Initial code drop of WebRTC r1804
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..90ec22b
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1 @@
+.svn
diff --git a/trunk/.gitignore b/trunk/.gitignore
new file mode 100644
index 0000000..c27b166
--- /dev/null
+++ b/trunk/.gitignore
@@ -0,0 +1,65 @@
+*.target.mk
+*.Makefile
+*.ncb
+*.ninja
+*.props
+*.pyc
+*.rules
+*.scons
+*.sdf
+*.sln
+*.suo
+*.targets
+*.user
+*.vcproj
+*.vcxproj
+*.vcxproj.filters
+*.vpj
+*.vpw
+*.vpwhistu
+*.vtg
+*.xcodeproj
+*~
+.*.sw?
+.DS_Store
+.cproject
+.gdb_history
+.gdbinit
+.metadata
+.project
+.pydevproject
+.settings
+/build
+/chromium_deps
+/gyp-mac-tool
+/Makefile
+/out
+/resources
+/src/supplement.gypi
+/testing
+/third_party/asan
+/third_party/cygwin
+/third_party/expat
+/third_party/gaeunit
+/third_party/gold
+/third_party/google-gflags/src
+/third_party/google-visualization-python
+/third_party/jsoncpp
+/third_party/libjingle
+/third_party/libjpeg
+/third_party/libjpeg_turbo
+/third_party/libsrtp
+/third_party/libvpx
+/third_party/libyuv
+/third_party/llvm-build
+/third_party/oauth2
+/third_party/protobuf
+/third_party/valgrind
+/third_party/yasm
+/tools/clang
+/tools/gyp
+/tools/python
+/tools/valgrind
+/tools/win
+/x86-generic_out/
+/xcodebuild
diff --git a/trunk/AUTHORS b/trunk/AUTHORS
new file mode 100644
index 0000000..c3d75d6
--- /dev/null
+++ b/trunk/AUTHORS
@@ -0,0 +1,6 @@
+# Names should be added to this file like so:
+# Name or Organization <email address>
+
+Google Inc.
+Mozilla Foundation
+Ben Strong <bstrong@gmail.com>
\ No newline at end of file
diff --git a/trunk/Android.mk b/trunk/Android.mk
new file mode 100644
index 0000000..b7c28e6
--- /dev/null
+++ b/trunk/Android.mk
@@ -0,0 +1,167 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+MY_WEBRTC_ROOT_PATH := $(call my-dir)
+
+# voice
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/neteq/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/cng/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g711/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g722/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/pcm16b/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/ilbc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_conference_mixer/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_device/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/media_file/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/rtp_rtcp/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/udp_transport/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/utility/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/source/Android.mk
+
+# video
+include $(MY_WEBRTC_ROOT_PATH)/src/common_video/jpeg/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_video/libyuv/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_capture/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/i420/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/vp8/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_processing/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_render/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/Android.mk
+
+# third party
+include $(MY_WEBRTC_ROOT_PATH)/libvpx.mk
+
+# build .so
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../external/webrtc/android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_audio_preprocessing
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_spl \
+ libwebrtc_resampler \
+ libwebrtc_apm \
+ libwebrtc_apm_utility \
+ libwebrtc_vad \
+ libwebrtc_ns \
+ libwebrtc_agc \
+ libwebrtc_aec \
+ libwebrtc_aecm \
+ libwebrtc_system_wrappers
+
+# Add Neon libraries.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_WHOLE_STATIC_LIBRARIES += \
+ libwebrtc_aecm_neon \
+ libwebrtc_ns_neon
+endif
+
+LOCAL_STATIC_LIBRARIES := \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
+###
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_system_wrappers \
+ libwebrtc_audio_device \
+ libwebrtc_pcm16b \
+ libwebrtc_cng \
+ libwebrtc_audio_coding \
+ libwebrtc_rtp_rtcp \
+ libwebrtc_media_file \
+ libwebrtc_udp_transport \
+ libwebrtc_utility \
+ libwebrtc_neteq \
+ libwebrtc_audio_conference_mixer \
+ libwebrtc_isac \
+ libwebrtc_ilbc \
+ libwebrtc_isacfix \
+ libwebrtc_g722 \
+ libwebrtc_g711 \
+ libwebrtc_voe_core \
+ libwebrtc_video_render \
+ libwebrtc_video_capture \
+ libwebrtc_i420 \
+ libwebrtc_video_coding \
+ libwebrtc_video_processing \
+ libwebrtc_vp8 \
+ libwebrtc_vie_core \
+ libwebrtc_yuv \
+ libwebrtc_jpeg \
+ libwebrtc_vpx
+
+# Add Neon libraries.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_WHOLE_STATIC_LIBRARIES += \
+ libwebrtc_isacfix_neon
+endif
+
+LOCAL_STATIC_LIBRARIES := \
+ libyuv_static
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport \
+ libjpeg \
+ libGLESv2 \
+ libOpenSLES \
+ libwebrtc_audio_preprocessing
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
+# test apps, they're for test only; all these test apps have LOCAL_MODULE_TAGS:=tests
+# voice engine test apps
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
+#include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/auto_test/Android.mk
+# video engien test apps
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/android_test/Android.mk
+#include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/test/auto_test/android/Android.mk
diff --git a/trunk/DEPS b/trunk/DEPS
new file mode 100644
index 0000000..0c8892e
--- /dev/null
+++ b/trunk/DEPS
@@ -0,0 +1,130 @@
+vars = {
+ # Use this googlecode_url variable only if there is an internal mirror for it.
+ # If you do not know, use the full path while defining your new deps entry.
+ "googlecode_url": "http://%s.googlecode.com/svn",
+ "chromium_trunk" : "http://src.chromium.org/svn/trunk",
+ "chromium_revision": "122775",
+
+ # External resources like video and audio files used for testing purposes.
+ # Downloaded on demand when needed.
+ "webrtc_resources_revision": "8",
+}
+
+# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
+# https; the latter can cause problems for users behind proxies.
+deps = {
+ "trunk/chromium_deps":
+ File(Var("chromium_trunk") + "/src/DEPS@" + Var("chromium_revision")),
+
+ "trunk/build":
+ Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
+
+ "trunk/testing":
+ Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
+
+ "trunk/testing/gmock":
+ From("trunk/chromium_deps", "src/testing/gmock"),
+
+ "trunk/testing/gtest":
+ From("trunk/chromium_deps", "src/testing/gtest"),
+
+ "trunk/third_party/expat":
+ Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
+
+ # Used by tools/quality_tracking.
+ "trunk/third_party/gaeunit":
+ "http://code.google.com/p/gaeunit.git@e16d5bd4",
+
+ "trunk/third_party/google-gflags/src":
+ (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
+
+ # Used by tools/quality_tracking/dashboard and tools/python_charts.
+ "trunk/third_party/google-visualization-python":
+ (Var("googlecode_url") % "google-visualization-python") + "/trunk@15",
+
+ "trunk/third_party/libjpeg":
+ Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
+
+ "trunk/third_party/libjpeg_turbo":
+ From("trunk/chromium_deps", "src/third_party/libjpeg_turbo"),
+
+ "trunk/third_party/libvpx/source/libvpx":
+ "http://git.chromium.org/webm/libvpx.git@v1.0.0",
+
+ "trunk/third_party/libyuv":
+ (Var("googlecode_url") % "libyuv") + "/trunk@190",
+
+ "trunk/third_party/protobuf":
+ Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
+
+ # Used by tools/quality_tracking.
+ "trunk/third_party/oauth2":
+ "http://github.com/simplegeo/python-oauth2.git@a83f4a29",
+
+ "trunk/third_party/yasm":
+ Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
+
+ "trunk/third_party/yasm/source/patched-yasm":
+ From("trunk/chromium_deps", "src/third_party/yasm/source/patched-yasm"),
+
+ "trunk/tools/clang":
+ Var("chromium_trunk") + "/src/tools/clang@" + Var("chromium_revision"),
+
+ "trunk/tools/gyp":
+ From("trunk/chromium_deps", "src/tools/gyp"),
+
+ "trunk/tools/python":
+ Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
+
+ "trunk/tools/valgrind":
+ Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
+
+ # Needed by build/common.gypi.
+ "trunk/tools/win/supalink":
+ Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
+}
+
+deps_os = {
+ "win": {
+ "trunk/third_party/cygwin":
+ Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
+
+ # Used by libjpeg-turbo.
+ "trunk/third_party/yasm/binaries":
+ From("trunk/chromium_deps", "src/third_party/yasm/binaries"),
+ },
+ "unix": {
+ "trunk/third_party/gold":
+ From("trunk/chromium_deps", "src/third_party/gold"),
+ },
+}
+
+hooks = [
+ {
+ # Create a supplement.gypi file under trunk/. This file will be picked up
+ # by gyp and we use it to set Chromium related variables (inside_chromium_build)
+ # to 0 and enable the standalone build.
+ "pattern": ".",
+ "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
+ },
+ {
+ # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
+ # zero seconds to run. If something changed, it downloads a prebuilt clang.
+ "pattern": ".",
+ "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
+ },
+ {
+ # Download test resources, i.e. video and audio files. If the latest
+ # version is already downloaded, this takes zero seconds to run.
+ # If a newer version or no current download exists, it will download
+ # the resources and extract them.
+ "pattern": ".",
+ "action": ["python", "trunk/tools/resources/update.py"],
+ },
+ {
+ # A change to a .gyp, .gypi, or to GYP itself should run the generator.
+ "pattern": ".",
+ "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/webrtc.gyp"],
+ },
+]
+
diff --git a/trunk/LICENSE b/trunk/LICENSE
new file mode 100644
index 0000000..dd4a345
--- /dev/null
+++ b/trunk/LICENSE
@@ -0,0 +1 @@
+Refer to src/LICENSE.
diff --git a/trunk/LICENSE_THIRD_PARTY b/trunk/LICENSE_THIRD_PARTY
new file mode 100644
index 0000000..d47c055
--- /dev/null
+++ b/trunk/LICENSE_THIRD_PARTY
@@ -0,0 +1 @@
+Refer to src/LICENSE_THIRD_PARTY.
diff --git a/trunk/OWNERS b/trunk/OWNERS
new file mode 100644
index 0000000..b110a52
--- /dev/null
+++ b/trunk/OWNERS
@@ -0,0 +1,5 @@
+henrika@webrtc.org
+niklas.enbom@webrtc.org
+andrew@webrtc.org
+tina.legrand@webrtc.org
+tommi@webrtc.org
\ No newline at end of file
diff --git a/trunk/PATENTS b/trunk/PATENTS
new file mode 100644
index 0000000..5cb83ec
--- /dev/null
+++ b/trunk/PATENTS
@@ -0,0 +1 @@
+Refer to src/PATENTS.
diff --git a/trunk/PRESUBMIT.py b/trunk/PRESUBMIT.py
new file mode 100644
index 0000000..56c8d5b
--- /dev/null
+++ b/trunk/PRESUBMIT.py
@@ -0,0 +1,144 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+def _LicenseHeader(input_api):
+ """Returns the license header regexp."""
+ license_header = (
+ r'.*? Copyright \(c\) %(year)s The WebRTC project authors\. '
+ r'All Rights Reserved\.\n'
+ r'.*?\n'
+ r'.*? Use of this source code is governed by a BSD-style license\n'
+ r'.*? that can be found in the LICENSE file in the root of the source\n'
+ r'.*? tree\. An additional intellectual property rights grant can be '
+ r'found\n'
+ r'.*? in the file PATENTS\. All contributing project authors may\n'
+ r'.*? be found in the AUTHORS file in the root of the source tree\.\n'
+ ) % {
+ 'year': input_api.time.strftime('%Y'),
+ }
+ return license_header
+
+def _CheckNoIOStreamInHeaders(input_api, output_api):
+ """Checks to make sure no .h files include <iostream>."""
+ files = []
+ pattern = input_api.re.compile(r'^#include\s*<iostream>',
+ input_api.re.MULTILINE)
+ for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
+ if not f.LocalPath().endswith('.h'):
+ continue
+ contents = input_api.ReadFile(f)
+ if pattern.search(contents):
+ files.append(f)
+
+ if len(files):
+ return [ output_api.PresubmitError(
+ 'Do not #include <iostream> in header files, since it inserts static ' +
+ 'initialization into every file including the header. Instead, ' +
+ '#include <ostream>. See http://crbug.com/94794',
+ files) ]
+ return []
+
+def _CheckNoFRIEND_TEST(input_api, output_api):
+ """Make sure that gtest's FRIEND_TEST() macro is not used, the
+ FRIEND_TEST_ALL_PREFIXES() macro from testsupport/gtest_prod_util.h should be
+ used instead since that allows for FLAKY_, FAILS_ and DISABLED_ prefixes."""
+ problems = []
+
+ file_filter = lambda f: f.LocalPath().endswith(('.cc', '.h'))
+ for f in input_api.AffectedFiles(file_filter=file_filter):
+ for line_num, line in f.ChangedContents():
+ if 'FRIEND_TEST(' in line:
+ problems.append(' %s:%d' % (f.LocalPath(), line_num))
+
+ if not problems:
+ return []
+ return [output_api.PresubmitPromptWarning('WebRTC\'s code should not use '
+ 'gtest\'s FRIEND_TEST() macro. Include testsupport/gtest_prod_util.h and '
+ 'use FRIEND_TEST_ALL_PREFIXES() instead.\n' + '\n'.join(problems))]
+
+def _CheckNewFilesLintClean(input_api, output_api, source_file_filter=None):
+ """Checks that all NEW '.cc' and '.h' files pass cpplint.py.
+ This check is based on _CheckChangeLintsClean in
+ depot_tools/presubmit_canned_checks.py but has less filters and only checks
+ added files."""
+ result = []
+
+ # Initialize cpplint.
+ import cpplint
+ # Access to a protected member _XX of a client class
+ # pylint: disable=W0212
+ cpplint._cpplint_state.ResetErrorCounts()
+
+ # Justifications for each filter:
+ #
+ # - build/header_guard : WebRTC coding style says they should be prefixed
+ # with WEBRTC_, which is not possible to configure in
+ # cpplint.py.
+ cpplint._SetFilters('-build/header_guard')
+
+ # Use the strictest verbosity level for cpplint.py (level 1) which is the
+ # default when running cpplint.py from command line.
+ # To make it possible to work with not-yet-converted code, we're only applying
+ # it to new (or moved/renamed) files.
+ verbosity_level = 1
+ files = []
+ for f in input_api.AffectedSourceFiles(source_file_filter):
+ # Note that moved/renamed files also count as added for svn.
+ if (f.Action() == 'A'):
+ files.append(f.AbsoluteLocalPath())
+ for file_name in files:
+ cpplint.ProcessFile(file_name, verbosity_level)
+
+ if cpplint._cpplint_state.error_count > 0:
+ if input_api.is_committing:
+ # TODO(kjellander): Change back to PresubmitError below when we're
+ # confident with the lint settings.
+ res_type = output_api.PresubmitPromptWarning
+ else:
+ res_type = output_api.PresubmitPromptWarning
+ result = [res_type('Changelist failed cpplint.py check.')]
+
+ return result
+
+def _CommonChecks(input_api, output_api):
+ """Checks common to both upload and commit."""
+ # TODO(kjellander): Use presubmit_canned_checks.PanProjectChecks too.
+ results = []
+ results.extend(input_api.canned_checks.CheckLongLines(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeHasNoTabs(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeHasNoStrayWhitespace(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeTodoHasOwner(
+ input_api, output_api))
+ results.extend(_CheckNewFilesLintClean(input_api, output_api))
+ results.extend(input_api.canned_checks.CheckLicense(
+ input_api, output_api, _LicenseHeader(input_api)))
+ results.extend(_CheckNoIOStreamInHeaders(input_api, output_api))
+ results.extend(_CheckNoFRIEND_TEST(input_api, output_api))
+ return results
+
+def CheckChangeOnUpload(input_api, output_api):
+ results = []
+ results.extend(_CommonChecks(input_api, output_api))
+ return results
+
+def CheckChangeOnCommit(input_api, output_api):
+ results = []
+ results.extend(_CommonChecks(input_api, output_api))
+ results.extend(input_api.canned_checks.CheckOwners(input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeWasUploaded(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeHasDescription(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeHasBugField(
+ input_api, output_api))
+ results.extend(input_api.canned_checks.CheckChangeHasTestField(
+ input_api, output_api))
+ return results
diff --git a/trunk/WATCHLISTS b/trunk/WATCHLISTS
new file mode 100644
index 0000000..1b592f5
--- /dev/null
+++ b/trunk/WATCHLISTS
@@ -0,0 +1,112 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Inspired by chromium.org:
+# http://dev.chromium.org/developers/contributing-code/watchlists
+
+{
+ 'WATCHLIST_DEFINITIONS': {
+ 'this_file': {
+ 'filepath': '^WATCHLISTS$',
+ },
+ 'all_src': {
+ 'filepath': 'src/.*',
+ },
+ 'root_files': {
+ # src/build/ and non-recursive contents of ./ and src/
+ 'filepath': '^[^/]*$|src/[^/]*$|src/build/.*',
+ },
+ 'documented_interfaces': {
+ 'filepath': 'src/[^/]*\.h$|'\
+ 'src/video_engine/main/interface/.*|'\
+ 'src/voice_engine/main/interface/.*',
+ },
+ 'build_files': {
+ 'filepath': '\.gyp$|\.gypi$|Android\.mk$',
+ },
+ 'java_files': {
+ 'filepath': '\.java$|\.xml$',
+ },
+ 'video_engine': {
+ 'filepath': 'src/video_engine/.*',
+ },
+ 'voice_engine': {
+ 'filepath': 'src/voice_engine/.*',
+ },
+ 'common_audio': {
+ 'filepath': 'src/common_audio/.*',
+ },
+ 'video_capture': {
+ 'filepath': 'src/modules/video_capture/.*',
+ },
+ 'video_render': {
+ 'filepath': 'src/modules/video_render/.*',
+ },
+ 'audio_device': {
+ 'filepath': 'src/modules/audio_device/.*',
+ },
+ 'audio_coding': {
+ 'filepath': 'src/modules/audio_coding/.*',
+ },
+ 'neteq': {
+ 'filepath': 'src/modules/audio_coding/neteq/.*',
+ },
+ 'audio_processing': {
+ 'filepath': 'src/modules/audio_processing/.*',
+ },
+ 'video_codecs': {
+ 'filepath': 'src/modules/video_coding/codecs/.*',
+ },
+ 'video_coding': {
+ 'filepath': 'src/modules/video_coding/.*',
+ },
+ 'rtp_rtcp': {
+ 'filepath': 'src/modules/rtp_rtcp/.*'
+ },
+ 'system_wrappers': {
+ 'filepath': 'src/system_wrappers/.*',
+ },
+ },
+
+ 'WATCHLISTS': {
+ 'this_file': [''],
+ 'all_src': ['tterriberry@mozilla.com',
+ 'giles@mozilla.com'],
+ 'root_files': ['andrew@webrtc.org',
+ 'niklas.enbom@webrtc.org'],
+ 'documented_interfaces': ['interface-changes@webrtc.org',
+ 'rwolff@gocast.it'],
+ 'build_files': ['leozwang@webrtc.org'],
+ 'java_files': ['leozwang@webrtc.org'],
+ 'common_audio': ['bjornv@webrtc.org',
+ 'andrew@webrtc.org'],
+ 'video_engine': ['mflodman@webrtc.org',
+ 'perkj@webrtc.org'],
+ 'voice_engine': ['henrika@webrtc.org'],
+ 'video_capture': ['mflodman@webrtc.org',
+ 'perkj@webrtc.org',
+ 'leozwang@webrtc.org'],
+ 'video_render': ['mflodman@webrtc.org',
+ 'perkj@webrtc.org',
+ 'leozwang@webrtc.org'],
+ 'audio_device': ['henrika@webrtc.org',
+ 'leozwang@webrtc.org'],
+ 'audio_coding': ['tina.legrand@webrtc.org'],
+ 'neteq': ['henrik.lundin@webrtc.org'],
+ 'audio_processing': ['andrew@webrtc.org',
+ 'bjornv@webrtc.org'],
+ 'video_codecs': ['henrik.lundin@webrtc.org',
+ 'pwestin@webrtc.org'],
+ 'video_coding': ['stefan@webrtc.org'],
+ 'rtp_rtcp': ['mflodman@webrtc.org',
+ 'pwestin@webrtc.org'],
+ 'system_wrappers': ['mflodman@webrtc.org',
+ 'henrika@webrtc.org',
+ 'andrew@webrtc.org'],
+ },
+}
diff --git a/trunk/android-webrtc.mk b/trunk/android-webrtc.mk
new file mode 100644
index 0000000..cd495b0
--- /dev/null
+++ b/trunk/android-webrtc.mk
@@ -0,0 +1,46 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# These defines will apply to all source files
+# Think again before changing it
+MY_WEBRTC_COMMON_DEFS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
+ '-DWEBRTC_ANDROID'
+# The following macros are used by modules,
+# we might need to re-organize them
+# '-DWEBRTC_ANDROID_OPENSLES' [module audio_device]
+# '-DNETEQ_VOICEENGINE_CODECS' [module audio_coding neteq]
+# '-DWEBRTC_MODULE_UTILITY_VIDEO' [module media_file] [module utility]
+ifeq ($(TARGET_ARCH),arm)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM'
+# '-DWEBRTC_DETECT_ARM_NEON' # only used in a build configuration without Neon
+# TODO(kma): figure out if the above define could be moved to NDK build only.
+
+# TODO(kma): test if the code under next two macros works with generic GCC compilers
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM_NEON'
+MY_ARM_CFLAGS_NEON := \
+ -flax-vector-conversions
+endif
+
+ifneq (,$(filter '-DWEBRTC_DETECT_ARM_NEON' '-DWEBRTC_ARCH_ARM_NEON', \
+ $(MY_WEBRTC_COMMON_DEFS)))
+WEBRTC_BUILD_NEON_LIBS := true
+endif
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM_V7A'
+endif
+
+endif # ifeq ($(TARGET_ARCH),arm)
\ No newline at end of file
diff --git a/trunk/codereview.settings b/trunk/codereview.settings
new file mode 100644
index 0000000..b70057e
--- /dev/null
+++ b/trunk/codereview.settings
@@ -0,0 +1,9 @@
+# This file is used by gcl to get repository specific information.
+CODE_REVIEW_SERVER: webrtc-codereview.appspot.com
+#CC_LIST:
+#VIEW_VC:
+#STATUS:
+TRY_ON_UPLOAD: False
+#TRYSERVER_SVN_URL:
+#GITCL_PREUPLOAD:
+#GITCL_PREDCOMMIT:
diff --git a/trunk/libvpx.mk b/trunk/libvpx.mk
new file mode 100644
index 0000000..07c04dc
--- /dev/null
+++ b/trunk/libvpx.mk
@@ -0,0 +1,107 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+MY_LIBVPX_PATH = ../libvpx
+
+LOCAL_SRC_FILES = \
+ $(MY_LIBVPX_PATH)/vp8/common/alloccommon.c \
+ $(MY_LIBVPX_PATH)/vp8/common/blockd.c \
+ $(MY_LIBVPX_PATH)/vp8/common/debugmodes.c \
+ $(MY_LIBVPX_PATH)/vp8/common/entropy.c \
+ $(MY_LIBVPX_PATH)/vp8/common/entropymode.c \
+ $(MY_LIBVPX_PATH)/vp8/common/entropymv.c \
+ $(MY_LIBVPX_PATH)/vp8/common/extend.c \
+ $(MY_LIBVPX_PATH)/vp8/common/filter.c \
+ $(MY_LIBVPX_PATH)/vp8/common/findnearmv.c \
+ $(MY_LIBVPX_PATH)/vp8/common/generic/systemdependent.c \
+ $(MY_LIBVPX_PATH)/vp8/common/idctllm.c \
+ $(MY_LIBVPX_PATH)/vp8/common/invtrans.c \
+ $(MY_LIBVPX_PATH)/vp8/common/loopfilter.c \
+ $(MY_LIBVPX_PATH)/vp8/common/loopfilter_filters.c \
+ $(MY_LIBVPX_PATH)/vp8/common/mbpitch.c \
+ $(MY_LIBVPX_PATH)/vp8/common/modecont.c \
+ $(MY_LIBVPX_PATH)/vp8/common/modecontext.c \
+ $(MY_LIBVPX_PATH)/vp8/common/quant_common.c \
+ $(MY_LIBVPX_PATH)/vp8/common/recon.c \
+ $(MY_LIBVPX_PATH)/vp8/common/reconinter.c \
+ $(MY_LIBVPX_PATH)/vp8/common/reconintra.c \
+ $(MY_LIBVPX_PATH)/vp8/common/reconintra4x4.c \
+ $(MY_LIBVPX_PATH)/vp8/common/setupintrarecon.c \
+ $(MY_LIBVPX_PATH)/vp8/common/swapyv12buffer.c \
+ $(MY_LIBVPX_PATH)/vp8/common/textblit.c \
+ $(MY_LIBVPX_PATH)/vp8/common/treecoder.c \
+ $(MY_LIBVPX_PATH)/vp8/vp8_cx_iface.c \
+ $(MY_LIBVPX_PATH)/vp8/vp8_dx_iface.c \
+ $(MY_LIBVPX_PATH)/vpx_config.c \
+ $(MY_LIBVPX_PATH)/vpx/src/vpx_codec.c \
+ $(MY_LIBVPX_PATH)/vpx/src/vpx_decoder.c \
+ $(MY_LIBVPX_PATH)/vpx/src/vpx_image.c \
+ $(MY_LIBVPX_PATH)/vpx_mem/vpx_mem.c \
+ $(MY_LIBVPX_PATH)/vpx_scale/generic/vpxscale.c \
+ $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12config.c \
+ $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12extend.c \
+ $(MY_LIBVPX_PATH)/vpx_scale/generic/gen_scalers.c \
+ $(MY_LIBVPX_PATH)/vpx_scale/generic/scalesystemdependent.c \
+ $(MY_LIBVPX_PATH)/vpx/src/vpx_encoder.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/bitstream.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/boolhuff.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/dct.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/encodeframe.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/encodeintra.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/encodemb.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/encodemv.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/ethreading.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/firstpass.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/generic/csystemdependent.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/mcomp.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/modecosts.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/pickinter.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/picklpf.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/psnr.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/quantize.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/ratectrl.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/rdopt.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/sad_c.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/segmentation.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/tokenize.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/treewriter.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/onyx_if.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/temporal_filter.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/variance_c.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/dboolhuff.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/decodemv.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/decodframe.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/dequantize.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/detokenize.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/generic/dsystemdependent.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/onyxd_if.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/reconintra_mt.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/threading.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/idct_blk.c \
+ $(MY_LIBVPX_PATH)/vp8/common/arm/arm_systemdependent.c \
+ $(MY_LIBVPX_PATH)/vp8/encoder/arm/arm_csystemdependent.c \
+ $(MY_LIBVPX_PATH)/vp8/decoder/arm/arm_dsystemdependent.c \
+
+LOCAL_CFLAGS := \
+ -DHAVE_CONFIG_H=vpx_config.h \
+ -include $(LOCAL_PATH)/third_party/libvpx/source/config/android/vpx_config.h
+
+LOCAL_MODULE := libwebrtc_vpx
+
+LOCAL_C_INCLUDES := \
+ external/libvpx \
+ external/libvpx/vpx_ports \
+ external/libvpx/vp8/common \
+ external/libvpx/vp8/encoder \
+ external/libvpx/vp8 \
+ external/libvpx/vpx_codec
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/license_template.txt b/trunk/license_template.txt
new file mode 100644
index 0000000..5a3e653
--- /dev/null
+++ b/trunk/license_template.txt
@@ -0,0 +1,10 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
diff --git a/trunk/peerconnection/DEPS b/trunk/peerconnection/DEPS
new file mode 100644
index 0000000..7f1e4fe
--- /dev/null
+++ b/trunk/peerconnection/DEPS
@@ -0,0 +1,132 @@
+vars = {
+ # Use this googlecode_url variable only if there is an internal mirror for it.
+ # If you do not know, use the full path while defining your new deps entry.
+ "googlecode_url": "http://%s.googlecode.com/svn",
+ "chromium_trunk" : "http://src.chromium.org/svn/trunk",
+ "chromium_revision": "120526",
+ "webrtc_revision": "1538",
+ "libjingle_revision": "115",
+}
+
+# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
+# https; the latter can cause problems for users behind proxies.
+deps = {
+ # WebRTC deps.
+ "trunk/src":
+ (Var("googlecode_url") % "webrtc") + "/trunk/src@" + Var("webrtc_revision"),
+
+ "trunk/tools":
+ (Var("googlecode_url") % "webrtc") + "/trunk/tools@" + Var("webrtc_revision"),
+
+ "trunk/test":
+ (Var("googlecode_url") % "webrtc") + "/trunk/test@" + Var("webrtc_revision"),
+
+ "trunk/third_party/google-gflags":
+ (Var("googlecode_url") % "webrtc") + "/trunk/third_party/google-gflags@" + Var("webrtc_revision"),
+
+ "trunk/third_party/libvpx":
+ (Var("googlecode_url") % "webrtc") + "/trunk/third_party/libvpx@" + Var("webrtc_revision"),
+
+ "trunk/build":
+ Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
+
+ "trunk/testing":
+ Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
+
+ "trunk/testing/gtest":
+ (Var("googlecode_url") % "googletest") + "/trunk@573",
+
+ "trunk/testing/gmock":
+ (Var("googlecode_url") % "googlemock") + "/trunk@386",
+
+ "trunk/tools/gyp":
+ (Var("googlecode_url") % "gyp") + "/trunk@1187",
+
+ # Needed by build/common.gypi.
+ "trunk/tools/win/supalink":
+ Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
+
+ "trunk/tools/clang/scripts":
+ Var("chromium_trunk") + "/src/tools/clang/scripts@" + Var("chromium_revision"),
+
+ "trunk/tools/python":
+ Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
+
+ "trunk/tools/valgrind":
+ Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
+
+ "trunk/third_party/protobuf/":
+ Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
+
+ "trunk/third_party/libvpx/source/libvpx":
+ "http://git.chromium.org/webm/libvpx.git@e479379a",
+
+ "trunk/third_party/libjpeg_turbo/":
+ Var("chromium_trunk") + "/deps/third_party/libjpeg_turbo@119959",
+
+ "trunk/third_party/libjpeg/":
+ Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
+
+ "trunk/third_party/yasm/":
+ Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
+
+ "trunk/third_party/expat/":
+ Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
+
+ "trunk/third_party/google-gflags/src":
+ (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
+
+ "trunk/third_party/yasm/source/patched-yasm":
+ Var("chromium_trunk") + "/deps/third_party/yasm/patched-yasm@73761",
+
+ # Used by libjpeg-turbo
+ "trunk/third_party/yasm/binaries":
+ Var("chromium_trunk") + "/deps/third_party/yasm/binaries@74228",
+
+ # TODO(andrew): roll to 164 after fixing:
+ # http://code.google.com/p/webrtc/issues/detail?id=267
+ "trunk/third_party/libyuv":
+ (Var("googlecode_url") % "libyuv") + "/trunk@121",
+
+ # libjingle deps.
+ "trunk/third_party/libjingle/source":
+ (Var("googlecode_url") % "libjingle") + "/trunk/@" + Var("libjingle_revision"),
+
+ "trunk/third_party/libsrtp/":
+ Var("chromium_trunk") + "/deps/third_party/libsrtp@119742",
+
+ "trunk/third_party/jsoncpp/":
+ Var("chromium_trunk") + "/src/third_party/jsoncpp@" + Var("chromium_revision"),
+
+ "trunk/third_party/jsoncpp/source":
+ "http://jsoncpp.svn.sourceforge.net/svnroot/jsoncpp/trunk/jsoncpp@248",
+}
+
+deps_os = {
+ "win": {
+ "trunk/third_party/cygwin/":
+ Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
+ }
+}
+
+hooks = [
+ {
+ # Create a supplement.gypi file under trunk/. This file will be picked up
+ # by gyp and we use it to set Chromium related variables (inside_chromium_build)
+ # to 0 and enable the standalone build.
+ "pattern": ".",
+ "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
+ },
+ {
+ # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
+ # zero seconds to run. If something changed, it downloads a prebuilt clang.
+ "pattern": ".",
+ "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
+ },
+ {
+ # A change to a .gyp, .gypi, or to GYP itself should run the generator.
+ "pattern": ".",
+ "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/peerconnection.gyp"],
+ },
+]
+
diff --git a/trunk/peerconnection/OWNERS b/trunk/peerconnection/OWNERS
new file mode 100644
index 0000000..1527445
--- /dev/null
+++ b/trunk/peerconnection/OWNERS
@@ -0,0 +1,5 @@
+henrike@webrtc.org
+mallinath@webrtc.org
+perkj@webrtc.org
+wu@webrtc.org
+tommi@webrtc.org
diff --git a/trunk/peerconnection/README b/trunk/peerconnection/README
new file mode 100644
index 0000000..bfe1234
--- /dev/null
+++ b/trunk/peerconnection/README
@@ -0,0 +1,10 @@
+This folder can be used to pull together the chromium version of webrtc
+and libjingle, and build the peerconnection sample client and server.
+
+Follow the instructions for WebRTC here:
+http://www.webrtc.org/reference/getting-started
+
+but substitute this "gclient config" command:
+gclient config --name trunk
+http://webrtc.googlecode.com/svn/trunk/peerconnection
+
diff --git a/trunk/peerconnection/peerconnection.gyp b/trunk/peerconnection/peerconnection.gyp
new file mode 100644
index 0000000..e928194
--- /dev/null
+++ b/trunk/peerconnection/peerconnection.gyp
@@ -0,0 +1,113 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ 'src/build/common.gypi', ],
+ 'variables': {
+ 'peerconnection_sample': 'third_party/libjingle/source/talk/examples/peerconnection',
+ },
+
+ 'targets': [
+ {
+ 'target_name': 'peerconnection_server',
+ 'type': 'executable',
+ 'sources': [
+ '<(peerconnection_sample)/server/data_socket.cc',
+ '<(peerconnection_sample)/server/data_socket.h',
+ '<(peerconnection_sample)/server/main.cc',
+ '<(peerconnection_sample)/server/peer_channel.cc',
+ '<(peerconnection_sample)/server/peer_channel.h',
+ '<(peerconnection_sample)/server/utils.cc',
+ '<(peerconnection_sample)/server/utils.h',
+ ],
+ 'include_dirs': [
+ 'third_party/libjingle/source',
+ ],
+ },
+ ],
+ 'conditions': [
+ ['OS=="win"', {
+ 'targets': [
+ {
+ 'target_name': 'peerconnection_client',
+ 'type': 'executable',
+ 'sources': [
+ '<(peerconnection_sample)/client/conductor.cc',
+ '<(peerconnection_sample)/client/conductor.h',
+ '<(peerconnection_sample)/client/defaults.cc',
+ '<(peerconnection_sample)/client/defaults.h',
+ '<(peerconnection_sample)/client/main.cc',
+ '<(peerconnection_sample)/client/main_wnd.cc',
+ '<(peerconnection_sample)/client/main_wnd.h',
+ '<(peerconnection_sample)/client/peer_connection_client.cc',
+ '<(peerconnection_sample)/client/peer_connection_client.h',
+ 'third_party/libjingle/source/talk/base/win32socketinit.cc',
+ 'third_party/libjingle/source/talk/base/win32socketserver.cc',
+ ],
+ 'msvs_settings': {
+ 'VCLinkerTool': {
+ 'SubSystem': '2', # Windows
+ },
+ },
+ 'dependencies': [
+ 'third_party/libjingle/libjingle.gyp:libjingle_app',
+ ],
+ 'include_dirs': [
+ 'src',
+ 'src/modules/interface',
+ 'third_party/libjingle/source',
+ ],
+ },
+ ], # targets
+ }, ], # OS="win"
+ ['OS=="linux"', {
+ 'targets': [
+ {
+ 'target_name': 'peerconnection_client',
+ 'type': 'executable',
+ 'sources': [
+ '<(peerconnection_sample)/client/conductor.cc',
+ '<(peerconnection_sample)/client/conductor.h',
+ '<(peerconnection_sample)/client/defaults.cc',
+ '<(peerconnection_sample)/client/defaults.h',
+ '<(peerconnection_sample)/client/linux/main.cc',
+ '<(peerconnection_sample)/client/linux/main_wnd.cc',
+ '<(peerconnection_sample)/client/linux/main_wnd.h',
+ '<(peerconnection_sample)/client/peer_connection_client.cc',
+ '<(peerconnection_sample)/client/peer_connection_client.h',
+ ],
+ 'dependencies': [
+ 'third_party/libjingle/libjingle.gyp:libjingle_app',
+ # TODO(tommi): Switch to this and remove specific gtk dependency
+ # sections below for cflags and link_settings.
+ # '<(DEPTH)/build/linux/system.gyp:gtk',
+ ],
+ 'include_dirs': [
+ 'src',
+ 'src/modules/interface',
+ 'third_party/libjingle/source',
+ ],
+ 'cflags': [
+ '<!@(pkg-config --cflags gtk+-2.0)',
+ ],
+ 'link_settings': {
+ 'ldflags': [
+ '<!@(pkg-config --libs-only-L --libs-only-other gtk+-2.0 gthread-2.0)',
+ ],
+ 'libraries': [
+ '<!@(pkg-config --libs-only-l gtk+-2.0 gthread-2.0)',
+ '-lX11',
+ '-lXext',
+ ],
+ },
+ },
+ ], # targets
+ }, ], # OS="linux"
+ ],
+
+}
diff --git a/trunk/peerconnection/third_party/libjingle/libjingle.gyp b/trunk/peerconnection/third_party/libjingle/libjingle.gyp
new file mode 100644
index 0000000..48939bf
--- /dev/null
+++ b/trunk/peerconnection/third_party/libjingle/libjingle.gyp
@@ -0,0 +1,698 @@
+# Copyright (c) 2012 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+ # TODO(wu): Use the libjingle.gyp from chromium and get rid of this file.
+ 'variables': {
+ 'no_libjingle_logging%': 0,
+ 'conditions': [
+ ['inside_chromium_build==1', {
+ 'overrides': 'overrides',
+ },{
+ 'overrides': 'source',
+ }],
+ ],
+ },
+ 'target_defaults': {
+ 'defines': [
+ 'FEATURE_ENABLE_SSL',
+ 'FEATURE_ENABLE_VOICEMAIL', # TODO(ncarter): Do we really need this?
+ '_USE_32BIT_TIME_T',
+ 'SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS',
+ 'EXPAT_RELATIVE_PATH',
+ 'JSONCPP_RELATIVE_PATH',
+ 'WEBRTC_RELATIVE_PATH',
+ 'HAVE_WEBRTC',
+ 'HAVE_WEBRTC_VIDEO',
+ 'HAVE_WEBRTC_VOICE',
+ 'HAVE_SRTP',
+ 'SRTP_RELATIVE_PATH',
+ ],
+ 'configurations': {
+ 'Debug': {
+ 'defines': [
+ # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
+ # _DEBUG and remove this define. See below as well.
+ '_DEBUG',
+ ],
+ }
+ },
+ 'dependencies': [
+ '../expat/expat.gyp:expat',
+ ],
+ 'export_dependent_settings': [
+ '../expat/expat.gyp:expat',
+ ],
+ 'include_dirs': [
+ '../../third_party/libyuv/include/',
+ ],
+ 'direct_dependent_settings': {
+ 'defines': [
+ 'FEATURE_ENABLE_SSL',
+ 'FEATURE_ENABLE_VOICEMAIL',
+ 'EXPAT_RELATIVE_PATH',
+ 'WEBRTC_RELATIVE_PATH',
+ ],
+ 'conditions': [
+ ['OS=="win"', {
+ 'link_settings': {
+ 'libraries': [
+ '-lsecur32.lib',
+ '-lcrypt32.lib',
+ '-liphlpapi.lib',
+ ],
+ },
+ }],
+ ['OS=="win"', {
+ 'include_dirs': [
+ '../third_party/platformsdk_win7/files/Include',
+ ],
+ 'defines': [
+ '_CRT_SECURE_NO_WARNINGS', # Suppres warnings about _vsnprinf
+ ],
+ }],
+ ['OS=="linux"', {
+ 'defines': [
+ 'LINUX',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'defines': [
+ 'OSX',
+ ],
+ }],
+ ['os_posix == 1', {
+ 'defines': [
+ 'POSIX',
+ ],
+ }],
+ ['OS=="openbsd" or OS=="freebsd"', {
+ 'defines': [
+ 'BSD',
+ ],
+ }],
+ ['no_libjingle_logging==1', {
+ 'defines': [
+ 'NO_LIBJINGLE_LOGGING',
+ ],
+ }],
+ ],
+ },
+ 'all_dependent_settings': {
+ 'configurations': {
+ 'Debug': {
+ 'defines': [
+ # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
+ # _DEBUG and remove this define. See above as well.
+ '_DEBUG',
+ ],
+ }
+ },
+ },
+ 'conditions': [
+ ['inside_chromium_build==1', {
+ 'defines': [
+ 'NO_SOUND_SYSTEM',
+ ],
+ 'include_dirs': [
+ '<(overrides)',
+ 'source',
+ '../..', # the third_party folder for webrtc includes
+ '../../third_party/expat/files',
+ ],
+ 'direct_dependent_settings': {
+ 'defines': [
+ 'NO_SOUND_SYSTEM',
+ ],
+ 'include_dirs': [
+ '<(overrides)',
+ 'source',
+ '../../third_party/expat/files'
+ ],
+ },
+ 'dependencies': [
+ '../../base/base.gyp:base',
+ '../../net/net.gyp:net',
+ ],
+ },{
+ 'include_dirs': [
+ # the third_party folder for webrtc/ includes (non-chromium).
+ '../../src',
+ 'source',
+ '../../third_party/expat/files',
+ ],
+ }],
+ ['OS=="win"', {
+ 'include_dirs': [
+ '../third_party/platformsdk_win7/files/Include',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'defines': [
+ 'LINUX',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'defines': [
+ 'OSX',
+ ],
+ }],
+ ['os_posix == 1', {
+ 'defines': [
+ 'POSIX',
+ ],
+ }],
+ ['OS=="openbsd" or OS=="freebsd"', {
+ 'defines': [
+ 'BSD',
+ ],
+ }],
+ ],
+ },
+ 'targets': [
+ {
+ 'target_name': 'libjingle',
+ 'type': 'static_library',
+ 'sources': [
+ '<(overrides)/talk/base/basictypes.h',
+ '<(overrides)/talk/base/constructormagic.h',
+
+ # Need to override logging.h because we need
+ # SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to work.
+ # TODO(sergeyu): push SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to
+ # libjingle and remove this override.
+ '<(overrides)/talk/base/logging.h',
+
+ '<(overrides)/talk/base/scoped_ptr.h',
+
+ # Libjingle's QName is not threadsafe, so we need to use our own version
+ # here.
+ # TODO(sergeyu): Fix QName in Libjingle.
+ '<(overrides)/talk/xmllite/qname.cc',
+ '<(overrides)/talk/xmllite/qname.h',
+
+ 'source/talk/base/Equifax_Secure_Global_eBusiness_CA-1.h',
+ 'source/talk/base/asyncfile.cc',
+ 'source/talk/base/asyncfile.h',
+ 'source/talk/base/asynchttprequest.cc',
+ 'source/talk/base/asynchttprequest.h',
+ 'source/talk/base/asyncpacketsocket.h',
+ 'source/talk/base/asyncsocket.cc',
+ 'source/talk/base/asyncsocket.h',
+ 'source/talk/base/asynctcpsocket.cc',
+ 'source/talk/base/asynctcpsocket.h',
+ 'source/talk/base/asyncudpsocket.cc',
+ 'source/talk/base/asyncudpsocket.h',
+ 'source/talk/base/autodetectproxy.cc',
+ 'source/talk/base/autodetectproxy.h',
+ 'source/talk/base/base64.cc',
+ 'source/talk/base/base64.h',
+ 'source/talk/base/basicdefs.h',
+ 'source/talk/base/basicpacketsocketfactory.cc',
+ 'source/talk/base/basicpacketsocketfactory.h',
+ 'source/talk/base/bytebuffer.cc',
+ 'source/talk/base/bytebuffer.h',
+ 'source/talk/base/byteorder.h',
+ 'source/talk/base/checks.cc',
+ 'source/talk/base/checks.h',
+ 'source/talk/base/common.cc',
+ 'source/talk/base/common.h',
+ 'source/talk/base/criticalsection.h',
+ 'source/talk/base/cryptstring.h',
+ 'source/talk/base/diskcache.cc',
+ 'source/talk/base/diskcache.h',
+ 'source/talk/base/event.cc',
+ 'source/talk/base/event.h',
+ 'source/talk/base/fileutils.cc',
+ 'source/talk/base/fileutils.h',
+ 'source/talk/base/firewallsocketserver.cc',
+ 'source/talk/base/firewallsocketserver.h',
+ 'source/talk/base/flags.cc',
+ 'source/talk/base/flags.h',
+ 'source/talk/base/helpers.cc',
+ 'source/talk/base/helpers.h',
+ 'source/talk/base/host.cc',
+ 'source/talk/base/host.h',
+ 'source/talk/base/httpbase.cc',
+ 'source/talk/base/httpbase.h',
+ 'source/talk/base/httpclient.h',
+ 'source/talk/base/httpclient.cc',
+ 'source/talk/base/httpcommon-inl.h',
+ 'source/talk/base/httpcommon.cc',
+ 'source/talk/base/httpcommon.h',
+ 'source/talk/base/httprequest.cc',
+ 'source/talk/base/httprequest.h',
+ 'source/talk/base/ipaddress.cc',
+ 'source/talk/base/ipaddress.h',
+ 'source/talk/base/json.cc',
+ 'source/talk/base/json.h',
+ 'source/talk/base/linked_ptr.h',
+ 'source/talk/base/logging.cc',
+ 'source/talk/base/md5.h',
+ 'source/talk/base/md5c.c',
+ 'source/talk/base/messagehandler.cc',
+ 'source/talk/base/messagehandler.h',
+ 'source/talk/base/messagequeue.cc',
+ 'source/talk/base/messagequeue.h',
+ 'source/talk/base/nethelpers.cc',
+ 'source/talk/base/nethelpers.h',
+ 'source/talk/base/network.cc',
+ 'source/talk/base/network.h',
+ 'source/talk/base/pathutils.cc',
+ 'source/talk/base/pathutils.h',
+ 'source/talk/base/physicalsocketserver.cc',
+ 'source/talk/base/physicalsocketserver.h',
+ 'source/talk/base/proxydetect.cc',
+ 'source/talk/base/proxydetect.h',
+ 'source/talk/base/proxyinfo.cc',
+ 'source/talk/base/proxyinfo.h',
+ 'source/talk/base/ratetracker.cc',
+ 'source/talk/base/ratetracker.h',
+ 'source/talk/base/sec_buffer.h',
+ 'source/talk/base/signalthread.cc',
+ 'source/talk/base/signalthread.h',
+ 'source/talk/base/sigslot.h',
+ 'source/talk/base/sigslotrepeater.h',
+ 'source/talk/base/socket.h',
+ 'source/talk/base/socketadapters.cc',
+ 'source/talk/base/socketadapters.h',
+ 'source/talk/base/socketaddress.cc',
+ 'source/talk/base/socketaddress.h',
+ 'source/talk/base/socketaddresspair.cc',
+ 'source/talk/base/socketaddresspair.h',
+ 'source/talk/base/socketfactory.h',
+ 'source/talk/base/socketpool.cc',
+ 'source/talk/base/socketpool.h',
+ 'source/talk/base/socketserver.h',
+ 'source/talk/base/socketstream.cc',
+ 'source/talk/base/socketstream.h',
+ 'source/talk/base/ssladapter.cc',
+ 'source/talk/base/ssladapter.h',
+ 'source/talk/base/sslsocketfactory.cc',
+ 'source/talk/base/sslsocketfactory.h',
+ 'source/talk/base/stream.cc',
+ 'source/talk/base/stream.h',
+ 'source/talk/base/stringdigest.cc',
+ 'source/talk/base/stringdigest.h',
+ 'source/talk/base/stringencode.cc',
+ 'source/talk/base/stringencode.h',
+ 'source/talk/base/stringutils.cc',
+ 'source/talk/base/stringutils.h',
+ 'source/talk/base/task.cc',
+ 'source/talk/base/task.h',
+ 'source/talk/base/taskparent.cc',
+ 'source/talk/base/taskparent.h',
+ 'source/talk/base/taskrunner.cc',
+ 'source/talk/base/taskrunner.h',
+ 'source/talk/base/thread.cc',
+ 'source/talk/base/thread.h',
+ 'source/talk/base/time.cc',
+ 'source/talk/base/time.h',
+ 'source/talk/base/urlencode.cc',
+ 'source/talk/base/urlencode.h',
+ 'source/talk/base/worker.cc',
+ 'source/talk/base/worker.h',
+ 'source/talk/xmllite/xmlbuilder.cc',
+ 'source/talk/xmllite/xmlbuilder.h',
+ 'source/talk/xmllite/xmlconstants.cc',
+ 'source/talk/xmllite/xmlconstants.h',
+ 'source/talk/xmllite/xmlelement.cc',
+ 'source/talk/xmllite/xmlelement.h',
+ 'source/talk/xmllite/xmlnsstack.cc',
+ 'source/talk/xmllite/xmlnsstack.h',
+ 'source/talk/xmllite/xmlparser.cc',
+ 'source/talk/xmllite/xmlparser.h',
+ 'source/talk/xmllite/xmlprinter.cc',
+ 'source/talk/xmllite/xmlprinter.h',
+ 'source/talk/xmpp/asyncsocket.h',
+ 'source/talk/xmpp/constants.cc',
+ 'source/talk/xmpp/constants.h',
+ 'source/talk/xmpp/jid.cc',
+ 'source/talk/xmpp/jid.h',
+ 'source/talk/xmpp/plainsaslhandler.h',
+ 'source/talk/xmpp/prexmppauth.h',
+ 'source/talk/xmpp/ratelimitmanager.cc',
+ 'source/talk/xmpp/ratelimitmanager.h',
+ 'source/talk/xmpp/saslcookiemechanism.h',
+ 'source/talk/xmpp/saslhandler.h',
+ 'source/talk/xmpp/saslmechanism.cc',
+ 'source/talk/xmpp/saslmechanism.h',
+ 'source/talk/xmpp/saslplainmechanism.h',
+ 'source/talk/xmpp/xmppclient.cc',
+ 'source/talk/xmpp/xmppclient.h',
+ 'source/talk/xmpp/xmppclientsettings.h',
+ 'source/talk/xmpp/xmppengine.h',
+ 'source/talk/xmpp/xmppengineimpl.cc',
+ 'source/talk/xmpp/xmppengineimpl.h',
+ 'source/talk/xmpp/xmppengineimpl_iq.cc',
+ 'source/talk/xmpp/xmpplogintask.cc',
+ 'source/talk/xmpp/xmpplogintask.h',
+ 'source/talk/xmpp/xmppstanzaparser.cc',
+ 'source/talk/xmpp/xmppstanzaparser.h',
+ 'source/talk/xmpp/xmpptask.cc',
+ 'source/talk/xmpp/xmpptask.h',
+ ],
+ 'conditions': [
+ ['inside_chromium_build==0', {
+ 'sources': [
+ 'source/talk/sound/automaticallychosensoundsystem.h',
+ 'source/talk/sound/platformsoundsystem.cc',
+ 'source/talk/sound/platformsoundsystem.h',
+ 'source/talk/sound/platformsoundsystemfactory.cc',
+ 'source/talk/sound/platformsoundsystemfactory.h',
+ 'source/talk/sound/sounddevicelocator.h',
+ 'source/talk/sound/soundinputstreaminterface.h',
+ 'source/talk/sound/soundoutputstreaminterface.h',
+ 'source/talk/sound/soundsystemfactory.h',
+ 'source/talk/sound/soundsysteminterface.cc',
+ 'source/talk/sound/soundsysteminterface.h',
+ 'source/talk/sound/soundsystemproxy.cc',
+ 'source/talk/sound/soundsystemproxy.h',
+ ],
+ 'conditions' : [
+ ['OS=="linux"', {
+ 'sources': [
+ 'source/talk/sound/alsasoundsystem.cc',
+ 'source/talk/sound/alsasoundsystem.h',
+ 'source/talk/sound/alsasymboltable.cc',
+ 'source/talk/sound/alsasymboltable.h',
+ 'source/talk/sound/linuxsoundsystem.cc',
+ 'source/talk/sound/linuxsoundsystem.h',
+ 'source/talk/sound/pulseaudiosoundsystem.cc',
+ 'source/talk/sound/pulseaudiosoundsystem.h',
+ 'source/talk/sound/pulseaudiosymboltable.cc',
+ 'source/talk/sound/pulseaudiosymboltable.h',
+ ],
+ }],
+ ],
+ }],
+ ['OS=="win"', {
+ 'sources': [
+ '<(overrides)/talk/base/win32socketinit.cc',
+ 'source/talk/base/schanneladapter.cc',
+ 'source/talk/base/schanneladapter.h',
+ 'source/talk/base/win32.h',
+ 'source/talk/base/win32.cc',
+ 'source/talk/base/win32filesystem.cc',
+ 'source/talk/base/win32filesystem.h',
+ 'source/talk/base/win32window.h',
+ 'source/talk/base/win32window.cc',
+ 'source/talk/base/win32securityerrors.cc',
+ 'source/talk/base/winfirewall.cc',
+ 'source/talk/base/winfirewall.h',
+ 'source/talk/base/winping.cc',
+ 'source/talk/base/winping.h',
+ ],
+ }],
+ ['os_posix == 1', {
+ 'sources': [
+ 'source/talk/base/latebindingsymboltable.cc',
+ 'source/talk/base/latebindingsymboltable.h',
+ 'source/talk/base/sslstreamadapter.cc',
+ 'source/talk/base/sslstreamadapter.h',
+ 'source/talk/base/unixfilesystem.cc',
+ 'source/talk/base/unixfilesystem.h',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'sources': [
+ 'source/talk/base/linux.cc',
+ 'source/talk/base/linux.h',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'sources': [
+ 'source/talk/base/macconversion.cc',
+ 'source/talk/base/macconversion.h',
+ 'source/talk/base/macutils.cc',
+ 'source/talk/base/macutils.h',
+ ],
+ }],
+ ],
+ 'dependencies': [
+ '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+ ],
+ },
+ # This has to be is a separate project due to a bug in MSVS:
+ # https://connect.microsoft.com/VisualStudio/feedback/details/368272/duplicate-cpp-filename-in-c-project-visual-studio-2008
+ # We have two files named "constants.cc" and MSVS doesn't handle this
+ # properly.
+ {
+ 'target_name': 'libjingle_p2p',
+ 'type': 'static_library',
+ 'sources': [
+ 'source/talk/p2p/base/candidate.h',
+ 'source/talk/p2p/base/common.h',
+ 'source/talk/p2p/base/constants.cc',
+ 'source/talk/p2p/base/constants.h',
+ 'source/talk/p2p/base/p2ptransport.cc',
+ 'source/talk/p2p/base/p2ptransport.h',
+ 'source/talk/p2p/base/p2ptransportchannel.cc',
+ 'source/talk/p2p/base/p2ptransportchannel.h',
+ 'source/talk/p2p/base/port.cc',
+ 'source/talk/p2p/base/port.h',
+ 'source/talk/p2p/base/portallocator.cc',
+ 'source/talk/p2p/base/portallocator.h',
+ 'source/talk/p2p/base/portallocatorsessionproxy.cc',
+ 'source/talk/p2p/base/portallocatorsessionproxy.h',
+ 'source/talk/p2p/base/portproxy.cc',
+ 'source/talk/p2p/base/portproxy.h',
+ 'source/talk/p2p/base/pseudotcp.cc',
+ 'source/talk/p2p/base/pseudotcp.h',
+ 'source/talk/p2p/base/rawtransport.cc',
+ 'source/talk/p2p/base/rawtransport.h',
+ 'source/talk/p2p/base/rawtransportchannel.cc',
+ 'source/talk/p2p/base/rawtransportchannel.h',
+ 'source/talk/p2p/base/relayport.cc',
+ 'source/talk/p2p/base/relayport.h',
+ 'source/talk/p2p/base/session.cc',
+ 'source/talk/p2p/base/session.h',
+ 'source/talk/p2p/base/sessionclient.h',
+ 'source/talk/p2p/base/sessiondescription.cc',
+ 'source/talk/p2p/base/sessiondescription.h',
+ 'source/talk/p2p/base/sessionid.h',
+ 'source/talk/p2p/base/sessionmanager.cc',
+ 'source/talk/p2p/base/sessionmanager.h',
+ 'source/talk/p2p/base/sessionmessages.cc',
+ 'source/talk/p2p/base/sessionmessages.h',
+ 'source/talk/p2p/base/parsing.cc',
+ 'source/talk/p2p/base/parsing.h',
+ 'source/talk/p2p/base/stun.cc',
+ 'source/talk/p2p/base/stun.h',
+ 'source/talk/p2p/base/stunport.cc',
+ 'source/talk/p2p/base/stunport.h',
+ 'source/talk/p2p/base/stunrequest.cc',
+ 'source/talk/p2p/base/stunrequest.h',
+ 'source/talk/p2p/base/tcpport.cc',
+ 'source/talk/p2p/base/tcpport.h',
+ 'source/talk/p2p/base/transport.cc',
+ 'source/talk/p2p/base/transport.h',
+ 'source/talk/p2p/base/transportchannel.cc',
+ 'source/talk/p2p/base/transportchannel.h',
+ 'source/talk/p2p/base/transportchannelimpl.h',
+ 'source/talk/p2p/base/transportchannelproxy.cc',
+ 'source/talk/p2p/base/transportchannelproxy.h',
+ 'source/talk/p2p/base/udpport.cc',
+ 'source/talk/p2p/base/udpport.h',
+ 'source/talk/p2p/client/basicportallocator.cc',
+ 'source/talk/p2p/client/basicportallocator.h',
+ 'source/talk/p2p/client/httpportallocator.cc',
+ 'source/talk/p2p/client/httpportallocator.h',
+ 'source/talk/p2p/client/sessionmanagertask.h',
+ 'source/talk/p2p/client/sessionsendtask.h',
+ 'source/talk/p2p/client/socketmonitor.cc',
+ 'source/talk/p2p/client/socketmonitor.h',
+ 'source/talk/session/phone/audiomonitor.cc',
+ 'source/talk/session/phone/audiomonitor.h',
+ 'source/talk/session/phone/call.cc',
+ 'source/talk/session/phone/call.h',
+ 'source/talk/session/phone/channel.cc',
+ 'source/talk/session/phone/channel.h',
+ 'source/talk/session/phone/channelmanager.cc',
+ 'source/talk/session/phone/channelmanager.h',
+ 'source/talk/session/phone/codec.cc',
+ 'source/talk/session/phone/codec.h',
+ 'source/talk/session/phone/cryptoparams.h',
+ 'source/talk/session/phone/currentspeakermonitor.cc',
+ 'source/talk/session/phone/currentspeakermonitor.h',
+ 'source/talk/session/phone/devicemanager.cc',
+ 'source/talk/session/phone/devicemanager.h',
+ 'source/talk/session/phone/filemediaengine.cc',
+ 'source/talk/session/phone/filemediaengine.h',
+ 'source/talk/session/phone/mediachannel.h',
+ 'source/talk/session/phone/mediaengine.cc',
+ 'source/talk/session/phone/mediaengine.h',
+ 'source/talk/session/phone/mediamessages.cc',
+ 'source/talk/session/phone/mediamessages.h',
+ 'source/talk/session/phone/mediamonitor.cc',
+ 'source/talk/session/phone/mediamonitor.h',
+ 'source/talk/session/phone/mediasession.cc',
+ 'source/talk/session/phone/mediasession.h',
+ 'source/talk/session/phone/mediasessionclient.cc',
+ 'source/talk/session/phone/mediasessionclient.h',
+ 'source/talk/session/phone/mediasink.h',
+ 'source/talk/session/phone/rtcpmuxfilter.cc',
+ 'source/talk/session/phone/rtcpmuxfilter.h',
+ 'source/talk/session/phone/rtpdump.cc',
+ 'source/talk/session/phone/rtpdump.h',
+ 'source/talk/session/phone/rtputils.cc',
+ 'source/talk/session/phone/rtputils.h',
+ 'source/talk/session/phone/soundclip.cc',
+ 'source/talk/session/phone/soundclip.h',
+ 'source/talk/session/phone/srtpfilter.cc',
+ 'source/talk/session/phone/srtpfilter.h',
+ 'source/talk/session/phone/ssrcmuxfilter.cc',
+ 'source/talk/session/phone/ssrcmuxfilter.h',
+ 'source/talk/session/phone/streamparams.cc',
+ 'source/talk/session/phone/videocapturer.cc',
+ 'source/talk/session/phone/videocapturer.h',
+ 'source/talk/session/phone/videocommon.cc',
+ 'source/talk/session/phone/videocommon.h',
+ 'source/talk/session/phone/videoframe.cc',
+ 'source/talk/session/phone/videoframe.h',
+ 'source/talk/session/phone/voicechannel.h',
+ 'source/talk/session/phone/webrtcpassthroughrender.cc',
+ 'source/talk/session/phone/webrtccommon.h',
+ 'source/talk/session/phone/webrtcvideocapturer.cc',
+ 'source/talk/session/phone/webrtcvideocapturer.h',
+ 'source/talk/session/phone/webrtcvideoengine.cc',
+ 'source/talk/session/phone/webrtcvideoengine.h',
+ 'source/talk/session/phone/webrtcvideoframe.cc',
+ 'source/talk/session/phone/webrtcvideoframe.h',
+ 'source/talk/session/phone/webrtcvie.h',
+ 'source/talk/session/phone/webrtcvoe.h',
+ 'source/talk/session/phone/webrtcvoiceengine.cc',
+ 'source/talk/session/phone/webrtcvoiceengine.h',
+ 'source/talk/session/tunnel/pseudotcpchannel.cc',
+ 'source/talk/session/tunnel/pseudotcpchannel.h',
+ 'source/talk/session/tunnel/tunnelsessionclient.cc',
+ 'source/talk/session/tunnel/tunnelsessionclient.h',
+ ],
+ 'conditions': [
+ ['OS=="win"', {
+ 'sources': [
+ 'source/talk/session/phone/gdivideorenderer.cc',
+ 'source/talk/session/phone/gdivideorenderer.h',
+ 'source/talk/session/phone/win32devicemanager.cc',
+ 'source/talk/session/phone/win32devicemanager.h',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'sources': [
+ 'source/talk/session/phone/gtkvideorenderer.cc',
+ 'source/talk/session/phone/gtkvideorenderer.cc',
+ 'source/talk/session/phone/gtkvideorenderer.h',
+ 'source/talk/session/phone/libudevsymboltable.cc',
+ 'source/talk/session/phone/libudevsymboltable.h',
+ 'source/talk/session/phone/linuxdevicemanager.cc',
+ 'source/talk/session/phone/linuxdevicemanager.h',
+ 'source/talk/session/phone/v4llookup.cc',
+ 'source/talk/session/phone/v4llookup.h',
+ ],
+ 'include_dirs': [
+ 'source/talk/third_party/libudev',
+ ],
+ 'cflags': [
+ '<!@(pkg-config --cflags gtk+-2.0)',
+ ],
+ }],
+ ['inside_chromium_build==1', {
+ 'dependencies': [
+ '../../third_party/webrtc/modules/modules.gyp:audio_device',
+ '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
+ '../../third_party/webrtc/modules/modules.gyp:video_render_module',
+ '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
+ '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
+ '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+ 'libjingle',
+ ],
+ }, {
+ 'dependencies': [
+ '../../src/modules/modules.gyp:audio_device',
+ '../../src/modules/modules.gyp:video_capture_module',
+ '../../src/modules/modules.gyp:video_render_module',
+ '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '../../src/video_engine/video_engine.gyp:video_engine_core',
+ '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
+ '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+ 'libjingle',
+ ],
+ } ], # inside_chromium_build
+ ], # conditions
+ },
+ # seperate project for app
+ {
+ 'target_name': 'libjingle_app',
+ 'type': '<(library)',
+ 'sources': [
+ 'source/talk/app/webrtc/audiotrackimpl.cc',
+ 'source/talk/app/webrtc/audiotrackimpl.h',
+ 'source/talk/app/webrtc/mediastream.h',
+ 'source/talk/app/webrtc/mediastreamhandler.cc',
+ 'source/talk/app/webrtc/mediastreamhandler.h',
+ 'source/talk/app/webrtc/mediastreamimpl.cc',
+ 'source/talk/app/webrtc/mediastreamimpl.h',
+ 'source/talk/app/webrtc/mediastreamprovider.h',
+ 'source/talk/app/webrtc/mediastreamproxy.cc',
+ 'source/talk/app/webrtc/mediastreamproxy.h',
+ 'source/talk/app/webrtc/mediastreamtrackproxy.cc',
+ 'source/talk/app/webrtc/mediastreamtrackproxy.h',
+ 'source/talk/app/webrtc/mediatrackimpl.h',
+ 'source/talk/app/webrtc/notifierimpl.h',
+ 'source/talk/app/webrtc/peerconnection.h',
+ 'source/talk/app/webrtc/peerconnectionfactoryimpl.cc',
+ 'source/talk/app/webrtc/peerconnectionfactoryimpl.h',
+ 'source/talk/app/webrtc/peerconnectionimpl.cc',
+ 'source/talk/app/webrtc/peerconnectionimpl.h',
+ 'source/talk/app/webrtc/peerconnectionsignaling.cc',
+ 'source/talk/app/webrtc/peerconnectionsignaling.h',
+ 'source/talk/app/webrtc/portallocatorfactory.cc',
+ 'source/talk/app/webrtc/portallocatorfactory.h',
+ 'source/talk/app/webrtc/roaperrorcodes.h',
+ 'source/talk/app/webrtc/roapmessages.cc',
+ 'source/talk/app/webrtc/roapmessages.h',
+ 'source/talk/app/webrtc/roapsession.cc',
+ 'source/talk/app/webrtc/roapsession.h',
+ 'source/talk/app/webrtc/sessiondescriptionprovider.h',
+ 'source/talk/app/webrtc/streamcollectionimpl.h',
+ 'source/talk/app/webrtc/videorendererimpl.cc',
+ 'source/talk/app/webrtc/videotrackimpl.cc',
+ 'source/talk/app/webrtc/videotrackimpl.h',
+ 'source/talk/app/webrtc/webrtcjson.cc',
+ 'source/talk/app/webrtc/webrtcjson.h',
+ 'source/talk/app/webrtc/webrtcsdp.cc',
+ 'source/talk/app/webrtc/webrtcsdp.h',
+ 'source/talk/app/webrtc/webrtcsession.cc',
+ 'source/talk/app/webrtc/webrtcsession.h',
+ 'source/talk/app/webrtc/webrtcsessionobserver.h',
+ ],
+ 'dependencies': [
+ '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+ '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+ ],
+ 'conditions': [
+ ['inside_chromium_build==1', {
+ 'dependencies': [
+ '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
+ '../../third_party/webrtc/modules/modules.gyp:video_render_module',
+ '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
+ '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
+ '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ 'libjingle_p2p',
+ ],
+ }, {
+ 'dependencies': [
+ '../../src/modules/modules.gyp:video_capture_module',
+ '../../src/modules/modules.gyp:video_render_module',
+ '../../src/video_engine/video_engine.gyp:video_engine_core',
+ '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
+ '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ 'libjingle_p2p',
+ ],
+ } ], # inside_chromium_build
+ ], # conditions
+ },
+ ],
+}
diff --git a/trunk/src/LICENSE b/trunk/src/LICENSE
new file mode 100644
index 0000000..4c41b7b
--- /dev/null
+++ b/trunk/src/LICENSE
@@ -0,0 +1,29 @@
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+
+ * Neither the name of Google nor the names of its contributors may
+ be used to endorse or promote products derived from this software
+ without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/src/LICENSE_THIRD_PARTY b/trunk/src/LICENSE_THIRD_PARTY
new file mode 100644
index 0000000..e19708a
--- /dev/null
+++ b/trunk/src/LICENSE_THIRD_PARTY
@@ -0,0 +1,27 @@
+This source tree contains third party source code which is governed by third
+party licenses. This file contains references to files which are under other
+licenses than the one provided in the LICENSE file in the root of the source
+tree.
+
+Files governed by third party licenses:
+common_audio/signal_processing/spl_sqrt_floor.c
+common_audio/signal_processing/spl_sqrt_floor.s
+modules/audio_coding/codecs/G711/main/source/g711.h
+modules/audio_coding/codecs/G711/main/source/g711.c
+modules/audio_coding/codecs/G722/main/source/g722_decode.h
+modules/audio_coding/codecs/G722/main/source/g722_decode.c
+modules/audio_coding/codecs/G722/main/source/g722_encode.h
+modules/audio_coding/codecs/G722/main/source/g722_encode.c
+modules/audio_coding/codecs/iSAC/main/source/fft.c
+modules/audio_device/main/source/Mac/portaudio/pa_memorybarrier.h
+modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.h
+modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.c
+modules/audio_processing/utility/fft4g.c
+modules/audio_processing/aec/aec_rdft.c
+system_wrappers/interface/fix_interlocked_exchange_pointer_windows.h
+system_wrappers/interface/scoped_ptr.h
+system_wrappers/interface/scoped_refptr.h
+system_wrappers/source/condition_variable_windows.cc
+system_wrappers/source/spreadsortlib/constants.hpp
+system_wrappers/source/spreadsortlib/spreadsort.hpp
+system_wrappers/source/thread_windows_set_name.h
diff --git a/trunk/src/PATENTS b/trunk/src/PATENTS
new file mode 100644
index 0000000..190607a
--- /dev/null
+++ b/trunk/src/PATENTS
@@ -0,0 +1,24 @@
+Additional IP Rights Grant (Patents)
+
+"This implementation" means the copyrightable works distributed by
+Google as part of the WebRTC code package.
+
+Google hereby grants to you a perpetual, worldwide, non-exclusive,
+no-charge, irrevocable (except as stated in this section) patent
+license to make, have made, use, offer to sell, sell, import,
+transfer, and otherwise run, modify and propagate the contents of this
+implementation of the WebRTC code package, where such license applies
+only to those patent claims, both currently owned by Google and
+acquired in the future, licensable by Google that are necessarily
+infringed by this implementation of the WebRTC code package. This
+grant does not include claims that would be infringed only as a
+consequence of further modification of this implementation. If you or
+your agent or exclusive licensee institute or order or agree to the
+institution of patent litigation against any entity (including a
+cross-claim or counterclaim in a lawsuit) alleging that this
+implementation of the WebRTC code package or any code incorporated
+within this implementation of the WebRTC code package constitutes
+direct or contributory patent infringement, or inducement of patent
+infringement, then any patent rights granted to you under this License
+for this implementation of the WebRTC code package shall terminate as
+of the date such litigation is filed.
diff --git a/trunk/src/README.chromium b/trunk/src/README.chromium
new file mode 100644
index 0000000..246c13d
--- /dev/null
+++ b/trunk/src/README.chromium
@@ -0,0 +1,13 @@
+Name: WebRTC
+URL: http://www.webrtc.org
+Version: 90
+License: BSD
+License File: LICENSE
+
+Description:
+WebRTC provides real time voice and video processing
+functionality to enable the implementation of
+PeerConnection/MediaStream.
+
+Third party code used in this project is described
+in the file LICENSE_THIRD_PARTY.
diff --git a/trunk/src/build/common.gypi b/trunk/src/build/common.gypi
new file mode 100644
index 0000000..74b6003
--- /dev/null
+++ b/trunk/src/build/common.gypi
@@ -0,0 +1,180 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# This file contains common settings for building WebRTC components.
+
+{
+ 'variables': {
+ # These variables need to be nested in order to use them in a conditions
+ # block to set other variables.
+ 'variables': {
+ # This will be set to zero in the supplement.gypi triggered by a gclient
+ # hook in the standalone build.
+ 'build_with_chromium%': 1,
+ },
+
+ 'build_with_chromium%': '<(build_with_chromium)',
+
+ # The Chromium common.gypi we use treats all gyp files without
+ # chromium_code==1 as third party code. This disables many of the
+ # preferred warning settings.
+ #
+ # We can set this here to have WebRTC code treated as Chromium code. Our
+ # third party code will still have the reduced warning settings.
+ 'chromium_code': 1,
+
+ # Adds video support to dependencies shared by voice and video engine.
+ # This should normally be enabled; the intended use is to disable only
+ # when building voice engine exclusively.
+ 'enable_video%': 1,
+
+ # Selects fixed-point code where possible.
+ # TODO(andrew): we'd like to set this based on the target OS/architecture.
+ 'prefer_fixed_point%': 0,
+
+ # Enable data logging. Produces text files with data logged within engines
+ # which can be easily parsed for offline processing.
+ 'enable_data_logging%': 0,
+
+ # Disable these to not build components which can be externally provided.
+ 'build_libjpeg%': 1,
+ 'build_libyuv%': 1,
+
+ 'conditions': [
+ ['OS=="win"', {
+ # TODO(andrew, perkj): does this need to be here?
+ # Path needed to build Direct Show base classes on Windows.
+ # The code is included in the Windows SDK.
+ 'direct_show_base_classes':
+ 'C:/Program Files/Microsoft SDKs/Windows/v7.1/Samples/multimedia/directshow/baseclasses/',
+ }],
+ ['build_with_chromium==1', {
+ # Exclude pulse audio on Chromium since its prerequisites don't require
+ # pulse audio.
+ 'include_pulse_audio%': 0,
+
+ # Exclude internal ADM since Chromium uses its own IO handling.
+ 'include_internal_audio_device%': 0,
+
+ # Exclude internal VCM in Chromium build.
+ 'include_internal_video_capture%': 0,
+
+ # Exclude internal video render module in Chromium build.
+ 'include_internal_video_render%': 0,
+
+ # Disable the use of protocol buffers in production code.
+ 'enable_protobuf%': 0,
+
+ 'webrtc_root%': '<(DEPTH)/third_party/webrtc',
+ }, {
+ # Settings for the standalone (not-in-Chromium) build.
+
+ 'include_pulse_audio%': 1,
+
+ 'include_internal_audio_device%': 1,
+
+ 'include_internal_video_capture%': 1,
+
+ 'include_internal_video_render%': 1,
+
+ 'enable_protobuf%': 1,
+
+ 'webrtc_root%': '<(DEPTH)/src',
+
+ 'conditions': [
+ ['OS=="mac"', {
+ # TODO(andrew): clang is the default on Mac. For now, disable the
+ # Chrome plugins, which causes a flood of chromium-style warnings.
+ # Investigate enabling the plugins:
+ # http://code.google.com/p/webrtc/issues/detail?id=163
+ 'clang_use_chrome_plugins%': 0,
+ }],
+ ],
+ }],
+ ], # conditions
+ },
+ 'target_defaults': {
+ 'include_dirs': [
+ '..','../..', # common_types.h, typedefs.h
+ ],
+ 'conditions': [
+ ['build_with_chromium==1', {
+ 'defines': [
+ # Changes settings for Chromium build.
+ 'WEBRTC_CHROMIUM_BUILD',
+ ],
+ }, {
+ 'conditions': [
+ ['os_posix==1', {
+ 'cflags': [
+ '-Wextra',
+ # We need to repeat some flags from Chromium's common.gypi here
+ # that get overridden by -Wextra.
+ '-Wno-unused-parameter',
+ '-Wno-missing-field-initializers',
+ ],
+ 'cflags_cc': [
+ # This is enabled for clang; enable for gcc as well.
+ '-Woverloaded-virtual',
+ ],
+ }],
+ ],
+ }],
+ ['OS=="linux"', {
+ 'defines': [
+ 'WEBRTC_TARGET_PC',
+ 'WEBRTC_LINUX',
+ 'WEBRTC_THREAD_RR',
+ # TODO(andrew): can we select this automatically?
+ # Define this if the Linux system does not support CLOCK_MONOTONIC.
+ #'WEBRTC_CLOCK_TYPE_REALTIME',
+ ],
+ }],
+ ['OS=="mac"', {
+ # TODO(andrew): what about PowerPC?
+ # Setup for Intel
+ 'defines': [
+ 'WEBRTC_TARGET_MAC_INTEL',
+ 'WEBRTC_MAC_INTEL',
+ 'WEBRTC_MAC',
+ 'WEBRTC_THREAD_RR',
+ 'WEBRTC_CLOCK_TYPE_REALTIME',
+ ],
+ }],
+ ['OS=="win"', {
+ 'defines': [
+ 'WEBRTC_TARGET_PC',
+ ],
+ # TODO(andrew): remove this block when possible.
+ # 4389: Signed/unsigned mismatch.
+ # 4373: MSVC legacy warning for ignoring const / volatile in
+ # signatures. TODO(phoglund): get rid of 4373 supression when
+ # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
+ 'msvs_disabled_warnings': [4389, 4373],
+ }],
+ ['OS=="android"', {
+ 'defines': [
+ 'WEBRTC_TARGET_PC',
+ 'WEBRTC_LINUX',
+ # TODO(leozwang): Investigate CLOCK_REALTIME and CLOCK_MONOTONIC
+ # support on Android. Keep WEBRTC_CLOCK_TYPE_REALTIME for now,
+ # remove it after I verify that CLOCK_MONOTONIC is fully functional
+ # with condition and event functions in system_wrappers.
+ 'WEBRTC_CLOCK_TYPE_REALTIME',
+ 'WEBRTC_THREAD_RR',
+ 'WEBRTC_ANDROID',
+ 'WEBRTC_ARM_INLINE_CALLS',
+ # TODO(leozwang): move WEBRTC_ARCH_ARM to typedefs.h.
+ 'WEBRTC_ARCH_ARM',
+ 'WEBRTC_ANDROID_OPENSLES',
+ ],
+ }],
+ ], # conditions
+ }, # target_defaults
+}
+
diff --git a/trunk/src/build/merge_libs.gyp b/trunk/src/build/merge_libs.gyp
new file mode 100644
index 0000000..d23bf11
--- /dev/null
+++ b/trunk/src/build/merge_libs.gyp
@@ -0,0 +1,43 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ 'common.gypi', ],
+ 'targets': [
+ {
+ 'target_name': 'no_op',
+ 'type': 'executable',
+ 'dependencies': [
+ '../video_engine/video_engine.gyp:video_engine_core',
+ ],
+ 'sources': [ 'no_op.cc', ],
+ },
+ {
+ 'target_name': 'merged_lib',
+ 'type': 'none',
+ 'dependencies': [
+ 'no_op',
+ ],
+ 'actions': [
+ {
+ 'variables': {
+ 'output_lib_name': 'webrtc',
+ 'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)_<(OS)_<(target_arch)_<(CONFIGURATION_NAME)<(STATIC_LIB_SUFFIX)',
+ },
+ 'action_name': 'merge_libs',
+ 'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op<(EXECUTABLE_SUFFIX)'],
+ 'outputs': ['<(output_lib)'],
+ 'action': ['python',
+ 'merge_libs.py',
+ '<(PRODUCT_DIR)',
+ '<(output_lib)',],
+ },
+ ],
+ },
+ ],
+}
diff --git a/trunk/src/build/merge_libs.py b/trunk/src/build/merge_libs.py
new file mode 100644
index 0000000..31c5efb
--- /dev/null
+++ b/trunk/src/build/merge_libs.py
@@ -0,0 +1,58 @@
+#!/usr/bin/env python
+
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Searches for libraries and/or object files on the specified path and
+# merges them into a single library.
+
+import subprocess
+import sys
+
+if __name__ == '__main__':
+ if len(sys.argv) != 3:
+ sys.stderr.write('Usage: ' + sys.argv[0] + ' <search_path> <output_lib>\n')
+ sys.exit(2)
+
+ search_path = sys.argv[1]
+ output_lib = sys.argv[2]
+
+ from subprocess import call, PIPE
+ if sys.platform.startswith('linux'):
+ call(["rm -f " + output_lib], shell=True)
+ call(["rm -rf " + search_path + "/obj.target/*do_not_use*"], shell=True)
+ call(["ar crs " + output_lib + " $(find " + search_path +
+ "/obj.target -name *\.o)"], shell=True)
+ call(["ar crs " + output_lib + " $(find " + search_path +
+ "/obj/gen -name *\.o)"], shell=True)
+
+ elif sys.platform == 'darwin':
+ call(["rm -f " + output_lib], shell=True)
+ call(["rm -f " + search_path + "/*do_not_use*"], shell=True)
+ call(["libtool -static -v -o " + output_lib + " " + search_path + "/*.a"],
+ shell=True)
+
+ elif sys.platform == 'win32':
+ # We need to execute a batch file to set some environment variables for the
+ # lib command. VS 8 uses vsvars.bat and VS 9 uses vsvars32.bat. It's
+ # required that at least one of them is in the system PATH. We try both and
+ # suppress stderr and stdout to fail silently.
+ call(["vsvars.bat"], stderr=PIPE, stdout=PIPE, shell=True)
+ call(["vsvars32.bat"], stderr=PIPE, stdout=PIPE, shell=True)
+ call(["del " + output_lib], shell=True)
+ call(["del /F /S /Q " + search_path + "/lib/*do_not_use*"],
+ shell=True)
+ call(["lib /OUT:" + output_lib + " " + search_path + "/lib/*.lib"],
+ shell=True)
+
+ else:
+ sys.stderr.write('Platform not supported: %r\n\n' % sys.platform)
+ sys.exit(1)
+
+ sys.exit(0)
+
diff --git a/trunk/src/build/no_op.cc b/trunk/src/build/no_op.cc
new file mode 100644
index 0000000..4cce7f6
--- /dev/null
+++ b/trunk/src/build/no_op.cc
@@ -0,0 +1,3 @@
+int main() {
+ return 0;
+}
diff --git a/trunk/src/build/protoc.gypi b/trunk/src/build/protoc.gypi
new file mode 100644
index 0000000..70bf71e
--- /dev/null
+++ b/trunk/src/build/protoc.gypi
@@ -0,0 +1,97 @@
+# Copyright (c) 2011 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+# It was necessary to copy this file to WebRTC, because the path to
+# build/common.gypi is different for the standalone and Chromium builds. Gyp
+# doesn't permit conditional inclusion or variable expansion in include paths.
+# http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
+
+# This file is meant to be included into a target to provide a rule
+# to invoke protoc in a consistent manner.
+#
+# To use this, create a gyp target with the following form:
+# {
+# 'target_name': 'my_proto_lib',
+# 'type': 'static_library',
+# 'sources': [
+# 'foo.proto',
+# 'bar.proto',
+# ],
+# 'variables': {
+# # Optional, see below: 'proto_in_dir': '.'
+# 'proto_out_dir': 'dir/for/my_proto_lib'
+# },
+# 'includes': ['path/to/this/gypi/file'],
+# }
+# If necessary, you may add normal .cc files to the sources list or other gyp
+# dependencies. The proto headers are guaranteed to be generated before any
+# source files, even within this target, are compiled.
+#
+# The 'proto_in_dir' variable must be the relative path to the
+# directory containing the .proto files. If left out, it defaults to '.'.
+#
+# The 'proto_out_dir' variable specifies the path suffix that output
+# files are generated under. Targets that gyp-depend on my_proto_lib
+# will be able to include the resulting proto headers with an include
+# like:
+# #include "dir/for/my_proto_lib/foo.pb.h"
+#
+# Implementation notes:
+# A proto_out_dir of foo/bar produces
+# <(SHARED_INTERMEDIATE_DIR)/protoc_out/foo/bar/{file1,file2}.pb.{cc,h}
+# <(SHARED_INTERMEDIATE_DIR)/pyproto/foo/bar/{file1,file2}_pb2.py
+
+{
+ 'variables': {
+ 'protoc': '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
+ 'cc_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out/<(proto_out_dir)',
+ 'py_dir': '<(PRODUCT_DIR)/pyproto/<(proto_out_dir)',
+ 'proto_in_dir%': '.',
+ },
+ 'rules': [
+ {
+ 'rule_name': 'genproto',
+ 'extension': 'proto',
+ 'inputs': [
+ '<(protoc)',
+ ],
+ 'outputs': [
+ '<(py_dir)/<(RULE_INPUT_ROOT)_pb2.py',
+ '<(cc_dir)/<(RULE_INPUT_ROOT).pb.cc',
+ '<(cc_dir)/<(RULE_INPUT_ROOT).pb.h',
+ ],
+ 'action': [
+ '<(protoc)',
+ '--proto_path=<(proto_in_dir)',
+ # Naively you'd use <(RULE_INPUT_PATH) here, but protoc requires
+ # --proto_path is a strict prefix of the path given as an argument.
+ '<(proto_in_dir)/<(RULE_INPUT_ROOT)<(RULE_INPUT_EXT)',
+ '--cpp_out=<(cc_dir)',
+ '--python_out=<(py_dir)',
+ ],
+ 'message': 'Generating C++ and Python code from <(RULE_INPUT_PATH)',
+ 'process_outputs_as_sources': 1,
+ },
+ ],
+ 'dependencies': [
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+ ],
+ 'include_dirs': [
+ '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
+ ]
+ },
+ 'export_dependent_settings': [
+ # The generated headers reference headers within protobuf_lite,
+ # so dependencies must be able to find those headers too.
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+ ],
+ # This target exports a hard dependency because it generates header
+ # files.
+ 'hard_dependency': 1,
+}
diff --git a/trunk/src/common_audio/OWNERS b/trunk/src/common_audio/OWNERS
new file mode 100644
index 0000000..84582f2
--- /dev/null
+++ b/trunk/src/common_audio/OWNERS
@@ -0,0 +1,4 @@
+bjornv@webrtc.org
+tina.legrand@webrtc.org
+jan.skoglund@webrtc.org
+andrew@webrtc.org
diff --git a/trunk/src/common_audio/common_audio.gyp b/trunk/src/common_audio/common_audio.gyp
new file mode 100644
index 0000000..3d3da3f
--- /dev/null
+++ b/trunk/src/common_audio/common_audio.gyp
@@ -0,0 +1,16 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ 'signal_processing/signal_processing.gypi',
+ 'resampler/resampler.gypi',
+ 'vad/vad.gypi',
+ ],
+}
diff --git a/trunk/src/common_audio/resampler/Android.mk b/trunk/src/common_audio/resampler/Android.mk
new file mode 100644
index 0000000..b1d630a
--- /dev/null
+++ b/trunk/src/common_audio/resampler/Android.mk
@@ -0,0 +1,47 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_resampler
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := resampler.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/resampler/include/resampler.h b/trunk/src/common_audio/resampler/include/resampler.h
new file mode 100644
index 0000000..38e6bd3
--- /dev/null
+++ b/trunk/src/common_audio/resampler/include/resampler.h
@@ -0,0 +1,116 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
+#define WEBRTC_RESAMPLER_RESAMPLER_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+// TODO(andrew): the implementation depends on the exact values of this enum.
+// It should be rewritten in a less fragile way.
+enum ResamplerType
+{
+ // 4 MSB = Number of channels
+ // 4 LSB = Synchronous or asynchronous
+
+ kResamplerSynchronous = 0x10,
+ kResamplerAsynchronous = 0x11,
+ kResamplerSynchronousStereo = 0x20,
+ kResamplerAsynchronousStereo = 0x21,
+ kResamplerInvalid = 0xff
+};
+
+// TODO(andrew): doesn't need to be part of the interface.
+enum ResamplerMode
+{
+ kResamplerMode1To1,
+ kResamplerMode1To2,
+ kResamplerMode1To3,
+ kResamplerMode1To4,
+ kResamplerMode1To6,
+ kResamplerMode1To12,
+ kResamplerMode2To3,
+ kResamplerMode2To11,
+ kResamplerMode4To11,
+ kResamplerMode8To11,
+ kResamplerMode11To16,
+ kResamplerMode11To32,
+ kResamplerMode2To1,
+ kResamplerMode3To1,
+ kResamplerMode4To1,
+ kResamplerMode6To1,
+ kResamplerMode12To1,
+ kResamplerMode3To2,
+ kResamplerMode11To2,
+ kResamplerMode11To4,
+ kResamplerMode11To8
+};
+
+class Resampler
+{
+
+public:
+ Resampler();
+ // TODO(andrew): use an init function instead.
+ Resampler(int inFreq, int outFreq, ResamplerType type);
+ ~Resampler();
+
+ // Reset all states
+ int Reset(int inFreq, int outFreq, ResamplerType type);
+
+ // Reset all states if any parameter has changed
+ int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
+
+ // Synchronous resampling, all output samples are written to samplesOut
+ int Push(const WebRtc_Word16* samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
+ int maxLen, int &outLen);
+
+ // Asynchronous resampling, input
+ int Insert(WebRtc_Word16* samplesIn, int lengthIn);
+
+ // Asynchronous resampling output, remaining samples are buffered
+ int Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen);
+
+private:
+ // Generic pointers since we don't know what states we'll need
+ void* state1_;
+ void* state2_;
+ void* state3_;
+
+ // Storage if needed
+ WebRtc_Word16* in_buffer_;
+ WebRtc_Word16* out_buffer_;
+ int in_buffer_size_;
+ int out_buffer_size_;
+ int in_buffer_size_max_;
+ int out_buffer_size_max_;
+
+ // State
+ int my_in_frequency_khz_;
+ int my_out_frequency_khz_;
+ ResamplerMode my_mode_;
+ ResamplerType my_type_;
+
+ // Extra instance for stereo
+ Resampler* slave_left_;
+ Resampler* slave_right_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
diff --git a/trunk/src/common_audio/resampler/resampler.cc b/trunk/src/common_audio/resampler/resampler.cc
new file mode 100644
index 0000000..2db27b1
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler.cc
@@ -0,0 +1,1084 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "signal_processing_library.h"
+#include "resampler.h"
+
+
+namespace webrtc
+{
+
+Resampler::Resampler()
+{
+ state1_ = NULL;
+ state2_ = NULL;
+ state3_ = NULL;
+ in_buffer_ = NULL;
+ out_buffer_ = NULL;
+ in_buffer_size_ = 0;
+ out_buffer_size_ = 0;
+ in_buffer_size_max_ = 0;
+ out_buffer_size_max_ = 0;
+ // we need a reset before we will work
+ my_in_frequency_khz_ = 0;
+ my_out_frequency_khz_ = 0;
+ my_mode_ = kResamplerMode1To1;
+ my_type_ = kResamplerInvalid;
+ slave_left_ = NULL;
+ slave_right_ = NULL;
+}
+
+Resampler::Resampler(int inFreq, int outFreq, ResamplerType type)
+{
+ state1_ = NULL;
+ state2_ = NULL;
+ state3_ = NULL;
+ in_buffer_ = NULL;
+ out_buffer_ = NULL;
+ in_buffer_size_ = 0;
+ out_buffer_size_ = 0;
+ in_buffer_size_max_ = 0;
+ out_buffer_size_max_ = 0;
+ // we need a reset before we will work
+ my_in_frequency_khz_ = 0;
+ my_out_frequency_khz_ = 0;
+ my_mode_ = kResamplerMode1To1;
+ my_type_ = kResamplerInvalid;
+ slave_left_ = NULL;
+ slave_right_ = NULL;
+
+ Reset(inFreq, outFreq, type);
+}
+
+Resampler::~Resampler()
+{
+ if (state1_)
+ {
+ free(state1_);
+ }
+ if (state2_)
+ {
+ free(state2_);
+ }
+ if (state3_)
+ {
+ free(state3_);
+ }
+ if (in_buffer_)
+ {
+ free(in_buffer_);
+ }
+ if (out_buffer_)
+ {
+ free(out_buffer_);
+ }
+ if (slave_left_)
+ {
+ delete slave_left_;
+ }
+ if (slave_right_)
+ {
+ delete slave_right_;
+ }
+}
+
+int Resampler::ResetIfNeeded(int inFreq, int outFreq, ResamplerType type)
+{
+ int tmpInFreq_kHz = inFreq / 1000;
+ int tmpOutFreq_kHz = outFreq / 1000;
+
+ if ((tmpInFreq_kHz != my_in_frequency_khz_) || (tmpOutFreq_kHz != my_out_frequency_khz_)
+ || (type != my_type_))
+ {
+ return Reset(inFreq, outFreq, type);
+ } else
+ {
+ return 0;
+ }
+}
+
+int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
+{
+
+ if (state1_)
+ {
+ free(state1_);
+ state1_ = NULL;
+ }
+ if (state2_)
+ {
+ free(state2_);
+ state2_ = NULL;
+ }
+ if (state3_)
+ {
+ free(state3_);
+ state3_ = NULL;
+ }
+ if (in_buffer_)
+ {
+ free(in_buffer_);
+ in_buffer_ = NULL;
+ }
+ if (out_buffer_)
+ {
+ free(out_buffer_);
+ out_buffer_ = NULL;
+ }
+ if (slave_left_)
+ {
+ delete slave_left_;
+ slave_left_ = NULL;
+ }
+ if (slave_right_)
+ {
+ delete slave_right_;
+ slave_right_ = NULL;
+ }
+
+ in_buffer_size_ = 0;
+ out_buffer_size_ = 0;
+ in_buffer_size_max_ = 0;
+ out_buffer_size_max_ = 0;
+
+ // This might be overridden if parameters are not accepted.
+ my_type_ = type;
+
+ // Start with a math exercise, Euclid's algorithm to find the gcd:
+
+ int a = inFreq;
+ int b = outFreq;
+ int c = a % b;
+ while (c != 0)
+ {
+ a = b;
+ b = c;
+ c = a % b;
+ }
+ // b is now the gcd;
+
+ // We need to track what domain we're in.
+ my_in_frequency_khz_ = inFreq / 1000;
+ my_out_frequency_khz_ = outFreq / 1000;
+
+ // Scale with GCD
+ inFreq = inFreq / b;
+ outFreq = outFreq / b;
+
+ // Do we need stereo?
+ if ((my_type_ & 0xf0) == 0x20)
+ {
+ // Change type to mono
+ type = static_cast<ResamplerType>(
+ ((static_cast<int>(type) & 0x0f) + 0x10));
+ slave_left_ = new Resampler(inFreq, outFreq, type);
+ slave_right_ = new Resampler(inFreq, outFreq, type);
+ }
+
+ if (inFreq == outFreq)
+ {
+ my_mode_ = kResamplerMode1To1;
+ } else if (inFreq == 1)
+ {
+ switch (outFreq)
+ {
+ case 2:
+ my_mode_ = kResamplerMode1To2;
+ break;
+ case 3:
+ my_mode_ = kResamplerMode1To3;
+ break;
+ case 4:
+ my_mode_ = kResamplerMode1To4;
+ break;
+ case 6:
+ my_mode_ = kResamplerMode1To6;
+ break;
+ case 12:
+ my_mode_ = kResamplerMode1To12;
+ break;
+ default:
+ my_type_ = kResamplerInvalid;
+ return -1;
+ }
+ } else if (outFreq == 1)
+ {
+ switch (inFreq)
+ {
+ case 2:
+ my_mode_ = kResamplerMode2To1;
+ break;
+ case 3:
+ my_mode_ = kResamplerMode3To1;
+ break;
+ case 4:
+ my_mode_ = kResamplerMode4To1;
+ break;
+ case 6:
+ my_mode_ = kResamplerMode6To1;
+ break;
+ case 12:
+ my_mode_ = kResamplerMode12To1;
+ break;
+ default:
+ my_type_ = kResamplerInvalid;
+ return -1;
+ }
+ } else if ((inFreq == 2) && (outFreq == 3))
+ {
+ my_mode_ = kResamplerMode2To3;
+ } else if ((inFreq == 2) && (outFreq == 11))
+ {
+ my_mode_ = kResamplerMode2To11;
+ } else if ((inFreq == 4) && (outFreq == 11))
+ {
+ my_mode_ = kResamplerMode4To11;
+ } else if ((inFreq == 8) && (outFreq == 11))
+ {
+ my_mode_ = kResamplerMode8To11;
+ } else if ((inFreq == 3) && (outFreq == 2))
+ {
+ my_mode_ = kResamplerMode3To2;
+ } else if ((inFreq == 11) && (outFreq == 2))
+ {
+ my_mode_ = kResamplerMode11To2;
+ } else if ((inFreq == 11) && (outFreq == 4))
+ {
+ my_mode_ = kResamplerMode11To4;
+ } else if ((inFreq == 11) && (outFreq == 16))
+ {
+ my_mode_ = kResamplerMode11To16;
+ } else if ((inFreq == 11) && (outFreq == 32))
+ {
+ my_mode_ = kResamplerMode11To32;
+ } else if ((inFreq == 11) && (outFreq == 8))
+ {
+ my_mode_ = kResamplerMode11To8;
+ } else
+ {
+ my_type_ = kResamplerInvalid;
+ return -1;
+ }
+
+ // Now create the states we need
+ switch (my_mode_)
+ {
+ case kResamplerMode1To1:
+ // No state needed;
+ break;
+ case kResamplerMode1To2:
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode1To3:
+ state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+ WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
+ break;
+ case kResamplerMode1To4:
+ // 1:2
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:4
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode1To6:
+ // 1:2
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:6
+ state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+ WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
+ break;
+ case kResamplerMode1To12:
+ // 1:2
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:4
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ // 4:12
+ state3_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+ WebRtcSpl_ResetResample16khzTo48khz(
+ (WebRtcSpl_State16khzTo48khz*) state3_);
+ break;
+ case kResamplerMode2To3:
+ // 2:6
+ state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+ WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
+ // 6:3
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode2To11:
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+ state2_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
+ WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state2_);
+ break;
+ case kResamplerMode4To11:
+ state1_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
+ WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state1_);
+ break;
+ case kResamplerMode8To11:
+ state1_ = malloc(sizeof(WebRtcSpl_State16khzTo22khz));
+ WebRtcSpl_ResetResample16khzTo22khz((WebRtcSpl_State16khzTo22khz *)state1_);
+ break;
+ case kResamplerMode11To16:
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+ state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+ WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
+ break;
+ case kResamplerMode11To32:
+ // 11 -> 22
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+ // 22 -> 16
+ state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+ WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
+
+ // 16 -> 32
+ state3_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
+
+ break;
+ case kResamplerMode2To1:
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode3To1:
+ state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+ WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
+ break;
+ case kResamplerMode4To1:
+ // 4:2
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:1
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode6To1:
+ // 6:2
+ state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+ WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
+ // 2:1
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode12To1:
+ // 12:4
+ state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+ WebRtcSpl_ResetResample48khzTo16khz(
+ (WebRtcSpl_State48khzTo16khz*) state1_);
+ // 4:2
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:1
+ state3_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
+ case kResamplerMode3To2:
+ // 3:6
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 6:2
+ state2_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+ WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state2_);
+ break;
+ case kResamplerMode11To2:
+ state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
+ WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
+
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+
+ break;
+ case kResamplerMode11To4:
+ state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
+ WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
+ break;
+ case kResamplerMode11To8:
+ state1_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+ WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state1_);
+ break;
+
+ }
+
+ return 0;
+}
+
+// Synchronous resampling, all output samples are written to samplesOut
+int Resampler::Push(const WebRtc_Word16 * samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
+ int maxLen, int &outLen)
+{
+ // Check that the resampler is not in asynchronous mode
+ if (my_type_ & 0x0f)
+ {
+ return -1;
+ }
+
+ // Do we have a stereo signal?
+ if ((my_type_ & 0xf0) == 0x20)
+ {
+
+ // Split up the signal and call the slave object for each channel
+
+ WebRtc_Word16* left = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
+ WebRtc_Word16* right = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
+ WebRtc_Word16* out_left = (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
+ WebRtc_Word16* out_right =
+ (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
+ int res = 0;
+ for (int i = 0; i < lengthIn; i += 2)
+ {
+ left[i >> 1] = samplesIn[i];
+ right[i >> 1] = samplesIn[i + 1];
+ }
+
+ // It's OK to overwrite the local parameter, since it's just a copy
+ lengthIn = lengthIn / 2;
+
+ int actualOutLen_left = 0;
+ int actualOutLen_right = 0;
+ // Do resampling for right channel
+ res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2, actualOutLen_left);
+ res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2, actualOutLen_right);
+ if (res || (actualOutLen_left != actualOutLen_right))
+ {
+ free(left);
+ free(right);
+ free(out_left);
+ free(out_right);
+ return -1;
+ }
+
+ // Reassemble the signal
+ for (int i = 0; i < actualOutLen_left; i++)
+ {
+ samplesOut[i * 2] = out_left[i];
+ samplesOut[i * 2 + 1] = out_right[i];
+ }
+ outLen = 2 * actualOutLen_left;
+
+ free(left);
+ free(right);
+ free(out_left);
+ free(out_right);
+
+ return 0;
+ }
+
+ // Containers for temp samples
+ WebRtc_Word16* tmp;
+ WebRtc_Word16* tmp_2;
+ // tmp data for resampling routines
+ WebRtc_Word32* tmp_mem;
+
+ switch (my_mode_)
+ {
+ case kResamplerMode1To1:
+ memcpy(samplesOut, samplesIn, lengthIn * sizeof(WebRtc_Word16));
+ outLen = lengthIn;
+ break;
+ case kResamplerMode1To2:
+ if (maxLen < (lengthIn * 2))
+ {
+ return -1;
+ }
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+ outLen = lengthIn * 2;
+ return 0;
+ case kResamplerMode1To3:
+
+ // We can only handle blocks of 160 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 160) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < (lengthIn * 3))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 160)
+ {
+ WebRtcSpl_Resample16khzTo48khz(samplesIn + i, samplesOut + i * 3,
+ (WebRtcSpl_State16khzTo48khz *)state1_,
+ tmp_mem);
+ }
+ outLen = lengthIn * 3;
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode1To4:
+ if (maxLen < (lengthIn * 4))
+ {
+ return -1;
+ }
+
+ tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+ // 1:2
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+ // 2:4
+ WebRtcSpl_UpsampleBy2(tmp, lengthIn * 2, samplesOut, (WebRtc_Word32*)state2_);
+ outLen = lengthIn * 4;
+ free(tmp);
+ return 0;
+ case kResamplerMode1To6:
+ // We can only handle blocks of 80 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 80) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < (lengthIn * 6))
+ {
+ return -1;
+ }
+
+ //1:2
+
+ tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+ outLen = lengthIn * 2;
+
+ for (int i = 0; i < outLen; i += 160)
+ {
+ WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
+ (WebRtcSpl_State16khzTo48khz *)state2_,
+ tmp_mem);
+ }
+ outLen = outLen * 3;
+ free(tmp_mem);
+ free(tmp);
+
+ return 0;
+ case kResamplerMode1To12:
+ // We can only handle blocks of 40 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 40) != 0) {
+ return -1;
+ }
+ if (maxLen < (lengthIn * 12)) {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*) malloc(336 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*) malloc(sizeof(WebRtc_Word16) * 4 * lengthIn);
+ //1:2
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut,
+ (WebRtc_Word32*) state1_);
+ outLen = lengthIn * 2;
+ //2:4
+ WebRtcSpl_UpsampleBy2(samplesOut, outLen, tmp, (WebRtc_Word32*) state2_);
+ outLen = outLen * 2;
+ // 4:12
+ for (int i = 0; i < outLen; i += 160) {
+ // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
+ // as input and outputs a resampled block of 480 samples. The
+ // data is now actually in 32 kHz sampling rate, despite the
+ // function name, and with a resampling factor of three becomes
+ // 96 kHz.
+ WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
+ (WebRtcSpl_State16khzTo48khz*) state3_,
+ tmp_mem);
+ }
+ outLen = outLen * 3;
+ free(tmp_mem);
+ free(tmp);
+
+ return 0;
+ case kResamplerMode2To3:
+ if (maxLen < (lengthIn * 3 / 2))
+ {
+ return -1;
+ }
+ // 2:6
+ // We can only handle blocks of 160 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 160) != 0)
+ {
+ return -1;
+ }
+ tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 3));
+ tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+ for (int i = 0; i < lengthIn; i += 160)
+ {
+ WebRtcSpl_Resample16khzTo48khz(samplesIn + i, tmp + i * 3,
+ (WebRtcSpl_State16khzTo48khz *)state1_,
+ tmp_mem);
+ }
+ lengthIn = lengthIn * 3;
+ // 6:3
+ WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
+ outLen = lengthIn / 2;
+ free(tmp);
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode2To11:
+
+ // We can only handle blocks of 80 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 80) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 11) / 2))
+ {
+ return -1;
+ }
+ tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+ // 1:2
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+ lengthIn *= 2;
+
+ tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 80)
+ {
+ WebRtcSpl_Resample8khzTo22khz(tmp + i, samplesOut + (i * 11) / 4,
+ (WebRtcSpl_State8khzTo22khz *)state2_,
+ tmp_mem);
+ }
+ outLen = (lengthIn * 11) / 4;
+ free(tmp_mem);
+ free(tmp);
+ return 0;
+ case kResamplerMode4To11:
+
+ // We can only handle blocks of 80 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 80) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 11) / 4))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 80)
+ {
+ WebRtcSpl_Resample8khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 4,
+ (WebRtcSpl_State8khzTo22khz *)state1_,
+ tmp_mem);
+ }
+ outLen = (lengthIn * 11) / 4;
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode8To11:
+ // We can only handle blocks of 160 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 160) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 11) / 8))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(88 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 160)
+ {
+ WebRtcSpl_Resample16khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 8,
+ (WebRtcSpl_State16khzTo22khz *)state1_,
+ tmp_mem);
+ }
+ outLen = (lengthIn * 11) / 8;
+ free(tmp_mem);
+ return 0;
+
+ case kResamplerMode11To16:
+ // We can only handle blocks of 110 samples
+ if ((lengthIn % 110) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 16) / 11))
+ {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
+
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+
+ for (int i = 0; i < (lengthIn * 2); i += 220)
+ {
+ WebRtcSpl_Resample22khzTo16khz(tmp + i, samplesOut + (i / 220) * 160,
+ (WebRtcSpl_State22khzTo16khz *)state2_,
+ tmp_mem);
+ }
+
+ outLen = (lengthIn * 16) / 11;
+
+ free(tmp_mem);
+ free(tmp);
+ return 0;
+
+ case kResamplerMode11To32:
+
+ // We can only handle blocks of 110 samples
+ if ((lengthIn % 110) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 32) / 11))
+ {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
+
+ // 11 -> 22 kHz in samplesOut
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+
+ // 22 -> 16 in tmp
+ for (int i = 0; i < (lengthIn * 2); i += 220)
+ {
+ WebRtcSpl_Resample22khzTo16khz(samplesOut + i, tmp + (i / 220) * 160,
+ (WebRtcSpl_State22khzTo16khz *)state2_,
+ tmp_mem);
+ }
+
+ // 16 -> 32 in samplesOut
+ WebRtcSpl_UpsampleBy2(tmp, (lengthIn * 16) / 11, samplesOut,
+ (WebRtc_Word32*)state3_);
+
+ outLen = (lengthIn * 32) / 11;
+
+ free(tmp_mem);
+ free(tmp);
+ return 0;
+
+ case kResamplerMode2To1:
+ if (maxLen < (lengthIn / 2))
+ {
+ return -1;
+ }
+ WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+ outLen = lengthIn / 2;
+ return 0;
+ case kResamplerMode3To1:
+ // We can only handle blocks of 480 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 480) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < (lengthIn / 3))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 480)
+ {
+ WebRtcSpl_Resample48khzTo16khz(samplesIn + i, samplesOut + i / 3,
+ (WebRtcSpl_State48khzTo16khz *)state1_,
+ tmp_mem);
+ }
+ outLen = lengthIn / 3;
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode4To1:
+ if (maxLen < (lengthIn / 4))
+ {
+ return -1;
+ }
+ tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * lengthIn / 2);
+ // 4:2
+ WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+ // 2:1
+ WebRtcSpl_DownsampleBy2(tmp, lengthIn / 2, samplesOut, (WebRtc_Word32*)state2_);
+ outLen = lengthIn / 4;
+ free(tmp);
+ return 0;
+
+ case kResamplerMode6To1:
+ // We can only handle blocks of 480 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 480) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < (lengthIn / 6))
+ {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
+
+ for (int i = 0; i < lengthIn; i += 480)
+ {
+ WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
+ (WebRtcSpl_State48khzTo16khz *)state1_,
+ tmp_mem);
+ }
+ outLen = lengthIn / 3;
+ free(tmp_mem);
+ WebRtcSpl_DownsampleBy2(tmp, outLen, samplesOut, (WebRtc_Word32*)state2_);
+ free(tmp);
+ outLen = outLen / 2;
+ return 0;
+ case kResamplerMode12To1:
+ // We can only handle blocks of 480 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 480) != 0) {
+ return -1;
+ }
+ if (maxLen < (lengthIn / 12)) {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*) malloc(496 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
+ tmp_2 = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 6);
+ // 12:4
+ for (int i = 0; i < lengthIn; i += 480) {
+ // WebRtcSpl_Resample48khzTo16khz() takes a block of 480 samples
+ // as input and outputs a resampled block of 160 samples. The
+ // data is now actually in 96 kHz sampling rate, despite the
+ // function name, and with a resampling factor of 1/3 becomes
+ // 32 kHz.
+ WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
+ (WebRtcSpl_State48khzTo16khz*) state1_,
+ tmp_mem);
+ }
+ outLen = lengthIn / 3;
+ free(tmp_mem);
+ // 4:2
+ WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2,
+ (WebRtc_Word32*) state2_);
+ outLen = outLen / 2;
+ free(tmp);
+ // 2:1
+ WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
+ (WebRtc_Word32*) state3_);
+ free(tmp_2);
+ outLen = outLen / 2;
+ return 0;
+ case kResamplerMode3To2:
+ if (maxLen < (lengthIn * 2 / 3))
+ {
+ return -1;
+ }
+ // 3:6
+ tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 2));
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+ lengthIn *= 2;
+ // 6:2
+ // We can only handle blocks of 480 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 480) != 0)
+ {
+ free(tmp);
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+ for (int i = 0; i < lengthIn; i += 480)
+ {
+ WebRtcSpl_Resample48khzTo16khz(tmp + i, samplesOut + i / 3,
+ (WebRtcSpl_State48khzTo16khz *)state2_,
+ tmp_mem);
+ }
+ outLen = lengthIn / 3;
+ free(tmp);
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode11To2:
+ // We can only handle blocks of 220 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 220) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 2) / 11))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*)malloc((lengthIn * 4) / 11 * sizeof(WebRtc_Word16));
+
+ for (int i = 0; i < lengthIn; i += 220)
+ {
+ WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
+ (WebRtcSpl_State22khzTo8khz *)state1_,
+ tmp_mem);
+ }
+ lengthIn = (lengthIn * 4) / 11;
+
+ WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
+ outLen = lengthIn / 2;
+
+ free(tmp_mem);
+ free(tmp);
+ return 0;
+ case kResamplerMode11To4:
+ // We can only handle blocks of 220 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 220) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 4) / 11))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 220)
+ {
+ WebRtcSpl_Resample22khzTo8khz(samplesIn + i, samplesOut + (i * 4) / 11,
+ (WebRtcSpl_State22khzTo8khz *)state1_,
+ tmp_mem);
+ }
+ outLen = (lengthIn * 4) / 11;
+ free(tmp_mem);
+ return 0;
+ case kResamplerMode11To8:
+ // We can only handle blocks of 160 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 220) != 0)
+ {
+ return -1;
+ }
+ if (maxLen < ((lengthIn * 8) / 11))
+ {
+ return -1;
+ }
+ tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+
+ for (int i = 0; i < lengthIn; i += 220)
+ {
+ WebRtcSpl_Resample22khzTo16khz(samplesIn + i, samplesOut + (i * 8) / 11,
+ (WebRtcSpl_State22khzTo16khz *)state1_,
+ tmp_mem);
+ }
+ outLen = (lengthIn * 8) / 11;
+ free(tmp_mem);
+ return 0;
+ break;
+
+ }
+ return 0;
+}
+
+// Asynchronous resampling, input
+int Resampler::Insert(WebRtc_Word16 * samplesIn, int lengthIn)
+{
+ if (my_type_ != kResamplerAsynchronous)
+ {
+ return -1;
+ }
+ int sizeNeeded, tenMsblock;
+
+ // Determine need for size of outBuffer
+ sizeNeeded = out_buffer_size_ + ((lengthIn + in_buffer_size_) * my_out_frequency_khz_)
+ / my_in_frequency_khz_;
+ if (sizeNeeded > out_buffer_size_max_)
+ {
+ // Round the value upwards to complete 10 ms blocks
+ tenMsblock = my_out_frequency_khz_ * 10;
+ sizeNeeded = (sizeNeeded / tenMsblock + 1) * tenMsblock;
+ out_buffer_ = (WebRtc_Word16*)realloc(out_buffer_, sizeNeeded * sizeof(WebRtc_Word16));
+ out_buffer_size_max_ = sizeNeeded;
+ }
+
+ // If we need to use inBuffer, make sure all input data fits there.
+
+ tenMsblock = my_in_frequency_khz_ * 10;
+ if (in_buffer_size_ || (lengthIn % tenMsblock))
+ {
+ // Check if input buffer size is enough
+ if ((in_buffer_size_ + lengthIn) > in_buffer_size_max_)
+ {
+ // Round the value upwards to complete 10 ms blocks
+ sizeNeeded = ((in_buffer_size_ + lengthIn) / tenMsblock + 1) * tenMsblock;
+ in_buffer_ = (WebRtc_Word16*)realloc(in_buffer_,
+ sizeNeeded * sizeof(WebRtc_Word16));
+ in_buffer_size_max_ = sizeNeeded;
+ }
+ // Copy in data to input buffer
+ memcpy(in_buffer_ + in_buffer_size_, samplesIn, lengthIn * sizeof(WebRtc_Word16));
+
+ // Resample all available 10 ms blocks
+ int lenOut;
+ int dataLenToResample = (in_buffer_size_ / tenMsblock) * tenMsblock;
+ Push(in_buffer_, dataLenToResample, out_buffer_ + out_buffer_size_,
+ out_buffer_size_max_ - out_buffer_size_, lenOut);
+ out_buffer_size_ += lenOut;
+
+ // Save the rest
+ memmove(in_buffer_, in_buffer_ + dataLenToResample,
+ (in_buffer_size_ - dataLenToResample) * sizeof(WebRtc_Word16));
+ in_buffer_size_ -= dataLenToResample;
+ } else
+ {
+ // Just resample
+ int lenOut;
+ Push(in_buffer_, lengthIn, out_buffer_ + out_buffer_size_,
+ out_buffer_size_max_ - out_buffer_size_, lenOut);
+ out_buffer_size_ += lenOut;
+ }
+
+ return 0;
+}
+
+// Asynchronous resampling output, remaining samples are buffered
+int Resampler::Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen)
+{
+ if (my_type_ != kResamplerAsynchronous)
+ {
+ return -1;
+ }
+
+ // Check that we have enough data
+ if (desiredLen <= out_buffer_size_)
+ {
+ // Give out the date
+ memcpy(samplesOut, out_buffer_, desiredLen * sizeof(WebRtc_Word32));
+
+ // Shuffle down remaining
+ memmove(out_buffer_, out_buffer_ + desiredLen,
+ (out_buffer_size_ - desiredLen) * sizeof(WebRtc_Word16));
+
+ // Update remaining size
+ out_buffer_size_ -= desiredLen;
+
+ return 0;
+ } else
+ {
+ return -1;
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/common_audio/resampler/resampler.gypi b/trunk/src/common_audio/resampler/resampler.gypi
new file mode 100644
index 0000000..69f9b0e
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler.gypi
@@ -0,0 +1,55 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'resampler',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/resampler.h',
+ 'resampler.cc',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets' : [
+ {
+ 'target_name': 'resampler_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'resampler',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'resampler_unittest.cc',
+ ],
+ }, # resampler_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/resampler/resampler_unittest.cc b/trunk/src/common_audio/resampler/resampler_unittest.cc
new file mode 100644
index 0000000..9b1061a
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler_unittest.cc
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+#include "common_audio/resampler/include/resampler.h"
+
+// TODO(andrew): this is a work-in-progress. Many more tests are needed.
+
+namespace webrtc {
+namespace {
+const ResamplerType kTypes[] = {
+ kResamplerSynchronous,
+ kResamplerAsynchronous,
+ kResamplerSynchronousStereo,
+ kResamplerAsynchronousStereo
+ // kResamplerInvalid excluded
+};
+const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
+
+// Rates we must support.
+const int kMaxRate = 96000;
+const int kRates[] = {
+ 8000,
+ 16000,
+ 32000,
+ 44000,
+ 48000,
+ kMaxRate
+};
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+const int kMaxChannels = 2;
+const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
+
+// TODO(andrew): should we be supporting these combinations?
+bool ValidRates(int in_rate, int out_rate) {
+ // Not the most compact notation, for clarity.
+ if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
+ (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
+ return false;
+ }
+
+ return true;
+}
+
+class ResamplerTest : public testing::Test {
+ protected:
+ ResamplerTest();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ Resampler rs_;
+ int16_t data_in_[kDataSize];
+ int16_t data_out_[kDataSize];
+};
+
+ResamplerTest::ResamplerTest() {}
+
+void ResamplerTest::SetUp() {
+ // Initialize input data with anything. The tests are content independent.
+ memset(data_in_, 1, sizeof(data_in_));
+}
+
+void ResamplerTest::TearDown() {}
+
+TEST_F(ResamplerTest, Reset) {
+ // The only failure mode for the constructor is if Reset() fails. For the
+ // time being then (until an Init function is added), we rely on Reset()
+ // to test the constructor.
+
+ // Check that all required combinations are supported.
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ for (size_t k = 0; k < kTypesSize; ++k) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
+ << ", type: " << kTypes[k];
+ SCOPED_TRACE(ss.str());
+ if (ValidRates(kRates[i], kRates[j]))
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+ else
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+ }
+ }
+ }
+}
+
+// TODO(tlegrand): Replace code inside the two tests below with a function
+// with number of channels and ResamplerType as input.
+TEST_F(ResamplerTest, Synchronous) {
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+ SCOPED_TRACE(ss.str());
+
+ if (ValidRates(kRates[i], kRates[j])) {
+ int in_length = kRates[i] / 100;
+ int out_length = 0;
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+ EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+ out_length));
+ EXPECT_EQ(kRates[j] / 100, out_length);
+ } else {
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+ }
+ }
+ }
+}
+
+TEST_F(ResamplerTest, SynchronousStereo) {
+ // Number of channels is 2, stereo mode.
+ const int kChannels = 2;
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+ SCOPED_TRACE(ss.str());
+
+ if (ValidRates(kRates[i], kRates[j])) {
+ int in_length = kChannels * kRates[i] / 100;
+ int out_length = 0;
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
+ kResamplerSynchronousStereo));
+ EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+ out_length));
+ EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
+ } else {
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
+ kResamplerSynchronousStereo));
+ }
+ }
+ }
+}
+} // namespace
+} // namespace webrtc
diff --git a/trunk/src/common_audio/signal_processing/Android.mk b/trunk/src/common_audio/signal_processing/Android.mk
new file mode 100644
index 0000000..3ff066c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/Android.mk
@@ -0,0 +1,100 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_spl
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ auto_corr_to_refl_coef.c \
+ auto_correlation.c \
+ complex_fft.c \
+ copy_set_operations.c \
+ division_operations.c \
+ dot_product_with_scale.c \
+ energy.c \
+ filter_ar.c \
+ filter_ma_fast_q12.c \
+ get_hanning_window.c \
+ get_scaling_square.c \
+ ilbc_specific_functions.c \
+ levinson_durbin.c \
+ lpc_to_refl_coef.c \
+ min_max_operations.c \
+ randomization_functions.c \
+ refl_coef_to_lpc.c \
+ resample.c \
+ resample_48khz.c \
+ resample_by_2.c \
+ resample_by_2_internal.c \
+ resample_fractional.c \
+ spl_sqrt.c \
+ spl_version.c \
+ splitting_filter.c \
+ sqrt_of_one_minus_x_squared.c \
+ vector_scaling_operations.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../..
+
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+LOCAL_SRC_FILES += \
+ min_max_operations_neon.c \
+ cross_correlation_neon.s \
+ downsample_fast_neon.s
+LOCAL_CFLAGS += \
+ $(MY_ARM_CFLAGS_NEON)
+else
+LOCAL_SRC_FILES += \
+ cross_correlation.c \
+ downsample_fast.c
+endif
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+LOCAL_SRC_FILES += \
+ filter_ar_fast_q12_armv7.s
+else
+LOCAL_SRC_FILES += \
+ filter_ar_fast_q12.c
+endif
+
+ifeq ($(TARGET_ARCH),arm)
+LOCAL_SRC_FILES += \
+ complex_bit_reverse_arm.s \
+ spl_sqrt_floor.s
+else
+LOCAL_SRC_FILES += \
+ complex_bit_reverse.c \
+ spl_sqrt_floor.c
+endif
+
+LOCAL_SHARED_LIBRARIES := libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c b/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
new file mode 100644
index 0000000..b7e8858
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_AutoCorrToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32 *R, int use_order, WebRtc_Word16 *K)
+{
+ int i, n;
+ WebRtc_Word16 tmp;
+ G_CONST WebRtc_Word32 *rptr;
+ WebRtc_Word32 L_num, L_den;
+ WebRtc_Word16 *acfptr, *pptr, *wptr, *p1ptr, *w1ptr, ACF[WEBRTC_SPL_MAX_LPC_ORDER],
+ P[WEBRTC_SPL_MAX_LPC_ORDER], W[WEBRTC_SPL_MAX_LPC_ORDER];
+
+ // Initialize loop and pointers.
+ acfptr = ACF;
+ rptr = R;
+ pptr = P;
+ p1ptr = &P[1];
+ w1ptr = &W[1];
+ wptr = w1ptr;
+
+ // First loop; n=0. Determine shifting.
+ tmp = WebRtcSpl_NormW32(*R);
+ *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
+ *pptr++ = *acfptr++;
+
+ // Initialize ACF, P and W.
+ for (i = 1; i <= use_order; i++)
+ {
+ *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
+ *wptr++ = *acfptr;
+ *pptr++ = *acfptr++;
+ }
+
+ // Compute reflection coefficients.
+ for (n = 1; n <= use_order; n++, K++)
+ {
+ tmp = WEBRTC_SPL_ABS_W16(*p1ptr);
+ if (*P < tmp)
+ {
+ for (i = n; i <= use_order; i++)
+ *K++ = 0;
+
+ return;
+ }
+
+ // Division: WebRtcSpl_div(tmp, *P)
+ *K = 0;
+ if (tmp != 0)
+ {
+ L_num = tmp;
+ L_den = *P;
+ i = 15;
+ while (i--)
+ {
+ (*K) <<= 1;
+ L_num <<= 1;
+ if (L_num >= L_den)
+ {
+ L_num -= L_den;
+ (*K)++;
+ }
+ }
+ if (*p1ptr > 0)
+ *K = -*K;
+ }
+
+ // Last iteration; don't do Schur recursion.
+ if (n == use_order)
+ return;
+
+ // Schur recursion.
+ pptr = P;
+ wptr = w1ptr;
+ tmp = (WebRtc_Word16)(((WebRtc_Word32)*p1ptr * (WebRtc_Word32)*K + 16384) >> 15);
+ *pptr = WEBRTC_SPL_ADD_SAT_W16( *pptr, tmp );
+ pptr++;
+ for (i = 1; i <= use_order - n; i++)
+ {
+ tmp = (WebRtc_Word16)(((WebRtc_Word32)*wptr * (WebRtc_Word32)*K + 16384) >> 15);
+ *pptr = WEBRTC_SPL_ADD_SAT_W16( *(pptr+1), tmp );
+ pptr++;
+ tmp = (WebRtc_Word16)(((WebRtc_Word32)*pptr * (WebRtc_Word32)*K + 16384) >> 15);
+ *wptr = WEBRTC_SPL_ADD_SAT_W16( *wptr, tmp );
+ wptr++;
+ }
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/auto_correlation.c b/trunk/src/common_audio/signal_processing/auto_correlation.c
new file mode 100644
index 0000000..a00fde4
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/auto_correlation.c
@@ -0,0 +1,141 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_AutoCorrelation().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector,
+ int in_vector_length,
+ int order,
+ WebRtc_Word32* result,
+ int* scale)
+{
+ WebRtc_Word32 sum;
+ int i, j;
+ WebRtc_Word16 smax; // Sample max
+ G_CONST WebRtc_Word16* xptr1;
+ G_CONST WebRtc_Word16* xptr2;
+ WebRtc_Word32* resultptr;
+ int scaling = 0;
+
+#ifdef _ARM_OPT_
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+ WebRtc_Word16 loops4;
+#endif
+
+ if (order < 0)
+ order = in_vector_length;
+
+ // Find the max. sample
+ smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
+
+ // In order to avoid overflow when computing the sum we should scale the samples so that
+ // (in_vector_length * smax * smax) will not overflow.
+
+ if (smax == 0)
+ {
+ scaling = 0;
+ } else
+ {
+ int nbits = WebRtcSpl_GetSizeInBits(in_vector_length); // # of bits in the sum loop
+ int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
+
+ if (t > nbits)
+ {
+ scaling = 0;
+ } else
+ {
+ scaling = nbits - t;
+ }
+
+ }
+
+ resultptr = result;
+
+ // Perform the actual correlation calculation
+ for (i = 0; i < order + 1; i++)
+ {
+ int loops = (in_vector_length - i);
+ sum = 0;
+ xptr1 = in_vector;
+ xptr2 = &in_vector[i];
+#ifndef _ARM_OPT_
+ for (j = loops; j > 0; j--)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling);
+ }
+#else
+ loops4 = (loops >> 2) << 2;
+
+ if (scaling == 0)
+ {
+ for (j = 0; j < loops4; j = j + 4)
+ {
+ sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+ xptr1++;
+ xptr2++;
+ }
+
+ for (j = loops4; j < loops; j++)
+ {
+ sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+ xptr1++;
+ xptr2++;
+ }
+ }
+ else
+ {
+ for (j = 0; j < loops4; j = j + 4)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+ xptr1++;
+ xptr2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+ xptr1++;
+ xptr2++;
+ }
+
+ for (j = loops4; j < loops; j++)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+ xptr1++;
+ xptr2++;
+ }
+ }
+
+#endif
+ *resultptr++ = sum;
+ }
+
+ *scale = scaling;
+
+ return order + 1;
+}
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse.c b/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
new file mode 100644
index 0000000..02fde1e
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
@@ -0,0 +1,109 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+
+/* Tables for data buffer indexes that are bit reversed and thus need to be
+ * swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
+ * operations, while index_7[{1, 3, 5, ...}] are for the right side of the
+ * operation. Same for index_8.
+ */
+
+/* Indexes for the case of stages == 7. */
+static const int16_t index_7[112] = {
+ 1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
+ 12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
+ 23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
+ 37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
+ 51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
+ 81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
+ 103, 115, 111, 123
+};
+
+/* Indexes for the case of stages == 8. */
+static const int16_t index_8[240] = {
+ 1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
+ 11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
+ 40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
+ 30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
+ 148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
+ 51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
+ 124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
+ 75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
+ 234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
+ 166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
+ 115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
+ 193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
+ 149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
+ 213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
+ 203, 211, 207, 243, 215, 235, 223, 251, 239, 247
+};
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
+ /* For any specific value of stages, we know exactly the indexes that are
+ * bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
+ * stages are 7 and 8, so we use tables to save unnecessary iterations and
+ * calculations for these two cases.
+ */
+ if (stages == 7 || stages == 8) {
+ int m = 0;
+ int length = 112;
+ const int16_t* index = index_7;
+
+ if (stages == 8) {
+ length = 240;
+ index = index_8;
+ }
+
+ /* Decimation in time. Swap the elements with bit-reversed indexes. */
+ for (m = 0; m < length; m += 2) {
+ /* We declare a int32_t* type pointer, to load both the 16-bit real
+ * and imaginary elements from complex_data in one instruction, reducing
+ * complexity.
+ */
+ int32_t* complex_data_ptr = (int32_t*)complex_data;
+ int32_t temp = 0;
+
+ temp = complex_data_ptr[index[m]]; /* Real and imaginary */
+ complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
+ complex_data_ptr[index[m + 1]] = temp;
+ }
+ }
+ else {
+ int m = 0, mr = 0, l = 0;
+ int n = 1 << stages;
+ int nn = n - 1;
+
+ /* Decimation in time - re-order data */
+ for (m = 1; m <= nn; ++m) {
+ int32_t* complex_data_ptr = (int32_t*)complex_data;
+ int32_t temp = 0;
+
+ /* Find out indexes that are bit-reversed. */
+ l = n;
+ do {
+ l >>= 1;
+ } while (l > nn - mr);
+ mr = (mr & (l - 1)) + l;
+
+ if (mr <= m) {
+ continue;
+ }
+
+ /* Swap the elements with bit-reversed indexes.
+ * This is similar to the loop in the stages == 7 or 8 cases.
+ */
+ temp = complex_data_ptr[m]; /* Real and imaginary */
+ complex_data_ptr[m] = complex_data_ptr[mr];
+ complex_data_ptr[mr] = temp;
+ }
+ }
+}
+
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s b/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
new file mode 100644
index 0000000..4828077
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
@@ -0,0 +1,126 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_ComplexBitReverse(), optimized
+@ for ARMv5 platforms.
+@ Reference C code is in file complex_bit_reverse.c. Bit-exact.
+
+.arch armv5
+
+.global WebRtcSpl_ComplexBitReverse
+
+.align 2
+
+WebRtcSpl_ComplexBitReverse:
+.fnstart
+
+ push {r4-r7}
+
+ cmp r1, #7
+ adr r3, index_7 @ Table pointer.
+ mov r4, #112 @ Number of interations.
+ beq PRE_LOOP_STAGES_7_OR_8
+
+ cmp r1, #8
+ adr r3, index_8 @ Table pointer.
+ mov r4, #240 @ Number of interations.
+ beq PRE_LOOP_STAGES_7_OR_8
+
+ mov r3, #1 @ Initialize m.
+ mov r1, r3, asl r1 @ n = 1 << stages;
+ subs r6, r1, #1 @ nn = n - 1;
+ ble END
+
+ mov r5, r0 @ &complex_data
+ mov r4, #0 @ ml
+
+LOOP_GENERIC:
+ rsb r12, r4, r6 @ l > nn - mr
+ mov r2, r1 @ n
+
+LOOP_SHIFT:
+ asr r2, #1 @ l >>= 1;
+ cmp r2, r12
+ bgt LOOP_SHIFT
+
+ sub r12, r2, #1
+ and r4, r12, r4
+ add r4, r2 @ mr = (mr & (l - 1)) + l;
+ cmp r4, r3 @ mr <= m ?
+ ble UPDATE_REGISTERS
+
+ mov r12, r4, asl #2
+ ldr r7, [r5, #4] @ complex_data[2 * m, 2 * m + 1].
+ @ Offset 4 due to m incrementing from 1.
+ ldr r2, [r0, r12] @ complex_data[2 * mr, 2 * mr + 1].
+ str r7, [r0, r12]
+ str r2, [r5, #4]
+
+UPDATE_REGISTERS:
+ add r3, r3, #1
+ add r5, #4
+ cmp r3, r1
+ bne LOOP_GENERIC
+
+ b END
+
+PRE_LOOP_STAGES_7_OR_8:
+ add r4, r3, r4, asl #1
+
+LOOP_STAGES_7_OR_8:
+ ldrsh r2, [r3], #2 @ index[m]
+ ldrsh r5, [r3], #2 @ index[m + 1]
+ ldr r1, [r0, r2] @ complex_data[index[m], index[m] + 1]
+ ldr r12, [r0, r5] @ complex_data[index[m + 1], index[m + 1] + 1]
+ cmp r3, r4
+ str r1, [r0, r5]
+ str r12, [r0, r2]
+ bne LOOP_STAGES_7_OR_8
+
+END:
+ pop {r4-r7}
+ bx lr
+
+.fnend
+
+
+@ The index tables. Note the values are doubles of the actual indexes for 16-bit
+@ elements, different from the generic C code. It actually provides byte offsets
+@ for the indexes.
+
+.align 2
+index_7: @ Indexes for stages == 7.
+ .hword 4, 256, 8, 128, 12, 384, 16, 64, 20, 320, 24, 192, 28, 448, 36, 288
+ .hword 40, 160, 44, 416, 48, 96, 52, 352, 56, 224, 60, 480, 68, 272, 72, 144
+ .hword 76, 400, 84, 336, 88, 208, 92, 464, 100, 304, 104, 176, 108, 432, 116
+ .hword 368, 120, 240, 124, 496, 132, 264, 140, 392, 148, 328, 152, 200, 156
+ .hword 456, 164, 296, 172, 424, 180, 360, 184, 232, 188, 488, 196, 280, 204
+ .hword 408, 212, 344, 220, 472, 228, 312, 236, 440, 244, 376, 252, 504, 268
+ .hword 388, 276, 324, 284, 452, 300, 420, 308, 356, 316, 484, 332, 404, 348
+ .hword 468, 364, 436, 380, 500, 412, 460, 444, 492
+
+index_8: @ Indexes for stages == 8.
+ .hword 4, 512, 8, 256, 12, 768, 16, 128, 20, 640, 24, 384, 28, 896, 32, 64
+ .hword 36, 576, 40, 320, 44, 832, 48, 192, 52, 704, 56, 448, 60, 960, 68, 544
+ .hword 72, 288, 76, 800, 80, 160, 84, 672, 88, 416, 92, 928, 100, 608, 104
+ .hword 352, 108, 864, 112, 224, 116, 736, 120, 480, 124, 992, 132, 528, 136
+ .hword 272, 140, 784, 148, 656, 152, 400, 156, 912, 164, 592, 168, 336, 172
+ .hword 848, 176, 208, 180, 720, 184, 464, 188, 976, 196, 560, 200, 304, 204
+ .hword 816, 212, 688, 216, 432, 220, 944, 228, 624, 232, 368, 236, 880, 244
+ .hword 752, 248, 496, 252, 1008, 260, 520, 268, 776, 276, 648, 280, 392, 284
+ .hword 904, 292, 584, 296, 328, 300, 840, 308, 712, 312, 456, 316, 968, 324
+ .hword 552, 332, 808, 340, 680, 344, 424, 348, 936, 356, 616, 364, 872, 372
+ .hword 744, 376, 488, 380, 1000, 388, 536, 396, 792, 404, 664, 412, 920, 420
+ .hword 600, 428, 856, 436, 728, 440, 472, 444, 984, 452, 568, 460, 824, 468
+ .hword 696, 476, 952, 484, 632, 492, 888, 500, 760, 508, 1016, 524, 772, 532
+ .hword 644, 540, 900, 548, 580, 556, 836, 564, 708, 572, 964, 588, 804, 596
+ .hword 676, 604, 932, 620, 868, 628, 740, 636, 996, 652, 788, 668, 916, 684
+ .hword 852, 692, 724, 700, 980, 716, 820, 732, 948, 748, 884, 764, 1012, 796
+ .hword 908, 812, 844, 828, 972, 860, 940, 892, 1004, 956, 988
diff --git a/trunk/src/common_audio/signal_processing/complex_fft.c b/trunk/src/common_audio/signal_processing/complex_fft.c
new file mode 100644
index 0000000..1e8503c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_fft.c
@@ -0,0 +1,425 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_ComplexFFT().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define CFFTSFT 14
+#define CFFTRND 1
+#define CFFTRND2 16384
+
+#define CIFFTSFT 14
+#define CIFFTRND 1
+
+static const WebRtc_Word16 kSinTable1024[] = {
+ 0, 201, 402, 603, 804, 1005, 1206, 1406,
+ 1607, 1808, 2009, 2209, 2410, 2610, 2811, 3011,
+ 3211, 3411, 3611, 3811, 4011, 4210, 4409, 4608,
+ 4807, 5006, 5205, 5403, 5601, 5799, 5997, 6195,
+ 6392, 6589, 6786, 6982, 7179, 7375, 7571, 7766,
+ 7961, 8156, 8351, 8545, 8739, 8932, 9126, 9319,
+ 9511, 9703, 9895, 10087, 10278, 10469, 10659, 10849,
+ 11038, 11227, 11416, 11604, 11792, 11980, 12166, 12353,
+ 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
+ 14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268,
+ 15446, 15623, 15799, 15975, 16150, 16325, 16499, 16672,
+ 16845, 17017, 17189, 17360, 17530, 17699, 17868, 18036,
+ 18204, 18371, 18537, 18702, 18867, 19031, 19194, 19357,
+ 19519, 19680, 19840, 20000, 20159, 20317, 20474, 20631,
+ 20787, 20942, 21096, 21249, 21402, 21554, 21705, 21855,
+ 22004, 22153, 22301, 22448, 22594, 22739, 22883, 23027,
+ 23169, 23311, 23452, 23592, 23731, 23869, 24006, 24143,
+ 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
+ 25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198,
+ 26318, 26437, 26556, 26673, 26789, 26905, 27019, 27132,
+ 27244, 27355, 27466, 27575, 27683, 27790, 27896, 28001,
+ 28105, 28208, 28309, 28410, 28510, 28608, 28706, 28802,
+ 28897, 28992, 29085, 29177, 29268, 29358, 29446, 29534,
+ 29621, 29706, 29790, 29873, 29955, 30036, 30116, 30195,
+ 30272, 30349, 30424, 30498, 30571, 30643, 30713, 30783,
+ 30851, 30918, 30984, 31049,
+ 31113, 31175, 31236, 31297,
+ 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
+ 31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097,
+ 32137, 32176, 32213, 32249, 32284, 32318, 32350, 32382,
+ 32412, 32441, 32468, 32495, 32520, 32544, 32567, 32588,
+ 32609, 32628, 32646, 32662, 32678, 32692, 32705, 32717,
+ 32727, 32736, 32744, 32751, 32757, 32761, 32764, 32766,
+ 32767, 32766, 32764, 32761, 32757, 32751, 32744, 32736,
+ 32727, 32717, 32705, 32692, 32678, 32662, 32646, 32628,
+ 32609, 32588, 32567, 32544, 32520, 32495, 32468, 32441,
+ 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
+ 32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833,
+ 31785, 31735, 31684, 31633, 31580, 31525, 31470, 31413,
+ 31356, 31297, 31236, 31175, 31113, 31049, 30984, 30918,
+ 30851, 30783, 30713, 30643, 30571, 30498, 30424, 30349,
+ 30272, 30195, 30116, 30036, 29955, 29873, 29790, 29706,
+ 29621, 29534, 29446, 29358, 29268, 29177, 29085, 28992,
+ 28897, 28802, 28706, 28608, 28510, 28410, 28309, 28208,
+ 28105, 28001, 27896, 27790, 27683, 27575, 27466, 27355,
+ 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
+ 26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456,
+ 25329, 25201, 25072, 24942, 24811, 24679, 24546, 24413,
+ 24278, 24143, 24006, 23869, 23731, 23592, 23452, 23311,
+ 23169, 23027, 22883, 22739, 22594, 22448, 22301, 22153,
+ 22004, 21855, 21705, 21554, 21402, 21249, 21096, 20942,
+ 20787, 20631, 20474, 20317, 20159, 20000, 19840, 19680,
+ 19519, 19357, 19194, 19031, 18867, 18702, 18537, 18371,
+ 18204, 18036, 17868, 17699, 17530, 17360, 17189, 17017,
+ 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
+ 15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191,
+ 14009, 13827, 13645, 13462, 13278, 13094, 12909, 12724,
+ 12539, 12353, 12166, 11980, 11792, 11604, 11416, 11227,
+ 11038, 10849, 10659, 10469, 10278, 10087, 9895, 9703,
+ 9511, 9319, 9126, 8932, 8739, 8545, 8351, 8156,
+ 7961, 7766, 7571, 7375, 7179, 6982, 6786, 6589,
+ 6392, 6195, 5997, 5799, 5601, 5403, 5205, 5006,
+ 4807, 4608, 4409, 4210, 4011, 3811, 3611, 3411,
+ 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
+ 1607, 1406, 1206, 1005, 804, 603, 402, 201,
+ 0, -201, -402, -603, -804, -1005, -1206, -1406,
+ -1607, -1808, -2009, -2209, -2410, -2610, -2811, -3011,
+ -3211, -3411, -3611, -3811, -4011, -4210, -4409, -4608,
+ -4807, -5006, -5205, -5403, -5601, -5799, -5997, -6195,
+ -6392, -6589, -6786, -6982, -7179, -7375, -7571, -7766,
+ -7961, -8156, -8351, -8545, -8739, -8932, -9126, -9319,
+ -9511, -9703, -9895, -10087, -10278, -10469, -10659, -10849,
+ -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
+ -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
+ -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
+ -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
+ -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
+ -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
+ -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
+ -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
+ -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
+ -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
+ -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
+ -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
+ -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
+ -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
+ -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
+ -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
+ -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
+ -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
+ -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
+ -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
+ -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
+ -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
+ -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
+ -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
+ -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
+ -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
+ -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
+ -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
+ -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
+ -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
+ -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
+ -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
+ -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
+ -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
+ -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
+ -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
+ -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
+ -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
+ -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
+ -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
+ -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
+ -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
+ -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
+ -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
+ -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
+ -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
+ -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
+ -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
+ -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
+ -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
+ -11038, -10849, -10659, -10469, -10278, -10087, -9895, -9703,
+ -9511, -9319, -9126, -8932, -8739, -8545, -8351, -8156,
+ -7961, -7766, -7571, -7375, -7179, -6982, -6786, -6589,
+ -6392, -6195, -5997, -5799, -5601, -5403, -5205, -5006,
+ -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
+ -3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808,
+ -1607, -1406, -1206, -1005, -804, -603, -402, -201
+};
+
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+ int i, j, l, k, istep, n, m;
+ WebRtc_Word16 wr, wi;
+ WebRtc_Word32 tr32, ti32, qr32, qi32;
+
+ /* The 1024-value is a constant given from the size of kSinTable1024[],
+ * and should not be changed depending on the input parameter 'stages'
+ */
+ n = 1 << stages;
+ if (n > 1024)
+ return -1;
+
+ l = 1;
+ k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+ depending on the input parameter 'stages' */
+
+ if (mode == 0)
+ {
+ // mode==0: Low-complexity and Low-accuracy mode
+ while (l < n)
+ {
+ istep = l << 1;
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = -kSinTable1024[j];
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
+
+ ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
+
+ qr32 = (WebRtc_Word32)frfi[2 * i];
+ qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
+ }
+ }
+
+ --k;
+ l = istep;
+
+ }
+
+ } else
+ {
+ // mode==1: High-complexity and High-accuracy mode
+ while (l < n)
+ {
+ istep = l << 1;
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = -kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ WebRtc_Word32 wri;
+ WebRtc_Word32 frfi_r;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+ "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+ "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+ __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+ "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+ __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+ "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+
+#else
+ tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
+
+ ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
+#endif
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CFFTSFT);
+ ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CFFTSFT);
+
+ qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
+ qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
+
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
+ }
+ }
+
+ --k;
+ l = istep;
+ }
+ }
+ return 0;
+}
+
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+ int i, j, l, k, istep, n, m, scale, shift;
+ WebRtc_Word16 wr, wi;
+ WebRtc_Word32 tr32, ti32, qr32, qi32;
+ WebRtc_Word32 tmp32, round2;
+
+ /* The 1024-value is a constant given from the size of kSinTable1024[],
+ * and should not be changed depending on the input parameter 'stages'
+ */
+ n = 1 << stages;
+ if (n > 1024)
+ return -1;
+
+ scale = 0;
+
+ l = 1;
+ k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+ depending on the input parameter 'stages' */
+
+ while (l < n)
+ {
+ // variable scaling, depending upon data
+ shift = 0;
+ round2 = 8192;
+
+ tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
+ if (tmp32 > 13573)
+ {
+ shift++;
+ scale++;
+ round2 <<= 1;
+ }
+ if (tmp32 > 27146)
+ {
+ shift++;
+ scale++;
+ round2 <<= 1;
+ }
+
+ istep = l << 1;
+
+ if (mode == 0)
+ {
+ // mode==0: Low-complexity and Low-accuracy mode
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = kSinTable1024[j];
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
+ - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
+
+ ti32 = WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
+ + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
+
+ qr32 = (WebRtc_Word32)frfi[2 * i];
+ qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
+ }
+ }
+ } else
+ {
+ // mode==1: High-complexity and High-accuracy mode
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ WebRtc_Word32 wri;
+ WebRtc_Word32 frfi_r;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+ "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+ "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+ __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+ "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+ __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+ "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+#else
+
+ tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
+
+ ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
+#endif
+ tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CIFFTSFT);
+ ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CIFFTSFT);
+
+ qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
+ qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
+
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
+ shift+CIFFTSFT);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 - ti32 + round2), shift + CIFFTSFT);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
+ shift + CIFFTSFT);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 + ti32 + round2), shift + CIFFTSFT);
+ }
+ }
+
+ }
+ --k;
+ l = istep;
+ }
+ return scale;
+}
diff --git a/trunk/src/common_audio/signal_processing/copy_set_operations.c b/trunk/src/common_audio/signal_processing/copy_set_operations.c
new file mode 100644
index 0000000..8247337
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/copy_set_operations.c
@@ -0,0 +1,108 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the implementation of functions
+ * WebRtcSpl_MemSetW16()
+ * WebRtcSpl_MemSetW32()
+ * WebRtcSpl_MemCpyReversedOrder()
+ * WebRtcSpl_CopyFromEndW16()
+ * WebRtcSpl_ZerosArrayW16()
+ * WebRtcSpl_ZerosArrayW32()
+ * WebRtcSpl_OnesArrayW16()
+ * WebRtcSpl_OnesArrayW32()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+
+
+void WebRtcSpl_MemSetW16(WebRtc_Word16 *ptr, WebRtc_Word16 set_value, int length)
+{
+ int j;
+ WebRtc_Word16 *arrptr = ptr;
+
+ for (j = length; j > 0; j--)
+ {
+ *arrptr++ = set_value;
+ }
+}
+
+void WebRtcSpl_MemSetW32(WebRtc_Word32 *ptr, WebRtc_Word32 set_value, int length)
+{
+ int j;
+ WebRtc_Word32 *arrptr = ptr;
+
+ for (j = length; j > 0; j--)
+ {
+ *arrptr++ = set_value;
+ }
+}
+
+void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* dest, WebRtc_Word16* source, int length)
+{
+ int j;
+ WebRtc_Word16* destPtr = dest;
+ WebRtc_Word16* sourcePtr = source;
+
+ for (j = 0; j < length; j++)
+ {
+ *destPtr-- = *sourcePtr++;
+ }
+}
+
+WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in,
+ WebRtc_Word16 length,
+ WebRtc_Word16 samples,
+ WebRtc_Word16 *vector_out)
+{
+ // Copy the last <samples> of the input vector to vector_out
+ WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
+
+ return samples;
+}
+
+WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+ WebRtcSpl_MemSetW16(vector, 0, length);
+ return length;
+}
+
+WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+ WebRtcSpl_MemSetW32(vector, 0, length);
+ return length;
+}
+
+WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word16 *tmpvec = vector;
+ for (i = 0; i < length; i++)
+ {
+ *tmpvec++ = 1;
+ }
+ return length;
+}
+
+WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word32 *tmpvec = vector;
+ for (i = 0; i < length; i++)
+ {
+ *tmpvec++ = 1;
+ }
+ return length;
+}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation.c b/trunk/src/common_audio/signal_processing/cross_correlation.c
new file mode 100644
index 0000000..726a749
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/cross_correlation.c
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_CrossCorrelation().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+/* TODO(kma): Clean up the code in this file, and break it up for
+ * various platforms (Xscale, ARM/Neon etc.).
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation, WebRtc_Word16* seq1,
+ WebRtc_Word16* seq2, WebRtc_Word16 dim_seq,
+ WebRtc_Word16 dim_cross_correlation,
+ WebRtc_Word16 right_shifts,
+ WebRtc_Word16 step_seq2)
+{
+ int i, j;
+ WebRtc_Word16* seq1Ptr;
+ WebRtc_Word16* seq2Ptr;
+ WebRtc_Word32* CrossCorrPtr;
+
+#ifdef _XSCALE_OPT_
+
+#ifdef _WIN32
+#pragma message("NOTE: _XSCALE_OPT_ optimizations are used (overrides _ARM_OPT_ and requires /QRxscale compiler flag)")
+#endif
+
+ __int64 macc40;
+
+ int iseq1[250];
+ int iseq2[250];
+ int iseq3[250];
+ int * iseq1Ptr;
+ int * iseq2Ptr;
+ int * iseq3Ptr;
+ int len, i_len;
+
+ seq1Ptr = seq1;
+ iseq1Ptr = iseq1;
+ for(i = 0; i < ((dim_seq + 1) >> 1); i++)
+ {
+ *iseq1Ptr = (unsigned short)*seq1Ptr++;
+ *iseq1Ptr++ |= (WebRtc_Word32)*seq1Ptr++ << 16;
+
+ }
+
+ if(dim_seq%2)
+ {
+ *(iseq1Ptr-1) &= 0x0000ffff;
+ }
+ *iseq1Ptr = 0;
+ iseq1Ptr++;
+ *iseq1Ptr = 0;
+ iseq1Ptr++;
+ *iseq1Ptr = 0;
+
+ if(step_seq2 < 0)
+ {
+ seq2Ptr = seq2 - dim_cross_correlation + 1;
+ CrossCorrPtr = &cross_correlation[dim_cross_correlation - 1];
+ }
+ else
+ {
+ seq2Ptr = seq2;
+ CrossCorrPtr = cross_correlation;
+ }
+
+ len = dim_seq + dim_cross_correlation - 1;
+ i_len = (len + 1) >> 1;
+ iseq2Ptr = iseq2;
+
+ iseq3Ptr = iseq3;
+ for(i = 0; i < i_len; i++)
+ {
+ *iseq2Ptr = (unsigned short)*seq2Ptr++;
+ *iseq3Ptr = (unsigned short)*seq2Ptr;
+ *iseq2Ptr++ |= (WebRtc_Word32)*seq2Ptr++ << 16;
+ *iseq3Ptr++ |= (WebRtc_Word32)*seq2Ptr << 16;
+ }
+
+ if(len % 2)
+ {
+ iseq2[i_len - 1] &= 0x0000ffff;
+ iseq3[i_len - 1] = 0;
+ }
+ else
+ iseq3[i_len - 1] &= 0x0000ffff;
+
+ iseq2[i_len] = 0;
+ iseq3[i_len] = 0;
+ iseq2[i_len + 1] = 0;
+ iseq3[i_len + 1] = 0;
+ iseq2[i_len + 2] = 0;
+ iseq3[i_len + 2] = 0;
+
+ // Set pointer to start value
+ iseq2Ptr = iseq2;
+ iseq3Ptr = iseq3;
+
+ i_len = (dim_seq + 7) >> 3;
+ for (i = 0; i < dim_cross_correlation; i++)
+ {
+
+ iseq1Ptr = iseq1;
+
+ macc40 = 0;
+
+ _WriteCoProcessor(macc40, 0);
+
+ if((i & 1))
+ {
+ iseq3Ptr = iseq3 + (i >> 1);
+ for (j = i_len; j > 0; j--)
+ {
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+ }
+ }
+ else
+ {
+ iseq2Ptr = iseq2 + (i >> 1);
+ for (j = i_len; j > 0; j--)
+ {
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+ _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+ }
+
+ }
+
+ macc40 = _ReadCoProcessor(0);
+ *CrossCorrPtr = (WebRtc_Word32)(macc40 >> right_shifts);
+ CrossCorrPtr += step_seq2;
+ }
+#else // #ifdef _XSCALE_OPT_
+#ifdef _ARM_OPT_
+ WebRtc_Word16 dim_seq8 = (dim_seq >> 3) << 3;
+#endif
+
+ CrossCorrPtr = cross_correlation;
+
+ for (i = 0; i < dim_cross_correlation; i++)
+ {
+ // Set the pointer to the static vector, set the pointer to the sliding vector
+ // and initialize cross_correlation
+ seq1Ptr = seq1;
+ seq2Ptr = seq2 + (step_seq2 * i);
+ (*CrossCorrPtr) = 0;
+
+#ifndef _ARM_OPT_
+#ifdef _WIN32
+#pragma message("NOTE: default implementation is used")
+#endif
+ // Perform the cross correlation
+ for (j = 0; j < dim_seq; j++)
+ {
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr), right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+#else
+#ifdef _WIN32
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+#endif
+ if (right_shifts == 0)
+ {
+ // Perform the optimized cross correlation
+ for (j = 0; j < dim_seq8; j = j + 8)
+ {
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+
+ for (j = dim_seq8; j < dim_seq; j++)
+ {
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+ }
+ else // right_shifts != 0
+
+ {
+ // Perform the optimized cross correlation
+ for (j = 0; j < dim_seq8; j = j + 8)
+ {
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+
+ for (j = dim_seq8; j < dim_seq; j++)
+ {
+ (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+ right_shifts);
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+ }
+#endif
+ CrossCorrPtr++;
+ }
+#endif
+}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation_neon.s b/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
new file mode 100644
index 0000000..e9b1c69
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
@@ -0,0 +1,168 @@
+@
+@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ cross_correlation_neon.s
+@ This file contains the function WebRtcSpl_CrossCorrelation(),
+@ optimized for ARM Neon platform.
+@
+@ Reference Ccode at end of this file.
+@ Output is bit-exact with the reference C code, but not with the generic
+@ C code in file cross_correlation.c, due to reduction of shift operations
+@ from using Neon registers.
+
+@ Register usage:
+@
+@ r0: *cross_correlation (function argument)
+@ r1: *seq1 (function argument)
+@ r2: *seq2 (function argument)
+@ r3: dim_seq (function argument); then, total iteration of LOOP_DIM_SEQ
+@ r4: counter for LOOP_DIM_CROSS_CORRELATION
+@ r5: seq2_ptr
+@ r6: seq1_ptr
+@ r7: Total iteration of LOOP_DIM_SEQ_RESIDUAL
+@ r8, r9, r10, r11, r12: scratch
+
+.arch armv7-a
+.fpu neon
+
+.align 2
+.global WebRtcSpl_CrossCorrelation
+
+WebRtcSpl_CrossCorrelation:
+
+.fnstart
+
+.save {r4-r11}
+ push {r4-r11}
+
+ @ Put the shift value (-right_shifts) into a Neon register.
+ ldrsh r10, [sp, #36]
+ rsb r10, r10, #0
+ mov r8, r10, asr #31
+ vmov.32 d16, r10, r8
+
+ @ Initialize loop counters.
+ and r7, r3, #7 @ inner_loop_len2 = dim_seq % 8;
+ asr r3, r3, #3 @ inner_loop_len1 = dim_seq / 8;
+ ldrsh r4, [sp, #32] @ dim_cross_correlation
+
+LOOP_DIM_CROSS_CORRELATION:
+ vmov.i32 q9, #0
+ vmov.i32 q14, #0
+ movs r8, r3 @ inner_loop_len1
+ mov r6, r1 @ seq1_ptr
+ mov r5, r2 @ seq2_ptr
+ ble POST_LOOP_DIM_SEQ
+
+LOOP_DIM_SEQ:
+ vld1.16 {d20, d21}, [r6]! @ seq1_ptr
+ vld1.16 {d22, d23}, [r5]! @ seq2_ptr
+ subs r8, r8, #1
+ vmull.s16 q12, d20, d22
+ vmull.s16 q13, d21, d23
+ vpadal.s32 q9, q12
+ vpadal.s32 q14, q13
+ bgt LOOP_DIM_SEQ
+
+POST_LOOP_DIM_SEQ:
+ movs r10, r7 @ Loop counter
+ mov r12, #0
+ mov r8, #0
+ ble POST_LOOP_DIM_SEQ_RESIDUAL
+
+LOOP_DIM_SEQ_RESIDUAL:
+ ldrh r11, [r6], #2
+ ldrh r9, [r5], #2
+ smulbb r11, r11, r9
+ adds r8, r8, r11
+ adc r12, r12, r11, asr #31
+ subs r10, #1
+ bgt LOOP_DIM_SEQ_RESIDUAL
+
+POST_LOOP_DIM_SEQ_RESIDUAL: @ Sum the results up and do the shift.
+ vadd.i64 d18, d19
+ vadd.i64 d28, d29
+ vadd.i64 d18, d28
+ vmov.32 d17[0], r8
+ vmov.32 d17[1], r12
+ vadd.i64 d17, d18
+ vshl.s64 d17, d16
+ vst1.32 d17[0], [r0]! @ Store the output
+
+ ldr r8, [sp, #40] @ step_seq2
+ add r2, r8, lsl #1 @ prepare for seq2_ptr(r5) in the next loop.
+
+ subs r4, #1
+ bgt LOOP_DIM_CROSS_CORRELATION
+
+ pop {r4-r11}
+ bx lr
+
+.fnend
+
+
+@ TODO(kma): Place this piece of reference code into a C code file.
+@ void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation,
+@ WebRtc_Word16* seq1,
+@ WebRtc_Word16* seq2,
+@ WebRtc_Word16 dim_seq,
+@ WebRtc_Word16 dim_cross_correlation,
+@ WebRtc_Word16 right_shifts,
+@ WebRtc_Word16 step_seq2) {
+@ int i = 0;
+@ int j = 0;
+@ int inner_loop_len1 = dim_seq >> 3;
+@ int inner_loop_len2 = dim_seq - (inner_loop_len1 << 3);
+@
+@ assert(dim_cross_correlation > 0);
+@ assert(dim_seq > 0);
+@
+@ for (i = 0; i < dim_cross_correlation; i++) {
+@ int16_t *seq1_ptr = seq1;
+@ int16_t *seq2_ptr = seq2 + (step_seq2 * i);
+@ int64_t sum = 0;
+@
+@ for (j = inner_loop_len1; j > 0; j -= 1) {
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ }
+@
+@ // Calculate the rest of the samples.
+@ for (j = inner_loop_len2; j > 0; j -= 1) {
+@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@ seq1_ptr++;
+@ seq2_ptr++;
+@ }
+@
+@ *cross_correlation++ = (int32_t)(sum >> right_shifts);
+@ }
+@ }
diff --git a/trunk/src/common_audio/signal_processing/division_operations.c b/trunk/src/common_audio/signal_processing/division_operations.c
new file mode 100644
index 0000000..b143373
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/division_operations.c
@@ -0,0 +1,144 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the divisions
+ * WebRtcSpl_DivU32U16()
+ * WebRtcSpl_DivW32W16()
+ * WebRtcSpl_DivW32W16ResW16()
+ * WebRtcSpl_DivResultInQ31()
+ * WebRtcSpl_DivW32HiLow()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den)
+{
+ // Guard against division with 0
+ if (den != 0)
+ {
+ return (WebRtc_UWord32)(num / den);
+ } else
+ {
+ return (WebRtc_UWord32)0xFFFFFFFF;
+ }
+}
+
+WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den)
+{
+ // Guard against division with 0
+ if (den != 0)
+ {
+ return (WebRtc_Word32)(num / den);
+ } else
+ {
+ return (WebRtc_Word32)0x7FFFFFFF;
+ }
+}
+
+WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den)
+{
+ // Guard against division with 0
+ if (den != 0)
+ {
+ return (WebRtc_Word16)(num / den);
+ } else
+ {
+ return (WebRtc_Word16)0x7FFF;
+ }
+}
+
+WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den)
+{
+ WebRtc_Word32 L_num = num;
+ WebRtc_Word32 L_den = den;
+ WebRtc_Word32 div = 0;
+ int k = 31;
+ int change_sign = 0;
+
+ if (num == 0)
+ return 0;
+
+ if (num < 0)
+ {
+ change_sign++;
+ L_num = -num;
+ }
+ if (den < 0)
+ {
+ change_sign++;
+ L_den = -den;
+ }
+ while (k--)
+ {
+ div <<= 1;
+ L_num <<= 1;
+ if (L_num >= L_den)
+ {
+ L_num -= L_den;
+ div++;
+ }
+ }
+ if (change_sign == 1)
+ {
+ div = -div;
+ }
+ return div;
+}
+
+WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
+ WebRtc_Word16 den_low)
+{
+ WebRtc_Word16 approx, tmp_hi, tmp_low, num_hi, num_low;
+ WebRtc_Word32 tmpW32;
+
+ approx = (WebRtc_Word16)WebRtcSpl_DivW32W16((WebRtc_Word32)0x1FFFFFFF, den_hi);
+ // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
+
+ // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
+ tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
+ + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
+ // tmpW32 = den * approx
+
+ tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
+
+ // Store tmpW32 in hi and low format
+ tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ // tmpW32 = 1/den in Q29
+ tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
+ >> 15)) << 1);
+
+ // 1/den in hi and low format
+ tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ // Store num in hi and low format
+ num_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(num, 16);
+ num_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((num
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
+
+ // num * (1/den) by 32 bit multiplication (result in Q28)
+
+ tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
+ >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
+
+ // Put result in Q31 (convert from Q28)
+ tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
+
+ return tmpW32;
+}
diff --git a/trunk/src/common_audio/signal_processing/dot_product_with_scale.c b/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
new file mode 100644
index 0000000..6e085fd
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_DotProductWithScale().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
+ int length, int scaling)
+{
+ WebRtc_Word32 sum;
+ int i;
+#ifdef _ARM_OPT_
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+ WebRtc_Word16 len4 = (length >> 2) << 2;
+#endif
+
+ sum = 0;
+
+#ifndef _ARM_OPT_
+ for (i = 0; i < length; i++)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling);
+ }
+#else
+ if (scaling == 0)
+ {
+ for (i = 0; i < len4; i = i + 4)
+ {
+ sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ }
+
+ for (i = len4; i < length; i++)
+ {
+ sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ }
+ }
+ else
+ {
+ for (i = 0; i < len4; i = i + 4)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+ vector1++;
+ vector2++;
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+ vector1++;
+ vector2++;
+ }
+
+ for (i = len4; i < length; i++)
+ {
+ sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+ vector1++;
+ vector2++;
+ }
+ }
+#endif
+
+ return sum;
+}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast.c b/trunk/src/common_audio/signal_processing/downsample_fast.c
new file mode 100644
index 0000000..526cdca
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/downsample_fast.c
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+
+// TODO(Bjornv): Change the function parameter order to WebRTC code style.
+int WebRtcSpl_DownsampleFast(const int16_t* data_in,
+ int data_in_length,
+ int16_t* data_out,
+ int data_out_length,
+ const int16_t* __restrict coefficients,
+ int coefficients_length,
+ int factor,
+ int delay) {
+ int i = 0;
+ int j = 0;
+ int32_t out_s32 = 0;
+ int endpos = delay + factor * (data_out_length - 1) + 1;
+
+ // Return error if any of the running conditions doesn't meet.
+ if (data_out_length <= 0 || coefficients_length <= 0
+ || data_in_length < endpos) {
+ return -1;
+ }
+
+ for (i = delay; i < endpos; i += factor) {
+ out_s32 = 2048; // Round value, 0.5 in Q12.
+
+ for (j = 0; j < coefficients_length; j++) {
+ out_s32 += coefficients[j] * data_in[i - j]; // Q12.
+ }
+
+ out_s32 >>= 12; // Q0.
+
+ // Saturate and store the output.
+ *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
+ }
+
+ return 0;
+}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast_neon.s b/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
new file mode 100644
index 0000000..906b0a1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
@@ -0,0 +1,222 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_DownsampleFast(), optimized for
+@ ARM Neon platform. The description header can be found in
+@ signal_processing_library.h
+@
+@ The reference C code is in file downsample_fast.c. Bit-exact.
+
+.arch armv7-a
+.fpu neon
+
+.align 2
+.global WebRtcSpl_DownsampleFast
+
+WebRtcSpl_DownsampleFast:
+
+.fnstart
+
+.save {r4-r11}
+ push {r4-r11}
+
+ cmp r3, #0 @ data_out_length <= 0?
+ movle r0, #-1
+ ble END
+
+ ldrsh r12, [sp, #44]
+ ldr r5, [sp, #40] @ r5: factor
+ add r4, r12, #1 @ r4: delay + 1
+ sub r3, r3, #1 @ r3: data_out_length - 1
+ smulbb r3, r5, r3
+ ldr r8, [sp, #32] @ &coefficients[0]
+ mov r9, r12 @ Iteration counter for outer loops.
+ add r3, r4 @ delay + factor * (out_length-1) +1
+
+ cmp r3, r1 @ data_in_length < endpos?
+ movgt r0, #-1
+ bgt END
+
+ @ Initializations.
+ sub r3, r5, asl #3
+ add r11, r0, r12, asl #1 @ &data_in[delay]
+ ldr r0, [sp, #36] @ coefficients_length
+ add r3, r5 @ endpos - factor * 7
+
+ cmp r0, #0 @ coefficients_length <= 0 ?
+ movle r0, #-1
+ ble END
+
+ add r8, r0, asl #1 @ &coeffieient[coefficients_length]
+ cmp r9, r3
+ bge POST_LOOP_ENDPOS @ branch when Iteration < 8 times.
+
+@
+@ First part, unroll the loop 8 times, with 3 subcases (factor == 2, 4, others)
+@
+ mov r4, #-2
+
+ @ Direct program flow to the right channel.
+
+ @ r10 is an offset to &data_in[] in the loop. After an iteration, we need to
+ @ move the pointer back to original after advancing 16 bytes by a vld1, and
+ @ then move 2 bytes forward to increment one more sample.
+ cmp r5, #2
+ moveq r10, #-14
+ beq LOOP_ENDPOS_FACTOR2 @ Branch when factor == 2
+
+ @ Similar here, for r10, we need to move the pointer back to original after
+ @ advancing 32 bytes, then move 2 bytes forward to increment one sample.
+ cmp r5, #4
+ moveq r10, #-30
+ beq LOOP_ENDPOS_FACTOR4 @ Branch when factor == 4
+
+ @ For r10, we need to move the pointer back to original after advancing
+ @ (factor * 7 * 2) bytes, then move 2 bytes forward to increment one sample.
+ mov r10, r5, asl #4
+ rsb r10, #2
+ add r10, r5, asl #1
+ lsl r5, #1 @ r5 = factor * sizeof(data_in)
+
+@ The general case (factor != 2 && factor != 4)
+LOOP_ENDPOS_GENERAL:
+ @ Initializations.
+ vmov.i32 q2, #2048
+ vmov.i32 q3, #2048
+ sub r7, r8, #2
+ sub r12, r0, #1 @ coefficients_length - 1
+ sub r1, r11, r12, asl #1 @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_GENERAL:
+ vld1.16 {d2[], d3[]}, [r7], r4 @ coefficients[j]
+ vld1.16 d0[0], [r1], r5 @ data_in[i - j]
+ vld1.16 d0[1], [r1], r5 @ data_in[i + factor - j]
+ vld1.16 d0[2], [r1], r5 @ data_in[i + factor * 2 - j]
+ vld1.16 d0[3], [r1], r5 @ data_in[i + factor * 3 - j]
+ vld1.16 d1[0], [r1], r5 @ data_in[i + factor * 4 - j]
+ vld1.16 d1[1], [r1], r5 @ data_in[i + factor * 5 - j]
+ vld1.16 d1[2], [r1], r5 @ data_in[i + factor * 6 - j]
+ vld1.16 d1[3], [r1], r10 @ data_in[i + factor * 7 - j]
+ subs r12, #1
+ vmlal.s16 q2, d0, d2
+ vmlal.s16 q3, d1, d3
+ bge LOOP_COEFF_LENGTH_GENERAL
+
+ @ Shift, saturate, and store the result.
+ vqshrn.s32 d0, q2, #12
+ vqshrn.s32 d1, q3, #12
+ vst1.16 {d0, d1}, [r2]!
+
+ add r11, r5, asl #3 @ r11 -> &data_in[i + factor * 8]
+ add r9, r5, asl #2 @ Counter i = delay + factor * 8.
+ cmp r9, r3 @ i < endpos - factor * 7 ?
+ blt LOOP_ENDPOS_GENERAL
+ asr r5, #1 @ Restore r5 to the value of factor.
+ b POST_LOOP_ENDPOS
+
+@ The case for factor == 2.
+LOOP_ENDPOS_FACTOR2:
+ @ Initializations.
+ vmov.i32 q2, #2048
+ vmov.i32 q3, #2048
+ sub r7, r8, #2
+ sub r12, r0, #1 @ coefficients_length - 1
+ sub r1, r11, r12, asl #1 @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_FACTOR2:
+ vld1.16 {d16[], d17[]}, [r7], r4 @ coefficients[j]
+ vld2.16 {d0, d1}, [r1]! @ data_in[]
+ vld2.16 {d2, d3}, [r1], r10 @ data_in[]
+ subs r12, #1
+ vmlal.s16 q2, d0, d16
+ vmlal.s16 q3, d2, d17
+ bge LOOP_COEFF_LENGTH_FACTOR2
+
+ @ Shift, saturate, and store the result.
+ vqshrn.s32 d0, q2, #12
+ vqshrn.s32 d1, q3, #12
+ vst1.16 {d0, d1}, [r2]!
+
+ add r11, r5, asl #4 @ r11 -> &data_in[i + factor * 8]
+ add r9, r5, asl #3 @ Counter i = delay + factor * 8.
+ cmp r9, r3 @ i < endpos - factor * 7 ?
+ blt LOOP_ENDPOS_FACTOR2
+ b POST_LOOP_ENDPOS
+
+@ The case for factor == 4.
+LOOP_ENDPOS_FACTOR4:
+ @ Initializations.
+ vmov.i32 q2, #2048
+ vmov.i32 q3, #2048
+ sub r7, r8, #2
+ sub r12, r0, #1 @ coefficients_length - 1
+ sub r1, r11, r12, asl #1 @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_FACTOR4:
+ vld1.16 {d16[], d17[]}, [r7], r4 @ coefficients[j]
+ vld4.16 {d0, d1, d2, d3}, [r1]! @ data_in[]
+ vld4.16 {d18, d19, d20, d21}, [r1], r10 @ data_in[]
+ subs r12, #1
+ vmlal.s16 q2, d0, d16
+ vmlal.s16 q3, d18, d17
+ bge LOOP_COEFF_LENGTH_FACTOR4
+
+ @ Shift, saturate, and store the result.
+ vqshrn.s32 d0, q2, #12
+ vqshrn.s32 d1, q3, #12
+ vst1.16 {d0, d1}, [r2]!
+
+ add r11, r5, asl #4 @ r11 -> &data_in[i + factor * 8]
+ add r9, r5, asl #3 @ Counter i = delay + factor * 8.
+ cmp r9, r3 @ i < endpos - factor * 7 ?
+ blt LOOP_ENDPOS_FACTOR4
+
+@
+@ Second part, do the rest iterations (if any).
+@
+
+POST_LOOP_ENDPOS:
+ add r3, r5, asl #3
+ sub r3, r5 @ Restore r3 to endpos.
+ cmp r9, r3
+ movge r0, #0
+ bge END
+
+LOOP2_ENDPOS:
+ @ Initializations.
+ mov r7, r8
+ sub r12, r0, #1 @ coefficients_length - 1
+ sub r6, r11, r12, asl #1 @ &data_in[i - j]
+
+ mov r1, #2048
+
+LOOP2_COEFF_LENGTH:
+ ldrsh r4, [r7, #-2]! @ coefficients[j]
+ ldrsh r10, [r6], #2 @ data_in[i - j]
+ smlabb r1, r4, r10, r1
+ subs r12, #1
+ bge LOOP2_COEFF_LENGTH
+
+ @ Shift, saturate, and store the result.
+ ssat r1, #16, r1, asr #12
+ strh r1, [r2], #2
+
+ add r11, r5, asl #1 @ r11 -> &data_in[i + factor]
+ add r9, r5 @ Counter i = delay + factor.
+ cmp r9, r3 @ i < endpos?
+ blt LOOP2_ENDPOS
+
+ mov r0, #0
+
+END:
+ pop {r4-r11}
+ bx lr
+
+.fnend
diff --git a/trunk/src/common_audio/signal_processing/energy.c b/trunk/src/common_audio/signal_processing/energy.c
new file mode 100644
index 0000000..e8fdf94
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/energy.c
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_Energy().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector, int vector_length, int* scale_factor)
+{
+ WebRtc_Word32 en = 0;
+ int i;
+ int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
+ int looptimes = vector_length;
+ WebRtc_Word16 *vectorptr = vector;
+
+ for (i = 0; i < looptimes; i++)
+ {
+ en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
+ vectorptr++;
+ }
+ *scale_factor = scaling;
+
+ return en;
+}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar.c b/trunk/src/common_audio/signal_processing/filter_ar.c
new file mode 100644
index 0000000..24e83a6
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar.c
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_FilterAR().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* a,
+ int a_length,
+ G_CONST WebRtc_Word16* x,
+ int x_length,
+ WebRtc_Word16* state,
+ int state_length,
+ WebRtc_Word16* state_low,
+ int state_low_length,
+ WebRtc_Word16* filtered,
+ WebRtc_Word16* filtered_low,
+ int filtered_low_length)
+{
+ WebRtc_Word32 o;
+ WebRtc_Word32 oLOW;
+ int i, j, stop;
+ G_CONST WebRtc_Word16* x_ptr = &x[0];
+ WebRtc_Word16* filteredFINAL_ptr = filtered;
+ WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
+
+ for (i = 0; i < x_length; i++)
+ {
+ // Calculate filtered[i] and filtered_low[i]
+ G_CONST WebRtc_Word16* a_ptr = &a[1];
+ WebRtc_Word16* filtered_ptr = &filtered[i - 1];
+ WebRtc_Word16* filtered_low_ptr = &filtered_low[i - 1];
+ WebRtc_Word16* state_ptr = &state[state_length - 1];
+ WebRtc_Word16* state_low_ptr = &state_low[state_length - 1];
+
+ o = (WebRtc_Word32)(*x_ptr++) << 12;
+ oLOW = (WebRtc_Word32)0;
+
+ stop = (i < a_length) ? i + 1 : a_length;
+ for (j = 1; j < stop; j++)
+ {
+ o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
+ oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
+ }
+ for (j = i + 1; j < a_length; j++)
+ {
+ o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
+ oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
+ }
+
+ o += (oLOW >> 12);
+ *filteredFINAL_ptr = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
+ *filteredFINAL_LOW_ptr++ = (WebRtc_Word16)(o - ((WebRtc_Word32)(*filteredFINAL_ptr++)
+ << 12));
+ }
+
+ // Save the filter state
+ if (x_length >= state_length)
+ {
+ WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
+ WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
+ } else
+ {
+ for (i = 0; i < state_length - x_length; i++)
+ {
+ state[i] = state[i + x_length];
+ state_low[i] = state_low[i + x_length];
+ }
+ for (i = 0; i < x_length; i++)
+ {
+ state[state_length - x_length + i] = filtered[i];
+ state[state_length - x_length + i] = filtered_low[i];
+ }
+ }
+
+ return x_length;
+}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
new file mode 100644
index 0000000..0402302
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include <assert.h>
+
+#include "signal_processing_library.h"
+
+// TODO(bjornv): Change the return type to report errors.
+
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+ int16_t* data_out,
+ const int16_t* __restrict coefficients,
+ int coefficients_length,
+ int data_length) {
+ int i = 0;
+ int j = 0;
+
+ assert(data_length > 0);
+ assert(coefficients_length > 1);
+
+ for (i = 0; i < data_length; i++) {
+ int32_t output = 0;
+ int32_t sum = 0;
+
+ for (j = coefficients_length - 1; j > 0; j--) {
+ sum += coefficients[j] * data_out[i - j];
+ }
+
+ output = coefficients[0] * data_in[i];
+ output -= sum;
+
+ // Saturate and store the output.
+ output = WEBRTC_SPL_SAT(134215679, output, -134217728);
+ data_out[i] = (int16_t)((output + 2048) >> 12);
+ }
+}
+
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
new file mode 100644
index 0000000..5591bb8
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
@@ -0,0 +1,223 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_FilterARFastQ12(), optimized for
+@ ARMv7 platform. The description header can be found in
+@ signal_processing_library.h
+@
+@ Output is bit-exact with the generic C code as in filter_ar_fast_q12.c, and
+@ the reference C code at end of this file.
+
+@ Assumptions:
+@ (1) data_length > 0
+@ (2) coefficients_length > 1
+
+@ Register usage:
+@
+@ r0: &data_in[i]
+@ r1: &data_out[i], for result ouput
+@ r2: &coefficients[0]
+@ r3: coefficients_length
+@ r4: Iteration counter for the outer loop.
+@ r5: data_out[j] as multiplication inputs
+@ r6: Calculated value for output data_out[]; interation counter for inner loop
+@ r7: Partial sum of a filtering multiplication results
+@ r8: Partial sum of a filtering multiplication results
+@ r9: &data_out[], for filtering input; data_in[i]
+@ r10: coefficients[j]
+@ r11: Scratch
+@ r12: &coefficients[j]
+
+.arch armv7-a
+
+.align 2
+.global WebRtcSpl_FilterARFastQ12
+
+WebRtcSpl_FilterARFastQ12:
+
+.fnstart
+
+.save {r4-r11}
+ push {r4-r11}
+
+ ldrsh r12, [sp, #32] @ data_length
+ subs r4, r12, #1
+ beq ODD_LENGTH @ jump if data_length == 1
+
+LOOP_LENGTH:
+ add r12, r2, r3, lsl #1
+ sub r12, #4 @ &coefficients[coefficients_length - 2]
+ sub r9, r1, r3, lsl #1
+ add r9, #2 @ &data_out[i - coefficients_length + 1]
+ ldr r5, [r9], #4 @ data_out[i - coefficients_length + {1,2}]
+
+ mov r7, #0 @ sum1
+ mov r8, #0 @ sum2
+ subs r6, r3, #3 @ Iteration counter for inner loop.
+ beq ODD_A_LENGTH @ branch if coefficients_length == 3
+ blt POST_LOOP_A_LENGTH @ branch if coefficients_length == 2
+
+LOOP_A_LENGTH:
+ ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
+ subs r6, #2
+ smlatt r8, r10, r5, r8 @ sum2 += coefficients[j] * data_out[i - j + 1];
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
+ smlabt r7, r10, r5, r7 @ coefficients[j - 1] * data_out[i - j + 1];
+ ldr r5, [r9], #4 @ data_out[i - j + 2], data_out[i - j + 3]
+ smlabb r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 2];
+ bgt LOOP_A_LENGTH
+ blt POST_LOOP_A_LENGTH
+
+ODD_A_LENGTH:
+ ldrsh r10, [r12, #2] @ Filter coefficients coefficients[2]
+ sub r12, #2 @ &coefficients[0]
+ smlabb r7, r10, r5, r7 @ sum1 += coefficients[2] * data_out[i - 2];
+ smlabt r8, r10, r5, r8 @ sum2 += coefficients[2] * data_out[i - 1];
+ ldr r5, [r9, #-2] @ data_out[i - 1], data_out[i]
+
+POST_LOOP_A_LENGTH:
+ ldr r10, [r12] @ coefficients[0], coefficients[1]
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
+
+ ldr r9, [r0], #4 @ data_in[i], data_in[i + 1]
+ smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
+ sub r6, r7 @ output1 -= sum1;
+
+ sbfx r11, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r11
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1], #2 @ Store data_out[i]
+
+ smlatb r8, r10, r6, r8 @ sum2 += coefficients[1] * data_out[i];
+ smulbt r6, r10, r9 @ output2 = coefficients[0] * data_in[i + 1];
+ sub r6, r8 @ output1 -= sum1;
+
+ sbfx r11, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r11
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1], #2 @ Store data_out[i + 1]
+
+ subs r4, #2
+ bgt LOOP_LENGTH
+ blt END @ For even data_length, it's done. Jump to END.
+
+@ Process i = data_length -1, for the case of an odd length.
+ODD_LENGTH:
+ add r12, r2, r3, lsl #1
+ sub r12, #4 @ &coefficients[coefficients_length - 2]
+ sub r9, r1, r3, lsl #1
+ add r9, #2 @ &data_out[i - coefficients_length + 1]
+ mov r7, #0 @ sum1
+ mov r8, #0 @ sum1
+ subs r6, r3, #2 @ inner loop counter
+ beq EVEN_A_LENGTH @ branch if coefficients_length == 2
+
+LOOP2_A_LENGTH:
+ ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
+ ldr r5, [r9], #4 @ data_out[i - j], data_out[i - j + 1]
+ subs r6, #2
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
+ smlabt r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 1];
+ bgt LOOP2_A_LENGTH
+ addlt r12, #2
+ blt POST_LOOP2_A_LENGTH
+
+EVEN_A_LENGTH:
+ ldrsh r10, [r12, #2] @ Filter coefficients coefficients[1]
+ ldrsh r5, [r9] @ data_out[i - 1]
+ smlabb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
+
+POST_LOOP2_A_LENGTH:
+ ldrsh r10, [r12] @ Filter coefficients coefficients[0]
+ ldrsh r9, [r0] @ data_in[i]
+ smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
+ sub r6, r7 @ output1 -= sum1;
+ sub r6, r8 @ output1 -= sum1;
+ sbfx r8, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r8
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1] @ Store the data_out[i]
+
+END:
+ pop {r4-r11}
+ bx lr
+
+.fnend
+
+
+@Reference C code:
+@
+@void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
+@ int16_t* data_out,
+@ int16_t* __restrict coefficients,
+@ int coefficients_length,
+@ int data_length) {
+@ int i = 0;
+@ int j = 0;
+@
+@ for (i = 0; i < data_length - 1; i += 2) {
+@ int32_t output1 = 0;
+@ int32_t sum1 = 0;
+@ int32_t output2 = 0;
+@ int32_t sum2 = 0;
+@
+@ for (j = coefficients_length - 1; j > 2; j -= 2) {
+@ sum1 += coefficients[j] * data_out[i - j];
+@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
+@ sum2 += coefficients[j] * data_out[i - j + 1];
+@ sum2 += coefficients[j - 1] * data_out[i - j + 2];
+@ }
+@
+@ if (j == 2) {
+@ sum1 += coefficients[2] * data_out[i - 2];
+@ sum2 += coefficients[2] * data_out[i - 1];
+@ }
+@
+@ sum1 += coefficients[1] * data_out[i - 1];
+@ output1 = coefficients[0] * data_in[i];
+@ output1 -= sum1;
+@ // Saturate and store the output.
+@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@
+@ sum2 += coefficients[1] * data_out[i];
+@ output2 = coefficients[0] * data_in[i + 1];
+@ output2 -= sum2;
+@ // Saturate and store the output.
+@ output2 = WEBRTC_SPL_SAT(134215679, output2, -134217728);
+@ data_out[i + 1] = (int16_t)((output2 + 2048) >> 12);
+@ }
+@
+@ if (i == data_length - 1) {
+@ int32_t output1 = 0;
+@ int32_t sum1 = 0;
+@
+@ for (j = coefficients_length - 1; j > 1; j -= 2) {
+@ sum1 += coefficients[j] * data_out[i - j];
+@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
+@ }
+@
+@ if (j == 1) {
+@ sum1 += coefficients[1] * data_out[i - 1];
+@ }
+@
+@ output1 = coefficients[0] * data_in[i];
+@ output1 -= sum1;
+@ // Saturate and store the output.
+@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@ }
+@}
diff --git a/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
new file mode 100644
index 0000000..19ad9b1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_FilterMAFastQ12().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_ptr,
+ WebRtc_Word16* out_ptr,
+ WebRtc_Word16* B,
+ WebRtc_Word16 B_length,
+ WebRtc_Word16 length)
+{
+ WebRtc_Word32 o;
+ int i, j;
+ for (i = 0; i < length; i++)
+ {
+ G_CONST WebRtc_Word16* b_ptr = &B[0];
+ G_CONST WebRtc_Word16* x_ptr = &in_ptr[i];
+
+ o = (WebRtc_Word32)0;
+
+ for (j = 0; j < B_length; j++)
+ {
+ o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
+ }
+
+ // If output is higher than 32768, saturate it. Same with negative side
+ // 2^27 = 134217728, which corresponds to 32768 in Q12
+
+ // Saturate the output
+ o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
+
+ *out_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
+ }
+ return;
+}
diff --git a/trunk/src/common_audio/signal_processing/get_hanning_window.c b/trunk/src/common_audio/signal_processing/get_hanning_window.c
new file mode 100644
index 0000000..6d67e60
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/get_hanning_window.c
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_GetHanningWindow().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// Hanning table with 256 entries
+static const WebRtc_Word16 kHanningTable[] = {
+ 1, 2, 6, 10, 15, 22, 30, 39,
+ 50, 62, 75, 89, 104, 121, 138, 157,
+ 178, 199, 222, 246, 271, 297, 324, 353,
+ 383, 413, 446, 479, 513, 549, 586, 624,
+ 663, 703, 744, 787, 830, 875, 920, 967,
+ 1015, 1064, 1114, 1165, 1218, 1271, 1325, 1381,
+ 1437, 1494, 1553, 1612, 1673, 1734, 1796, 1859,
+ 1924, 1989, 2055, 2122, 2190, 2259, 2329, 2399,
+ 2471, 2543, 2617, 2691, 2765, 2841, 2918, 2995,
+ 3073, 3152, 3232, 3312, 3393, 3475, 3558, 3641,
+ 3725, 3809, 3895, 3980, 4067, 4154, 4242, 4330,
+ 4419, 4509, 4599, 4689, 4781, 4872, 4964, 5057,
+ 5150, 5244, 5338, 5432, 5527, 5622, 5718, 5814,
+ 5910, 6007, 6104, 6202, 6299, 6397, 6495, 6594,
+ 6693, 6791, 6891, 6990, 7090, 7189, 7289, 7389,
+ 7489, 7589, 7690, 7790, 7890, 7991, 8091, 8192,
+ 8293, 8393, 8494, 8594, 8694, 8795, 8895, 8995,
+ 9095, 9195, 9294, 9394, 9493, 9593, 9691, 9790,
+ 9889, 9987, 10085, 10182, 10280, 10377, 10474, 10570,
+10666, 10762, 10857, 10952, 11046, 11140, 11234, 11327,
+11420, 11512, 11603, 11695, 11785, 11875, 11965, 12054,
+12142, 12230, 12317, 12404, 12489, 12575, 12659, 12743,
+12826, 12909, 12991, 13072, 13152, 13232, 13311, 13389,
+13466, 13543, 13619, 13693, 13767, 13841, 13913, 13985,
+14055, 14125, 14194, 14262, 14329, 14395, 14460, 14525,
+14588, 14650, 14711, 14772, 14831, 14890, 14947, 15003,
+15059, 15113, 15166, 15219, 15270, 15320, 15369, 15417,
+15464, 15509, 15554, 15597, 15640, 15681, 15721, 15760,
+15798, 15835, 15871, 15905, 15938, 15971, 16001, 16031,
+16060, 16087, 16113, 16138, 16162, 16185, 16206, 16227,
+16246, 16263, 16280, 16295, 16309, 16322, 16334, 16345,
+16354, 16362, 16369, 16374, 16378, 16382, 16383, 16384
+};
+
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
+{
+ int jj;
+ WebRtc_Word16 *vptr1;
+
+ WebRtc_Word32 index;
+ WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
+
+ factor = WebRtcSpl_DivW32W16(factor, size);
+ if (size < 513)
+ index = (WebRtc_Word32)-0x200000;
+ else
+ index = (WebRtc_Word32)-0x100000;
+ vptr1 = v;
+
+ for (jj = 0; jj < size; jj++)
+ {
+ index += factor;
+ (*vptr1++) = kHanningTable[index >> 22];
+ }
+
+}
diff --git a/trunk/src/common_audio/signal_processing/get_scaling_square.c b/trunk/src/common_audio/signal_processing/get_scaling_square.c
new file mode 100644
index 0000000..dccbf33
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/get_scaling_square.c
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_GetScalingSquare().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16 *in_vector, int in_vector_length, int times)
+{
+ int nbits = WebRtcSpl_GetSizeInBits(times);
+ int i;
+ WebRtc_Word16 smax = -1;
+ WebRtc_Word16 sabs;
+ WebRtc_Word16 *sptr = in_vector;
+ int t;
+ int looptimes = in_vector_length;
+
+ for (i = looptimes; i > 0; i--)
+ {
+ sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
+ smax = (sabs > smax ? sabs : smax);
+ }
+ t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+ if (smax == 0)
+ {
+ return 0; // Since norm(0) returns 0
+ } else
+ {
+ return (t > nbits) ? 0 : nbits - t;
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c b/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
new file mode 100644
index 0000000..5a9e577
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the iLBC specific functions
+ * WebRtcSpl_ScaleAndAddVectorsWithRound()
+ * WebRtcSpl_ReverseOrderMultArrayElements()
+ * WebRtcSpl_ElementwiseVectorMult()
+ * WebRtcSpl_AddVectorsAndShift()
+ * WebRtcSpl_AddAffineVectorToVector()
+ * WebRtcSpl_AffineTransformVector()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16 *vector1, WebRtc_Word16 scale1,
+ WebRtc_Word16 *vector2, WebRtc_Word16 scale2,
+ WebRtc_Word16 right_shifts, WebRtc_Word16 *out,
+ WebRtc_Word16 vector_length)
+{
+ int i;
+ WebRtc_Word16 roundVal;
+ roundVal = 1 << right_shifts;
+ roundVal = roundVal >> 1;
+ for (i = 0; i < vector_length; i++)
+ {
+ out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(vector1[i], scale1)
+ + WEBRTC_SPL_MUL_16_16(vector2[i], scale2) + roundVal) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
+ G_CONST WebRtc_Word16 *win,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+ WebRtc_Word16 *outptr = out;
+ G_CONST WebRtc_Word16 *inptr = in;
+ G_CONST WebRtc_Word16 *winptr = win;
+ for (i = 0; i < vector_length; i++)
+ {
+ (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
+ *winptr--, right_shifts);
+ }
+}
+
+void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
+ G_CONST WebRtc_Word16 *win, WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+ WebRtc_Word16 *outptr = out;
+ G_CONST WebRtc_Word16 *inptr = in;
+ G_CONST WebRtc_Word16 *winptr = win;
+ for (i = 0; i < vector_length; i++)
+ {
+ (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
+ *winptr++, right_shifts);
+ }
+}
+
+void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in1,
+ G_CONST WebRtc_Word16 *in2, WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+ WebRtc_Word16 *outptr = out;
+ G_CONST WebRtc_Word16 *in1ptr = in1;
+ G_CONST WebRtc_Word16 *in2ptr = in2;
+ for (i = vector_length; i > 0; i--)
+ {
+ (*outptr++) = (WebRtc_Word16)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
+ WebRtc_Word16 gain, WebRtc_Word32 add_constant,
+ WebRtc_Word16 right_shifts, int vector_length)
+{
+ WebRtc_Word16 *inPtr;
+ WebRtc_Word16 *outPtr;
+ int i;
+
+ inPtr = in;
+ outPtr = out;
+ for (i = 0; i < vector_length; i++)
+ {
+ (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+ + (WebRtc_Word32)add_constant) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
+ WebRtc_Word16 gain, WebRtc_Word32 add_constant,
+ WebRtc_Word16 right_shifts, int vector_length)
+{
+ WebRtc_Word16 *inPtr;
+ WebRtc_Word16 *outPtr;
+ int i;
+
+ inPtr = in;
+ outPtr = out;
+ for (i = 0; i < vector_length; i++)
+ {
+ (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+ + (WebRtc_Word32)add_constant) >> right_shifts);
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/include/signal_processing_library.h b/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
new file mode 100644
index 0000000..87b8f29
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
@@ -0,0 +1,1667 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes all of the fix point signal processing library (SPL) function
+ * descriptions and declarations.
+ * For specific function calls, see bottom of file.
+ */
+
+#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+#include <string.h>
+#include "typedefs.h"
+
+#ifdef ARM_WINM
+#include <Armintr.h> // intrinsic file for windows mobile
+#endif
+
+// Macros specific for the fixed point implementation
+#define WEBRTC_SPL_WORD16_MAX 32767
+#define WEBRTC_SPL_WORD16_MIN -32768
+#define WEBRTC_SPL_WORD32_MAX (WebRtc_Word32)0x7fffffff
+#define WEBRTC_SPL_WORD32_MIN (WebRtc_Word32)0x80000000
+#define WEBRTC_SPL_MAX_LPC_ORDER 14
+#define WEBRTC_SPL_MAX_SEED_USED 0x80000000L
+#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
+#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
+#define WEBRTC_SPL_ABS_W16(a) \
+ (((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
+#define WEBRTC_SPL_ABS_W32(a) \
+ (((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
+
+#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
+#define WEBRTC_SPL_GET_BYTE(a, nr) (((WebRtc_Word8 *)a)[nr])
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ (((WebRtc_Word8 *)d_ptr)[index] = (val))
+#elif defined WEBRTC_BIG_ENDIAN
+#define WEBRTC_SPL_GET_BYTE(a, nr) \
+ ((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
+ ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
+ & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
+#else
+#define WEBRTC_SPL_GET_BYTE(a,nr) \
+ ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
+ ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
+ & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
+ ((val) << (8 * ((index) & 0x1)))
+#endif
+
+#define WEBRTC_SPL_MUL(a, b) \
+ ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UMUL(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
+#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_16_16(a, b) \
+ ((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
+ (((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_32_16(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
+#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_MUL_16_U16(a, b) \
+ ((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_DIV(a, b) \
+ ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UDIV(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
+
+#ifndef WEBRTC_ARCH_ARM_V7A
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+ ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+ (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
+ + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
+ ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
+ + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
+#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
+ ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
+ (WebRtc_Word16)(a32 >> 16)), b32) + \
+ (WEBRTC_SPL_MUL_16_32_RSFT16(( \
+ (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
+#endif
+
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
+
+#ifdef ARM_WINM
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+ _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
+#endif
+
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
+ (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+ ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
+ (((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
+
+// C + the 32 most significant bits of A * B
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+ (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
+
+#define WEBRTC_SPL_ADD_SAT_W32(a, b) WebRtcSpl_AddSatW32(a, b)
+#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
+#define WEBRTC_SPL_MUL_32_16(a, b) ((a) * (b))
+
+#define WEBRTC_SPL_SUB_SAT_W32(a, b) WebRtcSpl_SubSatW32(a, b)
+#define WEBRTC_SPL_ADD_SAT_W16(a, b) WebRtcSpl_AddSatW16(a, b)
+#define WEBRTC_SPL_SUB_SAT_W16(a, b) WebRtcSpl_SubSatW16(a, b)
+
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_IS_NEG(a) ((a) & 0x80000000)
+// Shifting with negative numbers allowed
+// Positive means left shift
+#define WEBRTC_SPL_SHIFT_W16(x, c) \
+ (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) \
+ (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+
+// Shifting with negative numbers not allowed
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_RSHIFT_W16(x, c) ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W16(x, c) ((x) << (c))
+#define WEBRTC_SPL_RSHIFT_W32(x, c) ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
+
+#define WEBRTC_SPL_RSHIFT_U16(x, c) ((WebRtc_UWord16)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U16(x, c) ((WebRtc_UWord16)(x) << (c))
+#define WEBRTC_SPL_RSHIFT_U32(x, c) ((WebRtc_UWord32)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U32(x, c) ((WebRtc_UWord32)(x) << (c))
+
+#define WEBRTC_SPL_VNEW(t, n) (t *) malloc (sizeof (t) * (n))
+#define WEBRTC_SPL_FREE free
+
+#define WEBRTC_SPL_RAND(a) \
+ ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
+ memcpy(v1, v2, (length) * sizeof(char))
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+ memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
+ memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+// inline functions:
+#include "spl_inl.h"
+
+// Get SPL Version
+WebRtc_Word16 WebRtcSpl_get_version(char* version,
+ WebRtc_Word16 length_in_bytes);
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16* in_vector,
+ int in_vector_length,
+ int times);
+
+// Copy and set operations. Implementation in copy_set_operations.c.
+// Descriptions at bottom of file.
+void WebRtcSpl_MemSetW16(WebRtc_Word16* vector,
+ WebRtc_Word16 set_value,
+ int vector_length);
+void WebRtcSpl_MemSetW32(WebRtc_Word32* vector,
+ WebRtc_Word32 set_value,
+ int vector_length);
+void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* out_vector,
+ WebRtc_Word16* in_vector,
+ int vector_length);
+WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16* in_vector,
+ WebRtc_Word16 in_vector_length,
+ WebRtc_Word16 samples,
+ WebRtc_Word16* out_vector);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16* vector,
+ WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32* vector,
+ WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16* vector,
+ WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32* vector,
+ WebRtc_Word16 vector_length);
+// End: Copy and set operations.
+
+// Minimum and maximum operations. Implementation in min_max_operations.c.
+// Descriptions at bottom of file.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+
+WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16* vector,
+ WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector,
+ WebRtc_Word16 length);
+// End: Minimum and maximum operations.
+
+// Vector scaling operations. Implementation in vector_scaling_operations.c.
+// Description at bottom of file.
+void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16* out_vector,
+ WebRtc_Word16 vector_length,
+ G_CONST WebRtc_Word16* in_vector,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
+ WebRtc_Word16 vector_length,
+ G_CONST WebRtc_Word32* in_vector,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
+ WebRtc_Word16 vector_length,
+ G_CONST WebRtc_Word32* in_vector,
+ WebRtc_Word16 right_shifts);
+
+void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
+ WebRtc_Word16* out_vector,
+ WebRtc_Word16 gain,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16* in_vector,
+ WebRtc_Word16* out_vector,
+ WebRtc_Word16 gain,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16* in_vector1,
+ WebRtc_Word16 gain1, int right_shifts1,
+ G_CONST WebRtc_Word16* in_vector2,
+ WebRtc_Word16 gain2, int right_shifts2,
+ WebRtc_Word16* out_vector,
+ int vector_length);
+// End: Vector scaling operations.
+
+// iLBC specific functions. Implementations in ilbc_specific_functions.c.
+// Description at bottom of file.
+void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16* in_vector1,
+ WebRtc_Word16 scale1,
+ WebRtc_Word16* in_vector2,
+ WebRtc_Word16 scale2,
+ WebRtc_Word16 right_shifts,
+ WebRtc_Word16* out_vector,
+ WebRtc_Word16 vector_length);
+void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16* out_vector,
+ G_CONST WebRtc_Word16* in_vector,
+ G_CONST WebRtc_Word16* window,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16* out_vector,
+ G_CONST WebRtc_Word16* in_vector,
+ G_CONST WebRtc_Word16* window,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16* out_vector,
+ G_CONST WebRtc_Word16* in_vector1,
+ G_CONST WebRtc_Word16* in_vector2,
+ WebRtc_Word16 vector_length,
+ WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16* out_vector,
+ WebRtc_Word16* in_vector,
+ WebRtc_Word16 gain,
+ WebRtc_Word32 add_constant,
+ WebRtc_Word16 right_shifts,
+ int vector_length);
+void WebRtcSpl_AffineTransformVector(WebRtc_Word16* out_vector,
+ WebRtc_Word16* in_vector,
+ WebRtc_Word16 gain,
+ WebRtc_Word32 add_constant,
+ WebRtc_Word16 right_shifts,
+ int vector_length);
+// End: iLBC specific functions.
+
+// Signal processing operations. Descriptions at bottom of this file.
+int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* vector,
+ int vector_length, int order,
+ WebRtc_Word32* result_vector,
+ int* scale);
+WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32* auto_corr,
+ WebRtc_Word16* lpc_coef,
+ WebRtc_Word16* refl_coef,
+ WebRtc_Word16 order);
+void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16* refl_coef,
+ int use_order,
+ WebRtc_Word16* lpc_coef);
+void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* lpc_coef,
+ int use_order,
+ WebRtc_Word16* refl_coef);
+void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32* auto_corr,
+ int use_order,
+ WebRtc_Word16* refl_coef);
+void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_corr,
+ WebRtc_Word16* vector1,
+ WebRtc_Word16* vector2,
+ WebRtc_Word16 dim_vector,
+ WebRtc_Word16 dim_cross_corr,
+ WebRtc_Word16 right_shifts,
+ WebRtc_Word16 step_vector2);
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16* window, WebRtc_Word16 size);
+void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
+ int vector_length,
+ WebRtc_Word16* out_vector);
+// End: Signal processing operations.
+
+// Randomization functions. Implementations collected in randomization_functions.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+ WebRtc_Word16 vector_length,
+ WebRtc_UWord32* seed);
+// End: Randomization functions.
+
+// Math functions
+WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value);
+WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value);
+
+// Divisions. Implementations collected in division_operations.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den);
+WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den);
+WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
+ WebRtc_Word16 den_low);
+// End: Divisions.
+
+WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector,
+ int vector_length,
+ int* scale_factor);
+
+WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16* vector1,
+ WebRtc_Word16* vector2,
+ int vector_length,
+ int scaling);
+
+// Filter operations.
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
+ G_CONST WebRtc_Word16* in_vector, int in_vector_length,
+ WebRtc_Word16* filter_state, int filter_state_length,
+ WebRtc_Word16* filter_state_low,
+ int filter_state_low_length, WebRtc_Word16* out_vector,
+ WebRtc_Word16* out_vector_low, int out_vector_low_length);
+
+void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
+ WebRtc_Word16* out_vector,
+ WebRtc_Word16* ma_coef,
+ WebRtc_Word16 ma_coef_length,
+ WebRtc_Word16 vector_length);
+
+// Performs a AR filtering on a vector in Q12
+// Input:
+// - data_in : Input samples
+// - data_out : State information in positions
+// data_out[-order] .. data_out[-1]
+// - coefficients : Filter coefficients (in Q12)
+// - coefficients_length: Number of coefficients (order+1)
+// - data_length : Number of samples to be filtered
+// Output:
+// - data_out : Filtered samples
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+ int16_t* data_out,
+ const int16_t* __restrict coefficients,
+ int coefficients_length,
+ int data_length);
+
+// Performs a MA down sampling filter on a vector
+// Input:
+// - data_in : Input samples (state in positions
+// data_in[-order] .. data_in[-1])
+// - data_in_length : Number of samples in |data_in| to be filtered.
+// This must be at least
+// |delay| + |factor|*(|out_vector_length|-1) + 1)
+// - data_out_length : Number of down sampled samples desired
+// - coefficients : Filter coefficients (in Q12)
+// - coefficients_length: Number of coefficients (order+1)
+// - factor : Decimation factor
+// - delay : Delay of filter (compensated for in out_vector)
+// Output:
+// - data_out : Filtered samples
+// Return value : 0 if OK, -1 if |in_vector| is too short
+int WebRtcSpl_DownsampleFast(const int16_t* data_in,
+ int data_in_length,
+ int16_t* data_out,
+ int data_out_length,
+ const int16_t* __restrict coefficients,
+ int coefficients_length,
+ int factor,
+ int delay);
+
+// End: Filter operations.
+
+// FFT operations
+
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
+
+// Treat a 16-bit complex data buffer |complex_data| as an array of 32-bit
+// values, and swap elements whose indexes are bit-reverses of each other.
+//
+// Input:
+// - complex_data : Complex data buffer containing 2^|stages| real
+// elements interleaved with 2^|stages| imaginary
+// elements: [Re Im Re Im Re Im....]
+// - stages : Number of FFT stages. Must be at least 3 and at most
+// 10, since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// Output:
+// - complex_data : The complex data buffer.
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
+
+// End: FFT operations
+
+/************************************************************
+ *
+ * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
+ *
+ ************************************************************/
+
+/*******************************************************************
+ * resample.c
+ *
+ * Includes the following resampling combinations
+ * 22 kHz -> 16 kHz
+ * 16 kHz -> 22 kHz
+ * 22 kHz -> 8 kHz
+ * 8 kHz -> 22 kHz
+ *
+ ******************************************************************/
+
+// state structure for 22 -> 16 resampler
+typedef struct
+{
+ WebRtc_Word32 S_22_44[8];
+ WebRtc_Word32 S_44_32[8];
+ WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State22khzTo16khz;
+
+void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
+ WebRtc_Word16* out,
+ WebRtcSpl_State22khzTo16khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
+
+// state structure for 16 -> 22 resampler
+typedef struct
+{
+ WebRtc_Word32 S_16_32[8];
+ WebRtc_Word32 S_32_22[8];
+} WebRtcSpl_State16khzTo22khz;
+
+void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
+ WebRtc_Word16* out,
+ WebRtcSpl_State16khzTo22khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
+
+// state structure for 22 -> 8 resampler
+typedef struct
+{
+ WebRtc_Word32 S_22_22[16];
+ WebRtc_Word32 S_22_16[8];
+ WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State22khzTo8khz;
+
+void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State22khzTo8khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
+
+// state structure for 8 -> 22 resampler
+typedef struct
+{
+ WebRtc_Word32 S_8_16[8];
+ WebRtc_Word32 S_16_11[8];
+ WebRtc_Word32 S_11_22[8];
+} WebRtcSpl_State8khzTo22khz;
+
+void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State8khzTo22khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
+
+/*******************************************************************
+ * resample_fractional.c
+ * Functions for internal use in the other resample functions
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 32 kHz
+ * 32 kHz -> 24 kHz
+ * 44 kHz -> 32 kHz
+ *
+ ******************************************************************/
+
+void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+ const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+ const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+ const WebRtc_Word32 K);
+
+/*******************************************************************
+ * resample_48khz.c
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 16 kHz
+ * 16 kHz -> 48 kHz
+ * 48 kHz -> 8 kHz
+ * 8 kHz -> 48 kHz
+ *
+ ******************************************************************/
+
+typedef struct
+{
+ WebRtc_Word32 S_48_48[16];
+ WebRtc_Word32 S_48_32[8];
+ WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State48khzTo16khz;
+
+void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State48khzTo16khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
+
+typedef struct
+{
+ WebRtc_Word32 S_16_32[8];
+ WebRtc_Word32 S_32_24[8];
+ WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State16khzTo48khz;
+
+void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State16khzTo48khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
+
+typedef struct
+{
+ WebRtc_Word32 S_48_24[8];
+ WebRtc_Word32 S_24_24[16];
+ WebRtc_Word32 S_24_16[8];
+ WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State48khzTo8khz;
+
+void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State48khzTo8khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
+
+typedef struct
+{
+ WebRtc_Word32 S_8_16[8];
+ WebRtc_Word32 S_16_12[8];
+ WebRtc_Word32 S_12_24[8];
+ WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State8khzTo48khz;
+
+void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State8khzTo48khz* state,
+ WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
+
+/*******************************************************************
+ * resample_by_2.c
+ *
+ * Includes down and up sampling by a factor of two.
+ *
+ ******************************************************************/
+
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState);
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
+ WebRtc_Word32* filtState);
+
+/************************************************************
+ * END OF RESAMPLING FUNCTIONS
+ ************************************************************/
+void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data,
+ WebRtc_Word16* low_band,
+ WebRtc_Word16* high_band,
+ WebRtc_Word32* filter_state1,
+ WebRtc_Word32* filter_state2);
+void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
+ const WebRtc_Word16* high_band,
+ WebRtc_Word16* out_data,
+ WebRtc_Word32* filter_state1,
+ WebRtc_Word32* filter_state2);
+
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+//
+// WebRtcSpl_AddSatW16(...)
+// WebRtcSpl_AddSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
+// the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+// - var1 : Input variable 1
+// - var2 : Input variable 2
+//
+// Return value : Added and saturated value
+//
+
+//
+// WebRtcSpl_SubSatW16(...)
+// WebRtcSpl_SubSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
+// of the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+// - var1 : Input variable 1
+// - var2 : Input variable 2
+//
+// Returned value : Subtracted and saturated value
+//
+
+//
+// WebRtcSpl_GetSizeInBits(...)
+//
+// Returns the # of bits that are needed at the most to represent the number
+// specified by the |value| parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bits needed to represent |value|
+//
+
+//
+// WebRtcSpl_NormW32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormW16(...)
+//
+// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormU32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the unsigned
+// 32-bit number specified by the |value| parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_GetScalingSquare(...)
+//
+// Returns the # of bits required to scale the samples specified in the
+// |in_vector| parameter so that, if the squares of the samples are added the
+// # of times specified by the |times| parameter, the 32-bit addition will not
+// overflow (result in WebRtc_Word32).
+//
+// Input:
+// - in_vector : Input vector to check scaling on
+// - in_vector_length : Samples in |in_vector|
+// - times : Number of additions to be performed
+//
+// Return value : Number of right bit shifts needed to avoid
+// overflow in the addition calculation
+//
+
+//
+// WebRtcSpl_MemSetW16(...)
+//
+// Sets all the values in the WebRtc_Word16 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+// - vector : Pointer to the WebRtc_Word16 vector
+// - set_value : Value specified
+// - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemSetW32(...)
+//
+// Sets all the values in the WebRtc_Word32 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+// - vector : Pointer to the WebRtc_Word16 vector
+// - set_value : Value specified
+// - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemCpyReversedOrder(...)
+//
+// Copies all the values from the source WebRtc_Word16 vector |in_vector| to a
+// destination WebRtc_Word16 vector |out_vector|. It is done in reversed order,
+// meaning that the first sample of |in_vector| is copied to the last sample of
+// the |out_vector|. The procedure continues until the last sample of
+// |in_vector| has been copied to the first sample of |out_vector|. This
+// creates a reversed vector. Used in e.g. prediction in iLBC.
+//
+// Input:
+// - in_vector : Pointer to the first sample in a WebRtc_Word16 vector
+// of length |length|
+// - vector_length : Number of elements to copy
+//
+// Output:
+// - out_vector : Pointer to the last sample in a WebRtc_Word16 vector
+// of length |length|
+//
+
+//
+// WebRtcSpl_CopyFromEndW16(...)
+//
+// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
+// to the vector |out_vector|.
+//
+// Input:
+// - in_vector : Input vector
+// - in_vector_length : Number of samples in |in_vector|
+// - samples : Number of samples to extract (from right side)
+// from |in_vector|
+//
+// Output:
+// - out_vector : Vector with the requested samples
+//
+// Return value : Number of copied samples in |out_vector|
+//
+
+//
+// WebRtcSpl_ZerosArrayW16(...)
+// WebRtcSpl_ZerosArrayW32(...)
+//
+// Inserts the value "zero" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+// - vector_length : Number of samples in vector
+//
+// Output:
+// - vector : Vector containing all zeros
+//
+// Return value : Number of samples in vector
+//
+
+//
+// WebRtcSpl_OnesArrayW16(...)
+// WebRtcSpl_OnesArrayW32(...)
+//
+// Inserts the value "one" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+// - vector_length : Number of samples in vector
+//
+// Output:
+// - vector : Vector containing all ones
+//
+// Return value : Number of samples in vector
+//
+
+//
+// WebRtcSpl_MinValueW16(...)
+// WebRtcSpl_MinValueW32(...)
+//
+// Returns the minimum value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxValueW16(...)
+// WebRtcSpl_MaxValueW32(...)
+//
+// Returns the maximum value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Maximum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsValueW16(...)
+// WebRtcSpl_MaxAbsValueW32(...)
+//
+// Returns the largest absolute value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsIndexW16(...)
+//
+// Returns the vector index to the largest absolute value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Index to maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MinIndexW16(...)
+// WebRtcSpl_MinIndexW32(...)
+//
+// Returns the vector index to the minimum sample value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Index to minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxIndexW16(...)
+// WebRtcSpl_MaxIndexW32(...)
+//
+// Returns the vector index to the maximum sample value of a vector
+//
+// Input:
+// - vector : Input vector
+// - vector_length : Number of samples in vector
+//
+// Return value : Index to maximum sample value in vector
+//
+
+//
+// WebRtcSpl_VectorBitShiftW16(...)
+// WebRtcSpl_VectorBitShiftW32(...)
+//
+// Bit shifts all the values in a vector up or downwards. Different calls for
+// WebRtc_Word16 and WebRtc_Word32 vectors respectively.
+//
+// Input:
+// - vector_length : Length of vector
+// - in_vector : Pointer to the vector that should be bit shifted
+// - right_shifts : Number of right bit shifts (negative value gives left
+// shifts)
+//
+// Output:
+// - out_vector : Pointer to the result vector (can be the same as
+// |in_vector|)
+//
+
+//
+// WebRtcSpl_VectorBitShiftW32ToW16(...)
+//
+// Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
+// stores the result as a WebRtc_Word16 vector
+//
+// Input:
+// - vector_length : Length of vector
+// - in_vector : Pointer to the vector that should be bit shifted
+// - right_shifts : Number of right bit shifts (negative value gives left
+// shifts)
+//
+// Output:
+// - out_vector : Pointer to the result vector (can be the same as
+// |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVector(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (gain*in_vector[k])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Scaling gain
+// - vector_length : Elements in the |in_vector|
+// - right_shifts : Number of right bit shifts applied
+//
+// Output:
+// - out_vector : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVectorWithSat(...)
+//
+// Performs the vector operation:
+// out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Scaling gain
+// - vector_length : Elements in the |in_vector|
+// - right_shifts : Number of right bit shifts applied
+//
+// Output:
+// - out_vector : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectors(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
+// + (gain2*in_vector2[k])>>right_shifts2
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - gain1 : Gain to be used for vector 1
+// - right_shifts1 : Right bit shift to be used for vector 1
+// - in_vector2 : Input vector 2
+// - gain2 : Gain to be used for vector 2
+// - right_shifts2 : Right bit shift to be used for vector 2
+// - vector_length : Elements in the input vectors
+//
+// Output:
+// - out_vector : Output vector
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectorsWithRound(...)
+//
+// Performs the vector operation:
+//
+// out_vector[k] = ((scale1*in_vector1[k]) + (scale2*in_vector2[k])
+// + round_value) >> right_shifts
+//
+// where:
+//
+// round_value = (1<<right_shifts)>>1
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - scale1 : Gain to be used for vector 1
+// - in_vector2 : Input vector 2
+// - scale2 : Gain to be used for vector 2
+// - right_shifts : Number of right bit shifts to be applied
+// - vector_length : Number of elements in the input vectors
+//
+// Output:
+// - out_vector : Output vector
+//
+
+//
+// WebRtcSpl_ReverseOrderMultArrayElements(...)
+//
+// Performs the vector operation:
+// out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - window : Window vector (should be reversed). The pointer
+// should be set to the last value in the vector
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in |in_vector|
+//
+// Output:
+// - out_vector : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ElementwiseVectorMult(...)
+//
+// Performs the vector operation:
+// out_vector[n] = (in_vector[n]*window[n])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - window : Window vector.
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in |in_vector|
+//
+// Output:
+// - out_vector : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_AddVectorsAndShift(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - in_vector2 : Input vector 2
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in |in_vector1| and |in_vector2|
+//
+// Output:
+// - out_vector : Output vector (can be same as |in_vector1|)
+//
+
+//
+// WebRtcSpl_AddAffineVectorToVector(...)
+//
+// Adds an affine transformed vector to another vector |out_vector|, i.e,
+// performs
+// out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Gain value, used to multiply the in vector with
+// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
+// but others can be used as well
+// - right_shifts : Number of right bit shifts (0-16)
+// - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+// - out_vector : Vector with the output
+//
+
+//
+// WebRtcSpl_AffineTransformVector(...)
+//
+// Affine transforms a vector, i.e, performs
+// out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Gain value, used to multiply the in vector with
+// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
+// but others can be used as well
+// - right_shifts : Number of right bit shifts (0-16)
+// - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+// - out_vector : Vector with the output
+//
+
+//
+// WebRtcSpl_AutoCorrelation(...)
+//
+// A 32-bit fix-point implementation of auto-correlation computation
+//
+// Input:
+// - vector : Vector to calculate autocorrelation upon
+// - vector_length : Length (in samples) of |vector|
+// - order : The order up to which the autocorrelation should be
+// calculated
+//
+// Output:
+// - result_vector : auto-correlation values (values should be seen
+// relative to each other since the absolute values
+// might have been down shifted to avoid overflow)
+//
+// - scale : The number of left shifts required to obtain the
+// auto-correlation in Q0
+//
+// Return value : Number of samples in |result_vector|, i.e., (order+1)
+//
+
+//
+// WebRtcSpl_LevinsonDurbin(...)
+//
+// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
+// does NOT use the 64 bit class
+//
+// Input:
+// - auto_corr : Vector with autocorrelation values of length >=
+// |use_order|+1
+// - use_order : The LPC filter order (support up to order 20)
+//
+// Output:
+// - lpc_coef : lpc_coef[0..use_order] LPC coefficients in Q12
+// - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
+// Q15
+//
+// Return value : 1 for stable 0 for unstable
+//
+
+//
+// WebRtcSpl_ReflCoefToLpc(...)
+//
+// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
+// This version is a 16 bit operation.
+//
+// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
+// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
+// "rare" cases even if the reflection coefficients are stable.
+//
+// Input:
+// - refl_coef : Reflection coefficients in Q15 that should be converted
+// to LPC coefficients
+// - use_order : Number of coefficients in |refl_coef|
+//
+// Output:
+// - lpc_coef : LPC coefficients in Q12
+//
+
+//
+// WebRtcSpl_LpcToReflCoef(...)
+//
+// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
+// This version is a 16 bit operation.
+// The conversion is implemented by the step-down algorithm.
+//
+// Input:
+// - lpc_coef : LPC coefficients in Q12, that should be converted to
+// reflection coefficients
+// - use_order : Number of coefficients in |lpc_coef|
+//
+// Output:
+// - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_AutoCorrToReflCoef(...)
+//
+// Calculates reflection coefficients (16 bit) from auto-correlation values
+//
+// Input:
+// - auto_corr : Auto-correlation values
+// - use_order : Number of coefficients wanted be calculated
+//
+// Output:
+// - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_CrossCorrelation(...)
+//
+// Calculates the cross-correlation between two sequences |vector1| and
+// |vector2|. |vector1| is fixed and |vector2| slides as the pointer is
+// increased with the amount |step_vector2|
+//
+// Input:
+// - vector1 : First sequence (fixed throughout the correlation)
+// - vector2 : Second sequence (slides |step_vector2| for each
+// new correlation)
+// - dim_vector : Number of samples to use in the cross-correlation
+// - dim_cross_corr : Number of cross-correlations to calculate (the
+// start position for |vector2| is updated for each
+// new one)
+// - right_shifts : Number of right bit shifts to use. This will
+// become the output Q-domain.
+// - step_vector2 : How many (positive or negative) steps the
+// |vector2| pointer should be updated for each new
+// cross-correlation value.
+//
+// Output:
+// - cross_corr : The cross-correlation in Q(-right_shifts)
+//
+
+//
+// WebRtcSpl_GetHanningWindow(...)
+//
+// Creates (the first half of) a Hanning window. Size must be at least 1 and
+// at most 512.
+//
+// Input:
+// - size : Length of the requested Hanning window (1 to 512)
+//
+// Output:
+// - window : Hanning vector in Q14.
+//
+
+//
+// WebRtcSpl_SqrtOfOneMinusXSquared(...)
+//
+// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
+// |in_vector|. Input and output values are in Q15.
+//
+// Inputs:
+// - in_vector : Values to calculate sqrt(1 - x^2) of
+// - vector_length : Length of vector |in_vector|
+//
+// Output:
+// - out_vector : Output values in Q15
+//
+
+//
+// WebRtcSpl_IncreaseSeed(...)
+//
+// Increases the seed (and returns the new value)
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : The new seed value
+//
+
+//
+// WebRtcSpl_RandU(...)
+//
+// Produces a uniformly distributed value in the WebRtc_Word16 range
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : Uniformly distributed value in the range
+// [Word16_MIN...Word16_MAX]
+//
+
+//
+// WebRtcSpl_RandN(...)
+//
+// Produces a normal distributed value in the WebRtc_Word16 range
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : N(0,1) value in the Q13 domain
+//
+
+//
+// WebRtcSpl_RandUArray(...)
+//
+// Produces a uniformly distributed vector with elements in the WebRtc_Word16
+// range
+//
+// Input:
+// - vector_length : Samples wanted in the vector
+// - seed : Seed for random calculation
+//
+// Output:
+// - vector : Vector with the uniform values
+// - seed : Updated seed value
+//
+// Return value : Number of samples in vector, i.e., |vector_length|
+//
+
+//
+// WebRtcSpl_Sqrt(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// A sixth order Taylor Series expansion is used here to compute the square
+// root of a number y^0.5 = (1+x)^0.5
+// where
+// x = y-1
+// = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+// 0.5 <= x < 1
+//
+// Input:
+// - value : Value to calculate sqrt of
+//
+// Return value : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_SqrtFloor(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// An iterative 4 cylce/bit routine
+//
+// Input:
+// - value : Value to calculate sqrt of
+//
+// Return value : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_DivU32U16(...)
+//
+// Divides a WebRtc_UWord32 |num| by a WebRtc_UWord16 |den|.
+//
+// If |den|==0, (WebRtc_UWord32)0xFFFFFFFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a WebRtc_UWord32), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|.
+//
+// If |den|==0, (WebRtc_Word32)0x7FFFFFFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a WebRtc_Word32), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16ResW16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// result is less than 32768, otherwise an unpredictable result will occur.
+//
+// If |den|==0, (WebRtc_Word16)0x7FFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a WebRtc_Word16), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivResultInQ31(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// absolute value of the denominator is larger than the numerator, otherwise
+// an unpredictable result will occur.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division in Q31.
+//
+
+//
+// WebRtcSpl_DivW32HiLow(...)
+//
+// Divides a WebRtc_Word32 |num| by a denominator in hi, low format. The
+// absolute value of the denominator has to be larger (or equal to) the
+// numerator.
+//
+// Input:
+// - num : Numerator
+// - den_hi : High part of denominator
+// - den_low : Low part of denominator
+//
+// Return value : Divided value in Q31
+//
+
+//
+// WebRtcSpl_Energy(...)
+//
+// Calculates the energy of a vector
+//
+// Input:
+// - vector : Vector which the energy should be calculated on
+// - vector_length : Number of samples in vector
+//
+// Output:
+// - scale_factor : Number of left bit shifts needed to get the physical
+// energy value, i.e, to get the Q0 value
+//
+// Return value : Energy value in Q(-|scale_factor|)
+//
+
+//
+// WebRtcSpl_FilterAR(...)
+//
+// Performs a 32-bit AR filtering on a vector in Q12
+//
+// Input:
+// - ar_coef : AR-coefficient vector (values in Q12),
+// ar_coef[0] must be 4096.
+// - ar_coef_length : Number of coefficients in |ar_coef|.
+// - in_vector : Vector to be filtered.
+// - in_vector_length : Number of samples in |in_vector|.
+// - filter_state : Current state (higher part) of the filter.
+// - filter_state_length : Length (in samples) of |filter_state|.
+// - filter_state_low : Current state (lower part) of the filter.
+// - filter_state_low_length : Length (in samples) of |filter_state_low|.
+// - out_vector_low_length : Maximum length (in samples) of
+// |out_vector_low|.
+//
+// Output:
+// - filter_state : Updated state (upper part) vector.
+// - filter_state_low : Updated state (lower part) vector.
+// - out_vector : Vector containing the upper part of the
+// filtered values.
+// - out_vector_low : Vector containing the lower part of the
+// filtered values.
+//
+// Return value : Number of samples in the |out_vector|.
+//
+
+//
+// WebRtcSpl_FilterMAFastQ12(...)
+//
+// Performs a MA filtering on a vector in Q12
+//
+// Input:
+// - in_vector : Input samples (state in positions
+// in_vector[-order] .. in_vector[-1])
+// - ma_coef : Filter coefficients (in Q12)
+// - ma_coef_length : Number of B coefficients (order+1)
+// - vector_length : Number of samples to be filtered
+//
+// Output:
+// - out_vector : Filtered samples
+//
+
+
+//
+// WebRtcSpl_DotProductWithScale(...)
+//
+// Calculates the dot product between two (WebRtc_Word16) vectors
+//
+// Input:
+// - vector1 : Vector 1
+// - vector2 : Vector 2
+// - vector_length : Number of samples used in the dot product
+// - scaling : The number of right bit shifts to apply on each term
+// during calculation to avoid overflow, i.e., the
+// output will be in Q(-|scaling|)
+//
+// Return value : The dot product in Q(-scaling)
+//
+
+//
+// WebRtcSpl_ComplexIFFT(...)
+//
+// Complex Inverse FFT
+//
+// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
+// is in bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With X as the input complex vector, y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+// M-1
+// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+// i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// Input:
+// - vector : In pointer to complex vector containing 2^|stages|
+// real elements interleaved with 2^|stages| imaginary
+// elements.
+// [ReImReImReIm....]
+// The elements are in Q(-scale) domain, see more on Return
+// Value below.
+//
+// - stages : Number of FFT stages. Must be at least 3 and at most 10,
+// since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// - mode : This parameter gives the user to choose how the FFT
+// should work.
+// mode==0: Low-complexity and Low-accuracy mode
+// mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+// - vector : Out pointer to the FFT vector (the same as input).
+//
+// Return Value : The scale value that tells the number of left bit shifts
+// that the elements in the |vector| should be shifted with
+// in order to get Q0 values, i.e. the physically correct
+// values. The scale parameter is always 0 or positive,
+// except if N>1024 (|stages|>10), which returns a scale
+// value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexFFT(...)
+//
+// Complex FFT
+//
+// Computes a complex 2^|stages|-point FFT on the input vector, which is in
+// bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With x as the input complex vector, Y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+// M-1
+// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+// i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// This routine prevents overflow by scaling by 2 before each FFT stage. This is
+// a fixed scaling, for proper normalization - there will be log2(n) passes, so
+// this results in an overall factor of 1/n, distributed to maximize arithmetic
+// accuracy.
+//
+// Input:
+// - vector : In pointer to complex vector containing 2^|stages| real
+// elements interleaved with 2^|stages| imaginary elements.
+// [ReImReImReIm....]
+// The output is in the Q0 domain.
+//
+// - stages : Number of FFT stages. Must be at least 3 and at most 10,
+// since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// - mode : This parameter gives the user to choose how the FFT
+// should work.
+// mode==0: Low-complexity and Low-accuracy mode
+// mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+// - vector : The output FFT vector is in the Q0 domain.
+//
+// Return value : The scale parameter is always 0, except if N>1024,
+// which returns a scale value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_AnalysisQMF(...)
+//
+// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
+// current version has F = 8000, therefore, a super-wideband audio signal is
+// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
+//
+// Input:
+// - in_data : Wide band speech signal, 320 samples (10 ms)
+//
+// Input & Output:
+// - filter_state1 : Filter state for first All-pass filter
+// - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+// - low_band : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
+// - high_band : Upper-band signal 8-16 kHz band (flipped in frequency
+// domain), 160 samples (10 ms)
+//
+
+//
+// WebRtcSpl_SynthesisQMF(...)
+//
+// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
+// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
+// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
+// kHz audio.
+//
+// Input:
+// - low_band : The signal with the 0-8 kHz band, 160 samples (10 ms)
+// - high_band : The signal with the 8-16 kHz band, 160 samples (10 ms)
+//
+// Input & Output:
+// - filter_state1 : Filter state for first All-pass filter
+// - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+// - out_data : Super-wideband speech signal, 0-16 kHz
+//
+
+// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+//
+// Input:
+// - value32 : The value of a 32-bit word.
+//
+// Output:
+// - out16 : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+//
+// Input:
+// - a : The value of the first 16-bit word.
+// - b : The value of the second 16-bit word.
+// - c : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
+
+// WebRtc_Word16 WebRtcSpl_get_version(...)
+//
+// This function gives the version string of the Signal Processing Library.
+//
+// Input:
+// - length_in_bytes : The size of Allocated space (in Bytes) where
+// the version number is written to (in string format).
+//
+// Output:
+// - version : Pointer to a buffer where the version number is written to.
+//
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl.h b/trunk/src/common_audio/signal_processing/include/spl_inl.h
new file mode 100644
index 0000000..23b3209
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/spl_inl.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_H_
+#define WEBRTC_SPL_SPL_INL_H_
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+#include "spl_inl_armv7.h"
+#else
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+ WebRtc_Word16 out16 = (WebRtc_Word16) value32;
+
+ if (value32 > 32767)
+ out16 = 32767;
+ else if (value32 < -32768)
+ out16 = -32768;
+
+ return out16;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sum;
+
+ // perform long addition
+ l_sum = l_var1 + l_var2;
+
+ // check for under or overflow
+ if (WEBRTC_SPL_IS_NEG(l_var1)) {
+ if (WEBRTC_SPL_IS_NEG(l_var2) && !WEBRTC_SPL_IS_NEG(l_sum)) {
+ l_sum = (WebRtc_Word32)0x80000000;
+ }
+ } else {
+ if (!WEBRTC_SPL_IS_NEG(l_var2) && WEBRTC_SPL_IS_NEG(l_sum)) {
+ l_sum = (WebRtc_Word32)0x7FFFFFFF;
+ }
+ }
+
+ return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+ WebRtc_Word16 var2) {
+ return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_diff;
+
+ // perform subtraction
+ l_diff = l_var1 - l_var2;
+
+ // check for underflow
+ if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
+ l_diff = (WebRtc_Word32)0x80000000;
+ // check for overflow
+ if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
+ l_diff = (WebRtc_Word32)0x7FFFFFFF;
+
+ return l_diff;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+ int bits;
+
+ if (0xFFFF0000 & n) {
+ bits = 16;
+ } else {
+ bits = 0;
+ }
+ if (0x0000FF00 & (n >> bits)) bits += 8;
+ if (0x000000F0 & (n >> bits)) bits += 4;
+ if (0x0000000C & (n >> bits)) bits += 2;
+ if (0x00000002 & (n >> bits)) bits += 1;
+ if (0x00000001 & (n >> bits)) bits += 1;
+
+ return bits;
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+ int zeros;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ if (!(0xFFFF8000 & a)) {
+ zeros = 16;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xFF800000 & (a << zeros))) zeros += 8;
+ if (!(0xF8000000 & (a << zeros))) zeros += 4;
+ if (!(0xE0000000 & (a << zeros))) zeros += 2;
+ if (!(0xC0000000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+ int zeros;
+
+ if (a == 0) return 0;
+
+ if (!(0xFFFF0000 & a)) {
+ zeros = 16;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xFF000000 & (a << zeros))) zeros += 8;
+ if (!(0xF0000000 & (a << zeros))) zeros += 4;
+ if (!(0xC0000000 & (a << zeros))) zeros += 2;
+ if (!(0x80000000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+ int zeros;
+
+ if (a <= 0) a ^= 0xFFFF;
+
+ if (!(0xFF80 & a)) {
+ zeros = 8;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xF800 & (a << zeros))) zeros += 4;
+ if (!(0xE000 & (a << zeros))) zeros += 2;
+ if (!(0xC000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+ int16_t b,
+ int32_t c) {
+ return (a * b + c);
+}
+
+#endif // WEBRTC_ARCH_ARM_V7A
+
+#endif // WEBRTC_SPL_SPL_INL_H_
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h b/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
new file mode 100644
index 0000000..5b19c2c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions for ARM processors in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
+#define WEBRTC_SPL_SPL_INL_ARMV7_H_
+
+// TODO(kma): Replace some assembly code with GCC intrinsics
+// (e.g. __builtin_clz).
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
+ WebRtc_Word32 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
+ WebRtc_Word16 b,
+ WebRtc_Word32 c) {
+ WebRtc_Word32 tmp;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
+ __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(WebRtc_Word32 a,
+ WebRtc_Word32 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+ return tmp;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+ int16_t b,
+ int32_t c) {
+ int32_t tmp = 0;
+ __asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
+ return tmp;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ WebRtc_Word32 s_sum;
+
+ __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
+
+ return (WebRtc_Word16) s_sum;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sum;
+
+ __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
+
+ return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+ WebRtc_Word16 var2) {
+ WebRtc_Word32 s_sub;
+
+ __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
+
+ return (WebRtc_Word16)s_sub;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sub;
+
+ __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
+
+ return l_sub;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+ WebRtc_Word32 tmp;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(n));
+
+ return (WebRtc_Word16)(32 - tmp);
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+ WebRtc_Word32 tmp;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp - 1;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+ int tmp;
+
+ if (a == 0) return 0;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+ WebRtc_Word32 tmp;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp - 17;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+ WebRtc_Word16 out16;
+
+ __asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
+
+ return out16;
+}
+#endif // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/trunk/src/common_audio/signal_processing/levinson_durbin.c b/trunk/src/common_audio/signal_processing/levinson_durbin.c
new file mode 100644
index 0000000..4e11cdb
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/levinson_durbin.c
@@ -0,0 +1,259 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_LevinsonDurbin().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define SPL_LEVINSON_MAXORDER 20
+
+WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32 *R, WebRtc_Word16 *A, WebRtc_Word16 *K,
+ WebRtc_Word16 order)
+{
+ WebRtc_Word16 i, j;
+ // Auto-correlation coefficients in high precision
+ WebRtc_Word16 R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
+ // LPC coefficients in high precision
+ WebRtc_Word16 A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
+ // LPC coefficients for next iteration
+ WebRtc_Word16 A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
+ // Reflection coefficient in high precision
+ WebRtc_Word16 K_hi, K_low;
+ // Prediction gain Alpha in high precision and with scale factor
+ WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp;
+ WebRtc_Word16 tmp_hi, tmp_low;
+ WebRtc_Word32 temp1W32, temp2W32, temp3W32;
+ WebRtc_Word16 norm;
+
+ // Normalize the autocorrelation R[0]...R[order+1]
+
+ norm = WebRtcSpl_NormW32(R[0]);
+
+ for (i = order; i >= 0; i--)
+ {
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
+ // Put R in hi and low format
+ R_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
+ }
+
+ // K = A[1] = -R[1] / R[0]
+
+ temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
+ + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
+ temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
+ temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
+ // Put back the sign on R[1]
+ if (temp2W32 > 0)
+ {
+ temp1W32 = -temp1W32;
+ }
+
+ // Put K in hi and low format
+ K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+ // Store first reflection coefficient
+ K[0] = K_hi;
+
+ temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); // A[1] in Q27
+
+ // Put A[1] in hi and low format
+ A_hi[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
+
+ // Alpha = R[0] * (1-K^2)
+
+ temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
+ << 1); // temp1W32 = k^2 in Q31
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+ temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
+
+ // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
+ tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ // Calculate Alpha in Q31
+ temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
+ + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
+ + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
+
+ // Normalize Alpha and put it in hi and low format
+
+ Alpha_exp = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
+ Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+ // Perform the iterative calculations in the Levinson-Durbin algorithm
+
+ for (i = 2; i <= order; i++)
+ {
+ /* ----
+ temp1W32 = R[i] + > R[j]*A[i-j]
+ /
+ ----
+ j=1..i-1
+ */
+
+ temp1W32 = 0;
+
+ for (j = 1; j < i; j++)
+ {
+ // temp1W32 is in Q31
+ temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
+ + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
+ + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
+ }
+
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
+ temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)
+ + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
+
+ // K = -temp1W32 / Alpha
+ temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
+ temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
+
+ // Put the sign of temp1W32 back again
+ if (temp1W32 > 0)
+ {
+ temp3W32 = -temp3W32;
+ }
+
+ // Use the Alpha shifts from earlier to de-normalize
+ norm = WebRtcSpl_NormW32(temp3W32);
+ if ((Alpha_exp <= norm) || (temp3W32 == 0))
+ {
+ temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
+ } else
+ {
+ if (temp3W32 > 0)
+ {
+ temp3W32 = (WebRtc_Word32)0x7fffffffL;
+ } else
+ {
+ temp3W32 = (WebRtc_Word32)0x80000000L;
+ }
+ }
+
+ // Put K on hi and low format
+ K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+ K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+ // Store Reflection coefficient in Q15
+ K[i - 1] = K_hi;
+
+ // Test for unstable filter.
+ // If unstable return 0 and let the user decide what to do in that case
+
+ if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32750)
+ {
+ return 0; // Unstable filter
+ }
+
+ /*
+ Compute updated LPC coefficient: Anew[i]
+ Anew[j]= A[j] + K*A[i-j] for j=1..i-1
+ Anew[i]= K
+ */
+
+ for (j = 1; j < i; j++)
+ {
+ // temp1W32 = A[j] in Q27
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16)
+ + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);
+
+ // temp1W32 += K*A[i-j] in Q27
+ temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
+ + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
+ + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
+
+ // Put Anew in hi and low format
+ A_upd_hi[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
+ }
+
+ // temp3W32 = K in Q27 (Convert from Q31 to Q27)
+ temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);
+
+ // Store Anew in hi and low format
+ A_upd_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+ A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
+
+ // Alpha = Alpha * (1-K^2)
+
+ temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
+ + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+ temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // 1 - K*K in Q31
+
+ // Convert 1- K^2 in hi and low format
+ tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ // Calculate Alpha = Alpha * (1-K^2) in Q31
+ temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
+ + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
+ + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
+
+ // Normalize Alpha and store it on hi and low format
+
+ norm = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
+
+ Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+ - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+ // Update the total normalization of Alpha
+ Alpha_exp = Alpha_exp + norm;
+
+ // Update A[]
+
+ for (j = 1; j <= i; j++)
+ {
+ A_hi[j] = A_upd_hi[j];
+ A_low[j] = A_upd_low[j];
+ }
+ }
+
+ /*
+ Set A[0] to 1.0 and store the A[i] i=1...order in Q12
+ (Convert from Q27 and use rounding)
+ */
+
+ A[0] = 4096;
+
+ for (i = 1; i <= order; i++)
+ {
+ // temp1W32 in Q27
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16)
+ + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
+ // Round and store upper word
+ A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32<<1)+(WebRtc_Word32)32768, 16);
+ }
+ return 1; // Stable filters
+}
diff --git a/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c b/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
new file mode 100644
index 0000000..2cb83c2
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_LpcToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50
+
+void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* a16, int use_order, WebRtc_Word16* k16)
+{
+ int m, k;
+ WebRtc_Word32 tmp32[SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER];
+ WebRtc_Word32 tmp_inv_denom32;
+ WebRtc_Word16 tmp_inv_denom16;
+
+ k16[use_order - 1] = WEBRTC_SPL_LSHIFT_W16(a16[use_order], 3); //Q12<<3 => Q15
+ for (m = use_order - 1; m > 0; m--)
+ {
+ // (1 - k^2) in Q30
+ tmp_inv_denom32 = ((WebRtc_Word32)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
+ // (1 - k^2) in Q15
+ tmp_inv_denom16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_inv_denom32, 15);
+
+ for (k = 1; k <= m; k++)
+ {
+ // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
+
+ // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
+ tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16)
+ - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
+
+ tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
+ }
+
+ for (k = 1; k < m; k++)
+ {
+ a16[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q13>>1 => Q12
+ }
+
+ tmp32[m] = WEBRTC_SPL_SAT(8191, tmp32[m], -8191);
+ k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
+ }
+ return;
+}
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations.c b/trunk/src/common_audio/signal_processing/min_max_operations.c
new file mode 100644
index 0000000..57eaff7
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/min_max_operations.c
@@ -0,0 +1,265 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the implementation of functions
+ * WebRtcSpl_MaxAbsValueW16()
+ * WebRtcSpl_MaxAbsIndexW16()
+ * WebRtcSpl_MaxAbsValueW32()
+ * WebRtcSpl_MaxValueW16()
+ * WebRtcSpl_MaxIndexW16()
+ * WebRtcSpl_MaxValueW32()
+ * WebRtcSpl_MaxIndexW32()
+ * WebRtcSpl_MinValueW16()
+ * WebRtcSpl_MinIndexW16()
+ * WebRtcSpl_MinValueW32()
+ * WebRtcSpl_MinIndexW32()
+ *
+ * The description header can be found in signal_processing_library.h.
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#if !(defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
+// Maximum absolute value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word32 tempMax = 0;
+ WebRtc_Word32 absVal;
+ WebRtc_Word16 totMax;
+ int i;
+ G_CONST WebRtc_Word16 *tmpvector = vector;
+
+ for (i = 0; i < length; i++)
+ {
+ absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
+ if (absVal > tempMax)
+ {
+ tempMax = absVal;
+ }
+ tmpvector++;
+ }
+ totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
+ return totMax;
+}
+
+#endif
+
+// Index of maximum absolute value in a word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 tempMax;
+ WebRtc_Word16 absTemp;
+ WebRtc_Word16 tempMaxIndex = 0;
+ WebRtc_Word16 i = 0;
+ G_CONST WebRtc_Word16 *tmpvector = vector;
+
+ tempMax = WEBRTC_SPL_ABS_W16(*tmpvector);
+ tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ absTemp = WEBRTC_SPL_ABS_W16(*tmpvector);
+ tmpvector++;
+ if (absTemp > tempMax)
+ {
+ tempMax = absTemp;
+ tempMaxIndex = i;
+ }
+ }
+ return tempMaxIndex;
+}
+
+// Maximum absolute value of word32 vector.
+WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+ WebRtc_UWord32 tempMax = 0;
+ WebRtc_UWord32 absVal;
+ WebRtc_Word32 retval;
+ int i;
+ G_CONST WebRtc_Word32 *tmpvector = vector;
+
+ for (i = 0; i < length; i++)
+ {
+ absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
+ if (absVal > tempMax)
+ {
+ tempMax = absVal;
+ }
+ tmpvector++;
+ }
+ retval = (WebRtc_Word32)(WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD32_MAX));
+ return retval;
+}
+
+// Maximum value of word16 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 tempMax;
+ WebRtc_Word16 i;
+ G_CONST WebRtc_Word16 *tmpvector = vector;
+
+ tempMax = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ > tempMax)
+ tempMax = vector[i];
+ }
+ return tempMax;
+}
+#else
+#pragma message(">> WebRtcSpl_MaxValueW16 is excluded from this build")
+#endif
+
+// Index of maximum value in a word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 tempMax;
+ WebRtc_Word16 tempMaxIndex = 0;
+ WebRtc_Word16 i = 0;
+ G_CONST WebRtc_Word16 *tmpvector = vector;
+
+ tempMax = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ > tempMax)
+ {
+ tempMax = vector[i];
+ tempMaxIndex = i;
+ }
+ }
+ return tempMaxIndex;
+}
+
+// Maximum value of word32 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word32 tempMax;
+ WebRtc_Word16 i;
+ G_CONST WebRtc_Word32 *tmpvector = vector;
+
+ tempMax = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ > tempMax)
+ tempMax = vector[i];
+ }
+ return tempMax;
+}
+#else
+#pragma message(">> WebRtcSpl_MaxValueW32 is excluded from this build")
+#endif
+
+// Index of maximum value in a word32 vector.
+WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word32 tempMax;
+ WebRtc_Word16 tempMaxIndex = 0;
+ WebRtc_Word16 i = 0;
+ G_CONST WebRtc_Word32 *tmpvector = vector;
+
+ tempMax = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ > tempMax)
+ {
+ tempMax = vector[i];
+ tempMaxIndex = i;
+ }
+ }
+ return tempMaxIndex;
+}
+
+// Minimum value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 tempMin;
+ WebRtc_Word16 i;
+ G_CONST WebRtc_Word16 *tmpvector = vector;
+
+ // Find the minimum value
+ tempMin = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ < tempMin)
+ tempMin = (vector[i]);
+ }
+ return tempMin;
+}
+
+// Index of minimum value in a word16 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word16 tempMin;
+ WebRtc_Word16 tempMinIndex = 0;
+ WebRtc_Word16 i = 0;
+ G_CONST WebRtc_Word16* tmpvector = vector;
+
+ // Find index of smallest value
+ tempMin = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ < tempMin)
+ {
+ tempMin = vector[i];
+ tempMinIndex = i;
+ }
+ }
+ return tempMinIndex;
+}
+#else
+#pragma message(">> WebRtcSpl_MinIndexW16 is excluded from this build")
+#endif
+
+// Minimum value of word32 vector.
+WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+ WebRtc_Word32 tempMin;
+ WebRtc_Word16 i;
+ G_CONST WebRtc_Word32 *tmpvector = vector;
+
+ // Find the minimum value
+ tempMin = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ < tempMin)
+ tempMin = (vector[i]);
+ }
+ return tempMin;
+}
+
+// Index of minimum value in a word32 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+ WebRtc_Word32 tempMin;
+ WebRtc_Word16 tempMinIndex = 0;
+ WebRtc_Word16 i = 0;
+ G_CONST WebRtc_Word32 *tmpvector = vector;
+
+ // Find index of smallest value
+ tempMin = *tmpvector++;
+ for (i = 1; i < length; i++)
+ {
+ if (*tmpvector++ < tempMin)
+ {
+ tempMin = vector[i];
+ tempMinIndex = i;
+ }
+ }
+ return tempMinIndex;
+}
+#else
+#pragma message(">> WebRtcSpl_MinIndexW32 is excluded from this build")
+#endif
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations_neon.c b/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
new file mode 100644
index 0000000..158bcc1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
+#include <arm_neon.h>
+
+#include "signal_processing_library.h"
+
+// Maximum absolute value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+ WebRtc_Word16 length) {
+ WebRtc_Word32 temp_max = 0;
+ WebRtc_Word32 abs_val;
+ WebRtc_Word16 tot_max;
+ int i;
+
+ __asm__("vmov.i16 d25, #0" : : : "d25");
+
+ for (i = 0; i < length - 7; i += 8) {
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&vector[i]) : "q13");
+ __asm__("vabs.s16 q13, q13" : : : "q13");
+ __asm__("vpmax.s16 d26, d27" : : : "q13");
+ __asm__("vpmax.s16 d25, d26" : : : "d25", "d26");
+ }
+ __asm__("vpmax.s16 d25, d25" : : : "d25");
+ __asm__("vpmax.s16 d25, d25" : : : "d25");
+ __asm__("vmov.s16 %0, d25[0]" : "=r"(temp_max): : "d25");
+
+ for (; i < length; i++) {
+ abs_val = WEBRTC_SPL_ABS_W32((vector[i]));
+ if (abs_val > temp_max) {
+ temp_max = abs_val;
+ }
+ }
+ tot_max = (WebRtc_Word16)WEBRTC_SPL_MIN(temp_max, WEBRTC_SPL_WORD16_MAX);
+ return tot_max;
+}
+
+#endif
diff --git a/trunk/src/common_audio/signal_processing/randomization_functions.c b/trunk/src/common_audio/signal_processing/randomization_functions.c
new file mode 100644
index 0000000..04271ad
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/randomization_functions.c
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the randomization functions
+ * WebRtcSpl_IncreaseSeed()
+ * WebRtcSpl_RandU()
+ * WebRtcSpl_RandN()
+ * WebRtcSpl_RandUArray()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+static const WebRtc_Word16 kRandNTable[] = {
+ 9178, -7260, 40, 10189, 4894, -3531, -13779, 14764,
+ -4008, -8884, -8990, 1008, 7368, 5184, 3251, -5817,
+ -9786, 5963, 1770, 8066, -7135, 10772, -2298, 1361,
+ 6484, 2241, -8633, 792, 199, -3344, 6553, -10079,
+ -15040, 95, 11608, -12469, 14161, -4176, 2476, 6403,
+ 13685, -16005, 6646, 2239, 10916, -3004, -602, -3141,
+ 2142, 14144, -5829, 5305, 8209, 4713, 2697, -5112,
+ 16092, -1210, -2891, -6631, -5360, -11878, -6781, -2739,
+ -6392, 536, 10923, 10872, 5059, -4748, -7770, 5477,
+ 38, -1025, -2892, 1638, 6304, 14375, -11028, 1553,
+ -1565, 10762, -393, 4040, 5257, 12310, 6554, -4799,
+ 4899, -6354, 1603, -1048, -2220, 8247, -186, -8944,
+ -12004, 2332, 4801, -4933, 6371, 131, 8614, -5927,
+ -8287, -22760, 4033, -15162, 3385, 3246, 3153, -5250,
+ 3766, 784, 6494, -62, 3531, -1582, 15572, 662,
+ -3952, -330, -3196, 669, 7236, -2678, -6569, 23319,
+ -8645, -741, 14830, -15976, 4903, 315, -11342, 10311,
+ 1858, -7777, 2145, 5436, 5677, -113, -10033, 826,
+ -1353, 17210, 7768, 986, -1471, 8291, -4982, 8207,
+ -14911, -6255, -2449, -11881, -7059, -11703, -4338, 8025,
+ 7538, -2823, -12490, 9470, -1613, -2529, -10092, -7807,
+ 9480, 6970, -12844, 5123, 3532, 4816, 4803, -8455,
+ -5045, 14032, -4378, -1643, 5756, -11041, -2732, -16618,
+ -6430, -18375, -3320, 6098, 5131, -4269, -8840, 2482,
+ -7048, 1547, -21890, -6505, -7414, -424, -11722, 7955,
+ 1653, -17299, 1823, 473, -9232, 3337, 1111, 873,
+ 4018, -8982, 9889, 3531, -11763, -3799, 7373, -4539,
+ 3231, 7054, -8537, 7616, 6244, 16635, 447, -2915,
+ 13967, 705, -2669, -1520, -1771, -16188, 5956, 5117,
+ 6371, -9936, -1448, 2480, 5128, 7550, -8130, 5236,
+ 8213, -6443, 7707, -1950, -13811, 7218, 7031, -3883,
+ 67, 5731, -2874, 13480, -3743, 9298, -3280, 3552,
+ -4425, -18, -3785, -9988, -5357, 5477, -11794, 2117,
+ 1416, -9935, 3376, 802, -5079, -8243, 12652, 66,
+ 3653, -2368, 6781, -21895, -7227, 2487, 7839, -385,
+ 6646, -7016, -4658, 5531, -1705, 834, 129, 3694,
+ -1343, 2238, -22640, -6417, -11139, 11301, -2945, -3494,
+ -5626, 185, -3615, -2041, -7972, -3106, -60, -23497,
+ -1566, 17064, 3519, 2518, 304, -6805, -10269, 2105,
+ 1936, -426, -736, -8122, -1467, 4238, -6939, -13309,
+ 360, 7402, -7970, 12576, 3287, 12194, -6289, -16006,
+ 9171, 4042, -9193, 9123, -2512, 6388, -4734, -8739,
+ 1028, -5406, -1696, 5889, -666, -4736, 4971, 3565,
+ 9362, -6292, 3876, -3652, -19666, 7523, -4061, 391,
+ -11773, 7502, -3763, 4929, -9478, 13278, 2805, 4496,
+ 7814, 16419, 12455, -14773, 2127, -2746, 3763, 4847,
+ 3698, 6978, 4751, -6957, -3581, -45, 6252, 1513,
+ -4797, -7925, 11270, 16188, -2359, -5269, 9376, -10777,
+ 7262, 20031, -6515, -2208, -5353, 8085, -1341, -1303,
+ 7333, 5576, 3625, 5763, -7931, 9833, -3371, -10305,
+ 6534, -13539, -9971, 997, 8464, -4064, -1495, 1857,
+ 13624, 5458, 9490, -11086, -4524, 12022, -550, -198,
+ 408, -8455, -7068, 10289, 9712, -3366, 9028, -7621,
+ -5243, 2362, 6909, 4672, -4933, -1799, 4709, -4563,
+ -62, -566, 1624, -7010, 14730, -17791, -3697, -2344,
+ -1741, 7099, -9509, -6855, -1989, 3495, -2289, 2031,
+ 12784, 891, 14189, -3963, -5683, 421, -12575, 1724,
+ -12682, -5970, -8169, 3143, -1824, -5488, -5130, 8536,
+ 12799, 794, 5738, 3459, -11689, -258, -3738, -3775,
+ -8742, 2333, 8312, -9383, 10331, 13119, 8398, 10644,
+ -19433, -6446, -16277, -11793, 16284, 9345, 15222, 15834,
+ 2009, -7349, 130, -14547, 338, -5998, 3337, 21492,
+ 2406, 7703, -951, 11196, -564, 3406, 2217, 4806,
+ 2374, -5797, 11839, 8940, -11874, 18213, 2855, 10492
+};
+
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
+{
+ seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
+ return seed[0];
+}
+
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
+{
+ return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
+}
+
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
+{
+ return kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
+}
+
+// Creates an array of uniformly distributed variables
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+ WebRtc_Word16 vector_length,
+ WebRtc_UWord32* seed)
+{
+ int i;
+ for (i = 0; i < vector_length; i++)
+ {
+ vector[i] = WebRtcSpl_RandU(seed);
+ }
+ return vector_length;
+}
diff --git a/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c b/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
new file mode 100644
index 0000000..d07804d
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_ReflCoefToLpc().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16 *k, int use_order, WebRtc_Word16 *a)
+{
+ WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+ WebRtc_Word16 *aptr, *aptr2, *anyptr;
+ G_CONST WebRtc_Word16 *kptr;
+ int m, i;
+
+ kptr = k;
+ *a = 4096; // i.e., (Word16_MAX >> 3)+1.
+ *any = *a;
+ a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
+
+ for (m = 1; m < use_order; m++)
+ {
+ kptr++;
+ aptr = a;
+ aptr++;
+ aptr2 = &a[m];
+ anyptr = any;
+ anyptr++;
+
+ any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
+ for (i = 0; i < m; i++)
+ {
+ *anyptr = (*aptr)
+ + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
+ anyptr++;
+ aptr++;
+ aptr2--;
+ }
+
+ aptr = a;
+ anyptr = any;
+ for (i = 0; i < (m + 2); i++)
+ {
+ *aptr = *anyptr;
+ aptr++;
+ anyptr++;
+ }
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample.c b/trunk/src/common_audio/signal_processing/resample.c
new file mode 100644
index 0000000..19d1778
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample.c
@@ -0,0 +1,505 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling functions for 22 kHz.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+// Declaration of internally used functions
+static void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In, WebRtc_Word16 *Out,
+ const WebRtc_Word32 K);
+
+void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+ const WebRtc_Word32 K);
+
+// interpolation coefficients
+static const WebRtc_Word16 kCoefficients32To22[5][9] = {
+ {127, -712, 2359, -6333, 23456, 16775, -3695, 945, -154},
+ {-39, 230, -830, 2785, 32366, -2324, 760, -218, 38},
+ {117, -663, 2222, -6133, 26634, 13070, -3174, 831, -137},
+ {-77, 457, -1677, 5958, 31175, -4136, 1405, -408, 71},
+ { 98, -560, 1900, -5406, 29240, 9423, -2480, 663, -110}
+};
+
+//////////////////////
+// 22 kHz -> 16 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_22_16 5
+
+// 22 -> 16 resampler
+void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State22khzTo16khz* state, WebRtc_Word32* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_22_16; k++)
+ {
+ ///// 22 --> 44 /////
+ // WebRtc_Word16 in[220/SUB_BLOCKS_22_16]
+ // WebRtc_Word32 out[440/SUB_BLOCKS_22_16]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
+
+ ///// 44 --> 32 /////
+ // WebRtc_Word32 in[440/SUB_BLOCKS_22_16]
+ // WebRtc_Word32 out[320/SUB_BLOCKS_22_16]
+ /////
+ // copy state to and from input array
+ tmpmem[8] = state->S_44_32[0];
+ tmpmem[9] = state->S_44_32[1];
+ tmpmem[10] = state->S_44_32[2];
+ tmpmem[11] = state->S_44_32[3];
+ tmpmem[12] = state->S_44_32[4];
+ tmpmem[13] = state->S_44_32[5];
+ tmpmem[14] = state->S_44_32[6];
+ tmpmem[15] = state->S_44_32[7];
+ state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
+ state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
+ state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
+ state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
+ state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
+ state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
+ state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
+ state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
+
+ WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
+
+ ///// 32 --> 16 /////
+ // WebRtc_Word32 in[320/SUB_BLOCKS_22_16]
+ // WebRtc_Word32 out[160/SUB_BLOCKS_22_16]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
+
+ // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
+ in += 220 / SUB_BLOCKS_22_16;
+ out += 160 / SUB_BLOCKS_22_16;
+ }
+}
+
+// initialize state of 22 -> 16 resampler
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_22_44[k] = 0;
+ state->S_44_32[k] = 0;
+ state->S_32_16[k] = 0;
+ }
+}
+
+//////////////////////
+// 16 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_16_22 4
+
+// 16 -> 22 resampler
+void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State16khzTo22khz* state, WebRtc_Word32* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_16_22; k++)
+ {
+ ///// 16 --> 32 /////
+ // WebRtc_Word16 in[160/SUB_BLOCKS_16_22]
+ // WebRtc_Word32 out[320/SUB_BLOCKS_16_22]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
+
+ ///// 32 --> 22 /////
+ // WebRtc_Word32 in[320/SUB_BLOCKS_16_22]
+ // WebRtc_Word32 out[220/SUB_BLOCKS_16_22]
+ /////
+ // copy state to and from input array
+ tmpmem[0] = state->S_32_22[0];
+ tmpmem[1] = state->S_32_22[1];
+ tmpmem[2] = state->S_32_22[2];
+ tmpmem[3] = state->S_32_22[3];
+ tmpmem[4] = state->S_32_22[4];
+ tmpmem[5] = state->S_32_22[5];
+ tmpmem[6] = state->S_32_22[6];
+ tmpmem[7] = state->S_32_22[7];
+ state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
+ state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
+ state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
+ state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
+ state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
+ state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
+ state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
+ state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
+
+ WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
+
+ // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
+ in += 160 / SUB_BLOCKS_16_22;
+ out += 220 / SUB_BLOCKS_16_22;
+ }
+}
+
+// initialize state of 16 -> 22 resampler
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_16_32[k] = 0;
+ state->S_32_22[k] = 0;
+ }
+}
+
+//////////////////////
+// 22 kHz -> 8 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_22_8 2
+
+// 22 -> 8 resampler
+void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State22khzTo8khz* state, WebRtc_Word32* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_22_8; k++)
+ {
+ ///// 22 --> 22 lowpass /////
+ // WebRtc_Word16 in[220/SUB_BLOCKS_22_8]
+ // WebRtc_Word32 out[220/SUB_BLOCKS_22_8]
+ /////
+ WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
+
+ ///// 22 --> 16 /////
+ // WebRtc_Word32 in[220/SUB_BLOCKS_22_8]
+ // WebRtc_Word32 out[160/SUB_BLOCKS_22_8]
+ /////
+ // copy state to and from input array
+ tmpmem[8] = state->S_22_16[0];
+ tmpmem[9] = state->S_22_16[1];
+ tmpmem[10] = state->S_22_16[2];
+ tmpmem[11] = state->S_22_16[3];
+ tmpmem[12] = state->S_22_16[4];
+ tmpmem[13] = state->S_22_16[5];
+ tmpmem[14] = state->S_22_16[6];
+ tmpmem[15] = state->S_22_16[7];
+ state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
+ state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
+ state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
+ state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
+ state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
+ state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
+ state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
+ state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
+
+ WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
+
+ ///// 16 --> 8 /////
+ // WebRtc_Word32 in[160/SUB_BLOCKS_22_8]
+ // WebRtc_Word32 out[80/SUB_BLOCKS_22_8]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
+
+ // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
+ in += 220 / SUB_BLOCKS_22_8;
+ out += 80 / SUB_BLOCKS_22_8;
+ }
+}
+
+// initialize state of 22 -> 8 resampler
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_22_22[k] = 0;
+ state->S_22_22[k + 8] = 0;
+ state->S_22_16[k] = 0;
+ state->S_16_8[k] = 0;
+ }
+}
+
+//////////////////////
+// 8 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_8_22 2
+
+// 8 -> 22 resampler
+void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State8khzTo22khz* state, WebRtc_Word32* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_8_22; k++)
+ {
+ ///// 8 --> 16 /////
+ // WebRtc_Word16 in[80/SUB_BLOCKS_8_22]
+ // WebRtc_Word32 out[160/SUB_BLOCKS_8_22]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
+
+ ///// 16 --> 11 /////
+ // WebRtc_Word32 in[160/SUB_BLOCKS_8_22]
+ // WebRtc_Word32 out[110/SUB_BLOCKS_8_22]
+ /////
+ // copy state to and from input array
+ tmpmem[10] = state->S_16_11[0];
+ tmpmem[11] = state->S_16_11[1];
+ tmpmem[12] = state->S_16_11[2];
+ tmpmem[13] = state->S_16_11[3];
+ tmpmem[14] = state->S_16_11[4];
+ tmpmem[15] = state->S_16_11[5];
+ tmpmem[16] = state->S_16_11[6];
+ tmpmem[17] = state->S_16_11[7];
+ state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
+ state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
+ state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
+ state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
+ state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
+ state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
+ state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
+ state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
+
+ WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
+
+ ///// 11 --> 22 /////
+ // WebRtc_Word32 in[110/SUB_BLOCKS_8_22]
+ // WebRtc_Word16 out[220/SUB_BLOCKS_8_22]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
+
+ // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
+ in += 80 / SUB_BLOCKS_8_22;
+ out += 220 / SUB_BLOCKS_8_22;
+ }
+}
+
+// initialize state of 8 -> 22 resampler
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_8_16[k] = 0;
+ state->S_16_11[k] = 0;
+ state->S_11_22[k] = 0;
+ }
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToInt(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
+ const WebRtc_Word16* coef_ptr, WebRtc_Word32* out1,
+ WebRtc_Word32* out2)
+{
+ WebRtc_Word32 tmp1 = 16384;
+ WebRtc_Word32 tmp2 = 16384;
+ WebRtc_Word16 coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ *out1 = tmp1 + coef * in1[8];
+ *out2 = tmp2 + coef * in2[-8];
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToShort(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
+ const WebRtc_Word16* coef_ptr, WebRtc_Word16* out1,
+ WebRtc_Word16* out2)
+{
+ WebRtc_Word32 tmp1 = 16384;
+ WebRtc_Word32 tmp2 = 16384;
+ WebRtc_Word16 coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ tmp1 += coef * in1[8];
+ tmp2 += coef * in2[-8];
+
+ // scale down, round and saturate
+ tmp1 >>= 15;
+ if (tmp1 > (WebRtc_Word32)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ tmp2 >>= 15;
+ if (tmp2 > (WebRtc_Word32)0x00007FFF)
+ tmp2 = 0x00007FFF;
+ if (tmp2 < (WebRtc_Word32)0xFFFF8000)
+ tmp2 = 0xFFFF8000;
+ *out1 = (WebRtc_Word16)tmp1;
+ *out2 = (WebRtc_Word16)tmp2;
+}
+
+// Resampling ratio: 11/16
+// input: WebRtc_Word32 (normalized, not saturated) :: size 16 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 11 * K
+// K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32* In,
+ WebRtc_Word32* Out,
+ const WebRtc_Word32 K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (16 input samples -> 11 output samples);
+ // process in sub blocks of size 16 samples.
+ WebRtc_Word32 m;
+
+ for (m = 0; m < K; m++)
+ {
+ // first output sample
+ Out[0] = ((WebRtc_Word32)In[3] << 15) + (1 << 14);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+ // update pointers
+ In += 16;
+ Out += 11;
+ }
+}
+
+// Resampling ratio: 11/16
+// input: WebRtc_Word32 (normalized, not saturated) :: size 16 * K
+// output: WebRtc_Word16 (saturated) :: size 11 * K
+// K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In,
+ WebRtc_Word16 *Out,
+ const WebRtc_Word32 K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (16 input samples -> 11 output samples);
+ // process in sub blocks of size 16 samples.
+ WebRtc_Word32 tmp;
+ WebRtc_Word32 m;
+
+ for (m = 0; m < K; m++)
+ {
+ // first output sample
+ tmp = In[3];
+ if (tmp > (WebRtc_Word32)0x00007FFF)
+ tmp = 0x00007FFF;
+ if (tmp < (WebRtc_Word32)0xFFFF8000)
+ tmp = 0xFFFF8000;
+ Out[0] = (WebRtc_Word16)tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+ // update pointers
+ In += 16;
+ Out += 11;
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_48khz.c b/trunk/src/common_audio/signal_processing/resample_48khz.c
new file mode 100644
index 0000000..31cbe6b
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_48khz.c
@@ -0,0 +1,186 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains resampling functions between 48 kHz and nb/wb.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+////////////////////////////
+///// 48 kHz -> 16 kHz /////
+////////////////////////////
+
+// 48 -> 16 resampler
+void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State48khzTo16khz* state, WebRtc_Word32* tmpmem)
+{
+ ///// 48 --> 48(LP) /////
+ // WebRtc_Word16 in[480]
+ // WebRtc_Word32 out[480]
+ /////
+ WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
+
+ ///// 48 --> 32 /////
+ // WebRtc_Word32 in[480]
+ // WebRtc_Word32 out[320]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(WebRtc_Word32));
+ memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(WebRtc_Word32));
+ WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
+
+ ///// 32 --> 16 /////
+ // WebRtc_Word32 in[320]
+ // WebRtc_Word16 out[160]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
+}
+
+// initialize state of 48 -> 16 resampler
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
+{
+ memset(state->S_48_48, 0, 16 * sizeof(WebRtc_Word32));
+ memset(state->S_48_32, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_32_16, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+///// 16 kHz -> 48 kHz /////
+////////////////////////////
+
+// 16 -> 48 resampler
+void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State16khzTo48khz* state, WebRtc_Word32* tmpmem)
+{
+ ///// 16 --> 32 /////
+ // WebRtc_Word16 in[160]
+ // WebRtc_Word32 out[320]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
+
+ ///// 32 --> 24 /////
+ // WebRtc_Word32 in[320]
+ // WebRtc_Word32 out[240]
+ // copy state to and from input array
+ /////
+ memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(WebRtc_Word32));
+ memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(WebRtc_Word32));
+ WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
+
+ ///// 24 --> 48 /////
+ // WebRtc_Word32 in[240]
+ // WebRtc_Word16 out[480]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 16 -> 48 resampler
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
+{
+ memset(state->S_16_32, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_32_24, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+///// 48 kHz -> 8 kHz /////
+////////////////////////////
+
+// 48 -> 8 resampler
+void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State48khzTo8khz* state, WebRtc_Word32* tmpmem)
+{
+ ///// 48 --> 24 /////
+ // WebRtc_Word16 in[480]
+ // WebRtc_Word32 out[240]
+ /////
+ WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
+
+ ///// 24 --> 24(LP) /////
+ // WebRtc_Word32 in[240]
+ // WebRtc_Word32 out[240]
+ /////
+ WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
+
+ ///// 24 --> 16 /////
+ // WebRtc_Word32 in[240]
+ // WebRtc_Word32 out[160]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(WebRtc_Word32));
+ memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(WebRtc_Word32));
+ WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
+
+ ///// 16 --> 8 /////
+ // WebRtc_Word32 in[160]
+ // WebRtc_Word16 out[80]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
+}
+
+// initialize state of 48 -> 8 resampler
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
+{
+ memset(state->S_48_24, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_24_24, 0, 16 * sizeof(WebRtc_Word32));
+ memset(state->S_24_16, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_16_8, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+///// 8 kHz -> 48 kHz /////
+////////////////////////////
+
+// 8 -> 48 resampler
+void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+ WebRtcSpl_State8khzTo48khz* state, WebRtc_Word32* tmpmem)
+{
+ ///// 8 --> 16 /////
+ // WebRtc_Word16 in[80]
+ // WebRtc_Word32 out[160]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
+
+ ///// 16 --> 12 /////
+ // WebRtc_Word32 in[160]
+ // WebRtc_Word32 out[120]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(WebRtc_Word32));
+ memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(WebRtc_Word32));
+ WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
+
+ ///// 12 --> 24 /////
+ // WebRtc_Word32 in[120]
+ // WebRtc_Word16 out[240]
+ /////
+ WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
+
+ ///// 24 --> 48 /////
+ // WebRtc_Word32 in[240]
+ // WebRtc_Word16 out[480]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 8 -> 48 resampler
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
+{
+ memset(state->S_8_16, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_16_12, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_12_24, 0, 8 * sizeof(WebRtc_Word32));
+ memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2.c b/trunk/src/common_audio/signal_processing/resample_by_2.c
new file mode 100644
index 0000000..e239db7
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2.c
@@ -0,0 +1,181 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+
+// allpass filter coefficients.
+static const WebRtc_UWord32 kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
+static const WebRtc_UWord32 kResampleAllpass2[3] =
+ {12199, 37471 << 15, 60255 << 15};
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: state + ((diff * tbl_value) >> 16)
+
+static __inline WebRtc_Word32 MUL_ACCUM_1(WebRtc_Word32 tbl_value,
+ WebRtc_Word32 diff,
+ WebRtc_Word32 state) {
+ WebRtc_Word32 result;
+ __asm__("smlawb %r0, %r1, %r2, %r3": "=r"(result): "r"(diff),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
+//
+// The reason to introduce this function is that, in case we can't use smlawb
+// instruction (in MUL_ACCUM_1) due to input value range, we can still use
+// smmla to save some cycles.
+
+static __inline WebRtc_Word32 MUL_ACCUM_2(WebRtc_Word32 tbl_value,
+ WebRtc_Word32 diff,
+ WebRtc_Word32 state) {
+ WebRtc_Word32 result;
+ __asm__("smmla %r0, %r1, %r2, %r3": "=r"(result): "r"(diff << 1),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+#else
+
+// allpass filter coefficients.
+static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
+static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+#endif // WEBRTC_ARCH_ARM_V7A
+
+
+// decimator
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState) {
+ WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+ WebRtc_Word16 i;
+
+ register WebRtc_Word32 state0 = filtState[0];
+ register WebRtc_Word32 state1 = filtState[1];
+ register WebRtc_Word32 state2 = filtState[2];
+ register WebRtc_Word32 state3 = filtState[3];
+ register WebRtc_Word32 state4 = filtState[4];
+ register WebRtc_Word32 state5 = filtState[5];
+ register WebRtc_Word32 state6 = filtState[6];
+ register WebRtc_Word32 state7 = filtState[7];
+
+ for (i = (len >> 1); i > 0; i--) {
+ // lower allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
+
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState) {
+ WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+ WebRtc_Word16 i;
+
+ register WebRtc_Word32 state0 = filtState[0];
+ register WebRtc_Word32 state1 = filtState[1];
+ register WebRtc_Word32 state2 = filtState[2];
+ register WebRtc_Word32 state3 = filtState[3];
+ register WebRtc_Word32 state4 = filtState[4];
+ register WebRtc_Word32 state5 = filtState[5];
+ register WebRtc_Word32 state6 = filtState[6];
+ register WebRtc_Word32 state7 = filtState[7];
+
+ for (i = len; i > 0; i--) {
+ // lower allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
+ state2 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state3 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+
+ // upper allpass filter
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
+ state6 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state7 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.c b/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
new file mode 100644
index 0000000..cbd2395
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
@@ -0,0 +1,679 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file contains some internal resampling functions.
+ *
+ */
+
+#include "resample_by_2_internal.h"
+
+// allpass filter coefficients.
+static const WebRtc_Word16 kResampleAllpass[2][3] = {
+ {821, 6110, 12382},
+ {3050, 9368, 15063}
+};
+
+//
+// decimator
+// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
+// output: WebRtc_Word16 (saturated) (of length len/2)
+// state: filter state array; length = 8
+
+void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+ WebRtc_Word32 *state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ len >>= 1;
+
+ // lower allpass filter (operates on even input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // divide by two and store temporarily
+ in[i << 1] = (state[3] >> 1);
+ }
+
+ in++;
+
+ // upper allpass filter (operates on odd input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // divide by two and store temporarily
+ in[i << 1] = (state[7] >> 1);
+ }
+
+ in--;
+
+ // combine allpass outputs
+ for (i = 0; i < len; i += 2)
+ {
+ // divide by two, add both allpass outputs and round
+ tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
+ tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
+ if (tmp0 > (WebRtc_Word32)0x00007FFF)
+ tmp0 = 0x00007FFF;
+ if (tmp0 < (WebRtc_Word32)0xFFFF8000)
+ tmp0 = 0xFFFF8000;
+ out[i] = (WebRtc_Word16)tmp0;
+ if (tmp1 > (WebRtc_Word32)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i + 1] = (WebRtc_Word16)tmp1;
+ }
+}
+
+//
+// decimator
+// input: WebRtc_Word16
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len/2)
+// state: filter state array; length = 8
+
+void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in,
+ WebRtc_Word32 len,
+ WebRtc_Word32 *out,
+ WebRtc_Word32 *state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ len >>= 1;
+
+ // lower allpass filter (operates on even input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // divide by two and store temporarily
+ out[i] = (state[3] >> 1);
+ }
+
+ in++;
+
+ // upper allpass filter (operates on odd input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // divide by two and store temporarily
+ out[i] += (state[7] >> 1);
+ }
+
+ in--;
+}
+
+//
+// interpolator
+// input: WebRtc_Word16
+// output: WebRtc_Word32 (normalized, not saturated) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+ WebRtc_Word32 *state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[7] >> 15;
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 15;
+ }
+}
+
+//
+// interpolator
+// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+ WebRtc_Word32 *state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[7];
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3];
+ }
+}
+
+//
+// interpolator
+// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word16 (saturated) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+ WebRtc_Word32 *state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, saturate and store
+ tmp1 = state[7] >> 15;
+ if (tmp1 > (WebRtc_Word32)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i << 1] = (WebRtc_Word16)tmp1;
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, saturate and store
+ tmp1 = state[3] >> 15;
+ if (tmp1 > (WebRtc_Word32)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i << 1] = (WebRtc_Word16)tmp1;
+ }
+}
+
+// lowpass filter
+// input: WebRtc_Word16
+// output: WebRtc_Word32 (normalized, not saturated)
+// state: filter state array; length = 8
+void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len, WebRtc_Word32* out,
+ WebRtc_Word32* state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ len >>= 1;
+
+ // lower allpass filter: odd input -> even output samples
+ in++;
+ // initial state of polyphase delay element
+ tmp0 = state[12];
+ for (i = 0; i < len; i++)
+ {
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 1;
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ }
+ in--;
+
+ // upper allpass filter: even input -> even output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+ }
+
+ // switch to odd output samples
+ out++;
+
+ // lower allpass filter: even input -> odd output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[9];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[8] + diff * kResampleAllpass[1][0];
+ state[8] = tmp0;
+ diff = tmp1 - state[10];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[9] + diff * kResampleAllpass[1][1];
+ state[9] = tmp1;
+ diff = tmp0 - state[11];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[11] = state[10] + diff * kResampleAllpass[1][2];
+ state[10] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[11] >> 1;
+ }
+
+ // upper allpass filter: odd input -> odd output samples
+ in++;
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[13];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[12] + diff * kResampleAllpass[0][0];
+ state[12] = tmp0;
+ diff = tmp1 - state[14];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[13] + diff * kResampleAllpass[0][1];
+ state[13] = tmp1;
+ diff = tmp0 - state[15];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[15] = state[14] + diff * kResampleAllpass[0][2];
+ state[14] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+ }
+}
+
+// lowpass filter
+// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word32 (normalized, not saturated)
+// state: filter state array; length = 8
+void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
+ WebRtc_Word32* state)
+{
+ WebRtc_Word32 tmp0, tmp1, diff;
+ WebRtc_Word32 i;
+
+ len >>= 1;
+
+ // lower allpass filter: odd input -> even output samples
+ in++;
+ // initial state of polyphase delay element
+ tmp0 = state[12];
+ for (i = 0; i < len; i++)
+ {
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 1;
+ tmp0 = in[i << 1];
+ }
+ in--;
+
+ // upper allpass filter: even input -> even output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+ }
+
+ // switch to odd output samples
+ out++;
+
+ // lower allpass filter: even input -> odd output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[9];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[8] + diff * kResampleAllpass[1][0];
+ state[8] = tmp0;
+ diff = tmp1 - state[10];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[9] + diff * kResampleAllpass[1][1];
+ state[9] = tmp1;
+ diff = tmp0 - state[11];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[11] = state[10] + diff * kResampleAllpass[1][2];
+ state[10] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[11] >> 1;
+ }
+
+ // upper allpass filter: odd input -> odd output samples
+ in++;
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[13];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[12] + diff * kResampleAllpass[0][0];
+ state[12] = tmp0;
+ diff = tmp1 - state[14];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[13] + diff * kResampleAllpass[0][1];
+ state[13] = tmp1;
+ diff = tmp0 - state[15];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[15] = state[14] + diff * kResampleAllpass[0][2];
+ state[14] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.h b/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
new file mode 100644
index 0000000..b6ac9f0
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file contains some internal resampling functions.
+ *
+ */
+
+#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+
+#include "typedefs.h"
+
+/*******************************************************************
+ * resample_by_2_fast.c
+ * Functions for internal use in the other resample functions
+ ******************************************************************/
+void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+ WebRtc_Word32 *state);
+
+void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
+ WebRtc_Word32 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
+ WebRtc_Word32 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+ WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len,
+ WebRtc_Word16 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len,
+ WebRtc_Word32* out, WebRtc_Word32* state);
+
+void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
+ WebRtc_Word32* state);
+
+#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/trunk/src/common_audio/signal_processing/resample_fractional.c b/trunk/src/common_audio/signal_processing/resample_fractional.c
new file mode 100644
index 0000000..51003d4
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_fractional.c
@@ -0,0 +1,242 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling functions between 48, 44, 32 and 24 kHz.
+ * The description headers can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// interpolation coefficients
+static const WebRtc_Word16 kCoefficients48To32[2][8] = {
+ {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
+ {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
+};
+
+static const WebRtc_Word16 kCoefficients32To24[3][8] = {
+ {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
+ {386, -381, -2646, 19062, 19062, -2646, -381, 386},
+ {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
+};
+
+static const WebRtc_Word16 kCoefficients44To32[4][9] = {
+ {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
+ {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
+ {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
+ {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
+};
+
+// Resampling ratio: 2/3
+// input: WebRtc_Word32 (normalized, not saturated) :: size 3 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 2 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+ const WebRtc_Word32 K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (3 input samples -> 2 output samples);
+ // process in sub blocks of size 3 samples.
+ WebRtc_Word32 tmp;
+ WebRtc_Word32 m;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+ tmp += kCoefficients48To32[0][0] * In[0];
+ tmp += kCoefficients48To32[0][1] * In[1];
+ tmp += kCoefficients48To32[0][2] * In[2];
+ tmp += kCoefficients48To32[0][3] * In[3];
+ tmp += kCoefficients48To32[0][4] * In[4];
+ tmp += kCoefficients48To32[0][5] * In[5];
+ tmp += kCoefficients48To32[0][6] * In[6];
+ tmp += kCoefficients48To32[0][7] * In[7];
+ Out[0] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients48To32[1][0] * In[1];
+ tmp += kCoefficients48To32[1][1] * In[2];
+ tmp += kCoefficients48To32[1][2] * In[3];
+ tmp += kCoefficients48To32[1][3] * In[4];
+ tmp += kCoefficients48To32[1][4] * In[5];
+ tmp += kCoefficients48To32[1][5] * In[6];
+ tmp += kCoefficients48To32[1][6] * In[7];
+ tmp += kCoefficients48To32[1][7] * In[8];
+ Out[1] = tmp;
+
+ // update pointers
+ In += 3;
+ Out += 2;
+ }
+}
+
+// Resampling ratio: 3/4
+// input: WebRtc_Word32 (normalized, not saturated) :: size 4 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 3 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+ const WebRtc_Word32 K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (4 input samples -> 3 output samples);
+ // process in sub blocks of size 4 samples.
+ WebRtc_Word32 m;
+ WebRtc_Word32 tmp;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[0][0] * In[0];
+ tmp += kCoefficients32To24[0][1] * In[1];
+ tmp += kCoefficients32To24[0][2] * In[2];
+ tmp += kCoefficients32To24[0][3] * In[3];
+ tmp += kCoefficients32To24[0][4] * In[4];
+ tmp += kCoefficients32To24[0][5] * In[5];
+ tmp += kCoefficients32To24[0][6] * In[6];
+ tmp += kCoefficients32To24[0][7] * In[7];
+ Out[0] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[1][0] * In[1];
+ tmp += kCoefficients32To24[1][1] * In[2];
+ tmp += kCoefficients32To24[1][2] * In[3];
+ tmp += kCoefficients32To24[1][3] * In[4];
+ tmp += kCoefficients32To24[1][4] * In[5];
+ tmp += kCoefficients32To24[1][5] * In[6];
+ tmp += kCoefficients32To24[1][6] * In[7];
+ tmp += kCoefficients32To24[1][7] * In[8];
+ Out[1] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[2][0] * In[2];
+ tmp += kCoefficients32To24[2][1] * In[3];
+ tmp += kCoefficients32To24[2][2] * In[4];
+ tmp += kCoefficients32To24[2][3] * In[5];
+ tmp += kCoefficients32To24[2][4] * In[6];
+ tmp += kCoefficients32To24[2][5] * In[7];
+ tmp += kCoefficients32To24[2][6] * In[8];
+ tmp += kCoefficients32To24[2][7] * In[9];
+ Out[2] = tmp;
+
+ // update pointers
+ In += 4;
+ Out += 3;
+ }
+}
+
+//
+// fractional resampling filters
+// Fout = 11/16 * Fin
+// Fout = 8/11 * Fin
+//
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_ResampDotProduct(const WebRtc_Word32 *in1, const WebRtc_Word32 *in2,
+ const WebRtc_Word16 *coef_ptr, WebRtc_Word32 *out1,
+ WebRtc_Word32 *out2)
+{
+ WebRtc_Word32 tmp1 = 16384;
+ WebRtc_Word32 tmp2 = 16384;
+ WebRtc_Word16 coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ *out1 = tmp1 + coef * in1[8];
+ *out2 = tmp2 + coef * in2[-8];
+}
+
+// Resampling ratio: 8/11
+// input: WebRtc_Word32 (normalized, not saturated) :: size 11 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 8 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+ const WebRtc_Word32 K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (11 input samples -> 8 output samples);
+ // process in sub blocks of size 11 samples.
+ WebRtc_Word32 tmp;
+ WebRtc_Word32 m;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+
+ // first output sample
+ Out[0] = ((WebRtc_Word32)In[3] << 15) + tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ tmp += kCoefficients44To32[3][0] * In[5];
+ tmp += kCoefficients44To32[3][1] * In[6];
+ tmp += kCoefficients44To32[3][2] * In[7];
+ tmp += kCoefficients44To32[3][3] * In[8];
+ tmp += kCoefficients44To32[3][4] * In[9];
+ tmp += kCoefficients44To32[3][5] * In[10];
+ tmp += kCoefficients44To32[3][6] * In[11];
+ tmp += kCoefficients44To32[3][7] * In[12];
+ tmp += kCoefficients44To32[3][8] * In[13];
+ Out[4] = tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
+
+ // update pointers
+ In += 11;
+ Out += 8;
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/signal_processing.gypi b/trunk/src/common_audio/signal_processing/signal_processing.gypi
new file mode 100644
index 0000000..c67bf7c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/signal_processing.gypi
@@ -0,0 +1,85 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'signal_processing',
+ 'type': '<(library)',
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/signal_processing_library.h',
+ 'include/spl_inl.h',
+ 'auto_corr_to_refl_coef.c',
+ 'auto_correlation.c',
+ 'complex_fft.c',
+ 'complex_bit_reverse.c',
+ 'copy_set_operations.c',
+ 'cross_correlation.c',
+ 'division_operations.c',
+ 'dot_product_with_scale.c',
+ 'downsample_fast.c',
+ 'energy.c',
+ 'filter_ar.c',
+ 'filter_ar_fast_q12.c',
+ 'filter_ma_fast_q12.c',
+ 'get_hanning_window.c',
+ 'get_scaling_square.c',
+ 'ilbc_specific_functions.c',
+ 'levinson_durbin.c',
+ 'lpc_to_refl_coef.c',
+ 'min_max_operations.c',
+ 'randomization_functions.c',
+ 'refl_coef_to_lpc.c',
+ 'resample.c',
+ 'resample_48khz.c',
+ 'resample_by_2.c',
+ 'resample_by_2_internal.c',
+ 'resample_by_2_internal.h',
+ 'resample_fractional.c',
+ 'spl_sqrt.c',
+ 'spl_sqrt_floor.c',
+ 'spl_version.c',
+ 'splitting_filter.c',
+ 'sqrt_of_one_minus_x_squared.c',
+ 'vector_scaling_operations.c',
+ ],
+ }, # spl
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'signal_processing_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'signal_processing',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'signal_processing_unittest.cc',
+ ],
+ }, # spl_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc b/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
new file mode 100644
index 0000000..f6d1123
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
@@ -0,0 +1,454 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+#include "gtest/gtest.h"
+
+class SplTest : public testing::Test {
+ protected:
+ virtual ~SplTest() {
+ }
+ void SetUp() {
+ }
+ void TearDown() {
+ }
+};
+
+TEST_F(SplTest, MacroTest) {
+ // Macros with inputs.
+ int A = 10;
+ int B = 21;
+ int a = -3;
+ int b = WEBRTC_SPL_WORD32_MAX;
+ int nr = 2;
+ int d_ptr2 = 0;
+
+ EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
+
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
+ EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
+ WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
+ EXPECT_EQ(65536, d_ptr2);
+
+ EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
+ EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
+ EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
+ b = WEBRTC_SPL_WORD16_MAX >> 1;
+ EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_RSFT16(a, b));
+ EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
+ EXPECT_EQ(16382u, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
+ EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
+ EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
+
+ a = b;
+ b = -3;
+ EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
+ EXPECT_EQ(0u, WEBRTC_SPL_UDIV(a, b));
+
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
+ EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
+ EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
+
+ int a32 = WEBRTC_SPL_WORD32_MAX;
+ int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
+ int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
+ EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
+ EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
+
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
+ EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
+
+ EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
+
+ EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
+
+ EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
+ EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
+ EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
+
+ EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
+
+ // Shifting with negative numbers allowed
+ int shift_amount = 1; // Workaround compiler warning using variable here.
+ // Positive means left shift
+ EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, shift_amount));
+ EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
+
+ // Shifting with negative numbers not allowed
+ // We cannot do casting here due to signed/unsigned problem
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
+
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
+ EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
+ EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1));
+
+ EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
+}
+
+TEST_F(SplTest, InlineTest) {
+ WebRtc_Word16 a = 121;
+ WebRtc_Word16 b = -17;
+ WebRtc_Word32 A = 111121;
+ WebRtc_Word32 B = -1711;
+ char bVersion[8];
+
+ EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
+ EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
+
+ EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
+ EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
+
+ EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
+ EXPECT_EQ(14, WebRtcSpl_NormW32(A));
+ EXPECT_EQ(4, WebRtcSpl_NormW16(B));
+ EXPECT_EQ(15, WebRtcSpl_NormU32(A));
+
+ EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
+}
+
+TEST_F(SplTest, MathOperationsTest) {
+ int A = 117;
+ WebRtc_Word32 num = 117;
+ WebRtc_Word32 den = -5;
+ WebRtc_UWord16 denU = 5;
+ EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
+ EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
+
+
+ EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
+ EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
+ EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
+}
+
+TEST_F(SplTest, BasicArrayOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_UWord8 b8[kVectorSize];
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+
+ WebRtc_UWord8 bTmp8[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+ WebRtc_Word32 bTmp32[kVectorSize];
+
+ WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b16[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b16[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(1, b16[kk]);
+ }
+ WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b32[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b32[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(1, b32[kk]);
+ }
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ bTmp8[kk] = (WebRtc_Word8)kk;
+ bTmp16[kk] = (WebRtc_Word16)kk;
+ bTmp32[kk] = (WebRtc_Word32)kk;
+ }
+ WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b8[kk], bTmp8[kk]);
+ }
+ WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[kk], bTmp16[kk]);
+ }
+// WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(b32[kk], bTmp32[kk]);
+// }
+ EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
+ for (int kk = 0; kk < 2; ++kk) {
+ EXPECT_EQ(kk+2, bTmp16[kk]);
+ }
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b32[kk] = B[kk];
+ b16[kk] = (WebRtc_Word16)B[kk];
+ }
+ WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
+ }
+
+ WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[3-kk], bTmp16[kk]);
+ }
+}
+
+TEST_F(SplTest, MinMaxOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, -1100};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = B[kk];
+ b32[kk] = B[kk];
+ }
+
+ EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, kVectorSize));
+ EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, kVectorSize));
+ EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, kVectorSize));
+ EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, kVectorSize));
+ EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, kVectorSize));
+ EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, kVectorSize));
+
+ EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, kVectorSize));
+ EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, kVectorSize));
+
+ EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
+}
+
+TEST_F(SplTest, VectorOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_Word16 a16[kVectorSize];
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ a16[kk] = B[kk];
+ b16[kk] = B[kk];
+ }
+
+ WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
+ }
+
+ WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, kVectorSize, 2, 2, 0);
+ for (int kk = 0; kk < 2; ++kk) {
+ EXPECT_EQ(614236, b32[kk]);
+ }
+// EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
+ EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, kVectorSize, 2));
+
+ WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
+ }
+ WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
+ }
+
+ WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
+ for (int kk = 0; kk < kVectorSize - 1; ++kk) {
+ EXPECT_EQ(32767, bTmp16[kk]);
+ }
+ EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
+}
+
+TEST_F(SplTest, EstimatorsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = B[kk];
+ b32[kk] = B[kk];
+ }
+
+ EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
+}
+
+TEST_F(SplTest, FilterTest) {
+ const int kVectorSize = 4;
+ const int kFilterOrder = 3;
+ WebRtc_Word16 A[] = {1, 2, 33, 100};
+ WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
+ WebRtc_Word16 B[] = {4, 12, 133, 110};
+ WebRtc_Word16 data_in[kVectorSize];
+ WebRtc_Word16 data_out[kVectorSize];
+ WebRtc_Word16 bTmp16Low[kVectorSize];
+ WebRtc_Word16 bState[kVectorSize];
+ WebRtc_Word16 bStateLow[kVectorSize];
+
+ WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
+ WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ data_in[kk] = A[kk];
+ data_out[kk] = 0;
+ }
+
+ // MA filters.
+ // Note that the input data has |kFilterOrder| states before the actual
+ // data (one sample).
+ WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
+ kFilterOrder + 1, 1);
+ EXPECT_EQ(0, data_out[0]);
+ // AR filters.
+ // Note that the output data has |kFilterOrder| states before the actual
+ // data (one sample).
+ WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
+ kFilterOrder + 1, 1);
+ EXPECT_EQ(0, data_out[kFilterOrder]);
+
+ EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
+ 5,
+ data_in,
+ kVectorSize,
+ bState,
+ kVectorSize,
+ bStateLow,
+ kVectorSize,
+ data_out,
+ bTmp16Low,
+ kVectorSize));
+}
+
+TEST_F(SplTest, RandTest) {
+ const int kVectorSize = 4;
+ WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_UWord32 bSeed = 100000;
+
+ EXPECT_EQ(464449057u, WebRtcSpl_IncreaseSeed(&bSeed));
+ EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
+ EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
+ EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(BU[kk], b16[kk]);
+ }
+}
+
+TEST_F(SplTest, SignalProcessingTest) {
+ const int kVectorSize = 4;
+ int A[] = {1, 2, 33, 100};
+ const WebRtc_Word16 kHanning[4] = { 2399, 8192, 13985, 16384 };
+ WebRtc_Word16 b16[kVectorSize];
+
+ WebRtc_Word16 bTmp16[kVectorSize];
+ WebRtc_Word32 bTmp32[kVectorSize];
+
+ int bScale = 0;
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+
+ EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, kVectorSize, 1, bTmp32, &bScale));
+ // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
+// WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
+//// for (int kk = 0; kk < kVectorSize; ++kk) {
+//// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+//// }
+// WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
+//// for (int kk = 0; kk < kVectorSize; ++kk) {
+//// EXPECT_EQ(a16[kk], b16[kk]);
+//// }
+// WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
+//// for (int kk = 0; kk < kVectorSize; ++kk) {
+//// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+//// }
+
+ WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(kHanning[kk], bTmp16[kk]);
+ }
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+ EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
+ EXPECT_EQ(0, bScale);
+}
+
+TEST_F(SplTest, FFTTest) {
+ WebRtc_Word16 B[] = {1, 2, 33, 100,
+ 2, 3, 34, 101,
+ 3, 4, 35, 102,
+ 4, 5, 36, 103};
+
+ EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
+// for (int kk = 0; kk < 16; ++kk) {
+// EXPECT_EQ(A[kk], B[kk]);
+// }
+ EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
+// for (int kk = 0; kk < 16; ++kk) {
+// EXPECT_EQ(A[kk], B[kk]);
+// }
+ WebRtcSpl_ComplexBitReverse(B, 3);
+ for (int kk = 0; kk < 16; ++kk) {
+ //EXPECT_EQ(A[kk], B[kk]);
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt.c b/trunk/src/common_audio/signal_processing/spl_sqrt.c
new file mode 100644
index 0000000..cfe2cd3
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt.c
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_Sqrt().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in);
+
+WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in)
+{
+
+ WebRtc_Word16 x_half, t16;
+ WebRtc_Word32 A, B, x2;
+
+ /* The following block performs:
+ y=in/2
+ x=y-2^30
+ x_half=x/2^31
+ t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ + 0.875*((x_half)^5)
+ */
+
+ B = in;
+
+ B = WEBRTC_SPL_RSHIFT_W32(B, 1); // B = in/2
+ B = B - ((WebRtc_Word32)0x40000000); // B = in/2 - 1/2
+ x_half = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2
+ B = B + ((WebRtc_Word32)0x40000000); // B = 1 + x/2
+ B = B + ((WebRtc_Word32)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
+
+ x2 = ((WebRtc_Word32)x_half) * ((WebRtc_Word32)x_half) * 2; // A = (x/2)^2
+ A = -x2; // A = -(x/2)^2
+ B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
+
+ A = WEBRTC_SPL_RSHIFT_W32(A, 16);
+ A = A * A * 2; // A = (x/2)^4
+ t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+ B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
+
+ t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+ A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
+ t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+ B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+ t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(x2, 16);
+ A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
+
+ B = B + (A >> 1); // B = B + 0.5*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+ B = B + ((WebRtc_Word32)32768); // Round off bit
+
+ return B;
+}
+
+WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value)
+{
+ /*
+ Algorithm:
+
+ Six term Taylor Series is used here to compute the square root of a number
+ y^0.5 = (1+x)^0.5 where x = y-1
+ = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+ 0.5 <= x < 1
+
+ Example of how the algorithm works, with ut=sqrt(in), and
+ with in=73632 and ut=271 (even shift value case):
+
+ in=73632
+ y= in/131072
+ x=y-1
+ t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+ ut=t*(1/sqrt(2))*512
+
+ or:
+
+ in=73632
+ in2=73632*2^14
+ y= in2/2^31
+ x=y-1
+ t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+ ut=t*(1/sqrt(2))
+ ut2=ut*2^9
+
+ which gives:
+
+ in = 73632
+ in2 = 1206386688
+ y = 0.56176757812500
+ x = -0.43823242187500
+ t = 0.74973506527313
+ ut = 0.53014274874797
+ ut2 = 2.714330873589594e+002
+
+ or:
+
+ in=73632
+ in2=73632*2^14
+ y=in2/2
+ x=y-2^30
+ x_half=x/2^31
+ t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ + 0.875*((x_half)^5)
+ ut=t*(1/sqrt(2))
+ ut2=ut*2^9
+
+ which gives:
+
+ in = 73632
+ in2 = 1206386688
+ y = 603193344
+ x = -470548480
+ x_half = -0.21911621093750
+ t = 0.74973506527313
+ ut = 0.53014274874797
+ ut2 = 2.714330873589594e+002
+
+ */
+
+ WebRtc_Word16 x_norm, nshift, t16, sh;
+ WebRtc_Word32 A;
+
+ WebRtc_Word16 k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
+
+ A = value;
+
+ if (A == 0)
+ return (WebRtc_Word32)0; // sqrt(0) = 0
+
+ sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
+ A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
+ if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
+ {
+ A = A + ((WebRtc_Word32)32768); // Round off bit
+ } else
+ {
+ A = WEBRTC_SPL_WORD32_MAX;
+ }
+
+ x_norm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
+
+ nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
+ nshift = -nshift; // Negate the power for later de-normalization
+
+ A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16);
+ A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
+ A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
+
+ if ((-2 * nshift) == sh)
+ { // Even shift value case
+
+ t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
+
+ A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
+ A = A + ((WebRtc_Word32)32768); // Round off
+ A = A & ((WebRtc_Word32)0x7fff0000); // Round off
+
+ A = WEBRTC_SPL_RSHIFT_W32(A, 15); // A = A>>16
+
+ } else
+ {
+ A = WEBRTC_SPL_RSHIFT_W32(A, 16); // A = A>>16
+ }
+
+ A = A & ((WebRtc_Word32)0x0000ffff);
+ A = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
+
+ return A;
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
new file mode 100644
index 0000000..f0e8ae2
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
@@ -0,0 +1,54 @@
+/*
+ * Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
+ * trunk/third_party_mods/sqrt_floor.
+ *
+ * Minor modifications in code style for WebRTC, 2012.
+ */
+
+#include "signal_processing_library.h"
+
+/*
+ * Algorithm:
+ * Successive approximation of the equation (root + delta) ^ 2 = N
+ * until delta < 1. If delta < 1 we have the integer part of SQRT (N).
+ * Use delta = 2^i for i = 15 .. 0.
+ *
+ * Output precision is 16 bits. Note for large input values (close to
+ * 0x7FFFFFFF), bit 15 (the highest bit of the low 16-bit half word)
+ * contains the MSB information (a non-sign value). Do with caution
+ * if you need to cast the output to int16_t type.
+ *
+ * If the input value is negative, it returns 0.
+ */
+
+#define WEBRTC_SPL_SQRT_ITER(N) \
+ try1 = root + (1 << (N)); \
+ if (value >= try1 << (N)) \
+ { \
+ value -= try1 << (N); \
+ root |= 2 << (N); \
+ }
+
+int32_t WebRtcSpl_SqrtFloor(int32_t value)
+{
+ int32_t root = 0, try1;
+
+ WEBRTC_SPL_SQRT_ITER (15);
+ WEBRTC_SPL_SQRT_ITER (14);
+ WEBRTC_SPL_SQRT_ITER (13);
+ WEBRTC_SPL_SQRT_ITER (12);
+ WEBRTC_SPL_SQRT_ITER (11);
+ WEBRTC_SPL_SQRT_ITER (10);
+ WEBRTC_SPL_SQRT_ITER ( 9);
+ WEBRTC_SPL_SQRT_ITER ( 8);
+ WEBRTC_SPL_SQRT_ITER ( 7);
+ WEBRTC_SPL_SQRT_ITER ( 6);
+ WEBRTC_SPL_SQRT_ITER ( 5);
+ WEBRTC_SPL_SQRT_ITER ( 4);
+ WEBRTC_SPL_SQRT_ITER ( 3);
+ WEBRTC_SPL_SQRT_ITER ( 2);
+ WEBRTC_SPL_SQRT_ITER ( 1);
+ WEBRTC_SPL_SQRT_ITER ( 0);
+
+ return root >> 1;
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
new file mode 100644
index 0000000..cfd9ed0
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
@@ -0,0 +1,88 @@
+@ Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
+@ trunk/third_party_mods/sqrt_floor.
+@
+@ Minor modifications in code style for WebRTC, 2012.
+@ Output is bit-exact with the reference C code in spl_sqrt_floor.c.
+
+@ Input : r0 32 bit unsigned integer
+@ Output: r0 = INT (SQRT (r0)), precision is 16 bits
+@ Registers touched: r1, r2
+
+.global WebRtcSpl_SqrtFloor
+
+.align 2
+WebRtcSpl_SqrtFloor:
+.fnstart
+ mov r1, #3 << 30
+ mov r2, #1 << 30
+
+ @ unroll for i = 0 .. 15
+
+ cmp r0, r2, ror #2 * 0
+ subhs r0, r0, r2, ror #2 * 0
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 1
+ subhs r0, r0, r2, ror #2 * 1
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 2
+ subhs r0, r0, r2, ror #2 * 2
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 3
+ subhs r0, r0, r2, ror #2 * 3
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 4
+ subhs r0, r0, r2, ror #2 * 4
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 5
+ subhs r0, r0, r2, ror #2 * 5
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 6
+ subhs r0, r0, r2, ror #2 * 6
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 7
+ subhs r0, r0, r2, ror #2 * 7
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 8
+ subhs r0, r0, r2, ror #2 * 8
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 9
+ subhs r0, r0, r2, ror #2 * 9
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 10
+ subhs r0, r0, r2, ror #2 * 10
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 11
+ subhs r0, r0, r2, ror #2 * 11
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 12
+ subhs r0, r0, r2, ror #2 * 12
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 13
+ subhs r0, r0, r2, ror #2 * 13
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 14
+ subhs r0, r0, r2, ror #2 * 14
+ adc r2, r1, r2, lsl #1
+
+ cmp r0, r2, ror #2 * 15
+ subhs r0, r0, r2, ror #2 * 15
+ adc r2, r1, r2, lsl #1
+
+ bic r0, r2, #3 << 30 @ for rounding add: cmp r0, r2 adc r2, #1
+ bx lr
+
+.fnend
diff --git a/trunk/src/common_audio/signal_processing/spl_version.c b/trunk/src/common_audio/signal_processing/spl_version.c
new file mode 100644
index 0000000..936925e
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_version.c
@@ -0,0 +1,25 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_get_version().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+
+WebRtc_Word16 WebRtcSpl_get_version(char* version, WebRtc_Word16 length_in_bytes)
+{
+ strncpy(version, "1.2.0", length_in_bytes);
+ return 0;
+}
diff --git a/trunk/src/common_audio/signal_processing/splitting_filter.c b/trunk/src/common_audio/signal_processing/splitting_filter.c
new file mode 100644
index 0000000..f1acf67
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/splitting_filter.c
@@ -0,0 +1,198 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the splitting filter functions.
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// Number of samples in a low/high-band frame.
+enum
+{
+ kBandFrameLength = 160
+};
+
+// QMF filter coefficients in Q16.
+static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
+static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcSpl_AllPassQMF(...)
+//
+// Allpass filter used by the analysis and synthesis parts of the QMF filter.
+//
+// Input:
+// - in_data : Input data sequence (Q10)
+// - data_length : Length of data sequence (>2)
+// - filter_coefficients : Filter coefficients (length 3, Q16)
+//
+// Input & Output:
+// - filter_state : Filter state (length 6, Q10).
+//
+// Output:
+// - out_data : Output data sequence (Q10), length equal to
+// |data_length|
+//
+
+void WebRtcSpl_AllPassQMF(WebRtc_Word32* in_data, const WebRtc_Word16 data_length,
+ WebRtc_Word32* out_data, const WebRtc_UWord16* filter_coefficients,
+ WebRtc_Word32* filter_state)
+{
+ // The procedure is to filter the input with three first order all pass filters
+ // (cascade operations).
+ //
+ // a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
+ // y[n] = ----------- ----------- ----------- x[n]
+ // 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
+ //
+ // The input vector |filter_coefficients| includes these three filter coefficients.
+ // The filter state contains the in_data state, in_data[-1], followed by
+ // the out_data state, out_data[-1]. This is repeated for each cascade.
+ // The first cascade filter will filter the |in_data| and store the output in
+ // |out_data|. The second will the take the |out_data| as input and make an
+ // intermediate storage in |in_data|, to save memory. The third, and final, cascade
+ // filter operation takes the |in_data| (which is the output from the previous cascade
+ // filter) and store the output in |out_data|.
+ // Note that the input vector values are changed during the process.
+ WebRtc_Word16 k;
+ WebRtc_Word32 diff;
+ // First all-pass cascade; filter from in_data to out_data.
+
+ // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
+ // vector position n. Then the final output will be y[n] = y_3[n]
+
+ // First loop, use the states stored in memory.
+ // "diff" should be safe from wrap around since max values are 2^25
+ diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[1]); // = (x[0] - y_1[-1])
+ // y_1[0] = x[-1] + a_1 * (x[0] - y_1[-1])
+ out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
+
+ // For the remaining loops, use previous values.
+ for (k = 1; k < data_length; k++)
+ {
+ diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (x[n] - y_1[n-1])
+ // y_1[n] = x[n-1] + a_1 * (x[n] - y_1[n-1])
+ out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
+ }
+
+ // Update states.
+ filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
+ filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+
+ // Second all-pass cascade; filter from out_data to in_data.
+ diff = WEBRTC_SPL_SUB_SAT_W32(out_data[0], filter_state[3]); // = (y_1[0] - y_2[-1])
+ // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+ in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
+ for (k = 1; k < data_length; k++)
+ {
+ diff = WEBRTC_SPL_SUB_SAT_W32(out_data[k], in_data[k - 1]); // =(y_1[n] - y_2[n-1])
+ // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+ in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
+ }
+
+ filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+ filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+
+ // Third all-pass cascade; filter from in_data to out_data.
+ diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[5]); // = (y_2[0] - y[-1])
+ // y[0] = y_2[-1] + a_3 * (y_2[0] - y[-1])
+ out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
+ for (k = 1; k < data_length; k++)
+ {
+ diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (y_2[n] - y[n-1])
+ // y[n] = y_2[n-1] + a_3 * (y_2[n] - y[n-1])
+ out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
+ }
+ filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+ filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
+}
+
+void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data, WebRtc_Word16* low_band,
+ WebRtc_Word16* high_band, WebRtc_Word32* filter_state1,
+ WebRtc_Word32* filter_state2)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word16 k;
+ WebRtc_Word32 tmp;
+ WebRtc_Word32 half_in1[kBandFrameLength];
+ WebRtc_Word32 half_in2[kBandFrameLength];
+ WebRtc_Word32 filter1[kBandFrameLength];
+ WebRtc_Word32 filter2[kBandFrameLength];
+
+ // Split even and odd samples. Also shift them to Q10.
+ for (i = 0, k = 0; i < kBandFrameLength; i++, k += 2)
+ {
+ half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10);
+ half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10);
+ }
+
+ // All pass filter even and odd samples, independently.
+ WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter1,
+ filter_state1);
+ WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter2,
+ filter_state2);
+
+ // Take the sum and difference of filtered version of odd and even
+ // branches to get upper & lower band.
+ for (i = 0; i < kBandFrameLength; i++)
+ {
+ tmp = filter1[i] + filter2[i] + 1024;
+ tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
+ low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+
+ tmp = filter1[i] - filter2[i] + 1024;
+ tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
+ high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+ }
+}
+
+void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band, const WebRtc_Word16* high_band,
+ WebRtc_Word16* out_data, WebRtc_Word32* filter_state1,
+ WebRtc_Word32* filter_state2)
+{
+ WebRtc_Word32 tmp;
+ WebRtc_Word32 half_in1[kBandFrameLength];
+ WebRtc_Word32 half_in2[kBandFrameLength];
+ WebRtc_Word32 filter1[kBandFrameLength];
+ WebRtc_Word32 filter2[kBandFrameLength];
+ WebRtc_Word16 i;
+ WebRtc_Word16 k;
+
+ // Obtain the sum and difference channels out of upper and lower-band channels.
+ // Also shift to Q10 domain.
+ for (i = 0; i < kBandFrameLength; i++)
+ {
+ tmp = (WebRtc_Word32)low_band[i] + (WebRtc_Word32)high_band[i];
+ half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
+ tmp = (WebRtc_Word32)low_band[i] - (WebRtc_Word32)high_band[i];
+ half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
+ }
+
+ // all-pass filter the sum and difference channels
+ WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter2,
+ filter_state1);
+ WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter1,
+ filter_state2);
+
+ // The filtered signals are even and odd samples of the output. Combine
+ // them. The signals are Q10 should shift them back to Q0 and take care of
+ // saturation.
+ for (i = 0, k = 0; i < kBandFrameLength; i++)
+ {
+ tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+
+ tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+ }
+
+}
diff --git a/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
new file mode 100644
index 0000000..9fb2c73
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_SqrtOfOneMinusXSquared().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16 *xQ15, int vector_length,
+ WebRtc_Word16 *yQ15)
+{
+ WebRtc_Word32 sq;
+ int m;
+ WebRtc_Word16 tmp;
+
+ for (m = 0; m < vector_length; m++)
+ {
+ tmp = xQ15[m];
+ sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
+ sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
+ sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
+ yQ15[m] = (WebRtc_Word16)sq;
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/vector_scaling_operations.c b/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
new file mode 100644
index 0000000..20d239c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
@@ -0,0 +1,151 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the functions
+ * WebRtcSpl_VectorBitShiftW16()
+ * WebRtcSpl_VectorBitShiftW32()
+ * WebRtcSpl_VectorBitShiftW32ToW16()
+ * WebRtcSpl_ScaleVector()
+ * WebRtcSpl_ScaleVectorWithSat()
+ * WebRtcSpl_ScaleAndAddVectors()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16 *res,
+ WebRtc_Word16 length,
+ G_CONST WebRtc_Word16 *in,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+
+ if (right_shifts > 0)
+ {
+ for (i = length; i > 0; i--)
+ {
+ (*res++) = ((*in++) >> right_shifts);
+ }
+ } else
+ {
+ for (i = length; i > 0; i--)
+ {
+ (*res++) = ((*in++) << (-right_shifts));
+ }
+ }
+}
+
+void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
+ WebRtc_Word16 vector_length,
+ G_CONST WebRtc_Word32 *in_vector,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+
+ if (right_shifts > 0)
+ {
+ for (i = vector_length; i > 0; i--)
+ {
+ (*out_vector++) = ((*in_vector++) >> right_shifts);
+ }
+ } else
+ {
+ for (i = vector_length; i > 0; i--)
+ {
+ (*out_vector++) = ((*in_vector++) << (-right_shifts));
+ }
+ }
+}
+
+void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
+ WebRtc_Word16 length,
+ G_CONST WebRtc_Word32 *in,
+ WebRtc_Word16 right_shifts)
+{
+ int i;
+
+ if (right_shifts >= 0)
+ {
+ for (i = length; i > 0; i--)
+ {
+ (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
+ }
+ } else
+ {
+ WebRtc_Word16 left_shifts = -right_shifts;
+ for (i = length; i > 0; i--)
+ {
+ (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
+ }
+ }
+}
+
+void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
+ WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
+ WebRtc_Word16 right_shifts)
+{
+ // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+ int i;
+ G_CONST WebRtc_Word16 *inptr;
+ WebRtc_Word16 *outptr;
+
+ inptr = in_vector;
+ outptr = out_vector;
+
+ for (i = 0; i < in_vector_length; i++)
+ {
+ (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
+ }
+}
+
+void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
+ WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
+ WebRtc_Word16 right_shifts)
+{
+ // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+ int i;
+ WebRtc_Word32 tmpW32;
+ G_CONST WebRtc_Word16 *inptr;
+ WebRtc_Word16 *outptr;
+
+ inptr = in_vector;
+ outptr = out_vector;
+
+ for (i = 0; i < in_vector_length; i++)
+ {
+ tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
+ (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
+ }
+}
+
+void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16 *in1, WebRtc_Word16 gain1, int shift1,
+ G_CONST WebRtc_Word16 *in2, WebRtc_Word16 gain2, int shift2,
+ WebRtc_Word16 *out, int vector_length)
+{
+ // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
+ int i;
+ G_CONST WebRtc_Word16 *in1ptr;
+ G_CONST WebRtc_Word16 *in2ptr;
+ WebRtc_Word16 *outptr;
+
+ in1ptr = in1;
+ in2ptr = in2;
+ outptr = out;
+
+ for (i = 0; i < vector_length; i++)
+ {
+ (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
+ + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
+ }
+}
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
new file mode 100644
index 0000000..b587380
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
@@ -0,0 +1,704 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the Q14 radix-8 tables used in ARM9e optimizations.
+ *
+ */
+
+extern const int s_Q14S_8;
+const int s_Q14S_8 = 1024;
+extern const unsigned short t_Q14S_8[2032];
+const unsigned short t_Q14S_8[2032] = {
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+ 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+ 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+ 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+ 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+ 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+ 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
+ 0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
+ 0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
+ 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+ 0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
+ 0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
+ 0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
+ 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+ 0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
+ 0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
+ 0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
+ 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+ 0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
+ 0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
+ 0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
+ 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+ 0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
+ 0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
+ 0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
+ 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+ 0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
+ 0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
+ 0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
+ 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+ 0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
+ 0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
+ 0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
+ 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+ 0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
+ 0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
+ 0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x3e69,0x0192 ,0x3f36,0x00c9 ,0x3d9a,0x025b ,
+ 0x3cc8,0x0324 ,0x3e69,0x0192 ,0x3b1e,0x04b5 ,
+ 0x3b1e,0x04b5 ,0x3d9a,0x025b ,0x388e,0x070e ,
+ 0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
+ 0x37af,0x07d6 ,0x3bf4,0x03ed ,0x3334,0x0bb7 ,
+ 0x35eb,0x0964 ,0x3b1e,0x04b5 ,0x306c,0x0e06 ,
+ 0x341e,0x0af1 ,0x3a46,0x057e ,0x2d93,0x1050 ,
+ 0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
+ 0x306c,0x0e06 ,0x388e,0x070e ,0x27b3,0x14d2 ,
+ 0x2e88,0x0f8d ,0x37af,0x07d6 ,0x24ae,0x1709 ,
+ 0x2c9d,0x1112 ,0x36ce,0x089d ,0x219c,0x1937 ,
+ 0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
+ 0x28b2,0x1413 ,0x3505,0x0a2b ,0x1b56,0x1d79 ,
+ 0x26b3,0x1590 ,0x341e,0x0af1 ,0x1824,0x1f8c ,
+ 0x24ae,0x1709 ,0x3334,0x0bb7 ,0x14ea,0x2193 ,
+ 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+ 0x2093,0x19ef ,0x315b,0x0d41 ,0x0e61,0x257e ,
+ 0x1e7e,0x1b5d ,0x306c,0x0e06 ,0x0b14,0x2760 ,
+ 0x1c64,0x1cc6 ,0x2f7b,0x0eca ,0x07c4,0x2935 ,
+ 0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
+ 0x1824,0x1f8c ,0x2d93,0x1050 ,0x011c,0x2cb2 ,
+ 0x15fe,0x20e7 ,0x2c9d,0x1112 ,0xfdc7,0x2e5a ,
+ 0x13d5,0x223d ,0x2ba4,0x11d3 ,0xfa73,0x2ff2 ,
+ 0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
+ 0x0f79,0x24da ,0x29af,0x1354 ,0xf3d2,0x32ef ,
+ 0x0d48,0x2620 ,0x28b2,0x1413 ,0xf087,0x3453 ,
+ 0x0b14,0x2760 ,0x27b3,0x14d2 ,0xed41,0x35a5 ,
+ 0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
+ 0x06a9,0x29ce ,0x25b1,0x164c ,0xe6cb,0x3812 ,
+ 0x0471,0x2afb ,0x24ae,0x1709 ,0xe39c,0x392b ,
+ 0x0239,0x2c21 ,0x23a9,0x17c4 ,0xe077,0x3a30 ,
+ 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+ 0xfdc7,0x2e5a ,0x219c,0x1937 ,0xda4f,0x3bfd ,
+ 0xfb8f,0x2f6c ,0x2093,0x19ef ,0xd74e,0x3cc5 ,
+ 0xf957,0x3076 ,0x1f89,0x1aa7 ,0xd45c,0x3d78 ,
+ 0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
+ 0xf4ec,0x3274 ,0x1d72,0x1c12 ,0xcea5,0x3e9d ,
+ 0xf2b8,0x3368 ,0x1c64,0x1cc6 ,0xcbe2,0x3f0f ,
+ 0xf087,0x3453 ,0x1b56,0x1d79 ,0xc932,0x3f6b ,
+ 0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
+ 0xec2b,0x3612 ,0x1935,0x1edc ,0xc40c,0x3fe1 ,
+ 0xea02,0x36e5 ,0x1824,0x1f8c ,0xc197,0x3ffb ,
+ 0xe7dc,0x37b0 ,0x1711,0x203a ,0xbf38,0x3fff ,
+ 0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
+ 0xe39c,0x392b ,0x14ea,0x2193 ,0xbabf,0x3fc4 ,
+ 0xe182,0x39db ,0x13d5,0x223d ,0xb8a6,0x3f85 ,
+ 0xdf6d,0x3a82 ,0x12bf,0x22e7 ,0xb6a5,0x3f30 ,
+ 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+ 0xdb52,0x3bb6 ,0x1091,0x2435 ,0xb2f2,0x3e45 ,
+ 0xd94d,0x3c42 ,0x0f79,0x24da ,0xb140,0x3daf ,
+ 0xd74e,0x3cc5 ,0x0e61,0x257e ,0xafa9,0x3d03 ,
+ 0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
+ 0xd363,0x3daf ,0x0c2e,0x26c1 ,0xacd0,0x3b6d ,
+ 0xd178,0x3e15 ,0x0b14,0x2760 ,0xab8e,0x3a82 ,
+ 0xcf94,0x3e72 ,0x09fa,0x27fe ,0xaa6a,0x3984 ,
+ 0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
+ 0xcbe2,0x3f0f ,0x07c4,0x2935 ,0xa87b,0x374b ,
+ 0xca15,0x3f4f ,0x06a9,0x29ce ,0xa7b1,0x3612 ,
+ 0xc851,0x3f85 ,0x058d,0x2a65 ,0xa705,0x34c6 ,
+ 0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
+ 0xc4e2,0x3fd4 ,0x0355,0x2b8f ,0xa60b,0x31f8 ,
+ 0xc338,0x3fec ,0x0239,0x2c21 ,0xa5bc,0x3076 ,
+ 0xc197,0x3ffb ,0x011c,0x2cb2 ,0xa58d,0x2ee4 ,
+ 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+ 0xbe73,0x3ffb ,0xfee4,0x2dcf ,0xa58d,0x2b8f ,
+ 0xbcf0,0x3fec ,0xfdc7,0x2e5a ,0xa5bc,0x29ce ,
+ 0xbb77,0x3fd4 ,0xfcab,0x2ee4 ,0xa60b,0x27fe ,
+ 0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
+ 0xb8a6,0x3f85 ,0xfa73,0x2ff2 ,0xa705,0x2435 ,
+ 0xb74d,0x3f4f ,0xf957,0x3076 ,0xa7b1,0x223d ,
+ 0xb600,0x3f0f ,0xf83c,0x30f9 ,0xa87b,0x203a ,
+ 0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
+ 0xb388,0x3e72 ,0xf606,0x31f8 ,0xaa6a,0x1c12 ,
+ 0xb25e,0x3e15 ,0xf4ec,0x3274 ,0xab8e,0x19ef ,
+ 0xb140,0x3daf ,0xf3d2,0x32ef ,0xacd0,0x17c4 ,
+ 0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
+ 0xaf28,0x3cc5 ,0xf19f,0x33df ,0xafa9,0x1354 ,
+ 0xae2e,0x3c42 ,0xf087,0x3453 ,0xb140,0x1112 ,
+ 0xad41,0x3bb6 ,0xef6f,0x34c6 ,0xb2f2,0x0eca ,
+ 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+ 0xab8e,0x3a82 ,0xed41,0x35a5 ,0xb6a5,0x0a2b ,
+ 0xaac8,0x39db ,0xec2b,0x3612 ,0xb8a6,0x07d6 ,
+ 0xaa0f,0x392b ,0xeb16,0x367d ,0xbabf,0x057e ,
+ 0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
+ 0xa8c5,0x37b0 ,0xe8ef,0x374b ,0xbf38,0x00c9 ,
+ 0xa834,0x36e5 ,0xe7dc,0x37b0 ,0xc197,0xfe6e ,
+ 0xa7b1,0x3612 ,0xe6cb,0x3812 ,0xc40c,0xfc13 ,
+ 0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
+ 0xa6d3,0x3453 ,0xe4aa,0x38cf ,0xc932,0xf763 ,
+ 0xa678,0x3368 ,0xe39c,0x392b ,0xcbe2,0xf50f ,
+ 0xa62c,0x3274 ,0xe28e,0x3984 ,0xcea5,0xf2bf ,
+ 0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
+ 0xa5bc,0x3076 ,0xe077,0x3a30 ,0xd45c,0xee2d ,
+ 0xa599,0x2f6c ,0xdf6d,0x3a82 ,0xd74e,0xebed ,
+ 0xa585,0x2e5a ,0xde64,0x3ad3 ,0xda4f,0xe9b4 ,
+ 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+ 0xa585,0x2c21 ,0xdc57,0x3b6d ,0xe077,0xe559 ,
+ 0xa599,0x2afb ,0xdb52,0x3bb6 ,0xe39c,0xe33a ,
+ 0xa5bc,0x29ce ,0xda4f,0x3bfd ,0xe6cb,0xe124 ,
+ 0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
+ 0xa62c,0x2760 ,0xd84d,0x3c85 ,0xed41,0xdd19 ,
+ 0xa678,0x2620 ,0xd74e,0x3cc5 ,0xf087,0xdb26 ,
+ 0xa6d3,0x24da ,0xd651,0x3d03 ,0xf3d2,0xd93f ,
+ 0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
+ 0xa7b1,0x223d ,0xd45c,0x3d78 ,0xfa73,0xd59b ,
+ 0xa834,0x20e7 ,0xd363,0x3daf ,0xfdc7,0xd3df ,
+ 0xa8c5,0x1f8c ,0xd26d,0x3de3 ,0x011c,0xd231 ,
+ 0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
+ 0xaa0f,0x1cc6 ,0xd085,0x3e45 ,0x07c4,0xcf07 ,
+ 0xaac8,0x1b5d ,0xcf94,0x3e72 ,0x0b14,0xcd8c ,
+ 0xab8e,0x19ef ,0xcea5,0x3e9d ,0x0e61,0xcc21 ,
+ 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+ 0xad41,0x1709 ,0xcccc,0x3eeb ,0x14ea,0xc983 ,
+ 0xae2e,0x1590 ,0xcbe2,0x3f0f ,0x1824,0xc850 ,
+ 0xaf28,0x1413 ,0xcafb,0x3f30 ,0x1b56,0xc731 ,
+ 0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
+ 0xb140,0x1112 ,0xc932,0x3f6b ,0x219c,0xc52d ,
+ 0xb25e,0x0f8d ,0xc851,0x3f85 ,0x24ae,0xc44a ,
+ 0xb388,0x0e06 ,0xc772,0x3f9c ,0x27b3,0xc37b ,
+ 0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
+ 0xb600,0x0af1 ,0xc5ba,0x3fc4 ,0x2d93,0xc21d ,
+ 0xb74d,0x0964 ,0xc4e2,0x3fd4 ,0x306c,0xc18e ,
+ 0xb8a6,0x07d6 ,0xc40c,0x3fe1 ,0x3334,0xc115 ,
+ 0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
+ 0xbb77,0x04b5 ,0xc266,0x3ff5 ,0x388e,0xc064 ,
+ 0xbcf0,0x0324 ,0xc197,0x3ffb ,0x3b1e,0xc02c ,
+ 0xbe73,0x0192 ,0xc0ca,0x3fff ,0x3d9a,0xc00b ,
+ 0x4000,0x0000 ,0x3f9b,0x0065 ,0x3f36,0x00c9 ,
+ 0x3ed0,0x012e ,0x3e69,0x0192 ,0x3e02,0x01f7 ,
+ 0x3d9a,0x025b ,0x3d31,0x02c0 ,0x3cc8,0x0324 ,
+ 0x3c5f,0x0388 ,0x3bf4,0x03ed ,0x3b8a,0x0451 ,
+ 0x3b1e,0x04b5 ,0x3ab2,0x051a ,0x3a46,0x057e ,
+ 0x39d9,0x05e2 ,0x396b,0x0646 ,0x38fd,0x06aa ,
+ 0x388e,0x070e ,0x381f,0x0772 ,0x37af,0x07d6 ,
+ 0x373f,0x0839 ,0x36ce,0x089d ,0x365d,0x0901 ,
+ 0x35eb,0x0964 ,0x3578,0x09c7 ,0x3505,0x0a2b ,
+ 0x3492,0x0a8e ,0x341e,0x0af1 ,0x33a9,0x0b54 ,
+ 0x3334,0x0bb7 ,0x32bf,0x0c1a ,0x3249,0x0c7c ,
+ 0x31d2,0x0cdf ,0x315b,0x0d41 ,0x30e4,0x0da4 ,
+ 0x306c,0x0e06 ,0x2ff4,0x0e68 ,0x2f7b,0x0eca ,
+ 0x2f02,0x0f2b ,0x2e88,0x0f8d ,0x2e0e,0x0fee ,
+ 0x2d93,0x1050 ,0x2d18,0x10b1 ,0x2c9d,0x1112 ,
+ 0x2c21,0x1173 ,0x2ba4,0x11d3 ,0x2b28,0x1234 ,
+ 0x2aaa,0x1294 ,0x2a2d,0x12f4 ,0x29af,0x1354 ,
+ 0x2931,0x13b4 ,0x28b2,0x1413 ,0x2833,0x1473 ,
+ 0x27b3,0x14d2 ,0x2733,0x1531 ,0x26b3,0x1590 ,
+ 0x2632,0x15ee ,0x25b1,0x164c ,0x252f,0x16ab ,
+ 0x24ae,0x1709 ,0x242b,0x1766 ,0x23a9,0x17c4 ,
+ 0x2326,0x1821 ,0x22a3,0x187e ,0x221f,0x18db ,
+ 0x219c,0x1937 ,0x2117,0x1993 ,0x2093,0x19ef ,
+ 0x200e,0x1a4b ,0x1f89,0x1aa7 ,0x1f04,0x1b02 ,
+ 0x1e7e,0x1b5d ,0x1df8,0x1bb8 ,0x1d72,0x1c12 ,
+ 0x1ceb,0x1c6c ,0x1c64,0x1cc6 ,0x1bdd,0x1d20 ,
+ 0x1b56,0x1d79 ,0x1ace,0x1dd3 ,0x1a46,0x1e2b ,
+ 0x19be,0x1e84 ,0x1935,0x1edc ,0x18ad,0x1f34 ,
+ 0x1824,0x1f8c ,0x179b,0x1fe3 ,0x1711,0x203a ,
+ 0x1688,0x2091 ,0x15fe,0x20e7 ,0x1574,0x213d ,
+ 0x14ea,0x2193 ,0x145f,0x21e8 ,0x13d5,0x223d ,
+ 0x134a,0x2292 ,0x12bf,0x22e7 ,0x1234,0x233b ,
+ 0x11a8,0x238e ,0x111d,0x23e2 ,0x1091,0x2435 ,
+ 0x1005,0x2488 ,0x0f79,0x24da ,0x0eed,0x252c ,
+ 0x0e61,0x257e ,0x0dd4,0x25cf ,0x0d48,0x2620 ,
+ 0x0cbb,0x2671 ,0x0c2e,0x26c1 ,0x0ba1,0x2711 ,
+ 0x0b14,0x2760 ,0x0a87,0x27af ,0x09fa,0x27fe ,
+ 0x096d,0x284c ,0x08df,0x289a ,0x0852,0x28e7 ,
+ 0x07c4,0x2935 ,0x0736,0x2981 ,0x06a9,0x29ce ,
+ 0x061b,0x2a1a ,0x058d,0x2a65 ,0x04ff,0x2ab0 ,
+ 0x0471,0x2afb ,0x03e3,0x2b45 ,0x0355,0x2b8f ,
+ 0x02c7,0x2bd8 ,0x0239,0x2c21 ,0x01aa,0x2c6a ,
+ 0x011c,0x2cb2 ,0x008e,0x2cfa ,0x0000,0x2d41 ,
+ 0xff72,0x2d88 ,0xfee4,0x2dcf ,0xfe56,0x2e15 ,
+ 0xfdc7,0x2e5a ,0xfd39,0x2e9f ,0xfcab,0x2ee4 ,
+ 0xfc1d,0x2f28 ,0xfb8f,0x2f6c ,0xfb01,0x2faf ,
+ 0xfa73,0x2ff2 ,0xf9e5,0x3034 ,0xf957,0x3076 ,
+ 0xf8ca,0x30b8 ,0xf83c,0x30f9 ,0xf7ae,0x3139 ,
+ 0xf721,0x3179 ,0xf693,0x31b9 ,0xf606,0x31f8 ,
+ 0xf579,0x3236 ,0xf4ec,0x3274 ,0xf45f,0x32b2 ,
+ 0xf3d2,0x32ef ,0xf345,0x332c ,0xf2b8,0x3368 ,
+ 0xf22c,0x33a3 ,0xf19f,0x33df ,0xf113,0x3419 ,
+ 0xf087,0x3453 ,0xeffb,0x348d ,0xef6f,0x34c6 ,
+ 0xeee3,0x34ff ,0xee58,0x3537 ,0xedcc,0x356e ,
+ 0xed41,0x35a5 ,0xecb6,0x35dc ,0xec2b,0x3612 ,
+ 0xeba1,0x3648 ,0xeb16,0x367d ,0xea8c,0x36b1 ,
+ 0xea02,0x36e5 ,0xe978,0x3718 ,0xe8ef,0x374b ,
+ 0xe865,0x377e ,0xe7dc,0x37b0 ,0xe753,0x37e1 ,
+ 0xe6cb,0x3812 ,0xe642,0x3842 ,0xe5ba,0x3871 ,
+ 0xe532,0x38a1 ,0xe4aa,0x38cf ,0xe423,0x38fd ,
+ 0xe39c,0x392b ,0xe315,0x3958 ,0xe28e,0x3984 ,
+ 0xe208,0x39b0 ,0xe182,0x39db ,0xe0fc,0x3a06 ,
+ 0xe077,0x3a30 ,0xdff2,0x3a59 ,0xdf6d,0x3a82 ,
+ 0xdee9,0x3aab ,0xde64,0x3ad3 ,0xdde1,0x3afa ,
+ 0xdd5d,0x3b21 ,0xdcda,0x3b47 ,0xdc57,0x3b6d ,
+ 0xdbd5,0x3b92 ,0xdb52,0x3bb6 ,0xdad1,0x3bda ,
+ 0xda4f,0x3bfd ,0xd9ce,0x3c20 ,0xd94d,0x3c42 ,
+ 0xd8cd,0x3c64 ,0xd84d,0x3c85 ,0xd7cd,0x3ca5 ,
+ 0xd74e,0x3cc5 ,0xd6cf,0x3ce4 ,0xd651,0x3d03 ,
+ 0xd5d3,0x3d21 ,0xd556,0x3d3f ,0xd4d8,0x3d5b ,
+ 0xd45c,0x3d78 ,0xd3df,0x3d93 ,0xd363,0x3daf ,
+ 0xd2e8,0x3dc9 ,0xd26d,0x3de3 ,0xd1f2,0x3dfc ,
+ 0xd178,0x3e15 ,0xd0fe,0x3e2d ,0xd085,0x3e45 ,
+ 0xd00c,0x3e5c ,0xcf94,0x3e72 ,0xcf1c,0x3e88 ,
+ 0xcea5,0x3e9d ,0xce2e,0x3eb1 ,0xcdb7,0x3ec5 ,
+ 0xcd41,0x3ed8 ,0xcccc,0x3eeb ,0xcc57,0x3efd ,
+ 0xcbe2,0x3f0f ,0xcb6e,0x3f20 ,0xcafb,0x3f30 ,
+ 0xca88,0x3f40 ,0xca15,0x3f4f ,0xc9a3,0x3f5d ,
+ 0xc932,0x3f6b ,0xc8c1,0x3f78 ,0xc851,0x3f85 ,
+ 0xc7e1,0x3f91 ,0xc772,0x3f9c ,0xc703,0x3fa7 ,
+ 0xc695,0x3fb1 ,0xc627,0x3fbb ,0xc5ba,0x3fc4 ,
+ 0xc54e,0x3fcc ,0xc4e2,0x3fd4 ,0xc476,0x3fdb ,
+ 0xc40c,0x3fe1 ,0xc3a1,0x3fe7 ,0xc338,0x3fec ,
+ 0xc2cf,0x3ff1 ,0xc266,0x3ff5 ,0xc1fe,0x3ff8 ,
+ 0xc197,0x3ffb ,0xc130,0x3ffd ,0xc0ca,0x3fff ,
+ 0xc065,0x4000 ,0xc000,0x4000 ,0xbf9c,0x4000 ,
+ 0xbf38,0x3fff ,0xbed5,0x3ffd ,0xbe73,0x3ffb ,
+ 0xbe11,0x3ff8 ,0xbdb0,0x3ff5 ,0xbd50,0x3ff1 ,
+ 0xbcf0,0x3fec ,0xbc91,0x3fe7 ,0xbc32,0x3fe1 ,
+ 0xbbd4,0x3fdb ,0xbb77,0x3fd4 ,0xbb1b,0x3fcc ,
+ 0xbabf,0x3fc4 ,0xba64,0x3fbb ,0xba09,0x3fb1 ,
+ 0xb9af,0x3fa7 ,0xb956,0x3f9c ,0xb8fd,0x3f91 ,
+ 0xb8a6,0x3f85 ,0xb84f,0x3f78 ,0xb7f8,0x3f6b ,
+ 0xb7a2,0x3f5d ,0xb74d,0x3f4f ,0xb6f9,0x3f40 ,
+ 0xb6a5,0x3f30 ,0xb652,0x3f20 ,0xb600,0x3f0f ,
+ 0xb5af,0x3efd ,0xb55e,0x3eeb ,0xb50e,0x3ed8 ,
+ 0xb4be,0x3ec5 ,0xb470,0x3eb1 ,0xb422,0x3e9d ,
+ 0xb3d5,0x3e88 ,0xb388,0x3e72 ,0xb33d,0x3e5c ,
+ 0xb2f2,0x3e45 ,0xb2a7,0x3e2d ,0xb25e,0x3e15 ,
+ 0xb215,0x3dfc ,0xb1cd,0x3de3 ,0xb186,0x3dc9 ,
+ 0xb140,0x3daf ,0xb0fa,0x3d93 ,0xb0b5,0x3d78 ,
+ 0xb071,0x3d5b ,0xb02d,0x3d3f ,0xafeb,0x3d21 ,
+ 0xafa9,0x3d03 ,0xaf68,0x3ce4 ,0xaf28,0x3cc5 ,
+ 0xaee8,0x3ca5 ,0xaea9,0x3c85 ,0xae6b,0x3c64 ,
+ 0xae2e,0x3c42 ,0xadf2,0x3c20 ,0xadb6,0x3bfd ,
+ 0xad7b,0x3bda ,0xad41,0x3bb6 ,0xad08,0x3b92 ,
+ 0xacd0,0x3b6d ,0xac98,0x3b47 ,0xac61,0x3b21 ,
+ 0xac2b,0x3afa ,0xabf6,0x3ad3 ,0xabc2,0x3aab ,
+ 0xab8e,0x3a82 ,0xab5b,0x3a59 ,0xab29,0x3a30 ,
+ 0xaaf8,0x3a06 ,0xaac8,0x39db ,0xaa98,0x39b0 ,
+ 0xaa6a,0x3984 ,0xaa3c,0x3958 ,0xaa0f,0x392b ,
+ 0xa9e3,0x38fd ,0xa9b7,0x38cf ,0xa98d,0x38a1 ,
+ 0xa963,0x3871 ,0xa93a,0x3842 ,0xa912,0x3812 ,
+ 0xa8eb,0x37e1 ,0xa8c5,0x37b0 ,0xa89f,0x377e ,
+ 0xa87b,0x374b ,0xa857,0x3718 ,0xa834,0x36e5 ,
+ 0xa812,0x36b1 ,0xa7f1,0x367d ,0xa7d0,0x3648 ,
+ 0xa7b1,0x3612 ,0xa792,0x35dc ,0xa774,0x35a5 ,
+ 0xa757,0x356e ,0xa73b,0x3537 ,0xa71f,0x34ff ,
+ 0xa705,0x34c6 ,0xa6eb,0x348d ,0xa6d3,0x3453 ,
+ 0xa6bb,0x3419 ,0xa6a4,0x33df ,0xa68e,0x33a3 ,
+ 0xa678,0x3368 ,0xa664,0x332c ,0xa650,0x32ef ,
+ 0xa63e,0x32b2 ,0xa62c,0x3274 ,0xa61b,0x3236 ,
+ 0xa60b,0x31f8 ,0xa5fb,0x31b9 ,0xa5ed,0x3179 ,
+ 0xa5e0,0x3139 ,0xa5d3,0x30f9 ,0xa5c7,0x30b8 ,
+ 0xa5bc,0x3076 ,0xa5b2,0x3034 ,0xa5a9,0x2ff2 ,
+ 0xa5a1,0x2faf ,0xa599,0x2f6c ,0xa593,0x2f28 ,
+ 0xa58d,0x2ee4 ,0xa588,0x2e9f ,0xa585,0x2e5a ,
+ 0xa581,0x2e15 ,0xa57f,0x2dcf ,0xa57e,0x2d88 ,
+ 0xa57e,0x2d41 ,0xa57e,0x2cfa ,0xa57f,0x2cb2 ,
+ 0xa581,0x2c6a ,0xa585,0x2c21 ,0xa588,0x2bd8 ,
+ 0xa58d,0x2b8f ,0xa593,0x2b45 ,0xa599,0x2afb ,
+ 0xa5a1,0x2ab0 ,0xa5a9,0x2a65 ,0xa5b2,0x2a1a ,
+ 0xa5bc,0x29ce ,0xa5c7,0x2981 ,0xa5d3,0x2935 ,
+ 0xa5e0,0x28e7 ,0xa5ed,0x289a ,0xa5fb,0x284c ,
+ 0xa60b,0x27fe ,0xa61b,0x27af ,0xa62c,0x2760 ,
+ 0xa63e,0x2711 ,0xa650,0x26c1 ,0xa664,0x2671 ,
+ 0xa678,0x2620 ,0xa68e,0x25cf ,0xa6a4,0x257e ,
+ 0xa6bb,0x252c ,0xa6d3,0x24da ,0xa6eb,0x2488 ,
+ 0xa705,0x2435 ,0xa71f,0x23e2 ,0xa73b,0x238e ,
+ 0xa757,0x233b ,0xa774,0x22e7 ,0xa792,0x2292 ,
+ 0xa7b1,0x223d ,0xa7d0,0x21e8 ,0xa7f1,0x2193 ,
+ 0xa812,0x213d ,0xa834,0x20e7 ,0xa857,0x2091 ,
+ 0xa87b,0x203a ,0xa89f,0x1fe3 ,0xa8c5,0x1f8c ,
+ 0xa8eb,0x1f34 ,0xa912,0x1edc ,0xa93a,0x1e84 ,
+ 0xa963,0x1e2b ,0xa98d,0x1dd3 ,0xa9b7,0x1d79 ,
+ 0xa9e3,0x1d20 ,0xaa0f,0x1cc6 ,0xaa3c,0x1c6c ,
+ 0xaa6a,0x1c12 ,0xaa98,0x1bb8 ,0xaac8,0x1b5d ,
+ 0xaaf8,0x1b02 ,0xab29,0x1aa7 ,0xab5b,0x1a4b ,
+ 0xab8e,0x19ef ,0xabc2,0x1993 ,0xabf6,0x1937 ,
+ 0xac2b,0x18db ,0xac61,0x187e ,0xac98,0x1821 ,
+ 0xacd0,0x17c4 ,0xad08,0x1766 ,0xad41,0x1709 ,
+ 0xad7b,0x16ab ,0xadb6,0x164c ,0xadf2,0x15ee ,
+ 0xae2e,0x1590 ,0xae6b,0x1531 ,0xaea9,0x14d2 ,
+ 0xaee8,0x1473 ,0xaf28,0x1413 ,0xaf68,0x13b4 ,
+ 0xafa9,0x1354 ,0xafeb,0x12f4 ,0xb02d,0x1294 ,
+ 0xb071,0x1234 ,0xb0b5,0x11d3 ,0xb0fa,0x1173 ,
+ 0xb140,0x1112 ,0xb186,0x10b1 ,0xb1cd,0x1050 ,
+ 0xb215,0x0fee ,0xb25e,0x0f8d ,0xb2a7,0x0f2b ,
+ 0xb2f2,0x0eca ,0xb33d,0x0e68 ,0xb388,0x0e06 ,
+ 0xb3d5,0x0da4 ,0xb422,0x0d41 ,0xb470,0x0cdf ,
+ 0xb4be,0x0c7c ,0xb50e,0x0c1a ,0xb55e,0x0bb7 ,
+ 0xb5af,0x0b54 ,0xb600,0x0af1 ,0xb652,0x0a8e ,
+ 0xb6a5,0x0a2b ,0xb6f9,0x09c7 ,0xb74d,0x0964 ,
+ 0xb7a2,0x0901 ,0xb7f8,0x089d ,0xb84f,0x0839 ,
+ 0xb8a6,0x07d6 ,0xb8fd,0x0772 ,0xb956,0x070e ,
+ 0xb9af,0x06aa ,0xba09,0x0646 ,0xba64,0x05e2 ,
+ 0xbabf,0x057e ,0xbb1b,0x051a ,0xbb77,0x04b5 ,
+ 0xbbd4,0x0451 ,0xbc32,0x03ed ,0xbc91,0x0388 ,
+ 0xbcf0,0x0324 ,0xbd50,0x02c0 ,0xbdb0,0x025b ,
+ 0xbe11,0x01f7 ,0xbe73,0x0192 ,0xbed5,0x012e ,
+ 0xbf38,0x00c9 ,0xbf9c,0x0065 };
+
+
+extern const int s_Q14R_8;
+const int s_Q14R_8 = 1024;
+extern const unsigned short t_Q14R_8[2032];
+const unsigned short t_Q14R_8[2032] = {
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+ 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+ 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+ 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+ 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+ 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+ 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
+ 0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
+ 0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
+ 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+ 0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
+ 0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
+ 0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
+ 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+ 0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
+ 0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
+ 0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
+ 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+ 0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
+ 0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
+ 0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
+ 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+ 0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
+ 0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
+ 0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
+ 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+ 0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
+ 0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
+ 0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
+ 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+ 0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
+ 0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
+ 0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
+ 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+ 0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
+ 0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
+ 0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
+ 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+ 0x3ffb,0x0192 ,0x3fff,0x00c9 ,0x3ff5,0x025b ,
+ 0x3fec,0x0324 ,0x3ffb,0x0192 ,0x3fd4,0x04b5 ,
+ 0x3fd4,0x04b5 ,0x3ff5,0x025b ,0x3f9c,0x070e ,
+ 0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
+ 0x3f85,0x07d6 ,0x3fe1,0x03ed ,0x3eeb,0x0bb7 ,
+ 0x3f4f,0x0964 ,0x3fd4,0x04b5 ,0x3e72,0x0e06 ,
+ 0x3f0f,0x0af1 ,0x3fc4,0x057e ,0x3de3,0x1050 ,
+ 0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
+ 0x3e72,0x0e06 ,0x3f9c,0x070e ,0x3c85,0x14d2 ,
+ 0x3e15,0x0f8d ,0x3f85,0x07d6 ,0x3bb6,0x1709 ,
+ 0x3daf,0x1112 ,0x3f6b,0x089d ,0x3ad3,0x1937 ,
+ 0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
+ 0x3cc5,0x1413 ,0x3f30,0x0a2b ,0x38cf,0x1d79 ,
+ 0x3c42,0x1590 ,0x3f0f,0x0af1 ,0x37b0,0x1f8c ,
+ 0x3bb6,0x1709 ,0x3eeb,0x0bb7 ,0x367d,0x2193 ,
+ 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+ 0x3a82,0x19ef ,0x3e9d,0x0d41 ,0x33df,0x257e ,
+ 0x39db,0x1b5d ,0x3e72,0x0e06 ,0x3274,0x2760 ,
+ 0x392b,0x1cc6 ,0x3e45,0x0eca ,0x30f9,0x2935 ,
+ 0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
+ 0x37b0,0x1f8c ,0x3de3,0x1050 ,0x2dcf,0x2cb2 ,
+ 0x36e5,0x20e7 ,0x3daf,0x1112 ,0x2c21,0x2e5a ,
+ 0x3612,0x223d ,0x3d78,0x11d3 ,0x2a65,0x2ff2 ,
+ 0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
+ 0x3453,0x24da ,0x3d03,0x1354 ,0x26c1,0x32ef ,
+ 0x3368,0x2620 ,0x3cc5,0x1413 ,0x24da,0x3453 ,
+ 0x3274,0x2760 ,0x3c85,0x14d2 ,0x22e7,0x35a5 ,
+ 0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
+ 0x3076,0x29ce ,0x3bfd,0x164c ,0x1edc,0x3812 ,
+ 0x2f6c,0x2afb ,0x3bb6,0x1709 ,0x1cc6,0x392b ,
+ 0x2e5a,0x2c21 ,0x3b6d,0x17c4 ,0x1aa7,0x3a30 ,
+ 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+ 0x2c21,0x2e5a ,0x3ad3,0x1937 ,0x164c,0x3bfd ,
+ 0x2afb,0x2f6c ,0x3a82,0x19ef ,0x1413,0x3cc5 ,
+ 0x29ce,0x3076 ,0x3a30,0x1aa7 ,0x11d3,0x3d78 ,
+ 0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
+ 0x2760,0x3274 ,0x3984,0x1c12 ,0x0d41,0x3e9d ,
+ 0x2620,0x3368 ,0x392b,0x1cc6 ,0x0af1,0x3f0f ,
+ 0x24da,0x3453 ,0x38cf,0x1d79 ,0x089d,0x3f6b ,
+ 0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
+ 0x223d,0x3612 ,0x3812,0x1edc ,0x03ed,0x3fe1 ,
+ 0x20e7,0x36e5 ,0x37b0,0x1f8c ,0x0192,0x3ffb ,
+ 0x1f8c,0x37b0 ,0x374b,0x203a ,0xff37,0x3fff ,
+ 0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
+ 0x1cc6,0x392b ,0x367d,0x2193 ,0xfa82,0x3fc4 ,
+ 0x1b5d,0x39db ,0x3612,0x223d ,0xf82a,0x3f85 ,
+ 0x19ef,0x3a82 ,0x35a5,0x22e7 ,0xf5d5,0x3f30 ,
+ 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+ 0x1709,0x3bb6 ,0x34c6,0x2435 ,0xf136,0x3e45 ,
+ 0x1590,0x3c42 ,0x3453,0x24da ,0xeeee,0x3daf ,
+ 0x1413,0x3cc5 ,0x33df,0x257e ,0xecac,0x3d03 ,
+ 0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
+ 0x1112,0x3daf ,0x32ef,0x26c1 ,0xe83c,0x3b6d ,
+ 0x0f8d,0x3e15 ,0x3274,0x2760 ,0xe611,0x3a82 ,
+ 0x0e06,0x3e72 ,0x31f8,0x27fe ,0xe3ee,0x3984 ,
+ 0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
+ 0x0af1,0x3f0f ,0x30f9,0x2935 ,0xdfc6,0x374b ,
+ 0x0964,0x3f4f ,0x3076,0x29ce ,0xddc3,0x3612 ,
+ 0x07d6,0x3f85 ,0x2ff2,0x2a65 ,0xdbcb,0x34c6 ,
+ 0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
+ 0x04b5,0x3fd4 ,0x2ee4,0x2b8f ,0xd802,0x31f8 ,
+ 0x0324,0x3fec ,0x2e5a,0x2c21 ,0xd632,0x3076 ,
+ 0x0192,0x3ffb ,0x2dcf,0x2cb2 ,0xd471,0x2ee4 ,
+ 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+ 0xfe6e,0x3ffb ,0x2cb2,0x2dcf ,0xd11c,0x2b8f ,
+ 0xfcdc,0x3fec ,0x2c21,0x2e5a ,0xcf8a,0x29ce ,
+ 0xfb4b,0x3fd4 ,0x2b8f,0x2ee4 ,0xce08,0x27fe ,
+ 0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
+ 0xf82a,0x3f85 ,0x2a65,0x2ff2 ,0xcb3a,0x2435 ,
+ 0xf69c,0x3f4f ,0x29ce,0x3076 ,0xc9ee,0x223d ,
+ 0xf50f,0x3f0f ,0x2935,0x30f9 ,0xc8b5,0x203a ,
+ 0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
+ 0xf1fa,0x3e72 ,0x27fe,0x31f8 ,0xc67c,0x1c12 ,
+ 0xf073,0x3e15 ,0x2760,0x3274 ,0xc57e,0x19ef ,
+ 0xeeee,0x3daf ,0x26c1,0x32ef ,0xc493,0x17c4 ,
+ 0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
+ 0xebed,0x3cc5 ,0x257e,0x33df ,0xc2fd,0x1354 ,
+ 0xea70,0x3c42 ,0x24da,0x3453 ,0xc251,0x1112 ,
+ 0xe8f7,0x3bb6 ,0x2435,0x34c6 ,0xc1bb,0x0eca ,
+ 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+ 0xe611,0x3a82 ,0x22e7,0x35a5 ,0xc0d0,0x0a2b ,
+ 0xe4a3,0x39db ,0x223d,0x3612 ,0xc07b,0x07d6 ,
+ 0xe33a,0x392b ,0x2193,0x367d ,0xc03c,0x057e ,
+ 0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
+ 0xe074,0x37b0 ,0x203a,0x374b ,0xc001,0x00c9 ,
+ 0xdf19,0x36e5 ,0x1f8c,0x37b0 ,0xc005,0xfe6e ,
+ 0xddc3,0x3612 ,0x1edc,0x3812 ,0xc01f,0xfc13 ,
+ 0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
+ 0xdb26,0x3453 ,0x1d79,0x38cf ,0xc095,0xf763 ,
+ 0xd9e0,0x3368 ,0x1cc6,0x392b ,0xc0f1,0xf50f ,
+ 0xd8a0,0x3274 ,0x1c12,0x3984 ,0xc163,0xf2bf ,
+ 0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
+ 0xd632,0x3076 ,0x1aa7,0x3a30 ,0xc288,0xee2d ,
+ 0xd505,0x2f6c ,0x19ef,0x3a82 ,0xc33b,0xebed ,
+ 0xd3df,0x2e5a ,0x1937,0x3ad3 ,0xc403,0xe9b4 ,
+ 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+ 0xd1a6,0x2c21 ,0x17c4,0x3b6d ,0xc5d0,0xe559 ,
+ 0xd094,0x2afb ,0x1709,0x3bb6 ,0xc6d5,0xe33a ,
+ 0xcf8a,0x29ce ,0x164c,0x3bfd ,0xc7ee,0xe124 ,
+ 0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
+ 0xcd8c,0x2760 ,0x14d2,0x3c85 ,0xca5b,0xdd19 ,
+ 0xcc98,0x2620 ,0x1413,0x3cc5 ,0xcbad,0xdb26 ,
+ 0xcbad,0x24da ,0x1354,0x3d03 ,0xcd11,0xd93f ,
+ 0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
+ 0xc9ee,0x223d ,0x11d3,0x3d78 ,0xd00e,0xd59b ,
+ 0xc91b,0x20e7 ,0x1112,0x3daf ,0xd1a6,0xd3df ,
+ 0xc850,0x1f8c ,0x1050,0x3de3 ,0xd34e,0xd231 ,
+ 0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
+ 0xc6d5,0x1cc6 ,0x0eca,0x3e45 ,0xd6cb,0xcf07 ,
+ 0xc625,0x1b5d ,0x0e06,0x3e72 ,0xd8a0,0xcd8c ,
+ 0xc57e,0x19ef ,0x0d41,0x3e9d ,0xda82,0xcc21 ,
+ 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+ 0xc44a,0x1709 ,0x0bb7,0x3eeb ,0xde6d,0xc983 ,
+ 0xc3be,0x1590 ,0x0af1,0x3f0f ,0xe074,0xc850 ,
+ 0xc33b,0x1413 ,0x0a2b,0x3f30 ,0xe287,0xc731 ,
+ 0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
+ 0xc251,0x1112 ,0x089d,0x3f6b ,0xe6c9,0xc52d ,
+ 0xc1eb,0x0f8d ,0x07d6,0x3f85 ,0xe8f7,0xc44a ,
+ 0xc18e,0x0e06 ,0x070e,0x3f9c ,0xeb2e,0xc37b ,
+ 0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
+ 0xc0f1,0x0af1 ,0x057e,0x3fc4 ,0xefb0,0xc21d ,
+ 0xc0b1,0x0964 ,0x04b5,0x3fd4 ,0xf1fa,0xc18e ,
+ 0xc07b,0x07d6 ,0x03ed,0x3fe1 ,0xf449,0xc115 ,
+ 0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
+ 0xc02c,0x04b5 ,0x025b,0x3ff5 ,0xf8f2,0xc064 ,
+ 0xc014,0x0324 ,0x0192,0x3ffb ,0xfb4b,0xc02c ,
+ 0xc005,0x0192 ,0x00c9,0x3fff ,0xfda5,0xc00b ,
+ 0x4000,0x0000 ,0x4000,0x0065 ,0x3fff,0x00c9 ,
+ 0x3ffd,0x012e ,0x3ffb,0x0192 ,0x3ff8,0x01f7 ,
+ 0x3ff5,0x025b ,0x3ff1,0x02c0 ,0x3fec,0x0324 ,
+ 0x3fe7,0x0388 ,0x3fe1,0x03ed ,0x3fdb,0x0451 ,
+ 0x3fd4,0x04b5 ,0x3fcc,0x051a ,0x3fc4,0x057e ,
+ 0x3fbb,0x05e2 ,0x3fb1,0x0646 ,0x3fa7,0x06aa ,
+ 0x3f9c,0x070e ,0x3f91,0x0772 ,0x3f85,0x07d6 ,
+ 0x3f78,0x0839 ,0x3f6b,0x089d ,0x3f5d,0x0901 ,
+ 0x3f4f,0x0964 ,0x3f40,0x09c7 ,0x3f30,0x0a2b ,
+ 0x3f20,0x0a8e ,0x3f0f,0x0af1 ,0x3efd,0x0b54 ,
+ 0x3eeb,0x0bb7 ,0x3ed8,0x0c1a ,0x3ec5,0x0c7c ,
+ 0x3eb1,0x0cdf ,0x3e9d,0x0d41 ,0x3e88,0x0da4 ,
+ 0x3e72,0x0e06 ,0x3e5c,0x0e68 ,0x3e45,0x0eca ,
+ 0x3e2d,0x0f2b ,0x3e15,0x0f8d ,0x3dfc,0x0fee ,
+ 0x3de3,0x1050 ,0x3dc9,0x10b1 ,0x3daf,0x1112 ,
+ 0x3d93,0x1173 ,0x3d78,0x11d3 ,0x3d5b,0x1234 ,
+ 0x3d3f,0x1294 ,0x3d21,0x12f4 ,0x3d03,0x1354 ,
+ 0x3ce4,0x13b4 ,0x3cc5,0x1413 ,0x3ca5,0x1473 ,
+ 0x3c85,0x14d2 ,0x3c64,0x1531 ,0x3c42,0x1590 ,
+ 0x3c20,0x15ee ,0x3bfd,0x164c ,0x3bda,0x16ab ,
+ 0x3bb6,0x1709 ,0x3b92,0x1766 ,0x3b6d,0x17c4 ,
+ 0x3b47,0x1821 ,0x3b21,0x187e ,0x3afa,0x18db ,
+ 0x3ad3,0x1937 ,0x3aab,0x1993 ,0x3a82,0x19ef ,
+ 0x3a59,0x1a4b ,0x3a30,0x1aa7 ,0x3a06,0x1b02 ,
+ 0x39db,0x1b5d ,0x39b0,0x1bb8 ,0x3984,0x1c12 ,
+ 0x3958,0x1c6c ,0x392b,0x1cc6 ,0x38fd,0x1d20 ,
+ 0x38cf,0x1d79 ,0x38a1,0x1dd3 ,0x3871,0x1e2b ,
+ 0x3842,0x1e84 ,0x3812,0x1edc ,0x37e1,0x1f34 ,
+ 0x37b0,0x1f8c ,0x377e,0x1fe3 ,0x374b,0x203a ,
+ 0x3718,0x2091 ,0x36e5,0x20e7 ,0x36b1,0x213d ,
+ 0x367d,0x2193 ,0x3648,0x21e8 ,0x3612,0x223d ,
+ 0x35dc,0x2292 ,0x35a5,0x22e7 ,0x356e,0x233b ,
+ 0x3537,0x238e ,0x34ff,0x23e2 ,0x34c6,0x2435 ,
+ 0x348d,0x2488 ,0x3453,0x24da ,0x3419,0x252c ,
+ 0x33df,0x257e ,0x33a3,0x25cf ,0x3368,0x2620 ,
+ 0x332c,0x2671 ,0x32ef,0x26c1 ,0x32b2,0x2711 ,
+ 0x3274,0x2760 ,0x3236,0x27af ,0x31f8,0x27fe ,
+ 0x31b9,0x284c ,0x3179,0x289a ,0x3139,0x28e7 ,
+ 0x30f9,0x2935 ,0x30b8,0x2981 ,0x3076,0x29ce ,
+ 0x3034,0x2a1a ,0x2ff2,0x2a65 ,0x2faf,0x2ab0 ,
+ 0x2f6c,0x2afb ,0x2f28,0x2b45 ,0x2ee4,0x2b8f ,
+ 0x2e9f,0x2bd8 ,0x2e5a,0x2c21 ,0x2e15,0x2c6a ,
+ 0x2dcf,0x2cb2 ,0x2d88,0x2cfa ,0x2d41,0x2d41 ,
+ 0x2cfa,0x2d88 ,0x2cb2,0x2dcf ,0x2c6a,0x2e15 ,
+ 0x2c21,0x2e5a ,0x2bd8,0x2e9f ,0x2b8f,0x2ee4 ,
+ 0x2b45,0x2f28 ,0x2afb,0x2f6c ,0x2ab0,0x2faf ,
+ 0x2a65,0x2ff2 ,0x2a1a,0x3034 ,0x29ce,0x3076 ,
+ 0x2981,0x30b8 ,0x2935,0x30f9 ,0x28e7,0x3139 ,
+ 0x289a,0x3179 ,0x284c,0x31b9 ,0x27fe,0x31f8 ,
+ 0x27af,0x3236 ,0x2760,0x3274 ,0x2711,0x32b2 ,
+ 0x26c1,0x32ef ,0x2671,0x332c ,0x2620,0x3368 ,
+ 0x25cf,0x33a3 ,0x257e,0x33df ,0x252c,0x3419 ,
+ 0x24da,0x3453 ,0x2488,0x348d ,0x2435,0x34c6 ,
+ 0x23e2,0x34ff ,0x238e,0x3537 ,0x233b,0x356e ,
+ 0x22e7,0x35a5 ,0x2292,0x35dc ,0x223d,0x3612 ,
+ 0x21e8,0x3648 ,0x2193,0x367d ,0x213d,0x36b1 ,
+ 0x20e7,0x36e5 ,0x2091,0x3718 ,0x203a,0x374b ,
+ 0x1fe3,0x377e ,0x1f8c,0x37b0 ,0x1f34,0x37e1 ,
+ 0x1edc,0x3812 ,0x1e84,0x3842 ,0x1e2b,0x3871 ,
+ 0x1dd3,0x38a1 ,0x1d79,0x38cf ,0x1d20,0x38fd ,
+ 0x1cc6,0x392b ,0x1c6c,0x3958 ,0x1c12,0x3984 ,
+ 0x1bb8,0x39b0 ,0x1b5d,0x39db ,0x1b02,0x3a06 ,
+ 0x1aa7,0x3a30 ,0x1a4b,0x3a59 ,0x19ef,0x3a82 ,
+ 0x1993,0x3aab ,0x1937,0x3ad3 ,0x18db,0x3afa ,
+ 0x187e,0x3b21 ,0x1821,0x3b47 ,0x17c4,0x3b6d ,
+ 0x1766,0x3b92 ,0x1709,0x3bb6 ,0x16ab,0x3bda ,
+ 0x164c,0x3bfd ,0x15ee,0x3c20 ,0x1590,0x3c42 ,
+ 0x1531,0x3c64 ,0x14d2,0x3c85 ,0x1473,0x3ca5 ,
+ 0x1413,0x3cc5 ,0x13b4,0x3ce4 ,0x1354,0x3d03 ,
+ 0x12f4,0x3d21 ,0x1294,0x3d3f ,0x1234,0x3d5b ,
+ 0x11d3,0x3d78 ,0x1173,0x3d93 ,0x1112,0x3daf ,
+ 0x10b1,0x3dc9 ,0x1050,0x3de3 ,0x0fee,0x3dfc ,
+ 0x0f8d,0x3e15 ,0x0f2b,0x3e2d ,0x0eca,0x3e45 ,
+ 0x0e68,0x3e5c ,0x0e06,0x3e72 ,0x0da4,0x3e88 ,
+ 0x0d41,0x3e9d ,0x0cdf,0x3eb1 ,0x0c7c,0x3ec5 ,
+ 0x0c1a,0x3ed8 ,0x0bb7,0x3eeb ,0x0b54,0x3efd ,
+ 0x0af1,0x3f0f ,0x0a8e,0x3f20 ,0x0a2b,0x3f30 ,
+ 0x09c7,0x3f40 ,0x0964,0x3f4f ,0x0901,0x3f5d ,
+ 0x089d,0x3f6b ,0x0839,0x3f78 ,0x07d6,0x3f85 ,
+ 0x0772,0x3f91 ,0x070e,0x3f9c ,0x06aa,0x3fa7 ,
+ 0x0646,0x3fb1 ,0x05e2,0x3fbb ,0x057e,0x3fc4 ,
+ 0x051a,0x3fcc ,0x04b5,0x3fd4 ,0x0451,0x3fdb ,
+ 0x03ed,0x3fe1 ,0x0388,0x3fe7 ,0x0324,0x3fec ,
+ 0x02c0,0x3ff1 ,0x025b,0x3ff5 ,0x01f7,0x3ff8 ,
+ 0x0192,0x3ffb ,0x012e,0x3ffd ,0x00c9,0x3fff ,
+ 0x0065,0x4000 ,0x0000,0x4000 ,0xff9b,0x4000 ,
+ 0xff37,0x3fff ,0xfed2,0x3ffd ,0xfe6e,0x3ffb ,
+ 0xfe09,0x3ff8 ,0xfda5,0x3ff5 ,0xfd40,0x3ff1 ,
+ 0xfcdc,0x3fec ,0xfc78,0x3fe7 ,0xfc13,0x3fe1 ,
+ 0xfbaf,0x3fdb ,0xfb4b,0x3fd4 ,0xfae6,0x3fcc ,
+ 0xfa82,0x3fc4 ,0xfa1e,0x3fbb ,0xf9ba,0x3fb1 ,
+ 0xf956,0x3fa7 ,0xf8f2,0x3f9c ,0xf88e,0x3f91 ,
+ 0xf82a,0x3f85 ,0xf7c7,0x3f78 ,0xf763,0x3f6b ,
+ 0xf6ff,0x3f5d ,0xf69c,0x3f4f ,0xf639,0x3f40 ,
+ 0xf5d5,0x3f30 ,0xf572,0x3f20 ,0xf50f,0x3f0f ,
+ 0xf4ac,0x3efd ,0xf449,0x3eeb ,0xf3e6,0x3ed8 ,
+ 0xf384,0x3ec5 ,0xf321,0x3eb1 ,0xf2bf,0x3e9d ,
+ 0xf25c,0x3e88 ,0xf1fa,0x3e72 ,0xf198,0x3e5c ,
+ 0xf136,0x3e45 ,0xf0d5,0x3e2d ,0xf073,0x3e15 ,
+ 0xf012,0x3dfc ,0xefb0,0x3de3 ,0xef4f,0x3dc9 ,
+ 0xeeee,0x3daf ,0xee8d,0x3d93 ,0xee2d,0x3d78 ,
+ 0xedcc,0x3d5b ,0xed6c,0x3d3f ,0xed0c,0x3d21 ,
+ 0xecac,0x3d03 ,0xec4c,0x3ce4 ,0xebed,0x3cc5 ,
+ 0xeb8d,0x3ca5 ,0xeb2e,0x3c85 ,0xeacf,0x3c64 ,
+ 0xea70,0x3c42 ,0xea12,0x3c20 ,0xe9b4,0x3bfd ,
+ 0xe955,0x3bda ,0xe8f7,0x3bb6 ,0xe89a,0x3b92 ,
+ 0xe83c,0x3b6d ,0xe7df,0x3b47 ,0xe782,0x3b21 ,
+ 0xe725,0x3afa ,0xe6c9,0x3ad3 ,0xe66d,0x3aab ,
+ 0xe611,0x3a82 ,0xe5b5,0x3a59 ,0xe559,0x3a30 ,
+ 0xe4fe,0x3a06 ,0xe4a3,0x39db ,0xe448,0x39b0 ,
+ 0xe3ee,0x3984 ,0xe394,0x3958 ,0xe33a,0x392b ,
+ 0xe2e0,0x38fd ,0xe287,0x38cf ,0xe22d,0x38a1 ,
+ 0xe1d5,0x3871 ,0xe17c,0x3842 ,0xe124,0x3812 ,
+ 0xe0cc,0x37e1 ,0xe074,0x37b0 ,0xe01d,0x377e ,
+ 0xdfc6,0x374b ,0xdf6f,0x3718 ,0xdf19,0x36e5 ,
+ 0xdec3,0x36b1 ,0xde6d,0x367d ,0xde18,0x3648 ,
+ 0xddc3,0x3612 ,0xdd6e,0x35dc ,0xdd19,0x35a5 ,
+ 0xdcc5,0x356e ,0xdc72,0x3537 ,0xdc1e,0x34ff ,
+ 0xdbcb,0x34c6 ,0xdb78,0x348d ,0xdb26,0x3453 ,
+ 0xdad4,0x3419 ,0xda82,0x33df ,0xda31,0x33a3 ,
+ 0xd9e0,0x3368 ,0xd98f,0x332c ,0xd93f,0x32ef ,
+ 0xd8ef,0x32b2 ,0xd8a0,0x3274 ,0xd851,0x3236 ,
+ 0xd802,0x31f8 ,0xd7b4,0x31b9 ,0xd766,0x3179 ,
+ 0xd719,0x3139 ,0xd6cb,0x30f9 ,0xd67f,0x30b8 ,
+ 0xd632,0x3076 ,0xd5e6,0x3034 ,0xd59b,0x2ff2 ,
+ 0xd550,0x2faf ,0xd505,0x2f6c ,0xd4bb,0x2f28 ,
+ 0xd471,0x2ee4 ,0xd428,0x2e9f ,0xd3df,0x2e5a ,
+ 0xd396,0x2e15 ,0xd34e,0x2dcf ,0xd306,0x2d88 ,
+ 0xd2bf,0x2d41 ,0xd278,0x2cfa ,0xd231,0x2cb2 ,
+ 0xd1eb,0x2c6a ,0xd1a6,0x2c21 ,0xd161,0x2bd8 ,
+ 0xd11c,0x2b8f ,0xd0d8,0x2b45 ,0xd094,0x2afb ,
+ 0xd051,0x2ab0 ,0xd00e,0x2a65 ,0xcfcc,0x2a1a ,
+ 0xcf8a,0x29ce ,0xcf48,0x2981 ,0xcf07,0x2935 ,
+ 0xcec7,0x28e7 ,0xce87,0x289a ,0xce47,0x284c ,
+ 0xce08,0x27fe ,0xcdca,0x27af ,0xcd8c,0x2760 ,
+ 0xcd4e,0x2711 ,0xcd11,0x26c1 ,0xccd4,0x2671 ,
+ 0xcc98,0x2620 ,0xcc5d,0x25cf ,0xcc21,0x257e ,
+ 0xcbe7,0x252c ,0xcbad,0x24da ,0xcb73,0x2488 ,
+ 0xcb3a,0x2435 ,0xcb01,0x23e2 ,0xcac9,0x238e ,
+ 0xca92,0x233b ,0xca5b,0x22e7 ,0xca24,0x2292 ,
+ 0xc9ee,0x223d ,0xc9b8,0x21e8 ,0xc983,0x2193 ,
+ 0xc94f,0x213d ,0xc91b,0x20e7 ,0xc8e8,0x2091 ,
+ 0xc8b5,0x203a ,0xc882,0x1fe3 ,0xc850,0x1f8c ,
+ 0xc81f,0x1f34 ,0xc7ee,0x1edc ,0xc7be,0x1e84 ,
+ 0xc78f,0x1e2b ,0xc75f,0x1dd3 ,0xc731,0x1d79 ,
+ 0xc703,0x1d20 ,0xc6d5,0x1cc6 ,0xc6a8,0x1c6c ,
+ 0xc67c,0x1c12 ,0xc650,0x1bb8 ,0xc625,0x1b5d ,
+ 0xc5fa,0x1b02 ,0xc5d0,0x1aa7 ,0xc5a7,0x1a4b ,
+ 0xc57e,0x19ef ,0xc555,0x1993 ,0xc52d,0x1937 ,
+ 0xc506,0x18db ,0xc4df,0x187e ,0xc4b9,0x1821 ,
+ 0xc493,0x17c4 ,0xc46e,0x1766 ,0xc44a,0x1709 ,
+ 0xc426,0x16ab ,0xc403,0x164c ,0xc3e0,0x15ee ,
+ 0xc3be,0x1590 ,0xc39c,0x1531 ,0xc37b,0x14d2 ,
+ 0xc35b,0x1473 ,0xc33b,0x1413 ,0xc31c,0x13b4 ,
+ 0xc2fd,0x1354 ,0xc2df,0x12f4 ,0xc2c1,0x1294 ,
+ 0xc2a5,0x1234 ,0xc288,0x11d3 ,0xc26d,0x1173 ,
+ 0xc251,0x1112 ,0xc237,0x10b1 ,0xc21d,0x1050 ,
+ 0xc204,0x0fee ,0xc1eb,0x0f8d ,0xc1d3,0x0f2b ,
+ 0xc1bb,0x0eca ,0xc1a4,0x0e68 ,0xc18e,0x0e06 ,
+ 0xc178,0x0da4 ,0xc163,0x0d41 ,0xc14f,0x0cdf ,
+ 0xc13b,0x0c7c ,0xc128,0x0c1a ,0xc115,0x0bb7 ,
+ 0xc103,0x0b54 ,0xc0f1,0x0af1 ,0xc0e0,0x0a8e ,
+ 0xc0d0,0x0a2b ,0xc0c0,0x09c7 ,0xc0b1,0x0964 ,
+ 0xc0a3,0x0901 ,0xc095,0x089d ,0xc088,0x0839 ,
+ 0xc07b,0x07d6 ,0xc06f,0x0772 ,0xc064,0x070e ,
+ 0xc059,0x06aa ,0xc04f,0x0646 ,0xc045,0x05e2 ,
+ 0xc03c,0x057e ,0xc034,0x051a ,0xc02c,0x04b5 ,
+ 0xc025,0x0451 ,0xc01f,0x03ed ,0xc019,0x0388 ,
+ 0xc014,0x0324 ,0xc00f,0x02c0 ,0xc00b,0x025b ,
+ 0xc008,0x01f7 ,0xc005,0x0192 ,0xc003,0x012e ,
+ 0xc001,0x00c9 ,0xc000,0x0065 };
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
new file mode 100644
index 0000000..13fbd9f
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
@@ -0,0 +1,27 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the Q14 radix-2 tables used in ARM9E optimization routines.
+ *
+ */
+
+extern const unsigned short t_Q14S_rad8[2];
+const unsigned short t_Q14S_rad8[2] = { 0x0000,0x2d41 };
+
+//extern const int t_Q30S_rad8[2];
+//const int t_Q30S_rad8[2] = { 0x00000000,0x2d413ccd };
+
+extern const unsigned short t_Q14R_rad8[2];
+const unsigned short t_Q14R_rad8[2] = { 0x2d41,0x2d41 };
+
+//extern const int t_Q30R_rad8[2];
+//const int t_Q30R_rad8[2] = { 0x2d413ccd,0x2d413ccd };
diff --git a/trunk/src/common_audio/vad/Android.mk b/trunk/src/common_audio/vad/Android.mk
new file mode 100644
index 0000000..b7be3f0
--- /dev/null
+++ b/trunk/src/common_audio/vad/Android.mk
@@ -0,0 +1,50 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_vad
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ webrtc_vad.c \
+ vad_core.c \
+ vad_filterbank.c \
+ vad_gmm.c \
+ vad_sp.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/vad/include/webrtc_vad.h b/trunk/src/common_audio/vad/include/webrtc_vad.h
new file mode 100644
index 0000000..3f9e402
--- /dev/null
+++ b/trunk/src/common_audio/vad/include/webrtc_vad.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the VAD API calls. Specific function calls are given below.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+
+#include "typedefs.h"
+
+typedef struct WebRtcVadInst VadInst;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/****************************************************************************
+ * WebRtcVad_AssignSize(...)
+ *
+ * This functions get the size needed for storing the instance for encoder
+ * and decoder, respectively
+ *
+ * Input/Output:
+ * - size_in_bytes : Pointer to integer where the size is returned
+ *
+ * Return value : 0
+ */
+WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes);
+
+/****************************************************************************
+ * WebRtcVad_Assign(...)
+ *
+ * This functions Assigns memory for the instances.
+ *
+ * Input:
+ * - vad_inst_addr : Address to where to assign memory
+ * Output:
+ * - vad_inst : Pointer to the instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr);
+
+// Creates an instance to the VAD structure.
+//
+// - handle [o] : Pointer to the VAD instance that should be created.
+//
+// returns : 0 - (OK), -1 - (Error)
+int WebRtcVad_Create(VadInst** handle);
+
+// Frees the dynamic memory of a specified VAD instance.
+//
+// - handle [i] : Pointer to VAD instance that should be freed.
+//
+// returns : 0 - (OK), -1 - (NULL pointer in)
+int WebRtcVad_Free(VadInst* handle);
+
+/****************************************************************************
+ * WebRtcVad_Init(...)
+ *
+ * This function initializes a VAD instance
+ *
+ * Input:
+ * - vad_inst : Instance that should be initialized
+ *
+ * Output:
+ * - vad_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcVad_Init(VadInst *vad_inst);
+
+/****************************************************************************
+ * WebRtcVad_set_mode(...)
+ *
+ * This function initializes a VAD instance
+ *
+ * Input:
+ * - vad_inst : VAD instance
+ * - mode : Aggressiveness setting (0, 1, 2, or 3)
+ *
+ * Output:
+ * - vad_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcVad_set_mode(VadInst *vad_inst, int mode);
+
+/****************************************************************************
+ * WebRtcVad_Process(...)
+ *
+ * This functions does a VAD for the inserted speech frame
+ *
+ * Input
+ * - vad_inst : VAD Instance. Needs to be initiated before call.
+ * - fs : sampling frequency (Hz): 8000, 16000, or 32000
+ * - speech_frame : Pointer to speech frame buffer
+ * - frame_length : Length of speech frame buffer in number of samples
+ *
+ * Output:
+ * - vad_inst : Updated VAD instance
+ *
+ * Return value : 1 - Active Voice
+ * 0 - Non-active Voice
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
+ WebRtc_Word16 fs,
+ WebRtc_Word16 *speech_frame,
+ WebRtc_Word16 frame_length);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
diff --git a/trunk/src/common_audio/vad/vad.gypi b/trunk/src/common_audio/vad/vad.gypi
new file mode 100644
index 0000000..f15e651
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad.gypi
@@ -0,0 +1,69 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'vad',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/webrtc_vad.h',
+ 'webrtc_vad.c',
+ 'vad_core.c',
+ 'vad_core.h',
+ 'vad_defines.h',
+ 'vad_filterbank.c',
+ 'vad_filterbank.h',
+ 'vad_gmm.c',
+ 'vad_gmm.h',
+ 'vad_sp.c',
+ 'vad_sp.h',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets' : [
+ {
+ 'target_name': 'vad_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'vad',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'vad_core_unittest.cc',
+ 'vad_filterbank_unittest.cc',
+ 'vad_gmm_unittest.cc',
+ 'vad_sp_unittest.cc',
+ 'vad_unittest.cc',
+ 'vad_unittest.h',
+ ],
+ }, # vad_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/vad/vad_core.c b/trunk/src/common_audio/vad/vad_core.c
new file mode 100644
index 0000000..2c6e6a9
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core.c
@@ -0,0 +1,663 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_core.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+#include "vad_filterbank.h"
+#include "vad_gmm.h"
+#include "vad_sp.h"
+
+// Spectrum Weighting
+static const WebRtc_Word16 kSpectrumWeight[6] = { 6, 8, 10, 12, 14, 16 };
+static const WebRtc_Word16 kNoiseUpdateConst = 655; // Q15
+static const WebRtc_Word16 kSpeechUpdateConst = 6554; // Q15
+static const WebRtc_Word16 kBackEta = 154; // Q8
+// Minimum difference between the two models, Q5
+static const WebRtc_Word16 kMinimumDifference[6] = {
+ 544, 544, 576, 576, 576, 576 };
+// Upper limit of mean value for speech model, Q7
+static const WebRtc_Word16 kMaximumSpeech[6] = {
+ 11392, 11392, 11520, 11520, 11520, 11520 };
+// Minimum value for mean value
+static const WebRtc_Word16 kMinimumMean[2] = { 640, 768 };
+// Upper limit of mean value for noise model, Q7
+static const WebRtc_Word16 kMaximumNoise[6] = {
+ 9216, 9088, 8960, 8832, 8704, 8576 };
+// Start values for the Gaussian models, Q7
+// Weights for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataWeights[12] = {
+ 34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
+// Weights for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataWeights[12] = {
+ 48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
+// Means for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataMeans[12] = {
+ 6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
+// Means for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataMeans[12] = {
+ 8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
+};
+// Stds for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataStds[12] = {
+ 378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
+// Stds for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataStds[12] = {
+ 555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
+
+// Constants used in GmmProbability().
+//
+// Maximum number of counted speech (VAD = 1) frames in a row.
+static const int16_t kMaxSpeechFrames = 6;
+// Minimum standard deviation for both speech and noise.
+static const int16_t kMinStd = 384;
+
+// Constants in WebRtcVad_InitCore().
+// Default aggressiveness mode.
+static const short kDefaultMode = 0;
+static const int kInitCheck = 42;
+
+// Calculates the probabilities for both speech and background noise using
+// Gaussian Mixture Models. A hypothesis-test is performed to decide which type
+// of signal is most probable.
+//
+// - inst [i/o] : Pointer to VAD instance
+// - feature_vector [i] : Feature vector = log10(energy in frequency band)
+// - total_power [i] : Total power in audio frame.
+// - frame_length [i] : Number of input samples
+//
+// - returns : the VAD decision (0 - noise, 1 - speech).
+static int16_t GmmProbability(VadInstT *inst, WebRtc_Word16 *feature_vector,
+ WebRtc_Word16 total_power, int frame_length)
+{
+ int n, k;
+ WebRtc_Word16 backval;
+ WebRtc_Word16 h0, h1;
+ WebRtc_Word16 ratvec, xval;
+ WebRtc_Word16 vadflag;
+ WebRtc_Word16 shifts0, shifts1;
+ WebRtc_Word16 tmp16, tmp16_1, tmp16_2;
+ WebRtc_Word16 diff, nr, pos;
+ WebRtc_Word16 nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
+ WebRtc_Word16 delt, ndelt;
+ WebRtc_Word16 maxspe, maxmu;
+ WebRtc_Word16 deltaN[NUM_TABLE_VALUES], deltaS[NUM_TABLE_VALUES];
+ WebRtc_Word16 ngprvec[NUM_TABLE_VALUES], sgprvec[NUM_TABLE_VALUES];
+ WebRtc_Word32 h0test, h1test;
+ WebRtc_Word32 tmp32_1, tmp32_2;
+ WebRtc_Word32 dotVal;
+ WebRtc_Word32 nmid, smid;
+ WebRtc_Word32 probn[NUM_MODELS], probs[NUM_MODELS];
+ WebRtc_Word16 *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr,
+ *sstd1ptr, *sstd2ptr;
+ WebRtc_Word16 overhead1, overhead2, individualTest, totalTest;
+
+ // Set the thresholds to different values based on frame length
+ if (frame_length == 80)
+ {
+ // 80 input samples
+ overhead1 = inst->over_hang_max_1[0];
+ overhead2 = inst->over_hang_max_2[0];
+ individualTest = inst->individual[0];
+ totalTest = inst->total[0];
+ } else if (frame_length == 160)
+ {
+ // 160 input samples
+ overhead1 = inst->over_hang_max_1[1];
+ overhead2 = inst->over_hang_max_2[1];
+ individualTest = inst->individual[1];
+ totalTest = inst->total[1];
+ } else
+ {
+ // 240 input samples
+ overhead1 = inst->over_hang_max_1[2];
+ overhead2 = inst->over_hang_max_2[2];
+ individualTest = inst->individual[2];
+ totalTest = inst->total[2];
+ }
+
+ if (total_power > MIN_ENERGY)
+ { // If signal present at all
+
+ // Set pointers to the gaussian parameters
+ nmean1ptr = &inst->noise_means[0];
+ nmean2ptr = &inst->noise_means[NUM_CHANNELS];
+ smean1ptr = &inst->speech_means[0];
+ smean2ptr = &inst->speech_means[NUM_CHANNELS];
+ nstd1ptr = &inst->noise_stds[0];
+ nstd2ptr = &inst->noise_stds[NUM_CHANNELS];
+ sstd1ptr = &inst->speech_stds[0];
+ sstd2ptr = &inst->speech_stds[NUM_CHANNELS];
+
+ vadflag = 0;
+ dotVal = 0;
+ for (n = 0; n < NUM_CHANNELS; n++)
+ { // For all channels
+
+ pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
+ xval = feature_vector[n];
+
+ // Probability for Noise, Q7 * Q20 = Q27
+ tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
+ &deltaN[pos]);
+ probn[0] = (WebRtc_Word32)(kNoiseDataWeights[n] * tmp32_1);
+ tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
+ &deltaN[pos + 1]);
+ probn[1] = (WebRtc_Word32)(kNoiseDataWeights[n + NUM_CHANNELS] * tmp32_1);
+ h0test = probn[0] + probn[1]; // Q27
+ h0 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
+
+ // Probability for Speech
+ tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
+ &deltaS[pos]);
+ probs[0] = (WebRtc_Word32)(kSpeechDataWeights[n] * tmp32_1);
+ tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
+ &deltaS[pos + 1]);
+ probs[1] = (WebRtc_Word32)(kSpeechDataWeights[n + NUM_CHANNELS] * tmp32_1);
+ h1test = probs[0] + probs[1]; // Q27
+ h1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
+
+ // Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
+ shifts0 = WebRtcSpl_NormW32(h0test);
+ shifts1 = WebRtcSpl_NormW32(h1test);
+
+ if ((h0test > 0) && (h1test > 0))
+ {
+ ratvec = shifts0 - shifts1;
+ } else if (h1test > 0)
+ {
+ ratvec = 31 - shifts1;
+ } else if (h0test > 0)
+ {
+ ratvec = shifts0 - 31;
+ } else
+ {
+ ratvec = 0;
+ }
+
+ // VAD decision with spectrum weighting
+ dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
+
+ // Individual channel test
+ if ((ratvec << 2) > individualTest)
+ {
+ vadflag = 1;
+ }
+
+ // Probabilities used when updating model
+ if (h0 > 0)
+ {
+ tmp32_1 = probn[0] & 0xFFFFF000; // Q27
+ tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
+ ngprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h0);
+ ngprvec[pos + 1] = 16384 - ngprvec[pos];
+ } else
+ {
+ ngprvec[pos] = 16384;
+ ngprvec[pos + 1] = 0;
+ }
+
+ // Probabilities used when updating model
+ if (h1 > 0)
+ {
+ tmp32_1 = probs[0] & 0xFFFFF000;
+ tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
+ sgprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h1);
+ sgprvec[pos + 1] = 16384 - sgprvec[pos];
+ } else
+ {
+ sgprvec[pos] = 0;
+ sgprvec[pos + 1] = 0;
+ }
+ }
+
+ // Overall test
+ if (dotVal >= totalTest)
+ {
+ vadflag |= 1;
+ }
+
+ // Set pointers to the means and standard deviations.
+ nmean1ptr = &inst->noise_means[0];
+ smean1ptr = &inst->speech_means[0];
+ nstd1ptr = &inst->noise_stds[0];
+ sstd1ptr = &inst->speech_stds[0];
+
+ maxspe = 12800;
+
+ // Update the model's parameters
+ for (n = 0; n < NUM_CHANNELS; n++)
+ {
+
+ pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
+
+ // Get min value in past which is used for long term correction
+ backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
+
+ // Compute the "global" mean, that is the sum of the two means weighted
+ nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
+ nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS],
+ *(nmean1ptr+NUM_CHANNELS));
+ tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
+
+ for (k = 0; k < NUM_MODELS; k++)
+ {
+
+ nr = pos + k;
+
+ nmean2ptr = nmean1ptr + k * NUM_CHANNELS;
+ smean2ptr = smean1ptr + k * NUM_CHANNELS;
+ nstd2ptr = nstd1ptr + k * NUM_CHANNELS;
+ sstd2ptr = sstd1ptr + k * NUM_CHANNELS;
+ nmk = *nmean2ptr;
+ smk = *smean2ptr;
+ nsk = *nstd2ptr;
+ ssk = *sstd2ptr;
+
+ // Update noise mean vector if the frame consists of noise only
+ nmk2 = nmk;
+ if (!vadflag)
+ {
+ // deltaN = (x-mu)/sigma^2
+ // ngprvec[k] = probn[k]/(probn[0] + probn[1])
+
+ delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
+ deltaN[nr], 11); // Q14*Q11
+ nmk2 = nmk + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
+ kNoiseUpdateConst,
+ 22); // Q7+(Q14*Q15>>22)
+ }
+
+ // Long term correction of the noise mean
+ ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
+ ndelt -= tmp16_1; // Q8 - Q8
+ nmk3 = nmk2 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
+ kBackEta,
+ 9); // Q7+(Q8*Q8)>>9
+
+ // Control that the noise mean does not drift to much
+ tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
+ if (nmk3 < tmp16)
+ nmk3 = tmp16;
+ tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
+ if (nmk3 > tmp16)
+ nmk3 = tmp16;
+ *nmean2ptr = nmk3;
+
+ if (vadflag)
+ {
+ // Update speech mean vector:
+ // deltaS = (x-mu)/sigma^2
+ // sgprvec[k] = probn[k]/(probn[0] + probn[1])
+
+ delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
+ deltaS[nr],
+ 11); // (Q14*Q11)>>11=Q14
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
+ kSpeechUpdateConst,
+ 21) + 1;
+ smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
+
+ // Control that the speech mean does not drift to much
+ maxmu = maxspe + 640;
+ if (smk2 < kMinimumMean[k])
+ smk2 = kMinimumMean[k];
+ if (smk2 > maxmu)
+ smk2 = maxmu;
+
+ *smean2ptr = smk2;
+
+ // (Q7>>3) = Q4
+ tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
+
+ tmp16 = feature_vector[n] - tmp16; // Q4
+ tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
+ tmp32_2 = tmp32_1 - (WebRtc_Word32)4096; // Q12
+ tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
+ tmp32_1 = (WebRtc_Word32)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
+
+ tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
+
+ // 0.1 * Q20 / Q7 = Q13
+ if (tmp32_2 > 0)
+ tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
+ else
+ {
+ tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
+ tmp16 = -tmp16;
+ }
+ // divide by 4 giving an update factor of 0.025
+ tmp16 += 128; // Rounding
+ ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
+ // Division with 8 plus Q7
+ if (ssk < kMinStd)
+ ssk = kMinStd;
+ *sstd2ptr = ssk;
+ } else
+ {
+ // Update GMM variance vectors
+ // deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
+ tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
+
+ // (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
+ tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
+ tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
+ tmp32_2 = (WebRtc_Word32)(tmp16 * tmp32_1);
+ tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
+ // Q20 * approx 0.001 (2^-10=0.0009766)
+
+ // Q20 / Q7 = Q13
+ tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
+ if (tmp32_1 > 0)
+ tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
+ else
+ {
+ tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
+ tmp16 = -tmp16;
+ }
+ tmp16 += 32; // Rounding
+ nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
+
+ if (nsk < kMinStd)
+ nsk = kMinStd;
+
+ *nstd2ptr = nsk;
+ }
+ }
+
+ // Separate models if they are too close - nmid in Q14
+ nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
+ nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS], *nmean2ptr);
+
+ // smid in Q14
+ smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
+ smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+NUM_CHANNELS], *smean2ptr);
+
+ // diff = "global" speech mean - "global" noise mean
+ diff = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 9);
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
+ diff -= tmp16;
+
+ if (diff < kMinimumDifference[n])
+ {
+
+ tmp16 = kMinimumDifference[n] - diff; // Q5
+
+ // tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
+ // tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
+ tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
+ tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
+
+ // First Gauss, speech model
+ tmp16 = tmp16_1 + *smean1ptr;
+ *smean1ptr = tmp16;
+ smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
+
+ // Second Gauss, speech model
+ tmp16 = tmp16_1 + *smean2ptr;
+ *smean2ptr = tmp16;
+ smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+NUM_CHANNELS]);
+
+ // First Gauss, noise model
+ tmp16 = *nmean1ptr - tmp16_2;
+ *nmean1ptr = tmp16;
+
+ nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
+
+ // Second Gauss, noise model
+ tmp16 = *nmean2ptr - tmp16_2;
+ *nmean2ptr = tmp16;
+ nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+NUM_CHANNELS]);
+ }
+
+ // Control that the speech & noise means do not drift to much
+ maxspe = kMaximumSpeech[n];
+ tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 7);
+ if (tmp16_2 > maxspe)
+ { // Upper limit of speech model
+ tmp16_2 -= maxspe;
+
+ *smean1ptr -= tmp16_2;
+ *smean2ptr -= tmp16_2;
+ }
+
+ tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
+ if (tmp16_2 > kMaximumNoise[n])
+ {
+ tmp16_2 -= kMaximumNoise[n];
+
+ *nmean1ptr -= tmp16_2;
+ *nmean2ptr -= tmp16_2;
+ }
+
+ nmean1ptr++;
+ smean1ptr++;
+ nstd1ptr++;
+ sstd1ptr++;
+ }
+ inst->frame_counter++;
+ } else
+ {
+ vadflag = 0;
+ }
+
+ // Hangover smoothing
+ if (!vadflag)
+ {
+ if (inst->over_hang > 0)
+ {
+ vadflag = 2 + inst->over_hang;
+ inst->over_hang = inst->over_hang - 1;
+ }
+ inst->num_of_speech = 0;
+ } else
+ {
+ inst->num_of_speech = inst->num_of_speech + 1;
+ if (inst->num_of_speech > kMaxSpeechFrames)
+ {
+ inst->num_of_speech = kMaxSpeechFrames;
+ inst->over_hang = overhead2;
+ } else
+ inst->over_hang = overhead1;
+ }
+ return vadflag;
+}
+
+// Initialize the VAD. Set aggressiveness mode to default value.
+int WebRtcVad_InitCore(VadInstT* self) {
+ int i;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ // Initialization of general struct variables.
+ self->vad = 1; // Speech active (=1).
+ self->frame_counter = 0;
+ self->over_hang = 0;
+ self->num_of_speech = 0;
+
+ // Initialization of downsampling filter state.
+ memset(self->downsampling_filter_states, 0,
+ sizeof(self->downsampling_filter_states));
+
+ // Read initial PDF parameters.
+ for (i = 0; i < NUM_TABLE_VALUES; i++) {
+ self->noise_means[i] = kNoiseDataMeans[i];
+ self->speech_means[i] = kSpeechDataMeans[i];
+ self->noise_stds[i] = kNoiseDataStds[i];
+ self->speech_stds[i] = kSpeechDataStds[i];
+ }
+
+ // Initialize Index and Minimum value vectors.
+ for (i = 0; i < 16 * NUM_CHANNELS; i++) {
+ self->low_value_vector[i] = 10000;
+ self->index_vector[i] = 0;
+ }
+
+ // Initialize splitting filter states.
+ memset(self->upper_state, 0, sizeof(self->upper_state));
+ memset(self->lower_state, 0, sizeof(self->lower_state));
+
+ // Initialize high pass filter states.
+ memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
+
+ // Initialize mean value memory, for WebRtcVad_FindMinimum().
+ for (i = 0; i < NUM_CHANNELS; i++) {
+ self->mean_value[i] = 1600;
+ }
+
+ // Set aggressiveness mode to default (=|kDefaultMode|).
+ if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
+ return -1;
+ }
+
+ self->init_flag = kInitCheck;
+
+ return 0;
+}
+
+// Set aggressiveness mode
+int WebRtcVad_set_mode_core(VadInstT *inst, int mode)
+{
+
+ if (mode == 0)
+ {
+ // Quality mode
+ inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
+ inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
+ inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
+ inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
+ inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
+ inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
+
+ inst->individual[0] = INDIVIDUAL_10MS_Q;
+ inst->individual[1] = INDIVIDUAL_20MS_Q;
+ inst->individual[2] = INDIVIDUAL_30MS_Q;
+
+ inst->total[0] = TOTAL_10MS_Q;
+ inst->total[1] = TOTAL_20MS_Q;
+ inst->total[2] = TOTAL_30MS_Q;
+ } else if (mode == 1)
+ {
+ // Low bitrate mode
+ inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
+ inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
+ inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
+ inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
+ inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
+ inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
+
+ inst->individual[0] = INDIVIDUAL_10MS_LBR;
+ inst->individual[1] = INDIVIDUAL_20MS_LBR;
+ inst->individual[2] = INDIVIDUAL_30MS_LBR;
+
+ inst->total[0] = TOTAL_10MS_LBR;
+ inst->total[1] = TOTAL_20MS_LBR;
+ inst->total[2] = TOTAL_30MS_LBR;
+ } else if (mode == 2)
+ {
+ // Aggressive mode
+ inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
+ inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
+ inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
+ inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
+ inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
+ inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
+
+ inst->individual[0] = INDIVIDUAL_10MS_AGG;
+ inst->individual[1] = INDIVIDUAL_20MS_AGG;
+ inst->individual[2] = INDIVIDUAL_30MS_AGG;
+
+ inst->total[0] = TOTAL_10MS_AGG;
+ inst->total[1] = TOTAL_20MS_AGG;
+ inst->total[2] = TOTAL_30MS_AGG;
+ } else if (mode == 3)
+ {
+ // Very aggressive mode
+ inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
+ inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
+ inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
+ inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
+ inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
+ inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
+
+ inst->individual[0] = INDIVIDUAL_10MS_VAG;
+ inst->individual[1] = INDIVIDUAL_20MS_VAG;
+ inst->individual[2] = INDIVIDUAL_30MS_VAG;
+
+ inst->total[0] = TOTAL_10MS_VAG;
+ inst->total[1] = TOTAL_20MS_VAG;
+ inst->total[2] = TOTAL_30MS_VAG;
+ } else
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// Calculate VAD decision by first extracting feature values and then calculate
+// probability for both speech and background noise.
+
+WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+ int frame_length)
+{
+ WebRtc_Word16 len, vad;
+ WebRtc_Word16 speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
+ WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+
+ // Downsample signal 32->16->8 before doing VAD
+ WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
+ frame_length);
+ len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+
+ WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
+ len = WEBRTC_SPL_RSHIFT_W16(len, 1);
+
+ // Do VAD on an 8 kHz signal
+ vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+ return vad;
+}
+
+WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+ int frame_length)
+{
+ WebRtc_Word16 len, vad;
+ WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+ // Wideband: Downsample signal before doing VAD
+ WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
+ frame_length);
+
+ len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+ vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+ return vad;
+}
+
+WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+ int frame_length)
+{
+ WebRtc_Word16 feature_vector[NUM_CHANNELS], total_power;
+
+ // Get power in the bands
+ total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
+ feature_vector);
+
+ // Make a VAD
+ inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
+
+ return inst->vad;
+}
diff --git a/trunk/src/common_audio/vad/vad_core.h b/trunk/src/common_audio/vad/vad_core.h
new file mode 100644
index 0000000..c82fbce
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core.h
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the descriptions of the core VAD calls.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
+
+#include "typedefs.h"
+#include "vad_defines.h"
+
+typedef struct VadInstT_
+{
+
+ WebRtc_Word16 vad;
+ WebRtc_Word32 downsampling_filter_states[4];
+ WebRtc_Word16 noise_means[NUM_TABLE_VALUES];
+ WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
+ WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
+ WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
+ // TODO(bjornv): Change to |frame_count|.
+ WebRtc_Word32 frame_counter;
+ WebRtc_Word16 over_hang; // Over Hang
+ WebRtc_Word16 num_of_speech;
+ // TODO(bjornv): Change to |age_vector|.
+ WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
+ WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
+ // TODO(bjornv): Change to |median|.
+ WebRtc_Word16 mean_value[NUM_CHANNELS];
+ WebRtc_Word16 upper_state[5];
+ WebRtc_Word16 lower_state[5];
+ WebRtc_Word16 hp_filter_state[4];
+ WebRtc_Word16 over_hang_max_1[3];
+ WebRtc_Word16 over_hang_max_2[3];
+ WebRtc_Word16 individual[3];
+ WebRtc_Word16 total[3];
+
+ int init_flag;
+
+} VadInstT;
+
+// Initializes the core VAD component. The default aggressiveness mode is
+// controlled by |kDefaultMode| in vad_core.c.
+//
+// - self [i/o] : Instance that should be initialized
+//
+// returns : 0 (OK), -1 (NULL pointer in or if the default mode can't be
+// set)
+int WebRtcVad_InitCore(VadInstT* self);
+
+/****************************************************************************
+ * WebRtcVad_set_mode_core(...)
+ *
+ * This function changes the VAD settings
+ *
+ * Input:
+ * - inst : VAD instance
+ * - mode : Aggressiveness degree
+ * 0 (High quality) - 3 (Highly aggressive)
+ *
+ * Output:
+ * - inst : Changed instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcVad_set_mode_core(VadInstT* inst, int mode);
+
+/****************************************************************************
+ * WebRtcVad_CalcVad32khz(...)
+ * WebRtcVad_CalcVad16khz(...)
+ * WebRtcVad_CalcVad8khz(...)
+ *
+ * Calculate probability for active speech and make VAD decision.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - speech_frame : Input speech frame
+ * - frame_length : Number of input samples
+ *
+ * Output:
+ * - inst : Updated filter states etc.
+ *
+ * Return value : VAD decision
+ * 0 - No active speech
+ * 1-6 - Active speech
+ */
+WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+ int frame_length);
+WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+ int frame_length);
+WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+ int frame_length);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/trunk/src/common_audio/vad/vad_core_unittest.cc b/trunk/src/common_audio/vad/vad_core_unittest.cc
new file mode 100644
index 0000000..141b796
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "vad_unittest.h"
+
+extern "C" {
+#include "vad_core.h"
+}
+
+namespace {
+
+TEST_F(VadTest, InitCore) {
+ // Test WebRtcVad_InitCore().
+ VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+ // NULL pointer test.
+ EXPECT_EQ(-1, WebRtcVad_InitCore(NULL));
+
+ // Verify return = 0 for non-NULL pointer.
+ EXPECT_EQ(0, WebRtcVad_InitCore(self));
+ // Verify init_flag is set.
+ EXPECT_EQ(42, self->init_flag);
+
+ free(self);
+}
+
+TEST_F(VadTest, set_mode_core) {
+ VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+ // TODO(bjornv): Add NULL pointer check if we take care of it in
+ // vad_core.c
+
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ // Test WebRtcVad_set_mode_core().
+ // Invalid modes should return -1.
+ EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, -1));
+ EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, 1000));
+ // Valid modes should return 0.
+ for (size_t j = 0; j < kModesSize; ++j) {
+ EXPECT_EQ(0, WebRtcVad_set_mode_core(self, kModes[j]));
+ }
+
+ free(self);
+}
+
+TEST_F(VadTest, CalcVad) {
+ VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+ int16_t speech[kMaxFrameLength];
+
+ // TODO(bjornv): Add NULL pointer check if we take care of it in
+ // vad_core.c
+
+ // Test WebRtcVad_CalcVadXXkhz()
+ // Verify that all zeros in gives VAD = 0 out.
+ memset(speech, 0, sizeof(speech));
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+ if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+ EXPECT_EQ(0, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+ }
+ if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+ EXPECT_EQ(0, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+ }
+ if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+ EXPECT_EQ(0, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+ }
+ }
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+ speech[i] = (i * i);
+ }
+ for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+ if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+ EXPECT_EQ(1, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+ }
+ if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+ EXPECT_EQ(1, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+ }
+ if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+ EXPECT_EQ(1, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+ }
+ }
+
+ free(self);
+}
+} // namespace
diff --git a/trunk/src/common_audio/vad/vad_defines.h b/trunk/src/common_audio/vad/vad_defines.h
new file mode 100644
index 0000000..5d1539d
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_defines.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the macros used in VAD.
+ */
+
+#ifndef WEBRTC_VAD_DEFINES_H_
+#define WEBRTC_VAD_DEFINES_H_
+
+#define NUM_CHANNELS 6 // Eight frequency bands
+#define NUM_MODELS 2 // Number of Gaussian models
+#define NUM_TABLE_VALUES NUM_CHANNELS * NUM_MODELS
+
+#define MIN_ENERGY 10
+#define ALPHA1 6553 // 0.2 in Q15
+#define ALPHA2 32439 // 0.99 in Q15
+// Mode 0, Quality thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_Q 24
+#define INDIVIDUAL_20MS_Q 21 // (log10(2)*66)<<2 ~=16
+#define INDIVIDUAL_30MS_Q 24
+
+#define TOTAL_10MS_Q 57
+#define TOTAL_20MS_Q 48
+#define TOTAL_30MS_Q 57
+
+#define OHMAX1_10MS_Q 8 // Max Overhang 1
+#define OHMAX2_10MS_Q 14 // Max Overhang 2
+#define OHMAX1_20MS_Q 4 // Max Overhang 1
+#define OHMAX2_20MS_Q 7 // Max Overhang 2
+#define OHMAX1_30MS_Q 3
+#define OHMAX2_30MS_Q 5
+
+// Mode 1, Low bitrate thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_LBR 37
+#define INDIVIDUAL_20MS_LBR 32
+#define INDIVIDUAL_30MS_LBR 37
+
+#define TOTAL_10MS_LBR 100
+#define TOTAL_20MS_LBR 80
+#define TOTAL_30MS_LBR 100
+
+#define OHMAX1_10MS_LBR 8 // Max Overhang 1
+#define OHMAX2_10MS_LBR 14 // Max Overhang 2
+#define OHMAX1_20MS_LBR 4
+#define OHMAX2_20MS_LBR 7
+
+#define OHMAX1_30MS_LBR 3
+#define OHMAX2_30MS_LBR 5
+
+// Mode 2, Very aggressive thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_AGG 82
+#define INDIVIDUAL_20MS_AGG 78
+#define INDIVIDUAL_30MS_AGG 82
+
+#define TOTAL_10MS_AGG 285 //580
+#define TOTAL_20MS_AGG 260
+#define TOTAL_30MS_AGG 285
+
+#define OHMAX1_10MS_AGG 6 // Max Overhang 1
+#define OHMAX2_10MS_AGG 9 // Max Overhang 2
+#define OHMAX1_20MS_AGG 3
+#define OHMAX2_20MS_AGG 5
+
+#define OHMAX1_30MS_AGG 2
+#define OHMAX2_30MS_AGG 3
+
+// Mode 3, Super aggressive thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_VAG 94
+#define INDIVIDUAL_20MS_VAG 94
+#define INDIVIDUAL_30MS_VAG 94
+
+#define TOTAL_10MS_VAG 1100 //1700
+#define TOTAL_20MS_VAG 1050
+#define TOTAL_30MS_VAG 1100
+
+#define OHMAX1_10MS_VAG 6 // Max Overhang 1
+#define OHMAX2_10MS_VAG 9 // Max Overhang 2
+#define OHMAX1_20MS_VAG 3
+#define OHMAX2_20MS_VAG 5
+
+#define OHMAX1_30MS_VAG 2
+#define OHMAX2_30MS_VAG 3
+
+#endif // WEBRTC_VAD_DEFINES_H_
diff --git a/trunk/src/common_audio/vad/vad_filterbank.c b/trunk/src/common_audio/vad/vad_filterbank.c
new file mode 100644
index 0000000..2f5db44
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_filterbank.c
@@ -0,0 +1,335 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_filterbank.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+
+// Constants used in LogOfEnergy().
+static const int16_t kLogConst = 24660; // 160*log10(2) in Q9.
+static const int16_t kLogEnergyIntPart = 14336; // 14 in Q10
+
+// Coefficients used by HighPassFilter, Q14.
+static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
+static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
+
+// Allpass filter coefficients, upper and lower, in Q15.
+// Upper: 0.64, Lower: 0.17
+static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
+
+// Adjustment for division with two in SplitFilter.
+static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
+
+// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
+// sampled at 500 Hz.
+//
+// - data_in [i] : Input audio data sampled at 500 Hz.
+// - data_length [i] : Length of input and output data.
+// - filter_state [i/o] : State of the filter.
+// - data_out [o] : Output audio data in the frequency interval
+// 80 - 250 Hz.
+static void HighPassFilter(const int16_t* data_in, int data_length,
+ int16_t* filter_state, int16_t* data_out) {
+ int i;
+ const int16_t* in_ptr = data_in;
+ int16_t* out_ptr = data_out;
+ int32_t tmp32 = 0;
+
+
+ // The sum of the absolute values of the impulse response:
+ // The zero/pole-filter has a max amplification of a single sample of: 1.4546
+ // Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
+ // The all-zero section has a max amplification of a single sample of: 1.6189
+ // Impulse response: 0.4047 -0.8094 0.4047 0 0 0
+ // The all-pole section has a max amplification of a single sample of: 1.9931
+ // Impulse response: 1.0000 0.4734 -0.1189 -0.2187 -0.0627 0.04532
+
+ for (i = 0; i < data_length; i++) {
+ // All-zero section (filter coefficients in Q14).
+ tmp32 = WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], *in_ptr);
+ tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
+ tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]);
+ filter_state[1] = filter_state[0];
+ filter_state[0] = *in_ptr++;
+
+ // All-pole section (filter coefficients in Q14).
+ tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]);
+ tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
+ filter_state[3] = filter_state[2];
+ filter_state[2] = (int16_t) (tmp32 >> 14);
+ *out_ptr++ = filter_state[2];
+ }
+}
+
+// All pass filtering of |data_in|, used before splitting the signal into two
+// frequency bands (low pass vs high pass).
+// Note that |data_in| and |data_out| can NOT correspond to the same address.
+//
+// - data_in [i] : Input audio signal given in Q0.
+// - data_length [i] : Length of input and output data.
+// - filter_coefficient [i] : Given in Q15.
+// - filter_state [i/o] : State of the filter given in Q(-1).
+// - data_out [o] : Output audio signal given in Q(-1).
+static void AllPassFilter(const int16_t* data_in, int data_length,
+ int16_t filter_coefficient, int16_t* filter_state,
+ int16_t* data_out) {
+ // The filter can only cause overflow (in the w16 output variable)
+ // if more than 4 consecutive input numbers are of maximum value and
+ // has the the same sign as the impulse responses first taps.
+ // First 6 taps of the impulse response:
+ // 0.6399 0.5905 -0.3779 0.2418 -0.1547 0.0990
+
+ int i;
+ int16_t tmp16 = 0;
+ int32_t tmp32 = 0;
+ int32_t state32 = ((int32_t) (*filter_state) << 16); // Q15
+
+ for (i = 0; i < data_length; i++) {
+ tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficient, *data_in);
+ tmp16 = (int16_t) (tmp32 >> 16); // Q(-1)
+ *data_out++ = tmp16;
+ state32 = (((int32_t) (*data_in)) << 14); // Q14
+ state32 -= WEBRTC_SPL_MUL_16_16(filter_coefficient, tmp16); // Q14
+ state32 <<= 1; // Q15.
+ data_in += 2;
+ }
+
+ *filter_state = (int16_t) (state32 >> 16); // Q(-1)
+}
+
+// Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to
+// an upper (high pass) part and a lower (low pass) part respectively.
+//
+// - data_in [i] : Input audio data to be split into two frequency bands.
+// - data_length [i] : Length of |data_in|.
+// - upper_state [i/o] : State of the upper filter, given in Q(-1).
+// - lower_state [i/o] : State of the lower filter, given in Q(-1).
+// - hp_data_out [o] : Output audio data of the upper half of the spectrum.
+// The length is |data_length| / 2.
+// - lp_data_out [o] : Output audio data of the lower half of the spectrum.
+// The length is |data_length| / 2.
+static void SplitFilter(const int16_t* data_in, int data_length,
+ int16_t* upper_state, int16_t* lower_state,
+ int16_t* hp_data_out, int16_t* lp_data_out) {
+ int i;
+ int half_length = data_length >> 1; // Downsampling by 2.
+ int16_t tmp_out;
+
+ // All-pass filtering upper branch.
+ AllPassFilter(&data_in[0], half_length, kAllPassCoefsQ15[0], upper_state,
+ hp_data_out);
+
+ // All-pass filtering lower branch.
+ AllPassFilter(&data_in[1], half_length, kAllPassCoefsQ15[1], lower_state,
+ lp_data_out);
+
+ // Make LP and HP signals.
+ for (i = 0; i < half_length; i++) {
+ tmp_out = *hp_data_out;
+ *hp_data_out++ -= *lp_data_out;
+ *lp_data_out++ += tmp_out;
+ }
+}
+
+// Calculates the energy of |data_in| in dB, and also updates an overall
+// |total_energy| if necessary.
+//
+// - data_in [i] : Input audio data for energy calculation.
+// - data_length [i] : Length of input data.
+// - offset [i] : Offset value added to |log_energy|.
+// - total_energy [i/o] : An external energy updated with the energy of
+// |data_in|.
+// NOTE: |total_energy| is only updated if
+// |total_energy| <= MIN_ENERGY.
+// - log_energy [o] : 10 * log10("energy of |data_in|") given in Q4.
+static void LogOfEnergy(const int16_t* data_in, int data_length,
+ int16_t offset, int16_t* total_energy,
+ int16_t* log_energy) {
+ // |tot_rshifts| accumulates the number of right shifts performed on |energy|.
+ int tot_rshifts = 0;
+ // The |energy| will be normalized to 15 bits. We use unsigned integer because
+ // we eventually will mask out the fractional part.
+ uint32_t energy = 0;
+
+ assert(data_in != NULL);
+ assert(data_length > 0);
+
+ energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
+ &tot_rshifts);
+
+ if (energy != 0) {
+ // By construction, normalizing to 15 bits is equivalent with 17 leading
+ // zeros of an unsigned 32 bit value.
+ int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
+ // In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
+ // (14 << 10), which is what we initialize |log2_energy| with. For a more
+ // detailed derivations, see below.
+ int16_t log2_energy = kLogEnergyIntPart;
+
+ tot_rshifts += normalizing_rshifts;
+ // Normalize |energy| to 15 bits.
+ // |tot_rshifts| is now the total number of right shifts performed on
+ // |energy| after normalization. This means that |energy| is in
+ // Q(-tot_rshifts).
+ if (normalizing_rshifts < 0) {
+ energy <<= -normalizing_rshifts;
+ } else {
+ energy >>= normalizing_rshifts;
+ }
+
+ // Calculate the energy of |data_in| in dB, in Q4.
+ //
+ // 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
+ // 160 * log10(|energy| * 2^|tot_rshifts|) =
+ // 160 * log10(2) * log2(|energy| * 2^|tot_rshifts|) =
+ // 160 * log10(2) * (log2(|energy|) + log2(2^|tot_rshifts|)) =
+ // (160 * log10(2)) * (log2(|energy|) + |tot_rshifts|) =
+ // |kLogConst| * (|log2_energy| + |tot_rshifts|)
+ //
+ // We know by construction that |energy| is normalized to 15 bits. Hence,
+ // |energy| = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
+ // Further, we'd like |log2_energy| in Q10
+ // log2(|energy|) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
+ // 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
+ // 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
+ // (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
+ // (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
+ //
+ // Note that frac_Q15 = (|energy| & 0x00003FFF)
+
+ // Calculate and add the fractional part to |log2_energy|.
+ log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
+
+ // |kLogConst| is in Q9, |log2_energy| in Q10 and |tot_rshifts| in Q0.
+ // Note that we in our derivation above have accounted for an output in Q4.
+ *log_energy = (int16_t) (WEBRTC_SPL_MUL_16_16_RSFT(
+ kLogConst, log2_energy, 19) +
+ WEBRTC_SPL_MUL_16_16_RSFT(tot_rshifts, kLogConst, 9));
+
+ if (*log_energy < 0) {
+ *log_energy = 0;
+ }
+ } else {
+ *log_energy = offset;
+ return;
+ }
+
+ *log_energy += offset;
+
+ // Update the approximate |total_energy| with the energy of |data_in|, if
+ // |total_energy| has not exceeded MIN_ENERGY. |total_energy| is used as an
+ // energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
+ if (*total_energy <= MIN_ENERGY) {
+ if (tot_rshifts >= 0) {
+ // We know by construction that the |energy| > MIN_ENERGY in Q0, so add an
+ // arbitrary value such that |total_energy| exceeds MIN_ENERGY.
+ *total_energy += MIN_ENERGY + 1;
+ } else {
+ // By construction |energy| is represented by 15 bits, hence any number of
+ // right shifted |energy| will fit in an int16_t. In addition, adding the
+ // value to |total_energy| is wrap around safe as long as
+ // MIN_ENERGY < 8192.
+ *total_energy += (int16_t) (energy >> -tot_rshifts); // Q0.
+ }
+ }
+}
+
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
+ int data_length, int16_t* features) {
+ int16_t total_energy = 0;
+ // We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
+ // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
+ // have at most 120 samples after the first split and at most 60 samples after
+ // the second split.
+ int16_t hp_120[120], lp_120[120];
+ int16_t hp_60[60], lp_60[60];
+ const int half_data_length = data_length >> 1;
+ int length = half_data_length; // |data_length| / 2, corresponds to
+ // bandwidth = 2000 Hz after downsampling.
+
+ // Initialize variables for the first SplitFilter().
+ int frequency_band = 0;
+ const int16_t* in_ptr = data_in; // [0 - 4000] Hz.
+ int16_t* hp_out_ptr = hp_120; // [2000 - 4000] Hz.
+ int16_t* lp_out_ptr = lp_120; // [0 - 2000] Hz.
+
+ assert(data_length >= 0);
+ assert(data_length <= 240);
+ assert(4 < NUM_CHANNELS - 1); // Checking maximum |frequency_band|.
+
+ // Split at 2000 Hz and downsample.
+ SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
+ &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+ // For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
+ frequency_band = 1;
+ in_ptr = hp_120; // [2000 - 4000] Hz.
+ hp_out_ptr = hp_60; // [3000 - 4000] Hz.
+ lp_out_ptr = lp_60; // [2000 - 3000] Hz.
+ SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+ &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+ // Energy in 3000 Hz - 4000 Hz.
+ length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
+
+ LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
+
+ // Energy in 2000 Hz - 3000 Hz.
+ LogOfEnergy(lp_60, length, kOffsetVector[4], &total_energy, &features[4]);
+
+ // For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
+ frequency_band = 2;
+ in_ptr = lp_120; // [0 - 2000] Hz.
+ hp_out_ptr = hp_60; // [1000 - 2000] Hz.
+ lp_out_ptr = lp_60; // [0 - 1000] Hz.
+ length = half_data_length; // |data_length| / 2 <=> bandwidth = 2000 Hz.
+ SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+ &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+ // Energy in 1000 Hz - 2000 Hz.
+ length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
+ LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
+
+ // For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
+ frequency_band = 3;
+ in_ptr = lp_60; // [0 - 1000] Hz.
+ hp_out_ptr = hp_120; // [500 - 1000] Hz.
+ lp_out_ptr = lp_120; // [0 - 500] Hz.
+ SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+ &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+ // Energy in 500 Hz - 1000 Hz.
+ length >>= 1; // |data_length| / 8 <=> bandwidth = 500 Hz.
+ LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
+
+ // For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
+ frequency_band = 4;
+ in_ptr = lp_120; // [0 - 500] Hz.
+ hp_out_ptr = hp_60; // [250 - 500] Hz.
+ lp_out_ptr = lp_60; // [0 - 250] Hz.
+ SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+ &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+ // Energy in 250 Hz - 500 Hz.
+ length >>= 1; // |data_length| / 16 <=> bandwidth = 250 Hz.
+ LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
+
+ // Remove 0 Hz - 80 Hz, by high pass filtering the lower band.
+ HighPassFilter(lp_60, length, self->hp_filter_state, hp_120);
+
+ // Energy in 80 Hz - 250 Hz.
+ LogOfEnergy(hp_120, length, kOffsetVector[0], &total_energy, &features[0]);
+
+ return total_energy;
+}
diff --git a/trunk/src/common_audio/vad/vad_filterbank.h b/trunk/src/common_audio/vad/vad_filterbank.h
new file mode 100644
index 0000000..0c5c00c
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_filterbank.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes feature calculating functionality used in vad_core.c.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+
+#include "typedefs.h"
+#include "vad_core.h"
+
+// Takes |data_length| samples of |data_in| and calculates the logarithm of the
+// energy of each of the |NUM_CHANNELS| = 6 frequency bands used by the VAD:
+// 80 Hz - 250 Hz
+// 250 Hz - 500 Hz
+// 500 Hz - 1000 Hz
+// 1000 Hz - 2000 Hz
+// 2000 Hz - 3000 Hz
+// 3000 Hz - 4000 Hz
+//
+// The values are given in Q4 and written to |features|. Further, an approximate
+// overall energy is returned. The return value is used in
+// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
+// the threshold MIN_ENERGY.
+//
+// - self [i/o] : State information of the VAD.
+// - data_in [i] : Input audio data, for feature extraction.
+// - data_length [i] : Audio data size, in number of samples.
+// - features [o] : 10 * log10(energy in each frequency band), Q4.
+// - returns : Total energy of the signal (NOTE! This value is not
+// exact. It is only used in a comparison.)
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
+ int data_length, int16_t* features);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/trunk/src/common_audio/vad/vad_filterbank_unittest.cc b/trunk/src/common_audio/vad/vad_filterbank_unittest.cc
new file mode 100644
index 0000000..320fda9
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_filterbank_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "vad_unittest.h"
+
+extern "C" {
+#include "vad_core.h"
+#include "vad_defines.h"
+#include "vad_filterbank.h"
+}
+
+namespace {
+
+enum { kNumValidFrameLengths = 3 };
+
+TEST_F(VadTest, vad_filterbank) {
+ VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+ static const int16_t kReference[kNumValidFrameLengths] = { 48, 11, 11 };
+ static const int16_t kFeatures[kNumValidFrameLengths * NUM_CHANNELS] = {
+ 1213, 759, 587, 462, 434, 272,
+ 1479, 1385, 1291, 1200, 1103, 1099,
+ 1732, 1692, 1681, 1629, 1436, 1436
+ };
+ static const int16_t kOffsetVector[NUM_CHANNELS] = {
+ 368, 368, 272, 176, 176, 176 };
+ int16_t features[NUM_CHANNELS];
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ int16_t speech[kMaxFrameLength];
+ for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+ speech[i] = (i * i);
+ }
+
+ int frame_length_index = 0;
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+ if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+ EXPECT_EQ(kReference[frame_length_index],
+ WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+ features));
+ for (int k = 0; k < NUM_CHANNELS; ++k) {
+ EXPECT_EQ(kFeatures[k + frame_length_index * NUM_CHANNELS],
+ features[k]);
+ }
+ frame_length_index++;
+ }
+ }
+ EXPECT_EQ(kNumValidFrameLengths, frame_length_index);
+
+ // Verify that all zeros in gives kOffsetVector out.
+ memset(speech, 0, sizeof(speech));
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+ if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+ EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+ features));
+ for (int k = 0; k < NUM_CHANNELS; ++k) {
+ EXPECT_EQ(kOffsetVector[k], features[k]);
+ }
+ }
+ }
+
+ // Verify that all ones in gives kOffsetVector out. Any other constant input
+ // will have a small impact in the sub bands.
+ for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+ speech[i] = 1;
+ }
+ for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+ if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
+ features));
+ for (int k = 0; k < NUM_CHANNELS; ++k) {
+ EXPECT_EQ(kOffsetVector[k], features[k]);
+ }
+ }
+ }
+
+ free(self);
+}
+} // namespace
diff --git a/trunk/src/common_audio/vad/vad_gmm.c b/trunk/src/common_audio/vad/vad_gmm.c
new file mode 100644
index 0000000..20a703a
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_gmm.c
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_gmm.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+
+static const int32_t kCompVar = 22005;
+static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.
+
+// For a normal distribution, the probability of |input| is calculated and
+// returned (in Q20). The formula for normal distributed probability is
+//
+// 1 / s * exp(-(x - m)^2 / (2 * s^2))
+//
+// where the parameters are given in the following Q domains:
+// m = |mean| (Q7)
+// s = |std| (Q7)
+// x = |input| (Q4)
+// in addition to the probability we output |delta| (in Q11) used when updating
+// the noise/speech model.
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+ int16_t mean,
+ int16_t std,
+ int16_t* delta) {
+ int16_t tmp16, inv_std, inv_std2, exp_value = 0;
+ int32_t tmp32;
+
+ // Calculate |inv_std| = 1 / s, in Q10.
+ // 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
+ // Q-domain: Q17 / Q7 = Q10.
+ tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
+ inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
+
+ // Calculate |inv_std2| = 1 / s^2, in Q14.
+ tmp16 = (inv_std >> 2); // Q10 -> Q8.
+ // Q-domain: (Q8 * Q8) >> 2 = Q14.
+ inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2);
+ // TODO(bjornv): Investigate if changing to
+ // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6);
+ // gives better accuracy.
+
+ tmp16 = (input << 3); // Q4 -> Q7
+ tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
+
+ // To be used later, when updating noise/speech model.
+ // |delta| = (x - m) / s^2, in Q11.
+ // Q-domain: (Q14 * Q7) >> 10 = Q11.
+ *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inv_std2, tmp16, 10);
+
+ // Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
+ // division by two with one shift.
+ // Q-domain: (Q11 * Q7) >> 8 = Q10.
+ tmp32 = WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9);
+
+ // If the exponent is small enough to give a non-zero probability we calculate
+ // |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
+ // ~= exp2(-log2(exp(1)) * |tmp32|).
+ if (tmp32 < kCompVar) {
+ // Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
+ // Q-domain: (Q12 * Q10) >> 12 = Q10.
+ tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLog2Exp, (int16_t) tmp32, 12);
+ tmp16 = -tmp16;
+ exp_value = (0x0400 | (tmp16 & 0x03FF));
+ tmp16 ^= 0xFFFF;
+ tmp16 >>= 10;
+ tmp16 += 1;
+ // Get |exp_value| = exp(-|tmp32|) in Q10.
+ exp_value >>= tmp16;
+ }
+
+ // Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
+ // Q-domain: Q10 * Q10 = Q20.
+ return WEBRTC_SPL_MUL_16_16(inv_std, exp_value);
+}
diff --git a/trunk/src/common_audio/vad/vad_gmm.h b/trunk/src/common_audio/vad/vad_gmm.h
new file mode 100644
index 0000000..2333af7
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_gmm.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Gaussian probability calculations internally used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+
+#include "typedefs.h"
+
+// Calculates the probability for |input|, given that |input| comes from a
+// normal distribution with mean and standard deviation (|mean|, |std|).
+//
+// Inputs:
+// - input : input sample in Q4.
+// - mean : mean input in the statistical model, Q7.
+// - std : standard deviation, Q7.
+//
+// Output:
+//
+// - delta : input used when updating the model, Q11.
+// |delta| = (|input| - |mean|) / |std|^2.
+//
+// Return:
+// (probability for |input|) =
+// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+ int16_t mean,
+ int16_t std,
+ int16_t* delta);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
diff --git a/trunk/src/common_audio/vad/vad_gmm_unittest.cc b/trunk/src/common_audio/vad/vad_gmm_unittest.cc
new file mode 100644
index 0000000..205435a
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_gmm_unittest.cc
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "vad_unittest.h"
+
+extern "C" {
+#include "vad_gmm.h"
+}
+
+namespace {
+
+TEST_F(VadTest, vad_gmm) {
+ int16_t delta = 0;
+ // Input value at mean.
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(0, 0, 128, &delta));
+ EXPECT_EQ(0, delta);
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(16, 128, 128, &delta));
+ EXPECT_EQ(0, delta);
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(-16, -128, 128, &delta));
+ EXPECT_EQ(0, delta);
+
+ // Largest possible input to give non-zero probability.
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(59, 0, 128, &delta));
+ EXPECT_EQ(7552, delta);
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(75, 128, 128, &delta));
+ EXPECT_EQ(7552, delta);
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(-75, -128, 128, &delta));
+ EXPECT_EQ(-7552, delta);
+
+ // Too large input, should give zero probability.
+ EXPECT_EQ(0, WebRtcVad_GaussianProbability(105, 0, 128, &delta));
+ EXPECT_EQ(13440, delta);
+}
+} // namespace
diff --git a/trunk/src/common_audio/vad/vad_sp.c b/trunk/src/common_audio/vad/vad_sp.c
new file mode 100644
index 0000000..4fface3
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_sp.c
@@ -0,0 +1,181 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_sp.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+
+// Allpass filter coefficients, upper and lower, in Q13.
+// Upper: 0.64, Lower: 0.17.
+static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13
+
+// TODO(bjornv): Move this function to vad_filterbank.c.
+// Downsampling filter based on splitting filter and allpass functions.
+void WebRtcVad_Downsampling(int16_t* signal_in,
+ int16_t* signal_out,
+ int32_t* filter_state,
+ int in_length) {
+ int16_t tmp16_1 = 0, tmp16_2 = 0;
+ int32_t tmp32_1 = filter_state[0];
+ int32_t tmp32_2 = filter_state[1];
+ int n = 0;
+ int half_length = (in_length >> 1); // Downsampling by 2 gives half length.
+
+ // Filter coefficients in Q13, filter state in Q0.
+ for (n = 0; n < half_length; n++) {
+ // All-pass filtering upper branch.
+ tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
+ *signal_out = tmp16_1;
+ tmp32_1 = (int32_t) (*signal_in++) -
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
+
+ // All-pass filtering lower branch.
+ tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
+ *signal_out++ += tmp16_2;
+ tmp32_2 = (int32_t) (*signal_in++) -
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
+ }
+ // Store the filter states.
+ filter_state[0] = tmp32_1;
+ filter_state[1] = tmp32_2;
+}
+
+// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
+// smallest values the last 100 frames. Then calculates and returns the median
+// of the five smallest values.
+int16_t WebRtcVad_FindMinimum(VadInstT* self,
+ int16_t feature_value,
+ int channel) {
+ int i = 0, j = 0;
+ int position = -1;
+ // Offset to beginning of the 16 minimum values in memory.
+ int offset = (channel << 4);
+ int16_t current_median = 1600;
+ int16_t alpha = 0;
+ int32_t tmp32 = 0;
+ // Pointer to memory for the 16 minimum values and the age of each value of
+ // the |channel|.
+ int16_t* age_ptr = &self->index_vector[offset];
+ int16_t* value_ptr = &self->low_value_vector[offset];
+ int16_t *p1, *p2, *p3;
+
+ assert(channel < NUM_CHANNELS);
+
+ // Each value in |low_value_vector| is getting 1 loop older.
+ // Update age of each value in |age_ptr|, and remove old values.
+ for (i = 0; i < 16; i++) {
+ p3 = age_ptr + i;
+ if (*p3 != 100) {
+ *p3 += 1;
+ } else {
+ p1 = value_ptr + i + 1;
+ p2 = p3 + 1;
+ for (j = i; j < 16; j++) {
+ *(value_ptr + j) = *p1++;
+ *(age_ptr + j) = *p2++;
+ }
+ *(age_ptr + 15) = 101;
+ *(value_ptr + 15) = 10000;
+ }
+ }
+
+ // Check if |feature_value| is smaller than any of the values in
+ // |low_value_vector|. If so, find the |position| where to insert the new
+ // value.
+ if (feature_value < *(value_ptr + 7)) {
+ if (feature_value < *(value_ptr + 3)) {
+ if (feature_value < *(value_ptr + 1)) {
+ if (feature_value < *value_ptr) {
+ position = 0;
+ } else {
+ position = 1;
+ }
+ } else if (feature_value < *(value_ptr + 2)) {
+ position = 2;
+ } else {
+ position = 3;
+ }
+ } else if (feature_value < *(value_ptr + 5)) {
+ if (feature_value < *(value_ptr + 4)) {
+ position = 4;
+ } else {
+ position = 5;
+ }
+ } else if (feature_value < *(value_ptr + 6)) {
+ position = 6;
+ } else {
+ position = 7;
+ }
+ } else if (feature_value < *(value_ptr + 15)) {
+ if (feature_value < *(value_ptr + 11)) {
+ if (feature_value < *(value_ptr + 9)) {
+ if (feature_value < *(value_ptr + 8)) {
+ position = 8;
+ } else {
+ position = 9;
+ }
+ } else if (feature_value < *(value_ptr + 10)) {
+ position = 10;
+ } else {
+ position = 11;
+ }
+ } else if (feature_value < *(value_ptr + 13)) {
+ if (feature_value < *(value_ptr + 12)) {
+ position = 12;
+ } else {
+ position = 13;
+ }
+ } else if (feature_value < *(value_ptr + 14)) {
+ position = 14;
+ } else {
+ position = 15;
+ }
+ }
+
+ // If we have a new small value, put it in the correct position and shift
+ // larger values up.
+ if (position > -1) {
+ for (i = 15; i > position; i--) {
+ j = i - 1;
+ *(value_ptr + i) = *(value_ptr + j);
+ *(age_ptr + i) = *(age_ptr + j);
+ }
+ *(value_ptr + position) = feature_value;
+ *(age_ptr + position) = 1;
+ }
+
+ // Get |current_median|.
+ if (self->frame_counter > 2) {
+ current_median = *(value_ptr + 2);
+ } else if (self->frame_counter > 0) {
+ current_median = *value_ptr;
+ }
+
+ // Smooth the median value.
+ if (self->frame_counter > 0) {
+ if (current_median < self->mean_value[channel]) {
+ alpha = (int16_t) ALPHA1; // 0.2 in Q15.
+ } else {
+ alpha = (int16_t) ALPHA2; // 0.99 in Q15.
+ }
+ }
+ tmp32 = WEBRTC_SPL_MUL_16_16(alpha + 1, self->mean_value[channel]);
+ tmp32 += WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, current_median);
+ tmp32 += 16384;
+ self->mean_value[channel] = (int16_t) (tmp32 >> 15);
+
+ return self->mean_value[channel];
+}
diff --git a/trunk/src/common_audio/vad/vad_sp.h b/trunk/src/common_audio/vad/vad_sp.h
new file mode 100644
index 0000000..9e8b204
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_sp.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This file includes specific signal processing tools used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+
+#include "typedefs.h"
+#include "vad_core.h"
+
+// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
+//
+// Inputs:
+// - signal_in : Input signal.
+// - in_length : Length of input signal in samples.
+//
+// Input & Output:
+// - filter_state : Current filter states of the two all-pass filters. The
+// |filter_state| is updated after all samples have been
+// processed.
+//
+// Output:
+// - signal_out : Downsampled signal (of length |in_length| / 2).
+void WebRtcVad_Downsampling(int16_t* signal_in,
+ int16_t* signal_out,
+ int32_t* filter_state,
+ int in_length);
+
+// Updates and returns the smoothed feature minimum. As minimum we use the
+// median of the five smallest feature values in a 100 frames long window.
+// As long as |handle->frame_counter| is zero, that is, we haven't received any
+// "valid" data, FindMinimum() outputs the default value of 1600.
+//
+// Inputs:
+// - feature_value : New feature value to update with.
+// - channel : Channel number.
+//
+// Input & Output:
+// - handle : State information of the VAD.
+//
+// Returns:
+// : Smoothed minimum value for a moving window.
+int16_t WebRtcVad_FindMinimum(VadInstT* handle,
+ int16_t feature_value,
+ int channel);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
diff --git a/trunk/src/common_audio/vad/vad_sp_unittest.cc b/trunk/src/common_audio/vad/vad_sp_unittest.cc
new file mode 100644
index 0000000..03c844e
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_sp_unittest.cc
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "vad_unittest.h"
+
+extern "C" {
+#include "vad_core.h"
+#include "vad_defines.h"
+#include "vad_sp.h"
+}
+
+namespace {
+
+TEST_F(VadTest, vad_sp) {
+ VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+ int16_t zeros[kMaxFrameLength] = { 0 };
+ int32_t state[2] = { 0 };
+ int16_t data_in[kMaxFrameLength];
+ int16_t data_out[kMaxFrameLength];
+
+ // We expect the first value to be 1600 as long as |frame_counter| is zero,
+ // which is true for the first iteration.
+ static const int16_t kReferenceMin[32] = {
+ 1600, 720, 509, 512, 532, 552, 570, 588,
+ 606, 624, 642, 659, 675, 691, 707, 723,
+ 1600, 544, 502, 522, 542, 561, 579, 597,
+ 615, 633, 651, 667, 683, 699, 715, 731
+ };
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+ data_in[i] = (i * i);
+ }
+ // Input values all zeros, expect all zeros out.
+ WebRtcVad_Downsampling(zeros, data_out, state,
+ static_cast<int>(kMaxFrameLength));
+ EXPECT_EQ(0, state[0]);
+ EXPECT_EQ(0, state[1]);
+ for (int16_t i = 0; i < kMaxFrameLength / 2; ++i) {
+ EXPECT_EQ(0, data_out[i]);
+ }
+ // Make a simple non-zero data test.
+ WebRtcVad_Downsampling(data_in, data_out, state,
+ static_cast<int>(kMaxFrameLength));
+ EXPECT_EQ(207, state[0]);
+ EXPECT_EQ(2270, state[1]);
+
+ ASSERT_EQ(0, WebRtcVad_InitCore(self));
+ // TODO(bjornv): Replace this part of the test with taking values from an
+ // array and calculate the reference value here. Make sure the values are not
+ // ordered.
+ for (int16_t i = 0; i < 16; ++i) {
+ int16_t value = 500 * (i + 1);
+ for (int j = 0; j < NUM_CHANNELS; ++j) {
+ // Use values both above and below initialized value.
+ EXPECT_EQ(kReferenceMin[i], WebRtcVad_FindMinimum(self, value, j));
+ EXPECT_EQ(kReferenceMin[i + 16], WebRtcVad_FindMinimum(self, 12000, j));
+ }
+ self->frame_counter++;
+ }
+
+ free(self);
+}
+} // namespace
diff --git a/trunk/src/common_audio/vad/vad_unittest.cc b/trunk/src/common_audio/vad/vad_unittest.cc
new file mode 100644
index 0000000..60ee208
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_unittest.cc
@@ -0,0 +1,131 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_unittest.h"
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "webrtc_vad.h"
+
+VadTest::VadTest() {}
+
+void VadTest::SetUp() {}
+
+void VadTest::TearDown() {}
+
+// Returns true if the rate and frame length combination is valid.
+bool VadTest::ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) {
+ if (rate == 8000) {
+ if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
+ return true;
+ }
+ return false;
+ } else if (rate == 16000) {
+ if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
+ return true;
+ }
+ return false;
+ }
+ if (rate == 32000) {
+ if (frame_length == 320 || frame_length == 640 || frame_length == 960) {
+ return true;
+ }
+ return false;
+ }
+
+ return false;
+}
+
+namespace {
+
+TEST_F(VadTest, ApiTest) {
+ // This API test runs through the APIs for all possible valid and invalid
+ // combinations.
+
+ VadInst* handle = NULL;
+ int16_t zeros[kMaxFrameLength] = { 0 };
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ int16_t speech[kMaxFrameLength];
+ for (int16_t i = 0; i < kMaxFrameLength; i++) {
+ speech[i] = (i * i);
+ }
+
+ // Null instance tests
+ EXPECT_EQ(-1, WebRtcVad_Create(NULL));
+ EXPECT_EQ(-1, WebRtcVad_Init(NULL));
+ EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
+ EXPECT_EQ(-1, WebRtcVad_Free(NULL));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, kModes[0]));
+ EXPECT_EQ(-1, WebRtcVad_Process(NULL, kRates[0], speech, kFrameLengths[0]));
+
+ // WebRtcVad_AssignSize tests
+ int handle_size_bytes = 0;
+ EXPECT_EQ(0, WebRtcVad_AssignSize(&handle_size_bytes));
+ EXPECT_EQ(576, handle_size_bytes);
+
+ // WebRtcVad_Assign tests
+ void* tmp_handle = malloc(handle_size_bytes);
+ EXPECT_EQ(-1, WebRtcVad_Assign(&handle, NULL));
+ EXPECT_EQ(0, WebRtcVad_Assign(&handle, tmp_handle));
+ EXPECT_EQ(handle, tmp_handle);
+ free(tmp_handle);
+
+ // WebRtcVad_Create()
+ ASSERT_EQ(0, WebRtcVad_Create(&handle));
+
+ // Not initialized tests
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], speech, kFrameLengths[0]));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0]));
+
+ // WebRtcVad_Init() test
+ ASSERT_EQ(0, WebRtcVad_Init(handle));
+
+ // WebRtcVad_set_mode() invalid modes tests
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0] - 1));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[kModesSize - 1] + 1));
+
+ // WebRtcVad_Process() tests
+ // NULL speech pointer
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], NULL, kFrameLengths[0]));
+ // Invalid sampling rate
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, 9999, speech, kFrameLengths[0]));
+ // All zeros as input should work
+ EXPECT_EQ(0, WebRtcVad_Process(handle, kRates[0], zeros, kFrameLengths[0]));
+ for (size_t k = 0; k < kModesSize; k++) {
+ // Test valid modes
+ EXPECT_EQ(0, WebRtcVad_set_mode(handle, kModes[k]));
+ // Loop through sampling rate and frame length combinations
+ for (size_t i = 0; i < kRatesSize; i++) {
+ for (size_t j = 0; j < kFrameLengthsSize; j++) {
+ if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
+ EXPECT_EQ(1, WebRtcVad_Process(handle,
+ kRates[i],
+ speech,
+ kFrameLengths[j]));
+ } else {
+ EXPECT_EQ(-1, WebRtcVad_Process(handle,
+ kRates[i],
+ speech,
+ kFrameLengths[j]));
+ }
+ }
+ }
+ }
+
+ EXPECT_EQ(0, WebRtcVad_Free(handle));
+}
+
+// TODO(bjornv): Add a process test, run on file.
+
+} // namespace
diff --git a/trunk/src/common_audio/vad/vad_unittest.h b/trunk/src/common_audio/vad/vad_unittest.h
new file mode 100644
index 0000000..f68427c
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_unittest.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
+#define WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
+
+#include <stddef.h> // size_t
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+
+namespace {
+
+// Modes we support
+const int kModes[] = { 0, 1, 2, 3 };
+const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
+
+// Rates we support.
+const int16_t kRates[] = { 8000, 12000, 16000, 24000, 32000 };
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+
+// Frame lengths we support.
+const int16_t kMaxFrameLength = 960;
+const int16_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640,
+ kMaxFrameLength };
+const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
+
+} // namespace
+
+class VadTest : public ::testing::Test {
+ protected:
+ VadTest();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ // Returns true if the rate and frame length combination is valid.
+ bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length);
+};
+
+#endif // WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
diff --git a/trunk/src/common_audio/vad/webrtc_vad.c b/trunk/src/common_audio/vad/webrtc_vad.c
new file mode 100644
index 0000000..26941e6
--- /dev/null
+++ b/trunk/src/common_audio/vad/webrtc_vad.c
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes the VAD API calls. For a specific function call description,
+ * see webrtc_vad.h
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "webrtc_vad.h"
+#include "vad_core.h"
+
+static const int kInitCheck = 42;
+
+WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes)
+{
+ *size_in_bytes = sizeof(VadInstT) * 2 / sizeof(WebRtc_Word16);
+ return 0;
+}
+
+WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr)
+{
+
+ if (vad_inst == NULL)
+ {
+ return -1;
+ }
+
+ if (vad_inst_addr != NULL)
+ {
+ *vad_inst = (VadInst*)vad_inst_addr;
+ return 0;
+ } else
+ {
+ return -1;
+ }
+}
+
+int WebRtcVad_Create(VadInst** handle) {
+ VadInstT* self = NULL;
+
+ if (handle == NULL) {
+ return -1;
+ }
+
+ *handle = NULL;
+ self = (VadInstT*) malloc(sizeof(VadInstT));
+ *handle = (VadInst*) self;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->init_flag = 0;
+
+ return 0;
+}
+
+int WebRtcVad_Free(VadInst* handle) {
+ if (handle == NULL) {
+ return -1;
+ }
+
+ free(handle);
+
+ return 0;
+}
+
+int WebRtcVad_Init(VadInst* handle) {
+ // Initialize the core VAD component.
+ return WebRtcVad_InitCore((VadInstT*) handle);
+}
+
+int WebRtcVad_set_mode(VadInst *vad_inst, int mode)
+{
+ VadInstT* vad_ptr;
+
+ if (vad_inst == NULL)
+ {
+ return -1;
+ }
+
+ vad_ptr = (VadInstT*)vad_inst;
+ if (vad_ptr->init_flag != kInitCheck)
+ {
+ return -1;
+ }
+
+ return WebRtcVad_set_mode_core((VadInstT*)vad_inst, mode);
+}
+
+WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
+ WebRtc_Word16 fs,
+ WebRtc_Word16 *speech_frame,
+ WebRtc_Word16 frame_length)
+{
+ WebRtc_Word16 vad;
+ VadInstT* vad_ptr;
+
+ if (vad_inst == NULL)
+ {
+ return -1;
+ }
+
+ vad_ptr = (VadInstT*)vad_inst;
+ if (vad_ptr->init_flag != kInitCheck)
+ {
+ return -1;
+ }
+
+ if (speech_frame == NULL)
+ {
+ return -1;
+ }
+
+ if (fs == 32000)
+ {
+ if ((frame_length != 320) && (frame_length != 640) && (frame_length != 960))
+ {
+ return -1;
+ }
+ vad = WebRtcVad_CalcVad32khz((VadInstT*)vad_inst, speech_frame, frame_length);
+
+ } else if (fs == 16000)
+ {
+ if ((frame_length != 160) && (frame_length != 320) && (frame_length != 480))
+ {
+ return -1;
+ }
+ vad = WebRtcVad_CalcVad16khz((VadInstT*)vad_inst, speech_frame, frame_length);
+
+ } else if (fs == 8000)
+ {
+ if ((frame_length != 80) && (frame_length != 160) && (frame_length != 240))
+ {
+ return -1;
+ }
+ vad = WebRtcVad_CalcVad8khz((VadInstT*)vad_inst, speech_frame, frame_length);
+
+ } else
+ {
+ return -1; // Not a supported sampling frequency
+ }
+
+ if (vad > 0)
+ {
+ return 1;
+ } else if (vad == 0)
+ {
+ return 0;
+ } else
+ {
+ return -1;
+ }
+}
diff --git a/trunk/src/common_types.h b/trunk/src/common_types.h
new file mode 100644
index 0000000..9b3a0ce
--- /dev/null
+++ b/trunk/src/common_types.h
@@ -0,0 +1,573 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_TYPES_H
+#define WEBRTC_COMMON_TYPES_H
+
+#include "typedefs.h"
+
+#ifdef WEBRTC_EXPORT
+ #define WEBRTC_DLLEXPORT _declspec(dllexport)
+#elif WEBRTC_DLL
+ #define WEBRTC_DLLEXPORT _declspec(dllimport)
+#else
+ #define WEBRTC_DLLEXPORT
+#endif
+
+#ifndef NULL
+ #define NULL 0
+#endif
+
+#define RTP_PAYLOAD_NAME_SIZE 32
+
+namespace webrtc {
+
+class InStream
+{
+public:
+ virtual int Read(void *buf,int len) = 0;
+ virtual int Rewind() {return -1;}
+ virtual ~InStream() {}
+protected:
+ InStream() {}
+};
+
+class OutStream
+{
+public:
+ virtual bool Write(const void *buf,int len) = 0;
+ virtual int Rewind() {return -1;}
+ virtual ~OutStream() {}
+protected:
+ OutStream() {}
+};
+
+enum TraceModule
+{
+ // not a module, triggered from the engine code
+ kTraceVoice = 0x0001,
+ // not a module, triggered from the engine code
+ kTraceVideo = 0x0002,
+ // not a module, triggered from the utility code
+ kTraceUtility = 0x0003,
+ kTraceRtpRtcp = 0x0004,
+ kTraceTransport = 0x0005,
+ kTraceSrtp = 0x0006,
+ kTraceAudioCoding = 0x0007,
+ kTraceAudioMixerServer = 0x0008,
+ kTraceAudioMixerClient = 0x0009,
+ kTraceFile = 0x000a,
+ kTraceAudioProcessing = 0x000b,
+ kTraceVideoCoding = 0x0010,
+ kTraceVideoMixer = 0x0011,
+ kTraceAudioDevice = 0x0012,
+ kTraceVideoRenderer = 0x0014,
+ kTraceVideoCapture = 0x0015,
+ kTraceVideoPreocessing = 0x0016
+};
+
+enum TraceLevel
+{
+ kTraceNone = 0x0000, // no trace
+ kTraceStateInfo = 0x0001,
+ kTraceWarning = 0x0002,
+ kTraceError = 0x0004,
+ kTraceCritical = 0x0008,
+ kTraceApiCall = 0x0010,
+ kTraceDefault = 0x00ff,
+
+ kTraceModuleCall = 0x0020,
+ kTraceMemory = 0x0100, // memory info
+ kTraceTimer = 0x0200, // timing info
+ kTraceStream = 0x0400, // "continuous" stream of data
+
+ // used for debug purposes
+ kTraceDebug = 0x0800, // debug
+ kTraceInfo = 0x1000, // debug info
+
+ kTraceAll = 0xffff
+};
+
+// External Trace API
+class TraceCallback
+{
+public:
+ virtual void Print(const TraceLevel level,
+ const char *traceString,
+ const int length) = 0;
+protected:
+ virtual ~TraceCallback() {}
+ TraceCallback() {}
+};
+
+
+enum FileFormats
+{
+ kFileFormatWavFile = 1,
+ kFileFormatCompressedFile = 2,
+ kFileFormatAviFile = 3,
+ kFileFormatPreencodedFile = 4,
+ kFileFormatPcm16kHzFile = 7,
+ kFileFormatPcm8kHzFile = 8,
+ kFileFormatPcm32kHzFile = 9
+};
+
+
+enum ProcessingTypes
+{
+ kPlaybackPerChannel = 0,
+ kPlaybackAllChannelsMixed,
+ kRecordingPerChannel,
+ kRecordingAllChannelsMixed
+};
+
+// Encryption enums
+enum CipherTypes
+{
+ kCipherNull = 0,
+ kCipherAes128CounterMode = 1
+};
+
+enum AuthenticationTypes
+{
+ kAuthNull = 0,
+ kAuthHmacSha1 = 3
+};
+
+enum SecurityLevels
+{
+ kNoProtection = 0,
+ kEncryption = 1,
+ kAuthentication = 2,
+ kEncryptionAndAuthentication = 3
+};
+
+// Interface for encrypting and decrypting regular data and rtp/rtcp packets.
+// Implement this interface if you wish to provide an encryption scheme to
+// the voice or video engines.
+class Encryption
+{
+public:
+ // Encrypt the given data.
+ //
+ // Args:
+ // channel: The channel to encrypt data for.
+ // in_data: The data to encrypt. This data is bytes_in bytes long.
+ // out_data: The buffer to write the encrypted data to. You may write more
+ // bytes of encrypted data than what you got as input, up to a maximum
+ // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or
+ // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine.
+ // bytes_in: The number of bytes in the input buffer.
+ // bytes_out: The number of bytes written in out_data.
+ virtual void encrypt(
+ int channel,
+ unsigned char* in_data,
+ unsigned char* out_data,
+ int bytes_in,
+ int* bytes_out) = 0;
+
+ // Decrypts the given data. This should reverse the effects of encrypt().
+ //
+ // Args:
+ // channel_no: The channel to decrypt data for.
+ // in_data: The data to decrypt. This data is bytes_in bytes long.
+ // out_data: The buffer to write the decrypted data to. You may write more
+ // bytes of decrypted data than what you got as input, up to a maximum
+ // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or
+ // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine.
+ // bytes_in: The number of bytes in the input buffer.
+ // bytes_out: The number of bytes written in out_data.
+ virtual void decrypt(
+ int channel,
+ unsigned char* in_data,
+ unsigned char* out_data,
+ int bytes_in,
+ int* bytes_out) = 0;
+
+ // Encrypts a RTCP packet. Otherwise, this method has the same contract as
+ // encrypt().
+ virtual void encrypt_rtcp(
+ int channel,
+ unsigned char* in_data,
+ unsigned char* out_data,
+ int bytes_in,
+ int* bytes_out) = 0;
+
+ // Decrypts a RTCP packet. Otherwise, this method has the same contract as
+ // decrypt().
+ virtual void decrypt_rtcp(
+ int channel,
+ unsigned char* in_data,
+ unsigned char* out_data,
+ int bytes_in,
+ int* bytes_out) = 0;
+
+protected:
+ virtual ~Encryption() {}
+ Encryption() {}
+};
+
+// External transport callback interface
+class Transport
+{
+public:
+ virtual int SendPacket(int channel, const void *data, int len) = 0;
+ virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
+
+protected:
+ virtual ~Transport() {}
+ Transport() {}
+};
+
+// ==================================================================
+// Voice specific types
+// ==================================================================
+
+// Each codec supported can be described by this structure.
+struct CodecInst
+{
+ int pltype;
+ char plname[RTP_PAYLOAD_NAME_SIZE];
+ int plfreq;
+ int pacsize;
+ int channels;
+ int rate;
+};
+
+enum FrameType
+{
+ kFrameEmpty = 0,
+ kAudioFrameSpeech = 1,
+ kAudioFrameCN = 2,
+ kVideoFrameKey = 3, // independent frame
+ kVideoFrameDelta = 4, // depends on the previus frame
+ kVideoFrameGolden = 5, // depends on a old known previus frame
+ kVideoFrameAltRef = 6
+};
+
+// RTP
+enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
+
+enum RTPDirections
+{
+ kRtpIncoming = 0,
+ kRtpOutgoing
+};
+
+enum PayloadFrequencies
+{
+ kFreq8000Hz = 8000,
+ kFreq16000Hz = 16000,
+ kFreq32000Hz = 32000
+};
+
+enum VadModes // degree of bandwidth reduction
+{
+ kVadConventional = 0, // lowest reduction
+ kVadAggressiveLow,
+ kVadAggressiveMid,
+ kVadAggressiveHigh // highest reduction
+};
+
+struct NetworkStatistics // NETEQ statistics
+{
+ // current jitter buffer size in ms
+ WebRtc_UWord16 currentBufferSize;
+ // preferred (optimal) buffer size in ms
+ WebRtc_UWord16 preferredBufferSize;
+ // adding extra delay due to "peaky jitter"
+ bool jitterPeaksFound;
+ // loss rate (network + late) in percent (in Q14)
+ WebRtc_UWord16 currentPacketLossRate;
+ // late loss rate in percent (in Q14)
+ WebRtc_UWord16 currentDiscardRate;
+ // fraction (of original stream) of synthesized speech inserted through
+ // expansion (in Q14)
+ WebRtc_UWord16 currentExpandRate;
+ // fraction of synthesized speech inserted through pre-emptive expansion
+ // (in Q14)
+ WebRtc_UWord16 currentPreemptiveRate;
+ // fraction of data removed through acceleration (in Q14)
+ WebRtc_UWord16 currentAccelerateRate;
+ // clock-drift in parts-per-million (negative or positive)
+ int32_t clockDriftPPM;
+ // average packet waiting time in the jitter buffer (ms)
+ int meanWaitingTimeMs;
+ // median packet waiting time in the jitter buffer (ms)
+ int medianWaitingTimeMs;
+ // min packet waiting time in the jitter buffer (ms)
+ int minWaitingTimeMs;
+ // max packet waiting time in the jitter buffer (ms)
+ int maxWaitingTimeMs;
+};
+
+typedef struct
+{
+ int min; // minumum
+ int max; // maximum
+ int average; // average
+} StatVal;
+
+typedef struct // All levels are reported in dBm0
+{
+ StatVal speech_rx; // long-term speech levels on receiving side
+ StatVal speech_tx; // long-term speech levels on transmitting side
+ StatVal noise_rx; // long-term noise/silence levels on receiving side
+ StatVal noise_tx; // long-term noise/silence levels on transmitting side
+} LevelStatistics;
+
+typedef struct // All levels are reported in dB
+{
+ StatVal erl; // Echo Return Loss
+ StatVal erle; // Echo Return Loss Enhancement
+ StatVal rerl; // RERL = ERL + ERLE
+ // Echo suppression inside EC at the point just before its NLP
+ StatVal a_nlp;
+} EchoStatistics;
+
+enum TelephoneEventDetectionMethods
+{
+ kInBand = 0,
+ kOutOfBand = 1,
+ kInAndOutOfBand = 2
+};
+
+enum NsModes // type of Noise Suppression
+{
+ kNsUnchanged = 0, // previously set mode
+ kNsDefault, // platform default
+ kNsConference, // conferencing default
+ kNsLowSuppression, // lowest suppression
+ kNsModerateSuppression,
+ kNsHighSuppression,
+ kNsVeryHighSuppression, // highest suppression
+};
+
+enum AgcModes // type of Automatic Gain Control
+{
+ kAgcUnchanged = 0, // previously set mode
+ kAgcDefault, // platform default
+ // adaptive mode for use when analog volume control exists (e.g. for
+ // PC softphone)
+ kAgcAdaptiveAnalog,
+ // scaling takes place in the digital domain (e.g. for conference servers
+ // and embedded devices)
+ kAgcAdaptiveDigital,
+ // can be used on embedded devices where the the capture signal is level
+ // is predictable
+ kAgcFixedDigital
+};
+
+// EC modes
+enum EcModes // type of Echo Control
+{
+ kEcUnchanged = 0, // previously set mode
+ kEcDefault, // platform default
+ kEcConference, // conferencing default (aggressive AEC)
+ kEcAec, // Acoustic Echo Cancellation
+ kEcAecm, // AEC mobile
+};
+
+// AECM modes
+enum AecmModes // mode of AECM
+{
+ kAecmQuietEarpieceOrHeadset = 0,
+ // Quiet earpiece or headset use
+ kAecmEarpiece, // most earpiece use
+ kAecmLoudEarpiece, // Loud earpiece or quiet speakerphone use
+ kAecmSpeakerphone, // most speakerphone use (default)
+ kAecmLoudSpeakerphone // Loud speakerphone
+};
+
+// AGC configuration
+typedef struct
+{
+ unsigned short targetLeveldBOv;
+ unsigned short digitalCompressionGaindB;
+ bool limiterEnable;
+} AgcConfig; // AGC configuration parameters
+
+enum StereoChannel
+{
+ kStereoLeft = 0,
+ kStereoRight,
+ kStereoBoth
+};
+
+// Audio device layers
+enum AudioLayers
+{
+ kAudioPlatformDefault = 0,
+ kAudioWindowsWave = 1,
+ kAudioWindowsCore = 2,
+ kAudioLinuxAlsa = 3,
+ kAudioLinuxPulse = 4
+};
+
+enum NetEqModes // NetEQ playout configurations
+{
+ // Optimized trade-off between low delay and jitter robustness for two-way
+ // communication.
+ kNetEqDefault = 0,
+ // Improved jitter robustness at the cost of increased delay. Can be
+ // used in one-way communication.
+ kNetEqStreaming = 1,
+ // Optimzed for decodability of fax signals rather than for perceived audio
+ // quality.
+ kNetEqFax = 2,
+};
+
+enum NetEqBgnModes // NetEQ Background Noise (BGN) configurations
+{
+ // BGN is always on and will be generated when the incoming RTP stream
+ // stops (default).
+ kBgnOn = 0,
+ // The BGN is faded to zero (complete silence) after a few seconds.
+ kBgnFade = 1,
+ // BGN is not used at all. Silence is produced after speech extrapolation
+ // has faded.
+ kBgnOff = 2,
+};
+
+enum OnHoldModes // On Hold direction
+{
+ kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state.
+ kHoldSendOnly, // Put only sending in on-hold state.
+ kHoldPlayOnly // Put only playing in on-hold state.
+};
+
+enum AmrMode
+{
+ kRfc3267BwEfficient = 0,
+ kRfc3267OctetAligned = 1,
+ kRfc3267FileStorage = 2,
+};
+
+// ==================================================================
+// Video specific types
+// ==================================================================
+
+// Raw video types
+enum RawVideoType
+{
+ kVideoI420 = 0,
+ kVideoYV12 = 1,
+ kVideoYUY2 = 2,
+ kVideoUYVY = 3,
+ kVideoIYUV = 4,
+ kVideoARGB = 5,
+ kVideoRGB24 = 6,
+ kVideoRGB565 = 7,
+ kVideoARGB4444 = 8,
+ kVideoARGB1555 = 9,
+ kVideoMJPEG = 10,
+ kVideoNV12 = 11,
+ kVideoNV21 = 12,
+ kVideoBGRA = 13,
+ kVideoUnknown = 99
+};
+
+// Video codec
+enum { kConfigParameterSize = 128};
+enum { kPayloadNameSize = 32};
+enum { kMaxSimulcastStreams = 4};
+enum { kMaxTemporalStreams = 4};
+
+enum VideoCodecComplexity
+{
+ kComplexityNormal = 0,
+ kComplexityHigh = 1,
+ kComplexityHigher = 2,
+ kComplexityMax = 3
+};
+
+enum VideoCodecProfile
+{
+ kProfileBase = 0x00,
+ kProfileMain = 0x01
+};
+
+enum VP8ResilienceMode {
+ kResilienceOff, // The stream produced by the encoder requires a
+ // recovery frame (typically a key frame) to be
+ // decodable after a packet loss.
+ kResilientStream, // A stream produced by the encoder is resilient to
+ // packet losses, but packets within a frame subsequent
+ // to a loss can't be decoded.
+ kResilientFrames // Same as kResilientStream but with added resilience
+ // within a frame.
+};
+
+// VP8 specific
+struct VideoCodecVP8
+{
+ bool pictureLossIndicationOn;
+ bool feedbackModeOn;
+ VideoCodecComplexity complexity;
+ VP8ResilienceMode resilience;
+ unsigned char numberOfTemporalLayers;
+};
+
+// Unknown specific
+struct VideoCodecGeneric
+{
+};
+
+// Video codec types
+enum VideoCodecType
+{
+ kVideoCodecVP8,
+ kVideoCodecI420,
+ kVideoCodecRED,
+ kVideoCodecULPFEC,
+ kVideoCodecUnknown
+};
+
+union VideoCodecUnion
+{
+ VideoCodecVP8 VP8;
+ VideoCodecGeneric Generic;
+};
+
+
+// Simulcast is when the same stream is encoded multiple times with different
+// settings such as resolution.
+struct SimulcastStream
+{
+ unsigned short width;
+ unsigned short height;
+ unsigned char numberOfTemporalLayers;
+ unsigned int maxBitrate;
+ unsigned int qpMax; // minimum quality
+};
+
+// Common video codec properties
+struct VideoCodec
+{
+ VideoCodecType codecType;
+ char plName[kPayloadNameSize];
+ unsigned char plType;
+
+ unsigned short width;
+ unsigned short height;
+
+ unsigned int startBitrate;
+ unsigned int maxBitrate;
+ unsigned int minBitrate;
+ unsigned char maxFramerate;
+
+ VideoCodecUnion codecSpecific;
+
+ unsigned int qpMax;
+ unsigned char numberOfSimulcastStreams;
+ SimulcastStream simulcastStream[kMaxSimulcastStreams];
+};
+} // namespace webrtc
+#endif // WEBRTC_COMMON_TYPES_H
diff --git a/trunk/src/common_video/OWNERS b/trunk/src/common_video/OWNERS
new file mode 100644
index 0000000..7183cf2
--- /dev/null
+++ b/trunk/src/common_video/OWNERS
@@ -0,0 +1,4 @@
+stefan@webrtc.org
+mikhal@webrtc.org
+marpan@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/trunk/src/common_video/common_video.gyp b/trunk/src/common_video/common_video.gyp
new file mode 100644
index 0000000..3adb56c
--- /dev/null
+++ b/trunk/src/common_video/common_video.gyp
@@ -0,0 +1,15 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ 'libyuv/libyuv.gypi',
+ 'jpeg/jpeg.gypi',
+ ],
+}
diff --git a/trunk/src/common_video/interface/video_image.h b/trunk/src/common_video/interface/video_image.h
new file mode 100644
index 0000000..82e7b15
--- /dev/null
+++ b/trunk/src/common_video/interface/video_image.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
+#define COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
+
+#include "typedefs.h"
+#include <stdlib.h>
+
+namespace webrtc
+{
+
+enum VideoFrameType
+{
+ kKeyFrame = 0,
+ kDeltaFrame = 1,
+ kGoldenFrame = 2,
+ kAltRefFrame = 3,
+ kSkipFrame = 4
+};
+
+class RawImage
+{
+public:
+ RawImage() : _width(0), _height(0), _timeStamp(0), _buffer(NULL),
+ _length(0), _size(0) {}
+
+ RawImage(WebRtc_UWord8* buffer, WebRtc_UWord32 length,
+ WebRtc_UWord32 size) :
+ _width(0), _height(0), _timeStamp(0),
+ _buffer(buffer), _length(length), _size(size) {}
+
+ WebRtc_UWord32 _width;
+ WebRtc_UWord32 _height;
+ WebRtc_UWord32 _timeStamp;
+ WebRtc_UWord8* _buffer;
+ WebRtc_UWord32 _length;
+ WebRtc_UWord32 _size;
+};
+
+class EncodedImage
+{
+public:
+ EncodedImage() :
+ _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
+ _frameType(kDeltaFrame), _buffer(NULL), _length(0),
+ _size(0), _completeFrame(false) {}
+
+ EncodedImage(WebRtc_UWord8* buffer,
+ WebRtc_UWord32 length,
+ WebRtc_UWord32 size) :
+ _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
+ _frameType(kDeltaFrame), _buffer(buffer), _length(length),
+ _size(size), _completeFrame(false) {}
+
+ WebRtc_UWord32 _encodedWidth;
+ WebRtc_UWord32 _encodedHeight;
+ WebRtc_UWord32 _timeStamp;
+ VideoFrameType _frameType;
+ WebRtc_UWord8* _buffer;
+ WebRtc_UWord32 _length;
+ WebRtc_UWord32 _size;
+ bool _completeFrame;
+};
+
+} // namespace webrtc
+
+#endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
diff --git a/trunk/src/common_video/jpeg/Android.mk b/trunk/src/common_video/jpeg/Android.mk
new file mode 100644
index 0000000..b5bc243
--- /dev/null
+++ b/trunk/src/common_video/jpeg/Android.mk
@@ -0,0 +1,42 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_jpeg
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ jpeg.cc \
+ data_manager.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../ \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../../ \
+ external/jpeg
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_video/jpeg/data_manager.cc b/trunk/src/common_video/jpeg/data_manager.cc
new file mode 100644
index 0000000..a5a7a48
--- /dev/null
+++ b/trunk/src/common_video/jpeg/data_manager.cc
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_video/jpeg/data_manager.h"
+
+namespace webrtc
+{
+
+typedef struct
+{
+ jpeg_source_mgr mgr;
+ JOCTET* next_input_byte;
+ size_t bytes_in_buffer; /* # of byte spaces remaining in buffer */
+} DataSrcMgr;
+
+void
+jpegSetSrcBuffer(j_decompress_ptr cinfo, JOCTET* srcBuffer, size_t bufferSize)
+{
+ DataSrcMgr* src;
+ if (cinfo->src == NULL)
+ { /* first time for this JPEG object? */
+ cinfo->src = (struct jpeg_source_mgr *)
+ (*cinfo->mem->alloc_small) ((j_common_ptr) cinfo,
+ JPOOL_PERMANENT, sizeof(DataSrcMgr));
+ }
+
+ // Setting required functionality
+ src = (DataSrcMgr*) cinfo->src;
+ src->mgr.init_source = initSrc;;
+ src->mgr.fill_input_buffer = fillInputBuffer;
+ src->mgr.skip_input_data = skipInputData;
+ src->mgr.resync_to_restart = jpeg_resync_to_restart; // use default
+ src->mgr.term_source = termSource;
+ // setting buffer/src
+ src->bytes_in_buffer = bufferSize;
+ src->next_input_byte = srcBuffer;
+
+}
+
+
+void
+initSrc(j_decompress_ptr cinfo)
+{
+ DataSrcMgr *src = (DataSrcMgr*)cinfo->src;
+ src->mgr.next_input_byte = src->next_input_byte;
+ src->mgr.bytes_in_buffer = src->bytes_in_buffer;
+}
+
+boolean
+fillInputBuffer(j_decompress_ptr cinfo)
+{
+ return false;
+}
+
+
+void
+skipInputData(j_decompress_ptr cinfo, long num_bytes)
+{
+ DataSrcMgr* src = (DataSrcMgr*)cinfo->src;
+ if (num_bytes > 0)
+ {
+ if ((unsigned long)num_bytes > src->mgr.bytes_in_buffer)
+ src->mgr.bytes_in_buffer = 0;
+ else
+ {
+ src->mgr.next_input_byte += num_bytes;
+ src->mgr.bytes_in_buffer -= num_bytes;
+ }
+ }
+}
+
+
+void
+termSource (j_decompress_ptr cinfo)
+{
+ //
+}
+
+} // end of namespace webrtc
diff --git a/trunk/src/common_video/jpeg/data_manager.h b/trunk/src/common_video/jpeg/data_manager.h
new file mode 100644
index 0000000..61609ec
--- /dev/null
+++ b/trunk/src/common_video/jpeg/data_manager.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Jpeg source data manager
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER
+#define WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER
+
+#include <stdio.h>
+extern "C" {
+#if defined(USE_SYSTEM_LIBJPEG)
+#include <jpeglib.h>
+#else
+#include "jpeglib.h"
+#endif
+}
+
+namespace webrtc
+{
+
+// Source manager:
+
+
+// a general function that will set these values
+void
+jpegSetSrcBuffer(j_decompress_ptr cinfo, JOCTET* srcBuffer, size_t bufferSize);
+
+
+// Initialize source. This is called by jpeg_read_header() before any
+// data is actually read.
+
+void
+initSrc(j_decompress_ptr cinfo);
+
+
+// Fill input buffer
+// This is called whenever bytes_in_buffer has reached zero and more
+// data is wanted.
+
+boolean
+fillInputBuffer(j_decompress_ptr cinfo);
+
+// Skip input data
+// Skip num_bytes worth of data.
+
+void
+skipInputData(j_decompress_ptr cinfo, long num_bytes);
+
+
+
+
+// Terminate source
+void
+termSource (j_decompress_ptr cinfo);
+
+} // end of namespace webrtc
+
+
+#endif /* WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER */
diff --git a/trunk/src/common_video/jpeg/include/jpeg.h b/trunk/src/common_video/jpeg/include/jpeg.h
new file mode 100644
index 0000000..05e759c
--- /dev/null
+++ b/trunk/src/common_video/jpeg/include/jpeg.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_JPEG
+#define WEBRTC_COMMON_VIDEO_JPEG
+
+#include "typedefs.h"
+#include "video_image.h"
+
+// jpeg forward declaration
+struct jpeg_compress_struct;
+struct jpeg_decompress_struct;
+
+namespace webrtc
+{
+
+class JpegEncoder
+{
+public:
+ JpegEncoder();
+ ~JpegEncoder();
+
+// SetFileName
+// Input:
+// - fileName - Pointer to input vector (should be less than 256) to which the
+// compressed file will be written to
+// Output:
+// - 0 : OK
+// - (-1) : Error
+ WebRtc_Word32 SetFileName(const char* fileName);
+
+// Encode an I420 image. The encoded image is saved to a file
+//
+// Input:
+// - inputImage : Image to be encoded
+//
+// Output:
+// - 0 : OK
+// - (-1) : Error
+ WebRtc_Word32 Encode(const RawImage& inputImage);
+
+private:
+
+ jpeg_compress_struct* _cinfo;
+ char _fileName[256];
+};
+
+class JpegDecoder
+{
+ public:
+ JpegDecoder();
+ ~JpegDecoder();
+
+// Decodes a JPEG-stream
+// Supports 1 image component. 3 interleaved image components,
+// YCbCr sub-sampling 4:4:4, 4:2:2, 4:2:0.
+//
+// Input:
+// - inputImage : encoded image to be decoded.
+// - outputImage : RawImage to store decoded output
+//
+// Output:
+// - 0 : OK
+// - (-1) : Error
+ WebRtc_Word32 Decode(const EncodedImage& inputImage,
+ RawImage& outputImage);
+ private:
+ jpeg_decompress_struct* _cinfo;
+};
+
+
+}
+#endif /* WEBRTC_COMMON_VIDEO_JPEG */
diff --git a/trunk/src/common_video/jpeg/jpeg.cc b/trunk/src/common_video/jpeg/jpeg.cc
new file mode 100644
index 0000000..00586d4
--- /dev/null
+++ b/trunk/src/common_video/jpeg/jpeg.cc
@@ -0,0 +1,375 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(WIN32)
+ #include <basetsd.h>
+#endif
+#include <setjmp.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "common_video/jpeg/include/jpeg.h"
+#include "common_video/jpeg/data_manager.h"
+
+extern "C" {
+#if defined(USE_SYSTEM_LIBJPEG)
+#include <jpeglib.h>
+#else
+#include "jpeglib.h"
+#endif
+}
+
+
+namespace webrtc
+{
+
+// Error handler
+struct myErrorMgr {
+
+ struct jpeg_error_mgr pub;
+ jmp_buf setjmp_buffer;
+};
+typedef struct myErrorMgr * myErrorPtr;
+
+METHODDEF(void)
+MyErrorExit (j_common_ptr cinfo)
+{
+ myErrorPtr myerr = (myErrorPtr) cinfo->err;
+
+ // Return control to the setjmp point
+ longjmp(myerr->setjmp_buffer, 1);
+}
+
+JpegEncoder::JpegEncoder()
+{
+ _cinfo = new jpeg_compress_struct;
+ strcpy(_fileName, "Snapshot.jpg");
+}
+
+JpegEncoder::~JpegEncoder()
+{
+ if (_cinfo != NULL)
+ {
+ delete _cinfo;
+ _cinfo = NULL;
+ }
+}
+
+
+WebRtc_Word32
+JpegEncoder::SetFileName(const char* fileName)
+{
+ if (!fileName)
+ {
+ return -1;
+ }
+
+ if (fileName)
+ {
+ strncpy(_fileName, fileName, 256);
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+JpegEncoder::Encode(const RawImage& inputImage)
+{
+ if (inputImage._buffer == NULL || inputImage._size == 0)
+ {
+ return -1;
+ }
+ if (inputImage._width < 1 || inputImage._height < 1)
+ {
+ return -1;
+ }
+
+ FILE* outFile = NULL;
+
+ const WebRtc_UWord32 width = inputImage._width;
+ const WebRtc_UWord32 height = inputImage._height;
+
+ // Set error handler
+ myErrorMgr jerr;
+ _cinfo->err = jpeg_std_error(&jerr.pub);
+ jerr.pub.error_exit = MyErrorExit;
+ // Establish the setjmp return context
+ if (setjmp(jerr.setjmp_buffer))
+ {
+ // If we get here, the JPEG code has signaled an error.
+ jpeg_destroy_compress(_cinfo);
+ if (outFile != NULL)
+ {
+ fclose(outFile);
+ }
+ return -1;
+ }
+
+ if ((outFile = fopen(_fileName, "wb")) == NULL)
+ {
+ return -2;
+ }
+ // Create a compression object
+ jpeg_create_compress(_cinfo);
+
+ // Setting destination file
+ jpeg_stdio_dest(_cinfo, outFile);
+
+ // Set parameters for compression
+ _cinfo->in_color_space = JCS_YCbCr;
+ jpeg_set_defaults(_cinfo);
+
+ _cinfo->image_width = width;
+ _cinfo->image_height = height;
+ _cinfo->input_components = 3;
+
+ _cinfo->comp_info[0].h_samp_factor = 2; // Y
+ _cinfo->comp_info[0].v_samp_factor = 2;
+ _cinfo->comp_info[1].h_samp_factor = 1; // U
+ _cinfo->comp_info[1].v_samp_factor = 1;
+ _cinfo->comp_info[2].h_samp_factor = 1; // V
+ _cinfo->comp_info[2].v_samp_factor = 1;
+ _cinfo->raw_data_in = TRUE;
+
+ WebRtc_UWord32 height16 = (height + 15) & ~15;
+ WebRtc_UWord8* imgPtr = inputImage._buffer;
+ WebRtc_UWord8* origImagePtr = NULL;
+ if (height16 != height)
+ {
+ // Copy image to an adequate size buffer
+ WebRtc_UWord32 requiredSize = height16 * width * 3 >> 1;
+ origImagePtr = new WebRtc_UWord8[requiredSize];
+ memset(origImagePtr, 0, requiredSize);
+ memcpy(origImagePtr, inputImage._buffer, inputImage._length);
+ imgPtr = origImagePtr;
+ }
+
+ jpeg_start_compress(_cinfo, TRUE);
+
+ JSAMPROW y[16],u[8],v[8];
+ JSAMPARRAY data[3];
+
+ data[0] = y;
+ data[1] = u;
+ data[2] = v;
+
+ WebRtc_UWord32 i, j;
+
+ for (j = 0; j < height; j += 16)
+ {
+ for (i = 0; i < 16; i++)
+ {
+ y[i] = (JSAMPLE*)imgPtr + width * (i + j);
+
+ if (i % 2 == 0)
+ {
+ u[i / 2] = (JSAMPLE*) imgPtr + width * height +
+ width / 2 * ((i + j) / 2);
+ v[i / 2] = (JSAMPLE*) imgPtr + width * height +
+ width * height / 4 + width / 2 * ((i + j) / 2);
+ }
+ }
+ jpeg_write_raw_data(_cinfo, data, 16);
+ }
+
+ jpeg_finish_compress(_cinfo);
+ jpeg_destroy_compress(_cinfo);
+
+ fclose(outFile);
+
+ if (origImagePtr != NULL)
+ {
+ delete [] origImagePtr;
+ }
+
+ return 0;
+}
+
+JpegDecoder::JpegDecoder()
+{
+ _cinfo = new jpeg_decompress_struct;
+}
+
+JpegDecoder::~JpegDecoder()
+{
+ if (_cinfo != NULL)
+ {
+ delete _cinfo;
+ _cinfo = NULL;
+ }
+}
+
+WebRtc_Word32
+JpegDecoder::Decode(const EncodedImage& inputImage,
+ RawImage& outputImage)
+{
+
+ WebRtc_UWord8* tmpBuffer = NULL;
+ // Set error handler
+ myErrorMgr jerr;
+ _cinfo->err = jpeg_std_error(&jerr.pub);
+ jerr.pub.error_exit = MyErrorExit;
+
+ // Establish the setjmp return context
+ if (setjmp(jerr.setjmp_buffer))
+ {
+ if (_cinfo->is_decompressor)
+ {
+ jpeg_destroy_decompress(_cinfo);
+ }
+ if (tmpBuffer != NULL)
+ {
+ delete [] tmpBuffer;
+ }
+ return -1;
+ }
+
+ _cinfo->out_color_space = JCS_YCbCr;
+
+ // Create decompression object
+ jpeg_create_decompress(_cinfo);
+
+ // Specify data source
+ jpegSetSrcBuffer(_cinfo, (JOCTET*) inputImage._buffer, inputImage._size);
+
+ // Read header data
+ jpeg_read_header(_cinfo, TRUE);
+
+ _cinfo->raw_data_out = TRUE;
+ jpeg_start_decompress(_cinfo);
+
+ // Check header
+ if (_cinfo->num_components == 4)
+ {
+ return -2; // not supported
+ }
+ if (_cinfo->progressive_mode == 1)
+ {
+ return -2; // not supported
+ }
+
+
+ WebRtc_UWord32 height = _cinfo->image_height;
+ WebRtc_UWord32 width = _cinfo->image_width;
+
+ // Making sure width and height are even
+ if (height % 2)
+ {
+ height++;
+ }
+ if (width % 2)
+ {
+ width++;
+ }
+
+ WebRtc_UWord32 height16 = (height + 15) & ~15;
+ WebRtc_UWord32 stride = (width + 15) & ~15;
+ WebRtc_UWord32 uvStride = ((((stride + 1) >> 1) + 15) & ~15);
+
+ WebRtc_UWord32 tmpRequiredSize = stride * height16 +
+ 2 * (uvStride * ((height16 + 1) >> 1));
+ WebRtc_UWord32 requiredSize = width * height * 3 >> 1;
+
+ // verify sufficient buffer size
+ if (outputImage._buffer && outputImage._size < requiredSize)
+ {
+ delete [] outputImage._buffer;
+ outputImage._buffer = NULL;
+ }
+
+ if (outputImage._buffer == NULL)
+ {
+ outputImage._buffer = new WebRtc_UWord8[requiredSize];
+ outputImage._size = requiredSize;
+ }
+
+ WebRtc_UWord8* outPtr = outputImage._buffer;
+
+ if (tmpRequiredSize > requiredSize)
+ {
+ tmpBuffer = new WebRtc_UWord8[(int) (tmpRequiredSize)];
+ outPtr = tmpBuffer;
+ }
+
+ JSAMPROW y[16],u[8],v[8];
+ JSAMPARRAY data[3];
+ data[0] = y;
+ data[1] = u;
+ data[2] = v;
+
+ WebRtc_UWord32 hInd, i;
+ WebRtc_UWord32 numScanLines = 16;
+ WebRtc_UWord32 numLinesProcessed = 0;
+
+ while (_cinfo->output_scanline < _cinfo->output_height)
+ {
+ hInd = _cinfo->output_scanline;
+ for (i = 0; i < numScanLines; i++)
+ {
+ y[i] = outPtr + stride * (i + hInd);
+
+ if (i % 2 == 0)
+ {
+ u[i / 2] = outPtr + stride * height16 +
+ stride / 2 * ((i + hInd) / 2);
+ v[i / 2] = outPtr + stride * height16 +
+ stride * height16 / 4 +
+ stride / 2 * ((i + hInd) / 2);
+ }
+ }
+ // Processes exactly one iMCU row per call
+ numLinesProcessed = jpeg_read_raw_data(_cinfo, data, numScanLines);
+ // Error in read
+ if (numLinesProcessed == 0)
+ {
+ jpeg_abort((j_common_ptr)_cinfo);
+ return -1;
+ }
+ }
+
+ if (tmpRequiredSize > requiredSize)
+ {
+ WebRtc_UWord8* dstFramePtr = outputImage._buffer;
+ WebRtc_UWord8* tmpPtr = outPtr;
+
+ for (WebRtc_UWord32 p = 0; p < 3; p++)
+ {
+ const WebRtc_UWord32 h = (p == 0) ? height : height >> 1;
+ const WebRtc_UWord32 h16 = (p == 0) ? height16 : height16 >> 1;
+ const WebRtc_UWord32 w = (p == 0) ? width : width >> 1;
+ const WebRtc_UWord32 s = (p == 0) ? stride : stride >> 1;
+
+ for (WebRtc_UWord32 i = 0; i < h; i++)
+ {
+ memcpy(dstFramePtr, tmpPtr, w);
+ dstFramePtr += w;
+ tmpPtr += s;
+ }
+ tmpPtr += (h16 - h) * s;
+ }
+ }
+
+ if (tmpBuffer != NULL)
+ {
+ delete [] tmpBuffer;
+ }
+ // Setting output Image parameter
+ outputImage._width = width;
+ outputImage._height = height;
+ outputImage._length = requiredSize;
+ outputImage._timeStamp = inputImage._timeStamp;
+
+ jpeg_finish_decompress(_cinfo);
+ jpeg_destroy_decompress(_cinfo);
+ return 0;
+}
+
+
+}
diff --git a/trunk/src/common_video/jpeg/jpeg.gypi b/trunk/src/common_video/jpeg/jpeg.gypi
new file mode 100644
index 0000000..1738a3d
--- /dev/null
+++ b/trunk/src/common_video/jpeg/jpeg.gypi
@@ -0,0 +1,90 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'variables': {
+ 'use_libjpeg_turbo%': '<(use_libjpeg_turbo)',
+ },
+ 'targets': [
+ {
+ 'target_name': 'webrtc_jpeg',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_libyuv',
+ ],
+ 'include_dirs': [
+ 'include',
+ '<(webrtc_root)',
+ '<(webrtc_root)/common_video/interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ '<(webrtc_root)/common_video/interface',
+ ],
+ },
+ 'conditions': [
+ ['build_libjpeg==1', {
+ 'conditions': [
+ ['build_with_chromium==1', {
+ 'dependencies': [
+ '<(libjpeg_gyp_path):libjpeg',
+ ],
+ }, {
+ 'conditions': [
+ ['use_libjpeg_turbo==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/libjpeg_turbo/libjpeg.gyp:libjpeg',
+ ],
+ }, {
+ 'dependencies': [
+ '<(DEPTH)/third_party/libjpeg/libjpeg.gyp:libjpeg',
+ ],
+ }],
+ ],
+ }],
+ ],
+ }, {
+ # Need to add a directory normally exported by libjpeg.gyp.
+ 'include_dirs': [ '<(DEPTH)/third_party/libjpeg', ],
+ }],
+ ],
+ 'sources': [
+ 'include/jpeg.h',
+ 'data_manager.cc',
+ 'data_manager.h',
+ 'jpeg.cc',
+ ],
+ },
+ ], # targets
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'jpeg_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'webrtc_jpeg',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'jpeg_unittest.cc',
+ ],
+ },
+ ] # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_video/jpeg/jpeg_unittest.cc b/trunk/src/common_video/jpeg/jpeg_unittest.cc
new file mode 100644
index 0000000..fdcbe34
--- /dev/null
+++ b/trunk/src/common_video/jpeg/jpeg_unittest.cc
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+#include <string>
+
+#include "common_video/jpeg/include/jpeg.h"
+#include "common_video/interface/video_image.h"
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+const unsigned int kImageWidth = 640;
+const unsigned int kImageHeight = 480;
+
+class JpegTest: public testing::Test {
+ protected:
+ JpegTest()
+ : input_filename_(webrtc::test::ProjectRootPath() +
+ "test/data/common_video/jpeg/webrtc_logo.jpg"),
+ decoded_filename_(webrtc::test::OutputPath() + "TestJpegDec.yuv"),
+ encoded_filename_(webrtc::test::OutputPath() + "TestJpegEnc.jpg"),
+ encoded_buffer_(NULL) {}
+ virtual ~JpegTest() {}
+
+ void SetUp() {
+ encoder_ = new JpegEncoder();
+ decoder_ = new JpegDecoder();
+ }
+
+ void TearDown() {
+ if (encoded_buffer_ != NULL) {
+ if (encoded_buffer_->_buffer != NULL) {
+ delete [] encoded_buffer_->_buffer;
+ }
+ delete encoded_buffer_;
+ }
+ delete encoder_;
+ delete decoder_;
+ }
+
+ // Reads an encoded image. Caller will have to deallocate the memory of this
+ // object and it's _buffer byte array.
+ EncodedImage* ReadEncodedImage(std::string input_filename) {
+ FILE* open_file = fopen(input_filename.c_str(), "rb");
+ assert(open_file != NULL);
+ size_t length = webrtc::test::GetFileSize(input_filename);
+ EncodedImage* encoded_buffer = new EncodedImage();
+ encoded_buffer->_buffer = new WebRtc_UWord8[length];
+ encoded_buffer->_size = length;
+ encoded_buffer->_length = length;
+ if (fread(encoded_buffer->_buffer, 1, length, open_file) != length) {
+ ADD_FAILURE() << "Error reading file:" << input_filename;
+ }
+ fclose(open_file);
+ return encoded_buffer;
+ }
+
+ std::string input_filename_;
+ std::string decoded_filename_;
+ std::string encoded_filename_;
+ EncodedImage* encoded_buffer_;
+ JpegEncoder* encoder_;
+ JpegDecoder* decoder_;
+};
+
+TEST_F(JpegTest, Decode) {
+ encoded_buffer_ = ReadEncodedImage(input_filename_);
+ RawImage image_buffer;
+ EXPECT_EQ(0, decoder_->Decode(*encoded_buffer_, image_buffer));
+ EXPECT_GT(image_buffer._length, 0u);
+ EXPECT_EQ(kImageWidth, image_buffer._width);
+ EXPECT_EQ(kImageHeight, image_buffer._height);
+ delete [] image_buffer._buffer;
+}
+
+TEST_F(JpegTest, EncodeInvalidInputs) {
+ RawImage empty;
+ empty._width = 164;
+ empty._height = 164;
+ EXPECT_EQ(-1, encoder_->SetFileName(0));
+ EXPECT_EQ(-1, encoder_->Encode(empty));
+
+ empty._buffer = new WebRtc_UWord8[10];
+ empty._size = 0;
+ EXPECT_EQ(-1, encoder_->Encode(empty));
+
+ empty._size = 10;
+ empty._height = 0;
+ EXPECT_EQ(-1, encoder_->Encode(empty));
+
+ empty._height = 164;
+ empty._width = 0;
+ EXPECT_EQ(-1, encoder_->Encode(empty));
+ delete[] empty._buffer;
+}
+
+TEST_F(JpegTest, Encode) {
+ // Decode our input image then encode it again to a new file:
+ encoded_buffer_ = ReadEncodedImage(input_filename_);
+ RawImage image_buffer;
+ EXPECT_EQ(0, decoder_->Decode(*encoded_buffer_, image_buffer));
+
+ EXPECT_EQ(0, encoder_->SetFileName(encoded_filename_.c_str()));
+ EXPECT_EQ(0, encoder_->Encode(image_buffer));
+
+ // Save decoded image to file.
+ FILE* save_file = fopen(decoded_filename_.c_str(), "wb");
+ fwrite(image_buffer._buffer, 1, image_buffer._length, save_file);
+ fclose(save_file);
+
+ delete[] image_buffer._buffer;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/Android.mk b/trunk/src/common_video/libyuv/Android.mk
new file mode 100644
index 0000000..90f6250
--- /dev/null
+++ b/trunk/src/common_video/libyuv/Android.mk
@@ -0,0 +1,42 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_yuv
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ libyuv.cc \
+ scaler.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../libyuv/files/include
+
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
\ No newline at end of file
diff --git a/trunk/src/common_video/libyuv/include/libyuv.h b/trunk/src/common_video/libyuv/include/libyuv.h
new file mode 100644
index 0000000..bf106f1
--- /dev/null
+++ b/trunk/src/common_video/libyuv/include/libyuv.h
@@ -0,0 +1,194 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * WebRTC's Wrapper to libyuv.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
+#define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
+
+#include "common_types.h" // RawVideoTypes.
+#include "typedefs.h"
+
+namespace webrtc {
+
+// TODO(mikhal): 1. Sync libyuv and WebRtc meaning of stride.
+// 2. Reorder parameters for consistency.
+
+// Supported video types.
+enum VideoType {
+ kUnknown,
+ kI420,
+ kIYUV,
+ kRGB24,
+ kABGR,
+ kARGB,
+ kARGB4444,
+ kRGB565,
+ kARGB1555,
+ kYUY2,
+ kYV12,
+ kUYVY,
+ kMJPG,
+ kNV21,
+ kNV12,
+ kBGRA,
+};
+
+// Conversion between the RawVideoType and the LibYuv videoType.
+// TODO(wu): Consolidate types into one type throughout WebRtc.
+VideoType RawVideoTypeToCommonVideoVideoType(RawVideoType type);
+
+// Supported rotation
+// Direction of rotation - clockwise.
+enum VideoRotationMode {
+ kRotateNone = 0,
+ kRotate90 = 90,
+ kRotate180 = 180,
+ kRotate270 = 270,
+};
+
+// Calculate the required buffer size.
+// Input:
+// - type - The type of the designated video frame.
+// - width - frame width in pixels.
+// - height - frame height in pixels.
+// Return value: The required size in bytes to accommodate the specified
+// video frame or -1 in case of an error .
+int CalcBufferSize(VideoType type, int width, int height);
+
+// Compute required buffer size when converting from one type to another.
+// Input:
+// - src_video_type - Type of the existing video frame.
+// - dst_video_type - Type of the designated video frame.
+// - length - length in bytes of the data.
+// Return value: The required size in bytes to accommodate the specified
+// converted video frame or -1 in case of an error.
+int CalcBufferSize(VideoType src_video_type,
+ VideoType dst_video_type,
+ int length);
+// TODO (mikhal): Merge the two functions above.
+
+
+// Convert To I420
+// Input:
+// - src_video_type : Type of input video.
+// - src_frame : Pointer to a source frame.
+// - crop_x/crop_y : Starting positions for cropping (0 for no crop).
+// - src/dst_width : src/dst width in pixels.
+// - src/dst_height : src/dst height in pixels.
+// - sample_size : Required only for the parsing of MJPG (set to 0 else).
+// - dst_stride : Number of bytes in a row of the dst Y plane.
+// - rotate : Rotation mode of output image.
+// Output:
+// - dst_frame : Pointer to a destination frame.
+// Return value: 0 if OK, < 0 otherwise.
+
+int ConvertToI420(VideoType src_video_type,
+ const uint8_t* src_frame,
+ int crop_x, int crop_y,
+ int src_width, int src_height,
+ int sample_size,
+ int dst_width, int dst_height, int dst_stride,
+ VideoRotationMode rotation,
+ uint8_t* dst_frame);
+
+// Convert From I420
+// Input:
+// - src_frame : Pointer to a source frame.
+// - src_stride : Number of bytes in a row of the src Y plane.
+// - dst_video_type : Type of output video.
+// - dst_sample_size : Required only for the parsing of MJPG.
+// - width : Width in pixels.
+// - height : Height in pixels.
+// - dst_frame : Pointer to a destination frame.
+// Return value: 0 if OK, < 0 otherwise.
+int ConvertFromI420(const uint8_t* src_frame, int src_stride,
+ VideoType dst_video_type, int dst_sample_size,
+ int width, int height,
+ uint8_t* dst_frame);
+// ConvertFrom YV12.
+// Interface - same as above.
+int ConvertFromYV12(const uint8_t* src_frame, int src_stride,
+ VideoType dst_video_type, int dst_sample_size,
+ int width, int height,
+ uint8_t* dst_frame);
+
+// The following list describes designated conversion functions which
+// are not covered by the previous general functions.
+// Input and output descriptions mostly match the above descriptions, and are
+// therefore omitted.
+// Possible additional input value - dst_stride - stride of the dst frame.
+
+int ConvertI420ToARGB4444(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width,
+ int height,
+ int dst_stride);
+int ConvertI420ToRGB565(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width,
+ int height);
+int ConvertI420ToARGB1555(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width,
+ int height,
+ int dst_stride);
+int ConvertRGB24ToARGB(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height,
+ int dst_stride);
+int ConvertNV12ToRGB565(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height);
+
+// Mirror functions
+// The following 2 functions perform mirroring on a given image
+// (LeftRight/UpDown).
+// Input:
+// - width : Image width in pixels.
+// - height : Image height in pixels.
+// - src_frame : Pointer to a source frame.
+// - dst_frame : Pointer to a destination frame.
+// Return value: 0 if OK, < 0 otherwise.
+int MirrorI420LeftRight(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height);
+int MirrorI420UpDown(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height);
+
+// Mirror functions + conversion
+// Input:
+// - src_frame : Pointer to source frame.
+// - dst_frame : Pointer to destination frame.
+// - src_width : Width of input buffer.
+// - src_height : Height of input buffer.
+// - src_color_space : Color space to convert from, I420 if no
+// conversion should be done.
+// Return value: 0 if OK, < 0 otherwise.
+int ConvertToI420AndMirrorUpDown(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int src_width,
+ int src_height,
+ VideoType src_video_type);
+
+// Compute PSNR for an I420 frame (all planes).
+double I420PSNR(const uint8_t* ref_frame,
+ const uint8_t* test_frame,
+ int width, int height);
+// Compute SSIM for an I420 frame (all planes).
+double I420SSIM(const uint8_t* ref_frame,
+ const uint8_t* test_frame,
+ int width, int height);
+}
+
+#endif // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
diff --git a/trunk/src/common_video/libyuv/include/scaler.h b/trunk/src/common_video/libyuv/include/scaler.h
new file mode 100644
index 0000000..8f432f3
--- /dev/null
+++ b/trunk/src/common_video/libyuv/include/scaler.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Interface to the LibYuv scaling functionality
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
+#define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Supported scaling types
+enum ScaleMethod {
+ kScalePoint, // no interpolation
+ kScaleBilinear,
+ kScaleBox
+};
+
+// TODO (mikhal): Have set return the expected value of the dst_frame, such
+// that the user can allocate memory for Scale().
+class Scaler {
+ public:
+ Scaler();
+ ~Scaler();
+
+ // Set interpolation properties:
+ //
+ // Return value: 0 - OK
+ // -1 - parameter error
+ int Set(int src_width, int src_height,
+ int dst_width, int dst_height,
+ VideoType src_video_type, VideoType dst_video_type,
+ ScaleMethod method);
+
+ // Scale frame
+ // Memory is allocated by user. If dst_frame is not of sufficient size,
+ // the frame will be reallocated to the appropriate size.
+ // Return value: 0 - OK,
+ // -1 - parameter error
+ // -2 - scaler not set
+ int Scale(const uint8_t* src_frame,
+ uint8_t*& dst_frame,
+ int& dst_size);
+
+ private:
+ // Determine if the VideoTypes are currently supported.
+ bool SupportedVideoType(VideoType src_video_type,
+ VideoType dst_video_type);
+
+ ScaleMethod method_;
+ int src_width_;
+ int src_height_;
+ int dst_width_;
+ int dst_height_;
+ bool set_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
diff --git a/trunk/src/common_video/libyuv/libyuv.cc b/trunk/src/common_video/libyuv/libyuv.cc
new file mode 100644
index 0000000..d1ced7a
--- /dev/null
+++ b/trunk/src/common_video/libyuv/libyuv.cc
@@ -0,0 +1,450 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_video/libyuv/include/libyuv.h"
+
+#include <assert.h>
+
+// LibYuv includes
+#ifdef WEBRTC_ANDROID
+#include "libyuv/files/include/libyuv.h"
+#else
+#include "third_party/libyuv/include/libyuv.h"
+#endif
+
+namespace webrtc {
+
+VideoType RawVideoTypeToCommonVideoVideoType(RawVideoType type) {
+ switch (type) {
+ case kVideoI420:
+ return kI420;
+ case kVideoIYUV:
+ return kIYUV;
+ case kVideoRGB24:
+ return kRGB24;
+ case kVideoARGB:
+ return kARGB;
+ case kVideoARGB4444:
+ return kARGB4444;
+ case kVideoRGB565:
+ return kRGB565;
+ case kVideoARGB1555:
+ return kARGB1555;
+ case kVideoYUY2:
+ return kYUY2;
+ case kVideoYV12:
+ return kYV12;
+ case kVideoUYVY:
+ return kUYVY;
+ case kVideoNV21:
+ return kNV21;
+ case kVideoNV12:
+ return kNV12;
+ case kVideoBGRA:
+ return kBGRA;
+ case kVideoMJPEG:
+ return kMJPG;
+ default:
+ assert(false);
+ }
+ return kUnknown;
+}
+
+int CalcBufferSize(VideoType type, int width, int height) {
+ int bits_per_pixel = 32;
+ switch (type) {
+ case kI420:
+ case kNV12:
+ case kNV21:
+ case kIYUV:
+ case kYV12:
+ bits_per_pixel = 12;
+ break;
+ case kARGB4444:
+ case kRGB565:
+ case kARGB1555:
+ case kYUY2:
+ case kUYVY:
+ bits_per_pixel = 16;
+ break;
+ case kRGB24:
+ bits_per_pixel = 24;
+ break;
+ case kBGRA:
+ case kARGB:
+ bits_per_pixel = 32;
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+ return (width * height * bits_per_pixel) / 8; // bytes
+}
+
+int CalcBufferSize(VideoType src_video_type,
+ VideoType dst_video_type,
+ int length) {
+ int src_bits_per_pixel = 32;
+ switch (src_video_type) {
+ case kI420:
+ case kNV12:
+ case kNV21:
+ case kIYUV:
+ case kYV12:
+ src_bits_per_pixel = 12;
+ break;
+ case kARGB4444:
+ case kRGB565:
+ case kARGB1555:
+ case kYUY2:
+ case kUYVY:
+ src_bits_per_pixel = 16;
+ break;
+ case kRGB24:
+ src_bits_per_pixel = 24;
+ break;
+ case kBGRA:
+ case kARGB:
+ src_bits_per_pixel = 32;
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+
+ int dst_bits_per_pixel = 32;
+ switch (dst_video_type) {
+ case kI420:
+ case kIYUV:
+ case kYV12:
+ dst_bits_per_pixel = 12;
+ break;
+ case kARGB4444:
+ case kRGB565:
+ case kARGB1555:
+ case kYUY2:
+ case kUYVY:
+ dst_bits_per_pixel = 16;
+ break;
+ case kRGB24:
+ dst_bits_per_pixel = 24;
+ break;
+ case kBGRA:
+ case kARGB:
+ dst_bits_per_pixel = 32;
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+ return (length * dst_bits_per_pixel) / src_bits_per_pixel;
+}
+
+int ConvertI420ToARGB4444(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height,
+ int dst_stride) {
+ if (dst_stride == 0 || dst_stride == width)
+ dst_stride = 2 * width;
+ const uint8_t* yplane = src_frame;
+ const uint8_t* uplane = src_frame + width * height;
+ const uint8_t* vplane = uplane + (width * height / 4);
+
+ return libyuv::I420ToARGB4444(yplane, width,
+ uplane, width / 2,
+ vplane, width / 2,
+ dst_frame, dst_stride,
+ width, height);
+}
+
+int ConvertI420ToRGB565(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height) {
+ int abs_height = (height < 0) ? -height : height;
+ const uint8_t* yplane = src_frame;
+ const uint8_t* uplane = src_frame + width * abs_height;
+ const uint8_t* vplane = uplane + (width * abs_height / 4);
+
+ return libyuv::I420ToRGB565(yplane, width,
+ uplane, width / 2,
+ vplane, width / 2,
+ dst_frame, width,
+ width, height);
+}
+
+int ConvertI420ToARGB1555(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height,
+ int dst_stride) {
+ if (dst_stride == 0 || dst_stride == width)
+ dst_stride = 2 * width;
+ else if (dst_stride < 2 * width)
+ return -1;
+
+ const uint8_t* yplane = src_frame;
+ const uint8_t* uplane = src_frame + width * height;
+ const uint8_t* vplane = uplane + (width * height / 4);
+
+ return libyuv::I420ToARGB1555(yplane, width,
+ uplane, width / 2,
+ vplane, width / 2,
+ dst_frame, dst_stride,
+ width, height);
+}
+
+int ConvertNV12ToRGB565(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height) {
+ const uint8_t* yplane = src_frame;
+ const uint8_t* uvInterlaced = src_frame + (width * height);
+
+ return libyuv::NV12ToRGB565(yplane, width,
+ uvInterlaced, width / 2,
+ dst_frame, width,
+ width, height);
+}
+
+int ConvertRGB24ToARGB(const uint8_t* src_frame, uint8_t* dst_frame,
+ int width, int height, int dst_stride) {
+ if (dst_stride == 0 || dst_stride == width)
+ dst_stride = width;
+ // Stride - currently webrtc style
+ return libyuv::RGB24ToARGB(src_frame, width,
+ dst_frame, dst_stride,
+ width, height);
+}
+
+libyuv::RotationMode ConvertRotationMode(VideoRotationMode rotation) {
+ switch(rotation) {
+ case kRotateNone:
+ return libyuv::kRotate0;
+ case kRotate90:
+ return libyuv::kRotate90;
+ case kRotate180:
+ return libyuv::kRotate180;
+ case kRotate270:
+ return libyuv::kRotate270;
+ }
+ assert(false);
+ return libyuv::kRotate0;
+}
+
+int ConvertVideoType(VideoType video_type) {
+ switch(video_type) {
+ case kUnknown:
+ return libyuv::FOURCC_ANY;
+ case kI420:
+ return libyuv::FOURCC_I420;
+ case kIYUV: // same as KYV12
+ case kYV12:
+ return libyuv::FOURCC_YV12;
+ case kRGB24:
+ return libyuv::FOURCC_24BG;
+ case kABGR:
+ return libyuv::FOURCC_ABGR;
+ case kARGB4444:
+ case kRGB565:
+ case kARGB1555:
+ // TODO(mikhal): Not supported;
+ assert(false);
+ return libyuv::FOURCC_ANY;
+ case kYUY2:
+ return libyuv::FOURCC_YUY2;
+ case kUYVY:
+ return libyuv::FOURCC_UYVY;
+ case kMJPG:
+ return libyuv::FOURCC_MJPG;
+ case kNV21:
+ return libyuv::FOURCC_NV21;
+ case kNV12:
+ return libyuv::FOURCC_NV12;
+ case kARGB:
+ return libyuv::FOURCC_ARGB;
+ case kBGRA:
+ return libyuv::FOURCC_BGRA;
+ }
+ assert(false);
+ return libyuv::FOURCC_ANY;
+}
+
+int ConvertToI420(VideoType src_video_type,
+ const uint8_t* src_frame,
+ int crop_x, int crop_y,
+ int src_width, int src_height,
+ int sample_size,
+ int dst_width, int dst_height, int dst_stride,
+ VideoRotationMode rotation,
+ uint8_t* dst_frame) {
+ // All sanity tests are conducted within LibYuv.
+ uint8_t* dst_yplane = dst_frame;
+ uint8_t* dst_uplane = dst_yplane + dst_width * dst_height;
+ uint8_t* dst_vplane = dst_uplane + (dst_width * dst_height / 4);
+ return libyuv::ConvertToI420(src_frame, sample_size,
+ dst_yplane, dst_stride,
+ dst_uplane, (dst_stride + 1) / 2,
+ dst_vplane, (dst_stride + 1) / 2,
+ crop_x, crop_y,
+ src_width, src_height,
+ dst_width, dst_height,
+ ConvertRotationMode(rotation),
+ ConvertVideoType(src_video_type));
+}
+
+int ConvertFromI420(const uint8_t* src_frame, int src_stride,
+ VideoType dst_video_type, int dst_sample_size,
+ int width, int height,
+ uint8_t* dst_frame) {
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_yplane + width * height;
+ const uint8_t* src_vplane = src_uplane + (width * height / 4);
+ return libyuv::ConvertFromI420(src_yplane, src_stride,
+ src_uplane, (src_stride + 1) / 2,
+ src_vplane, (src_stride + 1) / 2,
+ dst_frame, dst_sample_size,
+ width, height,
+ ConvertVideoType(dst_video_type));
+}
+
+int ConvertFromYV12(const uint8_t* src_frame, int src_stride,
+ VideoType dst_video_type, int dst_sample_size,
+ int width, int height,
+ uint8_t* dst_frame) {
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_yplane + width * height;
+ const uint8_t* src_vplane = src_uplane + (width * height / 4);
+ // YV12 = Y, V, U
+ return libyuv::ConvertFromI420(src_yplane, src_stride,
+ src_vplane, (src_stride + 1) / 2,
+ src_uplane, (src_stride + 1) / 2,
+ dst_frame, dst_sample_size,
+ width, height,
+ ConvertVideoType(dst_video_type));
+}
+
+int MirrorI420LeftRight(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int width, int height) {
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_yplane + width * height;
+ const uint8_t* src_vplane = src_uplane + (width * height / 4);
+ uint8_t* dst_yplane = dst_frame;
+ uint8_t* dst_uplane = dst_yplane + width * height;
+ uint8_t* dst_vplane = dst_uplane + (width * height / 4);
+ return libyuv::I420Mirror(src_yplane, width,
+ src_uplane, width / 2,
+ src_vplane, width / 2,
+ dst_yplane, width,
+ dst_uplane, width / 2,
+ dst_vplane, width / 2,
+ width, height);
+}
+
+int MirrorI420UpDown(const uint8_t* src_frame, uint8_t* dst_frame,
+ int width, int height) {
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_frame + width * height;
+ const uint8_t* src_vplane = src_uplane + (width * height / 4);
+ uint8_t* dst_yplane = dst_frame;
+ uint8_t* dst_uplane = dst_frame + width * height;
+ uint8_t* dst_vplane = dst_uplane + (width * height / 4);
+
+ // Inserting negative height flips the frame.
+ return libyuv::I420Copy(src_yplane, width,
+ src_uplane, width / 2,
+ src_vplane, width / 2,
+ dst_yplane, width,
+ dst_uplane, width / 2,
+ dst_vplane, width / 2,
+ width, -height);
+}
+
+int ConvertToI420AndMirrorUpDown(const uint8_t* src_frame,
+ uint8_t* dst_frame,
+ int src_width, int src_height,
+ VideoType src_video_type) {
+ if (src_video_type != kI420 && src_video_type != kYV12)
+ return -1;
+ // TODO(mikhal): Use a more general convert function - with negative height
+ // Inserting negative height flips the frame.
+ // Using I420Copy with a negative height.
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_frame + src_width * src_height;
+ const uint8_t* src_vplane = src_uplane + (src_width * src_height / 4);
+ uint8_t* dst_yplane = dst_frame;
+ uint8_t* dst_uplane = dst_frame + src_width * src_height;
+ uint8_t* dst_vplane = dst_uplane + (src_width * src_height / 4);
+ if (src_video_type == kYV12) {
+ // Switch U and V
+ dst_vplane = dst_frame + src_width * src_height;
+ dst_uplane = dst_vplane + (src_width * src_height / 4);
+ }
+ // Inserting negative height flips the frame.
+ return libyuv::I420Copy(src_yplane, src_width,
+ src_uplane, src_width / 2,
+ src_vplane, src_width / 2,
+ dst_yplane, src_width,
+ dst_uplane, src_width / 2,
+ dst_vplane, src_width / 2,
+ src_width, -src_height);
+}
+
+// Compute PSNR for an I420 frame (all planes)
+double I420PSNR(const uint8_t* ref_frame,
+ const uint8_t* test_frame,
+ int width, int height) {
+ if (!ref_frame || !test_frame)
+ return -1;
+ else if (height < 0 || width < 0)
+ return -1;
+ const uint8_t* src_y_a = ref_frame;
+ const uint8_t* src_u_a = src_y_a + width * height;
+ const uint8_t* src_v_a = src_u_a + (width * height / 4);
+ const uint8_t* src_y_b = test_frame;
+ const uint8_t* src_u_b = src_y_b + width * height;
+ const uint8_t* src_v_b = src_u_b + (width * height / 4);
+ int stride_y = width;
+ int stride_uv = (width + 1) / 2;
+ double psnr = libyuv::I420Psnr(src_y_a, stride_y,
+ src_u_a, stride_uv,
+ src_v_a, stride_uv,
+ src_y_b, stride_y,
+ src_u_b, stride_uv,
+ src_v_b, stride_uv,
+ width, height);
+ // LibYuv sets the max psnr value to 128, we restrict it to 48.
+ // In case of 0 mse in one frame, 128 can skew the results significantly.
+ return (psnr > 48.0) ? 48.0 : psnr;
+}
+// Compute SSIM for an I420 frame (all planes)
+double I420SSIM(const uint8_t* ref_frame,
+ const uint8_t* test_frame,
+ int width, int height) {
+ if (!ref_frame || !test_frame)
+ return -1;
+ else if (height < 0 || width < 0)
+ return -1;
+ const uint8_t* src_y_a = ref_frame;
+ const uint8_t* src_u_a = src_y_a + width * height;
+ const uint8_t* src_v_a = src_u_a + (width * height / 4);
+ const uint8_t* src_y_b = test_frame;
+ const uint8_t* src_u_b = src_y_b + width * height;
+ const uint8_t* src_v_b = src_u_b + (width * height / 4);
+ int stride_y = width;
+ int stride_uv = (width + 1) / 2;
+ return libyuv::I420Ssim(src_y_a, stride_y,
+ src_u_a, stride_uv,
+ src_v_a, stride_uv,
+ src_y_b, stride_y,
+ src_u_b, stride_uv,
+ src_v_b, stride_uv,
+ width, height);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/libyuv.gypi b/trunk/src/common_video/libyuv/libyuv.gypi
new file mode 100644
index 0000000..bace5a2
--- /dev/null
+++ b/trunk/src/common_video/libyuv/libyuv.gypi
@@ -0,0 +1,55 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_libyuv',
+ 'type': '<(library)',
+ 'conditions': [
+ ['build_libyuv==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv'
+ ],
+ }, {
+ # Need to add a directory normally exported by libyuv.gyp.
+ 'include_dirs': [ '<(DEPTH)/third_party/libyuv/include', ],
+ }],
+ ],
+ 'sources': [
+ 'include/libyuv.h',
+ 'include/scaler.h',
+ 'libyuv.cc',
+ 'scaler.cc',
+ ],
+ 'include_dirs': [
+ '<(DEPTH)',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'libyuv_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'webrtc_libyuv',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'libyuv_unittest.cc',
+ 'scaler_unittest.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
diff --git a/trunk/src/common_video/libyuv/libyuv_unittest.cc b/trunk/src/common_video/libyuv/libyuv_unittest.cc
new file mode 100644
index 0000000..07225ec
--- /dev/null
+++ b/trunk/src/common_video/libyuv/libyuv_unittest.cc
@@ -0,0 +1,247 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <string.h>
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "gtest/gtest.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+int PrintFrame(const uint8_t* frame, int width, int height) {
+ if (frame == NULL)
+ return -1;
+ int k = 0;
+ for (int i = 0; i < height; i++) {
+ for (int j = 0; j < width; j++) {
+ printf("%d ", frame[k++]);
+ }
+ printf(" \n");
+ }
+ printf(" \n");
+ return 0;
+}
+
+int PrintFrame(const uint8_t* frame, int width,
+ int height, const char* str) {
+ if (frame == NULL)
+ return -1;
+ printf("%s %dx%d \n", str, width, height);
+
+ const uint8_t* frame_y = frame;
+ const uint8_t* frame_u = frame_y + width * height;
+ const uint8_t* frame_v = frame_u + width * height / 4;
+
+ int ret = 0;
+ ret += PrintFrame(frame_y, width, height);
+ ret += PrintFrame(frame_u, width / 2, height / 2);
+ ret += PrintFrame(frame_v, width / 2, height / 2);
+
+ return ret;
+}
+
+void CreateImage(int width, int height,
+ uint8_t* frame, int offset,
+ int height_factor, int width_factor) {
+ if (frame == NULL)
+ return;
+ for (int i = 0; i < height; i++) {
+ for (int j = 0; j < width; j++) {
+ *frame = static_cast<uint8_t>((i + offset) * height_factor
+ + j * width_factor);
+ frame++;
+ }
+ }
+}
+
+class TestLibYuv : public ::testing::Test {
+ protected:
+ TestLibYuv();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ FILE* source_file_;
+ const int width_;
+ const int height_;
+ const int frame_length_;
+};
+
+// TODO (mikhal): Use scoped_ptr when handling buffers.
+TestLibYuv::TestLibYuv()
+ : source_file_(NULL),
+ width_(352),
+ height_(288),
+ frame_length_(CalcBufferSize(kI420, 352, 288)) {
+}
+
+void TestLibYuv::SetUp() {
+ const std::string input_file_name = webrtc::test::ProjectRootPath() +
+ "resources/foreman_cif.yuv";
+ source_file_ = fopen(input_file_name.c_str(), "rb");
+ ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
+ input_file_name << "\n";
+}
+
+void TestLibYuv::TearDown() {
+ if (source_file_ != NULL) {
+ ASSERT_EQ(0, fclose(source_file_));
+ }
+ source_file_ = NULL;
+}
+
+TEST_F(TestLibYuv, ConvertSanityTest) {
+ // TODO(mikhal)
+}
+
+TEST_F(TestLibYuv, ConvertTest) {
+ // Reading YUV frame - testing on the first frame of the foreman sequence
+ int j = 0;
+ std::string output_file_name = webrtc::test::OutputPath() +
+ "LibYuvTest_conversion.yuv";
+ FILE* output_file = fopen(output_file_name.c_str(), "wb");
+ ASSERT_TRUE(output_file != NULL);
+
+ double psnr = 0;
+
+ uint8_t* orig_buffer = new uint8_t[frame_length_];
+ EXPECT_GT(fread(orig_buffer, 1, frame_length_, source_file_), 0U);
+
+ // printf("\nConvert #%d I420 <-> RGB24\n", j);
+ uint8_t* res_rgb_buffer2 = new uint8_t[width_ * height_ * 3];
+ uint8_t* res_i420_buffer = new uint8_t[frame_length_];
+
+ EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_, kRGB24, 0,
+ width_, height_, res_rgb_buffer2));
+
+ EXPECT_EQ(0, ConvertToI420(kRGB24, res_rgb_buffer2, 0, 0, width_, height_,
+ 0, width_, height_, width_, kRotateNone,
+ res_i420_buffer));
+
+ fwrite(res_i420_buffer, frame_length_, 1, output_file);
+ psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
+ // Optimization Speed- quality trade-off => 45 dB only (platform dependant).
+ EXPECT_GT(ceil(psnr), 44);
+ j++;
+ delete [] res_rgb_buffer2;
+
+ // printf("\nConvert #%d I420 <-> UYVY\n", j);
+ uint8_t* out_uyvy_buffer = new uint8_t[width_ * height_ * 2];
+ EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_,
+ kUYVY, 0, width_, height_, out_uyvy_buffer));
+ EXPECT_EQ(0, ConvertToI420(kUYVY, out_uyvy_buffer, 0, 0, width_, height_,
+ 0, width_, height_, width_,kRotateNone, res_i420_buffer));
+ psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
+ EXPECT_EQ(48.0, psnr);
+ fwrite(res_i420_buffer, frame_length_, 1, output_file);
+
+ j++;
+ delete [] out_uyvy_buffer;
+
+ // printf("\nConvert #%d I420 <-> I420 \n", j);
+ uint8_t* out_i420_buffer = new uint8_t[width_ * height_ * 3 / 2 ];
+ EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer, 0, 0, width_, height_,
+ 0, width_, height_, width_,
+ kRotateNone, out_i420_buffer));
+ EXPECT_EQ(0, ConvertFromI420(out_i420_buffer, width_, kI420, 0,
+ width_, height_, res_i420_buffer));
+ fwrite(res_i420_buffer, frame_length_, 1, output_file);
+ psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
+ EXPECT_EQ(48.0, psnr);
+ j++;
+ delete [] out_i420_buffer;
+
+ // printf("\nConvert #%d I420 <-> YV12\n", j);
+ uint8_t* outYV120Buffer = new uint8_t[frame_length_];
+
+ EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_, kYV12, 0,
+ width_, height_, outYV120Buffer));
+ EXPECT_EQ(0, ConvertFromYV12(outYV120Buffer, width_,
+ kI420, 0,
+ width_, height_,
+ res_i420_buffer));
+ fwrite(res_i420_buffer, frame_length_, 1, output_file);
+
+ psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
+ EXPECT_EQ(48.0, psnr);
+ j++;
+ delete [] outYV120Buffer;
+
+ // printf("\nConvert #%d I420 <-> YUY2\n", j);
+ uint8_t* out_yuy2_buffer = new uint8_t[width_ * height_ * 2];
+ EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_,
+ kYUY2, 0, width_, height_, out_yuy2_buffer));
+
+ EXPECT_EQ(0, ConvertToI420(kYUY2, out_yuy2_buffer, 0, 0, width_, height_,
+ 0, width_, height_, width_,
+ kRotateNone, res_i420_buffer));
+
+ fwrite(res_i420_buffer, frame_length_, 1, output_file);
+ psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
+ EXPECT_EQ(48.0, psnr);
+ ASSERT_EQ(0, fclose(output_file));
+ delete [] out_yuy2_buffer;
+ delete [] res_i420_buffer;
+ delete [] orig_buffer;
+}
+
+TEST_F(TestLibYuv, MirrorTest) {
+ // TODO (mikhal): Add an automated test to confirm output.
+ std::string str;
+ int width = 16;
+ int height = 8;
+ int factor_y = 1;
+ int factor_u = 1;
+ int factor_v = 1;
+ int start_buffer_offset = 10;
+ int length = webrtc::CalcBufferSize(kI420, width, height);
+
+ uint8_t* test_frame = new uint8_t[length];
+ memset(test_frame, 255, length);
+
+ // Create input frame
+ uint8_t* in_frame = test_frame;
+ uint8_t* in_frame_cb = in_frame + width * height;
+ uint8_t* in_frame_cr = in_frame_cb + (width * height) / 4;
+ CreateImage(width, height, in_frame, 10, factor_y, 1); // Y
+ CreateImage(width / 2, height / 2, in_frame_cb, 100, factor_u, 1); // Cb
+ CreateImage(width / 2, height / 2, in_frame_cr, 200, factor_v, 1); // Cr
+ EXPECT_EQ(0, PrintFrame(test_frame, width, height, "InputFrame"));
+
+ uint8_t* test_frame2 = new uint8_t[length + start_buffer_offset * 2];
+ memset(test_frame2, 255, length + start_buffer_offset * 2);
+ uint8_t* out_frame = test_frame2;
+
+ // LeftRight
+ std::cout << "Test Mirror function: LeftRight" << std::endl;
+ EXPECT_EQ(0, MirrorI420LeftRight(in_frame, out_frame, width, height));
+ EXPECT_EQ(0, PrintFrame(test_frame2, width, height, "OutputFrame"));
+ EXPECT_EQ(0, MirrorI420LeftRight(out_frame, test_frame, width, height));
+
+ EXPECT_EQ(0, memcmp(in_frame, test_frame, length));
+
+ // UpDown
+ std::cout << "Test Mirror function: UpDown" << std::endl;
+ EXPECT_EQ(0, MirrorI420UpDown(in_frame, out_frame, width, height));
+ EXPECT_EQ(0, PrintFrame(test_frame2, width, height, "OutputFrame"));
+ EXPECT_EQ(0, MirrorI420UpDown(out_frame, test_frame, width, height));
+
+ EXPECT_EQ(0, memcmp(in_frame, test_frame, length));
+
+ // TODO(mikhal): Write to a file, and ask to look at the file.
+
+ std::cout << "Do the mirrored frames look correct?" << std::endl;
+ delete [] test_frame;
+ delete [] test_frame2;
+}
+
+} // namespace
diff --git a/trunk/src/common_video/libyuv/scaler.cc b/trunk/src/common_video/libyuv/scaler.cc
new file mode 100644
index 0000000..a12462c
--- /dev/null
+++ b/trunk/src/common_video/libyuv/scaler.cc
@@ -0,0 +1,105 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_video/libyuv/include/scaler.h"
+
+// LibYuv
+#ifdef WEBRTC_ANDROID
+#include "libyuv/files/include/libyuv.h"
+#else
+#include "third_party/libyuv/include/libyuv.h"
+#endif
+
+namespace webrtc {
+
+Scaler::Scaler()
+ : method_(kScaleBox),
+ src_width_(0),
+ src_height_(0),
+ dst_width_(0),
+ dst_height_(0),
+ set_(false) {}
+
+Scaler::~Scaler() {}
+
+int Scaler::Set(int src_width, int src_height,
+ int dst_width, int dst_height,
+ VideoType src_video_type, VideoType dst_video_type,
+ ScaleMethod method) {
+ set_ = false;
+ if (src_width < 1 || src_height < 1 || dst_width < 1 || dst_height < 1)
+ return -1;
+
+ if (!SupportedVideoType(src_video_type, dst_video_type))
+ return -1;
+
+ src_width_ = src_width;
+ src_height_ = src_height;
+ dst_width_ = dst_width;
+ dst_height_ = dst_height;
+ method_ = method;
+ set_ = true;
+ return 0;
+}
+
+int Scaler::Scale(const uint8_t* src_frame,
+ uint8_t*& dst_frame,
+ int& dst_size) {
+ if (src_frame == NULL)
+ return -1;
+ if (!set_)
+ return -2;
+
+ // Making sure that destination frame is of sufficient size
+ int required_dst_size = dst_width_ * dst_height_ * 3 / 2;
+ if (dst_frame && required_dst_size > dst_size) {
+ // allocated buffer is too small
+ delete [] dst_frame;
+ dst_frame = NULL;
+ }
+ if (dst_frame == NULL) {
+ dst_frame = new uint8_t[required_dst_size];
+ dst_size = required_dst_size;
+ }
+
+ // Converting to planes:
+ const uint8_t* src_yplane = src_frame;
+ const uint8_t* src_uplane = src_frame + src_width_ * src_height_;
+ const uint8_t* src_vplane = src_uplane + src_width_ * src_height_ / 4;
+
+ uint8_t* dst_yplane = dst_frame;
+ uint8_t* dst_uplane = dst_frame + dst_width_ * dst_height_;
+ uint8_t* dst_vplane = dst_uplane + dst_width_ * dst_height_ / 4;
+
+ return libyuv::I420Scale(src_yplane, src_width_,
+ src_uplane, src_width_ / 2,
+ src_vplane, src_width_ / 2,
+ src_width_, src_height_,
+ dst_yplane, dst_width_,
+ dst_uplane, dst_width_ / 2,
+ dst_vplane, dst_width_ / 2,
+ dst_width_, dst_height_,
+ libyuv::FilterMode(method_));
+}
+
+// TODO(mikhal): Add support for more types.
+bool Scaler::SupportedVideoType(VideoType src_video_type,
+ VideoType dst_video_type) {
+ if (src_video_type != dst_video_type)
+ return false;
+
+ if ((src_video_type == kI420) || (src_video_type == kIYUV) ||
+ (src_video_type == kYV12))
+ return true;
+
+ return false;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/scaler_unittest.cc b/trunk/src/common_video/libyuv/scaler_unittest.cc
new file mode 100644
index 0000000..899ad89
--- /dev/null
+++ b/trunk/src/common_video/libyuv/scaler_unittest.cc
@@ -0,0 +1,239 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <string.h>
+
+#include "common_video/libyuv/include/scaler.h"
+#include "gtest/gtest.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+class TestScaler : public ::testing::Test {
+ protected:
+ TestScaler();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ void ScaleSequence(ScaleMethod method,
+ FILE* source_file, std::string out_name,
+ int src_width, int src_height,
+ int dst_width, int dst_height);
+
+ Scaler test_scaler_;
+ FILE* source_file_;
+ uint8_t* test_buffer_;
+ const int width_;
+ const int height_;
+ const int frame_length_;
+};
+
+
+// TODO (mikhal): Use scoped_ptr when handling buffers.
+TestScaler::TestScaler()
+ : source_file_(NULL),
+ width_(352),
+ height_(288),
+ frame_length_(CalcBufferSize(kI420, 352, 288)) {
+}
+
+void TestScaler::SetUp() {
+ const std::string input_file_name =
+ webrtc::test::ResourcePath("foreman_cif", "yuv");
+ source_file_ = fopen(input_file_name.c_str(), "rb");
+ ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
+ input_file_name << "\n";
+ test_buffer_ = new uint8_t[frame_length_];
+}
+
+void TestScaler::TearDown() {
+ if (source_file_ != NULL) {
+ ASSERT_EQ(0, fclose(source_file_));
+ }
+ source_file_ = NULL;
+ delete [] test_buffer_;
+}
+
+TEST_F(TestScaler, ScaleWithoutSettingValues) {
+ int size = 100;
+ EXPECT_EQ(-2, test_scaler_.Scale(test_buffer_, test_buffer_, size));
+}
+
+TEST_F(TestScaler, ScaleBadInitialValues) {
+ EXPECT_EQ(-1, test_scaler_.Set(0, 288, 352, 288, kI420, kI420, kScalePoint));
+ EXPECT_EQ(-1, test_scaler_.Set(704, 0, 352, 288, kI420, kI420, kScaleBox));
+ EXPECT_EQ(-1, test_scaler_.Set(704, 576, 352, 0, kI420, kI420,
+ kScaleBilinear));
+ EXPECT_EQ(-1, test_scaler_.Set(704, 576, 0, 288, kI420, kI420, kScalePoint));
+}
+
+TEST_F(TestScaler, ScaleSendingNullSourcePointer) {
+ int size = 0;
+ EXPECT_EQ(-1, test_scaler_.Scale(NULL, test_buffer_, size));
+}
+
+TEST_F(TestScaler, ScaleSendingBufferTooSmall) {
+ // Sending a buffer which is too small (should reallocate and update size)
+ EXPECT_EQ(0, test_scaler_.Set(352, 288, 144, 288, kI420, kI420, kScalePoint));
+ uint8_t* test_buffer2 = NULL;
+ int size = 0;
+ EXPECT_GT(fread(test_buffer_, 1, frame_length_, source_file_), 0U);
+ EXPECT_EQ(0, test_scaler_.Scale(test_buffer_, test_buffer2, size));
+ EXPECT_EQ(144 * 288 * 3 / 2, size);
+ delete [] test_buffer2;
+}
+
+//TODO (mikhal): Converge the test into one function that accepts the method.
+TEST_F(TestScaler, PointScaleTest) {
+ ScaleMethod method = kScalePoint;
+ std::string out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_PointScale_176_144.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ / 2, height_ / 2);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_320_240.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 320, 240);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_704_576.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ * 2, height_ * 2);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_300_200.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 300, 200);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_400_300.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 400, 300);
+}
+
+TEST_F(TestScaler, BiLinearScaleTest) {
+ ScaleMethod method = kScaleBilinear;
+ std::string out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BilinearScale_176_144.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ / 2, height_ / 2);
+ out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BilinearScale_320_240.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 320, 240);
+ out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BilinearScale_704_576.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ * 2, height_ * 2);
+ out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BilinearScale_300_200.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 300, 200);
+ out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BilinearScale_400_300.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 400, 300);
+}
+
+TEST_F(TestScaler, BoxScaleTest) {
+ ScaleMethod method = kScaleBox;
+ std::string out_name = webrtc::test::OutputPath() +
+ "LibYuvTest_BoxScale_176_144.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ / 2, height_ / 2);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_320_240.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 320, 240);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_704_576.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ width_ * 2, height_ * 2);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_300_200.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 300, 200);
+ out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_400_300.yuv";
+ ScaleSequence(method,
+ source_file_, out_name,
+ width_, height_,
+ 400, 300);
+}
+
+// TODO (mikhal): Move part to a separate scale test.
+void TestScaler::ScaleSequence(ScaleMethod method,
+ FILE* source_file, std::string out_name,
+ int src_width, int src_height,
+ int dst_width, int dst_height) {
+ FILE* output_file;
+ EXPECT_EQ(0, test_scaler_.Set(src_width, src_height,
+ dst_width, dst_height,
+ kI420, kI420, method));
+
+ output_file = fopen(out_name.c_str(), "wb");
+ ASSERT_TRUE(output_file != NULL);
+
+ rewind(source_file);
+
+ int out_required_size = dst_width * dst_height * 3 / 2;
+ int in_required_size = src_height * src_width * 3 / 2;
+ uint8_t* input_buffer = new uint8_t[in_required_size];
+ uint8_t* output_buffer = new uint8_t[out_required_size];
+
+ int64_t start_clock, total_clock;
+ total_clock = 0;
+ int frame_count = 0;
+
+ // Running through entire sequence
+ while (feof(source_file) == 0) {
+ if ((size_t)in_required_size !=
+ fread(input_buffer, 1, in_required_size, source_file))
+ break;
+
+ start_clock = TickTime::MillisecondTimestamp();
+ EXPECT_EQ(0, test_scaler_.Scale(input_buffer, output_buffer,
+ out_required_size));
+ total_clock += TickTime::MillisecondTimestamp() - start_clock;
+ fwrite(output_buffer, out_required_size, 1, output_file);
+ frame_count++;
+ }
+
+ if (frame_count) {
+ printf("Scaling[%d %d] => [%d %d]: ",
+ src_width, src_height, dst_width, dst_height);
+ printf("Average time per frame[ms]: %.2lf\n",
+ (static_cast<double>(total_clock) / frame_count));
+ }
+ ASSERT_EQ(0, fclose(output_file));
+ delete [] input_buffer;
+ delete [] output_buffer;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/engine_configurations.h b/trunk/src/engine_configurations.h
new file mode 100644
index 0000000..49a2f8c
--- /dev/null
+++ b/trunk/src/engine_configurations.h
@@ -0,0 +1,149 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_ENGINE_CONFIGURATIONS_H_
+#define WEBRTC_ENGINE_CONFIGURATIONS_H_
+
+// ============================================================================
+// Voice and Video
+// ============================================================================
+
+// Don't link in socket support in Chrome
+#ifdef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_EXTERNAL_TRANSPORT
+#endif
+
+// Optional to enable stand-alone
+// #define WEBRTC_EXTERNAL_TRANSPORT
+
+// ----------------------------------------------------------------------------
+// [Voice] Codec settings
+// ----------------------------------------------------------------------------
+
+#define WEBRTC_CODEC_ISAC // floating-point iSAC implementation (default)
+// #define WEBRTC_CODEC_ISACFX // fix-point iSAC implementation
+#define WEBRTC_CODEC_AVT
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_CODEC_ILBC
+#define WEBRTC_CODEC_G722
+#define WEBRTC_CODEC_PCM16
+#define WEBRTC_CODEC_RED
+#endif
+
+// ----------------------------------------------------------------------------
+// [Video] Codec settings
+// ----------------------------------------------------------------------------
+
+#define VIDEOCODEC_I420
+#define VIDEOCODEC_VP8
+
+// ============================================================================
+// VoiceEngine
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Settings for VoiceEngine
+// ----------------------------------------------------------------------------
+
+#define WEBRTC_VOICE_ENGINE_AGC // Near-end AGC
+#define WEBRTC_VOICE_ENGINE_ECHO // Near-end AEC
+#define WEBRTC_VOICE_ENGINE_NR // Near-end NS
+#define WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION // Typing detection
+#endif
+
+// ----------------------------------------------------------------------------
+// VoiceEngine sub-APIs
+// ----------------------------------------------------------------------------
+
+#define WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+#define WEBRTC_VOICE_ENGINE_CODEC_API
+#define WEBRTC_VOICE_ENGINE_DTMF_API
+#define WEBRTC_VOICE_ENGINE_HARDWARE_API
+#define WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+#define WEBRTC_VOICE_ENGINE_NETWORK_API
+#define WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+#define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+#define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+#define WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+#define WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+#define WEBRTC_VOICE_ENGINE_FILE_API
+#endif
+
+// ============================================================================
+// VideoEngine
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Settings for special VideoEngine configurations
+// ----------------------------------------------------------------------------
+// ----------------------------------------------------------------------------
+// VideoEngine sub-API:s
+// ----------------------------------------------------------------------------
+
+#define WEBRTC_VIDEO_ENGINE_CAPTURE_API
+#define WEBRTC_VIDEO_ENGINE_CODEC_API
+#define WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+#define WEBRTC_VIDEO_ENGINE_NETWORK_API
+#define WEBRTC_VIDEO_ENGINE_RENDER_API
+#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+// #define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_VIDEO_ENGINE_FILE_API
+#endif
+
+// ============================================================================
+// Platform specific configurations
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// VideoEngine Windows
+// ----------------------------------------------------------------------------
+
+#if defined(_WIN32)
+ // #define DIRECTDRAW_RENDERING
+ #define DIRECT3D9_RENDERING // Requires DirectX 9.
+#endif
+
+// ----------------------------------------------------------------------------
+// VideoEngine MAC
+// ----------------------------------------------------------------------------
+
+#if defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
+ // #define CARBON_RENDERING
+ #define COCOA_RENDERING
+#endif
+
+// ----------------------------------------------------------------------------
+// VideoEngine Mobile iPhone
+// ----------------------------------------------------------------------------
+
+#if defined(MAC_IPHONE)
+ #define EAGL_RENDERING
+#endif
+
+// ----------------------------------------------------------------------------
+// Deprecated
+// ----------------------------------------------------------------------------
+
+// #define WEBRTC_CODEC_G729
+// #define WEBRTC_DTMF_DETECTION
+// #define WEBRTC_SRTP
+// #define WEBRTC_SRTP_ALLOW_ROC_ITERATION
+
+#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
diff --git a/trunk/src/modules/audio_coding/codecs/OWNERS b/trunk/src/modules/audio_coding/codecs/OWNERS
new file mode 100644
index 0000000..e1e6256
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/OWNERS
@@ -0,0 +1,3 @@
+tina.legrand@webrtc.org
+turaj@webrtc.org
+jan.skoglund@webrtc.org
diff --git a/trunk/src/modules/audio_coding/codecs/cng/Android.mk b/trunk/src/modules/audio_coding/codecs/cng/Android.mk
new file mode 100644
index 0000000..32671cc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_cng
+LOCAL_MODULE_TAGS := optional
+LOCAL_GENERATED_SOURCES :=
+LOCAL_SRC_FILES := \
+ webrtc_cng.c \
+ cng_helpfuns.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng.gypi b/trunk/src/modules/audio_coding/codecs/cng/cng.gypi
new file mode 100644
index 0000000..2903cc2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/cng.gypi
@@ -0,0 +1,57 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'CNG',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/webrtc_cng.h',
+ 'webrtc_cng.c',
+ 'cng_helpfuns.c',
+ 'cng_helpfuns.h',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'cng_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'CNG',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'cng_unittest.cc',
+ ],
+ }, # CNG_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c
new file mode 100644
index 0000000..2e9029f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include "webrtc_cng.h"
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "cng_helpfuns.h"
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+void WebRtcCng_K2a16(
+ WebRtc_Word16 *k, /* Q15. */
+ int useOrder,
+ WebRtc_Word16 *a /* Q12. */
+)
+{
+ WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER+1];
+ WebRtc_Word16 *aptr, *aptr2, *anyptr;
+ G_CONST WebRtc_Word16 *kptr;
+ int m, i;
+
+ kptr = k;
+ *a = 4096; /* i.e., (Word16_MAX >> 3)+1 */
+ *any = *a;
+ a[1] = (*k+4) >> 3;
+ for( m=1; m<useOrder; m++ )
+ {
+ kptr++;
+ aptr = a;
+ aptr++;
+ aptr2 = &a[m];
+ anyptr = any;
+ anyptr++;
+
+ any[m+1] = (*kptr+4) >> 3;
+ for( i=0; i<m; i++ ) {
+ *anyptr++ = (*aptr++) + (WebRtc_Word16)( (( (WebRtc_Word32)(*aptr2--) * (WebRtc_Word32)*kptr )+16384) >> 15);
+ }
+
+ aptr = a;
+ anyptr = any;
+ for( i=0; i<(m+2); i++ ){
+ *aptr++ = *anyptr++;
+ }
+ }
+}
+
+
+#ifdef __cplusplus
+}
+#endif
+
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h
new file mode 100644
index 0000000..fd8d6dc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
+
+extern WebRtc_Word32 lpc_lagwinTbl_fixw32[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+void WebRtcCng_K2a16(WebRtc_Word16 *k, int useOrder, WebRtc_Word16 *a);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc b/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc
new file mode 100644
index 0000000..6a4edc0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "webrtc_cng.h"
+#include "gtest/gtest.h"
+
+TEST(CngTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h
new file mode 100644
index 0000000..d405e3a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h
@@ -0,0 +1,236 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
+
+#include "typedefs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define WEBRTC_CNG_MAX_LPC_ORDER 12
+#define WEBRTC_CNG_MAX_OUTSIZE_ORDER 640
+
+/* Define Error codes */
+
+/* 6100 Encoder */
+#define CNG_ENCODER_MEMORY_ALLOCATION_FAILED 6110
+#define CNG_ENCODER_NOT_INITIATED 6120
+#define CNG_DISALLOWED_LPC_ORDER 6130
+#define CNG_DISALLOWED_FRAME_SIZE 6140
+#define CNG_DISALLOWED_SAMPLING_FREQUENCY 6150
+/* 6200 Decoder */
+#define CNG_DECODER_MEMORY_ALLOCATION_FAILED 6210
+#define CNG_DECODER_NOT_INITIATED 6220
+
+
+typedef struct WebRtcCngEncInst CNG_enc_inst;
+typedef struct WebRtcCngDecInst CNG_dec_inst;
+
+
+/****************************************************************************
+ * WebRtcCng_Version(...)
+ *
+ * These functions returns the version name (string must be at least
+ * 500 characters long)
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_Version(char *version);
+
+/****************************************************************************
+ * WebRtcCng_AssignSizeEnc/Dec(...)
+ *
+ * These functions get the size needed for storing the instance for encoder
+ * and decoder, respectively
+ *
+ * Input/Output:
+ * - sizeinbytes : Pointer to integer where the size is returned
+ *
+ * Return value : 0
+ */
+
+WebRtc_Word16 WebRtcCng_AssignSizeEnc(int *sizeinbytes);
+WebRtc_Word16 WebRtcCng_AssignSizeDec(int *sizeinbytes);
+
+
+/****************************************************************************
+ * WebRtcCng_AssignEnc/Dec(...)
+ *
+ * These functions Assignes memory for the instances.
+ *
+ * Input:
+ * - CNG_inst_Addr : Adress to where to assign memory
+ * Output:
+ * - inst : Pointer to the instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_AssignEnc(CNG_enc_inst **inst, void *CNG_inst_Addr);
+WebRtc_Word16 WebRtcCng_AssignDec(CNG_dec_inst **inst, void *CNG_inst_Addr);
+
+
+/****************************************************************************
+ * WebRtcCng_CreateEnc/Dec(...)
+ *
+ * These functions create an instance to the specified structure
+ *
+ * Input:
+ * - XXX_inst : Pointer to created instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_CreateEnc(CNG_enc_inst **cng_inst);
+WebRtc_Word16 WebRtcCng_CreateDec(CNG_dec_inst **cng_inst);
+
+
+/****************************************************************************
+ * WebRtcCng_InitEnc/Dec(...)
+ *
+ * This function initializes a instance
+ *
+ * Input:
+ * - cng_inst : Instance that should be initialized
+ *
+ * - fs : 8000 for narrowband and 16000 for wideband
+ * - interval : generate SID data every interval ms
+ * - quality : Number of refl. coefs, maximum allowed is 12
+ *
+ * Output:
+ * - cng_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_InitEnc(CNG_enc_inst *cng_inst,
+ WebRtc_Word16 fs,
+ WebRtc_Word16 interval,
+ WebRtc_Word16 quality);
+WebRtc_Word16 WebRtcCng_InitDec(CNG_dec_inst *cng_dec_inst);
+
+
+/****************************************************************************
+ * WebRtcCng_FreeEnc/Dec(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+
+WebRtc_Word16 WebRtcCng_FreeEnc(CNG_enc_inst *cng_inst);
+WebRtc_Word16 WebRtcCng_FreeDec(CNG_dec_inst *cng_inst);
+
+
+
+/****************************************************************************
+ * WebRtcCng_Encode(...)
+ *
+ * These functions analyzes background noise
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance
+ * - speech : Signal to be analyzed
+ * - nrOfSamples : Size of speech vector
+ * - forceSID : not zero to force SID frame and reset
+ *
+ * Output:
+ * - bytesOut : Nr of bytes to transmit, might be 0
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_Encode(CNG_enc_inst *cng_inst,
+ WebRtc_Word16 *speech,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_UWord8* SIDdata,
+ WebRtc_Word16 *bytesOut,
+ WebRtc_Word16 forceSID);
+
+
+/****************************************************************************
+ * WebRtcCng_UpdateSid(...)
+ *
+ * These functions updates the CN state, when a new SID packet arrives
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ * - SID : SID packet, all headers removed
+ * - length : Length in bytes of SID packet
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcCng_UpdateSid(CNG_dec_inst *cng_inst,
+ WebRtc_UWord8 *SID,
+ WebRtc_Word16 length);
+
+
+/****************************************************************************
+ * WebRtcCng_Generate(...)
+ *
+ * These functions generates CN data when needed
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ * - outData : pointer to area to write CN data
+ * - nrOfSamples : How much data to generate
+ * - new_period : >0 if a new period of CNG, will reset history
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcCng_Generate(CNG_dec_inst *cng_inst,
+ WebRtc_Word16 * outData,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_Word16 new_period);
+
+
+/*****************************************************************************
+ * WebRtcCng_GetErrorCodeEnc/Dec(...)
+ *
+ * This functions can be used to check the error code of a CNG instance. When
+ * a function returns -1 a error code will be set for that instance. The
+ * function below extract the code of the last error that occurred in the
+ * specified instance.
+ *
+ * Input:
+ * - CNG_inst : CNG enc/dec instance
+ *
+ * Return value : Error code
+ */
+
+WebRtc_Word16 WebRtcCng_GetErrorCodeEnc(CNG_enc_inst *cng_inst);
+WebRtc_Word16 WebRtcCng_GetErrorCodeDec(CNG_dec_inst *cng_inst);
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc b/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc
new file mode 100644
index 0000000..b247831
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc
@@ -0,0 +1,225 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * CNG.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+#include "stdafx.h"
+#include "webrtc_cng.h"
+#include "webrtc_vad.h"
+
+CNG_enc_inst *e_inst;
+CNG_dec_inst *d_inst;
+
+VadInst *vinst;
+//#define ASSIGN
+
+short anaSpeech[WEBRTC_CNG_MAX_OUTSIZE_ORDER], genSpeech[WEBRTC_CNG_MAX_OUTSIZE_ORDER], state[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
+unsigned char SIDpkt[114];
+
+int main(int argc, char* argv[])
+{
+ FILE * infile, *outfile, *statefile;
+ short res=0,errtype;
+ /*float time=0.0;*/
+
+ WebRtcVad_Create(&vinst);
+ WebRtcVad_Init(vinst);
+
+ short size;
+ int samps=0;
+
+ if (argc < 5){
+ printf("Usage:\n CNG.exe infile outfile samplingfreq(Hz) interval(ms) order\n\n");
+ return(0);
+ }
+
+ infile=fopen(argv[1],"rb");
+ if (infile==NULL){
+ printf("file %s does not exist\n",argv[1]);
+ return(0);
+ }
+ outfile=fopen(argv[2],"wb");
+ statefile=fopen("CNGVAD.d","wb");
+ if (outfile==NULL){
+ printf("file %s could not be created\n",argv[2]);
+ return(0);
+ }
+
+ unsigned int fs=16000;
+ short frameLen=fs/50;
+
+#ifndef ASSIGN
+ res=WebRtcCng_CreateEnc(&e_inst);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ res=WebRtcCng_CreateDec(&d_inst);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeDec(d_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+#else
+
+ // Test the Assign-functions
+ int Esize, Dsize;
+ void *Eaddr, *Daddr;
+
+ res=WebRtcCng_AssignSizeEnc(&Esize);
+ res=WebRtcCng_AssignSizeDec(&Dsize);
+ Eaddr=malloc(Esize);
+ Daddr=malloc(Dsize);
+
+ res=WebRtcCng_AssignEnc(&e_inst, Eaddr);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+ res=WebRtcCng_AssignDec(&d_inst, Daddr);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeDec(d_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+#endif
+
+ res=WebRtcCng_InitEnc(e_inst,atoi(argv[3]),atoi(argv[4]),atoi(argv[5]));
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+ res=WebRtcCng_InitDec(d_inst);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeDec(d_inst);
+ fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+
+ static bool firstSilent=true;
+
+ int numSamp=0;
+ int speech=0;
+ int silent=0;
+ long cnt=0;
+
+ while(fread(anaSpeech,2,frameLen,infile)==frameLen){
+
+ cnt++;
+ if (cnt==60){
+ cnt=60;
+ }
+ /* time+=(float)frameLen/fs;
+ numSamp+=frameLen;
+ float temp[640];
+ for(unsigned int j=0;j<frameLen;j++)
+ temp[j]=(float)anaSpeech[j]; */
+
+ // if(!WebRtcVad_Process(vinst, fs, anaSpeech, frameLen)){
+
+
+ if(1){ // Do CNG coding of entire file
+
+ // if(!((anaSpeech[0]==0)&&(anaSpeech[1]==0)&&(anaSpeech[2]==0))){
+ if(firstSilent){
+ res = WebRtcCng_Encode(e_inst, anaSpeech, frameLen/2, SIDpkt,&size,1);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+
+ firstSilent=false;
+
+ res=WebRtcCng_Encode(e_inst, &anaSpeech[frameLen/2], frameLen/2, SIDpkt,&size,1);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+
+ }
+ else{
+ res=WebRtcCng_Encode(e_inst, anaSpeech, frameLen/2, SIDpkt,&size,0);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ res=WebRtcCng_Encode(e_inst, &anaSpeech[frameLen/2], frameLen/2, SIDpkt,&size,0);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
+ fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ }
+
+ if(size>0){
+ res=WebRtcCng_UpdateSid(d_inst,SIDpkt, size);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeDec(d_inst);
+ fprintf(stderr,"\n\n Error in decoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ }
+ res=WebRtcCng_Generate(d_inst,genSpeech, frameLen,0);
+ if (res < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcCng_GetErrorCodeDec(d_inst);
+ fprintf(stderr,"\n\n Error in decoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ memcpy(state,anaSpeech,2*frameLen);
+ }
+ else{
+ firstSilent=true;
+ memcpy(genSpeech,anaSpeech,2*frameLen);
+
+ memset(anaSpeech,0,frameLen*2);
+ memset(state,0,frameLen*2);
+
+ }
+ fwrite(genSpeech,2,frameLen,outfile);
+ fwrite(state,2,frameLen,statefile);
+
+ }
+
+ fclose(infile);
+ fclose(outfile);
+ fclose(statefile);
+ return 0;
+}
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc
new file mode 100644
index 0000000..995e510
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// stdafx.cpp : source file that includes just the standard includes
+// CNG.pch will be the pre-compiled header
+// stdafx.obj will contain the pre-compiled type information
+
+#include "stdafx.h"
+
+// TODO: reference any additional headers you need in STDAFX.H
+// and not in this file
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h
new file mode 100644
index 0000000..dd6c445
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// stdafx.h : include file for standard system include files,
+// or project specific include files that are used frequently, but
+// are changed infrequently
+//
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_TEST_STDAFX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_TEST_STDAFX_H_
+
+#if _MSC_VER > 1000
+#pragma once
+#endif // _MSC_VER > 1000
+
+#define WIN32_LEAN_AND_MEAN // Exclude rarely-used stuff from Windows headers
+
+#include <stdio.h>
+
+// TODO: reference additional headers your program requires here
+
+//{{AFX_INSERT_LOCATION}}
+// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
+
+#endif // !defined(AFX_STDAFX_H__DE2097A7_569B_42A0_A615_41BF352D6FFB__INCLUDED_)
diff --git a/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c b/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c
new file mode 100644
index 0000000..65f6672
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -0,0 +1,732 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include <string.h>
+#include <stdlib.h>
+
+#include "webrtc_cng.h"
+#include "signal_processing_library.h"
+#include "cng_helpfuns.h"
+#include "stdio.h"
+
+
+typedef struct WebRtcCngDecInst_t_ {
+
+ WebRtc_UWord32 dec_seed;
+ WebRtc_Word32 dec_target_energy;
+ WebRtc_Word32 dec_used_energy;
+ WebRtc_Word16 dec_target_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_used_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_filtstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_filtstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_Efiltstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_EfiltstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 dec_order;
+ WebRtc_Word16 dec_target_scale_factor; /*Q29*/
+ WebRtc_Word16 dec_used_scale_factor; /*Q29*/
+ WebRtc_Word16 target_scale_factor; /* Q13 */
+ WebRtc_Word16 errorcode;
+ WebRtc_Word16 initflag;
+
+} WebRtcCngDecInst_t;
+
+
+typedef struct WebRtcCngEncInst_t_ {
+
+ WebRtc_Word16 enc_nrOfCoefs;
+ WebRtc_Word16 enc_sampfreq;
+ WebRtc_Word16 enc_interval;
+ WebRtc_Word16 enc_msSinceSID;
+ WebRtc_Word32 enc_Energy;
+ WebRtc_Word16 enc_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word32 enc_corrVector[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 enc_filtstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 enc_filtstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_UWord32 enc_seed;
+ WebRtc_Word16 errorcode;
+ WebRtc_Word16 initflag;
+
+} WebRtcCngEncInst_t;
+
+const WebRtc_Word32 WebRtcCng_kDbov[94]={
+ 1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
+ 271562548, 215709799, 171344384, 136103682, 108110997, 85875618,
+ 68213428, 54183852, 43039763, 34187699, 27156255, 21570980,
+ 17134438, 13610368, 10811100, 8587562, 6821343, 5418385,
+ 4303976, 3418770, 2715625, 2157098, 1713444, 1361037,
+ 1081110, 858756, 682134, 541839, 430398, 341877,
+ 271563, 215710, 171344, 136104, 108111, 85876,
+ 68213, 54184, 43040, 34188, 27156, 21571,
+ 17134, 13610, 10811, 8588, 6821, 5418,
+ 4304, 3419, 2716, 2157, 1713, 1361,
+ 1081, 859, 682, 542, 430, 342,
+ 272, 216, 171, 136, 108, 86,
+ 68, 54, 43, 34, 27, 22,
+ 17, 14, 11, 9, 7, 5,
+ 4, 3, 3, 2, 2, 1,
+ 1, 1, 1, 1
+};
+const WebRtc_Word16 WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
+ 32702, 32636, 32570, 32505, 32439, 32374,
+ 32309, 32244, 32179, 32114, 32049, 31985
+};
+
+/****************************************************************************
+ * WebRtcCng_Version(...)
+ *
+ * These functions returns the version name (string must be at least
+ * 500 characters long)
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_Version(char *version)
+{
+ strcpy((char*)version,(const char*)"1.2.0\n");
+ return(0);
+}
+
+
+/****************************************************************************
+ * WebRtcCng_AssignSizeEnc/Dec(...)
+ *
+ * These functions get the size needed for storing the instance for encoder
+ * and decoder, respectively
+ *
+ * Input/Output:
+ * - sizeinbytes : Pointer to integer where the size is returned
+ *
+ * Return value : 0
+ */
+
+WebRtc_Word16 WebRtcCng_AssignSizeEnc(int *sizeinbytes)
+{
+ *sizeinbytes=sizeof(WebRtcCngEncInst_t)*2/sizeof(WebRtc_Word16);
+ return(0);
+}
+
+WebRtc_Word16 WebRtcCng_AssignSizeDec(int *sizeinbytes)
+{
+ *sizeinbytes=sizeof(WebRtcCngDecInst_t)*2/sizeof(WebRtc_Word16);
+ return(0);
+}
+
+
+/****************************************************************************
+ * WebRtcCng_AssignEnc/Dec(...)
+ *
+ * These functions Assignes memory for the instances.
+ *
+ * Input:
+ * - CNG_inst_Addr : Adress to where to assign memory
+ * Output:
+ * - inst : Pointer to the instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_AssignEnc(CNG_enc_inst **inst, void *CNG_inst_Addr)
+{
+ if (CNG_inst_Addr!=NULL) {
+ *inst = (CNG_enc_inst*)CNG_inst_Addr;
+ (*(WebRtcCngEncInst_t**) inst)->errorcode = 0;
+ (*(WebRtcCngEncInst_t**) inst)->initflag = 0;
+ return(0);
+ } else {
+ /* The memory could not be allocated */
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcCng_AssignDec(CNG_dec_inst **inst, void *CNG_inst_Addr)
+{
+ if (CNG_inst_Addr!=NULL) {
+ *inst = (CNG_dec_inst*)CNG_inst_Addr;
+ (*(WebRtcCngDecInst_t**) inst)->errorcode = 0;
+ (*(WebRtcCngDecInst_t**) inst)->initflag = 0;
+ return(0);
+ } else {
+ /* The memory could not be allocated */
+ return(-1);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcCng_CreateEnc/Dec(...)
+ *
+ * These functions create an instance to the specified structure
+ *
+ * Input:
+ * - XXX_inst : Pointer to created instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_CreateEnc(CNG_enc_inst **cng_inst)
+{
+ *cng_inst=(CNG_enc_inst*)malloc(sizeof(WebRtcCngEncInst_t));
+ if(cng_inst!=NULL) {
+ (*(WebRtcCngEncInst_t**) cng_inst)->errorcode = 0;
+ (*(WebRtcCngEncInst_t**) cng_inst)->initflag = 0;
+ return(0);
+ }
+ else {
+ /* The memory could not be allocated */
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcCng_CreateDec(CNG_dec_inst **cng_inst)
+{
+ *cng_inst=(CNG_dec_inst*)malloc(sizeof(WebRtcCngDecInst_t));
+ if(cng_inst!=NULL) {
+ (*(WebRtcCngDecInst_t**) cng_inst)->errorcode = 0;
+ (*(WebRtcCngDecInst_t**) cng_inst)->initflag = 0;
+ return(0);
+ }
+ else {
+ /* The memory could not be allocated */
+ return(-1);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcCng_InitEnc/Dec(...)
+ *
+ * This function initializes a instance
+ *
+ * Input:
+ * - cng_inst : Instance that should be initialized
+ *
+ * - fs : 8000 for narrowband and 16000 for wideband
+ * - interval : generate SID data every interval ms
+ * - quality : TBD
+ *
+ * Output:
+ * - cng_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+
+WebRtc_Word16 WebRtcCng_InitEnc(CNG_enc_inst *cng_inst,
+ WebRtc_Word16 fs,
+ WebRtc_Word16 interval,
+ WebRtc_Word16 quality)
+{
+ int i;
+
+ WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
+
+ memset(inst, 0, sizeof(WebRtcCngEncInst_t));
+
+ /* Check LPC order */
+
+ if (quality>WEBRTC_CNG_MAX_LPC_ORDER) {
+ inst->errorcode = CNG_DISALLOWED_LPC_ORDER;
+ return (-1);
+ }
+
+ if (fs<=0) {
+ inst->errorcode = CNG_DISALLOWED_SAMPLING_FREQUENCY;
+ return (-1);
+ }
+
+ inst->enc_sampfreq=fs;
+ inst->enc_interval=interval;
+ inst->enc_nrOfCoefs=quality;
+ inst->enc_msSinceSID=0;
+ inst->enc_seed=7777; /*For debugging only*/
+ inst->enc_Energy=0;
+ for(i=0;i<(WEBRTC_CNG_MAX_LPC_ORDER+1);i++){
+ inst->enc_reflCoefs[i]=0;
+ inst->enc_corrVector[i]=0;
+ }
+ inst->initflag=1;
+
+ return(0);
+}
+
+WebRtc_Word16 WebRtcCng_InitDec(CNG_dec_inst *cng_inst)
+{
+ int i;
+
+ WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
+
+ memset(inst, 0, sizeof(WebRtcCngDecInst_t));
+ inst->dec_seed=7777; /*For debugging only*/
+ inst->dec_order=5;
+ inst->dec_target_scale_factor=0;
+ inst->dec_used_scale_factor=0;
+ for(i=0;i<(WEBRTC_CNG_MAX_LPC_ORDER+1);i++){
+ inst->dec_filtstate[i]=0;
+ inst->dec_target_reflCoefs[i]=0;
+ inst->dec_used_reflCoefs[i]=0;
+ }
+ inst->dec_target_reflCoefs[0]=0;
+ inst->dec_used_reflCoefs[0]=0;
+ inst ->dec_used_energy=0;
+ inst->initflag=1;
+
+ return(0);
+}
+
+/****************************************************************************
+ * WebRtcCng_FreeEnc/Dec(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+
+WebRtc_Word16 WebRtcCng_FreeEnc(CNG_enc_inst *cng_inst)
+{
+ free(cng_inst);
+ return(0);
+}
+
+WebRtc_Word16 WebRtcCng_FreeDec(CNG_dec_inst *cng_inst)
+{
+ free(cng_inst);
+ return(0);
+}
+
+
+
+/****************************************************************************
+ * WebRtcCng_Encode(...)
+ *
+ * These functions analyzes background noise
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance
+ * - speech : Signal (noise) to be analyzed
+ * - nrOfSamples : Size of speech vector
+ * - bytesOut : Nr of bytes to transmit, might be 0
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcCng_Encode(CNG_enc_inst *cng_inst,
+ WebRtc_Word16 *speech,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_UWord8* SIDdata,
+ WebRtc_Word16* bytesOut,
+ WebRtc_Word16 forceSID)
+{
+ WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
+
+ WebRtc_Word16 arCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word32 corrVector[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 refCs[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 hanningW[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
+ WebRtc_Word16 ReflBeta=19661; /*0.6 in q15*/
+ WebRtc_Word16 ReflBetaComp=13107; /*0.4 in q15*/
+ WebRtc_Word32 outEnergy;
+ int outShifts;
+ int i, stab;
+ int acorrScale;
+ int index;
+ WebRtc_Word16 ind,factor;
+ WebRtc_Word32 *bptr, blo, bhi;
+ WebRtc_Word16 negate;
+ const WebRtc_Word16 *aptr;
+
+ WebRtc_Word16 speechBuf[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
+
+
+ /* check if encoder initiated */
+ if (inst->initflag != 1) {
+ inst->errorcode = CNG_ENCODER_NOT_INITIATED;
+ return (-1);
+ }
+
+
+ /* check framesize */
+ if (nrOfSamples>WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
+ inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
+ return (-1);
+ }
+
+
+ for(i=0;i<nrOfSamples;i++){
+ speechBuf[i]=speech[i];
+ }
+
+ factor=nrOfSamples;
+
+ /* Calculate energy and a coefficients */
+ outEnergy =WebRtcSpl_Energy(speechBuf, nrOfSamples, &outShifts);
+ while(outShifts>0){
+ if(outShifts>5){ /*We can only do 5 shifts without destroying accuracy in division factor*/
+ outEnergy<<=(outShifts-5);
+ outShifts=5;
+ }
+ else{
+ factor/=2;
+ outShifts--;
+ }
+ }
+ outEnergy=WebRtcSpl_DivW32W16(outEnergy,factor);
+
+ if (outEnergy > 1){
+ /* Create Hanning Window */
+ WebRtcSpl_GetHanningWindow(hanningW, nrOfSamples/2);
+ for( i=0;i<(nrOfSamples/2);i++ )
+ hanningW[nrOfSamples-i-1]=hanningW[i];
+
+ WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, nrOfSamples, 14);
+
+ WebRtcSpl_AutoCorrelation( speechBuf, nrOfSamples, inst->enc_nrOfCoefs, corrVector, &acorrScale );
+
+ if( *corrVector==0 )
+ *corrVector = WEBRTC_SPL_WORD16_MAX;
+
+ /* Adds the bandwidth expansion */
+ aptr = WebRtcCng_kCorrWindow;
+ bptr = corrVector;
+
+ // (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700)
+ for( ind=0; ind<inst->enc_nrOfCoefs; ind++ )
+ {
+ // The below code multiplies the 16 b corrWindow values (Q15) with
+ // the 32 b corrvector (Q0) and shifts the result down 15 steps.
+
+ negate = *bptr<0;
+ if( negate )
+ *bptr = -*bptr;
+
+ blo = (WebRtc_Word32)*aptr * (*bptr & 0xffff);
+ bhi = ((blo >> 16) & 0xffff) + ((WebRtc_Word32)(*aptr++) * ((*bptr >> 16) & 0xffff));
+ blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
+
+ *bptr = (( (bhi>>16) & 0x7fff) << 17) | ((WebRtc_UWord32)blo >> 15);
+ if( negate )
+ *bptr = -*bptr;
+ bptr++;
+ }
+
+ // end of bandwidth expansion
+
+ stab=WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, inst->enc_nrOfCoefs);
+
+ if(!stab){
+ // disregard from this frame
+ *bytesOut=0;
+ return(0);
+ }
+
+ }
+ else {
+ for(i=0;i<inst->enc_nrOfCoefs; i++)
+ refCs[i]=0;
+ }
+
+ if(forceSID){
+ /*Read instantaneous values instead of averaged*/
+ for(i=0;i<inst->enc_nrOfCoefs;i++)
+ inst->enc_reflCoefs[i]=refCs[i];
+ inst->enc_Energy=outEnergy;
+ }
+ else{
+ /*Average history with new values*/
+ for(i=0;i<(inst->enc_nrOfCoefs);i++){
+ inst->enc_reflCoefs[i]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->enc_reflCoefs[i],ReflBeta,15);
+ inst->enc_reflCoefs[i]+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i],ReflBetaComp,15);
+ }
+ inst->enc_Energy=(outEnergy>>2)+(inst->enc_Energy>>1)+(inst->enc_Energy>>2);
+ }
+
+
+ if(inst->enc_Energy<1){
+ inst->enc_Energy=1;
+ }
+
+ if((inst->enc_msSinceSID>(inst->enc_interval-1))||forceSID){
+
+ /* Search for best dbov value */
+ index=0;
+ for(i=1;i<93;i++){
+ /* Always round downwards */
+ if((inst->enc_Energy-WebRtcCng_kDbov[i])>0){
+ index=i;
+ break;
+ }
+ }
+ if((i==93)&&(index==0))
+ index=94;
+ SIDdata[0]=index;
+
+
+ /* Quantize coefs with tweak for WebRtc implementation of RFC3389 */
+ if(inst->enc_nrOfCoefs==WEBRTC_CNG_MAX_LPC_ORDER){
+ for(i=0;i<inst->enc_nrOfCoefs;i++){
+ SIDdata[i+1]=((inst->enc_reflCoefs[i]+128)>>8); /* Q15 to Q7*/ /* +127 */
+ }
+ }else{
+ for(i=0;i<inst->enc_nrOfCoefs;i++){
+ SIDdata[i+1]=(127+((inst->enc_reflCoefs[i]+128)>>8)); /* Q15 to Q7*/ /* +127 */
+ }
+ }
+
+ inst->enc_msSinceSID=0;
+ *bytesOut=inst->enc_nrOfCoefs+1;
+
+ inst->enc_msSinceSID+=(1000*nrOfSamples)/inst->enc_sampfreq;
+ return(inst->enc_nrOfCoefs+1);
+ }else{
+ inst->enc_msSinceSID+=(1000*nrOfSamples)/inst->enc_sampfreq;
+ *bytesOut=0;
+ return(0);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcCng_UpdateSid(...)
+ *
+ * These functions updates the CN state, when a new SID packet arrives
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ * - SID : SID packet, all headers removed
+ * - length : Length in bytes of SID packet
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcCng_UpdateSid(CNG_dec_inst *cng_inst,
+ WebRtc_UWord8 *SID,
+ WebRtc_Word16 length)
+{
+
+ WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
+ WebRtc_Word16 refCs[WEBRTC_CNG_MAX_LPC_ORDER];
+ WebRtc_Word32 targetEnergy;
+ int i;
+
+ if (inst->initflag != 1) {
+ inst->errorcode = CNG_DECODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ /*Throw away reflection coefficients of higher order than we can handle*/
+ if(length> (WEBRTC_CNG_MAX_LPC_ORDER+1))
+ length=WEBRTC_CNG_MAX_LPC_ORDER+1;
+
+ inst->dec_order=length-1;
+
+ if(SID[0]>93)
+ SID[0]=93;
+ targetEnergy=WebRtcCng_kDbov[SID[0]];
+ /* Take down target energy to 75% */
+ targetEnergy=targetEnergy>>1;
+ targetEnergy+=targetEnergy>>2;
+
+ inst->dec_target_energy=targetEnergy;
+
+ /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389 */
+ if(inst->dec_order==WEBRTC_CNG_MAX_LPC_ORDER){
+ for(i=0;i<(inst->dec_order);i++){
+ refCs[i]=SID[i+1]<<8; /* Q7 to Q15*/
+ inst->dec_target_reflCoefs[i]=refCs[i];
+ }
+ }else{
+ for(i=0;i<(inst->dec_order);i++){
+ refCs[i]=(SID[i+1]-127)<<8; /* Q7 to Q15*/
+ inst->dec_target_reflCoefs[i]=refCs[i];
+ }
+ }
+
+ for(i=(inst->dec_order);i<WEBRTC_CNG_MAX_LPC_ORDER;i++){
+ refCs[i]=0;
+ inst->dec_target_reflCoefs[i]=refCs[i];
+ }
+
+ return(0);
+}
+
+
+/****************************************************************************
+ * WebRtcCng_Generate(...)
+ *
+ * These functions generates CN data when needed
+ *
+ * Input:
+ * - cng_inst : Pointer to created instance that should be freed
+ * - outData : pointer to area to write CN data
+ * - nrOfSamples : How much data to generate
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcCng_Generate(CNG_dec_inst *cng_inst,
+ WebRtc_Word16 *outData,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_Word16 new_period)
+{
+ WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
+
+ int i;
+ WebRtc_Word16 excitation[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
+ WebRtc_Word16 low[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
+ WebRtc_Word16 lpPoly[WEBRTC_CNG_MAX_LPC_ORDER+1];
+ WebRtc_Word16 ReflBetaStd=26214; /*0.8 in q15*/
+ WebRtc_Word16 ReflBetaCompStd=6553; /*0.2in q15*/
+ WebRtc_Word16 ReflBetaNewP=19661; /*0.6 in q15*/
+ WebRtc_Word16 ReflBetaCompNewP=13107; /*0.4 in q15*/
+ WebRtc_Word16 Beta,BetaC, tmp1, tmp2, tmp3;
+ WebRtc_Word32 targetEnergy;
+ WebRtc_Word16 En;
+ WebRtc_Word16 temp16;
+
+ if (nrOfSamples>WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
+ inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
+ return (-1);
+ }
+
+
+ if (new_period) {
+ inst->dec_used_scale_factor=inst->dec_target_scale_factor;
+ Beta=ReflBetaNewP;
+ BetaC=ReflBetaCompNewP;
+ } else {
+ Beta=ReflBetaStd;
+ BetaC=ReflBetaCompStd;
+ }
+
+ /*Here we use a 0.5 weighting, should possibly be modified to 0.6*/
+ tmp1=inst->dec_used_scale_factor<<2; /* Q13->Q15 */
+ tmp2=inst->dec_target_scale_factor<<2; /* Q13->Q15 */
+ tmp3=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp1,Beta,15);
+ tmp3+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp2,BetaC,15);
+ inst->dec_used_scale_factor=tmp3>>2; /* Q15->Q13 */
+
+ inst->dec_used_energy=inst->dec_used_energy>>1;
+ inst->dec_used_energy+=inst->dec_target_energy>>1;
+
+
+ /* Do the same for the reflection coeffs */
+ for (i=0;i<WEBRTC_CNG_MAX_LPC_ORDER;i++) {
+ inst->dec_used_reflCoefs[i]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_used_reflCoefs[i],Beta,15);
+ inst->dec_used_reflCoefs[i]+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_target_reflCoefs[i],BetaC,15);
+ }
+
+ /* Compute the polynomial coefficients */
+ WebRtcCng_K2a16(inst->dec_used_reflCoefs, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
+
+ /***/
+
+ targetEnergy=inst->dec_used_energy;
+
+ // Calculate scaling factor based on filter energy
+ En=8192; //1.0 in Q13
+ for (i=0; i<(WEBRTC_CNG_MAX_LPC_ORDER); i++) {
+
+ // Floating point value for reference
+ // E*=1.0-((float)inst->dec_used_reflCoefs[i]/32768.0)*((float)inst->dec_used_reflCoefs[i]/32768.0);
+
+ // Same in fixed point
+ temp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_used_reflCoefs[i],inst->dec_used_reflCoefs[i],15); // K(i).^2 in Q15
+ temp16=0x7fff - temp16; // 1 - K(i).^2 in Q15
+ En=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(En,temp16,15);
+
+ }
+
+ //float scaling= sqrt(E*inst->dec_target_energy/((1<<24)));
+
+ //Calculate sqrt(En*target_energy/exctiation energy)
+
+ targetEnergy=WebRtcSpl_Sqrt(inst->dec_used_energy);
+
+ En=(WebRtc_Word16)WebRtcSpl_Sqrt(En)<<6; //We are missing a factor sqrt(2) here
+ En=(En*3)>>1; //1.5 estimates sqrt(2)
+
+ inst->dec_used_scale_factor=(WebRtc_Word16)((En*targetEnergy)>>12);
+
+
+ /***/
+
+ /*Generate excitation*/
+ /*Excitation energy per sample is 2.^24 - Q13 N(0,1) */
+ for(i=0;i<nrOfSamples;i++){
+ excitation[i]=WebRtcSpl_RandN(&inst->dec_seed)>>1;
+ }
+
+ /*Scale to correct energy*/
+ WebRtcSpl_ScaleVector(excitation, excitation, inst->dec_used_scale_factor, nrOfSamples, 13);
+
+ WebRtcSpl_FilterAR(
+ lpPoly, /* Coefficients in Q12 */
+ WEBRTC_CNG_MAX_LPC_ORDER+1,
+ excitation, /* Speech samples */
+ nrOfSamples,
+ inst->dec_filtstate, /* State preservation */
+ WEBRTC_CNG_MAX_LPC_ORDER,
+ inst->dec_filtstateLow, /* State preservation */
+ WEBRTC_CNG_MAX_LPC_ORDER,
+ outData, /* Filtered speech samples */
+ low,
+ nrOfSamples
+ );
+
+ return(0);
+
+}
+
+
+
+/****************************************************************************
+ * WebRtcCng_GetErrorCodeEnc/Dec(...)
+ *
+ * This functions can be used to check the error code of a CNG instance. When
+ * a function returns -1 a error code will be set for that instance. The
+ * function below extract the code of the last error that occured in the
+ * specified instance.
+ *
+ * Input:
+ * - CNG_inst : CNG enc/dec instance
+ *
+ * Return value : Error code
+ */
+
+WebRtc_Word16 WebRtcCng_GetErrorCodeEnc(CNG_enc_inst *cng_inst)
+{
+
+ /* typecast pointer to real structure */
+ WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
+
+ return inst->errorcode;
+}
+
+WebRtc_Word16 WebRtcCng_GetErrorCodeDec(CNG_dec_inst *cng_inst)
+{
+
+ /* typecast pointer to real structure */
+ WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
+
+ return inst->errorcode;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/Android.mk b/trunk/src/modules/audio_coding/codecs/g711/Android.mk
new file mode 100644
index 0000000..779bb7c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_g711
+LOCAL_MODULE_TAGS := optional
+LOCAL_GENERATED_SOURCES :=
+LOCAL_SRC_FILES := \
+ g711_interface.c \
+ g711.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../../..
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.c b/trunk/src/modules/audio_coding/codecs/g711/g711.c
new file mode 100644
index 0000000..954f377
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/g711.c
@@ -0,0 +1,83 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.c - A-law and u-law transcoding routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place this code in the
+ * public domain for the benefit of all mankind - even the slimy
+ * ones who might try to proprietize my work and use it to my
+ * detriment.
+ *
+ * $Id: g711.c,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed unused include files
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+/*! \file */
+
+#include "g711.h"
+#include "typedefs.h"
+
+/* Copied from the CCITT G.711 specification */
+static const WebRtc_UWord8 ulaw_to_alaw_table[256] =
+{
+ 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36, 37,
+ 58, 59, 56, 57, 62, 63, 60, 61, 50, 51, 48, 49, 54, 55, 52, 53,
+ 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6, 7, 4, 26,
+ 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17, 22, 23, 20, 21, 106,
+ 104, 105, 110, 111, 108, 109, 98, 99, 96, 97, 102, 103, 100, 101, 122, 120,
+ 126, 127, 124, 125, 114, 115, 112, 113, 118, 119, 116, 117, 75, 73, 79, 77,
+ 66, 67, 64, 65, 70, 71, 68, 69, 90, 91, 88, 89, 94, 95, 92, 93,
+ 82, 82, 83, 83, 80, 80, 81, 81, 86, 86, 87, 87, 84, 84, 85, 85,
+ 170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
+ 186, 187, 184, 185, 190, 191, 188, 189, 178, 179, 176, 177, 182, 183, 180, 181,
+ 138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 154,
+ 155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 234,
+ 232, 233, 238, 239, 236, 237, 226, 227, 224, 225, 230, 231, 228, 229, 250, 248,
+ 254, 255, 252, 253, 242, 243, 240, 241, 246, 247, 244, 245, 203, 201, 207, 205,
+ 194, 195, 192, 193, 198, 199, 196, 197, 218, 219, 216, 217, 222, 223, 220, 221,
+ 210, 210, 211, 211, 208, 208, 209, 209, 214, 214, 215, 215, 212, 212, 213, 213
+};
+
+/* These transcoding tables are copied from the CCITT G.711 specification. To achieve
+ optimal results, do not change them. */
+
+static const WebRtc_UWord8 alaw_to_ulaw_table[256] =
+{
+ 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36, 37,
+ 57, 58, 55, 56, 61, 62, 59, 60, 49, 50, 47, 48, 53, 54, 51, 52,
+ 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6, 7, 4, 5,
+ 26, 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17, 22, 23, 20, 21,
+ 98, 99, 96, 97, 102, 103, 100, 101, 93, 93, 92, 92, 95, 95, 94, 94,
+ 116, 118, 112, 114, 124, 126, 120, 122, 106, 107, 104, 105, 110, 111, 108, 109,
+ 72, 73, 70, 71, 76, 77, 74, 75, 64, 65, 63, 63, 68, 69, 66, 67,
+ 86, 87, 84, 85, 90, 91, 88, 89, 79, 79, 78, 78, 82, 83, 80, 81,
+ 170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
+ 185, 186, 183, 184, 189, 190, 187, 188, 177, 178, 175, 176, 181, 182, 179, 180,
+ 138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 133,
+ 154, 155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149,
+ 226, 227, 224, 225, 230, 231, 228, 229, 221, 221, 220, 220, 223, 223, 222, 222,
+ 244, 246, 240, 242, 252, 254, 248, 250, 234, 235, 232, 233, 238, 239, 236, 237,
+ 200, 201, 198, 199, 204, 205, 202, 203, 192, 193, 191, 191, 196, 197, 194, 195,
+ 214, 215, 212, 213, 218, 219, 216, 217, 207, 207, 206, 206, 210, 211, 208, 209
+};
+
+WebRtc_UWord8 alaw_to_ulaw(WebRtc_UWord8 alaw)
+{
+ return alaw_to_ulaw_table[alaw];
+}
+/*- End of function --------------------------------------------------------*/
+
+WebRtc_UWord8 ulaw_to_alaw(WebRtc_UWord8 ulaw)
+{
+ return ulaw_to_alaw_table[ulaw];
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.gypi b/trunk/src/modules/audio_coding/codecs/g711/g711.gypi
new file mode 100644
index 0000000..763e6db
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/g711.gypi
@@ -0,0 +1,65 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'G711',
+ 'type': '<(library)',
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/g711_interface.h',
+ 'g711_interface.c',
+ 'g711.c',
+ 'g711.h',
+ ],
+ },
+ ], # targets
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'g711_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'G711',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'g711_unittest.cc',
+ ],
+ },
+ {
+ 'target_name': 'g711_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'G711',
+ ],
+ 'sources': [
+ 'test/testG711.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.h b/trunk/src/modules/audio_coding/codecs/g711/g711.h
new file mode 100644
index 0000000..cd5e3d7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/g711.h
@@ -0,0 +1,382 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.h - In line A-law and u-law conversion routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place this code in the
+ * public domain for the benefit of all mankind - even the slimy
+ * ones who might try to proprietize my work and use it to my
+ * detriment.
+ *
+ * $Id: g711.h,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Two changes to make implementation bitexact with ITU-T reference implementation
+ */
+
+/*! \file */
+
+/*! \page g711_page A-law and mu-law handling
+Lookup tables for A-law and u-law look attractive, until you consider the impact
+on the CPU cache. If it causes a substantial area of your processor cache to get
+hit too often, cache sloshing will severely slow things down. The main reason
+these routines are slow in C, is the lack of direct access to the CPU's "find
+the first 1" instruction. A little in-line assembler fixes that, and the
+conversion routines can be faster than lookup tables, in most real world usage.
+A "find the first 1" instruction is available on most modern CPUs, and is a
+much underused feature.
+
+If an assembly language method of bit searching is not available, these routines
+revert to a method that can be a little slow, so the cache thrashing might not
+seem so bad :(
+
+Feel free to submit patches to add fast "find the first 1" support for your own
+favourite processor.
+
+Look up tables are used for transcoding between A-law and u-law, since it is
+difficult to achieve the precise transcoding procedure laid down in the G.711
+specification by other means.
+*/
+
+#if !defined(_G711_H_)
+#define _G711_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "typedefs.h"
+
+#if defined(__i386__)
+/*! \brief Find the bit position of the highest set bit in a word
+ \param bits The word to be searched
+ \return The bit number of the highest set bit, or -1 if the word is zero. */
+static __inline__ int top_bit(unsigned int bits)
+{
+ int res;
+
+ __asm__ __volatile__(" movl $-1,%%edx;\n"
+ " bsrl %%eax,%%edx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the bit position of the lowest set bit in a word
+ \param bits The word to be searched
+ \return The bit number of the lowest set bit, or -1 if the word is zero. */
+static __inline__ int bottom_bit(unsigned int bits)
+{
+ int res;
+
+ __asm__ __volatile__(" movl $-1,%%edx;\n"
+ " bsfl %%eax,%%edx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+/*- End of function --------------------------------------------------------*/
+#elif defined(__x86_64__)
+static __inline__ int top_bit(unsigned int bits)
+{
+ int res;
+
+ __asm__ __volatile__(" movq $-1,%%rdx;\n"
+ " bsrq %%rax,%%rdx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int bottom_bit(unsigned int bits)
+{
+ int res;
+
+ __asm__ __volatile__(" movq $-1,%%rdx;\n"
+ " bsfq %%rax,%%rdx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+/*- End of function --------------------------------------------------------*/
+#else
+static __inline int top_bit(unsigned int bits)
+{
+ int i;
+
+ if (bits == 0)
+ return -1;
+ i = 0;
+ if (bits & 0xFFFF0000)
+ {
+ bits &= 0xFFFF0000;
+ i += 16;
+ }
+ if (bits & 0xFF00FF00)
+ {
+ bits &= 0xFF00FF00;
+ i += 8;
+ }
+ if (bits & 0xF0F0F0F0)
+ {
+ bits &= 0xF0F0F0F0;
+ i += 4;
+ }
+ if (bits & 0xCCCCCCCC)
+ {
+ bits &= 0xCCCCCCCC;
+ i += 2;
+ }
+ if (bits & 0xAAAAAAAA)
+ {
+ bits &= 0xAAAAAAAA;
+ i += 1;
+ }
+ return i;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline int bottom_bit(unsigned int bits)
+{
+ int i;
+
+ if (bits == 0)
+ return -1;
+ i = 32;
+ if (bits & 0x0000FFFF)
+ {
+ bits &= 0x0000FFFF;
+ i -= 16;
+ }
+ if (bits & 0x00FF00FF)
+ {
+ bits &= 0x00FF00FF;
+ i -= 8;
+ }
+ if (bits & 0x0F0F0F0F)
+ {
+ bits &= 0x0F0F0F0F;
+ i -= 4;
+ }
+ if (bits & 0x33333333)
+ {
+ bits &= 0x33333333;
+ i -= 2;
+ }
+ if (bits & 0x55555555)
+ {
+ bits &= 0x55555555;
+ i -= 1;
+ }
+ return i;
+}
+/*- End of function --------------------------------------------------------*/
+#endif
+
+/* N.B. It is tempting to use look-up tables for A-law and u-law conversion.
+ * However, you should consider the cache footprint.
+ *
+ * A 64K byte table for linear to x-law and a 512 byte table for x-law to
+ * linear sound like peanuts these days, and shouldn't an array lookup be
+ * real fast? No! When the cache sloshes as badly as this one will, a tight
+ * calculation may be better. The messiest part is normally finding the
+ * segment, but a little inline assembly can fix that on an i386, x86_64 and
+ * many other modern processors.
+ */
+
+/*
+ * Mu-law is basically as follows:
+ *
+ * Biased Linear Input Code Compressed Code
+ * ------------------------ ---------------
+ * 00000001wxyza 000wxyz
+ * 0000001wxyzab 001wxyz
+ * 000001wxyzabc 010wxyz
+ * 00001wxyzabcd 011wxyz
+ * 0001wxyzabcde 100wxyz
+ * 001wxyzabcdef 101wxyz
+ * 01wxyzabcdefg 110wxyz
+ * 1wxyzabcdefgh 111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz. * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+//#define ULAW_ZEROTRAP /* turn on the trap as per the MIL-STD */
+#define ULAW_BIAS 0x84 /* Bias for linear code. */
+
+/*! \brief Encode a linear sample to u-law
+ \param linear The sample to encode.
+ \return The u-law value.
+*/
+static __inline WebRtc_UWord8 linear_to_ulaw(int linear)
+{
+ WebRtc_UWord8 u_val;
+ int mask;
+ int seg;
+
+ /* Get the sign and the magnitude of the value. */
+ if (linear < 0)
+ {
+ /* WebRtc, tlegrand: -1 added to get bitexact to reference implementation */
+ linear = ULAW_BIAS - linear - 1;
+ mask = 0x7F;
+ }
+ else
+ {
+ linear = ULAW_BIAS + linear;
+ mask = 0xFF;
+ }
+
+ seg = top_bit(linear | 0xFF) - 7;
+
+ /*
+ * Combine the sign, segment, quantization bits,
+ * and complement the code word.
+ */
+ if (seg >= 8)
+ u_val = (WebRtc_UWord8) (0x7F ^ mask);
+ else
+ u_val = (WebRtc_UWord8) (((seg << 4) | ((linear >> (seg + 3)) & 0xF)) ^ mask);
+#ifdef ULAW_ZEROTRAP
+ /* Optional ITU trap */
+ if (u_val == 0)
+ u_val = 0x02;
+#endif
+ return u_val;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Decode an u-law sample to a linear value.
+ \param ulaw The u-law sample to decode.
+ \return The linear value.
+*/
+static __inline WebRtc_Word16 ulaw_to_linear(WebRtc_UWord8 ulaw)
+{
+ int t;
+
+ /* Complement to obtain normal u-law value. */
+ ulaw = ~ulaw;
+ /*
+ * Extract and bias the quantization bits. Then
+ * shift up by the segment number and subtract out the bias.
+ */
+ t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int) ulaw & 0x70) >> 4);
+ return (WebRtc_Word16) ((ulaw & 0x80) ? (ULAW_BIAS - t) : (t - ULAW_BIAS));
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * A-law is basically as follows:
+ *
+ * Linear Input Code Compressed Code
+ * ----------------- ---------------
+ * 0000000wxyza 000wxyz
+ * 0000001wxyza 001wxyz
+ * 000001wxyzab 010wxyz
+ * 00001wxyzabc 011wxyz
+ * 0001wxyzabcd 100wxyz
+ * 001wxyzabcde 101wxyz
+ * 01wxyzabcdef 110wxyz
+ * 1wxyzabcdefg 111wxyz
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+#define ALAW_AMI_MASK 0x55
+
+/*! \brief Encode a linear sample to A-law
+ \param linear The sample to encode.
+ \return The A-law value.
+*/
+static __inline WebRtc_UWord8 linear_to_alaw(int linear)
+{
+ int mask;
+ int seg;
+
+ if (linear >= 0)
+ {
+ /* Sign (bit 7) bit = 1 */
+ mask = ALAW_AMI_MASK | 0x80;
+ }
+ else
+ {
+ /* Sign (bit 7) bit = 0 */
+ mask = ALAW_AMI_MASK;
+ /* WebRtc, tlegrand: Changed from -8 to -1 to get bitexact to reference
+ * implementation */
+ linear = -linear - 1;
+ }
+
+ /* Convert the scaled magnitude to segment number. */
+ seg = top_bit(linear | 0xFF) - 7;
+ if (seg >= 8)
+ {
+ if (linear >= 0)
+ {
+ /* Out of range. Return maximum value. */
+ return (WebRtc_UWord8) (0x7F ^ mask);
+ }
+ /* We must be just a tiny step below zero */
+ return (WebRtc_UWord8) (0x00 ^ mask);
+ }
+ /* Combine the sign, segment, and quantization bits. */
+ return (WebRtc_UWord8) (((seg << 4) | ((linear >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^ mask);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Decode an A-law sample to a linear value.
+ \param alaw The A-law sample to decode.
+ \return The linear value.
+*/
+static __inline WebRtc_Word16 alaw_to_linear(WebRtc_UWord8 alaw)
+{
+ int i;
+ int seg;
+
+ alaw ^= ALAW_AMI_MASK;
+ i = ((alaw & 0x0F) << 4);
+ seg = (((int) alaw & 0x70) >> 4);
+ if (seg)
+ i = (i + 0x108) << (seg - 1);
+ else
+ i += 8;
+ return (WebRtc_Word16) ((alaw & 0x80) ? i : -i);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Transcode from A-law to u-law, using the procedure defined in G.711.
+ \param alaw The A-law sample to transcode.
+ \return The best matching u-law value.
+*/
+WebRtc_UWord8 alaw_to_ulaw(WebRtc_UWord8 alaw);
+
+/*! \brief Transcode from u-law to A-law, using the procedure defined in G.711.
+ \param alaw The u-law sample to transcode.
+ \return The best matching A-law value.
+*/
+WebRtc_UWord8 ulaw_to_alaw(WebRtc_UWord8 ulaw);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c b/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c
new file mode 100644
index 0000000..a49abdb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c
@@ -0,0 +1,171 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include <string.h>
+#include "g711.h"
+#include "g711_interface.h"
+#include "typedefs.h"
+
+WebRtc_Word16 WebRtcG711_EncodeA(void *state,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded)
+{
+ int n;
+ WebRtc_UWord16 tempVal, tempVal2;
+
+ // Set and discard to avoid getting warnings
+ (void)(state = NULL);
+
+ // Sanity check of input length
+ if (len < 0) {
+ return (-1);
+ }
+
+ // Loop over all samples
+ for (n = 0; n < len; n++) {
+ tempVal = (WebRtc_UWord16)linear_to_alaw(speechIn[n]);
+
+#ifdef WEBRTC_BIG_ENDIAN
+ if ((n & 0x1) == 1) {
+ encoded[n>>1]|=((WebRtc_UWord16)tempVal);
+ } else {
+ encoded[n>>1]=((WebRtc_UWord16)tempVal)<<8;
+ }
+#else
+ if ((n & 0x1) == 1) {
+ tempVal2 |= ((WebRtc_UWord16) tempVal) << 8;
+ encoded[n >> 1] |= ((WebRtc_UWord16) tempVal) << 8;
+ } else {
+ tempVal2 = ((WebRtc_UWord16) tempVal);
+ encoded[n >> 1] = ((WebRtc_UWord16) tempVal);
+ }
+#endif
+ }
+ return (len);
+}
+
+WebRtc_Word16 WebRtcG711_EncodeU(void *state,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded)
+{
+ int n;
+ WebRtc_UWord16 tempVal;
+
+ // Set and discard to avoid getting warnings
+ (void)(state = NULL);
+
+ // Sanity check of input length
+ if (len < 0) {
+ return (-1);
+ }
+
+ // Loop over all samples
+ for (n = 0; n < len; n++) {
+ tempVal = (WebRtc_UWord16)linear_to_ulaw(speechIn[n]);
+
+ #ifdef WEBRTC_BIG_ENDIAN
+ if ((n & 0x1) == 1) {
+ encoded[n>>1]|=((WebRtc_UWord16)tempVal);
+ } else {
+ encoded[n>>1]=((WebRtc_UWord16)tempVal)<<8;
+ }
+ #else
+ if ((n & 0x1) == 1) {
+ encoded[n >> 1] |= ((WebRtc_UWord16) tempVal) << 8;
+ } else {
+ encoded[n >> 1] = ((WebRtc_UWord16) tempVal);
+ }
+ #endif
+ }
+ return (len);
+}
+
+WebRtc_Word16 WebRtcG711_DecodeA(void *state,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ int n;
+ WebRtc_UWord16 tempVal;
+
+ // Set and discard to avoid getting warnings
+ (void)(state = NULL);
+
+ // Sanity check of input length
+ if (len < 0) {
+ return (-1);
+ }
+
+ for (n = 0; n < len; n++) {
+ #ifdef WEBRTC_BIG_ENDIAN
+ if ((n & 0x1) == 1) {
+ tempVal=((WebRtc_UWord16)encoded[n>>1] & 0xFF);
+ } else {
+ tempVal=((WebRtc_UWord16)encoded[n>>1] >> 8);
+ }
+ #else
+ if ((n & 0x1) == 1) {
+ tempVal = (encoded[n >> 1] >> 8);
+ } else {
+ tempVal = (encoded[n >> 1] & 0xFF);
+ }
+ #endif
+ decoded[n] = (WebRtc_Word16) alaw_to_linear(tempVal);
+ }
+
+ *speechType = 1;
+ return (len);
+}
+
+WebRtc_Word16 WebRtcG711_DecodeU(void *state,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ int n;
+ WebRtc_UWord16 tempVal;
+
+ // Set and discard to avoid getting warnings
+ (void)(state = NULL);
+
+ // Sanity check of input length
+ if (len < 0) {
+ return (-1);
+ }
+
+ for (n = 0; n < len; n++) {
+ #ifdef WEBRTC_BIG_ENDIAN
+ if ((n & 0x1) == 1) {
+ tempVal=((WebRtc_UWord16)encoded[n>>1] & 0xFF);
+ } else {
+ tempVal=((WebRtc_UWord16)encoded[n>>1] >> 8);
+ }
+ #else
+ if ((n & 0x1) == 1) {
+ tempVal = (encoded[n >> 1] >> 8);
+ } else {
+ tempVal = (encoded[n >> 1] & 0xFF);
+ }
+ #endif
+ decoded[n] = (WebRtc_Word16) ulaw_to_linear(tempVal);
+ }
+
+ *speechType = 1;
+ return (len);
+}
+
+WebRtc_Word16 WebRtcG711_Version(char* version, WebRtc_Word16 lenBytes)
+{
+ strncpy(version, "2.0.0", lenBytes);
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc b/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc
new file mode 100644
index 0000000..c903bed
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "g711_interface.h"
+#include "gtest/gtest.h"
+
+TEST(G711Test, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h b/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h
new file mode 100644
index 0000000..25a9903
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h
@@ -0,0 +1,148 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
+
+#include "typedefs.h"
+
+// Comfort noise constants
+#define G711_WEBRTC_SPEECH 1
+#define G711_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG711_EncodeA(...)
+ *
+ * This function encodes a G711 A-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ * - state : Dummy state to make this codec look more like
+ * other codecs
+ * - speechIn : Input speech vector
+ * - len : Samples in speechIn
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG711_EncodeA(void *state,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded);
+
+/****************************************************************************
+ * WebRtcG711_EncodeU(...)
+ *
+ * This function encodes a G711 U-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ * - state : Dummy state to make this codec look more like
+ * other codecs
+ * - speechIn : Input speech vector
+ * - len : Samples in speechIn
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG711_EncodeU(void *state,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded);
+
+/****************************************************************************
+ * WebRtcG711_DecodeA(...)
+ *
+ * This function decodes a packet G711 A-law frame.
+ *
+ * Input:
+ * - state : Dummy state to make this codec look more like
+ * other codecs
+ * - encoded : Encoded data
+ * - len : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - speechType : 1 normal, 2 CNG (for G711 it should
+ * always return 1 since G711 does not have a
+ * built-in DTX/CNG scheme)
+ *
+ * Return value : >0 - Samples in decoded vector
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG711_DecodeA(void *state,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+
+/****************************************************************************
+ * WebRtcG711_DecodeU(...)
+ *
+ * This function decodes a packet G711 U-law frame.
+ *
+ * Input:
+ * - state : Dummy state to make this codec look more like
+ * other codecs
+ * - encoded : Encoded data
+ * - len : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - speechType : 1 normal, 2 CNG (for G711 it should
+ * always return 1 since G711 does not have a
+ * built-in DTX/CNG scheme)
+ *
+ * Return value : >0 - Samples in decoded vector
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG711_DecodeU(void *state,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+
+/**********************************************************************
+* WebRtcG711_Version(...)
+*
+* This function gives the version string of the G.711 codec.
+*
+* Input:
+* - lenBytes: the size of Allocated space (in Bytes) where
+* the version number is written to (in string format).
+*
+* Output:
+* - version: Pointer to a buffer where the version number is
+* written to.
+*
+*/
+
+WebRtc_Word16 WebRtcG711_Version(char* version, WebRtc_Word16 lenBytes);
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc b/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc
new file mode 100644
index 0000000..747f922
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -0,0 +1,169 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * testG711.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "g711_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G711 1000
+
+
+/* function for reading audio data from PCM file */
+int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
+
+ short k, rlen, status = 0;
+
+ rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
+ if (rlen < length) {
+ for (k = rlen; k < length; k++)
+ data[k] = 0;
+ status = 1;
+ }
+
+ return status;
+}
+
+int main(int argc, char* argv[])
+{
+ char inname[80], outname[40], bitname[40];
+ FILE *inp, *outp, *bitp = NULL;
+ int framecnt, endfile;
+
+ WebRtc_Word16 framelength = 80;
+
+ int err;
+
+ /* Runtime statistics */
+ double starttime;
+ double runtime;
+ double length_file;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 shortdata[480];
+ WebRtc_Word16 decoded[480];
+ WebRtc_Word16 streamdata[500];
+ WebRtc_Word16 speechType[1];
+ char law[2];
+ char versionNumber[40];
+
+ /* handling wrong input arguments in the command line */
+ if ((argc!=5) && (argc!=6)) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ printf("\nG.711 test application\n\n");
+ printf("Usage:\n\n");
+ printf("./testG711.exe framelength law infile outfile \n\n");
+ printf("framelength: Framelength in samples.\n");
+ printf("law : Coding law, A och u.\n");
+ printf("infile : Normal speech input file\n");
+ printf("outfile : Speech output file\n\n");
+ printf("outbits : Output bitstream file [optional]\n\n");
+ exit(0);
+
+ }
+
+ /* Get version and print */
+ WebRtcG711_Version(versionNumber, 40);
+
+ printf("-----------------------------------\n");
+ printf("G.711 version: %s\n\n", versionNumber);
+ /* Get frame length */
+ framelength = atoi(argv[1]);
+
+ /* Get compression law */
+ strcpy(law, argv[2]);
+
+ /* Get Input and Output files */
+ sscanf(argv[3], "%s", inname);
+ sscanf(argv[4], "%s", outname);
+ if (argc==6) {
+ sscanf(argv[5], "%s", bitname);
+ if ((bitp = fopen(bitname,"wb")) == NULL) {
+ printf(" G.711: Cannot read file %s.\n", bitname);
+ exit(1);
+ }
+ }
+
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" G.711: Cannot read file %s.\n", inname);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" G.711: Cannot write file %s.\n", outname);
+ exit(1);
+ }
+ printf("\nInput: %s\nOutput: %s\n", inname, outname);
+ if (argc==6) {
+ printf("\nBitfile: %s\n", bitname);
+ }
+
+ starttime = clock()/(double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
+
+ /* Initialize encoder and decoder */
+ framecnt= 0;
+ endfile = 0;
+ while (endfile == 0) {
+ framecnt++;
+ /* Read speech block */
+ endfile = readframe(shortdata, inp, framelength);
+
+ /* G.711 encoding */
+ if (!strcmp(law,"A")) {
+ /* A-law encoding */
+ stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
+ if (argc==6){
+ /* Write bits to file */
+ fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
+ }
+ err = WebRtcG711_DecodeA(NULL, streamdata, stream_len, decoded, speechType);
+ } else if (!strcmp(law,"u")){
+ /* u-law encoding */
+ stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
+ if (argc==6){
+ /* Write bits to file */
+ fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
+ }
+ err = WebRtcG711_DecodeU(NULL, streamdata, stream_len, decoded, speechType);
+ } else {
+ printf("Wrong law mode\n");
+ exit (1);
+ }
+ if (stream_len < 0 || err < 0) {
+ /* exit if returned with error */
+ printf("Error in encoder/decoder\n");
+ } else {
+ /* Write coded speech to file */
+ fwrite(decoded,sizeof(short),framelength,outp);
+ }
+ }
+
+
+ runtime = (double)(clock()/(double)CLOCKS_PER_SEC_G711-starttime);
+ length_file = ((double)framecnt*(double)framelength/8000);
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run G.711: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+ printf("---------------------END----------------------\n");
+
+ fclose(inp);
+ fclose(outp);
+
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/g722/Android.mk b/trunk/src/modules/audio_coding/codecs/g722/Android.mk
new file mode 100644
index 0000000..39dea9e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_g722
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ g722_interface.c \
+ g722_encode.c \
+ g722_decode.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../../..
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722.gypi b/trunk/src/modules/audio_coding/codecs/g722/g722.gypi
new file mode 100644
index 0000000..c1e55d9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722.gypi
@@ -0,0 +1,65 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+ 'targets': [
+ {
+ 'target_name': 'G722',
+ 'type': '<(library)',
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/g722_interface.h',
+ 'g722_interface.c',
+ 'g722_encode.c',
+ 'g722_decode.c',
+ 'g722_enc_dec.h',
+ ],
+ },
+ ], # targets
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'g722_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'G722',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'g722_unittest.cc',
+ ],
+ },
+ {
+ 'target_name': 'G722Test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'G722',
+ ],
+ 'sources': [
+ 'test/testG722.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c b/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c
new file mode 100644
index 0000000..499cc8f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c
@@ -0,0 +1,410 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_decode.c - The ITU G.722 codec, decode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based in part on a single channel G.722 codec which is:
+ *
+ * Copyright (c) CMU 1993
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_decode.c,v 1.15 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Added saturation check on output
+ */
+
+/*! \file */
+
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdio.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "typedefs.h"
+#include "g722_enc_dec.h"
+
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+{
+ WebRtc_Word16 amp16;
+
+ /* Hopefully this is optimised for the common case - not clipping */
+ amp16 = (WebRtc_Word16) amp;
+ if (amp == amp16)
+ return amp16;
+ if (amp > WEBRTC_INT16_MAX)
+ return WEBRTC_INT16_MAX;
+ return WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(g722_decode_state_t *s, int band, int d);
+
+static void block4(g722_decode_state_t *s, int band, int d)
+{
+ int wd1;
+ int wd2;
+ int wd3;
+ int i;
+
+ /* Block 4, RECONS */
+ s->band[band].d[0] = d;
+ s->band[band].r[0] = saturate(s->band[band].s + d);
+
+ /* Block 4, PARREC */
+ s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+ /* Block 4, UPPOL2 */
+ for (i = 0; i < 3; i++)
+ s->band[band].sg[i] = s->band[band].p[i] >> 15;
+ wd1 = saturate(s->band[band].a[1] << 2);
+
+ wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
+ if (wd2 > 32767)
+ wd2 = 32767;
+ wd3 = (s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128;
+ wd3 += (wd2 >> 7);
+ wd3 += (s->band[band].a[2]*32512) >> 15;
+ if (wd3 > 12288)
+ wd3 = 12288;
+ else if (wd3 < -12288)
+ wd3 = -12288;
+ s->band[band].ap[2] = wd3;
+
+ /* Block 4, UPPOL1 */
+ s->band[band].sg[0] = s->band[band].p[0] >> 15;
+ s->band[band].sg[1] = s->band[band].p[1] >> 15;
+ wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
+ wd2 = (s->band[band].a[1]*32640) >> 15;
+
+ s->band[band].ap[1] = saturate(wd1 + wd2);
+ wd3 = saturate(15360 - s->band[band].ap[2]);
+ if (s->band[band].ap[1] > wd3)
+ s->band[band].ap[1] = wd3;
+ else if (s->band[band].ap[1] < -wd3)
+ s->band[band].ap[1] = -wd3;
+
+ /* Block 4, UPZERO */
+ wd1 = (d == 0) ? 0 : 128;
+ s->band[band].sg[0] = d >> 15;
+ for (i = 1; i < 7; i++)
+ {
+ s->band[band].sg[i] = s->band[band].d[i] >> 15;
+ wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
+ wd3 = (s->band[band].b[i]*32640) >> 15;
+ s->band[band].bp[i] = saturate(wd2 + wd3);
+ }
+
+ /* Block 4, DELAYA */
+ for (i = 6; i > 0; i--)
+ {
+ s->band[band].d[i] = s->band[band].d[i - 1];
+ s->band[band].b[i] = s->band[band].bp[i];
+ }
+
+ for (i = 2; i > 0; i--)
+ {
+ s->band[band].r[i] = s->band[band].r[i - 1];
+ s->band[band].p[i] = s->band[band].p[i - 1];
+ s->band[band].a[i] = s->band[band].ap[i];
+ }
+
+ /* Block 4, FILTEP */
+ wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+ wd1 = (s->band[band].a[1]*wd1) >> 15;
+ wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+ wd2 = (s->band[band].a[2]*wd2) >> 15;
+ s->band[band].sp = saturate(wd1 + wd2);
+
+ /* Block 4, FILTEZ */
+ s->band[band].sz = 0;
+ for (i = 6; i > 0; i--)
+ {
+ wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+ s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+ }
+ s->band[band].sz = saturate(s->band[band].sz);
+
+ /* Block 4, PREDIC */
+ s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
+ int rate,
+ int options)
+{
+ if (s == NULL)
+ {
+ if ((s = (g722_decode_state_t *) malloc(sizeof(*s))) == NULL)
+ return NULL;
+ }
+ memset(s, 0, sizeof(*s));
+ if (rate == 48000)
+ s->bits_per_sample = 6;
+ else if (rate == 56000)
+ s->bits_per_sample = 7;
+ else
+ s->bits_per_sample = 8;
+ if ((options & G722_SAMPLE_RATE_8000))
+ s->eight_k = TRUE;
+ if ((options & G722_PACKED) && s->bits_per_sample != 8)
+ s->packed = TRUE;
+ else
+ s->packed = FALSE;
+ s->band[0].det = 32;
+ s->band[1].det = 8;
+ return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_decode_release(g722_decode_state_t *s)
+{
+ free(s);
+ return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
+ const WebRtc_UWord8 g722_data[], int len)
+{
+ static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
+ static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
+ 7, 6, 5, 4, 3, 2, 1, 0 };
+ static const int ilb[32] =
+ {
+ 2048, 2093, 2139, 2186, 2233, 2282, 2332,
+ 2383, 2435, 2489, 2543, 2599, 2656, 2714,
+ 2774, 2834, 2896, 2960, 3025, 3091, 3158,
+ 3228, 3298, 3371, 3444, 3520, 3597, 3676,
+ 3756, 3838, 3922, 4008
+ };
+ static const int wh[3] = {0, -214, 798};
+ static const int rh2[4] = {2, 1, 2, 1};
+ static const int qm2[4] = {-7408, -1616, 7408, 1616};
+ static const int qm4[16] =
+ {
+ 0, -20456, -12896, -8968,
+ -6288, -4240, -2584, -1200,
+ 20456, 12896, 8968, 6288,
+ 4240, 2584, 1200, 0
+ };
+ static const int qm5[32] =
+ {
+ -280, -280, -23352, -17560,
+ -14120, -11664, -9752, -8184,
+ -6864, -5712, -4696, -3784,
+ -2960, -2208, -1520, -880,
+ 23352, 17560, 14120, 11664,
+ 9752, 8184, 6864, 5712,
+ 4696, 3784, 2960, 2208,
+ 1520, 880, 280, -280
+ };
+ static const int qm6[64] =
+ {
+ -136, -136, -136, -136,
+ -24808, -21904, -19008, -16704,
+ -14984, -13512, -12280, -11192,
+ -10232, -9360, -8576, -7856,
+ -7192, -6576, -6000, -5456,
+ -4944, -4464, -4008, -3576,
+ -3168, -2776, -2400, -2032,
+ -1688, -1360, -1040, -728,
+ 24808, 21904, 19008, 16704,
+ 14984, 13512, 12280, 11192,
+ 10232, 9360, 8576, 7856,
+ 7192, 6576, 6000, 5456,
+ 4944, 4464, 4008, 3576,
+ 3168, 2776, 2400, 2032,
+ 1688, 1360, 1040, 728,
+ 432, 136, -432, -136
+ };
+ static const int qmf_coeffs[12] =
+ {
+ 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
+ };
+
+ int dlowt;
+ int rlow;
+ int ihigh;
+ int dhigh;
+ int rhigh;
+ int xout1;
+ int xout2;
+ int wd1;
+ int wd2;
+ int wd3;
+ int code;
+ int outlen;
+ int i;
+ int j;
+
+ outlen = 0;
+ rhigh = 0;
+ for (j = 0; j < len; )
+ {
+ if (s->packed)
+ {
+ /* Unpack the code bits */
+ if (s->in_bits < s->bits_per_sample)
+ {
+ s->in_buffer |= (g722_data[j++] << s->in_bits);
+ s->in_bits += 8;
+ }
+ code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
+ s->in_buffer >>= s->bits_per_sample;
+ s->in_bits -= s->bits_per_sample;
+ }
+ else
+ {
+ code = g722_data[j++];
+ }
+
+ switch (s->bits_per_sample)
+ {
+ default:
+ case 8:
+ wd1 = code & 0x3F;
+ ihigh = (code >> 6) & 0x03;
+ wd2 = qm6[wd1];
+ wd1 >>= 2;
+ break;
+ case 7:
+ wd1 = code & 0x1F;
+ ihigh = (code >> 5) & 0x03;
+ wd2 = qm5[wd1];
+ wd1 >>= 1;
+ break;
+ case 6:
+ wd1 = code & 0x0F;
+ ihigh = (code >> 4) & 0x03;
+ wd2 = qm4[wd1];
+ break;
+ }
+ /* Block 5L, LOW BAND INVQBL */
+ wd2 = (s->band[0].det*wd2) >> 15;
+ /* Block 5L, RECONS */
+ rlow = s->band[0].s + wd2;
+ /* Block 6L, LIMIT */
+ if (rlow > 16383)
+ rlow = 16383;
+ else if (rlow < -16384)
+ rlow = -16384;
+
+ /* Block 2L, INVQAL */
+ wd2 = qm4[wd1];
+ dlowt = (s->band[0].det*wd2) >> 15;
+
+ /* Block 3L, LOGSCL */
+ wd2 = rl42[wd1];
+ wd1 = (s->band[0].nb*127) >> 7;
+ wd1 += wl[wd2];
+ if (wd1 < 0)
+ wd1 = 0;
+ else if (wd1 > 18432)
+ wd1 = 18432;
+ s->band[0].nb = wd1;
+
+ /* Block 3L, SCALEL */
+ wd1 = (s->band[0].nb >> 6) & 31;
+ wd2 = 8 - (s->band[0].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[0].det = wd3 << 2;
+
+ block4(s, 0, dlowt);
+
+ if (!s->eight_k)
+ {
+ /* Block 2H, INVQAH */
+ wd2 = qm2[ihigh];
+ dhigh = (s->band[1].det*wd2) >> 15;
+ /* Block 5H, RECONS */
+ rhigh = dhigh + s->band[1].s;
+ /* Block 6H, LIMIT */
+ if (rhigh > 16383)
+ rhigh = 16383;
+ else if (rhigh < -16384)
+ rhigh = -16384;
+
+ /* Block 2H, INVQAH */
+ wd2 = rh2[ihigh];
+ wd1 = (s->band[1].nb*127) >> 7;
+ wd1 += wh[wd2];
+ if (wd1 < 0)
+ wd1 = 0;
+ else if (wd1 > 22528)
+ wd1 = 22528;
+ s->band[1].nb = wd1;
+
+ /* Block 3H, SCALEH */
+ wd1 = (s->band[1].nb >> 6) & 31;
+ wd2 = 10 - (s->band[1].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[1].det = wd3 << 2;
+
+ block4(s, 1, dhigh);
+ }
+
+ if (s->itu_test_mode)
+ {
+ amp[outlen++] = (WebRtc_Word16) (rlow << 1);
+ amp[outlen++] = (WebRtc_Word16) (rhigh << 1);
+ }
+ else
+ {
+ if (s->eight_k)
+ {
+ amp[outlen++] = (WebRtc_Word16) (rlow << 1);
+ }
+ else
+ {
+ /* Apply the receive QMF */
+ for (i = 0; i < 22; i++)
+ s->x[i] = s->x[i + 2];
+ s->x[22] = rlow + rhigh;
+ s->x[23] = rlow - rhigh;
+
+ xout1 = 0;
+ xout2 = 0;
+ for (i = 0; i < 12; i++)
+ {
+ xout2 += s->x[2*i]*qmf_coeffs[i];
+ xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
+ }
+ /* We shift by 12 to allow for the QMF filters (DC gain = 4096), less 1
+ to allow for the 15 bit input to the G.722 algorithm. */
+ /* WebRtc, tlegrand: added saturation */
+ amp[outlen++] = saturate(xout1 >> 11);
+ amp[outlen++] = saturate(xout2 >> 11);
+ }
+ }
+ }
+ return outlen;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h b/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h
new file mode 100644
index 0000000..d2d19b0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -0,0 +1,158 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722.h - The ITU G.722 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based on a single channel G.722 codec which is:
+ *
+ ***** Copyright (c) CMU 1993 *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Added new defines for minimum and maximum values of short int
+ */
+
+
+/*! \file */
+
+#if !defined(_G722_ENC_DEC_H_)
+#define _G722_ENC_DEC_H_
+
+/*! \page g722_page G.722 encoding and decoding
+\section g722_page_sec_1 What does it do?
+The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
+specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
+
+To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
+support an option for the linear audio to be an 8k samples/second stream. In this mode the
+codec is considerably faster, and still fully compatible with wideband terminals using G.722.
+
+\section g722_page_sec_2 How does it work?
+???.
+*/
+
+#define WEBRTC_INT16_MAX 32767
+#define WEBRTC_INT16_MIN -32768
+
+enum
+{
+ G722_SAMPLE_RATE_8000 = 0x0001,
+ G722_PACKED = 0x0002
+};
+
+typedef struct
+{
+ /*! TRUE if the operating in the special ITU test mode, with the band split filters
+ disabled. */
+ int itu_test_mode;
+ /*! TRUE if the G.722 data is packed */
+ int packed;
+ /*! TRUE if encode from 8k samples/second */
+ int eight_k;
+ /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+ int bits_per_sample;
+
+ /*! Signal history for the QMF */
+ int x[24];
+
+ struct
+ {
+ int s;
+ int sp;
+ int sz;
+ int r[3];
+ int a[3];
+ int ap[3];
+ int p[3];
+ int d[7];
+ int b[7];
+ int bp[7];
+ int sg[7];
+ int nb;
+ int det;
+ } band[2];
+
+ unsigned int in_buffer;
+ int in_bits;
+ unsigned int out_buffer;
+ int out_bits;
+} g722_encode_state_t;
+
+typedef struct
+{
+ /*! TRUE if the operating in the special ITU test mode, with the band split filters
+ disabled. */
+ int itu_test_mode;
+ /*! TRUE if the G.722 data is packed */
+ int packed;
+ /*! TRUE if decode to 8k samples/second */
+ int eight_k;
+ /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+ int bits_per_sample;
+
+ /*! Signal history for the QMF */
+ int x[24];
+
+ struct
+ {
+ int s;
+ int sp;
+ int sz;
+ int r[3];
+ int a[3];
+ int ap[3];
+ int p[3];
+ int d[7];
+ int b[7];
+ int bp[7];
+ int sg[7];
+ int nb;
+ int det;
+ } band[2];
+
+ unsigned int in_buffer;
+ int in_bits;
+ unsigned int out_buffer;
+ int out_bits;
+} g722_decode_state_t;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
+ int rate,
+ int options);
+int WebRtc_g722_encode_release(g722_encode_state_t *s);
+int WebRtc_g722_encode(g722_encode_state_t *s,
+ WebRtc_UWord8 g722_data[],
+ const WebRtc_Word16 amp[],
+ int len);
+
+g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
+ int rate,
+ int options);
+int WebRtc_g722_decode_release(g722_decode_state_t *s);
+int WebRtc_g722_decode(g722_decode_state_t *s,
+ WebRtc_Word16 amp[],
+ const WebRtc_UWord8 g722_data[],
+ int len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c b/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c
new file mode 100644
index 0000000..7487b64
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c
@@ -0,0 +1,434 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_encode.c - The ITU G.722 codec, encode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based on a single channel 64kbps only G.722 codec which is:
+ *
+ ***** Copyright (c) CMU 1993 *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_encode.c,v 1.14 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdio.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "typedefs.h"
+#include "g722_enc_dec.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+{
+ WebRtc_Word16 amp16;
+
+ /* Hopefully this is optimised for the common case - not clipping */
+ amp16 = (WebRtc_Word16) amp;
+ if (amp == amp16)
+ return amp16;
+ if (amp > WEBRTC_INT16_MAX)
+ return WEBRTC_INT16_MAX;
+ return WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(g722_encode_state_t *s, int band, int d)
+{
+ int wd1;
+ int wd2;
+ int wd3;
+ int i;
+
+ /* Block 4, RECONS */
+ s->band[band].d[0] = d;
+ s->band[band].r[0] = saturate(s->band[band].s + d);
+
+ /* Block 4, PARREC */
+ s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+ /* Block 4, UPPOL2 */
+ for (i = 0; i < 3; i++)
+ s->band[band].sg[i] = s->band[band].p[i] >> 15;
+ wd1 = saturate(s->band[band].a[1] << 2);
+
+ wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
+ if (wd2 > 32767)
+ wd2 = 32767;
+ wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128);
+ wd3 += (s->band[band].a[2]*32512) >> 15;
+ if (wd3 > 12288)
+ wd3 = 12288;
+ else if (wd3 < -12288)
+ wd3 = -12288;
+ s->band[band].ap[2] = wd3;
+
+ /* Block 4, UPPOL1 */
+ s->band[band].sg[0] = s->band[band].p[0] >> 15;
+ s->band[band].sg[1] = s->band[band].p[1] >> 15;
+ wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
+ wd2 = (s->band[band].a[1]*32640) >> 15;
+
+ s->band[band].ap[1] = saturate(wd1 + wd2);
+ wd3 = saturate(15360 - s->band[band].ap[2]);
+ if (s->band[band].ap[1] > wd3)
+ s->band[band].ap[1] = wd3;
+ else if (s->band[band].ap[1] < -wd3)
+ s->band[band].ap[1] = -wd3;
+
+ /* Block 4, UPZERO */
+ wd1 = (d == 0) ? 0 : 128;
+ s->band[band].sg[0] = d >> 15;
+ for (i = 1; i < 7; i++)
+ {
+ s->band[band].sg[i] = s->band[band].d[i] >> 15;
+ wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
+ wd3 = (s->band[band].b[i]*32640) >> 15;
+ s->band[band].bp[i] = saturate(wd2 + wd3);
+ }
+
+ /* Block 4, DELAYA */
+ for (i = 6; i > 0; i--)
+ {
+ s->band[band].d[i] = s->band[band].d[i - 1];
+ s->band[band].b[i] = s->band[band].bp[i];
+ }
+
+ for (i = 2; i > 0; i--)
+ {
+ s->band[band].r[i] = s->band[band].r[i - 1];
+ s->band[band].p[i] = s->band[band].p[i - 1];
+ s->band[band].a[i] = s->band[band].ap[i];
+ }
+
+ /* Block 4, FILTEP */
+ wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+ wd1 = (s->band[band].a[1]*wd1) >> 15;
+ wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+ wd2 = (s->band[band].a[2]*wd2) >> 15;
+ s->band[band].sp = saturate(wd1 + wd2);
+
+ /* Block 4, FILTEZ */
+ s->band[band].sz = 0;
+ for (i = 6; i > 0; i--)
+ {
+ wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+ s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+ }
+ s->band[band].sz = saturate(s->band[band].sz);
+
+ /* Block 4, PREDIC */
+ s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
+ int rate, int options)
+{
+ if (s == NULL)
+ {
+ if ((s = (g722_encode_state_t *) malloc(sizeof(*s))) == NULL)
+ return NULL;
+ }
+ memset(s, 0, sizeof(*s));
+ if (rate == 48000)
+ s->bits_per_sample = 6;
+ else if (rate == 56000)
+ s->bits_per_sample = 7;
+ else
+ s->bits_per_sample = 8;
+ if ((options & G722_SAMPLE_RATE_8000))
+ s->eight_k = TRUE;
+ if ((options & G722_PACKED) && s->bits_per_sample != 8)
+ s->packed = TRUE;
+ else
+ s->packed = FALSE;
+ s->band[0].det = 32;
+ s->band[1].det = 8;
+ return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_encode_release(g722_encode_state_t *s)
+{
+ free(s);
+ return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* WebRtc, tlegrand:
+ * Only define the following if bit-exactness with reference implementation
+ * is needed. Will only have any effect if input signal is saturated.
+ */
+//#define RUN_LIKE_REFERENCE_G722
+#ifdef RUN_LIKE_REFERENCE_G722
+WebRtc_Word16 limitValues (WebRtc_Word16 rl)
+{
+
+ WebRtc_Word16 yl;
+
+ yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
+
+ return (yl);
+}
+#endif
+
+int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
+ const WebRtc_Word16 amp[], int len)
+{
+ static const int q6[32] =
+ {
+ 0, 35, 72, 110, 150, 190, 233, 276,
+ 323, 370, 422, 473, 530, 587, 650, 714,
+ 786, 858, 940, 1023, 1121, 1219, 1339, 1458,
+ 1612, 1765, 1980, 2195, 2557, 2919, 0, 0
+ };
+ static const int iln[32] =
+ {
+ 0, 63, 62, 31, 30, 29, 28, 27,
+ 26, 25, 24, 23, 22, 21, 20, 19,
+ 18, 17, 16, 15, 14, 13, 12, 11,
+ 10, 9, 8, 7, 6, 5, 4, 0
+ };
+ static const int ilp[32] =
+ {
+ 0, 61, 60, 59, 58, 57, 56, 55,
+ 54, 53, 52, 51, 50, 49, 48, 47,
+ 46, 45, 44, 43, 42, 41, 40, 39,
+ 38, 37, 36, 35, 34, 33, 32, 0
+ };
+ static const int wl[8] =
+ {
+ -60, -30, 58, 172, 334, 538, 1198, 3042
+ };
+ static const int rl42[16] =
+ {
+ 0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
+ };
+ static const int ilb[32] =
+ {
+ 2048, 2093, 2139, 2186, 2233, 2282, 2332,
+ 2383, 2435, 2489, 2543, 2599, 2656, 2714,
+ 2774, 2834, 2896, 2960, 3025, 3091, 3158,
+ 3228, 3298, 3371, 3444, 3520, 3597, 3676,
+ 3756, 3838, 3922, 4008
+ };
+ static const int qm4[16] =
+ {
+ 0, -20456, -12896, -8968,
+ -6288, -4240, -2584, -1200,
+ 20456, 12896, 8968, 6288,
+ 4240, 2584, 1200, 0
+ };
+ static const int qm2[4] =
+ {
+ -7408, -1616, 7408, 1616
+ };
+ static const int qmf_coeffs[12] =
+ {
+ 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
+ };
+ static const int ihn[3] = {0, 1, 0};
+ static const int ihp[3] = {0, 3, 2};
+ static const int wh[3] = {0, -214, 798};
+ static const int rh2[4] = {2, 1, 2, 1};
+
+ int dlow;
+ int dhigh;
+ int el;
+ int wd;
+ int wd1;
+ int ril;
+ int wd2;
+ int il4;
+ int ih2;
+ int wd3;
+ int eh;
+ int mih;
+ int i;
+ int j;
+ /* Low and high band PCM from the QMF */
+ int xlow;
+ int xhigh;
+ int g722_bytes;
+ /* Even and odd tap accumulators */
+ int sumeven;
+ int sumodd;
+ int ihigh;
+ int ilow;
+ int code;
+
+ g722_bytes = 0;
+ xhigh = 0;
+ for (j = 0; j < len; )
+ {
+ if (s->itu_test_mode)
+ {
+ xlow =
+ xhigh = amp[j++] >> 1;
+ }
+ else
+ {
+ if (s->eight_k)
+ {
+ /* We shift by 1 to allow for the 15 bit input to the G.722 algorithm. */
+ xlow = amp[j++] >> 1;
+ }
+ else
+ {
+ /* Apply the transmit QMF */
+ /* Shuffle the buffer down */
+ for (i = 0; i < 22; i++)
+ s->x[i] = s->x[i + 2];
+ s->x[22] = amp[j++];
+ s->x[23] = amp[j++];
+
+ /* Discard every other QMF output */
+ sumeven = 0;
+ sumodd = 0;
+ for (i = 0; i < 12; i++)
+ {
+ sumodd += s->x[2*i]*qmf_coeffs[i];
+ sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
+ }
+ /* We shift by 12 to allow for the QMF filters (DC gain = 4096), plus 1
+ to allow for us summing two filters, plus 1 to allow for the 15 bit
+ input to the G.722 algorithm. */
+ xlow = (sumeven + sumodd) >> 14;
+ xhigh = (sumeven - sumodd) >> 14;
+
+#ifdef RUN_LIKE_REFERENCE_G722
+ /* The following lines are only used to verify bit-exactness
+ * with reference implementation of G.722. Higher precision
+ * is achieved without limiting the values.
+ */
+ xlow = limitValues(xlow);
+ xhigh = limitValues(xhigh);
+#endif
+ }
+ }
+ /* Block 1L, SUBTRA */
+ el = saturate(xlow - s->band[0].s);
+
+ /* Block 1L, QUANTL */
+ wd = (el >= 0) ? el : -(el + 1);
+
+ for (i = 1; i < 30; i++)
+ {
+ wd1 = (q6[i]*s->band[0].det) >> 12;
+ if (wd < wd1)
+ break;
+ }
+ ilow = (el < 0) ? iln[i] : ilp[i];
+
+ /* Block 2L, INVQAL */
+ ril = ilow >> 2;
+ wd2 = qm4[ril];
+ dlow = (s->band[0].det*wd2) >> 15;
+
+ /* Block 3L, LOGSCL */
+ il4 = rl42[ril];
+ wd = (s->band[0].nb*127) >> 7;
+ s->band[0].nb = wd + wl[il4];
+ if (s->band[0].nb < 0)
+ s->band[0].nb = 0;
+ else if (s->band[0].nb > 18432)
+ s->band[0].nb = 18432;
+
+ /* Block 3L, SCALEL */
+ wd1 = (s->band[0].nb >> 6) & 31;
+ wd2 = 8 - (s->band[0].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[0].det = wd3 << 2;
+
+ block4(s, 0, dlow);
+
+ if (s->eight_k)
+ {
+ /* Just leave the high bits as zero */
+ code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
+ }
+ else
+ {
+ /* Block 1H, SUBTRA */
+ eh = saturate(xhigh - s->band[1].s);
+
+ /* Block 1H, QUANTH */
+ wd = (eh >= 0) ? eh : -(eh + 1);
+ wd1 = (564*s->band[1].det) >> 12;
+ mih = (wd >= wd1) ? 2 : 1;
+ ihigh = (eh < 0) ? ihn[mih] : ihp[mih];
+
+ /* Block 2H, INVQAH */
+ wd2 = qm2[ihigh];
+ dhigh = (s->band[1].det*wd2) >> 15;
+
+ /* Block 3H, LOGSCH */
+ ih2 = rh2[ihigh];
+ wd = (s->band[1].nb*127) >> 7;
+ s->band[1].nb = wd + wh[ih2];
+ if (s->band[1].nb < 0)
+ s->band[1].nb = 0;
+ else if (s->band[1].nb > 22528)
+ s->band[1].nb = 22528;
+
+ /* Block 3H, SCALEH */
+ wd1 = (s->band[1].nb >> 6) & 31;
+ wd2 = 10 - (s->band[1].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[1].det = wd3 << 2;
+
+ block4(s, 1, dhigh);
+ code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
+ }
+
+ if (s->packed)
+ {
+ /* Pack the code bits */
+ s->out_buffer |= (code << s->out_bits);
+ s->out_bits += s->bits_per_sample;
+ if (s->out_bits >= 8)
+ {
+ g722_data[g722_bytes++] = (WebRtc_UWord8) (s->out_buffer & 0xFF);
+ s->out_bits -= 8;
+ s->out_buffer >>= 8;
+ }
+ }
+ else
+ {
+ g722_data[g722_bytes++] = (WebRtc_UWord8) code;
+ }
+ }
+ return g722_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c b/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c
new file mode 100644
index 0000000..d559014
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c
@@ -0,0 +1,115 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+
+#include <stdlib.h>
+#include <string.h>
+#include "g722_interface.h"
+#include "g722_enc_dec.h"
+#include "typedefs.h"
+
+
+WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+ *G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
+ if (*G722enc_inst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+{
+ // Create and/or reset the G.722 encoder
+ // Bitrate 64 kbps and wideband mode (2)
+ G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
+ (g722_encode_state_t*) G722enc_inst, 64000, 2);
+ if (G722enc_inst == NULL) {
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+ // Free encoder memory
+ return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
+}
+
+WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded)
+{
+ unsigned char *codechar = (unsigned char*) encoded;
+ // Encode the input speech vector
+ return WebRtc_g722_encode((g722_encode_state_t*) G722enc_inst,
+ codechar, speechIn, len);
+}
+
+WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+ *G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
+ if (*G722dec_inst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
+{
+ // Create and/or reset the G.722 decoder
+ // Bitrate 64 kbps and wideband mode (2)
+ G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
+ (g722_decode_state_t*) G722dec_inst, 64000, 2);
+ if (G722dec_inst == NULL) {
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+ // Free encoder memory
+ return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
+}
+
+WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ // Decode the G.722 encoder stream
+ *speechType=G722_WEBRTC_SPEECH;
+ return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
+ decoded, (WebRtc_UWord8*) encoded, len);
+}
+
+WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len)
+{
+ // Get version string
+ char version[30] = "2.0.0\n";
+ if (strlen(version) < (unsigned int)len)
+ {
+ strcpy(versionStr, version);
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc b/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc
new file mode 100644
index 0000000..a828edd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "g722_interface.h"
+#include "gtest/gtest.h"
+
+TEST(G722Test, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h b/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h
new file mode 100644
index 0000000..e50d66f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -0,0 +1,190 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
+
+#include "typedefs.h"
+
+/*
+ * Solution to support multiple instances
+ */
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/*
+ * Comfort noise constants
+ */
+
+#define G722_WEBRTC_SPEECH 1
+#define G722_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+/****************************************************************************
+ * WebRtcG722_CreateEncoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ * - G722enc_inst : G722 instance for encoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_EncoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ * - G722enc_inst : G722 instance, i.e. the user that should receive
+ * be initialized
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_FreeEncoder(...)
+ *
+ * Free the memory used for G722 encoder
+ *
+ * Input:
+ * - G722enc_inst : G722 instance for encoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+
+
+
+/****************************************************************************
+ * WebRtcG722_Encode(...)
+ *
+ * This function encodes G722 encoded data.
+ *
+ * Input:
+ * - G722enc_inst : G722 instance, i.e. the user that should encode
+ * a packet
+ * - speechIn : Input speech vector
+ * - len : Samples in speechIn
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded);
+
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ * - G722dec_inst : G722 instance for decoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_DecoderInit(...)
+ *
+ * This function initializes a G729 instance
+ *
+ * Input:
+ * - G729_decinst_t : G729 instance, i.e. the user that should receive
+ * be initialized
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_FreeDecoder(...)
+ *
+ * Free the memory used for G722 decoder
+ *
+ * Input:
+ * - G722dec_inst : G722 instance for decoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_Decode(...)
+ *
+ * This function decodes a packet with G729 frame(s). Output speech length
+ * will be a multiple of 80 samples (80*frames/packet).
+ *
+ * Input:
+ * - G722dec_inst : G722 instance, i.e. the user that should decode
+ * a packet
+ * - encoded : Encoded G722 frame(s)
+ * - len : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - speechType : 1 normal, 2 CNG (Since G722 does not have its own
+ * DTX/CNG scheme it should always return 1)
+ *
+ * Return value : >0 - Samples in decoded vector
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len);
+
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc b/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc
new file mode 100644
index 0000000..9ef8f2d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include "typedefs.h"
+
+/* include API */
+#include "g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722 100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/* function for reading audio data from PCM file */
+int readframe(WebRtc_Word16 *data, FILE *inp, int length)
+{
+ short k, rlen, status = 0;
+
+ rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
+ if (rlen < length) {
+ for (k = rlen; k < length; k++)
+ data[k] = 0;
+ status = 1;
+ }
+
+ return status;
+}
+
+int main(int argc, char* argv[])
+{
+ char inname[60], outbit[40], outname[40];
+ FILE *inp, *outbitp, *outp;
+
+ int framecnt, endfile;
+ WebRtc_Word16 framelength = 160;
+ G722EncInst *G722enc_inst;
+ G722DecInst *G722dec_inst;
+ int err;
+
+ /* Runtime statistics */
+ double starttime;
+ double runtime = 0;
+ double length_file;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 shortdata[960];
+ WebRtc_Word16 decoded[960];
+ WebRtc_Word16 streamdata[80*3];
+ WebRtc_Word16 speechType[1];
+
+ /* handling wrong input arguments in the command line */
+ if (argc!=5) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ printf("Usage:\n\n");
+ printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+ printf("with:\n");
+ printf("framelength : Framelength in samples.\n\n");
+ printf("infile : Normal speech input file\n\n");
+ printf("outbitfile : Bitstream output file\n\n");
+ printf("outspeechfile: Speech output file\n\n");
+ exit(0);
+
+ }
+
+ /* Get frame length */
+ framelength = atoi(argv[1]);
+
+ /* Get Input and Output files */
+ sscanf(argv[2], "%s", inname);
+ sscanf(argv[3], "%s", outbit);
+ sscanf(argv[4], "%s", outname);
+
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" G.722: Cannot read file %s.\n", inname);
+ exit(1);
+ }
+ if ((outbitp = fopen(outbit,"wb")) == NULL) {
+ printf(" G.722: Cannot write file %s.\n", outbit);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" G.722: Cannot write file %s.\n", outname);
+ exit(1);
+ }
+ printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit, outname);
+
+ /* Create and init */
+ WebRtcG722_CreateEncoder((G722EncInst **)&G722enc_inst);
+ WebRtcG722_CreateDecoder((G722DecInst **)&G722dec_inst);
+ WebRtcG722_EncoderInit((G722EncInst *)G722enc_inst);
+ WebRtcG722_DecoderInit((G722DecInst *)G722dec_inst);
+
+
+ /* Initialize encoder and decoder */
+ framecnt = 0;
+ endfile = 0;
+ while (endfile == 0) {
+ framecnt++;
+
+ /* Read speech block */
+ endfile = readframe(shortdata, inp, framelength);
+
+ /* Start clock before call to encoder and decoder */
+ starttime = clock()/(double)CLOCKS_PER_SEC_G722;
+
+ /* G.722 encoding + decoding */
+ stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
+ err = WebRtcG722_Decode((G722DecInst *)G722dec_inst, streamdata, stream_len, decoded, speechType);
+
+ /* Stop clock after call to encoder and decoder */
+ runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
+
+ if (stream_len < 0 || err < 0) {
+ /* exit if returned with error */
+ printf("Error in encoder/decoder\n");
+ } else {
+ /* Write coded bits to file */
+ fwrite(streamdata,sizeof(short),stream_len/2,outbitp);
+ /* Write coded speech to file */
+ fwrite(decoded,sizeof(short),framelength,outp);
+ }
+ }
+
+ WebRtcG722_FreeEncoder((G722EncInst *)G722enc_inst);
+ WebRtcG722_FreeDecoder((G722DecInst *)G722dec_inst);
+
+ length_file = ((double)framecnt*(double)framelength/16000);
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run G.722: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+ printf("---------------------END----------------------\n");
+
+ fclose(inp);
+ fclose(outbitp);
+ fclose(outp);
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h
new file mode 100644
index 0000000..28e9429
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h
@@ -0,0 +1,633 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
+
+/*
+ * Define the fixpoint numeric formats
+ */
+#include "typedefs.h"
+
+
+typedef struct {
+ void *dummy;
+} ISACFIX_MainStruct;
+
+
+#if defined(__cplusplus)
+extern "C" {
+#endif
+
+
+ /**************************************************************************
+ * WebRtcIsacfix_AssignSize(...)
+ *
+ * Functions used when malloc is not allowed
+ * Output the number of bytes needed to allocate for iSAC struct.
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes);
+
+ /**************************************************************************
+ * WebRtcIsacfix_Assign(...)
+ *
+ * Functions used when malloc is not allowed, it
+ * places a struct at the given address.
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ * - ISACFIX_inst_Addr : address of the memory where a space is
+ * for iSAC structure.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst,
+ void *ISACFIX_inst_Addr);
+
+ /****************************************************************************
+ * WebRtcIsacfix_Create(...)
+ *
+ * This function creates an ISAC instance, which will contain the state
+ * information for one coding/decoding channel.
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_Free(...)
+ *
+ * This function frees the ISAC instance created at the beginning.
+ *
+ * Input:
+ * - ISAC_main_inst : a ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_EncoderInit(...)
+ *
+ * This function initializes an ISAC instance prior to the encoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - CodingMode : 0 - Bit rate and frame length are automatically
+ * adjusted to available bandwidth on
+ * transmission channel.
+ * 1 - User sets a frame length and a target bit
+ * rate which is taken as the maximum short-term
+ * average bit rate.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 CodingMode);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_Encode(...)
+ *
+ * This function encodes 10ms frame(s) and inserts it into a package.
+ * Input speech length has to be 160 samples (10ms). The encoder buffers those
+ * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * 0 - The buffer didn't reach the chosen framesize
+ * so it keeps buffering speech samples.
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_Word16 *speechIn,
+ WebRtc_Word16 *encoded);
+
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_EncodeNb(...)
+ *
+ * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
+ * it into a package. Input speech length has to be 80 samples (10ms). The encoder
+ * interpolates into wide-band (16 kHz sampling) buffers those
+ * 10ms frames until it reaches the chosen Framesize (480 or 960 wide-band samples
+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * 0 - The buffer didn't reach the chosen framesize
+ * so it keeps buffering speech samples.
+ * -1 - Error
+ */
+
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_Word16 *speechIn,
+ WebRtc_Word16 *encoded);
+#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_DecoderInit(...)
+ *
+ * This function initializes an ISAC instance prior to the decoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ *
+ * Return value
+ * : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_UpdateBwEstimate1(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 arr_ts);
+
+ /****************************************************************************
+ * WebRtcIsacfix_UpdateBwEstimate(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - send_ts : the send time of the packet from RTP header,
+ * in samples.
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts);
+
+ /****************************************************************************
+ * WebRtcIsacfix_Decode(...)
+ *
+ * This function decodes an ISAC frame. Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_DecodeNb(...)
+ *
+ * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
+ * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_DecodePlcNb(...)
+ *
+ * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
+ * Output speech length will be "240*noOfLostFrames" samples
+ * that equevalent of "30*noOfLostFrames" millisecond.
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames (240 sample=30ms) to produce
+ * NOTE! Maximum number is 2 (480 samples = 60ms)
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames );
+#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+
+
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_DecodePlc(...)
+ *
+ * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
+ * Output speech length will be "480*noOfLostFrames" samples
+ * that is equevalent of "30*noOfLostFrames" millisecond.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames (480sample = 30ms)
+ * to produce
+ * NOTE! Maximum number is 2 (960 samples = 60ms)
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames );
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_ReadFrameLen(...)
+ *
+ * This function returns the length of the frame represented in the packet.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
+ WebRtc_Word16* frameLength);
+
+ /****************************************************************************
+ * WebRtcIsacfix_Control(...)
+ *
+ * This function sets the limit on the short-term average bit rate and the
+ * frame length. Should be used only in Instantaneous mode.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rate : limit on the short-term average bit rate,
+ * in bits/second (between 10000 and 32000)
+ * - framesize : number of milliseconds per frame (30 or 60)
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 rate,
+ WebRtc_Word16 framesize);
+
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_ControlBwe(...)
+ *
+ * This function sets the initial values of bottleneck and frame-size if
+ * iSAC is used in channel-adaptive mode. Through this API, users can
+ * enforce a frame-size for all values of bottleneck. Then iSAC will not
+ * automatically change the frame-size.
+ *
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rateBPS : initial value of bottleneck in bits/second
+ * 10000 <= rateBPS <= 32000 is accepted
+ * - frameSizeMs : number of milliseconds per frame (30 or 60)
+ * - enforceFrameSize : 1 to enforce the given frame-size through out
+ * the adaptation process, 0 to let iSAC change
+ * the frame-size if required.
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 rateBPS,
+ WebRtc_Word16 frameSizeMs,
+ WebRtc_Word16 enforceFrameSize);
+
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_version(...)
+ *
+ * This function returns the version number.
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ */
+
+ void WebRtcIsacfix_version(char *version);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_GetErrorCode(...)
+ *
+ * This function can be used to check the error code of an iSAC instance. When
+ * a function returns -1 a error code will be set for that instance. The
+ * function below extract the code of the last error that occured in the
+ * specified instance.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance
+ *
+ * Return value : Error code
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_GetUplinkBw(...)
+ *
+ * This function return iSAC send bitrate
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : <0 Error code
+ * else bitrate
+ */
+
+ WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_SetMaxPayloadSize(...)
+ *
+ * This function sets a limit for the maximum payload size of iSAC. The same
+ * value is used both for 30 and 60 msec packets.
+ * The absolute max will be valid until next time the function is called.
+ * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxPayloadBytes : maximum size of the payload in bytes
+ * valid values are between 100 and 400 bytes
+ *
+ *
+ * Return value : 0 if sucessful
+ * -1 if error happens
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 maxPayloadBytes);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_SetMaxRate(...)
+ *
+ * This function sets the maximum rate which the codec may not exceed for a
+ * singel packet. The maximum rate is set in bits per second.
+ * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
+ * per 30 msec).
+ * It is possible to set a maximum rate between 32000 and 53400 bits per second.
+ *
+ * The rate limit is valid until next time the function is called.
+ *
+ * NOTE! Packet size will never go above the value set if calling
+ * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxRateInBytes : maximum rate in bits per second,
+ * valid values are 32000 to 53400 bits
+ *
+ * Return value : 0 if sucessful
+ * -1 if error happens
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word32 maxRate);
+
+ /****************************************************************************
+ * WebRtcIsacfix_CreateInternal(...)
+ *
+ * This function creates the memory that is used to store data in the encoder
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_FreeInternal(...)
+ *
+ * This function frees the internal memory for storing encoder data.
+ *
+ * Input:
+ * - ISAC_main_inst : an ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_GetNewBitStream(...)
+ *
+ * This function returns encoded data, with the recieved bwe-index in the
+ * stream. It should always return a complete packet, i.e. only called once
+ * even for 60 msec frames
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - bweIndex : index of bandwidth estimate to put in new bitstream
+ * - scale : factor for rate change (0.4 ~=> half the rate, 1 no change).
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 bweIndex,
+ float scale,
+ WebRtc_Word16 *encoded);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_GetDownLinkBwIndex(...)
+ *
+ * This function returns index representing the Bandwidth estimate from
+ * other side to this side.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Output:
+ * - rateIndex : Bandwidth estimate to transmit to other side.
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
+ WebRtc_Word16* rateIndex);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_UpdateUplinkBw(...)
+ *
+ * This function takes an index representing the Bandwidth estimate from
+ * this side to other side and updates BWE.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ * - rateIndex : Bandwidth estimate from other side.
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
+ WebRtc_Word16 rateIndex);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_ReadBwIndex(...)
+ *
+ * This function returns the index of the Bandwidth estimate from the bitstream.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - rateIndex : Bandwidth estimate in bitstream
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
+ WebRtc_Word16* rateIndex);
+
+
+ /****************************************************************************
+ * WebRtcIsacfix_GetNewFrameLen(...)
+ *
+ * This function return the next frame length (in samples) of iSAC.
+ *
+ * Input:
+ * -ISAC_main_inst : iSAC instance
+ *
+ * Return value : frame lenght in samples
+ */
+
+ WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
+
+
+#if defined(__cplusplus)
+}
+#endif
+
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
new file mode 100644
index 0000000..c81d13a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
@@ -0,0 +1,143 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+#############################
+# Build the non-neon library.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_isacfix
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ arith_routines.c \
+ arith_routines_hist.c \
+ arith_routines_logist.c \
+ bandwidth_estimator.c \
+ decode.c \
+ decode_bwe.c \
+ decode_plc.c \
+ encode.c \
+ entropy_coding.c \
+ fft.c \
+ filterbank_tables.c \
+ filterbanks.c \
+ filters.c \
+ initialize.c \
+ isacfix.c \
+ lattice.c \
+ lpc_masking_model.c \
+ lpc_tables.c \
+ pitch_estimator.c \
+ pitch_filter.c \
+ pitch_gain_tables.c \
+ pitch_lag_tables.c \
+ spectrum_ar_model_tables.c \
+ transform.c
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+LOCAL_SRC_FILES += \
+ lattice_armv7.S
+else
+LOCAL_SRC_FILES += \
+ lattice_c.c
+endif
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#########################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_isacfix_neon
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ filters_neon.c \
+ lattice_neon.S #.S extention is for including a header file in assembly.
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ -mfpu=neon \
+ -mfloat-abi=softfp \
+ -flax-vector-conversions
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
+
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+###########################
+# isac test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= ../test/kenny.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../../../..
+
+LOCAL_STATIC_LIBRARIES := \
+ libwebrtc_isacfix \
+ libwebrtc_spl
+
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_STATIC_LIBRARIES += \
+ libwebrtc_isacfix_neon
+endif
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils
+
+LOCAL_MODULE:= webrtc_isac_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include $(BUILD_NATIVE_TEST)
+endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c
new file mode 100644
index 0000000..ee62bad
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routins.c
+ *
+ * This C file contains a function for finalizing the bitstream
+ * after arithmetic coding.
+ *
+ */
+
+#include "arith_routins.h"
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncTerminate(...)
+ *
+ * Final call to the arithmetic coder for an encoder call. This function
+ * terminates and return byte stream.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ *
+ * Return value : number of bytes in the stream
+ */
+WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData)
+{
+ WebRtc_UWord16 *streamPtr;
+ WebRtc_UWord16 negCarry;
+
+ /* point to the right place in the stream buffer */
+ streamPtr = streamData->stream + streamData->stream_index;
+
+ /* find minimum length (determined by current interval width) */
+ if ( streamData->W_upper > 0x01FFFFFF )
+ {
+ streamData->streamval += 0x01000000;
+
+ /* if result is less than the added value we must take care of the carry */
+ if (streamData->streamval < 0x01000000)
+ {
+ /* propagate carry */
+ if (streamData->full == 0) {
+ /* Add value to current value */
+ negCarry = *streamPtr;
+ negCarry += 0x0100;
+ *streamPtr = negCarry;
+
+ /* if value is too big, propagate carry to next byte, and so on */
+ while (!(negCarry))
+ {
+ negCarry = *--streamPtr;
+ negCarry++;
+ *streamPtr = negCarry;
+ }
+ } else {
+ /* propagate carry by adding one to the previous byte in the
+ * stream if that byte is 0xFFFF we need to propagate the carry
+ * furhter back in the stream */
+ while ( !(++(*--streamPtr)) );
+ }
+
+ /* put pointer back to the old value */
+ streamPtr = streamData->stream + streamData->stream_index;
+ }
+ /* write remaining data to bitstream, if "full == 0" first byte has data */
+ if (streamData->full == 0) {
+ *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+ streamData->full = 1;
+ } else {
+ *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
+ WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
+ streamData->full = 0;
+ }
+ }
+ else
+ {
+ streamData->streamval += 0x00010000;
+
+ /* if result is less than the added value we must take care of the carry */
+ if (streamData->streamval < 0x00010000)
+ {
+ /* propagate carry */
+ if (streamData->full == 0) {
+ /* Add value to current value */
+ negCarry = *streamPtr;
+ negCarry += 0x0100;
+ *streamPtr = negCarry;
+
+ /* if value to big, propagate carry to next byte, and so on */
+ while (!(negCarry))
+ {
+ negCarry = *--streamPtr;
+ negCarry++;
+ *streamPtr = negCarry;
+ }
+ } else {
+ /* Add carry to previous byte */
+ while ( !(++(*--streamPtr)) );
+ }
+
+ /* put pointer back to the old value */
+ streamPtr = streamData->stream + streamData->stream_index;
+ }
+ /* write remaining data (2 bytes) to bitstream */
+ if (streamData->full) {
+ *streamPtr++ = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 16);
+ } else {
+ *streamPtr++ |= (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+ *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 8)
+ & 0xFF00;
+ }
+ }
+
+ /* calculate stream length in bytes */
+ return (((streamPtr - streamData->stream)<<1) + !(streamData->full));
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c
new file mode 100644
index 0000000..14f1add
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c
@@ -0,0 +1,404 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routinshist.c
+ *
+ * This C file contains arithmetic encoding and decoding.
+ *
+ */
+
+#include "arith_routins.h"
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncHistMulti(...)
+ *
+ * Encode the histogram interval
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - data : data vector
+ * - cdf : array of cdf arrays
+ * - lenData : data vector length
+ *
+ * Return value : 0 if ok
+ * <0 if error detected
+ */
+int WebRtcIsacfix_EncHistMulti(Bitstr_enc *streamData,
+ const WebRtc_Word16 *data,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_Word16 lenData)
+{
+ WebRtc_UWord32 W_lower;
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 W_upper_LSB;
+ WebRtc_UWord32 W_upper_MSB;
+ WebRtc_UWord16 *streamPtr;
+ WebRtc_UWord16 negCarry;
+ WebRtc_UWord16 *maxStreamPtr;
+ WebRtc_UWord16 *streamPtrCarry;
+ WebRtc_UWord32 cdfLo;
+ WebRtc_UWord32 cdfHi;
+ int k;
+
+
+ /* point to beginning of stream buffer
+ * and set maximum streamPtr value */
+ streamPtr = streamData->stream + streamData->stream_index;
+ maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1;
+
+ W_upper = streamData->W_upper;
+
+ for (k = lenData; k > 0; k--)
+ {
+ /* fetch cdf_lower and cdf_upper from cdf tables */
+ cdfLo = (WebRtc_UWord32) *(*cdf + (WebRtc_UWord32)*data);
+ cdfHi = (WebRtc_UWord32) *(*cdf++ + (WebRtc_UWord32)*data++ + 1);
+
+ /* update interval */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = WEBRTC_SPL_RSHIFT_W32(W_upper, 16);
+ W_lower = WEBRTC_SPL_UMUL(W_upper_MSB, cdfLo);
+ W_lower += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfLo);
+ W_upper = WEBRTC_SPL_UMUL(W_upper_MSB, cdfHi);
+ W_upper += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfHi);
+
+ /* shift interval such that it begins at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamData->streamval += W_lower;
+
+ /* handle carry */
+ if (streamData->streamval < W_lower)
+ {
+ /* propagate carry */
+ streamPtrCarry = streamPtr;
+ if (streamData->full == 0) {
+ negCarry = *streamPtrCarry;
+ negCarry += 0x0100;
+ *streamPtrCarry = negCarry;
+ while (!(negCarry))
+ {
+ negCarry = *--streamPtrCarry;
+ negCarry++;
+ *streamPtrCarry = negCarry;
+ }
+ } else {
+ while ( !(++(*--streamPtrCarry)) );
+ }
+ }
+
+ /* renormalize interval, store most significant byte of streamval and update streamval
+ * W_upper < 2^24 */
+ while ( !(W_upper & 0xFF000000) )
+ {
+ W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
+ if (streamData->full == 0) {
+ *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
+ streamData->full = 1;
+ } else {
+ *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
+ WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
+ streamData->full = 0;
+ }
+
+ if( streamPtr > maxStreamPtr ) {
+ return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
+ }
+ streamData->streamval = WEBRTC_SPL_LSHIFT_W32(streamData->streamval, 8);
+ }
+ }
+
+ /* calculate new stream_index */
+ streamData->stream_index = streamPtr - streamData->stream;
+ streamData->W_upper = W_upper;
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecHistBisectMulti(...)
+ *
+ * Function to decode more symbols from the arithmetic bytestream, using
+ * method of bisection cdf tables should be of size 2^k-1 (which corresponds
+ * to an alphabet size of 2^k-2)
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - cdf : array of cdf arrays
+ * - cdfSize : array of cdf table sizes+1 (power of two: 2^k)
+ * - lenData : data vector length
+ *
+ * Output:
+ * - data : data vector
+ *
+ * Return value : number of bytes in the stream
+ * <0 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(WebRtc_Word16 *data,
+ Bitstr_dec *streamData,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_UWord16 *cdfSize,
+ const WebRtc_Word16 lenData)
+{
+ WebRtc_UWord32 W_lower = 0;
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord32 W_upper_LSB;
+ WebRtc_UWord32 W_upper_MSB;
+ WebRtc_UWord32 streamval;
+ const WebRtc_UWord16 *streamPtr;
+ const WebRtc_UWord16 *cdfPtr;
+ WebRtc_Word16 sizeTmp;
+ int k;
+
+
+ streamPtr = streamData->stream + streamData->stream_index;
+ W_upper = streamData->W_upper;
+
+ /* Error check: should not be possible in normal operation */
+ if (W_upper == 0) {
+ return -2;
+ }
+
+ /* first time decoder is called for this stream */
+ if (streamData->stream_index == 0)
+ {
+ /* read first word from bytestream */
+ streamval = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)*streamPtr++, 16);
+ streamval |= *streamPtr++;
+ } else {
+ streamval = streamData->streamval;
+ }
+
+ for (k = lenData; k > 0; k--)
+ {
+ /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = WEBRTC_SPL_RSHIFT_W32(W_upper, 16);
+
+ /* start halfway the cdf range */
+ sizeTmp = WEBRTC_SPL_RSHIFT_W16(*cdfSize++, 1);
+ cdfPtr = *cdf + (sizeTmp - 1);
+
+ /* method of bisection */
+ for ( ;; )
+ {
+ W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
+ W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+ sizeTmp = WEBRTC_SPL_RSHIFT_W16(sizeTmp, 1);
+ if (sizeTmp == 0) {
+ break;
+ }
+
+ if (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ cdfPtr += sizeTmp;
+ } else {
+ W_upper = W_tmp;
+ cdfPtr -= sizeTmp;
+ }
+ }
+ if (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ *data++ = cdfPtr - *cdf++;
+ } else {
+ W_upper = W_tmp;
+ *data++ = cdfPtr - *cdf++ - 1;
+ }
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamval -= W_lower;
+
+ /* renormalize interval and update streamval */
+ /* W_upper < 2^24 */
+ while ( !(W_upper & 0xFF000000) )
+ {
+ /* read next byte from stream */
+ if (streamData->full == 0) {
+ streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) |
+ (*streamPtr++ & 0x00FF);
+ streamData->full = 1;
+ } else {
+ streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) |
+ WEBRTC_SPL_RSHIFT_W16(*streamPtr, 8);
+ streamData->full = 0;
+ }
+ W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
+ }
+
+
+ /* Error check: should not be possible in normal operation */
+ if (W_upper == 0) {
+ return -2;
+ }
+
+ }
+
+ streamData->stream_index = streamPtr - streamData->stream;
+ streamData->W_upper = W_upper;
+ streamData->streamval = streamval;
+
+ if ( W_upper > 0x01FFFFFF ) {
+ return (streamData->stream_index*2 - 3 + !streamData->full);
+ } else {
+ return (streamData->stream_index*2 - 2 + !streamData->full);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecHistOneStepMulti(...)
+ *
+ * Function to decode more symbols from the arithmetic bytestream, taking
+ * single step up or down at a time.
+ * cdf tables can be of arbitrary size, but large tables may take a lot of
+ * iterations.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - cdf : array of cdf arrays
+ * - initIndex : vector of initial cdf table search entries
+ * - lenData : data vector length
+ *
+ * Output:
+ * - data : data vector
+ *
+ * Return value : number of bytes in original stream
+ * <0 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(WebRtc_Word16 *data,
+ Bitstr_dec *streamData,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_UWord16 *initIndex,
+ const WebRtc_Word16 lenData)
+{
+ WebRtc_UWord32 W_lower;
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord32 W_upper_LSB;
+ WebRtc_UWord32 W_upper_MSB;
+ WebRtc_UWord32 streamval;
+ const WebRtc_UWord16 *streamPtr;
+ const WebRtc_UWord16 *cdfPtr;
+ int k;
+
+
+ streamPtr = streamData->stream + streamData->stream_index;
+ W_upper = streamData->W_upper;
+ /* Error check: Should not be possible in normal operation */
+ if (W_upper == 0) {
+ return -2;
+ }
+
+ /* Check if it is the first time decoder is called for this stream */
+ if (streamData->stream_index == 0)
+ {
+ /* read first word from bytestream */
+ streamval = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
+ streamval |= *streamPtr++;
+ } else {
+ streamval = streamData->streamval;
+ }
+
+ for (k = lenData; k > 0; k--)
+ {
+ /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
+
+ /* start at the specified table entry */
+ cdfPtr = *cdf + (*initIndex++);
+ W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
+ W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+
+ if (streamval > W_tmp)
+ {
+ for ( ;; )
+ {
+ W_lower = W_tmp;
+
+ /* range check */
+ if (cdfPtr[0] == 65535) {
+ return -3;
+ }
+
+ W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *++cdfPtr);
+ W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+
+ if (streamval <= W_tmp) {
+ break;
+ }
+ }
+ W_upper = W_tmp;
+ *data++ = cdfPtr - *cdf++ - 1;
+ } else {
+ for ( ;; )
+ {
+ W_upper = W_tmp;
+ --cdfPtr;
+
+ /* range check */
+ if (cdfPtr < *cdf) {
+ return -3;
+ }
+
+ W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
+ W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+
+ if (streamval > W_tmp) {
+ break;
+ }
+ }
+ W_lower = W_tmp;
+ *data++ = cdfPtr - *cdf++;
+ }
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamval -= W_lower;
+
+ /* renormalize interval and update streamval */
+ /* W_upper < 2^24 */
+ while ( !(W_upper & 0xFF000000) )
+ {
+ /* read next byte from stream */
+ if (streamData->full == 0) {
+ streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) | (*streamPtr++ & 0x00FF);
+ streamData->full = 1;
+ } else {
+ streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) | (*streamPtr >> 8);
+ streamData->full = 0;
+ }
+ W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
+ }
+ }
+
+ streamData->stream_index = streamPtr - streamData->stream;
+ streamData->W_upper = W_upper;
+ streamData->streamval = streamval;
+
+ /* find number of bytes in original stream (determined by current interval width) */
+ if ( W_upper > 0x01FFFFFF ) {
+ return (streamData->stream_index*2 - 3 + !streamData->full);
+ } else {
+ return (streamData->stream_index*2 - 2 + !streamData->full);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c
new file mode 100644
index 0000000..39c437e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c
@@ -0,0 +1,404 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routinslogist.c
+ *
+ * This C file contains arithmetic encode and decode logistic
+ *
+ */
+
+#include "arith_routins.h"
+
+
+/* Tables for piecewise linear cdf functions: y = k*x */
+
+/* x Points for function piecewise() in Q15 */
+static const WebRtc_Word32 kHistEdges[51] = {
+ -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
+ -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644,
+ -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428,
+ 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500,
+ 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572,
+ 327680
+};
+
+
+/* k Points for function piecewise() in Q0 */
+static const WebRtc_UWord16 kCdfSlope[51] = {
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088,
+ 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312,
+ 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 2,
+ 0
+};
+
+/* y Points for function piecewise() in Q0 */
+static const WebRtc_UWord16 kCdfLogistic[51] = {
+ 0, 2, 4, 6, 8, 10, 12, 14, 16, 18,
+ 20, 22, 24, 29, 38, 57, 92, 153, 279, 559,
+ 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636,
+ 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514,
+ 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534,
+ 65535
+};
+
+
+/****************************************************************************
+ * WebRtcIsacfix_Piecewise(...)
+ *
+ * Piecewise linear function
+ *
+ * Input:
+ * - xinQ15 : input value x in Q15
+ *
+ * Return value : korresponding y-value in Q0
+ */
+
+
+static __inline WebRtc_UWord16 WebRtcIsacfix_Piecewise(WebRtc_Word32 xinQ15) {
+ WebRtc_Word32 ind;
+ WebRtc_Word32 qtmp1;
+ WebRtc_UWord16 qtmp2;
+
+ /* Find index for x-value */
+ qtmp1 = WEBRTC_SPL_SAT(kHistEdges[50],xinQ15,kHistEdges[0]);
+ ind = WEBRTC_SPL_MUL(5, qtmp1 - kHistEdges[0]);
+ ind = WEBRTC_SPL_RSHIFT_W32(ind, 16);
+
+ /* Calculate corresponding y-value ans return*/
+ qtmp1 = qtmp1 - kHistEdges[ind];
+ qtmp2 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(
+ WEBRTC_SPL_UMUL_32_16(qtmp1,kCdfSlope[ind]), 15);
+ return (kCdfLogistic[ind] + qtmp2);
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_EncLogisticMulti2(...)
+ *
+ * Arithmetic coding of spectrum.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - dataQ7 : data vector in Q7
+ * - envQ8 : side info vector defining the width of the pdf
+ * in Q8
+ * - lenData : data vector length
+ *
+ * Return value : 0 if ok,
+ * <0 otherwise.
+ */
+int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
+ WebRtc_Word16 *dataQ7,
+ const WebRtc_UWord16 *envQ8,
+ const WebRtc_Word16 lenData)
+{
+ WebRtc_UWord32 W_lower;
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord16 W_upper_LSB;
+ WebRtc_UWord16 W_upper_MSB;
+ WebRtc_UWord16 *streamPtr;
+ WebRtc_UWord16 *maxStreamPtr;
+ WebRtc_UWord16 *streamPtrCarry;
+ WebRtc_UWord16 negcarry;
+ WebRtc_UWord32 cdfLo;
+ WebRtc_UWord32 cdfHi;
+ int k;
+
+ /* point to beginning of stream buffer
+ * and set maximum streamPtr value */
+ streamPtr = streamData->stream + streamData->stream_index;
+ maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1;
+ W_upper = streamData->W_upper;
+
+ for (k = 0; k < lenData; k++)
+ {
+ /* compute cdf_lower and cdf_upper by evaluating the
+ * WebRtcIsacfix_Piecewise linear cdf */
+ cdfLo = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8));
+ cdfHi = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8));
+
+ /* test and clip if probability gets too small */
+ while ((cdfLo + 1) >= cdfHi) {
+ /* clip */
+ if (*dataQ7 > 0) {
+ *dataQ7 -= 128;
+ cdfHi = cdfLo;
+ cdfLo = WebRtcIsacfix_Piecewise(
+ WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8));
+ } else {
+ *dataQ7 += 128;
+ cdfLo = cdfHi;
+ cdfHi = WebRtcIsacfix_Piecewise(
+ WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8));
+ }
+ }
+
+ dataQ7++;
+ /* increment only once per 4 iterations */
+ envQ8 += (k & 1) & (k >> 1);
+
+
+ /* update interval */
+ W_upper_LSB = (WebRtc_UWord16)W_upper;
+ W_upper_MSB = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
+ W_lower = WEBRTC_SPL_UMUL_32_16(cdfLo, W_upper_MSB);
+ W_lower += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfLo, W_upper_LSB);
+ W_upper = WEBRTC_SPL_UMUL_32_16(cdfHi, W_upper_MSB);
+ W_upper += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfHi, W_upper_LSB);
+
+ /* shift interval such that it begins at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamData->streamval += W_lower;
+
+ /* handle carry */
+ if (streamData->streamval < W_lower)
+ {
+ /* propagate carry */
+ streamPtrCarry = streamPtr;
+ if (streamData->full == 0) {
+ negcarry = *streamPtrCarry;
+ negcarry += 0x0100;
+ *streamPtrCarry = negcarry;
+ while (!(negcarry))
+ {
+ negcarry = *--streamPtrCarry;
+ negcarry++;
+ *streamPtrCarry = negcarry;
+ }
+ } else {
+ while (!(++(*--streamPtrCarry)));
+ }
+ }
+
+ /* renormalize interval, store most significant byte of streamval and update streamval
+ * W_upper < 2^24 */
+ while ( !(W_upper & 0xFF000000) )
+ {
+ W_upper = WEBRTC_SPL_LSHIFT_U32(W_upper, 8);
+ if (streamData->full == 0) {
+ *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(
+ streamData->streamval, 24);
+ streamData->full = 1;
+ } else {
+ *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_U32(
+ WEBRTC_SPL_RSHIFT_U32(streamData->streamval, 24), 8);
+ streamData->full = 0;
+ }
+
+ if( streamPtr > maxStreamPtr )
+ return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
+
+ streamData->streamval = WEBRTC_SPL_LSHIFT_U32(streamData->streamval, 8);
+ }
+ }
+
+ /* calculate new stream_index */
+ streamData->stream_index = streamPtr - streamData->stream;
+ streamData->W_upper = W_upper;
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecLogisticMulti2(...)
+ *
+ * Arithmetic decoding of spectrum.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - envQ8 : side info vector defining the width of the pdf
+ * in Q8
+ * - lenData : data vector length
+ *
+ * Input/Output:
+ * - dataQ7 : input: dither vector, output: data vector
+ *
+ * Return value : number of bytes in the stream so far
+ * -1 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(WebRtc_Word16 *dataQ7,
+ Bitstr_dec *streamData,
+ const WebRtc_Word32 *envQ8,
+ const WebRtc_Word16 lenData)
+{
+ WebRtc_UWord32 W_lower;
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord16 W_upper_LSB;
+ WebRtc_UWord16 W_upper_MSB;
+ WebRtc_UWord32 streamVal;
+ WebRtc_UWord16 cdfTmp;
+ WebRtc_Word32 res;
+ WebRtc_Word32 inSqrt;
+ WebRtc_Word32 newRes;
+ const WebRtc_UWord16 *streamPtr;
+ WebRtc_Word16 candQ7;
+ WebRtc_Word16 envCount;
+ WebRtc_UWord16 tmpARSpecQ8 = 0;
+ int k, i;
+
+
+ /* point to beginning of stream buffer */
+ streamPtr = streamData->stream + streamData->stream_index;
+ W_upper = streamData->W_upper;
+
+ /* Check if it is first time decoder is called for this stream */
+ if (streamData->stream_index == 0)
+ {
+ /* read first word from bytestream */
+ streamVal = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
+ streamVal |= *streamPtr++;
+
+ } else {
+ streamVal = streamData->streamval;
+ }
+
+
+ res = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,
+ WEBRTC_SPL_RSHIFT_W16(WebRtcSpl_GetSizeInBits(envQ8[0]), 1));
+ envCount = 0;
+
+ /* code assumes lenData%4 == 0 */
+ for (k = 0; k < lenData; k += 4)
+ {
+ int k4;
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ inSqrt = envQ8[envCount];
+ i = 10;
+
+ /* For safty reasons */
+ if (inSqrt < 0)
+ inSqrt=-inSqrt;
+
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+ do
+ {
+ res = newRes;
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+ } while (newRes != res && i-- > 0);
+
+ tmpARSpecQ8 = (WebRtc_UWord16)newRes;
+
+ for(k4 = 0; k4 < 4; k4++)
+ {
+ /* find the integer *data for which streamVal lies in [W_lower+1, W_upper] */
+ W_upper_LSB = (WebRtc_UWord16) (W_upper & 0x0000FFFF);
+ W_upper_MSB = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
+
+ /* find first candidate by inverting the logistic cdf
+ * Input dither value collected from io-stream */
+ candQ7 = - *dataQ7 + 64;
+ cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
+
+ W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
+ W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
+
+ if (streamVal > W_tmp)
+ {
+ W_lower = W_tmp;
+ candQ7 += 128;
+ cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
+
+ W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
+ W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
+
+ while (streamVal > W_tmp)
+ {
+ W_lower = W_tmp;
+ candQ7 += 128;
+ cdfTmp = WebRtcIsacfix_Piecewise(
+ WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
+
+ W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
+ W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
+
+ /* error check */
+ if (W_lower == W_tmp) {
+ return -1;
+ }
+ }
+ W_upper = W_tmp;
+
+ /* Output value put in dataQ7: another sample decoded */
+ *dataQ7 = candQ7 - 64;
+ }
+ else
+ {
+ W_upper = W_tmp;
+ candQ7 -= 128;
+ cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
+
+ W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
+ W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
+
+ while ( !(streamVal > W_tmp) )
+ {
+ W_upper = W_tmp;
+ candQ7 -= 128;
+ cdfTmp = WebRtcIsacfix_Piecewise(
+ WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
+
+ W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
+ W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
+
+ /* error check */
+ if (W_upper == W_tmp){
+ return -1;
+ }
+ }
+ W_lower = W_tmp;
+
+ /* Output value put in dataQ7: another sample decoded */
+ *dataQ7 = candQ7 + 64;
+ }
+
+ dataQ7++;
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamVal -= W_lower;
+
+ /* renormalize interval and update streamVal
+ * W_upper < 2^24 */
+ while ( !(W_upper & 0xFF000000) )
+ {
+ /* read next byte from stream */
+ if (streamData->full == 0) {
+ streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | (*streamPtr++ & 0x00FF);
+ streamData->full = 1;
+ } else {
+ streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) |
+ WEBRTC_SPL_RSHIFT_U16(*streamPtr, 8);
+ streamData->full = 0;
+ }
+ W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
+ }
+ }
+ envCount++;
+ }
+
+ streamData->stream_index = streamPtr - streamData->stream;
+ streamData->W_upper = W_upper;
+ streamData->streamval = streamVal;
+
+ /* find number of bytes in original stream (determined by current interval width) */
+ if ( W_upper > 0x01FFFFFF )
+ return (streamData->stream_index*2 - 3 + !streamData->full);
+ else
+ return (streamData->stream_index*2 - 2 + !streamData->full);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h
new file mode 100644
index 0000000..9aa49da
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h
@@ -0,0 +1,160 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routins.h
+ *
+ * Functions for arithmetic coding.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
+
+#include "structs.h"
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncLogisticMulti2(...)
+ *
+ * Arithmetic coding of spectrum.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - dataQ7 : data vector in Q7
+ * - envQ8 : side info vector defining the width of the pdf
+ * in Q8
+ * - lenData : data vector length
+ *
+ * Return value : 0 if ok,
+ * <0 otherwise.
+ */
+int WebRtcIsacfix_EncLogisticMulti2(
+ Bitstr_enc *streamData,
+ WebRtc_Word16 *dataQ7,
+ const WebRtc_UWord16 *env,
+ const WebRtc_Word16 lenData);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncTerminate(...)
+ *
+ * Final call to the arithmetic coder for an encoder call. This function
+ * terminates and return byte stream.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ *
+ * Return value : number of bytes in the stream
+ */
+WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecLogisticMulti2(...)
+ *
+ * Arithmetic decoding of spectrum.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - envQ8 : side info vector defining the width of the pdf
+ * in Q8
+ * - lenData : data vector length
+ *
+ * Input/Output:
+ * - dataQ7 : input: dither vector, output: data vector, in Q7
+ *
+ * Return value : number of bytes in the stream so far
+ * <0 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(
+ WebRtc_Word16 *data,
+ Bitstr_dec *streamData,
+ const WebRtc_Word32 *env,
+ const WebRtc_Word16 lenData);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncHistMulti(...)
+ *
+ * Encode the histogram interval
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - data : data vector
+ * - cdf : array of cdf arrays
+ * - lenData : data vector length
+ *
+ * Return value : 0 if ok
+ * <0 if error detected
+ */
+int WebRtcIsacfix_EncHistMulti(
+ Bitstr_enc *streamData,
+ const WebRtc_Word16 *data,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_Word16 lenData);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecHistBisectMulti(...)
+ *
+ * Function to decode more symbols from the arithmetic bytestream, using
+ * method of bisection.
+ * C df tables should be of size 2^k-1 (which corresponds to an
+ * alphabet size of 2^k-2)
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - cdf : array of cdf arrays
+ * - cdfSize : array of cdf table sizes+1 (power of two: 2^k)
+ * - lenData : data vector length
+ *
+ * Output:
+ * - data : data vector
+ *
+ * Return value : number of bytes in the stream
+ * <0 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(
+ WebRtc_Word16 *data,
+ Bitstr_dec *streamData,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_UWord16 *cdfSize,
+ const WebRtc_Word16 lenData);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecHistOneStepMulti(...)
+ *
+ * Function to decode more symbols from the arithmetic bytestream, taking
+ * single step up or down at a time.
+ * cdf tables can be of arbitrary size, but large tables may take a lot of
+ * iterations.
+ *
+ * Input:
+ * - streamData : in-/output struct containing bitstream
+ * - cdf : array of cdf arrays
+ * - initIndex : vector of initial cdf table search entries
+ * - lenData : data vector length
+ *
+ * Output:
+ * - data : data vector
+ *
+ * Return value : number of bytes in original stream
+ * <0 if error detected
+ */
+WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(
+ WebRtc_Word16 *data,
+ Bitstr_dec *streamData,
+ const WebRtc_UWord16 **cdf,
+ const WebRtc_UWord16 *initIndex,
+ const WebRtc_Word16 lenData);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c
new file mode 100644
index 0000000..a274b66
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c
@@ -0,0 +1,1022 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * bandwidth_estimator.c
+ *
+ * This file contains the code for the Bandwidth Estimator designed
+ * for iSAC.
+ *
+ * NOTE! Castings needed for C55, do not remove!
+ *
+ */
+
+#include "bandwidth_estimator.h"
+#include "settings.h"
+
+
+/* array of quantization levels for bottle neck info; Matlab code: */
+/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
+static const WebRtc_Word16 kQRateTable[12] = {
+ 10000, 11115, 12355, 13733, 15265, 16967,
+ 18860, 20963, 23301, 25900, 28789, 32000
+};
+
+/* 0.1 times the values in the table kQRateTable */
+/* values are in Q16 */
+static const WebRtc_Word32 KQRate01[12] = {
+ 65536000, 72843264, 80969728, 90000589, 100040704, 111194931,
+ 123600896, 137383117, 152705434, 169738240, 188671590, 209715200
+};
+
+/* Bits per Bytes Seconds
+ * 8 bits/byte * 1000 msec/sec * 1/framelength (in msec)->bits/byte*sec
+ * frame length will either be 30 or 60 msec. 8738 is 1/60 in Q19 and 1/30 in Q18
+ * The following number is either in Q15 or Q14 depending on the current frame length */
+static const WebRtc_Word32 kBitsByteSec = 4369000;
+
+/* Received header rate. First value is for 30 ms packets and second for 60 ms */
+static const WebRtc_Word16 kRecHeaderRate[2] = {
+ 9333, 4666
+};
+
+/* Inverted minimum and maximum bandwidth in Q30.
+ minBwInv 30 ms, maxBwInv 30 ms,
+ minBwInv 60 ms, maxBwInv 69 ms
+*/
+static const WebRtc_Word32 kInvBandwidth[4] = {
+ 55539, 25978,
+ 73213, 29284
+};
+
+/* Number of samples in 25 msec */
+static const WebRtc_Word32 kSamplesIn25msec = 400;
+
+
+/****************************************************************************
+ * WebRtcIsacfix_InitBandwidthEstimator(...)
+ *
+ * This function initializes the struct for the bandwidth estimator
+ *
+ * Input/Output:
+ * - bweStr : Struct containing bandwidth information.
+ *
+ * Return value : 0
+ */
+WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr)
+{
+ bweStr->prevFrameSizeMs = INIT_FRAME_LEN;
+ bweStr->prevRtpNumber = 0;
+ bweStr->prevSendTime = 0;
+ bweStr->prevArrivalTime = 0;
+ bweStr->prevRtpRate = 1;
+ bweStr->lastUpdate = 0;
+ bweStr->lastReduction = 0;
+ bweStr->countUpdates = -9;
+
+ /* INIT_BN_EST = 20000
+ * INIT_BN_EST_Q7 = 2560000
+ * INIT_HDR_RATE = 4666
+ * INIT_REC_BN_EST_Q5 = 789312
+ *
+ * recBwInv = 1/(INIT_BN_EST + INIT_HDR_RATE) in Q30
+ * recBwAvg = INIT_BN_EST + INIT_HDR_RATE in Q5
+ */
+ bweStr->recBwInv = 43531;
+ bweStr->recBw = INIT_BN_EST;
+ bweStr->recBwAvgQ = INIT_BN_EST_Q7;
+ bweStr->recBwAvg = INIT_REC_BN_EST_Q5;
+ bweStr->recJitter = (WebRtc_Word32) 327680; /* 10 in Q15 */
+ bweStr->recJitterShortTerm = 0;
+ bweStr->recJitterShortTermAbs = (WebRtc_Word32) 40960; /* 5 in Q13 */
+ bweStr->recMaxDelay = (WebRtc_Word32) 10;
+ bweStr->recMaxDelayAvgQ = (WebRtc_Word32) 5120; /* 10 in Q9 */
+ bweStr->recHeaderRate = INIT_HDR_RATE;
+ bweStr->countRecPkts = 0;
+ bweStr->sendBwAvg = INIT_BN_EST_Q7;
+ bweStr->sendMaxDelayAvg = (WebRtc_Word32) 5120; /* 10 in Q9 */
+
+ bweStr->countHighSpeedRec = 0;
+ bweStr->highSpeedRec = 0;
+ bweStr->countHighSpeedSent = 0;
+ bweStr->highSpeedSend = 0;
+ bweStr->inWaitPeriod = 0;
+
+ /* Find the inverse of the max bw and min bw in Q30
+ * (1 / (MAX_ISAC_BW + INIT_HDR_RATE) in Q30
+ * (1 / (MIN_ISAC_BW + INIT_HDR_RATE) in Q30
+ */
+ bweStr->maxBwInv = kInvBandwidth[3];
+ bweStr->minBwInv = kInvBandwidth[2];
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_UpdateUplinkBwImpl(...)
+ *
+ * This function updates bottle neck rate received from other side in payload
+ * and calculates a new bottle neck to send to the other side.
+ *
+ * Input/Output:
+ * - bweStr : struct containing bandwidth information.
+ * - rtpNumber : value from RTP packet, from NetEq
+ * - frameSize : length of signal frame in ms, from iSAC decoder
+ * - sendTime : value in RTP header giving send time in samples
+ * - arrivalTime : value given by timeGetTime() time of arrival in
+ * samples of packet from NetEq
+ * - pksize : size of packet in bytes, from NetEq
+ * - Index : integer (range 0...23) indicating bottle neck &
+ * jitter as estimated by other side
+ *
+ * Return value : 0 if everything went fine,
+ * -1 otherwise
+ */
+WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
+ const WebRtc_UWord16 rtpNumber,
+ const WebRtc_Word16 frameSize,
+ const WebRtc_UWord32 sendTime,
+ const WebRtc_UWord32 arrivalTime,
+ const WebRtc_Word16 pksize,
+ const WebRtc_UWord16 Index)
+{
+ WebRtc_UWord16 weight = 0;
+ WebRtc_UWord32 currBwInv = 0;
+ WebRtc_UWord16 recRtpRate;
+ WebRtc_UWord32 arrTimeProj;
+ WebRtc_Word32 arrTimeDiff;
+ WebRtc_Word32 arrTimeNoise;
+ WebRtc_Word32 arrTimeNoiseAbs;
+ WebRtc_Word32 sendTimeDiff;
+
+ WebRtc_Word32 delayCorrFactor = DELAY_CORRECTION_MED;
+ WebRtc_Word32 lateDiff = 0;
+ WebRtc_Word16 immediateSet = 0;
+ WebRtc_Word32 frameSizeSampl;
+
+ WebRtc_Word32 temp;
+ WebRtc_Word32 msec;
+ WebRtc_UWord32 exponent;
+ WebRtc_UWord32 reductionFactor;
+ WebRtc_UWord32 numBytesInv;
+ WebRtc_Word32 sign;
+
+ WebRtc_UWord32 byteSecondsPerBit;
+ WebRtc_UWord32 tempLower;
+ WebRtc_UWord32 tempUpper;
+ WebRtc_Word32 recBwAvgInv;
+ WebRtc_Word32 numPktsExpected;
+
+ WebRtc_Word16 errCode;
+
+ /* UPDATE ESTIMATES FROM OTHER SIDE */
+
+ /* The function also checks if Index has a valid value */
+ errCode = WebRtcIsacfix_UpdateUplinkBwRec(bweStr, Index);
+ if (errCode <0) {
+ return(errCode);
+ }
+
+
+ /* UPDATE ESTIMATES ON THIS SIDE */
+
+ /* Bits per second per byte * 1/30 or 1/60 */
+ if (frameSize == 60) {
+ /* If frameSize changed since last call, from 30 to 60, recalculate some values */
+ if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) {
+ bweStr->countUpdates = 10;
+ bweStr->recHeaderRate = kRecHeaderRate[1];
+
+ bweStr->maxBwInv = kInvBandwidth[3];
+ bweStr->minBwInv = kInvBandwidth[2];
+ bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
+ }
+
+ /* kBitsByteSec is in Q15 */
+ recRtpRate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
+ (WebRtc_Word32)pksize), 15) + bweStr->recHeaderRate;
+
+ } else {
+ /* If frameSize changed since last call, from 60 to 30, recalculate some values */
+ if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) {
+ bweStr->countUpdates = 10;
+ bweStr->recHeaderRate = kRecHeaderRate[0];
+
+ bweStr->maxBwInv = kInvBandwidth[1];
+ bweStr->minBwInv = kInvBandwidth[0];
+ bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
+ }
+
+ /* kBitsByteSec is in Q14 */
+ recRtpRate = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
+ (WebRtc_Word32)pksize), 14) + bweStr->recHeaderRate;
+ }
+
+
+ /* Check for timer wrap-around */
+ if (arrivalTime < bweStr->prevArrivalTime) {
+ bweStr->prevArrivalTime = arrivalTime;
+ bweStr->lastUpdate = arrivalTime;
+ bweStr->lastReduction = arrivalTime + FS3;
+
+ bweStr->countRecPkts = 0;
+
+ /* store frame size */
+ bweStr->prevFrameSizeMs = frameSize;
+
+ /* store far-side transmission rate */
+ bweStr->prevRtpRate = recRtpRate;
+
+ /* store far-side RTP time stamp */
+ bweStr->prevRtpNumber = rtpNumber;
+
+ return 0;
+ }
+
+ bweStr->countRecPkts++;
+
+ /* Calculate framesize in msec */
+ frameSizeSampl = WEBRTC_SPL_MUL_16_16((WebRtc_Word16)SAMPLES_PER_MSEC, frameSize);
+
+ /* Check that it's not one of the first 9 packets */
+ if ( bweStr->countUpdates > 0 ) {
+
+ /* Stay in Wait Period for 1.5 seconds (no updates in wait period) */
+ if(bweStr->inWaitPeriod) {
+ if ((arrivalTime - bweStr->startWaitPeriod)> FS_1_HALF) {
+ bweStr->inWaitPeriod = 0;
+ }
+ }
+
+ /* If not been updated for a long time, reduce the BN estimate */
+
+ /* Check send time difference between this packet and previous received */
+ sendTimeDiff = sendTime - bweStr->prevSendTime;
+ if (sendTimeDiff <= WEBRTC_SPL_LSHIFT_W32(frameSizeSampl, 1)) {
+
+ /* Only update if 3 seconds has past since last update */
+ if ((arrivalTime - bweStr->lastUpdate) > FS3) {
+
+ /* Calculate expected number of received packets since last update */
+ numPktsExpected = WEBRTC_SPL_UDIV(arrivalTime - bweStr->lastUpdate, frameSizeSampl);
+
+ /* If received number of packets is more than 90% of expected (922 = 0.9 in Q10): */
+ /* do the update, else not */
+ if(WEBRTC_SPL_LSHIFT_W32(bweStr->countRecPkts, 10) > WEBRTC_SPL_MUL_16_16(922, numPktsExpected)) {
+ /* Q4 chosen to approx dividing by 16 */
+ msec = (arrivalTime - bweStr->lastReduction);
+
+ /* the number below represents 13 seconds, highly unlikely
+ but to insure no overflow when reduction factor is multiplied by recBw inverse */
+ if (msec > 208000) {
+ msec = 208000;
+ }
+
+ /* Q20 2^(negative number: - 76/1048576) = .99995
+ product is Q24 */
+ exponent = WEBRTC_SPL_UMUL(0x0000004C, msec);
+
+ /* do the approx with positive exponent so that value is actually rf^-1
+ and multiply by bw inverse */
+ reductionFactor = WEBRTC_SPL_RSHIFT_U32(0x01000000 | (exponent & 0x00FFFFFF),
+ WEBRTC_SPL_RSHIFT_U32(exponent, 24));
+
+ /* reductionFactor in Q13 */
+ reductionFactor = WEBRTC_SPL_RSHIFT_U32(reductionFactor, 11);
+
+ if ( reductionFactor != 0 ) {
+ bweStr->recBwInv = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recBwInv, (WebRtc_Word32)reductionFactor);
+ bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwInv, 13);
+
+ } else {
+ /* recBwInv = 1 / (INIT_BN_EST + INIT_HDR_RATE) in Q26 (Q30??)*/
+ bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
+ WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
+ }
+
+ /* reset time-since-update counter */
+ bweStr->lastReduction = arrivalTime;
+ } else {
+ /* Delay last reduction with 3 seconds */
+ bweStr->lastReduction = arrivalTime + FS3;
+ bweStr->lastUpdate = arrivalTime;
+ bweStr->countRecPkts = 0;
+ }
+ }
+ } else {
+ bweStr->lastReduction = arrivalTime + FS3;
+ bweStr->lastUpdate = arrivalTime;
+ bweStr->countRecPkts = 0;
+ }
+
+
+ /* update only if previous packet was not lost */
+ if ( rtpNumber == bweStr->prevRtpNumber + 1 ) {
+ arrTimeDiff = arrivalTime - bweStr->prevArrivalTime;
+
+ if (!(bweStr->highSpeedSend && bweStr->highSpeedRec)) {
+ if (arrTimeDiff > frameSizeSampl) {
+ if (sendTimeDiff > 0) {
+ lateDiff = arrTimeDiff - sendTimeDiff -
+ WEBRTC_SPL_LSHIFT_W32(frameSizeSampl, 1);
+ } else {
+ lateDiff = arrTimeDiff - frameSizeSampl;
+ }
+
+ /* 8000 is 1/2 second (in samples at FS) */
+ if (lateDiff > 8000) {
+ delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MAX;
+ bweStr->inWaitPeriod = 1;
+ bweStr->startWaitPeriod = arrivalTime;
+ immediateSet = 1;
+ } else if (lateDiff > 5120) {
+ delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MED;
+ immediateSet = 1;
+ bweStr->inWaitPeriod = 1;
+ bweStr->startWaitPeriod = arrivalTime;
+ }
+ }
+ }
+
+ if ((bweStr->prevRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->recBwAvg, 5)) &&
+ (recRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwAvg, 5)) &&
+ !bweStr->inWaitPeriod) {
+
+ /* test if still in initiation period and increment counter */
+ if (bweStr->countUpdates++ > 99) {
+ /* constant weight after initiation part, 0.01 in Q13 */
+ weight = (WebRtc_UWord16) 82;
+ } else {
+ /* weight decreases with number of updates, 1/countUpdates in Q13 */
+ weight = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
+ (WebRtc_Word32)(8192 + WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->countUpdates, 1)),
+ (WebRtc_Word16)bweStr->countUpdates);
+ }
+
+ /* Bottle Neck Estimation */
+
+ /* limit outliers, if more than 25 ms too much */
+ if (arrTimeDiff > frameSizeSampl + kSamplesIn25msec) {
+ arrTimeDiff = frameSizeSampl + kSamplesIn25msec;
+ }
+
+ /* don't allow it to be less than frame rate - 10 ms */
+ if (arrTimeDiff < frameSizeSampl - FRAMESAMPLES_10ms) {
+ arrTimeDiff = frameSizeSampl - FRAMESAMPLES_10ms;
+ }
+
+ /* compute inverse receiving rate for last packet, in Q19 */
+ numBytesInv = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
+ (WebRtc_Word32)(524288 + WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)pksize + HEADER_SIZE), 1)),
+ (WebRtc_Word16)(pksize + HEADER_SIZE));
+
+ /* 8389 is ~ 1/128000 in Q30 */
+ byteSecondsPerBit = WEBRTC_SPL_MUL_16_16(arrTimeDiff, 8389);
+
+ /* get upper N bits */
+ tempUpper = WEBRTC_SPL_RSHIFT_U32(byteSecondsPerBit, 15);
+
+ /* get lower 15 bits */
+ tempLower = byteSecondsPerBit & 0x00007FFF;
+
+ tempUpper = WEBRTC_SPL_MUL(tempUpper, numBytesInv);
+ tempLower = WEBRTC_SPL_MUL(tempLower, numBytesInv);
+ tempLower = WEBRTC_SPL_RSHIFT_U32(tempLower, 15);
+
+ currBwInv = tempUpper + tempLower;
+ currBwInv = WEBRTC_SPL_RSHIFT_U32(currBwInv, 4);
+
+ /* Limit inv rate. Note that minBwInv > maxBwInv! */
+ if(currBwInv < bweStr->maxBwInv) {
+ currBwInv = bweStr->maxBwInv;
+ } else if(currBwInv > bweStr->minBwInv) {
+ currBwInv = bweStr->minBwInv;
+ }
+
+ /* update bottle neck rate estimate */
+ bweStr->recBwInv = WEBRTC_SPL_UMUL(weight, currBwInv) +
+ WEBRTC_SPL_UMUL((WebRtc_UWord32) 8192 - weight, bweStr->recBwInv);
+
+ /* Shift back to Q30 from Q40 (actual used bits shouldn't be more than 27 based on minBwInv)
+ up to 30 bits used with Q13 weight */
+ bweStr->recBwInv = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwInv, 13);
+
+ /* reset time-since-update counter */
+ bweStr->lastUpdate = arrivalTime;
+ bweStr->lastReduction = arrivalTime + FS3;
+ bweStr->countRecPkts = 0;
+
+ /* to save resolution compute the inverse of recBwAvg in Q26 by left shifting numerator to 2^31
+ and NOT right shifting recBwAvg 5 bits to an integer
+ At max 13 bits are used
+ shift to Q5 */
+ recBwAvgInv = WEBRTC_SPL_UDIV((WebRtc_UWord32)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
+ bweStr->recBwAvg);
+
+ /* Calculate Projected arrival time difference */
+
+ /* The numerator of the quotient can be 22 bits so right shift inv by 4 to avoid overflow
+ result in Q22 */
+ arrTimeProj = WEBRTC_SPL_MUL((WebRtc_Word32)8000, recBwAvgInv);
+ /* shift to Q22 */
+ arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 4);
+ /* complete calulation */
+ arrTimeProj = WEBRTC_SPL_MUL(((WebRtc_Word32)pksize + HEADER_SIZE), arrTimeProj);
+ /* shift to Q10 */
+ arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 12);
+
+ /* difference between projected and actual arrival time differences */
+ /* Q9 (only shift arrTimeDiff by 5 to simulate divide by 16 (need to revisit if change sampling rate) DH */
+ if (WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) > (WebRtc_Word32)arrTimeProj) {
+ arrTimeNoise = WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) - arrTimeProj;
+ sign = 1;
+ } else {
+ arrTimeNoise = arrTimeProj - WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6);
+ sign = -1;
+ }
+
+ /* Q9 */
+ arrTimeNoiseAbs = arrTimeNoise;
+
+ /* long term averaged absolute jitter, Q15 */
+ weight = WEBRTC_SPL_RSHIFT_W32(weight, 3);
+ bweStr->recJitter = WEBRTC_SPL_MUL(weight, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 5))
+ + WEBRTC_SPL_MUL(1024 - weight, bweStr->recJitter);
+
+ /* remove the fractional portion */
+ bweStr->recJitter = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitter, 10);
+
+ /* Maximum jitter is 10 msec in Q15 */
+ if (bweStr->recJitter > (WebRtc_Word32)327680) {
+ bweStr->recJitter = (WebRtc_Word32)327680;
+ }
+
+ /* short term averaged absolute jitter */
+ /* Calculation in Q13 products in Q23 */
+ bweStr->recJitterShortTermAbs = WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 3)) +
+ WEBRTC_SPL_MUL(973, bweStr->recJitterShortTermAbs);
+ bweStr->recJitterShortTermAbs = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTermAbs , 10);
+
+ /* short term averaged jitter */
+ /* Calculation in Q13 products in Q23 */
+ bweStr->recJitterShortTerm = WEBRTC_SPL_MUL(205, WEBRTC_SPL_LSHIFT_W32(arrTimeNoise, 3)) * sign +
+ WEBRTC_SPL_MUL(3891, bweStr->recJitterShortTerm);
+
+ if (bweStr->recJitterShortTerm < 0) {
+ temp = -bweStr->recJitterShortTerm;
+ temp = WEBRTC_SPL_RSHIFT_W32(temp, 12);
+ bweStr->recJitterShortTerm = -temp;
+ } else {
+ bweStr->recJitterShortTerm = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 12);
+ }
+ }
+ }
+ } else {
+ /* reset time-since-update counter when receiving the first 9 packets */
+ bweStr->lastUpdate = arrivalTime;
+ bweStr->lastReduction = arrivalTime + FS3;
+ bweStr->countRecPkts = 0;
+ bweStr->countUpdates++;
+ }
+
+ /* Limit to minimum or maximum bottle neck rate (in Q30) */
+ if (bweStr->recBwInv > bweStr->minBwInv) {
+ bweStr->recBwInv = bweStr->minBwInv;
+ } else if (bweStr->recBwInv < bweStr->maxBwInv) {
+ bweStr->recBwInv = bweStr->maxBwInv;
+ }
+
+
+ /* store frame length */
+ bweStr->prevFrameSizeMs = frameSize;
+
+ /* store far-side transmission rate */
+ bweStr->prevRtpRate = recRtpRate;
+
+ /* store far-side RTP time stamp */
+ bweStr->prevRtpNumber = rtpNumber;
+
+ /* Replace bweStr->recMaxDelay by the new value (atomic operation) */
+ if (bweStr->prevArrivalTime != 0xffffffff) {
+ bweStr->recMaxDelay = WEBRTC_SPL_MUL(3, bweStr->recJitter);
+ }
+
+ /* store arrival time stamp */
+ bweStr->prevArrivalTime = arrivalTime;
+ bweStr->prevSendTime = sendTime;
+
+ /* Replace bweStr->recBw by the new value */
+ bweStr->recBw = WEBRTC_SPL_UDIV(1073741824, bweStr->recBwInv) - bweStr->recHeaderRate;
+
+ if (immediateSet) {
+ /* delay correction factor is in Q10 */
+ bweStr->recBw = WEBRTC_SPL_UMUL(delayCorrFactor, bweStr->recBw);
+ bweStr->recBw = WEBRTC_SPL_RSHIFT_U32(bweStr->recBw, 10);
+
+ if (bweStr->recBw < (WebRtc_Word32) MIN_ISAC_BW) {
+ bweStr->recBw = (WebRtc_Word32) MIN_ISAC_BW;
+ }
+
+ bweStr->recBwAvg = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw + bweStr->recHeaderRate, 5);
+
+ bweStr->recBwAvgQ = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw, 7);
+
+ bweStr->recJitterShortTerm = 0;
+
+ bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, bweStr->recBw + bweStr->recHeaderRate);
+
+ immediateSet = 0;
+ }
+
+
+ return 0;
+}
+
+/* This function updates the send bottle neck rate */
+/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
+/* returns 0 if everything went fine, -1 otherwise */
+WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
+ const WebRtc_Word16 Index)
+{
+ WebRtc_UWord16 RateInd;
+
+ if ( (Index < 0) || (Index > 23) ) {
+ return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
+ }
+
+ /* UPDATE ESTIMATES FROM OTHER SIDE */
+
+ if ( Index > 11 ) {
+ RateInd = Index - 12;
+ /* compute the jitter estimate as decoded on the other side in Q9 */
+ /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MAX_ISAC_MD */
+ bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
+ WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MAX_ISAC_MD, 9));
+ bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
+
+ } else {
+ RateInd = Index;
+ /* compute the jitter estimate as decoded on the other side in Q9 */
+ /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MIN_ISAC_MD */
+ bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
+ WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MIN_ISAC_MD,9));
+ bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
+
+ }
+
+
+ /* compute the BN estimate as decoded on the other side */
+ /* sendBwAvg = 0.9 * sendBwAvg + 0.1 * kQRateTable[RateInd]; */
+ bweStr->sendBwAvg = WEBRTC_SPL_UMUL(461, bweStr->sendBwAvg) +
+ WEBRTC_SPL_UMUL(51, WEBRTC_SPL_LSHIFT_U32(kQRateTable[RateInd], 7));
+ bweStr->sendBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 9);
+
+
+ if (WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7) > 28000 && !bweStr->highSpeedSend) {
+ bweStr->countHighSpeedSent++;
+
+ /* approx 2 seconds with 30ms frames */
+ if (bweStr->countHighSpeedSent >= 66) {
+ bweStr->highSpeedSend = 1;
+ }
+ } else if (!bweStr->highSpeedSend) {
+ bweStr->countHighSpeedSent = 0;
+ }
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
+ *
+ * This function calculates and returns the bandwidth/jitter estimation code
+ * (integer 0...23) to put in the sending iSAC payload.
+ *
+ * Input:
+ * - bweStr : BWE struct
+ *
+ * Return:
+ * bandwith and jitter index (0..23)
+ */
+WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr)
+{
+ WebRtc_Word32 rate;
+ WebRtc_Word32 maxDelay;
+ WebRtc_UWord16 rateInd;
+ WebRtc_UWord16 maxDelayBit;
+ WebRtc_Word32 tempTerm1;
+ WebRtc_Word32 tempTerm2;
+ WebRtc_Word32 tempTermX;
+ WebRtc_Word32 tempTermY;
+ WebRtc_Word32 tempMin;
+ WebRtc_Word32 tempMax;
+
+ /* Get Rate Index */
+
+ /* Get unquantized rate. Always returns 10000 <= rate <= 32000 */
+ rate = WebRtcIsacfix_GetDownlinkBandwidth(bweStr);
+
+ /* Compute the averaged BN estimate on this side */
+
+ /* recBwAvg = 0.9 * recBwAvg + 0.1 * (rate + bweStr->recHeaderRate), 0.9 and 0.1 in Q9 */
+ bweStr->recBwAvg = WEBRTC_SPL_UMUL(922, bweStr->recBwAvg) +
+ WEBRTC_SPL_UMUL(102, WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)rate + bweStr->recHeaderRate, 5));
+ bweStr->recBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 10);
+
+ /* find quantization index that gives the closest rate after averaging */
+ for (rateInd = 1; rateInd < 12; rateInd++) {
+ if (rate <= kQRateTable[rateInd]){
+ break;
+ }
+ }
+
+ /* find closest quantization index, and update quantized average by taking: */
+ /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */
+
+ /* 0.9 times recBwAvgQ in Q16 */
+ /* 461/512 - 25/65536 =0.900009 */
+ tempTerm1 = WEBRTC_SPL_MUL(bweStr->recBwAvgQ, 25);
+ tempTerm1 = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 7);
+ tempTermX = WEBRTC_SPL_UMUL(461, bweStr->recBwAvgQ) - tempTerm1;
+
+ /* rate in Q16 */
+ tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)rate, 16);
+
+ /* 0.1 * kQRateTable[rateInd] = KQRate01[rateInd] */
+ tempTerm1 = tempTermX + KQRate01[rateInd] - tempTermY;
+ tempTerm2 = tempTermY - tempTermX - KQRate01[rateInd-1];
+
+ /* Compare (0.9 * recBwAvgQ + 0.1 * kQRateTable[rateInd] - rate) >
+ (rate - 0.9 * recBwAvgQ - 0.1 * kQRateTable[rateInd-1]) */
+ if (tempTerm1 > tempTerm2) {
+ rateInd--;
+ }
+
+ /* Update quantized average by taking: */
+ /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */
+
+ /* Add 0.1 times kQRateTable[rateInd], in Q16 */
+ tempTermX += KQRate01[rateInd];
+
+ /* Shift back to Q7 */
+ bweStr->recBwAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTermX, 9);
+
+ /* Count consecutive received bandwidth above 28000 kbps (28000 in Q7 = 3584000) */
+ /* If 66 high estimates in a row, set highSpeedRec to one */
+ /* 66 corresponds to ~2 seconds in 30 msec mode */
+ if ((bweStr->recBwAvgQ > 3584000) && !bweStr->highSpeedRec) {
+ bweStr->countHighSpeedRec++;
+ if (bweStr->countHighSpeedRec >= 66) {
+ bweStr->highSpeedRec = 1;
+ }
+ } else if (!bweStr->highSpeedRec) {
+ bweStr->countHighSpeedRec = 0;
+ }
+
+ /* Get Max Delay Bit */
+
+ /* get unquantized max delay */
+ maxDelay = WebRtcIsacfix_GetDownlinkMaxDelay(bweStr);
+
+ /* Update quantized max delay average */
+ tempMax = 652800; /* MAX_ISAC_MD * 0.1 in Q18 */
+ tempMin = 130560; /* MIN_ISAC_MD * 0.1 in Q18 */
+ tempTermX = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recMaxDelayAvgQ, (WebRtc_Word32)461);
+ tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)maxDelay, 18);
+
+ tempTerm1 = tempTermX + tempMax - tempTermY;
+ tempTerm2 = tempTermY - tempTermX - tempMin;
+
+ if ( tempTerm1 > tempTerm2) {
+ maxDelayBit = 0;
+ tempTerm1 = tempTermX + tempMin;
+
+ /* update quantized average, shift back to Q9 */
+ bweStr->recMaxDelayAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 9);
+ } else {
+ maxDelayBit = 12;
+ tempTerm1 = tempTermX + tempMax;
+
+ /* update quantized average, shift back to Q9 */
+ bweStr->recMaxDelayAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 9);
+ }
+
+ /* Return bandwitdh and jitter index (0..23) */
+ return (WebRtc_UWord16)(rateInd + maxDelayBit);
+}
+
+/* get the bottle neck rate from far side to here, as estimated on this side */
+WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
+{
+ WebRtc_UWord32 recBw;
+ WebRtc_Word32 jitter_sign; /* Q8 */
+ WebRtc_Word32 bw_adjust; /* Q16 */
+ WebRtc_Word32 rec_jitter_short_term_abs_inv; /* Q18 */
+ WebRtc_Word32 temp;
+
+ /* Q18 rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
+ 2^18 then needs to be shifted 13 bits to 2^31 */
+ rec_jitter_short_term_abs_inv = WEBRTC_SPL_UDIV(0x80000000, bweStr->recJitterShortTermAbs);
+
+ /* Q27 = 9 + 18 */
+ jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (WebRtc_Word32)rec_jitter_short_term_abs_inv);
+
+ if (jitter_sign < 0) {
+ temp = -jitter_sign;
+ temp = WEBRTC_SPL_RSHIFT_W32(temp, 19);
+ jitter_sign = -temp;
+ } else {
+ jitter_sign = WEBRTC_SPL_RSHIFT_W32(jitter_sign, 19);
+ }
+
+ /* adjust bw proportionally to negative average jitter sign */
+ //bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign);
+ //Q8 -> Q16 .15 +.15 * jitter^2 first term is .15 in Q16 latter term is Q8*Q8*Q8
+ //38 in Q8 ~.15 9830 in Q16 ~.15
+ temp = 9830 + WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL(38, WEBRTC_SPL_MUL(jitter_sign, jitter_sign))), 8);
+
+ if (jitter_sign < 0) {
+ temp = WEBRTC_SPL_MUL(jitter_sign, temp);
+ temp = -temp;
+ temp = WEBRTC_SPL_RSHIFT_W32(temp, 8);
+ bw_adjust = (WebRtc_UWord32)65536 + temp; /* (1 << 16) + temp; */
+ } else {
+ bw_adjust = (WebRtc_UWord32)65536 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(jitter_sign, temp), 8);/* (1 << 16) - ((jitter_sign * temp) >> 8); */
+ }
+
+ //make sure following multiplication won't overflow
+ //bw adjust now Q14
+ bw_adjust = WEBRTC_SPL_RSHIFT_W32(bw_adjust, 2);//see if good resolution is maintained
+
+ /* adjust Rate if jitter sign is mostly constant */
+ recBw = WEBRTC_SPL_UMUL(bweStr->recBw, bw_adjust);
+
+ recBw = WEBRTC_SPL_RSHIFT_W32(recBw, 14);
+
+ /* limit range of bottle neck rate */
+ if (recBw < MIN_ISAC_BW) {
+ recBw = MIN_ISAC_BW;
+ } else if (recBw > MAX_ISAC_BW) {
+ recBw = MAX_ISAC_BW;
+ }
+
+ return (WebRtc_UWord16) recBw;
+}
+
+/* Returns the mmax delay (in ms) */
+WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
+{
+ WebRtc_Word16 recMaxDelay;
+
+ recMaxDelay = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(bweStr->recMaxDelay, 15);
+
+ /* limit range of jitter estimate */
+ if (recMaxDelay < MIN_ISAC_MD) {
+ recMaxDelay = MIN_ISAC_MD;
+ } else if (recMaxDelay > MAX_ISAC_MD) {
+ recMaxDelay = MAX_ISAC_MD;
+ }
+
+ return recMaxDelay;
+}
+
+/* get the bottle neck rate from here to far side, as estimated by far side */
+WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bweStr)
+{
+ WebRtc_Word16 send_bw;
+
+ send_bw = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7);
+
+ /* limit range of bottle neck rate */
+ if (send_bw < MIN_ISAC_BW) {
+ send_bw = MIN_ISAC_BW;
+ } else if (send_bw > MAX_ISAC_BW) {
+ send_bw = MAX_ISAC_BW;
+ }
+
+ return send_bw;
+}
+
+
+
+/* Returns the max delay value from the other side in ms */
+WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bweStr)
+{
+ WebRtc_Word16 send_max_delay;
+
+ send_max_delay = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
+
+ /* limit range of jitter estimate */
+ if (send_max_delay < MIN_ISAC_MD) {
+ send_max_delay = MIN_ISAC_MD;
+ } else if (send_max_delay > MAX_ISAC_MD) {
+ send_max_delay = MAX_ISAC_MD;
+ }
+
+ return send_max_delay;
+}
+
+
+
+
+/*
+ * update long-term average bitrate and amount of data in buffer
+ * returns minimum payload size (bytes)
+ */
+WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
+ WebRtc_Word16 StreamSize, /* bytes in bitstream */
+ const WebRtc_Word16 FrameSamples, /* samples per frame */
+ const WebRtc_Word16 BottleNeck, /* bottle neck rate; excl headers (bps) */
+ const WebRtc_Word16 DelayBuildUp) /* max delay from bottle neck buffering (ms) */
+{
+ WebRtc_Word32 MinRate = 0;
+ WebRtc_UWord16 MinBytes;
+ WebRtc_Word16 TransmissionTime;
+ WebRtc_Word32 inv_Q12;
+ WebRtc_Word32 den;
+
+
+ /* first 10 packets @ low rate, then INIT_BURST_LEN packets @ fixed rate of INIT_RATE bps */
+ if (State->InitCounter > 0) {
+ if (State->InitCounter-- <= INIT_BURST_LEN) {
+ MinRate = INIT_RATE;
+ } else {
+ MinRate = 0;
+ }
+ } else {
+ /* handle burst */
+ if (State->BurstCounter) {
+ if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
+ /* max bps derived from BottleNeck and DelayBuildUp values */
+ inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples));
+ MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp, inv_Q12), 3)), BottleNeck);
+ } else {
+ /* max bps derived from StillBuffered and DelayBuildUp values */
+ inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples);
+ if (DelayBuildUp > State->StillBuffered) {
+ MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp - State->StillBuffered, inv_Q12), 3)), BottleNeck);
+ } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
+ /* MinRate will be negative here */
+ MinRate = 0;
+ } else {
+ MinRate = WEBRTC_SPL_MUL((512 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(den, inv_Q12), 3)), BottleNeck);
+ }
+ //if (MinRate < 1.04 * BottleNeck)
+ // MinRate = 1.04 * BottleNeck;
+ //Q9
+ if (MinRate < WEBRTC_SPL_MUL(532, BottleNeck)) {
+ MinRate += WEBRTC_SPL_MUL(22, BottleNeck);
+ }
+ }
+
+ State->BurstCounter--;
+ }
+ }
+
+
+ /* convert rate from bits/second to bytes/packet */
+ //round and shift before conversion
+ MinRate += 256;
+ MinRate = WEBRTC_SPL_RSHIFT_W32(MinRate, 9);
+ MinBytes = (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
+
+ /* StreamSize will be adjusted if less than MinBytes */
+ if (StreamSize < MinBytes) {
+ StreamSize = MinBytes;
+ }
+
+ /* keep track of when bottle neck was last exceeded by at least 1% */
+ //517/512 ~ 1.01
+ if (WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, FS8), FrameSamples) > (WEBRTC_SPL_MUL(517, BottleNeck) >> 9)) {
+ if (State->PrevExceed) {
+ /* bottle_neck exceded twice in a row, decrease ExceedAgo */
+ State->ExceedAgo -= WEBRTC_SPL_DIV(BURST_INTERVAL, BURST_LEN - 1);
+ if (State->ExceedAgo < 0) {
+ State->ExceedAgo = 0;
+ }
+ } else {
+ State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
+ State->PrevExceed = 1;
+ }
+ } else {
+ State->PrevExceed = 0;
+ State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
+ }
+
+ /* set burst flag if bottle neck not exceeded for long time */
+ if ((State->ExceedAgo > BURST_INTERVAL) && (State->BurstCounter == 0)) {
+ if (State->PrevExceed) {
+ State->BurstCounter = BURST_LEN - 1;
+ } else {
+ State->BurstCounter = BURST_LEN;
+ }
+ }
+
+
+ /* Update buffer delay */
+ TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck); /* ms */
+ State->StillBuffered += TransmissionTime;
+ State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); //>>4 = SAMPLES_PER_MSEC /* ms */
+ if (State->StillBuffered < 0) {
+ State->StillBuffered = 0;
+ }
+
+ if (State->StillBuffered > 2000) {
+ State->StillBuffered = 2000;
+ }
+
+ return MinBytes;
+}
+
+
+/*
+ * update long-term average bitrate and amount of data in buffer
+ */
+void WebRtcIsacfix_UpdateRateModel(RateModel *State,
+ WebRtc_Word16 StreamSize, /* bytes in bitstream */
+ const WebRtc_Word16 FrameSamples, /* samples per frame */
+ const WebRtc_Word16 BottleNeck) /* bottle neck rate; excl headers (bps) */
+{
+ WebRtc_Word16 TransmissionTime;
+
+ /* avoid the initial "high-rate" burst */
+ State->InitCounter = 0;
+
+ /* Update buffer delay */
+ TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck); /* ms */
+ State->StillBuffered += TransmissionTime;
+ State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
+ if (State->StillBuffered < 0) {
+ State->StillBuffered = 0;
+ }
+
+}
+
+
+void WebRtcIsacfix_InitRateModel(RateModel *State)
+{
+ State->PrevExceed = 0; /* boolean */
+ State->ExceedAgo = 0; /* ms */
+ State->BurstCounter = 0; /* packets */
+ State->InitCounter = INIT_BURST_LEN + 10; /* packets */
+ State->StillBuffered = 1; /* ms */
+}
+
+
+
+
+
+WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framesamples)
+{
+ WebRtc_Word16 new_framesamples;
+
+ new_framesamples = current_framesamples;
+
+ /* find new framelength */
+ switch(current_framesamples) {
+ case 480:
+ if (bottle_neck < Thld_30_60) {
+ new_framesamples = 960;
+ }
+ break;
+ case 960:
+ if (bottle_neck >= Thld_60_30) {
+ new_framesamples = 480;
+ }
+ break;
+ default:
+ new_framesamples = -1; /* Error */
+ }
+
+ return new_framesamples;
+}
+
+WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples)
+{
+ WebRtc_Word16 s2nr = 0;
+
+ /* find new SNR value */
+ //consider BottleNeck to be in Q10 ( * 1 in Q10)
+ switch(framesamples) {
+ case 480:
+ /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/
+ s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+ break;
+ case 960:
+ /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
+ s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+ break;
+ default:
+ s2nr = -1; /* Error */
+ }
+
+ return s2nr; //return in Q10
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h
new file mode 100644
index 0000000..76a50f8
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * bandwidth_estimator.h
+ *
+ * This header file contains the API for the Bandwidth Estimator
+ * designed for iSAC.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
+
+#include "structs.h"
+
+
+/****************************************************************************
+ * WebRtcIsacfix_InitBandwidthEstimator(...)
+ *
+ * This function initializes the struct for the bandwidth estimator
+ *
+ * Input/Output:
+ * - bwest_str : Struct containing bandwidth information.
+ *
+ * Return value : 0
+ */
+
+WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
+
+
+/****************************************************************************
+ * WebRtcIsacfix_UpdateUplinkBwImpl(...)
+ *
+ * This function updates bottle neck rate received from other side in payload
+ * and calculates a new bottle neck to send to the other side.
+ *
+ * Input/Output:
+ * - bweStr : struct containing bandwidth information.
+ * - rtpNumber : value from RTP packet, from NetEq
+ * - frameSize : length of signal frame in ms, from iSAC decoder
+ * - sendTime : value in RTP header giving send time in samples
+ * - arrivalTime : value given by timeGetTime() time of arrival in
+ * samples of packet from NetEq
+ * - pksize : size of packet in bytes, from NetEq
+ * - Index : integer (range 0...23) indicating bottle neck &
+ * jitter as estimated by other side
+ *
+ * Return value : 0 if everything went fine,
+ * -1 otherwise
+ */
+
+WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bwest_str,
+ const WebRtc_UWord16 rtp_number,
+ const WebRtc_Word16 frameSize,
+ const WebRtc_UWord32 send_ts,
+ const WebRtc_UWord32 arr_ts,
+ const WebRtc_Word16 pksize,
+ const WebRtc_UWord16 Index);
+
+/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
+WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
+ const WebRtc_Word16 Index);
+
+/****************************************************************************
+ * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
+ *
+ * This function calculates and returns the bandwidth/jitter estimation code
+ * (integer 0...23) to put in the sending iSAC payload.
+ *
+ * Input:
+ * - bweStr : BWE struct
+ *
+ * Return:
+ * bandwith and jitter index (0..23)
+ */
+WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bwest_str);
+
+/* Returns the bandwidth estimation (in bps) */
+WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bwest_str);
+
+/* Returns the bandwidth that iSAC should send with in bps */
+WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str);
+
+/* Returns the max delay (in ms) */
+WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
+
+/* Returns the max delay value from the other side in ms */
+WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
+
+/*
+ * update amount of data in bottle neck buffer and burst handling
+ * returns minimum payload size (bytes)
+ */
+WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
+ WebRtc_Word16 StreamSize, /* bytes in bitstream */
+ const WebRtc_Word16 FrameLen, /* ms per frame */
+ const WebRtc_Word16 BottleNeck, /* bottle neck rate; excl headers (bps) */
+ const WebRtc_Word16 DelayBuildUp); /* max delay from bottle neck buffering (ms) */
+
+/*
+ * update long-term average bitrate and amount of data in buffer
+ */
+void WebRtcIsacfix_UpdateRateModel(RateModel *State,
+ WebRtc_Word16 StreamSize, /* bytes in bitstream */
+ const WebRtc_Word16 FrameSamples, /* samples per frame */
+ const WebRtc_Word16 BottleNeck); /* bottle neck rate; excl headers (bps) */
+
+
+void WebRtcIsacfix_InitRateModel(RateModel *State);
+
+/* Returns the new framelength value (input argument: bottle_neck) */
+WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framelength);
+
+/* Returns the new SNR value (input argument: bottle_neck) */
+//returns snr in Q10
+WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples);
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h
new file mode 100644
index 0000000..154f2e9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h
@@ -0,0 +1,183 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * codec.h
+ *
+ * This header file contains the calls to the internal encoder
+ * and decoder functions.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
+
+#include "structs.h"
+
+
+int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
+ Bitstr_dec *streamdata,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts);
+
+WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16 *signal_out16,
+ ISACFIX_DecInst_t *ISACdec_obj,
+ WebRtc_Word16 *current_framesamples);
+
+WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16 *decoded,
+ ISACFIX_DecInst_t *ISACdec_obj,
+ WebRtc_Word16 *current_framesample );
+
+int WebRtcIsacfix_EncodeImpl(WebRtc_Word16 *in,
+ ISACFIX_EncInst_t *ISACenc_obj,
+ BwEstimatorstr *bw_estimatordata,
+ WebRtc_Word16 CodingMode);
+
+int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t *ISACenc_obj,
+ int BWnumber,
+ float scale);
+
+/************************** initialization functions *************************/
+
+void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata);
+void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec *maskdata);
+
+void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr *prefiltdata);
+
+void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr *postfiltdata);
+
+void WebRtcIsacfix_InitPitchFilter(PitchFiltstr *pitchfiltdata);
+
+void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct *State);
+
+void WebRtcIsacfix_InitPlc( PLCstr *State );
+
+
+/**************************** transform functions ****************************/
+
+void WebRtcIsacfix_InitTransform();
+
+
+void WebRtcIsacfix_Time2Spec(WebRtc_Word16 *inre1Q9,
+ WebRtc_Word16 *inre2Q9,
+ WebRtc_Word16 *outre,
+ WebRtc_Word16 *outim);
+
+
+
+void WebRtcIsacfix_Spec2Time(WebRtc_Word16 *inreQ7,
+ WebRtc_Word16 *inimQ7,
+ WebRtc_Word32 *outre1Q16,
+ WebRtc_Word32 *outre2Q16);
+
+
+
+
+/***************************** filterbank functions **************************/
+
+
+
+void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16 *in,
+ WebRtc_Word16 *LP16,
+ WebRtc_Word16 *HP16,
+ PreFiltBankstr *prefiltdata);
+
+void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16 *tempin_ch1,
+ WebRtc_Word16 *tempin_ch2,
+ WebRtc_Word16 *out16,
+ PostFiltBankstr *postfiltdata);
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+
+void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16 *in,
+ WebRtc_Word16 *LP16,
+ WebRtc_Word16 *HP16,
+ PreFiltBankstr *prefiltdata);
+
+void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16 *tempin_ch1,
+ WebRtc_Word16 *tempin_ch2,
+ WebRtc_Word16 *out16,
+ PostFiltBankstr *postfiltdata,
+ WebRtc_Word16 len);
+
+#endif
+
+/************************* normalized lattice filters ************************/
+
+
+void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
+ WebRtc_Word32 *stateGQ15,
+ WebRtc_Word16 *lat_inQ0,
+ WebRtc_Word16 *filt_coefQ15,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 lo_hi,
+ WebRtc_Word16 *lat_outQ9);
+
+void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
+ WebRtc_Word16 *stateGQ0,
+ WebRtc_Word32 *lat_inQ25,
+ WebRtc_Word16 *filt_coefQ15,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 lo_hi,
+ WebRtc_Word16 *lat_outQ0);
+
+int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
+ const WebRtc_Word16* __restrict x,
+ WebRtc_Word16 N,
+ WebRtc_Word16 order,
+ WebRtc_Word16* __restrict scale);
+
+void WebRtcIsacfix_FilterMaLoopC(int16_t input0,
+ int16_t input1,
+ int32_t input2,
+ int32_t* ptr0,
+ int32_t* ptr1,
+ int32_t* ptr2);
+
+// Functions for ARM-Neon platforms, in place of the above two generic C ones.
+#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+int WebRtcIsacfix_AutocorrNeon(WebRtc_Word32* __restrict r,
+ const WebRtc_Word16* __restrict x,
+ WebRtc_Word16 N,
+ WebRtc_Word16 order,
+ WebRtc_Word16* __restrict scale);
+
+void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
+ int16_t input1,
+ int32_t input2,
+ int32_t* ptr0,
+ int32_t* ptr1,
+ int32_t* ptr2);
+#endif
+
+/**** Function pointers associated with
+ **** WebRtcIsacfix_AutocorrC() / WebRtcIsacfix_AutocorrNeon()
+ **** and WebRtcIsacfix_FilterMaLoopC() / WebRtcIsacfix_FilterMaLoopNeon().
+ ****/
+
+typedef int (*AutocorrFix)(WebRtc_Word32* __restrict r,
+ const WebRtc_Word16* __restrict x,
+ WebRtc_Word16 N,
+ WebRtc_Word16 order,
+ WebRtc_Word16* __restrict scale);
+extern AutocorrFix WebRtcIsacfix_AutocorrFix;
+
+typedef void (*FilterMaLoopFix)(int16_t input0,
+ int16_t input1,
+ int32_t input2,
+ int32_t* ptr0,
+ int32_t* ptr1,
+ int32_t* ptr2);
+extern FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c
new file mode 100644
index 0000000..2e15e7a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * decode.c
+ *
+ * This C file contains the internal decoding function.
+ *
+ */
+
+#include <string.h>
+
+#include "bandwidth_estimator.h"
+#include "codec.h"
+#include "entropy_coding.h"
+#include "pitch_estimator.h"
+#include "settings.h"
+#include "structs.h"
+
+
+
+
+WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16 *signal_out16,
+ ISACFIX_DecInst_t *ISACdec_obj,
+ WebRtc_Word16 *current_framesamples)
+{
+ int k;
+ int err;
+ WebRtc_Word16 BWno;
+ WebRtc_Word16 len = 0;
+
+ WebRtc_Word16 model;
+
+
+ WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES/2];
+ WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES/2];
+
+ WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES/2];
+ WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES/2];
+
+ WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
+ WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
+ WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+
+ WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
+ WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
+ WebRtc_Word16 AvgPitchGain_Q12;
+
+ WebRtc_Word16 tmp_1, tmp_2;
+ WebRtc_Word32 tmp32a, tmp32b;
+ WebRtc_Word16 gainQ13;
+
+
+ WebRtc_Word16 frame_nb; /* counter */
+ WebRtc_Word16 frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
+ WebRtc_Word16 processed_samples;
+
+ /* PLC */
+ WebRtc_Word16 overlapWin[ 240 ];
+
+ (ISACdec_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
+ (ISACdec_obj->bitstr_obj).streamval = 0;
+ (ISACdec_obj->bitstr_obj).stream_index = 0;
+ (ISACdec_obj->bitstr_obj).full = 1;
+
+
+ /* decode framelength and BW estimation - not used, only for stream pointer*/
+ err = WebRtcIsacfix_DecodeFrameLen(&ISACdec_obj->bitstr_obj, current_framesamples);
+ if (err<0) // error check
+ return err;
+
+ frame_mode = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
+ processed_samples = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
+
+ err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
+ if (err<0) // error check
+ return err;
+
+ /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames bundled together (60ms) */
+ for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
+
+ /* decode & dequantize pitch parameters */
+ err = WebRtcIsacfix_DecodePitchGain(&(ISACdec_obj->bitstr_obj), PitchGains_Q12);
+ if (err<0) // error check
+ return err;
+
+ err = WebRtcIsacfix_DecodePitchLag(&ISACdec_obj->bitstr_obj, PitchGains_Q12, PitchLags_Q7);
+ if (err<0) // error check
+ return err;
+
+ AvgPitchGain_Q12 = (WebRtc_Word16)(((WebRtc_Word32)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2);
+
+ /* decode & dequantize FiltCoef */
+ err = WebRtcIsacfix_DecodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
+ &ISACdec_obj->bitstr_obj, &model);
+
+ if (err<0) // error check
+ return err;
+
+ /* decode & dequantize spectrum */
+ len = WebRtcIsacfix_DecodeSpec(&ISACdec_obj->bitstr_obj, Vector_Word16_1, Vector_Word16_2, AvgPitchGain_Q12);
+ if (len < 0) // error check
+ return len;
+
+ // Why does this need Q16 in and out? /JS
+ WebRtcIsacfix_Spec2Time(Vector_Word16_1, Vector_Word16_2, Vector_Word32_1, Vector_Word32_2);
+
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ Vector_Word16_1[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(Vector_Word32_1[k]+64, 7); //Q16 -> Q9
+ }
+
+ /* ---- If this is recovery frame ---- */
+ if( (ISACdec_obj->plcstr_obj).used == PLC_WAS_USED )
+ {
+ (ISACdec_obj->plcstr_obj).used = PLC_NOT_USED;
+ if( (ISACdec_obj->plcstr_obj).B < 1000 )
+ {
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 4000;
+ }
+
+ ISACdec_obj->plcstr_obj.decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
+ ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
+ ISACdec_obj->plcstr_obj.pitchCycles = 0;
+
+ PitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
+
+ /* ---- Add-overlap ---- */
+ WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
+ for( k = 0; k < RECOVERY_OVERLAP; k++ )
+ Vector_Word16_1[k] = WEBRTC_SPL_ADD_SAT_W16(
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
+
+
+
+ }
+
+ /* --- Store side info --- */
+ if( frame_nb == frame_mode )
+ {
+ /* --- LPC info */
+ WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).lofilt_coefQ15, &lofilt_coefQ15[(SUBFRAMES-1)*ORDERLO], ORDERLO );
+ WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).hifilt_coefQ15, &hifilt_coefQ15[(SUBFRAMES-1)*ORDERHI], ORDERHI );
+ (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0] = gain_lo_hiQ17[(SUBFRAMES-1) * 2];
+ (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1] = gain_lo_hiQ17[(SUBFRAMES-1) * 2 + 1];
+
+ /* --- LTP info */
+ (ISACdec_obj->plcstr_obj).AvgPitchGain_Q12 = PitchGains_Q12[3];
+ (ISACdec_obj->plcstr_obj).lastPitchGain_Q12 = PitchGains_Q12[3];
+ (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = PitchLags_Q7[3];
+
+ if( PitchLags_Q7[3] < 3000 )
+ (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 += PitchLags_Q7[3];
+
+ WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvIn, Vector_Word16_1, FRAMESAMPLES/2 );
+
+ }
+ /* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */
+
+ /* inverse pitch filter */
+ WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2, &ISACdec_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 4);
+
+ if( frame_nb == frame_mode )
+ {
+ WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvOut, &(Vector_Word16_2[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10)]), PITCH_MAX_LAG );
+ }
+
+
+ /* reduce gain to compensate for pitch enhancer */
+ /* gain = 1.0f - 0.45f * AvgPitchGain; */
+ tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
+ tmp32b = 262144 - tmp32a; // Q18
+ gainQ13 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q13
+
+ for (k = 0; k < FRAMESAMPLES/2; k++)
+ {
+ Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(Vector_Word16_2[k], gainQ13), 3); // Q25
+ }
+
+
+ /* perceptual post-filtering (using normalized lattice filter) */
+ WebRtcIsacfix_NormLatticeFilterAr(ORDERLO, (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0,
+ Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1);
+
+ /* --- Store Highpass Residual --- */
+ for (k = 0; k < FRAMESAMPLES/2; k++)
+ Vector_Word32_1[k] = WEBRTC_SPL_LSHIFT_W32(Vector_Word32_2[k], 9); // Q16 -> Q25
+
+ for( k = 0; k < PITCH_MAX_LAG + 10; k++ )
+ (ISACdec_obj->plcstr_obj).prevHP[k] = Vector_Word32_1[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10) + k];
+
+
+ WebRtcIsacfix_NormLatticeFilterAr(ORDERHI, (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0,
+ Vector_Word32_1, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2);
+
+ /* recombine the 2 bands */
+
+ /* Form the polyphase signals, and compensate for DC offset */
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1)); /* Construct a new upper channel signal*/
+ tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k])); /* Construct a new lower channel signal*/
+ Vector_Word16_1[k] = tmp_1;
+ Vector_Word16_2[k] = tmp_2;
+ }
+
+ WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1, Vector_Word16_2, signal_out16 + frame_nb * processed_samples, &ISACdec_obj->postfiltbankstr_obj);
+
+ }
+ return len;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c
new file mode 100644
index 0000000..68c6003
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * decode_bwe.c
+ *
+ * This C file contains the internal decode bandwidth estimate function.
+ *
+ */
+
+
+#include "bandwidth_estimator.h"
+#include "codec.h"
+#include "entropy_coding.h"
+#include "structs.h"
+
+
+
+
+int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
+ Bitstr_dec *streamdata,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts)
+{
+ WebRtc_Word16 index;
+ WebRtc_Word16 frame_samples;
+ int err;
+
+ /* decode framelength */
+ err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples);
+ /* error check */
+ if (err<0) {
+ return err;
+ }
+
+ /* decode BW estimation */
+ err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index);
+ /* error check */
+ if (err<0) {
+ return err;
+ }
+
+ /* Update BWE with received data */
+ err = WebRtcIsacfix_UpdateUplinkBwImpl(
+ bwest_str,
+ rtp_seq_number,
+ (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
+ send_ts,
+ arr_ts,
+ (WebRtc_Word16) packet_size, /* in bytes */
+ index);
+
+ /* error check */
+ if (err<0) {
+ return err;
+ }
+
+ /* Succesful */
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c
new file mode 100644
index 0000000..de51658
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c
@@ -0,0 +1,830 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * decode_plc.c
+ *
+ * Packet Loss Concealment.
+ *
+ */
+
+#include <string.h>
+
+#include "settings.h"
+#include "entropy_coding.h"
+#include "pitch_estimator.h"
+#include "bandwidth_estimator.h"
+#include "structs.h"
+#include "codec.h"
+
+
+#define NO_OF_PRIMES 8
+#define NOISE_FILTER_LEN 30
+
+/*
+ * function to decode the bitstream
+ * returns the total number of bytes in the stream
+ */
+
+static WebRtc_Word16 plc_filterma_Fast(
+ WebRtc_Word16 *In, /* (i) Vector to be filtered. InOut[-orderCoef+1]
+ to InOut[-1] contains state */
+ WebRtc_Word16 *Out, /* (o) Filtered vector */
+ WebRtc_Word16 *B, /* (i) The filter coefficients (in Q0) */
+ WebRtc_Word16 Blen, /* (i) Number of B coefficients */
+ WebRtc_Word16 len, /* (i) Number of samples to be filtered */
+ WebRtc_Word16 reduceDecay,
+ WebRtc_Word16 decay,
+ WebRtc_Word16 rshift )
+{
+ int i, j;
+ WebRtc_Word32 o;
+ WebRtc_Word32 lim;
+
+ lim = WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 15 + rshift )-1;
+
+ for (i = 0; i < len; i++)
+ {
+ G_CONST WebRtc_Word16 *b_ptr = &B[0];
+ G_CONST WebRtc_Word16 *x_ptr = &In[i];
+
+ o = (WebRtc_Word32)0;
+
+ for (j = 0;j < Blen; j++)
+ {
+ o = WEBRTC_SPL_ADD_SAT_W32( o, WEBRTC_SPL_MUL_16_16( *b_ptr, *x_ptr) );
+ b_ptr++;
+ x_ptr--;
+ }
+
+ /* to round off correctly */
+ o = WEBRTC_SPL_ADD_SAT_W32( o, WEBRTC_SPL_LSHIFT_W32( 1, (rshift-1) ) );
+
+ /* saturate according to the domain of the filter coefficients */
+ o = WEBRTC_SPL_SAT((WebRtc_Word32)lim, o, (WebRtc_Word32)-lim);
+
+ /* o should be in the range of WebRtc_Word16 */
+ o = WEBRTC_SPL_RSHIFT_W32( o, rshift );
+
+ /* decay the output signal; this is specific to plc */
+ *Out++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16)o, decay, 15); // ((o + (WebRtc_Word32)2048) >> 12);
+
+ /* change the decay */
+ decay -= reduceDecay;
+ if( decay < 0 )
+ decay = 0;
+ }
+ return( decay );
+}
+
+
+
+
+
+
+
+
+static __inline WebRtc_Word32 log2_Q8_T( WebRtc_UWord32 x ) {
+
+ WebRtc_Word32 zeros, lg2;
+ WebRtc_Word16 frac;
+
+ zeros=WebRtcSpl_NormU32(x);
+ frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
+ /* log2(magn(i)) */
+
+ lg2= (WEBRTC_SPL_LSHIFT_W16((31-zeros), 8)+frac);
+ return lg2;
+
+}
+
+static __inline WebRtc_Word16 exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
+
+ WebRtc_Word16 tmp16_1, tmp16_2;
+
+ tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
+ tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+ if(tmp16_1>0)
+ return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+ else
+ return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+
+}
+
+
+/*
+ This is a fixed-point version of the above code with limLow = 700 and limHigh = 5000,
+ hard-coded. The values 700 and 5000 were experimentally obtained.
+
+ The function implements membership values for two sets. The mebership functions are
+ of second orders corresponding to half-bell-shapped pulses.
+*/
+static void MemshipValQ15( WebRtc_Word16 in, WebRtc_Word16 *A, WebRtc_Word16 *B )
+{
+ WebRtc_Word16 x;
+
+ in -= 700; /* translate the lowLim to 0, limHigh = 5000 - 700, M = 2150 */
+
+ if( in <= 2150 )
+ {
+ if( in > 0 )
+ {
+ /* b = in^2 / (2 * M^2), a = 1 - b in Q0.
+ We have to compute in Q15 */
+
+ /* x = in / 2150 {in Q15} = x * 15.2409 {in Q15} =
+ x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999 */
+
+ /* we are sure that x is in the range of WebRtc_Word16 */
+ x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
+ WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
+ /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
+ be in correct domain and one more for the division by 2 */
+ *B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
+ *A = WEBRTC_SPL_WORD16_MAX - *B;
+ }
+ else
+ {
+ *B = 0;
+ *A = WEBRTC_SPL_WORD16_MAX;
+ }
+ }
+ else
+ {
+ if( in < 4300 )
+ {
+ /* This is a mirror case of the above */
+ in = 4300 - in;
+ x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
+ WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
+ /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
+ be in correct domain and one more for the division by 2 */
+ *A = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
+ *B = WEBRTC_SPL_WORD16_MAX - *A;
+
+ }
+ else
+ {
+ *A = 0;
+ *B = WEBRTC_SPL_WORD16_MAX;
+ }
+ }
+}
+
+
+
+
+static void LinearResampler( WebRtc_Word16 *in, WebRtc_Word16 *out, WebRtc_Word16 lenIn, WebRtc_Word16 lenOut )
+{
+ WebRtc_Word32 n;
+ WebRtc_Word16 resOut, i, j, relativePos, diff; /* */
+ WebRtc_UWord16 udiff;
+
+ if( lenIn == lenOut )
+ {
+ WEBRTC_SPL_MEMCPY_W16( out, in, lenIn );
+ return;
+ }
+
+ n = WEBRTC_SPL_MUL_16_16( (WebRtc_Word16)(lenIn-1), RESAMP_RES );
+ resOut = WebRtcSpl_DivW32W16ResW16( n, (WebRtc_Word16)(lenOut-1) );
+
+ out[0] = in[0];
+ for( i = 1, j = 0, relativePos = 0; i < lenOut; i++ )
+ {
+
+ relativePos += resOut;
+ while( relativePos > RESAMP_RES )
+ {
+ j++;
+ relativePos -= RESAMP_RES;
+ }
+
+
+ /* an overflow may happen and the differce in sample values may
+ * require more than 16 bits. We like to avoid 32 bit arithmatic
+ * as much as possible */
+
+ if( (in[ j ] > 0) && (in[j + 1] < 0) )
+ {
+ udiff = (WebRtc_UWord16)(in[ j ] - in[j + 1]);
+ out[ i ] = in[ j ] - (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
+ }
+ else
+ {
+ if( (in[j] < 0) && (in[j+1] > 0) )
+ {
+ udiff = (WebRtc_UWord16)( in[j + 1] - in[ j ] );
+ out[ i ] = in[ j ] + (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
+ }
+ else
+ {
+ diff = in[ j + 1 ] - in[ j ];
+ out[ i ] = in[ j ] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
+ }
+ }
+ }
+}
+
+
+
+
+
+WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16 *signal_out16,
+ ISACFIX_DecInst_t *ISACdec_obj,
+ WebRtc_Word16 *current_framesamples )
+{
+ int subframecnt;
+ WebRtc_Word16 len = 0;
+
+ WebRtc_Word16* Vector_Word16_1;
+ WebRtc_Word16 Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
+ WebRtc_Word16* Vector_Word16_2;
+ WebRtc_Word16 Vector_Word16_Extended_2[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
+
+ WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES_HALF];
+ WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES_HALF];
+
+ WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
+ WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
+
+ WebRtc_Word16 pitchLags_Q7[PITCH_SUBFRAMES];
+ WebRtc_Word16 pitchGains_Q12[PITCH_SUBFRAMES];
+
+ WebRtc_Word16 tmp_1, tmp_2;
+ WebRtc_Word32 tmp32a, tmp32b;
+ WebRtc_Word16 gainQ13;
+
+ WebRtc_Word16 myDecayRate;
+
+ /* ---------- PLC variables ------------ */
+ WebRtc_Word16 lag0, i, k, noiseIndex;
+ WebRtc_Word16 stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10];
+
+ WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+
+ WebRtc_Word16 nLP, pLP, wNoisyLP, wPriodicLP, tmp16, minIdx;
+ WebRtc_Word32 nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff;
+ WebRtc_Word16 noise1, rshift;
+
+
+ WebRtc_Word16 ltpGain, pitchGain, myVoiceIndicator, myAbs, maxAbs;
+ WebRtc_Word32 varIn, varOut, logVarIn, logVarOut, Q, logMaxAbs;
+ int rightShiftIn, rightShiftOut;
+
+
+ /* ------------------------------------- */
+
+
+ myDecayRate = (DECAY_RATE);
+ Vector_Word16_1 = &Vector_Word16_Extended_1[NOISE_FILTER_LEN];
+ Vector_Word16_2 = &Vector_Word16_Extended_2[NOISE_FILTER_LEN];
+
+
+ /* ----- Simply Copy Previous LPC parameters ------ */
+ for( subframecnt = 0; subframecnt < SUBFRAMES; subframecnt++ )
+ {
+ /* lower Band */
+ WEBRTC_SPL_MEMCPY_W16(&lofilt_coefQ15[ subframecnt * ORDERLO ],
+ (ISACdec_obj->plcstr_obj).lofilt_coefQ15, ORDERLO);
+ gain_lo_hiQ17[2*subframecnt] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0];
+
+ /* Upper Band */
+ WEBRTC_SPL_MEMCPY_W16(&hifilt_coefQ15[ subframecnt * ORDERHI ],
+ (ISACdec_obj->plcstr_obj).hifilt_coefQ15, ORDERHI);
+ gain_lo_hiQ17[2*subframecnt + 1] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1];
+ }
+
+
+
+
+ lag0 = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 + 64, 7 ) + 1;
+
+
+ if( (ISACdec_obj->plcstr_obj).used != PLC_WAS_USED )
+ {
+ (ISACdec_obj->plcstr_obj).pitchCycles = 0;
+
+ (ISACdec_obj->plcstr_obj).lastPitchLP =
+ &((ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0]);
+ minCorr = WEBRTC_SPL_WORD32_MAX;
+
+ if ( (FRAMESAMPLES_HALF - 2*lag0 - 10) > 0 )
+ {
+ minIdx = 11;
+ for( i = 0; i < 21; i++ )
+ {
+ corr = 0;
+ for( k = 0; k < lag0; k++ )
+ {
+ corr = WEBRTC_SPL_ADD_SAT_W32( corr, WEBRTC_SPL_ABS_W32(
+ WEBRTC_SPL_SUB_SAT_W16(
+ (ISACdec_obj->plcstr_obj).lastPitchLP[k],
+ (ISACdec_obj->plcstr_obj).prevPitchInvIn[
+ FRAMESAMPLES_HALF - 2*lag0 - 10 + i + k ] ) ) );
+ }
+ if( corr < minCorr )
+ {
+ minCorr = corr;
+ minIdx = i;
+ }
+ }
+ (ISACdec_obj->plcstr_obj).prevPitchLP =
+ &( (ISACdec_obj->plcstr_obj).prevPitchInvIn[
+ FRAMESAMPLES_HALF - lag0*2 - 10 + minIdx] );
+ }
+ else
+ {
+ (ISACdec_obj->plcstr_obj).prevPitchLP =
+ (ISACdec_obj->plcstr_obj).lastPitchLP;
+ }
+ pitchGain = (ISACdec_obj->plcstr_obj).lastPitchGain_Q12;
+
+ WebRtcSpl_AutoCorrelation(
+ &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0],
+ lag0, 0, &varIn, &rightShiftIn);
+ WebRtcSpl_AutoCorrelation(
+ &(ISACdec_obj->plcstr_obj).prevPitchInvOut[PITCH_MAX_LAG + 10 - lag0],
+ lag0, 0, &varOut, &rightShiftOut);
+
+ maxAbs = 0;
+ for( i = 0; i< lag0; i++)
+ {
+ myAbs = WEBRTC_SPL_ABS_W16(
+ (ISACdec_obj->plcstr_obj).prevPitchInvOut[
+ PITCH_MAX_LAG + 10 - lag0 + i] );
+ maxAbs = (myAbs > maxAbs)? myAbs:maxAbs;
+ }
+ logVarIn = log2_Q8_T( (WebRtc_UWord32)( varIn ) ) +
+ (WebRtc_Word32)(rightShiftIn << 8);
+ logVarOut = log2_Q8_T( (WebRtc_UWord32)( varOut ) ) +
+ (WebRtc_Word32)(rightShiftOut << 8);
+ logMaxAbs = log2_Q8_T( (WebRtc_UWord32)( maxAbs ) );
+
+ ltpGain = (WebRtc_Word16)(logVarOut - logVarIn);
+ Q = 2 * logMaxAbs - ( logVarOut - 1512 );
+
+ /*
+ * ---
+ * We are computing sqrt( (VarIn/lag0) / var( noise ) )
+ * var( noise ) is almost 256. we have already computed log2( VarIn ) in Q8
+ * so we actually compute 2^( 0.5*(log2( VarIn ) - log2( lag0 ) - log2( var(noise ) ) ).
+ * Note that put log function is in Q8 but the exponential function is in Q10.
+ * --
+ */
+
+ logVarIn -= log2_Q8_T( (WebRtc_UWord32)( lag0 ) );
+ tmp16 = (WebRtc_Word16)((logVarIn<<1) - (4<<10) );
+ rightShiftIn = 0;
+ if( tmp16 > 4096 )
+ {
+ tmp16 -= 4096;
+ tmp16 = exp2_Q10_T( tmp16 );
+ tmp16 >>= 6;
+ }
+ else
+ tmp16 = exp2_Q10_T( tmp16 )>>10;
+
+ (ISACdec_obj->plcstr_obj).std = tmp16 - 4;
+
+ if( (ltpGain < 110) || (ltpGain > 230) )
+ {
+ if( ltpGain < 100 && (pitchGain < 1800) )
+ {
+ (ISACdec_obj->plcstr_obj).A = WEBRTC_SPL_WORD16_MAX;
+ }
+ else
+ {
+ (ISACdec_obj->plcstr_obj).A = ((ltpGain < 110) && (Q < 800)
+ )? WEBRTC_SPL_WORD16_MAX:0;
+ }
+ (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX -
+ (ISACdec_obj->plcstr_obj).A;
+ }
+ else
+ {
+ if( (pitchGain < 450) || (pitchGain > 1600) )
+ {
+ (ISACdec_obj->plcstr_obj).A = ((pitchGain < 450)
+ )? WEBRTC_SPL_WORD16_MAX:0;
+ (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX -
+ (ISACdec_obj->plcstr_obj).A;
+ }
+ else
+ {
+ myVoiceIndicator = ltpGain * 2 + pitchGain;
+ MemshipValQ15( myVoiceIndicator,
+ &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B );
+ }
+ }
+
+
+
+ myVoiceIndicator = ltpGain * 16 + pitchGain * 2 + (pitchGain >> 8);
+ MemshipValQ15( myVoiceIndicator,
+ &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B );
+
+
+
+ (ISACdec_obj->plcstr_obj).stretchLag = lag0;
+ (ISACdec_obj->plcstr_obj).pitchIndex = 0;
+
+ }
+ else
+ {
+ myDecayRate = (DECAY_RATE<<2);
+ }
+
+ if( (ISACdec_obj->plcstr_obj).B < 1000 )
+ {
+ myDecayRate += (DECAY_RATE<<3);
+ }
+
+ /* ------------ reconstructing the residual signal ------------------ */
+
+ LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP,
+ stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
+ /* inverse pitch filter */
+
+ pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
+ ((ISACdec_obj->plcstr_obj).stretchLag<<7);
+ pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
+ pitchGains_Q12[2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ pitchGains_Q12[3], 1010, 10 );
+ pitchGains_Q12[1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ pitchGains_Q12[2], 1010, 10 );
+ pitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ pitchGains_Q12[1], 1010, 10 );
+
+
+ /* most of the time either B or A are zero so seperating */
+ if( (ISACdec_obj->plcstr_obj).B == 0 )
+ {
+ for( i = 0; i < FRAMESAMPLES_HALF; i++ )
+ {
+ /* --- Low Pass */
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+ Vector_Word16_1[i] = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
+
+ /* --- Highpass */
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+ Vector_Word16_2[i] = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
+
+ }
+ for( i = 1; i < NOISE_FILTER_LEN; i++ )
+ {
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+ Vector_Word16_Extended_1[ i ] = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
+
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+ Vector_Word16_Extended_2[ i ] = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
+ }
+ plc_filterma_Fast(Vector_Word16_1, Vector_Word16_Extended_1,
+ &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF -
+ NOISE_FILTER_LEN], (WebRtc_Word16) NOISE_FILTER_LEN,
+ (WebRtc_Word16) FRAMESAMPLES_HALF, (WebRtc_Word16)(5),
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise, (WebRtc_Word16)(6));
+
+ maxCoeff = WebRtcSpl_MaxAbsValueW32(
+ &(ISACdec_obj->plcstr_obj).prevHP[
+ PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN], NOISE_FILTER_LEN );
+
+ rshift = 0;
+ while( maxCoeff > WEBRTC_SPL_WORD16_MAX )
+ {
+ maxCoeff = WEBRTC_SPL_RSHIFT_W32(maxCoeff, 1);
+ rshift++;
+ }
+ for( i = 0; i < NOISE_FILTER_LEN; i++ ) {
+ Vector_Word16_1[ FRAMESAMPLES_HALF - NOISE_FILTER_LEN + i] =
+ (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (ISACdec_obj->plcstr_obj).prevHP[
+ PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN + i], rshift);
+ }
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise = plc_filterma_Fast(
+ Vector_Word16_2,
+ Vector_Word16_Extended_2,
+ &Vector_Word16_1[FRAMESAMPLES_HALF - NOISE_FILTER_LEN],
+ (WebRtc_Word16) NOISE_FILTER_LEN,
+ (WebRtc_Word16) FRAMESAMPLES_HALF,
+ (WebRtc_Word16) (5),
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise,
+ (WebRtc_Word16) (7) );
+
+ for( i = 0; i < FRAMESAMPLES_HALF; i++ )
+ Vector_Word32_2[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)Vector_Word16_Extended_2[i], rshift );
+
+ Vector_Word16_1 = Vector_Word16_Extended_1;
+ }
+ else
+ {
+ if( (ISACdec_obj->plcstr_obj).A == 0 )
+ {
+ /* ------ Periodic Vector --- */
+ for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
+ {
+ /* --- Lowpass */
+ pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+
+ /* --- Highpass */
+ pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
+ (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
+ (ISACdec_obj->plcstr_obj).stretchLag +
+ (ISACdec_obj->plcstr_obj).pitchIndex] );
+
+ /* --- lower the muliplier (more decay at next sample) --- */
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate);
+ if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 )
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0;
+
+ (ISACdec_obj->plcstr_obj).pitchIndex++;
+
+ if( (ISACdec_obj->plcstr_obj).pitchIndex ==
+ (ISACdec_obj->plcstr_obj).stretchLag )
+ {
+ (ISACdec_obj->plcstr_obj).pitchIndex = 0;
+ (ISACdec_obj->plcstr_obj).pitchCycles++;
+
+ if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) )
+ {
+ (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1;
+ }
+ else
+ {
+ (ISACdec_obj->plcstr_obj).stretchLag = lag0;
+ }
+
+ (ISACdec_obj->plcstr_obj).stretchLag = (
+ (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG
+ )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag;
+
+ LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP,
+ stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
+
+ LinearResampler( (ISACdec_obj->plcstr_obj).prevPitchLP,
+ stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
+
+ switch( (ISACdec_obj->plcstr_obj).pitchCycles )
+ {
+ case 1:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)((
+ (WebRtc_Word32)stretchPitchLP[k]* 3 +
+ (WebRtc_Word32)stretchPitchLP1[k])>>2);
+ }
+ break;
+ }
+ case 2:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)((
+ (WebRtc_Word32)stretchPitchLP[k] +
+ (WebRtc_Word32)stretchPitchLP1[k] )>>1);
+ }
+ break;
+ }
+ case 3:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)((stretchPitchLP[k] +
+ (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
+ }
+ break;
+ }
+ }
+
+ if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 )
+ {
+ myDecayRate += 35; //(myDecayRate>>1);
+ (ISACdec_obj->plcstr_obj).pitchCycles = 0;
+ }
+
+ }
+
+ /* ------ Sum the noisy and periodic signals ------ */
+ Vector_Word16_1[i] = pLP;
+ Vector_Word32_2[i] = pHP;
+ }
+ }
+ else
+ {
+ for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
+ {
+
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+
+ noise1 = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
+
+ nLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (WebRtc_Word16)((noise1)*(ISACdec_obj->plcstr_obj).std),
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise, 15 );
+
+ /* --- Highpass */
+ (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
+ (ISACdec_obj->plcstr_obj).seed );
+ noise1 = WEBRTC_SPL_RSHIFT_W16(
+ (ISACdec_obj->plcstr_obj).seed, 11 ) - 8;
+
+ nHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise,
+ (WebRtc_Word32)(noise1*(ISACdec_obj->plcstr_obj).std) );
+
+ /* --- lower the muliplier (more decay at next sample) --- */
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise -= (myDecayRate);
+ if( (ISACdec_obj->plcstr_obj).decayCoeffNoise < 0 )
+ (ISACdec_obj->plcstr_obj).decayCoeffNoise = 0;
+
+ /* ------ Periodic Vector --- */
+ /* --- Lowpass */
+ pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+
+ /* --- Highpass */
+ pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
+ (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
+ (ISACdec_obj->plcstr_obj).stretchLag +
+ (ISACdec_obj->plcstr_obj).pitchIndex] );
+
+ /* --- lower the muliplier (more decay at next sample) --- */
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate);
+ if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 )
+ {
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0;
+ }
+
+ /* ------ Weighting the noisy and periodic vectors ------- */
+ wNoisyLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
+ (ISACdec_obj->plcstr_obj).A, nLP, 15 ) );
+ wNoisyHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
+ (ISACdec_obj->plcstr_obj).A, (nHP) ) );
+
+ wPriodicLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
+ (ISACdec_obj->plcstr_obj).B, pLP, 15));
+ wPriodicHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
+ (ISACdec_obj->plcstr_obj).B, pHP));
+
+ (ISACdec_obj->plcstr_obj).pitchIndex++;
+
+ if((ISACdec_obj->plcstr_obj).pitchIndex ==
+ (ISACdec_obj->plcstr_obj).stretchLag)
+ {
+ (ISACdec_obj->plcstr_obj).pitchIndex = 0;
+ (ISACdec_obj->plcstr_obj).pitchCycles++;
+
+ if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) )
+ (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1;
+ else
+ (ISACdec_obj->plcstr_obj).stretchLag = lag0;
+
+ (ISACdec_obj->plcstr_obj).stretchLag = (
+ (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG
+ )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag;
+ LinearResampler(
+ (ISACdec_obj->plcstr_obj).lastPitchLP,
+ stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
+
+ LinearResampler((ISACdec_obj->plcstr_obj).prevPitchLP,
+ stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
+
+ switch((ISACdec_obj->plcstr_obj).pitchCycles)
+ {
+ case 1:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)((
+ (WebRtc_Word32)stretchPitchLP[k]* 3 +
+ (WebRtc_Word32)stretchPitchLP1[k] )>>2);
+ }
+ break;
+ }
+ case 2:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)((
+ (WebRtc_Word32)stretchPitchLP[k] +
+ (WebRtc_Word32)stretchPitchLP1[k])>>1);
+ }
+ break;
+ }
+ case 3:
+ {
+ for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
+ {
+ stretchPitchLP[k] = (WebRtc_Word16)(
+ (stretchPitchLP[k] +
+ (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
+ }
+ break;
+ }
+ }
+
+ if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 )
+ {
+ myDecayRate += 55; //(myDecayRate>>1);
+ (ISACdec_obj->plcstr_obj).pitchCycles = 0;
+ }
+ }
+
+ /* ------ Sum the noisy and periodic signals ------ */
+ Vector_Word16_1[i] = (WebRtc_Word16)WEBRTC_SPL_ADD_SAT_W16(
+ wNoisyLP, wPriodicLP );
+ Vector_Word32_2[i] = (WebRtc_Word32)WEBRTC_SPL_ADD_SAT_W32(
+ wNoisyHP, wPriodicHP );
+ }
+ }
+ }
+ /* ----------------- residual signal is reconstructed ------------------ */
+
+ k = (ISACdec_obj->plcstr_obj).pitchIndex;
+ /* --- Write one pitch cycle for recovery block --- */
+
+ for( i = 0; i < RECOVERY_OVERLAP; i++ )
+ {
+ (ISACdec_obj->plcstr_obj).overlapLP[i] = (WebRtc_Word16)(
+ WEBRTC_SPL_MUL_16_16_RSFT(stretchPitchLP[k],
+ (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15) );
+ k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
+ }
+
+ (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = (ISACdec_obj->plcstr_obj).stretchLag << 7;
+
+
+ /* --- Inverse Pitch Filter --- */
+ WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2,
+ &ISACdec_obj->pitchfiltstr_obj, pitchLags_Q7, pitchGains_Q12, 4);
+
+ /* reduce gain to compensate for pitch enhancer */
+ /* gain = 1.0f - 0.45f * AvgPitchGain; */
+ tmp32a = WEBRTC_SPL_MUL_16_16_RSFT((ISACdec_obj->plcstr_obj).AvgPitchGain_Q12,
+ 29, 0); // Q18
+ tmp32b = 262144 - tmp32a; // Q18
+ gainQ13 = (WebRtc_Word16) (tmp32b >> 5); // Q13
+
+ /* perceptual post-filtering (using normalized lattice filter) */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++)
+ Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(
+ Vector_Word16_2[k], gainQ13) << 3; // Q25
+
+
+ WebRtcIsacfix_NormLatticeFilterAr(ORDERLO,
+ (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0,
+ Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1);
+
+ WebRtcIsacfix_NormLatticeFilterAr(ORDERHI,
+ (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0,
+ Vector_Word32_2, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2);
+
+ /* recombine the 2 bands */
+
+ /* Form the polyphase signals, and compensate for DC offset */
+ for (k=0;k<FRAMESAMPLES_HALF;k++)
+ {
+ /* Construct a new upper channel signal*/
+ tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
+ ((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1));
+ /* Construct a new lower channel signal*/
+ tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
+ ((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k]));
+ Vector_Word16_1[k] = tmp_1;
+ Vector_Word16_2[k] = tmp_2;
+ }
+
+
+ WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1,
+ Vector_Word16_2, signal_out16, &ISACdec_obj->postfiltbankstr_obj);
+
+ (ISACdec_obj->plcstr_obj).used = PLC_WAS_USED;
+ *current_framesamples = 480;
+
+ return len;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
new file mode 100644
index 0000000..cb531e5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
@@ -0,0 +1,626 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * encode.c
+ *
+ * Encoding function for the iSAC coder.
+ *
+ */
+
+#include "arith_routins.h"
+#include "bandwidth_estimator.h"
+#include "codec.h"
+#include "pitch_gain_tables.h"
+#include "pitch_lag_tables.h"
+#include "entropy_coding.h"
+#include "lpc_tables.h"
+#include "lpc_masking_model.h"
+#include "pitch_estimator.h"
+#include "structs.h"
+#include <stdio.h>
+
+
+int WebRtcIsacfix_EncodeImpl(WebRtc_Word16 *in,
+ ISACFIX_EncInst_t *ISACenc_obj,
+ BwEstimatorstr *bw_estimatordata,
+ WebRtc_Word16 CodingMode)
+{
+ WebRtc_Word16 stream_length = 0;
+ WebRtc_Word16 usefulstr_len = 0;
+ int k;
+ WebRtc_Word16 BWno;
+
+ WebRtc_Word16 lofilt_coefQ15[(ORDERLO)*SUBFRAMES];
+ WebRtc_Word16 hifilt_coefQ15[(ORDERHI)*SUBFRAMES];
+ WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
+
+ WebRtc_Word16 LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD];
+ WebRtc_Word16 LP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
+ WebRtc_Word16 HP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
+
+ WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
+ WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
+ WebRtc_Word16 AvgPitchGain_Q12;
+
+ WebRtc_Word16 frame_mode; /* 0 for 30ms, 1 for 60ms */
+ WebRtc_Word16 processed_samples;
+ int status;
+
+ WebRtc_Word32 bits_gainsQ11;
+ WebRtc_Word16 MinBytes;
+ WebRtc_Word16 bmodel;
+
+ transcode_obj transcodingParam;
+ WebRtc_Word16 payloadLimitBytes;
+ WebRtc_Word16 arithLenBeforeEncodingDFT;
+ WebRtc_Word16 iterCntr;
+
+ /* copy new frame length and bottle neck rate only for the first 10 ms data */
+ if (ISACenc_obj->buffer_index == 0) {
+ /* set the framelength for the next packet */
+ ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength;
+ }
+
+ frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */
+ processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */
+
+ /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */
+ /**************************************************************************************/
+ /* fill the buffer with 10ms input data */
+ for(k=0; k<FRAMESAMPLES_10ms; k++) {
+ ISACenc_obj->data_buffer_fix[k + ISACenc_obj->buffer_index] = in[k];
+ }
+ /* if buffersize is not equal to current framesize, and end of file is not reached yet, */
+ /* increase index and go back to main to get more speech samples */
+ if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) {
+ ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms;
+ return 0;
+ }
+ /* if buffer reached the right size, reset index and continue with encoding the frame */
+ ISACenc_obj->buffer_index = 0;
+
+ /* end of buffer function */
+ /**************************/
+
+ /* encoding */
+ /************/
+
+ if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 )
+ {
+ /* reset bitstream */
+ ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
+ ISACenc_obj->bitstr_obj.streamval = 0;
+ ISACenc_obj->bitstr_obj.stream_index = 0;
+ ISACenc_obj->bitstr_obj.full = 1;
+
+ if (CodingMode == 0) {
+ ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata);
+ ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata);
+ }
+ if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
+ ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
+ ISACenc_obj->current_framesamples);
+ }
+
+ // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
+ // 901/1024 is 0.87988281250000
+ ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
+ ISACenc_obj->current_framesamples);
+
+ /* encode frame length */
+ status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj);
+ if (status < 0)
+ {
+ /* Wrong frame size */
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+
+ /* Save framelength for multiple packets memory */
+ if (ISACenc_obj->SaveEnc_ptr != NULL) {
+ (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples;
+ }
+
+ /* bandwidth estimation and coding */
+ BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata);
+ status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
+ if (status < 0)
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+ }
+
+ /* split signal in two bands */
+ WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj );
+
+ /* estimate pitch parameters and pitch-filter lookahead signal */
+ WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP,
+ &ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */
+
+ /* Set where to store data in multiple packets memory */
+ if (ISACenc_obj->SaveEnc_ptr != NULL) {
+ if (frame_mode == 0 || ISACenc_obj->frame_nb == 0)
+ {
+ (ISACenc_obj->SaveEnc_ptr)->startIdx = 0;
+ }
+ else
+ {
+ (ISACenc_obj->SaveEnc_ptr)->startIdx = 1;
+ }
+ }
+
+ /* quantize & encode pitch parameters */
+ status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
+ if (status < 0)
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+ status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
+ if (status < 0)
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+ AvgPitchGain_Q12 = WEBRTC_SPL_RSHIFT_W32(PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3], 2);
+
+ /* find coefficients for perceptual pre-filters */
+ WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj,
+ ISACenc_obj->s2nr, PitchGains_Q12,
+ gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/
+
+ // record LPC Gains for possible bit-rate reduction
+ for(k = 0; k < KLT_ORDER_GAIN; k++)
+ {
+ transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
+ }
+
+ /* code LPC model and shape - gains not quantized yet */
+ status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
+ &bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam);
+ if (status < 0)
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+ arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
+
+ /* low-band filtering */
+ WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15,
+ LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */
+
+ /* pitch filter */
+ WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */
+
+ /* high-band filtering */
+ WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15,
+ HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/
+
+ /* transform */
+ WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/
+
+ /* Save data for multiple packets memory */
+ if (ISACenc_obj->SaveEnc_ptr != NULL) {
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
+ (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
+ }
+ (ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12;
+ }
+
+ /* quantization and lossless coding */
+ status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
+ if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+
+ if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
+ {
+ // it is a 60ms and we are in the first 30ms
+ // then the limit at this point should be half of the assigned value
+ payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1;
+ }
+ else if (frame_mode == 0)
+ {
+ // it is a 30ms frame
+ payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3;
+ }
+ else
+ {
+ // this is the second half of a 60ms frame.
+ payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes
+ }
+
+ iterCntr = 0;
+ while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) ||
+ (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH))
+ {
+ WebRtc_Word16 arithLenDFTByte;
+ WebRtc_Word16 bytesLeftQ5;
+ WebRtc_Word16 ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25};
+
+ // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs
+ // such as DTMF, sweep-sine, ...
+ //
+ // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14
+ // WebRtc_Word16 scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755};
+
+
+ // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit
+ // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403
+ //
+ WebRtc_Word16 scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500};
+ WebRtc_Word16 idx;
+
+ if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION)
+ {
+ // We were not able to limit the payload size
+
+ if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
+ {
+ // This was the first 30ms of a 60ms frame. Although the payload is larger than it
+ // should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed
+ // the limit.
+ ISACenc_obj->frame_nb = 1;
+ return 0;
+ }
+ else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1))
+ {
+ ISACenc_obj->frame_nb = 0;
+ }
+
+ if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
+ {
+ return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
+ }
+ else
+ {
+ return status;
+ }
+ }
+ if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
+ {
+ arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT;
+ bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5;
+
+ // bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive)
+ // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6
+ // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8
+ // to avoid division we do more simplification.
+ //
+ // values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i]
+ // and the corresponding scale is chosen
+
+ // we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte;
+ idx = 4;
+ idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 2:-2;
+ idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 1:-1;
+ idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 0:-1;
+ }
+ else
+ {
+ // we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not
+ // trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive
+ // case.
+ idx = 0;
+ }
+ // scale FFT coefficients to reduce the bit-rate
+ for(k = 0; k < FRAMESAMPLES_HALF; k++)
+ {
+ LP16a[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
+ LPandHP[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
+ }
+
+ // Save data for multiple packets memory
+ if (ISACenc_obj->SaveEnc_ptr != NULL)
+ {
+ for(k = 0; k < FRAMESAMPLES_HALF; k++)
+ {
+ (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
+ (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
+ }
+ }
+
+ // scale the unquantized LPC gains and save the scaled version for the future use
+ for(k = 0; k < KLT_ORDER_GAIN; k++)
+ {
+ gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; //
+ transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
+ }
+
+ // reset the bit-stream object to the state which it had before encoding LPC Gains
+ ISACenc_obj->bitstr_obj.full = transcodingParam.full;
+ ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
+ ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval;
+ ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
+ ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord;
+ ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord;
+
+
+ // quantize and encode LPC gain
+ WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
+ arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
+ status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
+ if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
+ {
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ // If this is the second 30ms of a 60ms frame reset this such that in the next call
+ // encoder starts fresh.
+ ISACenc_obj->frame_nb = 0;
+ }
+ return status;
+ }
+ iterCntr++;
+ }
+
+ if (frame_mode == 1 && ISACenc_obj->frame_nb == 0)
+ /* i.e. 60 ms framesize and just processed the first 30ms, */
+ /* go back to main function to buffer the other 30ms speech frame */
+ {
+ ISACenc_obj->frame_nb = 1;
+ return 0;
+ }
+ else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
+ {
+ ISACenc_obj->frame_nb = 0;
+ /* also update the framelength for next packet, in Adaptive mode only */
+ if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
+ ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
+ ISACenc_obj->current_framesamples);
+ }
+ }
+
+
+ /* complete arithmetic coding */
+ stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
+ /* can this be negative? */
+
+ if(CodingMode == 0)
+ {
+
+ /* update rate model and get minimum number of bytes in this packet */
+ MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
+ ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay);
+
+ /* if bitstream is too short, add garbage at the end */
+
+ /* Store length of coded data */
+ usefulstr_len = stream_length;
+
+ /* Make sure MinBytes does not exceed packet size limit */
+ if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) {
+ MinBytes = ISACenc_obj->payloadLimitBytes30;
+ } else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) {
+ MinBytes = ISACenc_obj->payloadLimitBytes60;
+ }
+
+ /* Make sure we don't allow more than 255 bytes of garbage data.
+ We store the length of the garbage data in 8 bits in the bitstream,
+ 255 is the max garbage lenght we can signal using 8 bits. */
+ if( MinBytes > usefulstr_len + 255 ) {
+ MinBytes = usefulstr_len + 255;
+ }
+
+ /* Save data for creation of multiple bitstreams */
+ if (ISACenc_obj->SaveEnc_ptr != NULL) {
+ (ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes;
+ }
+
+ while (stream_length < MinBytes)
+ {
+ if (stream_length & 0x0001){
+ ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
+ ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] |= (WebRtc_UWord16)(ISACenc_obj->bitstr_seed & 0xFF);
+ } else {
+ ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
+ ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] = WEBRTC_SPL_LSHIFT_U16(ISACenc_obj->bitstr_seed, 8);
+ }
+ stream_length++;
+ }
+
+ /* to get the real stream_length, without garbage */
+ if (usefulstr_len & 0x0001) {
+ ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00;
+ ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF;
+ }
+ else {
+ ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF;
+ ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += WEBRTC_SPL_LSHIFT_U16((MinBytes - usefulstr_len) & 0x00FF, 8);
+ }
+ }
+ else
+ {
+ /* update rate model */
+ WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
+ ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck);
+ }
+ return stream_length;
+}
+
+/* This function is used to create a new bitstream with new BWE.
+ The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl()
+ is used. The data needed is taken from the struct, where it was stored
+ when calling the encoder. */
+int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t *ISACenc_obj,
+ int BWnumber,
+ float scale)
+{
+ int ii;
+ int status;
+ WebRtc_Word16 BWno = BWnumber;
+ int stream_length = 0;
+
+ WebRtc_Word16 model;
+ const WebRtc_UWord16 *Q_PitchGain_cdf_ptr[1];
+ const WebRtc_UWord16 **cdf;
+ const ISAC_SaveEncData_t *SaveEnc_str;
+ WebRtc_Word32 tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
+ WebRtc_Word16 tmpLPCindex_g[KLT_ORDER_GAIN<<1];
+ WebRtc_Word16 tmp_fre[FRAMESAMPLES];
+ WebRtc_Word16 tmp_fim[FRAMESAMPLES];
+
+ SaveEnc_str = ISACenc_obj->SaveEnc_ptr;
+
+ /* Check if SaveEnc memory exists */
+ if (SaveEnc_str == NULL) {
+ return (-1);
+ }
+
+ /* Sanity Check - possible values for BWnumber is 0 - 23 */
+ if ((BWnumber < 0) || (BWnumber > 23)) {
+ return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
+ }
+
+ /* reset bitstream */
+ ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
+ ISACenc_obj->bitstr_obj.streamval = 0;
+ ISACenc_obj->bitstr_obj.stream_index = 0;
+ ISACenc_obj->bitstr_obj.full = 1;
+
+ /* encode frame length */
+ status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj);
+ if (status < 0) {
+ /* Wrong frame size */
+ return status;
+ }
+
+ /* encode bandwidth estimate */
+ status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
+ if (status < 0) {
+ return status;
+ }
+
+ /* Transcoding */
+ /* If scale < 1, rescale data to produce lower bitrate signal */
+ if ((0.0 < scale) && (scale < 1.0)) {
+ /* Compensate LPC gain */
+ for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
+ tmpLPCcoeffs_g[ii] = (WebRtc_Word32) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]);
+ }
+
+ /* Scale DFT */
+ for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
+ tmp_fre[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fre[ii]) ;
+ tmp_fim[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fim[ii]) ;
+ }
+ } else {
+ for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
+ tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii];
+ }
+
+ for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
+ tmp_fre[ii] = SaveEnc_str->fre[ii];
+ tmp_fim[ii] = SaveEnc_str->fim[ii];
+ }
+ }
+
+ /* Loop over number of 30 msec */
+ for (ii = 0; ii <= SaveEnc_str->startIdx; ii++)
+ {
+
+ /* encode pitch gains */
+ *Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
+ status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii],
+ Q_PitchGain_cdf_ptr, 1);
+ if (status < 0) {
+ return status;
+ }
+
+ /* entropy coding of quantization pitch lags */
+ /* voicing classificiation */
+ if (SaveEnc_str->meanGain[ii] <= 819) {
+ cdf = WebRtcIsacfix_kPitchLagPtrLo;
+ } else if (SaveEnc_str->meanGain[ii] <= 1638) {
+ cdf = WebRtcIsacfix_kPitchLagPtrMid;
+ } else {
+ cdf = WebRtcIsacfix_kPitchLagPtrHi;
+ }
+ status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj,
+ &SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES);
+ if (status < 0) {
+ return status;
+ }
+
+ /* LPC */
+ /* entropy coding of model number */
+ model = 0;
+ status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model,
+ WebRtcIsacfix_kModelCdfPtr, 1);
+ if (status < 0) {
+ return status;
+ }
+
+ /* entropy coding of quantization indices - LPC shape only */
+ status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii],
+ WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE);
+ if (status < 0) {
+ return status;
+ }
+
+ /* If transcoding, get new LPC gain indices */
+ if (scale < 1.0) {
+ WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]);
+ }
+
+ /* entropy coding of quantization indices - LPC gain */
+ status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii],
+ WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
+ if (status < 0) {
+ return status;
+ }
+
+ /* quantization and lossless coding */
+ status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF],
+ &ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]);
+ if (status < 0) {
+ return status;
+ }
+ }
+
+ /* complete arithmetic coding */
+ stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
+
+ return stream_length;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c
new file mode 100644
index 0000000..0b64d83
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c
@@ -0,0 +1,2072 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * entropy_coding.c
+ *
+ * This file contains all functions used to arithmetically
+ * encode the iSAC bistream.
+ *
+ */
+
+#include <stddef.h>
+
+#include "arith_routins.h"
+#include "spectrum_ar_model_tables.h"
+#include "pitch_gain_tables.h"
+#include "pitch_lag_tables.h"
+#include "entropy_coding.h"
+#include "lpc_tables.h"
+#include "settings.h"
+#include "signal_processing_library.h"
+
+
+/*
+ This function implements the fix-point correspondant function to lrint.
+
+ FLP: (WebRtc_Word32)floor(flt+.499999999999)
+ FIP: (fixVal+roundVal)>>qDomain
+
+ where roundVal = 2^(qDomain-1) = 1<<(qDomain-1)
+
+*/
+static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal, WebRtc_Word16 qDomain) {
+ WebRtc_Word32 intgr;
+ WebRtc_Word32 roundVal;
+
+ roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, qDomain-1);
+ intgr = WEBRTC_SPL_RSHIFT_W32(fixVal+roundVal, qDomain);
+
+ return intgr;
+}
+
+/*
+ __inline WebRtc_UWord32 stepwise(WebRtc_Word32 dinQ10) {
+
+ WebRtc_Word32 ind, diQ10, dtQ10;
+
+ diQ10 = dinQ10;
+ if (diQ10 < DPMIN_Q10)
+ diQ10 = DPMIN_Q10;
+ if (diQ10 >= DPMAX_Q10)
+ diQ10 = DPMAX_Q10 - 1;
+
+ dtQ10 = diQ10 - DPMIN_Q10;*/ /* Q10 + Q10 = Q10 */
+/* ind = (dtQ10 * 5) >> 10; */ /* 2^10 / 5 = 0.2 in Q10 */
+/* Q10 -> Q0 */
+
+/* return rpointsFIX_Q10[ind];
+
+ }
+*/
+
+/* logN(x) = logN(2)*log2(x) = 0.6931*log2(x). Output in Q8. */
+/* The input argument X to logN(X) is 2^17 times higher than the
+ input floating point argument Y to log(Y), since the X value
+ is a Q17 value. This can be compensated for after the call, by
+ subraction a value Z for each Q-step. One Q-step means that
+ X gets 2 thimes higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
+ 177.445678 should be subtracted (since logN() returns a Q8 value).
+ For a X value in Q17, the value 177.445678*17 = 3017 should be
+ subtracted */
+static WebRtc_Word16 CalcLogN(WebRtc_Word32 arg) {
+ WebRtc_Word16 zeros, log2, frac, logN;
+
+ zeros=WebRtcSpl_NormU32(arg);
+ frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_LSHIFT_W32(arg, zeros)&0x7FFFFFFF, 23);
+ log2=(WebRtc_Word16)(WEBRTC_SPL_LSHIFT_W32(31-zeros, 8)+frac); // log2(x) in Q8
+ logN=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
+ logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x.
+
+ return logN;
+}
+
+
+/*
+ expN(x) = 2^(a*x), where a = log2(e) ~= 1.442695
+
+ Input: Q8 (WebRtc_Word16)
+ Output: Q17 (WebRtc_Word32)
+
+ a = log2(e) = log2(exp(1)) ~= 1.442695 ==> a = 23637 in Q14 (1.442688)
+ To this value, 700 is added or subtracted in order to get an average error
+ nearer zero, instead of always same-sign.
+*/
+
+static WebRtc_Word32 CalcExpN(WebRtc_Word16 x) {
+ WebRtc_Word16 ax, axINT, axFRAC;
+ WebRtc_Word16 exp16;
+ WebRtc_Word32 exp;
+
+ if (x>=0) {
+ // ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
+ ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
+ axINT = WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
+ axFRAC = ax&0x00FF;
+ exp16 = WEBRTC_SPL_LSHIFT_W32(1, axINT); //Q0
+ axFRAC = axFRAC+256; //Q8
+ exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q0*Q8 = Q8
+ exp = WEBRTC_SPL_LSHIFT_W32(exp, 9); //Q17
+ } else {
+ // ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
+ ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
+ ax = -ax;
+ axINT = 1 + WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
+ axFRAC = 0x00FF - (ax&0x00FF);
+ exp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(32768, axINT); //Q15
+ axFRAC = axFRAC+256; //Q8
+ exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q15*Q8 = Q23
+ exp = WEBRTC_SPL_RSHIFT_W32(exp, 6); //Q17
+ }
+
+ return exp;
+}
+
+
+/* compute correlation from power spectrum */
+static void CalcCorrelation(WebRtc_Word32 *PSpecQ12, WebRtc_Word32 *CorrQ7)
+{
+ WebRtc_Word32 summ[FRAMESAMPLES/8];
+ WebRtc_Word32 diff[FRAMESAMPLES/8];
+ WebRtc_Word32 sum;
+ int k, n;
+
+ for (k = 0; k < FRAMESAMPLES/8; k++) {
+ summ[k] = WEBRTC_SPL_RSHIFT_W32(PSpecQ12[k] + PSpecQ12[FRAMESAMPLES/4-1 - k] + 16, 5);
+ diff[k] = WEBRTC_SPL_RSHIFT_W32(PSpecQ12[k] - PSpecQ12[FRAMESAMPLES/4-1 - k] + 16, 5);
+ }
+
+ sum = 2;
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += summ[n];
+ CorrQ7[0] = sum;
+
+ for (k = 0; k < AR_ORDER; k += 2) {
+ sum = 0;
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], diff[n]) + 256, 9);
+ CorrQ7[k+1] = sum;
+ }
+
+ for (k=1; k<AR_ORDER; k+=2) {
+ sum = 0;
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], summ[n]) + 256, 9);
+ CorrQ7[k+1] = sum;
+ }
+}
+
+
+/* compute inverse AR power spectrum */
+static void CalcInvArSpec(const WebRtc_Word16 *ARCoefQ12,
+ const WebRtc_Word32 gainQ10,
+ WebRtc_Word32 *CurveQ16)
+{
+ WebRtc_Word32 CorrQ11[AR_ORDER+1];
+ WebRtc_Word32 sum, tmpGain;
+ WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
+ const WebRtc_Word16 *CS_ptrQ9;
+ int k, n;
+ WebRtc_Word16 round, shftVal = 0, sh;
+
+ sum = 0;
+ for (n = 0; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
+ CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
+
+ /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
+ if(gainQ10>400000){
+ tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
+ round = 32;
+ shftVal = 6;
+ } else {
+ tmpGain = gainQ10;
+ round = 256;
+ shftVal = 9;
+ }
+
+ for (k = 1; k < AR_ORDER+1; k++) {
+ sum = 16384;
+ for (n = k; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
+ CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
+ }
+ sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ CurveQ16[n] = sum;
+
+ for (k = 1; k < AR_ORDER; k += 2) {
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], CorrQ11[k+1]) + 2, 2);
+ }
+
+ CS_ptrQ9 = WebRtcIsacfix_kCos[0];
+
+ /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
+ sh=WebRtcSpl_NormW32(CorrQ11[1]);
+ if (CorrQ11[1]==0) /* Use next correlation */
+ sh=WebRtcSpl_NormW32(CorrQ11[2]);
+
+ if (sh<9)
+ shftVal = 9 - sh;
+ else
+ shftVal = 0;
+
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
+ for (k = 2; k < AR_ORDER; k += 2) {
+ CS_ptrQ9 = WebRtcIsacfix_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
+ }
+
+ for (k=0; k<FRAMESAMPLES/8; k++) {
+ CurveQ16[FRAMESAMPLES/4-1 - k] = CurveQ16[k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
+ CurveQ16[k] += WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
+ }
+}
+
+static void CalcRootInvArSpec(const WebRtc_Word16 *ARCoefQ12,
+ const WebRtc_Word32 gainQ10,
+ WebRtc_UWord16 *CurveQ8)
+{
+ WebRtc_Word32 CorrQ11[AR_ORDER+1];
+ WebRtc_Word32 sum, tmpGain;
+ WebRtc_Word32 summQ16[FRAMESAMPLES/8];
+ WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
+
+ const WebRtc_Word16 *CS_ptrQ9;
+ int k, n, i;
+ WebRtc_Word16 round, shftVal = 0, sh;
+ WebRtc_Word32 res, in_sqrt, newRes;
+
+ sum = 0;
+ for (n = 0; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
+ CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
+
+ /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
+ if(gainQ10>400000){
+ tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
+ round = 32;
+ shftVal = 6;
+ } else {
+ tmpGain = gainQ10;
+ round = 256;
+ shftVal = 9;
+ }
+
+ for (k = 1; k < AR_ORDER+1; k++) {
+ sum = 16384;
+ for (n = k; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
+ CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
+ }
+ sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ summQ16[n] = sum;
+
+ for (k = 1; k < (AR_ORDER); k += 2) {
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ summQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(CorrQ11[k+1],WebRtcIsacfix_kCos[k][n]) + 2, 2);
+ }
+
+ CS_ptrQ9 = WebRtcIsacfix_kCos[0];
+
+ /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
+ sh=WebRtcSpl_NormW32(CorrQ11[1]);
+ if (CorrQ11[1]==0) /* Use next correlation */
+ sh=WebRtcSpl_NormW32(CorrQ11[2]);
+
+ if (sh<9)
+ shftVal = 9 - sh;
+ else
+ shftVal = 0;
+
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
+ for (k = 2; k < AR_ORDER; k += 2) {
+ CS_ptrQ9 = WebRtcIsacfix_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
+ }
+
+ in_sqrt = summQ16[0] + WEBRTC_SPL_LSHIFT_W32(diffQ16[0], shftVal);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = WEBRTC_SPL_LSHIFT_W32(1, WEBRTC_SPL_RSHIFT_W16(WebRtcSpl_GetSizeInBits(in_sqrt), 1));
+
+ for (k = 0; k < FRAMESAMPLES/8; k++)
+ {
+ in_sqrt = summQ16[k] + WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
+ i = 10;
+
+ /* make in_sqrt positive to prohibit sqrt of negative values */
+ if(in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ do
+ {
+ res = newRes;
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ } while (newRes != res && i-- > 0);
+
+ CurveQ8[k] = (WebRtc_Word16)newRes;
+ }
+ for (k = FRAMESAMPLES/8; k < FRAMESAMPLES/4; k++) {
+
+ in_sqrt = summQ16[FRAMESAMPLES/4-1 - k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[FRAMESAMPLES/4-1 - k], shftVal);
+ i = 10;
+
+ /* make in_sqrt positive to prohibit sqrt of negative values */
+ if(in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ do
+ {
+ res = newRes;
+ newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ } while (newRes != res && i-- > 0);
+
+ CurveQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+}
+
+
+
+/* generate array of dither samples in Q7 */
+static void GenerateDitherQ7(WebRtc_Word16 *bufQ7,
+ WebRtc_UWord32 seed,
+ WebRtc_Word16 length,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ int k;
+ WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14, shft;
+
+ if (AvgPitchGain_Q12 < 614) /* this threshold should be equal to that in decode_spec() */
+ {
+ for (k = 0; k < length-2; k += 3)
+ {
+ /* new random unsigned WebRtc_Word32 */
+ seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 (Q7) */
+ dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25); // * 128/4294967295
+
+ /* new random unsigned WebRtc_Word32 */
+ seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 */
+ dither2_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(seed + 16777216, 25);
+
+ shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 15);
+ if (shft < 5)
+ {
+ bufQ7[k] = dither1_Q7;
+ bufQ7[k+1] = dither2_Q7;
+ bufQ7[k+2] = 0;
+ }
+ else if (shft < 10)
+ {
+ bufQ7[k] = dither1_Q7;
+ bufQ7[k+1] = 0;
+ bufQ7[k+2] = dither2_Q7;
+ }
+ else
+ {
+ bufQ7[k] = 0;
+ bufQ7[k+1] = dither1_Q7;
+ bufQ7[k+2] = dither2_Q7;
+ }
+ }
+ }
+ else
+ {
+ dither_gain_Q14 = (WebRtc_Word16)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12));
+
+ /* dither on half of the coefficients */
+ for (k = 0; k < length-1; k += 2)
+ {
+ /* new random unsigned WebRtc_Word32 */
+ seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 */
+ dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25);
+
+ /* dither sample is placed in either even or odd index */
+ shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 1); /* either 0 or 1 */
+
+ bufQ7[k + shft] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(dither_gain_Q14, dither1_Q7) + 8192, 14);
+ bufQ7[k + 1 - shft] = 0;
+ }
+ }
+}
+
+
+
+
+/*
+ * function to decode the complex spectrum from the bitstream
+ * returns the total number of bytes in the stream
+ */
+WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
+ WebRtc_Word16 *frQ7,
+ WebRtc_Word16 *fiQ7,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ WebRtc_Word16 data[FRAMESAMPLES];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES/4];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 gainQ10;
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word16 len;
+ int k;
+
+ /* create dither signal */
+ GenerateDitherQ7(data, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12); /* Dither is output in vector 'Data' */
+
+ /* decode model parameters */
+ if (WebRtcIsacfix_DecodeRcCoef(streamdata, RCQ15) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ if (WebRtcIsacfix_DecodeGain2(streamdata, &gain2_Q10) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* compute inverse AR power spectrum */
+ CalcInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* arithmetic decoding of spectrum */
+ /* 'data' input and output. Input = Dither */
+ len = WebRtcIsacfix_DecLogisticMulti2(data, streamdata, invARSpec2_Q16, (WebRtc_Word16)FRAMESAMPLES);
+
+ if (len<1)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* subtract dither and scale down spectral samples with low SNR */
+ if (AvgPitchGain_Q12 <= 614)
+ {
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)30, 10),
+ (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2195456, 16));
+ *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
+ *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
+ *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
+ *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
+ }
+ }
+ else
+ {
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)36, 10),
+ (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2654208, 16));
+ *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
+ *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
+ *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
+ *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
+ }
+ }
+
+ return len;
+}
+
+
+int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
+ const WebRtc_Word16 *fi,
+ Bitstr_enc *streamdata,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ WebRtc_Word16 dataQ7[FRAMESAMPLES];
+ WebRtc_Word32 PSpec[FRAMESAMPLES/4];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES/4];
+ WebRtc_Word32 CorrQ7[AR_ORDER+1];
+ WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word16 val;
+ WebRtc_Word32 nrg;
+ WebRtc_UWord32 sum;
+ WebRtc_Word16 lft_shft;
+ WebRtc_Word16 status;
+ int k, n, j;
+
+
+ /* create dither_float signal */
+ GenerateDitherQ7(dataQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
+
+ /* add dither and quantize, and compute power spectrum */
+ /* Vector dataQ7 contains Dither in Q7 */
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ val = ((*fr++ + dataQ7[k] + 64) & 0xFF80) - dataQ7[k]; /* Data = Dither */
+ dataQ7[k] = val; /* New value in Data */
+ sum = WEBRTC_SPL_UMUL(val, val);
+
+ val = ((*fi++ + dataQ7[k+1] + 64) & 0xFF80) - dataQ7[k+1]; /* Data = Dither */
+ dataQ7[k+1] = val; /* New value in Data */
+ sum += WEBRTC_SPL_UMUL(val, val);
+
+ val = ((*fr++ + dataQ7[k+2] + 64) & 0xFF80) - dataQ7[k+2]; /* Data = Dither */
+ dataQ7[k+2] = val; /* New value in Data */
+ sum += WEBRTC_SPL_UMUL(val, val);
+
+ val = ((*fi++ + dataQ7[k+3] + 64) & 0xFF80) - dataQ7[k+3]; /* Data = Dither */
+ dataQ7[k+3] = val; /* New value in Data */
+ sum += WEBRTC_SPL_UMUL(val, val);
+
+ PSpec[k>>2] = WEBRTC_SPL_RSHIFT_U32(sum, 2);
+ }
+
+ /* compute correlation from power spectrum */
+ CalcCorrelation(PSpec, CorrQ7);
+
+
+ /* find AR coefficients */
+ /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
+ lft_shft = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
+
+ if (lft_shft > 0) {
+ for (k=0; k<AR_ORDER+1; k++)
+ CorrQ7_norm[k] = WEBRTC_SPL_LSHIFT_W32(CorrQ7[k], lft_shft);
+ } else {
+ for (k=0; k<AR_ORDER+1; k++)
+ CorrQ7_norm[k] = WEBRTC_SPL_RSHIFT_W32(CorrQ7[k], -lft_shft);
+ }
+
+ /* find RC coefficients */
+ WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
+
+ /* quantize & code RC Coef */
+ status = WebRtcIsacfix_EncodeRcCoef(RCQ15, streamdata);
+ if (status < 0) {
+ return status;
+ }
+
+ /* RC -> AR coefficients */
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ /* compute ARCoef' * Corr * ARCoef in Q19 */
+ nrg = 0;
+ for (j = 0; j <= AR_ORDER; j++) {
+ for (n = 0; n <= j; n++)
+ nrg += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(ARCoefQ12[j], WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CorrQ7_norm[j-n], ARCoefQ12[n]) + 256, 9)) + 4, 3);
+ for (n = j+1; n <= AR_ORDER; n++)
+ nrg += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(ARCoefQ12[j], WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CorrQ7_norm[n-j], ARCoefQ12[n]) + 256, 9)) + 4, 3);
+ }
+
+ if (lft_shft > 0)
+ nrg = WEBRTC_SPL_RSHIFT_W32(nrg, lft_shft);
+ else
+ nrg = WEBRTC_SPL_LSHIFT_W32(nrg, -lft_shft);
+
+ if(nrg>131072)
+ gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES >> 2, nrg); /* also shifts 31 bits to the left! */
+ else
+ gain2_Q10 = WEBRTC_SPL_RSHIFT_W32(FRAMESAMPLES, 2);
+
+ /* quantize & code gain2_Q10 */
+ if (WebRtcIsacfix_EncodeGain2(&gain2_Q10, streamdata))
+ return -1;
+
+ /* compute inverse AR magnitude spectrum */
+ CalcRootInvArSpec(ARCoefQ12, gain2_Q10, invARSpecQ8);
+
+
+ /* arithmetic coding of spectrum */
+ status = WebRtcIsacfix_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, (WebRtc_Word16)FRAMESAMPLES);
+ if ( status )
+ return( status );
+
+ return 0;
+}
+
+
+/* Matlab's LAR definition */
+static void Rc2LarFix(const WebRtc_Word16 *rcQ15, WebRtc_Word32 *larQ17, WebRtc_Word16 order) {
+
+ /*
+
+ This is a piece-wise implemenetation of a rc2lar-function (all values in the comment
+ are Q15 values and are based on [0 24956/32768 30000/32768 32500/32768], i.e.
+ [0.76159667968750 0.91552734375000 0.99182128906250]
+
+ x0 x1 a k x0(again) b
+ ==================================================================================
+ 0.00 0.76: 0 2.625997508581 0 0
+ 0.76 0.91: 2.000012018559 7.284502668663 0.761596679688 -3.547841027073
+ 0.91 0.99: 3.121320351712 31.115835041229 0.915527343750 -25.366077452148
+ 0.99 1.00: 5.495270168700 686.663805654056 0.991821289063 -675.552510708011
+
+ The implementation is y(x)= a + (x-x0)*k, but this can be simplified to
+
+ y(x) = a-x0*k + x*k = b + x*k, where b = a-x0*k
+
+ akx=[0 2.625997508581 0
+ 2.000012018559 7.284502668663 0.761596679688
+ 3.121320351712 31.115835041229 0.915527343750
+ 5.495270168700 686.663805654056 0.991821289063];
+
+ b = akx(:,1) - akx(:,3).*akx(:,2)
+
+ [ 0.0
+ -3.547841027073
+ -25.366077452148
+ -675.552510708011]
+
+ */
+
+ int k;
+ WebRtc_Word16 rc;
+ WebRtc_Word32 larAbsQ17;
+
+ for (k = 0; k < order; k++) {
+
+ rc = WEBRTC_SPL_ABS_W16(rcQ15[k]); //Q15
+
+ /* Calculate larAbsQ17 in Q17 from rc in Q15 */
+
+ if (rc<24956) { //0.7615966 in Q15
+ // (Q15*Q13)>>11 = Q17
+ larAbsQ17 = WEBRTC_SPL_MUL_16_16_RSFT(rc, 21512, 11);
+ } else if (rc<30000) { //0.91552734375 in Q15
+ // Q17 + (Q15*Q12)>>10 = Q17
+ larAbsQ17 = -465024 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 29837, 10);
+ } else if (rc<32500) { //0.99182128906250 in Q15
+ // Q17 + (Q15*Q10)>>8 = Q17
+ larAbsQ17 = -3324784 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 31863, 8);
+ } else {
+ // Q17 + (Q15*Q5)>>3 = Q17
+ larAbsQ17 = -88546020 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 21973, 3);
+ }
+
+ if (rcQ15[k]>0) {
+ larQ17[k] = larAbsQ17;
+ } else {
+ larQ17[k] = -larAbsQ17;
+ }
+ }
+}
+
+
+static void Lar2RcFix(const WebRtc_Word32 *larQ17, WebRtc_Word16 *rcQ15, WebRtc_Word16 order) {
+
+ /*
+ This is a piece-wise implemenetation of a lar2rc-function
+ See comment in Rc2LarFix() about details.
+ */
+
+ int k;
+ WebRtc_Word16 larAbsQ11;
+ WebRtc_Word32 rc;
+
+ for (k = 0; k < order; k++) {
+
+ larAbsQ11 = (WebRtc_Word16) WEBRTC_SPL_ABS_W32(WEBRTC_SPL_RSHIFT_W32(larQ17[k]+32,6)); //Q11
+
+ if (larAbsQ11<4097) { //2.000012018559 in Q11
+ // Q11*Q16>>12 = Q15
+ rc = WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24957, 12);
+ } else if (larAbsQ11<6393) { //3.121320351712 in Q11
+ // (Q11*Q17 + Q13)>>13 = Q15
+ rc = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(larAbsQ11, 17993) + 130738688), 13);
+ } else if (larAbsQ11<11255) { //5.495270168700 in Q11
+ // (Q11*Q19 + Q30)>>15 = Q15
+ rc = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(larAbsQ11, 16850) + 875329820), 15);
+ } else {
+ // (Q11*Q24>>16 + Q19)>>4 = Q15
+ rc = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24433, 16)) + 515804), 4);
+ }
+
+ if (larQ17[k]<=0) {
+ rc = -rc;
+ }
+
+ rcQ15[k] = (WebRtc_Word16) rc; // Q15
+ }
+}
+
+static void Poly2LarFix(WebRtc_Word16 *lowbandQ15,
+ WebRtc_Word16 orderLo,
+ WebRtc_Word16 *hibandQ15,
+ WebRtc_Word16 orderHi,
+ WebRtc_Word16 Nsub,
+ WebRtc_Word32 *larsQ17) {
+
+ int k, n;
+ WebRtc_Word32 *outpQ17;
+ WebRtc_Word16 orderTot;
+ WebRtc_Word32 larQ17[MAX_ORDER]; // Size 7+6 is enough
+
+ orderTot = (orderLo + orderHi);
+ outpQ17 = larsQ17;
+ for (k = 0; k < Nsub; k++) {
+
+ Rc2LarFix(lowbandQ15, larQ17, orderLo);
+
+ for (n = 0; n < orderLo; n++)
+ outpQ17[n] = larQ17[n]; //Q17
+
+ Rc2LarFix(hibandQ15, larQ17, orderHi);
+
+ for (n = 0; n < orderHi; n++)
+ outpQ17[n + orderLo] = larQ17[n]; //Q17;
+
+ outpQ17 += orderTot;
+ lowbandQ15 += orderLo;
+ hibandQ15 += orderHi;
+ }
+}
+
+
+static void Lar2polyFix(WebRtc_Word32 *larsQ17,
+ WebRtc_Word16 *lowbandQ15,
+ WebRtc_Word16 orderLo,
+ WebRtc_Word16 *hibandQ15,
+ WebRtc_Word16 orderHi,
+ WebRtc_Word16 Nsub) {
+
+ int k, n;
+ WebRtc_Word16 orderTot;
+ WebRtc_Word16 *outplQ15, *outphQ15;
+ WebRtc_Word32 *inpQ17;
+ WebRtc_Word16 rcQ15[7+6];
+
+ orderTot = (orderLo + orderHi);
+ outplQ15 = lowbandQ15;
+ outphQ15 = hibandQ15;
+ inpQ17 = larsQ17;
+ for (k = 0; k < Nsub; k++) {
+
+ /* gains not handled here as in the FLP version */
+
+ /* Low band */
+ Lar2RcFix(&inpQ17[0], rcQ15, orderLo);
+ for (n = 0; n < orderLo; n++)
+ outplQ15[n] = rcQ15[n]; // Refl. coeffs
+
+ /* High band */
+ Lar2RcFix(&inpQ17[orderLo], rcQ15, orderHi);
+ for (n = 0; n < orderHi; n++)
+ outphQ15[n] = rcQ15[n]; // Refl. coeffs
+
+ inpQ17 += orderTot;
+ outplQ15 += orderLo;
+ outphQ15 += orderHi;
+ }
+}
+
+int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *LPCCoef_loQ15,
+ WebRtc_Word16 *LPCCoef_hiQ15,
+ Bitstr_dec *streamdata,
+ WebRtc_Word16 *outmodel) {
+
+ WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_GAIN+KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
+ int err;
+
+ err = WebRtcIsacfix_DecodeLpcCoef(streamdata, larsQ17, gain_lo_hiQ17, outmodel);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_LPC;
+
+ Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES);
+
+ return 0;
+}
+
+/* decode & dequantize LPC Coef */
+int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
+ WebRtc_Word32 *LPCCoefQ17,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *outmodel)
+{
+ int j, k, n;
+ int err;
+ WebRtc_Word16 pos, pos2, posg, poss, offsg, offss, offs2;
+ WebRtc_Word16 gainpos;
+ WebRtc_Word16 model;
+ WebRtc_Word16 index_QQ[KLT_ORDER_SHAPE];
+ WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+ WebRtc_Word16 tmpcoeffs_sQ10[KLT_ORDER_SHAPE];
+ WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
+ WebRtc_Word32 tmpcoeffs2_sQ18[KLT_ORDER_SHAPE];
+ WebRtc_Word32 sumQQ;
+ WebRtc_Word16 sumQQ16;
+ WebRtc_Word32 tmp32;
+
+
+
+ /* entropy decoding of model number */
+ err = WebRtcIsacfix_DecHistOneStepMulti(&model, streamdata, WebRtcIsacfix_kModelCdfPtr, WebRtcIsacfix_kModelInitIndex, 1);
+ if (err<0) // error check
+ return err;
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsacfix_DecHistOneStepMulti(index_QQ, streamdata, WebRtcIsacfix_kCdfShapePtr[model], WebRtcIsacfix_kInitIndexShape[model], KLT_ORDER_SHAPE);
+ if (err<0) // error check
+ return err;
+ /* find quantization levels for coefficients */
+ for (k=0; k<KLT_ORDER_SHAPE; k++) {
+ tmpcoeffs_sQ10[WebRtcIsacfix_kSelIndShape[k]] = WebRtcIsacfix_kLevelsShapeQ10[WebRtcIsacfix_kOfLevelsShape[model]+WebRtcIsacfix_kOffsetShape[model][k] + index_QQ[k]];
+ }
+
+ err = WebRtcIsacfix_DecHistOneStepMulti(index_QQ, streamdata, WebRtcIsacfix_kCdfGainPtr[model], WebRtcIsacfix_kInitIndexGain[model], KLT_ORDER_GAIN);
+ if (err<0) // error check
+ return err;
+ /* find quantization levels for coefficients */
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+ tmpcoeffs_gQ17[WebRtcIsacfix_kSelIndGain[k]] = WebRtcIsacfix_kLevelsGainQ17[WebRtcIsacfix_kOfLevelsGain[model]+ WebRtcIsacfix_kOffsetGain[model][k] + index_QQ[k]];
+ }
+
+
+ /* inverse KLT */
+
+ /* left transform */ // Transpose matrix!
+ offsg = 0;
+ offss = 0;
+ posg = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = offsg;
+ pos2 = offs2;
+ for (n=0; n<2; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[model][pos2], tmpcoeffs_gQ17[pos]<<5)); // (Q15*Q17)>>(16-5) = Q21
+ pos++;
+ pos2++;
+ }
+ tmpcoeffs2_gQ21[posg] = sumQQ; //Q21
+ posg++;
+ offs2 += 2;
+ }
+ offs2 = 0;
+
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = offss;
+ pos2 = offs2;
+ for (n=0; n<LPC_SHAPE_ORDER; n++) {
+ sumQQ += WEBRTC_SPL_MUL_16_16_RSFT(tmpcoeffs_sQ10[pos], WebRtcIsacfix_kT1ShapeQ15[model][pos2], 7); // (Q10*Q15)>>7 = Q18
+ pos++;
+ pos2++;
+ }
+ tmpcoeffs2_sQ18[poss] = sumQQ; //Q18
+ poss++;
+ offs2 += LPC_SHAPE_ORDER;
+ }
+ offsg += 2;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* right transform */ // Transpose matrix
+ offsg = 0;
+ offss = 0;
+ posg = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[model][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
+ pos += 2;
+ pos2 += SUBFRAMES;
+
+ }
+ tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
+ posg++;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[model][pos2], tmpcoeffs2_sQ18[pos])); // (Q15*Q18)>>16 = Q17
+ pos += LPC_SHAPE_ORDER;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_sQ17[poss] = sumQQ;
+ poss++;
+ }
+ offsg += 2;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* scaling, mean addition, and gain restoration */
+ gainpos = 0;
+ posg = 0;poss = 0;pos=0;
+ for (k=0; k<SUBFRAMES; k++) {
+
+ /* log gains */
+ sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+ sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
+ sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
+ gain_lo_hiQ17[gainpos] = sumQQ; //Q17
+ gainpos++;
+ posg++;
+
+ sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+ sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
+ sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
+ gain_lo_hiQ17[gainpos] = sumQQ; //Q17
+ gainpos++;
+ posg++;
+
+ /* lo band LAR coeffs */
+ for (n=0; n<ORDERLO; n++, pos++, poss++) {
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(31208, tmpcoeffs_sQ17[poss]); // (Q16*Q17)>>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16
+ tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17
+ LPCCoefQ17[pos] = tmp32;
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<ORDERHI; n++, pos++, poss++) {
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]), 3); // ((Q13*Q17)>>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13
+ tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17
+ LPCCoefQ17[pos] = tmp32;
+ }
+ }
+
+
+ *outmodel=model;
+
+ return 0;
+}
+
+/* estimate codel length of LPC Coef */
+static int EstCodeLpcCoef(WebRtc_Word32 *LPCCoefQ17,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *model,
+ WebRtc_Word32 *sizeQ11,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData,
+ transcode_obj *transcodingParam) {
+ int j, k, n;
+ WebRtc_Word16 posQQ, pos2QQ, gainpos;
+ WebRtc_Word16 pos, pos2, poss, posg, offsg, offss, offs2;
+ WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN], index_sQQ[KLT_ORDER_SHAPE];
+ WebRtc_Word16 index_ovr_gQQ[KLT_ORDER_GAIN], index_ovr_sQQ[KLT_ORDER_SHAPE];
+ WebRtc_Word32 BitsQQ;
+
+ WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
+ WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs2_sQ17[KLT_ORDER_SHAPE];
+ WebRtc_Word32 sumQQ;
+ WebRtc_Word32 tmp32;
+ WebRtc_Word16 sumQQ16;
+ int status = 0;
+
+ /* write LAR coefficients to statistics file */
+ /* Save data for creation of multiple bitstreams (and transcoding) */
+ if (encData != NULL) {
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+ encData->LPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k];
+ }
+ }
+
+ /* log gains, mean removal and scaling */
+ posg = 0;poss = 0;pos=0; gainpos=0;
+
+ for (k=0; k<SUBFRAMES; k++) {
+ /* log gains */
+
+ /* The input argument X to logN(X) is 2^17 times higher than the
+ input floating point argument Y to log(Y), since the X value
+ is a Q17 value. This can be compensated for after the call, by
+ subraction a value Z for each Q-step. One Q-step means that
+ X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
+ 177.445678 should be subtracted (since logN() returns a Q8 value).
+ For a X value in Q17, the value 177.445678*17 = 3017 should be
+ subtracted */
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+
+ /* lo band LAR coeffs */
+ for (n=0; n<ORDERLO; n++, poss++, pos++) {
+ tmp32 = LPCCoefQ17[pos] - WebRtcIsacfix_kMeansShapeQ17[0][poss]; //Q17
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(17203, tmp32<<3); // tmp32 = 2.1*tmp32
+ tmpcoeffs_sQ17[poss] = tmp32; //Q17
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<ORDERHI; n++, poss++, pos++) {
+ tmp32 = LPCCoefQ17[pos] - WebRtcIsacfix_kMeansShapeQ17[0][poss]; //Q17
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(14746, tmp32<<1); // tmp32 = 0.45*tmp32
+ tmpcoeffs_sQ17[poss] = tmp32; //Q17
+ }
+
+ }
+
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ offss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<2; n++) {
+ sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
+ pos++;
+ pos2 += 2;
+ }
+ tmpcoeffs2_gQ21[posg] = sumQQ;
+ posg++;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = offss;
+ pos2 = k;
+ for (n=0; n<LPC_SHAPE_ORDER; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1ShapeQ15[0][pos2], tmpcoeffs_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
+ pos++;
+ pos2 += LPC_SHAPE_ORDER;
+ }
+ tmpcoeffs2_sQ17[poss] = sumQQ; //Q17
+ poss++;
+ }
+ offsg += 2;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offss = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
+ pos += 2;
+ pos2++;
+ }
+ tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
+ posg++;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[0][pos2], tmpcoeffs2_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
+ pos += LPC_SHAPE_ORDER;
+ pos2++;
+ }
+ tmpcoeffs_sQ17[poss] = sumQQ;
+ poss++;
+ }
+ offs2 += SUBFRAMES;
+ offsg += 2;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* quantize coefficients */
+
+ BitsQQ = 0;
+ for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
+ {
+ posQQ = WebRtcIsacfix_kSelIndGain[k];
+ pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+
+ index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
+ if (index_gQQ[k] < 0) {
+ index_gQQ[k] = 0;
+ }
+ else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
+ index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
+ }
+ index_ovr_gQQ[k] = WebRtcIsacfix_kOffsetGain[0][k]+index_gQQ[k];
+ posQQ = WebRtcIsacfix_kOfLevelsGain[0] + index_ovr_gQQ[k];
+
+ /* Save data for creation of multiple bitstreams */
+ if (encData != NULL) {
+ encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k];
+ }
+
+ /* determine number of bits */
+ sumQQ = WebRtcIsacfix_kCodeLenGainQ11[posQQ]; //Q11
+ BitsQQ += sumQQ;
+ }
+
+ for (k=0; k<KLT_ORDER_SHAPE; k++) //ATTN: ok?
+ {
+ index_sQQ[k] = (WebRtc_Word16)(CalcLrIntQ(tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]], 17) + WebRtcIsacfix_kQuantMinShape[k]); //ATTN: ok?
+
+ if (index_sQQ[k] < 0)
+ index_sQQ[k] = 0;
+ else if (index_sQQ[k] > WebRtcIsacfix_kMaxIndShape[k])
+ index_sQQ[k] = WebRtcIsacfix_kMaxIndShape[k];
+ index_ovr_sQQ[k] = WebRtcIsacfix_kOffsetShape[0][k]+index_sQQ[k];
+
+ posQQ = WebRtcIsacfix_kOfLevelsShape[0] + index_ovr_sQQ[k];
+ sumQQ = WebRtcIsacfix_kCodeLenShapeQ11[posQQ]; //Q11
+ BitsQQ += sumQQ;
+ }
+
+
+
+ *model = 0;
+ *sizeQ11=BitsQQ;
+
+ /* entropy coding of model number */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, model, WebRtcIsacfix_kModelCdfPtr, 1);
+ if (status < 0) {
+ return status;
+ }
+
+ /* entropy coding of quantization indices - shape only */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, index_sQQ, WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE);
+ if (status < 0) {
+ return status;
+ }
+
+ /* Save data for creation of multiple bitstreams */
+ if (encData != NULL) {
+ for (k=0; k<KLT_ORDER_SHAPE; k++)
+ {
+ encData->LPCindex_s[KLT_ORDER_SHAPE*encData->startIdx + k] = index_sQQ[k];
+ }
+ }
+ /* save the state of the bitstream object 'streamdata' for the possible bit-rate reduction */
+ transcodingParam->full = streamdata->full;
+ transcodingParam->stream_index = streamdata->stream_index;
+ transcodingParam->streamval = streamdata->streamval;
+ transcodingParam->W_upper = streamdata->W_upper;
+ transcodingParam->beforeLastWord = streamdata->stream[streamdata->stream_index-1];
+ transcodingParam->lastWord = streamdata->stream[streamdata->stream_index];
+
+ /* entropy coding of index */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
+ if (status < 0) {
+ return status;
+ }
+
+ /* find quantization levels for shape coefficients */
+ for (k=0; k<KLT_ORDER_SHAPE; k++) {
+ tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]] = WEBRTC_SPL_MUL(128, WebRtcIsacfix_kLevelsShapeQ10[WebRtcIsacfix_kOfLevelsShape[0]+index_ovr_sQQ[k]]);
+
+ }
+ /* inverse KLT */
+
+ /* left transform */ // Transpose matrix!
+ offss = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = offss;
+ pos2 = offs2;
+ for (n=0; n<LPC_SHAPE_ORDER; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1ShapeQ15[0][pos2], tmpcoeffs_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
+ pos++;
+ pos2++;
+ }
+ tmpcoeffs2_sQ17[poss] = sumQQ;
+
+ poss++;
+ offs2 += LPC_SHAPE_ORDER;
+ }
+ offss += LPC_SHAPE_ORDER;
+ }
+
+
+ /* right transform */ // Transpose matrix
+ offss = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[0][pos2], tmpcoeffs2_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
+ pos += LPC_SHAPE_ORDER;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_sQ17[poss] = sumQQ;
+ poss++;
+ }
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* scaling, mean addition, and gain restoration */
+ poss = 0;pos=0;
+ for (k=0; k<SUBFRAMES; k++) {
+
+ /* lo band LAR coeffs */
+ for (n=0; n<ORDERLO; n++, pos++, poss++) {
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(31208, tmpcoeffs_sQ17[poss]); // (Q16*Q17)>>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16
+ tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17
+ LPCCoefQ17[pos] = tmp32;
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<ORDERHI; n++, pos++, poss++) {
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]), 3); // ((Q13*Q17)>>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13
+ tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17
+ LPCCoefQ17[pos] = tmp32;
+ }
+
+ }
+
+ //to update tmpcoeffs_gQ17 to the proper state
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+ tmpcoeffs_gQ17[WebRtcIsacfix_kSelIndGain[k]] = WebRtcIsacfix_kLevelsGainQ17[WebRtcIsacfix_kOfLevelsGain[0]+index_ovr_gQQ[k]];
+ }
+
+
+
+ /* find quantization levels for coefficients */
+
+ /* left transform */
+ offsg = 0;
+ posg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = offsg;
+ pos2 = offs2;
+ for (n=0; n<2; n++) {
+ sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][pos2], tmpcoeffs_gQ17[pos])<<1); // (Q15*Q17)>>(16-1) = Q17
+ pos++;
+ pos2++;
+ }
+ tmpcoeffs2_gQ21[posg] = WEBRTC_SPL_LSHIFT_W32(sumQQ, 4); //Q17<<4 = Q21
+ posg++;
+ offs2 += 2;
+ }
+ offsg += 2;
+ }
+
+ /* right transform */ // Transpose matrix
+ offsg = 0;
+ posg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
+ pos += 2;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
+ posg++;
+ }
+ offsg += 2;
+ }
+
+ /* scaling, mean addition, and gain restoration */
+ posg = 0;
+ gainpos = 0;
+ for (k=0; k<2*SUBFRAMES; k++) {
+
+ sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
+ sumQQ16 += WebRtcIsacfix_kMeansGainQ8[0][posg];
+ sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
+ gain_lo_hiQ17[gainpos] = sumQQ; //Q17
+
+ gainpos++;
+ pos++;posg++;
+ }
+
+ return 0;
+}
+
+int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData) {
+ int j, k, n;
+ WebRtc_Word16 posQQ, pos2QQ, gainpos;
+ WebRtc_Word16 pos, pos2, posg, offsg, offs2;
+ WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN];
+
+ WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+ WebRtc_Word32 sumQQ;
+ int status = 0;
+
+ /* write LAR coefficients to statistics file */
+ /* Save data for creation of multiple bitstreams (and transcoding) */
+ if (encData != NULL) {
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+ encData->LPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k];
+ }
+ }
+
+ /* log gains, mean removal and scaling */
+ posg = 0; pos = 0; gainpos = 0;
+
+ for (k=0; k<SUBFRAMES; k++) {
+ /* log gains */
+
+ /* The input argument X to logN(X) is 2^17 times higher than the
+ input floating point argument Y to log(Y), since the X value
+ is a Q17 value. This can be compensated for after the call, by
+ subraction a value Z for each Q-step. One Q-step means that
+ X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
+ 177.445678 should be subtracted (since logN() returns a Q8 value).
+ For a X value in Q17, the value 177.445678*17 = 3017 should be
+ subtracted */
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+ }
+
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<2; n++) {
+ sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
+ pos++;
+ pos2 += 2;
+ }
+ tmpcoeffs2_gQ21[posg] = sumQQ;
+ posg++;
+ }
+ offsg += 2;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
+ pos += 2;
+ pos2++;
+ }
+ tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
+ posg++;
+ }
+ offsg += 2;
+ offs2 += SUBFRAMES;
+ }
+
+ /* quantize coefficients */
+
+ for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
+ {
+ posQQ = WebRtcIsacfix_kSelIndGain[k];
+ pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+
+ index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
+ if (index_gQQ[k] < 0) {
+ index_gQQ[k] = 0;
+ }
+ else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
+ index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
+ }
+
+ /* Save data for creation of multiple bitstreams */
+ if (encData != NULL) {
+ encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k];
+ }
+ }
+
+ /* entropy coding of index */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
+ if (status < 0) {
+ return status;
+ }
+
+ return 0;
+}
+
+
+int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *LPCCoef_loQ15,
+ WebRtc_Word16 *LPCCoef_hiQ15,
+ WebRtc_Word16 *model,
+ WebRtc_Word32 *sizeQ11,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData,
+ transcode_obj *transcodeParam)
+{
+ int status = 0;
+ WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
+ // = (6+12)*6 == 108
+
+ Poly2LarFix(LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES, larsQ17);
+
+ status = EstCodeLpcCoef(larsQ17, gain_lo_hiQ17, model, sizeQ11, streamdata, encData, transcodeParam);
+ if (status < 0) {
+ return (status);
+ }
+
+ Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES);
+
+ return 0;
+}
+
+
+/* decode & dequantize RC */
+int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata, WebRtc_Word16 *RCQ15)
+{
+ int k, err;
+ WebRtc_Word16 index[AR_ORDER];
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsacfix_DecHistOneStepMulti(index, streamdata, WebRtcIsacfix_kRcCdfPtr, WebRtcIsacfix_kRcInitInd, AR_ORDER);
+ if (err<0) // error check
+ return err;
+
+ /* find quantization levels for reflection coefficients */
+ for (k=0; k<AR_ORDER; k++)
+ {
+ RCQ15[k] = *(WebRtcIsacfix_kRcLevPtr[k] + index[k]);
+ }
+
+ return 0;
+}
+
+
+
+/* quantize & code RC */
+int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15, Bitstr_enc *streamdata)
+{
+ int k;
+ WebRtc_Word16 index[AR_ORDER];
+ int status;
+
+ /* quantize reflection coefficients (add noise feedback?) */
+ for (k=0; k<AR_ORDER; k++)
+ {
+ index[k] = WebRtcIsacfix_kRcInitInd[k];
+
+ if (RCQ15[k] > WebRtcIsacfix_kRcBound[index[k]])
+ {
+ while (RCQ15[k] > WebRtcIsacfix_kRcBound[index[k] + 1])
+ index[k]++;
+ }
+ else
+ {
+ while (RCQ15[k] < WebRtcIsacfix_kRcBound[--index[k]]) ;
+ }
+
+ RCQ15[k] = *(WebRtcIsacfix_kRcLevPtr[k] + index[k]);
+ }
+
+
+ /* entropy coding of quantization indices */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, index, WebRtcIsacfix_kRcCdfPtr, AR_ORDER);
+
+ /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
+ return status;
+}
+
+
+/* decode & dequantize squared Gain */
+int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata, WebRtc_Word32 *gainQ10)
+{
+ int err;
+ WebRtc_Word16 index;
+
+ /* entropy decoding of quantization index */
+ err = WebRtcIsacfix_DecHistOneStepMulti(
+ &index,
+ streamdata,
+ WebRtcIsacfix_kGainPtr,
+ WebRtcIsacfix_kGainInitInd,
+ 1);
+ /* error check */
+ if (err<0) {
+ return err;
+ }
+
+ /* find quantization level */
+ *gainQ10 = WebRtcIsacfix_kGain2Lev[index];
+
+ return 0;
+}
+
+
+
+/* quantize & code squared Gain */
+int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gainQ10, Bitstr_enc *streamdata)
+{
+ WebRtc_Word16 index;
+ int status = 0;
+
+ /* find quantization index */
+ index = WebRtcIsacfix_kGainInitInd[0];
+ if (*gainQ10 > WebRtcIsacfix_kGain2Bound[index])
+ {
+ while (*gainQ10 > WebRtcIsacfix_kGain2Bound[index + 1])
+ index++;
+ }
+ else
+ {
+ while (*gainQ10 < WebRtcIsacfix_kGain2Bound[--index]) ;
+ }
+
+ /* dequantize */
+ *gainQ10 = WebRtcIsacfix_kGain2Lev[index];
+
+ /* entropy coding of quantization index */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, &index, WebRtcIsacfix_kGainPtr, 1);
+
+ /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
+ return status;
+}
+
+
+/* code and decode Pitch Gains and Lags functions */
+
+/* decode & dequantize Pitch Gains */
+int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata, WebRtc_Word16 *PitchGains_Q12)
+{
+ int err;
+ WebRtc_Word16 index_comb;
+ const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
+
+ /* entropy decoding of quantization indices */
+ *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
+ err = WebRtcIsacfix_DecHistBisectMulti(&index_comb, streamdata, pitch_gain_cdf_ptr, WebRtcIsacfix_kCdfTableSizeGain, 1);
+ /* error check, Q_mean_Gain.. tables are of size 144 */
+ if ((err<0) || (index_comb<0) || (index_comb>144))
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN;
+
+ /* unquantize back to pitch gains by table look-up */
+ PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb];
+ PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb];
+ PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb];
+ PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb];
+
+ return 0;
+}
+
+
+/* quantize & code Pitch Gains */
+int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12, Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
+{
+ int k,j;
+ WebRtc_Word16 SQ15[PITCH_SUBFRAMES];
+ WebRtc_Word16 index[3];
+ WebRtc_Word16 index_comb;
+ const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
+ WebRtc_Word32 CQ17;
+ int status = 0;
+
+
+ /* get the approximate arcsine (almost linear)*/
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ SQ15[k] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
+
+
+ /* find quantization index; only for the first three transform coefficients */
+ for (k=0; k<3; k++)
+ {
+ /* transform */
+ CQ17=0;
+ for (j=0; j<PITCH_SUBFRAMES; j++) {
+ CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], SQ15[j],10); // Q17
+ }
+
+ index[k] = (WebRtc_Word16)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
+
+ /* check that the index is not outside the boundaries of the table */
+ if (index[k] < WebRtcIsacfix_kLowerlimiGain[k]) index[k] = WebRtcIsacfix_kLowerlimiGain[k];
+ else if (index[k] > WebRtcIsacfix_kUpperlimitGain[k]) index[k] = WebRtcIsacfix_kUpperlimitGain[k];
+ index[k] -= WebRtcIsacfix_kLowerlimiGain[k];
+ }
+
+ /* calculate unique overall index */
+ index_comb = (WebRtc_Word16)(WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[0], index[0]) +
+ WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[1], index[1]) + index[2]);
+
+ /* unquantize back to pitch gains by table look-up */
+ // (Y)
+ PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb];
+ PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb];
+ PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb];
+ PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb];
+
+
+ /* entropy coding of quantization pitch gains */
+ *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
+ status = WebRtcIsacfix_EncHistMulti(streamdata, &index_comb, pitch_gain_cdf_ptr, 1);
+ if (status < 0) {
+ return status;
+ }
+
+ /* Save data for creation of multiple bitstreams */
+ if (encData != NULL) {
+ encData->pitchGain_index[encData->startIdx] = index_comb;
+ }
+
+ return 0;
+}
+
+
+
+/* Pitch LAG */
+
+
+/* decode & dequantize Pitch Lags */
+int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
+ WebRtc_Word16 *PitchGain_Q12,
+ WebRtc_Word16 *PitchLags_Q7)
+{
+ int k, err;
+ WebRtc_Word16 index[PITCH_SUBFRAMES];
+ const WebRtc_Word16 *mean_val2Q10, *mean_val4Q10;
+
+ const WebRtc_Word16 *lower_limit;
+ const WebRtc_UWord16 *init_index;
+ const WebRtc_UWord16 *cdf_size;
+ const WebRtc_UWord16 **cdf;
+
+ WebRtc_Word32 meangainQ12;
+ WebRtc_Word32 CQ11, CQ10,tmp32a,tmp32b;
+ WebRtc_Word16 shft,tmp16a,tmp16c;
+
+ meangainQ12=0;
+ for (k = 0; k < 4; k++)
+ meangainQ12 += PitchGain_Q12[k];
+
+ meangainQ12 = WEBRTC_SPL_RSHIFT_W32(meangainQ12, 2); // Get average
+
+ /* voicing classificiation */
+ if (meangainQ12 <= 819) { // mean_gain < 0.2
+ shft = -1; // StepSize=2.0;
+ cdf = WebRtcIsacfix_kPitchLagPtrLo;
+ cdf_size = WebRtcIsacfix_kPitchLagSizeLo;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo;
+ lower_limit = WebRtcIsacfix_kLowerLimitLo;
+ init_index = WebRtcIsacfix_kInitIndLo;
+ } else if (meangainQ12 <= 1638) { // mean_gain < 0.4
+ shft = 0; // StepSize=1.0;
+ cdf = WebRtcIsacfix_kPitchLagPtrMid;
+ cdf_size = WebRtcIsacfix_kPitchLagSizeMid;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid;
+ lower_limit = WebRtcIsacfix_kLowerLimitMid;
+ init_index = WebRtcIsacfix_kInitIndMid;
+ } else {
+ shft = 1; // StepSize=0.5;
+ cdf = WebRtcIsacfix_kPitchLagPtrHi;
+ cdf_size = WebRtcIsacfix_kPitchLagSizeHi;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi;
+ lower_limit = WebRtcIsacfix_kLowerLimitHi;
+ init_index = WebRtcIsacfix_kInitIndHi;
+ }
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsacfix_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1);
+ if ((err<0) || (index[0]<0)) // error check
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
+
+ err = WebRtcIsacfix_DecHistOneStepMulti(index+1, streamdata, cdf+1, init_index, 3);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
+
+
+ /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
+ CQ11 = ((WebRtc_Word32)index[0] + lower_limit[0]); // Q0
+ CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32a = WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11);
+ tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);
+ PitchLags_Q7[k] = tmp16a;
+ }
+
+ CQ10 = mean_val2Q10[index[1]];
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
+ tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
+ PitchLags_Q7[k] += tmp16c;
+ }
+
+ CQ10 = mean_val4Q10[index[3]];
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
+ tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
+ PitchLags_Q7[k] += tmp16c;
+ }
+
+ return 0;
+}
+
+
+
+/* quantize & code Pitch Lags */
+int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagsQ7,WebRtc_Word16 *PitchGain_Q12,
+ Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
+{
+ int k, j;
+ WebRtc_Word16 index[PITCH_SUBFRAMES];
+ WebRtc_Word32 meangainQ12, CQ17;
+ WebRtc_Word32 CQ11, CQ10,tmp32a;
+
+ const WebRtc_Word16 *mean_val2Q10,*mean_val4Q10;
+ const WebRtc_Word16 *lower_limit, *upper_limit;
+ const WebRtc_UWord16 **cdf;
+ WebRtc_Word16 shft, tmp16a, tmp16b, tmp16c;
+ WebRtc_Word32 tmp32b;
+ int status = 0;
+
+ /* compute mean pitch gain */
+ meangainQ12=0;
+ for (k = 0; k < 4; k++)
+ meangainQ12 += PitchGain_Q12[k];
+
+ meangainQ12 = WEBRTC_SPL_RSHIFT_W32(meangainQ12, 2);
+
+ /* Save data for creation of multiple bitstreams */
+ if (encData != NULL) {
+ encData->meanGain[encData->startIdx] = meangainQ12;
+ }
+
+ /* voicing classificiation */
+ if (meangainQ12 <= 819) { // mean_gain < 0.2
+ shft = -1; // StepSize=2.0;
+ cdf = WebRtcIsacfix_kPitchLagPtrLo;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo;
+ lower_limit = WebRtcIsacfix_kLowerLimitLo;
+ upper_limit = WebRtcIsacfix_kUpperLimitLo;
+ } else if (meangainQ12 <= 1638) { // mean_gain < 0.4
+ shft = 0; // StepSize=1.0;
+ cdf = WebRtcIsacfix_kPitchLagPtrMid;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid;
+ lower_limit = WebRtcIsacfix_kLowerLimitMid;
+ upper_limit = WebRtcIsacfix_kUpperLimitMid;
+ } else {
+ shft = 1; // StepSize=0.5;
+ cdf = WebRtcIsacfix_kPitchLagPtrHi;
+ mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi;
+ mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi;
+ lower_limit = WebRtcIsacfix_kLowerLimitHi;
+ upper_limit = WebRtcIsacfix_kUpperLimitHi;
+ }
+
+ /* find quantization index */
+ for (k=0; k<4; k++)
+ {
+ /* transform */
+ CQ17=0;
+ for (j=0; j<PITCH_SUBFRAMES; j++)
+ CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], PitchLagsQ7[j],2); // Q17
+
+ CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize
+
+ /* quantize */
+ tmp16b = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(CQ17 + 65536, 17 );
+ index[k] = tmp16b;
+
+ /* check that the index is not outside the boundaries of the table */
+ if (index[k] < lower_limit[k]) index[k] = lower_limit[k];
+ else if (index[k] > upper_limit[k]) index[k] = upper_limit[k];
+ index[k] -= lower_limit[k];
+
+ /* Save data for creation of multiple bitstreams */
+ if(encData != NULL) {
+ encData->pitchIndex[PITCH_SUBFRAMES*encData->startIdx + k] = index[k];
+ }
+ }
+
+ /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
+ CQ11 = (index[0] + lower_limit[0]); // Q0
+ CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
+
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32a = WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11); // Q12
+ tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);// Q7
+ PitchLagsQ7[k] = tmp16a;
+ }
+
+ CQ10 = mean_val2Q10[index[1]];
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
+ tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
+ PitchLagsQ7[k] += tmp16c;
+ }
+
+ CQ10 = mean_val4Q10[index[3]];
+ for (k=0; k<PITCH_SUBFRAMES; k++) {
+ tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
+ tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
+ PitchLagsQ7[k] += tmp16c;
+ }
+
+ /* entropy coding of quantization pitch lags */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES);
+
+ /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
+ return status;
+}
+
+
+
+/* Routines for inband signaling of bandwitdh estimation */
+/* Histograms based on uniform distribution of indices */
+/* Move global variables later! */
+
+
+/* cdf array for frame length indicator */
+const WebRtc_UWord16 kFrameLenCdf[4] = {
+ 0, 21845, 43690, 65535};
+
+/* pointer to cdf array for frame length indicator */
+const WebRtc_UWord16 *kFrameLenCdfPtr[1] = {kFrameLenCdf};
+
+/* initial cdf index for decoder of frame length indicator */
+const WebRtc_UWord16 kFrameLenInitIndex[1] = {1};
+
+
+int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
+ WebRtc_Word16 *framesamples)
+{
+
+ int err;
+ WebRtc_Word16 frame_mode;
+
+ err = 0;
+ /* entropy decoding of frame length [1:30ms,2:60ms] */
+ err = WebRtcIsacfix_DecHistOneStepMulti(&frame_mode, streamdata, kFrameLenCdfPtr, kFrameLenInitIndex, 1);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH;
+
+ switch(frame_mode) {
+ case 1:
+ *framesamples = 480; /* 30ms */
+ break;
+ case 2:
+ *framesamples = 960; /* 60ms */
+ break;
+ default:
+ err = -ISAC_DISALLOWED_FRAME_MODE_DECODER;
+ }
+
+ return err;
+}
+
+
+int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr_enc *streamdata) {
+
+ int status;
+ WebRtc_Word16 frame_mode;
+
+ status = 0;
+ frame_mode = 0;
+ /* entropy coding of frame length [1:480 samples,2:960 samples] */
+ switch(framesamples) {
+ case 480:
+ frame_mode = 1;
+ break;
+ case 960:
+ frame_mode = 2;
+ break;
+ default:
+ status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER;
+ }
+
+ if (status < 0)
+ return status;
+
+ status = WebRtcIsacfix_EncHistMulti(streamdata, &frame_mode, kFrameLenCdfPtr, 1);
+
+ return status;
+}
+
+/* cdf array for estimated bandwidth */
+const WebRtc_UWord16 kBwCdf[25] = {
+ 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
+ 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
+ 62804, 65535};
+
+/* pointer to cdf array for estimated bandwidth */
+const WebRtc_UWord16 *kBwCdfPtr[1] = {kBwCdf};
+
+/* initial cdf index for decoder of estimated bandwidth*/
+const WebRtc_UWord16 kBwInitIndex[1] = {7};
+
+
+int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata, WebRtc_Word16 *BWno) {
+
+ int err;
+ WebRtc_Word16 BWno32;
+
+ /* entropy decoding of sender's BW estimation [0..23] */
+ err = WebRtcIsacfix_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
+ *BWno = (WebRtc_Word16)BWno32;
+ return err;
+
+}
+
+
+int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno, Bitstr_enc *streamdata)
+{
+ int status = 0;
+ /* entropy encoding of receiver's BW estimation [0..23] */
+ status = WebRtcIsacfix_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1);
+
+ return status;
+}
+
+/* estimate codel length of LPC Coef */
+void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *index_gQQ) {
+ int j, k, n;
+ WebRtc_Word16 posQQ, pos2QQ;
+ WebRtc_Word16 pos, pos2, posg, offsg, offs2, gainpos;
+ WebRtc_Word32 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
+ WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
+ WebRtc_Word32 sumQQ;
+
+
+ /* log gains, mean removal and scaling */
+ posg = 0;pos=0; gainpos=0;
+
+ for (k=0; k<SUBFRAMES; k++) {
+ /* log gains */
+
+ /* The input argument X to logN(X) is 2^17 times higher than the
+ input floating point argument Y to log(Y), since the X value
+ is a Q17 value. This can be compensated for after the call, by
+ subraction a value Z for each Q-step. One Q-step means that
+ X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
+ 177.445678 should be subtracted (since logN() returns a Q8 value).
+ For a X value in Q17, the value 177.445678*17 = 3017 should be
+ subtracted */
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+
+ tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
+ tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
+ posg++; gainpos++;
+
+ }
+
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<2; n++) {
+ sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
+ pos++;
+ pos2 += 2;
+ }
+ tmpcoeffs2_gQ21[posg] = sumQQ;
+ posg++;
+ }
+
+ offsg += 2;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<2; k++) {
+ sumQQ = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
+ pos += 2;
+ pos2++;
+ }
+ tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
+ posg++;
+ }
+ offsg += 2;
+ offs2 += SUBFRAMES;
+ }
+
+ /* quantize coefficients */
+ for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
+ {
+ posQQ = WebRtcIsacfix_kSelIndGain[k];
+ pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
+
+ index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
+ if (index_gQQ[k] < 0) {
+ index_gQQ[k] = 0;
+ }
+ else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
+ index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h
new file mode 100644
index 0000000..298ea22
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * entropy_coding.h
+ *
+ * This header file contains all of the functions used to arithmetically
+ * encode the iSAC bistream
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
+
+#include "structs.h"
+
+/* decode complex spectrum (return number of bytes in stream) */
+WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
+ WebRtc_Word16 *frQ7,
+ WebRtc_Word16 *fiQ7,
+ WebRtc_Word16 AvgPitchGain_Q12);
+
+/* encode complex spectrum */
+int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
+ const WebRtc_Word16 *fi,
+ Bitstr_enc *streamdata,
+ WebRtc_Word16 AvgPitchGain_Q12);
+
+
+/* decode & dequantize LPC Coef */
+int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
+ WebRtc_Word32 *LPCCoefQ17,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *outmodel);
+
+int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *LPCCoef_loQ15,
+ WebRtc_Word16 *LPCCoef_hiQ15,
+ Bitstr_dec *streamdata,
+ WebRtc_Word16 *outmodel);
+
+/* quantize & code LPC Coef */
+int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *LPCCoef_loQ15,
+ WebRtc_Word16 *LPCCoef_hiQ15,
+ WebRtc_Word16 *model,
+ WebRtc_Word32 *sizeQ11,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData,
+ transcode_obj *transcodeParam);
+
+int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData);
+/* decode & dequantize RC */
+int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata,
+ WebRtc_Word16 *RCQ15);
+
+/* quantize & code RC */
+int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15,
+ Bitstr_enc *streamdata);
+
+/* decode & dequantize squared Gain */
+int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata,
+ WebRtc_Word32 *Gain2);
+
+/* quantize & code squared Gain (input is squared gain) */
+int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gain2,
+ Bitstr_enc *streamdata);
+
+int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData);
+
+int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagQ7,
+ WebRtc_Word16 *PitchGain_Q12,
+ Bitstr_enc *streamdata,
+ ISAC_SaveEncData_t* encData);
+
+int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
+ WebRtc_Word16 *PitchGain_Q12);
+
+int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
+ WebRtc_Word16 *PitchGain_Q12,
+ WebRtc_Word16 *PitchLagQ7);
+
+int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
+ WebRtc_Word16 *framelength);
+
+
+int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framelength,
+ Bitstr_enc *streamdata);
+
+int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata,
+ WebRtc_Word16 *BWno);
+
+
+int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno,
+ Bitstr_enc *streamdata);
+
+void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *tmpcoeffs_gQ6,
+ WebRtc_Word16 *index_gQQ);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c
new file mode 100644
index 0000000..fff35c4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c
@@ -0,0 +1,415 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * fft.c
+ *
+ * Fast Fourier Transform
+ *
+ */
+
+
+#include "fft.h"
+
+const WebRtc_Word16 kSortTabFft[240] = {
+ 0, 60, 120, 180, 20, 80, 140, 200, 40, 100, 160, 220,
+ 4, 64, 124, 184, 24, 84, 144, 204, 44, 104, 164, 224,
+ 8, 68, 128, 188, 28, 88, 148, 208, 48, 108, 168, 228,
+ 12, 72, 132, 192, 32, 92, 152, 212, 52, 112, 172, 232,
+ 16, 76, 136, 196, 36, 96, 156, 216, 56, 116, 176, 236,
+ 1, 61, 121, 181, 21, 81, 141, 201, 41, 101, 161, 221,
+ 5, 65, 125, 185, 25, 85, 145, 205, 45, 105, 165, 225,
+ 9, 69, 129, 189, 29, 89, 149, 209, 49, 109, 169, 229,
+ 13, 73, 133, 193, 33, 93, 153, 213, 53, 113, 173, 233,
+ 17, 77, 137, 197, 37, 97, 157, 217, 57, 117, 177, 237,
+ 2, 62, 122, 182, 22, 82, 142, 202, 42, 102, 162, 222,
+ 6, 66, 126, 186, 26, 86, 146, 206, 46, 106, 166, 226,
+ 10, 70, 130, 190, 30, 90, 150, 210, 50, 110, 170, 230,
+ 14, 74, 134, 194, 34, 94, 154, 214, 54, 114, 174, 234,
+ 18, 78, 138, 198, 38, 98, 158, 218, 58, 118, 178, 238,
+ 3, 63, 123, 183, 23, 83, 143, 203, 43, 103, 163, 223,
+ 7, 67, 127, 187, 27, 87, 147, 207, 47, 107, 167, 227,
+ 11, 71, 131, 191, 31, 91, 151, 211, 51, 111, 171, 231,
+ 15, 75, 135, 195, 35, 95, 155, 215, 55, 115, 175, 235,
+ 19, 79, 139, 199, 39, 99, 159, 219, 59, 119, 179, 239
+};
+
+/* Cosine table in Q14 */
+const WebRtc_Word16 kCosTabFfftQ14[240] = {
+ 16384, 16378, 16362, 16333, 16294, 16244, 16182, 16110, 16026, 15931, 15826, 15709,
+ 15582, 15444, 15296, 15137, 14968, 14788, 14598, 14399, 14189, 13970, 13741, 13502,
+ 13255, 12998, 12733, 12458, 12176, 11885, 11585, 11278, 10963, 10641, 10311, 9974,
+ 9630, 9280, 8923, 8561, 8192, 7818, 7438, 7053, 6664, 6270, 5872, 5469,
+ 5063, 4653, 4240, 3825, 3406, 2986, 2563, 2139, 1713, 1285, 857, 429,
+ 0, -429, -857, -1285, -1713, -2139, -2563, -2986, -3406, -3825, -4240, -4653,
+ -5063, -5469, -5872, -6270, -6664, -7053, -7438, -7818, -8192, -8561, -8923, -9280,
+ -9630, -9974, -10311, -10641, -10963, -11278, -11585, -11885, -12176, -12458, -12733, -12998,
+ -13255, -13502, -13741, -13970, -14189, -14399, -14598, -14788, -14968, -15137, -15296, -15444,
+ -15582, -15709, -15826, -15931, -16026, -16110, -16182, -16244, -16294, -16333, -16362, -16378,
+ -16384, -16378, -16362, -16333, -16294, -16244, -16182, -16110, -16026, -15931, -15826, -15709,
+ -15582, -15444, -15296, -15137, -14968, -14788, -14598, -14399, -14189, -13970, -13741, -13502,
+ -13255, -12998, -12733, -12458, -12176, -11885, -11585, -11278, -10963, -10641, -10311, -9974,
+ -9630, -9280, -8923, -8561, -8192, -7818, -7438, -7053, -6664, -6270, -5872, -5469,
+ -5063, -4653, -4240, -3825, -3406, -2986, -2563, -2139, -1713, -1285, -857, -429,
+ 0, 429, 857, 1285, 1713, 2139, 2563, 2986, 3406, 3825, 4240, 4653,
+ 5063, 5469, 5872, 6270, 6664, 7053, 7438, 7818, 8192, 8561, 8923, 9280,
+ 9630, 9974, 10311, 10641, 10963, 11278, 11585, 11885, 12176, 12458, 12733, 12998,
+ 13255, 13502, 13741, 13970, 14189, 14399, 14598, 14788, 14968, 15137, 15296, 15444,
+ 15582, 15709, 15826, 15931, 16026, 16110, 16182, 16244, 16294, 16333, 16362, 16378
+};
+
+
+
+/* Uses 16x16 mul, without rounding, which is faster. Uses WEBRTC_SPL_MUL_16_16_RSFT */
+WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign) {
+
+ WebRtc_Word16 dd, ee, ff, gg, hh, ii;
+ WebRtc_Word16 k0, k1, k2, k3, k4, kk;
+ WebRtc_Word16 tmp116, tmp216;
+
+ WebRtc_Word16 ccc1Q14, ccc2Q14, ccc3Q14, sss1Q14, sss2Q14, sss3Q14;
+ WebRtc_Word16 sss60Q14, ccc72Q14, sss72Q14;
+ WebRtc_Word16 aaQx, ajQx, akQx, ajmQx, ajpQx, akmQx, akpQx;
+ WebRtc_Word16 bbQx, bjQx, bkQx, bjmQx, bjpQx, bkmQx, bkpQx;
+
+ WebRtc_Word16 ReDATAQx[240], ImDATAQx[240];
+
+ sss60Q14 = kCosTabFfftQ14[20];
+ ccc72Q14 = kCosTabFfftQ14[48];
+ sss72Q14 = kCosTabFfftQ14[12];
+
+ if (iSign < 0) {
+ sss72Q14 = -sss72Q14;
+ sss60Q14 = -sss60Q14;
+ }
+ /* Complexity is: 10 cycles */
+
+ /* compute fourier transform */
+
+ // transform for factor of 4
+ for (kk=0; kk<60; kk++) {
+ k0 = kk;
+ k1 = k0 + 60;
+ k2 = k1 + 60;
+ k3 = k2 + 60;
+
+ akpQx = RexQx[k0] + RexQx[k2];
+ akmQx = RexQx[k0] - RexQx[k2];
+ ajpQx = RexQx[k1] + RexQx[k3];
+ ajmQx = RexQx[k1] - RexQx[k3];
+ bkpQx = ImxQx[k0] + ImxQx[k2];
+ bkmQx = ImxQx[k0] - ImxQx[k2];
+ bjpQx = ImxQx[k1] + ImxQx[k3];
+ bjmQx = ImxQx[k1] - ImxQx[k3];
+
+ RexQx[k0] = akpQx + ajpQx;
+ ImxQx[k0] = bkpQx + bjpQx;
+ ajpQx = akpQx - ajpQx;
+ bjpQx = bkpQx - bjpQx;
+ if (iSign < 0) {
+ akpQx = akmQx + bjmQx;
+ bkpQx = bkmQx - ajmQx;
+ akmQx -= bjmQx;
+ bkmQx += ajmQx;
+ } else {
+ akpQx = akmQx - bjmQx;
+ bkpQx = bkmQx + ajmQx;
+ akmQx += bjmQx;
+ bkmQx -= ajmQx;
+ }
+
+ ccc1Q14 = kCosTabFfftQ14[kk];
+ ccc2Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(2, kk)];
+ ccc3Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(3, kk)];
+ sss1Q14 = kCosTabFfftQ14[kk+60];
+ sss2Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(2, kk)+60];
+ sss3Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(3, kk)+60];
+ if (iSign==1) {
+ sss1Q14 = -sss1Q14;
+ sss2Q14 = -sss2Q14;
+ sss3Q14 = -sss3Q14;
+ }
+
+ //Do several multiplications like Q14*Q16>>14 = Q16
+ // RexQ16[k1] = akpQ16 * ccc1Q14 - bkpQ16 * sss1Q14;
+ // RexQ16[k2] = ajpQ16 * ccc2Q14 - bjpQ16 * sss2Q14;
+ // RexQ16[k3] = akmQ16 * ccc3Q14 - bkmQ16 * sss3Q14;
+ // ImxQ16[k1] = akpQ16 * sss1Q14 + bkpQ16 * ccc1Q14;
+ // ImxQ16[k2] = ajpQ16 * sss2Q14 + bjpQ16 * ccc2Q14;
+ // ImxQ16[k3] = akmQ16 * sss3Q14 + bkmQ16 * ccc3Q14;
+
+ RexQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul)
+ RexQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14);
+ RexQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14);
+ ImxQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14);
+ ImxQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14);
+ ImxQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, akmQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, bkmQx, 14);
+ //This mul segment needs 6*8 = 48 cycles for 16x16 muls, but 6*20 = 120 cycles for 16x32 muls
+
+
+ }
+ /* Complexity is: 51+48 = 99 cycles for 16x16 muls, but 51+120 = 171 cycles for 16x32 muls*/
+
+ // transform for factor of 3
+ kk=0;
+ k1=20;
+ k2=40;
+
+ for (hh=0; hh<4; hh++) {
+ for (ii=0; ii<20; ii++) {
+ akQx = RexQx[kk];
+ bkQx = ImxQx[kk];
+ ajQx = RexQx[k1] + RexQx[k2];
+ bjQx = ImxQx[k1] + ImxQx[k2];
+ RexQx[kk] = akQx + ajQx;
+ ImxQx[kk] = bkQx + bjQx;
+ tmp116 = WEBRTC_SPL_RSHIFT_W16(ajQx, 1);
+ tmp216 = WEBRTC_SPL_RSHIFT_W16(bjQx, 1);
+ akQx = akQx - tmp116;
+ bkQx = bkQx - tmp216;
+ tmp116 = RexQx[k1] - RexQx[k2];
+ tmp216 = ImxQx[k1] - ImxQx[k2];
+
+ ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp116, 14); // Q14*Qx>>14 = Qx
+ bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp216, 14); // Q14*Qx>>14 = Qx
+ RexQx[k1] = akQx - bjQx;
+ RexQx[k2] = akQx + bjQx;
+ ImxQx[k1] = bkQx + ajQx;
+ ImxQx[k2] = bkQx - ajQx;
+
+ kk++;
+ k1++;
+ k2++;
+ }
+ /* Complexity : (31+6)*20 = 740 cycles for 16x16 muls, but (31+18)*20 = 980 cycles for 16x32 muls*/
+ kk=kk+40;
+ k1=k1+40;
+ k2=k2+40;
+ }
+ /* Complexity : 4*(740+3) = 2972 cycles for 16x16 muls, but 4*(980+3) = 3932 cycles for 16x32 muls*/
+
+ /* multiply by rotation factor for odd factor 3 or 5 (not for 4)
+ Same code (duplicated) for both ii=2 and ii=3 */
+ kk = 1;
+ ee = 0;
+ ff = 0;
+
+ for (gg=0; gg<19; gg++) {
+ kk += 20;
+ ff = ff+4;
+ for (hh=0; hh<2; hh++) {
+ ee = ff + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(hh, ff);
+ dd = ee + 60;
+ ccc2Q14 = kCosTabFfftQ14[ee];
+ sss2Q14 = kCosTabFfftQ14[dd];
+ if (iSign==1) {
+ sss2Q14 = -sss2Q14;
+ }
+ for (ii=0; ii<4; ii++) {
+ akQx = RexQx[kk];
+ bkQx = ImxQx[kk];
+ RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - // Q14*Qx>>14 = Qx
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
+ ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + // Q14*Qx>>14 = Qx
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
+
+
+ kk += 60;
+ }
+ kk = kk - 220;
+ }
+ // Complexity: 2*(13+5+4*13+2) = 144 for 16x16 muls, but 2*(13+5+4*33+2) = 304 cycles for 16x32 muls
+ kk = kk - 59;
+ }
+ // Complexity: 19*144 = 2736 for 16x16 muls, but 19*304 = 5776 cycles for 16x32 muls
+
+ // transform for factor of 5
+ kk = 0;
+ ccc2Q14 = kCosTabFfftQ14[96];
+ sss2Q14 = kCosTabFfftQ14[84];
+ if (iSign==1) {
+ sss2Q14 = -sss2Q14;
+ }
+
+ for (hh=0; hh<4; hh++) {
+ for (ii=0; ii<12; ii++) {
+ k1 = kk + 4;
+ k2 = k1 + 4;
+ k3 = k2 + 4;
+ k4 = k3 + 4;
+
+ akpQx = RexQx[k1] + RexQx[k4];
+ akmQx = RexQx[k1] - RexQx[k4];
+ bkpQx = ImxQx[k1] + ImxQx[k4];
+ bkmQx = ImxQx[k1] - ImxQx[k4];
+ ajpQx = RexQx[k2] + RexQx[k3];
+ ajmQx = RexQx[k2] - RexQx[k3];
+ bjpQx = ImxQx[k2] + ImxQx[k3];
+ bjmQx = ImxQx[k2] - ImxQx[k3];
+ aaQx = RexQx[kk];
+ bbQx = ImxQx[kk];
+ RexQx[kk] = aaQx + akpQx + ajpQx;
+ ImxQx[kk] = bbQx + bkpQx + bjpQx;
+
+ akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, akpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) + aaQx;
+ bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bkpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14) + bbQx;
+ ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, akmQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajmQx, 14);
+ bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bkmQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjmQx, 14);
+ // 32+4*8=64 or 32+4*20=112
+
+ RexQx[k1] = akQx - bjQx;
+ RexQx[k4] = akQx + bjQx;
+ ImxQx[k1] = bkQx + ajQx;
+ ImxQx[k4] = bkQx - ajQx;
+
+ akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, ajpQx, 14) + aaQx;
+ bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkpQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bjpQx, 14) + bbQx;
+ ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akmQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, ajmQx, 14);
+ bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkmQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bjmQx, 14);
+ // 8+4*8=40 or 8+4*20=88
+
+ RexQx[k2] = akQx - bjQx;
+ RexQx[k3] = akQx + bjQx;
+ ImxQx[k2] = bkQx + ajQx;
+ ImxQx[k3] = bkQx - ajQx;
+
+ kk = k4 + 4;
+ }
+ // Complexity: 12*(64+40+10) = 1368 for 16x16 muls, but 12*(112+88+10) = 2520 cycles for 16x32 muls
+ kk -= 239;
+ }
+ // Complexity: 4*1368 = 5472 for 16x16 muls, but 4*2520 = 10080 cycles for 16x32 muls
+
+ /* multiply by rotation factor for odd factor 3 or 5 (not for 4)
+ Same code (duplicated) for both ii=2 and ii=3 */
+ kk = 1;
+ ee=0;
+
+ for (gg=0; gg<3; gg++) {
+ kk += 4;
+ dd = 12 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(12, gg);
+ ff = 0;
+ for (hh=0; hh<4; hh++) {
+ ff = ff+dd;
+ ee = ff+60;
+ for (ii=0; ii<12; ii++) {
+ akQx = RexQx[kk];
+ bkQx = ImxQx[kk];
+
+ ccc2Q14 = kCosTabFfftQ14[ff];
+ sss2Q14 = kCosTabFfftQ14[ee];
+
+ if (iSign==1) {
+ sss2Q14 = -sss2Q14;
+ }
+
+ RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) -
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
+ ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
+
+ kk += 20;
+ }
+ kk = kk - 236;
+ // Complexity: 12*(12+12) = 288 for 16x16 muls, but 12*(12+32) = 528 cycles for 16x32 muls
+ }
+ kk = kk - 19;
+ // Complexity: 4*288+6 for 16x16 muls, but 4*528+6 cycles for 16x32 muls
+ }
+ // Complexity: 3*4*288+6 = 3462 for 16x16 muls, but 3*4*528+6 = 6342 cycles for 16x32 muls
+
+
+ // last transform for factor of 4 */
+ for (kk=0; kk<240; kk=kk+4) {
+ k1 = kk + 1;
+ k2 = k1 + 1;
+ k3 = k2 + 1;
+
+ akpQx = RexQx[kk] + RexQx[k2];
+ akmQx = RexQx[kk] - RexQx[k2];
+ ajpQx = RexQx[k1] + RexQx[k3];
+ ajmQx = RexQx[k1] - RexQx[k3];
+ bkpQx = ImxQx[kk] + ImxQx[k2];
+ bkmQx = ImxQx[kk] - ImxQx[k2];
+ bjpQx = ImxQx[k1] + ImxQx[k3];
+ bjmQx = ImxQx[k1] - ImxQx[k3];
+ RexQx[kk] = akpQx + ajpQx;
+ ImxQx[kk] = bkpQx + bjpQx;
+ ajpQx = akpQx - ajpQx;
+ bjpQx = bkpQx - bjpQx;
+ if (iSign < 0) {
+ akpQx = akmQx + bjmQx;
+ bkpQx = bkmQx - ajmQx;
+ akmQx -= bjmQx;
+ bkmQx += ajmQx;
+ } else {
+ akpQx = akmQx - bjmQx;
+ bkpQx = bkmQx + ajmQx;
+ akmQx += bjmQx;
+ bkmQx -= ajmQx;
+ }
+ RexQx[k1] = akpQx;
+ RexQx[k2] = ajpQx;
+ RexQx[k3] = akmQx;
+ ImxQx[k1] = bkpQx;
+ ImxQx[k2] = bjpQx;
+ ImxQx[k3] = bkmQx;
+ }
+ // Complexity: 60*45 = 2700 for 16x16 muls, but 60*45 = 2700 cycles for 16x32 muls
+
+ /* permute the results to normal order */
+ for (ii=0; ii<240; ii++) {
+ ReDATAQx[ii]=RexQx[ii];
+ ImDATAQx[ii]=ImxQx[ii];
+ }
+ // Complexity: 240*2=480 cycles
+
+ for (ii=0; ii<240; ii++) {
+ RexQx[ii]=ReDATAQx[kSortTabFft[ii]];
+ ImxQx[ii]=ImDATAQx[kSortTabFft[ii]];
+ }
+ // Complexity: 240*2*2=960 cycles
+
+ // Total complexity:
+ // 16x16 16x32
+ // Complexity: 10 10
+ // Complexity: 99 171
+ // Complexity: 2972 3932
+ // Complexity: 2736 5776
+ // Complexity: 5472 10080
+ // Complexity: 3462 6342
+ // Complexity: 2700 2700
+ // Complexity: 480 480
+ // Complexity: 960 960
+ // =======================
+ // 18891 30451
+ //
+ // If this FFT is called 2 time each frame, i.e. 67 times per second, it will correspond to
+ // a C54 complexity of 67*18891/1000000 = 1.27 MIPS with 16x16-muls, and 67*30451/1000000 =
+ // = 2.04 MIPS with 16x32-muls. Note that this routine somtimes is called 6 times during the
+ // encoding of a frame, i.e. the max complexity would be 7/2*1.27 = 4.4 MIPS for the 16x16 mul case.
+
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h
new file mode 100644
index 0000000..efa116e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*--------------------------------*-C-*---------------------------------*
+ * File:
+ * fft.h
+ * ---------------------------------------------------------------------*
+ * Re[]: real value array
+ * Im[]: imaginary value array
+ * nTotal: total number of complex values
+ * nPass: number of elements involved in this pass of transform
+ * nSpan: nspan/nPass = number of bytes to increment pointer
+ * in Re[] and Im[]
+ * isign: exponent: +1 = forward -1 = reverse
+ * scaling: normalizing constant by which the final result is *divided*
+ * scaling == -1, normalize by total dimension of the transform
+ * scaling < -1, normalize by the square-root of the total dimension
+ *
+ * ----------------------------------------------------------------------
+ * See the comments in the code for correct usage!
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
+
+
+#include "structs.h"
+
+
+WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign);
+
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c
new file mode 100644
index 0000000..87c62aa
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filterbank_tables.c
+ *
+ * This file contains variables that are used in
+ * filterbanks.c
+ *
+ */
+
+#include "filterbank_tables.h"
+#include "settings.h"
+
+
+/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
+ * In float, they are:
+ * {-1.94895953203325f, 0.94984516000000f, -0.05101826139794f, 0.05015484000000f};
+ */
+const WebRtc_Word16 WebRtcIsacfix_kHpStCoeffInQ30[8] = {
+ -31932, 16189, /* Q30 hi/lo pair */
+ 15562, 17243, /* Q30 hi/lo pair */
+ -26748, -17186, /* Q35 hi/lo pair */
+ 26296, -27476 /* Q35 hi/lo pair */
+};
+
+/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
+ * In float, they are:
+ * {-1.99701049409000f, 0.99714204490000f, 0.01701049409000f, -0.01704204490000f};
+ */
+const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut1Q30[8] = {
+ -32719, -1306, /* Q30 hi/lo pair */
+ 16337, 11486, /* Q30 hi/lo pair */
+ 8918, 26078, /* Q35 hi/lo pair */
+ -8935, 3956 /* Q35 hi/lo pair */
+};
+
+/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
+ * In float, they are:
+ * {-1.98645294509837f, 0.98672435560000f, 0.00645294509837f, -0.00662435560000f};
+ */
+const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut2Q30[8] = {
+ -32546, -2953, /* Q30 hi/lo pair */
+ 16166, 32233, /* Q30 hi/lo pair */
+ 3383, 13217, /* Q35 hi/lo pair */
+ -3473, -4597 /* Q35 hi/lo pair */
+};
+
+/* The upper channel all-pass filter factors */
+const WebRtc_Word16 WebRtcIsacfix_kUpperApFactorsQ15[2] = {
+ 1137, 12537
+};
+
+/* The lower channel all-pass filter factors */
+const WebRtc_Word16 WebRtcIsacfix_kLowerApFactorsQ15[2] = {
+ 5059, 24379
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h
new file mode 100644
index 0000000..b6be4f0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filterbank_tables.h
+ *
+ * Header file for variables that are defined in
+ * filterbank_tables.c.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
+
+#include "typedefs.h"
+
+/********************* Coefficient Tables ************************/
+
+/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+extern const WebRtc_Word16 WebRtcIsacfix_kHpStCoeffInQ30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
+
+/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+extern const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut1Q30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
+
+/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+extern const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut2Q30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
+
+/* The upper channel all-pass filter factors */
+extern const WebRtc_Word16 WebRtcIsacfix_kUpperApFactorsQ15[2];
+
+/* The lower channel all-pass filter factors */
+extern const WebRtc_Word16 WebRtcIsacfix_kLowerApFactorsQ15[2];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c
new file mode 100644
index 0000000..a53aa66
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c
@@ -0,0 +1,326 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filterbanks.c
+ *
+ * This file contains function
+ * WebRtcIsacfix_SplitAndFilter, and WebRtcIsacfix_FilterAndCombine
+ * which implement filterbanks that produce decimated lowpass and
+ * highpass versions of a signal, and performs reconstruction.
+ *
+ */
+
+#include "codec.h"
+#include "filterbank_tables.h"
+#include "settings.h"
+
+
+static void AllpassFilter2FixDec16(WebRtc_Word16 *InOut16, //Q0
+ const WebRtc_Word16 *APSectionFactors, //Q15
+ WebRtc_Word16 lengthInOut,
+ WebRtc_Word16 NumberOfSections,
+ WebRtc_Word32 *FilterState) //Q16
+{
+ int n, j;
+ WebRtc_Word32 a, b;
+
+ for (j=0; j<NumberOfSections; j++) {
+ for (n=0;n<lengthInOut;n++) {
+
+
+ a = WEBRTC_SPL_MUL_16_16(APSectionFactors[j], InOut16[n]); //Q15*Q0=Q15
+ a = WEBRTC_SPL_LSHIFT_W32(a, 1); // Q15 -> Q16
+ b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16
+ a = WEBRTC_SPL_MUL_16_16_RSFT(-APSectionFactors[j], (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16), 0); //Q15*Q0=Q15
+ FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(WEBRTC_SPL_LSHIFT_W32(a,1), WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n],16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
+ InOut16[n] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
+
+ }
+ }
+
+}
+
+
+static void HighpassFilterFixDec32(
+ WebRtc_Word16 *io, /* Q0:input Q0: Output */
+ WebRtc_Word16 len, /* length of input, Input */
+ const WebRtc_Word16 *coeff, /* Coeff: [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
+ WebRtc_Word32 *state) /* Q4:filter state Input/Output */
+{
+ int k;
+ WebRtc_Word32 a, b, c, in;
+
+
+
+ for (k=0; k<len; k++) {
+ in = (WebRtc_Word32)io[k];
+ /* Q35 * Q4 = Q39 ; shift 32 bit => Q7 */
+ a = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*2], coeff[2*2+1], state[0]);
+ b = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*3], coeff[2*3+1], state[1]);
+
+ c = ((WebRtc_Word32)in) + WEBRTC_SPL_RSHIFT_W32(a+b, 7); // Q0
+ //c = WEBRTC_SPL_RSHIFT_W32(c, 1); // Q-1
+ io[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(c); // Write output as Q0
+
+ /* Q30 * Q4 = Q34 ; shift 32 bit => Q2 */
+ a = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*0], coeff[2*0+1], state[0]);
+ b = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*1], coeff[2*1+1], state[1]);
+
+ c = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in, 2) - a - b; // New state in Q2
+ c= (WebRtc_Word32)WEBRTC_SPL_SAT((WebRtc_Word32)536870911, c, (WebRtc_Word32)-536870912); // Check for wrap-around
+
+ state[1] = state[0];
+ state[0] = WEBRTC_SPL_LSHIFT_W32(c, 2); // Write state as Q4
+
+ }
+}
+
+
+void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16 *pin,
+ WebRtc_Word16 *LP16,
+ WebRtc_Word16 *HP16,
+ PreFiltBankstr *prefiltdata)
+{
+ /* Function WebRtcIsacfix_SplitAndFilter */
+ /* This function creates low-pass and high-pass decimated versions of part of
+ the input signal, and part of the signal in the input 'lookahead buffer'. */
+
+ int k;
+
+ WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2 + QLOOKAHEAD];
+ WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2 + QLOOKAHEAD];
+ WebRtc_Word32 tmpState[WEBRTC_SPL_MUL_16_16(2,(QORDER-1))]; /* 4 */
+
+
+ /* High pass filter */
+ HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix);
+
+
+ /* First Channel */
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ tempin_ch1[QLOOKAHEAD + k] = pin[1+WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+ for (k=0;k<QLOOKAHEAD;k++) {
+ tempin_ch1[k]=prefiltdata->INLABUF1_fix[k];
+ prefiltdata->INLABUF1_fix[k]=pin[FRAMESAMPLES+1-WEBRTC_SPL_MUL_16_16(2, QLOOKAHEAD)+WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+
+ /* Second Channel. This is exactly like the first channel, except that the
+ even samples are now filtered instead (lower channel). */
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ tempin_ch2[QLOOKAHEAD+k] = pin[WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+ for (k=0;k<QLOOKAHEAD;k++) {
+ tempin_ch2[k]=prefiltdata->INLABUF2_fix[k];
+ prefiltdata->INLABUF2_fix[k]=pin[FRAMESAMPLES-WEBRTC_SPL_MUL_16_16(2, QLOOKAHEAD)+WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+
+
+ /*obtain polyphase components by forward all-pass filtering through each channel */
+ /* The all pass filtering automatically updates the filter states which are exported in the
+ prefiltdata structure */
+ AllpassFilter2FixDec16(tempin_ch1,WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_fix);
+ AllpassFilter2FixDec16(tempin_ch2,WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_fix);
+
+ for (k=0;k<WEBRTC_SPL_MUL_16_16(2, (QORDER-1));k++)
+ tmpState[k] = prefiltdata->INSTAT1_fix[k];
+ AllpassFilter2FixDec16(tempin_ch1 + FRAMESAMPLES/2,WebRtcIsacfix_kUpperApFactorsQ15, QLOOKAHEAD , NUMBEROFCHANNELAPSECTIONS, tmpState);
+ for (k=0;k<WEBRTC_SPL_MUL_16_16(2, (QORDER-1));k++)
+ tmpState[k] = prefiltdata->INSTAT2_fix[k];
+ AllpassFilter2FixDec16(tempin_ch2 + FRAMESAMPLES/2,WebRtcIsacfix_kLowerApFactorsQ15, QLOOKAHEAD , NUMBEROFCHANNELAPSECTIONS, tmpState);
+
+
+ /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
+ for (k=0; k<FRAMESAMPLES/2 + QLOOKAHEAD; k++) {
+ WebRtc_Word32 tmp1, tmp2, tmp3;
+ tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
+ tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
+ tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
+ LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
+ tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
+ HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
+ }
+
+}/*end of WebRtcIsacfix_SplitAndFilter */
+
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+
+/* Without lookahead */
+void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16 *pin,
+ WebRtc_Word16 *LP16,
+ WebRtc_Word16 *HP16,
+ PreFiltBankstr *prefiltdata)
+{
+ /* Function WebRtcIsacfix_SplitAndFilter2 */
+ /* This function creates low-pass and high-pass decimated versions of part of
+ the input signal. */
+
+ int k;
+
+ WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2];
+ WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2];
+
+
+ /* High pass filter */
+ HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix);
+
+
+ /* First Channel */
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ tempin_ch1[k] = pin[1+WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+
+ /* Second Channel. This is exactly like the first channel, except that the
+ even samples are now filtered instead (lower channel). */
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ tempin_ch2[k] = pin[WEBRTC_SPL_MUL_16_16(2, k)];
+ }
+
+
+ /*obtain polyphase components by forward all-pass filtering through each channel */
+ /* The all pass filtering automatically updates the filter states which are exported in the
+ prefiltdata structure */
+ AllpassFilter2FixDec16(tempin_ch1,WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_fix);
+ AllpassFilter2FixDec16(tempin_ch2,WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_fix);
+
+
+ /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ WebRtc_Word32 tmp1, tmp2, tmp3;
+ tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
+ tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
+ tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
+ LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
+ tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
+ HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
+ }
+
+}/*end of WebRtcIsacfix_SplitAndFilter */
+
+#endif
+
+
+
+//////////////////////////////////////////////////////////
+////////// Combining
+/* Function WebRtcIsacfix_FilterAndCombine */
+/* This is a decoder function that takes the decimated
+ length FRAMESAMPLES/2 input low-pass and
+ high-pass signals and creates a reconstructed fullband
+ output signal of length FRAMESAMPLES. WebRtcIsacfix_FilterAndCombine
+ is the sibling function of WebRtcIsacfix_SplitAndFilter */
+/* INPUTS:
+ inLP: a length FRAMESAMPLES/2 array of input low-pass
+ samples.
+ inHP: a length FRAMESAMPLES/2 array of input high-pass
+ samples.
+ postfiltdata: input data structure containing the filterbank
+ states from the previous decoding iteration.
+ OUTPUTS:
+ Out: a length FRAMESAMPLES array of output reconstructed
+ samples (fullband) based on the input low-pass and
+ high-pass signals.
+ postfiltdata: the input data structure containing the filterbank
+ states is updated for the next decoding iteration */
+void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16 *tempin_ch1,
+ WebRtc_Word16 *tempin_ch2,
+ WebRtc_Word16 *out16,
+ PostFiltBankstr *postfiltdata)
+{
+ int k;
+ WebRtc_Word16 in[FRAMESAMPLES];
+
+ /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
+ that were used as a lower channel at the encoding side. So at the decoder, the
+ corresponding all-pass filter factors for each channel are swapped.*/
+
+ AllpassFilter2FixDec16(tempin_ch1, WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_fix);
+
+ /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
+ at the decoder are swapped from the ones at the encoder, the 'upper' channel
+ all-pass filter factors (kUpperApFactors) are used to filter this new lower channel signal */
+
+ AllpassFilter2FixDec16(tempin_ch2, WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_fix);
+
+ /* Merge outputs to form the full length output signal.*/
+ for (k=0;k<FRAMESAMPLES/2;k++) {
+ in[WEBRTC_SPL_MUL_16_16(2, k)]=tempin_ch2[k];
+ in[WEBRTC_SPL_MUL_16_16(2, k)+1]=tempin_ch1[k];
+ }
+
+ /* High pass filter */
+ HighpassFilterFixDec32(in, FRAMESAMPLES, WebRtcIsacfix_kHPStCoeffOut1Q30, postfiltdata->HPstates1_fix);
+ HighpassFilterFixDec32(in, FRAMESAMPLES, WebRtcIsacfix_kHPStCoeffOut2Q30, postfiltdata->HPstates2_fix);
+
+ for (k=0;k<FRAMESAMPLES;k++) {
+ out16[k] = in[k];
+ }
+}
+
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+/* Function WebRtcIsacfix_FilterAndCombine */
+/* This is a decoder function that takes the decimated
+ length len/2 input low-pass and
+ high-pass signals and creates a reconstructed fullband
+ output signal of length len. WebRtcIsacfix_FilterAndCombine
+ is the sibling function of WebRtcIsacfix_SplitAndFilter */
+/* INPUTS:
+ inLP: a length len/2 array of input low-pass
+ samples.
+ inHP: a length len/2 array of input high-pass
+ samples.
+ postfiltdata: input data structure containing the filterbank
+ states from the previous decoding iteration.
+ OUTPUTS:
+ Out: a length len array of output reconstructed
+ samples (fullband) based on the input low-pass and
+ high-pass signals.
+ postfiltdata: the input data structure containing the filterbank
+ states is updated for the next decoding iteration */
+void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16 *tempin_ch1,
+ WebRtc_Word16 *tempin_ch2,
+ WebRtc_Word16 *out16,
+ PostFiltBankstr *postfiltdata,
+ WebRtc_Word16 len)
+{
+ int k;
+ WebRtc_Word16 in[FRAMESAMPLES];
+
+ /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
+ that were used as a lower channel at the encoding side. So at the decoder, the
+ corresponding all-pass filter factors for each channel are swapped.*/
+
+ AllpassFilter2FixDec16(tempin_ch1, WebRtcIsacfix_kLowerApFactorsQ15,(WebRtc_Word16) (len/2), NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_fix);
+
+ /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
+ at the decoder are swapped from the ones at the encoder, the 'upper' channel
+ all-pass filter factors (kUpperApFactors) are used to filter this new lower channel signal */
+
+ AllpassFilter2FixDec16(tempin_ch2, WebRtcIsacfix_kUpperApFactorsQ15, (WebRtc_Word16) (len/2), NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_fix);
+
+ /* Merge outputs to form the full length output signal.*/
+ for (k=0;k<len/2;k++) {
+ in[WEBRTC_SPL_MUL_16_16(2, k)]=tempin_ch2[k];
+ in[WEBRTC_SPL_MUL_16_16(2, k)+1]=tempin_ch1[k];
+ }
+
+ /* High pass filter */
+ HighpassFilterFixDec32(in, len, WebRtcIsacfix_kHPStCoeffOut1Q30, postfiltdata->HPstates1_fix);
+ HighpassFilterFixDec32(in, len, WebRtcIsacfix_kHPStCoeffOut2Q30, postfiltdata->HPstates2_fix);
+
+ for (k=0;k<len;k++) {
+ out16[k] = in[k];
+ }
+}
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c
new file mode 100644
index 0000000..6ee0477
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c
@@ -0,0 +1,122 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filters.c
+ *
+ * This file contains function WebRtcIsacfix_AutocorrC,
+ * AllpassFilterForDec32, and WebRtcIsacfix_DecimateAllpass32
+ *
+ */
+
+#include <string.h>
+
+#include "pitch_estimator.h"
+#include "lpc_masking_model.h"
+#include "codec.h"
+
+// Autocorrelation function in fixed point.
+// NOTE! Different from SPLIB-version in how it scales the signal.
+int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
+ const WebRtc_Word16* __restrict x,
+ WebRtc_Word16 N,
+ WebRtc_Word16 order,
+ WebRtc_Word16* __restrict scale) {
+ int i = 0;
+ int j = 0;
+ int16_t scaling = 0;
+ int32_t sum = 0;
+ uint32_t temp = 0;
+ int64_t prod = 0;
+
+ // Calculate r[0].
+ for (i = 0; i < N; i++) {
+ prod += WEBRTC_SPL_MUL_16_16(x[i], x[i]);
+ }
+
+ // Calculate scaling (the value of shifting).
+ temp = (uint32_t)(prod >> 31);
+ if(temp == 0) {
+ scaling = 0;
+ } else {
+ scaling = 32 - WebRtcSpl_NormU32(temp);
+ }
+ r[0] = (int32_t)(prod >> scaling);
+
+ // Perform the actual correlation calculation.
+ for (i = 1; i < order + 1; i++) {
+ prod = 0;
+ for (j = 0; j < N - i; j++) {
+ prod += WEBRTC_SPL_MUL_16_16(x[j], x[i + j]);
+ }
+ sum = (int32_t)(prod >> scaling);
+ r[i] = sum;
+ }
+
+ *scale = scaling;
+
+ return(order + 1);
+}
+
+static const WebRtc_Word32 kApUpperQ15[ALLPASSSECTIONS] = { 1137, 12537 };
+static const WebRtc_Word32 kApLowerQ15[ALLPASSSECTIONS] = { 5059, 24379 };
+
+
+static void AllpassFilterForDec32(WebRtc_Word16 *InOut16, //Q0
+ const WebRtc_Word32 *APSectionFactors, //Q15
+ WebRtc_Word16 lengthInOut,
+ WebRtc_Word32 *FilterState) //Q16
+{
+ int n, j;
+ WebRtc_Word32 a, b;
+
+ for (j=0; j<ALLPASSSECTIONS; j++) {
+ for (n=0;n<lengthInOut;n+=2){
+ a = WEBRTC_SPL_MUL_16_32_RSFT16(InOut16[n], APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
+ a = WEBRTC_SPL_LSHIFT_W32(a, 1); // Q15 -> Q16
+ b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16
+ a = WEBRTC_SPL_MUL_16_32_RSFT16(
+ (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16),
+ -APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
+ FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(
+ WEBRTC_SPL_LSHIFT_W32(a,1),
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n], 16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
+ InOut16[n] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
+ }
+ }
+}
+
+
+
+
+void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
+ WebRtc_Word32 *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
+ WebRtc_Word16 N, /* number of input samples */
+ WebRtc_Word16 *out) /* array of size N/2 */
+{
+ int n;
+ WebRtc_Word16 data_vec[PITCH_FRAME_LEN];
+
+ /* copy input */
+ memcpy(data_vec+1, in, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (N-1)));
+
+
+ data_vec[0] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)],16); //the z^(-1) state
+ state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)] = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)in[N-1],16);
+
+
+
+ AllpassFilterForDec32(data_vec+1, kApUpperQ15, N, state_in);
+ AllpassFilterForDec32(data_vec, kApLowerQ15, N, state_in+ALLPASSSECTIONS);
+
+ for (n=0;n<N/2;n++) {
+ out[n]=WEBRTC_SPL_ADD_SAT_W16(data_vec[WEBRTC_SPL_MUL_16_16(2, n)], data_vec[WEBRTC_SPL_MUL_16_16(2, n)+1]);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c
new file mode 100644
index 0000000..8270359
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c
@@ -0,0 +1,167 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filters_neon.c
+ *
+ * This file contains function WebRtcIsacfix_AutocorrNeon, optimized for
+ * ARM Neon platform.
+ *
+ */
+
+#include <arm_neon.h>
+#include <assert.h>
+
+#include "codec.h"
+
+// Autocorrelation function in fixed point.
+// NOTE! Different from SPLIB-version in how it scales the signal.
+int WebRtcIsacfix_AutocorrNeon(
+ WebRtc_Word32* __restrict r,
+ const WebRtc_Word16* __restrict x,
+ WebRtc_Word16 N,
+ WebRtc_Word16 order,
+ WebRtc_Word16* __restrict scale) {
+
+ // The 1st for loop assumed N % 4 == 0.
+ assert(N % 4 == 0);
+
+ int i = 0;
+ int zeros_low = 0;
+ int zeros_high = 0;
+ int16_t scaling = 0;
+ int32_t sum = 0;
+
+ // Step 1, calculate r[0] and how much scaling is needed.
+
+ int16x4_t reg16x4;
+ int64x1_t reg64x1a;
+ int64x1_t reg64x1b;
+ int32x4_t reg32x4;
+ int64x2_t reg64x2 = vdupq_n_s64(0); // zeros
+
+ // Loop over the samples and do:
+ // sum += WEBRTC_SPL_MUL_16_16(x[i], x[i]);
+ for (i = 0; i < N; i += 4) {
+ reg16x4 = vld1_s16(&x[i]);
+ reg32x4 = vmull_s16(reg16x4, reg16x4);
+ reg64x2 = vpadalq_s32(reg64x2, reg32x4);
+ }
+ reg64x1a = vget_low_s64(reg64x2);
+ reg64x1b = vget_high_s64(reg64x2);
+ reg64x1a = vadd_s64(reg64x1a, reg64x1b);
+
+ // Calculate the value of shifting (scaling).
+ __asm__ __volatile__(
+ "vmov %[z_l], %[z_h], %P[reg]\n\t"
+ "clz %[z_l], %[z_l]\n\t"
+ "clz %[z_h], %[z_h]\n\t"
+ :[z_l]"+r"(zeros_low),
+ [z_h]"+r"(zeros_high)
+ :[reg]"w"(reg64x1a)
+ );
+ if (zeros_high != 32) {
+ scaling = (32 - zeros_high + 1);
+ } else if (zeros_low == 0) {
+ scaling = 1;
+ }
+ reg64x1b = -scaling;
+ reg64x1a = vshl_s64(reg64x1a, reg64x1b);
+
+ // Record the result.
+ r[0] = (int32_t)vget_lane_s64(reg64x1a, 0);
+
+
+ // Step 2, perform the actual correlation calculation.
+
+ /* Original C code (for the rest of the function):
+ for (i = 1; i < order + 1; i++) {
+ prod = 0;
+ for (j = 0; j < N - i; j++) {
+ prod += WEBRTC_SPL_MUL_16_16(x[j], x[i + j]);
+ }
+ sum = (int32_t)(prod >> scaling);
+ r[i] = sum;
+ }
+ */
+
+ for (i = 1; i < order + 1; i++) {
+ int32_t prod_lower = 0;
+ int32_t prod_upper = 0;
+ int16_t* ptr0 = &x[0];
+ int16_t* ptr1 = &x[i];
+ int32_t tmp = 0;
+
+ // Initialize the sum (q9) to zero.
+ __asm__ __volatile__("vmov.i32 q9, #0\n\t":::"q9");
+
+ // Calculate the major block of the samples (a multiple of 8).
+ for (; ptr0 < &x[N - i - 7];) {
+ __asm__ __volatile__(
+ "vld1.16 {d20, d21}, [%[ptr0]]!\n\t"
+ "vld1.16 {d22, d23}, [%[ptr1]]!\n\t"
+ "vmull.s16 q12, d20, d22\n\t"
+ "vmull.s16 q13, d21, d23\n\t"
+ "vpadal.s32 q9, q12\n\t"
+ "vpadal.s32 q9, q13\n\t"
+
+ // Specify constraints.
+ :[ptr0]"+r"(ptr0),
+ [ptr1]"+r"(ptr1)
+ :
+ :"d18", "d19", "d20", "d21", "d22", "d23", "d24", "d25", "d26", "d27"
+ );
+ }
+
+ // Calculate the rest of the samples.
+ for (; ptr0 < &x[N - i]; ptr0++, ptr1++) {
+ __asm__ __volatile__(
+ "smulbb %[tmp], %[ptr0], %[ptr1]\n\t"
+ "adds %[prod_lower], %[prod_lower], %[tmp]\n\t"
+ "adc %[prod_upper], %[prod_upper], %[tmp], asr #31\n\t"
+
+ // Specify constraints.
+ :[prod_lower]"+r"(prod_lower),
+ [prod_upper]"+r"(prod_upper),
+ [tmp]"+r"(tmp)
+ :[ptr0]"r"(*ptr0),
+ [ptr1]"r"(*ptr1)
+ );
+ }
+
+ // Sum the results up, and do shift.
+ __asm__ __volatile__(
+ "vadd.i64 d18, d19\n\t"
+ "vmov.32 d17[0], %[prod_lower]\n\t"
+ "vmov.32 d17[1], %[prod_upper]\n\t"
+ "vadd.i64 d17, d18\n\t"
+ "mov %[tmp], %[scaling], asr #31\n\t"
+ "vmov.32 d16, %[scaling], %[tmp]\n\t"
+ "vshl.s64 d17, d16\n\t"
+ "vmov.32 %[sum], d17[0]\n\t"
+
+ // Specify constraints.
+ :[sum]"=r"(sum),
+ [tmp]"+r"(tmp)
+ :[prod_upper]"r"(prod_upper),
+ [prod_lower]"r"(prod_lower),
+ [scaling]"r"(-scaling)
+ :"d16", "d17", "d18", "d19"
+ );
+
+ // Record the result.
+ r[i] = sum;
+ }
+
+ // Record the result.
+ *scale = scaling;
+
+ return(order + 1);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c
new file mode 100644
index 0000000..4d11af5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c
@@ -0,0 +1,175 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * initialize.c
+ *
+ * Internal initfunctions
+ *
+ */
+
+#include "codec.h"
+#include "structs.h"
+#include "pitch_estimator.h"
+
+
+void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata) {
+
+ int k;
+
+ for (k = 0; k < WINLEN; k++) {
+ maskdata->DataBufferLoQ0[k] = (WebRtc_Word16) 0;
+ maskdata->DataBufferHiQ0[k] = (WebRtc_Word16) 0;
+ }
+ for (k = 0; k < ORDERLO+1; k++) {
+ maskdata->CorrBufLoQQ[k] = (WebRtc_Word32) 0;
+ maskdata->CorrBufLoQdom[k] = 0;
+
+ maskdata->PreStateLoGQ15[k] = 0;
+
+ }
+ for (k = 0; k < ORDERHI+1; k++) {
+ maskdata->CorrBufHiQQ[k] = (WebRtc_Word32) 0;
+ maskdata->CorrBufHiQdom[k] = 0;
+ maskdata->PreStateHiGQ15[k] = 0;
+ }
+
+ maskdata->OldEnergy = 10;
+
+ return;
+}
+
+void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec *maskdata) {
+
+ int k;
+
+ for (k = 0; k < ORDERLO+1; k++)
+ {
+ maskdata->PostStateLoGQ0[k] = 0;
+ }
+ for (k = 0; k < ORDERHI+1; k++)
+ {
+ maskdata->PostStateHiGQ0[k] = 0;
+ }
+
+ maskdata->OldEnergy = 10;
+
+ return;
+}
+
+
+
+
+
+
+
+void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr *prefiltdata)
+{
+ int k;
+
+ for (k = 0; k < QLOOKAHEAD; k++) {
+ prefiltdata->INLABUF1_fix[k] = 0;
+ prefiltdata->INLABUF2_fix[k] = 0;
+ }
+ for (k = 0; k < WEBRTC_SPL_MUL_16_16(2,(QORDER-1)); k++) {
+
+ prefiltdata->INSTAT1_fix[k] = 0;
+ prefiltdata->INSTAT2_fix[k] = 0;
+ }
+
+ /* High pass filter states */
+ prefiltdata->HPstates_fix[0] = 0;
+ prefiltdata->HPstates_fix[1] = 0;
+
+ return;
+}
+
+void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr *postfiltdata)
+{
+ int k;
+
+ for (k = 0; k < WEBRTC_SPL_MUL_16_16(2, POSTQORDER); k++) {
+
+ postfiltdata->STATE_0_LOWER_fix[k] = 0;
+ postfiltdata->STATE_0_UPPER_fix[k] = 0;
+ }
+
+ /* High pass filter states */
+
+ postfiltdata->HPstates1_fix[0] = 0;
+ postfiltdata->HPstates1_fix[1] = 0;
+
+ postfiltdata->HPstates2_fix[0] = 0;
+ postfiltdata->HPstates2_fix[1] = 0;
+
+ return;
+}
+
+
+void WebRtcIsacfix_InitPitchFilter(PitchFiltstr *pitchfiltdata)
+{
+ int k;
+
+ for (k = 0; k < PITCH_BUFFSIZE; k++)
+ pitchfiltdata->ubufQQ[k] = 0;
+ for (k = 0; k < (PITCH_DAMPORDER); k++)
+ pitchfiltdata->ystateQQ[k] = 0;
+
+ pitchfiltdata->oldlagQ7 = 6400; /* 50.0 in Q7 */
+ pitchfiltdata->oldgainQ12 = 0;
+}
+
+void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct *State)
+{
+ int k;
+
+ for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++) {
+ State->dec_buffer16[k] = 0;
+ }
+ for (k = 0; k < WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)+1; k++) {
+ State->decimator_state32[k] = 0;
+ }
+
+ for (k = 0; k < QLOOKAHEAD; k++)
+ State->inbuf[k] = 0;
+
+ WebRtcIsacfix_InitPitchFilter(&(State->PFstr_wght));
+
+ WebRtcIsacfix_InitPitchFilter(&(State->PFstr));
+}
+
+
+void WebRtcIsacfix_InitPlc( PLCstr *State )
+{
+ State->decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX;
+ State->decayCoeffNoise = WEBRTC_SPL_WORD16_MAX;
+
+ State->used = PLC_WAS_USED;
+
+ WebRtcSpl_ZerosArrayW16(State->overlapLP, RECOVERY_OVERLAP);
+ WebRtcSpl_ZerosArrayW16(State->lofilt_coefQ15, ORDERLO);
+ WebRtcSpl_ZerosArrayW16(State->hifilt_coefQ15, ORDERHI );
+
+ State->AvgPitchGain_Q12 = 0;
+ State->lastPitchGain_Q12 = 0;
+ State->lastPitchLag_Q7 = 0;
+ State->gain_lo_hiQ17[0]=State->gain_lo_hiQ17[1] = 0;
+ WebRtcSpl_ZerosArrayW16(State->prevPitchInvIn, FRAMESAMPLES/2);
+ WebRtcSpl_ZerosArrayW16(State->prevPitchInvOut, PITCH_MAX_LAG + 10 );
+ WebRtcSpl_ZerosArrayW32(State->prevHP, PITCH_MAX_LAG + 10 );
+ State->pitchCycles = 0;
+ State->A = 0;
+ State->B = 0;
+ State->pitchIndex = 0;
+ State->stretchLag = 240;
+ State->seed = 4447;
+
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c
new file mode 100644
index 0000000..3a37785
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c
@@ -0,0 +1,1529 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * isacfix.c
+ *
+ * This C file contains the functions for the ISAC API
+ *
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "isacfix.h"
+#include "bandwidth_estimator.h"
+#include "codec.h"
+#include "entropy_coding.h"
+#include "structs.h"
+
+
+/**************************************************************************
+ * WebRtcIsacfix_AssignSize(...)
+ *
+ * Functions used when malloc is not allowed
+ * Returns number of bytes needed to allocate for iSAC struct.
+ *
+ */
+
+WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes) {
+ *sizeinbytes=sizeof(ISACFIX_SubStruct)*2/sizeof(WebRtc_Word16);
+ return(0);
+}
+
+/***************************************************************************
+ * WebRtcIsacfix_Assign(...)
+ *
+ * Functions used when malloc is not allowed
+ * Place struct at given address
+ *
+ * If successful, Return 0, else Return -1
+ */
+
+WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) {
+ if (ISACFIX_inst_Addr!=NULL) {
+ *inst = (ISACFIX_MainStruct*)ISACFIX_inst_Addr;
+ (*(ISACFIX_SubStruct**)inst)->errorcode = 0;
+ (*(ISACFIX_SubStruct**)inst)->initflag = 0;
+ (*(ISACFIX_SubStruct**)inst)->ISACenc_obj.SaveEnc_ptr = NULL;
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+
+#ifndef ISACFIX_NO_DYNAMIC_MEM
+
+/****************************************************************************
+ * WebRtcIsacfix_Create(...)
+ *
+ * This function creates a ISAC instance, which will contain the state
+ * information for one coding/decoding channel.
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst)
+{
+ ISACFIX_SubStruct *tempo;
+ tempo = malloc(1 * sizeof(ISACFIX_SubStruct));
+ *ISAC_main_inst = (ISACFIX_MainStruct *)tempo;
+ if (*ISAC_main_inst!=NULL) {
+ (*(ISACFIX_SubStruct**)ISAC_main_inst)->errorcode = 0;
+ (*(ISACFIX_SubStruct**)ISAC_main_inst)->initflag = 0;
+ (*(ISACFIX_SubStruct**)ISAC_main_inst)->ISACenc_obj.SaveEnc_ptr = NULL;
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_CreateInternal(...)
+ *
+ * This function creates the memory that is used to store data in the encoder
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Allocate memory for storing encoder data */
+ ISAC_inst->ISACenc_obj.SaveEnc_ptr = malloc(1 * sizeof(ISAC_SaveEncData_t));
+
+ if (ISAC_inst->ISACenc_obj.SaveEnc_ptr!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+
+#endif
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_Free(...)
+ *
+ * This function frees the ISAC instance created at the beginning.
+ *
+ * Input:
+ * - ISAC_main_inst : a ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ free(ISAC_main_inst);
+ return(0);
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_FreeInternal(...)
+ *
+ * This function frees the internal memory for storing encoder data.
+ *
+ * Input:
+ * - ISAC_main_inst : a ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Release memory */
+ free(ISAC_inst->ISACenc_obj.SaveEnc_ptr);
+
+ return(0);
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_EncoderInit(...)
+ *
+ * This function initializes a ISAC instance prior to the encoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - CodingMode : 0 -> Bit rate and frame length are automatically
+ * adjusted to available bandwidth on
+ * transmission channel.
+ * 1 -> User sets a frame length and a target bit
+ * rate which is taken as the maximum short-term
+ * average bit rate.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 CodingMode)
+{
+ int k;
+ WebRtc_Word16 statusInit;
+ ISACFIX_SubStruct *ISAC_inst;
+
+ statusInit = 0;
+ /* typecast pointer to rela structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* flag encoder init */
+ ISAC_inst->initflag |= 2;
+
+ if (CodingMode == 0)
+ /* Adaptive mode */
+ ISAC_inst->ISACenc_obj.new_framelength = INITIAL_FRAMESAMPLES;
+ else if (CodingMode == 1)
+ /* Instantaneous mode */
+ ISAC_inst->ISACenc_obj.new_framelength = 480; /* default for I-mode */
+ else {
+ ISAC_inst->errorcode = ISAC_DISALLOWED_CODING_MODE;
+ statusInit = -1;
+ }
+
+ ISAC_inst->CodingMode = CodingMode;
+
+ WebRtcIsacfix_InitMaskingEnc(&ISAC_inst->ISACenc_obj.maskfiltstr_obj);
+ WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACenc_obj.prefiltbankstr_obj);
+ WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACenc_obj.pitchfiltstr_obj);
+ WebRtcIsacfix_InitPitchAnalysis(&ISAC_inst->ISACenc_obj.pitchanalysisstr_obj);
+
+
+ WebRtcIsacfix_InitBandwidthEstimator(&ISAC_inst->bwestimator_obj);
+ WebRtcIsacfix_InitRateModel(&ISAC_inst->ISACenc_obj.rate_data_obj);
+
+
+ ISAC_inst->ISACenc_obj.buffer_index = 0;
+ ISAC_inst->ISACenc_obj.frame_nb = 0;
+ ISAC_inst->ISACenc_obj.BottleNeck = 32000; /* default for I-mode */
+ ISAC_inst->ISACenc_obj.MaxDelay = 10; /* default for I-mode */
+ ISAC_inst->ISACenc_obj.current_framesamples = 0;
+ ISAC_inst->ISACenc_obj.s2nr = 0;
+ ISAC_inst->ISACenc_obj.MaxBits = 0;
+ ISAC_inst->ISACenc_obj.bitstr_seed = 4447;
+ ISAC_inst->ISACenc_obj.payloadLimitBytes30 = STREAM_MAXW16_30MS << 1;
+ ISAC_inst->ISACenc_obj.payloadLimitBytes60 = STREAM_MAXW16_60MS << 1;
+ ISAC_inst->ISACenc_obj.maxPayloadBytes = STREAM_MAXW16_60MS << 1;
+ ISAC_inst->ISACenc_obj.maxRateInBytes = STREAM_MAXW16_30MS << 1;
+ ISAC_inst->ISACenc_obj.enforceFrameSize = 0;
+
+ /* Init the bistream data area to zero */
+ for (k=0; k<STREAM_MAXW16_60MS; k++){
+ ISAC_inst->ISACenc_obj.bitstr_obj.stream[k] = 0;
+ }
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACenc_obj.interpolatorstr_obj);
+#endif
+
+ // Initiaze function pointers.
+ WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrC;
+ WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopC;
+
+#ifdef WEBRTC_ARCH_ARM_NEON
+ WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrNeon;
+ WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopNeon;
+#endif
+
+ return statusInit;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_Encode(...)
+ *
+ * This function encodes 10ms frame(s) and inserts it into a package.
+ * Input speech length has to be 160 samples (10ms). The encoder buffers those
+ * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : 0 - The buffer didn't reach the chosen framesize
+ * so it keeps buffering speech samples.
+ * : -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_Word16 *speechIn,
+ WebRtc_Word16 *encoded)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ WebRtc_Word16 stream_len;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+
+ /* typecast pointer to rela structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+
+ /* check if encoder initiated */
+ if ((ISAC_inst->initflag & 2) != 2) {
+ ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechIn,
+ &ISAC_inst->ISACenc_obj,
+ &ISAC_inst->bwestimator_obj,
+ ISAC_inst->CodingMode);
+ if (stream_len<0) {
+ ISAC_inst->errorcode = - stream_len;
+ return -1;
+ }
+
+
+ /* convert from bytes to WebRtc_Word16 */
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0;k<(stream_len+1)>>1;k++) {
+ encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
+ | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
+ }
+
+#else
+ WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
+#endif
+
+
+
+ return stream_len;
+
+}
+
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_EncodeNb(...)
+ *
+ * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
+ * it into a package. Input speech length has to be 80 samples (10ms). The encoder
+ * interpolates into wide-band (16 kHz sampling) buffers those
+ * 10ms frames until it reaches the chosen Framesize (480 or 960 wide-band samples
+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : 0 - The buffer didn't reach the chosen framesize
+ * so it keeps buffering speech samples.
+ * : -1 - Error
+ */
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_Word16 *speechIn,
+ WebRtc_Word16 *encoded)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ WebRtc_Word16 stream_len;
+ WebRtc_Word16 speechInWB[FRAMESAMPLES_10ms];
+ WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES_10ms/2];
+ WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES_10ms/2];
+
+ int k;
+
+
+ /* typecast pointer to rela structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+
+ /* check if encoder initiated */
+ if ((ISAC_inst->initflag & 2) != 2) {
+ ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
+ return (-1);
+ }
+
+
+ /* Oversample to WB */
+
+ /* Form polyphase signals, and compensate for DC offset */
+ for (k=0;k<FRAMESAMPLES_10ms/2;k++) {
+ Vector_Word16_1[k] = speechIn[k] + 1;
+ Vector_Word16_2[k] = speechIn[k];
+ }
+ WebRtcIsacfix_FilterAndCombine2(Vector_Word16_1, Vector_Word16_2, speechInWB, &ISAC_inst->ISACenc_obj.interpolatorstr_obj, FRAMESAMPLES_10ms);
+
+
+ /* Encode WB signal */
+ stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechInWB,
+ &ISAC_inst->ISACenc_obj,
+ &ISAC_inst->bwestimator_obj,
+ ISAC_inst->CodingMode);
+ if (stream_len<0) {
+ ISAC_inst->errorcode = - stream_len;
+ return -1;
+ }
+
+
+ /* convert from bytes to WebRtc_Word16 */
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0;k<(stream_len+1)>>1;k++) {
+ encoded[k] = (WebRtc_Word16)(((WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8)
+ | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
+ }
+
+#else
+ WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
+#endif
+
+
+
+ return stream_len;
+}
+#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
+
+
+/****************************************************************************
+ * WebRtcIsacfix_GetNewBitStream(...)
+ *
+ * This function returns encoded data, with the recieved bwe-index in the
+ * stream. It should always return a complete packet, i.e. only called once
+ * even for 60 msec frames
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - bweIndex : index of bandwidth estimate to put in new bitstream
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 bweIndex,
+ float scale,
+ WebRtc_Word16 *encoded)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ WebRtc_Word16 stream_len;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+
+ /* typecast pointer to rela structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+
+ /* check if encoder initiated */
+ if ((ISAC_inst->initflag & 2) != 2) {
+ ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ stream_len = WebRtcIsacfix_EncodeStoredData(&ISAC_inst->ISACenc_obj,
+ bweIndex,
+ scale);
+ if (stream_len<0) {
+ ISAC_inst->errorcode = - stream_len;
+ return -1;
+ }
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0;k<(stream_len+1)>>1;k++) {
+ encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
+ | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
+ }
+
+#else
+ WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
+#endif
+
+ return stream_len;
+
+}
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecoderInit(...)
+ *
+ * This function initializes a ISAC instance prior to the decoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ *
+ * Return value
+ * : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* flag decoder init */
+ ISAC_inst->initflag |= 1;
+
+
+ WebRtcIsacfix_InitMaskingDec(&ISAC_inst->ISACdec_obj.maskfiltstr_obj);
+ WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACdec_obj.postfiltbankstr_obj);
+ WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACdec_obj.pitchfiltstr_obj);
+
+ /* TS */
+ WebRtcIsacfix_InitPlc( &ISAC_inst->ISACdec_obj.plcstr_obj );
+
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACdec_obj.decimatorstr_obj);
+#endif
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_UpdateBwEstimate1(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 arr_ts)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ Bitstr_dec streamdata;
+ WebRtc_UWord16 partOfStream[5];
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ /* Set stream pointer to point at partOfStream */
+ streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Sanity check of packet length */
+ if (packet_size <= 0) {
+ /* return error code if the packet length is null or less */
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ } else if (packet_size > (STREAM_MAXW16<<1)) {
+ /* return error code if length of stream is too long */
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ /* check if decoder initiated */
+ if ((ISAC_inst->initflag & 1) != 1) {
+ ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+ streamdata.full = 1;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<5; k++) {
+ streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+#else
+ memcpy(streamdata.stream, encoded, 5);
+#endif
+
+ if (packet_size == 0)
+ {
+ /* return error code if the packet length is null */
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ }
+
+ err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj,
+ &streamdata,
+ packet_size,
+ rtp_seq_number,
+ 0,
+ arr_ts);
+
+
+ if (err < 0)
+ {
+ /* return error code if something went wrong */
+ ISAC_inst->errorcode = -err;
+ return -1;
+ }
+
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_UpdateBwEstimate(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - send_ts : Send Time Stamp from RTP header
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ Bitstr_dec streamdata;
+ WebRtc_UWord16 partOfStream[5];
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ /* Set stream pointer to point at partOfStream */
+ streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Sanity check of packet length */
+ if (packet_size <= 0) {
+ /* return error code if the packet length is null or less */
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ } else if (packet_size > (STREAM_MAXW16<<1)) {
+ /* return error code if length of stream is too long */
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ /* check if decoder initiated */
+ if ((ISAC_inst->initflag & 1) != 1) {
+ ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+ streamdata.full = 1;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<5; k++) {
+ streamdata.stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+#else
+ memcpy(streamdata.stream, encoded, 5);
+#endif
+
+ if (packet_size == 0)
+ {
+ /* return error code if the packet length is null */
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ }
+
+ err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj,
+ &streamdata,
+ packet_size,
+ rtp_seq_number,
+ send_ts,
+ arr_ts);
+
+ if (err < 0)
+ {
+ /* return error code if something went wrong */
+ ISAC_inst->errorcode = -err;
+ return -1;
+ }
+
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_Decode(...)
+ *
+ * This function decodes a ISAC frame. Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+
+WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ /* number of samples (480 or 960), output from decoder */
+ /* that were actually used in the encoder/decoder (determined on the fly) */
+ WebRtc_Word16 number_of_samples;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 declen = 0;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* check if decoder initiated */
+ if ((ISAC_inst->initflag & 1) != 1) {
+ ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ /* Sanity check of packet length */
+ if (len <= 0) {
+ /* return error code if the packet length is null or less */
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ } else if (len > (STREAM_MAXW16<<1)) {
+ /* return error code if length of stream is too long */
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
+
+ /* convert bitstream from WebRtc_Word16 to bytes */
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<(len>>1); k++) {
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+ if (len & 0x0001)
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
+#endif
+
+ /* added for NetEq purposes (VAD/DTX related) */
+ *speechType=1;
+
+ declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
+
+ if (declen < 0) {
+ /* Some error inside the decoder */
+ ISAC_inst->errorcode = -declen;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * MAX_FRAMESAMPLES);
+ return -1;
+ }
+
+ /* error check */
+
+ if (declen & 0x0001) {
+ if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+ return -1;
+ }
+ } else {
+ if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+ return -1;
+ }
+ }
+
+ return number_of_samples;
+}
+
+
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecodeNb(...)
+ *
+ * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
+ * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+ const WebRtc_UWord16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ /* twice the number of samples (480 or 960), output from decoder */
+ /* that were actually used in the encoder/decoder (determined on the fly) */
+ WebRtc_Word16 number_of_samples;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 declen = 0;
+ WebRtc_Word16 dummy[FRAMESAMPLES/2];
+
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* check if decoder initiated */
+ if ((ISAC_inst->initflag & 1) != 1) {
+ ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ if (len == 0)
+ { /* return error code if the packet length is null */
+
+ ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
+ return -1;
+ }
+
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
+
+ /* convert bitstream from WebRtc_Word16 to bytes */
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<(len>>1); k++) {
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+ if (len & 0x0001)
+ (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
+#endif
+
+ /* added for NetEq purposes (VAD/DTX related) */
+ *speechType=1;
+
+ declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
+
+ if (declen < 0) {
+ /* Some error inside the decoder */
+ ISAC_inst->errorcode = -declen;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * FRAMESAMPLES);
+ return -1;
+ }
+
+ /* error check */
+
+ if (declen & 0x0001) {
+ if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+ return -1;
+ }
+ } else {
+ if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
+ ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
+ memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
+ return -1;
+ }
+ }
+
+ WebRtcIsacfix_SplitAndFilter2(decoded, decoded, dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
+
+ if (number_of_samples>FRAMESAMPLES) {
+ WebRtcIsacfix_SplitAndFilter2(decoded + FRAMESAMPLES, decoded + FRAMESAMPLES/2,
+ dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
+ }
+
+ return number_of_samples/2;
+}
+#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecodePlcNb(...)
+ *
+ * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
+ * Output speech length will be "240*noOfLostFrames" samples
+ * that is equevalent of "30*noOfLostFrames" millisecond.
+ *
+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames (240 sample=30ms) to produce
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames )
+{
+ WebRtc_Word16 no_of_samples, declen, k, ok;
+ WebRtc_Word16 outframeNB[FRAMESAMPLES];
+ WebRtc_Word16 outframeWB[FRAMESAMPLES];
+ WebRtc_Word16 dummy[FRAMESAMPLES/2];
+
+
+ ISACFIX_SubStruct *ISAC_inst;
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
+ if (noOfLostFrames > 2){
+ noOfLostFrames = 2;
+ }
+
+ k = 0;
+ declen = 0;
+ while( noOfLostFrames > 0 )
+ {
+ ok = WebRtcIsacfix_DecodePlcImpl( outframeWB, &ISAC_inst->ISACdec_obj, &no_of_samples );
+ if(ok)
+ return -1;
+
+ WebRtcIsacfix_SplitAndFilter2(outframeWB, &(outframeNB[k*240]), dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
+
+ declen += no_of_samples;
+ noOfLostFrames--;
+ k++;
+ }
+
+ declen>>=1;
+
+ for (k=0;k<declen;k++) {
+ decoded[k] = outframeNB[k];
+ }
+
+ return declen;
+}
+#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
+
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_DecodePlc(...)
+ *
+ * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
+ * Output speech length will be "480*noOfLostFrames" samples
+ * that is equevalent of "30*noOfLostFrames" millisecond.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames (480sample = 30ms)
+ * to produce
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames)
+{
+
+ WebRtc_Word16 no_of_samples, declen, k, ok;
+ WebRtc_Word16 outframe16[MAX_FRAMESAMPLES];
+
+ ISACFIX_SubStruct *ISAC_inst;
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
+ if (noOfLostFrames > 2) {
+ noOfLostFrames = 2;
+ }
+ k = 0;
+ declen = 0;
+ while( noOfLostFrames > 0 )
+ {
+ ok = WebRtcIsacfix_DecodePlcImpl( &(outframe16[k*480]), &ISAC_inst->ISACdec_obj, &no_of_samples );
+ if(ok)
+ return -1;
+ declen += no_of_samples;
+ noOfLostFrames--;
+ k++;
+ }
+
+ for (k=0;k<declen;k++) {
+ decoded[k] = outframe16[k];
+ }
+
+ return declen;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_Control(...)
+ *
+ * This function sets the limit on the short-term average bit rate and the
+ * frame length. Should be used only in Instantaneous mode.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rate : limit on the short-term average bit rate,
+ * in bits/second (between 10000 and 32000)
+ * - framesize : number of milliseconds per frame (30 or 60)
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 rate,
+ WebRtc_Word16 framesize)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ if (ISAC_inst->CodingMode == 0)
+ {
+ /* in adaptive mode */
+ ISAC_inst->errorcode = ISAC_MODE_MISMATCH;
+ return -1;
+ }
+
+
+ if (rate >= 10000 && rate <= 32000)
+ ISAC_inst->ISACenc_obj.BottleNeck = rate;
+ else {
+ ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK;
+ return -1;
+ }
+
+
+
+ if (framesize == 30 || framesize == 60)
+ ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * framesize;
+ else {
+ ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
+ return -1;
+ }
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_ControlBwe(...)
+ *
+ * This function sets the initial values of bottleneck and frame-size if
+ * iSAC is used in channel-adaptive mode. Through this API, users can
+ * enforce a frame-size for all values of bottleneck. Then iSAC will not
+ * automatically change the frame-size.
+ *
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rateBPS : initial value of bottleneck in bits/second
+ * 10000 <= rateBPS <= 32000 is accepted
+ * For default bottleneck set rateBPS = 0
+ * - frameSizeMs : number of milliseconds per frame (30 or 60)
+ * - enforceFrameSize : 1 to enforce the given frame-size through out
+ * the adaptation process, 0 to let iSAC change
+ * the frame-size if required.
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 rateBPS,
+ WebRtc_Word16 frameSizeMs,
+ WebRtc_Word16 enforceFrameSize)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ /* Typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if ((ISAC_inst->initflag & 2) != 2) {
+ ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
+ return (-1);
+ }
+
+ /* Check that we are in channel-adaptive mode, otherwise, return -1 */
+ if (ISAC_inst->CodingMode != 0) {
+ ISAC_inst->errorcode = ISAC_MODE_MISMATCH;
+ return (-1);
+ }
+
+ /* Set struct variable if enforceFrameSize is set. ISAC will then keep the */
+ /* chosen frame size. */
+ ISAC_inst->ISACenc_obj.enforceFrameSize = (enforceFrameSize != 0)? 1:0;
+
+ /* Set initial rate, if value between 10000 and 32000, */
+ /* if rateBPS is 0, keep the default initial bottleneck value (15000) */
+ if ((rateBPS >= 10000) && (rateBPS <= 32000)) {
+ ISAC_inst->bwestimator_obj.sendBwAvg = (((WebRtc_UWord32)rateBPS) << 7);
+ } else if (rateBPS != 0) {
+ ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK;
+ return -1;
+ }
+
+ /* Set initial framesize. If enforceFrameSize is set the frame size will not change */
+ if ((frameSizeMs == 30) || (frameSizeMs == 60)) {
+ ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * frameSizeMs;
+ } else {
+ ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
+ return -1;
+ }
+
+ return 0;
+}
+
+
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_GetDownLinkBwIndex(...)
+ *
+ * This function returns index representing the Bandwidth estimate from
+ * other side to this side.
+ *
+ * Input:
+ * - ISAC_main_inst: iSAC struct
+ *
+ * Output:
+ * - rateIndex : Bandwidth estimate to transmit to other side.
+ *
+ */
+
+WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
+ WebRtc_Word16* rateIndex)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Call function to get Bandwidth Estimate */
+ *rateIndex = WebRtcIsacfix_GetDownlinkBwIndexImpl(&ISAC_inst->bwestimator_obj);
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_UpdateUplinkBw(...)
+ *
+ * This function takes an index representing the Bandwidth estimate from
+ * this side to other side and updates BWE.
+ *
+ * Input:
+ * - ISAC_main_inst: iSAC struct
+ * - rateIndex : Bandwidth estimate from other side.
+ *
+ */
+
+WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
+ WebRtc_Word16 rateIndex)
+{
+ WebRtc_Word16 err = 0;
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ /* Call function to update BWE with received Bandwidth Estimate */
+ err = WebRtcIsacfix_UpdateUplinkBwRec(&ISAC_inst->bwestimator_obj, rateIndex);
+ if (err < 0) {
+ ISAC_inst->errorcode = -err;
+ return (-1);
+ }
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_ReadFrameLen(...)
+ *
+ * This function returns the length of the frame represented in the packet.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ *
+ */
+
+WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
+ WebRtc_Word16* frameLength)
+{
+ Bitstr_dec streamdata;
+ WebRtc_UWord16 partOfStream[5];
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ /* Set stream pointer to point at partOfStream */
+ streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+ streamdata.full = 1;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<5; k++) {
+ streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+#else
+ memcpy(streamdata.stream, encoded, 5);
+#endif
+
+ /* decode frame length */
+ err = WebRtcIsacfix_DecodeFrameLen(&streamdata, frameLength);
+ if (err<0) // error check
+ return err;
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_ReadBwIndex(...)
+ *
+ * This function returns the index of the Bandwidth estimate from the bitstream.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ * - rateIndex : Bandwidth estimate in bitstream
+ *
+ */
+
+WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
+ WebRtc_Word16* rateIndex)
+{
+ Bitstr_dec streamdata;
+ WebRtc_UWord16 partOfStream[5];
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ /* Set stream pointer to point at partOfStream */
+ streamdata.stream = (WebRtc_UWord16 *)partOfStream;
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+ streamdata.full = 1;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<5; k++) {
+ streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ }
+#else
+ memcpy(streamdata.stream, encoded, 5);
+#endif
+
+ /* decode frame length, needed to get to the rateIndex in the bitstream */
+ err = WebRtcIsacfix_DecodeFrameLen(&streamdata, rateIndex);
+ if (err<0) // error check
+ return err;
+
+ /* decode BW estimation */
+ err = WebRtcIsacfix_DecodeSendBandwidth(&streamdata, rateIndex);
+ if (err<0) // error check
+ return err;
+
+ return 0;
+}
+
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_GetErrorCode(...)
+ *
+ * This function can be used to check the error code of an iSAC instance. When
+ * a function returns -1 a error code will be set for that instance. The
+ * function below extract the code of the last error that occured in the
+ * specified instance.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance
+ *
+ * Return value : Error code
+ */
+
+WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ return ISAC_inst->errorcode;
+}
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_GetUplinkBw(...)
+ *
+ * This function returns the inst quantized iSAC send bitrate
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : bitrate
+ */
+
+WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+ BwEstimatorstr * bw = (BwEstimatorstr*)&(ISAC_inst->bwestimator_obj);
+
+ return (WebRtc_Word32) WebRtcIsacfix_GetUplinkBandwidth(bw);
+}
+
+/****************************************************************************
+ * WebRtcIsacfix_GetNewFrameLen(...)
+ *
+ * This function return the next frame length (in samples) of iSAC.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : frame lenght in samples
+ */
+
+WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst)
+{
+ ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+ return ISAC_inst->ISACenc_obj.new_framelength;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_SetMaxPayloadSize(...)
+ *
+ * This function sets a limit for the maximum payload size of iSAC. The same
+ * value is used both for 30 and 60 msec packets.
+ * The absolute max will be valid until next time the function is called.
+ * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxPayloadBytes : maximum size of the payload in bytes
+ * valid values are between 100 and 400 bytes
+ *
+ *
+ * Return value : 0 if sucessful
+ * -1 if error happens
+ */
+
+WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word16 maxPayloadBytes)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ if((maxPayloadBytes < 100) || (maxPayloadBytes > 400))
+ {
+ /* maxPayloadBytes is out of valid range */
+ return -1;
+ }
+ else
+ {
+ /* Set new absolute max, which will not change unless this function
+ is called again with a new value */
+ ISAC_inst->ISACenc_obj.maxPayloadBytes = maxPayloadBytes;
+
+ /* Set new maximum values for 30 and 60 msec packets */
+ if (maxPayloadBytes < ISAC_inst->ISACenc_obj.maxRateInBytes) {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxPayloadBytes;
+ } else {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxRateInBytes;
+ }
+
+ if ( maxPayloadBytes < (ISAC_inst->ISACenc_obj.maxRateInBytes << 1)) {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes60 = maxPayloadBytes;
+ } else {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (ISAC_inst->ISACenc_obj.maxRateInBytes << 1);
+ }
+ }
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsacfix_SetMaxRate(...)
+ *
+ * This function sets the maximum rate which the codec may not exceed for a
+ * singel packet. The maximum rate is set in bits per second.
+ * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
+ * per 30 msec).
+ * It is possible to set a maximum rate between 32000 and 53400 bits per second.
+ *
+ * The rate limit is valid until next time the function is called.
+ *
+ * NOTE! Packet size will never go above the value set if calling
+ * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxRateInBytes : maximum rate in bits per second,
+ * valid values are 32000 to 53400 bits
+ *
+ * Return value : 0 if sucessful
+ * -1 if error happens
+ */
+
+WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
+ WebRtc_Word32 maxRate)
+{
+ ISACFIX_SubStruct *ISAC_inst;
+ WebRtc_Word16 maxRateInBytes;
+
+ /* typecast pointer to real structure */
+ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
+
+ if((maxRate < 32000) || (maxRate > 53400))
+ {
+ /* maxRate is out of valid range */
+ return -1;
+ }
+ else
+ {
+ /* Calculate maximum number of bytes per 30 msec packets for the given
+ maximum rate. Multiply with 30/1000 to get number of bits per 30 msec,
+ divide by 8 to get number of bytes per 30 msec:
+ maxRateInBytes = floor((maxRate * 30/1000) / 8); */
+ maxRateInBytes = (WebRtc_Word16)( WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_MUL(maxRate, 3), 800) );
+
+ /* Store the value for usage in the WebRtcIsacfix_SetMaxPayloadSize-function */
+ ISAC_inst->ISACenc_obj.maxRateInBytes = maxRateInBytes;
+
+ /* For 30 msec packets: if the new limit is below the maximum
+ payload size, set a new limit */
+ if (maxRateInBytes < ISAC_inst->ISACenc_obj.maxPayloadBytes) {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxRateInBytes;
+ } else {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxPayloadBytes;
+ }
+
+ /* For 60 msec packets: if the new limit (times 2) is below the
+ maximum payload size, set a new limit */
+ if ( (maxRateInBytes << 1) < ISAC_inst->ISACenc_obj.maxPayloadBytes) {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (maxRateInBytes << 1);
+ } else {
+ ISAC_inst->ISACenc_obj.payloadLimitBytes60 = ISAC_inst->ISACenc_obj.maxPayloadBytes;
+ }
+ }
+
+ return 0;
+}
+
+
+
+/****************************************************************************
+ * WebRtcIsacfix_version(...)
+ *
+ * This function returns the version number.
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ */
+
+void WebRtcIsacfix_version(char *version)
+{
+ strcpy(version, "3.6.0");
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi
new file mode 100644
index 0000000..a13d1f5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi
@@ -0,0 +1,82 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'iSACFix',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/isacfix.h',
+ 'arith_routines.c',
+ 'arith_routines_hist.c',
+ 'arith_routines_logist.c',
+ 'bandwidth_estimator.c',
+ 'decode.c',
+ 'decode_bwe.c',
+ 'decode_plc.c',
+ 'encode.c',
+ 'entropy_coding.c',
+ 'fft.c',
+ 'filterbank_tables.c',
+ 'filterbanks.c',
+ 'filters.c',
+ 'initialize.c',
+ 'isacfix.c',
+ 'lattice.c',
+ 'lattice_c.c',
+ 'lpc_masking_model.c',
+ 'lpc_tables.c',
+ 'pitch_estimator.c',
+ 'pitch_filter.c',
+ 'pitch_gain_tables.c',
+ 'pitch_lag_tables.c',
+ 'spectrum_ar_model_tables.c',
+ 'transform.c',
+ 'arith_routins.h',
+ 'bandwidth_estimator.h',
+ 'codec.h',
+ 'entropy_coding.h',
+ 'fft.h',
+ 'filterbank_tables.h',
+ 'lpc_masking_model.h',
+ 'lpc_tables.h',
+ 'pitch_estimator.h',
+ 'pitch_gain_tables.h',
+ 'pitch_lag_tables.h',
+ 'settings.h',
+ 'spectrum_ar_model_tables.h',
+ 'structs.h',
+ ],
+ 'conditions': [
+ ['OS!="win"', {
+ 'defines': [
+ 'WEBRTC_LINUX',
+ ],
+ }],
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c
new file mode 100644
index 0000000..8822c6e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c
@@ -0,0 +1,313 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lattice.c
+ *
+ * Contains the normalized lattice filter routines (MA and AR) for iSAC codec
+ *
+ */
+
+#include "codec.h"
+#include "settings.h"
+
+#define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32) \
+ ((WebRtc_Word32)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
+/* This macro is FORBIDDEN to use elsewhere than in a function in this file and
+ its corresponding neon version. It might give unpredictable results, since a
+ general WebRtc_Word32*WebRtc_Word32 multiplication results in a 64 bit value.
+ The result is then shifted just 16 steps to the right, giving need for 48
+ bits, i.e. in the generel case, it will NOT fit in a WebRtc_Word32. In the
+ cases used in here, the WebRtc_Word32 will be enough, since (for a good
+ reason) the involved multiplicands aren't big enough to overflow a
+ WebRtc_Word32 after shifting right 16 bits. I have compared the result of a
+ multiplication between t32 and tmp32, done in two ways:
+ 1) Using (WebRtc_Word32) (((float)(tmp32))*((float)(tmp32b))/65536.0);
+ 2) Using LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
+ By running 25 files, I haven't found any bigger diff than 64 - this was in the
+ case when method 1) gave 650235648 and 2) gave 650235712.
+*/
+
+/* Function prototype: filtering ar_g_Q0[] and ar_f_Q0[] through an AR filter
+ with coefficients cth_Q15[] and sth_Q15[].
+ Implemented for both generic and ARMv7 platforms.
+ */
+void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0,
+ int16_t* ar_f_Q0,
+ int16_t* cth_Q15,
+ int16_t* sth_Q15,
+ int16_t order_coef);
+
+/* Inner loop used for function WebRtcIsacfix_NormLatticeFilterMa().
+ It does:
+ for 0 <= n < HALF_SUBFRAMELEN - 1:
+ *ptr2 = input2 * (*ptr2) + input0 * (*ptr0));
+ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
+*/
+void WebRtcIsacfix_FilterMaLoopC(int16_t input0, // Filter coefficient
+ int16_t input1, // Filter coefficient
+ int32_t input2, // Inverse coeff. (1/input1)
+ int32_t* ptr0, // Sample buffer
+ int32_t* ptr1, // Sample buffer
+ int32_t* ptr2) { // Sample buffer
+ int n = 0;
+
+ // Separate the 32-bit variable input2 into two 16-bit integers (high 16 and
+ // low 16 bits), for using LATTICE_MUL_32_32_RSFT16 in the loop.
+ int16_t t16a = (int16_t)(input2 >> 16);
+ int16_t t16b = (int16_t)input2;
+ if (t16b < 0) t16a++;
+
+ // The loop filtering the samples *ptr0, *ptr1, *ptr2 with filter coefficients
+ // input0, input1, and input2.
+ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++, ptr0++, ptr1++, ptr2++) {
+ int32_t tmp32a = 0;
+ int32_t tmp32b = 0;
+
+ // Calculate *ptr2 = input2 * (*ptr2 + input0 * (*ptr0));
+ tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr0); // Q15 * Q15 >> 15 = Q15
+ tmp32b = *ptr2 + tmp32a; // Q15 + Q15 = Q15
+ *ptr2 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
+
+ // Calculate *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
+ tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input1, *ptr0); // Q15*Q15>>15 = Q15
+ tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr2); // Q15*Q15>>15 = Q15
+ *ptr1 = tmp32a + tmp32b; // Q15 + Q15 = Q15
+ }
+}
+
+// Declare a function pointer.
+FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
+
+/* filter the signal using normalized lattice filter */
+/* MA filter */
+void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
+ WebRtc_Word32 *stateGQ15,
+ WebRtc_Word16 *lat_inQ0,
+ WebRtc_Word16 *filt_coefQ15,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 lo_hi,
+ WebRtc_Word16 *lat_outQ9)
+{
+ WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
+ WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
+
+ int u, i, k, n;
+ WebRtc_Word16 temp2,temp3;
+ WebRtc_Word16 ord_1 = orderCoef+1;
+ WebRtc_Word32 inv_cthQ16[MAX_AR_MODEL_ORDER];
+
+ WebRtc_Word32 gain32, fQtmp;
+ WebRtc_Word16 gain16;
+ WebRtc_Word16 gain_sh;
+
+ WebRtc_Word32 tmp32, tmp32b;
+ WebRtc_Word32 fQ15vec[HALF_SUBFRAMELEN];
+ WebRtc_Word32 gQ15[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
+ WebRtc_Word16 sh;
+ WebRtc_Word16 t16a;
+ WebRtc_Word16 t16b;
+
+ for (u=0;u<SUBFRAMES;u++)
+ {
+ int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
+
+ /* set the Direct Form coefficients */
+ temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
+ temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u)+lo_hi;
+
+ /* compute lattice filter coefficients */
+ memcpy(sthQ15, &filt_coefQ15[temp2], orderCoef * sizeof(WebRtc_Word16));
+
+ WebRtcSpl_SqrtOfOneMinusXSquared(sthQ15, orderCoef, cthQ15);
+
+ /* compute the gain */
+ gain32 = gain_lo_hiQ17[temp3];
+ gain_sh = WebRtcSpl_NormW32(gain32);
+ gain32 = WEBRTC_SPL_LSHIFT_W32(gain32, gain_sh); //Q(17+gain_sh)
+
+ for (k=0;k<orderCoef;k++)
+ {
+ gain32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[k], gain32); //Q15*Q(17+gain_sh)>>15 = Q(17+gain_sh)
+ inv_cthQ16[k] = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, cthQ15[k]); // 1/cth[k] in Q31/Q15 = Q16
+ }
+ gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(gain32, 16); //Q(1+gain_sh)
+
+ /* normalized lattice filter */
+ /*****************************/
+
+ /* initial conditions */
+ for (i=0;i<HALF_SUBFRAMELEN;i++)
+ {
+ fQ15vec[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
+ gQ15[0][i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
+ }
+
+
+ fQtmp = fQ15vec[0];
+
+ /* get the state of f&g for the first input, for all orders */
+ for (i=1;i<ord_1;i++)
+ {
+ // Calculate f[i][0] = inv_cth[i-1]*(f[i-1][0] + sth[i-1]*stateG[i-1]);
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], stateGQ15[i-1]);//Q15*Q15>>15 = Q15
+ tmp32b= fQtmp + tmp32; //Q15+Q15=Q15
+ tmp32 = inv_cthQ16[i-1]; //Q16
+ t16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+ t16b = (WebRtc_Word16) (tmp32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)t16a), 16));
+ if (t16b<0) t16a++;
+ tmp32 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
+ fQtmp = tmp32; // Q15
+
+ // Calculate g[i][0] = cth[i-1]*stateG[i-1] + sth[i-1]* f[i][0];
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[i-1], stateGQ15[i-1]); //Q15*Q15>>15 = Q15
+ tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], fQtmp); //Q15*Q15>>15 = Q15
+ tmp32 = tmp32 + tmp32b;//Q15+Q15 = Q15
+ gQ15[i][0] = tmp32; // Q15
+ }
+
+ /* filtering */
+ /* save the states */
+ for(k=0;k<orderCoef;k++)
+ {
+ // for 0 <= n < HALF_SUBFRAMELEN - 1:
+ // f[k+1][n+1] = inv_cth[k]*(f[k][n+1] + sth[k]*g[k][n]);
+ // g[k+1][n+1] = cth[k]*g[k][n] + sth[k]* f[k+1][n+1];
+ WebRtcIsacfix_FilterMaLoopFix(sthQ15[k], cthQ15[k], inv_cthQ16[k],
+ &gQ15[k][0], &gQ15[k+1][1], &fQ15vec[1]);
+ }
+
+ fQ15vec[0] = fQtmp;
+
+ for(n=0;n<HALF_SUBFRAMELEN;n++)
+ {
+ //gain32 = WEBRTC_SPL_RSHIFT_W32(gain32, gain_sh); // Q(17+gain_sh) -> Q17
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(gain16, fQ15vec[n]); //Q(1+gain_sh)*Q15>>16 = Q(gain_sh)
+ sh = 9-gain_sh; //number of needed shifts to reach Q9
+ t16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh);
+ lat_outQ9[n + temp1] = t16a;
+ }
+
+ /* save the states */
+ for (i=0;i<ord_1;i++)
+ {
+ stateGQ15[i] = gQ15[i][HALF_SUBFRAMELEN-1];
+ }
+ //process next frame
+ }
+
+ return;
+}
+
+
+
+
+
+/* ----------------AR filter-------------------------*/
+/* filter the signal using normalized lattice filter */
+void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
+ WebRtc_Word16 *stateGQ0,
+ WebRtc_Word32 *lat_inQ25,
+ WebRtc_Word16 *filt_coefQ15,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 lo_hi,
+ WebRtc_Word16 *lat_outQ0)
+{
+ int ii,n,k,i,u;
+ WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
+ WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
+ WebRtc_Word32 tmp32;
+
+
+ WebRtc_Word16 tmpAR;
+ WebRtc_Word16 ARfQ0vec[HALF_SUBFRAMELEN];
+ WebRtc_Word16 ARgQ0vec[MAX_AR_MODEL_ORDER+1];
+
+ WebRtc_Word32 inv_gain32;
+ WebRtc_Word16 inv_gain16;
+ WebRtc_Word16 den16;
+ WebRtc_Word16 sh;
+
+ WebRtc_Word16 temp2,temp3;
+ WebRtc_Word16 ord_1 = orderCoef+1;
+
+ for (u=0;u<SUBFRAMES;u++)
+ {
+ int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
+
+ //set the denominator and numerator of the Direct Form
+ temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
+ temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u) + lo_hi;
+
+ for (ii=0; ii<orderCoef; ii++) {
+ sthQ15[ii] = filt_coefQ15[temp2+ii];
+ }
+
+ WebRtcSpl_SqrtOfOneMinusXSquared(sthQ15, orderCoef, cthQ15);
+
+ /* Simulation of the 25 files shows that maximum value in
+ the vector gain_lo_hiQ17[] is 441344, which means that
+ it is log2((2^31)/441344) = 12.2 shifting bits from
+ saturation. Therefore, it should be safe to use Q27 instead
+ of Q17. */
+
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(gain_lo_hiQ17[temp3], 10); // Q27
+
+ for (k=0;k<orderCoef;k++) {
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[k], tmp32); // Q15*Q27>>15 = Q27
+ }
+
+ sh = WebRtcSpl_NormW32(tmp32); // tmp32 is the gain
+ den16 = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh-16); //Q(27+sh-16) = Q(sh+11) (all 16 bits are value bits)
+ inv_gain32 = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, den16); // 1/gain in Q31/Q(sh+11) = Q(20-sh)
+
+ //initial conditions
+ inv_gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inv_gain32, 2); // 1/gain in Q(20-sh-2) = Q(18-sh)
+
+ for (i=0;i<HALF_SUBFRAMELEN;i++)
+ {
+
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(lat_inQ25[i + temp1], 1); //Q25->Q26
+ tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh)
+ tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0
+
+ ARfQ0vec[i] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ }
+
+ for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
+ {
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(cthQ15[i],ARfQ0vec[0])) - (WEBRTC_SPL_MUL_16_16(sthQ15[i],stateGQ0[i])) + 16384), 15);
+ tmpAR = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sthQ15[i],ARfQ0vec[0])) + (WEBRTC_SPL_MUL_16_16(cthQ15[i], stateGQ0[i])) + 16384), 15);
+ ARgQ0vec[i+1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ ARfQ0vec[0] = tmpAR;
+ }
+ ARgQ0vec[0] = ARfQ0vec[0];
+
+ // Filter ARgQ0vec[] and ARfQ0vec[] through coefficients cthQ15[] and sthQ15[].
+ WebRtcIsacfix_FilterArLoop(ARgQ0vec, ARfQ0vec, cthQ15, sthQ15, orderCoef);
+
+ for(n=0;n<HALF_SUBFRAMELEN;n++)
+ {
+ lat_outQ0[n + temp1] = ARfQ0vec[n];
+ }
+
+
+ /* cannot use memcpy in the following */
+
+ for (i=0;i<ord_1;i++)
+ {
+ stateGQ0[i] = ARgQ0vec[i];
+ }
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S
new file mode 100644
index 0000000..1cd3a76
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S
@@ -0,0 +1,82 @@
+@
+@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ Contains a function for the core loop in the normalized lattice AR
+@ filter routine for iSAC codec, optimized for ARMv7 platforms.
+@
+@ Output is bit-exact with the reference C code in lattic_c.c
+@
+@ Register usage:
+@
+@ r0: &ar_g_Q0
+@ r1: &ar_f_Q0
+@ r2: &cth_Q15
+@ r3: &sth_Q15
+@ r4: out loop counter
+@ r5: tmpAR
+@ r9: inner loop counter
+@ r12: constant #16384
+@ r6, r7, r8, r10, r11: scratch
+
+#include "settings.h"
+
+.arch armv7-a
+.global WebRtcIsacfix_FilterArLoop
+.align 2
+
+WebRtcIsacfix_FilterArLoop:
+.fnstart
+
+.save {r4-r11}
+ push {r4-r11}
+
+ add r1, #2 @ &ar_f_Q0[1]
+ mov r12, #16384
+ mov r4, #HALF_SUBFRAMELEN
+ sub r4, #1 @ Outer loop counter = HALF_SUBFRAMELEN - 1
+
+HALF_SUBFRAME_LOOP: @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
+
+ ldr r9, [sp, #32] @ Restore the inner loop counter to order_coef
+ ldrh r5, [r1] @ tmpAR = ar_f_Q0[n+1]
+ add r0, r9, asl #1 @ Restore r0 to &ar_g_Q0[order_coef]
+ add r2, r9, asl #1 @ Restore r2 to &cth_Q15[order_coef]
+ add r3, r9, asl #1 @ Restore r3 to &sth_Q15[order_coef]
+
+ORDER_COEF_LOOP: @ for(k = order_coef - 1 ; k >= 0; k--)
+
+ ldrh r7, [r3, #-2]! @ sth_Q15[k]
+ ldrh r6, [r2, #-2]! @ cth_Q15[k]
+
+ ldrh r8, [r0, #-2] @ ar_g_Q0[k]
+ smlabb r11, r7, r5, r12 @ sth_Q15[k] * tmpAR + 16384
+ smlabb r10, r6, r5, r12 @ cth_Q15[k] * tmpAR + 16384
+ smulbb r7, r7, r8 @ sth_Q15[k] * ar_g_Q0[k]
+ smlabb r11, r6, r8, r11 @ cth_Q15[k]*ar_g_Q0[k]+(sth_Q15[k]*tmpAR+16384)
+
+ sub r10, r10, r7 @ cth_Q15[k]*tmpAR+16384-(sth_Q15[k]*ar_g_Q0[k])
+ ssat r11, #16, r11, asr #15
+ ssat r5, #16, r10, asr #15
+ strh r11, [r0], #-2 @ Output: ar_g_Q0[k+1]
+
+ subs r9, #1
+ bgt ORDER_COEF_LOOP
+
+ strh r5, [r0] @ Output: ar_g_Q0[0] = tmpAR;
+ strh r5, [r1], #2 @ Output: ar_f_Q0[n+1] = tmpAR;
+
+ subs r4, #1
+ bne HALF_SUBFRAME_LOOP
+
+ pop {r4-r11}
+ bx lr
+
+.fnend
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c
new file mode 100644
index 0000000..80ccf39
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Contains the core loop function for the lattice filter AR routine
+ * for iSAC codec.
+ *
+ */
+
+#include "settings.h"
+#include "signal_processing_library.h"
+#include "typedefs.h"
+
+/* Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
+ * cth_Q15[] and sth_Q15[].
+ */
+void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples
+ int16_t* ar_f_Q0, // Input samples
+ int16_t* cth_Q15, // Filter coefficients
+ int16_t* sth_Q15, // Filter coefficients
+ int16_t order_coef) { // order of the filter
+ int n = 0;
+
+ for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) {
+ int k = 0;
+ int16_t tmpAR = 0;
+ int32_t tmp32 = 0;
+ int32_t tmp32_2 = 0;
+
+ tmpAR = ar_f_Q0[n + 1];
+ for (k = order_coef - 1; k >= 0; k--) {
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(cth_Q15[k], tmpAR))
+ - (WEBRTC_SPL_MUL_16_16(sth_Q15[k], ar_g_Q0[k])) + 16384), 15);
+ tmp32_2 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sth_Q15[k], tmpAR))
+ + (WEBRTC_SPL_MUL_16_16(cth_Q15[k], ar_g_Q0[k])) + 16384), 15);
+ tmpAR = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32);
+ ar_g_Q0[k + 1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32_2);
+ }
+ ar_f_Q0[n + 1] = tmpAR;
+ ar_g_Q0[0] = tmpAR;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S
new file mode 100644
index 0000000..a59b6e3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S
@@ -0,0 +1,155 @@
+@
+@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS. All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ lattice_neon.s
+@
+@ Contains a function for the core loop in the normalized lattice MA
+@ filter routine for iSAC codec, optimized for ARM Neon platform.
+@ void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
+@ int16_t input1,
+@ int32_t input2,
+@ int32_t* ptr0,
+@ int32_t* ptr1,
+@ int32_t* __restrict ptr2);
+@ It calculates
+@ *ptr2 = input2 * (*ptr2) + input0 * (*ptr0));
+@ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
+@ in Q15 domain.
+@
+@ Reference code in lattice.c.
+@ Output is not bit-exact with the reference C code, due to the replacement
+@ of WEBRTC_SPL_MUL_16_32_RSFT15 and LATTICE_MUL_32_32_RSFT16 with Neon
+@ instructions, smulwb, and smull. Speech quality was not degraded by
+@ testing speech and tone vectors.
+
+.arch armv7-a
+.fpu neon
+
+#include "settings.h"
+
+.global WebRtcIsacfix_FilterMaLoopNeon
+
+.align 2
+
+WebRtcIsacfix_FilterMaLoopNeon:
+.fnstart
+
+.save {r4-r8}
+ push {r4-r8}
+
+ vdup.32 d28, r0 @ Initialize Neon register with input0
+ vdup.32 d29, r1 @ Initialize Neon register with input1
+ vdup.32 d30, r2 @ Initialize Neon register with input2
+ ldr r4, [sp, #20] @ ptr1
+ ldr r12, [sp, #24] @ ptr2
+
+ @ Number of loop iterations after unrolling: r5 = (HALF_SUBFRAMELEN - 1) >> 2
+ @ Leftover samples after the loop, in r6:
+ @ r6 = (HALF_SUBFRAMELEN - 1) - (HALF_SUBFRAMELEN - 1) >> 2 << 2
+ mov r6, #HALF_SUBFRAMELEN
+ sub r6, #1
+ lsr r5, r6, #2
+ sub r6, r5, lsl #2
+
+ @ First r5 iterations in a loop.
+
+LOOP:
+ vld1.32 {d0, d1}, [r3]! @ *ptr0
+
+ vmull.s32 q10, d0, d28 @ tmp32a = input0 * (*ptr0)
+ vmull.s32 q11, d1, d28 @ tmp32a = input0 * (*ptr0)
+ vmull.s32 q12, d0, d29 @ input1 * (*ptr0)
+ vmull.s32 q13, d1, d29 @ input1 * (*ptr0)
+
+ vrshrn.i64 d4, q10, #15
+ vrshrn.i64 d5, q11, #15
+
+ vld1.32 {d2, d3}, [r12] @ *ptr2
+ vadd.i32 q3, q2, q1 @ tmp32b = *ptr2 + tmp32a
+
+ vrshrn.i64 d0, q12, #15
+
+ vmull.s32 q10, d6, d30 @ input2 * (*ptr2 + tmp32b)
+ vmull.s32 q11, d7, d30 @ input2 * (*ptr2 + tmp32b)
+
+ vrshrn.i64 d16, q10, #16
+ vrshrn.i64 d17, q11, #16
+
+ vmull.s32 q10, d16, d28 @ input0 * (*ptr2)
+ vmull.s32 q11, d17, d28 @ input0 * (*ptr2)
+
+ vrshrn.i64 d1, q13, #15
+ vrshrn.i64 d18, q10, #15
+ vrshrn.i64 d19, q11, #15
+
+ vst1.32 {d16, d17}, [r12]! @ *ptr2
+
+ vadd.i32 q9, q0, q9
+ subs r5, #1
+ vst1.32 {d18, d19}, [r4]! @ *ptr1
+
+ bgt LOOP
+
+ @ Check how many samples still need to be processed.
+ subs r6, #2
+ blt LAST_SAMPLE
+
+ @ Process two more samples:
+ vld1.32 d0, [r3]! @ *ptr0
+
+ vmull.s32 q11, d0, d28 @ tmp32a = input0 * (*ptr0)
+ vmull.s32 q13, d0, d29 @ input1 * (*ptr0)
+
+ vld1.32 d18, [r12] @ *ptr2
+ vrshrn.i64 d4, q11, #15
+
+ vadd.i32 d7, d4, d18 @ tmp32b = *ptr2 + tmp32a
+ vmull.s32 q11, d7, d30 @ input2 * (*ptr2 + tmp32b)
+ vrshrn.i64 d16, q11, #16
+
+ vmull.s32 q11, d16, d28 @ input0 * (*ptr2)
+ vst1.32 d16, [r12]! @ *ptr2
+
+ vrshrn.i64 d0, q13, #15
+ vrshrn.i64 d19, q11, #15
+ vadd.i32 d19, d0, d19
+
+ vst1.32 d19, [r4]! @ *ptr1
+
+ @ If there's still one more sample, process it here.
+LAST_SAMPLE:
+ cmp r6, #1
+ bne END
+
+ @ *ptr2 = input2 * (*ptr2 + input0 * (*ptr0));
+
+ ldr r7, [r3] @ *ptr0
+ ldr r8, [r12] @ *ptr2
+
+ smulwb r5, r7, r0 @ tmp32a = *ptr0 * input0 >> 16
+ add r8, r8, r5, lsl #1 @ tmp32b = *ptr2 + (tmp32a << 1)
+ smull r5, r6, r8, r2 @ tmp32b * input2, in 64 bits
+ lsl r6, #16
+ add r6, r5, lsr #16 @ Only take the middle 32 bits
+ str r6, [r12] @ Output (*ptr2, as 32 bits)
+
+ @ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
+
+ smulwb r5, r7, r1 @ tmp32a = *ptr0 * input1 >> 16
+ smulwb r6, r6, r0 @ tmp32b = *ptr2 * input0 >> 16
+ lsl r5, r5, #1
+ add r5, r6, lsl #1
+ str r5, [r4] @ Output (*ptr1)
+
+END:
+ pop {r4-r8}
+ bx lr
+
+.fnend
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c
new file mode 100644
index 0000000..4fa8ebb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c
@@ -0,0 +1,1035 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_masking_model.c
+ *
+ * LPC analysis and filtering functions
+ *
+ */
+
+#include "codec.h"
+#include "entropy_coding.h"
+#include "settings.h"
+
+
+/* The conversion is implemented by the step-down algorithm */
+void WebRtcSpl_AToK_JSK(
+ WebRtc_Word16 *a16, /* Q11 */
+ WebRtc_Word16 useOrder,
+ WebRtc_Word16 *k16 /* Q15 */
+ )
+{
+ int m, k;
+ WebRtc_Word32 tmp32[MAX_AR_MODEL_ORDER];
+ WebRtc_Word32 tmp32b;
+ WebRtc_Word32 tmp_inv_denum32;
+ WebRtc_Word16 tmp_inv_denum16;
+
+ k16[useOrder-1]= WEBRTC_SPL_LSHIFT_W16(a16[useOrder], 4); //Q11<<4 => Q15
+
+ for (m=useOrder-1; m>0; m--) {
+ tmp_inv_denum32 = ((WebRtc_Word32) 1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]); // (1 - k^2) in Q30
+ tmp_inv_denum16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp_inv_denum32, 15); // (1 - k^2) in Q15
+
+ for (k=1; k<=m; k++) {
+ tmp32b = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16) -
+ WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
+
+ tmp32[k] = WebRtcSpl_DivW32W16(tmp32b, tmp_inv_denum16); //Q27/Q15 = Q12
+ }
+
+ for (k=1; k<m; k++) {
+ a16[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q12>>1 => Q11
+ }
+
+ tmp32[m] = WEBRTC_SPL_SAT(4092, tmp32[m], -4092);
+ k16[m-1] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmp32[m], 3); //Q12<<3 => Q15
+ }
+
+ return;
+}
+
+
+
+
+
+WebRtc_Word16 WebRtcSpl_LevinsonW32_JSK(
+ WebRtc_Word32 *R, /* (i) Autocorrelation of length >= order+1 */
+ WebRtc_Word16 *A, /* (o) A[0..order] LPC coefficients (Q11) */
+ WebRtc_Word16 *K, /* (o) K[0...order-1] Reflection coefficients (Q15) */
+ WebRtc_Word16 order /* (i) filter order */
+ ) {
+ WebRtc_Word16 i, j;
+ WebRtc_Word16 R_hi[LEVINSON_MAX_ORDER+1], R_low[LEVINSON_MAX_ORDER+1];
+ /* Aurocorr coefficients in high precision */
+ WebRtc_Word16 A_hi[LEVINSON_MAX_ORDER+1], A_low[LEVINSON_MAX_ORDER+1];
+ /* LPC coefficients in high precicion */
+ WebRtc_Word16 A_upd_hi[LEVINSON_MAX_ORDER+1], A_upd_low[LEVINSON_MAX_ORDER+1];
+ /* LPC coefficients for next iteration */
+ WebRtc_Word16 K_hi, K_low; /* reflection coefficient in high precision */
+ WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp; /* Prediction gain Alpha in high precision
+ and with scale factor */
+ WebRtc_Word16 tmp_hi, tmp_low;
+ WebRtc_Word32 temp1W32, temp2W32, temp3W32;
+ WebRtc_Word16 norm;
+
+ /* Normalize the autocorrelation R[0]...R[order+1] */
+
+ norm = WebRtcSpl_NormW32(R[0]);
+
+ for (i=order;i>=0;i--) {
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
+ /* Put R in hi and low format */
+ R_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
+ }
+
+ /* K = A[1] = -R[1] / R[0] */
+
+ temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16) +
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); /* R[1] in Q31 */
+ temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); /* abs R[1] */
+ temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); /* abs(R[1])/R[0] in Q31 */
+ /* Put back the sign on R[1] */
+ if (temp2W32 > 0) {
+ temp1W32 = -temp1W32;
+ }
+
+ /* Put K in hi and low format */
+ K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+ /* Store first reflection coefficient */
+ K[0] = K_hi;
+
+ temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); /* A[1] in Q27 */
+
+ /* Put A[1] in hi and low format */
+ A_hi[1] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
+
+ /* Alpha = R[0] * (1-K^2) */
+
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, K_low), 14) +
+ WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1); /* temp1W32 = k^2 in Q31 */
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */
+ temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; /* temp1W32 = (1 - K[0]*K[0]) in Q31 */
+
+ /* Store temp1W32 = 1 - K[0]*K[0] on hi and low format */
+ tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ /* Calculate Alpha in Q31 */
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low), 15) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi), 15) ), 1);
+
+ /* Normalize Alpha and put it in hi and low format */
+
+ Alpha_exp = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
+ Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+ /* Perform the iterative calculations in the
+ Levinson Durbin algorithm */
+
+ for (i=2; i<=order; i++)
+ {
+
+ /* ----
+ \
+ temp1W32 = R[i] + > R[j]*A[i-j]
+ /
+ ----
+ j=1..i-1
+ */
+
+ temp1W32 = 0;
+
+ for(j=1; j<i; j++) {
+ /* temp1W32 is in Q31 */
+ temp1W32 += (WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]), 1) +
+ WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]), 15) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]), 15) ), 1));
+ }
+
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
+ temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16) +
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
+
+ /* K = -temp1W32 / Alpha */
+ temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* abs(temp1W32) */
+ temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); /* abs(temp1W32)/Alpha */
+
+ /* Put the sign of temp1W32 back again */
+ if (temp1W32 > 0) {
+ temp3W32 = -temp3W32;
+ }
+
+ /* Use the Alpha shifts from earlier to denormalize */
+ norm = WebRtcSpl_NormW32(temp3W32);
+ if ((Alpha_exp <= norm)||(temp3W32==0)) {
+ temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
+ } else {
+ if (temp3W32 > 0)
+ {
+ temp3W32 = (WebRtc_Word32)0x7fffffffL;
+ } else
+ {
+ temp3W32 = (WebRtc_Word32)0x80000000L;
+ }
+ }
+
+ /* Put K on hi and low format */
+ K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+ K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+ /* Store Reflection coefficient in Q15 */
+ K[i-1] = K_hi;
+
+ /* Test for unstable filter. If unstable return 0 and let the
+ user decide what to do in that case
+ */
+
+ if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32740) {
+ return(-i); /* Unstable filter */
+ }
+
+ /*
+ Compute updated LPC coefficient: Anew[i]
+ Anew[j]= A[j] + K*A[i-j] for j=1..i-1
+ Anew[i]= K
+ */
+
+ for(j=1; j<i; j++)
+ {
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16) +
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1); /* temp1W32 = A[j] in Q27 */
+
+ temp1W32 += WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j]) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]), 15) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]), 15) ), 1); /* temp1W32 += K*A[i-j] in Q27 */
+
+ /* Put Anew in hi and low format */
+ A_upd_hi[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
+ }
+
+ temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4); /* temp3W32 = K in Q27 (Convert from Q31 to Q27) */
+
+ /* Store Anew in hi and low format */
+ A_upd_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+ A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
+
+ /* Alpha = Alpha * (1-K^2) */
+
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, K_low), 14) +
+ WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1); /* K*K in Q31 */
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */
+ temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; /* 1 - K*K in Q31 */
+
+ /* Convert 1- K^2 in hi and low format */
+ tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+ /* Calculate Alpha = Alpha * (1-K^2) in Q31 */
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low), 15) +
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi), 15)), 1);
+
+ /* Normalize Alpha and store it on hi and low format */
+
+ norm = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
+
+ Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+ Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+ /* Update the total nomalization of Alpha */
+ Alpha_exp = Alpha_exp + norm;
+
+ /* Update A[] */
+
+ for(j=1; j<=i; j++)
+ {
+ A_hi[j] =A_upd_hi[j];
+ A_low[j] =A_upd_low[j];
+ }
+ }
+
+ /*
+ Set A[0] to 1.0 and store the A[i] i=1...order in Q12
+ (Convert from Q27 and use rounding)
+ */
+
+ A[0] = 2048;
+
+ for(i=1; i<=order; i++) {
+ /* temp1W32 in Q27 */
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16) +
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
+ /* Round and store upper word */
+ A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32+(WebRtc_Word32)32768, 16);
+ }
+ return(1); /* Stable filters */
+}
+
+
+
+
+
+/* window */
+/* Matlab generation of floating point code:
+ * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid;
+ * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end
+ * All values are multiplyed with 2^21 in fixed point code.
+ */
+static const WebRtc_Word16 kWindowAutocorr[WINLEN] = {
+ 0, 0, 0, 0, 0, 1, 1, 2, 2, 3, 5, 6,
+ 8, 10, 12, 14, 17, 20, 24, 28, 33, 38, 43, 49,
+ 56, 63, 71, 79, 88, 98, 108, 119, 131, 143, 157, 171,
+ 186, 202, 219, 237, 256, 275, 296, 318, 341, 365, 390, 416,
+ 444, 472, 502, 533, 566, 600, 635, 671, 709, 748, 789, 831,
+ 875, 920, 967, 1015, 1065, 1116, 1170, 1224, 1281, 1339, 1399, 1461,
+ 1525, 1590, 1657, 1726, 1797, 1870, 1945, 2021, 2100, 2181, 2263, 2348,
+ 2434, 2523, 2614, 2706, 2801, 2898, 2997, 3099, 3202, 3307, 3415, 3525,
+ 3637, 3751, 3867, 3986, 4106, 4229, 4354, 4481, 4611, 4742, 4876, 5012,
+ 5150, 5291, 5433, 5578, 5725, 5874, 6025, 6178, 6333, 6490, 6650, 6811,
+ 6974, 7140, 7307, 7476, 7647, 7820, 7995, 8171, 8349, 8529, 8711, 8894,
+ 9079, 9265, 9453, 9642, 9833, 10024, 10217, 10412, 10607, 10803, 11000, 11199,
+ 11398, 11597, 11797, 11998, 12200, 12401, 12603, 12805, 13008, 13210, 13412, 13614,
+ 13815, 14016, 14216, 14416, 14615, 14813, 15009, 15205, 15399, 15591, 15782, 15971,
+ 16157, 16342, 16524, 16704, 16881, 17056, 17227, 17395, 17559, 17720, 17877, 18030,
+ 18179, 18323, 18462, 18597, 18727, 18851, 18970, 19082, 19189, 19290, 19384, 19471,
+ 19551, 19623, 19689, 19746, 19795, 19835, 19867, 19890, 19904, 19908, 19902, 19886,
+ 19860, 19823, 19775, 19715, 19644, 19561, 19465, 19357, 19237, 19102, 18955, 18793,
+ 18618, 18428, 18223, 18004, 17769, 17518, 17252, 16970, 16672, 16357, 16025, 15677,
+ 15311, 14929, 14529, 14111, 13677, 13225, 12755, 12268, 11764, 11243, 10706, 10152,
+ 9583, 8998, 8399, 7787, 7162, 6527, 5883, 5231, 4576, 3919, 3265, 2620,
+ 1990, 1386, 825, 333
+};
+
+
+/* By using a hearing threshold level in dB of -28 dB (higher value gives more noise),
+ the H_T_H (in float) can be calculated as:
+ H_T_H = pow(10.0, 0.05 * (-28.0)) = 0.039810717055350
+ In Q19, H_T_H becomes round(0.039810717055350*2^19) ~= 20872, i.e.
+ H_T_H = 20872/524288.0, and H_T_HQ19 = 20872;
+*/
+
+
+/* The bandwidth expansion vectors are created from:
+ kPolyVecLo=[0.900000,0.810000,0.729000,0.656100,0.590490,0.531441,0.478297,0.430467,0.387420,0.348678,0.313811,0.282430];
+ kPolyVecHi=[0.800000,0.640000,0.512000,0.409600,0.327680,0.262144];
+ round(kPolyVecLo*32768)
+ round(kPolyVecHi*32768)
+*/
+static const WebRtc_Word16 kPolyVecLo[12] = {
+ 29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425, 10283, 9255
+};
+static const WebRtc_Word16 kPolyVecHi[6] = {
+ 26214, 20972, 16777, 13422, 10737, 8590
+};
+
+static __inline WebRtc_Word32 log2_Q8_LPC( WebRtc_UWord32 x ) {
+
+ WebRtc_Word32 zeros, lg2;
+ WebRtc_Word16 frac;
+
+ zeros=WebRtcSpl_NormU32(x);
+ frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
+
+ /* log2(x) */
+
+ lg2= (WEBRTC_SPL_LSHIFT_W16((31-zeros), 8)+frac);
+ return lg2;
+
+}
+
+static const WebRtc_Word16 kMulPitchGain = -25; /* 200/256 in Q5 */
+static const WebRtc_Word16 kChngFactor = 3523; /* log10(2)*10/4*0.4/1.4=log10(2)/1.4= 0.2150 in Q14 */
+static const WebRtc_Word16 kExp2 = 11819; /* 1/log(2) */
+
+void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input, const WebRtc_Word16 *pitchGains_Q12,
+ WebRtc_UWord32 *oldEnergy, WebRtc_Word16 *varscale)
+{
+ int k;
+ WebRtc_UWord32 nrgQ[4];
+ WebRtc_Word16 nrgQlog[4];
+ WebRtc_Word16 tmp16, chng1, chng2, chng3, chng4, tmp, chngQ, oldNrgQlog, pgQ, pg3;
+ WebRtc_Word32 expPg32;
+ WebRtc_Word16 expPg, divVal;
+ WebRtc_Word16 tmp16_1, tmp16_2;
+
+ /* Calculate energies of first and second frame halfs */
+ nrgQ[0]=0;
+ for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES/4 + QLOOKAHEAD) / 2; k++) {
+ nrgQ[0] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
+ }
+ nrgQ[1]=0;
+ for ( ; k < (FRAMESAMPLES/2 + QLOOKAHEAD) / 2; k++) {
+ nrgQ[1] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
+ }
+ nrgQ[2]=0;
+ for ( ; k < (WEBRTC_SPL_MUL_16_16(FRAMESAMPLES, 3)/4 + QLOOKAHEAD) / 2; k++) {
+ nrgQ[2] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
+ }
+ nrgQ[3]=0;
+ for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) {
+ nrgQ[3] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
+ }
+
+ for ( k=0; k<4; k++) {
+ nrgQlog[k] = (WebRtc_Word16)log2_Q8_LPC(nrgQ[k]); /* log2(nrgQ) */
+ }
+ oldNrgQlog = (WebRtc_Word16)log2_Q8_LPC(*oldEnergy);
+
+ /* Calculate average level change */
+ chng1 = WEBRTC_SPL_ABS_W16(nrgQlog[3]-nrgQlog[2]);
+ chng2 = WEBRTC_SPL_ABS_W16(nrgQlog[2]-nrgQlog[1]);
+ chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]);
+ chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog);
+ tmp = chng1+chng2+chng3+chng4;
+ chngQ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
+ chngQ += 2926; /* + 1.0/1.4 in Q12 */
+
+ /* Find average pitch gain */
+ pgQ = 0;
+ for (k=0; k<4; k++)
+ {
+ pgQ += pitchGains_Q12[k];
+ }
+
+ pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
+ pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18 */
+ pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10 kMulPitchGain = -25 = -200 in Q-3. */
+
+ tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
+ if (tmp16<0) {
+ tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
+ tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
+ if (tmp16_1<0)
+ expPg=(WebRtc_Word16) -WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+ else
+ expPg=(WebRtc_Word16) -WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+ } else
+ expPg = (WebRtc_Word16) -16384; /* 1 in Q14, since 2^0=1 */
+
+ expPg32 = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W16((WebRtc_Word32)expPg, 8); /* Q22 */
+ divVal = WebRtcSpl_DivW32W16ResW16(expPg32, chngQ); /* Q22/Q12=Q10 */
+
+ tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,divVal,13);/* Q13*Q10>>13 => Q10*/
+ if (tmp16<0) {
+ tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
+ tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
+ if (tmp16_1<0)
+ expPg=(WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+ else
+ expPg=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+ } else
+ expPg = (WebRtc_Word16) 16384; /* 1 in Q14, since 2^0=1 */
+
+ *varscale = expPg-1;
+ *oldEnergy = nrgQ[3];
+}
+
+
+
+static __inline WebRtc_Word16 exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
+
+ WebRtc_Word16 tmp16_1, tmp16_2;
+
+ tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
+ tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+ if(tmp16_1>0)
+ return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+ else
+ return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+
+}
+
+
+// Declare a function pointer.
+AutocorrFix WebRtcIsacfix_AutocorrFix;
+
+void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
+ WebRtc_Word16 *inHiQ0,
+ MaskFiltstr_enc *maskdata,
+ WebRtc_Word16 snrQ10,
+ const WebRtc_Word16 *pitchGains_Q12,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *lo_coeffQ15,
+ WebRtc_Word16 *hi_coeffQ15)
+{
+ int k, n, j, ii;
+ WebRtc_Word16 pos1, pos2;
+ WebRtc_Word16 sh_lo, sh_hi, sh, ssh, shMem;
+ WebRtc_Word16 varscaleQ14;
+
+ WebRtc_Word16 tmpQQlo, tmpQQhi;
+ WebRtc_Word32 tmp32;
+ WebRtc_Word16 tmp16,tmp16b;
+
+ WebRtc_Word16 polyHI[ORDERHI+1];
+ WebRtc_Word16 rcQ15_lo[ORDERLO], rcQ15_hi[ORDERHI];
+
+
+ WebRtc_Word16 DataLoQ6[WINLEN], DataHiQ6[WINLEN];
+ WebRtc_Word32 corrloQQ[ORDERLO+2];
+ WebRtc_Word32 corrhiQQ[ORDERHI+1];
+ WebRtc_Word32 corrlo2QQ[ORDERLO+1];
+ WebRtc_Word16 scale;
+ WebRtc_Word16 QdomLO, QdomHI, newQdomHI, newQdomLO;
+
+ WebRtc_Word32 round;
+ WebRtc_Word32 res_nrgQQ;
+ WebRtc_Word32 sqrt_nrg;
+
+ WebRtc_Word32 aSQR32;
+
+ /* less-noise-at-low-frequencies factor */
+ WebRtc_Word16 aaQ14;
+
+ /* Multiplication with 1/sqrt(12) ~= 0.28901734104046 can be done by convertion to
+ Q15, i.e. round(0.28901734104046*32768) = 9471, and use 9471/32768.0 ~= 0.289032
+ */
+ WebRtc_Word16 snrq, shft;
+
+ WebRtc_Word16 tmp16a;
+ WebRtc_Word32 tmp32a, tmp32b, tmp32c;
+
+ WebRtc_Word16 a_LOQ11[ORDERLO+1];
+ WebRtc_Word16 k_vecloQ15[ORDERLO];
+ WebRtc_Word16 a_HIQ12[ORDERHI+1];
+ WebRtc_Word16 k_vechiQ15[ORDERHI];
+
+ WebRtc_Word16 stab;
+
+ snrq=snrQ10;
+
+ /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/
+ tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
+ tmp16b = exp2_Q10_T(tmp16); // Q10
+ snrq = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
+
+ /* change quallevel depending on pitch gains and level fluctuations */
+ WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14);
+
+ /* less-noise-at-low-frequencies factor */
+ /* Calculation of 0.35 * (0.5 + 0.5 * varscale) in fixpoint:
+ With 0.35 in Q16 (0.35 ~= 22938/65536.0 = 0.3500061) and varscaleQ14 in Q14,
+ we get Q16*Q14>>16 = Q14
+ */
+ aaQ14 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(22938, (8192 + WEBRTC_SPL_RSHIFT_W32(varscaleQ14, 1)))
+ + ((WebRtc_Word32)32768)), 16);
+
+ /* Calculate tmp = (1.0 + aa*aa); in Q12 */
+ tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
+ tmpQQlo = 4096 + WEBRTC_SPL_RSHIFT_W16(tmp16, 1); // Q12 + Q13>>1 = Q12
+
+ /* Calculate tmp = (1.0+aa) * (1.0+aa); */
+ tmp16 = 8192 + WEBRTC_SPL_RSHIFT_W16(aaQ14, 1); // 1+a in Q13
+ tmpQQhi = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
+
+ /* replace data in buffer by new look-ahead data */
+ for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) {
+ maskdata->DataBufferLoQ0[pos1 + WINLEN - QLOOKAHEAD] = inLoQ0[pos1];
+ }
+
+ for (k = 0; k < SUBFRAMES; k++) {
+
+ /* Update input buffer and multiply signal with window */
+ for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
+ maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2];
+ maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2];
+ DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+ maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+ DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+ maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+ }
+ pos2 = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(k, UPDATE)/2);
+ for (n = 0; n < UPDATE/2; n++, pos1++) {
+ maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2];
+ maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++];
+ DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+ maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+ DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
+ maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+ }
+
+ /* Get correlation coefficients */
+ /* The highest absolute value measured inside DataLo in the test set
+ For DataHi, corresponding value was 160.
+
+ This means that it should be possible to represent the input values
+ to WebRtcSpl_AutoCorrelation() as Q6 values (since 307*2^6 =
+ 19648). Of course, Q0 will also work, but due to the low energy in
+ DataLo and DataHi, the outputted autocorrelation will be more accurate
+ and mimic the floating point code better, by being in an high as possible
+ Q-domain.
+ */
+
+ WebRtcIsacfix_AutocorrFix(corrloQQ,DataLoQ6,WINLEN, ORDERLO+1, &scale);
+ QdomLO = 12-scale; // QdomLO is the Q-domain of corrloQQ
+ sh_lo = WebRtcSpl_NormW32(corrloQQ[0]);
+ QdomLO += sh_lo;
+ for (ii=0; ii<ORDERLO+2; ii++) {
+ corrloQQ[ii] = WEBRTC_SPL_LSHIFT_W32(corrloQQ[ii], sh_lo);
+ }
+ /* It is investigated whether it was possible to use 16 bits for the
+ 32-bit vector corrloQQ, but it didn't work. */
+
+ WebRtcIsacfix_AutocorrFix(corrhiQQ,DataHiQ6,WINLEN, ORDERHI, &scale);
+
+ QdomHI = 12-scale; // QdomHI is the Q-domain of corrhiQQ
+ sh_hi = WebRtcSpl_NormW32(corrhiQQ[0]);
+ QdomHI += sh_hi;
+ for (ii=0; ii<ORDERHI+1; ii++) {
+ corrhiQQ[ii] = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[ii], sh_hi);
+ }
+
+ /* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */
+
+ /* Calculate corrlo2[0] = tmpQQlo * corrlo[0] - 2.0*tmpQQlo * corrlo[1];*/
+ corrlo2QQ[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[0]), 1)- // Q(12+QdomLO-16)>>1 = Q(QdomLO-5)
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, corrloQQ[1]), 2); // 2*Q(14+QdomLO-16)>>3 = Q(QdomLO-2)>>2 = Q(QdomLO-5)
+
+ /* Calculate corrlo2[n] = tmpQQlo * corrlo[n] - tmpQQlo * (corrlo[n-1] + corrlo[n+1]);*/
+ for (n = 1; n <= ORDERLO; n++) {
+
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n-1], 1) + WEBRTC_SPL_RSHIFT_W32(corrloQQ[n+1], 1); // Q(QdomLO-1)
+ corrlo2QQ[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[n]), 1)- // Q(12+QdomLO-16)>>1 = Q(QdomLO-5)
+ WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, tmp32), 2); // Q(14+QdomLO-1-16)>>2 = Q(QdomLO-3)>>2 = Q(QdomLO-5)
+
+ }
+ QdomLO -= 5;
+
+ /* Calculate corrhi[n] = tmpQQhi * corrhi[n]; */
+ for (n = 0; n <= ORDERHI; n++) {
+ corrhiQQ[n] = WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQhi, corrhiQQ[n]); // Q(12+QdomHI-16) = Q(QdomHI-4)
+ }
+ QdomHI -= 4;
+
+ /* add white noise floor */
+ /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) */
+ /* Calculate corrlo2[0] += 9.5367431640625e-7; and
+ corrhi[0] += 9.5367431640625e-7, where the constant is 1/2^20 */
+
+ tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomLO-20);
+ corrlo2QQ[0] += tmp32;
+ tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomHI-20);
+ corrhiQQ[0] += tmp32;
+
+ /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) before the following
+ code segment, where we want to make sure we get a 1-bit margin */
+ for (n = 0; n <= ORDERLO; n++) {
+ corrlo2QQ[n] = WEBRTC_SPL_RSHIFT_W32(corrlo2QQ[n], 1); // Make sure we have a 1-bit margin
+ }
+ QdomLO -= 1; // Now, corrlo2QQ is in Q(QdomLO), with a 1-bit margin
+
+ for (n = 0; n <= ORDERHI; n++) {
+ corrhiQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], 1); // Make sure we have a 1-bit margin
+ }
+ QdomHI -= 1; // Now, corrhiQQ is in Q(QdomHI), with a 1-bit margin
+
+
+ newQdomLO = QdomLO;
+
+ for (n = 0; n <= ORDERLO; n++) {
+ WebRtc_Word32 tmp, tmpB, tmpCorr;
+ WebRtc_Word16 alpha=328; //0.01 in Q15
+ WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
+ WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q15
+
+ if (maskdata->CorrBufLoQQ[n] != 0) {
+ shMem=WebRtcSpl_NormW32(maskdata->CorrBufLoQQ[n]);
+ sh = QdomLO - maskdata->CorrBufLoQdom[n];
+ if (sh<=shMem) {
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], sh); // Get CorrBufLoQQ to same domain as corrlo2
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp);
+ } else if ((sh-shMem)<7){
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufLoQQ as much as possible
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, (sh-shMem)), tmp); // Shift alpha the number of times required to get tmp in QdomLO
+ } else {
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, 6), tmp); // Shift alpha as much as possible without overflow the number of times required to get tmp in QdomHI
+ tmpCorr = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n], sh-shMem-6);
+ tmp = tmp + tmpCorr;
+ maskdata->CorrBufLoQQ[n] = tmp;
+ newQdomLO = QdomLO-(sh-shMem-6);
+ maskdata->CorrBufLoQdom[n] = newQdomLO;
+ }
+ } else
+ tmp = 0;
+
+ tmp = tmp + corrlo2QQ[n];
+
+ maskdata->CorrBufLoQQ[n] = tmp;
+ maskdata->CorrBufLoQdom[n] = QdomLO;
+
+ tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp);
+ tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, corrlo2QQ[n]);
+ corrlo2QQ[n] = tmp + tmpB;
+ }
+ if( newQdomLO!=QdomLO) {
+ for (n = 0; n <= ORDERLO; n++) {
+ if (maskdata->CorrBufLoQdom[n] != newQdomLO)
+ corrloQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n], maskdata->CorrBufLoQdom[n]-newQdomLO);
+ }
+ QdomLO = newQdomLO;
+ }
+
+
+ newQdomHI = QdomHI;
+
+ for (n = 0; n <= ORDERHI; n++) {
+ WebRtc_Word32 tmp, tmpB, tmpCorr;
+ WebRtc_Word16 alpha=328; //0.01 in Q15
+ WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
+ WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q1
+ if (maskdata->CorrBufHiQQ[n] != 0) {
+ shMem=WebRtcSpl_NormW32(maskdata->CorrBufHiQQ[n]);
+ sh = QdomHI - maskdata->CorrBufHiQdom[n];
+ if (sh<=shMem) {
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], sh); // Get CorrBufHiQQ to same domain as corrhi
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp);
+ tmpCorr = corrhiQQ[n];
+ tmp = tmp + tmpCorr;
+ maskdata->CorrBufHiQQ[n] = tmp;
+ maskdata->CorrBufHiQdom[n] = QdomHI;
+ } else if ((sh-shMem)<7) {
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, (sh-shMem)), tmp); // Shift alpha the number of times required to get tmp in QdomHI
+ tmpCorr = corrhiQQ[n];
+ tmp = tmp + tmpCorr;
+ maskdata->CorrBufHiQQ[n] = tmp;
+ maskdata->CorrBufHiQdom[n] = QdomHI;
+ } else {
+ tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
+ tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, 6), tmp); // Shift alpha as much as possible without overflow the number of times required to get tmp in QdomHI
+ tmpCorr = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], sh-shMem-6);
+ tmp = tmp + tmpCorr;
+ maskdata->CorrBufHiQQ[n] = tmp;
+ newQdomHI = QdomHI-(sh-shMem-6);
+ maskdata->CorrBufHiQdom[n] = newQdomHI;
+ }
+ } else {
+ tmp = corrhiQQ[n];
+ tmpCorr = tmp;
+ maskdata->CorrBufHiQQ[n] = tmp;
+ maskdata->CorrBufHiQdom[n] = QdomHI;
+ }
+
+ tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp);
+ tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, tmpCorr);
+ corrhiQQ[n] = tmp + tmpB;
+ }
+
+ if( newQdomHI!=QdomHI) {
+ for (n = 0; n <= ORDERHI; n++) {
+ if (maskdata->CorrBufHiQdom[n] != newQdomHI)
+ corrhiQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], maskdata->CorrBufHiQdom[n]-newQdomHI);
+ }
+ QdomHI = newQdomHI;
+ }
+
+ stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, ORDERLO);
+
+ if (stab<0) { // If unstable use lower order
+ a_LOQ11[0]=2048;
+ for (n = 1; n <= ORDERLO; n++) {
+ a_LOQ11[n]=0;
+ }
+
+ stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, 8);
+ }
+
+
+ WebRtcSpl_LevinsonDurbin(corrhiQQ, a_HIQ12, k_vechiQ15, ORDERHI);
+
+ /* bandwidth expansion */
+ for (n = 1; n <= ORDERLO; n++) {
+ a_LOQ11[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]);
+ }
+
+
+ polyHI[0] = a_HIQ12[0];
+ for (n = 1; n <= ORDERHI; n++) {
+ a_HIQ12[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]);
+ polyHI[n] = a_HIQ12[n];
+ }
+
+ /* Normalize the corrlo2 vector */
+ sh = WebRtcSpl_NormW32(corrlo2QQ[0]);
+ for (n = 0; n <= ORDERLO; n++) {
+ corrlo2QQ[n] = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[n], sh);
+ }
+ QdomLO += sh; /* Now, corrlo2QQ is still in Q(QdomLO) */
+
+
+ /* residual energy */
+
+ sh_lo = 31;
+ res_nrgQQ = 0;
+ for (j = 0; j <= ORDERLO; j++)
+ {
+ for (n = 0; n < j; n++)
+ {
+ WebRtc_Word16 index, diff, sh_corr;
+
+ index = j - n; //WEBRTC_SPL_ABS_W16(j-n);
+
+ /* Calculation of res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n]; */
+ /* corrlo2QQ is in Q(QdomLO) */
+ tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_LOQ11[j], a_LOQ11[n])); // Q11*Q11 = Q22
+ // multiply by 2 as loop only on half of the matrix. a_LOQ11 gone through bandwidth
+ // expation so the following shift is safe.
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(tmp32, 1);
+ sh = WebRtcSpl_NormW32(tmp32);
+ aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(22+sh)
+ sh_corr = WebRtcSpl_NormW32(corrlo2QQ[index]);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[index], sh_corr);
+ tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(22+sh)*Q(QdomLO+sh_corr)>>32 = Q(22+sh+QdomLO+sh_corr-32) = Q(sh+QdomLO+sh_corr-10)
+ sh = sh+QdomLO+sh_corr-10;
+
+ diff = sh_lo-sh;
+
+ round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
+ if (diff==0)
+ round = 0;
+ if (diff>=31) {
+ res_nrgQQ = tmp32;
+ sh_lo = sh;
+ } else if (diff>0) {
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
+ sh_lo = sh-1;
+ } else if (diff>-31){
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
+ sh_lo = sh_lo-1;
+ }
+ sh = WebRtcSpl_NormW32(res_nrgQQ);
+ res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
+ sh_lo += sh;
+ }
+ n = j;
+ {
+ WebRtc_Word16 index, diff, sh_corr;
+
+ index = 0; //WEBRTC_SPL_ABS_W16(j-n);
+
+ /* Calculation of res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n]; */
+ /* corrlo2QQ is in Q(QdomLO) */
+ tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_LOQ11[j], a_LOQ11[n]); // Q11*Q11 = Q22
+ sh = WebRtcSpl_NormW32(tmp32);
+ aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(22+sh)
+ sh_corr = WebRtcSpl_NormW32(corrlo2QQ[index]);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[index], sh_corr);
+ tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(22+sh)*Q(QdomLO+sh_corr)>>32 = Q(22+sh+QdomLO+sh_corr-32) = Q(sh+QdomLO+sh_corr-10)
+ sh = sh+QdomLO+sh_corr-10;
+ diff = sh_lo-sh;
+
+ round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
+ if (diff==0)
+ round = 0;
+ if (diff>=31) {
+ res_nrgQQ = tmp32;
+ sh_lo = sh;
+ } else if (diff>0) {
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
+ sh_lo = sh-1;
+ } else if (diff>-31){
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
+ sh_lo = sh_lo-1;
+ }
+ sh = WebRtcSpl_NormW32(res_nrgQQ);
+ res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
+ sh_lo += sh;
+ }
+ }
+ /* Convert to reflection coefficients */
+
+
+ WebRtcSpl_AToK_JSK(a_LOQ11, ORDERLO, rcQ15_lo);
+
+ if (sh_lo & 0x0001) {
+ res_nrgQQ=WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1);
+ sh_lo-=1;
+ }
+
+
+ if( res_nrgQQ > 0 )
+ {
+ sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ);
+
+ /* add hearing threshold and compute the gain */
+ /* lo_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
+
+
+ //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17); // Q14
+ tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1); // H_T_HQ19=65536 (16-17=-1) ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1); // sqrt_nrg is in Qssh
+ ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1); // sqrt_nrg is in Qssh
+ sh = ssh - 14;
+ tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
+ tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator)
+ tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0); //Q24 (numerator)
+
+ sh = WebRtcSpl_NormW32(tmp32c);
+ shft = 16 - sh;
+ tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator)
+
+ tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
+ sh = ssh-shft-7;
+ *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17
+ }
+ else
+ {
+ *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17); // Gains in Q17
+ }
+ gain_lo_hiQ17++;
+
+ /* copy coefficients to output array */
+ for (n = 0; n < ORDERLO; n++) {
+ *lo_coeffQ15 = (WebRtc_Word16) (rcQ15_lo[n]);
+ lo_coeffQ15++;
+ }
+ /* residual energy */
+ res_nrgQQ = 0;
+ sh_hi = 31;
+
+
+ for (j = 0; j <= ORDERHI; j++)
+ {
+ for (n = 0; n < j; n++)
+ {
+ WebRtc_Word16 index, diff, sh_corr;
+
+ index = j-n; //WEBRTC_SPL_ABS_W16(j-n);
+
+ /* Calculation of res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n] * 2; for j != n */
+ /* corrhiQQ is in Q(QdomHI) */
+ tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_HIQ12[j], a_HIQ12[n])); // Q12*Q12 = Q24
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(tmp32, 1);
+ sh = WebRtcSpl_NormW32(tmp32);
+ aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(24+sh)
+ sh_corr = WebRtcSpl_NormW32(corrhiQQ[index]);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[index],sh_corr);
+ tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(24+sh)*Q(QdomHI+sh_corr)>>32 = Q(24+sh+QdomHI+sh_corr-32) = Q(sh+QdomHI+sh_corr-8)
+ sh = sh+QdomHI+sh_corr-8;
+ diff = sh_hi-sh;
+
+ round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
+ if (diff==0)
+ round = 0;
+ if (diff>=31) {
+ res_nrgQQ = tmp32;
+ sh_hi = sh;
+ } else if (diff>0) {
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
+ sh_hi = sh-1;
+ } else if (diff>-31){
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
+ sh_hi = sh_hi-1;
+ }
+
+ sh = WebRtcSpl_NormW32(res_nrgQQ);
+ res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
+ sh_hi += sh;
+ }
+
+ n = j;
+ {
+ WebRtc_Word16 index, diff, sh_corr;
+
+ index = 0; //n-j; //WEBRTC_SPL_ABS_W16(j-n);
+
+ /* Calculation of res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n];*/
+ /* corrhiQQ is in Q(QdomHI) */
+ tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_HIQ12[j], a_HIQ12[n])); // Q12*Q12 = Q24
+ sh = WebRtcSpl_NormW32(tmp32);
+ aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(24+sh)
+ sh_corr = WebRtcSpl_NormW32(corrhiQQ[index]);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[index],sh_corr);
+ tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(24+sh)*Q(QdomHI+sh_corr)>>32 = Q(24+sh+QdomHI+sh_corr-32) = Q(sh+QdomHI+sh_corr-8)
+ sh = sh+QdomHI+sh_corr-8;
+ diff = sh_hi-sh;
+
+ round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
+ if (diff==0)
+ round = 0;
+ if (diff>=31) {
+ res_nrgQQ = tmp32;
+ sh_hi = sh;
+ } else if (diff>0) {
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
+ sh_hi = sh-1;
+ } else if (diff>-31){
+ res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
+ sh_hi = sh_hi-1;
+ }
+
+ sh = WebRtcSpl_NormW32(res_nrgQQ);
+ res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
+ sh_hi += sh;
+ }
+ }
+
+ /* Convert to reflection coefficients */
+ WebRtcSpl_LpcToReflCoef(polyHI, ORDERHI, rcQ15_hi);
+
+ if (sh_hi & 0x0001) {
+ res_nrgQQ=WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1);
+ sh_hi-=1;
+ }
+
+
+ if( res_nrgQQ > 0 )
+ {
+ sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ);
+
+
+ /* add hearing threshold and compute the gain */
+ /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
+
+ //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17); // Q14
+ tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1); // H_T_HQ19=65536 (16-17=-1)
+
+ ssh= WEBRTC_SPL_RSHIFT_W32(sh_hi, 1); // sqrt_nrg is in Qssh
+ sh = ssh - 14;
+ tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
+ tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator)
+ tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0); //Q24 (numerator)
+
+ sh = WebRtcSpl_NormW32(tmp32c);
+ shft = 16 - sh;
+ tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator)
+
+ tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
+ sh = ssh-shft-7;
+ *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17
+ }
+ else
+ {
+ *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17); // Gains in Q17
+ }
+ gain_lo_hiQ17++;
+
+
+ /* copy coefficients to output array */
+ for (n = 0; n < ORDERHI; n++) {
+ *hi_coeffQ15 = rcQ15_hi[n];
+ hi_coeffQ15++;
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h
new file mode 100644
index 0000000..9a64844
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_masking_model.h
+ *
+ * LPC functions
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
+
+#include "structs.h"
+
+void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input,
+ const WebRtc_Word16 *pitchGains_Q12,
+ WebRtc_UWord32 *oldEnergy,
+ WebRtc_Word16 *varscale);
+
+void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
+ WebRtc_Word16 *inHiQ0,
+ MaskFiltstr_enc *maskdata,
+ WebRtc_Word16 snrQ10,
+ const WebRtc_Word16 *pitchGains_Q12,
+ WebRtc_Word32 *gain_lo_hiQ17,
+ WebRtc_Word16 *lo_coeffQ15,
+ WebRtc_Word16 *hi_coeffQ15);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c
new file mode 100644
index 0000000..90cc9af
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c
@@ -0,0 +1,1280 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_tables.c
+ *
+ * Coding tables for the KLT coefficients
+ *
+ */
+
+
+#include "settings.h"
+#include "lpc_tables.h"
+
+/* indices of KLT coefficients used */
+const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 10, 11};
+
+const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 10, 11, 12, 13, 14, 15, 16, 17, 18, 19,
+ 20, 21, 22, 23, 24, 25, 26, 27, 28, 29,
+ 30, 31, 32, 33, 34, 35, 36, 37, 38, 39,
+ 40, 41, 42, 43, 44, 45, 46, 47, 48, 49,
+ 50, 51, 52, 53, 54, 55, 56, 57, 58, 59,
+ 60, 61, 62, 63, 64, 65, 66, 67, 68, 69,
+ 70, 71, 72, 73, 74, 75, 76, 77, 78, 79,
+ 80, 81, 82, 83, 84, 85, 86, 87, 88, 89,
+ 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,
+ 100, 101, 102, 103, 104, 105, 106, 107
+};
+
+/* cdf array for model indicator */
+const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[4] = {
+ 0, 15434, 37548, 65535
+};
+
+/* pointer to cdf array for model indicator */
+const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1] = {
+ WebRtcIsacfix_kModelCdf
+};
+
+/* initial cdf index for decoder of model indicator */
+const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1] = {
+ 1
+};
+
+/* offset to go from rounded value to quantization index */
+const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12] ={
+ 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 1, 1, 1, 1, 2, 2, 2, 3, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 2, 2, 3, 0, 0, 0, 0,
+ 1, 0, 1, 1, 1, 1, 1, 1, 1, 2,
+ 2, 4, 3, 5, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 2, 1, 2, 2, 3, 4,
+ 4, 7, 0, 0, 1, 1, 1, 1, 1, 1,
+ 1, 2, 3, 2, 3, 4, 4, 5, 7, 13,
+ 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
+ 5, 6, 7, 11, 9, 13, 12, 26
+};
+
+/* maximum quantization index */
+const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12] = {
+ 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 2, 2, 2, 2, 4, 4, 5, 6, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 2, 2,
+ 2, 2, 3, 4, 5, 7, 0, 0, 0, 0,
+ 2, 0, 2, 2, 2, 2, 3, 2, 2, 4,
+ 4, 6, 6, 9, 0, 0, 0, 0, 2, 2,
+ 2, 2, 2, 2, 3, 2, 4, 4, 7, 7,
+ 9, 13, 0, 0, 2, 2, 2, 2, 2, 2,
+ 3, 4, 5, 4, 6, 8, 8, 10, 16, 25,
+ 0, 2, 2, 4, 5, 4, 4, 4, 7, 8,
+ 9, 10, 13, 19, 17, 23, 25, 49
+};
+
+/* index offset */
+const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[3][12] = {
+ { 0, 7, 20, 29, 44, 55, 75, 88, 120, 143, 200, 253},
+ { 0, 7, 19, 27, 42, 53, 73, 86, 117, 140, 197, 249},
+ { 0, 7, 20, 28, 44, 55, 75, 89, 121, 145, 202, 257}
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[3][108] = {
+ {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 11, 14, 17, 20, 23, 28, 33, 39, 46, 47,
+ 48, 49, 50, 52, 53, 54, 55, 56, 58, 61,
+ 64, 67, 70, 74, 79, 85, 93, 94, 95, 96,
+ 97, 100, 101, 104, 107, 110, 113, 117, 120, 123,
+ 128, 133, 140, 147, 157, 158, 159, 160, 161, 164,
+ 167, 170, 173, 176, 179, 183, 186, 191, 196, 204,
+ 212, 222, 236, 237, 238, 241, 244, 247, 250, 253,
+ 256, 260, 265, 271, 276, 283, 292, 301, 312, 329,
+ 355, 356, 359, 362, 367, 373, 378, 383, 388, 396,
+ 405, 415, 426, 440, 460, 478, 502, 528
+ },
+ {
+ 0, 1, 2, 3, 4, 6, 7, 8, 9, 11,
+ 13, 16, 19, 22, 26, 29, 34, 39, 45, 46,
+ 47, 48, 49, 50, 51, 52, 53, 55, 57, 60,
+ 63, 66, 70, 73, 78, 84, 91, 92, 93, 94,
+ 95, 96, 97, 99, 102, 105, 108, 111, 114, 118,
+ 123, 128, 134, 141, 151, 152, 153, 154, 156, 159,
+ 162, 165, 168, 171, 174, 177, 181, 186, 194, 200,
+ 208, 218, 233, 234, 235, 236, 239, 242, 245, 248,
+ 251, 254, 258, 263, 270, 277, 288, 297, 308, 324,
+ 349, 351, 354, 357, 361, 366, 372, 378, 383, 390,
+ 398, 407, 420, 431, 450, 472, 496, 524
+ },
+ {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 11,
+ 14, 17, 20, 23, 26, 29, 34, 40, 47, 48,
+ 49, 50, 51, 52, 53, 54, 55, 58, 61, 64,
+ 67, 70, 73, 77, 82, 88, 96, 97, 98, 99,
+ 101, 102, 104, 107, 110, 113, 116, 119, 122, 125,
+ 129, 134, 141, 150, 160, 161, 162, 163, 166, 168,
+ 171, 174, 177, 180, 183, 186, 190, 195, 201, 208,
+ 216, 226, 243, 244, 245, 248, 251, 254, 257, 260,
+ 263, 268, 273, 278, 284, 291, 299, 310, 323, 340,
+ 366, 368, 371, 374, 379, 383, 389, 394, 399, 406,
+ 414, 422, 433, 445, 461, 480, 505, 533
+ }
+};
+
+/* initial cdf index for KLT coefficients */
+const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[3][12] = {
+ { 3, 6, 4, 7, 5, 10, 6, 16, 11, 28, 26, 69},
+ { 3, 6, 4, 7, 5, 10, 6, 15, 11, 28, 26, 69},
+ { 3, 6, 4, 8, 5, 10, 7, 16, 12, 28, 27, 70}
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[3][108] = {
+ {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 1, 1, 1, 1, 2, 2, 3, 3, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
+ 1, 1, 2, 2, 3, 4, 0, 0, 0, 0,
+ 1, 0, 1, 1, 1, 1, 2, 1, 1, 2,
+ 2, 3, 3, 5, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 2, 1, 2, 2, 4, 4,
+ 5, 7, 0, 0, 1, 1, 1, 1, 1, 1,
+ 2, 2, 3, 2, 3, 4, 4, 5, 8, 13,
+ 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
+ 5, 5, 7, 10, 9, 12, 13, 25
+ },
+ {
+ 0, 0, 0, 0, 1, 0, 0, 0, 1, 1,
+ 1, 1, 1, 2, 1, 2, 2, 3, 0, 0,
+ 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
+ 1, 2, 1, 2, 3, 3, 0, 0, 0, 0,
+ 0, 0, 1, 1, 1, 1, 1, 1, 2, 2,
+ 2, 3, 3, 5, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 1, 1, 2, 2, 4, 3, 4,
+ 5, 7, 0, 0, 0, 1, 1, 1, 1, 1,
+ 1, 2, 2, 3, 3, 5, 4, 5, 8, 12,
+ 1, 1, 1, 2, 2, 3, 3, 2, 3, 4,
+ 4, 6, 5, 9, 11, 12, 14, 25
+ },
+ {
+ 0, 0, 0, 0, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 1, 2, 3, 3, 0, 0,
+ 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
+ 1, 1, 2, 2, 3, 4, 0, 0, 0, 1,
+ 0, 1, 1, 1, 1, 1, 1, 1, 1, 2,
+ 2, 3, 4, 5, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 1, 1, 2, 2, 3, 3, 4,
+ 5, 8, 0, 0, 1, 1, 1, 1, 1, 1,
+ 2, 2, 2, 3, 3, 4, 5, 6, 8, 13,
+ 1, 1, 1, 2, 2, 3, 2, 2, 3, 4,
+ 4, 5, 6, 8, 9, 12, 14, 25
+ }
+};
+
+/* offsets for quantizer representation levels*/
+const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3] = {
+ 0, 392, 779
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3] = {
+ 0, 578, 1152
+};
+
+/* quantizer representation levels */
+
+
+
+const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176] = {
+ -364547,-231664,-102123,-573,104807,238257,368823,-758583,-640135,-510291
+ ,-377662,-252785,-113177,2627,112906,248601,389461,522691,644517,763974
+ ,-538963,-368179,-245823,-106095,-890,104299,241111,350730,493190,-800763
+ ,-646230,-510239,-382115,-248623,-111829,-2983,113852,251105,388114,519757
+ ,644048,774712,896334,1057931,-770009,-635310,-503690,-375087,-248106,-108525
+ ,-105,108259,243472,377948,519271,-1160885,-1032044,-914636,-777593,-647891
+ ,-518408,-388028,-254321,-115293,-598,117849,251296,385367,515510,652727
+ ,777432,920363,1038625,1153637,1316836,-632287,-505430,-379430,-248458,-118013
+ ,-888,118762,250266,381650,513327,652169,766471,932113,-2107480,-1971030
+ ,-1827020,-1698263,-1558670,-1436314,-1305377,-1172252,-1047355,-914202,-779651,-651001
+ ,-520999,-390394,-255761,-123490,-1893,126839,256703,385493,518607,651760
+ ,782750,908693,1044131,1163865,1311066,1424177,1582628,1709823,1831740,1955391
+ ,-1423044,-1288917,-1181281,-1043222,-911770,-780354,-646799,-522664,-386721,-258266
+ ,-128060,-1101,128233,259996,390336,519590,649290,778701,908010,1040796
+ ,1161235,1306889,1441882,-4446002,-4301031,-4194304,-4080591,-3947740,-3808975,-3686530
+ ,-3567839,-3383251,-3287089,-3136577,-3017405,-2869860,-2751321,-2619984,-2482932,-2354790
+ ,-2223147,-2090669,-1964135,-1831208,-1706697,-1570817,-1446008,-1305386,-1175773,-1046066
+ ,-915356,-785120,-653614,-524331,-393767,-260442,-130187,-799,128841,261466
+ ,393616,520542,652117,784613,914159,1045399,1181072,1308971,1442502,1570346
+ ,1693912,1843986,1966014,2090474,2224869,2364593,2475934,2628403,2752512,2856640
+ ,-4192441,-4063232,-3917821,-3799195,-3666233,-3519199,-3411021,-3269192,-3135684,-3008826
+ ,-2880875,-2747342,-2620981,-2494872,-2354979,-2229718,-2098939,-1964971,-1835399,-1703452
+ ,-1572806,-1440482,-1311794,-1179338,-1046521,-919823,-785914,-655335,-523416,-395507
+ ,-264833,-132184,-2546,131698,256217,391372,522688,651248,789964,909618
+ ,1035305,1179145,1313824,1436934,1552353,1693722,1815508,1972826,2096328,2228224
+ ,2359296,2490368,2598848,-6160384,-6029312,-5881382,-5767168,-5636096,-5505024,-5373952
+ ,-5228418,-5110384,-4954923,-4880576,-4710990,-4587364,-4471340,-4333905,-4211513,-4051293
+ ,-3907927,-3800105,-3675961,-3538640,-3413663,-3271148,-3152105,-3019103,-2869647,-2744015
+ ,-2620639,-2479385,-2364211,-2227611,-2095427,-1974497,-1834168,-1703561,-1568961,-1439826
+ ,-1309192,-1174050,-1050191,-917836,-786015,-656943,-518934,-394831,-257708,-128041
+ ,1610,128991,264442,393977,521383,653849,788164,918641,1049122,1181971
+ ,1308934,1439505,1571808,1706305,1836318,1966235,2097269,2228990,2357005,2490292
+ ,2617400,2749680,2881234,3014880,3145637,3276467,3409099,3536637,3671493,3802918
+ ,3929740,4065036,4194143,4325999,4456126,4586857,4717194,4843923,4978676,5110913
+ ,5245281,5371394,5499780,5633779,5762611,5897682,6028688,6167546,6296465,6421682
+ ,6548882,6682074,6809432,6941956,7078143,7204509,7334296,7475137,7609896,7732044
+ ,7861604,8002039,8131670,8259222,8390299,8522399,8650037,8782348,8908402,9037815
+ ,9164594,9300338,9434679,9574500,9699702,9833934,9948152,10083972,10244937,10332822
+ ,10485760,10600122,10760754,10892964,11010048,11111004,11272192,11403264,11525091,11624984
+ ,11796480,11915146,-393216,-262144,-101702,-740,100568,262144,393216,-786432
+ ,-655360,-524288,-383907,-243301,-94956,-156,95547,269629,416691,524288
+ ,655360,-393216,-262144,-88448,-37,87318,262144,393216,524288,-917504
+ ,-786432,-655360,-495894,-373308,-267503,-93211,4119,91308,250895,393216
+ ,526138,655360,786432,917504,-786432,-655360,-524288,-393216,-262144,-83497
+ ,222,86893,240922,393216,524288,-1048576,-917504,-790472,-655360,-508639
+ ,-383609,-262016,-95550,-3775,96692,256797,364847,534906,655360,786432
+ ,889679,1048576,1179648,1310720,1441792,-655360,-524288,-377684,-248408,-93690
+ ,1261,95441,227519,393216,524288,655360,786432,917504,-2097152,-1966080
+ ,-1809470,-1703936,-1572864,-1441792,-1314289,-1195149,-1056205,-917504,-809951,-657769
+ ,-521072,-383788,-248747,-106350,-2944,105550,243408,388548,521064,628732
+ ,786432,885456,1064548,1179648,1310720,1441792,1572864,1703936,1835008,-1441792
+ ,-1310720,-1179648,-1037570,-888492,-767774,-646634,-519935,-373458,-248029,-111915
+ ,760,111232,247735,379432,507672,672699,786432,917504,1048576,1179648
+ ,1310720,1441792,-4456448,-4325376,-4194304,-4063232,-3932160,-3801088,-3670016,-3538944
+ ,-3407872,-3276800,-3145728,-3014656,-2883584,-2752512,-2647002,-2490368,-2359296,-2228224
+ ,-2097152,-1951753,-1835008,-1703936,-1594177,-1462001,-1289150,-1160774,-1025917,-924928
+ ,-782509,-641294,-516191,-386630,-251910,-118886,5210,121226,253949,386008
+ ,517973,649374,780064,917783,1052462,1183856,1290593,1419389,1556641,1699884
+ ,1835008,1988314,2090470,2228224,2359296,2490368,2621440,2752512,2883584,-3801088
+ ,-3643514,-3539937,-3409931,-3263294,-3145658,-3012952,-2879230,-2752359,-2622556,-2483471
+ ,-2357556,-2226500,-2093112,-1965892,-1833664,-1701035,-1567767,-1440320,-1310556,-1178339
+ ,-1049625,-916812,-786477,-655277,-525050,-393773,-264828,-130696,-480,132126
+ ,260116,394197,527846,652294,785563,917183,1049511,1175958,1308161,1438759
+ ,1572253,1698835,1828535,1967072,2089391,2212798,2348901,2461547,2621440,2752512
+ ,2883584,-7309870,-7203780,-7062699,-6939106,-6790819,-6672036,-6553600,-6422317,-6288422
+ ,-6164694,-6026456,-5901410,-5754168,-5621459,-5502710,-5369686,-5240454,-5120712,-4976140
+ ,-4847970,-4723070,-4589083,-4450923,-4324680,-4189892,-4065551,-3931803,-3800209,-3668539
+ ,-3539395,-3404801,-3277470,-3141389,-3016710,-2885724,-2752612,-2618541,-2486762,-2354153
+ ,-2225059,-2094984,-1968194,-1830895,-1699508,-1575743,-1444516,-1308683,-1179714,-1053088
+ ,-917981,-783707,-653900,-524980,-395409,-260309,-131948,-3452,132113,263241
+ ,392185,522597,654134,788288,919810,1045795,1179210,1314201,1444235,1574447
+ ,1705193,1834009,1967332,2098102,2229019,2359147,2489859,2619878,2754966,2879671
+ ,3014438,3146143,3276733,3405958,3542196,3667493,3798815,3932961,4062458,4187125
+ ,4322346,4454875,4587752,4716809,4848274,4975027,5111957,5242215,5373085,5501158
+ ,5640140,5762918,5895358,6024008,6157906,6290628,6422713,6546339,6675888,6815606
+ ,6955288,7077501,7211630,7337893,7473635,7607175,7728310,7866475,7999658,8127888
+ ,8241758,8386483,8522550,8641582,8771915,8922139,9038632,9179385,9313426,9437184
+ ,9568256,9699328,9830400,9952933,10120004,10223616,10354688,10474645,10616832,-393216
+ ,-262144,-85425,-121,82533,262144,393216,-786432,-655360,-524288,-379928
+ ,-222821,-95200,287,95541,227093,393216,493567,655360,786432,-393216
+ ,-262144,-86805,510,86722,262144,393216,524288,-1048576,-917504,-786432
+ ,-624456,-529951,-395071,-241627,-101168,81,99975,241605,393216,524288
+ ,655360,786432,917504,-786432,-655360,-524288,-393216,-230359,-95619,-137
+ ,94425,226222,393216,524288,-1179648,-1048576,-917504,-773841,-655360,-492258
+ ,-379715,-244707,-103621,-434,104523,242680,381575,523659,650565,786432
+ ,917504,1048576,1179648,1310720,-786432,-629344,-524288,-376757,-242858,-101932
+ ,-2715,107155,239212,366480,514943,655360,786432,917504,-2228224,-2097152
+ ,-1966080,-1835008,-1703936,-1572864,-1441792,-1284584,-1179648,-1048819,-934658,-777181
+ ,-626371,-515660,-377493,-248975,-113036,436,113584,248354,379718,512475
+ ,653932,796494,917504,1048576,1179648,1310720,1441792,1572864,1703936,1835008
+ ,-1572864,-1441792,-1297608,-1161159,-1032316,-917092,-779770,-647384,-515529,-384269
+ ,-250003,-119252,1053,118111,249512,380545,512039,648101,770656,907003
+ ,1021725,1178082,1310720,1441792,-4587520,-4456448,-4325376,-4194304,-4063232,-3932160
+ ,-3801088,-3670016,-3538944,-3407872,-3276800,-3145728,-2999335,-2883584,-2752512,-2621440
+ ,-2490368,-2359296,-2228224,-2112691,-1966080,-1848781,-1709830,-1566109,-1438427,-1303530
+ ,-1176124,-1040936,-913876,-784585,-652025,-518361,-385267,-256342,-127297,-2733
+ ,125422,257792,389363,519911,651106,783805,909407,1044143,1174156,1309267
+ ,1436173,1553771,1708958,1814083,1967036,2095386,2255169,2359296,2478303,2621440
+ ,2752512,-4456448,-4325376,-4194304,-4063232,-3932160,-3797524,-3670016,-3560250,-3413217
+ ,-3257719,-3166416,-2986626,-2878000,-2781144,-2625383,-2495465,-2346792,-2230930,-2077063
+ ,-1949225,-1819274,-1697261,-1568664,-1443074,-1304302,-1175289,-1043794,-913423,-785561
+ ,-652104,-522835,-392667,-260517,-130088,-2,129509,260990,391931,522470
+ ,655770,784902,917093,1046445,1176951,1303121,1441362,1565401,1702022,1822856
+ ,1952852,2090384,2214607,2338436,2457483,2621440,-8781824,-8650752,-8519680,-8388608
+ ,-8260828,-8126464,-8003337,-7859030,-7750057,-7602176,-7471104,-7340032,-7193045,-7090588
+ ,-6946816,-6843344,-6676635,-6557575,-6447804,-6277614,-6159736,-6035729,-5884723,-5739567
+ ,-5634818,-5489867,-5372864,-5243300,-5098939,-4988639,-4856258,-4728494,-4591717,-4447428
+ ,-4322409,-4192918,-4062638,-3934141,-3797545,-3673373,-3531587,-3407391,-3277404,-3147797
+ ,-3013578,-2886548,-2749811,-2616428,-2490949,-2361301,-2228482,-2096883,-1964343,-1831754
+ ,-1702201,-1572495,-1442012,-1309242,-1182451,-1048996,-916905,-786510,-657079,-524730
+ ,-393672,-261313,-128743,166,130678,261334,393287,524155,655570,786839
+ ,917353,1052167,1179013,1309360,1442634,1571153,1703961,1832027,1965014,2097912
+ ,2224861,2355341,2490455,2623051,2753484,2877015,3015783,3144157,3273705,3405255
+ ,3542006,3669580,3802417,3935413,4065088,4190896,4333521,4456355,4579781,4713832
+ ,4845707,4978625,5113278,5243817,5382318,5500592,5638135,5761179,5900822,6029270
+ ,6186398,6297816,6436435,6559163,6666389,6806548,6950461,7086078,7195777,7350973
+ ,7480132,7614852,7743514,7847288,8014762,8126464,8257536,8388608,8519680,8650752
+ ,8781824,8912896,9043968,9175040,9306112,9437184
+};
+
+
+
+const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735] = {
+ 0, 0, -1, 0, 0, 1, 0, 1, 0, -821
+ , 1, -763, -1, 656, -620, 0, 633, -636, 4, 615
+ , -630, 1, 649, -1773, -670, 5, 678, 1810, -1876, -676
+ , 0, 691, 1843, -1806, -743, -1, 749, 1795, 2920, -2872
+ , -1761, -772, -3, 790, 1763, 2942, 0, 0, 0, 0
+ , -792, 2, 0, 0, 1, 0, -854, 0, -702, -1
+ , 662, -624, -5, 638, -611, -6, 638, -647, 0, 651
+ , -685, -4, 679, 2123, -1814, -693, 0, 664, 1791, -1735
+ , -737, 0, 771, 1854, 2873, -2867, -1842, -793, -1, 821
+ , 1826, 2805, 3922, 0, 0, 0, -1, -779, 1, 786
+ , 1, -708, 0, 789, -799, 1, 797, -663, 2, 646
+ , -600, 3, 609, -600, 1, 658, 1807, -627, -3, 612
+ , -625, 3, 632, -1732, -674, 1, 672, 2048, -1768, -715
+ , 0, 724, 1784, -3881, -3072, -1774, -719, -1, 730, 1811
+ , -2963, -1829, -806, -1, 816, 1795, 3050, -5389, -3784, -2942
+ , -1893, -865, -12, 867, 1885, 2945, 3928, -2, 1, 4
+ , 0, -694, 2, 665, -598, 5, 587, -599, -1, 661
+ , -656, -7, 611, -607, 5, 603, -618, -4, 620, -1794
+ , -645, -2, 654, -655, -1, 658, -1801, -700, 5, 707
+ , 1927, -1752, -745, -8, 752, 1843, -2838, -1781, -801, 11
+ , 796, 1811, 2942, 3866, -3849, -3026, -1848, -819, 2, 827
+ , 1825, 2963, -3873, -2904, -1869, -910, -6, 903, 1902, 2885
+ , 3978, 5286, -7168, -6081, -4989, -3968, -2963, -1970, -943, -2
+ , 953, 1951, 2968, 3974, 5009, 6032, -2, 3, -1024, 2
+ , 1024, -637, 1, 669, -613, -7, 630, -603, 4, 612
+ , -612, 0, 590, -645, -11, 627, -657, -2, 671, 1849
+ , -1853, -694, 2, 702, 1838, -3304, -1780, -736, -8, 732
+ , 1772, -1709, -755, -6, 760, 1780, -2994, -1780, -800, 8
+ , 819, 1830, 2816, -4096, -2822, -1881, -851, -4, 855, 1872
+ , 2840, 3899, -3908, -2904, -1878, -887, 6, 897, 1872, 2942
+ , 4008, -4992, -3881, -2933, -1915, -928, 1, 937, 1919, 2900
+ , 4009, 4881, -6848, -6157, -5065, -3981, -2983, -1972, -978, -1
+ , 968, 1979, 2988, 4008, 5007, 6108, 7003, 8051, 9027,-13272
+ ,-12012,-11228,-10213, -9261, -8084, -7133, -6075, -5052, -4050, -3036
+ , -2014, -996, -4, 1007, 2031, 3038, 4049, 5074, 6134, 7069
+ , 8094, 9069, 10212, 11049, 12104, 51, -1024, -13, 1024, -609
+ , -107, 613, -2048, -687, -95, 667, 2048, -3072, -1724, -785
+ , -34, 732, 1819, -2048, -703, -26, 681, 2048, -2048, -686
+ , -9, 665, 2048, -2048, -702, 37, 748, 1723, -4096, -2786
+ , -1844, -837, 37, 811, 1742, 3072, -4096, -2783, -1848, -881
+ , 39, 898, 1843, 2792, 3764, -5120, -4096, -2923, -1833, -852
+ , -14, 862, 1824, 2834, 4096, -6144, -5120, -3914, -2842, -1870
+ , -886, -27, 888, 1929, 2931, 4051, -7168, -6144, -5120, -3866
+ , -2933, -1915, -927, 64, 933, 1902, 2929, 3912, 5063, 6144
+ ,-11264,-10240, -9216, -8192, -7086, -6144, -5039, -3972, -2943, -1929
+ , -941, 3, 938, 1942, 2959, 3933, 4905, 6088, 6983, 8192
+ , -9216, -8192, -7202, -6088, -4983, -4019, -2955, -1975, -966, 17
+ , 997, 1981, 2967, 3990, 4948, 6022, 6967, 8192,-13312,-12288
+ ,-11264,-10240, -9216, -8049, -6997, -6040, -5026, -4043, -3029, -2034
+ , -1015, -23, 984, 1997, 3010, 4038, 5002, 6015, 6946, 8061
+ , 9216, 10240,-12381,-11264,-10240, -9060, -8058, -7153, -6085, -5075
+ , -4051, -3042, -2037, -1017, -5, 1007, 2028, 3035, 4050, 5088
+ , 6111, 7160, 8156, 9215, 10095, 11229, 12202, 13016,-26624,-25600
+ ,-24582,-23671,-22674,-21400,-20355,-19508,-18315,-17269,-16361,-15299
+ ,-14363,-13294,-12262,-11237,-10203, -9227, -8165, -7156, -6116, -5122
+ , -4076, -3056, -2043, -1020, -8, 1027, 2047, 3065, 4110, 5130
+ , 6125, 7168, 8195, 9206, 10230, 11227, 12256, 13304, 14281, 15316
+ , 16374, 17382, 18428, 19388, 20361, 21468, 22448, 23781, 0, 0
+ , -1, 0, -2, 1024, 0, 0, 0, -1, 1024, -1024
+ , 1, -1024, 4, 1024, -1024, 2, 1024, -1024, 2, 1024
+ , -2048, -1024, -4, 1024, -1024, 2, 1024, -2048, -1024, -3
+ , 1024, 2048, -2048, -1024, 4, 1024, 2048, -3072, -2048, -1024
+ , -1, 662, 2048, 0, 1, 0, 0, 1, -2, -2
+ , 0, 2, 1024, -1, 1024, -1024, 4, 1024, -1024, 1
+ , 1024, -1024, 1, 1024, -2048, -781, -4, 844, -807, -5
+ , 866, -2048, -726, -13, 777, 2048, -2048, -643, -4, 617
+ , 2048, 3072, -3072, -2048, -629, 1, 630, 2048, 3072, 0
+ , -1, 1, -2, 2, 1, -1024, 5, -1024, 6, 1024
+ , -1024, 4, 1024, -1024, 1, 1024, -1024, -9, 1024, -673
+ , -7, 655, -2048, -665, -15, 716, -2048, -647, 4, 640
+ , 2048, -2048, -615, -1, 635, 2048, -2048, -613, 10, 637
+ , 2048, 3072, -3072, -2048, -647, -3, 641, 2048, 3072, -5120
+ , -4096, -3072, -2048, -681, 6, 685, 2048, 3072, 4096, 1
+ , 1, 0, -1, 1024, -1024, -3, 1024, -1024, 6, 1024
+ , -1024, -1, 769, -733, 0, 1024, -876, -2, 653, -1024
+ , -4, 786, -596, -13, 595, -634, -2, 638, 2048, -2048
+ , -620, -5, 620, 2048, -4096, -3072, -2048, -639, 11, 655
+ , 2048, 3072, -3072, -2048, -659, 5, 663, 2048, -3072, -1823
+ , -687, 22, 695, 2048, 3072, 4096, -4096, -3072, -1848, -715
+ , -3, 727, 1816, 3072, 4096, 5120, -8192, -7168, -6144, -5120
+ , -4096, -2884, -1771, -756, -14, 775, 1844, 3072, 4096, 5120
+ , 6144, -1, 1, 0, -1024, 2, 815, -768, 2, 708
+ , -1024, -3, 693, -661, -7, 607, -643, -5, 609, -624
+ , 3, 631, -682, -3, 691, 2048, -2048, -640, 5, 650
+ , 2048, -3072, -2048, -701, 9, 704, 2048, 3072, -3072, -2048
+ , -670, 10, 674, 2048, 3072, -5120, -4096, -3072, -1749, -738
+ , 0, 733, 1811, 3072, 4096, 5120, -4096, -3072, -1873, -753
+ , 0, 756, 1874, 3072, 4096, -5120, -4096, -2900, -1838, -793
+ , -6, 793, 1868, 2837, 4096, 5120, -7168, -6144, -5120, -4096
+ , -2832, -1891, -828, 1, 828, 1901, 2823, 3912, 5120, 6144
+ , 7168, 8192,-13312,-12288,-11264,-10240, -9216, -8192, -7168, -6144
+ , -5120, -3976, -3004, -1911, -869, 7, 869, 1932, 3024, 3992
+ , 5009, 6144, 7168, 8192, 9216, 10240, 11264, -4, 1024, -629
+ , -22, 609, -623, 9, 640, -2048, -768, 1, 682, -2048
+ , -741, 49, 722, 2048, -3072, -1706, -808, -20, 768, 1750
+ , -1684, -727, -29, 788, 1840, 3033, -1758, -784, 0, 801
+ , 1702, -3072, -1813, -814, 38, 820, 1884, 2927, -4096, -3241
+ , -1839, -922, 25, 882, 1886, 2812, -4096, -2982, -1923, -894
+ , 84, 912, 1869, 2778, 4096, -4928, -3965, -2902, -1920, -883
+ , 3, 917, 1953, 2921, 3957, 4922, 6144, 7168, -5120, -3916
+ , -2897, -1949, -930, 31, 959, 1934, 2901, 3851, 5120, -9216
+ , -8192, -7046, -6029, -5030, -4034, -2980, -1969, -1013, -76, 963
+ , 1963, 2901, 3929, 4893, 6270, 7168, 8192, 9216,-12288,-11264
+ ,-10240, -9216, -8192, -6846, -6123, -5108, -4008, -3000, -1963, -954
+ , -6, 958, 1992, 3009, 4020, 5085, 6097, 7168, 8192, 9216
+ ,-11264,-10139, -9194, -8127, -7156, -6102, -5053, -4049, -3036, -2025
+ , -1009, -34, 974, 1984, 3034, 4028, 5138, 6000, 7057, 8166
+ , 9070, 10033, 11360, 12288,-13312,-12288,-10932,-10190, -9120, -8123
+ , -7128, -6103, -5074, -4081, -3053, -2029, -989, -4, 1010, 2028
+ , 3051, 4073, 5071, 6099, 7132, 8147, 9295, 10159, 11023, 12263
+ , 13312, 14336,-25600,-24576,-23552,-22529,-21504,-20480,-19456,-18637
+ ,-17425,-16165,-15316,-14327,-13606,-12135,-11182,-10107, -9153, -8144
+ , -7146, -6160, -5129, -4095, -3064, -2038, -1025, 1, 1031, 2072
+ , 3074, 4088, 5123, 6149, 7157, 8173, 9198, 10244, 11250, 12268
+ , 13263, 14289, 15351, 16370, 17402, 18413, 19474, 20337, 21386, 22521
+ , 23367, 24350, 0, 0, 0, 0, 0, 0, 0, 0
+ , -1024, 0, 1024, -1024, 0, 1024, -1024, 0, 1024, -1024
+ , 0, 1024, -1024, 0, 1024, -773, 0, 1024, -674, 0
+ , 645, -2048, -745, 0, 628, 2048, -2048, -712, 0, 681
+ , 2048, 3072, -3072, -2048, -673, 0, 682, 1964, 3257, 0
+ , 0, 0, 0, 0, 0, 0, 0, -1024, 0, 1024
+ , -1024, 0, 1024, -1024, 0, 1024, -705, 0, 623, -771
+ , 0, 1024, -786, 0, 688, -631, 0, 652, 2048, -2048
+ , -627, -1, 666, 2048, -3072, -1756, -694, 0, 674, 2048
+ , -3098, -1879, -720, 5, 694, 1886, 2958, 4096, 0, 0
+ , 0, 0, 1024, 0, 0, 1024, -769, 0, 1024, -1024
+ , 0, 1024, -1024, 0, 1024, -817, 0, 734, -786, 0
+ , 651, -638, 0, 637, -623, 0, 671, -652, 0, 619
+ , 2048, -2048, -670, -1, 663, 2048, -1908, -680, 1, 686
+ , 2048, 3072, 4096, -4096, -3072, -1833, -711, 0, 727, 1747
+ , 3072, 4096, -4096, -2971, -1826, -762, 2, 766, 1832, 2852
+ , 3928, 5079, 0, 0, 0, -1024, 0, 1024, -1024, 0
+ , -656, 0, 1024, -599, 0, 620, -1024, 0, 1024, -603
+ , 0, 622, -643, 0, 660, -599, 0, 611, -641, -1
+ , 651, 2048, -2048, -648, -2, 647, 1798, -3072, -2048, -672
+ , 2, 670, 2048, -3072, -1780, -694, -1, 706, 1751, 3072
+ , -3072, -1862, -757, 7, 739, 1798, 3072, 4096, -5120, -4096
+ , -3253, -1811, -787, 3, 782, 1887, 3123, 4096, -7252, -6144
+ , -5354, -4060, -2864, -1863, -820, -11, 847, 1903, 2970, 3851
+ , 4921, 5957, 7168, 8192, 9306, 0, 0, -1024, 0, 1024
+ , -726, 0, 706, -692, 0, 593, -598, 0, 616, -624
+ , 0, 616, -605, 0, 613, -2048, -652, 1, 635, 2048
+ , -2048, -647, -1, 660, 2048, -1811, -668, -2, 685, 2048
+ , -1796, -731, -2, 730, 1702, 3072, -3072, -1766, -747, -4
+ , 756, 1770, 3072, -4096, -3024, -1762, -783, 4, 771, 1781
+ , 3072, -5120, -4057, -2807, -1832, -822, 0, 816, 1804, 2851
+ , 3949, 5120, -6144, -4899, -3927, -2920, -1893, -874, -2, 868
+ , 1881, 2905, 3960, 4912, 6144, -9216, -8192, -7168, -6225, -4963
+ , -3943, -2956, -1890, -902, 0, 897, 1914, 2916, 3984, 4990
+ , 6050, 7168,-11264,-10217, -9114, -8132, -7035, -5988, -4984, -4000
+ , -2980, -1962, -927, 7, 931, 1956, 2981, 4031, 4972, 6213
+ , 7227, 8192, 9216, 10240, 11170, 12288, 13312, 14336, 0, 1024
+ , -557, 1, 571, -606, -4, 612, -1676, -707, 10, 673
+ , 2048, -2048, -727, 5, 686, -3072, -1772, -755, 12, 716
+ , 1877, -1856, -786, 2, 786, 1712, -1685, -818, -16, 863
+ , 1729, -3072, -1762, -857, 3, 866, 1838, 2841, -3862, -2816
+ , -1864, -925, -2, 923, 1897, 2779, -2782, -1838, -920, -28
+ , 931, 1951, 2835, 3804, -4815, -4001, -2940, -1934, -959, -22
+ , 975, 1957, 2904, 3971, 4835, -5148, -3892, -2944, -1953, -986
+ , -11, 989, 1968, 2939, 3949, 4947, 5902, -9216, -8192, -6915
+ , -6004, -4965, -4013, -3009, -1977, -987, -1, 982, 1972, 3000
+ , 3960, 4939, 5814, -8976, -7888, -7084, -5955, -5043, -4009, -2991
+ , -2002, -1000, -8, 993, 2011, 3023, 4026, 5028, 6023, 7052
+ , 8014, 9216,-11240,-10036, -9125, -8118, -7105, -6062, -5048, -4047
+ , -3044, -2025, -1009, -1, 1011, 2023, 3042, 4074, 5085, 6108
+ , 7119, 8142, 9152, 10114, 11141, 12250, 13307,-15360,-14099,-13284
+ ,-12291,-11223,-10221, -9152, -8147, -7128, -6104, -5077, -4072, -3062
+ , -2033, -1020, 7, 1018, 2038, 3059, 4081, 5084, 6109, 7102
+ , 8128, 9134, 10125, 11239, 12080,-23552,-22528,-21504,-20480,-19456
+ ,-18159,-17240,-16291,-15364,-14285,-13305,-12271,-11233,-10217, -9198
+ , -8175, -7157, -6134, -5122, -4089, -3071, -2047, -1018, 3, 1026
+ , 2041, 3077, 4090, 5108, 6131, 7150, 8172, 9175, 10196, 11272
+ , 12303, 13273, 14328, 15332, 16334, 17381, 18409, 19423, 20423, 21451
+ , 22679, 23391, 24568, 25600, 26589
+};
+
+/* cdf tables for quantizer indices */
+const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212] = {
+ 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17,
+ 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426,
+ 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510,
+ 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603,
+ 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4,
+ 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535,
+ 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592,
+ 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, 65522,
+ 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525,
+ 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117,
+ 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792,
+ 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, 64329,
+ 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, 0, 8,
+ 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966,
+ 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, 65435,
+ 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59,
+ 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583,
+ 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747,
+ 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, 49797,
+ 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, 64509, 64798,
+ 65045, 65162, 65288, 65363, 65447, 65506, 65522, 65531, 65533, 65535,
+ 0, 4, 6, 25, 38, 71, 138, 264, 519, 808,
+ 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108,
+ 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, 48747,
+ 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, 64275, 64673,
+ 64936, 65112, 65217, 65334, 65426, 65464, 65477, 65489, 65518, 65527,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, 12,
+ 14, 16, 21, 33, 50, 71, 84, 92, 105, 138,
+ 180, 255, 318, 377, 435, 473, 511, 590, 682, 758,
+ 913, 1097, 1256, 1449, 1671, 1884, 2169, 2445, 2772, 3157,
+ 3563, 3944, 4375, 4848, 5334, 5820, 6448, 7101, 7716, 8378,
+ 9102, 9956, 10752, 11648, 12707, 13670, 14758, 15910, 17187, 18472,
+ 19627, 20649, 21951, 23169, 24283, 25552, 26862, 28227, 29391, 30764,
+ 31882, 33213, 34432, 35600, 36910, 38116, 39464, 40729, 41872, 43144,
+ 44371, 45514, 46762, 47813, 48968, 50069, 51032, 51974, 52908, 53737,
+ 54603, 55445, 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264,
+ 60607, 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776,
+ 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, 64551,
+ 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, 65426, 65447,
+ 65472, 65485, 65487, 65489, 65502, 65510, 65512, 65514, 65516, 65518,
+ 65522, 65531, 65533, 65535, 0, 2, 4, 6, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 10, 222, 65321,
+ 65513, 65528, 65531, 65533, 65535, 0, 2, 4, 50, 65476,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 12,
+ 38, 544, 64936, 65509, 65523, 65525, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 10, 1055, 64508, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 10, 12, 123,
+ 3956, 62999, 65372, 65495, 65515, 65521, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 12, 53, 4707, 59445,
+ 65467, 65525, 65527, 65529, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 38, 40, 50, 67,
+ 96, 234, 929, 14345, 55750, 64866, 65389, 65462, 65514, 65517,
+ 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 15, 35, 91, 377, 1946,
+ 13618, 52565, 63714, 65184, 65465, 65520, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
+ 14, 16, 18, 20, 22, 24, 26, 28, 30, 32,
+ 34, 36, 38, 40, 42, 44, 46, 48, 50, 52,
+ 54, 82, 149, 362, 751, 1701, 4239, 12893, 38627, 55072,
+ 60875, 63071, 64158, 64702, 65096, 65283, 65412, 65473, 65494, 65505,
+ 65508, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533,
+ 65535, 0, 2, 15, 23, 53, 143, 260, 418, 698,
+ 988, 1353, 1812, 2411, 3144, 4015, 5143, 6401, 7611, 8999,
+ 10653, 12512, 14636, 16865, 19404, 22154, 24798, 27521, 30326, 33102,
+ 35790, 38603, 41415, 43968, 46771, 49435, 52152, 54715, 57143, 59481,
+ 61178, 62507, 63603, 64489, 64997, 65257, 65427, 65473, 65503, 65520,
+ 65529, 65531, 65533, 65535, 0, 3, 6, 9, 26, 32,
+ 44, 46, 64, 94, 111, 164, 205, 254, 327, 409,
+ 506, 608, 733, 885, 1093, 1292, 1482, 1742, 1993, 2329,
+ 2615, 3029, 3374, 3798, 4257, 4870, 5405, 5992, 6618, 7225,
+ 7816, 8418, 9051, 9761, 10532, 11380, 12113, 13010, 13788, 14594,
+ 15455, 16361, 17182, 18088, 18997, 20046, 20951, 21968, 22947, 24124,
+ 25296, 26547, 27712, 28775, 29807, 30835, 31709, 32469, 33201, 34014,
+ 34876, 35773, 36696, 37620, 38558, 39547, 40406, 41277, 42367, 43290,
+ 44445, 45443, 46510, 47684, 48973, 50157, 51187, 52242, 53209, 54083,
+ 55006, 55871, 56618, 57293, 57965, 58556, 59222, 59722, 60180, 60554,
+ 60902, 61250, 61554, 61837, 62100, 62372, 62631, 62856, 63078, 63324,
+ 63557, 63768, 63961, 64089, 64235, 64352, 64501, 64633, 64770, 64887,
+ 65001, 65059, 65121, 65188, 65246, 65302, 65346, 65390, 65428, 65463,
+ 65477, 65506, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 109, 65332, 65531, 65533,
+ 65535, 0, 2, 4, 6, 8, 25, 1817, 63874, 65511,
+ 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 907, 65014,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 10,
+ 12, 132, 2743, 62708, 65430, 65525, 65527, 65529, 65531, 65533,
+ 65535, 0, 2, 4, 6, 8, 35, 3743, 61666, 65485,
+ 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 23,
+ 109, 683, 6905, 58417, 64911, 65398, 65497, 65518, 65525, 65527,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 53, 510,
+ 10209, 55212, 64573, 65441, 65522, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
+ 22, 32, 90, 266, 1037, 3349, 14468, 50488, 62394, 64685,
+ 65341, 65480, 65514, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
+ 65533, 65535, 0, 2, 4, 6, 9, 16, 37, 106,
+ 296, 748, 1868, 5733, 18897, 45553, 60165, 63949, 64926, 65314,
+ 65441, 65508, 65524, 65529, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 18, 20, 22, 24,
+ 26, 28, 30, 32, 34, 36, 38, 40, 42, 44,
+ 46, 48, 50, 83, 175, 344, 667, 1293, 2337, 4357,
+ 8033, 14988, 28600, 43244, 52011, 57042, 59980, 61779, 63065, 63869,
+ 64390, 64753, 64988, 65164, 65326, 65422, 65462, 65492, 65506, 65522,
+ 65524, 65526, 65531, 65533, 65535, 0, 2, 4, 6, 8,
+ 10, 12, 14, 16, 25, 39, 48, 55, 62, 65,
+ 85, 106, 139, 169, 194, 252, 323, 485, 688, 1074,
+ 1600, 2544, 3863, 5733, 8303, 11397, 15529, 20273, 25734, 31455,
+ 36853, 41891, 46410, 50306, 53702, 56503, 58673, 60479, 61880, 62989,
+ 63748, 64404, 64852, 65124, 65309, 65424, 65480, 65524, 65528, 65533,
+ 65535, 0, 2, 4, 6, 8, 10, 12, 14, 21,
+ 23, 25, 27, 29, 31, 39, 41, 43, 48, 60,
+ 72, 79, 106, 136, 166, 187, 224, 252, 323, 381,
+ 427, 478, 568, 660, 783, 912, 1046, 1175, 1365, 1567,
+ 1768, 2024, 2347, 2659, 3049, 3529, 4033, 4623, 5281, 5925,
+ 6726, 7526, 8417, 9468, 10783, 12141, 13571, 15222, 16916, 18659,
+ 20350, 22020, 23725, 25497, 27201, 29026, 30867, 32632, 34323, 36062,
+ 37829, 39466, 41144, 42654, 43981, 45343, 46579, 47759, 49013, 50171,
+ 51249, 52283, 53245, 54148, 54938, 55669, 56421, 57109, 57791, 58464,
+ 59092, 59674, 60105, 60653, 61083, 61407, 61757, 62095, 62388, 62649,
+ 62873, 63157, 63358, 63540, 63725, 63884, 64046, 64155, 64278, 64426,
+ 64548, 64654, 64806, 64906, 64994, 65077, 65137, 65215, 65277, 65324,
+ 65354, 65409, 65437, 65455, 65462, 65490, 65495, 65499, 65508, 65511,
+ 65513, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
+ 65533, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059] = {
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4,
+ 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0,
+ 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779,
+ 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0,
+ 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576,
+ 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535,
+ 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611,
+ 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809,
+ 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535,
+ 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531,
+ 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527,
+ 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0,
+ 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005,
+ 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38,
+ 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804,
+ 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535,
+ 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522,
+ 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0,
+ 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780,
+ 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84,
+ 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150,
+ 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949,
+ 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200,
+ 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670,
+ 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535,
+ 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535,
+ 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419,
+ 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531,
+ 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29,
+ 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535,
+ 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2,
+ 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29,
+ 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13,
+ 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535,
+ 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324,
+ 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4,
+ 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371,
+ 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, 65150,
+ 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533,
+ 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533,
+ 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2,
+ 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535,
+ 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544,
+ 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412,
+ 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783,
+ 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6,
+ 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2,
+ 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487,
+ 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
+ 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474,
+ 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167,
+ 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495,
+ 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13,
+ 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495,
+ 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4,
+ 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834,
+ 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, 64907,
+ 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25,
+ 42, 46, 50, 75, 113, 147, 281, 448, 657, 909,
+ 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169,
+ 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, 52058,
+ 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, 64547, 64924,
+ 65158, 65313, 65430, 65481, 65518, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65533, 65535, 0, 2, 65535, 0,
+ 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
+ 65535, 0, 2, 4, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 4, 65531, 65533, 65535, 0, 2, 4, 65531,
+ 65533, 65535, 0, 2, 4, 6, 65524, 65533, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65533,
+ 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0,
+ 2, 65533, 65535, 0, 2, 4, 65532, 65535, 0, 6,
+ 65523, 65535, 0, 2, 15, 65530, 65533, 65535, 0, 2,
+ 35, 65493, 65531, 65533, 65535, 0, 2, 4, 158, 65382,
+ 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 2, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 9, 65512, 65535, 0, 2,
+ 12, 65529, 65535, 0, 2, 73, 65434, 65533, 65535, 0,
+ 2, 240, 65343, 65533, 65535, 0, 2, 476, 65017, 65531,
+ 65533, 65535, 0, 2, 4, 1046, 64686, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 1870, 63898, 65529, 65531, 65533,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65532, 65535, 0, 6, 65533, 65535, 0, 6, 65523, 65535,
+ 0, 2, 65532, 65535, 0, 137, 65439, 65535, 0, 576,
+ 64899, 65533, 65535, 0, 2, 289, 65299, 65533, 65535, 0,
+ 2, 4, 6, 880, 64134, 65531, 65533, 65535, 0, 2,
+ 4, 1853, 63347, 65533, 65535, 0, 2, 6, 2516, 61762,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 9, 3980, 61380,
+ 65503, 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8,
+ 10, 12, 61, 6393, 59859, 65466, 65527, 65529, 65531, 65533,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, 65532,
+ 65535, 0, 3, 65529, 65535, 0, 2, 65529, 65535, 0,
+ 61, 65453, 65535, 0, 234, 65313, 65535, 0, 503, 65138,
+ 65535, 0, 155, 65402, 65533, 65535, 0, 2, 1058, 64554,
+ 65533, 65535, 0, 2, 4, 3138, 62109, 65531, 65533, 65535,
+ 0, 2, 4, 2031, 63339, 65531, 65533, 65535, 0, 2,
+ 4, 6, 9, 4155, 60778, 65523, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 41, 6189, 59269, 65490, 65531, 65533, 65535,
+ 0, 2, 4, 6, 210, 8789, 57043, 65400, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 26, 453, 10086,
+ 55499, 64948, 65483, 65524, 65527, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
+ 114, 1014, 11202, 52670, 64226, 65356, 65503, 65514, 65523, 65525,
+ 65527, 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 15,
+ 65301, 65535, 0, 152, 64807, 65535, 0, 2, 3328, 63308,
+ 65535, 0, 2, 4050, 59730, 65533, 65535, 0, 2, 164,
+ 10564, 61894, 65529, 65535, 0, 15, 6712, 59831, 65076, 65532,
+ 65535, 0, 32, 7712, 57449, 65459, 65535, 0, 2, 210,
+ 7849, 53110, 65021, 65523, 65535, 0, 2, 12, 1081, 13883,
+ 48262, 62870, 65477, 65535, 0, 2, 88, 847, 6145, 37852,
+ 62012, 65454, 65533, 65535, 0, 9, 47, 207, 1823, 14522,
+ 45521, 61069, 64891, 65481, 65528, 65531, 65533, 65535, 0, 2,
+ 9, 488, 2881, 12758, 38703, 58412, 64420, 65410, 65533, 65535,
+ 0, 2, 4, 6, 61, 333, 1891, 6486, 19720, 43188,
+ 57547, 62472, 64796, 65421, 65497, 65523, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 10, 12, 29, 117, 447,
+ 1528, 6138, 21242, 43133, 56495, 62432, 64746, 65362, 65500, 65529,
+ 65531, 65533, 65535, 0, 2, 18, 105, 301, 760, 1490,
+ 3472, 7568, 15002, 26424, 40330, 53029, 60048, 62964, 64274, 64890,
+ 65337, 65445, 65489, 65513, 65527, 65530, 65533, 65535, 0, 2,
+ 4, 6, 41, 102, 409, 853, 2031, 4316, 7302, 11328,
+ 16869, 24825, 34926, 43481, 50877, 56126, 59874, 62103, 63281, 63857,
+ 64166, 64675, 65382, 65522, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 18, 29, 38, 53,
+ 58, 96, 181, 503, 1183, 2849, 5590, 8600, 11379, 13942,
+ 16478, 19453, 22638, 26039, 29411, 32921, 37596, 41433, 44998, 48560,
+ 51979, 55106, 57666, 59892, 61485, 62616, 63484, 64018, 64375, 64685,
+ 64924, 65076, 65278, 65395, 65471, 65509, 65529, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 7,
+ 65519, 65535, 0, 2, 14, 65491, 65533, 65535, 0, 2,
+ 81, 65427, 65531, 65533, 65535, 0, 2, 4, 312, 65293,
+ 65528, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
+ 65535, 0, 5, 65523, 65535, 0, 2, 65533, 65535, 0,
+ 7, 65526, 65535, 0, 46, 65464, 65533, 65535, 0, 2,
+ 120, 65309, 65533, 65535, 0, 2, 5, 362, 65097, 65533,
+ 65535, 0, 2, 18, 1164, 64785, 65528, 65531, 65533, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0,
+ 65535, 0, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65530, 65535,
+ 0, 2, 65523, 65535, 0, 69, 65477, 65535, 0, 141,
+ 65459, 65535, 0, 194, 65325, 65533, 65535, 0, 2, 543,
+ 64912, 65533, 65535, 0, 5, 1270, 64301, 65529, 65531, 65533,
+ 65535, 0, 2, 4, 12, 2055, 63538, 65508, 65531, 65533,
+ 65535, 0, 2, 7, 102, 3775, 61970, 65429, 65526, 65528,
+ 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65535, 0, 9, 65533, 65535, 0,
+ 25, 65512, 65535, 0, 2, 65533, 65535, 0, 44, 65480,
+ 65535, 0, 48, 65475, 65535, 0, 162, 65373, 65535, 0,
+ 637, 64806, 65533, 65535, 0, 2, 935, 64445, 65533, 65535,
+ 0, 2, 4, 1662, 64083, 65533, 65535, 0, 2, 12,
+ 3036, 62469, 65521, 65533, 65535, 0, 2, 120, 5405, 60468,
+ 65469, 65531, 65533, 65535, 0, 2, 4, 18, 254, 6663,
+ 58999, 65272, 65528, 65533, 65535, 0, 2, 4, 9, 12,
+ 67, 591, 8981, 56781, 64564, 65365, 65508, 65524, 65526, 65529,
+ 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 2, 65533,
+ 65535, 0, 9, 65526, 65535, 0, 14, 65503, 65535, 0,
+ 127, 65390, 65535, 0, 517, 64990, 65535, 0, 178, 65330,
+ 65535, 0, 2, 1055, 64533, 65533, 65535, 0, 2, 1558,
+ 63942, 65533, 65535, 0, 2, 2205, 63173, 65533, 65535, 0,
+ 25, 4493, 60862, 65505, 65533, 65535, 0, 2, 48, 5890,
+ 59442, 65482, 65533, 65535, 0, 2, 4, 127, 7532, 58191,
+ 65394, 65533, 65535, 0, 2, 5, 32, 550, 10388, 54924,
+ 65046, 65510, 65531, 65533, 65535, 0, 2, 4, 30, 150,
+ 1685, 14340, 51375, 63619, 65288, 65503, 65528, 65533, 65535, 0,
+ 2, 4, 6, 8, 28, 97, 473, 2692, 15407, 50020,
+ 62880, 65064, 65445, 65508, 65531, 65533, 65535, 0, 2, 4,
+ 12, 32, 79, 150, 372, 907, 2184, 5868, 18207, 45431,
+ 59856, 64031, 65096, 65401, 65481, 65507, 65521, 65523, 65525, 65527,
+ 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 182, 65491,
+ 65535, 0, 877, 64286, 65535, 0, 9, 2708, 63612, 65533,
+ 65535, 0, 2, 6038, 59532, 65535, 0, 2, 92, 5500,
+ 60539, 65533, 65535, 0, 268, 8908, 56512, 65385, 65535, 0,
+ 129, 13110, 52742, 65036, 65535, 0, 2, 806, 14003, 51929,
+ 64732, 65523, 65535, 0, 7, 92, 2667, 18159, 47678, 62610,
+ 65355, 65535, 0, 32, 1836, 19676, 48237, 61677, 64960, 65526,
+ 65535, 0, 21, 159, 967, 5668, 22782, 44709, 58317, 64020,
+ 65406, 65528, 65535, 0, 7, 162, 1838, 8328, 23929, 43014,
+ 56394, 63374, 65216, 65484, 65521, 65535, 0, 2, 4, 6,
+ 28, 268, 1120, 3613, 10688, 24185, 40989, 54917, 61684, 64510,
+ 65403, 65530, 65535, 0, 2, 16, 44, 139, 492, 1739,
+ 5313, 13558, 26766, 41566, 52446, 58937, 62815, 64480, 65201, 65454,
+ 65524, 65533, 65535, 0, 7, 25, 76, 263, 612, 1466,
+ 3325, 6832, 12366, 20152, 29466, 39255, 47360, 53506, 57740, 60726,
+ 62845, 64131, 64882, 65260, 65459, 65521, 65528, 65530, 65535, 0,
+ 2, 4, 14, 48, 136, 312, 653, 1240, 2369, 4327,
+ 7028, 10759, 15449, 21235, 28027, 35386, 42938, 49562, 54990, 59119,
+ 62086, 63916, 64863, 65249, 65445, 65493, 65523, 65535, 0, 2,
+ 4, 6, 8, 10, 12, 21, 83, 208, 409, 723,
+ 1152, 1868, 2951, 4463, 6460, 8979, 11831, 15195, 18863, 22657,
+ 26762, 30881, 34963, 39098, 43054, 47069, 50620, 53871, 56821, 59386,
+ 61340, 62670, 63512, 64023, 64429, 64750, 64944, 65126, 65279, 65366,
+ 65413, 65445, 65473, 65505, 65510, 65521, 65528, 65530, 65535
+};
+
+/* pointers to cdf tables for quantizer indices */
+const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[3][12] = {
+ { WebRtcIsacfix_kCdfGain +0 +0, WebRtcIsacfix_kCdfGain +0 +8, WebRtcIsacfix_kCdfGain +0 +22,
+ WebRtcIsacfix_kCdfGain +0 +32, WebRtcIsacfix_kCdfGain +0 +48, WebRtcIsacfix_kCdfGain +0 +60,
+ WebRtcIsacfix_kCdfGain +0 +81, WebRtcIsacfix_kCdfGain +0 +95, WebRtcIsacfix_kCdfGain +0 +128,
+ WebRtcIsacfix_kCdfGain +0 +152, WebRtcIsacfix_kCdfGain +0 +210, WebRtcIsacfix_kCdfGain +0 +264
+ },
+ { WebRtcIsacfix_kCdfGain +404 +0, WebRtcIsacfix_kCdfGain +404 +8, WebRtcIsacfix_kCdfGain +404 +21,
+ WebRtcIsacfix_kCdfGain +404 +30, WebRtcIsacfix_kCdfGain +404 +46, WebRtcIsacfix_kCdfGain +404 +58,
+ WebRtcIsacfix_kCdfGain +404 +79, WebRtcIsacfix_kCdfGain +404 +93, WebRtcIsacfix_kCdfGain +404 +125,
+ WebRtcIsacfix_kCdfGain +404 +149, WebRtcIsacfix_kCdfGain +404 +207, WebRtcIsacfix_kCdfGain +404 +260
+ },
+ { WebRtcIsacfix_kCdfGain +803 +0, WebRtcIsacfix_kCdfGain +803 +8, WebRtcIsacfix_kCdfGain +803 +22,
+ WebRtcIsacfix_kCdfGain +803 +31, WebRtcIsacfix_kCdfGain +803 +48, WebRtcIsacfix_kCdfGain +803 +60,
+ WebRtcIsacfix_kCdfGain +803 +81, WebRtcIsacfix_kCdfGain +803 +96, WebRtcIsacfix_kCdfGain +803 +129,
+ WebRtcIsacfix_kCdfGain +803 +154, WebRtcIsacfix_kCdfGain +803 +212, WebRtcIsacfix_kCdfGain +803 +268
+ }
+};
+
+const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[3][108] = {
+ { WebRtcIsacfix_kCdfShape +0 +0, WebRtcIsacfix_kCdfShape +0 +2, WebRtcIsacfix_kCdfShape +0 +4,
+ WebRtcIsacfix_kCdfShape +0 +6, WebRtcIsacfix_kCdfShape +0 +8, WebRtcIsacfix_kCdfShape +0 +10,
+ WebRtcIsacfix_kCdfShape +0 +12, WebRtcIsacfix_kCdfShape +0 +14, WebRtcIsacfix_kCdfShape +0 +16,
+ WebRtcIsacfix_kCdfShape +0 +18, WebRtcIsacfix_kCdfShape +0 +21, WebRtcIsacfix_kCdfShape +0 +25,
+ WebRtcIsacfix_kCdfShape +0 +29, WebRtcIsacfix_kCdfShape +0 +33, WebRtcIsacfix_kCdfShape +0 +37,
+ WebRtcIsacfix_kCdfShape +0 +43, WebRtcIsacfix_kCdfShape +0 +49, WebRtcIsacfix_kCdfShape +0 +56,
+ WebRtcIsacfix_kCdfShape +0 +64, WebRtcIsacfix_kCdfShape +0 +66, WebRtcIsacfix_kCdfShape +0 +68,
+ WebRtcIsacfix_kCdfShape +0 +70, WebRtcIsacfix_kCdfShape +0 +72, WebRtcIsacfix_kCdfShape +0 +75,
+ WebRtcIsacfix_kCdfShape +0 +77, WebRtcIsacfix_kCdfShape +0 +79, WebRtcIsacfix_kCdfShape +0 +81,
+ WebRtcIsacfix_kCdfShape +0 +83, WebRtcIsacfix_kCdfShape +0 +86, WebRtcIsacfix_kCdfShape +0 +90,
+ WebRtcIsacfix_kCdfShape +0 +94, WebRtcIsacfix_kCdfShape +0 +98, WebRtcIsacfix_kCdfShape +0 +102,
+ WebRtcIsacfix_kCdfShape +0 +107, WebRtcIsacfix_kCdfShape +0 +113, WebRtcIsacfix_kCdfShape +0 +120,
+ WebRtcIsacfix_kCdfShape +0 +129, WebRtcIsacfix_kCdfShape +0 +131, WebRtcIsacfix_kCdfShape +0 +133,
+ WebRtcIsacfix_kCdfShape +0 +135, WebRtcIsacfix_kCdfShape +0 +137, WebRtcIsacfix_kCdfShape +0 +141,
+ WebRtcIsacfix_kCdfShape +0 +143, WebRtcIsacfix_kCdfShape +0 +147, WebRtcIsacfix_kCdfShape +0 +151,
+ WebRtcIsacfix_kCdfShape +0 +155, WebRtcIsacfix_kCdfShape +0 +159, WebRtcIsacfix_kCdfShape +0 +164,
+ WebRtcIsacfix_kCdfShape +0 +168, WebRtcIsacfix_kCdfShape +0 +172, WebRtcIsacfix_kCdfShape +0 +178,
+ WebRtcIsacfix_kCdfShape +0 +184, WebRtcIsacfix_kCdfShape +0 +192, WebRtcIsacfix_kCdfShape +0 +200,
+ WebRtcIsacfix_kCdfShape +0 +211, WebRtcIsacfix_kCdfShape +0 +213, WebRtcIsacfix_kCdfShape +0 +215,
+ WebRtcIsacfix_kCdfShape +0 +217, WebRtcIsacfix_kCdfShape +0 +219, WebRtcIsacfix_kCdfShape +0 +223,
+ WebRtcIsacfix_kCdfShape +0 +227, WebRtcIsacfix_kCdfShape +0 +231, WebRtcIsacfix_kCdfShape +0 +235,
+ WebRtcIsacfix_kCdfShape +0 +239, WebRtcIsacfix_kCdfShape +0 +243, WebRtcIsacfix_kCdfShape +0 +248,
+ WebRtcIsacfix_kCdfShape +0 +252, WebRtcIsacfix_kCdfShape +0 +258, WebRtcIsacfix_kCdfShape +0 +264,
+ WebRtcIsacfix_kCdfShape +0 +273, WebRtcIsacfix_kCdfShape +0 +282, WebRtcIsacfix_kCdfShape +0 +293,
+ WebRtcIsacfix_kCdfShape +0 +308, WebRtcIsacfix_kCdfShape +0 +310, WebRtcIsacfix_kCdfShape +0 +312,
+ WebRtcIsacfix_kCdfShape +0 +316, WebRtcIsacfix_kCdfShape +0 +320, WebRtcIsacfix_kCdfShape +0 +324,
+ WebRtcIsacfix_kCdfShape +0 +328, WebRtcIsacfix_kCdfShape +0 +332, WebRtcIsacfix_kCdfShape +0 +336,
+ WebRtcIsacfix_kCdfShape +0 +341, WebRtcIsacfix_kCdfShape +0 +347, WebRtcIsacfix_kCdfShape +0 +354,
+ WebRtcIsacfix_kCdfShape +0 +360, WebRtcIsacfix_kCdfShape +0 +368, WebRtcIsacfix_kCdfShape +0 +378,
+ WebRtcIsacfix_kCdfShape +0 +388, WebRtcIsacfix_kCdfShape +0 +400, WebRtcIsacfix_kCdfShape +0 +418,
+ WebRtcIsacfix_kCdfShape +0 +445, WebRtcIsacfix_kCdfShape +0 +447, WebRtcIsacfix_kCdfShape +0 +451,
+ WebRtcIsacfix_kCdfShape +0 +455, WebRtcIsacfix_kCdfShape +0 +461, WebRtcIsacfix_kCdfShape +0 +468,
+ WebRtcIsacfix_kCdfShape +0 +474, WebRtcIsacfix_kCdfShape +0 +480, WebRtcIsacfix_kCdfShape +0 +486,
+ WebRtcIsacfix_kCdfShape +0 +495, WebRtcIsacfix_kCdfShape +0 +505, WebRtcIsacfix_kCdfShape +0 +516,
+ WebRtcIsacfix_kCdfShape +0 +528, WebRtcIsacfix_kCdfShape +0 +543, WebRtcIsacfix_kCdfShape +0 +564,
+ WebRtcIsacfix_kCdfShape +0 +583, WebRtcIsacfix_kCdfShape +0 +608, WebRtcIsacfix_kCdfShape +0 +635
+ },
+ { WebRtcIsacfix_kCdfShape +686 +0, WebRtcIsacfix_kCdfShape +686 +2, WebRtcIsacfix_kCdfShape +686 +4,
+ WebRtcIsacfix_kCdfShape +686 +6, WebRtcIsacfix_kCdfShape +686 +8, WebRtcIsacfix_kCdfShape +686 +11,
+ WebRtcIsacfix_kCdfShape +686 +13, WebRtcIsacfix_kCdfShape +686 +15, WebRtcIsacfix_kCdfShape +686 +17,
+ WebRtcIsacfix_kCdfShape +686 +20, WebRtcIsacfix_kCdfShape +686 +23, WebRtcIsacfix_kCdfShape +686 +27,
+ WebRtcIsacfix_kCdfShape +686 +31, WebRtcIsacfix_kCdfShape +686 +35, WebRtcIsacfix_kCdfShape +686 +40,
+ WebRtcIsacfix_kCdfShape +686 +44, WebRtcIsacfix_kCdfShape +686 +50, WebRtcIsacfix_kCdfShape +686 +56,
+ WebRtcIsacfix_kCdfShape +686 +63, WebRtcIsacfix_kCdfShape +686 +65, WebRtcIsacfix_kCdfShape +686 +67,
+ WebRtcIsacfix_kCdfShape +686 +69, WebRtcIsacfix_kCdfShape +686 +71, WebRtcIsacfix_kCdfShape +686 +73,
+ WebRtcIsacfix_kCdfShape +686 +75, WebRtcIsacfix_kCdfShape +686 +77, WebRtcIsacfix_kCdfShape +686 +79,
+ WebRtcIsacfix_kCdfShape +686 +82, WebRtcIsacfix_kCdfShape +686 +85, WebRtcIsacfix_kCdfShape +686 +89,
+ WebRtcIsacfix_kCdfShape +686 +93, WebRtcIsacfix_kCdfShape +686 +97, WebRtcIsacfix_kCdfShape +686 +102,
+ WebRtcIsacfix_kCdfShape +686 +106, WebRtcIsacfix_kCdfShape +686 +112, WebRtcIsacfix_kCdfShape +686 +119,
+ WebRtcIsacfix_kCdfShape +686 +127, WebRtcIsacfix_kCdfShape +686 +129, WebRtcIsacfix_kCdfShape +686 +131,
+ WebRtcIsacfix_kCdfShape +686 +133, WebRtcIsacfix_kCdfShape +686 +135, WebRtcIsacfix_kCdfShape +686 +137,
+ WebRtcIsacfix_kCdfShape +686 +139, WebRtcIsacfix_kCdfShape +686 +142, WebRtcIsacfix_kCdfShape +686 +146,
+ WebRtcIsacfix_kCdfShape +686 +150, WebRtcIsacfix_kCdfShape +686 +154, WebRtcIsacfix_kCdfShape +686 +158,
+ WebRtcIsacfix_kCdfShape +686 +162, WebRtcIsacfix_kCdfShape +686 +167, WebRtcIsacfix_kCdfShape +686 +173,
+ WebRtcIsacfix_kCdfShape +686 +179, WebRtcIsacfix_kCdfShape +686 +186, WebRtcIsacfix_kCdfShape +686 +194,
+ WebRtcIsacfix_kCdfShape +686 +205, WebRtcIsacfix_kCdfShape +686 +207, WebRtcIsacfix_kCdfShape +686 +209,
+ WebRtcIsacfix_kCdfShape +686 +211, WebRtcIsacfix_kCdfShape +686 +214, WebRtcIsacfix_kCdfShape +686 +218,
+ WebRtcIsacfix_kCdfShape +686 +222, WebRtcIsacfix_kCdfShape +686 +226, WebRtcIsacfix_kCdfShape +686 +230,
+ WebRtcIsacfix_kCdfShape +686 +234, WebRtcIsacfix_kCdfShape +686 +238, WebRtcIsacfix_kCdfShape +686 +242,
+ WebRtcIsacfix_kCdfShape +686 +247, WebRtcIsacfix_kCdfShape +686 +253, WebRtcIsacfix_kCdfShape +686 +262,
+ WebRtcIsacfix_kCdfShape +686 +269, WebRtcIsacfix_kCdfShape +686 +278, WebRtcIsacfix_kCdfShape +686 +289,
+ WebRtcIsacfix_kCdfShape +686 +305, WebRtcIsacfix_kCdfShape +686 +307, WebRtcIsacfix_kCdfShape +686 +309,
+ WebRtcIsacfix_kCdfShape +686 +311, WebRtcIsacfix_kCdfShape +686 +315, WebRtcIsacfix_kCdfShape +686 +319,
+ WebRtcIsacfix_kCdfShape +686 +323, WebRtcIsacfix_kCdfShape +686 +327, WebRtcIsacfix_kCdfShape +686 +331,
+ WebRtcIsacfix_kCdfShape +686 +335, WebRtcIsacfix_kCdfShape +686 +340, WebRtcIsacfix_kCdfShape +686 +346,
+ WebRtcIsacfix_kCdfShape +686 +354, WebRtcIsacfix_kCdfShape +686 +362, WebRtcIsacfix_kCdfShape +686 +374,
+ WebRtcIsacfix_kCdfShape +686 +384, WebRtcIsacfix_kCdfShape +686 +396, WebRtcIsacfix_kCdfShape +686 +413,
+ WebRtcIsacfix_kCdfShape +686 +439, WebRtcIsacfix_kCdfShape +686 +442, WebRtcIsacfix_kCdfShape +686 +446,
+ WebRtcIsacfix_kCdfShape +686 +450, WebRtcIsacfix_kCdfShape +686 +455, WebRtcIsacfix_kCdfShape +686 +461,
+ WebRtcIsacfix_kCdfShape +686 +468, WebRtcIsacfix_kCdfShape +686 +475, WebRtcIsacfix_kCdfShape +686 +481,
+ WebRtcIsacfix_kCdfShape +686 +489, WebRtcIsacfix_kCdfShape +686 +498, WebRtcIsacfix_kCdfShape +686 +508,
+ WebRtcIsacfix_kCdfShape +686 +522, WebRtcIsacfix_kCdfShape +686 +534, WebRtcIsacfix_kCdfShape +686 +554,
+ WebRtcIsacfix_kCdfShape +686 +577, WebRtcIsacfix_kCdfShape +686 +602, WebRtcIsacfix_kCdfShape +686 +631
+ },
+ { WebRtcIsacfix_kCdfShape +1368 +0, WebRtcIsacfix_kCdfShape +1368 +2, WebRtcIsacfix_kCdfShape +1368 +4,
+ WebRtcIsacfix_kCdfShape +1368 +6, WebRtcIsacfix_kCdfShape +1368 +8, WebRtcIsacfix_kCdfShape +1368 +10,
+ WebRtcIsacfix_kCdfShape +1368 +12, WebRtcIsacfix_kCdfShape +1368 +14, WebRtcIsacfix_kCdfShape +1368 +16,
+ WebRtcIsacfix_kCdfShape +1368 +20, WebRtcIsacfix_kCdfShape +1368 +24, WebRtcIsacfix_kCdfShape +1368 +28,
+ WebRtcIsacfix_kCdfShape +1368 +32, WebRtcIsacfix_kCdfShape +1368 +36, WebRtcIsacfix_kCdfShape +1368 +40,
+ WebRtcIsacfix_kCdfShape +1368 +44, WebRtcIsacfix_kCdfShape +1368 +50, WebRtcIsacfix_kCdfShape +1368 +57,
+ WebRtcIsacfix_kCdfShape +1368 +65, WebRtcIsacfix_kCdfShape +1368 +67, WebRtcIsacfix_kCdfShape +1368 +69,
+ WebRtcIsacfix_kCdfShape +1368 +71, WebRtcIsacfix_kCdfShape +1368 +73, WebRtcIsacfix_kCdfShape +1368 +75,
+ WebRtcIsacfix_kCdfShape +1368 +77, WebRtcIsacfix_kCdfShape +1368 +79, WebRtcIsacfix_kCdfShape +1368 +81,
+ WebRtcIsacfix_kCdfShape +1368 +85, WebRtcIsacfix_kCdfShape +1368 +89, WebRtcIsacfix_kCdfShape +1368 +93,
+ WebRtcIsacfix_kCdfShape +1368 +97, WebRtcIsacfix_kCdfShape +1368 +101, WebRtcIsacfix_kCdfShape +1368 +105,
+ WebRtcIsacfix_kCdfShape +1368 +110, WebRtcIsacfix_kCdfShape +1368 +116, WebRtcIsacfix_kCdfShape +1368 +123,
+ WebRtcIsacfix_kCdfShape +1368 +132, WebRtcIsacfix_kCdfShape +1368 +134, WebRtcIsacfix_kCdfShape +1368 +136,
+ WebRtcIsacfix_kCdfShape +1368 +138, WebRtcIsacfix_kCdfShape +1368 +141, WebRtcIsacfix_kCdfShape +1368 +143,
+ WebRtcIsacfix_kCdfShape +1368 +146, WebRtcIsacfix_kCdfShape +1368 +150, WebRtcIsacfix_kCdfShape +1368 +154,
+ WebRtcIsacfix_kCdfShape +1368 +158, WebRtcIsacfix_kCdfShape +1368 +162, WebRtcIsacfix_kCdfShape +1368 +166,
+ WebRtcIsacfix_kCdfShape +1368 +170, WebRtcIsacfix_kCdfShape +1368 +174, WebRtcIsacfix_kCdfShape +1368 +179,
+ WebRtcIsacfix_kCdfShape +1368 +185, WebRtcIsacfix_kCdfShape +1368 +193, WebRtcIsacfix_kCdfShape +1368 +203,
+ WebRtcIsacfix_kCdfShape +1368 +214, WebRtcIsacfix_kCdfShape +1368 +216, WebRtcIsacfix_kCdfShape +1368 +218,
+ WebRtcIsacfix_kCdfShape +1368 +220, WebRtcIsacfix_kCdfShape +1368 +224, WebRtcIsacfix_kCdfShape +1368 +227,
+ WebRtcIsacfix_kCdfShape +1368 +231, WebRtcIsacfix_kCdfShape +1368 +235, WebRtcIsacfix_kCdfShape +1368 +239,
+ WebRtcIsacfix_kCdfShape +1368 +243, WebRtcIsacfix_kCdfShape +1368 +247, WebRtcIsacfix_kCdfShape +1368 +251,
+ WebRtcIsacfix_kCdfShape +1368 +256, WebRtcIsacfix_kCdfShape +1368 +262, WebRtcIsacfix_kCdfShape +1368 +269,
+ WebRtcIsacfix_kCdfShape +1368 +277, WebRtcIsacfix_kCdfShape +1368 +286, WebRtcIsacfix_kCdfShape +1368 +297,
+ WebRtcIsacfix_kCdfShape +1368 +315, WebRtcIsacfix_kCdfShape +1368 +317, WebRtcIsacfix_kCdfShape +1368 +319,
+ WebRtcIsacfix_kCdfShape +1368 +323, WebRtcIsacfix_kCdfShape +1368 +327, WebRtcIsacfix_kCdfShape +1368 +331,
+ WebRtcIsacfix_kCdfShape +1368 +335, WebRtcIsacfix_kCdfShape +1368 +339, WebRtcIsacfix_kCdfShape +1368 +343,
+ WebRtcIsacfix_kCdfShape +1368 +349, WebRtcIsacfix_kCdfShape +1368 +355, WebRtcIsacfix_kCdfShape +1368 +361,
+ WebRtcIsacfix_kCdfShape +1368 +368, WebRtcIsacfix_kCdfShape +1368 +376, WebRtcIsacfix_kCdfShape +1368 +385,
+ WebRtcIsacfix_kCdfShape +1368 +397, WebRtcIsacfix_kCdfShape +1368 +411, WebRtcIsacfix_kCdfShape +1368 +429,
+ WebRtcIsacfix_kCdfShape +1368 +456, WebRtcIsacfix_kCdfShape +1368 +459, WebRtcIsacfix_kCdfShape +1368 +463,
+ WebRtcIsacfix_kCdfShape +1368 +467, WebRtcIsacfix_kCdfShape +1368 +473, WebRtcIsacfix_kCdfShape +1368 +478,
+ WebRtcIsacfix_kCdfShape +1368 +485, WebRtcIsacfix_kCdfShape +1368 +491, WebRtcIsacfix_kCdfShape +1368 +497,
+ WebRtcIsacfix_kCdfShape +1368 +505, WebRtcIsacfix_kCdfShape +1368 +514, WebRtcIsacfix_kCdfShape +1368 +523,
+ WebRtcIsacfix_kCdfShape +1368 +535, WebRtcIsacfix_kCdfShape +1368 +548, WebRtcIsacfix_kCdfShape +1368 +565,
+ WebRtcIsacfix_kCdfShape +1368 +585, WebRtcIsacfix_kCdfShape +1368 +611, WebRtcIsacfix_kCdfShape +1368 +640
+ }
+};
+
+/* code length for all coefficients using different models */
+
+const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392] = {
+ 25189, 16036, 8717, 358, 8757, 15706, 21456, 24397, 18502, 17559
+ , 13794, 11088, 7480, 873, 6603, 11636, 14627, 16805, 19132, 26624
+ , 26624, 19408, 13751, 7280, 583, 7591, 15178, 23773, 28672, 25189
+ , 19045, 16442, 13412, 10397, 5893, 1338, 6376, 9992, 12074, 13853
+ , 15781, 19821, 22819, 28672, 28672, 25189, 19858, 15781, 11262, 5477
+ , 1298, 5632, 11814, 17234, 22020, 28672, 19677, 18125, 16587, 14521
+ , 13032, 11196, 9249, 5411, 2495, 4994, 7975, 10234, 12308, 13892
+ , 15148, 17944, 21725, 23917, 25189, 19539, 16293, 11531, 7808, 4475
+ , 2739, 4872, 8089, 11314, 14992, 18105, 23257, 26624, 25189, 23257
+ , 23257, 20982, 18697, 18023, 16338, 16036, 14539, 13695, 13146, 11763
+ , 10754, 9074, 7260, 5584, 4430, 5553, 6848, 8344, 10141, 11636
+ , 12535, 13416, 14342, 15477, 17296, 19282, 22349, 23773, 28672, 28672
+ , 26624, 23773, 21456, 18023, 15118, 13362, 11212, 9293, 8043, 6985
+ , 5908, 5721, 5853, 6518, 7316, 8360, 9716, 11289, 12912, 14652
+ , 16969, 19858, 23773, 26624, 28013, 30720, 30720, 28672, 25426, 23141
+ , 25426, 23773, 20720, 19408, 18697, 19282, 16859, 16338, 16026, 15377
+ , 15021, 14319, 14251, 13937, 13260, 13017, 12332, 11703, 11430, 10359
+ , 10128, 9405, 8757, 8223, 7974, 7859, 7646, 7673, 7997, 8580
+ , 8880, 9061, 9866, 10397, 11358, 12200, 13244, 14157, 15021, 16026
+ , 16490, 18697, 18479, 20011, 19677, 20720, 24576, 26276, 30720, 30720
+ , 28672, 30720, 24068, 25189, 22437, 20345, 18479, 16396, 16026, 14928
+ , 13877, 13450, 12696, 12766, 11626, 11098, 10159, 9998, 9437, 9275
+ , 8783, 8552, 8629, 8488, 8522, 8454, 8571, 8775, 8915, 9427
+ , 9483, 9851, 10260, 10933, 11131, 11974, 12560, 13833, 15080, 16304
+ , 17491, 19017, 18697, 19408, 22020, 25189, 25426, 22819, 26276, 30720
+ , 30720, 30720, 30720, 30720, 30720, 28672, 30720, 30720, 30720, 30720
+ , 28013, 25426, 24397, 23773, 25189, 26624, 25189, 22437, 21725, 20011
+ , 20527, 20720, 20771, 22020, 22020, 19858, 19408, 19972, 17866, 17360
+ , 17791, 17219, 16805, 16927, 16067, 16162, 15661, 15178, 15021, 15209
+ , 14845, 14570, 14490, 14490, 13733, 13617, 13794, 13577, 13312, 12824
+ , 13032, 12683, 12189, 12469, 12109, 11940, 11636, 11617, 11932, 12294
+ , 11578, 11775, 12039, 11654, 11560, 11439, 11909, 11421, 12029, 11513
+ , 11773, 11899, 11560, 11805, 11476, 11664, 11963, 11647, 11754, 11963
+ , 11703, 12211, 11932, 12074, 12469, 12535, 12560, 12912, 12783, 12866
+ , 12884, 13378, 13957, 13775, 13635, 14019, 14545, 15240, 15520, 15554
+ , 15697, 16490, 16396, 17281, 16599, 16969, 17963, 16859, 16983, 16805
+ , 17099, 18210, 17219, 17646, 17700, 17646, 18297, 17425, 18479, 17791
+ , 17718, 19282, 18672, 20173, 20982, 21725, 21456, 23773, 23257, 25189
+ , 30720, 30720, 25189, 26624, 30720, 30720, 30720, 30720, 28672, 26276
+ , 30720, 30720
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 28672
+ , 0, 26624, 1, 23773, 22819, 4, 20982, 18598, 10, 19282
+ , 16587, 22, 16442, 26624, 13126, 60, 14245, 26624, 26624, 12736
+ , 79, 12912, 25189, 22819, 9563, 249, 9474, 22349, 28672, 23257
+ , 17944, 7980, 434, 8181, 16431, 26624, 0, 0, 0, 0
+ , 28672, 0, 0, 0, 0, 0, 28672, 0, 22437, 3
+ , 22437, 20982, 5, 20982, 16442, 22, 16752, 13814, 49, 14646
+ , 11645, 116, 11734, 26624, 28672, 10613, 158, 11010, 24397, 19539
+ , 8046, 453, 7709, 19017, 28672, 23257, 15110, 6770, 758, 6523
+ , 14108, 24397, 28672, 0, 0, 0, 0, 28672, 0, 28672
+ , 0, 26624, 1, 28672, 28672, 1, 26624, 24397, 2, 23257
+ , 21725, 4, 20982, 17158, 18, 17281, 28672, 15178, 35, 15209
+ , 12343, 92, 12320, 26624, 10344, 189, 10217, 30720, 22020, 9033
+ , 322, 8549, 23773, 28672, 30720, 20622, 7666, 473, 7806, 20527
+ , 24397, 14135, 5995, 960, 6018, 14872, 23773, 26624, 20928, 16293
+ , 10636, 4926, 1588, 5256, 11088, 18043, 25189, 0, 0, 0
+ , 0, 24397, 1, 25189, 20720, 5, 21456, 21209, 3, 25189
+ , 20982, 5, 21456, 15818, 30, 15410, 13794, 60, 13416, 28672
+ , 11162, 142, 11025, 9337, 231, 10094, 23773, 8338, 405, 7930
+ , 26624, 19677, 6787, 613, 7318, 19161, 28672, 16442, 6319, 932
+ , 5748, 15312, 25189, 28672, 28672, 28672, 13998, 5513, 1263, 5146
+ , 14024, 24397, 22819, 15818, 9460, 4447, 2122, 4681, 9970, 15945
+ , 22349, 28672, 30720, 22622, 19017, 14872, 10689, 7405, 4473, 2983
+ , 4783, 7894, 11186, 14964, 18210, 24397, 0, 0, 30720, 0
+ , 30720, 21456, 3, 23773, 14964, 39, 14757, 14179, 53, 13751
+ , 14928, 36, 15272, 12430, 79, 13228, 9135, 285, 9077, 28672
+ , 28672, 8377, 403, 7919, 26624, 28672, 23257, 7068, 560, 7473
+ , 20345, 19677, 6770, 720, 6464, 18697, 25189, 16249, 5779, 1087
+ , 5494, 15209, 22819, 30720, 20622, 12601, 5240, 1419, 5091, 12095
+ , 19408, 26624, 22819, 16805, 10683, 4812, 2056, 4293, 9836, 16026
+ , 24397, 25189, 18409, 13833, 8681, 4503, 2653, 4220, 8329, 13853
+ , 19132, 26624, 25189, 20771, 17219, 12630, 9520, 6733, 4565, 3657
+ , 4817, 7069, 10058, 13212, 16805, 21209, 26624, 26276, 28672, 28672
+ , 26276, 23257, 20173, 19282, 16538, 15051, 12811, 10754, 9267, 7547
+ , 6270, 5407, 5214, 6057, 7054, 8226, 9488, 10806, 12793, 14442
+ , 16442, 19677, 22099, 26276, 28672, 0, 30720, 0, 30720, 11920
+ , 56, 20720, 30720, 6766, 355, 13130, 30720, 30720, 22180, 5589
+ , 736, 7902, 26624, 30720, 7634, 354, 9721, 30720, 30720, 9027
+ , 246, 10117, 30720, 30720, 9630, 453, 6709, 23257, 30720, 25683
+ , 14228, 6127, 1271, 4615, 15178, 30720, 30720, 23504, 12382, 5739
+ , 2015, 3492, 10560, 22020, 26624, 30720, 30720, 23257, 13192, 4873
+ , 1527, 5001, 12445, 22020, 30720, 30720, 30720, 30720, 19344, 10761
+ , 4051, 1927, 5281, 10594, 17866, 28672, 30720, 30720, 30720, 21869
+ , 15554, 10060, 5979, 2710, 3085, 7889, 14646, 21725, 28672, 30720
+ , 30720, 30720, 30720, 30720, 30720, 30720, 22719, 17425, 13212, 8083
+ , 4439, 2820, 4305, 8136, 12988, 17425, 21151, 28672, 28672, 30720
+ , 30720, 30720, 28672, 20527, 19282, 14412, 10513, 7407, 5079, 3744
+ , 4115, 6308, 9621, 13599, 17040, 22349, 28672, 30720, 30720, 30720
+ , 30720, 30720, 30720, 29522, 19282, 14545, 11485, 9093, 6760, 5262
+ , 4672, 4970, 6005, 7852, 9732, 12343, 14672, 19161, 22819, 25189
+ , 30720, 30720, 28672, 30720, 30720, 20720, 18125, 14388, 12007, 9825
+ , 8092, 7064, 6069, 5903, 5932, 6359, 7169, 8310, 9324, 10711
+ , 11867, 13096, 14157, 16338, 17040, 19161, 21725, 23773, 30720, 30720
+ , 26276, 25426, 24397, 28672, 28672, 23257, 22020, 22349, 18297, 17646
+ , 16983, 16431, 16162, 15021, 15178, 13751, 12142, 10895, 10193, 9632
+ , 9086, 8896, 8823, 8735, 8591, 8754, 8649, 8361, 8329, 8522
+ , 8373, 8739, 8993, 9657, 10454, 11279, 11899, 12614, 14024, 14273
+ , 15477, 15240, 16649, 17866, 18697, 21151, 22099
+};
+
+/* left KLT transforms */
+const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[3][4] = {
+ { -26130, 19773, 19773, 26130 },
+ { -26664, 19046, 19046, 26664 },
+ { -23538, 22797, 22797, 23538 }
+};
+
+
+
+const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[3][324] = {
+ { 52,16,168,7,439,-138,-89,306,671,882,
+ 157,1301,291,1598,-3571,-1943,-1119,32404,96,-12,
+ 379,-64,-307,345,-836,539,1045,2541,-2865,-992,
+ 1683,-4717,5808,7427,30599,2319,183,-73,451,481,
+ 933,-198,781,-397,1244,-777,3690,-2414,149,-1356,
+ -2593,-31140,8289,-1737,-202,-14,-214,360,501,450,
+ -245,-7,797,3638,-2804,3042,-337,22137,-22103,2264,
+ 6838,-3381,305,172,263,-195,-355,351,179,513,
+ 2234,3343,5509,7531,19075,-17740,-16836,2244,-629,-1505,
+ -153,108,124,-324,2694,-124,1492,-850,5347,4285,
+ 7439,-10229,-22822,-12467,-12891,3645,822,-232,131,13,
+ 374,565,536,4681,1294,-1935,1926,-5734,-10643,26462,
+ -12480,-5589,-1038,-2468,964,-704,-247,-106,186,-558,
+ -4050,3760,2972,2141,-7393,6294,26740,11991,-3251,5461,
+ 5341,1574,2208,-51,-552,-297,-753,-154,2068,-5371,
+ 3578,4106,28043,-10533,8041,2353,2389,4609,3410,1906,
+ 351,-249,18,-15,1117,539,2870,9084,17585,-24528,
+ -366,-6490,2009,-3170,2942,1116,-232,1672,1065,606,
+ -399,-388,-518,38,3728,28948,-11936,4543,4104,-4441,
+ 1545,-4044,1485,622,-68,186,-473,135,-280,125,
+ -546,-1813,6989,6606,23711,19376,-2636,2870,-4553,-1687,
+ 878,-375,205,-208,-409,-108,-200,-45,-1670,-337,
+ 8213,-5524,-2334,5240,-12939,-26205,5937,-1582,-592,-959,
+ -5374,2449,3400,559,349,-492,668,12379,-27684,3419,
+ 5117,4415,-297,-8270,-1252,-3490,-1272,-1199,-3159,191,
+ 630,488,-797,-3071,12912,-27783,-10249,1047,647,619,
+ 111,-3722,-915,-1055,-502,5,-1384,-306,221,68,
+ 5219,13173,-26474,-11663,-5626,927,806,-1127,236,-589,
+ -522,-230,-312,-315,-428,-573,426,192,-11830,-26883,
+ -14121,-2785,-1429,-109,410,-832,-302,539,-459,104,
+ 1,-530,-202,-289,153,116,30082,-12944,-671,20,
+ 649,98,103,215,234,0,280,-51,-169,298,
+ 31,230,-73,-51
+ },
+ { -154,-7,-192,61,-739,-389,-947,-162,-60,94,
+ 511,-716,1520,-1428,4168,-2214,1816,32270,-123,-77,
+ -199,-99,-42,-588,203,-240,-930,-35,1580,234,
+ 3206,-5507,-1495,-10946,30000,-2667,-136,-176,-240,-175,
+ -204,-661,-1796,-1039,-1271,498,3143,734,2663,2699,
+ -8127,29333,10495,2356,-72,113,-91,118,-2840,-723,
+ -1733,-1158,-389,-2116,-3054,-3,-5179,8071,29546,6308,
+ 5657,-3178,-186,-294,-473,-635,1213,-983,-1437,-1715,
+ -1094,1280,-92,-9573,948,29576,-7060,-5921,2954,1349,
+ -337,-108,-1099,962,418,-413,-1149,-334,1241,3975,
+ -6825,26725,-14377,7051,-4772,-1707,2335,2008,-150,570,
+ 1371,42,-1649,-619,2039,3369,-1225,1583,-2755,-15207,
+ -27504,-4855,-4304,1495,2733,1324,15,-448,403,353,
+ 3016,-1242,2338,2673,2064,-7496,-30447,-3686,5833,-1301,
+ -2455,2122,1519,608,43,-653,773,-3072,912,-1537,
+ 4505,10284,30237,1549,3200,-691,205,1702,658,1014,
+ 1499,148,79,-322,-1162,-4639,-813,7536,3204,29109,
+ -10747,-26,1611,2286,2114,2561,1022,372,348,207,
+ 1062,-1088,-443,-9849,2381,5671,29097,-7612,-2927,3853,
+ 194,1155,275,1438,1438,1312,581,888,-784,906,
+ 112,-11103,25104,14438,-9311,-3068,1210,368,370,-940,
+ -2434,-1148,1925,392,657,258,-526,1475,-2281,-4265,
+ -1880,1534,2185,-1472,959,-30934,6306,3114,-4109,1768,
+ -2612,-703,45,644,2185,2033,5670,7211,19114,-22427,
+ 6432,5150,-4090,-2694,3860,1245,-596,293,1829,369,
+ -319,229,-3256,2170,-6374,-26216,-4570,-16053,-5766,-262,
+ -2006,2873,-1477,147,378,-1544,-344,-544,-985,-481,
+ 4210,4542,30757,-7291,-4863,1529,-2079,-628,-603,-783,
+ -408,1646,697,808,-620,-292,181,158,-13313,-29173,
+ 5984,-1262,859,-1776,-558,-24,-883,-1421,739,210,
+ -531,-285,131,-160,-246,-56,29345,-13706,-2859,-2966,
+ -300,-970,-2382,-268,-103,-636,-12,-62,-691,-253,
+ -147,-127,27,66
+ },
+ { 55,-212,-198,489,-274,81,682,399,328,-934,
+ -389,-37,1357,-3632,5276,6581,-9493,-29921,29,-45,
+ 2,190,172,-15,311,-130,-1085,-25,324,-684,
+ 3223,-6580,4485,-5280,-29521,9933,82,-320,-530,229,
+ -705,-533,-414,848,-1842,-4473,1390,-857,6717,-6692,
+ 4648,29397,576,8339,-68,-85,238,-330,264,-1012,
+ -381,-203,-3384,-3329,3906,6810,3790,-6250,28312,-8078,
+ 8089,1565,160,-569,-612,-613,-1063,-1928,-1125,3421,
+ -7481,-7484,4942,-6984,4330,-25591,-10574,-6982,5682,-1781,
+ -308,89,178,-1715,-420,-3530,-5776,1219,-8617,-7137,
+ 7015,4981,24875,12657,-5408,-3356,-785,-1972,326,-858,
+ -506,-3382,-986,-6258,-2259,4015,-8374,-10482,3127,23826,
+ -14126,-514,-5417,2178,-2912,-17,-587,80,67,-5881,
+ -1702,-5351,-4481,398,-10156,-225,20727,-15460,-11603,7752,
+ 3660,1714,-2001,-359,499,-527,-1225,-7820,-1297,-6326,
+ -8526,7900,-18328,13311,-17488,-2926,-196,-17,2281,873,
+ 480,-160,-624,471,780,-8729,1707,-14262,-20647,1721,
+ 18590,-2206,-1214,-1066,312,-2602,783,-412,-113,49,
+ -119,1305,-2371,-15132,-1833,-18252,20295,-8316,2227,341,
+ -2074,-702,3082,-262,-465,-198,430,30,-70,-788,
+ 2342,-25132,-4863,19783,-484,2137,2811,-1906,799,1586,
+ 962,-734,-191,-30,-129,-93,-1126,1729,5860,-2030,
+ 8953,603,-3338,-10869,-1144,22070,12130,10513,3191,-6881,
+ -3514,2090,711,-666,1843,-5997,-5681,2921,-17641,-2801,
+ 4969,18590,7169,12214,8587,4405,3008,-1074,-371,-77,
+ 253,331,-5611,5014,13152,-1985,18483,-1696,8043,20463,
+ 2381,-393,1688,-1205,618,1220,457,248,-83,176,
+ 7920,-13676,-22139,-3038,17402,2036,844,3258,994,719,
+ 2087,-44,426,494,12,-91,46,5,-14204,22912,
+ -18156,-361,442,2298,-829,2229,386,1433,1335,1323,
+ 55,-592,-139,49,-12,-57,27783,17134,350,-282,
+ 552,158,142,2488,465,329,1087,118,143,10,
+ 56,65,-15,-31
+ }
+};
+
+/* right KLT transforms */
+const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[3][36] = {
+ { 4775, -14892, 20313, -17104, 10533, -3613, -6782, 16044, -8889,
+ -11019, 21330, -10720, 13193, -15678, -11101, 14461, 12250, -13096,
+ -16951, 2167, 16066, 15569, -702, -16754, -19195, -12823, -4321,
+ 5128, 13348, 17825, 13232, 13404, 13494, 13490, 13383, 13261
+ },
+ { -3725, 11408, -18493, 20031, -13097, 3865, 9344, -19294, 10740,
+ 8856, -18432, 8982, 13975, -14444, -11930, 11774, 14285, -13594,
+ -16323, -4, 16340, 15609, 359, -17220, -18401, -13471, -4643,
+ 5225, 13375, 18053, 13124, 13463, 13621, 13583, 13393, 13072
+ },
+ { -3513, 11402, -17883, 19504, -14399, 4885, 8702, -19513, 12046,
+ 8533, -18110, 8447, 12778, -14838, -12444, 13177, 14107, -12759,
+ -17268, 914, 15822, 15661, 838, -16686, -18907, -12936, -4820,
+ 4175, 12398, 18830, 12913, 13215, 13433, 13572, 13601, 13518
+ }
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[3][36] = {
+ { 4400, -11512, 17205, -19470, 14770, -5345, 9784, -19222, 11228,
+ 6842, -18371, 9909, 14191, -13496, -11563, 14015, 11827, -14839,
+ -15439, 948, 17802, 14827, -2053, -17132, 18723, 14516, 4135,
+ -6822, -13869, -16016, 12975, 13341, 13563, 13603, 13478, 13296
+ },
+ { 5420, -14215, 19060, -18073, 11709, -3911, 9645, -18335, 7717,
+ 10842, -19283, 9777, 14898, -12555, -13661, 11668, 13520, -13733,
+ -15936, -1358, 15671, 16728, 328, -17100, 17527, 13973, 5587,
+ -5194, -14165, -17677, 12970, 13446, 13693, 13660, 13462, 13015
+ },
+ { 4386, -12426, 18019, -18895, 13894, -5034, 9713, -19270, 10283,
+ 8692, -18439, 9317, 13992, -13454, -13241, 12850, 13366, -13336,
+ -16334, -498, 15976, 16213, -114, -16987, 18191, 13659, 4958,
+ -5116, -13444, -18021, 12911, 13424, 13718, 13674, 13464, 13054
+ }
+};
+
+/* means of log gains and LAR coefficients*/
+const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[3][12] = {
+ { -1758, -1370, -1758, -1373, -1757, -1375,
+ -1758, -1374, -1758, -1373, -1755, -1370
+ },
+ { -1569, -1224, -1569, -1225, -1569, -1227,
+ -1569, -1226, -1567, -1225, -1565, -1224
+ },
+ { -1452, -957, -1447, -951, -1438, -944,
+ -1431, -938, -1419, -931, -1406, -926
+ }
+};
+
+
+const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108] = {
+ { -119581, 34418, -44193, 11112, -4428, 18906, 9222, 8068, 1953, 5425,
+ 1871, 1689, 109933, 33751, 10471, -2566, 1090, 2320, -119219, 33728,
+ -43759, 11450, -4870, 19117, 9174, 8037, 1972, 5331, 1872, 1843,
+ 109899, 34301, 10629, -2316, 1272, 2562, -118608, 32318, -44012, 11591,
+ -4914, 18932, 9456, 8088, 1900, 5419, 1723, 1853, 109963, 35059,
+ 10745, -2335, 1161, 2520, -119174, 32107, -44462, 11635, -4694, 18611,
+ 9757, 8108, 1969, 5486, 1673, 1777, 109636, 34907, 10643, -2406,
+ 1034, 2420, -118597, 32320, -44590, 10854, -4569, 18821, 9701, 7866,
+ 2003, 5577, 1732, 1626, 109913, 34448, 10714, -2752, 990, 2228,
+ -118138, 32996, -44352, 10334, -3772, 18488, 9464, 7865, 2208, 5540,
+ 1745, 1664, 109880, 33381, 10640, -2779, 980, 2054
+ },
+ { -146328, 46370, 1047, 26431, 10035, 13933, 6415, 14359, -2368, 6661,
+ 2269, 1764, 96623, 7802, 4163, 10742, 1643, 2954, -146871, 46561, 1127,
+ 26225, 10113, 14096, 6771, 14323, -2037, 6788, 2297, 1761, 96324, 8382,
+ 4309, 10450, 1695, 3016, -146502, 46475, 1580, 26118, 10487, 14179, 6622,
+ 14439, -2034, 6757, 2342, 1761, 95869, 8966, 4347, 10358, 1999, 2855,
+ -146958, 47717, 826, 25952, 10263, 14061, 5266, 13681, -2417, 6582, 2047,
+ 1608, 96257, 9107, 4452, 10301, 1792, 2676, -146992, 47123, 446, 25822,
+ 10405, 14292, 5140, 13804, -2403, 6496, 1834, 1735, 97489, 9253, 4414,
+ 10684, 1549, 2721, -145811, 46182, 901, 26482, 10241, 14524, 6075, 14514,
+ -2147, 6691, 2196, 1899, 97011, 8178, 4102, 10758, 1638, 2869
+ },
+ { -166617, 46969, -43908, 17726, 6330, 25615, 6913, 5450, -2301, 1984,
+ 507, 2883, 149998, 28709, 19333, 16703, 11093, 8965, -168254, 46604,
+ -44315, 17862, 6474, 25746, 7018, 5373, -2343, 1930, 513, 2819, 150391,
+ 28627, 19194, 16678, 10998, 8929, -169093, 46084, -44767, 17427, 6401,
+ 25674, 7147, 5472, -2336, 1820, 491, 2802, 149860, 28430, 19064, 16524,
+ 10898, 8875, -170205, 46189, -44877, 17403, 6190, 25209, 7035, 5673, -2173,
+ 1894, 574, 2756, 148830, 28230, 18819, 16418, 10789, 8811, -171263, 45045,
+ -44834, 16858, 6103, 24726, 7014, 5713, -2103, 1877, 518, 2729, 147073,
+ 27744, 18629, 16277, 10690, 8703, -171720, 44153, -45062, 15951, 5872,
+ 24429, 7044, 5585, -2082, 1807, 519, 2769, 144791, 27402, 18490, 16126,
+ 10548, 8635
+ }
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h
new file mode 100644
index 0000000..4f2e0e7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h
@@ -0,0 +1,98 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_tables.h
+ *
+ * header file for coding tables for the LPC coefficients
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
+
+#include "typedefs.h"
+
+
+/* indices of KLT coefficients used */
+extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108];
+
+/* cdf array for model indicator */
+extern const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS+1];
+
+/* pointer to cdf array for model indicator */
+extern const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1];
+
+/* initial cdf index for decoder of model indicator */
+extern const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1];
+
+/* offset to go from rounded value to quantization index */
+extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12];
+
+extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108];
+
+/* maximum quantization index */
+extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108];
+
+/* index offset */
+extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[KLT_NUM_MODELS][108];
+
+/* initial cdf index for KLT coefficients */
+extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[KLT_NUM_MODELS][108];
+
+/* offsets for quantizer representation levels */
+extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3];
+
+/* quantizer representation levels */
+extern const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176];
+
+extern const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735];
+
+/* cdf tables for quantizer indices */
+extern const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212];
+
+extern const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059];
+
+/* pointers to cdf tables for quantizer indices */
+extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
+
+/* code length for all coefficients using different models */
+extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392];
+
+extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577];
+
+/* left KLT transforms */
+extern const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4];
+
+extern const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324];
+
+/* right KLT transforms */
+extern const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36];
+
+extern const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36];
+
+/* means of log gains and LAR coefficients */
+extern const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[KLT_NUM_MODELS][12];
+
+extern const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c
new file mode 100644
index 0000000..1702098
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c
@@ -0,0 +1,519 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.c
+ *
+ * Pitch filter functions
+ *
+ */
+
+#include <string.h>
+
+#include "signal_processing_library.h"
+#include "pitch_estimator.h"
+
+/* log2[0.2, 0.5, 0.98] in Q8 */
+static const WebRtc_Word16 kLogLagWinQ8[3] = {
+ -594, -256, -7
+};
+
+/* [1 -0.75 0.25] in Q12 */
+static const WebRtc_Word16 kACoefQ12[3] = {
+ 4096, -3072, 1024
+};
+
+
+
+static __inline WebRtc_Word32 Log2Q8( WebRtc_UWord32 x ) {
+
+ WebRtc_Word32 zeros, lg2;
+ WebRtc_Word16 frac;
+
+ zeros=WebRtcSpl_NormU32(x);
+ frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)(WEBRTC_SPL_LSHIFT_W32(x, zeros))&0x7FFFFFFF), 23);
+ /* log2(magn(i)) */
+
+ lg2= (WEBRTC_SPL_LSHIFT_W32((31-zeros), 8)+frac);
+ return lg2;
+
+}
+
+static __inline WebRtc_Word16 Exp2Q10(WebRtc_Word16 x) { // Both in and out in Q10
+
+ WebRtc_Word16 tmp16_1, tmp16_2;
+
+ tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
+ tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
+ if(tmp16_1>0)
+ return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
+ else
+ return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
+
+}
+
+
+
+/* 1D parabolic interpolation . All input and output values are in Q8 */
+static __inline void Intrp1DQ8(WebRtc_Word32 *x, WebRtc_Word32 *fx, WebRtc_Word32 *y, WebRtc_Word32 *fy) {
+
+ WebRtc_Word16 sign1=1, sign2=1;
+ WebRtc_Word32 r32, q32, t32, nom32, den32;
+ WebRtc_Word16 t16, tmp16, tmp16_1;
+
+ if ((fx[0]>0) && (fx[2]>0)) {
+ r32=fx[1]-fx[2];
+ q32=fx[0]-fx[1];
+ nom32=q32+r32;
+ den32=WEBRTC_SPL_MUL_32_16((q32-r32), 2);
+ if (nom32<0)
+ sign1=-1;
+ if (den32<0)
+ sign2=-1;
+
+ /* t = (q32+r32)/(2*(q32-r32)) = (fx[0]-fx[1] + fx[1]-fx[2])/(2 * fx[0]-fx[1] - (fx[1]-fx[2]))*/
+ /* (Signs are removed because WebRtcSpl_DivResultInQ31 can't handle negative numbers) */
+ t32=WebRtcSpl_DivResultInQ31(WEBRTC_SPL_MUL_32_16(nom32, sign1),WEBRTC_SPL_MUL_32_16(den32, sign2)); /* t in Q31, without signs */
+
+ t16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(t32, 23); /* Q8 */
+ t16=t16*sign1*sign2; /* t in Q8 with signs */
+
+ *y = x[0]+t16; /* Q8 */
+ // *y = x[1]+t16; /* Q8 */
+
+ /* The following code calculates fy in three steps */
+ /* fy = 0.5 * t * (t-1) * fx[0] + (1-t*t) * fx[1] + 0.5 * t * (t+1) * fx[2]; */
+
+ /* Part I: 0.5 * t * (t-1) * fx[0] */
+ tmp16_1=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16,t16); /* Q8*Q8=Q16 */
+ tmp16_1 = WEBRTC_SPL_RSHIFT_W16(tmp16_1,2); /* Q16>>2 = Q14 */
+ t16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16, 64); /* Q8<<6 = Q14 */
+ tmp16 = tmp16_1-t16;
+ *fy = WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[0]); /* (Q14 * Q8 >>15)/2 = Q8 */
+
+ /* Part II: (1-t*t) * fx[1] */
+ tmp16 = 16384-tmp16_1; /* 1 in Q14 - Q14 */
+ *fy += WEBRTC_SPL_MUL_16_32_RSFT14(tmp16, fx[1]);/* Q14 * Q8 >> 14 = Q8 */
+
+ /* Part III: 0.5 * t * (t+1) * fx[2] */
+ tmp16 = tmp16_1+t16;
+ *fy += WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[2]);/* (Q14 * Q8 >>15)/2 = Q8 */
+ } else {
+ *y = x[0];
+ *fy= fx[1];
+ }
+}
+
+
+static void FindFour32(WebRtc_Word32 *in, WebRtc_Word16 length, WebRtc_Word16 *bestind)
+{
+ WebRtc_Word32 best[4]= {-100, -100, -100, -100};
+ WebRtc_Word16 k;
+
+ for (k=0; k<length; k++) {
+ if (in[k] > best[3]) {
+ if (in[k] > best[2]) {
+ if (in[k] > best[1]) {
+ if (in[k] > best[0]) { // The Best
+ best[3] = best[2];
+ bestind[3] = bestind[2];
+ best[2] = best[1];
+ bestind[2] = bestind[1];
+ best[1] = best[0];
+ bestind[1] = bestind[0];
+ best[0] = in[k];
+ bestind[0] = k;
+ } else { // 2nd best
+ best[3] = best[2];
+ bestind[3] = bestind[2];
+ best[2] = best[1];
+ bestind[2] = bestind[1];
+ best[1] = in[k];
+ bestind[1] = k;
+ }
+ } else { // 3rd best
+ best[3] = best[2];
+ bestind[3] = bestind[2];
+ best[2] = in[k];
+ bestind[2] = k;
+ }
+ } else { // 4th best
+ best[3] = in[k];
+ bestind[3] = k;
+ }
+ }
+ }
+}
+
+
+
+
+
+static void PCorr2Q32(const WebRtc_Word16 *in, WebRtc_Word32 *logcorQ8)
+{
+ WebRtc_Word16 scaling,n,k;
+ WebRtc_Word32 ysum32,csum32, lys, lcs;
+ WebRtc_Word32 prod32, oneQ8;
+
+
+ const WebRtc_Word16 *x, *inptr;
+
+ oneQ8 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, 8); // 1.00 in Q8
+
+ x = in + PITCH_MAX_LAG/2 + 2;
+ scaling = WebRtcSpl_GetScalingSquare ((WebRtc_Word16 *) in, PITCH_CORR_LEN2, PITCH_CORR_LEN2);
+ ysum32 = 1;
+ csum32 = 0;
+ x = in + PITCH_MAX_LAG/2 + 2;
+ for (n = 0; n < PITCH_CORR_LEN2; n++) {
+ ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[n],(WebRtc_Word16) in[n], scaling); // Q0
+ csum32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) x[n],(WebRtc_Word16) in[n], scaling); // Q0
+ }
+
+ logcorQ8 += PITCH_LAG_SPAN2 - 1;
+
+ lys=Log2Q8((WebRtc_UWord32) ysum32); // Q8
+ lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
+
+ if (csum32>0) {
+
+ lcs=Log2Q8((WebRtc_UWord32) csum32); // 2log(csum) in Q8
+
+ if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2 in Q8
+ *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum))
+ } else {
+ *logcorQ8 = oneQ8; // 1.00
+ }
+
+ } else {
+ *logcorQ8 = 0;
+ }
+
+
+ for (k = 1; k < PITCH_LAG_SPAN2; k++) {
+ inptr = &in[k];
+ ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[k-1],(WebRtc_Word16) in[k-1], scaling);
+ ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1],(WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1], scaling);
+ csum32 = 0;
+ prod32 = WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) x[0],(WebRtc_Word16) inptr[0], scaling);
+
+ for (n = 1; n < PITCH_CORR_LEN2; n++) {
+ csum32 += prod32;
+ prod32 = WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) x[n],(WebRtc_Word16) inptr[n], scaling);
+ }
+
+ csum32 += prod32;
+ logcorQ8--;
+
+ lys=Log2Q8((WebRtc_UWord32)ysum32); // Q8
+ lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
+
+ if (csum32>0) {
+
+ lcs=Log2Q8((WebRtc_UWord32) csum32); // 2log(csum) in Q8
+
+ if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2
+ *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum))
+ } else {
+ *logcorQ8 = oneQ8; // 1.00
+ }
+
+ } else {
+ *logcorQ8 = 0;
+ }
+ }
+}
+
+
+
+void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in, /* Q0 */
+ PitchAnalysisStruct *State,
+ WebRtc_Word16 *lagsQ7 /* Q7 */
+ )
+{
+ WebRtc_Word16 buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+ WebRtc_Word32 *crrvecQ8_1,*crrvecQ8_2;
+ WebRtc_Word32 cv1q[PITCH_LAG_SPAN2+2],cv2q[PITCH_LAG_SPAN2+2], peakvq[PITCH_LAG_SPAN2+2];
+ int k;
+ WebRtc_Word16 peaks_indq;
+ WebRtc_Word16 peakiq[PITCH_LAG_SPAN2];
+ WebRtc_Word32 corr;
+ WebRtc_Word32 corr32, corr_max32, corr_max_o32;
+ WebRtc_Word16 npkq;
+ WebRtc_Word16 best4q[4]={0,0,0,0};
+ WebRtc_Word32 xq[3],yq[1],fyq[1];
+ WebRtc_Word32 *fxq;
+ WebRtc_Word32 best_lag1q, best_lag2q;
+ WebRtc_Word32 tmp32a,tmp32b,lag32,ratq;
+ WebRtc_Word16 start;
+ WebRtc_Word16 oldgQ12, tmp16a, tmp16b, gain_bias16,tmp16c, tmp16d, bias16;
+ WebRtc_Word32 tmp32c,tmp32d, tmp32e;
+ WebRtc_Word16 old_lagQ;
+ WebRtc_Word32 old_lagQ8;
+ WebRtc_Word32 lagsQ8[4];
+
+ old_lagQ = State->PFstr_wght.oldlagQ7; // Q7
+ old_lagQ8= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)old_lagQ,1); //Q8
+
+ oldgQ12= State->PFstr_wght.oldgainQ12;
+
+ crrvecQ8_1=&cv1q[1];
+ crrvecQ8_2=&cv2q[1];
+
+
+ /* copy old values from state buffer */
+ memcpy(buf_dec16, State->dec_buffer16, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2)));
+
+ /* decimation; put result after the old values */
+ WebRtcIsacfix_DecimateAllpass32(in, State->decimator_state32, PITCH_FRAME_LEN,
+ &buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
+
+ /* low-pass filtering */
+ start= PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2;
+ WebRtcSpl_FilterARFastQ12(&buf_dec16[start],&buf_dec16[start],(WebRtc_Word16*)kACoefQ12,3, PITCH_FRAME_LEN/2);
+
+ /* copy end part back into state buffer */
+ for (k = 0; k < (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2); k++)
+ State->dec_buffer16[k] = buf_dec16[k+PITCH_FRAME_LEN/2];
+
+
+ /* compute correlation for first and second half of the frame */
+ PCorr2Q32(buf_dec16, crrvecQ8_1);
+ PCorr2Q32(buf_dec16 + PITCH_CORR_STEP2, crrvecQ8_2);
+
+
+ /* bias towards pitch lag of previous frame */
+ tmp32a = Log2Q8((WebRtc_UWord32) old_lagQ8) - 2304; // log2(0.5*oldlag) in Q8
+ tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0;
+ gain_bias16 = (WebRtc_Word16) tmp32b; //Q12
+ if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12
+
+
+ for (k = 0; k < PITCH_LAG_SPAN2; k++)
+ {
+ if (crrvecQ8_1[k]>0) {
+ tmp32b = Log2Q8((WebRtc_UWord32) (k + (PITCH_MIN_LAG/2-2)));
+ tmp16a = (WebRtc_Word16) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
+ tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10
+ tmp16b = (WebRtc_Word16) tmp32c; // Q10 & <8
+ tmp32d = WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 177 , 8); // mult with ln2 in Q8
+ tmp16c = (WebRtc_Word16) tmp32d; // Q10 & <4
+ tmp16d = Exp2Q10((WebRtc_Word16) -tmp16c); //Q10
+ tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10 & * 0.5
+ bias16 = (WebRtc_Word16) (1024 + tmp32c); // Q10
+ tmp32b = Log2Q8((WebRtc_UWord32) bias16) - 2560; // Q10 in -> Q8 out with 10*2^8 offset
+ crrvecQ8_1[k] += tmp32b ; // -10*2^8 offset
+ }
+ }
+
+ /* taper correlation functions */
+ for (k = 0; k < 3; k++) {
+ crrvecQ8_1[k] += kLogLagWinQ8[k];
+ crrvecQ8_2[k] += kLogLagWinQ8[k];
+
+ crrvecQ8_1[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k];
+ crrvecQ8_2[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k];
+ }
+
+
+ /* Make zeropadded corr vectors */
+ cv1q[0]=0;
+ cv2q[0]=0;
+ cv1q[PITCH_LAG_SPAN2+1]=0;
+ cv2q[PITCH_LAG_SPAN2+1]=0;
+ corr_max32 = 0;
+
+ for (k = 1; k <= PITCH_LAG_SPAN2; k++)
+ {
+
+
+ corr32=crrvecQ8_1[k-1];
+ if (corr32 > corr_max32)
+ corr_max32 = corr32;
+
+ corr32=crrvecQ8_2[k-1];
+ corr32 += -4; // Compensate for later (log2(0.99))
+
+ if (corr32 > corr_max32)
+ corr_max32 = corr32;
+
+ }
+
+ /* threshold value to qualify as a peak */
+ // corr_max32 += -726; // log(0.14)/log(2.0) in Q8
+ corr_max32 += -1000; // log(0.14)/log(2.0) in Q8
+ corr_max_o32 = corr_max32;
+
+
+ /* find peaks in corr1 */
+ peaks_indq = 0;
+ for (k = 1; k <= PITCH_LAG_SPAN2; k++)
+ {
+ corr32=cv1q[k];
+ if (corr32>corr_max32) { // Disregard small peaks
+ if ((corr32>=cv1q[k-1]) && (corr32>cv1q[k+1])) { // Peak?
+ peakvq[peaks_indq] = corr32;
+ peakiq[peaks_indq++] = k;
+ }
+ }
+ }
+
+
+ /* find highest interpolated peak */
+ corr_max32=0;
+ best_lag1q =0;
+ if (peaks_indq > 0) {
+ FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
+ npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
+
+ for (k=0;k<npkq;k++) {
+
+ lag32 = peakiq[best4q[k]];
+ fxq = &cv1q[peakiq[best4q[k]]-1];
+ xq[0]= lag32;
+ xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8);
+ Intrp1DQ8(xq, fxq, yq, fyq);
+
+ tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
+ /* Bias towards short lags */
+ /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */
+ tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -42, 8);
+ tmp32c= tmp32b + 256;
+ *fyq += tmp32c;
+ if (*fyq > corr_max32) {
+ corr_max32 = *fyq;
+ best_lag1q = *yq;
+ }
+ }
+ tmp32a = best_lag1q - OFFSET_Q8;
+ tmp32b = WEBRTC_SPL_LSHIFT_W32(tmp32a, 1);
+ lagsQ8[0] = tmp32b + PITCH_MIN_LAG_Q8;
+ lagsQ8[1] = lagsQ8[0];
+ } else {
+ lagsQ8[0] = old_lagQ8;
+ lagsQ8[1] = lagsQ8[0];
+ }
+
+ /* Bias towards constant pitch */
+ tmp32a = lagsQ8[0] - PITCH_MIN_LAG_Q8;
+ ratq = WEBRTC_SPL_RSHIFT_W32(tmp32a, 1) + OFFSET_Q8;
+
+ for (k = 1; k <= PITCH_LAG_SPAN2; k++)
+ {
+ tmp32a = WEBRTC_SPL_LSHIFT_W32(k, 7); // 0.5*k Q8
+ tmp32b = (WebRtc_Word32) (WEBRTC_SPL_LSHIFT_W32(tmp32a, 1)) - ratq; // Q8
+ tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32b, (WebRtc_Word16) tmp32b, 8); // Q8
+
+ tmp32b = (WebRtc_Word32) tmp32c + (WebRtc_Word32) WEBRTC_SPL_RSHIFT_W32(ratq, 1); // (k-r)^2 + 0.5 * r Q8
+ tmp32c = Log2Q8((WebRtc_UWord32) tmp32a) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
+ tmp32d = Log2Q8((WebRtc_UWord32) tmp32b) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
+ tmp32e = tmp32c -tmp32d;
+
+ cv2q[k] += WEBRTC_SPL_RSHIFT_W32(tmp32e, 1);
+
+ }
+
+ /* find peaks in corr2 */
+ corr_max32 = corr_max_o32;
+ peaks_indq = 0;
+
+ for (k = 1; k <= PITCH_LAG_SPAN2; k++)
+ {
+ corr=cv2q[k];
+ if (corr>corr_max32) { // Disregard small peaks
+ if ((corr>=cv2q[k-1]) && (corr>cv2q[k+1])) { // Peak?
+ peakvq[peaks_indq] = corr;
+ peakiq[peaks_indq++] = k;
+ }
+ }
+ }
+
+
+
+ /* find highest interpolated peak */
+ corr_max32 = 0;
+ best_lag2q =0;
+ if (peaks_indq > 0) {
+
+ FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
+ npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
+ for (k=0;k<npkq;k++) {
+
+ lag32 = peakiq[best4q[k]];
+ fxq = &cv2q[peakiq[best4q[k]]-1];
+
+ xq[0]= lag32;
+ xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8);
+ Intrp1DQ8(xq, fxq, yq, fyq);
+
+ /* Bias towards short lags */
+ /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */
+ tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
+ tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -82, 8);
+ tmp32c= tmp32b + 256;
+ *fyq += tmp32c;
+ if (*fyq > corr_max32) {
+ corr_max32 = *fyq;
+ best_lag2q = *yq;
+ }
+ }
+
+ tmp32a = best_lag2q - OFFSET_Q8;
+ tmp32b = WEBRTC_SPL_LSHIFT_W32(tmp32a, 1);
+ lagsQ8[2] = tmp32b + PITCH_MIN_LAG_Q8;
+ lagsQ8[3] = lagsQ8[2];
+ } else {
+ lagsQ8[2] = lagsQ8[0];
+ lagsQ8[3] = lagsQ8[0];
+ }
+
+ lagsQ7[0]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[0], 1);
+ lagsQ7[1]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[1], 1);
+ lagsQ7[2]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[2], 1);
+ lagsQ7[3]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[3], 1);
+
+
+}
+
+
+
+void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *inn, /* PITCH_FRAME_LEN samples */
+ WebRtc_Word16 *outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+ PitchAnalysisStruct *State,
+ WebRtc_Word16 *PitchLags_Q7,
+ WebRtc_Word16 *PitchGains_Q12)
+{
+ WebRtc_Word16 inbufQ0[PITCH_FRAME_LEN + QLOOKAHEAD];
+ WebRtc_Word16 k;
+
+ /* inital pitch estimate */
+ WebRtcIsacfix_InitialPitch(inn, State, PitchLags_Q7);
+
+
+ /* Calculate gain */
+ WebRtcIsacfix_PitchFilterGains(inn, &(State->PFstr_wght), PitchLags_Q7, PitchGains_Q12);
+
+ /* concatenate previous input's end and current input */
+ for (k = 0; k < QLOOKAHEAD; k++) {
+ inbufQ0[k] = State->inbuf[k];
+ }
+ for (k = 0; k < PITCH_FRAME_LEN; k++) {
+ inbufQ0[k+QLOOKAHEAD] = (WebRtc_Word16) inn[k];
+ }
+
+ /* lookahead pitch filtering for masking analysis */
+ WebRtcIsacfix_PitchFilter(inbufQ0, outQ0, &(State->PFstr), PitchLags_Q7,PitchGains_Q12, 2);
+
+
+ /* store last part of input */
+ for (k = 0; k < QLOOKAHEAD; k++) {
+ State->inbuf[k] = inbufQ0[k + PITCH_FRAME_LEN];
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h
new file mode 100644
index 0000000..afdc978
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.h
+ *
+ * Pitch functions
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
+
+#include "structs.h"
+
+
+
+void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *in, /* PITCH_FRAME_LEN samples */
+ WebRtc_Word16 *outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+ PitchAnalysisStruct *State,
+ WebRtc_Word16 *lagsQ7,
+ WebRtc_Word16 *PitchGains_Q12);
+
+
+void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in,
+ PitchAnalysisStruct *State,
+ WebRtc_Word16 *qlags);
+
+void WebRtcIsacfix_PitchFilter(WebRtc_Word16 *indatFix,
+ WebRtc_Word16 *outdatQQ,
+ PitchFiltstr *pfp,
+ WebRtc_Word16 *lagsQ7,
+ WebRtc_Word16 *gainsQ12,
+ WebRtc_Word16 type);
+
+void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16 *indatQ0,
+ PitchFiltstr *pfp,
+ WebRtc_Word16 *lagsQ7,
+ WebRtc_Word16 *gainsQ12);
+
+
+
+void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
+ WebRtc_Word32 *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
+ WebRtc_Word16 N, /* number of input samples */
+ WebRtc_Word16 *out); /* array of size N/2 */
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c
new file mode 100644
index 0000000..87af153
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c
@@ -0,0 +1,331 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimatorfilter.c
+ *
+ * Pitch filter functions
+ *
+ */
+
+#include <string.h>
+
+#include "pitch_estimator.h"
+
+
+/* Filter coefficicients in Q15 */
+static const WebRtc_Word16 kDampFilter[PITCH_DAMPORDER] = {
+ -2294, 8192, 20972, 8192, -2294
+};
+
+/* Interpolation coefficients; generated by design_pitch_filter.m.
+ * Coefficients are stored in Q14.
+ */
+static const WebRtc_Word16 kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+ {-367, 1090, -2706, 9945, 10596, -3318, 1626, -781, 287},
+ {-325, 953, -2292, 7301, 12963, -3320, 1570, -743, 271},
+ {-240, 693, -1622, 4634, 14809, -2782, 1262, -587, 212},
+ {-125, 358, -817, 2144, 15982, -1668, 721, -329, 118},
+ { 0, 0, -1, 1, 16380, 1, -1, 0, 0},
+ { 118, -329, 721, -1668, 15982, 2144, -817, 358, -125},
+ { 212, -587, 1262, -2782, 14809, 4634, -1622, 693, -240},
+ { 271, -743, 1570, -3320, 12963, 7301, -2292, 953, -325}
+};
+
+
+
+
+static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal, WebRtc_Word16 qDomain) {
+ WebRtc_Word32 intgr;
+ WebRtc_Word32 roundVal;
+
+ roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, qDomain-1);
+ intgr = WEBRTC_SPL_RSHIFT_W32(fixVal+roundVal, qDomain);
+
+ return intgr;
+}
+
+void WebRtcIsacfix_PitchFilter(WebRtc_Word16 *indatQQ, /* Q10 if type is 1 or 4, Q0 if type is 2 */
+ WebRtc_Word16 *outdatQQ,
+ PitchFiltstr *pfp,
+ WebRtc_Word16 *lagsQ7,
+ WebRtc_Word16 *gainsQ12,
+ WebRtc_Word16 type)
+{
+ int k, n, m, ind;
+ WebRtc_Word16 sign = 1;
+ WebRtc_Word16 inystateQQ[PITCH_DAMPORDER];
+ WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE+QLOOKAHEAD];
+ WebRtc_Word16 Gain = 21299; /* 1.3 in Q14 */
+ WebRtc_Word16 DivFactor = 6553; /* 0.2 in Q15 */
+ WebRtc_Word16 oldLagQ7, oldGainQ12,
+ lagdeltaQ7, curLagQ7,
+ gaindeltaQ12, curGainQ12;
+ WebRtc_Word16 tmpW16, indW16=0, frcQQ, cnt=0, pos, pos2;
+ const WebRtc_Word16 *fracoeffQQ=NULL;
+ WebRtc_Word32 tmpW32;
+
+ if (type==4)
+ sign = -1;
+
+ /* Set up buffer and states */
+ memcpy(ubufQQ, pfp->ubufQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
+ memcpy(inystateQQ, pfp->ystateQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_DAMPORDER));
+
+ /* Get old lag and gain value from memory */
+ oldLagQ7 = pfp->oldlagQ7;
+ oldGainQ12 = pfp->oldgainQ12;
+
+ if (type==4) {
+ /* make output more periodic */
+ /* Fixed 1.3 = 21299 in Q14 */
+ for (k=0;k<PITCH_SUBFRAMES;k++)
+ gainsQ12[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainsQ12[k], Gain, 14);
+ }
+
+ /* No interpolation if pitch lag step is big */
+ if ((WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(lagsQ7[0], 3), 1) < oldLagQ7) ||
+ (lagsQ7[0] > WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(oldLagQ7, 3), 1))) {
+ oldLagQ7 = lagsQ7[0];
+ oldGainQ12 = gainsQ12[0];
+ }
+
+ ind=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdeltaQ7 = lagsQ7[k]-oldLagQ7;
+ lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(lagdeltaQ7,DivFactor,15);
+ curLagQ7 = oldLagQ7;
+ gaindeltaQ12 = gainsQ12[k]-oldGainQ12;
+ gaindeltaQ12 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gaindeltaQ12,DivFactor,15);
+
+ curGainQ12 = oldGainQ12;
+ oldLagQ7 = lagsQ7[k];
+ oldGainQ12 = gainsQ12[k];
+
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+
+ if (cnt == 0) { /* Update parameters */
+
+ curGainQ12 += gaindeltaQ12;
+ curLagQ7 += lagdeltaQ7;
+ indW16 = (WebRtc_Word16)CalcLrIntQ(curLagQ7,7);
+ tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16,7);
+ tmpW16 -= curLagQ7;
+ frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16,4);
+ frcQQ += 4;
+
+ if(frcQQ==PITCH_FRACS)
+ frcQQ=0;
+ fracoeffQQ = kIntrpCoef[frcQQ];
+
+ cnt=12;
+ }
+
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystateQQ[m] = inystateQQ[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - (indW16 + 2);
+
+ tmpW32=0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos2+m], fracoeffQQ[m]);
+
+ /* Saturate to avoid overflow in tmpW16 */
+ tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104);
+ tmpW32 += 8192;
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
+
+ inystateQQ[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(curGainQ12, tmpW16,12); /* Multiply with gain */
+
+ /* Low pass filter */
+ tmpW32=0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ tmpW32 += WEBRTC_SPL_MUL_16_16(inystateQQ[m], kDampFilter[m]);
+
+ /* Saturate to avoid overflow in tmpW16 */
+ tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208);
+ tmpW32 += 16384;
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,15);
+
+ /* Subtract from input and update buffer */
+ tmpW32 = indatQQ[ind] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
+ outdatQQ[ind] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
+ tmpW32 = indatQQ[ind] + (WebRtc_Word32)outdatQQ[ind];
+ ubufQQ[pos] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
+
+ ind++;
+ cnt--;
+ }
+ }
+
+
+ /* Export buffer and states */
+ memcpy(pfp->ubufQQ, ubufQQ+PITCH_FRAME_LEN, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
+ memcpy(pfp->ystateQQ, inystateQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_DAMPORDER));
+
+ pfp->oldlagQ7 = oldLagQ7;
+ pfp->oldgainQ12 = oldGainQ12;
+
+ if (type==2) {
+ /* Filter look-ahead segment */
+ for (n=0;n<QLOOKAHEAD;n++) {
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystateQQ[m] = inystateQQ[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2= pos - (indW16 + 2);
+
+ tmpW32=0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos2+m], fracoeffQQ[m]);
+
+ if (tmpW32<536862720) {//536870912)
+ tmpW32 += 8192;
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
+ } else
+ tmpW16= 32767;
+ inystateQQ[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(curGainQ12, tmpW16,12); /* Multiply with gain */
+
+ /* Low pass filter */
+ tmpW32=0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ tmpW32 += WEBRTC_SPL_MUL_16_16(inystateQQ[m], kDampFilter[m]);
+ if (tmpW32<1073725440) { //1073741824)
+ tmpW32 += 16384;
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,15);
+ } else
+ tmpW16 = 32767;
+
+ /* Subtract from input and update buffer */
+ tmpW32 = indatQQ[ind] - (WebRtc_Word32)tmpW16;
+ outdatQQ[ind] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
+ tmpW32 = indatQQ[ind] + (WebRtc_Word32)outdatQQ[ind];
+ ubufQQ[pos] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
+
+ ind++;
+ }
+
+ }
+
+
+}
+
+
+void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16 *indatQ0,
+ PitchFiltstr *pfp,
+ WebRtc_Word16 *lagsQ7,
+ WebRtc_Word16 *gainsQ12)
+{
+ int k, n, m, ind;
+
+ WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE];
+ WebRtc_Word16 oldLagQ7,lagdeltaQ7, curLagQ7;
+ WebRtc_Word16 DivFactor = 6553;
+ const WebRtc_Word16 *fracoeffQQ = NULL;
+ WebRtc_Word16 scale;
+ WebRtc_Word16 cnt=0, pos, pos3QQ, frcQQ, indW16 = 0, tmpW16;
+ WebRtc_Word32 tmpW32, tmp2W32, csum1QQ, esumxQQ;
+
+ /* Set up buffer and states */
+ memcpy(ubufQQ, pfp->ubufQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
+ oldLagQ7 = pfp->oldlagQ7;
+
+ /* No interpolation if pitch lag step is big */
+ if ((WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(lagsQ7[0], 3), 1) < oldLagQ7) ||
+ (lagsQ7[0] > WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(oldLagQ7, 3), 1))) {
+ oldLagQ7 = lagsQ7[0];
+ }
+
+ ind=0;
+ scale=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdeltaQ7 = lagsQ7[k]-oldLagQ7;
+ lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(lagdeltaQ7,DivFactor,15);
+ curLagQ7 = oldLagQ7;
+ oldLagQ7 = lagsQ7[k];
+
+ csum1QQ=1;
+ esumxQQ=1;
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+
+ if (cnt == 0) { /* Update parameters */
+ curLagQ7 += lagdeltaQ7;
+ indW16 = (WebRtc_Word16)CalcLrIntQ(curLagQ7,7);
+ tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16,7);
+ tmpW16 -= curLagQ7;
+ frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16,4);
+ frcQQ += 4;
+
+ if(frcQQ==PITCH_FRACS)
+ frcQQ=0;
+ fracoeffQQ = kIntrpCoef[frcQQ];
+
+ cnt=12;
+ }
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos3QQ = pos - (indW16 + 4);
+
+ tmpW32=0;
+ for (m=0;m<PITCH_FRACORDER;m++){
+ tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos3QQ+m], fracoeffQQ[m]);
+ }
+
+ /* Subtract from input and update buffer */
+ ubufQQ[pos] = indatQ0[ind];
+
+ tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32);
+ tmpW32 += 8192;
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
+ tmpW32 = WEBRTC_SPL_MUL_16_16(tmpW16, tmpW16);
+
+ if ((tmp2W32>1073700000) || (csum1QQ>1073700000) || (tmpW32>1073700000) || (esumxQQ>1073700000)) {//2^30
+ scale++;
+ csum1QQ = WEBRTC_SPL_RSHIFT_W32(csum1QQ,1);
+ esumxQQ = WEBRTC_SPL_RSHIFT_W32(esumxQQ,1);
+ }
+ tmp2W32 = WEBRTC_SPL_RSHIFT_W32(tmp2W32,scale);
+ csum1QQ += tmp2W32;
+ tmpW32 = WEBRTC_SPL_RSHIFT_W32(tmpW32,scale);
+ esumxQQ += tmpW32;
+
+ ind++;
+ cnt--;
+ }
+
+ if (csum1QQ<esumxQQ) {
+ tmp2W32=WebRtcSpl_DivResultInQ31(csum1QQ,esumxQQ);
+
+ /* Gain should be half the correlation */
+ tmpW32=WEBRTC_SPL_RSHIFT_W32(tmp2W32,20);
+ } else
+ tmpW32=4096;
+ gainsQ12[k]=(WebRtc_Word16)WEBRTC_SPL_SAT(PITCH_MAX_GAIN_Q12, tmpW32, 0);
+
+ }
+
+ /* Export buffer and states */
+ memcpy(pfp->ubufQQ, ubufQQ+PITCH_FRAME_LEN, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
+ pfp->oldlagQ7 = lagsQ7[PITCH_SUBFRAMES-1];
+ pfp->oldgainQ12 = gainsQ12[PITCH_SUBFRAMES-1];
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c
new file mode 100644
index 0000000..50ea658
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c
@@ -0,0 +1,149 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_gain_tables.c
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#include "pitch_gain_tables.h"
+
+
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+
+/* cdf for quantized pitch filter gains */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255] = {
+ 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956,
+ 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411,
+ 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722,
+ 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227,
+ 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098,
+ 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705,
+ 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029,
+ 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767,
+ 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132,
+ 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467,
+ 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165,
+ 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860,
+ 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901,
+ 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525,
+ 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535
+};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3] = {
+ -7, -2, -1
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3] = {
+ 0, 3, 1
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2] = {
+ 18, 3
+};
+
+/* size of cdf table */
+const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1] = {
+ 256
+};
+
+/* mean values of pitch filter gains in FIXED point Q12 */
+const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144] = {
+ 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839,
+ 1843, 1843, 1843, 1843, 1843, 1843, 1843, 814, 846, 1092, 1013,
+ 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843, 1843,
+ 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263,
+ 1380, 1447, 1559, 1676, 1645, 1749, 1843, 1843, 1843, 1843, 81,
+ 477, 563, 611, 706, 806, 849, 1012, 1192, 1128, 1330, 1489,
+ 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488,
+ 575, 602, 741, 890, 835, 1079, 1196, 1182, 1376, 1519, 1506,
+ 1680, 1843, 0, 47, 97, 69, 289, 381, 385, 474, 617,
+ 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0,
+ 0, 0, 112, 120, 190, 283, 442, 343, 526, 809, 684,
+ 935, 1134, 1020, 1265, 1506, 0, 0, 0, 0, 0, 0,
+ 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020,
+ 1265
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144] = {
+ 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784,
+ 1606, 1843, 1843, 1711, 1843, 1843, 1814, 1389, 1275, 1040, 1564,
+ 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720, 1475,
+ 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253,
+ 1111, 1495, 1343, 1178, 1770, 1465, 1234, 1814, 1581, 1342, 1040,
+ 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277, 1094, 981,
+ 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648,
+ 540, 948, 744, 572, 1009, 844, 636, 1234, 934, 685, 1342,
+ 1217, 984, 537, 318, 124, 603, 423, 350, 687, 479, 322,
+ 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27,
+ 0, 397, 222, 38, 513, 271, 124, 624, 325, 157, 737,
+ 484, 233, 849, 597, 343, 27, 0, 0, 141, 0, 0,
+ 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343,
+ 87
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144] = {
+ 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639,
+ 1393, 1760, 1525, 1285, 1656, 1419, 1176, 1835, 1718, 1475, 1841,
+ 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299, 1040,
+ 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260,
+ 1115, 1398, 1151, 1025, 1172, 1080, 790, 1176, 928, 677, 1475,
+ 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057, 893, 800,
+ 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830,
+ 710, 875, 751, 601, 795, 642, 583, 790, 544, 475, 677,
+ 474, 140, 987, 750, 482, 697, 573, 450, 691, 487, 303,
+ 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484,
+ 229, 624, 348, 153, 441, 261, 136, 397, 166, 51, 283,
+ 27, 0, 168, 0, 0, 484, 229, 0, 370, 57, 0,
+ 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0,
+ 0
+};
+
+
+const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144] = {
+ 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434,
+ 1656, 843, 1092, 1336, 504, 757, 1007, 1843, 1843, 1843, 1838,
+ 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821, 1092,
+ 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268,
+ 1409, 805, 961, 1131, 444, 670, 843, 0, 249, 504, 1425,
+ 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490, 704, 867,
+ 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073,
+ 1283, 610, 759, 939, 278, 464, 645, 0, 200, 270, 0,
+ 0, 0, 935, 1163, 1410, 528, 790, 1068, 377, 499, 717,
+ 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935,
+ 1182, 343, 551, 735, 161, 262, 423, 0, 55, 27, 0,
+ 0, 0, 0, 0, 0, 430, 684, 935, 87, 377, 597,
+ 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0
+};
+
+
+
+/* transform matrix in Q12*/
+const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4] = {
+ { -2048, -2048, -2048, -2048 },
+ { 2748, 916, -916, -2748 },
+ { 2048, -2048, -2048, 2048 },
+ { 916, -2748, 2748, -916 }
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h
new file mode 100644
index 0000000..788e553
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_gain_tables.h
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
+
+#include "typedefs.h"
+
+
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+/* cdf for quantized pitch filter gains */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3];
+extern const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3];
+extern const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2];
+
+/* mean values of pitch filter gains in Q12*/
+extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144];
+extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144];
+extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144];
+extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144];
+
+/* size of cdf table */
+extern const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1];
+
+/* transform matrix */
+extern const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c
new file mode 100644
index 0000000..81700e4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c
@@ -0,0 +1,306 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_lag_tables.c
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#include "settings.h"
+#include "pitch_lag_tables.h"
+
+
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+
+/* tables for use with small pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127] = {
+ 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070,
+ 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269,
+ 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358,
+ 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213,
+ 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734,
+ 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288,
+ 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082,
+ 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273,
+ 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726,
+ 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359,
+ 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363,
+ 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856,
+ 65152, 65535, 65535, 65535, 65535, 65535, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20] = {
+ 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983,
+ 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2] = {
+ 0, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10] = {
+ 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535
+};
+
+const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4] = {
+ WebRtcIsacfix_kPitchLagCdf1Lo,
+ WebRtcIsacfix_kPitchLagCdf2Lo,
+ WebRtcIsacfix_kPitchLagCdf3Lo,
+ WebRtcIsacfix_kPitchLagCdf4Lo
+};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1] = {
+ 128
+};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4] = {
+ -140, -9, 0, -4
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4] = {
+ -20, 9, 0, 4
+};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3] = {
+ 10, 1, 5
+};
+
+/* mean values of pitch filter lags in Q10 */
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19] = {
+ -17627, -16207, -14409, -12319, -10253, -8200, -6054, -3986, -1948, -19,
+ 1937, 3974, 6064, 8155, 10229, 12270, 14296, 16127, 17520
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9] = {
+ -7949, -6063, -4036, -1941, 38, 1977, 4060, 6059
+};
+
+
+
+/* tables for use with medium pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255] = {
+ 0, 28, 61, 88, 121, 149, 233, 331, 475, 559,
+ 624, 661, 689, 712, 745, 791, 815, 843, 866, 922,
+ 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513,
+ 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770,
+ 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319,
+ 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185,
+ 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675,
+ 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421,
+ 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227,
+ 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861,
+ 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821,
+ 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285,
+ 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836,
+ 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741,
+ 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651,
+ 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048,
+ 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064,
+ 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755,
+ 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973,
+ 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602,
+ 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719,
+ 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688,
+ 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454,
+ 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902,
+ 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36] = {
+ 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608,
+ 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007,
+ 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949,
+ 65039, 65115, 65223, 65360, 65474, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2] = {
+ 0, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20] = {
+ 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347,
+ 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535
+};
+
+const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4] = {
+ WebRtcIsacfix_kPitchLagCdf1Mid,
+ WebRtcIsacfix_kPitchLagCdf2Mid,
+ WebRtcIsacfix_kPitchLagCdf3Mid,
+ WebRtcIsacfix_kPitchLagCdf4Mid
+};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1] = {
+ 256
+};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4] = {
+ -280, -17, 0, -9
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4] = {
+ -40, 17, 0, 9
+};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3] = {
+ 18, 1, 10
+};
+
+/* mean values of pitch filter lags in Q10 */
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35] = {
+ -17297, -16250, -15416, -14343, -13341, -12363, -11270,
+ -10355, -9122, -8217, -7172, -6083, -5102, -4004, -3060,
+ -1982, -952, -18, 935, 1976, 3040, 4032,
+ 5082, 6065, 7257, 8202, 9264, 10225, 11242,
+ 12234, 13337, 14336, 15374, 16187, 17347
+};
+
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19] = {
+ -8811, -8081, -7203, -6003, -5057, -4025, -2983, -1964,
+ -891, 29, 921, 1920, 2988, 4064, 5187, 6079, 7173, 8074, 8849
+};
+
+
+/* tables for use with large pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511] = {
+ 0, 7, 18, 33, 69, 105, 156, 228, 315, 612,
+ 680, 691, 709, 724, 735, 738, 742, 746, 749, 753,
+ 756, 760, 764, 774, 782, 785, 789, 796, 800, 803,
+ 807, 814, 818, 822, 829, 832, 847, 854, 858, 869,
+ 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064,
+ 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136,
+ 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401,
+ 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477,
+ 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542,
+ 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665,
+ 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759,
+ 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987,
+ 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251,
+ 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378,
+ 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935,
+ 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127,
+ 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427,
+ 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760,
+ 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271,
+ 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904,
+ 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382,
+ 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022,
+ 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271,
+ 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024,
+ 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842,
+ 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536,
+ 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358,
+ 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324,
+ 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392,
+ 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384,
+ 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614,
+ 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555,
+ 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463,
+ 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649,
+ 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068,
+ 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588,
+ 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168,
+ 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077,
+ 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173,
+ 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894,
+ 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607,
+ 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078,
+ 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566,
+ 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693,
+ 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839,
+ 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558,
+ 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115,
+ 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68] = {
+ 0, 7, 11, 22, 37, 52, 56, 59, 81, 85,
+ 89, 96, 115, 130, 137, 152, 170, 181, 193, 200,
+ 207, 233, 237, 259, 289, 318, 363, 433, 592, 992,
+ 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584,
+ 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402,
+ 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476,
+ 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2] = {
+ 0, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35] = {
+ 0, 7, 19, 30, 41, 48, 63, 74, 82, 96,
+ 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341,
+ 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453,
+ 65465, 65476, 65490, 65509, 65528, 65535
+};
+
+const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4] = {
+ WebRtcIsacfix_kPitchLagCdf1Hi,
+ WebRtcIsacfix_kPitchLagCdf2Hi,
+ WebRtcIsacfix_kPitchLagCdf3Hi,
+ WebRtcIsacfix_kPitchLagCdf4Hi
+};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1] = {
+ 512
+};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4] = {
+ -552, -34, 0, -16
+};
+
+const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4] = {
+ -80, 32, 0, 17
+};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3] = {
+ 34, 1, 18
+};
+
+/* mean values of pitch filter lags */
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67] = {
+ -17482, -16896, -16220, -15929, -15329, -14848, -14336, -13807, -13312, -12800, -12218, -11720,
+ -11307, -10649, -10396, -9742, -9148, -8668, -8297, -7718, -7155, -6656, -6231, -5600, -5129,
+ -4610, -4110, -3521, -3040, -2525, -2016, -1506, -995, -477, -5, 469, 991, 1510, 2025, 2526, 3079,
+ 3555, 4124, 4601, 5131, 5613, 6194, 6671, 7140, 7645, 8207, 8601, 9132, 9728, 10359, 10752, 11302,
+ 11776, 12288, 12687, 13204, 13759, 14295, 14810, 15360, 15764, 16350
+};
+
+
+const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34] = {
+ -8175, -7659, -7205, -6684, -6215, -5651, -5180, -4566, -4087, -3536, -3096,
+ -2532, -1990, -1482, -959, -440, 11, 451, 954, 1492, 2020, 2562, 3059,
+ 3577, 4113, 4618, 5134, 5724, 6060, 6758, 7015, 7716, 8066, 8741
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h
new file mode 100644
index 0000000..9517c29
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_lag_tables.h
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
+
+
+#include "typedefs.h"
+
+
+/********************* Pitch Filter Lag Coefficient Tables ************************/
+
+/* tables for use with small pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10];
+
+extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4];
+extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3];
+
+/* mean values of pitch filter lags */
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19];
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9];
+
+
+
+/* tables for use with medium pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20];
+
+extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4];
+extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3];
+
+/* mean values of pitch filter lags */
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35];
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19];
+
+
+/* tables for use with large pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2];
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35];
+
+extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4];
+extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3];
+
+/* mean values of pitch filter lags */
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67];
+extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34];
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h
new file mode 100644
index 0000000..da88ba2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * settings.h
+ *
+ * Declaration of #defines used in the iSAC codec
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
+
+
+/* sampling frequency (Hz) */
+#define FS 16000
+/* 1.5 times Sampling frequency */
+#define FS_1_HALF (WebRtc_UWord32) 24000
+/* Three times Sampling frequency */
+#define FS3 (WebRtc_UWord32) 48000
+/* Eight times Sampling frequency */
+#define FS8 (WebRtc_UWord32) 128000
+
+/* number of samples per frame (either 480 (30ms) or 960 (60ms)) */
+#define INITIAL_FRAMESAMPLES 960
+
+/* miliseconds */
+#define FRAMESIZE 30
+/* number of samples per frame processed in the encoder (30ms) */
+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
+#define FRAMESAMPLES_HALF 240
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms 160 /* ((10*FS)/1000) */
+/* Number of samples per 1 ms */
+#define SAMPLES_PER_MSEC 16
+/* number of subframes */
+#define SUBFRAMES 6
+/* length of a subframe */
+#define UPDATE 80
+/* length of half a subframe (low/high band) */
+#define HALF_SUBFRAMELEN 40 /* (UPDATE/2) */
+/* samples of look ahead (in a half-band, so actually half the samples of look ahead @ FS) */
+#define QLOOKAHEAD 24 /* 3 ms */
+
+/* order of AR model in spectral entropy coder */
+#define AR_ORDER 6
+#define MAX_ORDER 13
+#define LEVINSON_MAX_ORDER 12
+
+/* window length (masking analysis) */
+#define WINLEN 256
+/* order of low-band pole filter used to approximate masking curve */
+#define ORDERLO 12
+/* order of hi-band pole filter used to approximate masking curve */
+#define ORDERHI 6
+
+#define KLT_NUM_AVG_GAIN 0
+#define KLT_NUM_AVG_SHAPE 0
+#define KLT_NUM_MODELS 3
+#define LPC_SHAPE_ORDER 18 /* (ORDERLO + ORDERHI) */
+
+#define KLT_ORDER_GAIN 12 /* (2 * SUBFRAMES) */
+#define KLT_ORDER_SHAPE 108 /* (LPC_SHAPE_ORDER * SUBFRAMES) */
+
+
+
+/* order for post_filter_bank */
+#define POSTQORDER 3
+/* order for pre-filterbank */
+#define QORDER 3
+/* for decimator */
+#define ALLPASSSECTIONS 2
+/* The number of composite all-pass filter factors */
+#define NUMBEROFCOMPOSITEAPSECTIONS 4
+
+/* The number of all-pass filter factors in an upper or lower channel*/
+#define NUMBEROFCHANNELAPSECTIONS 2
+
+
+
+#define DPMIN_Q10 -10240 /* -10.00 in Q10 */
+#define DPMAX_Q10 10240 /* 10.00 in Q10 */
+#define MINBITS_Q10 10240 /* 10.0 in Q10 */
+
+
+/* array size for byte stream in number of Word16. */
+#define STREAM_MAXW16 300 /* The old maximum size still needed for the decoding */
+#define STREAM_MAXW16_30MS 100 /* 100 Word16 = 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
+#define STREAM_MAXW16_60MS 200 /* 200 Word16 = 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
+
+
+/* storage size for bit counts */
+//#define BIT_COUNTER_SIZE 30
+/* maximum order of any AR model or filter */
+#define MAX_AR_MODEL_ORDER 12
+
+/* Maximum number of iterations allowed to limit payload size */
+#define MAX_PAYLOAD_LIMIT_ITERATION 1
+
+/* Bandwidth estimator */
+
+#define MIN_ISAC_BW 10000 /* Minimum bandwidth in bits per sec */
+#define MAX_ISAC_BW 32000 /* Maxmum bandwidth in bits per sec */
+#define MIN_ISAC_MD 5 /* Minimum Max Delay in ?? */
+#define MAX_ISAC_MD 25 /* Maxmum Max Delay in ?? */
+#define DELAY_CORRECTION_MAX 717
+#define DELAY_CORRECTION_MED 819
+#define Thld_30_60 18000
+#define Thld_60_30 27000
+
+/* assumed header size; we don't know the exact number (header compression may be used) */
+#define HEADER_SIZE 35 /* bytes */
+#define INIT_FRAME_LEN 60
+#define INIT_BN_EST 20000
+#define INIT_BN_EST_Q7 2560000 /* 20 kbps in Q7 */
+#define INIT_REC_BN_EST_Q5 789312 /* INIT_BN_EST + INIT_HDR_RATE in Q5 */
+
+/* 8738 in Q18 is ~ 1/30 */
+/* #define INIT_HDR_RATE (((HEADER_SIZE * 8 * 1000) * 8738) >> NUM_BITS_TO_SHIFT (INIT_FRAME_LEN)) */
+#define INIT_HDR_RATE 4666
+/* number of packets in a row for a high rate burst */
+#define BURST_LEN 3
+/* ms, max time between two full bursts */
+#define BURST_INTERVAL 800
+/* number of packets in a row for initial high rate burst */
+#define INIT_BURST_LEN 5
+/* bits/s, rate for the first BURST_LEN packets */
+#define INIT_RATE 10240000 /* INIT_BN_EST in Q9 */
+
+
+/* For pitch analysis */
+#define PITCH_FRAME_LEN 240 /* (FRAMESAMPLES/2) 30 ms */
+#define PITCH_MAX_LAG 140 /* 57 Hz */
+#define PITCH_MIN_LAG 20 /* 400 Hz */
+#define PITCH_MIN_LAG_Q8 5120 /* 256 * PITCH_MIN_LAG */
+#define OFFSET_Q8 768 /* 256 * 3 */
+
+#define PITCH_MAX_GAIN_Q12 1843 /* 0.45 */
+#define PITCH_LAG_SPAN2 65 /* (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5) */
+#define PITCH_CORR_LEN2 60 /* 15 ms */
+#define PITCH_CORR_STEP2 60 /* (PITCH_FRAME_LEN/4) */
+#define PITCH_SUBFRAMES 4
+#define PITCH_SUBFRAME_LEN 60 /* (PITCH_FRAME_LEN/PITCH_SUBFRAMES) */
+
+/* For pitch filter */
+#define PITCH_BUFFSIZE 190 /* (PITCH_MAX_LAG + 50) Extra 50 for fraction and LP filters */
+#define PITCH_INTBUFFSIZE 430 /* (PITCH_FRAME_LEN+PITCH_BUFFSIZE) */
+#define PITCH_FRACS 8
+#define PITCH_FRACORDER 9
+#define PITCH_DAMPORDER 5
+
+
+/* Order of high pass filter */
+#define HPORDER 2
+
+
+/* PLC */
+#define DECAY_RATE 10 /* Q15, 20% of decay every lost frame apllied linearly sample by sample*/
+#define PLC_WAS_USED 1
+#define PLC_NOT_USED 3
+#define RECOVERY_OVERLAP 80
+#define RESAMP_RES 256
+#define RESAMP_RES_BIT 8
+
+
+
+/* Define Error codes */
+/* 6000 General */
+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
+#define ISAC_MODE_MISMATCH 6020
+#define ISAC_DISALLOWED_BOTTLENECK 6030
+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
+/* 6200 Bandwidth estimator */
+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
+/* 6400 Encoder */
+#define ISAC_ENCODER_NOT_INITIATED 6410
+#define ISAC_DISALLOWED_CODING_MODE 6420
+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
+/* 6600 Decoder */
+#define ISAC_DECODER_NOT_INITIATED 6610
+#define ISAC_EMPTY_PACKET 6620
+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
+#define ISAC_LENGTH_MISMATCH 6730
+/* 6800 Call setup formats */
+#define ISAC_INCOMPATIBLE_FORMATS 6810
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c
new file mode 100644
index 0000000..81b932f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c
@@ -0,0 +1,193 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * spectrum_ar_model_tables.c
+ *
+ * This file contains tables with AR coefficients, Gain coefficients
+ * and cosine tables.
+ *
+ */
+
+#include "spectrum_ar_model_tables.h"
+#include "settings.h"
+
+/********************* AR Coefficient Tables ************************/
+
+/* cdf for quantized reflection coefficient 1 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12] = {
+ 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531,
+ 65533, 65535
+};
+
+/* cdf for quantized reflection coefficient 2 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12] = {
+ 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531,
+ 65533, 65535
+};
+
+/* cdf for quantized reflection coefficient 3 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12] = {
+ 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531,
+ 65533, 65535
+};
+
+/* cdf for quantized reflection coefficient 4 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12] = {
+ 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531,
+ 65533, 65535
+};
+
+/* cdf for quantized reflection coefficient 5 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12] = {
+ 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531,
+ 65533, 65535
+};
+
+/* cdf for quantized reflection coefficient 6 */
+const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12] = {
+ 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531,
+ 65533, 65535
+};
+
+/* representation levels for quantized reflection coefficient 1 */
+const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11] = {
+ -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 2 */
+const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11] = {
+ -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 3 */
+const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11] = {
+ -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 4 */
+const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11] = {
+ -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 5 */
+const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11] = {
+ -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 6 */
+const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11] = {
+ -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
+};
+
+/* quantization boundary levels for reflection coefficients */
+const WebRtc_Word16 WebRtcIsacfix_kRcBound[12] = {
+ -32768, -31441, -27566, -21458, -13612, -4663,
+ 4663, 13612, 21458, 27566, 31441, 32767
+};
+
+/* initial index for AR reflection coefficient quantizer and cdf table search */
+const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[6] = {
+ 5, 5, 5, 5, 5, 5
+};
+
+/* pointers to AR cdf tables */
+const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER] = {
+ WebRtcIsacfix_kRc1Cdf,
+ WebRtcIsacfix_kRc2Cdf,
+ WebRtcIsacfix_kRc3Cdf,
+ WebRtcIsacfix_kRc4Cdf,
+ WebRtcIsacfix_kRc5Cdf,
+ WebRtcIsacfix_kRc6Cdf
+};
+
+/* pointers to AR representation levels tables */
+const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = {
+ WebRtcIsacfix_kRc1Levels,
+ WebRtcIsacfix_kRc2Levels,
+ WebRtcIsacfix_kRc3Levels,
+ WebRtcIsacfix_kRc4Levels,
+ WebRtcIsacfix_kRc5Levels,
+ WebRtcIsacfix_kRc6Levels
+};
+
+
+/******************** GAIN Coefficient Tables ***********************/
+
+/* cdf for Gain coefficient */
+const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19] = {
+ 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172,
+ 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535
+};
+
+/* representation levels for quantized squared Gain coefficient */
+const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18] = {
+ 128, 128, 128, 128, 128, 215, 364, 709, 1268,
+ 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000
+};
+
+/* quantization boundary levels for squared Gain coefficient */
+const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19] = {
+ 0, 21, 35, 59, 99, 166, 280, 475, 815, 1414,
+ 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF
+};
+
+/* pointers to Gain cdf table */
+const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1] = {
+ WebRtcIsacfix_kGainCdf
+};
+
+/* gain initial index for gain quantizer and cdf table search */
+const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1] = {
+ 11
+};
+
+
+/************************* Cosine Tables ****************************/
+
+/* cosine table */
+const WebRtc_Word16 WebRtcIsacfix_kCos[6][60] = {
+ { 512, 512, 511, 510, 508, 507, 505, 502, 499, 496,
+ 493, 489, 485, 480, 476, 470, 465, 459, 453, 447,
+ 440, 433, 426, 418, 410, 402, 394, 385, 376, 367,
+ 357, 348, 338, 327, 317, 306, 295, 284, 273, 262,
+ 250, 238, 226, 214, 202, 190, 177, 165, 152, 139,
+ 126, 113, 100, 87, 73, 60, 47, 33, 20, 7 },
+ { 512, 510, 508, 503, 498, 491, 483, 473, 462, 450,
+ 437, 422, 406, 389, 371, 352, 333, 312, 290, 268,
+ 244, 220, 196, 171, 145, 120, 93, 67, 40, 13,
+ -13, -40, -67, -93, -120, -145, -171, -196, -220, -244,
+ -268, -290, -312, -333, -352, -371, -389, -406, -422, -437,
+ -450, -462, -473, -483, -491, -498, -503, -508, -510, -512 },
+ { 512, 508, 502, 493, 480, 465, 447, 426, 402, 376,
+ 348, 317, 284, 250, 214, 177, 139, 100, 60, 20,
+ -20, -60, -100, -139, -177, -214, -250, -284, -317, -348,
+ -376, -402, -426, -447, -465, -480, -493, -502, -508, -512,
+ -512, -508, -502, -493, -480, -465, -447, -426, -402, -376,
+ -348, -317, -284, -250, -214, -177, -139, -100, -60, -20 },
+ { 511, 506, 495, 478, 456, 429, 398, 362, 322, 279,
+ 232, 183, 133, 80, 27, -27, -80, -133, -183, -232,
+ -279, -322, -362, -398, -429, -456, -478, -495, -506, -511,
+ -511, -506, -495, -478, -456, -429, -398, -362, -322, -279,
+ -232, -183, -133, -80, -27, 27, 80, 133, 183, 232,
+ 279, 322, 362, 398, 429, 456, 478, 495, 506, 511 },
+ { 511, 502, 485, 459, 426, 385, 338, 284, 226, 165,
+ 100, 33, -33, -100, -165, -226, -284, -338, -385, -426,
+ -459, -485, -502, -511, -511, -502, -485, -459, -426, -385,
+ -338, -284, -226, -165, -100, -33, 33, 100, 165, 226,
+ 284, 338, 385, 426, 459, 485, 502, 511, 511, 502,
+ 485, 459, 426, 385, 338, 284, 226, 165, 100, 33 },
+ { 510, 498, 473, 437, 389, 333, 268, 196, 120, 40,
+ -40, -120, -196, -268, -333, -389, -437, -473, -498, -510,
+ -510, -498, -473, -437, -389, -333, -268, -196, -120, -40,
+ 40, 120, 196, 268, 333, 389, 437, 473, 498, 510,
+ 510, 498, 473, 437, 389, 333, 268, 196, 120, 40,
+ -40, -120, -196, -268, -333, -389, -437, -473, -498, -510 }
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h
new file mode 100644
index 0000000..b506d0e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * spectrum_ar_model_tables.h
+ *
+ * This file contains definitions of tables with AR coefficients,
+ * Gain coefficients and cosine tables.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+
+#include "typedefs.h"
+#include "settings.h"
+
+
+/********************* AR Coefficient Tables ************************/
+/* cdf for quantized reflection coefficient 1 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12];
+
+/* cdf for quantized reflection coefficient 2 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12];
+
+/* cdf for quantized reflection coefficient 3 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12];
+
+/* cdf for quantized reflection coefficient 4 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12];
+
+/* cdf for quantized reflection coefficient 5 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12];
+
+/* cdf for quantized reflection coefficient 6 */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12];
+
+/* representation levels for quantized reflection coefficient 1 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11];
+
+/* representation levels for quantized reflection coefficient 2 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11];
+
+/* representation levels for quantized reflection coefficient 3 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11];
+
+/* representation levels for quantized reflection coefficient 4 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11];
+
+/* representation levels for quantized reflection coefficient 5 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11];
+
+/* representation levels for quantized reflection coefficient 6 */
+extern const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11];
+
+/* quantization boundary levels for reflection coefficients */
+extern const WebRtc_Word16 WebRtcIsacfix_kRcBound[12];
+
+/* initial indices for AR reflection coefficient quantizer and cdf table search */
+extern const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[AR_ORDER];
+
+/* pointers to AR cdf tables */
+extern const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
+
+/* pointers to AR representation levels tables */
+extern const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER];
+
+
+/******************** GAIN Coefficient Tables ***********************/
+/* cdf for Gain coefficient */
+extern const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19];
+
+/* representation levels for quantized Gain coefficient */
+extern const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18];
+
+/* squared quantization boundary levels for Gain coefficient */
+extern const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19];
+
+/* pointer to Gain cdf table */
+extern const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1];
+
+/* Gain initial index for gain quantizer and cdf table search */
+extern const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1];
+
+/************************* Cosine Tables ****************************/
+/* Cosine table */
+extern const WebRtc_Word16 WebRtcIsacfix_kCos[6][60];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h
new file mode 100644
index 0000000..54dffa9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h
@@ -0,0 +1,382 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
+
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+#include "settings.h"
+
+/* Bitstream struct for decoder */
+typedef struct Bitstreamstruct_dec {
+
+ WebRtc_UWord16 *stream; /* Pointer to bytestream to decode */
+ WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
+ WebRtc_UWord32 streamval;
+ WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
+ WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
+ /* 1 - both bytes are empty (we just filled the previous memory */
+
+} Bitstr_dec;
+
+/* Bitstream struct for encoder */
+typedef struct Bitstreamstruct_enc {
+
+ WebRtc_UWord16 stream[STREAM_MAXW16_60MS]; /* Vector for adding encoded bytestream */
+ WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
+ WebRtc_UWord32 streamval;
+ WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
+ WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
+ /* 1 - both bytes are empty (we just filled the previous memory */
+
+} Bitstr_enc;
+
+
+typedef struct {
+
+ WebRtc_Word16 DataBufferLoQ0[WINLEN];
+ WebRtc_Word16 DataBufferHiQ0[WINLEN];
+
+ WebRtc_Word32 CorrBufLoQQ[ORDERLO+1];
+ WebRtc_Word32 CorrBufHiQQ[ORDERHI+1];
+
+ WebRtc_Word16 CorrBufLoQdom[ORDERLO+1];
+ WebRtc_Word16 CorrBufHiQdom[ORDERHI+1];
+
+ WebRtc_Word32 PreStateLoGQ15[ORDERLO+1];
+ WebRtc_Word32 PreStateHiGQ15[ORDERHI+1];
+
+ WebRtc_UWord32 OldEnergy;
+
+} MaskFiltstr_enc;
+
+
+
+typedef struct {
+
+ WebRtc_Word16 PostStateLoGQ0[ORDERLO+1];
+ WebRtc_Word16 PostStateHiGQ0[ORDERHI+1];
+
+ WebRtc_UWord32 OldEnergy;
+
+} MaskFiltstr_dec;
+
+
+
+
+
+
+
+
+typedef struct {
+
+ //state vectors for each of the two analysis filters
+
+ WebRtc_Word32 INSTAT1_fix[2*(QORDER-1)];
+ WebRtc_Word32 INSTAT2_fix[2*(QORDER-1)];
+ WebRtc_Word16 INLABUF1_fix[QLOOKAHEAD];
+ WebRtc_Word16 INLABUF2_fix[QLOOKAHEAD];
+
+ /* High pass filter */
+ WebRtc_Word32 HPstates_fix[HPORDER];
+
+} PreFiltBankstr;
+
+
+typedef struct {
+
+ //state vectors for each of the two analysis filters
+ WebRtc_Word32 STATE_0_LOWER_fix[2*POSTQORDER];
+ WebRtc_Word32 STATE_0_UPPER_fix[2*POSTQORDER];
+
+ /* High pass filter */
+
+ WebRtc_Word32 HPstates1_fix[HPORDER];
+ WebRtc_Word32 HPstates2_fix[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+
+
+ /* data buffer for pitch filter */
+ WebRtc_Word16 ubufQQ[PITCH_BUFFSIZE];
+
+ /* low pass state vector */
+ WebRtc_Word16 ystateQQ[PITCH_DAMPORDER];
+
+ /* old lag and gain */
+ WebRtc_Word16 oldlagQ7;
+ WebRtc_Word16 oldgainQ12;
+
+} PitchFiltstr;
+
+
+
+typedef struct {
+
+ //for inital estimator
+ WebRtc_Word16 dec_buffer16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2];
+ WebRtc_Word32 decimator_state32[2*ALLPASSSECTIONS+1];
+ WebRtc_Word16 inbuf[QLOOKAHEAD];
+
+ PitchFiltstr PFstr_wght;
+ PitchFiltstr PFstr;
+
+
+} PitchAnalysisStruct;
+
+
+typedef struct {
+ /* Parameters used in PLC to avoid re-computation */
+
+ /* --- residual signals --- */
+ WebRtc_Word16 prevPitchInvIn[FRAMESAMPLES/2];
+ WebRtc_Word16 prevPitchInvOut[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90
+ WebRtc_Word32 prevHP[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90
+
+
+ WebRtc_Word16 decayCoeffPriodic; /* how much to supress a sample */
+ WebRtc_Word16 decayCoeffNoise;
+ WebRtc_Word16 used; /* if PLC is used */
+
+
+ WebRtc_Word16 *lastPitchLP; // [FRAMESAMPLES/2]; saved 240;
+
+
+ /* --- LPC side info --- */
+ WebRtc_Word16 lofilt_coefQ15[ ORDERLO ];
+ WebRtc_Word16 hifilt_coefQ15[ ORDERHI ];
+ WebRtc_Word32 gain_lo_hiQ17[2];
+
+ /* --- LTP side info --- */
+ WebRtc_Word16 AvgPitchGain_Q12;
+ WebRtc_Word16 lastPitchGain_Q12;
+ WebRtc_Word16 lastPitchLag_Q7;
+
+ /* --- Add-overlap in recovery packet --- */
+ WebRtc_Word16 overlapLP[ RECOVERY_OVERLAP ]; // [FRAMESAMPLES/2]; saved 160
+
+ WebRtc_Word16 pitchCycles;
+ WebRtc_Word16 A;
+ WebRtc_Word16 B;
+ WebRtc_Word16 pitchIndex;
+ WebRtc_Word16 stretchLag;
+ WebRtc_Word16 *prevPitchLP; // [ FRAMESAMPLES/2 ]; saved 240
+ WebRtc_Word16 seed;
+
+ WebRtc_Word16 std;
+} PLCstr;
+
+
+
+/* Have instance of struct together with other iSAC structs */
+typedef struct {
+
+ WebRtc_Word16 prevFrameSizeMs; /* Previous frame size (in ms) */
+ WebRtc_UWord16 prevRtpNumber; /* Previous RTP timestamp from received packet */
+ /* (in samples relative beginning) */
+ WebRtc_UWord32 prevSendTime; /* Send time for previous packet, from RTP header */
+ WebRtc_UWord32 prevArrivalTime; /* Arrival time for previous packet (in ms using timeGetTime()) */
+ WebRtc_UWord16 prevRtpRate; /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+ WebRtc_UWord32 lastUpdate; /* Time since the last update of the Bottle Neck estimate (in samples) */
+ WebRtc_UWord32 lastReduction; /* Time sinse the last reduction (in samples) */
+ WebRtc_Word32 countUpdates; /* How many times the estimate was update in the beginning */
+
+ /* The estimated bottle neck rate from there to here (in bits/s) */
+ WebRtc_UWord32 recBw;
+ WebRtc_UWord32 recBwInv;
+ WebRtc_UWord32 recBwAvg;
+ WebRtc_UWord32 recBwAvgQ;
+
+ WebRtc_UWord32 minBwInv;
+ WebRtc_UWord32 maxBwInv;
+
+ /* The estimated mean absolute jitter value, as seen on this side (in ms) */
+ WebRtc_Word32 recJitter;
+ WebRtc_Word32 recJitterShortTerm;
+ WebRtc_Word32 recJitterShortTermAbs;
+ WebRtc_Word32 recMaxDelay;
+ WebRtc_Word32 recMaxDelayAvgQ;
+
+
+ WebRtc_Word16 recHeaderRate; /* (assumed) bitrate for headers (bps) */
+
+ WebRtc_UWord32 sendBwAvg; /* The estimated bottle neck rate from here to there (in bits/s) */
+ WebRtc_Word32 sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen on the other siee (in ms) */
+
+
+ WebRtc_Word16 countRecPkts; /* number of packets received since last update */
+ WebRtc_Word16 highSpeedRec; /* flag for marking that a high speed network has been detected downstream */
+
+ /* number of consecutive pkts sent during which the bwe estimate has
+ remained at a value greater than the downstream threshold for determining highspeed network */
+ WebRtc_Word16 countHighSpeedRec;
+
+ /* flag indicating bwe should not adjust down immediately for very late pckts */
+ WebRtc_Word16 inWaitPeriod;
+
+ /* variable holding the time of the start of a window of time when
+ bwe should not adjust down immediately for very late pckts */
+ WebRtc_UWord32 startWaitPeriod;
+
+ /* number of consecutive pkts sent during which the bwe estimate has
+ remained at a value greater than the upstream threshold for determining highspeed network */
+ WebRtc_Word16 countHighSpeedSent;
+
+ /* flag indicated the desired number of packets over threshold rate have been sent and
+ bwe will assume the connection is over broadband network */
+ WebRtc_Word16 highSpeedSend;
+
+
+
+
+} BwEstimatorstr;
+
+
+typedef struct {
+
+ /* boolean, flags if previous packet exceeded B.N. */
+ WebRtc_Word16 PrevExceed;
+ /* ms */
+ WebRtc_Word16 ExceedAgo;
+ /* packets left to send in current burst */
+ WebRtc_Word16 BurstCounter;
+ /* packets */
+ WebRtc_Word16 InitCounter;
+ /* ms remaining in buffer when next packet will be sent */
+ WebRtc_Word16 StillBuffered;
+
+} RateModel;
+
+/* The following strutc is used to store data from encoding, to make it
+ fast and easy to construct a new bitstream with a different Bandwidth
+ estimate. All values (except framelength and minBytes) is double size to
+ handle 60 ms of data.
+*/
+typedef struct {
+
+ /* Used to keep track of if it is first or second part of 60 msec packet */
+ int startIdx;
+
+ /* Frame length in samples */
+ WebRtc_Word16 framelength;
+
+ /* Pitch Gain */
+ WebRtc_Word16 pitchGain_index[2];
+
+ /* Pitch Lag */
+ WebRtc_Word32 meanGain[2];
+ WebRtc_Word16 pitchIndex[PITCH_SUBFRAMES*2];
+
+ /* LPC */
+ WebRtc_Word32 LPCcoeffs_g[12*2]; /* KLT_ORDER_GAIN = 12 */
+ WebRtc_Word16 LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
+ WebRtc_Word16 LPCindex_g[12*2]; /* KLT_ORDER_GAIN = 12 */
+
+ /* Encode Spec */
+ WebRtc_Word16 fre[FRAMESAMPLES];
+ WebRtc_Word16 fim[FRAMESAMPLES];
+ WebRtc_Word16 AvgPitchGain[2];
+
+ /* Used in adaptive mode only */
+ int minBytes;
+
+} ISAC_SaveEncData_t;
+
+typedef struct {
+
+ Bitstr_enc bitstr_obj;
+ MaskFiltstr_enc maskfiltstr_obj;
+ PreFiltBankstr prefiltbankstr_obj;
+ PitchFiltstr pitchfiltstr_obj;
+ PitchAnalysisStruct pitchanalysisstr_obj;
+ RateModel rate_data_obj;
+
+ WebRtc_Word16 buffer_index;
+ WebRtc_Word16 current_framesamples;
+
+ WebRtc_Word16 data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES
+
+ WebRtc_Word16 frame_nb;
+ WebRtc_Word16 BottleNeck;
+ WebRtc_Word16 MaxDelay;
+ WebRtc_Word16 new_framelength;
+ WebRtc_Word16 s2nr;
+ WebRtc_UWord16 MaxBits;
+
+ WebRtc_Word16 bitstr_seed;
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ PostFiltBankstr interpolatorstr_obj;
+#endif
+
+ ISAC_SaveEncData_t *SaveEnc_ptr;
+ WebRtc_Word16 payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
+ WebRtc_Word16 payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
+ WebRtc_Word16 maxPayloadBytes; /* Maximum allowed number of bits for both 30 and 60 msec packet */
+ WebRtc_Word16 maxRateInBytes; /* Maximum allowed rate in bytes per 30 msec packet */
+ WebRtc_Word16 enforceFrameSize; /* If set iSAC will never change packet size */
+
+} ISACFIX_EncInst_t;
+
+
+typedef struct {
+
+ Bitstr_dec bitstr_obj;
+ MaskFiltstr_dec maskfiltstr_obj;
+ PostFiltBankstr postfiltbankstr_obj;
+ PitchFiltstr pitchfiltstr_obj;
+ PLCstr plcstr_obj; /* TS; for packet loss concealment */
+
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ PreFiltBankstr decimatorstr_obj;
+#endif
+
+} ISACFIX_DecInst_t;
+
+
+
+typedef struct {
+
+ ISACFIX_EncInst_t ISACenc_obj;
+ ISACFIX_DecInst_t ISACdec_obj;
+ BwEstimatorstr bwestimator_obj;
+ WebRtc_Word16 CodingMode; /* 0 = adaptive; 1 = instantaneous */
+ WebRtc_Word16 errorcode;
+ WebRtc_Word16 initflag; /* 0 = nothing initiated; 1 = encoder or decoder */
+ /* not initiated; 2 = all initiated */
+} ISACFIX_SubStruct;
+
+
+typedef struct {
+ WebRtc_Word32 lpcGains[12]; /* 6 lower-band & 6 upper-band we may need to double it for 60*/
+ /* */
+ WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
+ WebRtc_UWord32 streamval;
+ WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
+ WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
+ /* 1 - both bytes are empty (we just filled the previous memory */
+ WebRtc_UWord16 beforeLastWord;
+ WebRtc_UWord16 lastWord;
+} transcode_obj;
+
+
+//Bitstr_enc myBitStr;
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c
new file mode 100644
index 0000000..56ef9f2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c
@@ -0,0 +1,296 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * WebRtcIsacfix_kTransform.c
+ *
+ * Transform functions
+ *
+ */
+
+#include "fft.h"
+#include "codec.h"
+#include "settings.h"
+
+
+/* Cosine table 1 in Q14 */
+static const WebRtc_Word16 kCosTab1[FRAMESAMPLES/2] = {
+ 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270,
+ 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880,
+ 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218,
+ 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295,
+ 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128,
+ 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736,
+ 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143,
+ 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377,
+ 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467,
+ 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447,
+ 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351,
+ 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214,
+ 0, -214, -429, -643, -857, -1072, -1285, -1499, -1713, -1926,
+ -2139, -2351, -2563, -2775, -2986, -3196, -3406, -3616, -3825, -4033,
+ -4240, -4447, -4653, -4859, -5063, -5266, -5469, -5671, -5872, -6071,
+ -6270, -6467, -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006,
+ -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456, -9630, -9803,
+ -9974, -10143, -10311, -10477, -10641, -10803, -10963, -11121, -11278, -11433,
+ -11585, -11736, -11885, -12031, -12176, -12318, -12458, -12597, -12733, -12867,
+ -12998, -13128, -13255, -13380, -13502, -13623, -13741, -13856, -13970, -14081,
+ -14189, -14295, -14399, -14500, -14598, -14694, -14788, -14879, -14968, -15053,
+ -15137, -15218, -15296, -15371, -15444, -15515, -15582, -15647, -15709, -15769,
+ -15826, -15880, -15931, -15980, -16026, -16069, -16110, -16147, -16182, -16214,
+ -16244, -16270, -16294, -16315, -16333, -16349, -16362, -16371, -16378, -16383
+};
+
+
+/* Sine table 1 in Q14 */
+static const WebRtc_Word16 kSinTab1[FRAMESAMPLES/2] = {
+ 0, 214, 429, 643, 857, 1072, 1285, 1499, 1713, 1926,
+ 2139, 2351, 2563, 2775, 2986, 3196, 3406, 3616, 3825, 4033,
+ 4240, 4447, 4653, 4859, 5063, 5266, 5469, 5671, 5872, 6071,
+ 6270, 6467, 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006,
+ 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456, 9630, 9803,
+ 9974, 10143, 10311, 10477, 10641, 10803, 10963, 11121, 11278, 11433,
+ 11585, 11736, 11885, 12031, 12176, 12318, 12458, 12597, 12733, 12867,
+ 12998, 13128, 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081,
+ 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879, 14968, 15053,
+ 15137, 15218, 15296, 15371, 15444, 15515, 15582, 15647, 15709, 15769,
+ 15826, 15880, 15931, 15980, 16026, 16069, 16110, 16147, 16182, 16214,
+ 16244, 16270, 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383,
+ 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270,
+ 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880,
+ 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218,
+ 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295,
+ 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128,
+ 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736,
+ 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143,
+ 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377,
+ 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467,
+ 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447,
+ 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351,
+ 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214
+};
+
+
+/* Cosine table 2 in Q14 */
+static const WebRtc_Word16 kCosTab2[FRAMESAMPLES/4] = {
+ 107, -322, 536, -750, 965, -1179, 1392, -1606, 1819, -2032,
+ 2245, -2457, 2669, -2880, 3091, -3301, 3511, -3720, 3929, -4137,
+ 4344, -4550, 4756, -4961, 5165, -5368, 5570, -5771, 5971, -6171,
+ 6369, -6566, 6762, -6957, 7150, -7342, 7534, -7723, 7912, -8099,
+ 8285, -8469, 8652, -8833, 9013, -9191, 9368, -9543, 9717, -9889,
+ 10059, -10227, 10394, -10559, 10722, -10883, 11042, -11200, 11356, -11509,
+ 11661, -11810, 11958, -12104, 12247, -12389, 12528, -12665, 12800, -12933,
+ 13063, -13192, 13318, -13441, 13563, -13682, 13799, -13913, 14025, -14135,
+ 14242, -14347, 14449, -14549, 14647, -14741, 14834, -14924, 15011, -15095,
+ 15178, -15257, 15334, -15408, 15480, -15549, 15615, -15679, 15739, -15798,
+ 15853, -15906, 15956, -16003, 16048, -16090, 16129, -16165, 16199, -16229,
+ 16257, -16283, 16305, -16325, 16342, -16356, 16367, -16375, 16381, -16384
+};
+
+
+/* Sine table 2 in Q14 */
+static const WebRtc_Word16 kSinTab2[FRAMESAMPLES/4] = {
+ 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257,
+ 16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853,
+ 15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178,
+ 15095, -15011, 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242,
+ 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318, 13192, -13063,
+ 12933, -12800, 12665, -12528, 12389, -12247, 12104, -11958, 11810, -11661,
+ 11509, -11356, 11200, -11042, 10883, -10722, 10559, -10394, 10227, -10059,
+ 9889, -9717, 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285,
+ 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762, 6566, -6369,
+ 6171, -5971, 5771, -5570, 5368, -5165, 4961, -4756, 4550, -4344,
+ 4137, -3929, 3720, -3511, 3301, -3091, 2880, -2669, 2457, -2245,
+ 2032, -1819, 1606, -1392, 1179, -965, 750, -536, 322, -107
+};
+
+
+
+void WebRtcIsacfix_Time2Spec(WebRtc_Word16 *inre1Q9,
+ WebRtc_Word16 *inre2Q9,
+ WebRtc_Word16 *outreQ7,
+ WebRtc_Word16 *outimQ7)
+{
+
+ int k;
+ WebRtc_Word32 tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2];
+ WebRtc_Word16 tmp1rQ14, tmp1iQ14;
+ WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
+ WebRtc_Word32 v1Q16, v2Q16;
+ WebRtc_Word16 factQ19, sh;
+
+ /* Multiply with complex exponentials and combine into one complex vector */
+ factQ19 = 16921; // 0.5/sqrt(240) in Q19 is round(.5/sqrt(240)*(2^19)) = 16921
+ for (k = 0; k < FRAMESAMPLES/2; k++) {
+ tmp1rQ14 = kCosTab1[k];
+ tmp1iQ14 = kSinTab1[k];
+ xrQ16 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(tmp1rQ14, inre1Q9[k]) + WEBRTC_SPL_MUL_16_16(tmp1iQ14, inre2Q9[k]), 7);
+ xiQ16 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(tmp1rQ14, inre2Q9[k]) - WEBRTC_SPL_MUL_16_16(tmp1iQ14, inre1Q9[k]), 7);
+ tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xrQ16)+4, 3); // (Q16*Q19>>16)>>3 = Q16
+ tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xiQ16)+4, 3); // (Q16*Q19>>16)>>3 = Q16
+ }
+
+
+ xrQ16 = WebRtcSpl_MaxAbsValueW32(tmpreQ16, FRAMESAMPLES/2);
+ yrQ16 = WebRtcSpl_MaxAbsValueW32(tmpimQ16, FRAMESAMPLES/2);
+ if (yrQ16>xrQ16) {
+ xrQ16 = yrQ16;
+ }
+
+ sh = WebRtcSpl_NormW32(xrQ16);
+ sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh)
+ //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh)
+
+ //"Fastest" vectors
+ if (sh>=0) {
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpreQ16[k], sh); //Q(16+sh)
+ inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpimQ16[k], sh); //Q(16+sh)
+ }
+ } else {
+ WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpreQ16[k]+round, -sh); //Q(16+sh)
+ inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpimQ16[k]+round, -sh); //Q(16+sh)
+ }
+ }
+
+ /* Get DFT */
+ WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); // real call
+
+ //"Fastest" vectors
+ if (sh>=0) {
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre1Q9[k], sh); //Q(16+sh) -> Q16
+ tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre2Q9[k], sh); //Q(16+sh) -> Q16
+ }
+ } else {
+ for (k=0; k<FRAMESAMPLES/2; k++) {
+ tmpreQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre1Q9[k], -sh); //Q(16+sh) -> Q16
+ tmpimQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre2Q9[k], -sh); //Q(16+sh) -> Q16
+ }
+ }
+
+
+ /* Use symmetry to separate into two complex vectors and center frames in time around zero */
+ for (k = 0; k < FRAMESAMPLES/4; k++) {
+ xrQ16 = tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k];
+ yiQ16 = -tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k];
+ xiQ16 = tmpimQ16[k] - tmpimQ16[FRAMESAMPLES/2 - 1 - k];
+ yrQ16 = tmpimQ16[k] + tmpimQ16[FRAMESAMPLES/2 - 1 - k];
+ tmp1rQ14 = kCosTab2[k];
+ tmp1iQ14 = kSinTab2[k];
+ v1Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xiQ16);
+ v2Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xiQ16);
+ outreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v1Q16, 9);
+ outimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v2Q16, 9);
+ v1Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yiQ16);
+ v2Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yiQ16);
+ outreQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v1Q16, 9); //CalcLrIntQ(v1Q16, 9);
+ outimQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v2Q16, 9); //CalcLrIntQ(v2Q16, 9);
+
+ }
+}
+
+
+void WebRtcIsacfix_Spec2Time(WebRtc_Word16 *inreQ7, WebRtc_Word16 *inimQ7, WebRtc_Word32 *outre1Q16, WebRtc_Word32 *outre2Q16)
+{
+
+ int k;
+ WebRtc_Word16 tmp1rQ14, tmp1iQ14;
+ WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
+ WebRtc_Word32 tmpInRe, tmpInIm, tmpInRe2, tmpInIm2;
+ WebRtc_Word16 factQ11;
+ WebRtc_Word16 sh;
+
+ for (k = 0; k < FRAMESAMPLES/4; k++) {
+ /* Move zero in time to beginning of frames */
+ tmp1rQ14 = kCosTab2[k];
+ tmp1iQ14 = kSinTab2[k];
+
+ tmpInRe = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[k], 9); // Q7 -> Q16
+ tmpInIm = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[k], 9); // Q7 -> Q16
+ tmpInRe2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[FRAMESAMPLES/2 - 1 - k], 9); // Q7 -> Q16
+ tmpInIm2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[FRAMESAMPLES/2 - 1 - k], 9); // Q7 -> Q16
+
+ xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm);
+ xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe);
+ yrQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm2) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe2);
+ yiQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe2) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm2);
+
+ /* Combine into one vector, z = x + j * y */
+ outre1Q16[k] = xrQ16 - yiQ16;
+ outre1Q16[FRAMESAMPLES/2 - 1 - k] = xrQ16 + yiQ16;
+ outre2Q16[k] = xiQ16 + yrQ16;
+ outre2Q16[FRAMESAMPLES/2 - 1 - k] = -xiQ16 + yrQ16;
+ }
+
+ /* Get IDFT */
+ tmpInRe = WebRtcSpl_MaxAbsValueW32(outre1Q16, 240);
+ tmpInIm = WebRtcSpl_MaxAbsValueW32(outre2Q16, 240);
+ if (tmpInIm>tmpInRe) {
+ tmpInRe = tmpInIm;
+ }
+
+ sh = WebRtcSpl_NormW32(tmpInRe);
+ sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh)
+ //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh)
+
+ //"Fastest" vectors
+ if (sh>=0) {
+ for (k=0; k<240; k++) {
+ inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre1Q16[k], sh); //Q(16+sh)
+ inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre2Q16[k], sh); //Q(16+sh)
+ }
+ } else {
+ WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
+ for (k=0; k<240; k++) {
+ inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre1Q16[k]+round, -sh); //Q(16+sh)
+ inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre2Q16[k]+round, -sh); //Q(16+sh)
+ }
+ }
+
+ WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); // real call
+
+ //"Fastest" vectors
+ if (sh>=0) {
+ for (k=0; k<240; k++) {
+ outre1Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inreQ7[k], sh); //Q(16+sh) -> Q16
+ outre2Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inimQ7[k], sh); //Q(16+sh) -> Q16
+ }
+ } else {
+ for (k=0; k<240; k++) {
+ outre1Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inreQ7[k], -sh); //Q(16+sh) -> Q16
+ outre2Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inimQ7[k], -sh); //Q(16+sh) -> Q16
+ }
+ }
+
+ /* Divide through by the normalizing constant: */
+ /* scale all values with 1/240, i.e. with 273 in Q16 */
+ /* 273/65536 ~= 0.0041656 */
+ /* 1/240 ~= 0.0041666 */
+ for (k=0; k<240; k++) {
+ outre1Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre1Q16[k]);
+ outre2Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre2Q16[k]);
+ }
+
+ /* Demodulate and separate */
+ factQ11 = 31727; // sqrt(240) in Q11 is round(15.49193338482967*2048) = 31727
+ for (k = 0; k < FRAMESAMPLES/2; k++) {
+ tmp1rQ14 = kCosTab1[k];
+ tmp1iQ14 = kSinTab1[k];
+ xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre1Q16[k]) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre2Q16[k]);
+ xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre2Q16[k]) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre1Q16[k]);
+ xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xrQ16);
+ xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xiQ16);
+ outre2Q16[k] = xiQ16;
+ outre1Q16[k] = xrQ16;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc
new file mode 100644
index 0000000..753acd7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc
@@ -0,0 +1,173 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <ctype.h>
+
+//#include "isac_codec.h"
+//#include "isac_structs.h"
+#include "isacfix.h"
+
+
+#define NUM_CODECS 1
+
+int main(int argc, char* argv[])
+{
+ FILE *inFileList;
+ FILE *audioFile;
+ FILE *outFile;
+ char audioFileName[501];
+ short audioBuff[960];
+ short encoded[600];
+ short startAudio;
+ short encodedLen;
+ ISACFIX_MainStruct *isac_struct;
+ unsigned long int hist[601];
+
+ // reset the histogram
+ for(short n=0; n < 601; n++)
+ {
+ hist[n] = 0;
+ }
+
+
+ inFileList = fopen(argv[1], "r");
+ if(inFileList == NULL)
+ {
+ printf("Could not open the input file.\n");
+ getchar();
+ exit(-1);
+ }
+ outFile = fopen(argv[2], "w");
+ if(outFile == NULL)
+ {
+ printf("Could not open the histogram file.\n");
+ getchar();
+ exit(-1);
+ }
+
+ short frameSizeMsec = 30;
+ if(argc > 3)
+ {
+ frameSizeMsec = atoi(argv[3]);
+ }
+
+ short audioOffset = 0;
+ if(argc > 4)
+ {
+ audioOffset = atoi(argv[4]);
+ }
+ int ok;
+ ok = WebRtcIsacfix_Create(&isac_struct);
+ // instantaneous mode
+ ok |= WebRtcIsacfix_EncoderInit(isac_struct, 1);
+ // is not used but initialize
+ ok |= WebRtcIsacfix_DecoderInit(isac_struct);
+ ok |= WebRtcIsacfix_Control(isac_struct, 32000, frameSizeMsec);
+
+ if(ok != 0)
+ {
+ printf("\nProblem in seting up iSAC\n");
+ exit(-1);
+ }
+
+ while( fgets(audioFileName, 500, inFileList) != NULL )
+ {
+ // remove trailing white-spaces and any Cntrl character
+ if(strlen(audioFileName) == 0)
+ {
+ continue;
+ }
+ short n = strlen(audioFileName) - 1;
+ while(isspace(audioFileName[n]) || iscntrl(audioFileName[n]))
+ {
+ audioFileName[n] = '\0';
+ n--;
+ if(n < 0)
+ {
+ break;
+ }
+ }
+
+ // remove leading spaces
+ if(strlen(audioFileName) == 0)
+ {
+ continue;
+ }
+ n = 0;
+ while((isspace(audioFileName[n]) || iscntrl(audioFileName[n])) &&
+ (audioFileName[n] != '\0'))
+ {
+ n++;
+ }
+ memmove(audioFileName, &audioFileName[n], 500 - n);
+ if(strlen(audioFileName) == 0)
+ {
+ continue;
+ }
+ audioFile = fopen(audioFileName, "rb");
+ if(audioFile == NULL)
+ {
+ printf("\nCannot open %s!!!!!\n", audioFileName);
+ exit(0);
+ }
+
+ if(audioOffset > 0)
+ {
+ fseek(audioFile, (audioOffset<<1), SEEK_SET);
+ }
+
+ while(fread(audioBuff, sizeof(short), (480*frameSizeMsec/30), audioFile) >= (480*frameSizeMsec/30))
+ {
+ startAudio = 0;
+ do
+ {
+ encodedLen = WebRtcIsacfix_Encode(isac_struct,
+ &audioBuff[startAudio], encoded);
+ startAudio += 160;
+ } while(encodedLen == 0);
+
+ if(encodedLen < 0)
+ {
+ printf("\nEncoding Error!!!\n");
+ exit(0);
+ }
+ hist[encodedLen]++;
+ }
+ fclose(audioFile);
+ }
+ fclose(inFileList);
+ unsigned long totalFrames = 0;
+ for(short n=0; n < 601; n++)
+ {
+ totalFrames += hist[n];
+ fprintf(outFile, "%10lu\n", hist[n]);
+ }
+ fclose(outFile);
+
+ short topTenCntr = 0;
+ printf("\nTotal number of Frames %lu\n\n", totalFrames);
+ printf("Payload Len # occurences\n");
+ printf("=========== ============\n");
+
+ for(short n = 600; (n >= 0) && (topTenCntr < 10); n--)
+ {
+ if(hist[n] > 0)
+ {
+ topTenCntr++;
+ printf(" %3d %3d\n", n, hist[n]);
+ }
+ }
+ WebRtcIsacfix_Free(isac_struct);
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc
new file mode 100644
index 0000000..2791db4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc
@@ -0,0 +1,260 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+ Stand Alone test application for ISACFIX and ISAC LC
+
+******************************************************************/
+
+#include <string.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include "typedefs.h"
+
+#include "isacfix.h"
+ISACFIX_MainStruct *ISACfix_inst;
+
+#define FS 16000
+
+
+typedef struct {
+ WebRtc_UWord32 arrival_time; /* samples */
+ WebRtc_UWord32 sample_count; /* samples */
+ WebRtc_UWord16 rtp_number;
+} BottleNeckModel;
+
+void get_arrival_time(int current_framesamples, /* samples */
+ int packet_size, /* bytes */
+ int bottleneck, /* excluding headers; bits/s */
+ BottleNeckModel *BN_data)
+{
+ const int HeaderSize = 35;
+ int HeaderRate;
+
+ HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
+
+ /* everything in samples */
+ BN_data->sample_count = BN_data->sample_count + current_framesamples;
+
+ BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+
+ if (BN_data->arrival_time < BN_data->sample_count)
+ BN_data->arrival_time = BN_data->sample_count;
+
+ BN_data->rtp_number++;
+}
+
+/*
+#ifdef __cplusplus
+extern "C" {
+#endif
+*/
+int main(int argc, char* argv[]){
+
+ /* Parameters */
+ FILE *pInFile, *pOutFile, *pChcFile;
+ WebRtc_Word8 inFile[40];
+ WebRtc_Word8 outFile[40];
+ WebRtc_Word8 chcFile[40];
+ WebRtc_Word8 codec[10];
+ WebRtc_Word16 bitrt, spType, size;
+ WebRtc_UWord16 frameLen;
+ WebRtc_Word16 sigOut[1000], sigIn[1000];
+ WebRtc_UWord16 bitStream[500]; /* double to 32 kbps for 60 ms */
+
+ WebRtc_Word16 chc, ok;
+ int noOfCalls, cdlen;
+ WebRtc_Word16 noOfLostFrames;
+ int err, errtype;
+
+ BottleNeckModel BN_data;
+
+ int totalbits =0;
+ int totalsmpls =0;
+
+ /*End Parameters*/
+
+
+ if ((argc==6)||(argc==7) ){
+
+ strcpy(codec,argv[5]);
+
+ if(argc==7){
+ if (!_stricmp("isac",codec)){
+ bitrt = atoi(argv[6]);
+ if ( (bitrt<10000)&&(bitrt>32000)){
+ printf("Error: Supported bit rate in the range 10000-32000 bps!\n");
+ exit(-1);
+ }
+
+ }else{
+ printf("Error: Codec not recognized. Check spelling!\n");
+ exit(-1);
+ }
+
+ } else {
+ printf("Error: Codec not recognized. Check spelling!\n");
+ exit(-1);
+ }
+ } else {
+ printf("Error: Wrong number of input parameter!\n\n");
+ exit(-1);
+ }
+
+ frameLen = atoi(argv[4]);
+ strcpy(chcFile,argv[3]);
+ strcpy(outFile,argv[2]);
+ strcpy(inFile,argv[1]);
+
+ /* Open file streams */
+ if( (pInFile = fopen(inFile,"rb")) == NULL ) {
+ printf( "Error: Did not find input file!\n" );
+ exit(-1);
+ }
+ strcat(outFile,"_");
+ strcat(outFile, argv[4]);
+ strcat(outFile,"_");
+ strcat(outFile, codec);
+
+ if (argc==7){
+ strcat(outFile,"_");
+ strcat(outFile, argv[6]);
+ }
+ if (_stricmp("none", chcFile)){
+ strcat(outFile,"_");
+ strcat(outFile, "plc");
+ }
+
+ strcat(outFile, ".otp");
+
+ if (_stricmp("none", chcFile)){
+ if( (pChcFile = fopen(chcFile,"rb")) == NULL ) {
+ printf( "Error: Did not find channel file!\n" );
+ exit(-1);
+ }
+ }
+ /******************************************************************/
+ if (!_stricmp("isac", codec)){ /* ISAC */
+ if ((frameLen!=480)&&(frameLen!=960)) {
+ printf("Error: ISAC only supports 480 and 960 samples per frame (not %d)\n", frameLen);
+ exit(-1);
+ }
+ if( (pOutFile = fopen(outFile,"wb")) == NULL ) {
+ printf( "Could not open output file!\n" );
+ exit(-1);
+ }
+ ok=WebRtcIsacfix_Create(&ISACfix_inst);
+ if (ok!=0) {
+ printf("Couldn't allocate memory for iSAC fix instance\n");
+ exit(-1);
+ }
+
+ BN_data.arrival_time = 0;
+ BN_data.sample_count = 0;
+ BN_data.rtp_number = 0;
+
+ WebRtcIsacfix_EncoderInit(ISACfix_inst,1);
+ WebRtcIsacfix_DecoderInit(ISACfix_inst);
+ err = WebRtcIsacfix_Control(ISACfix_inst, bitrt, (frameLen>>4));
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ /* loop over frame */
+ while (fread(sigIn,sizeof(WebRtc_Word16),frameLen,pInFile) == frameLen) {
+
+ noOfCalls=0;
+ cdlen=0;
+ while (cdlen<=0) {
+ cdlen=WebRtcIsacfix_Encode(ISACfix_inst,&sigIn[noOfCalls*160],(WebRtc_Word16*)bitStream);
+ if(cdlen==-1){
+ errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
+ printf("\n\nError in encoder: %d.\n\n", errtype);
+ exit(-1);
+ }
+ noOfCalls++;
+ }
+
+
+ if(_stricmp("none", chcFile)){
+ if (fread(&chc,sizeof(WebRtc_Word16),1,pChcFile)!=1) /* packet may be lost */
+ break;
+ } else {
+ chc = 1; /* packets never lost */
+ }
+
+ /* simulate packet handling through NetEq and the modem */
+ get_arrival_time(frameLen, cdlen, bitrt, &BN_data);
+
+ if (chc){ /* decode */
+
+ err = WebRtcIsacfix_UpdateBwEstimate1(ISACfix_inst,
+ bitStream,
+ cdlen,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ size = WebRtcIsacfix_Decode(ISACfix_inst, bitStream, cdlen, sigOut, &spType);
+ if(size<=0){
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ exit(-1);
+ }
+ } else { /* PLC */
+ if (frameLen == 480){
+ noOfLostFrames = 1;
+ } else{
+ noOfLostFrames = 2;
+ }
+ size = WebRtcIsacfix_DecodePlc(ISACfix_inst, sigOut, noOfLostFrames );
+ if(size<=0){
+ errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ exit(-1);
+ }
+ }
+
+ /* Write decoded speech to file */
+ fwrite(sigOut,sizeof(short),size,pOutFile);
+
+ totalbits += 8 * cdlen;
+ totalsmpls += size;
+
+ }
+ /******************************************************************/
+ }
+
+// printf("\n\ntotal bits = %d bits", totalbits);
+ printf("\nmeasured average bitrate = %0.3f kbits/s", (double)totalbits * 16 / totalsmpls);
+ printf("\n");
+
+
+ fclose(pInFile);
+ fclose(pOutFile);
+ if (_stricmp("none", chcFile)){
+ fclose(pChcFile);
+ }
+
+ if (!_stricmp("isac", codec)){
+ WebRtcIsacfix_Free(ISACfix_inst);
+ }
+
+ return 0;
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt
new file mode 100644
index 0000000..05f7410
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt
@@ -0,0 +1,3 @@
+bottlenecks.txt
+lowrates.txt
+tworates.txt
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt
new file mode 100644
index 0000000..f26b7af
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt
@@ -0,0 +1,31 @@
+DTMF_16kHz_long.pcm
+DTMF_16kHz_short.pcm
+F00.INP
+F01.INP
+F02.INP
+F03.INP
+F04.INP
+F05.INP
+F06.INP
+longtest.pcm
+ltest_speech_clean.pcm
+ltest_music.pcm
+ltest_speech_noisy.pcm
+misc2.pcm
+purenb.pcm
+sawsweep_380_60.pcm
+sinesweep.pcm
+sinesweep_half.pcm
+speechmusic.pcm
+speechmusic_nb.pcm
+speechoffice0dB.pcm
+speech_and_misc_NB.pcm
+speech_and_misc_WB.pcm
+testM4.pcm
+testM4D_rev.pcm
+testM4D.pcm
+testfile.pcm
+tone_cisco.pcm
+tone_cisco_long.pcm
+wb_contspeech.pcm
+wb_speech_office25db.pcm
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt
new file mode 100644
index 0000000..08bbde3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt
@@ -0,0 +1,6 @@
+DTMF_16kHz_short.pcm
+ltest_speech_noisy.pcm
+misc2.pcm
+sinesweep.pcm
+speechmusic.pcm
+tone_cisco.pcm
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls
new file mode 100644
index 0000000..f0889ef
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls
Binary files differ
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt
new file mode 100644
index 0000000..96b87c0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt
@@ -0,0 +1,481 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+diff ../dataqa350/i30_1DTMF_16kHz_long.pcm ../dataqa351/i30_1DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_1DTMF_16kHz_long.pcm ../dataqa351/i60_1DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_2DTMF_16kHz_long.pcm ../dataqa351/i30_2DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_2DTMF_16kHz_long.pcm ../dataqa351/i60_2DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_3DTMF_16kHz_long.pcm ../dataqa351/i30_3DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_3DTMF_16kHz_long.pcm ../dataqa351/i60_3DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_4DTMF_16kHz_long.pcm ../dataqa351/i30_4DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_4DTMF_16kHz_long.pcm ../dataqa351/i60_4DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_5DTMF_16kHz_long.pcm ../dataqa351/i30_5DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_5DTMF_16kHz_long.pcm ../dataqa351/i60_5DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_6DTMF_16kHz_long.pcm ../dataqa351/i30_6DTMF_16kHz_long.pcm
+diff ../dataqa350/i60_6DTMF_16kHz_long.pcm ../dataqa351/i60_6DTMF_16kHz_long.pcm
+diff ../dataqa350/a1DTMF_16kHz_long.pcm ../dataqa351/a1DTMF_16kHz_long.pcm
+diff ../dataqa350/a2DTMF_16kHz_long.pcm ../dataqa351/a2DTMF_16kHz_long.pcm
+diff ../dataqa350/a3DTMF_16kHz_long.pcm ../dataqa351/a3DTMF_16kHz_long.pcm
+diff ../dataqa350/i30_7DTMF_16kHz_short.pcm ../dataqa351/i30_7DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_7DTMF_16kHz_short.pcm ../dataqa351/i60_7DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_8DTMF_16kHz_short.pcm ../dataqa351/i30_8DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_8DTMF_16kHz_short.pcm ../dataqa351/i60_8DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_9DTMF_16kHz_short.pcm ../dataqa351/i30_9DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_9DTMF_16kHz_short.pcm ../dataqa351/i60_9DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_10DTMF_16kHz_short.pcm ../dataqa351/i30_10DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_10DTMF_16kHz_short.pcm ../dataqa351/i60_10DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_11DTMF_16kHz_short.pcm ../dataqa351/i30_11DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_11DTMF_16kHz_short.pcm ../dataqa351/i60_11DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_12DTMF_16kHz_short.pcm ../dataqa351/i30_12DTMF_16kHz_short.pcm
+diff ../dataqa350/i60_12DTMF_16kHz_short.pcm ../dataqa351/i60_12DTMF_16kHz_short.pcm
+diff ../dataqa350/a4DTMF_16kHz_short.pcm ../dataqa351/a4DTMF_16kHz_short.pcm
+diff ../dataqa350/a5DTMF_16kHz_short.pcm ../dataqa351/a5DTMF_16kHz_short.pcm
+diff ../dataqa350/a6DTMF_16kHz_short.pcm ../dataqa351/a6DTMF_16kHz_short.pcm
+diff ../dataqa350/i30_13F00.INP ../dataqa350/i30_13F00.INP
+diff ../dataqa350/i60_13F00.INP ../dataqa350/i60_13F00.INP
+diff ../dataqa350/i30_14F00.INP ../dataqa350/i30_14F00.INP
+diff ../dataqa350/i60_14F00.INP ../dataqa350/i60_14F00.INP
+diff ../dataqa350/i30_15F00.INP ../dataqa350/i30_15F00.INP
+diff ../dataqa350/i60_15F00.INP ../dataqa350/i60_15F00.INP
+diff ../dataqa350/i30_16F00.INP ../dataqa350/i30_16F00.INP
+diff ../dataqa350/i60_16F00.INP ../dataqa350/i60_16F00.INP
+diff ../dataqa350/i30_17F00.INP ../dataqa350/i30_17F00.INP
+diff ../dataqa350/i60_17F00.INP ../dataqa350/i60_17F00.INP
+diff ../dataqa350/i30_18F00.INP ../dataqa350/i30_18F00.INP
+diff ../dataqa350/i60_18F00.INP ../dataqa350/i60_18F00.INP
+diff ../dataqa350/a7F00.INP ../dataqa350/a7F00.INP
+diff ../dataqa350/a8F00.INP ../dataqa350/a8F00.INP
+diff ../dataqa350/a9F00.INP ../dataqa350/a9F00.INP
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+diff ../dataqa350/i60_19F01.INP ../dataqa350/i60_19F01.INP
+diff ../dataqa350/i30_20F01.INP ../dataqa350/i30_20F01.INP
+diff ../dataqa350/i60_20F01.INP ../dataqa350/i60_20F01.INP
+diff ../dataqa350/i30_21F01.INP ../dataqa350/i30_21F01.INP
+diff ../dataqa350/i60_21F01.INP ../dataqa350/i60_21F01.INP
+diff ../dataqa350/i30_22F01.INP ../dataqa350/i30_22F01.INP
+diff ../dataqa350/i60_22F01.INP ../dataqa350/i60_22F01.INP
+diff ../dataqa350/i30_23F01.INP ../dataqa350/i30_23F01.INP
+diff ../dataqa350/i60_23F01.INP ../dataqa350/i60_23F01.INP
+diff ../dataqa350/i30_24F01.INP ../dataqa350/i30_24F01.INP
+diff ../dataqa350/i60_24F01.INP ../dataqa350/i60_24F01.INP
+diff ../dataqa350/a10F01.INP ../dataqa350/a10F01.INP
+diff ../dataqa350/a11F01.INP ../dataqa350/a11F01.INP
+diff ../dataqa350/a12F01.INP ../dataqa350/a12F01.INP
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+diff ../dataqa350/i60_25F02.INP ../dataqa350/i60_25F02.INP
+diff ../dataqa350/i30_26F02.INP ../dataqa350/i30_26F02.INP
+diff ../dataqa350/i60_26F02.INP ../dataqa350/i60_26F02.INP
+diff ../dataqa350/i30_27F02.INP ../dataqa350/i30_27F02.INP
+diff ../dataqa350/i60_27F02.INP ../dataqa350/i60_27F02.INP
+diff ../dataqa350/i30_28F02.INP ../dataqa350/i30_28F02.INP
+diff ../dataqa350/i60_28F02.INP ../dataqa350/i60_28F02.INP
+diff ../dataqa350/i30_29F02.INP ../dataqa350/i30_29F02.INP
+diff ../dataqa350/i60_29F02.INP ../dataqa350/i60_29F02.INP
+diff ../dataqa350/i30_30F02.INP ../dataqa350/i30_30F02.INP
+diff ../dataqa350/i60_30F02.INP ../dataqa350/i60_30F02.INP
+diff ../dataqa350/a13F02.INP ../dataqa350/a13F02.INP
+diff ../dataqa350/a14F02.INP ../dataqa350/a14F02.INP
+diff ../dataqa350/a15F02.INP ../dataqa350/a15F02.INP
+diff ../dataqa350/i30_31F03.INP ../dataqa350/i30_31F03.INP
+diff ../dataqa350/i60_31F03.INP ../dataqa350/i60_31F03.INP
+diff ../dataqa350/i30_32F03.INP ../dataqa350/i30_32F03.INP
+diff ../dataqa350/i60_32F03.INP ../dataqa350/i60_32F03.INP
+diff ../dataqa350/i30_33F03.INP ../dataqa350/i30_33F03.INP
+diff ../dataqa350/i60_33F03.INP ../dataqa350/i60_33F03.INP
+diff ../dataqa350/i30_34F03.INP ../dataqa350/i30_34F03.INP
+diff ../dataqa350/i60_34F03.INP ../dataqa350/i60_34F03.INP
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+diff ../dataqa350/i60_35F03.INP ../dataqa350/i60_35F03.INP
+diff ../dataqa350/i30_36F03.INP ../dataqa350/i30_36F03.INP
+diff ../dataqa350/i60_36F03.INP ../dataqa350/i60_36F03.INP
+diff ../dataqa350/a16F03.INP ../dataqa350/a16F03.INP
+diff ../dataqa350/a17F03.INP ../dataqa350/a17F03.INP
+diff ../dataqa350/a18F03.INP ../dataqa350/a18F03.INP
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+diff ../dataqa350/i60_37F04.INP ../dataqa350/i60_37F04.INP
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+diff ../dataqa350/i60_38F04.INP ../dataqa350/i60_38F04.INP
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+diff ../dataqa350/i60_39F04.INP ../dataqa350/i60_39F04.INP
+diff ../dataqa350/i30_40F04.INP ../dataqa350/i30_40F04.INP
+diff ../dataqa350/i60_40F04.INP ../dataqa350/i60_40F04.INP
+diff ../dataqa350/i30_41F04.INP ../dataqa350/i30_41F04.INP
+diff ../dataqa350/i60_41F04.INP ../dataqa350/i60_41F04.INP
+diff ../dataqa350/i30_42F04.INP ../dataqa350/i30_42F04.INP
+diff ../dataqa350/i60_42F04.INP ../dataqa350/i60_42F04.INP
+diff ../dataqa350/a19F04.INP ../dataqa350/a19F04.INP
+diff ../dataqa350/a20F04.INP ../dataqa350/a20F04.INP
+diff ../dataqa350/a21F04.INP ../dataqa350/a21F04.INP
+diff ../dataqa350/i30_43F05.INP ../dataqa350/i30_43F05.INP
+diff ../dataqa350/i60_43F05.INP ../dataqa350/i60_43F05.INP
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+diff ../dataqa350/i60_44F05.INP ../dataqa350/i60_44F05.INP
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+diff ../dataqa350/i60_45F05.INP ../dataqa350/i60_45F05.INP
+diff ../dataqa350/i30_46F05.INP ../dataqa350/i30_46F05.INP
+diff ../dataqa350/i60_46F05.INP ../dataqa350/i60_46F05.INP
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+diff ../dataqa350/i60_47F05.INP ../dataqa350/i60_47F05.INP
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+diff ../dataqa350/a22F05.INP ../dataqa350/a22F05.INP
+diff ../dataqa350/a23F05.INP ../dataqa350/a23F05.INP
+diff ../dataqa350/a24F05.INP ../dataqa350/a24F05.INP
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+diff ../dataqa350/i60_49F06.INP ../dataqa350/i60_49F06.INP
+diff ../dataqa350/i30_50F06.INP ../dataqa350/i30_50F06.INP
+diff ../dataqa350/i60_50F06.INP ../dataqa350/i60_50F06.INP
+diff ../dataqa350/i30_51F06.INP ../dataqa350/i30_51F06.INP
+diff ../dataqa350/i60_51F06.INP ../dataqa350/i60_51F06.INP
+diff ../dataqa350/i30_52F06.INP ../dataqa350/i30_52F06.INP
+diff ../dataqa350/i60_52F06.INP ../dataqa350/i60_52F06.INP
+diff ../dataqa350/i30_53F06.INP ../dataqa350/i30_53F06.INP
+diff ../dataqa350/i60_53F06.INP ../dataqa350/i60_53F06.INP
+diff ../dataqa350/i30_54F06.INP ../dataqa350/i30_54F06.INP
+diff ../dataqa350/i60_54F06.INP ../dataqa350/i60_54F06.INP
+diff ../dataqa350/a25F06.INP ../dataqa350/a25F06.INP
+diff ../dataqa350/a26F06.INP ../dataqa350/a26F06.INP
+diff ../dataqa350/a27F06.INP ../dataqa350/a27F06.INP
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+diff ../dataqa350/i60_55longtest.pcm ../dataqa351/i60_55longtest.pcm
+diff ../dataqa350/i30_56longtest.pcm ../dataqa351/i30_56longtest.pcm
+diff ../dataqa350/i60_56longtest.pcm ../dataqa351/i60_56longtest.pcm
+diff ../dataqa350/i30_57longtest.pcm ../dataqa351/i30_57longtest.pcm
+diff ../dataqa350/i60_57longtest.pcm ../dataqa351/i60_57longtest.pcm
+diff ../dataqa350/i30_58longtest.pcm ../dataqa351/i30_58longtest.pcm
+diff ../dataqa350/i60_58longtest.pcm ../dataqa351/i60_58longtest.pcm
+diff ../dataqa350/i30_59longtest.pcm ../dataqa351/i30_59longtest.pcm
+diff ../dataqa350/i60_59longtest.pcm ../dataqa351/i60_59longtest.pcm
+diff ../dataqa350/i30_60longtest.pcm ../dataqa351/i30_60longtest.pcm
+diff ../dataqa350/i60_60longtest.pcm ../dataqa351/i60_60longtest.pcm
+diff ../dataqa350/a28longtest.pcm ../dataqa351/a28longtest.pcm
+diff ../dataqa350/a29longtest.pcm ../dataqa351/a29longtest.pcm
+diff ../dataqa350/a30longtest.pcm ../dataqa351/a30longtest.pcm
+diff ../dataqa350/i30_61ltest_speech_clean.pcm ../dataqa351/i30_61ltest_speech_clean.pcm
+diff ../dataqa350/i60_61ltest_speech_clean.pcm ../dataqa351/i60_61ltest_speech_clean.pcm
+diff ../dataqa350/i30_62ltest_speech_clean.pcm ../dataqa351/i30_62ltest_speech_clean.pcm
+diff ../dataqa350/i60_62ltest_speech_clean.pcm ../dataqa351/i60_62ltest_speech_clean.pcm
+diff ../dataqa350/i30_63ltest_speech_clean.pcm ../dataqa351/i30_63ltest_speech_clean.pcm
+diff ../dataqa350/i60_63ltest_speech_clean.pcm ../dataqa351/i60_63ltest_speech_clean.pcm
+diff ../dataqa350/i30_64ltest_speech_clean.pcm ../dataqa351/i30_64ltest_speech_clean.pcm
+diff ../dataqa350/i60_64ltest_speech_clean.pcm ../dataqa351/i60_64ltest_speech_clean.pcm
+diff ../dataqa350/i30_65ltest_speech_clean.pcm ../dataqa351/i30_65ltest_speech_clean.pcm
+diff ../dataqa350/i60_65ltest_speech_clean.pcm ../dataqa351/i60_65ltest_speech_clean.pcm
+diff ../dataqa350/i30_66ltest_speech_clean.pcm ../dataqa351/i30_66ltest_speech_clean.pcm
+diff ../dataqa350/i60_66ltest_speech_clean.pcm ../dataqa351/i60_66ltest_speech_clean.pcm
+diff ../dataqa350/a31ltest_speech_clean.pcm ../dataqa351/a31ltest_speech_clean.pcm
+diff ../dataqa350/a32ltest_speech_clean.pcm ../dataqa351/a32ltest_speech_clean.pcm
+diff ../dataqa350/a33ltest_speech_clean.pcm ../dataqa351/a33ltest_speech_clean.pcm
+diff ../dataqa350/i30_67ltest_music.pcm ../dataqa351/i30_67ltest_music.pcm
+diff ../dataqa350/i60_67ltest_music.pcm ../dataqa351/i60_67ltest_music.pcm
+diff ../dataqa350/i30_68ltest_music.pcm ../dataqa351/i30_68ltest_music.pcm
+diff ../dataqa350/i60_68ltest_music.pcm ../dataqa351/i60_68ltest_music.pcm
+diff ../dataqa350/i30_69ltest_music.pcm ../dataqa351/i30_69ltest_music.pcm
+diff ../dataqa350/i60_69ltest_music.pcm ../dataqa351/i60_69ltest_music.pcm
+diff ../dataqa350/i30_70ltest_music.pcm ../dataqa351/i30_70ltest_music.pcm
+diff ../dataqa350/i60_70ltest_music.pcm ../dataqa351/i60_70ltest_music.pcm
+diff ../dataqa350/i30_71ltest_music.pcm ../dataqa351/i30_71ltest_music.pcm
+diff ../dataqa350/i60_71ltest_music.pcm ../dataqa351/i60_71ltest_music.pcm
+diff ../dataqa350/i30_72ltest_music.pcm ../dataqa351/i30_72ltest_music.pcm
+diff ../dataqa350/i60_72ltest_music.pcm ../dataqa351/i60_72ltest_music.pcm
+diff ../dataqa350/a34ltest_music.pcm ../dataqa351/a34ltest_music.pcm
+diff ../dataqa350/a35ltest_music.pcm ../dataqa351/a35ltest_music.pcm
+diff ../dataqa350/a36ltest_music.pcm ../dataqa351/a36ltest_music.pcm
+diff ../dataqa350/i30_73ltest_speech_noisy.pcm ../dataqa351/i30_73ltest_speech_noisy.pcm
+diff ../dataqa350/i60_73ltest_speech_noisy.pcm ../dataqa351/i60_73ltest_speech_noisy.pcm
+diff ../dataqa350/i30_74ltest_speech_noisy.pcm ../dataqa351/i30_74ltest_speech_noisy.pcm
+diff ../dataqa350/i60_74ltest_speech_noisy.pcm ../dataqa351/i60_74ltest_speech_noisy.pcm
+diff ../dataqa350/i30_75ltest_speech_noisy.pcm ../dataqa351/i30_75ltest_speech_noisy.pcm
+diff ../dataqa350/i60_75ltest_speech_noisy.pcm ../dataqa351/i60_75ltest_speech_noisy.pcm
+diff ../dataqa350/i30_76ltest_speech_noisy.pcm ../dataqa351/i30_76ltest_speech_noisy.pcm
+diff ../dataqa350/i60_76ltest_speech_noisy.pcm ../dataqa351/i60_76ltest_speech_noisy.pcm
+diff ../dataqa350/i30_77ltest_speech_noisy.pcm ../dataqa351/i30_77ltest_speech_noisy.pcm
+diff ../dataqa350/i60_77ltest_speech_noisy.pcm ../dataqa351/i60_77ltest_speech_noisy.pcm
+diff ../dataqa350/i30_78ltest_speech_noisy.pcm ../dataqa351/i30_78ltest_speech_noisy.pcm
+diff ../dataqa350/i60_78ltest_speech_noisy.pcm ../dataqa351/i60_78ltest_speech_noisy.pcm
+diff ../dataqa350/a37ltest_speech_noisy.pcm ../dataqa351/a37ltest_speech_noisy.pcm
+diff ../dataqa350/a38ltest_speech_noisy.pcm ../dataqa351/a38ltest_speech_noisy.pcm
+diff ../dataqa350/a39ltest_speech_noisy.pcm ../dataqa351/a39ltest_speech_noisy.pcm
+diff ../dataqa350/i30_79misc2.pcm ../dataqa351/i30_79misc2.pcm
+diff ../dataqa350/i60_79misc2.pcm ../dataqa351/i60_79misc2.pcm
+diff ../dataqa350/i30_80misc2.pcm ../dataqa351/i30_80misc2.pcm
+diff ../dataqa350/i60_80misc2.pcm ../dataqa351/i60_80misc2.pcm
+diff ../dataqa350/i30_81misc2.pcm ../dataqa351/i30_81misc2.pcm
+diff ../dataqa350/i60_81misc2.pcm ../dataqa351/i60_81misc2.pcm
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+diff ../dataqa350/i60_82misc2.pcm ../dataqa351/i60_82misc2.pcm
+diff ../dataqa350/i30_83misc2.pcm ../dataqa351/i30_83misc2.pcm
+diff ../dataqa350/i60_83misc2.pcm ../dataqa351/i60_83misc2.pcm
+diff ../dataqa350/i30_84misc2.pcm ../dataqa351/i30_84misc2.pcm
+diff ../dataqa350/i60_84misc2.pcm ../dataqa351/i60_84misc2.pcm
+diff ../dataqa350/a40misc2.pcm ../dataqa351/a40misc2.pcm
+diff ../dataqa350/a41misc2.pcm ../dataqa351/a41misc2.pcm
+diff ../dataqa350/a42misc2.pcm ../dataqa351/a42misc2.pcm
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+diff ../dataqa350/i30_86purenb.pcm ../dataqa351/i30_86purenb.pcm
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+diff ../dataqa350/i30_87purenb.pcm ../dataqa351/i30_87purenb.pcm
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+diff ../dataqa350/i60_88purenb.pcm ../dataqa351/i60_88purenb.pcm
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+diff ../dataqa350/i60_89purenb.pcm ../dataqa351/i60_89purenb.pcm
+diff ../dataqa350/i30_90purenb.pcm ../dataqa351/i30_90purenb.pcm
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+diff ../dataqa350/a43purenb.pcm ../dataqa351/a43purenb.pcm
+diff ../dataqa350/a44purenb.pcm ../dataqa351/a44purenb.pcm
+diff ../dataqa350/a45purenb.pcm ../dataqa351/a45purenb.pcm
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+diff ../dataqa350/i60_91sawsweep_380_60.pcm ../dataqa351/i60_91sawsweep_380_60.pcm
+diff ../dataqa350/i30_92sawsweep_380_60.pcm ../dataqa351/i30_92sawsweep_380_60.pcm
+diff ../dataqa350/i60_92sawsweep_380_60.pcm ../dataqa351/i60_92sawsweep_380_60.pcm
+diff ../dataqa350/i30_93sawsweep_380_60.pcm ../dataqa351/i30_93sawsweep_380_60.pcm
+diff ../dataqa350/i60_93sawsweep_380_60.pcm ../dataqa351/i60_93sawsweep_380_60.pcm
+diff ../dataqa350/i30_94sawsweep_380_60.pcm ../dataqa351/i30_94sawsweep_380_60.pcm
+diff ../dataqa350/i60_94sawsweep_380_60.pcm ../dataqa351/i60_94sawsweep_380_60.pcm
+diff ../dataqa350/i30_95sawsweep_380_60.pcm ../dataqa351/i30_95sawsweep_380_60.pcm
+diff ../dataqa350/i60_95sawsweep_380_60.pcm ../dataqa351/i60_95sawsweep_380_60.pcm
+diff ../dataqa350/i30_96sawsweep_380_60.pcm ../dataqa351/i30_96sawsweep_380_60.pcm
+diff ../dataqa350/i60_96sawsweep_380_60.pcm ../dataqa351/i60_96sawsweep_380_60.pcm
+diff ../dataqa350/a46sawsweep_380_60.pcm ../dataqa351/a46sawsweep_380_60.pcm
+diff ../dataqa350/a47sawsweep_380_60.pcm ../dataqa351/a47sawsweep_380_60.pcm
+diff ../dataqa350/a48sawsweep_380_60.pcm ../dataqa351/a48sawsweep_380_60.pcm
+diff ../dataqa350/i30_97sinesweep.pcm ../dataqa351/i30_97sinesweep.pcm
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+diff ../dataqa350/i30_98sinesweep.pcm ../dataqa351/i30_98sinesweep.pcm
+diff ../dataqa350/i60_98sinesweep.pcm ../dataqa351/i60_98sinesweep.pcm
+diff ../dataqa350/i30_99sinesweep.pcm ../dataqa351/i30_99sinesweep.pcm
+diff ../dataqa350/i60_99sinesweep.pcm ../dataqa351/i60_99sinesweep.pcm
+diff ../dataqa350/i30_100sinesweep.pcm ../dataqa351/i30_100sinesweep.pcm
+diff ../dataqa350/i60_100sinesweep.pcm ../dataqa351/i60_100sinesweep.pcm
+diff ../dataqa350/i30_101sinesweep.pcm ../dataqa351/i30_101sinesweep.pcm
+diff ../dataqa350/i60_101sinesweep.pcm ../dataqa351/i60_101sinesweep.pcm
+diff ../dataqa350/i30_102sinesweep.pcm ../dataqa351/i30_102sinesweep.pcm
+diff ../dataqa350/i60_102sinesweep.pcm ../dataqa351/i60_102sinesweep.pcm
+diff ../dataqa350/a49sinesweep.pcm ../dataqa351/a49sinesweep.pcm
+diff ../dataqa350/a50sinesweep.pcm ../dataqa351/a50sinesweep.pcm
+diff ../dataqa350/a51sinesweep.pcm ../dataqa351/a51sinesweep.pcm
+diff ../dataqa350/i30_103sinesweep_half.pcm ../dataqa351/i30_103sinesweep_half.pcm
+diff ../dataqa350/i60_103sinesweep_half.pcm ../dataqa351/i60_103sinesweep_half.pcm
+diff ../dataqa350/i30_104sinesweep_half.pcm ../dataqa351/i30_104sinesweep_half.pcm
+diff ../dataqa350/i60_104sinesweep_half.pcm ../dataqa351/i60_104sinesweep_half.pcm
+diff ../dataqa350/i30_105sinesweep_half.pcm ../dataqa351/i30_105sinesweep_half.pcm
+diff ../dataqa350/i60_105sinesweep_half.pcm ../dataqa351/i60_105sinesweep_half.pcm
+diff ../dataqa350/i30_106sinesweep_half.pcm ../dataqa351/i30_106sinesweep_half.pcm
+diff ../dataqa350/i60_106sinesweep_half.pcm ../dataqa351/i60_106sinesweep_half.pcm
+diff ../dataqa350/i30_107sinesweep_half.pcm ../dataqa351/i30_107sinesweep_half.pcm
+diff ../dataqa350/i60_107sinesweep_half.pcm ../dataqa351/i60_107sinesweep_half.pcm
+diff ../dataqa350/i30_108sinesweep_half.pcm ../dataqa351/i30_108sinesweep_half.pcm
+diff ../dataqa350/i60_108sinesweep_half.pcm ../dataqa351/i60_108sinesweep_half.pcm
+diff ../dataqa350/a52sinesweep_half.pcm ../dataqa351/a52sinesweep_half.pcm
+diff ../dataqa350/a53sinesweep_half.pcm ../dataqa351/a53sinesweep_half.pcm
+diff ../dataqa350/a54sinesweep_half.pcm ../dataqa351/a54sinesweep_half.pcm
+diff ../dataqa350/i30_109speechmusic.pcm ../dataqa351/i30_109speechmusic.pcm
+diff ../dataqa350/i60_109speechmusic.pcm ../dataqa351/i60_109speechmusic.pcm
+diff ../dataqa350/i30_110speechmusic.pcm ../dataqa351/i30_110speechmusic.pcm
+diff ../dataqa350/i60_110speechmusic.pcm ../dataqa351/i60_110speechmusic.pcm
+diff ../dataqa350/i30_111speechmusic.pcm ../dataqa351/i30_111speechmusic.pcm
+diff ../dataqa350/i60_111speechmusic.pcm ../dataqa351/i60_111speechmusic.pcm
+diff ../dataqa350/i30_112speechmusic.pcm ../dataqa351/i30_112speechmusic.pcm
+diff ../dataqa350/i60_112speechmusic.pcm ../dataqa351/i60_112speechmusic.pcm
+diff ../dataqa350/i30_113speechmusic.pcm ../dataqa351/i30_113speechmusic.pcm
+diff ../dataqa350/i60_113speechmusic.pcm ../dataqa351/i60_113speechmusic.pcm
+diff ../dataqa350/i30_114speechmusic.pcm ../dataqa351/i30_114speechmusic.pcm
+diff ../dataqa350/i60_114speechmusic.pcm ../dataqa351/i60_114speechmusic.pcm
+diff ../dataqa350/a55speechmusic.pcm ../dataqa351/a55speechmusic.pcm
+diff ../dataqa350/a56speechmusic.pcm ../dataqa351/a56speechmusic.pcm
+diff ../dataqa350/a57speechmusic.pcm ../dataqa351/a57speechmusic.pcm
+diff ../dataqa350/i30_115speechmusic_nb.pcm ../dataqa351/i30_115speechmusic_nb.pcm
+diff ../dataqa350/i60_115speechmusic_nb.pcm ../dataqa351/i60_115speechmusic_nb.pcm
+diff ../dataqa350/i30_116speechmusic_nb.pcm ../dataqa351/i30_116speechmusic_nb.pcm
+diff ../dataqa350/i60_116speechmusic_nb.pcm ../dataqa351/i60_116speechmusic_nb.pcm
+diff ../dataqa350/i30_117speechmusic_nb.pcm ../dataqa351/i30_117speechmusic_nb.pcm
+diff ../dataqa350/i60_117speechmusic_nb.pcm ../dataqa351/i60_117speechmusic_nb.pcm
+diff ../dataqa350/i30_118speechmusic_nb.pcm ../dataqa351/i30_118speechmusic_nb.pcm
+diff ../dataqa350/i60_118speechmusic_nb.pcm ../dataqa351/i60_118speechmusic_nb.pcm
+diff ../dataqa350/i30_119speechmusic_nb.pcm ../dataqa351/i30_119speechmusic_nb.pcm
+diff ../dataqa350/i60_119speechmusic_nb.pcm ../dataqa351/i60_119speechmusic_nb.pcm
+diff ../dataqa350/i30_120speechmusic_nb.pcm ../dataqa351/i30_120speechmusic_nb.pcm
+diff ../dataqa350/i60_120speechmusic_nb.pcm ../dataqa351/i60_120speechmusic_nb.pcm
+diff ../dataqa350/a58speechmusic_nb.pcm ../dataqa351/a58speechmusic_nb.pcm
+diff ../dataqa350/a59speechmusic_nb.pcm ../dataqa351/a59speechmusic_nb.pcm
+diff ../dataqa350/a60speechmusic_nb.pcm ../dataqa351/a60speechmusic_nb.pcm
+diff ../dataqa350/i30_121speechoffice0dB.pcm ../dataqa351/i30_121speechoffice0dB.pcm
+diff ../dataqa350/i60_121speechoffice0dB.pcm ../dataqa351/i60_121speechoffice0dB.pcm
+diff ../dataqa350/i30_122speechoffice0dB.pcm ../dataqa351/i30_122speechoffice0dB.pcm
+diff ../dataqa350/i60_122speechoffice0dB.pcm ../dataqa351/i60_122speechoffice0dB.pcm
+diff ../dataqa350/i30_123speechoffice0dB.pcm ../dataqa351/i30_123speechoffice0dB.pcm
+diff ../dataqa350/i60_123speechoffice0dB.pcm ../dataqa351/i60_123speechoffice0dB.pcm
+diff ../dataqa350/i30_124speechoffice0dB.pcm ../dataqa351/i30_124speechoffice0dB.pcm
+diff ../dataqa350/i60_124speechoffice0dB.pcm ../dataqa351/i60_124speechoffice0dB.pcm
+diff ../dataqa350/i30_125speechoffice0dB.pcm ../dataqa351/i30_125speechoffice0dB.pcm
+diff ../dataqa350/i60_125speechoffice0dB.pcm ../dataqa351/i60_125speechoffice0dB.pcm
+diff ../dataqa350/i30_126speechoffice0dB.pcm ../dataqa351/i30_126speechoffice0dB.pcm
+diff ../dataqa350/i60_126speechoffice0dB.pcm ../dataqa351/i60_126speechoffice0dB.pcm
+diff ../dataqa350/a61speechoffice0dB.pcm ../dataqa351/a61speechoffice0dB.pcm
+diff ../dataqa350/a62speechoffice0dB.pcm ../dataqa351/a62speechoffice0dB.pcm
+diff ../dataqa350/a63speechoffice0dB.pcm ../dataqa351/a63speechoffice0dB.pcm
+diff ../dataqa350/i30_127speech_and_misc_NB.pcm ../dataqa351/i30_127speech_and_misc_NB.pcm
+diff ../dataqa350/i60_127speech_and_misc_NB.pcm ../dataqa351/i60_127speech_and_misc_NB.pcm
+diff ../dataqa350/i30_128speech_and_misc_NB.pcm ../dataqa351/i30_128speech_and_misc_NB.pcm
+diff ../dataqa350/i60_128speech_and_misc_NB.pcm ../dataqa351/i60_128speech_and_misc_NB.pcm
+diff ../dataqa350/i30_129speech_and_misc_NB.pcm ../dataqa351/i30_129speech_and_misc_NB.pcm
+diff ../dataqa350/i60_129speech_and_misc_NB.pcm ../dataqa351/i60_129speech_and_misc_NB.pcm
+diff ../dataqa350/i30_130speech_and_misc_NB.pcm ../dataqa351/i30_130speech_and_misc_NB.pcm
+diff ../dataqa350/i60_130speech_and_misc_NB.pcm ../dataqa351/i60_130speech_and_misc_NB.pcm
+diff ../dataqa350/i30_131speech_and_misc_NB.pcm ../dataqa351/i30_131speech_and_misc_NB.pcm
+diff ../dataqa350/i60_131speech_and_misc_NB.pcm ../dataqa351/i60_131speech_and_misc_NB.pcm
+diff ../dataqa350/i30_132speech_and_misc_NB.pcm ../dataqa351/i30_132speech_and_misc_NB.pcm
+diff ../dataqa350/i60_132speech_and_misc_NB.pcm ../dataqa351/i60_132speech_and_misc_NB.pcm
+diff ../dataqa350/a64speech_and_misc_NB.pcm ../dataqa351/a64speech_and_misc_NB.pcm
+diff ../dataqa350/a65speech_and_misc_NB.pcm ../dataqa351/a65speech_and_misc_NB.pcm
+diff ../dataqa350/a66speech_and_misc_NB.pcm ../dataqa351/a66speech_and_misc_NB.pcm
+diff ../dataqa350/i30_133speech_and_misc_WB.pcm ../dataqa351/i30_133speech_and_misc_WB.pcm
+diff ../dataqa350/i60_133speech_and_misc_WB.pcm ../dataqa351/i60_133speech_and_misc_WB.pcm
+diff ../dataqa350/i30_134speech_and_misc_WB.pcm ../dataqa351/i30_134speech_and_misc_WB.pcm
+diff ../dataqa350/i60_134speech_and_misc_WB.pcm ../dataqa351/i60_134speech_and_misc_WB.pcm
+diff ../dataqa350/i30_135speech_and_misc_WB.pcm ../dataqa351/i30_135speech_and_misc_WB.pcm
+diff ../dataqa350/i60_135speech_and_misc_WB.pcm ../dataqa351/i60_135speech_and_misc_WB.pcm
+diff ../dataqa350/i30_136speech_and_misc_WB.pcm ../dataqa351/i30_136speech_and_misc_WB.pcm
+diff ../dataqa350/i60_136speech_and_misc_WB.pcm ../dataqa351/i60_136speech_and_misc_WB.pcm
+diff ../dataqa350/i30_137speech_and_misc_WB.pcm ../dataqa351/i30_137speech_and_misc_WB.pcm
+diff ../dataqa350/i60_137speech_and_misc_WB.pcm ../dataqa351/i60_137speech_and_misc_WB.pcm
+diff ../dataqa350/i30_138speech_and_misc_WB.pcm ../dataqa351/i30_138speech_and_misc_WB.pcm
+diff ../dataqa350/i60_138speech_and_misc_WB.pcm ../dataqa351/i60_138speech_and_misc_WB.pcm
+diff ../dataqa350/a67speech_and_misc_WB.pcm ../dataqa351/a67speech_and_misc_WB.pcm
+diff ../dataqa350/a68speech_and_misc_WB.pcm ../dataqa351/a68speech_and_misc_WB.pcm
+diff ../dataqa350/a69speech_and_misc_WB.pcm ../dataqa351/a69speech_and_misc_WB.pcm
+diff ../dataqa350/i30_139testM4.pcm ../dataqa351/i30_139testM4.pcm
+diff ../dataqa350/i60_139testM4.pcm ../dataqa351/i60_139testM4.pcm
+diff ../dataqa350/i30_140testM4.pcm ../dataqa351/i30_140testM4.pcm
+diff ../dataqa350/i60_140testM4.pcm ../dataqa351/i60_140testM4.pcm
+diff ../dataqa350/i30_141testM4.pcm ../dataqa351/i30_141testM4.pcm
+diff ../dataqa350/i60_141testM4.pcm ../dataqa351/i60_141testM4.pcm
+diff ../dataqa350/i30_142testM4.pcm ../dataqa351/i30_142testM4.pcm
+diff ../dataqa350/i60_142testM4.pcm ../dataqa351/i60_142testM4.pcm
+diff ../dataqa350/i30_143testM4.pcm ../dataqa351/i30_143testM4.pcm
+diff ../dataqa350/i60_143testM4.pcm ../dataqa351/i60_143testM4.pcm
+diff ../dataqa350/i30_144testM4.pcm ../dataqa351/i30_144testM4.pcm
+diff ../dataqa350/i60_144testM4.pcm ../dataqa351/i60_144testM4.pcm
+diff ../dataqa350/a70testM4.pcm ../dataqa351/a70testM4.pcm
+diff ../dataqa350/a71testM4.pcm ../dataqa351/a71testM4.pcm
+diff ../dataqa350/a72testM4.pcm ../dataqa351/a72testM4.pcm
+diff ../dataqa350/i30_145testM4D_rev.pcm ../dataqa351/i30_145testM4D_rev.pcm
+diff ../dataqa350/i60_145testM4D_rev.pcm ../dataqa351/i60_145testM4D_rev.pcm
+diff ../dataqa350/i30_146testM4D_rev.pcm ../dataqa351/i30_146testM4D_rev.pcm
+diff ../dataqa350/i60_146testM4D_rev.pcm ../dataqa351/i60_146testM4D_rev.pcm
+diff ../dataqa350/i30_147testM4D_rev.pcm ../dataqa351/i30_147testM4D_rev.pcm
+diff ../dataqa350/i60_147testM4D_rev.pcm ../dataqa351/i60_147testM4D_rev.pcm
+diff ../dataqa350/i30_148testM4D_rev.pcm ../dataqa351/i30_148testM4D_rev.pcm
+diff ../dataqa350/i60_148testM4D_rev.pcm ../dataqa351/i60_148testM4D_rev.pcm
+diff ../dataqa350/i30_149testM4D_rev.pcm ../dataqa351/i30_149testM4D_rev.pcm
+diff ../dataqa350/i60_149testM4D_rev.pcm ../dataqa351/i60_149testM4D_rev.pcm
+diff ../dataqa350/i30_150testM4D_rev.pcm ../dataqa351/i30_150testM4D_rev.pcm
+diff ../dataqa350/i60_150testM4D_rev.pcm ../dataqa351/i60_150testM4D_rev.pcm
+diff ../dataqa350/a73testM4D_rev.pcm ../dataqa351/a73testM4D_rev.pcm
+diff ../dataqa350/a74testM4D_rev.pcm ../dataqa351/a74testM4D_rev.pcm
+diff ../dataqa350/a75testM4D_rev.pcm ../dataqa351/a75testM4D_rev.pcm
+diff ../dataqa350/i30_151testM4D.pcm ../dataqa351/i30_151testM4D.pcm
+diff ../dataqa350/i60_151testM4D.pcm ../dataqa351/i60_151testM4D.pcm
+diff ../dataqa350/i30_152testM4D.pcm ../dataqa351/i30_152testM4D.pcm
+diff ../dataqa350/i60_152testM4D.pcm ../dataqa351/i60_152testM4D.pcm
+diff ../dataqa350/i30_153testM4D.pcm ../dataqa351/i30_153testM4D.pcm
+diff ../dataqa350/i60_153testM4D.pcm ../dataqa351/i60_153testM4D.pcm
+diff ../dataqa350/i30_154testM4D.pcm ../dataqa351/i30_154testM4D.pcm
+diff ../dataqa350/i60_154testM4D.pcm ../dataqa351/i60_154testM4D.pcm
+diff ../dataqa350/i30_155testM4D.pcm ../dataqa351/i30_155testM4D.pcm
+diff ../dataqa350/i60_155testM4D.pcm ../dataqa351/i60_155testM4D.pcm
+diff ../dataqa350/i30_156testM4D.pcm ../dataqa351/i30_156testM4D.pcm
+diff ../dataqa350/i60_156testM4D.pcm ../dataqa351/i60_156testM4D.pcm
+diff ../dataqa350/a76testM4D.pcm ../dataqa351/a76testM4D.pcm
+diff ../dataqa350/a77testM4D.pcm ../dataqa351/a77testM4D.pcm
+diff ../dataqa350/a78testM4D.pcm ../dataqa351/a78testM4D.pcm
+diff ../dataqa350/i30_157testfile.pcm ../dataqa351/i30_157testfile.pcm
+diff ../dataqa350/i60_157testfile.pcm ../dataqa351/i60_157testfile.pcm
+diff ../dataqa350/i30_158testfile.pcm ../dataqa351/i30_158testfile.pcm
+diff ../dataqa350/i60_158testfile.pcm ../dataqa351/i60_158testfile.pcm
+diff ../dataqa350/i30_159testfile.pcm ../dataqa351/i30_159testfile.pcm
+diff ../dataqa350/i60_159testfile.pcm ../dataqa351/i60_159testfile.pcm
+diff ../dataqa350/i30_160testfile.pcm ../dataqa351/i30_160testfile.pcm
+diff ../dataqa350/i60_160testfile.pcm ../dataqa351/i60_160testfile.pcm
+diff ../dataqa350/i30_161testfile.pcm ../dataqa351/i30_161testfile.pcm
+diff ../dataqa350/i60_161testfile.pcm ../dataqa351/i60_161testfile.pcm
+diff ../dataqa350/i30_162testfile.pcm ../dataqa351/i30_162testfile.pcm
+diff ../dataqa350/i60_162testfile.pcm ../dataqa351/i60_162testfile.pcm
+diff ../dataqa350/a79testfile.pcm ../dataqa351/a79testfile.pcm
+diff ../dataqa350/a80testfile.pcm ../dataqa351/a80testfile.pcm
+diff ../dataqa350/a81testfile.pcm ../dataqa351/a81testfile.pcm
+diff ../dataqa350/i30_163tone_cisco.pcm ../dataqa351/i30_163tone_cisco.pcm
+diff ../dataqa350/i60_163tone_cisco.pcm ../dataqa351/i60_163tone_cisco.pcm
+diff ../dataqa350/i30_164tone_cisco.pcm ../dataqa351/i30_164tone_cisco.pcm
+diff ../dataqa350/i60_164tone_cisco.pcm ../dataqa351/i60_164tone_cisco.pcm
+diff ../dataqa350/i30_165tone_cisco.pcm ../dataqa351/i30_165tone_cisco.pcm
+diff ../dataqa350/i60_165tone_cisco.pcm ../dataqa351/i60_165tone_cisco.pcm
+diff ../dataqa350/i30_166tone_cisco.pcm ../dataqa351/i30_166tone_cisco.pcm
+diff ../dataqa350/i60_166tone_cisco.pcm ../dataqa351/i60_166tone_cisco.pcm
+diff ../dataqa350/i30_167tone_cisco.pcm ../dataqa351/i30_167tone_cisco.pcm
+diff ../dataqa350/i60_167tone_cisco.pcm ../dataqa351/i60_167tone_cisco.pcm
+diff ../dataqa350/i30_168tone_cisco.pcm ../dataqa351/i30_168tone_cisco.pcm
+diff ../dataqa350/i60_168tone_cisco.pcm ../dataqa351/i60_168tone_cisco.pcm
+diff ../dataqa350/a82tone_cisco.pcm ../dataqa351/a82tone_cisco.pcm
+diff ../dataqa350/a83tone_cisco.pcm ../dataqa351/a83tone_cisco.pcm
+diff ../dataqa350/a84tone_cisco.pcm ../dataqa351/a84tone_cisco.pcm
+diff ../dataqa350/i30_169tone_cisco_long.pcm ../dataqa351/i30_169tone_cisco_long.pcm
+diff ../dataqa350/i60_169tone_cisco_long.pcm ../dataqa351/i60_169tone_cisco_long.pcm
+diff ../dataqa350/i30_170tone_cisco_long.pcm ../dataqa351/i30_170tone_cisco_long.pcm
+diff ../dataqa350/i60_170tone_cisco_long.pcm ../dataqa351/i60_170tone_cisco_long.pcm
+diff ../dataqa350/i30_171tone_cisco_long.pcm ../dataqa351/i30_171tone_cisco_long.pcm
+diff ../dataqa350/i60_171tone_cisco_long.pcm ../dataqa351/i60_171tone_cisco_long.pcm
+diff ../dataqa350/i30_172tone_cisco_long.pcm ../dataqa351/i30_172tone_cisco_long.pcm
+diff ../dataqa350/i60_172tone_cisco_long.pcm ../dataqa351/i60_172tone_cisco_long.pcm
+diff ../dataqa350/i30_173tone_cisco_long.pcm ../dataqa351/i30_173tone_cisco_long.pcm
+diff ../dataqa350/i60_173tone_cisco_long.pcm ../dataqa351/i60_173tone_cisco_long.pcm
+diff ../dataqa350/i30_174tone_cisco_long.pcm ../dataqa351/i30_174tone_cisco_long.pcm
+diff ../dataqa350/i60_174tone_cisco_long.pcm ../dataqa351/i60_174tone_cisco_long.pcm
+diff ../dataqa350/a85tone_cisco_long.pcm ../dataqa351/a85tone_cisco_long.pcm
+diff ../dataqa350/a86tone_cisco_long.pcm ../dataqa351/a86tone_cisco_long.pcm
+diff ../dataqa350/a87tone_cisco_long.pcm ../dataqa351/a87tone_cisco_long.pcm
+diff ../dataqa350/i30_175wb_contspeech.pcm ../dataqa351/i30_175wb_contspeech.pcm
+diff ../dataqa350/i60_175wb_contspeech.pcm ../dataqa351/i60_175wb_contspeech.pcm
+diff ../dataqa350/i30_176wb_contspeech.pcm ../dataqa351/i30_176wb_contspeech.pcm
+diff ../dataqa350/i60_176wb_contspeech.pcm ../dataqa351/i60_176wb_contspeech.pcm
+diff ../dataqa350/i30_177wb_contspeech.pcm ../dataqa351/i30_177wb_contspeech.pcm
+diff ../dataqa350/i60_177wb_contspeech.pcm ../dataqa351/i60_177wb_contspeech.pcm
+diff ../dataqa350/i30_178wb_contspeech.pcm ../dataqa351/i30_178wb_contspeech.pcm
+diff ../dataqa350/i60_178wb_contspeech.pcm ../dataqa351/i60_178wb_contspeech.pcm
+diff ../dataqa350/i30_179wb_contspeech.pcm ../dataqa351/i30_179wb_contspeech.pcm
+diff ../dataqa350/i60_179wb_contspeech.pcm ../dataqa351/i60_179wb_contspeech.pcm
+diff ../dataqa350/i30_180wb_contspeech.pcm ../dataqa351/i30_180wb_contspeech.pcm
+diff ../dataqa350/i60_180wb_contspeech.pcm ../dataqa351/i60_180wb_contspeech.pcm
+diff ../dataqa350/a88wb_contspeech.pcm ../dataqa351/a88wb_contspeech.pcm
+diff ../dataqa350/a89wb_contspeech.pcm ../dataqa351/a89wb_contspeech.pcm
+diff ../dataqa350/a90wb_contspeech.pcm ../dataqa351/a90wb_contspeech.pcm
+diff ../dataqa350/i30_181wb_speech_office25db.pcm ../dataqa351/i30_181wb_speech_office25db.pcm
+diff ../dataqa350/i60_181wb_speech_office25db.pcm ../dataqa351/i60_181wb_speech_office25db.pcm
+diff ../dataqa350/i30_182wb_speech_office25db.pcm ../dataqa351/i30_182wb_speech_office25db.pcm
+diff ../dataqa350/i60_182wb_speech_office25db.pcm ../dataqa351/i60_182wb_speech_office25db.pcm
+diff ../dataqa350/i30_183wb_speech_office25db.pcm ../dataqa351/i30_183wb_speech_office25db.pcm
+diff ../dataqa350/i60_183wb_speech_office25db.pcm ../dataqa351/i60_183wb_speech_office25db.pcm
+diff ../dataqa350/i30_184wb_speech_office25db.pcm ../dataqa351/i30_184wb_speech_office25db.pcm
+diff ../dataqa350/i60_184wb_speech_office25db.pcm ../dataqa351/i60_184wb_speech_office25db.pcm
+diff ../dataqa350/i30_185wb_speech_office25db.pcm ../dataqa351/i30_185wb_speech_office25db.pcm
+diff ../dataqa350/i60_185wb_speech_office25db.pcm ../dataqa351/i60_185wb_speech_office25db.pcm
+diff ../dataqa350/i30_186wb_speech_office25db.pcm ../dataqa351/i30_186wb_speech_office25db.pcm
+diff ../dataqa350/i60_186wb_speech_office25db.pcm ../dataqa351/i60_186wb_speech_office25db.pcm
+diff ../dataqa350/a91wb_speech_office25db.pcm ../dataqa351/a91wb_speech_office25db.pcm
+diff ../dataqa350/a92wb_speech_office25db.pcm ../dataqa351/a92wb_speech_office25db.pcm
+diff ../dataqa350/a93wb_speech_office25db.pcm ../dataqa351/a93wb_speech_office25db.pcm
+diff ../dataqa350/a30_1DTMF_16kHz_short.pcm ../dataqa351/a30_1DTMF_16kHz_short.pcm
+diff ../dataqa350/a60_1DTMF_16kHz_short.pcm ../dataqa351/a60_1DTMF_16kHz_short.pcm
+diff ../dataqa350/a30_2ltest_speech_noisy.pcm ../dataqa351/a30_2ltest_speech_noisy.pcm
+diff ../dataqa350/a60_2ltest_speech_noisy.pcm ../dataqa351/a60_2ltest_speech_noisy.pcm
+diff ../dataqa350/a30_3misc2.pcm ../dataqa351/a30_3misc2.pcm
+diff ../dataqa350/a60_3misc2.pcm ../dataqa351/a60_3misc2.pcm
+diff ../dataqa350/a30_4sinesweep.pcm ../dataqa351/a30_4sinesweep.pcm
+diff ../dataqa350/a60_4sinesweep.pcm ../dataqa351/a60_4sinesweep.pcm
+diff ../dataqa350/a30_5speechmusic.pcm ../dataqa351/a30_5speechmusic.pcm
+diff ../dataqa350/a60_5speechmusic.pcm ../dataqa351/a60_5speechmusic.pcm
+diff ../dataqa350/a30_6tone_cisco.pcm ../dataqa351/a30_6tone_cisco.pcm
+diff ../dataqa350/a60_6tone_cisco.pcm ../dataqa351/a60_6tone_cisco.pcm
+diff ../dataqa350/a60_7tone_cisco.pcm ../dataqa351/a60_7tone_cisco.pcm
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt
new file mode 100644
index 0000000..9e3629b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt
@@ -0,0 +1,20 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logplc.txt
+echo "START PLC TEST" > $LOGFILE
+
+OUTDIR1=../dataqaplc_0
+OUTDIR2=../dataqaplc_1
+
+diff $OUTDIR1/outplc1.pcm $OUTDIR2/outplc1.pcm
+diff $OUTDIR1/outplc2.pcm $OUTDIR2/outplc2.pcm
+diff $OUTDIR1/outplc3.pcm $OUTDIR2/outplc3.pcm
+diff $OUTDIR1/outplc4.pcm $OUTDIR2/outplc4.pcm
+diff $OUTDIR1/outplc5.pcm $OUTDIR2/outplc5.pcm
+diff $OUTDIR1/outplc6.pcm $OUTDIR2/outplc6.pcm
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt
new file mode 100644
index 0000000..eeffc0c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt
@@ -0,0 +1,61 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logNormal.txt
+echo "START ISAC TEST" > $LOGFILE
+echo >> $LOGFILE
+
+ISAC=../Release/kenny.exe
+ISACFIXFLOAT=../Release/testFixFloat.exe
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+OUTDIR=../dataqa
+mkdir -p $OUTDIR
+
+TARGETRATE=(10000 15000 20000 25000 30000 32000)
+#echo ${CHANNELFILES[1]}
+
+index1=0
+index2=0
+
+for file in $INFILES # loop over all input files
+ do
+
+ for rate in ${TARGETRATE[*]}
+ do
+ let "index1=index1+1"
+ $ISAC -I $rate -FL 30 $INDIR/"$file" $OUTDIR/i30_$index1"$file" >> $LOGFILE
+ $ISAC -I $rate -FL 60 $INDIR/"$file" $OUTDIR/i60_$index1"$file" >> $LOGFILE
+ done
+ for channel in $CHANNELFILES
+ do
+ let "index2=index2+1"
+ $ISAC $INDIR/$channel $INDIR/"$file" $OUTDIR/a$index2"$file" >> $LOGFILE
+ done
+
+done
+
+index1=0
+
+for file in $SUBSET # loop over the subset of input files
+ do
+ let "index1=index1+1"
+ $ISAC $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL $INDIR/"$file" $OUTDIR/a30_$index1"$file" >> $LOGFILE
+ $ISAC $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
+done
+
+let "index1=index1+1"
+ $ISAC $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
+
+# Run fault test
+
+#./runiSACfault.txt
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt
new file mode 100644
index 0000000..605595c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt
@@ -0,0 +1,45 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logNB.txt
+echo "START NARROWBAND TEST" > $LOGFILE
+echo >> $LOGFILE
+
+ISAC=../Release/kenny.exe
+ISACFIXFLOAT=../Release/testFixFloat.exe
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+OUTDIR=../dataqaNB
+mkdir -p $OUTDIR
+
+TARGETRATE=(10000 15000 20000 25000 30000 32000)
+#echo ${CHANNELFILES[1]}
+
+index1=0
+index2=0
+
+# Narrowband Interfaces
+
+for file in $SUBSET # loop over all input files
+ do
+ for rate in ${TARGETRATE[*]}
+ do
+ let "index1=index1+1"
+ $ISAC $rate -FL 30 -NB 1 $INDIR/"$file" $OUTDIR/nb130_$index1"$file" >> $LOGFILE
+ $ISAC $rate -FL 60 -NB 1 $INDIR/"$file" $OUTDIR/nb160_$index1"$file" >> $LOGFILE
+ $ISAC $rate -FL 30 -NB 2 $INDIR/"$file" $OUTDIR/nb230_$index1"$file" >> $LOGFILE
+ $ISAC $rate -FL 60 -NB 2 $INDIR/"$file" $OUTDIR/nb260_$index1"$file" >> $LOGFILE
+ $ISAC $rate -FL 30 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc30_$index1"$file" >> $LOGFILE
+ $ISAC $rate -FL 60 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc60_$index1"$file" >> $LOGFILE
+ done
+
+done
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt
new file mode 100644
index 0000000..6bee6f7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt
@@ -0,0 +1,23 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logplc.txt
+echo "START PLC TEST" > $LOGFILE
+
+ISAC=../Release/kenny.exe
+
+INDIR=../data/orig
+OUTDIR=../dataqaplc_0
+mkdir -p $OUTDIR
+
+$ISAC 12000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc1.pcm
+$ISAC 20000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc2.pcm
+$ISAC 32000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc3.pcm
+$ISAC 12000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc4.pcm
+$ISAC 20000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc5.pcm
+$ISAC 32000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc6.pcm
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt
new file mode 100644
index 0000000..d8403e0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt
@@ -0,0 +1,23 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGG=loggRate.txt
+OUTDIR=../dataqaRate
+mkdir -p $OUTDIR
+
+../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_1.pcm > $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_2.pcm >> $LOGG
+../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_3.pcm >> $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_4.pcm >> $LOGG
+../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_5.pcm >> $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_6.pcm >> $LOGG
+../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_7.pcm >> $LOGG
+
+../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_11.pcm >> $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_12.pcm >> $LOGG
+../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_13.pcm >> $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_14.pcm >> $LOGG
+../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_15.pcm >> $LOGG
+../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_16.pcm >> $LOGG
+../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_17.pcm >> $LOGG
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt
new file mode 100644
index 0000000..f4d9478
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt
@@ -0,0 +1,40 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logfault.txt
+echo "START FAULT TEST" > $LOGFILE
+
+ISAC=../Release/kenny.exe
+ISACFIXFLOAT=../Release/testFixFloat.exe
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+OUTDIR=../dataqaft
+mkdir -p $OUTDIR
+
+TARGETRATE=(10000 15000 20000 25000 30000 32000)
+FAULTTEST=(1 2 3 4 5 6 7 9)
+
+index1=0
+
+file=wb_contspeech.pcm
+
+# Fault test
+for testnr in ${FAULTTEST[*]}
+ do
+ $ISAC 32000 -F $testnr $INDIR/"$file" $OUTDIR/ft$testnr"$file" >> $LOGFILE
+done
+
+# Fault test number 10, error in bitstream
+ $ISAC 32000 -F 10 $INDIR/"$file" $OUTDIR/ft10_"$file" >> $LOGFILE
+ $ISAC 32000 -F 10 -PL 10 $INDIR/"$file" $OUTDIR/ft10plc_"$file" >> $LOGFILE
+ $ISAC 32000 -F 10 -NB 1 $INDIR/"$file" $OUTDIR/ft10nb1_"$file" >> $LOGFILE
+ $ISAC 32000 -F 10 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/ft10nb2_"$file" >> $LOGFILE
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt
new file mode 100644
index 0000000..c9e02df
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt
@@ -0,0 +1,47 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logfxfl.txt
+echo "START FIX-FLOAT TEST" > $LOGFILE
+
+
+ISACFIXFLOAT=../testFixFloat.exe
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+OUTDIR=../dataqafxfl
+mkdir -p $OUTDIR
+
+index1=0
+
+for file in $INFILES # loop over all input files
+ do
+
+ for channel in $CHANNELFILES
+ do
+ let "index1=index1+1"
+
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
+ done
+
+done
+
+index1=0
+
+for file in $SUBSET # loop over the subset of input files
+ do
+ let "index1=index1+1"
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
+done
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c
new file mode 100644
index 0000000..1b9c44c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c
@@ -0,0 +1,847 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* kenny.c - Main function for the iSAC coder */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <ctype.h>
+
+#include "isacfix.h"
+
+
+/* Defines */
+#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
+#define MAX_FRAMESAMPLES 960 /* max number of samples per frame (= 60 ms frame) */
+#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
+#define FS 16000 /* sampling frequency (Hz) */
+
+/* Function for reading audio data from PCM file */
+int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
+
+ short k, rlen, status = 0;
+
+ rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
+ if (rlen < length) {
+ for (k = rlen; k < length; k++)
+ data[k] = 0;
+ status = 1;
+ }
+
+ return status;
+}
+
+/* Struct for bottleneck model */
+typedef struct {
+ WebRtc_UWord32 send_time; /* samples */
+ WebRtc_UWord32 arrival_time; /* samples */
+ WebRtc_UWord32 sample_count; /* samples */
+ WebRtc_UWord16 rtp_number;
+} BottleNeckModel;
+
+void get_arrival_time(int current_framesamples, /* samples */
+ int packet_size, /* bytes */
+ int bottleneck, /* excluding headers; bits/s */
+ BottleNeckModel *BN_data)
+{
+ const int HeaderSize = 35;
+ int HeaderRate;
+
+ HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
+
+ /* everything in samples */
+ BN_data->sample_count = BN_data->sample_count + current_framesamples;
+
+ BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+ BN_data->send_time += current_framesamples;
+
+ if (BN_data->arrival_time < BN_data->sample_count)
+ BN_data->arrival_time = BN_data->sample_count;
+
+ BN_data->rtp_number++;
+}
+
+void get_arrival_time2(int current_framesamples,
+ int current_delay,
+ BottleNeckModel *BN_data)
+{
+ if (current_delay == -1)
+ //dropped packet
+ {
+ BN_data->arrival_time += current_framesamples;
+ }
+ else if (current_delay != -2)
+ {
+ //
+ BN_data->arrival_time += (current_framesamples + ((FS/1000) * current_delay));
+ }
+ //else
+ //current packet has same timestamp as previous packet
+
+ BN_data->rtp_number++;
+}
+
+int main(int argc, char* argv[])
+{
+
+ char inname[100], outname[100], outbitsname[100], bottleneck_file[100];
+ FILE *inp, *outp, *f_bn, *outbits;
+ int endfile;
+
+ int i, errtype, h = 0, k, packetLossPercent = 0;
+ WebRtc_Word16 CodingMode;
+ WebRtc_Word16 bottleneck;
+ WebRtc_Word16 framesize = 30; /* ms */
+ int cur_framesmpls, err = 0, lostPackets = 0;
+
+ /* Runtime statistics */
+ double starttime, runtime, length_file;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 framecnt, declen = 0;
+ WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
+ WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
+ WebRtc_UWord16 streamdata[500];
+ WebRtc_Word16 speechType[1];
+ WebRtc_Word16 prevFrameSize = 1;
+ WebRtc_Word16 rateBPS = 0;
+ WebRtc_Word16 fixedFL = 0;
+ WebRtc_Word16 payloadSize = 0;
+ WebRtc_Word32 payloadRate = 0;
+ int setControlBWE = 0;
+ int readLoss;
+ FILE *plFile = NULL;
+
+ char version_number[20];
+ char tmpBit[5] = ".bit";
+
+ double kbps;
+ int totalbits =0;
+ int totalsmpls =0;
+#ifdef _DEBUG
+ FILE *fy;
+#endif
+ WebRtc_Word16 testNum, testCE;
+
+ FILE *fp_gns = NULL;
+ int gns = 0;
+ int cur_delay = 0;
+ char gns_file[100];
+
+ int nbTest = 0;
+ WebRtc_Word16 lostFrame;
+ float scale = (float)0.7;
+ /* only one structure used for ISAC encoder */
+ ISACFIX_MainStruct *ISAC_main_inst;
+
+ /* For fault test 10, garbage data */
+ FILE *seedfile;
+ unsigned int random_seed = (unsigned int) time(NULL);//1196764538
+
+ BottleNeckModel BN_data;
+ f_bn = NULL;
+
+#ifdef _DEBUG
+ fy = fopen("bit_rate.dat", "w");
+ fclose(fy);
+ fy = fopen("bytes_frames.dat", "w");
+ fclose(fy);
+#endif
+
+ readLoss = 0;
+ packetLossPercent = 0;
+
+ /* Handling wrong input arguments in the command line */
+ if ((argc<3) || (argc>21)) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ WebRtcIsacfix_version(version_number);
+ printf("iSAC version %s \n\n", version_number);
+
+ printf("Usage:\n\n");
+ printf("./kenny.exe [-F num][-I] bottleneck_value infile outfile \n\n");
+ printf("with:\n");
+ printf("[-I] :if -I option is specified, the coder will use\n");
+ printf(" an instantaneous Bottleneck value. If not, it\n");
+ printf(" will be an adaptive Bottleneck value.\n\n");
+ printf("bottleneck_value :the value of the bottleneck provided either\n");
+ printf(" as a fixed value (e.g. 25000) or\n");
+ printf(" read from a file (e.g. bottleneck.txt)\n\n");
+ printf("[-INITRATE num] :Set a new value for initial rate. Note! Only used"
+ " in adaptive mode.\n\n");
+ printf("[-FL num] :Set (initial) frame length in msec. Valid length"
+ " are 30 and 60 msec.\n\n");
+ printf("[-FIXED_FL] :Frame length will be fixed to initial value.\n\n");
+ printf("[-MAX num] :Set the limit for the payload size of iSAC"
+ " in bytes. \n");
+ printf(" Minimum 100, maximum 400.\n\n");
+ printf("[-MAXRATE num] :Set the maxrate for iSAC in bits per second. \n");
+ printf(" Minimum 32000, maximum 53400.\n\n");
+ printf("[-F num] :if -F option is specified, the test function\n");
+ printf(" will run the iSAC API fault scenario specified"
+ " by the\n");
+ printf(" supplied number.\n");
+ printf(" F 1 - Call encoder prior to init encoder call\n");
+ printf(" F 2 - Call decoder prior to init decoder call\n");
+ printf(" F 3 - Call decoder prior to encoder call\n");
+ printf(" F 4 - Call decoder with a too short coded"
+ " sequence\n");
+ printf(" F 5 - Call decoder with a too long coded"
+ " sequence\n");
+ printf(" F 6 - Call decoder with random bit stream\n");
+ printf(" F 7 - Call init encoder/decoder at random"
+ " during a call\n");
+ printf(" F 8 - Call encoder/decoder without having"
+ " allocated memory for \n");
+ printf(" encoder/decoder instance\n");
+ printf(" F 9 - Call decodeB without calling decodeA\n");
+ printf(" F 10 - Call decodeB with garbage data\n");
+ printf("[-PL num] : if -PL option is specified 0<num<100 will "
+ "specify the\n");
+ printf(" percentage of packet loss\n\n");
+ printf("[-G file] : if -G option is specified the file given is"
+ " a .gns file\n");
+ printf(" that represents a network profile\n\n");
+ printf("[-NB num] : if -NB option, use the narrowband interfaces\n");
+ printf(" num=1 => encode with narrowband encoder"
+ " (infile is narrowband)\n");
+ printf(" num=2 => decode with narrowband decoder"
+ " (outfile is narrowband)\n\n");
+ printf("[-CE num] : Test of APIs used by Conference Engine.\n");
+ printf(" CE 1 - createInternal, freeInternal,"
+ " getNewBitstream \n");
+ printf(" CE 2 - transcode, getBWE \n");
+ printf(" CE 3 - getSendBWE, setSendBWE. \n\n");
+ printf("[-RTP_INIT num] : if -RTP_INIT option is specified num will be"
+ " the initial\n");
+ printf(" value of the rtp sequence number.\n\n");
+ printf("infile : Normal speech input file\n\n");
+ printf("outfile : Speech output file\n\n");
+ printf("Example usage : \n\n");
+ printf("./kenny.exe -I bottleneck.txt speechIn.pcm speechOut.pcm\n\n");
+ exit(0);
+
+ }
+
+ /* Print version number */
+ WebRtcIsacfix_version(version_number);
+ printf("iSAC version %s \n\n", version_number);
+
+ /* Loop over all command line arguments */
+ CodingMode = 0;
+ testNum = 0;
+ testCE = 0;
+ for (i = 1; i < argc-2;i++) {
+ /* Instantaneous mode */
+ if (!strcmp ("-I", argv[i])) {
+ printf("\nInstantaneous BottleNeck\n");
+ CodingMode = 1;
+ i++;
+ }
+
+ /* Set (initial) bottleneck value */
+ if (!strcmp ("-INITRATE", argv[i])) {
+ rateBPS = atoi(argv[i + 1]);
+ setControlBWE = 1;
+ if ((rateBPS < 10000) || (rateBPS > 32000)) {
+ printf("\n%d is not a initial rate. "
+ "Valid values are in the range 10000 to 32000.\n", rateBPS);
+ exit(0);
+ }
+ printf("\nNew initial rate: %d\n", rateBPS);
+ i++;
+ }
+
+ /* Set (initial) framelength */
+ if (!strcmp ("-FL", argv[i])) {
+ framesize = atoi(argv[i + 1]);
+ if ((framesize != 30) && (framesize != 60)) {
+ printf("\n%d is not a valid frame length. "
+ "Valid length are 30 and 60 msec.\n", framesize);
+ exit(0);
+ }
+ printf("\nFrame Length: %d\n", framesize);
+ i++;
+ }
+
+ /* Fixed frame length */
+ if (!strcmp ("-FIXED_FL", argv[i])) {
+ fixedFL = 1;
+ setControlBWE = 1;
+ }
+
+ /* Set maximum allowed payload size in bytes */
+ if (!strcmp ("-MAX", argv[i])) {
+ payloadSize = atoi(argv[i + 1]);
+ printf("Maximum Payload Size: %d\n", payloadSize);
+ i++;
+ }
+
+ /* Set maximum rate in bytes */
+ if (!strcmp ("-MAXRATE", argv[i])) {
+ payloadRate = atoi(argv[i + 1]);
+ printf("Maximum Rate in kbps: %d\n", payloadRate);
+ i++;
+ }
+
+ /* Test of fault scenarious */
+ if (!strcmp ("-F", argv[i])) {
+ testNum = atoi(argv[i + 1]);
+ printf("\nFault test: %d\n", testNum);
+ if (testNum < 1 || testNum > 10) {
+ printf("\n%d is not a valid Fault Scenario number."
+ " Valid Fault Scenarios are numbered 1-10.\n", testNum);
+ exit(0);
+ }
+ i++;
+ }
+
+ /* Packet loss test */
+ if (!strcmp ("-PL", argv[i])) {
+ if( isdigit( *argv[i+1] ) ) {
+ packetLossPercent = atoi( argv[i+1] );
+ if( (packetLossPercent < 0) | (packetLossPercent > 100) ) {
+ printf( "\nInvalid packet loss perentage \n" );
+ exit( 0 );
+ }
+ if( packetLossPercent > 0 ) {
+ printf( "\nSimulating %d %% of independent packet loss\n",
+ packetLossPercent );
+ } else {
+ printf( "\nNo Packet Loss Is Simulated \n" );
+ }
+ readLoss = 0;
+ } else {
+ readLoss = 1;
+ plFile = fopen( argv[i+1], "rb" );
+ if( plFile == NULL ) {
+ printf( "\n couldn't open the frameloss file: %s\n", argv[i+1] );
+ exit( 0 );
+ }
+ printf( "\nSimulating packet loss through the given "
+ "channel file: %s\n", argv[i+1] );
+ }
+ i++;
+ }
+
+ /* Random packetlosses */
+ if (!strcmp ("-rnd", argv[i])) {
+ srand(time(NULL) );
+ printf( "\n Random pattern in lossed packets \n" );
+ }
+
+ /* Use gns file */
+ if (!strcmp ("-G", argv[i])) {
+ sscanf(argv[i + 1], "%s", gns_file);
+ fp_gns = fopen(gns_file, "rb");
+ if (fp_gns == NULL) {
+ printf("Cannot read file %s.\n", gns_file);
+ exit(0);
+ }
+ gns = 1;
+ i++;
+ }
+
+ /* Run Narrowband interfaces (either encoder or decoder) */
+ if (!strcmp ("-NB", argv[i])) {
+ nbTest = atoi(argv[i + 1]);
+ i++;
+ }
+
+ /* Run Conference Engine APIs */
+ if (!strcmp ("-CE", argv[i])) {
+ testCE = atoi(argv[i + 1]);
+ if (testCE==1 || testCE==2) {
+ i++;
+ scale = (float)atof( argv[i+1] );
+ } else if (testCE < 1 || testCE > 3) {
+ printf("\n%d is not a valid CE-test number, valid Fault "
+ "Scenarios are numbered 1-3\n", testCE);
+ exit(0);
+ }
+ i++;
+ }
+
+ /* Set initial RTP number */
+ if (!strcmp ("-RTP_INIT", argv[i])) {
+ i++;
+ }
+ }
+
+ /* Get Bottleneck value */
+ /* Gns files and bottleneck should not and can not be used simultaneously */
+ bottleneck = atoi(argv[CodingMode+1]);
+ if (bottleneck == 0 && gns == 0) {
+ sscanf(argv[CodingMode+1], "%s", bottleneck_file);
+ f_bn = fopen(bottleneck_file, "rb");
+ if (f_bn == NULL) {
+ printf("No value provided for BottleNeck and cannot read file %s\n", bottleneck_file);
+ exit(0);
+ } else {
+ int aux_var;
+ printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
+ if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+ exit(0);
+ }
+ }
+ bottleneck = (WebRtc_Word16)aux_var;
+ /* Bottleneck is a cosine function
+ * Matlab code for writing the bottleneck file:
+ * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
+ * fid = fopen('bottleneck.txt', 'wb');
+ * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
+ */
+ }
+ } else {
+ f_bn = NULL;
+ printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+ }
+
+ if (CodingMode == 0) {
+ printf("\nAdaptive BottleNeck\n");
+ }
+
+ /* Get Input and Output files */
+ sscanf(argv[argc-2], "%s", inname);
+ sscanf(argv[argc-1], "%s", outname);
+
+ /* Add '.bit' to output bitstream file */
+ while ((int)outname[h] != 0) {
+ outbitsname[h] = outname[h];
+ h++;
+ }
+ for (k=0; k<5; k++) {
+ outbitsname[h] = tmpBit[k];
+ h++;
+ }
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" iSAC: Cannot read file %s\n", inname);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" iSAC: Cannot write file %s\n", outname);
+ exit(1);
+ }
+
+ if ((outbits = fopen(outbitsname,"wb")) == NULL) {
+ printf(" iSAC: Cannot write file %s\n", outbitsname);
+ exit(1);
+ }
+ printf("\nInput:%s\nOutput:%s\n\n", inname, outname);
+
+ /* Error test number 10, garbage data */
+ if (testNum == 10) {
+ /* Test to run decoder with garbage data */
+ srand(random_seed);
+
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ printf("Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "%u\n", random_seed);
+ fclose(seedfile);
+ }
+ }
+
+ /* Runtime statistics */
+ starttime = clock()/(double)CLOCKS_PER_SEC;
+
+ /* Initialize the ISAC and BN structs */
+ if (testNum != 8)
+ {
+ if(1){
+ err =WebRtcIsacfix_Create(&ISAC_main_inst);
+ }else{
+ /* Test the Assign functions */
+ int sss;
+ void *ppp;
+ err =WebRtcIsacfix_AssignSize(&sss);
+ ppp=malloc(sss);
+ err =WebRtcIsacfix_Assign(&ISAC_main_inst,ppp);
+ }
+ /* Error check */
+ if (err < 0) {
+ printf("\n\n Error in create.\n\n");
+ }
+ if (testCE == 1) {
+ err = WebRtcIsacfix_CreateInternal(ISAC_main_inst);
+ /* Error check */
+ if (err < 0) {
+ printf("\n\n Error in createInternal.\n\n");
+ }
+ }
+ }
+
+ /* Init of bandwidth data */
+ BN_data.send_time = 0;
+ BN_data.arrival_time = 0;
+ BN_data.sample_count = 0;
+ BN_data.rtp_number = 0;
+
+ /* Initialize encoder and decoder */
+ framecnt= 0;
+ endfile = 0;
+ if (testNum != 1) {
+ WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
+ }
+ if (testNum != 2) {
+ WebRtcIsacfix_DecoderInit(ISAC_main_inst);
+ }
+
+ if (CodingMode == 1) {
+ err = WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in control: %d.\n\n", errtype);
+ }
+ } else if(setControlBWE == 1) {
+ err = WebRtcIsacfix_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
+ }
+
+ if (payloadSize != 0) {
+ err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ }
+ if (payloadRate != 0) {
+ err = WebRtcIsacfix_SetMaxRate(ISAC_main_inst, payloadRate);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
+ exit(EXIT_FAILURE);
+ }
+ }
+
+ *speechType = 1;
+
+
+ while (endfile == 0) {
+
+ if(testNum == 7 && (rand()%2 == 0)) {
+ err = WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
+ /* Error check */
+ if (err < 0) {
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in encoderinit: %d.\n\n", errtype);
+ }
+
+ err = WebRtcIsacfix_DecoderInit(ISAC_main_inst);
+ /* Error check */
+ if (err < 0) {
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in decoderinit: %d.\n\n", errtype);
+ }
+ }
+
+
+ cur_framesmpls = 0;
+ while (1) {
+ /* Read 10 ms speech block */
+ if (nbTest != 1) {
+ endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
+ } else {
+ endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
+ }
+
+ if (testNum == 7) {
+ srand(time(NULL));
+ }
+
+ /* iSAC encoding */
+ if (!(testNum == 3 && framecnt == 0)) {
+ if (nbTest != 1) {
+ short bwe;
+
+ /* Encode */
+ stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
+ shortdata,
+ (WebRtc_Word16*)streamdata);
+
+ /* If packet is ready, and CE testing, call the different API functions
+ from the internal API. */
+ if (stream_len>0) {
+ if (testCE == 1) {
+ err = WebRtcIsacfix_ReadBwIndex((WebRtc_Word16*)streamdata, &bwe);
+ stream_len = WebRtcIsacfix_GetNewBitStream(
+ ISAC_main_inst,
+ bwe,
+ scale,
+ (WebRtc_Word16*)streamdata);
+ } else if (testCE == 2) {
+ /* transcode function not supported */
+ } else if (testCE == 3) {
+ /* Only for Function testing. The functions should normally
+ not be used in this way */
+
+ err = WebRtcIsacfix_GetDownLinkBwIndex(ISAC_main_inst, &bwe);
+ /* Error Check */
+ if (err < 0) {
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in getSendBWE: %d.\n", errtype);
+ }
+
+ err = WebRtcIsacfix_UpdateUplinkBw(ISAC_main_inst, bwe);
+ /* Error Check */
+ if (err < 0) {
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in setBWE: %d.\n", errtype);
+ }
+
+ }
+ }
+ } else {
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ stream_len = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
+ shortdata,
+ streamdata);
+#else
+ stream_len = -1;
+#endif
+ }
+ }
+ else
+ {
+ break;
+ }
+
+ if (stream_len < 0 || err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in encoder: %d.\n", errtype);
+ } else {
+ fwrite(streamdata, sizeof(char), stream_len, outbits);
+ }
+
+ cur_framesmpls += FRAMESAMPLES_10ms;
+
+ /* read next bottleneck rate */
+ if (f_bn != NULL) {
+ int aux_var;
+ if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+ exit(0);
+ }
+ }
+ bottleneck = (WebRtc_Word16)aux_var;
+ if (CodingMode == 1) {
+ WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
+ }
+ }
+
+ /* exit encoder loop if the encoder returned a bitstream */
+ if (stream_len != 0) break;
+ }
+
+ /* make coded sequence to short be inreasing */
+ /* the length the decoder expects */
+ if (testNum == 4) {
+ stream_len += 10;
+ }
+
+ /* make coded sequence to long be decreasing */
+ /* the length the decoder expects */
+ if (testNum == 5) {
+ stream_len -= 10;
+ }
+
+ if (testNum == 6) {
+ srand(time(NULL));
+ for (i = 0; i < stream_len; i++ ) {
+ streamdata[i] = rand();
+ }
+ }
+
+ /* set pointer to beginning of file */
+ if (fp_gns != NULL) {
+ if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
+ fseek(fp_gns, 0L, SEEK_SET);
+ if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
+ exit(0);
+ }
+ }
+ }
+
+ /* simulate packet handling through NetEq and the modem */
+ if (!(testNum == 3 && framecnt == 0)) {
+ if (gns == 0) {
+ get_arrival_time(cur_framesmpls, stream_len, bottleneck,
+ &BN_data);
+ } else {
+ get_arrival_time2(cur_framesmpls, cur_delay, &BN_data);
+ }
+ }
+
+ /* packet not dropped */
+ if (cur_delay != -1) {
+
+ /* Error test number 10, garbage data */
+ if (testNum == 10) {
+ for ( i = 0; i < stream_len; i++) {
+ streamdata[i] = (short) (streamdata[i] + (short) rand());
+ }
+ }
+
+ if (testNum != 9) {
+ err = WebRtcIsacfix_UpdateBwEstimate(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.send_time,
+ BN_data.arrival_time);
+
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in decoder: %d.\n", errtype);
+ }
+ }
+#ifdef _DEBUG
+ fprintf(stderr," \rframe = %7d", framecnt);
+#endif
+
+ if( readLoss == 1 ) {
+ if( fread( &lostFrame, sizeof(WebRtc_Word16), 1, plFile ) != 1 ) {
+ rewind( plFile );
+ }
+ lostFrame = !lostFrame;
+ } else {
+ lostFrame = (rand()%100 < packetLossPercent);
+ }
+
+
+
+ /* iSAC decoding */
+ if( lostFrame && framecnt > 0) {
+ if (nbTest !=2) {
+ declen = WebRtcIsacfix_DecodePlc(ISAC_main_inst,
+ decoded, prevFrameSize );
+ } else {
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ declen = WebRtcIsacfix_DecodePlcNb(ISAC_main_inst, decoded,
+ prevFrameSize );
+#else
+ declen = -1;
+#endif
+ }
+ lostPackets++;
+ } else {
+ if (nbTest !=2 ) {
+ short FL;
+ /* Call getFramelen, only used here for function test */
+ err = WebRtcIsacfix_ReadFrameLen((WebRtc_Word16*)streamdata, &FL);
+ declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
+ decoded, speechType );
+ /* Error check */
+ if (err<0 || declen<0 || FL!=declen) {
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in decode_B/or getFrameLen: %d.\n", errtype);
+ }
+ prevFrameSize = declen/480;
+
+ } else {
+#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
+ declen = WebRtcIsacfix_DecodeNb( ISAC_main_inst, streamdata,
+ stream_len, decoded, speechType );
+#else
+ declen = -1;
+#endif
+ prevFrameSize = declen/240;
+ }
+ }
+
+ if (declen <= 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
+ printf("\nError in decoder: %d.\n", errtype);
+ }
+
+ /* Write decoded speech frame to file */
+ fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
+ // fprintf( ratefile, "%f \n", stream_len / ( ((double)declen)/
+ // ((double)FS) ) * 8 );
+ } else {
+ lostPackets++;
+ }
+ framecnt++;
+
+ totalsmpls += declen;
+ totalbits += 8 * stream_len;
+ kbps = ((double) FS) / ((double) cur_framesmpls) * 8.0 *
+ stream_len / 1000.0;// kbits/s
+
+ /* Error test number 10, garbage data */
+ if (testNum == 10) {
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ printf( "Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "ok\n\n");
+ fclose(seedfile);
+ }
+ }
+
+#ifdef _DEBUG
+
+ fy = fopen("bit_rate.dat", "a");
+ fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
+ fclose(fy);
+
+#endif /* _DEBUG */
+
+ }
+ printf("\nLost Frames %d ~ %4.1f%%\n", lostPackets,
+ (double)lostPackets/(double)framecnt*100.0 );
+ printf("\n\ntotal bits = %d bits", totalbits);
+ printf("\nmeasured average bitrate = %0.3f kbits/s",
+ (double)totalbits *(FS/1000) / totalsmpls);
+ printf("\n");
+
+#ifdef _DEBUG
+ /* fprintf(stderr,"\n\ntotal bits = %d bits", totalbits);
+ fprintf(stderr,"\nmeasured average bitrate = %0.3f kbits/s",
+ (double)totalbits *(FS/1000) / totalsmpls);
+ fprintf(stderr,"\n");
+ */
+#endif /* _DEBUG */
+
+ /* Runtime statistics */
+
+
+ runtime = (double)(((double)clock()/(double)CLOCKS_PER_SEC)-starttime);
+ length_file = ((double)framecnt*(double)declen/FS);
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n",
+ runtime, (100*runtime/length_file));
+ printf("\n\n_______________________________________________\n");
+
+ fclose(inp);
+ fclose(outp);
+ fclose(outbits);
+
+ if ( testCE == 1) {
+ WebRtcIsacfix_FreeInternal(ISAC_main_inst);
+ }
+ WebRtcIsacfix_Free(ISAC_main_inst);
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c
new file mode 100644
index 0000000..57c30ca
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c
@@ -0,0 +1,693 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * test_iSACfixfloat.c
+ *
+ * Test compatibility and quality between floating- and fixed-point code
+ * */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "isac.h"
+#include "isacfix.h"
+
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms 160
+/* sampling frequency (Hz) */
+#define FS 16000
+
+
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC 1000
+
+
+
+// FILE *histfile, *ratefile;
+
+
+/* function for reading audio data from PCM file */
+int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
+
+ short k, rlen, status = 0;
+
+ rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
+ if (rlen < length) {
+ for (k = rlen; k < length; k++)
+ data[k] = 0;
+ status = 1;
+ }
+
+ return status;
+}
+
+typedef struct {
+ WebRtc_UWord32 send_time; /* samples */
+ WebRtc_UWord32 arrival_time; /* samples */
+ WebRtc_UWord32 sample_count; /* samples */
+ WebRtc_UWord16 rtp_number;
+} BottleNeckModel;
+
+void get_arrival_time(int current_framesamples, /* samples */
+ int packet_size, /* bytes */
+ int bottleneck, /* excluding headers; bits/s */
+ BottleNeckModel *BN_data)
+{
+ const int HeaderSize = 35;
+ int HeaderRate;
+
+ HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
+
+ /* everything in samples */
+ BN_data->sample_count = BN_data->sample_count + current_framesamples;
+
+ BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+ BN_data->send_time += current_framesamples;
+
+ if (BN_data->arrival_time < BN_data->sample_count)
+ BN_data->arrival_time = BN_data->sample_count;
+
+ BN_data->rtp_number++;
+}
+
+
+
+int main(int argc, char* argv[])
+{
+
+ char inname[50], outname[50], bottleneck_file[50], bitfilename[60], bitending[10]="_bits.pcm";
+ FILE *inp, *outp, *f_bn, *bitsp;
+ int framecnt, endfile;
+
+
+ int i,j,errtype, plc=0;
+ WebRtc_Word16 CodingMode;
+ WebRtc_Word16 bottleneck;
+
+ WebRtc_Word16 framesize = 30; /* ms */
+ //WebRtc_Word16 framesize = 60; /* To invoke cisco complexity case at frame 2252 */
+
+ int cur_framesmpls, err;
+
+ /* Runtime statistics */
+ double starttime;
+ double runtime;
+ double length_file;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 declen;
+
+ WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
+ WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
+ WebRtc_UWord16 streamdata[600];
+ WebRtc_Word16 speechType[1];
+
+// WebRtc_Word16 *iSACstruct;
+
+ char version_number[20];
+ int mode=-1, tmp, nbTest=0; /*,sss;*/
+
+#ifdef _DEBUG
+ FILE *fy;
+ double kbps;
+ int totalbits =0;
+ int totalsmpls =0;
+#endif /* _DEBUG */
+
+
+
+
+ /* only one structure used for ISAC encoder */
+ ISAC_MainStruct *ISAC_main_inst;
+ ISACFIX_MainStruct *ISACFIX_main_inst;
+
+ BottleNeckModel BN_data;
+ f_bn = NULL;
+
+#ifdef _DEBUG
+ fy = fopen("bit_rate.dat", "w");
+ fclose(fy);
+ fy = fopen("bytes_frames.dat", "w");
+ fclose(fy);
+#endif /* _DEBUG */
+
+
+//histfile = fopen("histo.dat", "ab");
+//ratefile = fopen("rates.dat", "ab");
+
+ /* handling wrong input arguments in the command line */
+ if ((argc<6) || (argc>10)) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ WebRtcIsacfix_version(version_number);
+ printf("iSAC version %s \n\n", version_number);
+
+ printf("Usage:\n\n");
+ printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
+ printf("with:\n");
+
+ printf("[-I] : if -I option is specified, the coder will use\n");
+ printf(" an instantaneous Bottleneck value. If not, it\n");
+ printf(" will be an adaptive Bottleneck value.\n\n");
+ printf("bottleneck_value : the value of the bottleneck provided either\n");
+ printf(" as a fixed value (e.g. 25000) or\n");
+ printf(" read from a file (e.g. bottleneck.txt)\n\n");
+ printf("[-m] mode : Mode (encoder - decoder):\n");
+ printf(" : 0 - float - float \n");
+ printf(" : 1 - float - fix \n");
+ printf(" : 2 - fix - float \n");
+ printf(" : 3 - fix - fix \n");
+ printf("[-PLC] : Test PLC packetlosses\n");
+ printf("[-NB] num : Test NB interfaces, num=1 encNB, num=2 decNB\n");
+ printf("infile : Normal speech input file\n\n");
+ printf("outfile : Speech output file\n\n");
+ printf("Example usage:\n\n");
+ printf("./kenny.exe -I bottleneck.txt -m 1 speechIn.pcm speechOut.pcm\n\n");
+ exit(0);
+
+ }
+
+
+ printf("--------------------START---------------------\n\n");
+ WebRtcIsac_version(version_number);
+ printf("iSAC FLOAT version %s \n", version_number);
+ WebRtcIsacfix_version(version_number);
+ printf("iSAC FIX version %s \n\n", version_number);
+
+ CodingMode = 0;
+ tmp=1;
+ for (i = 1; i < argc;i++)
+ {
+ if (!strcmp ("-I", argv[i]))
+ {
+ printf("\nInstantaneous BottleNeck\n");
+ CodingMode = 1;
+ i++;
+ tmp=0;
+ }
+
+ if (!strcmp ("-m", argv[i])) {
+ mode=atoi(argv[i+1]);
+ i++;
+ }
+
+ if (!strcmp ("-PLC", argv[i]))
+ {
+ plc=1;
+ }
+
+ if (!strcmp ("-NB", argv[i]))
+ {
+ nbTest = atoi(argv[i + 1]);
+ i++;
+ }
+
+ }
+
+ if(mode<0) {
+ printf("\nError! Mode must be set: -m 0 \n");
+ exit(0);
+ }
+
+ if (CodingMode == 0)
+ {
+ printf("\nAdaptive BottleNeck\n");
+ }
+
+
+
+ /* Get Bottleneck value */
+ bottleneck = atoi(argv[2-tmp]);
+ if (bottleneck == 0)
+ {
+ sscanf(argv[2-tmp], "%s", bottleneck_file);
+ f_bn = fopen(bottleneck_file, "rb");
+ if (f_bn == NULL)
+ {
+ printf("No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
+ exit(0);
+ }
+ else {
+ printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
+ if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ fscanf(f_bn, "%d", &bottleneck);
+ }
+
+ /* Bottleneck is a cosine function
+ * Matlab code for writing the bottleneck file:
+ * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
+ * fid = fopen('bottleneck.txt', 'wb');
+ * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
+ */
+ }
+ }
+ else
+ {
+ printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+ }
+
+
+
+ /* Get Input and Output files */
+ sscanf(argv[argc-2], "%s", inname);
+ sscanf(argv[argc-1], "%s", outname);
+
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" iSAC: Cannot read file %s.\n", inname);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" iSAC: Cannot write file %s.\n", outname);
+ exit(1);
+ }
+ printf("\nInput:%s\nOutput:%s\n", inname, outname);
+
+ i=0;
+ while (outname[i]!='\0') {
+ bitfilename[i]=outname[i];
+ i++;
+ }
+ i-=4;
+ for (j=0;j<9;j++, i++)
+ bitfilename[i]=bitending[j];
+ bitfilename[i]='\0';
+ if ((bitsp = fopen(bitfilename,"wb")) == NULL) {
+ printf(" iSAC: Cannot read file %s.\n", bitfilename);
+ exit(1);
+ }
+ printf("Bitstream:%s\n\n", bitfilename);
+
+
+
+ starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+
+
+ /* Initialize the ISAC and BN structs */
+ WebRtcIsac_create(&ISAC_main_inst);
+/* WebRtcIsacfix_AssignSize(&sss);
+ iSACstruct=malloc(sss);
+ WebRtcIsacfix_Assign(&ISACFIX_main_inst,iSACstruct);*/
+ WebRtcIsacfix_Create(&ISACFIX_main_inst);
+
+ BN_data.send_time = 0;
+ BN_data.arrival_time = 0;
+ BN_data.sample_count = 0;
+ BN_data.rtp_number = 0;
+
+ /* Initialize encoder and decoder */
+ framecnt= 0;
+ endfile = 0;
+
+ if (mode==0) { /* Encode using FLOAT, decode using FLOAT */
+
+ printf("Coding mode: Encode using FLOAT, decode using FLOAT \n\n");
+
+ /* Init iSAC FLOAT */
+ WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+ WebRtcIsac_DecoderInit(ISAC_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ // exit(EXIT_FAILURE);
+ }
+ }
+
+ } else if (mode==1) { /* Encode using FLOAT, decode using FIX */
+
+ printf("Coding mode: Encode using FLOAT, decode using FIX \n\n");
+
+ /* Init iSAC FLOAT */
+ WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+ WebRtcIsac_DecoderInit(ISAC_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ // exit(EXIT_FAILURE);
+ }
+ }
+
+ /* Init iSAC FIX */
+ WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+ WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ }
+ } else if (mode==2) { /* Encode using FIX, decode using FLOAT */
+
+ printf("Coding mode: Encode using FIX, decode using FLOAT \n\n");
+
+ /* Init iSAC FLOAT */
+ WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+ WebRtcIsac_DecoderInit(ISAC_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ }
+
+ /* Init iSAC FIX */
+ WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+ WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ }
+ } else if (mode==3) {
+
+ printf("Coding mode: Encode using FIX, decode using FIX \n\n");
+
+ WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+ WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+ if (CodingMode == 1) {
+ err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\n Error in initialization: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ }
+
+ } else
+ printf("Mode must be value between 0 and 3\n");
+ *speechType = 1;
+
+//#define BI_TEST 1
+#ifdef BI_TEST
+ err = WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_main_inst, 300);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\n Error in setMaxPayloadSize: %d.\n\n", errtype);
+ fclose(inp);
+ fclose(outp);
+ fclose(bitsp);
+ return(EXIT_FAILURE);
+ }
+#endif
+
+
+ while (endfile == 0) {
+
+ cur_framesmpls = 0;
+ while (1) {
+ /* Read 10 ms speech block */
+ if (nbTest != 1)
+ endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
+ else
+ endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
+
+ /* iSAC encoding */
+
+ if (mode==0 || mode ==1) {
+ stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, streamdata);
+ if (stream_len < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in encoder: %d.\n\n", errtype);
+ // exit(EXIT_FAILURE);
+ }
+ } else if (mode==2 || mode==3) {
+ /* iSAC encoding */
+ if (nbTest != 1)
+ stream_len = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata, streamdata);
+ else
+ stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
+
+ if (stream_len < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\nError in encoder: %d.\n\n", errtype);
+ // exit(EXIT_FAILURE);
+ }
+ }
+
+ cur_framesmpls += FRAMESAMPLES_10ms;
+
+ /* read next bottleneck rate */
+ if (f_bn != NULL) {
+ if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ fscanf(f_bn, "%d", &bottleneck);
+ }
+ if (CodingMode == 1) {
+ if (mode==0 || mode==1)
+ WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ else if (mode==2 || mode==3)
+ WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+ }
+ }
+
+ /* exit encoder loop if the encoder returned a bitstream */
+ if (stream_len != 0) break;
+ }
+
+ fwrite(streamdata, 1, stream_len, bitsp); /* NOTE! Writes bytes to file */
+
+ /* simulate packet handling through NetEq and the modem */
+ get_arrival_time(cur_framesmpls, stream_len, bottleneck,
+ &BN_data);
+//*****************************
+ if (1){
+ if (mode==0) {
+ err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ /* iSAC decoding */
+ declen = WebRtcIsac_Decode(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ if (declen <= 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ } else if (mode==1) {
+
+ err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+ err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+
+ declen = WebRtcIsac_Decode(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+
+ /* iSAC decoding */
+ if (plc && (framecnt+1)%10 == 0) {
+ if (nbTest !=2 )
+ declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
+ else
+ declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
+ } else {
+ if (nbTest !=2 )
+ declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ else
+ declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ }
+
+ if (declen <= 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ } else if (mode==2) {
+ err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+
+ err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.arrival_time);
+
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ /* iSAC decoding */
+ declen = WebRtcIsac_Decode(ISAC_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ if (declen <= 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ } else if (mode==3) {
+ err = WebRtcIsacfix_UpdateBwEstimate(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ BN_data.rtp_number,
+ BN_data.send_time,
+ BN_data.arrival_time);
+
+ if (err < 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ /* iSAC decoding */
+
+ if (plc && (framecnt+1)%10 == 0) {
+ if (nbTest !=2 )
+ declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
+ else
+ declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
+ } else {
+ if (nbTest !=2 )
+ declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ else
+ declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
+ streamdata,
+ stream_len,
+ decoded,
+ speechType);
+ }
+ if (declen <= 0) {
+ /* exit if returned with error */
+ errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+ printf("\n\nError in decoder: %d.\n\n", errtype);
+ //exit(EXIT_FAILURE);
+ }
+ }
+
+ /* Write decoded speech frame to file */
+ fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
+ }
+
+ fprintf(stderr," \rframe = %d", framecnt);
+ framecnt++;
+
+
+
+#ifdef _DEBUG
+
+ totalsmpls += declen;
+ totalbits += 8 * stream_len;
+ kbps = ((double) FS) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
+ fy = fopen("bit_rate.dat", "a");
+ fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
+ fclose(fy);
+
+#endif /* _DEBUG */
+
+ }
+
+#ifdef _DEBUG
+ printf("\n\ntotal bits = %d bits", totalbits);
+ printf("\nmeasured average bitrate = %0.3f kbits/s", (double)totalbits *(FS/1000) / totalsmpls);
+ printf("\n");
+#endif /* _DEBUG */
+
+ /* Runtime statistics */
+ runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
+ length_file = ((double)framecnt*(double)declen/FS);
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+ printf("---------------------END----------------------\n");
+
+ fclose(inp);
+ fclose(outp);
+
+ WebRtcIsac_Free(ISAC_main_inst);
+ WebRtcIsacfix_Free(ISACFIX_main_inst);
+
+
+
+// fclose(histfile);
+// fclose(ratefile);
+
+ return 0;
+
+}
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi
new file mode 100644
index 0000000..abe2454
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi
@@ -0,0 +1,71 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ # simple kenny
+ {
+ 'target_name': 'iSACtest',
+ 'type': 'executable',
+ 'dependencies': [
+ 'iSAC',
+ ],
+ 'include_dirs': [
+ './main/test',
+ './main/interface',
+ './main/util',
+ ],
+ 'sources': [
+ './main/test/simpleKenny.c',
+ './main/util/utility.c',
+ ],
+ },
+ # ReleaseTest-API
+ {
+ 'target_name': 'iSACAPITest',
+ 'type': 'executable',
+ 'dependencies': [
+ 'iSAC',
+ ],
+ 'include_dirs': [
+ './main/test',
+ './main/interface',
+ './main/util',
+ ],
+ 'sources': [
+ './main/test/ReleaseTest-API/ReleaseTest-API.cc',
+ './main/util/utility.c',
+ ],
+ },
+ # SwitchingSampRate
+ {
+ 'target_name': 'iSACSwitchSampRateTest',
+ 'type': 'executable',
+ 'dependencies': [
+ 'iSAC',
+ ],
+ 'include_dirs': [
+ './main/test',
+ './main/interface',
+ '../../../../common_audio/signal_processing/include',
+ './main/util',
+ ],
+ 'sources': [
+ './main/test/SwitchingSampRate/SwitchingSampRate.cc',
+ './main/util/utility.c',
+ ],
+ },
+
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi
new file mode 100644
index 0000000..627342f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi
@@ -0,0 +1,33 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ # kenny
+ {
+ 'target_name': 'iSACFixtest',
+ 'type': 'executable',
+ 'dependencies': [
+ 'iSACFix',
+ ],
+ 'include_dirs': [
+ './fix/test',
+ './fix/interface',
+ ],
+ 'sources': [
+ './fix/test/kenny.c',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h
new file mode 100644
index 0000000..03c260b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h
@@ -0,0 +1,729 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
+
+/*
+ * Define the fixed-point numeric formats
+ */
+#include "typedefs.h"
+
+typedef struct WebRtcISACStruct ISACStruct;
+
+enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
+
+
+#if defined(__cplusplus)
+extern "C" {
+#endif
+
+ /******************************************************************************
+ * WebRtcIsac_AssignSize(...)
+ *
+ * This function returns the size of the ISAC instance, so that the instance
+ * can be created outside iSAC.
+ *
+ * Input:
+ * - samplingRate : sampling rate of the input/output audio.
+ *
+ * Output:
+ * - sizeinbytes : number of bytes needed to allocate for the
+ * instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_AssignSize(
+ int* sizeinbytes);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Assign(...)
+ *
+ * This function assignes the memory already created to the ISAC instance.
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ * - samplingRate : sampling rate of the input/output audio.
+ * - ISAC_inst_Addr : the already allocated memory, where we put the
+ * iSAC structure.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_Assign(
+ ISACStruct** ISAC_main_inst,
+ void* ISAC_inst_Addr);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Create(...)
+ *
+ * This function creates an ISAC instance, which will contain the state
+ * information for one coding/decoding channel.
+ *
+ * Input:
+ * - *ISAC_main_inst : a pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_Create(
+ ISACStruct** ISAC_main_inst);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Free(...)
+ *
+ * This function frees the ISAC instance created at the beginning.
+ *
+ * Input:
+ * - ISAC_main_inst : an ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_Free(
+ ISACStruct* ISAC_main_inst);
+
+
+ /******************************************************************************
+ * WebRtcIsac_EncoderInit(...)
+ *
+ * This function initializes an ISAC instance prior to the encoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - CodingMode : 0 -> Bit rate and frame length are
+ * automatically adjusted to available bandwidth
+ * on transmission channel, just valid if codec
+ * is created to work in wideband mode.
+ * 1 -> User sets a frame length and a target bit
+ * rate which is taken as the maximum
+ * short-term average bit rate.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_EncoderInit(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 CodingMode);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Encode(...)
+ *
+ * This function encodes 10ms audio blocks and inserts it into a package.
+ * Input speech length has 160 samples if operating at 16 kHz sampling
+ * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
+ * input audio until the whole frame is buffered then proceeds with encoding.
+ *
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : 0 - The buffer didn't reach the chosen
+ * frame-size so it keeps buffering speech
+ * samples.
+ * : -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_Encode(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_Word16* speechIn,
+ WebRtc_Word16* encoded);
+
+
+ /******************************************************************************
+ * WebRtcIsac_DecoderInit(...)
+ *
+ * This function initializes an ISAC instance prior to the decoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ *
+ * Return value
+ * : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_DecoderInit(
+ ISACStruct* ISAC_main_inst);
+
+
+ /******************************************************************************
+ * WebRtcIsac_UpdateBwEstimate(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - send_ts : the RTP send timestamp, given in samples
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Decode(...)
+ *
+ * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
+ * of the output audio could be either 480 or 960 samples, equivalent to
+ * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
+ * output audio is 960 samples, which is 30 ms.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - len : bytes in encoded vector.
+ *
+ * Output:
+ * - decoded : The decoded vector.
+ *
+ * Return value : >0 - number of samples in decoded vector.
+ * -1 - Error.
+ */
+
+ WebRtc_Word16 WebRtcIsac_Decode(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType);
+
+
+ /******************************************************************************
+ * WebRtcIsac_DecodePlc(...)
+ *
+ * This function conducts PLC for ISAC frame(s). Output speech length
+ * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
+ * the output is multiple of 480 samples if operating at 16 kHz and multiple
+ * of 960 if operating at 32 kHz.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames to produce.
+ *
+ * Output:
+ * - decoded : The decoded vector.
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_DecodePlc(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16 noOfLostFrames);
+
+
+ /******************************************************************************
+ * WebRtcIsac_Control(...)
+ *
+ * This function sets the limit on the short-term average bit-rate and the
+ * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
+ * rate, an average bit-rate between 10000 to 32000 bps is valid and a
+ * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
+ * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rate : limit on the short-term average bit rate,
+ * in bits/second.
+ * - framesize : frame-size in millisecond.
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_Control(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 rate,
+ WebRtc_Word16 framesize);
+
+
+ /******************************************************************************
+ * WebRtcIsac_ControlBwe(...)
+ *
+ * This function sets the initial values of bottleneck and frame-size if
+ * iSAC is used in channel-adaptive mode. Therefore, this API is not
+ * applicable if the codec is created to operate in super-wideband mode.
+ *
+ * Through this API, users can enforce a frame-size for all values of
+ * bottleneck. Then iSAC will not automatically change the frame-size.
+ *
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rateBPS : initial value of bottleneck in bits/second
+ * 10000 <= rateBPS <= 56000 is accepted
+ * For default bottleneck set rateBPS = 0
+ * - frameSizeMs : number of milliseconds per frame (30 or 60)
+ * - enforceFrameSize : 1 to enforce the given frame-size through
+ * out the adaptation process, 0 to let iSAC
+ * change the frame-size if required.
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIsac_ControlBwe(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 rateBPS,
+ WebRtc_Word16 frameSizeMs,
+ WebRtc_Word16 enforceFrameSize);
+
+
+ /******************************************************************************
+ * WebRtcIsac_ReadFrameLen(...)
+ *
+ * This function returns the length of the frame represented in the packet.
+ *
+ * Input:
+ * - encoded : Encoded bit-stream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsac_ReadFrameLen(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_Word16* encoded,
+ WebRtc_Word16* frameLength);
+
+
+ /******************************************************************************
+ * WebRtcIsac_version(...)
+ *
+ * This function returns the version number.
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ */
+
+ void WebRtcIsac_version(
+ char *version);
+
+
+ /******************************************************************************
+ * WebRtcIsac_GetErrorCode(...)
+ *
+ * This function can be used to check the error code of an iSAC instance. When
+ * a function returns -1 a error code will be set for that instance. The
+ * function below extract the code of the last error that occurred in the
+ * specified instance.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance
+ *
+ * Return value : Error code
+ */
+
+ WebRtc_Word16 WebRtcIsac_GetErrorCode(
+ ISACStruct* ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsac_GetUplinkBw(...)
+ *
+ * This function outputs the target bottleneck of the codec. In
+ * channel-adaptive mode, the target bottleneck is specified through in-band
+ * signalling retreived by bandwidth estimator.
+ * In channel-independent, also called instantaneous mode, the target
+ * bottleneck is provided to the encoder by calling xxx_control(...). If
+ * xxx_control is never called the default values is returned. The default
+ * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec,
+ * and it is 56000 bits/sec for 32 kHz sampling rate.
+ * Note that the output is the iSAC internal operating bottleneck which might
+ * differ slightly from the one provided through xxx_control().
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Output:
+ * - *bottleneck : bottleneck in bits/sec
+ *
+ * Return value : -1 if error happens
+ * 0 bit-rates computed correctly.
+ */
+
+ WebRtc_Word16 WebRtcIsac_GetUplinkBw(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32* bottleneck);
+
+
+ /******************************************************************************
+ * WebRtcIsac_SetMaxPayloadSize(...)
+ *
+ * This function sets a limit for the maximum payload size of iSAC. The same
+ * value is used both for 30 and 60 ms packets. If the encoder sampling rate
+ * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
+ * encoder sampling rate is 32 kHz the maximum payload size is between 120
+ * and 600 bytes.
+ *
+ * If an out of range limit is used, the function returns -1, but the closest
+ * valid value will be applied.
+ *
+ * ---------------
+ * IMPORTANT NOTES
+ * ---------------
+ * The size of a packet is limited to the minimum of 'max-payload-size' and
+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
+ * 170 bytes, i.e. min(170, 300).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxPayloadBytes : maximum size of the payload in bytes
+ * valid values are between 120 and 400 bytes
+ * if encoder sampling rate is 16 kHz. For
+ * 32 kHz encoder sampling rate valid values
+ * are between 120 and 600 bytes.
+ *
+ * Return value : 0 if successful
+ * -1 if error happens
+ */
+
+ WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 maxPayloadBytes);
+
+
+ /******************************************************************************
+ * WebRtcIsac_SetMaxRate(...)
+ *
+ * This function sets the maximum rate which the codec may not exceed for
+ * any signal packet. The maximum rate is defined and payload-size per
+ * frame-size in bits per second.
+ *
+ * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
+ * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
+ * if the encoder sampling rate is 32 kHz.
+ *
+ * It is possible to set a maximum rate between 32000 and 53400 bits/sec
+ * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
+ *
+ * If an out of range limit is used, the function returns -1, but the closest
+ * valid value will be applied.
+ *
+ * ---------------
+ * IMPORTANT NOTES
+ * ---------------
+ * The size of a packet is limited to the minimum of 'max-payload-size' and
+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
+ * 170 bytes, min(170, 300).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxRate : maximum rate in bits per second,
+ * valid values are 32000 to 53400 bits/sec in
+ * wideband mode, and 32000 to 160000 bits/sec in
+ * super-wideband mode.
+ *
+ * Return value : 0 if successful
+ * -1 if error happens
+ */
+
+ WebRtc_Word16 WebRtcIsac_SetMaxRate(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 maxRate);
+
+
+ /******************************************************************************
+ * WebRtcIsac_DecSampRate()
+ * Return the sampling rate of the decoded audio.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : enumerator representing sampling frequency
+ * associated with the decoder, i.e. the
+ * sampling rate of the decoded audio.
+ *
+ */
+
+ enum IsacSamplingRate WebRtcIsac_DecSampRate(
+ ISACStruct* ISAC_main_inst);
+
+
+ /******************************************************************************
+ * WebRtcIsac_EncSampRate()
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : enumerator representing sampling frequency
+ * associated with the encoder, the input audio
+ * is expected to be sampled at this rate.
+ *
+ */
+
+ enum IsacSamplingRate WebRtcIsac_EncSampRate(
+ ISACStruct* ISAC_main_inst);
+
+
+ /******************************************************************************
+ * WebRtcIsac_SetDecSampRate()
+ * Set the sampling rate of the decoder. Initialization of the decoder WILL
+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
+ * which is set when the instance is created.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - sampRate : enumerator specifying the sampling rate.
+ *
+ * Return value : 0 if successful
+ * -1 if failed.
+ */
+
+ WebRtc_Word16 WebRtcIsac_SetDecSampRate(
+ ISACStruct* ISAC_main_inst,
+ enum IsacSamplingRate sampRate);
+
+
+ /******************************************************************************
+ * WebRtcIsac_SetEncSampRate()
+ * Set the sampling rate of the encoder. Initialization of the encoder WILL
+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
+ * which is set when the instance is created. The encoding-mode and the
+ * bottleneck remain unchanged by this call, however, the maximum rate and
+ * maximum payload-size will reset to their default value.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - sampRate : enumerator specifying the sampling rate.
+ *
+ * Return value : 0 if successful
+ * -1 if failed.
+ */
+
+ WebRtc_Word16 WebRtcIsac_SetEncSampRate(
+ ISACStruct* ISAC_main_inst,
+ enum IsacSamplingRate sampRate);
+
+
+
+ /******************************************************************************
+ * WebRtcIsac_GetNewBitStream(...)
+ *
+ * This function returns encoded data, with the recieved bwe-index in the
+ * stream. If the rate is set to a value less than bottleneck of codec
+ * the new bistream will be re-encoded with the given target rate.
+ * It should always return a complete packet, i.e. only called once
+ * even for 60 msec frames.
+ *
+ * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
+ * NOTE 2! Currently not implemented for SWB mode.
+ * NOTE 3! Rates larger than the bottleneck of the codec will be limited
+ * to the current bottleneck.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - bweIndex : Index of bandwidth estimate to put in new
+ * bitstream
+ * - rate : target rate of the transcoder is bits/sec.
+ * Valid values are the accepted rate in iSAC,
+ * i.e. 10000 to 56000.
+ * - isRCU : if the new bit-stream is an RCU stream.
+ * Note that the rate parameter always indicates
+ * the target rate of the main paylaod, regardless
+ * of 'isRCU' value.
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error or called in SWB mode
+ * NOTE! No error code is written to
+ * the struct since it is only allowed to read
+ * the struct.
+ */
+ WebRtc_Word16 WebRtcIsac_GetNewBitStream(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 bweIndex,
+ WebRtc_Word16 jitterInfo,
+ WebRtc_Word32 rate,
+ WebRtc_Word16* encoded,
+ WebRtc_Word16 isRCU);
+
+
+
+ /****************************************************************************
+ * WebRtcIsac_GetDownLinkBwIndex(...)
+ *
+ * This function returns index representing the Bandwidth estimate from
+ * other side to this side.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Output:
+ * - bweIndex : Bandwidth estimate to transmit to other side.
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* bweIndex,
+ WebRtc_Word16* jitterInfo);
+
+
+ /****************************************************************************
+ * WebRtcIsac_UpdateUplinkBw(...)
+ *
+ * This function takes an index representing the Bandwidth estimate from
+ * this side to other side and updates BWE.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ * - bweIndex : Bandwidth estimate from other side.
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 bweIndex);
+
+
+ /****************************************************************************
+ * WebRtcIsac_ReadBwIndex(...)
+ *
+ * This function returns the index of the Bandwidth estimate from the bitstream.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ * - bweIndex : Bandwidth estimate in bitstream
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsac_ReadBwIndex(
+ const WebRtc_Word16* encoded,
+ WebRtc_Word16* bweIndex);
+
+
+
+ /*******************************************************************************
+ * WebRtcIsac_GetNewFrameLen(...)
+ *
+ * returns the frame lenght (in samples) of the next packet. In the case of channel-adaptive
+ * mode, iSAC decides on its frame lenght based on the estimated bottleneck
+ * this allows a user to prepare for the next packet (at the encoder)
+ *
+ * The primary usage is in CE to make the iSAC works in channel-adaptive mode
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Return Value : frame lenght in samples
+ *
+ */
+
+ WebRtc_Word16 WebRtcIsac_GetNewFrameLen(
+ ISACStruct* ISAC_main_inst);
+
+
+ /****************************************************************************
+ * WebRtcIsac_GetRedPayload(...)
+ *
+ * Populates "encoded" with the redundant payload of the recently encoded
+ * frame. This function has to be called once that WebRtcIsac_Encode(...)
+ * returns a positive value. Regardless of the frame-size this function will
+ * be called only once after encoding is completed.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : -1 - Error
+ *
+ *
+ */
+ WebRtc_Word16 WebRtcIsac_GetRedPayload(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* encoded);
+
+
+ /****************************************************************************
+ * WebRtcIsac_DecodeRcu(...)
+ *
+ * This function decodes a redundant (RCU) iSAC frame. Function is called in
+ * NetEq with a stored RCU payload i case of packet loss. Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC RCU frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+ WebRtc_Word16 WebRtcIsac_DecodeRcu(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType);
+
+
+#if defined(__cplusplus)
+}
+#endif
+
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
new file mode 100644
index 0000000..07b2a31
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
@@ -0,0 +1,66 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_isac
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ arith_routines.c \
+ arith_routines_hist.c \
+ arith_routines_logist.c \
+ bandwidth_estimator.c \
+ crc.c \
+ decode.c \
+ decode_bwe.c \
+ encode.c \
+ encode_lpc_swb.c \
+ entropy_coding.c \
+ fft.c \
+ filter_functions.c \
+ filterbank_tables.c \
+ intialize.c \
+ isac.c \
+ filterbanks.c \
+ pitch_lag_tables.c \
+ lattice.c \
+ lpc_gain_swb_tables.c \
+ lpc_analysis.c \
+ lpc_shape_swb12_tables.c \
+ lpc_shape_swb16_tables.c \
+ lpc_tables.c \
+ pitch_estimator.c \
+ pitch_filter.c \
+ pitch_gain_tables.c \
+ spectrum_ar_model_tables.c \
+ transform.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c
new file mode 100644
index 0000000..31c441a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "arith_routines.h"
+#include "settings.h"
+
+
+/*
+ * terminate and return byte stream;
+ * returns the number of bytes in the stream
+ */
+int WebRtcIsac_EncTerminate(Bitstr *streamdata) /* in-/output struct containing bitstream */
+{
+ WebRtc_UWord8 *stream_ptr;
+
+
+ /* point to the right place in the stream buffer */
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+
+ /* find minimum length (determined by current interval width) */
+ if ( streamdata->W_upper > 0x01FFFFFF )
+ {
+ streamdata->streamval += 0x01000000;
+ /* add carry to buffer */
+ if (streamdata->streamval < 0x01000000)
+ {
+ /* propagate carry */
+ while ( !(++(*--stream_ptr)) );
+ /* put pointer back to the old value */
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ }
+ /* write remaining data to bitstream */
+ *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+ }
+ else
+ {
+ streamdata->streamval += 0x00010000;
+ /* add carry to buffer */
+ if (streamdata->streamval < 0x00010000)
+ {
+ /* propagate carry */
+ while ( !(++(*--stream_ptr)) );
+ /* put pointer back to the old value */
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ }
+ /* write remaining data to bitstream */
+ *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+ *stream_ptr++ = (WebRtc_UWord8) ((streamdata->streamval >> 16) & 0x00FF);
+ }
+
+ /* calculate stream length */
+ return (int)(stream_ptr - streamdata->stream);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h
new file mode 100644
index 0000000..8e5f496
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routines.h
+ *
+ * Functions for arithmetic coding.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
+
+#include "structs.h"
+
+
+int WebRtcIsac_EncLogisticMulti2(
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ WebRtc_Word16 *dataQ7, /* input: data vector */
+ const WebRtc_UWord16 *env, /* input: side info vector defining the width of the pdf */
+ const int N, /* input: data vector length */
+ const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
+
+/* returns the number of bytes in the stream */
+int WebRtcIsac_EncTerminate(Bitstr *streamdata); /* in-/output struct containing bitstream */
+
+/* returns the number of bytes in the stream so far */
+int WebRtcIsac_DecLogisticMulti2(
+ WebRtc_Word16 *data, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 *env, /* input: side info vector defining the width of the pdf */
+ const WebRtc_Word16 *dither, /* input: dither vector */
+ const int N, /* input: data vector length */
+ const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
+
+void WebRtcIsac_EncHistMulti(
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const int *data, /* input: data vector */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const int N); /* input: data vector length */
+
+int WebRtcIsac_DecHistBisectMulti(
+ int *data, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
+ const int N); /* input: data vector length */
+
+int WebRtcIsac_DecHistOneStepMulti(
+ int *data, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const WebRtc_UWord16 *init_index,/* input: vector of initial cdf table search entries */
+ const int N); /* input: data vector length */
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c
new file mode 100644
index 0000000..f4a13d6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c
@@ -0,0 +1,291 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "settings.h"
+#include "arith_routines.h"
+
+
+/*
+ * code symbols into arithmetic bytestream
+ */
+void WebRtcIsac_EncHistMulti(Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const int *data, /* input: data vector */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const int N) /* input: data vector length */
+{
+ WebRtc_UWord32 W_lower, W_upper;
+ WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
+ WebRtc_UWord8 *stream_ptr;
+ WebRtc_UWord8 *stream_ptr_carry;
+ WebRtc_UWord32 cdf_lo, cdf_hi;
+ int k;
+
+
+ /* point to beginning of stream buffer */
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ W_upper = streamdata->W_upper;
+
+ for (k=N; k>0; k--)
+ {
+ /* fetch cdf_lower and cdf_upper from cdf tables */
+ cdf_lo = (WebRtc_UWord32) *(*cdf + *data);
+ cdf_hi = (WebRtc_UWord32) *(*cdf++ + *data++ + 1);
+
+ /* update interval */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = W_upper >> 16;
+ W_lower = W_upper_MSB * cdf_lo;
+ W_lower += (W_upper_LSB * cdf_lo) >> 16;
+ W_upper = W_upper_MSB * cdf_hi;
+ W_upper += (W_upper_LSB * cdf_hi) >> 16;
+
+ /* shift interval such that it begins at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamdata->streamval += W_lower;
+
+ /* handle carry */
+ if (streamdata->streamval < W_lower)
+ {
+ /* propagate carry */
+ stream_ptr_carry = stream_ptr;
+ while (!(++(*--stream_ptr_carry)));
+ }
+
+ /* renormalize interval, store most significant byte of streamval and update streamval */
+ while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
+ {
+ W_upper <<= 8;
+ *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+ streamdata->streamval <<= 8;
+ }
+ }
+
+ /* calculate new stream_index */
+ streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
+ streamdata->W_upper = W_upper;
+
+ return;
+}
+
+
+
+/*
+ * function to decode more symbols from the arithmetic bytestream, using method of bisection
+ * cdf tables should be of size 2^k-1 (which corresponds to an alphabet size of 2^k-2)
+ */
+int WebRtcIsac_DecHistBisectMulti(int *data, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
+ const int N) /* input: data vector length */
+{
+ WebRtc_UWord32 W_lower, W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
+ WebRtc_UWord32 streamval;
+ const WebRtc_UWord8 *stream_ptr;
+ const WebRtc_UWord16 *cdf_ptr;
+ int size_tmp;
+ int k;
+
+ W_lower = 0; //to remove warning -DH
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ W_upper = streamdata->W_upper;
+ if (W_upper == 0)
+ /* Should not be possible in normal operation */
+ return -2;
+
+ if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
+ {
+ /* read first word from bytestream */
+ streamval = *stream_ptr << 24;
+ streamval |= *++stream_ptr << 16;
+ streamval |= *++stream_ptr << 8;
+ streamval |= *++stream_ptr;
+ } else {
+ streamval = streamdata->streamval;
+ }
+
+ for (k=N; k>0; k--)
+ {
+ /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = W_upper >> 16;
+
+ /* start halfway the cdf range */
+ size_tmp = *cdf_size++ >> 1;
+ cdf_ptr = *cdf + (size_tmp - 1);
+
+ /* method of bisection */
+ for ( ;; )
+ {
+ W_tmp = W_upper_MSB * *cdf_ptr;
+ W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
+ size_tmp >>= 1;
+ if (size_tmp == 0) break;
+ if (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ cdf_ptr += size_tmp;
+ } else {
+ W_upper = W_tmp;
+ cdf_ptr -= size_tmp;
+ }
+ }
+ if (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ *data++ = (int)(cdf_ptr - *cdf++);
+ } else {
+ W_upper = W_tmp;
+ *data++ = (int)(cdf_ptr - *cdf++ - 1);
+ }
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamval -= W_lower;
+
+ /* renormalize interval and update streamval */
+ while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
+ {
+ /* read next byte from stream */
+ streamval = (streamval << 8) | *++stream_ptr;
+ W_upper <<= 8;
+ }
+
+ if (W_upper == 0)
+ /* Should not be possible in normal operation */
+ return -2;
+
+
+ }
+
+ streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
+ streamdata->W_upper = W_upper;
+ streamdata->streamval = streamval;
+
+
+ /* find number of bytes in original stream (determined by current interval width) */
+ if ( W_upper > 0x01FFFFFF )
+ return streamdata->stream_index - 2;
+ else
+ return streamdata->stream_index - 1;
+}
+
+
+
+/*
+ * function to decode more symbols from the arithmetic bytestream, taking single step up or
+ * down at a time
+ * cdf tables can be of arbitrary size, but large tables may take a lot of iterations
+ */
+int WebRtcIsac_DecHistOneStepMulti(int *data, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
+ const WebRtc_UWord16 *init_index, /* input: vector of initial cdf table search entries */
+ const int N) /* input: data vector length */
+{
+ WebRtc_UWord32 W_lower, W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
+ WebRtc_UWord32 streamval;
+ const WebRtc_UWord8 *stream_ptr;
+ const WebRtc_UWord16 *cdf_ptr;
+ int k;
+
+
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ W_upper = streamdata->W_upper;
+ if (W_upper == 0)
+ /* Should not be possible in normal operation */
+ return -2;
+
+ if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
+ {
+ /* read first word from bytestream */
+ streamval = *stream_ptr << 24;
+ streamval |= *++stream_ptr << 16;
+ streamval |= *++stream_ptr << 8;
+ streamval |= *++stream_ptr;
+ } else {
+ streamval = streamdata->streamval;
+ }
+
+
+ for (k=N; k>0; k--)
+ {
+ /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = W_upper >> 16;
+
+ /* start at the specified table entry */
+ cdf_ptr = *cdf + (*init_index++);
+ W_tmp = W_upper_MSB * *cdf_ptr;
+ W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
+ if (streamval > W_tmp)
+ {
+ for ( ;; )
+ {
+ W_lower = W_tmp;
+ if (cdf_ptr[0]==65535)
+ /* range check */
+ return -3;
+ W_tmp = W_upper_MSB * *++cdf_ptr;
+ W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
+ if (streamval <= W_tmp) break;
+ }
+ W_upper = W_tmp;
+ *data++ = (int)(cdf_ptr - *cdf++ - 1);
+ } else {
+ for ( ;; )
+ {
+ W_upper = W_tmp;
+ --cdf_ptr;
+ if (cdf_ptr<*cdf) {
+ /* range check */
+ return -3;
+ }
+ W_tmp = W_upper_MSB * *cdf_ptr;
+ W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
+ if (streamval > W_tmp) break;
+ }
+ W_lower = W_tmp;
+ *data++ = (int)(cdf_ptr - *cdf++);
+ }
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+ /* add integer to bitstream */
+ streamval -= W_lower;
+
+ /* renormalize interval and update streamval */
+ while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
+ {
+ /* read next byte from stream */
+ streamval = (streamval << 8) | *++stream_ptr;
+ W_upper <<= 8;
+ }
+ }
+
+ streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
+ streamdata->W_upper = W_upper;
+ streamdata->streamval = streamval;
+
+
+ /* find number of bytes in original stream (determined by current interval width) */
+ if ( W_upper > 0x01FFFFFF )
+ return streamdata->stream_index - 2;
+ else
+ return streamdata->stream_index - 1;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c
new file mode 100644
index 0000000..422855a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c
@@ -0,0 +1,294 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * arith_routines.h
+ *
+ * This file contains functions for arithmatically encoding and
+ * decoding DFT coefficients.
+ *
+ */
+
+
+#include "arith_routines.h"
+
+
+
+static const WebRtc_Word32 kHistEdgesQ15[51] = {
+ -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
+ -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644,
+ -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428,
+ 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500,
+ 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572,
+ 327680};
+
+
+static const int kCdfSlopeQ0[51] = { /* Q0 */
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088,
+ 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312,
+ 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 2, 0};
+
+
+static const int kCdfQ16[51] = { /* Q16 */
+ 0, 2, 4, 6, 8, 10, 12, 14, 16, 18,
+ 20, 22, 24, 29, 38, 57, 92, 153, 279, 559,
+ 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636,
+ 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514,
+ 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534,
+ 65535};
+
+
+
+/* function to be converted to fixed point */
+static __inline WebRtc_UWord32 piecewise(WebRtc_Word32 xinQ15) {
+
+ WebRtc_Word32 ind, qtmp1, qtmp2, qtmp3;
+ WebRtc_UWord32 tmpUW32;
+
+
+ qtmp2 = xinQ15;
+
+ if (qtmp2 < kHistEdgesQ15[0]) {
+ qtmp2 = kHistEdgesQ15[0];
+ }
+ if (qtmp2 > kHistEdgesQ15[50]) {
+ qtmp2 = kHistEdgesQ15[50];
+ }
+
+ qtmp1 = qtmp2 - kHistEdgesQ15[0]; /* Q15 - Q15 = Q15 */
+ ind = (qtmp1 * 5) >> 16; /* 2^16 / 5 = 0.4 in Q15 */
+ /* Q15 -> Q0 */
+ qtmp1 = qtmp2 - kHistEdgesQ15[ind]; /* Q15 - Q15 = Q15 */
+ qtmp2 = kCdfSlopeQ0[ind] * qtmp1; /* Q0 * Q15 = Q15 */
+ qtmp3 = qtmp2>>15; /* Q15 -> Q0 */
+
+ tmpUW32 = kCdfQ16[ind] + qtmp3; /* Q0 + Q0 = Q0 */
+ return tmpUW32;
+}
+
+
+
+int WebRtcIsac_EncLogisticMulti2(
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ WebRtc_Word16 *dataQ7, /* input: data vector */
+ const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
+ const int N, /* input: data vector length / 2 */
+ const WebRtc_Word16 isSWB12kHz)
+{
+ WebRtc_UWord32 W_lower, W_upper;
+ WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
+ WebRtc_UWord8 *stream_ptr;
+ WebRtc_UWord8 *maxStreamPtr;
+ WebRtc_UWord8 *stream_ptr_carry;
+ WebRtc_UWord32 cdf_lo, cdf_hi;
+ int k;
+
+ /* point to beginning of stream buffer */
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ W_upper = streamdata->W_upper;
+
+ maxStreamPtr = streamdata->stream + STREAM_SIZE_MAX_60 - 1;
+ for (k = 0; k < N; k++)
+ {
+ /* compute cdf_lower and cdf_upper by evaluating the piecewise linear cdf */
+ cdf_lo = piecewise((*dataQ7 - 64) * *envQ8);
+ cdf_hi = piecewise((*dataQ7 + 64) * *envQ8);
+
+ /* test and clip if probability gets too small */
+ while (cdf_lo+1 >= cdf_hi) {
+ /* clip */
+ if (*dataQ7 > 0) {
+ *dataQ7 -= 128;
+ cdf_hi = cdf_lo;
+ cdf_lo = piecewise((*dataQ7 - 64) * *envQ8);
+ } else {
+ *dataQ7 += 128;
+ cdf_lo = cdf_hi;
+ cdf_hi = piecewise((*dataQ7 + 64) * *envQ8);
+ }
+ }
+
+ dataQ7++;
+ // increment only once per 4 iterations for SWB-16kHz or WB
+ // increment only once per 2 iterations for SWB-12kHz
+ envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1));
+
+
+ /* update interval */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = W_upper >> 16;
+ W_lower = W_upper_MSB * cdf_lo;
+ W_lower += (W_upper_LSB * cdf_lo) >> 16;
+ W_upper = W_upper_MSB * cdf_hi;
+ W_upper += (W_upper_LSB * cdf_hi) >> 16;
+
+ /* shift interval such that it begins at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamdata->streamval += W_lower;
+
+ /* handle carry */
+ if (streamdata->streamval < W_lower)
+ {
+ /* propagate carry */
+ stream_ptr_carry = stream_ptr;
+ while (!(++(*--stream_ptr_carry)));
+ }
+
+ /* renormalize interval, store most significant byte of streamval and update streamval */
+ while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
+ {
+ W_upper <<= 8;
+ *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
+
+ if(stream_ptr > maxStreamPtr)
+ {
+ return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
+ }
+ streamdata->streamval <<= 8;
+ }
+ }
+
+ /* calculate new stream_index */
+ streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
+ streamdata->W_upper = W_upper;
+
+ return 0;
+}
+
+
+
+int WebRtcIsac_DecLogisticMulti2(
+ WebRtc_Word16 *dataQ7, /* output: data vector */
+ Bitstr *streamdata, /* in-/output struct containing bitstream */
+ const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
+ const WebRtc_Word16 *ditherQ7,/* input: dither vector */
+ const int N, /* input: data vector length */
+ const WebRtc_Word16 isSWB12kHz)
+{
+ WebRtc_UWord32 W_lower, W_upper;
+ WebRtc_UWord32 W_tmp;
+ WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
+ WebRtc_UWord32 streamval;
+ const WebRtc_UWord8 *stream_ptr;
+ WebRtc_UWord32 cdf_tmp;
+ WebRtc_Word16 candQ7;
+ int k;
+
+ stream_ptr = streamdata->stream + streamdata->stream_index;
+ W_upper = streamdata->W_upper;
+ if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
+ {
+ /* read first word from bytestream */
+ streamval = *stream_ptr << 24;
+ streamval |= *++stream_ptr << 16;
+ streamval |= *++stream_ptr << 8;
+ streamval |= *++stream_ptr;
+ } else {
+ streamval = streamdata->streamval;
+ }
+
+
+ for (k = 0; k < N; k++)
+ {
+ /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
+ W_upper_LSB = W_upper & 0x0000FFFF;
+ W_upper_MSB = W_upper >> 16;
+
+ /* find first candidate by inverting the logistic cdf */
+ candQ7 = - *ditherQ7 + 64;
+ cdf_tmp = piecewise(candQ7 * *envQ8);
+
+ W_tmp = W_upper_MSB * cdf_tmp;
+ W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
+ if (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ candQ7 += 128;
+ cdf_tmp = piecewise(candQ7 * *envQ8);
+
+ W_tmp = W_upper_MSB * cdf_tmp;
+ W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
+ while (streamval > W_tmp)
+ {
+ W_lower = W_tmp;
+ candQ7 += 128;
+ cdf_tmp = piecewise(candQ7 * *envQ8);
+
+ W_tmp = W_upper_MSB * cdf_tmp;
+ W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
+
+ /* error check */
+ if (W_lower == W_tmp) return -1;
+ }
+ W_upper = W_tmp;
+
+ /* another sample decoded */
+ *dataQ7 = candQ7 - 64;
+ }
+ else
+ {
+ W_upper = W_tmp;
+ candQ7 -= 128;
+ cdf_tmp = piecewise(candQ7 * *envQ8);
+
+ W_tmp = W_upper_MSB * cdf_tmp;
+ W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
+ while ( !(streamval > W_tmp) )
+ {
+ W_upper = W_tmp;
+ candQ7 -= 128;
+ cdf_tmp = piecewise(candQ7 * *envQ8);
+
+ W_tmp = W_upper_MSB * cdf_tmp;
+ W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
+
+ /* error check */
+ if (W_upper == W_tmp) return -1;
+ }
+ W_lower = W_tmp;
+
+ /* another sample decoded */
+ *dataQ7 = candQ7 + 64;
+ }
+ ditherQ7++;
+ dataQ7++;
+ // increment only once per 4 iterations for SWB-16kHz or WB
+ // increment only once per 2 iterations for SWB-12kHz
+ envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1));
+
+ /* shift interval to start at zero */
+ W_upper -= ++W_lower;
+
+ /* add integer to bitstream */
+ streamval -= W_lower;
+
+ /* renormalize interval and update streamval */
+ while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
+ {
+ /* read next byte from stream */
+ streamval = (streamval << 8) | *++stream_ptr;
+ W_upper <<= 8;
+ }
+ }
+
+ streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
+ streamdata->W_upper = W_upper;
+ streamdata->streamval = streamval;
+
+ /* find number of bytes in original stream (determined by current interval width) */
+ if ( W_upper > 0x01FFFFFF )
+ return streamdata->stream_index - 2;
+ else
+ return streamdata->stream_index - 1;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c
new file mode 100644
index 0000000..d0a50c5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c
@@ -0,0 +1,1020 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * BwEstimator.c
+ *
+ * This file contains the code for the Bandwidth Estimator designed
+ * for iSAC.
+ *
+ */
+
+#include "bandwidth_estimator.h"
+#include "settings.h"
+#include "isac.h"
+
+#include <math.h>
+
+/* array of quantization levels for bottle neck info; Matlab code: */
+/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
+static const float kQRateTableWb[12] =
+{
+ 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f,
+ 18859.8f, 20963.3f, 23301.4f, 25900.3f, 28789.0f, 32000.0f};
+
+
+static const float kQRateTableSwb[24] =
+{
+ 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f,
+ 18859.8f, 20963.3f, 23153.1f, 25342.9f, 27532.7f, 29722.5f,
+ 31912.3f, 34102.1f, 36291.9f, 38481.7f, 40671.4f, 42861.2f,
+ 45051.0f, 47240.8f, 49430.6f, 51620.4f, 53810.2f, 56000.0f,
+};
+
+
+
+
+WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
+ BwEstimatorstr* bwest_str,
+ enum IsacSamplingRate encoderSampRate,
+ enum IsacSamplingRate decoderSampRate)
+{
+ switch(encoderSampRate)
+ {
+ case kIsacWideband:
+ {
+ bwest_str->send_bw_avg = INIT_BN_EST_WB;
+ break;
+ }
+ case kIsacSuperWideband:
+ {
+ bwest_str->send_bw_avg = INIT_BN_EST_SWB;
+ break;
+ }
+ }
+
+ switch(decoderSampRate)
+ {
+ case kIsacWideband:
+ {
+ bwest_str->prev_frame_length = INIT_FRAME_LEN_WB;
+ bwest_str->rec_bw_inv = 1.0f /
+ (INIT_BN_EST_WB + INIT_HDR_RATE_WB);
+ bwest_str->rec_bw = (WebRtc_Word32)INIT_BN_EST_WB;
+ bwest_str->rec_bw_avg_Q = INIT_BN_EST_WB;
+ bwest_str->rec_bw_avg = INIT_BN_EST_WB + INIT_HDR_RATE_WB;
+ bwest_str->rec_header_rate = INIT_HDR_RATE_WB;
+ break;
+ }
+ case kIsacSuperWideband:
+ {
+ bwest_str->prev_frame_length = INIT_FRAME_LEN_SWB;
+ bwest_str->rec_bw_inv = 1.0f /
+ (INIT_BN_EST_SWB + INIT_HDR_RATE_SWB);
+ bwest_str->rec_bw = (WebRtc_Word32)INIT_BN_EST_SWB;
+ bwest_str->rec_bw_avg_Q = INIT_BN_EST_SWB;
+ bwest_str->rec_bw_avg = INIT_BN_EST_SWB + INIT_HDR_RATE_SWB;
+ bwest_str->rec_header_rate = INIT_HDR_RATE_SWB;
+ break;
+ }
+ }
+
+ bwest_str->prev_rec_rtp_number = 0;
+ bwest_str->prev_rec_arr_ts = 0;
+ bwest_str->prev_rec_send_ts = 0;
+ bwest_str->prev_rec_rtp_rate = 1.0f;
+ bwest_str->last_update_ts = 0;
+ bwest_str->last_reduction_ts = 0;
+ bwest_str->count_tot_updates_rec = -9;
+ bwest_str->rec_jitter = 10.0f;
+ bwest_str->rec_jitter_short_term = 0.0f;
+ bwest_str->rec_jitter_short_term_abs = 5.0f;
+ bwest_str->rec_max_delay = 10.0f;
+ bwest_str->rec_max_delay_avg_Q = 10.0f;
+ bwest_str->num_pkts_rec = 0;
+
+ bwest_str->send_max_delay_avg = 10.0f;
+
+ bwest_str->hsn_detect_rec = 0;
+
+ bwest_str->num_consec_rec_pkts_over_30k = 0;
+
+ bwest_str->hsn_detect_snd = 0;
+
+ bwest_str->num_consec_snt_pkts_over_30k = 0;
+
+ bwest_str->in_wait_period = 0;
+
+ bwest_str->change_to_WB = 0;
+
+ bwest_str->numConsecLatePkts = 0;
+ bwest_str->consecLatency = 0;
+ bwest_str->inWaitLatePkts = 0;
+ bwest_str->senderTimestamp = 0;
+ bwest_str->receiverTimestamp = 0;
+ return 0;
+}
+
+/* This function updates both bottle neck rates */
+/* Parameters: */
+/* rtp_number - value from RTP packet, from NetEq */
+/* frame length - length of signal frame in ms, from iSAC decoder */
+/* send_ts - value in RTP header giving send time in samples */
+/* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
+/* pksize - size of packet in bytes, from NetEq */
+/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
+/* returns 0 if everything went fine, -1 otherwise */
+WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
+ BwEstimatorstr *bwest_str,
+ const WebRtc_UWord16 rtp_number,
+ const WebRtc_Word32 frame_length,
+ const WebRtc_UWord32 send_ts,
+ const WebRtc_UWord32 arr_ts,
+ const WebRtc_Word32 pksize
+ /*, const WebRtc_UWord16 Index*/)
+{
+ float weight = 0.0f;
+ float curr_bw_inv = 0.0f;
+ float rec_rtp_rate;
+ float t_diff_proj;
+ float arr_ts_diff;
+ float send_ts_diff;
+ float arr_time_noise;
+ float arr_time_noise_abs;
+
+ float delay_correction_factor = 1;
+ float late_diff = 0.0f;
+ int immediate_set = 0;
+ int num_pkts_expected;
+
+
+ // We have to adjust the header-rate if the first packet has a
+ // frame-size different than the initialized value.
+ if ( frame_length != bwest_str->prev_frame_length )
+ {
+ bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f *
+ 1000.0f / (float)frame_length; /* bits/s */
+ }
+
+ /* UPDATE ESTIMATES ON THIS SIDE */
+ /* compute far-side transmission rate */
+ rec_rtp_rate = ((float)pksize * 8.0f * 1000.0f / (float)frame_length) +
+ bwest_str->rec_header_rate;
+ // rec_rtp_rate packet bits/s + header bits/s
+
+ /* check for timer wrap-around */
+ if (arr_ts < bwest_str->prev_rec_arr_ts)
+ {
+ bwest_str->prev_rec_arr_ts = arr_ts;
+ bwest_str->last_update_ts = arr_ts;
+ bwest_str->last_reduction_ts = arr_ts + 3*FS;
+ bwest_str->num_pkts_rec = 0;
+
+ /* store frame length */
+ bwest_str->prev_frame_length = frame_length;
+
+ /* store far-side transmission rate */
+ bwest_str->prev_rec_rtp_rate = rec_rtp_rate;
+
+ /* store far-side RTP time stamp */
+ bwest_str->prev_rec_rtp_number = rtp_number;
+
+ return 0;
+ }
+
+ bwest_str->num_pkts_rec++;
+
+ /* check that it's not one of the first 9 packets */
+ if ( bwest_str->count_tot_updates_rec > 0 )
+ {
+ if(bwest_str->in_wait_period > 0 )
+ {
+ bwest_str->in_wait_period--;
+ }
+
+ bwest_str->inWaitLatePkts -= ((bwest_str->inWaitLatePkts > 0)? 1:0);
+ send_ts_diff = (float)(send_ts - bwest_str->prev_rec_send_ts);
+
+ if (send_ts_diff <= (16 * frame_length)*2)
+ //doesn't allow for a dropped packet, not sure necessary to be
+ // that strict -DH
+ {
+ /* if not been updated for a long time, reduce the BN estimate */
+ if((WebRtc_UWord32)(arr_ts - bwest_str->last_update_ts) *
+ 1000.0f / FS > 3000)
+ {
+ //how many frames should have been received since the last
+ // update if too many have been dropped or there have been
+ // big delays won't allow this reduction may no longer need
+ // the send_ts_diff here
+ num_pkts_expected = (int)(((float)(arr_ts -
+ bwest_str->last_update_ts) * 1000.0f /(float) FS) /
+ (float)frame_length);
+
+ if(((float)bwest_str->num_pkts_rec/(float)num_pkts_expected) >
+ 0.9)
+ {
+ float inv_bitrate = (float) pow( 0.99995,
+ (double)((WebRtc_UWord32)(arr_ts -
+ bwest_str->last_reduction_ts)*1000.0f/FS) );
+
+ if ( inv_bitrate )
+ {
+ bwest_str->rec_bw_inv /= inv_bitrate;
+
+ //precautionary, likely never necessary
+ if (bwest_str->hsn_detect_snd &&
+ bwest_str->hsn_detect_rec)
+ {
+ if (bwest_str->rec_bw_inv > 0.000066f)
+ {
+ bwest_str->rec_bw_inv = 0.000066f;
+ }
+ }
+ }
+ else
+ {
+ bwest_str->rec_bw_inv = 1.0f /
+ (INIT_BN_EST_WB + INIT_HDR_RATE_WB);
+ }
+ /* reset time-since-update counter */
+ bwest_str->last_reduction_ts = arr_ts;
+ }
+ else
+ //reset here?
+ {
+ bwest_str->last_reduction_ts = arr_ts + 3*FS;
+ bwest_str->last_update_ts = arr_ts;
+ bwest_str->num_pkts_rec = 0;
+ }
+ }
+ }
+ else
+ {
+ bwest_str->last_reduction_ts = arr_ts + 3*FS;
+ bwest_str->last_update_ts = arr_ts;
+ bwest_str->num_pkts_rec = 0;
+ }
+
+
+ /* temporarily speed up adaptation if frame length has changed */
+ if ( frame_length != bwest_str->prev_frame_length )
+ {
+ bwest_str->count_tot_updates_rec = 10;
+ bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f *
+ 1000.0f / (float)frame_length; /* bits/s */
+
+ bwest_str->rec_bw_inv = 1.0f /((float)bwest_str->rec_bw +
+ bwest_str->rec_header_rate);
+ }
+
+ ////////////////////////
+ arr_ts_diff = (float)(arr_ts - bwest_str->prev_rec_arr_ts);
+
+ if (send_ts_diff > 0 )
+ {
+ late_diff = arr_ts_diff - send_ts_diff;
+ }
+ else
+ {
+ late_diff = arr_ts_diff - (float)(16 * frame_length);
+ }
+
+ if((late_diff > 0) && !bwest_str->inWaitLatePkts)
+ {
+ bwest_str->numConsecLatePkts++;
+ bwest_str->consecLatency += late_diff;
+ }
+ else
+ {
+ bwest_str->numConsecLatePkts = 0;
+ bwest_str->consecLatency = 0;
+ }
+ if(bwest_str->numConsecLatePkts > 50)
+ {
+ float latencyMs = bwest_str->consecLatency/(FS/1000);
+ float averageLatencyMs = latencyMs / bwest_str->numConsecLatePkts;
+ delay_correction_factor = frame_length / (frame_length + averageLatencyMs);
+ immediate_set = 1;
+ bwest_str->inWaitLatePkts = (WebRtc_Word16)((bwest_str->consecLatency/(FS/1000)) / 30);// + 150;
+ bwest_str->start_wait_period = arr_ts;
+ }
+ ///////////////////////////////////////////////
+
+
+
+ /* update only if previous packet was not lost */
+ if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 )
+ {
+
+
+ if (!(bwest_str->hsn_detect_snd && bwest_str->hsn_detect_rec))
+ {
+ if ((arr_ts_diff > (float)(16 * frame_length)))
+ {
+ //1/2 second
+ if ((late_diff > 8000.0f) && !bwest_str->in_wait_period)
+ {
+ delay_correction_factor = 0.7f;
+ bwest_str->in_wait_period = 55;
+ bwest_str->start_wait_period = arr_ts;
+ immediate_set = 1;
+ }
+ //320 ms
+ else if (late_diff > 5120.0f && !bwest_str->in_wait_period)
+ {
+ delay_correction_factor = 0.8f;
+ immediate_set = 1;
+ bwest_str->in_wait_period = 44;
+ bwest_str->start_wait_period = arr_ts;
+ }
+ }
+ }
+
+
+ if ((bwest_str->prev_rec_rtp_rate > bwest_str->rec_bw_avg) &&
+ (rec_rtp_rate > bwest_str->rec_bw_avg) &&
+ !bwest_str->in_wait_period)
+ {
+ /* test if still in initiation period and increment counter */
+ if (bwest_str->count_tot_updates_rec++ > 99)
+ {
+ /* constant weight after initiation part */
+ weight = 0.01f;
+ }
+ else
+ {
+ /* weight decreases with number of updates */
+ weight = 1.0f / (float) bwest_str->count_tot_updates_rec;
+ }
+ /* Bottle Neck Estimation */
+
+ /* limit outliers */
+ /* if more than 25 ms too much */
+ if (arr_ts_diff > frame_length * FS/1000 + 400.0f)
+ {
+ // in samples, why 25ms??
+ arr_ts_diff = frame_length * FS/1000 + 400.0f;
+ }
+ if(arr_ts_diff < (frame_length * FS/1000) - 160.0f)
+ {
+ /* don't allow it to be less than frame rate - 10 ms */
+ arr_ts_diff = (float)frame_length * FS/1000 - 160.0f;
+ }
+
+ /* compute inverse receiving rate for last packet */
+ curr_bw_inv = arr_ts_diff / ((float)(pksize + HEADER_SIZE) *
+ 8.0f * FS); // (180+35)*8*16000 = 27.5 Mbit....
+
+
+ if(curr_bw_inv <
+ (1.0f / (MAX_ISAC_BW + bwest_str->rec_header_rate)))
+ {
+ // don't allow inv rate to be larger than MAX
+ curr_bw_inv = (1.0f /
+ (MAX_ISAC_BW + bwest_str->rec_header_rate));
+ }
+
+ /* update bottle neck rate estimate */
+ bwest_str->rec_bw_inv = weight * curr_bw_inv +
+ (1.0f - weight) * bwest_str->rec_bw_inv;
+
+ /* reset time-since-update counter */
+ bwest_str->last_update_ts = arr_ts;
+ bwest_str->last_reduction_ts = arr_ts + 3 * FS;
+ bwest_str->num_pkts_rec = 0;
+
+ /* Jitter Estimation */
+ /* projected difference between arrival times */
+ t_diff_proj = ((float)(pksize + HEADER_SIZE) * 8.0f *
+ 1000.0f) / bwest_str->rec_bw_avg;
+
+
+ // difference between projected and actual
+ // arrival time differences
+ arr_time_noise = (float)(arr_ts_diff*1000.0f/FS) -
+ t_diff_proj;
+ arr_time_noise_abs = (float) fabs( arr_time_noise );
+
+ /* long term averaged absolute jitter */
+ bwest_str->rec_jitter = weight * arr_time_noise_abs +
+ (1.0f - weight) * bwest_str->rec_jitter;
+ if (bwest_str->rec_jitter > 10.0f)
+ {
+ bwest_str->rec_jitter = 10.0f;
+ }
+ /* short term averaged absolute jitter */
+ bwest_str->rec_jitter_short_term_abs = 0.05f *
+ arr_time_noise_abs + 0.95f *
+ bwest_str->rec_jitter_short_term_abs;
+
+ /* short term averaged jitter */
+ bwest_str->rec_jitter_short_term = 0.05f * arr_time_noise +
+ 0.95f * bwest_str->rec_jitter_short_term;
+ }
+ }
+ }
+ else
+ {
+ // reset time-since-update counter when
+ // receiving the first 9 packets
+ bwest_str->last_update_ts = arr_ts;
+ bwest_str->last_reduction_ts = arr_ts + 3*FS;
+ bwest_str->num_pkts_rec = 0;
+
+ bwest_str->count_tot_updates_rec++;
+ }
+
+ /* limit minimum bottle neck rate */
+ if (bwest_str->rec_bw_inv > 1.0f / ((float)MIN_ISAC_BW +
+ bwest_str->rec_header_rate))
+ {
+ bwest_str->rec_bw_inv = 1.0f / ((float)MIN_ISAC_BW +
+ bwest_str->rec_header_rate);
+ }
+
+ // limit maximum bitrate
+ if (bwest_str->rec_bw_inv < 1.0f / ((float)MAX_ISAC_BW +
+ bwest_str->rec_header_rate))
+ {
+ bwest_str->rec_bw_inv = 1.0f / ((float)MAX_ISAC_BW +
+ bwest_str->rec_header_rate);
+ }
+
+ /* store frame length */
+ bwest_str->prev_frame_length = frame_length;
+
+ /* store far-side transmission rate */
+ bwest_str->prev_rec_rtp_rate = rec_rtp_rate;
+
+ /* store far-side RTP time stamp */
+ bwest_str->prev_rec_rtp_number = rtp_number;
+
+ // Replace bwest_str->rec_max_delay by the new
+ // value (atomic operation)
+ bwest_str->rec_max_delay = 3.0f * bwest_str->rec_jitter;
+
+ /* store send and arrival time stamp */
+ bwest_str->prev_rec_arr_ts = arr_ts ;
+ bwest_str->prev_rec_send_ts = send_ts;
+
+ /* Replace bwest_str->rec_bw by the new value (atomic operation) */
+ bwest_str->rec_bw = (WebRtc_Word32)(1.0f / bwest_str->rec_bw_inv -
+ bwest_str->rec_header_rate);
+
+ if (immediate_set)
+ {
+ bwest_str->rec_bw = (WebRtc_Word32) (delay_correction_factor *
+ (float) bwest_str->rec_bw);
+
+ if (bwest_str->rec_bw < (WebRtc_Word32) MIN_ISAC_BW)
+ {
+ bwest_str->rec_bw = (WebRtc_Word32) MIN_ISAC_BW;
+ }
+
+ bwest_str->rec_bw_avg = bwest_str->rec_bw +
+ bwest_str->rec_header_rate;
+
+ bwest_str->rec_bw_avg_Q = (float) bwest_str->rec_bw;
+
+ bwest_str->rec_jitter_short_term = 0.0f;
+
+ bwest_str->rec_bw_inv = 1.0f / (bwest_str->rec_bw +
+ bwest_str->rec_header_rate);
+
+ bwest_str->count_tot_updates_rec = 1;
+
+ immediate_set = 0;
+ bwest_str->consecLatency = 0;
+ bwest_str->numConsecLatePkts = 0;
+ }
+
+ return 0;
+}
+
+
+/* This function updates the send bottle neck rate */
+/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
+/* returns 0 if everything went fine, -1 otherwise */
+WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word16 index,
+ enum IsacSamplingRate encoderSamplingFreq)
+{
+ if((index < 0) || (index > 23))
+ {
+ return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
+ }
+
+ /* UPDATE ESTIMATES FROM OTHER SIDE */
+ if(encoderSamplingFreq == kIsacWideband)
+ {
+ if(index > 11)
+ {
+ index -= 12;
+ /* compute the jitter estimate as decoded on the other side */
+ bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
+ 0.1f * (float)MAX_ISAC_MD;
+ }
+ else
+ {
+ /* compute the jitter estimate as decoded on the other side */
+ bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
+ 0.1f * (float)MIN_ISAC_MD;
+ }
+
+ /* compute the BN estimate as decoded on the other side */
+ bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg +
+ 0.1f * kQRateTableWb[index];
+ }
+ else
+ {
+ /* compute the BN estimate as decoded on the other side */
+ bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg +
+ 0.1f * kQRateTableSwb[index];
+ }
+
+ if (bwest_str->send_bw_avg > (float) 28000 && !bwest_str->hsn_detect_snd)
+ {
+ bwest_str->num_consec_snt_pkts_over_30k++;
+
+ if (bwest_str->num_consec_snt_pkts_over_30k >= 66)
+ {
+ //approx 2 seconds with 30ms frames
+ bwest_str->hsn_detect_snd = 1;
+ }
+ }
+ else if (!bwest_str->hsn_detect_snd)
+ {
+ bwest_str->num_consec_snt_pkts_over_30k = 0;
+ }
+ return 0;
+}
+
+// called when there is upper-band bit-stream to update jitter
+// statistics.
+WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word32 index)
+{
+ if((index < 0) || (index > 23))
+ {
+ return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
+ }
+
+ if(index > 0)
+ {
+ /* compute the jitter estimate as decoded on the other side */
+ bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
+ 0.1f * (float)MAX_ISAC_MD;
+ }
+ else
+ {
+ /* compute the jitter estimate as decoded on the other side */
+ bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
+ 0.1f * (float)MIN_ISAC_MD;
+ }
+
+ return 0;
+}
+
+
+
+// Returns the bandwidth/jitter estimation code (integer 0...23)
+// to put in the sending iSAC payload
+WebRtc_UWord16
+WebRtcIsac_GetDownlinkBwJitIndexImpl(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word16* bottleneckIndex,
+ WebRtc_Word16* jitterInfo,
+ enum IsacSamplingRate decoderSamplingFreq)
+{
+ float MaxDelay;
+ //WebRtc_UWord16 MaxDelayBit;
+
+ float rate;
+ float r;
+ float e1, e2;
+ const float weight = 0.1f;
+ const float* ptrQuantizationTable;
+ WebRtc_Word16 addJitterInfo;
+ WebRtc_Word16 minInd;
+ WebRtc_Word16 maxInd;
+ WebRtc_Word16 midInd;
+
+ /* Get Max Delay Bit */
+ /* get unquantized max delay */
+ MaxDelay = (float)WebRtcIsac_GetDownlinkMaxDelay(bwest_str);
+
+ if ( ((1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight *
+ MAX_ISAC_MD - MaxDelay) > (MaxDelay - (1.f-weight) *
+ bwest_str->rec_max_delay_avg_Q - weight * MIN_ISAC_MD) )
+ {
+ jitterInfo[0] = 0;
+ /* update quantized average */
+ bwest_str->rec_max_delay_avg_Q =
+ (1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight *
+ (float)MIN_ISAC_MD;
+ }
+ else
+ {
+ jitterInfo[0] = 1;
+ /* update quantized average */
+ bwest_str->rec_max_delay_avg_Q =
+ (1.f-weight) * bwest_str->rec_max_delay_avg_Q + weight *
+ (float)MAX_ISAC_MD;
+ }
+
+ // Get unquantized rate.
+ rate = (float)WebRtcIsac_GetDownlinkBandwidth(bwest_str);
+
+ /* Get Rate Index */
+ if(decoderSamplingFreq == kIsacWideband)
+ {
+ ptrQuantizationTable = kQRateTableWb;
+ addJitterInfo = 1;
+ maxInd = 11;
+ }
+ else
+ {
+ ptrQuantizationTable = kQRateTableSwb;
+ addJitterInfo = 0;
+ maxInd = 23;
+ }
+
+ minInd = 0;
+ while(maxInd > minInd + 1)
+ {
+ midInd = (maxInd + minInd) >> 1;
+ if(rate > ptrQuantizationTable[midInd])
+ {
+ minInd = midInd;
+ }
+ else
+ {
+ maxInd = midInd;
+ }
+ }
+ // Chose the index which gives results an average which is closest
+ // to rate
+ r = (1 - weight) * bwest_str->rec_bw_avg_Q - rate;
+ e1 = weight * ptrQuantizationTable[minInd] + r;
+ e2 = weight * ptrQuantizationTable[maxInd] + r;
+ e1 = (e1 > 0)? e1:-e1;
+ e2 = (e2 > 0)? e2:-e2;
+ if(e1 < e2)
+ {
+ bottleneckIndex[0] = minInd;
+ }
+ else
+ {
+ bottleneckIndex[0] = maxInd;
+ }
+
+ bwest_str->rec_bw_avg_Q = (1 - weight) * bwest_str->rec_bw_avg_Q +
+ weight * ptrQuantizationTable[bottleneckIndex[0]];
+ bottleneckIndex[0] += jitterInfo[0] * 12 * addJitterInfo;
+
+ bwest_str->rec_bw_avg = (1 - weight) * bwest_str->rec_bw_avg + weight *
+ (rate + bwest_str->rec_header_rate);
+
+ return 0;
+}
+
+
+
+/* get the bottle neck rate from far side to here, as estimated on this side */
+WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
+{
+ WebRtc_Word32 rec_bw;
+ float jitter_sign;
+ float bw_adjust;
+
+ /* create a value between -1.0 and 1.0 indicating "average sign" of jitter */
+ jitter_sign = bwest_str->rec_jitter_short_term /
+ bwest_str->rec_jitter_short_term_abs;
+
+ /* adjust bw proportionally to negative average jitter sign */
+ bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign);
+
+ /* adjust Rate if jitter sign is mostly constant */
+ rec_bw = (WebRtc_Word32)(bwest_str->rec_bw * bw_adjust);
+
+ /* limit range of bottle neck rate */
+ if (rec_bw < MIN_ISAC_BW)
+ {
+ rec_bw = MIN_ISAC_BW;
+ }
+ else if (rec_bw > MAX_ISAC_BW)
+ {
+ rec_bw = MAX_ISAC_BW;
+ }
+ return rec_bw;
+}
+
+/* Returns the max delay (in ms) */
+WebRtc_Word32
+WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
+{
+ WebRtc_Word32 rec_max_delay;
+
+ rec_max_delay = (WebRtc_Word32)(bwest_str->rec_max_delay);
+
+ /* limit range of jitter estimate */
+ if (rec_max_delay < MIN_ISAC_MD)
+ {
+ rec_max_delay = MIN_ISAC_MD;
+ }
+ else if (rec_max_delay > MAX_ISAC_MD)
+ {
+ rec_max_delay = MAX_ISAC_MD;
+ }
+ return rec_max_delay;
+}
+
+/* get the bottle neck rate from here to far side, as estimated by far side */
+void
+WebRtcIsac_GetUplinkBandwidth(
+ const BwEstimatorstr* bwest_str,
+ WebRtc_Word32* bitRate)
+{
+ /* limit range of bottle neck rate */
+ if (bwest_str->send_bw_avg < MIN_ISAC_BW)
+ {
+ *bitRate = MIN_ISAC_BW;
+ }
+ else if (bwest_str->send_bw_avg > MAX_ISAC_BW)
+ {
+ *bitRate = MAX_ISAC_BW;
+ }
+ else
+ {
+ *bitRate = (WebRtc_Word32)(bwest_str->send_bw_avg);
+ }
+ return;
+}
+
+/* Returns the max delay value from the other side in ms */
+WebRtc_Word32
+WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str)
+{
+ WebRtc_Word32 send_max_delay;
+
+ send_max_delay = (WebRtc_Word32)(bwest_str->send_max_delay_avg);
+
+ /* limit range of jitter estimate */
+ if (send_max_delay < MIN_ISAC_MD)
+ {
+ send_max_delay = MIN_ISAC_MD;
+ }
+ else if (send_max_delay > MAX_ISAC_MD)
+ {
+ send_max_delay = MAX_ISAC_MD;
+ }
+ return send_max_delay;
+}
+
+
+/*
+ * update long-term average bitrate and amount of data in buffer
+ * returns minimum payload size (bytes)
+ */
+int WebRtcIsac_GetMinBytes(
+ RateModel* State,
+ int StreamSize, /* bytes in bitstream */
+ const int FrameSamples, /* samples per frame */
+ const double BottleNeck, /* bottle neck rate; excl headers (bps) */
+ const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
+ enum ISACBandwidth bandwidth
+ /*,WebRtc_Word16 frequentLargePackets*/)
+{
+ double MinRate = 0.0;
+ int MinBytes;
+ double TransmissionTime;
+ int burstInterval = BURST_INTERVAL;
+
+ // first 10 packets @ low rate, then INIT_BURST_LEN packets @
+ // fixed rate of INIT_RATE bps
+ if (State->InitCounter > 0)
+ {
+ if (State->InitCounter-- <= INIT_BURST_LEN)
+ {
+ if(bandwidth == isac8kHz)
+ {
+ MinRate = INIT_RATE_WB;
+ }
+ else
+ {
+ MinRate = INIT_RATE_SWB;
+ }
+ }
+ else
+ {
+ MinRate = 0;
+ }
+ }
+ else
+ {
+ /* handle burst */
+ if (State->BurstCounter)
+ {
+ if (State->StillBuffered < (1.0 - 1.0/BURST_LEN) * DelayBuildUp)
+ {
+ /* max bps derived from BottleNeck and DelayBuildUp values */
+ MinRate = (1.0 + (FS/1000) * DelayBuildUp /
+ (double)(BURST_LEN * FrameSamples)) * BottleNeck;
+ }
+ else
+ {
+ // max bps derived from StillBuffered and DelayBuildUp
+ // values
+ MinRate = (1.0 + (FS/1000) * (DelayBuildUp -
+ State->StillBuffered) / (double)FrameSamples) * BottleNeck;
+ if (MinRate < 1.04 * BottleNeck)
+ {
+ MinRate = 1.04 * BottleNeck;
+ }
+ }
+ State->BurstCounter--;
+ }
+ }
+
+
+ /* convert rate from bits/second to bytes/packet */
+ MinBytes = (int) (MinRate * FrameSamples / (8.0 * FS));
+
+ /* StreamSize will be adjusted if less than MinBytes */
+ if (StreamSize < MinBytes)
+ {
+ StreamSize = MinBytes;
+ }
+
+ /* keep track of when bottle neck was last exceeded by at least 1% */
+ if (StreamSize * 8.0 * FS / FrameSamples > 1.01 * BottleNeck) {
+ if (State->PrevExceed) {
+ /* bottle_neck exceded twice in a row, decrease ExceedAgo */
+ State->ExceedAgo -= /*BURST_INTERVAL*/ burstInterval / (BURST_LEN - 1);
+ if (State->ExceedAgo < 0)
+ State->ExceedAgo = 0;
+ }
+ else
+ {
+ State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */
+ State->PrevExceed = 1;
+ }
+ }
+ else
+ {
+ State->PrevExceed = 0;
+ State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */
+ }
+
+ /* set burst flag if bottle neck not exceeded for long time */
+ if ((State->ExceedAgo > burstInterval) &&
+ (State->BurstCounter == 0))
+ {
+ if (State->PrevExceed)
+ {
+ State->BurstCounter = BURST_LEN - 1;
+ }
+ else
+ {
+ State->BurstCounter = BURST_LEN;
+ }
+ }
+
+
+ /* Update buffer delay */
+ TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */
+ State->StillBuffered += TransmissionTime;
+ State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
+ if (State->StillBuffered < 0.0)
+ {
+ State->StillBuffered = 0.0;
+ }
+
+ return MinBytes;
+}
+
+
+/*
+ * update long-term average bitrate and amount of data in buffer
+ */
+void WebRtcIsac_UpdateRateModel(
+ RateModel *State,
+ int StreamSize, /* bytes in bitstream */
+ const int FrameSamples, /* samples per frame */
+ const double BottleNeck) /* bottle neck rate; excl headers (bps) */
+{
+ double TransmissionTime;
+
+ /* avoid the initial "high-rate" burst */
+ State->InitCounter = 0;
+
+ /* Update buffer delay */
+ TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */
+ State->StillBuffered += TransmissionTime;
+ State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
+ if (State->StillBuffered < 0.0)
+ State->StillBuffered = 0.0;
+
+}
+
+
+void WebRtcIsac_InitRateModel(
+ RateModel *State)
+{
+ State->PrevExceed = 0; /* boolean */
+ State->ExceedAgo = 0; /* ms */
+ State->BurstCounter = 0; /* packets */
+ State->InitCounter = INIT_BURST_LEN + 10; /* packets */
+ State->StillBuffered = 1.0; /* ms */
+}
+
+int WebRtcIsac_GetNewFrameLength(
+ double bottle_neck,
+ int current_framesamples)
+{
+ int new_framesamples;
+
+ const int Thld_20_30 = 20000;
+
+ //const int Thld_30_20 = 30000;
+ const int Thld_30_20 = 1000000; // disable 20 ms frames
+
+ const int Thld_30_60 = 18000;
+ //const int Thld_30_60 = 0; // disable 60 ms frames
+
+ const int Thld_60_30 = 27000;
+
+
+ new_framesamples = current_framesamples;
+
+ /* find new framelength */
+ switch(current_framesamples) {
+ case 320:
+ if (bottle_neck < Thld_20_30)
+ new_framesamples = 480;
+ break;
+ case 480:
+ if (bottle_neck < Thld_30_60)
+ new_framesamples = 960;
+ else if (bottle_neck > Thld_30_20)
+ new_framesamples = 320;
+ break;
+ case 960:
+ if (bottle_neck >= Thld_60_30)
+ new_framesamples = 480;
+ break;
+ }
+
+ return new_framesamples;
+}
+
+double WebRtcIsac_GetSnr(
+ double bottle_neck,
+ int framesamples)
+{
+ double s2nr;
+
+ const double a_20 = -30.0;
+ const double b_20 = 0.8;
+ const double c_20 = 0.0;
+
+ const double a_30 = -23.0;
+ const double b_30 = 0.48;
+ const double c_30 = 0.0;
+
+ const double a_60 = -23.0;
+ const double b_60 = 0.53;
+ const double c_60 = 0.0;
+
+
+ /* find new SNR value */
+ switch(framesamples) {
+ case 320:
+ s2nr = a_20 + b_20 * bottle_neck * 0.001 + c_20 * bottle_neck *
+ bottle_neck * 0.000001;
+ break;
+ case 480:
+ s2nr = a_30 + b_30 * bottle_neck * 0.001 + c_30 * bottle_neck *
+ bottle_neck * 0.000001;
+ break;
+ case 960:
+ s2nr = a_60 + b_60 * bottle_neck * 0.001 + c_60 * bottle_neck *
+ bottle_neck * 0.000001;
+ break;
+ default:
+ s2nr = 0;
+ }
+
+ return s2nr;
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h
new file mode 100644
index 0000000..5604d7b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * bandwidth_estimator.h
+ *
+ * This header file contains the API for the Bandwidth Estimator
+ * designed for iSAC.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
+
+#include "structs.h"
+#include "settings.h"
+
+
+#define MIN_ISAC_BW 10000
+#define MIN_ISAC_BW_LB 10000
+#define MIN_ISAC_BW_UB 25000
+
+#define MAX_ISAC_BW 56000
+#define MAX_ISAC_BW_UB 32000
+#define MAX_ISAC_BW_LB 32000
+
+#define MIN_ISAC_MD 5
+#define MAX_ISAC_MD 25
+
+// assumed header size, in bytes; we don't know the exact number
+// (header compression may be used)
+#define HEADER_SIZE 35
+
+// Initial Frame-Size, in ms, for Wideband & Super-Wideband Mode
+#define INIT_FRAME_LEN_WB 60
+#define INIT_FRAME_LEN_SWB 30
+
+// Initial Bottleneck Estimate, in bits/sec, for
+// Wideband & Super-wideband mode
+#define INIT_BN_EST_WB 20e3f
+#define INIT_BN_EST_SWB 56e3f
+
+// Initial Header rate (header rate depends on frame-size),
+// in bits/sec, for Wideband & Super-Wideband mode.
+#define INIT_HDR_RATE_WB \
+ ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_WB)
+#define INIT_HDR_RATE_SWB \
+ ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_SWB)
+
+// number of packets in a row for a high rate burst
+#define BURST_LEN 3
+
+// ms, max time between two full bursts
+#define BURST_INTERVAL 500
+
+// number of packets in a row for initial high rate burst
+#define INIT_BURST_LEN 5
+
+// bits/s, rate for the first BURST_LEN packets
+#define INIT_RATE_WB INIT_BN_EST_WB
+#define INIT_RATE_SWB INIT_BN_EST_SWB
+
+
+#if defined(__cplusplus)
+extern "C" {
+#endif
+
+ /* This function initializes the struct */
+ /* to be called before using the struct for anything else */
+ /* returns 0 if everything went fine, -1 otherwise */
+ WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
+ BwEstimatorstr* bwest_str,
+ enum IsacSamplingRate encoderSampRate,
+ enum IsacSamplingRate decoderSampRate);
+
+ /* This function updates the receiving estimate */
+ /* Parameters: */
+ /* rtp_number - value from RTP packet, from NetEq */
+ /* frame length - length of signal frame in ms, from iSAC decoder */
+ /* send_ts - value in RTP header giving send time in samples */
+ /* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
+ /* pksize - size of packet in bytes, from NetEq */
+ /* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
+ /* returns 0 if everything went fine, -1 otherwise */
+ WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
+ BwEstimatorstr* bwest_str,
+ const WebRtc_UWord16 rtp_number,
+ const WebRtc_Word32 frame_length,
+ const WebRtc_UWord32 send_ts,
+ const WebRtc_UWord32 arr_ts,
+ const WebRtc_Word32 pksize);
+
+ /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
+ WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word16 Index,
+ enum IsacSamplingRate encoderSamplingFreq);
+
+ /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
+ WebRtc_UWord16 WebRtcIsac_GetDownlinkBwJitIndexImpl(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word16* bottleneckIndex,
+ WebRtc_Word16* jitterInfo,
+ enum IsacSamplingRate decoderSamplingFreq);
+
+ /* Returns the bandwidth estimation (in bps) */
+ WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth(
+ const BwEstimatorstr *bwest_str);
+
+ /* Returns the max delay (in ms) */
+ WebRtc_Word32 WebRtcIsac_GetDownlinkMaxDelay(
+ const BwEstimatorstr *bwest_str);
+
+ /* Returns the bandwidth that iSAC should send with in bps */
+ void WebRtcIsac_GetUplinkBandwidth(
+ const BwEstimatorstr* bwest_str,
+ WebRtc_Word32* bitRate);
+
+ /* Returns the max delay value from the other side in ms */
+ WebRtc_Word32 WebRtcIsac_GetUplinkMaxDelay(
+ const BwEstimatorstr *bwest_str);
+
+
+ /*
+ * update amount of data in bottle neck buffer and burst handling
+ * returns minimum payload size (bytes)
+ */
+ int WebRtcIsac_GetMinBytes(
+ RateModel* State,
+ int StreamSize, /* bytes in bitstream */
+ const int FrameLen, /* ms per frame */
+ const double BottleNeck, /* bottle neck rate; excl headers (bps) */
+ const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
+ enum ISACBandwidth bandwidth
+ /*,WebRtc_Word16 frequentLargePackets*/);
+
+ /*
+ * update long-term average bitrate and amount of data in buffer
+ */
+ void WebRtcIsac_UpdateRateModel(
+ RateModel* State,
+ int StreamSize, /* bytes in bitstream */
+ const int FrameSamples, /* samples per frame */
+ const double BottleNeck); /* bottle neck rate; excl headers (bps) */
+
+
+ void WebRtcIsac_InitRateModel(
+ RateModel *State);
+
+ /* Returns the new framelength value (input argument: bottle_neck) */
+ int WebRtcIsac_GetNewFrameLength(
+ double bottle_neck,
+ int current_framelength);
+
+ /* Returns the new SNR value (input argument: bottle_neck) */
+ double WebRtcIsac_GetSnr(
+ double bottle_neck,
+ int new_framelength);
+
+
+ WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
+ BwEstimatorstr* bwest_str,
+ WebRtc_Word32 index);
+
+#if defined(__cplusplus)
+}
+#endif
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h
new file mode 100644
index 0000000..6af27ea
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h
@@ -0,0 +1,292 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * codec.h
+ *
+ * This header file contains the calls to the internal encoder
+ * and decoder functions.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
+
+#include "structs.h"
+
+int WebRtcIsac_EstimateBandwidth(
+ BwEstimatorstr* bwest_str,
+ Bitstr* streamdata,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts,
+ enum IsacSamplingRate encoderSampRate,
+ enum IsacSamplingRate decoderSampRate);
+
+int WebRtcIsac_DecodeLb(
+ float* signal_out,
+ ISACLBDecStruct* ISACdec_obj,
+ WebRtc_Word16* current_framesamples,
+ WebRtc_Word16 isRCUPayload);
+
+int WebRtcIsac_DecodeRcuLb(
+ float* signal_out,
+ ISACLBDecStruct* ISACdec_obj,
+ WebRtc_Word16* current_framesamples);
+
+int WebRtcIsac_EncodeLb(
+ float* in,
+ ISACLBEncStruct* ISACencLB_obj,
+ WebRtc_Word16 codingMode,
+ WebRtc_Word16 bottleneckIndex);
+
+int WebRtcIsac_EncodeStoredDataLb(
+ const ISAC_SaveEncData_t* ISACSavedEnc_obj,
+ Bitstr* ISACBitStr_obj,
+ int BWnumber,
+ float scale);
+
+
+int WebRtcIsac_EncodeStoredDataUb12(
+ const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
+ Bitstr* bitStream,
+ WebRtc_Word32 jitterInfo,
+ float scale);
+
+int WebRtcIsac_EncodeStoredDataUb16(
+ const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
+ Bitstr* bitStream,
+ WebRtc_Word32 jitterInfo,
+ float scale);
+
+
+WebRtc_Word16 WebRtcIsac_GetRedPayloadUb(
+ const ISACUBSaveEncDataStruct* ISACSavedEncObj,
+ Bitstr* bitStreamObj,
+ enum ISACBandwidth bandwidth);
+/******************************************************************************
+ * WebRtcIsac_RateAllocation()
+ * Internal function to perform a rate-allocation for upper and lower-band,
+ * given a total rate.
+ *
+ * Input:
+ * - inRateBitPerSec : a total bit-rate in bits/sec.
+ *
+ * Output:
+ * - rateLBBitPerSec : a bit-rate allocated to the lower-band
+ * in bits/sec.
+ * - rateUBBitPerSec : a bit-rate allocated to the upper-band
+ * in bits/sec.
+ *
+ * Return value : 0 if rate allocation has been successful.
+ * -1 if failed to allocate rates.
+ */
+
+WebRtc_Word16
+WebRtcIsac_RateAllocation(
+ WebRtc_Word32 inRateBitPerSec,
+ double* rateLBBitPerSec,
+ double* rateUBBitPerSec,
+ enum ISACBandwidth* bandwidthKHz);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeUb16()
+ *
+ * Decode the upper-band if the codec is in 0-16 kHz mode.
+ *
+ * Input/Output:
+ * -ISACdec_obj : pointer to the upper-band decoder object. The
+ * bit-stream is stored inside the decoder object.
+ *
+ * Output:
+ * -signal_out : decoded audio, 480 samples 30 ms.
+ *
+ * Return value : >0 number of decoded bytes.
+ * <0 if an error occurred.
+ */
+int WebRtcIsac_DecodeUb16(
+ float* signal_out,
+ ISACUBDecStruct* ISACdec_obj,
+ WebRtc_Word16 isRCUPayload);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeUb12()
+ *
+ * Decode the upper-band if the codec is in 0-12 kHz mode.
+ *
+ * Input/Output:
+ * -ISACdec_obj : pointer to the upper-band decoder object. The
+ * bit-stream is stored inside the decoder object.
+ *
+ * Output:
+ * -signal_out : decoded audio, 480 samples 30 ms.
+ *
+ * Return value : >0 number of decoded bytes.
+ * <0 if an error occurred.
+ */
+int WebRtcIsac_DecodeUb12(
+ float* signal_out,
+ ISACUBDecStruct* ISACdec_obj,
+ WebRtc_Word16 isRCUPayload);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeUb16()
+ *
+ * Encode the upper-band if the codec is in 0-16 kHz mode.
+ *
+ * Input:
+ * -in : upper-band audio, 160 samples (10 ms).
+ *
+ * Input/Output:
+ * -ISACdec_obj : pointer to the upper-band encoder object. The
+ * bit-stream is stored inside the encoder object.
+ *
+ * Return value : >0 number of encoded bytes.
+ * <0 if an error occurred.
+ */
+int WebRtcIsac_EncodeUb16(
+ float* in,
+ ISACUBEncStruct* ISACenc_obj,
+ WebRtc_Word32 jitterInfo);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeUb12()
+ *
+ * Encode the upper-band if the codec is in 0-12 kHz mode.
+ *
+ * Input:
+ * -in : upper-band audio, 160 samples (10 ms).
+ *
+ * Input/Output:
+ * -ISACdec_obj : pointer to the upper-band encoder object. The
+ * bit-stream is stored inside the encoder object.
+ *
+ * Return value : >0 number of encoded bytes.
+ * <0 if an error occurred.
+ */
+int WebRtcIsac_EncodeUb12(
+ float* in,
+ ISACUBEncStruct* ISACenc_obj,
+ WebRtc_Word32 jitterInfo);
+
+/************************** initialization functions *************************/
+
+void WebRtcIsac_InitMasking(MaskFiltstr *maskdata);
+
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr *prefiltdata);
+
+void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata);
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr *pitchfiltdata);
+
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct *State);
+
+
+/**************************** transform functions ****************************/
+
+void WebRtcIsac_InitTransform();
+
+void WebRtcIsac_Time2Spec(double *inre1,
+ double *inre2,
+ WebRtc_Word16 *outre,
+ WebRtc_Word16 *outim,
+ FFTstr *fftstr_obj);
+
+void WebRtcIsac_Spec2time(double *inre,
+ double *inim,
+ double *outre1,
+ double *outre2,
+ FFTstr *fftstr_obj);
+
+
+/******************************* filter functions ****************************/
+
+void WebRtcIsac_AllPoleFilter(double *InOut,
+ double *Coef,
+ int lengthInOut,
+ int orderCoef);
+
+void WebRtcIsac_AllZeroFilter(double *In,
+ double *Coef,
+ int lengthInOut,
+ int orderCoef,
+ double *Out);
+
+void WebRtcIsac_ZeroPoleFilter(double *In,
+ double *ZeroCoef,
+ double *PoleCoef,
+ int lengthInOut,
+ int orderCoef,
+ double *Out);
+
+
+/***************************** filterbank functions **************************/
+
+void WebRtcIsac_SplitAndFilter(double *in,
+ double *LP,
+ double *HP,
+ double *LP_la,
+ double *HP_la,
+ PreFiltBankstr *prefiltdata);
+
+
+void WebRtcIsac_FilterAndCombine(double *InLP,
+ double *InHP,
+ double *Out,
+ PostFiltBankstr *postfiltdata);
+
+
+
+void WebRtcIsac_SplitAndFilterFloat(float *in,
+ float *LP,
+ float *HP,
+ double *LP_la,
+ double *HP_la,
+ PreFiltBankstr *prefiltdata);
+
+
+void WebRtcIsac_FilterAndCombineFloat(float *InLP,
+ float *InHP,
+ float *Out,
+ PostFiltBankstr *postfiltdata);
+
+
+/************************* normalized lattice filters ************************/
+
+void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
+ float *stateF,
+ float *stateG,
+ float *lat_in,
+ double *filtcoeflo,
+ double *lat_out);
+
+void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
+ float *stateF,
+ float *stateG,
+ double *lat_in,
+ double *lo_filt_coef,
+ float *lat_out);
+
+void WebRtcIsac_Dir2Lat(double *a,
+ int orderCoef,
+ float *sth,
+ float *cth);
+
+void WebRtcIsac_AutoCorr(double *r,
+ const double *x,
+ int N,
+ int order);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c
new file mode 100644
index 0000000..098e4b7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "crc.h"
+#include <stdlib.h>
+#include "signal_processing_library.h"
+
+#define POLYNOMIAL 0x04c11db7L
+
+
+static const WebRtc_UWord32 kCrcTable[256] = {
+ 0, 0x4c11db7, 0x9823b6e, 0xd4326d9, 0x130476dc, 0x17c56b6b,
+ 0x1a864db2, 0x1e475005, 0x2608edb8, 0x22c9f00f, 0x2f8ad6d6, 0x2b4bcb61,
+ 0x350c9b64, 0x31cd86d3, 0x3c8ea00a, 0x384fbdbd, 0x4c11db70, 0x48d0c6c7,
+ 0x4593e01e, 0x4152fda9, 0x5f15adac, 0x5bd4b01b, 0x569796c2, 0x52568b75,
+ 0x6a1936c8, 0x6ed82b7f, 0x639b0da6, 0x675a1011, 0x791d4014, 0x7ddc5da3,
+ 0x709f7b7a, 0x745e66cd, 0x9823b6e0, 0x9ce2ab57, 0x91a18d8e, 0x95609039,
+ 0x8b27c03c, 0x8fe6dd8b, 0x82a5fb52, 0x8664e6e5, 0xbe2b5b58, 0xbaea46ef,
+ 0xb7a96036, 0xb3687d81, 0xad2f2d84, 0xa9ee3033, 0xa4ad16ea, 0xa06c0b5d,
+ 0xd4326d90, 0xd0f37027, 0xddb056fe, 0xd9714b49, 0xc7361b4c, 0xc3f706fb,
+ 0xceb42022, 0xca753d95, 0xf23a8028, 0xf6fb9d9f, 0xfbb8bb46, 0xff79a6f1,
+ 0xe13ef6f4, 0xe5ffeb43, 0xe8bccd9a, 0xec7dd02d, 0x34867077, 0x30476dc0,
+ 0x3d044b19, 0x39c556ae, 0x278206ab, 0x23431b1c, 0x2e003dc5, 0x2ac12072,
+ 0x128e9dcf, 0x164f8078, 0x1b0ca6a1, 0x1fcdbb16, 0x18aeb13, 0x54bf6a4,
+ 0x808d07d, 0xcc9cdca, 0x7897ab07, 0x7c56b6b0, 0x71159069, 0x75d48dde,
+ 0x6b93dddb, 0x6f52c06c, 0x6211e6b5, 0x66d0fb02, 0x5e9f46bf, 0x5a5e5b08,
+ 0x571d7dd1, 0x53dc6066, 0x4d9b3063, 0x495a2dd4, 0x44190b0d, 0x40d816ba,
+ 0xaca5c697, 0xa864db20, 0xa527fdf9, 0xa1e6e04e, 0xbfa1b04b, 0xbb60adfc,
+ 0xb6238b25, 0xb2e29692, 0x8aad2b2f, 0x8e6c3698, 0x832f1041, 0x87ee0df6,
+ 0x99a95df3, 0x9d684044, 0x902b669d, 0x94ea7b2a, 0xe0b41de7, 0xe4750050,
+ 0xe9362689, 0xedf73b3e, 0xf3b06b3b, 0xf771768c, 0xfa325055, 0xfef34de2,
+ 0xc6bcf05f, 0xc27dede8, 0xcf3ecb31, 0xcbffd686, 0xd5b88683, 0xd1799b34,
+ 0xdc3abded, 0xd8fba05a, 0x690ce0ee, 0x6dcdfd59, 0x608edb80, 0x644fc637,
+ 0x7a089632, 0x7ec98b85, 0x738aad5c, 0x774bb0eb, 0x4f040d56, 0x4bc510e1,
+ 0x46863638, 0x42472b8f, 0x5c007b8a, 0x58c1663d, 0x558240e4, 0x51435d53,
+ 0x251d3b9e, 0x21dc2629, 0x2c9f00f0, 0x285e1d47, 0x36194d42, 0x32d850f5,
+ 0x3f9b762c, 0x3b5a6b9b, 0x315d626, 0x7d4cb91, 0xa97ed48, 0xe56f0ff,
+ 0x1011a0fa, 0x14d0bd4d, 0x19939b94, 0x1d528623, 0xf12f560e, 0xf5ee4bb9,
+ 0xf8ad6d60, 0xfc6c70d7, 0xe22b20d2, 0xe6ea3d65, 0xeba91bbc, 0xef68060b,
+ 0xd727bbb6, 0xd3e6a601, 0xdea580d8, 0xda649d6f, 0xc423cd6a, 0xc0e2d0dd,
+ 0xcda1f604, 0xc960ebb3, 0xbd3e8d7e, 0xb9ff90c9, 0xb4bcb610, 0xb07daba7,
+ 0xae3afba2, 0xaafbe615, 0xa7b8c0cc, 0xa379dd7b, 0x9b3660c6, 0x9ff77d71,
+ 0x92b45ba8, 0x9675461f, 0x8832161a, 0x8cf30bad, 0x81b02d74, 0x857130c3,
+ 0x5d8a9099, 0x594b8d2e, 0x5408abf7, 0x50c9b640, 0x4e8ee645, 0x4a4ffbf2,
+ 0x470cdd2b, 0x43cdc09c, 0x7b827d21, 0x7f436096, 0x7200464f, 0x76c15bf8,
+ 0x68860bfd, 0x6c47164a, 0x61043093, 0x65c52d24, 0x119b4be9, 0x155a565e,
+ 0x18197087, 0x1cd86d30, 0x29f3d35, 0x65e2082, 0xb1d065b, 0xfdc1bec,
+ 0x3793a651, 0x3352bbe6, 0x3e119d3f, 0x3ad08088, 0x2497d08d, 0x2056cd3a,
+ 0x2d15ebe3, 0x29d4f654, 0xc5a92679, 0xc1683bce, 0xcc2b1d17, 0xc8ea00a0,
+ 0xd6ad50a5, 0xd26c4d12, 0xdf2f6bcb, 0xdbee767c, 0xe3a1cbc1, 0xe760d676,
+ 0xea23f0af, 0xeee2ed18, 0xf0a5bd1d, 0xf464a0aa, 0xf9278673, 0xfde69bc4,
+ 0x89b8fd09, 0x8d79e0be, 0x803ac667, 0x84fbdbd0, 0x9abc8bd5, 0x9e7d9662,
+ 0x933eb0bb, 0x97ffad0c, 0xafb010b1, 0xab710d06, 0xa6322bdf, 0xa2f33668,
+ 0xbcb4666d, 0xb8757bda, 0xb5365d03, 0xb1f740b4
+};
+
+
+
+
+/****************************************************************************
+ * WebRtcIsac_GetCrc(...)
+ *
+ * This function returns a 32 bit CRC checksum of a bit stream
+ *
+ * Input:
+ * - bitstream : payload bitstream
+ * - len_bitstream_in_bytes : number of 8-bit words in the bit stream
+ *
+ * Output:
+ * - crc : checksum
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsac_GetCrc(const WebRtc_Word16* bitstream,
+ WebRtc_Word16 len_bitstream_in_bytes,
+ WebRtc_UWord32* crc)
+{
+ WebRtc_UWord8* bitstream_ptr_uw8;
+ WebRtc_UWord32 crc_state;
+ int byte_cntr;
+ int crc_tbl_indx;
+
+ /* Sanity Check. */
+ if (bitstream == NULL) {
+ return -1;
+ }
+ /* cast to UWord8 pointer */
+ bitstream_ptr_uw8 = (WebRtc_UWord8 *)bitstream;
+
+ /* initialize */
+ crc_state = 0xFFFFFFFF;
+
+ for (byte_cntr = 0; byte_cntr < len_bitstream_in_bytes; byte_cntr++) {
+ crc_tbl_indx = (WEBRTC_SPL_RSHIFT_U32(crc_state, 24) ^
+ bitstream_ptr_uw8[byte_cntr]) & 0xFF;
+ crc_state = WEBRTC_SPL_LSHIFT_U32(crc_state, 8) ^ kCrcTable[crc_tbl_indx];
+ }
+
+ *crc = ~crc_state;
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h
new file mode 100644
index 0000000..0151278
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * crc.h
+ *
+ * Checksum functions
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
+
+#include "typedefs.h"
+
+/****************************************************************************
+ * WebRtcIsac_GetCrc(...)
+ *
+ * This function returns a 32 bit CRC checksum of a bit stream
+ *
+ * Input:
+ * - encoded : payload bit stream
+ * - no_of_word8s : number of 8-bit words in the bit stream
+ *
+ * Output:
+ * - crc : checksum
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsac_GetCrc(
+ const WebRtc_Word16* encoded,
+ WebRtc_Word16 no_of_word8s,
+ WebRtc_UWord32* crc);
+
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c
new file mode 100644
index 0000000..25634b0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c
@@ -0,0 +1,330 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * decode_B.c
+ *
+ * This file contains definition of funtions for decoding.
+ * Decoding of lower-band, including normal-decoding and RCU decoding.
+ * Decoding of upper-band, including 8-12 kHz, when the bandwidth is
+ * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz.
+ *
+ */
+
+
+#include "codec.h"
+#include "entropy_coding.h"
+#include "pitch_estimator.h"
+#include "bandwidth_estimator.h"
+#include "structs.h"
+#include "settings.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+
+/*
+ * function to decode the bitstream
+ * returns the total number of bytes in the stream
+ */
+int
+WebRtcIsac_DecodeLb(
+ float* signal_out,
+ ISACLBDecStruct* ISACdecLB_obj,
+ WebRtc_Word16* current_framesamples,
+ WebRtc_Word16 isRCUPayload)
+{
+ int k, model;
+ int len, err;
+ WebRtc_Word16 bandwidthInd;
+
+ float LP_dec_float[FRAMESAMPLES_HALF];
+ float HP_dec_float[FRAMESAMPLES_HALF];
+
+ double LPw[FRAMESAMPLES_HALF];
+ double HPw[FRAMESAMPLES_HALF];
+ double LPw_pf[FRAMESAMPLES_HALF];
+
+ double lo_filt_coef[(ORDERLO+1)*SUBFRAMES];
+ double hi_filt_coef[(ORDERHI+1)*SUBFRAMES];
+
+ double real_f[FRAMESAMPLES_HALF];
+ double imag_f[FRAMESAMPLES_HALF];
+
+ double PitchLags[4];
+ double PitchGains[4];
+ double AvgPitchGain;
+ WebRtc_Word16 PitchGains_Q12[4];
+ WebRtc_Word16 AvgPitchGain_Q12;
+
+ float gain;
+
+ int frame_nb; /* counter */
+ int frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
+ int processed_samples;
+
+ (ISACdecLB_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
+ (ISACdecLB_obj->bitstr_obj).streamval = 0;
+ (ISACdecLB_obj->bitstr_obj).stream_index = 0;
+
+ len = 0;
+
+ /* decode framelength and BW estimation - not used,
+ only for stream pointer*/
+ err = WebRtcIsac_DecodeFrameLen(&ISACdecLB_obj->bitstr_obj,
+ current_framesamples);
+ if (err < 0) { // error check
+ return err;
+ }
+
+ /* frame_mode: 0, or 1 */
+ frame_mode = *current_framesamples/MAX_FRAMESAMPLES;
+ /* processed_samples: either 320 (20ms) or 480 (30, 60 ms) */
+ processed_samples = *current_framesamples/(frame_mode+1);
+
+ err = WebRtcIsac_DecodeSendBW(&ISACdecLB_obj->bitstr_obj, &bandwidthInd);
+ if (err < 0) { // error check
+ return err;
+ }
+
+ /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames
+ bundled together (60ms) */
+ for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
+ /* decode & dequantize pitch parameters */
+ err = WebRtcIsac_DecodePitchGain(&(ISACdecLB_obj->bitstr_obj),
+ PitchGains_Q12);
+ if (err < 0) { // error check
+ return err;
+ }
+
+ err = WebRtcIsac_DecodePitchLag(&ISACdecLB_obj->bitstr_obj,
+ PitchGains_Q12, PitchLags);
+ if (err < 0) { // error check
+ return err;
+ }
+
+ AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
+ PitchGains_Q12[2] + PitchGains_Q12[3])>>2;
+
+ /* decode & dequantize FiltCoef */
+ err = WebRtcIsac_DecodeLpc(&ISACdecLB_obj->bitstr_obj,
+ lo_filt_coef,hi_filt_coef, &model);
+ if (err < 0) { // error check
+ return err;
+ }
+ /* decode & dequantize spectrum */
+ len = WebRtcIsac_DecodeSpecLb(&ISACdecLB_obj->bitstr_obj,
+ real_f, imag_f, AvgPitchGain_Q12);
+ if (len < 0) { // error check
+ return len;
+ }
+
+ /* inverse transform */
+ WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw,
+ &ISACdecLB_obj->fftstr_obj);
+
+ /* convert PitchGains back to FLOAT for pitchfilter_post */
+ for (k = 0; k < 4; k++) {
+ PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
+ }
+
+ if(isRCUPayload)
+ {
+ for (k = 0; k < 240; k++) {
+ LPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
+ HPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
+ }
+ }
+
+ /* inverse pitch filter */
+ WebRtcIsac_PitchfilterPost(LPw, LPw_pf,
+ &ISACdecLB_obj->pitchfiltstr_obj, PitchLags, PitchGains);
+ /* convert AvgPitchGain back to FLOAT for computation of gain */
+ AvgPitchGain = ((float)AvgPitchGain_Q12)/4096;
+ gain = 1.0f - 0.45f * (float)AvgPitchGain;
+
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ /* reduce gain to compensate for pitch enhancer */
+ LPw_pf[ k ] *= gain;
+ }
+
+ if(isRCUPayload)
+ {
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ /* compensation for transcoding gain changes*/
+ LPw_pf[k] *= RCU_TRANSCODING_SCALE;
+ HPw[k] *= RCU_TRANSCODING_SCALE;
+ }
+ }
+
+ /* perceptual post-filtering (using normalized lattice filter) */
+ WebRtcIsac_NormLatticeFilterAr(ORDERLO,
+ ISACdecLB_obj->maskfiltstr_obj.PostStateLoF,
+ (ISACdecLB_obj->maskfiltstr_obj).PostStateLoG,
+ LPw_pf, lo_filt_coef, LP_dec_float);
+ WebRtcIsac_NormLatticeFilterAr(ORDERHI,
+ ISACdecLB_obj->maskfiltstr_obj.PostStateHiF,
+ (ISACdecLB_obj->maskfiltstr_obj).PostStateHiG,
+ HPw, hi_filt_coef, HP_dec_float);
+
+ /* recombine the 2 bands */
+ WebRtcIsac_FilterAndCombineFloat( LP_dec_float, HP_dec_float,
+ signal_out + frame_nb * processed_samples,
+ &ISACdecLB_obj->postfiltbankstr_obj);
+ }
+
+ return len;
+}
+
+
+/*
+ * This decode function is called when the codec is operating in 16 kHz
+ * bandwidth to decode the upperband, i.e. 8-16 kHz.
+ *
+ * Contrary to lower-band, the upper-band (8-16 kHz) is not split in
+ * frequency, but split to 12 sub-frames, i.e. twice as lower-band.
+ */
+int
+WebRtcIsac_DecodeUb16(
+ float* signal_out,
+ ISACUBDecStruct* ISACdecUB_obj,
+ WebRtc_Word16 isRCUPayload)
+{
+ int len, err;
+
+ double halfFrameFirst[FRAMESAMPLES_HALF];
+ double halfFrameSecond[FRAMESAMPLES_HALF];
+
+ double percepFilterParam[(UB_LPC_ORDER+1) * (SUBFRAMES<<1) +
+ (UB_LPC_ORDER+1)];
+
+ double real_f[FRAMESAMPLES_HALF];
+ double imag_f[FRAMESAMPLES_HALF];
+
+ len = 0;
+
+ /* decode & dequantize FiltCoef */
+ memset(percepFilterParam, 0, sizeof(percepFilterParam));
+ err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj,
+ percepFilterParam, isac16kHz);
+ if (err < 0) { // error check
+ return err;
+ }
+
+ /* decode & dequantize spectrum */
+ len = WebRtcIsac_DecodeSpecUB16(&ISACdecUB_obj->bitstr_obj, real_f,
+ imag_f);
+ if (len < 0) { // error check
+ return len;
+ }
+ if(isRCUPayload)
+ {
+ int n;
+ for(n = 0; n < 240; n++)
+ {
+ real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
+ imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
+ }
+ }
+
+ /* inverse transform */
+ WebRtcIsac_Spec2time(real_f, imag_f, halfFrameFirst, halfFrameSecond,
+ &ISACdecUB_obj->fftstr_obj);
+
+ /* perceptual post-filtering (using normalized lattice filter) */
+ WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
+ ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
+ (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameFirst,
+ &percepFilterParam[(UB_LPC_ORDER+1)], signal_out);
+
+ WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
+ ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
+ (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameSecond,
+ &percepFilterParam[(UB_LPC_ORDER + 1) * SUBFRAMES + (UB_LPC_ORDER+1)],
+ &signal_out[FRAMESAMPLES_HALF]);
+
+ return len;
+}
+
+/*
+ * This decode function is called when the codec operates at 0-12 kHz
+ * bandwidth to decode the upperband, i.e. 8-12 kHz.
+ *
+ * At the encoder the upper-band is split into two band, 8-12 kHz & 12-16
+ * kHz, and only 8-12 kHz is encoded. At the decoder, 8-12 kHz band is
+ * reconstructed and 12-16 kHz replaced with zeros. Then two bands
+ * are combined, to reconstruct the upperband 8-16 kHz.
+ */
+int
+WebRtcIsac_DecodeUb12(
+ float* signal_out,
+ ISACUBDecStruct* ISACdecUB_obj,
+ WebRtc_Word16 isRCUPayload)
+{
+ int len, err;
+
+ float LP_dec_float[FRAMESAMPLES_HALF];
+ float HP_dec_float[FRAMESAMPLES_HALF];
+
+ double LPw[FRAMESAMPLES_HALF];
+ double HPw[FRAMESAMPLES_HALF];
+
+ double percepFilterParam[(UB_LPC_ORDER+1)*SUBFRAMES];
+
+ double real_f[FRAMESAMPLES_HALF];
+ double imag_f[FRAMESAMPLES_HALF];
+
+ len = 0;
+
+ /* decode & dequantize FiltCoef */
+ err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj,
+ percepFilterParam, isac12kHz);
+ if(err < 0) { // error check
+ return err;
+ }
+
+ /* decode & dequantize spectrum */
+ len = WebRtcIsac_DecodeSpecUB12(&ISACdecUB_obj->bitstr_obj,
+ real_f, imag_f);
+ if(len < 0) { // error check
+ return len;
+ }
+
+ if(isRCUPayload)
+ {
+ int n;
+ for(n = 0; n < 240; n++)
+ {
+ real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
+ imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
+ }
+ }
+
+ /* inverse transform */
+ WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw, &ISACdecUB_obj->fftstr_obj);
+
+ /* perceptual post-filtering (using normalized lattice filter) */
+ WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
+ ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
+ (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, LPw,
+ percepFilterParam, LP_dec_float);
+
+ /* Zerro for upper-band */
+ memset(HP_dec_float, 0, sizeof(float) * (FRAMESAMPLES_HALF));
+
+ /* recombine the 2 bands */
+ WebRtcIsac_FilterAndCombineFloat(HP_dec_float, LP_dec_float, signal_out,
+ &ISACdecUB_obj->postfiltbankstr_obj);
+
+
+
+ return len;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
new file mode 100644
index 0000000..cdac7fa
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "structs.h"
+#include "bandwidth_estimator.h"
+#include "entropy_coding.h"
+#include "codec.h"
+
+
+int
+WebRtcIsac_EstimateBandwidth(
+ BwEstimatorstr* bwest_str,
+ Bitstr* streamdata,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts,
+ enum IsacSamplingRate encoderSampRate,
+ enum IsacSamplingRate decoderSampRate)
+{
+ WebRtc_Word16 index;
+ WebRtc_Word16 frame_samples;
+ WebRtc_UWord32 sendTimestampIn16kHz;
+ WebRtc_UWord32 arrivalTimestampIn16kHz;
+ WebRtc_UWord32 diffSendTime;
+ WebRtc_UWord32 diffArrivalTime;
+ int err;
+
+ /* decode framelength and BW estimation */
+ err = WebRtcIsac_DecodeFrameLen(streamdata, &frame_samples);
+ if(err < 0) // error check
+ {
+ return err;
+ }
+ err = WebRtcIsac_DecodeSendBW(streamdata, &index);
+ if(err < 0) // error check
+ {
+ return err;
+ }
+
+ /* UPDATE ESTIMATES FROM OTHER SIDE */
+ err = WebRtcIsac_UpdateUplinkBwImpl(bwest_str, index, encoderSampRate);
+ if(err < 0)
+ {
+ return err;
+ }
+
+ // We like BWE to work at 16 kHz sampling rate,
+ // therefore, we have to change the timestamps accordingly.
+ // translate the send timestamp if required
+ diffSendTime = (WebRtc_UWord32)((WebRtc_UWord32)send_ts -
+ (WebRtc_UWord32)bwest_str->senderTimestamp);
+ bwest_str->senderTimestamp = send_ts;
+
+ diffArrivalTime = (WebRtc_UWord32)((WebRtc_UWord32)arr_ts -
+ (WebRtc_UWord32)bwest_str->receiverTimestamp);
+ bwest_str->receiverTimestamp = arr_ts;
+
+ if(decoderSampRate == kIsacSuperWideband)
+ {
+ diffArrivalTime = (WebRtc_UWord32)diffArrivalTime >> 1;
+ diffSendTime = (WebRtc_UWord32)diffSendTime >> 1;
+ }
+ // arrival timestamp in 16 kHz
+ arrivalTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
+ bwest_str->prev_rec_arr_ts + (WebRtc_UWord32)diffArrivalTime);
+ // send timestamp in 16 kHz
+ sendTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
+ bwest_str->prev_rec_send_ts + (WebRtc_UWord32)diffSendTime);
+
+ err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number,
+ (frame_samples * 1000) / FS, sendTimestampIn16kHz,
+ arrivalTimestampIn16kHz, packet_size);
+ // error check
+ if(err < 0)
+ {
+ return err;
+ }
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c
new file mode 100644
index 0000000..75cd726
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c
@@ -0,0 +1,1451 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * encode.c
+ *
+ * This file contains definition of funtions for encoding.
+ * Decoding of upper-band, including 8-12 kHz, when the bandwidth is
+ * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz.
+ *
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "structs.h"
+#include "codec.h"
+#include "pitch_estimator.h"
+#include "entropy_coding.h"
+#include "arith_routines.h"
+#include "pitch_gain_tables.h"
+#include "pitch_lag_tables.h"
+#include "spectrum_ar_model_tables.h"
+#include "lpc_tables.h"
+#include "lpc_analysis.h"
+#include "bandwidth_estimator.h"
+#include "lpc_shape_swb12_tables.h"
+#include "lpc_shape_swb16_tables.h"
+#include "lpc_gain_swb_tables.h"
+
+
+#define UB_LOOKAHEAD 24
+
+/*
+ Rate allocation tables of lower and upper-band bottleneck for
+ 12kHz & 16kHz bandwidth.
+
+ 12 kHz bandwidth
+ -----------------
+ The overall bottleneck of the coder is between 38 kbps and 45 kbps. We have
+ considered 7 enteries, uniformly distributed in this interval, i.e. 38,
+ 39.17, 40.33, 41.5, 42.67, 43.83 and 45. For every entery, the lower-band
+ and the upper-band bottlenecks are specified in
+ 'kLowerBandBitRate12' and 'kUpperBandBitRate12'
+ tables, respectively. E.g. the overall rate of 41.5 kbps corresponts to a
+ bottleneck of 31 kbps for lower-band and 27 kbps for upper-band. Given an
+ overall bottleneck of the codec, we use linear interpolation to get
+ lower-band and upper-band bottlenecks.
+
+ 16 kHz bandwidth
+ -----------------
+ The overall bottleneck of the coder is between 50 kbps and 56 kbps. We have
+ considered 7 enteries, uniformly distributed in this interval, i.e. 50, 51.2,
+ 52.4, 53.6, 54.8 and 56. For every entery, the lower-band and the upper-band
+ bottlenecks are specified in 'kLowerBandBitRate16' and
+ 'kUpperBandBitRate16' tables, respectively. E.g. the overall rate
+ of 53.6 kbps corresponts to a bottleneck of 32 kbps for lower-band and 30
+ kbps for upper-band. Given an overall bottleneck of the codec, we use linear
+ interpolation to get lower-band and upper-band bottlenecks.
+
+*/
+
+// 38 39.17 40.33 41.5 42.67 43.83 45
+static const WebRtc_Word16 kLowerBandBitRate12[7] = {
+ 29000, 30000, 30000, 31000, 31000, 32000, 32000};
+static const WebRtc_Word16 kUpperBandBitRate12[7] = {
+ 25000, 25000, 27000, 27000, 29000, 29000, 32000};
+
+// 50 51.2 52.4 53.6 54.8 56
+static const WebRtc_Word16 kLowerBandBitRate16[6] = {
+ 31000, 31000, 32000, 32000, 32000, 32000};
+static const WebRtc_Word16 kUpperBandBitRate16[6] = {
+ 28000, 29000, 29000, 30000, 31000, 32000};
+
+/******************************************************************************
+ * WebRtcIsac_RateAllocation()
+ * Internal function to perform a rate-allocation for upper and lower-band,
+ * given a total rate.
+ *
+ * Input:
+ * - inRateBitPerSec : a total bottleneck in bits/sec.
+ *
+ * Output:
+ * - rateLBBitPerSec : a bottleneck allocated to the lower-band
+ * in bits/sec.
+ * - rateUBBitPerSec : a bottleneck allocated to the upper-band
+ * in bits/sec.
+ *
+ * Return value : 0 if rate allocation has been successful.
+ * -1 if failed to allocate rates.
+ */
+
+WebRtc_Word16
+WebRtcIsac_RateAllocation(
+ WebRtc_Word32 inRateBitPerSec,
+ double* rateLBBitPerSec,
+ double* rateUBBitPerSec,
+ enum ISACBandwidth* bandwidthKHz)
+{
+ WebRtc_Word16 idx;
+ double idxD;
+ double idxErr;
+ if(inRateBitPerSec < 38000)
+ {
+ // If the given overall bottleneck is less than 38000 then
+ // then codec has to operate in wideband mode, i.e. 8 kHz
+ // bandwidth.
+ *rateLBBitPerSec = (WebRtc_Word16)((inRateBitPerSec > 32000)?
+ 32000:inRateBitPerSec);
+ *rateUBBitPerSec = 0;
+ *bandwidthKHz = isac8kHz;
+ }
+ else if((inRateBitPerSec >= 38000) && (inRateBitPerSec < 50000))
+ {
+ // At a bottleneck between 38 and 50 kbps the codec is operating
+ // at 12 kHz bandwidth. Using xxxBandBitRate12[] to calculates
+ // upper/lower bottleneck
+
+ // find the bottlenecks by linear interpolation
+ // step is (45000 - 38000)/6.0 we use the inverse of it.
+ const double stepSizeInv = 8.5714286e-4;
+ idxD = (inRateBitPerSec - 38000) * stepSizeInv;
+ idx = (idxD >= 6)? 6:((WebRtc_Word16)idxD);
+ idxErr = idxD - idx;
+ *rateLBBitPerSec = kLowerBandBitRate12[idx];
+ *rateUBBitPerSec = kUpperBandBitRate12[idx];
+
+ if(idx < 6)
+ {
+ *rateLBBitPerSec += (WebRtc_Word16)(idxErr *
+ (kLowerBandBitRate12[idx + 1] -
+ kLowerBandBitRate12[idx]));
+ *rateUBBitPerSec += (WebRtc_Word16)(idxErr *
+ (kUpperBandBitRate12[idx + 1] -
+ kUpperBandBitRate12[idx]));
+ }
+
+ *bandwidthKHz = isac12kHz;
+ }
+ else if((inRateBitPerSec >= 50000) && (inRateBitPerSec <= 56000))
+ {
+ // A bottleneck between 50 and 56 kbps corresponds to bandwidth
+ // of 16 kHz. Using xxxBandBitRate16[] to calculates
+ // upper/lower bottleneck
+
+ // find the bottlenecks by linear interpolation
+ // step is (56000 - 50000)/5 we use the inverse of it
+ const double stepSizeInv = 8.3333333e-4;
+ idxD = (inRateBitPerSec - 50000) * stepSizeInv;
+ idx = (idxD >= 5)? 5:((WebRtc_Word16)idxD);
+ idxErr = idxD - idx;
+ *rateLBBitPerSec = kLowerBandBitRate16[idx];
+ *rateUBBitPerSec = kUpperBandBitRate16[idx];
+
+ if(idx < 5)
+ {
+ *rateLBBitPerSec += (WebRtc_Word16)(idxErr *
+ (kLowerBandBitRate16[idx + 1] -
+ kLowerBandBitRate16[idx]));
+
+ *rateUBBitPerSec += (WebRtc_Word16)(idxErr *
+ (kUpperBandBitRate16[idx + 1] -
+ kUpperBandBitRate16[idx]));
+ }
+
+ *bandwidthKHz = isac16kHz;
+ }
+ else
+ {
+ // Out-of-range botlteneck value.
+ return -1;
+ }
+
+ // limit the values.
+ *rateLBBitPerSec = (*rateLBBitPerSec > 32000)? 32000:*rateLBBitPerSec;
+ *rateUBBitPerSec = (*rateUBBitPerSec > 32000)? 32000:*rateUBBitPerSec;
+
+ return 0;
+}
+
+
+
+int
+WebRtcIsac_EncodeLb(
+ float* in,
+ ISACLBEncStruct* ISACencLB_obj,
+ WebRtc_Word16 codingMode,
+ WebRtc_Word16 bottleneckIndex)
+{
+ int stream_length = 0;
+ int err;
+ int k;
+ int iterCntr;
+
+ double lofilt_coef[(ORDERLO+1)*SUBFRAMES];
+ double hifilt_coef[(ORDERHI+1)*SUBFRAMES];
+ float LP[FRAMESAMPLES_HALF];
+ float HP[FRAMESAMPLES_HALF];
+
+ double LP_lookahead[FRAMESAMPLES_HALF];
+ double HP_lookahead[FRAMESAMPLES_HALF];
+ double LP_lookahead_pf[FRAMESAMPLES_HALF + QLOOKAHEAD];
+ double LPw[FRAMESAMPLES_HALF];
+
+ double HPw[FRAMESAMPLES_HALF];
+ double LPw_pf[FRAMESAMPLES_HALF];
+ WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
+ WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
+
+ double PitchLags[4];
+ double PitchGains[4];
+ WebRtc_Word16 PitchGains_Q12[4];
+ WebRtc_Word16 AvgPitchGain_Q12;
+
+ int frame_mode; /* 0 for 30ms, 1 for 60ms */
+ int processed_samples, status = 0;
+
+ double bits_gains;
+ int bmodel;
+
+ transcode_obj transcodingParam;
+ double bytesLeftSpecCoding;
+ WebRtc_UWord16 payloadLimitBytes;
+
+ /* copy new frame length and bottle neck rate only for the first
+ 10 ms data */
+ if (ISACencLB_obj->buffer_index == 0) {
+ /* set the framelength for the next packet */
+ ISACencLB_obj->current_framesamples = ISACencLB_obj->new_framelength;
+ }
+ /* frame_mode is 0 (30 ms) or 1 (60 ms) */
+ frame_mode = ISACencLB_obj->current_framesamples/MAX_FRAMESAMPLES;
+ /* processed_samples: 480 (30, 60 ms) */
+ processed_samples = ISACencLB_obj->current_framesamples/(frame_mode+1);
+
+ /* buffer speech samples (by 10ms packet) until the framelength */
+ /* is reached (30 or 60 ms) */
+ /****************************************************************/
+
+ /* fill the buffer with 10ms input data */
+ for (k = 0; k < FRAMESAMPLES_10ms; k++) {
+ ISACencLB_obj->data_buffer_float[k + ISACencLB_obj->buffer_index] =
+ in[k];
+ }
+
+ /* if buffersize is not equal to current framesize then increase index
+ and return. We do no encoding untill we have enough audio. */
+ if (ISACencLB_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) {
+ ISACencLB_obj->buffer_index += FRAMESAMPLES_10ms;
+ return 0;
+ }
+ /* if buffer reached the right size, reset index and continue with
+ encoding the frame */
+ ISACencLB_obj->buffer_index = 0;
+
+ /* end of buffer function */
+ /**************************/
+
+ /* encoding */
+ /************/
+
+ if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0 ) {
+ // This is to avoid Linux warnings until we change 'int' to 'Word32'
+ // at all places.
+ int intVar;
+ /* reset bitstream */
+ ISACencLB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
+ ISACencLB_obj->bitstr_obj.streamval = 0;
+ ISACencLB_obj->bitstr_obj.stream_index = 0;
+
+ if((codingMode == 0) && (frame_mode == 0) &&
+ (ISACencLB_obj->enforceFrameSize == 0)) {
+ ISACencLB_obj->new_framelength =
+ WebRtcIsac_GetNewFrameLength(ISACencLB_obj->bottleneck,
+ ISACencLB_obj->current_framesamples);
+ }
+
+ ISACencLB_obj->s2nr = WebRtcIsac_GetSnr(
+ ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples);
+
+ /* encode frame length */
+ status = WebRtcIsac_EncodeFrameLen(
+ ISACencLB_obj->current_framesamples, &ISACencLB_obj->bitstr_obj);
+ if (status < 0) {
+ /* Wrong frame size */
+ return status;
+ }
+ /* Save framelength for multiple packets memory */
+ ISACencLB_obj->SaveEnc_obj.framelength =
+ ISACencLB_obj->current_framesamples;
+
+ /* To be used for Redundant Coding */
+ ISACencLB_obj->lastBWIdx = bottleneckIndex;
+ intVar = (int)bottleneckIndex;
+ WebRtcIsac_EncodeReceiveBw(&intVar, &ISACencLB_obj->bitstr_obj);
+ }
+
+ /* split signal in two bands */
+ WebRtcIsac_SplitAndFilterFloat(ISACencLB_obj->data_buffer_float, LP, HP,
+ LP_lookahead, HP_lookahead, &ISACencLB_obj->prefiltbankstr_obj );
+
+ /* estimate pitch parameters and pitch-filter lookahead signal */
+ WebRtcIsac_PitchAnalysis(LP_lookahead, LP_lookahead_pf,
+ &ISACencLB_obj->pitchanalysisstr_obj, PitchLags, PitchGains);
+
+ /* encode in FIX Q12 */
+
+ /* convert PitchGain to Fixed point */
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+ PitchGains_Q12[k] = (WebRtc_Word16)(PitchGains[k] * 4096.0);
+ }
+
+ /* Set where to store data in multiple packets memory */
+ if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0)
+ {
+ ISACencLB_obj->SaveEnc_obj.startIdx = 0;
+ } else {
+ ISACencLB_obj->SaveEnc_obj.startIdx = 1;
+ }
+
+ /* quantize & encode pitch parameters */
+ WebRtcIsac_EncodePitchGain(PitchGains_Q12, &ISACencLB_obj->bitstr_obj,
+ &ISACencLB_obj->SaveEnc_obj);
+ WebRtcIsac_EncodePitchLag(PitchLags, PitchGains_Q12,
+ &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
+
+ AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
+ PitchGains_Q12[2] + PitchGains_Q12[3])>>2;
+
+ /* find coefficients for perceptual pre-filters */
+ WebRtcIsac_GetLpcCoefLb(LP_lookahead_pf, HP_lookahead,
+ &ISACencLB_obj->maskfiltstr_obj, ISACencLB_obj->s2nr,
+ PitchGains_Q12, lofilt_coef, hifilt_coef);
+
+ /* code LPC model and shape - gains not quantized yet */
+ WebRtcIsac_EncodeLpcLb(lofilt_coef, hifilt_coef, &bmodel, &bits_gains,
+ &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
+
+ /* convert PitchGains back to FLOAT for pitchfilter_pre */
+ for (k = 0; k < 4; k++) {
+ PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
+ }
+
+ /* Store the state of arithmetic coder before coding LPC gains */
+ transcodingParam.W_upper = ISACencLB_obj->bitstr_obj.W_upper;
+ transcodingParam.stream_index = ISACencLB_obj->bitstr_obj.stream_index;
+ transcodingParam.streamval = ISACencLB_obj->bitstr_obj.streamval;
+ transcodingParam.stream[0] = ISACencLB_obj->bitstr_obj.stream[
+ ISACencLB_obj->bitstr_obj.stream_index - 2];
+ transcodingParam.stream[1] = ISACencLB_obj->bitstr_obj.stream[
+ ISACencLB_obj->bitstr_obj.stream_index - 1];
+ transcodingParam.stream[2] = ISACencLB_obj->bitstr_obj.stream[
+ ISACencLB_obj->bitstr_obj.stream_index];
+
+ /* Store LPC Gains before encoding them */
+ for(k = 0; k < SUBFRAMES; k++) {
+ transcodingParam.loFiltGain[k] = lofilt_coef[(LPC_LOBAND_ORDER+1)*k];
+ transcodingParam.hiFiltGain[k] = hifilt_coef[(LPC_HIBAND_ORDER+1)*k];
+ }
+
+ /* Code gains */
+ WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, bmodel,
+ &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
+
+ /* Get the correct value for the payload limit and calculate the
+ number of bytes left for coding the spectrum.*/
+ if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) {
+ /* It is a 60ms and we are in the first 30ms then the limit at
+ this point should be half of the assigned value */
+ payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 >> 1;
+ }
+ else if (frame_mode == 0) {
+ /* It is a 30ms frame */
+ /* Subract 3 because termination process may add 3 bytes */
+ payloadLimitBytes = ISACencLB_obj->payloadLimitBytes30 - 3;
+ } else {
+ /* This is the second half of a 60ms frame. */
+ /* Subract 3 because termination process may add 3 bytes */
+ payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 - 3;
+ }
+ bytesLeftSpecCoding = payloadLimitBytes - transcodingParam.stream_index;
+
+ /* perceptual pre-filtering (using normalized lattice filter) */
+ /* low-band filtering */
+ WebRtcIsac_NormLatticeFilterMa(ORDERLO,
+ ISACencLB_obj->maskfiltstr_obj.PreStateLoF,
+ ISACencLB_obj->maskfiltstr_obj.PreStateLoG, LP, lofilt_coef, LPw);
+ /* high-band filtering */
+ WebRtcIsac_NormLatticeFilterMa(ORDERHI,
+ ISACencLB_obj->maskfiltstr_obj.PreStateHiF,
+ ISACencLB_obj->maskfiltstr_obj.PreStateHiG, HP, hifilt_coef, HPw);
+
+
+ /* pitch filter */
+ WebRtcIsac_PitchfilterPre(LPw, LPw_pf, &ISACencLB_obj->pitchfiltstr_obj,
+ PitchLags, PitchGains);
+
+ /* transform */
+ WebRtcIsac_Time2Spec(LPw_pf, HPw, fre, fim, &ISACencLB_obj->fftstr_obj);
+
+
+ /* Save data for multiple packets memory */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ ISACencLB_obj->SaveEnc_obj.fre[k +
+ ISACencLB_obj->SaveEnc_obj.startIdx*FRAMESAMPLES_HALF] = fre[k];
+ ISACencLB_obj->SaveEnc_obj.fim[k +
+ ISACencLB_obj->SaveEnc_obj.startIdx*FRAMESAMPLES_HALF] = fim[k];
+ }
+ ISACencLB_obj->SaveEnc_obj.AvgPitchGain[
+ ISACencLB_obj->SaveEnc_obj.startIdx] = AvgPitchGain_Q12;
+
+ /* quantization and lossless coding */
+ err = WebRtcIsac_EncodeSpecLb(fre, fim, &ISACencLB_obj->bitstr_obj,
+ AvgPitchGain_Q12);
+ if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ /* There has been an error but it was not too large payload
+ (we can cure too large payload) */
+ if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) {
+ /* If this is the second 30ms of a 60ms frame reset
+ this such that in the next call encoder starts fresh. */
+ ISACencLB_obj->frame_nb = 0;
+ }
+ return err;
+ }
+ iterCntr = 0;
+ while((ISACencLB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
+ (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ double bytesSpecCoderUsed;
+ double transcodeScale;
+
+ if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
+ /* We were not able to limit the payload size */
+ if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) {
+ /* This was the first 30ms of a 60ms frame. Although
+ the payload is larger than it should be but we let
+ the second 30ms be encoded. Maybe together we
+ won't exceed the limit. */
+ ISACencLB_obj->frame_nb = 1;
+ return 0;
+ } else if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 1)) {
+ ISACencLB_obj->frame_nb = 0;
+ }
+
+ if(err != -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
+ return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
+ } else {
+ return status;
+ }
+ }
+
+ if(err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
+ bytesSpecCoderUsed = STREAM_SIZE_MAX;
+ // being coservative
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
+ } else {
+ bytesSpecCoderUsed = ISACencLB_obj->bitstr_obj.stream_index -
+ transcodingParam.stream_index;
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
+ }
+
+ /* To be safe, we reduce the scale depending on
+ the number of iterations. */
+ transcodeScale *= (1.0 - (0.9 * (double)iterCntr /
+ (double)MAX_PAYLOAD_LIMIT_ITERATION));
+
+ /* Scale the LPC Gains */
+ for (k = 0; k < SUBFRAMES; k++) {
+ lofilt_coef[(LPC_LOBAND_ORDER+1) * k] =
+ transcodingParam.loFiltGain[k] * transcodeScale;
+ hifilt_coef[(LPC_HIBAND_ORDER+1) * k] =
+ transcodingParam.hiFiltGain[k] * transcodeScale;
+ transcodingParam.loFiltGain[k] =
+ lofilt_coef[(LPC_LOBAND_ORDER+1) * k];
+ transcodingParam.hiFiltGain[k] =
+ hifilt_coef[(LPC_HIBAND_ORDER+1) * k];
+ }
+
+ /* Scale DFT coefficients */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale);
+ fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale);
+ }
+
+ /* Save data for multiple packets memory */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ ISACencLB_obj->SaveEnc_obj.fre[k +
+ ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF] =
+ fre[k];
+ ISACencLB_obj->SaveEnc_obj.fim[k +
+ ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF] =
+ fim[k];
+ }
+
+ /* Re-store the state of arithmetic coder before coding LPC gains */
+ ISACencLB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
+ ISACencLB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
+ ISACencLB_obj->bitstr_obj.streamval = transcodingParam.streamval;
+ ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
+ transcodingParam.stream[0];
+ ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
+ transcodingParam.stream[1];
+ ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
+ transcodingParam.stream[2];
+
+ /* Code gains */
+ WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, bmodel,
+ &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
+
+ /* Update the number of bytes left for encoding the spectrum */
+ bytesLeftSpecCoding = payloadLimitBytes -
+ transcodingParam.stream_index;
+
+ /* Encode the spectrum */
+ err = WebRtcIsac_EncodeSpecLb(fre, fim, &ISACencLB_obj->bitstr_obj,
+ AvgPitchGain_Q12);
+ if((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ /* There has been an error but it was not too large
+ payload (we can cure too large payload) */
+ if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) {
+ /* If this is the second 30ms of a 60ms frame reset
+ this such that in the next call encoder starts fresh. */
+ ISACencLB_obj->frame_nb = 0;
+ }
+ return err;
+ }
+ iterCntr++;
+ }
+
+ /* i.e. 60 ms framesize and just processed the first 30ms, */
+ /* go back to main function to buffer the other 30ms speech frame */
+ if (frame_mode == 1)
+ {
+ if(ISACencLB_obj->frame_nb == 0)
+ {
+ ISACencLB_obj->frame_nb = 1;
+ return 0;
+ }
+ else if(ISACencLB_obj->frame_nb == 1)
+ {
+ ISACencLB_obj->frame_nb = 0;
+ /* also update the framelength for next packet,
+ in Adaptive mode only */
+ if (codingMode == 0 && (ISACencLB_obj->enforceFrameSize == 0))
+ {
+ ISACencLB_obj->new_framelength =
+ WebRtcIsac_GetNewFrameLength(ISACencLB_obj->bottleneck,
+ ISACencLB_obj->current_framesamples);
+ }
+ }
+ }
+ else
+ {
+ ISACencLB_obj->frame_nb = 0;
+ }
+
+ /* complete arithmetic coding */
+ stream_length = WebRtcIsac_EncTerminate(&ISACencLB_obj->bitstr_obj);
+
+ return stream_length;
+}
+
+int
+WebRtcIsac_EncodeUb16(
+ float* in,
+ ISACUBEncStruct* ISACencUB_obj,
+ WebRtc_Word32 jitterInfo)
+{
+ int err;
+ int k;
+
+ double lpcVecs[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ double percepFilterParams[(1 + UB_LPC_ORDER) * (SUBFRAMES<<1) +
+ (1 + UB_LPC_ORDER)];
+
+ double LP_lookahead[FRAMESAMPLES];
+ WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
+ WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
+
+ int status = 0;
+
+ double varscale[2];
+ double corr[SUBFRAMES<<1][UB_LPC_ORDER + 1];
+ double lpcGains[SUBFRAMES<<1];
+ transcode_obj transcodingParam;
+ double bytesLeftSpecCoding;
+ WebRtc_UWord16 payloadLimitBytes;
+ WebRtc_UWord16 iterCntr;
+ double s2nr;
+
+ /* buffer speech samples (by 10ms packet) until the framelength is */
+ /* reached (30 or 60 ms) */
+ /*********************************************************************/
+
+ /* fill the buffer with 10ms input data */
+ for (k = 0; k < FRAMESAMPLES_10ms; k++) {
+ ISACencUB_obj->data_buffer_float[k + ISACencUB_obj->buffer_index] =
+ in[k];
+ }
+
+ /* if buffersize is not equal to current framesize, and end of file is
+ not reached yet, we don't do encoding unless we have the whole frame */
+ if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) {
+ ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms;
+ return 0;
+ }
+
+ /* end of buffer function */
+ /**************************/
+
+ /* encoding */
+ /************/
+
+ /* reset bitstream */
+ ISACencUB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
+ ISACencUB_obj->bitstr_obj.streamval = 0;
+ ISACencUB_obj->bitstr_obj.stream_index = 0;
+
+ /* bandwidth estimation and coding */
+ /* To be used for Redundant Coding */
+ WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj);
+
+ status = WebRtcIsac_EncodeBandwidth(isac16kHz,
+ &ISACencUB_obj->bitstr_obj);
+ if (status < 0) {
+ return status;
+ }
+
+ s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck,
+ FRAMESAMPLES);
+
+ memcpy(lpcVecs, ISACencUB_obj->lastLPCVec, UB_LPC_ORDER * sizeof(double));
+
+ for (k = 0; k < FRAMESAMPLES; k++) {
+ LP_lookahead[k] = ISACencUB_obj->data_buffer_float[UB_LOOKAHEAD + k];
+ }
+
+ /* find coefficients for perceptual pre-filters */
+ WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj,
+ &lpcVecs[UB_LPC_ORDER], corr, varscale, isac16kHz);
+
+ memcpy(ISACencUB_obj->lastLPCVec,
+ &lpcVecs[(UB16_LPC_VEC_PER_FRAME - 1) * (UB_LPC_ORDER)],
+ sizeof(double) * UB_LPC_ORDER);
+
+ /* code LPC model and shape - gains not quantized yet */
+ WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj,
+ percepFilterParams, isac16kHz, &ISACencUB_obj->SaveEnc_obj);
+
+
+ // the first set of lpc parameters are from the last sub-frame of
+ // the previous frame. so we don't care about them
+ WebRtcIsac_GetLpcGain(s2nr, &percepFilterParams[UB_LPC_ORDER + 1],
+ (SUBFRAMES<<1), lpcGains, corr, varscale);
+
+ /* Store the state of arithmetic coder before coding LPC gains */
+ transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index;
+ transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper;
+ transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval;
+ transcodingParam.stream[0] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index - 2];
+ transcodingParam.stream[1] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index - 1];
+ transcodingParam.stream[2] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index];
+
+ /* Store LPC Gains before encoding them */
+ for(k = 0; k < SUBFRAMES; k++) {
+ transcodingParam.loFiltGain[k] = lpcGains[k];
+ transcodingParam.hiFiltGain[k] = lpcGains[SUBFRAMES + k];
+ }
+
+ // Store the gains for multiple encoding
+ memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, (SUBFRAMES << 1) * sizeof(double));
+
+ WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj,
+ ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
+ WebRtcIsac_EncodeLpcGainUb(&lpcGains[SUBFRAMES], &ISACencUB_obj->bitstr_obj,
+ &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]);
+
+ /* Get the correct value for the payload limit and calculate the number of
+ bytes left for coding the spectrum. It is a 30ms frame
+ Subract 3 because termination process may add 3 bytes */
+ payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes -
+ ISACencUB_obj->numBytesUsed - 3;
+ bytesLeftSpecCoding = payloadLimitBytes -
+ ISACencUB_obj->bitstr_obj.stream_index;
+
+ for (k = 0; k < (SUBFRAMES<<1); k++) {
+ percepFilterParams[k*(UB_LPC_ORDER + 1) + (UB_LPC_ORDER + 1)] =
+ lpcGains[k];
+ }
+
+ /* perceptual pre-filtering (using normalized lattice filter) */
+ /* first half-frame filtering */
+ WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoG,
+ &ISACencUB_obj->data_buffer_float[0],
+ &percepFilterParams[UB_LPC_ORDER + 1],
+ &LP_lookahead[0]);
+
+ /* Second half-frame filtering */
+ WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoG,
+ &ISACencUB_obj->data_buffer_float[FRAMESAMPLES_HALF],
+ &percepFilterParams[(UB_LPC_ORDER + 1) + SUBFRAMES *
+ (UB_LPC_ORDER + 1)], &LP_lookahead[FRAMESAMPLES_HALF]);
+
+ WebRtcIsac_Time2Spec(&LP_lookahead[0], &LP_lookahead[FRAMESAMPLES_HALF],
+ fre, fim, &ISACencUB_obj->fftstr_obj);
+
+ //Store FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ // Prepare the audio buffer for the next packet
+ // move the last 3 ms to the beginning of the buffer
+ memcpy(ISACencUB_obj->data_buffer_float,
+ &ISACencUB_obj->data_buffer_float[FRAMESAMPLES],
+ LB_TOTAL_DELAY_SAMPLES * sizeof(float));
+ // start writing with 3 ms delay to compensate for the delay
+ // of the lower-band.
+ ISACencUB_obj->buffer_index = LB_TOTAL_DELAY_SAMPLES;
+
+ // Save the bit-stream object at this point for FEC.
+ memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
+ &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
+
+ /* quantization and lossless coding */
+ err = WebRtcIsac_EncodeSpecUB16(fre, fim, &ISACencUB_obj->bitstr_obj);
+ if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ return err;
+ }
+
+ iterCntr = 0;
+ while((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
+ (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ double bytesSpecCoderUsed;
+ double transcodeScale;
+
+ if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
+ /* We were not able to limit the payload size */
+ return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
+ }
+
+ if (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
+ bytesSpecCoderUsed = STREAM_SIZE_MAX;
+ // being conservative
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
+ } else {
+ bytesSpecCoderUsed = ISACencUB_obj->bitstr_obj.stream_index -
+ transcodingParam.stream_index;
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
+ }
+
+ /* To be safe, we reduce the scale depending on the
+ number of iterations. */
+ transcodeScale *= (1.0 - (0.9 * (double)iterCntr/
+ (double)MAX_PAYLOAD_LIMIT_ITERATION));
+
+ /* Scale the LPC Gains */
+ for (k = 0; k < SUBFRAMES; k++) {
+ transcodingParam.loFiltGain[k] *= transcodeScale;
+ transcodingParam.hiFiltGain[k] *= transcodeScale;
+ }
+
+ /* Scale DFT coefficients */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale + 0.5);
+ fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale + 0.5);
+ }
+
+ //Store FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+
+ /* Store the state of arithmetic coder before coding LPC gains */
+ ISACencUB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
+
+ ISACencUB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
+
+ ISACencUB_obj->bitstr_obj.streamval = transcodingParam.streamval;
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
+ transcodingParam.stream[0];
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
+ transcodingParam.stream[1];
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
+ transcodingParam.stream[2];
+
+ // Store the gains for multiple encoding
+ memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains,
+ (SUBFRAMES << 1) * sizeof(double));
+
+ WebRtcIsac_EncodeLpcGainUb(transcodingParam.loFiltGain,
+ &ISACencUB_obj->bitstr_obj,
+ ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
+ WebRtcIsac_EncodeLpcGainUb(transcodingParam.hiFiltGain,
+ &ISACencUB_obj->bitstr_obj,
+ &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]);
+
+ /* Update the number of bytes left for encoding the spectrum */
+ bytesLeftSpecCoding = payloadLimitBytes -
+ ISACencUB_obj->bitstr_obj.stream_index;
+
+ // Save the bit-stream object at this point for FEC.
+ memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
+ &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
+
+ /* Encode the spectrum */
+ err = WebRtcIsac_EncodeSpecUB16(fre, fim, &ISACencUB_obj->bitstr_obj);
+ if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ /* There has been an error but it was not too large payload
+ (we can cure too large payload) */
+ return err;
+ }
+ iterCntr++;
+ }
+
+ /* complete arithmetic coding */
+ return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj);
+}
+
+
+int
+WebRtcIsac_EncodeUb12(
+ float* in,
+ ISACUBEncStruct* ISACencUB_obj,
+ WebRtc_Word32 jitterInfo)
+{
+ int err;
+ int k;
+ int iterCntr;
+
+ double lpcVecs[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
+
+ double percepFilterParams[(1 + UB_LPC_ORDER) * SUBFRAMES];
+ float LP[FRAMESAMPLES_HALF];
+ float HP[FRAMESAMPLES_HALF];
+
+ double LP_lookahead[FRAMESAMPLES_HALF];
+ double HP_lookahead[FRAMESAMPLES_HALF];
+ double LPw[FRAMESAMPLES_HALF];
+
+ double HPw[FRAMESAMPLES_HALF];
+ WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
+ WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
+
+ int status = 0;
+
+ double varscale[1];
+
+ double corr[UB_LPC_GAIN_DIM][UB_LPC_ORDER + 1];
+ double lpcGains[SUBFRAMES];
+ transcode_obj transcodingParam;
+ double bytesLeftSpecCoding;
+ WebRtc_UWord16 payloadLimitBytes;
+ double s2nr;
+
+ /* buffer speech samples (by 10ms packet) until the framelength is */
+ /* reached (30 or 60 ms) */
+ /********************************************************************/
+
+ /* fill the buffer with 10ms input data */
+ for (k=0; k<FRAMESAMPLES_10ms; k++) {
+ ISACencUB_obj->data_buffer_float[k + ISACencUB_obj->buffer_index] =
+ in[k];
+ }
+
+ /* if buffer-size is not equal to current frame-size then increase the
+ index and return. We do the encoding when we have enough audio. */
+ if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) {
+ ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms;
+ return 0;
+ }
+ /* if buffer reached the right size, reset index and continue
+ with encoding the frame */
+ ISACencUB_obj->buffer_index = 0;
+
+ /* end of buffer function */
+ /**************************/
+
+ /* encoding */
+ /************/
+
+ /* reset bitstream */
+ ISACencUB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
+ ISACencUB_obj->bitstr_obj.streamval = 0;
+ ISACencUB_obj->bitstr_obj.stream_index = 0;
+
+ /* bandwidth estimation and coding */
+ /* To be used for Redundant Coding */
+ WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj);
+
+ status = WebRtcIsac_EncodeBandwidth(isac12kHz,
+ &ISACencUB_obj->bitstr_obj);
+ if (status < 0) {
+ return status;
+ }
+
+
+ s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck,
+ FRAMESAMPLES);
+
+ /* split signal in two bands */
+ WebRtcIsac_SplitAndFilterFloat(ISACencUB_obj->data_buffer_float, HP, LP,
+ HP_lookahead, LP_lookahead, &ISACencUB_obj->prefiltbankstr_obj);
+
+ /* find coefficients for perceptual pre-filters */
+ WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj,
+ lpcVecs, corr, varscale, isac12kHz);
+
+ /* code LPC model and shape - gains not quantized yet */
+ WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj,
+ percepFilterParams, isac12kHz, &ISACencUB_obj->SaveEnc_obj);
+
+ WebRtcIsac_GetLpcGain(s2nr, percepFilterParams, SUBFRAMES, lpcGains,
+ corr, varscale);
+
+ /* Store the state of arithmetic coder before coding LPC gains */
+ transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper;
+
+ transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index;
+
+ transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval;
+
+ transcodingParam.stream[0] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index - 2];
+
+ transcodingParam.stream[1] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index - 1];
+
+ transcodingParam.stream[2] = ISACencUB_obj->bitstr_obj.stream[
+ ISACencUB_obj->bitstr_obj.stream_index];
+
+ /* Store LPC Gains before encoding them */
+ for(k = 0; k < SUBFRAMES; k++) {
+ transcodingParam.loFiltGain[k] = lpcGains[k];
+ }
+
+ // Store the gains for multiple encoding
+ memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, SUBFRAMES *
+ sizeof(double));
+
+ WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj,
+ ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
+
+ for(k = 0; k < SUBFRAMES; k++) {
+ percepFilterParams[k*(UB_LPC_ORDER + 1)] = lpcGains[k];
+ }
+
+ /* perceptual pre-filtering (using normalized lattice filter) */
+ /* low-band filtering */
+ WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
+ ISACencUB_obj->maskfiltstr_obj.PreStateLoG, LP, percepFilterParams,
+ LPw);
+
+ /* Get the correct value for the payload limit and calculate the number
+ of bytes left for coding the spectrum. It is a 30ms frame Subract 3
+ because termination process may add 3 bytes */
+ payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes -
+ ISACencUB_obj->numBytesUsed - 3;
+ bytesLeftSpecCoding = payloadLimitBytes -
+ ISACencUB_obj->bitstr_obj.stream_index;
+
+ memset(HPw, 0, sizeof(double) * FRAMESAMPLES_HALF);
+
+ /* transform */
+ WebRtcIsac_Time2Spec(LPw, HPw, fre, fim, &ISACencUB_obj->fftstr_obj);
+
+ //Store real FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ //Store imaginary FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ // Save the bit-stream object at this point for FEC.
+ memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
+ &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
+
+ /* quantization and lossless coding */
+ err = WebRtcIsac_EncodeSpecUB12(fre, fim, &ISACencUB_obj->bitstr_obj);
+ if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ /* There has been an error but it was not too large
+ payload (we can cure too large payload) */
+ return err;
+ }
+ iterCntr = 0;
+ while((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
+ (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ double bytesSpecCoderUsed;
+ double transcodeScale;
+
+ if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
+ /* We were not able to limit the payload size */
+ return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
+ }
+
+ if (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
+ bytesSpecCoderUsed = STREAM_SIZE_MAX;
+ // being coservative
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
+ } else {
+ bytesSpecCoderUsed = ISACencUB_obj->bitstr_obj.stream_index -
+ transcodingParam.stream_index;
+ transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
+ }
+
+ /* To be safe, we reduce the scale depending on the
+ number of iterations. */
+ transcodeScale *= (1.0 - (0.9 * (double)iterCntr/
+ (double)MAX_PAYLOAD_LIMIT_ITERATION));
+
+ /* Scale the LPC Gains */
+ for (k = 0; k < SUBFRAMES; k++) {
+ transcodingParam.loFiltGain[k] *= transcodeScale;
+ }
+
+ /* Scale DFT coefficients */
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale + 0.5);
+ fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale + 0.5);
+ }
+
+ //Store real FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+ //Store imaginary FFT coefficients for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
+ FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
+
+
+ /* Re-store the state of arithmetic coder before coding LPC gains */
+ ISACencUB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
+
+ ISACencUB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
+
+ ISACencUB_obj->bitstr_obj.streamval = transcodingParam.streamval;
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
+ transcodingParam.stream[0];
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
+ transcodingParam.stream[1];
+
+ ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
+ transcodingParam.stream[2];
+
+ // Store the gains for multiple encoding
+ memcpy(&ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, SUBFRAMES *
+ sizeof(double));
+
+ // encode LPC gain and store quantization indices. HAving quantization
+ // indices reduces transcoding complexity if 'scale factor' is 1.
+ WebRtcIsac_EncodeLpcGainUb(transcodingParam.loFiltGain,
+ &ISACencUB_obj->bitstr_obj,
+ ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
+
+ // Save the bit-stream object at this point for FEC.
+ memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
+ &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
+
+ /* Update the number of bytes left for encoding the spectrum */
+ bytesLeftSpecCoding = payloadLimitBytes -
+ ISACencUB_obj->bitstr_obj.stream_index;
+
+ /* Encode the spectrum */
+ err = WebRtcIsac_EncodeSpecUB12(fre, fim,
+ &ISACencUB_obj->bitstr_obj);
+ if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
+ /* There has been an error but it was not too large payload
+ (we can cure too large payload) */
+ return err;
+ }
+ iterCntr++;
+ }
+
+ /* complete arithmetic coding */
+ return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj);
+}
+
+
+
+
+
+
+/* This function is used to create a new bitstream with new BWE.
+ The same data as previously encoded with the function WebRtcIsac_Encoder().
+ The data needed is taken from the struct, where it was stored
+ when calling the encoder. */
+
+int WebRtcIsac_EncodeStoredDataLb(
+ const ISAC_SaveEncData_t* ISACSavedEnc_obj,
+ Bitstr* ISACBitStr_obj,
+ int BWnumber,
+ float scale)
+{
+ int ii;
+ int status;
+ int BWno = BWnumber;
+
+ const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
+ const WebRtc_UWord16 **cdf;
+
+ double tmpLPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
+ double tmpLPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
+ int tmpLPCindex_g[12*2];
+ WebRtc_Word16 tmp_fre[FRAMESAMPLES], tmp_fim[FRAMESAMPLES];
+
+ /* Sanity Check - possible values for BWnumber is 0 - 23 */
+ if ((BWnumber < 0) || (BWnumber > 23)) {
+ return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
+ }
+
+ /* reset bitstream */
+ ISACBitStr_obj->W_upper = 0xFFFFFFFF;
+ ISACBitStr_obj->streamval = 0;
+ ISACBitStr_obj->stream_index = 0;
+
+ /* encode frame length */
+ status = WebRtcIsac_EncodeFrameLen(ISACSavedEnc_obj->framelength,
+ ISACBitStr_obj);
+ if (status < 0) {
+ /* Wrong frame size */
+ return status;
+ }
+
+ /* Transcoding */
+ if ((scale > 0.0) && (scale < 1.0)) {
+ /* Compensate LPC gain */
+ for (ii = 0;
+ ii < ((ORDERLO + 1)* SUBFRAMES * (1 + ISACSavedEnc_obj->startIdx));
+ ii++) {
+ tmpLPCcoeffs_lo[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_lo[ii];
+ }
+ for (ii = 0;
+ ii < ((ORDERHI + 1) * SUBFRAMES *(1 + ISACSavedEnc_obj->startIdx));
+ ii++) {
+ tmpLPCcoeffs_hi[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_hi[ii];
+ }
+ /* Scale DFT */
+ for (ii = 0;
+ ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx));
+ ii++) {
+ tmp_fre[ii] = (WebRtc_Word16)((scale) *
+ (float)ISACSavedEnc_obj->fre[ii]) ;
+ tmp_fim[ii] = (WebRtc_Word16)((scale) *
+ (float)ISACSavedEnc_obj->fim[ii]) ;
+ }
+ } else {
+ for (ii = 0;
+ ii < (KLT_ORDER_GAIN * (1 + ISACSavedEnc_obj->startIdx));
+ ii++) {
+ tmpLPCindex_g[ii] = ISACSavedEnc_obj->LPCindex_g[ii];
+ }
+ for (ii = 0;
+ ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx));
+ ii++) {
+ tmp_fre[ii] = ISACSavedEnc_obj->fre[ii];
+ tmp_fim[ii] = ISACSavedEnc_obj->fim[ii];
+ }
+ }
+
+ /* encode bandwidth estimate */
+ WebRtcIsac_EncodeReceiveBw(&BWno, ISACBitStr_obj);
+
+ /* Loop over number of 30 msec */
+ for (ii = 0; ii <= ISACSavedEnc_obj->startIdx; ii++) {
+ /* encode pitch gains */
+ *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
+ WebRtcIsac_EncHistMulti(ISACBitStr_obj,
+ &ISACSavedEnc_obj->pitchGain_index[ii], WebRtcIsac_kQPitchGainCdf_ptr, 1);
+
+ /* entropy coding of quantization pitch lags */
+ /* voicing classificiation */
+ if (ISACSavedEnc_obj->meanGain[ii] < 0.2) {
+ cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
+ } else if (ISACSavedEnc_obj->meanGain[ii] < 0.4) {
+ cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
+ } else {
+ cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
+ }
+ WebRtcIsac_EncHistMulti(ISACBitStr_obj,
+ &ISACSavedEnc_obj->pitchIndex[PITCH_SUBFRAMES*ii], cdf,
+ PITCH_SUBFRAMES);
+
+ /* LPC */
+ /* entropy coding of model number */
+ WebRtcIsac_EncHistMulti(ISACBitStr_obj,
+ &ISACSavedEnc_obj->LPCmodel[ii], WebRtcIsac_kQKltModelCdfPtr, 1);
+
+ /* entropy coding of quantization indices - LPC shape only */
+ WebRtcIsac_EncHistMulti(ISACBitStr_obj,
+ &ISACSavedEnc_obj->LPCindex_s[KLT_ORDER_SHAPE*ii],
+ WebRtcIsac_kQKltCdfPtrShape[ISACSavedEnc_obj->LPCmodel[ii]],
+ KLT_ORDER_SHAPE);
+
+ /* If transcoding, get new LPC gain indices */
+ if (scale < 1.0) {
+ WebRtcIsac_TranscodeLPCCoef(&tmpLPCcoeffs_lo[(ORDERLO+1) *
+ SUBFRAMES*ii], &tmpLPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*ii],
+ ISACSavedEnc_obj->LPCmodel[ii],
+ &tmpLPCindex_g[KLT_ORDER_GAIN * ii]);
+ }
+
+ /* entropy coding of quantization indices - LPC gain */
+ WebRtcIsac_EncHistMulti(ISACBitStr_obj,
+ &tmpLPCindex_g[KLT_ORDER_GAIN*ii], WebRtcIsac_kQKltCdfPtrGain[
+ ISACSavedEnc_obj->LPCmodel[ii]], KLT_ORDER_GAIN);
+
+ /* quantization and lossless coding */
+ status = WebRtcIsac_EncodeSpecLb(&tmp_fre[ii*FRAMESAMPLES_HALF],
+ &tmp_fim[ii*FRAMESAMPLES_HALF], ISACBitStr_obj,
+ ISACSavedEnc_obj->AvgPitchGain[ii]);
+ if (status < 0) {
+ return status;
+ }
+ }
+
+ /* complete arithmetic coding */
+ return WebRtcIsac_EncTerminate(ISACBitStr_obj);
+}
+
+
+
+
+int WebRtcIsac_EncodeStoredDataUb12(
+ const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
+ Bitstr* bitStream,
+ WebRtc_Word32 jitterInfo,
+ float scale)
+{
+ int n;
+ int err;
+ double lpcGain[SUBFRAMES];
+ WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
+ WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+
+ /* reset bitstream */
+ bitStream->W_upper = 0xFFFFFFFF;
+ bitStream->streamval = 0;
+ bitStream->stream_index = 0;
+
+ // Encode jitter index
+ WebRtcIsac_EncodeJitterInfo(jitterInfo, bitStream);
+
+ err = WebRtcIsac_EncodeBandwidth(isac12kHz, bitStream);
+ if(err < 0)
+ {
+ return err;
+ }
+
+ // Encode LPC-shape
+ WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->indexLPCShape,
+ WebRtcIsac_kLpcShapeCdfMatUb12, UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
+
+
+ // we only consider scales between zero and one.
+ if((scale <= 0.0) || (scale > 1.0))
+ {
+ scale = 1.0f;
+ }
+
+ if(scale == 1.0f)
+ {
+ //memcpy(lpcGain, ISACSavedEnc_obj->lpcGain, SUBFRAMES * sizeof(double));
+ WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->lpcGainIndex,
+ WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
+ // store FFT coefficients
+ err = WebRtcIsac_EncodeSpecUB12(ISACSavedEnc_obj->realFFT,
+ ISACSavedEnc_obj->imagFFT, bitStream);
+ }
+ else
+ {
+ /* scale lpc gain and FFT coefficients */
+ for(n = 0; n < SUBFRAMES; n++)
+ {
+ lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n];
+ }
+ // store lpc gain
+ WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream);
+ for(n = 0; n < FRAMESAMPLES_HALF; n++)
+ {
+ realFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->realFFT[n] + 0.5f);
+ imagFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->imagFFT[n] + 0.5f);
+ }
+ // store FFT coefficients
+ err = WebRtcIsac_EncodeSpecUB12(realFFT, imagFFT, bitStream);
+ }
+ if(err < 0)
+ {
+ // error happened while encoding FFT coefficients.
+ return err;
+ }
+
+ /* complete arithmetic coding */
+ return WebRtcIsac_EncTerminate(bitStream);
+}
+
+
+int
+WebRtcIsac_EncodeStoredDataUb16(
+ const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
+ Bitstr* bitStream,
+ WebRtc_Word32 jitterInfo,
+ float scale)
+{
+ int n;
+ int err;
+ double lpcGain[SUBFRAMES << 1];
+ WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
+ WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+
+ /* reset bitstream */
+ bitStream->W_upper = 0xFFFFFFFF;
+ bitStream->streamval = 0;
+ bitStream->stream_index = 0;
+
+ // Encode jitter index
+ WebRtcIsac_EncodeJitterInfo(jitterInfo, bitStream);
+
+ err = WebRtcIsac_EncodeBandwidth(isac16kHz, bitStream);
+ if(err < 0)
+ {
+ return err;
+ }
+
+ WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->indexLPCShape,
+ WebRtcIsac_kLpcShapeCdfMatUb16, UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
+
+ // we only consider scales between zero and one.
+ if((scale <= 0.0) || (scale > 1.0))
+ {
+ scale = 1.0f;
+ }
+
+ if(scale == 1.0f)
+ {
+ // store gains
+ WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->lpcGainIndex,
+ WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
+ WebRtcIsac_EncHistMulti(bitStream, &ISACSavedEnc_obj->lpcGainIndex[SUBFRAMES],
+ WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
+ // store FFT coefficients
+ err = WebRtcIsac_EncodeSpecUB16(ISACSavedEnc_obj->realFFT,
+ ISACSavedEnc_obj->imagFFT, bitStream);
+
+ }
+ else
+ {
+ /* Scale Gain */
+ for(n = 0; n < SUBFRAMES; n++)
+ {
+ lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n];
+ lpcGain[n + SUBFRAMES] = scale * ISACSavedEnc_obj->lpcGain[n + SUBFRAMES];
+ }
+ // store lpc gain
+ WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream);
+ WebRtcIsac_StoreLpcGainUb(&lpcGain[SUBFRAMES], bitStream);
+ /* scale FFT coefficients */
+ for(n = 0; n < FRAMESAMPLES_HALF; n++)
+ {
+ realFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->realFFT[n] + 0.5f);
+ imagFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->imagFFT[n] + 0.5f);
+ }
+ // store FFT coefficients
+ err = WebRtcIsac_EncodeSpecUB16(realFFT, imagFFT, bitStream);
+ }
+
+ if(err < 0)
+ {
+ // error happened while encoding FFT coefficients.
+ return err;
+ }
+
+ /* complete arithmetic coding */
+ return WebRtcIsac_EncTerminate(bitStream);
+}
+
+
+WebRtc_Word16
+WebRtcIsac_GetRedPayloadUb(
+ const ISACUBSaveEncDataStruct* ISACSavedEncObj,
+ Bitstr* bitStreamObj,
+ enum ISACBandwidth bandwidth)
+{
+ int n;
+ WebRtc_Word16 status;
+ WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
+ WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+
+ // store bit-stream object.
+ memcpy(bitStreamObj, &ISACSavedEncObj->bitStreamObj, sizeof(Bitstr));
+
+ // Scale FFT coefficients.
+ for(n = 0; n < FRAMESAMPLES_HALF; n++)
+ {
+ realFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->realFFT[n] *
+ RCU_TRANSCODING_SCALE_UB + 0.5);
+ imagFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->imagFFT[n] *
+ RCU_TRANSCODING_SCALE_UB + 0.5);
+ }
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ status = WebRtcIsac_EncodeSpecUB12(realFFT, imagFFT, bitStreamObj);
+ break;
+ }
+ case isac16kHz:
+ {
+ status = WebRtcIsac_EncodeSpecUB16(realFFT, imagFFT, bitStreamObj);
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(status < 0)
+ {
+ // error happened
+ return status;
+ }
+ else
+ {
+ // terminate entropy coding
+ return WebRtcIsac_EncTerminate(bitStreamObj);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c
new file mode 100644
index 0000000..2bf4c36
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c
@@ -0,0 +1,708 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * code_LPC_UB.c
+ *
+ * This file contains definition of functions used to
+ * encode LPC parameters (Shape & gain) of the upper band.
+ *
+ */
+
+#include "encode_lpc_swb.h"
+#include "typedefs.h"
+#include "settings.h"
+
+#include "lpc_shape_swb12_tables.h"
+#include "lpc_shape_swb16_tables.h"
+#include "lpc_gain_swb_tables.h"
+
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+/******************************************************************************
+ * WebRtcIsac_RemoveLarMean()
+ *
+ * Remove the means from LAR coefficients.
+ *
+ * Input:
+ * -lar : pointer to lar vectors. LAR vectors are
+ * concatenated.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -lar : pointer to mean-removed LAR:s.
+ *
+ *
+ */
+WebRtc_Word16
+WebRtcIsac_RemoveLarMean(
+ double* lar,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 coeffCntr;
+ WebRtc_Word16 vecCntr;
+ WebRtc_Word16 numVec;
+ const double* meanLAR;
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ numVec = UB_LPC_VEC_PER_FRAME;
+ meanLAR = WebRtcIsac_kMeanLarUb12;
+ break;
+ }
+ case isac16kHz:
+ {
+ numVec = UB16_LPC_VEC_PER_FRAME;
+ meanLAR = WebRtcIsac_kMeanLarUb16;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ for(vecCntr = 0; vecCntr < numVec; vecCntr++)
+ {
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ // REMOVE MEAN
+ *lar++ -= meanLAR[coeffCntr];
+ }
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateIntraVec()
+ *
+ * Remove the correlation amonge the components of LAR vectors. If LAR vectors
+ * of one frame are put in a matrix where each column is a LAR vector of a
+ * sub-frame, then this is equivalent to multiplying the LAR matrix with
+ * a decorrelting mtrix from left.
+ *
+ * Input:
+ * -inLar : pointer to mean-removed LAR vecrtors.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : decorrelated LAR vectors.
+ */
+WebRtc_Word16
+WebRtcIsac_DecorrelateIntraVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth)
+{
+ const double* ptrData;
+ const double* ptrRow;
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+ WebRtc_Word16 larVecCntr;
+ WebRtc_Word16 numVec;
+ const double* decorrMat;
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ decorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0];
+ numVec = UB_LPC_VEC_PER_FRAME;
+ break;
+ }
+ case isac16kHz:
+ {
+ decorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0];
+ numVec = UB16_LPC_VEC_PER_FRAME;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ //
+ // decorrMat * data
+ //
+ // data is assumed to contain 'numVec' of LAR
+ // vectors (mean removed) each of dimension 'UB_LPC_ORDER'
+ // concatenated one after the other.
+ //
+
+ ptrData = data;
+ for(larVecCntr = 0; larVecCntr < numVec; larVecCntr++)
+ {
+ for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++)
+ {
+ ptrRow = &decorrMat[rowCntr * UB_LPC_ORDER];
+ *out = 0;
+ for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++)
+ {
+ *out += ptrData[colCntr] * ptrRow[colCntr];
+ }
+ out++;
+ }
+ ptrData += UB_LPC_ORDER;
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateInterVec()
+ *
+ * Remover the correlation among mean-removed LAR vectors. If LAR vectors
+ * of one frame are put in a matrix where each column is a LAR vector of a
+ * sub-frame, then this is equivalent to multiplying the LAR matrix with
+ * a decorrelting mtrix from right.
+ *
+ * Input:
+ * -data : pointer to matrix of LAR vectors. The matrix
+ * is stored column-wise.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : decorrelated LAR vectors.
+ */
+WebRtc_Word16
+WebRtcIsac_DecorrelateInterVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 coeffCntr;
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+ const double* decorrMat;
+ WebRtc_Word16 interVecDim;
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ decorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0];
+ interVecDim = UB_LPC_VEC_PER_FRAME;
+ break;
+ }
+ case isac16kHz:
+ {
+ decorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0];
+ interVecDim = UB16_LPC_VEC_PER_FRAME;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ //
+ // data * decorrMat
+ //
+ // data is of size 'interVecDim' * 'UB_LPC_ORDER'
+ // That is 'interVecDim' of LAR vectors (mean removed)
+ // in columns each of dimension 'UB_LPC_ORDER'.
+ // matrix is stored column-wise.
+ //
+
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ for(colCntr = 0; colCntr < interVecDim; colCntr++)
+ {
+ out[coeffCntr + colCntr * UB_LPC_ORDER] = 0;
+ for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
+ {
+ out[coeffCntr + colCntr * UB_LPC_ORDER] +=
+ data[coeffCntr + rowCntr * UB_LPC_ORDER] *
+ decorrMat[rowCntr * interVecDim + colCntr];
+ }
+ }
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_QuantizeUncorrLar()
+ *
+ * Quantize the uncorrelated parameters.
+ *
+ * Input:
+ * -data : uncorrelated LAR vectors.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -data : quantized version of the input.
+ * -idx : pointer to quantization indices.
+ */
+double
+WebRtcIsac_QuantizeUncorrLar(
+ double* data,
+ int* recIdx,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 cntr;
+ WebRtc_Word32 idx;
+ WebRtc_Word16 interVecDim;
+ const double* leftRecPoint;
+ double quantizationStepSize;
+ const WebRtc_Word16* numQuantCell;
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12;
+ quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12;
+ numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb12;
+ interVecDim = UB_LPC_VEC_PER_FRAME;
+ break;
+ }
+ case isac16kHz:
+ {
+ leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16;
+ quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16;
+ numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb16;
+ interVecDim = UB16_LPC_VEC_PER_FRAME;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ //
+ // Quantize the parametrs.
+ //
+ for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++)
+ {
+ idx = (WebRtc_Word32)floor((*data - leftRecPoint[cntr]) /
+ quantizationStepSize + 0.5);
+ if(idx < 0)
+ {
+ idx = 0;
+ }
+ else if(idx >= numQuantCell[cntr])
+ {
+ idx = numQuantCell[cntr] - 1;
+ }
+
+ *data++ = leftRecPoint[cntr] + idx * quantizationStepSize;
+ *recIdx++ = idx;
+ }
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_DequantizeLpcParam()
+ *
+ * Get the quantized value of uncorrelated LARs given the quantization indices.
+ *
+ * Input:
+ * -idx : pointer to quantiztion indices.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : pointer to quantized values.
+ */
+WebRtc_Word16
+WebRtcIsac_DequantizeLpcParam(
+ const int* idx,
+ double* out,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 cntr;
+ WebRtc_Word16 interVecDim;
+ const double* leftRecPoint;
+ double quantizationStepSize;
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12;
+ quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12;
+ interVecDim = UB_LPC_VEC_PER_FRAME;
+ break;
+ }
+ case isac16kHz:
+ {
+ leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16;
+ quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16;
+ interVecDim = UB16_LPC_VEC_PER_FRAME;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ //
+ // Dequantize given the quantization indices
+ //
+
+ for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++)
+ {
+ *out++ = leftRecPoint[cntr] + *idx++ * quantizationStepSize;
+ }
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateIntraVec()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateIntraVec().
+ *
+ * Input:
+ * -data : uncorrelated parameters.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : correlated parametrs.
+ */
+WebRtc_Word16
+WebRtcIsac_CorrelateIntraVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 vecCntr;
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+ WebRtc_Word16 numVec;
+ const double* ptrData;
+ const double* intraVecDecorrMat;
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ numVec = UB_LPC_VEC_PER_FRAME;
+ intraVecDecorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0];
+ break;
+ }
+ case isac16kHz:
+ {
+ numVec = UB16_LPC_VEC_PER_FRAME;
+ intraVecDecorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0];
+ break;
+ }
+ default:
+ return -1;
+ }
+
+
+ ptrData = data;
+ for(vecCntr = 0; vecCntr < numVec; vecCntr++)
+ {
+ for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++)
+ {
+ *out = 0;
+ for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++)
+ {
+ *out += ptrData[rowCntr] *
+ intraVecDecorrMat[rowCntr * UB_LPC_ORDER + colCntr];
+ }
+ out++;
+ }
+ ptrData += UB_LPC_ORDER;
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateInterVec()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateInterVec().
+ *
+ * Input:
+ * -data
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : correlated parametrs.
+ */
+WebRtc_Word16
+WebRtcIsac_CorrelateInterVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 coeffCntr;
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+ WebRtc_Word16 interVecDim;
+ double myVec[UB16_LPC_VEC_PER_FRAME];
+ const double* interVecDecorrMat;
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ interVecDim = UB_LPC_VEC_PER_FRAME;
+ interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0];
+ break;
+ }
+ case isac16kHz:
+ {
+ interVecDim = UB16_LPC_VEC_PER_FRAME;
+ interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0];
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
+ {
+ myVec[rowCntr] = 0;
+ for(colCntr = 0; colCntr < interVecDim; colCntr++)
+ {
+ myVec[rowCntr] += data[coeffCntr + colCntr * UB_LPC_ORDER] * //*ptrData *
+ interVecDecorrMat[rowCntr * interVecDim + colCntr];
+ //ptrData += UB_LPC_ORDER;
+ }
+ }
+
+ for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
+ {
+ out[coeffCntr + rowCntr * UB_LPC_ORDER] = myVec[rowCntr];
+ }
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_AddLarMean()
+ *
+ * This is the inverse of WebRtcIsac_RemoveLarMean()
+ *
+ * Input:
+ * -data : pointer to mean-removed LAR:s.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -data : pointer to LARs.
+ */
+WebRtc_Word16
+WebRtcIsac_AddLarMean(
+ double* data,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 coeffCntr;
+ WebRtc_Word16 vecCntr;
+ WebRtc_Word16 numVec;
+ const double* meanLAR;
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ numVec = UB_LPC_VEC_PER_FRAME;
+ meanLAR = WebRtcIsac_kMeanLarUb12;
+ break;
+ }
+ case isac16kHz:
+ {
+ numVec = UB16_LPC_VEC_PER_FRAME;
+ meanLAR = WebRtcIsac_kMeanLarUb16;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ for(vecCntr = 0; vecCntr < numVec; vecCntr++)
+ {
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ *data++ += meanLAR[coeffCntr];
+ }
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_ToLogDomainRemoveMean()
+ *
+ * Transform the LPC gain to log domain then remove the mean value.
+ *
+ * Input:
+ * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains
+ *
+ * Output:
+ * -lpcGain : mean-removed in log domain.
+ */
+WebRtc_Word16
+WebRtcIsac_ToLogDomainRemoveMean(
+ double* data)
+{
+ WebRtc_Word16 coeffCntr;
+ for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
+ {
+ data[coeffCntr] = log(data[coeffCntr]) - WebRtcIsac_kMeanLpcGain;
+ }
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateLPGain()
+ *
+ * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like
+ * multiplying gain vector with decorrelating matrix.
+ *
+ * Input:
+ * -data : LPC gain in log-domain with mean removed.
+ *
+ * Output:
+ * -out : decorrelated parameters.
+ */
+WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
+ const double* data,
+ double* out)
+{
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+
+ for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++)
+ {
+ *out = 0;
+ for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++)
+ {
+ *out += data[rowCntr] * WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr];
+ }
+ out++;
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_QuantizeLpcGain()
+ *
+ * Quantize the decorrelated log-domain gains.
+ *
+ * Input:
+ * -lpcGain : uncorrelated LPC gains.
+ *
+ * Output:
+ * -idx : quantization indices
+ * -lpcGain : quantized value of the inpt.
+ */
+double WebRtcIsac_QuantizeLpcGain(
+ double* data,
+ int* idx)
+{
+ WebRtc_Word16 coeffCntr;
+ for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
+ {
+ *idx = (int)floor((*data - WebRtcIsac_kLeftRecPointLpcGain[coeffCntr]) /
+ WebRtcIsac_kQSizeLpcGain + 0.5);
+
+ if(*idx < 0)
+ {
+ *idx = 0;
+ }
+ else if(*idx >= WebRtcIsac_kNumQCellLpcGain[coeffCntr])
+ {
+ *idx = WebRtcIsac_kNumQCellLpcGain[coeffCntr] - 1;
+ }
+ *data = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx *
+ WebRtcIsac_kQSizeLpcGain;
+
+ data++;
+ idx++;
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_DequantizeLpcGain()
+ *
+ * Get the quantized values given the quantization indices.
+ *
+ * Input:
+ * -idx : pointer to quantization indices.
+ *
+ * Output:
+ * -lpcGains : quantized values of the given parametes.
+ */
+WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
+ const int* idx,
+ double* out)
+{
+ WebRtc_Word16 coeffCntr;
+ for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
+ {
+ *out = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx *
+ WebRtcIsac_kQSizeLpcGain;
+ out++;
+ idx++;
+ }
+ return 0;
+}
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateLpcGain()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateLPGain().
+ *
+ * Input:
+ * -data : decorrelated parameters.
+ *
+ * Output:
+ * -out : correlated parameters.
+ */
+WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
+ const double* data,
+ double* out)
+{
+ WebRtc_Word16 rowCntr;
+ WebRtc_Word16 colCntr;
+
+ for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++)
+ {
+ *out = 0;
+ for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++)
+ {
+ *out += WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr] * data[colCntr];
+ }
+ out++;
+ }
+
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_AddMeanToLinearDomain()
+ *
+ * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean().
+ *
+ * Input:
+ * -lpcGain : LPC gain in log-domain & mean removed
+ *
+ * Output:
+ * -lpcGain : LPC gain in normal domain.
+ */
+WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
+ double* lpcGains)
+{
+ WebRtc_Word16 coeffCntr;
+ for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
+ {
+ lpcGains[coeffCntr] = exp(lpcGains[coeffCntr] + WebRtcIsac_kMeanLpcGain);
+ }
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h
new file mode 100644
index 0000000..e7f1a76
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h
@@ -0,0 +1,283 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * encode_lpc_swb.h
+ *
+ * This file contains declaration of functions used to
+ * encode LPC parameters (Shape & gain) of the upper band.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
+
+#include "typedefs.h"
+#include "settings.h"
+#include "structs.h"
+
+
+/******************************************************************************
+ * WebRtcIsac_RemoveLarMean()
+ *
+ * Remove the means from LAR coefficients.
+ *
+ * Input:
+ * -lar : pointer to lar vectors. LAR vectors are
+ * concatenated.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -lar : pointer to mean-removed LAR:s.
+ *
+ *
+ */
+WebRtc_Word16 WebRtcIsac_RemoveLarMean(
+ double* lar,
+ WebRtc_Word16 bandwidth);
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateIntraVec()
+ *
+ * Remove the correlation amonge the components of LAR vectors. If LAR vectors
+ * of one frame are put in a matrix where each column is a LAR vector of a
+ * sub-frame, then this is equivalent to multiplying the LAR matrix with
+ * a decorrelting mtrix from left.
+ *
+ * Input:
+ * -inLar : pointer to mean-removed LAR vecrtors.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : decorrelated LAR vectors.
+ */
+WebRtc_Word16 WebRtcIsac_DecorrelateIntraVec(
+ const double* inLAR,
+ double* out,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateInterVec()
+ *
+ * Remover the correlation among mean-removed LAR vectors. If LAR vectors
+ * of one frame are put in a matrix where each column is a LAR vector of a
+ * sub-frame, then this is equivalent to multiplying the LAR matrix with
+ * a decorrelting mtrix from right.
+ *
+ * Input:
+ * -data : pointer to matrix of LAR vectors. The matrix
+ * is stored column-wise.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : decorrelated LAR vectors.
+ */
+WebRtc_Word16 WebRtcIsac_DecorrelateInterVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_QuantizeUncorrLar()
+ *
+ * Quantize the uncorrelated parameters.
+ *
+ * Input:
+ * -data : uncorrelated LAR vectors.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -data : quantized version of the input.
+ * -idx : pointer to quantization indices.
+ */
+double WebRtcIsac_QuantizeUncorrLar(
+ double* data,
+ int* idx,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateIntraVec()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateIntraVec().
+ *
+ * Input:
+ * -data : uncorrelated parameters.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : correlated parametrs.
+ */
+WebRtc_Word16 WebRtcIsac_CorrelateIntraVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateInterVec()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateInterVec().
+ *
+ * Input:
+ * -data
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : correlated parametrs.
+ */
+WebRtc_Word16 WebRtcIsac_CorrelateInterVec(
+ const double* data,
+ double* out,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_AddLarMean()
+ *
+ * This is the inverse of WebRtcIsac_RemoveLarMean()
+ *
+ * Input:
+ * -data : pointer to mean-removed LAR:s.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -data : pointer to LARs.
+ */
+WebRtc_Word16 WebRtcIsac_AddLarMean(
+ double* data,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_DequantizeLpcParam()
+ *
+ * Get the quantized value of uncorrelated LARs given the quantization indices.
+ *
+ * Input:
+ * -idx : pointer to quantiztion indices.
+ * -bandwidth : indicates if the given LAR vectors belong
+ * to SWB-12kHz or SWB-16kHz.
+ *
+ * Output:
+ * -out : pointer to quantized values.
+ */
+WebRtc_Word16 WebRtcIsac_DequantizeLpcParam(
+ const int* idx,
+ double* out,
+ WebRtc_Word16 bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_ToLogDomainRemoveMean()
+ *
+ * Transform the LPC gain to log domain then remove the mean value.
+ *
+ * Input:
+ * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains
+ *
+ * Output:
+ * -lpcGain : mean-removed in log domain.
+ */
+WebRtc_Word16 WebRtcIsac_ToLogDomainRemoveMean(
+ double* lpGains);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecorrelateLPGain()
+ *
+ * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like
+ * multiplying gain vector with decorrelating matrix.
+ *
+ * Input:
+ * -data : LPC gain in log-domain with mean removed.
+ *
+ * Output:
+ * -out : decorrelated parameters.
+ */
+WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
+ const double* data,
+ double* out);
+
+
+/******************************************************************************
+ * WebRtcIsac_QuantizeLpcGain()
+ *
+ * Quantize the decorrelated log-domain gains.
+ *
+ * Input:
+ * -lpcGain : uncorrelated LPC gains.
+ *
+ * Output:
+ * -idx : quantization indices
+ * -lpcGain : quantized value of the inpt.
+ */
+double WebRtcIsac_QuantizeLpcGain(
+ double* lpGains,
+ int* idx);
+
+
+/******************************************************************************
+ * WebRtcIsac_DequantizeLpcGain()
+ *
+ * Get the quantized values given the quantization indices.
+ *
+ * Input:
+ * -idx : pointer to quantization indices.
+ *
+ * Output:
+ * -lpcGains : quantized values of the given parametes.
+ */
+WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
+ const int* idx,
+ double* lpGains);
+
+
+/******************************************************************************
+ * WebRtcIsac_CorrelateLpcGain()
+ *
+ * This is the inverse of WebRtcIsac_DecorrelateLPGain().
+ *
+ * Input:
+ * -data : decorrelated parameters.
+ *
+ * Output:
+ * -out : correlated parameters.
+ */
+WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
+ const double* data,
+ double* out);
+
+
+/******************************************************************************
+ * WebRtcIsac_AddMeanToLinearDomain()
+ *
+ * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean().
+ *
+ * Input:
+ * -lpcGain : LPC gain in log-domain & mean removed
+ *
+ * Output:
+ * -lpcGain : LPC gain in normal domain.
+ */
+WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
+ double* lpcGains);
+
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c
new file mode 100644
index 0000000..a729944
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c
@@ -0,0 +1,2748 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * entropy_coding.c
+ *
+ * This header file defines all of the functions used to arithmetically
+ * encode the iSAC bistream
+ *
+ */
+
+
+#include "entropy_coding.h"
+#include "settings.h"
+#include "arith_routines.h"
+#include "signal_processing_library.h"
+#include "spectrum_ar_model_tables.h"
+#include "lpc_tables.h"
+#include "pitch_gain_tables.h"
+#include "pitch_lag_tables.h"
+#include "encode_lpc_swb.h"
+#include "lpc_shape_swb12_tables.h"
+#include "lpc_shape_swb16_tables.h"
+#include "lpc_gain_swb_tables.h"
+#include "os_specific_inline.h"
+
+#include <math.h>
+#include <string.h>
+
+static const WebRtc_UWord16 kLpcVecPerSegmentUb12 = 5;
+static const WebRtc_UWord16 kLpcVecPerSegmentUb16 = 4;
+
+/* coefficients for the stepwise rate estimation */
+static const WebRtc_Word32 kRPointsQ10[100] = {
+ 14495, 14295, 14112, 13944, 13788, 13643, 13459, 13276, 13195, 13239,
+ 13243, 13191, 13133, 13216, 13263, 13330, 13316, 13242, 13191, 13106,
+ 12942, 12669, 12291, 11840, 11361, 10795, 10192, 9561, 8934, 8335,
+ 7750, 7161, 6589, 6062, 5570, 5048, 4548, 4069, 3587, 3143,
+ 2717, 2305, 1915, 1557, 1235, 963, 720, 541, 423, 366,
+ 369, 435, 561, 750, 1001, 1304, 1626, 1989, 2381, 2793,
+ 3219, 3656, 4134, 4612, 5106, 5629, 6122, 6644, 7216, 7801,
+ 8386, 8987, 9630, 10255, 10897, 11490, 11950, 12397, 12752, 12999,
+ 13175, 13258, 13323, 13290, 13296, 13335, 13113, 13255, 13347, 13355,
+ 13298, 13247, 13313, 13155, 13267, 13313, 13374, 13446, 13525, 13609};
+
+
+/* cdf array for encoder bandwidth (12 vs 16 kHz) indicator */
+static const WebRtc_UWord16 kOneBitEqualProbCdf[3] = {
+ 0, 32768, 65535 };
+
+/* pointer to cdf array for encoder bandwidth (12 vs 16 kHz) indicator */
+static const WebRtc_UWord16 *kOneBitEqualProbCdf_ptr[1] = {
+ kOneBitEqualProbCdf };
+
+/* initial cdf index for decoder of encoded bandwidth (12 vs 16 kHz) indicator */
+static const WebRtc_UWord16 kOneBitEqualProbInitIndex[1] = {1};
+
+
+/* coefficients for the stepwise rate estimation */
+
+
+static const WebRtc_Word32 acnQ10 = 426;
+static const WebRtc_Word32 bcnQ10 = -581224;
+static const WebRtc_Word32 ccnQ10 = 722631;
+static const WebRtc_Word32 lbcnQ10 = -402874;
+#define DPMIN_Q10 -10240 // -10.00 in Q10
+#define DPMAX_Q10 10240 // 10.00 in Q10
+#define MINBITS_Q10 10240 /* 10.0 in Q10 */
+#define IS_SWB_12KHZ 1
+
+__inline WebRtc_UWord32 stepwise(WebRtc_Word32 dinQ10) {
+
+ WebRtc_Word32 ind, diQ10, dtQ10;
+
+ diQ10 = dinQ10;
+ if (diQ10 < DPMIN_Q10)
+ diQ10 = DPMIN_Q10;
+ if (diQ10 >= DPMAX_Q10)
+ diQ10 = DPMAX_Q10 - 1;
+
+ dtQ10 = diQ10 - DPMIN_Q10; /* Q10 + Q10 = Q10 */
+ ind = (dtQ10 * 5) >> 10; /* 2^10 / 5 = 0.2 in Q10 */
+ /* Q10 -> Q0 */
+
+ return kRPointsQ10[ind];
+}
+
+
+__inline short log2_Q10_B( int x )
+{
+ int zeros;
+ short frac;
+
+ zeros = WebRtcSpl_NormU32( x );
+ frac = ((unsigned int)(x << zeros) & 0x7FFFFFFF) >> 21;
+ return (short) (((31 - zeros) << 10) + frac);
+}
+
+
+
+/* compute correlation from power spectrum */
+static void WebRtcIsac_FindCorrelation(WebRtc_Word32 *PSpecQ12, WebRtc_Word32 *CorrQ7)
+{
+ WebRtc_Word32 summ[FRAMESAMPLES/8];
+ WebRtc_Word32 diff[FRAMESAMPLES/8];
+ const WebRtc_Word16 *CS_ptrQ9;
+ WebRtc_Word32 sum;
+ int k, n;
+
+ for (k = 0; k < FRAMESAMPLES/8; k++) {
+ summ[k] = (PSpecQ12[k] + PSpecQ12[FRAMESAMPLES_QUARTER-1 - k] + 16) >> 5;
+ diff[k] = (PSpecQ12[k] - PSpecQ12[FRAMESAMPLES_QUARTER-1 - k] + 16) >> 5;
+ }
+
+ sum = 2;
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += summ[n];
+ CorrQ7[0] = sum;
+
+ for (k = 0; k < AR_ORDER; k += 2) {
+ sum = 0;
+ CS_ptrQ9 = WebRtcIsac_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += (CS_ptrQ9[n] * diff[n] + 256) >> 9;
+ CorrQ7[k+1] = sum;
+ }
+
+ for (k=1; k<AR_ORDER; k+=2) {
+ sum = 0;
+ CS_ptrQ9 = WebRtcIsac_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ sum += (CS_ptrQ9[n] * summ[n] + 256) >> 9;
+ CorrQ7[k+1] = sum;
+ }
+}
+
+/* compute inverse AR power spectrum */
+/* Changed to the function used in iSAC FIX for compatibility reasons */
+static void WebRtcIsac_FindInvArSpec(const WebRtc_Word16 *ARCoefQ12,
+ const WebRtc_Word32 gainQ10,
+ WebRtc_Word32 *CurveQ16)
+{
+ WebRtc_Word32 CorrQ11[AR_ORDER+1];
+ WebRtc_Word32 sum, tmpGain;
+ WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
+ const WebRtc_Word16 *CS_ptrQ9;
+ int k, n;
+ WebRtc_Word16 round, shftVal = 0, sh;
+
+ sum = 0;
+ for (n = 0; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
+ CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
+
+ /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
+ if(gainQ10>400000){
+ tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
+ round = 32;
+ shftVal = 6;
+ } else {
+ tmpGain = gainQ10;
+ round = 256;
+ shftVal = 9;
+ }
+
+ for (k = 1; k < AR_ORDER+1; k++) {
+ sum = 16384;
+ for (n = k; n < AR_ORDER+1; n++)
+ sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
+ sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
+ CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
+ }
+ sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ CurveQ16[n] = sum;
+
+ for (k = 1; k < AR_ORDER; k += 2) {
+ //CS_ptrQ9 = WebRtcIsac_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL(WebRtcIsac_kCos[k][n], CorrQ11[k+1]) + 2, 2);
+ }
+
+ CS_ptrQ9 = WebRtcIsac_kCos[0];
+
+ /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
+ sh=WebRtcSpl_NormW32(CorrQ11[1]);
+ if (CorrQ11[1]==0) /* Use next correlation */
+ sh=WebRtcSpl_NormW32(CorrQ11[2]);
+
+ if (sh<9)
+ shftVal = 9 - sh;
+ else
+ shftVal = 0;
+
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
+ for (k = 2; k < AR_ORDER; k += 2) {
+ CS_ptrQ9 = WebRtcIsac_kCos[k];
+ for (n = 0; n < FRAMESAMPLES/8; n++)
+ diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
+ }
+
+ for (k=0; k<FRAMESAMPLES/8; k++) {
+ CurveQ16[FRAMESAMPLES_QUARTER-1 - k] = CurveQ16[k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
+ CurveQ16[k] += WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
+ }
+}
+
+/* generate array of dither samples in Q7 */
+static void GenerateDitherQ7Lb(WebRtc_Word16 *bufQ7,
+ WebRtc_UWord32 seed,
+ int length,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ int k, shft;
+ WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14;
+
+ if (AvgPitchGain_Q12 < 614) /* this threshold should be equal to that in decode_spec() */
+ {
+ for (k = 0; k < length-2; k += 3)
+ {
+ /* new random unsigned int */
+ seed = (seed * 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 (Q7) */
+ dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25); // * 128/4294967295
+
+ /* new random unsigned int */
+ seed = (seed * 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 */
+ dither2_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25);
+
+ shft = (seed >> 25) & 15;
+ if (shft < 5)
+ {
+ bufQ7[k] = dither1_Q7;
+ bufQ7[k+1] = dither2_Q7;
+ bufQ7[k+2] = 0;
+ }
+ else if (shft < 10)
+ {
+ bufQ7[k] = dither1_Q7;
+ bufQ7[k+1] = 0;
+ bufQ7[k+2] = dither2_Q7;
+ }
+ else
+ {
+ bufQ7[k] = 0;
+ bufQ7[k+1] = dither1_Q7;
+ bufQ7[k+2] = dither2_Q7;
+ }
+ }
+ }
+ else
+ {
+ dither_gain_Q14 = (WebRtc_Word16)(22528 - 10 * AvgPitchGain_Q12);
+
+ /* dither on half of the coefficients */
+ for (k = 0; k < length-1; k += 2)
+ {
+ /* new random unsigned int */
+ seed = (seed * 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 */
+ dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25);
+
+ /* dither sample is placed in either even or odd index */
+ shft = (seed >> 25) & 1; /* either 0 or 1 */
+
+ bufQ7[k + shft] = (((dither_gain_Q14 * dither1_Q7) + 8192)>>14);
+ bufQ7[k + 1 - shft] = 0;
+ }
+ }
+}
+
+
+
+/******************************************************************************
+ * GenerateDitherQ7LbUB()
+ *
+ * generate array of dither samples in Q7 There are less zeros in dither
+ * vector compared to GenerateDitherQ7Lb.
+ *
+ * A uniform random number generator with the range of [-64 64] is employed
+ * but the generated dithers are scaled by 0.35, a heuristic scaling.
+ *
+ * Input:
+ * -seed : the initial seed for the random number generator.
+ * -length : the number of dither values to be generated.
+ *
+ * Output:
+ * -bufQ7 : pointer to a buffer where dithers are written to.
+ */
+static void GenerateDitherQ7LbUB(
+ WebRtc_Word16 *bufQ7,
+ WebRtc_UWord32 seed,
+ int length)
+{
+ int k;
+ for (k = 0; k < length; k++) {
+ /* new random unsigned int */
+ seed = (seed * 196314165) + 907633515;
+
+ /* fixed-point dither sample between -64 and 64 (Q7) */
+ // * 128/4294967295
+ bufQ7[k] = (WebRtc_Word16)(((int)seed + 16777216)>>25);
+
+ // scale by 0.35
+ bufQ7[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(bufQ7[k],
+ 2048, 13);
+ }
+}
+
+
+
+/*
+ * function to decode the complex spectrum from the bit stream
+ * returns the total number of bytes in the stream
+ */
+int WebRtcIsac_DecodeSpecLb(Bitstr *streamdata,
+ double *fr,
+ double *fi,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ WebRtc_Word16 DitherQ7[FRAMESAMPLES];
+ WebRtc_Word16 data[FRAMESAMPLES];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 gainQ10;
+ WebRtc_Word32 gain2_Q10, res;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ int k, len, i;
+
+ /* create dither signal */
+ GenerateDitherQ7Lb(DitherQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
+
+ /* decode model parameters */
+ if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic decoding of spectrum */
+ if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8, DitherQ7,
+ FRAMESAMPLES, !IS_SWB_12KHZ)) <1)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* subtract dither and scale down spectral samples with low SNR */
+ if (AvgPitchGain_Q12 <= 614)
+ {
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ gainQ10 = WebRtcSpl_DivW32W16ResW16(30 << 10,
+ (WebRtc_Word16)((invARSpec2_Q16[k>>2] + (32768 + (33 << 16))) >> 16));
+ *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
+ *fi++ = (double)((data[k+1] * gainQ10 + 512) >> 10) / 128.0;
+ *fr++ = (double)((data[k+2] * gainQ10 + 512) >> 10) / 128.0;
+ *fi++ = (double)((data[k+3] * gainQ10 + 512) >> 10) / 128.0;
+ }
+ }
+ else
+ {
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ gainQ10 = WebRtcSpl_DivW32W16ResW16(36 << 10,
+ (WebRtc_Word16)((invARSpec2_Q16[k>>2] + (32768 + (40 << 16))) >> 16));
+ *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
+ *fi++ = (double)((data[k+1] * gainQ10 + 512) >> 10) / 128.0;
+ *fr++ = (double)((data[k+2] * gainQ10 + 512) >> 10) / 128.0;
+ *fi++ = (double)((data[k+3] * gainQ10 + 512) >> 10) / 128.0;
+ }
+ }
+
+ return len;
+}
+
+/******************************************************************************
+ * WebRtcIsac_DecodeSpecUB16()
+ * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
+ * This function is called when the codec is in 0-16 kHz bandwidth.
+ * The decoded DFT coefficient can be transformed to time domain by
+ * WebRtcIsac_Time2Spec().
+ *
+ * Input:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are written to.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are written to.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_DecodeSpecUB16(
+ Bitstr* streamdata,
+ double* fr,
+ double* fi)
+{
+ WebRtc_Word16 DitherQ7[FRAMESAMPLES];
+ WebRtc_Word16 data[FRAMESAMPLES];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word32 gain2_Q10, res;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ int k, len, i, j;
+
+ /* create dither signal */
+ GenerateDitherQ7LbUB(DitherQ7, streamdata->W_upper, FRAMESAMPLES);
+
+ /* decode model parameters */
+ if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic decoding of spectrum */
+ if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8,
+ DitherQ7, FRAMESAMPLES, !IS_SWB_12KHZ)) <1)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ /* re-arrange DFT coefficients and scale down */
+ for (j = 0, k = 0; k < FRAMESAMPLES; k += 4, j++)
+ {
+ fr[j] = (double)data[ k ] / 128.0;
+ fi[j] = (double)data[k+1] / 128.0;
+ fr[(FRAMESAMPLES_HALF) - 1 - j] = (double)data[k+2] / 128.0;
+ fi[(FRAMESAMPLES_HALF) - 1 - j] = (double)data[k+3] / 128.0;
+
+ }
+ return len;
+}
+
+
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeSpecUB12()
+ * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
+ * This function is called when the codec is in 0-12 kHz bandwidth.
+ * The decoded DFT coefficient can be transformed to time domain by
+ * WebRtcIsac_Time2Spec().
+ *
+ * Input:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are written to.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are written to.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_DecodeSpecUB12(
+ Bitstr *streamdata,
+ double *fr,
+ double *fi)
+{
+ WebRtc_Word16 DitherQ7[FRAMESAMPLES];
+ WebRtc_Word16 data[FRAMESAMPLES];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word32 res;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ int k, len, i;
+
+ /* create dither signal */
+ GenerateDitherQ7LbUB(DitherQ7, streamdata->W_upper, FRAMESAMPLES);
+
+ /* decode model parameters */
+ if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic decoding of spectrum */
+ if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata,
+ invARSpecQ8, DitherQ7, (FRAMESAMPLES_HALF), IS_SWB_12KHZ)) < 1)
+ {
+ return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
+ }
+
+ for (k = 0, i = 0; k < FRAMESAMPLES_HALF; k += 4)
+ {
+ fr[i] = (double)data[ k ] / 128.0;
+ fi[i] = (double)data[k+1] / 128.0;
+ i++;
+ fr[i] = (double)data[k+2] / 128.0;
+ fi[i] = (double)data[k+3] / 128.0;
+ i++;
+ }
+
+ // The second half of real and imaginary coefficients is zero. This is
+ // due to using the old FFT module which requires two signals as input
+ // while in 0-12 kHz mode we only have 8-12 kHz band, and the second signal
+ // is set to zero
+ memset(&fr[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * sizeof(double));
+ memset(&fi[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * sizeof(double));
+
+ return len;
+}
+
+
+
+
+
+int WebRtcIsac_EncodeSpecLb(const WebRtc_Word16 *fr,
+ const WebRtc_Word16 *fi,
+ Bitstr *streamdata,
+ WebRtc_Word16 AvgPitchGain_Q12)
+{
+ WebRtc_Word16 ditherQ7[FRAMESAMPLES];
+ WebRtc_Word16 dataQ7[FRAMESAMPLES];
+ WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 CorrQ7[AR_ORDER+1];
+ WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word16 val;
+ WebRtc_Word32 nrg, res;
+ WebRtc_UWord32 sum;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ WebRtc_Word16 err;
+ WebRtc_UWord32 nrg_u32;
+ int shift_var;
+ int k, n, j, i;
+
+
+ /* create dither_float signal */
+ GenerateDitherQ7Lb(ditherQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
+
+ /* add dither and quantize, and compute power spectrum */
+ for (k = 0; k < FRAMESAMPLES; k += 4)
+ {
+ val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
+ dataQ7[k] = val;
+ sum = val * val;
+
+ val = ((*fi++ + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
+ dataQ7[k+1] = val;
+ sum += val * val;
+
+ val = ((*fr++ + ditherQ7[k+2] + 64) & 0xFF80) - ditherQ7[k+2];
+ dataQ7[k+2] = val;
+ sum += val * val;
+
+ val = ((*fi++ + ditherQ7[k+3] + 64) & 0xFF80) - ditherQ7[k+3];
+ dataQ7[k+3] = val;
+ sum += val * val;
+
+ PSpec[k>>2] = sum >> 2;
+ }
+
+ /* compute correlation from power spectrum */
+ WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
+
+
+ /* find AR coefficients */
+ /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
+ shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
+
+ if (shift_var > 0) {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] << shift_var;
+ }
+ } else {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
+ }
+ }
+
+ /* find RC coefficients */
+ WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
+
+ /* quantize & code RC Coefficient */
+ WebRtcIsac_EncodeRc(RCQ15, streamdata);
+
+ /* RC -> AR coefficients */
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ /* compute ARCoef' * Corr * ARCoef in Q19 */
+ nrg = 0;
+ for (j = 0; j <= AR_ORDER; j++) {
+ for (n = 0; n <= j; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
+ }
+ for (n = j+1; n <= AR_ORDER; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
+ }
+ }
+
+ nrg_u32 = (WebRtc_UWord32)nrg;
+ if (shift_var > 0) {
+ nrg_u32 = nrg_u32 >> shift_var;
+ } else {
+ nrg_u32 = nrg_u32 << (-shift_var);
+ }
+
+ if (nrg_u32 > 0x7FFFFFFF)
+ nrg = 0x7FFFFFFF;
+ else
+ nrg = (WebRtc_Word32)nrg_u32;
+
+ gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
+
+ /* quantize & code gain2_Q10 */
+ if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
+ return -1;
+ }
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic coding of spectrum */
+ err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
+ FRAMESAMPLES, !IS_SWB_12KHZ);
+ if (err < 0)
+ {
+ return (err);
+ }
+
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeSpecUB16()
+ * Quantize and encode real and imaginary part of the DFT coefficients.
+ * This function is called when the codec is in 0-16 kHz bandwidth.
+ * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
+ *
+ *
+ * Input:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are stored.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are stored.
+ *
+ * Output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_EncodeSpecUB16(
+ const WebRtc_Word16* fr,
+ const WebRtc_Word16* fi,
+ Bitstr* streamdata)
+{
+ WebRtc_Word16 ditherQ7[FRAMESAMPLES];
+ WebRtc_Word16 dataQ7[FRAMESAMPLES];
+ WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 CorrQ7[AR_ORDER+1];
+ WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word16 val;
+ WebRtc_Word32 nrg, res;
+ WebRtc_UWord32 sum;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ WebRtc_Word16 err;
+ WebRtc_UWord32 nrg_u32;
+ int shift_var;
+ int k, n, j, i;
+
+ /* create dither_float signal */
+ GenerateDitherQ7LbUB(ditherQ7, streamdata->W_upper, FRAMESAMPLES);
+
+ /* add dither and quantize, and compute power spectrum */
+ for (j = 0, k = 0; k < FRAMESAMPLES; k += 4, j++)
+ {
+ val = ((fr[j] + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
+ dataQ7[k] = val;
+ sum = val * val;
+
+ val = ((fi[j] + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
+ dataQ7[k+1] = val;
+ sum += val * val;
+
+ val = ((fr[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k+2] + 64) &
+ 0xFF80) - ditherQ7[k+2];
+ dataQ7[k+2] = val;
+ sum += val * val;
+
+ val = ((fi[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k+3] + 64) &
+ 0xFF80) - ditherQ7[k+3];
+ dataQ7[k+3] = val;
+ sum += val * val;
+
+ PSpec[k>>2] = sum >> 2;
+ }
+
+ /* compute correlation from power spectrum */
+ WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
+
+
+ /* find AR coefficients
+ number of bit shifts to 14-bit normalize CorrQ7[0]
+ (leaving room for sign) */
+ shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
+
+ if (shift_var > 0) {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] << shift_var;
+ }
+ } else {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
+ }
+ }
+
+ /* find RC coefficients */
+ WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
+
+ /* quantize & code RC Coef */
+ WebRtcIsac_EncodeRc(RCQ15, streamdata);
+
+ /* RC -> AR coefficients */
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+ /* compute ARCoef' * Corr * ARCoef in Q19 */
+ nrg = 0;
+ for (j = 0; j <= AR_ORDER; j++) {
+ for (n = 0; n <= j; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] +
+ 256) >> 9) + 4 ) >> 3;
+ }
+ for (n = j+1; n <= AR_ORDER; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] +
+ 256) >> 9) + 4 ) >> 3;
+ }
+ }
+ nrg_u32 = (WebRtc_UWord32)nrg;
+ if (shift_var > 0) {
+ nrg_u32 = nrg_u32 >> shift_var;
+ } else {
+ nrg_u32 = nrg_u32 << (-shift_var);
+ }
+
+ if (nrg_u32 > 0x7FFFFFFF)
+ nrg = 0x7FFFFFFF;
+ else
+ nrg = (WebRtc_Word32)nrg_u32;
+
+ gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
+
+ /* quantize & code gain2_Q10 */
+ if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
+ return -1;
+ }
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic coding of spectrum */
+ err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
+ FRAMESAMPLES, !IS_SWB_12KHZ);
+ if (err < 0)
+ {
+ return (err);
+ }
+
+ return 0;
+}
+
+
+
+
+int WebRtcIsac_EncodeSpecUB12(const WebRtc_Word16 *fr,
+ const WebRtc_Word16 *fi,
+ Bitstr *streamdata)
+{
+ WebRtc_Word16 ditherQ7[FRAMESAMPLES];
+ WebRtc_Word16 dataQ7[FRAMESAMPLES];
+ WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
+ WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
+ WebRtc_Word32 CorrQ7[AR_ORDER+1];
+ WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
+ WebRtc_Word16 RCQ15[AR_ORDER];
+ WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
+ WebRtc_Word32 gain2_Q10;
+ WebRtc_Word16 val;
+ WebRtc_Word32 nrg, res;
+ WebRtc_UWord32 sum;
+ WebRtc_Word32 in_sqrt;
+ WebRtc_Word32 newRes;
+ WebRtc_Word16 err;
+ int shift_var;
+ int k, n, j, i;
+ WebRtc_UWord32 nrg_u32;
+
+ /* create dither_float signal */
+ GenerateDitherQ7LbUB(ditherQ7, streamdata->W_upper, FRAMESAMPLES);
+
+ /* add dither and quantize, and compute power spectrum */
+ for (k = 0, j = 0; k < (FRAMESAMPLES_HALF); k += 4)
+ {
+ val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
+ dataQ7[k] = val;
+ sum = (val) * (val);
+
+ val = ((*fi++ + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
+ dataQ7[k+1] = val;
+ sum += (val) * (val);
+
+ if(j < FRAMESAMPLES_QUARTER)
+ {
+ PSpec[j] = sum >> 1;
+ j++;
+ }
+
+ val = ((*fr++ + ditherQ7[k+2] + 64) & 0xFF80) - ditherQ7[k+2];
+ dataQ7[k+2] = val;
+ sum = (val) * (val);
+
+ val = ((*fi++ + ditherQ7[k+3] + 64) & 0xFF80) - ditherQ7[k+3];
+ dataQ7[k+3] = val;
+ sum += (val) * (val);
+
+ if(j < FRAMESAMPLES_QUARTER)
+ {
+ PSpec[j] = sum >> 1;
+ j++;
+ }
+ }
+ /* compute correlation from power spectrum */
+ WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
+
+
+ /* find AR coefficients */
+ /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
+ shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
+
+ if (shift_var > 0) {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] << shift_var;
+ }
+ } else {
+ for (k=0; k<AR_ORDER+1; k++) {
+ CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
+ }
+ }
+
+ /* find RC coefficients */
+ WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
+
+ /* quantize & code RC Coef */
+ WebRtcIsac_EncodeRc(RCQ15, streamdata);
+
+
+ /* RC -> AR coefficients */
+ WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
+
+
+ /* compute ARCoef' * Corr * ARCoef in Q19 */
+ nrg = 0;
+ for (j = 0; j <= AR_ORDER; j++) {
+ for (n = 0; n <= j; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
+ }
+ for (n = j+1; n <= AR_ORDER; n++) {
+ nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
+ }
+ }
+
+ nrg_u32 = (WebRtc_UWord32)nrg;
+ if (shift_var > 0) {
+ nrg_u32 = nrg_u32 >> shift_var;
+ } else {
+ nrg_u32 = nrg_u32 << (-shift_var);
+ }
+
+ if (nrg_u32 > 0x7FFFFFFF) {
+ nrg = 0x7FFFFFFF;
+ } else {
+ nrg = (WebRtc_Word32)nrg_u32;
+ }
+
+ gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
+
+ /* quantize & code gain2_Q10 */
+ if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
+ return -1;
+ }
+
+ /* compute inverse AR power spectrum */
+ WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
+
+ /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
+ res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
+ {
+ in_sqrt = invARSpec2_Q16[k];
+ i = 10;
+
+ /* Negative values make no sense for a real sqrt-function. */
+ if (in_sqrt<0)
+ in_sqrt=-in_sqrt;
+
+ newRes = (in_sqrt / res + res) >> 1;
+ do
+ {
+ res = newRes;
+ newRes = (in_sqrt / res + res) >> 1;
+ } while (newRes != res && i-- > 0);
+
+ invARSpecQ8[k] = (WebRtc_Word16)newRes;
+ }
+
+ /* arithmetic coding of spectrum */
+ err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
+ (FRAMESAMPLES_HALF), IS_SWB_12KHZ);
+ if (err < 0)
+ {
+ return (err);
+ }
+
+ return 0;
+}
+
+
+
+/* step-up */
+void WebRtcIsac_Rc2Poly(double *RC, int N, double *a)
+{
+ int m, k;
+ double tmp[MAX_AR_MODEL_ORDER];
+
+ a[0] = 1.0;
+ tmp[0] = 1.0;
+ for (m=1; m<=N; m++) {
+ /* copy */
+ for (k=1; k<m; k++)
+ tmp[k] = a[k];
+
+ a[m] = RC[m-1];
+ for (k=1; k<m; k++)
+ a[k] += RC[m-1] * tmp[m-k];
+ }
+ return;
+}
+
+/* step-down */
+void WebRtcIsac_Poly2Rc(double *a, int N, double *RC)
+{
+ int m, k;
+ double tmp[MAX_AR_MODEL_ORDER];
+ double tmp_inv;
+
+ RC[N-1] = a[N];
+ for (m=N-1; m>0; m--) {
+ tmp_inv = 1.0 / (1.0 - RC[m]*RC[m]);
+ for (k=1; k<=m; k++)
+ tmp[k] = (a[k] - RC[m] * a[m-k+1]) * tmp_inv;
+
+ for (k=1; k<m; k++)
+ a[k] = tmp[k];
+
+ RC[m-1] = tmp[m];
+ }
+ return;
+}
+
+
+#define MAX_ORDER 100
+
+
+void WebRtcIsac_Rc2Lar(const double *refc, double *lar, int order) { /* Matlab's LAR definition */
+
+ int k;
+
+ for (k = 0; k < order; k++) {
+ lar[k] = log((1 + refc[k]) / (1 - refc[k]));
+ }
+
+}
+
+
+void WebRtcIsac_Lar2Rc(const double *lar, double *refc, int order) {
+
+ int k;
+ double tmp;
+
+ for (k = 0; k < order; k++) {
+ tmp = exp(lar[k]);
+ refc[k] = (tmp - 1) / (tmp + 1);
+ }
+}
+
+void WebRtcIsac_Poly2Lar(double *lowband, int orderLo, double *hiband, int orderHi, int Nsub, double *lars) {
+
+ int k, n, orderTot;
+ double poly[MAX_ORDER], lar[MAX_ORDER], rc[MAX_ORDER], *inpl, *inph, *outp;
+
+ orderTot = (orderLo + orderHi + 2);
+ inpl = lowband;
+ inph = hiband;
+ outp = lars;
+ poly[0] = 1.0;
+ for (k = 0; k < Nsub; k++) {
+ /* gains */
+ outp[0] = inpl[0];
+ outp[1] = inph[0];
+
+ /* Low band */
+ for (n = 1; n <= orderLo; n++)
+ poly[n] = inpl[n];
+ WebRtcIsac_Poly2Rc(poly, orderLo, rc);
+ WebRtcIsac_Rc2Lar(rc, lar, orderLo);
+ for (n = 0; n < orderLo; n++)
+ outp[n + 2] = lar[n];
+
+ /* High band */
+ for (n = 1; n <= orderHi; n++)
+ poly[n] = inph[n];
+ WebRtcIsac_Poly2Rc(poly, orderHi, rc);
+ WebRtcIsac_Rc2Lar(rc, lar, orderHi);
+ for (n = 0; n < orderHi; n++)
+ outp[n + orderLo + 2] = lar[n];
+
+ inpl += orderLo + 1;
+ inph += orderHi + 1;
+ outp += orderTot;
+ }
+}
+
+
+
+WebRtc_Word16
+WebRtcIsac_Poly2LarUB(
+ double* lpcVecs,
+ WebRtc_Word16 bandwidth)
+{
+ double poly[MAX_ORDER];
+ double rc[MAX_ORDER];
+ double* ptrIO;
+ WebRtc_Word16 vecCntr;
+ WebRtc_Word16 vecSize;
+ WebRtc_Word16 numVec;
+
+ vecSize = UB_LPC_ORDER;
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ numVec = UB_LPC_VEC_PER_FRAME;
+ break;
+ }
+ case isac16kHz:
+ {
+ numVec = UB16_LPC_VEC_PER_FRAME;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ ptrIO = lpcVecs;
+ poly[0] = 1.0;
+ for(vecCntr = 0; vecCntr < numVec; vecCntr++)
+ {
+ memcpy(&poly[1], ptrIO, sizeof(double) * vecSize);
+ WebRtcIsac_Poly2Rc(poly, vecSize, rc);
+ WebRtcIsac_Rc2Lar(rc, ptrIO, vecSize);
+ ptrIO += vecSize;
+ }
+ return 0;
+}
+
+
+
+void WebRtcIsac_Lar2Poly(double *lars, double *lowband, int orderLo, double *hiband, int orderHi, int Nsub) {
+
+ int k, n, orderTot;
+ double poly[MAX_ORDER], lar[MAX_ORDER], rc[MAX_ORDER], *outpl, *outph, *inp;
+
+ orderTot = (orderLo + orderHi + 2);
+ outpl = lowband;
+ outph = hiband;
+ inp = lars;
+ for (k = 0; k < Nsub; k++) {
+ /* gains */
+ outpl[0] = inp[0];
+ outph[0] = inp[1];
+
+ /* Low band */
+ for (n = 0; n < orderLo; n++)
+ lar[n] = inp[n + 2];
+ WebRtcIsac_Lar2Rc(lar, rc, orderLo);
+ WebRtcIsac_Rc2Poly(rc, orderLo, poly);
+ for (n = 1; n <= orderLo; n++)
+ outpl[n] = poly[n];
+
+ /* High band */
+ for (n = 0; n < orderHi; n++)
+ lar[n] = inp[n + orderLo + 2];
+ WebRtcIsac_Lar2Rc(lar, rc, orderHi);
+ WebRtcIsac_Rc2Poly(rc, orderHi, poly);
+ for (n = 1; n <= orderHi; n++)
+ outph[n] = poly[n];
+
+ outpl += orderLo + 1;
+ outph += orderHi + 1;
+ inp += orderTot;
+ }
+}
+
+// assumes 2 LAR vectors interpolates to 'numPolyVec' A-polynomials
+void
+WebRtcIsac_Lar2PolyInterpolUB(
+ double* larVecs,
+ double* percepFilterParams,
+ int numPolyVecs) // includes the first and the last point of the interval
+{
+
+ int polyCntr, coeffCntr;
+ double larInterpol[UB_LPC_ORDER];
+ double rc[UB_LPC_ORDER];
+ double delta[UB_LPC_ORDER];
+
+ // calculate the step-size for linear interpolation coefficients
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ delta[coeffCntr] = (larVecs[UB_LPC_ORDER + coeffCntr] -
+ larVecs[coeffCntr]) / (numPolyVecs - 1);
+ }
+
+ for(polyCntr = 0; polyCntr < numPolyVecs; polyCntr++)
+ {
+ for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
+ {
+ larInterpol[coeffCntr] = larVecs[coeffCntr] +
+ delta[coeffCntr] * polyCntr;
+ }
+ WebRtcIsac_Lar2Rc(larInterpol, rc, UB_LPC_ORDER);
+
+ // convert to A-polynomial, the following function returns A[0] = 1;
+ // which is written where gains had to be written. Then we write the
+ // gain (outside this function). This way we say a memcpy
+ WebRtcIsac_Rc2Poly(rc, UB_LPC_ORDER, percepFilterParams);
+ percepFilterParams += (UB_LPC_ORDER + 1);
+ }
+}
+
+int WebRtcIsac_DecodeLpc(Bitstr *streamdata, double *LPCCoef_lo, double *LPCCoef_hi, int *outmodel) {
+
+ double lars[KLT_ORDER_GAIN + KLT_ORDER_SHAPE];
+ int err;
+
+ err = WebRtcIsac_DecodeLpcCoef(streamdata, lars, outmodel);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_LPC;
+
+ WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES);
+
+ return 0;
+}
+
+WebRtc_Word16
+WebRtcIsac_DecodeInterpolLpcUb(
+ Bitstr* streamdata,
+ double* percepFilterParams,
+ WebRtc_Word16 bandwidth)
+{
+
+ double lpcCoeff[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ int err;
+ int interpolCntr;
+ int subframeCntr;
+ WebRtc_Word16 numSegments;
+ WebRtc_Word16 numVecPerSegment;
+ WebRtc_Word16 numGains;
+
+ double percepFilterGains[SUBFRAMES<<1];
+ double* ptrOutParam = percepFilterParams;
+
+ err = WebRtcIsac_DecodeLpcCoefUB(streamdata, lpcCoeff, percepFilterGains,
+ bandwidth);
+
+ // error check
+ if (err<0)
+ {
+ return -ISAC_RANGE_ERROR_DECODE_LPC;
+ }
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ numGains = SUBFRAMES;
+ numSegments = UB_LPC_VEC_PER_FRAME - 1;
+ numVecPerSegment = kLpcVecPerSegmentUb12;
+ break;
+ }
+ case isac16kHz:
+ {
+ numGains = SUBFRAMES << 1;
+ numSegments = UB16_LPC_VEC_PER_FRAME - 1;
+ numVecPerSegment = kLpcVecPerSegmentUb16;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+
+
+ for(interpolCntr = 0; interpolCntr < numSegments; interpolCntr++)
+ {
+ WebRtcIsac_Lar2PolyInterpolUB(
+ &lpcCoeff[interpolCntr * UB_LPC_ORDER], ptrOutParam,
+ numVecPerSegment + 1);
+
+ ptrOutParam += ((numVecPerSegment) *
+ (UB_LPC_ORDER + 1));
+ }
+
+ ptrOutParam = percepFilterParams;
+
+ if(bandwidth == isac16kHz)
+ {
+ ptrOutParam += (1 + UB_LPC_ORDER);
+ }
+
+ for(subframeCntr = 0; subframeCntr < numGains; subframeCntr++)
+ {
+ *ptrOutParam = percepFilterGains[subframeCntr];
+ ptrOutParam += (1 + UB_LPC_ORDER);
+ }
+
+ return 0;
+}
+
+
+/* decode & dequantize LPC Coef */
+int WebRtcIsac_DecodeLpcCoef(Bitstr *streamdata, double *LPCCoef, int *outmodel)
+{
+ int j, k, n, model, pos, pos2, posg, poss, offsg, offss, offs2;
+ int index_g[KLT_ORDER_GAIN], index_s[KLT_ORDER_SHAPE];
+ double tmpcoeffs_g[KLT_ORDER_GAIN],tmpcoeffs_s[KLT_ORDER_SHAPE];
+ double tmpcoeffs2_g[KLT_ORDER_GAIN], tmpcoeffs2_s[KLT_ORDER_SHAPE];
+ double sum;
+ int err;
+
+
+ /* entropy decoding of model number */
+ err = WebRtcIsac_DecHistOneStepMulti(&model, streamdata, WebRtcIsac_kQKltModelCdfPtr, WebRtcIsac_kQKltModelInitIndex, 1);
+ if (err<0) // error check
+ return err;
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata, WebRtcIsac_kQKltCdfPtrShape[model], WebRtcIsac_kQKltInitIndexShape[model], KLT_ORDER_SHAPE);
+ if (err<0) // error check
+ return err;
+ err = WebRtcIsac_DecHistOneStepMulti(index_g, streamdata, WebRtcIsac_kQKltCdfPtrGain[model], WebRtcIsac_kQKltInitIndexGain[model], KLT_ORDER_GAIN);
+ if (err<0) // error check
+ return err;
+
+
+ /* find quantization levels for coefficients */
+ for (k=0; k<KLT_ORDER_SHAPE; k++) {
+ tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] = WebRtcIsac_kQKltLevelsShape[WebRtcIsac_kQKltOfLevelsShape[model]+WebRtcIsac_kQKltOffsetShape[model][k] + index_s[k]];
+ }
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+ tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+ WebRtcIsac_kQKltOffsetGain[model][k] + index_g[k]];
+ }
+
+
+ /* inverse KLT */
+
+ /* left transform */ // Transpose matrix!
+ offsg = 0;
+ offss = 0;
+ posg = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = offsg;
+ pos2 = offs2;
+ for (n=0; n<LPC_GAIN_ORDER; n++)
+ sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2++];
+ tmpcoeffs2_g[posg++] = sum;
+ offs2 += LPC_GAIN_ORDER;
+ }
+ offs2 = 0;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = offss;
+ pos2 = offs2;
+ for (n=0; n<LPC_SHAPE_ORDER; n++)
+ sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[model][pos2++];
+ tmpcoeffs2_s[poss++] = sum;
+ offs2 += LPC_SHAPE_ORDER;
+ }
+ offsg += LPC_GAIN_ORDER;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+
+ /* right transform */ // Transpose matrix
+ offsg = 0;
+ offss = 0;
+ posg = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2];
+ pos += LPC_GAIN_ORDER;
+ pos2 += SUBFRAMES;
+
+ }
+ tmpcoeffs_g[posg++] = sum;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[model][pos2];
+ pos += LPC_SHAPE_ORDER;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_s[poss++] = sum;
+ }
+ offsg += LPC_GAIN_ORDER;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+
+ /* scaling, mean addition, and gain restoration */
+ posg = 0;poss = 0;pos=0;
+ for (k=0; k<SUBFRAMES; k++) {
+
+ /* log gains */
+ LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[model][posg];
+ LPCCoef[pos] = exp(LPCCoef[pos]);
+ pos++;posg++;
+ LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[model][posg];
+ LPCCoef[pos] = exp(LPCCoef[pos]);
+ pos++;posg++;
+
+ /* lo band LAR coeffs */
+ for (n=0; n<LPC_LOBAND_ORDER; n++, pos++, poss++) {
+ LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[model][poss];
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<LPC_HIBAND_ORDER; n++, pos++, poss++) {
+ LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[model][poss];
+ }
+ }
+
+
+ *outmodel=model;
+
+ return 0;
+}
+
+/* estimate codel length of LPC Coef */
+void WebRtcIsac_EncodeLar(double *LPCCoef, int *model, double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
+ int j, k, n, bmodel, pos, pos2, poss, posg, offsg, offss, offs2;
+ int index_g[KLT_ORDER_GAIN], index_s[KLT_ORDER_SHAPE];
+ int index_ovr_g[KLT_ORDER_GAIN], index_ovr_s[KLT_ORDER_SHAPE];
+ double Bits;
+ double tmpcoeffs_g[KLT_ORDER_GAIN], tmpcoeffs_s[KLT_ORDER_SHAPE];
+ double tmpcoeffs2_g[KLT_ORDER_GAIN], tmpcoeffs2_s[KLT_ORDER_SHAPE];
+ double sum;
+
+ /* Only one LPC model remains in iSAC. Tables for other models are saved for compatibility reasons. */
+ bmodel = 0;
+
+ /* log gains, mean removal and scaling */
+ posg = 0;poss = 0;pos=0;
+
+ for (k=0; k<SUBFRAMES; k++) {
+ /* log gains */
+ tmpcoeffs_g[posg] = log(LPCCoef[pos]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[bmodel][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;pos++;
+
+ tmpcoeffs_g[posg] = log(LPCCoef[pos]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[bmodel][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;pos++;
+
+ /* lo band LAR coeffs */
+ for (n=0; n<LPC_LOBAND_ORDER; n++, poss++, pos++) {
+ tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[bmodel][poss];
+ tmpcoeffs_s[poss] *= LPC_LOBAND_SCALE;
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<LPC_HIBAND_ORDER; n++, poss++, pos++) {
+ tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[bmodel][poss];
+ tmpcoeffs_s[poss] *= LPC_HIBAND_SCALE;
+ }
+ }
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ offss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<LPC_GAIN_ORDER; n++) {
+ sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[bmodel][pos2];
+ pos2 += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs2_g[posg++] = sum;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = offss;
+ pos2 = k;
+ for (n=0; n<LPC_SHAPE_ORDER; n++) {
+ sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[bmodel][pos2];
+ pos2 += LPC_SHAPE_ORDER;
+ }
+ tmpcoeffs2_s[poss++] = sum;
+ }
+ offsg += LPC_GAIN_ORDER;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offss = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[bmodel][pos2++];
+ pos += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs_g[posg++] = sum;
+ }
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[bmodel][pos2++];
+ pos += LPC_SHAPE_ORDER;
+ }
+ tmpcoeffs_s[poss++] = sum;
+ }
+ offs2 += SUBFRAMES;
+ offsg += LPC_GAIN_ORDER;
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* quantize coefficients */
+
+ Bits = 0.0;
+ for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
+ {
+ pos = WebRtcIsac_kQKltSelIndGain[k];
+ pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
+ index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k]; //ATTN: ok?
+ if (index_g[k] < 0) {
+ index_g[k] = 0;
+ }
+ else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k])
+ index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
+ index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[bmodel][k]+index_g[k];
+ pos = WebRtcIsac_kQKltOfLevelsGain[bmodel] + index_ovr_g[k];
+
+ /* determine number of bits */
+ sum = WebRtcIsac_kQKltCodeLenGain[pos];
+ Bits += sum;
+ }
+
+ for (k=0; k<KLT_ORDER_SHAPE; k++) //ATTN: ok?
+ {
+ index_s[k] = (WebRtcIsac_lrint(tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] / KLT_STEPSIZE)) + WebRtcIsac_kQKltQuantMinShape[k]; //ATTN: ok?
+ if (index_s[k] < 0)
+ index_s[k] = 0;
+ else if (index_s[k] > WebRtcIsac_kQKltMaxIndShape[k])
+ index_s[k] = WebRtcIsac_kQKltMaxIndShape[k];
+ index_ovr_s[k] = WebRtcIsac_kQKltOffsetShape[bmodel][k]+index_s[k];
+ pos = WebRtcIsac_kQKltOfLevelsShape[bmodel] + index_ovr_s[k];
+ sum = WebRtcIsac_kQKltCodeLenShape[pos];
+ Bits += sum;
+ }
+
+
+ /* Only one model remains in this version of the code, model = 0 */
+ *model=bmodel;
+ *size=Bits;
+
+ /* entropy coding of model number */
+ WebRtcIsac_EncHistMulti(streamdata, model, WebRtcIsac_kQKltModelCdfPtr, 1);
+
+ /* entropy coding of quantization indices - shape only */
+ WebRtcIsac_EncHistMulti(streamdata, index_s, WebRtcIsac_kQKltCdfPtrShape[bmodel], KLT_ORDER_SHAPE);
+
+ /* Save data for creation of multiple bit streams */
+ encData->LPCmodel[encData->startIdx] = 0;
+ for (k=0; k<KLT_ORDER_SHAPE; k++)
+ {
+ encData->LPCindex_s[KLT_ORDER_SHAPE*encData->startIdx + k] = index_s[k];
+ }
+
+ /* find quantization levels for shape coefficients */
+ for (k=0; k<KLT_ORDER_SHAPE; k++) {
+ tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] = WebRtcIsac_kQKltLevelsShape[WebRtcIsac_kQKltOfLevelsShape[bmodel]+index_ovr_s[k]];
+ }
+ /* inverse KLT */
+ /* left transform */ // Transpose matrix!
+ offss = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = offss;
+ pos2 = offs2;
+ for (n=0; n<LPC_SHAPE_ORDER; n++)
+ sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[bmodel][pos2++];
+ tmpcoeffs2_s[poss++] = sum;
+ offs2 += LPC_SHAPE_ORDER;
+ }
+ offss += LPC_SHAPE_ORDER;
+ }
+
+
+ /* right transform */ // Transpose matrix
+ offss = 0;
+ poss = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ poss = offss;
+ for (k=0; k<LPC_SHAPE_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[bmodel][pos2];
+ pos += LPC_SHAPE_ORDER;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_s[poss++] = sum;
+ }
+ offss += LPC_SHAPE_ORDER;
+ }
+
+ /* scaling, mean addition, and gain restoration */
+ poss = 0;pos=0;
+ for (k=0; k<SUBFRAMES; k++) {
+
+ /* log gains */
+ pos+=2;
+
+ /* lo band LAR coeffs */
+ for (n=0; n<LPC_LOBAND_ORDER; n++, pos++, poss++) {
+ LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[bmodel][poss];
+ }
+
+ /* hi band LAR coeffs */
+ for (n=0; n<LPC_HIBAND_ORDER; n++, pos++, poss++) {
+ LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE;
+ LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[bmodel][poss];
+ }
+
+ }
+
+}
+
+
+void WebRtcIsac_EncodeLpcLb(double *LPCCoef_lo, double *LPCCoef_hi, int *model,
+ double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
+
+ double lars[KLT_ORDER_GAIN+KLT_ORDER_SHAPE];
+ int k;
+
+ WebRtcIsac_Poly2Lar(LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES, lars);
+ WebRtcIsac_EncodeLar(lars, model, size, streamdata, encData);
+ WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES);
+ /* Save data for creation of multiple bit streams (and transcoding) */
+ for (k=0; k<(ORDERLO+1)*SUBFRAMES; k++) {
+ encData->LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*encData->startIdx + k] = LPCCoef_lo[k];
+ }
+ for (k=0; k<(ORDERHI+1)*SUBFRAMES; k++) {
+ encData->LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*encData->startIdx + k] = LPCCoef_hi[k];
+ }
+}
+
+
+WebRtc_Word16
+WebRtcIsac_EncodeLpcUB(
+ double* lpcVecs,
+ Bitstr* streamdata,
+ double* interpolLPCCoeff,
+ WebRtc_Word16 bandwidth,
+ ISACUBSaveEncDataStruct* encData)
+{
+
+ double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ int idx[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ int interpolCntr;
+
+ WebRtcIsac_Poly2LarUB(lpcVecs, bandwidth);
+ WebRtcIsac_RemoveLarMean(lpcVecs, bandwidth);
+ WebRtcIsac_DecorrelateIntraVec(lpcVecs, U, bandwidth);
+ WebRtcIsac_DecorrelateInterVec(U, lpcVecs, bandwidth);
+ WebRtcIsac_QuantizeUncorrLar(lpcVecs, idx, bandwidth);
+
+ WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth);
+ WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth);
+ WebRtcIsac_AddLarMean(lpcVecs, bandwidth);
+
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ // Stor the indices to be used for multiple encoding.
+ memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME *
+ sizeof(int));
+ WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb12,
+ UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
+ for(interpolCntr = 0; interpolCntr < UB_INTERPOL_SEGMENTS; interpolCntr++)
+ {
+ WebRtcIsac_Lar2PolyInterpolUB(lpcVecs,
+ interpolLPCCoeff, kLpcVecPerSegmentUb12 + 1);
+ lpcVecs += UB_LPC_ORDER;
+ interpolLPCCoeff += (kLpcVecPerSegmentUb12 * (UB_LPC_ORDER + 1));
+ }
+ break;
+ }
+ case isac16kHz:
+ {
+ // Stor the indices to be used for multiple encoding.
+ memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME *
+ sizeof(int));
+ WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb16,
+ UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
+ for(interpolCntr = 0; interpolCntr < UB16_INTERPOL_SEGMENTS; interpolCntr++)
+ {
+ WebRtcIsac_Lar2PolyInterpolUB(lpcVecs,
+ interpolLPCCoeff, kLpcVecPerSegmentUb16 + 1);
+ lpcVecs += UB_LPC_ORDER;
+ interpolLPCCoeff += (kLpcVecPerSegmentUb16 * (UB_LPC_ORDER + 1));
+ }
+ break;
+ }
+ default:
+ return -1;
+ }
+ return 0;
+}
+
+void WebRtcIsac_EncodeLpcGainLb(double *LPCCoef_lo, double *LPCCoef_hi, int model, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
+
+ int j, k, n, pos, pos2, posg, offsg, offs2;
+ int index_g[KLT_ORDER_GAIN];
+ int index_ovr_g[KLT_ORDER_GAIN];
+ double tmpcoeffs_g[KLT_ORDER_GAIN];
+ double tmpcoeffs2_g[KLT_ORDER_GAIN];
+ double sum;
+
+ /* log gains, mean removal and scaling */
+ posg = 0;
+ for (k=0; k<SUBFRAMES; k++) {
+ tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER+1)*k]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;
+ tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER+1)*k]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;
+ }
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<LPC_GAIN_ORDER; n++) {
+ sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2];
+ pos2 += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs2_g[posg++] = sum;
+ }
+ offsg += LPC_GAIN_ORDER;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2++];
+ pos += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs_g[posg++] = sum;
+ }
+ offs2 += SUBFRAMES;
+ offsg += LPC_GAIN_ORDER;
+ }
+
+
+ /* quantize coefficients */
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+
+ /* get index */
+ pos = WebRtcIsac_kQKltSelIndGain[k];
+ pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
+ index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k];
+ if (index_g[k] < 0) {
+ index_g[k] = 0;
+ }
+ else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) {
+ index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
+ }
+ index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[model][k]+index_g[k];
+
+ /* find quantization levels for coefficients */
+ tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+index_ovr_g[k]];
+
+ /* Save data for creation of multiple bit streams */
+ encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_g[k];
+ }
+
+
+ /* entropy coding of quantization indices - gain */
+ WebRtcIsac_EncHistMulti(streamdata, index_g, WebRtcIsac_kQKltCdfPtrGain[model], KLT_ORDER_GAIN);
+
+ /* find quantization levels for coefficients */
+
+ /* left transform */
+ offsg = 0;
+ posg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ offs2 = 0;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = offsg;
+ pos2 = offs2;
+ for (n=0; n<LPC_GAIN_ORDER; n++)
+ sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2++];
+ tmpcoeffs2_g[posg++] = sum;
+ offs2 += LPC_GAIN_ORDER;
+ }
+ offsg += LPC_GAIN_ORDER;
+ }
+
+ /* right transform */ // Transpose matrix
+ offsg = 0;
+ posg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = j;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2];
+ pos += LPC_GAIN_ORDER;
+ pos2 += SUBFRAMES;
+ }
+ tmpcoeffs_g[posg++] = sum;
+ }
+ offsg += LPC_GAIN_ORDER;
+ }
+
+
+ /* scaling, mean addition, and gain restoration */
+ posg = 0;
+ for (k=0; k<SUBFRAMES; k++) {
+ sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
+ sum += WebRtcIsac_kLpcMeansGain[model][posg];
+ LPCCoef_lo[k*(LPC_LOBAND_ORDER+1)] = exp(sum);
+ pos++;posg++;
+ sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
+ sum += WebRtcIsac_kLpcMeansGain[model][posg];
+ LPCCoef_hi[k*(LPC_HIBAND_ORDER+1)] = exp(sum);
+ pos++;posg++;
+ }
+
+}
+
+void
+WebRtcIsac_EncodeLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata,
+ int* lpcGainIndex)
+{
+ double U[UB_LPC_GAIN_DIM];
+ int idx[UB_LPC_GAIN_DIM];
+ WebRtcIsac_ToLogDomainRemoveMean(lpGains);
+ WebRtcIsac_DecorrelateLPGain(lpGains, U);
+ WebRtcIsac_QuantizeLpcGain(U, idx);
+ // Store the index for re-encoding for FEC.
+ memcpy(lpcGainIndex, idx, UB_LPC_GAIN_DIM * sizeof(int));
+ WebRtcIsac_CorrelateLpcGain(U, lpGains);
+ WebRtcIsac_AddMeanToLinearDomain(lpGains);
+ WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
+}
+
+
+void
+WebRtcIsac_StoreLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata)
+{
+ double U[UB_LPC_GAIN_DIM];
+ int idx[UB_LPC_GAIN_DIM];
+ WebRtcIsac_ToLogDomainRemoveMean(lpGains);
+ WebRtcIsac_DecorrelateLPGain(lpGains, U);
+ WebRtcIsac_QuantizeLpcGain(U, idx);
+ WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
+}
+
+
+
+WebRtc_Word16
+WebRtcIsac_DecodeLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata)
+{
+ double U[UB_LPC_GAIN_DIM];
+ int idx[UB_LPC_GAIN_DIM];
+ int err;
+ err = WebRtcIsac_DecHistOneStepMulti(idx, streamdata,
+ WebRtcIsac_kLpcGainCdfMat, WebRtcIsac_kLpcGainEntropySearch,
+ UB_LPC_GAIN_DIM);
+ if(err < 0)
+ {
+ return -1;
+ }
+ WebRtcIsac_DequantizeLpcGain(idx, U);
+ WebRtcIsac_CorrelateLpcGain(U, lpGains);
+ WebRtcIsac_AddMeanToLinearDomain(lpGains);
+ return 0;
+}
+
+
+
+/* decode & dequantize RC */
+int WebRtcIsac_DecodeRc(Bitstr *streamdata, WebRtc_Word16 *RCQ15)
+{
+ int k, err;
+ int index[AR_ORDER];
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsac_DecHistOneStepMulti(index, streamdata, WebRtcIsac_kQArRcCdfPtr,
+ WebRtcIsac_kQArRcInitIndex, AR_ORDER);
+ if (err<0) // error check
+ return err;
+
+ /* find quantization levels for reflection coefficients */
+ for (k=0; k<AR_ORDER; k++)
+ {
+ RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]);
+ }
+
+ return 0;
+}
+
+
+
+/* quantize & code RC */
+void WebRtcIsac_EncodeRc(WebRtc_Word16 *RCQ15, Bitstr *streamdata)
+{
+ int k;
+ int index[AR_ORDER];
+
+ /* quantize reflection coefficients (add noise feedback?) */
+ for (k=0; k<AR_ORDER; k++)
+ {
+ index[k] = WebRtcIsac_kQArRcInitIndex[k];
+
+ if (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k]])
+ {
+ while (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k] + 1])
+ index[k]++;
+ }
+ else
+ {
+ while (RCQ15[k] < WebRtcIsac_kQArBoundaryLevels[--index[k]]) ;
+ }
+
+ RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]);
+ }
+
+
+ /* entropy coding of quantization indices */
+ WebRtcIsac_EncHistMulti(streamdata, index, WebRtcIsac_kQArRcCdfPtr, AR_ORDER);
+}
+
+
+/* decode & dequantize squared Gain */
+int WebRtcIsac_DecodeGain2(Bitstr *streamdata, WebRtc_Word32 *gainQ10)
+{
+ int index, err;
+
+ /* entropy decoding of quantization index */
+ err = WebRtcIsac_DecHistOneStepMulti(&index, streamdata, WebRtcIsac_kQGainCdf_ptr,
+ WebRtcIsac_kQGainInitIndex, 1);
+ if (err<0) // error check
+ return err;
+
+ /* find quantization level */
+ *gainQ10 = WebRtcIsac_kQGain2Levels[index];
+
+ return 0;
+}
+
+
+
+/* quantize & code squared Gain */
+int WebRtcIsac_EncodeGain2(WebRtc_Word32 *gainQ10, Bitstr *streamdata)
+{
+ int index;
+
+
+ /* find quantization index */
+ index = WebRtcIsac_kQGainInitIndex[0];
+ if (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index])
+ {
+ while (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index + 1])
+ index++;
+ }
+ else
+ {
+ while (*gainQ10 < WebRtcIsac_kQGain2BoundaryLevels[--index]) ;
+ }
+
+ /* dequantize */
+ *gainQ10 = WebRtcIsac_kQGain2Levels[index];
+
+
+ /* entropy coding of quantization index */
+ WebRtcIsac_EncHistMulti(streamdata, &index, WebRtcIsac_kQGainCdf_ptr, 1);
+
+ return 0;
+}
+
+
+/* code and decode Pitch Gains and Lags functions */
+
+/* decode & dequantize Pitch Gains */
+int WebRtcIsac_DecodePitchGain(Bitstr *streamdata, WebRtc_Word16 *PitchGains_Q12)
+{
+ int index_comb, err;
+ const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
+
+ /* entropy decoding of quantization indices */
+ *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
+ err = WebRtcIsac_DecHistBisectMulti(&index_comb, streamdata, WebRtcIsac_kQPitchGainCdf_ptr, WebRtcIsac_kQCdfTableSizeGain, 1);
+ /* error check, Q_mean_Gain.. tables are of size 144 */
+ if ((err<0) || (index_comb<0) || (index_comb>144))
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN;
+
+ /* unquantize back to pitch gains by table look-up */
+ PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb];
+ PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb];
+ PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb];
+ PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb];
+
+ return 0;
+}
+
+
+/* quantize & code Pitch Gains */
+void WebRtcIsac_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12, Bitstr *streamdata, ISAC_SaveEncData_t* encData)
+{
+ int k,j;
+ double C;
+ double S[PITCH_SUBFRAMES];
+ int index[3];
+ int index_comb;
+ const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
+ double PitchGains[PITCH_SUBFRAMES] = {0,0,0,0};
+
+ /* take the asin */
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ {
+ PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
+ S[k] = asin(PitchGains[k]);
+ }
+
+
+ /* find quantization index; only for the first three transform coefficients */
+ for (k=0; k<3; k++)
+ {
+ /* transform */
+ C = 0.0;
+ for (j=0; j<PITCH_SUBFRAMES; j++)
+ C += WebRtcIsac_kTransform[k][j] * S[j];
+
+ /* quantize */
+ index[k] = WebRtcIsac_lrint(C / PITCH_GAIN_STEPSIZE);
+
+ /* check that the index is not outside the boundaries of the table */
+ if (index[k] < WebRtcIsac_kIndexLowerLimitGain[k]) index[k] = WebRtcIsac_kIndexLowerLimitGain[k];
+ else if (index[k] > WebRtcIsac_kIndexUpperLimitGain[k]) index[k] = WebRtcIsac_kIndexUpperLimitGain[k];
+ index[k] -= WebRtcIsac_kIndexLowerLimitGain[k];
+ }
+
+ /* calculate unique overall index */
+ index_comb = WebRtcIsac_kIndexMultsGain[0] * index[0] + WebRtcIsac_kIndexMultsGain[1] * index[1] + index[2];
+
+ /* unquantize back to pitch gains by table look-up */
+ PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb];
+ PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb];
+ PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb];
+ PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb];
+
+ /* entropy coding of quantization pitch gains */
+ *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
+ WebRtcIsac_EncHistMulti(streamdata, &index_comb, WebRtcIsac_kQPitchGainCdf_ptr, 1);
+ encData->pitchGain_index[encData->startIdx] = index_comb;
+
+}
+
+
+
+/* Pitch LAG */
+
+
+/* decode & dequantize Pitch Lags */
+int WebRtcIsac_DecodePitchLag(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12, double *PitchLags)
+{
+ int k, err;
+ double StepSize;
+ double C;
+ int index[PITCH_SUBFRAMES];
+ double mean_gain;
+ const double *mean_val2, *mean_val3, *mean_val4;
+ const WebRtc_Word16 *lower_limit;
+ const WebRtc_UWord16 *init_index;
+ const WebRtc_UWord16 *cdf_size;
+ const WebRtc_UWord16 **cdf;
+
+ //(Y)
+ double PitchGain[4]={0,0,0,0};
+ //
+
+ /* compute mean pitch gain */
+ mean_gain = 0.0;
+ for (k = 0; k < 4; k++)
+ {
+ //(Y)
+ PitchGain[k] = ((float)PitchGain_Q12[k])/4096;
+ //(Y)
+ mean_gain += PitchGain[k];
+ }
+ mean_gain /= 4.0;
+
+ /* voicing classificiation */
+ if (mean_gain < 0.2) {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeLo;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
+ cdf_size = WebRtcIsac_kQPitchLagCdfSizeLo;
+ mean_val2 = WebRtcIsac_kQMeanLag2Lo;
+ mean_val3 = WebRtcIsac_kQMeanLag3Lo;
+ mean_val4 = WebRtcIsac_kQMeanLag4Lo;
+ lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo;
+ init_index = WebRtcIsac_kQInitIndexLagLo;
+ } else if (mean_gain < 0.4) {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeMid;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
+ cdf_size = WebRtcIsac_kQPitchLagCdfSizeMid;
+ mean_val2 = WebRtcIsac_kQMeanLag2Mid;
+ mean_val3 = WebRtcIsac_kQMeanLag3Mid;
+ mean_val4 = WebRtcIsac_kQMeanLag4Mid;
+ lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid;
+ init_index = WebRtcIsac_kQInitIndexLagMid;
+ } else {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeHi;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
+ cdf_size = WebRtcIsac_kQPitchLagCdfSizeHi;
+ mean_val2 = WebRtcIsac_kQMeanLag2Hi;
+ mean_val3 = WebRtcIsac_kQMeanLag3Hi;
+ mean_val4 = WebRtcIsac_kQMeanLag4Hi;
+ lower_limit = WebRtcIsac_kQindexLowerLimitLagHi;
+ init_index = WebRtcIsac_kQInitIndexLagHi;
+ }
+
+ /* entropy decoding of quantization indices */
+ err = WebRtcIsac_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1);
+ if ((err<0) || (index[0]<0)) // error check
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
+
+ err = WebRtcIsac_DecHistOneStepMulti(index+1, streamdata, cdf+1, init_index, 3);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
+
+
+ /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
+ C = (index[0] + lower_limit[0]) * StepSize;
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C;
+ C = mean_val2[index[1]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C;
+ C = mean_val3[index[2]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C;
+ C = mean_val4[index[3]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C;
+
+ return 0;
+}
+
+
+
+/* quantize & code Pitch Lags */
+void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData)
+{
+ int k, j;
+ double StepSize;
+ double C;
+ int index[PITCH_SUBFRAMES];
+ double mean_gain;
+ const double *mean_val2, *mean_val3, *mean_val4;
+ const WebRtc_Word16 *lower_limit, *upper_limit;
+ const WebRtc_UWord16 **cdf;
+
+ //(Y)
+ double PitchGain[4]={0,0,0,0};
+ //
+
+ /* compute mean pitch gain */
+ mean_gain = 0.0;
+ for (k = 0; k < 4; k++)
+ {
+ //(Y)
+ PitchGain[k] = ((float)PitchGain_Q12[k])/4096;
+ //(Y)
+ mean_gain += PitchGain[k];
+ }
+ mean_gain /= 4.0;
+
+ /* Save data for creation of multiple bit streams */
+ encData->meanGain[encData->startIdx] = mean_gain;
+
+ /* voicing classification */
+ if (mean_gain < 0.2) {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeLo;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
+ mean_val2 = WebRtcIsac_kQMeanLag2Lo;
+ mean_val3 = WebRtcIsac_kQMeanLag3Lo;
+ mean_val4 = WebRtcIsac_kQMeanLag4Lo;
+ lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo;
+ upper_limit = WebRtcIsac_kQIndexUpperLimitLagLo;
+ } else if (mean_gain < 0.4) {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeMid;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
+ mean_val2 = WebRtcIsac_kQMeanLag2Mid;
+ mean_val3 = WebRtcIsac_kQMeanLag3Mid;
+ mean_val4 = WebRtcIsac_kQMeanLag4Mid;
+ lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid;
+ upper_limit = WebRtcIsac_kQIndexUpperLimitLagMid;
+ } else {
+ StepSize = WebRtcIsac_kQPitchLagStepsizeHi;
+ cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
+ mean_val2 = WebRtcIsac_kQMeanLag2Hi;
+ mean_val3 = WebRtcIsac_kQMeanLag3Hi;
+ mean_val4 = WebRtcIsac_kQMeanLag4Hi;
+ lower_limit = WebRtcIsac_kQindexLowerLimitLagHi;
+ upper_limit = WebRtcIsac_kQindexUpperLimitLagHi;
+ }
+
+
+ /* find quantization index */
+ for (k=0; k<4; k++)
+ {
+ /* transform */
+ C = 0.0;
+ for (j=0; j<PITCH_SUBFRAMES; j++)
+ C += WebRtcIsac_kTransform[k][j] * PitchLags[j];
+
+ /* quantize */
+ index[k] = WebRtcIsac_lrint(C / StepSize);
+
+ /* check that the index is not outside the boundaries of the table */
+ if (index[k] < lower_limit[k]) index[k] = lower_limit[k];
+ else if (index[k] > upper_limit[k]) index[k] = upper_limit[k];
+ index[k] -= lower_limit[k];
+
+ /* Save data for creation of multiple bit streams */
+ encData->pitchIndex[PITCH_SUBFRAMES*encData->startIdx + k] = index[k];
+ }
+
+ /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
+ C = (index[0] + lower_limit[0]) * StepSize;
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C;
+ C = mean_val2[index[1]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C;
+ C = mean_val3[index[2]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C;
+ C = mean_val4[index[3]];
+ for (k=0; k<PITCH_SUBFRAMES; k++)
+ PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C;
+
+
+ /* entropy coding of quantization pitch lags */
+ WebRtcIsac_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES);
+
+}
+
+
+
+/* Routines for in-band signaling of bandwidth estimation */
+/* Histograms based on uniform distribution of indices */
+/* Move global variables later! */
+
+
+/* cdf array for frame length indicator */
+const WebRtc_UWord16 WebRtcIsac_kFrameLengthCdf[4] = {
+ 0, 21845, 43690, 65535};
+
+/* pointer to cdf array for frame length indicator */
+const WebRtc_UWord16 *WebRtcIsac_kFrameLengthCdf_ptr[1] = {WebRtcIsac_kFrameLengthCdf};
+
+/* initial cdf index for decoder of frame length indicator */
+const WebRtc_UWord16 WebRtcIsac_kFrameLengthInitIndex[1] = {1};
+
+
+int WebRtcIsac_DecodeFrameLen(Bitstr *streamdata,
+ WebRtc_Word16 *framesamples)
+{
+
+ int frame_mode, err;
+
+ err = 0;
+ /* entropy decoding of frame length [1:30ms,2:60ms] */
+ err = WebRtcIsac_DecHistOneStepMulti(&frame_mode, streamdata, WebRtcIsac_kFrameLengthCdf_ptr, WebRtcIsac_kFrameLengthInitIndex, 1);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH;
+
+ switch(frame_mode) {
+ case 1:
+ *framesamples = 480; /* 30ms */
+ break;
+ case 2:
+ *framesamples = 960; /* 60ms */
+ break;
+ default:
+ err = -ISAC_DISALLOWED_FRAME_MODE_DECODER;
+ }
+
+ return err;
+}
+
+int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr *streamdata) {
+
+ int frame_mode, status;
+
+ status = 0;
+ frame_mode = 0;
+ /* entropy coding of frame length [1:480 samples,2:960 samples] */
+ switch(framesamples) {
+ case 480:
+ frame_mode = 1;
+ break;
+ case 960:
+ frame_mode = 2;
+ break;
+ default:
+ status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER;
+ }
+
+ if (status < 0)
+ return status;
+
+ WebRtcIsac_EncHistMulti(streamdata, &frame_mode, WebRtcIsac_kFrameLengthCdf_ptr, 1);
+
+ return status;
+}
+
+/* cdf array for estimated bandwidth */
+static const WebRtc_UWord16 kBwCdf[25] = {
+ 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
+ 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
+ 62804, 65535};
+
+/* pointer to cdf array for estimated bandwidth */
+static const WebRtc_UWord16 *kBwCdfPtr[1] = { kBwCdf };
+
+/* initial cdf index for decoder of estimated bandwidth*/
+static const WebRtc_UWord16 kBwInitIndex[1] = { 7 };
+
+
+int WebRtcIsac_DecodeSendBW(Bitstr *streamdata, WebRtc_Word16 *BWno) {
+
+ int BWno32, err;
+
+ /* entropy decoding of sender's BW estimation [0..23] */
+ err = WebRtcIsac_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1);
+ if (err<0) // error check
+ return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
+ *BWno = (WebRtc_Word16)BWno32;
+ return err;
+
+}
+
+void WebRtcIsac_EncodeReceiveBw(int *BWno, Bitstr *streamdata) {
+
+ /* entropy encoding of receiver's BW estimation [0..23] */
+ WebRtcIsac_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1);
+
+}
+
+
+/* estimate code length of LPC Coef */
+void WebRtcIsac_TranscodeLPCCoef(double *LPCCoef_lo, double *LPCCoef_hi, int model,
+ int *index_g) {
+
+ int j, k, n, pos, pos2, posg, offsg, offs2;
+ int index_ovr_g[KLT_ORDER_GAIN];
+ double tmpcoeffs_g[KLT_ORDER_GAIN];
+ double tmpcoeffs2_g[KLT_ORDER_GAIN];
+ double sum;
+
+ /* log gains, mean removal and scaling */
+ posg = 0;
+ for (k=0; k<SUBFRAMES; k++) {
+ tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER+1)*k]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;
+ tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER+1)*k]);
+ tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
+ tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
+ posg++;
+ }
+
+ /* KLT */
+
+ /* left transform */
+ offsg = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = offsg;
+ pos2 = k;
+ for (n=0; n<LPC_GAIN_ORDER; n++) {
+ sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2];
+ pos2 += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs2_g[posg++] = sum;
+ }
+ offsg += LPC_GAIN_ORDER;
+ }
+
+ /* right transform */
+ offsg = 0;
+ offs2 = 0;
+ for (j=0; j<SUBFRAMES; j++) {
+ posg = offsg;
+ for (k=0; k<LPC_GAIN_ORDER; k++) {
+ sum = 0;
+ pos = k;
+ pos2 = offs2;
+ for (n=0; n<SUBFRAMES; n++) {
+ sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2++];
+ pos += LPC_GAIN_ORDER;
+ }
+ tmpcoeffs_g[posg++] = sum;
+ }
+ offs2 += SUBFRAMES;
+ offsg += LPC_GAIN_ORDER;
+ }
+
+
+ /* quantize coefficients */
+ for (k=0; k<KLT_ORDER_GAIN; k++) {
+
+ /* get index */
+ pos = WebRtcIsac_kQKltSelIndGain[k];
+ pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
+ index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k];
+ if (index_g[k] < 0) {
+ index_g[k] = 0;
+ }
+ else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) {
+ index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
+ }
+ index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[model][k]+index_g[k];
+
+ /* find quantization levels for coefficients */
+ tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+index_ovr_g[k]];
+ }
+}
+
+
+/* decode & dequantize LPC Coef */
+int
+WebRtcIsac_DecodeLpcCoefUB(
+ Bitstr* streamdata,
+ double* lpcVecs,
+ double* percepFilterGains,
+ WebRtc_Word16 bandwidth)
+{
+ int index_s[KLT_ORDER_SHAPE];
+
+ double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ int err;
+
+ /* entropy decoding of quantization indices */
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata,
+ WebRtcIsac_kLpcShapeCdfMatUb12, WebRtcIsac_kLpcShapeEntropySearchUb12,
+ UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
+ break;
+ }
+ case isac16kHz:
+ {
+ err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata,
+ WebRtcIsac_kLpcShapeCdfMatUb16, WebRtcIsac_kLpcShapeEntropySearchUb16,
+ UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if (err<0) // error check
+ {
+ return err;
+ }
+
+ WebRtcIsac_DequantizeLpcParam(index_s, lpcVecs, bandwidth);
+ WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth);
+ WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth);
+ WebRtcIsac_AddLarMean(lpcVecs, bandwidth);
+
+
+ WebRtcIsac_DecodeLpcGainUb(percepFilterGains, streamdata);
+
+ if(bandwidth == isac16kHz)
+ {
+ // decode another set of Gains
+ WebRtcIsac_DecodeLpcGainUb(&percepFilterGains[SUBFRAMES], streamdata);
+ }
+
+ return 0;
+}
+
+WebRtc_Word16
+WebRtcIsac_EncodeBandwidth(
+ enum ISACBandwidth bandwidth,
+ Bitstr* streamData)
+{
+ int bandwidthMode;
+ switch(bandwidth)
+ {
+ case isac12kHz:
+ {
+ bandwidthMode = 0;
+ break;
+ }
+ case isac16kHz:
+ {
+ bandwidthMode = 1;
+ break;
+ }
+ default:
+ return -ISAC_DISALLOWED_ENCODER_BANDWIDTH;
+ }
+
+ WebRtcIsac_EncHistMulti(streamData, &bandwidthMode,
+ kOneBitEqualProbCdf_ptr, 1);
+ return 0;
+}
+
+WebRtc_Word16
+WebRtcIsac_DecodeBandwidth(
+ Bitstr* streamData,
+ enum ISACBandwidth* bandwidth)
+{
+ int bandwidthMode;
+
+ if(WebRtcIsac_DecHistOneStepMulti(&bandwidthMode, streamData,
+ kOneBitEqualProbCdf_ptr,
+ kOneBitEqualProbInitIndex, 1) < 0)
+ {
+ // error check
+ return -ISAC_RANGE_ERROR_DECODE_BANDWITH;
+ }
+
+ switch(bandwidthMode)
+ {
+ case 0:
+ {
+ *bandwidth = isac12kHz;
+ break;
+ }
+ case 1:
+ {
+ *bandwidth = isac16kHz;
+ break;
+ }
+ default:
+ return -ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER;
+ }
+ return 0;
+}
+
+WebRtc_Word16
+WebRtcIsac_EncodeJitterInfo(
+ WebRtc_Word32 jitterIndex,
+ Bitstr* streamData)
+{
+ // This is to avoid LINUX warning until we change 'int' to
+ // 'Word32'
+ int intVar;
+
+ if((jitterIndex < 0) || (jitterIndex > 1))
+ {
+ return -1;
+ }
+ intVar = (int)(jitterIndex);
+ // Use the same CDF table as for bandwidth
+ // both take two values with equal probability
+ WebRtcIsac_EncHistMulti(streamData, &intVar,
+ kOneBitEqualProbCdf_ptr, 1);
+ return 0;
+
+}
+
+WebRtc_Word16
+WebRtcIsac_DecodeJitterInfo(
+ Bitstr* streamData,
+ WebRtc_Word32* jitterInfo)
+{
+ int intVar;
+
+ // Use the same CDF table as for bandwidth
+ // both take two values with equal probability
+ if(WebRtcIsac_DecHistOneStepMulti(&intVar, streamData,
+ kOneBitEqualProbCdf_ptr,
+ kOneBitEqualProbInitIndex, 1) < 0)
+ {
+ // error check
+ return -ISAC_RANGE_ERROR_DECODE_BANDWITH;
+ }
+ *jitterInfo = (WebRtc_Word16)(intVar);
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h
new file mode 100644
index 0000000..8446bcf
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h
@@ -0,0 +1,412 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * entropy_coding.h
+ *
+ * This header file declares all of the functions used to arithmetically
+ * encode the iSAC bistream
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
+
+#include "structs.h"
+
+/* decode complex spectrum (return number of bytes in stream) */
+int WebRtcIsac_DecodeSpecLb(Bitstr *streamdata,
+ double *fr,
+ double *fi,
+ WebRtc_Word16 AvgPitchGain_Q12);
+
+/******************************************************************************
+ * WebRtcIsac_DecodeSpecUB16()
+ * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
+ * This function is called when the codec is in 0-16 kHz bandwidth.
+ * The decoded DFT coefficient can be transformed to time domain by
+ * WebRtcIsac_Time2Spec().
+ *
+ * Input:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are written to.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are written to.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_DecodeSpecUB16(
+ Bitstr* streamdata,
+ double* fr,
+ double* fi);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeSpecUB12()
+ * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
+ * This function is called when the codec is in 0-12 kHz bandwidth.
+ * The decoded DFT coefficient can be transformed to time domain by
+ * WebRtcIsac_Time2Spec().
+ *
+ * Input:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are written to.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are written to.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_DecodeSpecUB12(
+ Bitstr* streamdata,
+ double* fr,
+ double* fi);
+
+
+/* encode complex spectrum */
+int WebRtcIsac_EncodeSpecLb(const WebRtc_Word16* fr,
+ const WebRtc_Word16* fi,
+ Bitstr* streamdata,
+ WebRtc_Word16 AvgPitchGain_Q12);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeSpecUB16()
+ * Quantize and encode real and imaginary part of the DFT coefficients.
+ * This function is called when the codec is in 0-16 kHz bandwidth.
+ * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
+ *
+ *
+ * Input:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are stored.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are stored.
+ *
+ * Output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_EncodeSpecUB16(
+ const WebRtc_Word16* fr,
+ const WebRtc_Word16* fi,
+ Bitstr* streamdata);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeSpecUB12()
+ * Quantize and encode real and imaginary part of the DFT coefficients.
+ * This function is called when the codec is in 0-12 kHz bandwidth.
+ * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
+ *
+ *
+ * Input:
+ * -*fr : pointer to a buffer where the real part of DFT
+ * coefficients are stored.
+ * -*fi : pointer to a buffer where the imaginary part
+ * of DFT coefficients are stored.
+ *
+ * Output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Return value : < 0 if an error occures
+ * 0 if succeeded.
+ */
+int WebRtcIsac_EncodeSpecUB12(
+ const WebRtc_Word16* fr,
+ const WebRtc_Word16* fi,
+ Bitstr* streamdata);
+
+
+/* decode & dequantize LPC Coef */
+int WebRtcIsac_DecodeLpcCoef(Bitstr *streamdata, double *LPCCoef, int *outmodel);
+int WebRtcIsac_DecodeLpcCoefUB(
+ Bitstr* streamdata,
+ double* lpcVecs,
+ double* percepFilterGains,
+ WebRtc_Word16 bandwidth);
+
+int WebRtcIsac_DecodeLpc(Bitstr *streamdata, double *LPCCoef_lo, double *LPCCoef_hi, int *outmodel);
+
+/* quantize & code LPC Coef */
+void WebRtcIsac_EncodeLpcLb(double *LPCCoef_lo, double *LPCCoef_hi, int *model, double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
+void WebRtcIsac_EncodeLpcGainLb(double *LPCCoef_lo, double *LPCCoef_hi, int model, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
+
+/******************************************************************************
+ * WebRtcIsac_EncodeLpcUB()
+ * Encode LPC parameters, given as A-polynomial, of upper-band. The encoding
+ * is performed in LAR domain.
+ * For the upper-band, we compute and encode LPC of some sub-frames, LPC of
+ * other sub-frames are computed by linear interpolation, in LAR domain. This
+ * function performs the interpolation and returns the LPC of all sub-frames.
+ *
+ * Inputs:
+ * - lpcCoef : a buffer containing A-polynomials of sub-frames
+ * (excluding first coefficient that is 1).
+ * - bandwidth : specifies if the codec is operating at 0-12 kHz
+ * or 0-16 kHz mode.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - interpolLPCCoeff : Decoded and interpolated LPC (A-polynomial)
+ * of all sub-frames.
+ * If LP analysis is of order K, and there are N
+ * sub-frames then this is a buffer of size
+ * (k + 1) * N, each vector starts with the LPC gain
+ * of the corresponding sub-frame. The LPC gains
+ * are encoded and inserted after this function is
+ * called. The first A-coefficient which is 1 is not
+ * included.
+ *
+ * Return value : 0 if encoding is successful,
+ * <0 if failed to encode.
+ */
+WebRtc_Word16 WebRtcIsac_EncodeLpcUB(
+ double* lpcCoeff,
+ Bitstr* streamdata,
+ double* interpolLPCCoeff,
+ WebRtc_Word16 bandwidth,
+ ISACUBSaveEncDataStruct* encData);
+
+/******************************************************************************
+ * WebRtcIsac_DecodeInterpolLpcUb()
+ * Decode LPC coefficients and interpolate to get the coefficients fo all
+ * sub-frmaes.
+ *
+ * Inputs:
+ * - bandwidth : spepecifies if the codec is in 0-12 kHz or
+ * 0-16 kHz mode.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - percepFilterParam : Decoded and interpolated LPC (A-polynomial) of
+ * all sub-frames.
+ * If LP analysis is of order K, and there are N
+ * sub-frames then this is a buffer of size
+ * (k + 1) * N, each vector starts with the LPC gain
+ * of the corresponding sub-frame. The LPC gains
+ * are encoded and inserted after this function is
+ * called. The first A-coefficient which is 1 is not
+ * included.
+ *
+ * Return value : 0 if encoding is successful,
+ * <0 if failed to encode.
+ */
+WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(
+ Bitstr* streamdata,
+ double* percepFilterParam,
+ WebRtc_Word16 bandwidth);
+
+/* decode & dequantize RC */
+int WebRtcIsac_DecodeRc(Bitstr *streamdata, WebRtc_Word16 *RCQ15);
+
+/* quantize & code RC */
+void WebRtcIsac_EncodeRc(WebRtc_Word16 *RCQ15, Bitstr *streamdata);
+
+/* decode & dequantize squared Gain */
+int WebRtcIsac_DecodeGain2(Bitstr *streamdata, WebRtc_Word32 *Gain2);
+
+/* quantize & code squared Gain (input is squared gain) */
+int WebRtcIsac_EncodeGain2(WebRtc_Word32 *gain2, Bitstr *streamdata);
+
+void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
+
+void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
+
+int WebRtcIsac_DecodePitchGain(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12);
+int WebRtcIsac_DecodePitchLag(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12, double *PitchLag);
+
+int WebRtcIsac_DecodeFrameLen(Bitstr *streamdata, WebRtc_Word16 *framelength);
+int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framelength, Bitstr *streamdata);
+int WebRtcIsac_DecodeSendBW(Bitstr *streamdata, WebRtc_Word16 *BWno);
+void WebRtcIsac_EncodeReceiveBw(int *BWno, Bitstr *streamdata);
+
+/* step-down */
+void WebRtcIsac_Poly2Rc(double *a, int N, double *RC);
+
+/* step-up */
+void WebRtcIsac_Rc2Poly(double *RC, int N, double *a);
+
+void WebRtcIsac_TranscodeLPCCoef(double *LPCCoef_lo, double *LPCCoef_hi, int model,
+ int *index_g);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeLpcGainUb()
+ * Encode LPC gains of sub-Frames.
+ *
+ * Input/outputs:
+ * - lpGains : a buffer which contains 'SUBFRAME' number of
+ * LP gains to be encoded. The input values are
+ * overwritten by the quantized values.
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - lpcGainIndex : quantization indices for lpc gains, these will
+ * be stored to be used for FEC.
+ */
+void WebRtcIsac_EncodeLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata,
+ int* lpcGainIndex);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeLpcGainUb()
+ * Store LPC gains of sub-Frames in 'streamdata'.
+ *
+ * Input:
+ * - lpGains : a buffer which contains 'SUBFRAME' number of
+ * LP gains to be encoded.
+ * Input/outputs:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ */
+void WebRtcIsac_StoreLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeLpcGainUb()
+ * Decode the LPC gain of sub-frames.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - lpGains : a buffer where decoded LPC gians will be stored.
+ *
+ * Return value : 0 if succeeded.
+ * <0 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(
+ double* lpGains,
+ Bitstr* streamdata);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeBandwidth()
+ * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz.
+ *
+ * Input:
+ * - bandwidth : an enumerator specifying if the codec in is
+ * 0-12 kHz or 0-16 kHz mode.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Return value : 0 if succeeded.
+ * <0 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_EncodeBandwidth(
+ enum ISACBandwidth bandwidth,
+ Bitstr* streamData);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeBandwidth()
+ * Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz
+ * or 0-16 kHz.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - bandwidth : an enumerator specifying if the codec is in
+ * 0-12 kHz or 0-16 kHz mode.
+ *
+ * Return value : 0 if succeeded.
+ * <0 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_DecodeBandwidth(
+ Bitstr* streamData,
+ enum ISACBandwidth* bandwidth);
+
+
+/******************************************************************************
+ * WebRtcIsac_EncodeJitterInfo()
+ * Decode the jitter information.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Input:
+ * - jitterInfo : one bit of info specifying if the channel is
+ * in high/low jitter. Zero indicates low jitter
+ * and one indicates high jitter.
+ *
+ * Return value : 0 if succeeded.
+ * <0 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(
+ WebRtc_Word32 jitterIndex,
+ Bitstr* streamData);
+
+
+/******************************************************************************
+ * WebRtcIsac_DecodeJitterInfo()
+ * Decode the jitter information.
+ *
+ * Input/output:
+ * - streamdata : pointer to a stucture containg the encoded
+ * data and theparameters needed for entropy
+ * coding.
+ *
+ * Output:
+ * - jitterInfo : one bit of info specifying if the channel is
+ * in high/low jitter. Zero indicates low jitter
+ * and one indicates high jitter.
+ *
+ * Return value : 0 if succeeded.
+ * <0 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(
+ Bitstr* streamData,
+ WebRtc_Word32* jitterInfo);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c
new file mode 100644
index 0000000..c824798
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c
@@ -0,0 +1,947 @@
+/*
+ * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
+ * Queen's Univ at Kingston (Canada)
+ *
+ * Permission to use, copy, modify, and distribute this software for
+ * any purpose without fee is hereby granted, provided that this
+ * entire notice is included in all copies of any software which is
+ * or includes a copy or modification of this software and in all
+ * copies of the supporting documentation for such software.
+ *
+ * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTY. IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
+ * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
+ * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
+ * FITNESS FOR ANY PARTICULAR PURPOSE.
+ *
+ * All of which is to say that you can do what you like with this
+ * source code provided you don't try to sell it as your own and you
+ * include an unaltered copy of this message (including the
+ * copyright).
+ *
+ * It is also implicitly understood that bug fixes and improvements
+ * should make their way back to the general Internet community so
+ * that everyone benefits.
+ *
+ * Changes:
+ * Trivial type modifications by the WebRTC authors.
+ */
+
+
+/*
+ * File:
+ * WebRtcIsac_Fftn.c
+ *
+ * Public:
+ * WebRtcIsac_Fftn / fftnf ();
+ *
+ * Private:
+ * WebRtcIsac_Fftradix / fftradixf ();
+ *
+ * Descript:
+ * multivariate complex Fourier transform, computed in place
+ * using mixed-radix Fast Fourier Transform algorithm.
+ *
+ * Fortran code by:
+ * RC Singleton, Stanford Research Institute, Sept. 1968
+ *
+ * translated by f2c (version 19950721).
+ *
+ * int WebRtcIsac_Fftn (int ndim, const int dims[], REAL Re[], REAL Im[],
+ * int iSign, double scaling);
+ *
+ * NDIM = the total number dimensions
+ * DIMS = a vector of array sizes
+ * if NDIM is zero then DIMS must be zero-terminated
+ *
+ * RE and IM hold the real and imaginary components of the data, and return
+ * the resulting real and imaginary Fourier coefficients. Multidimensional
+ * data *must* be allocated contiguously. There is no limit on the number
+ * of dimensions.
+ *
+ * ISIGN = the sign of the complex exponential (ie, forward or inverse FFT)
+ * the magnitude of ISIGN (normally 1) is used to determine the
+ * correct indexing increment (see below).
+ *
+ * SCALING = normalizing constant by which the final result is *divided*
+ * if SCALING == -1, normalize by total dimension of the transform
+ * if SCALING < -1, normalize by the square-root of the total dimension
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * int dims[3] = {n1,n2,n3}
+ * WebRtcIsac_Fftn (3, dims, Re, Im, 1, scaling);
+ *
+ *-----------------------------------------------------------------------*
+ * int WebRtcIsac_Fftradix (REAL Re[], REAL Im[], size_t nTotal, size_t nPass,
+ * size_t nSpan, int iSign, size_t max_factors,
+ * size_t max_perm);
+ *
+ * RE, IM - see above documentation
+ *
+ * Although there is no limit on the number of dimensions, WebRtcIsac_Fftradix() must
+ * be called once for each dimension, but the calls may be in any order.
+ *
+ * NTOTAL = the total number of complex data values
+ * NPASS = the dimension of the current variable
+ * NSPAN/NPASS = the spacing of consecutive data values while indexing the
+ * current variable
+ * ISIGN - see above documentation
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n1, n1, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n2, n1*n2, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n3, n1*n2*n3, 1, maxf, maxp);
+ *
+ * single-variate transform,
+ * NTOTAL = N = NSPAN = (number of complex data values),
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n, n, n, 1, maxf, maxp);
+ *
+ * The data can also be stored in a single array with alternating real and
+ * imaginary parts, the magnitude of ISIGN is changed to 2 to give correct
+ * indexing increment, and data [0] and data [1] used to pass the initial
+ * addresses for the sequences of real and imaginary values,
+ *
+ * example:
+ * REAL data [2*NTOTAL];
+ * WebRtcIsac_Fftradix ( &data[0], &data[1], NTOTAL, nPass, nSpan, 2, maxf, maxp);
+ *
+ * for temporary allocation:
+ *
+ * MAX_FACTORS >= the maximum prime factor of NPASS
+ * MAX_PERM >= the number of prime factors of NPASS. In addition,
+ * if the square-free portion K of NPASS has two or more prime
+ * factors, then MAX_PERM >= (K-1)
+ *
+ * storage in FACTOR for a maximum of 15 prime factors of NPASS. if NPASS
+ * has more than one square-free factor, the product of the square-free
+ * factors must be <= 210 array storage for maximum prime factor of 23 the
+ * following two constants should agree with the array dimensions.
+ *
+ *----------------------------------------------------------------------*/
+#include "fft.h"
+
+#include <stdlib.h>
+#include <math.h>
+
+
+
+/* double precision routine */
+static int
+WebRtcIsac_Fftradix (double Re[], double Im[],
+ size_t nTotal, size_t nPass, size_t nSpan, int isign,
+ int max_factors, unsigned int max_perm,
+ FFTstr *fftstate);
+
+
+
+#ifndef M_PI
+# define M_PI 3.14159265358979323846264338327950288
+#endif
+
+#ifndef SIN60
+# define SIN60 0.86602540378443865 /* sin(60 deg) */
+# define COS72 0.30901699437494742 /* cos(72 deg) */
+# define SIN72 0.95105651629515357 /* sin(72 deg) */
+#endif
+
+# define REAL double
+# define FFTN WebRtcIsac_Fftn
+# define FFTNS "fftn"
+# define FFTRADIX WebRtcIsac_Fftradix
+# define FFTRADIXS "fftradix"
+
+
+int WebRtcIsac_Fftns(unsigned int ndim, const int dims[],
+ double Re[],
+ double Im[],
+ int iSign,
+ double scaling,
+ FFTstr *fftstate)
+{
+
+ size_t nSpan, nPass, nTotal;
+ unsigned int i;
+ int ret, max_factors, max_perm;
+
+ /*
+ * tally the number of elements in the data array
+ * and determine the number of dimensions
+ */
+ nTotal = 1;
+ if (ndim && dims [0])
+ {
+ for (i = 0; i < ndim; i++)
+ {
+ if (dims [i] <= 0)
+ {
+ return -1;
+ }
+ nTotal *= dims [i];
+ }
+ }
+ else
+ {
+ ndim = 0;
+ for (i = 0; dims [i]; i++)
+ {
+ if (dims [i] <= 0)
+ {
+ return -1;
+ }
+ nTotal *= dims [i];
+ ndim++;
+ }
+ }
+
+ /* determine maximum number of factors and permuations */
+#if 1
+ /*
+ * follow John Beale's example, just use the largest dimension and don't
+ * worry about excess allocation. May be someone else will do it?
+ */
+ max_factors = max_perm = 1;
+ for (i = 0; i < ndim; i++)
+ {
+ nSpan = dims [i];
+ if ((int)nSpan > max_factors)
+ {
+ max_factors = (int)nSpan;
+ }
+ if ((int)nSpan > max_perm)
+ {
+ max_perm = (int)nSpan;
+ }
+ }
+#else
+ /* use the constants used in the original Fortran code */
+ max_factors = 23;
+ max_perm = 209;
+#endif
+ /* loop over the dimensions: */
+ nPass = 1;
+ for (i = 0; i < ndim; i++)
+ {
+ nSpan = dims [i];
+ nPass *= nSpan;
+ ret = FFTRADIX (Re, Im, nTotal, nSpan, nPass, iSign,
+ max_factors, max_perm, fftstate);
+ /* exit, clean-up already done */
+ if (ret)
+ return ret;
+ }
+
+ /* Divide through by the normalizing constant: */
+ if (scaling && scaling != 1.0)
+ {
+ if (iSign < 0) iSign = -iSign;
+ if (scaling < 0.0)
+ {
+ scaling = (double)nTotal;
+ if (scaling < -1.0)
+ scaling = sqrt (scaling);
+ }
+ scaling = 1.0 / scaling; /* multiply is often faster */
+ for (i = 0; i < nTotal; i += iSign)
+ {
+ Re [i] *= scaling;
+ Im [i] *= scaling;
+ }
+ }
+ return 0;
+}
+
+/*
+ * singleton's mixed radix routine
+ *
+ * could move allocation out to WebRtcIsac_Fftn(), but leave it here so that it's
+ * possible to make this a standalone function
+ */
+
+static int FFTRADIX (REAL Re[],
+ REAL Im[],
+ size_t nTotal,
+ size_t nPass,
+ size_t nSpan,
+ int iSign,
+ int max_factors,
+ unsigned int max_perm,
+ FFTstr *fftstate)
+{
+ int ii, mfactor, kspan, ispan, inc;
+ int j, jc, jf, jj, k, k1, k2, k3, k4, kk, kt, nn, ns, nt;
+
+
+ REAL radf;
+ REAL c1, c2, c3, cd, aa, aj, ak, ajm, ajp, akm, akp;
+ REAL s1, s2, s3, sd, bb, bj, bk, bjm, bjp, bkm, bkp;
+
+ REAL *Rtmp = NULL; /* temp space for real part*/
+ REAL *Itmp = NULL; /* temp space for imaginary part */
+ REAL *Cos = NULL; /* Cosine values */
+ REAL *Sin = NULL; /* Sine values */
+
+ REAL s60 = SIN60; /* sin(60 deg) */
+ REAL c72 = COS72; /* cos(72 deg) */
+ REAL s72 = SIN72; /* sin(72 deg) */
+ REAL pi2 = M_PI; /* use PI first, 2 PI later */
+
+
+ fftstate->SpaceAlloced = 0;
+ fftstate->MaxPermAlloced = 0;
+
+
+ // initialize to avoid warnings
+ k3 = c2 = c3 = s2 = s3 = 0.0;
+
+ if (nPass < 2)
+ return 0;
+
+ /* allocate storage */
+ if (fftstate->SpaceAlloced < max_factors * sizeof (REAL))
+ {
+#ifdef SUN_BROKEN_REALLOC
+ if (!fftstate->SpaceAlloced) /* first time */
+ {
+ fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+ }
+ else
+ {
+#endif
+ fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+#ifdef SUN_BROKEN_REALLOC
+ }
+#endif
+ }
+ else
+ {
+ /* allow full use of alloc'd space */
+ max_factors = fftstate->SpaceAlloced / sizeof (REAL);
+ }
+ if (fftstate->MaxPermAlloced < max_perm)
+ {
+#ifdef SUN_BROKEN_REALLOC
+ if (!fftstate->MaxPermAlloced) /* first time */
+ else
+#endif
+ fftstate->MaxPermAlloced = max_perm;
+ }
+ else
+ {
+ /* allow full use of alloc'd space */
+ max_perm = fftstate->MaxPermAlloced;
+ }
+ if (fftstate->Tmp0 == NULL || fftstate->Tmp1 == NULL || fftstate->Tmp2 == NULL || fftstate->Tmp3 == NULL
+ || fftstate->Perm == NULL) {
+ return -1;
+ }
+
+ /* assign pointers */
+ Rtmp = (REAL *) fftstate->Tmp0;
+ Itmp = (REAL *) fftstate->Tmp1;
+ Cos = (REAL *) fftstate->Tmp2;
+ Sin = (REAL *) fftstate->Tmp3;
+
+ /*
+ * Function Body
+ */
+ inc = iSign;
+ if (iSign < 0) {
+ s72 = -s72;
+ s60 = -s60;
+ pi2 = -pi2;
+ inc = -inc; /* absolute value */
+ }
+
+ /* adjust for strange increments */
+ nt = inc * (int)nTotal;
+ ns = inc * (int)nSpan;
+ kspan = ns;
+
+ nn = nt - inc;
+ jc = ns / (int)nPass;
+ radf = pi2 * (double) jc;
+ pi2 *= 2.0; /* use 2 PI from here on */
+
+ ii = 0;
+ jf = 0;
+ /* determine the factors of n */
+ mfactor = 0;
+ k = (int)nPass;
+ while (k % 16 == 0) {
+ mfactor++;
+ fftstate->factor [mfactor - 1] = 4;
+ k /= 16;
+ }
+ j = 3;
+ jj = 9;
+ do {
+ while (k % jj == 0) {
+ mfactor++;
+ fftstate->factor [mfactor - 1] = j;
+ k /= jj;
+ }
+ j += 2;
+ jj = j * j;
+ } while (jj <= k);
+ if (k <= 4) {
+ kt = mfactor;
+ fftstate->factor [mfactor] = k;
+ if (k != 1)
+ mfactor++;
+ } else {
+ if (k - (k / 4 << 2) == 0) {
+ mfactor++;
+ fftstate->factor [mfactor - 1] = 2;
+ k /= 4;
+ }
+ kt = mfactor;
+ j = 2;
+ do {
+ if (k % j == 0) {
+ mfactor++;
+ fftstate->factor [mfactor - 1] = j;
+ k /= j;
+ }
+ j = ((j + 1) / 2 << 1) + 1;
+ } while (j <= k);
+ }
+ if (kt) {
+ j = kt;
+ do {
+ mfactor++;
+ fftstate->factor [mfactor - 1] = fftstate->factor [j - 1];
+ j--;
+ } while (j);
+ }
+
+ /* test that mfactors is in range */
+ if (mfactor > NFACTOR)
+ {
+ return -1;
+ }
+
+ /* compute fourier transform */
+ for (;;) {
+ sd = radf / (double) kspan;
+ cd = sin(sd);
+ cd = 2.0 * cd * cd;
+ sd = sin(sd + sd);
+ kk = 0;
+ ii++;
+
+ switch (fftstate->factor [ii - 1]) {
+ case 2:
+ /* transform for factor of 2 (including rotation factor) */
+ kspan /= 2;
+ k1 = kspan + 2;
+ do {
+ do {
+ k2 = kk + kspan;
+ ak = Re [k2];
+ bk = Im [k2];
+ Re [k2] = Re [kk] - ak;
+ Im [k2] = Im [kk] - bk;
+ Re [kk] += ak;
+ Im [kk] += bk;
+ kk = k2 + kspan;
+ } while (kk < nn);
+ kk -= nn;
+ } while (kk < jc);
+ if (kk >= kspan)
+ goto Permute_Results_Label; /* exit infinite loop */
+ do {
+ c1 = 1.0 - cd;
+ s1 = sd;
+ do {
+ do {
+ do {
+ k2 = kk + kspan;
+ ak = Re [kk] - Re [k2];
+ bk = Im [kk] - Im [k2];
+ Re [kk] += Re [k2];
+ Im [kk] += Im [k2];
+ Re [k2] = c1 * ak - s1 * bk;
+ Im [k2] = s1 * ak + c1 * bk;
+ kk = k2 + kspan;
+ } while (kk < (nt-1));
+ k2 = kk - nt;
+ c1 = -c1;
+ kk = k1 - k2;
+ } while (kk > k2);
+ ak = c1 - (cd * c1 + sd * s1);
+ s1 = sd * c1 - cd * s1 + s1;
+ c1 = 2.0 - (ak * ak + s1 * s1);
+ s1 *= c1;
+ c1 *= ak;
+ kk += jc;
+ } while (kk < k2);
+ k1 += inc + inc;
+ kk = (k1 - kspan + 1) / 2 + jc - 1;
+ } while (kk < (jc + jc));
+ break;
+
+ case 4: /* transform for factor of 4 */
+ ispan = kspan;
+ kspan /= 4;
+
+ do {
+ c1 = 1.0;
+ s1 = 0.0;
+ do {
+ do {
+ k1 = kk + kspan;
+ k2 = k1 + kspan;
+ k3 = k2 + kspan;
+ akp = Re [kk] + Re [k2];
+ akm = Re [kk] - Re [k2];
+ ajp = Re [k1] + Re [k3];
+ ajm = Re [k1] - Re [k3];
+ bkp = Im [kk] + Im [k2];
+ bkm = Im [kk] - Im [k2];
+ bjp = Im [k1] + Im [k3];
+ bjm = Im [k1] - Im [k3];
+ Re [kk] = akp + ajp;
+ Im [kk] = bkp + bjp;
+ ajp = akp - ajp;
+ bjp = bkp - bjp;
+ if (iSign < 0) {
+ akp = akm + bjm;
+ bkp = bkm - ajm;
+ akm -= bjm;
+ bkm += ajm;
+ } else {
+ akp = akm - bjm;
+ bkp = bkm + ajm;
+ akm += bjm;
+ bkm -= ajm;
+ }
+ /* avoid useless multiplies */
+ if (s1 == 0.0) {
+ Re [k1] = akp;
+ Re [k2] = ajp;
+ Re [k3] = akm;
+ Im [k1] = bkp;
+ Im [k2] = bjp;
+ Im [k3] = bkm;
+ } else {
+ Re [k1] = akp * c1 - bkp * s1;
+ Re [k2] = ajp * c2 - bjp * s2;
+ Re [k3] = akm * c3 - bkm * s3;
+ Im [k1] = akp * s1 + bkp * c1;
+ Im [k2] = ajp * s2 + bjp * c2;
+ Im [k3] = akm * s3 + bkm * c3;
+ }
+ kk = k3 + kspan;
+ } while (kk < nt);
+
+ c2 = c1 - (cd * c1 + sd * s1);
+ s1 = sd * c1 - cd * s1 + s1;
+ c1 = 2.0 - (c2 * c2 + s1 * s1);
+ s1 *= c1;
+ c1 *= c2;
+ /* values of c2, c3, s2, s3 that will get used next time */
+ c2 = c1 * c1 - s1 * s1;
+ s2 = 2.0 * c1 * s1;
+ c3 = c2 * c1 - s2 * s1;
+ s3 = c2 * s1 + s2 * c1;
+ kk = kk - nt + jc;
+ } while (kk < kspan);
+ kk = kk - kspan + inc;
+ } while (kk < jc);
+ if (kspan == jc)
+ goto Permute_Results_Label; /* exit infinite loop */
+ break;
+
+ default:
+ /* transform for odd factors */
+#ifdef FFT_RADIX4
+ return -1;
+ break;
+#else /* FFT_RADIX4 */
+ k = fftstate->factor [ii - 1];
+ ispan = kspan;
+ kspan /= k;
+
+ switch (k) {
+ case 3: /* transform for factor of 3 (optional code) */
+ do {
+ do {
+ k1 = kk + kspan;
+ k2 = k1 + kspan;
+ ak = Re [kk];
+ bk = Im [kk];
+ aj = Re [k1] + Re [k2];
+ bj = Im [k1] + Im [k2];
+ Re [kk] = ak + aj;
+ Im [kk] = bk + bj;
+ ak -= 0.5 * aj;
+ bk -= 0.5 * bj;
+ aj = (Re [k1] - Re [k2]) * s60;
+ bj = (Im [k1] - Im [k2]) * s60;
+ Re [k1] = ak - bj;
+ Re [k2] = ak + bj;
+ Im [k1] = bk + aj;
+ Im [k2] = bk - aj;
+ kk = k2 + kspan;
+ } while (kk < (nn - 1));
+ kk -= nn;
+ } while (kk < kspan);
+ break;
+
+ case 5: /* transform for factor of 5 (optional code) */
+ c2 = c72 * c72 - s72 * s72;
+ s2 = 2.0 * c72 * s72;
+ do {
+ do {
+ k1 = kk + kspan;
+ k2 = k1 + kspan;
+ k3 = k2 + kspan;
+ k4 = k3 + kspan;
+ akp = Re [k1] + Re [k4];
+ akm = Re [k1] - Re [k4];
+ bkp = Im [k1] + Im [k4];
+ bkm = Im [k1] - Im [k4];
+ ajp = Re [k2] + Re [k3];
+ ajm = Re [k2] - Re [k3];
+ bjp = Im [k2] + Im [k3];
+ bjm = Im [k2] - Im [k3];
+ aa = Re [kk];
+ bb = Im [kk];
+ Re [kk] = aa + akp + ajp;
+ Im [kk] = bb + bkp + bjp;
+ ak = akp * c72 + ajp * c2 + aa;
+ bk = bkp * c72 + bjp * c2 + bb;
+ aj = akm * s72 + ajm * s2;
+ bj = bkm * s72 + bjm * s2;
+ Re [k1] = ak - bj;
+ Re [k4] = ak + bj;
+ Im [k1] = bk + aj;
+ Im [k4] = bk - aj;
+ ak = akp * c2 + ajp * c72 + aa;
+ bk = bkp * c2 + bjp * c72 + bb;
+ aj = akm * s2 - ajm * s72;
+ bj = bkm * s2 - bjm * s72;
+ Re [k2] = ak - bj;
+ Re [k3] = ak + bj;
+ Im [k2] = bk + aj;
+ Im [k3] = bk - aj;
+ kk = k4 + kspan;
+ } while (kk < (nn-1));
+ kk -= nn;
+ } while (kk < kspan);
+ break;
+
+ default:
+ if (k != jf) {
+ jf = k;
+ s1 = pi2 / (double) k;
+ c1 = cos(s1);
+ s1 = sin(s1);
+ if (jf > max_factors){
+ return -1;
+ }
+ Cos [jf - 1] = 1.0;
+ Sin [jf - 1] = 0.0;
+ j = 1;
+ do {
+ Cos [j - 1] = Cos [k - 1] * c1 + Sin [k - 1] * s1;
+ Sin [j - 1] = Cos [k - 1] * s1 - Sin [k - 1] * c1;
+ k--;
+ Cos [k - 1] = Cos [j - 1];
+ Sin [k - 1] = -Sin [j - 1];
+ j++;
+ } while (j < k);
+ }
+ do {
+ do {
+ k1 = kk;
+ k2 = kk + ispan;
+ ak = aa = Re [kk];
+ bk = bb = Im [kk];
+ j = 1;
+ k1 += kspan;
+ do {
+ k2 -= kspan;
+ j++;
+ Rtmp [j - 1] = Re [k1] + Re [k2];
+ ak += Rtmp [j - 1];
+ Itmp [j - 1] = Im [k1] + Im [k2];
+ bk += Itmp [j - 1];
+ j++;
+ Rtmp [j - 1] = Re [k1] - Re [k2];
+ Itmp [j - 1] = Im [k1] - Im [k2];
+ k1 += kspan;
+ } while (k1 < k2);
+ Re [kk] = ak;
+ Im [kk] = bk;
+ k1 = kk;
+ k2 = kk + ispan;
+ j = 1;
+ do {
+ k1 += kspan;
+ k2 -= kspan;
+ jj = j;
+ ak = aa;
+ bk = bb;
+ aj = 0.0;
+ bj = 0.0;
+ k = 1;
+ do {
+ k++;
+ ak += Rtmp [k - 1] * Cos [jj - 1];
+ bk += Itmp [k - 1] * Cos [jj - 1];
+ k++;
+ aj += Rtmp [k - 1] * Sin [jj - 1];
+ bj += Itmp [k - 1] * Sin [jj - 1];
+ jj += j;
+ if (jj > jf) {
+ jj -= jf;
+ }
+ } while (k < jf);
+ k = jf - j;
+ Re [k1] = ak - bj;
+ Im [k1] = bk + aj;
+ Re [k2] = ak + bj;
+ Im [k2] = bk - aj;
+ j++;
+ } while (j < k);
+ kk += ispan;
+ } while (kk < nn);
+ kk -= nn;
+ } while (kk < kspan);
+ break;
+ }
+
+ /* multiply by rotation factor (except for factors of 2 and 4) */
+ if (ii == mfactor)
+ goto Permute_Results_Label; /* exit infinite loop */
+ kk = jc;
+ do {
+ c2 = 1.0 - cd;
+ s1 = sd;
+ do {
+ c1 = c2;
+ s2 = s1;
+ kk += kspan;
+ do {
+ do {
+ ak = Re [kk];
+ Re [kk] = c2 * ak - s2 * Im [kk];
+ Im [kk] = s2 * ak + c2 * Im [kk];
+ kk += ispan;
+ } while (kk < nt);
+ ak = s1 * s2;
+ s2 = s1 * c2 + c1 * s2;
+ c2 = c1 * c2 - ak;
+ kk = kk - nt + kspan;
+ } while (kk < ispan);
+ c2 = c1 - (cd * c1 + sd * s1);
+ s1 += sd * c1 - cd * s1;
+ c1 = 2.0 - (c2 * c2 + s1 * s1);
+ s1 *= c1;
+ c2 *= c1;
+ kk = kk - ispan + jc;
+ } while (kk < kspan);
+ kk = kk - kspan + jc + inc;
+ } while (kk < (jc + jc));
+ break;
+#endif /* FFT_RADIX4 */
+ }
+ }
+
+ /* permute the results to normal order---done in two stages */
+ /* permutation for square factors of n */
+Permute_Results_Label:
+ fftstate->Perm [0] = ns;
+ if (kt) {
+ k = kt + kt + 1;
+ if (mfactor < k)
+ k--;
+ j = 1;
+ fftstate->Perm [k] = jc;
+ do {
+ fftstate->Perm [j] = fftstate->Perm [j - 1] / fftstate->factor [j - 1];
+ fftstate->Perm [k - 1] = fftstate->Perm [k] * fftstate->factor [j - 1];
+ j++;
+ k--;
+ } while (j < k);
+ k3 = fftstate->Perm [k];
+ kspan = fftstate->Perm [1];
+ kk = jc;
+ k2 = kspan;
+ j = 1;
+ if (nPass != nTotal) {
+ /* permutation for multivariate transform */
+ Permute_Multi_Label:
+ do {
+ do {
+ k = kk + jc;
+ do {
+ /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+ ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+ bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+ kk += inc;
+ k2 += inc;
+ } while (kk < (k-1));
+ kk += ns - jc;
+ k2 += ns - jc;
+ } while (kk < (nt-1));
+ k2 = k2 - nt + kspan;
+ kk = kk - nt + jc;
+ } while (k2 < (ns-1));
+ do {
+ do {
+ k2 -= fftstate->Perm [j - 1];
+ j++;
+ k2 = fftstate->Perm [j] + k2;
+ } while (k2 > fftstate->Perm [j - 1]);
+ j = 1;
+ do {
+ if (kk < (k2-1))
+ goto Permute_Multi_Label;
+ kk += jc;
+ k2 += kspan;
+ } while (k2 < (ns-1));
+ } while (kk < (ns-1));
+ } else {
+ /* permutation for single-variate transform (optional code) */
+ Permute_Single_Label:
+ do {
+ /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+ ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+ bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+ kk += inc;
+ k2 += kspan;
+ } while (k2 < (ns-1));
+ do {
+ do {
+ k2 -= fftstate->Perm [j - 1];
+ j++;
+ k2 = fftstate->Perm [j] + k2;
+ } while (k2 >= fftstate->Perm [j - 1]);
+ j = 1;
+ do {
+ if (kk < k2)
+ goto Permute_Single_Label;
+ kk += inc;
+ k2 += kspan;
+ } while (k2 < (ns-1));
+ } while (kk < (ns-1));
+ }
+ jc = k3;
+ }
+
+ if ((kt << 1) + 1 >= mfactor)
+ return 0;
+ ispan = fftstate->Perm [kt];
+ /* permutation for square-free factors of n */
+ j = mfactor - kt;
+ fftstate->factor [j] = 1;
+ do {
+ fftstate->factor [j - 1] *= fftstate->factor [j];
+ j--;
+ } while (j != kt);
+ kt++;
+ nn = fftstate->factor [kt - 1] - 1;
+ if (nn > (int) max_perm) {
+ return -1;
+ }
+ j = jj = 0;
+ for (;;) {
+ k = kt + 1;
+ k2 = fftstate->factor [kt - 1];
+ kk = fftstate->factor [k - 1];
+ j++;
+ if (j > nn)
+ break; /* exit infinite loop */
+ jj += kk;
+ while (jj >= k2) {
+ jj -= k2;
+ k2 = kk;
+ k++;
+ kk = fftstate->factor [k - 1];
+ jj += kk;
+ }
+ fftstate->Perm [j - 1] = jj;
+ }
+ /* determine the permutation cycles of length greater than 1 */
+ j = 0;
+ for (;;) {
+ do {
+ j++;
+ kk = fftstate->Perm [j - 1];
+ } while (kk < 0);
+ if (kk != j) {
+ do {
+ k = kk;
+ kk = fftstate->Perm [k - 1];
+ fftstate->Perm [k - 1] = -kk;
+ } while (kk != j);
+ k3 = kk;
+ } else {
+ fftstate->Perm [j - 1] = -j;
+ if (j == nn)
+ break; /* exit infinite loop */
+ }
+ }
+ max_factors *= inc;
+ /* reorder a and b, following the permutation cycles */
+ for (;;) {
+ j = k3 + 1;
+ nt -= ispan;
+ ii = nt - inc + 1;
+ if (nt < 0)
+ break; /* exit infinite loop */
+ do {
+ do {
+ j--;
+ } while (fftstate->Perm [j - 1] < 0);
+ jj = jc;
+ do {
+ kspan = jj;
+ if (jj > max_factors) {
+ kspan = max_factors;
+ }
+ jj -= kspan;
+ k = fftstate->Perm [j - 1];
+ kk = jc * k + ii + jj;
+ k1 = kk + kspan - 1;
+ k2 = 0;
+ do {
+ k2++;
+ Rtmp [k2 - 1] = Re [k1];
+ Itmp [k2 - 1] = Im [k1];
+ k1 -= inc;
+ } while (k1 != (kk-1));
+ do {
+ k1 = kk + kspan - 1;
+ k2 = k1 - jc * (k + fftstate->Perm [k - 1]);
+ k = -fftstate->Perm [k - 1];
+ do {
+ Re [k1] = Re [k2];
+ Im [k1] = Im [k2];
+ k1 -= inc;
+ k2 -= inc;
+ } while (k1 != (kk-1));
+ kk = k2 + 1;
+ } while (k != j);
+ k1 = kk + kspan - 1;
+ k2 = 0;
+ do {
+ k2++;
+ Re [k1] = Rtmp [k2 - 1];
+ Im [k1] = Itmp [k2 - 1];
+ k1 -= inc;
+ } while (k1 != (kk-1));
+ } while (jj);
+ } while (j != 1);
+ }
+ return 0; /* exit point here */
+}
+/* ---------------------- end-of-file (c source) ---------------------- */
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h
new file mode 100644
index 0000000..a42f57b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*--------------------------------*-C-*---------------------------------*
+ * File:
+ * fftn.h
+ * ---------------------------------------------------------------------*
+ * Re[]: real value array
+ * Im[]: imaginary value array
+ * nTotal: total number of complex values
+ * nPass: number of elements involved in this pass of transform
+ * nSpan: nspan/nPass = number of bytes to increment pointer
+ * in Re[] and Im[]
+ * isign: exponent: +1 = forward -1 = reverse
+ * scaling: normalizing constant by which the final result is *divided*
+ * scaling == -1, normalize by total dimension of the transform
+ * scaling < -1, normalize by the square-root of the total dimension
+ *
+ * ----------------------------------------------------------------------
+ * See the comments in the code for correct usage!
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
+
+
+#include "structs.h"
+
+
+/* double precision routine */
+
+
+int WebRtcIsac_Fftns (unsigned int ndim, const int dims[], double Re[], double Im[],
+ int isign, double scaling, FFTstr *fftstate);
+
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c
new file mode 100644
index 0000000..33024a3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+#include "pitch_estimator.h"
+#include "lpc_analysis.h"
+#include "codec.h"
+
+
+
+void WebRtcIsac_AllPoleFilter(double *InOut, double *Coef, int lengthInOut, int orderCoef){
+
+ /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
+ double scal;
+ double sum;
+ int n,k;
+
+ //if (fabs(Coef[0]-1.0)<0.001) {
+ if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
+ {
+ for(n = 0; n < lengthInOut; n++)
+ {
+ sum = Coef[1] * InOut[-1];
+ for(k = 2; k <= orderCoef; k++){
+ sum += Coef[k] * InOut[-k];
+ }
+ *InOut++ -= sum;
+ }
+ }
+ else
+ {
+ scal = 1.0 / Coef[0];
+ for(n=0;n<lengthInOut;n++)
+ {
+ *InOut *= scal;
+ for(k=1;k<=orderCoef;k++){
+ *InOut -= scal*Coef[k]*InOut[-k];
+ }
+ InOut++;
+ }
+ }
+}
+
+
+void WebRtcIsac_AllZeroFilter(double *In, double *Coef, int lengthInOut, int orderCoef, double *Out){
+
+ /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
+
+ int n, k;
+ double tmp;
+
+ for(n = 0; n < lengthInOut; n++)
+ {
+ tmp = In[0] * Coef[0];
+
+ for(k = 1; k <= orderCoef; k++){
+ tmp += Coef[k] * In[-k];
+ }
+
+ *Out++ = tmp;
+ In++;
+ }
+}
+
+
+
+void WebRtcIsac_ZeroPoleFilter(double *In, double *ZeroCoef, double *PoleCoef, int lengthInOut, int orderCoef, double *Out){
+
+ /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
+ /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
+
+ WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
+ WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
+}
+
+
+void WebRtcIsac_AutoCorr(
+ double *r,
+ const double *x,
+ int N,
+ int order
+ )
+{
+ int lag, n;
+ double sum, prod;
+ const double *x_lag;
+
+ for (lag = 0; lag <= order; lag++)
+ {
+ sum = 0.0f;
+ x_lag = &x[lag];
+ prod = x[0] * x_lag[0];
+ for (n = 1; n < N - lag; n++) {
+ sum += prod;
+ prod = x[n] * x_lag[n];
+ }
+ sum += prod;
+ r[lag] = sum;
+ }
+
+}
+
+
+void WebRtcIsac_BwExpand(double *out, double *in, double coef, short length) {
+ int i;
+ double chirp;
+
+ chirp = coef;
+
+ out[0] = in[0];
+ for (i = 1; i < length; i++) {
+ out[i] = chirp * in[i];
+ chirp *= coef;
+ }
+}
+
+void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata) {
+
+ double tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
+ double corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
+ double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
+ double rho=0.9, *inp, *dp, *dp2;
+ double whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+ double weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+ double *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
+ int k, n, endpos, start;
+
+ /* Set up buffer and states */
+ memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
+ memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
+ memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
+
+ dp=weoutbuf;
+ dp2=whoutbuf;
+ for (k=0;k<PITCH_WLPCORDER;k++) {
+ *dp++ = wfdata->weostate[k];
+ *dp2++ = wfdata->whostate[k];
+ opol[k]=0.0;
+ }
+ opol[0]=1.0;
+ opol[PITCH_WLPCORDER]=0.0;
+ weo=dp;
+ who=dp2;
+
+ endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
+ inp=tmpbuffer + PITCH_WLPCBUFLEN;
+
+ for (n=0; n<PITCH_SUBFRAMES; n++) {
+ /* Windowing */
+ start=endpos-PITCH_WLPCWINLEN;
+ for (k=0; k<PITCH_WLPCWINLEN; k++) {
+ ext[k]=wfdata->window[k]*tmpbuffer[start+k];
+ }
+
+ /* Get LPC polynomial */
+ WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
+ corr[0]=1.01*corr[0]+1.0; /* White noise correction */
+ WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
+ WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
+
+ /* Filtering */
+ WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
+ WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
+
+ inp+=PITCH_SUBFRAME_LEN;
+ endpos+=PITCH_SUBFRAME_LEN;
+ weo+=PITCH_SUBFRAME_LEN;
+ who+=PITCH_SUBFRAME_LEN;
+ }
+
+ /* Export filter states */
+ for (k=0;k<PITCH_WLPCORDER;k++) {
+ wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
+ wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
+ }
+
+ /* Export output data */
+ memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+ memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+}
+
+
+static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
+static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
+
+
+
+void WebRtcIsac_AllpassFilterForDec(double *InOut,
+ const double *APSectionFactors,
+ int lengthInOut,
+ double *FilterState)
+{
+ //This performs all-pass filtering--a series of first order all-pass sections are used
+ //to filter the input in a cascade manner.
+ int n,j;
+ double temp;
+ for (j=0; j<ALLPASSSECTIONS; j++){
+ for (n=0;n<lengthInOut;n+=2){
+ temp = InOut[n]; //store input
+ InOut[n] = FilterState[j] + APSectionFactors[j]*temp;
+ FilterState[j] = -APSectionFactors[j]*InOut[n] + temp;
+ }
+ }
+}
+
+void WebRtcIsac_DecimateAllpass(const double *in,
+ double *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
+ int N, /* number of input samples */
+ double *out) /* array of size N/2 */
+{
+ int n;
+ double data_vec[PITCH_FRAME_LEN];
+
+ /* copy input */
+ memcpy(data_vec+1, in, sizeof(double) * (N-1));
+
+ data_vec[0] = state_in[2*ALLPASSSECTIONS]; //the z^(-1) state
+ state_in[2*ALLPASSSECTIONS] = in[N-1];
+
+ WebRtcIsac_AllpassFilterForDec(data_vec+1, APupper, N, state_in);
+ WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N, state_in+ALLPASSSECTIONS);
+
+ for (n=0;n<N/2;n++)
+ out[n] = data_vec[2*n] + data_vec[2*n+1];
+
+}
+
+
+
+/* create high-pass filter ocefficients
+ * z = 0.998 * exp(j*2*pi*35/8000);
+ * p = 0.94 * exp(j*2*pi*140/8000);
+ * HP_b = [1, -2*real(z), abs(z)^2];
+ * HP_a = [1, -2*real(p), abs(p)^2]; */
+static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
+static const double b_coef[2] = {-1.99524591718270, 0.99600400000000};
+static const float a_coef_float[2] = { 1.86864659625574f, -0.88360000000000f};
+static const float b_coef_float[2] = {-1.99524591718270f, 0.99600400000000f};
+
+/* second order high-pass filter */
+void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N)
+{
+ int k;
+
+ for (k=0; k<N; k++) {
+ *out = *in + state[1];
+ state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
+ state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
+ }
+}
+
+void WebRtcIsac_Highpass_float(const float *in, double *out, double *state, int N)
+{
+ int k;
+
+ for (k=0; k<N; k++) {
+ *out = (double)*in + state[1];
+ state[1] = state[0] + b_coef_float[0] * *in + a_coef_float[0] * *out;
+ state[0] = b_coef_float[1] * (double)*in++ + a_coef_float[1] * *out++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c
new file mode 100644
index 0000000..0f844af
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* filterbank_tables.c*/
+/* This file contains variables that are used in filterbanks.c*/
+
+#include "filterbank_tables.h"
+#include "settings.h"
+
+/* The composite all-pass filter factors */
+const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
+ 0.03470000000000f, 0.15440000000000f, 0.38260000000000f, 0.74400000000000f};
+
+/* The upper channel all-pass filter factors */
+const float WebRtcIsac_kUpperApFactorsFloat[2] = {
+ 0.03470000000000f, 0.38260000000000f};
+
+/* The lower channel all-pass filter factors */
+const float WebRtcIsac_kLowerApFactorsFloat[2] = {
+ 0.15440000000000f, 0.74400000000000f};
+
+/* The matrix for transforming the backward composite state to upper channel state */
+const float WebRtcIsac_kTransform1Float[8] = {
+ -0.00158678506084f, 0.00127157815343f, -0.00104805672709f, 0.00084837248079f,
+ 0.00134467983258f, -0.00107756549387f, 0.00088814793277f, -0.00071893072525f};
+
+/* The matrix for transforming the backward composite state to lower channel state */
+const float WebRtcIsac_kTransform2Float[8] = {
+ -0.00170686041697f, 0.00136780109829f, -0.00112736532350f, 0.00091257055385f,
+ 0.00103094281812f, -0.00082615076557f, 0.00068092756088f, -0.00055119165484f};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h
new file mode 100644
index 0000000..e8fda5e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filterbank_tables.h
+ *
+ * Header file for variables that are defined in
+ * filterbank_tables.c.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
+
+#include "structs.h"
+
+/********************* Coefficient Tables ************************/
+/* The number of composite all-pass filter factors */
+#define NUMBEROFCOMPOSITEAPSECTIONS 4
+
+/* The number of all-pass filter factors in an upper or lower channel*/
+#define NUMBEROFCHANNELAPSECTIONS 2
+
+/* The composite all-pass filter factors */
+extern const float WebRtcIsac_kCompositeApFactorsFloat[4];
+
+/* The upper channel all-pass filter factors */
+extern const float WebRtcIsac_kUpperApFactorsFloat[2];
+
+/* The lower channel all-pass filter factors */
+extern const float WebRtcIsac_kLowerApFactorsFloat[2];
+
+/* The matrix for transforming the backward composite state to upper channel state */
+extern const float WebRtcIsac_kTransform1Float[8];
+
+/* The matrix for transforming the backward composite state to lower channel state */
+extern const float WebRtcIsac_kTransform2Float[8];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c
new file mode 100644
index 0000000..671fd32
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c
@@ -0,0 +1,346 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * filterbanks.c
+ *
+ * This file contains function WebRtcIsac_AllPassFilter2Float,
+ * WebRtcIsac_SplitAndFilter, and WebRtcIsac_FilterAndCombine
+ * which implement filterbanks that produce decimated lowpass and
+ * highpass versions of a signal, and performs reconstruction.
+ *
+ */
+
+#include "settings.h"
+#include "filterbank_tables.h"
+#include "codec.h"
+
+/* This function performs all-pass filtering--a series of first order all-pass
+ * sections are used to filter the input in a cascade manner.
+ * The input is overwritten!!
+ */
+static void WebRtcIsac_AllPassFilter2Float(float *InOut, const float *APSectionFactors,
+ int lengthInOut, int NumberOfSections,
+ float *FilterState)
+{
+ int n, j;
+ float temp;
+ for (j=0; j<NumberOfSections; j++){
+ for (n=0;n<lengthInOut;n++){
+ temp = FilterState[j] + APSectionFactors[j] * InOut[n];
+ FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
+ InOut[n] = temp;
+ }
+ }
+}
+
+/* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+static const float kHpStCoefInFloat[4] =
+{-1.94895953203325f, 0.94984516000000f, -0.05101826139794f, 0.05015484000000f};
+
+/* Function WebRtcIsac_SplitAndFilter
+ * This function creates low-pass and high-pass decimated versions of part of
+ the input signal, and part of the signal in the input 'lookahead buffer'.
+
+ INPUTS:
+ in: a length FRAMESAMPLES array of input samples
+ prefiltdata: input data structure containing the filterbank states
+ and lookahead samples from the previous encoding
+ iteration.
+ OUTPUTS:
+ LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
+ have been phase equalized. The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+ HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
+ have been phase equalized. The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+
+ LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
+ These samples are not phase equalized. They are computed
+ from the samples in the in[] array.
+ HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
+ that are not phase equalized. They are computed from
+ the in[] vector.
+ prefiltdata: this input data structure's filterbank state and
+ lookahead sample buffers are updated for the next
+ encoding iteration.
+*/
+void WebRtcIsac_SplitAndFilterFloat(float *pin, float *LP, float *HP,
+ double *LP_la, double *HP_la,
+ PreFiltBankstr *prefiltdata)
+{
+ int k,n;
+ float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+ float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+ float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
+ float tempinoutvec[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
+ float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
+ float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
+ float in[FRAMESAMPLES];
+ float ftmp;
+
+
+ /* High pass filter */
+
+ for (k=0;k<FRAMESAMPLES;k++) {
+ in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
+ kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
+ ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
+ kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
+ prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
+ prefiltdata->HPstates_float[0] = ftmp;
+ }
+
+ /*
+ % backwards all-pass filtering to obtain zero-phase
+ [tmp1(N2+LA:-1:LA+1, 1), state1] = filter(Q.coef, Q.coef(end:-1:1), in(N:-2:2));
+ tmp1(LA:-1:1) = filter(Q.coef, Q.coef(end:-1:1), Q.LookAheadBuf1, state1);
+ Q.LookAheadBuf1 = in(N:-2:N-2*LA+2);
+ */
+ /*Backwards all-pass filter the odd samples of the input (upper channel)
+ to eventually obtain zero phase. The composite all-pass filter (comprised of both
+ the upper and lower channel all-pass filsters in series) is used for the
+ filtering. */
+
+ /* First Channel */
+
+ /*initial state of composite filter is zero */
+ for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
+ CompositeAPFilterState[k] = 0.0;
+ }
+ /* put every other sample of input into a temporary vector in reverse (backward) order*/
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ tempinoutvec[k] = in[FRAMESAMPLES-1-2*k];
+ }
+
+ /* now all-pass filter the backwards vector. Output values overwrite the input vector. */
+ WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+ /* save the backwards filtered output for later forward filtering,
+ but write it in forward order*/
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ tempin_ch1[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
+ }
+
+ /* save the backwards filter state becaue it will be transformed
+ later into a forward state */
+ for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+ ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
+ }
+
+ /* now backwards filter the samples in the lookahead buffer. The samples were
+ placed there in the encoding of the previous frame. The output samples
+ overwrite the input samples */
+ WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF1_float,
+ WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,
+ NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+ /* save the output, but write it in forward order */
+ /* write the lookahead samples for the next encoding iteration. Every other
+ sample at the end of the input frame is written in reverse order for the
+ lookahead length. Exported in the prefiltdata structure. */
+ for (k=0;k<QLOOKAHEAD;k++) {
+ tempin_ch1[QLOOKAHEAD-1-k]=prefiltdata->INLABUF1_float[k];
+ prefiltdata->INLABUF1_float[k]=in[FRAMESAMPLES-1-2*k];
+ }
+
+ /* Second Channel. This is exactly like the first channel, except that the
+ even samples are now filtered instead (lower channel). */
+ for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
+ CompositeAPFilterState[k] = 0.0;
+ }
+
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ tempinoutvec[k] = in[FRAMESAMPLES-2-2*k];
+ }
+
+ WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ tempin_ch2[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
+ }
+
+ for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+ ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
+ }
+
+
+ WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF2_float,
+ WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,NUMBEROFCOMPOSITEAPSECTIONS,
+ CompositeAPFilterState);
+
+ for (k=0;k<QLOOKAHEAD;k++) {
+ tempin_ch2[QLOOKAHEAD-1-k]=prefiltdata->INLABUF2_float[k];
+ prefiltdata->INLABUF2_float[k]=in[FRAMESAMPLES-2-2*k];
+ }
+
+ /* Transform filter states from backward to forward */
+ /*At this point, each of the states of the backwards composite filters for the
+ two channels are transformed into forward filtering states for the corresponding
+ forward channel filters. Each channel's forward filtering state from the previous
+ encoding iteration is added to the transformed state to get a proper forward state */
+
+ /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is multiplied by a
+ NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4) transform matrix to get the
+ new state that is added to the previous 2x1 input state */
+
+ for (k=0;k<NUMBEROFCHANNELAPSECTIONS;k++){ /* k is row variable */
+ for (n=0; n<NUMBEROFCOMPOSITEAPSECTIONS;n++){/* n is column variable */
+ prefiltdata->INSTAT1_float[k] += ForTransform_CompositeAPFilterState[n]*
+ WebRtcIsac_kTransform1Float[k*NUMBEROFCHANNELAPSECTIONS+n];
+ prefiltdata->INSTAT2_float[k] += ForTransform_CompositeAPFilterState2[n]*
+ WebRtcIsac_kTransform2Float[k*NUMBEROFCHANNELAPSECTIONS+n];
+ }
+ }
+
+ /*obtain polyphase components by forward all-pass filtering through each channel */
+ /* the backward filtered samples are now forward filtered with the corresponding channel filters */
+ /* The all pass filtering automatically updates the filter states which are exported in the
+ prefiltdata structure */
+ WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_float);
+ WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_float);
+
+ /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
+ for (k=0; k<FRAMESAMPLES_HALF; k++) {
+ LP[k] = 0.5f*(tempin_ch1[k] + tempin_ch2[k]);/* low pass signal*/
+ HP[k] = 0.5f*(tempin_ch1[k] - tempin_ch2[k]);/* high pass signal*/
+ }
+
+ /* Lookahead LP and HP signals */
+ /* now create low pass and high pass signals of the input vector. However, no
+ backwards filtering is performed, and hence no phase equalization is involved.
+ Also, the input contains some samples that are lookahead samples. The high pass
+ and low pass signals that are created are used outside this function for analysis
+ (not encoding) purposes */
+
+ /* set up input */
+ for (k=0; k<FRAMESAMPLES_HALF; k++) {
+ tempin_ch1[k]=in[2*k+1];
+ tempin_ch2[k]=in[2*k];
+ }
+
+ /* the input filter states are passed in and updated by the all-pass filtering routine and
+ exported in the prefiltdata structure*/
+ WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA1_float);
+ WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA2_float);
+
+ for (k=0; k<FRAMESAMPLES_HALF; k++) {
+ LP_la[k] = (float)(0.5f*(tempin_ch1[k] + tempin_ch2[k])); /*low pass */
+ HP_la[k] = (double)(0.5f*(tempin_ch1[k] - tempin_ch2[k])); /* high pass */
+ }
+
+
+}/*end of WebRtcIsac_SplitAndFilter */
+
+
+/* Combining */
+
+/* HPstcoeff_out_1 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+static const float kHpStCoefOut1Float[4] =
+{-1.99701049409000f, 0.99714204490000f, 0.01701049409000f, -0.01704204490000f};
+
+/* HPstcoeff_out_2 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+static const float kHpStCoefOut2Float[4] =
+{-1.98645294509837f, 0.98672435560000f, 0.00645294509837f, -0.00662435560000f};
+
+
+/* Function WebRtcIsac_FilterAndCombine */
+/* This is a decoder function that takes the decimated
+ length FRAMESAMPLES_HALF input low-pass and
+ high-pass signals and creates a reconstructed fullband
+ output signal of length FRAMESAMPLES. WebRtcIsac_FilterAndCombine
+ is the sibling function of WebRtcIsac_SplitAndFilter */
+/* INPUTS:
+ inLP: a length FRAMESAMPLES_HALF array of input low-pass
+ samples.
+ inHP: a length FRAMESAMPLES_HALF array of input high-pass
+ samples.
+ postfiltdata: input data structure containing the filterbank
+ states from the previous decoding iteration.
+ OUTPUTS:
+ Out: a length FRAMESAMPLES array of output reconstructed
+ samples (fullband) based on the input low-pass and
+ high-pass signals.
+ postfiltdata: the input data structure containing the filterbank
+ states is updated for the next decoding iteration */
+void WebRtcIsac_FilterAndCombineFloat(float *InLP,
+ float *InHP,
+ float *Out,
+ PostFiltBankstr *postfiltdata)
+{
+ int k;
+ float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
+ float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
+ float ftmp, ftmp2;
+
+ /* Form the polyphase signals*/
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ tempin_ch1[k]=InLP[k]+InHP[k]; /* Construct a new upper channel signal*/
+ tempin_ch2[k]=InLP[k]-InHP[k]; /* Construct a new lower channel signal*/
+ }
+
+
+ /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
+ that were used as a lower channel at the encoding side. So at the decoder, the
+ corresponding all-pass filter factors for each channel are swapped.*/
+ WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kLowerApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_float);
+
+ /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
+ at the decoder are swapped from the ones at the encoder, the 'upper' channel
+ all-pass filter factors (WebRtcIsac_kUpperApFactorsFloat) are used to filter this new
+ lower channel signal */
+ WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kUpperApFactorsFloat,
+ FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_float);
+
+
+ /* Merge outputs to form the full length output signal.*/
+ for (k=0;k<FRAMESAMPLES_HALF;k++) {
+ Out[2*k]=tempin_ch2[k];
+ Out[2*k+1]=tempin_ch1[k];
+ }
+
+
+ /* High pass filter */
+
+ for (k=0;k<FRAMESAMPLES;k++) {
+ ftmp2 = Out[k] + kHpStCoefOut1Float[2] * postfiltdata->HPstates1_float[0] +
+ kHpStCoefOut1Float[3] * postfiltdata->HPstates1_float[1];
+ ftmp = Out[k] - kHpStCoefOut1Float[0] * postfiltdata->HPstates1_float[0] -
+ kHpStCoefOut1Float[1] * postfiltdata->HPstates1_float[1];
+ postfiltdata->HPstates1_float[1] = postfiltdata->HPstates1_float[0];
+ postfiltdata->HPstates1_float[0] = ftmp;
+ Out[k] = ftmp2;
+ }
+
+ for (k=0;k<FRAMESAMPLES;k++) {
+ ftmp2 = Out[k] + kHpStCoefOut2Float[2] * postfiltdata->HPstates2_float[0] +
+ kHpStCoefOut2Float[3] * postfiltdata->HPstates2_float[1];
+ ftmp = Out[k] - kHpStCoefOut2Float[0] * postfiltdata->HPstates2_float[0] -
+ kHpStCoefOut2Float[1] * postfiltdata->HPstates2_float[1];
+ postfiltdata->HPstates2_float[1] = postfiltdata->HPstates2_float[0];
+ postfiltdata->HPstates2_float[0] = ftmp;
+ Out[k] = ftmp2;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c
new file mode 100644
index 0000000..6df034d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c
@@ -0,0 +1,175 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* encode.c - Encoding function for the iSAC coder */
+
+#include "structs.h"
+#include "codec.h"
+#include "pitch_estimator.h"
+
+#include <math.h>
+
+void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
+
+ int k;
+
+ for (k = 0; k < WINLEN; k++) {
+ maskdata->DataBufferLo[k] = 0.0;
+ maskdata->DataBufferHi[k] = 0.0;
+ }
+ for (k = 0; k < ORDERLO+1; k++) {
+ maskdata->CorrBufLo[k] = 0.0;
+ maskdata->PreStateLoF[k] = 0.0;
+ maskdata->PreStateLoG[k] = 0.0;
+ maskdata->PostStateLoF[k] = 0.0;
+ maskdata->PostStateLoG[k] = 0.0;
+ }
+ for (k = 0; k < ORDERHI+1; k++) {
+ maskdata->CorrBufHi[k] = 0.0;
+ maskdata->PreStateHiF[k] = 0.0;
+ maskdata->PreStateHiG[k] = 0.0;
+ maskdata->PostStateHiF[k] = 0.0;
+ maskdata->PostStateHiG[k] = 0.0;
+ }
+
+ maskdata->OldEnergy = 10.0;
+
+ /* fill tables for transforms */
+ WebRtcIsac_InitTransform();
+
+ return;
+}
+
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr *prefiltdata)
+{
+ int k;
+
+ for (k = 0; k < QLOOKAHEAD; k++) {
+ prefiltdata->INLABUF1[k] = 0;
+ prefiltdata->INLABUF2[k] = 0;
+
+ prefiltdata->INLABUF1_float[k] = 0;
+ prefiltdata->INLABUF2_float[k] = 0;
+ }
+ for (k = 0; k < 2*(QORDER-1); k++) {
+ prefiltdata->INSTAT1[k] = 0;
+ prefiltdata->INSTAT2[k] = 0;
+ prefiltdata->INSTATLA1[k] = 0;
+ prefiltdata->INSTATLA2[k] = 0;
+
+ prefiltdata->INSTAT1_float[k] = 0;
+ prefiltdata->INSTAT2_float[k] = 0;
+ prefiltdata->INSTATLA1_float[k] = 0;
+ prefiltdata->INSTATLA2_float[k] = 0;
+ }
+
+ /* High pass filter states */
+ prefiltdata->HPstates[0] = 0.0;
+ prefiltdata->HPstates[1] = 0.0;
+
+ prefiltdata->HPstates_float[0] = 0.0f;
+ prefiltdata->HPstates_float[1] = 0.0f;
+
+ return;
+}
+
+void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata)
+{
+ int k;
+
+ for (k = 0; k < 2*POSTQORDER; k++) {
+ postfiltdata->STATE_0_LOWER[k] = 0;
+ postfiltdata->STATE_0_UPPER[k] = 0;
+
+ postfiltdata->STATE_0_LOWER_float[k] = 0;
+ postfiltdata->STATE_0_UPPER_float[k] = 0;
+ }
+
+ /* High pass filter states */
+ postfiltdata->HPstates1[0] = 0.0;
+ postfiltdata->HPstates1[1] = 0.0;
+
+ postfiltdata->HPstates2[0] = 0.0;
+ postfiltdata->HPstates2[1] = 0.0;
+
+ postfiltdata->HPstates1_float[0] = 0.0f;
+ postfiltdata->HPstates1_float[1] = 0.0f;
+
+ postfiltdata->HPstates2_float[0] = 0.0f;
+ postfiltdata->HPstates2_float[1] = 0.0f;
+
+ return;
+}
+
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr *pitchfiltdata)
+{
+ int k;
+
+ for (k = 0; k < PITCH_BUFFSIZE; k++) {
+ pitchfiltdata->ubuf[k] = 0.0;
+ }
+ pitchfiltdata->ystate[0] = 0.0;
+ for (k = 1; k < (PITCH_DAMPORDER); k++) {
+ pitchfiltdata->ystate[k] = 0.0;
+ }
+ pitchfiltdata->oldlagp[0] = 50.0;
+ pitchfiltdata->oldgainp[0] = 0.0;
+}
+
+void WebRtcIsac_InitWeightingFilter(WeightFiltstr *wfdata)
+{
+ int k;
+ double t, dtmp, dtmp2, denum, denum2;
+
+ for (k=0;k<PITCH_WLPCBUFLEN;k++)
+ wfdata->buffer[k]=0.0;
+
+ for (k=0;k<PITCH_WLPCORDER;k++) {
+ wfdata->istate[k]=0.0;
+ wfdata->weostate[k]=0.0;
+ wfdata->whostate[k]=0.0;
+ }
+
+ /* next part should be in Matlab, writing to a global table */
+ t = 0.5;
+ denum = 1.0 / ((double) PITCH_WLPCWINLEN);
+ denum2 = denum * denum;
+ for (k=0;k<PITCH_WLPCWINLEN;k++) {
+ dtmp = PITCH_WLPCASYM * t * denum + (1-PITCH_WLPCASYM) * t * t * denum2;
+ dtmp *= 3.14159265;
+ dtmp2 = sin(dtmp);
+ wfdata->window[k] = dtmp2 * dtmp2;
+ t++;
+ }
+}
+
+/* clear all buffers */
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct *State)
+{
+ int k;
+
+ for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++)
+ State->dec_buffer[k] = 0.0;
+ for (k = 0; k < 2*ALLPASSSECTIONS+1; k++)
+ State->decimator_state[k] = 0.0;
+ for (k = 0; k < 2; k++)
+ State->hp_state[k] = 0.0;
+ for (k = 0; k < QLOOKAHEAD; k++)
+ State->whitened_buf[k] = 0.0;
+ for (k = 0; k < QLOOKAHEAD; k++)
+ State->inbuf[k] = 0.0;
+
+ WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
+
+ WebRtcIsac_InitPitchFilter(&(State->PFstr));
+
+ WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
new file mode 100644
index 0000000..67db1e4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
@@ -0,0 +1,2793 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * isac.c
+ *
+ * This C file contains the functions for the ISAC API
+ *
+ */
+
+#include "isac.h"
+#include "bandwidth_estimator.h"
+#include "crc.h"
+#include "entropy_coding.h"
+#include "codec.h"
+#include "structs.h"
+#include "signal_processing_library.h"
+#include "lpc_shape_swb16_tables.h"
+#include "os_specific_inline.h"
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <math.h>
+
+#define BIT_MASK_DEC_INIT 0x0001
+#define BIT_MASK_ENC_INIT 0x0002
+
+#define LEN_CHECK_SUM_WORD8 4
+#define MAX_NUM_LAYERS 10
+
+
+/****************************************************************************
+ * UpdatePayloadSizeLimit()
+ *
+ * Call this function to update the limit on the payload size. The limit on
+ * payload size might change i) if a user ''directly changes the limit by
+ * calling xxx_setMaxPayloadSize() or xxx_setMaxRate(), or ii) indirectly
+ * when bandwidth is changing. The latter might be the result of bandwidth
+ * adaptation, or direct change of the bottleneck in instantaneous mode.
+ *
+ * This function takes the current overall limit on payload, and translate it
+ * to the limits on lower and upper-band. If the codec is in wideband mode
+ * then the overall limit and the limit on the lower-band is the same.
+ * Otherwise, a fraction of the limit should be allocated to lower-band
+ * leaving some room for the upper-band bit-stream. That is why an update
+ * of limit is required every time that the bandwidth is changing.
+ *
+ */
+static void UpdatePayloadSizeLimit(
+ ISACMainStruct *instISAC)
+{
+ WebRtc_Word16 lim30MsPayloadBytes;
+ WebRtc_Word16 lim60MsPayloadBytes;
+
+ lim30MsPayloadBytes = WEBRTC_SPL_MIN(
+ (instISAC->maxPayloadSizeBytes),
+ (instISAC->maxRateBytesPer30Ms));
+
+ lim60MsPayloadBytes = WEBRTC_SPL_MIN(
+ (instISAC->maxPayloadSizeBytes),
+ (instISAC->maxRateBytesPer30Ms << 1));
+
+ // The only time that iSAC will have 60 ms
+ // frame-size is when operating in wideband so
+ // there is no upper-band bit-stream
+
+ if(instISAC->bandwidthKHz == isac8kHz)
+ {
+ // at 8 kHz there is no upper-band bit-stream
+ // therefore the lower-band limit is as the overall
+ // limit.
+ instISAC->instLB.ISACencLB_obj.payloadLimitBytes60 =
+ lim60MsPayloadBytes;
+ instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
+ lim30MsPayloadBytes;
+ }
+ else
+ {
+ // when in super-wideband, we only have 30 ms frames
+ // Do a rate allocation for the given limit.
+ if(lim30MsPayloadBytes > 250)
+ {
+ // 4/5 to lower-band the rest for upper-band
+ instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
+ (lim30MsPayloadBytes << 2) / 5;
+ }
+ else if(lim30MsPayloadBytes > 200)
+ {
+ // for the interval of 200 to 250 the share of
+ // upper-band linearly grows from 20 to 50;
+ instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
+ (lim30MsPayloadBytes << 1) / 5 + 100;
+ }
+ else
+ {
+ // allocate only 20 for upper-band
+ instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
+ lim30MsPayloadBytes - 20;
+ }
+ instISAC->instUB.ISACencUB_obj.maxPayloadSizeBytes =
+ lim30MsPayloadBytes;
+ }
+}
+
+
+/****************************************************************************
+ * UpdateBottleneck()
+ *
+ * This function updates the bottleneck only if the codec is operating in
+ * channel-adaptive mode. Furthermore, as the update of bottleneck might
+ * result in an update of bandwidth, therefore, the bottlenech should be
+ * updated just right before the first 10ms of a frame is pushed into encoder.
+ *
+ */
+static void UpdateBottleneck(
+ ISACMainStruct *instISAC)
+{
+ // read the bottleneck from bandwidth estimator for the
+ // first 10 ms audio. This way, if there is a change
+ // in bandwidth upper and lower-band will be in sync.
+ if((instISAC->codingMode == 0) &&
+ (instISAC->instLB.ISACencLB_obj.buffer_index == 0) &&
+ (instISAC->instLB.ISACencLB_obj.frame_nb == 0))
+ {
+ WebRtc_Word32 bottleneck;
+ WebRtcIsac_GetUplinkBandwidth(&(instISAC->bwestimator_obj),
+ &bottleneck);
+
+ // Adding hysteresis when increasing signal bandwidth
+ if((instISAC->bandwidthKHz == isac8kHz)
+ && (bottleneck > 37000)
+ && (bottleneck < 41000))
+ {
+ bottleneck = 37000;
+ }
+
+ // switching from 12 kHz to 16 kHz is not allowed at this revision
+ // If we let this happen, we have to take care of buffer_index and
+ // the last LPC vector.
+ if((instISAC->bandwidthKHz != isac16kHz) &&
+ (bottleneck > 46000))
+ {
+ bottleneck = 46000;
+ }
+
+ // we might need a rate allocation.
+ if(instISAC->encoderSamplingRateKHz == kIsacWideband)
+ {
+ // wideband is the only choise we have here.
+ instISAC->instLB.ISACencLB_obj.bottleneck =
+ (bottleneck > 32000)? 32000:bottleneck;
+ instISAC->bandwidthKHz = isac8kHz;
+ }
+ else
+ {
+ // do the rate-allosation and get the new bandwidth.
+ enum ISACBandwidth bandwidth;
+ WebRtcIsac_RateAllocation(bottleneck,
+ &(instISAC->instLB.ISACencLB_obj.bottleneck),
+ &(instISAC->instUB.ISACencUB_obj.bottleneck),
+ &bandwidth);
+ if(bandwidth != isac8kHz)
+ {
+ instISAC->instLB.ISACencLB_obj.new_framelength = 480;
+ }
+ if(bandwidth != instISAC->bandwidthKHz)
+ {
+ // bandwidth is changing.
+ instISAC->bandwidthKHz = bandwidth;
+ UpdatePayloadSizeLimit(instISAC);
+ if(bandwidth == isac12kHz)
+ {
+ instISAC->instLB.ISACencLB_obj.buffer_index = 0;
+ }
+ // currently we don't let the bandwidth to switch to 16 kHz
+ // if in adaptive mode. If we let this happen, we have to take
+ // car of buffer_index and the last LPC vector.
+ }
+ }
+ }
+}
+
+
+/****************************************************************************
+ * GetSendBandwidthInfo()
+ *
+ * This is called to get the bandwidth info. This info is the bandwidth and
+ * and the jitter of 'there-to-here' channel, estimated 'here.' These info
+ * is signaled in an in-band fashion to the otherside.
+ *
+ * The call to the bandwidth estimator trigers a recursive averaging which
+ * has to be synchronized between encoder & decoder, therefore. The call to
+ * BWE should be once per packet. As the BWE info is inserted into bit-stream
+ * we need a valid info right before the encodeLB function is going to
+ * generating a bit-stream. That is when lower-band buffer has already 20ms
+ * of audio, and the 3rd block of 10ms is going to be injected into encoder.
+ *
+ * Inputs:
+ * - instISAC : iSAC instance.
+ *
+ * Outputs:
+ * - bandwidthIndex : an index which has to be encoded in
+ * lower-band bit-stream, indicating the
+ * bandwidth of there-to-here channel.
+ * - jitterInfo : this indicates if the jitter is high
+ * or low and it is encoded in upper-band
+ * bit-stream.
+ *
+ */
+static void GetSendBandwidthInfo(
+ ISACMainStruct* instISAC,
+ WebRtc_Word16* bandwidthIndex,
+ WebRtc_Word16* jitterInfo)
+{
+ if((instISAC->instLB.ISACencLB_obj.buffer_index ==
+ (FRAMESAMPLES_10ms << 1)) &&
+ (instISAC->instLB.ISACencLB_obj.frame_nb == 0))
+ {
+ /* bandwidth estimation and coding */
+ WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj),
+ bandwidthIndex, jitterInfo, instISAC->decoderSamplingRateKHz);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_AssignSize(...)
+ *
+ * This function returns the size of the ISAC instance, so that the instance
+ * can be created out side iSAC.
+ *
+ * Output:
+ * - sizeinbytes : number of bytes needed to allocate for the
+ * instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_AssignSize(
+ int *sizeInBytes)
+{
+ *sizeInBytes = sizeof(ISACMainStruct) * 2 / sizeof(WebRtc_Word16);
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_Assign(...)
+ *
+ * This function assignes the memory already created to the ISAC instance.
+ *
+ * Input:
+ * - ISAC_main_inst : address of the pointer to the coder instance.
+ * - instISAC_Addr : the already allocaded memeory, where we put the
+ * iSAC struct
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_Assign(
+ ISACStruct** ISAC_main_inst,
+ void* instISAC_Addr)
+{
+ if(instISAC_Addr != NULL)
+ {
+ ISACMainStruct* instISAC = (ISACMainStruct*)instISAC_Addr;
+ instISAC->errorCode = 0;
+ instISAC->initFlag = 0;
+
+ // Assign the address
+ *ISAC_main_inst = (ISACStruct*)instISAC_Addr;
+
+ // Default is wideband.
+ instISAC->encoderSamplingRateKHz = kIsacWideband;
+ instISAC->decoderSamplingRateKHz = kIsacWideband;
+ instISAC->bandwidthKHz = isac8kHz;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_Create(...)
+ *
+ * This function creates an ISAC instance, which will contain the state
+ * information for one coding/decoding channel.
+ *
+ * Input:
+ * - ISAC_main_inst : address of the pointer to the coder instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_Create(
+ ISACStruct** ISAC_main_inst)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1);
+ *ISAC_main_inst = (ISACStruct*)instISAC;
+ if(*ISAC_main_inst != NULL)
+ {
+ instISAC->errorCode = 0;
+ instISAC->initFlag = 0;
+ // Default is wideband
+ instISAC->bandwidthKHz = isac8kHz;
+ instISAC->encoderSamplingRateKHz = kIsacWideband;
+ instISAC->decoderSamplingRateKHz = kIsacWideband;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_Free(...)
+ *
+ * This function frees the ISAC instance created at the beginning.
+ *
+ * Input:
+ * - ISAC_main_inst : a ISAC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_Free(
+ ISACStruct* ISAC_main_inst)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+ WEBRTC_SPL_FREE(instISAC);
+ return 0;
+}
+
+
+/****************************************************************************
+ * EncoderInitLb(...) - internal function for initialization of
+ * Lower Band
+ * EncoderInitUb(...) - internal function for initialization of
+ * Upper Band
+ * WebRtcIsac_EncoderInit(...) - API function
+ *
+ * This function initializes a ISAC instance prior to the encoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - CodingMode : 0 -> Bit rate and frame length are automatically
+ * adjusted to available bandwidth on
+ * transmission channel, applicable just to
+ * wideband mode.
+ * 1 -> User sets a frame length and a target bit
+ * rate which is taken as the maximum
+ * short-term average bit rate.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+static WebRtc_Word16 EncoderInitLb(
+ ISACLBStruct* instLB,
+ WebRtc_Word16 codingMode,
+ enum IsacSamplingRate sampRate)
+{
+ WebRtc_Word16 statusInit = 0;
+ int k;
+
+ /* Init stream vector to zero */
+ for(k=0; k < STREAM_SIZE_MAX_60; k++)
+ {
+ instLB->ISACencLB_obj.bitstr_obj.stream[k] = 0;
+ }
+
+ if((codingMode == 1) || (sampRate == kIsacSuperWideband))
+ {
+ // 30 ms frame-size if either in super-wideband or
+ // instanteneous mode (I-mode)
+ instLB->ISACencLB_obj.new_framelength = 480;
+ }
+ else
+ {
+ instLB->ISACencLB_obj.new_framelength = INITIAL_FRAMESAMPLES;
+ }
+
+ WebRtcIsac_InitMasking(&instLB->ISACencLB_obj.maskfiltstr_obj);
+ WebRtcIsac_InitPreFilterbank(&instLB->ISACencLB_obj.prefiltbankstr_obj);
+ WebRtcIsac_InitPitchFilter(&instLB->ISACencLB_obj.pitchfiltstr_obj);
+ WebRtcIsac_InitPitchAnalysis(
+ &instLB->ISACencLB_obj.pitchanalysisstr_obj);
+
+
+ instLB->ISACencLB_obj.buffer_index = 0;
+ instLB->ISACencLB_obj.frame_nb = 0;
+ /* default for I-mode */
+ instLB->ISACencLB_obj.bottleneck = 32000;
+ instLB->ISACencLB_obj.current_framesamples = 0;
+ instLB->ISACencLB_obj.s2nr = 0;
+ instLB->ISACencLB_obj.payloadLimitBytes30 = STREAM_SIZE_MAX_30;
+ instLB->ISACencLB_obj.payloadLimitBytes60 = STREAM_SIZE_MAX_60;
+ instLB->ISACencLB_obj.maxPayloadBytes = STREAM_SIZE_MAX_60;
+ instLB->ISACencLB_obj.maxRateInBytes = STREAM_SIZE_MAX_30;
+ instLB->ISACencLB_obj.enforceFrameSize = 0;
+ /* invalid value prevents getRedPayload to
+ run before encoder is called */
+ instLB->ISACencLB_obj.lastBWIdx = -1;
+ return statusInit;
+}
+
+static WebRtc_Word16 EncoderInitUb(
+ ISACUBStruct* instUB,
+ WebRtc_Word16 bandwidth)
+{
+ WebRtc_Word16 statusInit = 0;
+ int k;
+
+ /* Init stream vector to zero */
+ for(k = 0; k < STREAM_SIZE_MAX_60; k++)
+ {
+ instUB->ISACencUB_obj.bitstr_obj.stream[k] = 0;
+ }
+
+ WebRtcIsac_InitMasking(&instUB->ISACencUB_obj.maskfiltstr_obj);
+ WebRtcIsac_InitPreFilterbank(&instUB->ISACencUB_obj.prefiltbankstr_obj);
+
+ if(bandwidth == isac16kHz)
+ {
+ instUB->ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES;
+ }
+ else
+ {
+ instUB->ISACencUB_obj.buffer_index = 0;
+ }
+ /* default for I-mode */
+ instUB->ISACencUB_obj.bottleneck = 32000;
+ // These store the limits for the wideband + super-wideband bit-stream.
+ instUB->ISACencUB_obj.maxPayloadSizeBytes = STREAM_SIZE_MAX_30 << 1;
+ // This has to be updated after each lower-band encoding to guarantee
+ // a correct payload-limitation.
+ instUB->ISACencUB_obj.numBytesUsed = 0;
+ memset(instUB->ISACencUB_obj.data_buffer_float, 0,
+ (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES) * sizeof(float));
+
+ memcpy(&(instUB->ISACencUB_obj.lastLPCVec),
+ WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
+
+ return statusInit;
+}
+
+
+WebRtc_Word16 WebRtcIsac_EncoderInit(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 codingMode)
+{
+ ISACMainStruct *instISAC;
+ WebRtc_Word16 status;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if((codingMode != 0) && (codingMode != 1))
+ {
+ instISAC->errorCode = ISAC_DISALLOWED_CODING_MODE;
+ return -1;
+ }
+ // default bottleneck
+ instISAC->bottleneck = MAX_ISAC_BW;
+
+ if(instISAC->encoderSamplingRateKHz == kIsacWideband)
+ {
+ instISAC->bandwidthKHz = isac8kHz;
+ instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
+ instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
+ }
+ else
+ {
+ instISAC->bandwidthKHz = isac16kHz;
+ instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
+ instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
+ }
+
+ // Channel-adaptive = 0; Instantaneous (Channel-independent) = 1;
+ instISAC->codingMode = codingMode;
+
+ WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
+ instISAC->encoderSamplingRateKHz,
+ instISAC->decoderSamplingRateKHz);
+
+ WebRtcIsac_InitRateModel(&instISAC->rate_data_obj);
+ /* default for I-mode */
+ instISAC->MaxDelay = 10.0;
+
+ status = EncoderInitLb(&instISAC->instLB, codingMode,
+ instISAC->encoderSamplingRateKHz);
+ if(status < 0)
+ {
+ instISAC->errorCode = -status;
+ return -1;
+ }
+
+ if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ // Initialize encoder filter-bank.
+ memset(instISAC->analysisFBState1, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+ memset(instISAC->analysisFBState2, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+
+ status = EncoderInitUb(&(instISAC->instUB),
+ instISAC->bandwidthKHz);
+ if(status < 0)
+ {
+ instISAC->errorCode = -status;
+ return -1;
+ }
+ }
+ // Initializtion is successful, set the flag
+ instISAC->initFlag |= BIT_MASK_ENC_INIT;
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_Encode(...)
+ *
+ * This function encodes 10ms frame(s) and inserts it into a package.
+ * Input speech length has to be 160 samples (10ms). The encoder buffers those
+ * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - speechIn : input speech vector.
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ * Return value:
+ * : >0 - Length (in bytes) of coded data
+ * : 0 - The buffer didn't reach the chosen
+ * frameSize so it keeps buffering speech
+ * samples.
+ * : -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_Encode(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_Word16* speechIn,
+ WebRtc_Word16* encoded)
+{
+ ISACMainStruct* instISAC;
+ ISACLBStruct* instLB;
+ ISACUBStruct* instUB;
+
+ float inFrame[FRAMESAMPLES_10ms];
+ WebRtc_Word16 speechInLB[FRAMESAMPLES_10ms];
+ WebRtc_Word16 speechInUB[FRAMESAMPLES_10ms];
+ WebRtc_Word16 streamLenLB = 0;
+ WebRtc_Word16 streamLenUB = 0;
+ WebRtc_Word16 streamLen = 0;
+ WebRtc_Word16 k = 0;
+ WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
+ int garbageLen = 0;
+ WebRtc_Word32 bottleneck = 0;
+ WebRtc_Word16 bottleneckIdx = 0;
+ WebRtc_Word16 jitterInfo = 0;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+ instLB = &(instISAC->instLB);
+ instUB = &(instISAC->instUB);
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ WebRtcSpl_AnalysisQMF(speechIn, speechInLB, speechInUB,
+ instISAC->analysisFBState1, instISAC->analysisFBState2);
+
+ /* convert from fixed to floating point */
+ for(k = 0; k < FRAMESAMPLES_10ms; k++)
+ {
+ inFrame[k] = (float)speechInLB[k];
+ }
+ }
+ else
+ {
+ for(k = 0; k < FRAMESAMPLES_10ms; k++)
+ {
+ inFrame[k] = (float) speechIn[k];
+ }
+ }
+
+ /* add some noise to avoid denormal numbers */
+ inFrame[0] += (float)1.23455334e-3;
+ inFrame[1] -= (float)2.04324239e-3;
+ inFrame[2] += (float)1.90854954e-3;
+ inFrame[9] += (float)1.84854878e-3;
+
+
+ // This function will update the bottleneck if required
+ UpdateBottleneck(instISAC);
+
+ // Get the bandwith information which has to be sent to the other side
+ GetSendBandwidthInfo(instISAC, &bottleneckIdx, &jitterInfo);
+
+ //
+ // ENCODE LOWER-BAND
+ //
+ streamLenLB = WebRtcIsac_EncodeLb(inFrame, &instLB->ISACencLB_obj,
+ instISAC->codingMode, bottleneckIdx);
+
+ if(streamLenLB < 0)
+ {
+ return -1;
+ }
+
+ if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ instUB = &(instISAC->instUB);
+
+ // convert to float
+ for(k = 0; k < FRAMESAMPLES_10ms; k++)
+ {
+ inFrame[k] = (float) speechInUB[k];
+ }
+
+ /* add some noise to avoid denormal numbers */
+ inFrame[0] += (float)1.23455334e-3;
+ inFrame[1] -= (float)2.04324239e-3;
+ inFrame[2] += (float)1.90854954e-3;
+ inFrame[9] += (float)1.84854878e-3;
+
+ // Tell to upper-band the number of bytes used so far.
+ // This is for payload limitation.
+ instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
+ LEN_CHECK_SUM_WORD8;
+
+ //
+ // ENCODE UPPER-BAND
+ //
+ switch(instISAC->bandwidthKHz)
+ {
+ case isac12kHz:
+ {
+ streamLenUB = WebRtcIsac_EncodeUb12(inFrame,
+ &instUB->ISACencUB_obj,
+ jitterInfo);
+ break;
+ }
+ case isac16kHz:
+ {
+ streamLenUB = WebRtcIsac_EncodeUb16(inFrame,
+ &instUB->ISACencUB_obj,
+ jitterInfo);
+ break;
+ }
+ case isac8kHz:
+ {
+ streamLenUB = 0;
+ break;
+ }
+ }
+
+ if((streamLenUB < 0) &&
+ (streamLenUB != -ISAC_PAYLOAD_LARGER_THAN_LIMIT))
+ {
+ // an error has happened but this is not the error due to a
+ // bit-stream larger than the limit
+ return -1;
+ }
+
+ if(streamLenLB == 0)
+ {
+ return 0;
+ }
+
+ // One bite is allocated for the length. According to older decoders
+ // so the length bit-stream plus one byte for size and
+ // LEN_CHECK_SUM_WORD8 for the checksum should be less than or equal
+ // to 255.
+ if((streamLenUB > (255 - (LEN_CHECK_SUM_WORD8 + 1))) ||
+ (streamLenUB == -ISAC_PAYLOAD_LARGER_THAN_LIMIT))
+ {
+ // we have got a too long bit-stream we skip the upper-band
+ // bit-stream for this frame.
+ streamLenUB = 0;
+ }
+
+ memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
+ streamLenLB);
+ streamLen = streamLenLB;
+ if(streamLenUB > 0)
+ {
+ ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)(streamLenUB + 1 +
+ LEN_CHECK_SUM_WORD8);
+ memcpy(&ptrEncodedUW8[streamLenLB + 1],
+ instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
+ streamLen += ptrEncodedUW8[streamLenLB];
+ }
+ else
+ {
+ ptrEncodedUW8[streamLenLB] = 0;
+ }
+ }
+ else
+ {
+ if(streamLenLB == 0)
+ {
+ return 0;
+ }
+ memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
+ streamLenLB);
+ streamLenUB = 0;
+ streamLen = streamLenLB;
+ }
+
+ // Add Garbage if required.
+ WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj, &bottleneck);
+ if(instISAC->codingMode == 0)
+ {
+ int minBytes;
+ int limit;
+ WebRtc_UWord8* ptrGarbage;
+
+ instISAC->MaxDelay = (double)WebRtcIsac_GetUplinkMaxDelay(
+ &instISAC->bwestimator_obj);
+
+ /* update rate model and get minimum number of bytes in this packet */
+ minBytes = WebRtcIsac_GetMinBytes(&(instISAC->rate_data_obj),
+ streamLen, instISAC->instLB.ISACencLB_obj.current_framesamples,
+ bottleneck, instISAC->MaxDelay, instISAC->bandwidthKHz);
+
+ /* Make sure MinBytes does not exceed packet size limit */
+ if(instISAC->bandwidthKHz == isac8kHz)
+ {
+ if(instLB->ISACencLB_obj.current_framesamples == FRAMESAMPLES)
+ {
+ limit = instLB->ISACencLB_obj.payloadLimitBytes30;
+ }
+ else
+ {
+ limit = instLB->ISACencLB_obj.payloadLimitBytes60;
+ }
+ }
+ else
+ {
+ limit = instUB->ISACencUB_obj.maxPayloadSizeBytes;
+ }
+ minBytes = (minBytes > limit)? limit:minBytes;
+
+ /* Make sure we don't allow more than 255 bytes of garbage data.
+ We store the length of the garbage data in 8 bits in the bitstream,
+ 255 is the max garbage length we can signal using 8 bits. */
+ if((instISAC->bandwidthKHz == isac8kHz) ||
+ (streamLenUB == 0))
+ {
+ ptrGarbage = &ptrEncodedUW8[streamLenLB];
+ limit = streamLen + 255;
+ }
+ else
+ {
+ ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB];
+ limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]);
+ }
+ minBytes = (minBytes > limit)? limit:minBytes;
+
+ garbageLen = (minBytes > streamLen)? (minBytes - streamLen):0;
+
+ /* Save data for creation of multiple bitstreams */
+ //ISACencLB_obj->SaveEnc_obj.minBytes = MinBytes;
+
+ /* if bitstream is too short, add garbage at the end */
+ if(garbageLen > 0)
+ {
+ for(k = 0; k < garbageLen; k++)
+ {
+ ptrGarbage[k] = (WebRtc_UWord8)(rand() & 0xFF);
+ }
+
+ // for a correct length of the upper-band bit-stream together
+ // with the garbage. Garbage is embeded in upper-band bit-stream.
+ // That is the only way to preserve backward compatibility.
+ if((instISAC->bandwidthKHz == isac8kHz) ||
+ (streamLenUB == 0))
+ {
+ ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)garbageLen;
+ }
+ else
+ {
+ ptrEncodedUW8[streamLenLB] += (WebRtc_UWord8)garbageLen;
+ // write the length of the garbage at the end of the upper-band
+ // bit-stream, if exists. This helps for sanity check.
+ ptrEncodedUW8[streamLenLB + 1 + streamLenUB] = (WebRtc_UWord8)garbageLen;
+
+ }
+
+ streamLen += garbageLen;
+ }
+ }
+ else
+ {
+ /* update rate model */
+ WebRtcIsac_UpdateRateModel(&instISAC->rate_data_obj, streamLen,
+ instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck);
+ garbageLen = 0;
+ }
+
+ // Generate CRC if required.
+ if((instISAC->bandwidthKHz != isac8kHz) &&
+ (streamLenUB > 0))
+ {
+ WebRtc_UWord32 crc;
+
+ WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
+ streamLenUB + garbageLen, &crc);
+#ifndef WEBRTC_BIG_ENDIAN
+ for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
+ {
+ ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
+ (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+ }
+#else
+ memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
+ LEN_CHECK_SUM_WORD8);
+#endif
+ }
+
+ return streamLen;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_GetNewBitStream(...)
+ *
+ * This function returns encoded data, with the recieved bwe-index in the
+ * stream. If the rate is set to a value less than bottleneck of codec
+ * the new bistream will be re-encoded with the given target rate.
+ * It should always return a complete packet, i.e. only called once
+ * even for 60 msec frames.
+ *
+ * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
+ * NOTE 3! Rates larger than the bottleneck of the codec will be limited
+ * to the current bottleneck.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - bweIndex : Index of bandwidth estimate to put in new
+ * bitstream
+ * - rate : target rate of the transcoder is bits/sec.
+ * Valid values are the accepted rate in iSAC,
+ * i.e. 10000 to 56000.
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error or called in SWB mode
+ * NOTE! No error code is written to
+ * the struct since it is only allowed to read
+ * the struct.
+ */
+WebRtc_Word16 WebRtcIsac_GetNewBitStream(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 bweIndex,
+ WebRtc_Word16 jitterInfo,
+ WebRtc_Word32 rate,
+ WebRtc_Word16* encoded,
+ WebRtc_Word16 isRCU)
+{
+ Bitstr iSACBitStreamInst; /* Local struct for bitstream handling */
+ WebRtc_Word16 streamLenLB;
+ WebRtc_Word16 streamLenUB;
+ WebRtc_Word16 totalStreamLen;
+ double gain2;
+ double gain1;
+ float scale;
+ enum ISACBandwidth bandwidthKHz;
+ double rateLB;
+ double rateUB;
+ WebRtc_Word32 currentBN;
+ ISACMainStruct* instISAC;
+ WebRtc_UWord8* encodedPtrUW8 = (WebRtc_UWord8*)encoded;
+ WebRtc_UWord32 crc;
+#ifndef WEBRTC_BIG_ENDIAN
+ WebRtc_Word16 k;
+#endif
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ return -1;
+ }
+
+ // Get the bottleneck of this iSAC and limit the
+ // given rate to the current bottleneck.
+ WebRtcIsac_GetUplinkBw(ISAC_main_inst, ¤tBN);
+ if(rate > currentBN)
+ {
+ rate = currentBN;
+ }
+
+ if(WebRtcIsac_RateAllocation(rate, &rateLB, &rateUB, &bandwidthKHz) < 0)
+ {
+ return -1;
+ }
+
+ // Cannot transcode from 16 kHz to 12 kHz
+ if((bandwidthKHz == isac12kHz) &&
+ (instISAC->bandwidthKHz == isac16kHz))
+ {
+ return -1;
+ }
+
+ // These gains are in dB
+ // gain for the given rate.
+ gain1 = WebRtcIsac_GetSnr(rateLB,
+ instISAC->instLB.ISACencLB_obj.current_framesamples);
+ // gain of this iSAC
+ gain2 = WebRtcIsac_GetSnr(
+ instISAC->instLB.ISACencLB_obj.bottleneck,
+ instISAC->instLB.ISACencLB_obj.current_framesamples);
+
+ // scale is the ratio of two gains in normal domain.
+ scale = (float)pow(10, (gain1 - gain2) / 20.0);
+ // change the scale if this is a RCU bit-stream.
+ scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE):scale;
+
+ streamLenLB = WebRtcIsac_EncodeStoredDataLb(
+ &instISAC->instLB.ISACencLB_obj.SaveEnc_obj, &iSACBitStreamInst,
+ bweIndex, scale);
+
+ if(streamLenLB < 0)
+ {
+ return -1;
+ }
+
+ /* convert from bytes to WebRtc_Word16 */
+ memcpy(encoded, iSACBitStreamInst.stream, streamLenLB);
+
+ if(bandwidthKHz == isac8kHz)
+ {
+ return streamLenLB;
+ }
+
+ totalStreamLen = streamLenLB;
+ // super-wideband is always at 30ms.
+ // These gains are in dB
+ // gain for the given rate.
+ gain1 = WebRtcIsac_GetSnr(rateUB, FRAMESAMPLES);
+ // gain of this iSAC
+ gain2 = WebRtcIsac_GetSnr(
+ instISAC->instUB.ISACencUB_obj.bottleneck, FRAMESAMPLES);
+
+ // scale is the ratio of two gains in normal domain.
+ scale = (float)pow(10, (gain1 - gain2) / 20.0);
+
+ // change the scale if this is a RCU bit-stream.
+ scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE_UB):scale;
+
+ switch(instISAC->bandwidthKHz)
+ {
+ case isac12kHz:
+ {
+ streamLenUB = WebRtcIsac_EncodeStoredDataUb12(
+ &(instISAC->instUB.ISACencUB_obj.SaveEnc_obj),
+ &iSACBitStreamInst, jitterInfo, scale);
+ break;
+ }
+ case isac16kHz:
+ {
+ streamLenUB = WebRtcIsac_EncodeStoredDataUb16(
+ &(instISAC->instUB.ISACencUB_obj.SaveEnc_obj),
+ &iSACBitStreamInst, jitterInfo, scale);
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(streamLenUB < 0)
+ {
+ return -1;
+ }
+
+ if(streamLenUB + 1 + LEN_CHECK_SUM_WORD8 > 255)
+ {
+ return streamLenLB;
+ }
+
+ totalStreamLen = streamLenLB + streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+ encodedPtrUW8[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+
+ memcpy(&encodedPtrUW8[streamLenLB+1], iSACBitStreamInst.stream,
+ streamLenUB);
+
+ WebRtcIsac_GetCrc((WebRtc_Word16*)(&(encodedPtrUW8[streamLenLB + 1])),
+ streamLenUB, &crc);
+#ifndef WEBRTC_BIG_ENDIAN
+ for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
+ {
+ encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
+ (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+ }
+#else
+ memcpy(&encodedPtrUW8[streamLenLB + streamLenUB + 1], &crc,
+ LEN_CHECK_SUM_WORD8);
+#endif
+
+
+ return totalStreamLen;
+}
+
+
+/****************************************************************************
+ * DecoderInitLb(...) - internal function for initialization of
+ * Lower Band
+ * DecoderInitUb(...) - internal function for initialization of
+ * Upper Band
+ * WebRtcIsac_DecoderInit(...) - API function
+ *
+ * This function initializes a ISAC instance prior to the decoder calls.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ *
+ * Return value
+ * : 0 - Ok
+ * -1 - Error
+ */
+static WebRtc_Word16 DecoderInitLb(
+ ISACLBStruct* instISAC)
+{
+ int i;
+ /* Init stream vector to zero */
+ for (i=0; i<STREAM_SIZE_MAX_60; i++)
+ {
+ instISAC->ISACdecLB_obj.bitstr_obj.stream[i] = 0;
+ }
+
+ WebRtcIsac_InitMasking(&instISAC->ISACdecLB_obj.maskfiltstr_obj);
+ WebRtcIsac_InitPostFilterbank(
+ &instISAC->ISACdecLB_obj.postfiltbankstr_obj);
+ WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
+
+ return (0);
+}
+
+static WebRtc_Word16 DecoderInitUb(
+ ISACUBStruct* instISAC)
+{
+ int i;
+ /* Init stream vector to zero */
+ for (i = 0; i < STREAM_SIZE_MAX_60; i++)
+ {
+ instISAC->ISACdecUB_obj.bitstr_obj.stream[i] = 0;
+ }
+
+ WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
+ WebRtcIsac_InitPostFilterbank(
+ &instISAC->ISACdecUB_obj.postfiltbankstr_obj);
+ return (0);
+}
+
+WebRtc_Word16 WebRtcIsac_DecoderInit(
+ ISACStruct *ISAC_main_inst)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if(DecoderInitLb(&instISAC->instLB) < 0)
+ {
+ return -1;
+ }
+
+ if(instISAC->decoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ memset(instISAC->synthesisFBState1, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+ memset(instISAC->synthesisFBState2, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+
+ if(DecoderInitUb(&(instISAC->instUB)) < 0)
+ {
+ return -1;
+ }
+ }
+
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
+ instISAC->encoderSamplingRateKHz,
+ instISAC->decoderSamplingRateKHz);
+ }
+
+ instISAC->initFlag |= BIT_MASK_DEC_INIT;
+
+ instISAC->resetFlag_8kHz = 0;
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_UpdateBwEstimate(...)
+ *
+ * This function updates the estimate of the bandwidth.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s).
+ * - packet_size : size of the packet.
+ * - rtp_seq_number : the RTP number of the packet.
+ * - arr_ts : the arrival time of the packet (from NetEq)
+ * in samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts)
+{
+ ISACMainStruct *instISAC;
+ Bitstr streamdata;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* check if decoder initiated */
+ if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
+ BIT_MASK_DEC_INIT)
+ {
+ instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
+ return -1;
+ }
+
+ if(packet_size <= 0)
+ {
+ /* return error code if the packet length is null */
+ instISAC->errorCode = ISAC_EMPTY_PACKET;
+ return -1;
+ }
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for(k = 0; k < 10; k++)
+ {
+ streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
+ ((k&1) << 3)) & 0xFF);
+ }
+#else
+ memcpy(streamdata.stream, encoded, 10);
+#endif
+
+ err = WebRtcIsac_EstimateBandwidth(&instISAC->bwestimator_obj, &streamdata,
+ packet_size, rtp_seq_number, send_ts, arr_ts,
+ instISAC->encoderSamplingRateKHz,
+ instISAC->decoderSamplingRateKHz);
+
+ if(err < 0)
+ {
+ /* return error code if something went wrong */
+ instISAC->errorCode = -err;
+ return -1;
+ }
+
+ return 0;
+}
+
+static WebRtc_Word16 Decode(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word16 lenEncodedBytes,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType,
+ WebRtc_Word16 isRCUPayload)
+{
+ /* number of samples (480 or 960), output from decoder
+ that were actually used in the encoder/decoder
+ (determined on the fly) */
+ ISACMainStruct* instISAC;
+ ISACUBDecStruct* decInstUB;
+ ISACLBDecStruct* decInstLB;
+
+ WebRtc_Word16 numSamplesLB;
+ WebRtc_Word16 numSamplesUB;
+ WebRtc_Word16 speechIdx;
+ float outFrame[MAX_FRAMESAMPLES];
+ WebRtc_Word16 outFrameLB[MAX_FRAMESAMPLES];
+ WebRtc_Word16 outFrameUB[MAX_FRAMESAMPLES];
+ WebRtc_Word16 numDecodedBytesLB;
+ WebRtc_Word16 numDecodedBytesUB;
+ WebRtc_Word16 lenEncodedLBBytes;
+ WebRtc_Word16 validChecksum = 1;
+ WebRtc_Word16 k;
+ WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
+ WebRtc_UWord16 numLayer;
+ WebRtc_Word16 totSizeBytes;
+ WebRtc_Word16 err;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+ decInstUB = &(instISAC->instUB.ISACdecUB_obj);
+ decInstLB = &(instISAC->instLB.ISACdecLB_obj);
+
+ /* check if decoder initiated */
+ if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
+ BIT_MASK_DEC_INIT)
+ {
+ instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
+ return -1;
+ }
+
+ if(lenEncodedBytes <= 0)
+ {
+ /* return error code if the packet length is null */
+ instISAC->errorCode = ISAC_EMPTY_PACKET;
+ return -1;
+ }
+
+ // the size of the rncoded lower-band is bounded by
+ // STREAM_SIZE_MAX,
+ // If a payload with the size larger than STREAM_SIZE_MAX
+ // is received, it is not considered erroneous.
+ lenEncodedLBBytes = (lenEncodedBytes > STREAM_SIZE_MAX)
+ ? STREAM_SIZE_MAX:lenEncodedBytes;
+
+ // Copy to lower-band bit-stream structure
+ memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, ptrEncodedUW8,
+ lenEncodedLBBytes);
+
+ // Regardless of that the current codec is setup to work in
+ // wideband or super-wideband, the decoding of the lower-band
+ // has to be performed.
+ numDecodedBytesLB = WebRtcIsac_DecodeLb(outFrame, decInstLB,
+ &numSamplesLB, isRCUPayload);
+
+ // Check for error
+ if((numDecodedBytesLB < 0) ||
+ (numDecodedBytesLB > lenEncodedLBBytes) ||
+ (numSamplesLB > MAX_FRAMESAMPLES))
+ {
+ instISAC->errorCode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ // Error Check, we accept multi-layer bit-stream
+ // This will limit number of iterations of the
+ // while loop. Even withouut this the number of iterations
+ // is limited.
+ numLayer = 1;
+ totSizeBytes = numDecodedBytesLB;
+ while(totSizeBytes != lenEncodedBytes)
+ {
+ if((totSizeBytes > lenEncodedBytes) ||
+ (ptrEncodedUW8[totSizeBytes] == 0) ||
+ (numLayer > MAX_NUM_LAYERS))
+ {
+ instISAC->errorCode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+ totSizeBytes += ptrEncodedUW8[totSizeBytes];
+ numLayer++;
+ }
+
+ if(instISAC->decoderSamplingRateKHz == kIsacWideband)
+ {
+ for(k = 0; k < numSamplesLB; k++)
+ {
+ if(outFrame[k] > 32767)
+ {
+ decoded[k] = 32767;
+ }
+ else if(outFrame[k] < -32768)
+ {
+ decoded[k] = -32768;
+ }
+ else
+ {
+ decoded[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
+ }
+ }
+ numSamplesUB = 0;
+ }
+ else
+ {
+ WebRtc_UWord32 crc;
+ // We don't accept larger than 30ms (480 samples at lower-band)
+ // frame-size.
+ for(k = 0; k < numSamplesLB; k++)
+ {
+ if(outFrame[k] > 32767)
+ {
+ outFrameLB[k] = 32767;
+ }
+ else if(outFrame[k] < -32768)
+ {
+ outFrameLB[k] = -32768;
+ }
+ else
+ {
+ outFrameLB[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
+ }
+ }
+
+ //numSamplesUB = numSamplesLB;
+
+ // Check for possible error, and if upper-band stream exist.
+ if(numDecodedBytesLB == lenEncodedBytes)
+ {
+ // Decoding was successful. No super-wideband bitstream
+ // exists.
+ numSamplesUB = numSamplesLB;
+ memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
+
+ // Prepare for the potential increase of signal bandwidth
+ instISAC->resetFlag_8kHz = 2;
+ }
+ else
+ {
+ // this includes the check sum and the bytes that stores the
+ // length
+ WebRtc_Word16 lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
+
+ // Is this garbage or valid super-wideband bit-stream?
+ // Check if checksum is valid
+ if(lenNextStream <= (LEN_CHECK_SUM_WORD8 + 1))
+ {
+ // such a small second layer cannot be super-wideband layer.
+ // It must be a short garbage.
+ validChecksum = 0;
+ }
+ else
+ {
+ // Run CRC to see if the checksum match.
+ WebRtcIsac_GetCrc((WebRtc_Word16*)(
+ &ptrEncodedUW8[numDecodedBytesLB + 1]),
+ lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
+
+ validChecksum = 1;
+ for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
+ {
+ validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) ==
+ ptrEncodedUW8[numDecodedBytesLB + lenNextStream -
+ LEN_CHECK_SUM_WORD8 + k]);
+ }
+ }
+
+ if(!validChecksum)
+ {
+ // this is a garbage, we have received a wideband
+ // bit-stream with garbage
+ numSamplesUB = numSamplesLB;
+ memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
+ }
+ else
+ {
+ // A valid super-wideband biststream exists.
+ enum ISACBandwidth bandwidthKHz;
+ WebRtc_Word32 maxDelayBit;
+
+ //instISAC->bwestimator_obj.incomingStreamSampFreq =
+ // kIsacSuperWideband;
+ // If we have super-wideband bit-stream, we cannot
+ // have 60 ms frame-size.
+ if(numSamplesLB > FRAMESAMPLES)
+ {
+ instISAC->errorCode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ // the rest of the bit-stream contains the upper-band
+ // bit-stream curently this is the only thing there,
+ // however, we might add more layers.
+
+ // Have to exclude one byte where the length is stored
+ // and last 'LEN_CHECK_SUM_WORD8' bytes where the
+ // checksum is stored.
+ lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1);
+
+ memcpy(decInstUB->bitstr_obj.stream,
+ &ptrEncodedUW8[numDecodedBytesLB + 1], lenNextStream);
+
+ // THIS IS THE FIRST DECODING
+ decInstUB->bitstr_obj.W_upper = 0xFFFFFFFF;
+ decInstUB->bitstr_obj.streamval = 0;
+ decInstUB->bitstr_obj.stream_index = 0;
+
+ // Decode jitter infotmation
+ err = WebRtcIsac_DecodeJitterInfo(&decInstUB->bitstr_obj,
+ &maxDelayBit);
+ // error check
+ if(err < 0)
+ {
+ instISAC->errorCode = -err;
+ return -1;
+ }
+
+ // Update jitter info which is in the upper-band bit-stream
+ // only if the encoder is in super-wideband. Otherwise,
+ // the jitter info is already embeded in bandwidth index
+ // and has been updated.
+ if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ err = WebRtcIsac_UpdateUplinkJitter(
+ &(instISAC->bwestimator_obj), maxDelayBit);
+ if(err < 0)
+ {
+ instISAC->errorCode = -err;
+ return -1;
+ }
+ }
+
+ // decode bandwidth information
+ err = WebRtcIsac_DecodeBandwidth(&decInstUB->bitstr_obj,
+ &bandwidthKHz);
+ if(err < 0)
+ {
+ instISAC->errorCode = -err;
+ return -1;
+ }
+
+ switch(bandwidthKHz)
+ {
+ case isac12kHz:
+ {
+ numDecodedBytesUB = WebRtcIsac_DecodeUb12(outFrame,
+ decInstUB, isRCUPayload);
+
+ // Hang-over for transient alleviation -
+ // wait two frames to add the upper band going up from 8 kHz
+ if (instISAC->resetFlag_8kHz > 0)
+ {
+ if (instISAC->resetFlag_8kHz == 2)
+ {
+ // Silence first and a half frame
+ memset(outFrame, 0, MAX_FRAMESAMPLES *
+ sizeof(float));
+ }
+ else
+ {
+ const float rampStep = 2.0f / MAX_FRAMESAMPLES;
+ float rampVal = 0;
+ memset(outFrame, 0, (MAX_FRAMESAMPLES>>1) *
+ sizeof(float));
+
+ // Ramp up second half of second frame
+ for(k = MAX_FRAMESAMPLES/2; k < MAX_FRAMESAMPLES; k++)
+ {
+ outFrame[k] *= rampVal;
+ rampVal += rampStep;
+ }
+ }
+ instISAC->resetFlag_8kHz -= 1;
+ }
+
+ break;
+ }
+ case isac16kHz:
+ {
+ numDecodedBytesUB = WebRtcIsac_DecodeUb16(outFrame,
+ decInstUB, isRCUPayload);
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ // it might be less due to garbage.
+ if((numDecodedBytesUB != lenNextStream) &&
+ (numDecodedBytesUB != (lenNextStream - ptrEncodedUW8[
+ numDecodedBytesLB + 1 + numDecodedBytesUB])))
+ {
+ instISAC->errorCode = ISAC_LENGTH_MISMATCH;
+ return -1;
+ }
+
+ // If there is no error Upper-band always decodes
+ // 30 ms (480 samples)
+ numSamplesUB = FRAMESAMPLES;
+
+ // Convert to W16
+ for(k = 0; k < numSamplesUB; k++)
+ {
+ if(outFrame[k] > 32767)
+ {
+ outFrameUB[k] = 32767;
+ }
+ else if(outFrame[k] < -32768)
+ {
+ outFrameUB[k] = -32768;
+ }
+ else
+ {
+ outFrameUB[k] = (WebRtc_Word16)WebRtcIsac_lrint(
+ outFrame[k]);
+ }
+ }
+ }
+ }
+
+ speechIdx = 0;
+ while(speechIdx < numSamplesLB)
+ {
+ WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx],
+ &outFrameUB[speechIdx], &decoded[(speechIdx<<1)],
+ instISAC->synthesisFBState1, instISAC->synthesisFBState2);
+
+ speechIdx += FRAMESAMPLES_10ms;
+ }
+ }
+ *speechType = 0;
+ return (numSamplesLB + numSamplesUB);
+}
+
+
+
+
+
+
+
+/****************************************************************************
+ * WebRtcIsac_Decode(...)
+ *
+ * This function decodes a ISAC frame. Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the frameSize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcIsac_Decode(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word16 lenEncodedBytes,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType)
+{
+ WebRtc_Word16 isRCUPayload = 0;
+ return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
+ speechType, isRCUPayload);
+}
+
+/****************************************************************************
+ * WebRtcIsac_DecodeRcu(...)
+ *
+ * This function decodes a redundant (RCU) iSAC frame. Function is called in
+ * NetEq with a stored RCU payload i case of packet loss. Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the framesize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - encoded : encoded ISAC RCU frame(s)
+ * - len : bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded vector
+ * -1 - Error
+ */
+
+
+
+WebRtc_Word16 WebRtcIsac_DecodeRcu(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_UWord16* encoded,
+ WebRtc_Word16 lenEncodedBytes,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType)
+{
+ WebRtc_Word16 isRCUPayload = 1;
+ return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
+ speechType, isRCUPayload);
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_DecodePlc(...)
+ *
+ * This function conducts PLC for ISAC frame(s). Output speech length
+ * will be a multiple of 480 samples: 480 or 960 samples,
+ * depending on the frameSize (30 or 60 ms).
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - noOfLostFrames : Number of PLC frames to produce
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_DecodePlc(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* decoded,
+ WebRtc_Word16 noOfLostFrames)
+{
+ WebRtc_Word16 numSamples = 0;
+ ISACMainStruct* instISAC;
+
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
+ if(noOfLostFrames > 2)
+ {
+ noOfLostFrames = 2;
+ }
+
+ /* Get the number of samples per frame */
+ switch(instISAC->decoderSamplingRateKHz)
+ {
+ case kIsacWideband:
+ {
+ numSamples = 480 * noOfLostFrames;
+ break;
+ }
+ case kIsacSuperWideband:
+ {
+ numSamples = 960 * noOfLostFrames;
+ break;
+ }
+ }
+
+ /* Set output samples to zero */
+ memset(decoded, 0, numSamples * sizeof(WebRtc_Word16));
+ return numSamples;
+}
+
+
+/****************************************************************************
+ * ControlLb(...) - Internal function for controling Lower Band
+ * ControlUb(...) - Internal function for controling Upper Band
+ * WebRtcIsac_Control(...) - API function
+ *
+ * This function sets the limit on the short-term average bit rate and the
+ * frame length. Should be used only in Instantaneous mode.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rate : limit on the short-term average bit rate,
+ * in bits/second (between 10000 and 32000)
+ * - frameSize : number of milliseconds per frame (30 or 60)
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+static WebRtc_Word16 ControlLb(
+ ISACLBStruct* instISAC,
+ double rate,
+ WebRtc_Word16 frameSize)
+{
+ if((rate >= 10000) && (rate <= 32000))
+ {
+ instISAC->ISACencLB_obj.bottleneck = rate;
+ }
+ else
+ {
+ return -ISAC_DISALLOWED_BOTTLENECK;
+ }
+
+ if((frameSize == 30) || (frameSize == 60))
+ {
+ instISAC->ISACencLB_obj.new_framelength = (FS/1000) * frameSize;
+ }
+ else
+ {
+ return -ISAC_DISALLOWED_FRAME_LENGTH;
+ }
+
+ return 0;
+}
+
+static WebRtc_Word16 ControlUb(
+ ISACUBStruct* instISAC,
+ double rate)
+{
+ if((rate >= 10000) && (rate <= 32000))
+ {
+ instISAC->ISACencUB_obj.bottleneck = rate;
+ }
+ else
+ {
+ return -ISAC_DISALLOWED_BOTTLENECK;
+ }
+ return 0;
+}
+
+WebRtc_Word16 WebRtcIsac_Control(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 bottleneckBPS,
+ WebRtc_Word16 frameSize)
+{
+ ISACMainStruct *instISAC;
+ WebRtc_Word16 status;
+ double rateLB;
+ double rateUB;
+ enum ISACBandwidth bandwidthKHz;
+
+
+ /* Typecast pointer to real structure */
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if(instISAC->codingMode == 0)
+ {
+ /* in adaptive mode */
+ instISAC->errorCode = ISAC_MODE_MISMATCH;
+ return -1;
+ }
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ if(instISAC->encoderSamplingRateKHz == kIsacWideband)
+ {
+ // if the sampling rate is 16kHz then bandwith should be 8kHz,
+ // regardless of bottleneck.
+ bandwidthKHz = isac8kHz;
+ rateLB = (bottleneckBPS > 32000)? 32000:bottleneckBPS;
+ rateUB = 0;
+ }
+ else
+ {
+ if(WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB,
+ &bandwidthKHz) < 0)
+ {
+ return -1;
+ }
+ }
+
+ if((instISAC->encoderSamplingRateKHz == kIsacSuperWideband) &&
+ (frameSize != 30) &&
+ (bandwidthKHz != isac8kHz))
+ {
+ // Cannot have 60 ms in super-wideband
+ instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
+ return -1;
+ }
+
+ status = ControlLb(&instISAC->instLB, rateLB, frameSize);
+ if(status < 0)
+ {
+ instISAC->errorCode = -status;
+ return -1;
+ }
+ if(bandwidthKHz != isac8kHz)
+ {
+ status = ControlUb(&(instISAC->instUB), rateUB);
+ if(status < 0)
+ {
+ instISAC->errorCode = -status;
+ return -1;
+ }
+ }
+
+ //
+ // Check if bandwidth is changing from wideband to super-wideband
+ // then we have to synch data buffer of lower & upper-band. also
+ // clean up the upper-band data buffer.
+ //
+ if((instISAC->bandwidthKHz == isac8kHz) &&
+ (bandwidthKHz != isac8kHz))
+ {
+ memset(instISAC->instUB.ISACencUB_obj.data_buffer_float, 0,
+ sizeof(float) * (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES));
+
+ if(bandwidthKHz == isac12kHz)
+ {
+ instISAC->instUB.ISACencUB_obj.buffer_index =
+ instISAC->instLB.ISACencLB_obj.buffer_index;
+ }
+ else
+ {
+ instISAC->instUB.ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES +
+ instISAC->instLB.ISACencLB_obj.buffer_index;
+
+ memcpy(&(instISAC->instUB.ISACencUB_obj.lastLPCVec),
+ WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
+ }
+ }
+
+ // update the payload limit it the bandwidth is changing.
+ if(instISAC->bandwidthKHz != bandwidthKHz)
+ {
+ instISAC->bandwidthKHz = bandwidthKHz;
+ UpdatePayloadSizeLimit(instISAC);
+ }
+ instISAC->bottleneck = bottleneckBPS;
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_ControlBwe(...)
+ *
+ * This function sets the initial values of bottleneck and frame-size if
+ * iSAC is used in channel-adaptive mode. Through this API, users can
+ * enforce a frame-size for all values of bottleneck. Then iSAC will not
+ * automatically change the frame-size.
+ *
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance.
+ * - rateBPS : initial value of bottleneck in bits/second
+ * 10000 <= rateBPS <= 32000 is accepted
+ * For default bottleneck set rateBPS = 0
+ * - frameSizeMs : number of milliseconds per frame (30 or 60)
+ * - enforceFrameSize : 1 to enforce the given frame-size through out
+ * the adaptation process, 0 to let iSAC change
+ * the frame-size if required.
+ *
+ * Return value : 0 - ok
+ * -1 - Error
+ */
+WebRtc_Word16 WebRtcIsac_ControlBwe(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 bottleneckBPS,
+ WebRtc_Word16 frameSizeMs,
+ WebRtc_Word16 enforceFrameSize)
+{
+ ISACMainStruct *instISAC;
+ enum ISACBandwidth bandwidth;
+
+ /* Typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ /* Check that we are in channel-adaptive mode, otherwise, return (-1) */
+ if(instISAC->codingMode != 0)
+ {
+ instISAC->errorCode = ISAC_MODE_MISMATCH;
+ return -1;
+ }
+ if((frameSizeMs != 30) &&
+ (instISAC->encoderSamplingRateKHz == kIsacSuperWideband))
+ {
+ return -1;
+ }
+
+ /* Set struct variable if enforceFrameSize is set. ISAC will then */
+ /* keep the chosen frame size. */
+ if((enforceFrameSize != 0) /*||
+ (instISAC->samplingRateKHz == kIsacSuperWideband)*/)
+ {
+ instISAC->instLB.ISACencLB_obj.enforceFrameSize = 1;
+ }
+ else
+ {
+ instISAC->instLB.ISACencLB_obj.enforceFrameSize = 0;
+ }
+
+ /* Set initial rate, if value between 10000 and 32000, */
+ /* if rateBPS is 0, keep the default initial bottleneck value (15000) */
+ if(bottleneckBPS != 0)
+ {
+ double rateLB;
+ double rateUB;
+ if(WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB, &bandwidth) < 0)
+ {
+ return -1;
+ }
+ instISAC->bwestimator_obj.send_bw_avg = (float)bottleneckBPS;
+ instISAC->bandwidthKHz = bandwidth;
+ }
+
+ /* Set initial frameSize. If enforceFrameSize is set the frame size will
+ not change */
+ if(frameSizeMs != 0)
+ {
+ if((frameSizeMs == 30) || (frameSizeMs == 60))
+ {
+ instISAC->instLB.ISACencLB_obj.new_framelength = (FS/1000) *
+ frameSizeMs;
+ //instISAC->bwestimator_obj.rec_header_rate = ((float)HEADER_SIZE *
+ // 8.0f * 1000.0f / (float)frameSizeMs);
+ }
+ else
+ {
+ instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
+ return -1;
+ }
+ }
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_GetDownLinkBwIndex(...)
+ *
+ * This function returns index representing the Bandwidth estimate from
+ * other side to this side.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Output:
+ * - bweIndex : Bandwidth estimate to transmit to other side.
+ *
+ */
+WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* bweIndex,
+ WebRtc_Word16* jitterInfo)
+{
+ ISACMainStruct *instISAC;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
+ BIT_MASK_DEC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ /* Call function to get Bandwidth Estimate */
+ WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj),
+ bweIndex, jitterInfo, instISAC->decoderSamplingRateKHz);
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_UpdateUplinkBw(...)
+ *
+ * This function takes an index representing the Bandwidth estimate from
+ * this side to other side and updates BWE.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ * - rateIndex : Bandwidth estimate from other side.
+ *
+ * Return value : 0 - ok
+ * -1 - index out of range
+ */
+WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 bweIndex)
+{
+ ISACMainStruct *instISAC;
+ WebRtc_Word16 returnVal;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ /* Call function to get Bandwidth Estimate */
+ returnVal = WebRtcIsac_UpdateUplinkBwImpl(
+ &(instISAC->bwestimator_obj), bweIndex,
+ instISAC->encoderSamplingRateKHz);
+
+ if(returnVal < 0)
+ {
+ instISAC->errorCode = -returnVal;
+ return -1;
+ }
+ else
+ {
+ return 0;
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_ReadBwIndex(...)
+ *
+ * This function returns the index of the Bandwidth estimate from the
+ * bitstream.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ * - bweIndex : Bandwidth estimate in bitstream
+ *
+ */
+WebRtc_Word16 WebRtcIsac_ReadBwIndex(
+ const WebRtc_Word16* encoded,
+ WebRtc_Word16* bweIndex)
+{
+ Bitstr streamdata;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for(k = 0; k < 10; k++)
+ {
+ streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
+ ((k&1) << 3)) & 0xFF);
+ }
+#else
+ memcpy(streamdata.stream, encoded, 10);
+#endif
+
+ /* decode frame length */
+ err = WebRtcIsac_DecodeFrameLen(&streamdata, bweIndex);
+ if(err < 0)
+ {
+ return err;
+ }
+
+ /* decode BW estimation */
+ err = WebRtcIsac_DecodeSendBW(&streamdata, bweIndex);
+ if(err < 0)
+ {
+ return err;
+ }
+
+ return 0;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_ReadFrameLen(...)
+ *
+ * This function returns the length of the frame represented in the packet.
+ *
+ * Input:
+ * - encoded : Encoded bitstream
+ *
+ * Output:
+ * - frameLength : Length of frame in packet (in samples)
+ *
+ */
+WebRtc_Word16 WebRtcIsac_ReadFrameLen(
+ ISACStruct* ISAC_main_inst,
+ const WebRtc_Word16* encoded,
+ WebRtc_Word16* frameLength)
+{
+ Bitstr streamdata;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+ WebRtc_Word16 err;
+ ISACMainStruct* instISAC;
+
+ streamdata.W_upper = 0xFFFFFFFF;
+ streamdata.streamval = 0;
+ streamdata.stream_index = 0;
+
+#ifndef WEBRTC_BIG_ENDIAN
+ for (k=0; k<10; k++) {
+ streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
+ ((k&1) << 3)) & 0xFF);
+ }
+#else
+ memcpy(streamdata.stream, encoded, 10);
+#endif
+
+ /* decode frame length */
+ err = WebRtcIsac_DecodeFrameLen(&streamdata, frameLength);
+ if(err < 0) {
+ return -1;
+ }
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if(instISAC->decoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ // the decoded frame length indicates the number of samples in
+ // lower-band in this case, multiply by 2 to get the total number
+ // of samples.
+ *frameLength <<= 1;
+ }
+
+ return 0;
+}
+
+
+/*******************************************************************************
+ * WebRtcIsac_GetNewFrameLen(...)
+ *
+ * returns the frame lenght (in samples) of the next packet. In the case of
+ * channel-adaptive mode, iSAC decides on its frame lenght based on the
+ * estimated bottleneck this allows a user to prepare for the next packet
+ * (at the encoder).
+ *
+ * The primary usage is in CE to make the iSAC works in channel-adaptive mode
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Return Value : frame lenght in samples
+ *
+ */
+WebRtc_Word16 WebRtcIsac_GetNewFrameLen(
+ ISACStruct *ISAC_main_inst)
+{
+ ISACMainStruct *instISAC;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* Return new frame length */
+ if(instISAC->encoderSamplingRateKHz == kIsacWideband)
+ {
+ return (instISAC->instLB.ISACencLB_obj.new_framelength);
+ }
+ else
+ {
+ return ((instISAC->instLB.ISACencLB_obj.new_framelength) << 1);
+ }
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_GetErrorCode(...)
+ *
+ * This function can be used to check the error code of an iSAC instance.
+ * When a function returns -1 a error code will be set for that instance.
+ * The function below extract the code of the last error that occured in
+ * the specified instance.
+ *
+ * Input:
+ * - ISAC_main_inst : ISAC instance
+ *
+ * Return value : Error code
+ */
+WebRtc_Word16 WebRtcIsac_GetErrorCode(
+ ISACStruct *ISAC_main_inst)
+{
+ ISACMainStruct *instISAC;
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ return (instISAC->errorCode);
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_GetUplinkBw(...)
+ *
+ * This function outputs the target bottleneck of the codec. In
+ * channel-adaptive mode, the target bottleneck is specified through in-band
+ * signalling retreived by bandwidth estimator.
+ * In channel-independent, also called instantaneous mode, the target
+ * bottleneck is provided to the encoder by calling xxx_control(...) (if
+ * xxx_control is never called the default values is).
+ * Note that the output is the iSAC internal operating bottleneck whch might
+ * differ slightly from the one provided through xxx_control().
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Output:
+ * - *bottleneck : bottleneck in bits/sec
+ *
+ * Return value : -1 if error happens
+ * 0 bit-rates computed correctly.
+ */
+WebRtc_Word16 WebRtcIsac_GetUplinkBw(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32* bottleneck)
+{
+ ISACMainStruct* instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ if(instISAC->codingMode == 0)
+ {
+ // we are in adaptive mode then get the bottleneck from BWE
+ *bottleneck = (WebRtc_Word32)instISAC->bwestimator_obj.send_bw_avg;
+ }
+ else
+ {
+ *bottleneck = instISAC->bottleneck;
+ }
+
+ if((*bottleneck > 32000) && (*bottleneck < 38000))
+ {
+ *bottleneck = 32000;
+ }
+ else if((*bottleneck > 45000) && (*bottleneck < 50000))
+ {
+ *bottleneck = 45000;
+ }
+ else if(*bottleneck > 56000)
+ {
+ *bottleneck = 56000;
+ }
+
+ return 0;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_SetMaxPayloadSize(...)
+ *
+ * This function sets a limit for the maximum payload size of iSAC. The same
+ * value is used both for 30 and 60 ms packets. If the encoder sampling rate
+ * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
+ * encoder sampling rate is 32 kHz the maximum payload size is between 120
+ * and 600 bytes.
+ *
+ * ---------------
+ * IMPORTANT NOTES
+ * ---------------
+ * The size of a packet is limited to the minimum of 'max-payload-size' and
+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
+ * 170 bytes, i.e. min(170, 300).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxPayloadBytes : maximum size of the payload in bytes
+ * valid values are between 100 and 400 bytes
+ * if encoder sampling rate is 16 kHz. For
+ * 32 kHz encoder sampling rate valid values
+ * are between 100 and 600 bytes.
+ *
+ * Return value : 0 if successful
+ * -1 if error happens
+ */
+WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16 maxPayloadBytes)
+{
+ ISACMainStruct *instISAC;
+ WebRtc_Word16 status = 0;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+
+ if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
+ {
+ // sanity check
+ if(maxPayloadBytes < 120)
+ {
+ // maxRate is out of valid range
+ // set to the acceptable value and return -1.
+ maxPayloadBytes = 120;
+ status = -1;
+ }
+
+ /* sanity check */
+ if(maxPayloadBytes > STREAM_SIZE_MAX)
+ {
+ // maxRate is out of valid range
+ // set to the acceptable value and return -1.
+ maxPayloadBytes = STREAM_SIZE_MAX;
+ status = -1;
+ }
+ }
+ else
+ {
+ if(maxPayloadBytes < 120)
+ {
+ // max payload-size is out of valid range
+ // set to the acceptable value and return -1.
+ maxPayloadBytes = 120;
+ status = -1;
+ }
+ if(maxPayloadBytes > STREAM_SIZE_MAX_60)
+ {
+ // max payload-size is out of valid range
+ // set to the acceptable value and return -1.
+ maxPayloadBytes = STREAM_SIZE_MAX_60;
+ status = -1;
+ }
+ }
+ instISAC->maxPayloadSizeBytes = maxPayloadBytes;
+ UpdatePayloadSizeLimit(instISAC);
+ return status;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_SetMaxRate(...)
+ *
+ * This function sets the maximum rate which the codec may not exceed for
+ * any signal packet. The maximum rate is defined and payload-size per
+ * frame-size in bits per second.
+ *
+ * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
+ * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
+ * if the encoder sampling rate is 32 kHz.
+ *
+ * It is possible to set a maximum rate between 32000 and 53400 bits/sec
+ * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
+ *
+ * ---------------
+ * IMPORTANT NOTES
+ * ---------------
+ * The size of a packet is limited to the minimum of 'max-payload-size' and
+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
+ * 170 bytes, min(170, 300).
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - maxRate : maximum rate in bits per second,
+ * valid values are 32000 to 53400 bits/sec in
+ * wideband mode, and 32000 to 160000 bits/sec in
+ * super-wideband mode.
+ *
+ * Return value : 0 if successful
+ * -1 if error happens
+ */
+WebRtc_Word16 WebRtcIsac_SetMaxRate(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word32 maxRate)
+{
+ ISACMainStruct *instISAC;
+ WebRtc_Word16 maxRateInBytesPer30Ms;
+ WebRtc_Word16 status = 0;
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct *)ISAC_main_inst;
+
+ /* check if encoder initiated */
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ return -1;
+ }
+ /*
+ Calculate maximum number of bytes per 30 msec packets for the
+ given maximum rate. Multiply with 30/1000 to get number of
+ bits per 30 ms, divide by 8 to get number of bytes per 30 ms:
+ maxRateInBytes = floor((maxRate * 30/1000) / 8);
+ */
+ maxRateInBytesPer30Ms = (WebRtc_Word16)(maxRate*3/800);
+
+ if(instISAC->encoderSamplingRateKHz == kIsacWideband)
+ {
+ if(maxRate < 32000)
+ {
+ // max rate is out of valid range
+ // set to the acceptable value and return -1.
+ maxRateInBytesPer30Ms = 120;
+ status = -1;
+ }
+
+ if(maxRate > 53400)
+ {
+ // max rate is out of valid range
+ // set to the acceptable value and return -1.
+ maxRateInBytesPer30Ms = 200;
+ status = -1;
+ }
+ }
+ else
+ {
+ if(maxRateInBytesPer30Ms < 120)
+ {
+ // maxRate is out of valid range
+ // set to the acceptable value and return -1.
+ maxRateInBytesPer30Ms = 120;
+ status = -1;
+ }
+
+ if(maxRateInBytesPer30Ms > STREAM_SIZE_MAX)
+ {
+ // maxRate is out of valid range
+ // set to the acceptable value and return -1.
+ maxRateInBytesPer30Ms = STREAM_SIZE_MAX;
+ status = -1;
+ }
+ }
+ instISAC->maxRateBytesPer30Ms = maxRateInBytesPer30Ms;
+ UpdatePayloadSizeLimit(instISAC);
+ return status;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_GetRedPayload(...)
+ *
+ * Populates "encoded" with the redundant payload of the recently encoded
+ * frame. This function has to be called once that WebRtcIsac_Encode(...)
+ * returns a positive value. Regardless of the frame-size this function will
+ * be called only once after encoding is completed. The bit-stream is
+ * targeted for 16000 bit/sec.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC struct
+ *
+ * Output:
+ * - encoded : the encoded data vector
+ *
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * : -1 - Error
+ *
+ *
+ */
+WebRtc_Word16 WebRtcIsac_GetRedPayload(
+ ISACStruct* ISAC_main_inst,
+ WebRtc_Word16* encoded)
+{
+ ISACMainStruct* instISAC;
+ Bitstr iSACBitStreamInst;
+ WebRtc_Word16 streamLenLB;
+ WebRtc_Word16 streamLenUB;
+ WebRtc_Word16 streamLen;
+ WebRtc_Word16 totalLenUB;
+ WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
+#ifndef WEBRTC_BIG_ENDIAN
+ int k;
+#endif
+
+ /* typecast pointer to real structure */
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+
+ if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
+ }
+
+
+ iSACBitStreamInst.W_upper = 0xFFFFFFFF;
+ iSACBitStreamInst.streamval = 0;
+ iSACBitStreamInst.stream_index = 0;
+
+
+ streamLenLB = WebRtcIsac_EncodeStoredDataLb(
+ &instISAC->instLB.ISACencLB_obj.SaveEnc_obj,
+ &iSACBitStreamInst,
+ instISAC->instLB.ISACencLB_obj.lastBWIdx,
+ RCU_TRANSCODING_SCALE);
+
+ if(streamLenLB < 0)
+ {
+ return -1;
+ }
+
+ /* convert from bytes to WebRtc_Word16 */
+ memcpy(ptrEncodedUW8, iSACBitStreamInst.stream, streamLenLB);
+
+ streamLen = streamLenLB;
+
+ if(instISAC->bandwidthKHz == isac8kHz)
+ {
+ return streamLenLB;
+ }
+
+ streamLenUB = WebRtcIsac_GetRedPayloadUb(
+ &instISAC->instUB.ISACencUB_obj.SaveEnc_obj,
+ &iSACBitStreamInst, instISAC->bandwidthKHz);
+
+ if(streamLenUB < 0)
+ {
+ // an error has happened but this is not the error due to a
+ // bit-stream larger than the limit
+ return -1;
+ }
+
+ // We have one byte to write the total length of the upper band
+ // the length include the bitstream length, check-sum and the
+ // single byte where the length is written to. This is according to
+ // iSAC wideband and how the "garbage" is dealt.
+ totalLenUB = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+ if(totalLenUB > 255)
+ {
+ streamLenUB = 0;
+ }
+
+ // Generate CRC if required.
+ if((instISAC->bandwidthKHz != isac8kHz) &&
+ (streamLenUB > 0))
+ {
+ WebRtc_UWord32 crc;
+ streamLen += totalLenUB;
+ ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)totalLenUB;
+ memcpy(&ptrEncodedUW8[streamLenLB+1], iSACBitStreamInst.stream, streamLenUB);
+
+ WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
+ streamLenUB, &crc);
+#ifndef WEBRTC_BIG_ENDIAN
+ for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
+ {
+ ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
+ (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
+ }
+#else
+ memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
+ LEN_CHECK_SUM_WORD8);
+#endif
+ }
+
+
+ return streamLen;
+}
+
+
+/****************************************************************************
+ * WebRtcIsac_version(...)
+ *
+ * This function returns the version number.
+ *
+ * Output:
+ * - version : Pointer to character string
+ *
+ */
+void WebRtcIsac_version(char *version)
+{
+ strcpy(version, "4.3.0");
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_SetEncSampRate()
+ * Set the sampling rate of the encoder. Initialization of the encoder WILL
+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
+ * which is set when the instance is created. The encoding-mode and the
+ * bottleneck remain unchanged by this call, however, the maximum rate and
+ * maximum payload-size will reset to their default value.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - sampRate : enumerator specifying the sampling rate.
+ *
+ * Return value : 0 if successful
+ * -1 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_SetEncSampRate(
+ ISACStruct* ISAC_main_inst,
+ enum IsacSamplingRate sampRate)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if((sampRate != kIsacWideband) &&
+ (sampRate != kIsacSuperWideband))
+ {
+ // Sampling Frequency is not supported
+ instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
+ return -1;
+ }
+ else if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
+ BIT_MASK_ENC_INIT)
+ {
+ if(sampRate == kIsacWideband)
+ {
+ instISAC->bandwidthKHz = isac8kHz;
+ }
+ else
+ {
+ instISAC->bandwidthKHz = isac16kHz;
+ }
+ instISAC->encoderSamplingRateKHz = sampRate;
+ return 0;
+ }
+ else
+ {
+ ISACUBStruct* instUB = &(instISAC->instUB);
+ ISACLBStruct* instLB = &(instISAC->instLB);
+ double bottleneckLB;
+ double bottleneckUB;
+ WebRtc_Word32 bottleneck = instISAC->bottleneck;
+ WebRtc_Word16 codingMode = instISAC->codingMode;
+ WebRtc_Word16 frameSizeMs = instLB->ISACencLB_obj.new_framelength / (FS / 1000);
+
+ if((sampRate == kIsacWideband) &&
+ (instISAC->encoderSamplingRateKHz == kIsacSuperWideband))
+ {
+ // changing from super-wideband to wideband.
+ // we don't need to re-initialize the encoder of the
+ // lower-band.
+ instISAC->bandwidthKHz = isac8kHz;
+ if(codingMode == 1)
+ {
+ ControlLb(instLB,
+ (bottleneck > 32000)? 32000:bottleneck, FRAMESIZE);
+ }
+ instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
+ instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
+ }
+ else if((sampRate == kIsacSuperWideband) &&
+ (instISAC->encoderSamplingRateKHz == kIsacWideband))
+ {
+ if(codingMode == 1)
+ {
+ WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB,
+ &(instISAC->bandwidthKHz));
+ }
+
+ instISAC->bandwidthKHz = isac16kHz;
+ instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
+ instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
+
+ EncoderInitLb(instLB, codingMode, sampRate);
+ EncoderInitUb(instUB, instISAC->bandwidthKHz);
+
+ memset(instISAC->analysisFBState1, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+ memset(instISAC->analysisFBState2, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+
+ if(codingMode == 1)
+ {
+ instISAC->bottleneck = bottleneck;
+ ControlLb(instLB, bottleneckLB,
+ (instISAC->bandwidthKHz == isac8kHz)? frameSizeMs:FRAMESIZE);
+ if(instISAC->bandwidthKHz > isac8kHz)
+ {
+ ControlUb(instUB, bottleneckUB);
+ }
+ }
+ else
+ {
+ instLB->ISACencLB_obj.enforceFrameSize = 0;
+ instLB->ISACencLB_obj.new_framelength = FRAMESAMPLES;
+ }
+ }
+ instISAC->encoderSamplingRateKHz = sampRate;
+ return 0;
+ }
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_SetDecSampRate()
+ * Set the sampling rate of the decoder. Initialization of the decoder WILL
+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
+ * which is set when the instance is created.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ * - sampRate : enumerator specifying the sampling rate.
+ *
+ * Return value : 0 if successful
+ * -1 if failed.
+ */
+WebRtc_Word16 WebRtcIsac_SetDecSampRate(
+ ISACStruct* ISAC_main_inst,
+ enum IsacSamplingRate sampRate)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ if((sampRate != kIsacWideband) &&
+ (sampRate != kIsacSuperWideband))
+ {
+ // Sampling Frequency is not supported
+ instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
+ return -1;
+ }
+ else
+ {
+ if((instISAC->decoderSamplingRateKHz == kIsacWideband) &&
+ (sampRate == kIsacSuperWideband))
+ {
+ // switching from wideband to super-wideband at the decoder
+ // we need to reset the filter-bank and initialize
+ // upper-band decoder.
+ memset(instISAC->synthesisFBState1, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+ memset(instISAC->synthesisFBState2, 0,
+ FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
+
+ if(DecoderInitUb(&(instISAC->instUB)) < 0)
+ {
+ return -1;
+ }
+ }
+ instISAC->decoderSamplingRateKHz = sampRate;
+ return 0;
+ }
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_EncSampRate()
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : enumerator representing sampling frequency
+ * associated with the encoder, the input audio
+ * is expected to be sampled at this rate.
+ *
+ */
+enum IsacSamplingRate WebRtcIsac_EncSampRate(
+ ISACStruct* ISAC_main_inst)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ return instISAC->encoderSamplingRateKHz;
+}
+
+
+/******************************************************************************
+ * WebRtcIsac_DecSampRate()
+ * Return the sampling rate of the decoded audio.
+ *
+ * Input:
+ * - ISAC_main_inst : iSAC instance
+ *
+ * Return value : enumerator representing sampling frequency
+ * associated with the decoder, i.e. the
+ * sampling rate of the decoded audio.
+ *
+ */
+enum IsacSamplingRate WebRtcIsac_DecSampRate(
+ ISACStruct* ISAC_main_inst)
+{
+ ISACMainStruct* instISAC;
+
+ instISAC = (ISACMainStruct*)ISAC_main_inst;
+
+ return instISAC->decoderSamplingRateKHz;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi
new file mode 100644
index 0000000..d30be55
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi
@@ -0,0 +1,91 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'iSAC',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/isac.h',
+ 'arith_routines.c',
+ 'arith_routines_hist.c',
+ 'arith_routines_logist.c',
+ 'bandwidth_estimator.c',
+ 'crc.c',
+ 'decode.c',
+ 'decode_bwe.c',
+ 'encode.c',
+ 'encode_lpc_swb.c',
+ 'entropy_coding.c',
+ 'fft.c',
+ 'filter_functions.c',
+ 'filterbank_tables.c',
+ 'intialize.c',
+ 'isac.c',
+ 'filterbanks.c',
+ 'pitch_lag_tables.c',
+ 'lattice.c',
+ 'lpc_gain_swb_tables.c',
+ 'lpc_analysis.c',
+ 'lpc_shape_swb12_tables.c',
+ 'lpc_shape_swb16_tables.c',
+ 'lpc_tables.c',
+ 'pitch_estimator.c',
+ 'pitch_filter.c',
+ 'pitch_gain_tables.c',
+ 'spectrum_ar_model_tables.c',
+ 'transform.c',
+ 'arith_routines.h',
+ 'bandwidth_estimator.h',
+ 'codec.h',
+ 'crc.h',
+ 'encode_lpc_swb.h',
+ 'entropy_coding.h',
+ 'fft.h',
+ 'filterbank_tables.h',
+ 'lpc_gain_swb_tables.h',
+ 'lpc_analysis.h',
+ 'lpc_shape_swb12_tables.h',
+ 'lpc_shape_swb16_tables.h',
+ 'lpc_tables.h',
+ 'pitch_estimator.h',
+ 'pitch_gain_tables.h',
+ 'pitch_lag_tables.h',
+ 'settings.h',
+ 'spectrum_ar_model_tables.h',
+ 'structs.h',
+ 'os_specific_inline.h',
+ ],
+ 'conditions': [
+ ['OS!="win"', {
+ 'defines': [
+ 'WEBRTC_LINUX',
+ ],
+ }],
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c
new file mode 100644
index 0000000..a46135a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lattice.c
+ *
+ * contains the normalized lattice filter routines (MA and AR) for iSAC codec
+ *
+ */
+#include "settings.h"
+#include "codec.h"
+
+#include <math.h>
+#include <memory.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+/* filter the signal using normalized lattice filter */
+/* MA filter */
+void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
+ float *stateF,
+ float *stateG,
+ float *lat_in,
+ double *filtcoeflo,
+ double *lat_out)
+{
+ int n,k,i,u,temp1;
+ int ord_1 = orderCoef+1;
+ float sth[MAX_AR_MODEL_ORDER];
+ float cth[MAX_AR_MODEL_ORDER];
+ float inv_cth[MAX_AR_MODEL_ORDER];
+ double a[MAX_AR_MODEL_ORDER+1];
+ float f[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN], g[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
+ float gain1;
+
+ for (u=0;u<SUBFRAMES;u++)
+ {
+ /* set the Direct Form coefficients */
+ temp1 = u*ord_1;
+ a[0] = 1;
+ memcpy(a+1, filtcoeflo+temp1+1, sizeof(double) * (ord_1-1));
+
+ /* compute lattice filter coefficients */
+ WebRtcIsac_Dir2Lat(a,orderCoef,sth,cth);
+
+ /* compute the gain */
+ gain1 = (float)filtcoeflo[temp1];
+ for (k=0;k<orderCoef;k++)
+ {
+ gain1 *= cth[k];
+ inv_cth[k] = 1/cth[k];
+ }
+
+ /* normalized lattice filter */
+ /*****************************/
+
+ /* initial conditions */
+ for (i=0;i<HALF_SUBFRAMELEN;i++)
+ {
+ f[0][i] = lat_in[i + u * HALF_SUBFRAMELEN];
+ g[0][i] = lat_in[i + u * HALF_SUBFRAMELEN];
+ }
+
+ /* get the state of f&g for the first input, for all orders */
+ for (i=1;i<ord_1;i++)
+ {
+ f[i][0] = inv_cth[i-1]*(f[i-1][0] + sth[i-1]*stateG[i-1]);
+ g[i][0] = cth[i-1]*stateG[i-1] + sth[i-1]* f[i][0];
+ }
+
+ /* filtering */
+ for(k=0;k<orderCoef;k++)
+ {
+ for(n=0;n<(HALF_SUBFRAMELEN-1);n++)
+ {
+ f[k+1][n+1] = inv_cth[k]*(f[k][n+1] + sth[k]*g[k][n]);
+ g[k+1][n+1] = cth[k]*g[k][n] + sth[k]* f[k+1][n+1];
+ }
+ }
+
+ for(n=0;n<HALF_SUBFRAMELEN;n++)
+ {
+ lat_out[n + u * HALF_SUBFRAMELEN] = gain1 * f[orderCoef][n];
+ }
+
+ /* save the states */
+ for (i=0;i<ord_1;i++)
+ {
+ stateF[i] = f[i][HALF_SUBFRAMELEN-1];
+ stateG[i] = g[i][HALF_SUBFRAMELEN-1];
+ }
+ /* process next frame */
+ }
+
+ return;
+}
+
+
+/*///////////////////AR filter ///////////////////////////////*/
+/* filter the signal using normalized lattice filter */
+void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
+ float *stateF,
+ float *stateG,
+ double *lat_in,
+ double *lo_filt_coef,
+ float *lat_out)
+{
+ int n,k,i,u,temp1;
+ int ord_1 = orderCoef+1;
+ float sth[MAX_AR_MODEL_ORDER];
+ float cth[MAX_AR_MODEL_ORDER];
+ double a[MAX_AR_MODEL_ORDER+1];
+ float ARf[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN], ARg[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
+ float gain1,inv_gain1;
+
+ for (u=0;u<SUBFRAMES;u++)
+ {
+ /* set the denominator and numerator of the Direct Form */
+ temp1 = u*ord_1;
+ a[0] = 1;
+
+ memcpy(a+1, lo_filt_coef+temp1+1, sizeof(double) * (ord_1-1));
+
+ WebRtcIsac_Dir2Lat(a,orderCoef,sth,cth);
+
+ gain1 = (float)lo_filt_coef[temp1];
+ for (k=0;k<orderCoef;k++)
+ {
+ gain1 = cth[k]*gain1;
+ }
+
+ /* initial conditions */
+ inv_gain1 = 1/gain1;
+ for (i=0;i<HALF_SUBFRAMELEN;i++)
+ {
+ ARf[orderCoef][i] = (float)lat_in[i + u * HALF_SUBFRAMELEN]*inv_gain1;
+ }
+
+
+ for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
+ {
+ ARf[i][0] = cth[i]*ARf[i+1][0] - sth[i]*stateG[i];
+ ARg[i+1][0] = sth[i]*ARf[i+1][0] + cth[i]* stateG[i];
+ }
+ ARg[0][0] = ARf[0][0];
+
+ for(n=0;n<(HALF_SUBFRAMELEN-1);n++)
+ {
+ for(k=orderCoef-1;k>=0;k--)
+ {
+ ARf[k][n+1] = cth[k]*ARf[k+1][n+1] - sth[k]*ARg[k][n];
+ ARg[k+1][n+1] = sth[k]*ARf[k+1][n+1] + cth[k]* ARg[k][n];
+ }
+ ARg[0][n+1] = ARf[0][n+1];
+ }
+
+ memcpy(lat_out+u * HALF_SUBFRAMELEN, &(ARf[0][0]), sizeof(float) * HALF_SUBFRAMELEN);
+
+ /* cannot use memcpy in the following */
+ for (i=0;i<ord_1;i++)
+ {
+ stateF[i] = ARf[i][HALF_SUBFRAMELEN-1];
+ stateG[i] = ARg[i][HALF_SUBFRAMELEN-1];
+ }
+
+ }
+
+ return;
+}
+
+
+/* compute the reflection coefficients using the step-down procedure*/
+/* converts the direct form parameters to lattice form.*/
+/* a and b are vectors which contain the direct form coefficients,
+ according to
+ A(z) = a(1) + a(2)*z + a(3)*z^2 + ... + a(M+1)*z^M
+ B(z) = b(1) + b(2)*z + b(3)*z^2 + ... + b(M+1)*z^M
+*/
+
+void WebRtcIsac_Dir2Lat(double *a,
+ int orderCoef,
+ float *sth,
+ float *cth)
+{
+ int m, k;
+ float tmp[MAX_AR_MODEL_ORDER];
+ float tmp_inv, cth2;
+
+ sth[orderCoef-1] = (float)a[orderCoef];
+ cth2 = 1.0f - sth[orderCoef-1] * sth[orderCoef-1];
+ cth[orderCoef-1] = (float)sqrt(cth2);
+ for (m=orderCoef-1; m>0; m--)
+ {
+ tmp_inv = 1.0f / cth2;
+ for (k=1; k<=m; k++)
+ {
+ tmp[k] = ((float)a[k] - sth[m] * (float)a[m-k+1]) * tmp_inv;
+ }
+
+ for (k=1; k<m; k++)
+ {
+ a[k] = tmp[k];
+ }
+
+ sth[m-1] = tmp[m];
+ cth2 = 1 - sth[m-1] * sth[m-1];
+ cth[m-1] = (float)sqrt(cth2);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c
new file mode 100644
index 0000000..854b2d7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c
@@ -0,0 +1,535 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "lpc_analysis.h"
+#include "settings.h"
+#include "codec.h"
+#include "entropy_coding.h"
+
+#include <math.h>
+#include <string.h>
+
+#define LEVINSON_EPS 1.0e-10
+
+
+/* window */
+/* Matlab generation code:
+ * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid;
+ * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end
+ */
+static const double kLpcCorrWindow[WINLEN] = {
+ 0.00000000, 0.00000001, 0.00000004, 0.00000010, 0.00000020,
+ 0.00000035, 0.00000055, 0.00000083, 0.00000118, 0.00000163,
+ 0.00000218, 0.00000283, 0.00000361, 0.00000453, 0.00000558, 0.00000679,
+ 0.00000817, 0.00000973, 0.00001147, 0.00001342, 0.00001558,
+ 0.00001796, 0.00002058, 0.00002344, 0.00002657, 0.00002997,
+ 0.00003365, 0.00003762, 0.00004190, 0.00004651, 0.00005144, 0.00005673,
+ 0.00006236, 0.00006837, 0.00007476, 0.00008155, 0.00008875,
+ 0.00009636, 0.00010441, 0.00011290, 0.00012186, 0.00013128,
+ 0.00014119, 0.00015160, 0.00016252, 0.00017396, 0.00018594, 0.00019846,
+ 0.00021155, 0.00022521, 0.00023946, 0.00025432, 0.00026978,
+ 0.00028587, 0.00030260, 0.00031998, 0.00033802, 0.00035674,
+ 0.00037615, 0.00039626, 0.00041708, 0.00043863, 0.00046092, 0.00048396,
+ 0.00050775, 0.00053233, 0.00055768, 0.00058384, 0.00061080,
+ 0.00063858, 0.00066720, 0.00069665, 0.00072696, 0.00075813,
+ 0.00079017, 0.00082310, 0.00085692, 0.00089164, 0.00092728, 0.00096384,
+ 0.00100133, 0.00103976, 0.00107914, 0.00111947, 0.00116077,
+ 0.00120304, 0.00124630, 0.00129053, 0.00133577, 0.00138200,
+ 0.00142924, 0.00147749, 0.00152676, 0.00157705, 0.00162836, 0.00168070,
+ 0.00173408, 0.00178850, 0.00184395, 0.00190045, 0.00195799,
+ 0.00201658, 0.00207621, 0.00213688, 0.00219860, 0.00226137,
+ 0.00232518, 0.00239003, 0.00245591, 0.00252284, 0.00259079, 0.00265977,
+ 0.00272977, 0.00280078, 0.00287280, 0.00294582, 0.00301984,
+ 0.00309484, 0.00317081, 0.00324774, 0.00332563, 0.00340446,
+ 0.00348421, 0.00356488, 0.00364644, 0.00372889, 0.00381220, 0.00389636,
+ 0.00398135, 0.00406715, 0.00415374, 0.00424109, 0.00432920,
+ 0.00441802, 0.00450754, 0.00459773, 0.00468857, 0.00478001,
+ 0.00487205, 0.00496464, 0.00505775, 0.00515136, 0.00524542, 0.00533990,
+ 0.00543476, 0.00552997, 0.00562548, 0.00572125, 0.00581725,
+ 0.00591342, 0.00600973, 0.00610612, 0.00620254, 0.00629895,
+ 0.00639530, 0.00649153, 0.00658758, 0.00668341, 0.00677894, 0.00687413,
+ 0.00696891, 0.00706322, 0.00715699, 0.00725016, 0.00734266,
+ 0.00743441, 0.00752535, 0.00761540, 0.00770449, 0.00779254,
+ 0.00787947, 0.00796519, 0.00804963, 0.00813270, 0.00821431, 0.00829437,
+ 0.00837280, 0.00844949, 0.00852436, 0.00859730, 0.00866822,
+ 0.00873701, 0.00880358, 0.00886781, 0.00892960, 0.00898884,
+ 0.00904542, 0.00909923, 0.00915014, 0.00919805, 0.00924283, 0.00928436,
+ 0.00932252, 0.00935718, 0.00938821, 0.00941550, 0.00943890,
+ 0.00945828, 0.00947351, 0.00948446, 0.00949098, 0.00949294,
+ 0.00949020, 0.00948262, 0.00947005, 0.00945235, 0.00942938, 0.00940099,
+ 0.00936704, 0.00932738, 0.00928186, 0.00923034, 0.00917268,
+ 0.00910872, 0.00903832, 0.00896134, 0.00887763, 0.00878706,
+ 0.00868949, 0.00858478, 0.00847280, 0.00835343, 0.00822653, 0.00809199,
+ 0.00794970, 0.00779956, 0.00764145, 0.00747530, 0.00730103,
+ 0.00711857, 0.00692787, 0.00672888, 0.00652158, 0.00630597,
+ 0.00608208, 0.00584994, 0.00560962, 0.00536124, 0.00510493, 0.00484089,
+ 0.00456935, 0.00429062, 0.00400505, 0.00371310, 0.00341532,
+ 0.00311238, 0.00280511, 0.00249452, 0.00218184, 0.00186864,
+ 0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881
+};
+
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order)
+{
+
+ double sum, alpha;
+ int m, m_h, i;
+ alpha = 0; //warning -DH
+ a[0] = 1.0;
+ if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
+ for (i = 0; i < order; i++) {
+ k[i] = 0;
+ a[i+1] = 0;
+ }
+ } else {
+ a[1] = k[0] = -r[1]/r[0];
+ alpha = r[0] + r[1] * k[0];
+ for (m = 1; m < order; m++){
+ sum = r[m + 1];
+ for (i = 0; i < m; i++){
+ sum += a[i+1] * r[m - i];
+ }
+ k[m] = -sum / alpha;
+ alpha += k[m] * sum;
+ m_h = (m + 1) >> 1;
+ for (i = 0; i < m_h; i++){
+ sum = a[i+1] + k[m] * a[m - i];
+ a[m - i] += k[m] * a[i+1];
+ a[i+1] = sum;
+ }
+ a[m+1] = k[m];
+ }
+ }
+ return alpha;
+}
+
+
+//was static before, but didn't work with MEX file
+void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
+ double *oldEnergy, double *varscale)
+{
+ double nrg[4], chng, pg;
+ int k;
+
+ double pitchGains[4]={0,0,0,0};;
+
+ /* Calculate energies of first and second frame halfs */
+ nrg[0] = 0.0001;
+ for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES_QUARTER + QLOOKAHEAD) / 2; k++) {
+ nrg[0] += input[k]*input[k];
+ }
+ nrg[1] = 0.0001;
+ for ( ; k < (FRAMESAMPLES_HALF + QLOOKAHEAD) / 2; k++) {
+ nrg[1] += input[k]*input[k];
+ }
+ nrg[2] = 0.0001;
+ for ( ; k < (FRAMESAMPLES*3/4 + QLOOKAHEAD) / 2; k++) {
+ nrg[2] += input[k]*input[k];
+ }
+ nrg[3] = 0.0001;
+ for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) {
+ nrg[3] += input[k]*input[k];
+ }
+
+ /* Calculate average level change */
+ chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) +
+ fabs(10.0 * log10(nrg[2] / nrg[1])) +
+ fabs(10.0 * log10(nrg[1] / nrg[0])) +
+ fabs(10.0 * log10(nrg[0] / *oldEnergy)));
+
+
+ /* Find average pitch gain */
+ pg = 0.0;
+ for (k=0; k<4; k++)
+ {
+ pitchGains[k] = ((float)pitchGains_Q12[k])/4096;
+ pg += pitchGains[k];
+ }
+ pg *= 0.25;
+
+ /* If pitch gain is low and energy constant - increase noise level*/
+ /* Matlab code:
+ pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) ))
+ */
+ *varscale = 0.0 + 1.0 * exp( -1.4 * exp(-200.0 * pg*pg*pg) / (1.0 + 0.4 * chng) );
+
+ *oldEnergy = nrg[3];
+}
+
+void
+WebRtcIsac_GetVarsUB(
+ const double* input,
+ double* oldEnergy,
+ double* varscale)
+{
+ double nrg[4], chng;
+ int k;
+
+ /* Calculate energies of first and second frame halfs */
+ nrg[0] = 0.0001;
+ for (k = 0; k < (FRAMESAMPLES_QUARTER) / 2; k++) {
+ nrg[0] += input[k]*input[k];
+ }
+ nrg[1] = 0.0001;
+ for ( ; k < (FRAMESAMPLES_HALF) / 2; k++) {
+ nrg[1] += input[k]*input[k];
+ }
+ nrg[2] = 0.0001;
+ for ( ; k < (FRAMESAMPLES*3/4) / 2; k++) {
+ nrg[2] += input[k]*input[k];
+ }
+ nrg[3] = 0.0001;
+ for ( ; k < (FRAMESAMPLES) / 2; k++) {
+ nrg[3] += input[k]*input[k];
+ }
+
+ /* Calculate average level change */
+ chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) +
+ fabs(10.0 * log10(nrg[2] / nrg[1])) +
+ fabs(10.0 * log10(nrg[1] / nrg[0])) +
+ fabs(10.0 * log10(nrg[0] / *oldEnergy)));
+
+
+ /* If pitch gain is low and energy constant - increase noise level*/
+ /* Matlab code:
+ pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) ))
+ */
+ *varscale = exp( -1.4 / (1.0 + 0.4 * chng) );
+
+ *oldEnergy = nrg[3];
+}
+
+void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
+ double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
+ double *lo_coeff, double *hi_coeff)
+{
+ int k, n, j, pos1, pos2;
+ double varscale;
+
+ double DataLo[WINLEN], DataHi[WINLEN];
+ double corrlo[ORDERLO+2], corrlo2[ORDERLO+1];
+ double corrhi[ORDERHI+1];
+ double k_veclo[ORDERLO], k_vechi[ORDERHI];
+
+ double a_LO[ORDERLO+1], a_HI[ORDERHI+1];
+ double tmp, res_nrg;
+
+ double FwdA, FwdB;
+
+ /* hearing threshold level in dB; higher value gives more noise */
+ const double HearThresOffset = -28.0;
+
+ /* bandwdith expansion factors for low- and high band */
+ const double gammaLo = 0.9;
+ const double gammaHi = 0.8;
+
+ /* less-noise-at-low-frequencies factor */
+ double aa;
+
+
+ /* convert from dB to signal level */
+ const double H_T_H = pow(10.0, 0.05 * HearThresOffset);
+ double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46; /* divide by sqrt(12) */
+
+ /* change quallevel depending on pitch gains and level fluctuations */
+ WebRtcIsac_GetVars(inLo, pitchGains_Q12, &(maskdata->OldEnergy), &varscale);
+
+ /* less-noise-at-low-frequencies factor */
+ aa = 0.35 * (0.5 + 0.5 * varscale);
+
+ /* replace data in buffer by new look-ahead data */
+ for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++)
+ maskdata->DataBufferLo[pos1 + WINLEN - QLOOKAHEAD] = inLo[pos1];
+
+ for (k = 0; k < SUBFRAMES; k++) {
+
+ /* Update input buffer and multiply signal with window */
+ for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
+ maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 + UPDATE/2];
+ maskdata->DataBufferHi[pos1] = maskdata->DataBufferHi[pos1 + UPDATE/2];
+ DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
+ DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1];
+ }
+ pos2 = k * UPDATE/2;
+ for (n = 0; n < UPDATE/2; n++, pos1++) {
+ maskdata->DataBufferLo[pos1] = inLo[QLOOKAHEAD + pos2];
+ maskdata->DataBufferHi[pos1] = inHi[pos2++];
+ DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
+ DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1];
+ }
+
+ /* Get correlation coefficients */
+ WebRtcIsac_AutoCorr(corrlo, DataLo, WINLEN, ORDERLO+1); /* computing autocorrelation */
+ WebRtcIsac_AutoCorr(corrhi, DataHi, WINLEN, ORDERHI);
+
+
+ /* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */
+ corrlo2[0] = (1.0+aa*aa) * corrlo[0] - 2.0*aa * corrlo[1];
+ tmp = (1.0 + aa*aa);
+ for (n = 1; n <= ORDERLO; n++) {
+ corrlo2[n] = tmp * corrlo[n] - aa * (corrlo[n-1] + corrlo[n+1]);
+ }
+ tmp = (1.0+aa) * (1.0+aa);
+ for (n = 0; n <= ORDERHI; n++) {
+ corrhi[n] = tmp * corrhi[n];
+ }
+
+ /* add white noise floor */
+ corrlo2[0] += 1e-6;
+ corrhi[0] += 1e-6;
+
+
+ FwdA = 0.01;
+ FwdB = 0.01;
+
+ /* recursive filtering of correlation over subframes */
+ for (n = 0; n <= ORDERLO; n++) {
+ maskdata->CorrBufLo[n] = FwdA * maskdata->CorrBufLo[n] + corrlo2[n];
+ corrlo2[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufLo[n] + (1.0-FwdB) * corrlo2[n];
+ }
+ for (n = 0; n <= ORDERHI; n++) {
+ maskdata->CorrBufHi[n] = FwdA * maskdata->CorrBufHi[n] + corrhi[n];
+ corrhi[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufHi[n] + (1.0-FwdB) * corrhi[n];
+ }
+
+ /* compute prediction coefficients */
+ WebRtcIsac_LevDurb(a_LO, k_veclo, corrlo2, ORDERLO);
+ WebRtcIsac_LevDurb(a_HI, k_vechi, corrhi, ORDERHI);
+
+ /* bandwidth expansion */
+ tmp = gammaLo;
+ for (n = 1; n <= ORDERLO; n++) {
+ a_LO[n] *= tmp;
+ tmp *= gammaLo;
+ }
+
+ /* residual energy */
+ res_nrg = 0.0;
+ for (j = 0; j <= ORDERLO; j++) {
+ for (n = 0; n <= j; n++) {
+ res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n];
+ }
+ for (n = j+1; n <= ORDERLO; n++) {
+ res_nrg += a_LO[j] * corrlo2[n-j] * a_LO[n];
+ }
+ }
+
+ /* add hearing threshold and compute the gain */
+ *lo_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H);
+
+ /* copy coefficients to output array */
+ for (n = 1; n <= ORDERLO; n++) {
+ *lo_coeff++ = a_LO[n];
+ }
+
+
+ /* bandwidth expansion */
+ tmp = gammaHi;
+ for (n = 1; n <= ORDERHI; n++) {
+ a_HI[n] *= tmp;
+ tmp *= gammaHi;
+ }
+
+ /* residual energy */
+ res_nrg = 0.0;
+ for (j = 0; j <= ORDERHI; j++) {
+ for (n = 0; n <= j; n++) {
+ res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n];
+ }
+ for (n = j+1; n <= ORDERHI; n++) {
+ res_nrg += a_HI[j] * corrhi[n-j] * a_HI[n];
+ }
+ }
+
+ /* add hearing threshold and compute of the gain */
+ *hi_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H);
+
+ /* copy coefficients to output array */
+ for (n = 1; n <= ORDERHI; n++) {
+ *hi_coeff++ = a_HI[n];
+ }
+ }
+}
+
+
+
+/******************************************************************************
+ * WebRtcIsac_GetLpcCoefUb()
+ *
+ * Compute LP coefficients and correlation coefficients. At 12 kHz LP
+ * coefficients of the first and the last sub-frame is computed. At 16 kHz
+ * LP coefficients of 4th, 8th and 12th sub-frames are computed. We always
+ * compute correlation coefficients of all sub-frames.
+ *
+ * Inputs:
+ * -inSignal : Input signal
+ * -maskdata : a structure keeping signal from previous frame.
+ * -bandwidth : specifies if the codec is in 0-16 kHz mode or
+ * 0-12 kHz mode.
+ *
+ * Outputs:
+ * -lpCoeff : pointer to a buffer where A-polynomials are
+ * written to (first coeff is 1 and it is not
+ * written)
+ * -corrMat : a matrix where correlation coefficients of each
+ * sub-frame are written to one row.
+ * -varscale : a scale used to compute LPC gains.
+ */
+void
+WebRtcIsac_GetLpcCoefUb(
+ double* inSignal,
+ MaskFiltstr* maskdata,
+ double* lpCoeff,
+ double corrMat[][UB_LPC_ORDER + 1],
+ double* varscale,
+ WebRtc_Word16 bandwidth)
+{
+ int frameCntr, activeFrameCntr, n, pos1, pos2;
+ WebRtc_Word16 criterion1;
+ WebRtc_Word16 criterion2;
+ WebRtc_Word16 numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz));
+ double data[WINLEN];
+ double corrSubFrame[UB_LPC_ORDER+2];
+ double reflecCoeff[UB_LPC_ORDER];
+
+ double aPolynom[UB_LPC_ORDER+1];
+ double tmp;
+
+ /* bandwdith expansion factors */
+ const double gamma = 0.9;
+
+ /* change quallevel depending on pitch gains and level fluctuations */
+ WebRtcIsac_GetVarsUB(inSignal, &(maskdata->OldEnergy), varscale);
+
+ /* replace data in buffer by new look-ahead data */
+ for(frameCntr = 0, activeFrameCntr = 0; frameCntr < numSubFrames;
+ frameCntr++)
+ {
+ if(frameCntr == SUBFRAMES)
+ {
+ // we are in 16 kHz
+ varscale++;
+ WebRtcIsac_GetVarsUB(&inSignal[FRAMESAMPLES_HALF],
+ &(maskdata->OldEnergy), varscale);
+ }
+ /* Update input buffer and multiply signal with window */
+ for(pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++)
+ {
+ maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 +
+ UPDATE/2];
+ data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
+ }
+ pos2 = frameCntr * UPDATE/2;
+ for(n = 0; n < UPDATE/2; n++, pos1++, pos2++)
+ {
+ maskdata->DataBufferLo[pos1] = inSignal[pos2];
+ data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
+ }
+
+ /* Get correlation coefficients */
+ /* computing autocorrelation */
+ WebRtcIsac_AutoCorr(corrSubFrame, data, WINLEN, UB_LPC_ORDER+1);
+ memcpy(corrMat[frameCntr], corrSubFrame,
+ (UB_LPC_ORDER+1)*sizeof(double));
+
+ criterion1 = ((frameCntr == 0) || (frameCntr == (SUBFRAMES - 1))) &&
+ (bandwidth == isac12kHz);
+ criterion2 = (((frameCntr+1) % 4) == 0) &&
+ (bandwidth == isac16kHz);
+ if(criterion1 || criterion2)
+ {
+ /* add noise */
+ corrSubFrame[0] += 1e-6;
+ /* compute prediction coefficients */
+ WebRtcIsac_LevDurb(aPolynom, reflecCoeff, corrSubFrame,
+ UB_LPC_ORDER);
+
+ /* bandwidth expansion */
+ tmp = gamma;
+ for (n = 1; n <= UB_LPC_ORDER; n++)
+ {
+ *lpCoeff++ = aPolynom[n] * tmp;
+ tmp *= gamma;
+ }
+ activeFrameCntr++;
+ }
+ }
+}
+
+
+
+/******************************************************************************
+ * WebRtcIsac_GetLpcGain()
+ *
+ * Compute the LPC gains for each sub-frame, given the LPC of each sub-frame
+ * and the corresponding correlation coefficients.
+ *
+ * Inputs:
+ * -signal_noise_ratio : the desired SNR in dB.
+ * -numVecs : number of sub-frames
+ * -corrMat : a matrix of correlation coefficients where
+ * each row is a set of correlation coefficients of
+ * one sub-frame.
+ * -varscale : a scale computed when WebRtcIsac_GetLpcCoefUb()
+ * is called.
+ *
+ * Outputs:
+ * -gain : pointer to a buffer where LP gains are written.
+ *
+ */
+void
+WebRtcIsac_GetLpcGain(
+ double signal_noise_ratio,
+ const double* filtCoeffVecs,
+ int numVecs,
+ double* gain,
+ double corrMat[][UB_LPC_ORDER + 1],
+ const double* varscale)
+{
+ WebRtc_Word16 j, n;
+ WebRtc_Word16 subFrameCntr;
+ double aPolynom[ORDERLO + 1];
+ double res_nrg;
+
+ const double HearThresOffset = -28.0;
+ const double H_T_H = pow(10.0, 0.05 * HearThresOffset);
+ /* divide by sqrt(12) = 3.46 */
+ const double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46;
+
+ aPolynom[0] = 1;
+ for(subFrameCntr = 0; subFrameCntr < numVecs; subFrameCntr++)
+ {
+ if(subFrameCntr == SUBFRAMES)
+ {
+ // we are in second half of a SWB frame. use new varscale
+ varscale++;
+ }
+ memcpy(&aPolynom[1], &filtCoeffVecs[(subFrameCntr * (UB_LPC_ORDER + 1)) +
+ 1], sizeof(double) * UB_LPC_ORDER);
+
+ /* residual energy */
+ res_nrg = 0.0;
+ for(j = 0; j <= UB_LPC_ORDER; j++)
+ {
+ for(n = 0; n <= j; n++)
+ {
+ res_nrg += aPolynom[j] * corrMat[subFrameCntr][j-n] *
+ aPolynom[n];
+ }
+ for(n = j+1; n <= UB_LPC_ORDER; n++)
+ {
+ res_nrg += aPolynom[j] * corrMat[subFrameCntr][n-j] *
+ aPolynom[n];
+ }
+ }
+
+ /* add hearing threshold and compute the gain */
+ gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h
new file mode 100644
index 0000000..4eafeac
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_analysis.h
+ *
+ * LPC functions
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
+
+#include "settings.h"
+#include "structs.h"
+
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order);
+
+void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
+ double *oldEnergy, double *varscale);
+
+void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
+ double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
+ double *lo_coeff, double *hi_coeff);
+
+
+void WebRtcIsac_GetLpcGain(
+ double signal_noise_ratio,
+ const double* filtCoeffVecs,
+ int numVecs,
+ double* gain,
+ double corrLo[][UB_LPC_ORDER + 1],
+ const double* varscale);
+
+void WebRtcIsac_GetLpcCoefUb(
+ double* inSignal,
+ MaskFiltstr* maskdata,
+ double* lpCoeff,
+ double corr[][UB_LPC_ORDER + 1],
+ double* varscale,
+ WebRtc_Word16 bandwidth);
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c
new file mode 100644
index 0000000..25c69cb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * SWB_KLT_Tables_LPCGain.c
+ *
+ * This file defines tables used for entropy coding of LPC Gain
+ * of upper-band.
+ *
+ */
+
+#include "lpc_gain_swb_tables.h"
+#include "settings.h"
+#include "typedefs.h"
+
+const double WebRtcIsac_kQSizeLpcGain = 0.100000;
+
+const double WebRtcIsac_kMeanLpcGain = -3.3822;
+
+/*
+* The smallest reconstruction points for quantiztion of
+* LPC gains.
+*/
+const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES] =
+{
+ -0.800000, -1.000000, -1.200000, -2.200000, -3.000000, -12.700000
+};
+
+/*
+* Number of reconstruction points of quantizers for LPC Gains.
+*/
+const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES] =
+{
+ 17, 20, 25, 45, 77, 170
+};
+/*
+* Starting index for entropy decoder to search for the right interval,
+* one entry per LAR coefficient
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES] =
+{
+ 8, 10, 12, 22, 38, 85
+};
+
+/*
+* The following 6 vectors define CDF of 6 decorrelated LPC
+* gains.
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18] =
+{
+ 0, 10, 27, 83, 234, 568, 1601, 4683, 16830, 57534, 63437,
+ 64767, 65229, 65408, 65483, 65514, 65527, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21] =
+{
+ 0, 15, 33, 84, 185, 385, 807, 1619, 3529, 7850, 19488,
+ 51365, 62437, 64548, 65088, 65304, 65409, 65484, 65507, 65522, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26] =
+{
+ 0, 15, 29, 54, 89, 145, 228, 380, 652, 1493, 4260,
+ 12359, 34133, 50749, 57224, 60814, 62927, 64078, 64742, 65103, 65311, 65418,
+ 65473, 65509, 65521, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46] =
+{
+ 0, 8, 12, 16, 26, 42, 56, 76, 111, 164, 247,
+ 366, 508, 693, 1000, 1442, 2155, 3188, 4854, 7387, 11249, 17617,
+ 30079, 46711, 56291, 60127, 62140, 63258, 63954, 64384, 64690, 64891, 65031,
+ 65139, 65227, 65293, 65351, 65399, 65438, 65467, 65492, 65504, 65510, 65518,
+ 65523, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78] =
+{
+ 0, 17, 29, 39, 51, 70, 104, 154, 234, 324, 443,
+ 590, 760, 971, 1202, 1494, 1845, 2274, 2797, 3366, 4088, 4905,
+ 5899, 7142, 8683, 10625, 12983, 16095, 20637, 28216, 38859, 47237, 51537,
+ 54150, 56066, 57583, 58756, 59685, 60458, 61103, 61659, 62144, 62550, 62886,
+ 63186, 63480, 63743, 63954, 64148, 64320, 64467, 64600, 64719, 64837, 64939,
+ 65014, 65098, 65160, 65211, 65250, 65290, 65325, 65344, 65366, 65391, 65410,
+ 65430, 65447, 65460, 65474, 65487, 65494, 65501, 65509, 65513, 65518, 65520,
+ 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171] =
+{
+ 0, 10, 12, 14, 16, 18, 23, 29, 35, 42, 51,
+ 58, 65, 72, 78, 87, 96, 103, 111, 122, 134, 150,
+ 167, 184, 202, 223, 244, 265, 289, 315, 346, 379, 414,
+ 450, 491, 532, 572, 613, 656, 700, 751, 802, 853, 905,
+ 957, 1021, 1098, 1174, 1250, 1331, 1413, 1490, 1565, 1647, 1730,
+ 1821, 1913, 2004, 2100, 2207, 2314, 2420, 2532, 2652, 2783, 2921,
+ 3056, 3189, 3327, 3468, 3640, 3817, 3993, 4171, 4362, 4554, 4751,
+ 4948, 5142, 5346, 5566, 5799, 6044, 6301, 6565, 6852, 7150, 7470,
+ 7797, 8143, 8492, 8835, 9181, 9547, 9919, 10315, 10718, 11136, 11566,
+ 12015, 12482, 12967, 13458, 13953, 14432, 14903, 15416, 15936, 16452, 16967,
+ 17492, 18024, 18600, 19173, 19736, 20311, 20911, 21490, 22041, 22597, 23157,
+ 23768, 24405, 25034, 25660, 26280, 26899, 27614, 28331, 29015, 29702, 30403,
+ 31107, 31817, 32566, 33381, 34224, 35099, 36112, 37222, 38375, 39549, 40801,
+ 42074, 43350, 44626, 45982, 47354, 48860, 50361, 51845, 53312, 54739, 56026,
+ 57116, 58104, 58996, 59842, 60658, 61488, 62324, 63057, 63769, 64285, 64779,
+ 65076, 65344, 65430, 65500, 65517, 65535
+};
+
+/*
+* An array of pointers to CDFs of decorrelated LPC Gains
+*/
+const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES] =
+{
+ WebRtcIsac_kLpcGainCdfVec0, WebRtcIsac_kLpcGainCdfVec1,
+ WebRtcIsac_kLpcGainCdfVec2, WebRtcIsac_kLpcGainCdfVec3,
+ WebRtcIsac_kLpcGainCdfVec4, WebRtcIsac_kLpcGainCdfVec5
+};
+
+/*
+* A matrix to decorrellate LPC gains of subframes.
+*/
+const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES] =
+{
+ {-0.150860, 0.327872, 0.367220, 0.504613, 0.559270, 0.409234},
+ { 0.457128, -0.613591, -0.289283, -0.029734, 0.393760, 0.418240},
+ {-0.626043, 0.136489, -0.439118, -0.448323, 0.135987, 0.420869},
+ { 0.526617, 0.480187, 0.242552, -0.488754, -0.158713, 0.411331},
+ {-0.302587, -0.494953, 0.588112, -0.063035, -0.404290, 0.387510},
+ { 0.086378, 0.147714, -0.428875, 0.548300, -0.570121, 0.401391}
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h
new file mode 100644
index 0000000..1eba97c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * SWB_KLT_Tables_LPCGain.h
+ *
+ * This file declares tables used for entropy coding of LPC Gain
+ * of upper-band.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
+
+#include "settings.h"
+#include "typedefs.h"
+
+extern const double WebRtcIsac_kQSizeLpcGain;
+
+extern const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES];
+
+extern const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171];
+
+extern const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
+
+extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c
new file mode 100644
index 0000000..695d583
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c
@@ -0,0 +1,159 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * SWB_KLT_Tables.c
+ *
+ * This file defines tables used for entropy coding of LPC shape of
+ * upper-band signal if the bandwidth is 12 kHz.
+ *
+ */
+
+#include "lpc_shape_swb12_tables.h"
+#include "settings.h"
+#include "typedefs.h"
+
+/*
+* Mean value of LAR
+*/
+const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER] =
+{
+ 0.03748928306641, 0.09453441192543, -0.01112522344398, 0.03800237516842
+};
+
+/*
+* A rotation matrix to decorrelate intra-vector correlation,
+* i.e. correlation among components of LAR vector.
+*/
+const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER] =
+{
+ {-0.00075365493856, -0.05809964887743, -0.23397966154116, 0.97050367376411},
+ { 0.00625021257734, -0.17299965610679, 0.95977735920651, 0.22104179375008},
+ { 0.20543384258374, -0.96202143495696, -0.15301870801552, -0.09432375099565},
+ {-0.97865075648479, -0.20300322280841, -0.02581111653779, -0.01913568980258}
+};
+
+/*
+* A rotation matrix to remove correlation among LAR coefficients
+* of different LAR vectors. One might guess that decorrelation matrix
+* for the first component should differ from the second component
+* but we haven't observed a significant benefit of having different
+* decorrelation matrices for different components.
+*/
+const double WebRtcIsac_kInterVecDecorrMatUb12
+[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME] =
+{
+ { 0.70650597970460, -0.70770707262373},
+ {-0.70770707262373, -0.70650597970460}
+};
+
+/*
+* LAR quantization step-size.
+*/
+const double WebRtcIsac_kLpcShapeQStepSizeUb12 = 0.150000;
+
+/*
+* The smallest reconstruction points for quantiztion of LAR coefficients.
+*/
+const double WebRtcIsac_kLpcShapeLeftRecPointUb12
+[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME] =
+{
+ -0.900000, -1.050000, -1.350000, -1.800000, -1.350000, -1.650000,
+ -2.250000, -3.450000
+};
+
+/*
+* Number of reconstruction points of quantizers for LAR coefficients.
+*/
+const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
+{
+ 13, 15, 19, 27, 19, 24, 32, 48
+};
+
+/*
+* Starting index for entropy decoder to search for the right interval,
+* one entry per LAR coefficient
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
+{
+ 6, 7, 9, 13, 9, 12, 16, 24
+};
+
+/*
+* The following 8 vectors define CDF of 8 decorrelated LAR
+* coefficients.
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14] =
+{
+ 0, 13, 95, 418, 1687, 6498, 21317, 44200, 59029, 63849, 65147,
+ 65449, 65525, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16] =
+{
+ 0, 10, 59, 255, 858, 2667, 8200, 22609, 42988, 57202, 62947,
+ 64743, 65308, 65476, 65522, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20] =
+{
+ 0, 18, 40, 118, 332, 857, 2017, 4822, 11321, 24330, 41279,
+ 54342, 60637, 63394, 64659, 65184, 65398, 65482, 65518, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28] =
+{
+ 0, 21, 38, 90, 196, 398, 770, 1400, 2589, 4650, 8211,
+ 14933, 26044, 39592, 50814, 57452, 60971, 62884, 63995, 64621, 65019, 65273,
+ 65410, 65480, 65514, 65522, 65531, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20] =
+{
+ 0, 7, 46, 141, 403, 969, 2132, 4649, 10633, 24902, 43254,
+ 54665, 59928, 62674, 64173, 64938, 65293, 65464, 65523, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25] =
+{
+ 0, 7, 22, 72, 174, 411, 854, 1737, 3545, 6774, 13165,
+ 25221, 40980, 52821, 58714, 61706, 63472, 64437, 64989, 65287, 65430, 65503,
+ 65525, 65529, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33] =
+{
+ 0, 11, 21, 36, 65, 128, 228, 401, 707, 1241, 2126,
+ 3589, 6060, 10517, 18853, 31114, 42477, 49770, 54271, 57467, 59838, 61569,
+ 62831, 63772, 64433, 64833, 65123, 65306, 65419, 65466, 65499, 65519, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49] =
+{
+ 0, 14, 34, 67, 107, 167, 245, 326, 449, 645, 861,
+ 1155, 1508, 2003, 2669, 3544, 4592, 5961, 7583, 9887, 13256, 18765,
+ 26519, 34077, 40034, 44349, 47795, 50663, 53262, 55473, 57458, 59122, 60592,
+ 61742, 62690, 63391, 63997, 64463, 64794, 65045, 65207, 65309, 65394, 65443,
+ 65478, 65504, 65514, 65523, 65535
+};
+
+/*
+* An array of pointers to CDFs of decorrelated LARs
+*/
+const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
+{
+ WebRtcIsac_kLpcShapeCdfVec0Ub12, WebRtcIsac_kLpcShapeCdfVec1Ub12,
+ WebRtcIsac_kLpcShapeCdfVec2Ub12, WebRtcIsac_kLpcShapeCdfVec3Ub12,
+ WebRtcIsac_kLpcShapeCdfVec4Ub12, WebRtcIsac_kLpcShapeCdfVec5Ub12,
+ WebRtcIsac_kLpcShapeCdfVec6Ub12, WebRtcIsac_kLpcShapeCdfVec7Ub12
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h
new file mode 100644
index 0000000..1e93847
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_shape_swb12_tables.h
+ *
+ * This file declares tables used for entropy coding of LPC shape of
+ * upper-band signal if the bandwidth is 12 kHz.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
+
+#include "settings.h"
+#include "typedefs.h"
+
+extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER];
+
+extern const double WebRtcIsac_kMeanLpcGain;
+
+extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER];
+
+extern const double WebRtcIsac_kInterVecDecorrMatUb12
+[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME];
+
+extern const double WebRtcIsac_kLpcShapeQStepSizeUb12;
+
+extern const double WebRtcIsac_kLpcShapeLeftRecPointUb12
+[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME];
+
+
+extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
+
+extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
+[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c
new file mode 100644
index 0000000..89f4523
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c
@@ -0,0 +1,248 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * SWB16_KLT_Tables.c
+ *
+ * This file defines tables used for entropy coding of LPC shape of
+ * upper-band signal if the bandwidth is 16 kHz.
+ *
+ */
+
+#include "lpc_shape_swb16_tables.h"
+#include "settings.h"
+#include "typedefs.h"
+
+/*
+* Mean value of LAR
+*/
+const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER] =
+{
+0.454978, 0.364747, 0.102999, 0.104523
+};
+
+/*
+* A rotation matrix to decorrelate intra-vector correlation,
+* i.e. correlation among components of LAR vector.
+*/
+const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER] =
+{
+ {-0.020528, -0.085858, -0.002431, 0.996093},
+ {-0.033155, 0.036102, 0.998786, 0.004866},
+ { 0.202627, 0.974853, -0.028940, 0.088132},
+ {-0.978479, 0.202454, -0.039785, -0.002811}
+};
+
+/*
+* A rotation matrix to remove correlation among LAR coefficients
+* of different LAR vectors. One might guess that decorrelation matrix
+* for the first component should differ from the second component
+* but we haven't observed a significant benefit of having different
+* decorrelation matrices for different components.
+*/
+const double WebRtcIsac_kInterVecDecorrMatUb16
+[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME] =
+{
+ { 0.291675, -0.515786, 0.644927, 0.482658},
+ {-0.647220, 0.479712, 0.289556, 0.516856},
+ { 0.643084, 0.485489, -0.289307, 0.516763},
+ {-0.287185, -0.517823, -0.645389, 0.482553}
+};
+
+/*
+* The following 16 vectors define CDF of 16 decorrelated LAR
+* coefficients.
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14] =
+{
+ 0, 2, 20, 159, 1034, 5688, 20892, 44653,
+ 59849, 64485, 65383, 65518, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16] =
+{
+ 0, 1, 7, 43, 276, 1496, 6681, 21653,
+ 43891, 58859, 64022, 65248, 65489, 65529, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18] =
+{
+ 0, 1, 9, 54, 238, 933, 3192, 9461,
+ 23226, 42146, 56138, 62413, 64623, 65300, 65473, 65521,
+ 65533, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30] =
+{
+ 0, 2, 4, 8, 17, 36, 75, 155,
+ 329, 683, 1376, 2662, 5047, 9508, 17526, 29027,
+ 40363, 48997, 55096, 59180, 61789, 63407, 64400, 64967,
+ 65273, 65429, 65497, 65526, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16] =
+{
+ 0, 1, 10, 63, 361, 1785, 7407, 22242,
+ 43337, 58125, 63729, 65181, 65472, 65527, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17] =
+{
+ 0, 1, 7, 29, 134, 599, 2443, 8590,
+ 22962, 42635, 56911, 63060, 64940, 65408, 65513, 65531,
+ 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21] =
+{
+ 0, 1, 5, 16, 57, 191, 611, 1808,
+ 4847, 11755, 24612, 40910, 53789, 60698, 63729, 64924,
+ 65346, 65486, 65523, 65532, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36] =
+{
+ 0, 1, 4, 12, 25, 55, 104, 184,
+ 314, 539, 926, 1550, 2479, 3861, 5892, 8845,
+ 13281, 20018, 29019, 38029, 45581, 51557, 56057, 59284,
+ 61517, 63047, 64030, 64648, 65031, 65261, 65402, 65480,
+ 65518, 65530, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21] =
+{
+ 0, 1, 2, 7, 26, 103, 351, 1149,
+ 3583, 10204, 23846, 41711, 55361, 61917, 64382, 65186,
+ 65433, 65506, 65528, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21] =
+{
+ 0, 6, 19, 63, 205, 638, 1799, 4784,
+ 11721, 24494, 40803, 53805, 60886, 63822, 64931, 65333,
+ 65472, 65517, 65530, 65533, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28] =
+{
+ 0, 1, 3, 11, 31, 86, 221, 506,
+ 1101, 2296, 4486, 8477, 15356, 26079, 38941, 49952,
+ 57165, 61257, 63426, 64549, 65097, 65351, 65463, 65510,
+ 65526, 65532, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55] =
+{
+ 0, 3, 12, 23, 42, 65, 89, 115,
+ 150, 195, 248, 327, 430, 580, 784, 1099,
+ 1586, 2358, 3651, 5899, 9568, 14312, 19158, 23776,
+ 28267, 32663, 36991, 41153, 45098, 48680, 51870, 54729,
+ 57141, 59158, 60772, 62029, 63000, 63761, 64322, 64728,
+ 65000, 65192, 65321, 65411, 65463, 65496, 65514, 65523,
+ 65527, 65529, 65531, 65532, 65533, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26] =
+{
+ 0, 2, 4, 10, 21, 48, 114, 280,
+ 701, 1765, 4555, 11270, 24267, 41213, 54285, 61003,
+ 63767, 64840, 65254, 65421, 65489, 65514, 65526, 65532,
+ 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28] =
+{
+ 0, 1, 3, 6, 15, 36, 82, 196,
+ 453, 1087, 2557, 5923, 13016, 25366, 40449, 52582,
+ 59539, 62896, 64389, 65033, 65316, 65442, 65494, 65519,
+ 65529, 65533, 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34] =
+{
+ 0, 2, 4, 8, 18, 35, 73, 146,
+ 279, 524, 980, 1789, 3235, 5784, 10040, 16998,
+ 27070, 38543, 48499, 55421, 59712, 62257, 63748, 64591,
+ 65041, 65278, 65410, 65474, 65508, 65522, 65530, 65533,
+ 65534, 65535
+};
+
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71] =
+{
+ 0, 1, 2, 6, 13, 26, 55, 92,
+ 141, 191, 242, 296, 355, 429, 522, 636,
+ 777, 947, 1162, 1428, 1753, 2137, 2605, 3140,
+ 3743, 4409, 5164, 6016, 6982, 8118, 9451, 10993,
+ 12754, 14810, 17130, 19780, 22864, 26424, 30547, 35222,
+ 40140, 44716, 48698, 52056, 54850, 57162, 59068, 60643,
+ 61877, 62827, 63561, 64113, 64519, 64807, 65019, 65167,
+ 65272, 65343, 65399, 65440, 65471, 65487, 65500, 65509,
+ 65518, 65524, 65527, 65531, 65533, 65534, 65535
+};
+
+/*
+* An array of pointers to CDFs of decorrelated LARs
+*/
+const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = {
+ WebRtcIsac_kLpcShapeCdfVec01Ub16,
+ WebRtcIsac_kLpcShapeCdfVec1Ub16,
+ WebRtcIsac_kLpcShapeCdfVec2Ub16,
+ WebRtcIsac_kLpcShapeCdfVec3Ub16,
+ WebRtcIsac_kLpcShapeCdfVec4Ub16,
+ WebRtcIsac_kLpcShapeCdfVec5Ub16,
+ WebRtcIsac_kLpcShapeCdfVec6Ub16,
+ WebRtcIsac_kLpcShapeCdfVec7Ub16,
+ WebRtcIsac_kLpcShapeCdfVec8Ub16,
+ WebRtcIsac_kLpcShapeCdfVec01Ub160,
+ WebRtcIsac_kLpcShapeCdfVec01Ub161,
+ WebRtcIsac_kLpcShapeCdfVec01Ub162,
+ WebRtcIsac_kLpcShapeCdfVec01Ub163,
+ WebRtcIsac_kLpcShapeCdfVec01Ub164,
+ WebRtcIsac_kLpcShapeCdfVec01Ub165,
+ WebRtcIsac_kLpcShapeCdfVec01Ub166
+};
+
+/*
+* The smallest reconstruction points for quantiztion of LAR coefficients.
+*/
+const double WebRtcIsac_kLpcShapeLeftRecPointUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
+{
+ -0.8250, -0.9750, -1.1250, -2.1750, -0.9750, -1.1250, -1.4250,
+ -2.6250, -1.4250, -1.2750, -1.8750, -3.6750, -1.7250, -1.8750,
+ -2.3250, -5.4750
+};
+
+/*
+* Number of reconstruction points of quantizers for LAR coefficients.
+*/
+const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
+{
+ 13, 15, 17, 29, 15, 16, 20, 35, 20,
+ 20, 27, 54, 25, 27, 33, 70
+};
+
+/*
+* Starting index for entropy decoder to search for the right interval,
+* one entry per LAR coefficient
+*/
+const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
+{
+ 6, 7, 8, 14, 7, 8, 10, 17, 10,
+ 10, 13, 27, 12, 13, 16, 35
+};
+
+/*
+* LAR quantization step-size.
+*/
+const double WebRtcIsac_kLpcShapeQStepSizeUb16 = 0.150000;
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h
new file mode 100644
index 0000000..68d08b2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_shape_swb16_tables.h
+ *
+ * This file declares tables used for entropy coding of LPC shape of
+ * upper-band signal if the bandwidth is 16 kHz.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
+
+#include "settings.h"
+#include "typedefs.h"
+
+
+extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER];
+
+extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER];
+
+extern const double WebRtcIsac_kInterVecDecorrMatUb16
+[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
+
+extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+
+extern const double WebRtcIsac_kLpcShapeLeftRecPointUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+
+extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+
+extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
+[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+
+extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c
new file mode 100644
index 0000000..7df6121
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c
@@ -0,0 +1,1129 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* coding tables for the KLT coefficients */
+
+#include "lpc_tables.h"
+#include "settings.h"
+
+/* indices of KLT coefficients used */
+const WebRtc_UWord16 WebRtcIsac_kQKltSelIndGain[12] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 10, 11};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltSelIndShape[108] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 10, 11, 12, 13, 14, 15, 16, 17, 18, 19,
+ 20, 21, 22, 23, 24, 25, 26, 27, 28, 29,
+ 30, 31, 32, 33, 34, 35, 36, 37, 38, 39,
+ 40, 41, 42, 43, 44, 45, 46, 47, 48, 49,
+ 50, 51, 52, 53, 54, 55, 56, 57, 58, 59,
+ 60, 61, 62, 63, 64, 65, 66, 67, 68, 69,
+ 70, 71, 72, 73, 74, 75, 76, 77, 78, 79,
+ 80, 81, 82, 83, 84, 85, 86, 87, 88, 89,
+ 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,
+ 100, 101, 102, 103, 104, 105, 106, 107};
+
+/* cdf array for model indicator */
+const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[4] = {
+ 0, 15434, 37548, 65535};
+
+/* pointer to cdf array for model indicator */
+const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1] = {WebRtcIsac_kQKltModelCdf};
+
+/* initial cdf index for decoder of model indicator */
+const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1] = {1};
+
+/* offset to go from rounded value to quantization index */
+const short WebRtcIsac_kQKltQuantMinGain[12] = {
+ 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47};
+
+
+const short WebRtcIsac_kQKltQuantMinShape[108] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 1, 1, 1, 1, 2, 2, 2, 3, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 2, 2, 3, 0, 0, 0, 0,
+ 1, 0, 1, 1, 1, 1, 1, 1, 1, 2,
+ 2, 4, 3, 5, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 2, 1, 2, 2, 3, 4,
+ 4, 7, 0, 0, 1, 1, 1, 1, 1, 1,
+ 1, 2, 3, 2, 3, 4, 4, 5, 7, 13,
+ 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
+ 5, 6, 7, 11, 9, 13, 12, 26};
+
+/* maximum quantization index */
+const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12] = {
+ 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 2, 2, 2, 2, 4, 4, 5, 6, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 2, 2,
+ 2, 2, 3, 4, 5, 7, 0, 0, 0, 0,
+ 2, 0, 2, 2, 2, 2, 3, 2, 2, 4,
+ 4, 6, 6, 9, 0, 0, 0, 0, 2, 2,
+ 2, 2, 2, 2, 3, 2, 4, 4, 7, 7,
+ 9, 13, 0, 0, 2, 2, 2, 2, 2, 2,
+ 3, 4, 5, 4, 6, 8, 8, 10, 16, 25,
+ 0, 2, 2, 4, 5, 4, 4, 4, 7, 8,
+ 9, 10, 13, 19, 17, 23, 25, 49};
+
+/* index offset */
+const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[3][12] = {
+{ 0, 7, 20, 29, 44, 55, 75, 88, 120, 143,
+ 200, 253},
+{ 0, 7, 19, 27, 42, 53, 73, 86, 117, 140,
+ 197, 249},
+{ 0, 7, 20, 28, 44, 55, 75, 89, 121, 145,
+ 202, 257}};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[3][108] = {
+{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
+ 11, 14, 17, 20, 23, 28, 33, 39, 46, 47,
+ 48, 49, 50, 52, 53, 54, 55, 56, 58, 61,
+ 64, 67, 70, 74, 79, 85, 93, 94, 95, 96,
+ 97, 100, 101, 104, 107, 110, 113, 117, 120, 123,
+ 128, 133, 140, 147, 157, 158, 159, 160, 161, 164,
+ 167, 170, 173, 176, 179, 183, 186, 191, 196, 204,
+ 212, 222, 236, 237, 238, 241, 244, 247, 250, 253,
+ 256, 260, 265, 271, 276, 283, 292, 301, 312, 329,
+ 355, 356, 359, 362, 367, 373, 378, 383, 388, 396,
+ 405, 415, 426, 440, 460, 478, 502, 528},
+{ 0, 1, 2, 3, 4, 6, 7, 8, 9, 11,
+ 13, 16, 19, 22, 26, 29, 34, 39, 45, 46,
+ 47, 48, 49, 50, 51, 52, 53, 55, 57, 60,
+ 63, 66, 70, 73, 78, 84, 91, 92, 93, 94,
+ 95, 96, 97, 99, 102, 105, 108, 111, 114, 118,
+ 123, 128, 134, 141, 151, 152, 153, 154, 156, 159,
+ 162, 165, 168, 171, 174, 177, 181, 186, 194, 200,
+ 208, 218, 233, 234, 235, 236, 239, 242, 245, 248,
+ 251, 254, 258, 263, 270, 277, 288, 297, 308, 324,
+ 349, 351, 354, 357, 361, 366, 372, 378, 383, 390,
+ 398, 407, 420, 431, 450, 472, 496, 524},
+{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 11,
+ 14, 17, 20, 23, 26, 29, 34, 40, 47, 48,
+ 49, 50, 51, 52, 53, 54, 55, 58, 61, 64,
+ 67, 70, 73, 77, 82, 88, 96, 97, 98, 99,
+ 101, 102, 104, 107, 110, 113, 116, 119, 122, 125,
+ 129, 134, 141, 150, 160, 161, 162, 163, 166, 168,
+ 171, 174, 177, 180, 183, 186, 190, 195, 201, 208,
+ 216, 226, 243, 244, 245, 248, 251, 254, 257, 260,
+ 263, 268, 273, 278, 284, 291, 299, 310, 323, 340,
+ 366, 368, 371, 374, 379, 383, 389, 394, 399, 406,
+ 414, 422, 433, 445, 461, 480, 505, 533}};
+
+/* initial cdf index for KLT coefficients */
+const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[3][12] = {
+{ 3, 6, 4, 7, 5, 10, 6, 16, 11, 28,
+ 26, 69},
+{ 3, 6, 4, 7, 5, 10, 6, 15, 11, 28,
+ 26, 69},
+{ 3, 6, 4, 8, 5, 10, 7, 16, 12, 28,
+ 27, 70}};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[3][108] = {
+{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
+ 1, 1, 1, 1, 2, 2, 3, 3, 0, 0,
+ 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
+ 1, 1, 2, 2, 3, 4, 0, 0, 0, 0,
+ 1, 0, 1, 1, 1, 1, 2, 1, 1, 2,
+ 2, 3, 3, 5, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 2, 1, 2, 2, 4, 4,
+ 5, 7, 0, 0, 1, 1, 1, 1, 1, 1,
+ 2, 2, 3, 2, 3, 4, 4, 5, 8, 13,
+ 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
+ 5, 5, 7, 10, 9, 12, 13, 25},
+{ 0, 0, 0, 0, 1, 0, 0, 0, 1, 1,
+ 1, 1, 1, 2, 1, 2, 2, 3, 0, 0,
+ 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
+ 1, 2, 1, 2, 3, 3, 0, 0, 0, 0,
+ 0, 0, 1, 1, 1, 1, 1, 1, 2, 2,
+ 2, 3, 3, 5, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 1, 1, 2, 2, 4, 3, 4,
+ 5, 7, 0, 0, 0, 1, 1, 1, 1, 1,
+ 1, 2, 2, 3, 3, 5, 4, 5, 8, 12,
+ 1, 1, 1, 2, 2, 3, 3, 2, 3, 4,
+ 4, 6, 5, 9, 11, 12, 14, 25},
+{ 0, 0, 0, 0, 0, 0, 0, 0, 1, 1,
+ 1, 1, 1, 1, 1, 2, 3, 3, 0, 0,
+ 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
+ 1, 1, 2, 2, 3, 4, 0, 0, 0, 1,
+ 0, 1, 1, 1, 1, 1, 1, 1, 1, 2,
+ 2, 3, 4, 5, 0, 0, 0, 1, 1, 1,
+ 1, 1, 1, 1, 1, 2, 2, 3, 3, 4,
+ 5, 8, 0, 0, 1, 1, 1, 1, 1, 1,
+ 2, 2, 2, 3, 3, 4, 5, 6, 8, 13,
+ 1, 1, 1, 2, 2, 3, 2, 2, 3, 4,
+ 4, 5, 6, 8, 9, 12, 14, 25}};
+
+/* offsets for quantizer representation levels*/
+const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsGain[3] = {
+ 0, 392, 779};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsShape[3] = {
+ 0, 578, 1152};
+
+/* quantizer representation levels */
+const double WebRtcIsac_kQKltLevelsGain[1176] = {
+-2.78127126, -1.76745590, -0.77913790, -0.00437329, 0.79961206, 1.81775776, 2.81389782, -5.78753143, -4.88384084, -3.89320940,
+-2.88133610, -1.92859977, -0.86347396, 0.02003888, 0.86140400, 1.89667156, 2.97134967, 3.98781964, 4.91727277, 5.82865898,
+-4.11195874, -2.80898424, -1.87547977, -0.80943825, -0.00679084, 0.79573851, 1.83953397, 2.67586037, 3.76274082, -6.10933968,
+-4.93034581, -3.89281296, -2.91530625, -1.89684163, -0.85319130, -0.02275767, 0.86862017, 1.91578276, 2.96107339, 3.96543056,
+ 4.91369908, 5.91058154, 6.83848343, 8.07136925, -5.87470395, -4.84703049, -3.84284597, -2.86168446, -1.89290192, -0.82798145,
+-0.00080013, 0.82594974, 1.85754329, 2.88351798, 3.96172628, -8.85684885, -7.87387461, -6.97811862, -5.93256270, -4.94301439,
+-3.95513701, -2.96041544, -1.94031192, -0.87961478, -0.00456201, 0.89911505, 1.91723376, 2.94011511, 3.93302540, 4.97990967,
+ 5.93133404, 7.02181199, 7.92407762, 8.80155440, 10.04665814, -4.82396678, -3.85612158, -2.89482244, -1.89558408, -0.90036978,
+-0.00677823, 0.90607989, 1.90937981, 2.91175777, 3.91637730, 4.97565723, 5.84771228, 7.11145863, -16.07879840, -15.03776309,
+-13.93905670, -12.95671800, -11.89171202, -10.95820934, -9.95923714, -8.94357334, -7.99068299, -6.97481009, -5.94826231, -4.96673988,
+-3.97490466, -2.97846970, -1.95130435, -0.94215262, -0.01444043, 0.96770704, 1.95848598, 2.94107862, 3.95666119, 4.97253085,
+ 5.97191122, 6.93277360, 7.96608727, 8.87958779, 10.00264269, 10.86560820, 12.07449071, 13.04491775, 13.97507061, 14.91845261,
+-10.85696295, -9.83365357, -9.01245635, -7.95915145, -6.95625003, -5.95362618, -4.93468444, -3.98760978, -2.95044407, -1.97041277,
+-0.97701799, -0.00840234, 0.97834289, 1.98361415, 2.97802439, 3.96415871, 4.95369042, 5.94101770, 6.92756798, 7.94063998,
+ 8.85951828, 9.97077022, 11.00068503, -33.92030406, -32.81426422, -32.00000000, -31.13243639, -30.11886909, -29.06017570, -28.12598824,
+-27.22045482, -25.81215858, -25.07849962, -23.93018013, -23.02097643, -21.89529725, -20.99091085, -19.98889048, -18.94327044, -17.96562071,
+-16.96126218, -15.95054062, -14.98516200, -13.97101012, -13.02106500, -11.98438006, -11.03216748, -9.95930286, -8.97043946, -7.98085082,
+-6.98360995, -5.98998802, -4.98668173, -4.00032906, -3.00420619, -1.98701132, -0.99324682, -0.00609324, 0.98297834, 1.99483076,
+ 3.00305044, 3.97142097, 4.97525759, 5.98612258, 6.97448236, 7.97575900, 9.01086211, 9.98665542, 11.00541438, 11.98078628,
+ 12.92352471, 14.06849675, 14.99949430, 15.94904834, 16.97440321, 18.04040916, 18.88987609, 20.05312391, 21.00000000, 21.79443341,
+-31.98578825, -31.00000000, -29.89060567, -28.98555686, -27.97114102, -26.84935410, -26.02402230, -24.94195278, -23.92336849, -22.95552382,
+-21.97932836, -20.96055470, -19.99649553, -19.03436122, -17.96706525, -17.01139515, -16.01363516, -14.99154248, -14.00298333, -12.99630613,
+-11.99955519, -10.99000421, -10.00819092, -8.99763648, -7.98431793, -7.01769025, -5.99604690, -4.99980697, -3.99334671, -3.01748192,
+-2.02051217, -1.00848371, -0.01942358, 1.00477757, 1.95477872, 2.98593031, 3.98779079, 4.96862849, 6.02694771, 6.93983733,
+ 7.89874717, 8.99615862, 10.02367921, 10.96293452, 11.84351528, 12.92207187, 13.85122329, 15.05146877, 15.99371264, 17.00000000,
+ 18.00000000, 19.00000000, 19.82763573, -47.00000000, -46.00000000, -44.87138498, -44.00000000, -43.00000000, -42.00000000, -41.00000000,
+-39.88966612, -38.98913239, -37.80306486, -37.23584325, -35.94200288, -34.99881301, -34.11361858, -33.06507360, -32.13129135, -30.90891364,
+-29.81511907, -28.99250380, -28.04535391, -26.99767800, -26.04418164, -24.95687851, -24.04865595, -23.03392645, -21.89366707, -20.93517364,
+-19.99388660, -18.91620943, -18.03749683, -16.99532379, -15.98683813, -15.06421479, -13.99359211, -12.99714098, -11.97022520, -10.98500279,
+-9.98834422, -8.95729330, -8.01232284, -7.00253661, -5.99681626, -5.01207817, -3.95914904, -3.01232178, -1.96615919, -0.97687670,
+ 0.01228030, 0.98412288, 2.01753544, 3.00580570, 3.97783510, 4.98846894, 6.01321400, 7.00867732, 8.00416375, 9.01771966,
+ 9.98637729, 10.98255180, 11.99194163, 13.01807333, 14.00999545, 15.00118556, 16.00089224, 17.00584148, 17.98251763, 18.99942091,
+ 19.96917690, 20.97839265, 21.98207297, 23.00171271, 23.99930737, 24.99746061, 26.00936304, 26.98240132, 28.01126868, 29.01395915,
+ 29.98153507, 31.01376711, 31.99876818, 33.00475317, 33.99753994, 34.99493913, 35.98933585, 36.95620160, 37.98428461, 38.99317544,
+ 40.01832073, 40.98048133, 41.95999283, 42.98232091, 43.96523612, 44.99574268, 45.99524194, 47.05464025, 48.03821548, 48.99354366,
+ 49.96400411, 50.98017973, 51.95184408, 52.96291806, 54.00194392, 54.96603783, 55.95623778, 57.03076595, 58.05889901, 58.99081551,
+ 59.97928121, 61.05071612, 62.03971580, 63.01286038, 64.01290338, 65.02074503, 65.99454594, 67.00399425, 67.96571257, 68.95305727,
+ 69.92030664, 70.95594862, 71.98088567, 73.04764124, 74.00285480, 75.02696330, 75.89837673, 76.93459997, 78.16266309, 78.83317543,
+ 80.00000000, 80.87251574, 82.09803524, 83.10671664, 84.00000000, 84.77023523, 86.00000000, 87.00000000, 87.92946897, 88.69159118,
+ 90.00000000, 90.90535270, -3.00000000, -2.00000000, -0.77592424, -0.00564307, 0.76727305, 2.00000000, 3.00000000, -6.00000000,
+-5.00000000, -4.00000000, -2.92897924, -1.85623684, -0.72445303, -0.00119184, 0.72896652, 2.05710416, 3.17909894, 4.00000000,
+ 5.00000000, -3.00000000, -2.00000000, -0.67480586, -0.00028016, 0.66618169, 2.00000000, 3.00000000, 4.00000000, -7.00000000,
+-6.00000000, -5.00000000, -3.78336783, -2.84811556, -2.04088844, -0.71114371, 0.03142493, 0.69662772, 1.91417930, 3.00000000,
+ 4.01411062, 5.00000000, 6.00000000, 7.00000000, -6.00000000, -5.00000000, -4.00000000, -3.00000000, -2.00000000, -0.63703469,
+ 0.00169604, 0.66294191, 1.83808563, 3.00000000, 4.00000000, -8.00000000, -7.00000000, -6.03082300, -5.00000000, -3.88061019,
+-2.92670084, -1.99902336, -0.72898996, -0.02880170, 0.73769927, 1.95920233, 2.78356263, 4.08100921, 5.00000000, 6.00000000,
+ 6.78771437, 8.00000000, 9.00000000, 10.00000000, 11.00000000, -5.00000000, -4.00000000, -2.88150384, -1.89520024, -0.71479482,
+ 0.00962397, 0.72816030, 1.73583550, 3.00000000, 4.00000000, 5.00000000, 6.00000000, 7.00000000, -16.00000000, -15.00000000,
+-13.80516401, -13.00000000, -12.00000000, -11.00000000, -10.02723144, -9.11825995, -8.05820112, -7.00000000, -6.17943541, -5.01837980,
+-3.97546169, -2.92806857, -1.89778775, -0.81138893, -0.02246016, 0.80528415, 1.85705214, 2.96438524, 3.97540151, 4.79684246,
+ 6.00000000, 6.75549513, 8.12185828, 9.00000000, 10.00000000, 11.00000000, 12.00000000, 13.00000000, 14.00000000, -11.00000000,
+-10.00000000, -9.00000000, -7.91603344, -6.77865892, -5.85765006, -4.93342332, -3.96679157, -2.84925552, -1.89230732, -0.85384229,
+ 0.00579591, 0.84863246, 1.89006713, 2.89483818, 3.87322971, 5.13228411, 6.00000000, 7.00000000, 8.00000000, 9.00000000,
+ 10.00000000, 11.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000, -29.00000000, -28.00000000, -27.00000000,
+-26.00000000, -25.00000000, -24.00000000, -23.00000000, -22.00000000, -21.00000000, -20.19501953, -19.00000000, -18.00000000, -17.00000000,
+-16.00000000, -14.89069633, -14.00000000, -13.00000000, -12.16260304, -11.15418282, -9.83543570, -8.85600407, -7.82712677, -7.05664308,
+-5.97007352, -4.89268438, -3.93822771, -2.94975269, -1.92192127, -0.90702480, 0.03974847, 0.92488359, 1.93747579, 2.94500522,
+ 3.95181797, 4.95433087, 5.95141808, 7.00212920, 8.02964757, 9.03210585, 9.84644504, 10.82907720, 11.87622530, 12.96908371,
+ 14.00000000, 15.16963413, 15.94902025, 17.00000000, 18.00000000, 19.00000000, 20.00000000, 21.00000000, 22.00000000, -29.00000000,
+-27.79780781, -27.00757888, -26.01571026, -24.89695568, -23.99946491, -22.98699614, -21.96678139, -20.99883532, -20.00851529, -18.94738054,
+-17.98672566, -16.98684787, -15.96917397, -14.99856852, -13.98974852, -12.97786927, -11.96110939, -10.98877093, -9.99875257, -8.99001359,
+-8.00799989, -6.99471760, -6.00034670, -4.99936372, -4.00581479, -3.00424577, -2.02047620, -0.99713266, -0.00366397, 1.00803955,
+ 1.98452687, 3.00748501, 4.02714611, 4.97661026, 5.99337271, 6.99754716, 8.00713602, 8.97184974, 9.98047901, 10.97685939,
+ 11.99533975, 12.96107876, 13.95061478, 15.00756776, 15.94078690, 16.88231059, 17.92069248, 18.78011047, 20.00000000, 21.00000000,
+ 22.00000000, -55.76988333, -54.96048193, -53.88411581, -52.94117980, -51.80983449, -50.90359699, -50.00000000, -48.99838741, -47.97685542,
+-47.03288597, -45.97820919, -45.02418374, -43.90081897, -42.88832512, -41.98234549, -40.96745512, -39.98148729, -39.06792854, -37.96493755,
+-36.98707870, -36.03416079, -35.01192444, -33.95785029, -32.99469087, -31.96633807, -31.01769053, -29.99727691, -28.99329690, -27.98873019,
+-27.00344273, -25.97657141, -25.00511074, -23.96689479, -23.01566842, -22.01632643, -21.00076343, -19.97788007, -18.97248680, -17.96076284,
+-16.97585453, -15.98345587, -15.01612745, -13.96862118, -12.96622055, -12.02196641, -11.02078103, -9.98445656, -9.00050060, -8.03442387,
+-7.00363761, -5.97921358, -4.98886269, -4.00528221, -3.01672947, -1.98599795, -1.00668518, -0.02633490, 1.00794139, 2.00837138,
+ 2.99213287, 3.98710216, 4.99064334, 6.01416391, 7.01759708, 7.97878151, 8.99665730, 10.02656114, 11.01863887, 12.01207901,
+ 13.00958725, 13.99237829, 15.00954971, 16.00724653, 17.00606559, 17.99886292, 18.99611967, 19.98808171, 21.01871930, 21.97014763,
+ 22.99833843, 24.00316842, 24.99949142, 25.98539601, 27.02480733, 27.98075377, 28.98266019, 30.00611445, 30.99409128, 31.94523141,
+ 32.97688339, 33.98800206, 35.00177074, 35.98639997, 36.98939428, 37.95644255, 39.00114054, 39.99492439, 40.99338254, 41.97050844,
+ 43.03085663, 43.96757668, 44.97800970, 45.95953358, 46.98109551, 47.99368477, 49.00141209, 49.94459923, 50.93298108, 51.99894661,
+ 53.06463883, 53.99704669, 55.02037199, 55.98368047, 57.01930954, 58.03813852, 58.96232502, 60.01644186, 61.03254711, 62.01086576,
+ 62.87962247, 63.98378413, 65.02189831, 65.93003954, 66.92439900, 68.07051633, 68.95928756, 70.03315022, 71.05579859, 72.00000000,
+ 73.00000000, 74.00000000, 75.00000000, 75.93485291, 77.20950456, 78.00000000, 79.00000000, 79.91519960, 81.00000000, -3.00000000,
+-2.00000000, -0.65174074, -0.00092112, 0.62967387, 2.00000000, 3.00000000, -6.00000000, -5.00000000, -4.00000000, -2.89861729,
+-1.69999061, -0.72632201, 0.00219241, 0.72891750, 1.73257865, 3.00000000, 3.76561508, 5.00000000, 6.00000000, -3.00000000,
+-2.00000000, -0.66227013, 0.00389373, 0.66163500, 2.00000000, 3.00000000, 4.00000000, -8.00000000, -7.00000000, -6.00000000,
+-4.76421796, -4.04320264, -3.01415201, -1.84346485, -0.77185048, 0.00061977, 0.76274524, 1.84330156, 3.00000000, 4.00000000,
+ 5.00000000, 6.00000000, 7.00000000, -6.00000000, -5.00000000, -4.00000000, -3.00000000, -1.75749611, -0.72951347, -0.00104394,
+ 0.72040315, 1.72594036, 3.00000000, 4.00000000, -9.00000000, -8.00000000, -7.00000000, -5.90394062, -5.00000000, -3.75562807,
+-2.89699407, -1.86696610, -0.79056636, -0.00330943, 0.79744554, 1.85149941, 2.91118681, 3.99520311, 4.96341987, 6.00000000,
+ 7.00000000, 8.00000000, 9.00000000, 10.00000000, -6.00000000, -4.80151529, -4.00000000, -2.87442856, -1.85285815, -0.77767592,
+-0.02071301, 0.81752572, 1.82503940, 2.79602150, 3.92870203, 5.00000000, 6.00000000, 7.00000000, -17.00000000, -16.00000000,
+-15.00000000, -14.00000000, -13.00000000, -12.00000000, -11.00000000, -9.80059874, -9.00000000, -8.00185204, -7.13087808, -5.92942149,
+-4.77883243, -3.93417708, -2.88004618, -1.89952522, -0.86239337, 0.00332274, 0.86657548, 1.89479279, 2.89701813, 3.90987417,
+ 4.98910145, 6.07676766, 7.00000000, 8.00000000, 9.00000000, 10.00000000, 11.00000000, 12.00000000, 13.00000000, 14.00000000,
+-12.00000000, -11.00000000, -9.89996262, -8.85894205, -7.87594823, -6.99685317, -5.94917589, -4.93914916, -3.93317670, -2.93174244,
+-1.90737478, -0.90982242, 0.00803316, 0.90111563, 1.90362879, 2.90332432, 3.90654662, 4.94461954, 5.87963665, 6.91988113,
+ 7.79514004, 8.98805413, 10.00000000, 11.00000000, -35.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000,
+-29.00000000, -28.00000000, -27.00000000, -26.00000000, -25.00000000, -24.00000000, -22.88310970, -22.00000000, -21.00000000, -20.00000000,
+-19.00000000, -18.00000000, -17.00000000, -16.11854974, -15.00000000, -14.10507667, -13.04497040, -11.94846700, -10.97432494, -9.94514368,
+-8.97311414, -7.94171496, -6.97232122, -5.98590548, -4.97455572, -3.95477903, -2.93935454, -1.95573532, -0.97120273, -0.02084826,
+ 0.95689153, 1.96679781, 2.97060165, 3.96660892, 4.96754331, 5.97996089, 6.93822411, 7.96618014, 8.95809791, 9.98891474,
+ 10.95713402, 11.85433084, 13.03831696, 13.84035295, 15.00729606, 15.98652872, 17.20557599, 18.00000000, 18.90794805, 20.00000000,
+ 21.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000, -28.97280602, -28.00000000, -27.16255057, -26.04078092,
+-24.85442050, -24.15783484, -22.78614956, -21.95739865, -21.21844626, -20.03008104, -19.03888543, -17.90460490, -17.02064693, -15.84673652,
+-14.87140709, -13.87996048, -12.94907251, -11.96795995, -11.00977925, -9.95103238, -8.96674655, -7.96351667, -6.96886200, -5.99335494,
+-4.97515534, -3.98891694, -2.99581150, -1.98758360, -0.99249128, -0.00001403, 0.98807868, 1.99119869, 2.99019366, 3.98612953,
+ 5.00312941, 5.98833080, 6.99686651, 7.98373889, 8.97942222, 9.94202752, 10.99671622, 11.94306164, 12.98539825, 13.90728690,
+ 14.89907642, 15.94836675, 16.89611342, 17.84084949, 18.74910958, 20.00000000, -67.00000000, -66.00000000, -65.00000000, -64.00000000,
+-63.02511977, -62.00000000, -61.06061493, -59.95964043, -59.12824439, -58.00000000, -57.00000000, -56.00000000, -54.87857996, -54.09689334,
+-53.00000000, -52.21057366, -50.93867921, -50.03032952, -49.19283867, -47.89439051, -46.99505692, -46.04895543, -44.89687413, -43.78942208,
+-42.99025156, -41.88436155, -40.99169704, -40.00320429, -38.90181498, -38.06029271, -37.05030818, -36.07554573, -35.03202233, -33.93117946,
+-32.97736655, -31.98942819, -30.99546798, -30.01511004, -28.97296525, -28.02561164, -26.94386985, -25.99632704, -25.00461143, -24.01578192,
+-22.99177609, -22.02261094, -20.97939001, -19.96176066, -19.00442980, -18.01529434, -17.00196902, -15.99794828, -14.98675055, -13.97517657,
+-12.98676283, -11.99718760, -11.00167809, -9.98872268, -9.02138474, -8.00320338, -6.99542797, -6.00059136, -5.01311763, -4.00336943,
+-3.00348281, -1.99365875, -0.98223019, 0.00126343, 0.99699237, 1.99381968, 3.00054436, 3.99898305, 5.00160508, 6.00310399,
+ 6.99885096, 8.02740039, 8.99515550, 9.98962151, 11.00642302, 11.98694516, 13.00018933, 13.97726018, 14.99186645, 16.00580131,
+ 16.97434224, 17.96982658, 19.00066438, 20.01228749, 21.00741822, 21.94988312, 23.00860212, 23.98801542, 24.97638417, 25.98003521,
+ 27.02336188, 27.99667029, 29.01014125, 30.02481912, 31.01415797, 31.97399854, 33.06214485, 33.99929330, 34.94095386, 35.96368372,
+ 36.96980925, 37.98389244, 39.01121235, 40.00715026, 41.06382894, 41.96618280, 43.01555590, 43.95430436, 45.01970038, 45.99967821,
+ 47.19847394, 48.04852502, 49.10609965, 50.04244122, 50.86051406, 51.92983796, 53.02781107, 54.06248545, 54.89942009, 56.08347165,
+ 57.06887956, 58.09671115, 59.07832400, 59.87005277, 61.14778499, 62.00000000, 63.00000000, 64.00000000, 65.00000000, 66.00000000,
+ 67.00000000, 68.00000000, 69.00000000, 70.00000000, 71.00000000, 72.00000000};
+
+const double WebRtcIsac_kQKltLevelsShape[1735] = {
+ 0.00032397, 0.00008053, -0.00061202, -0.00012620, 0.00030437, 0.00054764, -0.00027902, 0.00069360, 0.00029449, -0.80219239,
+ 0.00091089, -0.74514927, -0.00094283, 0.64030631, -0.60509119, 0.00035575, 0.61851665, -0.62129957, 0.00375219, 0.60054900,
+-0.61554359, 0.00054977, 0.63362016, -1.73118727, -0.65422341, 0.00524568, 0.66165298, 1.76785515, -1.83182018, -0.65997434,
+-0.00011887, 0.67524299, 1.79933938, -1.76344480, -0.72547708, -0.00133017, 0.73104704, 1.75305377, 2.85164534, -2.80423916,
+-1.71959639, -0.75419722, -0.00329945, 0.77196760, 1.72211069, 2.87339653, 0.00031089, -0.00015311, 0.00018201, -0.00035035,
+-0.77357251, 0.00154647, -0.00047625, -0.00045299, 0.00086590, 0.00044762, -0.83383829, 0.00024787, -0.68526258, -0.00122472,
+ 0.64643255, -0.60904942, -0.00448987, 0.62309184, -0.59626442, -0.00574132, 0.62296546, -0.63222115, 0.00013441, 0.63609545,
+-0.66911055, -0.00369971, 0.66346095, 2.07281301, -1.77184694, -0.67640425, -0.00010145, 0.64818392, 1.74948973, -1.69420224,
+-0.71943894, -0.00004680, 0.75303493, 1.81075983, 2.80610041, -2.80005755, -1.79866753, -0.77409777, -0.00084220, 0.80141293,
+ 1.78291081, 2.73954236, 3.82994169, 0.00015140, -0.00012766, -0.00034241, -0.00119125, -0.76113497, 0.00069246, 0.76722027,
+ 0.00132862, -0.69107530, 0.00010656, 0.77061578, -0.78012970, 0.00095947, 0.77828502, -0.64787758, 0.00217168, 0.63050167,
+-0.58601125, 0.00306596, 0.59466308, -0.58603410, 0.00059779, 0.64257970, 1.76512766, -0.61193600, -0.00259517, 0.59767574,
+-0.61026273, 0.00315811, 0.61725479, -1.69169719, -0.65816029, 0.00067575, 0.65576890, 2.00000000, -1.72689193, -0.69780808,
+-0.00040990, 0.70668487, 1.74198458, -3.79028154, -3.00000000, -1.73194459, -0.70179341, -0.00106695, 0.71302629, 1.76849782,
+-2.89332364, -1.78585007, -0.78731491, -0.00132610, 0.79692976, 1.75247009, 2.97828682, -5.26238694, -3.69559829, -2.87286122,
+-1.84908818, -0.84434577, -0.01167975, 0.84641753, 1.84087672, 2.87628156, 3.83556679, -0.00190204, 0.00092642, 0.00354385,
+-0.00012982, -0.67742785, 0.00229509, 0.64935672, -0.58444751, 0.00470733, 0.57299534, -0.58456202, -0.00097715, 0.64593607,
+-0.64060330, -0.00638534, 0.59680157, -0.59287537, 0.00490772, 0.58919707, -0.60306173, -0.00417464, 0.60562100, -1.75218757,
+-0.63018569, -0.00225922, 0.63863300, -0.63949939, -0.00126421, 0.64268914, -1.75851182, -0.68318060, 0.00510418, 0.69049211,
+ 1.88178506, -1.71136148, -0.72710534, -0.00815559, 0.73412917, 1.79996711, -2.77111145, -1.73940498, -0.78212945, 0.01074476,
+ 0.77688916, 1.76873972, 2.87281379, 3.77554698, -3.75832725, -2.95463235, -1.80451491, -0.80017226, 0.00149902, 0.80729206,
+ 1.78265046, 2.89391793, -3.78236148, -2.83640598, -1.82532067, -0.88844327, -0.00620952, 0.88208030, 1.85757631, 2.81712391,
+ 3.88430176, 5.16179367, -7.00000000, -5.93805408, -4.87172597, -3.87524433, -2.89399744, -1.92359563, -0.92136341, -0.00172725,
+ 0.93087018, 1.90528280, 2.89809686, 3.88085708, 4.89147740, 5.89078692, -0.00239502, 0.00312564, -1.00000000, 0.00178325,
+ 1.00000000, -0.62198029, 0.00143254, 0.65344051, -0.59851220, -0.00676987, 0.61510140, -0.58894151, 0.00385055, 0.59794203,
+-0.59808568, -0.00038214, 0.57625703, -0.63009713, -0.01107985, 0.61278758, -0.64206758, -0.00154369, 0.65480598, 1.80604162,
+-1.80909286, -0.67810514, 0.00205762, 0.68571097, 1.79453891, -3.22682422, -1.73808453, -0.71870305, -0.00738594, 0.71486172,
+ 1.73005326, -1.66891897, -0.73689615, -0.00616203, 0.74262409, 1.73807899, -2.92417482, -1.73866741, -0.78133871, 0.00764425,
+ 0.80027264, 1.78668732, 2.74992588, -4.00000000, -2.75578740, -1.83697516, -0.83117035, -0.00355191, 0.83527172, 1.82814700,
+ 2.77377675, 3.80718693, -3.81667698, -2.83575471, -1.83372350, -0.86579471, 0.00547578, 0.87582281, 1.82858793, 2.87265007,
+ 3.91405377, -4.87521600, -3.78999094, -2.86437014, -1.86964365, -0.90618018, 0.00128243, 0.91497811, 1.87374952, 2.83199819,
+ 3.91519130, 4.76632822, -6.68713448, -6.01252467, -4.94587936, -3.88795368, -2.91299088, -1.92592211, -0.95504570, -0.00089980,
+ 0.94565200, 1.93239633, 2.91832808, 3.91363475, 4.88920034, 5.96471415, 6.83905252, 7.86195009, 8.81571018, -12.96141759,
+-11.73039516, -10.96459719, -9.97382433, -9.04414433, -7.89460619, -6.96628608, -5.93236595, -4.93337924, -3.95479990, -2.96451499,
+-1.96635876, -0.97271229, -0.00402238, 0.98343930, 1.98348291, 2.96641164, 3.95456471, 4.95517089, 5.98975714, 6.90322073,
+ 7.90468849, 8.85639467, 9.97255498, 10.79006309, 11.81988596, 0.04950500, -1.00000000, -0.01226628, 1.00000000, -0.59479469,
+-0.10438305, 0.59822144, -2.00000000, -0.67109149, -0.09256692, 0.65171621, 2.00000000, -3.00000000, -1.68391999, -0.76681039,
+-0.03354151, 0.71509146, 1.77615472, -2.00000000, -0.68661511, -0.02497881, 0.66478398, 2.00000000, -2.00000000, -0.67032784,
+-0.00920582, 0.64892756, 2.00000000, -2.00000000, -0.68561894, 0.03641869, 0.73021611, 1.68293863, -4.00000000, -2.72024184,
+-1.80096059, -0.81696185, 0.03604685, 0.79232033, 1.70070730, 3.00000000, -4.00000000, -2.71795670, -1.80482986, -0.86001162,
+ 0.03764903, 0.87723968, 1.79970771, 2.72685932, 3.67589143, -5.00000000, -4.00000000, -2.85492548, -1.78996365, -0.83250358,
+-0.01376828, 0.84195506, 1.78161105, 2.76754458, 4.00000000, -6.00000000, -5.00000000, -3.82268811, -2.77563624, -1.82608163,
+-0.86486114, -0.02671886, 0.86693165, 1.88422879, 2.86248347, 3.95632216, -7.00000000, -6.00000000, -5.00000000, -3.77533988,
+-2.86391432, -1.87052039, -0.90513658, 0.06271236, 0.91083620, 1.85734756, 2.86031688, 3.82019418, 4.94420394, 6.00000000,
+-11.00000000, -10.00000000, -9.00000000, -8.00000000, -6.91952415, -6.00000000, -4.92044374, -3.87845165, -2.87392362, -1.88413020,
+-0.91915740, 0.00318517, 0.91602800, 1.89664838, 2.88925058, 3.84123856, 4.78988651, 5.94526812, 6.81953917, 8.00000000,
+-9.00000000, -8.00000000, -7.03319143, -5.94530963, -4.86669720, -3.92438007, -2.88620396, -1.92848070, -0.94365985, 0.01671855,
+ 0.97349410, 1.93419878, 2.89740109, 3.89662823, 4.83235583, 5.88106535, 6.80328232, 8.00000000, -13.00000000, -12.00000000,
+-11.00000000, -10.00000000, -9.00000000, -7.86033489, -6.83344055, -5.89844215, -4.90811454, -3.94841298, -2.95820490, -1.98627966,
+-0.99161468, -0.02286136, 0.96055651, 1.95052433, 2.93969396, 3.94304346, 4.88522624, 5.87434241, 6.78309433, 7.87244101,
+ 9.00000000, 10.00000000, -12.09117356, -11.00000000, -10.00000000, -8.84766108, -7.86934236, -6.98544896, -5.94233429, -4.95583292,
+-3.95575986, -2.97085529, -1.98955811, -0.99359873, -0.00485413, 0.98298870, 1.98093258, 2.96430203, 3.95540216, 4.96915010,
+ 5.96775124, 6.99236918, 7.96503302, 8.99864542, 9.85857723, 10.96541926, 11.91647197, 12.71060069, -26.00000000, -25.00000000,
+-24.00585596, -23.11642573, -22.14271284, -20.89800711, -19.87815799, -19.05036354, -17.88555651, -16.86471209, -15.97711073, -14.94012359,
+-14.02661226, -12.98243228, -11.97489256, -10.97402777, -9.96425624, -9.01085220, -7.97372506, -6.98795002, -5.97271328, -5.00191694,
+-3.98055849, -2.98458048, -1.99470442, -0.99656768, -0.00825666, 1.00272004, 1.99922218, 2.99357669, 4.01407905, 5.01003897,
+ 5.98115528, 7.00018958, 8.00338125, 8.98981046, 9.98990318, 10.96341479, 11.96866930, 12.99175139, 13.94580443, 14.95745083,
+ 15.98992869, 16.97484646, 17.99630043, 18.93396897, 19.88347741, 20.96532482, 21.92191032, 23.22314702, 0.00006846, 0.00014352,
+-0.00056203, 0.00027588, -0.00147678, 1.00000000, 0.00003823, 0.00001975, -0.00033710, -0.00096712, 1.00000000, -1.00000000,
+ 0.00067511, -1.00000000, 0.00342065, 1.00000000, -1.00000000, 0.00196254, 1.00000000, -1.00000000, 0.00201173, 1.00000000,
+-2.00000000, -1.00000000, -0.00381686, 1.00000000, -1.00000000, 0.00178037, 1.00000000, -2.00000000, -1.00000000, -0.00320274,
+ 1.00000000, 2.00000000, -2.00000000, -1.00000000, 0.00426519, 1.00000000, 2.00000000, -3.00000000, -2.00000000, -1.00000000,
+-0.00074072, 0.64654602, 2.00000000, 0.00031217, 0.00063348, 0.00020247, 0.00047891, 0.00122893, -0.00150669, -0.00148276,
+ 0.00016848, 0.00147085, 1.00000000, -0.00088160, 1.00000000, -1.00000000, 0.00381641, 1.00000000, -1.00000000, 0.00129816,
+ 1.00000000, -1.00000000, 0.00074903, 1.00000000, -2.00000000, -0.76230566, -0.00370764, 0.82467977, -0.78769346, -0.00492670,
+ 0.84532630, -2.00000000, -0.70943195, -0.01257613, 0.75905385, 2.00000000, -2.00000000, -0.62780445, -0.00408633, 0.60272506,
+ 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.61412985, 0.00102833, 0.61527589, 2.00000000, 3.00000000, 0.00012115,
+-0.00080909, 0.00071061, -0.00227957, 0.00179794, 0.00103827, -1.00000000, 0.00444757, -1.00000000, 0.00604068, 1.00000000,
+-1.00000000, 0.00427327, 1.00000000, -1.00000000, 0.00086662, 1.00000000, -1.00000000, -0.00837492, 1.00000000, -0.65715934,
+-0.00645342, 0.64004630, -2.00000000, -0.64987682, -0.01449567, 0.69893373, -2.00000000, -0.63221961, 0.00421765, 0.62452105,
+ 2.00000000, -2.00000000, -0.60027006, -0.00110630, 0.62033821, 2.00000000, -2.00000000, -0.59823932, 0.00928313, 0.62188520,
+ 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.63230286, -0.00248555, 0.62632575, 2.00000000, 3.00000000, -5.00000000,
+-4.00000000, -3.00000000, -2.00000000, -0.66521143, 0.00544305, 0.66930486, 2.00000000, 3.00000000, 4.00000000, 0.00077008,
+ 0.00061140, -0.00009317, -0.00049643, 1.00000000, -1.00000000, -0.00285084, 1.00000000, -1.00000000, 0.00601784, 1.00000000,
+-1.00000000, -0.00091887, 0.75122772, -0.71579859, -0.00043545, 1.00000000, -0.85571363, -0.00227654, 0.63816873, -1.00000000,
+-0.00393484, 0.76748004, -0.58223659, -0.01229777, 0.58080322, -0.61945902, -0.00232238, 0.62277938, 2.00000000, -2.00000000,
+-0.60595489, -0.00535702, 0.60547736, 2.00000000, -4.00000000, -3.00000000, -2.00000000, -0.62368122, 0.01112097, 0.63997294,
+ 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.64318217, 0.00515139, 0.64781184, 2.00000000, -3.00000000, -1.78031579,
+-0.67122588, 0.02153711, 0.67899877, 2.00000000, 3.00000000, 4.00000000, -4.00000000, -3.00000000, -1.80503233, -0.69835727,
+-0.00270770, 0.70999554, 1.77332849, 3.00000000, 4.00000000, 5.00000000, -8.00000000, -7.00000000, -6.00000000, -5.00000000,
+-4.00000000, -2.81600693, -1.72970368, -0.73779413, -0.01384841, 0.75694606, 1.80042618, 3.00000000, 4.00000000, 5.00000000,
+ 6.00000000, -0.00051787, 0.00059593, -0.00023319, -1.00000000, 0.00191861, 0.79547197, -0.75020995, 0.00217840, 0.69165833,
+-1.00000000, -0.00304964, 0.67698951, -0.64516943, -0.00657667, 0.59260129, -0.62819301, -0.00456626, 0.59426260, -0.60909519,
+ 0.00256476, 0.61660408, -0.66560131, -0.00293463, 0.67477566, 2.00000000, -2.00000000, -0.62484067, 0.00505116, 0.63491494,
+ 2.00000000, -3.00000000, -2.00000000, -0.68427246, 0.00924353, 0.68755774, 2.00000000, 3.00000000, -3.00000000, -2.00000000,
+-0.65390928, 0.01008025, 0.65849449, 2.00000000, 3.00000000, -5.00000000, -4.00000000, -3.00000000, -1.70848232, -0.72079538,
+-0.00007674, 0.71556176, 1.76815351, 3.00000000, 4.00000000, 5.00000000, -4.00000000, -3.00000000, -1.82887466, -0.73529886,
+ 0.00033458, 0.73847588, 1.83009515, 3.00000000, 4.00000000, -5.00000000, -4.00000000, -2.83203553, -1.79500085, -0.77452749,
+-0.00614320, 0.77416943, 1.82469471, 2.77034612, 4.00000000, 5.00000000, -7.00000000, -6.00000000, -5.00000000, -4.00000000,
+-2.76574798, -1.84700836, -0.80822297, 0.00054165, 0.80901445, 1.85687331, 2.75680191, 3.81986695, 5.00000000, 6.00000000,
+ 7.00000000, 8.00000000, -13.00000000, -12.00000000, -11.00000000, -10.00000000, -9.00000000, -8.00000000, -7.00000000, -6.00000000,
+-5.00000000, -3.88304817, -2.93396067, -1.86645989, -0.84825410, 0.00666207, 0.84853252, 1.88634684, 2.95282618, 3.89813287,
+ 4.89189079, 6.00000000, 7.00000000, 8.00000000, 9.00000000, 10.00000000, 11.00000000, -0.00344877, 1.00000000, -0.61413659,
+-0.02115630, 0.59438887, -0.60873054, 0.00844993, 0.62510557, -2.00000000, -0.75002947, 0.00120913, 0.66616051, -2.00000000,
+-0.72324691, 0.04760499, 0.70532533, 2.00000000, -3.00000000, -1.66577589, -0.78941380, -0.01909714, 0.74993685, 1.70945570,
+-1.64422308, -0.70992006, -0.02795108, 0.76990363, 1.79682243, 2.96233315, -1.71686461, -0.76572785, -0.00041846, 0.78174132,
+ 1.66217596, -3.00000000, -1.77033369, -0.79475091, 0.03709740, 0.80097076, 1.83947400, 2.85879773, -4.00000000, -3.16528651,
+-1.79564411, -0.90078981, 0.02403102, 0.86138856, 1.84207433, 2.74584048, -4.00000000, -2.91249347, -1.87804769, -0.87323549,
+ 0.08164382, 0.89037056, 1.82505263, 2.71336163, 4.00000000, -4.81262228, -3.87173565, -2.83424209, -1.87517938, -0.86199960,
+ 0.00268598, 0.89547657, 1.90713511, 2.85219071, 3.86417171, 4.80711781, 6.00000000, 7.00000000, -5.00000000, -3.82388480,
+-2.82875808, -1.90350457, -0.90795818, 0.03047007, 0.93676836, 1.88844957, 2.83269711, 3.76109686, 5.00000000, -9.00000000,
+-8.00000000, -6.88037957, -5.88776398, -4.91209139, -3.93902541, -2.90989221, -1.92281230, -0.98960535, -0.07440511, 0.94023957,
+ 1.91666262, 2.83340828, 3.83651295, 4.77839424, 6.12284019, 7.00000000, 8.00000000, 9.00000000, -12.00000000, -11.00000000,
+-10.00000000, -9.00000000, -8.00000000, -6.68554513, -5.97994708, -4.98789075, -3.91383581, -2.92952795, -1.91727195, -0.93148075,
+-0.00568870, 0.93515148, 1.94580068, 2.93838956, 3.92567644, 4.96573603, 5.95402763, 7.00000000, 8.00000000, 9.00000000,
+-11.00000000, -9.90096030, -8.97868124, -7.93663988, -6.98806055, -5.95937864, -4.93473664, -3.95454756, -2.96518446, -1.97711766,
+-0.98552111, -0.03317271, 0.95115775, 1.93785086, 2.96310779, 3.93322450, 5.01716212, 5.85909823, 6.89163669, 7.97492693,
+ 8.85698897, 9.79802946, 11.09373957, 12.00000000, -13.00000000, -12.00000000, -10.67579109, -9.95079100, -8.90576592, -7.93254656,
+-6.96112672, -5.96015798, -4.95493809, -3.98556269, -2.98182856, -1.98150255, -0.96551153, -0.00399791, 0.98644875, 1.98043830,
+ 2.97969033, 3.97728257, 4.95173541, 5.95649050, 6.96447378, 7.95591513, 9.07680954, 9.92093070, 10.76496555, 11.97525735,
+ 13.00000000, 14.00000000, -25.00000000, -24.00000000, -23.00000000, -22.00072357, -21.00000000, -20.00000000, -19.00000000, -18.20003462,
+-17.01648407, -15.78651996, -14.95660266, -13.99167850, -13.28722978, -11.85013840, -10.92025302, -9.87055810, -8.93841040, -7.95329867,
+-6.97819441, -6.01593394, -5.00905213, -3.99905285, -2.99171810, -1.99062796, -1.00112466, 0.00140492, 1.00701091, 2.02327185,
+ 3.00194633, 3.99188294, 5.00313145, 6.00448038, 6.98904951, 7.98158293, 8.98212774, 10.00363404, 10.98641678, 11.98034311,
+ 12.95176779, 13.95383703, 14.99084578, 15.98600642, 16.99406826, 17.98134623, 19.01793961, 19.86072639, 20.88465474, 21.99287082,
+ 22.81916620, 23.77946383, 0.00000234, 0.00000298, 0.00000048, 0.00002408, -0.00000165, -0.00001831, -0.00005703, -0.00000184,
+-1.00000000, 0.00001977, 1.00000000, -1.00000000, 0.00000010, 1.00000000, -1.00000000, -0.00001152, 1.00000000, -1.00000000,
+ 0.00000840, 1.00000000, -1.00000000, 0.00002353, 1.00000000, -0.75455603, -0.00001433, 1.00000000, -0.65859705, -0.00000703,
+ 0.62995860, -2.00000000, -0.72724652, -0.00033969, 0.61359174, 2.00000000, -2.00000000, -0.69510998, -0.00031410, 0.66467605,
+ 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.65738683, 0.00039019, 0.66554720, 1.91774106, 3.18089124, 0.00000070,
+ 0.00001152, -0.00000795, -0.00000058, -0.00003502, -0.00001508, -0.00004225, -0.00002165, -1.00000000, 0.00004391, 1.00000000,
+-1.00000000, 0.00001784, 1.00000000, -1.00000000, -0.00003678, 1.00000000, -0.68878314, -0.00013166, 0.60880149, -0.75291978,
+ 0.00006493, 1.00000000, -0.76757316, 0.00003057, 0.67140524, -0.61602267, -0.00014495, 0.63625803, 2.00000000, -2.00000000,
+-0.61253314, -0.00116483, 0.65071851, 2.00000000, -3.00000000, -1.71451667, -0.67799909, -0.00048294, 0.65846019, 2.00000000,
+-3.02497593, -1.83515395, -0.70317981, 0.00519701, 0.67780009, 1.84218153, 2.88846262, 4.00000000, 0.00001124, 0.00000588,
+-0.00000172, 0.00002835, 1.00000000, 0.00001012, -0.00008644, 1.00000000, -0.75115901, 0.00004347, 1.00000000, -1.00000000,
+ 0.00002800, 1.00000000, -1.00000000, -0.00006039, 1.00000000, -0.79763258, -0.00011907, 0.71713616, -0.76791870, -0.00007113,
+ 0.63583609, -0.62337806, 0.00012891, 0.62242094, -0.60837055, 0.00043216, 0.65515705, -0.63637782, -0.00019749, 0.60423967,
+ 2.00000000, -2.00000000, -0.65404827, -0.00089304, 0.64706660, 2.00000000, -1.86334076, -0.66410366, 0.00063219, 0.66968004,
+ 2.00000000, 3.00000000, 4.00000000, -4.00000000, -3.00000000, -1.79048834, -0.69451890, 0.00030677, 0.71009333, 1.70591343,
+ 3.00000000, 4.00000000, -4.00000000, -2.90176499, -1.78368781, -0.74425178, 0.00234068, 0.74847325, 1.78886822, 2.78478854,
+ 3.83608985, 4.95996151, 0.00002170, 0.00001281, 0.00002162, -1.00000000, -0.00007266, 1.00000000, -1.00000000, -0.00003250,
+-0.64088804, 0.00015239, 1.00000000, -0.58450370, -0.00008410, 0.60567186, -1.00000000, -0.00010752, 1.00000000, -0.58922508,
+-0.00017378, 0.60755779, -0.62797206, -0.00001016, 0.64432847, -0.58497934, -0.00001851, 0.59716791, -0.62642499, -0.00097386,
+ 0.63568558, 2.00000000, -2.00000000, -0.63236390, -0.00173361, 0.63142762, 1.75629192, -3.00000000, -2.00000000, -0.65596684,
+ 0.00209364, 0.65419742, 2.00000000, -3.00000000, -1.73856625, -0.67767521, -0.00119512, 0.68973603, 1.70985573, 3.00000000,
+-3.00000000, -1.81820220, -0.73974134, 0.00695869, 0.72216179, 1.75624461, 3.00000000, 4.00000000, -5.00000000, -4.00000000,
+-3.17718593, -1.76857567, -0.76822322, 0.00267400, 0.76414602, 1.84309221, 3.04940652, 4.00000000, -7.08189123, -6.00000000,
+-5.22882249, -3.96477958, -2.79653492, -1.81923435, -0.80050253, -0.01086663, 0.82708565, 1.85804900, 2.89996354, 3.76028554,
+ 4.80518081, 5.81738096, 7.00000000, 8.00000000, 9.08816091, -0.00002979, -0.00000333, -1.00000000, -0.00011532, 1.00000000,
+-0.70921122, -0.00005325, 0.68933188, -0.67581263, -0.00023107, 0.57868212, -0.58388312, -0.00020850, 0.60149012, -0.60912457,
+ 0.00001567, 0.60180554, -0.59130091, -0.00038863, 0.59908653, -2.00000000, -0.63697707, 0.00083913, 0.62040514, 2.00000000,
+-2.00000000, -0.63216238, -0.00081100, 0.64411071, 2.00000000, -1.76856259, -0.65266989, -0.00243486, 0.66888899, 2.00000000,
+-1.75427214, -0.71415385, -0.00226376, 0.71296778, 1.66182947, 3.00000000, -3.00000000, -1.72505821, -0.72920134, -0.00360424,
+ 0.73800767, 1.72848281, 3.00000000, -4.00000000, -2.95284408, -1.72025758, -0.76503859, 0.00418761, 0.75297139, 1.73959808,
+ 3.00000000, -5.00000000, -3.96232791, -2.74080544, -1.78897123, -0.80233505, -0.00002050, 0.79693417, 1.76182598, 2.78434458,
+ 3.85693287, 5.00000000, -6.00000000, -4.78439284, -3.83501790, -2.85203629, -1.84909573, -0.85382658, -0.00181019, 0.84735145,
+ 1.83676575, 2.83656843, 3.86722376, 4.79702431, 6.00000000, -9.00000000, -8.00000000, -7.00000000, -6.07957292, -4.84677515,
+-3.85093972, -2.88683139, -1.84596391, -0.88058034, -0.00008692, 0.87554746, 1.86933183, 2.84729990, 3.89029797, 4.87311773,
+ 5.90844023, 7.00000000, -11.00000000, -9.97745420, -8.90015761, -7.94187517, -6.86987726, -5.84795335, -4.86693435, -3.90601819,
+-2.91031804, -1.91620096, -0.90497055, 0.00659199, 0.90926869, 1.90980821, 2.91070850, 3.93685967, 4.85581177, 6.06727337,
+ 7.05801043, 8.00000000, 9.00000000, 10.00000000, 10.90825787, 12.00000000, 13.00000000, 14.00000000, -0.00008918, 1.00000000,
+-0.54405938, 0.00120348, 0.55781920, -0.59227786, -0.00349602, 0.59777231, -1.63717598, -0.69048065, 0.00999281, 0.65770558,
+ 2.00000000, -2.00000000, -0.71013571, 0.00454518, 0.66991065, -3.00000000, -1.73004867, -0.73743921, 0.01162454, 0.69964842,
+ 1.83319587, -1.81225491, -0.76806000, 0.00164742, 0.76780397, 1.67168896, -1.64564794, -0.79903361, -0.01522880, 0.84277926,
+ 1.68873752, -3.00000000, -1.72063244, -0.83687428, 0.00246724, 0.84618697, 1.79464483, 2.77447025, -3.77118426, -2.75025539,
+-1.82050448, -0.90373722, -0.00187780, 0.90102245, 1.85249394, 2.71364180, -2.71720889, -1.79466125, -0.89860801, -0.02725825,
+ 0.90877329, 1.90542096, 2.76847902, 3.71496428, -4.70257302, -3.90746659, -2.87078421, -1.88858709, -0.93608993, -0.02157425,
+ 0.95181182, 1.91155682, 2.83614575, 3.87820801, 4.72172277, -5.02764544, -3.80066801, -2.87484378, -1.90707477, -0.96326017,
+-0.01060091, 0.96558851, 1.92191548, 2.86970759, 3.85655474, 4.83135970, 5.76387469, -9.00000000, -8.00000000, -6.75261776,
+-5.86333393, -4.84846871, -3.91871758, -2.93827286, -1.93050320, -0.96359634, -0.00141931, 0.95926312, 1.92541870, 2.93009411,
+ 3.86699087, 4.82315929, 5.67815206, -8.76594345, -7.70350451, -6.91784020, -5.81539490, -4.92526872, -3.91513203, -2.92134949,
+-1.95465646, -0.97638102, -0.00742564, 0.96948714, 1.96401112, 2.95256722, 3.93146353, 4.90991357, 5.88139022, 6.88640588,
+ 7.82610489, 9.00000000, -10.97611369, -9.80036760, -8.91109518, -7.92809404, -6.93865353, -5.91965899, -4.92957669, -3.95206224,
+-2.97308718, -1.97778214, -0.98552568, -0.00063212, 0.98686014, 1.97511867, 2.97114218, 3.97854244, 4.96578513, 5.96457765,
+ 6.95180187, 7.95163483, 8.93760897, 9.87666900, 10.88024562, 11.96270158, 12.99519291, -15.00000000, -13.76826291, -12.97229116,
+-12.00334834, -10.95980884, -9.98190891, -8.93798503, -7.95621309, -6.96109479, -5.96056649, -4.95843419, -3.97688640, -2.98989576,
+-1.98533395, -0.99580972, 0.00694370, 0.99421120, 1.99033132, 2.98751217, 3.98549580, 4.96482394, 5.96623233, 6.93564626,
+ 7.93772467, 8.92015276, 9.88785129, 10.97606096, 11.79686057, -23.00000000, -22.00000000, -21.00000000, -20.00000000, -19.00000000,
+-17.73310977, -16.83574096, -15.90889480, -15.00437366, -13.95007272, -12.99296117, -11.98334751, -10.96970820, -9.97775151, -8.98193840,
+-7.98378966, -6.98887770, -5.99059477, -5.00228769, -3.99355850, -2.99947486, -1.99897483, -0.99375857, 0.00324880, 1.00215912,
+ 1.99277083, 3.00503747, 3.99390482, 4.98854283, 5.98753219, 6.98245347, 7.98089893, 8.95960522, 9.95663648, 11.00810285,
+ 12.01421617, 12.96208687, 13.99227766, 14.97230040, 15.95114804, 16.97347393, 17.97794884, 18.96777118, 19.94446034, 20.94799029,
+ 22.14740083, 22.84288347, 23.99212109, 25.00000000, 25.96562658};
+
+/* cdf tables for quantizer indices */
+const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[1212] = {
+ 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17,
+ 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426,
+ 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510,
+ 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603,
+ 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4,
+ 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535,
+ 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592,
+ 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, 65522,
+ 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525,
+ 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117,
+ 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792,
+ 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, 64329,
+ 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, 0, 8,
+ 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966,
+ 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, 65435,
+ 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59,
+ 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583,
+ 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747,
+ 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, 49797,
+ 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, 64509, 64798,
+ 65045, 65162, 65288, 65363, 65447, 65506, 65522, 65531, 65533, 65535,
+ 0, 4, 6, 25, 38, 71, 138, 264, 519, 808,
+ 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108,
+ 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, 48747,
+ 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, 64275, 64673,
+ 64936, 65112, 65217, 65334, 65426, 65464, 65477, 65489, 65518, 65527,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, 12,
+ 14, 16, 21, 33, 50, 71, 84, 92, 105, 138,
+ 180, 255, 318, 377, 435, 473, 511, 590, 682, 758,
+ 913, 1097, 1256, 1449, 1671, 1884, 2169, 2445, 2772, 3157,
+ 3563, 3944, 4375, 4848, 5334, 5820, 6448, 7101, 7716, 8378,
+ 9102, 9956, 10752, 11648, 12707, 13670, 14758, 15910, 17187, 18472,
+ 19627, 20649, 21951, 23169, 24283, 25552, 26862, 28227, 29391, 30764,
+ 31882, 33213, 34432, 35600, 36910, 38116, 39464, 40729, 41872, 43144,
+ 44371, 45514, 46762, 47813, 48968, 50069, 51032, 51974, 52908, 53737,
+ 54603, 55445, 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264,
+ 60607, 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776,
+ 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, 64551,
+ 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, 65426, 65447,
+ 65472, 65485, 65487, 65489, 65502, 65510, 65512, 65514, 65516, 65518,
+ 65522, 65531, 65533, 65535, 0, 2, 4, 6, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 10, 222, 65321,
+ 65513, 65528, 65531, 65533, 65535, 0, 2, 4, 50, 65476,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 12,
+ 38, 544, 64936, 65509, 65523, 65525, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 10, 1055, 64508, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 10, 12, 123,
+ 3956, 62999, 65372, 65495, 65515, 65521, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 12, 53, 4707, 59445,
+ 65467, 65525, 65527, 65529, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 38, 40, 50, 67,
+ 96, 234, 929, 14345, 55750, 64866, 65389, 65462, 65514, 65517,
+ 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 15, 35, 91, 377, 1946,
+ 13618, 52565, 63714, 65184, 65465, 65520, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
+ 14, 16, 18, 20, 22, 24, 26, 28, 30, 32,
+ 34, 36, 38, 40, 42, 44, 46, 48, 50, 52,
+ 54, 82, 149, 362, 751, 1701, 4239, 12893, 38627, 55072,
+ 60875, 63071, 64158, 64702, 65096, 65283, 65412, 65473, 65494, 65505,
+ 65508, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533,
+ 65535, 0, 2, 15, 23, 53, 143, 260, 418, 698,
+ 988, 1353, 1812, 2411, 3144, 4015, 5143, 6401, 7611, 8999,
+ 10653, 12512, 14636, 16865, 19404, 22154, 24798, 27521, 30326, 33102,
+ 35790, 38603, 41415, 43968, 46771, 49435, 52152, 54715, 57143, 59481,
+ 61178, 62507, 63603, 64489, 64997, 65257, 65427, 65473, 65503, 65520,
+ 65529, 65531, 65533, 65535, 0, 3, 6, 9, 26, 32,
+ 44, 46, 64, 94, 111, 164, 205, 254, 327, 409,
+ 506, 608, 733, 885, 1093, 1292, 1482, 1742, 1993, 2329,
+ 2615, 3029, 3374, 3798, 4257, 4870, 5405, 5992, 6618, 7225,
+ 7816, 8418, 9051, 9761, 10532, 11380, 12113, 13010, 13788, 14594,
+ 15455, 16361, 17182, 18088, 18997, 20046, 20951, 21968, 22947, 24124,
+ 25296, 26547, 27712, 28775, 29807, 30835, 31709, 32469, 33201, 34014,
+ 34876, 35773, 36696, 37620, 38558, 39547, 40406, 41277, 42367, 43290,
+ 44445, 45443, 46510, 47684, 48973, 50157, 51187, 52242, 53209, 54083,
+ 55006, 55871, 56618, 57293, 57965, 58556, 59222, 59722, 60180, 60554,
+ 60902, 61250, 61554, 61837, 62100, 62372, 62631, 62856, 63078, 63324,
+ 63557, 63768, 63961, 64089, 64235, 64352, 64501, 64633, 64770, 64887,
+ 65001, 65059, 65121, 65188, 65246, 65302, 65346, 65390, 65428, 65463,
+ 65477, 65506, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529,
+ 65531, 65533, 65535, 0, 2, 4, 109, 65332, 65531, 65533,
+ 65535, 0, 2, 4, 6, 8, 25, 1817, 63874, 65511,
+ 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 907, 65014,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 10,
+ 12, 132, 2743, 62708, 65430, 65525, 65527, 65529, 65531, 65533,
+ 65535, 0, 2, 4, 6, 8, 35, 3743, 61666, 65485,
+ 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 23,
+ 109, 683, 6905, 58417, 64911, 65398, 65497, 65518, 65525, 65527,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 6, 53, 510,
+ 10209, 55212, 64573, 65441, 65522, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
+ 22, 32, 90, 266, 1037, 3349, 14468, 50488, 62394, 64685,
+ 65341, 65480, 65514, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
+ 65533, 65535, 0, 2, 4, 6, 9, 16, 37, 106,
+ 296, 748, 1868, 5733, 18897, 45553, 60165, 63949, 64926, 65314,
+ 65441, 65508, 65524, 65529, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 18, 20, 22, 24,
+ 26, 28, 30, 32, 34, 36, 38, 40, 42, 44,
+ 46, 48, 50, 83, 175, 344, 667, 1293, 2337, 4357,
+ 8033, 14988, 28600, 43244, 52011, 57042, 59980, 61779, 63065, 63869,
+ 64390, 64753, 64988, 65164, 65326, 65422, 65462, 65492, 65506, 65522,
+ 65524, 65526, 65531, 65533, 65535, 0, 2, 4, 6, 8,
+ 10, 12, 14, 16, 25, 39, 48, 55, 62, 65,
+ 85, 106, 139, 169, 194, 252, 323, 485, 688, 1074,
+ 1600, 2544, 3863, 5733, 8303, 11397, 15529, 20273, 25734, 31455,
+ 36853, 41891, 46410, 50306, 53702, 56503, 58673, 60479, 61880, 62989,
+ 63748, 64404, 64852, 65124, 65309, 65424, 65480, 65524, 65528, 65533,
+ 65535, 0, 2, 4, 6, 8, 10, 12, 14, 21,
+ 23, 25, 27, 29, 31, 39, 41, 43, 48, 60,
+ 72, 79, 106, 136, 166, 187, 224, 252, 323, 381,
+ 427, 478, 568, 660, 783, 912, 1046, 1175, 1365, 1567,
+ 1768, 2024, 2347, 2659, 3049, 3529, 4033, 4623, 5281, 5925,
+ 6726, 7526, 8417, 9468, 10783, 12141, 13571, 15222, 16916, 18659,
+ 20350, 22020, 23725, 25497, 27201, 29026, 30867, 32632, 34323, 36062,
+ 37829, 39466, 41144, 42654, 43981, 45343, 46579, 47759, 49013, 50171,
+ 51249, 52283, 53245, 54148, 54938, 55669, 56421, 57109, 57791, 58464,
+ 59092, 59674, 60105, 60653, 61083, 61407, 61757, 62095, 62388, 62649,
+ 62873, 63157, 63358, 63540, 63725, 63884, 64046, 64155, 64278, 64426,
+ 64548, 64654, 64806, 64906, 64994, 65077, 65137, 65215, 65277, 65324,
+ 65354, 65409, 65437, 65455, 65462, 65490, 65495, 65499, 65508, 65511,
+ 65513, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
+ 65533, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[2059] = {
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4,
+ 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0,
+ 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779,
+ 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0,
+ 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576,
+ 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535,
+ 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611,
+ 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809,
+ 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535,
+ 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531,
+ 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527,
+ 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0,
+ 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005,
+ 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38,
+ 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804,
+ 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535,
+ 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522,
+ 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0,
+ 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780,
+ 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84,
+ 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150,
+ 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949,
+ 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200,
+ 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670,
+ 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535,
+ 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535,
+ 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419,
+ 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531,
+ 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29,
+ 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535,
+ 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2,
+ 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29,
+ 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13,
+ 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535,
+ 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324,
+ 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4,
+ 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371,
+ 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, 65150,
+ 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533,
+ 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533,
+ 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2,
+ 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535,
+ 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544,
+ 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412,
+ 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783,
+ 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6,
+ 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2,
+ 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487,
+ 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
+ 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474,
+ 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167,
+ 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495,
+ 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13,
+ 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495,
+ 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4,
+ 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834,
+ 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, 64907,
+ 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25,
+ 42, 46, 50, 75, 113, 147, 281, 448, 657, 909,
+ 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169,
+ 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, 52058,
+ 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, 64547, 64924,
+ 65158, 65313, 65430, 65481, 65518, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65533, 65535, 0, 2, 65535, 0,
+ 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
+ 65535, 0, 2, 4, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 4, 65531, 65533, 65535, 0, 2, 4, 65531,
+ 65533, 65535, 0, 2, 4, 6, 65524, 65533, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65533,
+ 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0,
+ 2, 65533, 65535, 0, 2, 4, 65532, 65535, 0, 6,
+ 65523, 65535, 0, 2, 15, 65530, 65533, 65535, 0, 2,
+ 35, 65493, 65531, 65533, 65535, 0, 2, 4, 158, 65382,
+ 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 2, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 9, 65512, 65535, 0, 2,
+ 12, 65529, 65535, 0, 2, 73, 65434, 65533, 65535, 0,
+ 2, 240, 65343, 65533, 65535, 0, 2, 476, 65017, 65531,
+ 65533, 65535, 0, 2, 4, 1046, 64686, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 1870, 63898, 65529, 65531, 65533,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65532, 65535, 0, 6, 65533, 65535, 0, 6, 65523, 65535,
+ 0, 2, 65532, 65535, 0, 137, 65439, 65535, 0, 576,
+ 64899, 65533, 65535, 0, 2, 289, 65299, 65533, 65535, 0,
+ 2, 4, 6, 880, 64134, 65531, 65533, 65535, 0, 2,
+ 4, 1853, 63347, 65533, 65535, 0, 2, 6, 2516, 61762,
+ 65529, 65531, 65533, 65535, 0, 2, 4, 9, 3980, 61380,
+ 65503, 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8,
+ 10, 12, 61, 6393, 59859, 65466, 65527, 65529, 65531, 65533,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, 65532,
+ 65535, 0, 3, 65529, 65535, 0, 2, 65529, 65535, 0,
+ 61, 65453, 65535, 0, 234, 65313, 65535, 0, 503, 65138,
+ 65535, 0, 155, 65402, 65533, 65535, 0, 2, 1058, 64554,
+ 65533, 65535, 0, 2, 4, 3138, 62109, 65531, 65533, 65535,
+ 0, 2, 4, 2031, 63339, 65531, 65533, 65535, 0, 2,
+ 4, 6, 9, 4155, 60778, 65523, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 41, 6189, 59269, 65490, 65531, 65533, 65535,
+ 0, 2, 4, 6, 210, 8789, 57043, 65400, 65528, 65531,
+ 65533, 65535, 0, 2, 4, 6, 8, 26, 453, 10086,
+ 55499, 64948, 65483, 65524, 65527, 65529, 65531, 65533, 65535, 0,
+ 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
+ 114, 1014, 11202, 52670, 64226, 65356, 65503, 65514, 65523, 65525,
+ 65527, 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 15,
+ 65301, 65535, 0, 152, 64807, 65535, 0, 2, 3328, 63308,
+ 65535, 0, 2, 4050, 59730, 65533, 65535, 0, 2, 164,
+ 10564, 61894, 65529, 65535, 0, 15, 6712, 59831, 65076, 65532,
+ 65535, 0, 32, 7712, 57449, 65459, 65535, 0, 2, 210,
+ 7849, 53110, 65021, 65523, 65535, 0, 2, 12, 1081, 13883,
+ 48262, 62870, 65477, 65535, 0, 2, 88, 847, 6145, 37852,
+ 62012, 65454, 65533, 65535, 0, 9, 47, 207, 1823, 14522,
+ 45521, 61069, 64891, 65481, 65528, 65531, 65533, 65535, 0, 2,
+ 9, 488, 2881, 12758, 38703, 58412, 64420, 65410, 65533, 65535,
+ 0, 2, 4, 6, 61, 333, 1891, 6486, 19720, 43188,
+ 57547, 62472, 64796, 65421, 65497, 65523, 65529, 65531, 65533, 65535,
+ 0, 2, 4, 6, 8, 10, 12, 29, 117, 447,
+ 1528, 6138, 21242, 43133, 56495, 62432, 64746, 65362, 65500, 65529,
+ 65531, 65533, 65535, 0, 2, 18, 105, 301, 760, 1490,
+ 3472, 7568, 15002, 26424, 40330, 53029, 60048, 62964, 64274, 64890,
+ 65337, 65445, 65489, 65513, 65527, 65530, 65533, 65535, 0, 2,
+ 4, 6, 41, 102, 409, 853, 2031, 4316, 7302, 11328,
+ 16869, 24825, 34926, 43481, 50877, 56126, 59874, 62103, 63281, 63857,
+ 64166, 64675, 65382, 65522, 65531, 65533, 65535, 0, 2, 4,
+ 6, 8, 10, 12, 14, 16, 18, 29, 38, 53,
+ 58, 96, 181, 503, 1183, 2849, 5590, 8600, 11379, 13942,
+ 16478, 19453, 22638, 26039, 29411, 32921, 37596, 41433, 44998, 48560,
+ 51979, 55106, 57666, 59892, 61485, 62616, 63484, 64018, 64375, 64685,
+ 64924, 65076, 65278, 65395, 65471, 65509, 65529, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
+ 0, 65535, 0, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
+ 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 7,
+ 65519, 65535, 0, 2, 14, 65491, 65533, 65535, 0, 2,
+ 81, 65427, 65531, 65533, 65535, 0, 2, 4, 312, 65293,
+ 65528, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
+ 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
+ 65535, 0, 5, 65523, 65535, 0, 2, 65533, 65535, 0,
+ 7, 65526, 65535, 0, 46, 65464, 65533, 65535, 0, 2,
+ 120, 65309, 65533, 65535, 0, 2, 5, 362, 65097, 65533,
+ 65535, 0, 2, 18, 1164, 64785, 65528, 65531, 65533, 65535,
+ 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0,
+ 65535, 0, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65530, 65535,
+ 0, 2, 65523, 65535, 0, 69, 65477, 65535, 0, 141,
+ 65459, 65535, 0, 194, 65325, 65533, 65535, 0, 2, 543,
+ 64912, 65533, 65535, 0, 5, 1270, 64301, 65529, 65531, 65533,
+ 65535, 0, 2, 4, 12, 2055, 63538, 65508, 65531, 65533,
+ 65535, 0, 2, 7, 102, 3775, 61970, 65429, 65526, 65528,
+ 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2,
+ 65533, 65535, 0, 2, 65535, 0, 9, 65533, 65535, 0,
+ 25, 65512, 65535, 0, 2, 65533, 65535, 0, 44, 65480,
+ 65535, 0, 48, 65475, 65535, 0, 162, 65373, 65535, 0,
+ 637, 64806, 65533, 65535, 0, 2, 935, 64445, 65533, 65535,
+ 0, 2, 4, 1662, 64083, 65533, 65535, 0, 2, 12,
+ 3036, 62469, 65521, 65533, 65535, 0, 2, 120, 5405, 60468,
+ 65469, 65531, 65533, 65535, 0, 2, 4, 18, 254, 6663,
+ 58999, 65272, 65528, 65533, 65535, 0, 2, 4, 9, 12,
+ 67, 591, 8981, 56781, 64564, 65365, 65508, 65524, 65526, 65529,
+ 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 2, 65533,
+ 65535, 0, 9, 65526, 65535, 0, 14, 65503, 65535, 0,
+ 127, 65390, 65535, 0, 517, 64990, 65535, 0, 178, 65330,
+ 65535, 0, 2, 1055, 64533, 65533, 65535, 0, 2, 1558,
+ 63942, 65533, 65535, 0, 2, 2205, 63173, 65533, 65535, 0,
+ 25, 4493, 60862, 65505, 65533, 65535, 0, 2, 48, 5890,
+ 59442, 65482, 65533, 65535, 0, 2, 4, 127, 7532, 58191,
+ 65394, 65533, 65535, 0, 2, 5, 32, 550, 10388, 54924,
+ 65046, 65510, 65531, 65533, 65535, 0, 2, 4, 30, 150,
+ 1685, 14340, 51375, 63619, 65288, 65503, 65528, 65533, 65535, 0,
+ 2, 4, 6, 8, 28, 97, 473, 2692, 15407, 50020,
+ 62880, 65064, 65445, 65508, 65531, 65533, 65535, 0, 2, 4,
+ 12, 32, 79, 150, 372, 907, 2184, 5868, 18207, 45431,
+ 59856, 64031, 65096, 65401, 65481, 65507, 65521, 65523, 65525, 65527,
+ 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 182, 65491,
+ 65535, 0, 877, 64286, 65535, 0, 9, 2708, 63612, 65533,
+ 65535, 0, 2, 6038, 59532, 65535, 0, 2, 92, 5500,
+ 60539, 65533, 65535, 0, 268, 8908, 56512, 65385, 65535, 0,
+ 129, 13110, 52742, 65036, 65535, 0, 2, 806, 14003, 51929,
+ 64732, 65523, 65535, 0, 7, 92, 2667, 18159, 47678, 62610,
+ 65355, 65535, 0, 32, 1836, 19676, 48237, 61677, 64960, 65526,
+ 65535, 0, 21, 159, 967, 5668, 22782, 44709, 58317, 64020,
+ 65406, 65528, 65535, 0, 7, 162, 1838, 8328, 23929, 43014,
+ 56394, 63374, 65216, 65484, 65521, 65535, 0, 2, 4, 6,
+ 28, 268, 1120, 3613, 10688, 24185, 40989, 54917, 61684, 64510,
+ 65403, 65530, 65535, 0, 2, 16, 44, 139, 492, 1739,
+ 5313, 13558, 26766, 41566, 52446, 58937, 62815, 64480, 65201, 65454,
+ 65524, 65533, 65535, 0, 7, 25, 76, 263, 612, 1466,
+ 3325, 6832, 12366, 20152, 29466, 39255, 47360, 53506, 57740, 60726,
+ 62845, 64131, 64882, 65260, 65459, 65521, 65528, 65530, 65535, 0,
+ 2, 4, 14, 48, 136, 312, 653, 1240, 2369, 4327,
+ 7028, 10759, 15449, 21235, 28027, 35386, 42938, 49562, 54990, 59119,
+ 62086, 63916, 64863, 65249, 65445, 65493, 65523, 65535, 0, 2,
+ 4, 6, 8, 10, 12, 21, 83, 208, 409, 723,
+ 1152, 1868, 2951, 4463, 6460, 8979, 11831, 15195, 18863, 22657,
+ 26762, 30881, 34963, 39098, 43054, 47069, 50620, 53871, 56821, 59386,
+ 61340, 62670, 63512, 64023, 64429, 64750, 64944, 65126, 65279, 65366,
+ 65413, 65445, 65473, 65505, 65510, 65521, 65528, 65530, 65535};
+
+/* pointers to cdf tables for quantizer indices */
+const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[3][12] = {
+{WebRtcIsac_kQKltCdfGain +0 +0, WebRtcIsac_kQKltCdfGain +0 +8, WebRtcIsac_kQKltCdfGain +0 +22, WebRtcIsac_kQKltCdfGain +0 +32, WebRtcIsac_kQKltCdfGain +0 +48, WebRtcIsac_kQKltCdfGain +0 +60, WebRtcIsac_kQKltCdfGain +0 +81, WebRtcIsac_kQKltCdfGain +0 +95, WebRtcIsac_kQKltCdfGain +0 +128, WebRtcIsac_kQKltCdfGain +0 +152,
+WebRtcIsac_kQKltCdfGain +0 +210, WebRtcIsac_kQKltCdfGain +0 +264},
+{WebRtcIsac_kQKltCdfGain +404 +0, WebRtcIsac_kQKltCdfGain +404 +8, WebRtcIsac_kQKltCdfGain +404 +21, WebRtcIsac_kQKltCdfGain +404 +30, WebRtcIsac_kQKltCdfGain +404 +46, WebRtcIsac_kQKltCdfGain +404 +58, WebRtcIsac_kQKltCdfGain +404 +79, WebRtcIsac_kQKltCdfGain +404 +93, WebRtcIsac_kQKltCdfGain +404 +125, WebRtcIsac_kQKltCdfGain +404 +149,
+WebRtcIsac_kQKltCdfGain +404 +207, WebRtcIsac_kQKltCdfGain +404 +260},
+{WebRtcIsac_kQKltCdfGain +803 +0, WebRtcIsac_kQKltCdfGain +803 +8, WebRtcIsac_kQKltCdfGain +803 +22, WebRtcIsac_kQKltCdfGain +803 +31, WebRtcIsac_kQKltCdfGain +803 +48, WebRtcIsac_kQKltCdfGain +803 +60, WebRtcIsac_kQKltCdfGain +803 +81, WebRtcIsac_kQKltCdfGain +803 +96, WebRtcIsac_kQKltCdfGain +803 +129, WebRtcIsac_kQKltCdfGain +803 +154,
+WebRtcIsac_kQKltCdfGain +803 +212, WebRtcIsac_kQKltCdfGain +803 +268}};
+
+const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[3][108] = {
+{WebRtcIsac_kQKltCdfShape +0 +0, WebRtcIsac_kQKltCdfShape +0 +2, WebRtcIsac_kQKltCdfShape +0 +4, WebRtcIsac_kQKltCdfShape +0 +6, WebRtcIsac_kQKltCdfShape +0 +8, WebRtcIsac_kQKltCdfShape +0 +10, WebRtcIsac_kQKltCdfShape +0 +12, WebRtcIsac_kQKltCdfShape +0 +14, WebRtcIsac_kQKltCdfShape +0 +16, WebRtcIsac_kQKltCdfShape +0 +18,
+WebRtcIsac_kQKltCdfShape +0 +21, WebRtcIsac_kQKltCdfShape +0 +25, WebRtcIsac_kQKltCdfShape +0 +29, WebRtcIsac_kQKltCdfShape +0 +33, WebRtcIsac_kQKltCdfShape +0 +37, WebRtcIsac_kQKltCdfShape +0 +43, WebRtcIsac_kQKltCdfShape +0 +49, WebRtcIsac_kQKltCdfShape +0 +56, WebRtcIsac_kQKltCdfShape +0 +64, WebRtcIsac_kQKltCdfShape +0 +66,
+WebRtcIsac_kQKltCdfShape +0 +68, WebRtcIsac_kQKltCdfShape +0 +70, WebRtcIsac_kQKltCdfShape +0 +72, WebRtcIsac_kQKltCdfShape +0 +75, WebRtcIsac_kQKltCdfShape +0 +77, WebRtcIsac_kQKltCdfShape +0 +79, WebRtcIsac_kQKltCdfShape +0 +81, WebRtcIsac_kQKltCdfShape +0 +83, WebRtcIsac_kQKltCdfShape +0 +86, WebRtcIsac_kQKltCdfShape +0 +90,
+WebRtcIsac_kQKltCdfShape +0 +94, WebRtcIsac_kQKltCdfShape +0 +98, WebRtcIsac_kQKltCdfShape +0 +102, WebRtcIsac_kQKltCdfShape +0 +107, WebRtcIsac_kQKltCdfShape +0 +113, WebRtcIsac_kQKltCdfShape +0 +120, WebRtcIsac_kQKltCdfShape +0 +129, WebRtcIsac_kQKltCdfShape +0 +131, WebRtcIsac_kQKltCdfShape +0 +133, WebRtcIsac_kQKltCdfShape +0 +135,
+WebRtcIsac_kQKltCdfShape +0 +137, WebRtcIsac_kQKltCdfShape +0 +141, WebRtcIsac_kQKltCdfShape +0 +143, WebRtcIsac_kQKltCdfShape +0 +147, WebRtcIsac_kQKltCdfShape +0 +151, WebRtcIsac_kQKltCdfShape +0 +155, WebRtcIsac_kQKltCdfShape +0 +159, WebRtcIsac_kQKltCdfShape +0 +164, WebRtcIsac_kQKltCdfShape +0 +168, WebRtcIsac_kQKltCdfShape +0 +172,
+WebRtcIsac_kQKltCdfShape +0 +178, WebRtcIsac_kQKltCdfShape +0 +184, WebRtcIsac_kQKltCdfShape +0 +192, WebRtcIsac_kQKltCdfShape +0 +200, WebRtcIsac_kQKltCdfShape +0 +211, WebRtcIsac_kQKltCdfShape +0 +213, WebRtcIsac_kQKltCdfShape +0 +215, WebRtcIsac_kQKltCdfShape +0 +217, WebRtcIsac_kQKltCdfShape +0 +219, WebRtcIsac_kQKltCdfShape +0 +223,
+WebRtcIsac_kQKltCdfShape +0 +227, WebRtcIsac_kQKltCdfShape +0 +231, WebRtcIsac_kQKltCdfShape +0 +235, WebRtcIsac_kQKltCdfShape +0 +239, WebRtcIsac_kQKltCdfShape +0 +243, WebRtcIsac_kQKltCdfShape +0 +248, WebRtcIsac_kQKltCdfShape +0 +252, WebRtcIsac_kQKltCdfShape +0 +258, WebRtcIsac_kQKltCdfShape +0 +264, WebRtcIsac_kQKltCdfShape +0 +273,
+WebRtcIsac_kQKltCdfShape +0 +282, WebRtcIsac_kQKltCdfShape +0 +293, WebRtcIsac_kQKltCdfShape +0 +308, WebRtcIsac_kQKltCdfShape +0 +310, WebRtcIsac_kQKltCdfShape +0 +312, WebRtcIsac_kQKltCdfShape +0 +316, WebRtcIsac_kQKltCdfShape +0 +320, WebRtcIsac_kQKltCdfShape +0 +324, WebRtcIsac_kQKltCdfShape +0 +328, WebRtcIsac_kQKltCdfShape +0 +332,
+WebRtcIsac_kQKltCdfShape +0 +336, WebRtcIsac_kQKltCdfShape +0 +341, WebRtcIsac_kQKltCdfShape +0 +347, WebRtcIsac_kQKltCdfShape +0 +354, WebRtcIsac_kQKltCdfShape +0 +360, WebRtcIsac_kQKltCdfShape +0 +368, WebRtcIsac_kQKltCdfShape +0 +378, WebRtcIsac_kQKltCdfShape +0 +388, WebRtcIsac_kQKltCdfShape +0 +400, WebRtcIsac_kQKltCdfShape +0 +418,
+WebRtcIsac_kQKltCdfShape +0 +445, WebRtcIsac_kQKltCdfShape +0 +447, WebRtcIsac_kQKltCdfShape +0 +451, WebRtcIsac_kQKltCdfShape +0 +455, WebRtcIsac_kQKltCdfShape +0 +461, WebRtcIsac_kQKltCdfShape +0 +468, WebRtcIsac_kQKltCdfShape +0 +474, WebRtcIsac_kQKltCdfShape +0 +480, WebRtcIsac_kQKltCdfShape +0 +486, WebRtcIsac_kQKltCdfShape +0 +495,
+WebRtcIsac_kQKltCdfShape +0 +505, WebRtcIsac_kQKltCdfShape +0 +516, WebRtcIsac_kQKltCdfShape +0 +528, WebRtcIsac_kQKltCdfShape +0 +543, WebRtcIsac_kQKltCdfShape +0 +564, WebRtcIsac_kQKltCdfShape +0 +583, WebRtcIsac_kQKltCdfShape +0 +608, WebRtcIsac_kQKltCdfShape +0 +635},
+{WebRtcIsac_kQKltCdfShape +686 +0, WebRtcIsac_kQKltCdfShape +686 +2, WebRtcIsac_kQKltCdfShape +686 +4, WebRtcIsac_kQKltCdfShape +686 +6, WebRtcIsac_kQKltCdfShape +686 +8, WebRtcIsac_kQKltCdfShape +686 +11, WebRtcIsac_kQKltCdfShape +686 +13, WebRtcIsac_kQKltCdfShape +686 +15, WebRtcIsac_kQKltCdfShape +686 +17, WebRtcIsac_kQKltCdfShape +686 +20,
+WebRtcIsac_kQKltCdfShape +686 +23, WebRtcIsac_kQKltCdfShape +686 +27, WebRtcIsac_kQKltCdfShape +686 +31, WebRtcIsac_kQKltCdfShape +686 +35, WebRtcIsac_kQKltCdfShape +686 +40, WebRtcIsac_kQKltCdfShape +686 +44, WebRtcIsac_kQKltCdfShape +686 +50, WebRtcIsac_kQKltCdfShape +686 +56, WebRtcIsac_kQKltCdfShape +686 +63, WebRtcIsac_kQKltCdfShape +686 +65,
+WebRtcIsac_kQKltCdfShape +686 +67, WebRtcIsac_kQKltCdfShape +686 +69, WebRtcIsac_kQKltCdfShape +686 +71, WebRtcIsac_kQKltCdfShape +686 +73, WebRtcIsac_kQKltCdfShape +686 +75, WebRtcIsac_kQKltCdfShape +686 +77, WebRtcIsac_kQKltCdfShape +686 +79, WebRtcIsac_kQKltCdfShape +686 +82, WebRtcIsac_kQKltCdfShape +686 +85, WebRtcIsac_kQKltCdfShape +686 +89,
+WebRtcIsac_kQKltCdfShape +686 +93, WebRtcIsac_kQKltCdfShape +686 +97, WebRtcIsac_kQKltCdfShape +686 +102, WebRtcIsac_kQKltCdfShape +686 +106, WebRtcIsac_kQKltCdfShape +686 +112, WebRtcIsac_kQKltCdfShape +686 +119, WebRtcIsac_kQKltCdfShape +686 +127, WebRtcIsac_kQKltCdfShape +686 +129, WebRtcIsac_kQKltCdfShape +686 +131, WebRtcIsac_kQKltCdfShape +686 +133,
+WebRtcIsac_kQKltCdfShape +686 +135, WebRtcIsac_kQKltCdfShape +686 +137, WebRtcIsac_kQKltCdfShape +686 +139, WebRtcIsac_kQKltCdfShape +686 +142, WebRtcIsac_kQKltCdfShape +686 +146, WebRtcIsac_kQKltCdfShape +686 +150, WebRtcIsac_kQKltCdfShape +686 +154, WebRtcIsac_kQKltCdfShape +686 +158, WebRtcIsac_kQKltCdfShape +686 +162, WebRtcIsac_kQKltCdfShape +686 +167,
+WebRtcIsac_kQKltCdfShape +686 +173, WebRtcIsac_kQKltCdfShape +686 +179, WebRtcIsac_kQKltCdfShape +686 +186, WebRtcIsac_kQKltCdfShape +686 +194, WebRtcIsac_kQKltCdfShape +686 +205, WebRtcIsac_kQKltCdfShape +686 +207, WebRtcIsac_kQKltCdfShape +686 +209, WebRtcIsac_kQKltCdfShape +686 +211, WebRtcIsac_kQKltCdfShape +686 +214, WebRtcIsac_kQKltCdfShape +686 +218,
+WebRtcIsac_kQKltCdfShape +686 +222, WebRtcIsac_kQKltCdfShape +686 +226, WebRtcIsac_kQKltCdfShape +686 +230, WebRtcIsac_kQKltCdfShape +686 +234, WebRtcIsac_kQKltCdfShape +686 +238, WebRtcIsac_kQKltCdfShape +686 +242, WebRtcIsac_kQKltCdfShape +686 +247, WebRtcIsac_kQKltCdfShape +686 +253, WebRtcIsac_kQKltCdfShape +686 +262, WebRtcIsac_kQKltCdfShape +686 +269,
+WebRtcIsac_kQKltCdfShape +686 +278, WebRtcIsac_kQKltCdfShape +686 +289, WebRtcIsac_kQKltCdfShape +686 +305, WebRtcIsac_kQKltCdfShape +686 +307, WebRtcIsac_kQKltCdfShape +686 +309, WebRtcIsac_kQKltCdfShape +686 +311, WebRtcIsac_kQKltCdfShape +686 +315, WebRtcIsac_kQKltCdfShape +686 +319, WebRtcIsac_kQKltCdfShape +686 +323, WebRtcIsac_kQKltCdfShape +686 +327,
+WebRtcIsac_kQKltCdfShape +686 +331, WebRtcIsac_kQKltCdfShape +686 +335, WebRtcIsac_kQKltCdfShape +686 +340, WebRtcIsac_kQKltCdfShape +686 +346, WebRtcIsac_kQKltCdfShape +686 +354, WebRtcIsac_kQKltCdfShape +686 +362, WebRtcIsac_kQKltCdfShape +686 +374, WebRtcIsac_kQKltCdfShape +686 +384, WebRtcIsac_kQKltCdfShape +686 +396, WebRtcIsac_kQKltCdfShape +686 +413,
+WebRtcIsac_kQKltCdfShape +686 +439, WebRtcIsac_kQKltCdfShape +686 +442, WebRtcIsac_kQKltCdfShape +686 +446, WebRtcIsac_kQKltCdfShape +686 +450, WebRtcIsac_kQKltCdfShape +686 +455, WebRtcIsac_kQKltCdfShape +686 +461, WebRtcIsac_kQKltCdfShape +686 +468, WebRtcIsac_kQKltCdfShape +686 +475, WebRtcIsac_kQKltCdfShape +686 +481, WebRtcIsac_kQKltCdfShape +686 +489,
+WebRtcIsac_kQKltCdfShape +686 +498, WebRtcIsac_kQKltCdfShape +686 +508, WebRtcIsac_kQKltCdfShape +686 +522, WebRtcIsac_kQKltCdfShape +686 +534, WebRtcIsac_kQKltCdfShape +686 +554, WebRtcIsac_kQKltCdfShape +686 +577, WebRtcIsac_kQKltCdfShape +686 +602, WebRtcIsac_kQKltCdfShape +686 +631},
+{WebRtcIsac_kQKltCdfShape +1368 +0, WebRtcIsac_kQKltCdfShape +1368 +2, WebRtcIsac_kQKltCdfShape +1368 +4, WebRtcIsac_kQKltCdfShape +1368 +6, WebRtcIsac_kQKltCdfShape +1368 +8, WebRtcIsac_kQKltCdfShape +1368 +10, WebRtcIsac_kQKltCdfShape +1368 +12, WebRtcIsac_kQKltCdfShape +1368 +14, WebRtcIsac_kQKltCdfShape +1368 +16, WebRtcIsac_kQKltCdfShape +1368 +20,
+WebRtcIsac_kQKltCdfShape +1368 +24, WebRtcIsac_kQKltCdfShape +1368 +28, WebRtcIsac_kQKltCdfShape +1368 +32, WebRtcIsac_kQKltCdfShape +1368 +36, WebRtcIsac_kQKltCdfShape +1368 +40, WebRtcIsac_kQKltCdfShape +1368 +44, WebRtcIsac_kQKltCdfShape +1368 +50, WebRtcIsac_kQKltCdfShape +1368 +57, WebRtcIsac_kQKltCdfShape +1368 +65, WebRtcIsac_kQKltCdfShape +1368 +67,
+WebRtcIsac_kQKltCdfShape +1368 +69, WebRtcIsac_kQKltCdfShape +1368 +71, WebRtcIsac_kQKltCdfShape +1368 +73, WebRtcIsac_kQKltCdfShape +1368 +75, WebRtcIsac_kQKltCdfShape +1368 +77, WebRtcIsac_kQKltCdfShape +1368 +79, WebRtcIsac_kQKltCdfShape +1368 +81, WebRtcIsac_kQKltCdfShape +1368 +85, WebRtcIsac_kQKltCdfShape +1368 +89, WebRtcIsac_kQKltCdfShape +1368 +93,
+WebRtcIsac_kQKltCdfShape +1368 +97, WebRtcIsac_kQKltCdfShape +1368 +101, WebRtcIsac_kQKltCdfShape +1368 +105, WebRtcIsac_kQKltCdfShape +1368 +110, WebRtcIsac_kQKltCdfShape +1368 +116, WebRtcIsac_kQKltCdfShape +1368 +123, WebRtcIsac_kQKltCdfShape +1368 +132, WebRtcIsac_kQKltCdfShape +1368 +134, WebRtcIsac_kQKltCdfShape +1368 +136, WebRtcIsac_kQKltCdfShape +1368 +138,
+WebRtcIsac_kQKltCdfShape +1368 +141, WebRtcIsac_kQKltCdfShape +1368 +143, WebRtcIsac_kQKltCdfShape +1368 +146, WebRtcIsac_kQKltCdfShape +1368 +150, WebRtcIsac_kQKltCdfShape +1368 +154, WebRtcIsac_kQKltCdfShape +1368 +158, WebRtcIsac_kQKltCdfShape +1368 +162, WebRtcIsac_kQKltCdfShape +1368 +166, WebRtcIsac_kQKltCdfShape +1368 +170, WebRtcIsac_kQKltCdfShape +1368 +174,
+WebRtcIsac_kQKltCdfShape +1368 +179, WebRtcIsac_kQKltCdfShape +1368 +185, WebRtcIsac_kQKltCdfShape +1368 +193, WebRtcIsac_kQKltCdfShape +1368 +203, WebRtcIsac_kQKltCdfShape +1368 +214, WebRtcIsac_kQKltCdfShape +1368 +216, WebRtcIsac_kQKltCdfShape +1368 +218, WebRtcIsac_kQKltCdfShape +1368 +220, WebRtcIsac_kQKltCdfShape +1368 +224, WebRtcIsac_kQKltCdfShape +1368 +227,
+WebRtcIsac_kQKltCdfShape +1368 +231, WebRtcIsac_kQKltCdfShape +1368 +235, WebRtcIsac_kQKltCdfShape +1368 +239, WebRtcIsac_kQKltCdfShape +1368 +243, WebRtcIsac_kQKltCdfShape +1368 +247, WebRtcIsac_kQKltCdfShape +1368 +251, WebRtcIsac_kQKltCdfShape +1368 +256, WebRtcIsac_kQKltCdfShape +1368 +262, WebRtcIsac_kQKltCdfShape +1368 +269, WebRtcIsac_kQKltCdfShape +1368 +277,
+WebRtcIsac_kQKltCdfShape +1368 +286, WebRtcIsac_kQKltCdfShape +1368 +297, WebRtcIsac_kQKltCdfShape +1368 +315, WebRtcIsac_kQKltCdfShape +1368 +317, WebRtcIsac_kQKltCdfShape +1368 +319, WebRtcIsac_kQKltCdfShape +1368 +323, WebRtcIsac_kQKltCdfShape +1368 +327, WebRtcIsac_kQKltCdfShape +1368 +331, WebRtcIsac_kQKltCdfShape +1368 +335, WebRtcIsac_kQKltCdfShape +1368 +339,
+WebRtcIsac_kQKltCdfShape +1368 +343, WebRtcIsac_kQKltCdfShape +1368 +349, WebRtcIsac_kQKltCdfShape +1368 +355, WebRtcIsac_kQKltCdfShape +1368 +361, WebRtcIsac_kQKltCdfShape +1368 +368, WebRtcIsac_kQKltCdfShape +1368 +376, WebRtcIsac_kQKltCdfShape +1368 +385, WebRtcIsac_kQKltCdfShape +1368 +397, WebRtcIsac_kQKltCdfShape +1368 +411, WebRtcIsac_kQKltCdfShape +1368 +429,
+WebRtcIsac_kQKltCdfShape +1368 +456, WebRtcIsac_kQKltCdfShape +1368 +459, WebRtcIsac_kQKltCdfShape +1368 +463, WebRtcIsac_kQKltCdfShape +1368 +467, WebRtcIsac_kQKltCdfShape +1368 +473, WebRtcIsac_kQKltCdfShape +1368 +478, WebRtcIsac_kQKltCdfShape +1368 +485, WebRtcIsac_kQKltCdfShape +1368 +491, WebRtcIsac_kQKltCdfShape +1368 +497, WebRtcIsac_kQKltCdfShape +1368 +505,
+WebRtcIsac_kQKltCdfShape +1368 +514, WebRtcIsac_kQKltCdfShape +1368 +523, WebRtcIsac_kQKltCdfShape +1368 +535, WebRtcIsac_kQKltCdfShape +1368 +548, WebRtcIsac_kQKltCdfShape +1368 +565, WebRtcIsac_kQKltCdfShape +1368 +585, WebRtcIsac_kQKltCdfShape +1368 +611, WebRtcIsac_kQKltCdfShape +1368 +640}};
+
+/* code length for all coefficients using different models */
+const double WebRtcIsac_kQKltCodeLenGain[392] = {
+ 12.29956028, 7.83007500, 4.25642781, 0.17489215, 4.27591254, 7.66908312, 10.47643804, 11.91253716, 9.03421572, 8.57373525,
+ 6.73555740, 5.41409855, 3.65237863, 0.42623449, 3.22418399, 5.68145719, 7.14201900, 8.20558413, 9.34178852, 13.00000000,
+ 13.00000000, 9.47643804, 6.71459778, 3.55472644, 0.28457419, 3.70652835, 7.41128536, 11.60768258, 14.00000000, 12.29956028,
+ 9.29956028, 8.02845645, 6.54878889, 5.07667251, 2.87749552, 0.65310542, 3.11316029, 4.87911416, 5.89540125, 6.76398581,
+ 7.70537925, 9.67807191, 11.14201900, 14.00000000, 14.00000000, 12.29956028, 9.69621925, 7.70537925, 5.49915812, 2.67441345,
+ 0.63381441, 2.74999773, 5.76877882, 8.41503750, 10.75207249, 14.00000000, 9.60768258, 8.85025288, 8.09913319, 7.09010692,
+ 6.36337538, 5.46667027, 4.51618422, 2.64189829, 1.21843537, 2.43823474, 3.89409149, 4.99718498, 6.00989604, 6.78325414,
+ 7.39637366, 8.76159526, 10.60768258, 11.67807191, 12.29956028, 9.54056838, 7.95560588, 5.63040265, 3.81264793, 2.18521728,
+ 1.33727600, 2.37909290, 3.94981123, 5.52426657, 7.32051990, 8.84012866, 11.35614381, 13.00000000, 12.29956028, 11.35614381,
+ 11.35614381, 10.24511250, 9.12963528, 8.80032766, 7.97763219, 7.83007500, 7.09913319, 6.68711704, 6.41879942, 5.74379131,
+ 5.25096862, 4.43061904, 3.54492969, 2.72664147, 2.16306204, 2.71142226, 3.34357514, 4.07444556, 4.95151313, 5.68145719,
+ 6.12041675, 6.55085135, 7.00282052, 7.55705650, 8.44541115, 9.41503750, 10.91253716, 11.60768258, 14.00000000, 14.00000000,
+ 13.00000000, 11.60768258, 10.47643804, 8.80032766, 7.38161450, 6.52426657, 5.47447919, 4.53749773, 3.92719747, 3.41058292,
+ 2.88495635, 2.79344346, 2.85805254, 3.18261657, 3.57216340, 4.08225499, 4.74438125, 5.51215997, 6.30477171, 7.15450995,
+ 8.28575448, 9.69621925, 11.60768258, 13.00000000, 13.67807191, 15.00000000, 15.00000000, 14.00000000, 12.41503750, 11.29956028,
+ 12.41503750, 11.60768258, 10.11735695, 9.47643804, 9.12963528, 9.41503750, 8.23181568, 7.97763219, 7.82507432, 7.50814690,
+ 7.33466408, 6.99157138, 6.95834085, 6.80524315, 6.47447919, 6.35614381, 6.02128954, 5.71459778, 5.58109327, 5.05821876,
+ 4.94539568, 4.59220115, 4.27591254, 4.01522554, 3.89376424, 3.83760867, 3.73321346, 3.74674342, 3.90493270, 4.18942837,
+ 4.33599724, 4.42446075, 4.81760565, 5.07667251, 5.54570071, 5.95697272, 6.46667027, 6.91253716, 7.33466408, 7.82507432,
+ 8.05163277, 9.12963528, 9.02272008, 9.77118131, 9.60768258, 10.11735695, 12.00000000, 12.83007500, 15.00000000, 15.00000000,
+ 14.00000000, 15.00000000, 11.75207249, 12.29956028, 10.95560588, 9.93391081, 9.02272008, 8.00564656, 7.82507432, 7.28919357,
+ 6.77599833, 6.56745810, 6.19910010, 6.23347109, 5.67694524, 5.41879942, 4.96039548, 4.88170777, 4.60768258, 4.52883287,
+ 4.28876323, 4.17583679, 4.21332197, 4.14474217, 4.16119001, 4.12809476, 4.18501706, 4.28489599, 4.35299136, 4.60286019,
+ 4.63040265, 4.81017544, 5.00989604, 5.33822190, 5.43489792, 5.84644797, 6.13272126, 6.75444729, 7.36337538, 7.96108101,
+ 8.54056838, 9.28575448, 9.12963528, 9.47643804, 10.75207249, 12.29956028, 12.41503750, 11.14201900, 12.83007500, 15.00000000,
+ 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 14.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000,
+ 13.67807191, 12.41503750, 11.91253716, 11.60768258, 12.29956028, 13.00000000, 12.29956028, 10.95560588, 10.60768258, 9.77118131,
+ 10.02272008, 10.11735695, 10.14201900, 10.75207249, 10.75207249, 9.69621925, 9.47643804, 9.75207249, 8.72387559, 8.47643804,
+ 8.68711704, 8.40754296, 8.20558413, 8.26529038, 7.84518189, 7.89147554, 7.64685317, 7.41128536, 7.33466408, 7.42635281,
+ 7.24845594, 7.11430363, 7.07518750, 7.07518750, 6.70537925, 6.64906082, 6.73555740, 6.62931259, 6.50015411, 6.26190774,
+ 6.36337538, 6.19264508, 5.95151313, 6.08860801, 5.91253716, 5.83007500, 5.68145719, 5.67244736, 5.82632286, 6.00282052,
+ 5.65348627, 5.74970158, 5.87846648, 5.69052365, 5.64464890, 5.58531476, 5.81512466, 5.57688409, 5.87329553, 5.62170514,
+ 5.74851759, 5.81017544, 5.64464890, 5.76398581, 5.60339522, 5.69507833, 5.84139031, 5.68711704, 5.73908047, 5.84139031,
+ 5.71459778, 5.96245305, 5.82632286, 5.89540125, 6.08860801, 6.12041675, 6.13272126, 6.30477171, 6.24177679, 6.28232358,
+ 6.29091619, 6.53239445, 6.81512466, 6.72620440, 6.65792533, 6.84518189, 7.10215454, 7.44157929, 7.57793523, 7.59485854,
+ 7.66460965, 8.05163277, 8.00564656, 8.43775758, 8.10518224, 8.28575448, 8.77118131, 8.23181568, 8.29264087, 8.20558413,
+ 8.34894831, 8.89147554, 8.40754296, 8.61629571, 8.64244800, 8.61629571, 8.93391081, 8.50814690, 9.02272008, 8.68711704,
+ 8.65127185, 9.41503750, 9.11735695, 9.85025288, 10.24511250, 10.60768258, 10.47643804, 11.60768258, 11.35614381, 12.29956028,
+ 15.00000000, 15.00000000, 12.29956028, 13.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 14.00000000, 12.83007500,
+ 15.00000000, 15.00000000};
+
+const double WebRtcIsac_kQKltCodeLenShape[578] = {
+ 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000,
+ 0.00011007, 13.00000000, 0.00066056, 11.60768258, 11.14201900, 0.00185034, 10.24511250, 9.08113676, 0.00480700, 9.41503750,
+ 8.09913319, 0.01084946, 8.02845645, 13.00000000, 6.40941295, 0.02926496, 6.95560588, 13.00000000, 13.00000000, 6.21864029,
+ 0.03861814, 6.30477171, 12.29956028, 11.14201900, 4.66964328, 0.12158980, 4.62604734, 10.91253716, 14.00000000, 11.35614381,
+ 8.76159526, 3.89671219, 0.21179147, 3.99472634, 8.02272008, 13.00000000, 0.00002201, 0.00002201, 0.00002201, 0.00002201,
+ 14.00000000, 0.00011007, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000, 0.00011007, 10.95560588, 0.00147568,
+ 10.95560588, 10.24511250, 0.00240150, 10.24511250, 8.02845645, 0.01056115, 8.17982104, 6.74497143, 0.02381629, 7.15137706,
+ 5.68598330, 0.05650076, 5.72970467, 13.00000000, 14.00000000, 5.18221688, 0.07697435, 5.37611851, 11.91253716, 9.54056838,
+ 3.92853764, 0.22143514, 3.76428491, 9.28575448, 14.00000000, 11.35614381, 7.37794818, 3.30585980, 0.37001735, 3.18521728,
+ 6.88886433, 11.91253716, 14.00000000, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000, 0.00019814, 14.00000000,
+ 0.00002201, 13.00000000, 0.00028621, 14.00000000, 14.00000000, 0.00028621, 13.00000000, 11.91253716, 0.00094690, 11.35614381,
+ 10.60768258, 0.00213692, 10.24511250, 8.37794818, 0.00863317, 8.43775758, 14.00000000, 7.41128536, 0.01698415, 7.42635281,
+ 6.02702021, 0.04514485, 6.01558154, 13.00000000, 5.05090284, 0.09207659, 4.98877274, 15.00000000, 10.75207249, 4.41081703,
+ 0.15733047, 4.17424617, 11.60768258, 14.00000000, 15.00000000, 10.06926266, 3.74320161, 0.23091117, 3.81141115, 10.02272008,
+ 11.91253716, 6.90196792, 2.92703003, 0.46874039, 2.93846004, 7.26190774, 11.60768258, 13.00000000, 10.21864029, 7.95560588,
+ 5.19345038, 2.40520888, 0.77554605, 2.56628417, 5.41409855, 8.81017544, 12.29956028, 0.00002201, 0.00002201, 0.00002201,
+ 0.00002201, 11.91253716, 0.00068259, 12.29956028, 10.11735695, 0.00233535, 10.47643804, 10.35614381, 0.00140957, 12.29956028,
+ 10.24511250, 0.00222511, 10.47643804, 7.72387559, 0.01475842, 7.52426657, 6.73555740, 0.02924249, 6.55085135, 14.00000000,
+ 5.45021533, 0.06930886, 5.38345116, 4.55913083, 0.11289841, 4.92853764, 11.60768258, 4.07148162, 0.19798859, 3.87200568,
+ 13.00000000, 9.60768258, 3.31393725, 0.29937064, 3.57321111, 9.35614381, 14.00000000, 8.02845645, 3.08542800, 0.45503557,
+ 2.80678268, 7.47643804, 12.29956028, 14.00000000, 14.00000000, 14.00000000, 6.83509307, 2.69166097, 0.61673447, 2.51266238,
+ 6.84771516, 11.91253716, 11.14201900, 7.72387559, 4.61899789, 2.17136763, 1.03592993, 2.28586183, 4.86814304, 7.78568088,
+ 10.91253716, 14.00000000, 15.00000000, 11.04580369, 9.28575448, 7.26190774, 5.21946023, 3.61575588, 2.18431651, 1.45666604,
+ 2.33566383, 3.85470467, 5.46181107, 7.30651304, 8.89147554, 11.91253716, 0.00002201, 0.00002201, 15.00000000, 0.00011007,
+ 15.00000000, 10.47643804, 0.00149771, 11.60768258, 7.30651304, 0.01903486, 7.20558413, 6.92318440, 0.02587674, 6.71459778,
+ 7.28919357, 0.01756340, 7.45696818, 6.06926266, 0.03841465, 6.45890338, 4.46035649, 0.13896157, 4.43204392, 14.00000000,
+ 14.00000000, 4.09010692, 0.19672654, 3.86653665, 13.00000000, 14.00000000, 11.35614381, 3.45117809, 0.27340929, 3.64878468,
+ 9.93391081, 9.60768258, 3.30585980, 0.35178287, 3.15607895, 9.12963528, 12.29956028, 7.93391081, 2.82180202, 0.53064436,
+ 2.68258739, 7.42635281, 11.14201900, 15.00000000, 10.06926266, 6.15294265, 2.55861197, 0.69308389, 2.48602573, 5.90592231,
+ 9.47643804, 13.00000000, 11.14201900, 8.20558413, 5.21618324, 2.34973357, 1.00379135, 2.09611815, 4.80278331, 7.82507432,
+ 11.91253716, 12.29956028, 8.98877274, 6.75444729, 4.23886435, 2.19859476, 1.29528579, 2.06032897, 4.06704711, 6.76398581,
+ 9.34178852, 13.00000000, 12.29956028, 10.14201900, 8.40754296, 6.16710999, 4.64850859, 3.28768796, 2.22892326, 1.78568088,
+ 2.35209193, 3.45141888, 4.91121176, 6.45117809, 8.20558413, 10.35614381, 13.00000000, 12.83007500, 14.00000000, 14.00000000,
+ 12.83007500, 11.35614381, 9.85025288, 9.41503750, 8.07518750, 7.34894831, 6.25516616, 5.25096862, 4.52477322, 3.68513357,
+ 3.06152306, 2.64011320, 2.54608637, 2.95765662, 3.44445223, 4.01665007, 4.63258525, 5.27633906, 6.24678325, 7.05163277,
+ 8.02845645, 9.60768258, 10.79054663, 12.83007500, 14.00000000, 0.00002201, 15.00000000, 0.00011007, 15.00000000, 5.82009091,
+ 0.02710994, 10.11735695, 15.00000000, 3.30346709, 0.17317845, 6.41128536, 15.00000000, 15.00000000, 10.83007500, 2.72897475,
+ 0.35935964, 3.85853144, 13.00000000, 15.00000000, 3.72766182, 0.17268211, 4.74674342, 15.00000000, 15.00000000, 4.40754296,
+ 0.11993823, 4.93997965, 15.00000000, 15.00000000, 4.70193743, 0.22110152, 3.27591254, 11.35614381, 15.00000000, 12.54056838,
+ 6.94743195, 2.99157138, 0.62079088, 2.25338071, 7.41128536, 15.00000000, 15.00000000, 11.47643804, 6.04580369, 2.80232255,
+ 0.98380109, 1.70502034, 5.15607895, 10.75207249, 13.00000000, 15.00000000, 15.00000000, 11.35614381, 6.44157929, 2.37955105,
+ 0.74567258, 2.44181848, 6.07667251, 10.75207249, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 9.44541115, 5.25432568,
+ 1.97815248, 0.94086682, 2.57885561, 5.17265730, 8.72387559, 14.00000000, 15.00000000, 15.00000000, 15.00000000, 10.67807191,
+ 7.59485854, 4.91187431, 2.91934900, 1.32321648, 1.50614455, 3.85215911, 7.15137706, 10.60768258, 14.00000000, 15.00000000,
+ 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 11.09310940, 8.50814690, 6.45117809, 3.94675287,
+ 2.16730774, 1.37674720, 2.10215454, 3.97278511, 6.34178852, 8.50814690, 10.32757466, 14.00000000, 14.00000000, 15.00000000,
+ 15.00000000, 15.00000000, 14.00000000, 10.02272008, 9.41503750, 7.03710399, 5.13349379, 3.61683574, 2.47999526, 1.82788507,
+ 2.00945280, 3.08020557, 4.69793233, 6.64025044, 8.32051990, 10.91253716, 14.00000000, 15.00000000, 15.00000000, 15.00000000,
+ 15.00000000, 15.00000000, 15.00000000, 14.41503750, 9.41503750, 7.10215454, 5.60768258, 4.44014496, 3.30064444, 2.56941678,
+ 2.28103909, 2.42694459, 2.93206152, 3.83383692, 4.75207249, 6.02702021, 7.16394964, 9.35614381, 11.14201900, 12.29956028,
+ 15.00000000, 15.00000000, 14.00000000, 15.00000000, 15.00000000, 10.11735695, 8.85025288, 7.02558541, 5.86300889, 4.79726396,
+ 3.95117259, 3.44925321, 2.96348293, 2.88219459, 2.89671219, 3.10518224, 3.50065237, 4.05748549, 4.55291677, 5.23016216,
+ 5.79420675, 6.39452048, 6.91253716, 7.97763219, 8.32051990, 9.35614381, 10.60768258, 11.60768258, 15.00000000, 15.00000000,
+ 12.83007500, 12.41503750, 11.91253716, 14.00000000, 14.00000000, 11.35614381, 10.75207249, 10.91253716, 8.93391081, 8.61629571,
+ 8.29264087, 8.02272008, 7.89147554, 7.33466408, 7.41128536, 6.71459778, 5.92853764, 5.31964048, 4.97693875, 4.70308379,
+ 4.43632704, 4.34357514, 4.30825648, 4.26529038, 4.19465917, 4.27420773, 4.22315577, 4.08262792, 4.06667818, 4.16119001,
+ 4.08860801, 4.26698468, 4.39128315, 4.71517590, 5.10442472, 5.50714538, 5.81017544, 6.15922208};
+
+/* left KLT transforms */
+const double WebRtcIsac_kKltT1Gain[3][4] = {
+{-0.79742827, 0.60341375, 0.60341375, 0.79742827},
+{-0.81372390, 0.58125159, 0.58125159, 0.81372390},
+{-0.71832547, 0.69570721, 0.69570721, 0.71832547}};
+
+const double WebRtcIsac_kKltT1Shape[3][324] = {
+{ 0.00159597, 0.00049320, 0.00513821, 0.00021066, 0.01338581, -0.00422367, -0.00272072, 0.00935107, 0.02047622, 0.02691189,
+ 0.00478236, 0.03969702, 0.00886698, 0.04877604, -0.10898362, -0.05930891, -0.03415047, 0.98889721, 0.00293558, -0.00035282,
+ 0.01156321, -0.00195341, -0.00937631, 0.01052213, -0.02551163, 0.01644059, 0.03189927, 0.07754773, -0.08742313, -0.03026338,
+ 0.05136248, -0.14395974, 0.17725040, 0.22664856, 0.93380230, 0.07076411, 0.00557890, -0.00222834, 0.01377569, 0.01466808,
+ 0.02847361, -0.00603178, 0.02382480, -0.01210452, 0.03797267, -0.02371480, 0.11260335, -0.07366682, 0.00453436, -0.04136941,
+-0.07912843, -0.95031418, 0.25295337, -0.05302216, -0.00617554, -0.00044040, -0.00653778, 0.01097838, 0.01529174, 0.01374431,
+-0.00748512, -0.00020034, 0.02432713, 0.11101570, -0.08556891, 0.09282249, -0.01029446, 0.67556443, -0.67454300, 0.06910063,
+ 0.20866865, -0.10318050, 0.00932175, 0.00524058, 0.00803610, -0.00594676, -0.01082578, 0.01069906, 0.00546768, 0.01565291,
+ 0.06816200, 0.10201227, 0.16812734, 0.22984074, 0.58213170, -0.54138651, -0.51379962, 0.06847390, -0.01920037, -0.04592324,
+-0.00467394, 0.00328858, 0.00377424, -0.00987448, 0.08222096, -0.00377301, 0.04551941, -0.02592517, 0.16317082, 0.13077530,
+ 0.22702921, -0.31215289, -0.69645962, -0.38047101, -0.39339411, 0.11124777, 0.02508035, -0.00708074, 0.00400344, 0.00040331,
+ 0.01142402, 0.01725406, 0.01635170, 0.14285366, 0.03949233, -0.05905676, 0.05877154, -0.17497577, -0.32479440, 0.80754464,
+-0.38085603, -0.17055430, -0.03168622, -0.07531451, 0.02942002, -0.02148095, -0.00754114, -0.00322372, 0.00567812, -0.01701521,
+-0.12358320, 0.11473564, 0.09070136, 0.06533068, -0.22560802, 0.19209022, 0.81605094, 0.36592275, -0.09919829, 0.16667122,
+ 0.16300725, 0.04803807, 0.06739263, -0.00156752, -0.01685302, -0.00905240, -0.02297836, -0.00469939, 0.06310613, -0.16391930,
+ 0.10919511, 0.12529293, 0.85581322, -0.32145522, 0.24539076, 0.07181839, 0.07289591, 0.14066759, 0.10406711, 0.05815518,
+ 0.01072680, -0.00759339, 0.00053486, -0.00044865, 0.03407361, 0.01645348, 0.08758579, 0.27722240, 0.53665485, -0.74853376,
+-0.01118192, -0.19805430, 0.06130619, -0.09675299, 0.08978480, 0.03405255, -0.00706867, 0.05102045, 0.03250746, 0.01849966,
+-0.01216314, -0.01184187, -0.01579288, 0.00114807, 0.11376166, 0.88342114, -0.36425379, 0.13863190, 0.12524180, -0.13553892,
+ 0.04715856, -0.12341103, 0.04531568, 0.01899360, -0.00206897, 0.00567768, -0.01444163, 0.00411946, -0.00855896, 0.00381663,
+-0.01664861, -0.05534280, 0.21328278, 0.20161162, 0.72360394, 0.59130708, -0.08043791, 0.08757349, -0.13893918, -0.05147377,
+ 0.02680690, -0.01144070, 0.00625162, -0.00634215, -0.01248947, -0.00329455, -0.00609625, -0.00136305, -0.05097048, -0.01029851,
+ 0.25065384, -0.16856837, -0.07123372, 0.15992623, -0.39487617, -0.79972301, 0.18118185, -0.04826639, -0.01805578, -0.02927253,
+-0.16400618, 0.07472763, 0.10376449, 0.01705406, 0.01065801, -0.01500498, 0.02039914, 0.37776349, -0.84484186, 0.10434286,
+ 0.15616990, 0.13474456, -0.00906238, -0.25238368, -0.03820885, -0.10650905, -0.03880833, -0.03660028, -0.09640894, 0.00583314,
+ 0.01922097, 0.01489911, -0.02431117, -0.09372217, 0.39404721, -0.84786223, -0.31277121, 0.03193850, 0.01974060, 0.01887901,
+ 0.00337911, -0.11359599, -0.02792521, -0.03220184, -0.01533311, 0.00015962, -0.04225043, -0.00933965, 0.00675311, 0.00206060,
+ 0.15926771, 0.40199829, -0.80792558, -0.35591604, -0.17169764, 0.02830436, 0.02459982, -0.03438589, 0.00718705, -0.01798329,
+-0.01594508, -0.00702430, -0.00952419, -0.00962701, -0.01307212, -0.01749740, 0.01299602, 0.00587270, -0.36103108, -0.82039266,
+-0.43092844, -0.08500097, -0.04361674, -0.00333482, 0.01250434, -0.02538295, -0.00921797, 0.01645071, -0.01400872, 0.00317607,
+ 0.00003277, -0.01617646, -0.00616863, -0.00882661, 0.00466157, 0.00353237, 0.91803104, -0.39503305, -0.02048964, 0.00060125,
+ 0.01980634, 0.00300109, 0.00313880, 0.00657337, 0.00715163, 0.00000261, 0.00854276, -0.00154825, -0.00516128, 0.00909527,
+ 0.00095609, 0.00701196, -0.00221867, -0.00156741},
+{-0.00469582, -0.00020403, -0.00587134, 0.00185153, -0.02256479, -0.01185761, -0.02891481, -0.00493792, -0.00182344, 0.00285962,
+ 0.01558059, -0.02185140, 0.04639438, -0.04357142, 0.12718613, -0.06756136, 0.05542227, 0.98480184, -0.00374376, -0.00236433,
+-0.00607169, -0.00303290, -0.00127243, -0.01794845, 0.00620033, -0.00732704, -0.02837749, -0.00107164, 0.04820548, 0.00713300,
+ 0.09784244, -0.16806261, -0.04563341, -0.33406041, 0.91554083, -0.08139655, -0.00415851, -0.00538193, -0.00731198, -0.00534534,
+-0.00623075, -0.02016943, -0.05480133, -0.03172290, -0.03879603, 0.01518441, 0.09591688, 0.02238470, 0.08126640, 0.08236821,
+-0.24802119, 0.89516402, 0.32029647, 0.07188887, -0.00220366, 0.00344025, -0.00277284, 0.00358963, -0.08668007, -0.02205910,
+-0.05289669, -0.03535201, -0.01188017, -0.06456872, -0.09321006, -0.00009617, -0.15804070, 0.24632041, 0.90166119, 0.19250690,
+ 0.17264619, -0.09699155, -0.00567329, -0.00897700, -0.01442565, -0.01939390, 0.03702127, -0.02999862, -0.04385696, -0.05232394,
+-0.03339177, 0.03905964, -0.00281424, -0.29213275, 0.02892968, 0.90257613, -0.21546058, -0.18070946, 0.09014567, 0.04117230,
+-0.01029696, -0.00329116, -0.03354346, 0.02937079, 0.01274208, -0.01260649, -0.03505571, -0.01020645, 0.03787209, 0.12132165,
+-0.20826840, 0.81556933, -0.43874351, 0.21518682, -0.14564290, -0.05210031, 0.07124563, 0.06127983, -0.00457321, 0.01740496,
+ 0.04185176, 0.00128036, -0.05033693, -0.01890046, 0.06221734, 0.10280078, -0.03738531, 0.04830209, -0.08408293, -0.46409009,
+-0.83936263, -0.14817619, -0.13135927, 0.04563506, 0.08340661, 0.04040200, 0.00044396, -0.01365972, 0.01228951, 0.01078273,
+ 0.09205406, -0.03791500, 0.07135889, 0.08158339, 0.06298278, -0.22875755, -0.92917558, -0.11248260, 0.17801883, -0.03971674,
+-0.07491915, 0.06477287, 0.04635713, 0.01856159, 0.00130895, -0.01991604, 0.02358176, -0.09376056, 0.02782280, -0.04691559,
+ 0.13749249, 0.31383132, 0.92274602, 0.04727419, 0.09765196, -0.02108945, 0.00626005, 0.05193322, 0.02009133, 0.03094066,
+ 0.04573470, 0.00451733, 0.00240169, -0.00982355, -0.03546208, -0.14156875, -0.02480689, 0.22997442, 0.09778317, 0.88834235,
+-0.32797611, -0.00079977, 0.04917079, 0.06977359, 0.06451185, 0.07816204, 0.03119314, 0.01136506, 0.01062006, 0.00632783,
+ 0.03241828, -0.03318847, -0.01350502, -0.30055361, 0.07265375, 0.17308022, 0.88795796, -0.23231020, -0.08932700, 0.11759604,
+ 0.00590705, 0.03525351, 0.00840466, 0.04389942, 0.04387629, 0.04003275, 0.01772966, 0.02709780, -0.02393282, 0.02766178,
+ 0.00342983, -0.33882220, 0.76612668, 0.44061716, -0.28414784, -0.09364014, 0.03694060, 0.01124120, 0.01130268, -0.02869682,
+-0.07428963, -0.03504754, 0.05874942, 0.01196795, 0.02003875, 0.00787152, -0.01605561, 0.04501257, -0.06959958, -0.13015784,
+-0.05738065, 0.04681625, 0.06668700, -0.04492094, 0.02927765, -0.94404277, 0.19243952, 0.09504337, -0.12540826, 0.05394317,
+-0.07972638, -0.02145188, 0.00136427, 0.01964678, 0.06667373, 0.06204535, 0.17302394, 0.22005905, 0.58329964, -0.68440447,
+ 0.19628796, 0.15718011, -0.12481840, -0.08222507, 0.11780870, 0.03798206, -0.01818866, 0.00892766, 0.05582263, 0.01126832,
+-0.00973589, 0.00697442, -0.09937902, 0.06621185, -0.19452202, -0.80004569, -0.13946094, -0.48990700, -0.17595191, -0.00798873,
+-0.06121856, 0.08768040, -0.04507631, 0.00448896, 0.01153941, -0.04711652, -0.01050749, -0.01660047, -0.03007159, -0.01468906,
+ 0.12848053, 0.13859838, 0.93863771, -0.22250065, -0.14841278, 0.04666032, -0.06344813, -0.01915105, -0.01840150, -0.02389410,
+-0.01245496, 0.05023402, 0.02125840, 0.02467318, -0.01893022, -0.00889647, 0.00551817, 0.00481915, -0.40626968, -0.89028236,
+ 0.18261687, -0.03852330, 0.02621926, -0.05420122, -0.01704117, -0.00072893, -0.02694170, -0.04335124, 0.02256467, 0.00642301,
+-0.01619484, -0.00871160, 0.00400065, -0.00488820, -0.00752173, -0.00170603, 0.89554989, -0.41825934, -0.08725803, -0.09051404,
+-0.00916236, -0.02959065, -0.07268075, -0.00816626, -0.00314215, -0.01941078, -0.00036782, -0.00188655, -0.02107724, -0.00771657,
+-0.00448194, -0.00387517, 0.00082998, 0.00202471},
+{ 0.00167296, -0.00647772, -0.00604394, 0.01490810, -0.00837664, 0.00246438, 0.02082153, 0.01216715, 0.01001396, -0.02850860,
+-0.01187868, -0.00113289, 0.04140237, -0.11084998, 0.16102260, 0.20084170, -0.28969446, -0.91312256, 0.00087788, -0.00136895,
+ 0.00004622, 0.00578894, 0.00524119, -0.00044550, 0.00948906, -0.00396910, -0.03312197, -0.00075487, 0.00987494, -0.02088734,
+ 0.09835550, -0.20080342, 0.13687782, -0.16111863, -0.90089988, 0.30312999, 0.00248784, -0.00975419, -0.01617200, 0.00699371,
+-0.02151635, -0.01625774, -0.01262800, 0.02588781, -0.05620764, -0.13651454, 0.04242442, -0.02615307, 0.20497288, -0.20422909,
+ 0.14184406, 0.89712919, 0.01758042, 0.25447787, -0.00207668, -0.00260329, 0.00724812, -0.01007749, 0.00806242, -0.03089729,
+-0.01161934, -0.00618676, -0.10327342, -0.10160272, 0.11919283, 0.20781533, 0.11564869, -0.19072476, 0.86402008, -0.24650846,
+ 0.24684161, 0.04775750, 0.00486888, -0.01735569, -0.01868000, -0.01870386, -0.03243262, -0.05883701, -0.03433371, 0.10441236,
+-0.22831067, -0.22837988, 0.15082544, -0.21313767, 0.13215611, -0.78096079, -0.32270595, -0.21307018, 0.17339271, -0.05435742,
+-0.00940813, 0.00272520, 0.00542917, -0.05232991, -0.01280809, -0.10773627, -0.17626479, 0.03719285, -0.26297104, -0.21780618,
+ 0.21406665, 0.15202177, 0.75911044, 0.38627481, -0.16504189, -0.10242997, -0.02394939, -0.06018959, 0.00994733, -0.02617197,
+-0.01543723, -0.10320051, -0.03010481, -0.19098072, -0.06893233, 0.12253174, -0.25556092, -0.31989059, 0.09542655, 0.72712041,
+-0.43108921, -0.01568072, -0.16532685, 0.06646835, -0.08885408, -0.00050364, -0.01791050, 0.00245405, 0.00204794, -0.17948691,
+-0.05193881, -0.16329387, -0.13676259, 0.01214133, -0.30994612, -0.00687734, 0.63254090, -0.47180795, -0.35409214, 0.23658315,
+ 0.11170294, 0.05229887, -0.06107035, -0.01094212, 0.01523854, -0.01608284, -0.03739206, -0.23864328, -0.03958494, -0.19305719,
+-0.26019058, 0.24108257, -0.55933566, 0.40623396, -0.53367968, -0.08930957, -0.00599383, -0.00050845, 0.06960811, 0.02664961,
+ 0.01464197, -0.00486781, -0.01905736, 0.01437578, 0.02379930, -0.26639588, 0.05208876, -0.43525002, -0.63009424, 0.05251889,
+ 0.56732782, -0.06731164, -0.03705909, -0.03253946, 0.00950673, -0.07941760, 0.02388267, -0.01258409, -0.00343524, 0.00148711,
+-0.00362107, 0.03981813, -0.07235214, -0.46180041, -0.05595288, -0.55699317, 0.61935853, -0.25379716, 0.06796783, 0.01039267,
+-0.06329171, -0.02143024, 0.09406929, -0.00799203, -0.01419805, -0.00603024, 0.01313145, 0.00091161, -0.00212107, -0.02405340,
+ 0.07146405, -0.76695326, -0.14841817, 0.60372663, -0.01478424, 0.06522462, 0.08580016, -0.05817981, 0.02438942, 0.04840904,
+ 0.02934363, -0.02239678, -0.00582247, -0.00091312, -0.00394148, -0.00285276, -0.03435745, 0.05277435, 0.17882781, -0.06194164,
+ 0.27321118, 0.01840179, -0.10188148, -0.33168524, -0.03491221, 0.67351789, 0.37017376, 0.32083717, 0.09737800, -0.20998084,
+-0.10725041, 0.06379186, 0.02169903, -0.02031584, 0.05623799, -0.18300962, -0.17337803, 0.08915172, -0.53835537, -0.08547263,
+ 0.15163321, 0.56732906, 0.21878115, 0.37274266, 0.26206918, 0.13443927, 0.09178695, -0.03276324, -0.01131664, -0.00236369,
+ 0.00772568, 0.01008805, -0.17122615, 0.15301569, 0.40135484, -0.06058913, 0.56405128, -0.05176853, 0.24544337, 0.62448073,
+ 0.07265009, -0.01198695, 0.05151774, -0.03678498, 0.01886154, 0.03724094, 0.01393667, 0.00758055, -0.00254297, 0.00537118,
+ 0.24169707, -0.41735970, -0.67564355, -0.09270478, 0.53106033, 0.06214579, 0.02574404, 0.09943837, 0.03032542, 0.02194476,
+ 0.06369772, -0.00133741, 0.01301113, 0.01508494, 0.00036111, -0.00278870, 0.00139205, 0.00015792, -0.43347887, 0.69923146,
+-0.55406563, -0.01102231, 0.01347767, 0.07012139, -0.02530164, 0.06803192, 0.01177196, 0.04374491, 0.04073027, 0.04037438,
+ 0.00167330, -0.01807065, -0.00425562, 0.00149653, -0.00035119, -0.00172888, 0.84785495, 0.52289580, 0.01067734, -0.00859194,
+ 0.01685964, 0.00481442, 0.00434738, 0.07592695, 0.01419942, 0.01005336, 0.03316937, 0.00360465, 0.00435039, 0.00029122,
+ 0.00171268, 0.00198919, -0.00046889, -0.00094176}};
+
+/* right KLT transforms */
+const double WebRtcIsac_kKltT2Gain[3][36] = {
+{ 0.14572837, -0.45446306, 0.61990621, -0.52197033, 0.32145074, -0.11026900, -0.20698282, 0.48962182, -0.27127933, -0.33627476,
+ 0.65094037, -0.32715751, 0.40262573, -0.47844405, -0.33876075, 0.44130653, 0.37383966, -0.39964662, -0.51730480, 0.06611973,
+ 0.49030187, 0.47512886, -0.02141226, -0.51129451, -0.58578569, -0.39132064, -0.13187771, 0.15649421, 0.40735596, 0.54396897,
+ 0.40381276, 0.40904942, 0.41179766, 0.41167576, 0.40840251, 0.40468132},
+{-0.11368135, 0.34815515, -0.56434996, 0.61130763, -0.39970336, 0.11795708, 0.28514257, -0.58879243, 0.32775812, 0.27024886,
+-0.56251299, 0.27411037, 0.42649186, -0.44080232, -0.36408215, 0.35932457, 0.43592895, -0.41484213, -0.49813030, -0.00012592,
+ 0.49865688, 0.47634953, 0.01094246, -0.52552726, -0.56154082, -0.41110686, -0.14170764, 0.15946614, 0.40818082, 0.55094554,
+ 0.40051601, 0.41084781, 0.41567800, 0.41450700, 0.40871872, 0.39891823},
+{-0.10719481, 0.34796287, -0.54573957, 0.59521001, -0.43943367, 0.14907223, 0.26554957, -0.59549939, 0.36760692, 0.26040652,
+-0.55268701, 0.25778784, 0.38994096, -0.45282773, -0.37975656, 0.40213055, 0.43052647, -0.38937904, -0.52698359, 0.02788094,
+ 0.48284286, 0.47792474, 0.02557759, -0.50922240, -0.57699826, -0.39476779, -0.14708238, 0.12742149, 0.37835245, 0.57464021,
+ 0.39408127, 0.40327462, 0.40993655, 0.41419345, 0.41506301, 0.41253853}};
+
+const double WebRtcIsac_kKltT2Shape[3][36] = {
+{ 0.13427386, -0.35132558, 0.52506528, -0.59419077, 0.45075085, -0.16312057, 0.29857439, -0.58660147, 0.34265431, 0.20879510,
+-0.56063262, 0.30238345, 0.43308283, -0.41186999, -0.35288681, 0.42768996, 0.36094634, -0.45284910, -0.47116680, 0.02893449,
+ 0.54326135, 0.45249040, -0.06264420, -0.52283830, 0.57137758, 0.44298139, 0.12617554, -0.20819946, -0.42324603, -0.48876443,
+ 0.39597050, 0.40713935, 0.41389880, 0.41512486, 0.41130400, 0.40575001},
+{ 0.16540737, -0.43379435, 0.58165221, -0.55154773, 0.35734028, -0.11935912, 0.29434254, -0.55954817, 0.23549804, 0.33087258,
+-0.58848503, 0.29835834, 0.45464789, -0.38316155, -0.41689708, 0.35607296, 0.41260747, -0.41910198, -0.48633899, -0.04144955,
+ 0.47824583, 0.51050942, 0.01000345, -0.52184032, 0.53488229, 0.42641051, 0.17049774, -0.15849613, -0.43229355, -0.53945045,
+ 0.39582002, 0.41033103, 0.41788713, 0.41688080, 0.41081697, 0.39719658},
+{ 0.13386268, -0.37919915, 0.54989123, -0.57663572, 0.42402636, -0.15362720, 0.29641577, -0.58806770, 0.31381040, 0.26524954,
+-0.56271012, 0.28431868, 0.42699898, -0.41058922, -0.40408270, 0.39215865, 0.40788513, -0.40699735, -0.49846482, -0.01521208,
+ 0.48756040, 0.49479418, -0.00347672, -0.51841384, 0.55513106, 0.41683793, 0.15131217, -0.15613621, -0.41029341, -0.54996461,
+ 0.39402116, 0.40965305, 0.41862791, 0.41730770, 0.41089648, 0.39837262}};
+
+/* means of log gains and LAR coefficients*/
+const double WebRtcIsac_kLpcMeansGain[3][12] = {
+{-6.86881911, -5.35075273, -6.86792680, -5.36200897, -6.86401538, -5.36921533, -6.86802969, -5.36893966, -6.86538097, -5.36315063,
+-6.85535304, -5.35155315},
+{-6.12914600, -4.78070092, -6.12971780, -4.78382183, -6.12858525, -4.79362198, -6.12926491, -4.79017481, -6.12102401, -4.78346122,
+-6.11441152, -4.78019228},
+{-5.67273484, -3.73876311, -5.65246094, -3.71407895, -5.61716443, -3.68814580, -5.58804560, -3.66334094, -5.54189577, -3.63845640,
+-5.49293185, -3.61760203}};
+
+const double WebRtcIsac_kLpcMeansShape[3][108] = {
+{-0.91232981, 0.26258634, -0.33716701, 0.08477430, -0.03378426, 0.14423909, 0.07036185, 0.06155019, 0.01490385, 0.04138740,
+ 0.01427317, 0.01288970, 0.83872106, 0.25750199, 0.07988929, -0.01957923, 0.00831390, 0.01770300, -0.90957164, 0.25732216,
+-0.33385344, 0.08735740, -0.03715332, 0.14584917, 0.06998990, 0.06131968, 0.01504379, 0.04067339, 0.01428039, 0.01406460,
+ 0.83846243, 0.26169862, 0.08109025, -0.01767055, 0.00970539, 0.01954310, -0.90490803, 0.24656405, -0.33578607, 0.08843286,
+-0.03749139, 0.14443959, 0.07214669, 0.06170993, 0.01449947, 0.04134309, 0.01314762, 0.01413471, 0.83895203, 0.26748062,
+ 0.08197507, -0.01781298, 0.00885967, 0.01922394, -0.90922472, 0.24495889, -0.33921540, 0.08877169, -0.03581332, 0.14199172,
+ 0.07444032, 0.06185940, 0.01502054, 0.04185113, 0.01276579, 0.01355457, 0.83645358, 0.26631720, 0.08119697, -0.01835449,
+ 0.00788512, 0.01846446, -0.90482253, 0.24658310, -0.34019734, 0.08281090, -0.03486038, 0.14359248, 0.07401336, 0.06001471,
+ 0.01528421, 0.04254560, 0.01321472, 0.01240799, 0.83857127, 0.26281654, 0.08174380, -0.02099842, 0.00755176, 0.01699448,
+-0.90132307, 0.25174308, -0.33838268, 0.07883863, -0.02877906, 0.14105407, 0.07220290, 0.06000352, 0.01684879, 0.04226844,
+ 0.01331331, 0.01269244, 0.83832138, 0.25467485, 0.08118028, -0.02120528, 0.00747832, 0.01567212},
+{-1.11639718, 0.35377266, 0.00798929, 0.20165280, 0.07656104, 0.10629964, 0.04894160, 0.10955305, -0.01806405, 0.05082282,
+ 0.01730794, 0.01345957, 0.73717782, 0.05952284, 0.03176204, 0.08195122, 0.01253148, 0.02253385, -1.12053537, 0.35523538,
+ 0.00859646, 0.20007706, 0.07715852, 0.10754596, 0.05165976, 0.10927703, -0.01554395, 0.05178866, 0.01752534, 0.01343468,
+ 0.73489046, 0.06395167, 0.03287798, 0.07972374, 0.01293550, 0.02300929, -1.11772179, 0.35457623, 0.01205524, 0.19926481,
+ 0.08000866, 0.10817921, 0.05052481, 0.11016167, -0.01552091, 0.05155510, 0.01787163, 0.01343778, 0.73142568, 0.06840830,
+ 0.03316828, 0.07902608, 0.01525042, 0.02178127, -1.12120164, 0.36405233, 0.00630305, 0.19799738, 0.07829690, 0.10727588,
+ 0.04017317, 0.10437949, -0.01844109, 0.05021700, 0.01561726, 0.01226571, 0.73438044, 0.06947982, 0.03396317, 0.07858683,
+ 0.01367105, 0.02041955, -1.12146187, 0.35952226, 0.00340090, 0.19700813, 0.07938222, 0.10904137, 0.03921216, 0.10531403,
+-0.01833415, 0.04956231, 0.01399539, 0.01323582, 0.74378099, 0.07059589, 0.03367692, 0.08151462, 0.01182040, 0.02075577,
+-1.11245254, 0.35234230, 0.00687490, 0.20204252, 0.07813186, 0.11081259, 0.04634665, 0.11073238, -0.01637954, 0.05104577,
+ 0.01675122, 0.01448696, 0.74013627, 0.06239059, 0.03129412, 0.08207461, 0.01249475, 0.02189238},
+{-1.27118948, 0.35834331, -0.33499347, 0.13524073, 0.04829079, 0.19542773, 0.05273835, 0.04157974, -0.01755227, 0.01513442,
+ 0.00386630, 0.02199463, 1.14439142, 0.21903073, 0.14750213, 0.12743356, 0.08463334, 0.06839691, -1.28367777, 0.35556287,
+-0.33809405, 0.13627881, 0.04939309, 0.19642571, 0.05354373, 0.04099247, -0.01787481, 0.01472425, 0.00391474, 0.02150716,
+ 1.14739079, 0.21840872, 0.14643624, 0.12724347, 0.08390642, 0.06811938, -1.29007667, 0.35159558, -0.34154267, 0.13295849,
+ 0.04883602, 0.19587595, 0.05452759, 0.04174703, -0.01782110, 0.01388270, 0.00374754, 0.02138105, 1.14333767, 0.21690116,
+ 0.14544599, 0.12606728, 0.08314168, 0.06771389, -1.29856471, 0.35239315, -0.34238732, 0.13277553, 0.04722712, 0.19233156,
+ 0.05366901, 0.04328110, -0.01657749, 0.01444736, 0.00438108, 0.02102563, 1.13548397, 0.21537812, 0.14357377, 0.12525845,
+ 0.08230994, 0.06722511, -1.30663540, 0.34366563, -0.34205544, 0.12861679, 0.04655851, 0.18864359, 0.05351285, 0.04358693,
+-0.01604498, 0.01431907, 0.00395326, 0.02082299, 1.12207794, 0.21167325, 0.14212491, 0.12418671, 0.08155467, 0.06639789,
+-1.31011673, 0.33686271, -0.34379843, 0.12169569, 0.04480323, 0.18637557, 0.05374078, 0.04260827, -0.01588226, 0.01378294,
+ 0.00396009, 0.02112406, 1.10466984, 0.20905894, 0.14107033, 0.12303074, 0.08047136, 0.06588031}};
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h
new file mode 100644
index 0000000..604d963
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * lpc_tables.h
+ *
+ * header file for coding tables for the LPC coefficients
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
+
+#include "structs.h"
+
+#include "settings.h"
+
+#define KLT_STEPSIZE 1.00000000
+#define KLT_NUM_AVG_GAIN 0
+#define KLT_NUM_AVG_SHAPE 0
+#define KLT_NUM_MODELS 3
+#define LPC_GAIN_SCALE 4.000f
+#define LPC_LOBAND_SCALE 2.100f
+#define LPC_LOBAND_ORDER ORDERLO
+#define LPC_HIBAND_SCALE 0.450f
+#define LPC_HIBAND_ORDER ORDERHI
+#define LPC_GAIN_ORDER 2
+
+#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
+
+#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES)
+#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES)
+/* indices of KLT coefficients used */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltSelIndGain[12];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltSelIndShape[108];
+
+/* cdf array for model indicator */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
+
+/* pointer to cdf array for model indicator */
+extern const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1];
+
+/* initial cdf index for decoder of model indicator */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1];
+
+/* offset to go from rounded value to quantization index */
+extern const short WebRtcIsac_kQKltQuantMinGain[12];
+
+extern const short WebRtcIsac_kQKltQuantMinShape[108];
+
+/* maximum quantization index */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108];
+
+/* index offset */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[KLT_NUM_MODELS][108];
+
+/* initial cdf index for KLT coefficients */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[KLT_NUM_MODELS][108];
+
+/* offsets for quantizer representation levels */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsGain[3];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsShape[3];
+
+/* quantizer representation levels */
+extern const double WebRtcIsac_kQKltLevelsGain[1176];
+
+extern const double WebRtcIsac_kQKltLevelsShape[1735];
+
+/* cdf tables for quantizer indices */
+extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[1212];
+
+extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[2059];
+
+/* pointers to cdf tables for quantizer indices */
+extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[KLT_NUM_MODELS][12];
+
+extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[KLT_NUM_MODELS][108];
+
+/* code length for all coefficients using different models */
+extern const double WebRtcIsac_kQKltCodeLenGain[392];
+
+extern const double WebRtcIsac_kQKltCodeLenShape[578];
+
+/* left KLT transforms */
+extern const double WebRtcIsac_kKltT1Gain[KLT_NUM_MODELS][4];
+
+extern const double WebRtcIsac_kKltT1Shape[KLT_NUM_MODELS][324];
+
+/* right KLT transforms */
+extern const double WebRtcIsac_kKltT2Gain[KLT_NUM_MODELS][36];
+
+extern const double WebRtcIsac_kKltT2Shape[KLT_NUM_MODELS][36];
+
+/* means of log gains and LAR coefficients */
+extern const double WebRtcIsac_kLpcMeansGain[KLT_NUM_MODELS][12];
+
+extern const double WebRtcIsac_kLpcMeansShape[KLT_NUM_MODELS][108];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h
new file mode 100644
index 0000000..c469c2e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+
+#include <math.h>
+#include "typedefs.h"
+
+// TODO(turaj): switch to WEBRTC_POSIX when available
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#define WebRtcIsac_lrint lrint
+#elif (defined(WEBRTC_ARCH_X86) && defined(WIN32))
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+ long int x_int;
+
+ __asm {
+ fld x_dbl
+ fistp x_int
+ };
+
+ return x_int;
+}
+#else // Do a slow but correct implementation of lrint
+
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+ long int x_int;
+ x_int = (long int)floor(x_dbl + 0.499999999999);
+ return x_int;
+}
+
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c
new file mode 100644
index 0000000..75525f6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c
@@ -0,0 +1,622 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pitch_estimator.h"
+
+#include <math.h>
+#include <memory.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+static const double kInterpolWin[8] = {-0.00067556028640, 0.02184247643159, -0.12203175715679, 0.60086484101160,
+ 0.60086484101160, -0.12203175715679, 0.02184247643159, -0.00067556028640};
+
+/* interpolation filter */
+__inline static void IntrepolFilter(double *data_ptr, double *intrp)
+{
+ *intrp = kInterpolWin[0] * data_ptr[-3];
+ *intrp += kInterpolWin[1] * data_ptr[-2];
+ *intrp += kInterpolWin[2] * data_ptr[-1];
+ *intrp += kInterpolWin[3] * data_ptr[0];
+ *intrp += kInterpolWin[4] * data_ptr[1];
+ *intrp += kInterpolWin[5] * data_ptr[2];
+ *intrp += kInterpolWin[6] * data_ptr[3];
+ *intrp += kInterpolWin[7] * data_ptr[4];
+}
+
+
+/* 2D parabolic interpolation */
+/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
+__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
+{
+ double c, b[2], A[2][2];
+ double t1, t2, d;
+ double delta1, delta2;
+
+
+ // double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
+ // should result in: delta1 = 0.5; delta2 = 0.0; peak_val = 1.0
+
+ c = T[1][1];
+ b[0] = 0.5 * (T[1][2] + T[2][1] - T[0][1] - T[1][0]);
+ b[1] = 0.5 * (T[1][0] + T[2][1] - T[0][1] - T[1][2]);
+ A[0][1] = -0.5 * (T[0][1] + T[2][1] - T[1][0] - T[1][2]);
+ t1 = 0.5 * (T[0][0] + T[2][2]) - c;
+ t2 = 0.5 * (T[2][0] + T[0][2]) - c;
+ d = (T[0][1] + T[1][2] + T[1][0] + T[2][1]) - 4.0 * c - t1 - t2;
+ A[0][0] = -t1 - 0.5 * d;
+ A[1][1] = -t2 - 0.5 * d;
+
+ /* deal with singularities or ill-conditioned cases */
+ if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
+ *peak_val = T[1][1];
+ return;
+ }
+
+ /* Cholesky decomposition: replace A by upper-triangular factor */
+ A[0][0] = sqrt(A[0][0]);
+ A[0][1] = A[0][1] / A[0][0];
+ A[1][1] = sqrt(A[1][1] - A[0][1] * A[0][1]);
+
+ /* compute [x; y] = -0.5 * inv(A) * b */
+ t1 = b[0] / A[0][0];
+ t2 = (b[1] - t1 * A[0][1]) / A[1][1];
+ delta2 = t2 / A[1][1];
+ delta1 = 0.5 * (t1 - delta2 * A[0][1]) / A[0][0];
+ delta2 *= 0.5;
+
+ /* limit norm */
+ t1 = delta1 * delta1 + delta2 * delta2;
+ if (t1 > 1.0) {
+ delta1 /= t1;
+ delta2 /= t1;
+ }
+
+ *peak_val = 0.5 * (b[0] * delta1 + b[1] * delta2) + c;
+
+ *x += delta1;
+ *y += delta2;
+}
+
+
+static void PCorr(const double *in, double *outcorr)
+{
+ double sum, ysum, prod;
+ const double *x, *inptr;
+ int k, n;
+
+ //ysum = 1e-6; /* use this with float (i.s.o. double)! */
+ ysum = 1e-13;
+ sum = 0.0;
+ x = in + PITCH_MAX_LAG/2 + 2;
+ for (n = 0; n < PITCH_CORR_LEN2; n++) {
+ ysum += in[n] * in[n];
+ sum += x[n] * in[n];
+ }
+
+ outcorr += PITCH_LAG_SPAN2 - 1; /* index of last element in array */
+ *outcorr = sum / sqrt(ysum);
+
+ for (k = 1; k < PITCH_LAG_SPAN2; k++) {
+ ysum -= in[k-1] * in[k-1];
+ ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
+ sum = 0.0;
+ inptr = &in[k];
+ prod = x[0] * inptr[0];
+ for (n = 1; n < PITCH_CORR_LEN2; n++) {
+ sum += prod;
+ prod = x[n] * inptr[n];
+ }
+ sum += prod;
+ outcorr--;
+ *outcorr = sum / sqrt(ysum);
+ }
+}
+
+
+void WebRtcIsac_InitializePitch(const double *in,
+ const double old_lag,
+ const double old_gain,
+ PitchAnalysisStruct *State,
+ double *lags)
+{
+ double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+ double ratio, log_lag, gain_bias;
+ double bias;
+ double corrvec1[PITCH_LAG_SPAN2];
+ double corrvec2[PITCH_LAG_SPAN2];
+ int m, k;
+ // Allocating 10 extra entries at the begining of the CorrSurf
+ double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
+ double* CorrSurf[2*PITCH_BW+3];
+ double *CorrSurfPtr1, *CorrSurfPtr2;
+ double LagWin[3] = {0.2, 0.5, 0.98};
+ int ind1, ind2, peaks_ind, peak, max_ind;
+ int peaks[PITCH_MAX_NUM_PEAKS];
+ double adj, gain_tmp;
+ double corr, corr_max;
+ double intrp_a, intrp_b, intrp_c, intrp_d;
+ double peak_vals[PITCH_MAX_NUM_PEAKS];
+ double lags1[PITCH_MAX_NUM_PEAKS];
+ double lags2[PITCH_MAX_NUM_PEAKS];
+ double T[3][3];
+ int row;
+
+ for(k = 0; k < 2*PITCH_BW+3; k++)
+ {
+ CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
+ }
+ /* reset CorrSurf matrix */
+ memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
+
+ //warnings -DH
+ max_ind = 0;
+ peak = 0;
+
+ /* copy old values from state buffer */
+ memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+ /* decimation; put result after the old values */
+ WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
+ &buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
+
+ /* low-pass filtering */
+ for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
+ buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
+
+ /* copy end part back into state buffer */
+ memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+ /* compute correlation for first and second half of the frame */
+ PCorr(buf_dec, corrvec1);
+ PCorr(buf_dec + PITCH_CORR_STEP2, corrvec2);
+
+ /* bias towards pitch lag of previous frame */
+ log_lag = log(0.5 * old_lag);
+ gain_bias = 4.0 * old_gain * old_gain;
+ if (gain_bias > 0.8) gain_bias = 0.8;
+ for (k = 0; k < PITCH_LAG_SPAN2; k++)
+ {
+ ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
+ bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
+ corrvec1[k] *= bias;
+ }
+
+ /* taper correlation functions */
+ for (k = 0; k < 3; k++) {
+ gain_tmp = LagWin[k];
+ corrvec1[k] *= gain_tmp;
+ corrvec2[k] *= gain_tmp;
+ corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+ corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+ }
+
+ corr_max = 0.0;
+ /* fill middle row of correlation surface */
+ ind1 = 0;
+ ind2 = 0;
+ CorrSurfPtr1 = &CorrSurf[PITCH_BW][2];
+ for (k = 0; k < PITCH_LAG_SPAN2; k++) {
+ corr = corrvec1[ind1++] + corrvec2[ind2++];
+ CorrSurfPtr1[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ }
+ }
+ /* fill first and last rows of correlation surface */
+ ind1 = 0;
+ ind2 = PITCH_BW;
+ CorrSurfPtr1 = &CorrSurf[0][2];
+ CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
+ for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
+ ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+ adj = 0.2 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
+ corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+ CorrSurfPtr1[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ }
+ corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+ CorrSurfPtr2[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+ }
+ }
+ /* fill second and next to last rows of correlation surface */
+ ind1 = 0;
+ ind2 = PITCH_BW-1;
+ CorrSurfPtr1 = &CorrSurf[1][2];
+ CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
+ for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
+ ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+ adj = 0.9 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
+ corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+ CorrSurfPtr1[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ }
+ corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+ CorrSurfPtr2[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+ }
+ }
+ /* fill remainder of correlation surface */
+ for (m = 2; m < PITCH_BW; m++) {
+ ind1 = 0;
+ ind2 = PITCH_BW - m; /* always larger than ind1 */
+ CorrSurfPtr1 = &CorrSurf[m][2];
+ CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
+ for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
+ ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+ adj = ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
+ corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+ CorrSurfPtr1[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ }
+ corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+ CorrSurfPtr2[k] = corr;
+ if (corr > corr_max) {
+ corr_max = corr; /* update maximum */
+ max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+ }
+ }
+ }
+
+ /* threshold value to qualify as a peak */
+ corr_max *= 0.6;
+
+ peaks_ind = 0;
+ /* find peaks */
+ for (m = 1; m < PITCH_BW+1; m++) {
+ if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+ CorrSurfPtr1 = &CorrSurf[m][2];
+ for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
+ corr = CorrSurfPtr1[k];
+ if (corr > corr_max) {
+ if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+ if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+ /* found a peak; store index into matrix */
+ peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+ }
+ }
+ }
+ }
+ }
+ for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
+ if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+ CorrSurfPtr1 = &CorrSurf[m][2];
+ for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
+ corr = CorrSurfPtr1[k];
+ if (corr > corr_max) {
+ if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+ if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+ /* found a peak; store index into matrix */
+ peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+ if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+ }
+ }
+ }
+ }
+ }
+
+ if (peaks_ind > 0) {
+ /* examine each peak */
+ CorrSurfPtr1 = &CorrSurf[0][0];
+ for (k = 0; k < peaks_ind; k++) {
+ peak = peaks[k];
+
+ /* compute four interpolated values around current peak */
+ IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
+ IntrepolFilter(&CorrSurfPtr1[peak - 1 ], &intrp_b);
+ IntrepolFilter(&CorrSurfPtr1[peak ], &intrp_c);
+ IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
+
+ /* determine maximum of the interpolated values */
+ corr = CorrSurfPtr1[peak];
+ corr_max = intrp_a;
+ if (intrp_b > corr_max) corr_max = intrp_b;
+ if (intrp_c > corr_max) corr_max = intrp_c;
+ if (intrp_d > corr_max) corr_max = intrp_d;
+
+ /* determine where the peak sits and fill a 3x3 matrix around it */
+ row = peak / (PITCH_LAG_SPAN2+4);
+ lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+ lags2[k] = (double) (lags1[k] + PITCH_BW - row);
+ if ( corr > corr_max ) {
+ T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+ T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+ T[1][1] = corr;
+ T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+ T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+ T[1][0] = intrp_a;
+ T[0][1] = intrp_b;
+ T[2][1] = intrp_c;
+ T[1][2] = intrp_d;
+ } else {
+ if (intrp_a == corr_max) {
+ lags1[k] -= 0.5;
+ lags2[k] += 0.5;
+ IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
+ IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
+ T[1][1] = intrp_a;
+ T[0][2] = intrp_b;
+ T[2][2] = intrp_c;
+ T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
+ T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+ T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+ T[1][2] = corr;
+ } else if (intrp_b == corr_max) {
+ lags1[k] -= 0.5;
+ lags2[k] -= 0.5;
+ IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
+ T[2][0] = intrp_a;
+ T[1][1] = intrp_b;
+ IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
+ T[2][2] = intrp_d;
+ T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+ T[0][1] = CorrSurfPtr1[peak - 1];
+ T[2][1] = corr;
+ T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+ } else if (intrp_c == corr_max) {
+ lags1[k] += 0.5;
+ lags2[k] += 0.5;
+ T[0][0] = intrp_a;
+ IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
+ T[1][1] = intrp_c;
+ T[0][2] = intrp_d;
+ IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
+ T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+ T[0][1] = corr;
+ T[2][1] = CorrSurfPtr1[peak + 1];
+ T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+ } else {
+ lags1[k] += 0.5;
+ lags2[k] -= 0.5;
+ T[0][0] = intrp_b;
+ T[2][0] = intrp_c;
+ T[1][1] = intrp_d;
+ IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
+ IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
+ T[1][0] = corr;
+ T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+ T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+ T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
+ }
+ }
+
+ /* 2D parabolic interpolation gives more accurate lags and peak value */
+ Intrpol2D(T, &lags1[k], &lags2[k], &peak_vals[k]);
+ }
+
+ /* determine the highest peak, after applying a bias towards short lags */
+ corr_max = 0.0;
+ for (k = 0; k < peaks_ind; k++) {
+ corr = peak_vals[k] * pow(PITCH_PEAK_DECAY, log(lags1[k] + lags2[k]));
+ if (corr > corr_max) {
+ corr_max = corr;
+ peak = k;
+ }
+ }
+
+ lags1[peak] *= 2.0;
+ lags2[peak] *= 2.0;
+
+ if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
+ if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
+ if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
+ if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
+
+ /* store lags of highest peak in output array */
+ lags[0] = lags1[peak];
+ lags[1] = lags1[peak];
+ lags[2] = lags2[peak];
+ lags[3] = lags2[peak];
+ }
+ else
+ {
+ row = max_ind / (PITCH_LAG_SPAN2+4);
+ lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+ lags2[0] = (double) (lags1[0] + PITCH_BW - row);
+
+ if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
+ if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
+ if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
+ if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
+
+ /* store lags of highest peak in output array */
+ lags[0] = lags1[0];
+ lags[1] = lags1[0];
+ lags[2] = lags2[0];
+ lags[3] = lags2[0];
+ }
+}
+
+
+
+/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
+ * t = (0:4)';
+ * A = [t.^0, t.^1, t.^2, t.^3, t.^4];
+ * [Q, dummy] = qr(A);
+ * P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
+static const double kWeight[5][5] = {
+ { 0.29714285714286, -0.30857142857143, -0.05714285714286, 0.05142857142857, 0.01714285714286},
+ {-0.30857142857143, 0.67428571428571, -0.27142857142857, -0.14571428571429, 0.05142857142857},
+ {-0.05714285714286, -0.27142857142857, 0.65714285714286, -0.27142857142857, -0.05714285714286},
+ { 0.05142857142857, -0.14571428571429, -0.27142857142857, 0.67428571428571, -0.30857142857143},
+ { 0.01714285714286, 0.05142857142857, -0.05714285714286, -0.30857142857143, 0.29714285714286}
+};
+
+
+void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
+ double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+ PitchAnalysisStruct *State,
+ double *lags,
+ double *gains)
+{
+ double HPin[PITCH_FRAME_LEN];
+ double Weighted[PITCH_FRAME_LEN];
+ double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
+ double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
+ double out_G[PITCH_FRAME_LEN + QLOOKAHEAD]; // could be removed by using out instead
+ double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
+ double old_lag, old_gain;
+ double nrg_wht, tmp;
+ double Wnrg, Wfluct, Wgain;
+ double H[4][4];
+ double grad[4];
+ double dG[4];
+ int k, m, n, iter;
+
+ /* high pass filtering using second order pole-zero filter */
+ WebRtcIsac_Highpass(in, HPin, State->hp_state, PITCH_FRAME_LEN);
+
+ /* copy from state into buffer */
+ memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
+
+ /* compute weighted and whitened signals */
+ WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
+
+ /* copy from buffer into state */
+ memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
+
+ old_lag = State->PFstr_wght.oldlagp[0];
+ old_gain = State->PFstr_wght.oldgainp[0];
+
+ /* inital pitch estimate */
+ WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
+
+
+ /* Iterative optimization of lags - to be done */
+
+ /* compute energy of whitened signal */
+ nrg_wht = 0.0;
+ for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
+ nrg_wht += Whitened[k] * Whitened[k];
+
+
+ /* Iterative optimization of gains */
+
+ /* set weights for energy, gain fluctiation, and spectral gain penalty functions */
+ Wnrg = 1.0 / nrg_wht;
+ Wgain = 0.005;
+ Wfluct = 3.0;
+
+ /* set initial gains */
+ for (k = 0; k < 4; k++)
+ gains[k] = PITCH_MAX_GAIN_06;
+
+ /* two iterations should be enough */
+ for (iter = 0; iter < 2; iter++) {
+ /* compute Jacobian of pre-filter output towards gains */
+ WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
+
+ /* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
+ for (k = 0; k < 4; k++) {
+ tmp = 0.0;
+ for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+ tmp += out_G[n] * out_dG[k][n];
+ grad[k] = tmp * Wnrg;
+ }
+ for (k = 0; k < 4; k++) {
+ for (m = 0; m <= k; m++) {
+ tmp = 0.0;
+ for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+ tmp += out_dG[m][n] * out_dG[k][n];
+ H[k][m] = tmp * Wnrg;
+ }
+ }
+
+ /* add gradient and Hessian (lower triangle) for dampening fast gain changes */
+ for (k = 0; k < 4; k++) {
+ tmp = kWeight[k+1][0] * old_gain;
+ for (m = 0; m < 4; m++)
+ tmp += kWeight[k+1][m+1] * gains[m];
+ grad[k] += tmp * Wfluct;
+ }
+ for (k = 0; k < 4; k++) {
+ for (m = 0; m <= k; m++) {
+ H[k][m] += kWeight[k+1][m+1] * Wfluct;
+ }
+ }
+
+ /* add gradient and Hessian for dampening gain */
+ for (k = 0; k < 3; k++) {
+ tmp = 1.0 / (1 - gains[k]);
+ grad[k] += tmp * tmp * Wgain;
+ H[k][k] += 2.0 * tmp * (tmp * tmp * Wgain);
+ }
+ tmp = 1.0 / (1 - gains[3]);
+ grad[3] += 1.33 * (tmp * tmp * Wgain);
+ H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
+
+
+ /* compute Cholesky factorization of Hessian
+ * by overwritting the upper triangle; scale factors on diagonal
+ * (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
+ H[0][1] = H[1][0] / H[0][0];
+ H[0][2] = H[2][0] / H[0][0];
+ H[0][3] = H[3][0] / H[0][0];
+ H[1][1] -= H[0][0] * H[0][1] * H[0][1];
+ H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
+ H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
+ H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
+ H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
+ H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
+
+ /* Compute update as delta_gains = -inv(H) * grad */
+ /* copy and negate */
+ for (k = 0; k < 4; k++)
+ dG[k] = -grad[k];
+ /* back substitution */
+ dG[1] -= dG[0] * H[0][1];
+ dG[2] -= dG[0] * H[0][2] + dG[1] * H[1][2];
+ dG[3] -= dG[0] * H[0][3] + dG[1] * H[1][3] + dG[2] * H[2][3];
+ /* scale */
+ for (k = 0; k < 4; k++)
+ dG[k] /= H[k][k];
+ /* back substitution */
+ dG[2] -= dG[3] * H[2][3];
+ dG[1] -= dG[3] * H[1][3] + dG[2] * H[1][2];
+ dG[0] -= dG[3] * H[0][3] + dG[2] * H[0][2] + dG[1] * H[0][1];
+
+ /* update gains and check range */
+ for (k = 0; k < 4; k++) {
+ gains[k] += dG[k];
+ if (gains[k] > PITCH_MAX_GAIN)
+ gains[k] = PITCH_MAX_GAIN;
+ else if (gains[k] < 0.0)
+ gains[k] = 0.0;
+ }
+ }
+
+ /* update state for next frame */
+ WebRtcIsac_PitchfilterPre(Whitened, out, &(State->PFstr_wght), lags, gains);
+
+ /* concatenate previous input's end and current input */
+ memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
+ memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
+
+ /* lookahead pitch filtering for masking analysis */
+ WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);
+
+ /* store last part of input */
+ for (k = 0; k < QLOOKAHEAD; k++)
+ State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h
new file mode 100644
index 0000000..f5d9356
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.h
+ *
+ * Pitch functions
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+
+#include "structs.h"
+
+
+
+void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
+ double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+ PitchAnalysisStruct *State,
+ double *lags,
+ double *gains);
+
+void WebRtcIsac_InitializePitch(const double *in,
+ const double old_lag,
+ const double old_gain,
+ PitchAnalysisStruct *State,
+ double *lags);
+
+void WebRtcIsac_PitchfilterPre(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains);
+
+void WebRtcIsac_PitchfilterPost(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains);
+
+void WebRtcIsac_PitchfilterPre_la(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains);
+
+void WebRtcIsac_PitchfilterPre_gains(double *indat,
+ double *outdat,
+ double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains);
+
+void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata);
+
+void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N);
+
+void WebRtcIsac_DecimateAllpass(const double *in,
+ double *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
+ int N, /* number of input samples */
+ double *out); /* array of size N/2 */
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c
new file mode 100644
index 0000000..ccc8d21
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c
@@ -0,0 +1,469 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pitch_estimator.h"
+#include "os_specific_inline.h"
+
+#include <stdlib.h>
+#include <memory.h>
+#include <math.h>
+
+static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25, -0.07};
+
+/* interpolation coefficients; generated by design_pitch_filter.m */
+static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+ {-0.02239172458614, 0.06653315052934, -0.16515880017569, 0.60701333734125, 0.64671399919202, -0.20249000396417, 0.09926548334755, -0.04765933793109, 0.01754159521746},
+ {-0.01985640750434, 0.05816126837866, -0.13991265473714, 0.44560418147643, 0.79117042386876, -0.20266133815188, 0.09585268418555, -0.04533310458084, 0.01654127246314},
+ {-0.01463300534216, 0.04229888475060, -0.09897034715253, 0.28284326017787, 0.90385267956632, -0.16976950138649, 0.07704272393639, -0.03584218578311, 0.01295781500709},
+ {-0.00764851320885, 0.02184035544377, -0.04985561057281, 0.13083306574393, 0.97545011664662, -0.10177807997561, 0.04400901776474, -0.02010737175166, 0.00719783432422},
+ {-0.00000000000000, 0.00000000000000, -0.00000000000001, 0.00000000000001, 0.99999999999999, 0.00000000000001, -0.00000000000001, 0.00000000000000, -0.00000000000000},
+ { 0.00719783432422, -0.02010737175166, 0.04400901776474, -0.10177807997562, 0.97545011664663, 0.13083306574393, -0.04985561057280, 0.02184035544377, -0.00764851320885},
+ { 0.01295781500710, -0.03584218578312, 0.07704272393640, -0.16976950138650, 0.90385267956634, 0.28284326017785, -0.09897034715252, 0.04229888475059, -0.01463300534216},
+ { 0.01654127246315, -0.04533310458085, 0.09585268418557, -0.20266133815190, 0.79117042386878, 0.44560418147640, -0.13991265473712, 0.05816126837865, -0.01985640750433}
+};
+
+
+void WebRtcIsac_PitchfilterPre(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains)
+{
+
+ double ubuf[PITCH_INTBUFFSIZE];
+ const double *fracoeff = NULL;
+ double curgain, curlag, gaindelta, lagdelta;
+ double sum, inystate[PITCH_DAMPORDER];
+ double ftmp, oldlag, oldgain;
+ int k, n, m, pos, ind, pos2, Li, frc;
+
+ Li = 0;
+ /* Set up buffer and states */
+ memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
+
+ oldlag = *pfp->oldlagp;
+ oldgain = *pfp->oldgainp;
+
+ /* No interpolation if pitch lag step is big */
+ if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
+ oldlag = lags[0];
+ oldgain = gains[0];
+ }
+
+ ind=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
+ curlag=oldlag ;
+ gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
+ curgain=oldgain ;
+ oldlag=lags[k];
+ oldgain=gains[k];
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+ if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
+ curgain += gaindelta;
+ curlag += lagdelta;
+ Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
+ ftmp = Li - (curlag+PITCH_FILTDELAY);
+ frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
+ fracoeff = kIntrpCoef[frc];
+ }
+
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystate[m] = inystate[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+ sum=0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+
+ /* Low pass filter */
+ sum=0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Subtract from input and update buffer */
+ outdat[ind] = indat[ind] - sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+ ind++;
+ }
+ }
+
+ /* Export buffer and states */
+ memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
+
+ *pfp->oldlagp = oldlag;
+ *pfp->oldgainp = oldgain;
+
+}
+
+
+void WebRtcIsac_PitchfilterPre_la(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains)
+{
+ double ubuf[PITCH_INTBUFFSIZE+QLOOKAHEAD];
+ const double *fracoeff = NULL;
+ double curgain, curlag, gaindelta, lagdelta;
+ double sum, inystate[PITCH_DAMPORDER];
+ double ftmp;
+ double oldlag, oldgain;
+ int k, n, m, pos, ind, pos2, Li, frc;
+
+ Li = 0;
+ /* Set up buffer and states */
+ memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
+
+ oldlag = *pfp->oldlagp;
+ oldgain = *pfp->oldgainp;
+
+ /* No interpolation if pitch lag step is big */
+ if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
+ oldlag = lags[0];
+ oldgain = gains[0];
+ }
+
+
+ ind=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
+ curlag=oldlag ;
+ gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
+ curgain=oldgain ;
+ oldlag=lags[k];
+ oldgain=gains[k];
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+ if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
+ curgain += gaindelta;
+ curlag += lagdelta;
+ Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
+ ftmp = Li - (curlag+PITCH_FILTDELAY);
+ frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
+ fracoeff = kIntrpCoef[frc];
+ }
+
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystate[m] = inystate[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+ sum=0.0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+
+ /* Low pass filter */
+ sum=0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Subtract from input and update buffer */
+ outdat[ind] = indat[ind] - sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+ ind++;
+ }
+ }
+
+ /* Export buffer and states */
+ memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
+
+ *pfp->oldlagp = oldlag;
+ *pfp->oldgainp = oldgain;
+
+
+ /* Filter look-ahead segment */
+ for (n=0;n<QLOOKAHEAD;n++) {
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystate[m] = inystate[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+ sum=0.0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+
+ /* Low pass filter */
+ sum=0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Subtract from input and update buffer */
+ outdat[ind] = indat[ind] - sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+ ind++;
+ }
+}
+
+
+void WebRtcIsac_PitchfilterPre_gains(double *indat,
+ double *outdat,
+ double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains)
+{
+ double ubuf[PITCH_INTBUFFSIZE+QLOOKAHEAD];
+ double inystate_dG[4][PITCH_DAMPORDER];
+ double gain_mult[4];
+ const double *fracoeff = NULL;
+ double curgain, curlag, gaindelta, lagdelta;
+ double sum, sum2, inystate[PITCH_DAMPORDER];
+ double ftmp, oldlag, oldgain;
+ int k, n, m, m_tmp, j, pos, ind, pos2, Li, frc;
+
+ Li = 0;
+
+ /* Set up buffer and states */
+ memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
+
+ /* clear some buffers */
+ for (k = 0; k < 4; k++) {
+ gain_mult[k] = 0.0;
+ for (n = 0; n < PITCH_DAMPORDER; n++)
+ inystate_dG[k][n] = 0.0;
+ }
+
+ oldlag = *pfp->oldlagp;
+ oldgain = *pfp->oldgainp;
+
+ /* No interpolation if pitch lag step is big */
+ if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
+ oldlag = lags[0];
+ oldgain = gains[0];
+ gain_mult[0] = 1.0;
+ }
+
+
+ ind=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
+ curlag=oldlag ;
+ gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
+ curgain=oldgain ;
+ oldlag=lags[k];
+ oldgain=gains[k];
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+ if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
+ curgain += gaindelta;
+ curlag += lagdelta;
+ Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
+ ftmp = Li - (curlag+PITCH_FILTDELAY);
+ frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
+ fracoeff = kIntrpCoef[frc];
+ gain_mult[k] += 0.2;
+ if (gain_mult[k] > 1.0) gain_mult[k] = 1.0;
+ if (k > 0) gain_mult[k-1] -= 0.2;
+ }
+
+ /* shift low pass filter states */
+ for (m=PITCH_DAMPORDER-1;m>0;m--) {
+ inystate[m] = inystate[m-1];
+ for (j = 0; j < 4; j++)
+ inystate_dG[j][m] = inystate_dG[j][m-1];
+ }
+
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+
+ /* Filter to get fractional pitch */
+ sum=0.0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+ m_tmp = (Li-ind > 0) ? Li-ind : 0;
+ for (j = 0; j < k+1; j++) {
+ /* filter */
+ sum2 = 0.0;
+ for (m = PITCH_FRACORDER-1; m >= m_tmp; m--)
+ sum2 += out_dG[j][ind-Li + m] * fracoeff[m];
+ inystate_dG[j][0] = gain_mult[j] * sum + curgain * sum2;
+ }
+
+ /* Low pass filter */
+ sum=0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Subtract from input and update buffer */
+ outdat[ind] = indat[ind] - sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+
+ for (j = 0; j < k+1; j++) {
+ sum = 0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum -= inystate_dG[j][m] * kDampFilter[m];
+ out_dG[j][ind] = sum;
+ }
+ for (j = k+1; j < 4; j++)
+ out_dG[j][ind] = 0.0;
+
+
+ ind++;
+ }
+ }
+
+ /* Filter look-ahead segment */
+ for (n=0;n<QLOOKAHEAD;n++) {
+ /* shift low pass filter states */
+ for (m=PITCH_DAMPORDER-1;m>0;m--) {
+ inystate[m] = inystate[m-1];
+ for (j = 0; j < 4; j++)
+ inystate_dG[j][m] = inystate_dG[j][m-1];
+ }
+
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+
+ /* Filter to get fractional pitch */
+ sum=0.0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+ m_tmp = (Li-ind > 0) ? Li-ind : 0;
+ for (j = 0; (j<k+1)&&(j<4); j++) {
+ /* filter */
+ sum2 = 0.0;
+ for (m = PITCH_FRACORDER-1; m >= m_tmp; m--)
+ sum2 += out_dG[j][ind-Li + m] * fracoeff[m];
+ inystate_dG[j][0] = gain_mult[j] * sum + curgain * sum2;
+ }
+
+ /* Low pass filter */
+ sum=0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Subtract from input and update buffer */
+ outdat[ind] = indat[ind] - sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+
+ for (j = 0; (j<k+1)&&(j<4); j++) {
+ sum = 0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum -= inystate_dG[j][m] * kDampFilter[m];
+ out_dG[j][ind] = sum;
+ }
+
+ ind++;
+ }
+}
+
+
+void WebRtcIsac_PitchfilterPost(double *indat,
+ double *outdat,
+ PitchFiltstr *pfp,
+ double *lags,
+ double *gains)
+{
+
+ double ubuf[PITCH_INTBUFFSIZE];
+ const double *fracoeff = NULL;
+ double curgain, curlag, gaindelta, lagdelta;
+ double sum, inystate[PITCH_DAMPORDER];
+ double ftmp, oldlag, oldgain;
+ int k, n, m, pos, ind, pos2, Li, frc;
+
+ Li = 0;
+
+ /* Set up buffer and states */
+ memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
+
+ oldlag = *pfp->oldlagp;
+ oldgain = *pfp->oldgainp;
+
+ /* make output more periodic */
+ for (k=0;k<PITCH_SUBFRAMES;k++)
+ gains[k] *= 1.3;
+
+ /* No interpolation if pitch lag step is big */
+ if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
+ oldlag = lags[0];
+ oldgain = gains[0];
+ }
+
+
+ ind=0;
+ for (k=0;k<PITCH_SUBFRAMES;k++) {
+
+ /* Calculate interpolation steps */
+ lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
+ curlag=oldlag ;
+ gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
+ curgain=oldgain ;
+ oldlag=lags[k];
+ oldgain=gains[k];
+
+ for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
+ if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
+ curgain += gaindelta;
+ curlag += lagdelta;
+ Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
+ ftmp = Li - (curlag+PITCH_FILTDELAY);
+ frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
+ fracoeff = kIntrpCoef[frc];
+ }
+
+ /* shift low pass filter state */
+ for (m=PITCH_DAMPORDER-1;m>0;m--)
+ inystate[m] = inystate[m-1];
+
+ /* Filter to get fractional pitch */
+ pos = ind + PITCH_BUFFSIZE;
+ pos2 = pos - Li;
+ sum=0.0;
+ for (m=0;m<PITCH_FRACORDER;m++)
+ sum += ubuf[pos2+m] * fracoeff[m];
+ inystate[0] = curgain * sum; /* Multiply with gain */
+
+ /* Low pass filter */
+ sum=0.0;
+ for (m=0;m<PITCH_DAMPORDER;m++)
+ sum += inystate[m] * kDampFilter[m];
+
+ /* Add to input and update buffer */
+ outdat[ind] = indat[ind] + sum;
+ ubuf[pos] = indat[ind] + outdat[ind];
+ ind++;
+ }
+ }
+
+ /* Export buffer and states */
+ memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
+ memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
+
+ *pfp->oldlagp = oldlag;
+ *pfp->oldgainp = oldgain;
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c
new file mode 100644
index 0000000..5d998a2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c
@@ -0,0 +1,105 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pitch_gain_tables.h"
+
+#include "settings.h"
+
+/* header file for coding tables for the pitch filter side-info in the entropy coder */
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+/* cdf for quantized pitch filter gains */
+const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255] = {
+ 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956,
+ 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411,
+ 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722,
+ 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227,
+ 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098,
+ 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705,
+ 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029,
+ 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767,
+ 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132,
+ 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467,
+ 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165,
+ 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860,
+ 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901,
+ 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525,
+ 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3] = {
+ -7, -2, -1};
+
+const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3] = {
+ 0, 3, 1};
+
+const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2] = {
+ 18, 3};
+
+/* size of cdf table */
+const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1] = {
+ 256};
+
+///////////////////////////FIXED POINT
+/* mean values of pitch filter gains in FIXED point */
+const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144] = {
+ 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839, 1843, 1843, 1843, 1843, 1843,
+ 1843, 1843, 814, 846, 1092, 1013, 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843,
+ 1843, 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263, 1380, 1447, 1559, 1676,
+ 1645, 1749, 1843, 1843, 1843, 1843, 81, 477, 563, 611, 706, 806, 849, 1012, 1192, 1128,
+ 1330, 1489, 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488, 575, 602, 741,
+ 890, 835, 1079, 1196, 1182, 1376, 1519, 1506, 1680, 1843, 0, 47, 97, 69, 289, 381,
+ 385, 474, 617, 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0, 0, 0,
+ 112, 120, 190, 283, 442, 343, 526, 809, 684, 935, 1134, 1020, 1265, 1506, 0, 0,
+ 0, 0, 0, 0, 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020, 1265};
+
+const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144] = {
+ 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784, 1606, 1843, 1843, 1711, 1843,
+ 1843, 1814, 1389, 1275, 1040, 1564, 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720,
+ 1475, 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253, 1111, 1495, 1343, 1178,
+ 1770, 1465, 1234, 1814, 1581, 1342, 1040, 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277,
+ 1094, 981, 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648, 540, 948, 744,
+ 572, 1009, 844, 636, 1234, 934, 685, 1342, 1217, 984, 537, 318, 124, 603, 423, 350,
+ 687, 479, 322, 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27, 0, 397,
+ 222, 38, 513, 271, 124, 624, 325, 157, 737, 484, 233, 849, 597, 343, 27, 0,
+ 0, 141, 0, 0, 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343, 87};
+
+const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144] = {
+ 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639, 1393, 1760, 1525, 1285, 1656,
+ 1419, 1176, 1835, 1718, 1475, 1841, 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299,
+ 1040, 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260, 1115, 1398, 1151, 1025,
+ 1172, 1080, 790, 1176, 928, 677, 1475, 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057,
+ 893, 800, 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830, 710, 875, 751,
+ 601, 795, 642, 583, 790, 544, 475, 677, 474, 140, 987, 750, 482, 697, 573, 450,
+ 691, 487, 303, 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484, 229, 624,
+ 348, 153, 441, 261, 136, 397, 166, 51, 283, 27, 0, 168, 0, 0, 484, 229,
+ 0, 370, 57, 0, 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0, 0};
+
+
+const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144] = {
+ 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434, 1656, 843, 1092, 1336, 504,
+ 757, 1007, 1843, 1843, 1843, 1838, 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821,
+ 1092, 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268, 1409, 805, 961, 1131,
+ 444, 670, 843, 0, 249, 504, 1425, 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490,
+ 704, 867, 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073, 1283, 610, 759,
+ 939, 278, 464, 645, 0, 200, 270, 0, 0, 0, 935, 1163, 1410, 528, 790, 1068,
+ 377, 499, 717, 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935, 1182, 343,
+ 551, 735, 161, 262, 423, 0, 55, 27, 0, 0, 0, 0, 0, 0, 430, 684,
+ 935, 87, 377, 597, 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h
new file mode 100644
index 0000000..f958f5d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_gain_tables.h
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
+
+#include "typedefs.h"
+
+/* header file for coding tables for the pitch filter side-info in the entropy coder */
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+/* cdf for quantized pitch filter gains */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3];
+
+extern const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3];
+extern const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2];
+
+/* mean values of pitch filter gains */
+//(Y)
+extern const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144];
+extern const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144];
+extern const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144];
+extern const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144];
+//(Y)
+
+/* size of cdf table */
+extern const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c
new file mode 100644
index 0000000..72a031e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c
@@ -0,0 +1,277 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pitch_lag_tables.h"
+#include "settings.h"
+
+/* header file for coding tables for the pitch filter side-info in the entropy coder */
+/********************* Pitch Filter Gain Coefficient Tables ************************/
+
+/* tables for use with small pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127] = {
+ 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070,
+ 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269,
+ 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358,
+ 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213,
+ 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734,
+ 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288,
+ 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082,
+ 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273,
+ 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726,
+ 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359,
+ 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363,
+ 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856,
+ 65152, 65535, 65535, 65535, 65535, 65535, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20] = {
+ 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983,
+ 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2] = {
+ 0, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10] = {
+ 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535};
+
+const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4] = {WebRtcIsac_kQPitchLagCdf1Lo, WebRtcIsac_kQPitchLagCdf2Lo, WebRtcIsac_kQPitchLagCdf3Lo, WebRtcIsac_kQPitchLagCdf4Lo};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1] = {128};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4] = {
+-140, -9, 0, -4};
+
+const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4] = {
+-20, 9, 0, 4};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3] = {
+ 10, 1, 5};
+
+/* mean values of pitch filter lags */
+const double WebRtcIsac_kQMeanLag2Lo[19] = {
+-17.21385070, -15.82678944, -14.07123081, -12.03003877, -10.01311864, -8.00794627, -5.91162987, -3.89231876, -1.90220980, -0.01879275,
+ 1.89144232, 3.88123171, 5.92146992, 7.96435361, 9.98923648, 11.98266347, 13.96101002, 15.74855713, 17.10976611};
+
+const double WebRtcIsac_kQMeanLag3Lo[1] = {
+ 0.00000000};
+
+const double WebRtcIsac_kQMeanLag4Lo[9] = {
+-7.76246496, -5.92083980, -3.94095226, -1.89502305, 0.03724681, 1.93054221, 3.96443467, 5.91726366, 7.78434291};
+
+const double WebRtcIsac_kQPitchLagStepsizeLo = 2.000000;
+
+
+/* tables for use with medium pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255] = {
+ 0, 28, 61, 88, 121, 149, 233, 331, 475, 559,
+ 624, 661, 689, 712, 745, 791, 815, 843, 866, 922,
+ 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513,
+ 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770,
+ 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319,
+ 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185,
+ 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675,
+ 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421,
+ 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227,
+ 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861,
+ 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821,
+ 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285,
+ 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836,
+ 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741,
+ 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651,
+ 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048,
+ 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064,
+ 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755,
+ 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973,
+ 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602,
+ 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719,
+ 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688,
+ 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454,
+ 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902,
+ 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36] = {
+ 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608,
+ 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007,
+ 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949,
+ 65039, 65115, 65223, 65360, 65474, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2] = {
+ 0, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20] = {
+ 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347,
+ 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535};
+
+const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4] = {WebRtcIsac_kQPitchLagCdf1Mid, WebRtcIsac_kQPitchLagCdf2Mid, WebRtcIsac_kQPitchLagCdf3Mid, WebRtcIsac_kQPitchLagCdf4Mid};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1] = {256};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4] = {
+-280, -17, 0, -9};
+
+const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4] = {
+-40, 17, 0, 9};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3] = {
+ 18, 1, 10};
+
+/* mean values of pitch filter lags */
+const double WebRtcIsac_kQMeanLag2Mid[35] = {
+-16.89183900, -15.86949778, -15.05476653, -14.00664348, -13.02793036, -12.07324237, -11.00542532, -10.11250602, -8.90792971, -8.02474753,
+-7.00426767, -5.94055287, -4.98251338, -3.91053158, -2.98820425, -1.93524245, -0.92978085, -0.01722509, 0.91317387, 1.92973955,
+ 2.96908851, 3.93728974, 4.96308471, 5.92244151, 7.08673497, 8.00993708, 9.04656316, 9.98538742, 10.97851694, 11.94772884,
+ 13.02426166, 14.00039951, 15.01347042, 15.80758023, 16.94086895};
+
+const double WebRtcIsac_kQMeanLag3Mid[1] = {
+ 0.00000000};
+
+const double WebRtcIsac_kQMeanLag4Mid[19] = {
+-8.60409403, -7.89198395, -7.03450280, -5.86260421, -4.93822322, -3.93078706, -2.91302322, -1.91824007, -0.87003282, 0.02822649,
+ 0.89951758, 1.87495484, 2.91802604, 3.96874074, 5.06571703, 5.93618227, 7.00520185, 7.88497726, 8.64160364};
+
+const double WebRtcIsac_kQPitchLagStepsizeMid = 1.000000;
+
+
+/* tables for use with large pitch gain */
+
+/* cdf for quantized pitch filter lags */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511] = {
+ 0, 7, 18, 33, 69, 105, 156, 228, 315, 612,
+ 680, 691, 709, 724, 735, 738, 742, 746, 749, 753,
+ 756, 760, 764, 774, 782, 785, 789, 796, 800, 803,
+ 807, 814, 818, 822, 829, 832, 847, 854, 858, 869,
+ 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064,
+ 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136,
+ 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401,
+ 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477,
+ 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542,
+ 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665,
+ 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759,
+ 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987,
+ 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251,
+ 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378,
+ 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935,
+ 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127,
+ 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427,
+ 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760,
+ 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271,
+ 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904,
+ 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382,
+ 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022,
+ 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271,
+ 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024,
+ 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842,
+ 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536,
+ 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358,
+ 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324,
+ 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392,
+ 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384,
+ 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614,
+ 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555,
+ 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463,
+ 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649,
+ 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068,
+ 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588,
+ 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168,
+ 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077,
+ 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173,
+ 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894,
+ 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607,
+ 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078,
+ 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566,
+ 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693,
+ 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839,
+ 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558,
+ 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115,
+ 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
+ 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68] = {
+ 0, 7, 11, 22, 37, 52, 56, 59, 81, 85,
+ 89, 96, 115, 130, 137, 152, 170, 181, 193, 200,
+ 207, 233, 237, 259, 289, 318, 363, 433, 592, 992,
+ 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584,
+ 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402,
+ 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476,
+ 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2] = {
+ 0, 65535};
+
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35] = {
+ 0, 7, 19, 30, 41, 48, 63, 74, 82, 96,
+ 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341,
+ 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453, 65465,
+ 65476, 65490, 65509, 65528, 65535};
+
+const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4] = {WebRtcIsac_kQPitchLagCdf1Hi, WebRtcIsac_kQPitchLagCdf2Hi, WebRtcIsac_kQPitchLagCdf3Hi, WebRtcIsac_kQPitchLagCdf4Hi};
+
+/* size of first cdf table */
+const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1] = {512};
+
+/* index limits and ranges */
+const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4] = {
+-552, -34, 0, -16};
+
+const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4] = {
+-80, 32, 0, 17};
+
+/* initial index for arithmetic decoder */
+const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3] = {
+ 34, 1, 18};
+
+/* mean values of pitch filter lags */
+const double WebRtcIsac_kQMeanLag2Hi[67] = {
+-17.07263295, -16.50000000, -15.83966081, -15.55613708, -14.96948007, -14.50000000, -14.00000000, -13.48377986, -13.00000000, -12.50000000,
+-11.93199636, -11.44530414, -11.04197641, -10.39910301, -10.15202337, -9.51322461, -8.93357741, -8.46456632, -8.10270672, -7.53751847,
+-6.98686404, -6.50000000, -6.08463150, -5.46872991, -5.00864717, -4.50163760, -4.01382410, -3.43856708, -2.96898001, -2.46554810,
+-1.96861004, -1.47106701, -0.97197237, -0.46561654, -0.00531409, 0.45767857, 0.96777907, 1.47507903, 1.97740425, 2.46695420,
+ 3.00695774, 3.47167185, 4.02712538, 4.49280007, 5.01087640, 5.48191963, 6.04916550, 6.51511058, 6.97297819, 7.46565499,
+ 8.01489405, 8.39912001, 8.91819757, 9.50000000, 10.11654065, 10.50000000, 11.03712583, 11.50000000, 12.00000000, 12.38964346,
+ 12.89466127, 13.43657881, 13.96013840, 14.46279912, 15.00000000, 15.39412269, 15.96662441};
+
+const double WebRtcIsac_kQMeanLag3Hi[1] = {
+ 0.00000000};
+
+const double WebRtcIsac_kQMeanLag4Hi[34] = {
+-7.98331221, -7.47988769, -7.03626557, -6.52708003, -6.06982173, -5.51856292, -5.05827033, -4.45909878, -3.99125864, -3.45308135,
+-3.02328139, -2.47297273, -1.94341995, -1.44699056, -0.93612243, -0.43012406, 0.01120357, 0.44054812, 0.93199883, 1.45669587,
+ 1.97218322, 2.50187419, 2.98748690, 3.49343202, 4.01660147, 4.50984306, 5.01402683, 5.58936797, 5.91787793, 6.59998900,
+ 6.85034315, 7.53503316, 7.87711194, 8.53631648};
+
+const double WebRtcIsac_kQPitchLagStepsizeHi = 0.500000;
+
+/* transform matrix */
+const double WebRtcIsac_kTransform[4][4] = {
+{-0.50000000, -0.50000000, -0.50000000, -0.50000000},
+{ 0.67082039, 0.22360680, -0.22360680, -0.67082039},
+{ 0.50000000, -0.50000000, -0.50000000, 0.50000000},
+{ 0.22360680, -0.67082039, 0.67082039, -0.22360680}};
+
+/* transpose transform matrix */
+const double WebRtcIsac_kTransformTranspose[4][4] = {
+{-0.50000000, 0.67082039, 0.50000000, 0.22360680},
+{-0.50000000, 0.22360680, -0.50000000, -0.67082039},
+{-0.50000000, -0.22360680, -0.50000000, 0.67082039},
+{-0.50000000, -0.67082039, 0.50000000, -0.22360680}};
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h
new file mode 100644
index 0000000..67b02e5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_lag_tables.h
+ *
+ * This file contains tables for the pitch filter side-info in the entropy coder.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
+
+#include "typedefs.h"
+/* header file for coding tables for the pitch filter side-info in the entropy coder */
+/********************* Pitch Filter Lag Coefficient Tables ************************/
+
+/* tables for use with small pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10];
+
+extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4];
+extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3];
+
+/* mean values of pitch filter lags */
+extern const double WebRtcIsac_kQMeanLag2Lo[19];
+extern const double WebRtcIsac_kQMeanLag3Lo[1];
+extern const double WebRtcIsac_kQMeanLag4Lo[9];
+
+extern const double WebRtcIsac_kQPitchLagStepsizeLo;
+
+
+/* tables for use with medium pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20];
+
+extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4];
+extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3];
+
+/* mean values of pitch filter lags */
+extern const double WebRtcIsac_kQMeanLag2Mid[35];
+extern const double WebRtcIsac_kQMeanLag3Mid[1];
+extern const double WebRtcIsac_kQMeanLag4Mid[19];
+
+extern const double WebRtcIsac_kQPitchLagStepsizeMid;
+
+
+/* tables for use with large pitch gain */
+
+/* cdfs for quantized pitch lags */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2];
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35];
+
+extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4];
+
+/* size of first cdf table */
+extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1];
+
+/* index limits and ranges */
+extern const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4];
+extern const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4];
+
+/* initial index for arithmetic decoder */
+extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3];
+
+/* mean values of pitch filter lags */
+extern const double WebRtcIsac_kQMeanLag2Hi[67];
+extern const double WebRtcIsac_kQMeanLag3Hi[1];
+extern const double WebRtcIsac_kQMeanLag4Hi[34];
+
+extern const double WebRtcIsac_kQPitchLagStepsizeHi;
+
+/* transform matrix */
+extern const double WebRtcIsac_kTransform[4][4];
+
+/* transpose transform matrix */
+extern const double WebRtcIsac_kTransformTranspose[4][4];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h
new file mode 100644
index 0000000..b7aed77
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h
@@ -0,0 +1,199 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * settings.h
+ *
+ * Declaration of #defines used in the iSAC codec
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+
+/* sampling frequency (Hz) */
+#define FS 16000
+
+/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
+#define INITIAL_FRAMESAMPLES 960
+
+
+#define MAXFFTSIZE 2048
+#define NFACTOR 11
+
+
+
+/* do not modify the following; this will have to be modified if we have a 20ms framesize option */
+/*************************************************************************************************/
+/* miliseconds */
+#define FRAMESIZE 30
+/* number of samples per frame processed in the encoder, 480 */
+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
+#define FRAMESAMPLES_HALF 240
+#define FRAMESAMPLES_QUARTER 120
+/*************************************************************************************************/
+
+
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms ((10*FS)/1000)
+#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
+/* number of samples in 30 ms frame */
+#define FRAMESAMPLES_30ms 480
+/* number of subframes */
+#define SUBFRAMES 6
+/* length of a subframe */
+#define UPDATE 80
+/* length of half a subframe (low/high band) */
+#define HALF_SUBFRAMELEN (UPDATE/2)
+/* samples of look ahead (in a half-band, so actually half the samples of look ahead @ FS) */
+#define QLOOKAHEAD 24 /* 3 ms */
+/* order of AR model in spectral entropy coder */
+#define AR_ORDER 6
+/* order of LP model in spectral entropy coder */
+#define LP_ORDER 0
+
+/* window length (masking analysis) */
+#define WINLEN 256
+/* order of low-band pole filter used to approximate masking curve */
+#define ORDERLO 12
+/* order of hi-band pole filter used to approximate masking curve */
+#define ORDERHI 6
+
+#define UB_LPC_ORDER 4
+#define UB_LPC_VEC_PER_FRAME 2
+#define UB16_LPC_VEC_PER_FRAME 4
+#define UB_ACTIVE_SUBFRAMES 2
+#define UB_MAX_LPC_ORDER 6
+#define UB_INTERPOL_SEGMENTS 1
+#define UB16_INTERPOL_SEGMENTS 3
+#define LB_TOTAL_DELAY_SAMPLES 48
+enum ISACBandwidth {isac8kHz = 8, isac12kHz = 12, isac16kHz = 16};
+enum ISACBand{isacLowerBand = 0, isacUpperBand = 1};
+#define UB_LPC_GAIN_DIM SUBFRAMES
+#define FB_STATE_SIZE_WORD32 6
+
+
+/* order for post_filter_bank */
+#define POSTQORDER 3
+/* order for pre-filterbank */
+#define QORDER 3
+/* another order */
+#define QORDER_ALL (POSTQORDER+QORDER-1)
+/* for decimator */
+#define ALLPASSSECTIONS 2
+
+
+/* array size for byte stream in number of bytes. */
+#define STREAM_SIZE_MAX 600 /* The old maximum size still needed for the decoding */
+#define STREAM_SIZE_MAX_30 200 /* 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
+#define STREAM_SIZE_MAX_60 400 /* 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
+
+/* storage size for bit counts */
+#define BIT_COUNTER_SIZE 30
+/* maximum order of any AR model or filter */
+#define MAX_AR_MODEL_ORDER 12//50
+
+
+/* For pitch analysis */
+#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms */
+#define PITCH_MAX_LAG 140 /* 57 Hz */
+#define PITCH_MIN_LAG 20 /* 400 Hz */
+#define PITCH_MAX_GAIN 0.45
+#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
+#define PITCH_MAX_GAIN_Q12 1843
+#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5)
+#define PITCH_CORR_LEN2 60 /* 15 ms */
+#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN/4)
+#define PITCH_BW 11 /* half the band width of correlation surface */
+#define PITCH_SUBFRAMES 4
+#define PITCH_GRAN_PER_SUBFRAME 5
+#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN/PITCH_SUBFRAMES)
+#define PITCH_UPDATE (PITCH_SUBFRAME_LEN/PITCH_GRAN_PER_SUBFRAME)
+/* maximum number of peaks to be examined in correlation surface */
+#define PITCH_MAX_NUM_PEAKS 10
+#define PITCH_PEAK_DECAY 0.85
+/* For weighting filter */
+#define PITCH_WLPCORDER 6
+#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
+#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
+#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
+/* For pitch filter */
+#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50) /* Extra 50 for fraction and LP filters */
+#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN+PITCH_BUFFSIZE)
+/* Max rel. step for interpolation */
+#define PITCH_UPSTEP 1.5
+/* Max rel. step for interpolation */
+#define PITCH_DOWNSTEP 0.67
+#define PITCH_FRACS 8
+#define PITCH_FRACORDER 9
+#define PITCH_DAMPORDER 5
+#define PITCH_FILTDELAY 1.5f
+/* stepsize for quantization of the pitch Gain */
+#define PITCH_GAIN_STEPSIZE 0.125
+
+
+
+/* Order of high pass filter */
+#define HPORDER 2
+
+/* some mathematical constants */
+#define LOG2EXP 1.44269504088896 /* log2(exp) */
+#define PI 3.14159265358979
+
+/* Maximum number of iterations allowed to limit payload size */
+#define MAX_PAYLOAD_LIMIT_ITERATION 5
+
+/* Redundant Coding */
+#define RCU_BOTTLENECK_BPS 16000
+#define RCU_TRANSCODING_SCALE 0.40f
+#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
+
+#define RCU_TRANSCODING_SCALE_UB 0.50f
+#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
+
+
+/* Define Error codes */
+/* 6000 General */
+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
+#define ISAC_MODE_MISMATCH 6020
+#define ISAC_DISALLOWED_BOTTLENECK 6030
+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
+#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
+
+/* 6200 Bandwidth estimator */
+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
+/* 6400 Encoder */
+#define ISAC_ENCODER_NOT_INITIATED 6410
+#define ISAC_DISALLOWED_CODING_MODE 6420
+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
+#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
+/* 6600 Decoder */
+#define ISAC_DECODER_NOT_INITIATED 6610
+#define ISAC_EMPTY_PACKET 6620
+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
+#define ISAC_LENGTH_MISMATCH 6730
+#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
+#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
+/* 6800 Call setup formats */
+#define ISAC_INCOMPATIBLE_FORMATS 6810
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c
new file mode 100644
index 0000000..92b9c4d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c
@@ -0,0 +1,138 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "spectrum_ar_model_tables.h"
+#include "settings.h"
+
+/********************* AR Coefficient Tables ************************/
+/* cdf for quantized reflection coefficient 1 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12] = {
+ 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531,
+ 65533, 65535};
+
+/* cdf for quantized reflection coefficient 2 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12] = {
+ 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531,
+ 65533, 65535};
+
+/* cdf for quantized reflection coefficient 3 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12] = {
+ 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531,
+ 65533, 65535};
+
+/* cdf for quantized reflection coefficient 4 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12] = {
+ 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531,
+ 65533, 65535};
+
+/* cdf for quantized reflection coefficient 5 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12] = {
+ 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531,
+ 65533, 65535};
+
+/* cdf for quantized reflection coefficient 6 */
+const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12] = {
+ 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531,
+ 65533, 65535};
+
+/* representation levels for quantized reflection coefficient 1 */
+const WebRtc_Word16 WebRtcIsac_kQArRc1Levels[11] = {
+ -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 2 */
+const WebRtc_Word16 WebRtcIsac_kQArRc2Levels[11] = {
+ -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 3 */
+const WebRtc_Word16 WebRtcIsac_kQArRc3Levels[11] = {
+-32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 4 */
+const WebRtc_Word16 WebRtcIsac_kQArRc4Levels[11] = {
+-32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 5 */
+const WebRtc_Word16 WebRtcIsac_kQArRc5Levels[11] = {
+-32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
+};
+
+/* representation levels for quantized reflection coefficient 6 */
+const WebRtc_Word16 WebRtcIsac_kQArRc6Levels[11] = {
+-32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
+};
+
+/* quantization boundary levels for reflection coefficients */
+const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12] = {
+-32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, 32767
+};
+
+/* initial index for AR reflection coefficient quantizer and cdf table search */
+const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[6] = {
+ 5, 5, 5, 5, 5, 5};
+
+/* pointers to AR cdf tables */
+const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER] = {
+ WebRtcIsac_kQArRc1Cdf, WebRtcIsac_kQArRc2Cdf, WebRtcIsac_kQArRc3Cdf,
+ WebRtcIsac_kQArRc4Cdf, WebRtcIsac_kQArRc5Cdf, WebRtcIsac_kQArRc6Cdf
+};
+
+/* pointers to AR representation levels tables */
+const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER] = {
+ WebRtcIsac_kQArRc1Levels, WebRtcIsac_kQArRc2Levels, WebRtcIsac_kQArRc3Levels,
+ WebRtcIsac_kQArRc4Levels, WebRtcIsac_kQArRc5Levels, WebRtcIsac_kQArRc6Levels
+};
+
+
+/******************** GAIN Coefficient Tables ***********************/
+/* cdf for Gain coefficient */
+const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19] = {
+ 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172,
+ 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535};
+
+/* representation levels for quantized squared Gain coefficient */
+const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18] = {
+// 17, 28, 46, 76, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
+ 128, 128, 128, 128, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
+/* quantization boundary levels for squared Gain coefficient */
+const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19] = {
+0, 21, 35, 59, 99, 166, 280, 475, 815, 1414, 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF};
+
+/* pointers to Gain cdf table */
+const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1] = {WebRtcIsac_kQGainCdf};
+
+/* Gain initial index for gain quantizer and cdf table search */
+const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1] = {11};
+
+/************************* Cosine Tables ****************************/
+/* Cosine table */
+const WebRtc_Word16 WebRtcIsac_kCos[6][60] = {
+{512, 512, 511, 510, 508, 507, 505, 502, 499, 496, 493, 489, 485, 480, 476, 470, 465, 459, 453, 447,
+440, 433, 426, 418, 410, 402, 394, 385, 376, 367, 357, 348, 338, 327, 317, 306, 295, 284, 273, 262,
+250, 238, 226, 214, 202, 190, 177, 165, 152, 139, 126, 113, 100, 87, 73, 60, 47, 33, 20, 7},
+{512, 510, 508, 503, 498, 491, 483, 473, 462, 450, 437, 422, 406, 389, 371, 352, 333, 312, 290, 268,
+244, 220, 196, 171, 145, 120, 93, 67, 40, 13, -13, -40, -67, -93, -120, -145, -171, -196, -220, -244,
+-268, -290, -312, -333, -352, -371, -389, -406, -422, -437, -450, -462, -473, -483, -491, -498, -503, -508, -510, -512},
+{512, 508, 502, 493, 480, 465, 447, 426, 402, 376, 348, 317, 284, 250, 214, 177, 139, 100, 60, 20,
+-20, -60, -100, -139, -177, -214, -250, -284, -317, -348, -376, -402, -426, -447, -465, -480, -493, -502, -508, -512,
+-512, -508, -502, -493, -480, -465, -447, -426, -402, -376, -348, -317, -284, -250, -214, -177, -139, -100, -60, -20},
+{511, 506, 495, 478, 456, 429, 398, 362, 322, 279, 232, 183, 133, 80, 27, -27, -80, -133, -183, -232,
+-279, -322, -362, -398, -429, -456, -478, -495, -506, -511, -511, -506, -495, -478, -456, -429, -398, -362, -322, -279,
+-232, -183, -133, -80, -27, 27, 80, 133, 183, 232, 279, 322, 362, 398, 429, 456, 478, 495, 506, 511},
+{511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33, -33, -100, -165, -226, -284, -338, -385, -426,
+-459, -485, -502, -511, -511, -502, -485, -459, -426, -385, -338, -284, -226, -165, -100, -33, 33, 100, 165, 226,
+284, 338, 385, 426, 459, 485, 502, 511, 511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33},
+{510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510,
+-510, -498, -473, -437, -389, -333, -268, -196, -120, -40, 40, 120, 196, 268, 333, 389, 437, 473, 498, 510,
+510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510}
+};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h
new file mode 100644
index 0000000..159245b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * spectrum_ar_model_tables.h
+ *
+ * This file contains definitions of tables with AR coefficients,
+ * Gain coefficients and cosine tables.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+
+#include "structs.h"
+
+/********************* AR Coefficient Tables ************************/
+/* cdf for quantized reflection coefficient 1 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12];
+
+/* cdf for quantized reflection coefficient 2 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12];
+
+/* cdf for quantized reflection coefficient 3 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12];
+
+/* cdf for quantized reflection coefficient 4 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12];
+
+/* cdf for quantized reflection coefficient 5 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12];
+
+/* cdf for quantized reflection coefficient 6 */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12];
+
+/* quantization boundary levels for reflection coefficients */
+extern const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12];
+
+/* initial indices for AR reflection coefficient quantizer and cdf table search */
+extern const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[AR_ORDER];
+
+/* pointers to AR cdf tables */
+extern const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
+
+/* pointers to AR representation levels tables */
+extern const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
+
+
+/******************** GAIN Coefficient Tables ***********************/
+/* cdf for Gain coefficient */
+extern const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19];
+
+/* representation levels for quantized Gain coefficient */
+extern const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18];
+
+/* squared quantization boundary levels for Gain coefficient */
+extern const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19];
+
+/* pointer to Gain cdf table */
+extern const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1];
+
+/* Gain initial index for gain quantizer and cdf table search */
+extern const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1];
+
+/************************* Cosine Tables ****************************/
+/* Cosine table */
+extern const WebRtc_Word16 WebRtcIsac_kCos[6][60];
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h
new file mode 100644
index 0000000..7523ad6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h
@@ -0,0 +1,478 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+
+
+#include "typedefs.h"
+#include "settings.h"
+#include "isac.h"
+
+typedef struct Bitstreamstruct {
+
+ WebRtc_UWord8 stream[STREAM_SIZE_MAX];
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 streamval;
+ WebRtc_UWord32 stream_index;
+
+} Bitstr;
+
+typedef struct {
+
+ double DataBufferLo[WINLEN];
+ double DataBufferHi[WINLEN];
+
+ double CorrBufLo[ORDERLO+1];
+ double CorrBufHi[ORDERHI+1];
+
+ float PreStateLoF[ORDERLO+1];
+ float PreStateLoG[ORDERLO+1];
+ float PreStateHiF[ORDERHI+1];
+ float PreStateHiG[ORDERHI+1];
+ float PostStateLoF[ORDERLO+1];
+ float PostStateLoG[ORDERLO+1];
+ float PostStateHiF[ORDERHI+1];
+ float PostStateHiG[ORDERHI+1];
+
+ double OldEnergy;
+
+} MaskFiltstr;
+
+
+typedef struct {
+
+ //state vectors for each of the two analysis filters
+ double INSTAT1[2*(QORDER-1)];
+ double INSTAT2[2*(QORDER-1)];
+ double INSTATLA1[2*(QORDER-1)];
+ double INSTATLA2[2*(QORDER-1)];
+ double INLABUF1[QLOOKAHEAD];
+ double INLABUF2[QLOOKAHEAD];
+
+ float INSTAT1_float[2*(QORDER-1)];
+ float INSTAT2_float[2*(QORDER-1)];
+ float INSTATLA1_float[2*(QORDER-1)];
+ float INSTATLA2_float[2*(QORDER-1)];
+ float INLABUF1_float[QLOOKAHEAD];
+ float INLABUF2_float[QLOOKAHEAD];
+
+ /* High pass filter */
+ double HPstates[HPORDER];
+ float HPstates_float[HPORDER];
+
+} PreFiltBankstr;
+
+
+typedef struct {
+
+ //state vectors for each of the two analysis filters
+ double STATE_0_LOWER[2*POSTQORDER];
+ double STATE_0_UPPER[2*POSTQORDER];
+
+ /* High pass filter */
+ double HPstates1[HPORDER];
+ double HPstates2[HPORDER];
+
+ float STATE_0_LOWER_float[2*POSTQORDER];
+ float STATE_0_UPPER_float[2*POSTQORDER];
+
+ float HPstates1_float[HPORDER];
+ float HPstates2_float[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+
+ //data buffer for pitch filter
+ double ubuf[PITCH_BUFFSIZE];
+
+ //low pass state vector
+ double ystate[PITCH_DAMPORDER];
+
+ //old lag and gain
+ double oldlagp[1];
+ double oldgainp[1];
+
+} PitchFiltstr;
+
+typedef struct {
+
+ //data buffer
+ double buffer[PITCH_WLPCBUFLEN];
+
+ //state vectors
+ double istate[PITCH_WLPCORDER];
+ double weostate[PITCH_WLPCORDER];
+ double whostate[PITCH_WLPCORDER];
+
+ //LPC window -> should be a global array because constant
+ double window[PITCH_WLPCWINLEN];
+
+} WeightFiltstr;
+
+typedef struct {
+
+ //for inital estimator
+ double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
+ PITCH_MAX_LAG/2 - PITCH_FRAME_LEN/2+2];
+ double decimator_state[2*ALLPASSSECTIONS+1];
+ double hp_state[2];
+
+ double whitened_buf[QLOOKAHEAD];
+
+ double inbuf[QLOOKAHEAD];
+
+ PitchFiltstr PFstr_wght;
+ PitchFiltstr PFstr;
+ WeightFiltstr Wghtstr;
+
+} PitchAnalysisStruct;
+
+
+
+/* Have instance of struct together with other iSAC structs */
+typedef struct {
+
+ /* Previous frame length (in ms) */
+ WebRtc_Word32 prev_frame_length;
+
+ /* Previous RTP timestamp from received
+ packet (in samples relative beginning) */
+ WebRtc_Word32 prev_rec_rtp_number;
+
+ /* Send timestamp for previous packet (in ms using timeGetTime()) */
+ WebRtc_UWord32 prev_rec_send_ts;
+
+ /* Arrival time for previous packet (in ms using timeGetTime()) */
+ WebRtc_UWord32 prev_rec_arr_ts;
+
+ /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+ float prev_rec_rtp_rate;
+
+ /* Time sinse the last update of the BN estimate (in ms) */
+ WebRtc_UWord32 last_update_ts;
+
+ /* Time sinse the last reduction (in ms) */
+ WebRtc_UWord32 last_reduction_ts;
+
+ /* How many times the estimate was update in the beginning */
+ WebRtc_Word32 count_tot_updates_rec;
+
+ /* The estimated bottle neck rate from there to here (in bits/s) */
+ WebRtc_Word32 rec_bw;
+ float rec_bw_inv;
+ float rec_bw_avg;
+ float rec_bw_avg_Q;
+
+ /* The estimated mean absolute jitter value,
+ as seen on this side (in ms) */
+ float rec_jitter;
+ float rec_jitter_short_term;
+ float rec_jitter_short_term_abs;
+ float rec_max_delay;
+ float rec_max_delay_avg_Q;
+
+ /* (assumed) bitrate for headers (bps) */
+ float rec_header_rate;
+
+ /* The estimated bottle neck rate from here to there (in bits/s) */
+ float send_bw_avg;
+
+ /* The estimated mean absolute jitter value, as seen on
+ the other siee (in ms) */
+ float send_max_delay_avg;
+
+ // number of packets received since last update
+ int num_pkts_rec;
+
+ int num_consec_rec_pkts_over_30k;
+
+ // flag for marking that a high speed network has been
+ // detected downstream
+ int hsn_detect_rec;
+
+ int num_consec_snt_pkts_over_30k;
+
+ // flag for marking that a high speed network has
+ // been detected upstream
+ int hsn_detect_snd;
+
+ WebRtc_UWord32 start_wait_period;
+
+ int in_wait_period;
+
+ int change_to_WB;
+
+ WebRtc_UWord32 senderTimestamp;
+ WebRtc_UWord32 receiverTimestamp;
+ //enum IsacSamplingRate incomingStreamSampFreq;
+ WebRtc_UWord16 numConsecLatePkts;
+ float consecLatency;
+ WebRtc_Word16 inWaitLatePkts;
+} BwEstimatorstr;
+
+
+typedef struct {
+
+ /* boolean, flags if previous packet exceeded B.N. */
+ int PrevExceed;
+ /* ms */
+ int ExceedAgo;
+ /* packets left to send in current burst */
+ int BurstCounter;
+ /* packets */
+ int InitCounter;
+ /* ms remaining in buffer when next packet will be sent */
+ double StillBuffered;
+
+} RateModel;
+
+
+typedef struct {
+
+ unsigned int SpaceAlloced;
+ unsigned int MaxPermAlloced;
+ double Tmp0[MAXFFTSIZE];
+ double Tmp1[MAXFFTSIZE];
+ double Tmp2[MAXFFTSIZE];
+ double Tmp3[MAXFFTSIZE];
+ int Perm[MAXFFTSIZE];
+ int factor [NFACTOR];
+
+} FFTstr;
+
+
+/* The following strutc is used to store data from encoding, to make it
+ fast and easy to construct a new bitstream with a different Bandwidth
+ estimate. All values (except framelength and minBytes) is double size to
+ handle 60 ms of data.
+*/
+typedef struct {
+
+ /* Used to keep track of if it is first or second part of 60 msec packet */
+ int startIdx;
+
+ /* Frame length in samples */
+ WebRtc_Word16 framelength;
+
+ /* Pitch Gain */
+ int pitchGain_index[2];
+
+ /* Pitch Lag */
+ double meanGain[2];
+ int pitchIndex[PITCH_SUBFRAMES*2];
+
+ /* LPC */
+ int LPCmodel[2];
+ int LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
+ int LPCindex_g[12*2]; /* KLT_ORDER_GAIN = 12 */
+ double LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
+ double LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
+
+ /* Encode Spec */
+ WebRtc_Word16 fre[FRAMESAMPLES];
+ WebRtc_Word16 fim[FRAMESAMPLES];
+ WebRtc_Word16 AvgPitchGain[2];
+
+ /* Used in adaptive mode only */
+ int minBytes;
+
+} ISAC_SaveEncData_t;
+
+
+typedef struct {
+
+ int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+ double lpcGain[SUBFRAMES<<1];
+ int lpcGainIndex[SUBFRAMES<<1];
+
+ Bitstr bitStreamObj;
+
+ WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
+ WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
+} ISACUBSaveEncDataStruct;
+
+
+
+typedef struct {
+
+ Bitstr bitstr_obj;
+ MaskFiltstr maskfiltstr_obj;
+ PreFiltBankstr prefiltbankstr_obj;
+ PitchFiltstr pitchfiltstr_obj;
+ PitchAnalysisStruct pitchanalysisstr_obj;
+ FFTstr fftstr_obj;
+ ISAC_SaveEncData_t SaveEnc_obj;
+
+ int buffer_index;
+ WebRtc_Word16 current_framesamples;
+
+ float data_buffer_float[FRAMESAMPLES_30ms];
+
+ int frame_nb;
+ double bottleneck;
+ WebRtc_Word16 new_framelength;
+ double s2nr;
+
+ /* Maximum allowed number of bits for a 30 msec packet */
+ WebRtc_Word16 payloadLimitBytes30;
+ /* Maximum allowed number of bits for a 30 msec packet */
+ WebRtc_Word16 payloadLimitBytes60;
+ /* Maximum allowed number of bits for both 30 and 60 msec packet */
+ WebRtc_Word16 maxPayloadBytes;
+ /* Maximum allowed rate in bytes per 30 msec packet */
+ WebRtc_Word16 maxRateInBytes;
+
+ /*---
+ If set to 1 iSAC will not addapt the frame-size, if used in
+ channel-adaptive mode. The initial value will be used for all rates.
+ ---*/
+ WebRtc_Word16 enforceFrameSize;
+
+ /*-----
+ This records the BWE index the encoder injected into the bit-stream.
+ It will be used in RCU. The same BWE index of main paylaod will be in
+ the redundant payload. We can not retrive it from BWE because it is
+ a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
+ called only once per each encode.
+ -----*/
+ WebRtc_Word16 lastBWIdx;
+} ISACLBEncStruct;
+
+typedef struct {
+
+ Bitstr bitstr_obj;
+ MaskFiltstr maskfiltstr_obj;
+ PreFiltBankstr prefiltbankstr_obj;
+ FFTstr fftstr_obj;
+ ISACUBSaveEncDataStruct SaveEnc_obj;
+
+ int buffer_index;
+ float data_buffer_float[MAX_FRAMESAMPLES +
+ LB_TOTAL_DELAY_SAMPLES];
+ double bottleneck;
+ /* Maximum allowed number of bits for a 30 msec packet */
+ //WebRtc_Word16 payloadLimitBytes30;
+ /* Maximum allowed number of bits for both 30 and 60 msec packet */
+ //WebRtc_Word16 maxPayloadBytes;
+ WebRtc_Word16 maxPayloadSizeBytes;
+
+ double lastLPCVec[UB_LPC_ORDER];
+ WebRtc_Word16 numBytesUsed;
+ WebRtc_Word16 lastJitterInfo;
+} ISACUBEncStruct;
+
+
+
+typedef struct {
+
+ Bitstr bitstr_obj;
+ MaskFiltstr maskfiltstr_obj;
+ PostFiltBankstr postfiltbankstr_obj;
+ PitchFiltstr pitchfiltstr_obj;
+ FFTstr fftstr_obj;
+
+} ISACLBDecStruct;
+
+typedef struct {
+
+ Bitstr bitstr_obj;
+ MaskFiltstr maskfiltstr_obj;
+ PostFiltBankstr postfiltbankstr_obj;
+ FFTstr fftstr_obj;
+
+} ISACUBDecStruct;
+
+
+
+typedef struct {
+
+ ISACLBEncStruct ISACencLB_obj;
+ ISACLBDecStruct ISACdecLB_obj;
+} ISACLBStruct;
+
+
+typedef struct {
+
+ ISACUBEncStruct ISACencUB_obj;
+ ISACUBDecStruct ISACdecUB_obj;
+} ISACUBStruct;
+
+/*
+ This struct is used to take a snapshot of the entropy coder and LPC gains
+ right before encoding LPC gains. This allows us to go back to that state
+ if we like to limit the payload size.
+*/
+typedef struct {
+ /* 6 lower-band & 6 upper-band */
+ double loFiltGain[SUBFRAMES];
+ double hiFiltGain[SUBFRAMES];
+ /* Upper boundary of interval W */
+ WebRtc_UWord32 W_upper;
+ WebRtc_UWord32 streamval;
+ /* Index to the current position in bytestream */
+ WebRtc_UWord32 stream_index;
+ WebRtc_UWord8 stream[3];
+} transcode_obj;
+
+
+typedef struct {
+ // lower-band codec instance
+ ISACLBStruct instLB;
+ // upper-band codec instance
+ ISACUBStruct instUB;
+
+ // Bandwidth Estimator and model for the rate.
+ BwEstimatorstr bwestimator_obj;
+ RateModel rate_data_obj;
+ double MaxDelay;
+
+ /* 0 = adaptive; 1 = instantaneous */
+ WebRtc_Word16 codingMode;
+
+ // overall bottleneck of the codec
+ WebRtc_Word32 bottleneck;
+
+ // QMF Filter state
+ WebRtc_Word32 analysisFBState1[FB_STATE_SIZE_WORD32];
+ WebRtc_Word32 analysisFBState2[FB_STATE_SIZE_WORD32];
+ WebRtc_Word32 synthesisFBState1[FB_STATE_SIZE_WORD32];
+ WebRtc_Word32 synthesisFBState2[FB_STATE_SIZE_WORD32];
+
+ // Error Code
+ WebRtc_Word16 errorCode;
+
+ // bandwidth of the encoded audio 8, 12 or 16 kHz
+ enum ISACBandwidth bandwidthKHz;
+ // Sampling rate of audio, encoder and decode, 8 or 16 kHz
+ enum IsacSamplingRate encoderSamplingRateKHz;
+ enum IsacSamplingRate decoderSamplingRateKHz;
+ // Flag to keep track of initializations, lower & upper-band
+ // encoder and decoder.
+ WebRtc_Word16 initFlag;
+
+ // Flag to to indicate signal bandwidth switch
+ WebRtc_Word16 resetFlag_8kHz;
+
+ // Maximum allowed rate, measured in Bytes per 30 ms.
+ WebRtc_Word16 maxRateBytesPer30Ms;
+ // Maximum allowed payload-size, measured in Bytes.
+ WebRtc_Word16 maxPayloadSizeBytes;
+} ISACMainStruct;
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c
new file mode 100644
index 0000000..97b801a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c
@@ -0,0 +1,131 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "settings.h"
+#include "fft.h"
+#include "codec.h"
+#include "os_specific_inline.h"
+#include <math.h>
+
+static double costab1[FRAMESAMPLES_HALF];
+static double sintab1[FRAMESAMPLES_HALF];
+static double costab2[FRAMESAMPLES_QUARTER];
+static double sintab2[FRAMESAMPLES_QUARTER];
+
+void WebRtcIsac_InitTransform()
+{
+ int k;
+ double fact, phase;
+
+ fact = PI / (FRAMESAMPLES_HALF);
+ phase = 0.0;
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ costab1[k] = cos(phase);
+ sintab1[k] = sin(phase);
+ phase += fact;
+ }
+
+ fact = PI * ((double) (FRAMESAMPLES_HALF - 1)) / ((double) FRAMESAMPLES_HALF);
+ phase = 0.5 * fact;
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
+ costab2[k] = cos(phase);
+ sintab2[k] = sin(phase);
+ phase += fact;
+ }
+}
+
+
+void WebRtcIsac_Time2Spec(double *inre1,
+ double *inre2,
+ WebRtc_Word16 *outreQ7,
+ WebRtc_Word16 *outimQ7,
+ FFTstr *fftstr_obj)
+{
+
+ int k;
+ int dims[1];
+ double tmp1r, tmp1i, xr, xi, yr, yi, fact;
+ double tmpre[FRAMESAMPLES_HALF], tmpim[FRAMESAMPLES_HALF];
+
+
+ dims[0] = FRAMESAMPLES_HALF;
+
+
+ /* Multiply with complex exponentials and combine into one complex vector */
+ fact = 0.5 / sqrt(FRAMESAMPLES_HALF);
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ tmp1r = costab1[k];
+ tmp1i = sintab1[k];
+ tmpre[k] = (inre1[k] * tmp1r + inre2[k] * tmp1i) * fact;
+ tmpim[k] = (inre2[k] * tmp1r - inre1[k] * tmp1i) * fact;
+ }
+
+
+ /* Get DFT */
+ WebRtcIsac_Fftns(1, dims, tmpre, tmpim, -1, 1.0, fftstr_obj);
+
+ /* Use symmetry to separate into two complex vectors and center frames in time around zero */
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
+ xr = tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k];
+ yi = -tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k];
+ xi = tmpim[k] - tmpim[FRAMESAMPLES_HALF - 1 - k];
+ yr = tmpim[k] + tmpim[FRAMESAMPLES_HALF - 1 - k];
+
+ tmp1r = costab2[k];
+ tmp1i = sintab2[k];
+ outreQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0);
+ outimQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0);
+ outreQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0);
+ outimQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1r + yi * tmp1i) * 128.0);
+ }
+}
+
+
+void WebRtcIsac_Spec2time(double *inre, double *inim, double *outre1, double *outre2, FFTstr *fftstr_obj)
+{
+
+ int k;
+ double tmp1r, tmp1i, xr, xi, yr, yi, fact;
+
+ int dims;
+
+ dims = FRAMESAMPLES_HALF;
+
+ for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
+ /* Move zero in time to beginning of frames */
+ tmp1r = costab2[k];
+ tmp1i = sintab2[k];
+ xr = inre[k] * tmp1r + inim[k] * tmp1i;
+ xi = inim[k] * tmp1r - inre[k] * tmp1i;
+ yr = -inim[FRAMESAMPLES_HALF - 1 - k] * tmp1r - inre[FRAMESAMPLES_HALF - 1 - k] * tmp1i;
+ yi = -inre[FRAMESAMPLES_HALF - 1 - k] * tmp1r + inim[FRAMESAMPLES_HALF - 1 - k] * tmp1i;
+
+ /* Combine into one vector, z = x + j * y */
+ outre1[k] = xr - yi;
+ outre1[FRAMESAMPLES_HALF - 1 - k] = xr + yi;
+ outre2[k] = xi + yr;
+ outre2[FRAMESAMPLES_HALF - 1 - k] = -xi + yr;
+ }
+
+
+ /* Get IDFT */
+ WebRtcIsac_Fftns(1, &dims, outre1, outre2, 1, FRAMESAMPLES_HALF, fftstr_obj);
+
+
+ /* Demodulate and separate */
+ fact = sqrt(FRAMESAMPLES_HALF);
+ for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+ tmp1r = costab1[k];
+ tmp1i = sintab1[k];
+ xr = (outre1[k] * tmp1r - outre2[k] * tmp1i) * fact;
+ outre2[k] = (outre2[k] * tmp1r + outre1[k] * tmp1i) * fact;
+ outre1[k] = xr;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt
new file mode 100644
index 0000000..3f05224
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt
@@ -0,0 +1,433 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+
+
+if [ "$1" = "x64" ] || [ "$2" = "x64" ] || [ "$#" -eq 0 ]
+ then
+ PLATFORM=_X64
+ ISAC=../x64/Release/ReleaseTest-API_2005.exe
+elif [ "$1" = "LINUX" ] || [ "$2" = "LINUX" ]
+ then
+ PLATFORM=_linux
+ ISAC=../ReleaseTest-API/isacswtest
+else
+ PLATFORM=_2005
+ ISAC=../win32/Release/ReleaseTest-API_2005.exe
+fi
+
+if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "wb" ]
+ then
+ LOGFILE=logNormal"$PLATFORM".txt
+ echo "START ISAC WB TEST" > $LOGFILE
+ echo >> $LOGFILE
+
+ INFILES=$(cat InputFiles.txt)
+ SUBSET=$(cat InputFilesFew.txt)
+ CHANNELFILES=$(cat ChannelFiles.txt)
+ CHANNELLIST=($(cat ChannelFiles.txt))
+ INDIR=../data/orig
+ OUTDIR=../dataqa"$PLATFORM"
+ mkdir -p $OUTDIR
+ rm -f $OUTDIR/*
+
+ idx=0
+ RATE=10000
+ FRAMESIZE=30
+
+
+ for file in $INFILES # loop over all input files
+ do
+
+ echo "Input file: " $file
+ echo "-----------------------------------"
+ echo "Instantaneous with RATE " $RATE ", and Frame-size " $FRAMESIZE
+ $ISAC -I -B $RATE -FL $FRAMESIZE -FS 16 $INDIR/"$file" $OUTDIR/i_"$FRAMESIZE"_"$RATE"_"$file" >> $LOGFILE
+ echo
+
+ name="${CHANNELLIST[$idx]}"
+ echo "Adaptive with channel file: " $name
+
+ $ISAC -B $INDIR/${CHANNELLIST[$idx]} -FS 16 $INDIR/"$file" $OUTDIR/a_${name%.*}_"$file" >> $LOGFILE
+
+ echo
+ echo
+
+# alternate between 30 & 60 ms.
+ if [ $FRAMESIZE -eq 30 ]
+ then
+ FRAMESIZE=60
+ else
+ FRAMESIZE=30
+ fi
+
+# rate between 10000 to 32000 bits/sec
+ if [ $RATE -le 30000 ]
+ then
+ let "RATE=RATE+2000"
+ else
+ let "RATE=10000"
+ fi
+
+# there are only three channel file
+ if [ $idx -ge 2 ]; then
+ idx=0
+ else
+ let "idx=idx+1"
+ fi
+
+ done
+
+ idx=0
+
+# loop over the subset of input files
+ for file in $SUBSET
+ do
+
+ if [ $idx -eq 0 ]; then
+ $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a30_"$file" >> $LOGFILE
+ idx=1
+ else
+ $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a60_"$file" >> $LOGFILE
+ idx=0
+ fi
+ done
+
+ $ISAC -B $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 -FS 16 $INDIR/"$file" $OUTDIR/a60_Init25kbps_"$file" >> $LOGFILE
+
+ echo
+ echo WIDEBAND DONE!
+ echo
+ echo
+fi
+
+if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "swb" ]
+ then
+
+ LOGFILE=logNormal_SWB"$PLATFORM".txt
+ echo "START ISAC SWB TEST" > $LOGFILE
+ echo >> $LOGFILE
+
+ echo STARTING TO TEST SUPER-WIDEBAND
+
+ INFILES=$(cat InputFilesSWB.txt)
+ INDIR=../data/origswb
+ OUTDIR=../dataqaswb"$PLATFORM"
+ mkdir -p $OUTDIR
+ rm -f $OUTDIR/*
+
+ for file in $INFILES
+ do
+ echo
+ echo "Input file: " $file
+ echo "--------------------------------"
+ for RATE in 12000 20000 32000 38000 45000 50000 56000
+ do
+
+ echo "Rate " $RATE
+ $ISAC -I -B $RATE -FL 30 -FS 32 $INDIR/"$file" $OUTDIR/swb_"$RATE"_"$file" >> $LOGFILE
+ echo
+
+ done
+
+ done
+fi
+
+if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "API" ]
+ then
+
+ LOGFILE_API=logNormal_API"$PLATFORM".txt
+ echo
+ echo
+ echo "START ISAC API TEST" > $LOGFILE_API
+ echo >> $LOGFILE_API
+ idx=1
+ echo " Test Enforcement of frame-size"
+ echo "========================================================================================"
+ mkdir -p ../FrameSizeLim"$PLATFORM"
+ rm -f ../FrameSizeLim"$PLATFORM"/*
+ echo
+ echo "-- No enforcement; BN 10000"
+ echo
+ $ISAC -B 10000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Now Enforce 30 ms frame size with the same bottleneck"
+ echo "There should not be any 60 ms frame"
+ echo
+ $ISAC -B 10000 -FL 30 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- No enforcement; BN 32000"
+ echo
+ $ISAC -B 32000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Now Enforce 60 ms frame size with the same bottleneck"
+ echo "There should not be any 30 ms frame"
+ echo
+ $ISAC -B 32000 -FL 60 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo
+ echo
+ echo
+
+ echo " Test Limiting of Payload Size and Rate"
+ echo "========================================================================================"
+ mkdir -p ../PayloadLim"$PLATFORM"
+ rm -f ../PayloadLim"$PLATFORM"/*
+ echo
+ echo
+ echo "-- No Limit, frame-size 60 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 60 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Payload-size limit of 250, frame-size 60 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 60 -FS 16 -MAX 250 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Rate limit of 33 kbps for 60 ms frame-size"
+ echo
+ $ISAC -I -B 32000 -FL 60 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+
+ echo "-- No Limit, frame-size 30 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Payload-size limit of 130, frame-size 30 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 16 -MAX 130 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Rate limit of 33 kbps for 30 ms frame-size, wideband"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+
+ echo "-- No limit for 32 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Payload limit of 130 bytes for 32 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 32 -MAX 130 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+
+ echo "-- No limit, Rate 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
+ echo
+ $ISAC -I -B 45000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Rate limit of 46 kbps for 42 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
+ echo
+ $ISAC -I -B 45000 -FL 30 -FS 32 -MAXRATE 46000 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Payload limit of 170 bytes for 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
+ echo
+ $ISAC -I -B 45000 -FL 30 -FS 32 -MAX 170 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+
+ echo "-- No limit for 56 kbps, 30 ms, SUPER-WIDEBAND, 16 kHz"
+ echo
+ $ISAC -I -B 56000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Payload limit of 200 bytes for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
+ echo
+ $ISAC -I -B 56000 -FL 30 -FS 32 -MAX 200 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+ echo "-- Rate limit of 57 kbps for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
+ echo
+ $ISAC -I -B 56000 -FL 30 -FS 32 -MAXRATE 57000 ../data/origswb/jstest_32.pcm \
+ ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+ echo
+ echo
+ echo
+
+ echo " Test Trans-Coding"
+ echo "========================================================================================"
+ mkdir -p ../Transcoding"$PLATFORM"
+ rm -f ../Transcoding"$PLATFORM"/*
+ echo
+ echo
+ echo "-- 20 kbps, 30 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 20000 -FL 30 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
+ ../Transcoding"$PLATFORM"/APITest_refTrans20WB.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 32 kbps trans-coding to 20 kbps, 30 ms, WIDEBAND"
+ echo
+ $ISAC -I -B 32000 -FL 30 -FS 16 -T 20000 ../Transcoding"$PLATFORM"/APITest_32T20.pcm \
+ ../data/orig/speech_and_misc_WB.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+
+ echo
+ echo
+ echo "-- 38 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 38000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../Transcoding"$PLATFORM"/APITest_refTrans38.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 45 kbps trans-coding to 38 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 45000 -FL 30 -FS 32 -T 38000 ../Transcoding"$PLATFORM"/APITest_45T38.pcm \
+ ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+
+ echo
+ echo
+ echo "-- 20 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 20000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../Transcoding"$PLATFORM"/APITest_refTrans20SWB.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+
+ echo
+ echo
+
+ echo "-- 45 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 45000 -FL 30 -FS 32 -T 20000 ../Transcoding"$PLATFORM"/APITest_45T20.pcm \
+ ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+ echo "-- 50 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 50000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
+ ../Transcoding"$PLATFORM"/APITest_refTrans50.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 56 kbps trans-coding to 50 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 56000 -FL 30 -FS 32 -T 50000 ../Transcoding"$PLATFORM"/APITest_56T50.pcm \
+ ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 56 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
+ echo
+ $ISAC -I -B 56000 -FL 30 -FS 32 -T 20000 ../Transcoding"$PLATFORM"/APITest_56T20.pcm \
+ ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo "________________________________________________________"
+ echo
+ echo
+ echo
+ echo
+ echo
+
+ echo " Test FEC"
+ echo "========================================================================================"
+ mkdir -p ../FEC"$PLATFORM"
+ rm -f ../FEC"$PLATFORM"/*
+ echo
+ echo
+ echo "-- 32 kbps with transcoding to 20kbps, 30 ms, WIDEBAND, 10% packet loss"
+ $ISAC -I -B 32000 -FL 30 -FS 16 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_WB30_T20.pcm \
+ ../data/orig/speech_and_misc_WB.pcm ../FEC"$PLATFORM"/APITest_PL10_WB30.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 32 kbps, 60 ms, WIDEBAND, 10% packet loss"
+ $ISAC -I -B 32000 -FL 60 -FS 16 -PL 10 ../data/orig/speech_and_misc_WB.pcm \
+ ../FEC"$PLATFORM"/APITest_PL10_WB60.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 32 kbps with transcoding to 20 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
+ $ISAC -I -B 32000 -FL 30 -FS 32 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz_T20.pcm \
+ ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 45 kbps with Trascoding to 38 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
+ $ISAC -I -B 45000 -FL 30 -FS 32 -PL 10 -T 38000 ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz_T38.pcm \
+ ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+
+ echo "-- 56 kbps with transcoding to 50 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
+ $ISAC -I -B 56000 -FL 30 -FS 32 -PL 10 -T 50000 ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz_T50.pcm \
+ ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz.pcm >> $LOGFILE_API
+ let "idx=idx+1"
+ echo
+ echo
+fi
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt
new file mode 100644
index 0000000..63829a4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt
@@ -0,0 +1,80 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character????? ??????
+if [ "$1" = "x64" ] || [ "$#" -eq 0 ]
+ then
+ PLATFORM=_X64
+ ISAC=../x64/Release/ReleaseTest-API_2005.exe
+elif [ "$1" = "2005" ]
+ then
+ PLATFORM=_2005
+ ISAC=../win32/Release/ReleaseTest-API_2005.exe
+elif [ "$1" == "LINUX" ]
+ then
+ PLATFORM=_linux
+ ISAC=../ReleaseTest-API/isacswtest
+else
+ echo Unknown Platform
+ exit 2
+fi
+
+LOGFILE=logfault$PLATFORM.txt
+echo "START FAULT TEST" > $LOGFILE
+
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+INDIRSWB=../data/origswb
+OUTDIR=../dataqaft$PLATFORM
+mkdir -p $OUTDIR
+
+#maximum Target rate for different bandwidth
+TARGETRATE=( 32000 32000 44000 56000 )
+SAMPFREQ=( 16 32 32 32 )
+FAULTTEST=(1 2 3 4 5 6 7 9)
+
+index1=0
+
+file_wb=../data/orig/16kHz.pcm
+file_swb=../data/origswb/32kHz.pcm
+
+for idx in 0 1 2 3
+ do
+# Fault test
+ echo
+ echo "Sampling Frequency " ${SAMPFREQ[idx]} "kHz, Rate " ${TARGETRATE[idx]} "bps."
+ echo "---------------------------------------------------"
+ if [ ${SAMPFREQ[idx]} -eq 16 ]; then
+ file=$file_wb
+ else
+ file=$file_swb
+ fi
+
+ for testnr in ${FAULTTEST[*]}
+ do
+ echo "Running Fault Test " $testnr
+ $ISAC -I -B "${TARGETRATE[idx]}" -F $testnr -FS "${SAMPFREQ[idx]}" "$file" \
+ $OUTDIR/ft"$testnr"_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
+ echo
+
+ done
+
+# Fault test number 10, error in bitstream
+ echo "Running Fault Test 10"
+ $ISAC -I -B "${TARGETRATE[idx]}" -F 10 -FS "${SAMPFREQ[idx]}" "$file" \
+ $OUTDIR/ft10_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
+ echo
+ echo "Running Fault Test 10 with packetloss"
+ $ISAC -I -B "${TARGETRATE[idx]}" -F 10 -PL 10 -FS "${SAMPFREQ[idx]}" "$file" \
+ $OUTDIR/ft10plc_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
+ echo
+done
+
+echo
+echo
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt
new file mode 100644
index 0000000..4cda78e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt
@@ -0,0 +1,47 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+LOGFILE=logfxfl.txt
+echo "START FIX-FLOAT TEST" > $LOGFILE
+
+
+ISACFIXFLOAT=../../../fix/test/testFixFloat.exe
+
+INFILES=$(cat InputFiles.txt)
+SUBSET=$(cat InputFilesFew.txt)
+CHANNELFILES=$(cat ChannelFiles.txt)
+CHANNELLIST=($(cat ChannelFiles.txt))
+INDIR=../data/orig
+OUTDIR=../dataqafxfl
+mkdir -p $OUTDIR
+
+index1=0
+
+for file in $INFILES # loop over all input files
+ do
+
+ for channel in $CHANNELFILES
+ do
+ let "index1=index1+1"
+
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
+ done
+
+done
+
+index1=0
+
+for file in $SUBSET # loop over the subset of input files
+ do
+ let "index1=index1+1"
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
+ $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
+done
+
+echo DONE!
+
+
+
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc
new file mode 100644
index 0000000..19cff9e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -0,0 +1,1050 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// ReleaseTest-API.cpp : Defines the entry point for the console application.
+//
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <ctype.h>
+#include <iostream>
+
+/* include API */
+#include "isac.h"
+#include "utility.h"
+
+/* Defines */
+#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
+#define MAX_FRAMESAMPLES 960 /* max number of samples per frame
+ (= 60 ms frame & 16 kHz) or
+ (= 30 ms frame & 32 kHz) */
+#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
+#define SWBFRAMESAMPLES_10ms 320
+//#define FS 16000 /* sampling frequency (Hz) */
+
+#ifdef WIN32
+#define CLOCKS_PER_SEC 1000 /* Runtime statistics */
+#endif
+
+
+
+
+using namespace std;
+
+int main(int argc, char* argv[])
+{
+
+ char inname[100], outname[100], bottleneck_file[100], vadfile[100];
+ FILE *inp, *outp, *f_bn=NULL, *vadp, *bandwidthp;
+ int framecnt, endfile;
+
+ int i, errtype, VADusage = 0, packetLossPercent = 0;
+ WebRtc_Word16 CodingMode;
+ WebRtc_Word32 bottleneck;
+ WebRtc_Word16 framesize = 30; /* ms */
+ int cur_framesmpls, err;
+
+ /* Runtime statistics */
+ double starttime, runtime, length_file;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 declen, lostFrame = 0, declenTC = 0;
+
+ WebRtc_Word16 shortdata[SWBFRAMESAMPLES_10ms];
+ WebRtc_Word16 vaddata[SWBFRAMESAMPLES_10ms*3];
+ WebRtc_Word16 decoded[MAX_FRAMESAMPLES << 1];
+ WebRtc_Word16 decodedTC[MAX_FRAMESAMPLES << 1];
+ WebRtc_UWord16 streamdata[500];
+ WebRtc_Word16 speechType[1];
+ WebRtc_Word16 rateBPS = 0;
+ WebRtc_Word16 fixedFL = 0;
+ WebRtc_Word16 payloadSize = 0;
+ WebRtc_Word32 payloadRate = 0;
+ int setControlBWE = 0;
+ short FL, testNum;
+ char version_number[20];
+ FILE *plFile;
+ WebRtc_Word32 sendBN;
+
+#ifdef _DEBUG
+ FILE *fy;
+ double kbps;
+#endif /* _DEBUG */
+ int totalbits =0;
+ int totalsmpls =0;
+
+ /* If use GNS file */
+ FILE *fp_gns = NULL;
+ char gns_file[100];
+ short maxStreamLen30 = 0;
+ short maxStreamLen60 = 0;
+ short sampFreqKHz = 32;
+ short samplesIn10Ms;
+ short useAssign = 0;
+ //FILE logFile;
+ bool doTransCoding = false;
+ WebRtc_Word32 rateTransCoding = 0;
+ WebRtc_UWord16 streamDataTransCoding[600];
+ WebRtc_Word16 streamLenTransCoding = 0;
+ FILE* transCodingFile = NULL;
+ FILE* transcodingBitstream = NULL;
+ WebRtc_UWord32 numTransCodingBytes = 0;
+
+ /* only one structure used for ISAC encoder */
+ ISACStruct* ISAC_main_inst;
+ ISACStruct* decoderTransCoding;
+
+ BottleNeckModel BN_data;
+
+#ifdef _DEBUG
+ fy = fopen("bit_rate.dat", "w");
+ fclose(fy);
+ fy = fopen("bytes_frames.dat", "w");
+ fclose(fy);
+#endif /* _DEBUG */
+
+ /* Handling wrong input arguments in the command line */
+ if((argc<3) || (argc>17)) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ WebRtcIsac_version(version_number);
+ printf("iSAC-swb version %s \n\n", version_number);
+
+ printf("Usage:\n\n");
+ printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
+ printf("with:\n");
+ printf("[-FS num] : sampling frequency in kHz, valid values are 16 & 32,\n");
+ printf(" with 16 as default.\n");
+ printf("[-I] : if -I option is specified, the coder will use\n");
+ printf(" an instantaneous Bottleneck value. If not, it\n");
+ printf(" will be an adaptive Bottleneck value.\n\n");
+ printf("[-assign] : Use Assign API.\n");
+ printf("[-B num] : the value of the bottleneck provided either\n");
+ printf(" as a fixed value in bits/sec (e.g. 25000) or\n");
+ printf(" read from a file (e.g. bottleneck.txt)\n\n");
+ printf("[-INITRATE num] : Set a new value for initial rate. Note! Only used in \n");
+ printf(" adaptive mode.\n\n");
+ printf("[-FL num] : Set (initial) frame length in msec. Valid length are \n");
+ printf(" 30 and 60 msec.\n\n");
+ printf("[-FIXED_FL] : Frame length will be fixed to initial value.\n\n");
+ printf("[-MAX num] : Set the limit for the payload size of iSAC in bytes. \n");
+ printf(" Minimum 100 maximum 400.\n\n");
+ printf("[-MAXRATE num] : Set the maxrate for iSAC in bits per second. \n");
+ printf(" Minimum 32000, maximum 53400.\n\n");
+ printf("[-F num] : if -F option is specified, the test function\n");
+ printf(" will run the iSAC API fault scenario specified by the\n");
+ printf(" supplied number.\n");
+ printf(" F 1 - Call encoder prior to init encoder call\n");
+ printf(" F 2 - Call decoder prior to init decoder call\n");
+ printf(" F 3 - Call decoder prior to encoder call\n");
+ printf(" F 4 - Call decoder with a too short coded sequence\n");
+ printf(" F 5 - Call decoder with a too long coded sequence\n");
+ printf(" F 6 - Call decoder with random bit stream\n");
+ printf(" F 7 - Call init encoder/decoder at random during a call\n");
+ printf(" F 8 - Call encoder/decoder without having allocated memory \n");
+ printf(" for encoder/decoder instance\n");
+ printf(" F 9 - Call decodeB without calling decodeA\n");
+ printf(" F 10 - Call decodeB with garbage data\n");
+ printf("[-PL num] : if -PL option is specified \n");
+ printf("[-T rate file] : test trans-coding with target bottleneck 'rate' bits/sec\n");
+ printf(" the output file is written to 'file'\n");
+ printf("[-LOOP num] : number of times to repeat coding the input file for stress testing\n");
+ //printf("[-CE num] : Test of APIs used by Conference Engine.\n");
+ //printf(" CE 1 - getNewBitstream, getBWE \n");
+ //printf(" (CE 2 - RESERVED for transcoding)\n");
+ //printf(" CE 3 - getSendBWE, setSendBWE. \n\n");
+ //printf("-L filename : write the logging info into file (appending)\n");
+ printf("infile : Normal speech input file\n\n");
+ printf("outfile : Speech output file\n\n");
+ exit(0);
+ }
+
+ /* Print version number */
+ printf("-------------------------------------------------\n");
+ WebRtcIsac_version(version_number);
+ printf("iSAC version %s \n\n", version_number);
+
+ /* Loop over all command line arguments */
+ CodingMode = 0;
+ testNum = 0;
+ useAssign = 0;
+ //logFile = NULL;
+ char transCodingFileName[500];
+ WebRtc_Word16 totFileLoop = 0;
+ WebRtc_Word16 numFileLoop = 0;
+ for (i = 1; i < argc-2;i++)
+ {
+ if(!strcmp("-LOOP", argv[i]))
+ {
+ i++;
+ totFileLoop = (WebRtc_Word16)atol(argv[i]);
+ if(totFileLoop <= 0)
+ {
+ fprintf(stderr, "Invalid number of runs for the given input file, %d.", totFileLoop);
+ exit(0);
+ }
+ }
+
+ if(!strcmp("-T", argv[i]))
+ {
+ doTransCoding = true;
+ i++;
+ rateTransCoding = atoi(argv[i]);
+ i++;
+ strcpy(transCodingFileName, argv[i]);
+ }
+
+ /*Should we use assign API*/
+ if(!strcmp("-assign", argv[i]))
+ {
+ useAssign = 1;
+ }
+
+ /* Set Sampling Rate */
+ if(!strcmp("-FS", argv[i]))
+ {
+ i++;
+ sampFreqKHz = atoi(argv[i]);
+ }
+
+ /* Instantaneous mode */
+ if(!strcmp ("-I", argv[i]))
+ {
+ printf("Instantaneous BottleNeck\n");
+ CodingMode = 1;
+ }
+
+ /* Set (initial) bottleneck value */
+ if(!strcmp ("-INITRATE", argv[i])) {
+ rateBPS = atoi(argv[i + 1]);
+ setControlBWE = 1;
+ if((rateBPS < 10000) || (rateBPS > 32000))
+ {
+ printf("\n%d is not a initial rate. Valid values are in the range 10000 to 32000.\n", rateBPS);
+ exit(0);
+ }
+ printf("New initial rate: %d\n", rateBPS);
+ i++;
+ }
+
+ /* Set (initial) framelength */
+ if(!strcmp ("-FL", argv[i])) {
+ framesize = atoi(argv[i + 1]);
+ if((framesize != 30) && (framesize != 60))
+ {
+ printf("\n%d is not a valid frame length. Valid length are 30 and 60 msec.\n", framesize);
+ exit(0);
+ }
+ setControlBWE = 1;
+ printf("Frame Length: %d\n", framesize);
+ i++;
+ }
+
+ /* Fixed frame length */
+ if(!strcmp ("-FIXED_FL", argv[i]))
+ {
+ fixedFL = 1;
+ setControlBWE = 1;
+ printf("Fixed Frame Length\n");
+ }
+
+ /* Set maximum allowed payload size in bytes */
+ if(!strcmp ("-MAX", argv[i])) {
+ payloadSize = atoi(argv[i + 1]);
+ printf("Maximum Payload Size: %d\n", payloadSize);
+ i++;
+ }
+
+ /* Set maximum rate in bytes */
+ if(!strcmp ("-MAXRATE", argv[i])) {
+ payloadRate = atoi(argv[i + 1]);
+ printf("Maximum Rate in kbps: %d\n", payloadRate);
+ i++;
+ }
+
+ /* Test of fault scenarious */
+ if(!strcmp ("-F", argv[i]))
+ {
+ testNum = atoi(argv[i + 1]);
+ printf("Fault test: %d\n", testNum);
+ if(testNum < 1 || testNum > 10)
+ {
+ printf("\n%d is not a valid Fault Scenario number. Valid Fault Scenarios are numbered 1-10.\n", testNum);
+ exit(0);
+ }
+ i++;
+ }
+
+ /* Packet loss test */
+ if(!strcmp ("-PL", argv[i]))
+ {
+ if( isdigit( *argv[i+1] ) )
+ {
+ packetLossPercent = atoi( argv[i+1] );
+ if( (packetLossPercent < 0) | (packetLossPercent > 100) )
+ {
+ printf( "\nInvalid packet loss perentage \n" );
+ exit( 0 );
+ }
+ if( packetLossPercent > 0 )
+ {
+ printf( "Simulating %d %% of independent packet loss\n", packetLossPercent );
+ }
+ else
+ {
+ printf( "\nNo Packet Loss Is Simulated \n" );
+ }
+ }
+ else
+ {
+ plFile = fopen( argv[i+1], "rb" );
+ if( plFile == NULL )
+ {
+ printf( "\n couldn't open the frameloss file: %s\n", argv[i+1] );
+ exit( 0 );
+ }
+ printf( "Simulating packet loss through the given channel file: %s\n", argv[i+1] );
+ }
+ i++;
+ }
+
+ /* Random packetlosses */
+ if(!strcmp ("-rnd", argv[i]))
+ {
+ srand((unsigned int)time(NULL) );
+ printf( "Random pattern in lossed packets \n" );
+ }
+
+ /* Use gns file */
+ if(!strcmp ("-G", argv[i]))
+ {
+ sscanf(argv[i + 1], "%s", gns_file);
+ fp_gns = fopen(gns_file, "rb");
+ if(fp_gns == NULL)
+ {
+ printf("Cannot read file %s.\n", gns_file);
+ exit(0);
+ }
+ i++;
+ }
+
+
+ // make it with '-B'
+ /* Get Bottleneck value */
+ if(!strcmp("-B", argv[i]))
+ {
+ i++;
+ bottleneck = atoi(argv[i]);
+ if(bottleneck == 0)
+ {
+ sscanf(argv[i], "%s", bottleneck_file);
+ f_bn = fopen(bottleneck_file, "rb");
+ if(f_bn == NULL)
+ {
+ printf("Error No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
+ exit(0);
+ }
+ else
+ {
+ printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
+ if(fscanf(f_bn, "%d", &bottleneck) == EOF)
+ {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+ exit(0);
+ }
+ }
+
+ /* Bottleneck is a cosine function
+ * Matlab code for writing the bottleneck file:
+ * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
+ * fid = fopen('bottleneck.txt', 'wb');
+ * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
+ */
+ }
+ }
+ else
+ {
+ printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+ }
+ }
+ /* Run Conference Engine APIs */
+ // Do not test it in the first release
+ //
+ // if(!strcmp ("-CE", argv[i]))
+ // {
+ // testCE = atoi(argv[i + 1]);
+ // if(testCE==1)
+ // {
+ // i++;
+ // scale = (float)atof( argv[i+1] );
+ // }
+ // else if(testCE == 2)
+ // {
+ // printf("\nCE-test 2 (transcoding) not implemented.\n");
+ // exit(0);
+ // }
+ // else if(testCE < 1 || testCE > 3)
+ // {
+ // printf("\n%d is not a valid CE-test number. Valid CE tests are 1-3.\n", testCE);
+ // exit(0);
+ // }
+ // printf("CE-test number: %d\n", testCE);
+ // i++;
+ // }
+ }
+
+ if(CodingMode == 0)
+ {
+ printf("\nAdaptive BottleNeck\n");
+ }
+
+ switch(sampFreqKHz)
+ {
+ case 16:
+ {
+ printf("iSAC Wideband.\n");
+ samplesIn10Ms = FRAMESAMPLES_10ms;
+ break;
+ }
+ case 32:
+ {
+ printf("iSAC Supper-Wideband.\n");
+ samplesIn10Ms = SWBFRAMESAMPLES_10ms;
+ break;
+ }
+ default:
+ printf("Unsupported sampling frequency %d kHz", sampFreqKHz);
+ exit(0);
+ }
+
+
+
+
+ /* Get Input and Output files */
+ sscanf(argv[argc-2], "%s", inname);
+ sscanf(argv[argc-1], "%s", outname);
+ printf("\nInput file: %s\n", inname);
+ printf("Output file: %s\n\n", outname);
+ if((inp = fopen(inname,"rb")) == NULL)
+ {
+ printf(" Error iSAC Cannot read file %s.\n", inname);
+ cout << flush;
+ exit(1);
+ }
+
+ if((outp = fopen(outname,"wb")) == NULL)
+ {
+ printf(" Error iSAC Cannot write file %s.\n", outname);
+ cout << flush;
+ getchar();
+ exit(1);
+ }
+ if(VADusage)
+ {
+ if((vadp = fopen(vadfile,"rb")) == NULL)
+ {
+ printf(" Error iSAC Cannot read file %s.\n", vadfile);
+ cout << flush;
+ exit(1);
+ }
+ }
+
+ if((bandwidthp = fopen("bwe.pcm","wb")) == NULL)
+ {
+ printf(" Error iSAC Cannot read file %s.\n", "bwe.pcm");
+ cout << flush;
+ exit(1);
+ }
+
+
+ starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+
+ /* Initialize the ISAC and BN structs */
+ if(testNum != 8)
+ {
+ if(!useAssign)
+ {
+ err =WebRtcIsac_Create(&ISAC_main_inst);
+ WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ }
+ else
+ {
+ /* Test the Assign functions */
+ int sss;
+ void *ppp;
+ err = WebRtcIsac_AssignSize(&sss);
+ ppp = malloc(sss);
+ err = WebRtcIsac_Assign(&ISAC_main_inst, ppp);
+ WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ }
+ /* Error check */
+ if(err < 0)
+ {
+ printf("\n\n Error in create.\n\n");
+ cout << flush;
+ exit(EXIT_FAILURE);
+ }
+ }
+ BN_data.arrival_time = 0;
+ BN_data.sample_count = 0;
+ BN_data.rtp_number = 0;
+
+ /* Initialize encoder and decoder */
+ framecnt= 0;
+ endfile = 0;
+
+ if(doTransCoding)
+ {
+ WebRtcIsac_Create(&decoderTransCoding);
+ WebRtcIsac_SetEncSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ WebRtcIsac_SetDecSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
+ WebRtcIsac_DecoderInit(decoderTransCoding);
+ transCodingFile = fopen(transCodingFileName, "wb");
+ if(transCodingFile == NULL)
+ {
+ printf("Could not open %s to output trans-coding.\n", transCodingFileName);
+ exit(0);
+ }
+ strcat(transCodingFileName, ".bit");
+ transcodingBitstream = fopen(transCodingFileName, "wb");
+ if(transcodingBitstream == NULL)
+ {
+ printf("Could not open %s to write the bit-stream of transcoder.\n", transCodingFileName);
+ exit(0);
+ }
+ }
+
+ if(testNum != 1)
+ {
+ if(WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode) < 0)
+ {
+ printf("Error could not initialize the encoder \n");
+ cout << flush;
+ return 0;
+ }
+ }
+ if(testNum != 2)
+ {
+ if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
+ {
+ printf("Error could not initialize the decoder \n");
+ cout << flush;
+ return 0;
+ }
+ }
+ if(CodingMode == 1)
+ {
+ err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in initialization (control): %d.\n\n", errtype);
+ cout << flush;
+ if(testNum == 0)
+ {
+ exit(EXIT_FAILURE);
+ }
+ }
+ }
+
+ if((setControlBWE) && (CodingMode == 0))
+ {
+ err = WebRtcIsac_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+
+ printf("\n\n Error in Control BWE: %d.\n\n", errtype);
+ cout << flush;
+ exit(EXIT_FAILURE);
+ }
+ }
+
+ if(payloadSize != 0)
+ {
+ err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
+ cout << flush;
+ exit(EXIT_FAILURE);
+ }
+ }
+ if(payloadRate != 0)
+ {
+ err = WebRtcIsac_SetMaxRate(ISAC_main_inst, payloadRate);
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
+ cout << flush;
+ exit(EXIT_FAILURE);
+ }
+ }
+
+ *speechType = 1;
+
+ cout << "\n" << flush;
+
+ length_file = 0;
+ WebRtc_Word16 bnIdxTC;
+ WebRtc_Word16 jitterInfoTC;
+ while (endfile == 0)
+ {
+ /* Call init functions at random, fault test number 7 */
+ if(testNum == 7 && (rand()%2 == 0))
+ {
+ err = WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+ /* Error check */
+ if(err < 0)
+ {
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in encoderinit: %d.\n\n", errtype);
+ cout << flush;
+ }
+
+ err = WebRtcIsac_DecoderInit(ISAC_main_inst);
+ /* Error check */
+ if(err < 0)
+ {
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\n Error in decoderinit: %d.\n\n", errtype);
+ cout << flush;
+ }
+ }
+
+ cur_framesmpls = 0;
+ while (1)
+ {
+ /* Read 10 ms speech block */
+ endfile = readframe(shortdata, inp, samplesIn10Ms);
+
+ if(endfile)
+ {
+ numFileLoop++;
+ if(numFileLoop < totFileLoop)
+ {
+ rewind(inp);
+ framecnt = 0;
+ fprintf(stderr, "\n");
+ endfile = readframe(shortdata, inp, samplesIn10Ms);
+ }
+ }
+
+ if(testNum == 7)
+ {
+ srand((unsigned int)time(NULL));
+ }
+
+ /* iSAC encoding */
+ if(!(testNum == 3 && framecnt == 0))
+ {
+ stream_len = WebRtcIsac_Encode(ISAC_main_inst,
+ shortdata,
+ (WebRtc_Word16*)streamdata);
+ if((payloadSize != 0) && (stream_len > payloadSize))
+ {
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+
+ printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
+ cout << flush;
+ }
+
+ WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
+
+ if(stream_len>0)
+ {
+ if(doTransCoding)
+ {
+ WebRtc_Word16 indexStream;
+ WebRtc_UWord8 auxUW8;
+
+ /************************* Main Transcoding stream *******************************/
+ WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, &jitterInfoTC);
+ streamLenTransCoding = WebRtcIsac_GetNewBitStream(
+ ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
+ (WebRtc_Word16*)streamDataTransCoding, false);
+ if(streamLenTransCoding < 0)
+ {
+ fprintf(stderr, "Error in trans-coding\n");
+ exit(0);
+ }
+ auxUW8 = (WebRtc_UWord8)(((streamLenTransCoding & 0xFF00) >> 8) & 0x00FF);
+ fwrite(&auxUW8, sizeof(WebRtc_UWord8),
+ 1, transcodingBitstream);
+
+ auxUW8 = (WebRtc_UWord8)(streamLenTransCoding & 0x00FF);
+ fwrite(&auxUW8, sizeof(WebRtc_UWord8),
+ 1, transcodingBitstream);
+
+ fwrite((WebRtc_UWord8*)streamDataTransCoding, sizeof(WebRtc_UWord8),
+ streamLenTransCoding, transcodingBitstream);
+
+ WebRtcIsac_ReadBwIndex((WebRtc_Word16*)streamDataTransCoding, &indexStream);
+ if(indexStream != bnIdxTC)
+ {
+ fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n");
+ exit(0);
+ }
+ numTransCodingBytes += streamLenTransCoding;
+ }
+ }
+ }
+ else
+ {
+ break;
+ }
+
+ if(stream_len < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ printf("\n\nError in encoder: %d.\n\n", errtype);
+ cout << flush;
+ }
+ cur_framesmpls += samplesIn10Ms;
+ /* exit encoder loop if the encoder returned a bitstream */
+ if(stream_len != 0) break;
+ }
+
+ /* read next bottleneck rate */
+ if(f_bn != NULL)
+ {
+ if(fscanf(f_bn, "%d", &bottleneck) == EOF)
+ {
+ /* Set pointer to beginning of file */
+ fseek(f_bn, 0L, SEEK_SET);
+ if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+ exit(0);
+ }
+ }
+ if(CodingMode == 1)
+ {
+ WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+ }
+ }
+
+ length_file += cur_framesmpls;
+ if(cur_framesmpls == (3 * samplesIn10Ms))
+ {
+ maxStreamLen30 = (stream_len > maxStreamLen30)? stream_len:maxStreamLen30;
+ }
+ else
+ {
+ maxStreamLen60 = (stream_len > maxStreamLen60)? stream_len:maxStreamLen60;
+ }
+
+ if(!lostFrame)
+ {
+ lostFrame = ((rand()%100) < packetLossPercent);
+ }
+ else
+ {
+ lostFrame = 0;
+ }
+
+ // RED.
+ if(lostFrame)
+ {
+ stream_len = WebRtcIsac_GetRedPayload(ISAC_main_inst,
+ (WebRtc_Word16*)streamdata);
+
+ if(doTransCoding)
+ {
+ streamLenTransCoding = WebRtcIsac_GetNewBitStream(
+ ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
+ (WebRtc_Word16*)streamDataTransCoding, true);
+ if(streamLenTransCoding < 0)
+ {
+ fprintf(stderr, "Error in RED trans-coding\n");
+ exit(0);
+ }
+ }
+ }
+
+ /* make coded sequence to short be inreasing */
+ /* the length the decoder expects */
+ if(testNum == 4)
+ {
+ stream_len += 10;
+ }
+
+ /* make coded sequence to long be decreasing */
+ /* the length the decoder expects */
+ if(testNum == 5)
+ {
+ stream_len -= 10;
+ }
+
+ if(testNum == 6)
+ {
+ srand((unsigned int)time(NULL));
+ for(i = 0; i < stream_len; i++)
+ {
+ streamdata[i] = rand();
+ }
+ }
+
+ if(VADusage){
+ readframe(vaddata, vadp, samplesIn10Ms*3);
+ }
+
+ /* simulate packet handling through NetEq and the modem */
+ if(!(testNum == 3 && framecnt == 0))
+ {
+ get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data,
+ sampFreqKHz*1000, sampFreqKHz*1000);
+ }
+
+ if(VADusage && (framecnt>10 && vaddata[0]==0))
+ {
+ BN_data.rtp_number--;
+ }
+ else
+ {
+ /* Error test number 10, garbage data */
+ if(testNum == 10)
+ {
+ /* Test to run decoder with garbage data */
+ for(i = 0; i < stream_len; i++)
+ {
+ streamdata[i] = (short) (streamdata[i]) + (short) rand();
+ }
+ }
+
+ if(testNum != 9)
+ {
+ err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, streamdata,
+ stream_len, BN_data.rtp_number, BN_data.sample_count,
+ BN_data.arrival_time);
+
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+
+ printf("Error: in decoder: %d.", errtype);
+ cout << flush;
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+
+ }
+ }
+
+ /* Call getFramelen, only used here for function test */
+ err = WebRtcIsac_ReadFrameLen(ISAC_main_inst,
+ (WebRtc_Word16*)streamdata, &FL);
+ if(err < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+ printf(" Error: in getFrameLen %d.", errtype);
+ cout << flush;
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+ }
+
+ // iSAC decoding
+
+ if(lostFrame)
+ {
+ declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, streamdata,
+ stream_len, decoded, speechType);
+
+ if(doTransCoding)
+ {
+ declenTC = WebRtcIsac_DecodeRcu(decoderTransCoding,
+ streamDataTransCoding, streamLenTransCoding,
+ decodedTC, speechType);
+ }
+ }
+ else
+ {
+ declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata,
+ stream_len, decoded, speechType);
+
+ if(doTransCoding)
+ {
+ declenTC = WebRtcIsac_Decode(decoderTransCoding,
+ streamDataTransCoding, streamLenTransCoding,
+ decodedTC, speechType);
+ }
+ }
+
+ if(declen < 0)
+ {
+ /* exit if returned with error */
+ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+ printf(" Error: in decoder %d.", errtype);
+ cout << flush;
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+ }
+
+ if(declenTC < 0)
+ {
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+ printf(" Error: in decoding the transcoded stream");
+ cout << flush;
+ if(testNum == 0)
+ {
+ printf("\n\n");
+ }
+
+ }
+ }
+ /* Write decoded speech frame to file */
+ if((declen > 0) && (numFileLoop == 0))
+ {
+ fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
+ }
+
+ if((declenTC > 0) && (numFileLoop == 0))
+ {
+ fwrite(decodedTC, sizeof(WebRtc_Word16), declen, transCodingFile);
+ }
+
+
+ fprintf(stderr, "\rframe = %5d ", framecnt);
+ fflush(stderr);
+ framecnt++;
+
+ /* Error test number 10, garbage data */
+ //if(testNum == 10)
+ //{
+ // /* Test to run decoder with garbage data */
+ // if( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL )
+ // {
+ // fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+ // }
+ // else
+ // {
+ // fprintf(seedfile, "ok\n\n");
+ // fclose(seedfile);
+ // }
+ //}
+ /* Error test number 10, garbage data */
+ //if(testNum == 10)
+ //{
+ // /* Test to run decoder with garbage data */
+ // for ( i = 0; i < stream_len; i++)
+ // {
+ // streamdata[i] = (short) (streamdata[i] + (short) rand());
+ // }
+ //}
+
+
+ totalsmpls += declen;
+ totalbits += 8 * stream_len;
+#ifdef _DEBUG
+ kbps = ((double) sampFreqKHz * 1000.) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
+ fy = fopen("bit_rate.dat", "a");
+ fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
+ fclose(fy);
+
+#endif /* _DEBUG */
+
+ }
+ printf("\n");
+ printf("total bits = %d bits\n", totalbits);
+ printf("measured average bitrate = %0.3f kbits/s\n",
+ (double)totalbits *(sampFreqKHz) / totalsmpls);
+ if(doTransCoding)
+ {
+ printf("Transcoding average bit-rate = %0.3f kbps\n",
+ (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
+ fclose(transCodingFile);
+ }
+ printf("\n");
+
+ /* Runtime statistics */
+ runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
+ length_file = length_file /(sampFreqKHz * 1000.);
+
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+
+ if(maxStreamLen30 != 0)
+ {
+ printf("Maximum payload size 30ms Frames %d bytes (%0.3f kbps)\n",
+ maxStreamLen30,
+ maxStreamLen30 * 8 / 30.);
+ }
+ if(maxStreamLen60 != 0)
+ {
+ printf("Maximum payload size 60ms Frames %d bytes (%0.3f kbps)\n",
+ maxStreamLen60,
+ maxStreamLen60 * 8 / 60.);
+ }
+ //fprintf(stderr, "\n");
+
+ fprintf(stderr, " %.1f s", length_file);
+ fprintf(stderr, " %0.1f kbps", (double)totalbits *(sampFreqKHz) / totalsmpls);
+ if(maxStreamLen30 != 0)
+ {
+ fprintf(stderr, " plmax-30ms %d bytes (%0.0f kbps)",
+ maxStreamLen30,
+ maxStreamLen30 * 8 / 30.);
+ }
+ if(maxStreamLen60 != 0)
+ {
+ fprintf(stderr, " plmax-60ms %d bytes (%0.0f kbps)",
+ maxStreamLen60,
+ maxStreamLen60 * 8 / 60.);
+ }
+ if(doTransCoding)
+ {
+ fprintf(stderr, " transcoding rate %.0f kbps",
+ (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
+ }
+
+ fclose(inp);
+ fclose(outp);
+ WebRtcIsac_Free(ISAC_main_inst);
+
+
+ exit(0);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc
new file mode 100644
index 0000000..e0eed12
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -0,0 +1,443 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// SwitchingSampRate.cpp : Defines the entry point for the console
+// application.
+//
+
+#include <iostream>
+#include "isac.h"
+#include "utility.h"
+#include "signal_processing_library.h"
+
+#define MAX_FILE_NAME 500
+#define MAX_NUM_CLIENTS 2
+
+
+#define NUM_CLIENTS 2
+
+using namespace std;
+
+int main(int argc, char* argv[])
+{
+ char fileNameWB[MAX_FILE_NAME];
+ char fileNameSWB[MAX_FILE_NAME];
+
+ char outFileName[MAX_NUM_CLIENTS][MAX_FILE_NAME];
+
+ FILE* inFile[MAX_NUM_CLIENTS];
+ FILE* outFile[MAX_NUM_CLIENTS];
+
+ ISACStruct* codecInstance[MAX_NUM_CLIENTS];
+ WebRtc_Word32 resamplerState[MAX_NUM_CLIENTS][8];
+
+ enum IsacSamplingRate encoderSampRate[MAX_NUM_CLIENTS];
+
+ int minBn = 16000;
+ int maxBn = 56000;
+
+ int bnWB = 32000;
+ int bnSWB = 56000;
+
+ strcpy(outFileName[0], "switchSampRate_out1.pcm");
+ strcpy(outFileName[1], "switchSampRate_out2.pcm");
+
+ short clientCntr;
+
+ unsigned int lenEncodedInBytes[MAX_NUM_CLIENTS];
+ unsigned int lenAudioIn10ms[MAX_NUM_CLIENTS];
+ unsigned int lenEncodedInBytesTmp[MAX_NUM_CLIENTS];
+ unsigned int lenAudioIn10msTmp[MAX_NUM_CLIENTS];
+ BottleNeckModel* packetData[MAX_NUM_CLIENTS];
+
+ char versionNumber[100];
+ short samplesIn10ms[MAX_NUM_CLIENTS];
+ int bottleneck[MAX_NUM_CLIENTS];
+
+ printf("\n\n");
+ printf("____________________________________________\n\n");
+ WebRtcIsac_version(versionNumber);
+ printf(" iSAC-swb version %s\n", versionNumber);
+ printf("____________________________________________\n");
+
+
+ fileNameWB[0] = '\0';
+ fileNameSWB[0] = '\0';
+
+ char myFlag[20];
+ strcpy(myFlag, "-wb");
+ // READ THE WIDEBAND AND SUPER-WIDEBAND FILE NAMES
+ if(readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0)
+ {
+ printf("No wideband file is specified");
+ }
+
+ strcpy(myFlag, "-swb");
+ if(readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0)
+ {
+ printf("No super-wideband file is specified");
+ }
+
+ // THE FIRST CLIENT STARTS IN WIDEBAND
+ encoderSampRate[0] = kIsacWideband;
+ OPEN_FILE_RB(inFile[0], fileNameWB);
+
+ // THE SECOND CLIENT STARTS IN SUPER-WIDEBAND
+ encoderSampRate[1] = kIsacSuperWideband;
+ OPEN_FILE_RB(inFile[1], fileNameSWB);
+
+ strcpy(myFlag, "-I");
+ short codingMode = readSwitch(argc, argv, myFlag);
+
+ for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
+ {
+ codecInstance[clientCntr] = NULL;
+
+ printf("\n");
+ printf("Client %d\n", clientCntr + 1);
+ printf("---------\n");
+ printf("Starting %s",
+ (encoderSampRate[clientCntr] == kIsacWideband)
+ ? "wideband":"super-wideband");
+
+ // Open output File Name
+ OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]);
+ printf("Output File...................... %s\n", outFileName[clientCntr]);
+
+ samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
+
+ if(codingMode == 1)
+ {
+ bottleneck[clientCntr] = (clientCntr)? bnSWB:bnWB;
+ }
+ else
+ {
+ bottleneck[clientCntr] = (clientCntr)? minBn:maxBn;
+ }
+
+ printf("Bottleneck....................... %0.3f kbits/sec \n",
+ bottleneck[clientCntr] / 1000.0);
+
+ // coding-mode
+ printf("Encoding Mode.................... %s\n",
+ (codingMode == 1)? "Channel-Independent (Instantaneous)":"Adaptive");
+
+ lenEncodedInBytes[clientCntr] = 0;
+ lenAudioIn10ms[clientCntr] = 0;
+ lenEncodedInBytesTmp[clientCntr] = 0;
+ lenAudioIn10msTmp[clientCntr] = 0;
+
+ packetData[clientCntr] = (BottleNeckModel*)new(BottleNeckModel);
+ if(packetData[clientCntr] == NULL)
+ {
+ printf("Could not allocate memory for packetData \n");
+ return -1;
+ }
+ memset(packetData[clientCntr], 0, sizeof(BottleNeckModel));
+ memset(resamplerState[clientCntr], 0, sizeof(WebRtc_Word32) * 8);
+ }
+
+ for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
+ {
+ // Create
+ if(WebRtcIsac_Create(&codecInstance[clientCntr]))
+ {
+ printf("Could not creat client %d\n", clientCntr + 1);
+ return -1;
+ }
+
+ WebRtcIsac_SetEncSampRate(codecInstance[clientCntr], encoderSampRate[clientCntr]);
+
+ WebRtcIsac_SetDecSampRate(codecInstance[clientCntr],
+ encoderSampRate[clientCntr + (1 - ((clientCntr & 1)<<1))]);
+
+ // Initialize Encoder
+ if(WebRtcIsac_EncoderInit(codecInstance[clientCntr],
+ codingMode) < 0)
+ {
+ printf("Could not initialize client, %d\n", clientCntr + 1);
+ return -1;
+ }
+
+ // Initialize Decoder
+ if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0)
+ {
+ printf("Could not initialize decoder of client %d\n",
+ clientCntr + 1);
+ return -1;
+ }
+
+ // setup Rate if in Instantaneous mode
+ if(codingMode != 0)
+ {
+ // ONLY Clients who are not in Adaptive mode
+ if(WebRtcIsac_Control(codecInstance[clientCntr],
+ bottleneck[clientCntr], 30) < 0)
+ {
+ printf("Could not setup bottleneck and frame-size for client %d\n",
+ clientCntr + 1);
+ return -1;
+ }
+ }
+ }
+
+
+ short streamLen;
+ short numSamplesRead;
+ short lenDecodedAudio;
+ short senderIdx;
+ short receiverIdx;
+
+ printf("\n");
+ short num10ms[MAX_NUM_CLIENTS];
+ memset(num10ms, 0, sizeof(short)*MAX_NUM_CLIENTS);
+ FILE* arrivalTimeFile1 = fopen("arrivalTime1.dat", "wb");
+ FILE* arrivalTimeFile2 = fopen("arrivalTime2.dat", "wb");
+ short numPrint[MAX_NUM_CLIENTS];
+ memset(numPrint, 0, sizeof(short) * MAX_NUM_CLIENTS);
+
+ // Audio Buffers
+ short silence10ms[10 * 32];
+ memset(silence10ms, 0, 320 * sizeof(short));
+ short audioBuff10ms[10 * 32];
+ short audioBuff60ms[60 * 32];
+ short resampledAudio60ms[60 * 32];
+
+ unsigned short bitStream[600+600];
+ short speechType[1];
+
+ short numSampFreqChanged = 0;
+ while(numSampFreqChanged < 10)
+ {
+ for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
+ {
+ // Encoding/decoding for this pair of clients, if there is
+ // audio for any of them
+ //if(audioLeft[clientCntr] || audioLeft[clientCntr + 1])
+ //{
+ //for(pairCntr = 0; pairCntr < 2; pairCntr++)
+ //{
+ senderIdx = clientCntr; // + pairCntr;
+ receiverIdx = 1 - clientCntr;// + (1 - pairCntr);
+
+ //if(num10ms[senderIdx] > 6)
+ //{
+ // printf("Too many frames read for client %d",
+ // senderIdx + 1);
+ // return -1;
+ //}
+
+ numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
+ samplesIn10ms[senderIdx], inFile[senderIdx]);
+ if(numSamplesRead != samplesIn10ms[senderIdx])
+ {
+ // file finished switch encoder sampling frequency.
+ printf("Changing Encoder Sampling frequency in client %d to ", senderIdx+1);
+ fclose(inFile[senderIdx]);
+ numSampFreqChanged++;
+ if(encoderSampRate[senderIdx] == kIsacWideband)
+ {
+ printf("super-wideband.\n");
+ OPEN_FILE_RB(inFile[senderIdx], fileNameSWB);
+ encoderSampRate[senderIdx] = kIsacSuperWideband;
+ }
+ else
+ {
+ printf("wideband.\n");
+ OPEN_FILE_RB(inFile[senderIdx], fileNameWB);
+ encoderSampRate[senderIdx] = kIsacWideband;
+ }
+ WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], encoderSampRate[senderIdx]);
+ WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], encoderSampRate[senderIdx]);
+
+ samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
+
+ numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
+ samplesIn10ms[senderIdx], inFile[senderIdx]);
+ if(numSamplesRead != samplesIn10ms[senderIdx])
+ {
+ printf(" File %s for client %d has not enough audio\n",
+ (encoderSampRate[senderIdx]==kIsacWideband)? "wideband":"super-wideband",
+ senderIdx + 1);
+ return -1;
+ }
+ }
+ num10ms[senderIdx]++;
+
+ // sanity check
+ //if(num10ms[senderIdx] > 6)
+ //{
+ // printf("Client %d has got more than 60 ms audio and encoded no packet.\n",
+ // senderIdx);
+ // return -1;
+ //}
+
+ // Encode
+
+
+ streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
+ audioBuff10ms, (short*)bitStream);
+ WebRtc_Word16 ggg;
+ if (streamLen > 0) {
+ if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],
+ (short *) bitStream, &ggg))<0)
+ printf("ERROR\n");
+ }
+
+ // Sanity check
+ if(streamLen < 0)
+ {
+ printf(" Encoder error in client %d \n", senderIdx + 1);
+ return -1;
+ }
+
+
+ if(streamLen > 0)
+ {
+ // Packet generated; model sending through a channel, do bandwidth
+ // estimation at the receiver and decode.
+ lenEncodedInBytes[senderIdx] += streamLen;
+ lenAudioIn10ms[senderIdx] += (unsigned int)num10ms[senderIdx];
+ lenEncodedInBytesTmp[senderIdx] += streamLen;
+ lenAudioIn10msTmp[senderIdx] += (unsigned int)num10ms[senderIdx];
+
+ // Print after ~5 sec.
+ if(lenAudioIn10msTmp[senderIdx] >= 100)
+ {
+ numPrint[senderIdx]++;
+ printf(" %d, %6.3f => %6.3f ", senderIdx+1,
+ bottleneck[senderIdx] / 1000.0,
+ lenEncodedInBytesTmp[senderIdx] * 0.8 /
+ lenAudioIn10msTmp[senderIdx]);
+
+ if(codingMode == 0)
+ {
+ WebRtc_Word32 bn;
+ WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn);
+ printf("[%d] ", bn);
+ }
+ //WebRtc_Word16 rateIndexLB;
+ //WebRtc_Word16 rateIndexUB;
+ //WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx],
+ // &rateIndexLB, &rateIndexUB);
+ //printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB);
+
+ cout << flush;
+ lenEncodedInBytesTmp[senderIdx] = 0;
+ lenAudioIn10msTmp[senderIdx] = 0;
+ //if(senderIdx == (NUM_CLIENTS - 1))
+ //{
+ printf(" %0.1f \n", lenAudioIn10ms[senderIdx] * 10. /1000);
+ //}
+
+ // After ~20 sec change the bottleneck.
+ // if((numPrint[senderIdx] == 4) && (codingMode == 0))
+ // {
+ // numPrint[senderIdx] = 0;
+ // if(codingMode == 0)
+ // {
+ // int newBottleneck = bottleneck[senderIdx] +
+ // (bottleneckChange[senderIdx] * 1000);
+
+ // if(bottleneckChange[senderIdx] > 0)
+ // {
+ // if(newBottleneck >maxBn)
+ // {
+ // bottleneckChange[senderIdx] = -1;
+ // newBottleneck = bottleneck[senderIdx] +
+ // (bottleneckChange[senderIdx] * 1000);
+ // if(newBottleneck > minBn)
+ // {
+ // bottleneck[senderIdx] = newBottleneck;
+ // }
+ // }
+ // else
+ // {
+ // bottleneck[senderIdx] = newBottleneck;
+ // }
+ // }
+ // else
+ // {
+ // if(newBottleneck < minBn)
+ // {
+ // bottleneckChange[senderIdx] = 1;
+ // newBottleneck = bottleneck[senderIdx] +
+ // (bottleneckChange[senderIdx] * 1000);
+ // if(newBottleneck < maxBn)
+ // {
+ // bottleneck[senderIdx] = newBottleneck;
+ // }
+ // }
+ // else
+ // {
+ // bottleneck[senderIdx] = newBottleneck;
+ // }
+ // }
+ // }
+ // }
+ }
+
+ // model a channel of given bottleneck, to get the receive timestamp
+ get_arrival_time(num10ms[senderIdx] * samplesIn10ms[senderIdx],
+ streamLen, bottleneck[senderIdx], packetData[senderIdx],
+ encoderSampRate[senderIdx]*1000, encoderSampRate[senderIdx]*1000);
+
+ // Write the arrival time.
+ if(senderIdx == 0)
+ {
+ fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
+ arrivalTimeFile1);
+ }
+ else
+ {
+ fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
+ arrivalTimeFile2);
+ }
+
+ // BWE
+ if(WebRtcIsac_UpdateBwEstimate(codecInstance[receiverIdx],
+ bitStream, streamLen, packetData[senderIdx]->rtp_number,
+ packetData[senderIdx]->sample_count,
+ packetData[senderIdx]->arrival_time) < 0)
+ {
+ printf(" BWE Error at client %d \n", receiverIdx + 1);
+ return -1;
+ }
+ /**/
+ // Decode
+ lenDecodedAudio = WebRtcIsac_Decode(
+ codecInstance[receiverIdx], bitStream, streamLen,
+ audioBuff60ms, speechType);
+ if(lenDecodedAudio < 0)
+ {
+ printf(" Decoder error in client %d \n", receiverIdx + 1);
+ return -1;
+ }
+
+
+ if(encoderSampRate[senderIdx] == kIsacWideband)
+ {
+ WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms,
+ resamplerState[receiverIdx]);
+ fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
+ outFile[receiverIdx]);
+ }
+ else
+ {
+ fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
+ outFile[receiverIdx]);
+ }
+ num10ms[senderIdx] = 0;
+ }
+ //}
+ //}
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h
new file mode 100644
index 0000000..d708ad1
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
+
+#include <stdio.h>
+#include <string.h>
+#include "utility.h"
+
+typedef struct
+{
+ FILE* res0to4FilePtr;
+ FILE* res4to8FilePtr;
+ FILE* res8to12FilePtr;
+ FILE* res8to16FilePtr;
+
+ FILE* res0to4DecFilePtr;
+ FILE* res4to8DecFilePtr;
+ FILE* res8to12DecFilePtr;
+ FILE* res8to16DecFilePtr;
+
+ FILE* in0to4FilePtr;
+ FILE* in4to8FilePtr;
+ FILE* in8to12FilePtr;
+ FILE* in8to16FilePtr;
+
+ FILE* out0to4FilePtr;
+ FILE* out4to8FilePtr;
+ FILE* out8to12FilePtr;
+ FILE* out8to16FilePtr;
+
+ FILE* fftFilePtr;
+ FILE* fftDecFilePtr;
+
+ FILE* arrivalTime;
+
+ float lastArrivalTime;
+
+ int prevPacketLost;
+ int currPacketLost;
+ int nextPacketLost;
+
+ //double residualSignal4kHZ[240];
+ int packetLossPercent;
+
+ int maxPayloadLB;
+ int maxPayloadUB;
+ int lbBytes;
+ int ubBytes;
+
+
+}debugStruct;
+
+
+#define PRINT_ENTROPY_INFO(obj) \
+ do \
+ { \
+ printf("%10u, %u; ", \
+ obj->bitstr_obj.streamval, obj->bitstr_obj.stream_index); \
+ } while(0)
+
+int setupDebugStruct(debugStruct* str);
+
+#endif
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c
new file mode 100644
index 0000000..d92ab89
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c
@@ -0,0 +1,634 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* kenny.c - Main function for the iSAC coder */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+
+#ifdef WIN32
+#include "windows.h"
+#define CLOCKS_PER_SEC 1000
+#endif
+
+#include <ctype.h>
+#include <math.h>
+
+/* include API */
+#include "isac.h"
+#include "utility.h"
+//#include "commonDefs.h"
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES_SWB 1920
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_SWB_10ms 320
+#define FRAMESAMPLES_WB_10ms 160
+
+/* sampling frequency (Hz) */
+#define FS_SWB 32000
+#define FS_WB 16000
+
+//#define CHANGE_OUTPUT_NAME
+
+#ifdef HAVE_DEBUG_INFO
+ #include "debugUtility.h"
+ debugStruct debugInfo;
+#endif
+
+unsigned long framecnt = 0;
+
+int main(int argc, char* argv[])
+{
+ //--- File IO ----
+ FILE* inp;
+ FILE* outp;
+ char inname[500];
+ char outname[500];
+
+ /* Runtime statistics */
+ double rate;
+ double rateRCU;
+ unsigned long totalbits = 0;
+ unsigned long totalBitsRCU = 0;
+ unsigned long totalsmpls =0;
+
+ WebRtc_Word32 bottleneck = 39;
+ WebRtc_Word16 frameSize = 30; /* ms */
+ WebRtc_Word16 codingMode = 1;
+ WebRtc_Word16 shortdata[FRAMESAMPLES_SWB_10ms];
+ WebRtc_Word16 decoded[MAX_FRAMESAMPLES_SWB];
+ //WebRtc_UWord16 streamdata[1000];
+ WebRtc_Word16 speechType[1];
+ WebRtc_Word16 payloadLimit;
+ WebRtc_Word32 rateLimit;
+ ISACStruct* ISAC_main_inst;
+
+ WebRtc_Word16 stream_len = 0;
+ WebRtc_Word16 declen;
+ WebRtc_Word16 err;
+ WebRtc_Word16 cur_framesmpls;
+ int endfile;
+#ifdef WIN32
+ double length_file;
+ double runtime;
+ char outDrive[10];
+ char outPath[500];
+ char outPrefix[500];
+ char outSuffix[500];
+ char bitrateFileName[500];
+ FILE* bitrateFile;
+ double starttime;
+ double rateLB = 0;
+ double rateUB = 0;
+#endif
+ FILE* histFile;
+ FILE* averageFile;
+ int sampFreqKHz;
+ int samplesIn10Ms;
+ WebRtc_Word16 maxStreamLen = 0;
+ char histFileName[500];
+ char averageFileName[500];
+ unsigned int hist[600];
+ unsigned int tmpSumStreamLen = 0;
+ unsigned int packetCntr = 0;
+ unsigned int lostPacketCntr = 0;
+ WebRtc_UWord16 payload[600];
+ WebRtc_UWord16 payloadRCU[600];
+ WebRtc_UWord16 packetLossPercent = 0;
+ WebRtc_Word16 rcuStreamLen = 0;
+ int onlyEncode;
+ int onlyDecode;
+
+
+ BottleNeckModel packetData;
+ packetData.arrival_time = 0;
+ packetData.sample_count = 0;
+ packetData.rtp_number = 0;
+ memset(hist, 0, sizeof(hist));
+
+ /* handling wrong input arguments in the command line */
+ if(argc < 5)
+ {
+ int size;
+ WebRtcIsac_AssignSize(&size);
+
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("Usage:\n\n");
+ printf("%s infile outfile -bn bottelneck [options] \n\n", argv[0]);
+ printf("with:\n");
+ printf("-I................... indicates encoding in instantaneous mode.\n");
+ printf("-bn bottleneck....... the value of the bottleneck in bit/sec, e.g. 39742,\n");
+ printf(" in instantaneous (channel-independent) mode.\n\n");
+ printf("infile............... Normal speech input file\n\n");
+ printf("outfile.............. Speech output file\n\n");
+ printf("OPTIONS\n");
+ printf("-------\n");
+ printf("-fs sampFreq......... sampling frequency of codec 16 or 32 (default) kHz.\n");
+ printf("-plim payloadLim..... payload limit in bytes,\n");
+ printf(" default is the maximum possible.\n");
+ printf("-rlim rateLim........ rate limit in bits/sec, \n");
+ printf(" default is the maimum possible.\n");
+ printf("-h file.............. record histogram and *append* to 'file'.\n");
+ printf("-ave file............ record average rate of 3 sec intervales and *append* to 'file'.\n");
+ printf("-ploss............... packet-loss percentage.\n");
+ printf("-enc................. do only encoding and store the bit-stream\n");
+ printf("-dec................. the input file is a bit-stream, decode it.\n");
+
+ printf("\n");
+ printf("Example usage:\n\n");
+ printf("%s speechIn.pcm speechOut.pcm -B 40000 -fs 32 \n\n", argv[0]);
+
+ printf("structure size %d bytes\n", size);
+
+ exit(0);
+ }
+
+
+
+ /* Get Bottleneck value */
+ bottleneck = readParamInt(argc, argv, "-bn", 50000);
+ fprintf(stderr,"\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+
+ /* Get Input and Output files */
+ sscanf(argv[1], "%s", inname);
+ sscanf(argv[2], "%s", outname);
+ codingMode = readSwitch(argc, argv, "-I");
+ sampFreqKHz = (WebRtc_Word16)readParamInt(argc, argv, "-fs", 32);
+ if(readParamString(argc, argv, "-h", histFileName, 500) > 0)
+ {
+ histFile = fopen(histFileName, "a");
+ if(histFile == NULL)
+ {
+ printf("cannot open hist file %s", histFileName);
+ exit(0);
+ }
+ }
+ else
+ {
+ // NO recording of hitstogram
+ histFile = NULL;
+ }
+
+
+ packetLossPercent = readParamInt(argc, argv, "-ploss", 0);
+
+ if(readParamString(argc, argv, "-ave", averageFileName, 500) > 0)
+ {
+ averageFile = fopen(averageFileName, "a");
+ if(averageFile == NULL)
+ {
+ printf("cannot open file to write rate %s", averageFileName);
+ exit(0);
+ }
+ }
+ else
+ {
+ averageFile = NULL;
+ }
+
+ onlyEncode = readSwitch(argc, argv, "-enc");
+ onlyDecode = readSwitch(argc, argv, "-dec");
+
+
+ switch(sampFreqKHz)
+ {
+ case 16:
+ {
+ samplesIn10Ms = 160;
+ break;
+ }
+ case 32:
+ {
+ samplesIn10Ms = 320;
+ break;
+ }
+ default:
+ printf("A sampling frequency of %d kHz is not supported,\
+valid values are 8 and 16.\n", sampFreqKHz);
+ exit(-1);
+ }
+ payloadLimit = (WebRtc_Word16)readParamInt(argc, argv, "-plim", 400);
+ rateLimit = readParamInt(argc, argv, "-rlim", 106800);
+
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" iSAC: Cannot read file %s.\n", inname);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" iSAC: Cannot write file %s.\n", outname);
+ exit(1);
+ }
+
+#ifdef WIN32
+ _splitpath(outname, outDrive, outPath, outPrefix, outSuffix);
+ _makepath(bitrateFileName, outDrive, outPath, "bitrate", ".txt");
+
+ bitrateFile = fopen(bitrateFileName, "a");
+ fprintf(bitrateFile, "% %%s \n", inname);
+#endif
+
+ printf("\n");
+ printf("Input.................... %s\n", inname);
+ printf("Output................... %s\n", outname);
+ printf("Encoding Mode............ %s\n",
+ (codingMode == 1)? "Channel-Independent":"Channel-Adaptive");
+ printf("Bottleneck............... %d bits/sec\n", bottleneck);
+ printf("Packet-loss Percentage... %d\n", packetLossPercent);
+ printf("\n");
+
+#ifdef WIN32
+ starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+#endif
+
+ /* Initialize the ISAC and BN structs */
+ err = WebRtcIsac_Create(&ISAC_main_inst);
+
+ WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
+ WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
+ /* Error check */
+ if (err < 0) {
+ fprintf(stderr,"\n\n Error in create.\n\n");
+ exit(EXIT_FAILURE);
+ }
+
+ framecnt = 0;
+ endfile = 0;
+
+ /* Initialize encoder and decoder */
+ if(WebRtcIsac_EncoderInit(ISAC_main_inst, codingMode) < 0)
+ {
+ printf("cannot initialize encoder\n");
+ return -1;
+ }
+ if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
+ {
+ printf("cannot initialize decoder\n");
+ return -1;
+ }
+
+ //{
+ // WebRtc_Word32 b1, b2;
+ // FILE* fileID = fopen("GetBNTest.txt", "w");
+ // b2 = 32100;
+ // while(b2 <= 52000)
+ // {
+ // WebRtcIsac_Control(ISAC_main_inst, b2, frameSize);
+ // WebRtcIsac_GetUplinkBw(ISAC_main_inst, &b1);
+ // fprintf(fileID, "%5d %5d\n", b2, b1);
+ // b2 += 10;
+ // }
+ //}
+
+ if(codingMode == 1)
+ {
+ if(WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0)
+ {
+ printf("cannot set bottleneck\n");
+ return -1;
+ }
+ }
+ else
+ {
+ if(WebRtcIsac_ControlBwe(ISAC_main_inst, 15000, 30, 1) < 0)
+ {
+ printf("cannot configure BWE\n");
+ return -1;
+ }
+ }
+
+ if(WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadLimit) < 0)
+ {
+ printf("cannot set maximum payload size %d.\n", payloadLimit);
+ return -1;
+ }
+
+ if (rateLimit < 106800) {
+ if(WebRtcIsac_SetMaxRate(ISAC_main_inst, rateLimit) < 0)
+ {
+ printf("cannot set the maximum rate %d.\n", rateLimit);
+ return -1;
+ }
+ }
+
+ //=====================================
+//#ifdef HAVE_DEBUG_INFO
+// if(setupDebugStruct(&debugInfo) < 0)
+// {
+// exit(1);
+// }
+//#endif
+
+ while (endfile == 0)
+ {
+ fprintf(stderr," \rframe = %7li", framecnt);
+
+ //============== Readind from the file and encoding =================
+ cur_framesmpls = 0;
+ stream_len = 0;
+
+
+ if(onlyDecode)
+ {
+ WebRtc_UWord8 auxUW8;
+ size_t auxSizet;
+ if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
+ {
+ break;
+ }
+ stream_len = ((WebRtc_UWord8)auxUW8) << 8;
+ if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
+ {
+ break;
+ }
+ stream_len |= (WebRtc_UWord16)auxUW8;
+ auxSizet = (size_t)stream_len;
+ if(fread(payload, 1, auxSizet, inp) < auxSizet)
+ {
+ printf("last payload is corrupted\n");
+ break;
+ }
+ }
+ else
+ {
+ while(stream_len == 0)
+ {
+ // Read 10 ms speech block
+ endfile = readframe(shortdata, inp, samplesIn10Ms);
+ if(endfile)
+ {
+ break;
+ }
+ cur_framesmpls += samplesIn10Ms;
+
+ //-------- iSAC encoding ---------
+ stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata,
+ (WebRtc_Word16*)payload);
+
+ if(stream_len < 0)
+ {
+ // exit if returned with error
+ //errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ fprintf(stderr,"\nError in encoder\n");
+ getchar();
+ exit(EXIT_FAILURE);
+ }
+
+
+ }
+ //===================================================================
+ if(endfile)
+ {
+ break;
+ }
+
+ rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, (WebRtc_Word16*)payloadRCU);
+
+ get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
+ sampFreqKHz * 1000, sampFreqKHz * 1000);
+ if(WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
+ payload, stream_len, packetData.rtp_number,
+ packetData.sample_count,
+ packetData.arrival_time) < 0)
+ {
+ printf(" BWE Error at client\n");
+ return -1;
+ }
+ }
+
+ if(endfile)
+ {
+ break;
+ }
+
+ maxStreamLen = (stream_len > maxStreamLen)? stream_len:maxStreamLen;
+ packetCntr++;
+
+ hist[stream_len]++;
+ if(averageFile != NULL)
+ {
+ tmpSumStreamLen += stream_len;
+ if(packetCntr == 100)
+ {
+ // kbps
+ fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
+ packetCntr = 0;
+ tmpSumStreamLen = 0;
+ }
+ }
+
+ if(onlyEncode)
+ {
+ WebRtc_UWord8 auxUW8;
+ auxUW8 = (WebRtc_UWord8)(((stream_len & 0x7F00) >> 8) & 0xFF);
+ fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
+
+ auxUW8 = (WebRtc_UWord8)(stream_len & 0xFF);
+ fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
+ fwrite(payload, 1, stream_len, outp);
+ }
+ else
+ {
+
+ //======================= iSAC decoding ===========================
+
+ if((rand() % 100) < packetLossPercent)
+ {
+ declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU,
+ rcuStreamLen, decoded, speechType);
+ lostPacketCntr++;
+ }
+ else
+ {
+ declen = WebRtcIsac_Decode(ISAC_main_inst, payload,
+ stream_len, decoded, speechType);
+ }
+ if(declen <= 0)
+ {
+ //errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+ fprintf(stderr,"\nError in decoder.\n");
+ getchar();
+ exit(1);
+ }
+
+ // Write decoded speech frame to file
+ fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
+ cur_framesmpls = declen;
+ }
+ // Update Statistics
+ framecnt++;
+ totalsmpls += cur_framesmpls;
+ if(stream_len > 0)
+ {
+ totalbits += 8 * stream_len;
+ }
+ if(rcuStreamLen > 0)
+ {
+ totalBitsRCU += 8 * rcuStreamLen;
+ }
+ }
+
+ rate = ((double)totalbits * (sampFreqKHz)) / (double)totalsmpls;
+ rateRCU = ((double)totalBitsRCU * (sampFreqKHz)) / (double)totalsmpls;
+
+ printf("\n\n");
+ printf("Sampling Rate......................... %d kHz\n", sampFreqKHz);
+ printf("Payload Limit......................... %d bytes \n", payloadLimit);
+ printf("Rate Limit............................ %d bits/sec \n", rateLimit);
+
+#ifdef WIN32
+#ifdef HAVE_DEBUG_INFO
+ rateLB = ((double)debugInfo.lbBytes * 8. *
+ (sampFreqKHz)) / (double)totalsmpls;
+ rateUB = ((double)debugInfo.ubBytes * 8. *
+ (sampFreqKHz)) / (double)totalsmpls;
+#endif
+
+ fprintf(bitrateFile, "%d %10u %d %6.3f %6.3f %6.3f\n",
+ sampFreqKHz,
+ framecnt,
+ bottleneck,
+ rateLB,
+ rateUB,
+ rate);
+ fclose(bitrateFile);
+#endif // WIN32
+
+ printf("\n");
+ printf("Measured bit-rate..................... %0.3f kbps\n", rate);
+ printf("Measured RCU bit-ratre................ %0.3f kbps\n", rateRCU);
+ printf("Maximum bit-rate/payloadsize.......... %0.3f / %d\n",
+ maxStreamLen * 8 / 0.03, maxStreamLen);
+ printf("Measured packet-loss.................. %0.1f%% \n",
+ 100.0f * (float)lostPacketCntr / (float)packetCntr);
+
+//#ifdef HAVE_DEBUG_INFO
+// printf("Measured lower-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
+// rateLB, (double)(rateLB) * 100. /(double)(rate));
+// printf("Measured upper-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
+// rateUB, (double)(rateUB) * 100. /(double)(rate));
+//
+// printf("Maximum payload lower-band............ %d bytes (%0.3f kbps)\n",
+// debugInfo.maxPayloadLB, debugInfo.maxPayloadLB * 8.0 / 0.03);
+// printf("Maximum payload upper-band............ %d bytes (%0.3f kbps)\n",
+// debugInfo.maxPayloadUB, debugInfo.maxPayloadUB * 8.0 / 0.03);
+//#endif
+
+ printf("\n");
+
+ /* Runtime statistics */
+#ifdef WIN32
+ runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
+ length_file = ((double)framecnt*(double)declen/(sampFreqKHz*1000));
+ printf("Length of speech file................ %.1f s\n", length_file);
+ printf("Time to run iSAC..................... %.2f s (%.2f %% of realtime)\n\n",
+ runtime, (100*runtime/length_file));
+#endif
+ printf("\n\n_______________________________________________\n");
+
+ if(histFile != NULL)
+ {
+ int n;
+ for(n = 0; n < 600; n++)
+ {
+ fprintf(histFile, "%6d ", hist[n]);
+ }
+ fprintf(histFile, "\n");
+ fclose(histFile);
+ }
+ if(averageFile != NULL)
+ {
+ if(packetCntr > 0)
+ {
+ fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
+ }
+ fprintf(averageFile, "\n");
+ fclose(averageFile);
+ }
+
+ fclose(inp);
+ fclose(outp);
+
+ WebRtcIsac_Free(ISAC_main_inst);
+
+
+#ifdef CHANGE_OUTPUT_NAME
+ {
+ char* p;
+ char myExt[50];
+ char bitRateStr[10];
+ char newOutName[500];
+ strcpy(newOutName, outname);
+
+ myExt[0] = '\0';
+ p = strchr(newOutName, '.');
+ if(p != NULL)
+ {
+ strcpy(myExt, p);
+ *p = '_';
+ p++;
+ *p = '\0';
+ }
+ else
+ {
+ strcat(newOutName, "_");
+ }
+ sprintf(bitRateStr, "%0.0fkbps", rate);
+ strcat(newOutName, bitRateStr);
+ strcat(newOutName, myExt);
+ rename(outname, newOutName);
+ }
+#endif
+ exit(0);
+}
+
+
+#ifdef HAVE_DEBUG_INFO
+int setupDebugStruct(debugStruct* str)
+{
+ str->prevPacketLost = 0;
+ str->currPacketLost = 0;
+
+ OPEN_FILE_WB(str->res0to4FilePtr, "Res0to4.dat");
+ OPEN_FILE_WB(str->res4to8FilePtr, "Res4to8.dat");
+ OPEN_FILE_WB(str->res8to12FilePtr, "Res8to12.dat");
+ OPEN_FILE_WB(str->res8to16FilePtr, "Res8to16.dat");
+
+ OPEN_FILE_WB(str->res0to4DecFilePtr, "Res0to4Dec.dat");
+ OPEN_FILE_WB(str->res4to8DecFilePtr, "Res4to8Dec.dat");
+ OPEN_FILE_WB(str->res8to12DecFilePtr, "Res8to12Dec.dat");
+ OPEN_FILE_WB(str->res8to16DecFilePtr, "Res8to16Dec.dat");
+
+ OPEN_FILE_WB(str->in0to4FilePtr, "in0to4.dat");
+ OPEN_FILE_WB(str->in4to8FilePtr, "in4to8.dat");
+ OPEN_FILE_WB(str->in8to12FilePtr, "in8to12.dat");
+ OPEN_FILE_WB(str->in8to16FilePtr, "in8to16.dat");
+
+ OPEN_FILE_WB(str->out0to4FilePtr, "out0to4.dat");
+ OPEN_FILE_WB(str->out4to8FilePtr, "out4to8.dat");
+ OPEN_FILE_WB(str->out8to12FilePtr, "out8to12.dat");
+ OPEN_FILE_WB(str->out8to16FilePtr, "out8to16.dat");
+ OPEN_FILE_WB(str->fftFilePtr, "riFFT.dat");
+ OPEN_FILE_WB(str->fftDecFilePtr, "riFFTDec.dat");
+
+ OPEN_FILE_WB(str->arrivalTime, NULL/*"ArivalTime.dat"*/);
+ str->lastArrivalTime = 0;
+
+ str->maxPayloadLB = 0;
+ str->maxPayloadUB = 0;
+ str->lbBytes = 0;
+ str->ubBytes = 0;
+
+ return 0;
+};
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c
new file mode 100644
index 0000000..0a2256a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c
@@ -0,0 +1,178 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "utility.h"
+
+/* function for reading audio data from PCM file */
+int
+readframe(
+ short* data,
+ FILE* inp,
+ int length)
+{
+ short k, rlen, status = 0;
+ unsigned char* ptrUChar;
+ ptrUChar = (unsigned char*)data;
+
+ rlen = (short)fread(data, sizeof(short), length, inp);
+ if (rlen < length) {
+ for (k = rlen; k < length; k++)
+ data[k] = 0;
+ status = 1;
+ }
+
+ // Assuming that our PCM files are written in Intel machines
+ for(k = 0; k < length; k++)
+ {
+ data[k] = (short)ptrUChar[k<<1] | ((((short)ptrUChar[(k<<1) + 1]) << 8) & 0xFF00);
+ }
+
+ return status;
+}
+
+short
+readSwitch(
+ int argc,
+ char* argv[],
+ char* strID)
+{
+ short n;
+ for(n = 0; n < argc; n++)
+ {
+ if(strcmp(argv[n], strID) == 0)
+ {
+ return 1;
+ }
+ }
+ return 0;
+}
+
+double
+readParamDouble(
+ int argc,
+ char* argv[],
+ char* strID,
+ double defaultVal)
+{
+ double returnVal = defaultVal;
+ short n;
+ for(n = 0; n < argc; n++)
+ {
+ if(strcmp(argv[n], strID) == 0)
+ {
+ n++;
+ if(n < argc)
+ {
+ returnVal = atof(argv[n]);
+ }
+ break;
+ }
+ }
+ return returnVal;
+}
+
+int
+readParamInt(
+ int argc,
+ char* argv[],
+ char* strID,
+ int defaultVal)
+{
+ int returnVal = defaultVal;
+ short n;
+ for(n = 0; n < argc; n++)
+ {
+ if(strcmp(argv[n], strID) == 0)
+ {
+ n++;
+ if(n < argc)
+ {
+ returnVal = atoi(argv[n]);
+ }
+ break;
+ }
+ }
+ return returnVal;
+}
+
+int
+readParamString(
+ int argc,
+ char* argv[],
+ char* strID,
+ char* stringParam,
+ int maxSize)
+{
+ int paramLenght = 0;
+ short n;
+ for(n = 0; n < argc; n++)
+ {
+ if(strcmp(argv[n], strID) == 0)
+ {
+ n++;
+ if(n < argc)
+ {
+ strncpy(stringParam, argv[n], maxSize);
+ paramLenght = (int)strlen(argv[n]);
+ }
+ break;
+ }
+ }
+ return paramLenght;
+}
+
+void
+get_arrival_time(
+ int current_framesamples, /* samples */
+ int packet_size, /* bytes */
+ int bottleneck, /* excluding headers; bits/s */
+ BottleNeckModel* BN_data,
+ short senderSampFreqHz,
+ short receiverSampFreqHz)
+{
+ unsigned int travelTimeMs;
+ const int headerSizeByte = 35;
+
+ int headerRate;
+
+ BN_data->whenPackGeneratedMs += (current_framesamples / (senderSampFreqHz / 1000));
+
+ headerRate = headerSizeByte * 8 * senderSampFreqHz / current_framesamples; /* bits/s */
+
+ /* everything in samples */
+ BN_data->sample_count = BN_data->sample_count + current_framesamples;
+
+ //travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) /
+ // (bottleneck + HeaderRate)
+ travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
+ / (double)(bottleneck + headerRate) + 0.5);
+
+ if(BN_data->whenPrevPackLeftMs > BN_data->whenPackGeneratedMs)
+ {
+ BN_data->whenPrevPackLeftMs += travelTimeMs;
+ }
+ else
+ {
+ BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs +
+ travelTimeMs;
+ }
+
+ BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
+ (receiverSampFreqHz / 1000));
+
+// if (BN_data->arrival_time < BN_data->sample_count)
+// BN_data->arrival_time = BN_data->sample_count;
+
+ BN_data->rtp_number++;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h
new file mode 100644
index 0000000..f9fba94
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h
@@ -0,0 +1,144 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
+
+#include <stdlib.h>
+#include <stdio.h>
+
+#if defined(__cplusplus)
+extern "C" {
+#endif
+
+#define OPEN_FILE_WB(filePtr, fullPath) \
+ do \
+ { \
+ if(fullPath != NULL) \
+ { \
+ filePtr = fopen(fullPath, "wb"); \
+ if(filePtr == NULL) \
+ { \
+ printf("could not open %s to write to.", fullPath); \
+ return -1; \
+ } \
+ } \
+ else \
+ { \
+ filePtr = NULL; \
+ } \
+ }while(0)
+
+#define OPEN_FILE_AB(filePtr, fullPath) \
+ do \
+ { \
+ if(fullPath != NULL) \
+ { \
+ filePtr = fopen(fullPath, "ab"); \
+ if(filePtr == NULL) \
+ { \
+ printf("could not open %s to write to.", fullPath); \
+ return -1; \
+ } \
+ } \
+ else \
+ { \
+ filePtr = NULL; \
+ } \
+ }while(0)
+
+#define OPEN_FILE_RB(filePtr, fullPath) \
+ do \
+ { \
+ if(fullPath != NULL) \
+ { \
+ filePtr = fopen(fullPath, "rb"); \
+ if(filePtr == NULL) \
+ { \
+ printf("could not open %s to read from.", fullPath); \
+ return -1; \
+ } \
+ } \
+ else \
+ { \
+ filePtr = NULL; \
+ } \
+ }while(0)
+
+#define WRITE_FILE_D(bufferPtr, len, filePtr) \
+ do \
+ { \
+ if(filePtr != NULL) \
+ { \
+ double dummy[1000]; \
+ int cntr; \
+ for(cntr = 0; cntr < (len); cntr++) \
+ { \
+ dummy[cntr] = (double)bufferPtr[cntr]; \
+ } \
+ fwrite(dummy, sizeof(double), len, filePtr); \
+ fflush(filePtr); \
+ } \
+ } while(0)
+
+ typedef struct {
+ unsigned int whenPackGeneratedMs;
+ unsigned int whenPrevPackLeftMs;
+ unsigned int sendTimeMs ; /* milisecond */
+ unsigned int arrival_time; /* samples */
+ unsigned int sample_count; /* samples, also used as "send time stamp" */
+ unsigned int rtp_number;
+ } BottleNeckModel;
+
+ void get_arrival_time(
+ int current_framesamples, /* samples */
+ int packet_size, /* bytes */
+ int bottleneck, /* excluding headers; bits/s */
+ BottleNeckModel* BN_data,
+ short senderSampFreqHz,
+ short receiverSampFreqHz);
+
+ /* function for reading audio data from PCM file */
+ int readframe(
+ short* data,
+ FILE* inp,
+ int length);
+
+ short readSwitch(
+ int argc,
+ char* argv[],
+ char* strID);
+
+ double readParamDouble(
+ int argc,
+ char* argv[],
+ char* strID,
+ double defaultVal);
+
+ int readParamInt(
+ int argc,
+ char* argv[],
+ char* strID,
+ int defaultVal);
+
+ int readParamString(
+ int argc,
+ char* argv[],
+ char* strID,
+ char* stringParam,
+ int maxSize);
+
+#if defined(__cplusplus)
+}
+#endif
+
+
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk b/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk
new file mode 100644
index 0000000..cbadcab
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk
@@ -0,0 +1,165 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_ilbc
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ abs_quant.c \
+ abs_quant_loop.c \
+ augmented_cb_corr.c \
+ bw_expand.c \
+ cb_construct.c \
+ cb_mem_energy.c \
+ cb_mem_energy_augmentation.c \
+ cb_mem_energy_calc.c \
+ cb_search.c \
+ cb_search_core.c \
+ cb_update_best_index.c \
+ chebyshev.c \
+ comp_corr.c \
+ constants.c \
+ create_augmented_vec.c \
+ decode.c \
+ decode_residual.c \
+ decoder_interpolate_lsf.c \
+ do_plc.c \
+ encode.c \
+ energy_inverse.c \
+ enh_upsample.c \
+ enhancer.c \
+ enhancer_interface.c \
+ filtered_cb_vecs.c \
+ frame_classify.c \
+ gain_dequant.c \
+ gain_quant.c \
+ get_cd_vec.c \
+ get_lsp_poly.c \
+ get_sync_seq.c \
+ hp_input.c \
+ hp_output.c \
+ ilbc.c \
+ index_conv_dec.c \
+ index_conv_enc.c \
+ init_decode.c \
+ init_encode.c \
+ interpolate.c \
+ interpolate_samples.c \
+ lpc_encode.c \
+ lsf_check.c \
+ lsf_interpolate_to_poly_dec.c \
+ lsf_interpolate_to_poly_enc.c \
+ lsf_to_lsp.c \
+ lsf_to_poly.c \
+ lsp_to_lsf.c \
+ my_corr.c \
+ nearest_neighbor.c \
+ pack_bits.c \
+ poly_to_lsf.c \
+ poly_to_lsp.c \
+ refiner.c \
+ simple_interpolate_lsf.c \
+ simple_lpc_analysis.c \
+ simple_lsf_dequant.c \
+ simple_lsf_quant.c \
+ smooth.c \
+ smooth_out_data.c \
+ sort_sq.c \
+ split_vq.c \
+ state_construct.c \
+ state_search.c \
+ swap_bytes.c \
+ unpack_bits.c \
+ vq3.c \
+ vq4.c \
+ window32_w32.c \
+ xcorr_coef.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+
+# iLBC test app
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= test/iLBC_test.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../../..
+
+LOCAL_STATIC_LIBRARIES := \
+ libwebrtc_ilbc \
+ libwebrtc_spl
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils
+
+LOCAL_MODULE:= webrtc_ilbc_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include $(BUILD_NATIVE_TEST)
+endif
+
+# iLBC_testLib test app
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= test/iLBC_testLib.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../../..
+
+LOCAL_STATIC_LIBRARIES := \
+ libwebrtc_ilbc \
+ libwebrtc_spl
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils
+
+LOCAL_MODULE:= webrtc_ilbc_testLib
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include $(BUILD_NATIVE_TEST)
+endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c
new file mode 100644
index 0000000..4a70c8b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "abs_quant_loop.h"
+
+
+/*----------------------------------------------------------------*
+ * predictive noise shaping encoding of scaled start state
+ * (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+ and idxVec, uses state_first as
+ input) */
+ WebRtc_Word16 *in, /* (i) vector to encode */
+ WebRtc_Word16 *weightDenum /* (i) denominator of synthesis filter */
+ ) {
+ WebRtc_Word16 *syntOut;
+ WebRtc_Word16 quantLen[2];
+
+ /* Stack based */
+ WebRtc_Word16 syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+ WebRtc_Word16 in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+ WebRtc_Word16 *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+
+ /* Initialize the buffers */
+ WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
+ syntOut = &syntOutBuf[LPC_FILTERORDER];
+ /* Start with zero state */
+ WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
+
+ /* Perform the quantization loop in two sections of length quantLen[i],
+ where the perceptual weighting filter is updated at the subframe
+ border */
+
+ if (iLBC_encbits->state_first) {
+ quantLen[0]=SUBL;
+ quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
+ } else {
+ quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
+ quantLen[1]=SUBL;
+ }
+
+ /* Calculate the weighted residual, switch perceptual weighting
+ filter at the subframe border */
+ WebRtcSpl_FilterARFastQ12(
+ in, in_weighted,
+ weightDenum, LPC_FILTERORDER+1, quantLen[0]);
+ WebRtcSpl_FilterARFastQ12(
+ &in[quantLen[0]], &in_weighted[quantLen[0]],
+ &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
+
+ WebRtcIlbcfix_AbsQuantLoop(
+ syntOut,
+ in_weighted,
+ weightDenum,
+ quantLen,
+ iLBC_encbits->idxVec);
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h
new file mode 100644
index 0000000..fa59593
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * predictive noise shaping encoding of scaled start state
+ * (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+ and idxVec, uses state_first as
+ input) */
+ WebRtc_Word16 *in, /* (i) vector to encode */
+ WebRtc_Word16 *weightDenum /* (i) denominator of synthesis filter */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
new file mode 100644
index 0000000..4eebc3e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "sort_sq.h"
+
+void WebRtcIlbcfix_AbsQuantLoop(
+ WebRtc_Word16 *syntOutIN,
+ WebRtc_Word16 *in_weightedIN,
+ WebRtc_Word16 *weightDenumIN,
+ WebRtc_Word16 *quantLenIN,
+ WebRtc_Word16 *idxVecIN
+ )
+{
+ int n, k1, k2;
+ WebRtc_Word16 index;
+ WebRtc_Word32 toQW32;
+ WebRtc_Word32 toQ32;
+ WebRtc_Word16 tmp16a;
+ WebRtc_Word16 xq;
+
+ WebRtc_Word16 *syntOut = syntOutIN;
+ WebRtc_Word16 *in_weighted = in_weightedIN;
+ WebRtc_Word16 *weightDenum = weightDenumIN;
+ WebRtc_Word16 *quantLen = quantLenIN;
+ WebRtc_Word16 *idxVec = idxVecIN;
+
+ n=0;
+
+ for(k1=0;k1<2;k1++) {
+ for(k2=0;k2<quantLen[k1];k2++){
+
+ /* Filter to get the predicted value */
+ WebRtcSpl_FilterARFastQ12(
+ syntOut, syntOut,
+ weightDenum, LPC_FILTERORDER+1, 1);
+
+ /* the quantizer */
+ toQW32 = (WebRtc_Word32)(*in_weighted) - (WebRtc_Word32)(*syntOut);
+
+ toQ32 = (((WebRtc_Word32)toQW32)<<2);
+
+ if (toQ32 > 32767) {
+ toQ32 = (WebRtc_Word32) 32767;
+ } else if (toQ32 < -32768) {
+ toQ32 = (WebRtc_Word32) -32768;
+ }
+
+ /* Quantize the state */
+ if (toQW32<(-7577)) {
+ /* To prevent negative overflow */
+ index=0;
+ } else if (toQW32>8151) {
+ /* To prevent positive overflow */
+ index=7;
+ } else {
+ /* Find the best quantization index
+ (state_sq3Tbl is in Q13 and toQ is in Q11)
+ */
+ WebRtcIlbcfix_SortSq(&xq, &index,
+ (WebRtc_Word16)toQ32,
+ WebRtcIlbcfix_kStateSq3, 8);
+ }
+
+ /* Store selected index */
+ (*idxVec++) = index;
+
+ /* Compute decoded sample and update of the prediction filter */
+ tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
+
+ *syntOut = (WebRtc_Word16) (tmp16a + (WebRtc_Word32)(*in_weighted) - toQW32);
+
+ n++;
+ syntOut++; in_weighted++;
+ }
+ /* Update perceptual weighting filter at subframe border */
+ weightDenum += 11;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
new file mode 100644
index 0000000..f506e8e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * predictive noise shaping encoding of scaled start state
+ * (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuantLoop(
+ WebRtc_Word16 *syntOutIN,
+ WebRtc_Word16 *in_weightedIN,
+ WebRtc_Word16 *weightDenumIN,
+ WebRtc_Word16 *quantLenIN,
+ WebRtc_Word16 *idxVecIN
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
new file mode 100644
index 0000000..6011e92
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "augmented_cb_corr.h"
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+ WebRtc_Word16 *target, /* (i) Target vector */
+ WebRtc_Word16 *buffer, /* (i) Memory buffer */
+ WebRtc_Word16 *interpSamples, /* (i) buffer with
+ interpolated samples */
+ WebRtc_Word32 *crossDot, /* (o) The cross correlation between
+ the target and the Augmented
+ vector */
+ WebRtc_Word16 low, /* (i) Lag to start from (typically
+ 20) */
+ WebRtc_Word16 high, /* (i) Lag to end at (typically 39) */
+ WebRtc_Word16 scale) /* (i) Scale factor to use for
+ the crossDot */
+{
+ int lagcount;
+ WebRtc_Word16 ilow;
+ WebRtc_Word16 *targetPtr;
+ WebRtc_Word32 *crossDotPtr;
+ WebRtc_Word16 *iSPtr=interpSamples;
+
+ /* Calculate the correlation between the target and the
+ interpolated codebook. The correlation is calculated in
+ 3 sections with the interpolated part in the middle */
+ crossDotPtr=crossDot;
+ for (lagcount=low; lagcount<=high; lagcount++) {
+
+ ilow = (WebRtc_Word16) (lagcount-4);
+
+ /* Compute dot product for the first (lagcount-4) samples */
+ (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
+
+ /* Compute dot product on the interpolated samples */
+ (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
+ targetPtr = target + lagcount;
+ iSPtr += lagcount-ilow;
+
+ /* Compute dot product for the remaining samples */
+ (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
+ crossDotPtr++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
new file mode 100644
index 0000000..8e097fe
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Calculate correlation between target and Augmented codebooks
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+ WebRtc_Word16 *target, /* (i) Target vector */
+ WebRtc_Word16 *buffer, /* (i) Memory buffer */
+ WebRtc_Word16 *interpSamples, /* (i) buffer with
+ interpolated samples */
+ WebRtc_Word32 *crossDot, /* (o) The cross correlation between
+ the target and the Augmented
+ vector */
+ WebRtc_Word16 low, /* (i) Lag to start from (typically
+ 20) */
+ WebRtc_Word16 high, /* (i) Lag to end at (typically 39 */
+ WebRtc_Word16 scale); /* (i) Scale factor to use for
+ the crossDot */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c
new file mode 100644
index 0000000..a2287aa
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+/* The output is in the same domain as the input */
+void WebRtcIlbcfix_BwExpand(
+ WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
+ WebRtc_Word16 *in, /* (i) the lpc coefficients before bandwidth
+ expansion */
+ WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
+ WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
+ ) {
+ int i;
+
+ out[0] = in[0];
+ for (i = 1; i < length; i++) {
+ /* out[i] = coef[i] * in[i] with rounding.
+ in[] and out[] are in Q12 and coef[] is in Q15
+ */
+ out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(coef[i], in[i])+16384)>>15);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h
new file mode 100644
index 0000000..c9f3fab
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_BwExpand(
+ WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
+ WebRtc_Word16 *in, /* (i) the lpc coefficients before bandwidth
+ expansion */
+ WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
+ WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c
new file mode 100644
index 0000000..094a7e4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "gain_dequant.h"
+#include "get_cd_vec.h"
+
+/*----------------------------------------------------------------*
+ * Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbConstruct(
+ WebRtc_Word16 *decvector, /* (o) Decoded vector */
+ WebRtc_Word16 *index, /* (i) Codebook indices */
+ WebRtc_Word16 *gain_index, /* (i) Gain quantization indices */
+ WebRtc_Word16 *mem, /* (i) Buffer for codevector construction */
+ WebRtc_Word16 lMem, /* (i) Length of buffer */
+ WebRtc_Word16 veclen /* (i) Length of vector */
+ ){
+ int j;
+ WebRtc_Word16 gain[CB_NSTAGES];
+ /* Stack based */
+ WebRtc_Word16 cbvec0[SUBL];
+ WebRtc_Word16 cbvec1[SUBL];
+ WebRtc_Word16 cbvec2[SUBL];
+ WebRtc_Word32 a32;
+ WebRtc_Word16 *gainPtr;
+
+ /* gain de-quantization */
+
+ gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
+ gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
+ gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
+
+ /* codebook vector construction and construction of total vector */
+
+ /* Stack based */
+ WebRtcIlbcfix_GetCbVec(cbvec0, mem, index[0], lMem, veclen);
+ WebRtcIlbcfix_GetCbVec(cbvec1, mem, index[1], lMem, veclen);
+ WebRtcIlbcfix_GetCbVec(cbvec2, mem, index[2], lMem, veclen);
+
+ gainPtr = &gain[0];
+ for (j=0;j<veclen;j++) {
+ a32 = WEBRTC_SPL_MUL_16_16(*gainPtr++, cbvec0[j]);
+ a32 += WEBRTC_SPL_MUL_16_16(*gainPtr++, cbvec1[j]);
+ a32 += WEBRTC_SPL_MUL_16_16(*gainPtr, cbvec2[j]);
+ gainPtr -= 2;
+ decvector[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(a32 + 8192, 14);
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h
new file mode 100644
index 0000000..bec759f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbConstruct(
+ WebRtc_Word16 *decvector, /* (o) Decoded vector */
+ WebRtc_Word16 *index, /* (i) Codebook indices */
+ WebRtc_Word16 *gain_index, /* (i) Gain quantization indices */
+ WebRtc_Word16 *mem, /* (i) Buffer for codevector construction */
+ WebRtc_Word16 lMem, /* (i) Length of buffer */
+ WebRtc_Word16 veclen /* (i) Length of vector */
+ );
+
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
new file mode 100644
index 0000000..8613fa2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "cb_mem_energy_calc.h"
+
+/*----------------------------------------------------------------*
+ * Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
+ * the vectors in the codebook memory that will be used in the
+ * following search for the best match.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbMemEnergy(
+ WebRtc_Word16 range,
+ WebRtc_Word16 *CB, /* (i) The CB memory (1:st section) */
+ WebRtc_Word16 *filteredCB, /* (i) The filtered CB memory (2:nd section) */
+ WebRtc_Word16 lMem, /* (i) Length of the CB memory */
+ WebRtc_Word16 lTarget, /* (i) Length of the target vector */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
+ ) {
+ WebRtc_Word16 *ppi, *ppo, *pp;
+ WebRtc_Word32 energy, tmp32;
+
+ /* Compute the energy and store it in a vector. Also the
+ * corresponding shift values are stored. The energy values
+ * are reused in all three stages. */
+
+ /* Calculate the energy in the first block of 'lTarget' sampels. */
+ ppi = CB+lMem-lTarget-1;
+ ppo = CB+lMem-1;
+
+ pp=CB+lMem-lTarget;
+ energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+ /* Normalize the energy and store the number of shifts */
+ energyShifts[0] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[0]);
+ energyW16[0] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+
+ /* Compute the energy of the rest of the cb memory
+ * by step wise adding and subtracting the next
+ * sample and the last sample respectively. */
+ WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
+
+ /* Next, precompute the energy values for the filtered cb section */
+ energy=0;
+ pp=filteredCB+lMem-lTarget;
+
+ energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+ /* Normalize the energy and store the number of shifts */
+ energyShifts[base_size] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[base_size]);
+ energyW16[base_size] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+
+ ppi = filteredCB + lMem - 1 - lTarget;
+ ppo = filteredCB + lMem - 1;
+
+ WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
new file mode 100644
index 0000000..1aa2b7b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+
+void WebRtcIlbcfix_CbMemEnergy(
+ WebRtc_Word16 range,
+ WebRtc_Word16 *CB, /* (i) The CB memory (1:st section) */
+ WebRtc_Word16 *filteredCB, /* (i) The filtered CB memory (2:nd section) */
+ WebRtc_Word16 lMem, /* (i) Length of the CB memory */
+ WebRtc_Word16 lTarget, /* (i) Length of the target vector */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
new file mode 100644
index 0000000..0c6f479
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+ WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
+ WebRtc_Word16 *CBmem, /* (i) The CB memory */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size, /* (i) Index to where the energy values should be stored */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
+ ){
+ WebRtc_Word32 energy, tmp32;
+ WebRtc_Word16 *ppe, *pp, *interpSamplesPtr;
+ WebRtc_Word16 *CBmemPtr, lagcount;
+ WebRtc_Word16 *enPtr=&energyW16[base_size-20];
+ WebRtc_Word16 *enShPtr=&energyShifts[base_size-20];
+ WebRtc_Word32 nrjRecursive;
+
+ CBmemPtr = CBmem+147;
+ interpSamplesPtr = interpSamples;
+
+ /* Compute the energy for the first (low-5) noninterpolated samples */
+ nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale);
+ ppe = CBmemPtr - 20;
+
+ for (lagcount=20; lagcount<=39; lagcount++) {
+
+ /* Update the energy recursively to save complexity */
+ nrjRecursive = nrjRecursive +
+ WEBRTC_SPL_MUL_16_16_RSFT(*ppe, *ppe, scale);
+ ppe--;
+ energy = nrjRecursive;
+
+ /* interpolation */
+ energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale);
+ interpSamplesPtr += 4;
+
+ /* Compute energy for the remaining samples */
+ pp = CBmemPtr - lagcount;
+ energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
+
+ /* Normalize the energy and store the number of shifts */
+ (*enShPtr) = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, (*enShPtr));
+ (*enPtr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+ enShPtr++;
+ enPtr++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
new file mode 100644
index 0000000..938b87e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+ WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
+ WebRtc_Word16 *CBmem, /* (i) The CB memory */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size, /* (i) Index to where the energy values should be stored */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
new file mode 100644
index 0000000..40bb708
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/* Compute the energy of the rest of the cb memory
+ * by step wise adding and subtracting the next
+ * sample and the last sample respectively */
+void WebRtcIlbcfix_CbMemEnergyCalc(
+ WebRtc_Word32 energy, /* (i) input start energy */
+ WebRtc_Word16 range, /* (i) number of iterations */
+ WebRtc_Word16 *ppi, /* (i) input pointer 1 */
+ WebRtc_Word16 *ppo, /* (i) input pointer 2 */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
+ )
+{
+ WebRtc_Word16 j,shft;
+ WebRtc_Word32 tmp;
+ WebRtc_Word16 *eSh_ptr;
+ WebRtc_Word16 *eW16_ptr;
+
+
+ eSh_ptr = &energyShifts[1+base_size];
+ eW16_ptr = &energyW16[1+base_size];
+
+ for(j=0;j<range-1;j++) {
+
+ /* Calculate next energy by a +/-
+ operation on the edge samples */
+ tmp = WEBRTC_SPL_MUL_16_16(*ppi, *ppi);
+ tmp -= WEBRTC_SPL_MUL_16_16(*ppo, *ppo);
+ energy += WEBRTC_SPL_RSHIFT_W32(tmp, scale);
+ energy = WEBRTC_SPL_MAX(energy, 0);
+
+ ppi--;
+ ppo--;
+
+ /* Normalize the energy into a WebRtc_Word16 and store
+ the number of shifts */
+
+ shft = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
+ *eSh_ptr++ = shft;
+
+ tmp = WEBRTC_SPL_LSHIFT_W32(energy, shft);
+ *eW16_ptr++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp, 16);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
new file mode 100644
index 0000000..ee2e285
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+
+void WebRtcIlbcfix_CbMemEnergyCalc(
+ WebRtc_Word32 energy, /* (i) input start energy */
+ WebRtc_Word16 range, /* (i) number of iterations */
+ WebRtc_Word16 *ppi, /* (i) input pointer 1 */
+ WebRtc_Word16 *ppo, /* (i) input pointer 2 */
+ WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
+ WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
+ WebRtc_Word16 scale, /* (i) The scaling of all energy values */
+ WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c
new file mode 100644
index 0000000..c51ccf7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -0,0 +1,396 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "gain_quant.h"
+#include "filtered_cb_vecs.h"
+#include "constants.h"
+#include "cb_mem_energy.h"
+#include "interpolate_samples.h"
+#include "cb_mem_energy_augmentation.h"
+#include "cb_search_core.h"
+#include "energy_inverse.h"
+#include "augmented_cb_corr.h"
+#include "cb_update_best_index.h"
+#include "create_augmented_vec.h"
+
+/*----------------------------------------------------------------*
+ * Search routine for codebook encoding and gain quantization.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) the encoder state structure */
+ WebRtc_Word16 *index, /* (o) Codebook indices */
+ WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
+ WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
+ WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
+ WebRtc_Word16 lMem, /* (i) Length of buffer */
+ WebRtc_Word16 lTarget, /* (i) Length of vector */
+ WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
+ WebRtc_Word16 block /* (i) the subblock number */
+ ) {
+ WebRtc_Word16 i, j, stage, range;
+ WebRtc_Word16 *pp, scale, tmp;
+ WebRtc_Word16 bits, temp1, temp2;
+ WebRtc_Word16 base_size;
+ WebRtc_Word32 codedEner, targetEner;
+ WebRtc_Word16 gains[CB_NSTAGES+1];
+ WebRtc_Word16 *cb_vecPtr;
+ WebRtc_Word16 indexOffset, sInd, eInd;
+ WebRtc_Word32 CritMax=0;
+ WebRtc_Word16 shTotMax=WEBRTC_SPL_WORD16_MIN;
+ WebRtc_Word16 bestIndex=0;
+ WebRtc_Word16 bestGain=0;
+ WebRtc_Word16 indexNew, CritNewSh;
+ WebRtc_Word32 CritNew;
+ WebRtc_Word32 *cDotPtr;
+ WebRtc_Word16 noOfZeros;
+ WebRtc_Word16 *gainPtr;
+ WebRtc_Word32 t32, tmpW32;
+ WebRtc_Word16 *WebRtcIlbcfix_kGainSq5_ptr;
+ /* Stack based */
+ WebRtc_Word16 CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
+ WebRtc_Word32 cDot[128];
+ WebRtc_Word32 Crit[128];
+ WebRtc_Word16 targetVec[SUBL+LPC_FILTERORDER];
+ WebRtc_Word16 cbvectors[CB_MEML];
+ WebRtc_Word16 codedVec[SUBL];
+ WebRtc_Word16 interpSamples[20*4];
+ WebRtc_Word16 interpSamplesFilt[20*4];
+ WebRtc_Word16 energyW16[CB_EXPAND*128];
+ WebRtc_Word16 energyShifts[CB_EXPAND*128];
+ WebRtc_Word16 *inverseEnergy=energyW16; /* Reuse memory */
+ WebRtc_Word16 *inverseEnergyShifts=energyShifts; /* Reuse memory */
+ WebRtc_Word16 *buf = &CBbuf[LPC_FILTERORDER];
+ WebRtc_Word16 *target = &targetVec[LPC_FILTERORDER];
+ WebRtc_Word16 *aug_vec = (WebRtc_Word16*)cDot; /* length [SUBL], reuse memory */
+
+ /* Determine size of codebook sections */
+
+ base_size=lMem-lTarget+1;
+ if (lTarget==SUBL) {
+ base_size=lMem-19;
+ }
+
+ /* weighting of the CB memory */
+ noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block];
+ WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER);
+ WebRtcSpl_FilterARFastQ12(
+ decResidual+noOfZeros, buf+noOfZeros,
+ weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]);
+
+ /* weighting of the target vector */
+ WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER);
+ WebRtcSpl_FilterARFastQ12(
+ intarget, target,
+ weightDenum, LPC_FILTERORDER+1, lTarget);
+
+ /* Store target, towards the end codedVec is calculated as
+ the initial target minus the remaining target */
+ WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget);
+
+ /* Find the highest absolute value to calculate proper
+ vector scale factor (so that it uses 12 bits) */
+ temp1 = WebRtcSpl_MaxAbsValueW16(buf, (WebRtc_Word16)lMem);
+ temp2 = WebRtcSpl_MaxAbsValueW16(target, (WebRtc_Word16)lTarget);
+
+ if ((temp1>0)&&(temp2>0)) {
+ temp1 = WEBRTC_SPL_MAX(temp1, temp2);
+ scale = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(temp1, temp1));
+ } else {
+ /* temp1 or temp2 is negative (maximum was -32768) */
+ scale = 30;
+ }
+
+ /* Scale to so that a mul-add 40 times does not overflow */
+ scale = scale - 25;
+ scale = WEBRTC_SPL_MAX(0, scale);
+
+ /* Compute energy of the original target */
+ targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale);
+
+ /* Prepare search over one more codebook section. This section
+ is created by filtering the original buffer with a filter. */
+ WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]);
+
+ range = WebRtcIlbcfix_kSearchRange[block][0];
+
+ if(lTarget == SUBL) {
+ /* Create the interpolated samples and store them for use in all stages */
+
+ /* First section, non-filtered half of the cb */
+ WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem);
+
+ /* Second section, filtered half of the cb */
+ WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem);
+
+ /* Compute the CB vectors' energies for the first cb section (non-filtered) */
+ WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf,
+ scale, 20, energyW16, energyShifts);
+
+ /* Compute the CB vectors' energies for the second cb section (filtered cb) */
+ WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors,
+ scale, (WebRtc_Word16)(base_size+20), energyW16, energyShifts);
+
+ /* Compute the CB vectors' energies and store them in the vector
+ * energyW16. Also the corresponding shift values are stored. The
+ * energy values are used in all three stages. */
+ WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+ lTarget, energyW16+20, energyShifts+20, scale, base_size);
+
+ } else {
+ /* Compute the CB vectors' energies and store them in the vector
+ * energyW16. Also the corresponding shift values are stored. The
+ * energy values are used in all three stages. */
+ WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+ lTarget, energyW16, energyShifts, scale, base_size);
+
+ /* Set the energy positions 58-63 and 122-127 to zero
+ (otherwise they are uninitialized) */
+ WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range));
+ WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range));
+ }
+
+ /* Calculate Inverse Energy (energyW16 is already normalized
+ and will contain the inverse energy in Q29 after this call */
+ WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND);
+
+ /* The gain value computed in the previous stage is used
+ * as an upper limit to what the next stage gain value
+ * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as
+ * the upper limit. */
+ gains[0] = 16384;
+
+ for (stage=0; stage<CB_NSTAGES; stage++) {
+
+ /* Set up memories */
+ range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+ /* initialize search measures */
+ CritMax=0;
+ shTotMax=-100;
+ bestIndex=0;
+ bestGain=0;
+
+ /* loop over lags 40+ in the first codebook section, full search */
+ cb_vecPtr = buf+lMem-lTarget;
+
+ /* Calculate all the cross correlations (augmented part of CB) */
+ if (lTarget==SUBL) {
+ WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem,
+ interpSamples, cDot,
+ 20, 39, scale);
+ cDotPtr=&cDot[20];
+ } else {
+ cDotPtr=cDot;
+ }
+ /* Calculate all the cross correlations (main part of CB) */
+ WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1);
+
+ /* Adjust the search range for the augmented vectors */
+ if (lTarget==SUBL) {
+ range=WebRtcIlbcfix_kSearchRange[block][stage]+20;
+ } else {
+ range=WebRtcIlbcfix_kSearchRange[block][stage];
+ }
+
+ indexOffset=0;
+
+ /* Search for best index in this part of the vector */
+ WebRtcIlbcfix_CbSearchCore(
+ cDot, range, stage, inverseEnergy,
+ inverseEnergyShifts, Crit,
+ &indexNew, &CritNew, &CritNewSh);
+
+ /* Update the global best index and the corresponding gain */
+ WebRtcIlbcfix_CbUpdateBestIndex(
+ CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew+indexOffset],
+ inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+ &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+ sInd=bestIndex-(WebRtc_Word16)(CB_RESRANGE>>1);
+ eInd=sInd+CB_RESRANGE;
+ if (sInd<0) {
+ eInd-=sInd;
+ sInd=0;
+ }
+ if (eInd>=range) {
+ eInd=range-1;
+ sInd=eInd-CB_RESRANGE;
+ }
+
+ range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+ if (lTarget==SUBL) {
+ i=sInd;
+ if (sInd<20) {
+ WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors+lMem,
+ interpSamplesFilt, cDot,
+ (WebRtc_Word16)(sInd+20), (WebRtc_Word16)(WEBRTC_SPL_MIN(39, (eInd+20))), scale);
+ i=20;
+ }
+
+ cDotPtr=&cDot[WEBRTC_SPL_MAX(0,(20-sInd))];
+ cb_vecPtr = cbvectors+lMem-20-i;
+
+ /* Calculate the cross correlations (main part of the filtered CB) */
+ WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-i+1), scale, -1);
+
+ } else {
+ cDotPtr = cDot;
+ cb_vecPtr = cbvectors+lMem-lTarget-sInd;
+
+ /* Calculate the cross correlations (main part of the filtered CB) */
+ WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-sInd+1), scale, -1);
+
+ }
+
+ /* Adjust the search range for the augmented vectors */
+ indexOffset=base_size+sInd;
+
+ /* Search for best index in this part of the vector */
+ WebRtcIlbcfix_CbSearchCore(
+ cDot, (WebRtc_Word16)(eInd-sInd+1), stage, inverseEnergy+indexOffset,
+ inverseEnergyShifts+indexOffset, Crit,
+ &indexNew, &CritNew, &CritNewSh);
+
+ /* Update the global best index and the corresponding gain */
+ WebRtcIlbcfix_CbUpdateBestIndex(
+ CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew],
+ inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+ &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+ index[stage] = bestIndex;
+
+
+ bestGain = WebRtcIlbcfix_GainQuant(bestGain,
+ (WebRtc_Word16)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
+
+ /* Extract the best (according to measure) codebook vector
+ Also adjust the index, so that the augmented vectors are last.
+ Above these vectors were first...
+ */
+
+ if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
+
+ if(index[stage]<base_size) {
+ pp=buf+lMem-lTarget-index[stage];
+ } else {
+ pp=cbvectors+lMem-lTarget-
+ index[stage]+base_size;
+ }
+
+ } else {
+
+ if (index[stage]<base_size) {
+ if (index[stage]>=20) {
+ /* Adjust index and extract vector */
+ index[stage]-=20;
+ pp=buf+lMem-lTarget-index[stage];
+ } else {
+ /* Adjust index and extract vector */
+ index[stage]+=(base_size-20);
+
+ WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-base_size+40),
+ buf+lMem, aug_vec);
+ pp = aug_vec;
+
+ }
+ } else {
+
+ if ((index[stage] - base_size) >= 20) {
+ /* Adjust index and extract vector */
+ index[stage]-=20;
+ pp=cbvectors+lMem-lTarget-
+ index[stage]+base_size;
+ } else {
+ /* Adjust index and extract vector */
+ index[stage]+=(base_size-20);
+ WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-2*base_size+40),
+ cbvectors+lMem, aug_vec);
+ pp = aug_vec;
+ }
+ }
+ }
+
+ /* Subtract the best codebook vector, according
+ to measure, from the target vector */
+
+ WebRtcSpl_AddAffineVectorToVector(target, pp, (WebRtc_Word16)(-bestGain), (WebRtc_Word32)8192, (WebRtc_Word16)14, (int)lTarget);
+
+ /* record quantized gain */
+ gains[stage+1] = bestGain;
+
+ } /* end of Main Loop. for (stage=0;... */
+
+ /* Calculte the coded vector (original target - what's left) */
+ for (i=0;i<lTarget;i++) {
+ codedVec[i]-=target[i];
+ }
+
+ /* Gain adjustment for energy matching */
+ codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale);
+
+ j=gain_index[0];
+
+ temp1 = (WebRtc_Word16)WebRtcSpl_NormW32(codedEner);
+ temp2 = (WebRtc_Word16)WebRtcSpl_NormW32(targetEner);
+
+ if(temp1 < temp2) {
+ bits = 16 - temp1;
+ } else {
+ bits = 16 - temp2;
+ }
+
+ tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(gains[1],gains[1], 14);
+
+ targetEner = WEBRTC_SPL_MUL_16_16(
+ WEBRTC_SPL_SHIFT_W32(targetEner, -bits), tmp);
+
+ tmpW32 = ((WebRtc_Word32)(gains[1]-1))<<1;
+
+ /* Pointer to the table that contains
+ gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
+ gainPtr=(WebRtc_Word16*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
+ temp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
+
+ WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[j];
+
+ /* targetEner and codedEner are in Q(-2*scale) */
+ for (i=gain_index[0];i<32;i++) {
+
+ /* Change the index if
+ (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
+ gainTbl[i] < 2*gain[0]
+ */
+
+ t32 = WEBRTC_SPL_MUL_16_16(temp1, (*gainPtr));
+ t32 = t32 - targetEner;
+ if (t32 < 0) {
+ if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
+ j=i;
+ WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[i];
+ }
+ }
+ gainPtr++;
+ }
+ gain_index[0]=j;
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h
new file mode 100644
index 0000000..e4ad4b5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+
+void WebRtcIlbcfix_CbSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) the encoder state structure */
+ WebRtc_Word16 *index, /* (o) Codebook indices */
+ WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
+ WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
+ WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
+ WebRtc_Word16 lMem, /* (i) Length of buffer */
+ WebRtc_Word16 lTarget, /* (i) Length of vector */
+ WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
+ WebRtc_Word16 block /* (i) the subblock number */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c
new file mode 100644
index 0000000..711e2df
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -0,0 +1,113 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+void WebRtcIlbcfix_CbSearchCore(
+ WebRtc_Word32 *cDot, /* (i) Cross Correlation */
+ WebRtc_Word16 range, /* (i) Search range */
+ WebRtc_Word16 stage, /* (i) Stage of this search */
+ WebRtc_Word16 *inverseEnergy, /* (i) Inversed energy */
+ WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
+ with the offset 2*16-29 */
+ WebRtc_Word32 *Crit, /* (o) The criteria */
+ WebRtc_Word16 *bestIndex, /* (o) Index that corresponds to
+ maximum criteria (in this
+ vector) */
+ WebRtc_Word32 *bestCrit, /* (o) Value of critera for the
+ chosen index */
+ WebRtc_Word16 *bestCritSh) /* (o) The domain of the chosen
+ criteria */
+{
+ WebRtc_Word32 maxW32, tmp32;
+ WebRtc_Word16 max, sh, tmp16;
+ int i;
+ WebRtc_Word32 *cDotPtr;
+ WebRtc_Word16 cDotSqW16;
+ WebRtc_Word16 *inverseEnergyPtr;
+ WebRtc_Word32 *critPtr;
+ WebRtc_Word16 *inverseEnergyShiftPtr;
+
+ /* Don't allow negative values for stage 0 */
+ if (stage==0) {
+ cDotPtr=cDot;
+ for (i=0;i<range;i++) {
+ *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr));
+ cDotPtr++;
+ }
+ }
+
+ /* Normalize cDot to WebRtc_Word16, calculate the square of cDot and store the upper WebRtc_Word16 */
+ maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
+
+ sh = (WebRtc_Word16)WebRtcSpl_NormW32(maxW32);
+ cDotPtr = cDot;
+ inverseEnergyPtr = inverseEnergy;
+ critPtr = Crit;
+ inverseEnergyShiftPtr=inverseEnergyShift;
+ max=WEBRTC_SPL_WORD16_MIN;
+
+ for (i=0;i<range;i++) {
+ /* Calculate cDot*cDot and put the result in a WebRtc_Word16 */
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(*cDotPtr,sh);
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32,16);
+ cDotSqW16 = (WebRtc_Word16)(((WebRtc_Word32)(tmp16)*(tmp16))>>16);
+
+ /* Calculate the criteria (cDot*cDot/energy) */
+ *critPtr=WEBRTC_SPL_MUL_16_16(cDotSqW16, (*inverseEnergyPtr));
+
+ /* Extract the maximum shift value under the constraint
+ that the criteria is not zero */
+ if ((*critPtr)!=0) {
+ max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max);
+ }
+
+ inverseEnergyPtr++;
+ inverseEnergyShiftPtr++;
+ critPtr++;
+ cDotPtr++;
+ }
+
+ /* If no max shifts still at initialization value, set shift to zero */
+ if (max==WEBRTC_SPL_WORD16_MIN) {
+ max = 0;
+ }
+
+ /* Modify the criterias, so that all of them use the same Q domain */
+ critPtr=Crit;
+ inverseEnergyShiftPtr=inverseEnergyShift;
+ for (i=0;i<range;i++) {
+ /* Guarantee that the shift value is less than 16
+ in order to simplify for DSP's (and guard against >31) */
+ tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr));
+
+ (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16);
+ critPtr++;
+ inverseEnergyShiftPtr++;
+ }
+
+ /* Find the index of the best value */
+ *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range);
+ *bestCrit = Crit[*bestIndex];
+
+ /* Calculate total shifts of this criteria */
+ *bestCritSh = 32 - 2*sh + max;
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h
new file mode 100644
index 0000000..e074c52
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+
+#include "defines.h"
+
+void WebRtcIlbcfix_CbSearchCore(
+ WebRtc_Word32 *cDot, /* (i) Cross Correlation */
+ WebRtc_Word16 range, /* (i) Search range */
+ WebRtc_Word16 stage, /* (i) Stage of this search */
+ WebRtc_Word16 *inverseEnergy, /* (i) Inversed energy */
+ WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
+ with the offset 2*16-29 */
+ WebRtc_Word32 *Crit, /* (o) The criteria */
+ WebRtc_Word16 *bestIndex, /* (o) Index that corresponds to
+ maximum criteria (in this
+ vector) */
+ WebRtc_Word32 *bestCrit, /* (o) Value of critera for the
+ chosen index */
+ WebRtc_Word16 *bestCritSh); /* (o) The domain of the chosen
+ criteria */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
new file mode 100644
index 0000000..bf85408
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "cb_update_best_index.h"
+#include "constants.h"
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+ WebRtc_Word32 CritNew, /* (i) New Potentially best Criteria */
+ WebRtc_Word16 CritNewSh, /* (i) Shift value of above Criteria */
+ WebRtc_Word16 IndexNew, /* (i) Index of new Criteria */
+ WebRtc_Word32 cDotNew, /* (i) Cross dot of new index */
+ WebRtc_Word16 invEnergyNew, /* (i) Inversed energy new index */
+ WebRtc_Word16 energyShiftNew, /* (i) Energy shifts of new index */
+ WebRtc_Word32 *CritMax, /* (i/o) Maximum Criteria (so far) */
+ WebRtc_Word16 *shTotMax, /* (i/o) Shifts of maximum criteria */
+ WebRtc_Word16 *bestIndex, /* (i/o) Index that corresponds to
+ maximum criteria */
+ WebRtc_Word16 *bestGain) /* (i/o) Gain in Q14 that corresponds
+ to maximum criteria */
+{
+ WebRtc_Word16 shOld, shNew, tmp16;
+ WebRtc_Word16 scaleTmp;
+ WebRtc_Word32 gainW32;
+
+ /* Normalize the new and old Criteria to the same domain */
+ if (CritNewSh>(*shTotMax)) {
+ shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax));
+ shNew=0;
+ } else {
+ shOld=0;
+ shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh);
+ }
+
+ /* Compare the two criterias. If the new one is better,
+ calculate the gain and store this index as the new best one
+ */
+
+ if (WEBRTC_SPL_RSHIFT_W32(CritNew, shNew)>
+ WEBRTC_SPL_RSHIFT_W32((*CritMax),shOld)) {
+
+ tmp16 = (WebRtc_Word16)WebRtcSpl_NormW32(cDotNew);
+ tmp16 = 16 - tmp16;
+
+ /* Calculate the gain in Q14
+ Compensate for inverseEnergyshift in Q29 and that the energy
+ value was stored in a WebRtc_Word16 (shifted down 16 steps)
+ => 29-14+16 = 31 */
+
+ scaleTmp = -energyShiftNew-tmp16+31;
+ scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
+
+ gainW32 = WEBRTC_SPL_MUL_16_16_RSFT(
+ ((WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16)), invEnergyNew, scaleTmp);
+
+ /* Check if criteria satisfies Gain criteria (max 1.3)
+ if it is larger set the gain to 1.3
+ (slightly different from FLP version)
+ */
+ if (gainW32>21299) {
+ *bestGain=21299;
+ } else if (gainW32<-21299) {
+ *bestGain=-21299;
+ } else {
+ *bestGain=(WebRtc_Word16)gainW32;
+ }
+
+ *CritMax=CritNew;
+ *shTotMax=CritNewSh;
+ *bestIndex = IndexNew;
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
new file mode 100644
index 0000000..9015187
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+
+#include "defines.h"
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+ WebRtc_Word32 CritNew, /* (i) New Potentially best Criteria */
+ WebRtc_Word16 CritNewSh, /* (i) Shift value of above Criteria */
+ WebRtc_Word16 IndexNew, /* (i) Index of new Criteria */
+ WebRtc_Word32 cDotNew, /* (i) Cross dot of new index */
+ WebRtc_Word16 invEnergyNew, /* (i) Inversed energy new index */
+ WebRtc_Word16 energyShiftNew, /* (i) Energy shifts of new index */
+ WebRtc_Word32 *CritMax, /* (i/o) Maximum Criteria (so far) */
+ WebRtc_Word16 *shTotMax, /* (i/o) Shifts of maximum criteria */
+ WebRtc_Word16 *bestIndex, /* (i/o) Index that corresponds to
+ maximum criteria */
+ WebRtc_Word16 *bestGain); /* (i/o) Gain in Q14 that corresponds
+ to maximum criteria */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c
new file mode 100644
index 0000000..90108ff
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*------------------------------------------------------------------*
+ * Calculate the Chevyshev polynomial series
+ * F(w) = 2*exp(-j5w)*C(x)
+ * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ * T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
+ /* (o) Result of C(x) */
+ WebRtc_Word16 x, /* (i) Value to the Chevyshev polynomial */
+ WebRtc_Word16 *f /* (i) The coefficients in the polynomial */
+ ) {
+ WebRtc_Word16 b1_high, b1_low; /* Use the high, low format to increase the accuracy */
+ WebRtc_Word32 b2;
+ WebRtc_Word32 tmp1W32;
+ WebRtc_Word32 tmp2W32;
+ int i;
+
+ b2 = (WebRtc_Word32)0x1000000; /* b2 = 1.0 (Q23) */
+ /* Calculate b1 = 2*x + f[1] */
+ tmp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x, 10);
+ tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[1], 14);
+
+ for (i = 2; i < 5; i++) {
+ tmp2W32 = tmp1W32;
+
+ /* Split b1 (in tmp1W32) into a high and low part */
+ b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
+ b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
+
+ /* Calculate 2*x*b1-b2+f[i] */
+ tmp1W32 = WEBRTC_SPL_LSHIFT_W32( (WEBRTC_SPL_MUL_16_16(b1_high, x) +
+ WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15)), 2);
+
+ tmp1W32 -= b2;
+ tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 14);
+
+ /* Update b2 for next round */
+ b2 = tmp2W32;
+ }
+
+ /* Split b1 (in tmp1W32) into a high and low part */
+ b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
+ b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
+
+ /* tmp1W32 = x*b1 - b2 + f[i]/2 */
+ tmp1W32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(b1_high, x), 1) +
+ WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15), 1);
+
+ tmp1W32 -= b2;
+ tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 13);
+
+ /* Handle overflows and set to maximum or minimum WebRtc_Word16 instead */
+ if (tmp1W32>((WebRtc_Word32)33553408)) {
+ return(WEBRTC_SPL_WORD16_MAX);
+ } else if (tmp1W32<((WebRtc_Word32)-33554432)) {
+ return(WEBRTC_SPL_WORD16_MIN);
+ } else {
+ return((WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 10));
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h
new file mode 100644
index 0000000..57aab99
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+
+#include "defines.h"
+
+/*------------------------------------------------------------------*
+ * Calculate the Chevyshev polynomial series
+ * F(w) = 2*exp(-j5w)*C(x)
+ * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ * T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
+ /* (o) Result of C(x) */
+ WebRtc_Word16 x, /* (i) Value to the Chevyshev polynomial */
+ WebRtc_Word16 *f /* (i) The coefficients in the polynomial */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c
new file mode 100644
index 0000000..3d7f93e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Compute cross correlation and pitch gain for pitch prediction
+ * of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(
+ WebRtc_Word32 *corr, /* (o) cross correlation */
+ WebRtc_Word32 *ener, /* (o) energy */
+ WebRtc_Word16 *buffer, /* (i) signal buffer */
+ WebRtc_Word16 lag, /* (i) pitch lag */
+ WebRtc_Word16 bLen, /* (i) length of buffer */
+ WebRtc_Word16 sRange, /* (i) correlation search length */
+ WebRtc_Word16 scale /* (i) number of rightshifts to use */
+ ){
+ WebRtc_Word16 *w16ptr;
+
+ w16ptr=&buffer[bLen-sRange-lag];
+
+ /* Calculate correlation and energy */
+ (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale);
+ (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale);
+
+ /* For zero energy set the energy to 0 in order to avoid potential
+ problems for coming divisions */
+ if (*ener == 0) {
+ *corr = 0;
+ *ener = 1;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h
new file mode 100644
index 0000000..cd46532
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Compute cross correlation and pitch gain for pitch prediction
+ * of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(
+ WebRtc_Word32 *corr, /* (o) cross correlation */
+ WebRtc_Word32 *ener, /* (o) energy */
+ WebRtc_Word16 *buffer, /* (i) signal buffer */
+ WebRtc_Word16 lag, /* (i) pitch lag */
+ WebRtc_Word16 bLen, /* (i) length of buffer */
+ WebRtc_Word16 sRange, /* (i) correlation search length */
+ WebRtc_Word16 scale /* (i) number of rightshifts to use */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m
new file mode 100644
index 0000000..f768194
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m
@@ -0,0 +1,49 @@
+clear;
+pack;
+%
+% Enter the path to YOUR executable and remember to define the perprocessor
+% variable PRINT_MIPS te get the instructions printed to the screen.
+%
+command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat';
+cout=' > st.txt'; %saves to matlab variable 'st'
+eval(strcat(command,cout));
+if(length(cout)>3)
+ load st.txt
+else
+ disp('No cout file to load')
+end
+
+% initialize vector to zero
+index = find(st(1:end,1)==-1);
+indexnonzero = find(st(1:end,1)>0);
+frames = length(index)-indexnonzero(1)+1;
+start = indexnonzero(1) - 1;
+functionOrder=max(st(:,2));
+new=zeros(frames,functionOrder);
+
+for i = 1:frames,
+ for j = index(start-1+i)+1:(index(start+i)-1),
+ new(i,st(j,2)) = new(i,st(j,2)) + st(j,1);
+ end
+end
+
+result=zeros(functionOrder,3);
+for i=1:functionOrder
+ nonzeroelements = find(new(1:end,i)>0);
+ result(i,1)=i;
+
+ % Compute each function's mean complexity
+ % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000;
+
+ % Compute each function's maximum complexity in encoding
+ % and decoding respectively and then add it together:
+ % result(i,3)=(max(new(1:end,i))/0.03)/1000000;
+ result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000;
+end
+
+result
+
+% Compute maximum complexity for a single frame (enc/dec separately and together)
+maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000
+maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000
+totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/constants.c b/trunk/src/modules/audio_coding/codecs/ilbc/constants.c
new file mode 100644
index 0000000..5ebe9be
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/constants.c
@@ -0,0 +1,666 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
+
+const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
+const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
+
+/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
+ the 80 sample start state
+*/
+const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
+ 1638, 1843, 2048, 1843, 1638
+};
+
+/* LP Filter coeffs used for downsampling */
+const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
+ -273, 512, 1297, 1696, 1297, 512, -273
+};
+
+/* Constants used in the LPC calculations */
+
+/* Hanning LPC window (in Q15) */
+const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
+ 6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
+ 932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
+ 3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
+ 7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240,
+ 11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277,
+ 16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325,
+ 21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893,
+ 26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538,
+ 29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906,
+ 32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767,
+ 32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037,
+ 31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788,
+ 29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238,
+ 25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730,
+ 21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704,
+ 16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647,
+ 11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050,
+ 6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359,
+ 3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932,
+ 795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6
+};
+
+/* Asymmetric LPC window (in Q15)*/
+const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
+ 2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
+ 280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
+ 1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
+ 2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684,
+ 3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631,
+ 5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891,
+ 8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398,
+ 10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080,
+ 13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859,
+ 16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653,
+ 18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380,
+ 21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962,
+ 24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323,
+ 26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394,
+ 28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115,
+ 30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436,
+ 31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317,
+ 32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734,
+ 32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939,
+ 26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571
+};
+
+/* Lag window for LPC (Q31) */
+const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
+ 2147483647, 2144885453, 2137754373, 2125918626, 2109459810,
+ 2088483140, 2063130336, 2033564590, 1999977009, 1962580174,
+ 1921610283};
+
+/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
+ * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
+ */
+const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
+ 32767, 29573, 26690, 24087,
+ 21739, 19619, 17707, 15980,
+ 14422, 13016, 11747};
+
+/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
+ * floating point vector {1 0.4222 0.4222^2... }
+ */
+const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
+ 32767, 13835, 5841, 2466, 1041, 440,
+ 186, 78, 33, 14, 6};
+
+/* LSF quantization Q13 domain */
+const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
+ 1273, 2238, 3696,
+ 3199, 5309, 8209,
+ 3606, 5671, 7829,
+ 2815, 5262, 8778,
+ 2608, 4027, 5493,
+ 1582, 3076, 5945,
+ 2983, 4181, 5396,
+ 2437, 4322, 6902,
+ 1861, 2998, 4613,
+ 2007, 3250, 5214,
+ 1388, 2459, 4262,
+ 2563, 3805, 5269,
+ 2036, 3522, 5129,
+ 1935, 4025, 6694,
+ 2744, 5121, 7338,
+ 2810, 4248, 5723,
+ 3054, 5405, 7745,
+ 1449, 2593, 4763,
+ 3411, 5128, 6596,
+ 2484, 4659, 7496,
+ 1668, 2879, 4818,
+ 1812, 3072, 5036,
+ 1638, 2649, 3900,
+ 2464, 3550, 4644,
+ 1853, 2900, 4158,
+ 2458, 4163, 5830,
+ 2556, 4036, 6254,
+ 2703, 4432, 6519,
+ 3062, 4953, 7609,
+ 1725, 3703, 6187,
+ 2221, 3877, 5427,
+ 2339, 3579, 5197,
+ 2021, 4633, 7037,
+ 2216, 3328, 4535,
+ 2961, 4739, 6667,
+ 2807, 3955, 5099,
+ 2788, 4501, 6088,
+ 1642, 2755, 4431,
+ 3341, 5282, 7333,
+ 2414, 3726, 5727,
+ 1582, 2822, 5269,
+ 2259, 3447, 4905,
+ 3117, 4986, 7054,
+ 1825, 3491, 5542,
+ 3338, 5736, 8627,
+ 1789, 3090, 5488,
+ 2566, 3720, 4923,
+ 2846, 4682, 7161,
+ 1950, 3321, 5976,
+ 1834, 3383, 6734,
+ 3238, 4769, 6094,
+ 2031, 3978, 5903,
+ 1877, 4068, 7436,
+ 2131, 4644, 8296,
+ 2764, 5010, 8013,
+ 2194, 3667, 6302,
+ 2053, 3127, 4342,
+ 3523, 6595, 10010,
+ 3134, 4457, 5748,
+ 3142, 5819, 9414,
+ 2223, 4334, 6353,
+ 2022, 3224, 4822,
+ 2186, 3458, 5544,
+ 2552, 4757, 6870,
+ 10905, 12917, 14578,
+ 9503, 11485, 14485,
+ 9518, 12494, 14052,
+ 6222, 7487, 9174,
+ 7759, 9186, 10506,
+ 8315, 12755, 14786,
+ 9609, 11486, 13866,
+ 8909, 12077, 13643,
+ 7369, 9054, 11520,
+ 9408, 12163, 14715,
+ 6436, 9911, 12843,
+ 7109, 9556, 11884,
+ 7557, 10075, 11640,
+ 6482, 9202, 11547,
+ 6463, 7914, 10980,
+ 8611, 10427, 12752,
+ 7101, 9676, 12606,
+ 7428, 11252, 13172,
+ 10197, 12955, 15842,
+ 7487, 10955, 12613,
+ 5575, 7858, 13621,
+ 7268, 11719, 14752,
+ 7476, 11744, 13795,
+ 7049, 8686, 11922,
+ 8234, 11314, 13983,
+ 6560, 11173, 14984,
+ 6405, 9211, 12337,
+ 8222, 12054, 13801,
+ 8039, 10728, 13255,
+ 10066, 12733, 14389,
+ 6016, 7338, 10040,
+ 6896, 8648, 10234,
+ 7538, 9170, 12175,
+ 7327, 12608, 14983,
+ 10516, 12643, 15223,
+ 5538, 7644, 12213,
+ 6728, 12221, 14253,
+ 7563, 9377, 12948,
+ 8661, 11023, 13401,
+ 7280, 8806, 11085,
+ 7723, 9793, 12333,
+ 12225, 14648, 16709,
+ 8768, 13389, 15245,
+ 10267, 12197, 13812,
+ 5301, 7078, 11484,
+ 7100, 10280, 11906,
+ 8716, 12555, 14183,
+ 9567, 12464, 15434,
+ 7832, 12305, 14300,
+ 7608, 10556, 12121,
+ 8913, 11311, 12868,
+ 7414, 9722, 11239,
+ 8666, 11641, 13250,
+ 9079, 10752, 12300,
+ 8024, 11608, 13306,
+ 10453, 13607, 16449,
+ 8135, 9573, 10909,
+ 6375, 7741, 10125,
+ 10025, 12217, 14874,
+ 6985, 11063, 14109,
+ 9296, 13051, 14642,
+ 8613, 10975, 12542,
+ 6583, 10414, 13534,
+ 6191, 9368, 13430,
+ 5742, 6859, 9260,
+ 7723, 9813, 13679,
+ 8137, 11291, 12833,
+ 6562, 8973, 10641,
+ 6062, 8462, 11335,
+ 6928, 8784, 12647,
+ 7501, 8784, 10031,
+ 8372, 10045, 12135,
+ 8191, 9864, 12746,
+ 5917, 7487, 10979,
+ 5516, 6848, 10318,
+ 6819, 9899, 11421,
+ 7882, 12912, 15670,
+ 9558, 11230, 12753,
+ 7752, 9327, 11472,
+ 8479, 9980, 11358,
+ 11418, 14072, 16386,
+ 7968, 10330, 14423,
+ 8423, 10555, 12162,
+ 6337, 10306, 14391,
+ 8850, 10879, 14276,
+ 6750, 11885, 15710,
+ 7037, 8328, 9764,
+ 6914, 9266, 13476,
+ 9746, 13949, 15519,
+ 11032, 14444, 16925,
+ 8032, 10271, 11810,
+ 10962, 13451, 15833,
+ 10021, 11667, 13324,
+ 6273, 8226, 12936,
+ 8543, 10397, 13496,
+ 7936, 10302, 12745,
+ 6769, 8138, 10446,
+ 6081, 7786, 11719,
+ 8637, 11795, 14975,
+ 8790, 10336, 11812,
+ 7040, 8490, 10771,
+ 7338, 10381, 13153,
+ 6598, 7888, 9358,
+ 6518, 8237, 12030,
+ 9055, 10763, 12983,
+ 6490, 10009, 12007,
+ 9589, 12023, 13632,
+ 6867, 9447, 10995,
+ 7930, 9816, 11397,
+ 10241, 13300, 14939,
+ 5830, 8670, 12387,
+ 9870, 11915, 14247,
+ 9318, 11647, 13272,
+ 6721, 10836, 12929,
+ 6543, 8233, 9944,
+ 8034, 10854, 12394,
+ 9112, 11787, 14218,
+ 9302, 11114, 13400,
+ 9022, 11366, 13816,
+ 6962, 10461, 12480,
+ 11288, 13333, 15222,
+ 7249, 8974, 10547,
+ 10566, 12336, 14390,
+ 6697, 11339, 13521,
+ 11851, 13944, 15826,
+ 6847, 8381, 11349,
+ 7509, 9331, 10939,
+ 8029, 9618, 11909,
+ 13973, 17644, 19647, 22474,
+ 14722, 16522, 20035, 22134,
+ 16305, 18179, 21106, 23048,
+ 15150, 17948, 21394, 23225,
+ 13582, 15191, 17687, 22333,
+ 11778, 15546, 18458, 21753,
+ 16619, 18410, 20827, 23559,
+ 14229, 15746, 17907, 22474,
+ 12465, 15327, 20700, 22831,
+ 15085, 16799, 20182, 23410,
+ 13026, 16935, 19890, 22892,
+ 14310, 16854, 19007, 22944,
+ 14210, 15897, 18891, 23154,
+ 14633, 18059, 20132, 22899,
+ 15246, 17781, 19780, 22640,
+ 16396, 18904, 20912, 23035,
+ 14618, 17401, 19510, 21672,
+ 15473, 17497, 19813, 23439,
+ 18851, 20736, 22323, 23864,
+ 15055, 16804, 18530, 20916,
+ 16490, 18196, 19990, 21939,
+ 11711, 15223, 21154, 23312,
+ 13294, 15546, 19393, 21472,
+ 12956, 16060, 20610, 22417,
+ 11628, 15843, 19617, 22501,
+ 14106, 16872, 19839, 22689,
+ 15655, 18192, 20161, 22452,
+ 12953, 15244, 20619, 23549,
+ 15322, 17193, 19926, 21762,
+ 16873, 18676, 20444, 22359,
+ 14874, 17871, 20083, 21959,
+ 11534, 14486, 19194, 21857,
+ 17766, 19617, 21338, 23178,
+ 13404, 15284, 19080, 23136,
+ 15392, 17527, 19470, 21953,
+ 14462, 16153, 17985, 21192,
+ 17734, 19750, 21903, 23783,
+ 16973, 19096, 21675, 23815,
+ 16597, 18936, 21257, 23461,
+ 15966, 17865, 20602, 22920,
+ 15416, 17456, 20301, 22972,
+ 18335, 20093, 21732, 23497,
+ 15548, 17217, 20679, 23594,
+ 15208, 16995, 20816, 22870,
+ 13890, 18015, 20531, 22468,
+ 13211, 15377, 19951, 22388,
+ 12852, 14635, 17978, 22680,
+ 16002, 17732, 20373, 23544,
+ 11373, 14134, 19534, 22707,
+ 17329, 19151, 21241, 23462,
+ 15612, 17296, 19362, 22850,
+ 15422, 19104, 21285, 23164,
+ 13792, 17111, 19349, 21370,
+ 15352, 17876, 20776, 22667,
+ 15253, 16961, 18921, 22123,
+ 14108, 17264, 20294, 23246,
+ 15785, 17897, 20010, 21822,
+ 17399, 19147, 20915, 22753,
+ 13010, 15659, 18127, 20840,
+ 16826, 19422, 22218, 24084,
+ 18108, 20641, 22695, 24237,
+ 18018, 20273, 22268, 23920,
+ 16057, 17821, 21365, 23665,
+ 16005, 17901, 19892, 23016,
+ 13232, 16683, 21107, 23221,
+ 13280, 16615, 19915, 21829,
+ 14950, 18575, 20599, 22511,
+ 16337, 18261, 20277, 23216,
+ 14306, 16477, 21203, 23158,
+ 12803, 17498, 20248, 22014,
+ 14327, 17068, 20160, 22006,
+ 14402, 17461, 21599, 23688,
+ 16968, 18834, 20896, 23055,
+ 15070, 17157, 20451, 22315,
+ 15419, 17107, 21601, 23946,
+ 16039, 17639, 19533, 21424,
+ 16326, 19261, 21745, 23673,
+ 16489, 18534, 21658, 23782,
+ 16594, 18471, 20549, 22807,
+ 18973, 21212, 22890, 24278,
+ 14264, 18674, 21123, 23071,
+ 15117, 16841, 19239, 23118,
+ 13762, 15782, 20478, 23230,
+ 14111, 15949, 20058, 22354,
+ 14990, 16738, 21139, 23492,
+ 13735, 16971, 19026, 22158,
+ 14676, 17314, 20232, 22807,
+ 16196, 18146, 20459, 22339,
+ 14747, 17258, 19315, 22437,
+ 14973, 17778, 20692, 23367,
+ 15715, 17472, 20385, 22349,
+ 15702, 18228, 20829, 23410,
+ 14428, 16188, 20541, 23630,
+ 16824, 19394, 21365, 23246,
+ 13069, 16392, 18900, 21121,
+ 12047, 16640, 19463, 21689,
+ 14757, 17433, 19659, 23125,
+ 15185, 16930, 19900, 22540,
+ 16026, 17725, 19618, 22399,
+ 16086, 18643, 21179, 23472,
+ 15462, 17248, 19102, 21196,
+ 17368, 20016, 22396, 24096,
+ 12340, 14475, 19665, 23362,
+ 13636, 16229, 19462, 22728,
+ 14096, 16211, 19591, 21635,
+ 12152, 14867, 19943, 22301,
+ 14492, 17503, 21002, 22728,
+ 14834, 16788, 19447, 21411,
+ 14650, 16433, 19326, 22308,
+ 14624, 16328, 19659, 23204,
+ 13888, 16572, 20665, 22488,
+ 12977, 16102, 18841, 22246,
+ 15523, 18431, 21757, 23738,
+ 14095, 16349, 18837, 20947,
+ 13266, 17809, 21088, 22839,
+ 15427, 18190, 20270, 23143,
+ 11859, 16753, 20935, 22486,
+ 12310, 17667, 21736, 23319,
+ 14021, 15926, 18702, 22002,
+ 12286, 15299, 19178, 21126,
+ 15703, 17491, 21039, 23151,
+ 12272, 14018, 18213, 22570,
+ 14817, 16364, 18485, 22598,
+ 17109, 19683, 21851, 23677,
+ 12657, 14903, 19039, 22061,
+ 14713, 16487, 20527, 22814,
+ 14635, 16726, 18763, 21715,
+ 15878, 18550, 20718, 22906
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
+const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
+
+const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
+ 2308, 3652, 5434, 7885,
+ 10255, 12559, 15160, 17513,
+ 20328, 22752};
+
+const WebRtc_Word16 WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
+ 31476, 29565, 25819, 18725, 10276,
+ 1236, -9049, -17600, -25884, -30618
+};
+
+/* Q14 */
+const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
+const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
+
+/*
+ cos(x) in Q15
+ WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0)
+ used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const WebRtc_Word16 WebRtcIlbcfix_kCos[64] = {
+ 32767, 32729, 32610, 32413, 32138, 31786, 31357, 30853,
+ 30274, 29622, 28899, 28106, 27246, 26320, 25330, 24279,
+ 23170, 22006, 20788, 19520, 18205, 16846, 15447, 14010,
+ 12540, 11039, 9512, 7962, 6393, 4808, 3212, 1608,
+ 0, -1608, -3212, -4808, -6393, -7962, -9512, -11039,
+ -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006,
+ -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622,
+ -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729
+};
+
+/*
+ Derivative in Q19, used to interpolate between the
+ WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
+*/
+const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[64] = {
+ -632, -1893, -3150, -4399, -5638, -6863, -8072, -9261,
+ -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
+ -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
+ -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726,
+ -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009,
+ -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637,
+ -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428,
+ -9261, -8072, -6863, -5638, -4399, -3150, -1893, -632};
+
+/*
+ Table in Q15, used for a2lsf conversion
+ WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
+*/
+
+const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
+ 32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
+ 29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
+ 21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
+ 10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425,
+ -5126, -6812, -8480, -10125, -11743, -13327, -14876,
+ -16384, -17846, -19260, -20621, -21926, -23170, -24351,
+ -25465, -26509, -27481, -28377, -29196, -29935, -30591,
+ -31164, -31651, -32051, -32364, -32588, -32723, -32760
+};
+
+/*
+ Derivative of y = acos(x) in Q12
+ used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[64] = {
+ -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
+ -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
+ -901, -861, -827, -797, -772, -750, -730, -713,
+ -699, -687, -677, -668, -662, -657, -654, -652,
+ -652, -654, -657, -662, -668, -677, -687, -699,
+ -713, -730, -750, -772, -797, -827, -861, -901,
+ -946, -998, -1059, -1132, -1219, -1322, -1450, -1608,
+ -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887
+};
+
+
+/* Tables for quantization of start state */
+
+/* State quantization tables */
+const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
+ -30473, -17838, -9257, -2537,
+ 3639, 10893, 19958, 32636
+};
+
+/* This table defines the limits for the selection of the freqg
+ less or equal than value 0 => index = 0
+ less or equal than value k => index = k
+*/
+const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[64] = {
+ 118, 163, 222, 305, 425, 604,
+ 851, 1174, 1617, 2222, 3080, 4191,
+ 5525, 7215, 9193, 11540, 14397, 17604,
+ 21204, 25209, 29863, 35720, 42531, 50375,
+ 59162, 68845, 80108, 93754, 110326, 129488,
+ 150654, 174328, 201962, 233195, 267843, 308239,
+ 354503, 405988, 464251, 531550, 608652, 697516,
+ 802526, 928793, 1080145, 1258120, 1481106, 1760881,
+ 2111111, 2546619, 3078825, 3748642, 4563142, 5573115,
+ 6887601, 8582108, 10797296, 14014513, 18625760, 25529599,
+ 37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kScale[64] = {
+ /* Values in Q16 */
+ 29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
+ 6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
+ 2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083,
+ /* Values in Q21 */
+ 32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609,
+ 16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435,
+ 8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798,
+ 3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747
+};
+
+/*frgq in fixpoint, but already computed like this:
+ for(i=0; i<64; i++){
+ a = (pow(10,frgq[i])/4.5);
+ WebRtcIlbcfix_kFrgQuantMod[i] = round(a);
+ }
+
+ Value 0 :36 in Q8
+ 37:58 in Q5
+ 59:63 in Q3
+*/
+const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[64] = {
+ /* First 37 values in Q8 */
+ 569, 671, 786, 916, 1077, 1278,
+ 1529, 1802, 2109, 2481, 2898, 3440,
+ 3943, 4535, 5149, 5778, 6464, 7208,
+ 7904, 8682, 9397, 10285, 11240, 12246,
+ 13313, 14382, 15492, 16735, 18131, 19693,
+ 21280, 22912, 24624, 26544, 28432, 30488,
+ 32720,
+ /* 22 values in Q5 */
+ 4383, 4684, 5012, 5363, 5739, 6146,
+ 6603, 7113, 7679, 8285, 9040, 9850,
+ 10838, 11882, 13103, 14467, 15950, 17669,
+ 19712, 22016, 24800, 28576,
+ /* 5 values in Q3 */
+ 8240, 9792, 12040, 15440, 22472
+};
+
+/* Constants for codebook search and creation */
+
+/* Expansion filter to get additional cb section.
+ * Q12 and reversed compared to flp
+ */
+const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
+ -140, 446, -755, 3302, 2922, -590, 343, -138};
+
+/* Weighting coefficients for short lags.
+ * [0.2 0.4 0.6 0.8] in Q15 */
+const WebRtc_Word16 WebRtcIlbcfix_kAlpha[4]={
+ 6554, 13107, 19661, 26214};
+
+/* Ranges for search and filters at different subframes */
+
+const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+ {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
+
+const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+
+/* Gain Quantization for the codebook gains of the 3 stages */
+
+/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
+const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[9]={
+ -16384, -10813, -5407, 0, 4096, 8192,
+ 12288, 16384, 32767};
+
+/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
+const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[17]={
+ -17203, -14746, -12288, -9830, -7373, -4915,
+ -2458, 0, 2458, 4915, 7373, 9830,
+ 12288, 14746, 17203, 19661, 32767};
+
+/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
+const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[33]={
+ 614, 1229, 1843, 2458, 3072, 3686,
+ 4301, 4915, 5530, 6144, 6758, 7373,
+ 7987, 8602, 9216, 9830, 10445, 11059,
+ 11674, 12288, 12902, 13517, 14131, 14746,
+ 15360, 15974, 16589, 17203, 17818, 18432,
+ 19046, 19661, 32767};
+
+/* Q14 gain_sq5Tbl squared in Q14 */
+const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[32] = {
+ 23, 92, 207, 368, 576, 829,
+ 1129, 1474, 1866, 2304, 2787, 3317,
+ 3893, 4516, 5184, 5897, 6658, 7464,
+ 8318, 9216, 10160, 11151, 12187, 13271,
+ 14400, 15574, 16796, 18062, 19377, 20736,
+ 22140, 23593
+};
+
+const WebRtc_Word16* const WebRtcIlbcfix_kGain[3] =
+{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
+
+
+/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
+
+const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
+ {0, 0, 0, 4096, 0, 0, 0},
+ {64, -315, 1181, 3531, -436, 77, -64},
+ {97, -509, 2464, 2464, -509, 97, -97},
+ {77, -436, 3531, 1181, -315, 64, -77}
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[3] = {
+ 4800, 16384, 27968 /* Q16 */
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+ 160, 480, 800, 1120, 1440, 1760, 2080, 2400 /* Q(-2) */
+};
+
+/* PLC table */
+
+const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
+ 839, 1343, 2048, 2998, 4247, 5849
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
+ 0, 5462, 10922, 16384, 21846, 27306
+};
+
+const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
+ 26667, 18729, 13653, 10258, 7901, 6214
+};
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/constants.h b/trunk/src/modules/audio_coding/codecs/ilbc/constants.h
new file mode 100644
index 0000000..f787f74
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/constants.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+
+#include "defines.h"
+#include "typedefs.h"
+
+/* high pass filters */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[];
+
+/* Window for start state decision */
+extern const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[];
+
+/* low pass filter used for downsampling */
+extern const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[];
+
+/* LPC analysis and quantization */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[];
+extern const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kLspMean[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kCos[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[];
+
+/* state quantization tables */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[];
+extern const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kScale[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[];
+
+/* Ranges for search and filters at different subframes */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[];
+
+/* gain quantization tables */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[];
+extern const WebRtc_Word16* const WebRtcIlbcfix_kGain[];
+
+/* adaptive codebook definitions */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kAlpha[];
+
+/* enhancer definitions */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
+extern const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[];
+
+/* PLC tables */
+
+extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[];
+extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[];
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
new file mode 100644
index 0000000..f021c4d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+ WebRtc_Word16 index, /* (i) Index for the augmented vector to be created */
+ WebRtc_Word16 *buffer, /* (i) Pointer to the end of the codebook memory that
+ is used for creation of the augmented codebook */
+ WebRtc_Word16 *cbVec /* (o) The construced codebook vector */
+ ) {
+ WebRtc_Word16 ilow;
+ WebRtc_Word16 *ppo, *ppi;
+ WebRtc_Word16 cbVecTmp[4];
+
+ ilow = index-4;
+
+ /* copy the first noninterpolated part */
+ ppo = buffer-index;
+ WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index);
+
+ /* interpolation */
+ ppo = buffer - 4;
+ ppi = buffer - index - 4;
+
+ /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) + ((ppo[k]*alphaTbl[3-k])>>15);
+ for k = 0..3
+ */
+ WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha, 4, 15);
+ WebRtcSpl_ReverseOrderMultArrayElements(cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[3], 4, 15);
+ WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, 4, 0);
+
+ /* copy the second noninterpolated part */
+ ppo = buffer - index;
+ WEBRTC_SPL_MEMCPY_W16(cbVec+index,ppo,(SUBL-index));
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
new file mode 100644
index 0000000..970a9be
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+ WebRtc_Word16 index, /* (i) Index for the augmented vector to be created */
+ WebRtc_Word16 *buffer, /* (i) Pointer to the end of the codebook memory that
+ is used for creation of the augmented codebook */
+ WebRtc_Word16 *cbVec /* (o) The construced codebook vector */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode.c b/trunk/src/modules/audio_coding/codecs/ilbc/decode.c
new file mode 100644
index 0000000..827532c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decode.c
@@ -0,0 +1,244 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "simple_lsf_dequant.h"
+#include "decoder_interpolate_lsf.h"
+#include "index_conv_dec.h"
+#include "do_plc.h"
+#include "constants.h"
+#include "enhancer_interface.h"
+#include "xcorr_coef.h"
+#include "lsf_check.h"
+#include "decode_residual.h"
+#include "unpack_bits.h"
+#include "hp_output.h"
+#ifndef WEBRTC_BIG_ENDIAN
+#include "swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ * main decoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecodeImpl(
+ WebRtc_Word16 *decblock, /* (o) decoded signal block */
+ WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ WebRtc_Word16 mode /* (i) 0: bad packet, PLC,
+ 1: normal */
+ ) {
+ int i;
+ WebRtc_Word16 order_plus_one;
+
+ WebRtc_Word16 last_bit;
+ WebRtc_Word16 *data;
+ /* Stack based */
+ WebRtc_Word16 decresidual[BLOCKL_MAX];
+ WebRtc_Word16 PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
+ WebRtc_Word16 syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+ WebRtc_Word16 PLClpc[LPC_FILTERORDER + 1];
+ iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
+
+ /* Reuse some buffers that are non overlapping in order to save stack memory */
+ data = &PLCresidual[LPC_FILTERORDER];
+
+ if (mode>0) { /* the data are good */
+
+ /* decode data */
+
+#ifndef WEBRTC_BIG_ENDIAN
+ WebRtcIlbcfix_SwapBytes((WebRtc_UWord16*)bytes, iLBCdec_inst->no_of_words);
+#endif
+
+ /* Unpacketize bits into parameters */
+
+ last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode);
+
+#ifndef WEBRTC_BIG_ENDIAN
+ /* Swap back so that the the input vector "bytes" is unchanged */
+ WebRtcIlbcfix_SwapBytes((WebRtc_UWord16*)bytes, iLBCdec_inst->no_of_words);
+#endif
+
+ /* Check for bit errors */
+ if (iLBCbits_inst->startIdx<1)
+ mode = 0;
+ if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3))
+ mode = 0;
+ if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5))
+ mode = 0;
+ if (last_bit==1)
+ mode = 0;
+
+ if (mode==1) { /* No bit errors was detected, continue decoding */
+ /* Stack based */
+ WebRtc_Word16 lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+ WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+ /* adjust index */
+ WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
+
+ /* decode the lsf */
+ WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (WebRtc_Word16*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
+ WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
+ WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
+ lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
+
+ /* Decode the residual using the cb and gain indexes */
+ WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst, decresidual, syntdenum);
+
+ /* preparing the plc for a future loss! */
+ WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 0,
+ decresidual, syntdenum + (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
+ (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
+
+ /* Use the output from doThePLC */
+ WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+ }
+
+ }
+
+ if (mode == 0) {
+ /* the data is bad (either a PLC call
+ * was made or a bit error was detected)
+ */
+
+ /* packet loss conceal */
+
+ WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 1,
+ decresidual, syntdenum, (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
+
+ WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+
+ order_plus_one = LPC_FILTERORDER + 1;
+
+ for (i = 0; i < iLBCdec_inst->nsub; i++) {
+ WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one),
+ PLClpc, order_plus_one);
+ }
+ }
+
+ if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */
+
+ /* Update the filter and filter coefficients if there was a packet loss */
+ if (iLBCdec_inst->prev_enh_pl==2) {
+ for (i=0;i<iLBCdec_inst->nsub;i++) {
+ WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]),
+ syntdenum, (LPC_FILTERORDER+1));
+ }
+ }
+
+ /* post filtering */
+ (*iLBCdec_inst).last_lag =
+ WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst);
+
+ /* synthesis filtering */
+
+ /* Set up the filter state */
+ WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+ if (iLBCdec_inst->mode==20) {
+ /* Enhancer has 40 samples delay */
+ i=0;
+ WebRtcSpl_FilterARFastQ12(
+ data, data,
+ iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
+ LPC_FILTERORDER+1, SUBL);
+
+ for (i=1; i < iLBCdec_inst->nsub; i++) {
+ WebRtcSpl_FilterARFastQ12(
+ data+i*SUBL, data+i*SUBL,
+ syntdenum+(i-1)*(LPC_FILTERORDER+1),
+ LPC_FILTERORDER+1, SUBL);
+ }
+
+ } else if (iLBCdec_inst->mode==30) {
+ /* Enhancer has 80 samples delay */
+ for (i=0; i < 2; i++) {
+ WebRtcSpl_FilterARFastQ12(
+ data+i*SUBL, data+i*SUBL,
+ iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1),
+ LPC_FILTERORDER+1, SUBL);
+ }
+ for (i=2; i < iLBCdec_inst->nsub; i++) {
+ WebRtcSpl_FilterARFastQ12(
+ data+i*SUBL, data+i*SUBL,
+ syntdenum+(i-2)*(LPC_FILTERORDER+1),
+ LPC_FILTERORDER+1, SUBL);
+ }
+ }
+
+ /* Save the filter state */
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+ } else { /* Enhancer not activated */
+ WebRtc_Word16 lag;
+
+ /* Find last lag (since the enhancer is not called to give this info) */
+ lag = 20;
+ if (iLBCdec_inst->mode==20) {
+ lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
+ &decresidual[iLBCdec_inst->blockl-60],
+ &decresidual[iLBCdec_inst->blockl-60-lag],
+ 60,
+ 80, lag, -1);
+ } else {
+ lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
+ &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
+ &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
+ ENH_BLOCKL,
+ 100, lag, -1);
+ }
+
+ /* Store lag (it is needed if next packet is lost) */
+ (*iLBCdec_inst).last_lag = (int)lag;
+
+ /* copy data and run synthesis filter */
+ WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
+
+ /* Set up the filter state */
+ WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+ for (i=0; i < iLBCdec_inst->nsub; i++) {
+ WebRtcSpl_FilterARFastQ12(
+ data+i*SUBL, data+i*SUBL,
+ syntdenum + i*(LPC_FILTERORDER+1),
+ LPC_FILTERORDER+1, SUBL);
+ }
+
+ /* Save the filter state */
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+ }
+
+ WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
+
+ /* High pass filter the signal (with upscaling a factor 2 and saturation) */
+ WebRtcIlbcfix_HpOutput(decblock, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+ iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+ iLBCdec_inst->blockl);
+
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum,
+ syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1));
+
+ iLBCdec_inst->prev_enh_pl=0;
+
+ if (mode==0) { /* PLC was used */
+ iLBCdec_inst->prev_enh_pl=1;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode.h b/trunk/src/modules/audio_coding/codecs/ilbc/decode.h
new file mode 100644
index 0000000..0252d9c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decode.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * main decoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecodeImpl(
+ WebRtc_Word16 *decblock, /* (o) decoded signal block */
+ WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ WebRtc_Word16 mode /* (i) 0: bad packet, PLC,
+ 1: normal */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c
new file mode 100644
index 0000000..4bc1cd3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -0,0 +1,189 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "state_construct.h"
+#include "cb_construct.h"
+#include "index_conv_dec.h"
+#include "do_plc.h"
+#include "constants.h"
+#include "enhancer_interface.h"
+#include "xcorr_coef.h"
+#include "lsf_check.h"
+
+
+/*----------------------------------------------------------------*
+ * frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecodeResidual(
+ iLBC_Dec_Inst_t *iLBCdec_inst,
+ /* (i/o) the decoder state structure */
+ iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
+ for the decoding */
+ WebRtc_Word16 *decresidual, /* (o) decoded residual frame */
+ WebRtc_Word16 *syntdenum /* (i) the decoded synthesis filter
+ coefficients */
+ ) {
+ WebRtc_Word16 meml_gotten, Nfor, Nback, diff, start_pos;
+ WebRtc_Word16 subcount, subframe;
+ WebRtc_Word16 *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
+ WebRtc_Word16 *memVec = iLBCdec_inst->prevResidual; /* Memory for codebook and filter state (reuse memory in state) */
+ WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN]; /* Memory for codebook */
+
+ diff = STATE_LEN - iLBCdec_inst->state_short_len;
+
+ if (iLBC_encbits->state_first == 1) {
+ start_pos = (iLBC_encbits->startIdx-1)*SUBL;
+ } else {
+ start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff;
+ }
+
+ /* decode scalar part of start state */
+
+ WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax,
+ iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)],
+ &decresidual[start_pos], iLBCdec_inst->state_short_len
+ );
+
+ if (iLBC_encbits->state_first) { /* put adaptive part in the end */
+
+ /* setup memory */
+
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCdec_inst->state_short_len));
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
+ iLBCdec_inst->state_short_len);
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(
+ &decresidual[start_pos+iLBCdec_inst->state_short_len],
+ iLBC_encbits->cb_index, iLBC_encbits->gain_index,
+ mem+CB_MEML-ST_MEM_L_TBL,
+ ST_MEM_L_TBL, (WebRtc_Word16)diff
+ );
+
+ }
+ else {/* put adaptive part in the beginning */
+
+ /* create reversed vectors for prediction */
+
+ WebRtcSpl_MemCpyReversedOrder(reverseDecresidual+diff,
+ &decresidual[(iLBC_encbits->startIdx+1)*SUBL-1-STATE_LEN], diff);
+
+ /* setup memory */
+
+ meml_gotten = iLBCdec_inst->state_short_len;
+ WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+ decresidual+start_pos, meml_gotten);
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(
+ reverseDecresidual,
+ iLBC_encbits->cb_index, iLBC_encbits->gain_index,
+ mem+CB_MEML-ST_MEM_L_TBL,
+ ST_MEM_L_TBL, diff
+ );
+
+ /* get decoded residual from reversed vector */
+
+ WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1],
+ reverseDecresidual, diff);
+ }
+
+ /* counter for predicted subframes */
+
+ subcount=1;
+
+ /* forward prediction of subframes */
+
+ Nfor = iLBCdec_inst->nsub-iLBC_encbits->startIdx-1;
+
+ if( Nfor > 0 ) {
+
+ /* setup memory */
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+ decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN);
+
+ /* loop over subframes to encode */
+
+ for (subframe=0; subframe<Nfor; subframe++) {
+
+ /* construct decoded vector */
+ WebRtcIlbcfix_CbConstruct(
+ &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL],
+ iLBC_encbits->cb_index+subcount*CB_NSTAGES,
+ iLBC_encbits->gain_index+subcount*CB_NSTAGES,
+ mem, MEM_LF_TBL, SUBL
+ );
+
+ /* update memory */
+ WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, CB_MEML-SUBL);
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+ &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL);
+
+ subcount++;
+ }
+
+ }
+
+ /* backward prediction of subframes */
+
+ Nback = iLBC_encbits->startIdx-1;
+
+ if( Nback > 0 ){
+
+ /* setup memory */
+
+ meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx);
+ if( meml_gotten > CB_MEML ) {
+ meml_gotten=CB_MEML;
+ }
+
+ WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+ decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+
+ /* loop over subframes to decode */
+
+ for (subframe=0; subframe<Nback; subframe++) {
+
+ /* construct decoded vector */
+ WebRtcIlbcfix_CbConstruct(
+ &reverseDecresidual[subframe*SUBL],
+ iLBC_encbits->cb_index+subcount*CB_NSTAGES,
+ iLBC_encbits->gain_index+subcount*CB_NSTAGES,
+ mem, MEM_LF_TBL, SUBL
+ );
+
+ /* update memory */
+ WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, CB_MEML-SUBL);
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+ &reverseDecresidual[subframe*SUBL], SUBL);
+
+ subcount++;
+ }
+
+ /* get decoded residual from reversed vector */
+ WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1,
+ reverseDecresidual, SUBL*Nback);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h
new file mode 100644
index 0000000..ea7208a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecodeResidual(
+ iLBC_Dec_Inst_t *iLBCdec_inst,
+ /* (i/o) the decoder state structure */
+ iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
+ for the decoding */
+ WebRtc_Word16 *decresidual, /* (o) decoded residual frame */
+ WebRtc_Word16 *syntdenum /* (i) the decoded synthesis filter
+ coefficients */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
new file mode 100644
index 0000000..eee3105
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.c
+
+******************************************************************/
+
+#include "lsf_interpolate_to_poly_dec.h"
+#include "bw_expand.h"
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+ WebRtc_Word16 *syntdenum, /* (o) synthesis filter coefficients */
+ WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
+ coefficients */
+ WebRtc_Word16 *lsfdeq, /* (i) dequantized lsf coefficients */
+ WebRtc_Word16 length, /* (i) length of lsf coefficient vector */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i) the decoder state structure */
+ ){
+ int i, pos, lp_length;
+ WebRtc_Word16 lp[LPC_FILTERORDER + 1], *lsfdeq2;
+
+ lsfdeq2 = lsfdeq + length;
+ lp_length = length + 1;
+
+ if (iLBCdec_inst->mode==30) {
+ /* subframe 1: Interpolation between old and first LSF */
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
+ WebRtcIlbcfix_kLsfWeight30ms[0], length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
+ WebRtcIlbcfix_BwExpand(weightdenum, lp, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+
+ /* subframes 2 to 6: interpolation between first and last LSF */
+
+ pos = lp_length;
+ for (i = 1; i < 6; i++) {
+ WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2,
+ WebRtcIlbcfix_kLsfWeight30ms[i], length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
+ WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+ pos += lp_length;
+ }
+ } else { /* iLBCdec_inst->mode=20 */
+ /* subframes 1 to 4: interpolation between old and new LSF */
+ pos = 0;
+ for (i = 0; i < iLBCdec_inst->nsub; i++) {
+ WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
+ WebRtcIlbcfix_kLsfWeight20ms[i], length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
+ WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
+ pos += lp_length;
+ }
+ }
+
+ /* update memory */
+
+ if (iLBCdec_inst->mode==30) {
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
+ } else {
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
new file mode 100644
index 0000000..3896ca9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+ WebRtc_Word16 *syntdenum, /* (o) synthesis filter coefficients */
+ WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
+ coefficients */
+ WebRtc_Word16 *lsfdeq, /* (i) dequantized lsf coefficients */
+ WebRtc_Word16 length, /* (i) length of lsf coefficient vector */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i) the decoder state structure */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/defines.h b/trunk/src/modules/audio_coding/codecs/ilbc/defines.h
new file mode 100644
index 0000000..bdeba01
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/defines.h
@@ -0,0 +1,219 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ define.h
+
+******************************************************************/
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+#include <string.h>
+
+/* general codec settings */
+
+#define FS 8000
+#define BLOCKL_20MS 160
+#define BLOCKL_30MS 240
+#define BLOCKL_MAX 240
+#define NSUB_20MS 4
+#define NSUB_30MS 6
+#define NSUB_MAX 6
+#define NASUB_20MS 2
+#define NASUB_30MS 4
+#define NASUB_MAX 4
+#define SUBL 40
+#define STATE_LEN 80
+#define STATE_SHORT_LEN_30MS 58
+#define STATE_SHORT_LEN_20MS 57
+
+/* LPC settings */
+
+#define LPC_FILTERORDER 10
+#define LPC_LOOKBACK 60
+#define LPC_N_20MS 1
+#define LPC_N_30MS 2
+#define LPC_N_MAX 2
+#define LPC_ASYMDIFF 20
+#define LSF_NSPLIT 3
+#define LSF_NUMBER_OF_STEPS 4
+#define LPC_HALFORDER 5
+#define COS_GRID_POINTS 60
+
+/* cb settings */
+
+#define CB_NSTAGES 3
+#define CB_EXPAND 2
+#define CB_MEML 147
+#define CB_FILTERLEN (2*4)
+#define CB_HALFFILTERLEN 4
+#define CB_RESRANGE 34
+#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
+#define CB_MAXGAIN_FIXQ14 21299
+
+/* enhancer */
+
+#define ENH_BLOCKL 80 /* block length */
+#define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
+#define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
+ in said second sequence */
+#define ENH_SLOP 2 /* max difference estimated and
+ correct pitch period */
+#define ENH_PLOCSL 8 /* pitch-estimates and
+ pitch-locations buffer length */
+#define ENH_OVERHANG 2
+#define ENH_UPS0 4 /* upsampling rate */
+#define ENH_FL0 3 /* 2*FLO+1 is the length of each filter */
+#define ENH_FLO_MULT2_PLUS1 7
+#define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
+#define ENH_CORRDIM (2*ENH_SLOP+1)
+#define ENH_NBLOCKS (BLOCKL/ENH_BLOCKL)
+#define ENH_NBLOCKS_EXTRA 5
+#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
+#define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
+#define ENH_BUFL_FILTEROVERHEAD 3
+#define ENH_A0 819 /* Q14 */
+#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
+#define ENH_A0DIV2 26843546 /* Q30 */
+
+/* PLC */
+
+/* Down sampling */
+
+#define FILTERORDER_DS_PLUS1 7
+#define DELAY_DS 3
+#define FACTOR_DS 2
+
+/* bit stream defs */
+
+#define NO_OF_BYTES_20MS 38
+#define NO_OF_BYTES_30MS 50
+#define NO_OF_WORDS_20MS 19
+#define NO_OF_WORDS_30MS 25
+#define STATE_BITS 3
+#define BYTE_LEN 8
+#define ULP_CLASSES 3
+
+/* help parameters */
+
+#define TWO_PI_FIX 25736 /* Q12 */
+
+/* Constants for codebook search and creation */
+
+#define ST_MEM_L_TBL 85
+#define MEM_LF_TBL 147
+
+
+/* Struct for the bits */
+typedef struct iLBC_bits_t_ {
+ WebRtc_Word16 lsf[LSF_NSPLIT*LPC_N_MAX];
+ WebRtc_Word16 cb_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB index */
+ WebRtc_Word16 gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
+ WebRtc_Word16 idxForMax;
+ WebRtc_Word16 state_first;
+ WebRtc_Word16 idxVec[STATE_SHORT_LEN_30MS];
+ WebRtc_Word16 firstbits;
+ WebRtc_Word16 startIdx;
+} iLBC_bits;
+
+/* type definition encoder instance */
+typedef struct iLBC_Enc_Inst_t_ {
+
+ /* flag for frame size mode */
+ WebRtc_Word16 mode;
+
+ /* basic parameters for different frame sizes */
+ WebRtc_Word16 blockl;
+ WebRtc_Word16 nsub;
+ WebRtc_Word16 nasub;
+ WebRtc_Word16 no_of_bytes, no_of_words;
+ WebRtc_Word16 lpc_n;
+ WebRtc_Word16 state_short_len;
+
+ /* analysis filter state */
+ WebRtc_Word16 anaMem[LPC_FILTERORDER];
+
+ /* Fix-point old lsf parameters for interpolation */
+ WebRtc_Word16 lsfold[LPC_FILTERORDER];
+ WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
+
+ /* signal buffer for LP analysis */
+ WebRtc_Word16 lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+ /* state of input HP filter */
+ WebRtc_Word16 hpimemx[2];
+ WebRtc_Word16 hpimemy[4];
+
+#ifdef SPLIT_10MS
+ WebRtc_Word16 weightdenumbuf[66];
+ WebRtc_Word16 past_samples[160];
+ WebRtc_UWord16 bytes[25];
+ WebRtc_Word16 section;
+ WebRtc_Word16 Nfor_flag;
+ WebRtc_Word16 Nback_flag;
+ WebRtc_Word16 start_pos;
+ WebRtc_Word16 diff;
+#endif
+
+} iLBC_Enc_Inst_t;
+
+/* type definition decoder instance */
+typedef struct iLBC_Dec_Inst_t_ {
+
+ /* flag for frame size mode */
+ WebRtc_Word16 mode;
+
+ /* basic parameters for different frame sizes */
+ WebRtc_Word16 blockl;
+ WebRtc_Word16 nsub;
+ WebRtc_Word16 nasub;
+ WebRtc_Word16 no_of_bytes, no_of_words;
+ WebRtc_Word16 lpc_n;
+ WebRtc_Word16 state_short_len;
+
+ /* synthesis filter state */
+ WebRtc_Word16 syntMem[LPC_FILTERORDER];
+
+ /* old LSF for interpolation */
+ WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
+
+ /* pitch lag estimated in enhancer and used in PLC */
+ int last_lag;
+
+ /* PLC state information */
+ int consPLICount, prev_enh_pl;
+ WebRtc_Word16 perSquare;
+
+ WebRtc_Word16 prevScale, prevPLI;
+ WebRtc_Word16 prevLag, prevLpc[LPC_FILTERORDER+1];
+ WebRtc_Word16 prevResidual[NSUB_MAX*SUBL];
+ WebRtc_Word16 seed;
+
+ /* previous synthesis filter parameters */
+
+ WebRtc_Word16 old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+ /* state of output HP filter */
+ WebRtc_Word16 hpimemx[2];
+ WebRtc_Word16 hpimemy[4];
+
+ /* enhancer state information */
+ int use_enhancer;
+ WebRtc_Word16 enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
+ WebRtc_Word16 enh_period[ENH_NBLOCKS_TOT];
+
+} iLBC_Dec_Inst_t;
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c
new file mode 100644
index 0000000..0dfae2b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -0,0 +1,308 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "comp_corr.h"
+#include "bw_expand.h"
+
+/*----------------------------------------------------------------*
+ * Packet loss concealment routine. Conceals a residual signal
+ * and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+ WebRtc_Word16 *PLCresidual, /* (o) concealed residual */
+ WebRtc_Word16 *PLClpc, /* (o) concealed LP parameters */
+ WebRtc_Word16 PLI, /* (i) packet loss indicator
+ 0 - no PL, 1 = PL */
+ WebRtc_Word16 *decresidual, /* (i) decoded residual */
+ WebRtc_Word16 *lpc, /* (i) decoded LPC (only used for no PL) */
+ WebRtc_Word16 inlag, /* (i) pitch lag */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) decoder instance */
+ ){
+ WebRtc_Word16 i, pick;
+ WebRtc_Word32 cross, ener, cross_comp, ener_comp = 0;
+ WebRtc_Word32 measure, maxMeasure, energy;
+ WebRtc_Word16 max, crossSquareMax, crossSquare;
+ WebRtc_Word16 j, lag, tmp1, tmp2, randlag;
+ WebRtc_Word16 shift1, shift2, shift3, shiftMax;
+ WebRtc_Word16 scale3;
+ WebRtc_Word16 corrLen;
+ WebRtc_Word32 tmpW32, tmp2W32;
+ WebRtc_Word16 use_gain;
+ WebRtc_Word16 tot_gain;
+ WebRtc_Word16 max_perSquare;
+ WebRtc_Word16 scale1, scale2;
+ WebRtc_Word16 totscale;
+ WebRtc_Word32 nom;
+ WebRtc_Word16 denom;
+ WebRtc_Word16 pitchfact;
+ WebRtc_Word16 use_lag;
+ int ind;
+ WebRtc_Word16 randvec[BLOCKL_MAX];
+
+ /* Packet Loss */
+ if (PLI == 1) {
+
+ (*iLBCdec_inst).consPLICount += 1;
+
+ /* if previous frame not lost,
+ determine pitch pred. gain */
+
+ if (iLBCdec_inst->prevPLI != 1) {
+
+ /* Maximum 60 samples are correlated, preserve as high accuracy
+ as possible without getting overflow */
+ max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (WebRtc_Word16)iLBCdec_inst->blockl);
+ scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
+ if (scale3 < 0) {
+ scale3 = 0;
+ }
+
+ /* Store scale for use when interpolating between the
+ * concealment and the received packet */
+ iLBCdec_inst->prevScale = scale3;
+
+ /* Search around the previous lag +/-3 to find the
+ best pitch period */
+ lag = inlag - 3;
+
+ /* Guard against getting outside the frame */
+ corrLen = WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
+
+ WebRtcIlbcfix_CompCorr( &cross, &ener,
+ iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
+
+ /* Normalize and store cross^2 and the number of shifts */
+ shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
+ crossSquareMax = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross, -shiftMax),
+ WEBRTC_SPL_SHIFT_W32(cross, -shiftMax), 15);
+
+ for (j=inlag-2;j<=inlag+3;j++) {
+ WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp,
+ iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3);
+
+ /* Use the criteria (corr*corr)/energy to compare if
+ this lag is better or not. To avoid the division,
+ do a cross multiplication */
+ shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
+ crossSquare = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1),
+ WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1), 15);
+
+ shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
+ measure = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_SHIFT_W32(ener, -shift2),
+ crossSquare);
+
+ shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15;
+ maxMeasure = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3),
+ crossSquareMax);
+
+ /* Calculate shift value, so that the two measures can
+ be put in the same Q domain */
+ if(((shiftMax<<1)+shift3) > ((shift1<<1)+shift2)) {
+ tmp1 = WEBRTC_SPL_MIN(31, (shiftMax<<1)+shift3-(shift1<<1)-shift2);
+ tmp2 = 0;
+ } else {
+ tmp1 = 0;
+ tmp2 = WEBRTC_SPL_MIN(31, (shift1<<1)+shift2-(shiftMax<<1)-shift3);
+ }
+
+ if ((measure>>tmp1) > (maxMeasure>>tmp2)) {
+ /* New lag is better => record lag, measure and domain */
+ lag = j;
+ crossSquareMax = crossSquare;
+ cross = cross_comp;
+ shiftMax = shift1;
+ ener = ener_comp;
+ }
+ }
+
+ /* Calculate the periodicity for the lag with the maximum correlation.
+
+ Definition of the periodicity:
+ abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2)))
+
+ Work in the Square domain to simplify the calculations
+ max_perSquare is less than 1 (in Q15)
+ */
+ tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+ &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+ corrLen, scale3);
+
+ if ((tmp2W32>0)&&(ener_comp>0)) {
+ /* norm energies to WebRtc_Word16, compute the product of the energies and
+ use the upper WebRtc_Word16 as the denominator */
+
+ scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp2W32)-16;
+ tmp1=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
+
+ scale2=(WebRtc_Word16)WebRtcSpl_NormW32(ener)-16;
+ tmp2=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, scale2);
+ denom=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp1, tmp2, 16); /* denom in Q(scale1+scale2-16) */
+
+ /* Square the cross correlation and norm it such that max_perSquare
+ will be in Q15 after the division */
+
+ totscale = scale1+scale2-1;
+ tmp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
+ tmp2 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
+
+ nom = WEBRTC_SPL_MUL_16_16(tmp1, tmp2);
+ max_perSquare = (WebRtc_Word16)WebRtcSpl_DivW32W16(nom, denom);
+
+ } else {
+ max_perSquare = 0;
+ }
+ }
+
+ /* previous frame lost, use recorded lag and gain */
+
+ else {
+ lag = iLBCdec_inst->prevLag;
+ max_perSquare = iLBCdec_inst->perSquare;
+ }
+
+ /* Attenuate signal and scale down pitch pred gain if
+ several frames lost consecutively */
+
+ use_gain = 32767; /* 1.0 in Q15 */
+
+ if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) {
+ use_gain = 29491; /* 0.9 in Q15 */
+ } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) {
+ use_gain = 22938; /* 0.7 in Q15 */
+ } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) {
+ use_gain = 16384; /* 0.5 in Q15 */
+ } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) {
+ use_gain = 0; /* 0.0 in Q15 */
+ }
+
+ /* Compute mixing factor of picth repeatition and noise:
+ for max_per>0.7 set periodicity to 1.0
+ 0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4)
+ max_per<0.4 set periodicity to 0.0
+ */
+
+ if (max_perSquare>7868) { /* periodicity > 0.7 (0.7^4=0.2401 in Q15) */
+ pitchfact = 32767;
+ } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */
+ /* find best index and interpolate from that */
+ ind = 5;
+ while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) {
+ ind--;
+ }
+ /* pitch fact is approximated by first order */
+ tmpW32 = (WebRtc_Word32)WebRtcIlbcfix_kPlcPitchFact[ind] +
+ WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kPlcPfSlope[ind], (max_perSquare-WebRtcIlbcfix_kPlcPerSqr[ind]), 11);
+
+ pitchfact = (WebRtc_Word16)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
+
+ } else { /* periodicity < 0.4 */
+ pitchfact = 0;
+ }
+
+ /* avoid repetition of same pitch cycle (buzzyness) */
+ use_lag = lag;
+ if (lag<80) {
+ use_lag = 2*lag;
+ }
+
+ /* compute concealed residual */
+ energy = 0;
+
+ for (i=0; i<iLBCdec_inst->blockl; i++) {
+
+ /* noise component - 52 < randlagFIX < 117 */
+ iLBCdec_inst->seed = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iLBCdec_inst->seed, 31821)+(WebRtc_Word32)13849);
+ randlag = 53 + (WebRtc_Word16)(iLBCdec_inst->seed & 63);
+
+ pick = i - randlag;
+
+ if (pick < 0) {
+ randvec[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
+ } else {
+ randvec[i] = iLBCdec_inst->prevResidual[pick];
+ }
+
+ /* pitch repeatition component */
+ pick = i - use_lag;
+
+ if (pick < 0) {
+ PLCresidual[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
+ } else {
+ PLCresidual[i] = PLCresidual[pick];
+ }
+
+ /* Attinuate total gain for each 10 ms */
+ if (i<80) {
+ tot_gain=use_gain;
+ } else if (i<160) {
+ tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(31130, use_gain, 15); /* 0.95*use_gain */
+ } else {
+ tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(29491, use_gain, 15); /* 0.9*use_gain */
+ }
+
+
+ /* mix noise and pitch repeatition */
+
+ PLCresidual[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tot_gain,
+ (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( (WEBRTC_SPL_MUL_16_16(pitchfact, PLCresidual[i]) +
+ WEBRTC_SPL_MUL_16_16((32767-pitchfact), randvec[i]) + 16384),
+ 15),
+ 15);
+
+ /* Shifting down the result one step extra to ensure that no overflow
+ will occur */
+ energy += WEBRTC_SPL_MUL_16_16_RSFT(PLCresidual[i],
+ PLCresidual[i], (iLBCdec_inst->prevScale+1));
+
+ }
+
+ /* less than 30 dB, use only noise */
+ if (energy < (WEBRTC_SPL_SHIFT_W32(((WebRtc_Word32)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) {
+ energy = 0;
+ for (i=0; i<iLBCdec_inst->blockl; i++) {
+ PLCresidual[i] = randvec[i];
+ }
+ }
+
+ /* use the old LPC */
+ WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1);
+
+ /* Update state in case there are multiple frame losses */
+ iLBCdec_inst->prevLag = lag;
+ iLBCdec_inst->perSquare = max_perSquare;
+ }
+
+ /* no packet loss, copy input */
+
+ else {
+ WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl);
+ WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1));
+ iLBCdec_inst->consPLICount = 0;
+ }
+
+ /* update state */
+ iLBCdec_inst->prevPLI = PLI;
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1));
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h
new file mode 100644
index 0000000..c5bcc52
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Packet loss concealment routine. Conceals a residual signal
+ * and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+ WebRtc_Word16 *PLCresidual, /* (o) concealed residual */
+ WebRtc_Word16 *PLClpc, /* (o) concealed LP parameters */
+ WebRtc_Word16 PLI, /* (i) packet loss indicator
+ 0 - no PL, 1 = PL */
+ WebRtc_Word16 *decresidual, /* (i) decoded residual */
+ WebRtc_Word16 *lpc, /* (i) decoded LPC (only used for no PL) */
+ WebRtc_Word16 inlag, /* (i) pitch lag */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) decoder instance */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt
new file mode 100644
index 0000000..d4fba08
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt
@@ -0,0 +1,10867 @@
+
+
+
+
+
+
+Network Working Group S. Andersen
+Request for Comments: 3951 Aalborg University
+Category: Experimental A. Duric
+ Telio
+ H. Astrom
+ R. Hagen
+ W. Kleijn
+ J. Linden
+ Global IP Sound
+ December 2004
+
+
+ Internet Low Bit Rate Codec (iLBC)
+
+Status of this Memo
+
+ This memo defines an Experimental Protocol for the Internet
+ community. It does not specify an Internet standard of any kind.
+ Discussion and suggestions for improvement are requested.
+ Distribution of this memo is unlimited.
+
+Copyright Notice
+
+ Copyright (C) The Internet Society (2004).
+
+Abstract
+
+ This document specifies a speech codec suitable for robust voice
+ communication over IP. The codec is developed by Global IP Sound
+ (GIPS). It is designed for narrow band speech and results in a
+ payload bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s
+ for 20 ms frames. The codec enables graceful speech quality
+ degradation in the case of lost frames, which occurs in connection
+ with lost or delayed IP packets.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 1]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+Table of Contents
+
+ 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
+ 2. Outline of the Codec . . . . . . . . . . . . . . . . . . . . . 5
+ 2.1. Encoder. . . . . . . . . . . . . . . . . . . . . . . . . 5
+ 2.2. Decoder. . . . . . . . . . . . . . . . . . . . . . . . . 7
+ 3. Encoder Principles . . . . . . . . . . . . . . . . . . . . . . 7
+ 3.1. Pre-processing . . . . . . . . . . . . . . . . . . . . . 9
+ 3.2. LPC Analysis and Quantization. . . . . . . . . . . . . . 9
+ 3.2.1. Computation of Autocorrelation Coefficients. . . 10
+ 3.2.2. Computation of LPC Coefficients. . . . . . . . . 11
+ 3.2.3. Computation of LSF Coefficients from LPC
+ Coefficients . . . . . . . . . . . . . . . . . . 11
+ 3.2.4. Quantization of LSF Coefficients . . . . . . . . 12
+ 3.2.5. Stability Check of LSF Coefficients. . . . . . . 13
+ 3.2.6. Interpolation of LSF Coefficients. . . . . . . . 13
+ 3.2.7. LPC Analysis and Quantization for 20 ms Frames . 14
+ 3.3. Calculation of the Residual. . . . . . . . . . . . . . . 15
+ 3.4. Perceptual Weighting Filter. . . . . . . . . . . . . . . 15
+ 3.5. Start State Encoder. . . . . . . . . . . . . . . . . . . 15
+ 3.5.1. Start State Estimation . . . . . . . . . . . . . 16
+ 3.5.2. All-Pass Filtering and Scale Quantization. . . . 17
+ 3.5.3. Scalar Quantization. . . . . . . . . . . . . . . 18
+ 3.6. Encoding the Remaining Samples . . . . . . . . . . . . . 19
+ 3.6.1. Codebook Memory. . . . . . . . . . . . . . . . . 20
+ 3.6.2. Perceptual Weighting of Codebook Memory
+ and Target . . . . . . . . . . . . . . . . . . . 22
+ 3.6.3. Codebook Creation. . . . . . . . . . . . . . . . 23
+ 3.6.3.1. Creation of a Base Codebook . . . . . . 23
+ 3.6.3.2. Codebook Expansion. . . . . . . . . . . 24
+ 3.6.3.3. Codebook Augmentation . . . . . . . . . 24
+ 3.6.4. Codebook Search. . . . . . . . . . . . . . . . . 26
+ 3.6.4.1. Codebook Search at Each Stage . . . . . 26
+ 3.6.4.2. Gain Quantization at Each Stage . . . . 27
+ 3.6.4.3. Preparation of Target for Next Stage. . 28
+ 3.7. Gain Correction Encoding . . . . . . . . . . . . . . . . 28
+ 3.8. Bitstream Definition . . . . . . . . . . . . . . . . . . 29
+ 4. Decoder Principles . . . . . . . . . . . . . . . . . . . . . . 32
+ 4.1. LPC Filter Reconstruction. . . . . . . . . . . . . . . . 33
+ 4.2. Start State Reconstruction . . . . . . . . . . . . . . . 33
+ 4.3. Excitation Decoding Loop . . . . . . . . . . . . . . . . 34
+ 4.4. Multistage Adaptive Codebook Decoding. . . . . . . . . . 35
+ 4.4.1. Construction of the Decoded Excitation Signal. . 35
+ 4.5. Packet Loss Concealment. . . . . . . . . . . . . . . . . 35
+ 4.5.1. Block Received Correctly and Previous Block
+ Also Received. . . . . . . . . . . . . . . . . . 35
+ 4.5.2. Block Not Received . . . . . . . . . . . . . . . 36
+
+
+
+
+Andersen, et al. Experimental [Page 2]
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+
+
+ 4.5.3. Block Received Correctly When Previous Block
+ Not Received . . . . . . . . . . . . . . . . . . 36
+ 4.6. Enhancement. . . . . . . . . . . . . . . . . . . . . . . 37
+ 4.6.1. Estimating the Pitch . . . . . . . . . . . . . . 39
+ 4.6.2. Determination of the Pitch-Synchronous
+ Sequences. . . . . . . . . . . . . . . . . . . . 39
+ 4.6.3. Calculation of the Smoothed Excitation . . . . . 41
+ 4.6.4. Enhancer Criterion . . . . . . . . . . . . . . . 41
+ 4.6.5. Enhancing the Excitation . . . . . . . . . . . . 42
+ 4.7. Synthesis Filtering. . . . . . . . . . . . . . . . . . . 43
+ 4.8. Post Filtering . . . . . . . . . . . . . . . . . . . . . 43
+ 5. Security Considerations. . . . . . . . . . . . . . . . . . . . 43
+ 6. Evaluation of the iLBC Implementations . . . . . . . . . . . . 43
+ 7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 43
+ 7.1. Normative References . . . . . . . . . . . . . . . . . . 43
+ 7.2. Informative References . . . . . . . . . . . . . . . . . 44
+ 8. ACKNOWLEDGEMENTS . . . . . . . . . . . . . . . . . . . . . . . 44
+ APPENDIX A: Reference Implementation . . . . . . . . . . . . . . . 45
+ A.1. iLBC_test.c. . . . . . . . . . . . . . . . . . . . . . . 46
+ A.2 iLBC_encode.h. . . . . . . . . . . . . . . . . . . . . . 52
+ A.3. iLBC_encode.c. . . . . . . . . . . . . . . . . . . . . . 53
+ A.4. iLBC_decode.h. . . . . . . . . . . . . . . . . . . . . . 63
+ A.5. iLBC_decode.c. . . . . . . . . . . . . . . . . . . . . . 64
+ A.6. iLBC_define.h. . . . . . . . . . . . . . . . . . . . . . 76
+ A.7. constants.h. . . . . . . . . . . . . . . . . . . . . . . 80
+ A.8. constants.c. . . . . . . . . . . . . . . . . . . . . . . 82
+ A.9. anaFilter.h. . . . . . . . . . . . . . . . . . . . . . . 96
+ A.10. anaFilter.c. . . . . . . . . . . . . . . . . . . . . . . 97
+ A.11. createCB.h . . . . . . . . . . . . . . . . . . . . . . . 98
+ A.12. createCB.c . . . . . . . . . . . . . . . . . . . . . . . 99
+ A.13. doCPLC.h . . . . . . . . . . . . . . . . . . . . . . . .104
+ A.14. doCPLC.c . . . . . . . . . . . . . . . . . . . . . . . .104
+ A.15. enhancer.h . . . . . . . . . . . . . . . . . . . . . . .109
+ A.16. enhancer.c . . . . . . . . . . . . . . . . . . . . . . .110
+ A.17. filter.h . . . . . . . . . . . . . . . . . . . . . . . .123
+ A.18. filter.c . . . . . . . . . . . . . . . . . . . . . . . .125
+ A.19. FrameClassify.h. . . . . . . . . . . . . . . . . . . . .128
+ A.20. FrameClassify.c. . . . . . . . . . . . . . . . . . . . .129
+ A.21. gainquant.h. . . . . . . . . . . . . . . . . . . . . . .131
+ A.22. gainquant.c. . . . . . . . . . . . . . . . . . . . . . .131
+ A.23. getCBvec.h . . . . . . . . . . . . . . . . . . . . . . .134
+ A.24. getCBvec.c . . . . . . . . . . . . . . . . . . . . . . .134
+ A.25. helpfun.h. . . . . . . . . . . . . . . . . . . . . . . .138
+ A.26. helpfun.c. . . . . . . . . . . . . . . . . . . . . . . .140
+ A.27. hpInput.h. . . . . . . . . . . . . . . . . . . . . . . .146
+ A.28. hpInput.c. . . . . . . . . . . . . . . . . . . . . . . .146
+ A.29. hpOutput.h . . . . . . . . . . . . . . . . . . . . . . .148
+ A.30. hpOutput.c . . . . . . . . . . . . . . . . . . . . . . .148
+
+
+
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+
+ A.31. iCBConstruct.h . . . . . . . . . . . . . . . . . . . . .149
+ A.32. iCBConstruct.c . . . . . . . . . . . . . . . . . . . . .150
+ A.33. iCBSearch.h. . . . . . . . . . . . . . . . . . . . . . .152
+ A.34. iCBSearch.c. . . . . . . . . . . . . . . . . . . . . . .153
+ A.35. LPCdecode.h. . . . . . . . . . . . . . . . . . . . . . .163
+ A.36. LPCdecode.c. . . . . . . . . . . . . . . . . . . . . . .164
+ A.37. LPCencode.h. . . . . . . . . . . . . . . . . . . . . . .167
+ A.38. LPCencode.c. . . . . . . . . . . . . . . . . . . . . . .167
+ A.39. lsf.h. . . . . . . . . . . . . . . . . . . . . . . . . .172
+ A.40. lsf.c. . . . . . . . . . . . . . . . . . . . . . . . . .172
+ A.41. packing.h. . . . . . . . . . . . . . . . . . . . . . . .178
+ A.42. packing.c. . . . . . . . . . . . . . . . . . . . . . . .179
+ A.43. StateConstructW.h. . . . . . . . . . . . . . . . . . . .182
+ A.44. StateConstructW.c. . . . . . . . . . . . . . . . . . . .183
+ A.45. StateSearchW.h . . . . . . . . . . . . . . . . . . . . .185
+ A.46. StateSearchW.c . . . . . . . . . . . . . . . . . . . . .186
+ A.47. syntFilter.h . . . . . . . . . . . . . . . . . . . . . .190
+ A.48. syntFilter.c . . . . . . . . . . . . . . . . . . . . . .190
+ Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . .192
+ Full Copyright Statement . . . . . . . . . . . . . . . . . . . . .194
+
+1. Introduction
+
+ This document contains the description of an algorithm for the coding
+ of speech signals sampled at 8 kHz. The algorithm, called iLBC, uses
+ a block-independent linear-predictive coding (LPC) algorithm and has
+ support for two basic frame lengths: 20 ms at 15.2 kbit/s and 30 ms
+ at 13.33 kbit/s. When the codec operates at block lengths of 20 ms,
+ it produces 304 bits per block, which SHOULD be packetized as in [1].
+ Similarly, for block lengths of 30 ms it produces 400 bits per block,
+ which SHOULD be packetized as in [1]. The two modes for the
+ different frame sizes operate in a very similar way. When they
+ differ it is explicitly stated in the text, usually with the notation
+ x/y, where x refers to the 20 ms mode and y refers to the 30 ms mode.
+
+ The described algorithm results in a speech coding system with a
+ controlled response to packet losses similar to what is known from
+ pulse code modulation (PCM) with packet loss concealment (PLC), such
+ as the ITU-T G.711 standard [4], which operates at a fixed bit rate
+ of 64 kbit/s. At the same time, the described algorithm enables
+ fixed bit rate coding with a quality-versus-bit rate tradeoff close
+ to state-of-the-art. A suitable RTP payload format for the iLBC
+ codec is specified in [1].
+
+ Some of the applications for which this coder is suitable are real
+ time communications such as telephony and videoconferencing,
+ streaming audio, archival, and messaging.
+
+
+
+
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+
+
+ Cable Television Laboratories (CableLabs(R)) has adopted iLBC as a
+ mandatory PacketCable(TM) audio codec standard for VoIP over Cable
+ applications [3].
+
+ This document is organized as follows. Section 2 gives a brief
+ outline of the codec. The specific encoder and decoder algorithms
+ are explained in sections 3 and 4, respectively. Appendix A provides
+ a c-code reference implementation.
+
+ The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
+ "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
+ document are to be interpreted as described in BCP 14, RFC 2119 [2].
+
+2. Outline of the Codec
+
+ The codec consists of an encoder and a decoder as described in
+ sections 2.1 and 2.2, respectively.
+
+ The essence of the codec is LPC and block-based coding of the LPC
+ residual signal. For each 160/240 (20 ms/30 ms) sample block, the
+ following major steps are performed: A set of LPC filters are
+ computed, and the speech signal is filtered through them to produce
+ the residual signal. The codec uses scalar quantization of the
+ dominant part, in terms of energy, of the residual signal for the
+ block. The dominant state is of length 57/58 (20 ms/30 ms) samples
+ and forms a start state for dynamic codebooks constructed from the
+ already coded parts of the residual signal. These dynamic codebooks
+ are used to code the remaining parts of the residual signal. By this
+ method, coding independence between blocks is achieved, resulting in
+ elimination of propagation of perceptual degradations due to packet
+ loss. The method facilitates high-quality packet loss concealment
+ (PLC).
+
+2.1. Encoder
+
+ The input to the encoder SHOULD be 16 bit uniform PCM sampled at 8
+ kHz. It SHOULD be partitioned into blocks of BLOCKL=160/240 samples
+ for the 20/30 ms frame size. Each block is divided into NSUB=4/6
+ consecutive sub-blocks of SUBL=40 samples each. For 30 ms frame
+ size, the encoder performs two LPC_FILTERORDER=10 linear-predictive
+ coding (LPC) analyses. The first analysis applies a smooth window
+ centered over the second sub-block and extending to the middle of the
+ fifth sub-block. The second LPC analysis applies a smooth asymmetric
+ window centered over the fifth sub-block and extending to the end of
+ the sixth sub-block. For 20 ms frame size, one LPC_FILTERORDER=10
+ linear-predictive coding (LPC) analysis is performed with a smooth
+ window centered over the third sub-frame.
+
+
+
+
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+
+
+ For each of the LPC analyses, a set of line-spectral frequencies
+ (LSFs) are obtained, quantized, and interpolated to obtain LSF
+ coefficients for each sub-block. Subsequently, the LPC residual is
+ computed by using the quantized and interpolated LPC analysis
+ filters.
+
+ The two consecutive sub-blocks of the residual exhibiting the maximal
+ weighted energy are identified. Within these two sub-blocks, the
+ start state (segment) is selected from two choices: the first 57/58
+ samples or the last 57/58 samples of the two consecutive sub-blocks.
+ The selected segment is the one of higher energy. The start state is
+ encoded with scalar quantization.
+
+ A dynamic codebook encoding procedure is used to encode 1) the 23/22
+ (20 ms/30 ms) remaining samples in the two sub-blocks containing the
+ start state; 2) the sub-blocks after the start state in time; and 3)
+ the sub-blocks before the start state in time. Thus, the encoding
+ target can be either the 23/22 samples remaining of the two sub-
+ blocks containing the start state or a 40-sample sub-block. This
+ target can consist of samples indexed forward in time or backward in
+ time, depending on the location of the start state.
+
+ The codebook coding is based on an adaptive codebook built from a
+ codebook memory that contains decoded LPC excitation samples from the
+ already encoded part of the block. These samples are indexed in the
+ same time direction as the target vector, ending at the sample
+ instant prior to the first sample instant represented in the target
+ vector. The codebook is used in CB_NSTAGES=3 stages in a successive
+ refinement approach, and the resulting three code vector gains are
+ encoded with 5-, 4-, and 3-bit scalar quantization, respectively.
+
+ The codebook search method employs noise shaping derived from the LPC
+ filters, and the main decision criterion is to minimize the squared
+ error between the target vector and the code vectors. Each code
+ vector in this codebook comes from one of CB_EXPAND=2 codebook
+ sections. The first section is filled with delayed, already encoded
+ residual vectors. The code vectors of the second codebook section
+ are constructed by predefined linear combinations of vectors in the
+ first section of the codebook.
+
+ As codebook encoding with squared-error matching is known to produce
+ a coded signal of less power than does the scalar quantized start
+ state signal, a gain re-scaling method is implemented by a refined
+ search for a better set of codebook gains in terms of power matching
+ after encoding. This is done by searching for a higher value of the
+ gain factor for the first stage codebook, as the subsequent stage
+ codebook gains are scaled by the first stage gain.
+
+
+
+
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+
+
+2.2. Decoder
+
+ Typically for packet communications, a jitter buffer placed at the
+ receiving end decides whether the packet containing an encoded signal
+ block has been received or lost. This logic is not part of the codec
+ described here. For each encoded signal block received the decoder
+ performs a decoding. For each lost signal block, the decoder
+ performs a PLC operation.
+
+ The decoding for each block starts by decoding and interpolating the
+ LPC coefficients. Subsequently the start state is decoded.
+
+ For codebook-encoded segments, each segment is decoded by
+ constructing the three code vectors given by the received codebook
+ indices in the same way that the code vectors were constructed in the
+ encoder. The three gain factors are also decoded and the resulting
+ decoded signal is given by the sum of the three codebook vectors
+ scaled with respective gain.
+
+ An enhancement algorithm is applied to the reconstructed excitation
+ signal. This enhancement augments the periodicity of voiced speech
+ regions. The enhancement is optimized under the constraint that the
+ modification signal (defined as the difference between the enhanced
+ excitation and the excitation signal prior to enhancement) has a
+ short-time energy that does not exceed a preset fraction of the
+ short-time energy of the excitation signal prior to enhancement.
+
+ A packet loss concealment (PLC) operation is easily embedded in the
+ decoder. The PLC operation can, e.g., be based on repeating LPC
+ filters and obtaining the LPC residual signal by using a long-term
+ prediction estimate from previous residual blocks.
+
+3. Encoder Principles
+
+ The following block diagram is an overview of all the components of
+ the iLBC encoding procedure. The description of the blocks contains
+ references to the section where that particular procedure is further
+ described.
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+ +-----------+ +---------+ +---------+
+ speech -> | 1. Pre P | -> | 2. LPC | -> | 3. Ana | ->
+ +-----------+ +---------+ +---------+
+
+ +---------------+ +--------------+
+ -> | 4. Start Sel | ->| 5. Scalar Qu | ->
+ +---------------+ +--------------+
+
+ +--------------+ +---------------+
+ -> |6. CB Search | -> | 7. Packetize | -> payload
+ | +--------------+ | +---------------+
+ ----<---------<------
+ sub-frame 0..2/4 (20 ms/30 ms)
+
+ Figure 3.1. Flow chart of the iLBC encoder
+
+ 1. Pre-process speech with a HP filter, if needed (section 3.1).
+
+ 2. Compute LPC parameters, quantize, and interpolate (section 3.2).
+
+ 3. Use analysis filters on speech to compute residual (section 3.3).
+
+ 4. Select position of 57/58-sample start state (section 3.5).
+
+ 5. Quantize the 57/58-sample start state with scalar quantization
+ (section 3.5).
+
+ 6. Search the codebook for each sub-frame. Start with 23/22 sample
+ block, then encode sub-blocks forward in time, and then encode
+ sub-blocks backward in time. For each block, the steps in Figure
+ 3.4 are performed (section 3.6).
+
+ 7. Packetize the bits into the payload specified in Table 3.2.
+
+ The input to the encoder SHOULD be 16-bit uniform PCM sampled at 8
+ kHz. Also it SHOULD be partitioned into blocks of BLOCKL=160/240
+ samples. Each block input to the encoder is divided into NSUB=4/6
+ consecutive sub-blocks of SUBL=40 samples each.
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 8]
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+
+
+ 0 39 79 119 159
+ +---------------------------------------+
+ | 1 | 2 | 3 | 4 |
+ +---------------------------------------+
+ 20 ms frame
+
+ 0 39 79 119 159 199 239
+ +-----------------------------------------------------------+
+ | 1 | 2 | 3 | 4 | 5 | 6 |
+ +-----------------------------------------------------------+
+ 30 ms frame
+ Figure 3.2. One input block to the encoder for 20 ms (with four sub-
+ frames) and 30 ms (with six sub-frames).
+
+3.1. Pre-processing
+
+ In some applications, the recorded speech signal contains DC level
+ and/or 50/60 Hz noise. If these components have not been removed
+ prior to the encoder call, they should be removed by a high-pass
+ filter. A reference implementation of this, using a filter with a
+ cutoff frequency of 90 Hz, can be found in Appendix A.28.
+
+3.2. LPC Analysis and Quantization
+
+ The input to the LPC analysis module is a possibly high-pass filtered
+ speech buffer, speech_hp, that contains 240/300 (LPC_LOOKBACK +
+ BLOCKL = 80/60 + 160/240 = 240/300) speech samples, where samples 0
+ through 79/59 are from the previous block and samples 80/60 through
+ 239/299 are from the current block. No look-ahead into the next
+ block is used. For the very first block processed, the look-back
+ samples are assumed to be zeros.
+
+ For each input block, the LPC analysis calculates one/two set(s) of
+ LPC_FILTERORDER=10 LPC filter coefficients using the autocorrelation
+ method and the Levinson-Durbin recursion. These coefficients are
+ converted to the Line Spectrum Frequency representation. In the 20
+ ms case, the single lsf set represents the spectral characteristics
+ as measured at the center of the third sub-block. For 30 ms frames,
+ the first set, lsf1, represents the spectral properties of the input
+ signal at the center of the second sub-block, and the other set,
+ lsf2, represents the spectral characteristics as measured at the
+ center of the fifth sub-block. The details of the computation for 30
+ ms frames are described in sections 3.2.1 through 3.2.6. Section
+ 3.2.7 explains how the LPC Analysis and Quantization differs for 20
+ ms frames.
+
+
+
+
+
+
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+
+
+3.2.1. Computation of Autocorrelation Coefficients
+
+ The first step in the LPC analysis procedure is to calculate
+ autocorrelation coefficients by using windowed speech samples. This
+ windowing is the only difference in the LPC analysis procedure for
+ the two sets of coefficients. For the first set, a 240-sample-long
+ standard symmetric Hanning window is applied to samples 0 through 239
+ of the input data. The first window, lpc_winTbl, is defined as
+
+ lpc_winTbl[i]= 0.5 * (1.0 - cos((2*PI*(i+1))/(BLOCKL+1)));
+ i=0,...,119
+ lpc_winTbl[i] = winTbl[BLOCKL - i - 1]; i=120,...,239
+
+ The windowed speech speech_hp_win1 is then obtained by multiplying
+ the first 240 samples of the input speech buffer with the window
+ coefficients:
+
+ speech_hp_win1[i] = speech_hp[i] * lpc_winTbl[i];
+ i=0,...,BLOCKL-1
+
+ From these 240 windowed speech samples, 11 (LPC_FILTERORDER + 1)
+ autocorrelation coefficients, acf1, are calculated:
+
+ acf1[lag] += speech_hp_win1[n] * speech_hp_win1[n + lag];
+ lag=0,...,LPC_FILTERORDER; n=0,...,BLOCKL-lag-1
+
+ In order to make the analysis more robust against numerical precision
+ problems, a spectral smoothing procedure is applied by windowing the
+ autocorrelation coefficients before the LPC coefficients are
+ computed. Also, a white noise floor is added to the autocorrelation
+ function by multiplying coefficient zero by 1.0001 (40dB below the
+ energy of the windowed speech signal). These two steps are
+ implemented by multiplying the autocorrelation coefficients with the
+ following window:
+
+ lpc_lagwinTbl[0] = 1.0001;
+ lpc_lagwinTbl[i] = exp(-0.5 * ((2 * PI * 60.0 * i) /FS)^2);
+ i=1,...,LPC_FILTERORDER
+ where FS=8000 is the sampling frequency
+
+ Then, the windowed acf function acf1_win is obtained by
+
+ acf1_win[i] = acf1[i] * lpc_lagwinTbl[i];
+ i=0,...,LPC_FILTERORDER
+
+ The second set of autocorrelation coefficients, acf2_win, are
+ obtained in a similar manner. The window, lpc_asymwinTbl, is applied
+ to samples 60 through 299, i.e., the entire current block. The
+
+
+
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+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ window consists of two segments, the first (samples 0 to 219) being
+ half a Hanning window with length 440 and the second a quarter of a
+ cycle of a cosine wave. By using this asymmetric window, an LPC
+ analysis centered in the fifth sub-block is obtained without the need
+ for any look-ahead, which would add delay. The asymmetric window is
+ defined as
+
+ lpc_asymwinTbl[i] = (sin(PI * (i + 1) / 441))^2; i=0,...,219
+
+ lpc_asymwinTbl[i] = cos((i - 220) * PI / 40); i=220,...,239
+
+ and the windowed speech is computed by
+
+ speech_hp_win2[i] = speech_hp[i + LPC_LOOKBACK] *
+ lpc_asymwinTbl[i]; i=0,....BLOCKL-1
+
+ The windowed autocorrelation coefficients are then obtained in
+ exactly the same way as for the first analysis instance.
+
+ The generation of the windows lpc_winTbl, lpc_asymwinTbl, and
+ lpc_lagwinTbl are typically done in advance, and the arrays are
+ stored in ROM rather than repeating the calculation for every block.
+
+3.2.2. Computation of LPC Coefficients
+
+ From the 2 x 11 smoothed autocorrelation coefficients, acf1_win and
+ acf2_win, the 2 x 11 LPC coefficients, lp1 and lp2, are calculated
+ in the same way for both analysis locations by using the well known
+ Levinson-Durbin recursion. The first LPC coefficient is always 1.0,
+ resulting in ten unique coefficients.
+
+ After determining the LPC coefficients, a bandwidth expansion
+ procedure is applied to smooth the spectral peaks in the
+ short-term spectrum. The bandwidth addition is obtained by the
+ following modification of the LPC coefficients:
+
+ lp1_bw[i] = lp1[i] * chirp^i; i=0,...,LPC_FILTERORDER
+ lp2_bw[i] = lp2[i] * chirp^i; i=0,...,LPC_FILTERORDER
+
+ where "chirp" is a real number between 0 and 1. It is RECOMMENDED to
+ use a value of 0.9.
+
+3.2.3. Computation of LSF Coefficients from LPC Coefficients
+
+ Thus far, two sets of LPC coefficients that represent the short-term
+ spectral characteristics of the speech signal for two different time
+ locations within the current block have been determined. These
+ coefficients SHOULD be quantized and interpolated. Before this is
+
+
+
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+
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+
+
+ done, it is advantageous to convert the LPC parameters into another
+ type of representation called Line Spectral Frequencies (LSF). The
+ LSF parameters are used because they are better suited for
+ quantization and interpolation than the regular LPC coefficients.
+ Many computationally efficient methods for calculating the LSFs from
+ the LPC coefficients have been proposed in the literature. The
+ detailed implementation of one applicable method can be found in
+ Appendix A.26. The two arrays of LSF coefficients obtained, lsf1 and
+ lsf2, are of dimension 10 (LPC_FILTERORDER).
+
+3.2.4. Quantization of LSF Coefficients
+
+ Because the LPC filters defined by the two sets of LSFs are also
+ needed in the decoder, the LSF parameters need to be quantized and
+ transmitted as side information. The total number of bits required
+ to represent the quantization of the two LSF representations for one
+ block of speech is 40, with 20 bits used for each of lsf1 and lsf2.
+
+ For computational and storage reasons, the LSF vectors are quantized
+ using three-split vector quantization (VQ). That is, the LSF vectors
+ are split into three sub-vectors that are each quantized with a
+ regular VQ. The quantized versions of lsf1 and lsf2, qlsf1 and
+ qlsf2, are obtained by using the same memoryless split VQ. The
+ length of each of these two LSF vectors is 10, and they are split
+ into three sub-vectors containing 3, 3, and 4 values, respectively.
+
+ For each of the sub-vectors, a separate codebook of quantized values
+ has been designed with a standard VQ training method for a large
+ database containing speech from a large number of speakers recorded
+ under various conditions. The size of each of the three codebooks
+ associated with the split definitions above is
+
+ int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
+
+ The actual values of the vector quantization codebook that must be
+ used can be found in the reference code of Appendix A. Both sets of
+ LSF coefficients, lsf1 and lsf2, are quantized with a standard
+ memoryless split vector quantization (VQ) structure using the squared
+ error criterion in the LSF domain. The split VQ quantization
+ consists of the following steps:
+
+ 1) Quantize the first three LSF coefficients (1 - 3) with a VQ
+ codebook of size 64.
+ 2) Quantize the next three LSF coefficients 4 - 6 with VQ a codebook
+ of size 128.
+ 3) Quantize the last four LSF coefficients (7 - 10) with a VQ
+ codebook of size 128.
+
+
+
+
+Andersen, et al. Experimental [Page 12]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ This procedure, repeated for lsf1 and lsf2, gives six quantization
+ indices and the quantized sets of LSF coefficients qlsf1 and qlsf2.
+ Each set of three indices is encoded with 6 + 7 + 7 = 20 bits. The
+ total number of bits used for LSF quantization in a block is thus 40
+ bits.
+
+3.2.5. Stability Check of LSF Coefficients
+
+ The LSF representation of the LPC filter has the convenient property
+ that the coefficients are ordered by increasing value, i.e., lsf(n-1)
+ < lsf(n), 0 < n < 10, if the corresponding synthesis filter is
+ stable. As we are employing a split VQ scheme, it is possible that
+ at the split boundaries the LSF coefficients are not ordered
+ correctly and hence that the corresponding LP filter is unstable. To
+ ensure that the filter used is stable, a stability check is performed
+ for the quantized LSF vectors. If it turns out that the coefficients
+ are not ordered appropriately (with a safety margin of 50 Hz to
+ ensure that formant peaks are not too narrow), they will be moved
+ apart. The detailed method for this can be found in Appendix A.40.
+ The same procedure is performed in the decoder. This ensures that
+ exactly the same LSF representations are used in both encoder and
+ decoder.
+
+3.2.6. Interpolation of LSF Coefficients
+
+ From the two sets of LSF coefficients that are computed for each
+ block of speech, different LSFs are obtained for each sub-block by
+ means of interpolation. This procedure is performed for the original
+ LSFs (lsf1 and lsf2), as well as the quantized versions qlsf1 and
+ qlsf2, as both versions are used in the encoder. Here follows a
+ brief summary of the interpolation scheme; the details are found in
+ the c-code of Appendix A. In the first sub-block, the average of the
+ second LSF vector from the previous block and the first LSF vector in
+ the current block is used. For sub-blocks two through five, the LSFs
+ used are obtained by linear interpolation from lsf1 (and qlsf1) to
+ lsf2 (and qlsf2), with lsf1 used in sub-block two and lsf2 in sub-
+ block five. In the last sub-block, lsf2 is used. For the very first
+ block it is assumed that the last LSF vector of the previous block is
+ equal to a predefined vector, lsfmeanTbl, obtained by calculating the
+ mean LSF vector of the LSF design database.
+
+ lsfmeanTbl[LPC_FILTERORDER] = {0.281738, 0.445801, 0.663330,
+ 0.962524, 1.251831, 1.533081, 1.850586, 2.137817,
+ 2.481445, 2.777344}
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 13]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ The interpolation method is standard linear interpolation in the LSF
+ domain. The interpolated LSF values are converted to LPC
+ coefficients for each sub-block. The unquantized and quantized LPC
+ coefficients form two sets of filters respectively. The unquantized
+ analysis filter for sub-block k is defined as follows
+
+ ___
+ \
+ Ak(z)= 1 + > ak(i)*z^(-i)
+ /__
+ i=1...LPC_FILTERORDER
+
+ The quantized analysis filter for sub-block k is defined as follows
+ ___
+ \
+ A~k(z)= 1 + > a~k(i)*z^(-i)
+ /__
+ i=1...LPC_FILTERORDER
+
+ A reference implementation of the lsf encoding is given in Appendix
+ A.38. A reference implementation of the corresponding decoding can
+ be found in Appendix A.36.
+
+3.2.7. LPC Analysis and Quantization for 20 ms Frames
+
+ As previously stated, the codec only calculates one set of LPC
+ parameters for the 20 ms frame size as opposed to two sets for 30 ms
+ frames. A single set of autocorrelation coefficients is calculated
+ on the LPC_LOOKBACK + BLOCKL = 80 + 160 = 240 samples. These samples
+ are windowed with the asymmetric window lpc_asymwinTbl, centered over
+ the third sub-frame, to form speech_hp_win. Autocorrelation
+ coefficients, acf, are calculated on the 240 samples in speech_hp_win
+ and then windowed exactly as in section 3.2.1 (resulting in
+ acf_win).
+
+ This single set of windowed autocorrelation coefficients is used to
+ calculate LPC coefficients, LSF coefficients, and quantized LSF
+ coefficients in exactly the same manner as in sections 3.2.3 through
+ 3.2.4. As for the 30 ms frame size, the ten LSF coefficients are
+ divided into three sub-vectors of size 3, 3, and 4 and quantized by
+ using the same scheme and codebook as in section 3.2.4 to finally get
+ 3 quantization indices. The quantized LSF coefficients are
+ stabilized with the algorithm described in section 3.2.5.
+
+ From the set of LSF coefficients computed for this block and those
+ from the previous block, different LSFs are obtained for each sub-
+ block by means of interpolation. The interpolation is done linearly
+ in the LSF domain over the four sub-blocks, so that the n-th sub-
+
+
+
+Andersen, et al. Experimental [Page 14]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ frame uses the weight (4-n)/4 for the LSF from old frame and the
+ weight n/4 of the LSF from the current frame. For the very first
+ block the mean LSF, lsfmeanTbl, is used as the LSF from the previous
+ block. Similarly as seen in section 3.2.6, both unquantized, A(z),
+ and quantized, A~(z), analysis filters are calculated for each of the
+ four sub-blocks.
+
+3.3. Calculation of the Residual
+
+ The block of speech samples is filtered by the quantized and
+ interpolated LPC analysis filters to yield the residual signal. In
+ particular, the corresponding LPC analysis filter for each 40 sample
+ sub-block is used to filter the speech samples for the same sub-
+ block. The filter memory at the end of each sub-block is carried
+ over to the LPC filter of the next sub-block. The signal at the
+ output of each LP analysis filter constitutes the residual signal for
+ the corresponding sub-block.
+
+ A reference implementation of the LPC analysis filters is given in
+ Appendix A.10.
+
+3.4. Perceptual Weighting Filter
+
+ In principle any good design of a perceptual weighting filter can be
+ applied in the encoder without compromising this codec definition.
+ However, it is RECOMMENDED to use the perceptual weighting filter Wk
+ for sub-block k specified below:
+
+ Wk(z)=1/Ak(z/LPC_CHIRP_WEIGHTDENUM), where
+ LPC_CHIRP_WEIGHTDENUM = 0.4222
+
+ This is a simple design with low complexity that is applied in the
+ LPC residual domain. Here Ak(z) is the filter obtained for sub-block
+ k from unquantized but interpolated LSF coefficients.
+
+3.5. Start State Encoder
+
+ The start state is quantized by using a common 6-bit scalar quantizer
+ for the block and a 3-bit scalar quantizer operating on scaled
+ samples in the weighted speech domain. In the following we describe
+ the state encoding in greater detail.
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 15]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+3.5.1. Start State Estimation
+
+ The two sub-blocks containing the start state are determined by
+ finding the two consecutive sub-blocks in the block having the
+ highest power. Advantageously, down-weighting is used in the
+ beginning and end of the sub-frames, i.e., the following measure is
+ computed (NSUB=4/6 for 20/30 ms frame size):
+
+ nsub=1,...,NSUB-1
+ ssqn[nsub] = 0.0;
+ for (i=(nsub-1)*SUBL; i<(nsub-1)*SUBL+5; i++)
+ ssqn[nsub] += sampEn_win[i-(nsub-1)*SUBL]*
+ residual[i]*residual[i];
+ for (i=(nsub-1)*SUBL+5; i<(nsub+1)*SUBL-5; i++)
+ ssqn[nsub] += residual[i]*residual[i];
+ for (i=(nsub+1)*SUBL-5; i<(nsub+1)*SUBL; i++)
+ ssqn[nsub] += sampEn_win[(nsub+1)*SUBL-i-1]*
+ residual[i]*residual[i];
+
+ where sampEn_win[5]={1/6, 2/6, 3/6, 4/6, 5/6}; MAY be used. The
+ sub-frame number corresponding to the maximum value of
+ ssqEn_win[nsub-1]*ssqn[nsub] is selected as the start state
+ indicator. A weighting of ssqEn_win[]={0.8,0.9,1.0,0.9,0.8} for 30
+ ms frames and ssqEn_win[]={0.9,1.0,0.9} for 20 ms frames; MAY
+ advantageously be used to bias the start state towards the middle of
+ the frame.
+
+ For 20 ms frames there are three possible positions for the two-sub-
+ block length maximum power segment; the start state position is
+ encoded with 2 bits. The start state position, start, MUST be
+ encoded as
+
+ start=1: start state in sub-frame 0 and 1
+ start=2: start state in sub-frame 1 and 2
+ start=3: start state in sub-frame 2 and 3
+
+ For 30 ms frames there are five possible positions of the two-sub-
+ block length maximum power segment, the start state position is
+ encoded with 3 bits. The start state position, start, MUST be
+ encoded as
+
+ start=1: start state in sub-frame 0 and 1
+ start=2: start state in sub-frame 1 and 2
+ start=3: start state in sub-frame 2 and 3
+ start=4: start state in sub-frame 3 and 4
+ start=5: start state in sub-frame 4 and 5
+
+
+
+
+
+Andersen, et al. Experimental [Page 16]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ Hence, in both cases, index 0 is not used. In order to shorten the
+ start state for bit rate efficiency, the start state is brought down
+ to STATE_SHORT_LEN=57 samples for 20 ms frames and STATE_SHORT_LEN=58
+ samples for 30 ms frames. The power of the first 23/22 and last
+ 23/22 samples of the two sub-frame blocks identified above is
+ computed as the sum of the squared signal sample values, and the
+ 23/22-sample segment with the lowest power is excluded from the start
+ state. One bit is transmitted to indicate which of the two possible
+ 57/58 sample segments is used. The start state position within the
+ two sub-frames determined above, state_first, MUST be encoded as
+
+ state_first=1: start state is first STATE_SHORT_LEN samples
+ state_first=0: start state is last STATE_SHORT_LEN samples
+
+3.5.2. All-Pass Filtering and Scale Quantization
+
+ The block of residual samples in the start state is first filtered by
+ an all-pass filter with the quantized LPC coefficients as denominator
+ and reversed quantized LPC coefficients as numerator. The purpose of
+ this phase-dispersion filter is to get a more even distribution of
+ the sample values in the residual signal. The filtering is performed
+ by circular convolution, where the initial filter memory is set to
+ zero.
+
+ res(0..(STATE_SHORT_LEN-1)) = uncoded start state residual
+ res((STATE_SHORT_LEN)..(2*STATE_SHORT_LEN-1)) = 0
+
+ Pk(z) = A~rk(z)/A~k(z), where
+ ___
+ \
+ A~rk(z)= z^(-LPC_FILTERORDER)+>a~k(i+1)*z^(i-(LPC_FILTERORDER-1))
+ /__
+ i=0...(LPC_FILTERORDER-1)
+
+ and A~k(z) is taken from the block where the start state begins
+
+ res -> Pk(z) -> filtered
+
+ ccres(k) = filtered(k) + filtered(k+STATE_SHORT_LEN),
+ k=0..(STATE_SHORT_LEN-1)
+
+ The all-pass filtered block is searched for its largest magnitude
+ sample. The 10-logarithm of this magnitude is quantized with a 6-bit
+ quantizer, state_frgqTbl, by finding the nearest representation.
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 17]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ This results in an index, idxForMax, corresponding to a quantized
+ value, qmax. The all-pass filtered residual samples in the block are
+ then multiplied with a scaling factor scal=4.5/(10^qmax) to yield
+ normalized samples.
+
+ state_frgqTbl[64] = {1.000085, 1.071695, 1.140395, 1.206868,
+ 1.277188, 1.351503, 1.429380, 1.500727, 1.569049,
+ 1.639599, 1.707071, 1.781531, 1.840799, 1.901550,
+ 1.956695, 2.006750, 2.055474, 2.102787, 2.142819,
+ 2.183592, 2.217962, 2.257177, 2.295739, 2.332967,
+ 2.369248, 2.402792, 2.435080, 2.468598, 2.503394,
+ 2.539284, 2.572944, 2.605036, 2.636331, 2.668939,
+ 2.698780, 2.729101, 2.759786, 2.789834, 2.818679,
+ 2.848074, 2.877470, 2.906899, 2.936655, 2.967804,
+ 3.000115, 3.033367, 3.066355, 3.104231, 3.141499,
+ 3.183012, 3.222952, 3.265433, 3.308441, 3.350823,
+ 3.395275, 3.442793, 3.490801, 3.542514, 3.604064,
+ 3.666050, 3.740994, 3.830749, 3.938770, 4.101764}
+
+3.5.3. Scalar Quantization
+
+ The normalized samples are quantized in the perceptually weighted
+ speech domain by a sample-by-sample scalar DPCM quantization as
+ depicted in Figure 3.3. Each sample in the block is filtered by a
+ weighting filter Wk(z), specified in section 3.4, to form a weighted
+ speech sample x[n]. The target sample d[n] is formed by subtracting
+ a predicted sample y[n], where the prediction filter is given by
+
+ Pk(z) = 1 - 1 / Wk(z).
+
+ +-------+ x[n] + d[n] +-----------+ u[n]
+ residual -->| Wk(z) |-------->(+)---->| Quantizer |------> quantized
+ +-------+ - /|\ +-----------+ | residual
+ | \|/
+ y[n] +--------------------->(+)
+ | |
+ | +------+ |
+ +--------| Pk(z)|<------+
+ +------+
+
+ Figure 3.3. Quantization of start state samples by DPCM in weighted
+ speech domain.
+
+ The coded state sample u[n] is obtained by quantizing d[n] with a 3-
+ bit quantizer with quantization table state_sq3Tbl.
+
+ state_sq3Tbl[8] = {-3.719849, -2.177490, -1.130005, -0.309692,
+ 0.444214, 1.329712, 2.436279, 3.983887}
+
+
+
+Andersen, et al. Experimental [Page 18]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ The quantized samples are transformed back to the residual domain by
+ 1) scaling with 1/scal; 2) time-reversing the scaled samples; 3)
+ filtering the time-reversed samples by the same all-pass filter, as
+ in section 3.5.2, by using circular convolution; and 4) time-
+ reversing the filtered samples. (More detail is in section 4.2.)
+
+ A reference implementation of the start-state encoding can be found
+ in Appendix A.46.
+
+3.6. Encoding the Remaining Samples
+
+ A dynamic codebook is used to encode 1) the 23/22 remaining samples
+ in the two sub-blocks containing the start state; 2) the sub-blocks
+ after the start state in time; and 3) the sub-blocks before the start
+ state in time. Thus, the encoding target can be either the 23/22
+ samples remaining of the 2 sub-blocks containing the start state, or
+ a 40-sample sub-block. This target can consist of samples that are
+ indexed forward in time or backward in time, depending on the
+ location of the start state. The length of the target is denoted by
+ lTarget.
+
+ The coding is based on an adaptive codebook that is built from a
+ codebook memory that contains decoded LPC excitation samples from the
+ already encoded part of the block. These samples are indexed in the
+ same time direction as is the target vector and end at the sample
+ instant prior to the first sample instant represented in the target
+ vector. The codebook memory has length lMem, which is equal to
+ CB_MEML=147 for the two/four 40-sample sub-blocks and 85 for the
+ 23/22-sample sub-block.
+
+ The following figure shows an overview of the encoding procedure.
+
+ +------------+ +---------------+ +-------------+
+ -> | 1. Decode | -> | 2. Mem setup | -> | 3. Perc. W. | ->
+ +------------+ +---------------+ +-------------+
+
+ +------------+ +-----------------+
+ -> | 4. Search | -> | 5. Upd. Target | ------------------>
+ | +------------+ +------------------ |
+ ----<-------------<-----------<----------
+ stage=0..2
+
+ +----------------+
+ -> | 6. Recalc G[0] | ---------------> gains and CB indices
+ +----------------+
+
+ Figure 3.4. Flow chart of the codebook search in the iLBC encoder.
+
+
+
+
+Andersen, et al. Experimental [Page 19]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ 1. Decode the part of the residual that has been encoded so far,
+ using the codebook without perceptual weighting.
+
+ 2. Set up the memory by taking data from the decoded residual. This
+ memory is used to construct codebooks. For blocks preceding the
+ start state, both the decoded residual and the target are time
+ reversed (section 3.6.1).
+ 3. Filter the memory + target with the perceptual weighting filter
+ (section 3.6.2).
+
+ 4. Search for the best match between the target and the codebook
+ vector. Compute the optimal gain for this match and quantize that
+ gain (section 3.6.4).
+
+ 5. Update the perceptually weighted target by subtracting the
+ contribution from the selected codebook vector from the
+ perceptually weighted memory (quantized gain times selected
+ vector). Repeat 4 and 5 for the two additional stages.
+
+ 6. Calculate the energy loss due to encoding of the residual. If
+ needed, compensate for this loss by an upscaling and
+ requantization of the gain for the first stage (section 3.7).
+
+ The following sections provide an in-depth description of the
+ different blocks of Figure 3.4.
+
+3.6.1. Codebook Memory
+
+ The codebook memory is based on the already encoded sub-blocks, so
+ the available data for encoding increases for each new sub-block that
+ has been encoded. Until enough sub-blocks have been encoded to fill
+ the codebook memory with data, it is padded with zeros. The
+ following figure shows an example of the order in which the sub-
+ blocks are encoded for the 30 ms frame size if the start state is
+ located in the last 58 samples of sub-block 2 and 3.
+
+ +-----------------------------------------------------+
+ | 5 | 1 |///|////////| 2 | 3 | 4 |
+ +-----------------------------------------------------+
+
+ Figure 3.5. The order from 1 to 5 in which the sub-blocks are
+ encoded. The slashed area is the start state.
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 20]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ The first target sub-block to be encoded is number 1, and the
+ corresponding codebook memory is shown in the following figure. As
+ the target vector comes before the start state in time, the codebook
+ memory and target vector are time reversed; thus, after the block has
+ been time reversed the search algorithm can be reused. As only the
+ start state has been encoded so far, the last samples of the codebook
+ memory are padded with zeros.
+
+ +-------------------------
+ |zeros|\\\\\\\\|\\\\| 1 |
+ +-------------------------
+
+ Figure 3.6. The codebook memory, length lMem=85 samples, and the
+ target vector 1, length 22 samples.
+
+ The next step is to encode sub-block 2 by using the memory that now
+ has increased since sub-block 1 has been encoded. The following
+ figure shows the codebook memory for encoding of sub-block 2.
+
+ +-----------------------------------
+ | zeros | 1 |///|////////| 2 |
+ +-----------------------------------
+
+ Figure 3.7. The codebook memory, length lMem=147 samples, and the
+ target vector 2, length 40 samples.
+
+ The next step is to encode sub-block 3 by using the memory which has
+ been increased yet again since sub-blocks 1 and 2 have been encoded,
+ but the sub-block still has to be padded with a few zeros. The
+ following figure shows the codebook memory for encoding of sub-block
+ 3.
+
+ +------------------------------------------
+ |zeros| 1 |///|////////| 2 | 3 |
+ +------------------------------------------
+
+ Figure 3.8. The codebook memory, length lMem=147 samples, and the
+ target vector 3, length 40 samples.
+
+ The next step is to encode sub-block 4 by using the memory which now
+ has increased yet again since sub-blocks 1, 2, and 3 have been
+ encoded. This time, the memory does not have to be padded with
+ zeros. The following figure shows the codebook memory for encoding
+ of sub-block 4.
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 21]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ +------------------------------------------
+ |1|///|////////| 2 | 3 | 4 |
+ +------------------------------------------
+
+ Figure 3.9. The codebook memory, length lMem=147 samples, and the
+ target vector 4, length 40 samples.
+
+ The final target sub-block to be encoded is number 5, and the
+ following figure shows the corresponding codebook memory. As the
+ target vector comes before the start state in time, the codebook
+ memory and target vector are time reversed.
+
+ +-------------------------------------------
+ | 3 | 2 |\\\\\\\\|\\\\| 1 | 5 |
+ +-------------------------------------------
+
+ Figure 3.10. The codebook memory, length lMem=147 samples, and the
+ target vector 5, length 40 samples.
+
+ For the case of 20 ms frames, the encoding procedure looks almost
+ exactly the same. The only difference is that the size of the start
+ state is 57 samples and that there are only three sub-blocks to be
+ encoded. The encoding order is the same as above, starting with the
+ 23-sample target and then encoding the two remaining 40-sample sub-
+ blocks, first going forward in time and then going backward in time
+ relative to the start state.
+
+3.6.2. Perceptual Weighting of Codebook Memory and Target
+
+ To provide a perceptual weighting of the coding error, a
+ concatenation of the codebook memory and the target to be coded is
+ all-pole filtered with the perceptual weighting filter specified in
+ section 3.4. The filter state of the weighting filter is set to
+ zero.
+
+ in(0..(lMem-1)) = unweighted codebook memory
+ in(lMem..(lMem+lTarget-1)) = unweighted target signal
+
+
+ in -> Wk(z) -> filtered,
+ where Wk(z) is taken from the sub-block of the target
+
+ weighted codebook memory = filtered(0..(lMem-1))
+ weighted target signal = filtered(lMem..(lMem+lTarget-1))
+
+ The codebook search is done with the weighted codebook memory and the
+ weighted target, whereas the decoding and the codebook memory update
+ uses the unweighted codebook memory.
+
+
+
+Andersen, et al. Experimental [Page 22]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+3.6.3. Codebook Creation
+
+ The codebook for the search is created from the perceptually weighted
+ codebook memory. It consists of two sections, where the first is
+ referred to as the base codebook and the second as the expanded
+ codebook, as it is created by linear combinations of the first. Each
+ of these two sections also has a subsection referred to as the
+ augmented codebook. The augmented codebook is only created and used
+ for the coding of the 40-sample sub-blocks and not for the 23/22-
+ sample sub-block case. The codebook size used for the different
+ sub-blocks and different stages are summarized in the table below.
+
+ Stage
+ 1 2 & 3
+ --------------------------------------------
+ 22 128 (64+0)*2 128 (64+0)*2
+ Sub- 1:st 40 256 (108+20)*2 128 (44+20)*2
+ Blocks 2:nd 40 256 (108+20)*2 256 (108+20)*2
+ 3:rd 40 256 (108+20)*2 256 (108+20)*2
+ 4:th 40 256 (108+20)*2 256 (108+20)*2
+
+ Table 3.1. Codebook sizes for the 30 ms mode.
+
+ Table 3.1 shows the codebook size for the different sub-blocks and
+ stages for 30 ms frames. Inside the parentheses it shows how the
+ number of codebook vectors is distributed, within the two sections,
+ between the base/expanded codebook and the augmented base/expanded
+ codebook. It should be interpreted in the following way:
+ (base/expanded cb + augmented base/expanded cb). The total number of
+ codebook vectors for a specific sub-block and stage is given by the
+ following formula:
+
+ Tot. cb vectors = base cb + aug. base cb + exp. cb + aug. exp. cb
+
+ The corresponding values to Figure 3.1 for 20 ms frames are only
+ slightly modified. The short sub-block is 23 instead of 22 samples,
+ and the 3:rd and 4:th sub-frame are not present.
+
+3.6.3.1. Creation of a Base Codebook
+
+ The base codebook is given by the perceptually weighted codebook
+ memory that is mentioned in section 3.5.3. The different codebook
+ vectors are given by sliding a window of length 23/22 or 40, given by
+ variable lTarget, over the lMem-long perceptually weighted codebook
+ memory. The indices are ordered so that the codebook vector
+ containing sample (lMem-lTarget-n) to (lMem-n-1) of the codebook
+
+
+
+
+
+Andersen, et al. Experimental [Page 23]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ memory vector has index n, where n=0..lMem-lTarget. Thus the total
+ number of base codebook vectors is lMem-lTarget+1, and the indices
+ are ordered from sample delay lTarget (23/22 or 40) to lMem+1 (86 or
+ 148).
+
+3.6.3.2. Codebook Expansion
+
+ The base codebook is expanded by a factor of 2, creating an
+ additional section in the codebook. This new section is obtained by
+ filtering the base codebook, base_cb, with a FIR filter with filter
+ length CB_FILTERLEN=8. The construction of the expanded codebook
+ compensates for the delay of four samples introduced by the FIR
+ filter.
+
+ cbfiltersTbl[CB_FILTERLEN]={-0.033691, 0.083740, -0.144043,
+ 0.713379, 0.806152, -0.184326,
+ 0.108887, -0.034180};
+
+ ___
+ \
+ exp_cb(k)= + > cbfiltersTbl(i)*x(k-i+4)
+ /__
+ i=0...(LPC_FILTERORDER-1)
+
+ where x(j) = base_cb(j) for j=0..lMem-1 and 0 otherwise
+
+ The individual codebook vectors of the new filtered codebook, exp_cb,
+ and their indices are obtained in the same fashion as described above
+ for the base codebook.
+
+3.6.3.3. Codebook Augmentation
+
+ For cases where encoding entire sub-blocks, i.e., cbveclen=40, the
+ base and expanded codebooks are augmented to increase codebook
+ richness. The codebooks are augmented by vectors produced by
+ interpolation of segments. The base and expanded codebook,
+ constructed above, consists of vectors corresponding to sample delays
+ in the range from cbveclen to lMem. The codebook augmentation
+ attempts to augment these codebooks with vectors corresponding to
+ sample delays from 20 to 39. However, not all of these samples are
+ present in the base codebook and expanded codebook, respectively.
+ Therefore, the augmentation vectors are constructed as linear
+ combinations between samples corresponding to sample delays in the
+ range 20 to 39. The general idea of this procedure is presented in
+ the following figures and text. The procedure is performed for both
+ the base codebook and the expanded codebook.
+
+
+
+
+
+Andersen, et al. Experimental [Page 24]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ - - ------------------------|
+ codebook memory |
+ - - ------------------------|
+ |-5-|---15---|-5-|
+ pi pp po
+
+ | | Codebook vector
+ |---15---|-5-|-----20-----| <- corresponding to
+ i ii iii sample delay 20
+
+ Figure 3.11. Generation of the first augmented codebook.
+
+ Figure 3.11 shows the codebook memory with pointers pi, pp, and po,
+ where pi points to sample 25, pp to sample 20, and po to sample 5.
+ Below the codebook memory, the augmented codebook vector
+ corresponding to sample delay 20 is drawn. Segment i consists of
+ fifteen samples from pointer pp and forward in time. Segment ii
+ consists of five interpolated samples from pi and forward and from po
+ and forward. The samples are linearly interpolated with weights
+ [0.0, 0.2, 0.4, 0.6, 0.8] for pi and weights [1.0, 0.8, 0.6, 0.4,
+ 0.2] for po. Segment iii consists of twenty samples from pp and
+ forward. The augmented codebook vector corresponding to sample delay
+ 21 is produced by moving pointers pp and pi one sample backward in
+ time. This gives us the following figure.
+
+ - - ------------------------|
+ codebook memory |
+ - - ------------------------|
+ |-5-|---16---|-5-|
+ pi pp po
+
+ | | Codebook vector
+ |---16---|-5-|-----19-----| <- corresponding to
+ i ii iii sample delay 21
+
+ Figure 3.12. Generation of the second augmented codebook.
+
+ Figure 3.12 shows the codebook memory with pointers pi, pp and po
+ where pi points to sample 26, pp to sample 21, and po to sample 5.
+ Below the codebook memory, the augmented codebook vector
+ corresponding to sample delay 21 is drawn. Segment i now consists of
+ sixteen samples from pp and forward. Segment ii consists of five
+ interpolated samples from pi and forward and from po and forward, and
+ the interpolation weights are the same throughout the procedure.
+ Segment iii consists of nineteen samples from pp and forward. The
+ same procedure of moving the two pointers is continued until the last
+ augmented vector corresponding to sample delay 39 has been created.
+ This gives a total of twenty new codebook vectors to each of the two
+
+
+
+Andersen, et al. Experimental [Page 25]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ sections. Thus the total number of codebook vectors for each of the
+ two sections, when including the augmented codebook, becomes lMem-
+ SUBL+1+SUBL/2. This is provided that augmentation is evoked, i.e.,
+ that lTarget=SUBL.
+
+3.6.4. Codebook Search
+
+ The codebook search uses the codebooks described in the sections
+ above to find the best match of the perceptually weighted target, see
+ section 3.6.2. The search method is a multi-stage gain-shape
+ matching performed as follows. At each stage the best shape vector
+ is identified, then the gain is calculated and quantized, and finally
+ the target is updated in preparation for the next codebook search
+ stage. The number of stages is CB_NSTAGES=3.
+
+ If the target is the 23/22-sample vector the codebooks are indexed so
+ that the base codebook is followed by the expanded codebook. If the
+ target is 40 samples the order is as follows: base codebook,
+ augmented base codebook, expanded codebook, and augmented expanded
+ codebook. The size of each codebook section and its corresponding
+ augmented section is given by Table 3.1 in section 3.6.3.
+
+ For example, when the second 40-sample sub-block is coded, indices 0
+ - 107 correspond to the base codebook, 108 - 127 correspond to the
+ augmented base codebook, 128 - 235 correspond to the expanded
+ codebook, and indices 236 - 255 correspond to the augmented expanded
+ codebook. The indices are divided in the same fashion for all stages
+ in the example. Only in the case of coding the first 40-sample sub-
+ block is there a difference between stages (see Table 3.1).
+
+3.6.4.1. Codebook Search at Each Stage
+
+ The codebooks are searched to find the best match to the target at
+ each stage. When the best match is found, the target is updated and
+ the next-stage search is started. The three chosen codebook vectors
+ and their corresponding gains constitute the encoded sub-block. The
+ best match is decided by the following three criteria:
+
+ 1. Compute the measure
+
+ (target*cbvec)^2 / ||cbvec||^2
+
+ for all codebook vectors, cbvec, and choose the codebook vector
+ maximizing the measure. The expression (target*cbvec) is the dot
+ product between the target vector to be coded and the codebook vector
+ for which we compute the measure. The norm, ||x||, is defined as the
+ square root of (x*x).
+
+
+
+
+Andersen, et al. Experimental [Page 26]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ 2. The absolute value of the gain, corresponding to the chosen
+ codebook vector, cbvec, must be smaller than a fixed limit,
+ CB_MAXGAIN=1.3:
+
+ |gain| < CB_MAXGAIN
+
+ where the gain is computed in the following way:
+
+ gain = (target*cbvec) / ||cbvec||^2
+
+ 3. For the first stage, the dot product of the chosen codebook vector
+ and target must be positive:
+
+ target*cbvec > 0
+
+ In practice the above criteria are used in a sequential search
+ through all codebook vectors. The best match is found by registering
+ a new max measure and index whenever the previously registered max
+ measure is surpassed and all other criteria are fulfilled. If none
+ of the codebook vectors fulfill (2) and (3), the first codebook
+ vector is selected.
+
+3.6.4.2. Gain Quantization at Each Stage
+
+ The gain follows as a result of the computation
+
+ gain = (target*cbvec) / ||cbvec||^2
+
+ for the optimal codebook vector found by the procedure in section
+ 3.6.4.1.
+
+ The three stages quantize the gain, using 5, 4, and 3 bits,
+ respectively. In the first stage, the gain is limited to positive
+ values. This gain is quantized by finding the nearest value in the
+ quantization table gain_sq5Tbl.
+
+ gain_sq5Tbl[32]={0.037476, 0.075012, 0.112488, 0.150024, 0.187500,
+ 0.224976, 0.262512, 0.299988, 0.337524, 0.375000,
+ 0.412476, 0.450012, 0.487488, 0.525024, 0.562500,
+ 0.599976, 0.637512, 0.674988, 0.712524, 0.750000,
+ 0.787476, 0.825012, 0.862488, 0.900024, 0.937500,
+ 0.974976, 1.012512, 1.049988, 1.087524, 1.125000,
+ 1.162476, 1.200012}
+
+ The gains of the subsequent two stages can be either positive or
+ negative. The gains are quantized by using a quantization table
+ times a scale factor. The second stage uses the table gain_sq4Tbl,
+ and the third stage uses gain_sq3Tbl. The scale factor equates 0.1
+
+
+
+Andersen, et al. Experimental [Page 27]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ or the absolute value of the quantized gain representation value
+ obtained in the previous stage, whichever is larger. Again, the
+ resulting gain index is the index to the nearest value of the
+ quantization table times the scale factor.
+
+ gainQ = scaleFact * gain_sqXTbl[index]
+
+ gain_sq4Tbl[16]={-1.049988, -0.900024, -0.750000, -0.599976,
+ -0.450012, -0.299988, -0.150024, 0.000000, 0.150024,
+ 0.299988, 0.450012, 0.599976, 0.750000, 0.900024,
+ 1.049988, 1.200012}
+
+ gain_sq3Tbl[8]={-1.000000, -0.659973, -0.330017,0.000000,
+ 0.250000, 0.500000, 0.750000, 1.00000}
+
+3.6.4.3. Preparation of Target for Next Stage
+
+ Before performing the search for the next stage, the perceptually
+ weighted target vector is updated by subtracting from it the selected
+ codebook vector (from the perceptually weighted codebook) times the
+ corresponding quantized gain.
+
+ target[i] = target[i] - gainQ * selected_vec[i];
+
+ A reference implementation of the codebook encoding is found in
+ Appendix A.34.
+
+3.7. Gain Correction Encoding
+
+ The start state is quantized in a relatively model independent manner
+ using 3 bits per sample. In contrast, the remaining parts of the
+ block are encoded by using an adaptive codebook. This codebook will
+ produce high matching accuracy whenever there is a high correlation
+ between the target and the best codebook vector. For unvoiced speech
+ segments and background noises, this is not necessarily so, which,
+ due to the nature of the squared error criterion, results in a coded
+ signal with less power than the target signal. As the coded start
+ state has good power matching to the target, the result is a power
+ fluctuation within the encoded frame. Perceptually, the main problem
+ with this is that the time envelope of the signal energy becomes
+ unsteady. To overcome this problem, the gains for the codebooks are
+ re-scaled after the codebook encoding by searching for a new gain
+ factor for the first stage codebook that provides better power
+ matching.
+
+ First, the energy for the target signal, tene, is computed along with
+ the energy for the coded signal, cene, given by the addition of the
+ three gain scaled codebook vectors. Because the gains of the second
+
+
+
+Andersen, et al. Experimental [Page 28]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ and third stage scale with the gain of the first stage, when the
+ first stage gain is changed from gain[0] to gain_sq5Tbl[i] the energy
+ of the coded signal changes from cene to
+
+ cene*(gain_sq5Tbl[i]*gain_sq5Tbl[i])/(gain[0]*gain[0])
+
+ where gain[0] is the gain for the first stage found in the original
+ codebook search. A refined search is performed by testing the gain
+ indices i=0 to 31, and as long as the new codebook energy as given
+ above is less than tene, the gain index for stage 1 is increased. A
+ restriction is applied so that the new gain value for stage 1 cannot
+ be more than two times higher than the original value found in the
+ codebook search. Note that by using this method we do not change the
+ shape of the encoded vector, only the gain or amplitude.
+
+3.8. Bitstream Definition
+
+ The total number of bits used to describe one frame of 20 ms speech
+ is 304, which fits in 38 bytes and results in a bit rate of 15.20
+ kbit/s. For the case of a frame length of 30 ms speech, the total
+ number of bits used is 400, which fits in 50 bytes and results in a
+ bit rate of 13.33 kbit/s. In the bitstream definition, the bits are
+ distributed into three classes according to their bit error or loss
+ sensitivity. The most sensitive bits (class 1) are placed first in
+ the bitstream for each frame. The less sensitive bits (class 2) are
+ placed after the class 1 bits. The least sensitive bits (class 3)
+ are placed at the end of the bitstream for each frame.
+
+ In the 20/30 ms frame length cases for each class, the following hold
+ true: The class 1 bits occupy a total of 6/8 bytes (48/64 bits), the
+ class 2 bits occupy 8/12 bytes (64/96 bits), and the class 3 bits
+ occupy 24/30 bytes (191/239 bits). This distribution of the bits
+ enables the use of uneven level protection (ULP) as is exploited in
+ the payload format definition for iLBC [1]. The detailed bit
+ allocation is shown in the table below. When a quantization index is
+ distributed between more classes, the more significant bits belong to
+ the lowest class.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 29]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ Bitstream structure:
+
+ ------------------------------------------------------------------+
+ Parameter | Bits Class <1,2,3> |
+ | 20 ms frame | 30 ms frame |
+ ----------------------------------+---------------+---------------+
+ Split 1 | 6 <6,0,0> | 6 <6,0,0> |
+ LSF 1 Split 2 | 7 <7,0,0> | 7 <7,0,0> |
+ LSF Split 3 | 7 <7,0,0> | 7 <7,0,0> |
+ ------------------+---------------+---------------+
+ Split 1 | NA (Not Appl.)| 6 <6,0,0> |
+ LSF 2 Split 2 | NA | 7 <7,0,0> |
+ Split 3 | NA | 7 <7,0,0> |
+ ------------------+---------------+---------------+
+ Sum | 20 <20,0,0> | 40 <40,0,0> |
+ ----------------------------------+---------------+---------------+
+ Block Class | 2 <2,0,0> | 3 <3,0,0> |
+ ----------------------------------+---------------+---------------+
+ Position 22 sample segment | 1 <1,0,0> | 1 <1,0,0> |
+ ----------------------------------+---------------+---------------+
+ Scale Factor State Coder | 6 <6,0,0> | 6 <6,0,0> |
+ ----------------------------------+---------------+---------------+
+ Sample 0 | 3 <0,1,2> | 3 <0,1,2> |
+ Quantized Sample 1 | 3 <0,1,2> | 3 <0,1,2> |
+ Residual : | : : | : : |
+ State : | : : | : : |
+ Samples : | : : | : : |
+ Sample 56 | 3 <0,1,2> | 3 <0,1,2> |
+ Sample 57 | NA | 3 <0,1,2> |
+ ------------------+---------------+---------------+
+ Sum | 171 <0,57,114>| 174 <0,58,116>|
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 7 <6,0,1> | 7 <4,2,1> |
+ CB for 22/23 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
+ sample block Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
+ ------------------+---------------+---------------+
+ Sum | 21 <6,0,15> | 21 <4,2,15> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 5 <2,0,3> | 5 <1,1,3> |
+ Gain for 22/23 Stage 2 | 4 <1,1,2> | 4 <1,1,2> |
+ sample block Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Sum | 12 <3,1,8> | 12 <2,2,8> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 8 <7,0,1> | 8 <6,1,1> |
+ sub-block 1 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
+ Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
+ ------------------+---------------+---------------+
+
+
+
+Andersen, et al. Experimental [Page 30]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ Stage 1 | 8 <0,0,8> | 8 <0,7,1> |
+ sub-block 2 Stage 2 | 8 <0,0,8> | 8 <0,0,8> |
+ Indices Stage 3 | 8 <0,0,8> | 8 <0,0,8> |
+ for CB ------------------+---------------+---------------+
+ sub-blocks Stage 1 | NA | 8 <0,7,1> |
+ sub-block 3 Stage 2 | NA | 8 <0,0,8> |
+ Stage 3 | NA | 8 <0,0,8> |
+ ------------------+---------------+---------------+
+ Stage 1 | NA | 8 <0,7,1> |
+ sub-block 4 Stage 2 | NA | 8 <0,0,8> |
+ Stage 3 | NA | 8 <0,0,8> |
+ ------------------+---------------+---------------+
+ Sum | 46 <7,0,39> | 94 <6,22,66> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 5 <1,2,2> | 5 <1,2,2> |
+ sub-block 1 Stage 2 | 4 <1,1,2> | 4 <1,2,1> |
+ Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Stage 1 | 5 <1,1,3> | 5 <0,2,3> |
+ sub-block 2 Stage 2 | 4 <0,2,2> | 4 <0,2,2> |
+ Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ Gains for ------------------+---------------+---------------+
+ sub-blocks Stage 1 | NA | 5 <0,1,4> |
+ sub-block 3 Stage 2 | NA | 4 <0,1,3> |
+ Stage 3 | NA | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Stage 1 | NA | 5 <0,1,4> |
+ sub-block 4 Stage 2 | NA | 4 <0,1,3> |
+ Stage 3 | NA | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Sum | 24 <3,6,15> | 48 <2,12,34> |
+ ----------------------------------+---------------+---------------+
+ Empty frame indicator | 1 <0,0,1> | 1 <0,0,1> |
+ -------------------------------------------------------------------
+ SUM 304 <48,64,192> 400 <64,96,240>
+
+ Table 3.2. The bitstream definition for iLBC for both the 20 ms
+ frame size mode and the 30 ms frame size mode.
+
+ When packetized into the payload, the bits MUST be sorted as follows:
+ All the class 1 bits in the order (from top to bottom) as specified
+ in the table, all the class 2 bits (from top to bottom), and all the
+ class 3 bits in the same sequential order. The last bit, the empty
+ frame indicator, SHOULD be set to zero by the encoder. If this bit
+ is set to 1 the decoder SHOULD treat the data as a lost frame. For
+ example, this bit can be set to 1 to indicate lost frame for file
+ storage format, as in [1].
+
+
+
+
+Andersen, et al. Experimental [Page 31]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4. Decoder Principles
+
+ This section describes the principles of each component of the
+ decoder algorithm.
+
+ +-------------+ +--------+ +---------------+
+ payload -> | 1. Get para | -> | 2. LPC | -> | 3. Sc Dequant | ->
+ +-------------+ +--------+ +---------------+
+
+ +-------------+ +------------------+
+ -> | 4. Mem setup| -> | 5. Construct res |------->
+ | +-------------+ +------------------- |
+ ---------<-----------<-----------<------------
+ Sub-frame 0...2/4 (20 ms/30 ms)
+
+ +----------------+ +----------+
+ -> | 6. Enhance res | -> | 7. Synth | ------------>
+ +----------------+ +----------+
+
+ +-----------------+
+ -> | 8. Post Process | ----------------> decoded speech
+ +-----------------+
+
+ Figure 4.1. Flow chart of the iLBC decoder. If a frame was lost,
+ steps 1 to 5 SHOULD be replaced by a PLC algorithm.
+
+ 1. Extract the parameters from the bitstream.
+
+ 2. Decode the LPC and interpolate (section 4.1).
+
+ 3. Construct the 57/58-sample start state (section 4.2).
+
+ 4. Set up the memory by using data from the decoded residual. This
+ memory is used for codebook construction. For blocks preceding
+ the start state, both the decoded residual and the target are time
+ reversed. Sub-frames are decoded in the same order as they were
+ encoded.
+
+ 5. Construct the residuals of this sub-frame (gain[0]*cbvec[0] +
+ gain[1]*cbvec[1] + gain[2]*cbvec[2]). Repeat 4 and 5 until the
+ residual of all sub-blocks has been constructed.
+
+ 6. Enhance the residual with the post filter (section 4.6).
+
+ 7. Synthesis of the residual (section 4.7).
+
+ 8. Post process with HP filter, if desired (section 4.8).
+
+
+
+
+Andersen, et al. Experimental [Page 32]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4.1. LPC Filter Reconstruction
+
+ The decoding of the LP filter parameters is very straightforward.
+ For a set of three/six indices, the corresponding LSF vector(s) are
+ found by simple table lookup. For each of the LSF vectors, the three
+ split vectors are concatenated to obtain qlsf1 and qlsf2,
+ respectively (in the 20 ms mode only one LSF vector, qlsf, is
+ constructed). The next step is the stability check described in
+ section 3.2.5 followed by the interpolation scheme described in
+ section 3.2.6 (3.2.7 for 20 ms frames). The only difference is that
+ only the quantized LSFs are known at the decoder, and hence the
+ unquantized LSFs are not processed.
+
+ A reference implementation of the LPC filter reconstruction is given
+ in Appendix A.36.
+
+4.2. Start State Reconstruction
+
+ The scalar encoded STATE_SHORT_LEN=58 (STATE_SHORT_LEN=57 in the 20
+ ms mode) state samples are reconstructed by 1) forming a set of
+ samples (by table lookup) from the index stream idxVec[n], 2)
+ multiplying the set with 1/scal=(10^qmax)/4.5, 3) time reversing the
+ 57/58 samples, 4) filtering the time reversed block with the
+ dispersion (all-pass) filter used in the encoder (as described in
+ section 3.5.2); this compensates for the phase distortion of the
+ earlier filter operation, and 5 reversing the 57/58 samples from the
+ previous step.
+
+ in(0..(STATE_SHORT_LEN-1)) = time reversed samples from table
+ look-up,
+ idxVecDec((STATE_SHORT_LEN-1)..0)
+
+ in(STATE_SHORT_LEN..(2*STATE_SHORT_LEN-1)) = 0
+
+ Pk(z) = A~rk(z)/A~k(z), where
+ ___
+ \
+ A~rk(z)= z^(-LPC_FILTERORDER) + > a~ki*z^(i-(LPC_FILTERORDER-1))
+ /__
+ i=0...(LPC_FILTERORDER-1)
+
+ and A~k(z) is taken from the block where the start state begins
+
+ in -> Pk(z) -> filtered
+
+ out(k) = filtered(STATE_SHORT_LEN-1-k) +
+ filtered(2*STATE_SHORT_LEN-1-k),
+ k=0..(STATE_SHORT_LEN-1)
+
+
+
+Andersen, et al. Experimental [Page 33]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ The remaining 23/22 samples in the state are reconstructed by the
+ same adaptive codebook technique described in section 4.3. The
+ location bit determines whether these are the first or the last 23/22
+ samples of the 80-sample state vector. If the remaining 23/22
+ samples are the first samples, then the scalar encoded
+ STATE_SHORT_LEN state samples are time-reversed before initialization
+ of the adaptive codebook memory vector.
+
+ A reference implementation of the start state reconstruction is given
+ in Appendix A.44.
+
+4.3. Excitation Decoding Loop
+
+ The decoding of the LPC excitation vector proceeds in the same order
+ in which the residual was encoded at the encoder. That is, after the
+ decoding of the entire 80-sample state vector, the forward sub-blocks
+ (corresponding to samples occurring after the state vector samples)
+ are decoded, and then the backward sub-blocks (corresponding to
+ samples occurring before the state vector) are decoded, resulting in
+ a fully decoded block of excitation signal samples.
+
+ In particular, each sub-block is decoded by using the multistage
+ adaptive codebook decoding module described in section 4.4. This
+ module relies upon an adaptive codebook memory constructed before
+ each run of the adaptive codebook decoding. The construction of the
+ adaptive codebook memory in the decoder is identical to the method
+ outlined in section 3.6.3, except that it is done on the codebook
+ memory without perceptual weighting.
+
+ For the initial forward sub-block, the last STATE_LEN=80 samples of
+ the length CB_LMEM=147 adaptive codebook memory are filled with the
+ samples of the state vector. For subsequent forward sub-blocks, the
+ first SUBL=40 samples of the adaptive codebook memory are discarded,
+ the remaining samples are shifted by SUBL samples toward the
+ beginning of the vector, and the newly decoded SUBL=40 samples are
+ placed at the end of the adaptive codebook memory. For backward
+ sub-blocks, the construction is similar, except that every vector of
+ samples involved is first time reversed.
+
+ A reference implementation of the excitation decoding loop is found
+ in Appendix A.5.
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 34]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4.4. Multistage Adaptive Codebook Decoding
+
+ The Multistage Adaptive Codebook Decoding module is used at both the
+ sender (encoder) and the receiver (decoder) ends to produce a
+ synthetic signal in the residual domain that is eventually used to
+ produce synthetic speech. The module takes the index values used to
+ construct vectors that are scaled and summed together to produce a
+ synthetic signal that is the output of the module.
+
+4.4.1. Construction of the Decoded Excitation Signal
+
+ The unpacked index values provided at the input to the module are
+ references to extended codebooks, which are constructed as described
+ in section 3.6.3, except that they are based on the codebook memory
+ without the perceptual weighting. The unpacked three indices are
+ used to look up three codebook vectors. The unpacked three gain
+ indices are used to decode the corresponding 3 gains. In this
+ decoding, the successive rescaling, as described in section 3.6.4.2,
+ is applied.
+
+ A reference implementation of the adaptive codebook decoding is
+ listed in Appendix A.32.
+
+4.5. Packet Loss Concealment
+
+ If packet loss occurs, the decoder receives a signal saying that
+ information regarding a block is lost. For such blocks it is
+ RECOMMENDED to use a Packet Loss Concealment (PLC) unit to create a
+ decoded signal that masks the effect of that packet loss. In the
+ following we will describe an example of a PLC unit that can be used
+ with the iLBC codec. As the PLC unit is used only at the decoder,
+ the PLC unit does not affect interoperability between
+ implementations. Other PLC implementations MAY therefore be used.
+
+ The PLC described operates on the LP filters and the excitation
+ signals and is based on the following principles:
+
+4.5.1. Block Received Correctly and Previous Block Also Received
+
+ If the block is received correctly, the PLC only records state
+ information of the current block that can be used in case the next
+ block is lost. The LP filter coefficients for each sub-block and the
+ entire decoded excitation signal are all saved in the decoder state
+ structure. All of this information will be needed if the following
+ block is lost.
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 35]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4.5.2. Block Not Received
+
+ If the block is not received, the block substitution is based on a
+ pitch-synchronous repetition of the excitation signal, which is
+ filtered by the last LP filter of the previous block. The previous
+ block's information is stored in the decoder state structure.
+
+ A correlation analysis is performed on the previous block's
+ excitation signal in order to detect the amount of pitch periodicity
+ and a pitch value. The correlation measure is also used to decide on
+ the voicing level (the degree to which the previous block's
+ excitation was a voiced or roughly periodic signal). The excitation
+ in the previous block is used to create an excitation for the block
+ to be substituted, such that the pitch of the previous block is
+ maintained. Therefore, the new excitation is constructed in a
+ pitch-synchronous manner. In order to avoid a buzzy-sounding
+ substituted block, a random excitation is mixed with the new pitch
+ periodic excitation, and the relative use of the two components is
+ computed from the correlation measure (voicing level).
+
+ For the block to be substituted, the newly constructed excitation
+ signal is then passed through the LP filter to produce the speech
+ that will be substituted for the lost block.
+
+ For several consecutive lost blocks, the packet loss concealment
+ continues in a similar manner. The correlation measure of the last
+ block received is still used along with the same pitch value. The LP
+ filters of the last block received are also used again. The energy
+ of the substituted excitation for consecutive lost blocks is
+ decreased, leading to a dampened excitation, and therefore to
+ dampened speech.
+
+4.5.3. Block Received Correctly When Previous Block Not Received
+
+ For the case in which a block is received correctly when the previous
+ block was not, the correctly received block's directly decoded speech
+ (based solely on the received block) is not used as the actual
+ output. The reason for this is that the directly decoded speech does
+ not necessarily smoothly merge into the synthetic speech generated
+ for the previous lost block. If the two signals are not smoothly
+ merged, an audible discontinuity is accidentally produced.
+ Therefore, a correlation analysis between the two blocks of
+ excitation signal (the excitation of the previous concealed block and
+ that of the current received block) is performed to find the best
+ phase match. Then a simple overlap-add procedure is performed to
+ merge the previous excitation smoothly into the current block's
+ excitation.
+
+
+
+
+Andersen, et al. Experimental [Page 36]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ The exact implementation of the packet loss concealment does not
+ influence interoperability of the codec.
+
+ A reference implementation of the packet loss concealment is
+ suggested in Appendix A.14. Exact compliance with this suggested
+ algorithm is not needed for a reference implementation to be fully
+ compatible with the overall codec specification.
+
+4.6. Enhancement
+
+ The decoder contains an enhancement unit that operates on the
+ reconstructed excitation signal. The enhancement unit increases the
+ perceptual quality of the reconstructed signal by reducing the
+ speech-correlated noise in the voiced speech segments. Compared to
+ traditional postfilters, the enhancer has an advantage in that it can
+ only modify the excitation signal slightly. This means that there is
+ no risk of over enhancement. The enhancer works very similarly for
+ both the 20 ms frame size mode and the 30 ms frame size mode.
+
+ For the mode with 20 ms frame size, the enhancer uses a memory of six
+ 80-sample excitation blocks prior in time plus the two new 80-sample
+ excitation blocks. For each block of 160 new unenhanced excitation
+ samples, 160 enhanced excitation samples are produced. The enhanced
+ excitation is 40-sample delayed compared to the unenhanced
+ excitation, as the enhancer algorithm uses lookahead.
+
+ For the mode with 30 ms frame size, the enhancer uses a memory of
+ five 80-sample excitation blocks prior in time plus the three new
+ 80-sample excitation blocks. For each block of 240 new unenhanced
+ excitation samples, 240 enhanced excitation samples are produced.
+ The enhanced excitation is 80-sample delayed compared to the
+ unenhanced excitation, as the enhancer algorithm uses lookahead.
+
+ Outline of Enhancer
+
+ The speech enhancement unit operates on sub-blocks of 80 samples,
+ which means that there are two/three 80 sample sub-blocks per frame.
+ Each of these two/three sub-blocks is enhanced separately, but in an
+ analogous manner.
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 37]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ unenhanced residual
+ |
+ | +---------------+ +--------------+
+ +-> | 1. Pitch Est | -> | 2. Find PSSQ | -------->
+ +---------------+ | +--------------+
+ +-----<-------<------<--+
+ +------------+ enh block 0..1/2 |
+ -> | 3. Smooth | |
+ +------------+ |
+ \ |
+ /\ |
+ / \ Already |
+ / 4. \----------->----------->-----------+ |
+ \Crit/ Fulfilled | |
+ \? / v |
+ \/ | |
+ \ +-----------------+ +---------+ | |
+ Not +->| 5. Use Constr. | -> | 6. Mix | ----->
+ Fulfilled +-----------------+ +---------+
+
+ ---------------> enhanced residual
+
+ Figure 4.2. Flow chart of the enhancer.
+
+ 1. Pitch estimation of each of the two/three new 80-sample blocks.
+
+ 2. Find the pitch-period-synchronous sequence n (for block k) by a
+ search around the estimated pitch value. Do this for n=1,2,3,
+ -1,-2,-3.
+
+ 3. Calculate the smoothed residual generated by the six pitch-
+ period-synchronous sequences from prior step.
+
+ 4. Check if the smoothed residual satisfies the criterion (section
+ 4.6.4).
+
+ 5. Use constraint to calculate mixing factor (section 4.6.5).
+
+ 6. Mix smoothed signal with unenhanced residual (pssq(n) n=0).
+
+ The main idea of the enhancer is to find three 80 sample blocks
+ before and three 80-sample blocks after the analyzed unenhanced sub-
+ block and to use these to improve the quality of the excitation in
+ that sub-block. The six blocks are chosen so that they have the
+ highest possible correlation with the unenhanced sub-block that is
+ being enhanced. In other words, the six blocks are pitch-period-
+ synchronous sequences to the unenhanced sub-block.
+
+
+
+
+Andersen, et al. Experimental [Page 38]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ A linear combination of the six pitch-period-synchronous sequences is
+ calculated that approximates the sub-block. If the squared error
+ between the approximation and the unenhanced sub-block is small
+ enough, the enhanced residual is set equal to this approximation.
+ For the cases when the squared error criterion is not fulfilled, a
+ linear combination of the approximation and the unenhanced residual
+ forms the enhanced residual.
+
+4.6.1. Estimating the Pitch
+
+ Pitch estimates are needed to determine the locations of the pitch-
+ period-synchronous sequences in a complexity-efficient way. For each
+ of the new two/three sub-blocks, a pitch estimate is calculated by
+ finding the maximum correlation in the range from lag 20 to lag 120.
+ These pitch estimates are used to narrow down the search for the best
+ possible pitch-period-synchronous sequences.
+
+4.6.2. Determination of the Pitch-Synchronous Sequences
+
+ Upon receiving the pitch estimates from the prior step, the enhancer
+ analyzes and enhances one 80-sample sub-block at a time. The pitch-
+ period-synchronous-sequences pssq(n) can be viewed as vectors of
+ length 80 samples each shifted n*lag samples from the current sub-
+ block. The six pitch-period-synchronous-sequences, pssq(-3) to
+ pssq(-1) and pssq(1) to pssq(3), are found one at a time by the steps
+ below:
+
+ 1) Calculate the estimate of the position of the pssq(n). For
+ pssq(n) in front of pssq(0) (n > 0), the location of the pssq(n)
+ is estimated by moving one pitch estimate forward in time from the
+ exact location of pssq(n-1). Similarly, pssq(n) behind pssq(0) (n
+ < 0) is estimated by moving one pitch estimate backward in time
+ from the exact location of pssq(n+1). If the estimated pssq(n)
+ vector location is totally within the enhancer memory (Figure
+ 4.3), steps 2, 3, and 4 are performed, otherwise the pssq(n) is
+ set to zeros.
+
+ 2) Compute the correlation between the unenhanced excitation and
+ vectors around the estimated location interval of pssq(n). The
+ correlation is calculated in the interval estimated location +/- 2
+ samples. This results in five correlation values.
+
+ 3) The five correlation values are upsampled by a factor of 4, by
+ using four simple upsampling filters (MA filters with coefficients
+ upsFilter1.. upsFilter4). Within these the maximum value is
+ found, which specifies the best pitch-period with a resolution of
+ a quarter of a sample.
+
+
+
+
+Andersen, et al. Experimental [Page 39]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ upsFilter1[7]={0.000000 0.000000 0.000000 1.000000
+ 0.000000 0.000000 0.000000}
+ upsFilter2[7]={0.015625 -0.076904 0.288330 0.862061
+ -0.106445 0.018799 -0.015625}
+ upsFilter3[7]={0.023682 -0.124268 0.601563 0.601563
+ -0.124268 0.023682 -0.023682}
+ upsFilter4[7]={0.018799 -0.106445 0.862061 0.288330
+ -0.076904 0.015625 -0.018799}
+
+ 4) Generate the pssq(n) vector by upsampling of the excitation memory
+ and extracting the sequence that corresponds to the lag delay that
+ was calculated in prior step.
+
+ With the steps above, all the pssq(n) can be found in an iterative
+ manner, first moving backward in time from pssq(0) and then forward
+ in time from pssq(0).
+
+
+ 0 159 319 479 639
+ +---------------------------------------------------------------+
+ | -5 | -4 | -3 | -2 | -1 | 0 | 1 | 2 |
+ +---------------------------------------------------------------+
+ |pssq 0 |
+ |pssq -1| |pssq 1 |
+ |pssq -2| |pssq 2 |
+ |pssq -3| |pssq 3 |
+
+ Figure 4.3. Enhancement for 20 ms frame size.
+
+ Figure 4.3 depicts pitch-period-synchronous sequences in the
+ enhancement of the first 80 sample block in the 20 ms frame size
+ mode. The unenhanced signal input is stored in the last two sub-
+ blocks (1 - 2), and the six other sub-blocks contain unenhanced
+ residual prior-in-time. We perform the enhancement algorithm on two
+ blocks of 80 samples, where the first of the two blocks consists of
+ the last 40 samples of sub-block 0 and the first 40 samples of sub-
+ block 1. The second 80-sample block consists of the last 40 samples
+ of sub-block 1 and the first 40 samples of sub-block 2.
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 40]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ 0 159 319 479 639
+ +---------------------------------------------------------------+
+ | -4 | -3 | -2 | -1 | 0 | 1 | 2 | 3 |
+ +---------------------------------------------------------------+
+ |pssq 0 |
+ |pssq -1| |pssq 1 |
+ |pssq -2| |pssq 2 |
+ |pssq -3| |pssq 3 |
+
+ Figure 4.4. Enhancement for 30 ms frame size.
+
+ Figure 4.4 depicts pitch-period-synchronous sequences in the
+ enhancement of the first 80-sample block in the 30 ms frame size
+ mode. The unenhanced signal input is stored in the last three sub-
+ blocks (1 - 3). The five other sub-blocks contain unenhanced
+ residual prior-in-time. The enhancement algorithm is performed on
+ the three 80 sample sub-blocks 0, 1, and 2.
+
+4.6.3. Calculation of the Smoothed Excitation
+
+ A linear combination of the six pssq(n) (n!=0) form a smoothed
+ approximation, z, of pssq(0). Most of the weight is put on the
+ sequences that are close to pssq(0), as these are likely to be most
+ similar to pssq(0). The smoothed vector is also rescaled so that the
+ energy of z is the same as the energy of pssq(0).
+
+ ___
+ \
+ y = > pssq(i) * pssq_weight(i)
+ /__
+ i=-3,-2,-1,1,2,3
+
+ pssq_weight(i) = 0.5*(1-cos(2*pi*(i+4)/(2*3+2)))
+
+ z = C * y, where C = ||pssq(0)||/||y||
+
+4.6.4. Enhancer Criterion
+
+ The criterion of the enhancer is that the enhanced excitation is not
+ allowed to differ much from the unenhanced excitation. This
+ criterion is checked for each 80-sample sub-block.
+
+ e < (b * ||pssq(0)||^2), where b=0.05 and (Constraint 1)
+
+ e = (pssq(0)-z)*(pssq(0)-z), and "*" means the dot product
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 41]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4.6.5. Enhancing the excitation
+
+ From the criterion in the previous section, it is clear that the
+ excitation is not allowed to change much. The purpose of this
+ constraint is to prevent the creation of an enhanced signal
+ significantly different from the original signal. This also means
+ that the constraint limits the numerical size of the errors that the
+ enhancement procedure can make. That is especially important in
+ unvoiced segments and background noise segments for which increased
+ periodicity could lead to lower perceived quality.
+
+ When the constraint in the prior section is not met, the enhanced
+ residual is instead calculated through a constrained optimization by
+ using the Lagrange multiplier technique. The new constraint is that
+
+ e = (b * ||pssq(0)||^2) (Constraint 2)
+
+ We distinguish two solution regions for the optimization: 1) the
+ region where the first constraint is fulfilled and 2) the region
+ where the first constraint is not fulfilled and the second constraint
+ must be used.
+
+ In the first case, where the second constraint is not needed, the
+ optimized re-estimated vector is simply z, the energy-scaled version
+ of y.
+
+ In the second case, where the second constraint is activated and
+ becomes an equality constraint, we have
+
+ z= A*y + B*pssq(0)
+
+ where
+
+ A = sqrt((b-b^2/4)*(w00*w00)/ (w11*w00 + w10*w10)) and
+
+ w11 = pssq(0)*pssq(0)
+ w00 = y*y
+ w10 = y*pssq(0) (* symbolizes the dot product)
+
+ and
+
+ B = 1 - b/2 - A * w10/w00
+
+ Appendix A.16 contains a listing of a reference implementation for
+ the enhancement method.
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 42]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+4.7. Synthesis Filtering
+
+ Upon decoding or PLC of the LP excitation block, the decoded speech
+ block is obtained by running the decoded LP synthesis filter,
+ 1/A~k(z), over the block. The synthesis filters have to be shifted
+ to compensate for the delay in the enhancer. For 20 ms frame size
+ mode, they SHOULD be shifted one 40-sample sub-block, and for 30 ms
+ frame size mode, they SHOULD be shifted two 40-sample sub-blocks.
+ The LP coefficients SHOULD be changed at the first sample of every
+ sub-block while keeping the filter state. For PLC blocks, one
+ solution is to apply the last LP coefficients of the last decoded
+ speech block for all sub-blocks.
+
+ The reference implementation for the synthesis filtering can be found
+ in Appendix A.48.
+
+4.8. Post Filtering
+
+ If desired, the decoded block can be filtered by a high-pass filter.
+ This removes the low frequencies of the decoded signal. A reference
+ implementation of this, with cutoff at 65 Hz, is shown in Appendix
+ A.30.
+
+5. Security Considerations
+
+ This algorithm for the coding of speech signals is not subject to any
+ known security consideration; however, its RTP payload format [1] is
+ subject to several considerations, which are addressed there.
+ Confidentiality of the media streams is achieved by encryption;
+ therefore external mechanisms, such as SRTP [5], MAY be used for that
+ purpose.
+
+6. Evaluation of the iLBC Implementations
+
+ It is possible and suggested to evaluate certain iLBC implementation
+ by utilizing methodology and tools available at
+ http://www.ilbcfreeware.org/evaluation.html
+
+7. References
+
+7.1. Normative References
+
+ [1] Duric, A. and S. Andersen, "Real-time Transport Protocol (RTP)
+ Payload Format for internet Low Bit Rate Codec (iLBC) Speech",
+ RFC 3952, December 2004.
+
+ [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
+ Levels", BCP 14, RFC 2119, March 1997.
+
+
+
+Andersen, et al. Experimental [Page 43]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ [3] PacketCable(TM) Audio/Video Codecs Specification, Cable
+ Television Laboratories, Inc.
+
+7.2. Informative References
+
+ [4] ITU-T Recommendation G.711, available online from the ITU
+ bookstore at http://www.itu.int.
+
+ [5] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norman,
+ "The Secure Real Time Transport Protocol (SRTP)", RFC 3711, March
+ 2004.
+
+8. Acknowledgements
+
+ This extensive work, besides listed authors, has the following
+ authors, who could not have been listed among "official" authors (due
+ to IESG restrictions in the number of authors who can be listed):
+
+ Manohar N. Murthi (Department of Electrical and Computer
+ Engineering, University of Miami), Fredrik Galschiodt, Julian
+ Spittka, and Jan Skoglund (Global IP Sound).
+
+ The authors are deeply indebted to the following people and thank
+ them sincerely:
+
+ Henry Sinnreich, Patrik Faltstrom, Alan Johnston, and Jean-
+ Francois Mule for great support of the iLBC initiative and for
+ valuable feedback and comments.
+
+ Peter Vary, Frank Mertz, and Christoph Erdmann (RWTH Aachen);
+ Vladimir Cuperman (Niftybox LLC); Thomas Eriksson (Chalmers Univ
+ of Tech), and Gernot Kubin (TU Graz), for thorough review of the
+ iLBC document and their valuable feedback and remarks.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 44]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+APPENDIX A. Reference Implementation
+
+ This appendix contains the complete c-code for a reference
+ implementation of encoder and decoder for the specified codec.
+
+ The c-code consists of the following files with highest-level
+ functions:
+
+ iLBC_test.c: main function for evaluation purpose
+ iLBC_encode.h: encoder header
+ iLBC_encode.c: encoder function
+ iLBC_decode.h: decoder header
+ iLBC_decode.c: decoder function
+
+ The following files contain global defines and constants:
+
+ iLBC_define.h: global defines
+ constants.h: global constants header
+ constants.c: global constants memory allocations
+
+ The following files contain subroutines:
+
+ anaFilter.h: lpc analysis filter header
+ anaFilter.c: lpc analysis filter function
+ createCB.h: codebook construction header
+ createCB.c: codebook construction function
+ doCPLC.h: packet loss concealment header
+ doCPLC.c: packet loss concealment function
+ enhancer.h: signal enhancement header
+ enhancer.c: signal enhancement function
+ filter.h: general filter header
+ filter.c: general filter functions
+ FrameClassify.h: start state classification header
+ FrameClassify.c: start state classification function
+ gainquant.h: gain quantization header
+ gainquant.c: gain quantization function
+ getCBvec.h: codebook vector construction header
+ getCBvec.c: codebook vector construction function
+ helpfun.h: general purpose header
+ helpfun.c: general purpose functions
+ hpInput.h: input high pass filter header
+ hpInput.c: input high pass filter function
+ hpOutput.h: output high pass filter header
+ hpOutput.c: output high pass filter function
+ iCBConstruct.h: excitation decoding header
+ iCBConstruct.c: excitation decoding function
+ iCBSearch.h: excitation encoding header
+ iCBSearch.c: excitation encoding function
+
+
+
+Andersen, et al. Experimental [Page 45]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ LPCdecode.h: lpc decoding header
+ LPCdecode.c: lpc decoding function
+ LPCencode.h: lpc encoding header
+ LPCencode.c: lpc encoding function
+ lsf.h: line spectral frequencies header
+ lsf.c: line spectral frequencies functions
+ packing.h: bitstream packetization header
+ packing.c: bitstream packetization functions
+ StateConstructW.h: state decoding header
+ StateConstructW.c: state decoding functions
+ StateSearchW.h: state encoding header
+ StateSearchW.c: state encoding function
+ syntFilter.h: lpc synthesis filter header
+ syntFilter.c: lpc synthesis filter function
+
+ The implementation is portable and should work on many different
+ platforms. However, it is not difficult to optimize the
+ implementation on particular platforms, an exercise left to the
+ reader.
+
+A.1. iLBC_test.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_test.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <stdlib.h>
+ #include <stdio.h>
+ #include <string.h>
+ #include "iLBC_define.h"
+ #include "iLBC_encode.h"
+ #include "iLBC_decode.h"
+
+ /* Runtime statistics */
+ #include <time.h>
+
+ #define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS/2)
+
+ /*----------------------------------------------------------------*
+ * Encoder interface function
+
+
+
+Andersen, et al. Experimental [Page 46]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ *---------------------------------------------------------------*/
+
+ short encode( /* (o) Number of bytes encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) Encoder instance */
+ short *encoded_data, /* (o) The encoded bytes */
+ short *data /* (i) The signal block to encode*/
+ ){
+ float block[BLOCKL_MAX];
+ int k;
+
+ /* convert signal to float */
+
+ for (k=0; k<iLBCenc_inst->blockl; k++)
+ block[k] = (float)data[k];
+
+ /* do the actual encoding */
+
+ iLBC_encode((unsigned char *)encoded_data, block, iLBCenc_inst);
+
+
+ return (iLBCenc_inst->no_of_bytes);
+ }
+
+ /*----------------------------------------------------------------*
+ * Decoder interface function
+ *---------------------------------------------------------------*/
+
+ short decode( /* (o) Number of decoded samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ short *decoded_data, /* (o) Decoded signal block*/
+ short *encoded_data, /* (i) Encoded bytes */
+ short mode /* (i) 0=PL, 1=Normal */
+ ){
+ int k;
+ float decblock[BLOCKL_MAX], dtmp;
+
+ /* check if mode is valid */
+
+ if (mode<0 || mode>1) {
+ printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
+
+ /* do actual decoding of block */
+
+ iLBC_decode(decblock, (unsigned char *)encoded_data,
+ iLBCdec_inst, mode);
+
+ /* convert to short */
+
+
+
+Andersen, et al. Experimental [Page 47]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (k=0; k<iLBCdec_inst->blockl; k++){
+ dtmp=decblock[k];
+
+ if (dtmp<MIN_SAMPLE)
+ dtmp=MIN_SAMPLE;
+ else if (dtmp>MAX_SAMPLE)
+ dtmp=MAX_SAMPLE;
+ decoded_data[k] = (short) dtmp;
+ }
+
+ return (iLBCdec_inst->blockl);
+ }
+
+ /*---------------------------------------------------------------*
+ * Main program to test iLBC encoding and decoding
+ *
+ * Usage:
+ * exefile_name.exe <infile> <bytefile> <outfile> <channel>
+ *
+ * <infile> : Input file, speech for encoder (16-bit pcm file)
+ * <bytefile> : Bit stream output from the encoder
+ * <outfile> : Output file, decoded speech (16-bit pcm file)
+ * <channel> : Bit error file, optional (16-bit)
+ * 1 - Packet received correctly
+ * 0 - Packet Lost
+ *
+ *--------------------------------------------------------------*/
+
+ int main(int argc, char* argv[])
+ {
+
+ /* Runtime statistics */
+
+ float starttime;
+ float runtime;
+ float outtime;
+
+ FILE *ifileid,*efileid,*ofileid, *cfileid;
+ short data[BLOCKL_MAX];
+ short encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
+ int len;
+ short pli, mode;
+ int blockcount = 0;
+ int packetlosscount = 0;
+
+ /* Create structs */
+ iLBC_Enc_Inst_t Enc_Inst;
+ iLBC_Dec_Inst_t Dec_Inst;
+
+
+
+Andersen, et al. Experimental [Page 48]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* get arguments and open files */
+
+ if ((argc!=5) && (argc!=6)) {
+ fprintf(stderr,
+ "\n*-----------------------------------------------*\n");
+ fprintf(stderr,
+ " %s <20,30> input encoded decoded (channel)\n\n",
+ argv[0]);
+ fprintf(stderr,
+ " mode : Frame size for the encoding/decoding\n");
+ fprintf(stderr,
+ " 20 - 20 ms\n");
+ fprintf(stderr,
+ " 30 - 30 ms\n");
+ fprintf(stderr,
+ " input : Speech for encoder (16-bit pcm file)\n");
+ fprintf(stderr,
+ " encoded : Encoded bit stream\n");
+ fprintf(stderr,
+ " decoded : Decoded speech (16-bit pcm file)\n");
+ fprintf(stderr,
+ " channel : Packet loss pattern, optional (16-bit)\n");
+ fprintf(stderr,
+ " 1 - Packet received correctly\n");
+ fprintf(stderr,
+ " 0 - Packet Lost\n");
+ fprintf(stderr,
+ "*-----------------------------------------------*\n\n");
+ exit(1);
+ }
+ mode=atoi(argv[1]);
+ if (mode != 20 && mode != 30) {
+ fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
+ argv[1]);
+ exit(2);
+ }
+ if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+ exit(2);}
+ if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open encoded file %s\n",
+ argv[3]); exit(1);}
+ if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open decoded file %s\n",
+ argv[4]); exit(1);}
+ if (argc==6) {
+ if( (cfileid=fopen(argv[5],"rb")) == NULL) {
+ fprintf(stderr, "Cannot open channel file %s\n",
+
+
+
+Andersen, et al. Experimental [Page 49]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ argv[5]);
+ exit(1);
+ }
+ } else {
+ cfileid=NULL;
+ }
+
+ /* print info */
+
+ fprintf(stderr, "\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* iLBC test program *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,"\nMode : %2d ms\n", mode);
+ fprintf(stderr,"Input file : %s\n", argv[2]);
+ fprintf(stderr,"Encoded file : %s\n", argv[3]);
+ fprintf(stderr,"Output file : %s\n", argv[4]);
+ if (argc==6) {
+ fprintf(stderr,"Channel file : %s\n", argv[5]);
+ }
+ fprintf(stderr,"\n");
+
+ /* Initialization */
+
+ initEncode(&Enc_Inst, mode);
+ initDecode(&Dec_Inst, mode, 1);
+
+ /* Runtime statistics */
+
+ starttime=clock()/(float)CLOCKS_PER_SEC;
+
+ /* loop over input blocks */
+
+ while (fread(data,sizeof(short),Enc_Inst.blockl,ifileid)==
+ Enc_Inst.blockl) {
+
+ blockcount++;
+
+ /* encoding */
+
+
+
+Andersen, et al. Experimental [Page 50]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+ len=encode(&Enc_Inst, encoded_data, data);
+ fprintf(stderr, "\r");
+
+ /* write byte file */
+
+ fwrite(encoded_data, sizeof(unsigned char), len, efileid);
+
+ /* get channel data if provided */
+ if (argc==6) {
+ if (fread(&pli, sizeof(short), 1, cfileid)) {
+ if ((pli!=0)&&(pli!=1)) {
+ fprintf(stderr, "Error in channel file\n");
+ exit(0);
+ }
+ if (pli==0) {
+ /* Packet loss -> remove info from frame */
+ memset(encoded_data, 0,
+ sizeof(short)*ILBCNOOFWORDS_MAX);
+ packetlosscount++;
+ }
+ } else {
+ fprintf(stderr, "Error. Channel file too short\n");
+ exit(0);
+ }
+ } else {
+ pli=1;
+ }
+
+ /* decoding */
+
+ fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+
+ len=decode(&Dec_Inst, decoded_data, encoded_data, pli);
+ fprintf(stderr, "\r");
+
+ /* write output file */
+
+ fwrite(decoded_data,sizeof(short),len,ofileid);
+ }
+
+ /* Runtime statistics */
+
+ runtime = (float)(clock()/(float)CLOCKS_PER_SEC-starttime);
+ outtime = (float)((float)blockcount*(float)mode/1000.0);
+ printf("\n\nLength of speech file: %.1f s\n", outtime);
+ printf("Packet loss : %.1f%%\n",
+ 100.0*(float)packetlosscount/(float)blockcount);
+
+
+
+Andersen, et al. Experimental [Page 51]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ printf("Time to run iLBC :");
+ printf(" %.1f s (%.1f %% of realtime)\n\n", runtime,
+ (100*runtime/outtime));
+
+ /* close files */
+
+ fclose(ifileid); fclose(efileid); fclose(ofileid);
+ if (argc==6) {
+ fclose(cfileid);
+ }
+ return(0);
+ }
+
+A.2. iLBC_encode.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_encode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ILBCENCODE_H
+ #define __iLBC_ILBCENCODE_H
+
+ #include "iLBC_define.h"
+
+ short initEncode( /* (o) Number of bytes
+ encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ int mode /* (i) frame size mode */
+ );
+
+ void iLBC_encode(
+
+ unsigned char *bytes, /* (o) encoded data bits iLBC */
+ float *block, /* (o) speech vector to
+ encode */
+ iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+ state */
+ );
+
+ #endif
+
+
+
+
+Andersen, et al. Experimental [Page 52]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.3. iLBC_encode.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_encode.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <stdlib.h>
+ #include <string.h>
+
+ #include "iLBC_define.h"
+ #include "LPCencode.h"
+ #include "FrameClassify.h"
+ #include "StateSearchW.h"
+ #include "StateConstructW.h"
+ #include "helpfun.h"
+ #include "constants.h"
+ #include "packing.h"
+ #include "iCBSearch.h"
+ #include "iCBConstruct.h"
+ #include "hpInput.h"
+ #include "anaFilter.h"
+ #include "syntFilter.h"
+
+ /*----------------------------------------------------------------*
+ * Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+ short initEncode( /* (o) Number of bytes
+ encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ int mode /* (i) frame size mode */
+ ){
+ iLBCenc_inst->mode = mode;
+ if (mode==30) {
+ iLBCenc_inst->blockl = BLOCKL_30MS;
+ iLBCenc_inst->nsub = NSUB_30MS;
+ iLBCenc_inst->nasub = NASUB_30MS;
+ iLBCenc_inst->lpc_n = LPC_N_30MS;
+ iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
+ iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
+
+
+
+Andersen, et al. Experimental [Page 53]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
+ /* ULP init */
+ iLBCenc_inst->ULP_inst=&ULP_30msTbl;
+ }
+ else if (mode==20) {
+ iLBCenc_inst->blockl = BLOCKL_20MS;
+ iLBCenc_inst->nsub = NSUB_20MS;
+ iLBCenc_inst->nasub = NASUB_20MS;
+ iLBCenc_inst->lpc_n = LPC_N_20MS;
+ iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
+ iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
+ iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
+ /* ULP init */
+ iLBCenc_inst->ULP_inst=&ULP_20msTbl;
+ }
+ else {
+ exit(2);
+ }
+
+ memset((*iLBCenc_inst).anaMem, 0,
+ LPC_FILTERORDER*sizeof(float));
+ memcpy((*iLBCenc_inst).lsfold, lsfmeanTbl,
+ LPC_FILTERORDER*sizeof(float));
+ memcpy((*iLBCenc_inst).lsfdeqold, lsfmeanTbl,
+ LPC_FILTERORDER*sizeof(float));
+ memset((*iLBCenc_inst).lpc_buffer, 0,
+ (LPC_LOOKBACK+BLOCKL_MAX)*sizeof(float));
+ memset((*iLBCenc_inst).hpimem, 0, 4*sizeof(float));
+
+ return (iLBCenc_inst->no_of_bytes);
+ }
+
+ /*----------------------------------------------------------------*
+ * main encoder function
+ *---------------------------------------------------------------*/
+
+ void iLBC_encode(
+ unsigned char *bytes, /* (o) encoded data bits iLBC */
+ float *block, /* (o) speech vector to
+ encode */
+ iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+ state */
+ ){
+
+ float data[BLOCKL_MAX];
+ float residual[BLOCKL_MAX], reverseResidual[BLOCKL_MAX];
+
+ int start, idxForMax, idxVec[STATE_LEN];
+
+
+
+Andersen, et al. Experimental [Page 54]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
+ int n, k, meml_gotten, Nfor, Nback, i, pos;
+ int gain_index[CB_NSTAGES*NASUB_MAX],
+ extra_gain_index[CB_NSTAGES];
+ int cb_index[CB_NSTAGES*NASUB_MAX],extra_cb_index[CB_NSTAGES];
+ int lsf_i[LSF_NSPLIT*LPC_N_MAX];
+ unsigned char *pbytes;
+ int diff, start_pos, state_first;
+ float en1, en2;
+ int index, ulp, firstpart;
+ int subcount, subframe;
+ float weightState[LPC_FILTERORDER];
+ float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+ float weightdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+ float decresidual[BLOCKL_MAX];
+
+ /* high pass filtering of input signal if such is not done
+ prior to calling this function */
+
+ hpInput(block, iLBCenc_inst->blockl,
+ data, (*iLBCenc_inst).hpimem);
+
+ /* otherwise simply copy */
+
+ /*memcpy(data,block,iLBCenc_inst->blockl*sizeof(float));*/
+
+ /* LPC of hp filtered input data */
+
+ LPCencode(syntdenum, weightdenum, lsf_i, data, iLBCenc_inst);
+
+
+ /* inverse filter to get residual */
+
+ for (n=0; n<iLBCenc_inst->nsub; n++) {
+ anaFilter(&data[n*SUBL], &syntdenum[n*(LPC_FILTERORDER+1)],
+ SUBL, &residual[n*SUBL], iLBCenc_inst->anaMem);
+ }
+
+ /* find state location */
+
+ start = FrameClassify(iLBCenc_inst, residual);
+
+ /* check if state should be in first or last part of the
+ two subframes */
+
+ diff = STATE_LEN - iLBCenc_inst->state_short_len;
+ en1 = 0;
+ index = (start-1)*SUBL;
+
+
+
+Andersen, et al. Experimental [Page 55]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
+ en1 += residual[index+i]*residual[index+i];
+ }
+ en2 = 0;
+ index = (start-1)*SUBL+diff;
+ for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
+ en2 += residual[index+i]*residual[index+i];
+ }
+
+
+ if (en1 > en2) {
+ state_first = 1;
+ start_pos = (start-1)*SUBL;
+ } else {
+ state_first = 0;
+ start_pos = (start-1)*SUBL + diff;
+ }
+
+ /* scalar quantization of state */
+
+ StateSearchW(iLBCenc_inst, &residual[start_pos],
+ &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
+ &weightdenum[(start-1)*(LPC_FILTERORDER+1)], &idxForMax,
+ idxVec, iLBCenc_inst->state_short_len, state_first);
+
+ StateConstructW(idxForMax, idxVec,
+ &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
+ &decresidual[start_pos], iLBCenc_inst->state_short_len);
+
+ /* predictive quantization in state */
+
+ if (state_first) { /* put adaptive part in the end */
+
+ /* setup memory */
+
+ memset(mem, 0,
+ (CB_MEML-iLBCenc_inst->state_short_len)*sizeof(float));
+ memcpy(mem+CB_MEML-iLBCenc_inst->state_short_len,
+ decresidual+start_pos,
+ iLBCenc_inst->state_short_len*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ /* encode sub-frames */
+
+ iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
+ &residual[start_pos+iLBCenc_inst->state_short_len],
+ mem+CB_MEML-stMemLTbl,
+ stMemLTbl, diff, CB_NSTAGES,
+
+
+
+Andersen, et al. Experimental [Page 56]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ &weightdenum[start*(LPC_FILTERORDER+1)],
+ weightState, 0);
+
+ /* construct decoded vector */
+
+ iCBConstruct(
+ &decresidual[start_pos+iLBCenc_inst->state_short_len],
+ extra_cb_index, extra_gain_index,
+ mem+CB_MEML-stMemLTbl,
+ stMemLTbl, diff, CB_NSTAGES);
+
+ }
+ else { /* put adaptive part in the beginning */
+
+ /* create reversed vectors for prediction */
+
+ for (k=0; k<diff; k++) {
+ reverseResidual[k] = residual[(start+1)*SUBL-1
+ -(k+iLBCenc_inst->state_short_len)];
+ }
+
+ /* setup memory */
+
+ meml_gotten = iLBCenc_inst->state_short_len;
+ for (k=0; k<meml_gotten; k++) {
+ mem[CB_MEML-1-k] = decresidual[start_pos + k];
+ }
+ memset(mem, 0, (CB_MEML-k)*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ /* encode sub-frames */
+
+ iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
+ reverseResidual, mem+CB_MEML-stMemLTbl, stMemLTbl,
+ diff, CB_NSTAGES,
+ &weightdenum[(start-1)*(LPC_FILTERORDER+1)],
+ weightState, 0);
+
+ /* construct decoded vector */
+
+ iCBConstruct(reverseDecresidual, extra_cb_index,
+ extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
+ diff, CB_NSTAGES);
+
+ /* get decoded residual from reversed vector */
+
+ for (k=0; k<diff; k++) {
+ decresidual[start_pos-1-k] = reverseDecresidual[k];
+
+
+
+Andersen, et al. Experimental [Page 57]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ }
+
+ /* counter for predicted sub-frames */
+
+ subcount=0;
+
+ /* forward prediction of sub-frames */
+
+ Nfor = iLBCenc_inst->nsub-start-1;
+
+
+ if ( Nfor > 0 ) {
+
+ /* setup memory */
+
+ memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
+ memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
+ STATE_LEN*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ /* loop over sub-frames to encode */
+
+ for (subframe=0; subframe<Nfor; subframe++) {
+
+ /* encode sub-frame */
+
+ iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
+ gain_index+subcount*CB_NSTAGES,
+ &residual[(start+1+subframe)*SUBL],
+ mem+CB_MEML-memLfTbl[subcount],
+ memLfTbl[subcount], SUBL, CB_NSTAGES,
+ &weightdenum[(start+1+subframe)*
+ (LPC_FILTERORDER+1)],
+ weightState, subcount+1);
+
+ /* construct decoded vector */
+
+ iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
+ cb_index+subcount*CB_NSTAGES,
+ gain_index+subcount*CB_NSTAGES,
+ mem+CB_MEML-memLfTbl[subcount],
+ memLfTbl[subcount], SUBL, CB_NSTAGES);
+
+ /* update memory */
+
+ memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
+ memcpy(mem+CB_MEML-SUBL,
+
+
+
+Andersen, et al. Experimental [Page 58]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ &decresidual[(start+1+subframe)*SUBL],
+ SUBL*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ subcount++;
+ }
+ }
+
+
+ /* backward prediction of sub-frames */
+
+ Nback = start-1;
+
+
+ if ( Nback > 0 ) {
+
+ /* create reverse order vectors */
+
+ for (n=0; n<Nback; n++) {
+ for (k=0; k<SUBL; k++) {
+ reverseResidual[n*SUBL+k] =
+ residual[(start-1)*SUBL-1-n*SUBL-k];
+ reverseDecresidual[n*SUBL+k] =
+ decresidual[(start-1)*SUBL-1-n*SUBL-k];
+ }
+ }
+
+ /* setup memory */
+
+ meml_gotten = SUBL*(iLBCenc_inst->nsub+1-start);
+
+
+ if ( meml_gotten > CB_MEML ) {
+ meml_gotten=CB_MEML;
+ }
+ for (k=0; k<meml_gotten; k++) {
+ mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
+ }
+ memset(mem, 0, (CB_MEML-k)*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ /* loop over sub-frames to encode */
+
+ for (subframe=0; subframe<Nback; subframe++) {
+
+ /* encode sub-frame */
+
+ iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
+
+
+
+Andersen, et al. Experimental [Page 59]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ gain_index+subcount*CB_NSTAGES,
+ &reverseResidual[subframe*SUBL],
+ mem+CB_MEML-memLfTbl[subcount],
+ memLfTbl[subcount], SUBL, CB_NSTAGES,
+ &weightdenum[(start-2-subframe)*
+ (LPC_FILTERORDER+1)],
+ weightState, subcount+1);
+
+ /* construct decoded vector */
+
+ iCBConstruct(&reverseDecresidual[subframe*SUBL],
+ cb_index+subcount*CB_NSTAGES,
+ gain_index+subcount*CB_NSTAGES,
+ mem+CB_MEML-memLfTbl[subcount],
+ memLfTbl[subcount], SUBL, CB_NSTAGES);
+
+ /* update memory */
+
+ memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
+ memcpy(mem+CB_MEML-SUBL,
+ &reverseDecresidual[subframe*SUBL],
+ SUBL*sizeof(float));
+ memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
+
+ subcount++;
+
+ }
+
+ /* get decoded residual from reversed vector */
+
+ for (i=0; i<SUBL*Nback; i++) {
+ decresidual[SUBL*Nback - i - 1] =
+ reverseDecresidual[i];
+ }
+ }
+ /* end encoding part */
+
+ /* adjust index */
+ index_conv_enc(cb_index);
+
+ /* pack bytes */
+
+ pbytes=bytes;
+ pos=0;
+
+ /* loop over the 3 ULP classes */
+
+ for (ulp=0; ulp<3; ulp++) {
+
+
+
+Andersen, et al. Experimental [Page 60]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+
+ /* LSF */
+ for (k=0; k<LSF_NSPLIT*iLBCenc_inst->lpc_n; k++) {
+ packsplit(&lsf_i[k], &firstpart, &lsf_i[k],
+ iLBCenc_inst->ULP_inst->lsf_bits[k][ulp],
+ iLBCenc_inst->ULP_inst->lsf_bits[k][ulp]+
+ iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+1]+
+ iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->lsf_bits[k][ulp], &pos);
+ }
+
+ /* Start block info */
+
+ packsplit(&start, &firstpart, &start,
+ iLBCenc_inst->ULP_inst->start_bits[ulp],
+ iLBCenc_inst->ULP_inst->start_bits[ulp]+
+ iLBCenc_inst->ULP_inst->start_bits[ulp+1]+
+ iLBCenc_inst->ULP_inst->start_bits[ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->start_bits[ulp], &pos);
+
+ packsplit(&state_first, &firstpart, &state_first,
+ iLBCenc_inst->ULP_inst->startfirst_bits[ulp],
+ iLBCenc_inst->ULP_inst->startfirst_bits[ulp]+
+ iLBCenc_inst->ULP_inst->startfirst_bits[ulp+1]+
+ iLBCenc_inst->ULP_inst->startfirst_bits[ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->startfirst_bits[ulp], &pos);
+
+ packsplit(&idxForMax, &firstpart, &idxForMax,
+ iLBCenc_inst->ULP_inst->scale_bits[ulp],
+ iLBCenc_inst->ULP_inst->scale_bits[ulp]+
+ iLBCenc_inst->ULP_inst->scale_bits[ulp+1]+
+ iLBCenc_inst->ULP_inst->scale_bits[ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->scale_bits[ulp], &pos);
+
+ for (k=0; k<iLBCenc_inst->state_short_len; k++) {
+ packsplit(idxVec+k, &firstpart, idxVec+k,
+ iLBCenc_inst->ULP_inst->state_bits[ulp],
+ iLBCenc_inst->ULP_inst->state_bits[ulp]+
+ iLBCenc_inst->ULP_inst->state_bits[ulp+1]+
+ iLBCenc_inst->ULP_inst->state_bits[ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->state_bits[ulp], &pos);
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 61]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* 23/22 (20ms/30ms) sample block */
+
+ for (k=0;k<CB_NSTAGES;k++) {
+ packsplit(extra_cb_index+k, &firstpart,
+ extra_cb_index+k,
+ iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
+ iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp]+
+ iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+1]+
+ iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
+ &pos);
+ }
+
+ for (k=0;k<CB_NSTAGES;k++) {
+ packsplit(extra_gain_index+k, &firstpart,
+ extra_gain_index+k,
+ iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
+ iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp]+
+ iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+1]+
+ iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
+ &pos);
+ }
+
+ /* The two/four (20ms/30ms) 40 sample sub-blocks */
+
+ for (i=0; i<iLBCenc_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ packsplit(cb_index+i*CB_NSTAGES+k, &firstpart,
+ cb_index+i*CB_NSTAGES+k,
+ iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
+ iLBCenc_inst->ULP_inst->cb_index[i][k][ulp]+
+ iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+1]+
+ iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
+ &pos);
+ }
+ }
+
+ for (i=0; i<iLBCenc_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ packsplit(gain_index+i*CB_NSTAGES+k, &firstpart,
+ gain_index+i*CB_NSTAGES+k,
+ iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
+ iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp]+
+
+
+
+Andersen, et al. Experimental [Page 62]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+1]+
+ iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+2]);
+ dopack( &pbytes, firstpart,
+ iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
+ &pos);
+ }
+ }
+ }
+
+ /* set the last bit to zero (otherwise the decoder
+ will treat it as a lost frame) */
+ dopack( &pbytes, 0, 1, &pos);
+ }
+
+A.4. iLBC_decode.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_decode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ILBCDECODE_H
+ #define __iLBC_ILBCDECODE_H
+
+ #include "iLBC_define.h"
+
+ short initDecode( /* (o) Number of decoded
+ samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ int mode, /* (i) frame size mode */
+ int use_enhancer /* (i) 1 to use enhancer
+ 0 to run without
+ enhancer */
+ );
+
+ void iLBC_decode(
+ float *decblock, /* (o) decoded signal block */
+ unsigned char *bytes, /* (i) encoded signal bits */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ int mode /* (i) 0: bad packet, PLC,
+ 1: normal */
+
+
+
+Andersen, et al. Experimental [Page 63]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ );
+
+ #endif
+
+A.5. iLBC_decode.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_decode.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <stdlib.h>
+
+ #include "iLBC_define.h"
+ #include "StateConstructW.h"
+ #include "LPCdecode.h"
+ #include "iCBConstruct.h"
+ #include "doCPLC.h"
+ #include "helpfun.h"
+ #include "constants.h"
+ #include "packing.h"
+ #include "string.h"
+ #include "enhancer.h"
+ #include "hpOutput.h"
+ #include "syntFilter.h"
+
+ /*----------------------------------------------------------------*
+ * Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+ short initDecode( /* (o) Number of decoded
+ samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ int mode, /* (i) frame size mode */
+ int use_enhancer /* (i) 1 to use enhancer
+ 0 to run without
+ enhancer */
+ ){
+ int i;
+
+ iLBCdec_inst->mode = mode;
+
+
+
+Andersen, et al. Experimental [Page 64]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ if (mode==30) {
+ iLBCdec_inst->blockl = BLOCKL_30MS;
+ iLBCdec_inst->nsub = NSUB_30MS;
+ iLBCdec_inst->nasub = NASUB_30MS;
+ iLBCdec_inst->lpc_n = LPC_N_30MS;
+ iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
+ iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
+ iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
+ /* ULP init */
+ iLBCdec_inst->ULP_inst=&ULP_30msTbl;
+ }
+ else if (mode==20) {
+ iLBCdec_inst->blockl = BLOCKL_20MS;
+ iLBCdec_inst->nsub = NSUB_20MS;
+ iLBCdec_inst->nasub = NASUB_20MS;
+ iLBCdec_inst->lpc_n = LPC_N_20MS;
+ iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
+ iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
+ iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
+ /* ULP init */
+ iLBCdec_inst->ULP_inst=&ULP_20msTbl;
+ }
+ else {
+ exit(2);
+ }
+
+ memset(iLBCdec_inst->syntMem, 0,
+ LPC_FILTERORDER*sizeof(float));
+ memcpy((*iLBCdec_inst).lsfdeqold, lsfmeanTbl,
+ LPC_FILTERORDER*sizeof(float));
+
+ memset(iLBCdec_inst->old_syntdenum, 0,
+ ((LPC_FILTERORDER + 1)*NSUB_MAX)*sizeof(float));
+ for (i=0; i<NSUB_MAX; i++)
+ iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]=1.0;
+
+ iLBCdec_inst->last_lag = 20;
+
+ iLBCdec_inst->prevLag = 120;
+ iLBCdec_inst->per = 0.0;
+ iLBCdec_inst->consPLICount = 0;
+ iLBCdec_inst->prevPLI = 0;
+ iLBCdec_inst->prevLpc[0] = 1.0;
+ memset(iLBCdec_inst->prevLpc+1,0,
+ LPC_FILTERORDER*sizeof(float));
+ memset(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX*sizeof(float));
+ iLBCdec_inst->seed=777;
+
+
+
+
+Andersen, et al. Experimental [Page 65]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ memset(iLBCdec_inst->hpomem, 0, 4*sizeof(float));
+
+ iLBCdec_inst->use_enhancer = use_enhancer;
+ memset(iLBCdec_inst->enh_buf, 0, ENH_BUFL*sizeof(float));
+ for (i=0;i<ENH_NBLOCKS_TOT;i++)
+ iLBCdec_inst->enh_period[i]=(float)40.0;
+
+ iLBCdec_inst->prev_enh_pl = 0;
+
+ return (iLBCdec_inst->blockl);
+ }
+
+ /*----------------------------------------------------------------*
+ * frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+ void Decode(
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ float *decresidual, /* (o) decoded residual frame */
+ int start, /* (i) location of start
+ state */
+ int idxForMax, /* (i) codebook index for the
+ maximum value */
+ int *idxVec, /* (i) codebook indexes for the
+ samples in the start
+ state */
+ float *syntdenum, /* (i) the decoded synthesis
+ filter coefficients */
+ int *cb_index, /* (i) the indexes for the
+ adaptive codebook */
+ int *gain_index, /* (i) the indexes for the
+ corresponding gains */
+ int *extra_cb_index, /* (i) the indexes for the
+ adaptive codebook part
+ of start state */
+ int *extra_gain_index, /* (i) the indexes for the
+ corresponding gains */
+ int state_first /* (i) 1 if non adaptive part
+ of start state comes
+ first 0 if that part
+ comes last */
+ ){
+ float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
+ int k, meml_gotten, Nfor, Nback, i;
+ int diff, start_pos;
+ int subcount, subframe;
+
+
+
+
+Andersen, et al. Experimental [Page 66]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ diff = STATE_LEN - iLBCdec_inst->state_short_len;
+
+ if (state_first == 1) {
+ start_pos = (start-1)*SUBL;
+ } else {
+ start_pos = (start-1)*SUBL + diff;
+ }
+
+ /* decode scalar part of start state */
+
+ StateConstructW(idxForMax, idxVec,
+ &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
+ &decresidual[start_pos], iLBCdec_inst->state_short_len);
+
+
+ if (state_first) { /* put adaptive part in the end */
+
+ /* setup memory */
+
+ memset(mem, 0,
+ (CB_MEML-iLBCdec_inst->state_short_len)*sizeof(float));
+ memcpy(mem+CB_MEML-iLBCdec_inst->state_short_len,
+ decresidual+start_pos,
+ iLBCdec_inst->state_short_len*sizeof(float));
+
+ /* construct decoded vector */
+
+ iCBConstruct(
+ &decresidual[start_pos+iLBCdec_inst->state_short_len],
+ extra_cb_index, extra_gain_index, mem+CB_MEML-stMemLTbl,
+ stMemLTbl, diff, CB_NSTAGES);
+
+ }
+ else {/* put adaptive part in the beginning */
+
+ /* create reversed vectors for prediction */
+
+ for (k=0; k<diff; k++) {
+ reverseDecresidual[k] =
+ decresidual[(start+1)*SUBL-1-
+ (k+iLBCdec_inst->state_short_len)];
+ }
+
+ /* setup memory */
+
+ meml_gotten = iLBCdec_inst->state_short_len;
+ for (k=0; k<meml_gotten; k++){
+ mem[CB_MEML-1-k] = decresidual[start_pos + k];
+
+
+
+Andersen, et al. Experimental [Page 67]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ memset(mem, 0, (CB_MEML-k)*sizeof(float));
+
+ /* construct decoded vector */
+
+ iCBConstruct(reverseDecresidual, extra_cb_index,
+ extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
+ diff, CB_NSTAGES);
+
+ /* get decoded residual from reversed vector */
+
+ for (k=0; k<diff; k++) {
+ decresidual[start_pos-1-k] = reverseDecresidual[k];
+ }
+ }
+
+ /* counter for predicted sub-frames */
+
+ subcount=0;
+
+ /* forward prediction of sub-frames */
+
+ Nfor = iLBCdec_inst->nsub-start-1;
+
+ if ( Nfor > 0 ){
+
+ /* setup memory */
+
+ memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
+ memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
+ STATE_LEN*sizeof(float));
+
+ /* loop over sub-frames to encode */
+
+ for (subframe=0; subframe<Nfor; subframe++) {
+
+ /* construct decoded vector */
+
+ iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
+ cb_index+subcount*CB_NSTAGES,
+ gain_index+subcount*CB_NSTAGES,
+ mem+CB_MEML-memLfTbl[subcount],
+ memLfTbl[subcount], SUBL, CB_NSTAGES);
+
+ /* update memory */
+
+ memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
+ memcpy(mem+CB_MEML-SUBL,
+
+
+
+Andersen, et al. Experimental [Page 68]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ &decresidual[(start+1+subframe)*SUBL],
+ SUBL*sizeof(float));
+
+ subcount++;
+
+ }
+
+ }
+
+ /* backward prediction of sub-frames */
+
+ Nback = start-1;
+
+ if ( Nback > 0 ) {
+
+ /* setup memory */
+
+ meml_gotten = SUBL*(iLBCdec_inst->nsub+1-start);
+
+ if ( meml_gotten > CB_MEML ) {
+ meml_gotten=CB_MEML;
+ }
+ for (k=0; k<meml_gotten; k++) {
+ mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
+ }
+ memset(mem, 0, (CB_MEML-k)*sizeof(float));
+
+ /* loop over subframes to decode */
+
+ for (subframe=0; subframe<Nback; subframe++) {
+
+ /* construct decoded vector */
+
+ iCBConstruct(&reverseDecresidual[subframe*SUBL],
+ cb_index+subcount*CB_NSTAGES,
+ gain_index+subcount*CB_NSTAGES,
+ mem+CB_MEML-memLfTbl[subcount], memLfTbl[subcount],
+ SUBL, CB_NSTAGES);
+
+ /* update memory */
+
+ memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
+ memcpy(mem+CB_MEML-SUBL,
+ &reverseDecresidual[subframe*SUBL],
+ SUBL*sizeof(float));
+
+ subcount++;
+ }
+
+
+
+Andersen, et al. Experimental [Page 69]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* get decoded residual from reversed vector */
+
+ for (i=0; i<SUBL*Nback; i++)
+ decresidual[SUBL*Nback - i - 1] =
+ reverseDecresidual[i];
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * main decoder function
+ *---------------------------------------------------------------*/
+
+ void iLBC_decode(
+ float *decblock, /* (o) decoded signal block */
+ unsigned char *bytes, /* (i) encoded signal bits */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ int mode /* (i) 0: bad packet, PLC,
+ 1: normal */
+ ){
+ float data[BLOCKL_MAX];
+ float lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+ float PLCresidual[BLOCKL_MAX], PLClpc[LPC_FILTERORDER + 1];
+ float zeros[BLOCKL_MAX], one[LPC_FILTERORDER + 1];
+ int k, i, start, idxForMax, pos, lastpart, ulp;
+ int lag, ilag;
+ float cc, maxcc;
+ int idxVec[STATE_LEN];
+ int check;
+ int gain_index[NASUB_MAX*CB_NSTAGES],
+ extra_gain_index[CB_NSTAGES];
+ int cb_index[CB_NSTAGES*NASUB_MAX], extra_cb_index[CB_NSTAGES];
+ int lsf_i[LSF_NSPLIT*LPC_N_MAX];
+ int state_first;
+ int last_bit;
+ unsigned char *pbytes;
+ float weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+ int order_plus_one;
+ float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+ float decresidual[BLOCKL_MAX];
+
+ if (mode>0) { /* the data are good */
+
+ /* decode data */
+
+ pbytes=bytes;
+ pos=0;
+
+
+
+
+Andersen, et al. Experimental [Page 70]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* Set everything to zero before decoding */
+
+ for (k=0; k<LSF_NSPLIT*LPC_N_MAX; k++) {
+ lsf_i[k]=0;
+ }
+ start=0;
+ state_first=0;
+ idxForMax=0;
+ for (k=0; k<iLBCdec_inst->state_short_len; k++) {
+ idxVec[k]=0;
+ }
+ for (k=0; k<CB_NSTAGES; k++) {
+ extra_cb_index[k]=0;
+ }
+ for (k=0; k<CB_NSTAGES; k++) {
+ extra_gain_index[k]=0;
+ }
+ for (i=0; i<iLBCdec_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ cb_index[i*CB_NSTAGES+k]=0;
+ }
+ }
+ for (i=0; i<iLBCdec_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ gain_index[i*CB_NSTAGES+k]=0;
+ }
+ }
+
+ /* loop over ULP classes */
+
+ for (ulp=0; ulp<3; ulp++) {
+
+ /* LSF */
+ for (k=0; k<LSF_NSPLIT*iLBCdec_inst->lpc_n; k++){
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->lsf_bits[k][ulp], &pos);
+ packcombine(&lsf_i[k], lastpart,
+ iLBCdec_inst->ULP_inst->lsf_bits[k][ulp]);
+ }
+
+ /* Start block info */
+
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->start_bits[ulp], &pos);
+ packcombine(&start, lastpart,
+ iLBCdec_inst->ULP_inst->start_bits[ulp]);
+
+ unpack( &pbytes, &lastpart,
+
+
+
+Andersen, et al. Experimental [Page 71]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ iLBCdec_inst->ULP_inst->startfirst_bits[ulp], &pos);
+ packcombine(&state_first, lastpart,
+ iLBCdec_inst->ULP_inst->startfirst_bits[ulp]);
+
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->scale_bits[ulp], &pos);
+ packcombine(&idxForMax, lastpart,
+ iLBCdec_inst->ULP_inst->scale_bits[ulp]);
+
+ for (k=0; k<iLBCdec_inst->state_short_len; k++) {
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->state_bits[ulp], &pos);
+ packcombine(idxVec+k, lastpart,
+ iLBCdec_inst->ULP_inst->state_bits[ulp]);
+ }
+
+ /* 23/22 (20ms/30ms) sample block */
+
+ for (k=0; k<CB_NSTAGES; k++) {
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp],
+ &pos);
+ packcombine(extra_cb_index+k, lastpart,
+ iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp]);
+ }
+ for (k=0; k<CB_NSTAGES; k++) {
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp],
+ &pos);
+ packcombine(extra_gain_index+k, lastpart,
+ iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp]);
+ }
+
+ /* The two/four (20ms/30ms) 40 sample sub-blocks */
+
+ for (i=0; i<iLBCdec_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ unpack( &pbytes, &lastpart,
+ iLBCdec_inst->ULP_inst->cb_index[i][k][ulp],
+ &pos);
+ packcombine(cb_index+i*CB_NSTAGES+k, lastpart,
+ iLBCdec_inst->ULP_inst->cb_index[i][k][ulp]);
+ }
+ }
+
+ for (i=0; i<iLBCdec_inst->nasub; i++) {
+ for (k=0; k<CB_NSTAGES; k++) {
+ unpack( &pbytes, &lastpart,
+
+
+
+Andersen, et al. Experimental [Page 72]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp],
+ &pos);
+ packcombine(gain_index+i*CB_NSTAGES+k, lastpart,
+ iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp]);
+ }
+ }
+ }
+ /* Extract last bit. If it is 1 this indicates an
+ empty/lost frame */
+ unpack( &pbytes, &last_bit, 1, &pos);
+
+ /* Check for bit errors or empty/lost frames */
+ if (start<1)
+ mode = 0;
+ if (iLBCdec_inst->mode==20 && start>3)
+ mode = 0;
+ if (iLBCdec_inst->mode==30 && start>5)
+ mode = 0;
+ if (last_bit==1)
+ mode = 0;
+
+ if (mode==1) { /* No bit errors was detected,
+ continue decoding */
+
+ /* adjust index */
+ index_conv_dec(cb_index);
+
+ /* decode the lsf */
+
+ SimplelsfDEQ(lsfdeq, lsf_i, iLBCdec_inst->lpc_n);
+ check=LSF_check(lsfdeq, LPC_FILTERORDER,
+ iLBCdec_inst->lpc_n);
+ DecoderInterpolateLSF(syntdenum, weightdenum,
+ lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
+
+ Decode(iLBCdec_inst, decresidual, start, idxForMax,
+ idxVec, syntdenum, cb_index, gain_index,
+ extra_cb_index, extra_gain_index,
+ state_first);
+
+ /* preparing the plc for a future loss! */
+
+ doThePLC(PLCresidual, PLClpc, 0, decresidual,
+ syntdenum +
+ (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
+ (*iLBCdec_inst).last_lag, iLBCdec_inst);
+
+
+
+
+
+Andersen, et al. Experimental [Page 73]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ memcpy(decresidual, PLCresidual,
+ iLBCdec_inst->blockl*sizeof(float));
+ }
+
+ }
+
+ if (mode == 0) {
+ /* the data is bad (either a PLC call
+ * was made or a severe bit error was detected)
+ */
+
+ /* packet loss conceal */
+
+ memset(zeros, 0, BLOCKL_MAX*sizeof(float));
+
+ one[0] = 1;
+ memset(one+1, 0, LPC_FILTERORDER*sizeof(float));
+
+ start=0;
+
+ doThePLC(PLCresidual, PLClpc, 1, zeros, one,
+ (*iLBCdec_inst).last_lag, iLBCdec_inst);
+ memcpy(decresidual, PLCresidual,
+ iLBCdec_inst->blockl*sizeof(float));
+
+ order_plus_one = LPC_FILTERORDER + 1;
+ for (i = 0; i < iLBCdec_inst->nsub; i++) {
+ memcpy(syntdenum+(i*order_plus_one), PLClpc,
+ order_plus_one*sizeof(float));
+ }
+ }
+
+ if (iLBCdec_inst->use_enhancer == 1) {
+
+ /* post filtering */
+
+ iLBCdec_inst->last_lag =
+ enhancerInterface(data, decresidual, iLBCdec_inst);
+
+ /* synthesis filtering */
+
+ if (iLBCdec_inst->mode==20) {
+ /* Enhancer has 40 samples delay */
+ i=0;
+ syntFilter(data + i*SUBL,
+ iLBCdec_inst->old_syntdenum +
+ (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
+ SUBL, iLBCdec_inst->syntMem);
+
+
+
+Andersen, et al. Experimental [Page 74]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (i=1; i < iLBCdec_inst->nsub; i++) {
+ syntFilter(data + i*SUBL,
+ syntdenum + (i-1)*(LPC_FILTERORDER+1),
+ SUBL, iLBCdec_inst->syntMem);
+ }
+ } else if (iLBCdec_inst->mode==30) {
+ /* Enhancer has 80 samples delay */
+ for (i=0; i < 2; i++) {
+ syntFilter(data + i*SUBL,
+ iLBCdec_inst->old_syntdenum +
+ (i+iLBCdec_inst->nsub-2)*(LPC_FILTERORDER+1),
+ SUBL, iLBCdec_inst->syntMem);
+ }
+ for (i=2; i < iLBCdec_inst->nsub; i++) {
+ syntFilter(data + i*SUBL,
+ syntdenum + (i-2)*(LPC_FILTERORDER+1), SUBL,
+ iLBCdec_inst->syntMem);
+ }
+ }
+
+ } else {
+
+ /* Find last lag */
+ lag = 20;
+ maxcc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
+ &decresidual[BLOCKL_MAX-ENH_BLOCKL-lag], ENH_BLOCKL);
+
+ for (ilag=21; ilag<120; ilag++) {
+ cc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
+ &decresidual[BLOCKL_MAX-ENH_BLOCKL-ilag],
+ ENH_BLOCKL);
+
+ if (cc > maxcc) {
+ maxcc = cc;
+ lag = ilag;
+ }
+ }
+ iLBCdec_inst->last_lag = lag;
+
+ /* copy data and run synthesis filter */
+
+ memcpy(data, decresidual,
+ iLBCdec_inst->blockl*sizeof(float));
+ for (i=0; i < iLBCdec_inst->nsub; i++) {
+ syntFilter(data + i*SUBL,
+ syntdenum + i*(LPC_FILTERORDER+1), SUBL,
+ iLBCdec_inst->syntMem);
+ }
+
+
+
+Andersen, et al. Experimental [Page 75]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+ /* high pass filtering on output if desired, otherwise
+ copy to out */
+
+ hpOutput(data, iLBCdec_inst->blockl,
+ decblock,iLBCdec_inst->hpomem);
+
+ /* memcpy(decblock,data,iLBCdec_inst->blockl*sizeof(float));*/
+
+ memcpy(iLBCdec_inst->old_syntdenum, syntdenum,
+
+ iLBCdec_inst->nsub*(LPC_FILTERORDER+1)*sizeof(float));
+
+ iLBCdec_inst->prev_enh_pl=0;
+
+ if (mode==0) { /* PLC was used */
+ iLBCdec_inst->prev_enh_pl=1;
+ }
+ }
+
+A.6. iLBC_define.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_define.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+ #include <string.h>
+
+ #ifndef __iLBC_ILBCDEFINE_H
+ #define __iLBC_ILBCDEFINE_H
+
+ /* general codec settings */
+
+ #define FS (float)8000.0
+ #define BLOCKL_20MS 160
+ #define BLOCKL_30MS 240
+ #define BLOCKL_MAX 240
+ #define NSUB_20MS 4
+ #define NSUB_30MS 6
+ #define NSUB_MAX 6
+ #define NASUB_20MS 2
+
+
+
+Andersen, et al. Experimental [Page 76]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #define NASUB_30MS 4
+ #define NASUB_MAX 4
+ #define SUBL 40
+ #define STATE_LEN 80
+ #define STATE_SHORT_LEN_30MS 58
+ #define STATE_SHORT_LEN_20MS 57
+
+ /* LPC settings */
+
+ #define LPC_FILTERORDER 10
+ #define LPC_CHIRP_SYNTDENUM (float)0.9025
+ #define LPC_CHIRP_WEIGHTDENUM (float)0.4222
+ #define LPC_LOOKBACK 60
+ #define LPC_N_20MS 1
+ #define LPC_N_30MS 2
+ #define LPC_N_MAX 2
+ #define LPC_ASYMDIFF 20
+ #define LPC_BW (float)60.0
+ #define LPC_WN (float)1.0001
+ #define LSF_NSPLIT 3
+ #define LSF_NUMBER_OF_STEPS 4
+ #define LPC_HALFORDER (LPC_FILTERORDER/2)
+
+ /* cb settings */
+
+ #define CB_NSTAGES 3
+ #define CB_EXPAND 2
+ #define CB_MEML 147
+ #define CB_FILTERLEN 2*4
+ #define CB_HALFFILTERLEN 4
+ #define CB_RESRANGE 34
+ #define CB_MAXGAIN (float)1.3
+
+ /* enhancer */
+
+ #define ENH_BLOCKL 80 /* block length */
+ #define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
+ #define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
+ in said second sequence */
+ #define ENH_SLOP 2 /* max difference estimated and
+ correct pitch period */
+ #define ENH_PLOCSL 20 /* pitch-estimates and pitch-
+ locations buffer length */
+ #define ENH_OVERHANG 2
+ #define ENH_UPS0 4 /* upsampling rate */
+ #define ENH_FL0 3 /* 2*FLO+1 is the length of
+ each filter */
+ #define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
+
+
+
+Andersen, et al. Experimental [Page 77]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #define ENH_CORRDIM (2*ENH_SLOP+1)
+ #define ENH_NBLOCKS (BLOCKL_MAX/ENH_BLOCKL)
+ #define ENH_NBLOCKS_EXTRA 5
+ #define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS +
+ ENH_NBLOCKS_EXTRA */
+ #define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
+ #define ENH_ALPHA0 (float)0.05
+
+ /* Down sampling */
+
+ #define FILTERORDER_DS 7
+ #define DELAY_DS 3
+ #define FACTOR_DS 2
+
+ /* bit stream defs */
+
+ #define NO_OF_BYTES_20MS 38
+ #define NO_OF_BYTES_30MS 50
+ #define NO_OF_WORDS_20MS 19
+ #define NO_OF_WORDS_30MS 25
+ #define STATE_BITS 3
+ #define BYTE_LEN 8
+ #define ULP_CLASSES 3
+
+ /* help parameters */
+
+ #define FLOAT_MAX (float)1.0e37
+ #define EPS (float)2.220446049250313e-016
+ #define PI (float)3.14159265358979323846
+ #define MIN_SAMPLE -32768
+ #define MAX_SAMPLE 32767
+ #define TWO_PI (float)6.283185307
+ #define PI2 (float)0.159154943
+
+ /* type definition encoder instance */
+ typedef struct iLBC_ULP_Inst_t_ {
+ int lsf_bits[6][ULP_CLASSES+2];
+ int start_bits[ULP_CLASSES+2];
+ int startfirst_bits[ULP_CLASSES+2];
+ int scale_bits[ULP_CLASSES+2];
+ int state_bits[ULP_CLASSES+2];
+ int extra_cb_index[CB_NSTAGES][ULP_CLASSES+2];
+ int extra_cb_gain[CB_NSTAGES][ULP_CLASSES+2];
+ int cb_index[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
+ int cb_gain[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
+ } iLBC_ULP_Inst_t;
+
+ /* type definition encoder instance */
+
+
+
+Andersen, et al. Experimental [Page 78]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ typedef struct iLBC_Enc_Inst_t_ {
+
+ /* flag for frame size mode */
+ int mode;
+
+ /* basic parameters for different frame sizes */
+ int blockl;
+ int nsub;
+ int nasub;
+ int no_of_bytes, no_of_words;
+ int lpc_n;
+ int state_short_len;
+ const iLBC_ULP_Inst_t *ULP_inst;
+
+ /* analysis filter state */
+ float anaMem[LPC_FILTERORDER];
+
+ /* old lsf parameters for interpolation */
+ float lsfold[LPC_FILTERORDER];
+ float lsfdeqold[LPC_FILTERORDER];
+
+ /* signal buffer for LP analysis */
+ float lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+ /* state of input HP filter */
+ float hpimem[4];
+
+ } iLBC_Enc_Inst_t;
+
+ /* type definition decoder instance */
+ typedef struct iLBC_Dec_Inst_t_ {
+
+ /* flag for frame size mode */
+ int mode;
+
+ /* basic parameters for different frame sizes */
+ int blockl;
+ int nsub;
+ int nasub;
+ int no_of_bytes, no_of_words;
+ int lpc_n;
+ int state_short_len;
+ const iLBC_ULP_Inst_t *ULP_inst;
+
+ /* synthesis filter state */
+ float syntMem[LPC_FILTERORDER];
+
+ /* old LSF for interpolation */
+
+
+
+Andersen, et al. Experimental [Page 79]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float lsfdeqold[LPC_FILTERORDER];
+
+ /* pitch lag estimated in enhancer and used in PLC */
+ int last_lag;
+
+ /* PLC state information */
+ int prevLag, consPLICount, prevPLI, prev_enh_pl;
+ float prevLpc[LPC_FILTERORDER+1];
+ float prevResidual[NSUB_MAX*SUBL];
+ float per;
+ unsigned long seed;
+
+ /* previous synthesis filter parameters */
+ float old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+ /* state of output HP filter */
+ float hpomem[4];
+
+ /* enhancer state information */
+ int use_enhancer;
+ float enh_buf[ENH_BUFL];
+ float enh_period[ENH_NBLOCKS_TOT];
+
+ } iLBC_Dec_Inst_t;
+
+ #endif
+
+A.7. constants.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_CONSTANTS_H
+ #define __iLBC_CONSTANTS_H
+
+ #include "iLBC_define.h"
+
+
+ /* ULP bit allocation */
+
+
+
+
+Andersen, et al. Experimental [Page 80]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ extern const iLBC_ULP_Inst_t ULP_20msTbl;
+ extern const iLBC_ULP_Inst_t ULP_30msTbl;
+
+ /* high pass filters */
+
+ extern float hpi_zero_coefsTbl[];
+ extern float hpi_pole_coefsTbl[];
+ extern float hpo_zero_coefsTbl[];
+ extern float hpo_pole_coefsTbl[];
+
+ /* low pass filters */
+ extern float lpFilt_coefsTbl[];
+
+ /* LPC analysis and quantization */
+
+ extern float lpc_winTbl[];
+ extern float lpc_asymwinTbl[];
+ extern float lpc_lagwinTbl[];
+ extern float lsfCbTbl[];
+ extern float lsfmeanTbl[];
+ extern int dim_lsfCbTbl[];
+ extern int size_lsfCbTbl[];
+ extern float lsf_weightTbl_30ms[];
+ extern float lsf_weightTbl_20ms[];
+
+ /* state quantization tables */
+
+ extern float state_sq3Tbl[];
+ extern float state_frgqTbl[];
+
+ /* gain quantization tables */
+
+ extern float gain_sq3Tbl[];
+ extern float gain_sq4Tbl[];
+ extern float gain_sq5Tbl[];
+
+ /* adaptive codebook definitions */
+
+ extern int search_rangeTbl[5][CB_NSTAGES];
+ extern int memLfTbl[];
+ extern int stMemLTbl;
+ extern float cbfiltersTbl[CB_FILTERLEN];
+
+ /* enhancer definitions */
+
+ extern float polyphaserTbl[];
+ extern float enh_plocsTbl[];
+
+
+
+
+Andersen, et al. Experimental [Page 81]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #endif
+
+A.8. constants.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "iLBC_define.h"
+
+ /* ULP bit allocation */
+
+ /* 20 ms frame */
+
+ const iLBC_ULP_Inst_t ULP_20msTbl = {
+ /* LSF */
+ { {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
+ {0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
+ /* Start state location, gain and samples */
+ {2,0,0,0,0},
+ {1,0,0,0,0},
+ {6,0,0,0,0},
+ {0,1,2,0,0},
+ /* extra CB index and extra CB gain */
+ {{6,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
+ {{2,0,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
+ /* CB index and CB gain */
+ { {{7,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
+ {{0,0,8,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
+ {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
+ {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}},
+ { {{1,2,2,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
+ {{1,1,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
+ {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
+ {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}}
+ };
+
+ /* 30 ms frame */
+
+ const iLBC_ULP_Inst_t ULP_30msTbl = {
+ /* LSF */
+
+
+
+Andersen, et al. Experimental [Page 82]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ { {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
+ {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0}},
+ /* Start state location, gain and samples */
+ {3,0,0,0,0},
+ {1,0,0,0,0},
+ {6,0,0,0,0},
+ {0,1,2,0,0},
+ /* extra CB index and extra CB gain */
+ {{4,2,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
+ {{1,1,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
+ /* CB index and CB gain */
+ { {{6,1,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
+ {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
+ {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
+ {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}}},
+ { {{1,2,2,0,0}, {1,2,1,0,0}, {0,0,3,0,0}},
+ {{0,2,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
+ {{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}},
+ {{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}}}
+ };
+
+ /* HP Filters */
+
+ float hpi_zero_coefsTbl[3] = {
+ (float)0.92727436, (float)-1.8544941, (float)0.92727436
+ };
+ float hpi_pole_coefsTbl[3] = {
+ (float)1.0, (float)-1.9059465, (float)0.9114024
+ };
+ float hpo_zero_coefsTbl[3] = {
+ (float)0.93980581, (float)-1.8795834, (float)0.93980581
+ };
+ float hpo_pole_coefsTbl[3] = {
+ (float)1.0, (float)-1.9330735, (float)0.93589199
+ };
+
+ /* LP Filter */
+
+ float lpFilt_coefsTbl[FILTERORDER_DS]={
+ (float)-0.066650, (float)0.125000, (float)0.316650,
+ (float)0.414063, (float)0.316650,
+ (float)0.125000, (float)-0.066650
+ };
+
+ /* State quantization tables */
+
+ float state_sq3Tbl[8] = {
+ (float)-3.719849, (float)-2.177490, (float)-1.130005,
+
+
+
+Andersen, et al. Experimental [Page 83]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)-0.309692, (float)0.444214, (float)1.329712,
+ (float)2.436279, (float)3.983887
+ };
+
+ float state_frgqTbl[64] = {
+ (float)1.000085, (float)1.071695, (float)1.140395,
+ (float)1.206868, (float)1.277188, (float)1.351503,
+ (float)1.429380, (float)1.500727, (float)1.569049,
+ (float)1.639599, (float)1.707071, (float)1.781531,
+ (float)1.840799, (float)1.901550, (float)1.956695,
+ (float)2.006750, (float)2.055474, (float)2.102787,
+ (float)2.142819, (float)2.183592, (float)2.217962,
+ (float)2.257177, (float)2.295739, (float)2.332967,
+ (float)2.369248, (float)2.402792, (float)2.435080,
+ (float)2.468598, (float)2.503394, (float)2.539284,
+ (float)2.572944, (float)2.605036, (float)2.636331,
+ (float)2.668939, (float)2.698780, (float)2.729101,
+ (float)2.759786, (float)2.789834, (float)2.818679,
+ (float)2.848074, (float)2.877470, (float)2.906899,
+ (float)2.936655, (float)2.967804, (float)3.000115,
+ (float)3.033367, (float)3.066355, (float)3.104231,
+ (float)3.141499, (float)3.183012, (float)3.222952,
+ (float)3.265433, (float)3.308441, (float)3.350823,
+ (float)3.395275, (float)3.442793, (float)3.490801,
+ (float)3.542514, (float)3.604064, (float)3.666050,
+ (float)3.740994, (float)3.830749, (float)3.938770,
+ (float)4.101764
+ };
+
+ /* CB tables */
+
+ int search_rangeTbl[5][CB_NSTAGES]={{58,58,58}, {108,44,44},
+ {108,108,108}, {108,108,108}, {108,108,108}};
+ int stMemLTbl=85;
+ int memLfTbl[NASUB_MAX]={147,147,147,147};
+
+ /* expansion filter(s) */
+
+ float cbfiltersTbl[CB_FILTERLEN]={
+ (float)-0.034180, (float)0.108887, (float)-0.184326,
+ (float)0.806152, (float)0.713379, (float)-0.144043,
+ (float)0.083740, (float)-0.033691
+ };
+
+ /* Gain Quantization */
+
+ float gain_sq3Tbl[8]={
+ (float)-1.000000, (float)-0.659973, (float)-0.330017,
+
+
+
+Andersen, et al. Experimental [Page 84]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.000000, (float)0.250000, (float)0.500000,
+ (float)0.750000, (float)1.00000};
+
+ float gain_sq4Tbl[16]={
+ (float)-1.049988, (float)-0.900024, (float)-0.750000,
+ (float)-0.599976, (float)-0.450012, (float)-0.299988,
+ (float)-0.150024, (float)0.000000, (float)0.150024,
+ (float)0.299988, (float)0.450012, (float)0.599976,
+ (float)0.750000, (float)0.900024, (float)1.049988,
+ (float)1.200012};
+
+ float gain_sq5Tbl[32]={
+ (float)0.037476, (float)0.075012, (float)0.112488,
+ (float)0.150024, (float)0.187500, (float)0.224976,
+ (float)0.262512, (float)0.299988, (float)0.337524,
+ (float)0.375000, (float)0.412476, (float)0.450012,
+ (float)0.487488, (float)0.525024, (float)0.562500,
+ (float)0.599976, (float)0.637512, (float)0.674988,
+ (float)0.712524, (float)0.750000, (float)0.787476,
+ (float)0.825012, (float)0.862488, (float)0.900024,
+ (float)0.937500, (float)0.974976, (float)1.012512,
+ (float)1.049988, (float)1.087524, (float)1.125000,
+ (float)1.162476, (float)1.200012};
+
+ /* Enhancer - Upsamling a factor 4 (ENH_UPS0 = 4) */
+ float polyphaserTbl[ENH_UPS0*(2*ENH_FL0+1)]={
+ (float)0.000000, (float)0.000000, (float)0.000000,
+ (float)1.000000,
+ (float)0.000000, (float)0.000000, (float)0.000000,
+ (float)0.015625, (float)-0.076904, (float)0.288330,
+ (float)0.862061,
+ (float)-0.106445, (float)0.018799, (float)-0.015625,
+ (float)0.023682, (float)-0.124268, (float)0.601563,
+ (float)0.601563,
+ (float)-0.124268, (float)0.023682, (float)-0.023682,
+ (float)0.018799, (float)-0.106445, (float)0.862061,
+ (float)0.288330,
+ (float)-0.076904, (float)0.015625, (float)-0.018799};
+
+ float enh_plocsTbl[ENH_NBLOCKS_TOT] = {(float)40.0, (float)120.0,
+ (float)200.0, (float)280.0, (float)360.0,
+ (float)440.0, (float)520.0, (float)600.0};
+
+ /* LPC analysis and quantization */
+
+ int dim_lsfCbTbl[LSF_NSPLIT] = {3, 3, 4};
+ int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
+
+
+
+
+Andersen, et al. Experimental [Page 85]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float lsfmeanTbl[LPC_FILTERORDER] = {
+ (float)0.281738, (float)0.445801, (float)0.663330,
+ (float)0.962524, (float)1.251831, (float)1.533081,
+ (float)1.850586, (float)2.137817, (float)2.481445,
+ (float)2.777344};
+
+ float lsf_weightTbl_30ms[6] = {(float)(1.0/2.0), (float)1.0,
+ (float)(2.0/3.0),
+ (float)(1.0/3.0), (float)0.0, (float)0.0};
+
+ float lsf_weightTbl_20ms[4] = {(float)(3.0/4.0), (float)(2.0/4.0),
+ (float)(1.0/4.0), (float)(0.0)};
+
+ /* Hanning LPC window */
+ float lpc_winTbl[BLOCKL_MAX]={
+ (float)0.000183, (float)0.000671, (float)0.001526,
+ (float)0.002716, (float)0.004242, (float)0.006104,
+ (float)0.008301, (float)0.010834, (float)0.013702,
+ (float)0.016907, (float)0.020416, (float)0.024261,
+ (float)0.028442, (float)0.032928, (float)0.037750,
+ (float)0.042877, (float)0.048309, (float)0.054047,
+ (float)0.060089, (float)0.066437, (float)0.073090,
+ (float)0.080017, (float)0.087219, (float)0.094727,
+ (float)0.102509, (float)0.110535, (float)0.118835,
+ (float)0.127411, (float)0.136230, (float)0.145294,
+ (float)0.154602, (float)0.164154, (float)0.173920,
+ (float)0.183899, (float)0.194122, (float)0.204529,
+ (float)0.215149, (float)0.225952, (float)0.236938,
+ (float)0.248108, (float)0.259460, (float)0.270966,
+ (float)0.282654, (float)0.294464, (float)0.306396,
+ (float)0.318481, (float)0.330688, (float)0.343018,
+ (float)0.355438, (float)0.367981, (float)0.380585,
+ (float)0.393280, (float)0.406067, (float)0.418884,
+ (float)0.431763, (float)0.444702, (float)0.457672,
+ (float)0.470673, (float)0.483704, (float)0.496735,
+ (float)0.509766, (float)0.522797, (float)0.535828,
+ (float)0.548798, (float)0.561768, (float)0.574677,
+ (float)0.587524, (float)0.600342, (float)0.613068,
+ (float)0.625732, (float)0.638306, (float)0.650787,
+ (float)0.663147, (float)0.675415, (float)0.687561,
+ (float)0.699585, (float)0.711487, (float)0.723206,
+ (float)0.734802, (float)0.746216, (float)0.757477,
+ (float)0.768585, (float)0.779480, (float)0.790192,
+ (float)0.800720, (float)0.811005, (float)0.821106,
+ (float)0.830994, (float)0.840668, (float)0.850067,
+ (float)0.859253, (float)0.868225, (float)0.876892,
+ (float)0.885345, (float)0.893524, (float)0.901428,
+ (float)0.909058, (float)0.916412, (float)0.923492,
+
+
+
+Andersen, et al. Experimental [Page 86]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.930267, (float)0.936768, (float)0.942963,
+ (float)0.948853, (float)0.954437, (float)0.959717,
+ (float)0.964691, (float)0.969360, (float)0.973694,
+ (float)0.977692, (float)0.981384, (float)0.984741,
+ (float)0.987762, (float)0.990479, (float)0.992828,
+ (float)0.994873, (float)0.996552, (float)0.997925,
+ (float)0.998932, (float)0.999603, (float)0.999969,
+ (float)0.999969, (float)0.999603, (float)0.998932,
+ (float)0.997925, (float)0.996552, (float)0.994873,
+ (float)0.992828, (float)0.990479, (float)0.987762,
+ (float)0.984741, (float)0.981384, (float)0.977692,
+ (float)0.973694, (float)0.969360, (float)0.964691,
+ (float)0.959717, (float)0.954437, (float)0.948853,
+ (float)0.942963, (float)0.936768, (float)0.930267,
+ (float)0.923492, (float)0.916412, (float)0.909058,
+ (float)0.901428, (float)0.893524, (float)0.885345,
+ (float)0.876892, (float)0.868225, (float)0.859253,
+ (float)0.850067, (float)0.840668, (float)0.830994,
+ (float)0.821106, (float)0.811005, (float)0.800720,
+ (float)0.790192, (float)0.779480, (float)0.768585,
+ (float)0.757477, (float)0.746216, (float)0.734802,
+ (float)0.723206, (float)0.711487, (float)0.699585,
+ (float)0.687561, (float)0.675415, (float)0.663147,
+ (float)0.650787, (float)0.638306, (float)0.625732,
+ (float)0.613068, (float)0.600342, (float)0.587524,
+ (float)0.574677, (float)0.561768, (float)0.548798,
+ (float)0.535828, (float)0.522797, (float)0.509766,
+ (float)0.496735, (float)0.483704, (float)0.470673,
+ (float)0.457672, (float)0.444702, (float)0.431763,
+ (float)0.418884, (float)0.406067, (float)0.393280,
+ (float)0.380585, (float)0.367981, (float)0.355438,
+ (float)0.343018, (float)0.330688, (float)0.318481,
+ (float)0.306396, (float)0.294464, (float)0.282654,
+ (float)0.270966, (float)0.259460, (float)0.248108,
+ (float)0.236938, (float)0.225952, (float)0.215149,
+ (float)0.204529, (float)0.194122, (float)0.183899,
+ (float)0.173920, (float)0.164154, (float)0.154602,
+ (float)0.145294, (float)0.136230, (float)0.127411,
+ (float)0.118835, (float)0.110535, (float)0.102509,
+ (float)0.094727, (float)0.087219, (float)0.080017,
+ (float)0.073090, (float)0.066437, (float)0.060089,
+ (float)0.054047, (float)0.048309, (float)0.042877,
+ (float)0.037750, (float)0.032928, (float)0.028442,
+ (float)0.024261, (float)0.020416, (float)0.016907,
+ (float)0.013702, (float)0.010834, (float)0.008301,
+ (float)0.006104, (float)0.004242, (float)0.002716,
+ (float)0.001526, (float)0.000671, (float)0.000183
+ };
+
+
+
+Andersen, et al. Experimental [Page 87]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* Asymmetric LPC window */
+ float lpc_asymwinTbl[BLOCKL_MAX]={
+ (float)0.000061, (float)0.000214, (float)0.000458,
+ (float)0.000824, (float)0.001282, (float)0.001831,
+ (float)0.002472, (float)0.003235, (float)0.004120,
+ (float)0.005066, (float)0.006134, (float)0.007294,
+ (float)0.008545, (float)0.009918, (float)0.011383,
+ (float)0.012939, (float)0.014587, (float)0.016357,
+ (float)0.018219, (float)0.020172, (float)0.022217,
+ (float)0.024353, (float)0.026611, (float)0.028961,
+ (float)0.031372, (float)0.033905, (float)0.036530,
+ (float)0.039276, (float)0.042084, (float)0.044983,
+ (float)0.047974, (float)0.051086, (float)0.054260,
+ (float)0.057526, (float)0.060883, (float)0.064331,
+ (float)0.067871, (float)0.071503, (float)0.075226,
+ (float)0.079010, (float)0.082916, (float)0.086884,
+ (float)0.090942, (float)0.095062, (float)0.099304,
+ (float)0.103607, (float)0.107971, (float)0.112427,
+ (float)0.116974, (float)0.121582, (float)0.126282,
+ (float)0.131073, (float)0.135895, (float)0.140839,
+ (float)0.145813, (float)0.150879, (float)0.156006,
+ (float)0.161224, (float)0.166504, (float)0.171844,
+ (float)0.177246, (float)0.182709, (float)0.188263,
+ (float)0.193848, (float)0.199524, (float)0.205231,
+ (float)0.211029, (float)0.216858, (float)0.222778,
+ (float)0.228729, (float)0.234741, (float)0.240814,
+ (float)0.246918, (float)0.253082, (float)0.259308,
+ (float)0.265564, (float)0.271881, (float)0.278259,
+ (float)0.284668, (float)0.291107, (float)0.297607,
+ (float)0.304138, (float)0.310730, (float)0.317322,
+ (float)0.323975, (float)0.330658, (float)0.337372,
+ (float)0.344147, (float)0.350922, (float)0.357727,
+ (float)0.364594, (float)0.371460, (float)0.378357,
+ (float)0.385284, (float)0.392212, (float)0.399170,
+ (float)0.406158, (float)0.413177, (float)0.420197,
+ (float)0.427246, (float)0.434296, (float)0.441376,
+ (float)0.448456, (float)0.455536, (float)0.462646,
+ (float)0.469757, (float)0.476868, (float)0.483978,
+ (float)0.491089, (float)0.498230, (float)0.505341,
+ (float)0.512451, (float)0.519592, (float)0.526703,
+ (float)0.533813, (float)0.540924, (float)0.548004,
+ (float)0.555084, (float)0.562164, (float)0.569244,
+ (float)0.576294, (float)0.583313, (float)0.590332,
+ (float)0.597321, (float)0.604309, (float)0.611267,
+ (float)0.618195, (float)0.625092, (float)0.631989,
+ (float)0.638855, (float)0.645660, (float)0.652466,
+ (float)0.659241, (float)0.665985, (float)0.672668,
+ (float)0.679352, (float)0.685974, (float)0.692566,
+
+
+
+Andersen, et al. Experimental [Page 88]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.699127, (float)0.705658, (float)0.712128,
+ (float)0.718536, (float)0.724945, (float)0.731262,
+ (float)0.737549, (float)0.743805, (float)0.750000,
+ (float)0.756134, (float)0.762238, (float)0.768280,
+ (float)0.774261, (float)0.780182, (float)0.786072,
+ (float)0.791870, (float)0.797638, (float)0.803314,
+ (float)0.808960, (float)0.814514, (float)0.820038,
+ (float)0.825470, (float)0.830841, (float)0.836151,
+ (float)0.841400, (float)0.846558, (float)0.851654,
+ (float)0.856689, (float)0.861633, (float)0.866516,
+ (float)0.871338, (float)0.876068, (float)0.880737,
+ (float)0.885315, (float)0.889801, (float)0.894226,
+ (float)0.898560, (float)0.902832, (float)0.907013,
+ (float)0.911102, (float)0.915100, (float)0.919037,
+ (float)0.922882, (float)0.926636, (float)0.930328,
+ (float)0.933899, (float)0.937408, (float)0.940796,
+ (float)0.944122, (float)0.947357, (float)0.950470,
+ (float)0.953522, (float)0.956482, (float)0.959351,
+ (float)0.962097, (float)0.964783, (float)0.967377,
+ (float)0.969849, (float)0.972229, (float)0.974518,
+ (float)0.976715, (float)0.978821, (float)0.980835,
+ (float)0.982727, (float)0.984528, (float)0.986237,
+ (float)0.987854, (float)0.989380, (float)0.990784,
+ (float)0.992096, (float)0.993317, (float)0.994415,
+ (float)0.995422, (float)0.996338, (float)0.997162,
+ (float)0.997864, (float)0.998474, (float)0.998962,
+ (float)0.999390, (float)0.999695, (float)0.999878,
+ (float)0.999969, (float)0.999969, (float)0.996918,
+ (float)0.987701, (float)0.972382, (float)0.951050,
+ (float)0.923889, (float)0.891022, (float)0.852631,
+ (float)0.809021, (float)0.760406, (float)0.707092,
+ (float)0.649445, (float)0.587799, (float)0.522491,
+ (float)0.453979, (float)0.382690, (float)0.309021,
+ (float)0.233459, (float)0.156433, (float)0.078461
+ };
+
+ /* Lag window for LPC */
+ float lpc_lagwinTbl[LPC_FILTERORDER + 1]={
+ (float)1.000100, (float)0.998890, (float)0.995569,
+ (float)0.990057, (float)0.982392,
+ (float)0.972623, (float)0.960816, (float)0.947047,
+ (float)0.931405, (float)0.913989, (float)0.894909};
+
+ /* LSF quantization*/
+ float lsfCbTbl[64 * 3 + 128 * 3 + 128 * 4] = {
+ (float)0.155396, (float)0.273193, (float)0.451172,
+ (float)0.390503, (float)0.648071, (float)1.002075,
+ (float)0.440186, (float)0.692261, (float)0.955688,
+
+
+
+Andersen, et al. Experimental [Page 89]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.343628, (float)0.642334, (float)1.071533,
+ (float)0.318359, (float)0.491577, (float)0.670532,
+ (float)0.193115, (float)0.375488, (float)0.725708,
+ (float)0.364136, (float)0.510376, (float)0.658691,
+ (float)0.297485, (float)0.527588, (float)0.842529,
+ (float)0.227173, (float)0.365967, (float)0.563110,
+ (float)0.244995, (float)0.396729, (float)0.636475,
+ (float)0.169434, (float)0.300171, (float)0.520264,
+ (float)0.312866, (float)0.464478, (float)0.643188,
+ (float)0.248535, (float)0.429932, (float)0.626099,
+ (float)0.236206, (float)0.491333, (float)0.817139,
+ (float)0.334961, (float)0.625122, (float)0.895752,
+ (float)0.343018, (float)0.518555, (float)0.698608,
+ (float)0.372803, (float)0.659790, (float)0.945435,
+ (float)0.176880, (float)0.316528, (float)0.581421,
+ (float)0.416382, (float)0.625977, (float)0.805176,
+ (float)0.303223, (float)0.568726, (float)0.915039,
+ (float)0.203613, (float)0.351440, (float)0.588135,
+ (float)0.221191, (float)0.375000, (float)0.614746,
+ (float)0.199951, (float)0.323364, (float)0.476074,
+ (float)0.300781, (float)0.433350, (float)0.566895,
+ (float)0.226196, (float)0.354004, (float)0.507568,
+ (float)0.300049, (float)0.508179, (float)0.711670,
+ (float)0.312012, (float)0.492676, (float)0.763428,
+ (float)0.329956, (float)0.541016, (float)0.795776,
+ (float)0.373779, (float)0.604614, (float)0.928833,
+ (float)0.210571, (float)0.452026, (float)0.755249,
+ (float)0.271118, (float)0.473267, (float)0.662476,
+ (float)0.285522, (float)0.436890, (float)0.634399,
+ (float)0.246704, (float)0.565552, (float)0.859009,
+ (float)0.270508, (float)0.406250, (float)0.553589,
+ (float)0.361450, (float)0.578491, (float)0.813843,
+ (float)0.342651, (float)0.482788, (float)0.622437,
+ (float)0.340332, (float)0.549438, (float)0.743164,
+ (float)0.200439, (float)0.336304, (float)0.540894,
+ (float)0.407837, (float)0.644775, (float)0.895142,
+ (float)0.294678, (float)0.454834, (float)0.699097,
+ (float)0.193115, (float)0.344482, (float)0.643188,
+ (float)0.275757, (float)0.420776, (float)0.598755,
+ (float)0.380493, (float)0.608643, (float)0.861084,
+ (float)0.222778, (float)0.426147, (float)0.676514,
+ (float)0.407471, (float)0.700195, (float)1.053101,
+ (float)0.218384, (float)0.377197, (float)0.669922,
+ (float)0.313232, (float)0.454102, (float)0.600952,
+ (float)0.347412, (float)0.571533, (float)0.874146,
+ (float)0.238037, (float)0.405396, (float)0.729492,
+ (float)0.223877, (float)0.412964, (float)0.822021,
+ (float)0.395264, (float)0.582153, (float)0.743896,
+
+
+
+Andersen, et al. Experimental [Page 90]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.247925, (float)0.485596, (float)0.720581,
+ (float)0.229126, (float)0.496582, (float)0.907715,
+ (float)0.260132, (float)0.566895, (float)1.012695,
+ (float)0.337402, (float)0.611572, (float)0.978149,
+ (float)0.267822, (float)0.447632, (float)0.769287,
+ (float)0.250610, (float)0.381714, (float)0.530029,
+ (float)0.430054, (float)0.805054, (float)1.221924,
+ (float)0.382568, (float)0.544067, (float)0.701660,
+ (float)0.383545, (float)0.710327, (float)1.149170,
+ (float)0.271362, (float)0.529053, (float)0.775513,
+ (float)0.246826, (float)0.393555, (float)0.588623,
+ (float)0.266846, (float)0.422119, (float)0.676758,
+ (float)0.311523, (float)0.580688, (float)0.838623,
+ (float)1.331177, (float)1.576782, (float)1.779541,
+ (float)1.160034, (float)1.401978, (float)1.768188,
+ (float)1.161865, (float)1.525146, (float)1.715332,
+ (float)0.759521, (float)0.913940, (float)1.119873,
+ (float)0.947144, (float)1.121338, (float)1.282471,
+ (float)1.015015, (float)1.557007, (float)1.804932,
+ (float)1.172974, (float)1.402100, (float)1.692627,
+ (float)1.087524, (float)1.474243, (float)1.665405,
+ (float)0.899536, (float)1.105225, (float)1.406250,
+ (float)1.148438, (float)1.484741, (float)1.796265,
+ (float)0.785645, (float)1.209839, (float)1.567749,
+ (float)0.867798, (float)1.166504, (float)1.450684,
+ (float)0.922485, (float)1.229858, (float)1.420898,
+ (float)0.791260, (float)1.123291, (float)1.409546,
+ (float)0.788940, (float)0.966064, (float)1.340332,
+ (float)1.051147, (float)1.272827, (float)1.556641,
+ (float)0.866821, (float)1.181152, (float)1.538818,
+ (float)0.906738, (float)1.373535, (float)1.607910,
+ (float)1.244751, (float)1.581421, (float)1.933838,
+ (float)0.913940, (float)1.337280, (float)1.539673,
+ (float)0.680542, (float)0.959229, (float)1.662720,
+ (float)0.887207, (float)1.430542, (float)1.800781,
+ (float)0.912598, (float)1.433594, (float)1.683960,
+ (float)0.860474, (float)1.060303, (float)1.455322,
+ (float)1.005127, (float)1.381104, (float)1.706909,
+ (float)0.800781, (float)1.363892, (float)1.829102,
+ (float)0.781860, (float)1.124390, (float)1.505981,
+ (float)1.003662, (float)1.471436, (float)1.684692,
+ (float)0.981323, (float)1.309570, (float)1.618042,
+ (float)1.228760, (float)1.554321, (float)1.756470,
+ (float)0.734375, (float)0.895752, (float)1.225586,
+ (float)0.841797, (float)1.055664, (float)1.249268,
+ (float)0.920166, (float)1.119385, (float)1.486206,
+ (float)0.894409, (float)1.539063, (float)1.828979,
+ (float)1.283691, (float)1.543335, (float)1.858276,
+
+
+
+Andersen, et al. Experimental [Page 91]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.676025, (float)0.933105, (float)1.490845,
+ (float)0.821289, (float)1.491821, (float)1.739868,
+ (float)0.923218, (float)1.144653, (float)1.580566,
+ (float)1.057251, (float)1.345581, (float)1.635864,
+ (float)0.888672, (float)1.074951, (float)1.353149,
+ (float)0.942749, (float)1.195435, (float)1.505493,
+ (float)1.492310, (float)1.788086, (float)2.039673,
+ (float)1.070313, (float)1.634399, (float)1.860962,
+ (float)1.253296, (float)1.488892, (float)1.686035,
+ (float)0.647095, (float)0.864014, (float)1.401855,
+ (float)0.866699, (float)1.254883, (float)1.453369,
+ (float)1.063965, (float)1.532593, (float)1.731323,
+ (float)1.167847, (float)1.521484, (float)1.884033,
+ (float)0.956055, (float)1.502075, (float)1.745605,
+ (float)0.928711, (float)1.288574, (float)1.479614,
+ (float)1.088013, (float)1.380737, (float)1.570801,
+ (float)0.905029, (float)1.186768, (float)1.371948,
+ (float)1.057861, (float)1.421021, (float)1.617432,
+ (float)1.108276, (float)1.312500, (float)1.501465,
+ (float)0.979492, (float)1.416992, (float)1.624268,
+ (float)1.276001, (float)1.661011, (float)2.007935,
+ (float)0.993042, (float)1.168579, (float)1.331665,
+ (float)0.778198, (float)0.944946, (float)1.235962,
+ (float)1.223755, (float)1.491333, (float)1.815674,
+ (float)0.852661, (float)1.350464, (float)1.722290,
+ (float)1.134766, (float)1.593140, (float)1.787354,
+ (float)1.051392, (float)1.339722, (float)1.531006,
+ (float)0.803589, (float)1.271240, (float)1.652100,
+ (float)0.755737, (float)1.143555, (float)1.639404,
+ (float)0.700928, (float)0.837280, (float)1.130371,
+ (float)0.942749, (float)1.197876, (float)1.669800,
+ (float)0.993286, (float)1.378296, (float)1.566528,
+ (float)0.801025, (float)1.095337, (float)1.298950,
+ (float)0.739990, (float)1.032959, (float)1.383667,
+ (float)0.845703, (float)1.072266, (float)1.543823,
+ (float)0.915649, (float)1.072266, (float)1.224487,
+ (float)1.021973, (float)1.226196, (float)1.481323,
+ (float)0.999878, (float)1.204102, (float)1.555908,
+ (float)0.722290, (float)0.913940, (float)1.340210,
+ (float)0.673340, (float)0.835938, (float)1.259521,
+ (float)0.832397, (float)1.208374, (float)1.394165,
+ (float)0.962158, (float)1.576172, (float)1.912842,
+ (float)1.166748, (float)1.370850, (float)1.556763,
+ (float)0.946289, (float)1.138550, (float)1.400391,
+ (float)1.035034, (float)1.218262, (float)1.386475,
+ (float)1.393799, (float)1.717773, (float)2.000244,
+ (float)0.972656, (float)1.260986, (float)1.760620,
+ (float)1.028198, (float)1.288452, (float)1.484619,
+
+
+
+Andersen, et al. Experimental [Page 92]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)0.773560, (float)1.258057, (float)1.756714,
+ (float)1.080322, (float)1.328003, (float)1.742676,
+ (float)0.823975, (float)1.450806, (float)1.917725,
+ (float)0.859009, (float)1.016602, (float)1.191895,
+ (float)0.843994, (float)1.131104, (float)1.645020,
+ (float)1.189697, (float)1.702759, (float)1.894409,
+ (float)1.346680, (float)1.763184, (float)2.066040,
+ (float)0.980469, (float)1.253784, (float)1.441650,
+ (float)1.338135, (float)1.641968, (float)1.932739,
+ (float)1.223267, (float)1.424194, (float)1.626465,
+ (float)0.765747, (float)1.004150, (float)1.579102,
+ (float)1.042847, (float)1.269165, (float)1.647461,
+ (float)0.968750, (float)1.257568, (float)1.555786,
+ (float)0.826294, (float)0.993408, (float)1.275146,
+ (float)0.742310, (float)0.950439, (float)1.430542,
+ (float)1.054321, (float)1.439819, (float)1.828003,
+ (float)1.072998, (float)1.261719, (float)1.441895,
+ (float)0.859375, (float)1.036377, (float)1.314819,
+ (float)0.895752, (float)1.267212, (float)1.605591,
+ (float)0.805420, (float)0.962891, (float)1.142334,
+ (float)0.795654, (float)1.005493, (float)1.468506,
+ (float)1.105347, (float)1.313843, (float)1.584839,
+ (float)0.792236, (float)1.221802, (float)1.465698,
+ (float)1.170532, (float)1.467651, (float)1.664063,
+ (float)0.838257, (float)1.153198, (float)1.342163,
+ (float)0.968018, (float)1.198242, (float)1.391235,
+ (float)1.250122, (float)1.623535, (float)1.823608,
+ (float)0.711670, (float)1.058350, (float)1.512085,
+ (float)1.204834, (float)1.454468, (float)1.739136,
+ (float)1.137451, (float)1.421753, (float)1.620117,
+ (float)0.820435, (float)1.322754, (float)1.578247,
+ (float)0.798706, (float)1.005005, (float)1.213867,
+ (float)0.980713, (float)1.324951, (float)1.512939,
+ (float)1.112305, (float)1.438843, (float)1.735596,
+ (float)1.135498, (float)1.356689, (float)1.635742,
+ (float)1.101318, (float)1.387451, (float)1.686523,
+ (float)0.849854, (float)1.276978, (float)1.523438,
+ (float)1.377930, (float)1.627563, (float)1.858154,
+ (float)0.884888, (float)1.095459, (float)1.287476,
+ (float)1.289795, (float)1.505859, (float)1.756592,
+ (float)0.817505, (float)1.384155, (float)1.650513,
+ (float)1.446655, (float)1.702148, (float)1.931885,
+ (float)0.835815, (float)1.023071, (float)1.385376,
+ (float)0.916626, (float)1.139038, (float)1.335327,
+ (float)0.980103, (float)1.174072, (float)1.453735,
+ (float)1.705688, (float)2.153809, (float)2.398315, (float)2.743408,
+ (float)1.797119, (float)2.016846, (float)2.445679, (float)2.701904,
+ (float)1.990356, (float)2.219116, (float)2.576416, (float)2.813477,
+
+
+
+Andersen, et al. Experimental [Page 93]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)1.849365, (float)2.190918, (float)2.611572, (float)2.835083,
+ (float)1.657959, (float)1.854370, (float)2.159058, (float)2.726196,
+ (float)1.437744, (float)1.897705, (float)2.253174, (float)2.655396,
+ (float)2.028687, (float)2.247314, (float)2.542358, (float)2.875854,
+ (float)1.736938, (float)1.922119, (float)2.185913, (float)2.743408,
+ (float)1.521606, (float)1.870972, (float)2.526855, (float)2.786987,
+ (float)1.841431, (float)2.050659, (float)2.463623, (float)2.857666,
+ (float)1.590088, (float)2.067261, (float)2.427979, (float)2.794434,
+ (float)1.746826, (float)2.057373, (float)2.320190, (float)2.800781,
+ (float)1.734619, (float)1.940552, (float)2.306030, (float)2.826416,
+ (float)1.786255, (float)2.204468, (float)2.457520, (float)2.795288,
+ (float)1.861084, (float)2.170532, (float)2.414551, (float)2.763672,
+ (float)2.001465, (float)2.307617, (float)2.552734, (float)2.811890,
+ (float)1.784424, (float)2.124146, (float)2.381592, (float)2.645508,
+ (float)1.888794, (float)2.135864, (float)2.418579, (float)2.861206,
+ (float)2.301147, (float)2.531250, (float)2.724976, (float)2.913086,
+ (float)1.837769, (float)2.051270, (float)2.261963, (float)2.553223,
+ (float)2.012939, (float)2.221191, (float)2.440186, (float)2.678101,
+ (float)1.429565, (float)1.858276, (float)2.582275, (float)2.845703,
+ (float)1.622803, (float)1.897705, (float)2.367310, (float)2.621094,
+ (float)1.581543, (float)1.960449, (float)2.515869, (float)2.736450,
+ (float)1.419434, (float)1.933960, (float)2.394653, (float)2.746704,
+ (float)1.721924, (float)2.059570, (float)2.421753, (float)2.769653,
+ (float)1.911011, (float)2.220703, (float)2.461060, (float)2.740723,
+ (float)1.581177, (float)1.860840, (float)2.516968, (float)2.874634,
+ (float)1.870361, (float)2.098755, (float)2.432373, (float)2.656494,
+ (float)2.059692, (float)2.279785, (float)2.495605, (float)2.729370,
+ (float)1.815674, (float)2.181519, (float)2.451538, (float)2.680542,
+ (float)1.407959, (float)1.768311, (float)2.343018, (float)2.668091,
+ (float)2.168701, (float)2.394653, (float)2.604736, (float)2.829346,
+ (float)1.636230, (float)1.865723, (float)2.329102, (float)2.824219,
+ (float)1.878906, (float)2.139526, (float)2.376709, (float)2.679810,
+ (float)1.765381, (float)1.971802, (float)2.195435, (float)2.586914,
+ (float)2.164795, (float)2.410889, (float)2.673706, (float)2.903198,
+ (float)2.071899, (float)2.331055, (float)2.645874, (float)2.907104,
+ (float)2.026001, (float)2.311523, (float)2.594849, (float)2.863892,
+ (float)1.948975, (float)2.180786, (float)2.514893, (float)2.797852,
+ (float)1.881836, (float)2.130859, (float)2.478149, (float)2.804199,
+ (float)2.238159, (float)2.452759, (float)2.652832, (float)2.868286,
+ (float)1.897949, (float)2.101685, (float)2.524292, (float)2.880127,
+ (float)1.856445, (float)2.074585, (float)2.541016, (float)2.791748,
+ (float)1.695557, (float)2.199097, (float)2.506226, (float)2.742676,
+ (float)1.612671, (float)1.877075, (float)2.435425, (float)2.732910,
+ (float)1.568848, (float)1.786499, (float)2.194580, (float)2.768555,
+ (float)1.953369, (float)2.164551, (float)2.486938, (float)2.874023,
+ (float)1.388306, (float)1.725342, (float)2.384521, (float)2.771851,
+ (float)2.115356, (float)2.337769, (float)2.592896, (float)2.864014,
+ (float)1.905762, (float)2.111328, (float)2.363525, (float)2.789307,
+
+
+
+Andersen, et al. Experimental [Page 94]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)1.882568, (float)2.332031, (float)2.598267, (float)2.827637,
+ (float)1.683594, (float)2.088745, (float)2.361938, (float)2.608643,
+ (float)1.874023, (float)2.182129, (float)2.536133, (float)2.766968,
+ (float)1.861938, (float)2.070435, (float)2.309692, (float)2.700562,
+ (float)1.722168, (float)2.107422, (float)2.477295, (float)2.837646,
+ (float)1.926880, (float)2.184692, (float)2.442627, (float)2.663818,
+ (float)2.123901, (float)2.337280, (float)2.553101, (float)2.777466,
+ (float)1.588135, (float)1.911499, (float)2.212769, (float)2.543945,
+ (float)2.053955, (float)2.370850, (float)2.712158, (float)2.939941,
+ (float)2.210449, (float)2.519653, (float)2.770386, (float)2.958618,
+ (float)2.199463, (float)2.474731, (float)2.718262, (float)2.919922,
+ (float)1.960083, (float)2.175415, (float)2.608032, (float)2.888794,
+ (float)1.953735, (float)2.185181, (float)2.428223, (float)2.809570,
+ (float)1.615234, (float)2.036499, (float)2.576538, (float)2.834595,
+ (float)1.621094, (float)2.028198, (float)2.431030, (float)2.664673,
+ (float)1.824951, (float)2.267456, (float)2.514526, (float)2.747925,
+ (float)1.994263, (float)2.229126, (float)2.475220, (float)2.833984,
+ (float)1.746338, (float)2.011353, (float)2.588257, (float)2.826904,
+ (float)1.562866, (float)2.135986, (float)2.471680, (float)2.687256,
+ (float)1.748901, (float)2.083496, (float)2.460938, (float)2.686279,
+ (float)1.758057, (float)2.131470, (float)2.636597, (float)2.891602,
+ (float)2.071289, (float)2.299072, (float)2.550781, (float)2.814331,
+ (float)1.839600, (float)2.094360, (float)2.496460, (float)2.723999,
+ (float)1.882202, (float)2.088257, (float)2.636841, (float)2.923096,
+ (float)1.957886, (float)2.153198, (float)2.384399, (float)2.615234,
+ (float)1.992920, (float)2.351196, (float)2.654419, (float)2.889771,
+ (float)2.012817, (float)2.262451, (float)2.643799, (float)2.903076,
+ (float)2.025635, (float)2.254761, (float)2.508423, (float)2.784058,
+ (float)2.316040, (float)2.589355, (float)2.794189, (float)2.963623,
+ (float)1.741211, (float)2.279541, (float)2.578491, (float)2.816284,
+ (float)1.845337, (float)2.055786, (float)2.348511, (float)2.822021,
+ (float)1.679932, (float)1.926514, (float)2.499756, (float)2.835693,
+ (float)1.722534, (float)1.946899, (float)2.448486, (float)2.728760,
+ (float)1.829834, (float)2.043213, (float)2.580444, (float)2.867676,
+ (float)1.676636, (float)2.071655, (float)2.322510, (float)2.704834,
+ (float)1.791504, (float)2.113525, (float)2.469727, (float)2.784058,
+ (float)1.977051, (float)2.215088, (float)2.497437, (float)2.726929,
+ (float)1.800171, (float)2.106689, (float)2.357788, (float)2.738892,
+ (float)1.827759, (float)2.170166, (float)2.525879, (float)2.852417,
+ (float)1.918335, (float)2.132813, (float)2.488403, (float)2.728149,
+ (float)1.916748, (float)2.225098, (float)2.542603, (float)2.857666,
+ (float)1.761230, (float)1.976074, (float)2.507446, (float)2.884521,
+ (float)2.053711, (float)2.367432, (float)2.608032, (float)2.837646,
+ (float)1.595337, (float)2.000977, (float)2.307129, (float)2.578247,
+ (float)1.470581, (float)2.031250, (float)2.375854, (float)2.647583,
+ (float)1.801392, (float)2.128052, (float)2.399780, (float)2.822876,
+ (float)1.853638, (float)2.066650, (float)2.429199, (float)2.751465,
+ (float)1.956299, (float)2.163696, (float)2.394775, (float)2.734253,
+
+
+
+Andersen, et al. Experimental [Page 95]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)1.963623, (float)2.275757, (float)2.585327, (float)2.865234,
+ (float)1.887451, (float)2.105469, (float)2.331787, (float)2.587402,
+ (float)2.120117, (float)2.443359, (float)2.733887, (float)2.941406,
+ (float)1.506348, (float)1.766968, (float)2.400513, (float)2.851807,
+ (float)1.664551, (float)1.981079, (float)2.375732, (float)2.774414,
+ (float)1.720703, (float)1.978882, (float)2.391479, (float)2.640991,
+ (float)1.483398, (float)1.814819, (float)2.434448, (float)2.722290,
+ (float)1.769043, (float)2.136597, (float)2.563721, (float)2.774414,
+ (float)1.810791, (float)2.049316, (float)2.373901, (float)2.613647,
+ (float)1.788330, (float)2.005981, (float)2.359131, (float)2.723145,
+ (float)1.785156, (float)1.993164, (float)2.399780, (float)2.832520,
+ (float)1.695313, (float)2.022949, (float)2.522583, (float)2.745117,
+ (float)1.584106, (float)1.965576, (float)2.299927, (float)2.715576,
+ (float)1.894897, (float)2.249878, (float)2.655884, (float)2.897705,
+ (float)1.720581, (float)1.995728, (float)2.299438, (float)2.557007,
+ (float)1.619385, (float)2.173950, (float)2.574219, (float)2.787964,
+ (float)1.883179, (float)2.220459, (float)2.474365, (float)2.825073,
+ (float)1.447632, (float)2.045044, (float)2.555542, (float)2.744873,
+ (float)1.502686, (float)2.156616, (float)2.653320, (float)2.846558,
+ (float)1.711548, (float)1.944092, (float)2.282959, (float)2.685791,
+ (float)1.499756, (float)1.867554, (float)2.341064, (float)2.578857,
+ (float)1.916870, (float)2.135132, (float)2.568237, (float)2.826050,
+ (float)1.498047, (float)1.711182, (float)2.223267, (float)2.755127,
+ (float)1.808716, (float)1.997559, (float)2.256470, (float)2.758545,
+ (float)2.088501, (float)2.402710, (float)2.667358, (float)2.890259,
+ (float)1.545044, (float)1.819214, (float)2.324097, (float)2.692993,
+ (float)1.796021, (float)2.012573, (float)2.505737, (float)2.784912,
+ (float)1.786499, (float)2.041748, (float)2.290405, (float)2.650757,
+ (float)1.938232, (float)2.264404, (float)2.529053, (float)2.796143
+ };
+
+A.9. anaFilter.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ anaFilter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ANAFILTER_H
+ #define __iLBC_ANAFILTER_H
+
+ void anaFilter(
+
+
+
+Andersen, et al. Experimental [Page 96]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *In, /* (i) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len,/* (i) Length of signal */
+ float *Out, /* (o) Filtered signal */
+ float *mem /* (i/o) Filter state */
+ );
+
+ #endif
+
+A.10. anaFilter.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ anaFilter.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <string.h>
+ #include "iLBC_define.h"
+
+ /*----------------------------------------------------------------*
+ * LP analysis filter.
+ *---------------------------------------------------------------*/
+
+ void anaFilter(
+ float *In, /* (i) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len,/* (i) Length of signal */
+ float *Out, /* (o) Filtered signal */
+ float *mem /* (i/o) Filter state */
+ ){
+ int i, j;
+ float *po, *pi, *pm, *pa;
+
+ po = Out;
+
+ /* Filter first part using memory from past */
+
+ for (i=0; i<LPC_FILTERORDER; i++) {
+ pi = &In[i];
+ pm = &mem[LPC_FILTERORDER-1];
+ pa = a;
+ *po=0.0;
+
+
+
+Andersen, et al. Experimental [Page 97]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (j=0; j<=i; j++) {
+ *po+=(*pa++)*(*pi--);
+ }
+ for (j=i+1; j<LPC_FILTERORDER+1; j++) {
+
+ *po+=(*pa++)*(*pm--);
+ }
+ po++;
+ }
+
+ /* Filter last part where the state is entirely
+ in the input vector */
+
+ for (i=LPC_FILTERORDER; i<len; i++) {
+ pi = &In[i];
+ pa = a;
+ *po=0.0;
+ for (j=0; j<LPC_FILTERORDER+1; j++) {
+ *po+=(*pa++)*(*pi--);
+ }
+ po++;
+ }
+
+ /* Update state vector */
+
+ memcpy(mem, &In[len-LPC_FILTERORDER],
+ LPC_FILTERORDER*sizeof(float));
+ }
+
+A.11. createCB.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ createCB.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_CREATECB_H
+ #define __iLBC_CREATECB_H
+
+ void filteredCBvecs(
+ float *cbvectors, /* (o) Codebook vector for the
+ higher section */
+
+
+
+Andersen, et al. Experimental [Page 98]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *mem, /* (i) Buffer to create codebook
+ vectors from */
+ int lMem /* (i) Length of buffer */
+ );
+
+ void searchAugmentedCB(
+ int low, /* (i) Start index for the search */
+ int high, /* (i) End index for the search */
+ int stage, /* (i) Current stage */
+ int startIndex, /* (i) CB index for the first
+ augmented vector */
+ float *target, /* (i) Target vector for encoding */
+ float *buffer, /* (i) Pointer to the end of the
+ buffer for augmented codebook
+ construction */
+ float *max_measure, /* (i/o) Currently maximum measure */
+ int *best_index,/* (o) Currently the best index */
+ float *gain, /* (o) Currently the best gain */
+ float *energy, /* (o) Energy of augmented
+ codebook vectors */
+ float *invenergy/* (o) Inv energy of aug codebook
+ vectors */
+ );
+
+ void createAugmentedVec(
+ int index, /* (i) Index for the aug vector
+ to be created */
+ float *buffer, /* (i) Pointer to the end of the
+ buffer for augmented codebook
+ construction */
+ float *cbVec /* (o) The construced codebook vector */
+ );
+
+ #endif
+
+A.12. createCB.c
+
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ createCB.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+
+
+Andersen, et al. Experimental [Page 99]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include <string.h>
+ #include <math.h>
+
+ /*----------------------------------------------------------------*
+ * Construct an additional codebook vector by filtering the
+ * initial codebook buffer. This vector is then used to expand
+ * the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+ void filteredCBvecs(
+ float *cbvectors, /* (o) Codebook vectors for the
+ higher section */
+ float *mem, /* (i) Buffer to create codebook
+ vector from */
+ int lMem /* (i) Length of buffer */
+ ){
+ int j, k;
+ float *pp, *pp1;
+ float tempbuff2[CB_MEML+CB_FILTERLEN];
+ float *pos;
+
+ memset(tempbuff2, 0, (CB_HALFFILTERLEN-1)*sizeof(float));
+ memcpy(&tempbuff2[CB_HALFFILTERLEN-1], mem, lMem*sizeof(float));
+ memset(&tempbuff2[lMem+CB_HALFFILTERLEN-1], 0,
+ (CB_HALFFILTERLEN+1)*sizeof(float));
+
+ /* Create codebook vector for higher section by filtering */
+
+ /* do filtering */
+ pos=cbvectors;
+ memset(pos, 0, lMem*sizeof(float));
+ for (k=0; k<lMem; k++) {
+ pp=&tempbuff2[k];
+ pp1=&cbfiltersTbl[CB_FILTERLEN-1];
+ for (j=0;j<CB_FILTERLEN;j++) {
+ (*pos)+=(*pp++)*(*pp1--);
+ }
+ pos++;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * Search the augmented part of the codebook to find the best
+ * measure.
+ *----------------------------------------------------------------*/
+
+
+
+
+Andersen, et al. Experimental [Page 100]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void searchAugmentedCB(
+ int low, /* (i) Start index for the search */
+ int high, /* (i) End index for the search */
+ int stage, /* (i) Current stage */
+ int startIndex, /* (i) Codebook index for the first
+ aug vector */
+ float *target, /* (i) Target vector for encoding */
+ float *buffer, /* (i) Pointer to the end of the buffer for
+ augmented codebook construction */
+ float *max_measure, /* (i/o) Currently maximum measure */
+ int *best_index,/* (o) Currently the best index */
+ float *gain, /* (o) Currently the best gain */
+ float *energy, /* (o) Energy of augmented codebook
+ vectors */
+ float *invenergy/* (o) Inv energy of augmented codebook
+ vectors */
+ ) {
+ int icount, ilow, j, tmpIndex;
+ float *pp, *ppo, *ppi, *ppe, crossDot, alfa;
+ float weighted, measure, nrjRecursive;
+ float ftmp;
+
+ /* Compute the energy for the first (low-5)
+ noninterpolated samples */
+ nrjRecursive = (float) 0.0;
+ pp = buffer - low + 1;
+ for (j=0; j<(low-5); j++) {
+ nrjRecursive += ( (*pp)*(*pp) );
+ pp++;
+ }
+ ppe = buffer - low;
+
+
+ for (icount=low; icount<=high; icount++) {
+
+ /* Index of the codebook vector used for retrieving
+ energy values */
+ tmpIndex = startIndex+icount-20;
+
+ ilow = icount-4;
+
+ /* Update the energy recursively to save complexity */
+ nrjRecursive = nrjRecursive + (*ppe)*(*ppe);
+ ppe--;
+ energy[tmpIndex] = nrjRecursive;
+
+ /* Compute cross dot product for the first (low-5)
+ samples */
+
+
+
+Andersen, et al. Experimental [Page 101]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ crossDot = (float) 0.0;
+ pp = buffer-icount;
+ for (j=0; j<ilow; j++) {
+ crossDot += target[j]*(*pp++);
+ }
+
+ /* interpolation */
+ alfa = (float) 0.2;
+ ppo = buffer-4;
+ ppi = buffer-icount-4;
+ for (j=ilow; j<icount; j++) {
+ weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
+ ppo++;
+ ppi++;
+ energy[tmpIndex] += weighted*weighted;
+ crossDot += target[j]*weighted;
+ alfa += (float)0.2;
+ }
+
+ /* Compute energy and cross dot product for the
+ remaining samples */
+ pp = buffer - icount;
+ for (j=icount; j<SUBL; j++) {
+ energy[tmpIndex] += (*pp)*(*pp);
+ crossDot += target[j]*(*pp++);
+ }
+
+ if (energy[tmpIndex]>0.0) {
+ invenergy[tmpIndex]=(float)1.0/(energy[tmpIndex]+EPS);
+ } else {
+ invenergy[tmpIndex] = (float) 0.0;
+ }
+
+ if (stage==0) {
+ measure = (float)-10000000.0;
+
+ if (crossDot > 0.0) {
+ measure = crossDot*crossDot*invenergy[tmpIndex];
+ }
+ }
+ else {
+ measure = crossDot*crossDot*invenergy[tmpIndex];
+ }
+
+ /* check if measure is better */
+ ftmp = crossDot*invenergy[tmpIndex];
+
+ if ((measure>*max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
+
+
+
+Andersen, et al. Experimental [Page 102]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ *best_index = tmpIndex;
+ *max_measure = measure;
+ *gain = ftmp;
+ }
+ }
+ }
+
+
+ /*----------------------------------------------------------------*
+ * Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+ void createAugmentedVec(
+ int index, /* (i) Index for the augmented vector
+ to be created */
+ float *buffer, /* (i) Pointer to the end of the buffer for
+ augmented codebook construction */
+ float *cbVec/* (o) The construced codebook vector */
+ ) {
+ int ilow, j;
+ float *pp, *ppo, *ppi, alfa, alfa1, weighted;
+
+ ilow = index-5;
+
+ /* copy the first noninterpolated part */
+
+ pp = buffer-index;
+ memcpy(cbVec,pp,sizeof(float)*index);
+
+ /* interpolation */
+
+ alfa1 = (float)0.2;
+ alfa = 0.0;
+ ppo = buffer-5;
+ ppi = buffer-index-5;
+ for (j=ilow; j<index; j++) {
+ weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
+ ppo++;
+ ppi++;
+ cbVec[j] = weighted;
+ alfa += alfa1;
+ }
+
+ /* copy the second noninterpolated part */
+
+ pp = buffer - index;
+ memcpy(cbVec+index,pp,sizeof(float)*(SUBL-index));
+
+
+
+Andersen, et al. Experimental [Page 103]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+A.13. doCPLC.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ doCPLC.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_DOLPC_H
+ #define __iLBC_DOLPC_H
+
+ void doThePLC(
+ float *PLCresidual, /* (o) concealed residual */
+ float *PLClpc, /* (o) concealed LP parameters */
+ int PLI, /* (i) packet loss indicator
+ 0 - no PL, 1 = PL */
+ float *decresidual, /* (i) decoded residual */
+ float *lpc, /* (i) decoded LPC (only used for no PL) */
+ int inlag, /* (i) pitch lag */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) decoder instance */
+ );
+
+ #endif
+
+A.14. doCPLC.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ doCPLC.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+ #include <stdio.h>
+
+
+
+Andersen, et al. Experimental [Page 104]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #include "iLBC_define.h"
+
+ /*----------------------------------------------------------------*
+ * Compute cross correlation and pitch gain for pitch prediction
+ * of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+ void compCorr(
+ float *cc, /* (o) cross correlation coefficient */
+ float *gc, /* (o) gain */
+ float *pm,
+ float *buffer, /* (i) signal buffer */
+ int lag, /* (i) pitch lag */
+ int bLen, /* (i) length of buffer */
+ int sRange /* (i) correlation search length */
+ ){
+ int i;
+ float ftmp1, ftmp2, ftmp3;
+
+ /* Guard against getting outside buffer */
+ if ((bLen-sRange-lag)<0) {
+ sRange=bLen-lag;
+ }
+
+ ftmp1 = 0.0;
+ ftmp2 = 0.0;
+ ftmp3 = 0.0;
+ for (i=0; i<sRange; i++) {
+ ftmp1 += buffer[bLen-sRange+i] *
+ buffer[bLen-sRange+i-lag];
+ ftmp2 += buffer[bLen-sRange+i-lag] *
+ buffer[bLen-sRange+i-lag];
+ ftmp3 += buffer[bLen-sRange+i] *
+ buffer[bLen-sRange+i];
+ }
+
+ if (ftmp2 > 0.0) {
+ *cc = ftmp1*ftmp1/ftmp2;
+ *gc = (float)fabs(ftmp1/ftmp2);
+ *pm=(float)fabs(ftmp1)/
+ ((float)sqrt(ftmp2)*(float)sqrt(ftmp3));
+ }
+ else {
+ *cc = 0.0;
+ *gc = 0.0;
+ *pm=0.0;
+ }
+ }
+
+
+
+Andersen, et al. Experimental [Page 105]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /*----------------------------------------------------------------*
+ * Packet loss concealment routine. Conceals a residual signal
+ * and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+ void doThePLC(
+ float *PLCresidual, /* (o) concealed residual */
+ float *PLClpc, /* (o) concealed LP parameters */
+ int PLI, /* (i) packet loss indicator
+ 0 - no PL, 1 = PL */
+ float *decresidual, /* (i) decoded residual */
+ float *lpc, /* (i) decoded LPC (only used for no PL) */
+ int inlag, /* (i) pitch lag */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) decoder instance */
+ ){
+ int lag=20, randlag;
+ float gain, maxcc;
+ float use_gain;
+ float gain_comp, maxcc_comp, per, max_per;
+ int i, pick, use_lag;
+ float ftmp, randvec[BLOCKL_MAX], pitchfact, energy;
+
+ /* Packet Loss */
+
+ if (PLI == 1) {
+
+ iLBCdec_inst->consPLICount += 1;
+
+ /* if previous frame not lost,
+ determine pitch pred. gain */
+
+ if (iLBCdec_inst->prevPLI != 1) {
+
+ /* Search around the previous lag to find the
+ best pitch period */
+
+ lag=inlag-3;
+ compCorr(&maxcc, &gain, &max_per,
+ iLBCdec_inst->prevResidual,
+ lag, iLBCdec_inst->blockl, 60);
+ for (i=inlag-2;i<=inlag+3;i++) {
+ compCorr(&maxcc_comp, &gain_comp, &per,
+ iLBCdec_inst->prevResidual,
+ i, iLBCdec_inst->blockl, 60);
+
+ if (maxcc_comp>maxcc) {
+ maxcc=maxcc_comp;
+
+
+
+Andersen, et al. Experimental [Page 106]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ gain=gain_comp;
+ lag=i;
+ max_per=per;
+ }
+ }
+
+ }
+
+ /* previous frame lost, use recorded lag and periodicity */
+
+ else {
+ lag=iLBCdec_inst->prevLag;
+ max_per=iLBCdec_inst->per;
+ }
+
+ /* downscaling */
+
+ use_gain=1.0;
+ if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320)
+ use_gain=(float)0.9;
+ else if (iLBCdec_inst->consPLICount*
+ iLBCdec_inst->blockl>2*320)
+ use_gain=(float)0.7;
+ else if (iLBCdec_inst->consPLICount*
+ iLBCdec_inst->blockl>3*320)
+ use_gain=(float)0.5;
+ else if (iLBCdec_inst->consPLICount*
+ iLBCdec_inst->blockl>4*320)
+ use_gain=(float)0.0;
+
+ /* mix noise and pitch repeatition */
+ ftmp=(float)sqrt(max_per);
+ if (ftmp>(float)0.7)
+ pitchfact=(float)1.0;
+ else if (ftmp>(float)0.4)
+ pitchfact=(ftmp-(float)0.4)/((float)0.7-(float)0.4);
+ else
+ pitchfact=0.0;
+
+
+ /* avoid repetition of same pitch cycle */
+ use_lag=lag;
+ if (lag<80) {
+ use_lag=2*lag;
+ }
+
+ /* compute concealed residual */
+
+
+
+
+Andersen, et al. Experimental [Page 107]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ energy = 0.0;
+ for (i=0; i<iLBCdec_inst->blockl; i++) {
+
+ /* noise component */
+
+ iLBCdec_inst->seed=(iLBCdec_inst->seed*69069L+1) &
+ (0x80000000L-1);
+ randlag = 50 + ((signed long) iLBCdec_inst->seed)%70;
+ pick = i - randlag;
+
+ if (pick < 0) {
+ randvec[i] =
+ iLBCdec_inst->prevResidual[
+ iLBCdec_inst->blockl+pick];
+ } else {
+ randvec[i] = randvec[pick];
+ }
+
+ /* pitch repeatition component */
+ pick = i - use_lag;
+
+ if (pick < 0) {
+ PLCresidual[i] =
+ iLBCdec_inst->prevResidual[
+ iLBCdec_inst->blockl+pick];
+ } else {
+ PLCresidual[i] = PLCresidual[pick];
+ }
+
+ /* mix random and periodicity component */
+
+ if (i<80)
+ PLCresidual[i] = use_gain*(pitchfact *
+ PLCresidual[i] +
+ ((float)1.0 - pitchfact) * randvec[i]);
+ else if (i<160)
+ PLCresidual[i] = (float)0.95*use_gain*(pitchfact *
+ PLCresidual[i] +
+ ((float)1.0 - pitchfact) * randvec[i]);
+ else
+ PLCresidual[i] = (float)0.9*use_gain*(pitchfact *
+ PLCresidual[i] +
+ ((float)1.0 - pitchfact) * randvec[i]);
+
+ energy += PLCresidual[i] * PLCresidual[i];
+ }
+
+ /* less than 30 dB, use only noise */
+
+
+
+Andersen, et al. Experimental [Page 108]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+
+ if (sqrt(energy/(float)iLBCdec_inst->blockl) < 30.0) {
+ gain=0.0;
+ for (i=0; i<iLBCdec_inst->blockl; i++) {
+ PLCresidual[i] = randvec[i];
+ }
+ }
+
+ /* use old LPC */
+
+ memcpy(PLClpc,iLBCdec_inst->prevLpc,
+ (LPC_FILTERORDER+1)*sizeof(float));
+
+ }
+
+ /* no packet loss, copy input */
+
+ else {
+ memcpy(PLCresidual, decresidual,
+ iLBCdec_inst->blockl*sizeof(float));
+ memcpy(PLClpc, lpc, (LPC_FILTERORDER+1)*sizeof(float));
+ iLBCdec_inst->consPLICount = 0;
+ }
+
+ /* update state */
+
+ if (PLI) {
+ iLBCdec_inst->prevLag = lag;
+ iLBCdec_inst->per=max_per;
+ }
+
+ iLBCdec_inst->prevPLI = PLI;
+ memcpy(iLBCdec_inst->prevLpc, PLClpc,
+ (LPC_FILTERORDER+1)*sizeof(float));
+ memcpy(iLBCdec_inst->prevResidual, PLCresidual,
+ iLBCdec_inst->blockl*sizeof(float));
+ }
+
+A.15. enhancer.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ enhancer.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+
+
+Andersen, et al. Experimental [Page 109]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ ******************************************************************/
+
+ #ifndef __ENHANCER_H
+ #define __ENHANCER_H
+
+ #include "iLBC_define.h"
+
+ float xCorrCoef(
+ float *target, /* (i) first array */
+ float *regressor, /* (i) second array */
+ int subl /* (i) dimension arrays */
+ );
+
+ int enhancerInterface(
+ float *out, /* (o) the enhanced recidual signal */
+ float *in, /* (i) the recidual signal to enhance */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) the decoder state structure */
+ );
+
+ #endif
+
+A.16. enhancer.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ enhancer.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include "filter.h"
+
+ /*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ * according to the squared-error criterion
+ *---------------------------------------------------------------*/
+
+ void NearestNeighbor(
+
+
+
+Andersen, et al. Experimental [Page 110]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ int *index, /* (o) index of array element closest
+ to value */
+ float *array, /* (i) data array */
+ float value,/* (i) value */
+ int arlength/* (i) dimension of data array */
+ ){
+ int i;
+ float bestcrit,crit;
+
+ crit=array[0]-value;
+ bestcrit=crit*crit;
+ *index=0;
+ for (i=1; i<arlength; i++) {
+ crit=array[i]-value;
+ crit=crit*crit;
+
+ if (crit<bestcrit) {
+ bestcrit=crit;
+ *index=i;
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+ void mycorr1(
+ float* corr, /* (o) correlation of seq1 and seq2 */
+ float* seq1, /* (i) first sequence */
+ int dim1, /* (i) dimension first seq1 */
+ const float *seq2, /* (i) second sequence */
+ int dim2 /* (i) dimension seq2 */
+ ){
+ int i,j;
+
+ for (i=0; i<=dim1-dim2; i++) {
+ corr[i]=0.0;
+ for (j=0; j<dim2; j++) {
+ corr[i] += seq1[i+j] * seq2[j];
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+
+
+
+Andersen, et al. Experimental [Page 111]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void enh_upsample(
+ float* useq1, /* (o) upsampled output sequence */
+ float* seq1,/* (i) unupsampled sequence */
+ int dim1, /* (i) dimension seq1 */
+ int hfl /* (i) polyphase filter length=2*hfl+1 */
+ ){
+ float *pu,*ps;
+ int i,j,k,q,filterlength,hfl2;
+ const float *polyp[ENH_UPS0]; /* pointers to
+ polyphase columns */
+ const float *pp;
+
+ /* define pointers for filter */
+
+ filterlength=2*hfl+1;
+
+ if ( filterlength > dim1 ) {
+ hfl2=(int) (dim1/2);
+ for (j=0; j<ENH_UPS0; j++) {
+ polyp[j]=polyphaserTbl+j*filterlength+hfl-hfl2;
+ }
+ hfl=hfl2;
+ filterlength=2*hfl+1;
+ }
+ else {
+ for (j=0; j<ENH_UPS0; j++) {
+ polyp[j]=polyphaserTbl+j*filterlength;
+ }
+ }
+
+ /* filtering: filter overhangs left side of sequence */
+
+ pu=useq1;
+ for (i=hfl; i<filterlength; i++) {
+ for (j=0; j<ENH_UPS0; j++) {
+ *pu=0.0;
+ pp = polyp[j];
+ ps = seq1+i;
+ for (k=0; k<=i; k++) {
+ *pu += *ps-- * *pp++;
+ }
+ pu++;
+ }
+ }
+
+ /* filtering: simple convolution=inner products */
+
+ for (i=filterlength; i<dim1; i++) {
+
+
+
+Andersen, et al. Experimental [Page 112]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (j=0;j<ENH_UPS0; j++){
+ *pu=0.0;
+ pp = polyp[j];
+ ps = seq1+i;
+ for (k=0; k<filterlength; k++) {
+ *pu += *ps-- * *pp++;
+ }
+ pu++;
+ }
+ }
+
+ /* filtering: filter overhangs right side of sequence */
+
+ for (q=1; q<=hfl; q++) {
+ for (j=0; j<ENH_UPS0; j++) {
+ *pu=0.0;
+ pp = polyp[j]+q;
+ ps = seq1+dim1-1;
+ for (k=0; k<filterlength-q; k++) {
+ *pu += *ps-- * *pp++;
+ }
+ pu++;
+ }
+ }
+ }
+
+
+ /*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+ void refiner(
+ float *seg, /* (o) segment array */
+ float *updStartPos, /* (o) updated start point */
+ float* idata, /* (i) original data buffer */
+ int idatal, /* (i) dimension of idata */
+ int centerStartPos, /* (i) beginning center segment */
+ float estSegPos,/* (i) estimated beginning other segment */
+ float period /* (i) estimated pitch period */
+ ){
+ int estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
+ int tloc,tloc2,i,st,en,fraction;
+ float vect[ENH_VECTL],corrVec[ENH_CORRDIM],maxv;
+ float corrVecUps[ENH_CORRDIM*ENH_UPS0];
+
+
+
+Andersen, et al. Experimental [Page 113]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* defining array bounds */
+
+ estSegPosRounded=(int)(estSegPos - 0.5);
+
+ searchSegStartPos=estSegPosRounded-ENH_SLOP;
+
+ if (searchSegStartPos<0) {
+ searchSegStartPos=0;
+ }
+ searchSegEndPos=estSegPosRounded+ENH_SLOP;
+
+ if (searchSegEndPos+ENH_BLOCKL >= idatal) {
+ searchSegEndPos=idatal-ENH_BLOCKL-1;
+ }
+ corrdim=searchSegEndPos-searchSegStartPos+1;
+
+ /* compute upsampled correlation (corr33) and find
+ location of max */
+
+ mycorr1(corrVec,idata+searchSegStartPos,
+ corrdim+ENH_BLOCKL-1,idata+centerStartPos,ENH_BLOCKL);
+ enh_upsample(corrVecUps,corrVec,corrdim,ENH_FL0);
+ tloc=0; maxv=corrVecUps[0];
+ for (i=1; i<ENH_UPS0*corrdim; i++) {
+
+ if (corrVecUps[i]>maxv) {
+ tloc=i;
+ maxv=corrVecUps[i];
+ }
+ }
+
+ /* make vector can be upsampled without ever running outside
+ bounds */
+
+ *updStartPos= (float)searchSegStartPos +
+ (float)tloc/(float)ENH_UPS0+(float)1.0;
+ tloc2=(int)(tloc/ENH_UPS0);
+
+ if (tloc>tloc2*ENH_UPS0) {
+ tloc2++;
+ }
+ st=searchSegStartPos+tloc2-ENH_FL0;
+
+ if (st<0) {
+ memset(vect,0,-st*sizeof(float));
+ memcpy(&vect[-st],idata, (ENH_VECTL+st)*sizeof(float));
+ }
+ else {
+
+
+
+Andersen, et al. Experimental [Page 114]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ en=st+ENH_VECTL;
+
+ if (en>idatal) {
+ memcpy(vect, &idata[st],
+ (ENH_VECTL-(en-idatal))*sizeof(float));
+ memset(&vect[ENH_VECTL-(en-idatal)], 0,
+ (en-idatal)*sizeof(float));
+ }
+ else {
+ memcpy(vect, &idata[st], ENH_VECTL*sizeof(float));
+ }
+ }
+ fraction=tloc2*ENH_UPS0-tloc;
+
+ /* compute the segment (this is actually a convolution) */
+
+ mycorr1(seg,vect,ENH_VECTL,polyphaserTbl+(2*ENH_FL0+1)*fraction,
+ 2*ENH_FL0+1);
+ }
+
+ /*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+ void smath(
+ float *odata, /* (o) smoothed output */
+ float *sseq,/* (i) said second sequence of waveforms */
+ int hl, /* (i) 2*hl+1 is sseq dimension */
+ float alpha0/* (i) max smoothing energy fraction */
+ ){
+ int i,k;
+ float w00,w10,w11,A,B,C,*psseq,err,errs;
+ float surround[BLOCKL_MAX]; /* shape contributed by other than
+ current */
+ float wt[2*ENH_HL+1]; /* waveform weighting to get
+ surround shape */
+ float denom;
+
+ /* create shape of contribution from all waveforms except the
+ current one */
+
+ for (i=1; i<=2*hl+1; i++) {
+ wt[i-1] = (float)0.5*(1 - (float)cos(2*PI*i/(2*hl+2)));
+ }
+ wt[hl]=0.0; /* for clarity, not used */
+ for (i=0; i<ENH_BLOCKL; i++) {
+ surround[i]=sseq[i]*wt[0];
+ }
+
+
+
+Andersen, et al. Experimental [Page 115]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (k=1; k<hl; k++) {
+ psseq=sseq+k*ENH_BLOCKL;
+ for(i=0;i<ENH_BLOCKL; i++) {
+ surround[i]+=psseq[i]*wt[k];
+ }
+ }
+ for (k=hl+1; k<=2*hl; k++) {
+ psseq=sseq+k*ENH_BLOCKL;
+ for(i=0;i<ENH_BLOCKL; i++) {
+ surround[i]+=psseq[i]*wt[k];
+ }
+ }
+
+ /* compute some inner products */
+
+ w00 = w10 = w11 = 0.0;
+ psseq=sseq+hl*ENH_BLOCKL; /* current block */
+ for (i=0; i<ENH_BLOCKL;i++) {
+ w00+=psseq[i]*psseq[i];
+ w11+=surround[i]*surround[i];
+ w10+=surround[i]*psseq[i];
+ }
+
+ if (fabs(w11) < 1.0) {
+ w11=1.0;
+ }
+ C = (float)sqrt( w00/w11);
+
+ /* first try enhancement without power-constraint */
+
+ errs=0.0;
+ psseq=sseq+hl*ENH_BLOCKL;
+ for (i=0; i<ENH_BLOCKL; i++) {
+ odata[i]=C*surround[i];
+ err=psseq[i]-odata[i];
+ errs+=err*err;
+ }
+
+ /* if constraint violated by first try, add constraint */
+
+ if (errs > alpha0 * w00) {
+ if ( w00 < 1) {
+ w00=1;
+ }
+ denom = (w11*w00-w10*w10)/(w00*w00);
+
+ if (denom > 0.0001) { /* eliminates numerical problems
+ for if smooth */
+
+
+
+Andersen, et al. Experimental [Page 116]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ A = (float)sqrt( (alpha0- alpha0*alpha0/4)/denom);
+ B = -alpha0/2 - A * w10/w00;
+ B = B+1;
+ }
+ else { /* essentially no difference between cycles;
+ smoothing not needed */
+ A= 0.0;
+ B= 1.0;
+ }
+
+ /* create smoothed sequence */
+
+ psseq=sseq+hl*ENH_BLOCKL;
+ for (i=0; i<ENH_BLOCKL; i++) {
+ odata[i]=A*surround[i]+B*psseq[i];
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+ void getsseq(
+ float *sseq, /* (o) the pitch-synchronous sequence */
+ float *idata, /* (i) original data */
+ int idatal, /* (i) dimension of data */
+ int centerStartPos, /* (i) where current block starts */
+ float *period, /* (i) rough-pitch-period array */
+ float *plocs, /* (i) where periods of period array
+ are taken */
+ int periodl, /* (i) dimension period array */
+ int hl /* (i) 2*hl+1 is the number of sequences */
+ ){
+ int i,centerEndPos,q;
+ float blockStartPos[2*ENH_HL+1];
+ int lagBlock[2*ENH_HL+1];
+ float plocs2[ENH_PLOCSL];
+ float *psseq;
+
+ centerEndPos=centerStartPos+ENH_BLOCKL-1;
+
+ /* present */
+
+ NearestNeighbor(lagBlock+hl,plocs,
+ (float)0.5*(centerStartPos+centerEndPos),periodl);
+
+ blockStartPos[hl]=(float)centerStartPos;
+
+
+
+Andersen, et al. Experimental [Page 117]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ psseq=sseq+ENH_BLOCKL*hl;
+ memcpy(psseq, idata+centerStartPos, ENH_BLOCKL*sizeof(float));
+
+ /* past */
+
+ for (q=hl-1; q>=0; q--) {
+ blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
+ NearestNeighbor(lagBlock+q,plocs,
+ blockStartPos[q]+
+ ENH_BLOCKL_HALF-period[lagBlock[q+1]], periodl);
+
+
+ if (blockStartPos[q]-ENH_OVERHANG>=0) {
+ refiner(sseq+q*ENH_BLOCKL, blockStartPos+q, idata,
+ idatal, centerStartPos, blockStartPos[q],
+ period[lagBlock[q+1]]);
+ } else {
+ psseq=sseq+q*ENH_BLOCKL;
+ memset(psseq, 0, ENH_BLOCKL*sizeof(float));
+ }
+ }
+
+ /* future */
+
+ for (i=0; i<periodl; i++) {
+ plocs2[i]=plocs[i]-period[i];
+ }
+ for (q=hl+1; q<=2*hl; q++) {
+ NearestNeighbor(lagBlock+q,plocs2,
+ blockStartPos[q-1]+ENH_BLOCKL_HALF,periodl);
+
+ blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
+ if (blockStartPos[q]+ENH_BLOCKL+ENH_OVERHANG<idatal) {
+ refiner(sseq+ENH_BLOCKL*q, blockStartPos+q, idata,
+ idatal, centerStartPos, blockStartPos[q],
+ period[lagBlock[q]]);
+ }
+ else {
+ psseq=sseq+q*ENH_BLOCKL;
+ memset(psseq, 0, ENH_BLOCKL*sizeof(float));
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+
+
+Andersen, et al. Experimental [Page 118]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void enhancer(
+ float *odata, /* (o) smoothed block, dimension blockl */
+ float *idata, /* (i) data buffer used for enhancing */
+ int idatal, /* (i) dimension idata */
+ int centerStartPos, /* (i) first sample current block
+ within idata */
+ float alpha0, /* (i) max correction-energy-fraction
+ (in [0,1]) */
+ float *period, /* (i) pitch period array */
+ float *plocs, /* (i) locations where period array
+ values valid */
+ int periodl /* (i) dimension of period and plocs */
+ ){
+ float sseq[(2*ENH_HL+1)*ENH_BLOCKL];
+
+ /* get said second sequence of segments */
+
+ getsseq(sseq,idata,idatal,centerStartPos,period,
+ plocs,periodl,ENH_HL);
+
+ /* compute the smoothed output from said second sequence */
+
+ smath(odata,sseq,ENH_HL,alpha0);
+
+ }
+
+ /*----------------------------------------------------------------*
+ * cross correlation
+ *---------------------------------------------------------------*/
+
+ float xCorrCoef(
+ float *target, /* (i) first array */
+ float *regressor, /* (i) second array */
+ int subl /* (i) dimension arrays */
+ ){
+ int i;
+ float ftmp1, ftmp2;
+
+ ftmp1 = 0.0;
+ ftmp2 = 0.0;
+ for (i=0; i<subl; i++) {
+ ftmp1 += target[i]*regressor[i];
+ ftmp2 += regressor[i]*regressor[i];
+ }
+
+ if (ftmp1 > 0.0) {
+ return (float)(ftmp1*ftmp1/ftmp2);
+ }
+
+
+
+Andersen, et al. Experimental [Page 119]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ else {
+ return (float)0.0;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+ int enhancerInterface(
+ float *out, /* (o) enhanced signal */
+ float *in, /* (i) unenhanced signal */
+ iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
+ ){
+ float *enh_buf, *enh_period;
+ int iblock, isample;
+ int lag=0, ilag, i, ioffset;
+ float cc, maxcc;
+ float ftmp1, ftmp2;
+ float *inPtr, *enh_bufPtr1, *enh_bufPtr2;
+ float plc_pred[ENH_BLOCKL];
+
+ float lpState[6], downsampled[(ENH_NBLOCKS*ENH_BLOCKL+120)/2];
+ int inLen=ENH_NBLOCKS*ENH_BLOCKL+120;
+ int start, plc_blockl, inlag;
+
+ enh_buf=iLBCdec_inst->enh_buf;
+ enh_period=iLBCdec_inst->enh_period;
+
+ memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
+ (ENH_BUFL-iLBCdec_inst->blockl)*sizeof(float));
+
+ memcpy(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
+ iLBCdec_inst->blockl*sizeof(float));
+
+ if (iLBCdec_inst->mode==30)
+ plc_blockl=ENH_BLOCKL;
+ else
+ plc_blockl=40;
+
+ /* when 20 ms frame, move processing one block */
+ ioffset=0;
+ if (iLBCdec_inst->mode==20) ioffset=1;
+
+ i=3-ioffset;
+ memmove(enh_period, &enh_period[i],
+ (ENH_NBLOCKS_TOT-i)*sizeof(float));
+
+
+
+
+Andersen, et al. Experimental [Page 120]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* Set state information to the 6 samples right before
+ the samples to be downsampled. */
+
+ memcpy(lpState,
+ enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-126,
+ 6*sizeof(float));
+
+ /* Down sample a factor 2 to save computations */
+
+ DownSample(enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-120,
+ lpFilt_coefsTbl, inLen-ioffset*ENH_BLOCKL,
+ lpState, downsampled);
+
+ /* Estimate the pitch in the down sampled domain. */
+ for (iblock = 0; iblock<ENH_NBLOCKS-ioffset; iblock++) {
+
+ lag = 10;
+ maxcc = xCorrCoef(downsampled+60+iblock*
+ ENH_BLOCKL_HALF, downsampled+60+iblock*
+ ENH_BLOCKL_HALF-lag, ENH_BLOCKL_HALF);
+ for (ilag=11; ilag<60; ilag++) {
+ cc = xCorrCoef(downsampled+60+iblock*
+ ENH_BLOCKL_HALF, downsampled+60+iblock*
+ ENH_BLOCKL_HALF-ilag, ENH_BLOCKL_HALF);
+
+ if (cc > maxcc) {
+ maxcc = cc;
+ lag = ilag;
+ }
+ }
+
+ /* Store the estimated lag in the non-downsampled domain */
+ enh_period[iblock+ENH_NBLOCKS_EXTRA+ioffset] = (float)lag*2;
+
+
+ }
+
+
+ /* PLC was performed on the previous packet */
+ if (iLBCdec_inst->prev_enh_pl==1) {
+
+ inlag=(int)enh_period[ENH_NBLOCKS_EXTRA+ioffset];
+
+ lag = inlag-1;
+ maxcc = xCorrCoef(in, in+lag, plc_blockl);
+ for (ilag=inlag; ilag<=inlag+1; ilag++) {
+ cc = xCorrCoef(in, in+ilag, plc_blockl);
+
+
+
+
+Andersen, et al. Experimental [Page 121]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ if (cc > maxcc) {
+ maxcc = cc;
+ lag = ilag;
+ }
+ }
+
+ enh_period[ENH_NBLOCKS_EXTRA+ioffset-1]=(float)lag;
+
+ /* compute new concealed residual for the old lookahead,
+ mix the forward PLC with a backward PLC from
+ the new frame */
+
+ inPtr=&in[lag-1];
+
+ enh_bufPtr1=&plc_pred[plc_blockl-1];
+
+ if (lag>plc_blockl) {
+ start=plc_blockl;
+ } else {
+ start=lag;
+ }
+
+ for (isample = start; isample>0; isample--) {
+ *enh_bufPtr1-- = *inPtr--;
+ }
+
+ enh_bufPtr2=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+ for (isample = (plc_blockl-1-lag); isample>=0; isample--) {
+ *enh_bufPtr1-- = *enh_bufPtr2--;
+ }
+
+ /* limit energy change */
+ ftmp2=0.0;
+ ftmp1=0.0;
+ for (i=0;i<plc_blockl;i++) {
+ ftmp2+=enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i]*
+ enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i];
+ ftmp1+=plc_pred[i]*plc_pred[i];
+ }
+ ftmp1=(float)sqrt(ftmp1/(float)plc_blockl);
+ ftmp2=(float)sqrt(ftmp2/(float)plc_blockl);
+ if (ftmp1>(float)2.0*ftmp2 && ftmp1>0.0) {
+ for (i=0;i<plc_blockl-10;i++) {
+ plc_pred[i]*=(float)2.0*ftmp2/ftmp1;
+ }
+ for (i=plc_blockl-10;i<plc_blockl;i++) {
+ plc_pred[i]*=(float)(i-plc_blockl+10)*
+ ((float)1.0-(float)2.0*ftmp2/ftmp1)/(float)(10)+
+
+
+
+Andersen, et al. Experimental [Page 122]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ (float)2.0*ftmp2/ftmp1;
+ }
+ }
+
+ enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+ for (i=0; i<plc_blockl; i++) {
+ ftmp1 = (float) (i+1) / (float) (plc_blockl+1);
+ *enh_bufPtr1 *= ftmp1;
+ *enh_bufPtr1 += ((float)1.0-ftmp1)*
+ plc_pred[plc_blockl-1-i];
+ enh_bufPtr1--;
+ }
+ }
+
+ if (iLBCdec_inst->mode==20) {
+ /* Enhancer with 40 samples delay */
+ for (iblock = 0; iblock<2; iblock++) {
+ enhancer(out+iblock*ENH_BLOCKL, enh_buf,
+ ENH_BUFL, (5+iblock)*ENH_BLOCKL+40,
+ ENH_ALPHA0, enh_period, enh_plocsTbl,
+ ENH_NBLOCKS_TOT);
+ }
+ } else if (iLBCdec_inst->mode==30) {
+ /* Enhancer with 80 samples delay */
+ for (iblock = 0; iblock<3; iblock++) {
+ enhancer(out+iblock*ENH_BLOCKL, enh_buf,
+ ENH_BUFL, (4+iblock)*ENH_BLOCKL,
+ ENH_ALPHA0, enh_period, enh_plocsTbl,
+ ENH_NBLOCKS_TOT);
+ }
+ }
+
+ return (lag*2);
+ }
+
+A.17. filter.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ filter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+
+
+
+Andersen, et al. Experimental [Page 123]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #ifndef __iLBC_FILTER_H
+ #define __iLBC_FILTER_H
+
+ void AllPoleFilter(
+ float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
+ InOut[-1] contain the state of the
+ filter (delayed samples). InOut[0] to
+ InOut[lengthInOut-1] contain the filter
+ input, on en exit InOut[-orderCoef] to
+ InOut[-1] is unchanged and InOut[0] to
+ InOut[lengthInOut-1] contain filtered
+ samples */
+ float *Coef,/* (i) filter coefficients, Coef[0] is assumed
+ to be 1.0 */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef /* (i) number of filter coefficients */
+ );
+
+ void AllZeroFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain
+ filter input samples */
+ float *Coef,/* (i) filter coefficients (Coef[0] is assumed
+ to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain the filter state, on exit Out[0]
+ to Out[lengthInOut-1] contain filtered
+ samples */
+ );
+
+ void ZeroPoleFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain filter
+ input samples In[-orderCoef] to In[-1]
+ contain state of all-zero section */
+ float *ZeroCoef,/* (i) filter coefficients for all-zero
+ section (ZeroCoef[0] is assumed to
+ be 1.0) */
+ float *PoleCoef,/* (i) filter coefficients for all-pole section
+ (ZeroCoef[0] is assumed to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain state of all-pole section. On
+ exit Out[0] to Out[lengthInOut-1]
+ contain filtered samples */
+ );
+
+
+
+
+Andersen, et al. Experimental [Page 124]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void DownSample (
+ float *In, /* (i) input samples */
+ float *Coef, /* (i) filter coefficients */
+ int lengthIn, /* (i) number of input samples */
+ float *state, /* (i) filter state */
+ float *Out /* (o) downsampled output */
+ );
+
+ #endif
+
+A.18. filter.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ filter.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "iLBC_define.h"
+
+ /*----------------------------------------------------------------*
+ * all-pole filter
+ *---------------------------------------------------------------*/
+
+ void AllPoleFilter(
+ float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
+ InOut[-1] contain the state of the
+ filter (delayed samples). InOut[0] to
+ InOut[lengthInOut-1] contain the filter
+ input, on en exit InOut[-orderCoef] to
+ InOut[-1] is unchanged and InOut[0] to
+ InOut[lengthInOut-1] contain filtered
+ samples */
+ float *Coef,/* (i) filter coefficients, Coef[0] is assumed
+ to be 1.0 */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef /* (i) number of filter coefficients */
+ ){
+ int n,k;
+
+ for(n=0;n<lengthInOut;n++){
+ for(k=1;k<=orderCoef;k++){
+ *InOut -= Coef[k]*InOut[-k];
+
+
+
+Andersen, et al. Experimental [Page 125]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ InOut++;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * all-zero filter
+ *---------------------------------------------------------------*/
+
+ void AllZeroFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain
+ filter input samples */
+ float *Coef,/* (i) filter coefficients (Coef[0] is assumed
+ to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain the filter state, on exit Out[0]
+ to Out[lengthInOut-1] contain filtered
+ samples */
+ ){
+ int n,k;
+
+ for(n=0;n<lengthInOut;n++){
+ *Out = Coef[0]*In[0];
+ for(k=1;k<=orderCoef;k++){
+ *Out += Coef[k]*In[-k];
+ }
+ Out++;
+ In++;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * pole-zero filter
+ *---------------------------------------------------------------*/
+
+ void ZeroPoleFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain
+ filter input samples In[-orderCoef] to
+ In[-1] contain state of all-zero
+ section */
+ float *ZeroCoef,/* (i) filter coefficients for all-zero
+ section (ZeroCoef[0] is assumed to
+ be 1.0) */
+ float *PoleCoef,/* (i) filter coefficients for all-pole section
+ (ZeroCoef[0] is assumed to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+
+
+
+Andersen, et al. Experimental [Page 126]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain state of all-pole section. On
+ exit Out[0] to Out[lengthInOut-1]
+ contain filtered samples */
+ ){
+ AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
+ AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
+ }
+
+ /*----------------------------------------------------------------*
+ * downsample (LP filter and decimation)
+ *---------------------------------------------------------------*/
+
+ void DownSample (
+ float *In, /* (i) input samples */
+ float *Coef, /* (i) filter coefficients */
+ int lengthIn, /* (i) number of input samples */
+ float *state, /* (i) filter state */
+ float *Out /* (o) downsampled output */
+ ){
+ float o;
+ float *Out_ptr = Out;
+ float *Coef_ptr, *In_ptr;
+ float *state_ptr;
+ int i, j, stop;
+
+ /* LP filter and decimate at the same time */
+
+ for (i = DELAY_DS; i < lengthIn; i+=FACTOR_DS)
+ {
+ Coef_ptr = &Coef[0];
+ In_ptr = &In[i];
+ state_ptr = &state[FILTERORDER_DS-2];
+
+ o = (float)0.0;
+
+ stop = (i < FILTERORDER_DS) ? i + 1 : FILTERORDER_DS;
+
+ for (j = 0; j < stop; j++)
+ {
+ o += *Coef_ptr++ * (*In_ptr--);
+ }
+ for (j = i + 1; j < FILTERORDER_DS; j++)
+ {
+ o += *Coef_ptr++ * (*state_ptr--);
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 127]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ *Out_ptr++ = o;
+ }
+
+ /* Get the last part (use zeros as input for the future) */
+
+ for (i=(lengthIn+FACTOR_DS); i<(lengthIn+DELAY_DS);
+ i+=FACTOR_DS) {
+
+ o=(float)0.0;
+
+ if (i<lengthIn) {
+ Coef_ptr = &Coef[0];
+ In_ptr = &In[i];
+ for (j=0; j<FILTERORDER_DS; j++) {
+ o += *Coef_ptr++ * (*Out_ptr--);
+ }
+ } else {
+ Coef_ptr = &Coef[i-lengthIn];
+ In_ptr = &In[lengthIn-1];
+ for (j=0; j<FILTERORDER_DS-(i-lengthIn); j++) {
+ o += *Coef_ptr++ * (*In_ptr--);
+ }
+ }
+ *Out_ptr++ = o;
+ }
+ }
+
+A.19. FrameClassify.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ FrameClassify.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_FRAMECLASSIFY_H
+ #define __iLBC_FRAMECLASSIFY_H
+
+ int FrameClassify( /* index to the max-energy sub-frame */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) the encoder state structure */
+ float *residual /* (i) lpc residual signal */
+ );
+
+
+
+Andersen, et al. Experimental [Page 128]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #endif
+
+A.20. FrameClassify.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ FrameClassify.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "iLBC_define.h"
+
+ /*---------------------------------------------------------------*
+ * Classification of subframes to localize start state
+ *--------------------------------------------------------------*/
+
+ int FrameClassify( /* index to the max-energy sub-frame */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) the encoder state structure */
+ float *residual /* (i) lpc residual signal */
+ ) {
+ float max_ssqEn, fssqEn[NSUB_MAX], bssqEn[NSUB_MAX], *pp;
+ int n, l, max_ssqEn_n;
+ const float ssqEn_win[NSUB_MAX-1]={(float)0.8,(float)0.9,
+ (float)1.0,(float)0.9,(float)0.8};
+ const float sampEn_win[5]={(float)1.0/(float)6.0,
+ (float)2.0/(float)6.0, (float)3.0/(float)6.0,
+ (float)4.0/(float)6.0, (float)5.0/(float)6.0};
+
+ /* init the front and back energies to zero */
+
+ memset(fssqEn, 0, NSUB_MAX*sizeof(float));
+ memset(bssqEn, 0, NSUB_MAX*sizeof(float));
+
+ /* Calculate front of first seqence */
+
+ n=0;
+ pp=residual;
+ for (l=0; l<5; l++) {
+ fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
+ pp++;
+ }
+ for (l=5; l<SUBL; l++) {
+
+
+
+Andersen, et al. Experimental [Page 129]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ fssqEn[n] += (*pp) * (*pp);
+ pp++;
+ }
+
+ /* Calculate front and back of all middle sequences */
+
+ for (n=1; n<iLBCenc_inst->nsub-1; n++) {
+ pp=residual+n*SUBL;
+ for (l=0; l<5; l++) {
+ fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
+ bssqEn[n] += (*pp) * (*pp);
+ pp++;
+ }
+ for (l=5; l<SUBL-5; l++) {
+ fssqEn[n] += (*pp) * (*pp);
+ bssqEn[n] += (*pp) * (*pp);
+ pp++;
+ }
+ for (l=SUBL-5; l<SUBL; l++) {
+ fssqEn[n] += (*pp) * (*pp);
+ bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
+ pp++;
+ }
+ }
+
+ /* Calculate back of last seqence */
+
+ n=iLBCenc_inst->nsub-1;
+ pp=residual+n*SUBL;
+ for (l=0; l<SUBL-5; l++) {
+ bssqEn[n] += (*pp) * (*pp);
+ pp++;
+ }
+ for (l=SUBL-5; l<SUBL; l++) {
+ bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
+ pp++;
+ }
+
+ /* find the index to the weighted 80 sample with
+ most energy */
+
+ if (iLBCenc_inst->mode==20) l=1;
+ else l=0;
+
+ max_ssqEn=(fssqEn[0]+bssqEn[1])*ssqEn_win[l];
+ max_ssqEn_n=1;
+ for (n=2; n<iLBCenc_inst->nsub; n++) {
+
+
+
+
+Andersen, et al. Experimental [Page 130]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ l++;
+ if ((fssqEn[n-1]+bssqEn[n])*ssqEn_win[l] > max_ssqEn) {
+ max_ssqEn=(fssqEn[n-1]+bssqEn[n]) *
+ ssqEn_win[l];
+ max_ssqEn_n=n;
+ }
+ }
+
+ return max_ssqEn_n;
+ }
+
+A.21. gainquant.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ gainquant.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_GAINQUANT_H
+ #define __iLBC_GAINQUANT_H
+
+ float gainquant(/* (o) quantized gain value */
+ float in, /* (i) gain value */
+ float maxIn,/* (i) maximum of gain value */
+ int cblen, /* (i) number of quantization indices */
+ int *index /* (o) quantization index */
+ );
+
+ float gaindequant( /* (o) quantized gain value */
+ int index, /* (i) quantization index */
+ float maxIn,/* (i) maximum of unquantized gain */
+ int cblen /* (i) number of quantization indices */
+ );
+
+ #endif
+
+A.22. gainquant.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+
+
+
+Andersen, et al. Experimental [Page 131]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ gainquant.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <string.h>
+ #include <math.h>
+ #include "constants.h"
+ #include "filter.h"
+
+ /*----------------------------------------------------------------*
+ * quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+ float gainquant(/* (o) quantized gain value */
+ float in, /* (i) gain value */
+ float maxIn,/* (i) maximum of gain value */
+ int cblen, /* (i) number of quantization indices */
+ int *index /* (o) quantization index */
+ ){
+ int i, tindex;
+ float minmeasure,measure, *cb, scale;
+
+ /* ensure a lower bound on the scaling factor */
+
+ scale=maxIn;
+
+ if (scale<0.1) {
+ scale=(float)0.1;
+ }
+
+ /* select the quantization table */
+
+ if (cblen == 8) {
+ cb = gain_sq3Tbl;
+ } else if (cblen == 16) {
+ cb = gain_sq4Tbl;
+ } else {
+ cb = gain_sq5Tbl;
+ }
+
+ /* select the best index in the quantization table */
+
+ minmeasure=10000000.0;
+ tindex=0;
+ for (i=0; i<cblen; i++) {
+
+
+
+Andersen, et al. Experimental [Page 132]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ measure=(in-scale*cb[i])*(in-scale*cb[i]);
+
+ if (measure<minmeasure) {
+ tindex=i;
+ minmeasure=measure;
+ }
+ }
+ *index=tindex;
+
+ /* return the quantized value */
+
+ return scale*cb[tindex];
+ }
+
+ /*----------------------------------------------------------------*
+ * decoder for quantized gains in the gain-shape coding of
+ * residual
+ *---------------------------------------------------------------*/
+
+ float gaindequant( /* (o) quantized gain value */
+ int index, /* (i) quantization index */
+ float maxIn,/* (i) maximum of unquantized gain */
+ int cblen /* (i) number of quantization indices */
+ ){
+ float scale;
+
+ /* obtain correct scale factor */
+
+ scale=(float)fabs(maxIn);
+
+ if (scale<0.1) {
+ scale=(float)0.1;
+ }
+
+ /* select the quantization table and return the decoded value */
+
+ if (cblen==8) {
+ return scale*gain_sq3Tbl[index];
+ } else if (cblen==16) {
+ return scale*gain_sq4Tbl[index];
+ }
+ else if (cblen==32) {
+ return scale*gain_sq5Tbl[index];
+ }
+
+ return 0.0;
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 133]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.23. getCBvec.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ getCBvec.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_GETCBVEC_H
+ #define __iLBC_GETCBVEC_H
+
+ void getCBvec(
+ float *cbvec, /* (o) Constructed codebook vector */
+ float *mem, /* (i) Codebook buffer */
+ int index, /* (i) Codebook index */
+ int lMem, /* (i) Length of codebook buffer */
+ int cbveclen/* (i) Codebook vector length */
+ );
+
+ #endif
+
+A.24. getCBvec.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ getCBvec.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include <string.h>
+
+ /*----------------------------------------------------------------*
+ * Construct codebook vector for given index.
+ *---------------------------------------------------------------*/
+
+ void getCBvec(
+
+
+
+Andersen, et al. Experimental [Page 134]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *cbvec, /* (o) Constructed codebook vector */
+ float *mem, /* (i) Codebook buffer */
+ int index, /* (i) Codebook index */
+ int lMem, /* (i) Length of codebook buffer */
+ int cbveclen/* (i) Codebook vector length */
+ ){
+ int j, k, n, memInd, sFilt;
+ float tmpbuf[CB_MEML];
+ int base_size;
+ int ilow, ihigh;
+ float alfa, alfa1;
+
+ /* Determine size of codebook sections */
+
+ base_size=lMem-cbveclen+1;
+
+ if (cbveclen==SUBL) {
+ base_size+=cbveclen/2;
+ }
+
+ /* No filter -> First codebook section */
+
+ if (index<lMem-cbveclen+1) {
+
+ /* first non-interpolated vectors */
+
+ k=index+cbveclen;
+ /* get vector */
+ memcpy(cbvec, mem+lMem-k, cbveclen*sizeof(float));
+
+ } else if (index < base_size) {
+
+ k=2*(index-(lMem-cbveclen+1))+cbveclen;
+
+ ihigh=k/2;
+ ilow=ihigh-5;
+
+ /* Copy first noninterpolated part */
+
+ memcpy(cbvec, mem+lMem-k/2, ilow*sizeof(float));
+
+ /* interpolation */
+
+ alfa1=(float)0.2;
+ alfa=0.0;
+ for (j=ilow; j<ihigh; j++) {
+ cbvec[j]=((float)1.0-alfa)*mem[lMem-k/2+j]+
+ alfa*mem[lMem-k+j];
+
+
+
+Andersen, et al. Experimental [Page 135]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ alfa+=alfa1;
+ }
+
+ /* Copy second noninterpolated part */
+
+ memcpy(cbvec+ihigh, mem+lMem-k+ihigh,
+ (cbveclen-ihigh)*sizeof(float));
+
+ }
+
+ /* Higher codebook section based on filtering */
+
+ else {
+
+ /* first non-interpolated vectors */
+
+ if (index-base_size<lMem-cbveclen+1) {
+ float tempbuff2[CB_MEML+CB_FILTERLEN+1];
+ float *pos;
+ float *pp, *pp1;
+
+ memset(tempbuff2, 0,
+ CB_HALFFILTERLEN*sizeof(float));
+ memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
+ lMem*sizeof(float));
+ memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
+ (CB_HALFFILTERLEN+1)*sizeof(float));
+
+ k=index-base_size+cbveclen;
+ sFilt=lMem-k;
+ memInd=sFilt+1-CB_HALFFILTERLEN;
+
+ /* do filtering */
+ pos=cbvec;
+ memset(pos, 0, cbveclen*sizeof(float));
+ for (n=0; n<cbveclen; n++) {
+ pp=&tempbuff2[memInd+n+CB_HALFFILTERLEN];
+ pp1=&cbfiltersTbl[CB_FILTERLEN-1];
+ for (j=0; j<CB_FILTERLEN; j++) {
+ (*pos)+=(*pp++)*(*pp1--);
+ }
+ pos++;
+ }
+ }
+
+ /* interpolated vectors */
+
+ else {
+
+
+
+Andersen, et al. Experimental [Page 136]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float tempbuff2[CB_MEML+CB_FILTERLEN+1];
+
+ float *pos;
+ float *pp, *pp1;
+ int i;
+
+ memset(tempbuff2, 0,
+ CB_HALFFILTERLEN*sizeof(float));
+ memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
+ lMem*sizeof(float));
+ memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
+ (CB_HALFFILTERLEN+1)*sizeof(float));
+
+ k=2*(index-base_size-
+ (lMem-cbveclen+1))+cbveclen;
+ sFilt=lMem-k;
+ memInd=sFilt+1-CB_HALFFILTERLEN;
+
+ /* do filtering */
+ pos=&tmpbuf[sFilt];
+ memset(pos, 0, k*sizeof(float));
+ for (i=0; i<k; i++) {
+ pp=&tempbuff2[memInd+i+CB_HALFFILTERLEN];
+ pp1=&cbfiltersTbl[CB_FILTERLEN-1];
+ for (j=0; j<CB_FILTERLEN; j++) {
+ (*pos)+=(*pp++)*(*pp1--);
+ }
+ pos++;
+ }
+
+ ihigh=k/2;
+ ilow=ihigh-5;
+
+ /* Copy first noninterpolated part */
+
+ memcpy(cbvec, tmpbuf+lMem-k/2,
+ ilow*sizeof(float));
+
+ /* interpolation */
+
+ alfa1=(float)0.2;
+ alfa=0.0;
+ for (j=ilow; j<ihigh; j++) {
+ cbvec[j]=((float)1.0-alfa)*
+ tmpbuf[lMem-k/2+j]+alfa*tmpbuf[lMem-k+j];
+ alfa+=alfa1;
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 137]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* Copy second noninterpolated part */
+
+ memcpy(cbvec+ihigh, tmpbuf+lMem-k+ihigh,
+ (cbveclen-ihigh)*sizeof(float));
+ }
+ }
+ }
+
+A.25. helpfun.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ helpfun.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HELPFUN_H
+ #define __iLBC_HELPFUN_H
+
+ void autocorr(
+ float *r, /* (o) autocorrelation vector */
+ const float *x, /* (i) data vector */
+ int N, /* (i) length of data vector */
+ int order /* largest lag for calculated
+ autocorrelations */
+ );
+
+ void window(
+ float *z, /* (o) the windowed data */
+ const float *x, /* (i) the original data vector */
+ const float *y, /* (i) the window */
+ int N /* (i) length of all vectors */
+ );
+
+ void levdurb(
+ float *a, /* (o) lpc coefficient vector starting
+ with 1.0 */
+ float *k, /* (o) reflection coefficients */
+ float *r, /* (i) autocorrelation vector */
+ int order /* (i) order of lpc filter */
+ );
+
+ void interpolate(
+
+
+
+Andersen, et al. Experimental [Page 138]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *out, /* (o) the interpolated vector */
+ float *in1, /* (i) the first vector for the
+ interpolation */
+ float *in2, /* (i) the second vector for the
+ interpolation */
+ float coef, /* (i) interpolation weights */
+ int length /* (i) length of all vectors */
+ );
+
+ void bwexpand(
+ float *out, /* (o) the bandwidth expanded lpc
+ coefficients */
+ float *in, /* (i) the lpc coefficients before bandwidth
+ expansion */
+ float coef, /* (i) the bandwidth expansion factor */
+ int length /* (i) the length of lpc coefficient vectors */
+ );
+
+ void vq(
+ float *Xq, /* (o) the quantized vector */
+ int *index, /* (o) the quantization index */
+ const float *CB,/* (i) the vector quantization codebook */
+ float *X, /* (i) the vector to quantize */
+ int n_cb, /* (i) the number of vectors in the codebook */
+ int dim /* (i) the dimension of all vectors */
+ );
+
+ void SplitVQ(
+ float *qX, /* (o) the quantized vector */
+ int *index, /* (o) a vector of indexes for all vector
+ codebooks in the split */
+ float *X, /* (i) the vector to quantize */
+ const float *CB,/* (i) the quantizer codebook */
+ int nsplit, /* the number of vector splits */
+ const int *dim, /* the dimension of X and qX */
+ const int *cbsize /* the number of vectors in the codebook */
+ );
+
+
+ void sort_sq(
+ float *xq, /* (o) the quantized value */
+ int *index, /* (o) the quantization index */
+ float x, /* (i) the value to quantize */
+ const float *cb,/* (i) the quantization codebook */
+ int cb_size /* (i) the size of the quantization codebook */
+ );
+
+ int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
+
+
+
+Andersen, et al. Experimental [Page 139]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ nonstable ones */
+ float *lsf, /* (i) a table of lsf vectors */
+ int dim, /* (i) the dimension of each lsf vector */
+ int NoAn /* (i) the number of lsf vectors in the
+ table */
+ );
+
+ #endif
+
+A.26. helpfun.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ helpfun.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+
+ /*----------------------------------------------------------------*
+ * calculation of auto correlation
+ *---------------------------------------------------------------*/
+
+ void autocorr(
+ float *r, /* (o) autocorrelation vector */
+ const float *x, /* (i) data vector */
+ int N, /* (i) length of data vector */
+ int order /* largest lag for calculated
+ autocorrelations */
+ ){
+ int lag, n;
+ float sum;
+
+ for (lag = 0; lag <= order; lag++) {
+ sum = 0;
+ for (n = 0; n < N - lag; n++) {
+ sum += x[n] * x[n+lag];
+ }
+ r[lag] = sum;
+ }
+
+
+
+Andersen, et al. Experimental [Page 140]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+ /*----------------------------------------------------------------*
+ * window multiplication
+ *---------------------------------------------------------------*/
+
+ void window(
+ float *z, /* (o) the windowed data */
+ const float *x, /* (i) the original data vector */
+ const float *y, /* (i) the window */
+ int N /* (i) length of all vectors */
+ ){
+ int i;
+
+ for (i = 0; i < N; i++) {
+ z[i] = x[i] * y[i];
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * levinson-durbin solution for lpc coefficients
+ *---------------------------------------------------------------*/
+
+ void levdurb(
+ float *a, /* (o) lpc coefficient vector starting
+ with 1.0 */
+ float *k, /* (o) reflection coefficients */
+ float *r, /* (i) autocorrelation vector */
+ int order /* (i) order of lpc filter */
+ ){
+ float sum, alpha;
+ int m, m_h, i;
+
+ a[0] = 1.0;
+
+ if (r[0] < EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
+ for (i = 0; i < order; i++) {
+ k[i] = 0;
+ a[i+1] = 0;
+ }
+ } else {
+ a[1] = k[0] = -r[1]/r[0];
+ alpha = r[0] + r[1] * k[0];
+ for (m = 1; m < order; m++){
+ sum = r[m + 1];
+ for (i = 0; i < m; i++){
+ sum += a[i+1] * r[m - i];
+ }
+
+
+
+Andersen, et al. Experimental [Page 141]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ k[m] = -sum / alpha;
+ alpha += k[m] * sum;
+ m_h = (m + 1) >> 1;
+ for (i = 0; i < m_h; i++){
+ sum = a[i+1] + k[m] * a[m - i];
+ a[m - i] += k[m] * a[i+1];
+ a[i+1] = sum;
+ }
+ a[m+1] = k[m];
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * interpolation between vectors
+ *---------------------------------------------------------------*/
+
+ void interpolate(
+ float *out, /* (o) the interpolated vector */
+ float *in1, /* (i) the first vector for the
+ interpolation */
+ float *in2, /* (i) the second vector for the
+ interpolation */
+ float coef, /* (i) interpolation weights */
+ int length /* (i) length of all vectors */
+ ){
+ int i;
+ float invcoef;
+
+ invcoef = (float)1.0 - coef;
+ for (i = 0; i < length; i++) {
+ out[i] = coef * in1[i] + invcoef * in2[i];
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+ void bwexpand(
+ float *out, /* (o) the bandwidth expanded lpc
+ coefficients */
+ float *in, /* (i) the lpc coefficients before bandwidth
+ expansion */
+ float coef, /* (i) the bandwidth expansion factor */
+ int length /* (i) the length of lpc coefficient vectors */
+ ){
+ int i;
+
+
+
+Andersen, et al. Experimental [Page 142]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float chirp;
+
+ chirp = coef;
+
+ out[0] = in[0];
+ for (i = 1; i < length; i++) {
+ out[i] = chirp * in[i];
+ chirp *= coef;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * vector quantization
+ *---------------------------------------------------------------*/
+
+ void vq(
+ float *Xq, /* (o) the quantized vector */
+ int *index, /* (o) the quantization index */
+ const float *CB,/* (i) the vector quantization codebook */
+ float *X, /* (i) the vector to quantize */
+ int n_cb, /* (i) the number of vectors in the codebook */
+ int dim /* (i) the dimension of all vectors */
+ ){
+ int i, j;
+ int pos, minindex;
+ float dist, tmp, mindist;
+
+ pos = 0;
+ mindist = FLOAT_MAX;
+ minindex = 0;
+ for (j = 0; j < n_cb; j++) {
+ dist = X[0] - CB[pos];
+ dist *= dist;
+ for (i = 1; i < dim; i++) {
+ tmp = X[i] - CB[pos + i];
+ dist += tmp*tmp;
+ }
+
+ if (dist < mindist) {
+ mindist = dist;
+ minindex = j;
+ }
+ pos += dim;
+ }
+ for (i = 0; i < dim; i++) {
+ Xq[i] = CB[minindex*dim + i];
+ }
+ *index = minindex;
+
+
+
+Andersen, et al. Experimental [Page 143]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+ /*----------------------------------------------------------------*
+ * split vector quantization
+ *---------------------------------------------------------------*/
+
+ void SplitVQ(
+ float *qX, /* (o) the quantized vector */
+ int *index, /* (o) a vector of indexes for all vector
+ codebooks in the split */
+ float *X, /* (i) the vector to quantize */
+ const float *CB,/* (i) the quantizer codebook */
+ int nsplit, /* the number of vector splits */
+ const int *dim, /* the dimension of X and qX */
+ const int *cbsize /* the number of vectors in the codebook */
+ ){
+ int cb_pos, X_pos, i;
+
+ cb_pos = 0;
+ X_pos= 0;
+ for (i = 0; i < nsplit; i++) {
+ vq(qX + X_pos, index + i, CB + cb_pos, X + X_pos,
+ cbsize[i], dim[i]);
+ X_pos += dim[i];
+ cb_pos += dim[i] * cbsize[i];
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * scalar quantization
+ *---------------------------------------------------------------*/
+
+ void sort_sq(
+ float *xq, /* (o) the quantized value */
+ int *index, /* (o) the quantization index */
+ float x, /* (i) the value to quantize */
+ const float *cb,/* (i) the quantization codebook */
+ int cb_size /* (i) the size of the quantization codebook */
+ ){
+ int i;
+
+ if (x <= cb[0]) {
+ *index = 0;
+ *xq = cb[0];
+ } else {
+ i = 0;
+ while ((x > cb[i]) && i < cb_size - 1) {
+ i++;
+
+
+
+Andersen, et al. Experimental [Page 144]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+ if (x > ((cb[i] + cb[i - 1])/2)) {
+ *index = i;
+ *xq = cb[i];
+ } else {
+ *index = i - 1;
+ *xq = cb[i - 1];
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+ int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
+ nonstable ones */
+ float *lsf, /* (i) a table of lsf vectors */
+ int dim, /* (i) the dimension of each lsf vector */
+ int NoAn /* (i) the number of lsf vectors in the
+ table */
+ ){
+ int k,n,m, Nit=2, change=0,pos;
+ float tmp;
+ static float eps=(float)0.039; /* 50 Hz */
+ static float eps2=(float)0.0195;
+ static float maxlsf=(float)3.14; /* 4000 Hz */
+ static float minlsf=(float)0.01; /* 0 Hz */
+
+ /* LSF separation check*/
+
+ for (n=0; n<Nit; n++) { /* Run through a couple of times */
+ for (m=0; m<NoAn; m++) { /* Number of analyses per frame */
+ for (k=0; k<(dim-1); k++) {
+ pos=m*dim+k;
+
+ if ((lsf[pos+1]-lsf[pos])<eps) {
+
+ if (lsf[pos+1]<lsf[pos]) {
+ tmp=lsf[pos+1];
+ lsf[pos+1]= lsf[pos]+eps2;
+ lsf[pos]= lsf[pos+1]-eps2;
+ } else {
+ lsf[pos]-=eps2;
+ lsf[pos+1]+=eps2;
+ }
+ change=1;
+
+
+
+Andersen, et al. Experimental [Page 145]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+
+ if (lsf[pos]<minlsf) {
+ lsf[pos]=minlsf;
+ change=1;
+ }
+
+ if (lsf[pos]>maxlsf) {
+ lsf[pos]=maxlsf;
+ change=1;
+ }
+ }
+ }
+ }
+
+ return change;
+ }
+
+A.27. hpInput.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ hpInput.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HPINPUT_H
+ #define __iLBC_HPINPUT_H
+
+ void hpInput(
+ float *In, /* (i) vector to filter */
+ int len, /* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ );
+
+ #endif
+
+A.28. hpInput.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+
+
+Andersen, et al. Experimental [Page 146]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ hpInput.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "constants.h"
+
+ /*----------------------------------------------------------------*
+ * Input high-pass filter
+ *---------------------------------------------------------------*/
+
+ void hpInput(
+ float *In, /* (i) vector to filter */
+ int len, /* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ ){
+ int i;
+ float *pi, *po;
+
+ /* all-zero section*/
+
+ pi = &In[0];
+ po = &Out[0];
+ for (i=0; i<len; i++) {
+ *po = hpi_zero_coefsTbl[0] * (*pi);
+ *po += hpi_zero_coefsTbl[1] * mem[0];
+ *po += hpi_zero_coefsTbl[2] * mem[1];
+
+ mem[1] = mem[0];
+ mem[0] = *pi;
+ po++;
+ pi++;
+
+ }
+
+ /* all-pole section*/
+
+ po = &Out[0];
+ for (i=0; i<len; i++) {
+ *po -= hpi_pole_coefsTbl[1] * mem[2];
+ *po -= hpi_pole_coefsTbl[2] * mem[3];
+
+ mem[3] = mem[2];
+ mem[2] = *po;
+ po++;
+
+
+
+Andersen, et al. Experimental [Page 147]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ }
+
+A.29. hpOutput.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ hpOutput.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HPOUTPUT_H
+ #define __iLBC_HPOUTPUT_H
+
+ void hpOutput(
+ float *In, /* (i) vector to filter */
+ int len,/* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ );
+
+ #endif
+
+A.30. hpOutput.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ hpOutput.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "constants.h"
+
+ /*----------------------------------------------------------------*
+ * Output high-pass filter
+ *---------------------------------------------------------------*/
+
+ void hpOutput(
+
+
+
+Andersen, et al. Experimental [Page 148]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *In, /* (i) vector to filter */
+ int len,/* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ ){
+ int i;
+ float *pi, *po;
+
+ /* all-zero section*/
+
+ pi = &In[0];
+ po = &Out[0];
+ for (i=0; i<len; i++) {
+ *po = hpo_zero_coefsTbl[0] * (*pi);
+ *po += hpo_zero_coefsTbl[1] * mem[0];
+ *po += hpo_zero_coefsTbl[2] * mem[1];
+
+ mem[1] = mem[0];
+ mem[0] = *pi;
+ po++;
+ pi++;
+
+ }
+
+ /* all-pole section*/
+
+ po = &Out[0];
+ for (i=0; i<len; i++) {
+ *po -= hpo_pole_coefsTbl[1] * mem[2];
+ *po -= hpo_pole_coefsTbl[2] * mem[3];
+
+ mem[3] = mem[2];
+ mem[2] = *po;
+ po++;
+ }
+ }
+
+A.31. iCBConstruct.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBConstruct.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+
+
+
+Andersen, et al. Experimental [Page 149]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ICBCONSTRUCT_H
+ #define __iLBC_ICBCONSTRUCT_H
+
+ void index_conv_enc(
+ int *index /* (i/o) Codebook indexes */
+ );
+
+ void index_conv_dec(
+ int *index /* (i/o) Codebook indexes */
+ );
+
+ void iCBConstruct(
+ float *decvector, /* (o) Decoded vector */
+ int *index, /* (i) Codebook indices */
+ int *gain_index,/* (i) Gain quantization indices */
+ float *mem, /* (i) Buffer for codevector construction */
+ int lMem, /* (i) Length of buffer */
+ int veclen, /* (i) Length of vector */
+ int nStages /* (i) Number of codebook stages */
+ );
+
+ #endif
+
+A.32. iCBConstruct.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBConstruct.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+
+ #include "iLBC_define.h"
+ #include "gainquant.h"
+ #include "getCBvec.h"
+
+ /*----------------------------------------------------------------*
+ * Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+
+
+
+Andersen, et al. Experimental [Page 150]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void index_conv_enc(
+ int *index /* (i/o) Codebook indexes */
+ ){
+ int k;
+
+ for (k=1; k<CB_NSTAGES; k++) {
+
+ if ((index[k]>=108)&&(index[k]<172)) {
+ index[k]-=64;
+ } else if (index[k]>=236) {
+ index[k]-=128;
+ } else {
+ /* ERROR */
+ }
+ }
+ }
+
+ void index_conv_dec(
+ int *index /* (i/o) Codebook indexes */
+ ){
+ int k;
+
+ for (k=1; k<CB_NSTAGES; k++) {
+
+ if ((index[k]>=44)&&(index[k]<108)) {
+ index[k]+=64;
+ } else if ((index[k]>=108)&&(index[k]<128)) {
+ index[k]+=128;
+ } else {
+ /* ERROR */
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+ void iCBConstruct(
+ float *decvector, /* (o) Decoded vector */
+ int *index, /* (i) Codebook indices */
+ int *gain_index,/* (i) Gain quantization indices */
+ float *mem, /* (i) Buffer for codevector construction */
+ int lMem, /* (i) Length of buffer */
+ int veclen, /* (i) Length of vector */
+ int nStages /* (i) Number of codebook stages */
+ ){
+ int j,k;
+
+
+
+Andersen, et al. Experimental [Page 151]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float gain[CB_NSTAGES];
+ float cbvec[SUBL];
+
+ /* gain de-quantization */
+
+ gain[0] = gaindequant(gain_index[0], 1.0, 32);
+ if (nStages > 1) {
+ gain[1] = gaindequant(gain_index[1],
+ (float)fabs(gain[0]), 16);
+ }
+ if (nStages > 2) {
+ gain[2] = gaindequant(gain_index[2],
+ (float)fabs(gain[1]), 8);
+ }
+
+ /* codebook vector construction and construction of
+ total vector */
+
+ getCBvec(cbvec, mem, index[0], lMem, veclen);
+ for (j=0;j<veclen;j++){
+ decvector[j] = gain[0]*cbvec[j];
+ }
+ if (nStages > 1) {
+ for (k=1; k<nStages; k++) {
+ getCBvec(cbvec, mem, index[k], lMem, veclen);
+ for (j=0;j<veclen;j++) {
+ decvector[j] += gain[k]*cbvec[j];
+ }
+ }
+ }
+ }
+
+A.33. iCBSearch.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBSearch.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ICBSEARCH_H
+ #define __iLBC_ICBSEARCH_H
+
+
+
+
+Andersen, et al. Experimental [Page 152]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void iCBSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) the encoder state structure */
+ int *index, /* (o) Codebook indices */
+ int *gain_index,/* (o) Gain quantization indices */
+ float *intarget,/* (i) Target vector for encoding */
+ float *mem, /* (i) Buffer for codebook construction */
+ int lMem, /* (i) Length of buffer */
+ int lTarget, /* (i) Length of vector */
+ int nStages, /* (i) Number of codebook stages */
+ float *weightDenum, /* (i) weighting filter coefficients */
+ float *weightState, /* (i) weighting filter state */
+ int block /* (i) the sub-block number */
+ );
+
+ #endif
+
+A.34. iCBSearch.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBSearch.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+
+ #include "iLBC_define.h"
+ #include "gainquant.h"
+ #include "createCB.h"
+ #include "filter.h"
+ #include "constants.h"
+
+ /*----------------------------------------------------------------*
+ * Search routine for codebook encoding and gain quantization.
+ *---------------------------------------------------------------*/
+
+ void iCBSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) the encoder state structure */
+ int *index, /* (o) Codebook indices */
+ int *gain_index,/* (o) Gain quantization indices */
+
+
+
+Andersen, et al. Experimental [Page 153]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *intarget,/* (i) Target vector for encoding */
+ float *mem, /* (i) Buffer for codebook construction */
+ int lMem, /* (i) Length of buffer */
+ int lTarget, /* (i) Length of vector */
+ int nStages, /* (i) Number of codebook stages */
+ float *weightDenum, /* (i) weighting filter coefficients */
+ float *weightState, /* (i) weighting filter state */
+ int block /* (i) the sub-block number */
+ ){
+ int i, j, icount, stage, best_index, range, counter;
+ float max_measure, gain, measure, crossDot, ftmp;
+ float gains[CB_NSTAGES];
+ float target[SUBL];
+ int base_index, sInd, eInd, base_size;
+ int sIndAug=0, eIndAug=0;
+ float buf[CB_MEML+SUBL+2*LPC_FILTERORDER];
+ float invenergy[CB_EXPAND*128], energy[CB_EXPAND*128];
+ float *pp, *ppi=0, *ppo=0, *ppe=0;
+ float cbvectors[CB_MEML];
+ float tene, cene, cvec[SUBL];
+ float aug_vec[SUBL];
+
+ memset(cvec,0,SUBL*sizeof(float));
+
+ /* Determine size of codebook sections */
+
+ base_size=lMem-lTarget+1;
+
+ if (lTarget==SUBL) {
+ base_size=lMem-lTarget+1+lTarget/2;
+ }
+
+ /* setup buffer for weighting */
+
+ memcpy(buf,weightState,sizeof(float)*LPC_FILTERORDER);
+ memcpy(buf+LPC_FILTERORDER,mem,lMem*sizeof(float));
+ memcpy(buf+LPC_FILTERORDER+lMem,intarget,lTarget*sizeof(float));
+
+ /* weighting */
+
+ AllPoleFilter(buf+LPC_FILTERORDER, weightDenum,
+ lMem+lTarget, LPC_FILTERORDER);
+
+ /* Construct the codebook and target needed */
+
+ memcpy(target, buf+LPC_FILTERORDER+lMem, lTarget*sizeof(float));
+
+ tene=0.0;
+
+
+
+Andersen, et al. Experimental [Page 154]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ for (i=0; i<lTarget; i++) {
+ tene+=target[i]*target[i];
+ }
+
+ /* Prepare search over one more codebook section. This section
+ is created by filtering the original buffer with a filter. */
+
+ filteredCBvecs(cbvectors, buf+LPC_FILTERORDER, lMem);
+
+ /* The Main Loop over stages */
+
+ for (stage=0; stage<nStages; stage++) {
+
+ range = search_rangeTbl[block][stage];
+
+ /* initialize search measure */
+
+ max_measure = (float)-10000000.0;
+ gain = (float)0.0;
+ best_index = 0;
+
+ /* Compute cross dot product between the target
+ and the CB memory */
+
+ crossDot=0.0;
+ pp=buf+LPC_FILTERORDER+lMem-lTarget;
+ for (j=0; j<lTarget; j++) {
+ crossDot += target[j]*(*pp++);
+ }
+
+ if (stage==0) {
+
+ /* Calculate energy in the first block of
+ 'lTarget' samples. */
+ ppe = energy;
+ ppi = buf+LPC_FILTERORDER+lMem-lTarget-1;
+ ppo = buf+LPC_FILTERORDER+lMem-1;
+
+ *ppe=0.0;
+ pp=buf+LPC_FILTERORDER+lMem-lTarget;
+ for (j=0; j<lTarget; j++) {
+ *ppe+=(*pp)*(*pp++);
+ }
+
+ if (*ppe>0.0) {
+ invenergy[0] = (float) 1.0 / (*ppe + EPS);
+ } else {
+ invenergy[0] = (float) 0.0;
+
+
+
+Andersen, et al. Experimental [Page 155]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ ppe++;
+
+ measure=(float)-10000000.0;
+
+ if (crossDot > 0.0) {
+ measure = crossDot*crossDot*invenergy[0];
+ }
+ }
+ else {
+ measure = crossDot*crossDot*invenergy[0];
+ }
+
+ /* check if measure is better */
+ ftmp = crossDot*invenergy[0];
+
+ if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
+ best_index = 0;
+ max_measure = measure;
+ gain = ftmp;
+ }
+
+ /* loop over the main first codebook section,
+ full search */
+
+ for (icount=1; icount<range; icount++) {
+
+ /* calculate measure */
+
+ crossDot=0.0;
+ pp = buf+LPC_FILTERORDER+lMem-lTarget-icount;
+
+ for (j=0; j<lTarget; j++) {
+ crossDot += target[j]*(*pp++);
+ }
+
+ if (stage==0) {
+ *ppe++ = energy[icount-1] + (*ppi)*(*ppi) -
+ (*ppo)*(*ppo);
+ ppo--;
+ ppi--;
+
+ if (energy[icount]>0.0) {
+ invenergy[icount] =
+ (float)1.0/(energy[icount]+EPS);
+ } else {
+ invenergy[icount] = (float) 0.0;
+ }
+
+
+
+Andersen, et al. Experimental [Page 156]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ measure=(float)-10000000.0;
+
+ if (crossDot > 0.0) {
+ measure = crossDot*crossDot*invenergy[icount];
+ }
+ }
+ else {
+ measure = crossDot*crossDot*invenergy[icount];
+ }
+
+ /* check if measure is better */
+ ftmp = crossDot*invenergy[icount];
+
+ if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
+ best_index = icount;
+ max_measure = measure;
+ gain = ftmp;
+ }
+ }
+
+ /* Loop over augmented part in the first codebook
+ * section, full search.
+ * The vectors are interpolated.
+ */
+
+ if (lTarget==SUBL) {
+
+ /* Search for best possible cb vector and
+ compute the CB-vectors' energy. */
+ searchAugmentedCB(20, 39, stage, base_size-lTarget/2,
+ target, buf+LPC_FILTERORDER+lMem,
+ &max_measure, &best_index, &gain, energy,
+ invenergy);
+ }
+
+ /* set search range for following codebook sections */
+
+ base_index=best_index;
+
+ /* unrestricted search */
+
+ if (CB_RESRANGE == -1) {
+ sInd=0;
+ eInd=range-1;
+ sIndAug=20;
+ eIndAug=39;
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 157]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* restricted search around best index from first
+ codebook section */
+
+ else {
+ /* Initialize search indices */
+ sIndAug=0;
+ eIndAug=0;
+ sInd=base_index-CB_RESRANGE/2;
+ eInd=sInd+CB_RESRANGE;
+
+ if (lTarget==SUBL) {
+
+ if (sInd<0) {
+
+ sIndAug = 40 + sInd;
+ eIndAug = 39;
+ sInd=0;
+
+ } else if ( base_index < (base_size-20) ) {
+
+ if (eInd > range) {
+ sInd -= (eInd-range);
+ eInd = range;
+ }
+ } else { /* base_index >= (base_size-20) */
+
+ if (sInd < (base_size-20)) {
+ sIndAug = 20;
+ sInd = 0;
+ eInd = 0;
+ eIndAug = 19 + CB_RESRANGE;
+
+ if(eIndAug > 39) {
+ eInd = eIndAug-39;
+ eIndAug = 39;
+ }
+ } else {
+ sIndAug = 20 + sInd - (base_size-20);
+ eIndAug = 39;
+ sInd = 0;
+ eInd = CB_RESRANGE - (eIndAug-sIndAug+1);
+ }
+ }
+
+ } else { /* lTarget = 22 or 23 */
+
+ if (sInd < 0) {
+ eInd -= sInd;
+
+
+
+Andersen, et al. Experimental [Page 158]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ sInd = 0;
+ }
+
+ if(eInd > range) {
+ sInd -= (eInd - range);
+ eInd = range;
+ }
+ }
+ }
+
+ /* search of higher codebook section */
+
+ /* index search range */
+ counter = sInd;
+ sInd += base_size;
+ eInd += base_size;
+
+
+ if (stage==0) {
+ ppe = energy+base_size;
+ *ppe=0.0;
+
+ pp=cbvectors+lMem-lTarget;
+ for (j=0; j<lTarget; j++) {
+ *ppe+=(*pp)*(*pp++);
+ }
+
+ ppi = cbvectors + lMem - 1 - lTarget;
+ ppo = cbvectors + lMem - 1;
+
+ for (j=0; j<(range-1); j++) {
+ *(ppe+1) = *ppe + (*ppi)*(*ppi) - (*ppo)*(*ppo);
+ ppo--;
+ ppi--;
+ ppe++;
+ }
+ }
+
+ /* loop over search range */
+
+ for (icount=sInd; icount<eInd; icount++) {
+
+ /* calculate measure */
+
+ crossDot=0.0;
+ pp=cbvectors + lMem - (counter++) - lTarget;
+
+ for (j=0;j<lTarget;j++) {
+
+
+
+Andersen, et al. Experimental [Page 159]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ crossDot += target[j]*(*pp++);
+ }
+
+ if (energy[icount]>0.0) {
+ invenergy[icount] =(float)1.0/(energy[icount]+EPS);
+ } else {
+ invenergy[icount] =(float)0.0;
+ }
+
+ if (stage==0) {
+
+ measure=(float)-10000000.0;
+
+ if (crossDot > 0.0) {
+ measure = crossDot*crossDot*
+ invenergy[icount];
+ }
+ }
+ else {
+ measure = crossDot*crossDot*invenergy[icount];
+ }
+
+ /* check if measure is better */
+ ftmp = crossDot*invenergy[icount];
+
+ if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
+ best_index = icount;
+ max_measure = measure;
+ gain = ftmp;
+ }
+ }
+
+ /* Search the augmented CB inside the limited range. */
+
+ if ((lTarget==SUBL)&&(sIndAug!=0)) {
+ searchAugmentedCB(sIndAug, eIndAug, stage,
+ 2*base_size-20, target, cbvectors+lMem,
+ &max_measure, &best_index, &gain, energy,
+ invenergy);
+ }
+
+ /* record best index */
+
+ index[stage] = best_index;
+
+ /* gain quantization */
+
+ if (stage==0){
+
+
+
+Andersen, et al. Experimental [Page 160]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+
+ if (gain<0.0){
+ gain = 0.0;
+ }
+
+ if (gain>CB_MAXGAIN) {
+ gain = (float)CB_MAXGAIN;
+ }
+ gain = gainquant(gain, 1.0, 32, &gain_index[stage]);
+ }
+ else {
+ if (stage==1) {
+ gain = gainquant(gain, (float)fabs(gains[stage-1]),
+ 16, &gain_index[stage]);
+ } else {
+ gain = gainquant(gain, (float)fabs(gains[stage-1]),
+ 8, &gain_index[stage]);
+ }
+ }
+
+ /* Extract the best (according to measure)
+ codebook vector */
+
+ if (lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
+
+ if (index[stage]<base_size) {
+ pp=buf+LPC_FILTERORDER+lMem-lTarget-index[stage];
+ } else {
+ pp=cbvectors+lMem-lTarget-
+ index[stage]+base_size;
+ }
+ } else {
+
+ if (index[stage]<base_size) {
+ if (index[stage]<(base_size-20)) {
+ pp=buf+LPC_FILTERORDER+lMem-
+ lTarget-index[stage];
+ } else {
+ createAugmentedVec(index[stage]-base_size+40,
+ buf+LPC_FILTERORDER+lMem,aug_vec);
+ pp=aug_vec;
+ }
+ } else {
+ int filterno, position;
+
+ filterno=index[stage]/base_size;
+ position=index[stage]-filterno*base_size;
+
+
+
+
+Andersen, et al. Experimental [Page 161]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+
+ if (position<(base_size-20)) {
+ pp=cbvectors+filterno*lMem-lTarget-
+ index[stage]+filterno*base_size;
+ } else {
+ createAugmentedVec(
+ index[stage]-(filterno+1)*base_size+40,
+ cbvectors+filterno*lMem,aug_vec);
+ pp=aug_vec;
+ }
+ }
+ }
+
+ /* Subtract the best codebook vector, according
+ to measure, from the target vector */
+
+ for (j=0;j<lTarget;j++) {
+ cvec[j] += gain*(*pp);
+ target[j] -= gain*(*pp++);
+ }
+
+ /* record quantized gain */
+
+ gains[stage]=gain;
+
+ }/* end of Main Loop. for (stage=0;... */
+
+ /* Gain adjustment for energy matching */
+ cene=0.0;
+ for (i=0; i<lTarget; i++) {
+ cene+=cvec[i]*cvec[i];
+ }
+ j=gain_index[0];
+
+ for (i=gain_index[0]; i<32; i++) {
+ ftmp=cene*gain_sq5Tbl[i]*gain_sq5Tbl[i];
+
+ if ((ftmp<(tene*gains[0]*gains[0])) &&
+ (gain_sq5Tbl[j]<(2.0*gains[0]))) {
+ j=i;
+ }
+ }
+ gain_index[0]=j;
+ }
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 162]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.35. LPCdecode.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPC_decode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LPC_DECODE_H
+ #define __iLBC_LPC_DECODE_H
+
+ void LSFinterpolate2a_dec(
+ float *a, /* (o) lpc coefficients for a sub-frame */
+ float *lsf1, /* (i) first lsf coefficient vector */
+ float *lsf2, /* (i) second lsf coefficient vector */
+ float coef, /* (i) interpolation weight */
+ int length /* (i) length of lsf vectors */
+ );
+
+ void SimplelsfDEQ(
+ float *lsfdeq, /* (o) dequantized lsf coefficients */
+ int *index, /* (i) quantization index */
+ int lpc_n /* (i) number of LPCs */
+ );
+
+ void DecoderInterpolateLSF(
+ float *syntdenum, /* (o) synthesis filter coefficients */
+ float *weightdenum, /* (o) weighting denumerator
+ coefficients */
+ float *lsfdeq, /* (i) dequantized lsf coefficients */
+ int length, /* (i) length of lsf coefficient vector */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i) the decoder state structure */
+ );
+
+ #endif
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 163]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.36. LPCdecode.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPC_decode.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+
+ #include "helpfun.h"
+ #include "lsf.h"
+ #include "iLBC_define.h"
+ #include "constants.h"
+
+ /*---------------------------------------------------------------*
+ * interpolation of lsf coefficients for the decoder
+ *--------------------------------------------------------------*/
+
+ void LSFinterpolate2a_dec(
+ float *a, /* (o) lpc coefficients for a sub-frame */
+ float *lsf1, /* (i) first lsf coefficient vector */
+ float *lsf2, /* (i) second lsf coefficient vector */
+ float coef, /* (i) interpolation weight */
+ int length /* (i) length of lsf vectors */
+ ){
+ float lsftmp[LPC_FILTERORDER];
+
+ interpolate(lsftmp, lsf1, lsf2, coef, length);
+ lsf2a(a, lsftmp);
+ }
+
+ /*---------------------------------------------------------------*
+ * obtain dequantized lsf coefficients from quantization index
+ *--------------------------------------------------------------*/
+
+ void SimplelsfDEQ(
+ float *lsfdeq, /* (o) dequantized lsf coefficients */
+ int *index, /* (i) quantization index */
+ int lpc_n /* (i) number of LPCs */
+ ){
+ int i, j, pos, cb_pos;
+
+
+
+Andersen, et al. Experimental [Page 164]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* decode first LSF */
+
+ pos = 0;
+ cb_pos = 0;
+ for (i = 0; i < LSF_NSPLIT; i++) {
+ for (j = 0; j < dim_lsfCbTbl[i]; j++) {
+ lsfdeq[pos + j] = lsfCbTbl[cb_pos +
+ (long)(index[i])*dim_lsfCbTbl[i] + j];
+ }
+ pos += dim_lsfCbTbl[i];
+ cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
+ }
+
+ if (lpc_n>1) {
+
+ /* decode last LSF */
+
+ pos = 0;
+ cb_pos = 0;
+ for (i = 0; i < LSF_NSPLIT; i++) {
+ for (j = 0; j < dim_lsfCbTbl[i]; j++) {
+ lsfdeq[LPC_FILTERORDER + pos + j] =
+ lsfCbTbl[cb_pos +
+ (long)(index[LSF_NSPLIT + i])*
+ dim_lsfCbTbl[i] + j];
+ }
+ pos += dim_lsfCbTbl[i];
+ cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+ void DecoderInterpolateLSF(
+ float *syntdenum, /* (o) synthesis filter coefficients */
+ float *weightdenum, /* (o) weighting denumerator
+ coefficients */
+ float *lsfdeq, /* (i) dequantized lsf coefficients */
+ int length, /* (i) length of lsf coefficient vector */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i) the decoder state structure */
+ ){
+ int i, pos, lp_length;
+ float lp[LPC_FILTERORDER + 1], *lsfdeq2;
+
+
+
+
+Andersen, et al. Experimental [Page 165]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ lsfdeq2 = lsfdeq + length;
+ lp_length = length + 1;
+
+ if (iLBCdec_inst->mode==30) {
+ /* sub-frame 1: Interpolation between old and first */
+
+ LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
+ lsf_weightTbl_30ms[0], length);
+ memcpy(syntdenum,lp,lp_length*sizeof(float));
+ bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM,
+ lp_length);
+
+ /* sub-frames 2 to 6: interpolation between first
+ and last LSF */
+
+ pos = lp_length;
+ for (i = 1; i < 6; i++) {
+ LSFinterpolate2a_dec(lp, lsfdeq, lsfdeq2,
+ lsf_weightTbl_30ms[i], length);
+ memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
+ bwexpand(weightdenum + pos, lp,
+ LPC_CHIRP_WEIGHTDENUM, lp_length);
+ pos += lp_length;
+ }
+ }
+ else {
+ pos = 0;
+ for (i = 0; i < iLBCdec_inst->nsub; i++) {
+ LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold,
+ lsfdeq, lsf_weightTbl_20ms[i], length);
+ memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
+ bwexpand(weightdenum+pos, lp, LPC_CHIRP_WEIGHTDENUM,
+ lp_length);
+ pos += lp_length;
+ }
+ }
+
+ /* update memory */
+
+ if (iLBCdec_inst->mode==30)
+ memcpy(iLBCdec_inst->lsfdeqold, lsfdeq2,
+ length*sizeof(float));
+ else
+ memcpy(iLBCdec_inst->lsfdeqold, lsfdeq,
+ length*sizeof(float));
+
+ }
+
+
+
+
+Andersen, et al. Experimental [Page 166]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.37. LPCencode.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPCencode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LPCENCOD_H
+ #define __iLBC_LPCENCOD_H
+
+ void LPCencode(
+ float *syntdenum, /* (i/o) synthesis filter coefficients
+ before/after encoding */
+ float *weightdenum, /* (i/o) weighting denumerator coefficients
+ before/after encoding */
+ int *lsf_index, /* (o) lsf quantization index */
+ float *data, /* (i) lsf coefficients to quantize */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ );
+
+ #endif
+
+A.38. LPCencode.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPCencode.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <string.h>
+
+ #include "iLBC_define.h"
+ #include "helpfun.h"
+ #include "lsf.h"
+ #include "constants.h"
+
+
+
+Andersen, et al. Experimental [Page 167]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /*----------------------------------------------------------------*
+ * lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+ void SimpleAnalysis(
+ float *lsf, /* (o) lsf coefficients */
+ float *data, /* (i) new data vector */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ){
+ int k, is;
+ float temp[BLOCKL_MAX], lp[LPC_FILTERORDER + 1];
+ float lp2[LPC_FILTERORDER + 1];
+ float r[LPC_FILTERORDER + 1];
+
+ is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+ memcpy(iLBCenc_inst->lpc_buffer+is,data,
+ iLBCenc_inst->blockl*sizeof(float));
+
+ /* No lookahead, last window is asymmetric */
+
+ for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
+
+ is = LPC_LOOKBACK;
+
+ if (k < (iLBCenc_inst->lpc_n - 1)) {
+ window(temp, lpc_winTbl,
+ iLBCenc_inst->lpc_buffer, BLOCKL_MAX);
+ } else {
+ window(temp, lpc_asymwinTbl,
+ iLBCenc_inst->lpc_buffer + is, BLOCKL_MAX);
+ }
+
+ autocorr(r, temp, BLOCKL_MAX, LPC_FILTERORDER);
+ window(r, r, lpc_lagwinTbl, LPC_FILTERORDER + 1);
+
+ levdurb(lp, temp, r, LPC_FILTERORDER);
+ bwexpand(lp2, lp, LPC_CHIRP_SYNTDENUM, LPC_FILTERORDER+1);
+
+ a2lsf(lsf + k*LPC_FILTERORDER, lp2);
+ }
+ is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+ memmove(iLBCenc_inst->lpc_buffer,
+ iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is,
+ is*sizeof(float));
+ }
+
+ /*----------------------------------------------------------------*
+
+
+
+Andersen, et al. Experimental [Page 168]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ * lsf interpolator and conversion from lsf to a coefficients
+ * (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+ void LSFinterpolate2a_enc(
+ float *a, /* (o) lpc coefficients */
+ float *lsf1,/* (i) first set of lsf coefficients */
+ float *lsf2,/* (i) second set of lsf coefficients */
+ float coef, /* (i) weighting coefficient to use between
+ lsf1 and lsf2 */
+ long length /* (i) length of coefficient vectors */
+ ){
+ float lsftmp[LPC_FILTERORDER];
+
+ interpolate(lsftmp, lsf1, lsf2, coef, length);
+ lsf2a(a, lsftmp);
+ }
+
+ /*----------------------------------------------------------------*
+ * lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+ void SimpleInterpolateLSF(
+ float *syntdenum, /* (o) the synthesis filter denominator
+ resulting from the quantized
+ interpolated lsf */
+ float *weightdenum, /* (o) the weighting filter denominator
+ resulting from the unquantized
+ interpolated lsf */
+ float *lsf, /* (i) the unquantized lsf coefficients */
+ float *lsfdeq, /* (i) the dequantized lsf coefficients */
+ float *lsfold, /* (i) the unquantized lsf coefficients of
+ the previous signal frame */
+ float *lsfdeqold, /* (i) the dequantized lsf coefficients of
+ the previous signal frame */
+ int length, /* (i) should equate LPC_FILTERORDER */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ){
+ int i, pos, lp_length;
+ float lp[LPC_FILTERORDER + 1], *lsf2, *lsfdeq2;
+
+ lsf2 = lsf + length;
+ lsfdeq2 = lsfdeq + length;
+ lp_length = length + 1;
+
+ if (iLBCenc_inst->mode==30) {
+ /* sub-frame 1: Interpolation between old and first
+
+
+
+Andersen, et al. Experimental [Page 169]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ set of lsf coefficients */
+
+ LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
+ lsf_weightTbl_30ms[0], length);
+ memcpy(syntdenum,lp,lp_length*sizeof(float));
+ LSFinterpolate2a_enc(lp, lsfold, lsf,
+ lsf_weightTbl_30ms[0], length);
+ bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM, lp_length);
+
+ /* sub-frame 2 to 6: Interpolation between first
+ and second set of lsf coefficients */
+
+ pos = lp_length;
+ for (i = 1; i < iLBCenc_inst->nsub; i++) {
+ LSFinterpolate2a_enc(lp, lsfdeq, lsfdeq2,
+ lsf_weightTbl_30ms[i], length);
+ memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
+
+ LSFinterpolate2a_enc(lp, lsf, lsf2,
+ lsf_weightTbl_30ms[i], length);
+ bwexpand(weightdenum + pos, lp,
+ LPC_CHIRP_WEIGHTDENUM, lp_length);
+ pos += lp_length;
+ }
+ }
+ else {
+ pos = 0;
+ for (i = 0; i < iLBCenc_inst->nsub; i++) {
+ LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
+ lsf_weightTbl_20ms[i], length);
+ memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
+ LSFinterpolate2a_enc(lp, lsfold, lsf,
+ lsf_weightTbl_20ms[i], length);
+ bwexpand(weightdenum+pos, lp,
+ LPC_CHIRP_WEIGHTDENUM, lp_length);
+ pos += lp_length;
+ }
+ }
+
+ /* update memory */
+
+ if (iLBCenc_inst->mode==30) {
+ memcpy(lsfold, lsf2, length*sizeof(float));
+ memcpy(lsfdeqold, lsfdeq2, length*sizeof(float));
+ }
+ else {
+ memcpy(lsfold, lsf, length*sizeof(float));
+ memcpy(lsfdeqold, lsfdeq, length*sizeof(float));
+
+
+
+Andersen, et al. Experimental [Page 170]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+ void SimplelsfQ(
+ float *lsfdeq, /* (o) dequantized lsf coefficients
+ (dimension FILTERORDER) */
+ int *index, /* (o) quantization index */
+ float *lsf, /* (i) the lsf coefficient vector to be
+ quantized (dimension FILTERORDER ) */
+ int lpc_n /* (i) number of lsf sets to quantize */
+ ){
+ /* Quantize first LSF with memoryless split VQ */
+ SplitVQ(lsfdeq, index, lsf, lsfCbTbl, LSF_NSPLIT,
+ dim_lsfCbTbl, size_lsfCbTbl);
+
+ if (lpc_n==2) {
+ /* Quantize second LSF with memoryless split VQ */
+ SplitVQ(lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
+ lsf + LPC_FILTERORDER, lsfCbTbl, LSF_NSPLIT,
+ dim_lsfCbTbl, size_lsfCbTbl);
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * lpc encoder
+ *---------------------------------------------------------------*/
+
+ void LPCencode(
+ float *syntdenum, /* (i/o) synthesis filter coefficients
+ before/after encoding */
+ float *weightdenum, /* (i/o) weighting denumerator
+ coefficients before/after
+ encoding */
+ int *lsf_index, /* (o) lsf quantization index */
+ float *data, /* (i) lsf coefficients to quantize */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ){
+ float lsf[LPC_FILTERORDER * LPC_N_MAX];
+ float lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+ int change=0;
+
+ SimpleAnalysis(lsf, data, iLBCenc_inst);
+ SimplelsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
+
+
+
+Andersen, et al. Experimental [Page 171]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ change=LSF_check(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
+ SimpleInterpolateLSF(syntdenum, weightdenum,
+ lsf, lsfdeq, iLBCenc_inst->lsfold,
+ iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
+ }
+
+A.39. lsf.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ lsf.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LSF_H
+ #define __iLBC_LSF_H
+
+ void a2lsf(
+ float *freq,/* (o) lsf coefficients */
+ float *a /* (i) lpc coefficients */
+ );
+
+ void lsf2a(
+ float *a_coef, /* (o) lpc coefficients */
+ float *freq /* (i) lsf coefficients */
+ );
+
+ #endif
+
+A.40. lsf.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ lsf.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <string.h>
+
+
+
+Andersen, et al. Experimental [Page 172]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ #include <math.h>
+
+ #include "iLBC_define.h"
+
+ /*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsf coefficients
+ *---------------------------------------------------------------*/
+
+ void a2lsf(
+ float *freq,/* (o) lsf coefficients */
+ float *a /* (i) lpc coefficients */
+ ){
+ float steps[LSF_NUMBER_OF_STEPS] =
+ {(float)0.00635, (float)0.003175, (float)0.0015875,
+ (float)0.00079375};
+ float step;
+ int step_idx;
+ int lsp_index;
+ float p[LPC_HALFORDER];
+ float q[LPC_HALFORDER];
+ float p_pre[LPC_HALFORDER];
+ float q_pre[LPC_HALFORDER];
+ float old_p, old_q, *old;
+ float *pq_coef;
+ float omega, old_omega;
+ int i;
+ float hlp, hlp1, hlp2, hlp3, hlp4, hlp5;
+
+ for (i=0; i<LPC_HALFORDER; i++) {
+ p[i] = (float)-1.0 * (a[i + 1] + a[LPC_FILTERORDER - i]);
+ q[i] = a[LPC_FILTERORDER - i] - a[i + 1];
+ }
+
+ p_pre[0] = (float)-1.0 - p[0];
+ p_pre[1] = - p_pre[0] - p[1];
+ p_pre[2] = - p_pre[1] - p[2];
+ p_pre[3] = - p_pre[2] - p[3];
+ p_pre[4] = - p_pre[3] - p[4];
+ p_pre[4] = p_pre[4] / 2;
+
+ q_pre[0] = (float)1.0 - q[0];
+ q_pre[1] = q_pre[0] - q[1];
+ q_pre[2] = q_pre[1] - q[2];
+ q_pre[3] = q_pre[2] - q[3];
+ q_pre[4] = q_pre[3] - q[4];
+ q_pre[4] = q_pre[4] / 2;
+
+ omega = 0.0;
+
+
+
+Andersen, et al. Experimental [Page 173]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ old_omega = 0.0;
+
+ old_p = FLOAT_MAX;
+ old_q = FLOAT_MAX;
+
+ /* Here we loop through lsp_index to find all the
+ LPC_FILTERORDER roots for omega. */
+
+ for (lsp_index = 0; lsp_index<LPC_FILTERORDER; lsp_index++) {
+
+ /* Depending on lsp_index being even or odd, we
+ alternatively solve the roots for the two LSP equations. */
+
+
+ if ((lsp_index & 0x1) == 0) {
+ pq_coef = p_pre;
+ old = &old_p;
+ } else {
+ pq_coef = q_pre;
+ old = &old_q;
+ }
+
+ /* Start with low resolution grid */
+
+ for (step_idx = 0, step = steps[step_idx];
+ step_idx < LSF_NUMBER_OF_STEPS;){
+
+ /* cos(10piw) + pq(0)cos(8piw) + pq(1)cos(6piw) +
+ pq(2)cos(4piw) + pq(3)cod(2piw) + pq(4) */
+
+ hlp = (float)cos(omega * TWO_PI);
+ hlp1 = (float)2.0 * hlp + pq_coef[0];
+ hlp2 = (float)2.0 * hlp * hlp1 - (float)1.0 +
+ pq_coef[1];
+ hlp3 = (float)2.0 * hlp * hlp2 - hlp1 + pq_coef[2];
+ hlp4 = (float)2.0 * hlp * hlp3 - hlp2 + pq_coef[3];
+ hlp5 = hlp * hlp4 - hlp3 + pq_coef[4];
+
+
+ if (((hlp5 * (*old)) <= 0.0) || (omega >= 0.5)){
+
+ if (step_idx == (LSF_NUMBER_OF_STEPS - 1)){
+
+ if (fabs(hlp5) >= fabs(*old)) {
+ freq[lsp_index] = omega - step;
+ } else {
+ freq[lsp_index] = omega;
+ }
+
+
+
+Andersen, et al. Experimental [Page 174]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+
+
+ if ((*old) >= 0.0){
+ *old = (float)-1.0 * FLOAT_MAX;
+ } else {
+ *old = FLOAT_MAX;
+ }
+
+ omega = old_omega;
+ step_idx = 0;
+
+ step_idx = LSF_NUMBER_OF_STEPS;
+ } else {
+
+ if (step_idx == 0) {
+ old_omega = omega;
+ }
+
+ step_idx++;
+ omega -= steps[step_idx];
+
+ /* Go back one grid step */
+
+ step = steps[step_idx];
+ }
+ } else {
+
+ /* increment omega until they are of different sign,
+ and we know there is at least one root between omega
+ and old_omega */
+ *old = hlp5;
+ omega += step;
+ }
+ }
+ }
+
+ for (i = 0; i<LPC_FILTERORDER; i++) {
+ freq[i] = freq[i] * TWO_PI;
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * conversion from lsf coefficients to lpc coefficients
+ *---------------------------------------------------------------*/
+
+ void lsf2a(
+ float *a_coef, /* (o) lpc coefficients */
+ float *freq /* (i) lsf coefficients */
+
+
+
+Andersen, et al. Experimental [Page 175]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ ){
+ int i, j;
+ float hlp;
+ float p[LPC_HALFORDER], q[LPC_HALFORDER];
+ float a[LPC_HALFORDER + 1], a1[LPC_HALFORDER],
+ a2[LPC_HALFORDER];
+ float b[LPC_HALFORDER + 1], b1[LPC_HALFORDER],
+ b2[LPC_HALFORDER];
+
+ for (i=0; i<LPC_FILTERORDER; i++) {
+ freq[i] = freq[i] * PI2;
+ }
+
+ /* Check input for ill-conditioned cases. This part is not
+ found in the TIA standard. It involves the following 2 IF
+ blocks. If "freq" is judged ill-conditioned, then we first
+ modify freq[0] and freq[LPC_HALFORDER-1] (normally
+ LPC_HALFORDER = 10 for LPC applications), then we adjust
+ the other "freq" values slightly */
+
+
+ if ((freq[0] <= 0.0) || (freq[LPC_FILTERORDER - 1] >= 0.5)){
+
+
+ if (freq[0] <= 0.0) {
+ freq[0] = (float)0.022;
+ }
+
+
+ if (freq[LPC_FILTERORDER - 1] >= 0.5) {
+ freq[LPC_FILTERORDER - 1] = (float)0.499;
+ }
+
+ hlp = (freq[LPC_FILTERORDER - 1] - freq[0]) /
+ (float) (LPC_FILTERORDER - 1);
+
+ for (i=1; i<LPC_FILTERORDER; i++) {
+ freq[i] = freq[i - 1] + hlp;
+ }
+ }
+
+ memset(a1, 0, LPC_HALFORDER*sizeof(float));
+ memset(a2, 0, LPC_HALFORDER*sizeof(float));
+ memset(b1, 0, LPC_HALFORDER*sizeof(float));
+ memset(b2, 0, LPC_HALFORDER*sizeof(float));
+ memset(a, 0, (LPC_HALFORDER+1)*sizeof(float));
+ memset(b, 0, (LPC_HALFORDER+1)*sizeof(float));
+
+
+
+
+Andersen, et al. Experimental [Page 176]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* p[i] and q[i] compute cos(2*pi*omega_{2j}) and
+ cos(2*pi*omega_{2j-1} in eqs. 4.2.2.2-1 and 4.2.2.2-2.
+ Note that for this code p[i] specifies the coefficients
+ used in .Q_A(z) while q[i] specifies the coefficients used
+ in .P_A(z) */
+
+ for (i=0; i<LPC_HALFORDER; i++) {
+ p[i] = (float)cos(TWO_PI * freq[2 * i]);
+ q[i] = (float)cos(TWO_PI * freq[2 * i + 1]);
+ }
+
+ a[0] = 0.25;
+ b[0] = 0.25;
+
+ for (i= 0; i<LPC_HALFORDER; i++) {
+ a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
+ b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
+ a2[i] = a1[i];
+ a1[i] = a[i];
+ b2[i] = b1[i];
+ b1[i] = b[i];
+ }
+
+ for (j=0; j<LPC_FILTERORDER; j++) {
+
+ if (j == 0) {
+ a[0] = 0.25;
+ b[0] = -0.25;
+ } else {
+ a[0] = b[0] = 0.0;
+ }
+
+ for (i=0; i<LPC_HALFORDER; i++) {
+ a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
+ b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
+ a2[i] = a1[i];
+ a1[i] = a[i];
+ b2[i] = b1[i];
+ b1[i] = b[i];
+ }
+
+ a_coef[j + 1] = 2 * (a[LPC_HALFORDER] + b[LPC_HALFORDER]);
+ }
+
+ a_coef[0] = 1.0;
+ }
+
+
+
+
+
+Andersen, et al. Experimental [Page 177]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.41. packing.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ packing.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __PACKING_H
+ #define __PACKING_H
+
+ void packsplit(
+ int *index, /* (i) the value to split */
+ int *firstpart, /* (o) the value specified by most
+ significant bits */
+ int *rest, /* (o) the value specified by least
+ significant bits */
+ int bitno_firstpart, /* (i) number of bits in most
+ significant part */
+ int bitno_total /* (i) number of bits in full range
+ of value */
+ );
+
+ void packcombine(
+ int *index, /* (i/o) the msb value in the
+ combined value out */
+ int rest, /* (i) the lsb value */
+ int bitno_rest /* (i) the number of bits in the
+ lsb part */
+ );
+
+ void dopack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to pack
+ new data, on exit pointer
+ to place in bitstream to
+ pack future data */
+ int index, /* (i) the value to pack */
+ int bitno, /* (i) the number of bits that the
+ value will fit within */
+ int *pos /* (i/o) write position in the
+ current byte */
+ );
+
+
+
+Andersen, et al. Experimental [Page 178]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ void unpack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to
+ unpack new data from, on
+ exit pointer to place in
+ bitstream to unpack future
+ data from */
+ int *index, /* (o) resulting value */
+ int bitno, /* (i) number of bits used to
+ represent the value */
+ int *pos /* (i/o) read position in the
+ current byte */
+ );
+
+ #endif
+
+A.42. packing.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ packing.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <stdlib.h>
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include "helpfun.h"
+ #include "string.h"
+
+ /*----------------------------------------------------------------*
+ * splitting an integer into first most significant bits and
+ * remaining least significant bits
+ *---------------------------------------------------------------*/
+
+ void packsplit(
+ int *index, /* (i) the value to split */
+ int *firstpart, /* (o) the value specified by most
+ significant bits */
+ int *rest, /* (o) the value specified by least
+ significant bits */
+
+
+
+Andersen, et al. Experimental [Page 179]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ int bitno_firstpart, /* (i) number of bits in most
+ significant part */
+ int bitno_total /* (i) number of bits in full range
+ of value */
+ ){
+ int bitno_rest = bitno_total-bitno_firstpart;
+
+ *firstpart = *index>>(bitno_rest);
+ *rest = *index-(*firstpart<<(bitno_rest));
+ }
+
+ /*----------------------------------------------------------------*
+ * combining a value corresponding to msb's with a value
+ * corresponding to lsb's
+ *---------------------------------------------------------------*/
+
+ void packcombine(
+ int *index, /* (i/o) the msb value in the
+ combined value out */
+ int rest, /* (i) the lsb value */
+ int bitno_rest /* (i) the number of bits in the
+ lsb part */
+ ){
+ *index = *index<<bitno_rest;
+ *index += rest;
+ }
+
+ /*----------------------------------------------------------------*
+ * packing of bits into bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+ void dopack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to pack
+ new data, on exit pointer
+ to place in bitstream to
+ pack future data */
+ int index, /* (i) the value to pack */
+ int bitno, /* (i) the number of bits that the
+ value will fit within */
+ int *pos /* (i/o) write position in the
+ current byte */
+ ){
+ int posLeft;
+
+ /* Clear the bits before starting in a new byte */
+
+ if ((*pos)==0) {
+
+
+
+Andersen, et al. Experimental [Page 180]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ **bitstream=0;
+ }
+
+ while (bitno>0) {
+
+ /* Jump to the next byte if end of this byte is reached*/
+
+ if (*pos==8) {
+ *pos=0;
+ (*bitstream)++;
+ **bitstream=0;
+ }
+
+ posLeft=8-(*pos);
+
+ /* Insert index into the bitstream */
+
+ if (bitno <= posLeft) {
+ **bitstream |= (unsigned char)(index<<(posLeft-bitno));
+ *pos+=bitno;
+ bitno=0;
+ } else {
+ **bitstream |= (unsigned char)(index>>(bitno-posLeft));
+
+ *pos=8;
+ index-=((index>>(bitno-posLeft))<<(bitno-posLeft));
+
+ bitno-=posLeft;
+ }
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+ void unpack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to
+ unpack new data from, on
+ exit pointer to place in
+ bitstream to unpack future
+ data from */
+ int *index, /* (o) resulting value */
+ int bitno, /* (i) number of bits used to
+ represent the value */
+ int *pos /* (i/o) read position in the
+ current byte */
+
+
+
+Andersen, et al. Experimental [Page 181]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ ){
+ int BitsLeft;
+
+ *index=0;
+
+ while (bitno>0) {
+
+ /* move forward in bitstream when the end of the
+ byte is reached */
+
+ if (*pos==8) {
+ *pos=0;
+ (*bitstream)++;
+ }
+
+ BitsLeft=8-(*pos);
+
+ /* Extract bits to index */
+
+ if (BitsLeft>=bitno) {
+ *index+=((((**bitstream)<<(*pos)) & 0xFF)>>(8-bitno));
+
+ *pos+=bitno;
+ bitno=0;
+ } else {
+
+ if ((8-bitno)>0) {
+ *index+=((((**bitstream)<<(*pos)) & 0xFF)>>
+ (8-bitno));
+ *pos=8;
+ } else {
+ *index+=(((int)(((**bitstream)<<(*pos)) & 0xFF))<<
+ (bitno-8));
+ *pos=8;
+ }
+ bitno-=BitsLeft;
+ }
+ }
+ }
+
+A.43. StateConstructW.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateConstructW.h
+
+
+
+
+Andersen, et al. Experimental [Page 182]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_STATECONSTRUCTW_H
+ #define __iLBC_STATECONSTRUCTW_H
+
+ void StateConstructW(
+ int idxForMax, /* (i) 6-bit index for the quantization of
+ max amplitude */
+ int *idxVec, /* (i) vector of quantization indexes */
+ float *syntDenum, /* (i) synthesis filter denumerator */
+ float *out, /* (o) the decoded state vector */
+ int len /* (i) length of a state vector */
+ );
+
+ #endif
+
+A.44. StateConstructW.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateConstructW.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include "filter.h"
+
+ /*----------------------------------------------------------------*
+ * decoding of the start state
+ *---------------------------------------------------------------*/
+
+ void StateConstructW(
+ int idxForMax, /* (i) 6-bit index for the quantization of
+ max amplitude */
+ int *idxVec, /* (i) vector of quantization indexes */
+ float *syntDenum, /* (i) synthesis filter denumerator */
+
+
+
+Andersen, et al. Experimental [Page 183]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *out, /* (o) the decoded state vector */
+ int len /* (i) length of a state vector */
+ ){
+ float maxVal, tmpbuf[LPC_FILTERORDER+2*STATE_LEN], *tmp,
+ numerator[LPC_FILTERORDER+1];
+ float foutbuf[LPC_FILTERORDER+2*STATE_LEN], *fout;
+ int k,tmpi;
+
+ /* decoding of the maximum value */
+
+ maxVal = state_frgqTbl[idxForMax];
+ maxVal = (float)pow(10,maxVal)/(float)4.5;
+
+ /* initialization of buffers and coefficients */
+
+ memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
+ memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
+ for (k=0; k<LPC_FILTERORDER; k++) {
+ numerator[k]=syntDenum[LPC_FILTERORDER-k];
+ }
+ numerator[LPC_FILTERORDER]=syntDenum[0];
+ tmp = &tmpbuf[LPC_FILTERORDER];
+ fout = &foutbuf[LPC_FILTERORDER];
+
+ /* decoding of the sample values */
+
+ for (k=0; k<len; k++) {
+ tmpi = len-1-k;
+ /* maxVal = 1/scal */
+ tmp[k] = maxVal*state_sq3Tbl[idxVec[tmpi]];
+ }
+
+ /* circular convolution with all-pass filter */
+
+ memset(tmp+len, 0, len*sizeof(float));
+ ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
+ LPC_FILTERORDER, fout);
+ for (k=0;k<len;k++) {
+ out[k] = fout[len-1-k]+fout[2*len-1-k];
+ }
+ }
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 184]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.45. StateSearchW.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateSearchW.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_STATESEARCHW_H
+ #define __iLBC_STATESEARCHW_H
+
+ void AbsQuantW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *in, /* (i) vector to encode */
+ float *syntDenum, /* (i) denominator of synthesis filter */
+ float *weightDenum, /* (i) denominator of weighting filter */
+ int *out, /* (o) vector of quantizer indexes */
+ int len, /* (i) length of vector to encode and
+ vector of quantizer indexes */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ );
+
+ void StateSearchW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *residual,/* (i) target residual vector */
+ float *syntDenum, /* (i) lpc synthesis filter */
+ float *weightDenum, /* (i) weighting filter denuminator */
+ int *idxForMax, /* (o) quantizer index for maximum
+ amplitude */
+ int *idxVec, /* (o) vector of quantization indexes */
+ int len, /* (i) length of all vectors */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ );
+
+
+ #endif
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 185]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.46. StateSearchW.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateSearchW.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include <math.h>
+ #include <string.h>
+
+ #include "iLBC_define.h"
+ #include "constants.h"
+ #include "filter.h"
+ #include "helpfun.h"
+
+ /*----------------------------------------------------------------*
+ * predictive noise shaping encoding of scaled start state
+ * (subrutine for StateSearchW)
+ *---------------------------------------------------------------*/
+
+ void AbsQuantW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *in, /* (i) vector to encode */
+ float *syntDenum, /* (i) denominator of synthesis filter */
+ float *weightDenum, /* (i) denominator of weighting filter */
+ int *out, /* (o) vector of quantizer indexes */
+ int len, /* (i) length of vector to encode and
+ vector of quantizer indexes */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ ){
+ float *syntOut;
+ float syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+ float toQ, xq;
+ int n;
+ int index;
+
+ /* initialization of buffer for filtering */
+
+ memset(syntOutBuf, 0, LPC_FILTERORDER*sizeof(float));
+
+
+
+
+Andersen, et al. Experimental [Page 186]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ /* initialization of pointer for filtering */
+
+ syntOut = &syntOutBuf[LPC_FILTERORDER];
+
+ /* synthesis and weighting filters on input */
+
+ if (state_first) {
+ AllPoleFilter (in, weightDenum, SUBL, LPC_FILTERORDER);
+ } else {
+ AllPoleFilter (in, weightDenum,
+ iLBCenc_inst->state_short_len-SUBL,
+ LPC_FILTERORDER);
+ }
+
+ /* encoding loop */
+
+ for (n=0; n<len; n++) {
+
+ /* time update of filter coefficients */
+
+ if ((state_first)&&(n==SUBL)){
+ syntDenum += (LPC_FILTERORDER+1);
+ weightDenum += (LPC_FILTERORDER+1);
+
+ /* synthesis and weighting filters on input */
+ AllPoleFilter (&in[n], weightDenum, len-n,
+ LPC_FILTERORDER);
+
+ } else if ((state_first==0)&&
+ (n==(iLBCenc_inst->state_short_len-SUBL))) {
+ syntDenum += (LPC_FILTERORDER+1);
+ weightDenum += (LPC_FILTERORDER+1);
+
+ /* synthesis and weighting filters on input */
+ AllPoleFilter (&in[n], weightDenum, len-n,
+ LPC_FILTERORDER);
+
+ }
+
+ /* prediction of synthesized and weighted input */
+
+ syntOut[n] = 0.0;
+ AllPoleFilter (&syntOut[n], weightDenum, 1,
+ LPC_FILTERORDER);
+
+ /* quantization */
+
+ toQ = in[n]-syntOut[n];
+
+
+
+Andersen, et al. Experimental [Page 187]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ sort_sq(&xq, &index, toQ, state_sq3Tbl, 8);
+ out[n]=index;
+ syntOut[n] = state_sq3Tbl[out[n]];
+
+ /* update of the prediction filter */
+
+ AllPoleFilter(&syntOut[n], weightDenum, 1,
+ LPC_FILTERORDER);
+ }
+ }
+
+ /*----------------------------------------------------------------*
+ * encoding of start state
+ *---------------------------------------------------------------*/
+
+ void StateSearchW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *residual,/* (i) target residual vector */
+ float *syntDenum, /* (i) lpc synthesis filter */
+ float *weightDenum, /* (i) weighting filter denuminator */
+ int *idxForMax, /* (o) quantizer index for maximum
+ amplitude */
+ int *idxVec, /* (o) vector of quantization indexes */
+ int len, /* (i) length of all vectors */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ ){
+ float dtmp, maxVal;
+ float tmpbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS];
+ float *tmp, numerator[1+LPC_FILTERORDER];
+ float foutbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS], *fout;
+ int k;
+ float qmax, scal;
+
+ /* initialization of buffers and filter coefficients */
+
+ memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
+ memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
+ for (k=0; k<LPC_FILTERORDER; k++) {
+ numerator[k]=syntDenum[LPC_FILTERORDER-k];
+ }
+ numerator[LPC_FILTERORDER]=syntDenum[0];
+ tmp = &tmpbuf[LPC_FILTERORDER];
+ fout = &foutbuf[LPC_FILTERORDER];
+
+ /* circular convolution with the all-pass filter */
+
+
+
+
+Andersen, et al. Experimental [Page 188]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ memcpy(tmp, residual, len*sizeof(float));
+ memset(tmp+len, 0, len*sizeof(float));
+ ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
+ LPC_FILTERORDER, fout);
+ for (k=0; k<len; k++) {
+ fout[k] += fout[k+len];
+ }
+
+ /* identification of the maximum amplitude value */
+
+ maxVal = fout[0];
+ for (k=1; k<len; k++) {
+
+ if (fout[k]*fout[k] > maxVal*maxVal){
+ maxVal = fout[k];
+ }
+ }
+ maxVal=(float)fabs(maxVal);
+
+ /* encoding of the maximum amplitude value */
+
+ if (maxVal < 10.0) {
+ maxVal = 10.0;
+ }
+ maxVal = (float)log10(maxVal);
+ sort_sq(&dtmp, idxForMax, maxVal, state_frgqTbl, 64);
+
+ /* decoding of the maximum amplitude representation value,
+ and corresponding scaling of start state */
+
+ maxVal=state_frgqTbl[*idxForMax];
+ qmax = (float)pow(10,maxVal);
+ scal = (float)(4.5)/qmax;
+ for (k=0; k<len; k++){
+ fout[k] *= scal;
+ }
+
+ /* predictive noise shaping encoding of scaled start state */
+
+ AbsQuantW(iLBCenc_inst, fout,syntDenum,
+ weightDenum,idxVec, len, state_first);
+ }
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 189]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+A.47. syntFilter.h
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ syntFilter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_SYNTFILTER_H
+ #define __iLBC_SYNTFILTER_H
+
+ void syntFilter(
+ float *Out, /* (i/o) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len, /* (i) Length of signal */
+ float *mem /* (i/o) Filter state */
+ );
+
+ #endif
+
+A.48. syntFilter.c
+
+ /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ syntFilter.c
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #include "iLBC_define.h"
+
+ /*----------------------------------------------------------------*
+ * LP synthesis filter.
+ *---------------------------------------------------------------*/
+
+ void syntFilter(
+ float *Out, /* (i/o) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len, /* (i) Length of signal */
+
+
+
+Andersen, et al. Experimental [Page 190]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ float *mem /* (i/o) Filter state */
+ ){
+ int i, j;
+ float *po, *pi, *pa, *pm;
+
+ po=Out;
+
+ /* Filter first part using memory from past */
+
+ for (i=0; i<LPC_FILTERORDER; i++) {
+ pi=&Out[i-1];
+ pa=&a[1];
+ pm=&mem[LPC_FILTERORDER-1];
+ for (j=1; j<=i; j++) {
+ *po-=(*pa++)*(*pi--);
+ }
+ for (j=i+1; j<LPC_FILTERORDER+1; j++) {
+ *po-=(*pa++)*(*pm--);
+ }
+ po++;
+ }
+
+ /* Filter last part where the state is entirely in
+ the output vector */
+
+ for (i=LPC_FILTERORDER; i<len; i++) {
+ pi=&Out[i-1];
+ pa=&a[1];
+ for (j=1; j<LPC_FILTERORDER+1; j++) {
+ *po-=(*pa++)*(*pi--);
+ }
+ po++;
+ }
+
+ /* Update state vector */
+
+ memcpy(mem, &Out[len-LPC_FILTERORDER],
+ LPC_FILTERORDER*sizeof(float));
+ }
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 191]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+Authors' Addresses
+
+ Soren Vang Andersen
+ Department of Communication Technology
+ Aalborg University
+ Fredrik Bajers Vej 7A
+ 9200 Aalborg
+ Denmark
+
+ Phone: ++45 9 6358627
+ EMail: sva@kom.auc.dk
+
+
+ Alan Duric
+ Telio AS
+ Stoperigt. 2
+ Oslo, N-0250
+ Norway
+
+ Phone: +47 21673555
+ EMail: alan.duric@telio.no
+
+
+ Henrik Astrom
+ Global IP Sound AB
+ Olandsgatan 42
+ Stockholm, S-11663
+ Sweden
+
+ Phone: +46 8 54553040
+ EMail: henrik.astrom@globalipsound.com
+
+
+ Roar Hagen
+ Global IP Sound AB
+ Olandsgatan 42
+ Stockholm, S-11663
+ Sweden
+
+ Phone: +46 8 54553040
+ EMail: roar.hagen@globalipsound.com
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 192]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+ W. Bastiaan Kleijn
+ Global IP Sound AB
+ Olandsgatan 42
+ Stockholm, S-11663
+ Sweden
+
+ Phone: +46 8 54553040
+ EMail: bastiaan.kleijn@globalipsound.com
+
+
+ Jan Linden
+ Global IP Sound Inc.
+ 900 Kearny Street, suite 500
+ San Francisco, CA-94133
+ USA
+
+ Phone: +1 415 397 2555
+ EMail: jan.linden@globalipsound.com
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 193]
+
+RFC 3951 Internet Low Bit Rate Codec December 2004
+
+
+Full Copyright Statement
+
+ Copyright (C) The Internet Society (2004).
+
+ This document is subject to the rights, licenses and restrictions
+ contained in BCP 78, and except as set forth therein, the authors
+ retain all their rights.
+
+ This document and the information contained herein are provided on an
+ "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
+ OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
+ ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
+ INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
+ INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
+ WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
+
+Intellectual Property
+
+ The IETF takes no position regarding the validity or scope of any
+ Intellectual Property Rights or other rights that might be claimed to
+ pertain to the implementation or use of the technology described in
+ this document or the extent to which any license under such rights
+ might or might not be available; nor does it represent that it has
+ made any independent effort to identify any such rights. Information
+ on the IETF's procedures with respect to rights in IETF Documents can
+ be found in BCP 78 and BCP 79.
+
+ Copies of IPR disclosures made to the IETF Secretariat and any
+ assurances of licenses to be made available, or the result of an
+ attempt made to obtain a general license or permission for the use of
+ such proprietary rights by implementers or users of this
+ specification can be obtained from the IETF on-line IPR repository at
+ http://www.ietf.org/ipr.
+
+ The IETF invites any interested party to bring to its attention any
+ copyrights, patents or patent applications, or other proprietary
+ rights that may cover technology that may be required to implement
+ this standard. Please address the information to the IETF at ietf-
+ ipr@ietf.org.
+
+
+Acknowledgement
+
+ Funding for the RFC Editor function is currently provided by the
+ Internet Society.
+
+
+
+
+
+
+Andersen, et al. Experimental [Page 194]
+
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt
new file mode 100644
index 0000000..eade7cd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt
@@ -0,0 +1,731 @@
+
+
+
+
+
+
+Network Working Group A. Duric
+Request for Comments: 3952 Telio
+Category: Experimental S. Andersen
+ Aalborg University
+ December 2004
+
+
+ Real-time Transport Protocol (RTP) Payload Format
+ for internet Low Bit Rate Codec (iLBC) Speech
+
+Status of this Memo
+
+ This memo defines an Experimental Protocol for the Internet
+ community. It does not specify an Internet standard of any kind.
+ Discussion and suggestions for improvement are requested.
+ Distribution of this memo is unlimited.
+
+Copyright Notice
+
+ Copyright (C) The Internet Society (2004).
+
+Abstract
+
+ This document describes the Real-time Transport Protocol (RTP)
+ payload format for the internet Low Bit Rate Codec (iLBC) Speech
+ developed by Global IP Sound (GIPS). Also, within the document there
+ are included necessary details for the use of iLBC with MIME and
+ Session Description Protocol (SDP).
+
+Table of Contents
+
+ 1. Introduction. . . . . . . . . . . . . . . . . . . . . . . . . . 2
+ 2. Background. . . . . . . . . . . . . . . . . . . . . . . . . . . 2
+ 3. RTP Payload Format. . . . . . . . . . . . . . . . . . . . . . . 3
+ 3.1. Bitstream definition . . . . . . . . . . . . . . . . . . . 3
+ 3.2. Multiple iLBC frames in a RTP packet . . . . . . . . . . . 6
+ 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
+ 4.1. Storage Mode . . . . . . . . . . . . . . . . . . . . . . . 7
+ 4.2. MIME registration of iLBC. . . . . . . . . . . . . . . . . 8
+ 5. Mapping to SDP Parameters . . . . . . . . . . . . . . . . . . . 9
+ 6. Security Considerations . . . . . . . . . . . . . . . . . . . . 11
+ 7. References. . . . . . . . . . . . . . . . . . . . . . . . . . . 11
+ 7.1. Normative References . . . . . . . . . . . . . . . . . . . 11
+ 7.2. Informative References . . . . . . . . . . . . . . . . . . 12
+ 8. Acknowledgements. . . . . . . . . . . . . . . . . . . . . . . . 12
+ Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
+ Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 13
+
+
+
+
+Duric & Andersen Experimental [Page 1]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+1. Introduction
+
+ This document describes how compressed iLBC speech, as produced by
+ the iLBC codec [1], may be formatted for use as an RTP payload type.
+ Methods are provided to packetize the codec data frames into RTP
+ packets. The sender may send one or more codec data frames per
+ packet depending on the application scenario or based on the
+ transport network condition, bandwidth restriction, delay
+ requirements and packet-loss tolerance.
+
+ The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
+ "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
+ document are to be interpreted as described in BCP 14, RFC 2119 [2].
+
+2. Background
+
+ Global IP Sound (GIPS) has developed a speech compression algorithm
+ for use in IP based communications [1]. The iLBC codec enables
+ graceful speech quality degradation in the case of lost frames, which
+ occurs in connection with lost or delayed IP packets.
+
+ This codec is suitable for real time communications such as,
+ telephony and videoconferencing, streaming audio, archival and
+ messaging.
+
+ The iLBC codec [1] is an algorithm that compresses each basic frame
+ (20 ms or 30 ms) of 8000 Hz, 16-bit sampled input speech, into output
+ frames with rate of 400 bits for 30 ms basic frame size and 304 bits
+ for 20 ms basic frame size.
+
+ The codec supports two basic frame lengths: 30 ms at 13.33 kbit/s and
+ 20 ms at 15.2 kbit/s, using a block independent linear-predictive
+ coding (LPC) algorithm. When the codec operates at block lengths of
+ 20 ms, it produces 304 bits per block which MUST be packetized in 38
+ bytes. Similarly, for block lengths of 30 ms it produces 400 bits
+ per block which MUST be packetized in 50 bytes. This algorithm
+ results in a speech coding system with a controlled response to
+ packet losses similar to what is known from pulse code modulation
+ (PCM) with a packet loss concealment (PLC), such as ITU-T G711
+ standard [7], which operates at a fixed bit rate of 64 kbit/s. At
+ the same time, this algorithm enables fixed bit rate coding with a
+ quality-versus-bit rate tradeoff close to what is known from code-
+ excited linear prediction (CELP).
+
+
+
+
+
+
+
+
+Duric & Andersen Experimental [Page 2]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+3. RTP Payload Format
+
+ The iLBC codec uses 20 or 30 ms frames and a sampling rate clock of 8
+ kHz, so the RTP timestamp MUST be in units of 1/8000 of a second. The
+ RTP payload for iLBC has the format shown in the figure bellow. No
+ addition header specific to this payload format is required.
+
+ This format is intended for the situations where the sender and the
+ receiver send one or more codec data frames per packet. The RTP
+ packet looks as follows:
+
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | RTP Header [3] |
+ +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+ | |
+ + one or more frames of iLBC [1] |
+ | |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ Figure 1, Packet format diagram
+
+ The RTP header of the packetized encoded iLBC speech has the expected
+ values as described in [3]. The usage of M bit SHOULD be as
+ specified in the applicable RTP profile, for example, RFC 3551 [4]
+ specifies that if the sender does not suppress silence (i.e., sends a
+ frame on every frame interval), the M bit will always be zero. When
+ more then one codec data frame is present in a single RTP packet, the
+ timestamp is, as always, the oldest data frame represented in the RTP
+ packet.
+
+ The assignment of an RTP payload type for this new packet format is
+ outside the scope of this document, and will not be specified here.
+ It is expected that the RTP profile for a particular class of
+ applications will assign a payload type for this encoding, or if that
+ is not done, then a payload type in the dynamic range shall be chosen
+ by the sender.
+
+3.1. Bitstream definition
+
+ The total number of bits used to describe one frame of 20 ms speech
+ is 304, which fits in 38 bytes and results in a bit rate of 15.20
+ kbit/s. For the case with a frame length of 30 ms speech the total
+ number of bits used is 400, which fits in 50 bytes and results in a
+ bit rate of 13.33 kbit/s. In the bitstream definition, the bits are
+ distributed into three classes according to their bit error or loss
+ sensitivity. The most sensitive bits (class 1) are placed first in
+
+
+
+Duric & Andersen Experimental [Page 3]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ the bitstream for each frame. The less sensitive bits (class 2) are
+ placed after the class 1 bits. The least sensitive bits (class 3)
+ are placed at the end of the bitstream for each frame.
+
+ Looking at the 20/30 ms frame length cases for each class: The class
+ 1 bits occupy a total of 6/8 bytes (48/64 bits), the class 2 bits
+ occupy 8/12 bytes (64/96 bits), and the class 3 bits occupy 24/30
+ bytes (191/239 bits). This distribution of the bits enables the use
+ of uneven level protection (ULP). The detailed bit allocation is
+ shown in the table below. When a quantization index is distributed
+ between more classes the more significant bits belong to the lowest
+ class.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Duric & Andersen Experimental [Page 4]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ Bitstream structure:
+
+ ------------------------------------------------------------------+
+ Parameter | Bits Class <1,2,3> |
+ | 20 ms frame | 30 ms frame |
+ ----------------------------------+---------------+---------------+
+ Split 1 | 6 <6,0,0> | 6 <6,0,0> |
+ LSF 1 Split 2 | 7 <7,0,0> | 7 <7,0,0> |
+ LSF Split 3 | 7 <7,0,0> | 7 <7,0,0> |
+ ------------------+---------------+---------------+
+ Split 1 | NA (Not Appl.)| 6 <6,0,0> |
+ LSF 2 Split 2 | NA | 7 <7,0,0> |
+ Split 3 | NA | 7 <7,0,0> |
+ ------------------+---------------+---------------+
+ Sum | 20 <20,0,0> | 40 <40,0,0> |
+ ----------------------------------+---------------+---------------+
+ Block Class. | 2 <2,0,0> | 3 <3,0,0> |
+ ----------------------------------+---------------+---------------+
+ Position 22 sample segment | 1 <1,0,0> | 1 <1,0,0> |
+ ----------------------------------+---------------+---------------+
+ Scale Factor State Coder | 6 <6,0,0> | 6 <6,0,0> |
+ ----------------------------------+---------------+---------------+
+ Sample 0 | 3 <0,1,2> | 3 <0,1,2> |
+ Quantized Sample 1 | 3 <0,1,2> | 3 <0,1,2> |
+ Residual : | : : | : : |
+ State : | : : | : : |
+ Samples : | : : | : : |
+ Sample 56 | 3 <0,1,2> | 3 <0,1,2> |
+ Sample 57 | NA | 3 <0,1,2> |
+ ------------------+---------------+---------------+
+ Sum | 171 <0,57,114>| 174 <0,58,116>|
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 7 <6,0,1> | 7 <4,2,1> |
+ CB for 22/23 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
+ sample block Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
+ ------------------+---------------+---------------+
+ Sum | 21 <6,0,15> | 21 <4,2,15> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 5 <2,0,3> | 5 <1,1,3> |
+ Gain for 22/23 Stage 2 | 4 <1,1,2> | 4 <1,1,2> |
+ sample block Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Sum | 12 <3,1,8> | 12 <2,2,8> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 8 <7,0,1> | 8 <6,1,1> |
+ sub-block 1 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
+ Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
+ ------------------+---------------+---------------+
+
+
+
+Duric & Andersen Experimental [Page 5]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ Stage 1 | 8 <0,0,8> | 8 <0,7,1> |
+ sub-block 2 Stage 2 | 8 <0,0,8> | 8 <0,0,8> |
+ Indices Stage 3 | 8 <0,0,8> | 8 <0,0,8> |
+ for CB ------------------+---------------+---------------+
+ sub-blocks Stage 1 | NA | 8 <0,7,1> |
+ sub-block 3 Stage 2 | NA | 8 <0,0,8> |
+ Stage 3 | NA | 8 <0,0,8> |
+ ------------------+---------------+---------------+
+ Stage 1 | NA | 8 <0,7,1> |
+ sub-block 4 Stage 2 | NA | 8 <0,0,8> |
+ Stage 3 | NA | 8 <0,0,8> |
+ ------------------+---------------+---------------+
+ Sum | 46 <7,0,39> | 94 <6,22,66> |
+ ----------------------------------+---------------+---------------+
+ Stage 1 | 5 <1,2,2> | 5 <1,2,2> |
+ sub-block 1 Stage 2 | 4 <1,1,2> | 4 <1,2,1> |
+ Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Stage 1 | 5 <1,1,3> | 5 <0,2,3> |
+ sub-block 2 Stage 2 | 4 <0,2,2> | 4 <0,2,2> |
+ Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
+ Gains for ------------------+---------------+---------------+
+ sub-blocks Stage 1 | NA | 5 <0,1,4> |
+ sub-block 3 Stage 2 | NA | 4 <0,1,3> |
+ Stage 3 | NA | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Stage 1 | NA | 5 <0,1,4> |
+ sub-block 4 Stage 2 | NA | 4 <0,1,3> |
+ Stage 3 | NA | 3 <0,0,3> |
+ ------------------+---------------+---------------+
+ Sum | 24 <3,6,15> | 48 <2,12,34> |
+ -------------------------------------------------------------------
+ Empty frame indicator | 1 <0,0,1> | 1 <0,0,1> |
+ -------------------------------------------------------------------
+ SUM 304 <48,64,192> 400 <64,96,240>
+
+ Table 3.1 The bitstream definition for iLBC.
+
+ When packetized into the payload, all the class 1 bits MUST be sorted
+ in order (from top and down) as they were specified in the table.
+ Additionally, all the class 2 bits MUST be sorted (from top and down)
+ and all the class 3 bits MUST be sorted in the same sequential order.
+
+3.2. Multiple iLBC frames in a RTP packet
+
+ More than one iLBC frame may be included in a single RTP packet by a
+ sender.
+
+
+
+
+Duric & Andersen Experimental [Page 6]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ It is important to observe that senders have the following additional
+ restrictions:
+
+ o SHOULD NOT include more iLBC frames in a single RTP packet than
+ will fit in the MTU of the RTP transport protocol.
+
+ o Frames MUST NOT be split between RTP packets.
+
+ o Frames of the different modes (20 ms and 30 ms) MUST NOT be
+ included within the same packet.
+
+ It is RECOMMENDED that the number of frames contained within an RTP
+ packet are consistent with the application. For example, in
+ telephony and other real time applications where delay is important,
+ the delay is lower depending on the amount of frames per packet
+ (i.e., fewer frames per packet, the lower the delay). Whereas for
+ bandwidth constrained links or delay insensitive streaming messaging
+ application, one or more frames per packet would be acceptable.
+
+ Information describing the number of frames contained in an RTP
+ packet is not transmitted as part of the RTP payload. The way to
+ determine the number of iLBC frames is to count the total number of
+ octets within the RTP packet, and divide the octet count by the
+ number of expected octets per frame (32/50 per frame).
+
+4. IANA Considerations
+
+ One new MIME sub-type as described in this section has been
+ registered.
+
+4.1. Storage Mode
+
+ The storage mode is used for storing speech frames (e.g., as a file
+ or email attachment).
+
+ +------------------+
+ | Header |
+ +------------------+
+ | Speech frame 1 |
+ +------------------+
+ : :
+ +------------------+
+ | Speech frame n |
+ +------------------+
+
+ Figure 2, Storage format diagram
+
+
+
+
+
+Duric & Andersen Experimental [Page 7]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ The file begins with a header that includes only a magic number to
+ identify that it is an iLBC file.
+
+ The magic number for iLBC file MUST correspond to the ASCII character
+ string:
+
+ o for 30 ms frame size mode:"#!iLBC30\n", or "0x23 0x21 0x69
+ 0x4C 0x42 0x43 0x33 0x30 0x0A" in hexadecimal form,
+
+ o for 20 ms frame size mode:"#!iLBC20\n", or "0x23 0x21 0x69
+ 0x4C 0x42 0x43 0x32 0x30 0x0A" in hexadecimal form.
+
+ After the header, follow the speech frames in consecutive order.
+
+ Speech frames lost in transmission MUST be stored as "empty frames",
+ as defined in [1].
+
+4.2. MIME Registration of iLBC
+
+ MIME media type name: audio
+
+ MIME subtype: iLBC
+
+ Optional parameters:
+
+ All of the parameters does apply for RTP transfer only.
+
+ maxptime:The maximum amount of media which can be encapsulated in
+ each packet, expressed as time in milliseconds. The time
+ SHALL be calculated as the sum of the time the media present
+ in the packet represents. The time SHOULD be a multiple of
+ the frame size. This attribute is probably only meaningful
+ for audio data, but may be used with other media types if it
+ makes sense. It is a media attribute, and is not dependent
+ on charset. Note that this attribute was introduced after
+ RFC 2327, and non updated implementations will ignore this
+ attribute.
+
+ mode: The iLBC operating frame mode (20 or 30 ms) that will be
+ encapsulated in each packet. Values can be 0, 20 and 30
+ (where 0 is reserved, 20 stands for preferred 20 ms frame
+ size and 30 stands for preferred 30 ms frame size).
+
+ ptime: Defined as usual for RTP audio (see [5]).
+
+ Encoding considerations:
+ This type is defined for transfer via both RTP (RFC 3550)
+ and stored-file methods as described in Section 4.1, of RFC
+
+
+
+Duric & Andersen Experimental [Page 8]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ 3952. Audio data is binary data, and must be encoded for
+ non-binary transport; the Base64 encoding is suitable for
+ email.
+
+ Security considerations:
+ See Section 6 of RFC 3952.
+
+ Public specification:
+ Please refer to RFC 3951 [1].
+
+ Additional information:
+ The following applies to stored-file transfer methods:
+
+ Magic number:
+ ASCII character string for:
+ o 30 ms frame size mode "#!iLBC30\n" (or 0x23 0x21
+ 0x69 0x4C 0x42 0x43 0x33 0x30 0x0A in hexadecimal)
+ o 20 ms frame size mode "#!iLBC20\n" (or 0x23 0x21
+ 0x69 0x4C 0x42 0x43 0x32 0x30 0x0A in hexadecimal)
+
+ File extensions: lbc, LBC
+ Macintosh file type code: none
+ Object identifier or OID: none
+
+ Person & email address to contact for further information:
+ alan.duric@telio.no
+
+ Intended usage: COMMON.
+ It is expected that many VoIP applications will use this
+ type.
+
+ Author/Change controller:
+ alan.duric@telio.no
+ IETF Audio/Video transport working group
+
+5. Mapping To SDP Parameters
+
+ The information carried in the MIME media type specification has a
+ specific mapping to fields in the Session Description Protocol (SDP)
+ [5], which is commonly used to describe RTP sessions. When SDP is
+ used to specify sessions employing the iLBC codec, the mapping is as
+ follows:
+
+ o The MIME type ("audio") goes in SDP "m=" as the media name.
+
+ o The MIME subtype (payload format name) goes in SDP "a=rtpmap" as
+ the encoding name.
+
+
+
+
+Duric & Andersen Experimental [Page 9]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
+ "a=maxptime" attributes, respectively.
+
+ o The parameter "mode" goes in the SDP "a=fmtp" attribute by copying
+ it directly from the MIME media type string as "mode=value".
+
+ When conveying information by SDP, the encoding name SHALL be "iLBC"
+ (the same as the MIME subtype).
+
+ An example of the media representation in SDP for describing iLBC
+ might be:
+
+ m=audio 49120 RTP/AVP 97
+ a=rtpmap:97 iLBC/8000
+
+ If 20 ms frame size mode is used, remote iLBC encoder SHALL receive
+ "mode" parameter in the SDP "a=fmtp" attribute by copying them
+ directly from the MIME media type string as a semicolon separated
+ with parameter=value, where parameter is "mode", and values can be 0
+ and 20 (where 0 is reserved and 20 stands for preferred 20 ms frame
+ size). An example of the media representation in SDP for describing
+ iLBC when 20 ms frame size mode is used might be:
+
+ m=audio 49120 RTP/AVP 97
+ a=rtpmap:97 iLBC/8000
+ a=fmtp:97 mode=20
+
+ It is important to emphasize the bi-directional character of the
+ "mode" parameter - both sides of a bi-directional session MUST use
+ the same "mode" value.
+
+ The offer contains the preferred mode of the offerer. The answerer
+ may agree to that mode by including the same mode in the answer, or
+ may include a different mode. The resulting mode used by both
+ parties SHALL be the lower of the bandwidth modes in the offer and
+ answer.
+
+ That is, an offer of "mode=20" receiving an answer of "mode=30" will
+ result in "mode=30" being used by both participants. Similarly, an
+ offer of "mode=30" and an answer of "mode=20" will result in
+ "mode=30" being used by both participants.
+
+ This is important when one end point utilizes a bandwidth constrained
+ link (e.g., 28.8k modem link or slower), where only the lower frame
+ size will work.
+
+
+
+
+
+
+Duric & Andersen Experimental [Page 10]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ Parameter ptime can not be used for the purpose of specifying iLBC
+ operating mode, due to fact that for the certain values it will be
+ impossible to distinguish which mode is about to be used (e.g., when
+ ptime=60, it would be impossible to distinguish if packet is carrying
+ 2 frames of 30 ms or 3 frames of 20 ms, etc.).
+
+ Note that the payload format (encoding) names are commonly shown in
+ upper case. MIME subtypes are commonly shown in lower case. These
+ names are case-insensitive in both places. Similarly, parameter
+ names are case-insensitive both in MIME types and in the default
+ mapping to the SDP a=fmtp attribute
+
+6. Security Considerations
+
+ RTP packets using the payload format defined in this specification
+ are subject to the general security considerations discussed in [3]
+ and any appropriate profile (e.g., [4]).
+
+ As this format transports encoded speech, the main security issues
+ include confidentiality and authentication of the speech itself. The
+ payload format itself does not have any built-in security mechanisms.
+ Confidentiality of the media streams is achieved by encryption,
+ therefore external mechanisms, such as SRTP [6], MAY be used for that
+ purpose. The data compression used with this payload format is
+ applied end-to-end; hence encryption may be performed after
+ compression with no conflict between the two operations.
+
+ A potential denial-of-service threat exists for data encoding using
+ compression techniques that have non-uniform receiver-end
+ computational load. The attacker can inject pathological datagrams
+ into the stream which are complex to decode and cause the receiver to
+ become overloaded. However, the encodings covered in this document
+ do not exhibit any significant non-uniformity.
+
+7. References
+
+7.1. Normative References
+
+ [1] Andersen, S., Duric, A., Astrom, H., Hagen, R., Kleijn, W., and
+ J. Linden, "Internet Low Bit Rate Codec (iLBC)", RFC 3951,
+ December 2004.
+
+ [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
+ Levels", BCP 14, RFC 2119, March 1997.
+
+ [3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
+ "RTP: A Transport Protocol for Real-Time Applications", STD 64,
+ RFC 3550, July 2003.
+
+
+
+Duric & Andersen Experimental [Page 11]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+ [4] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
+ Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
+
+ [5] Handley, M. and V. Jacobson, "SDP: Session Description
+ Protocol", RFC 2327, April 1998.
+
+ [6] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
+ Norrman, "The Secure Real-time Transport Protocol", RFC 3711,
+ March 2004.
+
+7.2. Informative References
+
+ [7] ITU-T Recommendation G.711, available online from the ITU
+ bookstore at http://www.itu.int.
+
+8. Acknowledgements
+
+ Henry Sinnreich, Patrik Faltstrom, Alan Johnston and Jean-Francois
+ Mule for great support of the iLBC initiative and for valuable
+ feedback and comments.
+
+Authors' Addresses
+
+ Alan Duric
+ Telio AS
+ Stoperigt. 2
+ Oslo, N-0250
+ Norway
+
+ Phone: +47 21673505
+ EMail: alan.duric@telio.no
+
+
+ Soren Vang Andersen
+ Department of Communication Technology
+ Aalborg University
+ Fredrik Bajers Vej 7A
+ 9200 Aalborg
+ Denmark
+
+ Phone: ++45 9 6358627
+ EMail: sva@kom.auc.dk
+
+
+
+
+
+
+
+
+
+Duric & Andersen Experimental [Page 12]
+
+RFC 3952 RTP Payload Format for iLBC Speech December 2004
+
+
+Full Copyright Statement
+
+ Copyright (C) The Internet Society (2004).
+
+ This document is subject to the rights, licenses and restrictions
+ contained in BCP 78, and except as set forth therein, the authors
+ retain all their rights.
+
+ This document and the information contained herein are provided on an
+ "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
+ OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
+ ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
+ INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
+ INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
+ WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
+
+Intellectual Property
+
+ The IETF takes no position regarding the validity or scope of any
+ Intellectual Property Rights or other rights that might be claimed to
+ pertain to the implementation or use of the technology described in
+ this document or the extent to which any license under such rights
+ might or might not be available; nor does it represent that it has
+ made any independent effort to identify any such rights. Information
+ on the IETF's procedures with respect to rights in IETF Documents can
+ be found in BCP 78 and BCP 79.
+
+ Copies of IPR disclosures made to the IETF Secretariat and any
+ assurances of licenses to be made available, or the result of an
+ attempt made to obtain a general license or permission for the use of
+ such proprietary rights by implementers or users of this
+ specification can be obtained from the IETF on-line IPR repository at
+ http://www.ietf.org/ipr.
+
+ The IETF invites any interested party to bring to its attention any
+ copyrights, patents or patent applications, or other proprietary
+ rights that may cover technology that may be required to implement
+ this standard. Please address the information to the IETF at ietf-
+ ipr@ietf.org.
+
+
+Acknowledgement
+
+ Funding for the RFC Editor function is currently provided by the
+ Internet Society.
+
+
+
+
+
+
+Duric & Andersen Experimental [Page 13]
+
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/encode.c
new file mode 100644
index 0000000..64e9eab
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/encode.c
@@ -0,0 +1,518 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "lpc_encode.h"
+#include "frame_classify.h"
+#include "state_search.h"
+#include "state_construct.h"
+#include "constants.h"
+#include "cb_search.h"
+#include "cb_construct.h"
+#include "index_conv_enc.h"
+#include "pack_bits.h"
+#include "hp_input.h"
+
+#ifdef SPLIT_10MS
+#include "unpack_bits.h"
+#include "index_conv_dec.h"
+#endif
+#ifndef WEBRTC_BIG_ENDIAN
+#include "swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ * main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+ WebRtc_UWord16 *bytes, /* (o) encoded data bits iLBC */
+ WebRtc_Word16 *block, /* (i) speech vector to encode */
+ iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+ state */
+ ){
+ int n, meml_gotten, Nfor, Nback;
+ WebRtc_Word16 diff, start_pos;
+ int index;
+ int subcount, subframe;
+ WebRtc_Word16 start_count, end_count;
+ WebRtc_Word16 *residual;
+ WebRtc_Word32 en1, en2;
+ WebRtc_Word16 scale, max;
+ WebRtc_Word16 *syntdenum;
+ WebRtc_Word16 *decresidual;
+ WebRtc_Word16 *reverseResidual;
+ WebRtc_Word16 *reverseDecresidual;
+ /* Stack based */
+ WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+ WebRtc_Word16 dataVec[BLOCKL_MAX + LPC_FILTERORDER];
+ WebRtc_Word16 memVec[CB_MEML+CB_FILTERLEN];
+ WebRtc_Word16 bitsMemory[sizeof(iLBC_bits)/sizeof(WebRtc_Word16)];
+ iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
+
+
+#ifdef SPLIT_10MS
+ WebRtc_Word16 *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
+ WebRtc_Word16 last_bit;
+#endif
+
+ WebRtc_Word16 *data = &dataVec[LPC_FILTERORDER];
+ WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN];
+
+ /* Reuse som buffers to save stack memory */
+ residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
+ syntdenum = mem; /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */
+ decresidual = residual; /* Already encoded residual is overwritten by the decoded version */
+ reverseResidual = data; /* data and reverseResidual are used non overlapping in the code */
+ reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */
+
+#ifdef SPLIT_10MS
+
+ WebRtcSpl_MemSetW16 ( (WebRtc_Word16 *) iLBCbits_inst, 0,
+ (WebRtc_Word16) (sizeof(iLBC_bits) / sizeof(WebRtc_Word16)) );
+
+ start_pos = iLBCenc_inst->start_pos;
+ diff = iLBCenc_inst->diff;
+
+ if (iLBCenc_inst->section != 0){
+ WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf,
+ SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+ /* Un-Packetize the frame into parameters */
+ last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+ if (last_bit)
+ return;
+ /* adjust index */
+ WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index);
+
+ if (iLBCenc_inst->section == 1){
+ /* Save first 80 samples of a 160/240 sample frame for 20/30msec */
+ WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80);
+ }
+ else{ // iLBCenc_inst->section == 2 AND mode = 30ms
+ /* Save second 80 samples of a 240 sample frame for 30msec */
+ WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80);
+ }
+ }
+ else{ // iLBCenc_inst->section == 0
+ /* form a complete frame of 160/240 for 20msec/30msec mode */
+ WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80);
+ WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples,
+ (iLBCenc_inst->mode * 8) - 80);
+ iLBCenc_inst->Nfor_flag = 0;
+ iLBCenc_inst->Nback_flag = 0;
+#else
+ /* copy input block to data*/
+ WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl);
+#endif
+
+ /* high pass filtering of input signal and scale down the residual (*0.5) */
+ WebRtcIlbcfix_HpInput(data, (WebRtc_Word16*)WebRtcIlbcfix_kHpInCoefs,
+ iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
+ iLBCenc_inst->blockl);
+
+ /* LPC of hp filtered input data */
+ WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data,
+ iLBCenc_inst);
+
+ /* Set up state */
+ WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER);
+
+ /* inverse filter to get residual */
+ for (n=0; n<iLBCenc_inst->nsub; n++ ) {
+ WebRtcSpl_FilterMAFastQ12(
+ &data[n*SUBL], &residual[n*SUBL],
+ &syntdenum[n*(LPC_FILTERORDER+1)],
+ LPC_FILTERORDER+1, SUBL);
+ }
+
+ /* Copy the state for next frame */
+ WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+ /* find state location */
+
+ iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual);
+
+ /* check if state should be in first or last part of the
+ two subframes */
+
+ index = (iLBCbits_inst->startIdx-1)*SUBL;
+ max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL);
+ scale=WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max,max));
+
+ /* Scale to maximum 25 bits so that the MAC won't cause overflow */
+ scale = scale - 25;
+ if(scale < 0) {
+ scale = 0;
+ }
+
+ diff = STATE_LEN - iLBCenc_inst->state_short_len;
+ en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+ iLBCenc_inst->state_short_len, scale);
+ index += diff;
+ en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+ iLBCenc_inst->state_short_len, scale);
+ if (en1 > en2) {
+ iLBCbits_inst->state_first = 1;
+ start_pos = (iLBCbits_inst->startIdx-1)*SUBL;
+ } else {
+ iLBCbits_inst->state_first = 0;
+ start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff;
+ }
+
+ /* scalar quantization of state */
+
+ WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos],
+ &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+ &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]);
+
+ WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec,
+ &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+ &decresidual[start_pos], iLBCenc_inst->state_short_len
+ );
+
+ /* predictive quantization in state */
+
+ if (iLBCbits_inst->state_first) { /* put adaptive part in the end */
+
+ /* setup memory */
+
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
+ decresidual+start_pos, iLBCenc_inst->state_short_len);
+
+ /* encode subframes */
+
+ WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+ &residual[start_pos+iLBCenc_inst->state_short_len],
+ mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+ &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0);
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(&decresidual[start_pos+iLBCenc_inst->state_short_len],
+ iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+ mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL,
+ diff
+ );
+
+ }
+ else { /* put adaptive part in the beginning */
+
+ /* create reversed vectors for prediction */
+
+ WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1],
+ &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff);
+
+ /* setup memory */
+
+ meml_gotten = iLBCenc_inst->state_short_len;
+ WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
+
+ /* encode subframes */
+ WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+ reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+ &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+ 0);
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(reverseDecresidual,
+ iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+ mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL,
+ diff
+ );
+
+ /* get decoded residual from reversed vector */
+
+ WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff);
+ }
+
+#ifdef SPLIT_10MS
+ iLBCenc_inst->start_pos = start_pos;
+ iLBCenc_inst->diff = diff;
+ iLBCenc_inst->section++;
+ /* adjust index */
+ WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+ /* Packetize the parameters into the frame */
+ WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+ WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+ SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+ return;
+ }
+#endif
+
+ /* forward prediction of subframes */
+
+ Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1;
+
+ /* counter for predicted subframes */
+#ifdef SPLIT_10MS
+ if (iLBCenc_inst->mode == 20)
+ {
+ subcount = 1;
+ }
+ if (iLBCenc_inst->mode == 30)
+ {
+ if (iLBCenc_inst->section == 1)
+ {
+ subcount = 1;
+ }
+ if (iLBCenc_inst->section == 2)
+ {
+ subcount = 3;
+ }
+ }
+#else
+ subcount=1;
+#endif
+
+ if( Nfor > 0 ){
+
+ /* setup memory */
+
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+ decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN);
+
+#ifdef SPLIT_10MS
+ if (iLBCenc_inst->Nfor_flag > 0)
+ {
+ for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++)
+ {
+ /* update memory */
+ WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+ WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+ &decresidual[(iLBCbits_inst->startIdx + 1 +
+ subframe) * SUBL], SUBL);
+ }
+ }
+
+ iLBCenc_inst->Nfor_flag++;
+
+ if (iLBCenc_inst->mode == 20)
+ {
+ start_count = 0;
+ end_count = Nfor;
+ }
+ if (iLBCenc_inst->mode == 30)
+ {
+ if (iLBCenc_inst->section == 1)
+ {
+ start_count = 0;
+ end_count = WEBRTC_SPL_MIN (Nfor, 2);
+ }
+ if (iLBCenc_inst->section == 2)
+ {
+ start_count = WEBRTC_SPL_MIN (Nfor, 2);
+ end_count = Nfor;
+ }
+ }
+#else
+ start_count = 0;
+ end_count = (WebRtc_Word16)Nfor;
+#endif
+
+ /* loop over subframes to encode */
+
+ for (subframe = start_count; subframe < end_count; subframe++){
+
+ /* encode subframe */
+
+ WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+ iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+ &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
+ mem, MEM_LF_TBL, SUBL,
+ &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
+ (WebRtc_Word16)subcount);
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(&decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
+ iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+ iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+ mem, MEM_LF_TBL,
+ SUBL
+ );
+
+ /* update memory */
+
+ WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, (CB_MEML-SUBL));
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+ &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL);
+
+ subcount++;
+ }
+ }
+
+#ifdef SPLIT_10MS
+ if ((iLBCenc_inst->section == 1) &&
+ (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2))
+ {
+ iLBCenc_inst->section++;
+ /* adjust index */
+ WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+ /* Packetize the parameters into the frame */
+ WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+ WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+ SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+ return;
+ }
+#endif
+
+ /* backward prediction of subframes */
+
+ Nback = iLBCbits_inst->startIdx-1;
+
+ if( Nback > 0 ){
+
+ /* create reverse order vectors
+ (The decresidual does not need to be copied since it is
+ contained in the same vector as the residual)
+ */
+
+ WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
+
+ /* setup memory */
+
+ meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx);
+ if( meml_gotten > CB_MEML ) {
+ meml_gotten=CB_MEML;
+ }
+
+ WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
+ WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
+
+#ifdef SPLIT_10MS
+ if (iLBCenc_inst->Nback_flag > 0)
+ {
+ for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++)
+ {
+ /* update memory */
+ WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+ WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+ &reverseDecresidual[subframe * SUBL], SUBL);
+ }
+ }
+
+ iLBCenc_inst->Nback_flag++;
+
+
+ if (iLBCenc_inst->mode == 20)
+ {
+ start_count = 0;
+ end_count = Nback;
+ }
+ if (iLBCenc_inst->mode == 30)
+ {
+ if (iLBCenc_inst->section == 1)
+ {
+ start_count = 0;
+ end_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
+ }
+ if (iLBCenc_inst->section == 2)
+ {
+ start_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
+ end_count = Nback;
+ }
+ }
+#else
+ start_count = 0;
+ end_count = (WebRtc_Word16)Nback;
+#endif
+
+ /* loop over subframes to encode */
+
+ for (subframe = start_count; subframe < end_count; subframe++){
+
+ /* encode subframe */
+
+ WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+ iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
+ mem, MEM_LF_TBL, SUBL,
+ &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
+ (WebRtc_Word16)subcount);
+
+ /* construct decoded vector */
+
+ WebRtcIlbcfix_CbConstruct(&reverseDecresidual[subframe*SUBL],
+ iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+ iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+ mem, MEM_LF_TBL, SUBL
+ );
+
+ /* update memory */
+
+ WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, (CB_MEML-SUBL));
+ WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+ &reverseDecresidual[subframe*SUBL], SUBL);
+
+ subcount++;
+
+ }
+
+ /* get decoded residual from reversed vector */
+
+ WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback);
+ }
+ /* end encoding part */
+
+ /* adjust index */
+
+ WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index);
+
+ /* Packetize the parameters into the frame */
+
+#ifdef SPLIT_10MS
+ if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){
+ WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+ }
+ else{
+ WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+ }
+#else
+ WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+#endif
+
+#ifndef WEBRTC_BIG_ENDIAN
+ /* Swap bytes for LITTLE ENDIAN since the packbits()
+ function assumes BIG_ENDIAN machine */
+#ifdef SPLIT_10MS
+ if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) ||
+ ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){
+ WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words);
+ }
+#else
+ WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words);
+#endif
+#endif
+
+#ifdef SPLIT_10MS
+ if (subcount == (iLBCenc_inst->nsub - 1))
+ {
+ iLBCenc_inst->section = 0;
+ }
+ else
+ {
+ iLBCenc_inst->section++;
+ WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+ SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+ }
+#endif
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/encode.h
new file mode 100644
index 0000000..b553f0c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/encode.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+ WebRtc_UWord16 *bytes, /* (o) encoded data bits iLBC */
+ WebRtc_Word16 *block, /* (i) speech vector to encode */
+ iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+ state */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c
new file mode 100644
index 0000000..d56069b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.c
+
+******************************************************************/
+
+/* Inverses the in vector in into Q29 domain */
+
+#include "energy_inverse.h"
+
+void WebRtcIlbcfix_EnergyInverse(
+ WebRtc_Word16 *energy, /* (i/o) Energy and inverse
+ energy (in Q29) */
+ int noOfEnergies) /* (i) The length of the energy
+ vector */
+{
+ WebRtc_Word32 Nom=(WebRtc_Word32)0x1FFFFFFF;
+ WebRtc_Word16 *energyPtr;
+ int i;
+
+ /* Set the minimum energy value to 16384 to avoid overflow */
+ energyPtr=energy;
+ for (i=0; i<noOfEnergies; i++) {
+ (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384);
+ energyPtr++;
+ }
+
+ /* Calculate inverse energy in Q29 */
+ energyPtr=energy;
+ for (i=0; i<noOfEnergies; i++) {
+ (*energyPtr) = (WebRtc_Word16)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
+ energyPtr++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h
new file mode 100644
index 0000000..db13589
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+
+#include "defines.h"
+
+/* Inverses the in vector in into Q29 domain */
+
+void WebRtcIlbcfix_EnergyInverse(
+ WebRtc_Word16 *energy, /* (i/o) Energy and inverse
+ energy (in Q29) */
+ int noOfEnergies); /* (i) The length of the energy
+ vector */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c
new file mode 100644
index 0000000..3343816
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+ WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
+ WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
+ ){
+ int j;
+ WebRtc_Word32 *pu1, *pu11;
+ WebRtc_Word16 *ps, *w16tmp;
+ const WebRtc_Word16 *pp;
+
+ /* filtering: filter overhangs left side of sequence */
+ pu1=useq1;
+ for (j=0;j<ENH_UPS0; j++) {
+ pu11=pu1;
+ /* i = 2 */
+ pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+ ps=seq1+2;
+ (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ pu11+=ENH_UPS0;
+ /* i = 3 */
+ pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+ ps=seq1+3;
+ (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ pu11+=ENH_UPS0;
+ /* i = 4 */
+ pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+ ps=seq1+4;
+ (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
+ pu1++;
+ }
+
+ /* filtering: simple convolution=inner products
+ (not needed since the sequence is so short)
+ */
+
+ /* filtering: filter overhangs right side of sequence */
+
+ /* Code with loops, which is equivivalent to the expanded version below
+
+ filterlength = 5;
+ hf1 = 2;
+ for(j=0;j<ENH_UPS0; j++){
+ pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j;
+ for(i=1; i<=hfl; i++){
+ *pu=0;
+ pp = polyp[j]+i;
+ ps = seq1+dim1-1;
+ for(k=0;k<filterlength-i;k++) {
+ *pu += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ }
+ pu+=ENH_UPS0;
+ }
+ }
+ */
+ pu1 = useq1 + 12;
+ w16tmp = seq1+4;
+ for (j=0;j<ENH_UPS0; j++) {
+ pu11 = pu1;
+ /* i = 1 */
+ pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2;
+ ps = w16tmp;
+ (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ pu11+=ENH_UPS0;
+ /* i = 2 */
+ pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3;
+ ps = w16tmp;
+ (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
+ pu11+=ENH_UPS0;
+
+ pu1++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h
new file mode 100644
index 0000000..53534cc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+ WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
+ WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c
new file mode 100644
index 0000000..b8f3335
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "get_sync_seq.h"
+#include "smooth.h"
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+ WebRtc_Word16 *odata, /* (o) smoothed block, dimension blockl */
+ WebRtc_Word16 *idata, /* (i) data buffer used for enhancing */
+ WebRtc_Word16 idatal, /* (i) dimension idata */
+ WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
+ WebRtc_Word16 *period, /* (i) pitch period array (pitch bward-in time) */
+ WebRtc_Word16 *plocs, /* (i) locations where period array values valid */
+ WebRtc_Word16 periodl /* (i) dimension of period and plocs */
+ ){
+ /* Stack based */
+ WebRtc_Word16 surround[ENH_BLOCKL];
+
+ WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
+
+ /* get said second sequence of segments */
+
+ WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs,
+ periodl, ENH_HL, surround);
+
+ /* compute the smoothed output from said second sequence */
+
+ WebRtcIlbcfix_Smooth(odata, idata+centerStartPos, surround);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h
new file mode 100644
index 0000000..e14f559
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+ WebRtc_Word16 *odata, /* (o) smoothed block, dimension blockl */
+ WebRtc_Word16 *idata, /* (i) data buffer used for enhancing */
+ WebRtc_Word16 idatal, /* (i) dimension idata */
+ WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
+ WebRtc_Word16 *period, /* (i) pitch period array (pitch bward-in time) */
+ WebRtc_Word16 *plocs, /* (i) locations where period array values valid */
+ WebRtc_Word16 periodl /* (i) dimension of period and plocs */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c
new file mode 100644
index 0000000..61b71d1
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -0,0 +1,381 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "xcorr_coef.h"
+#include "enhancer.h"
+#include "hp_output.h"
+
+
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
+ WebRtc_Word16 *out, /* (o) enhanced signal */
+ WebRtc_Word16 *in, /* (i) unenhanced signal */
+ iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
+ ){
+ int iblock;
+ int lag=20, tlag=20;
+ int inLen=iLBCdec_inst->blockl+120;
+ WebRtc_Word16 scale, scale1, plc_blockl;
+ WebRtc_Word16 *enh_buf, *enh_period;
+ WebRtc_Word32 tmp1, tmp2, max, new_blocks;
+ WebRtc_Word16 *enh_bufPtr1;
+ int i, k;
+ WebRtc_Word16 EnChange;
+ WebRtc_Word16 SqrtEnChange;
+ WebRtc_Word16 inc;
+ WebRtc_Word16 win;
+ WebRtc_Word16 *tmpW16ptr;
+ WebRtc_Word16 startPos;
+ WebRtc_Word16 *plc_pred;
+ WebRtc_Word16 *target, *regressor;
+ WebRtc_Word16 max16;
+ int shifts;
+ WebRtc_Word32 ener;
+ WebRtc_Word16 enerSh;
+ WebRtc_Word16 corrSh;
+ WebRtc_Word16 ind, sh;
+ WebRtc_Word16 start, stop;
+ /* Stack based */
+ WebRtc_Word16 totsh[3];
+ WebRtc_Word16 downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
+ WebRtc_Word32 corr32[50];
+ WebRtc_Word32 corrmax[3];
+ WebRtc_Word16 corr16[3];
+ WebRtc_Word16 en16[3];
+ WebRtc_Word16 lagmax[3];
+
+ plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
+ downsampled are non overlapping */
+ enh_buf=iLBCdec_inst->enh_buf;
+ enh_period=iLBCdec_inst->enh_period;
+
+ /* Copy in the new data into the enhancer buffer */
+
+ WEBRTC_SPL_MEMMOVE_W16(enh_buf, &enh_buf[iLBCdec_inst->blockl],
+ ENH_BUFL-iLBCdec_inst->blockl);
+
+ WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
+ iLBCdec_inst->blockl);
+
+ /* Set variables that are dependent on frame size */
+ if (iLBCdec_inst->mode==30) {
+ plc_blockl=ENH_BLOCKL;
+ new_blocks=3;
+ startPos=320; /* Start position for enhancement
+ (640-new_blocks*ENH_BLOCKL-80) */
+ } else {
+ plc_blockl=40;
+ new_blocks=2;
+ startPos=440; /* Start position for enhancement
+ (640-new_blocks*ENH_BLOCKL-40) */
+ }
+
+ /* Update the pitch prediction for each enhancer block, move the old ones */
+ WEBRTC_SPL_MEMMOVE_W16(enh_period, &enh_period[new_blocks],
+ (ENH_NBLOCKS_TOT-new_blocks));
+
+ k=WebRtcSpl_DownsampleFast(
+ enh_buf+ENH_BUFL-inLen, /* Input samples */
+ (WebRtc_Word16)(inLen+ENH_BUFL_FILTEROVERHEAD),
+ downsampled,
+ (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(inLen, 1),
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpFiltCoefs, /* Coefficients in Q12 */
+ FILTERORDER_DS_PLUS1, /* Length of filter (order-1) */
+ FACTOR_DS,
+ DELAY_DS);
+
+ /* Estimate the pitch in the down sampled domain. */
+ for(iblock = 0; iblock<new_blocks; iblock++){
+
+ /* references */
+ i=60+WEBRTC_SPL_MUL_16_16(iblock,ENH_BLOCKL_HALF);
+ target=downsampled+i;
+ regressor=downsampled+i-10;
+
+ /* scaling */
+ max16=WebRtcSpl_MaxAbsValueW16(®ressor[-50],
+ (WebRtc_Word16)(ENH_BLOCKL_HALF+50-1));
+ shifts = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max16, max16)) - 25;
+ shifts = WEBRTC_SPL_MAX(0, shifts);
+
+ /* compute cross correlation */
+ WebRtcSpl_CrossCorrelation(corr32, target, regressor,
+ ENH_BLOCKL_HALF, 50, (WebRtc_Word16)shifts, -1);
+
+ /* Find 3 highest correlations that should be compared for the
+ highest (corr*corr)/ener */
+
+ for (i=0;i<2;i++) {
+ lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
+ corrmax[i] = corr32[lagmax[i]];
+ start = lagmax[i] - 2;
+ stop = lagmax[i] + 2;
+ start = WEBRTC_SPL_MAX(0, start);
+ stop = WEBRTC_SPL_MIN(49, stop);
+ for (k=start; k<=stop; k++) {
+ corr32[k] = 0;
+ }
+ }
+ lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50);
+ corrmax[2] = corr32[lagmax[2]];
+
+ /* Calculate normalized corr^2 and ener */
+ for (i=0;i<3;i++) {
+ corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
+ ener = WebRtcSpl_DotProductWithScale(®ressor[-lagmax[i]],
+ ®ressor[-lagmax[i]],
+ ENH_BLOCKL_HALF, shifts);
+ enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
+ corr16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
+ corr16[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(corr16[i],
+ corr16[i], 16);
+ en16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
+ totsh[i] = enerSh - WEBRTC_SPL_LSHIFT_W32(corrSh, 1);
+ }
+
+ /* Compare lagmax[0..3] for the (corr^2)/ener criteria */
+ ind = 0;
+ for (i=1; i<3; i++) {
+ if (totsh[ind] > totsh[i]) {
+ sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]);
+ if ( WEBRTC_SPL_MUL_16_16(corr16[ind], en16[i]) <
+ WEBRTC_SPL_MUL_16_16_RSFT(corr16[i], en16[ind], sh)) {
+ ind = i;
+ }
+ } else {
+ sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]);
+ if (WEBRTC_SPL_MUL_16_16_RSFT(corr16[ind], en16[i], sh) <
+ WEBRTC_SPL_MUL_16_16(corr16[i], en16[ind])) {
+ ind = i;
+ }
+ }
+ }
+
+ lag = lagmax[ind] + 10;
+
+ /* Store the estimated lag in the non-downsampled domain */
+ enh_period[ENH_NBLOCKS_TOT-new_blocks+iblock] =
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(lag, 8);
+
+ /* Store the estimated lag for backward PLC */
+ if (iLBCdec_inst->prev_enh_pl==1) {
+ if (!iblock) {
+ tlag = WEBRTC_SPL_MUL_16_16(lag, 2);
+ }
+ } else {
+ if (iblock==1) {
+ tlag = WEBRTC_SPL_MUL_16_16(lag, 2);
+ }
+ }
+
+ lag = WEBRTC_SPL_MUL_16_16(lag, 2);
+ }
+
+ if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) {
+
+ /* Calculate the best lag of the new frame
+ This is used to interpolate backwards and mix with the PLC'd data
+ */
+
+ /* references */
+ target=in;
+ regressor=in+tlag-1;
+
+ /* scaling */
+ max16=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(plc_blockl+3-1));
+ if (max16>5000)
+ shifts=2;
+ else
+ shifts=0;
+
+ /* compute cross correlation */
+ WebRtcSpl_CrossCorrelation(corr32, target, regressor,
+ plc_blockl, 3, (WebRtc_Word16)shifts, 1);
+
+ /* find lag */
+ lag=WebRtcSpl_MaxIndexW32(corr32, 3);
+ lag+=tlag-1;
+
+ /* Copy the backward PLC to plc_pred */
+
+ if (iLBCdec_inst->prev_enh_pl==1) {
+ if (lag>plc_blockl) {
+ WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl);
+ } else {
+ WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag);
+ WEBRTC_SPL_MEMCPY_W16(
+ plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag],
+ (plc_blockl-lag));
+ }
+ } else {
+ int pos;
+
+ pos = plc_blockl;
+
+ while (lag<pos) {
+ WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag);
+ pos = pos - lag;
+ }
+ WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos);
+
+ }
+
+ if (iLBCdec_inst->prev_enh_pl==1) {
+ /* limit energy change
+ if energy in backward PLC is more than 4 times higher than the forward
+ PLC, then reduce the energy in the backward PLC vector:
+ sample 1...len-16 set energy of the to 4 times forward PLC
+ sample len-15..len interpolate between 4 times fw PLC and bw PLC energy
+
+ Note: Compared to floating point code there is a slight change,
+ the window is 16 samples long instead of 10 samples to simplify the
+ calculations
+ */
+
+ max=WebRtcSpl_MaxAbsValueW16(
+ &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
+ max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
+ max = WEBRTC_SPL_MAX(max, max16);
+ scale=22-(WebRtc_Word16)WebRtcSpl_NormW32(max);
+ scale=WEBRTC_SPL_MAX(scale,0);
+
+ tmp2 = WebRtcSpl_DotProductWithScale(
+ &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+ &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+ plc_blockl, scale);
+ tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred,
+ plc_blockl, scale);
+
+ /* Check the energy difference */
+ if ((tmp1>0)&&((tmp1>>2)>tmp2)) {
+ /* EnChange is now guaranteed to be <0.5
+ Calculate EnChange=tmp2/tmp1 in Q16
+ */
+
+ scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp1);
+ tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
+
+ tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
+ EnChange = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp2,
+ (WebRtc_Word16)tmp1);
+
+ /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
+ SqrtEnChange = (WebRtc_Word16)WebRtcSpl_SqrtFloor(
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)EnChange, 14));
+
+
+ /* Multiply first part of vector with 2*SqrtEnChange */
+ WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange,
+ (WebRtc_Word16)(plc_blockl-16), 14);
+
+ /* Calculate increase parameter for window part (16 last samples) */
+ /* (1-2*SqrtEnChange)/16 in Q15 */
+ inc=(2048-WEBRTC_SPL_RSHIFT_W16(SqrtEnChange, 3));
+
+ win=0;
+ tmpW16ptr=&plc_pred[plc_blockl-16];
+
+ for (i=16;i>0;i--) {
+ (*tmpW16ptr)=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (*tmpW16ptr), (SqrtEnChange+(win>>1)), 14);
+ /* multiply by (2.0*SqrtEnChange+win) */
+
+ win += inc;
+ tmpW16ptr++;
+ }
+ }
+
+ /* Make the linear interpolation between the forward PLC'd data
+ and the backward PLC'd data (from the new frame)
+ */
+
+ if (plc_blockl==40) {
+ inc=400; /* 1/41 in Q14 */
+ } else { /* plc_blockl==80 */
+ inc=202; /* 1/81 in Q14 */
+ }
+ win=0;
+ enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+ for (i=0; i<plc_blockl; i++) {
+ win+=inc;
+ *enh_bufPtr1 =
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*enh_bufPtr1), win, 14);
+ *enh_bufPtr1 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (16384-win), plc_pred[plc_blockl-1-i], 14);
+ enh_bufPtr1--;
+ }
+ } else {
+ WebRtc_Word16 *synt = &downsampled[LPC_FILTERORDER];
+
+ enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
+ WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
+
+ /* Clear fileter memory */
+ WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+ WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+ WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+
+ /* Initialize filter memory by filtering through 2 lags */
+ WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem,
+ LPC_FILTERORDER);
+ WebRtcSpl_FilterARFastQ12(
+ enh_bufPtr1,
+ synt,
+ &iLBCdec_inst->old_syntdenum[
+ (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+ LPC_FILTERORDER+1, (WebRtc_Word16)lag);
+
+ WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
+ LPC_FILTERORDER);
+ WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+ iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+ (WebRtc_Word16)lag);
+ WebRtcSpl_FilterARFastQ12(
+ enh_bufPtr1, synt,
+ &iLBCdec_inst->old_syntdenum[
+ (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+ LPC_FILTERORDER+1, (WebRtc_Word16)lag);
+
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
+ LPC_FILTERORDER);
+ WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
+ iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+ (WebRtc_Word16)lag);
+ }
+ }
+
+
+ /* Perform enhancement block by block */
+
+ for (iblock = 0; iblock<new_blocks; iblock++) {
+ WebRtcIlbcfix_Enhancer(out+WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL),
+ enh_buf,
+ ENH_BUFL,
+ (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL)+startPos),
+ enh_period,
+ (WebRtc_Word16*)WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+ }
+
+ return (lag);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h
new file mode 100644
index 0000000..37b27e2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
+ WebRtc_Word16 *out, /* (o) enhanced signal */
+ WebRtc_Word16 *in, /* (i) unenhanced signal */
+ iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
new file mode 100644
index 0000000..7cece26
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * Construct an additional codebook vector by filtering the
+ * initial codebook buffer. This vector is then used to expand
+ * the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+ WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
+ WebRtc_Word16 *CBmem, /* (i) Codebook memory that is filtered to create a
+ second CB section */
+ int lMem, /* (i) Length of codebook memory */
+ WebRtc_Word16 samples /* (i) Number of samples to filter */
+ ) {
+
+ /* Set up the memory, start with zero state */
+ WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN);
+ WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+ WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples);
+
+ /* Filter to obtain the filtered CB memory */
+
+ WebRtcSpl_FilterMAFastQ12(
+ CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
+ (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
new file mode 100644
index 0000000..c502e8f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct an additional codebook vector by filtering the
+ * initial codebook buffer. This vector is then used to expand
+ * the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+ WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
+ WebRtc_Word16 *CBmem, /* (i) Codebook memory that is filtered to create a
+ second CB section */
+ int lMem, /* (i) Length of codebook memory */
+ WebRtc_Word16 samples /* (i) Number of samples to filter */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c
new file mode 100644
index 0000000..ea3675e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * Classification of subframes to localize start state
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
+ /* (o) Index to the max-energy sub frame */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) the encoder state structure */
+ WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
+ ){
+ WebRtc_Word16 max, scale;
+ WebRtc_Word32 ssqEn[NSUB_MAX-1];
+ WebRtc_Word16 *ssqPtr;
+ WebRtc_Word32 *seqEnPtr;
+ WebRtc_Word32 maxW32;
+ WebRtc_Word16 scale1;
+ WebRtc_Word16 pos;
+ int n;
+
+ /*
+ Calculate the energy of each of the 80 sample blocks
+ in the draft the 4 first and last samples are windowed with 1/5...4/5
+ and 4/5...1/5 respectively. To simplify for the fixpoint we have changed
+ this to 0 0 1 1 and 1 1 0 0
+ */
+
+ max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl);
+ scale=WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max,max));
+
+ /* Scale to maximum 24 bits so that it won't overflow for 76 samples */
+ scale = scale-24;
+ scale1 = WEBRTC_SPL_MAX(0, scale);
+
+ /* Calculate energies */
+ ssqPtr=residualFIX + 2;
+ seqEnPtr=ssqEn;
+ for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+ (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1);
+ ssqPtr += 40;
+ seqEnPtr++;
+ }
+
+ /* Scale to maximum 20 bits in order to allow for the 11 bit window */
+ maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1));
+ scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
+ scale1 = WEBRTC_SPL_MAX(0, scale);
+
+ /* Window each 80 block with the ssqEn_winTbl window to give higher probability for
+ the blocks in the middle
+ */
+ seqEnPtr=ssqEn;
+ if (iLBCenc_inst->mode==20) {
+ ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
+ } else {
+ ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin;
+ }
+ for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+ (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
+ seqEnPtr++;
+ ssqPtr++;
+ }
+
+ /* Extract the best choise of start state */
+ pos = WebRtcSpl_MaxIndexW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1)) + 1;
+
+ return(pos);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h
new file mode 100644
index 0000000..faf4666
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+
+WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
+ /* (o) Index to the max-energy sub frame */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) the encoder state structure */
+ WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c
new file mode 100644
index 0000000..9450a80
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * decoder for quantized gains in the gain-shape coding of
+ * residual
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_GainDequant(
+ /* (o) quantized gain value (Q14) */
+ WebRtc_Word16 index, /* (i) quantization index */
+ WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
+ WebRtc_Word16 stage /* (i) The stage of the search */
+ ){
+ WebRtc_Word16 scale;
+ const WebRtc_Word16 *gain;
+
+ /* obtain correct scale factor */
+
+ scale=WEBRTC_SPL_ABS_W16(maxIn);
+ scale = WEBRTC_SPL_MAX(1638, scale); /* if lower than 0.1, set it to 0.1 */
+
+ /* select the quantization table and return the decoded value */
+ gain = WebRtcIlbcfix_kGain[stage];
+
+ return((WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, gain[index])+8192)>>14));
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h
new file mode 100644
index 0000000..28f2ceb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * decoder for quantized gains in the gain-shape coding of
+ * residual
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_GainDequant(
+ /* (o) quantized gain value (Q14) */
+ WebRtc_Word16 index, /* (i) quantization index */
+ WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
+ WebRtc_Word16 stage /* (i) The stage of the search */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c
new file mode 100644
index 0000000..bdf88a5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+ WebRtc_Word16 gain, /* (i) gain value Q14 */
+ WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
+ WebRtc_Word16 stage, /* (i) The stage of the search */
+ WebRtc_Word16 *index /* (o) quantization index */
+ ) {
+
+ WebRtc_Word16 scale, returnVal, cblen;
+ WebRtc_Word32 gainW32, measure1, measure2;
+ const WebRtc_Word16 *cbPtr, *cb;
+ int loc, noMoves, noChecks, i;
+
+ /* ensure a lower bound (0.1) on the scaling factor */
+
+ scale = WEBRTC_SPL_MAX(1638, maxIn);
+
+ /* select the quantization table and calculate
+ the length of the table and the number of
+ steps in the binary search that are needed */
+ cb = WebRtcIlbcfix_kGain[stage];
+ cblen = 32>>stage;
+ noChecks = 4-stage;
+
+ /* Multiply the gain with 2^14 to make the comparison
+ easier and with higher precision */
+ gainW32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)gain, 14);
+
+ /* Do a binary search, starting in the middle of the CB
+ loc - defines the current position in the table
+ noMoves - defines the number of steps to move in the CB in order
+ to get next CB location
+ */
+
+ loc = cblen>>1;
+ noMoves = loc;
+ cbPtr = cb + loc; /* Centre of CB */
+
+ for (i=noChecks;i>0;i--) {
+ noMoves>>=1;
+ measure1=WEBRTC_SPL_MUL_16_16(scale, (*cbPtr));
+
+ /* Move up if gain is larger, otherwise move down in table */
+ measure1 = measure1 - gainW32;
+
+ if (0>measure1) {
+ cbPtr+=noMoves;
+ loc+=noMoves;
+ } else {
+ cbPtr-=noMoves;
+ loc-=noMoves;
+ }
+ }
+
+ /* Check which value is the closest one: loc-1, loc or loc+1 */
+
+ measure1=WEBRTC_SPL_MUL_16_16(scale, (*cbPtr));
+ if (gainW32>measure1) {
+ /* Check against value above loc */
+ measure2=WEBRTC_SPL_MUL_16_16(scale, (*(cbPtr+1)));
+ if ((measure2-gainW32)<(gainW32-measure1)) {
+ loc+=1;
+ }
+ } else {
+ /* Check against value below loc */
+ measure2=WEBRTC_SPL_MUL_16_16(scale, (*(cbPtr-1)));
+ if ((gainW32-measure2)<=(measure1-gainW32)) {
+ loc-=1;
+ }
+ }
+
+ /* Guard against getting outside the table. The calculation above can give a location
+ which is one above the maximum value (in very rare cases) */
+ loc=WEBRTC_SPL_MIN(loc, (cblen-1));
+ *index=loc;
+
+ /* Calculate the quantized gain value (in Q14) */
+ returnVal=(WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, cb[loc])+8192)>>14);
+
+ /* return the quantized value */
+ return(returnVal);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h
new file mode 100644
index 0000000..a2f0596
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+ WebRtc_Word16 gain, /* (i) gain value Q14 */
+ WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
+ WebRtc_Word16 stage, /* (i) The stage of the search */
+ WebRtc_Word16 *index /* (o) quantization index */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c
new file mode 100644
index 0000000..aba3e31
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "create_augmented_vec.h"
+
+/*----------------------------------------------------------------*
+ * Construct codebook vector for given index.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetCbVec(
+ WebRtc_Word16 *cbvec, /* (o) Constructed codebook vector */
+ WebRtc_Word16 *mem, /* (i) Codebook buffer */
+ WebRtc_Word16 index, /* (i) Codebook index */
+ WebRtc_Word16 lMem, /* (i) Length of codebook buffer */
+ WebRtc_Word16 cbveclen /* (i) Codebook vector length */
+ ){
+ WebRtc_Word16 k, base_size;
+ WebRtc_Word16 lag;
+ /* Stack based */
+ WebRtc_Word16 tempbuff2[SUBL+5];
+
+ /* Determine size of codebook sections */
+
+ base_size=lMem-cbveclen+1;
+
+ if (cbveclen==SUBL) {
+ base_size+=WEBRTC_SPL_RSHIFT_W16(cbveclen,1);
+ }
+
+ /* No filter -> First codebook section */
+
+ if (index<lMem-cbveclen+1) {
+
+ /* first non-interpolated vectors */
+
+ k=index+cbveclen;
+ /* get vector */
+ WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen);
+
+ } else if (index < base_size) {
+
+ /* Calculate lag */
+
+ k=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (index-(lMem-cbveclen+1)))+cbveclen;
+
+ lag=WEBRTC_SPL_RSHIFT_W16(k, 1);
+
+ WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);
+
+ }
+
+ /* Higher codebbok section based on filtering */
+
+ else {
+
+ WebRtc_Word16 memIndTest;
+
+ /* first non-interpolated vectors */
+
+ if (index-base_size<lMem-cbveclen+1) {
+
+ /* Set up filter memory, stuff zeros outside memory buffer */
+
+ memIndTest = lMem-(index-base_size+cbveclen);
+
+ WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+ WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+ /* do filtering to get the codebook vector */
+
+ WebRtcSpl_FilterMAFastQ12(
+ &mem[memIndTest+4], cbvec, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
+ CB_FILTERLEN, cbveclen);
+ }
+
+ /* interpolated vectors */
+
+ else {
+ /* Stuff zeros outside memory buffer */
+ memIndTest = lMem-cbveclen-CB_FILTERLEN;
+ WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+ /* do filtering */
+ WebRtcSpl_FilterMAFastQ12(
+ &mem[memIndTest+7], tempbuff2, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
+ CB_FILTERLEN, (WebRtc_Word16)(cbveclen+5));
+
+ /* Calculate lag index */
+ lag = (cbveclen<<1)-20+index-base_size-lMem-1;
+
+ WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec);
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h
new file mode 100644
index 0000000..99b5d4e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+
+void WebRtcIlbcfix_GetCbVec(
+ WebRtc_Word16 *cbvec, /* (o) Constructed codebook vector */
+ WebRtc_Word16 *mem, /* (i) Codebook buffer */
+ WebRtc_Word16 index, /* (i) Codebook index */
+ WebRtc_Word16 lMem, /* (i) Length of codebook buffer */
+ WebRtc_Word16 cbveclen /* (i) Codebook vector length */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
new file mode 100644
index 0000000..c55e918
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ * b = -2.0 * lsp[2*i-2];
+ * f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ * for (j=i; j>=2; j--) {
+ * f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ * }
+ * f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(
+ WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
+ WebRtc_Word32 *f) /* (o) polonymial in Q24 */
+{
+ WebRtc_Word32 tmpW32;
+ int i, j;
+ WebRtc_Word16 high, low;
+ WebRtc_Word16 *lspPtr;
+ WebRtc_Word32 *fPtr;
+
+ lspPtr = lsp;
+ fPtr = f;
+ /* f[0] = 1.0 (Q24) */
+ (*fPtr) = (WebRtc_Word32)16777216;
+ fPtr++;
+
+ (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
+ fPtr++;
+ lspPtr+=2;
+
+ for(i=2; i<=5; i++)
+ {
+ (*fPtr) = fPtr[-2];
+
+ for(j=i; j>1; j--)
+ {
+ /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
+ high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1], 16);
+ low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1]-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)high),16), 1);
+
+ tmpW32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(high, (*lspPtr)), 2) +
+ WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(low, (*lspPtr), 15), 2);
+
+ (*fPtr) += fPtr[-2];
+ (*fPtr) -= tmpW32;
+ fPtr--;
+ }
+ (*fPtr) -= (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(*lspPtr), 10);
+
+ fPtr+=i;
+ lspPtr+=2;
+ }
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
new file mode 100644
index 0000000..b0520b4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ * b = -2.0 * lsp[2*i-2];
+ * f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ * for (j=i; j>=2; j--) {
+ * f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ * }
+ * f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(
+ WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
+ WebRtc_Word32 *f); /* (o) polonymial in Q24 */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c
new file mode 100644
index 0000000..ce72865
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -0,0 +1,108 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "refiner.h"
+#include "nearest_neighbor.h"
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+ WebRtc_Word16 *idata, /* (i) original data */
+ WebRtc_Word16 idatal, /* (i) dimension of data */
+ WebRtc_Word16 centerStartPos, /* (i) where current block starts */
+ WebRtc_Word16 *period, /* (i) rough-pitch-period array (Q-2) */
+ WebRtc_Word16 *plocs, /* (i) where periods of period array are taken (Q-2) */
+ WebRtc_Word16 periodl, /* (i) dimension period array */
+ WebRtc_Word16 hl, /* (i) 2*hl+1 is the number of sequences */
+ WebRtc_Word16 *surround /* (i/o) The contribution from this sequence
+ summed with earlier contributions */
+ ){
+ WebRtc_Word16 i,centerEndPos,q;
+ /* Stack based */
+ WebRtc_Word16 lagBlock[2*ENH_HL+1];
+ WebRtc_Word16 blockStartPos[2*ENH_HL+1]; /* Defines the position to search around (Q2) */
+ WebRtc_Word16 plocs2[ENH_PLOCSL];
+
+ centerEndPos=centerStartPos+ENH_BLOCKL-1;
+
+ /* present (find predicted lag from this position) */
+
+ WebRtcIlbcfix_NearestNeighbor(lagBlock+hl,plocs,
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (centerStartPos+centerEndPos)),
+ periodl);
+
+ blockStartPos[hl]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, centerStartPos);
+
+ /* past (find predicted position and perform a refined
+ search to find the best sequence) */
+
+ for(q=hl-1;q>=0;q--) {
+ blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
+
+ WebRtcIlbcfix_NearestNeighbor(lagBlock+q, plocs,
+ (WebRtc_Word16)(blockStartPos[q] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)-period[lagBlock[q+1]]),
+ periodl);
+
+ if((blockStartPos[q]-(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_OVERHANG))>=0) {
+
+ /* Find the best possible sequence in the 4 times upsampled
+ domain around blockStartPos+q */
+ WebRtcIlbcfix_Refiner(blockStartPos+q,idata,idatal,
+ centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[q]);
+
+ } else {
+ /* Don't add anything since this sequence would
+ be outside the buffer */
+ }
+ }
+
+ /* future (find predicted position and perform a refined
+ search to find the best sequence) */
+
+ for(i=0;i<periodl;i++) {
+ plocs2[i]=(plocs[i]-period[i]);
+ }
+
+ for(q=hl+1;q<=WEBRTC_SPL_MUL_16_16(2, hl);q++) {
+
+ WebRtcIlbcfix_NearestNeighbor(lagBlock+q,plocs2,
+ (WebRtc_Word16)(blockStartPos[q-1]+
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)),periodl);
+
+ blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
+
+ if( (blockStartPos[q]+(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, (ENH_BLOCKL+ENH_OVERHANG)))
+ <
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, idatal)) {
+
+ /* Find the best possible sequence in the 4 times upsampled
+ domain around blockStartPos+q */
+ WebRtcIlbcfix_Refiner(blockStartPos+q, idata, idatal,
+ centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[2*hl-q]);
+
+ }
+ else {
+ /* Don't add anything since this sequence would
+ be outside the buffer */
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h
new file mode 100644
index 0000000..a0ffd39
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+ WebRtc_Word16 *idata, /* (i) original data */
+ WebRtc_Word16 idatal, /* (i) dimension of data */
+ WebRtc_Word16 centerStartPos, /* (i) where current block starts */
+ WebRtc_Word16 *period, /* (i) rough-pitch-period array (Q-2) */
+ WebRtc_Word16 *plocs, /* (i) where periods of period array are taken (Q-2) */
+ WebRtc_Word16 periodl, /* (i) dimension period array */
+ WebRtc_Word16 hl, /* (i) 2*hl+1 is the number of sequences */
+ WebRtc_Word16 *surround /* (i/o) The contribution from this sequence
+ summed with earlier contributions */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c
new file mode 100644
index 0000000..f202f62
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * high-pass filter of input with *0.5 and saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpInput(
+ WebRtc_Word16 *signal, /* (i/o) signal vector */
+ WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
+ {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+ is assumed to be 1.0 */
+ WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
+ yhi[n-2] ylow[n-2] */
+ WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
+ WebRtc_Word16 len) /* (i) Number of samples to filter */
+{
+ int i;
+ WebRtc_Word32 tmpW32;
+ WebRtc_Word32 tmpW32b;
+
+ for (i=0; i<len; i++) {
+
+ /*
+ y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+ + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+ */
+
+ tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */
+ tmpW32 = (tmpW32>>15);
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */
+ tmpW32 = (tmpW32<<1);
+
+ tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); /* b[1]*x[i-1] */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); /* b[2]*x[i-2] */
+
+ /* Update state (input part) */
+ x[1] = x[0];
+ x[0] = signal[i];
+
+ /* Rounding in Q(12+1), i.e. add 2^12 */
+ tmpW32b = tmpW32 + 4096;
+
+ /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
+ tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)268435455, tmpW32b, (WebRtc_Word32)-268435456);
+
+ /* Convert back to Q0 and multiply with 0.5 */
+ signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 13);
+
+ /* Update state (filtered part) */
+ y[2] = y[0];
+ y[3] = y[1];
+
+ /* upshift tmpW32 by 3 with saturation */
+ if (tmpW32>268435455) {
+ tmpW32 = WEBRTC_SPL_WORD32_MAX;
+ } else if (tmpW32<-268435456) {
+ tmpW32 = WEBRTC_SPL_WORD32_MIN;
+ } else {
+ tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
+ }
+
+ y[0] = (WebRtc_Word16)(tmpW32 >> 16);
+ y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h
new file mode 100644
index 0000000..f56c4f7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+
+#include "defines.h"
+
+void WebRtcIlbcfix_HpInput(
+ WebRtc_Word16 *signal, /* (i/o) signal vector */
+ WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
+ {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+ is assumed to be 1.0 */
+ WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
+ yhi[n-2] ylow[n-2] */
+ WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
+ WebRtc_Word16 len); /* (i) Number of samples to filter */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c
new file mode 100644
index 0000000..8e1c919
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * high-pass filter of output and *2 with saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpOutput(
+ WebRtc_Word16 *signal, /* (i/o) signal vector */
+ WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
+ {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+ is assumed to be 1.0 */
+ WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
+ yhi[n-2] ylow[n-2] */
+ WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
+ WebRtc_Word16 len) /* (i) Number of samples to filter */
+{
+ int i;
+ WebRtc_Word32 tmpW32;
+ WebRtc_Word32 tmpW32b;
+
+ for (i=0; i<len; i++) {
+
+ /*
+ y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+ + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+ */
+
+ tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */
+ tmpW32 = (tmpW32>>15);
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */
+ tmpW32 = (tmpW32<<1);
+
+ tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); /* b[1]*x[i-1] */
+ tmpW32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); /* b[2]*x[i-2] */
+
+ /* Update state (input part) */
+ x[1] = x[0];
+ x[0] = signal[i];
+
+ /* Rounding in Q(12-1), i.e. add 2^10 */
+ tmpW32b = tmpW32 + 1024;
+
+ /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
+ tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)67108863, tmpW32b, (WebRtc_Word32)-67108864);
+
+ /* Convert back to Q0 and multiply with 2 */
+ signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11);
+
+ /* Update state (filtered part) */
+ y[2] = y[0];
+ y[3] = y[1];
+
+ /* upshift tmpW32 by 3 with saturation */
+ if (tmpW32>268435455) {
+ tmpW32 = WEBRTC_SPL_WORD32_MAX;
+ } else if (tmpW32<-268435456) {
+ tmpW32 = WEBRTC_SPL_WORD32_MIN;
+ } else {
+ tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
+ }
+
+ y[0] = (WebRtc_Word16)(tmpW32 >> 16);
+ y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
+
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h
new file mode 100644
index 0000000..c9a7426
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+
+#include "defines.h"
+
+void WebRtcIlbcfix_HpOutput(
+ WebRtc_Word16 *signal, /* (i/o) signal vector */
+ WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
+ {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+ is assumed to be 1.0 */
+ WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
+ yhi[n-2] ylow[n-2] */
+ WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
+ WebRtc_Word16 len); /* (i) Number of samples to filter */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c
new file mode 100644
index 0000000..75b64c4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -0,0 +1,255 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBCInterface.c
+
+******************************************************************/
+
+#include "ilbc.h"
+#include "defines.h"
+#include "init_encode.h"
+#include "encode.h"
+#include "init_decode.h"
+#include "decode.h"
+#include <stdlib.h>
+
+
+WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst, WebRtc_Word16 *ILBCENC_inst_Addr, WebRtc_Word16 *size) {
+ *iLBC_encinst=(iLBC_encinst_t*)ILBCENC_inst_Addr;
+ *size=sizeof(iLBC_Enc_Inst_t)/sizeof(WebRtc_Word16);
+ if (*iLBC_encinst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst, WebRtc_Word16 *ILBCDEC_inst_Addr, WebRtc_Word16 *size) {
+ *iLBC_decinst=(iLBC_decinst_t*)ILBCDEC_inst_Addr;
+ *size=sizeof(iLBC_Dec_Inst_t)/sizeof(WebRtc_Word16);
+ if (*iLBC_decinst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst) {
+ *iLBC_encinst=(iLBC_encinst_t*)malloc(sizeof(iLBC_Enc_Inst_t));
+ if (*iLBC_encinst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst) {
+ *iLBC_decinst=(iLBC_decinst_t*)malloc(sizeof(iLBC_Dec_Inst_t));
+ if (*iLBC_decinst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst) {
+ free(iLBC_encinst);
+ return(0);
+}
+
+WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst) {
+ free(iLBC_decinst);
+ return(0);
+}
+
+
+WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst, WebRtc_Word16 mode)
+{
+ if ((mode==20)||(mode==30)) {
+ WebRtcIlbcfix_InitEncode((iLBC_Enc_Inst_t*) iLBCenc_inst, mode);
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, WebRtc_Word16 *speechIn, WebRtc_Word16 len, WebRtc_Word16 *encoded) {
+
+ WebRtc_Word16 pos = 0;
+ WebRtc_Word16 encpos = 0;
+
+ if ((len != ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
+#ifdef SPLIT_10MS
+ (len != 80) &&
+#endif
+ (len != 2*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
+ (len != 3*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl))
+ {
+ /* A maximum of 3 frames/packet is allowed */
+ return(-1);
+ } else {
+
+ /* call encoder */
+ while (pos<len) {
+ WebRtcIlbcfix_EncodeImpl((WebRtc_UWord16*) &encoded[encpos], &speechIn[pos], (iLBC_Enc_Inst_t*) iLBCenc_inst);
+#ifdef SPLIT_10MS
+ pos += 80;
+ if(((iLBC_Enc_Inst_t*)iLBCenc_inst)->section == 0)
+#else
+ pos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl;
+#endif
+ encpos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->no_of_words;
+ }
+ return (encpos*2);
+ }
+}
+
+WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 mode) {
+ if ((mode==20)||(mode==30)) {
+ WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, mode, 1);
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst) {
+ WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 20, 1);
+ return(0);
+}
+WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst) {
+ WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 30, 1);
+ return(0);
+}
+
+
+WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ int i=0;
+ /* Allow for automatic switching between the frame sizes
+ (although you do get some discontinuity) */
+ if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+ /* ok, do nothing */
+ } else {
+ /* Test if the mode has changed */
+ if (((iLBC_Dec_Inst_t*)iLBCdec_inst)->mode==20) {
+ if ((len==NO_OF_BYTES_30MS)||
+ (len==2*NO_OF_BYTES_30MS)||
+ (len==3*NO_OF_BYTES_30MS)) {
+ WebRtcIlbcfix_InitDecode(((iLBC_Dec_Inst_t*)iLBCdec_inst), 30, ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
+ } else {
+ /* Unsupported frame length */
+ return(-1);
+ }
+ } else {
+ if ((len==NO_OF_BYTES_20MS)||
+ (len==2*NO_OF_BYTES_20MS)||
+ (len==3*NO_OF_BYTES_20MS)) {
+ WebRtcIlbcfix_InitDecode(((iLBC_Dec_Inst_t*)iLBCdec_inst), 20, ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
+ } else {
+ /* Unsupported frame length */
+ return(-1);
+ }
+ }
+ }
+
+ while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+ WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+ i++;
+ }
+ /* iLBC does not support VAD/CNG yet */
+ *speechType=1;
+ return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+}
+
+WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ int i=0;
+ if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+ /* ok, do nothing */
+ } else {
+ return(-1);
+ }
+
+ while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+ WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+ i++;
+ }
+ /* iLBC does not support VAD/CNG yet */
+ *speechType=1;
+ return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+}
+
+WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType)
+{
+ int i=0;
+ if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
+ (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
+ /* ok, do nothing */
+ } else {
+ return(-1);
+ }
+
+ while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
+ WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
+ i++;
+ }
+ /* iLBC does not support VAD/CNG yet */
+ *speechType=1;
+ return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+}
+
+WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
+ int i;
+ WebRtc_UWord16 dummy;
+
+ for (i=0;i<noOfLostFrames;i++) {
+ /* call decoder */
+ WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], &dummy, (iLBC_Dec_Inst_t*) iLBCdec_inst, 0);
+ }
+ return (noOfLostFrames*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
+}
+
+WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
+
+ /* Two input parameters not used, but needed for function pointers in NetEQ */
+ (void)(decoded = NULL);
+ (void)(noOfLostFrames = 0);
+
+ WebRtcSpl_MemSetW16(((iLBC_Dec_Inst_t*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
+ ((iLBC_Dec_Inst_t*)iLBCdec_inst)->prev_enh_pl = 2;
+
+ return (0);
+}
+
+void WebRtcIlbcfix_version(char *version)
+{
+ strcpy((char*)version, "1.1.1");
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi
new file mode 100644
index 0000000..57410c5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi
@@ -0,0 +1,190 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'iLBC',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ 'interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'interface',
+ ],
+ },
+ 'sources': [
+ 'interface/ilbc.h',
+ 'abs_quant.c',
+ 'abs_quant_loop.c',
+ 'augmented_cb_corr.c',
+ 'bw_expand.c',
+ 'cb_construct.c',
+ 'cb_mem_energy.c',
+ 'cb_mem_energy_augmentation.c',
+ 'cb_mem_energy_calc.c',
+ 'cb_search.c',
+ 'cb_search_core.c',
+ 'cb_update_best_index.c',
+ 'chebyshev.c',
+ 'comp_corr.c',
+ 'constants.c',
+ 'create_augmented_vec.c',
+ 'decode.c',
+ 'decode_residual.c',
+ 'decoder_interpolate_lsf.c',
+ 'do_plc.c',
+ 'encode.c',
+ 'energy_inverse.c',
+ 'enh_upsample.c',
+ 'enhancer.c',
+ 'enhancer_interface.c',
+ 'filtered_cb_vecs.c',
+ 'frame_classify.c',
+ 'gain_dequant.c',
+ 'gain_quant.c',
+ 'get_cd_vec.c',
+ 'get_lsp_poly.c',
+ 'get_sync_seq.c',
+ 'hp_input.c',
+ 'hp_output.c',
+ 'ilbc.c',
+ 'index_conv_dec.c',
+ 'index_conv_enc.c',
+ 'init_decode.c',
+ 'init_encode.c',
+ 'interpolate.c',
+ 'interpolate_samples.c',
+ 'lpc_encode.c',
+ 'lsf_check.c',
+ 'lsf_interpolate_to_poly_dec.c',
+ 'lsf_interpolate_to_poly_enc.c',
+ 'lsf_to_lsp.c',
+ 'lsf_to_poly.c',
+ 'lsp_to_lsf.c',
+ 'my_corr.c',
+ 'nearest_neighbor.c',
+ 'pack_bits.c',
+ 'poly_to_lsf.c',
+ 'poly_to_lsp.c',
+ 'refiner.c',
+ 'simple_interpolate_lsf.c',
+ 'simple_lpc_analysis.c',
+ 'simple_lsf_dequant.c',
+ 'simple_lsf_quant.c',
+ 'smooth.c',
+ 'smooth_out_data.c',
+ 'sort_sq.c',
+ 'split_vq.c',
+ 'state_construct.c',
+ 'state_search.c',
+ 'swap_bytes.c',
+ 'unpack_bits.c',
+ 'vq3.c',
+ 'vq4.c',
+ 'window32_w32.c',
+ 'xcorr_coef.c',
+ 'abs_quant.h',
+ 'abs_quant_loop.h',
+ 'augmented_cb_corr.h',
+ 'bw_expand.h',
+ 'cb_construct.h',
+ 'cb_mem_energy.h',
+ 'cb_mem_energy_augmentation.h',
+ 'cb_mem_energy_calc.h',
+ 'cb_search.h',
+ 'cb_search_core.h',
+ 'cb_update_best_index.h',
+ 'chebyshev.h',
+ 'comp_corr.h',
+ 'constants.h',
+ 'create_augmented_vec.h',
+ 'decode.h',
+ 'decode_residual.h',
+ 'decoder_interpolate_lsf.h',
+ 'do_plc.h',
+ 'encode.h',
+ 'energy_inverse.h',
+ 'enh_upsample.h',
+ 'enhancer.h',
+ 'enhancer_interface.h',
+ 'filtered_cb_vecs.h',
+ 'frame_classify.h',
+ 'gain_dequant.h',
+ 'gain_quant.h',
+ 'get_cd_vec.h',
+ 'get_lsp_poly.h',
+ 'get_sync_seq.h',
+ 'hp_input.h',
+ 'hp_output.h',
+ 'defines.h',
+ 'index_conv_dec.h',
+ 'index_conv_enc.h',
+ 'init_decode.h',
+ 'init_encode.h',
+ 'interpolate.h',
+ 'interpolate_samples.h',
+ 'lpc_encode.h',
+ 'lsf_check.h',
+ 'lsf_interpolate_to_poly_dec.h',
+ 'lsf_interpolate_to_poly_enc.h',
+ 'lsf_to_lsp.h',
+ 'lsf_to_poly.h',
+ 'lsp_to_lsf.h',
+ 'my_corr.h',
+ 'nearest_neighbor.h',
+ 'pack_bits.h',
+ 'poly_to_lsf.h',
+ 'poly_to_lsp.h',
+ 'refiner.h',
+ 'simple_interpolate_lsf.h',
+ 'simple_lpc_analysis.h',
+ 'simple_lsf_dequant.h',
+ 'simple_lsf_quant.h',
+ 'smooth.h',
+ 'smooth_out_data.h',
+ 'sort_sq.h',
+ 'split_vq.h',
+ 'state_construct.h',
+ 'state_search.h',
+ 'swap_bytes.h',
+ 'unpack_bits.h',
+ 'vq3.h',
+ 'vq4.h',
+ 'window32_w32.h',
+ 'xcorr_coef.h',
+ ], # sources
+ }, # iLBC
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'iLBCtest',
+ 'type': 'executable',
+ 'dependencies': [
+ 'iLBC',
+ ],
+ 'sources': [
+ 'test/iLBC_test.c',
+ ],
+ }, # iLBCtest
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c
new file mode 100644
index 0000000..0d6346a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.c
+
+******************************************************************/
+
+#include "defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(
+ WebRtc_Word16 *index /* (i/o) Codebook indexes */
+ ){
+ int k;
+
+ for (k=4;k<6;k++) {
+ /* Readjust the second and third codebook index for the first 40 sample
+ so that they look the same as the first (in terms of lag)
+ */
+ if ((index[k]>=44)&&(index[k]<108)) {
+ index[k]+=64;
+ } else if ((index[k]>=108)&&(index[k]<128)) {
+ index[k]+=128;
+ } else {
+ /* ERROR */
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h
new file mode 100644
index 0000000..f29ee23
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+
+#include "defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(
+ WebRtc_Word16 *index /* (i/o) Codebook indexes */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c
new file mode 100644
index 0000000..cbc04b6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ IiLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.c
+
+******************************************************************/
+
+#include "defines.h"
+/*----------------------------------------------------------------*
+ * Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(
+ WebRtc_Word16 *index /* (i/o) Codebook indexes */
+ ){
+ int k;
+
+ for (k=4;k<6;k++) {
+ /* Readjust the second and third codebook index so that it is
+ packetized into 7 bits (before it was put in lag-wise the same
+ way as for the first codebook which uses 8 bits)
+ */
+ if ((index[k]>=108)&&(index[k]<172)) {
+ index[k]-=64;
+ } else if (index[k]>=236) {
+ index[k]-=128;
+ } else {
+ /* ERROR */
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h
new file mode 100644
index 0000000..d28a6e2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(
+ WebRtc_Word16 *index /* (i/o) Codebook indexes */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c
new file mode 100644
index 0000000..b654f1e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -0,0 +1,98 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitDecode.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ WebRtc_Word16 mode, /* (i) frame size mode */
+ int use_enhancer /* (i) 1 to use enhancer
+ 0 to run without enhancer */
+ ) {
+ int i;
+
+ iLBCdec_inst->mode = mode;
+
+ /* Set all the variables that are dependent on the frame size mode */
+ if (mode==30) {
+ iLBCdec_inst->blockl = BLOCKL_30MS;
+ iLBCdec_inst->nsub = NSUB_30MS;
+ iLBCdec_inst->nasub = NASUB_30MS;
+ iLBCdec_inst->lpc_n = LPC_N_30MS;
+ iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
+ iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
+ iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
+ }
+ else if (mode==20) {
+ iLBCdec_inst->blockl = BLOCKL_20MS;
+ iLBCdec_inst->nsub = NSUB_20MS;
+ iLBCdec_inst->nasub = NASUB_20MS;
+ iLBCdec_inst->lpc_n = LPC_N_20MS;
+ iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
+ iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
+ iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
+ }
+ else {
+ return(-1);
+ }
+
+ /* Reset all the previous LSF to mean LSF */
+ WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+
+ /* Clear the synthesis filter memory */
+ WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+
+ /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */
+ WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX));
+ for (i=0; i<NSUB_MAX; i++) {
+ iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096;
+ }
+
+ /* Clear the variables that are used for the PLC */
+ iLBCdec_inst->last_lag = 20;
+ iLBCdec_inst->consPLICount = 0;
+ iLBCdec_inst->prevPLI = 0;
+ iLBCdec_inst->perSquare = 0;
+ iLBCdec_inst->prevLag = 120;
+ iLBCdec_inst->prevLpc[0] = 4096;
+ WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER);
+ WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX);
+
+ /* Initialize the seed for the random number generator */
+ iLBCdec_inst->seed = 777;
+
+ /* Set the filter state of the HP filter to 0 */
+ WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+ WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+
+ /* Set the variables that are used in the ehnahcer */
+ iLBCdec_inst->use_enhancer = use_enhancer;
+ WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD));
+ for (i=0;i<ENH_NBLOCKS_TOT;i++) {
+ iLBCdec_inst->enh_period[i]=160; /* Q(-4) */
+ }
+
+ iLBCdec_inst->prev_enh_pl = 0;
+
+ return (iLBCdec_inst->blockl);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h
new file mode 100644
index 0000000..3452f34
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitDecode.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ WebRtc_Word16 mode, /* (i) frame size mode */
+ int use_enhancer /* (i) 1 to use enhancer
+ 0 to run without enhancer */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c
new file mode 100644
index 0000000..e034bb0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ WebRtc_Word16 mode /* (i) frame size mode */
+ ){
+ iLBCenc_inst->mode = mode;
+
+ /* Set all the variables that are dependent on the frame size mode */
+ if (mode==30) {
+ iLBCenc_inst->blockl = BLOCKL_30MS;
+ iLBCenc_inst->nsub = NSUB_30MS;
+ iLBCenc_inst->nasub = NASUB_30MS;
+ iLBCenc_inst->lpc_n = LPC_N_30MS;
+ iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
+ iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
+ iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
+ }
+ else if (mode==20) {
+ iLBCenc_inst->blockl = BLOCKL_20MS;
+ iLBCenc_inst->nsub = NSUB_20MS;
+ iLBCenc_inst->nasub = NASUB_20MS;
+ iLBCenc_inst->lpc_n = LPC_N_20MS;
+ iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
+ iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
+ iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
+ }
+ else {
+ return(-1);
+ }
+
+ /* Clear the buffers and set the previous LSF and LSP to the mean value */
+ WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER);
+ WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+ WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+ WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX);
+
+ /* Set the filter state of the HP filter to 0 */
+ WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2);
+ WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4);
+
+#ifdef SPLIT_10MS
+ /*Zeroing the past samples for 10msec Split*/
+ WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160);
+ iLBCenc_inst->section = 0;
+#endif
+
+ return (iLBCenc_inst->no_of_bytes);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h
new file mode 100644
index 0000000..f1d1858
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ WebRtc_Word16 mode /* (i) frame size mode */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h
new file mode 100644
index 0000000..ca1d39e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h
@@ -0,0 +1,259 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * ilbc.h
+ *
+ * This header file contains all of the API's for iLBC.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
+
+/*
+ * Define the fixpoint numeric formats
+ */
+
+#include "typedefs.h"
+
+/*
+ * Solution to support multiple instances
+ * Customer has to cast instance to proper type
+ */
+
+typedef struct iLBC_encinst_t_ iLBC_encinst_t;
+
+typedef struct iLBC_decinst_t_ iLBC_decinst_t;
+
+/*
+ * Comfort noise constants
+ */
+
+#define ILBC_SPEECH 1
+#define ILBC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+ /****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions assigns the encoder/decoder instance to the specified
+ * memory location
+ *
+ * Input:
+ * - XXX_xxxinst : Pointer to created instance that should be
+ * assigned
+ * - ILBCXXX_inst_Addr : Pointer to the desired memory space
+ * - size : The size that this structure occupies (in Word16)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst,
+ WebRtc_Word16 *ILBCENC_inst_Addr,
+ WebRtc_Word16 *size);
+ WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst,
+ WebRtc_Word16 *ILBCDEC_inst_Addr,
+ WebRtc_Word16 *size);
+
+
+ /****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions create a instance to the specified structure
+ *
+ * Input:
+ * - XXX_inst : Pointer to created instance that should be created
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst);
+ WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_XxxFree(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ * - XXX_inst : Pointer to created instance that should be freed
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst);
+ WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst);
+
+
+ /****************************************************************************
+ * WebRtcIlbcfix_EncoderInit(...)
+ *
+ * This function initializes a iLBC instance
+ *
+ * Input:
+ * - iLBCenc_inst : iLBC instance, i.e. the user that should receive
+ * be initialized
+ * - frameLen : The frame length of the codec 20/30 (ms)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst,
+ WebRtc_Word16 frameLen);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_Encode(...)
+ *
+ * This function encodes one iLBC frame. Input speech length has be a
+ * multiple of the frame length.
+ *
+ * Input:
+ * - iLBCenc_inst : iLBC instance, i.e. the user that should encode
+ * a package
+ * - speechIn : Input speech vector
+ * - len : Samples in speechIn (160, 240, 320 or 480)
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst,
+ WebRtc_Word16 *speechIn,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *encoded);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_DecoderInit(...)
+ *
+ * This function initializes a iLBC instance with either 20 or 30 ms frames
+ * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
+ * not needed to specify the frame length with a variable.
+ *
+ * Input:
+ * - iLBC_decinst_t : iLBC instance, i.e. the user that should receive
+ * be initialized
+ * - frameLen : The frame length of the codec 20/30 (ms)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 frameLen);
+ WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst);
+ WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_Decode(...)
+ *
+ * This function decodes a packet with iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
+ *
+ * Input:
+ * - iLBCdec_inst : iLBC instance, i.e. the user that should decode
+ * a packet
+ * - encoded : Encoded iLBC frame(s)
+ * - len : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - speechType : 1 normal, 2 CNG
+ *
+ * Return value : >0 - Samples in decoded vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16* encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+ WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+ WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *encoded,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 *speechType);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_DecodePlc(...)
+ *
+ * This function conducts PLC for iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples.
+ *
+ * Input:
+ * - iLBCdec_inst : iLBC instance, i.e. the user that should perform
+ * a PLC
+ * - noOfLostFrames : Number of PLC frames to produce
+ *
+ * Output:
+ * - decoded : The "decoded" vector
+ *
+ * Return value : >0 - Samples in decoded PLC vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_NetEqPlc(...)
+ *
+ * This function updates the decoder when a packet loss has occured, but it
+ * does not produce any PLC data. Function can be used if another PLC method
+ * is used (i.e NetEq).
+ *
+ * Input:
+ * - iLBCdec_inst : iLBC instance that should be updated
+ * - noOfLostFrames : Number of lost frames
+ *
+ * Output:
+ * - decoded : The "decoded" vector (nothing in this case)
+ *
+ * Return value : >0 - Samples in decoded PLC vector
+ * -1 - Error
+ */
+
+ WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst,
+ WebRtc_Word16 *decoded,
+ WebRtc_Word16 noOfLostFrames);
+
+ /****************************************************************************
+ * WebRtcIlbcfix_version(...)
+ *
+ * This function returns the version number of iLBC
+ *
+ * Output:
+ * - version : Version number of iLBC (maximum 20 char)
+ */
+
+ void WebRtcIlbcfix_version(char *version);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c
new file mode 100644
index 0000000..11cb33c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+ WebRtc_Word16 *out, /* (o) output vector */
+ WebRtc_Word16 *in1, /* (i) first input vector */
+ WebRtc_Word16 *in2, /* (i) second input vector */
+ WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
+ WebRtc_Word16 length) /* (i) number of sample is vectors */
+{
+ int i;
+ WebRtc_Word16 invcoef;
+
+ /*
+ Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
+ */
+
+ invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
+ for (i = 0; i < length; i++) {
+ out[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(coef, in1[i]) + WEBRTC_SPL_MUL_16_16(invcoef, in2[i]))+8192,
+ 14);
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h
new file mode 100644
index 0000000..a12021c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+ WebRtc_Word16 *out, /* (o) output vector */
+ WebRtc_Word16 *in1, /* (i) first input vector */
+ WebRtc_Word16 *in2, /* (i) second input vector */
+ WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
+ WebRtc_Word16 length); /* (i) number of sample is vectors */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c
new file mode 100644
index 0000000..31eb52e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+void WebRtcIlbcfix_InterpolateSamples(
+ WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
+ WebRtc_Word16 *CBmem, /* (i) The CB memory */
+ WebRtc_Word16 lMem /* (i) Length of the CB memory */
+ ) {
+ WebRtc_Word16 *ppi, *ppo, i, j, temp1, temp2;
+ WebRtc_Word16 *tmpPtr;
+
+ /* Calculate the 20 vectors of interpolated samples (4 samples each)
+ that are used in the codebooks for lag 20 to 39 */
+ tmpPtr = interpSamples;
+ for (j=0; j<20; j++) {
+ temp1 = 0;
+ temp2 = 3;
+ ppo = CBmem+lMem-4;
+ ppi = CBmem+lMem-j-24;
+ for (i=0; i<4; i++) {
+
+ *tmpPtr++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) +
+ (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
+
+ ppo++;
+ ppi++;
+ temp1++;
+ temp2--;
+ }
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h
new file mode 100644
index 0000000..5c98aaf
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the interpolated samples for the Augmented CB
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_InterpolateSamples(
+ WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
+ WebRtc_Word16 *CBmem, /* (i) The CB memory */
+ WebRtc_Word16 lMem /* (i) Length of the CB memory */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c
new file mode 100644
index 0000000..73d67a0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "simple_lpc_analysis.h"
+#include "simple_interpolate_lsf.h"
+#include "simple_lsf_quant.h"
+#include "lsf_check.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+ WebRtc_Word16 *syntdenum, /* (i/o) synthesis filter coefficients
+ before/after encoding */
+ WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
+ before/after encoding */
+ WebRtc_Word16 *lsf_index, /* (o) lsf quantization index */
+ WebRtc_Word16 *data, /* (i) Speech to do LPC analysis on */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ) {
+ /* Stack based */
+ WebRtc_Word16 lsf[LPC_FILTERORDER * LPC_N_MAX];
+ WebRtc_Word16 lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+
+ /* Calculate LSF's from the input speech */
+ WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
+
+ /* Quantize the LSF's */
+ WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
+
+ /* Stableize the LSF's if needed */
+ WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
+
+ /* Calculate the synthesis and weighting filter coefficients from
+ the optimal LSF and the dequantized LSF */
+ WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum,
+ lsf, lsfdeq, iLBCenc_inst->lsfold,
+ iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h
new file mode 100644
index 0000000..36967a3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+ WebRtc_Word16 *syntdenum, /* (i/o) synthesis filter coefficients
+ before/after encoding */
+ WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
+ before/after encoding */
+ WebRtc_Word16 *lsf_index, /* (o) lsf quantization index */
+ WebRtc_Word16 *data, /* (i) Speech to do LPC analysis on */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c
new file mode 100644
index 0000000..7097d74
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(
+ WebRtc_Word16 *lsf, /* LSF parameters */
+ int dim, /* dimension of LSF */
+ int NoAn) /* No of analysis per frame */
+{
+ int k,n,m, Nit=2, change=0,pos;
+ const WebRtc_Word16 eps=319; /* 0.039 in Q13 (50 Hz)*/
+ const WebRtc_Word16 eps2=160; /* eps/2.0 in Q13;*/
+ const WebRtc_Word16 maxlsf=25723; /* 3.14; (4000 Hz)*/
+ const WebRtc_Word16 minlsf=82; /* 0.01; (0 Hz)*/
+
+ /* LSF separation check*/
+ for (n=0;n<Nit;n++) { /* Run through a 2 times */
+ for (m=0;m<NoAn;m++) { /* Number of analyses per frame */
+ for (k=0;k<(dim-1);k++) {
+ pos=m*dim+k;
+
+ /* Seperate coefficients with a safety margin of 50 Hz */
+ if ((lsf[pos+1]-lsf[pos])<eps) {
+
+ if (lsf[pos+1]<lsf[pos]) {
+ lsf[pos+1]= lsf[pos]+eps2;
+ lsf[pos]= lsf[pos+1]-eps2;
+ } else {
+ lsf[pos]-=eps2;
+ lsf[pos+1]+=eps2;
+ }
+ change=1;
+ }
+
+ /* Limit minimum and maximum LSF */
+ if (lsf[pos]<minlsf) {
+ lsf[pos]=minlsf;
+ change=1;
+ }
+
+ if (lsf[pos]>maxlsf) {
+ lsf[pos]=maxlsf;
+ change=1;
+ }
+ }
+ }
+ }
+
+ return change;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h
new file mode 100644
index 0000000..830bbed
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(
+ WebRtc_Word16 *lsf, /* LSF parameters */
+ int dim, /* dimension of LSF */
+ int NoAn); /* No of analysis per frame */
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
new file mode 100644
index 0000000..3bb23d0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.c
+
+******************************************************************/
+
+#include "interpolate.h"
+#include "lsf_to_poly.h"
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+ WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
+ WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
+ WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
+ WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+ lsf1 and lsf2 Q14 */
+ WebRtc_Word16 length /* (i) length of coefficient vectors */
+ ){
+ WebRtc_Word16 lsftmp[LPC_FILTERORDER];
+
+ /* interpolate LSF */
+ WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+ /* Compute the filter coefficients from the LSF */
+ WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
new file mode 100644
index 0000000..23fe3a7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+ WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
+ WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
+ WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
+ WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+ lsf1 and lsf2 Q14 */
+ WebRtc_Word16 length /* (i) length of coefficient vectors */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
new file mode 100644
index 0000000..3b0a34d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "interpolate.h"
+#include "lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ * lsf interpolator and conversion from lsf to a coefficients
+ * (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+ WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
+ WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
+ WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
+ WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+ lsf1 and lsf2 Q14 */
+ WebRtc_Word16 length /* (i) length of coefficient vectors */
+ ) {
+ /* Stack based */
+ WebRtc_Word16 lsftmp[LPC_FILTERORDER];
+
+ /* interpolate LSF */
+ WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+ /* Compute the filter coefficients from the LSF */
+ WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
new file mode 100644
index 0000000..1bbbb80
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lsf interpolator and conversion from lsf to a coefficients
+ * (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+ WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
+ WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
+ WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
+ WebRtc_Word16 coef, /* (i) weighting coefficient to use between
+ lsf1 and lsf2 Q14 */
+ WebRtc_Word16 length /* (i) length of coefficient vectors */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
new file mode 100644
index 0000000..84278a4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+ WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
+ WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+ WebRtc_Word16 m /* (i) number of coefficients */
+ ) {
+ WebRtc_Word16 i, k;
+ WebRtc_Word16 diff; /* difference, which is used for the
+ linear approximation (Q8) */
+ WebRtc_Word16 freq; /* normalized frequency in Q15 (0..1) */
+ WebRtc_Word32 tmpW32;
+
+ for(i=0; i<m; i++)
+ {
+ freq = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15);
+ /* 20861: 1.0/(2.0*PI) in Q17 */
+ /*
+ Upper 8 bits give the index k and
+ Lower 8 bits give the difference, which needs
+ to be approximated linearly
+ */
+ k = WEBRTC_SPL_RSHIFT_W16(freq, 8);
+ diff = (freq&0x00ff);
+
+ /* Guard against getting outside table */
+
+ if (k>63) {
+ k = 63;
+ }
+
+ /* Calculate linear approximation */
+ tmpW32 = WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kCosDerivative[k], diff);
+ lsp[i] = WebRtcIlbcfix_kCos[k]+(WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
new file mode 100644
index 0000000..db6549b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+ WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
+ WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+ WebRtc_Word16 m /* (i) number of coefficients */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
new file mode 100644
index 0000000..f1c4a9e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "lsf_to_lsp.h"
+#include "get_lsp_poly.h"
+#include "constants.h"
+
+void WebRtcIlbcfix_Lsf2Poly(
+ WebRtc_Word16 *a, /* (o) predictor coefficients (order = 10) in Q12 */
+ WebRtc_Word16 *lsf /* (i) line spectral frequencies in Q13 */
+ ) {
+ WebRtc_Word32 f[2][6]; /* f[0][] and f[1][] corresponds to
+ F1(z) and F2(z) respectivly */
+ WebRtc_Word32 *f1ptr, *f2ptr;
+ WebRtc_Word16 *a1ptr, *a2ptr;
+ WebRtc_Word32 tmpW32;
+ WebRtc_Word16 lsp[10];
+ int i;
+
+ /* Convert lsf to lsp */
+ WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER);
+
+ /* Get F1(z) and F2(z) from the lsp */
+ f1ptr=f[0];
+ f2ptr=f[1];
+ WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr);
+ WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr);
+
+ /* for i = 5 down to 1
+ Compute f1[i] += f1[i-1];
+ and f2[i] += f2[i-1];
+ */
+ f1ptr=&f[0][5];
+ f2ptr=&f[1][5];
+ for (i=5; i>0; i--)
+ {
+ (*f1ptr) += (*(f1ptr-1));
+ (*f2ptr) -= (*(f2ptr-1));
+ f1ptr--;
+ f2ptr--;
+ }
+
+ /* Get the A(z) coefficients
+ a[0] = 1.0
+ for i = 1 to 5
+ a[i] = (f1[i] + f2[i] + round)>>13;
+ for i = 1 to 5
+ a[11-i] = (f1[i] - f2[i] + round)>>13;
+ */
+ a[0]=4096;
+ a1ptr=&a[1];
+ a2ptr=&a[10];
+ f1ptr=&f[0][1];
+ f2ptr=&f[1][1];
+ for (i=5; i>0; i--)
+ {
+ tmpW32 = (*f1ptr) + (*f2ptr);
+ (*a1ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
+
+ tmpW32 = (*f1ptr) - (*f2ptr);
+ (*a2ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
+
+ a1ptr++;
+ a2ptr--;
+ f1ptr++;
+ f2ptr++;
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
new file mode 100644
index 0000000..a00693b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Convert from LSF coefficients to A coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Poly(
+ WebRtc_Word16 *a, /* (o) predictor coefficients (order = 10) in Q12 */
+ WebRtc_Word16 *lsf /* (i) line spectral frequencies in Q13 */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
new file mode 100644
index 0000000..134afbb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+ WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
+ WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
+ (ordered, so that lsf[i]<lsf[i+1]) */
+ WebRtc_Word16 m /* (i) Number of coefficients */
+ )
+{
+ WebRtc_Word16 i, k;
+ WebRtc_Word16 diff; /* diff between table value and desired value (Q15) */
+ WebRtc_Word16 freq; /* lsf/(2*pi) (Q16) */
+ WebRtc_Word16 *lspPtr, *lsfPtr, *cosTblPtr;
+ WebRtc_Word16 tmp;
+
+ /* set the index to maximum index value in WebRtcIlbcfix_kCos */
+ k = 63;
+
+ /*
+ Start with the highest LSP and then work the way down
+ For each LSP the lsf is calculated by first order approximation
+ of the acos(x) function
+ */
+ lspPtr = &lsp[9];
+ lsfPtr = &lsf[9];
+ cosTblPtr=(WebRtc_Word16*)&WebRtcIlbcfix_kCos[k];
+ for(i=m-1; i>=0; i--)
+ {
+ /*
+ locate value in the table, which is just above lsp[i],
+ basically an approximation to acos(x)
+ */
+ while( (((WebRtc_Word32)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
+ {
+ k-=1;
+ cosTblPtr--;
+ }
+
+ /* Calculate diff, which is used in the linear approximation of acos(x) */
+ diff = (*lspPtr)-(*cosTblPtr);
+
+ /*
+ The linear approximation of acos(lsp[i]) :
+ acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11)
+ */
+
+ /* tmp (linear offset) in Q16 */
+ tmp = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAcosDerivative[k],diff, 11);
+
+ /* freq in Q16 */
+ freq = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W16(k,9)+tmp;
+
+ /* lsf = freq*2*pi */
+ (*lsfPtr) = (WebRtc_Word16)(((WebRtc_Word32)freq*25736)>>15);
+
+ lsfPtr--;
+ lspPtr--;
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
new file mode 100644
index 0000000..97ba7e4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+ WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
+ WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
+ (ordered, so that lsf[i]<lsf[i+1]) */
+ WebRtc_Word16 m /* (i) Number of coefficients */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c
new file mode 100644
index 0000000..2162205
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(
+ WebRtc_Word32 *corr, /* (o) correlation of seq1 and seq2 */
+ WebRtc_Word16 *seq1, /* (i) first sequence */
+ WebRtc_Word16 dim1, /* (i) dimension first seq1 */
+ const WebRtc_Word16 *seq2, /* (i) second sequence */
+ WebRtc_Word16 dim2 /* (i) dimension seq2 */
+ ){
+ WebRtc_Word16 max, scale, loops;
+
+ /* Calculate correlation between the two sequences. Scale the
+ result of the multiplcication to maximum 26 bits in order
+ to avoid overflow */
+ max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
+ scale=WebRtcSpl_GetSizeInBits(max);
+
+ scale = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(2,scale)-26);
+ if (scale<0) {
+ scale=0;
+ }
+
+ loops=dim1-dim2+1;
+
+ /* Calculate the cross correlations */
+ WebRtcSpl_CrossCorrelation(corr, (WebRtc_Word16*)seq2, seq1, dim2, loops, scale, 1);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h
new file mode 100644
index 0000000..f588c53
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(
+ WebRtc_Word32 *corr, /* (o) correlation of seq1 and seq2 */
+ WebRtc_Word16 *seq1, /* (i) first sequence */
+ WebRtc_Word16 dim1, /* (i) dimension first seq1 */
+ const WebRtc_Word16 *seq2, /* (i) second sequence */
+ WebRtc_Word16 dim2 /* (i) dimension seq2 */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
new file mode 100644
index 0000000..ea9e1eb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ * according to the squared-error criterion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_NearestNeighbor(
+ WebRtc_Word16 *index, /* (o) index of array element closest to value */
+ WebRtc_Word16 *array, /* (i) data array (Q2) */
+ WebRtc_Word16 value, /* (i) value (Q2) */
+ WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
+ ){
+ int i;
+ WebRtc_Word16 diff;
+ /* Stack based */
+ WebRtc_Word32 crit[8];
+
+ /* Calculate square distance */
+ for(i=0;i<arlength;i++){
+ diff=array[i]-value;
+ crit[i]=WEBRTC_SPL_MUL_16_16(diff, diff);
+ }
+
+ /* Find the minimum square distance */
+ *index=WebRtcSpl_MinIndexW32(crit, (WebRtc_Word16)arlength);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
new file mode 100644
index 0000000..705e17a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ * according to the squared-error criterion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_NearestNeighbor(
+ WebRtc_Word16 *index, /* (o) index of array element closest to value */
+ WebRtc_Word16 *array, /* (i) data array (Q2) */
+ WebRtc_Word16 value, /* (i) value (Q2) */
+ WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c
new file mode 100644
index 0000000..3990fbe
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c
@@ -0,0 +1,251 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+ WebRtc_UWord16 *bitstream, /* (o) The packetized bitstream */
+ iLBC_bits *enc_bits, /* (i) Encoded bits */
+ WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
+ ){
+ WebRtc_UWord16 *bitstreamPtr;
+ int i, k;
+ WebRtc_Word16 *tmpPtr;
+
+ bitstreamPtr=bitstream;
+
+ /* Class 1 bits of ULP */
+ /* First WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[0])<<10; /* Bit 0..5 */
+ (*bitstreamPtr) |= (enc_bits->lsf[1])<<3; /* Bit 6..12 */
+ (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* Second WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[2]&0xF)<<12; /* Bit 0..3 */
+
+ if (mode==20) {
+ (*bitstreamPtr) |= (enc_bits->startIdx)<<10; /* Bit 4..5 */
+ (*bitstreamPtr) |= (enc_bits->state_first)<<9; /* Bit 6 */
+ (*bitstreamPtr) |= (enc_bits->idxForMax)<<3; /* Bit 7..12 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* Third WebRtc_Word16 */
+ (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12; /* Bit 0..2 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8; /* Bit 3..4 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7; /* Bit 5 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2; /* Bit 6..12 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2; /* Bit 13 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2; /* Bit 14 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4; /* Bit 15 */
+ } else { /* mode==30 */
+ (*bitstreamPtr) |= (enc_bits->lsf[3])<<6; /* Bit 4..9 */
+ (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* Third WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[4]&0x1)<<15; /* Bit 0 */
+ (*bitstreamPtr) |= (enc_bits->lsf[5])<<8; /* Bit 1..7 */
+ (*bitstreamPtr) |= (enc_bits->startIdx)<<5; /* Bit 8..10 */
+ (*bitstreamPtr) |= (enc_bits->state_first)<<4; /* Bit 11 */
+ (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 4:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7; /* Bit 2..5 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5; /* Bit 6 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5; /* Bit 7 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC); /* Bit 8..13 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3; /* Bit 14 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3; /* Bit 15 */
+ }
+ /* Class 2 bits of ULP */
+ /* 4:th to 6:th WebRtc_Word16 for 20 ms case
+ 5:th to 7:th WebRtc_Word16 for 30 ms case */
+ bitstreamPtr++;
+ tmpPtr=enc_bits->idxVec;
+ for (k=0; k<3; k++) {
+ (*bitstreamPtr) = 0;
+ for (i=15; i>=0; i--) {
+ (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ bitstreamPtr++;
+ }
+
+ if (mode==20) {
+ /* 7:th WebRtc_Word16 */
+ (*bitstreamPtr) = 0;
+ for (i=15; i>6; i--) {
+ (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4; /* Bit 9 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2; /* Bit 10..11 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1; /* Bit 12 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1; /* Bit 13 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2; /* Bit 14..15 */
+
+ } else { /* mode==30 */
+ /* 8:th WebRtc_Word16 */
+ (*bitstreamPtr) = 0;
+ for (i=15; i>5; i--) {
+ (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3; /* Bit 10..11 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8); /* Bit 12 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4); /* Bit 13 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2); /* Bit 14 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7; /* Bit 15 */
+ bitstreamPtr++;
+ /* 9:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2; /* Bit 6..12 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* 10:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8; /* Bit 4..5 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7; /* Bit 6..7 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3; /* Bit 8..9 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2; /* Bit 10..11 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1; /* Bit 12 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1; /* Bit 13 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3; /* Bit 14 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3; /* Bit 15 */
+ }
+ bitstreamPtr++;
+ /* Class 3 bits of ULP */
+ /* 8:th to 14:th WebRtc_Word16 for 20 ms case
+ 11:th to 17:th WebRtc_Word16 for 30 ms case */
+ tmpPtr=enc_bits->idxVec;
+ for (k=0; k<7; k++) {
+ (*bitstreamPtr) = 0;
+ for (i=14; i>=0; i-=2) {
+ (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
+ tmpPtr++;
+ }
+ bitstreamPtr++;
+ }
+
+ if (mode==20) {
+ /* 15:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+ (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13; /* Bit 2 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6; /* Bit 3..9 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* 16:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[2])&0x1))<<15;
+ /* Bit 0 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12; /* Bit 1..3 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10; /* Bit 4..5 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7; /* Bit 6..8 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6; /* Bit 9 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* 17:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[4])&0x1))<<15;
+ /* Bit 0 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8; /* Bit 1..7 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[6]); /* Bit 8..15 */
+ bitstreamPtr++;
+ /* 18:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[7]))<<8; /* Bit 0..7 */
+ (*bitstreamPtr) |= (enc_bits->cb_index[8]); /* Bit 8..15 */
+ bitstreamPtr++;
+ /* 19:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->gain_index[3])&0x3))<<14;
+ /* Bit 0..1 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12; /* Bit 2..3 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9; /* Bit 4..6 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6; /* Bit 7..9 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4; /* Bit 10..11 */
+ (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1; /* Bit 12..14 */
+ } else { /* mode==30 */
+ /* 18:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+ (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12; /* Bit 2..3 */
+ (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11; /* Bit 4 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4; /* Bit 5..11 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 19:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[2])&0x7)<<13;
+ /* Bit 0..2 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10; /* Bit 3..5 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8; /* Bit 6..7 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5; /* Bit 8..10 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4; /* Bit 11 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 20:th WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[4])&0x7)<<13;
+ /* Bit 0..2 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6; /* Bit 3..9 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5; /* Bit 10 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3; /* Bit 11..15 */
+ bitstreamPtr++;
+ /* 21:st WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[7])&0x7)<<13;
+ /* Bit 0..2 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5; /* Bit 3..10 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4; /* Bit 11 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 22:nd WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[10])&0xF)<<12;
+ /* Bit 0..3 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4; /* Bit 4..11 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3; /* Bit 12 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* 23:rd WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[13])&0x1F)<<11;
+ /* Bit 0..4 */
+ (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3; /* Bit 5..12 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1; /* Bit 13..14 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1); /* Bit 15 */
+ bitstreamPtr++;
+ /* 24:rd WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->gain_index[5]))<<13;
+ /* Bit 0..2 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10; /* Bit 3..5 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8; /* Bit 6..7 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5; /* Bit 8..10 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1; /* Bit 11..14 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2; /* Bit 15 */
+ bitstreamPtr++;
+ /* 25:rd WebRtc_Word16 */
+ (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->gain_index[10])&0x3)<<14;
+ /* Bit 0..1 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11; /* Bit 2..4 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7; /* Bit 5..8 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4; /* Bit 9..11 */
+ (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1; /* Bit 12..14 */
+ }
+ /* Last bit is automatically zero */
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h
new file mode 100644
index 0000000..ed3f224
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+ WebRtc_UWord16 *bitstream, /* (o) The packetized bitstream */
+ iLBC_bits *enc_bits, /* (i) Encoded bits */
+ WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
new file mode 100644
index 0000000..fe91851
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "poly_to_lsp.h"
+#include "lsp_to_lsf.h"
+
+void WebRtcIlbcfix_Poly2Lsf(
+ WebRtc_Word16 *lsf, /* (o) lsf coefficients (Q13) */
+ WebRtc_Word16 *a /* (i) A coefficients (Q12) */
+ ) {
+ WebRtc_Word16 lsp[10];
+ WebRtcIlbcfix_Poly2Lsp(a, lsp, (WebRtc_Word16*)WebRtcIlbcfix_kLspMean);
+ WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
new file mode 100644
index 0000000..0ea595e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsf(
+ WebRtc_Word16 *lsf, /* (o) lsf coefficients (Q13) */
+ WebRtc_Word16 *a /* (i) A coefficients (Q12) */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
new file mode 100644
index 0000000..29b4213
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
@@ -0,0 +1,156 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "chebyshev.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+ WebRtc_Word16 *a, /* (o) A coefficients in Q12 */
+ WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
+ WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
+ coefficients turn out to be unstable */
+ ) {
+ WebRtc_Word16 f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
+ WebRtc_Word16 *a_i_ptr, *a_10mi_ptr;
+ WebRtc_Word16 *f1ptr, *f2ptr;
+ WebRtc_Word32 tmpW32;
+ WebRtc_Word16 x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
+ WebRtc_Word16 shifts, sign;
+ int i, j;
+ int foundFreqs;
+ int fi_select;
+
+ /*
+ Calculate the two polynomials f1(z) and f2(z)
+ (the sum and the diff polynomial)
+ f1[0] = f2[0] = 1.0;
+ f1[i+1] = a[i+1] + a[10-i] - f1[i];
+ f2[i+1] = a[i+1] - a[10-i] - f1[i];
+ */
+
+ a_i_ptr = a + 1;
+ a_10mi_ptr = a + 10;
+ f1ptr = f[0];
+ f2ptr = f[1];
+ (*f1ptr) = 1024; /* 1.0 in Q10 */
+ (*f2ptr) = 1024; /* 1.0 in Q10 */
+ for (i = 0; i < 5; i++) {
+ (*(f1ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)+(*a_10mi_ptr)), 2) - (*f1ptr));
+ (*(f2ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)-(*a_10mi_ptr)), 2) + (*f2ptr));
+ a_i_ptr++;
+ a_10mi_ptr--;
+ f1ptr++;
+ f2ptr++;
+ }
+
+ /*
+ find the LSPs using the Chebychev pol. evaluation
+ */
+
+ fi_select = 0; /* selector between f1 and f2, start with f1 */
+
+ foundFreqs = 0;
+
+ xlow = WebRtcIlbcfix_kCosGrid[0];
+ ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+ /*
+ Iterate until all the 10 LSP's have been found or
+ all the grid points have been tried. If the 10 LSP's can
+ not be found, set the LSP vector to previous LSP
+ */
+
+ for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) {
+ xhigh = xlow;
+ yhigh = ylow;
+ xlow = WebRtcIlbcfix_kCosGrid[j];
+ ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+ if (WEBRTC_SPL_MUL_16_16(ylow, yhigh) <= 0) {
+ /* Run 4 times to reduce the interval */
+ for (i = 0; i < 4; i++) {
+ /* xmid =(xlow + xhigh)/2 */
+ xmid = WEBRTC_SPL_RSHIFT_W16(xlow, 1) + WEBRTC_SPL_RSHIFT_W16(xhigh, 1);
+ ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
+
+ if (WEBRTC_SPL_MUL_16_16(ylow, ymid) <= 0) {
+ yhigh = ymid;
+ xhigh = xmid;
+ } else {
+ ylow = ymid;
+ xlow = xmid;
+ }
+ }
+
+ /*
+ Calculater xint by linear interpolation:
+ xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow);
+ */
+
+ x = xhigh - xlow;
+ y = yhigh - ylow;
+
+ if (y == 0) {
+ xint = xlow;
+ } else {
+ sign = y;
+ y = WEBRTC_SPL_ABS_W16(y);
+ shifts = (WebRtc_Word16)WebRtcSpl_NormW32(y)-16;
+ y = WEBRTC_SPL_LSHIFT_W16(y, shifts);
+ y = (WebRtc_Word16)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
+
+ tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(x, y, (19-shifts));
+
+ /* y=(xhigh-xlow)/(yhigh-ylow) */
+ y = (WebRtc_Word16)(tmpW32&0xFFFF);
+
+ if (sign < 0) {
+ y = -y;
+ }
+ /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
+ tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(ylow, y, 10);
+ xint = xlow-(WebRtc_Word16)(tmpW32&0xFFFF);
+ }
+
+ /* Store the calculated lsp */
+ lsp[foundFreqs] = (WebRtc_Word16)xint;
+ foundFreqs++;
+
+ /* if needed, set xlow and ylow for next recursion */
+ if (foundFreqs<10) {
+ xlow = xint;
+ /* Swap between f1 and f2 (f[0][] and f[1][]) */
+ fi_select = ((fi_select+1)&0x1);
+
+ ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+ }
+ }
+ }
+
+ /* Check if M roots found, if not then use the old LSP */
+ if (foundFreqs < 10) {
+ WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10);
+ }
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
new file mode 100644
index 0000000..7eebb25
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+ WebRtc_Word16 *a, /* (o) A coefficients in Q12 */
+ WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
+ WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
+ coefficients turn out to be unstable */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c
new file mode 100644
index 0000000..9210092
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c
@@ -0,0 +1,150 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "enh_upsample.h"
+#include "my_corr.h"
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+ WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
+ WebRtc_Word16 *idata, /* (i) original data buffer */
+ WebRtc_Word16 idatal, /* (i) dimension of idata */
+ WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
+ WebRtc_Word16 estSegPos, /* (i) estimated beginning other segment (Q-2) */
+ WebRtc_Word16 *surround, /* (i/o) The contribution from this sequence
+ summed with earlier contributions */
+ WebRtc_Word16 gain /* (i) Gain to use for this sequence */
+ ){
+ WebRtc_Word16 estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
+ WebRtc_Word16 tloc,tloc2,i,st,en,fraction;
+
+ WebRtc_Word32 maxtemp, scalefact;
+ WebRtc_Word16 *filtStatePtr, *polyPtr;
+ /* Stack based */
+ WebRtc_Word16 filt[7];
+ WebRtc_Word32 corrVecUps[ENH_CORRDIM*ENH_UPS0];
+ WebRtc_Word32 corrVecTemp[ENH_CORRDIM];
+ WebRtc_Word16 vect[ENH_VECTL];
+ WebRtc_Word16 corrVec[ENH_CORRDIM];
+
+ /* defining array bounds */
+
+ estSegPosRounded=WEBRTC_SPL_RSHIFT_W16((estSegPos - 2),2);
+
+ searchSegStartPos=estSegPosRounded-ENH_SLOP;
+
+ if (searchSegStartPos<0) {
+ searchSegStartPos=0;
+ }
+ searchSegEndPos=estSegPosRounded+ENH_SLOP;
+
+ if(searchSegEndPos+ENH_BLOCKL >= idatal) {
+ searchSegEndPos=idatal-ENH_BLOCKL-1;
+ }
+ corrdim=searchSegEndPos-searchSegStartPos+1;
+
+ /* compute upsampled correlation and find
+ location of max */
+
+ WebRtcIlbcfix_MyCorr(corrVecTemp,idata+searchSegStartPos,
+ (WebRtc_Word16)(corrdim+ENH_BLOCKL-1),idata+centerStartPos,ENH_BLOCKL);
+
+ /* Calculate the rescaling factor for the correlation in order to
+ put the correlation in a WebRtc_Word16 vector instead */
+ maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (WebRtc_Word16)corrdim);
+
+ scalefact=WebRtcSpl_GetSizeInBits(maxtemp)-15;
+
+ if (scalefact>0) {
+ for (i=0;i<corrdim;i++) {
+ corrVec[i]=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(corrVecTemp[i], scalefact);
+ }
+ } else {
+ for (i=0;i<corrdim;i++) {
+ corrVec[i]=(WebRtc_Word16)corrVecTemp[i];
+ }
+ }
+ /* In order to guarantee that all values are initialized */
+ for (i=corrdim;i<ENH_CORRDIM;i++) {
+ corrVec[i]=0;
+ }
+
+ /* Upsample the correlation */
+ WebRtcIlbcfix_EnhUpsample(corrVecUps,corrVec);
+
+ /* Find maximum */
+ tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (WebRtc_Word16) (ENH_UPS0*corrdim));
+
+ /* make vector can be upsampled without ever running outside
+ bounds */
+ *updStartPos = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(searchSegStartPos,4) + tloc + 4;
+
+ tloc2 = WEBRTC_SPL_RSHIFT_W16((tloc+3), 2);
+
+ st=searchSegStartPos+tloc2-ENH_FL0;
+
+ /* initialize the vector to be filtered, stuff with zeros
+ when data is outside idata buffer */
+ if(st<0){
+ WebRtcSpl_MemSetW16(vect, 0, (WebRtc_Word16)(-st));
+ WEBRTC_SPL_MEMCPY_W16(&vect[-st], idata, (ENH_VECTL+st));
+ }
+ else{
+ en=st+ENH_VECTL;
+
+ if(en>idatal){
+ WEBRTC_SPL_MEMCPY_W16(vect, &idata[st],
+ (ENH_VECTL-(en-idatal)));
+ WebRtcSpl_MemSetW16(&vect[ENH_VECTL-(en-idatal)], 0,
+ (WebRtc_Word16)(en-idatal));
+ }
+ else {
+ WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
+ }
+ }
+ /* Calculate which of the 4 fractions to use */
+ fraction=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(tloc2,ENH_UPS0)-tloc;
+
+ /* compute the segment (this is actually a convolution) */
+
+ filtStatePtr = filt + 6;
+ polyPtr = (WebRtc_Word16*)WebRtcIlbcfix_kEnhPolyPhaser[fraction];
+ for (i=0;i<7;i++) {
+ *filtStatePtr-- = *polyPtr++;
+ }
+
+ WebRtcSpl_FilterMAFastQ12(
+ &vect[6], vect, filt,
+ ENH_FLO_MULT2_PLUS1, ENH_BLOCKL);
+
+ /* Add the contribution from this vector (scaled with gain) to the total surround vector */
+ WebRtcSpl_AddAffineVectorToVector(
+ surround, vect, gain,
+ (WebRtc_Word32)32768, 16, ENH_BLOCKL);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h
new file mode 100644
index 0000000..559555c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+ WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
+ WebRtc_Word16 *idata, /* (i) original data buffer */
+ WebRtc_Word16 idatal, /* (i) dimension of idata */
+ WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
+ WebRtc_Word16 estSegPos, /* (i) estimated beginning other segment (Q-2) */
+ WebRtc_Word16 *surround, /* (i/o) The contribution from this sequence
+ summed with earlier contributions */
+ WebRtc_Word16 gain /* (i) Gain to use for this sequence */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
new file mode 100644
index 0000000..ee5e643
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -0,0 +1,130 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "lsf_interpolate_to_poly_enc.h"
+#include "bw_expand.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+ WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
+ resulting from the quantized
+ interpolated lsf Q12 */
+ WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
+ resulting from the unquantized
+ interpolated lsf Q12 */
+ WebRtc_Word16 *lsf, /* (i) the unquantized lsf coefficients Q13 */
+ WebRtc_Word16 *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */
+ WebRtc_Word16 *lsfold, /* (i) the unquantized lsf coefficients of
+ the previous signal frame Q13 */
+ WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+ previous signal frame Q13 */
+ WebRtc_Word16 length, /* (i) should equate FILTERORDER */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ) {
+ int i, pos, lp_length;
+
+ WebRtc_Word16 *lsf2, *lsfdeq2;
+ /* Stack based */
+ WebRtc_Word16 lp[LPC_FILTERORDER + 1];
+
+ lsf2 = lsf + length;
+ lsfdeq2 = lsfdeq + length;
+ lp_length = length + 1;
+
+ if (iLBCenc_inst->mode==30) {
+ /* subframe 1: Interpolation between old and first set of
+ lsf coefficients */
+
+ /* Calculate Analysis/Syntehsis filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+ WebRtcIlbcfix_kLsfWeight30ms[0],
+ length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length);
+
+ /* Calculate Weighting filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+ WebRtcIlbcfix_kLsfWeight30ms[0],
+ length);
+ WebRtcIlbcfix_BwExpand(weightdenum, lp,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+ (WebRtc_Word16)lp_length);
+
+ /* subframe 2 to 6: Interpolation between first and second
+ set of lsf coefficients */
+
+ pos = lp_length;
+ for (i = 1; i < iLBCenc_inst->nsub; i++) {
+
+ /* Calculate Analysis/Syntehsis filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2,
+ WebRtcIlbcfix_kLsfWeight30ms[i],
+ length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+ /* Calculate Weighting filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2,
+ WebRtcIlbcfix_kLsfWeight30ms[i],
+ length);
+ WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+ (WebRtc_Word16)lp_length);
+
+ pos += lp_length;
+ }
+
+ /* update memory */
+
+ WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length);
+ WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length);
+
+ } else { /* iLBCenc_inst->mode==20 */
+ pos = 0;
+ for (i = 0; i < iLBCenc_inst->nsub; i++) {
+
+ /* Calculate Analysis/Syntehsis filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+ WebRtcIlbcfix_kLsfWeight20ms[i],
+ length);
+ WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+ /* Calculate Weighting filter from quantized LSF */
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+ WebRtcIlbcfix_kLsfWeight20ms[i],
+ length);
+ WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+ (WebRtc_Word16)lp_length);
+
+ pos += lp_length;
+ }
+
+ /* update memory */
+
+ WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
+ WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
+
+ }
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
new file mode 100644
index 0000000..8cdd7da
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+ WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
+ resulting from the quantized
+ interpolated lsf Q12 */
+ WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
+ resulting from the unquantized
+ interpolated lsf Q12 */
+ WebRtc_Word16 *lsf, /* (i) the unquantized lsf coefficients Q13 */
+ WebRtc_Word16 *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */
+ WebRtc_Word16 *lsfold, /* (i) the unquantized lsf coefficients of
+ the previous signal frame Q13 */
+ WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+ previous signal frame Q13 */
+ WebRtc_Word16 length, /* (i) should equate FILTERORDER */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
new file mode 100644
index 0000000..2d19edd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "window32_w32.h"
+#include "bw_expand.h"
+#include "poly_to_lsf.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+ WebRtc_Word16 *lsf, /* (o) lsf coefficients */
+ WebRtc_Word16 *data, /* (i) new block of speech */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ ) {
+ int k;
+ int scale;
+ WebRtc_Word16 is;
+ WebRtc_Word16 stability;
+ /* Stack based */
+ WebRtc_Word16 A[LPC_FILTERORDER + 1];
+ WebRtc_Word32 R[LPC_FILTERORDER + 1];
+ WebRtc_Word16 windowedData[BLOCKL_MAX];
+ WebRtc_Word16 rc[LPC_FILTERORDER];
+
+ is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+ WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
+
+ /* No lookahead, last window is asymmetric */
+
+ for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
+
+ is = LPC_LOOKBACK;
+
+ if (k < (iLBCenc_inst->lpc_n - 1)) {
+
+ /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */
+ WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15);
+ } else {
+
+ /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */
+ WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15);
+ }
+
+ /* Compute autocorrelation */
+ WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale);
+
+ /* Window autocorrelation vector */
+ WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 );
+
+ /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */
+ stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER);
+
+ /*
+ Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable
+ This should basically never happen...
+ */
+ if (stability!=1) {
+ A[0]=4096;
+ WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER);
+ }
+
+ /* Bandwidth expand the filter coefficients */
+ WebRtcIlbcfix_BwExpand(A, A, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
+
+ /* Convert from A to LSF representation */
+ WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
+ }
+
+ is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+ WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer,
+ iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
new file mode 100644
index 0000000..83c1e5b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+ WebRtc_Word16 *lsf, /* (o) lsf coefficients */
+ WebRtc_Word16 *data, /* (i) new block of speech */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
new file mode 100644
index 0000000..7b5efa0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+ WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients */
+ WebRtc_Word16 *index, /* (i) quantization index */
+ WebRtc_Word16 lpc_n /* (i) number of LPCs */
+ ){
+ int i, j, pos, cb_pos;
+
+ /* decode first LSF */
+
+ pos = 0;
+ cb_pos = 0;
+ for (i = 0; i < LSF_NSPLIT; i++) {
+ for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+ lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos +
+ WEBRTC_SPL_MUL_16_16(index[i], WebRtcIlbcfix_kLsfDimCb[i]) + j];
+ }
+ pos += WebRtcIlbcfix_kLsfDimCb[i];
+ cb_pos += WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kLsfSizeCb[i], WebRtcIlbcfix_kLsfDimCb[i]);
+ }
+
+ if (lpc_n>1) {
+ /* decode last LSF */
+ pos = 0;
+ cb_pos = 0;
+ for (i = 0; i < LSF_NSPLIT; i++) {
+ for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+ lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos +
+ WEBRTC_SPL_MUL_16_16(index[LSF_NSPLIT + i], WebRtcIlbcfix_kLsfDimCb[i]) + j];
+ }
+ pos += WebRtcIlbcfix_kLsfDimCb[i];
+ cb_pos += WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kLsfSizeCb[i], WebRtcIlbcfix_kLsfDimCb[i]);
+ }
+ }
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
new file mode 100644
index 0000000..efd3103
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+ WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients */
+ WebRtc_Word16 *index, /* (i) quantization index */
+ WebRtc_Word16 lpc_n /* (i) number of LPCs */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
new file mode 100644
index 0000000..aa27fb4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "split_vq.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+ WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
+ (dimension FILTERORDER) Q13 */
+ WebRtc_Word16 *index, /* (o) quantization index */
+ WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
+ quantized (dimension FILTERORDER) Q13 */
+ WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
+ ){
+
+ /* Quantize first LSF with memoryless split VQ */
+ WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
+
+ if (lpc_n==2) {
+ /* Quantize second LSF with memoryless split VQ */
+ WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
+ lsf + LPC_FILTERORDER, (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb,
+ (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
+ }
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
new file mode 100644
index 0000000..fd17b2e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+ WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
+ (dimension FILTERORDER) Q13 */
+ WebRtc_Word16 *index, /* (o) quantization index */
+ WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
+ quantized (dimension FILTERORDER) Q13 */
+ WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c
new file mode 100644
index 0000000..b606077
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c
@@ -0,0 +1,211 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "smooth_out_data.h"
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(
+ WebRtc_Word16 *odata, /* (o) smoothed output */
+ WebRtc_Word16 *current, /* (i) the un enhanced residual for
+ this block */
+ WebRtc_Word16 *surround /* (i) The approximation from the
+ surrounding sequences */
+ ) {
+ WebRtc_Word16 maxtot, scale, scale1, scale2;
+ WebRtc_Word16 A, B, C, denomW16;
+ WebRtc_Word32 B_W32, denom, num;
+ WebRtc_Word32 errs;
+ WebRtc_Word32 w00,w10,w11, endiff, crit;
+ WebRtc_Word32 w00prim, w10prim, w11_div_w00;
+ WebRtc_Word16 w11prim;
+ WebRtc_Word16 bitsw00, bitsw10, bitsw11;
+ WebRtc_Word32 w11w00, w10w10, w00w00;
+ WebRtc_Word16 max1, max2;
+
+ /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
+
+ w00 = w10 = w11 = 0;
+
+ max1=WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL);
+ max2=WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL);
+ maxtot=WEBRTC_SPL_MAX(max1, max2);
+
+ scale=WebRtcSpl_GetSizeInBits(maxtot);
+ scale = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2,scale)-26;
+ scale=WEBRTC_SPL_MAX(0, scale);
+
+ w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
+ w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale);
+ w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale);
+
+ if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX;
+ if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX;
+
+ /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim
+ is in Q16 */
+
+ bitsw00 = WebRtcSpl_GetSizeInBits(w00);
+ bitsw11 = WebRtcSpl_GetSizeInBits(w11);
+ bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10));
+ scale1 = 31 - bitsw00;
+ scale2 = 15 - bitsw11;
+
+ if (scale2>(scale1-16)) {
+ scale2 = scale1 - 16;
+ } else {
+ scale1 = scale2 + 16;
+ }
+
+ w00prim = WEBRTC_SPL_LSHIFT_W32(w00, scale1);
+ w11prim = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w11, scale2);
+
+ /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
+ if (w11prim>64) {
+ endiff = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)WebRtcSpl_DivW32W16(w00prim, w11prim), 6);
+ C = (WebRtc_Word16)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
+ } else {
+ C = 1;
+ }
+
+ /* first try enhancement without power-constraint */
+
+ errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C);
+
+
+
+ /* if constraint violated by first try, add constraint */
+
+ if ( (6-scale+scale1) > 31) {
+ crit=0;
+ } else {
+ /* crit = 0.05 * w00 (Result in Q-6) */
+ crit = WEBRTC_SPL_SHIFT_W32(
+ WEBRTC_SPL_MUL(ENH_A0, WEBRTC_SPL_RSHIFT_W32(w00prim, 14)),
+ -(6-scale+scale1));
+ }
+
+ if (errs > crit) {
+
+ if( w00 < 1) {
+ w00=1;
+ }
+
+ /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */
+
+ scale1 = bitsw00-15;
+ scale2 = bitsw11-15;
+
+ if (scale2>scale1) {
+ scale = scale2;
+ } else {
+ scale = scale1;
+ }
+
+ w11w00 = WEBRTC_SPL_MUL_16_16(
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w11, -scale),
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
+
+ w10w10 = WEBRTC_SPL_MUL_16_16(
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale),
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale));
+
+ w00w00 = WEBRTC_SPL_MUL_16_16(
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale),
+ (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
+
+ /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
+ if (w00w00>65536) {
+ endiff = (w11w00-w10w10);
+ endiff = WEBRTC_SPL_MAX(0, endiff);
+ /* denom is in Q16 */
+ denom = WebRtcSpl_DivW32W16(endiff, (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(w00w00, 16));
+ } else {
+ denom = 65536;
+ }
+
+ if( denom > 7){ /* eliminates numerical problems
+ for if smooth */
+
+ scale=WebRtcSpl_GetSizeInBits(denom)-15;
+
+ if (scale>0) {
+ /* denomW16 is in Q(16+scale) */
+ denomW16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(denom, scale);
+
+ /* num in Q(34-scale) */
+ num=WEBRTC_SPL_RSHIFT_W32(ENH_A0_MINUS_A0A0DIV4, scale);
+ } else {
+ /* denomW16 is in Q16 */
+ denomW16=(WebRtc_Word16)denom;
+
+ /* num in Q34 */
+ num=ENH_A0_MINUS_A0A0DIV4;
+ }
+
+ /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
+ A = (WebRtc_Word16)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
+
+ /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
+ scale1 = 31-bitsw10;
+ scale2 = 21-scale1;
+ w10prim = WEBRTC_SPL_LSHIFT_W32(w10, scale1);
+ w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2);
+ scale = bitsw00-scale2-15;
+
+ if (scale>0) {
+ w10prim=WEBRTC_SPL_RSHIFT_W32(w10prim, scale);
+ w00prim=WEBRTC_SPL_RSHIFT_W32(w00prim, scale);
+ }
+
+ if ((w00prim>0)&&(w10prim>0)) {
+ w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (WebRtc_Word16)w00prim);
+
+ if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
+ B_W32 = 0;
+ } else {
+ B_W32 = (WebRtc_Word32)1073741824 - (WebRtc_Word32)ENH_A0DIV2 -
+ WEBRTC_SPL_MUL(A, w11_div_w00);
+ }
+ B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B_W32, 16); /* B in Q14 */
+ } else {
+ /* No smoothing */
+ A = 0;
+ B = 16384; /* 1 in Q14 */
+ }
+ }
+ else{ /* essentially no difference between cycles;
+ smoothing not needed */
+
+ A = 0;
+ B = 16384; /* 1 in Q14 */
+ }
+
+ /* create smoothed sequence */
+
+ WebRtcSpl_ScaleAndAddVectors(surround, A, 9,
+ current, B, 14,
+ odata, ENH_BLOCKL);
+ }
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h
new file mode 100644
index 0000000..88ce805
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(
+ WebRtc_Word16 *odata, /* (o) smoothed output */
+ WebRtc_Word16 *current, /* (i) the un enhanced residual for
+ this block */
+ WebRtc_Word16 *surround /* (i) The approximation from the
+ surrounding sequences */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c
new file mode 100644
index 0000000..9bacd85
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
+ WebRtc_Word16 *odata,
+ WebRtc_Word16 *psseq,
+ WebRtc_Word16 *surround,
+ WebRtc_Word16 C)
+{
+ int i;
+
+ WebRtc_Word16 err;
+ WebRtc_Word32 errs;
+
+ for(i=0;i<80;i++) {
+ odata[i]= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(C, surround[i])+1024), 11);
+ }
+
+ errs=0;
+ for(i=0;i<80;i++) {
+ err=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
+ errs+=WEBRTC_SPL_MUL_16_16(err, err); /* errs in Q-6 */
+ }
+
+ return errs;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h
new file mode 100644
index 0000000..6fbe694
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * help function to WebRtcIlbcfix_Smooth()
+ *---------------------------------------------------------------*/
+
+WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
+ WebRtc_Word16 *odata,
+ WebRtc_Word16 *psseq,
+ WebRtc_Word16 *surround,
+ WebRtc_Word16 C);
+
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c
new file mode 100644
index 0000000..9276a7b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+ WebRtc_Word16 *xq, /* (o) the quantized value */
+ WebRtc_Word16 *index, /* (o) the quantization index */
+ WebRtc_Word16 x, /* (i) the value to quantize */
+ const WebRtc_Word16 *cb, /* (i) the quantization codebook */
+ WebRtc_Word16 cb_size /* (i) the size of the quantization codebook */
+ ){
+ int i;
+
+ if (x <= cb[0]) {
+ *index = 0;
+ *xq = cb[0];
+ } else {
+ i = 0;
+ while ((x > cb[i]) && (i < (cb_size-1))) {
+ i++;
+ }
+
+ if (x > WEBRTC_SPL_RSHIFT_W32(( (WebRtc_Word32)cb[i] + cb[i - 1] + 1),1)) {
+ *index = i;
+ *xq = cb[i];
+ } else {
+ *index = i - 1;
+ *xq = cb[i - 1];
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h
new file mode 100644
index 0000000..2863dc5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+ WebRtc_Word16 *xq, /* (o) the quantized value */
+ WebRtc_Word16 *index, /* (o) the quantization index */
+ WebRtc_Word16 x, /* (i) the value to quantize */
+ const WebRtc_Word16 *cb, /* (i) the quantization codebook */
+ WebRtc_Word16 cb_size /* (i) the size of the quantization codebook */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c
new file mode 100644
index 0000000..d908fa2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "vq3.h"
+#include "vq4.h"
+
+/*----------------------------------------------------------------*
+ * split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+ WebRtc_Word16 *qX, /* (o) the quantized vector in Q13 */
+ WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
+ codebooks in the split */
+ WebRtc_Word16 *X, /* (i) the vector to quantize */
+ WebRtc_Word16 *CB, /* (i) the quantizer codebook in Q13 */
+ WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
+ WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
+ ) {
+
+ WebRtc_Word16 *qXPtr, *indexPtr, *CBPtr, *XPtr;
+
+ /* Quantize X with the 3 vectror quantization tables */
+
+ qXPtr=qX;
+ indexPtr=index;
+ CBPtr=CB;
+ XPtr=X;
+ WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]);
+
+ qXPtr+=3;
+ indexPtr+=1;
+ CBPtr+=(dim[0]*cbsize[0]);
+ XPtr+=3;
+ WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]);
+
+ qXPtr+=3;
+ indexPtr+=1;
+ CBPtr+=(dim[1]*cbsize[1]);
+ XPtr+=3;
+ WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h
new file mode 100644
index 0000000..7264a21
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+ WebRtc_Word16 *qX, /* (o) the quantized vector in Q13 */
+ WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
+ codebooks in the split */
+ WebRtc_Word16 *X, /* (i) the vector to quantize */
+ WebRtc_Word16 *CB, /* (i) the quantizer codebook in Q13 */
+ WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
+ WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c
new file mode 100644
index 0000000..9d03cc3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * decoding of the start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+ WebRtc_Word16 idxForMax, /* (i) 6-bit index for the quantization of
+ max amplitude */
+ WebRtc_Word16 *idxVec, /* (i) vector of quantization indexes */
+ WebRtc_Word16 *syntDenum, /* (i) synthesis filter denumerator */
+ WebRtc_Word16 *Out_fix, /* (o) the decoded state vector */
+ WebRtc_Word16 len /* (i) length of a state vector */
+ ) {
+ int k;
+ WebRtc_Word16 maxVal;
+ WebRtc_Word16 *tmp1, *tmp2, *tmp3;
+ /* Stack based */
+ WebRtc_Word16 numerator[1+LPC_FILTERORDER];
+ WebRtc_Word16 sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+ WebRtc_Word16 sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+ WebRtc_Word16 *sampleVal = &sampleValVec[LPC_FILTERORDER];
+ WebRtc_Word16 *sampleMa = &sampleMaVec[LPC_FILTERORDER];
+ WebRtc_Word16 *sampleAr = &sampleValVec[LPC_FILTERORDER];
+
+ /* initialization of coefficients */
+
+ for (k=0; k<LPC_FILTERORDER+1; k++){
+ numerator[k] = syntDenum[LPC_FILTERORDER-k];
+ }
+
+ /* decoding of the maximum value */
+
+ maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax];
+
+ /* decoding of the sample values */
+ tmp1 = sampleVal;
+ tmp2 = &idxVec[len-1];
+
+ if (idxForMax<37) {
+ for(k=0; k<len; k++){
+ /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
+ maxVal is in Q8 and result is in Q(-1) */
+ (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)2097152) >> 22);
+ tmp1++;
+ tmp2--;
+ }
+ } else if (idxForMax<59) {
+ for(k=0; k<len; k++){
+ /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
+ maxVal is in Q5 and result is in Q(-1) */
+ (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)262144) >> 19);
+ tmp1++;
+ tmp2--;
+ }
+ } else {
+ for(k=0; k<len; k++){
+ /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
+ maxVal is in Q3 and result is in Q(-1) */
+ (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)65536) >> 17);
+ tmp1++;
+ tmp2--;
+ }
+ }
+
+ /* Set the rest of the data to zero */
+ WebRtcSpl_MemSetW16(&sampleVal[len], 0, len);
+
+ /* circular convolution with all-pass filter */
+
+ /* Set the state to zero */
+ WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER));
+
+ /* Run MA filter + AR filter */
+ WebRtcSpl_FilterMAFastQ12(
+ sampleVal, sampleMa,
+ numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(len + LPC_FILTERORDER));
+ WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
+ WebRtcSpl_FilterARFastQ12(
+ sampleMa, sampleAr,
+ syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*len));
+
+ tmp1 = &sampleAr[len-1];
+ tmp2 = &sampleAr[2*len-1];
+ tmp3 = Out_fix;
+ for(k=0;k<len;k++){
+ (*tmp3) = (*tmp1) + (*tmp2);
+ tmp1--;
+ tmp2--;
+ tmp3++;
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h
new file mode 100644
index 0000000..465699b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+
+/*----------------------------------------------------------------*
+ * Generate the start state from the quantized indexes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+ WebRtc_Word16 idxForMax, /* (i) 6-bit index for the quantization of
+ max amplitude */
+ WebRtc_Word16 *idxVec, /* (i) vector of quantization indexes */
+ WebRtc_Word16 *syntDenum, /* (i) synthesis filter denumerator */
+ WebRtc_Word16 *Out_fix, /* (o) the decoded state vector */
+ WebRtc_Word16 len /* (i) length of a state vector */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c
new file mode 100644
index 0000000..824a0ba
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.c
+
+******************************************************************/
+
+#include "defines.h"
+#include "constants.h"
+#include "abs_quant.h"
+
+/*----------------------------------------------------------------*
+ * encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
+ and idxVec, input state_first) */
+ WebRtc_Word16 *residual, /* (i) target residual vector */
+ WebRtc_Word16 *syntDenum, /* (i) lpc synthesis filter */
+ WebRtc_Word16 *weightDenum /* (i) weighting filter denuminator */
+ ) {
+ WebRtc_Word16 k, index;
+ WebRtc_Word16 maxVal;
+ WebRtc_Word16 scale, shift;
+ WebRtc_Word32 maxValsq;
+ WebRtc_Word16 scaleRes;
+ WebRtc_Word16 max;
+ int i;
+ /* Stack based */
+ WebRtc_Word16 numerator[1+LPC_FILTERORDER];
+ WebRtc_Word16 residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+ WebRtc_Word16 sampleMa[2*STATE_SHORT_LEN_30MS];
+ WebRtc_Word16 *residualLong = &residualLongVec[LPC_FILTERORDER];
+ WebRtc_Word16 *sampleAr = residualLong;
+
+ /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
+ max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
+ scaleRes = WebRtcSpl_GetSizeInBits(max)-12;
+ scaleRes = WEBRTC_SPL_MAX(0, scaleRes);
+ /* Set up the filter coefficients for the circular convolution */
+ for (i=0; i<LPC_FILTERORDER+1; i++) {
+ numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes);
+ }
+
+ /* Copy the residual to a temporary buffer that we can filter
+ * and set the remaining samples to zero.
+ */
+ WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len);
+ WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len);
+
+ /* Run the Zero-Pole filter (Ciurcular convolution) */
+ WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
+ WebRtcSpl_FilterMAFastQ12(
+ residualLong, sampleMa,
+ numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(iLBCenc_inst->state_short_len + LPC_FILTERORDER));
+ WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
+
+ WebRtcSpl_FilterARFastQ12(
+ sampleMa, sampleAr,
+ syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*iLBCenc_inst->state_short_len));
+
+ for(k=0;k<iLBCenc_inst->state_short_len;k++){
+ sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
+ }
+
+ /* Find maximum absolute value in the vector */
+ maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len);
+
+ /* Find the best index */
+
+ if ((((WebRtc_Word32)maxVal)<<scaleRes)<23170) {
+ maxValsq=((WebRtc_Word32)maxVal*maxVal)<<(2+2*scaleRes);
+ } else {
+ maxValsq=(WebRtc_Word32)WEBRTC_SPL_WORD32_MAX;
+ }
+
+ index=0;
+ for (i=0;i<63;i++) {
+
+ if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) {
+ index=i+1;
+ } else {
+ i=63;
+ }
+ }
+ iLBC_encbits->idxForMax=index;
+
+ /* Rescale the vector before quantization */
+ scale=WebRtcIlbcfix_kScale[index];
+
+ if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */
+ shift=4;
+ } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */
+ shift=9;
+ }
+
+ /* Set up vectors for AbsQuant and rescale it with the scale factor */
+ WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
+ iLBCenc_inst->state_short_len, (WebRtc_Word16)(shift-scaleRes));
+
+ /* Quantize the values in fout[] */
+ WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h
new file mode 100644
index 0000000..8b7f298
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
+ and idxVec, input state_first) */
+ WebRtc_Word16 *residual, /* (i) target residual vector */
+ WebRtc_Word16 *syntDenum, /* (i) lpc synthesis filter */
+ WebRtc_Word16 *weightDenum /* (i) weighting filter denuminator */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c
new file mode 100644
index 0000000..61b8b7b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+ WebRtc_UWord16 *sequence, /* (i/o) the sequence to swap */
+ WebRtc_Word16 wordLength /* (i) number or WebRtc_UWord16 to swap */
+ ) {
+ int k;
+ WebRtc_UWord16 temp=0;
+ for( k=wordLength; k>0; k-- ) {
+ temp = (*sequence >> 8)|(*sequence << 8);
+ *sequence++ = temp;
+ //*sequence++ = (*sequence >> 8) | (*sequence << 8);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h
new file mode 100644
index 0000000..2f2b3eb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+ WebRtc_UWord16 *sequence, /* (i/o) the sequence to swap */
+ WebRtc_Word16 wordLength /* (i) number or WebRtc_UWord16 to swap */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
new file mode 100644
index 0000000..d2c9ba5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -0,0 +1,225 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_test.c
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include "ilbc.h"
+
+/*---------------------------------------------------------------*
+ * Main program to test iLBC encoding and decoding
+ *
+ * Usage:
+ * exefile_name.exe <infile> <bytefile> <outfile> <channel>
+ *
+ * <infile> : Input file, speech for encoder (16-bit pcm file)
+ * <bytefile> : Bit stream output from the encoder
+ * <outfile> : Output file, decoded speech (16-bit pcm file)
+ * <channel> : Bit error file, optional (16-bit)
+ * 1 - Packet received correctly
+ * 0 - Packet Lost
+ *
+ *--------------------------------------------------------------*/
+
+#define BLOCKL_MAX 240
+#define ILBCNOOFWORDS_MAX 25
+
+
+int main(int argc, char* argv[])
+{
+
+ FILE *ifileid,*efileid,*ofileid, *cfileid;
+ WebRtc_Word16 data[BLOCKL_MAX];
+ WebRtc_Word16 encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
+ int len;
+ short pli, mode;
+ int blockcount = 0;
+ int packetlosscount = 0;
+ int frameLen;
+ WebRtc_Word16 speechType;
+ iLBC_encinst_t *Enc_Inst;
+ iLBC_decinst_t *Dec_Inst;
+
+#ifdef __ILBC_WITH_40BITACC
+ /* Doublecheck that long long exists */
+ if (sizeof(long)>=sizeof(long long)) {
+ fprintf(stderr, "40-bit simulation is not be supported on this platform\n");
+ exit(0);
+ }
+#endif
+
+ /* get arguments and open files */
+
+ if ((argc!=5) && (argc!=6)) {
+ fprintf(stderr,
+ "\n*-----------------------------------------------*\n");
+ fprintf(stderr,
+ " %s <20,30> input encoded decoded (channel)\n\n",
+ argv[0]);
+ fprintf(stderr,
+ " mode : Frame size for the encoding/decoding\n");
+ fprintf(stderr,
+ " 20 - 20 ms\n");
+ fprintf(stderr,
+ " 30 - 30 ms\n");
+ fprintf(stderr,
+ " input : Speech for encoder (16-bit pcm file)\n");
+ fprintf(stderr,
+ " encoded : Encoded bit stream\n");
+ fprintf(stderr,
+ " decoded : Decoded speech (16-bit pcm file)\n");
+ fprintf(stderr,
+ " channel : Packet loss pattern, optional (16-bit)\n");
+ fprintf(stderr,
+ " 1 - Packet received correctly\n");
+ fprintf(stderr,
+ " 0 - Packet Lost\n");
+ fprintf(stderr,
+ "*-----------------------------------------------*\n\n");
+ exit(1);
+ }
+ mode=atoi(argv[1]);
+ if (mode != 20 && mode != 30) {
+ fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
+ argv[1]);
+ exit(2);
+ }
+ if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+ exit(2);}
+ if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open encoded file file %s\n",
+ argv[3]); exit(1);}
+ if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open decoded file %s\n",
+ argv[4]); exit(1);}
+ if (argc==6) {
+ if( (cfileid=fopen(argv[5],"rb")) == NULL) {
+ fprintf(stderr, "Cannot open channel file %s\n",
+ argv[5]);
+ exit(1);
+ }
+ } else {
+ cfileid=NULL;
+ }
+
+ /* print info */
+
+ fprintf(stderr, "\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* iLBC test program *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,"\nMode : %2d ms\n", mode);
+ fprintf(stderr,"Input file : %s\n", argv[2]);
+ fprintf(stderr,"Encoded file : %s\n", argv[3]);
+ fprintf(stderr,"Output file : %s\n", argv[4]);
+ if (argc==6) {
+ fprintf(stderr,"Channel file : %s\n", argv[5]);
+ }
+ fprintf(stderr,"\n");
+
+ /* Create structs */
+ WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+ WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+
+ /* Initialization */
+
+ WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+ WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+ frameLen = mode*8;
+
+ /* loop over input blocks */
+
+ while (((WebRtc_Word16)fread(data,sizeof(WebRtc_Word16),frameLen,ifileid))==
+ frameLen) {
+
+ blockcount++;
+
+ /* encoding */
+
+ fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+ len=WebRtcIlbcfix_Encode(Enc_Inst, data, (WebRtc_Word16)frameLen, encoded_data);
+ fprintf(stderr, "\r");
+
+ /* write byte file */
+
+ fwrite(encoded_data, sizeof(WebRtc_Word16), ((len+1)/sizeof(WebRtc_Word16)), efileid);
+
+ /* get channel data if provided */
+ if (argc==6) {
+ if (fread(&pli, sizeof(WebRtc_Word16), 1, cfileid)) {
+ if ((pli!=0)&&(pli!=1)) {
+ fprintf(stderr, "Error in channel file\n");
+ exit(0);
+ }
+ if (pli==0) {
+ /* Packet loss -> remove info from frame */
+ memset(encoded_data, 0,
+ sizeof(WebRtc_Word16)*ILBCNOOFWORDS_MAX);
+ packetlosscount++;
+ }
+ } else {
+ fprintf(stderr, "Error. Channel file too short\n");
+ exit(0);
+ }
+ } else {
+ pli=1;
+ }
+
+ /* decoding */
+
+ fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+ if (pli==1) {
+ len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
+ (WebRtc_Word16)len, decoded_data,&speechType);
+ } else {
+ len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
+ }
+ fprintf(stderr, "\r");
+
+ /* write output file */
+
+ fwrite(decoded_data,sizeof(WebRtc_Word16),len,ofileid);
+ }
+
+ /* close files */
+
+ fclose(ifileid); fclose(efileid); fclose(ofileid);
+ if (argc==6) {
+ fclose(cfileid);
+ }
+
+ /* Free structs */
+ WebRtcIlbcfix_EncoderFree(Enc_Inst);
+ WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+
+ printf("\nDone with simulation\n\n");
+
+ return(0);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
new file mode 100644
index 0000000..ee5e484
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -0,0 +1,207 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+iLBC Speech Coder ANSI-C Source Code
+
+iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include "ilbc.h"
+
+//#define JUNK_DATA
+#ifdef JUNK_DATA
+#define SEED_FILE "randseed.txt"
+#endif
+
+
+/*----------------------------------------------------------------*
+* Main program to test iLBC encoding and decoding
+*
+* Usage:
+* exefile_name.exe <infile> <bytefile> <outfile>
+*
+*---------------------------------------------------------------*/
+
+int main(int argc, char* argv[])
+{
+ FILE *ifileid,*efileid,*ofileid, *chfileid;
+ short encoded_data[55], data[240], speechType;
+ short len, mode, pli;
+ int blockcount = 0;
+
+ iLBC_encinst_t *Enc_Inst;
+ iLBC_decinst_t *Dec_Inst;
+#ifdef JUNK_DATA
+ int i;
+ FILE *seedfile;
+ unsigned int random_seed = (unsigned int) time(NULL);//1196764538
+#endif
+
+ /* Create structs */
+ WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+ WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+ /* get arguments and open files */
+
+ if (argc != 6 ) {
+ fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+ argv[0]);
+ fprintf(stderr, "Example:\n");
+ fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+ exit(1);
+ }
+ mode=atoi(argv[1]);
+ if (mode != 20 && mode != 30) {
+ fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+ exit(2);
+ }
+ if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+ exit(2);}
+ if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open channelfile file %s\n",
+ argv[3]); exit(3);}
+ if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open output file %s\n",
+ argv[4]); exit(3);}
+ if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+ exit(2);
+ }
+ /* print info */
+ fprintf(stderr, "\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* iLBCtest *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+ fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+ fprintf(stderr,"\nMode : %2d ms\n", mode);
+#endif
+ fprintf(stderr,"\nInput file : %s\n", argv[2]);
+ fprintf(stderr,"Coded file : %s\n", argv[3]);
+ fprintf(stderr,"Output file : %s\n\n", argv[4]);
+ fprintf(stderr,"Channel file : %s\n\n", argv[5]);
+
+#ifdef JUNK_DATA
+ srand(random_seed);
+
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "%u\n", random_seed);
+ fclose(seedfile);
+ }
+#endif
+
+ /* Initialization */
+ WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+ WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+
+ /* loop over input blocks */
+#ifdef SPLIT_10MS
+ while(fread(data, sizeof(short), 80, ifileid) == 80) {
+#else
+ while((short)fread(data,sizeof(short),(mode<<3),ifileid)==(mode<<3)) {
+#endif
+ blockcount++;
+
+ /* encoding */
+ fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+#ifdef SPLIT_10MS
+ len=WebRtcIlbcfix_Encode(Enc_Inst, data, 80, encoded_data);
+#else
+ len=WebRtcIlbcfix_Encode(Enc_Inst, data, (short)(mode<<3), encoded_data);
+#endif
+ fprintf(stderr, "\r");
+
+#ifdef JUNK_DATA
+ for ( i = 0; i < len; i++) {
+ encoded_data[i] = (short) (encoded_data[i] + (short) rand());
+ }
+#endif
+ /* write byte file */
+ if(len != 0){ //len may be 0 in 10ms split case
+ fwrite(encoded_data,1,len,efileid);
+ }
+
+ if(len != 0){ //len may be 0 in 10ms split case
+ /* get channel data if provided */
+ if (argc==6) {
+ if (fread(&pli, sizeof(WebRtc_Word16), 1, chfileid)) {
+ if ((pli!=0)&&(pli!=1)) {
+ fprintf(stderr, "Error in channel file\n");
+ exit(0);
+ }
+ if (pli==0) {
+ /* Packet loss -> remove info from frame */
+ memset(encoded_data, 0, sizeof(WebRtc_Word16)*25);
+ }
+ } else {
+ fprintf(stderr, "Error. Channel file too short\n");
+ exit(0);
+ }
+ } else {
+ pli=1;
+ }
+
+ /* decoding */
+ fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+ if (pli==1) {
+ len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, &speechType);
+ } else {
+ len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
+ }
+ fprintf(stderr, "\r");
+
+ /* write output file */
+ fwrite(data,sizeof(short),len,ofileid);
+ }
+ }
+
+#ifdef JUNK_DATA
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "ok\n\n");
+ fclose(seedfile);
+ }
+#endif
+
+ /* free structs */
+ WebRtcIlbcfix_EncoderFree(Enc_Inst);
+ WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+ /* close files */
+ fclose(ifileid);
+ fclose(efileid);
+ fclose(ofileid);
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
new file mode 100644
index 0000000..f67945e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -0,0 +1,343 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "defines.h"
+#include "nit_encode.h"
+#include "encode.h"
+#include "init_decode.h"
+#include "decode.h"
+#include "constants.h"
+#include "ilbc.h"
+
+#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
+
+/* Runtime statistics */
+#include <time.h>
+/* #define CLOCKS_PER_SEC 1000 */
+
+/*----------------------------------------------------------------*
+ * Encoder interface function
+ *---------------------------------------------------------------*/
+
+short encode( /* (o) Number of bytes encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ WebRtc_Word16 *encoded_data, /* (o) The encoded bytes */
+ WebRtc_Word16 *data /* (i) The signal block to encode */
+ ){
+
+ /* do the actual encoding */
+ WebRtcIlbcfix_Encode((WebRtc_UWord16 *)encoded_data, data, iLBCenc_inst);
+
+ return (iLBCenc_inst->no_of_bytes);
+}
+
+/*----------------------------------------------------------------*
+ * Decoder interface function
+ *---------------------------------------------------------------*/
+
+short decode( /* (o) Number of decoded samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ short *decoded_data, /* (o) Decoded signal block */
+ short *encoded_data, /* (i) Encoded bytes */
+ short mode /* (i) 0=PL, 1=Normal */
+ ){
+
+ /* check if mode is valid */
+
+ if (mode<0 || mode>1) {
+ printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
+
+ /* do actual decoding of block */
+
+ WebRtcIlbcfix_Decode(decoded_data, (WebRtc_UWord16 *)encoded_data,
+ iLBCdec_inst, mode);
+
+ return (iLBCdec_inst->blockl);
+}
+
+/*----------------------------------------------------------------*
+ * Main program to test iLBC encoding and decoding
+ *
+ * Usage:
+ * exefile_name.exe <infile> <bytefile> <outfile> <channelfile>
+ *
+ *---------------------------------------------------------------*/
+
+#define MAXFRAMES 10000
+#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES)
+
+int main(int argc, char* argv[])
+{
+
+ /* Runtime statistics */
+
+ float starttime1, starttime2;
+ float runtime1, runtime2;
+ float outtime;
+
+ FILE *ifileid,*efileid,*ofileid, *chfileid;
+ short *inputdata, *encodeddata, *decodeddata;
+ short *channeldata;
+ int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
+ short mode;
+ iLBC_Enc_Inst_t Enc_Inst;
+ iLBC_Dec_Inst_t Dec_Inst;
+
+ short frameLen;
+ short count;
+#ifdef SPLIT_10MS
+ short size;
+#endif
+
+ inputdata=(short*) malloc(MAXFILELEN*sizeof(short));
+ if (inputdata==NULL) {
+ fprintf(stderr,"Could not allocate memory for vector\n");
+ exit(0);
+ }
+ encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short));
+ if (encodeddata==NULL) {
+ fprintf(stderr,"Could not allocate memory for vector\n");
+ free(inputdata);
+ exit(0);
+ }
+ decodeddata=(short*) malloc(MAXFILELEN*sizeof(short));
+ if (decodeddata==NULL) {
+ fprintf(stderr,"Could not allocate memory for vector\n");
+ free(inputdata);
+ free(encodeddata);
+ exit(0);
+ }
+ channeldata=(short*) malloc(MAXFRAMES*sizeof(short));
+ if (channeldata==NULL) {
+ fprintf(stderr,"Could not allocate memory for vector\n");
+ free(inputdata);
+ free(encodeddata);
+ free(decodeddata);
+ exit(0);
+ }
+
+ /* get arguments and open files */
+
+ if (argc != 6 ) {
+ fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+ argv[0]);
+ fprintf(stderr, "Example:\n");
+ fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+ exit(1);
+ }
+ mode=atoi(argv[1]);
+ if (mode != 20 && mode != 30) {
+ fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+ exit(2);
+ }
+ if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+ exit(2);}
+ if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open channelfile file %s\n",
+ argv[3]); exit(3);}
+ if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+ fprintf(stderr, "Cannot open output file %s\n",
+ argv[4]); exit(3);}
+ if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+ fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+ exit(2);}
+
+
+ /* print info */
+#ifndef PRINT_MIPS
+ fprintf(stderr, "\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* iLBCtest *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "* *\n");
+ fprintf(stderr,
+ "*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+ fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+ fprintf(stderr,"\nMode : %2d ms\n", mode);
+#endif
+ fprintf(stderr,"\nInput file : %s\n", argv[2]);
+ fprintf(stderr,"Coded file : %s\n", argv[3]);
+ fprintf(stderr,"Output file : %s\n\n", argv[4]);
+ fprintf(stderr,"Channel file : %s\n\n", argv[5]);
+#endif
+
+ /* Initialization */
+
+ WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode);
+ WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1);
+
+ /* extract the input file and channel file */
+
+#ifdef SPLIT_10MS
+ frameLen = (mode==20)? 80:160;
+ fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid);
+ Enc_Inst.section = 0;
+
+ while( fread(&inputdata[noOfBlocks*80], sizeof(short),
+ 80, ifileid) == 80 ) {
+ noOfBlocks++;
+ }
+
+ noOfBlocks += frameLen/80;
+ frameLen = 80;
+#else
+ frameLen = Enc_Inst.blockl;
+
+ while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
+ Enc_Inst.blockl,ifileid)==(WebRtc_UWord16)Enc_Inst.blockl){
+ noOfBlocks++;
+ }
+#endif
+
+
+ while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1)
+ && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) {
+ blockcount++;
+ }
+
+ if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) {
+ fprintf(stderr,"Channel file %s is too short\n", argv[4]);
+ free(inputdata);
+ free(encodeddata);
+ free(decodeddata);
+ free(channeldata);
+ exit(0);
+ }
+
+ count=0;
+
+ /* Runtime statistics */
+
+ starttime1 = clock()/(float)CLOCKS_PER_SEC;
+
+ /* Encoding loop */
+#ifdef PRINT_MIPS
+ printf("-1 -1\n");
+#endif
+
+#ifdef SPLIT_10MS
+ /* "Enc_Inst.section != 0" is to make sure we run through full
+ lengths of all vectors for 10ms split mode.
+ */
+ // while( (count < noOfBlocks) || (Enc_Inst.section != 0) ) {
+ while( count < blockcount * (Enc_Inst.blockl/frameLen) ) {
+
+ encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words *
+ (count/(Enc_Inst.nsub/2))],
+ &inputdata[frameLen * count] );
+#else
+ while (count < noOfBlocks) {
+ encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count],
+ &inputdata[frameLen * count] );
+#endif
+
+#ifdef PRINT_MIPS
+ printf("-1 -1\n");
+#endif
+
+ count++;
+ }
+
+ count=0;
+
+ /* Runtime statistics */
+
+ starttime2=clock()/(float)CLOCKS_PER_SEC;
+ runtime1 = (float)(starttime2-starttime1);
+
+ /* Decoding loop */
+
+ while (count < blockcount) {
+ if (channeldata[count]==1) {
+ /* Normal decoding */
+ decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl],
+ &encodeddata[Dec_Inst.no_of_words * count], 1);
+ } else if (channeldata[count]==0) {
+ /* PLC */
+ short emptydata[ILBCNOOFWORDS_MAX];
+ memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short));
+ decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl],
+ emptydata, 0);
+ noOfLostBlocks++;
+ } else {
+ printf("Error in channel file (values have to be either 1 or 0)\n");
+ exit(0);
+ }
+#ifdef PRINT_MIPS
+ printf("-1 -1\n");
+#endif
+
+ count++;
+ }
+
+ /* Runtime statistics */
+
+ runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2);
+
+ outtime = (float)((float)blockcount*
+ (float)mode/1000.0);
+
+#ifndef PRINT_MIPS
+ printf("\nLength of speech file: %.1f s\n", outtime);
+ printf("Lost frames : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount);
+
+ printf("Time to run iLBC_encode+iLBC_decode:");
+ printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2,
+ (100*(runtime1+runtime2)/outtime));
+
+ printf("Time in iLBC_encode :");
+ printf(" %.1f s (%.1f%% of total runtime)\n",
+ runtime1, 100.0*runtime1/(runtime1+runtime2));
+
+ printf("Time in iLBC_decode :");
+ printf(" %.1f s (%.1f%% of total runtime)\n\n",
+ runtime2, 100.0*runtime2/(runtime1+runtime2));
+#endif
+
+ /* Write data to files */
+ for (i=0; i<blockcount; i++) {
+ fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short),
+ Enc_Inst.no_of_words, efileid);
+ }
+ for (i=0;i<blockcount;i++) {
+ fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid);
+ }
+
+ /* return memory and close files */
+
+ free(inputdata);
+ free(encodeddata);
+ free(decodeddata);
+ free(channeldata);
+ fclose(ifileid); fclose(efileid); fclose(ofileid);
+ return(0);
+ }
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
new file mode 100644
index 0000000..a52f7c1
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
@@ -0,0 +1,73 @@
+#!/bin/bash
+(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
+
+#
+# This script can be used to verify the bit exactness of iLBC fixed-point version 1.0.6
+#
+
+INP=../../../../../../../test/data/audio_coding
+EXEP=../../../../../../../out/Release
+OUTP=./GeneratedFiles
+mkdir ./GeneratedFiles
+
+$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F03.INP $OUTP/F03.BIT20 $OUTP/F03.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F04.INP $OUTP/F04.BIT20 $OUTP/F04.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F05.INP $OUTP/F05.BIT20 $OUTP/F05.OUT20 $INP/clean.chn
+$EXEP/iLBCtest 20 $INP/F06.INP $OUTP/F06.BIT20 $OUTP/F06.OUT20 $INP/clean.chn
+
+$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F03.INP $OUTP/F03.BIT30 $OUTP/F03.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F04.INP $OUTP/F04.BIT30 $OUTP/F04.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F05.INP $OUTP/F05.BIT30 $OUTP/F05.OUT30 $INP/clean.chn
+$EXEP/iLBCtest 30 $INP/F06.INP $OUTP/F06.BIT30 $OUTP/F06.OUT30 $INP/clean.chn
+
+$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00_tlm10.OUT20 $INP/tlm10.chn
+$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01_tlm10.OUT20 $INP/tlm10.chn
+$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02_tlm10.OUT20 $INP/tlm10.chn
+$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00_tlm10.OUT30 $INP/tlm10.chn
+$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01_tlm10.OUT30 $INP/tlm10.chn
+$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02_tlm10.OUT30 $INP/tlm10.chn
+
+
+diff $OUTP/F00.BIT20 $INP/F00.BIT20
+diff $OUTP/F01.BIT20 $INP/F01.BIT20
+diff $OUTP/F02.BIT20 $INP/F02.BIT20
+diff $OUTP/F03.BIT20 $INP/F03.BIT20
+diff $OUTP/F04.BIT20 $INP/F04.BIT20
+diff $OUTP/F05.BIT20 $INP/F05.BIT20
+diff $OUTP/F06.BIT20 $INP/F06.BIT20
+diff $OUTP/F00.OUT20 $INP/F00.OUT20
+diff $OUTP/F01.OUT20 $INP/F01.OUT20
+diff $OUTP/F02.OUT20 $INP/F02.OUT20
+diff $OUTP/F03.OUT20 $INP/F03.OUT20
+diff $OUTP/F04.OUT20 $INP/F04.OUT20
+diff $OUTP/F05.OUT20 $INP/F05.OUT20
+diff $OUTP/F06.OUT20 $INP/F06.OUT20
+
+diff $OUTP/F00.BIT30 $INP/F00.BIT30
+diff $OUTP/F01.BIT30 $INP/F01.BIT30
+diff $OUTP/F02.BIT30 $INP/F02.BIT30
+diff $OUTP/F03.BIT30 $INP/F03.BIT30
+diff $OUTP/F04.BIT30 $INP/F04.BIT30
+diff $OUTP/F05.BIT30 $INP/F05.BIT30
+diff $OUTP/F06.BIT30 $INP/F06.BIT30
+diff $OUTP/F00.OUT30 $INP/F00.OUT30
+diff $OUTP/F01.OUT30 $INP/F01.OUT30
+diff $OUTP/F02.OUT30 $INP/F02.OUT30
+diff $OUTP/F03.OUT30 $INP/F03.OUT30
+diff $OUTP/F04.OUT30 $INP/F04.OUT30
+diff $OUTP/F05.OUT30 $INP/F05.OUT30
+diff $OUTP/F06.OUT30 $INP/F06.OUT30
+
+diff $OUTP/F00_tlm10.OUT20 $INP/F00_tlm10.OUT20
+diff $OUTP/F01_tlm10.OUT20 $INP/F01_tlm10.OUT20
+diff $OUTP/F02_tlm10.OUT20 $INP/F02_tlm10.OUT20
+diff $OUTP/F00_tlm10.OUT30 $INP/F00_tlm10.OUT30
+diff $OUTP/F01_tlm10.OUT30 $INP/F01_tlm10.OUT30
+diff $OUTP/F02_tlm10.OUT30 $INP/F02_tlm10.OUT30
+
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c
new file mode 100644
index 0000000..6c883a7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c
@@ -0,0 +1,239 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+ WebRtc_UWord16 *bitstream, /* (i) The packatized bitstream */
+ iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */
+ WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
+ ) {
+ WebRtc_UWord16 *bitstreamPtr;
+ int i, k;
+ WebRtc_Word16 *tmpPtr;
+
+ bitstreamPtr=bitstream;
+
+ /* First WebRtc_Word16 */
+ enc_bits->lsf[0] = (*bitstreamPtr)>>10; /* Bit 0..5 */
+ enc_bits->lsf[1] = ((*bitstreamPtr)>>3)&0x7F; /* Bit 6..12 */
+ enc_bits->lsf[2] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* Second WebRtc_Word16 */
+ enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */
+
+ if (mode==20) {
+ enc_bits->startIdx = ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */
+ enc_bits->state_first = ((*bitstreamPtr)>>9)&0x1; /* Bit 6 */
+ enc_bits->idxForMax = ((*bitstreamPtr)>>3)&0x3F; /* Bit 7..12 */
+ enc_bits->cb_index[0] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* Third WebRtc_Word16 */
+ enc_bits->cb_index[0] |= ((*bitstreamPtr)>>12)&0xE; /* Bit 0..2 */
+ enc_bits->gain_index[0] = ((*bitstreamPtr)>>8)&0x18; /* Bit 3..4 */
+ enc_bits->gain_index[1] = ((*bitstreamPtr)>>7)&0x8; /* Bit 5 */
+ enc_bits->cb_index[3] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
+ enc_bits->gain_index[3] = ((*bitstreamPtr)<<2)&0x10; /* Bit 13 */
+ enc_bits->gain_index[4] = ((*bitstreamPtr)<<2)&0x8; /* Bit 14 */
+ enc_bits->gain_index[6] = ((*bitstreamPtr)<<4)&0x10; /* Bit 15 */
+ } else { /* mode==30 */
+ enc_bits->lsf[3] = ((*bitstreamPtr)>>6)&0x3F; /* Bit 4..9 */
+ enc_bits->lsf[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* Third WebRtc_Word16 */
+ enc_bits->lsf[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
+ enc_bits->lsf[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */
+ enc_bits->startIdx = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+ enc_bits->state_first = ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
+ enc_bits->idxForMax = ((*bitstreamPtr)<<2)&0x3C; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 4:th WebRtc_Word16 */
+ enc_bits->idxForMax |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
+ enc_bits->cb_index[0] = ((*bitstreamPtr)>>7)&0x78; /* Bit 2..5 */
+ enc_bits->gain_index[0] = ((*bitstreamPtr)>>5)&0x10; /* Bit 6 */
+ enc_bits->gain_index[1] = ((*bitstreamPtr)>>5)&0x8; /* Bit 7 */
+ enc_bits->cb_index[3] = ((*bitstreamPtr))&0xFC; /* Bit 8..13 */
+ enc_bits->gain_index[3] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */
+ enc_bits->gain_index[4] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */
+ }
+ /* Class 2 bits of ULP */
+ /* 4:th to 6:th WebRtc_Word16 for 20 ms case
+ 5:th to 7:th WebRtc_Word16 for 30 ms case */
+ bitstreamPtr++;
+ tmpPtr=enc_bits->idxVec;
+ for (k=0; k<3; k++) {
+ for (i=15; i>=0; i--) {
+ (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ bitstreamPtr++;
+ }
+
+ if (mode==20) {
+ /* 7:th WebRtc_Word16 */
+ for (i=15; i>6; i--) {
+ (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ enc_bits->gain_index[1] |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9 */
+ enc_bits->gain_index[3] |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+ enc_bits->gain_index[4] |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12 */
+ enc_bits->gain_index[6] |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */
+ enc_bits->gain_index[7] = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
+
+ } else { /* mode==30 */
+ /* 8:th WebRtc_Word16 */
+ for (i=15; i>5; i--) {
+ (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
+ /* Bit 15-i */
+ tmpPtr++;
+ }
+ enc_bits->cb_index[0] |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */
+ enc_bits->gain_index[0] |= ((*bitstreamPtr))&0x8; /* Bit 12 */
+ enc_bits->gain_index[1] |= ((*bitstreamPtr))&0x4; /* Bit 13 */
+ enc_bits->cb_index[3] |= ((*bitstreamPtr))&0x2; /* Bit 14 */
+ enc_bits->cb_index[6] = ((*bitstreamPtr)<<7)&0x80; /* Bit 15 */
+ bitstreamPtr++;
+ /* 9:th WebRtc_Word16 */
+ enc_bits->cb_index[6] |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5 */
+ enc_bits->cb_index[9] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
+ enc_bits->cb_index[12] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* 10:th WebRtc_Word16 */
+ enc_bits->cb_index[12] |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
+ enc_bits->gain_index[3] |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5 */
+ enc_bits->gain_index[4] |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7 */
+ enc_bits->gain_index[6] = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9 */
+ enc_bits->gain_index[7] = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+ enc_bits->gain_index[9] = ((*bitstreamPtr)<<1)&0x10; /* Bit 12 */
+ enc_bits->gain_index[10] = ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */
+ enc_bits->gain_index[12] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */
+ enc_bits->gain_index[13] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */
+ }
+ bitstreamPtr++;
+ /* Class 3 bits of ULP */
+ /* 8:th to 14:th WebRtc_Word16 for 20 ms case
+ 11:th to 17:th WebRtc_Word16 for 30 ms case */
+ tmpPtr=enc_bits->idxVec;
+ for (k=0; k<7; k++) {
+ for (i=14; i>=0; i-=2) {
+ (*tmpPtr) |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/
+ tmpPtr++;
+ }
+ bitstreamPtr++;
+ }
+
+ if (mode==20) {
+ /* 15:th WebRtc_Word16 */
+ enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
+ enc_bits->cb_index[0] |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2 */
+ enc_bits->cb_index[1] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */
+ enc_bits->cb_index[2] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* 16:th WebRtc_Word16 */
+ enc_bits->cb_index[2] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
+ enc_bits->gain_index[0] |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3 */
+ enc_bits->gain_index[1] |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */
+ enc_bits->gain_index[2] = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8 */
+ enc_bits->cb_index[3] |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9 */
+ enc_bits->cb_index[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+ bitstreamPtr++;
+ /* 17:th WebRtc_Word16 */
+ enc_bits->cb_index[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
+ enc_bits->cb_index[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */
+ enc_bits->cb_index[6] = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
+ bitstreamPtr++;
+ /* 18:th WebRtc_Word16 */
+ enc_bits->cb_index[7] = (*bitstreamPtr)>>8; /* Bit 0..7 */
+ enc_bits->cb_index[8] = (*bitstreamPtr)&0xFF; /* Bit 8..15 */
+ bitstreamPtr++;
+ /* 19:th WebRtc_Word16 */
+ enc_bits->gain_index[3] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
+ enc_bits->gain_index[4] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */
+ enc_bits->gain_index[5] = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6 */
+ enc_bits->gain_index[6] |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9 */
+ enc_bits->gain_index[7] |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
+ enc_bits->gain_index[8] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+ } else { /* mode==30 */
+ /* 18:th WebRtc_Word16 */
+ enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
+ enc_bits->idxVec[57] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */
+ enc_bits->cb_index[0] |= ((*bitstreamPtr)>>11)&1; /* Bit 4 */
+ enc_bits->cb_index[1] = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
+ enc_bits->cb_index[2] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 19:th WebRtc_Word16 */
+ enc_bits->cb_index[2] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
+ enc_bits->gain_index[0] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */
+ enc_bits->gain_index[1] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */
+ enc_bits->gain_index[2] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+ enc_bits->cb_index[3] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
+ enc_bits->cb_index[4] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 20:th WebRtc_Word16 */
+ enc_bits->cb_index[4] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
+ enc_bits->cb_index[5] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */
+ enc_bits->cb_index[6] |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10 */
+ enc_bits->cb_index[7] = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
+ bitstreamPtr++;
+ /* 21:st WebRtc_Word16 */
+ enc_bits->cb_index[7] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
+ enc_bits->cb_index[8] = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
+ enc_bits->cb_index[9] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
+ enc_bits->cb_index[10] = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
+ bitstreamPtr++;
+ /* 22:nd WebRtc_Word16 */
+ enc_bits->cb_index[10] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */
+ enc_bits->cb_index[11] = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
+ enc_bits->cb_index[12] |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12 */
+ enc_bits->cb_index[13] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+ bitstreamPtr++;
+ /* 23:rd WebRtc_Word16 */
+ enc_bits->cb_index[13] |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4 */
+ enc_bits->cb_index[14] = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
+ enc_bits->gain_index[3] |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
+ enc_bits->gain_index[4] |= ((*bitstreamPtr)&0x1); /* Bit 15 */
+ bitstreamPtr++;
+ /* 24:rd WebRtc_Word16 */
+ enc_bits->gain_index[5] = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
+ enc_bits->gain_index[6] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */
+ enc_bits->gain_index[7] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */
+ enc_bits->gain_index[8] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+ enc_bits->gain_index[9] |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
+ enc_bits->gain_index[10] |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15 */
+ bitstreamPtr++;
+ /* 25:rd WebRtc_Word16 */
+ enc_bits->gain_index[10] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
+ enc_bits->gain_index[11] = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4 */
+ enc_bits->gain_index[12] |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8 */
+ enc_bits->gain_index[13] |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */
+ enc_bits->gain_index[14] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+ }
+ /* Last bit should be zero, otherwise it's an "empty" frame */
+ if (((*bitstreamPtr)&0x1) == 1) {
+ return(1);
+ } else {
+ return(0);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h
new file mode 100644
index 0000000..864865f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+ WebRtc_UWord16 *bitstream, /* (i) The packatized bitstream */
+ iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */
+ WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c
new file mode 100644
index 0000000..81d1bfa
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.c
+
+******************************************************************/
+
+#include "vq3.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+ WebRtc_Word16 *Xq, /* quantized vector (Q13) */
+ WebRtc_Word16 *index,
+ WebRtc_Word16 *CB, /* codebook in Q13 */
+ WebRtc_Word16 *X, /* vector to quantize (Q13) */
+ WebRtc_Word16 n_cb
+ ){
+ WebRtc_Word16 i, j;
+ WebRtc_Word16 pos, minindex=0;
+ WebRtc_Word16 tmp;
+ WebRtc_Word32 dist, mindist;
+
+ pos = 0;
+ mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+ /* Find the codebook with the lowest square distance */
+ for (j = 0; j < n_cb; j++) {
+ tmp = X[0] - CB[pos];
+ dist = WEBRTC_SPL_MUL_16_16(tmp, tmp);
+ for (i = 1; i < 3; i++) {
+ tmp = X[i] - CB[pos + i];
+ dist += WEBRTC_SPL_MUL_16_16(tmp, tmp);
+ }
+
+ if (dist < mindist) {
+ mindist = dist;
+ minindex = j;
+ }
+ pos += 3;
+ }
+
+ /* Store the quantized codebook and the index */
+ for (i = 0; i < 3; i++) {
+ Xq[i] = CB[minindex*3 + i];
+ }
+ *index = minindex;
+
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h
new file mode 100644
index 0000000..f2628e0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+
+#include "typedefs.h"
+
+/*----------------------------------------------------------------*
+ * Vector quantization of order 3 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+ WebRtc_Word16 *Xq, /* (o) the quantized vector (Q13) */
+ WebRtc_Word16 *index, /* (o) the quantization index */
+ WebRtc_Word16 *CB, /* (i) the vector quantization codebook (Q13) */
+ WebRtc_Word16 *X, /* (i) the vector to quantize (Q13) */
+ WebRtc_Word16 n_cb /* (i) the number of vectors in the codebook */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c
new file mode 100644
index 0000000..3d4c26d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.c
+
+******************************************************************/
+
+#include "vq4.h"
+#include "constants.h"
+
+/*----------------------------------------------------------------*
+ * vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+ WebRtc_Word16 *Xq, /* quantized vector (Q13) */
+ WebRtc_Word16 *index,
+ WebRtc_Word16 *CB, /* codebook in Q13 */
+ WebRtc_Word16 *X, /* vector to quantize (Q13) */
+ WebRtc_Word16 n_cb
+ ){
+ WebRtc_Word16 i, j;
+ WebRtc_Word16 pos, minindex=0;
+ WebRtc_Word16 tmp;
+ WebRtc_Word32 dist, mindist;
+
+ pos = 0;
+ mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+ /* Find the codebook with the lowest square distance */
+ for (j = 0; j < n_cb; j++) {
+ tmp = X[0] - CB[pos];
+ dist = WEBRTC_SPL_MUL_16_16(tmp, tmp);
+ for (i = 1; i < 4; i++) {
+ tmp = X[i] - CB[pos + i];
+ dist += WEBRTC_SPL_MUL_16_16(tmp, tmp);
+ }
+
+ if (dist < mindist) {
+ mindist = dist;
+ minindex = j;
+ }
+ pos += 4;
+ }
+
+ /* Store the quantized codebook and the index */
+ for (i = 0; i < 4; i++) {
+ Xq[i] = CB[minindex*4 + i];
+ }
+ *index = minindex;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h
new file mode 100644
index 0000000..1b8cff2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+
+#include "typedefs.h"
+
+/*----------------------------------------------------------------*
+ * Vector quantization of order 4 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+ WebRtc_Word16 *Xq, /* (o) the quantized vector (Q13) */
+ WebRtc_Word16 *index, /* (o) the quantization index */
+ WebRtc_Word16 *CB, /* (i) the vector quantization codebook (Q13) */
+ WebRtc_Word16 *X, /* (i) the vector to quantize (Q13) */
+ WebRtc_Word16 n_cb /* (i) the number of vectors in the codebook */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c
new file mode 100644
index 0000000..b0e8406
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(
+ WebRtc_Word32 *z, /* Output */
+ WebRtc_Word32 *x, /* Input (same domain as Output)*/
+ const WebRtc_Word32 *y, /* Q31 Window */
+ WebRtc_Word16 N /* length to process */
+ ) {
+ WebRtc_Word16 i;
+ WebRtc_Word16 x_low, x_hi, y_low, y_hi;
+ WebRtc_Word16 left_shifts;
+ WebRtc_Word32 temp;
+
+ left_shifts = (WebRtc_Word16)WebRtcSpl_NormW32(x[0]);
+ WebRtcSpl_VectorBitShiftW32(x, N, x, (WebRtc_Word16)(-left_shifts));
+
+
+ /* The double precision numbers use a special representation:
+ * w32 = hi<<16 + lo<<1
+ */
+ for (i = 0; i < N; i++) {
+ /* Extract higher bytes */
+ x_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(x[i], 16);
+ y_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(y[i], 16);
+
+ /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
+ temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_hi, 16);
+ x_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((x[i] - temp), 1);
+
+ temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y_hi, 16);
+ y_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((y[i] - temp), 1);
+
+ /* Calculate z by a 32 bit multiplication using both low and high from x and y */
+ temp = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(x_hi, y_hi), 1);
+ temp = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_hi, y_low, 14)));
+
+ z[i] = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_low, y_hi, 14)));
+ }
+
+ WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts);
+
+ return;
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h
new file mode 100644
index 0000000..121188a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(
+ WebRtc_Word32 *z, /* Output */
+ WebRtc_Word32 *x, /* Input (same domain as Output)*/
+ const WebRtc_Word32 *y, /* Q31 Window */
+ WebRtc_Word16 N /* length to process */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c
new file mode 100644
index 0000000..04170ad
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_XcorrCoef(
+ WebRtc_Word16 *target, /* (i) first array */
+ WebRtc_Word16 *regressor, /* (i) second array */
+ WebRtc_Word16 subl, /* (i) dimension arrays */
+ WebRtc_Word16 searchLen, /* (i) the search lenght */
+ WebRtc_Word16 offset, /* (i) samples offset between arrays */
+ WebRtc_Word16 step /* (i) +1 or -1 */
+ ){
+ int k;
+ WebRtc_Word16 maxlag;
+ WebRtc_Word16 pos;
+ WebRtc_Word16 max;
+ WebRtc_Word16 crossCorrScale, Energyscale;
+ WebRtc_Word16 crossCorrSqMod, crossCorrSqMod_Max;
+ WebRtc_Word32 crossCorr, Energy;
+ WebRtc_Word16 crossCorrmod, EnergyMod, EnergyMod_Max;
+ WebRtc_Word16 *tp, *rp;
+ WebRtc_Word16 *rp_beg, *rp_end;
+ WebRtc_Word16 totscale, totscale_max;
+ WebRtc_Word16 scalediff;
+ WebRtc_Word32 newCrit, maxCrit;
+ int shifts;
+
+ /* Initializations, to make sure that the first one is selected */
+ crossCorrSqMod_Max=0;
+ EnergyMod_Max=WEBRTC_SPL_WORD16_MAX;
+ totscale_max=-500;
+ maxlag=0;
+ pos=0;
+
+ /* Find scale value and start position */
+ if (step==1) {
+ max=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(subl+searchLen-1));
+ rp_beg = regressor;
+ rp_end = ®ressor[subl];
+ } else { /* step==-1 */
+ max=WebRtcSpl_MaxAbsValueW16(®ressor[-searchLen], (WebRtc_Word16)(subl+searchLen-1));
+ rp_beg = ®ressor[-1];
+ rp_end = ®ressor[subl-1];
+ }
+
+ /* Introduce a scale factor on the Energy in WebRtc_Word32 in
+ order to make sure that the calculation does not
+ overflow */
+
+ if (max>5000) {
+ shifts=2;
+ } else {
+ shifts=0;
+ }
+
+ /* Calculate the first energy, then do a +/- to get the other energies */
+ Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts);
+
+ for (k=0;k<searchLen;k++) {
+ tp = target;
+ rp = ®ressor[pos];
+
+ crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts);
+
+ if ((Energy>0)&&(crossCorr>0)) {
+
+ /* Put cross correlation and energy on 16 bit word */
+ crossCorrScale=(WebRtc_Word16)WebRtcSpl_NormW32(crossCorr)-16;
+ crossCorrmod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
+ Energyscale=(WebRtc_Word16)WebRtcSpl_NormW32(Energy)-16;
+ EnergyMod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
+
+ /* Square cross correlation and store upper WebRtc_Word16 */
+ crossCorrSqMod=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(crossCorrmod, crossCorrmod, 16);
+
+ /* Calculate the total number of (dynamic) right shifts that have
+ been performed on (crossCorr*crossCorr)/energy
+ */
+ totscale=Energyscale-(crossCorrScale<<1);
+
+ /* Calculate the shift difference in order to be able to compare the two
+ (crossCorr*crossCorr)/energy in the same domain
+ */
+ scalediff=totscale-totscale_max;
+ scalediff=WEBRTC_SPL_MIN(scalediff,31);
+ scalediff=WEBRTC_SPL_MAX(scalediff,-31);
+
+ /* Compute the cross multiplication between the old best criteria
+ and the new one to be able to compare them without using a
+ division */
+
+ if (scalediff<0) {
+ newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
+ maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod);
+ } else {
+ newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max);
+ maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod)>>scalediff;
+ }
+
+ /* Store the new lag value if the new criteria is larger
+ than previous largest criteria */
+
+ if (newCrit > maxCrit) {
+ crossCorrSqMod_Max = crossCorrSqMod;
+ EnergyMod_Max = EnergyMod;
+ totscale_max = totscale;
+ maxlag = k;
+ }
+ }
+ pos+=step;
+
+ /* Do a +/- to get the next energy */
+ Energy += step*(WEBRTC_SPL_RSHIFT_W32(
+ ((WebRtc_Word32)(*rp_end)*(*rp_end)) - ((WebRtc_Word32)(*rp_beg)*(*rp_beg)),
+ shifts));
+ rp_beg+=step;
+ rp_end+=step;
+ }
+
+ return(maxlag+offset);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h
new file mode 100644
index 0000000..ac885c4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.h
+
+******************************************************************/
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_XcorrCoef(
+ WebRtc_Word16 *target, /* (i) first array */
+ WebRtc_Word16 *regressor, /* (i) second array */
+ WebRtc_Word16 subl, /* (i) dimension arrays */
+ WebRtc_Word16 searchLen, /* (i) the search lenght */
+ WebRtc_Word16 offset, /* (i) samples offset between arrays */
+ WebRtc_Word16 step /* (i) +1 or -1 */
+ );
+
+#endif
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk b/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk
new file mode 100644
index 0000000..2e88fa6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk
@@ -0,0 +1,37 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_pcm16b
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := pcm16b.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../../..
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
new file mode 100644
index 0000000..e3cac4d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
+/*
+ * Define the fixpoint numeric formats
+ */
+
+#include "typedefs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcPcm16b_EncodeW16(...)
+ *
+ * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ * - speechIn16b : Input speech vector
+ * - len : Number of samples in speech vector
+ *
+ * Output:
+ * - speechOut16b : Encoded data vector (big endian 16 bit)
+ *
+ * Returned value : Size in bytes of speechOut16b
+ */
+
+WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speechOut16b);
+
+/****************************************************************************
+ * WebRtcPcm16b_Encode(...)
+ *
+ * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ * - speech16b : Input speech vector
+ * - len : Number of samples in speech vector
+ *
+ * Output:
+ * - speech8b : Encoded data vector (big endian 16 bit)
+ *
+ * Returned value : Size in bytes of speech8b
+ */
+
+WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
+ WebRtc_Word16 len,
+ unsigned char *speech8b);
+
+/****************************************************************************
+ * WebRtcPcm16b_DecodeW16(...)
+ *
+ * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ * - speechIn16b : Encoded data vector (big endian 16 bit)
+ * - len : Number of bytes in speechIn16b
+ *
+ * Output:
+ * - speechOut16b : Decoded speech vector
+ *
+ * Returned value : Samples in speechOut16b
+ */
+
+WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
+ WebRtc_Word16 *speechIn16b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speechOut16b,
+ WebRtc_Word16* speechType);
+
+/****************************************************************************
+ * WebRtcPcm16b_Decode(...)
+ *
+ * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ * - speech8b : Encoded data vector (big endian 16 bit)
+ * - len : Number of bytes in speech8b
+ *
+ * Output:
+ * - speech16b : Decoded speech vector
+ *
+ * Returned value : Samples in speech16b
+ */
+
+
+WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speech16b);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* PCM16B */
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c
new file mode 100644
index 0000000..0cff5dd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -0,0 +1,105 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include "pcm16b.h"
+
+#include <stdlib.h>
+
+#include "typedefs.h"
+
+#ifdef WEBRTC_BIG_ENDIAN
+#include "signal_processing_library.h"
+#endif
+
+#define HIGHEND 0xFF00
+#define LOWEND 0xFF
+
+
+
+/* Encoder with WebRtc_Word16 Output */
+WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speechOut16b)
+{
+#ifdef WEBRTC_BIG_ENDIAN
+ WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);
+#else
+ int i;
+ for (i=0;i<len;i++) {
+ speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|((((WebRtc_UWord16)speechIn16b[i])<<8)&0xFF00);
+ }
+#endif
+ return(len<<1);
+}
+
+
+/* Encoder with char Output (old version) */
+WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
+ WebRtc_Word16 len,
+ unsigned char *speech8b)
+{
+ WebRtc_Word16 samples=len*2;
+ WebRtc_Word16 pos;
+ WebRtc_Word16 short1;
+ WebRtc_Word16 short2;
+ for (pos=0;pos<len;pos++) {
+ short1=HIGHEND & speech16b[pos];
+ short2=LOWEND & speech16b[pos];
+ short1=short1>>8;
+ speech8b[pos*2]=(unsigned char) short1;
+ speech8b[pos*2+1]=(unsigned char) short2;
+ }
+ return(samples);
+}
+
+
+/* Decoder with WebRtc_Word16 Input instead of char when the WebRtc_Word16 Encoder is used */
+WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
+ WebRtc_Word16 *speechIn16b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speechOut16b,
+ WebRtc_Word16* speechType)
+{
+#ifdef WEBRTC_BIG_ENDIAN
+ WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(WebRtc_Word16)+1)>>1));
+#else
+ int i;
+ int samples=len>>1;
+
+ for (i=0;i<samples;i++) {
+ speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|(((WebRtc_UWord16)(speechIn16b[i]&0xFF))<<8);
+ }
+#endif
+
+ *speechType=1;
+
+ // Avoid warning.
+ (void)(inst = NULL);
+
+ return(len>>1);
+}
+
+/* "old" version of the decoder that uses char as input (not used in NetEq any more) */
+WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
+ WebRtc_Word16 len,
+ WebRtc_Word16 *speech16b)
+{
+ WebRtc_Word16 samples=len>>1;
+ WebRtc_Word16 pos;
+ WebRtc_Word16 shortval;
+ for (pos=0;pos<samples;pos++) {
+ shortval=((unsigned short) speech8b[pos*2]);
+ shortval=(shortval<<8)&HIGHEND;
+ shortval=shortval|(((unsigned short) speech8b[pos*2+1])&LOWEND);
+ speech16b[pos]=shortval;
+ }
+ return(samples);
+}
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
new file mode 100644
index 0000000..38754e6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
@@ -0,0 +1,52 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'PCM16B',
+ 'type': '<(library)',
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/pcm16b.h',
+ 'pcm16b.c',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'pcm16b_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'PCM16B',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'pcm16b_unittest.cc',
+ ],
+ }, # PCM16B_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc
new file mode 100644
index 0000000..eb910b3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "pcm16b.h"
+#include "gtest/gtest.h"
+
+TEST(Pcm16bTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/main/OWNERS b/trunk/src/modules/audio_coding/main/OWNERS
new file mode 100644
index 0000000..e1e6256
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/OWNERS
@@ -0,0 +1,3 @@
+tina.legrand@webrtc.org
+turaj@webrtc.org
+jan.skoglund@webrtc.org
diff --git a/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h b/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h
new file mode 100644
index 0000000..ed13689
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h
@@ -0,0 +1,888 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
+
+#include "audio_coding_module_typedefs.h"
+#include "module.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+
+// forward declarations
+struct CodecInst;
+
+#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 Khz
+
+// Callback class used for sending data ready to be packetized
+class AudioPacketizationCallback {
+ public:
+ virtual ~AudioPacketizationCallback() {}
+
+ virtual WebRtc_Word32 SendData(
+ FrameType frameType, WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData, WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation) = 0;
+};
+
+// Callback class used for inband Dtmf detection
+class AudioCodingFeedback {
+ public:
+ virtual ~AudioCodingFeedback() {}
+
+ virtual WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf,
+ const bool end) = 0;
+};
+
+// Callback class used for reporting VAD decision
+class ACMVADCallback {
+ public:
+ virtual ~ACMVADCallback() {}
+
+ virtual WebRtc_Word32 InFrameType(WebRtc_Word16 frameType) = 0;
+};
+
+// Callback class used for reporting receiver statistics
+class ACMVQMonCallback {
+ public:
+ virtual ~ACMVQMonCallback() {}
+
+ virtual WebRtc_Word32 NetEqStatistics(
+ const WebRtc_Word32 id, // current ACM id
+ const WebRtc_UWord16 MIUsValid, // valid voice duration in ms
+ const WebRtc_UWord16 MIUsReplaced, // concealed voice duration in ms
+ const WebRtc_UWord8 eventFlags, // concealed voice flags
+ const WebRtc_UWord16 delayMS) = 0; // average delay in ms
+};
+
+class AudioCodingModule: public Module {
+ protected:
+ AudioCodingModule() {}
+ virtual ~AudioCodingModule() {}
+
+ public:
+ ///////////////////////////////////////////////////////////////////////////
+ // Creation and destruction of a ACM
+ //
+ static AudioCodingModule* Create(const WebRtc_Word32 id);
+
+ static void Destroy(AudioCodingModule* module);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Utility functions
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_UWord8 NumberOfCodecs()
+ // Returns number of supported codecs.
+ //
+ // Return value:
+ // number of supported codecs.
+ ///
+ static WebRtc_UWord8 NumberOfCodecs();
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Codec()
+ // Get supported codec with list number.
+ //
+ // Input:
+ // -listId : list number.
+ //
+ // Output:
+ // -codec : a structure where the parameters of the codec,
+ // given by list number is written to.
+ //
+ // Return value:
+ // -1 if the list number (listId) is invalid.
+ // 0 if succeeded.
+ //
+ static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, CodecInst& codec);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Codec()
+ // Get supported codec with the given codec name and sampling frequency.
+ // If the sampling frequency is -1 then the search will be only based on
+ // codec name.
+ //
+ // Input:
+ // -payloadName : name of the codec.
+ // -samplingFreqHz : samling frequency of the codec.
+ //
+ // Output:
+ // -codec : a structure where the parameters of the codec,
+ // given by name is written to.
+ //
+ // Return value:
+ // -1 if the list number (listId) is invalid.
+ // 0 if succeeded.
+ //
+ static WebRtc_Word32 Codec(const char* payloadName, CodecInst& codec,
+ const WebRtc_Word32 samplingFreqHz = -1);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Codec()
+ //
+ // Returns the list number of the given codec name and sampling frequency.
+ // If the sampling frequency is -1 then the search will be only based on
+ // codec name.
+ //
+ // Input:
+ // -payloadName : name of the codec.
+ // -samplingFreqHz : samling frequency of the codec.
+ //
+ // Return value:
+ // if the codec is found, the index of the codec in the list,
+ // -1 if the codec is not found.
+ //
+ static WebRtc_Word32 Codec(const char* payloadName,
+ const WebRtc_Word32 samplingFreqHz = -1);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool IsCodecValid()
+ // Checks the validity of the parameters of the given codec.
+ //
+ // Input:
+ // -codec : the structur which keeps the parameters of the
+ // codec.
+ //
+ // Reurn value:
+ // true if the parameters are valid,
+ // false if any parameter is not valid.
+ //
+ static bool IsCodecValid(const CodecInst& codec);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Sender
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 InitializeSender()
+ // Any encoder-related state of ACM will be initialized to the
+ // same state when ACM is created. This will not interrupt or
+ // effect decoding functionality of ACM. ACM will lose all the
+ // encoding-related settings by calling this function.
+ // For instance, a send codec has to be registered again.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 InitializeSender() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ResetEncoder()
+ // This API resets the states of encoder. All the encoder settings, such as
+ // send-codec or VAD/DTX, will be preserved.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 ResetEncoder() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterSendCodec()
+ // Registers a codec, specified by "sendCodec," as sending codec.
+ // This API can be called multiple of times to register Codec. The last codec
+ // registered overwrites the previous ones.
+ // The API can also be used to change payload type for CNG and RED, which are
+ // registered by default to default payload types.
+ // Note that registering CNG and RED won't overwrite speech codecs.
+ // This API can be called to set/change the send payload-type, frame-size
+ // or encoding rate (if applicable for the codec).
+ //
+ // Input:
+ // -sendCodec : Parameters of the codec to be registered, c.f.
+ // common_types.h for the definition of
+ // CodecInst.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 RegisterSendCodec(const CodecInst& sendCodec) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SendCodec()
+ // Get parameters for the codec currently registered as send codec.
+ //
+ // Output:
+ // -currentSendCodec : parameters of the send codec.
+ //
+ // Return value:
+ // -1 if failed to get send codec,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 SendCodec(CodecInst& currentSendCodec) const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SendFrequency()
+ // Get the sampling frequency of the current encoder in Hertz.
+ //
+ // Return value:
+ // positive; sampling frequency [Hz] of the current encoder.
+ // -1 if an error has happened.
+ //
+ virtual WebRtc_Word32 SendFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Bitrate()
+ // Get encoding bit-rate in bits per second.
+ //
+ // Return value:
+ // positive; encoding rate in bits/sec,
+ // -1 if an error is happened.
+ //
+ virtual WebRtc_Word32 SendBitrate() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetReceivedEstimatedBandwidth()
+ // Set available bandwidth [bits/sec] of the up-link channel.
+ // This information is used for traffic shaping, and is currently only
+ // supported if iSAC is the send codec.
+ //
+ // Input:
+ // -bw : bandwidth in bits/sec estimated for
+ // up-link.
+ // Return value
+ // -1 if error occurred in setting the bandwidth,
+ // 0 bandwidth is set successfully.
+ //
+ virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
+ const WebRtc_Word32 bw) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterTransportCallback()
+ // Register a transport callback which will be called to deliver
+ // the encoded buffers whenever Process() is called and a
+ // bit-stream is ready.
+ //
+ // Input:
+ // -transport : pointer to the callback class
+ // transport->SendData() is called whenever
+ // Process() is called and bit-stream is ready
+ // to deliver.
+ //
+ // Return value:
+ // -1 if the transport callback could not be registered
+ // 0 if registration is successful.
+ //
+ virtual WebRtc_Word32 RegisterTransportCallback(
+ AudioPacketizationCallback* transport) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Add10MsData()
+ // Add 10MS of raw (PCM) audio data to the encoder. If the sampling
+ // frequency of the audio does not match the sampling frequency of the
+ // current encoder ACM will resample the audio.
+ //
+ // Input:
+ // -audioFrame : the input audio frame, containing raw audio
+ // sampling frequency etc.,
+ // c.f. module_common_types.h for definition of
+ // AudioFrame.
+ //
+ // Return value:
+ // 0 successfully added the frame.
+ // -1 some error occurred and data is not added.
+ // < -1 to add the frame to the buffer n samples had to be
+ // overwritten, -n is the return value in this case.
+ //
+ virtual WebRtc_Word32 Add10MsData(const AudioFrame& audioFrame) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (FEC) Forward Error Correction
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetFECStatus(const bool enable)
+ // configure FEC status i.e. on/off.
+ //
+ // RFC 2198 describes a solution which has a single payload type which
+ // signifies a packet with redundancy. That packet then becomes a container,
+ // encapsulating multiple payloads into a single RTP packet.
+ // Such a scheme is flexible, since any amount of redundancy may be
+ // encapsulated within a single packet. There is, however, a small overhead
+ // since each encapsulated payload must be preceded by a header indicating
+ // the type of data enclosed.
+ //
+ // This means that FEC is actually a RED scheme.
+ //
+ // Input:
+ // -enableFEC : if true FEC is enabled, otherwise FEC is
+ // disabled.
+ //
+ // Return value:
+ // -1 if failed to set FEC status,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 SetFECStatus(const bool enableFEC) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool FECStatus()
+ // Get FEC status
+ //
+ // Return value
+ // true if FEC is enabled,
+ // false if FEC is disabled.
+ //
+ virtual bool FECStatus() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (VAD) Voice Activity Detection
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetVAD()
+ // If DTX is enabled & the codec does not have internal DTX/VAD
+ // WebRtc VAD will be automatically enabled and 'enableVAD' is ignored.
+ //
+ // If DTX is disabled but VAD is enabled no DTX packets are send,
+ // regardless of whether the codec has internal DTX/VAD or not. In this
+ // case, WebRtc VAD is running to label frames as active/in-active.
+ //
+ // Inputs:
+ // -enableDTX : if true DTX is enabled,
+ // otherwise DTX is disabled.
+ // -enableVAD : if true VAD is enabled,
+ // otherwise VAD is disabled.
+ // -vadMode : determines the aggressiveness of VAD. A more
+ // aggressive mode results in more frames labeled
+ // as in-active, c.f. definition of
+ // ACMVADMode in audio_coding_module_typedefs.h
+ // for valid values.
+ //
+ // Return value:
+ // -1 if failed to set up VAD/DTX,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 SetVAD(const bool enableDTX = true,
+ const bool enableVAD = false,
+ const ACMVADMode vadMode = VADNormal) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 VAD()
+ // Get VAD status.
+ //
+ // Outputs:
+ // -dtxEnabled : is set to true if DTX is enabled, otherwise
+ // is set to false.
+ // -vadEnabled : is set to true if VAD is enabled, otherwise
+ // is set to false.
+ // -vadMode : is set to the current aggressiveness of VAD.
+ //
+ // Return value:
+ // -1 if fails to retrieve the setting of DTX/VAD,
+ // 0 if succeeeded.
+ //
+ virtual WebRtc_Word32 VAD(bool& dtxEnabled, bool& vadEnabled,
+ ACMVADMode& vadMode) const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ReplaceInternalDTXWithWebRtc()
+ // Used to replace codec internal DTX scheme with WebRtc. This is only
+ // supported for G729, where this call replaces AnnexB with WebRtc DTX.
+ //
+ // Input:
+ // -useWebRtcDTX : if false (default) the codec built-in DTX/VAD
+ // scheme is used, otherwise the internal DTX is
+ // replaced with WebRtc DTX/VAD.
+ //
+ // Return value:
+ // -1 if failed to replace codec internal DTX with WebRtc,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
+ const bool useWebRtcDTX = false) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 IsInternalDTXReplacedWithWebRtc()
+ // Get status if the codec internal DTX (when such exists) is replaced with
+ // WebRtc DTX. This is only supported for G729.
+ //
+ // Output:
+ // -usesWebRtcDTX : is set to true if the codec internal DTX is
+ // replaced with WebRtc DTX/VAD, otherwise it is set
+ // to false.
+ //
+ // Return value:
+ // -1 if failed to determine if codec internal DTX is replaced with WebRtc,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
+ bool& usesWebRtcDTX) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterVADCallback()
+ // Call this method to register a callback function which is called
+ // any time that ACM encounters an empty frame. That is a frame which is
+ // recognized inactive. Depending on the codec WebRtc VAD or internal codec
+ // VAD is employed to identify a frame as active/inactive.
+ //
+ // Input:
+ // -vadCallback : pointer to a callback function.
+ //
+ // Return value:
+ // -1 if failed to register the callback function.
+ // 0 if the callback function is registered successfully.
+ //
+ virtual WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vadCallback) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Receiver
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 InitializeReceiver()
+ // Any decoder-related state of ACM will be initialized to the
+ // same state when ACM is created. This will not interrupt or
+ // effect encoding functionality of ACM. ACM would lose all the
+ // decoding-related settings by calling this function.
+ // For instance, all registered codecs are deleted and have to be
+ // registered again.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 InitializeReceiver() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ResetDecoder()
+ // This API resets the states of decoders. ACM will not lose any
+ // decoder-related settings, such as registered codecs.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word32 ResetDecoder() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ReceiveFrequency()
+ // Get sampling frequency of the last received payload.
+ //
+ // Return value:
+ // non-negative the sampling frequency in Hertz.
+ // -1 if an error has occurred.
+ //
+ virtual WebRtc_Word32 ReceiveFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 PlayoutFrequency()
+ // Get sampling frequency of audio played out.
+ //
+ // Return value:
+ // the sampling frequency in Hertz.
+ //
+ virtual WebRtc_Word32 PlayoutFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterReceiveCodec()
+ // Register possible decoders, can be called multiple times for
+ // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
+ //
+ // Input:
+ // -receiveCodec : parameters of the codec to be registered, c.f.
+ // common_types.h for the definition of
+ // CodecInst.
+ //
+ // Return value:
+ // -1 if failed to register the codec
+ // 0 if the codec registered successfully.
+ //
+ virtual WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receiveCodec) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 UnregisterReceiveCodec()
+ // Unregister the codec currently registered with a specific payload type
+ // from the list of possible receive codecs.
+ //
+ // Input:
+ // -payloadType : The number representing the payload type to
+ // unregister.
+ //
+ // Output:
+ // -1 if the unregistration fails.
+ // 0 if the given codec is successfully unregistered.
+ //
+ virtual WebRtc_Word32 UnregisterReceiveCodec(
+ const WebRtc_Word16 receiveCodec) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ReceiveCodec()
+ // Get the codec associated with last received payload.
+ //
+ // Output:
+ // -currRcvCodec : parameters of the codec associated with the last
+ // received payload, c.f. common_types.h for
+ // the definition of CodecInst.
+ //
+ // Return value:
+ // -1 if failed to retrieve the codec,
+ // 0 if the codec is successfully retrieved.
+ //
+ virtual WebRtc_Word32 ReceiveCodec(CodecInst& currRcvCodec) const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 IncomingPacket()
+ // Call this function to insert a parsed RTP packet into ACM.
+ //
+ // Inputs:
+ // -incomingPayload : received payload.
+ // -payloadLengthByte : the length of payload in bytes.
+ // -rtpInfo : the relevant information retrieved from RTP
+ // header.
+ //
+ // Return value:
+ // -1 if failed to push in the payload
+ // 0 if payload is successfully pushed in.
+ //
+ virtual WebRtc_Word32 IncomingPacket(const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLengthByte,
+ const WebRtcRTPHeader& rtpInfo) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 IncomingPayload()
+ // Call this API to push incoming payloads when there is no rtp-info.
+ // The rtp-info will be created in ACM. One usage for this API is when
+ // pre-encoded files are pushed in ACM
+ //
+ // Inputs:
+ // -incomingPayload : received payload.
+ // -payloadLenghtByte : the length, in bytes, of the received payload.
+ // -payloadType : the payload-type. This specifies which codec has
+ // to be used to decode the payload.
+ // -timestamp : send timestamp of the payload. ACM starts with
+ // a random value and increment it by the
+ // packet-size, which is given when the codec in
+ // question is registered by RegisterReceiveCodec().
+ // Therefore, it is essential to have the timestamp
+ // if the frame-size differ from the registered
+ // value or if the incoming payload contains DTX
+ // packets.
+ //
+ // Return value:
+ // -1 if failed to push in the payload
+ // 0 if payload is successfully pushed in.
+ //
+ virtual WebRtc_Word32 IncomingPayload(const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLengthByte,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timestamp = 0) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetMinimumPlayoutDelay()
+ // Set Minimum playout delay, used for lip-sync.
+ //
+ // Input:
+ // -timeMs : minimum delay in milliseconds.
+ //
+ // Return value:
+ // -1 if failed to set the delay,
+ // 0 if the minimum delay is set.
+ //
+ virtual WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 timeMs) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterIncomingMessagesCallback()
+ // Used by the module to deliver messages to the codec module/application
+ // when a Dtmf tone is detected, as well as when it stopped.
+ //
+ // Inputs:
+ // -inMsgCallback : pointer to callback function which will be called
+ // if Dtmf is detected.
+ // -cpt : enables CPT (Call Progress Tone) detection for the
+ // specified country. c.f. definition of ACMCountries
+ // in audio_coding_module_typedefs.h for valid
+ // entries. The default value disables CPT
+ // detection.
+ //
+ // Return value:
+ // -1 if the message callback could not be registered
+ // 0 if registration is successful.
+ //
+ virtual WebRtc_Word32
+ RegisterIncomingMessagesCallback(
+ AudioCodingFeedback* inMsgCallback,
+ const ACMCountries cpt = ACMDisableCountryDetection) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetDtmfPlayoutStatus()
+ // Configure Dtmf playout, i.e. whether out-of-band
+ // Dtmf tones are played or not.
+ //
+ // Input:
+ // -enable : if true to enable playout out-of-band Dtmf tones,
+ // false to disable.
+ //
+ // Return value:
+ // -1 if the method fails, e.g. Dtmf playout is not supported.
+ // 0 if the status is set successfully.
+ //
+ virtual WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool DtmfPlayoutStatus()
+ // Get Dtmf playout status.
+ //
+ // Return value:
+ // true if out-of-band Dtmf tones are played,
+ // false if playout of Dtmf tones is disabled.
+ //
+ virtual bool DtmfPlayoutStatus() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetBackgroundNoiseMode()
+ // Sets the mode of the background noise playout in an event of long
+ // packetloss burst. For the valid modes see the declaration of
+ // ACMBackgroundNoiseMode in audio_coding_module_typedefs.h.
+ //
+ // Input:
+ // -mode : the mode for the background noise playout.
+ //
+ // Return value:
+ // -1 if failed to set the mode.
+ // 0 if succeeded in setting the mode.
+ //
+ virtual WebRtc_Word32 SetBackgroundNoiseMode(
+ const ACMBackgroundNoiseMode mode) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 BackgroundNoiseMode()
+ // Call this method to get the mode of the background noise playout.
+ // Playout of background noise is a result of a long packetloss burst.
+ // See ACMBackgroundNoiseMode in audio_coding_module_typedefs.h for
+ // possible modes.
+ //
+ // Output:
+ // -mode : a reference to ACMBackgroundNoiseMode enumerator.
+ //
+ // Return value:
+ // 0 if the output is a valid mode.
+ // -1 if ACM failed to output a valid mode.
+ //
+ virtual WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 PlayoutTimestamp()
+ // The send timestamp of an RTP packet is associated with the decoded
+ // audio of the packet in question. This function returns the timestamp of
+ // the latest audio obtained by calling PlayoutData10ms().
+ //
+ // Input:
+ // -timestamp : a reference to a WebRtc_UWord32 to receive the
+ // timestamp.
+ // Return value:
+ // 0 if the output is a correct timestamp.
+ // -1 if failed to output the correct timestamp.
+ //
+ //
+ virtual WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 DecoderEstimatedBandwidth()
+ // Get the estimate of the Bandwidth, in bits/second, based on the incoming
+ // stream. This API is useful in one-way communication scenarios, where
+ // the bandwidth information is sent in an out-of-band fashion.
+ // Currently only supported if iSAC is registered as a reciever.
+ //
+ // Return value:
+ // >0 bandwidth in bits/second.
+ // -1 if failed to get a bandwidth estimate.
+ //
+ virtual WebRtc_Word32 DecoderEstimatedBandwidth() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetPlayoutMode()
+ // Call this API to set the playout mode. Playout mode could be optimized
+ // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
+ // optimized to deliver highest audio quality while maintaining a minimum
+ // delay. In FAX mode, NetEQ is optimized to have few delay changes as
+ // possible and maintain a constant delay, perhaps large relative to voice
+ // mode, to avoid PLC. In streaming mode, we tolerate a little more delay
+ // to acheive better jitter robustness.
+ //
+ // Input:
+ // -mode : playout mode. Possible inputs are:
+ // "voice",
+ // "fax" and
+ // "streaming".
+ //
+ // Return value:
+ // -1 if failed to set the mode,
+ // 0 if succeeding.
+ //
+ virtual WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // AudioPlayoutMode PlayoutMode()
+ // Get playout mode, i.e. whether it is speech, FAX or streaming. See
+ // audio_coding_module_typedefs.h for definition of AudioPlayoutMode.
+ //
+ // Return value:
+ // voice: is for voice output,
+ // fax: a mode that is optimized for receiving FAX signals.
+ // In this mode NetEq tries to maintain a constant high
+ // delay to avoid PLC if possible.
+ // streaming: a mode that is suitable for streaminq. In this mode we
+ // accept longer delay to improve jitter robustness.
+ //
+ virtual AudioPlayoutMode PlayoutMode() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 PlayoutData10Ms(
+ // Get 10 milliseconds of raw audio data for playout, at the given sampling
+ // frequency. ACM will perform a resampling if required.
+ //
+ // Input:
+ // -desiredFreqHz : the desired sampling frequency, in Hertz, of the
+ // output audio. If set to -1, the function returns the
+ // audio at the current sampling frequency.
+ //
+ // Output:
+ // -audioFrame : output audio frame which contains raw audio data
+ // and other relevant parameters, c.f.
+ // module_common_types.h for the definition of
+ // AudioFrame.
+ //
+ // Return value:
+ // -1 if the function fails,
+ // 0 if the function succeeds.
+ //
+ virtual WebRtc_Word32
+ PlayoutData10Ms(const WebRtc_Word32 desiredFreqHz,
+ AudioFrame &audioFrame) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (CNG) Comfort Noise Generation
+ // Generate comfort noise when receiving DTX packets
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 SetReceiveVADMode()
+ // Configure VAD aggressiveness on the incoming stream.
+ //
+ // Input:
+ // -mode : aggressiveness of the VAD on incoming stream.
+ // See audio_coding_module_typedefs.h for the
+ // definition of ACMVADMode, and possible
+ // values for aggressiveness.
+ //
+ // Return value:
+ // -1 if fails to set the mode,
+ // 0 if the mode is set successfully.
+ //
+ virtual WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // ACMVADMode ReceiveVADMode()
+ // Get VAD aggressiveness on the incoming stream.
+ //
+ // Return value:
+ // aggressiveness of VAD, running on the incoming stream. A more
+ // aggressive mode means more audio frames will be labeled as in-active.
+ // See audio_coding_module_typedefs.h for the definition of
+ // ACMVADMode.
+ //
+ virtual ACMVADMode ReceiveVADMode() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Codec specific
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetISACMaxRate()
+ // Set the maximum instantaneous rate of iSAC. For a payload of B bits
+ // with a frame-size of T sec the instantaneous rate is B/T bist per
+ // second. Therefore, (B/T < maxRateBitPerSec) and
+ // (B < maxPayloadLenBytes * 8) are always satisfied for iSAC payloads,
+ // c.f SetISACMaxPayloadSize().
+ //
+ // Input:
+ // -maxRateBitPerSec : maximum instantaneous bit-rate given in bits/sec.
+ //
+ // Return value:
+ // -1 if failed to set the maximum rate.
+ // 0 if the maximum rate is set successfully.
+ //
+ virtual WebRtc_Word32 SetISACMaxRate(
+ const WebRtc_UWord32 maxRateBitPerSec) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetISACMaxPayloadSize()
+ // Set the maximum payload size of iSAC packets. No iSAC payload,
+ // regardless of its frame-size, may exceed the given limit. For
+ // an iSAC payload of size B bits and frame-size T sec we have;
+ // (B < maxPayloadLenBytes * 8) and (B/T < maxRateBitPerSec), c.f.
+ // SetISACMaxRate().
+ //
+ // Input:
+ // -maxPayloadLenBytes : maximum payload size in bytes.
+ //
+ // Return value:
+ // -1 if failed to set the maximm payload-size.
+ // 0 if the given linit is seet successfully.
+ //
+ virtual WebRtc_Word32 SetISACMaxPayloadSize(
+ const WebRtc_UWord16 maxPayloadLenBytes) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ConfigISACBandwidthEstimator()
+ // Call this function to configure the bandwidth estimator of ISAC.
+ // During the adaptation of bit-rate, iSAC atomatically adjusts the
+ // frame-size (either 30 or 60 ms) to save on RTP header. The initial
+ // frame-size can be specified by the first argument. The configuration also
+ // regards the initial estimate of bandwidths. The estimator starts from
+ // this point and converges to the actual bottleneck. This is given by the
+ // second parameter. Furthermore, it is also possible to control the
+ // adaptation of frame-size. This is specified by the last parameter.
+ //
+ // Input:
+ // -initFrameSizeMsec : initial frame-size in milisecods. For iSAC-wb
+ // 30 ms and 60 ms (default) are acceptable values,
+ // and for iSAC-swb 30 ms is the only acceptable
+ // value. Zero indiates default value.
+ // -initRateBitPerSec : initial estimate of the bandwidth. Values
+ // between 10000 and 58000 are acceptable.
+ // -enforceFrameSize : if true, the frame-size will not be adapted.
+ //
+ // Return value:
+ // -1 if failed to configure the bandwidth estimator,
+ // 0 if the configuration was successfully applied.
+ //
+ virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize = false) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // statistics
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 NetworkStatistics()
+ // Get network statistics.
+ //
+ // Input:
+ // -networkStatistics : a structure that contains network statistics.
+ //
+ // Return value:
+ // -1 if failed to set the network statistics,
+ // 0 if statistics are set successfully.
+ //
+ virtual WebRtc_Word32 NetworkStatistics(
+ ACMNetworkStatistics& networkStatistics) const = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
diff --git a/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h b/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
new file mode 100644
index 0000000..c0e06ef
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
@@ -0,0 +1,197 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+///////////////////////////////////////////////////////////////////////////
+// enum AudioPlayoutMode
+// An enumerator for different playout modes.
+//
+// -voice : This is the standard mode for VoIP calls. The trade-off
+// between low delay and jitter robustness is optimized
+// for high-quality two-way communication.
+// NetEQs packet loss concealment and signal processing
+// capabilities are fully employed.
+// -fax : The fax mode is optimized for decodability of fax signals
+// rather than for perceived audio quality. When this mode
+// is selected, NetEQ will do as few delay changes as possible,
+// trying to maintain a high and constant delay. Meanwhile,
+// the packet loss concealment efforts are reduced.
+//
+// -streaming : In the case of one-way communication such as passive
+// conference participant, a webinar, or a streaming application,
+// this mode can be used to improve the jitter robustness at
+// the cost of increased delay.
+//
+enum AudioPlayoutMode {
+ voice = 0,
+ fax = 1,
+ streaming = 2
+};
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMSpeechType
+// An enumerator for possible labels of a decoded frame.
+//
+// -normal : a normal speech frame. If VAD is enabled on the
+// incoming stream this label indicate that the
+// frame is active.
+// -PLC : a PLC frame. The corresponding packet was lost
+// and this frame generated by PLC techniques.
+// -CNG : the frame is comfort noise. This happens if VAD
+// is enabled at the sender and we have received
+// SID.
+// -PLCCNG : PLC will fade to comfort noise if the duration
+// of PLC is long. This labels such a case.
+// -VADPassive : the VAD at the receiver recognizes this frame as
+// passive.
+//
+enum ACMSpeechType {
+ normal = 0,
+ PLC = 1,
+ CNG = 2,
+ PLCCNG = 3,
+ VADPassive = 4
+};
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMVADMode
+// An enumerator for aggressiveness of VAD
+// -VADNormal : least aggressive mode.
+// -VADLowBitrate : more aggressive than "VADNormal" to save on
+// bit-rate.
+// -VADAggr : an aggressive mode.
+// -VADVeryAggr : the most agressive mode.
+//
+enum ACMVADMode {
+ VADNormal = 0,
+ VADLowBitrate = 1,
+ VADAggr = 2,
+ VADVeryAggr = 3
+};
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMCountries
+// An enumerator for countries, used when enabling CPT for a specific country.
+//
+enum ACMCountries {
+ ACMDisableCountryDetection = -1, // disable CPT detection
+ ACMUSA = 0,
+ ACMJapan,
+ ACMCanada,
+ ACMFrance,
+ ACMGermany,
+ ACMAustria,
+ ACMBelgium,
+ ACMUK,
+ ACMCzech,
+ ACMDenmark,
+ ACMFinland,
+ ACMGreece,
+ ACMHungary,
+ ACMIceland,
+ ACMIreland,
+ ACMItaly,
+ ACMLuxembourg,
+ ACMMexico,
+ ACMNorway,
+ ACMPoland,
+ ACMPortugal,
+ ACMSpain,
+ ACMSweden,
+ ACMTurkey,
+ ACMChina,
+ ACMHongkong,
+ ACMTaiwan,
+ ACMKorea,
+ ACMSingapore,
+ ACMNonStandard1
+// non-standard countries
+};
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMAMRPackingFormat
+// An enumerator for different bit-packing format of AMR codec according to
+// RFC 3267.
+//
+// -AMRUndefined : undefined.
+// -AMRBandwidthEfficient : bandwidth-efficient mode.
+// -AMROctetAlligned : Octet-alligned mode.
+// -AMRFileStorage : file-storage mode.
+//
+enum ACMAMRPackingFormat {
+ AMRUndefined = -1,
+ AMRBandwidthEfficient = 0,
+ AMROctetAlligned = 1,
+ AMRFileStorage = 2
+};
+
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Struct containing network statistics
+//
+// -currentBufferSize : current jitter buffer size in ms
+// -preferredBufferSize : preferred (optimal) buffer size in ms
+// -jitterPeaksFound : indicate if peaky-jitter mode is engaged, that is,
+// if severe but sparse network delays have occurred.
+// -currentPacketLossRate : loss rate (network + late) (in Q14)
+// -currentDiscardRate : late loss rate (in Q14)
+// -currentExpandRate : fraction (of original stream) of synthesized
+// speech inserted through expansion (in Q14)
+// -currentPreemptiveRate : fraction of synthesized speech inserted through
+// pre-emptive expansion (in Q14)
+// -currentAccelerateRate : fraction of data removed through acceleration
+// (in Q14)
+// -clockDriftPPM : clock-drift between sender and receiver in parts-
+// per-million. Positive means that receiver sample
+// rate is higher than sender sample rate.
+// -meanWaitingTimeMs : average packet waiting time in the buffer
+// -medianWaitingTimeMs : median packet waiting time in the buffer
+// -minWaitingTimeMs : min packet waiting time in the buffer
+// -maxWaitingTimeMs : max packet waiting time in the buffer
+typedef struct {
+ WebRtc_UWord16 currentBufferSize;
+ WebRtc_UWord16 preferredBufferSize;
+ bool jitterPeaksFound;
+ WebRtc_UWord16 currentPacketLossRate;
+ WebRtc_UWord16 currentDiscardRate;
+ WebRtc_UWord16 currentExpandRate;
+ WebRtc_UWord16 currentPreemptiveRate;
+ WebRtc_UWord16 currentAccelerateRate;
+ int32_t clockDriftPPM;
+ int meanWaitingTimeMs;
+ int medianWaitingTimeMs;
+ int minWaitingTimeMs;
+ int maxWaitingTimeMs;
+} ACMNetworkStatistics;
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Enumeration of background noise mode a mapping from NetEQ interface.
+//
+// -On : default "normal" behavior with eternal noise
+// -Fade : noise fades to zero after some time
+// -Off : background noise is always zero
+//
+enum ACMBackgroundNoiseMode {
+ On,
+ Fade,
+ Off
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/Android.mk b/trunk/src/modules/audio_coding/main/source/Android.mk
new file mode 100644
index 0000000..90214a9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/Android.mk
@@ -0,0 +1,67 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_audio_coding
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ acm_cng.cc \
+ acm_codec_database.cc \
+ acm_dtmf_detection.cc \
+ acm_dtmf_playout.cc \
+ acm_g722.cc \
+ acm_generic_codec.cc \
+ acm_ilbc.cc \
+ acm_isac.cc \
+ acm_neteq.cc \
+ acm_pcm16b.cc \
+ acm_pcma.cc \
+ acm_pcmu.cc \
+ acm_red.cc \
+ acm_resampler.cc \
+ audio_coding_module.cc \
+ audio_coding_module_impl.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../codecs/cng/include \
+ $(LOCAL_PATH)/../../codecs/g711/include \
+ $(LOCAL_PATH)/../../codecs/g722/include \
+ $(LOCAL_PATH)/../../codecs/ilbc/interface \
+ $(LOCAL_PATH)/../../codecs/iSAC/main/interface \
+ $(LOCAL_PATH)/../../codecs/iSAC/fix/interface \
+ $(LOCAL_PATH)/../../codecs/pcm16b/include \
+ $(LOCAL_PATH)/../../neteq/interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../../common_audio/resampler/include \
+ $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../../common_audio/vad/include \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amr.cc b/trunk/src/modules/audio_coding/main/source/acm_amr.cc
new file mode 100644
index 0000000..71cf603
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_amr.cc
@@ -0,0 +1,444 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_amr.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "audio_coding_module_typedefs.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_AMR
+// NOTE! GSM AMR is not included in the open-source package. The following
+// interface file is needed:
+//
+// /modules/audio_coding/codecs/amr/main/interface/amr_interface.h
+//
+// The API in the header file should match the one below.
+//
+// int16_t WebRtcAmr_CreateEnc(AMR_encinst_t_** encInst);
+// int16_t WebRtcAmr_CreateDec(AMR_decinst_t_** decInst);
+// int16_t WebRtcAmr_FreeEnc(AMR_encinst_t_* encInst);
+// int16_t WebRtcAmr_FreeDec(AMR_decinst_t_* decInst);
+// int16_t WebRtcAmr_Encode(AMR_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t*output,
+// int16_t mode);
+// int16_t WebRtcAmr_EncoderInit(AMR_encinst_t_* encInst,
+// int16_t dtxMode);
+// int16_t WebRtcAmr_EncodeBitmode(AMR_encinst_t_* encInst,
+// int format);
+// int16_t WebRtcAmr_Decode(AMR_decinst_t_* decInst);
+// int16_t WebRtcAmr_DecodePlc(AMR_decinst_t_* decInst);
+// int16_t WebRtcAmr_DecoderInit(AMR_decinst_t_* decInst);
+// int16_t WebRtcAmr_DecodeBitmode(AMR_decinst_t_* decInst,
+// int format);
+// void WebRtcAmr_Version(char *versionStr, short len);
+#include "amr_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_AMR
+ACMAMR::ACMAMR(WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _encodingMode(-1), // Invalid value.
+ _encodingRate(0), // Invalid value.
+ _encoderPackingFormat(AMRBandwidthEfficient),
+ _decoderPackingFormat(AMRBandwidthEfficient) {
+ return;
+}
+
+ACMAMR::~ACMAMR() {
+ return;
+}
+
+WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMR::EnableDTX() {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMR::DisableDTX() {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMR::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMR::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMAMR::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMAMR::DestructEncoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMAMR::DestructDecoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+ return -1;
+}
+
+void ACMAMR::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
+ ACMAMRPackingFormat /* packingFormat */) {
+ return -1;
+}
+
+ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const {
+ return AMRUndefined;
+}
+
+WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
+ ACMAMRPackingFormat /* packingFormat */) {
+ return -1;
+}
+
+ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
+ return AMRUndefined;
+}
+
+WebRtc_Word16 ACMAMR::UnregisterFromNetEqSafe(ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+
+#else //===================== Actual Implementation =======================
+
+#define WEBRTC_AMR_MR475 0
+#define WEBRTC_AMR_MR515 1
+#define WEBRTC_AMR_MR59 2
+#define WEBRTC_AMR_MR67 3
+#define WEBRTC_AMR_MR74 4
+#define WEBRTC_AMR_MR795 5
+#define WEBRTC_AMR_MR102 6
+#define WEBRTC_AMR_MR122 7
+
+ACMAMR::ACMAMR(WebRtc_Word16 codecID)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _encodingMode(-1), // invalid value
+ _encodingRate(0) { // invalid value
+ _codecID = codecID;
+ _hasInternalDTX = true;
+ _encoderPackingFormat = AMRBandwidthEfficient;
+ _decoderPackingFormat = AMRBandwidthEfficient;
+ return;
+}
+
+ACMAMR::~ACMAMR() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcAmr_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ WebRtcAmr_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) {
+ WebRtc_Word16 vadDecision = 1;
+ // sanity check, if the rate is set correctly. we might skip this
+ // sanity check. if rate is not set correctly, initialization flag
+ // should be false and should not be here.
+ if ((_encodingMode < WEBRTC_AMR_MR475) ||
+ (_encodingMode > WEBRTC_AMR_MR122)) {
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+ *bitStreamLenByte = WebRtcAmr_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead],
+ _frameLenSmpl,
+ (WebRtc_Word16*) bitStream,
+ _encodingMode);
+
+ // Update VAD, if internal DTX is used
+ if (_hasInternalDTX && _dtxEnabled) {
+ if (*bitStreamLenByte <= (7 * _frameLenSmpl / 160)) {
+ vadDecision = 0;
+ }
+ for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+ _vadLabel[n] = vadDecision;
+ }
+ }
+ // increment the read index
+ _inAudioIxRead += _frameLenSmpl;
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMAMR::EnableDTX() {
+ if (_dtxEnabled) {
+ return 0;
+ } else if (_encoderExist) { // check if encoder exist
+ // enable DTX
+ if (WebRtcAmr_EncoderInit(_encoderInstPtr, 1) < 0) {
+ return -1;
+ }
+ _dtxEnabled = true;
+ return 0;
+ } else {
+ return -1;
+ }
+}
+
+WebRtc_Word16 ACMAMR::DisableDTX() {
+ if (!_dtxEnabled) {
+ return 0;
+ } else if (_encoderExist) { // check if encoder exist
+ // disable DTX
+ if (WebRtcAmr_EncoderInit(_encoderInstPtr, 0) < 0) {
+ return -1;
+ }
+ _dtxEnabled = false;
+ return 0;
+ } else {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+WebRtc_Word16 ACMAMR::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+ WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
+ status += (WebRtcAmr_EncoderInit(
+ _encoderInstPtr, ((codecParams->enableDTX) ? 1 : 0)) < 0) ? -1 : 0;
+ status += (WebRtcAmr_EncodeBitmode(
+ _encoderInstPtr, _encoderPackingFormat) < 0) ? -1 : 0;
+ return (status < 0) ? -1 : 0;
+}
+
+WebRtc_Word16 ACMAMR::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ WebRtc_Word16 status =
+ ((WebRtcAmr_DecoderInit(_decoderInstPtr) < 0) ? -1 : 0);
+ status += WebRtcAmr_DecodeBitmode(_decoderInstPtr, _decoderPackingFormat);
+ return (status < 0) ? -1 : 0;
+}
+
+WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ // Todo:
+ // log error
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_AMR_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderAMR, codecInst.pltype, _decoderInstPtr,
+ 8000);
+ SET_AMR_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec* ACMAMR::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
+ return WebRtcAmr_CreateEnc(&_encoderInstPtr);
+}
+
+void ACMAMR::DestructEncoderSafe() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcAmr_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ // there is no encoder set the following
+ _encoderExist = false;
+ _encoderInitialized = false;
+ _encodingMode = -1; // invalid value
+ _encodingRate = 0; // invalid value
+}
+
+WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
+ return WebRtcAmr_CreateDec(&_decoderInstPtr);
+}
+
+void ACMAMR::DestructDecoderSafe() {
+ if (_decoderInstPtr != NULL) {
+ WebRtcAmr_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ // there is no encoder instance set the followings
+ _decoderExist = false;
+ _decoderInitialized = false;
+}
+
+WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 rate) {
+ switch (rate) {
+ case 4750: {
+ _encodingMode = WEBRTC_AMR_MR475;
+ _encodingRate = 4750;
+ break;
+ }
+ case 5150: {
+ _encodingMode = WEBRTC_AMR_MR515;
+ _encodingRate = 5150;
+ break;
+ }
+ case 5900: {
+ _encodingMode = WEBRTC_AMR_MR59;
+ _encodingRate = 5900;
+ break;
+ }
+ case 6700: {
+ _encodingMode = WEBRTC_AMR_MR67;
+ _encodingRate = 6700;
+ break;
+ }
+ case 7400: {
+ _encodingMode = WEBRTC_AMR_MR74;
+ _encodingRate = 7400;
+ break;
+ }
+ case 7950: {
+ _encodingMode = WEBRTC_AMR_MR795;
+ _encodingRate = 7950;
+ break;
+ }
+ case 10200: {
+ _encodingMode = WEBRTC_AMR_MR102;
+ _encodingRate = 10200;
+ break;
+ }
+ case 12200: {
+ _encodingMode = WEBRTC_AMR_MR122;
+ _encodingRate = 12200;
+ break;
+ }
+ default: {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMAMR::InternalDestructEncoderInst(void* ptrInst) {
+ // Free the memory where ptrInst is pointing to
+ if (ptrInst != NULL) {
+ WebRtcAmr_FreeEnc(reinterpret_cast<AMR_encinst_t_*>(ptrInst));
+ }
+ return;
+}
+
+WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
+ ACMAMRPackingFormat packingFormat) {
+ if ((packingFormat != AMRBandwidthEfficient) &&
+ (packingFormat != AMROctetAlligned) &&
+ (packingFormat != AMRFileStorage)) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Invalid AMR Encoder packing-format.");
+ return -1;
+ } else {
+ if (WebRtcAmr_EncodeBitmode(_encoderInstPtr, packingFormat) < 0) {
+ return -1;
+ } else {
+ _encoderPackingFormat = packingFormat;
+ return 0;
+ }
+ }
+}
+
+ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const {
+ return _encoderPackingFormat;
+}
+
+WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
+ ACMAMRPackingFormat packingFormat) {
+ if ((packingFormat != AMRBandwidthEfficient) &&
+ (packingFormat != AMROctetAlligned) &&
+ (packingFormat != AMRFileStorage)) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Invalid AMR decoder packing-format.");
+ return -1;
+ } else {
+ if (WebRtcAmr_DecodeBitmode(_decoderInstPtr, packingFormat) < 0) {
+ return -1;
+ } else {
+ _decoderPackingFormat = packingFormat;
+ return 0;
+ }
+ }
+}
+
+ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
+ return _decoderPackingFormat;
+}
+
+WebRtc_Word16 ACMAMR::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not "
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderAMR);
+}
+
+#endif
+
+}
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amr.h b/trunk/src/modules/audio_coding/main/source/acm_amr.h
new file mode 100644
index 0000000..cc342e7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_amr.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct AMR_encinst_t_;
+struct AMR_decinst_t_;
+
+namespace webrtc {
+
+enum ACMAMRPackingFormat;
+
+class ACMAMR: public ACMGenericCodec {
+ public:
+ ACMAMR(WebRtc_Word16 codecID);
+ ~ACMAMR();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 SetAMREncoderPackingFormat(
+ const ACMAMRPackingFormat packingFormat);
+
+ ACMAMRPackingFormat AMREncoderPackingFormat() const;
+
+ WebRtc_Word16 SetAMRDecoderPackingFormat(
+ const ACMAMRPackingFormat packingFormat);
+
+ ACMAMRPackingFormat AMRDecoderPackingFormat() const;
+
+ protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+
+ WebRtc_Word16 EnableDTX();
+
+ WebRtc_Word16 DisableDTX();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ AMR_encinst_t_* _encoderInstPtr;
+ AMR_decinst_t_* _decoderInstPtr;
+ WebRtc_Word16 _encodingMode;
+ WebRtc_Word16 _encodingRate;
+ ACMAMRPackingFormat _encoderPackingFormat;
+ ACMAMRPackingFormat _decoderPackingFormat;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc b/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc
new file mode 100644
index 0000000..fed2e3f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc
@@ -0,0 +1,450 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_amrwb.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "audio_coding_module_typedefs.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_AMRWB
+// NOTE! GSM AMR-wb is not included in the open-source package. The
+// following interface file is needed:
+//
+// /modules/audio_coding/codecs/amrwb/main/interface/amrwb_interface.h
+//
+// The API in the header file should match the one below.
+//
+// int16_t WebRtcAmrWb_CreateEnc(AMRWB_encinst_t_** encInst);
+// int16_t WebRtcAmrWb_CreateDec(AMRWB_decinst_t_** decInst);
+// int16_t WebRtcAmrWb_FreeEnc(AMRWB_encinst_t_* encInst);
+// int16_t WebRtcAmrWb_FreeDec(AMRWB_decinst_t_* decInst);
+// int16_t WebRtcAmrWb_Encode(AMRWB_encinst_t_* encInst, int16_t* input,
+// int16_t len, int16_t* output, int16_t mode);
+// int16_t WebRtcAmrWb_EncoderInit(AMRWB_encinst_t_* encInst,
+// int16_t dtxMode);
+// int16_t WebRtcAmrWb_EncodeBitmode(AMRWB_encinst_t_* encInst,
+// int format);
+// int16_t WebRtcAmrWb_Decode(AMRWB_decinst_t_* decInst);
+// int16_t WebRtcAmrWb_DecodePlc(AMRWB_decinst_t_* decInst);
+// int16_t WebRtcAmrWb_DecoderInit(AMRWB_decinst_t_* decInst);
+// int16_t WebRtcAmrWb_DecodeBitmode(AMRWB_decinst_t_* decInst,
+// int format);
+// void WebRtcAmrWb_Version(char *versionStr, short len);
+#include "amrwb_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_AMRWB
+ACMAMRwb::ACMAMRwb(WebRtc_Word16 /* codecID*/)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _encodingMode(-1), // invalid value
+ _encodingRate(0), // invalid value
+ _encoderPackingFormat(AMRBandwidthEfficient),
+ _decoderPackingFormat(AMRBandwidthEfficient) {
+ return;
+}
+
+ACMAMRwb::~ACMAMRwb() {
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalEncode(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMRwb::EnableDTX() {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMRwb::DisableDTX() {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec*
+ACMAMRwb::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMAMRwb::DestructEncoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMAMRwb::DestructDecoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
+ return -1;
+}
+
+void ACMAMRwb::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
+ ACMAMRPackingFormat /* packingFormat */) {
+ return -1;
+}
+
+ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const {
+ return AMRUndefined;
+}
+
+WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
+ ACMAMRPackingFormat /* packingFormat */) {
+ return -1;
+}
+
+ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
+ return AMRUndefined;
+}
+
+WebRtc_Word16 ACMAMRwb::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+#define AMRWB_MODE_7k 0
+#define AMRWB_MODE_9k 1
+#define AMRWB_MODE_12k 2
+#define AMRWB_MODE_14k 3
+#define AMRWB_MODE_16k 4
+#define AMRWB_MODE_18k 5
+#define AMRWB_MODE_20k 6
+#define AMRWB_MODE_23k 7
+#define AMRWB_MODE_24k 8
+
+ACMAMRwb::ACMAMRwb(WebRtc_Word16 codecID)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _encodingMode(-1), // invalid value
+ _encodingRate(0) { // invalid value
+ _codecID = codecID;
+ _hasInternalDTX = true;
+ _encoderPackingFormat = AMRBandwidthEfficient;
+ _decoderPackingFormat = AMRBandwidthEfficient;
+ return;
+}
+
+ACMAMRwb::~ACMAMRwb() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcAmrWb_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ WebRtcAmrWb_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalEncode(WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) {
+ WebRtc_Word16 vadDecision = 1;
+ // sanity check, if the rate is set correctly. we might skip this
+ // sanity check. if rate is not set correctly, initialization flag
+ // should be false and should not be here.
+ if ((_encodingMode < AMRWB_MODE_7k) || (_encodingMode > AMRWB_MODE_24k)) {
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+ *bitStreamLenByte = WebRtcAmrWb_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead],
+ _frameLenSmpl,
+ (WebRtc_Word16*) bitStream,
+ _encodingMode);
+
+ // Update VAD, if internal DTX is used
+ if (_hasInternalDTX && _dtxEnabled) {
+ if (*bitStreamLenByte <= (7 * _frameLenSmpl / 160)) {
+ vadDecision = 0;
+ }
+ for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
+ _vadLabel[n] = vadDecision;
+ }
+ }
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl;
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMAMRwb::EnableDTX() {
+ if (_dtxEnabled) {
+ return 0;
+ } else if (_encoderExist) { // check if encoder exist
+ // enable DTX
+ if (WebRtcAmrWb_EncoderInit(_encoderInstPtr, 1) < 0) {
+ return -1;
+ }
+ _dtxEnabled = true;
+ return 0;
+ } else {
+ return -1;
+ }
+}
+
+WebRtc_Word16 ACMAMRwb::DisableDTX() {
+ if (!_dtxEnabled) {
+ return 0;
+ } else if (_encoderExist) { // check if encoder exist
+ // disable DTX
+ if (WebRtcAmrWb_EncoderInit(_encoderInstPtr, 0) < 0) {
+ return -1;
+ }
+ _dtxEnabled = false;
+ return 0;
+ } else {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+WebRtc_Word16 ACMAMRwb::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+ // sanity check
+ if (_encoderInstPtr == NULL) {
+ return -1;
+ }
+
+ WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
+ status += (WebRtcAmrWb_EncoderInit(
+ _encoderInstPtr, ((codecParams->enableDTX) ? 1 : 0)) < 0) ? -1 : 0;
+ status += (WebRtcAmrWb_EncodeBitmode(
+ _encoderInstPtr, _encoderPackingFormat) < 0) ? -1 : 0;
+ return (status < 0) ? -1 : 0;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ WebRtc_Word16 status = WebRtcAmrWb_DecodeBitmode(_decoderInstPtr,
+ _decoderPackingFormat);
+ status += ((WebRtcAmrWb_DecoderInit(_decoderInstPtr) < 0) ? -1 : 0);
+ return (status < 0) ? -1 : 0;
+}
+
+WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_AMRWB_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderAMRWB, codecInst.pltype, _decoderInstPtr,
+ 16000);
+ SET_AMRWB_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec* ACMAMRwb::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
+ return WebRtcAmrWb_CreateEnc(&_encoderInstPtr);
+}
+
+void ACMAMRwb::DestructEncoderSafe() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcAmrWb_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ // there is no encoder set the following
+ _encoderExist = false;
+ _encoderInitialized = false;
+ _encodingMode = -1; // invalid value
+ _encodingRate = 0;
+}
+
+WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
+ return WebRtcAmrWb_CreateDec(&_decoderInstPtr);
+}
+
+void ACMAMRwb::DestructDecoderSafe() {
+ if (_decoderInstPtr != NULL) {
+ WebRtcAmrWb_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ // there is no encoder instance set the followings
+ _decoderExist = false;
+ _decoderInitialized = false;
+}
+
+WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 rate) {
+ switch (rate) {
+ case 7000: {
+ _encodingMode = AMRWB_MODE_7k;
+ _encodingRate = 7000;
+ break;
+ }
+ case 9000: {
+ _encodingMode = AMRWB_MODE_9k;
+ _encodingRate = 9000;
+ break;
+ }
+ case 12000: {
+ _encodingMode = AMRWB_MODE_12k;
+ _encodingRate = 12000;
+ break;
+ }
+ case 14000: {
+ _encodingMode = AMRWB_MODE_14k;
+ _encodingRate = 14000;
+ break;
+ }
+ case 16000: {
+ _encodingMode = AMRWB_MODE_16k;
+ _encodingRate = 16000;
+ break;
+ }
+ case 18000: {
+ _encodingMode = AMRWB_MODE_18k;
+ _encodingRate = 18000;
+ break;
+ }
+ case 20000: {
+ _encodingMode = AMRWB_MODE_20k;
+ _encodingRate = 20000;
+ break;
+ }
+ case 23000: {
+ _encodingMode = AMRWB_MODE_23k;
+ _encodingRate = 23000;
+ break;
+ }
+ case 24000: {
+ _encodingMode = AMRWB_MODE_24k;
+ _encodingRate = 24000;
+ break;
+ }
+ default: {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMAMRwb::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ WebRtcAmrWb_FreeEnc(static_cast<AMRWB_encinst_t_*>(ptrInst));
+ }
+ return;
+}
+
+WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
+ ACMAMRPackingFormat packingFormat) {
+ if ((packingFormat != AMRBandwidthEfficient) &&
+ (packingFormat != AMROctetAlligned) &&
+ (packingFormat != AMRFileStorage)) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Invalid AMRwb encoder packing-format.");
+ return -1;
+ } else {
+ if (WebRtcAmrWb_EncodeBitmode(_encoderInstPtr, packingFormat) < 0) {
+ return -1;
+ } else {
+ _encoderPackingFormat = packingFormat;
+ return 0;
+ }
+ }
+}
+
+ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const {
+ return _encoderPackingFormat;
+}
+
+WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
+ ACMAMRPackingFormat packingFormat) {
+ if ((packingFormat != AMRBandwidthEfficient) &&
+ (packingFormat != AMROctetAlligned) &&
+ (packingFormat != AMRFileStorage)) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Invalid AMRwb decoder packing-format.");
+ return -1;
+ } else {
+ if (WebRtcAmrWb_DecodeBitmode(_decoderInstPtr, packingFormat) < 0) {
+ return -1;
+ } else {
+ _decoderPackingFormat = packingFormat;
+ return 0;
+ }
+ }
+}
+
+ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
+ return _decoderPackingFormat;
+}
+
+WebRtc_Word16 ACMAMRwb::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not"
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderAMRWB);
+}
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amrwb.h b/trunk/src/modules/audio_coding/main/source/acm_amrwb.h
new file mode 100644
index 0000000..775f07f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_amrwb.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct AMRWB_encinst_t_;
+struct AMRWB_decinst_t_;
+
+namespace webrtc {
+
+enum ACMAMRPackingFormat;
+
+class ACMAMRwb: public ACMGenericCodec {
+ public:
+ ACMAMRwb(WebRtc_Word16 codecID);
+ ~ACMAMRwb();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams* codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams* codecParams);
+
+ WebRtc_Word16 SetAMRwbEncoderPackingFormat(
+ const ACMAMRPackingFormat packingFormat);
+
+ ACMAMRPackingFormat AMRwbEncoderPackingFormat() const;
+
+ WebRtc_Word16 SetAMRwbDecoderPackingFormat(
+ const ACMAMRPackingFormat packingFormat);
+
+ ACMAMRPackingFormat AMRwbDecoderPackingFormat() const;
+
+ protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
+
+ WebRtc_Word16 EnableDTX();
+
+ WebRtc_Word16 DisableDTX();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ AMRWB_encinst_t_* _encoderInstPtr;
+ AMRWB_decinst_t_* _decoderInstPtr;
+
+ WebRtc_Word16 _encodingMode;
+ WebRtc_Word16 _encodingRate;
+ ACMAMRPackingFormat _encoderPackingFormat;
+ ACMAMRPackingFormat _decoderPackingFormat;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_celt.cc b/trunk/src/modules/audio_coding/main/source/acm_celt.cc
new file mode 100644
index 0000000..5b12f32
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_celt.cc
@@ -0,0 +1,317 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_celt.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+// TODO(tlegrand): Add full paths.
+
+#ifdef WEBRTC_CODEC_CELT
+// NOTE! Celt is not included in the open-source package. Modify this file or
+// your codec API to match the function call and name of used Celt API file.
+#include "celt_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_CELT
+
+ACMCELT::ACMCELT(int16_t /* codecID */)
+ : enc_inst_ptr_(NULL),
+ dec_inst_ptr_(NULL),
+ sampling_freq_(0),
+ bitrate_(0),
+ channels_(1),
+ dec_channels_(1) {
+ return;
+}
+
+ACMCELT::~ACMCELT() {
+ return;
+}
+
+int16_t ACMCELT::InternalEncode(uint8_t* /* bitStream */,
+ int16_t* /* bitStreamLenByte */) {
+ return -1;
+}
+
+int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
+ int16_t /* bitStreamLenByte */,
+ int16_t* /* audio */,
+ int16_t* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMCELT::CreateInstance(void) {
+ return NULL;
+}
+
+int16_t ACMCELT::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMCELT::DestructEncoderSafe() {
+ return;
+}
+
+int16_t ACMCELT::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMCELT::DestructDecoderSafe() {
+ return;
+}
+
+void ACMCELT::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* /* netEq */,
+ int16_t /* payloadType */) {
+ return -1;
+}
+
+bool ACMCELT::IsTrueStereoCodec() {
+ return true;
+}
+
+int16_t ACMCELT::SetBitRateSafe(const int32_t /*rate*/) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+ACMCELT::ACMCELT(int16_t codecID)
+ : enc_inst_ptr_(NULL),
+ dec_inst_ptr_(NULL),
+ sampling_freq_(32000), // Default sampling frequency.
+ bitrate_(64000), // Default rate.
+ channels_(1), // Default send mono.
+ dec_channels_(1) { // Default receive mono.
+ // TODO(tlegrand): remove later when ACMGenericCodec has a new constructor.
+ _codecID = codecID;
+
+ return;
+}
+
+ACMCELT::~ACMCELT() {
+ if (enc_inst_ptr_ != NULL) {
+ WebRtcCelt_FreeEnc(enc_inst_ptr_);
+ enc_inst_ptr_ = NULL;
+ }
+ if (dec_inst_ptr_ != NULL) {
+ WebRtcCelt_FreeDec(dec_inst_ptr_);
+ dec_inst_ptr_ = NULL;
+ }
+ return;
+}
+
+int16_t ACMCELT::InternalEncode(uint8_t* bitStream, int16_t* bitStreamLenByte) {
+ *bitStreamLenByte = 0;
+
+ // Call Encoder.
+ *bitStreamLenByte = WebRtcCelt_Encode(enc_inst_ptr_,
+ &_inAudio[_inAudioIxRead],
+ bitStream);
+
+ // Increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer.
+ _inAudioIxRead += _frameLenSmpl * channels_;
+
+ if (*bitStreamLenByte < 0) {
+ // Error reported from the encoder.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Encode error for Celt");
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+ return *bitStreamLenByte;
+}
+
+int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
+ int16_t /* bitStreamLenByte */,
+ int16_t* /* audio */,
+ int16_t* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+ // Set bitrate and check that it is within the valid range.
+ int16_t status = SetBitRateSafe((codecParams->codecInstant).rate);
+ if (status < 0) {
+ return -1;
+ }
+
+ // If number of channels changed we need to re-create memory.
+ if (codecParams->codecInstant.channels != channels_) {
+ WebRtcCelt_FreeEnc(enc_inst_ptr_);
+ enc_inst_ptr_ = NULL;
+ // Store new number of channels.
+ channels_ = codecParams->codecInstant.channels;
+ if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) {
+ return -1;
+ }
+ }
+
+ // Initiate encoder.
+ if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
+ return 0;
+ } else {
+ return -1;
+ }
+}
+
+int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
+ // If number of channels changed we need to re-create memory.
+ if (codecParams->codecInstant.channels != dec_channels_) {
+ WebRtcCelt_FreeDec(dec_inst_ptr_);
+ dec_inst_ptr_ = NULL;
+ // Store new number of channels.
+ dec_channels_ = codecParams->codecInstant.channels;
+ if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
+ return -1;
+ }
+ }
+
+ // Initiate decoder.
+ if (WebRtcCelt_DecoderInit(dec_inst_ptr_) < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: init decoder failed for Celt.");
+ return -1;
+ }
+ return 0;
+}
+
+int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodecDef: Decoder uninitialized for Celt");
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" and "SET_CELT_FUNCTIONS" or "SET_CELTSLAVE_FUNCTIONS".
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderCELT_32, codecInst.pltype, dec_inst_ptr_,
+ 32000);
+ // If this is the master of NetEQ, regular decoder will be added, otherwise
+ // the slave decoder will be used.
+ if (_isMaster) {
+ SET_CELT_FUNCTIONS((codecDef));
+ } else {
+ SET_CELTSLAVE_FUNCTIONS((codecDef));
+ }
+ return 0;
+}
+
+ACMGenericCodec* ACMCELT::CreateInstance(void) {
+ return NULL;
+}
+
+int16_t ACMCELT::InternalCreateEncoder() {
+ if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, _noChannels) < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: create encoder failed for Celt");
+ return -1;
+ }
+ channels_ = _noChannels;
+ return 0;
+}
+
+void ACMCELT::DestructEncoderSafe() {
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if (enc_inst_ptr_ != NULL) {
+ WebRtcCelt_FreeEnc(enc_inst_ptr_);
+ enc_inst_ptr_ = NULL;
+ }
+}
+
+int16_t ACMCELT::InternalCreateDecoder() {
+ if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: create decoder failed for Celt");
+ return -1;
+ }
+
+ return 0;
+}
+
+void ACMCELT::DestructDecoderSafe() {
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if (dec_inst_ptr_ != NULL) {
+ WebRtcCelt_FreeDec(dec_inst_ptr_);
+ dec_inst_ptr_ = NULL;
+ }
+}
+
+void ACMCELT::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ WebRtcCelt_FreeEnc(static_cast<CELT_encinst_t*>(ptrInst));
+ }
+ return;
+}
+
+int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(
+ webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec: given payload-type does not match the stored "
+ "payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderCELT_32);
+}
+
+bool ACMCELT::IsTrueStereoCodec() {
+ return true;
+}
+
+int16_t ACMCELT::SetBitRateSafe(const int32_t rate) {
+ // Check that rate is in the valid range.
+ if ((rate >= 48000) && (rate <= 128000)) {
+ bitrate_ = rate;
+ return 0;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: Invalid rate Celt");
+ return -1;
+ }
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_celt.h b/trunk/src/modules/audio_coding/main/source/acm_celt.h
new file mode 100644
index 0000000..776bd7f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_celt.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct CELT_encinst_t_;
+struct CELT_decinst_t_;
+
+namespace webrtc {
+
+class ACMCELT : public ACMGenericCodec {
+ public:
+ ACMCELT(int16_t codecID);
+ ~ACMCELT();
+
+ ACMGenericCodec* CreateInstance(void);
+
+ int16_t InternalEncode(uint8_t* bitstream, int16_t* bitStreamLenByte);
+
+ int16_t InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ int16_t InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ protected:
+
+ WebRtc_Word16 DecodeSafe(
+ uint8_t* /* bitStream */,
+ int16_t /* bitStreamLenByte */,
+ int16_t* /* audio */,
+ int16_t* /* audioSamples */,
+ // TODO(leozwang): use int8_t here when WebRtc_Word8 is properly typed.
+ // http://code.google.com/p/webrtc/issues/detail?id=311
+ WebRtc_Word8* /* speechType */);
+
+ int32_t CodecDef(WebRtcNetEQ_CodecDef& codecDef, const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ int16_t InternalCreateEncoder();
+
+ int16_t InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ int16_t UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType);
+
+ bool IsTrueStereoCodec();
+
+ int16_t SetBitRateSafe(const int32_t rate);
+
+ CELT_encinst_t_* enc_inst_ptr_;
+ CELT_decinst_t_* dec_inst_ptr_;
+ uint16_t sampling_freq_;
+ int32_t bitrate_;
+ uint16_t channels_;
+ uint16_t dec_channels_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_cng.cc b/trunk/src/modules/audio_coding/main/source/acm_cng.cc
new file mode 100644
index 0000000..1bda752
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_cng.cc
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_cng.h"
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_cng.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+namespace webrtc {
+
+ACMCNG::ACMCNG(WebRtc_Word16 codecID) {
+ _encoderInstPtr = NULL;
+ _decoderInstPtr = NULL;
+ _codecID = codecID;
+ _sampFreqHz = ACMCodecDB::CodecFreq(_codecID);
+ return;
+}
+
+ACMCNG::~ACMCNG() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcCng_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ WebRtcCng_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+// CNG is not like a regular encoder, this function
+// should not be called normally
+// instead the following function is called from inside
+// ACMGenericCodec::ProcessFrameVADDTX
+WebRtc_Word16 ACMCNG::InternalEncode(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMCNG::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+// CNG is not like a regular encoder,
+// this function should not be called normally
+// instead the following function is called from inside
+// ACMGenericCodec::ProcessFrameVADDTX
+WebRtc_Word16 ACMCNG::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMCNG::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return WebRtcCng_InitDec(_decoderInstPtr);
+}
+
+WebRtc_Word32 ACMCNG::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ // TODO (tlegrand): log error
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_CNG_FUNCTION."
+ // Then return the structure back to NetEQ to add the codec to it's
+ // database.
+
+ if (_sampFreqHz == 8000 || _sampFreqHz == 16000 || _sampFreqHz == 32000) {
+ SET_CODEC_PAR((codecDef), kDecoderCNG, codecInst.pltype,
+ _decoderInstPtr, _sampFreqHz);
+ SET_CNG_FUNCTIONS((codecDef));
+ return 0;
+ } else {
+ return -1;
+ }
+}
+
+ACMGenericCodec* ACMCNG::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMCNG::InternalCreateEncoder() {
+ if (WebRtcCng_CreateEnc(&_encoderInstPtr) < 0) {
+ _encoderInstPtr = NULL;
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+void ACMCNG::DestructEncoderSafe() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcCng_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ _encoderExist = false;
+ _encoderInitialized = false;
+}
+
+WebRtc_Word16 ACMCNG::InternalCreateDecoder() {
+ if (WebRtcCng_CreateDec(&_decoderInstPtr) < 0) {
+ _decoderInstPtr = NULL;
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+void ACMCNG::DestructDecoderSafe() {
+ if (_decoderInstPtr != NULL) {
+ WebRtcCng_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ _decoderExist = false;
+ _decoderInitialized = false;
+}
+
+void ACMCNG::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ WebRtcCng_FreeEnc(static_cast<CNG_enc_inst*>(ptrInst));
+ }
+ return;
+}
+
+WebRtc_Word16 ACMCNG::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(
+ webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not "
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderCNG);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_cng.h b/trunk/src/modules/audio_coding/main/source/acm_cng.h
new file mode 100644
index 0000000..1781b2b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_cng.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct WebRtcCngEncInst;
+struct WebRtcCngDecInst;
+
+namespace webrtc {
+
+class ACMCNG: public ACMGenericCodec {
+ public:
+ ACMCNG(WebRtc_Word16 codecID);
+ ~ACMCNG();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 EnableDTX() {
+ return -1;
+ }
+
+ WebRtc_Word16 DisableDTX() {
+ return -1;
+ }
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ WebRtcCngEncInst* _encoderInstPtr;
+ WebRtcCngDecInst* _decoderInstPtr;
+ WebRtc_Word16 _sampFreqHz;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc b/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc
new file mode 100644
index 0000000..c623d29
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc
@@ -0,0 +1,998 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file generates databases with information about all supported audio
+ * codecs.
+ */
+
+// TODO(tlegrand): Change constant input pointers in all functions to constant
+// references, where appropriate.
+#include "acm_codec_database.h"
+
+#include <stdio.h>
+
+#include "acm_common_defs.h"
+#include "trace.h"
+
+// Includes needed to get version info and to create the codecs.
+// G.711, PCM mu-law and A-law.
+#include "acm_pcma.h"
+#include "acm_pcmu.h"
+#include "g711_interface.h"
+// CNG.
+#include "acm_cng.h"
+#include "webrtc_cng.h"
+// NetEQ.
+#include "webrtc_neteq.h"
+#ifdef WEBRTC_CODEC_ISAC
+ #include "acm_isac.h"
+ #include "acm_isac_macros.h"
+ #include "isac.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+ #include "acm_isac.h"
+ #include "acm_isac_macros.h"
+ #include "isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ #include "pcm16b.h"
+ #include "acm_pcm16b.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ #include "acm_ilbc.h"
+ #include "ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ #include "acm_amr.h"
+ #include "amr_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ #include "acm_amrwb.h"
+ #include "amrwb_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ #include "acm_celt.h"
+ #include "celt_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_G722
+ #include "acm_g722.h"
+ #include "g722_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ #include "acm_g7221.h"
+ #include "g7221_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ #include "acm_g7221c.h"
+ #include "g7221c_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_G729
+ #include "acm_g729.h"
+ #include "g729_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ #include "acm_g7291.h"
+ #include "g7291_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ #include "acm_gsmfr.h"
+ #include "gsmfr_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ #include "acm_speex.h"
+ #include "speex_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_AVT
+ #include "acm_dtmf_playout.h"
+#endif
+#ifdef WEBRTC_CODEC_RED
+ #include "acm_red.h"
+#endif
+
+namespace webrtc {
+
+// We dynamically allocate some of the dynamic payload types to the defined
+// codecs. Note! There are a limited number of payload types. If more codecs
+// are defined they will receive reserved fixed payload types (values 67-95).
+const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
+ 105, 107, 108, 109, 110, 111, 112, 113, 114, 115, 116, 117, 118, 119,
+ 120, 121, 122, 123, 124, 125, 126, 95, 94, 93, 92, 91, 90, 89,
+ 88, 87, 86, 85, 84, 83, 82, 81, 80, 79, 78, 77, 76, 75,
+ 74, 73, 72, 71, 70, 69, 68, 67
+};
+
+// Creates database with all supported codec at compile time.
+// Each entry needs the following parameters in the given order:
+// payload type, name, sampling frequency, packet size in samples,
+// default channel support, and default rate.
+#if (defined(WEBRTC_CODEC_PCM16) || \
+ defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) || \
+ defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G729_1) || \
+ defined(WEBRTC_CODEC_SPEEX) || defined(WEBRTC_CODEC_G722_1) || \
+ defined(WEBRTC_CODEC_G722_1C))
+static int count_database = 0;
+#endif
+
+const CodecInst ACMCodecDB::database_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ {103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate},
+# if (defined(WEBRTC_CODEC_ISAC))
+ {104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate},
+# endif
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ {kDynamicPayloadtypes[count_database++], "L16", 8000, 80, 1, 128000},
+ {kDynamicPayloadtypes[count_database++], "L16", 16000, 160, 1, 256000},
+ {kDynamicPayloadtypes[count_database++], "L16", 32000, 320, 1, 512000},
+#endif
+ // G.711, PCM mu-law and A-law.
+ {0, "PCMU", 8000, 160, 1, 64000},
+ {8, "PCMA", 8000, 160, 1, 64000},
+#ifdef WEBRTC_CODEC_ILBC
+ {102, "ILBC", 8000, 240, 1, 13300},
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ {kDynamicPayloadtypes[count_database++], "AMR", 8000, 160, 1, 12200},
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ {kDynamicPayloadtypes[count_database++], "AMR-WB", 16000, 320, 1, 20000},
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 2, 64000},
+#endif
+#ifdef WEBRTC_CODEC_G722
+ {9, "G722", 16000, 320, 1, 64000},
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 32000},
+ {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 24000},
+ {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 16000},
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 48000},
+ {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 32000},
+ {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 24000},
+#endif
+#ifdef WEBRTC_CODEC_G729
+ {18, "G729", 8000, 240, 1, 8000},
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ {kDynamicPayloadtypes[count_database++], "G7291", 16000, 320, 1, 32000},
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ {3, "GSM", 8000, 160, 1, 13200},
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ {kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
+ {kDynamicPayloadtypes[count_database++], "speex", 16000, 320, 1, 22000},
+#endif
+ // Comfort noise for three different sampling frequencies.
+ {13, "CN", 8000, 240, 1, 0},
+ {98, "CN", 16000, 480, 1, 0},
+ {99, "CN", 32000, 960, 1, 0},
+#ifdef WEBRTC_CODEC_AVT
+ {106, "telephone-event", 8000, 240, 1, 0},
+#endif
+#ifdef WEBRTC_CODEC_RED
+ {127, "red", 8000, 0, 1, 0},
+#endif
+ // To prevent compile errors due to trailing commas.
+ {-1, "Null", -1, -1, -1, -1}
+};
+
+// Create database with all codec settings at compile time.
+// Each entry needs the following parameters in the given order:
+// Number of allowed packet sizes, a vector with the allowed packet sizes,
+// Basic block samples, max number of channels that are supported.
+const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ {2, {kIsacPacSize480, kIsacPacSize960}, 0, 1},
+# if (defined(WEBRTC_CODEC_ISAC))
+ {1, {kIsacPacSize960}, 0, 1},
+# endif
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ {4, {80, 160, 240, 320}, 0, 2},
+ {4, {160, 320, 480, 640}, 0, 2},
+ {2, {320, 640}, 0, 2},
+#endif
+ // G.711, PCM mu-law and A-law.
+ {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+ {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+#ifdef WEBRTC_CODEC_ILBC
+ {4, {160, 240, 320, 480}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ {3, {160, 320, 480}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ {3, {320, 640, 960}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ {1, {320}, 0, 2},
+#endif
+#ifdef WEBRTC_CODEC_G722
+ {6, {160, 320, 480, 640, 800, 960}, 0, 2},
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ {1, {320}, 320, 2},
+ {1, {320}, 320, 2},
+ {1, {320}, 320, 2},
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ {1, {640}, 640, 2},
+ {1, {640}, 640, 2},
+ {1, {640}, 640, 2},
+#endif
+#ifdef WEBRTC_CODEC_G729
+ {6, {80, 160, 240, 320, 400, 480}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ {3, {320, 640, 960}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ {3, {160, 320, 480}, 160, 1},
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ {3, {160, 320, 480}, 0, 1},
+ {3, {320, 640, 960}, 0, 1},
+#endif
+ // Comfort noise for three different sampling frequencies.
+ {1, {240}, 240, 1},
+ {1, {480}, 480, 1},
+ {1, {960}, 960, 1},
+#ifdef WEBRTC_CODEC_AVT
+ {1, {240}, 240, 1},
+#endif
+#ifdef WEBRTC_CODEC_RED
+ {1, {0}, 0, 1},
+#endif
+ // To prevent compile errors due to trailing commas.
+ {-1, {-1}, -1, -1}
+};
+
+// Create a database of all NetEQ decoders at compile time.
+const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ kDecoderISAC,
+# if (defined(WEBRTC_CODEC_ISAC))
+ kDecoderISACswb,
+# endif
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ kDecoderPCM16B,
+ kDecoderPCM16Bwb,
+ kDecoderPCM16Bswb32kHz,
+#endif
+ // G.711, PCM mu-las and A-law.
+ kDecoderPCMu,
+ kDecoderPCMa,
+#ifdef WEBRTC_CODEC_ILBC
+ kDecoderILBC,
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ kDecoderAMR,
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ kDecoderAMRWB,
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ kDecoderCELT_32,
+#endif
+#ifdef WEBRTC_CODEC_G722
+ kDecoderG722,
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ kDecoderG722_1_32,
+ kDecoderG722_1_24,
+ kDecoderG722_1_16,
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ kDecoderG722_1C_48,
+ kDecoderG722_1C_32,
+ kDecoderG722_1C_24,
+#endif
+#ifdef WEBRTC_CODEC_G729
+ kDecoderG729,
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ kDecoderG729_1,
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ kDecoderGSMFR,
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ kDecoderSPEEX_8,
+ kDecoderSPEEX_16,
+#endif
+ // Comfort noise for three different sampling frequencies.
+ kDecoderCNG,
+ kDecoderCNG,
+ kDecoderCNG,
+#ifdef WEBRTC_CODEC_AVT
+ kDecoderAVT,
+#endif
+#ifdef WEBRTC_CODEC_RED
+ kDecoderRED,
+#endif
+ kDecoderReservedEnd
+};
+
+// Get codec information from database.
+// TODO(tlegrand): replace memcpy with a pointer to the data base memory.
+int ACMCodecDB::Codec(int codec_id, CodecInst* codec_inst) {
+ // Error check to see that codec_id is not out of bounds.
+ if ((codec_id < 0) || (codec_id >= kNumCodecs)) {
+ return -1;
+ }
+
+ // Copy database information for the codec to the output.
+ memcpy(codec_inst, &database_[codec_id], sizeof(CodecInst));
+
+ return 0;
+}
+
+// Enumerator for error codes when asking for codec database id.
+enum {
+ kInvalidCodec = -10,
+ kInvalidFrequency = -20,
+ kInvalidPayloadtype = -30,
+ kInvalidPacketSize = -40,
+ kInvalidRate = -50
+};
+
+// Gets the codec id number from the database. If there is some mismatch in
+// the codec settings, an error message will be recorded in the error string.
+// NOTE! Only the first mismatch found will be recorded in the error string.
+int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id,
+ char* err_message, int max_message_len_byte) {
+ int codec_id = ACMCodecDB::CodecNumber(codec_inst, mirror_id);
+
+ // Write error message if ACMCodecDB::CodecNumber() returned error.
+ if ((codec_id < 0) && (err_message != NULL)) {
+ char my_err_msg[1000];
+
+ if (codec_id == kInvalidCodec) {
+ sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, plname=%s "
+ "is not a valid codec", codec_inst->plname);
+ } else if (codec_id == kInvalidFrequency) {
+ sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, plfreq=%d "
+ "is not a valid frequency for the codec %s", codec_inst->plfreq,
+ codec_inst->plname);
+ } else if (codec_id == kInvalidPayloadtype) {
+ sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, payload "
+ "number %d is out of range for %s", codec_inst->pltype,
+ codec_inst->plname);
+ } else if (codec_id == kInvalidPacketSize) {
+ sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, Packet "
+ "size is out of range for %s", codec_inst->plname);
+ } else if (codec_id == kInvalidRate) {
+ sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, rate=%d "
+ "is not a valid rate for %s", codec_inst->rate,
+ codec_inst->plname);
+ } else {
+ // Other error
+ sprintf(my_err_msg, "invalid codec parameters to be registered, "
+ "ACMCodecDB::CodecNumber failed");
+ }
+
+ strncpy(err_message, my_err_msg, max_message_len_byte - 1);
+ // make sure that the message is null-terminated.
+ err_message[max_message_len_byte - 1] = '\0';
+ }
+
+ return codec_id;
+}
+
+// Gets the codec id number from the database. If there is some mismatch in
+// the codec settings, the function will return an error code.
+// NOTE! The first mismatch found will generate the return value.
+int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id) {
+ int codec_number = -1;
+ bool name_match = false;
+
+ // Looks for a matching payload name and frequency in the codec list.
+ // Need to check both since some codecs have several codec entries with
+ // different frequencies (like iSAC).
+ for (int i = 0; i < kNumCodecs; i++) {
+ if (STR_CASE_CMP(database_[i].plname, codec_inst->plname) == 0) {
+ // We have found a matching codec name in the list.
+ name_match = true;
+
+ // Checks if frequency match.
+ if (codec_inst->plfreq == database_[i].plfreq) {
+ codec_number = i;
+ break;
+ }
+ }
+ }
+
+ // Checks if the error is in the name or in the frequency.
+ if (codec_number == -1) {
+ if (!name_match) {
+ return kInvalidCodec;
+ } else {
+ return kInvalidFrequency;
+ }
+ }
+
+ // Checks the validity of payload type
+ if (!ValidPayloadType(codec_inst->pltype)) {
+ return kInvalidPayloadtype;
+ }
+
+ // Comfort Noise is special case, packet-size & rate is not checked.
+ if (STR_CASE_CMP(database_[codec_number].plname, "CN") == 0) {
+ *mirror_id = codec_number;
+ return codec_number;
+ }
+
+ // RED is special case, packet-size & rate is not checked.
+ if (STR_CASE_CMP(database_[codec_number].plname, "red") == 0) {
+ *mirror_id = codec_number;
+ return codec_number;
+ }
+
+ // Checks the validity of packet size.
+ if (codec_settings_[codec_number].num_packet_sizes > 0) {
+ bool packet_size_ok = false;
+ int i;
+ int packet_size_samples;
+ for (i = 0; i < codec_settings_[codec_number].num_packet_sizes; i++) {
+ packet_size_samples =
+ codec_settings_[codec_number].packet_sizes_samples[i];
+ if (codec_inst->pacsize == packet_size_samples) {
+ packet_size_ok = true;
+ break;
+ }
+ }
+
+ if (!packet_size_ok) {
+ return kInvalidPacketSize;
+ }
+ }
+
+ if (codec_inst->pacsize < 1) {
+ return kInvalidPacketSize;
+ }
+
+
+ // Check the validity of rate. Codecs with multiple rates have their own
+ // function for this.
+ *mirror_id = codec_number;
+ if (STR_CASE_CMP("isac", codec_inst->plname) == 0) {
+ if (IsISACRateValid(codec_inst->rate)) {
+ // Set mirrorID to iSAC WB which is only created once to be used both for
+ // iSAC WB and SWB, because they need to share struct.
+ *mirror_id = kISAC;
+ return codec_number;
+ } else {
+ return kInvalidRate;
+ }
+ } else if (STR_CASE_CMP("ilbc", codec_inst->plname) == 0) {
+ return IsILBCRateValid(codec_inst->rate, codec_inst->pacsize)
+ ? codec_number : kInvalidRate;
+ } else if (STR_CASE_CMP("amr", codec_inst->plname) == 0) {
+ return IsAMRRateValid(codec_inst->rate)
+ ? codec_number : kInvalidRate;
+ } else if (STR_CASE_CMP("amr-wb", codec_inst->plname) == 0) {
+ return IsAMRwbRateValid(codec_inst->rate)
+ ? codec_number : kInvalidRate;
+ } else if (STR_CASE_CMP("g7291", codec_inst->plname) == 0) {
+ return IsG7291RateValid(codec_inst->rate)
+ ? codec_number : kInvalidRate;
+ } else if (STR_CASE_CMP("speex", codec_inst->plname) == 0) {
+ return IsSpeexRateValid(codec_inst->rate)
+ ? codec_number : kInvalidRate;
+ } else if (STR_CASE_CMP("celt", codec_inst->plname) == 0) {
+ return IsCeltRateValid(codec_inst->rate)
+ ? codec_number : kInvalidRate;
+ }
+
+ return IsRateValid(codec_number, codec_inst->rate) ?
+ codec_number : kInvalidRate;
+}
+
+// Gets codec id number, and mirror id, from database for the receiver.
+int ACMCodecDB::ReceiverCodecNumber(const CodecInst* codec_inst,
+ int* mirror_id) {
+ int codec_number = -1;
+
+ // Looks for a matching payload name and frequency in the codec list.
+ // Need to check both since some codecs have several codec entries with
+ // different frequencies (like iSAC).
+ for (int i = 0; i < kNumCodecs; i++) {
+ if (STR_CASE_CMP(database_[i].plname, codec_inst->plname) == 0) {
+ // We have found a matching codec name in the list.
+
+ // Check if frequency match.
+ if (codec_inst->plfreq == database_[i].plfreq) {
+ codec_number = i;
+ *mirror_id = codec_number;
+
+ // Check if codec is iSAC, set mirrorID to iSAC WB which is only
+ // created once to be used both for iSAC WB and SWB, because they need
+ // to share struct.
+ if (STR_CASE_CMP(codec_inst->plname, "ISAC") == 0) {
+ *mirror_id = kISAC;
+ }
+
+ break;
+ }
+ }
+ }
+
+ return codec_number;
+}
+
+// Returns the codec sampling frequency for codec with id = "codec_id" in
+// database.
+int ACMCodecDB::CodecFreq(int codec_id) {
+ // Error check to see that codec_id is not out of bounds.
+ if (codec_id < 0 || codec_id >= kNumCodecs) {
+ return -1;
+ }
+
+ return database_[codec_id].plfreq;
+}
+
+// Returns the codec's basic coding block size in samples.
+int ACMCodecDB::BasicCodingBlock(int codec_id) {
+ // Error check to see that codec_id is not out of bounds.
+ if (codec_id < 0 || codec_id >= kNumCodecs) {
+ return -1;
+ }
+
+ return codec_settings_[codec_id].basic_block_samples;
+}
+
+// Returns the NetEQ decoder database.
+const WebRtcNetEQDecoder* ACMCodecDB::NetEQDecoders() {
+ return neteq_decoders_;
+}
+
+// All version numbers for the codecs in the database are listed in text.
+// TODO(tlegrand): change to use std::string.
+int ACMCodecDB::CodecsVersion(char* version, size_t* remaining_buffer_bytes,
+ size_t* position) {
+ const size_t kTemporaryBufferSize = 500;
+ const size_t kVersionBufferSize = 1000;
+ char versions_buffer[kVersionBufferSize];
+ char version_num_buf[kTemporaryBufferSize];
+ size_t len = *position;
+ size_t remaining_size = kVersionBufferSize;
+
+ versions_buffer[0] = '\0';
+
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ ACM_ISAC_VERSION(version_num_buf);
+ strncat(versions_buffer, "ISAC\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, "\n", remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, "L16\t\t1.0.0\n", remaining_size);
+#endif
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG711_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.711\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, "\n", remaining_size);
+#ifdef WEBRTC_CODEC_ILBC
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcIlbcfix_version(version_num_buf);
+ strncat(versions_buffer, "ILBC\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, "\n", remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcAmr_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "AMR\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcAmrWb_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "AMR-WB\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_G722
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG722_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.722\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG7221_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.722.1\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG7221c_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.722.1C\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_G729
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG729_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.729\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcG7291_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "G.729.1\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcGSMFR_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "GSM-FR\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcSpeex_Version(version_num_buf, kTemporaryBufferSize);
+ strncat(versions_buffer, "Speex\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#endif
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ WebRtcCng_Version(version_num_buf);
+ strncat(versions_buffer, "CNG\t\t", remaining_size);
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, version_num_buf, remaining_size);
+#ifdef WEBRTC_CODEC_AVT
+ remaining_size = kVersionBufferSize - strlen(versions_buffer);
+ strncat(versions_buffer, "Tone Generation\t1.0.0\n", remaining_size);
+#endif
+ strncpy(&version[len], versions_buffer, *remaining_buffer_bytes);
+ *position = strlen(version);
+ *remaining_buffer_bytes -= (*position - len);
+ if (*remaining_buffer_bytes < strlen(versions_buffer)) {
+ return -1;
+ }
+
+ return 0;
+}
+
+// Gets mirror id. The Id is used for codecs sharing struct for settings that
+// need different payload types.
+int ACMCodecDB::MirrorID(int codec_id) {
+ if (STR_CASE_CMP(database_[codec_id].plname, "isac") == 0) {
+ return kISAC;
+ } else {
+ return codec_id;
+ }
+}
+
+// Creates memory/instance for storing codec state.
+ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
+ // All we have support for right now.
+ if (!STR_CASE_CMP(codec_inst->plname, "ISAC")) {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ return new ACMISAC(kISAC);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "PCMU")) {
+ return new ACMPCMU(kPCMU);
+ } else if (!STR_CASE_CMP(codec_inst->plname, "PCMA")) {
+ return new ACMPCMA(kPCMA);
+ } else if (!STR_CASE_CMP(codec_inst->plname, "ILBC")) {
+#ifdef WEBRTC_CODEC_ILBC
+ return new ACMILBC(kILBC);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "AMR")) {
+#ifdef WEBRTC_CODEC_AMR
+ return new ACMAMR(kGSMAMR);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "AMR-WB")) {
+#ifdef WEBRTC_CODEC_AMRWB
+ return new ACMAMRwb(kGSMAMRWB);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "CELT")) {
+#ifdef WEBRTC_CODEC_CELT
+ return new ACMCELT(kCELT32);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "G722")) {
+#ifdef WEBRTC_CODEC_G722
+ return new ACMG722(kG722);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "G7221")) {
+ switch (codec_inst->plfreq) {
+ case 16000: {
+#ifdef WEBRTC_CODEC_G722_1
+ int codec_id;
+ switch (codec_inst->rate) {
+ case 16000 : {
+ codec_id = kG722_1_16;
+ break;
+ }
+ case 24000 : {
+ codec_id = kG722_1_24;
+ break;
+ }
+ case 32000 : {
+ codec_id = kG722_1_32;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ return new ACMG722_1(codec_id);
+ }
+#endif
+ }
+ case 32000: {
+#ifdef WEBRTC_CODEC_G722_1C
+ int codec_id;
+ switch (codec_inst->rate) {
+ case 24000 : {
+ codec_id = kG722_1C_24;
+ break;
+ }
+ case 32000 : {
+ codec_id = kG722_1C_32;
+ break;
+ }
+ case 48000 : {
+ codec_id = kG722_1C_48;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ return new ACMG722_1C(codec_id);
+ }
+#endif
+ }
+ }
+ } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
+ // For CN we need to check sampling frequency to know what codec to create.
+ int codec_id;
+ switch (codec_inst->plfreq) {
+ case 8000: {
+ codec_id = kCNNB;
+ break;
+ }
+ case 16000: {
+ codec_id = kCNWB;
+ break;
+ }
+ case 32000: {
+ codec_id = kCNSWB;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ }
+ return new ACMCNG(codec_id);
+ } else if (!STR_CASE_CMP(codec_inst->plname, "G729")) {
+#ifdef WEBRTC_CODEC_G729
+ return new ACMG729(kG729);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "G7291")) {
+#ifdef WEBRTC_CODEC_G729_1
+ return new ACMG729_1(kG729_1);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
+#ifdef WEBRTC_CODEC_SPEEX
+ int codec_id;
+ switch (codec_inst->plfreq) {
+ case 8000: {
+ codec_id = kSPEEX8;
+ break;
+ }
+ case 16000: {
+ codec_id = kSPEEX16;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ }
+ return new ACMSPEEX(codec_id);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
+ // For CN we need to check sampling frequency to know what codec to create.
+ int codec_id;
+ switch (codec_inst->plfreq) {
+ case 8000: {
+ codec_id = kCNNB;
+ break;
+ }
+ case 16000: {
+ codec_id = kCNWB;
+ break;
+ }
+ case 32000: {
+ codec_id = kCNSWB;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ }
+ return new ACMCNG(codec_id);
+ } else if (!STR_CASE_CMP(codec_inst->plname, "L16")) {
+#ifdef WEBRTC_CODEC_PCM16
+ // For L16 we need to check sampling frequency to know what codec to create.
+ int codec_id;
+ switch (codec_inst->plfreq) {
+ case 8000: {
+ codec_id = kPCM16B;
+ break;
+ }
+ case 16000: {
+ codec_id =kPCM16Bwb;
+ break;
+ }
+ case 32000: {
+ codec_id = kPCM16Bswb32kHz;
+ break;
+ }
+ default: {
+ return NULL;
+ }
+ }
+ return new ACMPCM16B(codec_id);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "telephone-event")) {
+#ifdef WEBRTC_CODEC_AVT
+ return new ACMDTMFPlayout(kAVT);
+#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "red")) {
+#ifdef WEBRTC_CODEC_RED
+ return new ACMRED(kRED);
+#endif
+ }
+ return NULL;
+}
+
+// Checks if the bitrate is valid for the codec.
+bool ACMCodecDB::IsRateValid(int codec_id, int rate) {
+ if (database_[codec_id].rate == rate) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+// Checks if the bitrate is valid for iSAC.
+bool ACMCodecDB::IsISACRateValid(int rate) {
+ if ((rate == -1) || ((rate <= 56000) && (rate >= 10000))) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+// Checks if the bitrate is valid for iLBC.
+bool ACMCodecDB::IsILBCRateValid(int rate, int frame_size_samples) {
+ if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
+ (rate == 13300)) {
+ return true;
+ } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
+ (rate == 15200)) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+// Check if the bitrate is valid for the GSM-AMR.
+bool ACMCodecDB::IsAMRRateValid(int rate) {
+ switch (rate) {
+ case 4750:
+ case 5150:
+ case 5900:
+ case 6700:
+ case 7400:
+ case 7950:
+ case 10200:
+ case 12200: {
+ return true;
+ }
+ default: {
+ return false;
+ }
+ }
+}
+
+// Check if the bitrate is valid for GSM-AMR-WB.
+bool ACMCodecDB::IsAMRwbRateValid(int rate) {
+ switch (rate) {
+ case 7000:
+ case 9000:
+ case 12000:
+ case 14000:
+ case 16000:
+ case 18000:
+ case 20000:
+ case 23000:
+ case 24000: {
+ return true;
+ }
+ default: {
+ return false;
+ }
+ }
+}
+
+// Check if the bitrate is valid for G.729.1.
+bool ACMCodecDB::IsG7291RateValid(int rate) {
+ switch (rate) {
+ case 8000:
+ case 12000:
+ case 14000:
+ case 16000:
+ case 18000:
+ case 20000:
+ case 22000:
+ case 24000:
+ case 26000:
+ case 28000:
+ case 30000:
+ case 32000: {
+ return true;
+ }
+ default: {
+ return false;
+ }
+ }
+}
+
+// Checks if the bitrate is valid for Speex.
+bool ACMCodecDB::IsSpeexRateValid(int rate) {
+ if (rate > 2000) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+// Checks if the bitrate is valid for Celt.
+bool ACMCodecDB::IsCeltRateValid(int rate) {
+ if ((rate >= 48000) && (rate <= 128000)) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+// Checks if the payload type is in the valid range.
+bool ACMCodecDB::ValidPayloadType(int payload_type) {
+ if ((payload_type < 0) || (payload_type > 127)) {
+ return false;
+ }
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_codec_database.h b/trunk/src/modules/audio_coding/main/source/acm_codec_database.h
new file mode 100644
index 0000000..6830e65
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_codec_database.h
@@ -0,0 +1,308 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file generates databases with information about all supported audio
+ * codecs.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
+
+#include "acm_generic_codec.h"
+#include "common_types.h"
+#include "webrtc_neteq.h"
+
+namespace webrtc {
+
+// TODO(tlegrand): replace class ACMCodecDB with a namespace.
+class ACMCodecDB {
+ public:
+ // Enum with array indexes for the supported codecs. NOTE! The order MUST
+ // be the same as when creating the database in acm_codec_database.cc.
+ enum {
+ kNone = -1
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ , kISAC
+# if (defined(WEBRTC_CODEC_ISAC))
+ , kISACSWB
+# endif
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ , kPCM16B
+ , kPCM16Bwb
+ , kPCM16Bswb32kHz
+#endif
+ , kPCMU
+ , kPCMA
+#ifdef WEBRTC_CODEC_ILBC
+ , kILBC
+#endif
+#ifdef WEBRTC_CODEC_AMR
+ , kGSMAMR
+#endif
+#ifdef WEBRTC_CODEC_AMRWB
+ , kGSMAMRWB
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ , kCELT32
+#endif
+#ifdef WEBRTC_CODEC_G722
+ , kG722
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ , kG722_1_32
+ , kG722_1_24
+ , kG722_1_16
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ , kG722_1C_48
+ , kG722_1C_32
+ , kG722_1C_24
+#endif
+#ifdef WEBRTC_CODEC_G729
+ , kG729
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ , kG729_1
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ , kGSMFR
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ , kSPEEX8
+ , kSPEEX16
+#endif
+ , kCNNB
+ , kCNWB
+ , kCNSWB
+#ifdef WEBRTC_CODEC_AVT
+ , kAVT
+#endif
+#ifdef WEBRTC_CODEC_RED
+ , kRED
+#endif
+ , kNumCodecs
+ };
+
+ // Set unsupported codecs to -1
+#ifndef WEBRTC_CODEC_ISAC
+ enum {kISACSWB = -1};
+# ifndef WEBRTC_CODEC_ISACFX
+ enum {kISAC = -1};
+# endif
+#endif
+#ifndef WEBRTC_CODEC_PCM16
+ enum {kPCM16B = -1};
+ enum {kPCM16Bwb = -1};
+ enum {kPCM16Bswb32kHz = -1};
+#endif
+ // 48 kHz not supported, always set to -1.
+ enum {kPCM16Bswb48kHz = -1};
+#ifndef WEBRTC_CODEC_ILBC
+ enum {kILBC = -1};
+#endif
+#ifndef WEBRTC_CODEC_AMR
+ enum {kGSMAMR = -1};
+#endif
+#ifndef WEBRTC_CODEC_AMRWB
+ enum {kGSMAMRWB = -1};
+#endif
+#ifndef WEBRTC_CODEC_CELT
+ enum {kCELT32 = -1};
+#endif
+#ifndef WEBRTC_CODEC_G722
+ enum {kG722 = -1};
+#endif
+#ifndef WEBRTC_CODEC_G722_1
+ enum {kG722_1_32 = -1};
+ enum {kG722_1_24 = -1};
+ enum {kG722_1_16 = -1};
+#endif
+#ifndef WEBRTC_CODEC_G722_1C
+ enum {kG722_1C_48 = -1};
+ enum {kG722_1C_32 = -1};
+ enum {kG722_1C_24 = -1};
+#endif
+#ifndef WEBRTC_CODEC_G729
+ enum {kG729 = -1};
+#endif
+#ifndef WEBRTC_CODEC_G729_1
+ enum {kG729_1 = -1};
+#endif
+#ifndef WEBRTC_CODEC_GSMFR
+ enum {kGSMFR = -1};
+#endif
+#ifndef WEBRTC_CODEC_SPEEX
+ enum {kSPEEX8 = -1};
+ enum {kSPEEX16 = -1};
+#endif
+#ifndef WEBRTC_CODEC_AVT
+ enum {kAVT = -1};
+#endif
+#ifndef WEBRTC_CODEC_RED
+ enum {kRED = -1};
+#endif
+
+ // kMaxNumCodecs - Maximum number of codecs that can be activated in one
+ // build.
+ // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
+ // These might need to be increased if adding a new codec to the database
+ static const int kMaxNumCodecs = 50;
+ static const int kMaxNumPacketSize = 6;
+
+ // Codec specific settings
+ //
+ // num_packet_sizes - number of allowed packet sizes.
+ // packet_sizes_samples - list of the allowed packet sizes.
+ // basic_block_samples - assigned a value different from 0 if the codec
+ // requires to be fed with a specific number of samples
+ // that can be different from packet size.
+ // channel_support - number of channels supported to encode;
+ // 1 = mono, 2 = stereo, etc.
+ struct CodecSettings {
+ int num_packet_sizes;
+ int packet_sizes_samples[kMaxNumPacketSize];
+ int basic_block_samples;
+ int channel_support;
+ };
+
+ // Gets codec information from database at the position in database given by
+ // [codec_id].
+ // Input:
+ // [codec_id] - number that specifies at what position in the database to
+ // get the information.
+ // Output:
+ // [codec_inst] - filled with information about the codec.
+ // Return:
+ // 0 if successful, otherwise -1.
+ static int Codec(int codec_id, CodecInst* codec_inst);
+
+ // Returns codec id and mirror id from database, given the information
+ // received in the input [codec_inst]. Mirror id is a number that tells
+ // where to find the codec's memory (instance). The number is either the
+ // same as codec id (most common), or a number pointing at a different
+ // entry in the database, if the codec has several entries with different
+ // payload types. This is used for codecs that must share one struct even if
+ // the payload type differs.
+ // One example is the codec iSAC which has the same struct for both 16 and
+ // 32 khz, but they have different entries in the database. Let's say the
+ // function is called with iSAC 32kHz. The function will return 1 as that is
+ // the entry in the data base, and [mirror_id] = 0, as that is the entry for
+ // iSAC 16 kHz, which holds the shared memory.
+ // Input:
+ // [codec_inst] - Information about the codec for which we require the
+ // database id.
+ // Output:
+ // [mirror_id] - mirror id, which most often is the same as the return
+ // value, see above.
+ // [err_message] - if present, in the event of a mismatch found between the
+ // input and the database, a descriptive error message is
+ // written here.
+ // [err_message] - if present, the length of error message is returned here.
+ // Return:
+ // codec id if successful, otherwise < 0.
+ static int CodecNumber(const CodecInst* codec_inst, int* mirror_id,
+ char* err_message, int max_message_len_byte);
+ static int CodecNumber(const CodecInst* codec_inst, int* mirror_id);
+ static int ReceiverCodecNumber(const CodecInst* codec_inst, int* mirror_id);
+
+ // Returns the codec sampling frequency for codec with id = "codec_id" in
+ // database.
+ // TODO(tlegrand): Check if function is needed, or if we can change
+ // to access database directly.
+ // Input:
+ // [codec_id] - number that specifies at what position in the database to
+ // get the information.
+ // Return:
+ // codec sampling frequency if successful, otherwise -1.
+ static int CodecFreq(int codec_id);
+
+ // Return the codec's basic coding block size in samples.
+ // TODO(tlegrand): Check if function is needed, or if we can change
+ // to access database directly.
+ // Input:
+ // [codec_id] - number that specifies at what position in the database to
+ // get the information.
+ // Return:
+ // codec basic block size if successful, otherwise -1.
+ static int BasicCodingBlock(int codec_id);
+
+ // Returns the NetEQ decoder database.
+ static const WebRtcNetEQDecoder* NetEQDecoders();
+
+ // All version numbers for the codecs in the database are listed in text.
+ // Input/Output:
+ // [version] - pointer to a char vector with minimum size 1000 bytes.
+ // Audio coding module's and all component's versions is
+ // written here.
+ // [remaining_buffer_bytes] - remaining space in buffer.
+ // [position] - current position to write at in buffer.
+ // Return:
+ // -1 if version information doesn't fit, 0 on success.
+ static int CodecsVersion(char* version, size_t* remaining_buffer_bytes,
+ size_t* position);
+
+ // Returns mirror id, which is a number that tells where to find the codec's
+ // memory (instance). It is either the same as codec id (most common), or a
+ // number pointing at a different entry in the database, if the codec have
+ // several entries with different payload types. This is used for codecs that
+ // must share struct even if the payload type differs.
+ // TODO(tlegrand): Check if function is needed, or if we can change
+ // to access database directly.
+ // Input:
+ // [codec_id] - number that specifies codec's position in the database.
+ // Return:
+ // Mirror id on success, otherwise -1.
+ static int MirrorID(int codec_id);
+
+ // Create memory/instance for storing codec state.
+ // Input:
+ // [codec_inst] - information about codec. Only name of codec, "plname", is
+ // used in this function.
+ static ACMGenericCodec* CreateCodecInstance(const CodecInst* codec_inst);
+
+ // Checks if the bitrate is valid for the codec.
+ // Input:
+ // [codec_id] - number that specifies codec's position in the database.
+ // [rate] - bitrate to check.
+ // [frame_size_samples] - (used for iLBC) specifies which frame size to go
+ // with the rate.
+ static bool IsRateValid(int codec_id, int rate);
+ static bool IsISACRateValid(int rate);
+ static bool IsILBCRateValid(int rate, int frame_size_samples);
+ static bool IsAMRRateValid(int rate);
+ static bool IsAMRwbRateValid(int rate);
+ static bool IsG7291RateValid(int rate);
+ static bool IsSpeexRateValid(int rate);
+ static bool IsCeltRateValid(int rate);
+
+ // Check if the payload type is valid, meaning that it is in the valid range
+ // of 0 to 127.
+ // Input:
+ // [payload_type] - payload type.
+ static bool ValidPayloadType(int payload_type);
+
+ // Databases with information about the supported codecs
+ // database_ - stored information about all codecs: payload type, name,
+ // sampling frequency, packet size in samples, default channel
+ // support, and default rate.
+ // codec_settings_ - stored codec settings: number of allowed packet sizes,
+ // a vector with the allowed packet sizes, basic block
+ // samples, and max number of channels that are supported.
+ // neteq_decoders_ - list of supported decoders in NetEQ.
+ static const CodecInst database_[kMaxNumCodecs];
+ static const CodecSettings codec_settings_[kMaxNumCodecs];
+ static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_common_defs.h b/trunk/src/modules/audio_coding/main/source/acm_common_defs.h
new file mode 100644
index 0000000..fd8dbd6
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_common_defs.h
@@ -0,0 +1,115 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
+
+#include <string.h>
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "typedefs.h"
+
+// Checks for enabled codecs, we prevent enabling codecs which are not
+// compatible.
+#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
+#error iSAC and iSACFX codecs cannot be enabled at the same time
+#endif
+
+#ifdef WIN32
+// OS-dependent case-insensitive string comparison
+#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+// OS-dependent case-insensitive string comparison
+#define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+namespace webrtc {
+
+// 60 ms is the maximum block size we support. An extra 20 ms is considered
+// for safety if process() method is not called when it should be, i.e. we
+// accept 20 ms of jitter. 80 ms @ 32 kHz (super wide-band) is 2560 samples.
+#define AUDIO_BUFFER_SIZE_W16 2560
+
+// There is one timestamp per each 10 ms of audio
+// the audio buffer, at max, may contain 32 blocks of 10ms
+// audio if the sampling frequency is 8000 Hz (80 samples per block).
+// Therefore, The size of the buffer where we keep timestamps
+// is defined as follows
+#define TIMESTAMP_BUFFER_SIZE_W32 (AUDIO_BUFFER_SIZE_W16/80)
+
+// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
+#define MAX_PAYLOAD_SIZE_BYTE 7680
+
+// General codec specific defines
+const int kIsacWbDefaultRate = 32000;
+const int kIsacSwbDefaultRate = 56000;
+const int kIsacPacSize480 = 480;
+const int kIsacPacSize960 = 960;
+
+// An encoded bit-stream is labeled by one of the following enumerators.
+//
+// kNoEncoding : There has been no encoding.
+// kActiveNormalEncoded : Active audio frame coded by the codec.
+// kPassiveNormalEncoded : Passive audio frame coded by the codec.
+// kPassiveDTXNB : Passive audio frame coded by narrow-band CN.
+// kPassiveDTXWB : Passive audio frame coded by wide-band CN.
+// kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN.
+//
+enum WebRtcACMEncodingType {
+ kNoEncoding,
+ kActiveNormalEncoded,
+ kPassiveNormalEncoded,
+ kPassiveDTXNB,
+ kPassiveDTXWB,
+ kPassiveDTXSWB
+};
+
+// A structure which contains codec parameters. For instance, used when
+// initializing encoder and decoder.
+//
+// codecInstant : c.f. common_types.h
+// enableDTX : set true to enable DTX. If codec does not have
+// internal DTX, this will enable VAD.
+// enableVAD : set true to enable VAD.
+// vadMode : VAD mode, c.f. audio_coding_module_typedefs.h
+// for possible values.
+struct WebRtcACMCodecParams {
+ CodecInst codecInstant;
+ bool enableDTX;
+ bool enableVAD;
+ ACMVADMode vadMode;
+};
+
+// A structure that encapsulates audio buffer and related parameters
+// used for synchronization of audio of two ACMs.
+//
+// inAudio : same as ACMGenericCodec::_inAudio
+// inAudioIxRead : same as ACMGenericCodec::_inAudioIxRead
+// inAudioIxWrite : same as ACMGenericCodec::_inAudioIxWrite
+// inTimestamp : same as ACMGenericCodec::_inTimestamp
+// inTimestampIxWrite : same as ACMGenericCodec::_inTImestampIxWrite
+// lastTimestamp : same as ACMGenericCodec::_lastTimestamp
+// lastInTimestamp : same as AudioCodingModuleImpl::_lastInTimestamp
+//
+struct WebRtcACMAudioBuff {
+ WebRtc_Word16 inAudio[AUDIO_BUFFER_SIZE_W16];
+ WebRtc_Word16 inAudioIxRead;
+ WebRtc_Word16 inAudioIxWrite;
+ WebRtc_UWord32 inTimestamp[TIMESTAMP_BUFFER_SIZE_W32];
+ WebRtc_Word16 inTimestampIxWrite;
+ WebRtc_UWord32 lastTimestamp;
+ WebRtc_UWord32 lastInTimestamp;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc
new file mode 100644
index 0000000..7669b07
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_dtmf_detection.h"
+#include "audio_coding_module_typedefs.h"
+
+namespace webrtc {
+
+ACMDTMFDetection::ACMDTMFDetection()
+ : _init(0) {}
+
+ACMDTMFDetection::~ACMDTMFDetection() {}
+
+WebRtc_Word16 ACMDTMFDetection::Enable(ACMCountries /* cpt */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFDetection::Disable() {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFDetection::Detect(
+ const WebRtc_Word16* /* inAudioBuff */,
+ const WebRtc_UWord16 /* inBuffLenWord16 */,
+ const WebRtc_Word32 /* inFreqHz */,
+ bool& /* toneDetected */,
+ WebRtc_Word16& /* tone */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFDetection::GetVersion(
+ WebRtc_Word8* /* version */,
+ WebRtc_UWord32& /* remainingBufferInBytes */,
+ WebRtc_UWord32& /* position */) {
+ return -1;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h
new file mode 100644
index 0000000..16aba84
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
+
+#include "acm_resampler.h"
+#include "audio_coding_module_typedefs.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class ACMDTMFDetection {
+ public:
+ ACMDTMFDetection();
+ ~ACMDTMFDetection();
+ WebRtc_Word16 Enable(ACMCountries cpt = ACMDisableCountryDetection);
+ WebRtc_Word16 Disable();
+ WebRtc_Word16 Detect(const WebRtc_Word16* inAudioBuff,
+ const WebRtc_UWord16 inBuffLenWord16,
+ const WebRtc_Word32 inFreqHz,
+ bool& toneDetected,
+ WebRtc_Word16& tone);
+
+ static WebRtc_Word16 GetVersion(WebRtc_Word8* version,
+ WebRtc_UWord32& remainingBufferInBytes,
+ WebRtc_UWord32& position);
+
+ private:
+ ACMResampler _resampler;
+ bool _init;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc
new file mode 100644
index 0000000..e748f8a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_dtmf_playout.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_AVT
+
+ACMDTMFPlayout::ACMDTMFPlayout(
+ WebRtc_Word16 /* codecID */) {
+ return;
+}
+
+ACMDTMFPlayout::~ACMDTMFPlayout() {
+ return;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
+ return -1;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+void ACMDTMFPlayout::DestructEncoderSafe() {
+ return;
+}
+
+void ACMDTMFPlayout::DestructDecoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+ACMDTMFPlayout::ACMDTMFPlayout(WebRtc_Word16 codecID) {
+ _codecID = codecID;
+}
+
+ACMDTMFPlayout::~ACMDTMFPlayout() {
+ return;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ // This codec does not need initialization,
+ // DTMFPlayout has no instance
+ return 0;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ // This codec does not need initialization,
+ // DTMFPlayout has no instance
+ return 0;
+}
+
+WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_AVT_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderAVT, codecInst.pltype, NULL, 8000);
+ SET_AVT_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
+ // DTMFPlayout has no instance
+ return 0;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
+ // DTMFPlayout has no instance
+ return 0;
+}
+
+void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptrInst */) {
+ // DTMFPlayout has no instance
+ return;
+}
+
+void ACMDTMFPlayout::DestructEncoderSafe() {
+ // DTMFPlayout has no instance
+ return;
+}
+
+void ACMDTMFPlayout::DestructDecoderSafe() {
+ // DTMFPlayout has no instance
+ return;
+}
+
+WebRtc_Word16 ACMDTMFPlayout::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not "
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderAVT);
+}
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h
new file mode 100644
index 0000000..a75d691
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc {
+
+class ACMDTMFPlayout: public ACMGenericCodec {
+ public:
+ ACMDTMFPlayout(WebRtc_Word16 codecID);
+ ~ACMDTMFPlayout();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void InternalDestructEncoderInst(void* ptrInst);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g722.cc b/trunk/src/modules/audio_coding/main/source/acm_g722.cc
new file mode 100644
index 0000000..40e431c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g722.cc
@@ -0,0 +1,337 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_g722.h"
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+#include "g722_interface.h"
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_G722
+
+ACMG722::ACMG722(WebRtc_Word16 /* codecID */)
+ : _ptrEncStr(NULL),
+ _ptrDecStr(NULL),
+ _encoderInstPtr(NULL),
+ _encoderInstPtrRight(NULL),
+ _decoderInstPtr(NULL) {
+ return;
+}
+
+ACMG722::~ACMG722() {
+ return;
+}
+
+WebRtc_Word32 ACMG722::Add10MsDataSafe(const WebRtc_UWord32 /* timestamp */,
+ const WebRtc_Word16* /* data */,
+ const WebRtc_UWord16 /* lengthSmpl */,
+ const WebRtc_UWord8 /* audioChannel */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722::InternalEncode(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMG722::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMG722::DestructEncoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMG722::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMG722::DestructDecoderSafe() {
+ return;
+}
+
+void ACMG722::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+WebRtc_Word16 ACMG722::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+// Encoder and decoder memory
+struct ACMG722EncStr {
+ G722EncInst* inst; // instance for left channel in case of stereo
+ G722EncInst* instRight; // instance for right channel in case of stereo
+};
+struct ACMG722DecStr {
+ G722DecInst* inst; // instance for left channel in case of stereo
+ G722DecInst* instRight; // instance for right channel in case of stereo
+};
+
+ACMG722::ACMG722(WebRtc_Word16 codecID)
+ : _encoderInstPtr(NULL),
+ _encoderInstPtrRight(NULL),
+ _decoderInstPtr(NULL) {
+ // Encoder
+ _ptrEncStr = new ACMG722EncStr;
+ if (_ptrEncStr != NULL) {
+ _ptrEncStr->inst = NULL;
+ _ptrEncStr->instRight = NULL;
+ }
+ // Decoder
+ _ptrDecStr = new ACMG722DecStr;
+ if (_ptrDecStr != NULL) {
+ _ptrDecStr->inst = NULL;
+ _ptrDecStr->instRight = NULL; // Not used
+ }
+ _codecID = codecID;
+ return;
+}
+
+ACMG722::~ACMG722() {
+ // Encoder
+ if (_ptrEncStr != NULL) {
+ if (_ptrEncStr->inst != NULL) {
+ WebRtcG722_FreeEncoder(_ptrEncStr->inst);
+ _ptrEncStr->inst = NULL;
+ }
+ if (_ptrEncStr->instRight != NULL) {
+ WebRtcG722_FreeEncoder(_ptrEncStr->instRight);
+ _ptrEncStr->instRight = NULL;
+ }
+ delete _ptrEncStr;
+ _ptrEncStr = NULL;
+ }
+ // Decoder
+ if (_ptrDecStr != NULL) {
+ if (_ptrDecStr->inst != NULL) {
+ WebRtcG722_FreeDecoder(_ptrDecStr->inst);
+ _ptrDecStr->inst = NULL;
+ }
+ if (_ptrDecStr->instRight != NULL) {
+ WebRtcG722_FreeDecoder(_ptrDecStr->instRight);
+ _ptrDecStr->instRight = NULL;
+ }
+ delete _ptrDecStr;
+ _ptrDecStr = NULL;
+ }
+ return;
+}
+
+WebRtc_Word32 ACMG722::Add10MsDataSafe(const WebRtc_UWord32 timestamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 lengthSmpl,
+ const WebRtc_UWord8 audioChannel) {
+ return ACMGenericCodec::Add10MsDataSafe((timestamp >> 1), data, lengthSmpl,
+ audioChannel);
+}
+
+WebRtc_Word16 ACMG722::InternalEncode(WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) {
+ // If stereo, split input signal in left and right channel before encoding
+ if (_noChannels == 2) {
+ WebRtc_Word16 leftChannel[960];
+ WebRtc_Word16 rightChannel[960];
+ WebRtc_UWord8 outLeft[480];
+ WebRtc_UWord8 outRight[480];
+ WebRtc_Word16 lenInBytes;
+ for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
+ leftChannel[j] = _inAudio[_inAudioIxRead + i];
+ rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
+ }
+ lenInBytes = WebRtcG722_Encode(_encoderInstPtr, leftChannel, _frameLenSmpl,
+ (WebRtc_Word16*) outLeft);
+ lenInBytes += WebRtcG722_Encode(_encoderInstPtrRight, rightChannel,
+ _frameLenSmpl, (WebRtc_Word16*) outRight);
+ *bitStreamLenByte = lenInBytes;
+
+ // Interleave the 4 bits per sample from left and right channel
+ for (int i = 0, j = 0; i < lenInBytes; i += 2, j++) {
+ bitStream[i] = (outRight[j] & 0xF0) + (outLeft[j] >> 4);
+ bitStream[i + 1] = ((outRight[j] & 0x0F) << 4) + (outLeft[j] & 0x0F);
+ }
+ } else {
+ *bitStreamLenByte = WebRtcG722_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead],
+ _frameLenSmpl,
+ (WebRtc_Word16*) bitStream);
+ }
+
+ // increment the read index this tell the caller how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl * _noChannels;
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMG722::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+ if (codecParams->codecInstant.channels == 2) {
+ // Create codec struct for right channel
+ if (_ptrEncStr->instRight == NULL) {
+ WebRtcG722_CreateEncoder(&_ptrEncStr->instRight);
+ if (_ptrEncStr->instRight == NULL) {
+ return -1;
+ }
+ }
+ _encoderInstPtrRight = _ptrEncStr->instRight;
+ if (WebRtcG722_EncoderInit(_encoderInstPtrRight) < 0) {
+ return -1;
+ }
+ }
+
+ return WebRtcG722_EncoderInit(_encoderInstPtr);
+}
+
+WebRtc_Word16 ACMG722::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return WebRtcG722_DecoderInit(_decoderInstPtr);
+}
+
+WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ // TODO: log error
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_G722_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderG722, codecInst.pltype, _decoderInstPtr,
+ 16000);
+ SET_G722_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec* ACMG722::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722::InternalCreateEncoder() {
+ if (_ptrEncStr == NULL) {
+ // this structure must be created at the costructor
+ // if it is still NULL then there is a probelm and
+ // we dont continue
+ return -1;
+ }
+ WebRtcG722_CreateEncoder(&_ptrEncStr->inst);
+ if (_ptrEncStr->inst == NULL) {
+ return -1;
+ }
+ _encoderInstPtr = _ptrEncStr->inst;
+ return 0;
+}
+
+void ACMG722::DestructEncoderSafe() {
+ if (_ptrEncStr != NULL) {
+ if (_ptrEncStr->inst != NULL) {
+ WebRtcG722_FreeEncoder(_ptrEncStr->inst);
+ _ptrEncStr->inst = NULL;
+ }
+ }
+ _encoderExist = false;
+ _encoderInitialized = false;
+}
+
+WebRtc_Word16 ACMG722::InternalCreateDecoder() {
+ if (_ptrDecStr == NULL) {
+ // this structure must be created at the costructor
+ // if it is still NULL then there is a probelm and
+ // we dont continue
+ return -1;
+ }
+
+ WebRtcG722_CreateDecoder(&_ptrDecStr->inst);
+ if (_ptrDecStr->inst == NULL) {
+ return -1;
+ }
+ _decoderInstPtr = _ptrDecStr->inst;
+ return 0;
+}
+
+void ACMG722::DestructDecoderSafe() {
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if (_ptrDecStr != NULL) {
+ if (_ptrDecStr->inst != NULL) {
+ WebRtcG722_FreeDecoder(_ptrDecStr->inst);
+ _ptrDecStr->inst = NULL;
+ }
+ }
+}
+
+void ACMG722::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ WebRtcG722_FreeEncoder(static_cast<G722EncInst*>(ptrInst));
+ }
+ return;
+}
+
+WebRtc_Word16 ACMG722::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not "
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderG722);
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g722.h b/trunk/src/modules/audio_coding/main/source/acm_g722.h
new file mode 100644
index 0000000..776d188
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g722.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
+
+#include "acm_generic_codec.h"
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+// forward declaration
+struct ACMG722EncStr;
+struct ACMG722DecStr;
+
+class ACMG722: public ACMGenericCodec {
+ public:
+ ACMG722(WebRtc_Word16 codecID);
+ ~ACMG722();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ WebRtc_Word32 Add10MsDataSafe(const WebRtc_UWord32 timestamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 lengthSmpl,
+ const WebRtc_UWord8 audioChannel);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ ACMG722EncStr* _ptrEncStr;
+ ACMG722DecStr* _ptrDecStr;
+
+ G722EncInst* _encoderInstPtr;
+ G722EncInst* _encoderInstPtrRight; // Prepared for stereo
+ G722DecInst* _decoderInstPtr;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221.cc b/trunk/src/modules/audio_coding/main/source/acm_g7221.cc
new file mode 100644
index 0000000..b82d931
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7221.cc
@@ -0,0 +1,528 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_g7221.h"
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_G722_1
+// NOTE! G.722.1 is not included in the open-source package. The following
+// interface file is needed:
+//
+// /modules/audio_coding/codecs/g7221/main/interface/g7221_interface.h
+//
+// The API in the header file should match the one below.
+//
+// int16_t WebRtcG7221_CreateEnc16(G722_1_16_encinst_t_** encInst);
+// int16_t WebRtcG7221_CreateEnc24(G722_1_24_encinst_t_** encInst);
+// int16_t WebRtcG7221_CreateEnc32(G722_1_32_encinst_t_** encInst);
+// int16_t WebRtcG7221_CreateDec16(G722_1_16_decinst_t_** decInst);
+// int16_t WebRtcG7221_CreateDec24(G722_1_24_decinst_t_** decInst);
+// int16_t WebRtcG7221_CreateDec32(G722_1_32_decinst_t_** decInst);
+//
+// int16_t WebRtcG7221_FreeEnc16(G722_1_16_encinst_t_** encInst);
+// int16_t WebRtcG7221_FreeEnc24(G722_1_24_encinst_t_** encInst);
+// int16_t WebRtcG7221_FreeEnc32(G722_1_32_encinst_t_** encInst);
+// int16_t WebRtcG7221_FreeDec16(G722_1_16_decinst_t_** decInst);
+// int16_t WebRtcG7221_FreeDec24(G722_1_24_decinst_t_** decInst);
+// int16_t WebRtcG7221_FreeDec32(G722_1_32_decinst_t_** decInst);
+//
+// int16_t WebRtcG7221_EncoderInit16(G722_1_16_encinst_t_* encInst);
+// int16_t WebRtcG7221_EncoderInit24(G722_1_24_encinst_t_* encInst);
+// int16_t WebRtcG7221_EncoderInit32(G722_1_32_encinst_t_* encInst);
+// int16_t WebRtcG7221_DecoderInit16(G722_1_16_decinst_t_* decInst);
+// int16_t WebRtcG7221_DecoderInit24(G722_1_24_decinst_t_* decInst);
+// int16_t WebRtcG7221_DecoderInit32(G722_1_32_decinst_t_* decInst);
+//
+// int16_t WebRtcG7221_Encode16(G722_1_16_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221_Encode24(G722_1_24_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221_Encode32(G722_1_32_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+//
+// int16_t WebRtcG7221_Decode16(G722_1_16_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221_Decode24(G722_1_24_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221_Decode32(G722_1_32_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+//
+// int16_t WebRtcG7221_DecodePlc16(G722_1_16_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// int16_t WebRtcG7221_DecodePlc24(G722_1_24_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// int16_t WebRtcG7221_DecodePlc32(G722_1_32_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// void WebRtcG7221_Version(char *versionStr, short len);
+#include "g7221_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_G722_1
+
+ACMG722_1::ACMG722_1(WebRtc_Word16 /* codecID */)
+ : _operationalRate(-1),
+ _encoderInstPtr(NULL),
+ _encoderInstPtrRight(NULL),
+ _decoderInstPtr(NULL),
+ _encoderInst16Ptr(NULL),
+ _encoderInst16PtrR(NULL),
+ _encoderInst24Ptr(NULL),
+ _encoderInst24PtrR(NULL),
+ _encoderInst32Ptr(NULL),
+ _encoderInst32PtrR(NULL),
+ _decoderInst16Ptr(NULL),
+ _decoderInst24Ptr(NULL),
+ _decoderInst32Ptr(NULL) {
+ return;
+}
+
+ACMG722_1::~ACMG722_1() {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1::InternalEncode(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMG722_1::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMG722_1::DestructEncoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMG722_1::DestructDecoderSafe() {
+ return;
+}
+
+void ACMG722_1::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */, WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+ACMG722_1::ACMG722_1(
+ WebRtc_Word16 codecID):
+ _encoderInstPtr(NULL),
+ _encoderInstPtrRight(NULL),
+ _decoderInstPtr(NULL),
+ _encoderInst16Ptr(NULL),
+ _encoderInst16PtrR(NULL),
+ _encoderInst24Ptr(NULL),
+ _encoderInst24PtrR(NULL),
+ _encoderInst32Ptr(NULL),
+ _encoderInst32PtrR(NULL),
+ _decoderInst16Ptr(NULL),
+ _decoderInst24Ptr(NULL),
+ _decoderInst32Ptr(NULL) {
+ _codecID = codecID;
+ if (_codecID == ACMCodecDB::kG722_1_16) {
+ _operationalRate = 16000;
+ } else if (_codecID == ACMCodecDB::kG722_1_24) {
+ _operationalRate = 24000;
+ } else if (_codecID == ACMCodecDB::kG722_1_32) {
+ _operationalRate = 32000;
+ } else {
+ _operationalRate = -1;
+ }
+ return;
+}
+
+ACMG722_1::~ACMG722_1() {
+ if (_encoderInstPtr != NULL) {
+ delete _encoderInstPtr;
+ _encoderInstPtr = NULL;
+ }
+ if (_encoderInstPtrRight != NULL) {
+ delete _encoderInstPtrRight;
+ _encoderInstPtrRight = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ delete _decoderInstPtr;
+ _decoderInstPtr = NULL;
+ }
+
+ switch (_operationalRate) {
+ case 16000: {
+ _encoderInst16Ptr = NULL;
+ _encoderInst16PtrR = NULL;
+ _decoderInst16Ptr = NULL;
+ break;
+ }
+ case 24000: {
+ _encoderInst24Ptr = NULL;
+ _encoderInst24PtrR = NULL;
+ _decoderInst24Ptr = NULL;
+ break;
+ }
+ case 32000: {
+ _encoderInst32Ptr = NULL;
+ _encoderInst32PtrR = NULL;
+ _decoderInst32Ptr = NULL;
+ break;
+ }
+ default: {
+ break;
+ }
+ }
+ return;
+}
+
+WebRtc_Word16 ACMG722_1::InternalEncode(WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) {
+ WebRtc_Word16 leftChannel[320];
+ WebRtc_Word16 rightChannel[320];
+ WebRtc_Word16 lenInBytes;
+ WebRtc_Word16 outB[160];
+
+ // If stereo, split input signal in left and right channel before encoding
+ if (_noChannels == 2) {
+ for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
+ leftChannel[j] = _inAudio[_inAudioIxRead + i];
+ rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
+ }
+ } else {
+ memcpy(leftChannel, &_inAudio[_inAudioIxRead], 320);
+ }
+
+ switch (_operationalRate) {
+ case 16000: {
+ Inst lenInBytes = WebRtcG7221_Encode16(_encoderInst16Ptr, leftChannel,
+ 320, &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221_Encode16(_encoderInst16PtrR, rightChannel,
+ 320, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ case 24000: {
+ lenInBytes = WebRtcG7221_Encode24(_encoderInst24Ptr, leftChannel, 320,
+ &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221_Encode24(_encoderInst24PtrR, rightChannel,
+ 320, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ case 32000: {
+ lenInBytes = WebRtcG7221_Encode32(_encoderInst32Ptr, leftChannel, 320,
+ &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221_Encode32(_encoderInst32PtrR, rightChannel,
+ 320, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitEncode: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+ memcpy(bitStream, outB, lenInBytes);
+ *bitStreamLenByte = lenInBytes;
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += 320 * _noChannels;
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMG722_1::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams) {
+ WebRtc_Word16 ret;
+
+ switch (_operationalRate) {
+ case 16000: {
+ ret = WebRtcG7221_EncoderInit16(_encoderInst16PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221_EncoderInit16(_encoderInst16Ptr);
+ }
+ case 24000: {
+ ret = WebRtcG7221_EncoderInit24(_encoderInst24PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221_EncoderInit24(_encoderInst24Ptr);
+ }
+ case 32000: {
+ ret = WebRtcG7221_EncoderInit32(_encoderInst32PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221_EncoderInit32(_encoderInst32Ptr);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError,Inst webrtc::kTraceAudioCoding,
+ _uniqueID, "InternalInitEncoder: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+}
+
+WebRtc_Word16 ACMG722_1::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ switch (_operationalRate) {
+ case 16000: {
+ return WebRtcG7221_DecoderInit16(_decoderInst16Ptr);
+ }
+ case 24000: {
+ return WebRtcG7221_DecoderInit24(_decoderInst24Ptr);
+ }
+ case 32000: {
+ return WebRtcG7221_DecoderInit32(_decoderInst32Ptr);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+}
+
+WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ // Todo:
+ // log error
+ return -1;
+ }
+ // NetEq has an array of pointers to WebRtcNetEQ_CodecDef.
+ // Get an entry of that array (neteq wrapper will allocate memory)
+ // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would
+ // be the index of the entry.
+ // Fill up the given structure by calling
+ // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION."
+ // Then return the structure back to NetEQ to add the codec to it's
+ // database.
+ switch (_operationalRate) {
+ case 16000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1_16, codecInst.pltype,
+ _decoderInst16Ptr, 16000);
+ SET_G722_1_16_FUNCTIONS((codecDef));
+ break;
+ }
+ case 24000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1_24, codecInst.pltype,
+ _decoderInst24Ptr, 16000);
+ SET_G722_1_24_FUNCTIONS((codecDef));
+ break;
+ }
+ case 32000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1_32, codecInst.pltype,
+ _decoderInst32Ptr, 16000);
+ SET_G722_1_32_FUNCTIONS((codecDef));
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodecDef: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+ACMGenericCodec* ACMG722_1::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
+ if ((_encoderInstPtr == NULL) || (_encoderInstPtrRight == NULL)) {
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 16000: {
+ WebRtcG7221_CreateEnc16(&_encoderInst16Ptr);
+ WebRtcG7221_CreateEnc16(&_encoderInst16PtrR);
+ break;
+ }
+ case 24000: {
+ WebRtcG7221_CreateEnc24(&_encoderInst24Ptr);
+ WebRtcG7221_CreateEnc24(&_encoderInst24PtrR);
+ break;
+ }
+ case 32000: {
+ WebRtcG7221_CreateEnc32(&_encoderInst32Ptr);
+ WebRtcG7221_CreateEnc32(&_encoderInst32PtrR);
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMG722_1::DestructEncoderSafe() {
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if (_encoderInstPtr != NULL) {
+ delete _encoderInstPtr;
+ _encoderInstPtr = NULL;
+ }
+ if (_encoderInstPtrRight != NULL) {
+ delete _encoderInstPtrRight;
+ _encoderInstPtrRight = NULL;
+ }
+ _encoderInst16Ptr = NULL;
+ _encoderInst24Ptr = NULL;
+ _encoderInst32Ptr = NULL;
+}
+
+WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
+ if (_decoderInstPtr == NULL) {
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 16000: {
+ WebRtcG7221_CreateDec16(&_decoderInst16Ptr);
+ break;
+ }
+ case 24000: {
+ WebRtcG7221_CreateDec24(&_decoderInst24Ptr);
+ break;
+ }
+ case 32000: {
+ WebRtcG7221_CreateDec32(&_decoderInst32Ptr);
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMG722_1::DestructDecoderSafe() {
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if (_decoderInstPtr != NULL) {
+ delete _decoderInstPtr;
+ _decoderInstPtr = NULL;
+ }
+ _decoderInst16Ptr = NULL;
+ _decoderInst24Ptr = NULL;
+ _decoderInst32Ptr = NULL;
+}
+
+void ACMG722_1::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ delete ptrInst;
+ }
+ return;
+}
+
+WebRtc_Word16 ACMG722_1::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not "
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 16000: {
+ return netEq->RemoveCodec(kDecoderG722_1_16);
+ }
+ case 24000: {
+ return netEq->RemoveCodec(kDecoderG722_1_24);
+ }
+ case 32000: {
+ return netEq->RemoveCodec(kDecoderG722_1_32);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "UnregisterFromNetEqSafe: Wrong rate for G722_1.");
+ return -1;
+ }
+ }
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221.h b/trunk/src/modules/audio_coding/main/source/acm_g7221.h
new file mode 100644
index 0000000..3130afd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7221.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct G722_1_16_encinst_t_;
+struct G722_1_16_decinst_t_;
+struct G722_1_24_encinst_t_;
+struct G722_1_24_decinst_t_;
+struct G722_1_32_encinst_t_;
+struct G722_1_32_decinst_t_;
+struct G722_1_Inst_t_;
+
+namespace webrtc {
+
+class ACMG722_1: public ACMGenericCodec {
+ public:
+ ACMG722_1(WebRtc_Word16 codecID);
+ ~ACMG722_1();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ protected:
+ WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ WebRtc_Word32 _operationalRate;
+
+ G722_1_Inst_t_* _encoderInstPtr;
+ G722_1_Inst_t_* _encoderInstPtrRight; //Used in stereo mode
+ G722_1_Inst_t_* _decoderInstPtr;
+
+ // Only one set of these pointer is valid at any instance
+ G722_1_16_encinst_t_* _encoderInst16Ptr;
+ G722_1_16_encinst_t_* _encoderInst16PtrR;
+ G722_1_24_encinst_t_* _encoderInst24Ptr;
+ G722_1_24_encinst_t_* _encoderInst24PtrR;
+ G722_1_32_encinst_t_* _encoderInst32Ptr;
+ G722_1_32_encinst_t_* _encoderInst32PtrR;
+
+ // Only one of these pointer is valid at any instance
+ G722_1_16_decinst_t_* _decoderInst16Ptr;
+ G722_1_24_decinst_t_* _decoderInst24Ptr;
+ G722_1_32_decinst_t_* _decoderInst32Ptr;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc b/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc
new file mode 100644
index 0000000..920965e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc
@@ -0,0 +1,532 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_g7221c.h"
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+#include "trace.h"
+
+#ifdef WEBRTC_CODEC_G722_1C
+// NOTE! G.722.1C is not included in the open-source package. The following
+// interface file is needed:
+//
+// /modules/audio_coding/codecs/g7221c/main/interface/g7221c_interface.h
+//
+// The API in the header file should match the one below.
+//
+// int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** encInst);
+// int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** encInst);
+// int16_t WebRtcG7221C_CreateEnc48(G722_1C_48_encinst_t_** encInst);
+// int16_t WebRtcG7221C_CreateDec24(G722_1C_24_decinst_t_** decInst);
+// int16_t WebRtcG7221C_CreateDec32(G722_1C_32_decinst_t_** decInst);
+// int16_t WebRtcG7221C_CreateDec48(G722_1C_48_decinst_t_** decInst);
+//
+// int16_t WebRtcG7221C_FreeEnc24(G722_1C_24_encinst_t_** encInst);
+// int16_t WebRtcG7221C_FreeEnc32(G722_1C_32_encinst_t_** encInst);
+// int16_t WebRtcG7221C_FreeEnc48(G722_1C_48_encinst_t_** encInst);
+// int16_t WebRtcG7221C_FreeDec24(G722_1C_24_decinst_t_** decInst);
+// int16_t WebRtcG7221C_FreeDec32(G722_1C_32_decinst_t_** decInst);
+// int16_t WebRtcG7221C_FreeDec48(G722_1C_48_decinst_t_** decInst);
+//
+// int16_t WebRtcG7221C_EncoderInit24(G722_1C_24_encinst_t_* encInst);
+// int16_t WebRtcG7221C_EncoderInit32(G722_1C_32_encinst_t_* encInst);
+// int16_t WebRtcG7221C_EncoderInit48(G722_1C_48_encinst_t_* encInst);
+// int16_t WebRtcG7221C_DecoderInit24(G722_1C_24_decinst_t_* decInst);
+// int16_t WebRtcG7221C_DecoderInit32(G722_1C_32_decinst_t_* decInst);
+// int16_t WebRtcG7221C_DecoderInit48(G722_1C_48_decinst_t_* decInst);
+//
+// int16_t WebRtcG7221C_Encode24(G722_1C_24_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221C_Encode32(G722_1C_32_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221C_Encode48(G722_1C_48_encinst_t_* encInst,
+// int16_t* input,
+// int16_t len,
+// int16_t* output);
+//
+// int16_t WebRtcG7221C_Decode24(G722_1C_24_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221C_Decode32(G722_1C_32_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+// int16_t WebRtcG7221C_Decode48(G722_1C_48_decinst_t_* decInst,
+// int16_t* bitstream,
+// int16_t len,
+// int16_t* output);
+//
+// int16_t WebRtcG7221C_DecodePlc24(G722_1C_24_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// int16_t WebRtcG7221C_DecodePlc32(G722_1C_32_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* decInst,
+// int16_t* output,
+// int16_t nrLostFrames);
+// void WebRtcG7221C_Version(char *versionStr, short len);
+#include "g7221c_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_G722_1C
+
+ACMG722_1C::ACMG722_1C(WebRtc_Word16 /* codecID */)
+ : _operationalRate(-1),
+ _encoderInstPtr(NULL),
+ _encoderInstPtrRight(NULL),
+ _decoderInstPtr(NULL),
+ _encoderInst24Ptr(NULL),
+ _encoderInst24PtrR(NULL),
+ _encoderInst32Ptr(NULL),
+ _encoderInst32PtrR(NULL),
+ _encoderInst48Ptr(NULL),
+ _encoderInst48PtrR(NULL),
+ _decoderInst24Ptr(NULL),
+ _decoderInst32Ptr(NULL),
+ _decoderInst48Ptr(NULL) {
+ return;
+}
+
+ACMG722_1C::~ACMG722_1C() {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
+}
+
+WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
+}
+
+ACMGenericCodec* ACMG722_1C::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
+ return -1;
+}
+
+void ACMG722_1C::DestructEncoderSafe() {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
+ return -1;
+}
+
+void ACMG722_1C::DestructDecoderSafe() {
+ return;
+}
+
+void ACMG722_1C::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
+}
+
+WebRtc_Word16 ACMG722_1C::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */) {
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+ACMG722_1C::ACMG722_1C(WebRtc_Word16 codecID) :
+ _encoderInstPtr(NULL), _encoderInstPtrRight(NULL), _decoderInstPtr(NULL),
+ _encoderInst24Ptr(NULL), _encoderInst24PtrR(NULL), _encoderInst32Ptr(NULL),
+ _encoderInst32PtrR(NULL), _encoderInst48Ptr(NULL), _encoderInst48PtrR(NULL),
+ _decoderInst24Ptr(NULL), _decoderInst32Ptr(NULL), _decoderInst48Ptr(NULL) {
+ _codecID = codecID;
+ if (_codecID == ACMCodecDB::kG722_1C_24) {
+ _operationalRate = 24000;
+ } else if (_codecID == ACMCodecDB::kG722_1C_32) {
+ _operationalRate = 32000;
+ } else if (_codecID == ACMCodecDB::kG722_1C_48) {
+ _operationalRate = 48000;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Wrong codec id for G722_1c.");
+ _operationalRate = -1;
+ }
+ return;
+}
+
+ACMG722_1C::~ACMG722_1C() {
+ if (_encoderInstPtr != NULL) {
+ delete _encoderInstPtr;
+ _encoderInstPtr = NULL;
+ }
+ if (_encoderInstPtrRight != NULL) {
+ delete _encoderInstPtrRight;
+ _encoderInstPtrRight = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ delete _decoderInstPtr;
+ _decoderInstPtr = NULL;
+ }
+
+ switch (_operationalRate) {
+ case 24000: {
+ _encoderInst24Ptr = NULL;
+ _encoderInst24PtrR = NULL;
+ _decoderInst24Ptr = NULL;
+ break;
+ }
+ case 32000: {
+ _encoderInst32Ptr = NULL;
+ _encoderInst32PtrR = NULL;
+ _decoderInst32Ptr = NULL;
+ break;
+ }
+ case 48000: {
+ _encoderInst48Ptr = NULL;
+ _encoderInst48PtrR = NULL;
+ _decoderInst48Ptr = NULL;
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Wrong rate for G722_1c.");
+ break;
+ }
+ }
+ return;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalEncode(WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) {
+ WebRtc_Word16 leftChannel[640];
+ WebRtc_Word16 rightChannel[640];
+ WebRtc_Word16 lenInBytes;
+ WebRtc_Word16 outB[240];
+
+ // If stereo, split input signal in left and right channel before encoding
+ if (_noChannels == 2) {
+ for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
+ leftChannel[j] = _inAudio[_inAudioIxRead + i];
+ rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
+ }
+ } else {
+ memcpy(leftChannel, &_inAudio[_inAudioIxRead], 640);
+ }
+
+ switch (_operationalRate) {
+ case 24000: {
+ lenInBytes = WebRtcG7221C_Encode24(_encoderInst24Ptr, leftChannel, 640,
+ &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221C_Encode24(_encoderInst24PtrR, rightChannel,
+ 640, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ case 32000: {
+ lenInBytes = WebRtcG7221C_Encode32(_encoderInst32Ptr, leftChannel, 640,
+ &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221C_Encode32(_encoderInst32PtrR, rightChannel,
+ 640, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ case 48000: {
+ lenInBytes = WebRtcG7221C_Encode48(_encoderInst48Ptr, leftChannel, 640,
+ &outB[0]);
+ if (_noChannels == 2) {
+ lenInBytes += WebRtcG7221C_Encode48(_encoderInst48PtrR, rightChannel,
+ 640, &outB[lenInBytes / 2]);
+ }
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+
+ memcpy(bitStream, outB, lenInBytes);
+ *bitStreamLenByte = lenInBytes;
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += 640 * _noChannels;
+
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */) {
+ return 0;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams) {
+ WebRtc_Word16 ret;
+
+ switch (_operationalRate) {
+ case 24000: {
+ ret = WebRtcG7221C_EncoderInit24(_encoderInst24PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221C_EncoderInit24(_encoderInst24Ptr);
+ }
+ case 32000: {
+ ret = WebRtcG7221C_EncoderInit32(_encoderInst32PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221C_EncoderInit32(_encoderInst32Ptr);
+ }
+ case 48000: {
+ ret = WebRtcG7221C_EncoderInit48(_encoderInst48PtrR);
+ if (ret < 0) {
+ return ret;
+ }
+ return WebRtcG7221C_EncoderInit48(_encoderInst48Ptr);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitEncode: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+}
+
+WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */) {
+ switch (_operationalRate) {
+ case 24000: {
+ return WebRtcG7221C_DecoderInit24(_decoderInst24Ptr);
+ }
+ case 32000: {
+ return WebRtcG7221C_DecoderInit32(_decoderInst32Ptr);
+ }
+ case 48000: {
+ return WebRtcG7221C_DecoderInit48(_decoderInst48Ptr);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+}
+
+WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+
+ if (!_decoderInitialized) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: decoder not initialized for G722_1c");
+ return -1;
+ }
+ // NetEq has an array of pointers to WebRtcNetEQ_CodecDef.
+ // get an entry of that array (neteq wrapper will allocate memory)
+ // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would
+ // be the index of the entry.
+ // Fill up the given structure by calling
+ // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION."
+ // Then return the structure back to NetEQ to add the codec to it's
+ // database.
+ switch (_operationalRate) {
+ case 24000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1C_24, codecInst.pltype,
+ _decoderInst24Ptr, 32000);
+ SET_G722_1C_24_FUNCTIONS((codecDef));
+ break;
+ }
+ case 32000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1C_32, codecInst.pltype,
+ _decoderInst32Ptr, 32000);
+ SET_G722_1C_32_FUNCTIONS((codecDef));
+ break;
+ }
+ case 48000: {
+ SET_CODEC_PAR((codecDef), kDecoderG722_1C_32, codecInst.pltype,
+ _decoderInst48Ptr, 32000);
+ SET_G722_1C_48_FUNCTIONS((codecDef));
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+ACMGenericCodec*
+ACMG722_1C::CreateInstance(void) {
+ return NULL;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
+ if ((_encoderInstPtr == NULL) || (_encoderInstPtrRight == NULL)) {
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 24000: {
+ WebRtcG7221C_CreateEnc24(&_encoderInst24Ptr);
+ WebRtcG7221C_CreateEnc24(&_encoderInst24PtrR);
+ break;
+ }
+ case 32000: {
+ WebRtcG7221C_CreateEnc32(&_encoderInst32Ptr);
+ WebRtcG7221C_CreateEnc32(&_encoderInst32PtrR);
+ break;
+ }
+ case 48000: {
+ WebRtcG7221C_CreateEnc48(&_encoderInst48Ptr);
+ WebRtcG7221C_CreateEnc48(&_encoderInst48PtrR);
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMG722_1C::DestructEncoderSafe() {
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if (_encoderInstPtr != NULL) {
+ delete _encoderInstPtr;
+ _encoderInstPtr = NULL;
+ }
+ if (_encoderInstPtrRight != NULL) {
+ delete _encoderInstPtrRight;
+ _encoderInstPtrRight = NULL;
+ }
+ _encoderInst24Ptr = NULL;
+ _encoderInst32Ptr = NULL;
+ _encoderInst48Ptr = NULL;
+}
+
+WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
+ if (_decoderInstPtr == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: cannot create decoder");
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 24000: {
+ WebRtcG7221C_CreateDec24(&_decoderInst24Ptr);
+ break;
+ }
+ case 32000: {
+ WebRtcG7221C_CreateDec32(&_decoderInst32Ptr);
+ break;
+ }
+ case 48000: {
+ WebRtcG7221C_CreateDec48(&_decoderInst48Ptr);
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: Wrong rate for G722_1c.");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+void ACMG722_1C::DestructDecoderSafe() {
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if (_decoderInstPtr != NULL) {
+ delete _decoderInstPtr;
+ _decoderInstPtr = NULL;
+ }
+ _decoderInst24Ptr = NULL;
+ _decoderInst32Ptr = NULL;
+ _decoderInst48Ptr = NULL;
+}
+
+void ACMG722_1C::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ delete ptrInst;
+ }
+ return;
+}
+
+WebRtc_Word16 ACMG722_1C::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) {
+ if (payloadType != _decoderParams.codecInstant.pltype) {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceAudioCoding,
+ _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not"
+ "match the stored payload type",
+ _decoderParams.codecInstant.plname, payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ switch (_operationalRate) {
+ case 24000: {
+ return netEq->RemoveCodec(kDecoderG722_1C_24);
+ }
+ case 32000: {
+ return netEq->RemoveCodec(kDecoderG722_1C_32);
+ }
+ case 48000: {
+ return netEq->RemoveCodec(kDecoderG722_1C_48);
+ }
+ default: {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Could not remove codec from NetEQ for G722_1c."
+ "Sampling frequency doesn't match");
+ return -1;
+ }
+ }
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221c.h b/trunk/src/modules/audio_coding/main/source/acm_g7221c.h
new file mode 100644
index 0000000..5129e4c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7221c.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct G722_1C_24_encinst_t_;
+struct G722_1C_24_decinst_t_;
+struct G722_1C_32_encinst_t_;
+struct G722_1C_32_decinst_t_;
+struct G722_1C_48_encinst_t_;
+struct G722_1C_48_decinst_t_;
+struct G722_1_Inst_t_;
+
+namespace webrtc {
+
+class ACMG722_1C : public ACMGenericCodec
+{
+public:
+ ACMG722_1C(WebRtc_Word16 codecID);
+ ~ACMG722_1C();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ WebRtc_Word32 _operationalRate;
+
+ G722_1_Inst_t_* _encoderInstPtr;
+ G722_1_Inst_t_* _encoderInstPtrRight; //Used in stereo mode
+ G722_1_Inst_t_* _decoderInstPtr;
+
+ // Only one set of these pointer is valid at any instance
+ G722_1C_24_encinst_t_* _encoderInst24Ptr;
+ G722_1C_24_encinst_t_* _encoderInst24PtrR;
+ G722_1C_32_encinst_t_* _encoderInst32Ptr;
+ G722_1C_32_encinst_t_* _encoderInst32PtrR;
+ G722_1C_48_encinst_t_* _encoderInst48Ptr;
+ G722_1C_48_encinst_t_* _encoderInst48PtrR;
+
+ // Only one of these pointer is valid at any instance
+ G722_1C_24_decinst_t_* _decoderInst24Ptr;
+ G722_1C_32_decinst_t_* _decoderInst32Ptr;
+ G722_1C_48_decinst_t_* _decoderInst48Ptr;
+};
+
+} // namespace webrtc;
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g729.cc b/trunk/src/modules/audio_coding/main/source/acm_g729.cc
new file mode 100644
index 0000000..bca2d72
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g729.cc
@@ -0,0 +1,546 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_g729.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_G729
+ // NOTE! G.729 is not included in the open-source package. The following
+ // interface file is needed:
+ //
+ // /modules/audio_coding/codecs/g729/main/interface/g729_interface.h
+ //
+ // The API in the header file should match the one below.
+ //
+ // int16_t WebRtcG729_CreateEnc(G729_encinst_t_** inst);
+ // int16_t WebRtcG729_CreateDec(G729_decinst_t_** inst);
+ // int16_t WebRtcG729_FreeEnc(G729_encinst_t_* inst);
+ // int16_t WebRtcG729_FreeDec(G729_decinst_t_* inst);
+ // int16_t WebRtcG729_Encode(G729_encinst_t_* encInst, int16_t* input,
+ // int16_t len, int16_t* output);
+ // int16_t WebRtcG729_EncoderInit(G729_encinst_t_* encInst, int16_t mode);
+ // int16_t WebRtcG729_Decode(G729_decinst_t_* decInst);
+ // int16_t WebRtcG729_DecodeBwe(G729_decinst_t_* decInst, int16_t* input);
+ // int16_t WebRtcG729_DecodePlc(G729_decinst_t_* decInst);
+ // int16_t WebRtcG729_DecoderInit(G729_decinst_t_* decInst);
+ // void WebRtcG729_Version(char *versionStr, short len);
+ #include "g729_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_G729
+
+ACMG729::ACMG729(WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL) {
+ return;
+}
+
+
+ACMG729::~ACMG729()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729::EnableDTX()
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729::DisableDTX()
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMG729::ReplaceInternalDTXSafe(
+ const bool /*replaceInternalDTX*/)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMG729::IsInternalDTXReplacedSafe(
+ bool* /* internalDTXReplaced */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMG729::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMG729::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMG729::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMG729::DestructDecoderSafe()
+{
+ return;
+}
+
+
+void
+ACMG729::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+
+#else //===================== Actual Implementation =======================
+
+ACMG729::ACMG729(
+ WebRtc_Word16 codecID):
+_encoderInstPtr(NULL),
+_decoderInstPtr(NULL)
+{
+ _codecID = codecID;
+ _hasInternalDTX = true;
+ return;
+}
+
+
+ACMG729::~ACMG729()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ // Delete encoder memory
+ WebRtcG729_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ // Delete decoder memory
+ WebRtcG729_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ // Initialize before entering the loop
+ WebRtc_Word16 noEncodedSamples = 0;
+ WebRtc_Word16 tmpLenByte = 0;
+ WebRtc_Word16 vadDecision = 0;
+ *bitStreamLenByte = 0;
+ while(noEncodedSamples < _frameLenSmpl)
+ {
+ // Call G.729 encoder with pointer to encoder memory, input
+ // audio, number of samples and bitsream
+ tmpLenByte = WebRtcG729_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead], 80,
+ (WebRtc_Word16*)(&(bitStream[*bitStreamLenByte])));
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += 80;
+
+ // sanity check
+ if(tmpLenByte < 0)
+ {
+ // error has happened
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+ // increment number of written bytes
+ *bitStreamLenByte += tmpLenByte;
+ switch(tmpLenByte)
+ {
+ case 0:
+ {
+ if(0 == noEncodedSamples)
+ {
+ // this is the first 10 ms in this packet and there is
+ // no data generated, perhaps DTX is enabled and the
+ // codec is not generating any bit-stream for this 10 ms.
+ // we do not continue encoding this frame.
+ return 0;
+ }
+ break;
+ }
+ case 2:
+ {
+ // check if G.729 internal DTX is enabled
+ if(_hasInternalDTX && _dtxEnabled)
+ {
+ vadDecision = 0;
+ for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
+ {
+ _vadLabel[n] = vadDecision;
+ }
+ }
+ // we got a SID and have to send out this packet no matter
+ // how much audio we have encoded
+ return *bitStreamLenByte;
+ }
+ case 10:
+ {
+ vadDecision = 1;
+ // this is a valid length just continue encoding
+ break;
+ }
+ default:
+ {
+ return -1;
+ }
+ }
+
+ // update number of encoded samples
+ noEncodedSamples += 80;
+ }
+
+ // update VAD decision vector
+ if(_hasInternalDTX && !vadDecision && _dtxEnabled)
+ {
+ for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
+ {
+ _vadLabel[n] = vadDecision;
+ }
+ }
+
+ // done encoding, return number of encoded bytes
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMG729::EnableDTX()
+{
+ if(_dtxEnabled)
+ {
+ // DTX already enabled, do nothing
+ return 0;
+ }
+ else if(_encoderExist)
+ {
+ // Re-init the G.729 encoder to turn on DTX
+ if(WebRtcG729_EncoderInit(_encoderInstPtr, 1) < 0)
+ {
+ return -1;
+ }
+ _dtxEnabled = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+WebRtc_Word16
+ACMG729::DisableDTX()
+{
+ if(!_dtxEnabled)
+ {
+ // DTX already dissabled, do nothing
+ return 0;
+ }
+ else if(_encoderExist)
+ {
+ // Re-init the G.729 decoder to turn off DTX
+ if(WebRtcG729_EncoderInit(_encoderInstPtr, 0) < 0)
+ {
+ return -1;
+ }
+ _dtxEnabled = false;
+ return 0;
+ }
+ else
+ {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+
+WebRtc_Word32
+ACMG729::ReplaceInternalDTXSafe(
+ const bool replaceInternalDTX)
+{
+ // This function is used to dissable the G.729 built in DTX and use an
+ // external instead.
+
+ if(replaceInternalDTX == _hasInternalDTX)
+ {
+ // Make sure we keep the DTX/VAD setting if possible
+ bool oldEnableDTX = _dtxEnabled;
+ bool oldEnableVAD = _vadEnabled;
+ ACMVADMode oldMode = _vadMode;
+ if (replaceInternalDTX)
+ {
+ // Disable internal DTX before enabling external DTX
+ DisableDTX();
+ }
+ else
+ {
+ // Disable external DTX before enabling internal
+ ACMGenericCodec::DisableDTX();
+ }
+ _hasInternalDTX = !replaceInternalDTX;
+ WebRtc_Word16 status = SetVADSafe(oldEnableDTX, oldEnableVAD, oldMode);
+ // Check if VAD status has changed from inactive to active, or if error was
+ // reported
+ if (status == 1) {
+ _vadEnabled = true;
+ return status;
+ } else if (status < 0) {
+ _hasInternalDTX = replaceInternalDTX;
+ return -1;
+ }
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMG729::IsInternalDTXReplacedSafe(
+ bool* internalDTXReplaced)
+{
+ // Get status of wether DTX is replaced or not
+ *internalDTXReplaced = !_hasInternalDTX;
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMG729::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ // This function is not used. G.729 decoder is called from inside NetEQ
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ // Init G.729 encoder
+ return WebRtcG729_EncoderInit(_encoderInstPtr,
+ ((codecParams->enableDTX)? 1:0));
+}
+
+
+WebRtc_Word16
+ACMG729::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // Init G.729 decoder
+ return WebRtcG729_DecoderInit(_decoderInstPtr);
+}
+
+
+WebRtc_Word32
+ACMG729::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ // Todo:
+ // log error
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderG729, codecInst.pltype,
+ _decoderInstPtr, 8000);
+ SET_G729_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMG729::CreateInstance(void)
+{
+ // Function not used
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMG729::InternalCreateEncoder()
+{
+ // Create encoder memory
+ return WebRtcG729_CreateEnc(&_encoderInstPtr);
+}
+
+
+void
+ACMG729::DestructEncoderSafe()
+{
+ // Free encoder memory
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcG729_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+}
+
+
+WebRtc_Word16
+ACMG729::InternalCreateDecoder()
+{
+ // Create decoder memory
+ return WebRtcG729_CreateDec(&_decoderInstPtr);
+}
+
+
+void
+ACMG729::DestructDecoderSafe()
+{
+ // Free decoder memory
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcG729_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+}
+
+
+void
+ACMG729::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WebRtcG729_FreeEnc((G729_encinst_t_*)ptrInst);
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ // Remove codec from the NetEQ database
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderG729);
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g729.h b/trunk/src/modules/audio_coding/main/source/acm_g729.h
new file mode 100644
index 0000000..50b648a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g729.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct G729_encinst_t_;
+struct G729_decinst_t_;
+
+namespace webrtc {
+
+class ACMG729 : public ACMGenericCodec
+{
+public:
+ ACMG729(WebRtc_Word16 codecID);
+ ~ACMG729();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 EnableDTX();
+
+ WebRtc_Word16 DisableDTX();
+
+ WebRtc_Word32 ReplaceInternalDTXSafe(
+ const bool replaceInternalDTX);
+
+ WebRtc_Word32 IsInternalDTXReplacedSafe(
+ bool* internalDTXReplaced);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ G729_encinst_t_* _encoderInstPtr;
+ G729_decinst_t_* _decoderInstPtr;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7291.cc b/trunk/src/modules/audio_coding/main/source/acm_g7291.cc
new file mode 100644
index 0000000..d832e9c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7291.cc
@@ -0,0 +1,500 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_g7291.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_G729_1
+ // NOTE! G.729.1 is not included in the open-source package. The following
+ // interface file is needed:
+ //
+ // /modules/audio_coding/codecs/g7291/main/interface/g7291_interface.h
+ //
+ // The API in the header file should match the one below.
+ //
+ // int16_t WebRtcG7291_Create(G729_1_inst_t_** inst);
+ // int16_t WebRtcG7291_Free(G729_1_inst_t_* inst);
+ // int16_t WebRtcG7291_Encode(G729_1_inst_t_* encInst, int16_t* input,
+ // int16_t* output, int16_t myRate,
+ // int16_t nrFrames);
+ // int16_t WebRtcG7291_EncoderInit(G729_1_inst_t_* encInst, int16_t myRate,
+ // int16_t flag8kHz, int16_t flagG729mode);
+ // int16_t WebRtcG7291_Decode(G729_1_inst_t_* decInst);
+ // int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
+ // int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
+ // int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
+ // void WebRtcG7291_Version(char *versionStr, short len);
+ #include "g7291_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_G729_1
+
+ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _myRate(32000),
+ _flag8kHz(0),
+ _flagG729mode(0) {
+ return;
+}
+
+
+ACMG729_1::~ACMG729_1()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729_1::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMG729_1::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMG729_1::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMG729_1::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMG729_1::DestructDecoderSafe()
+{
+ return;
+}
+
+
+void
+ACMG729_1::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729_1::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMG729_1::SetBitRateSafe(
+ const WebRtc_Word32 /*rate*/ )
+{
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+struct G729_1_inst_t_;
+
+ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _myRate(32000), // Default rate.
+ _flag8kHz(0),
+ _flagG729mode(0) {
+ // TODO(tlegrand): We should add codecID as a input variable to the
+ // constructor of ACMGenericCodec.
+ _codecID = codecID;
+ return;
+}
+
+ACMG729_1::~ACMG729_1()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcG7291_Free(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcG7291_Free(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+
+ // Initialize before entering the loop
+ WebRtc_Word16 noEncodedSamples = 0;
+ *bitStreamLenByte = 0;
+
+ WebRtc_Word16 byteLengthFrame = 0;
+
+ // Derive number of 20ms frames per encoded packet.
+ // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
+ WebRtc_Word16 n20msFrames = (_frameLenSmpl / 320);
+ // Byte length for the frame. +1 is for rate information.
+ byteLengthFrame = _myRate/(8*50) * n20msFrames + (1 - _flagG729mode);
+
+ // The following might be revised if we have G729.1 Annex C (support for DTX);
+ do
+ {
+ *bitStreamLenByte = WebRtcG7291_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
+ (WebRtc_Word16*)bitStream, _myRate, n20msFrames);
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += 160;
+
+ // sanity check
+ if(*bitStreamLenByte < 0)
+ {
+ // error has happened
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Encode error for G729_1");
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+ noEncodedSamples += 160;
+ } while(*bitStreamLenByte == 0);
+
+
+ // This criteria will change if we have Annex C.
+ if(*bitStreamLenByte != byteLengthFrame)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Encode error for G729_1");
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+
+ if(noEncodedSamples != _frameLenSmpl)
+ {
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMG729_1::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ //set the bit rate and initialize
+ _myRate = codecParams->codecInstant.rate;
+ return SetBitRateSafe( (WebRtc_UWord32)_myRate);
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: init decoder failed for G729_1");
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMG729_1::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: Decoder uninitialized for G729_1");
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype,
+ _decoderInstPtr, 16000);
+ SET_G729_1_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMG729_1::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalCreateEncoder()
+{
+ if (WebRtcG7291_Create(&_encoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: create encoder failed for G729_1");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMG729_1::DestructEncoderSafe()
+{
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcG7291_Free(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+}
+
+
+WebRtc_Word16
+ACMG729_1::InternalCreateDecoder()
+{
+ if (WebRtcG7291_Create(&_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: create decoder failed for G729_1");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMG729_1::DestructDecoderSafe()
+{
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcG7291_Free(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+}
+
+
+void
+ACMG729_1::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ //WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMG729_1::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec: given payload-type does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderG729_1);
+}
+
+WebRtc_Word16
+ACMG729_1::SetBitRateSafe(
+ const WebRtc_Word32 rate)
+{
+ //allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
+ // 22000, 24000, 26000, 28000, 30000, 32000};
+ // TODO(tlegrand): This check exists in one other place two. Should be
+ // possible to reuse code.
+ switch(rate)
+ {
+ case 8000:
+ {
+ _myRate = 8000;
+ break;
+ }
+ case 12000:
+ {
+ _myRate = 12000;
+ break;
+ }
+ case 14000:
+ {
+ _myRate = 14000;
+ break;
+ }
+ case 16000:
+ {
+ _myRate = 16000;
+ break;
+ }
+ case 18000:
+ {
+ _myRate = 18000;
+ break;
+ }
+ case 20000:
+ {
+ _myRate = 20000;
+ break;
+ }
+ case 22000:
+ {
+ _myRate = 22000;
+ break;
+ }
+ case 24000:
+ {
+ _myRate = 24000;
+ break;
+ }
+ case 26000:
+ {
+ _myRate = 26000;
+ break;
+ }
+ case 28000:
+ {
+ _myRate = 28000;
+ break;
+ }
+ case 30000:
+ {
+ _myRate = 30000;
+ break;
+ }
+ case 32000:
+ {
+ _myRate = 32000;
+ break;
+ }
+ default:
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: Invalid rate G729_1");
+ return -1;
+ }
+ }
+
+ // Re-init with new rate
+ if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz, _flagG729mode) >= 0)
+ {
+ _encoderParams.codecInstant.rate = _myRate;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7291.h b/trunk/src/modules/audio_coding/main/source/acm_g7291.h
new file mode 100644
index 0000000..cb27dc9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_g7291.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct G729_1_inst_t_;
+struct G729_1_inst_t_;
+
+namespace webrtc {
+
+class ACMG729_1: public ACMGenericCodec
+{
+public:
+ ACMG729_1(WebRtc_Word16 codecID);
+ ~ACMG729_1();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ //WebRtc_Word16 EnableDTX();
+ //
+ //WebRtc_Word16 DisableDTX();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 rate);
+
+ G729_1_inst_t_* _encoderInstPtr;
+ G729_1_inst_t_* _decoderInstPtr;
+
+ WebRtc_UWord16 _myRate;
+ WebRtc_Word16 _flag8kHz;
+ WebRtc_Word16 _flagG729mode;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc
new file mode 100644
index 0000000..0e212fb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -0,0 +1,1554 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <string.h>
+
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_generic_codec.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_vad.h"
+#include "webrtc_cng.h"
+
+namespace webrtc
+{
+
+// Enum for CNG
+enum
+{
+ kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
+ kNewCNGNumPLCParams = 8
+};
+
+#define ACM_SID_INTERVAL_MSEC 100
+
+// We set some of the variables to invalid values as a check point
+// if a proper initialization has happened. Another approach is
+// to initialize to a default codec that we are sure is always included.
+ACMGenericCodec::ACMGenericCodec()
+ : _inAudioIxWrite(0),
+ _inAudioIxRead(0),
+ _inTimestampIxWrite(0),
+ _inAudio(NULL),
+ _inTimestamp(NULL),
+ _frameLenSmpl(-1), // invalid value
+ _noChannels(1),
+ _codecID(-1), // invalid value
+ _noMissedSamples(0),
+ _encoderExist(false),
+ _decoderExist(false),
+ _encoderInitialized(false),
+ _decoderInitialized(false),
+ _registeredInNetEq(false),
+ _hasInternalDTX(false),
+ _ptrVADInst(NULL),
+ _vadEnabled(false),
+ _vadMode(VADNormal),
+ _dtxEnabled(false),
+ _ptrDTXInst(NULL),
+ _numLPCParams(kNewCNGNumPLCParams),
+ _sentCNPrevious(false),
+ _isMaster(true),
+ _netEqDecodeLock(NULL),
+ _codecWrapperLock(*RWLockWrapper::CreateRWLock()),
+ _lastEncodedTimestamp(0),
+ _lastTimestamp(0xD87F3F9F),
+ _isAudioBuffFresh(true),
+ _uniqueID(0) {
+ // Initialize VAD vector.
+ for (int i = 0; i < MAX_FRAME_SIZE_10MSEC; i++) {
+ _vadLabel[i] = 0;
+ }
+
+ // Nullify memory for encoder and decoder, and set payload type to an
+ // invalid value.
+ memset(&_encoderParams, 0, sizeof(WebRtcACMCodecParams));
+ _encoderParams.codecInstant.pltype = -1;
+ memset(&_decoderParams, 0, sizeof(WebRtcACMCodecParams));
+ _decoderParams.codecInstant.pltype = -1;
+}
+
+ACMGenericCodec::~ACMGenericCodec()
+{
+ // Check all the members which are pointers and
+ // if they are not NULL delete/free them.
+
+ if(_ptrVADInst != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst);
+ _ptrVADInst = NULL;
+ }
+
+ if (_inAudio != NULL)
+ {
+ delete [] _inAudio;
+ _inAudio = NULL;
+ }
+
+ if (_inTimestamp != NULL)
+ {
+ delete [] _inTimestamp;
+ _inTimestamp = NULL;
+ }
+ if(_ptrDTXInst != NULL)
+ {
+ WebRtcCng_FreeEnc(_ptrDTXInst);
+ _ptrDTXInst = NULL;
+ }
+ delete &_codecWrapperLock;
+}
+
+WebRtc_Word32
+ACMGenericCodec::Add10MsData(
+ const WebRtc_UWord32 timestamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 lengthSmpl,
+ const WebRtc_UWord8 audioChannel)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return Add10MsDataSafe(timestamp, data, lengthSmpl, audioChannel);
+}
+
+WebRtc_Word32
+ACMGenericCodec::Add10MsDataSafe(
+ const WebRtc_UWord32 timestamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 lengthSmpl,
+ const WebRtc_UWord8 audioChannel)
+{
+ // The codec expects to get data in correct sampling rate.
+ // get the sampling frequency of the codec
+ WebRtc_UWord16 plFreqHz;
+
+ if(EncoderSampFreq(plFreqHz) < 0)
+ {
+ // _codecID is not correct, perhaps the codec is not initialized yet.
+ return -1;
+ }
+
+ // Sanity check, if the length of the input corresponds to 10 ms.
+ if((plFreqHz / 100) != lengthSmpl)
+ {
+ // This is not 10 ms of audio, given the sampling frequency of the
+ // codec
+ return -1;
+ }
+ if(_lastTimestamp == timestamp)
+ {
+ // Same timestamp as the last time, overwrite.
+ if((_inAudioIxWrite >= lengthSmpl) && (_inTimestampIxWrite > 0))
+ {
+ _inAudioIxWrite -= lengthSmpl;
+ _inTimestampIxWrite--;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
+ "Adding 10ms with previous timestamp, \
+overwriting the previous 10ms");
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
+ "Adding 10ms with previous timestamp, this will sound bad");
+ }
+ }
+
+ _lastTimestamp = timestamp;
+
+ if ((_inAudioIxWrite + lengthSmpl*audioChannel) > AUDIO_BUFFER_SIZE_W16)
+ {
+ // Get the number of samples to be overwritten
+ WebRtc_Word16 missedSamples = _inAudioIxWrite + lengthSmpl*audioChannel -
+ AUDIO_BUFFER_SIZE_W16;
+
+ // Move the data (overwite the old data)
+ memmove(_inAudio, _inAudio + missedSamples,
+ (AUDIO_BUFFER_SIZE_W16 - lengthSmpl*audioChannel)*sizeof(WebRtc_Word16));
+ // Copy the new data
+ memcpy(_inAudio + (AUDIO_BUFFER_SIZE_W16 - lengthSmpl*audioChannel), data,
+ lengthSmpl*audioChannel * sizeof(WebRtc_Word16));
+
+ // Get the number of 10 ms blocks which are overwritten
+ WebRtc_Word16 missed10MsecBlocks =
+ (WebRtc_Word16)((missedSamples/audioChannel * 100) / plFreqHz);
+
+ // Move the timestamps
+ memmove(_inTimestamp, _inTimestamp + missed10MsecBlocks,
+ (_inTimestampIxWrite - missed10MsecBlocks) * sizeof(WebRtc_UWord32));
+ _inTimestampIxWrite -= missed10MsecBlocks;
+ _inTimestamp[_inTimestampIxWrite] = timestamp;
+ _inTimestampIxWrite++;
+
+ // Buffer is full
+ _inAudioIxWrite = AUDIO_BUFFER_SIZE_W16;
+ IncreaseNoMissedSamples(missedSamples);
+ _isAudioBuffFresh = false;
+ return -missedSamples;
+ }
+ memcpy(_inAudio + _inAudioIxWrite, data, lengthSmpl*audioChannel * sizeof(WebRtc_Word16));
+ _inAudioIxWrite += lengthSmpl*audioChannel;
+
+ assert(_inTimestampIxWrite < TIMESTAMP_BUFFER_SIZE_W32);
+ assert(_inTimestampIxWrite >= 0);
+
+ _inTimestamp[_inTimestampIxWrite] = timestamp;
+ _inTimestampIxWrite++;
+ _isAudioBuffFresh = false;
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::Encode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_UWord32* timeStamp,
+ WebRtcACMEncodingType* encodingType)
+{
+ WriteLockScoped lockCodec(_codecWrapperLock);
+ ReadLockScoped lockNetEq(*_netEqDecodeLock);
+ return EncodeSafe(bitStream, bitStreamLenByte,
+ timeStamp, encodingType);
+}
+
+
+WebRtc_Word16
+ACMGenericCodec::EncodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_UWord32* timeStamp,
+ WebRtcACMEncodingType* encodingType)
+{
+ // Do we have enough data to encode?
+ // we wait until we have a full frame to encode.
+ if(_inAudioIxWrite < _frameLenSmpl*_noChannels)
+ {
+ // There is not enough audio
+ *timeStamp = 0;
+ *bitStreamLenByte = 0;
+ // Doesn't really matter what this parameter set to
+ *encodingType = kNoEncoding;
+ return 0;
+ }
+
+ // Not all codecs accept the whole frame to be pushed into
+ // encoder at once.
+ const WebRtc_Word16 myBasicCodingBlockSmpl =
+ ACMCodecDB::BasicCodingBlock(_codecID);
+ if((myBasicCodingBlockSmpl < 0) ||
+ (!_encoderInitialized) ||
+ (!_encoderExist))
+ {
+ // This should not happen
+ *timeStamp = 0;
+ *bitStreamLenByte = 0;
+ *encodingType = kNoEncoding;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EncodeSafe: error, basic coding sample block is negative");
+ return -1;
+ }
+
+ // This makes the internal encoder read from the begining of the buffer
+ _inAudioIxRead = 0;
+ *timeStamp = _inTimestamp[0];
+
+ // Process the audio through VAD the function doesn't set _vadLabels.
+ // If VAD is disabled all labels are set to ONE (active)
+ WebRtc_Word16 status = 0;
+ WebRtc_Word16 dtxProcessedSamples = 0;
+
+ status = ProcessFrameVADDTX(bitStream, bitStreamLenByte,
+ &dtxProcessedSamples);
+
+ if(status < 0)
+ {
+ *timeStamp = 0;
+ *bitStreamLenByte = 0;
+ *encodingType = kNoEncoding;
+ }
+ else
+ {
+ if(dtxProcessedSamples > 0)
+ {
+ // Dtx have processed some samples may or may not a bit-stream
+ // is generated we should not do any encoding (normally there
+ // will be not enough data)
+
+ // Setting the following makes that the move of audio data
+ // and timestamps happen correctly
+ _inAudioIxRead = dtxProcessedSamples;
+ // This will let the owner of ACMGenericCodec to know that the
+ // generated bit-stream is DTX to use correct payload type
+ WebRtc_UWord16 sampFreqHz;
+ EncoderSampFreq(sampFreqHz);
+ if (sampFreqHz == 8000) {
+ *encodingType = kPassiveDTXNB;
+ } else if (sampFreqHz == 16000) {
+ *encodingType = kPassiveDTXWB;
+ } else if (sampFreqHz == 32000) {
+ *encodingType = kPassiveDTXSWB;
+ } else {
+ status = -1;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EncodeSafe: Wrong sampling frequency for DTX.");
+ }
+
+ // Transport empty frame if we have an empty bitstream
+ if ((*bitStreamLenByte == 0)
+ && (_sentCNPrevious || ((_inAudioIxWrite - _inAudioIxRead) <= 0))
+ )
+ {
+ // Makes sure we transmit an empty frame
+ *bitStreamLenByte = 1;
+ *encodingType = kNoEncoding;
+ }
+ _sentCNPrevious = true;
+ }
+ else
+ {
+ _sentCNPrevious = false;
+ // This will let the caller of the method to know if the frame is
+ // Active or non-Active The caller of the method knows that the
+ // stream is encoded by codec and can use the info for callbacks,
+ // if any registered.
+ if(myBasicCodingBlockSmpl == 0)
+ {
+ // This codec can handle all allowed frame sizes as basic
+ // coding block
+ status = InternalEncode(bitStream, bitStreamLenByte);
+
+ if(status < 0)
+ {
+ // TODO:
+ // Maybe reseting the encoder to be fresh for the next
+ // frame
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EncodeSafe: error in internalEncode");
+ *bitStreamLenByte = 0;
+ *encodingType = kNoEncoding;
+ }
+ }
+ else
+ {
+ // A basic-coding-block for this codec is defined so we loop
+ // over the audio with the steps of the basic-coding-block.
+ // It is not necessary that in each itteration
+ WebRtc_Word16 tmpBitStreamLenByte;
+
+ // Reset the variables which will be increamented in the loop
+ *bitStreamLenByte = 0;
+ bool done = false;
+ while(!done)
+ {
+ status = InternalEncode(&bitStream[*bitStreamLenByte],
+ &tmpBitStreamLenByte);
+ *bitStreamLenByte += tmpBitStreamLenByte;
+
+ // Guard Against errors and too large payloads
+ if((status < 0) ||
+ (*bitStreamLenByte > MAX_PAYLOAD_SIZE_BYTE))
+ {
+ // Error has happened if we are in the middle of a full
+ // frame we have to exit. Before exiting, whatever bits
+ // are in the buffer are probably corruptred. Anyways
+ // we ignore them.
+ *bitStreamLenByte = 0;
+ *encodingType = kNoEncoding;
+ // We might have come here because of the second
+ // condition.
+ status = -1;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding,
+ _uniqueID, "EncodeSafe: error in InternalEncode");
+ // break from the loop
+ break;
+ }
+
+ done = _inAudioIxRead >= _frameLenSmpl;
+ }
+ }
+ if(status >= 0)
+ {
+ *encodingType = (_vadLabel[0] == 1)?
+ kActiveNormalEncoded:kPassiveNormalEncoded;
+ // Transport empty frame if we have an empty bitsteram
+ if ((*bitStreamLenByte == 0) && ((_inAudioIxWrite - _inAudioIxRead) <= 0))
+ {
+ // Makes sure we transmit an empty frame
+ *bitStreamLenByte = 1;
+ *encodingType = kNoEncoding;
+ }
+ }
+ }
+ }
+
+ // Move the timestampe buffer according to the number of 10 ms blocks
+ // which are read.
+ WebRtc_UWord16 sampFreqHz;
+ EncoderSampFreq(sampFreqHz);
+
+ WebRtc_Word16 num10MsecBlocks =
+ (WebRtc_Word16)((_inAudioIxRead/_noChannels * 100) / sampFreqHz);
+ if(_inTimestampIxWrite > num10MsecBlocks)
+ {
+ memmove(_inTimestamp, _inTimestamp + num10MsecBlocks,
+ (_inTimestampIxWrite - num10MsecBlocks) * sizeof(WebRtc_Word32));
+ }
+ _inTimestampIxWrite -= num10MsecBlocks;
+
+ // We have to move the audio that is not encoded to the beginning
+ // of the buffer and accordingly adjust the read and write indices.
+ if(_inAudioIxRead < _inAudioIxWrite)
+ {
+ memmove(_inAudio, &_inAudio[_inAudioIxRead],
+ (_inAudioIxWrite - _inAudioIxRead)*sizeof(WebRtc_Word16));
+ }
+
+ _inAudioIxWrite -= _inAudioIxRead;
+
+ _inAudioIxRead = 0;
+ _lastEncodedTimestamp = *timeStamp;
+ return (status < 0) ? (-1):(*bitStreamLenByte);
+}
+
+WebRtc_Word16
+ACMGenericCodec::Decode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return DecodeSafe(bitStream, bitStreamLenByte, audio,
+ audioSamples, speechType);
+}
+
+bool
+ACMGenericCodec::EncoderInitialized()
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return _encoderInitialized;
+}
+
+bool
+ACMGenericCodec::DecoderInitialized()
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return _decoderInitialized;
+}
+
+
+WebRtc_Word32
+ACMGenericCodec::RegisterInNetEq(
+ ACMNetEQ* netEq,
+ const CodecInst& codecInst)
+{
+ WebRtcNetEQ_CodecDef codecDef;
+ WriteLockScoped wl(_codecWrapperLock);
+
+ if(CodecDef(codecDef, codecInst) < 0)
+ {
+ // Failed to register
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "RegisterInNetEq: error, failed to register");
+ _registeredInNetEq = false;
+ return -1;
+ }
+ else
+ {
+ if(netEq->AddCodec(&codecDef, _isMaster) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "RegisterInNetEq: error, failed to add codec");
+ _registeredInNetEq = false;
+ return -1;
+ }
+ // Registered
+ _registeredInNetEq = true;
+ return 0;
+ }
+}
+
+WebRtc_Word16
+ACMGenericCodec::EncoderParams(
+ WebRtcACMCodecParams* encParams)
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return EncoderParamsSafe(encParams);
+}
+
+WebRtc_Word16
+ACMGenericCodec::EncoderParamsSafe(
+ WebRtcACMCodecParams* encParams)
+{
+ // Codec parameters are valid only if the encoder is initialized
+ if(_encoderInitialized)
+ {
+ WebRtc_Word32 currentRate;
+ memcpy(encParams, &_encoderParams, sizeof(WebRtcACMCodecParams));
+ currentRate = encParams->codecInstant.rate;
+ CurrentRate(currentRate);
+ encParams->codecInstant.rate = currentRate;
+ return 0;
+ }
+ else
+ {
+ encParams->codecInstant.plname[0] = '\0';
+ encParams->codecInstant.pltype = -1;
+ encParams->codecInstant.pacsize = 0;
+ encParams->codecInstant.rate = 0;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EncoderParamsSafe: error, encoder not initialized");
+ return -1;
+ }
+}
+
+bool
+ACMGenericCodec::DecoderParams(
+ WebRtcACMCodecParams* decParams,
+ const WebRtc_UWord8 payloadType)
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return DecoderParamsSafe(decParams, payloadType);
+}
+
+bool
+ACMGenericCodec::DecoderParamsSafe(
+ WebRtcACMCodecParams* decParams,
+ const WebRtc_UWord8 payloadType)
+{
+ // Decoder parameters are valid only if decoder is initialized
+ if(_decoderInitialized)
+ {
+ if(payloadType == _decoderParams.codecInstant.pltype)
+ {
+ memcpy(decParams, &_decoderParams, sizeof(WebRtcACMCodecParams));
+ return true;
+ }
+ }
+
+ decParams->codecInstant.plname[0] = '\0';
+ decParams->codecInstant.pltype = -1;
+ decParams->codecInstant.pacsize = 0;
+ decParams->codecInstant.rate = 0;
+ return false;
+}
+
+WebRtc_Word16
+ACMGenericCodec::ResetEncoder()
+{
+ WriteLockScoped lockCodec(_codecWrapperLock);
+ ReadLockScoped lockNetEq(*_netEqDecodeLock);
+ return ResetEncoderSafe();
+}
+
+WebRtc_Word16
+ACMGenericCodec::ResetEncoderSafe()
+{
+ if(!_encoderExist || !_encoderInitialized)
+ {
+ // We don't reset if doesn't exists or not initialized yet
+ return 0;
+ }
+
+ _inAudioIxWrite = 0;
+ _inAudioIxRead = 0;
+ _inTimestampIxWrite = 0;
+ _noMissedSamples = 0;
+ _isAudioBuffFresh = true;
+ memset(_inAudio, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
+ memset(_inTimestamp, 0, TIMESTAMP_BUFFER_SIZE_W32 * sizeof(WebRtc_Word32));
+
+ // Store DTX/VAD params
+ bool enableVAD = _vadEnabled;
+ bool enableDTX = _dtxEnabled;
+ ACMVADMode mode = _vadMode;
+
+ // Reset the encoder
+ if(InternalResetEncoder() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "ResetEncoderSafe: error in reset encoder");
+ return -1;
+ }
+
+ // Disable DTX & VAD this deletes the states
+ // we like to have fresh start
+ DisableDTX();
+ DisableVAD();
+
+ // Set DTX/VAD
+ return SetVADSafe(enableDTX, enableVAD, mode);
+}
+
+WebRtc_Word16
+ACMGenericCodec::InternalResetEncoder()
+{
+ // For most of the codecs it is sufficient to
+ // call their internal initialization.
+ // There are some exceptions.
+ // ----
+ // For iSAC we don't want to lose BWE history,
+ // so for iSAC we have to over-write this function.
+ // ----
+ return InternalInitEncoder(&_encoderParams);
+}
+
+WebRtc_Word16
+ACMGenericCodec::InitEncoder(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization)
+{
+ WriteLockScoped lockCodec(_codecWrapperLock);
+ ReadLockScoped lockNetEq(*_netEqDecodeLock);
+ return InitEncoderSafe(codecParams, forceInitialization);
+}
+
+WebRtc_Word16
+ACMGenericCodec::InitEncoderSafe(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization)
+{
+ // Check if we got a valid set of parameters
+ int mirrorID;
+ int codecNumber =
+ ACMCodecDB::CodecNumber(&(codecParams->codecInstant), &mirrorID);
+
+ if(codecNumber < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitEncoderSafe: error, codec number negative");
+ return -1;
+ }
+ // Check if the parameters are for this codec
+ if((_codecID >= 0) && (_codecID != codecNumber) && (_codecID != mirrorID))
+ {
+ // The current codec is not the same as the one given by codecParams
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitEncoderSafe: current codec is not the same as the one given by codecParams");
+ return -1;
+ }
+
+ if(!CanChangeEncodingParam(codecParams->codecInstant))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitEncoderSafe: cannot change encoding parameters");
+ return -1;
+ }
+
+ if(_encoderInitialized && !forceInitialization)
+ {
+ // The encoder is already initialized
+ return 0;
+ }
+ WebRtc_Word16 status;
+ if(!_encoderExist)
+ {
+ _encoderInitialized = false;
+ status = CreateEncoder();
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitEncoderSafe: cannot create encoder");
+ return -1;
+ }
+ else
+ {
+ _encoderExist = true;
+ }
+ }
+ _frameLenSmpl = (codecParams->codecInstant).pacsize;
+ _noChannels = codecParams->codecInstant.channels;
+ status = InternalInitEncoder(codecParams);
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitEncoderSafe: error in init encoder");
+ _encoderInitialized = false;
+ return -1;
+ }
+ else
+ {
+ memcpy(&_encoderParams, codecParams, sizeof(WebRtcACMCodecParams));
+ _encoderInitialized = true;
+ if(_inAudio == NULL)
+ {
+ _inAudio = new WebRtc_Word16[AUDIO_BUFFER_SIZE_W16];
+ if(_inAudio == NULL)
+ {
+ return -1;
+ }
+ memset(_inAudio, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
+ }
+ if(_inTimestamp == NULL)
+ {
+ _inTimestamp = new WebRtc_UWord32[TIMESTAMP_BUFFER_SIZE_W32];
+ if(_inTimestamp == NULL)
+ {
+ return -1;
+ }
+ memset(_inTimestamp, 0, sizeof(WebRtc_UWord32) *
+ TIMESTAMP_BUFFER_SIZE_W32);
+ }
+ _isAudioBuffFresh = true;
+ }
+ status = SetVADSafe(codecParams->enableDTX, codecParams->enableVAD,
+ codecParams->vadMode);
+
+ return status;
+}
+
+bool
+ACMGenericCodec::CanChangeEncodingParam(
+ CodecInst& /*codecInst*/)
+{
+ return true;
+}
+
+WebRtc_Word16
+ACMGenericCodec::InitDecoder(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization)
+{
+ WriteLockScoped lockCodc(_codecWrapperLock);
+ WriteLockScoped lockNetEq(*_netEqDecodeLock);
+ return InitDecoderSafe(codecParams, forceInitialization);
+}
+
+WebRtc_Word16
+ACMGenericCodec::InitDecoderSafe(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization)
+{
+ int mirrorID;
+ // Check if we got a valid set of parameters
+ int codecNumber =
+ ACMCodecDB::ReceiverCodecNumber(&codecParams->codecInstant, &mirrorID);
+
+ if(codecNumber < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitDecoderSafe: error, invalid codec number");
+ return -1;
+ }
+ // Check if the parameters are for this codec
+ if((_codecID >= 0) && (_codecID != codecNumber) && (_codecID != mirrorID))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitDecoderSafe: current codec is not the same as the one given "
+ "by codecParams");
+ // The current codec is not the same as the one given by codecParams
+ return -1;
+ }
+
+
+ if(_decoderInitialized && !forceInitialization)
+ {
+ // The encoder is already initialized
+ return 0;
+ }
+
+ WebRtc_Word16 status;
+ if(!_decoderExist)
+ {
+ _decoderInitialized = false;
+ status = CreateDecoder();
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitDecoderSafe: cannot create decoder");
+ return -1;
+ }
+ else
+ {
+ _decoderExist = true;
+ }
+ }
+
+ status = InternalInitDecoder(codecParams);
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InitDecoderSafe: cannot init decoder");
+ _decoderInitialized = false;
+ return -1;
+ }
+ else
+ {
+ // Store the parameters
+ SaveDecoderParamSafe(codecParams);
+ _decoderInitialized = true;
+ }
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::ResetDecoder(WebRtc_Word16 payloadType)
+{
+ WriteLockScoped lockCodec(_codecWrapperLock);
+ WriteLockScoped lockNetEq(*_netEqDecodeLock);
+ return ResetDecoderSafe(payloadType);
+}
+
+WebRtc_Word16
+ACMGenericCodec::ResetDecoderSafe(WebRtc_Word16 payloadType)
+{
+ WebRtcACMCodecParams decoderParams;
+ if(!_decoderExist || !_decoderInitialized)
+ {
+ return 0;
+ }
+ // Initialization of the decoder should work for all
+ // the codec. If there is a codec that has to keep
+ // some states then we need to define a virtual and
+ // overwrite in that codec
+ DecoderParamsSafe(&decoderParams, (WebRtc_UWord8) payloadType);
+ return InternalInitDecoder(&decoderParams);
+}
+
+void
+ACMGenericCodec::ResetNoMissedSamples()
+{
+ WriteLockScoped cs(_codecWrapperLock);
+ _noMissedSamples = 0;
+}
+
+void
+ACMGenericCodec::IncreaseNoMissedSamples(
+ const WebRtc_Word16 noSamples)
+{
+ _noMissedSamples += noSamples;
+}
+
+// Get the number of missed samples, this can be public
+WebRtc_UWord32
+ACMGenericCodec::NoMissedSamples() const
+{
+ ReadLockScoped cs(_codecWrapperLock);
+ return _noMissedSamples;
+}
+void
+ACMGenericCodec::DestructEncoder()
+{
+ WriteLockScoped wl(_codecWrapperLock);
+
+ // Disable VAD and delete the instance
+ if(_ptrVADInst != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst);
+ _ptrVADInst = NULL;
+ }
+ _vadEnabled = false;
+ _vadMode = VADNormal;
+
+ //Disable DTX and delete the instance
+ _dtxEnabled = false;
+ if(_ptrDTXInst != NULL)
+ {
+ WebRtcCng_FreeEnc(_ptrDTXInst);
+ _ptrDTXInst = NULL;
+ }
+ _numLPCParams = kNewCNGNumPLCParams;
+
+ DestructEncoderSafe();
+}
+
+void
+ACMGenericCodec::DestructDecoder()
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ _decoderParams.codecInstant.pltype = -1;
+ DestructDecoderSafe();
+}
+
+WebRtc_Word16
+ACMGenericCodec::SetBitRate(
+ const WebRtc_Word32 bitRateBPS)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return SetBitRateSafe(bitRateBPS);
+}
+
+WebRtc_Word16
+ACMGenericCodec::SetBitRateSafe(
+ const WebRtc_Word32 bitRateBPS)
+{
+ // If the codec can change the bit-rate this function
+ // should be overwritten, otherewise the only acceptable
+ // value is the one that is in database.
+ CodecInst codecParams;
+ if(ACMCodecDB::Codec(_codecID, &codecParams) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: error in ACMCodecDB::Codec");
+ return -1;
+ }
+ if(codecParams.rate != bitRateBPS)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: rate value is not acceptable");
+ return -1;
+ }
+ else
+ {
+ return 0;
+ }
+}
+
+WebRtc_Word32
+ACMGenericCodec::GetEstimatedBandwidth()
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return GetEstimatedBandwidthSafe();
+}
+
+WebRtc_Word32
+ACMGenericCodec::GetEstimatedBandwidthSafe()
+{
+ // All codecs but iSAC will return -1
+ return -1;
+}
+
+WebRtc_Word32
+ACMGenericCodec::SetEstimatedBandwidth(
+ WebRtc_Word32 estimatedBandwidth)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return SetEstimatedBandwidthSafe(estimatedBandwidth);
+}
+
+WebRtc_Word32
+ACMGenericCodec::SetEstimatedBandwidthSafe(
+ WebRtc_Word32 /*estimatedBandwidth*/)
+{
+ // All codecs but iSAC will return -1
+ return -1;
+}
+
+WebRtc_Word32
+ACMGenericCodec::GetRedPayload(
+ WebRtc_UWord8* redPayload,
+ WebRtc_Word16* payloadBytes)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ return GetRedPayloadSafe(redPayload, payloadBytes);
+}
+
+WebRtc_Word32
+ACMGenericCodec::GetRedPayloadSafe(
+ WebRtc_UWord8* /* redPayload */,
+ WebRtc_Word16* /* payloadBytes */)
+{
+ return -1; // Do nothing by default
+}
+
+WebRtc_Word16
+ACMGenericCodec::CreateEncoder()
+{
+ WebRtc_Word16 status = 0;
+ if(!_encoderExist)
+ {
+ status = InternalCreateEncoder();
+ // We just created the codec and obviously it is not initialized
+ _encoderInitialized = false;
+ }
+
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CreateEncoder: error in internal create encoder");
+ _encoderExist = false;
+ }
+ else
+ {
+ _encoderExist = true;
+ }
+ return status;
+}
+
+WebRtc_Word16
+ACMGenericCodec::CreateDecoder()
+{
+ WebRtc_Word16 status = 0;
+ if(!_decoderExist)
+ {
+ status = InternalCreateDecoder();
+ // Decoder just created and obviously it is not initialized
+ _decoderInitialized = false;
+ }
+
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CreateDecoder: error in internal create decoder");
+ _decoderExist = false;
+ }
+ else
+ {
+ _decoderExist = true;
+ }
+ return status;
+}
+
+
+void ACMGenericCodec::DestructEncoderInst(void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WriteLockScoped lockCodec(_codecWrapperLock);
+ ReadLockScoped lockNetEq(*_netEqDecodeLock);
+ InternalDestructEncoderInst(ptrInst);
+ }
+}
+
+
+WebRtc_Word16
+ACMGenericCodec::AudioBuffer(
+ WebRtcACMAudioBuff& audioBuff)
+{
+ ReadLockScoped cs(_codecWrapperLock);
+ memcpy(audioBuff.inAudio, _inAudio,
+ AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
+ audioBuff.inAudioIxRead = _inAudioIxRead;
+ audioBuff.inAudioIxWrite = _inAudioIxWrite;
+ memcpy(audioBuff.inTimestamp, _inTimestamp,
+ TIMESTAMP_BUFFER_SIZE_W32*sizeof(WebRtc_UWord32));
+ audioBuff.inTimestampIxWrite = _inTimestampIxWrite;
+ audioBuff.lastTimestamp = _lastTimestamp;
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMGenericCodec::SetAudioBuffer(
+ WebRtcACMAudioBuff& audioBuff)
+{
+ WriteLockScoped cs(_codecWrapperLock);
+ memcpy(_inAudio, audioBuff.inAudio,
+ AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
+ _inAudioIxRead = audioBuff.inAudioIxRead;
+ _inAudioIxWrite = audioBuff.inAudioIxWrite;
+ memcpy(_inTimestamp, audioBuff.inTimestamp,
+ TIMESTAMP_BUFFER_SIZE_W32*sizeof(WebRtc_UWord32));
+ _inTimestampIxWrite = audioBuff.inTimestampIxWrite;
+ _lastTimestamp = audioBuff.lastTimestamp;
+ _isAudioBuffFresh = false;
+ return 0;
+}
+
+
+WebRtc_UWord32
+ACMGenericCodec::LastEncodedTimestamp() const
+{
+ ReadLockScoped cs(_codecWrapperLock);
+ return _lastEncodedTimestamp;
+}
+
+
+WebRtc_UWord32
+ACMGenericCodec::EarliestTimestamp() const
+{
+ ReadLockScoped cs(_codecWrapperLock);
+ return _inTimestamp[0];
+}
+
+
+WebRtc_Word16
+ACMGenericCodec::SetVAD(
+ const bool enableDTX,
+ const bool enableVAD,
+ const ACMVADMode mode)
+{
+ WriteLockScoped cs(_codecWrapperLock);
+ return SetVADSafe(enableDTX, enableVAD, mode);
+}
+
+
+WebRtc_Word16
+ACMGenericCodec::SetVADSafe(
+ const bool enableDTX,
+ const bool enableVAD,
+ const ACMVADMode mode)
+{
+ if(enableDTX)
+ {
+ // Make G729 AnnexB a special case
+ if (!STR_CASE_CMP(_encoderParams.codecInstant.plname, "G729") && !_hasInternalDTX)
+ {
+ if (ACMGenericCodec::EnableDTX() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetVADSafe: error in enable DTX");
+ return -1;
+ }
+ }
+ else
+ {
+ if(EnableDTX() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetVADSafe: error in enable DTX");
+ return -1;
+ }
+ }
+
+ if(_hasInternalDTX)
+ {
+ // Codec has internal DTX, practically we don't need WebRtc VAD,
+ // however, we let the user to turn it on if they need call-backs
+ // on silence. Store VAD mode for future even if VAD is off.
+ _vadMode = mode;
+ return (enableVAD)? EnableVAD(mode):DisableVAD();
+ }
+ else
+ {
+ // Codec does not have internal DTX so enabling DTX requires an
+ // active VAD. 'enableDTX == true' overwrites VAD status.
+ if(EnableVAD(mode) < 0)
+ {
+ // If we cannot create VAD we have to disable DTX
+ if(!_vadEnabled)
+ {
+ DisableDTX();
+ }
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetVADSafe: error in enable VAD");
+ return -1;
+ }
+
+ // Return '1', to let the caller know VAD was turned on, even if the
+ // function was called with VAD='false'
+ if (enableVAD == false) {
+ return 1;
+ } else {
+ return 0;
+ }
+ }
+ }
+ else
+ {
+ // Make G729 AnnexB a special case
+ if (!STR_CASE_CMP(_encoderParams.codecInstant.plname, "G729") && !_hasInternalDTX)
+ {
+ ACMGenericCodec::DisableDTX();
+ }
+ else
+ {
+ DisableDTX();
+ }
+ return (enableVAD)? EnableVAD(mode):DisableVAD();
+ }
+}
+
+WebRtc_Word16
+ACMGenericCodec::EnableDTX()
+{
+ if(_hasInternalDTX)
+ {
+ // We should not be here if we have internal DTX
+ // this function should be overwritten by the derived
+ // class in this case
+ return -1;
+ }
+ if(!_dtxEnabled)
+ {
+ if(WebRtcCng_CreateEnc(&_ptrDTXInst) < 0)
+ {
+ _ptrDTXInst = NULL;
+ return -1;
+ }
+ WebRtc_UWord16 freqHz;
+ EncoderSampFreq(freqHz);
+ if(WebRtcCng_InitEnc(_ptrDTXInst, (WebRtc_Word16)freqHz,
+ ACM_SID_INTERVAL_MSEC, _numLPCParams) < 0)
+ {
+ // Couldn't initialize, has to return -1, and free the memory
+ WebRtcCng_FreeEnc(_ptrDTXInst);
+ _ptrDTXInst = NULL;
+ return -1;
+ }
+ _dtxEnabled = true;
+ }
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::DisableDTX()
+{
+ if(_hasInternalDTX)
+ {
+ // We should not be here if we have internal DTX
+ // this function should be overwritten by the derived
+ // class in this case
+ return -1;
+ }
+ if(_ptrDTXInst != NULL)
+ {
+ WebRtcCng_FreeEnc(_ptrDTXInst);
+ _ptrDTXInst = NULL;
+ }
+ _dtxEnabled = false;
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::EnableVAD(
+ ACMVADMode mode)
+{
+ if((mode < VADNormal) || (mode > VADVeryAggr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EnableVAD: error in VAD mode range");
+ return -1;
+ }
+
+ if(!_vadEnabled)
+ {
+ if(WebRtcVad_Create(&_ptrVADInst) < 0)
+ {
+ _ptrVADInst = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EnableVAD: error in create VAD");
+ return -1;
+ }
+ if(WebRtcVad_Init(_ptrVADInst) < 0)
+ {
+ WebRtcVad_Free(_ptrVADInst);
+ _ptrVADInst = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EnableVAD: error in init VAD");
+ return -1;
+ }
+ }
+
+ // Set the vad mode to the given value
+ if(WebRtcVad_set_mode(_ptrVADInst, mode) < 0)
+ {
+ // We failed to set the mode and we have to return -1. If
+ // we already have a working VAD (_vadEnabled == true) then
+ // we leave it to work. otherwise, the following will be
+ // executed.
+ if(!_vadEnabled)
+ {
+ // We just created the instance but cannot set the mode
+ // we have to free the memomry.
+ WebRtcVad_Free(_ptrVADInst);
+ _ptrVADInst = NULL;
+ }
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
+ "EnableVAD: failed to set the VAD mode");
+ return -1;
+ }
+ _vadMode = mode;
+ _vadEnabled = true;
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::DisableVAD()
+{
+ if(_ptrVADInst != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst);
+ _ptrVADInst = NULL;
+ }
+ _vadEnabled = false;
+ return 0;
+}
+
+WebRtc_Word32
+ACMGenericCodec::ReplaceInternalDTX(
+ const bool replaceInternalDTX)
+{
+ WriteLockScoped cs(_codecWrapperLock);
+ return ReplaceInternalDTXSafe(replaceInternalDTX);
+}
+
+WebRtc_Word32
+ACMGenericCodec::ReplaceInternalDTXSafe(
+ const bool /* replaceInternalDTX */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMGenericCodec::IsInternalDTXReplaced(
+ bool* internalDTXReplaced)
+{
+ WriteLockScoped cs(_codecWrapperLock);
+ return IsInternalDTXReplacedSafe(internalDTXReplaced);
+}
+
+WebRtc_Word32
+ACMGenericCodec::IsInternalDTXReplacedSafe(
+ bool* internalDTXReplaced)
+{
+ *internalDTXReplaced = false;
+ return 0;
+}
+
+WebRtc_Word16
+ACMGenericCodec::ProcessFrameVADDTX(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_Word16* samplesProcessed)
+{
+ if(!_vadEnabled)
+ {
+ // VAD not enabled, set all vadLable[] to 1 (speech detected)
+ for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
+ {
+ _vadLabel[n] = 1;
+ }
+ *samplesProcessed = 0;
+ return 0;
+ }
+ WebRtc_UWord16 freqHz;
+ EncoderSampFreq(freqHz);
+
+ // Calculate number of samples in 10 ms blocks, and number ms in one frame
+ WebRtc_Word16 samplesIn10Msec = (WebRtc_Word16)(freqHz / 100);
+ WebRtc_Word32 frameLenMsec = (((WebRtc_Word32)_frameLenSmpl * 1000) / freqHz);
+ WebRtc_Word16 status;
+
+ // Vector for storing maximum 30 ms of mono audio at 32 kHz
+ WebRtc_Word16 audio[960];
+
+ // Calculate number of VAD-blocks to process, and number of samples in each block.
+ int noSamplesToProcess[2];
+ if (frameLenMsec == 40)
+ {
+ // 20 ms in each VAD block
+ noSamplesToProcess[0] = noSamplesToProcess[1] = 2*samplesIn10Msec;
+ }
+ else
+ {
+ // For 10-30 ms framesizes, second VAD block will be size zero ms,
+ // for 50 and 60 ms first VAD block will be 30 ms.
+ noSamplesToProcess[0] = (frameLenMsec > 30)? 3*samplesIn10Msec : _frameLenSmpl;
+ noSamplesToProcess[1] = _frameLenSmpl-noSamplesToProcess[0];
+ }
+
+ int offSet = 0;
+ int loops = (noSamplesToProcess[1]>0) ? 2 : 1;
+ for (int i=0; i<loops; i++) {
+ // If stereo, calculate mean of the two channels
+ if(_noChannels == 2) {
+ for (int j=0; j<noSamplesToProcess[i]; j++) {
+ audio[j] = (_inAudio[(offSet+j)*2]+_inAudio[(offSet+j)*2+1])/2;
+ }
+ offSet = noSamplesToProcess[0];
+ } else {
+ // Mono, copy data from _inAudio to continue work on
+ memcpy(audio, _inAudio, sizeof(WebRtc_Word16)*noSamplesToProcess[i]);
+ }
+
+ // Call VAD
+ status = WebRtcVad_Process(_ptrVADInst, (WebRtc_Word16)freqHz,
+ audio, noSamplesToProcess[i]);
+
+ _vadLabel[i] = status;
+
+ if(status < 0)
+ {
+ // This will force that the data be removed from the buffer
+ *samplesProcessed += noSamplesToProcess[i];
+ return -1;
+ }
+
+ // If VAD decision non-active, update DTX. NOTE! We only do this if the first part of
+ // a frame gets the VAD decision "inactive". Otherwise DTX might say it is time to
+ // transmit SID frame, but we will encode the whole frame, because the first part is
+ // active.
+ *samplesProcessed = 0;
+ if((status == 0) && (i==0) && _dtxEnabled && !_hasInternalDTX)
+ {
+ WebRtc_Word16 bitStreamLen;
+ WebRtc_Word16 num10MsecFrames = noSamplesToProcess[i] / samplesIn10Msec;
+ *bitStreamLenByte = 0;
+ for(WebRtc_Word16 n = 0; n < num10MsecFrames; n++)
+ {
+ // This block is (passive) && (vad enabled)
+ status = WebRtcCng_Encode(_ptrDTXInst, &audio[n*samplesIn10Msec],
+ samplesIn10Msec, bitStream, &bitStreamLen, 0);
+ if (status < 0) {
+ return -1;
+ }
+
+ *samplesProcessed += samplesIn10Msec*_noChannels;
+
+ // bitStreamLen will only be > 0 once per 100 ms
+ *bitStreamLenByte += bitStreamLen;
+ }
+
+
+ // Check if all samples got processed by the DTX
+ if(*samplesProcessed != noSamplesToProcess[i]*_noChannels) {
+ // Set to zero since something went wrong. Shouldn't happen.
+ *samplesProcessed = 0;
+ }
+ }
+
+ if(*samplesProcessed > 0)
+ {
+ // The block contains inactive speech, and is processed by DTX.
+ // Discontinue running VAD.
+ break;
+ }
+ }
+
+ return status;
+}
+
+WebRtc_Word16
+ACMGenericCodec::SamplesLeftToEncode()
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return (_frameLenSmpl <= _inAudioIxWrite)?
+ 0:(_frameLenSmpl - _inAudioIxWrite);
+}
+
+WebRtc_Word32
+ACMGenericCodec::UnregisterFromNetEq(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ if(!_registeredInNetEq)
+ {
+ return 0;
+ }
+ if(UnregisterFromNetEqSafe(netEq, payloadType) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "UnregisterFromNetEq: error, cannot unregister from NetEq");
+ _registeredInNetEq = true;
+ return -1;
+ }
+ else
+ {
+ _registeredInNetEq = false;
+ return 0;
+ }
+}
+
+void
+ACMGenericCodec::SetUniqueID(
+ const WebRtc_UWord32 id)
+{
+ _uniqueID = id;
+}
+
+bool
+ACMGenericCodec::IsAudioBufferFresh() const
+{
+ ReadLockScoped rl(_codecWrapperLock);
+ return _isAudioBuffFresh;
+}
+
+// This function is replaced by codec specific functions for some codecs
+WebRtc_Word16
+ACMGenericCodec::EncoderSampFreq(WebRtc_UWord16& sampFreqHz)
+{
+ WebRtc_Word32 f;
+ f = ACMCodecDB::CodecFreq(_codecID);
+ if(f < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EncoderSampFreq: codec frequency is negative");
+ return -1;
+ }
+ else
+ {
+ sampFreqHz = (WebRtc_UWord16)f;
+ return 0;
+ }
+}
+
+
+WebRtc_Word32
+ACMGenericCodec::ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 /* initFrameSizeMsec */,
+ const WebRtc_UWord16 /* initRateBitPerSec */,
+ const bool /* enforceFrameSize */)
+{
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
+ "The send-codec is not iSAC, failed to config iSAC bandwidth estimator.");
+ return -1;
+}
+
+WebRtc_Word32
+ACMGenericCodec::SetISACMaxRate(
+ const WebRtc_UWord32 /* maxRateBitPerSec */)
+{
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
+ "The send-codec is not iSAC, failed to set iSAC max rate.");
+ return -1;
+}
+
+WebRtc_Word32
+ACMGenericCodec::SetISACMaxPayloadSize(
+ const WebRtc_UWord16 /* maxPayloadLenBytes */)
+{
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
+ "The send-codec is not iSAC, failed to set iSAC max payload-size.");
+ return -1;
+}
+
+
+void
+ACMGenericCodec::SaveDecoderParam(
+ const WebRtcACMCodecParams* codecParams)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ SaveDecoderParamSafe(codecParams);
+}
+
+
+void
+ACMGenericCodec::SaveDecoderParamSafe(
+ const WebRtcACMCodecParams* codecParams)
+{
+ memcpy(&_decoderParams, codecParams, sizeof(WebRtcACMCodecParams));
+}
+
+WebRtc_Word16
+ACMGenericCodec::UpdateEncoderSampFreq(
+ WebRtc_UWord16 /* encoderSampFreqHz */)
+{
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "It is asked for a change in smapling frequency while the \
+current send-codec supports only one sampling rate.");
+ return -1;
+}
+
+
+void
+ACMGenericCodec::SetIsMaster(
+ bool isMaster)
+{
+ WriteLockScoped wl(_codecWrapperLock);
+ _isMaster = isMaster;
+}
+
+
+
+WebRtc_Word16
+ACMGenericCodec::REDPayloadISAC(
+ const WebRtc_Word32 /* isacRate */,
+ const WebRtc_Word16 /* isacBwEstimate */,
+ WebRtc_UWord8* /* payload */,
+ WebRtc_Word16* /* payloadLenBytes */)
+{
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error: REDPayloadISAC is an iSAC specific function");
+ return -1;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h
new file mode 100644
index 0000000..7f8ef8c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h
@@ -0,0 +1,1343 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
+
+#include "acm_common_defs.h"
+#include "audio_coding_module_typedefs.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+
+#define MAX_FRAME_SIZE_10MSEC 6
+
+// forward declaration
+struct WebRtcVadInst;
+struct WebRtcCngEncInst;
+
+namespace webrtc
+{
+
+// forward declaration
+struct CodecInst;
+class ACMNetEQ;
+
+class ACMGenericCodec
+{
+public:
+ ///////////////////////////////////////////////////////////////////////////
+ // Constructor of the class
+ //
+ ACMGenericCodec();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Destructor of the class.
+ //
+ virtual ~ACMGenericCodec();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // ACMGenericCodec* CreateInstance();
+ // The function will be used for FEC. It is not implemented yet.
+ //
+ virtual ACMGenericCodec* CreateInstance() = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 Encode()
+ // The function is called to perform an encoding of the audio stored in
+ // audio buffer. An encoding is performed only if enough audio, i.e. equal
+ // to the frame-size of the codec, exist. The audio frame will be processed
+ // by VAD and CN/DTX if required. There are few different cases.
+ //
+ // A) Neither VAD nor DTX is active; the frame is encoded by the encoder.
+ //
+ // B) VAD is enabled but not DTX; in this case the audio is processed by VAD
+ // and encoded by the encoder. The "*encodingType" will be either
+ // "activeNormalEncode" or "passiveNormalEncode" if frame is active or
+ // passive, respectively.
+ //
+ // C) DTX is enabled; if the codec has internal VAD/DTX we just encode the
+ // frame by the encoder. Otherwise, the frame is passed through VAD and
+ // if identified as passive, then it will be processed by CN/DTX. If the
+ // frame is active it will be encoded by the encoder.
+ //
+ // This function acquires the appropriate locks and calls EncodeSafe() for
+ // the actual processing.
+ //
+ // Outputs:
+ // -bitStream : a buffer where bit-stream will be written to.
+ // -bitStreamLenByte : contains the length of the bit-stream in
+ // bytes.
+ // -timeStamp : contains the RTP timestamp, this is the
+ // sampling time of the first sample encoded
+ // (measured in number of samples).
+ // -encodingType : contains the type of encoding applied on the
+ // audio samples. The alternatives are
+ // (c.f. acm_common_types.h)
+ // -kNoEncoding:
+ // there was not enough data to encode. or
+ // some error has happened that we could
+ // not do encoding.
+ // -kActiveNormalEncoded:
+ // the audio frame is active and encoded by
+ // the given codec.
+ // -kPassiveNormalEncoded:
+ // the audio frame is passive but coded with
+ // the given codec (NO DTX).
+ // -kPassiveDTXWB:
+ // The audio frame is passive and used
+ // wide-band CN to encode.
+ // -kPassiveDTXNB:
+ // The audio frame is passive and used
+ // narrow-band CN to encode.
+ //
+ // Return value:
+ // -1 if error is occurred, otherwise the length of the bit-stream in
+ // bytes.
+ //
+ WebRtc_Word16 Encode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_UWord32* timeStamp,
+ WebRtcACMEncodingType* encodingType);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 Decode()
+ // This function is used to decode a given bit-stream, without engaging
+ // NetEQ.
+ //
+ // This function acquires the appropriate locks and calls DecodeSafe() for
+ // the actual processing. Please note that this is not functional yet.
+ //
+ // Inputs:
+ // -bitStream : a buffer where bit-stream will be read.
+ // -bitStreamLenByte : the length of the bit-stream in bytes.
+ //
+ // Outputs:
+ // -audio : pointer to a buffer where the audio will written.
+ // -audioSamples : number of audio samples out of decoding the given
+ // bit-stream.
+ // -speechType : speech type (for future use).
+ //
+ // Return value:
+ // -1 if failed to decode,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 Decode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool EncoderInitialized();
+ //
+ // Return value:
+ // True if the encoder is successfully initialized,
+ // false otherwise.
+ //
+ bool EncoderInitialized();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool DecoderInitialized();
+ //
+ // Return value:
+ // True if the decoder is successfully initialized,
+ // false otherwise.
+ //
+ bool DecoderInitialized();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 EncoderParams()
+ // It is called to get encoder parameters. It will call
+ // EncoderParamsSafe() in turn.
+ //
+ // Output:
+ // -encParams : a buffer where the encoder parameters is
+ // written to. If the encoder is not
+ // initialized this buffer is filled with
+ // invalid values
+ // Return value:
+ // -1 if the encoder is not initialized,
+ // 0 otherwise.
+ //
+ //
+ WebRtc_Word16 EncoderParams(
+ WebRtcACMCodecParams *encParams);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 DecoderParams(...)
+ // It is called to get decoder parameters. It will call DecoderParamsSafe()
+ // in turn.
+ //
+ // Output:
+ // -decParams : a buffer where the decoder parameters is
+ // written to. If the decoder is not initialized
+ // this buffer is filled with invalid values
+ //
+ // Return value:
+ // -1 if the decoder is not initialized,
+ // 0 otherwise.
+ //
+ //
+ bool DecoderParams(
+ WebRtcACMCodecParams *decParams,
+ const WebRtc_UWord8 payloadType);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InitEncoder(...)
+ // This function is called to initialize the encoder with the given
+ // parameters.
+ //
+ // Input:
+ // -codecParams : parameters of encoder.
+ // -forceInitialization: if false the initialization is invoked only if
+ // the encoder is not initialized. If true the
+ // encoder is forced to (re)initialize.
+ //
+ // Return value:
+ // 0 if could initialize successfully,
+ // -1 if failed to initialize.
+ //
+ //
+ WebRtc_Word16 InitEncoder(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InitDecoder()
+ // This function is called to initialize the decoder with the given
+ // parameters. (c.f. acm_common_defs.h & common_types.h for the
+ // definition of the structure)
+ //
+ // Input:
+ // -codecParams : parameters of decoder.
+ // -forceInitialization: if false the initialization is invoked only
+ // if the decoder is not initialized. If true
+ // the encoder is forced to(re)initialize.
+ //
+ // Return value:
+ // 0 if could initialize successfully,
+ // -1 if failed to initialize.
+ //
+ //
+ WebRtc_Word16 InitDecoder(
+ WebRtcACMCodecParams* codecParams,
+ bool forceInitialization);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 RegisterInNetEq(...)
+ // This function is called to register the decoder in NetEq, with the given
+ // payload-type.
+ //
+ // Inputs:
+ // -netEq : pointer to NetEq Instance
+ // -codecInst : instance with of the codec settings of the codec
+ //
+ // Return values
+ // -1 if failed to register,
+ // 0 if successfully initialized.
+ //
+ WebRtc_Word32 RegisterInNetEq(
+ ACMNetEQ* netEq,
+ const CodecInst& codecInst);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 Add10MsData(...)
+ // This function is called to add 10 ms of audio to the audio buffer of
+ // the codec.
+ //
+ // Inputs:
+ // -timeStamp : the timestamp of the 10 ms audio. the timestamp
+ // is the sampling time of the
+ // first sample measured in number of samples.
+ // -data : a buffer that contains the audio. The codec
+ // expects to get the audio in correct sampling
+ // frequency
+ // -length : the length of the audio buffer
+ // -audioChannel : 0 for mono, 1 for stereo (not supported yet)
+ //
+ // Return values:
+ // -1 if failed
+ // 0 otherwise.
+ //
+ WebRtc_Word32 Add10MsData(
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8 audioChannel);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_UWord32 NoMissedSamples()
+ // This function returns the number of samples which are overwritten in
+ // the audio buffer. The audio samples are overwritten if the input audio
+ // buffer is full, but Add10MsData() is called. (We might remove this
+ // function if it is not used)
+ //
+ // Return Value:
+ // Number of samples which are overwritten.
+ //
+ WebRtc_UWord32 NoMissedSamples() const;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void ResetNoMissedSamples()
+ // This function resets the number of overwritten samples to zero.
+ // (We might remove this function if we remove NoMissedSamples())
+ //
+ void ResetNoMissedSamples();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 SetBitRate()
+ // The function is called to set the encoding rate.
+ //
+ // Input:
+ // -bitRateBPS : encoding rate in bits per second
+ //
+ // Return value:
+ // -1 if failed to set the rate, due to invalid input or given
+ // codec is not rate-adjustable.
+ // 0 if the rate is adjusted successfully
+ //
+ WebRtc_Word16 SetBitRate(const WebRtc_Word32 bitRateBPS);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // DestructEncoderInst()
+ // This API is used in conferencing. It will free the memory that is pointed
+ // by "ptrInst". "ptrInst" is a pointer to encoder instance, created and
+ // filled up by calling EncoderInst(...).
+ //
+ // Inputs:
+ // -ptrInst : pointer to an encoder instance to be deleted.
+ //
+ //
+ void DestructEncoderInst(
+ void* ptrInst);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 AudioBuffer()
+ // This is used when synchronization of codecs is required. There are cases
+ // that the audio buffers of two codecs have to be synched. By calling this
+ // function on can get the audio buffer and other related parameters, such
+ // as timestamps...
+ //
+ // Output:
+ // -audioBuff : a pointer to WebRtcACMAudioBuff where the audio
+ // buffer of this codec will be written to.
+ //
+ // Return value:
+ // -1 if fails to copy the audio buffer,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 AudioBuffer(
+ WebRtcACMAudioBuff& audioBuff);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_UWord32 EarliestTimestamp()
+ // Returns the timestamp of the first 10 ms in audio buffer. This is used
+ // to identify if a synchronization of two encoders is required.
+ //
+ // Return value:
+ // timestamp of the first 10 ms audio in the audio buffer.
+ //
+ WebRtc_UWord32 EarliestTimestamp() const;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 SetAudioBuffer()
+ // This function is called to set the audio buffer and the associated
+ // parameters to a given value.
+ //
+ // Return value:
+ // -1 if fails to copy the audio buffer,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
+
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 SetVAD()
+ // This is called to set VAD & DTX. If the codec has internal DTX that will
+ // be used. If DTX is enabled and the codec does not have internal DTX,
+ // WebRtc-VAD will be used to decide if the frame is active. If DTX is
+ // disabled but VAD is enabled. The audio is passed through VAD to label it
+ // as active or passive, but the frame is encoded normally. However the
+ // bit-stream is labeled properly so that ACM::Process() can use this
+ // information. In case of failure, the previous states of the VAD & DTX
+ // are kept.
+ //
+ // Inputs:
+ // -enableDTX : if true DTX will be enabled otherwise the DTX is
+ // disabled. If codec has internal DTX that will be
+ // used, otherwise WebRtc-CNG is used. In the latter
+ // case VAD is automatically activated.
+ // -enableVAD : if true WebRtc-VAD is enabled, otherwise VAD is
+ // disabled, except for the case that DTX is enabled
+ // but codec doesn't have internal DTX. In this case
+ // VAD is enabled regardless of the value of
+ // "enableVAD."
+ // -mode : this specifies the aggressiveness of VAD.
+ //
+ // Return value
+ // -1 if failed to set DTX & VAD as specified,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 SetVAD(
+ const bool enableDTX = true,
+ const bool enableVAD = false,
+ const ACMVADMode mode = VADNormal);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ReplaceInternalDTX()
+ // This is called to replace the codec internal DTX with WebRtc DTX.
+ // This is only valid for G729 where the user has possibility to replace
+ // AnnexB with WebRtc DTX. For other codecs this function has no effect.
+ //
+ // Input:
+ // -replaceInternalDTX : if true the internal DTX is replaced with WebRtc.
+ //
+ // Return value
+ // -1 if failed to replace internal DTX,
+ // 0 if succeeded.
+ //
+ WebRtc_Word32 ReplaceInternalDTX(const bool replaceInternalDTX);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 IsInternalDTXReplaced()
+ // This is called to check if the codec internal DTX is replaced by WebRtc DTX.
+ // This is only valid for G729 where the user has possibility to replace
+ // AnnexB with WebRtc DTX. For other codecs this function has no effect.
+ //
+ // Output:
+ // -internalDTXReplaced : if true the internal DTX is replaced with WebRtc.
+ //
+ // Return value
+ // -1 if failed to check if replace internal DTX or replacement not feasible,
+ // 0 if succeeded.
+ //
+ WebRtc_Word32 IsInternalDTXReplaced(bool* internalDTXReplaced);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void SetNetEqDecodeLock()
+ // Passes the NetEq lock to the codec.
+ //
+ // Input:
+ // -netEqDecodeLock : pointer to the lock associated with NetEQ of ACM.
+ //
+ void SetNetEqDecodeLock(
+ RWLockWrapper* netEqDecodeLock)
+ {
+ _netEqDecodeLock = netEqDecodeLock;
+ }
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool HasInternalDTX()
+ // Used to check if the codec has internal DTX.
+ //
+ // Return value:
+ // true if the codec has an internal DTX, e.g. G729,
+ // false otherwise.
+ //
+ bool HasInternalDTX() const
+ {
+ return _hasInternalDTX;
+ }
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 GetEstimatedBandwidth()
+ // Used to get decoder estimated bandwidth. Only iSAC will provide a value.
+ //
+ //
+ // Return value:
+ // -1 if fails to get decoder estimated bandwidth,
+ // >0 estimated bandwidth in bits/sec.
+ //
+ WebRtc_Word32 GetEstimatedBandwidth();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 SetEstimatedBandwidth()
+ // Used to set estiamted bandwidth sent out of band from other side. Only
+ // iSAC will have use for the value.
+ //
+ // Input:
+ // -estimatedBandwidth: estimated bandwidth in bits/sec
+ //
+ // Return value:
+ // -1 if fails to set estimated bandwidth,
+ // 0 on success.
+ //
+ WebRtc_Word32 SetEstimatedBandwidth(WebRtc_Word32 estimatedBandwidth);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 GetRedPayload()
+ // Used to get codec specific RED payload (if such is implemented).
+ // Currently only done in iSAC.
+ //
+ // Outputs:
+ // -redPayload : a pointer to the data for RED payload.
+ // -payloadBytes : number of bytes in RED payload.
+ //
+ // Return value:
+ // -1 if fails to get codec specific RED,
+ // 0 if succeeded.
+ //
+ WebRtc_Word32 GetRedPayload(
+ WebRtc_UWord8* redPayload,
+ WebRtc_Word16* payloadBytes);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 ResetEncoder()
+ // By calling this function you would re-initialize the encoder with the
+ // current parameters. All the settings, e.g. VAD/DTX, frame-size... should
+ // remain unchanged. (In case of iSAC we don't want to lose BWE history.)
+ //
+ // Return value
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 ResetEncoder();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 ResetEncoder()
+ // By calling this function you would re-initialize the decoder with the
+ // current parameters.
+ //
+ // Return value
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 ResetDecoder(
+ WebRtc_Word16 payloadType);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void DestructEncoder()
+ // This function is called to delete the encoder instance, if possible, to
+ // have a fresh start. For codecs where encoder and decoder share the same
+ // instance we cannot delete the encoder and instead we will initialize the
+ // encoder. We also delete VAD and DTX if they have been created.
+ //
+ void DestructEncoder();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void DestructDecoder()
+ // This function is called to delete the decoder instance, if possible, to
+ // have a fresh start. For codecs where encoder and decoder share the same
+ // instance we cannot delete the encoder and instead we will initialize the
+ // decoder. Before deleting decoder instance it has to be removed from the
+ // NetEq list.
+ //
+ void DestructDecoder();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 SamplesLeftToEncode()
+ // Returns the number of samples required to be able to do encoding.
+ //
+ // Return value:
+ // Number of samples.
+ //
+ WebRtc_Word16 SamplesLeftToEncode();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_UWord32 LastEncodedTimestamp()
+ // Returns the timestamp of the last frame it encoded.
+ //
+ // Return value:
+ // Timestamp.
+ //
+ WebRtc_UWord32 LastEncodedTimestamp() const;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 UnregisterFromNetEq()
+ // To remove the codec from NetEQ. If the codec (or the decoder instance)
+ // is going to be deleted, first the codec has to be removed from NetEq
+ // by calling this function.
+ //
+ // Input:
+ // -netEq : pointer to a NetEq instance that the codec
+ // has to be unregistered from.
+ //
+ // Output:
+ // -1 if failed to unregister the codec,
+ // 0 if the codec is successfully unregistered.
+ //
+ WebRtc_Word32 UnregisterFromNetEq(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // SetUniqueID()
+ // Set a unique ID for the codec to be used for tracing and debuging
+ //
+ // Input
+ // -id : A number to identify the codec.
+ //
+ void SetUniqueID(
+ const WebRtc_UWord32 id);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // IsAudioBufferFresh()
+ // Specifies if ever audio is injected to this codec.
+ //
+ // Return value
+ // -true; no audio is feed into this codec
+ // -false; audio has already been fed to the codec.
+ //
+ bool IsAudioBufferFresh() const;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // UpdateDecoderSampFreq()
+ // For most of the codecs this function does nothing. It must be
+ // implemented for those codecs that one codec instance serves as the
+ // decoder for different flavers of the codec. One example is iSAC. there,
+ // iSAC 16 kHz and iSAC 32 kHz are treated as two different codecs with
+ // different payload types, however, there is only one iSAC instance to
+ // decode. The reason for that is we would like to decode and encode with
+ // the same codec instance for bandwidth estimator to work.
+ //
+ // Each time that we receive a new payload type, we call this funtion to
+ // prepare the decoder associated with the new payload. Normally, decoders
+ // doesn't have to do anything. For iSAC the decoder has to change it's
+ // sampling rate. The input parameter specifies the current flaver of the
+ // codec in codec database. For instance, if we just got a SWB payload then
+ // the input parameter is ACMCodecDB::isacswb.
+ //
+ // Input:
+ // -codecId : the ID of the codec associated with the
+ // payload type that we just received.
+ //
+ // Return value:
+ // 0 if succeeded in updating the decoder.
+ // -1 if failed to update.
+ //
+ virtual WebRtc_Word16 UpdateDecoderSampFreq(
+ WebRtc_Word16 /* codecId */)
+ {
+ return 0;
+ }
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // UpdateEncoderSampFreq()
+ // Call this function to update the encoder sampling frequency. This
+ // is for codecs where one payload-name supports several encoder sampling
+ // frequencies. Otherwise, to change the sampling frequency we need to
+ // register new codec. ACM will consider that as registration of a new
+ // codec, not a change in parameter. For iSAC, switching from WB to SWB
+ // is treated as a change in parameter. Therefore, we need this function.
+ //
+ // Input:
+ // -encoderSampFreqHz : encoder sampling frequency.
+ //
+ // Return value:
+ // -1 if failed, or if this is meaningless for the given codec.
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 UpdateEncoderSampFreq(
+ WebRtc_UWord16 encoderSampFreqHz);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // EncoderSampFreq()
+ // Get the sampling frequency that the encoder (WebRtc wrapper) expects.
+ //
+ // Output:
+ // -sampFreqHz : sampling frequency, in Hertz, which the encoder
+ // should be fed with.
+ //
+ // Return value:
+ // -1 if failed to output sampling rate.
+ // 0 if the sample rate is returned successfully.
+ //
+ virtual WebRtc_Word16 EncoderSampFreq(
+ WebRtc_UWord16& sampFreqHz);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word32 ConfigISACBandwidthEstimator()
+ // Call this function to configure the bandwidth estimator of ISAC.
+ // During the adaptation of bit-rate, iSAC atomatically adjusts the
+ // frame-size (either 30 or 60 ms) to save on RTP header. The initial
+ // frame-size can be specified by the first argument. The configuration also
+ // regards the initial estimate of bandwidths. The estimator starts from
+ // this point and converges to the actual bottleneck. This is given by the
+ // second parameter. Furthermore, it is also possible to control the
+ // adaptation of frame-size. This is specified by the last parameter.
+ //
+ // Input:
+ // -initFrameSizeMsec : initial frame-size in milisecods. For iSAC-wb
+ // 30 ms and 60 ms (default) are acceptable values,
+ // and for iSAC-swb 30 ms is the only acceptable
+ // value. Zero indiates default value.
+ // -initRateBitPerSec : initial estimate of the bandwidth. Values
+ // between 10000 and 58000 are acceptable.
+ // -enforceFrameSize : if true, the frame-size will not be adapted.
+ //
+ // Return value:
+ // -1 if failed to configure the bandwidth estimator,
+ // 0 if the configuration was successfully applied.
+ //
+ virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // SetISACMaxPayloadSize()
+ // Set the maximum payload size of iSAC packets. No iSAC payload,
+ // regardless of its frame-size, may exceed the given limit. For
+ // an iSAC payload of size B bits and frame-size T sec we have;
+ // (B < maxPayloadLenBytes * 8) and (B/T < maxRateBitPerSec), c.f.
+ // SetISACMaxRate().
+ //
+ // Input:
+ // -maxPayloadLenBytes : maximum payload size in bytes.
+ //
+ // Return value:
+ // -1 if failed to set the maximm payload-size.
+ // 0 if the given linit is seet successfully.
+ //
+ virtual WebRtc_Word32 SetISACMaxPayloadSize(
+ const WebRtc_UWord16 maxPayloadLenBytes);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // SetISACMaxRate()
+ // Set the maximum instantaneous rate of iSAC. For a payload of B bits
+ // with a frame-size of T sec the instantaneous rate is B/T bist per
+ // second. Therefore, (B/T < maxRateBitPerSec) and
+ // (B < maxPayloadLenBytes * 8) are always satisfied for iSAC payloads,
+ // c.f SetISACMaxPayloadSize().
+ //
+ // Input:
+ // -maxRateBitPerSec : maximum instantaneous bit-rate given in bits/sec.
+ //
+ // Return value:
+ // -1 if failed to set the maximum rate.
+ // 0 if the maximum rate is set successfully.
+ //
+ virtual WebRtc_Word32 SetISACMaxRate(
+ const WebRtc_UWord32 maxRateBitPerSec);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // SaveDecoderParamS()
+ // Save the parameters of decoder.
+ //
+ // Input:
+ // -codecParams : pointer to a struct where the parameters of
+ // decoder is stored in.
+ //
+ void SaveDecoderParam(
+ const WebRtcACMCodecParams* codecParams);
+
+
+ WebRtc_Word32 FrameSize()
+ {
+ return _frameLenSmpl;
+ }
+
+ void SetIsMaster(bool isMaster);
+
+
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // REDPayloadISAC()
+ // This is an iSAC-specific function. The function is called to get RED
+ // paylaod from a default-encoder.
+ //
+ // Inputs:
+ // -isacRate : the target rate of the main payload. A RED
+ // paylaod is generated according to the rate of
+ // main paylaod. Note that we are not specifying the
+ // rate of RED payload, but the main payload.
+ // -isacBwEstimate : bandwidth information should be inserted in
+ // RED payload.
+ //
+ // Output:
+ // -payload : pointer to a buffer where the RED paylaod will
+ // written to.
+ // -paylaodLenBytes : a place-holder to write the length of the RED
+ // payload in Bytes.
+ //
+ // Return value:
+ // -1 if an error occures, otherwise the length of the payload (in Bytes)
+ // is returned.
+ //
+ //
+ virtual WebRtc_Word16 REDPayloadISAC(
+ const WebRtc_Word32 isacRate,
+ const WebRtc_Word16 isacBwEstimate,
+ WebRtc_UWord8* payload,
+ WebRtc_Word16* payloadLenBytes);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // IsTrueStereoCodec()
+ // Call to see if current encoder is a true stereo codec. This function
+ // should be overwritten for codecs which are true stereo codecs
+ // Return value:
+ // -true if stereo codec
+ // -false if not stereo codec.
+ //
+ virtual bool IsTrueStereoCodec() {
+ return false;
+ }
+
+protected:
+ ///////////////////////////////////////////////////////////////////////////
+ // All the functions with FunctionNameSafe(...) contain the actual
+ // implementation of FunctionName(...). FunctionName() acquires an
+ // appropriate lock and calls FunctionNameSafe() to do the actual work.
+ // Therefore, for the description of functionality, input/output arguments
+ // and return value we refer to FunctionName()
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See Encode() for the description of function, input(s)/output(s) and
+ // return value.
+ //
+ WebRtc_Word16 EncodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_UWord32* timeStamp,
+ WebRtcACMEncodingType* encodingType);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See Decode() for the description of function, input(s)/output(s) and
+ // return value.
+ //
+ virtual WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See Add10MsSafe() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ virtual WebRtc_Word32 Add10MsDataSafe(
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16* data,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8 audioChannel);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See RegisterInNetEq() for the description of function,
+ // input(s)/output(s) and return value.
+ //
+ virtual WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See EncoderParam() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ WebRtc_Word16 EncoderParamsSafe(
+ WebRtcACMCodecParams *encParams);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See DecoderParam for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ // Note:
+ // Any Class where a single instance handle several flavers of the
+ // same codec, therefore, several payload types are associated with
+ // the same instance have to implement this function.
+ //
+ // Currently only iSAC is implementing it. A single iSAC instance is
+ // used for decoding both WB & SWB stream. At one moment both WB & SWB
+ // can be registered as receive codec. Hence two payloads are associated
+ // with a single codec instance.
+ //
+ virtual bool DecoderParamsSafe(
+ WebRtcACMCodecParams *decParams,
+ const WebRtc_UWord8 payloadType);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See ResetEncoder() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ WebRtc_Word16 ResetEncoderSafe();
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See InitEncoder() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ WebRtc_Word16 InitEncoderSafe(
+ WebRtcACMCodecParams *codecParams,
+ bool forceInitialization);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See InitDecoder() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ WebRtc_Word16 InitDecoderSafe(
+ WebRtcACMCodecParams *codecParams,
+ bool forceInitialization);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See ResetDecoder() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ WebRtc_Word16 ResetDecoderSafe(
+ WebRtc_Word16 payloadType);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See DestructEncoder() for the description of function,
+ // input(s)/output(s) and return value.
+ //
+ virtual void DestructEncoderSafe() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See DestructDecoder() for the description of function,
+ // input(s)/output(s) and return value.
+ //
+ virtual void DestructDecoderSafe() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See SetBitRate() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ // Any codec that can change the bit-rate has to implement this.
+ //
+ virtual WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 bitRateBPS);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See GetEstimatedBandwidth() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ virtual WebRtc_Word32 GetEstimatedBandwidthSafe();
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See SetEstimatedBandwidth() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ virtual WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See GetRedPayload() for the description of function, input(s)/output(s)
+ // and return value.
+ //
+ virtual WebRtc_Word32 GetRedPayloadSafe(
+ WebRtc_UWord8* redPayload,
+ WebRtc_Word16* payloadBytes);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See SetVAD() for the description of function, input(s)/output(s) and
+ // return value.
+ //
+ WebRtc_Word16 SetVADSafe(
+ const bool enableDTX = true,
+ const bool enableVAD = false,
+ const ACMVADMode mode = VADNormal);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See ReplaceInternalDTX() for the description of function, input and
+ // return value.
+ //
+ virtual WebRtc_Word32 ReplaceInternalDTXSafe(
+ const bool replaceInternalDTX);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See IsInternalDTXReplaced() for the description of function, input and
+ // return value.
+ //
+ virtual WebRtc_Word32 IsInternalDTXReplacedSafe(
+ bool* internalDTXReplaced);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // See UnregisterFromNetEq() for the description of function,
+ // input(s)/output(s) and return value.
+ //
+ virtual WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType) = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 CreateEncoder()
+ // Creates the encoder instance.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 CreateEncoder();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 CreateDecoder()
+ // Creates the decoder instance.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 CreateDecoder();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 EnableVAD();
+ // Enables VAD with the given mode. The VAD instance will be created if
+ // it does not exists.
+ //
+ // Input:
+ // -mode : VAD mode c.f. audio_coding_module_typedefs.h for
+ // the options.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 EnableVAD(ACMVADMode mode);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 DisableVAD()
+ // Disables VAD.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 DisableVAD();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 EnableDTX()
+ // Enables DTX. This method should be overwritten for codecs which have
+ // internal DTX.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 EnableDTX();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 DisableDTX()
+ // Disables usage of DTX. This method should be overwritten for codecs which
+ // have internal DTX.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 DisableDTX();
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalEncode()
+ // This is a codec-specific function called in EncodeSafe() to actually
+ // encode a frame of audio.
+ //
+ // Outputs:
+ // -bitStream : pointer to a buffer where the bit-stream is
+ // written to.
+ // -bitStreamLenByte : the length of the bit-stream in byte, a negative
+ // value indicates error.
+ //
+ // Return value:
+ // -1 if failed,
+ // otherwise the length of the bit-stream is returned.
+ //
+ virtual WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte) = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalInitEncoder()
+ // This is a codec-specific function called in InitEncoderSafe(), it has to
+ // do all codec-specific operation to initialize the encoder given the
+ // encoder parameters.
+ //
+ // Input:
+ // -codecParams : pointer to a structure that contains parameters to
+ // initialize encoder.
+ // Set codecParam->CodecInst.rate to -1 for
+ // iSAC to operate in adaptive mode.
+ // (to do: if frame-length is -1 frame-length will be
+ // automatically adjusted, otherwise, given
+ // frame-length is forced)
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams) = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalInitDecoder()
+ // This is a codec-specific function called in InitDecoderSafe(), it has to
+ // do all codec-specific operation to initialize the decoder given the
+ // decoder parameters.
+ //
+ // Input:
+ // -codecParams : pointer to a structure that contains parameters to
+ // initialize encoder.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams) = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void IncreaseNoMissedSamples()
+ // This method is called to increase the number of samples that are
+ // overwritten in the audio buffer.
+ //
+ // Input:
+ // -noSamples : the number of overwritten samples is incremented
+ // by this value.
+ //
+ void IncreaseNoMissedSamples(
+ const WebRtc_Word16 noSamples);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalCreateEncoder()
+ // This is a codec-specific method called in CreateEncoderSafe() it is
+ // supposed to perform all codec-specific operations to create encoder
+ // instance.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 InternalCreateEncoder() = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalCreateDecoder()
+ // This is a codec-specific method called in CreateDecoderSafe() it is
+ // supposed to perform all codec-specific operations to create decoder
+ // instance.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 InternalCreateDecoder() = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // void InternalDestructEncoderInst()
+ // This is a codec-specific method, used in conferencing, called from
+ // DestructEncoderInst(). The input argument is pointer to encoder instance
+ // (codec instance for codecs that encoder and decoder share the same
+ // instance). This method is called to free the memory that "ptrInst" is
+ // pointing to.
+ //
+ // Input:
+ // -ptrInst : pointer to encoder instance.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual void InternalDestructEncoderInst(
+ void* ptrInst) = 0;
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 InternalResetEncoder()
+ // This method is called to reset the states of encoder. However, the
+ // current parameters, e.g. frame-length, should remain as they are. For
+ // most of the codecs a re-initialization of the encoder is what needs to
+ // be down. But for iSAC we like to keep the BWE history so we cannot
+ // re-initialize. As soon as such an API is implemented in iSAC this method
+ // has to be overwritten in ACMISAC class.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ virtual WebRtc_Word16 InternalResetEncoder();
+
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // WebRtc_Word16 ProcessFrameVADDTX()
+ // This function is called when a full frame of audio is available. It will
+ // break the audio frame into blocks such that each block could be processed
+ // by VAD & CN/DTX. If a frame is divided into two blocks then there are two
+ // cases. First, the first block is active, the second block will not be
+ // processed by CN/DTX but only by VAD and return to caller with
+ // '*samplesProcessed' set to zero. There, the audio frame will be encoded
+ // by the encoder. Second, the first block is inactive and is processed by
+ // CN/DTX, then we stop processing the next block and return to the caller
+ // which is EncodeSafe(), with "*samplesProcessed" equal to the number of
+ // samples in first block.
+ //
+ // Output:
+ // -bitStream : pointer to a buffer where DTX frame, if
+ // generated, will be written to.
+ // -bitStreamLenByte : contains the length of bit-stream in bytes, if
+ // generated. Zero if no bit-stream is generated.
+ // -noSamplesProcessed : contains no of samples that actually CN has
+ // processed. Those samples processed by CN will not
+ // be encoded by the encoder, obviously. If
+ // contains zero, it means that the frame has been
+ // identified as active by VAD. Note that
+ // "*noSamplesProcessed" might be non-zero but
+ // "*bitStreamLenByte" be zero.
+ //
+ // Return value:
+ // -1 if failed,
+ // 0 if succeeded.
+ //
+ WebRtc_Word16 ProcessFrameVADDTX(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_Word16* samplesProcessed);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // CanChangeEncodingParam()
+ // Check if the codec parameters can be changed. In conferencing normally
+ // codec parametrs cannot be changed. The exception is bit-rate of isac.
+ //
+ // return value:
+ // -true if codec parameters are allowed to change.
+ // -flase otherwise.
+ //
+ virtual bool CanChangeEncodingParam(CodecInst& codecInst);
+
+
+ ///////////////////////////////////////////////////////////////////////////
+ // CurrentRate()
+ // Call to get the current encoding rate of the encoder. This function
+ // should be overwritten for codecs whic automatically change their
+ // target rate. One example is iSAC. The output of the function is the
+ // current target rate.
+ //
+ // Output:
+ // -rateBitPerSec : the current target rate of the codec.
+ //
+ virtual void CurrentRate(
+ WebRtc_Word32& /* rateBitPerSec */)
+ {
+ return;
+ }
+
+ virtual void SaveDecoderParamSafe(
+ const WebRtcACMCodecParams* codecParams);
+
+ // &_inAudio[_inAudioIxWrite] always point to where new audio can be
+ // written to
+ WebRtc_Word16 _inAudioIxWrite;
+
+ // &_inAudio[_inAudioIxRead] points to where audio has to be read from
+ WebRtc_Word16 _inAudioIxRead;
+
+ WebRtc_Word16 _inTimestampIxWrite;
+
+ // Where the audio is stored before encoding,
+ // To save memory the following buffer can be allocated
+ // dynamically for 80ms depending on the sampling frequency
+ // of the codec.
+ WebRtc_Word16* _inAudio;
+ WebRtc_UWord32* _inTimestamp;
+
+ WebRtc_Word16 _frameLenSmpl;
+ WebRtc_UWord16 _noChannels;
+
+ // This will point to a static database of the supported codecs
+ WebRtc_Word16 _codecID;
+
+ // This will account for the No of samples were not encoded
+ // the case is rare, either samples are missed due to overwite
+ // at input buffer or due to encoding error
+ WebRtc_UWord32 _noMissedSamples;
+
+ // True if the encoder instance created
+ bool _encoderExist;
+ bool _decoderExist;
+ // True if the ecncoder instance initialized
+ bool _encoderInitialized;
+ bool _decoderInitialized;
+
+ bool _registeredInNetEq;
+
+ // VAD/DTX
+ bool _hasInternalDTX;
+ WebRtcVadInst* _ptrVADInst;
+ bool _vadEnabled;
+ ACMVADMode _vadMode;
+ WebRtc_Word16 _vadLabel[MAX_FRAME_SIZE_10MSEC];
+ bool _dtxEnabled;
+ WebRtcCngEncInst* _ptrDTXInst;
+ WebRtc_UWord8 _numLPCParams;
+ bool _sentCNPrevious;
+ bool _isMaster;
+
+ WebRtcACMCodecParams _encoderParams;
+ WebRtcACMCodecParams _decoderParams;
+
+ // Used as a global lock for all avaiable decoders
+ // so that no decoder is used when NetEQ decodes.
+ RWLockWrapper* _netEqDecodeLock;
+ // Used to lock wrapper internal data
+ // such as buffers and state variables.
+ RWLockWrapper& _codecWrapperLock;
+
+ WebRtc_UWord32 _lastEncodedTimestamp;
+ WebRtc_UWord32 _lastTimestamp;
+ bool _isAudioBuffFresh;
+ WebRtc_UWord32 _uniqueID;
+};
+
+} // namespace webrt
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc
new file mode 100644
index 0000000..c1f817c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -0,0 +1,416 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_gsmfr.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_GSMFR
+ // NOTE! GSM-FR is not included in the open-source package. The following
+ // interface file is needed:
+ //
+ // /modules/audio_coding/codecs/gsmfr/main/interface/gsmfr_interface.h
+ //
+ // The API in the header file should match the one below.
+ //
+ // int16_t WebRtcGSMFR_CreateEnc(GSMFR_encinst_t_** inst);
+ // int16_t WebRtcGSMFR_CreateDec(GSMFR_decinst_t_** inst);
+ // int16_t WebRtcGSMFR_FreeEnc(GSMFR_encinst_t_* inst);
+ // int16_t WebRtcGSMFR_FreeDec(GSMFR_decinst_t_* inst);
+ // int16_t WebRtcGSMFR_Encode(GSMFR_encinst_t_* encInst, int16_t* input,
+ // int16_t len, int16_t* output);
+ // int16_t WebRtcGSMFR_EncoderInit(GSMFR_encinst_t_* encInst, int16_t mode);
+ // int16_t WebRtcGSMFR_Decode(GSMFR_decinst_t_* decInst);
+ // int16_t WebRtcGSMFR_DecodeBwe(GSMFR_decinst_t_* decInst, int16_t* input);
+ // int16_t WebRtcGSMFR_DecodePlc(GSMFR_decinst_t_* decInst);
+ // int16_t WebRtcGSMFR_DecoderInit(GSMFR_decinst_t_* decInst);
+ // void WebRtcGSMFR_Version(char *versionStr, short len);
+ #include "gsmfr_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_GSMFR
+
+ACMGSMFR::ACMGSMFR(WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL) {
+ return;
+}
+
+
+ACMGSMFR::~ACMGSMFR()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::EnableDTX()
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::DisableDTX()
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMGSMFR::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMGSMFR::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMGSMFR::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMGSMFR::DestructDecoderSafe()
+{
+ return;
+}
+
+
+void
+ACMGSMFR::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+
+#else //===================== Actual Implementation =======================
+
+ACMGSMFR::ACMGSMFR(
+ WebRtc_Word16 codecID):
+_encoderInstPtr(NULL),
+_decoderInstPtr(NULL)
+{
+ _codecID = codecID;
+ _hasInternalDTX = true;
+ return;
+}
+
+
+ACMGSMFR::~ACMGSMFR()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcGSMFR_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcGSMFR_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ *bitStreamLenByte = WebRtcGSMFR_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl;
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::EnableDTX()
+{
+ if(_dtxEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist)
+ {
+ if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 1) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "EnableDTX: cannot init encoder for GSMFR");
+ return -1;
+ }
+ _dtxEnabled = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+WebRtc_Word16
+ACMGSMFR::DisableDTX()
+{
+ if(!_dtxEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist)
+ {
+ if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 0) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "DisableDTX: cannot init encoder for GSMFR");
+ return -1;
+ }
+ _dtxEnabled = false;
+ return 0;
+ }
+ else
+ {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, ((codecParams->enableDTX)? 1:0)) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitEncoder: cannot init encoder for GSMFR");
+ }
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ if (WebRtcGSMFR_DecoderInit(_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: cannot init decoder for GSMFR");
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMGSMFR::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodecDef: decoder is not initialized for GSMFR");
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_GSMFR_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderGSMFR, codecInst.pltype,
+ _decoderInstPtr, 8000);
+ SET_GSMFR_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMGSMFR::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalCreateEncoder()
+{
+ if (WebRtcGSMFR_CreateEnc(&_encoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: cannot create instance for GSMFR encoder");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMGSMFR::DestructEncoderSafe()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcGSMFR_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ _encoderExist = false;
+ _encoderInitialized = false;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::InternalCreateDecoder()
+{
+ if (WebRtcGSMFR_CreateDec(&_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: cannot create instance for GSMFR decoder");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMGSMFR::DestructDecoderSafe()
+{
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcGSMFR_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ _decoderExist = false;
+ _decoderInitialized = false;
+}
+
+
+void
+ACMGSMFR::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*)ptrInst);
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMGSMFR::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec: payload-type does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderGSMFR);
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h
new file mode 100644
index 0000000..43e05fc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct GSMFR_encinst_t_;
+struct GSMFR_decinst_t_;
+
+namespace webrtc {
+
+class ACMGSMFR : public ACMGenericCodec
+{
+public:
+ ACMGSMFR(WebRtc_Word16 codecID);
+ ~ACMGSMFR();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 EnableDTX();
+
+ WebRtc_Word16 DisableDTX();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ GSMFR_encinst_t_* _encoderInstPtr;
+ GSMFR_decinst_t_* _decoderInstPtr;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc b/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc
new file mode 100644
index 0000000..f16e2f8
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc
@@ -0,0 +1,390 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_ilbc.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_ILBC
+ #include "ilbc.h"
+#endif
+
+namespace webrtc
+{
+
+#ifndef WEBRTC_CODEC_ILBC
+
+ACMILBC::ACMILBC(WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL) {
+ return;
+}
+
+
+ACMILBC::~ACMILBC()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMILBC::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMILBC::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMILBC::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMILBC::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMILBC::DestructDecoderSafe()
+{
+ return;
+}
+
+
+void
+ACMILBC::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+WebRtc_Word16
+ACMILBC::SetBitRateSafe(const WebRtc_Word32 /* rate */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMILBC::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+
+#else //===================== Actual Implementation =======================
+
+
+ACMILBC::ACMILBC(
+ WebRtc_Word16 codecID):
+_encoderInstPtr(NULL),
+_decoderInstPtr(NULL)
+{
+ _codecID = codecID;
+ return;
+}
+
+
+ACMILBC::~ACMILBC()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ *bitStreamLenByte = WebRtcIlbcfix_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
+ if (*bitStreamLenByte < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: error in encode for ILBC");
+ return -1;
+ }
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl;
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMILBC::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ // initialize with a correct processing block length
+ if((160 == (codecParams->codecInstant).pacsize) ||
+ (320 == (codecParams->codecInstant).pacsize))
+ {
+ // processing block of 20ms
+ return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
+ }
+ else if((240 == (codecParams->codecInstant).pacsize) ||
+ (480 == (codecParams->codecInstant).pacsize))
+ {
+ // processing block of 30ms
+ return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitEncoder: invalid processing block");
+ return -1;
+ }
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalInitDecoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ // initialize with a correct processing block length
+ if((160 == (codecParams->codecInstant).pacsize) ||
+ (320 == (codecParams->codecInstant).pacsize))
+ {
+ // processing block of 20ms
+ return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 20);
+ }
+ else if((240 == (codecParams->codecInstant).pacsize) ||
+ (480 == (codecParams->codecInstant).pacsize))
+ {
+ // processing block of 30ms
+ return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 30);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: invalid processing block");
+ return -1;
+ }
+}
+
+
+WebRtc_Word32
+ACMILBC::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: decoder not initialized for ILBC");
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_ILBC_FUNCTION."
+ // Then return the structure back to NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderILBC, codecInst.pltype,
+ _decoderInstPtr, 8000);
+ SET_ILBC_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMILBC::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalCreateEncoder()
+{
+ if (WebRtcIlbcfix_EncoderCreate(&_encoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: cannot create instance for ILBC encoder");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMILBC::DestructEncoderSafe()
+{
+ _encoderInitialized = false;
+ _encoderExist = false;
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+}
+
+
+WebRtc_Word16
+ACMILBC::InternalCreateDecoder()
+{
+ if (WebRtcIlbcfix_DecoderCreate(&_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: cannot create instance for ILBC decoder");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMILBC::DestructDecoderSafe()
+{
+ _decoderInitialized = false;
+ _decoderExist = false;
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+}
+
+
+void
+ACMILBC::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WebRtcIlbcfix_EncoderFree((iLBC_encinst_t_*)ptrInst);
+ }
+ return;
+}
+
+WebRtc_Word16
+ACMILBC::SetBitRateSafe(const WebRtc_Word32 rate)
+{
+ // Check that rate is valid. No need to store the value
+ if (rate == 13300)
+ {
+ WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
+ }
+ else if (rate == 15200)
+ {
+ WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
+ }
+ else
+ {
+ return -1;
+ }
+ _encoderParams.codecInstant.rate = rate;
+
+ return 0;
+}
+
+WebRtc_Word16
+ACMILBC::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec: given payload-type does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderILBC);
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_ilbc.h b/trunk/src/modules/audio_coding/main/source/acm_ilbc.h
new file mode 100644
index 0000000..2d3e420
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_ilbc.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct iLBC_encinst_t_;
+struct iLBC_decinst_t_;
+
+namespace webrtc
+{
+
+class ACMILBC : public ACMGenericCodec
+{
+public:
+ ACMILBC(WebRtc_Word16 codecID);
+ ~ACMILBC();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+
+ WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 rate);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ iLBC_encinst_t_* _encoderInstPtr;
+ iLBC_decinst_t_* _decoderInstPtr;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac.cc b/trunk/src/modules/audio_coding/main/source/acm_isac.cc
new file mode 100644
index 0000000..adbb9c5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_isac.cc
@@ -0,0 +1,1241 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_isac.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+
+#ifdef WEBRTC_CODEC_ISAC
+ #include "acm_isac_macros.h"
+ #include "isac.h"
+#endif
+
+#ifdef WEBRTC_CODEC_ISACFX
+ #include "acm_isac_macros.h"
+ #include "isacfix.h"
+#endif
+
+namespace webrtc
+{
+
+// we need this otherwise we cannot use forward declaration
+// in the header file
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+struct ACMISACInst
+{
+ ACM_ISAC_STRUCT *inst;
+};
+#endif
+
+#define ISAC_MIN_RATE 10000
+#define ISAC_MAX_RATE 56000
+
+
+// How the scaling is computed. iSAC computes a gain based on the
+// bottleneck. It follows the following expression for that
+//
+// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0)
+// / 3.4641;
+//
+// Where for 30 ms framelength we have,
+//
+// a = -23; b = 0.48; c = 0;
+//
+// As the default encoder is operating at 32kbps we have the scale as
+//
+// S(BN_kbps) = G(BN_kbps) / G(32);
+
+#define ISAC_NUM_SUPPORTED_RATES 9
+const WebRtc_UWord16 isacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
+ 32000, 30000, 26000, 23000, 21000,
+ 19000, 17000, 15000, 12000};
+
+const float isacScale[ISAC_NUM_SUPPORTED_RATES] = {
+ 1.0f, 0.8954f, 0.7178f, 0.6081f, 0.5445f,
+ 0.4875f, 0.4365f, 0.3908f, 0.3311f};
+
+// Tables for bandwidth estimates
+#define NR_ISAC_BANDWIDTHS 24
+const WebRtc_Word32 isacRatesWB[NR_ISAC_BANDWIDTHS] =
+{
+ 10000, 11100, 12300, 13700, 15200, 16900,
+ 18800, 20900, 23300, 25900, 28700, 31900,
+ 10100, 11200, 12400, 13800, 15300, 17000,
+ 18900, 21000, 23400, 26000, 28800, 32000};
+
+
+const WebRtc_Word32 isacRatesSWB[NR_ISAC_BANDWIDTHS] =
+{
+ 10000, 11000, 12400, 13800, 15300, 17000,
+ 18900, 21000, 23200, 25400, 27600, 29800,
+ 32000, 34100, 36300, 38500, 40700, 42900,
+ 45100, 47300, 49500, 51700, 53900, 56000,
+};
+
+#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))
+
+ACMISAC::ACMISAC(WebRtc_Word16 /* codecID */)
+ : _codecInstPtr(NULL),
+ _isEncInitialized(false),
+ _isacCodingMode(CHANNEL_INDEPENDENT),
+ _enforceFrameSize(false),
+ _isacCurrentBN(32000),
+ _samplesIn10MsAudio(160) { // Initiates to 16 kHz mode.
+ // Initiate decoder parameters for the 32 kHz mode.
+ memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
+ _decoderParams32kHz.codecInstant.pltype = -1;
+
+ return;
+}
+
+
+ACMISAC::~ACMISAC()
+{
+ return;
+}
+
+
+ACMGenericCodec*
+ACMISAC::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalEncode(
+ WebRtc_UWord8* /* bitstream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMISAC::DestructDecoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMISAC::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word32
+ACMISAC::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+void
+ACMISAC::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+WebRtc_Word16
+ACMISAC::DeliverCachedIsacData(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */,
+ WebRtc_UWord32* /* timestamp */,
+ WebRtcACMEncodingType* /* encodingType */,
+ const WebRtc_UWord16 /* isacRate */,
+ const WebRtc_UWord8 /* isacBWestimate */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::Transcode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */,
+ WebRtc_Word16 /* qBWE */,
+ WebRtc_Word32 /* scale */,
+ bool /* isRED */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMISAC::SetBitRateSafe(
+ WebRtc_Word32 /* bitRate */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::GetEstimatedBandwidthSafe()
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::SetEstimatedBandwidthSafe(
+ WebRtc_Word32 /* estimatedBandwidth */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::GetRedPayloadSafe(
+ WebRtc_UWord8* /* redPayload */,
+ WebRtc_Word16* /* payloadBytes */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::UpdateDecoderSampFreq(
+ WebRtc_Word16 /* codecId */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMISAC::UpdateEncoderSampFreq(
+ WebRtc_UWord16 /* encoderSampFreqHz */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMISAC::EncoderSampFreq(
+ WebRtc_UWord16& /* sampFreqHz */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 /* initFrameSizeMsec */,
+ const WebRtc_UWord16 /* initRateBitPerSec */,
+ const bool /* enforceFrameSize */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::SetISACMaxPayloadSize(
+ const WebRtc_UWord16 /* maxPayloadLenBytes */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMISAC::SetISACMaxRate(
+ const WebRtc_UWord32 /* maxRateBitPerSec */)
+{
+ return -1;
+}
+
+
+void
+ACMISAC::UpdateFrameLen()
+{
+ return;
+}
+
+void
+ACMISAC::CurrentRate(
+ WebRtc_Word32& /*rateBitPerSec */)
+{
+ return;
+}
+
+bool
+ACMISAC::DecoderParamsSafe(
+ WebRtcACMCodecParams* /* decParams */,
+ const WebRtc_UWord8 /* payloadType */)
+{
+ return false;
+}
+
+void
+ACMISAC::SaveDecoderParamSafe(
+ const WebRtcACMCodecParams* /* codecParams */)
+{
+ return;
+}
+
+WebRtc_Word16
+ACMISAC::REDPayloadISAC(
+ const WebRtc_Word32 /* isacRate */,
+ const WebRtc_Word16 /* isacBwEstimate */,
+ WebRtc_UWord8* /* payload */,
+ WebRtc_Word16* /* payloadLenBytes */)
+{
+ return -1;
+}
+
+
+#else //===================== Actual Implementation =======================
+
+
+
+#ifdef WEBRTC_CODEC_ISACFX
+
+enum IsacSamplingRate
+{
+ kIsacWideband = 16,
+ kIsacSuperWideband = 32
+};
+
+static float
+ACMISACFixTranscodingScale(
+ WebRtc_UWord16 rate)
+{
+ // find the scale for transcoding, the scale is rounded
+ // downward
+ float scale = -1;
+ for(WebRtc_Word16 n=0; n < ISAC_NUM_SUPPORTED_RATES; n++)
+ {
+ if(rate >= isacSuportedRates[n])
+ {
+ scale = isacScale[n];
+ break;
+ }
+ }
+ return scale;
+}
+
+static void
+ACMISACFixGetSendBitrate(
+ ACM_ISAC_STRUCT* inst,
+ WebRtc_Word32* bottleNeck)
+{
+ *bottleNeck = WebRtcIsacfix_GetUplinkBw(inst);
+}
+
+static WebRtc_Word16
+ACMISACFixGetNewBitstream(
+ ACM_ISAC_STRUCT* inst,
+ WebRtc_Word16 BWEIndex,
+ WebRtc_Word16 /* jitterIndex */,
+ WebRtc_Word32 rate,
+ WebRtc_Word16* bitStream,
+ bool isRED)
+{
+ if (isRED)
+ {
+ // RED not supported with iSACFIX
+ return -1;
+ }
+ float scale = ACMISACFixTranscodingScale((WebRtc_UWord16)rate);
+ return WebRtcIsacfix_GetNewBitStream(inst, BWEIndex, scale, bitStream);
+}
+
+
+static WebRtc_Word16
+ACMISACFixGetSendBWE(
+ ACM_ISAC_STRUCT* inst,
+ WebRtc_Word16* rateIndex,
+ WebRtc_Word16* /* dummy */)
+{
+ WebRtc_Word16 localRateIndex;
+ WebRtc_Word16 status = WebRtcIsacfix_GetDownLinkBwIndex(inst, &localRateIndex);
+ if(status < 0)
+ {
+ return -1;
+ }
+ else
+ {
+ *rateIndex = localRateIndex;
+ return 0;
+ }
+}
+
+static WebRtc_Word16
+ACMISACFixControlBWE(
+ ACM_ISAC_STRUCT* inst,
+ WebRtc_Word32 rateBPS,
+ WebRtc_Word16 frameSizeMs,
+ WebRtc_Word16 enforceFrameSize)
+{
+ return WebRtcIsacfix_ControlBwe(inst, (WebRtc_Word16)rateBPS,
+ frameSizeMs, enforceFrameSize);
+}
+
+static WebRtc_Word16
+ACMISACFixControl(
+ ACM_ISAC_STRUCT* inst,
+ WebRtc_Word32 rateBPS,
+ WebRtc_Word16 frameSizeMs)
+{
+ return WebRtcIsacfix_Control(inst, (WebRtc_Word16)rateBPS,
+ frameSizeMs);
+}
+
+static IsacSamplingRate
+ACMISACFixGetEncSampRate(
+ ACM_ISAC_STRUCT* /* inst */)
+{
+ return kIsacWideband;
+}
+
+
+static IsacSamplingRate
+ACMISACFixGetDecSampRate(
+ ACM_ISAC_STRUCT* /* inst */)
+{
+ return kIsacWideband;
+}
+
+#endif
+
+
+
+
+
+
+ACMISAC::ACMISAC(WebRtc_Word16 codecID)
+ : _isEncInitialized(false),
+ _isacCodingMode(CHANNEL_INDEPENDENT),
+ _enforceFrameSize(false),
+ _isacCurrentBN(32000),
+ _samplesIn10MsAudio(160) { // Initiates to 16 kHz mode.
+ _codecID = codecID;
+
+ // Create codec instance.
+ _codecInstPtr = new ACMISACInst;
+ if (_codecInstPtr == NULL) {
+ return;
+ }
+ _codecInstPtr->inst = NULL;
+
+ // Initiate decoder parameters for the 32 kHz mode.
+ memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
+ _decoderParams32kHz.codecInstant.pltype = -1;
+
+ // TODO(tlegrand): Check if the following is really needed, now that
+ // ACMGenericCodec has been updated to initialize this value.
+ // Initialize values that can be used uninitialized otherwise
+ _decoderParams.codecInstant.pltype = -1;
+}
+
+
+ACMISAC::~ACMISAC()
+{
+ if (_codecInstPtr != NULL)
+ {
+ if(_codecInstPtr->inst != NULL)
+ {
+ ACM_ISAC_FREE(_codecInstPtr->inst);
+ _codecInstPtr->inst = NULL;
+ }
+ delete _codecInstPtr;
+ _codecInstPtr = NULL;
+ }
+ return;
+}
+
+
+ACMGenericCodec*
+ACMISAC::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ // ISAC takes 10ms audio everytime we call encoder, therefor,
+ // it should be treated like codecs with 'basic coding block'
+ // non-zero, and the following 'while-loop' should not be necessary.
+ // However, due to a mistake in the codec the frame-size might change
+ // at the first 10ms pushed in to iSAC if the bit-rate is low, this is
+ // sort of a bug in iSAC. to address this we treat iSAC as the
+ // following.
+
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+ *bitStreamLenByte = 0;
+ while((*bitStreamLenByte == 0) && (_inAudioIxRead < _frameLenSmpl))
+ {
+ if(_inAudioIxRead > _inAudioIxWrite)
+ {
+ // something is wrong.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "The actual fram-size of iSAC appears to be larger that expected. All audio \
+pushed in but no bit-stream is generated.");
+ return -1;
+ }
+ *bitStreamLenByte = ACM_ISAC_ENCODE(_codecInstPtr->inst,
+ &_inAudio[_inAudioIxRead], (WebRtc_Word16*)bitstream);
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _samplesIn10MsAudio;
+ }
+ if(*bitStreamLenByte == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
+ "ISAC Has encoded the whole frame but no bit-stream is generated.");
+ }
+
+ // a packet is generated iSAC, is set in adaptive mode may change
+ // the frame length and we like to update the bottleneck value as
+ // well, although updating bottleneck is not crucial
+ if((*bitStreamLenByte > 0) && (_isacCodingMode == ADAPTIVE))
+ {
+ //_frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
+ ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
+ }
+ UpdateFrameLen();
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMISAC::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ // if rate is set to -1 then iSAC has to be in adaptive mode
+ if(codecParams->codecInstant.rate == -1)
+ {
+ _isacCodingMode = ADAPTIVE;
+ }
+
+ // sanity check that rate is in acceptable range
+ else if((codecParams->codecInstant.rate >= ISAC_MIN_RATE) &&
+ (codecParams->codecInstant.rate <= ISAC_MAX_RATE))
+ {
+ _isacCodingMode = CHANNEL_INDEPENDENT;
+ _isacCurrentBN = codecParams->codecInstant.rate;
+ }
+ else
+ {
+ return -1;
+ }
+
+ // we need to set the encoder sampling frequency.
+ if(UpdateEncoderSampFreq((WebRtc_UWord16)codecParams->codecInstant.plfreq) < 0)
+ {
+ return -1;
+ }
+ if(ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0)
+ {
+ return -1;
+ }
+
+ // apply the frame-size and rate if operating in
+ // channel-independent mode
+ if(_isacCodingMode == CHANNEL_INDEPENDENT)
+ {
+ if(ACM_ISAC_CONTROL(_codecInstPtr->inst,
+ codecParams->codecInstant.rate,
+ codecParams->codecInstant.pacsize /
+ (codecParams->codecInstant.plfreq / 1000)) < 0)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ // We need this for adaptive case and has to be called
+ // after initialization
+ ACM_ISAC_GETSENDBITRATE(
+ _codecInstPtr->inst, &_isacCurrentBN);
+ }
+ _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
+ return 0;
+}
+
+WebRtc_Word16
+ACMISAC::InternalInitDecoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+
+ // set decoder sampling frequency.
+ if(codecParams->codecInstant.plfreq == 32000)
+ {
+ UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
+ }
+ else
+ {
+ UpdateDecoderSampFreq(ACMCodecDB::kISAC);
+ }
+
+ // in a one-way communication we may never register send-codec.
+ // However we like that the BWE to work properly so it has to
+ // be initialized. The BWE is initialized when iSAC encoder is initialized.
+ // Therefore, we need this.
+ if(!_encoderInitialized)
+ {
+ // Since we don't require a valid rate or a valid packet size when initializing
+ // the decoder, we set valid values before initializing encoder
+ codecParams->codecInstant.rate = kIsacWbDefaultRate;
+ codecParams->codecInstant.pacsize = kIsacPacSize960;
+ if(InternalInitEncoder(codecParams) < 0)
+ {
+ return -1;
+ }
+ _encoderInitialized = true;
+ }
+
+ return ACM_ISAC_DECODERINIT(_codecInstPtr->inst);
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalCreateDecoder()
+{
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+ WebRtc_Word16 status = ACM_ISAC_CREATE (&(_codecInstPtr->inst));
+
+ // specific to codecs with one instance for encoding and decoding
+ _encoderInitialized = false;
+ if(status < 0)
+ {
+ _encoderExist = false;
+ }
+ else
+ {
+ _encoderExist = true;
+ }
+ return status;
+}
+
+
+void
+ACMISAC::DestructDecoderSafe()
+{
+ // codec with shared instance cannot delete.
+ _decoderInitialized = false;
+ return;
+}
+
+
+WebRtc_Word16
+ACMISAC::InternalCreateEncoder()
+{
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+ WebRtc_Word16 status = ACM_ISAC_CREATE(&(_codecInstPtr->inst));
+
+ // specific to codecs with one instance for encoding and decoding
+ _decoderInitialized = false;
+ if(status < 0)
+ {
+ _decoderExist = false;
+ }
+ else
+ {
+ _decoderExist = true;
+ }
+ return status;
+}
+
+
+void
+ACMISAC::DestructEncoderSafe()
+{
+ // codec with shared instance cannot delete.
+ _encoderInitialized = false;
+ return;
+}
+
+
+WebRtc_Word32
+ACMISAC::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ // Sanity checks
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+ if (!_decoderInitialized || !_decoderExist)
+ {
+ // Todo:
+ // log error
+ return -1;
+ }
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_ISAC_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ if(codecInst.plfreq == 16000)
+ {
+ SET_CODEC_PAR((codecDef), kDecoderISAC, codecInst.pltype,
+ _codecInstPtr->inst, 16000);
+#ifdef WEBRTC_CODEC_ISAC
+ SET_ISAC_FUNCTIONS((codecDef));
+#else
+ SET_ISACfix_FUNCTIONS((codecDef));
+#endif
+ }
+ else
+ {
+#ifdef WEBRTC_CODEC_ISAC
+ SET_CODEC_PAR((codecDef), kDecoderISACswb, codecInst.pltype,
+ _codecInstPtr->inst, 32000);
+ SET_ISACSWB_FUNCTIONS((codecDef));
+#else
+ return -1;
+#endif
+ }
+
+ return 0;
+}
+
+
+void
+ACMISAC::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ ACM_ISAC_FREE((ACM_ISAC_STRUCT *)ptrInst);
+ }
+ return;
+}
+
+WebRtc_Word16
+ACMISAC::Transcode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_Word16 qBWE,
+ WebRtc_Word32 rate,
+ bool isRED)
+{
+ WebRtc_Word16 jitterInfo = 0;
+ // transcode from a higher rate to lower rate
+ // sanity check
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+
+ *bitStreamLenByte = ACM_ISAC_GETNEWBITSTREAM(_codecInstPtr->inst,
+ qBWE, jitterInfo, rate, (WebRtc_Word16*)bitStream, (isRED)? 1:0);
+
+ if(*bitStreamLenByte < 0)
+ {
+ // error happened
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+ else
+ {
+ return *bitStreamLenByte;
+ }
+}
+
+WebRtc_Word16
+ACMISAC::SetBitRateSafe(
+ WebRtc_Word32 bitRate)
+{
+ if (_codecInstPtr == NULL)
+ {
+ return -1;
+ }
+ WebRtc_UWord16 encoderSampFreq;
+ EncoderSampFreq(encoderSampFreq);
+ bool reinit = false;
+ // change the BN of iSAC
+ if(bitRate == -1)
+ {
+ // ADAPTIVE MODE
+ // Check if it was already in adaptive mode
+ if(_isacCodingMode != ADAPTIVE)
+ {
+ // was not in adaptive, then set the mode to adaptive
+ // and flag for re-initialization
+ _isacCodingMode = ADAPTIVE;
+ reinit = true;
+ }
+ }
+ // Sanity check if the rate valid
+ else if((bitRate >= ISAC_MIN_RATE) &&
+ (bitRate <= ISAC_MAX_RATE))
+ {
+ //check if it was in channel-independent mode before
+ if(_isacCodingMode != CHANNEL_INDEPENDENT)
+ {
+ // was not in channel independent, set the mode to
+ // channel-independent and flag for re-initialization
+ _isacCodingMode = CHANNEL_INDEPENDENT;
+ reinit = true;
+ }
+ // store the bottleneck
+ _isacCurrentBN = (WebRtc_UWord16)bitRate;
+ }
+ else
+ {
+ // invlaid rate
+ return -1;
+ }
+
+ WebRtc_Word16 status = 0;
+ if(reinit)
+ {
+ // initialize and check if it is successful
+ if(ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0)
+ {
+ // failed initialization
+ return -1;
+ }
+ }
+ if(_isacCodingMode == CHANNEL_INDEPENDENT)
+ {
+
+ status = ACM_ISAC_CONTROL(_codecInstPtr->inst, _isacCurrentBN,
+ (encoderSampFreq == 32000)? 30:(_frameLenSmpl / 16));
+ if(status < 0)
+ {
+ status = -1;
+ }
+ }
+
+ // Update encoder parameters
+ _encoderParams.codecInstant.rate = bitRate;
+
+ UpdateFrameLen();
+ return status;
+}
+
+
+WebRtc_Word32
+ACMISAC::GetEstimatedBandwidthSafe()
+{
+ WebRtc_Word16 bandwidthIndex;
+ WebRtc_Word16 delayIndex;
+ IsacSamplingRate sampRate;
+
+ // Get bandwidth information
+ ACM_ISAC_GETSENDBWE(_codecInstPtr->inst, &bandwidthIndex, &delayIndex);
+
+ // Validy check of index
+ if ((bandwidthIndex < 0) || (bandwidthIndex >= NR_ISAC_BANDWIDTHS))
+ {
+ return -1;
+ }
+
+ // Check sample frequency
+ sampRate = ACM_ISAC_GETDECSAMPRATE(_codecInstPtr->inst);
+ if(sampRate == kIsacWideband)
+ {
+ return isacRatesWB[bandwidthIndex];
+ }
+ else
+ {
+ return isacRatesSWB[bandwidthIndex];
+ }
+}
+
+WebRtc_Word32
+ACMISAC::SetEstimatedBandwidthSafe(
+ WebRtc_Word32 estimatedBandwidth)
+{
+ IsacSamplingRate sampRate;
+ WebRtc_Word16 bandwidthIndex;
+
+ // Check sample frequency and choose appropriate table
+ sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
+
+ if(sampRate == kIsacWideband)
+ {
+ // Search through the WB rate table to find the index
+
+ bandwidthIndex = NR_ISAC_BANDWIDTHS/2 - 1;
+ for (int i=0; i<(NR_ISAC_BANDWIDTHS/2); i++)
+ {
+ if (estimatedBandwidth == isacRatesWB[i])
+ {
+ bandwidthIndex = i;
+ break;
+ } else if (estimatedBandwidth == isacRatesWB[i+NR_ISAC_BANDWIDTHS/2])
+ {
+ bandwidthIndex = i + NR_ISAC_BANDWIDTHS/2;
+ break;
+ } else if (estimatedBandwidth < isacRatesWB[i])
+ {
+ bandwidthIndex = i;
+ break;
+ }
+ }
+ }
+ else
+ {
+ // Search through the SWB rate table to find the index
+ bandwidthIndex = NR_ISAC_BANDWIDTHS - 1;
+ for (int i=0; i<NR_ISAC_BANDWIDTHS; i++)
+ {
+ if(estimatedBandwidth <= isacRatesSWB[i])
+ {
+ bandwidthIndex = i;
+ break;
+ }
+ }
+ }
+
+ // Set iSAC Bandwidth Estimate
+ ACM_ISAC_SETBWE(_codecInstPtr->inst, bandwidthIndex);
+
+ return 0;
+}
+
+WebRtc_Word32
+ACMISAC::GetRedPayloadSafe(
+#if (!defined(WEBRTC_CODEC_ISAC))
+ WebRtc_UWord8* /* redPayload */,
+ WebRtc_Word16* /* payloadBytes */)
+{
+ return -1;
+#else
+ WebRtc_UWord8* redPayload,
+ WebRtc_Word16* payloadBytes)
+{
+
+ WebRtc_Word16 bytes = WebRtcIsac_GetRedPayload(_codecInstPtr->inst, (WebRtc_Word16*)redPayload);
+ if (bytes < 0)
+ {
+ return -1;
+ }
+ *payloadBytes = bytes;
+ return 0;
+#endif
+}
+
+
+WebRtc_Word16
+ACMISAC::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType == _decoderParams.codecInstant.pltype)
+ {
+ return netEq->RemoveCodec(kDecoderISAC);
+ }
+ else if(payloadType == _decoderParams32kHz.codecInstant.pltype)
+ {
+ return netEq->RemoveCodec(kDecoderISACswb);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type %d or %d",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype,
+ _decoderParams32kHz.codecInstant.pltype);
+
+ return -1;
+ }
+}
+
+
+WebRtc_Word16
+ACMISAC::UpdateDecoderSampFreq(
+#ifdef WEBRTC_CODEC_ISAC
+ WebRtc_Word16 codecId)
+{
+ if(ACMCodecDB::kISAC == codecId)
+ {
+ return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacWideband);
+ }
+ else if(ACMCodecDB::kISACSWB == codecId)
+ {
+ return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacSuperWideband);
+ }
+ else
+ {
+ return -1;
+ }
+
+#else
+ WebRtc_Word16 /* codecId */)
+{
+ return 0;
+#endif
+}
+
+
+WebRtc_Word16
+ACMISAC::UpdateEncoderSampFreq(
+#ifdef WEBRTC_CODEC_ISAC
+ WebRtc_UWord16 encoderSampFreqHz)
+{
+ WebRtc_UWord16 currentSampRateHz;
+ EncoderSampFreq(currentSampRateHz);
+
+ if(currentSampRateHz != encoderSampFreqHz)
+ {
+ if((encoderSampFreqHz != 16000) && (encoderSampFreqHz != 32000))
+ {
+ return -1;
+ }
+ else
+ {
+ _inAudioIxRead = 0;
+ _inAudioIxWrite = 0;
+ _inTimestampIxWrite = 0;
+ if(encoderSampFreqHz == 16000)
+ {
+ if(WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacWideband) < 0)
+ {
+ return -1;
+ }
+ _samplesIn10MsAudio = 160;
+ }
+ else
+ {
+
+ if(WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacSuperWideband) < 0)
+ {
+ return -1;
+ }
+ _samplesIn10MsAudio = 320;
+ }
+ _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
+ _encoderParams.codecInstant.pacsize = _frameLenSmpl;
+ _encoderParams.codecInstant.plfreq = encoderSampFreqHz;
+ return 0;
+ }
+ }
+#else
+ WebRtc_UWord16 /* codecId */)
+{
+#endif
+ return 0;
+}
+
+WebRtc_Word16
+ACMISAC::EncoderSampFreq(
+ WebRtc_UWord16& sampFreqHz)
+{
+ IsacSamplingRate sampRate;
+ sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
+ if(sampRate == kIsacSuperWideband)
+ {
+ sampFreqHz = 32000;
+ }
+ else
+ {
+ sampFreqHz = 16000;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+ACMISAC::ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize)
+{
+ WebRtc_Word16 status;
+ {
+ WebRtc_UWord16 sampFreqHz;
+ EncoderSampFreq(sampFreqHz);
+ // @TODO: at 32kHz we hardcode calling with 30ms and enforce
+ // the frame-size otherwise we might get error. Revise if
+ // control-bwe is changed.
+ if(sampFreqHz == 32000)
+ {
+ status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst,
+ initRateBitPerSec, 30, 1);
+ }
+ else
+ {
+ status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst,
+ initRateBitPerSec, initFrameSizeMsec, enforceFrameSize? 1:0);
+ }
+ }
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Coutn't config iSAC BWE.");
+ return -1;
+ }
+ UpdateFrameLen();
+ ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
+ return 0;
+}
+
+WebRtc_Word32
+ACMISAC::SetISACMaxPayloadSize(
+ const WebRtc_UWord16 maxPayloadLenBytes)
+{
+ return ACM_ISAC_SETMAXPAYLOADSIZE(_codecInstPtr->inst, maxPayloadLenBytes);
+}
+
+WebRtc_Word32
+ACMISAC::SetISACMaxRate(
+ const WebRtc_UWord32 maxRateBitPerSec)
+{
+ return ACM_ISAC_SETMAXRATE(_codecInstPtr->inst, maxRateBitPerSec);
+}
+
+
+void
+ACMISAC::UpdateFrameLen()
+{
+ _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
+ _encoderParams.codecInstant.pacsize = _frameLenSmpl;
+}
+
+void
+ACMISAC::CurrentRate(WebRtc_Word32& rateBitPerSec)
+{
+ if(_isacCodingMode == ADAPTIVE)
+ {
+ ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &rateBitPerSec);
+ }
+}
+
+
+bool
+ACMISAC::DecoderParamsSafe(
+ WebRtcACMCodecParams* decParams,
+ const WebRtc_UWord8 payloadType)
+{
+ if(_decoderInitialized)
+ {
+ if(payloadType == _decoderParams.codecInstant.pltype)
+ {
+ memcpy(decParams, &_decoderParams, sizeof(WebRtcACMCodecParams));
+ return true;
+ }
+ if(payloadType == _decoderParams32kHz.codecInstant.pltype)
+ {
+ memcpy(decParams, &_decoderParams32kHz,
+ sizeof(WebRtcACMCodecParams));
+ return true;
+ }
+ }
+ return false;
+}
+
+void
+ACMISAC::SaveDecoderParamSafe(
+ const WebRtcACMCodecParams* codecParams)
+{
+ // set decoder sampling frequency.
+ if(codecParams->codecInstant.plfreq == 32000)
+ {
+ memcpy(&_decoderParams32kHz, codecParams, sizeof(WebRtcACMCodecParams));
+ }
+ else
+ {
+ memcpy(&_decoderParams, codecParams, sizeof(WebRtcACMCodecParams));
+ }
+}
+
+
+WebRtc_Word16
+ACMISAC::REDPayloadISAC(
+ const WebRtc_Word32 isacRate,
+ const WebRtc_Word16 isacBwEstimate,
+ WebRtc_UWord8* payload,
+ WebRtc_Word16* payloadLenBytes)
+{
+ WebRtc_Word16 status;
+ ReadLockScoped rl(_codecWrapperLock);
+ status = Transcode(payload, payloadLenBytes, isacBwEstimate, isacRate, true);
+ return status;
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac.h b/trunk/src/modules/audio_coding/main/source/acm_isac.h
new file mode 100644
index 0000000..6013c61
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_isac.h
@@ -0,0 +1,153 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc
+{
+
+struct ACMISACInst;
+
+enum iSACCodingMode {ADAPTIVE, CHANNEL_INDEPENDENT};
+
+
+class ACMISAC : public ACMGenericCodec
+{
+public:
+ ACMISAC(WebRtc_Word16 codecID);
+ ~ACMISAC();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 DeliverCachedIsacData(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_UWord32* timestamp,
+ WebRtcACMEncodingType* encodingType,
+ const WebRtc_UWord16 isacRate,
+ const WebRtc_UWord8 isacBWestimate);
+
+ WebRtc_Word16 DeliverCachedData(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */,
+ WebRtc_UWord32* /* timestamp */,
+ WebRtcACMEncodingType* /* encodingType */)
+ {
+ return -1;
+ }
+
+ WebRtc_Word16 UpdateDecoderSampFreq(
+ WebRtc_Word16 codecId);
+
+ WebRtc_Word16 UpdateEncoderSampFreq(
+ WebRtc_UWord16 sampFreqHz);
+
+ WebRtc_Word16 EncoderSampFreq(
+ WebRtc_UWord16& sampFreqHz);
+
+ WebRtc_Word32 ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize);
+
+ WebRtc_Word32 SetISACMaxPayloadSize(
+ const WebRtc_UWord16 maxPayloadLenBytes);
+
+ WebRtc_Word32 SetISACMaxRate(
+ const WebRtc_UWord32 maxRateBitPerSec);
+
+ WebRtc_Word16 REDPayloadISAC(
+ const WebRtc_Word32 isacRate,
+ const WebRtc_Word16 isacBwEstimate,
+ WebRtc_UWord8* payload,
+ WebRtc_Word16* payloadLenBytes);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 bitRate);
+
+ WebRtc_Word32 GetEstimatedBandwidthSafe();
+
+ WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
+
+ WebRtc_Word32 GetRedPayloadSafe(
+ WebRtc_UWord8* redPayload,
+ WebRtc_Word16* payloadBytes);
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 Transcode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte,
+ WebRtc_Word16 qBWE,
+ WebRtc_Word32 rate,
+ bool isRED);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void CurrentRate(WebRtc_Word32& rateBitPerSec);
+
+ void UpdateFrameLen();
+
+ bool DecoderParamsSafe(
+ WebRtcACMCodecParams *decParams,
+ const WebRtc_UWord8 payloadType);
+
+ void SaveDecoderParamSafe(
+ const WebRtcACMCodecParams* codecParams);
+
+ ACMISACInst* _codecInstPtr;
+
+ bool _isEncInitialized;
+ iSACCodingMode _isacCodingMode;
+ bool _enforceFrameSize;
+ WebRtc_Word32 _isacCurrentBN;
+ WebRtc_UWord16 _samplesIn10MsAudio;
+ WebRtcACMCodecParams _decoderParams32kHz;
+};
+
+} //namespace
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h b/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h
new file mode 100644
index 0000000..9e5f55f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
+
+#include "engine_configurations.h"
+
+namespace webrtc
+{
+
+#ifdef WEBRTC_CODEC_ISAC
+# define ACM_ISAC_CREATE WebRtcIsac_Create
+# define ACM_ISAC_FREE WebRtcIsac_Free
+# define ACM_ISAC_ENCODERINIT WebRtcIsac_EncoderInit
+# define ACM_ISAC_ENCODE WebRtcIsac_Encode
+# define ACM_ISAC_DECODERINIT WebRtcIsac_DecoderInit
+# define ACM_ISAC_DECODE_BWE WebRtcIsac_UpdateBwEstimate
+# define ACM_ISAC_DECODE_B WebRtcIsac_Decode
+# define ACM_ISAC_DECODEPLC WebRtcIsac_DecodePlc
+# define ACM_ISAC_CONTROL WebRtcIsac_Control
+# define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe
+# define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen
+# define ACM_ISAC_VERSION WebRtcIsac_version
+# define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode
+# define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw
+# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize
+# define ACM_ISAC_SETMAXRATE WebRtcIsac_SetMaxRate
+# define ACM_ISAC_GETNEWBITSTREAM WebRtcIsac_GetNewBitStream
+# define ACM_ISAC_GETSENDBWE WebRtcIsac_GetDownLinkBwIndex
+# define ACM_ISAC_SETBWE WebRtcIsac_UpdateUplinkBw
+# define ACM_ISAC_GETBWE WebRtcIsac_ReadBwIndex
+# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsac_GetNewFrameLen
+# define ACM_ISAC_STRUCT ISACStruct
+# define ACM_ISAC_GETENCSAMPRATE WebRtcIsac_EncSampRate
+# define ACM_ISAC_GETDECSAMPRATE WebRtcIsac_DecSampRate
+#endif
+
+#ifdef WEBRTC_CODEC_ISACFX
+# define ACM_ISAC_CREATE WebRtcIsacfix_Create
+# define ACM_ISAC_FREE WebRtcIsacfix_Free
+# define ACM_ISAC_ENCODERINIT WebRtcIsacfix_EncoderInit
+# define ACM_ISAC_ENCODE WebRtcIsacfix_Encode
+# define ACM_ISAC_DECODERINIT WebRtcIsacfix_DecoderInit
+# define ACM_ISAC_DECODE_BWE WebRtcIsacfix_UpdateBwEstimate
+# define ACM_ISAC_DECODE_B WebRtcIsacfix_Decode
+# define ACM_ISAC_DECODEPLC WebRtcIsacfix_DecodePlc
+# define ACM_ISAC_CONTROL ACMISACFixControl // local Impl
+# define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl
+# define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen
+# define ACM_ISAC_VERSION WebRtcIsacfix_version
+# define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode
+# define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl
+# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize
+# define ACM_ISAC_SETMAXRATE WebRtcIsacfix_SetMaxRate
+# define ACM_ISAC_GETNEWBITSTREAM ACMISACFixGetNewBitstream // local Impl
+# define ACM_ISAC_GETSENDBWE ACMISACFixGetSendBWE // local Impl
+# define ACM_ISAC_SETBWE WebRtcIsacfix_UpdateUplinkBw
+# define ACM_ISAC_GETBWE WebRtcIsacfix_ReadBwIndex
+# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsacfix_GetNewFrameLen
+# define ACM_ISAC_STRUCT ISACFIX_MainStruct
+# define ACM_ISAC_GETENCSAMPRATE ACMISACFixGetEncSampRate // local Impl
+# define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl
+#endif
+
+} //namespace
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
+
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq.cc b/trunk/src/modules/audio_coding/main/source/acm_neteq.cc
new file mode 100644
index 0000000..be25918
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_neteq.cc
@@ -0,0 +1,1270 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include <algorithm> // sort
+#include <stdlib.h> // malloc
+#include <vector>
+
+#include "acm_neteq.h"
+#include "common_types.h"
+#include "critical_section_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "signal_processing_library.h"
+#include "tick_util.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_internal.h"
+
+namespace webrtc
+{
+
+#define RTP_HEADER_SIZE 12
+#define NETEQ_INIT_FREQ 8000
+#define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000)
+#define NETEQ_ERR_MSG_LEN_BYTE (WEBRTC_NETEQ_MAX_ERROR_NAME + 1)
+
+
+ACMNetEQ::ACMNetEQ()
+:
+_id(0),
+_currentSampFreqKHz(NETEQ_INIT_FREQ_KHZ),
+_avtPlayout(false),
+_playoutMode(voice),
+_netEqCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+_vadStatus(false),
+_vadMode(VADNormal),
+_decodeLock(RWLockWrapper::CreateRWLock()),
+_numSlaves(0),
+_receivedStereo(false),
+_masterSlaveInfo(NULL),
+_previousAudioActivity(AudioFrame::kVadUnknown),
+_extraDelay(0),
+_callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
+{
+ for(int n = 0; n < MAX_NUM_SLAVE_NETEQ + 1; n++)
+ {
+ _isInitialized[n] = false;
+ _ptrVADInst[n] = NULL;
+ _inst[n] = NULL;
+ _instMem[n] = NULL;
+ _netEqPacketBuffer[n] = NULL;
+ }
+}
+
+ACMNetEQ::~ACMNetEQ()
+{
+ {
+ CriticalSectionScoped lock(*_netEqCritSect);
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if (_instMem[idx] != NULL)
+ {
+ free(_instMem[idx]);
+ _instMem[idx] = NULL;
+ }
+ if (_netEqPacketBuffer[idx] != NULL)
+ {
+ free(_netEqPacketBuffer[idx]);
+ _netEqPacketBuffer[idx] = NULL;
+ }
+ if(_ptrVADInst[idx] != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst[idx]);
+ _ptrVADInst[idx] = NULL;
+ }
+ }
+ if(_masterSlaveInfo != NULL)
+ {
+ free(_masterSlaveInfo);
+ _masterSlaveInfo = NULL;
+ }
+ }
+ if(_netEqCritSect != NULL)
+ {
+ delete _netEqCritSect;
+ }
+
+ if(_decodeLock != NULL)
+ {
+ delete _decodeLock;
+ }
+
+ if(_callbackCritSect != NULL)
+ {
+ delete _callbackCritSect;
+ }
+}
+
+WebRtc_Word32
+ACMNetEQ::Init()
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(InitByIdxSafe(idx) < 0)
+ {
+ return -1;
+ }
+ // delete VAD instance and start fresh if required.
+ if(_ptrVADInst[idx] != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst[idx]);
+ _ptrVADInst[idx] = NULL;
+ }
+ if(_vadStatus)
+ {
+ // Has to enable VAD
+ if(EnableVADByIdxSafe(idx) < 0)
+ {
+ // Failed to enable VAD.
+ // Delete VAD instance, if it is created
+ if(_ptrVADInst[idx] != NULL)
+ {
+ WebRtcVad_Free(_ptrVADInst[idx]);
+ _ptrVADInst[idx] = NULL;
+ }
+ // We are at initialization of NetEq, if failed to
+ // enable VAD, we delete the NetEq instance.
+ if (_instMem[idx] != NULL) {
+ free(_instMem[idx]);
+ _instMem[idx] = NULL;
+ _inst[idx] = NULL;
+ }
+ _isInitialized[idx] = false;
+ return -1;
+ }
+ }
+ _isInitialized[idx] = true;
+ }
+ if (EnableVAD() == -1)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::InitByIdxSafe(
+ const WebRtc_Word16 idx)
+{
+ int memorySizeBytes;
+ if (WebRtcNetEQ_AssignSize(&memorySizeBytes) != 0)
+ {
+ LogError("AssignSize", idx);
+ return -1;
+ }
+
+ if(_instMem[idx] != NULL)
+ {
+ free(_instMem[idx]);
+ _instMem[idx] = NULL;
+ }
+ _instMem[idx] = malloc(memorySizeBytes);
+ if (_instMem[idx] == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "InitByIdxSafe: NetEq Initialization error: could not allocate memory for NetEq");
+ _isInitialized[idx] = false;
+ return -1;
+ }
+ if (WebRtcNetEQ_Assign(&_inst[idx], _instMem[idx]) != 0)
+ {
+ if (_instMem[idx] != NULL) {
+ free(_instMem[idx]);
+ _instMem[idx] = NULL;
+ }
+ LogError("Assign", idx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "InitByIdxSafe: NetEq Initialization error: could not Assign");
+ _isInitialized[idx] = false;
+ return -1;
+ }
+ if (WebRtcNetEQ_Init(_inst[idx], NETEQ_INIT_FREQ) != 0)
+ {
+ if (_instMem[idx] != NULL) {
+ free(_instMem[idx]);
+ _instMem[idx] = NULL;
+ }
+ LogError("Init", idx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "InitByIdxSafe: NetEq Initialization error: could not initialize NetEq");
+ _isInitialized[idx] = false;
+ return -1;
+ }
+ _isInitialized[idx] = true;
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::EnableVADByIdxSafe(
+ const WebRtc_Word16 idx)
+{
+ if(_ptrVADInst[idx] == NULL)
+ {
+ if(WebRtcVad_Create(&_ptrVADInst[idx]) < 0)
+ {
+ _ptrVADInst[idx] = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "EnableVADByIdxSafe: NetEq Initialization error: could not create VAD");
+ return -1;
+ }
+ }
+
+ if(WebRtcNetEQ_SetVADInstance(_inst[idx], _ptrVADInst[idx],
+ (WebRtcNetEQ_VADInitFunction) WebRtcVad_Init,
+ (WebRtcNetEQ_VADSetmodeFunction) WebRtcVad_set_mode,
+ (WebRtcNetEQ_VADFunction) WebRtcVad_Process) < 0)
+ {
+ LogError("setVADinstance", idx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "EnableVADByIdxSafe: NetEq Initialization error: could not set VAD instance");
+ return -1;
+ }
+
+ if(WebRtcNetEQ_SetVADMode(_inst[idx], _vadMode) < 0)
+ {
+ LogError("setVADmode", idx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "EnableVADByIdxSafe: NetEq Initialization error: could not set VAD mode");
+ return -1;
+ }
+ return 0;
+}
+
+
+
+
+WebRtc_Word32
+ACMNetEQ::AllocatePacketBuffer(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs)
+{
+ // Due to WebRtcNetEQ_GetRecommendedBufferSize
+ // the following has to be int otherwise we will have compiler error
+ // if not casted
+
+ CriticalSectionScoped lock(*_netEqCritSect);
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, idx) < 0)
+ {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::AllocatePacketBufferByIdxSafe(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs,
+ const WebRtc_Word16 idx)
+{
+ int maxNoPackets;
+ int bufferSizeInBytes;
+
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AllocatePacketBufferByIdxSafe: NetEq is not initialized.");
+ return -1;
+ }
+ if (WebRtcNetEQ_GetRecommendedBufferSize(_inst[idx], usedCodecs, noOfCodecs,
+ kTCPLargeJitter , &maxNoPackets, &bufferSizeInBytes)
+ != 0)
+ {
+ LogError("GetRecommendedBufferSize", idx);
+ return -1;
+ }
+ if(_netEqPacketBuffer[idx] != NULL)
+ {
+ free(_netEqPacketBuffer[idx]);
+ _netEqPacketBuffer[idx] = NULL;
+ }
+
+ _netEqPacketBuffer[idx] = (WebRtc_Word16 *)malloc(bufferSizeInBytes);
+ if (_netEqPacketBuffer[idx] == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AllocatePacketBufferByIdxSafe: NetEq Initialization error: could not allocate "
+ "memory for NetEq Packet Buffer");
+ return -1;
+
+ }
+ if (WebRtcNetEQ_AssignBuffer(_inst[idx], maxNoPackets, _netEqPacketBuffer[idx],
+ bufferSizeInBytes) != 0)
+ {
+ if (_netEqPacketBuffer[idx] != NULL) {
+ free(_netEqPacketBuffer[idx]);
+ _netEqPacketBuffer[idx] = NULL;
+ }
+ LogError("AssignBuffer", idx);
+ return -1;
+ }
+ return 0;
+}
+
+
+
+
+WebRtc_Word32
+ACMNetEQ::SetExtraDelay(
+ const WebRtc_Word32 delayInMS)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetExtraDelay: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_SetExtraDelay(_inst[idx], delayInMS) < 0)
+ {
+ LogError("SetExtraDelay", idx);
+ return -1;
+ }
+ }
+ _extraDelay = delayInMS;
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMNetEQ::SetAVTPlayout(
+ const bool enable)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if (_avtPlayout != enable)
+ {
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetAVTPlayout: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_SetAVTPlayout(_inst[idx], (enable) ? 1 : 0) < 0)
+ {
+ LogError("SetAVTPlayout", idx);
+ return -1;
+ }
+ }
+ }
+ _avtPlayout = enable;
+ return 0;
+}
+
+
+bool
+ACMNetEQ::AVTPlayout() const
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ return _avtPlayout;
+}
+
+WebRtc_Word32
+ACMNetEQ::CurrentSampFreqHz() const
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "CurrentSampFreqHz: NetEq is not initialized.");
+ return -1;
+ }
+ return (WebRtc_Word32)(1000*_currentSampFreqKHz);
+}
+
+
+WebRtc_Word32
+ACMNetEQ::SetPlayoutMode(
+ const AudioPlayoutMode mode)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(_playoutMode != mode)
+ {
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetPlayoutMode: NetEq is not initialized.");
+ return -1;
+ }
+
+ enum WebRtcNetEQPlayoutMode playoutMode = kPlayoutOff;
+ switch(mode)
+ {
+ case voice:
+ playoutMode = kPlayoutOn;
+ break;
+ case fax:
+ playoutMode = kPlayoutFax;
+ break;
+ case streaming:
+ playoutMode = kPlayoutStreaming;
+ break;
+ }
+ if(WebRtcNetEQ_SetPlayoutMode(_inst[idx], playoutMode) < 0)
+ {
+ LogError("SetPlayoutMode", idx);
+ return -1;
+ }
+ }
+ _playoutMode = mode;
+ }
+
+ return 0;
+}
+
+AudioPlayoutMode
+ACMNetEQ::PlayoutMode() const
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ return _playoutMode;
+}
+
+
+WebRtc_Word32
+ACMNetEQ::NetworkStatistics(
+ ACMNetworkStatistics* statistics) const
+{
+ WebRtcNetEQ_NetworkStatistics stats;
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "NetworkStatistics: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_GetNetworkStatistics(_inst[0], &stats) == 0)
+ {
+ statistics->currentAccelerateRate = stats.currentAccelerateRate;
+ statistics->currentBufferSize = stats.currentBufferSize;
+ statistics->jitterPeaksFound = (stats.jitterPeaksFound > 0);
+ statistics->currentDiscardRate = stats.currentDiscardRate;
+ statistics->currentExpandRate = stats.currentExpandRate;
+ statistics->currentPacketLossRate = stats.currentPacketLossRate;
+ statistics->currentPreemptiveRate = stats.currentPreemptiveRate;
+ statistics->preferredBufferSize = stats.preferredBufferSize;
+ statistics->clockDriftPPM = stats.clockDriftPPM;
+ }
+ else
+ {
+ LogError("getNetworkStatistics", 0);
+ return -1;
+ }
+ const int kArrayLen = 100;
+ int waiting_times[kArrayLen];
+ int waiting_times_len = WebRtcNetEQ_GetRawFrameWaitingTimes(
+ _inst[0], kArrayLen, waiting_times);
+ if (waiting_times_len > 0)
+ {
+ std::vector<int> waiting_times_vec(waiting_times,
+ waiting_times + waiting_times_len);
+ std::sort(waiting_times_vec.begin(), waiting_times_vec.end());
+ size_t size = waiting_times_vec.size();
+ assert(size == static_cast<size_t>(waiting_times_len));
+ if (size % 2 == 0)
+ {
+ statistics->medianWaitingTimeMs =
+ (waiting_times_vec[size / 2 - 1] +
+ waiting_times_vec[size / 2]) / 2;
+ }
+ else
+ {
+ statistics->medianWaitingTimeMs = waiting_times_vec[size / 2];
+ }
+ statistics->minWaitingTimeMs = waiting_times_vec.front();
+ statistics->maxWaitingTimeMs = waiting_times_vec.back();
+ double sum = 0;
+ for (size_t i = 0; i < size; ++i) {
+ sum += waiting_times_vec[i];
+ }
+ statistics->meanWaitingTimeMs = static_cast<int>(sum / size);
+ }
+ else if (waiting_times_len == 0)
+ {
+ statistics->meanWaitingTimeMs = -1;
+ statistics->medianWaitingTimeMs = -1;
+ statistics->minWaitingTimeMs = -1;
+ statistics->maxWaitingTimeMs = -1;
+ }
+ else
+ {
+ LogError("getRawFrameWaitingTimes", 0);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+ACMNetEQ::RecIn(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo)
+{
+ // translate to NetEq struct
+ WebRtcNetEQ_RTPInfo netEqRTPInfo;
+ netEqRTPInfo.payloadType = rtpInfo.header.payloadType;
+ netEqRTPInfo.sequenceNumber = rtpInfo.header.sequenceNumber;
+ netEqRTPInfo.timeStamp = rtpInfo.header.timestamp;
+ netEqRTPInfo.SSRC = rtpInfo.header.ssrc;
+ netEqRTPInfo.markerBit = rtpInfo.header.markerBit;
+
+ CriticalSectionScoped lock(*_netEqCritSect);
+ // Down-cast the time to (32-6)-bit since we only care about
+ // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
+ // we masked 6 most significant bits of 32-bit so we don't loose resolution
+ // when do the following multiplication.
+ const WebRtc_UWord32 nowInMs = static_cast<WebRtc_UWord32>(
+ TickTime::MillisecondTimestamp() & 0x03ffffff);
+ WebRtc_UWord32 recvTimestamp = static_cast<WebRtc_UWord32>
+ (_currentSampFreqKHz * nowInMs);
+
+ int status;
+
+ if(rtpInfo.type.Audio.channel == 1)
+ {
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecIn: NetEq is not initialized.");
+ return -1;
+ }
+ // PUSH into Master
+ status = WebRtcNetEQ_RecInRTPStruct(_inst[0], &netEqRTPInfo,
+ (WebRtc_UWord8 *)incomingPayload, (WebRtc_Word16)payloadLength,
+ recvTimestamp);
+ if(status < 0)
+ {
+ LogError("RecInRTPStruct", 0);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecIn: NetEq, error in pushing in Master");
+ return -1;
+ }
+ }
+ else if(rtpInfo.type.Audio.channel == 2)
+ {
+ if(!_isInitialized[1])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecIn: NetEq is not initialized.");
+ return -1;
+ }
+ // PUSH into Slave
+ status = WebRtcNetEQ_RecInRTPStruct(_inst[1], &netEqRTPInfo,
+ (WebRtc_UWord8 *)incomingPayload, (WebRtc_Word16)payloadLength,
+ recvTimestamp);
+ if(status < 0)
+ {
+ LogError("RecInRTPStruct", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecIn: NetEq, error in pushing in Slave");
+ return -1;
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecIn: NetEq, error invalid numbe of channels %d \
+(1, for Master stream, and 2, for slave stream, are valid values)",
+ rtpInfo.type.Audio.channel);
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+ACMNetEQ::RecOut(
+ AudioFrame& audioFrame)
+{
+ enum WebRtcNetEQOutputType type;
+ WebRtc_Word16 payloadLenSample;
+ enum WebRtcNetEQOutputType typeMaster;
+ enum WebRtcNetEQOutputType typeSlave;
+
+ WebRtc_Word16 payloadLenSampleSlave;
+
+ CriticalSectionScoped lockNetEq(*_netEqCritSect);
+
+ if(!_receivedStereo)
+ {
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq is not initialized.");
+ return -1;
+ }
+ {
+ WriteLockScoped lockCodec(*_decodeLock);
+ if(WebRtcNetEQ_RecOut(_inst[0], &(audioFrame._payloadData[0]),
+ &payloadLenSample) != 0)
+ {
+ LogError("RecOut", 0);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq, error in pulling out for mono case");
+
+ // Check for errors that can be recovered from:
+ // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
+ int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
+ if(errorCode != 2003)
+ {
+ // Cannot recover; return an error
+ return -1;
+ }
+ }
+ }
+ WebRtcNetEQ_GetSpeechOutputType(_inst[0], &type);
+ audioFrame._audioChannel = 1;
+ }
+ else
+ {
+ if(!_isInitialized[0] || !_isInitialized[1])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq is not initialized.");
+ return -1;
+ }
+ WebRtc_Word16 payloadMaster[480];
+ WebRtc_Word16 payloadSlave[480];
+ {
+ WriteLockScoped lockCodec(*_decodeLock);
+ if(WebRtcNetEQ_RecOutMasterSlave(_inst[0], payloadMaster,
+ &payloadLenSample, _masterSlaveInfo, 1) != 0)
+ {
+ LogError("RecOutMasterSlave", 0);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq, error in pulling out for master");
+
+ // Check for errors that can be recovered from:
+ // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
+ int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
+ if(errorCode != 2003)
+ {
+ // Cannot recover; return an error
+ return -1;
+ }
+ }
+ if(WebRtcNetEQ_RecOutMasterSlave(_inst[1], payloadSlave,
+ &payloadLenSampleSlave, _masterSlaveInfo, 0) != 0)
+ {
+ LogError("RecOutMasterSlave", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq, error in pulling out for slave");
+
+ // Check for errors that can be recovered from:
+ // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
+ int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
+ if(errorCode != 2003)
+ {
+ // Cannot recover; return an error
+ return -1;
+ }
+ }
+ }
+ if(payloadLenSample != payloadLenSampleSlave)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
+ "RecOut: mismatch between the lenght of the decoded \
+audio by Master (%d samples) and Slave (%d samples).",
+ payloadLenSample, payloadLenSampleSlave);
+ if(payloadLenSample > payloadLenSampleSlave)
+ {
+ memset(&payloadSlave[payloadLenSampleSlave], 0,
+ (payloadLenSample - payloadLenSampleSlave) * sizeof(WebRtc_Word16));
+ }
+ }
+
+ for(WebRtc_Word16 n = 0; n < payloadLenSample; n++)
+ {
+ audioFrame._payloadData[n<<1] = payloadMaster[n];
+ audioFrame._payloadData[(n<<1)+1] = payloadSlave[n];
+ }
+ audioFrame._audioChannel = 2;
+
+ WebRtcNetEQ_GetSpeechOutputType(_inst[0], &typeMaster);
+ WebRtcNetEQ_GetSpeechOutputType(_inst[1], &typeSlave);
+ if((typeMaster == kOutputNormal) ||
+ (typeSlave == kOutputNormal))
+ {
+ type = kOutputNormal;
+ }
+ else
+ {
+ type = typeMaster;
+ }
+ }
+
+ audioFrame._payloadDataLengthInSamples = static_cast<WebRtc_UWord16>(payloadLenSample);
+ // NetEq always returns 10 ms of audio.
+ _currentSampFreqKHz = static_cast<float>(audioFrame._payloadDataLengthInSamples) / 10.0f;
+ audioFrame._frequencyInHz = audioFrame._payloadDataLengthInSamples * 100;
+ if(_vadStatus)
+ {
+ if(type == kOutputVADPassive)
+ {
+ audioFrame._vadActivity = AudioFrame::kVadPassive;
+ audioFrame._speechType = AudioFrame::kNormalSpeech;
+ }
+ else if(type == kOutputNormal)
+ {
+ audioFrame._vadActivity = AudioFrame::kVadActive;
+ audioFrame._speechType = AudioFrame::kNormalSpeech;
+ }
+ else if(type == kOutputPLC)
+ {
+ audioFrame._vadActivity = _previousAudioActivity;
+ audioFrame._speechType = AudioFrame::kPLC;
+ }
+ else if(type == kOutputCNG)
+ {
+ audioFrame._vadActivity = AudioFrame::kVadPassive;
+ audioFrame._speechType = AudioFrame::kCNG;
+ }
+ else
+ {
+ audioFrame._vadActivity = AudioFrame::kVadPassive;
+ audioFrame._speechType = AudioFrame::kPLCCNG;
+ }
+ }
+ else
+ {
+ // Always return kVadUnknown when receive VAD is inactive
+ audioFrame._vadActivity = AudioFrame::kVadUnknown;
+
+ if(type == kOutputNormal)
+ {
+ audioFrame._speechType = AudioFrame::kNormalSpeech;
+ }
+ else if(type == kOutputPLC)
+ {
+ audioFrame._speechType = AudioFrame::kPLC;
+ }
+ else if(type == kOutputPLCtoCNG)
+ {
+ audioFrame._speechType = AudioFrame::kPLCCNG;
+ }
+ else if(type == kOutputCNG)
+ {
+ audioFrame._speechType = AudioFrame::kCNG;
+ }
+ else
+ {
+ // type is kOutputVADPassive which
+ // we don't expect to get if _vadStatus is false
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
+ "RecOut: NetEq returned kVadPassive while _vadStatus is false.");
+ audioFrame._vadActivity = AudioFrame::kVadUnknown;
+ audioFrame._speechType = AudioFrame::kNormalSpeech;
+ }
+ }
+ _previousAudioActivity = audioFrame._vadActivity;
+
+ return 0;
+}
+
+// When ACMGenericCodec has set the codec specific parameters in codecDef
+// it calls AddCodec() to add the new codec to the NetEQ database.
+WebRtc_Word32
+ACMNetEQ::AddCodec(
+ WebRtcNetEQ_CodecDef* codecDef,
+ bool toMaster)
+{
+ if (codecDef == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "ACMNetEQ::AddCodec: error, codecDef is NULL");
+ return -1;
+ }
+ CriticalSectionScoped lock(*_netEqCritSect);
+
+ WebRtc_Word16 idx;
+ if(toMaster)
+ {
+ idx = 0;
+ }
+ else
+ {
+ idx = 1;
+ }
+
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "ACMNetEQ::AddCodec: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_CodecDbAdd(_inst[idx], codecDef) < 0)
+ {
+ LogError("CodecDB_Add", idx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "ACMNetEQ::AddCodec: NetEq, error in adding codec");
+ return -1;
+ }
+ else
+ {
+ return 0;
+ }
+}
+
+// Creates a Word16 RTP packet out of a Word8 payload and an rtp info struct.
+// Must be byte order safe.
+void
+ACMNetEQ::RTPPack(
+ WebRtc_Word16* rtpPacket,
+ const WebRtc_Word8* payload,
+ const WebRtc_Word32 payloadLengthW8,
+ const WebRtcRTPHeader& rtpInfo)
+{
+ WebRtc_Word32 idx = 0;
+ WEBRTC_SPL_SET_BYTE(rtpPacket, (WebRtc_Word8)0x80, idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, rtpInfo.header.payloadType, idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.sequenceNumber), 1), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.sequenceNumber), 0), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.timestamp), 3), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.timestamp), 2), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.timestamp), 1), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.timestamp), 0), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.ssrc), 3), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.ssrc), 2), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.ssrc), 1), idx);
+ idx++;
+
+ WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
+ &(rtpInfo.header.ssrc), 0), idx);
+ idx++;
+
+ for (WebRtc_Word16 i=0; i < payloadLengthW8; i++)
+ {
+ WEBRTC_SPL_SET_BYTE(rtpPacket, payload[i], idx);
+ idx++;
+ }
+ if (payloadLengthW8 & 1)
+ {
+ // Our 16 bits buffer is one byte too large, set that
+ // last byte to zero.
+ WEBRTC_SPL_SET_BYTE(rtpPacket, 0x0, idx);
+ }
+}
+
+WebRtc_Word16
+ACMNetEQ::EnableVAD()
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if (_vadStatus)
+ {
+ return 0;
+ }
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetVADStatus: NetEq is not initialized.");
+ return -1;
+ }
+ // VAD was off and we have to turn it on
+ if(EnableVADByIdxSafe(idx) < 0)
+ {
+ return -1;
+ }
+
+ // Set previous VAD status to PASSIVE
+ _previousAudioActivity = AudioFrame::kVadPassive;
+ }
+ _vadStatus = true;
+ return 0;
+}
+
+
+ACMVADMode
+ACMNetEQ::VADMode() const
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ return _vadMode;
+}
+
+
+WebRtc_Word16
+ACMNetEQ::SetVADMode(
+ const ACMVADMode mode)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if((mode < VADNormal) || (mode > VADVeryAggr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetVADMode: NetEq error: could not set VAD mode, mode is not supported");
+ return -1;
+ }
+ else
+ {
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetVADMode: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_SetVADMode(_inst[idx], mode) < 0)
+ {
+ LogError("SetVADmode", idx);
+ return -1;
+ }
+ }
+ _vadMode = mode;
+ return 0;
+ }
+}
+
+
+WebRtc_Word32
+ACMNetEQ::FlushBuffers()
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "FlushBuffers: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_FlushBuffers(_inst[idx]) < 0)
+ {
+ LogError("FlushBuffers", idx);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMNetEQ::GetVersion(
+ char* version,
+ WebRtc_UWord32& remainingBufferInBytes,
+ WebRtc_UWord32& position)
+{
+ WebRtc_UWord32 len = position;
+ strncpy(&version[position], "NetEq\t\t", remainingBufferInBytes);
+ position = (WebRtc_UWord32)strlen(version);
+ remainingBufferInBytes -= (position - len);
+ len = position;
+
+ char myVersion[100];
+ if(WebRtcNetEQ_GetVersion(myVersion) < 0)
+ {
+ return -1;
+ }
+
+ strncpy(&version[position], myVersion, remainingBufferInBytes);
+ position = (WebRtc_UWord32)strlen(version);
+ remainingBufferInBytes -= (position - len);
+ len = position;
+
+ strncpy(&version[position], "\n", remainingBufferInBytes);
+ position = (WebRtc_UWord32)strlen(version);
+ remainingBufferInBytes -= (position - len);
+ len = position;
+
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::RemoveCodec(
+ WebRtcNetEQDecoder codecIdx,
+ bool isStereo)
+{
+ // sanity check
+ if((codecIdx <= kDecoderReservedStart) ||
+ (codecIdx >= kDecoderReservedEnd))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RemoveCodec: NetEq error: could not Remove Codec, codec index out of range");
+ return -1;
+ }
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RemoveCodec: NetEq is not initialized.");
+ return -1;
+ }
+
+ if(WebRtcNetEQ_CodecDbRemove(_inst[0], codecIdx) < 0)
+ {
+ LogError("CodecDB_Remove", 0);
+ return -1;
+ }
+
+ if(isStereo)
+ {
+ if(WebRtcNetEQ_CodecDbRemove(_inst[1], codecIdx) < 0)
+ {
+ LogError("CodecDB_Remove", 1);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::SetBackgroundNoiseMode(
+ const ACMBackgroundNoiseMode mode)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
+ {
+ if(!_isInitialized[idx])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetBackgroundNoiseMode: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_SetBGNMode(_inst[idx], (WebRtcNetEQBGNMode)mode) < 0)
+ {
+ LogError("SetBGNMode", idx);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word16
+ACMNetEQ::BackgroundNoiseMode(
+ ACMBackgroundNoiseMode& mode)
+{
+ WebRtcNetEQBGNMode myMode;
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(!_isInitialized[0])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "BackgroundNoiseMode: NetEq is not initialized.");
+ return -1;
+ }
+ if(WebRtcNetEQ_GetBGNMode(_inst[0], &myMode) < 0)
+ {
+ LogError("WebRtcNetEQ_GetBGNMode", 0);
+ return -1;
+ }
+ else
+ {
+ mode = (ACMBackgroundNoiseMode)myMode;
+ }
+ return 0;
+}
+
+void
+ACMNetEQ::SetUniqueId(
+ WebRtc_Word32 id)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ _id = id;
+}
+
+
+void
+ACMNetEQ::LogError(
+ const char* neteqFuncName,
+ const WebRtc_Word16 idx) const
+{
+ char errorName[NETEQ_ERR_MSG_LEN_BYTE];
+ char myFuncName[50];
+ int neteqErrorCode = WebRtcNetEQ_GetErrorCode(_inst[idx]);
+ WebRtcNetEQ_GetErrorName(neteqErrorCode, errorName, NETEQ_ERR_MSG_LEN_BYTE - 1);
+ strncpy(myFuncName, neteqFuncName, 49);
+ errorName[NETEQ_ERR_MSG_LEN_BYTE - 1] = '\0';
+ myFuncName[49] = '\0';
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "NetEq-%d Error in function %s, error-code: %d, error-string: %s",
+ idx,
+ myFuncName,
+ neteqErrorCode,
+ errorName);
+}
+
+
+WebRtc_Word32
+ACMNetEQ::PlayoutTimestamp(
+ WebRtc_UWord32& timestamp)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ if(WebRtcNetEQ_GetSpeechTimeStamp(_inst[0], ×tamp) < 0)
+ {
+ LogError("GetSpeechTimeStamp", 0);
+ return -1;
+ }
+ else
+ {
+ return 0;
+ }
+}
+
+WebRtc_Word16
+ACMNetEQ::AddSlave(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ const WebRtc_Word16 slaveIdx = 1;
+ if(_numSlaves < 1)
+ {
+ // initialize the receiver, this also sets up VAD.
+ if(InitByIdxSafe(slaveIdx) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not Initialize");
+ return -1;
+ }
+
+ // Allocate buffer.
+ if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, slaveIdx) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not Allocate Packet Buffer");
+ return -1;
+ }
+
+ if(_masterSlaveInfo != NULL)
+ {
+ free(_masterSlaveInfo);
+ _masterSlaveInfo = NULL;
+ }
+ int msInfoSize = WebRtcNetEQ_GetMasterSlaveInfoSize();
+ _masterSlaveInfo = malloc(msInfoSize);
+
+ if(_masterSlaveInfo == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not Allocate memory for Master-Slave Info");
+ return -1;
+ }
+
+ // We accept this as initialized NetEQ, the rest is to synchronize
+ // Slave with Master.
+ _numSlaves = 1;
+ _isInitialized[slaveIdx] = true;
+
+ // Set Slave delay as all other instances.
+ if(WebRtcNetEQ_SetExtraDelay(_inst[slaveIdx], _extraDelay) < 0)
+ {
+ LogError("SetExtraDelay", slaveIdx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not set delay");
+ return -1;
+ }
+
+ // Set AVT
+ if(WebRtcNetEQ_SetAVTPlayout(_inst[slaveIdx], (_avtPlayout) ? 1 : 0) < 0)
+ {
+ LogError("SetAVTPlayout", slaveIdx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not set AVT playout.");
+ return -1;
+ }
+
+ // Set Background Noise
+ WebRtcNetEQBGNMode currentMode;
+ if(WebRtcNetEQ_GetBGNMode(_inst[0], ¤tMode) < 0)
+ {
+ LogError("GetBGNMode", 0);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AAddSlave: AddSlave Failed, Could not Get BGN form Master.");
+ return -1;
+ }
+
+ if(WebRtcNetEQ_SetBGNMode(_inst[slaveIdx], (WebRtcNetEQBGNMode)currentMode) < 0)
+ {
+ LogError("SetBGNMode", slaveIdx);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not set BGN mode.");
+ return -1;
+ }
+
+ enum WebRtcNetEQPlayoutMode playoutMode = kPlayoutOff;
+ switch(_playoutMode)
+ {
+ case voice:
+ playoutMode = kPlayoutOn;
+ break;
+ case fax:
+ playoutMode = kPlayoutFax;
+ break;
+ case streaming:
+ playoutMode = kPlayoutStreaming;
+ break;
+ }
+ if(WebRtcNetEQ_SetPlayoutMode(_inst[slaveIdx], playoutMode) < 0)
+ {
+ LogError("SetPlayoutMode", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "AddSlave: AddSlave Failed, Could not Set Playout Mode.");
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+void
+ACMNetEQ::SetReceivedStereo(
+ bool receivedStereo)
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ _receivedStereo = receivedStereo;
+}
+
+WebRtc_UWord8
+ACMNetEQ::NumSlaves()
+{
+ CriticalSectionScoped lock(*_netEqCritSect);
+ return _numSlaves;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq.h b/trunk/src/modules/audio_coding/main/source/acm_neteq.h
new file mode 100644
index 0000000..677c622
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_neteq.h
@@ -0,0 +1,383 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
+
+#include "audio_coding_module.h"
+#include "audio_coding_module_typedefs.h"
+#include "engine_configurations.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+#include "webrtc_neteq.h"
+#include "webrtc_vad.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RWLockWrapper;
+struct CodecInst;
+enum AudioPlayoutMode;
+enum ACMSpeechType;
+
+#define MAX_NUM_SLAVE_NETEQ 1
+
+class ACMNetEQ
+{
+public:
+ // Constructor of the class
+ ACMNetEQ();
+
+ // Destructor of the class.
+ ~ACMNetEQ();
+
+ //
+ // GetVersion()
+ // Fills the version array with the NetEQ version and updates the
+ // remainingBufferInBytes and position variables accordingly.
+ //
+ // Output:
+ // - version : An array to be filled with the version
+ // data.
+ //
+ // Input/Output:
+ // - remainingBuffInBytes : The number of free bytes at the end of
+ // the version array.
+ // - position : Position where the free space starts.
+ //
+ // Return value : 0 if ok.
+ // -1 if NetEQ returned an error.
+ //
+ static WebRtc_Word32 GetVersion(
+ char* version,
+ WebRtc_UWord32& remainingBuffInBytes,
+ WebRtc_UWord32& position);
+
+ //
+ // Init()
+ // Allocates memory for NetEQ and VAD and initializes them.
+ //
+ // Return value : 0 if ok.
+ // -1 if NetEQ or VAD returned an error or
+ // if out of memory.
+ //
+ WebRtc_Word32 Init();
+
+ //
+ // RecIn()
+ // Gives the payload to NetEQ.
+ //
+ // Input:
+ // - incomingPayload : Incoming audio payload.
+ // - payloadLength : Length of incoming audio payload.
+ // - rtpInfo : RTP header for the incoming payload containing
+ // information about payload type, sequence number,
+ // timestamp, ssrc and marker bit.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 RecIn(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo);
+
+ //
+ // RecOut()
+ // Asks NetEQ for 10 ms of decoded audio.
+ //
+ // Input:
+ // -audioFrame : an audio frame were output data and
+ // associated parameters are written to.
+ //
+ // Return value : 0 if ok.
+ // -1 if NetEQ returned an error.
+ //
+ WebRtc_Word32 RecOut(
+ AudioFrame& audioFrame);
+
+ //
+ // AddCodec()
+ // Adds a new codec to the NetEQ codec database.
+ //
+ // Input:
+ // - codecDef : The codec to be added.
+ // - toMaster : true if the codec has to be added to Master
+ // NetEq, otherwise will be added to the Slave
+ // NetEQ.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 AddCodec(
+ WebRtcNetEQ_CodecDef *codecDef,
+ bool toMaster = true);
+
+ //
+ // AllocatePacketBuffer()
+ // Allocates the NetEQ packet buffer.
+ //
+ // Input:
+ // - usedCodecs : An array of the codecs to be used by NetEQ.
+ // - noOfCodecs : Number of codecs in usedCodecs.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 AllocatePacketBuffer(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs);
+
+ //
+ // SetExtraDelay()
+ // Sets an delayInMS milliseconds extra delay in NetEQ.
+ //
+ // Input:
+ // - delayInMS : Extra delay in milliseconds.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 SetExtraDelay(
+ const WebRtc_Word32 delayInMS);
+
+ //
+ // SetAVTPlayout()
+ // Enable/disable playout of AVT payloads.
+ //
+ // Input:
+ // - enable : Enable if true, disable if false.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 SetAVTPlayout(
+ const bool enable);
+
+ //
+ // AVTPlayout()
+ // Get the current AVT playout state.
+ //
+ // Return value : True if AVT playout is enabled.
+ // False if AVT playout is disabled.
+ //
+ bool AVTPlayout() const;
+
+ //
+ // CurrentSampFreqHz()
+ // Get the current sampling frequency in Hz.
+ //
+ // Return value : Sampling frequency in Hz.
+ //
+ WebRtc_Word32 CurrentSampFreqHz() const;
+
+ //
+ // SetPlayoutMode()
+ // Sets the playout mode to voice or fax.
+ //
+ // Input:
+ // - mode : The playout mode to be used, voice,
+ // fax, or streaming.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 SetPlayoutMode(
+ const AudioPlayoutMode mode);
+
+ //
+ // PlayoutMode()
+ // Get the current playout mode.
+ //
+ // Return value : The current playout mode.
+ //
+ AudioPlayoutMode PlayoutMode() const;
+
+ //
+ // NetworkStatistics()
+ // Get the current network statistics from NetEQ.
+ //
+ // Output:
+ // - statistics : The current network statistics.
+ //
+ // Return value : 0 if ok.
+ // <0 if NetEQ returned an error.
+ //
+ WebRtc_Word32 NetworkStatistics(
+ ACMNetworkStatistics* statistics) const;
+
+ //
+ // VADMode()
+ // Get the current VAD Mode.
+ //
+ // Return value : The current VAD mode.
+ //
+ ACMVADMode VADMode() const;
+
+ //
+ // SetVADMode()
+ // Set the VAD mode.
+ //
+ // Input:
+ // - mode : The new VAD mode.
+ //
+ // Return value : 0 if ok.
+ // -1 if an error occurred.
+ //
+ WebRtc_Word16 SetVADMode(
+ const ACMVADMode mode);
+
+ //
+ // DecodeLock()
+ // Get the decode lock used to protect decoder instances while decoding.
+ //
+ // Return value : Pointer to the decode lock.
+ //
+ RWLockWrapper* DecodeLock() const
+ {
+ return _decodeLock;
+ }
+
+ //
+ // FlushBuffers()
+ // Flushes the NetEQ packet and speech buffers.
+ //
+ // Return value : 0 if ok.
+ // -1 if NetEQ returned an error.
+ //
+ WebRtc_Word32 FlushBuffers();
+
+ //
+ // RemoveCodec()
+ // Removes a codec from the NetEQ codec database.
+ //
+ // Input:
+ // - codecIdx : Codec to be removed.
+ //
+ // Return value : 0 if ok.
+ // -1 if an error occurred.
+ //
+ WebRtc_Word16 RemoveCodec(
+ WebRtcNetEQDecoder codecIdx,
+ bool isStereo = false);
+
+
+ //
+ // SetBackgroundNoiseMode()
+ // Set the mode of the background noise.
+ //
+ // Input:
+ // - mode : an enumerator specifying the mode of the
+ // background noise.
+ //
+ // Return value : 0 if succeeded,
+ // -1 if failed to set the mode.
+ //
+ WebRtc_Word16 SetBackgroundNoiseMode(
+ const ACMBackgroundNoiseMode mode);
+
+ //
+ // BackgroundNoiseMode()
+ // return the mode of the background noise.
+ //
+ // Return value : The mode of background noise.
+ //
+ WebRtc_Word16 BackgroundNoiseMode(
+ ACMBackgroundNoiseMode& mode);
+
+ void SetUniqueId(
+ WebRtc_Word32 id);
+
+ WebRtc_Word32 PlayoutTimestamp(
+ WebRtc_UWord32& timestamp);
+
+ void SetReceivedStereo(
+ bool receivedStereo);
+
+ WebRtc_UWord8 NumSlaves();
+
+ enum JB {masterJB = 0, slaveJB = 1};
+
+ WebRtc_Word16 AddSlave(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs);
+
+private:
+ //
+ // RTPPack()
+ // Creates a Word16 RTP packet out of the payload data in Word16 and
+ // a WebRtcRTPHeader.
+ //
+ // Input:
+ // - payload : Payload to be packetized.
+ // - payloadLengthW8 : Length of the payload in bytes.
+ // - rtpInfo : RTP header struct.
+ //
+ // Output:
+ // - rtpPacket : The RTP packet.
+ //
+ static void RTPPack(
+ WebRtc_Word16* rtpPacket,
+ const WebRtc_Word8* payload,
+ const WebRtc_Word32 payloadLengthW8,
+ const WebRtcRTPHeader& rtpInfo);
+
+ void LogError(
+ const char* neteqFuncName,
+ const WebRtc_Word16 idx) const;
+
+ WebRtc_Word16 InitByIdxSafe(
+ const WebRtc_Word16 idx);
+
+ // EnableVAD()
+ // Enable VAD.
+ //
+ // Return value : 0 if ok.
+ // -1 if an error occurred.
+ //
+ WebRtc_Word16 EnableVAD();
+
+ WebRtc_Word16 EnableVADByIdxSafe(
+ const WebRtc_Word16 idx);
+
+ WebRtc_Word16 AllocatePacketBufferByIdxSafe(
+ const WebRtcNetEQDecoder* usedCodecs,
+ WebRtc_Word16 noOfCodecs,
+ const WebRtc_Word16 idx);
+
+ void* _inst[MAX_NUM_SLAVE_NETEQ + 1];
+ void* _instMem[MAX_NUM_SLAVE_NETEQ + 1];
+
+ WebRtc_Word16* _netEqPacketBuffer[MAX_NUM_SLAVE_NETEQ + 1];
+
+ WebRtc_Word32 _id;
+ float _currentSampFreqKHz;
+ bool _avtPlayout;
+ AudioPlayoutMode _playoutMode;
+ CriticalSectionWrapper* _netEqCritSect;
+
+ WebRtcVadInst* _ptrVADInst[MAX_NUM_SLAVE_NETEQ + 1];
+
+ bool _vadStatus;
+ ACMVADMode _vadMode;
+ RWLockWrapper* _decodeLock;
+ bool _isInitialized[MAX_NUM_SLAVE_NETEQ + 1];
+ WebRtc_UWord8 _numSlaves;
+ bool _receivedStereo;
+ void* _masterSlaveInfo;
+ AudioFrame::VADActivity _previousAudioActivity;
+ WebRtc_Word32 _extraDelay;
+
+ CriticalSectionWrapper* _callbackCritSect;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc b/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc
new file mode 100644
index 0000000..7feb2e0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc
@@ -0,0 +1,147 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains unit tests for ACM's NetEQ wrapper (class ACMNetEQ).
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/main/source/acm_codec_database.h"
+#include "modules/audio_coding/main/source/acm_neteq.h"
+#include "modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
+#include "modules/interface/module_common_types.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+class AcmNetEqTest : public ::testing::Test {
+ protected:
+ static const size_t kMaxPayloadLen = 5760; // 60 ms, 48 kHz, 16 bit samples.
+ static const int kPcm16WbPayloadType = 94;
+ AcmNetEqTest() {}
+ virtual void SetUp();
+ virtual void TearDown() {}
+
+ void InsertZeroPacket(uint16_t sequence_number,
+ uint32_t timestamp,
+ uint8_t payload_type,
+ uint32_t ssrc,
+ bool marker_bit,
+ size_t len_payload_bytes);
+ void PullData(int expected_num_samples);
+
+ ACMNetEQ neteq_;
+};
+
+void AcmNetEqTest::SetUp() {
+ ASSERT_EQ(0, neteq_.Init());
+ ASSERT_EQ(0, neteq_.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(),
+ ACMCodecDB::kNumCodecs));
+ WebRtcNetEQ_CodecDef codec_def;
+ SET_CODEC_PAR(codec_def, kDecoderPCM16Bwb, kPcm16WbPayloadType, NULL, 16000);
+ SET_PCM16B_WB_FUNCTIONS(codec_def);
+ ASSERT_EQ(0, neteq_.AddCodec(&codec_def, true));
+}
+
+void AcmNetEqTest::InsertZeroPacket(uint16_t sequence_number,
+ uint32_t timestamp,
+ uint8_t payload_type,
+ uint32_t ssrc,
+ bool marker_bit,
+ size_t len_payload_bytes) {
+ ASSERT_TRUE(len_payload_bytes <= kMaxPayloadLen);
+ uint16_t payload[kMaxPayloadLen] = {0};
+ WebRtcRTPHeader rtp_header;
+ rtp_header.header.sequenceNumber = sequence_number;
+ rtp_header.header.timestamp = timestamp;
+ rtp_header.header.ssrc = ssrc;
+ rtp_header.header.payloadType = payload_type;
+ rtp_header.header.markerBit = marker_bit;
+ rtp_header.type.Audio.channel = 1;
+ ASSERT_EQ(0, neteq_.RecIn(reinterpret_cast<WebRtc_Word8*>(payload),
+ len_payload_bytes, rtp_header));
+}
+
+void AcmNetEqTest::PullData(int expected_num_samples) {
+ AudioFrame out_frame;
+ ASSERT_EQ(0, neteq_.RecOut(out_frame));
+ ASSERT_EQ(expected_num_samples, out_frame._payloadDataLengthInSamples);
+}
+
+TEST_F(AcmNetEqTest, NetworkStatistics) {
+ // Use fax mode to avoid time-scaling. This is to simplify the testing of
+ // packet waiting times in the packet buffer.
+ neteq_.SetPlayoutMode(fax);
+ // Insert 31 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
+ int num_frames = 30;
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ int i, j;
+ for (i = 0; i < num_frames; ++i) {
+ InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
+ kPayloadBytes);
+ }
+ // Pull out data once.
+ PullData(kSamples);
+ // Insert one more packet (to produce different mean and median).
+ i = num_frames;
+ InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
+ kPayloadBytes);
+ // Pull out all data.
+ for (j = 1; j < num_frames + 1; ++j) {
+ PullData(kSamples);
+ }
+
+ ACMNetworkStatistics stats;
+ ASSERT_EQ(0, neteq_.NetworkStatistics(&stats));
+ EXPECT_EQ(0, stats.currentBufferSize);
+ EXPECT_EQ(0, stats.preferredBufferSize);
+ EXPECT_FALSE(stats.jitterPeaksFound);
+ EXPECT_EQ(0, stats.currentPacketLossRate);
+ EXPECT_EQ(0, stats.currentDiscardRate);
+ EXPECT_EQ(0, stats.currentExpandRate);
+ EXPECT_EQ(0, stats.currentPreemptiveRate);
+ EXPECT_EQ(0, stats.currentAccelerateRate);
+ EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off.
+ EXPECT_EQ(300, stats.maxWaitingTimeMs);
+ EXPECT_EQ(10, stats.minWaitingTimeMs);
+ EXPECT_EQ(159, stats.meanWaitingTimeMs);
+ EXPECT_EQ(160, stats.medianWaitingTimeMs);
+}
+
+TEST_F(AcmNetEqTest, TestZeroLengthWaitingTimesVector) {
+ // Insert one packet.
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ int i = 0;
+ InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
+ kPayloadBytes);
+ // Do not pull out any data.
+
+ ACMNetworkStatistics stats;
+ ASSERT_EQ(0, neteq_.NetworkStatistics(&stats));
+ EXPECT_EQ(0, stats.currentBufferSize);
+ EXPECT_EQ(0, stats.preferredBufferSize);
+ EXPECT_FALSE(stats.jitterPeaksFound);
+ EXPECT_EQ(0, stats.currentPacketLossRate);
+ EXPECT_EQ(0, stats.currentDiscardRate);
+ EXPECT_EQ(0, stats.currentExpandRate);
+ EXPECT_EQ(0, stats.currentPreemptiveRate);
+ EXPECT_EQ(0, stats.currentAccelerateRate);
+ EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off.
+ EXPECT_EQ(-1, stats.minWaitingTimeMs);
+ EXPECT_EQ(-1, stats.maxWaitingTimeMs);
+ EXPECT_EQ(-1, stats.meanWaitingTimeMs);
+ EXPECT_EQ(-1, stats.medianWaitingTimeMs);
+}
+
+} // namespace
diff --git a/trunk/src/modules/audio_coding/main/source/acm_opus.cc b/trunk/src/modules/audio_coding/main/source/acm_opus.cc
new file mode 100644
index 0000000..ebdf971
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_opus.cc
@@ -0,0 +1,477 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_opus.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_OPUS
+ // NOTE! Opus is not included in the open-source package. Modify this file or your codec
+ // API to match the function call and name of used Opus API file.
+ // #include "opus_interface.h"
+#endif
+
+namespace webrtc
+{
+
+#ifndef WEBRTC_CODEC_OPUS
+
+ACMOPUS::ACMOPUS(WebRtc_Word16 /* codecID */)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _mySampFreq(0),
+ _myRate(0),
+ _opusMode(0),
+ _flagVBR(0) {
+ return;
+}
+
+
+ACMOPUS::~ACMOPUS()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMOPUS::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMOPUS::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMOPUS::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+void
+ACMOPUS::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMOPUS::DestructDecoderSafe()
+{
+ return;
+}
+
+
+void
+ACMOPUS::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMOPUS::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMOPUS::SetBitRateSafe(
+ const WebRtc_Word32 /*rate*/ )
+{
+ return -1;
+}
+
+#else //===================== Actual Implementation =======================
+
+// Remove when integrating a real Opus wrapper
+extern WebRtc_Word16 WebRtcOpus_CreateEnc(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
+extern WebRtc_Word16 WebRtcOpus_CreateDec(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
+extern WebRtc_Word16 WebRtcOpus_FreeEnc(OPUS_inst_t_* inst);
+extern WebRtc_Word16 WebRtcOpus_FreeDec(OPUS_inst_t_* inst);
+extern WebRtc_Word16 WebRtcOpus_Encode(OPUS_inst_t_* encInst,
+ WebRtc_Word16* input,
+ WebRtc_Word16* output,
+ WebRtc_Word16 len,
+ WebRtc_Word16 byteLen);
+extern WebRtc_Word16 WebRtcOpus_EncoderInit(OPUS_inst_t_* encInst,
+ WebRtc_Word16 samplFreq,
+ WebRtc_Word16 mode,
+ WebRtc_Word16 vbrFlag);
+extern WebRtc_Word16 WebRtcOpus_Decode(OPUS_inst_t_* decInst);
+extern WebRtc_Word16 WebRtcOpus_DecodeBwe(OPUS_inst_t_* decInst, WebRtc_Word16* input);
+extern WebRtc_Word16 WebRtcOpus_DecodePlc(OPUS_inst_t_* decInst);
+extern WebRtc_Word16 WebRtcOpus_DecoderInit(OPUS_inst_t_* decInst);
+
+ACMOPUS::ACMOPUS(WebRtc_Word16 codecID)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _mySampFreq(48000), // Default sampling frequency.
+ _myRate(50000), // Default rate.
+ _opusMode(1), // Default mode is the hybrid mode.
+ _flagVBR(0) { // Default VBR off.
+ _codecID = codecID;
+
+ // Current implementation doesn't have DTX. That might change.
+ _hasInternalDTX = false;
+
+ return;
+}
+
+ACMOPUS::~ACMOPUS()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcOpus_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcOpus_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ WebRtc_Word16 noEncodedSamples = 0;
+ WebRtc_Word16 tmpLenByte = 0;
+ *bitStreamLenByte = 0;
+
+ WebRtc_Word16 byteLengthFrame = 0;
+
+ // Derive what byte-length is requested
+ byteLengthFrame = _myRate*_frameLenSmpl/(8*_mySampFreq);
+
+ // Call Encoder
+ *bitStreamLenByte = WebRtcOpus_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
+ (WebRtc_Word16*)bitStream, _frameLenSmpl, byteLengthFrame);
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl;
+
+ // sanity check
+ if(*bitStreamLenByte < 0)
+ {
+ // error has happened
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Encode error for Opus");
+ *bitStreamLenByte = 0;
+ return -1;
+ }
+
+ return *bitStreamLenByte;
+}
+
+
+
+WebRtc_Word16
+ACMOPUS::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ //set the bit rate and initialize
+ _myRate = codecParams->codecInstant.rate;
+ return SetBitRateSafe( (WebRtc_UWord32)_myRate);
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ if (WebRtcOpus_DecoderInit(_decoderInstPtr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalInitDecoder: init decoder failed for Opus");
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMOPUS::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: Decoder uninitialized for Opus");
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderOpus, codecInst.pltype,
+ _decoderInstPtr, 16000);
+ SET_OPUS_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMOPUS::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalCreateEncoder()
+{
+ if (WebRtcOpus_CreateEnc(&_encoderInstPtr, _mySampFreq) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateEncoder: create encoder failed for Opus");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMOPUS::DestructEncoderSafe()
+{
+ _encoderExist = false;
+ _encoderInitialized = false;
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcOpus_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+}
+
+
+WebRtc_Word16
+ACMOPUS::InternalCreateDecoder()
+{
+ if (WebRtcOpus_CreateDec(&_decoderInstPtr, _mySampFreq) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalCreateDecoder: create decoder failed for Opus");
+ return -1;
+ }
+ return 0;
+}
+
+
+void
+ACMOPUS::DestructDecoderSafe()
+{
+ _decoderExist = false;
+ _decoderInitialized = false;
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcOpus_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+}
+
+
+void
+ACMOPUS::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WebRtcOpus_FreeEnc((OPUS_inst_t*)ptrInst);
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMOPUS::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec: given payload-type does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+ return netEq->RemoveCodec(kDecoderOpus);
+}
+
+WebRtc_Word16
+ACMOPUS::SetBitRateSafe(
+ const WebRtc_Word32 rate)
+{
+ //allowed rates: {8000, 12000, 14000, 16000, 18000, 20000,
+ // 22000, 24000, 26000, 28000, 30000, 32000};
+ switch(rate)
+ {
+ case 8000:
+ {
+ _myRate = 8000;
+ break;
+ }
+ case 12000:
+ {
+ _myRate = 12000;
+ break;
+ }
+ case 14000:
+ {
+ _myRate = 14000;
+ break;
+ }
+ case 16000:
+ {
+ _myRate = 16000;
+ break;
+ }
+ case 18000:
+ {
+ _myRate = 18000;
+ break;
+ }
+ case 20000:
+ {
+ _myRate = 20000;
+ break;
+ }
+ case 22000:
+ {
+ _myRate = 22000;
+ break;
+ }
+ case 24000:
+ {
+ _myRate = 24000;
+ break;
+ }
+ case 26000:
+ {
+ _myRate = 26000;
+ break;
+ }
+ case 28000:
+ {
+ _myRate = 28000;
+ break;
+ }
+ case 30000:
+ {
+ _myRate = 30000;
+ break;
+ }
+ case 32000:
+ {
+ _myRate = 32000;
+ break;
+ }
+ default:
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: Invalid rate Opus");
+ return -1;
+ }
+ }
+
+ // Re-init with new rate
+ if (WebRtcOpus_EncoderInit(_encoderInstPtr, _mySampFreq, _opusMode, _flagVBR) >= 0)
+ {
+ _encoderParams.codecInstant.rate = _myRate;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_opus.h b/trunk/src/modules/audio_coding/main/source/acm_opus.h
new file mode 100644
index 0000000..278c369
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_opus.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct OPUS_inst_t_;
+struct OPUS_inst_t_;
+
+namespace webrtc
+{
+
+class ACMOPUS: public ACMGenericCodec
+{
+public:
+ ACMOPUS(WebRtc_Word16 codecID);
+ ~ACMOPUS();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 rate);
+
+ OPUS_inst_t_* _encoderInstPtr;
+ OPUS_inst_t_* _decoderInstPtr;
+
+ WebRtc_UWord16 _mySampFreq;
+ WebRtc_UWord16 _myRate;
+ WebRtc_Word16 _opusMode;
+ WebRtc_Word16 _flagVBR;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc
new file mode 100644
index 0000000..5c93e90
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc
@@ -0,0 +1,334 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_pcm16b.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_PCM16
+ #include "pcm16b.h"
+#endif
+
+namespace webrtc
+{
+
+#ifndef WEBRTC_CODEC_PCM16
+
+ACMPCM16B::ACMPCM16B(
+ WebRtc_Word16 /* codecID */)
+{
+ return;
+}
+
+
+ACMPCM16B::~ACMPCM16B()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+
+WebRtc_Word32
+ACMPCM16B::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+
+ACMGenericCodec*
+ACMPCM16B::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalCreateEncoder()
+{
+ return -1;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalCreateDecoder()
+{
+ return -1;
+}
+
+
+void
+ACMPCM16B::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+
+void
+ACMPCM16B::DestructEncoderSafe()
+{
+ return;
+}
+
+void
+ACMPCM16B::DestructDecoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+
+
+#else //===================== Actual Implementation =======================
+
+
+ACMPCM16B::ACMPCM16B(
+ WebRtc_Word16 codecID)
+{
+ _codecID = codecID;
+ _samplingFreqHz = ACMCodecDB::CodecFreq(_codecID);
+}
+
+
+ACMPCM16B::~ACMPCM16B()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ *bitStreamLenByte = WebRtcPcm16b_Encode(&_inAudio[_inAudioIxRead],
+ _frameLenSmpl*_noChannels,
+ bitStream);
+ // increment the read index to tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl*_noChannels;
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMPCM16B::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ switch(_samplingFreqHz)
+ {
+ case 8000:
+ {
+ SET_CODEC_PAR((codecDef), kDecoderPCM16B, codecInst.pltype,
+ NULL, 8000);
+ SET_PCM16B_FUNCTIONS((codecDef));
+ break;
+ }
+ case 16000:
+ {
+ SET_CODEC_PAR((codecDef), kDecoderPCM16Bwb, codecInst.pltype,
+ NULL, 16000);
+ SET_PCM16B_WB_FUNCTIONS((codecDef));
+ break;
+ }
+ case 32000:
+ {
+ SET_CODEC_PAR((codecDef), kDecoderPCM16Bswb32kHz,
+ codecInst.pltype, NULL, 32000);
+ SET_PCM16B_SWB32_FUNCTIONS((codecDef));
+ break;
+ }
+ default:
+ {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMPCM16B::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalCreateEncoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::InternalCreateDecoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+void
+ACMPCM16B::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ // PCM has no instance
+ return;
+}
+
+
+void
+ACMPCM16B::DestructEncoderSafe()
+{
+ // PCM has no instance
+ _encoderExist = false;
+ _encoderInitialized = false;
+ return;
+}
+
+void
+ACMPCM16B::DestructDecoderSafe()
+{
+ // PCM has no instance
+ _decoderExist = false;
+ _decoderInitialized = false;
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCM16B::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+
+ switch(_samplingFreqHz)
+ {
+ case 8000:
+ {
+ return netEq->RemoveCodec(kDecoderPCM16B);
+ }
+ case 16000:
+ {
+ return netEq->RemoveCodec(kDecoderPCM16Bwb);
+ }
+ case 32000:
+ {
+ return netEq->RemoveCodec(kDecoderPCM16Bswb32kHz);
+ }
+ default:
+ {
+ return -1;
+ }
+ }
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h
new file mode 100644
index 0000000..a81d8fe
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc
+{
+
+class ACMPCM16B : public ACMGenericCodec
+{
+public:
+ ACMPCM16B(WebRtc_Word16 codecID);
+ ~ACMPCM16B();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word32 _samplingFreqHz;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcma.cc b/trunk/src/modules/audio_coding/main/source/acm_pcma.cc
new file mode 100644
index 0000000..c86bd1c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcma.cc
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_pcma.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+// Codec interface
+#include "g711_interface.h"
+
+namespace webrtc
+{
+
+ACMPCMA::ACMPCMA(WebRtc_Word16 codecID)
+{
+ _codecID = codecID;
+}
+
+
+ACMPCMA::~ACMPCMA()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCMA::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ *bitStreamLenByte = WebRtcG711_EncodeA(NULL, &_inAudio[_inAudioIxRead],
+ _frameLenSmpl*_noChannels, (WebRtc_Word16*)bitStream);
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl*_noChannels;
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMPCMA::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMA::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMA::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word32 ACMPCMA::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_PCMA_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderPCMa, codecInst.pltype, NULL, 8000);
+ SET_PCMA_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMPCMA::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMPCMA::InternalCreateEncoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMA::InternalCreateDecoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+void
+ACMPCMA::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ // PCM has no instance
+ return;
+}
+
+
+void
+ACMPCMA::DestructEncoderSafe()
+{
+ // PCM has no instance
+ return;
+}
+
+
+void
+ACMPCMA::DestructDecoderSafe()
+{
+ // PCM has no instance
+ _decoderInitialized = false;
+ _decoderExist = false;
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCMA::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+
+ return netEq->RemoveCodec(kDecoderPCMa);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcma.h b/trunk/src/modules/audio_coding/main/source/acm_pcma.h
new file mode 100644
index 0000000..db25798
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcma.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc
+{
+
+class ACMPCMA : public ACMGenericCodec
+{
+public:
+ ACMPCMA(WebRtc_Word16 codecID);
+ ~ACMPCMA();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc b/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc
new file mode 100644
index 0000000..320ba5b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc
@@ -0,0 +1,167 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_pcmu.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+// Codec interface
+#include "g711_interface.h"
+
+namespace webrtc
+{
+
+ACMPCMU::ACMPCMU(WebRtc_Word16 codecID)
+{
+ _codecID = codecID;
+}
+
+
+ACMPCMU::~ACMPCMU()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCMU::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ *bitStreamLenByte = WebRtcG711_EncodeU(NULL, &_inAudio[_inAudioIxRead],
+ _frameLenSmpl*_noChannels, (WebRtc_Word16*)bitStream);
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _frameLenSmpl*_noChannels;
+ return *bitStreamLenByte;
+}
+
+
+WebRtc_Word16
+ACMPCMU::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMU::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMU::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMPCMU::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderPCMu, codecInst.pltype, NULL, 8000);
+ SET_PCMU_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMPCMU::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMPCMU::InternalCreateEncoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMPCMU::InternalCreateDecoder()
+{
+ // PCM has no instance
+ return 0;
+}
+
+
+void
+ACMPCMU::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ // PCM has no instance
+ return;
+}
+
+
+void
+ACMPCMU::DestructEncoderSafe()
+{
+ // PCM has no instance
+ _encoderExist = false;
+ _encoderInitialized = false;
+ return;
+}
+
+void ACMPCMU::DestructDecoderSafe()
+{
+ // PCM has no instance
+ _decoderInitialized = false;
+ _decoderExist = false;
+ return;
+}
+
+
+WebRtc_Word16
+ACMPCMU::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+
+ return netEq->RemoveCodec(kDecoderPCMu);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcmu.h b/trunk/src/modules/audio_coding/main/source/acm_pcmu.h
new file mode 100644
index 0000000..2fc4223
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_pcmu.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc
+{
+
+class ACMPCMU : public ACMGenericCodec
+{
+public:
+ ACMPCMU(WebRtc_Word16 codecID);
+ ~ACMPCMU();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_red.cc b/trunk/src/modules/audio_coding/main/source/acm_red.cc
new file mode 100644
index 0000000..88dbc24
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_red.cc
@@ -0,0 +1,163 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_red.h"
+#include "acm_neteq.h"
+#include "acm_common_defs.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+namespace webrtc
+{
+
+ACMRED::ACMRED(WebRtc_Word16 codecID)
+{
+ _codecID = codecID;
+}
+
+
+ACMRED::~ACMRED()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMRED::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ // RED is never used as an encoder
+ // RED has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMRED::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMRED::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // RED has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMRED::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ // This codec does not need initialization,
+ // RED has no instance
+ return 0;
+}
+
+
+WebRtc_Word32
+ACMRED::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ // Todo:
+ // log error
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+ SET_CODEC_PAR((codecDef), kDecoderRED, codecInst.pltype, NULL, 8000);
+ SET_RED_FUNCTIONS((codecDef));
+ return 0;
+}
+
+
+ACMGenericCodec*
+ACMRED::CreateInstance(void)
+{
+ return NULL;
+}
+
+
+WebRtc_Word16
+ACMRED::InternalCreateEncoder()
+{
+ // RED has no instance
+ return 0;
+}
+
+
+WebRtc_Word16
+ACMRED::InternalCreateDecoder()
+{
+ // RED has no instance
+ return 0;
+}
+
+
+void
+ACMRED::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ // RED has no instance
+ return;
+}
+
+
+void
+ACMRED::DestructEncoderSafe()
+{
+ // RED has no instance
+ return;
+}
+
+void ACMRED::DestructDecoderSafe()
+{
+ // RED has no instance
+ return;
+}
+
+
+WebRtc_Word16
+ACMRED::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+
+ return netEq->RemoveCodec(kDecoderRED);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_red.h b/trunk/src/modules/audio_coding/main/source/acm_red.h
new file mode 100644
index 0000000..d22cb8c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_red.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
+
+#include "acm_generic_codec.h"
+
+namespace webrtc
+{
+
+class ACMRED : public ACMGenericCodec
+{
+public:
+ ACMRED(WebRtc_Word16 codecID);
+ ~ACMRED();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_resampler.cc b/trunk/src/modules/audio_coding/main/source/acm_resampler.cc
new file mode 100644
index 0000000..1e36e73
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_resampler.cc
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "acm_resampler.h"
+#include "critical_section_wrapper.h"
+#include "resampler.h"
+#include "signal_processing_library.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+ACMResampler::ACMResampler():
+
+_resamplerCritSect(*CriticalSectionWrapper::CreateCriticalSection())
+{
+}
+
+ACMResampler::~ACMResampler()
+{
+
+ delete &_resamplerCritSect;
+}
+
+
+WebRtc_Word16
+ACMResampler::Resample10Msec(
+ const WebRtc_Word16* inAudio,
+ WebRtc_Word32 inFreqHz,
+ WebRtc_Word16* outAudio,
+ WebRtc_Word32 outFreqHz,
+ WebRtc_UWord8 numAudioChannels)
+{
+
+ CriticalSectionScoped cs(_resamplerCritSect);
+
+ if(inFreqHz == outFreqHz)
+ {
+ size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
+ memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
+ return static_cast<WebRtc_Word16>(inFreqHz / 100);
+ }
+
+ int maxLen = 480 * numAudioChannels; //max number of samples for 10ms at 48kHz
+ int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
+ int outLen;
+
+ WebRtc_Word32 ret;
+ ResamplerType type;
+ type = (numAudioChannels == 1)? kResamplerSynchronous:kResamplerSynchronousStereo;
+
+ ret = _resampler.ResetIfNeeded(inFreqHz,outFreqHz,type);
+ if (ret < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+ "Error in reset of resampler");
+ return -1;
+ }
+
+ ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
+ if (ret < 0 )
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+ "Error in resampler: resampler.Push");
+ return -1;
+ }
+
+ WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
+
+ return outAudioLenSmpl;
+
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_resampler.h b/trunk/src/modules/audio_coding/main/source/acm_resampler.h
new file mode 100644
index 0000000..246a4fe
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_resampler.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
+
+#include "resampler.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+
+class ACMResampler
+{
+public:
+ ACMResampler();
+ ~ACMResampler();
+
+ WebRtc_Word16 Resample10Msec(
+ const WebRtc_Word16* inAudio,
+ const WebRtc_Word32 inFreqHz,
+ WebRtc_Word16* outAudio,
+ const WebRtc_Word32 outFreqHz,
+ WebRtc_UWord8 numAudioChannels);
+
+private:
+
+ //Use the Resampler class
+ Resampler _resampler;
+ CriticalSectionWrapper& _resamplerCritSect;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_speex.cc b/trunk/src/modules/audio_coding/main/source/acm_speex.cc
new file mode 100644
index 0000000..acb2a3b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_speex.cc
@@ -0,0 +1,669 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_speex.h"
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+
+#ifdef WEBRTC_CODEC_SPEEX
+ // NOTE! Speex is not included in the open-source package. The following
+ // interface file is needed:
+ //
+ // /modules/audio_coding/codecs/speex/main/interface/speex_interface.h
+ //
+ // The API in the header file should match the one below.
+ //
+ // int16_t WebRtcSpeex_CreateEnc(SPEEX_encinst_t **SPEEXenc_inst,
+ // int32_t fs);
+ // int16_t WebRtcSpeex_FreeEnc(SPEEX_encinst_t *SPEEXenc_inst);
+ // int16_t WebRtcSpeex_CreateDec(SPEEX_decinst_t **SPEEXdec_inst,
+ // int32_t fs,
+ // int16_t enh_enabled);
+ // int16_t WebRtcSpeex_FreeDec(SPEEX_decinst_t *SPEEXdec_inst);
+ // int16_t WebRtcSpeex_Encode(SPEEX_encinst_t *SPEEXenc_inst,
+ // int16_t *speechIn,
+ // int32_t rate);
+ // int16_t WebRtcSpeex_EncoderInit(SPEEX_encinst_t *SPEEXenc_inst,
+ // int16_t vbr, int16_t complexity,
+ // int16_t vad_enable);
+ // int16_t WebRtcSpeex_GetBitstream(SPEEX_encinst_t *SPEEXenc_inst,
+ // int16_t *encoded);
+ // int16_t WebRtcSpeex_DecodePlc(SPEEX_decinst_t *SPEEXdec_inst,
+ // int16_t *decoded, int16_t noOfLostFrames);
+ // int16_t WebRtcSpeex_Decode(SPEEX_decinst_t *SPEEXdec_inst,
+ // int16_t *encoded, int16_t len,
+ // int16_t *decoded, int16_t *speechType);
+ // int16_t WebRtcSpeex_DecoderInit(SPEEX_decinst_t *SPEEXdec_inst);
+ // void WebRtcSpeex_Version(char *versionStr, short len);
+ #include "speex_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_SPEEX
+ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID*/)
+ : _encoderInstPtr(NULL),
+ _decoderInstPtr(NULL),
+ _complMode(0),
+ _vbrEnabled(false),
+ _encodingRate(-1),
+ _samplingFrequency(-1),
+ _samplesIn20MsAudio(-1) {
+ return;
+}
+
+ACMSPEEX::~ACMSPEEX()
+{
+ return;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalEncode(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16* /* bitStreamLenByte */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::EnableDTX()
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::DisableDTX()
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalInitEncoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ return -1;
+}
+
+WebRtc_Word32
+ACMSPEEX::CodecDef(
+ WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */)
+{
+ return -1;
+}
+
+ACMGenericCodec*
+ACMSPEEX::CreateInstance(void)
+{
+ return NULL;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalCreateEncoder()
+{
+ return -1;
+}
+
+void
+ACMSPEEX::DestructEncoderSafe()
+{
+ return;
+}
+
+
+WebRtc_Word16
+ACMSPEEX::InternalCreateDecoder()
+{
+ return -1;
+}
+
+void
+ACMSPEEX::DestructDecoderSafe()
+{
+ return;
+}
+
+WebRtc_Word16
+ACMSPEEX::SetBitRateSafe(
+ const WebRtc_Word32 /* rate */)
+{
+ return -1;
+}
+
+void
+ACMSPEEX::InternalDestructEncoderInst(
+ void* /* ptrInst */)
+{
+ return;
+}
+
+WebRtc_Word16
+ACMSPEEX::UnregisterFromNetEqSafe(
+ ACMNetEQ* /* netEq */,
+ WebRtc_Word16 /* payloadType */)
+{
+ return -1;
+}
+
+#ifdef UNUSEDSPEEX
+WebRtc_Word16
+ACMSPEEX::EnableVBR()
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::DisableVBR()
+{
+ return -1;
+}
+
+WebRtc_Word16
+ACMSPEEX::SetComplMode(
+ WebRtc_Word16 mode)
+{
+ return -1;
+}
+#endif
+
+#else //===================== Actual Implementation =======================
+
+ACMSPEEX::ACMSPEEX(WebRtc_Word16 codecID):
+_encoderInstPtr(NULL),
+_decoderInstPtr(NULL)
+{
+ _codecID = codecID;
+
+ // Set sampling frequency, frame size and rate Speex
+ if(_codecID == ACMCodecDB::kSPEEX8)
+ {
+ _samplingFrequency = 8000;
+ _samplesIn20MsAudio = 160;
+ _encodingRate = 11000;
+ }
+ else if(_codecID == ACMCodecDB::kSPEEX16)
+ {
+ _samplingFrequency = 16000;
+ _samplesIn20MsAudio = 320;
+ _encodingRate = 22000;
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Wrong codec id for Speex.");
+
+ _samplingFrequency = -1;
+ _samplesIn20MsAudio = -1;
+ _encodingRate = -1;
+ }
+
+ _hasInternalDTX = true;
+ _dtxEnabled = false;
+ _vbrEnabled = false;
+ _complMode = 3; // default complexity value
+
+ return;
+}
+
+ACMSPEEX::~ACMSPEEX()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcSpeex_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcSpeex_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalEncode(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16* bitStreamLenByte)
+{
+ WebRtc_Word16 status;
+ WebRtc_Word16 numEncodedSamples = 0;
+ WebRtc_Word16 n = 0;
+
+ while( numEncodedSamples < _frameLenSmpl)
+ {
+ status = WebRtcSpeex_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
+ _encodingRate);
+
+ // increment the read index this tell the caller that how far
+ // we have gone forward in reading the audio buffer
+ _inAudioIxRead += _samplesIn20MsAudio;
+ numEncodedSamples += _samplesIn20MsAudio;
+
+ if(status < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error in Speex encoder");
+ return status;
+ }
+
+ // Update VAD, if internal DTX is used
+ if(_hasInternalDTX && _dtxEnabled)
+ {
+ _vadLabel[n++] = status;
+ _vadLabel[n++] = status;
+ }
+
+ if(status == 0)
+ {
+ // This frame is detected as inactive. We need send whatever
+ // encoded so far.
+ *bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
+ (WebRtc_Word16*)bitStream);
+
+ return *bitStreamLenByte;
+ }
+ }
+
+ *bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
+ (WebRtc_Word16*)bitStream);
+ return *bitStreamLenByte;
+}
+
+WebRtc_Word16
+ACMSPEEX::DecodeSafe(
+ WebRtc_UWord8* /* bitStream */,
+ WebRtc_Word16 /* bitStreamLenByte */,
+ WebRtc_Word16* /* audio */,
+ WebRtc_Word16* /* audioSamples */,
+ WebRtc_Word8* /* speechType */)
+{
+ return 0;
+}
+
+WebRtc_Word16
+ACMSPEEX::EnableDTX()
+{
+ if(_dtxEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist) // check if encoder exist
+ {
+ // enable DTX
+ if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 1) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot enable DTX for Speex");
+ return -1;
+ }
+ _dtxEnabled = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word16
+ACMSPEEX::DisableDTX()
+{
+ if(!_dtxEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist) // check if encoder exist
+ {
+ // disable DTX
+ if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 0) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot disable DTX for Speex");
+ return -1;
+ }
+ _dtxEnabled = false;
+ return 0;
+ }
+ else
+ {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+
+ return 0;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalInitEncoder(
+ WebRtcACMCodecParams* codecParams)
+{
+ // sanity check
+ if (_encoderInstPtr == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot initialize Speex encoder, instance does not exist");
+ return -1;
+ }
+
+ WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
+ status += (WebRtcSpeex_EncoderInit(_encoderInstPtr, _vbrEnabled, _complMode, ((codecParams->enableDTX)? 1:0)) < 0)? -1:0;
+
+ if (status >= 0) {
+ return 0;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error in initialization of Speex encoder");
+ return -1;
+ }
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalInitDecoder(
+ WebRtcACMCodecParams* /* codecParams */)
+{
+ WebRtc_Word16 status;
+
+ // sanity check
+ if (_decoderInstPtr == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot initialize Speex decoder, instance does not exist");
+ return -1;
+ }
+ status = ((WebRtcSpeex_DecoderInit(_decoderInstPtr) < 0)? -1:0);
+
+ if (status >= 0) {
+ return 0;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error in initialization of Speex decoder");
+ return -1;
+ }
+}
+
+WebRtc_Word32
+ACMSPEEX::CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst)
+{
+ if (!_decoderInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error, Speex decoder is not initialized");
+ return -1;
+ }
+
+ // Fill up the structure by calling
+ // "SET_CODEC_PAR" & "SET_SPEEX_FUNCTION."
+ // Then call NetEQ to add the codec to it's
+ // database.
+
+ switch(_samplingFrequency)
+ {
+ case 8000:
+ {
+ SET_CODEC_PAR((codecDef), kDecoderSPEEX_8, codecInst.pltype,
+ _decoderInstPtr, 8000);
+ break;
+ }
+ case 16000:
+ {
+ SET_CODEC_PAR((codecDef), kDecoderSPEEX_16, codecInst.pltype,
+ _decoderInstPtr, 16000);
+ break;
+ }
+ default:
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Unsupported sampling frequency for Speex");
+
+ return -1;
+ }
+ }
+
+ SET_SPEEX_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec*
+ACMSPEEX::CreateInstance(void)
+{
+ return NULL;
+}
+
+WebRtc_Word16
+ACMSPEEX::InternalCreateEncoder()
+{
+ return WebRtcSpeex_CreateEnc(&_encoderInstPtr, _samplingFrequency);
+}
+
+void
+ACMSPEEX::DestructEncoderSafe()
+{
+ if(_encoderInstPtr != NULL)
+ {
+ WebRtcSpeex_FreeEnc(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ // there is no encoder set the following
+ _encoderExist = false;
+ _encoderInitialized = false;
+ _encodingRate = 0;
+}
+
+
+WebRtc_Word16
+ACMSPEEX::InternalCreateDecoder()
+{
+ return WebRtcSpeex_CreateDec(&_decoderInstPtr, _samplingFrequency, 1);
+}
+
+void
+ACMSPEEX::DestructDecoderSafe()
+{
+ if(_decoderInstPtr != NULL)
+ {
+ WebRtcSpeex_FreeDec(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ // there is no encoder instance set the followings
+ _decoderExist = false;
+ _decoderInitialized = false;
+}
+
+WebRtc_Word16
+ACMSPEEX::SetBitRateSafe(
+ const WebRtc_Word32 rate)
+{
+ // Check if changed rate
+ if (rate == _encodingRate) {
+ return 0;
+ } else if (rate > 2000) {
+ _encodingRate = rate;
+ _encoderParams.codecInstant.rate = rate;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Unsupported encoding rate for Speex");
+
+ return -1;
+ }
+
+ return 0;
+}
+
+
+void
+ACMSPEEX::InternalDestructEncoderInst(
+ void* ptrInst)
+{
+ if(ptrInst != NULL)
+ {
+ WebRtcSpeex_FreeEnc((SPEEX_encinst_t_*)ptrInst);
+ }
+ return;
+}
+
+
+WebRtc_Word16
+ACMSPEEX::UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType)
+{
+ if(payloadType != _decoderParams.codecInstant.pltype)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot unregister codec %s given payload-type %d does not match \
+the stored payload type",
+ _decoderParams.codecInstant.plname,
+ payloadType,
+ _decoderParams.codecInstant.pltype);
+ return -1;
+ }
+
+
+ switch(_samplingFrequency)
+ {
+ case 8000:
+ {
+ return netEq->RemoveCodec(kDecoderSPEEX_8);
+ }
+ case 16000:
+ {
+ return netEq->RemoveCodec(kDecoderSPEEX_16);
+ }
+ default:
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Could not unregister Speex from NetEQ. Sampling frequency doesn't match");
+ return -1;
+ }
+ }
+}
+
+
+#ifdef UNUSEDSPEEX
+
+// This API is currently not in use. If requested to be able to enable/disable VBR
+// an ACM API need to be added.
+WebRtc_Word16
+ACMSPEEX::EnableVBR()
+{
+ if(_vbrEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist) // check if encoder exist
+ {
+ // enable Variable Bit Rate (VBR)
+ if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 1, _complMode, (_dtxEnabled? 1:0)) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot enable VBR mode for Speex");
+
+ return -1;
+ }
+ _vbrEnabled = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+
+// This API is currently not in use. If requested to be able to enable/disable VBR
+// an ACM API need to be added.
+WebRtc_Word16
+ACMSPEEX::DisableVBR()
+{
+ if(!_vbrEnabled)
+ {
+ return 0;
+ }
+ else if(_encoderExist) // check if encoder exist
+ {
+ // disable DTX
+ if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, _complMode, (_dtxEnabled? 1:0)) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Cannot disable DTX for Speex");
+
+ return -1;
+ }
+ _vbrEnabled = false;
+ return 0;
+ }
+ else
+ {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+// This API is currently not in use. If requested to be able to set complexity
+// an ACM API need to be added.
+WebRtc_Word16
+ACMSPEEX::SetComplMode(
+ WebRtc_Word16 mode)
+{
+ // Check if new mode
+ if(mode == _complMode)
+ {
+ return 0;
+ }
+ else if(_encoderExist) // check if encoder exist
+ {
+ // Set new mode
+ if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, mode, (_dtxEnabled? 1:0)) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Error in complexity mode for Speex");
+ return -1;
+ }
+ _complMode = mode;
+ return 0;
+ }
+ else
+ {
+ // encoder doesn't exists, therefore disabling is harmless
+ return 0;
+ }
+}
+
+#endif
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_speex.h b/trunk/src/modules/audio_coding/main/source/acm_speex.h
new file mode 100644
index 0000000..7988e45
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/acm_speex.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct SPEEX_encinst_t_;
+struct SPEEX_decinst_t_;
+
+namespace webrtc {
+
+class ACMSPEEX : public ACMGenericCodec
+{
+public:
+ ACMSPEEX(WebRtc_Word16 codecID);
+ ~ACMSPEEX();
+ // for FEC
+ ACMGenericCodec* CreateInstance(void);
+
+ WebRtc_Word16 InternalEncode(
+ WebRtc_UWord8* bitstream,
+ WebRtc_Word16* bitStreamLenByte);
+
+ WebRtc_Word16 InternalInitEncoder(
+ WebRtcACMCodecParams *codecParams);
+
+ WebRtc_Word16 InternalInitDecoder(
+ WebRtcACMCodecParams *codecParams);
+
+protected:
+ WebRtc_Word16 DecodeSafe(
+ WebRtc_UWord8* bitStream,
+ WebRtc_Word16 bitStreamLenByte,
+ WebRtc_Word16* audio,
+ WebRtc_Word16* audioSamples,
+ WebRtc_Word8* speechType);
+
+ WebRtc_Word32 CodecDef(
+ WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst);
+
+ void DestructEncoderSafe();
+
+ void DestructDecoderSafe();
+
+ WebRtc_Word16 InternalCreateEncoder();
+
+ WebRtc_Word16 InternalCreateDecoder();
+
+ void InternalDestructEncoderInst(
+ void* ptrInst);
+
+ WebRtc_Word16 SetBitRateSafe(
+ const WebRtc_Word32 rate);
+
+ WebRtc_Word16 EnableDTX();
+
+ WebRtc_Word16 DisableDTX();
+
+#ifdef UNUSEDSPEEX
+ WebRtc_Word16 EnableVBR();
+
+ WebRtc_Word16 DisableVBR();
+
+ WebRtc_Word16 SetComplMode(
+ WebRtc_Word16 mode);
+#endif
+
+ WebRtc_Word16 UnregisterFromNetEqSafe(
+ ACMNetEQ* netEq,
+ WebRtc_Word16 payloadType);
+
+ SPEEX_encinst_t_* _encoderInstPtr;
+ SPEEX_decinst_t_* _decoderInstPtr;
+ WebRtc_Word16 _complMode;
+ bool _vbrEnabled;
+ WebRtc_Word32 _encodingRate;
+ WebRtc_Word16 _samplingFrequency;
+ WebRtc_UWord16 _samplesIn20MsAudio;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc b/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc
new file mode 100644
index 0000000..2cd959d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc
@@ -0,0 +1,136 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include "acm_dtmf_detection.h"
+#include "audio_coding_module.h"
+#include "audio_coding_module_impl.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+// Create module
+AudioCodingModule*
+AudioCodingModule::Create(
+ const WebRtc_Word32 id)
+{
+ return new AudioCodingModuleImpl(id);
+}
+
+// Destroy module
+void
+AudioCodingModule::Destroy(
+ AudioCodingModule* module)
+{
+ delete static_cast<AudioCodingModuleImpl*> (module);
+}
+
+// Get number of supported codecs
+WebRtc_UWord8 AudioCodingModule::NumberOfCodecs()
+{
+ return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs);
+}
+
+// Get supported codec param with id
+WebRtc_Word32
+AudioCodingModule::Codec(
+ const WebRtc_UWord8 listId,
+ CodecInst& codec)
+{
+ // Get the codec settings for the codec with the given list ID
+ return ACMCodecDB::Codec(listId, &codec);
+}
+
+// Get supported codec Param with name
+WebRtc_Word32
+AudioCodingModule::Codec(
+ const char* payloadName,
+ CodecInst& codec,
+ const WebRtc_Word32 samplingFreqHz)
+{
+ // Search through codec list for a matching name
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
+ {
+ // Store codec settings for codec number "codeCntr" in the output struct
+ ACMCodecDB::Codec(codecCntr, &codec);
+
+ if(!STR_CASE_CMP(codec.plname, payloadName))
+ {
+ // If samplingFreqHz is set (!= -1), check if frequency matches
+ if((samplingFreqHz == codec.plfreq) || (samplingFreqHz == -1))
+ {
+ // We found a match, return OK
+ return 0;
+ }
+ }
+ }
+
+ // if we are here we couldn't find anything
+ // set the params to unacceptable values
+ codec.plname[0] = '\0';
+ codec.pltype = -1;
+ codec.pacsize = 0;
+ codec.rate = 0;
+ codec.plfreq = 0;
+ return -1;
+}
+
+// Get supported codec Index with name, and frequency if needed
+WebRtc_Word32
+AudioCodingModule::Codec(
+ const char* payloadName,
+ const WebRtc_Word32 samplingFreqHz)
+{
+ CodecInst codec;
+
+ // Search through codec list for a matching name
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
+ {
+ // Temporally store codec settings for codec number "codeCntr" in "codec"
+ ACMCodecDB::Codec(codecCntr, &codec);
+
+ if(!STR_CASE_CMP(codec.plname, payloadName))
+ {
+ // If samplingFreqHz is set (!= -1), check if frequency matches
+ if((samplingFreqHz == codec.plfreq) || (samplingFreqHz == -1))
+ {
+ // We found a match, return codec list number (index)
+ return codecCntr;
+ }
+ }
+ }
+
+ // We did not find a matching codec in the list
+ return -1;
+}
+
+// Checks the validity of the parameters of the given codec
+bool
+AudioCodingModule::IsCodecValid(
+ const CodecInst& codec)
+{
+ int mirrorID;
+ char errMsg[500];
+
+ int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500);
+
+ if(codecNumber < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, errMsg);
+ return false;
+ }
+ else
+ {
+ return true;
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi b/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi
new file mode 100644
index 0000000..80114e0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -0,0 +1,152 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audio_coding_module',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'CNG',
+ 'G711',
+ 'G722',
+ 'iLBC',
+ 'iSAC',
+ 'iSACFix',
+ 'PCM16B',
+ 'NetEq',
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/common_audio/common_audio.gyp:vad',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/audio_coding_module.h',
+ '../interface/audio_coding_module_typedefs.h',
+ 'acm_amr.cc',
+ 'acm_amr.h',
+ 'acm_amrwb.cc',
+ 'acm_amrwb.h',
+ 'acm_celt.cc',
+ 'acm_celt.h',
+ 'acm_cng.cc',
+ 'acm_cng.h',
+ 'acm_codec_database.cc',
+ 'acm_codec_database.h',
+ 'acm_dtmf_detection.cc',
+ 'acm_dtmf_detection.h',
+ 'acm_dtmf_playout.cc',
+ 'acm_dtmf_playout.h',
+ 'acm_g722.cc',
+ 'acm_g722.h',
+ 'acm_g7221.cc',
+ 'acm_g7221.h',
+ 'acm_g7221c.cc',
+ 'acm_g7221c.h',
+ 'acm_g729.cc',
+ 'acm_g729.h',
+ 'acm_g7291.cc',
+ 'acm_g7291.h',
+ 'acm_generic_codec.cc',
+ 'acm_generic_codec.h',
+ 'acm_gsmfr.cc',
+ 'acm_gsmfr.h',
+ 'acm_ilbc.cc',
+ 'acm_ilbc.h',
+ 'acm_isac.cc',
+ 'acm_isac.h',
+ 'acm_isac_macros.h',
+ 'acm_neteq.cc',
+ 'acm_neteq.h',
+ 'acm_opus.cc',
+ 'acm_opus.h',
+ 'acm_speex.cc',
+ 'acm_speex.h',
+ 'acm_pcm16b.cc',
+ 'acm_pcm16b.h',
+ 'acm_pcma.cc',
+ 'acm_pcma.h',
+ 'acm_pcmu.cc',
+ 'acm_pcmu.h',
+ 'acm_red.cc',
+ 'acm_red.h',
+ 'acm_resampler.cc',
+ 'acm_resampler.h',
+ 'audio_coding_module.cc',
+ 'audio_coding_module_impl.cc',
+ 'audio_coding_module_impl.h',
+ ],
+ },
+ ],
+ # Exclude the test targets when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'audio_coding_module_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_coding_module',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'sources': [
+ '../test/ACMTest.cc',
+ '../test/APITest.cc',
+ '../test/Channel.cc',
+ '../test/EncodeDecodeTest.cc',
+ '../test/iSACTest.cc',
+ '../test/PCMFile.cc',
+ '../test/RTPFile.cc',
+ '../test/SpatialAudio.cc',
+ '../test/TestAllCodecs.cc',
+ '../test/Tester.cc',
+ '../test/TestFEC.cc',
+ '../test/TestStereo.cc',
+ '../test/TestVADDTX.cc',
+ '../test/TimedTrace.cc',
+ '../test/TwoWayCommunication.cc',
+ '../test/utility.cc',
+ ],
+ },
+ {
+ 'target_name': 'audio_coding_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_coding_module',
+ 'NetEq',
+ '<(webrtc_root)/common_audio/common_audio.gyp:vad',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'sources': [
+ 'acm_neteq_unittest.cc',
+ ],
+ }, # audio_coding_unittests
+ ],
+ }],
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
new file mode 100644
index 0000000..18b6f4f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -0,0 +1,2624 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_codec_database.h"
+#include "acm_common_defs.h"
+#include "acm_dtmf_detection.h"
+#include "acm_generic_codec.h"
+#include "acm_resampler.h"
+#include "audio_coding_module_impl.h"
+#include "critical_section_wrapper.h"
+#include "engine_configurations.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+
+#include <assert.h>
+#include <stdlib.h>
+
+#ifdef ACM_QA_TEST
+# include <stdio.h>
+#endif
+
+#ifdef TIMED_LOGGING
+ char message[500];
+ #include "../test/timedtrace.h"
+ #include <string.h>
+ #define LOGWITHTIME(logString) \
+ sprintf(message, logString, _id); \
+ _trace.TimedLogg(message);
+#else
+ #define LOGWITHTIME(logString)
+#endif
+
+namespace webrtc
+{
+
+enum {
+ kACMToneEnd = 999
+};
+
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+ const WebRtc_Word32 id):
+ _packetizationCallback(NULL),
+ _id(id),
+ _lastTimestamp(0),
+ _lastInTimestamp(0),
+ _vadEnabled(false),
+ _dtxEnabled(false),
+ _vadMode(VADNormal),
+ _stereoSend(false),
+ _prev_received_channel(0),
+ _expected_channels(1),
+ _currentSendCodecIdx(-1), // invalid value
+ _sendCodecRegistered(false),
+ _acmCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _vadCallback(NULL),
+ _lastRecvAudioCodecPlType(255),
+ _isFirstRED(true),
+ _fecEnabled(false),
+ _fragmentation(NULL),
+ _lastFECTimestamp(0),
+ _redPayloadType(255),
+ _receiveREDPayloadType(255), // invalid value
+ _previousPayloadType(255),
+ _dummyRTPHeader(NULL),
+ _recvPlFrameSizeSmpls(0),
+ _receiverInitialized(false),
+ _dtmfDetector(NULL),
+ _dtmfCallback(NULL),
+ _lastDetectedTone(kACMToneEnd),
+ _callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
+{
+ _lastTimestamp = 0xD87F3F9F;
+ _lastInTimestamp = 0xD87F3F9F;
+
+ // Nullify send codec memory, set payload type and set codec name to
+ // invalid values.
+ memset(&_sendCodecInst, 0, sizeof(CodecInst));
+ strncpy(_sendCodecInst.plname, "noCodecRegistered", 31);
+ _sendCodecInst.pltype = -1;
+
+ // Nullify memory for CNG, DTMF and RED.
+ memset(&_cngNB, 0, sizeof(CodecInst));
+ memset(&_cngWB, 0, sizeof(CodecInst));
+ memset(&_cngSWB, 0, sizeof(CodecInst));
+ memset(&_RED, 0, sizeof(CodecInst));
+ memset(&_DTMF, 0, sizeof(CodecInst));
+
+ for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
+ {
+ _codecs[i] = NULL;
+ _registeredPlTypes[i] = -1;
+ _stereoReceive[i] = false;
+ _slaveCodecs[i] = NULL;
+ _mirrorCodecIdx[i] = -1;
+ }
+
+ _netEq.SetUniqueId(_id);
+
+ // Allocate memory for RED
+ _redBuffer = new WebRtc_UWord8[MAX_PAYLOAD_SIZE_BYTE];
+ _fragmentation = new RTPFragmentationHeader;
+ _fragmentation->fragmentationVectorSize = 2;
+ _fragmentation->fragmentationOffset = new WebRtc_UWord32[2];
+ _fragmentation->fragmentationLength = new WebRtc_UWord32[2];
+ _fragmentation->fragmentationTimeDiff = new WebRtc_UWord16[2];
+ _fragmentation->fragmentationPlType = new WebRtc_UWord8[2];
+
+ // Register the default payload type for RED and for
+ // CNG for the three frequencies 8, 16 and 32 kHz
+ for (int i = (ACMCodecDB::kNumCodecs - 1); i>=0; i--)
+ {
+ if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "red") == 0))
+ {
+ _redPayloadType = ACMCodecDB::database_[i].pltype;
+ }
+ else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0))
+ {
+ if (ACMCodecDB::database_[i].plfreq == 8000)
+ {
+ memcpy(&_cngNB, &ACMCodecDB::database_[i], sizeof(_cngNB));
+ }
+ else if (ACMCodecDB::database_[i].plfreq == 16000)
+ {
+ memcpy(&_cngWB, &ACMCodecDB::database_[i], sizeof(_cngWB));
+ } else if (ACMCodecDB::database_[i].plfreq == 32000)
+ {
+ memcpy(&_cngSWB, &ACMCodecDB::database_[i], sizeof(_cngSWB));
+ }
+ }
+ }
+
+ if(InitializeReceiverSafe() < 0 )
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot initialize reciever");
+ }
+#ifdef TIMED_LOGGING
+ _trace.SetUp("TimedLogg.txt");
+#endif
+
+#ifdef ACM_QA_TEST
+ char fileName[500];
+ sprintf(fileName, "ACM_QA_incomingPL_%03d_%d%d%d%d%d%d.dat",
+ _id,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10);
+
+ _incomingPL = fopen(fileName, "wb");
+
+ sprintf(fileName, "ACM_QA_outgoingPL_%03d_%d%d%d%d%d%d.dat",
+ _id,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10);
+ _outgoingPL = fopen(fileName, "wb");
+#endif
+
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created");
+}
+
+AudioCodingModuleImpl::~AudioCodingModuleImpl()
+{
+ {
+ CriticalSectionScoped lock(*_acmCritSect);
+ _currentSendCodecIdx = -1;
+
+ for (int i=0; i < ACMCodecDB::kMaxNumCodecs; i++)
+ {
+ if (_codecs[i] != NULL)
+ {
+ assert(_mirrorCodecIdx[i] > -1);
+ if(_codecs[_mirrorCodecIdx[i]] != NULL)
+ {
+ delete _codecs[_mirrorCodecIdx[i]];
+ _codecs[_mirrorCodecIdx[i]] = NULL;
+ }
+ _codecs[i] = NULL;
+ }
+
+ if(_slaveCodecs[i] != NULL)
+ {
+ assert(_mirrorCodecIdx[i] > -1);
+ if(_slaveCodecs[_mirrorCodecIdx[i]] != NULL)
+ {
+ delete _slaveCodecs[_mirrorCodecIdx[i]];
+ _slaveCodecs[_mirrorCodecIdx[i]] = NULL;
+ }
+ _slaveCodecs[i] = NULL;
+ }
+ }
+
+ if(_dtmfDetector != NULL)
+ {
+ delete _dtmfDetector;
+ _dtmfDetector = NULL;
+ }
+ if(_dummyRTPHeader != NULL)
+ {
+ delete _dummyRTPHeader;
+ _dummyRTPHeader = NULL;
+ }
+ if(_redBuffer != NULL)
+ {
+ delete [] _redBuffer;
+ _redBuffer = NULL;
+ }
+ if(_fragmentation != NULL)
+ {
+ // Only need to delete fragmentation header, it will clean
+ // up it's own memory
+ delete _fragmentation;
+ _fragmentation = NULL;
+ }
+ }
+
+#ifdef ACM_QA_TEST
+ if(_incomingPL != NULL)
+ {
+ fclose(_incomingPL);
+ }
+
+ if(_outgoingPL != NULL)
+ {
+ fclose(_outgoingPL);
+ }
+#endif
+
+ delete _callbackCritSect;
+ _callbackCritSect = NULL;
+
+ delete _acmCritSect;
+ _acmCritSect = NULL;
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, _id, "Destroyed");
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::ChangeUniqueId(
+ const WebRtc_Word32 id)
+{
+ {
+ CriticalSectionScoped lock(*_acmCritSect);
+ _id = id;
+#ifdef ACM_QA_TEST
+ if(_incomingPL != NULL)
+ {
+ fclose(_incomingPL);
+ }
+
+ if(_outgoingPL != NULL)
+ {
+ fclose(_outgoingPL);
+ }
+
+ char fileName[500];
+ sprintf(fileName, "ACM_QA_incomingPL_%03d_%d%d%d%d%d%d.dat",
+ _id,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10);
+
+ _incomingPL = fopen(fileName, "wb");
+
+ sprintf(fileName, "ACM_QA_outgoingPL_%03d_%d%d%d%d%d%d.dat",
+ _id,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10,
+ rand() % 10);
+ _outgoingPL = fopen(fileName, "wb");
+#endif
+
+ for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
+ {
+ if(_codecs[i] != NULL)
+ {
+ _codecs[i]->SetUniqueID(id);
+ }
+ }
+ }
+
+ _netEq.SetUniqueId(_id);
+ return 0;
+}
+
+// returns the number of milliseconds until the module want a
+// worker thread to call Process
+WebRtc_Word32
+AudioCodingModuleImpl::TimeUntilNextProcess()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("TimeUntilNextProcess"))
+ {
+ return -1;
+ }
+ return _codecs[_currentSendCodecIdx]->SamplesLeftToEncode() /
+ (_sendCodecInst.plfreq / 1000);
+}
+
+// Process any pending tasks such as timeouts
+WebRtc_Word32
+AudioCodingModuleImpl::Process()
+{
+ WebRtc_UWord8 bitStream[2 * MAX_PAYLOAD_SIZE_BYTE]; // Make room for 1 RED payload
+ WebRtc_Word16 lengthBytes = 2 * MAX_PAYLOAD_SIZE_BYTE;
+ WebRtc_Word16 redLengthBytes = lengthBytes;
+ WebRtc_UWord32 rtpTimestamp;
+ WebRtc_Word16 status;
+ WebRtcACMEncodingType encodingType;
+ FrameType frameType = kAudioFrameSpeech;
+ WebRtc_UWord8 currentPayloadType = 0;
+ bool hasDataToSend = false;
+ bool fecActive = false;
+
+ // keep the scope of the ACM critical section limited
+ {
+ CriticalSectionScoped lock(*_acmCritSect);
+ if(!HaveValidEncoder("Process"))
+ {
+ return -1;
+ }
+
+ status = _codecs[_currentSendCodecIdx]->Encode(bitStream, &lengthBytes,
+ &rtpTimestamp, &encodingType);
+ if (status < 0) // Encode failed
+ {
+ // logging error
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Process(): Encoding Failed");
+ lengthBytes = 0;
+ return -1;
+ }
+ else if(status == 0)
+ {
+ // Not enough data
+ return 0;
+ }
+ else
+ {
+ switch(encodingType)
+ {
+ case kNoEncoding:
+ {
+ currentPayloadType = _previousPayloadType;
+ frameType = kFrameEmpty;
+ lengthBytes = 0;
+ break;
+ }
+ case kActiveNormalEncoded:
+ case kPassiveNormalEncoded:
+ {
+ currentPayloadType = (WebRtc_UWord8)_sendCodecInst.pltype;
+ frameType = kAudioFrameSpeech;
+ break;
+ }
+ case kPassiveDTXNB:
+ {
+ currentPayloadType = (WebRtc_UWord8)_cngNB.pltype;
+ frameType = kAudioFrameCN;
+ _isFirstRED = true;
+ break;
+ }
+ case kPassiveDTXWB:
+ {
+ currentPayloadType = (WebRtc_UWord8)_cngWB.pltype;
+ frameType = kAudioFrameCN;
+ _isFirstRED = true;
+ break;
+ }
+ case kPassiveDTXSWB:
+ {
+ currentPayloadType = (WebRtc_UWord8)_cngSWB.pltype;
+ frameType = kAudioFrameCN;
+ _isFirstRED = true;
+ break;
+ }
+ }
+ hasDataToSend = true;
+ _previousPayloadType = currentPayloadType;
+
+ // Redundancy encode is done here,
+ // the two bitstreams packetized into
+ // one RTP packet and the fragmentation points
+ // are set.
+ // Only apply RED on speech data.
+ if((_fecEnabled) &&
+ ((encodingType == kActiveNormalEncoded) ||
+ (encodingType == kPassiveNormalEncoded)))
+ {
+ // FEC is enabled within this scope.
+ //
+ // Note that, a special solution exists for iSAC since it is the only codec for
+ // which getRedPayload has a non-empty implementation.
+ //
+ // Summary of the FEC scheme below (use iSAC as example):
+ //
+ // 1st (_firstRED is true) encoded iSAC frame (primary #1) =>
+ // - call getRedPayload() and store redundancy for packet #1 in second
+ // fragment of RED buffer (old data)
+ // - drop the primary iSAC frame
+ // - don't call SendData
+ // 2nd (_firstRED is false) encoded iSAC frame (primary #2) =>
+ // - store primary #2 in 1st fragment of RED buffer and send the combined
+ // packet
+ // - the transmitted packet contains primary #2 (new) and reduncancy for
+ // packet #1 (old)
+ // - call getRedPayload() and store redundancy for packet #2 in second
+ // fragment of RED buffer
+ //
+ // ...
+ //
+ // Nth encoded iSAC frame (primary #N) =>
+ // - store primary #N in 1st fragment of RED buffer and send the combined
+ // packet
+ // - the transmitted packet contains primary #N (new) and reduncancy for
+ // packet #(N-1) (old)
+ // - call getRedPayload() and store redundancy for packet #N in second
+ // fragment of RED buffer
+ //
+ // For all other codecs, getRedPayload does nothing and returns -1 =>
+ // redundant data is only a copy.
+ //
+ // First combined packet contains : #2 (new) and #1 (old)
+ // Second combined packet contains: #3 (new) and #2 (old)
+ // Third combined packet contains : #4 (new) and #3 (old)
+ //
+ // Hence, even if every second packet is dropped, perfect reconstruction is
+ // possible.
+ fecActive = true;
+
+ hasDataToSend = false;
+ if(!_isFirstRED) // skip this part for the first packet in a RED session
+ {
+ // Rearrange bitStream such that FEC packets are included.
+ // Replace bitStream now that we have stored current bitStream.
+ memcpy(bitStream + _fragmentation->fragmentationOffset[1], _redBuffer,
+ _fragmentation->fragmentationLength[1]);
+ // Update the fragmentation time difference vector
+ WebRtc_UWord16 timeSinceLastTimestamp =
+ WebRtc_UWord16(rtpTimestamp - _lastFECTimestamp);
+
+ // Update fragmentation vectors
+ _fragmentation->fragmentationPlType[1] =
+ _fragmentation->fragmentationPlType[0];
+ _fragmentation->fragmentationTimeDiff[1] = timeSinceLastTimestamp;
+ hasDataToSend = true;
+ }
+
+ // Insert new packet length.
+ _fragmentation->fragmentationLength[0] = lengthBytes;
+
+ // Insert new packet payload type.
+ _fragmentation->fragmentationPlType[0] = currentPayloadType;
+ _lastFECTimestamp = rtpTimestamp;
+
+ // can be modified by the GetRedPayload() call if iSAC is utilized
+ redLengthBytes = lengthBytes;
+ // A fragmentation header is provided => packetization according to RFC 2198
+ // (RTP Payload for Redundant Audio Data) will be used.
+ // First fragment is the current data (new).
+ // Second fragment is the previous data (old).
+ lengthBytes =
+ static_cast<WebRtc_Word16> (_fragmentation->fragmentationLength[0] +
+ _fragmentation->fragmentationLength[1]);
+
+ // Get, and store, redundant data from the encoder based on the recently
+ // encoded frame.
+ // NOTE - only iSAC contains an implementation; all other codecs does nothing
+ // and returns -1.
+ if (_codecs[_currentSendCodecIdx]->GetRedPayload(_redBuffer,
+ &redLengthBytes) == -1)
+ {
+ // The codec was not iSAC => use current encoder output as redundant data
+ // instead (trivial FEC scheme)
+ memcpy(_redBuffer, bitStream, redLengthBytes);
+ }
+
+ _isFirstRED = false;
+ // Update payload type with RED payload type
+ currentPayloadType = _redPayloadType;
+ }
+ }
+ }
+
+ if(hasDataToSend)
+ {
+ CriticalSectionScoped lock(*_callbackCritSect);
+#ifdef ACM_QA_TEST
+ if(_outgoingPL != NULL)
+ {
+ fwrite(&rtpTimestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL);
+ fwrite(¤tPayloadType, sizeof(WebRtc_UWord8), 1, _outgoingPL);
+ fwrite(&lengthBytes, sizeof(WebRtc_Word16), 1, _outgoingPL);
+ }
+#endif
+
+ if(_packetizationCallback != NULL)
+ {
+ if (fecActive) {
+ _packetizationCallback->SendData(frameType, currentPayloadType,
+ rtpTimestamp, bitStream, lengthBytes, _fragmentation);
+ } else {
+ _packetizationCallback->SendData(frameType, currentPayloadType,
+ rtpTimestamp, bitStream, lengthBytes, NULL);
+ }
+ }
+
+ // This is for test
+ if(_vadCallback != NULL)
+ {
+ _vadCallback->InFrameType(((WebRtc_Word16)encodingType));
+ }
+ }
+ if (fecActive) {
+ _fragmentation->fragmentationLength[1] = redLengthBytes;
+ }
+ return lengthBytes;
+}
+
+
+
+
+/////////////////////////////////////////
+// Sender
+//
+
+// Initialize send codec
+WebRtc_Word32
+AudioCodingModuleImpl::InitializeSender()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ _sendCodecRegistered = false;
+ _currentSendCodecIdx = -1; // invalid value
+
+ _sendCodecInst.plname[0] = '\0';
+
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
+ {
+ if(_codecs[codecCntr] != NULL)
+ {
+ _codecs[codecCntr]->DestructEncoder();
+ }
+ }
+ // Initialize FEC/RED
+ _isFirstRED = true;
+ if(_fecEnabled)
+ {
+ if(_redBuffer != NULL)
+ {
+ memset(_redBuffer, 0, MAX_PAYLOAD_SIZE_BYTE);
+ }
+ if(_fragmentation != NULL)
+ {
+ _fragmentation->fragmentationVectorSize = 2;
+ _fragmentation->fragmentationOffset[0] = 0;
+ _fragmentation->fragmentationOffset[0] = MAX_PAYLOAD_SIZE_BYTE;
+ memset(_fragmentation->fragmentationLength, 0, sizeof(WebRtc_UWord32) * 2);
+ memset(_fragmentation->fragmentationTimeDiff, 0, sizeof(WebRtc_UWord16) * 2);
+ memset(_fragmentation->fragmentationPlType, 0, sizeof(WebRtc_UWord8) * 2);
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::ResetEncoder()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ if(!HaveValidEncoder("ResetEncoder"))
+ {
+ return -1;
+ }
+ return _codecs[_currentSendCodecIdx]->ResetEncoder();
+}
+
+void
+AudioCodingModuleImpl::UnregisterSendCodec()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ _sendCodecRegistered = false;
+ _currentSendCodecIdx = -1; // invalid value
+
+ return;
+}
+
+ACMGenericCodec*
+AudioCodingModuleImpl::CreateCodec(
+ const CodecInst& codec)
+{
+ ACMGenericCodec* myCodec = NULL;
+
+ myCodec = ACMCodecDB::CreateCodecInstance(&codec);
+ if(myCodec == NULL)
+ {
+ // Error, could not create the codec
+
+ // logging error
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "ACMCodecDB::CreateCodecInstance() failed in \
+CreateCodec()");
+ return myCodec;
+ }
+ myCodec->SetUniqueID(_id);
+ myCodec->SetNetEqDecodeLock(_netEq.DecodeLock());
+
+ return myCodec;
+}
+
+// can be called multiple times for Codec, CNG, RED
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterSendCodec(
+ const CodecInst& sendCodec)
+{
+ if((sendCodec.channels != 1) && (sendCodec.channels != 2))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Registering Send codec failed due to wrong number of channels, %d. Only\
+mono codecs are supported, i.e. channels=1.", sendCodec.channels);
+ return -1;
+ }
+
+ char errMsg[500];
+ int mirrorId;
+ int codecID = ACMCodecDB::CodecNumber(&sendCodec, &mirrorId, errMsg,
+ sizeof(errMsg));
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ // Check for reported errors from function CodecNumber()
+ if(codecID < 0)
+ {
+ if(!_sendCodecRegistered)
+ {
+ // This values has to be NULL if there is no codec registered
+ _currentSendCodecIdx = -1; // invalid value
+ }
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, errMsg);
+ // Failed to register Send Codec
+ return -1;
+ }
+
+ // telephone-event cannot be a send codec
+ if(!STR_CASE_CMP(sendCodec.plname, "telephone-event"))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "telephone-event cannot be registered as send codec");
+ return -1;
+ }
+
+ // RED can be registered with other payload type. If not registered a default
+ // payload type is used.
+ if(!STR_CASE_CMP(sendCodec.plname, "red"))
+ {
+ // Check if the payload-type is valid
+ if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Invalid payload-type %d for %s.",
+ sendCodec.pltype, sendCodec.plname);
+ return -1;
+ }
+ // Set RED payload type
+ _redPayloadType = (WebRtc_UWord8)sendCodec.pltype;
+ return 0;
+ }
+
+ // CNG can be registered with other payload type. If not registered the
+ // default payload types will be used: CNNB=13 (fixed), CNWB=97, CNSWB=98
+ if(!STR_CASE_CMP(sendCodec.plname, "CN"))
+ {
+ // CNG is registered
+ switch(sendCodec.plfreq)
+ {
+ case 8000:
+ {
+ memcpy(&_cngNB, &sendCodec, sizeof(_cngNB));
+ break;
+ }
+ case 16000:
+ {
+ memcpy(&_cngWB, &sendCodec, sizeof(_cngWB));
+ break;
+ }
+ case 32000:
+ {
+ memcpy(&_cngSWB, &sendCodec, sizeof(_cngSWB));
+ break;
+ }
+ default :
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RegisterSendCodec() failed, invalid frequency for CNG registeration");
+ return -1;
+ }
+ }
+
+ return 0;
+ }
+
+ // Check if the payload-type is valid
+ if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Invalid payload-type %d for %s.",
+ sendCodec.pltype, sendCodec.plname);
+ return -1;
+ }
+
+ // Check if codec supports the number of channels
+ if(ACMCodecDB::codec_settings_[codecID].channel_support < sendCodec.channels)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "%d number of channels not supportedn for %s.",
+ sendCodec.channels, sendCodec.plname);
+ return -1;
+ }
+
+ // Set Stereo
+ if (sendCodec.channels == 2)
+ {
+ _stereoSend = true;
+ }
+
+ // check if the codec is already registered as send codec
+ bool oldCodecFamily;
+ if(_sendCodecRegistered)
+ {
+ int sendCodecMirrorID;
+ int sendCodecID =
+ ACMCodecDB::CodecNumber(&_sendCodecInst, &sendCodecMirrorID);
+ assert(sendCodecID >= 0);
+ oldCodecFamily = (sendCodecID == codecID) || (mirrorId == sendCodecMirrorID);
+ }
+ else
+ {
+ oldCodecFamily = false;
+ }
+
+ // If new codec, register
+ if (!oldCodecFamily)
+ {
+ if(_codecs[mirrorId] == NULL)
+ {
+
+ _codecs[mirrorId] = CreateCodec(sendCodec);
+ if(_codecs[mirrorId] == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Create the codec");
+ return -1;
+ }
+ _mirrorCodecIdx[mirrorId] = mirrorId;
+ }
+
+ if(mirrorId != codecID)
+ {
+ _codecs[codecID] = _codecs[mirrorId];
+ _mirrorCodecIdx[codecID] = mirrorId;
+ }
+
+ ACMGenericCodec* tmpCodecPtr = _codecs[codecID];
+ WebRtc_Word16 status;
+ WebRtcACMCodecParams codecParams;
+
+ memcpy(&(codecParams.codecInstant), &sendCodec,
+ sizeof(CodecInst));
+ codecParams.enableVAD = _vadEnabled;
+ codecParams.enableDTX = _dtxEnabled;
+ codecParams.vadMode = _vadMode;
+ // force initialization
+ status = tmpCodecPtr->InitEncoder(&codecParams, true);
+
+ // Check if VAD was turned on, or if error is reported
+ if (status == 1) {
+ _vadEnabled = true;
+ } else if (status < 0)
+ {
+ // could not initialize the encoder
+
+ // Check if already have a registered codec
+ // Depending on that different messages are logged
+ if(!_sendCodecRegistered)
+ {
+ _currentSendCodecIdx = -1; // invalid value
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Initialize the encoder No Encoder is registered");
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Initialize the encoder, continue encoding \
+with the previously registered codec");
+ }
+ return -1;
+ }
+
+ // Everything is fine so we can replace the previous codec
+ // with this one
+ if(_sendCodecRegistered)
+ {
+ // If we change codec we start fresh with FEC.
+ // This is not strictly required by the standard.
+ _isFirstRED = true;
+
+ if(tmpCodecPtr->SetVAD(_dtxEnabled, _vadEnabled, _vadMode) < 0){
+ // SetVAD failed
+ _vadEnabled = false;
+ _dtxEnabled = false;
+ }
+
+ }
+
+ _currentSendCodecIdx = codecID;
+ _sendCodecRegistered = true;
+ memcpy(&_sendCodecInst, &sendCodec, sizeof(CodecInst));
+ _previousPayloadType = _sendCodecInst.pltype;
+ return 0;
+ }
+ else
+ {
+ // If codec is the same as already registers check if any parameters
+ // has changed compared to the current values.
+ // If any parameter is valid then apply it and record.
+ bool forceInit = false;
+
+ if(mirrorId != codecID)
+ {
+ _codecs[codecID] = _codecs[mirrorId];
+ _mirrorCodecIdx[codecID] = mirrorId;
+ }
+
+ // check the payload-type
+ if(sendCodec.pltype != _sendCodecInst.pltype)
+ {
+ // At this point check if the given payload type is valid.
+ // Record it later when the sampling frequency is changed
+ // successfully.
+ if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Out of range payload type");
+ return -1;
+ }
+
+ }
+
+ // If there is a codec that ONE instance of codec supports multiple
+ // sampling frequencies, then we need to take care of it here.
+ // one such a codec is iSAC. Both WB and SWB are encoded and decoded
+ // with one iSAC instance. Therefore, we need to update the encoder
+ // frequency if required.
+ if(_sendCodecInst.plfreq != sendCodec.plfreq)
+ {
+ forceInit = true;
+
+ // if sampling frequency is changed we have to start fresh with RED.
+ _isFirstRED = true;
+ }
+
+ // If packet size or number of channels has changed, we need to
+ // re-initialize the encoder.
+ if(_sendCodecInst.pacsize != sendCodec.pacsize)
+ {
+ forceInit = true;
+ }
+ if(_sendCodecInst.channels != sendCodec.channels)
+ {
+ forceInit = true;
+ }
+
+ if(forceInit)
+ {
+ WebRtcACMCodecParams codecParams;
+
+ memcpy(&(codecParams.codecInstant), &sendCodec,
+ sizeof(CodecInst));
+ codecParams.enableVAD = _vadEnabled;
+ codecParams.enableDTX = _dtxEnabled;
+ codecParams.vadMode = _vadMode;
+
+ // force initialization
+ if(_codecs[_currentSendCodecIdx]->InitEncoder(&codecParams, true) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Could not change the codec packet-size.");
+ return -1;
+ }
+
+ _sendCodecInst.plfreq = sendCodec.plfreq;
+ _sendCodecInst.pacsize = sendCodec.pacsize;
+ _sendCodecInst.channels = sendCodec.channels;
+ }
+
+ // If the change of sampling frequency has been successful then
+ // we store the payload-type.
+ _sendCodecInst.pltype = sendCodec.pltype;
+
+ // check if a change in Rate is required
+ if(sendCodec.rate != _sendCodecInst.rate)
+ {
+ if(_codecs[codecID]->SetBitRate(sendCodec.rate) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Could not change the codec rate.");
+ return -1;
+ }
+ _sendCodecInst.rate = sendCodec.rate;
+ }
+ _previousPayloadType = _sendCodecInst.pltype;
+
+ return 0;
+ }
+}
+
+// get current send codec
+WebRtc_Word32
+AudioCodingModuleImpl::SendCodec(
+ CodecInst& currentSendCodec) const
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "SendCodec()");
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!_sendCodecRegistered)
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "SendCodec Failed, no codec is registered");
+
+ return -1;
+ }
+ WebRtcACMCodecParams encoderParam;
+ _codecs[_currentSendCodecIdx]->EncoderParams(&encoderParam);
+ encoderParam.codecInstant.pltype = _sendCodecInst.pltype;
+ memcpy(¤tSendCodec, &(encoderParam.codecInstant),
+ sizeof(CodecInst));
+
+ return 0;
+}
+
+// get current send freq
+WebRtc_Word32
+AudioCodingModuleImpl::SendFrequency() const
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "SendFrequency()");
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!_sendCodecRegistered)
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "SendFrequency Failed, no codec is registered");
+
+ return -1;
+ }
+
+ return _sendCodecInst.plfreq;
+}
+
+// Get encode bitrate
+// Adaptive rate codecs return their current encode target rate, while other codecs
+// return there longterm avarage or their fixed rate.
+WebRtc_Word32
+AudioCodingModuleImpl::SendBitrate() const
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!_sendCodecRegistered)
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "SendBitrate Failed, no codec is registered");
+
+ return -1;
+ }
+
+ WebRtcACMCodecParams encoderParam;
+ _codecs[_currentSendCodecIdx]->EncoderParams(&encoderParam);
+
+ return encoderParam.codecInstant.rate;
+}
+
+// set available bandwidth, inform the encoder about the estimated bandwidth
+// received from the remote party
+WebRtc_Word32
+AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
+ const WebRtc_Word32 bw )
+{
+ return _codecs[_currentSendCodecIdx]->SetEstimatedBandwidth(bw);
+}
+
+// register a transport callback wich will be called to deliver
+// the encoded buffers
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterTransportCallback(
+ AudioPacketizationCallback* transport)
+{
+ CriticalSectionScoped lock(*_callbackCritSect);
+ _packetizationCallback = transport;
+ return 0;
+}
+
+// Used by the module to deliver messages to the codec module/appliation
+// AVT(DTMF)
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
+#ifndef WEBRTC_DTMF_DETECTION
+ AudioCodingFeedback* /* incomingMessagesCallback */,
+ const ACMCountries /* cpt */)
+{
+ return -1;
+#else
+ AudioCodingFeedback* incomingMessagesCallback,
+ const ACMCountries cpt)
+{
+ WebRtc_Word16 status = 0;
+
+ // Enter the critical section for callback
+ {
+ CriticalSectionScoped lock(*_callbackCritSect);
+ _dtmfCallback = incomingMessagesCallback;
+ }
+ // enter the ACM critical section to set up the DTMF class.
+ {
+ CriticalSectionScoped lock(*_acmCritSect);
+ // Check if the call is to disable or enable the callback
+ if(incomingMessagesCallback == NULL)
+ {
+ // callback is disabled, delete DTMF-detector class
+ if(_dtmfDetector != NULL)
+ {
+ delete _dtmfDetector;
+ _dtmfDetector = NULL;
+ }
+ status = 0;
+ }
+ else
+ {
+ status = 0;
+ if(_dtmfDetector == NULL)
+ {
+ _dtmfDetector = new(ACMDTMFDetection);
+ if(_dtmfDetector == NULL)
+ {
+ status = -1;
+ }
+ }
+ if(status >= 0)
+ {
+ status = _dtmfDetector->Enable(cpt);
+ if(status < 0)
+ {
+ // failed to initialize if DTMF-detection was not enabled before,
+ // delete the class, and set the callback to NULL and return -1.
+ delete _dtmfDetector;
+ _dtmfDetector = NULL;
+ }
+ }
+ }
+ }
+ // check if we failed in setting up the DTMF-detector class
+ if((status < 0))
+ {
+ // we failed, we cannot have the callback
+ CriticalSectionScoped lock(*_callbackCritSect);
+ _dtmfCallback = NULL;
+ }
+
+ return status;
+#endif
+}
+
+
+// Add 10MS of raw (PCM) audio data to the encoder
+WebRtc_Word32
+AudioCodingModuleImpl::Add10MsData(
+ const AudioFrame& audioFrame)
+{
+ // Do we have a codec registered?
+ CriticalSectionScoped lock(*_acmCritSect);
+ if(!HaveValidEncoder("Add10MsData"))
+ {
+ return -1;
+ }
+
+ if(audioFrame._payloadDataLengthInSamples == 0)
+ {
+ assert(false);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Add 10 ms audio, payload length is zero");
+ return -1;
+ }
+ // Allow for 8, 16, 32 and 48kHz input audio
+ if((audioFrame._frequencyInHz != 8000) &&
+ (audioFrame._frequencyInHz != 16000) &&
+ (audioFrame._frequencyInHz != 32000) &&
+ (audioFrame._frequencyInHz != 48000))
+ {
+ assert(false);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Add 10 ms audio, input frequency not valid");
+ return -1;
+ }
+
+
+ // If the length and frequency matches. We currently just support raw PCM
+ if((audioFrame._frequencyInHz/ 100) !=
+ audioFrame._payloadDataLengthInSamples)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Add 10 ms audio, input frequency and length doesn't \
+match");
+ return -1;
+ }
+
+ // Calculate the timestamp that should be pushed to codec.
+ // This might be different from the timestamp of the frame
+ // due to re-sampling
+ bool resamplingRequired =
+ ((WebRtc_Word32)audioFrame._frequencyInHz != _sendCodecInst.plfreq);
+
+ // If number of channels in audio doesn't match codec mode, we need
+ // either mono-to-stereo or stereo-to-mono conversion.
+ WebRtc_Word16 audio[WEBRTC_10MS_PCM_AUDIO];
+ int audio_channels = _sendCodecInst.channels;
+ if (audioFrame._audioChannel != _sendCodecInst.channels) {
+ if (_sendCodecInst.channels == 2) {
+ // Do mono-to-stereo conversion by copying each sample.
+ for (int k = 0; k < audioFrame._payloadDataLengthInSamples; k++) {
+ audio[k * 2] = audioFrame._payloadData[k];
+ audio[(k * 2) + 1] = audioFrame._payloadData[k];
+ }
+ } else if (_sendCodecInst.channels == 1) {
+ // Do stereo-to-mono conversion by creating the average of the stereo
+ // samples.
+ for (int k = 0; k < audioFrame._payloadDataLengthInSamples; k++) {
+ audio[k] = (audioFrame._payloadData[k * 2] +
+ audioFrame._payloadData[(k * 2) + 1]) >> 1;
+ }
+ }
+ } else {
+ // Copy payload data for future use.
+ size_t length = static_cast<size_t>(
+ audioFrame._payloadDataLengthInSamples * audio_channels);
+ memcpy(audio, audioFrame._payloadData, length * sizeof(WebRtc_UWord16));
+ }
+
+ WebRtc_UWord32 currentTimestamp;
+ WebRtc_Word32 status;
+ // if it is required, we have to do a resampling.
+ if(resamplingRequired)
+ {
+ WebRtc_Word16 resampledAudio[WEBRTC_10MS_PCM_AUDIO];
+ WebRtc_Word32 sendPlFreq = _sendCodecInst.plfreq;
+ WebRtc_UWord32 diffInputTimestamp;
+ WebRtc_Word16 newLengthSmpl;
+
+ // calculate the timestamp of this frame
+ if(_lastInTimestamp > audioFrame._timeStamp)
+ {
+ // a wrap around has happened
+ diffInputTimestamp = ((WebRtc_UWord32)0xFFFFFFFF - _lastInTimestamp)
+ + audioFrame._timeStamp;
+ }
+ else
+ {
+ diffInputTimestamp = audioFrame._timeStamp - _lastInTimestamp;
+ }
+ currentTimestamp = _lastTimestamp + (WebRtc_UWord32)(diffInputTimestamp *
+ ((double)_sendCodecInst.plfreq / (double)audioFrame._frequencyInHz));
+
+ newLengthSmpl = _inputResampler.Resample10Msec(
+ audio, audioFrame._frequencyInHz, resampledAudio, sendPlFreq,
+ audio_channels);
+
+ if(newLengthSmpl < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot add 10 ms audio, resmapling failed");
+ return -1;
+ }
+ status = _codecs[_currentSendCodecIdx]->Add10MsData(currentTimestamp,
+ resampledAudio, newLengthSmpl, audio_channels);
+ }
+ else
+ {
+ currentTimestamp = audioFrame._timeStamp;
+
+ status = _codecs[_currentSendCodecIdx]->Add10MsData(currentTimestamp,
+ audio, audioFrame._payloadDataLengthInSamples,
+ audio_channels);
+ }
+ _lastInTimestamp = audioFrame._timeStamp;
+ _lastTimestamp = currentTimestamp;
+ return status;
+}
+
+/////////////////////////////////////////
+// (FEC) Forward Error Correction
+//
+
+bool
+AudioCodingModuleImpl::FECStatus() const
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ return _fecEnabled;
+}
+
+// configure FEC status i.e on/off
+WebRtc_Word32
+AudioCodingModuleImpl::SetFECStatus(
+#ifdef WEBRTC_CODEC_RED
+ const bool enableFEC)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if (_fecEnabled != enableFEC)
+ {
+ // Reset the RED buffer
+ memset(_redBuffer, 0, MAX_PAYLOAD_SIZE_BYTE);
+
+ // Reset fragmentation buffers
+ _fragmentation->fragmentationVectorSize = 2;
+ _fragmentation->fragmentationOffset[0] = 0;
+ _fragmentation->fragmentationOffset[1] = MAX_PAYLOAD_SIZE_BYTE;
+ memset(_fragmentation->fragmentationLength, 0, sizeof(WebRtc_UWord32) * 2);
+ memset(_fragmentation->fragmentationTimeDiff, 0, sizeof(WebRtc_UWord16) * 2);
+ memset(_fragmentation->fragmentationPlType, 0, sizeof(WebRtc_UWord8) * 2);
+
+ // set _fecEnabled
+ _fecEnabled = enableFEC;
+ }
+ _isFirstRED = true; // Make sure we restart FEC
+ return 0;
+#else
+ const bool /* enableFEC */)
+{
+ _fecEnabled = false;
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
+ " WEBRTC_CODEC_RED is undefined => _fecEnabled = %d", _fecEnabled);
+ return -1;
+#endif
+}
+
+
+/////////////////////////////////////////
+// (VAD) Voice Activity Detection
+//
+
+WebRtc_Word32
+AudioCodingModuleImpl::SetVAD(
+ const bool enableDTX,
+ const bool enableVAD,
+ const ACMVADMode vadMode)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ // sanity check of the mode
+ if((vadMode != VADNormal) &&
+ (vadMode != VADLowBitrate) &&
+ (vadMode != VADAggr) &&
+ (vadMode != VADVeryAggr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Invalid VAD Mode %d, no change is made to VAD/DTX status",
+ (int)vadMode);
+ return -1;
+ }
+
+ // If a send codec is registered, set VAD/DTX for the codec
+ if(HaveValidEncoder("SetVAD")) {
+ WebRtc_Word16 status =
+ _codecs[_currentSendCodecIdx]->SetVAD(enableDTX, enableVAD, vadMode);
+ if(status == 1) {
+ // Vad was enabled;
+ _vadEnabled = true;
+ _dtxEnabled = enableDTX;
+ _vadMode = vadMode;
+
+ return 0;
+ } else if (status < 0) {
+ // SetVAD failed
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "SetVAD failed");
+
+ _vadEnabled = false;
+ _dtxEnabled = false;
+
+ return -1;
+ }
+ }
+
+ _vadEnabled = enableVAD;
+ _dtxEnabled = enableDTX;
+ _vadMode = vadMode;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::VAD(
+ bool& dtxEnabled,
+ bool& vadEnabled,
+ ACMVADMode& vadMode) const
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ dtxEnabled = _dtxEnabled;
+ vadEnabled = _vadEnabled;
+ vadMode = _vadMode;
+
+ return 0;
+}
+
+/////////////////////////////////////////
+// Receiver
+//
+
+WebRtc_Word32
+AudioCodingModuleImpl::InitializeReceiver()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ return InitializeReceiverSafe();
+}
+
+// Initialize receiver, resets codec database etc
+WebRtc_Word32
+AudioCodingModuleImpl::InitializeReceiverSafe()
+{
+ // If the receiver is already initialized then we
+ // also like to destruct decoders if any exist. After a call
+ // to this function, we should have a clean start-up.
+ if(_receiverInitialized)
+ {
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
+ {
+ if(UnregisterReceiveCodecSafe(codecCntr) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "InitializeReceiver() failed, Could not unregister codec");
+ return -1;
+ }
+ }
+ }
+ if (_netEq.Init() != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "InitializeReceiver() failed, Could not initialize NetEQ");
+ return -1;
+ }
+ _netEq.SetUniqueId(_id);
+ if (_netEq.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(),
+ ACMCodecDB::kNumCodecs) != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "NetEQ cannot allocatePacket Buffer");
+ return -1;
+ }
+
+ // Register RED and CN
+ int regInNeteq = 0;
+ for (int i = (ACMCodecDB::kNumCodecs - 1); i>-1; i--) {
+ if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "red") == 0)) {
+ regInNeteq = 1;
+ } else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0)) {
+ regInNeteq = 1;
+ }
+
+ if (regInNeteq == 1) {
+ if(RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i,
+ ACMNetEQ::masterJB) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot register master codec.");
+ return -1;
+ }
+ _registeredPlTypes[i] = ACMCodecDB::database_[i].pltype;
+ regInNeteq = 0;
+ }
+ }
+
+ _receiverInitialized = true;
+ return 0;
+}
+
+// Reset the decoder state
+WebRtc_Word32
+AudioCodingModuleImpl::ResetDecoder()
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
+ {
+ if((_codecs[codecCntr] != NULL) && (_registeredPlTypes[codecCntr] != -1))
+ {
+ if(_codecs[codecCntr]->ResetDecoder(_registeredPlTypes[codecCntr]) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "ResetDecoder failed:");
+ return -1;
+ }
+ }
+ }
+ return _netEq.FlushBuffers();
+}
+
+// get current receive freq
+WebRtc_Word32
+AudioCodingModuleImpl::ReceiveFrequency() const
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "ReceiveFrequency()");
+ WebRtcACMCodecParams codecParams;
+
+ CriticalSectionScoped lock(*_acmCritSect);
+ if(DecoderParamByPlType(_lastRecvAudioCodecPlType, codecParams) < 0)
+ {
+ return _netEq.CurrentSampFreqHz();
+ }
+ else
+ {
+ return codecParams.codecInstant.plfreq;
+ }
+}
+
+// get current playout freq
+WebRtc_Word32
+AudioCodingModuleImpl::PlayoutFrequency() const
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "PlayoutFrequency()");
+
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ return _netEq.CurrentSampFreqHz();
+}
+
+
+// register possible reveive codecs, can be called multiple times,
+// for codecs, CNG (NB, WB and SWB), DTMF, RED
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterReceiveCodec(
+ const CodecInst& receiveCodec)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(receiveCodec.channels > 2)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "More than 2 audio channel is not supported.");
+ return -1;
+ }
+
+ int mirrorId;
+ int codecId = ACMCodecDB::ReceiverCodecNumber(&receiveCodec, &mirrorId);
+
+ if(codecId < 0 || codecId >= ACMCodecDB::kNumCodecs)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Wrong codec params to be registered as receive codec");
+ return -1;
+ }
+ // Check if the payload-type is valid.
+ if(!ACMCodecDB::ValidPayloadType(receiveCodec.pltype))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Invalid payload-type %d for %s.",
+ receiveCodec.pltype, receiveCodec.plname);
+ return -1;
+ }
+
+ if(!_receiverInitialized)
+ {
+ if(InitializeReceiverSafe() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot initialize reciver, so failed registering a codec.");
+ return -1;
+ }
+ }
+
+ // If codec already registered, start with unregistering
+ if(_registeredPlTypes[codecId] != -1)
+ {
+ if(UnregisterReceiveCodecSafe(codecId) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot register master codec.");
+ return -1;
+ }
+ }
+
+ if(RegisterRecCodecMSSafe(receiveCodec, codecId, mirrorId,
+ ACMNetEQ::masterJB) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot register master codec.");
+ return -1;
+ }
+
+ // If receive stereo, make sure we have two instances of NetEQ, one for each channel
+ if(receiveCodec.channels == 2)
+ {
+ if(_netEq.NumSlaves() < 1)
+ {
+ if(_netEq.AddSlave(ACMCodecDB::NetEQDecoders(),
+ ACMCodecDB::kNumCodecs) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot Add Slave jitter buffer to NetEQ.");
+ return -1;
+ }
+
+ // Register RED and CN in slave.
+ bool reg_in_neteq = false;
+ for (int i = (ACMCodecDB::kNumCodecs - 1); i > -1; i--) {
+ if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "RED") == 0)) {
+ reg_in_neteq = true;
+ } else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0)) {
+ reg_in_neteq = true;
+ }
+
+ if (reg_in_neteq) {
+ if(RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i,
+ ACMNetEQ::slaveJB) < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot register slave codec.");
+ return -1;
+ }
+ _registeredPlTypes[i] = ACMCodecDB::database_[i].pltype;
+ reg_in_neteq = false;
+ }
+ }
+ }
+
+ if(RegisterRecCodecMSSafe(receiveCodec, codecId, mirrorId,
+ ACMNetEQ::slaveJB) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot register slave codec.");
+ return -1;
+ }
+
+ if((_stereoReceive[codecId] == false) &&
+ (_lastRecvAudioCodecPlType == receiveCodec.pltype))
+ {
+ _lastRecvAudioCodecPlType = -1;
+ }
+ _stereoReceive[codecId] = true;
+ }
+ else
+ {
+ _stereoReceive[codecId] = false;
+ }
+
+ _registeredPlTypes[codecId] = receiveCodec.pltype;
+
+ if(!STR_CASE_CMP(receiveCodec.plname, "RED"))
+ {
+ _receiveREDPayloadType = receiveCodec.pltype;
+ }
+ return 0;
+}
+
+
+
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterRecCodecMSSafe(
+ const CodecInst& receiveCodec,
+ WebRtc_Word16 codecId,
+ WebRtc_Word16 mirrorId,
+ ACMNetEQ::JB jitterBuffer)
+{
+ ACMGenericCodec** codecArray;
+ if(jitterBuffer == ACMNetEQ::masterJB)
+ {
+ codecArray = &_codecs[0];
+ }
+ else if(jitterBuffer == ACMNetEQ::slaveJB)
+ {
+ codecArray = &_slaveCodecs[0];
+ if (_codecs[codecId]->IsTrueStereoCodec()) {
+ // True stereo codecs need to use the same codec memory
+ // for both master and slave.
+ _slaveCodecs[mirrorId] = _codecs[mirrorId];
+ _mirrorCodecIdx[mirrorId] = mirrorId;
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "RegisterReceiveCodecMSSafe failed, jitterBuffer is neither master or slave ");
+ return -1;
+ }
+
+ if (codecArray[mirrorId] == NULL)
+ {
+ codecArray[mirrorId] = CreateCodec(receiveCodec);
+ if(codecArray[mirrorId] == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot create codec to register as receive codec");
+ return -1;
+ }
+ _mirrorCodecIdx[mirrorId] = mirrorId;
+ }
+ if(mirrorId != codecId)
+ {
+ codecArray[codecId] = codecArray[mirrorId];
+ _mirrorCodecIdx[codecId] = mirrorId;
+ }
+
+ codecArray[codecId]->SetIsMaster(jitterBuffer == ACMNetEQ::masterJB);
+
+ WebRtc_Word16 status = 0;
+ bool registerInNetEq = true;
+ WebRtcACMCodecParams codecParams;
+ memcpy(&(codecParams.codecInstant), &receiveCodec,
+ sizeof(CodecInst));
+ codecParams.enableVAD = false;
+ codecParams.enableDTX = false;
+ codecParams.vadMode = VADNormal;
+ if (!codecArray[codecId]->DecoderInitialized())
+ {
+ // force initialization
+ status = codecArray[codecId]->InitDecoder(&codecParams, true);
+ if(status < 0)
+ {
+ // could not initialize the decoder we don't want to
+ // continue if we could not initialize properly.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "could not initialize the receive codec, codec not registered");
+
+ return -1;
+ }
+ }
+ else if(mirrorId != codecId)
+ {
+ // Currently this only happens for iSAC.
+ // we have to store the decoder parameters
+
+ codecArray[codecId]->SaveDecoderParam(&codecParams);
+ }
+ if (registerInNetEq)
+ {
+ if(codecArray[codecId]->RegisterInNetEq(&_netEq, receiveCodec)
+ != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Receive codec could not be registered in NetEQ");
+
+ return -1;
+ }
+ // Guaranty that the same payload-type that is
+ // registered in NetEQ is stored in the codec.
+ codecArray[codecId]->SaveDecoderParam(&codecParams);
+ }
+
+ return status;
+}
+
+
+
+// Get current received codec
+WebRtc_Word32
+AudioCodingModuleImpl::ReceiveCodec(
+ CodecInst& currentReceiveCodec) const
+{
+ WebRtcACMCodecParams decoderParam;
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ for(int decCntr = 0; decCntr < ACMCodecDB::kMaxNumCodecs; decCntr++)
+ {
+ if(_codecs[decCntr] != NULL)
+ {
+ if(_codecs[decCntr]->DecoderInitialized())
+ {
+ if(_codecs[decCntr]->DecoderParams(&decoderParam,
+ _lastRecvAudioCodecPlType))
+ {
+ memcpy(¤tReceiveCodec, &decoderParam.codecInstant,
+ sizeof(CodecInst));
+ return 0;
+ }
+ }
+ }
+ }
+
+ // if we are here then we haven't found any codec
+ // set codec pltype to -1 to indicate that the structure
+ // is invalid and return -1.
+ currentReceiveCodec.pltype = -1;
+ return -1;
+}
+
+// Incoming packet from network parsed and ready for decode
+WebRtc_Word32
+AudioCodingModuleImpl::IncomingPacket(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo)
+{
+ if (payloadLength < 0)
+ {
+ // Log error
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, payload-length cannot be negative");
+ return -1;
+ }
+ {
+ // store the payload Type. this will be used to retrieve "received codec"
+ // and "received frequency."
+ CriticalSectionScoped lock(*_acmCritSect);
+#ifdef ACM_QA_TEST
+ if(_incomingPL != NULL)
+ {
+ fwrite(&rtpInfo.header.timestamp, sizeof(WebRtc_UWord32), 1, _incomingPL);
+ fwrite(&rtpInfo.header.payloadType, sizeof(WebRtc_UWord8), 1, _incomingPL);
+ fwrite(&payloadLength, sizeof(WebRtc_Word16), 1, _incomingPL);
+ }
+#endif
+
+ WebRtc_UWord8 myPayloadType;
+
+ // Check if this is an RED payload
+ if(rtpInfo.header.payloadType == _receiveREDPayloadType)
+ {
+ // get the primary payload-type.
+ myPayloadType = (WebRtc_UWord8)(incomingPayload[0] & 0x7F);
+ }
+ else
+ {
+ myPayloadType = rtpInfo.header.payloadType;
+ }
+
+ // If payload is audio, check if received payload is different from previous
+ if((!rtpInfo.type.Audio.isCNG) &&
+ (myPayloadType != _cngNB.pltype) &&
+ (myPayloadType != _cngWB.pltype) &&
+ (myPayloadType != _cngSWB.pltype))
+ {
+ // This is Audio not CNG
+
+ if(myPayloadType != _lastRecvAudioCodecPlType)
+ {
+ // We detect a change in payload type. It is necessary for iSAC
+ // we are going to use ONE iSAC instance for decoding both WB and
+ // SWB payloads. If payload is changed there might be a need to reset
+ // sampling rate of decoder. depending what we have received "now".
+ for(int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
+ {
+ if(_registeredPlTypes[i] == myPayloadType)
+ {
+ if(_codecs[i] == NULL)
+ {
+ // we found a payload type but the corresponding
+ // codec is NULL this should not happen
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, payload type found but corresponding "
+ "codec is NULL");
+ return -1;
+ }
+ _codecs[i]->UpdateDecoderSampFreq(i);
+ _netEq.SetReceivedStereo(_stereoReceive[i]);
+
+ // Store number of channels we expect to receive for the
+ // current payload type.
+ if (_stereoReceive[i]) {
+ _expected_channels = 2;
+ } else {
+ _expected_channels = 1;
+ }
+
+ // Reset previous received channel
+ _prev_received_channel = 0;
+
+ break;
+ }
+ }
+ }
+ _lastRecvAudioCodecPlType = myPayloadType;
+ }
+ }
+
+ // Check that number of received channels match the setup for the
+ // received codec.
+ if (_expected_channels == 2) {
+ if ((_prev_received_channel == 1) && (rtpInfo.type.Audio.channel == 1)) {
+ // We expect every second call to this function to be for channel 2,
+ // since we are in stereo-receive mode.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, payload is"
+ "mono, but codec registered as stereo.");
+ return -1;
+ }
+ _prev_received_channel = rtpInfo.type.Audio.channel;
+ } else if (rtpInfo.type.Audio.channel == 2) {
+ // Codec is registered as mono, but we receive a stereo packet.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, payload is"
+ "stereo, but codec registered as mono.");
+ return -1;
+ }
+
+ // Insert packet into NetEQ.
+ return _netEq.RecIn(incomingPayload, payloadLength, rtpInfo);
+}
+
+// Minimum playout delay (Used for lip-sync)
+WebRtc_Word32
+AudioCodingModuleImpl::SetMinimumPlayoutDelay(
+ const WebRtc_Word32 timeMs)
+{
+ if((timeMs < 0) || (timeMs > 1000))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Delay must be in the range of 0-1000 milliseconds.");
+ return -1;
+ }
+ return _netEq.SetExtraDelay(timeMs);
+}
+
+// Get Dtmf playout status
+bool
+AudioCodingModuleImpl::DtmfPlayoutStatus() const
+{
+#ifndef WEBRTC_CODEC_AVT
+ return false;
+#else
+ return _netEq.AVTPlayout();
+#endif
+}
+
+// configure Dtmf playout status i.e on/off
+// playout the incoming outband Dtmf tone
+WebRtc_Word32
+AudioCodingModuleImpl::SetDtmfPlayoutStatus(
+#ifndef WEBRTC_CODEC_AVT
+ const bool /* enable */)
+{
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
+ "SetDtmfPlayoutStatus() failed: AVT is not supported.");
+ return -1;
+#else
+ const bool enable)
+{
+ return _netEq.SetAVTPlayout(enable);
+#endif
+}
+
+// Estimate the Bandwidth based on the incoming stream
+// This is also done in the RTP module
+// need this for one way audio where the RTCP send the BW estimate
+WebRtc_Word32
+AudioCodingModuleImpl::DecoderEstimatedBandwidth() const
+{
+ CodecInst codecInst;
+ WebRtc_Word16 codecID = -1;
+ int plTypWB;
+ int plTypSWB;
+
+ // Get iSAC settings
+ for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
+ {
+ // Store codec settings for codec number "codeCntr" in the output struct
+ ACMCodecDB::Codec(codecCntr, &codecInst);
+
+ if(!STR_CASE_CMP(codecInst.plname, "isac"))
+ {
+ codecID = 1;
+ plTypWB = codecInst.pltype;
+
+ ACMCodecDB::Codec(codecCntr+1, &codecInst);
+ plTypSWB = codecInst.pltype;
+
+ break;
+ }
+ }
+
+ if(codecID < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "DecoderEstimatedBandwidth failed");
+ return -1;
+ }
+
+ if ((_lastRecvAudioCodecPlType == plTypWB) || (_lastRecvAudioCodecPlType == plTypSWB))
+ {
+ return _codecs[codecID]->GetEstimatedBandwidth();
+ } else {
+ return -1;
+ }
+}
+
+// Set playout mode for: voice, fax, or streaming
+WebRtc_Word32
+AudioCodingModuleImpl::SetPlayoutMode(
+ const AudioPlayoutMode mode)
+{
+ if((mode != voice) &&
+ (mode != fax) &&
+ (mode != streaming))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Invalid playout mode.");
+ return -1;
+ }
+ return _netEq.SetPlayoutMode(mode);
+}
+
+// Get playout mode voice, fax
+AudioPlayoutMode
+AudioCodingModuleImpl::PlayoutMode() const
+{
+ return _netEq.PlayoutMode();
+}
+
+
+// Get 10 milliseconds of raw audio data to play out
+// automatic resample to the requested frequency
+WebRtc_Word32
+AudioCodingModuleImpl::PlayoutData10Ms(
+ const WebRtc_Word32 desiredFreqHz,
+ AudioFrame& audioFrame)
+{
+ bool stereoMode;
+
+ // recOut always returns 10 ms
+ if (_netEq.RecOut(_audioFrame) != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "PlayoutData failed, RecOut Failed");
+ return -1;
+ }
+
+ audioFrame._audioChannel = _audioFrame._audioChannel;
+ audioFrame._vadActivity = _audioFrame._vadActivity;
+ audioFrame._speechType = _audioFrame._speechType;
+
+ stereoMode = (_audioFrame._audioChannel > 1);
+ //For stereo playout:
+ // Master and Slave samples are interleaved starting with Master
+
+ const WebRtc_UWord16 recvFreq = static_cast<WebRtc_UWord16>(_audioFrame._frequencyInHz);
+ bool toneDetected = false;
+ WebRtc_Word16 lastDetectedTone;
+ WebRtc_Word16 tone;
+
+ // limit the scope of ACM Critical section
+ // perhaps we don't need to have output resampler in
+ // critical section, it is supposed to be called in this
+ // function and no where else. However, it won't degrade complexity
+ {
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if ((recvFreq != desiredFreqHz) && (desiredFreqHz != -1))
+ {
+ // resample payloadData
+ WebRtc_Word16 tmpLen = _outputResampler.Resample10Msec(
+ _audioFrame._payloadData, recvFreq, audioFrame._payloadData, desiredFreqHz,
+ _audioFrame._audioChannel);
+
+ if(tmpLen < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "PlayoutData failed, resampler failed");
+ return -1;
+ }
+
+ //Set the payload data length from the resampler
+ audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)tmpLen;
+ // set the ssampling frequency
+ audioFrame._frequencyInHz = desiredFreqHz;
+ }
+ else
+ {
+ memcpy(audioFrame._payloadData, _audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel
+ * sizeof(WebRtc_Word16));
+ // set the payload length
+ audioFrame._payloadDataLengthInSamples = _audioFrame._payloadDataLengthInSamples;
+ // set the sampling frequency
+ audioFrame._frequencyInHz = recvFreq;
+ }
+
+ //Tone detection done for master channel
+ if(_dtmfDetector != NULL)
+ {
+ // Dtmf Detection
+ if(audioFrame._frequencyInHz == 8000)
+ {
+ // use audioFrame._payloadData then Dtmf detector doesn't
+ // need resampling
+ if(!stereoMode)
+ {
+ _dtmfDetector->Detect(audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples,
+ audioFrame._frequencyInHz, toneDetected, tone);
+ }
+ else
+ {
+ // we are in 8 kHz so the master channel needs only 80 samples
+ WebRtc_Word16 masterChannel[80];
+ for(int n = 0; n < 80; n++)
+ {
+ masterChannel[n] = audioFrame._payloadData[n<<1];
+ }
+ _dtmfDetector->Detect(masterChannel,
+ audioFrame._payloadDataLengthInSamples,
+ audioFrame._frequencyInHz, toneDetected, tone);
+ }
+ }
+ else
+ {
+ // Do the detection on the audio that we got from NetEQ (_audioFrame).
+ if(!stereoMode)
+ {
+ _dtmfDetector->Detect(_audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples, recvFreq,
+ toneDetected, tone);
+ }
+ else
+ {
+ WebRtc_Word16 masterChannel[WEBRTC_10MS_PCM_AUDIO];
+ for(int n = 0; n < _audioFrame._payloadDataLengthInSamples; n++)
+ {
+ masterChannel[n] = _audioFrame._payloadData[n<<1];
+ }
+ _dtmfDetector->Detect(masterChannel,
+ _audioFrame._payloadDataLengthInSamples, recvFreq,
+ toneDetected, tone);
+ }
+ }
+ }
+
+ // we want to do this while we are in _acmCritSect
+ // doesn't really need to initialize the following
+ // variable but Linux complains if we don't
+ lastDetectedTone = kACMToneEnd;
+ if(toneDetected)
+ {
+ lastDetectedTone = _lastDetectedTone;
+ _lastDetectedTone = tone;
+ }
+ }
+
+ if(toneDetected)
+ {
+ // we will deal with callback here, so enter callback critical
+ // section
+ CriticalSectionScoped lock(*_callbackCritSect);
+
+ if(_dtmfCallback != NULL)
+ {
+ if(tone != kACMToneEnd)
+ {
+ // just a tone
+ _dtmfCallback->IncomingDtmf((WebRtc_UWord8)tone, false);
+ }
+ else if((tone == kACMToneEnd) &&
+ (lastDetectedTone != kACMToneEnd))
+ {
+ // The tone is "END" and the previously detected tone is
+ // not "END," so call fir an end.
+ _dtmfCallback->IncomingDtmf((WebRtc_UWord8)lastDetectedTone,
+ true);
+ }
+ }
+ }
+
+ audioFrame._id = _id;
+ audioFrame._volume = -1;
+ audioFrame._energy = -1;
+ audioFrame._timeStamp = 0;
+
+ return 0;
+}
+
+/////////////////////////////////////////
+// (CNG) Comfort Noise Generation
+// Generate comfort noise when receiving DTX packets
+//
+
+// Get VAD aggressiveness on the incoming stream
+ACMVADMode
+AudioCodingModuleImpl::ReceiveVADMode() const
+{
+ return _netEq.VADMode();
+}
+
+// Configure VAD aggressiveness on the incoming stream
+WebRtc_Word16
+AudioCodingModuleImpl::SetReceiveVADMode(
+ const ACMVADMode mode)
+{
+ return _netEq.SetVADMode(mode);
+}
+
+/////////////////////////////////////////
+// statistics
+//
+
+WebRtc_Word32
+AudioCodingModuleImpl::NetworkStatistics(
+ ACMNetworkStatistics& statistics) const
+{
+ WebRtc_Word32 status;
+ status = _netEq.NetworkStatistics(&statistics);
+ return status;
+}
+
+void
+AudioCodingModuleImpl::DestructEncoderInst(
+ void* ptrInst)
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
+ "DestructEncoderInst()");
+ if(!HaveValidEncoder("DestructEncoderInst"))
+ {
+ return;
+ }
+
+ _codecs[_currentSendCodecIdx]->DestructEncoderInst(ptrInst);
+}
+
+WebRtc_Word16
+AudioCodingModuleImpl::AudioBuffer(
+ WebRtcACMAudioBuff& audioBuff)
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
+ "AudioBuffer()");
+ if(!HaveValidEncoder("AudioBuffer"))
+ {
+ return -1;
+ }
+
+ audioBuff.lastInTimestamp = _lastInTimestamp;
+ return _codecs[_currentSendCodecIdx]->AudioBuffer(audioBuff);
+}
+
+WebRtc_Word16
+AudioCodingModuleImpl::SetAudioBuffer(
+ WebRtcACMAudioBuff& audioBuff)
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
+ "SetAudioBuffer()");
+ if(!HaveValidEncoder("SetAudioBuffer"))
+ {
+ return -1;
+ }
+
+ return _codecs[_currentSendCodecIdx]->SetAudioBuffer(audioBuff);
+}
+
+
+WebRtc_UWord32
+AudioCodingModuleImpl::EarliestTimestamp() const
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
+ "EarliestTimestamp()");
+ if(!HaveValidEncoder("EarliestTimestamp"))
+ {
+ return -1;
+ }
+
+ return _codecs[_currentSendCodecIdx]->EarliestTimestamp();
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::RegisterVADCallback(
+ ACMVADCallback* vadCallback)
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
+ "RegisterVADCallback()");
+ CriticalSectionScoped lock(*_callbackCritSect);
+ _vadCallback = vadCallback;
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::IncomingPayload(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timestamp)
+{
+ if (payloadLength < 0)
+ {
+ // Log error in trace file.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, payload-length cannot be negative");
+ return -1;
+ }
+
+ if(_dummyRTPHeader == NULL)
+ {
+ // This is the first time that we are using _dummyRTPHeader
+ // so we have to create it.
+ WebRtcACMCodecParams codecParams;
+ _dummyRTPHeader = new WebRtcRTPHeader;
+ if (_dummyRTPHeader == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "IncomingPacket() Error, out of memory");
+ return -1;
+ }
+ _dummyRTPHeader->header.payloadType = payloadType;
+ // Don't matter in this case
+ _dummyRTPHeader->header.ssrc = 0;
+ _dummyRTPHeader->header.markerBit = false;
+ // start with random numbers
+ _dummyRTPHeader->header.sequenceNumber = rand();
+ _dummyRTPHeader->header.timestamp = (((WebRtc_UWord32)rand()) << 16) +
+ (WebRtc_UWord32)rand();
+ _dummyRTPHeader->type.Audio.channel = 1;
+
+ if(DecoderParamByPlType(payloadType, codecParams) < 0)
+ {
+ // we didn't find a codec with the given payload.
+ // something is wrong we exit, but we delete _dummyRTPHeader
+ // and set it to NULL to start clean next time
+ delete _dummyRTPHeader;
+ _dummyRTPHeader = NULL;
+ return -1;
+ }
+ _recvPlFrameSizeSmpls = codecParams.codecInstant.pacsize;
+ }
+
+ if(payloadType != _dummyRTPHeader->header.payloadType)
+ {
+ // payload type has changed since the last time we might need to
+ // update the frame-size
+ WebRtcACMCodecParams codecParams;
+ if(DecoderParamByPlType(payloadType, codecParams) < 0)
+ {
+ // we didn't find a codec with the given payload.
+ // something is wrong we exit
+ return -1;
+ }
+ _recvPlFrameSizeSmpls = codecParams.codecInstant.pacsize;
+ _dummyRTPHeader->header.payloadType = payloadType;
+ }
+
+ if(timestamp > 0)
+ {
+ _dummyRTPHeader->header.timestamp = timestamp;
+ }
+
+ // store the payload Type. this will be used to retrieve "received codec"
+ // and "received frequency."
+ _lastRecvAudioCodecPlType = payloadType;
+
+ // Insert in NetEQ
+ if(_netEq.RecIn(incomingPayload, payloadLength, (*_dummyRTPHeader)) < 0)
+ {
+ return -1;
+ }
+
+ // get ready for the next payload
+ _dummyRTPHeader->header.sequenceNumber++;
+ _dummyRTPHeader->header.timestamp += _recvPlFrameSizeSmpls;
+ return 0;
+}
+
+WebRtc_Word16
+AudioCodingModuleImpl::DecoderParamByPlType(
+ const WebRtc_UWord8 payloadType,
+ WebRtcACMCodecParams& codecParams) const
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ for(WebRtc_Word16 codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
+ {
+ if(_codecs[codecCntr] != NULL)
+ {
+ if(_codecs[codecCntr]->DecoderInitialized())
+ {
+ if(_codecs[codecCntr]->DecoderParams(&codecParams,
+ payloadType))
+ {
+ return 0;
+ }
+ }
+ }
+ }
+ // if we are here it means that we could not find a
+ // codec with that payload type. reset the values to
+ // not acceptable values and return -1;
+ codecParams.codecInstant.plname[0] = '\0';
+ codecParams.codecInstant.pacsize = 0;
+ codecParams.codecInstant.rate = 0;
+ codecParams.codecInstant.pltype = -1;
+ return -1;
+}
+
+
+
+WebRtc_Word16
+AudioCodingModuleImpl::DecoderListIDByPlName(
+ const char* payloadName,
+ const WebRtc_UWord16 sampFreqHz) const
+{
+ WebRtcACMCodecParams codecParams;
+ CriticalSectionScoped lock(*_acmCritSect);
+ for(WebRtc_Word16 codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
+ {
+ if((_codecs[codecCntr] != NULL))
+ {
+ if(_codecs[codecCntr]->DecoderInitialized())
+ {
+ assert(_registeredPlTypes[codecCntr] >= 0);
+ assert(_registeredPlTypes[codecCntr] <= 255);
+ _codecs[codecCntr]->DecoderParams(&codecParams,
+ (WebRtc_UWord8)_registeredPlTypes[codecCntr]);
+ if(!STR_CASE_CMP(codecParams.codecInstant.plname, payloadName))
+ {
+ // Check if the given sampling frequency matches.
+ // A zero sampling frequency means we matching the names
+ // is sufficient and we don't need to check for the
+ // frequencies.
+ // Currently it is only iSAC which has one name but two
+ // sampling frequencies.
+ if((sampFreqHz == 0) ||
+ (codecParams.codecInstant.plfreq == sampFreqHz))
+ {
+ return codecCntr;
+ }
+ }
+ }
+ }
+ }
+ // if we are here it means that we could not find a
+ // codec with that payload type. return -1;
+ return -1;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::LastEncodedTimestamp(WebRtc_UWord32& timestamp) const
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ if(!HaveValidEncoder("LastEncodedTimestamp"))
+ {
+ return -1;
+ }
+ timestamp = _codecs[_currentSendCodecIdx]->LastEncodedTimestamp();
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool useWebRtcDTX)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("ReplaceInternalDTXWithWebRtc"))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Cannot replace codec internal DTX when no send codec is registered.");
+ return -1;
+ }
+
+ WebRtc_Word32 res = _codecs[_currentSendCodecIdx]->ReplaceInternalDTX(useWebRtcDTX);
+ // Check if VAD is turned on, or if there is any error
+ if(res == 1)
+ {
+ _vadEnabled = true;
+ } else if(res < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Failed to set ReplaceInternalDTXWithWebRtc(%d)", useWebRtcDTX);
+ return res;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(bool& usesWebRtcDTX)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc"))
+ {
+ return -1;
+ }
+ if(_codecs[_currentSendCodecIdx]->IsInternalDTXReplaced(&usesWebRtcDTX) < 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+AudioCodingModuleImpl::SetISACMaxRate(
+ const WebRtc_UWord32 maxRateBitPerSec)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("SetISACMaxRate"))
+ {
+ return -1;
+ }
+
+ return _codecs[_currentSendCodecIdx]->SetISACMaxRate(maxRateBitPerSec);
+}
+
+
+WebRtc_Word32
+AudioCodingModuleImpl::SetISACMaxPayloadSize(
+ const WebRtc_UWord16 maxPayloadLenBytes)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("SetISACMaxPayloadSize"))
+ {
+ return -1;
+ }
+
+ return _codecs[_currentSendCodecIdx]->SetISACMaxPayloadSize(maxPayloadLenBytes);
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+
+ if(!HaveValidEncoder("ConfigISACBandwidthEstimator"))
+ {
+ return -1;
+ }
+
+ return _codecs[_currentSendCodecIdx]->ConfigISACBandwidthEstimator(
+ initFrameSizeMsec, initRateBitPerSec, enforceFrameSize);
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::SetBackgroundNoiseMode(
+ const ACMBackgroundNoiseMode mode)
+{
+ if((mode < On) ||
+ (mode > Off))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "The specified background noise is out of range.\n");
+ return -1;
+ }
+ return _netEq.SetBackgroundNoiseMode(mode);
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::BackgroundNoiseMode(
+ ACMBackgroundNoiseMode& mode)
+{
+ return _netEq.BackgroundNoiseMode(mode);
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::PlayoutTimestamp(
+ WebRtc_UWord32& timestamp)
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
+ "PlayoutTimestamp()");
+ return _netEq.PlayoutTimestamp(timestamp);
+}
+
+bool
+AudioCodingModuleImpl::HaveValidEncoder(
+ const char* callerName) const
+{
+ if((!_sendCodecRegistered) ||
+ (_currentSendCodecIdx < 0) ||
+ (_currentSendCodecIdx >= ACMCodecDB::kNumCodecs))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "%s failed: No send codec is registered.", callerName);
+ return false;
+ }
+ if((_currentSendCodecIdx < 0) ||
+ (_currentSendCodecIdx >= ACMCodecDB::kNumCodecs))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "%s failed: Send codec index out of range.", callerName);
+ return false;
+ }
+ if(_codecs[_currentSendCodecIdx] == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "%s failed: Send codec is NULL pointer.", callerName);
+ return false;
+ }
+ return true;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::UnregisterReceiveCodec(
+ const WebRtc_Word16 payloadType)
+{
+ CriticalSectionScoped lock(*_acmCritSect);
+ WebRtc_Word16 codecID;
+
+ // Search through the list of registered payload types
+ for (codecID = 0; codecID < ACMCodecDB::kMaxNumCodecs; codecID++)
+ {
+ if (_registeredPlTypes[codecID] == payloadType)
+ {
+ // we have found the codecID registered with the payload type
+ break;
+ }
+ }
+
+ if(codecID >= ACMCodecDB::kNumCodecs)
+ {
+ // payload type was not registered. No need to unregister
+ return 0;
+ }
+
+ // Unregister the codec with the given payload type
+ return UnregisterReceiveCodecSafe(codecID);
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::UnregisterReceiveCodecSafe(
+ const WebRtc_Word16 codecID)
+{
+ const WebRtcNetEQDecoder *neteqDecoder = ACMCodecDB::NetEQDecoders();
+ WebRtc_Word16 mirrorID = ACMCodecDB::MirrorID(codecID);
+ if(_codecs[codecID] != NULL)
+ {
+ if(_registeredPlTypes[codecID] != -1)
+ {
+ // before deleting the decoder instance unregister
+ // from NetEQ.
+ if(_netEq.RemoveCodec(neteqDecoder[codecID], _stereoReceive[codecID]) < 0)
+ {
+ CodecInst codecInst;
+ ACMCodecDB::Codec(codecID, &codecInst);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
+ "Unregistering %s-%d from NetEQ failed.",
+ codecInst.plname, codecInst.plfreq);
+ return -1;
+ }
+
+ // CN is a special case for NetEQ, all three sampling frequencies are
+ // deletad if one is deleted
+ if(STR_CASE_CMP(ACMCodecDB::database_[codecID].plname, "CN") == 0)
+ {
+ // Search codecs nearby in the database to unregister all CN.
+ for (int i=-2; i<3; i++)
+ {
+ if (STR_CASE_CMP(ACMCodecDB::database_[codecID+i].plname, "CN") == 0)
+ {
+ _codecs[codecID+i]->DestructDecoder();
+ if(_stereoReceive[codecID+i])
+ {
+ _slaveCodecs[codecID+i]->DestructDecoder();
+ }
+ _registeredPlTypes[codecID+i] = -1;
+ }
+ }
+ } else
+ {
+ if(codecID == mirrorID)
+ {
+ _codecs[codecID]->DestructDecoder();
+ if(_stereoReceive[codecID])
+ {
+ _slaveCodecs[codecID]->DestructDecoder();
+ }
+ }
+ }
+ }
+ }
+
+ if(_registeredPlTypes[codecID] == _receiveREDPayloadType)
+ {
+ // RED is going to be unregistered.
+ // set the following to an invalid value.
+ _receiveREDPayloadType = 255;
+ }
+ _registeredPlTypes[codecID] = -1;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioCodingModuleImpl::REDPayloadISAC(
+ const WebRtc_Word32 isacRate,
+ const WebRtc_Word16 isacBwEstimate,
+ WebRtc_UWord8* payload,
+ WebRtc_Word16* payloadLenByte)
+{
+ if(!HaveValidEncoder("EncodeData"))
+ {
+ return -1;
+ }
+ WebRtc_Word16 status;
+
+ status = _codecs[_currentSendCodecIdx]->REDPayloadISAC(isacRate, isacBwEstimate,
+ payload, payloadLenByte);
+
+ return status;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h
new file mode 100644
index 0000000..422b22e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -0,0 +1,384 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
+
+#include "acm_codec_database.h"
+#include "acm_neteq.h"
+#include "acm_resampler.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+
+namespace webrtc {
+
+class ACMDTMFDetection;
+class ACMGenericCodec;
+class CriticalSectionWrapper;
+class RWLockWrapper;
+
+//#define TIMED_LOGGING
+
+#ifdef TIMED_LOGGING
+ #include "../test/timedtrace.h"
+#endif
+
+#ifdef ACM_QA_TEST
+# include <stdio.h>
+#endif
+
+class AudioCodingModuleImpl : public AudioCodingModule
+{
+public:
+ // constructor
+ AudioCodingModuleImpl(
+ const WebRtc_Word32 id);
+
+ // destructor
+ ~AudioCodingModuleImpl();
+
+ // get version information for ACM and all components
+ WebRtc_Word32 Version(
+ char* version,
+ WebRtc_UWord32& remainingBufferInBytes,
+ WebRtc_UWord32& position) const;
+
+ // change the unique identifier of this object
+ virtual WebRtc_Word32 ChangeUniqueId(
+ const WebRtc_Word32 id);
+
+ // returns the number of milliseconds until the module want
+ // a worker thread to call Process
+ WebRtc_Word32 TimeUntilNextProcess();
+
+ // Process any pending tasks such as timeouts
+ WebRtc_Word32 Process();
+
+ // used in conference to go to and from active encoding, hence
+ // in and out of mix
+ WebRtc_Word32 SetMode(
+ const bool passive);
+
+
+
+ /////////////////////////////////////////
+ // Sender
+ //
+
+ // initialize send codec
+ WebRtc_Word32 InitializeSender();
+
+ // reset send codec
+ WebRtc_Word32 ResetEncoder();
+
+ // can be called multiple times for Codec, CNG, RED
+ WebRtc_Word32 RegisterSendCodec(
+ const CodecInst& sendCodec);
+
+ // get current send codec
+ WebRtc_Word32 SendCodec(
+ CodecInst& currentSendCodec) const;
+
+ // get current send freq
+ WebRtc_Word32 SendFrequency() const;
+
+ // Get encode bitrate
+ // Adaptive rate codecs return their current encode target rate, while other codecs
+ // return there longterm avarage or their fixed rate.
+ WebRtc_Word32 SendBitrate() const;
+
+ // set available bandwidth, inform the encoder about the
+ // estimated bandwidth received from the remote party
+ virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
+ const WebRtc_Word32 bw);
+
+ // register a transport callback which will be
+ // called to deliver the encoded buffers
+ WebRtc_Word32 RegisterTransportCallback(
+ AudioPacketizationCallback* transport);
+
+ // Used by the module to deliver messages to the codec module/application
+ // AVT(DTMF)
+ WebRtc_Word32 RegisterIncomingMessagesCallback(
+ AudioCodingFeedback* incomingMessagesCallback,
+ const ACMCountries cpt);
+
+ // Add 10MS of raw (PCM) audio data to the encoder
+ WebRtc_Word32 Add10MsData(
+ const AudioFrame& audioFrame);
+
+ // set background noise mode for NetEQ, on, off or fade
+ WebRtc_Word32 SetBackgroundNoiseMode(
+ const ACMBackgroundNoiseMode mode);
+
+ // get current background noise mode
+ WebRtc_Word32 BackgroundNoiseMode(
+ ACMBackgroundNoiseMode& mode);
+
+ /////////////////////////////////////////
+ // (FEC) Forward Error Correction
+ //
+
+ // configure FEC status i.e on/off
+ WebRtc_Word32 SetFECStatus(
+ const bool enable);
+
+ // Get FEC status
+ bool FECStatus() const;
+
+ /////////////////////////////////////////
+ // (VAD) Voice Activity Detection
+ // and
+ // (CNG) Comfort Noise Generation
+ //
+
+ WebRtc_Word32 SetVAD(
+ const bool enableDTX = true,
+ const bool enableVAD = false,
+ const ACMVADMode vadMode = VADNormal);
+
+ WebRtc_Word32 VAD(
+ bool& dtxEnabled,
+ bool& vadEnabled,
+ ACMVADMode& vadMode) const;
+
+ WebRtc_Word32 RegisterVADCallback(
+ ACMVADCallback* vadCallback);
+
+ // Get VAD aggressiveness on the incoming stream
+ ACMVADMode ReceiveVADMode() const;
+
+ // Configure VAD aggressiveness on the incoming stream
+ WebRtc_Word16 SetReceiveVADMode(
+ const ACMVADMode mode);
+
+
+ /////////////////////////////////////////
+ // Receiver
+ //
+
+ // initialize receiver, resets codec database etc
+ WebRtc_Word32 InitializeReceiver();
+
+ // reset the decoder state
+ WebRtc_Word32 ResetDecoder();
+
+ // get current receive freq
+ WebRtc_Word32 ReceiveFrequency() const;
+
+ // get current playout freq
+ WebRtc_Word32 PlayoutFrequency() const;
+
+ // register possible reveive codecs, can be called multiple times,
+ // for codecs, CNG, DTMF, RED
+ WebRtc_Word32 RegisterReceiveCodec(
+ const CodecInst& receiveCodec);
+
+ // get current received codec
+ WebRtc_Word32 ReceiveCodec(
+ CodecInst& currentReceiveCodec) const;
+
+ // incoming packet from network parsed and ready for decode
+ WebRtc_Word32 IncomingPacket(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo);
+
+ // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
+ // One usage for this API is when pre-encoded files are pushed in ACM.
+ WebRtc_Word32 IncomingPayload(
+ const WebRtc_Word8* incomingPayload,
+ const WebRtc_Word32 payloadLength,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timestamp = 0);
+
+ // Minimum playout dealy (Used for lip-sync)
+ WebRtc_Word32 SetMinimumPlayoutDelay(
+ const WebRtc_Word32 timeMs);
+
+ // configure Dtmf playout status i.e on/off playout the incoming outband Dtmf tone
+ WebRtc_Word32 SetDtmfPlayoutStatus(
+ const bool enable);
+
+ // Get Dtmf playout status
+ bool DtmfPlayoutStatus() const;
+
+ // Estimate the Bandwidth based on the incoming stream
+ // This is also done in the RTP module
+ // need this for one way audio where the RTCP send the BW estimate
+ WebRtc_Word32 DecoderEstimatedBandwidth() const;
+
+ // Set playout mode voice, fax
+ WebRtc_Word32 SetPlayoutMode(
+ const AudioPlayoutMode mode);
+
+ // Get playout mode voice, fax
+ AudioPlayoutMode PlayoutMode() const;
+
+ // Get playout timestamp
+ WebRtc_Word32 PlayoutTimestamp(
+ WebRtc_UWord32& timestamp);
+
+ // Get 10 milliseconds of raw audio data to play out
+ // automatic resample to the requested frequency if > 0
+ WebRtc_Word32 PlayoutData10Ms(
+ const WebRtc_Word32 desiredFreqHz,
+ AudioFrame &audioFrame);
+
+
+ /////////////////////////////////////////
+ // Statistics
+ //
+
+ WebRtc_Word32 NetworkStatistics(
+ ACMNetworkStatistics& statistics) const;
+
+ void DestructEncoderInst(void* ptrInst);
+
+ WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audioBuff);
+
+ // GET RED payload for iSAC. The method id called
+ // when 'this' ACM is default ACM.
+ WebRtc_Word32 REDPayloadISAC(
+ const WebRtc_Word32 isacRate,
+ const WebRtc_Word16 isacBwEstimate,
+ WebRtc_UWord8* payload,
+ WebRtc_Word16* payloadLenByte);
+
+ WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
+
+ WebRtc_UWord32 EarliestTimestamp() const;
+
+ WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
+
+ WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
+ const bool useWebRtcDTX);
+
+ WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
+ bool& usesWebRtcDTX);
+
+ WebRtc_Word32 SetISACMaxRate(
+ const WebRtc_UWord32 rateBitPerSec);
+
+ WebRtc_Word32 SetISACMaxPayloadSize(
+ const WebRtc_UWord16 payloadLenBytes);
+
+ WebRtc_Word32 ConfigISACBandwidthEstimator(
+ const WebRtc_UWord8 initFrameSizeMsec,
+ const WebRtc_UWord16 initRateBitPerSec,
+ const bool enforceFrameSize = false);
+
+ WebRtc_Word32 UnregisterReceiveCodec(
+ const WebRtc_Word16 payloadType);
+
+protected:
+ void UnregisterSendCodec();
+
+ WebRtc_Word32 UnregisterReceiveCodecSafe(
+ const WebRtc_Word16 codecID);
+
+ ACMGenericCodec* CreateCodec(
+ const CodecInst& codec);
+
+ WebRtc_Word16 DecoderParamByPlType(
+ const WebRtc_UWord8 payloadType,
+ WebRtcACMCodecParams& codecParams) const;
+
+ WebRtc_Word16 DecoderListIDByPlName(
+ const char* payloadName,
+ const WebRtc_UWord16 sampFreqHz = 0) const;
+
+ WebRtc_Word32 InitializeReceiverSafe();
+
+ bool HaveValidEncoder(const char* callerName) const;
+
+ WebRtc_Word32 RegisterRecCodecMSSafe(
+ const CodecInst& receiveCodec,
+ WebRtc_Word16 codecId,
+ WebRtc_Word16 mirrorId,
+ ACMNetEQ::JB jitterBuffer);
+
+private:
+ AudioPacketizationCallback* _packetizationCallback;
+ WebRtc_Word32 _id;
+ WebRtc_UWord32 _lastTimestamp;
+ WebRtc_UWord32 _lastInTimestamp;
+ CodecInst _sendCodecInst;
+ CodecInst _cngNB;
+ CodecInst _cngWB;
+ CodecInst _cngSWB;
+ CodecInst _RED;
+ CodecInst _DTMF;
+ bool _vadEnabled;
+ bool _dtxEnabled;
+ ACMVADMode _vadMode;
+ ACMGenericCodec* _codecs[ACMCodecDB::kMaxNumCodecs];
+ ACMGenericCodec* _slaveCodecs[ACMCodecDB::kMaxNumCodecs];
+ WebRtc_Word16 _mirrorCodecIdx[ACMCodecDB::kMaxNumCodecs];
+ bool _stereoReceive[ACMCodecDB::kMaxNumCodecs];
+ bool _stereoSend;
+ int _prev_received_channel;
+ int _expected_channels;
+ WebRtc_Word32 _currentSendCodecIdx;
+ bool _sendCodecRegistered;
+ ACMResampler _inputResampler;
+ ACMResampler _outputResampler;
+ ACMNetEQ _netEq;
+ CriticalSectionWrapper* _acmCritSect;
+ ACMVADCallback* _vadCallback;
+ WebRtc_UWord8 _lastRecvAudioCodecPlType;
+
+ // RED/FEC
+ bool _isFirstRED;
+ bool _fecEnabled;
+ WebRtc_UWord8* _redBuffer;
+ RTPFragmentationHeader* _fragmentation;
+ WebRtc_UWord32 _lastFECTimestamp;
+ WebRtc_UWord8 _redPayloadType;
+ // if no RED is registered as receive codec this
+ // will have an invalid value.
+ WebRtc_UWord8 _receiveREDPayloadType;
+
+ // This is to keep track of CN instances where we can send DTMFs
+ WebRtc_UWord8 _previousPayloadType;
+
+ // This keeps track of payload types associated with _codecs[].
+ // We define it as signed variable and initialize with -1 to indicate
+ // unused elements.
+ WebRtc_Word16 _registeredPlTypes[ACMCodecDB::kMaxNumCodecs];
+
+ // Used when payloads are pushed into ACM without any RTP info
+ // One example is when pre-encoded bit-stream is pushed from
+ // a file.
+ WebRtcRTPHeader* _dummyRTPHeader;
+ WebRtc_UWord16 _recvPlFrameSizeSmpls;
+
+ bool _receiverInitialized;
+ ACMDTMFDetection* _dtmfDetector;
+
+ AudioCodingFeedback* _dtmfCallback;
+ WebRtc_Word16 _lastDetectedTone;
+ CriticalSectionWrapper* _callbackCritSect;
+#ifdef TIMED_LOGGING
+ TimedTrace _trace;
+#endif
+
+ AudioFrame _audioFrame;
+
+#ifdef ACM_QA_TEST
+ FILE* _outgoingPL;
+ FILE* _incomingPL;
+#endif
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/trunk/src/modules/audio_coding/main/test/ACMTest.cc b/trunk/src/modules/audio_coding/main/test/ACMTest.cc
new file mode 100644
index 0000000..1bbac0e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/ACMTest.cc
@@ -0,0 +1,16 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ACMTest.h"
+
+ACMTest::~ACMTest()
+{
+}
+
diff --git a/trunk/src/modules/audio_coding/main/test/ACMTest.h b/trunk/src/modules/audio_coding/main/test/ACMTest.h
new file mode 100644
index 0000000..e965671
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/ACMTest.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef ACMTEST_H
+#define ACMTEST_H
+
+class ACMTest
+{
+public:
+ virtual ~ACMTest() =0;
+ virtual void Perform() =0;
+};
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/APITest.cc b/trunk/src/modules/audio_coding/main/test/APITest.cc
new file mode 100644
index 0000000..ed44178
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/APITest.cc
@@ -0,0 +1,1569 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cctype>
+#include <iostream>
+#include <ostream>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "APITest.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "event_wrapper.h"
+#include "gtest/gtest.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "testsupport/fileutils.h"
+#include "trace.h"
+#include "utility.h"
+
+namespace webrtc {
+
+#define TEST_DURATION_SEC 600
+
+#define NUMBER_OF_SENDER_TESTS 6
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+#define CHECK_THREAD_NULLITY(myThread, S) \
+ if(myThread != NULL) \
+ { \
+ unsigned int i; \
+ (myThread)->Start(i); \
+ } \
+ else \
+ { \
+ ADD_FAILURE() << S; \
+ }
+
+
+void
+APITest::Wait(WebRtc_UWord32 waitLengthMs)
+{
+ if(_randomTest)
+ {
+ return;
+ }
+ else
+ {
+ EventWrapper* myEvent = EventWrapper::Create();
+ myEvent->Wait(waitLengthMs);
+ delete myEvent;
+ return;
+ }
+}
+
+
+
+APITest::APITest():
+_acmA(NULL),
+_acmB(NULL),
+_channel_A2B(NULL),
+_channel_B2A(NULL),
+_writeToFile(true),
+_pullEventA(NULL),
+_pushEventA(NULL),
+_processEventA(NULL),
+_apiEventA(NULL),
+_pullEventB(NULL),
+_pushEventB(NULL),
+_processEventB(NULL),
+_apiEventB(NULL),
+_codecCntrA(0),
+_codecCntrB(0),
+_testCntrA(1),
+_testCntrB(1),
+_thereIsEncoderA(false),
+_thereIsEncoderB(false),
+_thereIsDecoderA(false),
+_thereIsDecoderB(false),
+_sendVADA(false),
+_sendDTXA(false),
+_sendVADModeA(VADNormal),
+_sendVADB(false),
+_sendDTXB(false),
+_sendVADModeB(VADNormal),
+_minDelayA(0),
+_minDelayB(0),
+_dotPositionA(0),
+_dotMoveDirectionA(1),
+_dotPositionB(39),
+_dotMoveDirectionB(-1),
+_dtmfCallback(NULL),
+_vadCallbackA(NULL),
+_vadCallbackB(NULL),
+_apiTestRWLock(*RWLockWrapper::CreateRWLock()),
+_randomTest(false),
+_testNumA(0),
+_testNumB(1)
+{
+ int n;
+ for( n = 0; n < 32; n++)
+ {
+ _payloadUsed[n] = false;
+ }
+
+ for(n = 0; n < 3; n++)
+ {
+ _receiveVADActivityA[n] = 0;
+ _receiveVADActivityB[n] = 0;
+ }
+
+ _movingDot[40] = '\0';
+
+ for(int n = 0; n <40; n++)
+ {
+ _movingDot[n] = ' ';
+ }
+}
+
+APITest::~APITest()
+{
+ DESTROY_ACM(_acmA);
+ DESTROY_ACM(_acmB);
+
+ DELETE_POINTER(_channel_A2B);
+ DELETE_POINTER(_channel_B2A);
+
+ DELETE_POINTER(_pushEventA);
+ DELETE_POINTER(_pullEventA);
+ DELETE_POINTER(_processEventA);
+ DELETE_POINTER(_apiEventA);
+
+ DELETE_POINTER(_pushEventB);
+ DELETE_POINTER(_pullEventB);
+ DELETE_POINTER(_processEventB);
+ DELETE_POINTER(_apiEventB);
+
+ _inFileA.Close();
+ _outFileA.Close();
+
+ _inFileB.Close();
+ _outFileB.Close();
+
+ DELETE_POINTER(_dtmfCallback);
+ DELETE_POINTER(_vadCallbackA);
+ DELETE_POINTER(_vadCallbackB);
+
+ delete &_apiTestRWLock;
+}
+
+
+
+//WebRtc_Word16
+//APITest::SetInFile(char* fileName, WebRtc_UWord16 frequencyHz)
+//{
+// return _inFile.Open(fileName, frequencyHz, "rb");
+//}
+//
+//WebRtc_Word16
+//APITest::SetOutFile(char* fileName, WebRtc_UWord16 frequencyHz)
+//{
+// return _outFile.Open(fileName, frequencyHz, "wb");
+//}
+
+WebRtc_Word16
+APITest::SetUp()
+{
+ _acmA = AudioCodingModule::Create(1);
+ _acmB = AudioCodingModule::Create(2);
+
+ CodecInst dummyCodec;
+ int lastPayloadType = 0;
+
+ WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
+ for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
+ {
+ AudioCodingModule::Codec(n, dummyCodec);
+ if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
+ (dummyCodec.plfreq == 32000))
+ {
+ continue;
+ }
+
+ printf("Register Receive Codec %s ", dummyCodec.plname);
+
+ if((n != 0) && !FixedPayloadTypeCodec(dummyCodec.plname))
+ {
+ // Check registration with an already occupied payload type
+ int currentPayloadType = dummyCodec.pltype;
+ dummyCodec.pltype = 97; //lastPayloadType;
+ CHECK_ERROR(_acmB->RegisterReceiveCodec(dummyCodec));
+ dummyCodec.pltype = currentPayloadType;
+ }
+
+ if((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname))
+ {
+ // test if re-registration works;
+ CodecInst nextCodec;
+ int currentPayloadType = dummyCodec.pltype;
+ AudioCodingModule::Codec(n + 1, nextCodec);
+ dummyCodec.pltype = nextCodec.pltype;
+ if(!FixedPayloadTypeCodec(nextCodec.plname))
+ {
+ _acmB->RegisterReceiveCodec(dummyCodec);
+ }
+ dummyCodec.pltype = currentPayloadType;
+ }
+
+ if((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname))
+ {
+ // test if un-registration works;
+ CodecInst nextCodec;
+ AudioCodingModule::Codec(n + 1, nextCodec);
+ nextCodec.pltype = dummyCodec.pltype;
+ if(!FixedPayloadTypeCodec(nextCodec.plname))
+ {
+ CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(nextCodec));
+ CHECK_ERROR_MT(_acmA->UnregisterReceiveCodec(nextCodec.pltype));
+ }
+ }
+
+
+ CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(dummyCodec));
+ printf(" side A done!");
+ CHECK_ERROR_MT(_acmB->RegisterReceiveCodec(dummyCodec));
+ printf(" side B done!\n");
+
+ if(!strcmp(dummyCodec.plname, "CN"))
+ {
+ CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
+ CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
+ }
+ lastPayloadType = dummyCodec.pltype;
+ if((lastPayloadType >= 96) && (lastPayloadType <= 127))
+ {
+ _payloadUsed[lastPayloadType - 96] = true;
+ }
+ }
+ _thereIsDecoderA = true;
+ _thereIsDecoderB = true;
+
+ // Register Send Codec
+ AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, dummyCodec);
+ CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
+ _thereIsEncoderA = true;
+ //
+ AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, dummyCodec);
+ CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
+ _thereIsEncoderB = true;
+
+ char fileName[500];
+ WebRtc_UWord16 frequencyHz;
+
+ printf("\n\nAPI Test\n");
+ printf("========\n");
+ printf("Hit enter to accept the default values indicated in []\n\n");
+
+ //--- Input A
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 32000;
+ printf("Enter input file at side A [%s]: ", fileName);
+ PCMFile::ChooseFile(fileName, 499, &frequencyHz);
+ _inFileA.Open(fileName, frequencyHz, "rb", true);
+
+ //--- Output A
+ std::string outputFileA = webrtc::test::OutputPath() + "outA.pcm";
+ strcpy(fileName, outputFileA.c_str());
+ printf("Enter output file at side A [%s]: ", fileName);
+ PCMFile::ChooseFile(fileName, 499, &frequencyHz);
+ _outFileA.Open(fileName, frequencyHz, "wb");
+
+ //--- Input B
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ printf("\n\nEnter input file at side B [%s]: ", fileName);
+ PCMFile::ChooseFile(fileName, 499, &frequencyHz);
+ _inFileB.Open(fileName, frequencyHz, "rb", true);
+
+ //--- Output B
+ std::string outputFileB = webrtc::test::OutputPath() + "outB.pcm";
+ strcpy(fileName, outputFileB.c_str());
+ printf("Enter output file at side B [%s]: ", fileName);
+ PCMFile::ChooseFile(fileName, 499, &frequencyHz);
+ _outFileB.Open(fileName, frequencyHz, "wb");
+
+ //--- Set A-to-B channel
+ _channel_A2B = new Channel(2);
+ CHECK_ERROR_MT(_acmA->RegisterTransportCallback(_channel_A2B));
+ _channel_A2B->RegisterReceiverACM(_acmB);
+
+ //--- Set B-to-A channel
+ _channel_B2A = new Channel(1);
+ CHECK_ERROR_MT(_acmB->RegisterTransportCallback(_channel_B2A));
+ _channel_B2A->RegisterReceiverACM(_acmA);
+
+ //--- EVENT TIMERS
+ // A
+ _pullEventA = EventWrapper::Create();
+ _pushEventA = EventWrapper::Create();
+ _processEventA = EventWrapper::Create();
+ _apiEventA = EventWrapper::Create();
+ // B
+ _pullEventB = EventWrapper::Create();
+ _pushEventB = EventWrapper::Create();
+ _processEventB = EventWrapper::Create();
+ _apiEventB = EventWrapper::Create();
+
+ //--- I/O params
+ // A
+ _outFreqHzA = _outFileA.SamplingFrequency();
+ // B
+ _outFreqHzB = _outFileB.SamplingFrequency();
+
+
+ //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt");
+
+ char print[11];
+
+ printf("\nRandom Test (y/n)?");
+ EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
+ print[10] = '\0';
+ if(strstr(print, "y") != NULL)
+ {
+ _randomTest = true;
+ _verbose = false;
+ _writeToFile = false;
+ Trace::CreateTrace();
+ Trace::SetTraceFile("ACMAPITest.txt");
+ //freopen("APITest_log.txt", "w", stdout);
+ }
+ else
+ {
+ Trace::CreateTrace();
+ Trace::SetTraceFile("ACMAPITest.txt", true);
+ _randomTest = false;
+ printf("\nPrint Tests (y/n)? ");
+ EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
+ print[10] = '\0';
+ if(strstr(print, "y") == NULL)
+ {
+ EXPECT_TRUE(freopen("APITest_log.txt", "w", stdout) != 0);
+ _verbose = false;
+ }
+ }
+
+#ifdef WEBRTC_DTMF_DETECTION
+ _dtmfCallback = new DTMFDetector;
+#endif
+ _vadCallbackA = new VADCallback;
+ _vadCallbackB = new VADCallback;
+
+ return 0;
+}
+
+bool
+APITest::PushAudioThreadA(void* obj)
+{
+ return static_cast<APITest*>(obj)->PushAudioRunA();
+}
+
+bool
+APITest::PushAudioThreadB(void* obj)
+{
+ return static_cast<APITest*>(obj)->PushAudioRunB();
+}
+
+bool
+APITest::PullAudioThreadA(void* obj)
+{
+ return static_cast<APITest*>(obj)->PullAudioRunA();
+}
+
+bool
+APITest::PullAudioThreadB(void* obj)
+{
+ return static_cast<APITest*>(obj)->PullAudioRunB();
+}
+
+bool
+APITest::ProcessThreadA(void* obj)
+{
+ return static_cast<APITest*>(obj)->ProcessRunA();
+}
+
+bool
+APITest::ProcessThreadB(void* obj)
+{
+ return static_cast<APITest*>(obj)->ProcessRunB();
+}
+
+bool
+APITest::APIThreadA(void* obj)
+{
+ return static_cast<APITest*>(obj)->APIRunA();
+}
+
+bool
+APITest::APIThreadB(void* obj)
+{
+ return static_cast<APITest*>(obj)->APIRunB();
+}
+
+bool
+APITest::PullAudioRunA()
+{
+ _pullEventA->Wait(100);
+ AudioFrame audioFrame;
+ if(_acmA->PlayoutData10Ms(_outFreqHzA, audioFrame) < 0)
+ {
+ bool thereIsDecoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsDecoder = _thereIsDecoderA;
+ }
+ if(thereIsDecoder)
+ {
+ fprintf(stderr, "\n>>>>>> cannot pull audio A <<<<<<<< \n");
+ }
+ }
+ else
+ {
+ if(_writeToFile)
+ {
+ _outFileA.Write10MsData(audioFrame);
+ }
+ _receiveVADActivityA[(int)audioFrame._vadActivity]++;
+ }
+ return true;
+}
+
+bool
+APITest::PullAudioRunB()
+{
+ _pullEventB->Wait(100);
+ AudioFrame audioFrame;
+ if(_acmB->PlayoutData10Ms(_outFreqHzB, audioFrame) < 0)
+ {
+ bool thereIsDecoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsDecoder = _thereIsDecoderB;
+ }
+ if(thereIsDecoder)
+ {
+ fprintf(stderr, "\n>>>>>> cannot pull audio B <<<<<<<< \n");
+ fprintf(stderr, "%d %d\n", _testNumA, _testNumB);
+ }
+ }
+ else
+ {
+ if(_writeToFile)
+ {
+ _outFileB.Write10MsData(audioFrame);
+ }
+ _receiveVADActivityB[(int)audioFrame._vadActivity]++;
+ }
+ return true;
+}
+
+bool
+APITest::PushAudioRunA()
+{
+ _pushEventA->Wait(100);
+ AudioFrame audioFrame;
+ _inFileA.Read10MsData(audioFrame);
+ if(_acmA->Add10MsData(audioFrame) < 0)
+ {
+ bool thereIsEncoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsEncoder = _thereIsEncoderA;
+ }
+ if(thereIsEncoder)
+ {
+ fprintf(stderr, "\n>>>> add10MsData at A failed <<<<\n");
+ }
+ }
+ return true;
+}
+
+bool
+APITest::PushAudioRunB()
+{
+ _pushEventB->Wait(100);
+ AudioFrame audioFrame;
+ _inFileB.Read10MsData(audioFrame);
+ if(_acmB->Add10MsData(audioFrame) < 0)
+ {
+ bool thereIsEncoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsEncoder = _thereIsEncoderB;
+ }
+
+ if(thereIsEncoder)
+ {
+ fprintf(stderr, "\n>>>> cannot add audio to B <<<<");
+ }
+ }
+
+ return true;
+}
+
+bool
+APITest::ProcessRunA()
+{
+ _processEventA->Wait(100);
+ if(_acmA->Process() < 0)
+ {
+ // do not print error message if there is no encoder
+ bool thereIsEncoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsEncoder = _thereIsEncoderA;
+ }
+
+ if(thereIsEncoder)
+ {
+ fprintf(stderr, "\n>>>>> Process Failed at A <<<<<\n");
+ }
+ }
+ return true;
+}
+
+bool
+APITest::ProcessRunB()
+{
+ _processEventB->Wait(100);
+ if(_acmB->Process() < 0)
+ {
+ bool thereIsEncoder;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ thereIsEncoder = _thereIsEncoderB;
+ }
+ if(thereIsEncoder)
+ {
+ fprintf(stderr, "\n>>>>> Process Failed at B <<<<<\n");
+ }
+ }
+ return true;
+}
+
+/*/
+ *
+ * In side A we test the APIs which are related to sender Side.
+ *
+/*/
+
+
+void
+APITest::RunTest(char thread)
+{
+ int testNum;
+ {
+ WriteLockScoped cs(_apiTestRWLock);
+ if(thread == 'A')
+ {
+ _testNumA = (_testNumB + 1 + (rand() % 6)) % 7;
+ testNum = _testNumA;
+
+ _movingDot[_dotPositionA] = ' ';
+ if(_dotPositionA == 0)
+ {
+ _dotMoveDirectionA = 1;
+ }
+ if(_dotPositionA == 19)
+ {
+ _dotMoveDirectionA = -1;
+ }
+ _dotPositionA += _dotMoveDirectionA;
+ _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0)? '>':'<';
+ }
+ else
+ {
+ _testNumB = (_testNumA + 1 + (rand() % 6)) % 7;
+ testNum = _testNumB;
+
+ _movingDot[_dotPositionB] = ' ';
+ if(_dotPositionB == 20)
+ {
+ _dotMoveDirectionB = 1;
+ }
+ if(_dotPositionB == 39)
+ {
+ _dotMoveDirectionB = -1;
+ }
+ _dotPositionB += _dotMoveDirectionB;
+ _movingDot[_dotPositionB] = (_dotMoveDirectionB > 0)? '>':'<';
+ }
+ //fprintf(stderr, "%c: %d \n", thread, testNum);
+ //fflush(stderr);
+ }
+ switch(testNum)
+ {
+ case 0:
+ CurrentCodec('A');
+ ChangeCodec('A');
+ break;
+ case 1:
+ TestPlayout('B');
+ break;
+ case 2:
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\nTesting Delay ...\n");
+ }
+ TestDelay('A');
+ break;
+ case 3:
+ TestSendVAD('A');
+ break;
+ case 4:
+ TestRegisteration('A');
+ break;
+ case 5:
+ TestReceiverVAD('A');
+ break;
+ case 6:
+#ifdef WEBRTC_DTMF_DETECTION
+ LookForDTMF('A');
+#endif
+ break;
+ default:
+ fprintf(stderr, "Wrong Test Number\n");
+ getchar();
+ exit(1);
+ }
+}
+
+
+
+bool
+APITest::APIRunA()
+{
+ _apiEventA->Wait(50);
+
+ bool randomTest;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ randomTest = _randomTest;
+ }
+ if(randomTest)
+ {
+ RunTest('A');
+ }
+ else
+ {
+ CurrentCodec('A');
+ ChangeCodec('A');
+ TestPlayout('B');
+ if(_codecCntrA == 0)
+ {
+ fprintf(stdout, "\nTesting Delay ...\n");
+ TestDelay('A');
+ }
+ // VAD TEST
+ TestSendVAD('A');
+ TestRegisteration('A');
+ TestReceiverVAD('A');
+#ifdef WEBRTC_DTMF_DETECTION
+ LookForDTMF('A');
+#endif
+ }
+ return true;
+}
+
+bool
+APITest::APIRunB()
+{
+ _apiEventB->Wait(50);
+ bool randomTest;
+ {
+ ReadLockScoped rl(_apiTestRWLock);
+ randomTest = _randomTest;
+ }
+ //_apiEventB->Wait(2000);
+ if(randomTest)
+ {
+ RunTest('B');
+ }
+
+ return true;
+}
+
+void
+APITest::Perform()
+{
+ SetUp();
+
+ //--- THREADS
+ // A
+ // PUSH
+ ThreadWrapper* myPushAudioThreadA = ThreadWrapper::CreateThread(PushAudioThreadA,
+ this, kNormalPriority, "PushAudioThreadA");
+ CHECK_THREAD_NULLITY(myPushAudioThreadA, "Unable to start A::PUSH thread");
+ // PULL
+ ThreadWrapper* myPullAudioThreadA = ThreadWrapper::CreateThread(PullAudioThreadA,
+ this, kNormalPriority, "PullAudioThreadA");
+ CHECK_THREAD_NULLITY(myPullAudioThreadA, "Unable to start A::PULL thread");
+ // Process
+ ThreadWrapper* myProcessThreadA = ThreadWrapper::CreateThread(ProcessThreadA,
+ this, kNormalPriority, "ProcessThreadA");
+ CHECK_THREAD_NULLITY(myProcessThreadA, "Unable to start A::Process thread");
+ // API
+ ThreadWrapper* myAPIThreadA = ThreadWrapper::CreateThread(APIThreadA,
+ this, kNormalPriority, "APIThreadA");
+ CHECK_THREAD_NULLITY(myAPIThreadA, "Unable to start A::API thread");
+ // B
+ // PUSH
+ ThreadWrapper* myPushAudioThreadB = ThreadWrapper::CreateThread(PushAudioThreadB,
+ this, kNormalPriority, "PushAudioThreadB");
+ CHECK_THREAD_NULLITY(myPushAudioThreadB, "Unable to start B::PUSH thread");
+ // PULL
+ ThreadWrapper* myPullAudioThreadB = ThreadWrapper::CreateThread(PullAudioThreadB,
+ this, kNormalPriority, "PullAudioThreadB");
+ CHECK_THREAD_NULLITY(myPullAudioThreadB, "Unable to start B::PULL thread");
+ // Process
+ ThreadWrapper* myProcessThreadB = ThreadWrapper::CreateThread(ProcessThreadB,
+ this, kNormalPriority, "ProcessThreadB");
+ CHECK_THREAD_NULLITY(myProcessThreadB, "Unable to start B::Process thread");
+ // API
+ ThreadWrapper* myAPIThreadB = ThreadWrapper::CreateThread(APIThreadB,
+ this, kNormalPriority, "APIThreadB");
+ CHECK_THREAD_NULLITY(myAPIThreadB, "Unable to start B::API thread");
+
+
+ //_apiEventA->StartTimer(true, 5000);
+ //_apiEventB->StartTimer(true, 5000);
+
+ _processEventA->StartTimer(true, 10);
+ _processEventB->StartTimer(true, 10);
+
+ _pullEventA->StartTimer(true, 10);
+ _pullEventB->StartTimer(true, 10);
+
+ _pushEventA->StartTimer(true, 10);
+ _pushEventB->StartTimer(true, 10);
+
+ // Keep main thread waiting for sender/receiver
+ // threads to complete
+ EventWrapper* completeEvent = EventWrapper::Create();
+ WebRtc_UWord64 startTime = TickTime::MillisecondTimestamp();
+ WebRtc_UWord64 currentTime;
+ do
+ {
+ {
+ //ReadLockScoped rl(_apiTestRWLock);
+ //fprintf(stderr, "\r%s", _movingDot);
+ }
+ //fflush(stderr);
+ completeEvent->Wait(50);
+ currentTime = TickTime::MillisecondTimestamp();
+ } while((currentTime - startTime) < 120000); // Run test in 2 minutes (120000 ms)
+
+ //completeEvent->Wait(0xFFFFFFFF);//(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
+ delete completeEvent;
+
+ myPushAudioThreadA->Stop();
+ myPullAudioThreadA->Stop();
+ myProcessThreadA->Stop();
+ myAPIThreadA->Stop();
+
+ delete myPushAudioThreadA;
+ delete myPullAudioThreadA;
+ delete myProcessThreadA;
+ delete myAPIThreadA;
+
+
+ myPushAudioThreadB->Stop();
+ myPullAudioThreadB->Stop();
+ myProcessThreadB->Stop();
+ myAPIThreadB->Stop();
+
+ delete myPushAudioThreadB;
+ delete myPullAudioThreadB;
+ delete myProcessThreadB;
+ delete myAPIThreadB;
+}
+
+
+void
+APITest::CheckVADStatus(char side)
+{
+
+ bool dtxEnabled;
+ bool vadEnabled;
+ ACMVADMode vadMode;
+
+ if(side == 'A')
+ {
+ _acmA->VAD(dtxEnabled, vadEnabled, vadMode);
+ _acmA->RegisterVADCallback(NULL);
+ _vadCallbackA->Reset();
+ _acmA->RegisterVADCallback(_vadCallbackA);
+
+ if(!_randomTest)
+ {
+ if(_verbose)
+ {
+ fprintf(stdout, "DTX %3s, VAD %3s, Mode %d",
+ dtxEnabled? "ON":"OFF",
+ vadEnabled? "ON":"OFF",
+ (int)vadMode);
+ Wait(5000);
+ fprintf(stdout, " => bit-rate %3.0f kbps\n",
+ _channel_A2B->BitRate());
+ }
+ else
+ {
+ Wait(5000);
+ fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
+ dtxEnabled? "ON":"OFF",
+ vadEnabled? "ON":"OFF",
+ (int)vadMode,
+ _channel_A2B->BitRate());
+ }
+ _vadCallbackA->PrintFrameTypes();
+ }
+
+ if(dtxEnabled != _sendDTXA)
+ {
+ fprintf(stderr, ">>> Error Enabling DTX <<<\n");
+ }
+ if((vadEnabled != _sendVADA) && (!dtxEnabled))
+ {
+ fprintf(stderr, ">>> Error Enabling VAD <<<\n");
+ }
+ if((vadMode != _sendVADModeA) && vadEnabled)
+ {
+ fprintf(stderr, ">>> Error setting VAD-mode <<<\n");
+ }
+ }
+ else
+ {
+ _acmB->VAD(dtxEnabled, vadEnabled, vadMode);
+
+ _acmB->RegisterVADCallback(NULL);
+ _vadCallbackB->Reset();
+ _acmB->RegisterVADCallback(_vadCallbackB);
+
+ if(!_randomTest)
+ {
+ if(_verbose)
+ {
+ fprintf(stdout, "DTX %3s, VAD %3s, Mode %d",
+ dtxEnabled? "ON":"OFF",
+ vadEnabled? "ON":"OFF",
+ (int)vadMode);
+ Wait(5000);
+ fprintf(stdout, " => bit-rate %3.0f kbps\n",
+ _channel_B2A->BitRate());
+ }
+ else
+ {
+ Wait(5000);
+ fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
+ dtxEnabled? "ON":"OFF",
+ vadEnabled? "ON":"OFF",
+ (int)vadMode,
+ _channel_B2A->BitRate());
+ }
+ _vadCallbackB->PrintFrameTypes();
+ }
+
+ if(dtxEnabled != _sendDTXB)
+ {
+ fprintf(stderr, ">>> Error Enabling DTX <<<\n");
+ }
+ if((vadEnabled != _sendVADB) && (!dtxEnabled))
+ {
+ fprintf(stderr, ">>> Error Enabling VAD <<<\n");
+ }
+ if((vadMode != _sendVADModeB) && vadEnabled)
+ {
+ fprintf(stderr, ">>> Error setting VAD-mode <<<\n");
+ }
+ }
+}
+
+// Set Min delay, get delay, playout timestamp
+void
+APITest::TestDelay(char side)
+{
+ AudioCodingModule* myACM;
+ Channel* myChannel;
+ WebRtc_Word32* myMinDelay;
+ EventWrapper* myEvent = EventWrapper::Create();
+
+ WebRtc_UWord32 inTimestamp = 0;
+ WebRtc_UWord32 outTimestamp = 0;
+ double estimDelay = 0;
+
+ double averageEstimDelay = 0;
+ double averageDelay = 0;
+
+ CircularBuffer estimDelayCB(100);
+ estimDelayCB.SetArithMean(true);
+
+ if(side == 'A')
+ {
+ myACM = _acmA;
+ myChannel = _channel_B2A;
+ myMinDelay = &_minDelayA;
+ }
+ else
+ {
+ myACM = _acmB;
+ myChannel = _channel_A2B;
+ myMinDelay = &_minDelayB;
+ }
+
+
+ CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
+
+
+ inTimestamp = myChannel->LastInTimestamp();
+ CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
+
+ if(!_randomTest)
+ {
+ myEvent->StartTimer(true, 30);
+ int n = 0;
+ int settlePoint = 5000;
+ while(n < settlePoint + 400)
+ {
+ myEvent->Wait(1000);
+
+ inTimestamp = myChannel->LastInTimestamp();
+ CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
+
+ //std::cout << outTimestamp << std::endl << std::flush;
+ estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
+ ((double)myACM->ReceiveFrequency() / 1000.0);
+
+ estimDelayCB.Update(estimDelay);
+
+ estimDelayCB.ArithMean(averageEstimDelay);
+ //printf("\n %6.1f \n", estimDelay);
+ //std::cout << " " << std::flush;
+
+ if(_verbose)
+ {
+ fprintf(stdout, "\rExpected: %4d, retreived: %6.1f, measured: %6.1f",
+ *myMinDelay, averageDelay, averageEstimDelay);
+ std::cout << " " << std::flush;
+ }
+ if((averageDelay > *myMinDelay) && (n < settlePoint))
+ {
+ settlePoint = n;
+ }
+ n++;
+ }
+ myEvent->StopTimer();
+ }
+
+ if((!_verbose) && (!_randomTest))
+ {
+ fprintf(stdout, "\nExpected: %4d, retreived: %6.1f, measured: %6.1f",
+ *myMinDelay, averageDelay, averageEstimDelay);
+ }
+
+ *myMinDelay = (rand() % 1000) + 1;
+
+ ACMNetworkStatistics networkStat;
+ CHECK_ERROR_MT(myACM->NetworkStatistics(networkStat));
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\nJitter Statistics at Side %c\n", side);
+ fprintf(stdout, "--------------------------------------\n");
+ fprintf(stdout, "buffer-size............. %d\n", networkStat.currentBufferSize);
+ fprintf(stdout, "Preferred buffer-size... %d\n", networkStat.preferredBufferSize);
+ fprintf(stdout, "Peaky jitter mode........%d\n", networkStat.jitterPeaksFound);
+ fprintf(stdout, "packet-size rate........ %d\n", networkStat.currentPacketLossRate);
+ fprintf(stdout, "discard rate............ %d\n", networkStat.currentDiscardRate);
+ fprintf(stdout, "expand rate............. %d\n", networkStat.currentExpandRate);
+ fprintf(stdout, "Preemptive rate......... %d\n", networkStat.currentPreemptiveRate);
+ fprintf(stdout, "Accelerate rate......... %d\n", networkStat.currentAccelerateRate);
+ fprintf(stdout, "Clock-drift............. %d\n", networkStat.clockDriftPPM);
+ fprintf(stdout, "Mean waiting time....... %d\n", networkStat.meanWaitingTimeMs);
+ fprintf(stdout, "Median waiting time..... %d\n", networkStat.medianWaitingTimeMs);
+ fprintf(stdout, "Min waiting time........ %d\n", networkStat.minWaitingTimeMs);
+ fprintf(stdout, "Max waiting time........ %d\n", networkStat.maxWaitingTimeMs);
+ }
+
+ CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
+
+ if(!_randomTest)
+ {
+ myEvent->Wait(500);
+ fprintf(stdout, "\n");
+ fprintf(stdout, "\n");
+ }
+ delete myEvent;
+}
+
+// Unregister a codec & register again.
+void
+APITest::TestRegisteration(char sendSide)
+{
+ AudioCodingModule* sendACM;
+ AudioCodingModule* receiveACM;
+ bool* thereIsDecoder;
+ EventWrapper* myEvent = EventWrapper::Create();
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\n");
+ fprintf(stdout, "---------------------------------------------------------\n");
+ fprintf(stdout, " Unregister/register Receive Codec\n");
+ fprintf(stdout, "---------------------------------------------------------\n");
+ }
+
+ switch(sendSide)
+ {
+ case 'A':
+ {
+ sendACM = _acmA;
+ receiveACM = _acmB;
+ thereIsDecoder = &_thereIsDecoderB;
+ break;
+ }
+ case 'B':
+ {
+ sendACM = _acmB;
+ receiveACM = _acmA;
+ thereIsDecoder = &_thereIsDecoderA;
+ break;
+ }
+ default:
+ fprintf(stderr, "Invalid sender-side in TestRegistration(%c)\n", sendSide);
+ exit(-1);
+ }
+
+ CodecInst myCodec;
+ if(sendACM->SendCodec(myCodec) < 0)
+ {
+ AudioCodingModule::Codec(_codecCntrA, myCodec);
+ }
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
+ fflush(stdout);
+ }
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsDecoder = false;
+ }
+ //myEvent->Wait(20);
+ CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+ Wait(1000);
+
+ int currentPayload = myCodec.pltype;
+
+ if(!FixedPayloadTypeCodec(myCodec.plname))
+ {
+ WebRtc_Word32 i;
+ for(i = 0; i < 32; i++)
+ {
+ if(!_payloadUsed[i])
+ {
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Register receive codec with new Payload, AUDIO BACK.\n");
+ }
+ //myCodec.pltype = i + 96;
+ //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+ //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+ //myEvent->Wait(20);
+ //{
+ // WriteLockScoped wl(_apiTestRWLock);
+ // *thereIsDecoder = true;
+ //}
+ Wait(1000);
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
+ }
+ //{
+ // WriteLockScoped wl(_apiTestRWLock);
+ // *thereIsDecoder = false;
+ //}
+ //myEvent->Wait(20);
+ //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+ Wait(1000);
+
+ myCodec.pltype = currentPayload;
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Register receive codec with default Payload, AUDIO BACK.\n");
+ fflush(stdout);
+ }
+ CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+ //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+ myEvent->Wait(20);
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsDecoder = true;
+ }
+ Wait(1000);
+
+ break;
+ }
+ }
+ if(i == 32)
+ {
+ CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsDecoder = true;
+ }
+ }
+ }
+ else
+ {
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Register receive codec with fixed Payload, AUDIO BACK.\n");
+ fflush(stdout);
+ }
+ CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+ //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+ //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+ myEvent->Wait(20);
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsDecoder = true;
+ }
+ }
+ delete myEvent;
+ if(!_randomTest)
+ {
+ fprintf(stdout, "---------------------------------------------------------\n");
+ }
+}
+
+// Playout Mode, background noise mode.
+// Receiver Frequency, playout frequency.
+void
+APITest::TestPlayout(char receiveSide)
+{
+ AudioCodingModule* receiveACM;
+ AudioPlayoutMode* playoutMode = NULL;
+ ACMBackgroundNoiseMode* bgnMode = NULL;
+ switch(receiveSide)
+ {
+ case 'A':
+ {
+ receiveACM = _acmA;
+ playoutMode = &_playoutModeA;
+ bgnMode = &_bgnModeA;
+ break;
+ }
+ case 'B':
+ {
+ receiveACM = _acmB;
+ playoutMode = &_playoutModeB;
+ bgnMode = &_bgnModeB;
+ break;
+ }
+ default:
+ receiveACM = _acmA;
+ }
+
+ WebRtc_Word32 receiveFreqHz = receiveACM->ReceiveFrequency();
+ WebRtc_Word32 playoutFreqHz = receiveACM->PlayoutFrequency();
+
+ CHECK_ERROR_MT(receiveFreqHz);
+ CHECK_ERROR_MT(playoutFreqHz);
+
+ char bgnString[25];
+ switch(*bgnMode)
+ {
+ case On:
+ {
+ *bgnMode = Fade;
+ strncpy(bgnString, "Fade", 25);
+ break;
+ }
+ case Fade:
+ {
+ *bgnMode = Off;
+ strncpy(bgnString, "OFF", 25);
+ break;
+ }
+ case Off:
+ {
+ *bgnMode = On;
+ strncpy(bgnString, "ON", 25);
+ break;
+ }
+ default:
+ *bgnMode = On;
+ strncpy(bgnString, "ON", 25);
+ }
+ CHECK_ERROR_MT(receiveACM->SetBackgroundNoiseMode(*bgnMode));
+ bgnString[24] = '\0';
+
+ char playoutString[25];
+ switch(*playoutMode)
+ {
+ case voice:
+ {
+ *playoutMode = fax;
+ strncpy(playoutString, "FAX", 25);
+ break;
+ }
+ case fax:
+ {
+ *playoutMode = streaming;
+ strncpy(playoutString, "Streaming", 25);
+ break;
+ }
+ case streaming:
+ {
+ *playoutMode = voice;
+ strncpy(playoutString, "Voice", 25);
+ break;
+ }
+ default:
+ *playoutMode = voice;
+ strncpy(playoutString, "Voice", 25);
+ }
+ CHECK_ERROR_MT(receiveACM->SetPlayoutMode(*playoutMode));
+ playoutString[24] = '\0';
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n");
+ fprintf(stdout, "In Side %c\n", receiveSide);
+ fprintf(stdout, "---------------------------------\n");
+ fprintf(stdout, "Receive Frequency....... %d Hz\n", receiveFreqHz);
+ fprintf(stdout, "Playout Frequency....... %d Hz\n", playoutFreqHz);
+ fprintf(stdout, "Audio Playout Mode...... %s\n", playoutString);
+ fprintf(stdout, "Background Noise Mode... %s\n", bgnString);
+ }
+}
+
+// set/get receiver VAD status & mode.
+void
+APITest::TestReceiverVAD(char side)
+{
+ AudioCodingModule* myACM;
+ int* myReceiveVADActivity;
+
+ if(side == 'A')
+ {
+ myACM = _acmA;
+ myReceiveVADActivity = _receiveVADActivityA;
+ }
+ else
+ {
+ myACM = _acmB;
+ myReceiveVADActivity = _receiveVADActivityB;
+ }
+
+ ACMVADMode mode = myACM->ReceiveVADMode();
+
+ CHECK_ERROR_MT(mode);
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\nCurrent Receive VAD at side %c\n", side);
+ fprintf(stdout, "----------------------------------\n");
+ fprintf(stdout, "mode.......... %d\n", (int)mode);
+ fprintf(stdout, "VAD Active.... %d\n", myReceiveVADActivity[0]);
+ fprintf(stdout, "VAD Passive... %d\n", myReceiveVADActivity[1]);
+ fprintf(stdout, "VAD Unknown... %d\n", myReceiveVADActivity[2]);
+ }
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\nChange Receive VAD at side %c\n\n", side);
+ }
+
+ switch(mode)
+ {
+ case VADNormal:
+ mode = VADAggr;
+ break;
+ case VADLowBitrate:
+ mode = VADVeryAggr;
+ break;
+ case VADAggr:
+ mode = VADLowBitrate;
+ break;
+ case VADVeryAggr:
+ mode = VADNormal;
+ break;
+ default:
+ mode = VADNormal;
+
+ CHECK_ERROR_MT(myACM->SetReceiveVADMode(mode));
+ }
+ for(int n = 0; n < 3; n++)
+ {
+ myReceiveVADActivity[n] = 0;
+ }
+}
+
+
+void
+APITest::TestSendVAD(char side)
+{
+ if(_randomTest)
+ {
+ return;
+ }
+
+ bool* vad;
+ bool* dtx;
+ ACMVADMode* mode;
+ Channel* myChannel;
+ AudioCodingModule* myACM;
+
+ CodecInst myCodec;
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\n");
+ fprintf(stdout, "-----------------------------------------------\n");
+ fprintf(stdout, " Test VAD API\n");
+ fprintf(stdout, "-----------------------------------------------\n");
+ }
+
+ if(side == 'A')
+ {
+ AudioCodingModule::Codec(_codecCntrA, myCodec);
+ vad = &_sendVADA;
+ dtx = &_sendDTXA;
+ mode = &_sendVADModeA;
+ myChannel = _channel_A2B;
+ myACM = _acmA;
+ }
+ else
+ {
+ AudioCodingModule::Codec(_codecCntrB, myCodec);
+ vad = &_sendVADB;
+ dtx = &_sendDTXB;
+ mode = &_sendVADModeB;
+ myChannel = _channel_B2A;
+ myACM = _acmB;
+ }
+
+ CheckVADStatus(side);
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\n");
+ }
+
+ switch(*mode)
+ {
+ case VADNormal:
+ *vad = true;
+ *dtx = true;
+ *mode = VADAggr;
+ break;
+ case VADLowBitrate:
+ *vad = true;
+ *dtx = true;
+ *mode = VADVeryAggr;
+ break;
+ case VADAggr:
+ *vad = true;
+ *dtx = true;
+ *mode = VADLowBitrate;
+ break;
+ case VADVeryAggr:
+ *vad = false;
+ *dtx = false;
+ *mode = VADNormal;
+ break;
+ default:
+ *mode = VADNormal;
+ }
+
+ *dtx = (myCodec.plfreq == 32000)? false:*dtx;
+
+ CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
+ myChannel->ResetStats();
+
+ CheckVADStatus(side);
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n");
+ fprintf(stdout, "-----------------------------------------------\n");
+ }
+
+ // Fault Test
+ CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)-1));
+ CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)4));
+
+
+
+}
+
+
+void
+APITest::CurrentCodec(char side)
+{
+ CodecInst myCodec;
+ if(side == 'A')
+ {
+ _acmA->SendCodec(myCodec);
+ }
+ else
+ {
+ _acmB->SendCodec(myCodec);
+ }
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\n");
+ fprintf(stdout, "Send codec in Side A\n");
+ fprintf(stdout, "----------------------------\n");
+ fprintf(stdout, "Name................. %s\n", myCodec.plname);
+ fprintf(stdout, "Sampling Frequency... %d\n", myCodec.plfreq);
+ fprintf(stdout, "Rate................. %d\n", myCodec.rate);
+ fprintf(stdout, "Payload-type......... %d\n", myCodec.pltype);
+ fprintf(stdout, "Packet-size.......... %d\n", myCodec.pacsize);
+ }
+
+ Wait(100);
+}
+
+void
+APITest::ChangeCodec(char side)
+{
+ CodecInst myCodec;
+ AudioCodingModule* myACM;
+ WebRtc_UWord8* codecCntr;
+ bool* thereIsEncoder;
+ bool* vad;
+ bool* dtx;
+ ACMVADMode* mode;
+ Channel* myChannel;
+ // Reset and Wait
+ if(!_randomTest)
+ {
+ fprintf(stdout, "Reset Encoder Side A \n");
+ }
+ if(side == 'A')
+ {
+ myACM = _acmA;
+ codecCntr = &_codecCntrA;
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ thereIsEncoder = &_thereIsEncoderA;
+ }
+ vad = &_sendVADA;
+ dtx = &_sendDTXA;
+ mode = &_sendVADModeA;
+ myChannel = _channel_A2B;
+ }
+ else
+ {
+ myACM = _acmB;
+ codecCntr = &_codecCntrB;
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ thereIsEncoder = &_thereIsEncoderB;
+ }
+ vad = &_sendVADB;
+ dtx = &_sendDTXB;
+ mode = &_sendVADModeB;
+ myChannel = _channel_B2A;
+ }
+
+ myACM->ResetEncoder();
+ Wait(100);
+
+ // Register the next codec
+ do
+ {
+ *codecCntr = (*codecCntr < AudioCodingModule::NumberOfCodecs() - 1)?
+ (*codecCntr + 1):0;
+
+ if(*codecCntr == 0)
+ {
+ //printf("Initialize Sender Side A \n");
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsEncoder = false;
+ }
+ CHECK_ERROR_MT(myACM->InitializeSender());
+ Wait(1000);
+
+ // After Initialization CN is lost, re-register them
+ if(AudioCodingModule::Codec("CN", myCodec, 8000) >= 0)
+ {
+ CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
+ }
+ if(AudioCodingModule::Codec("CN", myCodec, 16000) >= 0)
+ {
+ CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
+ }
+ // VAD & DTX are disabled after initialization
+ *vad = false;
+ *dtx = false;
+ _writeToFile = false;
+ }
+
+ AudioCodingModule::Codec(*codecCntr, myCodec);
+ } while(!STR_CASE_CMP(myCodec.plname, "CN") ||
+ !STR_CASE_CMP(myCodec.plname, "telephone-event") ||
+ !STR_CASE_CMP(myCodec.plname, "RED"));
+
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n====================================================================\n");
+ fprintf(stdout, " Registering New Codec %s, %d kHz, %d kbps\n",
+ myCodec.plname, myCodec.plfreq / 1000, myCodec.rate / 1000);
+ }
+ //std::cout<< std::flush;
+
+ // NO DTX for supe-wideband codec at this point
+ if(myCodec.plfreq == 32000)
+ {
+ *dtx = false;
+ CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
+
+ }
+
+ CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
+ myChannel->ResetStats();
+ {
+ WriteLockScoped wl(_apiTestRWLock);
+ *thereIsEncoder = true;
+ }
+ Wait(500);
+}
+
+
+void
+APITest::LookForDTMF(char side)
+{
+ if(!_randomTest)
+ {
+ fprintf(stdout, "\n\nLooking for DTMF Signal in Side %c\n", side);
+ fprintf(stdout, "----------------------------------------\n");
+ }
+
+ if(side == 'A')
+ {
+ _acmB->RegisterIncomingMessagesCallback(NULL);
+ _acmA->RegisterIncomingMessagesCallback(_dtmfCallback);
+ Wait(1000);
+ _acmA->RegisterIncomingMessagesCallback(NULL);
+ }
+ else
+ {
+ _acmA->RegisterIncomingMessagesCallback(NULL);
+ _acmB->RegisterIncomingMessagesCallback(_dtmfCallback);
+ Wait(1000);
+ _acmB->RegisterIncomingMessagesCallback(NULL);
+ }
+}
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/audio_coding/main/test/APITest.h b/trunk/src/modules/audio_coding/main/test/APITest.h
new file mode 100644
index 0000000..db0a87c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/APITest.h
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_TEST_H
+#define API_TEST_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+#include "event_wrapper.h"
+#include "utility.h"
+
+namespace webrtc {
+
+enum APITESTAction {TEST_CHANGE_CODEC_ONLY = 0, DTX_TEST = 1};
+
+class APITest : public ACMTest
+{
+public:
+ APITest();
+ ~APITest();
+
+ void Perform();
+private:
+ WebRtc_Word16 SetUp();
+
+ static bool PushAudioThreadA(void* obj);
+ static bool PullAudioThreadA(void* obj);
+ static bool ProcessThreadA(void* obj);
+ static bool APIThreadA(void* obj);
+
+ static bool PushAudioThreadB(void* obj);
+ static bool PullAudioThreadB(void* obj);
+ static bool ProcessThreadB(void* obj);
+ static bool APIThreadB(void* obj);
+
+ void CheckVADStatus(char side);
+
+ // Set Min delay, get delay, playout timestamp
+ void TestDelay(char side);
+
+ // Unregister a codec & register again.
+ void TestRegisteration(char side);
+
+ // Playout Mode, background noise mode.
+ // Receiver Frequency, playout frequency.
+ void TestPlayout(char receiveSide);
+
+ // set/get receiver VAD status & mode.
+ void TestReceiverVAD(char side);
+
+ //
+ void TestSendVAD(char side);
+
+ void CurrentCodec(char side);
+
+ void ChangeCodec(char side);
+
+ void Wait(WebRtc_UWord32 waitLengthMs);
+
+ void LookForDTMF(char side);
+
+ void RunTest(char thread);
+
+ bool PushAudioRunA();
+ bool PullAudioRunA();
+ bool ProcessRunA();
+ bool APIRunA();
+
+ bool PullAudioRunB();
+ bool PushAudioRunB();
+ bool ProcessRunB();
+ bool APIRunB();
+
+
+
+ //--- ACMs
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ //--- Channels
+ Channel* _channel_A2B;
+ Channel* _channel_B2A;
+
+ //--- I/O files
+ // A
+ PCMFile _inFileA;
+ PCMFile _outFileA;
+ // B
+ PCMFile _outFileB;
+ PCMFile _inFileB;
+
+ //--- I/O params
+ // A
+ WebRtc_Word32 _outFreqHzA;
+ // B
+ WebRtc_Word32 _outFreqHzB;
+
+ // Should we write to file.
+ // we might skip writing to file if we
+ // run the test for a long time.
+ bool _writeToFile;
+ //--- Events
+ // A
+ EventWrapper* _pullEventA; // pulling data from ACM
+ EventWrapper* _pushEventA; // pushing data to ACM
+ EventWrapper* _processEventA; // process
+ EventWrapper* _apiEventA; // API calls
+ // B
+ EventWrapper* _pullEventB; // pulling data from ACM
+ EventWrapper* _pushEventB; // pushing data to ACM
+ EventWrapper* _processEventB; // process
+ EventWrapper* _apiEventB; // API calls
+
+ // keep track of the codec in either side.
+ WebRtc_UWord8 _codecCntrA;
+ WebRtc_UWord8 _codecCntrB;
+
+ // keep track of tests
+ WebRtc_UWord8 _testCntrA;
+ WebRtc_UWord8 _testCntrB;
+
+ // Is set to true if there is no encoder in either side
+ bool _thereIsEncoderA;
+ bool _thereIsEncoderB;
+ bool _thereIsDecoderA;
+ bool _thereIsDecoderB;
+
+ bool _sendVADA;
+ bool _sendDTXA;
+ ACMVADMode _sendVADModeA;
+
+ bool _sendVADB;
+ bool _sendDTXB;
+ ACMVADMode _sendVADModeB;
+
+ WebRtc_Word32 _minDelayA;
+ WebRtc_Word32 _minDelayB;
+ bool _payloadUsed[32];
+
+ AudioPlayoutMode _playoutModeA;
+ AudioPlayoutMode _playoutModeB;
+
+ ACMBackgroundNoiseMode _bgnModeA;
+ ACMBackgroundNoiseMode _bgnModeB;
+
+
+ int _receiveVADActivityA[3];
+ int _receiveVADActivityB[3];
+ bool _verbose;
+
+ int _dotPositionA;
+ int _dotMoveDirectionA;
+ int _dotPositionB;
+ int _dotMoveDirectionB;
+
+ char _movingDot[41];
+
+ DTMFDetector* _dtmfCallback;
+ VADCallback* _vadCallbackA;
+ VADCallback* _vadCallbackB;
+ RWLockWrapper& _apiTestRWLock;
+ bool _randomTest;
+ int _testNumA;
+ int _testNumB;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/Channel.cc b/trunk/src/modules/audio_coding/main/test/Channel.cc
new file mode 100644
index 0000000..363b106
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/Channel.cc
@@ -0,0 +1,483 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <iostream>
+
+#include "audio_coding_module.h"
+#include "Channel.h"
+#include "tick_util.h"
+#include "typedefs.h"
+#include "common_types.h"
+
+namespace webrtc {
+
+WebRtc_Word32
+Channel::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation)
+{
+ WebRtcRTPHeader rtpInfo;
+ WebRtc_Word32 status;
+ WebRtc_UWord16 payloadDataSize = payloadSize;
+
+ rtpInfo.header.markerBit = false;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.sequenceNumber = _seqNo++;
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.timestamp = timeStamp;
+ if(frameType == kAudioFrameCN)
+ {
+ rtpInfo.type.Audio.isCNG = true;
+ }
+ else
+ {
+ rtpInfo.type.Audio.isCNG = false;
+ }
+ if(frameType == kFrameEmpty)
+ {
+ // Skip this frame
+ return 0;
+ }
+
+ rtpInfo.type.Audio.channel = 1;
+ // Treat fragmentation separately
+ if(fragmentation != NULL)
+ {
+ if((fragmentation->fragmentationTimeDiff[1] <= 0x3fff) && // silence for too long send only new data
+ (fragmentation->fragmentationVectorSize == 2))
+ {
+ // only 0x80 if we have multiple blocks
+ _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
+ WebRtc_UWord32 REDheader = (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
+ _payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
+ _payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
+ _payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);
+
+ _payloadData[4] = fragmentation->fragmentationPlType[0];
+ // copy the RED data
+ memcpy(_payloadData + 5,
+ payloadData + fragmentation->fragmentationOffset[1],
+ fragmentation->fragmentationLength[1]);
+ // copy the normal data
+ memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
+ payloadData + fragmentation->fragmentationOffset[0],
+ fragmentation->fragmentationLength[0]);
+ payloadDataSize += 5;
+ } else
+ {
+ // single block (newest one)
+ memcpy(_payloadData,
+ payloadData + fragmentation->fragmentationOffset[0],
+ fragmentation->fragmentationLength[0]);
+ payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
+ rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
+ }
+ }
+ else
+ {
+ memcpy(_payloadData, payloadData, payloadDataSize);
+ if(_isStereo)
+ {
+ if(_leftChannel)
+ {
+ memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
+ _leftChannel = false;
+ rtpInfo.type.Audio.channel = 1;
+ }
+ else
+ {
+ memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
+ _leftChannel = true;
+ rtpInfo.type.Audio.channel = 2;
+ }
+ }
+ }
+
+ _channelCritSect->Enter();
+ if(_saveBitStream)
+ {
+ //fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
+ }
+
+ if(!_isStereo)
+ {
+ CalcStatistics(rtpInfo, payloadSize);
+ }
+ _lastInTimestamp = timeStamp;
+ _totalBytes += payloadDataSize;
+ _channelCritSect->Leave();
+
+ if(_useFECTestWithPacketLoss)
+ {
+ _packetLoss += 1;
+ if(_packetLoss == 3)
+ {
+ _packetLoss = 0;
+ return 0;
+ }
+ }
+
+
+ //status = _receiverACM->IncomingPayload((WebRtc_Word8*)_payloadData, payloadSize, payloadType, timeStamp);
+ status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);
+
+ //delete [] payloadData;
+
+
+
+ return status;
+}
+
+void
+Channel::CalcStatistics(
+ WebRtcRTPHeader& rtpInfo,
+ WebRtc_UWord16 payloadSize)
+{
+ int n;
+ if((rtpInfo.header.payloadType != _lastPayloadType) &&
+ (_lastPayloadType != -1))
+ {
+ // payload-type is changed.
+ // we have to terminate the calculations on the previous payload type
+ // we ignore the last packet in that payload type just to make things
+ // easier.
+ for(n = 0; n < MAX_NUM_PAYLOADS; n++)
+ {
+ if(_lastPayloadType == _payloadStats[n].payloadType)
+ {
+ _payloadStats[n].newPacket = true;
+ break;
+ }
+ }
+ }
+ _lastPayloadType = rtpInfo.header.payloadType;
+
+ bool newPayload = true;
+ ACMTestPayloadStats* currentPayloadStr;
+ for(n = 0; n < MAX_NUM_PAYLOADS; n++)
+ {
+ if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
+ {
+ newPayload = false;
+ currentPayloadStr = &_payloadStats[n];
+ break;
+ }
+ }
+
+ if(!newPayload)
+ {
+ if(!currentPayloadStr->newPacket)
+ {
+ WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
+ (WebRtc_UWord32)currentPayloadStr->lastTimestamp);
+ assert(lastFrameSizeSample > 0);
+ int k = 0;
+ while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
+ lastFrameSizeSample) &&
+ (currentPayloadStr->frameSizeStats[k].frameSizeSample != 0))
+ {
+ k++;
+ }
+ ACMTestFrameSizeStats* currentFrameSizeStats =
+ &(currentPayloadStr->frameSizeStats[k]);
+ currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;
+
+ // increment the number of encoded samples.
+ currentFrameSizeStats->totalEncodedSamples +=
+ lastFrameSizeSample;
+ // increment the number of recveived packets
+ currentFrameSizeStats->numPackets++;
+ // increment the total number of bytes (this is based on
+ // the previous payload we don't know the frame-size of
+ // the current payload.
+ currentFrameSizeStats->totalPayloadLenByte +=
+ currentPayloadStr->lastPayloadLenByte;
+ // store the maximum payload-size (this is based on
+ // the previous payload we don't know the frame-size of
+ // the current payload.
+ if(currentFrameSizeStats->maxPayloadLen <
+ currentPayloadStr->lastPayloadLenByte)
+ {
+ currentFrameSizeStats->maxPayloadLen =
+ currentPayloadStr->lastPayloadLenByte;
+ }
+ // store the current values for the next time
+ currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
+ currentPayloadStr->lastPayloadLenByte = payloadSize;
+ }
+ else
+ {
+ currentPayloadStr->newPacket = false;
+ currentPayloadStr->lastPayloadLenByte = payloadSize;
+ currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
+ currentPayloadStr->payloadType = rtpInfo.header.payloadType;
+ }
+ }
+ else
+ {
+ n = 0;
+ while(_payloadStats[n].payloadType != -1)
+ {
+ n++;
+ }
+ // first packet
+ _payloadStats[n].newPacket = false;
+ _payloadStats[n].lastPayloadLenByte = payloadSize;
+ _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
+ _payloadStats[n].payloadType = rtpInfo.header.payloadType;
+ }
+}
+
+Channel::Channel(WebRtc_Word16 chID) :
+_receiverACM(NULL),
+_seqNo(0),
+_channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+_bitStreamFile(NULL),
+_saveBitStream(false),
+_lastPayloadType(-1),
+_isStereo(false),
+_leftChannel(true),
+_lastInTimestamp(0),
+_packetLoss(0),
+_useFECTestWithPacketLoss(false),
+_chID(chID),
+_beginTime(TickTime::MillisecondTimestamp()),
+_totalBytes(0)
+{
+ int n;
+ int k;
+ for(n = 0; n < MAX_NUM_PAYLOADS; n++)
+ {
+ _payloadStats[n].payloadType = -1;
+ _payloadStats[n].newPacket = true;
+ for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
+ {
+ _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+ _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+ _payloadStats[n].frameSizeStats[k].numPackets = 0;
+ _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+ _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+ }
+ }
+ if(chID >= 0)
+ {
+ _saveBitStream = true;
+ char bitStreamFileName[500];
+ sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
+ _bitStreamFile = fopen(bitStreamFileName, "wb");
+ }
+ else
+ {
+ _saveBitStream = false;
+ }
+}
+
+Channel::~Channel()
+{
+ delete _channelCritSect;
+}
+
+void
+Channel::RegisterReceiverACM(AudioCodingModule* acm)
+{
+ _receiverACM = acm;
+ return;
+}
+
+void
+Channel::ResetStats()
+{
+ int n;
+ int k;
+ _channelCritSect->Enter();
+ _lastPayloadType = -1;
+ for(n = 0; n < MAX_NUM_PAYLOADS; n++)
+ {
+ _payloadStats[n].payloadType = -1;
+ _payloadStats[n].newPacket = true;
+ for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
+ {
+ _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+ _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+ _payloadStats[n].frameSizeStats[k].numPackets = 0;
+ _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+ _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+ }
+ }
+ _beginTime = TickTime::MillisecondTimestamp();
+ _totalBytes = 0;
+ _channelCritSect->Leave();
+}
+
+WebRtc_Word16
+Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
+{
+ _channelCritSect->Enter();
+ int n;
+ payloadStats.payloadType = -1;
+ for(n = 0; n < MAX_NUM_PAYLOADS; n++)
+ {
+ if(_payloadStats[n].payloadType == codecInst.pltype)
+ {
+ memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
+ break;
+ }
+ }
+ if(payloadStats.payloadType == -1)
+ {
+ _channelCritSect->Leave();
+ return -1;
+ }
+ for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
+ {
+ if(payloadStats.frameSizeStats[n].frameSizeSample == 0)
+ {
+ _channelCritSect->Leave();
+ return 0;
+ }
+ payloadStats.frameSizeStats[n].usageLenSec =
+ (double)payloadStats.frameSizeStats[n].totalEncodedSamples
+ / (double)codecInst.plfreq;
+
+ payloadStats.frameSizeStats[n].rateBitPerSec =
+ payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
+ payloadStats.frameSizeStats[n].usageLenSec;
+
+ }
+ _channelCritSect->Leave();
+ return 0;
+}
+
+void
+Channel::Stats(WebRtc_UWord32* numPackets)
+{
+ _channelCritSect->Enter();
+ int k;
+ int n;
+ memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
+ for(k = 0; k < MAX_NUM_PAYLOADS; k++)
+ {
+ if(_payloadStats[k].payloadType == -1)
+ {
+ break;
+ }
+ numPackets[k] = 0;
+ for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
+ {
+ if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
+ {
+ break;
+ }
+ numPackets[k] +=
+ _payloadStats[k].frameSizeStats[n].numPackets;
+ }
+ }
+ _channelCritSect->Leave();
+}
+
+void
+Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
+{
+ _channelCritSect->Enter();
+
+ int k;
+ int n;
+ memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
+ for(k = 0; k < MAX_NUM_PAYLOADS; k++)
+ {
+ if(_payloadStats[k].payloadType == -1)
+ {
+ break;
+ }
+ payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
+ payloadLenByte[k] = 0;
+ for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
+ {
+ if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
+ {
+ break;
+ }
+ payloadLenByte[k] += (WebRtc_UWord16)
+ _payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
+ }
+ }
+
+ _channelCritSect->Leave();
+}
+
+
+void
+Channel::PrintStats(CodecInst& codecInst)
+{
+ ACMTestPayloadStats payloadStats;
+ Stats(codecInst, payloadStats);
+ printf("%s %d kHz\n",
+ codecInst.plname,
+ codecInst.plfreq / 1000);
+ printf("=====================================================\n");
+ if(payloadStats.payloadType == -1)
+ {
+ printf("No Packets are sent with payload-type %d (%s)\n\n",
+ codecInst.pltype,
+ codecInst.plname);
+ return;
+ }
+ for(int k = 0; k < MAX_NUM_FRAMESIZES; k++)
+ {
+ if(payloadStats.frameSizeStats[k].frameSizeSample == 0)
+ {
+ break;
+ }
+ printf("Frame-size.................... %d samples\n",
+ payloadStats.frameSizeStats[k].frameSizeSample);
+ printf("Average Rate.................. %.0f bits/sec\n",
+ payloadStats.frameSizeStats[k].rateBitPerSec);
+ printf("Maximum Payload-Size.......... %d Bytes\n",
+ payloadStats.frameSizeStats[k].maxPayloadLen);
+ printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
+ ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
+ (double)codecInst.plfreq) /
+ (double)payloadStats.frameSizeStats[k].frameSizeSample);
+ printf("Number of Packets............. %u\n",
+ (unsigned int)payloadStats.frameSizeStats[k].numPackets);
+ printf("Duration...................... %0.3f sec\n\n",
+ payloadStats.frameSizeStats[k].usageLenSec);
+
+ }
+
+}
+
+WebRtc_UWord32
+Channel::LastInTimestamp()
+{
+ WebRtc_UWord32 timestamp;
+ _channelCritSect->Enter();
+ timestamp = _lastInTimestamp;
+ _channelCritSect->Leave();
+ return timestamp;
+}
+
+double
+Channel::BitRate()
+{
+ double rate;
+ WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
+ _channelCritSect->Enter();
+ rate = ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
+ _channelCritSect->Leave();
+ return rate;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/Channel.h b/trunk/src/modules/audio_coding/main/test/Channel.h
new file mode 100644
index 0000000..375bec7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/Channel.h
@@ -0,0 +1,125 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef CHANNEL_H
+#define CHANNEL_H
+
+#include <stdio.h>
+
+#include "audio_coding_module.h"
+#include "critical_section_wrapper.h"
+#include "rw_lock_wrapper.h"
+
+namespace webrtc {
+
+#define MAX_NUM_PAYLOADS 50
+#define MAX_NUM_FRAMESIZES 6
+
+
+struct ACMTestFrameSizeStats
+{
+ WebRtc_UWord16 frameSizeSample;
+ WebRtc_Word16 maxPayloadLen;
+ WebRtc_UWord32 numPackets;
+ WebRtc_UWord64 totalPayloadLenByte;
+ WebRtc_UWord64 totalEncodedSamples;
+ double rateBitPerSec;
+ double usageLenSec;
+
+};
+
+struct ACMTestPayloadStats
+{
+ bool newPacket;
+ WebRtc_Word16 payloadType;
+ WebRtc_Word16 lastPayloadLenByte;
+ WebRtc_UWord32 lastTimestamp;
+ ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
+};
+
+class Channel: public AudioPacketizationCallback
+{
+public:
+
+ Channel(
+ WebRtc_Word16 chID = -1);
+ ~Channel();
+
+ WebRtc_Word32 SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ void RegisterReceiverACM(
+ AudioCodingModule *acm);
+
+ void ResetStats();
+
+ WebRtc_Word16 Stats(
+ CodecInst& codecInst,
+ ACMTestPayloadStats& payloadStats);
+
+ void Stats(
+ WebRtc_UWord32* numPackets);
+
+ void Stats(
+ WebRtc_UWord8* payloadLenByte,
+ WebRtc_UWord32* payloadType);
+
+ void PrintStats(
+ CodecInst& codecInst);
+
+ void SetIsStereo(bool isStereo)
+ {
+ _isStereo = isStereo;
+ }
+
+ WebRtc_UWord32 LastInTimestamp();
+
+ void SetFECTestWithPacketLoss(bool usePacketLoss)
+ {
+ _useFECTestWithPacketLoss = usePacketLoss;
+ }
+
+ double BitRate();
+
+private:
+ void CalcStatistics(
+ WebRtcRTPHeader& rtpInfo,
+ WebRtc_UWord16 payloadSize);
+
+ AudioCodingModule* _receiverACM;
+ WebRtc_UWord16 _seqNo;
+ // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
+ WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
+
+ CriticalSectionWrapper* _channelCritSect;
+ FILE* _bitStreamFile;
+ bool _saveBitStream;
+ WebRtc_Word16 _lastPayloadType;
+ ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
+ bool _isStereo;
+ WebRtcRTPHeader _rtpInfo;
+ bool _leftChannel;
+ WebRtc_UWord32 _lastInTimestamp;
+ // FEC Test variables
+ WebRtc_Word16 _packetLoss;
+ bool _useFECTestWithPacketLoss;
+ WebRtc_Word16 _chID;
+ WebRtc_UWord64 _beginTime;
+ WebRtc_UWord64 _totalBytes;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
new file mode 100644
index 0000000..74bb84c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -0,0 +1,407 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "EncodeDecodeTest.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "audio_coding_module.h"
+#include "common_types.h"
+#include "gtest/gtest.h"
+#include "trace.h"
+#include "testsupport/fileutils.h"
+#include "utility.h"
+
+namespace webrtc {
+
+TestPacketization::TestPacketization(RTPStream *rtpStream,
+ WebRtc_UWord16 frequency)
+ : _rtpStream(rtpStream),
+ _frequency(frequency),
+ _seqNo(0) {
+}
+
+TestPacketization::~TestPacketization() { }
+
+WebRtc_Word32 TestPacketization::SendData(
+ const FrameType /* frameType */,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* /* fragmentation */) {
+ _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
+ _frequency);
+ return 1;
+}
+
+Sender::Sender()
+ : _acm(NULL),
+ _pcmFile(),
+ _audioFrame(),
+ _payloadSize(0),
+ _timeStamp(0),
+ _packetization(NULL) {
+}
+
+void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
+ acm->InitializeSender();
+ struct CodecInst sendCodec;
+ int noOfCodecs = acm->NumberOfCodecs();
+ int codecNo;
+
+ if (testMode == 1) {
+ // Set the codec, input file, and parameters for the current test.
+ codecNo = codeId;
+ // Use same input file for now.
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ } else if (testMode == 0) {
+ // Set the codec, input file, and parameters for the current test.
+ codecNo = codeId;
+ acm->Codec(codecNo, sendCodec);
+ // Use same input file for now.
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ } else {
+ printf("List of supported codec.\n");
+ for (int n = 0; n < noOfCodecs; n++) {
+ acm->Codec(n, sendCodec);
+ printf("%d %s\n", n, sendCodec.plname);
+ }
+ printf("Choose your codec:");
+ ASSERT_GT(scanf("%d", &codecNo), 0);
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ }
+
+ acm->Codec(codecNo, sendCodec);
+ acm->RegisterSendCodec(sendCodec);
+ _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
+ if (acm->RegisterTransportCallback(_packetization) < 0) {
+ printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
+ codeId);
+ }
+
+ _acm = acm;
+ }
+
+void Sender::Teardown() {
+ _pcmFile.Close();
+ delete _packetization;
+}
+
+bool Sender::Add10MsData() {
+ if (!_pcmFile.EndOfFile()) {
+ _pcmFile.Read10MsData(_audioFrame);
+ WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
+ if (ok != 0) {
+ printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ return true;
+ }
+ return false;
+}
+
+bool Sender::Process() {
+ WebRtc_Word32 ok = _acm->Process();
+ if (ok < 0) {
+ printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ return true;
+}
+
+void Sender::Run() {
+ while (true) {
+ if (!Add10MsData()) {
+ break;
+ }
+ if (!Process()) { // This could be done in a processing thread
+ break;
+ }
+ }
+}
+
+Receiver::Receiver()
+ : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
+ _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
+}
+
+void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
+ struct CodecInst recvCodec;
+ int noOfCodecs;
+ acm->InitializeReceiver();
+
+ noOfCodecs = acm->NumberOfCodecs();
+ for (int i = 0; i < noOfCodecs; i++) {
+ acm->Codec((WebRtc_UWord8) i, recvCodec);
+ if (acm->RegisterReceiveCodec(recvCodec) != 0) {
+ printf("Unable to register codec: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ }
+
+ char filename[128];
+ _rtpStream = rtpStream;
+ int playSampFreq;
+
+ if (testMode == 1) {
+ playSampFreq=recvCodec.plfreq;
+ //output file for current run
+ sprintf(filename,"%s/out%dFile.pcm", webrtc::test::OutputPath().c_str(),
+ codeId);
+ _pcmFile.Open(filename, recvCodec.plfreq, "wb+");
+ } else if (testMode == 0) {
+ playSampFreq=32000;
+ //output file for current run
+ sprintf(filename, "%s/encodeDecode_out%d.pcm",
+ webrtc::test::OutputPath().c_str(), codeId);
+ _pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
+ } else {
+ printf("\nValid output frequencies:\n");
+ printf("8000\n16000\n32000\n-1,");
+ printf("which means output freq equal to received signal freq");
+ printf("\n\nChoose output sampling frequency: ");
+ ASSERT_GT(scanf("%d", &playSampFreq), 0);
+ sprintf(filename, "%s/outFile.pcm", webrtc::test::OutputPath().c_str());
+ _pcmFile.Open(filename, 32000, "wb+");
+ }
+
+ _realPayloadSizeBytes = 0;
+ _playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
+ _frequency = playSampFreq;
+ _acm = acm;
+ _firstTime = true;
+}
+
+void Receiver::Teardown() {
+ delete [] _playoutBuffer;
+ _pcmFile.Close();
+ if (testMode > 1)
+ Trace::ReturnTrace();
+}
+
+bool Receiver::IncomingPacket() {
+ if (!_rtpStream->EndOfFile()) {
+ if (_firstTime) {
+ _firstTime = false;
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+ _payloadSizeBytes, &_nextTime);
+ if (_realPayloadSizeBytes == 0) {
+ if (_rtpStream->EndOfFile()) {
+ _firstTime = true;
+ return true;
+ } else {
+ printf("Error in reading incoming payload.\n");
+ return false;
+ }
+ }
+ }
+
+ WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
+ _realPayloadSizeBytes, _rtpInfo);
+ if (ok != 0) {
+ printf("Error when inserting packet to ACM, for run: codecId: %d\n",
+ codeId);
+ exit(1);
+ }
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+ _payloadSizeBytes, &_nextTime);
+ if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+ _firstTime = true;
+ }
+ }
+ return true;
+}
+
+bool Receiver::PlayoutData() {
+ AudioFrame audioFrame;
+
+ if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
+ printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
+ codeId);
+ exit(1);
+ }
+ if (_playoutLengthSmpls == 0) {
+ return false;
+ }
+ _pcmFile.Write10MsData(audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples);
+ return true;
+}
+
+void Receiver::Run() {
+ WebRtc_UWord8 counter500Ms = 50;
+ WebRtc_UWord32 clock = 0;
+
+ while (counter500Ms > 0) {
+ if (clock == 0 || clock >= _nextTime) {
+ IncomingPacket();
+ if (clock == 0) {
+ clock = _nextTime;
+ }
+ }
+ if ((clock % 10) == 0) {
+ if (!PlayoutData()) {
+ clock++;
+ continue;
+ }
+ }
+ if (_rtpStream->EndOfFile()) {
+ counter500Ms--;
+ }
+ clock++;
+ }
+}
+
+EncodeDecodeTest::EncodeDecodeTest() {
+ _testMode = 2;
+ Trace::CreateTrace();
+ Trace::SetTraceFile("acm_encdec_test.txt");
+}
+
+EncodeDecodeTest::EncodeDecodeTest(int testMode) {
+ //testMode == 0 for autotest
+ //testMode == 1 for testing all codecs/parameters
+ //testMode > 1 for specific user-input test (as it was used before)
+ _testMode = testMode;
+ if(_testMode != 0) {
+ Trace::CreateTrace();
+ Trace::SetTraceFile("acm_encdec_test.txt");
+ }
+}
+
+void EncodeDecodeTest::Perform() {
+ if (_testMode == 0) {
+ printf("Running Encode/Decode Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ "---------- EncodeDecodeTest ----------");
+ }
+
+ int numCodecs = 1;
+ int codePars[3]; //freq, pacsize, rate
+ int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
+ //for a given codec
+
+ codePars[0] = 0;
+ codePars[1] = 0;
+ codePars[2] = 0;
+
+ if (_testMode == 1) {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ struct CodecInst sendCodecTmp;
+ numCodecs = acmTmp->NumberOfCodecs();
+ printf("List of supported codec.\n");
+ for(int n = 0; n < numCodecs; n++) {
+ acmTmp->Codec(n, sendCodecTmp);
+ if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
+ numPars[n] = 0;
+ } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
+ numPars[n] = 0;
+ } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
+ numPars[n] = 0;
+ } else {
+ numPars[n] = 1;
+ printf("%d %s\n", n, sendCodecTmp.plname);
+ }
+ }
+ AudioCodingModule::Destroy(acmTmp);
+ } else if (_testMode == 0) {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ numCodecs = acmTmp->NumberOfCodecs();
+ AudioCodingModule::Destroy(acmTmp);
+ struct CodecInst dummyCodec;
+
+ //chose range of testing for codecs/parameters
+ for(int i = 0 ; i < numCodecs ; i++) {
+ numPars[i] = 1;
+ acmTmp->Codec(i, dummyCodec);
+ if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0) {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
+ numPars[i] = 0;
+ }
+ }
+ } else {
+ numCodecs = 1;
+ numPars[0] = 1;
+ }
+
+ _receiver.testMode = _testMode;
+
+ //loop over all codecs:
+ for (int codeId = 0; codeId < numCodecs; codeId++) {
+ //only encode using real encoders, not telephone-event anc cn
+ for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
+ if (_testMode == 1) {
+ printf("\n");
+ printf("***FOR RUN: codeId: %d\n", codeId);
+ printf("\n");
+ } else if (_testMode == 0) {
+ printf(".");
+ }
+
+ EncodeToFile(1, codeId, codePars, _testMode);
+
+ AudioCodingModule *acm = AudioCodingModule::Create(10);
+ RTPFile rtpFile;
+ std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
+ rtpFile.Open(fileName.c_str(), "rb");
+
+ _receiver.codeId = codeId;
+
+ rtpFile.ReadHeader();
+ _receiver.Setup(acm, &rtpFile);
+ _receiver.Run();
+ _receiver.Teardown();
+ rtpFile.Close();
+ AudioCodingModule::Destroy(acm);
+
+ if (_testMode == 1) {
+ printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId);
+ }
+ }
+ }
+ if (_testMode == 0) {
+ printf("Done!\n");
+ }
+ if (_testMode == 1)
+ Trace::ReturnTrace();
+}
+
+void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
+ int testMode) {
+ AudioCodingModule *acm = AudioCodingModule::Create(0);
+ RTPFile rtpFile;
+ std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
+ rtpFile.Open(fileName.c_str(), "wb+");
+ rtpFile.WriteHeader();
+
+ //for auto_test and logging
+ _sender.testMode = testMode;
+ _sender.codeId = codeId;
+
+ _sender.Setup(acm, &rtpFile);
+ struct CodecInst sendCodecInst;
+ if (acm->SendCodec(sendCodecInst) >= 0) {
+ _sender.Run();
+ }
+ _sender.Teardown();
+ rtpFile.Close();
+ AudioCodingModule::Destroy(acm);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h
new file mode 100644
index 0000000..a730fea
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+
+#include "ACMTest.h"
+#include "audio_coding_module.h"
+#include "RTPFile.h"
+#include "PCMFile.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+#define MAX_INCOMING_PAYLOAD 8096
+
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization: public AudioPacketizationCallback {
+ public:
+ TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
+ ~TestPacketization();
+ virtual WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ private:
+ static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
+ WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
+ WebRtc_UWord32 ssrc);
+ RTPStream* _rtpStream;
+ WebRtc_Word32 _frequency;
+ WebRtc_Word16 _seqNo;
+};
+
+class Sender {
+ public:
+ Sender();
+ void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+ void Teardown();
+ void Run();
+ bool Add10MsData();
+ bool Process();
+
+ //for auto_test and logging
+ WebRtc_UWord8 testMode;
+ WebRtc_UWord8 codeId;
+
+ private:
+ AudioCodingModule* _acm;
+ PCMFile _pcmFile;
+ AudioFrame _audioFrame;
+ WebRtc_UWord16 _payloadSize;
+ WebRtc_UWord32 _timeStamp;
+ TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+ Receiver();
+ void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+ void Teardown();
+ void Run();
+ bool IncomingPacket();
+ bool PlayoutData();
+
+ //for auto_test and logging
+ WebRtc_UWord8 codeId;
+ WebRtc_UWord8 testMode;
+
+ private:
+ AudioCodingModule* _acm;
+ bool _rtpEOF;
+ RTPStream* _rtpStream;
+ PCMFile _pcmFile;
+ WebRtc_Word16* _playoutBuffer;
+ WebRtc_UWord16 _playoutLengthSmpls;
+ WebRtc_Word8 _incomingPayload[MAX_INCOMING_PAYLOAD];
+ WebRtc_UWord16 _payloadSizeBytes;
+ WebRtc_UWord16 _realPayloadSizeBytes;
+ WebRtc_Word32 _frequency;
+ bool _firstTime;
+ WebRtcRTPHeader _rtpInfo;
+ WebRtc_UWord32 _nextTime;
+};
+
+class EncodeDecodeTest: public ACMTest {
+ public:
+ EncodeDecodeTest();
+ EncodeDecodeTest(int testMode);
+ virtual void Perform();
+
+ WebRtc_UWord16 _playoutFreq;
+ WebRtc_UWord8 _testMode;
+
+ private:
+ void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
+
+ protected:
+ Sender _sender;
+ Receiver _receiver;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/PCMFile.cc b/trunk/src/modules/audio_coding/main/test/PCMFile.cc
new file mode 100644
index 0000000..520ddbe
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/PCMFile.cc
@@ -0,0 +1,302 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "PCMFile.h"
+
+#include <cctype>
+#include <stdio.h>
+#include <string.h>
+
+#include "gtest/gtest.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+PCMFile::PCMFile():
+_pcmFile(NULL),
+_nSamples10Ms(160),
+_frequency(16000),
+_endOfFile(false),
+_autoRewind(false),
+_rewinded(false),
+_timestamp(0),
+_readStereo(false),
+_saveStereo(false)
+{
+ _timestamp = (((WebRtc_UWord32)rand() & 0x0000FFFF) << 16) |
+ ((WebRtc_UWord32)rand() & 0x0000FFFF);
+}
+
+/*
+PCMFile::~PCMFile()
+{
+ if(_pcmFile != NULL)
+ {
+ fclose(_pcmFile);
+ _pcmFile = NULL;
+ }
+}
+*/
+
+WebRtc_Word16
+PCMFile::ChooseFile(
+ char* fileName,
+ WebRtc_Word16 maxLen)
+{
+ char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
+ //strcpy(_fileName, "in.pcm");
+ //printf("\n\nPlease enter the input file: ");
+ EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
+ tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
+ WebRtc_Word16 n = 0;
+
+ // removing leading spaces
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (tmpName[n] != 0) &&
+ (n < MAX_FILE_NAME_LENGTH_BYTE))
+ {
+ n++;
+ }
+ if(n > 0)
+ {
+ memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
+ }
+
+ //removing trailing spaces
+ n = (WebRtc_Word16)(strlen(tmpName) - 1);
+ if(n >= 0)
+ {
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (n >= 0))
+ {
+ n--;
+ }
+ }
+ if(n >= 0)
+ {
+ tmpName[n + 1] = '\0';
+ }
+
+ WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
+ if(len > maxLen)
+ {
+ return -1;
+ }
+ if(len > 0)
+ {
+ strncpy(fileName, tmpName, len+1);
+ }
+ return 0;
+}
+
+WebRtc_Word16
+PCMFile::ChooseFile(
+ char* fileName,
+ WebRtc_Word16 maxLen,
+ WebRtc_UWord16* frequencyHz)
+{
+ char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
+ //strcpy(_fileName, "in.pcm");
+ //printf("\n\nPlease enter the input file: ");
+ EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
+ tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
+ WebRtc_Word16 n = 0;
+
+ // removing leading spaces
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (tmpName[n] != 0) &&
+ (n < MAX_FILE_NAME_LENGTH_BYTE))
+ {
+ n++;
+ }
+ if(n > 0)
+ {
+ memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
+ }
+
+ //removing trailing spaces
+ n = (WebRtc_Word16)(strlen(tmpName) - 1);
+ if(n >= 0)
+ {
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (n >= 0))
+ {
+ n--;
+ }
+ }
+ if(n >= 0)
+ {
+ tmpName[n + 1] = '\0';
+ }
+
+ WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
+ if(len > maxLen)
+ {
+ return -1;
+ }
+ if(len > 0)
+ {
+ strncpy(fileName, tmpName, len+1);
+ }
+ printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequencyHz);
+ EXPECT_TRUE(fgets(tmpName, 10, stdin) != NULL);
+ WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
+ if(tmpFreq > 0)
+ {
+ *frequencyHz = tmpFreq;
+ }
+ return 0;
+}
+
+void
+PCMFile::Open(
+ const char* filename,
+ WebRtc_UWord16 frequency,
+ const char* mode,
+ bool autoRewind)
+{
+ if ((_pcmFile = fopen(filename, mode)) == NULL)
+ {
+ printf("Cannot open file %s.\n", filename);
+ ADD_FAILURE() << "Unable to read file";
+ }
+ _frequency = frequency;
+ _nSamples10Ms = (WebRtc_UWord16)(_frequency / 100);
+ _autoRewind = autoRewind;
+ _endOfFile = false;
+ _rewinded = false;
+}
+
+WebRtc_Word32
+PCMFile::SamplingFrequency() const
+{
+ return _frequency;
+}
+
+WebRtc_UWord16
+PCMFile::PayloadLength10Ms() const
+{
+ return _nSamples10Ms;
+}
+
+WebRtc_Word32
+PCMFile::Read10MsData(
+ AudioFrame& audioFrame)
+{
+ WebRtc_UWord16 noChannels = 1;
+ if (_readStereo)
+ {
+ noChannels = 2;
+ }
+
+ WebRtc_Word32 payloadSize = (WebRtc_Word32)fread(audioFrame._payloadData, sizeof(WebRtc_UWord16), _nSamples10Ms*noChannels, _pcmFile);
+ if (payloadSize < _nSamples10Ms*noChannels) {
+ for (int k = payloadSize; k < _nSamples10Ms*noChannels; k++)
+ {
+ audioFrame._payloadData[k] = 0;
+ }
+ if(_autoRewind)
+ {
+ rewind(_pcmFile);
+ _rewinded = true;
+ }
+ else
+ {
+ _endOfFile = true;
+ }
+ }
+ audioFrame._payloadDataLengthInSamples = _nSamples10Ms;
+ audioFrame._frequencyInHz = _frequency;
+ audioFrame._audioChannel = noChannels;
+ audioFrame._timeStamp = _timestamp;
+ _timestamp += _nSamples10Ms;
+ return _nSamples10Ms;
+}
+
+void
+PCMFile::Write10MsData(
+ AudioFrame& audioFrame)
+{
+ if(audioFrame._audioChannel == 1)
+ {
+ if(!_saveStereo)
+ {
+ fwrite(audioFrame._payloadData, sizeof(WebRtc_UWord16),
+ audioFrame._payloadDataLengthInSamples, _pcmFile);
+ }
+ else
+ {
+ WebRtc_Word16* stereoAudio = new WebRtc_Word16[2 *
+ audioFrame._payloadDataLengthInSamples];
+ int k;
+ for(k = 0; k < audioFrame._payloadDataLengthInSamples; k++)
+ {
+ stereoAudio[k<<1] = audioFrame._payloadData[k];
+ stereoAudio[(k<<1) + 1] = audioFrame._payloadData[k];
+ }
+ fwrite(stereoAudio, sizeof(WebRtc_Word16), 2*audioFrame._payloadDataLengthInSamples,
+ _pcmFile);
+ delete [] stereoAudio;
+ }
+ }
+ else
+ {
+ fwrite(audioFrame._payloadData, sizeof(WebRtc_Word16),
+ audioFrame._audioChannel * audioFrame._payloadDataLengthInSamples, _pcmFile);
+ }
+}
+
+
+void
+PCMFile::Write10MsData(
+ WebRtc_Word16* playoutBuffer,
+ WebRtc_UWord16 playoutLengthSmpls)
+{
+ fwrite(playoutBuffer, sizeof(WebRtc_UWord16), playoutLengthSmpls, _pcmFile);
+}
+
+
+void
+PCMFile::Close()
+{
+ fclose(_pcmFile);
+ _pcmFile = NULL;
+}
+
+void
+PCMFile::Rewind()
+{
+ rewind(_pcmFile);
+ _endOfFile = false;
+}
+
+bool
+PCMFile::Rewinded()
+{
+ return _rewinded;
+}
+
+void
+PCMFile::SaveStereo(
+ bool saveStereo)
+{
+ _saveStereo = saveStereo;
+}
+
+void
+PCMFile::ReadStereo(
+ bool readStereo)
+{
+ _readStereo = readStereo;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/PCMFile.h b/trunk/src/modules/audio_coding/main/test/PCMFile.h
new file mode 100644
index 0000000..a9cb9cf
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/PCMFile.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PCMFILE_H
+#define PCMFILE_H
+
+#include "typedefs.h"
+#include "module_common_types.h"
+#include <cstdio>
+#include <cstdlib>
+
+namespace webrtc {
+
+class PCMFile
+{
+public:
+ PCMFile();
+ ~PCMFile()
+ {
+ if(_pcmFile != NULL)
+ {
+ fclose(_pcmFile);
+ }
+ }
+ void Open(const char *filename, WebRtc_UWord16 frequency, const char *mode,
+ bool autoRewind = false);
+
+ WebRtc_Word32 Read10MsData(AudioFrame& audioFrame);
+
+ void Write10MsData(WebRtc_Word16 *playoutBuffer, WebRtc_UWord16 playoutLengthSmpls);
+ void Write10MsData(AudioFrame& audioFrame);
+
+ WebRtc_UWord16 PayloadLength10Ms() const;
+ WebRtc_Word32 SamplingFrequency() const;
+ void Close();
+ bool EndOfFile() const { return _endOfFile; }
+ void Rewind();
+ static WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen,
+ WebRtc_UWord16* frequencyHz);
+ static WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen);
+ bool Rewinded();
+ void SaveStereo(
+ bool saveStereo = true);
+ void ReadStereo(
+ bool readStereo = true);
+private:
+ FILE* _pcmFile;
+ WebRtc_UWord16 _nSamples10Ms;
+ WebRtc_Word32 _frequency;
+ bool _endOfFile;
+ bool _autoRewind;
+ bool _rewinded;
+ WebRtc_UWord32 _timestamp;
+ bool _readStereo;
+ bool _saveStereo;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/RTPFile.cc b/trunk/src/modules/audio_coding/main/test/RTPFile.cc
new file mode 100644
index 0000000..29d6a1d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/RTPFile.cc
@@ -0,0 +1,260 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "RTPFile.h"
+
+#include <stdlib.h>
+
+#ifdef WIN32
+# include <Winsock2.h>
+#else
+# include <arpa/inet.h>
+#endif
+
+#include "audio_coding_module.h"
+#include "engine_configurations.h"
+#include "gtest/gtest.h" // TODO (tlegrand): Consider removing usage of gtest.
+#include "rw_lock_wrapper.h"
+
+namespace webrtc {
+
+void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
+{
+ rtpInfo->header.payloadType = rtpHeader[1];
+ rtpInfo->header.sequenceNumber = (static_cast<WebRtc_UWord16>(rtpHeader[2])<<8) | rtpHeader[3];
+ rtpInfo->header.timestamp = (static_cast<WebRtc_UWord32>(rtpHeader[4])<<24) |
+ (static_cast<WebRtc_UWord32>(rtpHeader[5])<<16) |
+ (static_cast<WebRtc_UWord32>(rtpHeader[6])<<8) |
+ rtpHeader[7];
+ rtpInfo->header.ssrc = (static_cast<WebRtc_UWord32>(rtpHeader[8])<<24) |
+ (static_cast<WebRtc_UWord32>(rtpHeader[9])<<16) |
+ (static_cast<WebRtc_UWord32>(rtpHeader[10])<<8) |
+ rtpHeader[11];
+}
+
+void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader,
+ WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
+ WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc)
+{
+ rtpHeader[0]=(unsigned char)0x80;
+ rtpHeader[1]=(unsigned char)(payloadType & 0xFF);
+ rtpHeader[2]=(unsigned char)((seqNo>>8)&0xFF);
+ rtpHeader[3]=(unsigned char)((seqNo)&0xFF);
+ rtpHeader[4]=(unsigned char)((timeStamp>>24)&0xFF);
+ rtpHeader[5]=(unsigned char)((timeStamp>>16)&0xFF);
+
+ rtpHeader[6]=(unsigned char)((timeStamp>>8)&0xFF);
+ rtpHeader[7]=(unsigned char)(timeStamp & 0xFF);
+
+ rtpHeader[8]=(unsigned char)((ssrc>>24)&0xFF);
+ rtpHeader[9]=(unsigned char)((ssrc>>16)&0xFF);
+
+ rtpHeader[10]=(unsigned char)((ssrc>>8)&0xFF);
+ rtpHeader[11]=(unsigned char)(ssrc & 0xFF);
+}
+
+
+RTPPacket::RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
+ WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+ :
+payloadType(payloadType),
+timeStamp(timeStamp),
+seqNo(seqNo),
+payloadSize(payloadSize),
+frequency(frequency)
+{
+ if (payloadSize > 0)
+ {
+ this->payloadData = new WebRtc_UWord8[payloadSize];
+ memcpy(this->payloadData, payloadData, payloadSize);
+ }
+}
+
+RTPPacket::~RTPPacket()
+{
+ delete [] payloadData;
+}
+
+RTPBuffer::RTPBuffer()
+{
+ _queueRWLock = RWLockWrapper::CreateRWLock();
+}
+
+RTPBuffer::~RTPBuffer()
+{
+ delete _queueRWLock;
+}
+
+void
+RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+{
+ RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, payloadSize, frequency);
+ _queueRWLock->AcquireLockExclusive();
+ _rtpQueue.push(packet);
+ _queueRWLock->ReleaseLockExclusive();
+}
+
+WebRtc_UWord16
+RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset)
+{
+ _queueRWLock->AcquireLockShared();
+ RTPPacket *packet = _rtpQueue.front();
+ _rtpQueue.pop();
+ _queueRWLock->ReleaseLockShared();
+ rtpInfo->header.markerBit = 1;
+ rtpInfo->header.payloadType = packet->payloadType;
+ rtpInfo->header.sequenceNumber = packet->seqNo;
+ rtpInfo->header.ssrc = 0;
+ rtpInfo->header.timestamp = packet->timeStamp;
+ if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize)
+ {
+ memcpy(payloadData, packet->payloadData, packet->payloadSize);
+ }
+ else
+ {
+ return 0;
+ }
+ *offset = (packet->timeStamp/(packet->frequency/1000));
+
+ return packet->payloadSize;
+}
+
+bool
+RTPBuffer::EndOfFile() const
+{
+ _queueRWLock->AcquireLockShared();
+ bool eof = _rtpQueue.empty();
+ _queueRWLock->ReleaseLockShared();
+ return eof;
+}
+
+void RTPFile::Open(const char *filename, const char *mode)
+{
+ if ((_rtpFile = fopen(filename, mode)) == NULL)
+ {
+ printf("Cannot write file %s.\n", filename);
+ ADD_FAILURE() << "Unable to write file";
+ exit(1);
+ }
+}
+
+void RTPFile::Close()
+{
+ if (_rtpFile != NULL)
+ {
+ fclose(_rtpFile);
+ _rtpFile = NULL;
+ }
+}
+
+
+void RTPFile::WriteHeader()
+{
+ // Write data in a format that NetEQ and RTP Play can parse
+ fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
+ WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
+ fwrite(&dummy_variable, 4, 1, _rtpFile);
+ fwrite(&dummy_variable, 4, 1, _rtpFile);
+ fwrite(&dummy_variable, 4, 1, _rtpFile);
+ fwrite(&dummy_variable, 2, 1, _rtpFile);
+ fwrite(&dummy_variable, 2, 1, _rtpFile);
+ fflush(_rtpFile);
+}
+
+void RTPFile::ReadHeader()
+{
+ WebRtc_UWord32 start_sec, start_usec, source;
+ WebRtc_UWord16 port, padding;
+ char fileHeader[40];
+ EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
+ EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
+ start_sec=ntohl(start_sec);
+ EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
+ start_usec=ntohl(start_usec);
+ EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
+ source=ntohl(source);
+ EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
+ port=ntohs(port);
+ EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
+ padding=ntohs(padding);
+}
+
+void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
+{
+ /* write RTP packet to file */
+ WebRtc_UWord8 rtpHeader[12];
+ MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
+ WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
+ WebRtc_UWord16 plen = htons(12 + payloadSize);
+ WebRtc_UWord32 offsetMs;
+
+ offsetMs = (timeStamp/(frequency/1000));
+ offsetMs = htonl(offsetMs);
+ fwrite(&lengthBytes, 2, 1, _rtpFile);
+ fwrite(&plen, 2, 1, _rtpFile);
+ fwrite(&offsetMs, 4, 1, _rtpFile);
+ fwrite(rtpHeader, 12, 1, _rtpFile);
+ fwrite(payloadData, 1, payloadSize, _rtpFile);
+}
+
+WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset)
+{
+ WebRtc_UWord16 lengthBytes;
+ WebRtc_UWord16 plen;
+ WebRtc_UWord8 rtpHeader[12];
+ size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
+ /* Check if we have reached end of file. */
+ if ((read_len == 0) && feof(_rtpFile))
+ {
+ _rtpEOF = true;
+ return 0;
+ }
+ EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
+ EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
+ lengthBytes = ntohs(lengthBytes);
+ plen = ntohs(plen);
+ *offset = ntohl(*offset);
+ EXPECT_GT(plen, 11);
+
+ EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
+ ParseRTPHeader(rtpInfo, rtpHeader);
+ rtpInfo->type.Audio.isCNG = false;
+ rtpInfo->type.Audio.channel = 1;
+ EXPECT_EQ(lengthBytes, plen + 8);
+
+ if (plen == 0)
+ {
+ return 0;
+ }
+ if (payloadSize < (lengthBytes - 20))
+ {
+ return -1;
+ }
+ if (lengthBytes < 20)
+ {
+ return -1;
+ }
+ lengthBytes -= 20;
+ EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
+ return lengthBytes;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/RTPFile.h b/trunk/src/modules/audio_coding/main/test/RTPFile.h
new file mode 100644
index 0000000..e11b160
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/RTPFile.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTPFILE_H
+#define RTPFILE_H
+
+#include "audio_coding_module.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+#include "rw_lock_wrapper.h"
+#include <stdio.h>
+#include <queue>
+
+namespace webrtc {
+
+class RTPStream
+{
+public:
+ virtual ~RTPStream(){}
+
+ virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency) = 0;
+
+ // Returns the packet's payload size. Zero should be treated as an
+ // end-of-stream (in the case that EndOfFile() is true) or an error.
+ virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset) = 0;
+ virtual bool EndOfFile() const = 0;
+
+protected:
+ void MakeRTPheader(WebRtc_UWord8* rtpHeader,
+ WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
+ WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
+ void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader);
+};
+
+class RTPPacket
+{
+public:
+ RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
+ WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+ ~RTPPacket();
+ WebRtc_UWord8 payloadType;
+ WebRtc_UWord32 timeStamp;
+ WebRtc_Word16 seqNo;
+ WebRtc_UWord8* payloadData;
+ WebRtc_UWord16 payloadSize;
+ WebRtc_UWord32 frequency;
+};
+
+class RTPBuffer : public RTPStream
+{
+public:
+ RTPBuffer();
+ ~RTPBuffer();
+ void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+ WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset);
+ virtual bool EndOfFile() const;
+private:
+ RWLockWrapper* _queueRWLock;
+ std::queue<RTPPacket *> _rtpQueue;
+};
+
+class RTPFile : public RTPStream
+{
+public:
+ ~RTPFile(){}
+ RTPFile() : _rtpFile(NULL),_rtpEOF(false) {}
+ void Open(const char *outFilename, const char *mode);
+ void Close();
+ void WriteHeader();
+ void ReadHeader();
+ void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+ WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset);
+ bool EndOfFile() const { return _rtpEOF; }
+private:
+ FILE* _rtpFile;
+ bool _rtpEOF;
+};
+
+} // namespace webrtc
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc b/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc
new file mode 100644
index 0000000..e19ca44
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc
@@ -0,0 +1,246 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include <math.h>
+
+#include "common_types.h"
+#include "SpatialAudio.h"
+#include "trace.h"
+#include "testsupport/fileutils.h"
+#include "utility.h"
+
+namespace webrtc {
+
+#define NUM_PANN_COEFFS 10
+
+SpatialAudio::SpatialAudio(int testMode)
+{
+ _testMode = testMode;
+}
+
+SpatialAudio::~SpatialAudio()
+{
+ AudioCodingModule::Destroy(_acmLeft);
+ AudioCodingModule::Destroy(_acmRight);
+ AudioCodingModule::Destroy(_acmReceiver);
+ delete _channel;
+ _inFile.Close();
+ _outFile.Close();
+}
+
+WebRtc_Word16
+SpatialAudio::Setup()
+{
+ // Create ACMs and the Channel;
+ _acmLeft = AudioCodingModule::Create(1);
+ _acmRight = AudioCodingModule::Create(2);
+ _acmReceiver = AudioCodingModule::Create(3);
+ _channel = new Channel;
+
+ // Register callback for the sender side.
+ CHECK_ERROR(_acmLeft->RegisterTransportCallback(_channel));
+ CHECK_ERROR(_acmRight->RegisterTransportCallback(_channel));
+ // Register the receiver ACM in channel
+ _channel->RegisterReceiverACM(_acmReceiver);
+
+ char audioFileName[MAX_FILE_NAME_LENGTH_BYTE];
+ WebRtc_UWord16 sampFreqHz = 32000;
+
+ strncpy(audioFileName, "./test/data/audio_coding/testfile32kHz.pcm",
+ MAX_FILE_NAME_LENGTH_BYTE - 1);
+ if(_testMode == 1)
+ {
+ printf("Enter the input file [%s]: ", audioFileName);
+ PCMFile::ChooseFile(audioFileName, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
+ }
+ _inFile.Open(audioFileName, sampFreqHz, "rb", false);
+
+ if(_testMode == 0)
+ {
+ std::string outputFile = webrtc::test::OutputPath() +
+ "out_spatial_autotest.pcm";
+ strncpy(audioFileName, outputFile.c_str(),
+ MAX_FILE_NAME_LENGTH_BYTE - 1);
+ }
+ else if(_testMode == 1)
+ {
+ printf("\n");
+ std::string outputFile = webrtc::test::OutputPath() +
+ "testspatial_out.pcm";
+ strncpy(audioFileName, outputFile.c_str(),
+ MAX_FILE_NAME_LENGTH_BYTE - 1);
+ printf("Enter the output file [%s]: ", audioFileName);
+ PCMFile::ChooseFile(audioFileName, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
+ }
+ else
+ {
+ std::string outputFile = webrtc::test::OutputPath() +
+ "testspatial_out.pcm";
+ strncpy(audioFileName, outputFile.c_str(),
+ MAX_FILE_NAME_LENGTH_BYTE - 1);
+ }
+ _outFile.Open(audioFileName, sampFreqHz, "wb", false);
+ _outFile.SaveStereo(true);
+
+
+ // Register couple of codecs as receive codec
+ CodecInst codecInst;
+
+ _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
+ codecInst.channels = 2;
+ CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
+
+ _acmLeft->Codec((WebRtc_UWord8)3, codecInst);
+ codecInst.channels = 2;
+ CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
+
+ _acmLeft->Codec((WebRtc_UWord8)1, codecInst);
+ CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
+
+ _acmLeft->Codec((WebRtc_UWord8)4, codecInst);
+ CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
+
+ return 0;
+}
+
+void
+SpatialAudio::Perform()
+{
+ if(_testMode == 0)
+ {
+ printf("Running SpatialAudio Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- SpatialAudio ----------");
+ }
+
+ Setup();
+
+ CodecInst codecInst;
+ _acmLeft->Codec((WebRtc_UWord8)1, codecInst);
+ CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+ EncodeDecode();
+
+ WebRtc_Word16 pannCntr = 0;
+
+ double leftPanning[NUM_PANN_COEFFS] =
+ {1.00, 0.95, 0.90, 0.85, 0.80, 0.75, 0.70, 0.60, 0.55, 0.50};
+ double rightPanning[NUM_PANN_COEFFS] =
+ {0.50, 0.55, 0.60, 0.70, 0.75, 0.80, 0.85, 0.90, 0.95, 1.00};
+
+ while((pannCntr + 1) < NUM_PANN_COEFFS)
+ {
+ _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
+ codecInst.pacsize = 480;
+ CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+ CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+ EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+ pannCntr++;
+
+ // Change codec
+ _acmLeft->Codec((WebRtc_UWord8)3, codecInst);
+ codecInst.pacsize = 320;
+ CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+ CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+ EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+ pannCntr++;
+ if(_testMode == 0)
+ {
+ printf(".");
+ }
+ }
+
+ _acmLeft->Codec((WebRtc_UWord8)4, codecInst);
+ CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+ EncodeDecode();
+
+ _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
+ codecInst.pacsize = 480;
+ CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+ CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+ pannCntr = NUM_PANN_COEFFS -1;
+ while(pannCntr >= 0)
+ {
+ EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+ pannCntr--;
+ if(_testMode == 0)
+ {
+ printf(".");
+ }
+ }
+ if(_testMode == 0)
+ {
+ printf("Done!\n");
+ }
+}
+
+void
+SpatialAudio::EncodeDecode(
+ const double leftPanning,
+ const double rightPanning)
+{
+ AudioFrame audioFrame;
+ WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
+
+ const double rightToLeftRatio = rightPanning / leftPanning;
+
+ _channel->SetIsStereo(true);
+
+ while(!_inFile.EndOfFile())
+ {
+ _inFile.Read10MsData(audioFrame);
+ for(int n = 0; n < audioFrame._payloadDataLengthInSamples; n++)
+ {
+ audioFrame._payloadData[n] = (WebRtc_Word16)floor(
+ audioFrame._payloadData[n] * leftPanning + 0.5);
+ }
+ CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+ for(int n = 0; n < audioFrame._payloadDataLengthInSamples; n++)
+ {
+ audioFrame._payloadData[n] = (WebRtc_Word16)floor(
+ audioFrame._payloadData[n] * rightToLeftRatio + 0.5);
+ }
+ CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
+
+ CHECK_ERROR(_acmLeft->Process());
+ CHECK_ERROR(_acmRight->Process());
+
+ CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
+ _outFile.Write10MsData(audioFrame);
+ }
+ _inFile.Rewind();
+}
+
+void
+SpatialAudio::EncodeDecode()
+{
+ AudioFrame audioFrame;
+ WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
+
+ _channel->SetIsStereo(false);
+
+ while(!_inFile.EndOfFile())
+ {
+ _inFile.Read10MsData(audioFrame);
+ CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+ CHECK_ERROR(_acmLeft->Process());
+
+ CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
+ _outFile.Write10MsData(audioFrame);
+ }
+ _inFile.Rewind();
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/SpatialAudio.h b/trunk/src/modules/audio_coding/main/test/SpatialAudio.h
new file mode 100644
index 0000000..6a88327
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/SpatialAudio.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef ACM_TEST_SPATIAL_AUDIO_H
+#define ACM_TEST_SPATIAL_AUDIO_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+#include "audio_coding_module.h"
+#include "utility.h"
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+namespace webrtc {
+
+class SpatialAudio : public ACMTest
+{
+public:
+ SpatialAudio(int testMode);
+ ~SpatialAudio();
+
+ void Perform();
+private:
+ WebRtc_Word16 Setup();
+ void EncodeDecode(double leftPanning, double rightPanning);
+ void EncodeDecode();
+
+ AudioCodingModule* _acmLeft;
+ AudioCodingModule* _acmRight;
+ AudioCodingModule* _acmReceiver;
+ Channel* _channel;
+ PCMFile _inFile;
+ PCMFile _outFile;
+ int _testMode;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc
new file mode 100644
index 0000000..9d7f1e3
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -0,0 +1,872 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TestAllCodecs.h"
+
+#include <cassert>
+#include <iostream>
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "testsupport/fileutils.h"
+#include "trace.h"
+#include "utility.h"
+
+namespace webrtc {
+
+// Class for simulating packet handling
+TestPack::TestPack():
+_receiverACM(NULL),
+_seqNo(0),
+_timeStampDiff(0),
+_lastInTimestamp(0),
+_totalBytes(0),
+_payloadSize(0)
+{
+}
+TestPack::~TestPack()
+{
+}
+
+void
+TestPack::RegisterReceiverACM(AudioCodingModule* acm)
+{
+ _receiverACM = acm;
+ return;
+}
+WebRtc_Word32
+TestPack::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation)
+{
+ WebRtcRTPHeader rtpInfo;
+ WebRtc_Word32 status;
+ WebRtc_UWord16 payloadDataSize = payloadSize;
+
+ rtpInfo.header.markerBit = false;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.sequenceNumber = _seqNo++;
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.timestamp = timeStamp;
+ if(frameType == kAudioFrameCN)
+ {
+ rtpInfo.type.Audio.isCNG = true;
+ }
+ else
+ {
+ rtpInfo.type.Audio.isCNG = false;
+ }
+ if(frameType == kFrameEmpty)
+ {
+ // Skip this frame
+ return 0;
+ }
+
+ rtpInfo.type.Audio.channel = 1;
+ memcpy(_payloadData, payloadData, payloadDataSize);
+
+ status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);
+
+ _payloadSize = payloadDataSize;
+ _timeStampDiff = timeStamp - _lastInTimestamp;
+ _lastInTimestamp = timeStamp;
+ _totalBytes += payloadDataSize;
+ return status;
+}
+
+WebRtc_UWord16
+TestPack::GetPayloadSize()
+{
+ return _payloadSize;
+}
+
+
+WebRtc_UWord32
+TestPack::GetTimeStampDiff()
+{
+ return _timeStampDiff;
+}
+
+void
+TestPack::ResetPayloadSize()
+{
+ _payloadSize = 0;
+}
+
+TestAllCodecs::TestAllCodecs(int testMode):
+_acmA(NULL),
+_acmB(NULL),
+_channelA2B(NULL),
+_testCntr(0),
+_packSizeSamp(0),
+_packSizeBytes(0),
+_counter(0)
+{
+ // testMode = 0 for silent test (auto test)
+ _testMode = testMode;
+}
+
+TestAllCodecs::~TestAllCodecs()
+{
+ if(_acmA != NULL)
+ {
+ AudioCodingModule::Destroy(_acmA);
+ _acmA = NULL;
+ }
+ if(_acmB != NULL)
+ {
+ AudioCodingModule::Destroy(_acmB);
+ _acmB = NULL;
+ }
+ if(_channelA2B != NULL)
+ {
+ delete _channelA2B;
+ _channelA2B = NULL;
+ }
+}
+
+void TestAllCodecs::Perform()
+{
+
+ char file[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _inFileA.Open(file, 32000, "rb");
+
+ if(_testMode == 0)
+ {
+ printf("Running All Codecs Test");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TestAllCodecs ----------");
+ }
+
+ _acmA = AudioCodingModule::Create(0);
+ _acmB = AudioCodingModule::Create(1);
+
+ _acmA->InitializeReceiver();
+ _acmB->InitializeReceiver();
+
+ WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+ CodecInst myCodecParam;
+
+ for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+ {
+ _acmB->Codec(n, myCodecParam);
+ _acmB->RegisterReceiveCodec(myCodecParam);
+ }
+
+ // Create and connect the channel
+ _channelA2B = new TestPack;
+ _acmA->RegisterTransportCallback(_channelA2B);
+ _channelA2B->RegisterReceiverACM(_acmB);
+
+ // All codecs are tested for all allowed sampling frequencies, rates and packet sizes
+#ifdef WEBRTC_CODEC_GSMAMR
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecAMR[] = "AMR";
+ RegisterSendCodec('A', codecAMR, 8000, 4750, 160, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 4750, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 4750, 480, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5150, 160, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5150, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5150, 480, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5900, 160, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5900, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 5900, 480, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 6700, 160, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 6700, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 6700, 480, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7400, 160, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7400, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7400, 480, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7950, 160, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7950, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 7950, 480, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 10200, 160, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 10200, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 10200, 480, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 12200, 160, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 12200, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMR, 8000, 12200, 480, 3);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ char codecAMRWB[] = "AMR-WB";
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecAMRWB, 16000, 7000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 7000, 640, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 7000, 960, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 9000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 9000, 640, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 9000, 960, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 12000, 320, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 12000, 640, 6);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 12000, 960, 8);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 14000, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 14000, 640, 4);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 14000, 960, 5);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 16000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 16000, 640, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 16000, 960, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 18000, 320, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 18000, 640, 4);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 18000, 960, 5);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 20000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 20000, 640, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 20000, 960, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 23000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 23000, 640, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 23000, 960, 3);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 24000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 24000, 640, 2);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecAMRWB, 16000, 24000, 960, 2);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_G722
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG722[] = "G722";
+ RegisterSendCodec('A', codecG722, 16000, 64000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 480, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 640, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 800, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 960, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG7221_1[] = "G7221";
+ RegisterSendCodec('A', codecG7221_1, 16000, 32000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7221_1, 16000, 24000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7221_1, 16000, 16000, 320, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG7221_2[] = "G7221";
+ RegisterSendCodec('A', codecG7221_2, 32000, 48000, 640, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7221_2, 32000, 32000, 640, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7221_2, 32000, 24000, 640, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_G729
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG729[] = "G729";
+ RegisterSendCodec('A', codecG729, 8000, 8000, 80, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG729, 8000, 8000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG729, 8000, 8000, 240, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG729, 8000, 8000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG729, 8000, 8000, 400, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG729, 8000, 8000, 480, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG7291[] = "G7291";
+ RegisterSendCodec('A', codecG7291, 16000, 8000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 8000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 8000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 12000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 12000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 12000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 14000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 14000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 14000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 16000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 16000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 16000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 18000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 18000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 18000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 20000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 20000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 20000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 22000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 22000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 22000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 24000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 24000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 24000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 26000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 26000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 26000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 28000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 28000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 28000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 30000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 30000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 30000, 960, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 32000, 320, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 32000, 640, 1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecG7291, 16000, 32000, 960, 1);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecGSM[] = "GSM";
+ RegisterSendCodec('A', codecGSM, 8000, 13200, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecGSM, 8000, 13200, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecGSM, 8000, 13200, 480, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecILBC[] = "ILBC";
+ RegisterSendCodec('A', codecILBC, 8000, 13300, 240, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecILBC, 8000, 13300, 480, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecILBC, 8000, 15200, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecILBC, 8000, 15200, 320, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecISAC[] = "ISAC";
+ RegisterSendCodec('A', codecISAC, 16000, -1, 480, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 16000, -1, 960, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 16000, 15000, 480, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 16000, 32000, 960, -1);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecISAC, 32000, -1, 960, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 32000, 56000, 960, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 32000, 37000, 960, -1);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecISAC, 32000, 32000, 960, -1);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecL16[] = "L16";
+ RegisterSendCodec('A', codecL16, 8000, 128000, 80, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 240, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 320, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 480, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 640, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 640, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecPCMA[] = "PCMA";
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 240, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 400, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 480, 0);
+ Run(_channelA2B);
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ char codecPCMU[] = "PCMU";
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 240, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 400, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 480, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#ifdef WEBRTC_CODEC_SPEEX
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecSPEEX[] = "SPEEX";
+ RegisterSendCodec('A', codecSPEEX, 8000, 2400, 160, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecSPEEX, 8000, 8000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecSPEEX, 8000, 18200, 480, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecSPEEX, 16000, 4000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecSPEEX, 16000, 12800, 640, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecSPEEX, 16000, 34200, 960, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecCELT_32[] = "CELT";
+ RegisterSendCodec('A', codecCELT_32, 32000, 48000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecCELT_32, 32000, 64000, 320, 0);
+ Run(_channelA2B);
+ RegisterSendCodec('A', codecCELT_32, 32000, 128000, 320, 0);
+ Run(_channelA2B);
+ _outFileB.Close();
+#endif
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ } else {
+ printf("Done!\n");
+ }
+
+ /* Print out all codecs that were not tested in the run */
+ if(_testMode != 0) {
+ printf("The following codecs was not included in the test:\n");
+#ifndef WEBRTC_CODEC_GSMAMR
+ printf(" GSMAMR\n");
+#endif
+#ifndef WEBRTC_CODEC_GSMAMRWB
+ printf(" GSMAMR-wb\n");
+#endif
+#ifndef WEBRTC_CODEC_G722
+ printf(" G.722\n");
+#endif
+#ifndef WEBRTC_CODEC_G722_1
+ printf(" G.722.1\n");
+#endif
+#ifndef WEBRTC_CODEC_G722_1C
+ printf(" G.722.1C\n");
+#endif
+#ifndef WEBRTC_CODEC_G729
+ printf(" G.729\n");
+#endif
+#ifndef WEBRTC_CODEC_G729_1
+ printf(" G.729.1\n");
+#endif
+#ifndef WEBRTC_CODEC_GSMFR
+ printf(" GSMFR\n");
+#endif
+#ifndef WEBRTC_CODEC_ILBC
+ printf(" iLBC\n");
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+ printf(" ISAC float\n");
+#endif
+#ifndef WEBRTC_CODEC_ISACFX
+ printf(" ISAC fix\n");
+#endif
+#ifndef WEBRTC_CODEC_PCM16
+ printf(" PCM16\n");
+#endif
+#ifndef WEBRTC_CODEC_SPEEX
+ printf(" Speex\n");
+#endif
+
+ printf("\nTo complete the test, listen to the %d number of output files.\n", _testCntr);
+ }
+}
+
+// Register Codec to use in the test
+//
+// Input: side - which ACM to use, 'A' or 'B'
+// codecName - name to use when register the codec
+// samplingFreqHz - sampling frequency in Herz
+// rate - bitrate in bytes
+// packSize - packet size in samples
+// extraByte - if extra bytes needed compared to the bitrate
+// used when registering, can be an internal header
+// set to -1 if the codec is a variable rate codec
+WebRtc_Word16 TestAllCodecs::RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 samplingFreqHz,
+ int rate,
+ int packSize,
+ int extraByte)
+{
+ if(_testMode != 0) {
+ // Print out codec and settings
+ printf("codec: %s Freq: %d Rate: %d PackSize: %d", codecName, samplingFreqHz, rate, packSize);
+ }
+
+ // Store packetsize in samples, used to validate the recieved packet
+ _packSizeSamp = packSize;
+
+ // Store the expected packet size in bytes, used to validate the recieved packet
+ // If variable rate codec (extraByte == -1), set to -1 (65535)
+ if (extraByte != -1)
+ {
+ // Add 0.875 to always round up to a whole byte
+ _packSizeBytes = (WebRtc_UWord16)((float)(packSize*rate)/(float)(samplingFreqHz*8)+0.875)+extraByte;
+ }
+ else
+ {
+ // Packets will have a variable size
+ _packSizeBytes = -1;
+ }
+
+ // Set pointer to the ACM where to register the codec
+ AudioCodingModule* myACM;
+ switch(side)
+ {
+ case 'A':
+ {
+ myACM = _acmA;
+ break;
+ }
+ case 'B':
+ {
+ myACM = _acmB;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(myACM == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ CodecInst myCodecParam;
+
+ // Get all codec paramters before registering
+ CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
+ myCodecParam.rate = rate;
+ myCodecParam.pacsize = packSize;
+ CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
+
+ // initialization was succesful
+ return 0;
+}
+
+void TestAllCodecs::Run(TestPack* channel)
+{
+ AudioFrame audioFrame;
+
+ WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+ WebRtc_UWord16 recSize;
+ WebRtc_UWord32 timeStampDiff;
+ channel->ResetPayloadSize();
+ int errorCount = 0;
+
+ // Only run 1 second for each test case
+ while((_counter<1000)&& (!_inFileA.EndOfFile()))
+ {
+ // Add 10 msec to ACM
+ _inFileA.Read10MsData(audioFrame);
+ CHECK_ERROR(_acmA->Add10MsData(audioFrame));
+
+ // Run sender side of ACM
+ CHECK_ERROR(_acmA->Process());
+
+ // Verify that the received packet size matches the settings
+ recSize = channel->GetPayloadSize();
+ if (recSize) {
+ if ((recSize != _packSizeBytes) && (_packSizeBytes < 65535)) {
+ errorCount++;
+ }
+
+ // Verify that the timestamp is updated with expected length
+ timeStampDiff = channel->GetTimeStampDiff();
+ if ((_counter > 10) && (timeStampDiff != _packSizeSamp))
+ errorCount++;
+ }
+
+
+ // Run received side of ACM
+ CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
+
+ // Write output speech to file
+ _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
+ }
+
+ if (errorCount)
+ {
+ printf(" - test FAILED\n");
+ }
+ else if(_testMode != 0)
+ {
+ printf(" - test PASSED\n");
+ }
+
+ // Reset _counter
+ if (_counter == 1000) {
+ _counter = 0;
+ }
+ if (_inFileA.EndOfFile()) {
+ _inFileA.Rewind();
+ }
+}
+
+void TestAllCodecs::OpenOutFile(WebRtc_Word16 testNumber)
+{
+ char fileName[500];
+ sprintf(fileName, "%s/testallcodecs_out_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ _outFileB.Open(fileName, 32000, "wb");
+}
+
+void TestAllCodecs::DisplaySendReceiveCodec()
+{
+ CodecInst myCodecParam;
+ _acmA->SendCodec(myCodecParam);
+ printf("%s -> ", myCodecParam.plname);
+ _acmB->ReceiveCodec(myCodecParam);
+ printf("%s\n", myCodecParam.plname);
+}
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h
new file mode 100644
index 0000000..e0d621f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_ALL_CODECS_H
+#define TEST_ALL_CODECS_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+
+namespace webrtc {
+
+class TestPack : public AudioPacketizationCallback
+{
+public:
+ TestPack();
+ ~TestPack();
+
+ void RegisterReceiverACM(AudioCodingModule* acm);
+
+ virtual WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ WebRtc_UWord16 GetPayloadSize();
+ WebRtc_UWord32 GetTimeStampDiff();
+ void ResetPayloadSize();
+
+private:
+ AudioCodingModule* _receiverACM;
+ WebRtc_Word16 _seqNo;
+ WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
+ WebRtc_UWord32 _timeStampDiff;
+ WebRtc_UWord32 _lastInTimestamp;
+ WebRtc_UWord64 _totalBytes;
+ WebRtc_UWord16 _payloadSize;
+};
+
+class TestAllCodecs : public ACMTest
+{
+public:
+ TestAllCodecs(int testMode);
+ ~TestAllCodecs();
+
+ void Perform();
+private:
+ // The default value of '-1' indicates that the registration is based only on codec name
+ // and a sampling frequncy matching is not required. This is useful for codecs which support
+ // several sampling frequency.
+ WebRtc_Word16 RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 sampFreqHz,
+ int rate,
+ int packSize,
+ int extraByte);
+
+ void Run(TestPack* channel);
+ void OpenOutFile(WebRtc_Word16 testNumber);
+ void DisplaySendReceiveCodec();
+
+ WebRtc_Word32 SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ int _testMode;
+
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ TestPack* _channelA2B;
+
+ PCMFile _inFileA;
+ PCMFile _outFileB;
+ WebRtc_Word16 _testCntr;
+ WebRtc_UWord16 _packSizeSamp;
+ WebRtc_UWord16 _packSizeBytes;
+ int _counter;
+};
+
+#endif // TEST_ALL_CODECS_H
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestFEC.cc b/trunk/src/modules/audio_coding/main/test/TestFEC.cc
new file mode 100644
index 0000000..81ca91a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestFEC.cc
@@ -0,0 +1,627 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TestFEC.h"
+
+#include <cassert>
+#include <iostream>
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "trace.h"
+#include "testsupport/fileutils.h"
+#include "utility.h"
+
+namespace webrtc {
+
+TestFEC::TestFEC(int testMode):
+_acmA(NULL),
+_acmB(NULL),
+_channelA2B(NULL),
+_testCntr(0)
+{
+ _testMode = testMode;
+}
+
+TestFEC::~TestFEC()
+{
+ if(_acmA != NULL)
+ {
+ AudioCodingModule::Destroy(_acmA);
+ _acmA = NULL;
+ }
+ if(_acmB != NULL)
+ {
+ AudioCodingModule::Destroy(_acmB);
+ _acmB = NULL;
+ }
+ if(_channelA2B != NULL)
+ {
+ delete _channelA2B;
+ _channelA2B = NULL;
+ }
+}
+
+void TestFEC::Perform()
+{
+
+ if(_testMode == 0)
+ {
+ printf("Running FEC Test");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TestFEC ----------");
+ }
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _inFileA.Open(fileName, 32000, "rb");
+
+
+ bool fecEnabled;
+
+ _acmA = AudioCodingModule::Create(0);
+ _acmB = AudioCodingModule::Create(1);
+
+ _acmA->InitializeReceiver();
+ _acmB->InitializeReceiver();
+
+ WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+ CodecInst myCodecParam;
+ if(_testMode != 0)
+ {
+ printf("Registering codecs at receiver... \n");
+ }
+ for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+ {
+ _acmB->Codec(n, myCodecParam);
+ if(_testMode != 0)
+ {
+ printf("%s\n", myCodecParam.plname);
+ }
+ _acmB->RegisterReceiveCodec(myCodecParam);
+ }
+
+ // Create and connect the channel
+ _channelA2B = new Channel;
+ _acmA->RegisterTransportCallback(_channelA2B);
+ _channelA2B->RegisterReceiverACM(_acmB);
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+#ifndef WEBRTC_CODEC_G722
+ printf("G722 needs to be activated to run this test\n");
+ exit(-1);
+#endif
+ char nameG722[] = "G722";
+ RegisterSendCodec('A', nameG722, 16000);
+ char nameCN[] = "CN";
+ RegisterSendCodec('A', nameCN, 16000);
+ char nameRED[] = "RED";
+ RegisterSendCodec('A', nameRED);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ char nameISAC[] = "iSAC";
+ RegisterSendCodec('A',nameISAC, 16000);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADVeryAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+
+ RegisterSendCodec('A', nameISAC, 32000);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADVeryAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+
+ RegisterSendCodec('A', nameISAC, 32000);
+ OpenOutFile(_testCntr);
+ SetVAD(false, false, VADNormal);
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+
+ RegisterSendCodec('A', nameISAC, 16000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+ RegisterSendCodec('A', nameISAC, 32000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+ RegisterSendCodec('A', nameISAC, 16000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+
+
+
+
+ _channelA2B->SetFECTestWithPacketLoss(true);
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+
+ RegisterSendCodec('A',nameG722);
+ RegisterSendCodec('A', nameCN, 16000);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ RegisterSendCodec('A', nameISAC, 16000);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADVeryAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ RegisterSendCodec('A', nameISAC, 32000);
+ OpenOutFile(_testCntr);
+ SetVAD(true, true, VADVeryAggr);
+ _acmA->SetFECStatus(false);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+
+
+
+
+
+
+ if(_testMode != 0)
+ {
+ printf("=======================================================================\n");
+ printf("%d ",_testCntr++);
+ }
+ else
+ {
+ printf(".");
+ }
+ RegisterSendCodec('A', nameISAC, 32000);
+ OpenOutFile(_testCntr);
+ SetVAD(false, false, VADNormal);
+ _acmA->SetFECStatus(true);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+
+ RegisterSendCodec('A', nameISAC, 16000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+ RegisterSendCodec('A', nameISAC, 32000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+
+ RegisterSendCodec('A', nameISAC, 16000);
+ fecEnabled = _acmA->FECStatus();
+ if(_testMode != 0)
+ {
+ printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
+ DisplaySendReceiveCodec();
+ }
+ Run();
+ _outFileB.Close();
+
+
+
+ if(_testMode == 0)
+ {
+ printf("Done!\n");
+ }
+}
+
+WebRtc_Word32 TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode)
+{
+ if(_testMode != 0)
+ {
+ printf("DTX %s; VAD %s; VAD-Mode %d\n",
+ enableDTX? "ON":"OFF",
+ enableVAD? "ON":"OFF",
+ (WebRtc_Word16)vadMode);
+ }
+ return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
+}
+
+WebRtc_Word16 TestFEC::RegisterSendCodec(char side, char* codecName, WebRtc_Word32 samplingFreqHz)
+{
+ if(_testMode != 0)
+ {
+ if(samplingFreqHz > 0)
+ {
+ printf("Registering %s-%d for side %c\n", codecName, samplingFreqHz, side);
+ }
+ else
+ {
+ printf("Registering %s for side %c\n", codecName, side);
+ }
+ }
+ std::cout << std::flush;
+ AudioCodingModule* myACM;
+ switch(side)
+ {
+ case 'A':
+ {
+ myACM = _acmA;
+ break;
+ }
+ case 'B':
+ {
+ myACM = _acmB;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(myACM == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ CodecInst myCodecParam;
+
+ CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
+
+ CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
+
+ // initialization was succesful
+ return 0;
+}
+
+void TestFEC::Run()
+{
+ AudioFrame audioFrame;
+
+ WebRtc_UWord16 msecPassed = 0;
+ WebRtc_UWord32 secPassed = 0;
+ WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+
+ while(!_inFileA.EndOfFile())
+ {
+ _inFileA.Read10MsData(audioFrame);
+ CHECK_ERROR(_acmA->Add10MsData(audioFrame));
+ CHECK_ERROR(_acmA->Process());
+ CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
+ _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
+ msecPassed += 10;
+ if(msecPassed >= 1000)
+ {
+ msecPassed = 0;
+ secPassed++;
+ }
+ if(((secPassed%5) == 4) && (msecPassed == 0) && (_testCntr > 14))
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ _acmA->SetFECStatus(false);
+ printf("FEC currently %s\n",(_acmA->FECStatus()?"ON":"OFF"));
+ }
+ if(((secPassed%5) == 4) && (msecPassed >= 990) && (_testCntr > 14))
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ _acmA->SetFECStatus(true);
+ printf("FEC currently %s\n",(_acmA->FECStatus()?"ON":"OFF"));
+ }
+ }
+ _inFileA.Rewind();
+}
+
+void TestFEC::OpenOutFile(WebRtc_Word16 testNumber)
+{
+ char fileName[500];
+
+ if(_testMode == 0)
+ {
+ sprintf(fileName, "%s/TestFEC_autoFile_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ }
+ else
+ {
+ sprintf(fileName, "%s/TestFEC_outFile_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ }
+ _outFileB.Open(fileName, 32000, "wb");
+}
+
+void TestFEC::DisplaySendReceiveCodec()
+{
+ CodecInst myCodecParam;
+ _acmA->SendCodec(myCodecParam);
+ printf("%s -> ", myCodecParam.plname);
+ _acmB->ReceiveCodec(myCodecParam);
+ printf("%s\n", myCodecParam.plname);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestFEC.h b/trunk/src/modules/audio_coding/main/test/TestFEC.h
new file mode 100644
index 0000000..00e951f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestFEC.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FEC_H
+#define TEST_FEC_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+
+namespace webrtc {
+
+class TestFEC : public ACMTest
+{
+public:
+ TestFEC(int testMode);
+ ~TestFEC();
+
+ void Perform();
+private:
+ // The default value of '-1' indicates that the registration is based only on codec name
+ // and a sampling frequncy matching is not required. This is useful for codecs which support
+ // several sampling frequency.
+ WebRtc_Word16 RegisterSendCodec(char side, char* codecName, WebRtc_Word32 sampFreqHz = -1);
+ void Run();
+ void OpenOutFile(WebRtc_Word16 testNumber);
+ void DisplaySendReceiveCodec();
+ WebRtc_Word32 SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ Channel* _channelA2B;
+
+ PCMFile _inFileA;
+ PCMFile _outFileB;
+ WebRtc_Word16 _testCntr;
+ int _testMode;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TestStereo.cc b/trunk/src/modules/audio_coding/main/test/TestStereo.cc
new file mode 100644
index 0000000..2105d36
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestStereo.cc
@@ -0,0 +1,804 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TestStereo.h"
+
+#include <cassert>
+#include <iostream>
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "testsupport/fileutils.h"
+#include "trace.h"
+#include "utility.h"
+
+namespace webrtc {
+
+// Class for simulating packet handling
+TestPackStereo::TestPackStereo():
+_receiverACM(NULL),
+_seqNo(0),
+_timeStampDiff(0),
+_lastInTimestamp(0),
+_totalBytes(0),
+_payloadSize(0),
+_noChannels(1),
+_codecType(0)
+{
+}
+TestPackStereo::~TestPackStereo()
+{
+}
+
+void
+TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm)
+{
+ _receiverACM = acm;
+ return;
+}
+
+
+WebRtc_Word32
+TestPackStereo::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation)
+{
+ WebRtcRTPHeader rtpInfo;
+ WebRtc_Word32 status;
+ WebRtc_UWord16 payloadDataSize = payloadSize;
+ WebRtc_UWord8 payloadDataMaster[60 * 32 * 2 * 2];
+ WebRtc_UWord8 payloadDataSlave[60 * 32 * 2 * 2];
+
+ rtpInfo.header.markerBit = false;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.sequenceNumber = _seqNo++;
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.timestamp = timeStamp;
+ if(frameType == kFrameEmpty)
+ {
+ // Skip this frame
+ return 0;
+ }
+ if(frameType != kAudioFrameCN)
+ {
+ rtpInfo.type.Audio.isCNG = false;
+
+ // For stereo we need to call ACM with two incoming packets, one for each channel.
+ // Different packet-splitting depending on codec.
+ if (_codecType == 0) {
+ // one byte per sample
+ for (int i=0, j=0; i<payloadDataSize; i+=2, j++)
+ {
+ payloadDataMaster[j] = payloadData[i];
+ payloadDataSlave[j] = payloadData[i+1];
+ }
+ } else if (_codecType == 1) {
+ // two bytes per sample
+ for (int i=0, j=0; i<payloadDataSize; i+=4, j+=2)
+ {
+ payloadDataMaster[j] = payloadData[i];
+ payloadDataMaster[j+1] = payloadData[i+1];
+ payloadDataSlave[j] = payloadData[i+2];
+ payloadDataSlave[j+1] = payloadData[i+3];
+ }
+ } else if (_codecType == 2) {
+ // frameBased
+ memcpy(payloadDataMaster, &payloadData[0], payloadDataSize/2);
+ memcpy(payloadDataSlave, &payloadData[payloadDataSize/2], payloadDataSize/2);
+ } else if (_codecType == 3) {
+ // four bits per sample
+ for (int i=0, j=0; i<payloadDataSize; i+=2, j++)
+ {
+ payloadDataMaster[j] = (payloadData[i] & 0xF0) + (payloadData[i+1] >> 4);
+ payloadDataSlave[j] = ((payloadData[i] & 0x0F) << 4) + (payloadData[i+1] & 0x0F);
+ }
+ } else if (_codecType == 4) {
+ // True stereo, call both master and slave with whole stream.
+ memcpy(payloadDataMaster, payloadData, payloadSize);
+ memcpy(payloadDataSlave, payloadData, payloadSize);
+ payloadDataSize = payloadSize*2;
+ }
+ }
+ else
+ {
+ // If CNG packet, send the same packet to both master and slave.
+ rtpInfo.type.Audio.isCNG = true;
+ memcpy(payloadDataMaster, payloadData, payloadSize);
+ memcpy(payloadDataSlave, payloadData, payloadSize);
+ payloadDataSize = payloadSize*2;
+ }
+
+ if (_codecType != 5) {
+ // Call ACM with two packets, one for each channel
+ rtpInfo.type.Audio.channel = 1;
+ status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadDataMaster, payloadDataSize/2, rtpInfo);
+ rtpInfo.type.Audio.channel = 2;
+ status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadDataSlave, payloadDataSize/2, rtpInfo);
+ } else {
+ // Mono case, call ACM with one packet.
+ rtpInfo.type.Audio.channel = 1;
+ status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadData, payloadDataSize, rtpInfo);
+ }
+
+ if (frameType != kAudioFrameCN) {
+ _payloadSize = payloadDataSize;
+ } else {
+ _payloadSize = -1;
+ }
+ _timeStampDiff = timeStamp - _lastInTimestamp;
+ _lastInTimestamp = timeStamp;
+ _totalBytes += payloadDataSize;
+ return status;
+}
+
+WebRtc_UWord16
+TestPackStereo::GetPayloadSize()
+{
+ return _payloadSize;
+}
+
+
+WebRtc_UWord32
+TestPackStereo::GetTimeStampDiff()
+{
+ return _timeStampDiff;
+}
+
+void
+TestPackStereo::ResetPayloadSize()
+{
+ _payloadSize = 0;
+}
+
+void
+TestPackStereo::SetCodecType(int codecType)
+{
+ _codecType = codecType;
+}
+
+TestStereo::TestStereo(int testMode):
+_acmA(NULL),
+_acmB(NULL),
+_channelA2B(NULL),
+_testCntr(0),
+_packSizeSamp(0),
+_packSizeBytes(0),
+_counter(0)
+{
+ // testMode = 0 for silent test (auto test)
+ _testMode = testMode;
+}
+
+TestStereo::~TestStereo()
+{
+ if(_acmA != NULL)
+ {
+ AudioCodingModule::Destroy(_acmA);
+ _acmA = NULL;
+ }
+ if(_acmB != NULL)
+ {
+ AudioCodingModule::Destroy(_acmB);
+ _acmB = NULL;
+ }
+ if(_channelA2B != NULL)
+ {
+ delete _channelA2B;
+ _channelA2B = NULL;
+ }
+}
+
+void TestStereo::Perform()
+{
+ char file_name_stereo[500];
+ char file_name_mono[500];
+ WebRtc_UWord16 frequencyHz;
+ int audio_channels;
+ int codec_channels;
+
+ if(_testMode == 0)
+ {
+ printf("Running Stereo Test");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TestStereo ----------");
+ }
+
+ strcpy(file_name_stereo, "./test/data/audio_coding/teststereo32kHz.pcm");
+ strcpy(file_name_mono, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 32000;
+
+ _in_file_stereo.Open(file_name_stereo, frequencyHz, "rb");
+ _in_file_stereo.ReadStereo(true);
+ _in_file_mono.Open(file_name_mono, frequencyHz, "rb");
+ _in_file_mono.ReadStereo(false);
+
+ _acmA = AudioCodingModule::Create(0);
+ _acmB = AudioCodingModule::Create(1);
+
+ _acmA->InitializeReceiver();
+ _acmB->InitializeReceiver();
+
+ WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+ CodecInst myCodecParam;
+
+ // Register receiving codecs as stereo.
+ for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+ {
+ _acmB->Codec(n, myCodecParam);
+ if(!strcmp(myCodecParam.plname, "L16") ||
+ !strcmp(myCodecParam.plname, "PCMA")||
+ !strcmp(myCodecParam.plname, "PCMU")||
+ !strcmp(myCodecParam.plname, "G722")||
+ !strcmp(myCodecParam.plname, "CELT"))
+ {
+ myCodecParam.channels=2;
+ _acmB->RegisterReceiveCodec(myCodecParam);
+ }
+ }
+
+ // Create and connect the channel.
+ _channelA2B = new TestPackStereo;
+ _acmA->RegisterTransportCallback(_channelA2B);
+ _channelA2B->RegisterReceiverACM(_acmB);
+
+ //
+ // Test Stereo-To-Stereo for all codecs.
+ //
+ audio_channels = 2;
+ codec_channels = 2;
+
+ // All codecs are tested for all allowed sampling frequencies, rates and packet sizes
+#ifdef WEBRTC_CODEC_G722
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _channelA2B->SetCodecType(3);
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecG722[] = "G722";
+ RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 480, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 640, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 800, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 960, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _channelA2B->SetCodecType(1);
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecL16[] = "L16";
+ RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 240, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 480, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 640, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 640, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+#endif
+#define PCMA_AND_PCMU
+#ifdef PCMA_AND_PCMU
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _channelA2B->SetCodecType(0);
+ audio_channels = 2;
+ codec_channels = 2;
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecPCMA[] = "PCMA";
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 240, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 400, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 480, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecPCMU[] = "PCMU";
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 240, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 400, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 480, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ if(_testMode != 0) {
+ printf("=======================================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ } else {
+ printf(".");
+ }
+ _channelA2B->SetCodecType(4);
+ audio_channels = 2;
+ codec_channels = 2;
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ char codecCELT[] = "CELT";
+ RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecCELT, 32000, 128000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(true, true, VADNormal);
+ RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _acmA->SetVAD(false, false, VADNormal);
+ _outFileB.Close();
+#endif
+ //
+ // Test Mono-To-Stereo for all codecs.
+ //
+ audio_channels = 1;
+ codec_channels = 2;
+
+#ifdef WEBRTC_CODEC_G722
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ _channelA2B->SetCodecType(3);
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ _channelA2B->SetCodecType(1);
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef PCMA_AND_PCMU
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ _channelA2B->SetCodecType(0);
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+ _testCntr++;
+ _channelA2B->SetCodecType(4);
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+
+ //
+ // Test Stereo-To-Mono for all codecs.
+ //
+ audio_channels = 2;
+ codec_channels = 1;
+ _channelA2B->SetCodecType(5);
+
+ // Register receivers as mono.
+ for(WebRtc_UWord8 n = 0; n < numEncoders; n++) {
+ _acmB->Codec(n, myCodecParam);
+ if(!strcmp(myCodecParam.plname, "L16") ||
+ !strcmp(myCodecParam.plname, "PCMA")||
+ !strcmp(myCodecParam.plname, "PCMU")||
+ !strcmp(myCodecParam.plname, "G722")||
+ !strcmp(myCodecParam.plname, "CELT")) {
+ myCodecParam.channels = 1;
+ _acmB->RegisterReceiveCodec(myCodecParam);
+ }
+ }
+#ifdef WEBRTC_CODEC_G722
+ // Run stereo audio and mono codec.
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef PCMA_AND_PCMU
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ if(_testMode != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n",_testCntr + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
+ _testCntr++;
+ OpenOutFile(_testCntr);
+ RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
+ Run(_channelA2B, audio_channels, codec_channels);
+ _outFileB.Close();
+#endif
+
+ // Print out which codecs were tested, and which were not, in the run.
+ if(_testMode != 0) {
+ printf("\nThe following codecs was INCLUDED in the test:\n");
+#ifdef WEBRTC_CODEC_G722
+ printf(" G.722\n");
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ printf(" PCM16\n");
+#endif
+ printf(" G.711\n");
+
+ printf("\nTo complete the test, listen to the %d number of output files.\n", _testCntr);
+ } else {
+ printf("Done!\n");
+ }
+}
+
+// Register Codec to use in the test
+//
+// Input: side - which ACM to use, 'A' or 'B'
+// codecName - name to use when register the codec
+// samplingFreqHz - sampling frequency in Herz
+// rate - bitrate in bytes
+// packSize - packet size in samples
+// extraByte - if extra bytes needed compared to the bitrate
+// used when registering, can be an internal header
+// set to -1 if the codec is a variable rate codec
+WebRtc_Word16 TestStereo::RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 samplingFreqHz,
+ int rate,
+ int packSize,
+ int channels)
+{
+ if(_testMode != 0) {
+ // Print out codec and settings
+ printf("Codec: %s Freq: %d Rate: %d PackSize: %d", codecName, samplingFreqHz, rate, packSize);
+ }
+
+ // Store packetsize in samples, used to validate the recieved packet
+ _packSizeSamp = packSize;
+
+ // Store the expected packet size in bytes, used to validate the recieved packet
+ // Add 0.875 to always round up to a whole byte
+ _packSizeBytes = (WebRtc_UWord16)((float)(packSize*rate)/(float)(samplingFreqHz*8)+0.875);
+
+ // Set pointer to the ACM where to register the codec
+ AudioCodingModule* myACM;
+ switch(side)
+ {
+ case 'A':
+ {
+ myACM = _acmA;
+ break;
+ }
+ case 'B':
+ {
+ myACM = _acmB;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(myACM == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ CodecInst myCodecParam;
+
+ // Get all codec parameters before registering
+ CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
+ myCodecParam.rate = rate;
+ myCodecParam.pacsize = packSize;
+ // Start with register codec as mono, to test that changing to stereo works.
+ myCodecParam.channels = 1;
+ CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
+ // Register codec as stereo.
+ if (channels == 2) {
+ myCodecParam.channels = 2;
+ CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
+ }
+ // Initialization was successful.
+ return 0;
+}
+
+void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels)
+{
+ AudioFrame audioFrame;
+
+ WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+ WebRtc_UWord16 recSize;
+ WebRtc_UWord32 timeStampDiff;
+ channel->ResetPayloadSize();
+ int errorCount = 0;
+
+ // Only run 1 second for each test case
+ // TODO(tlegrand): either remove |_counter| or start using it as the comment
+ // above says. Now |_counter| is always 0.
+ while(_counter<1000)
+ {
+ // Add 10 msec to ACM
+ if (in_channels == 1) {
+ if (_in_file_mono.EndOfFile()) {
+ break;
+ }
+ _in_file_mono.Read10MsData(audioFrame);
+ } else {
+ if (_in_file_stereo.EndOfFile()) {
+ break;
+ }
+ _in_file_stereo.Read10MsData(audioFrame);
+ }
+ CHECK_ERROR(_acmA->Add10MsData(audioFrame));
+
+ // Run sender side of ACM
+ CHECK_ERROR(_acmA->Process());
+
+ // Verify that the received packet size matches the settings
+ recSize = channel->GetPayloadSize();
+ if ((0 < recSize) & (recSize < 65535)) {
+ if ((recSize != _packSizeBytes * out_channels) &&
+ (_packSizeBytes < 65535)) {
+ errorCount++;
+ }
+
+ // Verify that the timestamp is updated with expected length
+ timeStampDiff = channel->GetTimeStampDiff();
+ if ((_counter > 10) && (timeStampDiff != _packSizeSamp)) {
+ errorCount++;
+ }
+ }
+
+ // Run received side of ACM
+ CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
+
+ // Write output speech to file
+ _outFileB.Write10MsData(
+ audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel);
+ }
+
+ if (errorCount)
+ {
+ printf(" - test FAILED\n");
+ }
+ else if(_testMode != 0)
+ {
+ printf(" - test PASSED\n");
+ }
+
+ // Reset _counter
+ if (_counter == 1000) {
+ _counter = 0;
+ }
+ if (_in_file_mono.EndOfFile()) {
+ _in_file_mono.Rewind();
+ }
+ if (_in_file_stereo.EndOfFile()) {
+ _in_file_stereo.Rewind();
+ }
+}
+
+void TestStereo::OpenOutFile(WebRtc_Word16 testNumber)
+{
+ char fileName[500];
+ sprintf(fileName, "%s/teststereo_out_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ _outFileB.Open(fileName, 32000, "wb");
+}
+
+void TestStereo::DisplaySendReceiveCodec()
+{
+ CodecInst myCodecParam;
+ _acmA->SendCodec(myCodecParam);
+ if(_testMode != 0) {
+ printf("%s -> ", myCodecParam.plname);
+ }
+ _acmB->ReceiveCodec(myCodecParam);
+ if(_testMode != 0) {
+ printf("%s\n", myCodecParam.plname);
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestStereo.h b/trunk/src/modules/audio_coding/main/test/TestStereo.h
new file mode 100644
index 0000000..94a0c56
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestStereo.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_STEREO_H
+#define TEST_STEREO_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+
+namespace webrtc {
+
+class TestPackStereo : public AudioPacketizationCallback
+{
+public:
+ TestPackStereo();
+ ~TestPackStereo();
+
+ void RegisterReceiverACM(AudioCodingModule* acm);
+
+ virtual WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ WebRtc_UWord16 GetPayloadSize();
+ WebRtc_UWord32 GetTimeStampDiff();
+ void ResetPayloadSize();
+ void SetCodecType(int codecType);
+
+
+private:
+ AudioCodingModule* _receiverACM;
+ WebRtc_Word16 _seqNo;
+ WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
+ WebRtc_UWord32 _timeStampDiff;
+ WebRtc_UWord32 _lastInTimestamp;
+ WebRtc_UWord64 _totalBytes;
+ WebRtc_UWord16 _payloadSize;
+ WebRtc_UWord16 _noChannels;
+ int _codecType;
+};
+
+class TestStereo : public ACMTest
+{
+public:
+ TestStereo(int testMode);
+ ~TestStereo();
+
+ void Perform();
+private:
+ // The default value of '-1' indicates that the registration is based only on codec name
+ // and a sampling frequncy matching is not required. This is useful for codecs which support
+ // several sampling frequency.
+ WebRtc_Word16 RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 sampFreqHz,
+ int rate,
+ int packSize,
+ int channels);
+
+ void Run(TestPackStereo* channel, int in_channels, int out_channels);
+ void OpenOutFile(WebRtc_Word16 testNumber);
+ void DisplaySendReceiveCodec();
+
+ WebRtc_Word32 SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ int _testMode;
+
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ TestPackStereo* _channelA2B;
+
+ PCMFile _in_file_stereo;
+ PCMFile _in_file_mono;
+ PCMFile _outFileB;
+ WebRtc_Word16 _testCntr;
+ WebRtc_UWord16 _packSizeSamp;
+ WebRtc_UWord16 _packSizeBytes;
+ int _counter;
+ int _codecType;
+};
+
+} // namespace webrtc
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc b/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc
new file mode 100644
index 0000000..d7329fa
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc
@@ -0,0 +1,511 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TestVADDTX.h"
+
+#include <iostream>
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "testsupport/fileutils.h"
+#include "trace.h"
+#include "utility.h"
+
+namespace webrtc {
+
+TestVADDTX::TestVADDTX(int testMode):
+_acmA(NULL),
+_acmB(NULL),
+_channelA2B(NULL),
+_testResults(0)
+{
+ //testMode == 1 for more extensive testing
+ //testMode == 0 for quick test (autotest)
+ _testMode = testMode;
+}
+
+TestVADDTX::~TestVADDTX()
+{
+ if(_acmA != NULL)
+ {
+ AudioCodingModule::Destroy(_acmA);
+ _acmA = NULL;
+ }
+ if(_acmB != NULL)
+ {
+ AudioCodingModule::Destroy(_acmB);
+ _acmB = NULL;
+ }
+ if(_channelA2B != NULL)
+ {
+ delete _channelA2B;
+ _channelA2B = NULL;
+ }
+}
+
+void TestVADDTX::Perform()
+{
+ if(_testMode == 0)
+ {
+ printf("Running VAD/DTX Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ "---------- TestVADDTX ----------");
+ }
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _inFileA.Open(fileName, 32000, "rb");
+
+ _acmA = AudioCodingModule::Create(0);
+ _acmB = AudioCodingModule::Create(1);
+
+ _acmA->InitializeReceiver();
+ _acmB->InitializeReceiver();
+
+ WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
+ CodecInst myCodecParam;
+ if(_testMode != 0)
+ {
+ printf("Registering codecs at receiver... \n");
+ }
+ for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
+ {
+ _acmB->Codec(n, myCodecParam);
+ if(_testMode != 0)
+ {
+ printf("%s\n", myCodecParam.plname);
+ }
+ _acmB->RegisterReceiveCodec(myCodecParam);
+ }
+
+ // Create and connect the channel
+ _channelA2B = new Channel;
+ _acmA->RegisterTransportCallback(_channelA2B);
+ _channelA2B->RegisterReceiverACM(_acmB);
+
+ _acmA->RegisterVADCallback(&_monitor);
+
+
+ WebRtc_Word16 testCntr = 1;
+ WebRtc_Word16 testResults = 0;
+
+#ifdef WEBRTC_CODEC_ISAC
+ // Open outputfile
+ OpenOutFile(testCntr++);
+
+ // Register iSAC WB as send codec
+ char nameISAC[] = "ISAC";
+ RegisterSendCodec('A', nameISAC, 16000);
+
+ // Run the five test cased
+ runTestCases();
+
+ // Close file
+ _outFileB.Close();
+
+ // Open outputfile
+ OpenOutFile(testCntr++);
+
+ // Register iSAC SWB as send codec
+ RegisterSendCodec('A', nameISAC, 32000);
+
+ // Run the five test cased
+ runTestCases();
+
+ // Close file
+ _outFileB.Close();
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ // Open outputfile
+ OpenOutFile(testCntr++);
+
+ // Register iLBC as send codec
+ char nameILBC[] = "ilbc";
+ RegisterSendCodec('A', nameILBC);
+
+ // Run the five test cased
+ runTestCases();
+
+ // Close file
+ _outFileB.Close();
+
+#endif
+ if(_testMode) {
+ printf("Done!\n");
+ }
+
+ printf("VAD/DTX test completed with %d subtests failed\n", testResults);
+ if (testResults > 0)
+ {
+ printf("Press return\n\n");
+ getchar();
+ }
+}
+
+void TestVADDTX::runTestCases()
+{
+ if(_testMode != 0)
+ {
+ CodecInst myCodecParam;
+ _acmA->SendCodec(myCodecParam);
+ printf("%s\n", myCodecParam.plname);
+ }
+ else
+ {
+ printf(".");
+ }
+ // #1 DTX = OFF, VAD = ON, VADNormal
+ if(_testMode != 0)
+ printf("Test #1 ");
+ SetVAD(false, true, VADNormal);
+ Run();
+ _testResults += VerifyTest();
+
+ // #2 DTX = OFF, VAD = ON, VADAggr
+ if(_testMode != 0)
+ printf("Test #2 ");
+ SetVAD(false, true, VADAggr);
+ Run();
+ _testResults += VerifyTest();
+
+ // #3 DTX = ON, VAD = ON, VADLowBitrate
+ if(_testMode != 0)
+ printf("Test #3 ");
+ SetVAD(true, true, VADLowBitrate);
+ Run();
+ _testResults += VerifyTest();
+
+ // #4 DTX = ON, VAD = ON, VADVeryAggr
+ if(_testMode != 0)
+ printf("Test #4 ");
+ SetVAD(true, true, VADVeryAggr);
+ Run();
+ _testResults += VerifyTest();
+
+ // #5 DTX = ON, VAD = OFF, VADNormal
+ if(_testMode != 0)
+ printf("Test #5 ");
+ SetVAD(true, false, VADNormal);
+ Run();
+ _testResults += VerifyTest();
+
+}
+void TestVADDTX::runTestInternalDTX()
+{
+ // #6 DTX = ON, VAD = ON, VADNormal
+ if(_testMode != 0)
+ printf("Test #6 ");
+
+ SetVAD(true, true, VADNormal);
+ if(_acmA->ReplaceInternalDTXWithWebRtc(true) < 0) {
+ printf("Was not able to replace DTX since CN was not registered\n");
+ }
+ Run();
+ _testResults += VerifyTest();
+}
+
+void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
+{
+ bool dtxEnabled, vadEnabled;
+ ACMVADMode vadModeSet;
+
+ if (_acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode) < 0) {
+ assert(false);
+ }
+ if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
+ assert(false);
+ }
+
+ if(_testMode != 0)
+ {
+ if(statusDTX != dtxEnabled)
+ {
+ printf("DTX: %s not the same as requested: %s\n",
+ dtxEnabled? "ON":"OFF", dtxEnabled? "OFF":"ON");
+ }
+ if(((statusVAD == true) && (vadEnabled == false)) ||
+ ((statusVAD == false) && (vadEnabled == false) &&
+ (statusDTX == true)))
+ {
+ printf("VAD: %s not the same as requested: %s\n",
+ vadEnabled? "ON":"OFF", vadEnabled? "OFF":"ON");
+ }
+ if(vadModeSet != vadMode)
+ {
+ printf("VAD mode: %d not the same as requested: %d\n",
+ (WebRtc_Word16)vadModeSet, (WebRtc_Word16)vadMode);
+ }
+ }
+
+ // Requested VAD/DTX settings
+ _setStruct.statusDTX = statusDTX;
+ _setStruct.statusVAD = statusVAD;
+ _setStruct.vadMode = (ACMVADMode) vadMode;
+
+ // VAD settings after setting VAD in ACM
+ _getStruct.statusDTX = dtxEnabled;
+ _getStruct.statusVAD = vadEnabled;
+ _getStruct.vadMode = vadModeSet;
+
+}
+
+VADDTXstruct TestVADDTX::GetVAD()
+{
+ VADDTXstruct retStruct;
+ bool dtxEnabled, vadEnabled;
+ ACMVADMode vadModeSet;
+
+ if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
+ assert(false);
+ }
+
+ retStruct.statusDTX = dtxEnabled;
+ retStruct.statusVAD = vadEnabled;
+ retStruct.vadMode = vadModeSet;
+ return retStruct;
+}
+
+WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 samplingFreqHz,
+ WebRtc_Word32 rateKbps)
+{
+ if(_testMode != 0)
+ {
+ printf("Registering %s for side %c\n", codecName, side);
+ }
+ std::cout << std::flush;
+ AudioCodingModule* myACM;
+ switch(side)
+ {
+ case 'A':
+ {
+ myACM = _acmA;
+ break;
+ }
+ case 'B':
+ {
+ myACM = _acmB;
+ break;
+ }
+ default:
+ return -1;
+ }
+
+ if(myACM == NULL)
+ {
+ return -1;
+ }
+
+ CodecInst myCodecParam;
+ for(WebRtc_Word16 codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
+ codecCntr++)
+ {
+ CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, myCodecParam));
+ if(!STR_CASE_CMP(myCodecParam.plname, codecName))
+ {
+ if((samplingFreqHz == -1) || (myCodecParam.plfreq == samplingFreqHz))
+ {
+ if((rateKbps == -1) || (myCodecParam.rate == rateKbps))
+ {
+ break;
+ }
+ }
+ }
+ }
+
+ CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
+
+ // initialization was succesful
+ return 0;
+}
+
+void TestVADDTX::Run()
+{
+ AudioFrame audioFrame;
+
+ WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms();
+ WebRtc_UWord32 timestampA = 1;
+ WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+
+ while(!_inFileA.EndOfFile())
+ {
+ _inFileA.Read10MsData(audioFrame);
+ audioFrame._timeStamp = timestampA;
+ timestampA += SamplesIn10MsecA;
+ CHECK_ERROR(_acmA->Add10MsData(audioFrame));
+
+ CHECK_ERROR(_acmA->Process());
+
+ CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
+ _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
+ }
+#ifdef PRINT_STAT
+ _monitor.PrintStatistics(_testMode);
+#endif
+ _inFileA.Rewind();
+ _monitor.GetStatistics(_statCounter);
+ _monitor.ResetStatistics();
+}
+
+void TestVADDTX::OpenOutFile(WebRtc_Word16 testNumber)
+{
+ char fileName[500];
+ if(_testMode == 0)
+ {
+ sprintf(fileName, "%s/testVADDTX_autoFile_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ }
+ else
+ {
+ sprintf(fileName, "%s/testVADDTX_outFile_%02d.pcm",
+ webrtc::test::OutputPath().c_str(), testNumber);
+ }
+ _outFileB.Open(fileName, 16000, "wb");
+}
+
+
+WebRtc_Word16 TestVADDTX::VerifyTest()
+{
+ // Verify empty frame result
+ WebRtc_UWord8 statusEF = 0;
+ WebRtc_UWord8 vadPattern = 0;
+ WebRtc_UWord8 emptyFramePattern[6];
+ CodecInst myCodecParam;
+ _acmA->SendCodec(myCodecParam);
+ bool dtxInUse = true;
+ bool isReplaced = false;
+ if ((STR_CASE_CMP(myCodecParam.plname,"G729") == 0) ||
+ (STR_CASE_CMP(myCodecParam.plname,"G723") == 0) ||
+ (STR_CASE_CMP(myCodecParam.plname,"AMR") == 0) ||
+ (STR_CASE_CMP(myCodecParam.plname,"AMR-wb") == 0) ||
+ (STR_CASE_CMP(myCodecParam.plname,"speex") == 0))
+ {
+ _acmA->IsInternalDTXReplacedWithWebRtc(isReplaced);
+ if (!isReplaced)
+ {
+ dtxInUse = false;
+ }
+ }
+
+ // Check for error in VAD/DTX settings
+ if (_getStruct.statusDTX != _setStruct.statusDTX){
+ // DTX status doesn't match expected
+ vadPattern |= 4;
+ }
+ if (_getStruct.statusDTX){
+ if ((!_getStruct.statusVAD && dtxInUse) || (!dtxInUse && (_getStruct.statusVAD !=_setStruct.statusVAD)))
+ {
+ // Missmatch in VAD setting
+ vadPattern |= 2;
+ }
+ } else {
+ if (_getStruct.statusVAD != _setStruct.statusVAD){
+ // VAD status doesn't match expected
+ vadPattern |= 2;
+ }
+ }
+ if (_getStruct.vadMode != _setStruct.vadMode){
+ // VAD Mode doesn't match expected
+ vadPattern |= 1;
+ }
+
+ // Set expected empty frame pattern
+ int ii;
+ for (ii = 0; ii < 6; ii++) {
+ emptyFramePattern[ii] = 0;
+ }
+ emptyFramePattern[0] = 1; // "kNoEncoding", not important to check. Codecs with packetsize != 80 samples will get this output.
+ emptyFramePattern[1] = 1; // Expect to always receive some frames labeled "kActiveNormalEncoded"
+ emptyFramePattern[2] = (((!_getStruct.statusDTX && _getStruct.statusVAD) || (!dtxInUse && _getStruct.statusDTX))); // "kPassiveNormalEncoded"
+ emptyFramePattern[3] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 8000))); // "kPassiveDTXNB"
+ emptyFramePattern[4] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 16000))); // "kPassiveDTXWB"
+ emptyFramePattern[5] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 32000))); // "kPassiveDTXSWB"
+
+ // Check pattern 1-5 (skip 0)
+ for (int ii = 1; ii < 6; ii++)
+ {
+ if (emptyFramePattern[ii])
+ {
+ statusEF |= (_statCounter[ii] == 0);
+ }
+ else
+ {
+ statusEF |= (_statCounter[ii] > 0);
+ }
+ }
+ if ((statusEF == 0) && (vadPattern == 0))
+ {
+ if(_testMode != 0)
+ {
+ printf(" Test OK!\n");
+ }
+ return 0;
+ }
+ else
+ {
+ if (statusEF)
+ {
+ printf("\t\t\tUnexpected empty frame result!\n");
+ }
+ if (vadPattern)
+ {
+ printf("\t\t\tUnexpected SetVAD() result!\tDTX: %d\tVAD: %d\tMode: %d\n", (vadPattern >> 2) & 1, (vadPattern >> 1) & 1, vadPattern & 1);
+ }
+ return 1;
+ }
+}
+
+ActivityMonitor::ActivityMonitor()
+{
+ _counter[0] = _counter[1] = _counter[2] = _counter[3] = _counter[4] = _counter[5] = 0;
+}
+
+ActivityMonitor::~ActivityMonitor()
+{
+}
+
+WebRtc_Word32 ActivityMonitor::InFrameType(WebRtc_Word16 frameType)
+{
+ _counter[frameType]++;
+ return 0;
+}
+
+void ActivityMonitor::PrintStatistics(int testMode)
+{
+ if(testMode != 0)
+ {
+ printf("\n");
+ printf("kActiveNormalEncoded kPassiveNormalEncoded kPassiveDTXWB kPassiveDTXNB kPassiveDTXSWB kFrameEmpty\n");
+
+ printf("%19u", _counter[1]);
+ printf("%22u", _counter[2]);
+ printf("%14u", _counter[3]);
+ printf("%14u", _counter[4]);
+ printf("%14u", _counter[5]);
+ printf("%11u", _counter[0]);
+
+ printf("\n\n");
+ }
+}
+
+void ActivityMonitor::ResetStatistics()
+{
+ _counter[0] = _counter[1] = _counter[2] = _counter[3] = _counter[4] = _counter[5] = 0;
+}
+
+void ActivityMonitor::GetStatistics(WebRtc_UWord32* getCounter)
+{
+ for (int ii = 0; ii < 6; ii++)
+ {
+ getCounter[ii] = _counter[ii];
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestVADDTX.h b/trunk/src/modules/audio_coding/main/test/TestVADDTX.h
new file mode 100644
index 0000000..e8f9e1e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TestVADDTX.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_VAD_DTX_H
+#define TEST_VAD_DTX_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+
+namespace webrtc {
+
+typedef struct
+{
+ bool statusDTX;
+ bool statusVAD;
+ ACMVADMode vadMode;
+} VADDTXstruct;
+
+class ActivityMonitor : public ACMVADCallback
+{
+public:
+ ActivityMonitor();
+ ~ActivityMonitor();
+ WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
+ void PrintStatistics(int testMode);
+ void ResetStatistics();
+ void GetStatistics(WebRtc_UWord32* getCounter);
+private:
+ // counting according to
+ /*enum WebRtcACMEncodingType
+ {
+ kNoEncoding,
+ kActiveNormalEncoded,
+ kPassiveNormalEncoded,
+ kPassiveDTXNB,
+ kPassiveDTXWB,
+ kPassiveDTXSWB
+ };*/
+ WebRtc_UWord32 _counter[6];
+};
+
+class TestVADDTX : public ACMTest
+{
+public:
+ TestVADDTX(int testMode);
+ ~TestVADDTX();
+
+ void Perform();
+private:
+ // Registration can be based on codec name only, codec name and sampling frequency, or
+ // codec name, sampling frequency and rate.
+ WebRtc_Word16 RegisterSendCodec(char side,
+ char* codecName,
+ WebRtc_Word32 samplingFreqHz = -1,
+ WebRtc_Word32 rateKhz = -1);
+ void Run();
+ void OpenOutFile(WebRtc_Word16 testNumber);
+ void runTestCases();
+ void runTestInternalDTX();
+ void SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode);
+ VADDTXstruct GetVAD();
+ WebRtc_Word16 VerifyTest();//VADDTXstruct setDTX, VADDTXstruct getDTX);
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ Channel* _channelA2B;
+
+ PCMFile _inFileA;
+ PCMFile _outFileB;
+
+ ActivityMonitor _monitor;
+ WebRtc_UWord32 _statCounter[6];
+
+ int _testMode;
+ int _testResults;
+ VADDTXstruct _setStruct;
+ VADDTXstruct _getStruct;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/Tester.cc b/trunk/src/modules/audio_coding/main/test/Tester.cc
new file mode 100644
index 0000000..e35a0ec
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/Tester.cc
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string>
+#include <vector>
+
+#include "audio_coding_module.h"
+#include "trace.h"
+
+#include "APITest.h"
+#include "EncodeDecodeTest.h"
+#include "gtest/gtest.h"
+#include "iSACTest.h"
+#include "SpatialAudio.h"
+#include "TestAllCodecs.h"
+#include "TestFEC.h"
+#include "TestStereo.h"
+#include "TestVADDTX.h"
+#include "TwoWayCommunication.h"
+#include "testsupport/fileutils.h"
+
+using webrtc::AudioCodingModule;
+using webrtc::Trace;
+
+// Be sure to create the following directories before running the tests:
+// ./modules/audio_coding/main/test/res_tests
+// ./modules/audio_coding/main/test/res_autotests
+
+// Choose what tests to run by defining one or more of the following:
+#define ACM_AUTO_TEST // Most common codecs and settings will be tested
+//#define ACM_TEST_ENC_DEC // You decide what to test in run time.
+ // Used for debugging and for testing while implementing.
+//#define ACM_TEST_TWO_WAY // Debugging
+//#define ACM_TEST_ALL_ENC_DEC // Loop through all defined codecs and settings
+//#define ACM_TEST_STEREO // Run stereo and spatial audio tests
+//#define ACM_TEST_VAD_DTX // Run all VAD/DTX tests
+//#define ACM_TEST_FEC // Test FEC (also called RED)
+//#define ACM_TEST_CODEC_SPEC_API // Only iSAC has codec specfic APIs in this version
+//#define ACM_TEST_FULL_API // Test all APIs with threads (long test)
+
+
+void PopulateTests(std::vector<ACMTest*>* tests)
+{
+
+ Trace::CreateTrace();
+ std::string trace_file = webrtc::test::OutputPath() + "acm_trace.txt";
+ Trace::SetTraceFile(trace_file.c_str());
+
+ printf("The following tests will be executed:\n");
+#ifdef ACM_AUTO_TEST
+ printf(" ACM auto test\n");
+ tests->push_back(new webrtc::EncodeDecodeTest(0));
+ tests->push_back(new webrtc::TwoWayCommunication(0));
+ tests->push_back(new webrtc::TestAllCodecs(0));
+ tests->push_back(new webrtc::TestStereo(0));
+ tests->push_back(new webrtc::SpatialAudio(0));
+ tests->push_back(new webrtc::TestVADDTX(0));
+ tests->push_back(new webrtc::TestFEC(0));
+ tests->push_back(new webrtc::ISACTest(0));
+#endif
+#ifdef ACM_TEST_ENC_DEC
+ printf(" ACM encode-decode test\n");
+ tests->push_back(new webrtc::EncodeDecodeTest(2));
+#endif
+#ifdef ACM_TEST_TWO_WAY
+ printf(" ACM two-way communication test\n");
+ tests->push_back(new webrtc::TwoWayCommunication(1));
+#endif
+#ifdef ACM_TEST_ALL_ENC_DEC
+ printf(" ACM all codecs test\n");
+ tests->push_back(new webrtc::TestAllCodecs(1));
+#endif
+#ifdef ACM_TEST_STEREO
+ printf(" ACM stereo test\n");
+ tests->push_back(new webrtc::TestStereo(1));
+ tests->push_back(new webrtc::SpatialAudio(2));
+#endif
+#ifdef ACM_TEST_VAD_DTX
+ printf(" ACM VAD-DTX test\n");
+ tests->push_back(new webrtc::TestVADDTX(1));
+#endif
+#ifdef ACM_TEST_FEC
+ printf(" ACM FEC test\n");
+ tests->push_back(new webrtc::TestFEC(1));
+#endif
+#ifdef ACM_TEST_CODEC_SPEC_API
+ printf(" ACM codec API test\n");
+ tests->push_back(new webrtc::ISACTest(1));
+#endif
+#ifdef ACM_TEST_FULL_API
+ printf(" ACM full API test\n");
+ tests->push_back(new webrtc::APITest());
+#endif
+ printf("\n");
+}
+
+// TODO(kjellander): Make this a proper gtest instead of using this single test
+// to run all the tests.
+TEST(AudioCodingModuleTest, RunAllTests)
+{
+ std::vector<ACMTest*> tests;
+ PopulateTests(&tests);
+ std::vector<ACMTest*>::iterator it;
+ for (it=tests.begin() ; it < tests.end(); it++)
+ {
+ (*it)->Perform();
+ delete (*it);
+ }
+
+ Trace::ReturnTrace();
+ printf("ACM test completed\n");
+}
diff --git a/trunk/src/modules/audio_coding/main/test/TimedTrace.cc b/trunk/src/modules/audio_coding/main/test/TimedTrace.cc
new file mode 100644
index 0000000..6bf301f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TimedTrace.cc
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TimedTrace.h"
+#include <math.h>
+
+double TimedTrace::_timeEllapsedSec = 0;
+FILE* TimedTrace::_timedTraceFile = NULL;
+
+TimedTrace::TimedTrace()
+{
+
+}
+
+TimedTrace::~TimedTrace()
+{
+ if(_timedTraceFile != NULL)
+ {
+ fclose(_timedTraceFile);
+ }
+ _timedTraceFile = NULL;
+}
+
+WebRtc_Word16
+TimedTrace::SetUp(char* fileName)
+{
+ if(_timedTraceFile == NULL)
+ {
+ _timedTraceFile = fopen(fileName, "w");
+ }
+ if(_timedTraceFile == NULL)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+void
+TimedTrace::SetTimeEllapsed(double timeEllapsedSec)
+{
+ _timeEllapsedSec = timeEllapsedSec;
+}
+
+double
+TimedTrace::TimeEllapsed()
+{
+ return _timeEllapsedSec;
+}
+
+void
+TimedTrace::Tick10Msec()
+{
+ _timeEllapsedSec += 0.010;
+}
+
+void
+TimedTrace::TimedLogg(char* message)
+{
+ unsigned int minutes = (WebRtc_UWord32)floor(_timeEllapsedSec / 60.0);
+ double seconds = _timeEllapsedSec - minutes * 60;
+ //char myFormat[100] = "%8.2f, %3u:%05.2f: %s\n";
+ if(_timedTraceFile != NULL)
+ {
+ fprintf(_timedTraceFile, "%8.2f, %3u:%05.2f: %s\n",
+ _timeEllapsedSec,
+ minutes,
+ seconds,
+ message);
+ }
+}
diff --git a/trunk/src/modules/audio_coding/main/test/TimedTrace.h b/trunk/src/modules/audio_coding/main/test/TimedTrace.h
new file mode 100644
index 0000000..d37d287
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TimedTrace.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TIMED_TRACE_H
+#define TIMED_TRACE_H
+
+#include "typedefs.h"
+
+#include <cstdio>
+#include <cstdlib>
+
+
+class TimedTrace
+{
+public:
+ TimedTrace();
+ ~TimedTrace();
+
+ void SetTimeEllapsed(double myTime);
+ double TimeEllapsed();
+ void Tick10Msec();
+ WebRtc_Word16 SetUp(char* fileName);
+ void TimedLogg(char* message);
+
+private:
+ static double _timeEllapsedSec;
+ static FILE* _timedTraceFile;
+
+};
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc
new file mode 100644
index 0000000..c67733a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc
@@ -0,0 +1,524 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TwoWayCommunication.h"
+
+#include <cctype>
+#include <stdio.h>
+#include <string.h>
+
+#ifdef WIN32
+#include <Windows.h>
+#endif
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "gtest/gtest.h"
+#include "PCMFile.h"
+#include "trace.h"
+#include "testsupport/fileutils.h"
+#include "utility.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+TwoWayCommunication::TwoWayCommunication(int testMode)
+{
+ _testMode = testMode;
+}
+
+TwoWayCommunication::~TwoWayCommunication()
+{
+ AudioCodingModule::Destroy(_acmA);
+ AudioCodingModule::Destroy(_acmB);
+
+ AudioCodingModule::Destroy(_acmRefA);
+ AudioCodingModule::Destroy(_acmRefB);
+
+ delete _channel_A2B;
+ delete _channel_B2A;
+
+ delete _channelRef_A2B;
+ delete _channelRef_B2A;
+#ifdef WEBRTC_DTMF_DETECTION
+ if(_dtmfDetectorA != NULL)
+ {
+ delete _dtmfDetectorA;
+ }
+ if(_dtmfDetectorB != NULL)
+ {
+ delete _dtmfDetectorB;
+ }
+#endif
+ _inFileA.Close();
+ _inFileB.Close();
+ _outFileA.Close();
+ _outFileB.Close();
+ _outFileRefA.Close();
+ _outFileRefB.Close();
+}
+
+
+WebRtc_UWord8
+TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
+ WebRtc_UWord8* codecID_B)
+{
+ AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
+ WebRtc_UWord8 noCodec = tmpACM->NumberOfCodecs();
+ CodecInst codecInst;
+ printf("List of Supported Codecs\n");
+ printf("========================\n");
+ for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
+ {
+ tmpACM->Codec(codecCntr, codecInst);
+ printf("%d- %s\n", codecCntr, codecInst.plname);
+ }
+ printf("\nChoose a send codec for side A [0]: ");
+ char myStr[15] = "";
+ EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+ *codecID_A = (WebRtc_UWord8)atoi(myStr);
+
+ printf("\nChoose a send codec for side B [0]: ");
+ EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+ *codecID_B = (WebRtc_UWord8)atoi(myStr);
+
+ AudioCodingModule::Destroy(tmpACM);
+ printf("\n");
+ return 0;
+}
+
+WebRtc_Word16
+TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen,
+ WebRtc_UWord16* frequencyHz)
+{
+ char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
+ //strcpy(_fileName, "in.pcm");
+ //printf("\n\nPlease enter the input file: ");
+ EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
+ tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
+ WebRtc_Word16 n = 0;
+
+ // removing leading spaces
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (tmpName[n] != 0) &&
+ (n < MAX_FILE_NAME_LENGTH_BYTE))
+ {
+ n++;
+ }
+ if(n > 0)
+ {
+ memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
+ }
+
+ //removing trailing spaces
+ n = (WebRtc_Word16)(strlen(tmpName) - 1);
+ if(n >= 0)
+ {
+ while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
+ (n >= 0))
+ {
+ n--;
+ }
+ }
+ if(n >= 0)
+ {
+ tmpName[n + 1] = '\0';
+ }
+
+ WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
+ if(len > maxLen)
+ {
+ return -1;
+ }
+ if(len > 0)
+ {
+ strncpy(fileName, tmpName, len+1);
+ }
+ printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
+ *frequencyHz);
+ EXPECT_TRUE(fgets(tmpName, 6, stdin) != NULL);
+ WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
+ if(tmpFreq > 0)
+ {
+ *frequencyHz = tmpFreq;
+ }
+ return 0;
+}
+
+WebRtc_Word16 TwoWayCommunication::SetUp()
+{
+ _acmA = AudioCodingModule::Create(1);
+ _acmB = AudioCodingModule::Create(2);
+
+ _acmRefA = AudioCodingModule::Create(3);
+ _acmRefB = AudioCodingModule::Create(4);
+
+ WebRtc_UWord8 codecID_A;
+ WebRtc_UWord8 codecID_B;
+
+ ChooseCodec(&codecID_A, &codecID_B);
+ CodecInst codecInst_A;
+ CodecInst codecInst_B;
+ CodecInst dummyCodec;
+ _acmA->Codec(codecID_A, codecInst_A);
+ _acmB->Codec(codecID_B, codecInst_B);
+
+ _acmA->Codec(6, dummyCodec);
+
+ //--- Set A codecs
+ CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
+ CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
+#ifdef WEBRTC_DTMF_DETECTION
+ _dtmfDetectorA = new(DTMFDetector);
+ CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
+ ACMUSA));
+#endif
+ //--- Set ref-A codecs
+ CHECK_ERROR(_acmRefA->RegisterSendCodec(codecInst_A));
+ CHECK_ERROR(_acmRefA->RegisterReceiveCodec(codecInst_B));
+
+ //--- Set B codecs
+ CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
+ CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
+#ifdef WEBRTC_DTMF_DETECTION
+ _dtmfDetectorB = new(DTMFDetector);
+ CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
+ ACMUSA));
+#endif
+
+ //--- Set ref-B codecs
+ CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
+ CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
+
+ char fileName[500];
+ char refFileName[500];
+ WebRtc_UWord16 frequencyHz;
+
+ //--- Input A
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 32000;
+ printf("Enter input file at side A [%s]: ", fileName);
+ ChooseFile(fileName, 499, &frequencyHz);
+
+
+ _inFileA.Open(fileName, frequencyHz, "rb");
+
+ //--- Output A
+ std::string outputFileA = webrtc::test::OutputPath() + "outA.pcm";
+ strcpy(fileName, outputFileA.c_str());
+ frequencyHz = 16000;
+ printf("Enter output file at side A [%s]: ", fileName);
+ ChooseFile(fileName, 499, &frequencyHz);
+ _outFileA.Open(fileName, frequencyHz, "wb");
+ strcpy(refFileName, "ref_");
+ strcat(refFileName, fileName);
+ _outFileRefA.Open(refFileName, frequencyHz, "wb");
+
+ //--- Input B
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 32000;
+ printf("\n\nEnter input file at side B [%s]: ", fileName);
+ ChooseFile(fileName, 499, &frequencyHz);
+ _inFileB.Open(fileName, frequencyHz, "rb");
+
+ //--- Output B
+ std::string outputFileB = webrtc::test::OutputPath() + "outB.pcm";
+ strcpy(fileName, outputFileB.c_str());
+ frequencyHz = 16000;
+ printf("Enter output file at side B [%s]: ", fileName);
+ ChooseFile(fileName, 499, &frequencyHz);
+ _outFileB.Open(fileName, frequencyHz, "wb");
+ strcpy(refFileName, "ref_");
+ strcat(refFileName, fileName);
+ _outFileRefB.Open(refFileName, frequencyHz, "wb");
+
+ //--- Set A-to-B channel
+ _channel_A2B = new Channel;
+ _acmA->RegisterTransportCallback(_channel_A2B);
+ _channel_A2B->RegisterReceiverACM(_acmB);
+ //--- Do the same for the reference
+ _channelRef_A2B = new Channel;
+ _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+ _channelRef_A2B->RegisterReceiverACM(_acmRefB);
+
+ //--- Set B-to-A channel
+ _channel_B2A = new Channel;
+ _acmB->RegisterTransportCallback(_channel_B2A);
+ _channel_B2A->RegisterReceiverACM(_acmA);
+ //--- Do the same for reference
+ _channelRef_B2A = new Channel;
+ _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+ _channelRef_B2A->RegisterReceiverACM(_acmRefA);
+
+ // The clicks will be more obvious when we
+ // are in FAX mode.
+ _acmB->SetPlayoutMode(fax);
+ _acmRefB->SetPlayoutMode(fax);
+
+ return 0;
+}
+
+WebRtc_Word16 TwoWayCommunication::SetUpAutotest()
+{
+ _acmA = AudioCodingModule::Create(1);
+ _acmB = AudioCodingModule::Create(2);
+
+ _acmRefA = AudioCodingModule::Create(3);
+ _acmRefB = AudioCodingModule::Create(4);
+
+ CodecInst codecInst_A;
+ CodecInst codecInst_B;
+ CodecInst dummyCodec;
+
+ _acmA->Codec("ISAC", codecInst_A, 16000);
+ _acmB->Codec("L16", codecInst_B, 8000);
+ _acmA->Codec(6, dummyCodec);
+
+ //--- Set A codecs
+ CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
+ CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
+#ifdef WEBRTC_DTMF_DETECTION
+ _dtmfDetectorA = new(DTMFDetector);
+ CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
+ ACMUSA));
+#endif
+
+ //--- Set ref-A codecs
+ CHECK_ERROR(_acmRefA->RegisterSendCodec(codecInst_A));
+ CHECK_ERROR(_acmRefA->RegisterReceiveCodec(codecInst_B));
+
+ //--- Set B codecs
+ CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
+ CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
+#ifdef WEBRTC_DTMF_DETECTION
+ _dtmfDetectorB = new(DTMFDetector);
+ CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
+ ACMUSA));
+#endif
+
+ //--- Set ref-B codecs
+ CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
+ CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
+
+ char fileName[500];
+ char refFileName[500];
+ WebRtc_UWord16 frequencyHz;
+
+
+ //--- Input A
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 16000;
+ _inFileA.Open(fileName, frequencyHz, "rb");
+
+ //--- Output A
+ std::string outputFileA = webrtc::test::OutputPath() + "outAutotestA.pcm";
+ strcpy(fileName, outputFileA.c_str());
+ frequencyHz = 16000;
+ _outFileA.Open(fileName, frequencyHz, "wb");
+ std::string outputRefFileA = webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
+ strcpy(refFileName, outputRefFileA.c_str());
+ _outFileRefA.Open(refFileName, frequencyHz, "wb");
+
+ //--- Input B
+ strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
+ frequencyHz = 16000;
+ _inFileB.Open(fileName, frequencyHz, "rb");
+
+ //--- Output B
+ std::string outputFileB = webrtc::test::OutputPath() + "outAutotestB.pcm";
+ strcpy(fileName, outputFileB.c_str());
+ frequencyHz = 16000;
+ _outFileB.Open(fileName, frequencyHz, "wb");
+ std::string outputRefFileB = webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
+ strcpy(refFileName, outputRefFileB.c_str());
+ _outFileRefB.Open(refFileName, frequencyHz, "wb");
+
+ //--- Set A-to-B channel
+ _channel_A2B = new Channel;
+ _acmA->RegisterTransportCallback(_channel_A2B);
+ _channel_A2B->RegisterReceiverACM(_acmB);
+ //--- Do the same for the reference
+ _channelRef_A2B = new Channel;
+ _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+ _channelRef_A2B->RegisterReceiverACM(_acmRefB);
+
+ //--- Set B-to-A channel
+ _channel_B2A = new Channel;
+ _acmB->RegisterTransportCallback(_channel_B2A);
+ _channel_B2A->RegisterReceiverACM(_acmA);
+ //--- Do the same for reference
+ _channelRef_B2A = new Channel;
+ _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+ _channelRef_B2A->RegisterReceiverACM(_acmRefA);
+
+ // The clicks will be more obvious when we
+ // are in FAX mode.
+ _acmB->SetPlayoutMode(fax);
+ _acmRefB->SetPlayoutMode(fax);
+
+ return 0;
+}
+
+void
+TwoWayCommunication::Perform()
+{
+ if(_testMode == 0)
+ {
+ printf("Running TwoWayCommunication Test");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TwoWayCommunication ----------");
+ SetUpAutotest();
+ }
+ else
+ {
+ SetUp();
+ }
+ unsigned int msecPassed = 0;
+ unsigned int secPassed = 0;
+
+ WebRtc_Word32 outFreqHzA = _outFileA.SamplingFrequency();
+ WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
+
+ AudioFrame audioFrame;
+
+ CodecInst codecInst_B;
+ CodecInst dummy;
+
+ _acmB->SendCodec(codecInst_B);
+
+ if(_testMode != 0)
+ {
+ printf("\n");
+ printf("sec:msec A B\n");
+ printf("-------- ----- -----\n");
+ }
+
+ while(!_inFileA.EndOfFile() && !_inFileB.EndOfFile())
+ {
+ _inFileA.Read10MsData(audioFrame);
+ _acmA->Add10MsData(audioFrame);
+ _acmRefA->Add10MsData(audioFrame);
+
+ _inFileB.Read10MsData(audioFrame);
+ _acmB->Add10MsData(audioFrame);
+ _acmRefB->Add10MsData(audioFrame);
+
+
+ _acmA->Process();
+ _acmB->Process();
+ _acmRefA->Process();
+ _acmRefB->Process();
+
+ _acmA->PlayoutData10Ms(outFreqHzA, audioFrame);
+ _outFileA.Write10MsData(audioFrame);
+
+ _acmRefA->PlayoutData10Ms(outFreqHzA, audioFrame);
+ _outFileRefA.Write10MsData(audioFrame);
+
+ _acmB->PlayoutData10Ms(outFreqHzB, audioFrame);
+ _outFileB.Write10MsData(audioFrame);
+
+ _acmRefB->PlayoutData10Ms(outFreqHzB, audioFrame);
+ _outFileRefB.Write10MsData(audioFrame);
+
+ msecPassed += 10;
+ if(msecPassed >= 1000)
+ {
+ msecPassed = 0;
+ secPassed++;
+ }
+ if(((secPassed%5) == 4) && (msecPassed == 0))
+ {
+ if(_testMode != 0)
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ }
+ _acmA->ResetEncoder();
+ if(_testMode == 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- Errors epected");
+ printf(".");
+ }
+ else
+ {
+ printf("Reset Encoder (click in side B) ");
+ printf("Initialize Sender (no audio in side A)\n");
+ }
+ CHECK_ERROR(_acmB->InitializeSender());
+ }
+ if(((secPassed%5) == 4) && (msecPassed >= 990))
+ {
+ if(_testMode == 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "----- END: Errors epected");
+ printf(".");
+ }
+ else
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ printf(" ");
+ printf("Register Send Codec (audio back in side A)\n");
+ }
+ CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
+ CHECK_ERROR(_acmB->SendCodec(dummy));
+ }
+ if(((secPassed%7) == 6) && (msecPassed == 0))
+ {
+ CHECK_ERROR(_acmB->ResetDecoder());
+ if(_testMode == 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- Errors epected");
+ printf(".");
+ }
+ else
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ printf("Initialize Receiver (no audio in side A) ");
+ printf("Reset Decoder\n");
+ }
+ CHECK_ERROR(_acmA->InitializeReceiver());
+ }
+ if(((secPassed%7) == 6) && (msecPassed >= 990))
+ {
+ if(_testMode == 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "----- END: Errors epected");
+ printf(".");
+ }
+ else
+ {
+ printf("%3u:%3u ", secPassed, msecPassed);
+ printf("Register Receive Coded (audio back in side A)\n");
+ }
+ CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
+ }
+ //Sleep(9);
+ }
+ if(_testMode == 0)
+ {
+ printf("Done!\n");
+ }
+
+#ifdef WEBRTC_DTMF_DETECTION
+ printf("\nDTMF at Side A\n");
+ _dtmfDetectorA->PrintDetectedDigits();
+
+ printf("\nDTMF at Side B\n");
+ _dtmfDetectorB->PrintDetectedDigits();
+#endif
+
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h
new file mode 100644
index 0000000..0b33317
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TWO_WAY_COMMUNICATION_H
+#define TWO_WAY_COMMUNICATION_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+#include "audio_coding_module.h"
+#include "utility.h"
+
+namespace webrtc {
+
+class TwoWayCommunication : public ACMTest
+{
+public:
+ TwoWayCommunication(int testMode = 1);
+ ~TwoWayCommunication();
+
+ void Perform();
+private:
+ WebRtc_UWord8 ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecID_B);
+ WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen, WebRtc_UWord16* frequencyHz);
+ WebRtc_Word16 SetUp();
+ WebRtc_Word16 SetUpAutotest();
+
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ AudioCodingModule* _acmRefA;
+ AudioCodingModule* _acmRefB;
+
+ Channel* _channel_A2B;
+ Channel* _channel_B2A;
+
+ Channel* _channelRef_A2B;
+ Channel* _channelRef_B2A;
+
+ PCMFile _inFileA;
+ PCMFile _inFileB;
+
+ PCMFile _outFileA;
+ PCMFile _outFileB;
+
+ PCMFile _outFileRefA;
+ PCMFile _outFileRefB;
+
+ DTMFDetector* _dtmfDetectorA;
+ DTMFDetector* _dtmfDetectorB;
+
+ int _testMode;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/iSACTest.cc b/trunk/src/modules/audio_coding/main/test/iSACTest.cc
new file mode 100644
index 0000000..966c4c5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/iSACTest.cc
@@ -0,0 +1,598 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cctype>
+#include <stdio.h>
+#include <string.h>
+
+#if _WIN32
+#include <windows.h>
+#elif WEBRTC_LINUX
+#include <ctime>
+#else
+#include <sys/time.h>
+#include <time.h>
+#endif
+
+#include "event_wrapper.h"
+#include "iSACTest.h"
+#include "utility.h"
+#include "trace.h"
+#include "testsupport/fileutils.h"
+#include "tick_util.h"
+
+namespace webrtc {
+
+void SetISACConfigDefault(
+ ACMTestISACConfig& isacConfig)
+{
+ isacConfig.currentRateBitPerSec = 0;
+ isacConfig.currentFrameSizeMsec = 0;
+ isacConfig.maxRateBitPerSec = 0;
+ isacConfig.maxPayloadSizeByte = 0;
+ isacConfig.encodingMode = -1;
+ isacConfig.initRateBitPerSec = 0;
+ isacConfig.initFrameSizeInMsec = 0;
+ isacConfig.enforceFrameSize = false;
+ return;
+}
+
+
+WebRtc_Word16 SetISAConfig(
+ ACMTestISACConfig& isacConfig,
+ AudioCodingModule* acm,
+ int testMode)
+{
+
+ if((isacConfig.currentRateBitPerSec != 0) ||
+ (isacConfig.currentFrameSizeMsec != 0))
+ {
+ CodecInst sendCodec;
+ acm->SendCodec(sendCodec);
+ if(isacConfig.currentRateBitPerSec < 0)
+ {
+ sendCodec.rate = -1;
+ CHECK_ERROR(acm->RegisterSendCodec(sendCodec));
+ if(testMode != 0)
+ {
+ printf("ISAC-%s Registered in adaptive (channel-dependent) mode.\n",
+ (sendCodec.plfreq == 32000)? "swb":"wb");
+ }
+ }
+ else
+ {
+
+ if(isacConfig.currentRateBitPerSec != 0)
+ {
+ sendCodec.rate = isacConfig.currentRateBitPerSec;
+ }
+ if(isacConfig.currentFrameSizeMsec != 0)
+ {
+ sendCodec.pacsize = isacConfig.currentFrameSizeMsec *
+ (sendCodec.plfreq / 1000);
+ }
+ CHECK_ERROR(acm->RegisterSendCodec(sendCodec));
+ if(testMode != 0)
+ {
+ printf("Target rate is set to %d bit/sec with frame-size %d ms \n",
+ (int)isacConfig.currentRateBitPerSec,
+ (int)sendCodec.pacsize / (sendCodec.plfreq / 1000));
+ }
+ }
+ }
+
+ if(isacConfig.maxRateBitPerSec > 0)
+ {
+ CHECK_ERROR(acm->SetISACMaxRate(isacConfig.maxRateBitPerSec));
+ if(testMode != 0)
+ {
+ printf("Max rate is set to %u bit/sec\n",
+ isacConfig.maxRateBitPerSec);
+ }
+ }
+ if(isacConfig.maxPayloadSizeByte > 0)
+ {
+ CHECK_ERROR(acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte));
+ if(testMode != 0)
+ {
+ printf("Max payload-size is set to %u bit/sec\n",
+ isacConfig.maxPayloadSizeByte);
+ }
+ }
+ if((isacConfig.initFrameSizeInMsec != 0) ||
+ (isacConfig.initRateBitPerSec != 0))
+ {
+ CHECK_ERROR(acm->ConfigISACBandwidthEstimator(
+ (WebRtc_UWord8)isacConfig.initFrameSizeInMsec,
+ (WebRtc_UWord16)isacConfig.initRateBitPerSec,
+ isacConfig.enforceFrameSize));
+ if((isacConfig.initFrameSizeInMsec != 0) && (testMode != 0))
+ {
+ printf("Initialize BWE to %d msec frame-size\n",
+ isacConfig.initFrameSizeInMsec);
+ }
+ if((isacConfig.initRateBitPerSec != 0) && (testMode != 0))
+ {
+ printf("Initialize BWE to %u bit/sec send-bandwidth\n",
+ isacConfig.initRateBitPerSec);
+ }
+ }
+
+ return 0;
+}
+
+
+ISACTest::ISACTest(int testMode)
+{
+ _testMode = testMode;
+}
+
+ISACTest::~ISACTest()
+{
+ AudioCodingModule::Destroy(_acmA);
+ AudioCodingModule::Destroy(_acmB);
+
+ delete _channel_A2B;
+ delete _channel_B2A;
+}
+
+
+WebRtc_Word16
+ISACTest::Setup()
+{
+ int codecCntr;
+ CodecInst codecParam;
+
+ _acmA = AudioCodingModule::Create(1);
+ _acmB = AudioCodingModule::Create(2);
+
+ for(codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs(); codecCntr++)
+ {
+ AudioCodingModule::Codec(codecCntr, codecParam);
+ if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 16000)
+ {
+ memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
+ _idISAC16kHz = codecCntr;
+ }
+ if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 32000)
+ {
+ memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
+ _idISAC32kHz = codecCntr;
+ }
+ }
+
+ // register both iSAC-wb & iSAC-swb in both sides as receiver codecs
+ CHECK_ERROR(_acmA->RegisterReceiveCodec(_paramISAC16kHz));
+ CHECK_ERROR(_acmA->RegisterReceiveCodec(_paramISAC32kHz));
+ CHECK_ERROR(_acmB->RegisterReceiveCodec(_paramISAC16kHz));
+ CHECK_ERROR(_acmB->RegisterReceiveCodec(_paramISAC32kHz));
+
+ //--- Set A-to-B channel
+ _channel_A2B = new Channel;
+ CHECK_ERROR(_acmA->RegisterTransportCallback(_channel_A2B));
+ _channel_A2B->RegisterReceiverACM(_acmB);
+
+ //--- Set B-to-A channel
+ _channel_B2A = new Channel;
+ CHECK_ERROR(_acmB->RegisterTransportCallback(_channel_B2A));
+ _channel_B2A->RegisterReceiverACM(_acmA);
+
+ strncpy(_fileNameSWB, "./test/data/audio_coding/testfile32kHz.pcm",
+ MAX_FILE_NAME_LENGTH_BYTE);
+
+ _acmB->RegisterSendCodec(_paramISAC16kHz);
+ _acmA->RegisterSendCodec(_paramISAC32kHz);
+
+ if(_testMode != 0)
+ {
+ printf("Side A Send Codec\n");
+ printf("%s %d\n", _paramISAC32kHz.plname, _paramISAC32kHz.plfreq);
+
+ printf("Side B Send Codec\n");
+ printf("%s %d\n", _paramISAC16kHz.plname, _paramISAC16kHz.plfreq);
+ }
+
+ _inFileA.Open(_fileNameSWB, 32000, "rb");
+ std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
+ std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
+ _outFileA.Open(fileNameA.c_str(), 32000, "wb");
+ _outFileB.Open(fileNameB.c_str(), 32000, "wb");
+
+ while(!_inFileA.EndOfFile())
+ {
+ Run10ms();
+ }
+ CodecInst receiveCodec;
+ CHECK_ERROR(_acmA->ReceiveCodec(receiveCodec));
+ if(_testMode != 0)
+ {
+ printf("Side A Receive Codec\n");
+ printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
+ }
+
+ CHECK_ERROR(_acmB->ReceiveCodec(receiveCodec));
+ if(_testMode != 0)
+ {
+ printf("Side B Receive Codec\n");
+ printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
+ }
+
+ _inFileA.Close();
+ _outFileA.Close();
+ _outFileB.Close();
+
+ return 0;
+}
+
+
+void
+ISACTest::Perform()
+{
+ if(_testMode == 0)
+ {
+ printf("Running iSAC Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- iSACTest ----------");
+ }
+
+ Setup();
+
+ WebRtc_Word16 testNr = 0;
+ ACMTestISACConfig wbISACConfig;
+ ACMTestISACConfig swbISACConfig;
+
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = -1;
+ swbISACConfig.currentRateBitPerSec = -1;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ if (_testMode != 0)
+ {
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = -1;
+ swbISACConfig.currentRateBitPerSec = -1;
+ wbISACConfig.initRateBitPerSec = 13000;
+ wbISACConfig.initFrameSizeInMsec = 60;
+ swbISACConfig.initRateBitPerSec = 20000;
+ swbISACConfig.initFrameSizeInMsec = 30;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = 20000;
+ swbISACConfig.currentRateBitPerSec = 48000;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = 16000;
+ swbISACConfig.currentRateBitPerSec = 30000;
+ wbISACConfig.currentFrameSizeMsec = 60;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+ }
+
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ int user_input;
+ if((_testMode == 0) || (_testMode == 1))
+ {
+ swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
+ wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
+ }
+ else
+ {
+ printf("Enter the max payload-size for side A: ");
+ CHECK_ERROR(scanf("%d", &user_input));
+ swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
+ printf("Enter the max payload-size for side B: ");
+ CHECK_ERROR(scanf("%d", &user_input));
+ wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
+ }
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ _acmA->ResetEncoder();
+ _acmB->ResetEncoder();
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ if((_testMode == 0) || (_testMode == 1))
+ {
+ swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
+ wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
+ }
+ else
+ {
+ printf("Enter the max rate for side A: ");
+ CHECK_ERROR(scanf("%d", &user_input));
+ swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
+ printf("Enter the max rate for side B: ");
+ CHECK_ERROR(scanf("%d", &user_input));
+ wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
+ }
+
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+
+ testNr++;
+ if(_testMode == 0)
+ {
+ SwitchingSamplingRate(testNr, 4);
+ printf("Done!\n");
+ }
+ else
+ {
+ SwitchingSamplingRate(testNr, 80);
+ }
+}
+
+
+void
+ISACTest::Run10ms()
+{
+ AudioFrame audioFrame;
+
+ _inFileA.Read10MsData(audioFrame);
+ CHECK_ERROR(_acmA->Add10MsData(audioFrame));
+
+ CHECK_ERROR(_acmB->Add10MsData(audioFrame));
+
+ CHECK_ERROR(_acmA->Process());
+ CHECK_ERROR(_acmB->Process());
+
+ CHECK_ERROR(_acmA->PlayoutData10Ms(32000, audioFrame));
+ _outFileA.Write10MsData(audioFrame);
+
+ CHECK_ERROR(_acmB->PlayoutData10Ms(32000, audioFrame));
+ _outFileB.Write10MsData(audioFrame);
+}
+
+void
+ISACTest::EncodeDecode(
+ int testNr,
+ ACMTestISACConfig& wbISACConfig,
+ ACMTestISACConfig& swbISACConfig)
+{
+ if(_testMode == 0)
+ {
+ printf(".");
+ }
+ else
+ {
+ printf("\nTest %d:\n\n", testNr);
+ }
+ char fileNameOut[MAX_FILE_NAME_LENGTH_BYTE];
+
+ // Files in Side A
+ _inFileA.Open(_fileNameSWB, 32000, "rb", true);
+ if(_testMode == 0)
+ {
+ sprintf(fileNameOut,
+ "%s/out_iSACTest_%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "A",
+ testNr);
+ }
+ else
+ {
+ sprintf(fileNameOut,
+ "%s/out%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "A",
+ testNr);
+ }
+ _outFileA.Open(fileNameOut, 32000, "wb");
+
+ // Files in Side B
+ _inFileB.Open(_fileNameSWB, 32000, "rb", true);
+ if(_testMode == 0)
+ {
+ sprintf(fileNameOut,
+ "%s/out_iSACTest_%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "B",
+ testNr);
+ }
+ else
+ {
+ sprintf(fileNameOut,
+ "%s/out%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "B",
+ testNr);
+ }
+ _outFileB.Open(fileNameOut, 32000, "wb");
+
+ CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC16kHz));
+ CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
+
+ CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC32kHz));
+ CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
+ if(_testMode != 0)
+ {
+ printf("Side A Sending Super-Wideband \n");
+ printf("Side B Sending Wideband\n\n");
+ }
+
+ SetISAConfig(swbISACConfig, _acmA, _testMode);
+ SetISAConfig(wbISACConfig, _acmB, _testMode);
+
+ bool adaptiveMode = false;
+ if((swbISACConfig.currentRateBitPerSec == -1) ||
+ (wbISACConfig.currentRateBitPerSec == -1))
+ {
+ adaptiveMode = true;
+ }
+ _myTimer.Reset();
+ _channel_A2B->ResetStats();
+ _channel_B2A->ResetStats();
+
+ char currentTime[500];
+ if(_testMode == 2) printf("\n");
+ CodecInst sendCodec;
+ EventWrapper* myEvent = EventWrapper::Create();
+ myEvent->StartTimer(true, 10);
+ while(!(_inFileA.EndOfFile() || _inFileA.Rewinded()))
+ {
+ Run10ms();
+ _myTimer.Tick10ms();
+ _myTimer.CurrentTimeHMS(currentTime);
+ if(_testMode == 2) printf("\r%s ", currentTime);
+
+ if((adaptiveMode) && (_testMode != 0))
+ {
+ myEvent->Wait(5000);
+
+ _acmA->SendCodec(sendCodec);
+ if(_testMode == 2) printf("[%d] ", sendCodec.rate);
+ _acmB->SendCodec(sendCodec);
+ if(_testMode == 2) printf("[%d] ", sendCodec.rate);
+ }
+ }
+
+ if(_testMode != 0)
+ {
+ printf("\n\nSide A statistics\n\n");
+ _channel_A2B->PrintStats(_paramISAC32kHz);
+
+ printf("\n\nSide B statistics\n\n");
+ _channel_B2A->PrintStats(_paramISAC16kHz);
+ }
+
+ _channel_A2B->ResetStats();
+ _channel_B2A->ResetStats();
+
+ if(_testMode != 0) printf("\n");
+ _outFileA.Close();
+ _outFileB.Close();
+ _inFileA.Close();
+ _inFileB.Close();
+}
+
+void
+ISACTest::SwitchingSamplingRate(
+ int testNr,
+ int maxSampRateChange)
+{
+ char fileNameOut[MAX_FILE_NAME_LENGTH_BYTE];
+
+ // Files in Side A
+ _inFileA.Open(_fileNameSWB, 32000, "rb");
+ if(_testMode == 0)
+ {
+ sprintf(fileNameOut,
+ "%s/out_iSACTest_%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "A",
+ testNr);
+ }
+ else
+ {
+ printf("\nTest %d", testNr);
+ printf(" Alternate between WB and SWB at the sender Side\n\n");
+ sprintf(fileNameOut,
+ "%s/out%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "A",
+ testNr);
+ }
+ _outFileA.Open(fileNameOut, 32000, "wb", true);
+
+ // Files in Side B
+ _inFileB.Open(_fileNameSWB, 32000, "rb");
+ if(_testMode == 0)
+ {
+ sprintf(fileNameOut,
+ "%s/out_iSACTest_%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "B",
+ testNr);
+ }
+ else
+ {
+ sprintf(fileNameOut, "%s/out%s_%02d.pcm",
+ webrtc::test::OutputPath().c_str(),
+ "B",
+ testNr);
+ }
+ _outFileB.Open(fileNameOut, 32000, "wb", true);
+
+ CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
+ CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
+ if(_testMode != 0)
+ {
+ printf("Side A Sending Super-Wideband \n");
+ printf("Side B Sending Wideband\n");
+ }
+
+ int numSendCodecChanged = 0;
+ _myTimer.Reset();
+ char currentTime[50];
+ while(numSendCodecChanged < (maxSampRateChange<<1))
+ {
+ Run10ms();
+ _myTimer.Tick10ms();
+ _myTimer.CurrentTimeHMS(currentTime);
+ if(_testMode == 2) printf("\r%s", currentTime);
+ if(_inFileA.EndOfFile())
+ {
+ if(_inFileA.SamplingFrequency() == 16000)
+ {
+ if(_testMode != 0) printf("\nSide A switched to Send Super-Wideband\n");
+ _inFileA.Close();
+ _inFileA.Open(_fileNameSWB, 32000, "rb");
+ CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
+ }
+ else
+ {
+ if(_testMode != 0) printf("\nSide A switched to Send Wideband\n");
+ _inFileA.Close();
+ _inFileA.Open(_fileNameSWB, 32000, "rb");
+ CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC16kHz));
+ }
+ numSendCodecChanged++;
+ }
+
+ if(_inFileB.EndOfFile())
+ {
+ if(_inFileB.SamplingFrequency() == 16000)
+ {
+ if(_testMode != 0) printf("\nSide B switched to Send Super-Wideband\n");
+ _inFileB.Close();
+ _inFileB.Open(_fileNameSWB, 32000, "rb");
+ CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC32kHz));
+ }
+ else
+ {
+ if(_testMode != 0) printf("\nSide B switched to Send Wideband\n");
+ _inFileB.Close();
+ _inFileB.Open(_fileNameSWB, 32000, "rb");
+ CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
+ }
+ numSendCodecChanged++;
+ }
+ }
+ _outFileA.Close();
+ _outFileB.Close();
+ _inFileA.Close();
+ _inFileB.Close();
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/iSACTest.h b/trunk/src/modules/audio_coding/main/test/iSACTest.h
new file mode 100644
index 0000000..17bacad
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/iSACTest.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef ACM_ISAC_TEST_H
+#define ACM_ISAC_TEST_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+#include "audio_coding_module.h"
+#include "utility.h"
+#include "common_types.h"
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+#define NO_OF_CLIENTS 15
+
+namespace webrtc {
+
+struct ACMTestISACConfig
+{
+ WebRtc_Word32 currentRateBitPerSec;
+ WebRtc_Word16 currentFrameSizeMsec;
+ WebRtc_UWord32 maxRateBitPerSec;
+ WebRtc_Word16 maxPayloadSizeByte;
+ WebRtc_Word16 encodingMode;
+ WebRtc_UWord32 initRateBitPerSec;
+ WebRtc_Word16 initFrameSizeInMsec;
+ bool enforceFrameSize;
+};
+
+
+
+class ISACTest : public ACMTest
+{
+public:
+ ISACTest(int testMode);
+ ~ISACTest();
+
+ void Perform();
+private:
+ WebRtc_Word16 Setup();
+ WebRtc_Word16 SetupConference();
+ WebRtc_Word16 RunConference();
+
+
+ void Run10ms();
+
+ void EncodeDecode(
+ int testNr,
+ ACMTestISACConfig& wbISACConfig,
+ ACMTestISACConfig& swbISACConfig);
+
+ void TestBWE(
+ int testNr);
+
+ void SwitchingSamplingRate(
+ int testNr,
+ int maxSampRateChange);
+
+ AudioCodingModule* _acmA;
+ AudioCodingModule* _acmB;
+
+ Channel* _channel_A2B;
+ Channel* _channel_B2A;
+
+ PCMFile _inFileA;
+ PCMFile _inFileB;
+
+ PCMFile _outFileA;
+ PCMFile _outFileB;
+
+ WebRtc_UWord8 _idISAC16kHz;
+ WebRtc_UWord8 _idISAC32kHz;
+ CodecInst _paramISAC16kHz;
+ CodecInst _paramISAC32kHz;
+
+ char _fileNameWB[MAX_FILE_NAME_LENGTH_BYTE];
+ char _fileNameSWB[MAX_FILE_NAME_LENGTH_BYTE];
+
+ ACMTestTimer _myTimer;
+ int _testMode;
+
+ AudioCodingModule* _defaultACM32;
+ AudioCodingModule* _defaultACM16;
+
+ AudioCodingModule* _confACM[NO_OF_CLIENTS];
+ AudioCodingModule* _clientACM[NO_OF_CLIENTS];
+ Channel* _conf2Client[NO_OF_CLIENTS];
+ Channel* _client2Conf[NO_OF_CLIENTS];
+
+ PCMFile _clientOutFile[NO_OF_CLIENTS];
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/audio_coding/main/test/utility.cc b/trunk/src/modules/audio_coding/main/test/utility.cc
new file mode 100644
index 0000000..56acbf7
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/utility.cc
@@ -0,0 +1,434 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "utility.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "audio_coding_module.h"
+#include "common_types.h"
+#include "gtest/gtest.h"
+
+#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
+
+namespace webrtc {
+
+ACMTestTimer::ACMTestTimer() :
+_msec(0),
+_sec(0),
+_min(0),
+_hour(0)
+{
+ return;
+}
+
+ACMTestTimer::~ACMTestTimer()
+{
+ return;
+}
+
+void ACMTestTimer::Reset()
+{
+ _msec = 0;
+ _sec = 0;
+ _min = 0;
+ _hour = 0;
+ return;
+}
+void ACMTestTimer::Tick10ms()
+{
+ _msec += 10;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick1ms()
+{
+ _msec++;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick100ms()
+{
+ _msec += 100;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick1sec()
+{
+ _sec++;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::CurrentTimeHMS(char* currTime)
+{
+ sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min, (double)_sec + (double)_msec / 1000.);
+ return;
+}
+
+void ACMTestTimer::CurrentTime(
+ unsigned long& h,
+ unsigned char& m,
+ unsigned char& s,
+ unsigned short& ms)
+{
+ h = _hour;
+ m = _min;
+ s = _sec;
+ ms = _msec;
+ return;
+}
+
+void ACMTestTimer::Adjust()
+{
+ unsigned int n;
+ if(_msec >= 1000)
+ {
+ n = _msec / 1000;
+ _msec -= (1000 * n);
+ _sec += n;
+ }
+ if(_sec >= 60)
+ {
+ n = _sec / 60;
+ _sec -= (n * 60);
+ _min += n;
+ }
+ if(_min >= 60)
+ {
+ n = _min / 60;
+ _min -= (n * 60);
+ _hour += n;
+ }
+}
+
+
+WebRtc_Word16
+ChooseCodec(
+ CodecInst& codecInst)
+{
+
+ PrintCodecs();
+ //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
+ WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
+ WebRtc_Word8 codecID;
+ bool outOfRange = false;
+ char myStr[15] = "";
+ do
+ {
+ printf("\nChoose a codec [0]: ");
+ EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+ codecID = atoi(myStr);
+ if((codecID < 0) || (codecID >= noCodec))
+ {
+ printf("\nOut of range.\n");
+ outOfRange = true;
+ }
+ } while(outOfRange);
+
+ CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, codecInst));
+ return 0;
+}
+
+void
+PrintCodecs()
+{
+ WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
+
+ CodecInst codecInst;
+ printf("No Name [Hz] [bps]\n");
+ for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
+ {
+ AudioCodingModule::Codec(codecCntr, codecInst);
+ printf("%2d- %-18s %5d %6d\n",
+ codecCntr, codecInst.plname, codecInst.plfreq, codecInst.rate);
+ }
+
+}
+
+CircularBuffer::CircularBuffer(WebRtc_UWord32 len):
+_buff(NULL),
+_idx(0),
+_buffIsFull(false),
+_calcAvg(false),
+_calcVar(false),
+_sum(0),
+_sumSqr(0)
+{
+ _buff = new double[len];
+ if(_buff == NULL)
+ {
+ _buffLen = 0;
+ }
+ else
+ {
+ for(WebRtc_UWord32 n = 0; n < len; n++)
+ {
+ _buff[n] = 0;
+ }
+ _buffLen = len;
+ }
+}
+
+CircularBuffer::~CircularBuffer()
+{
+ if(_buff != NULL)
+ {
+ delete [] _buff;
+ _buff = NULL;
+ }
+}
+
+void
+CircularBuffer::Update(
+ const double newVal)
+{
+ assert(_buffLen > 0);
+
+ // store the value that is going to be overwritten
+ double oldVal = _buff[_idx];
+ // record the new value
+ _buff[_idx] = newVal;
+ // increment the index, to point to where we would
+ // write next
+ _idx++;
+ // it is a circular buffer, if we are at the end
+ // we have to cycle to the beginning
+ if(_idx >= _buffLen)
+ {
+ // flag that the buffer is filled up.
+ _buffIsFull = true;
+ _idx = 0;
+ }
+
+ // Update
+
+ if(_calcAvg)
+ {
+ // for the average we have to update
+ // the sum
+ _sum += (newVal - oldVal);
+ }
+
+ if(_calcVar)
+ {
+ // to calculate variance we have to update
+ // the sum of squares
+ _sumSqr += (double)(newVal - oldVal) * (double)(newVal + oldVal);
+ }
+}
+
+void
+CircularBuffer::SetArithMean(
+ bool enable)
+{
+ assert(_buffLen > 0);
+
+ if(enable && !_calcAvg)
+ {
+ WebRtc_UWord32 lim;
+ if(_buffIsFull)
+ {
+ lim = _buffLen;
+ }
+ else
+ {
+ lim = _idx;
+ }
+ _sum = 0;
+ for(WebRtc_UWord32 n = 0; n < lim; n++)
+ {
+ _sum += _buff[n];
+ }
+ }
+ _calcAvg = enable;
+}
+
+void
+CircularBuffer::SetVariance(
+ bool enable)
+{
+ assert(_buffLen > 0);
+
+ if(enable && !_calcVar)
+ {
+ WebRtc_UWord32 lim;
+ if(_buffIsFull)
+ {
+ lim = _buffLen;
+ }
+ else
+ {
+ lim = _idx;
+ }
+ _sumSqr = 0;
+ for(WebRtc_UWord32 n = 0; n < lim; n++)
+ {
+ _sumSqr += _buff[n] * _buff[n];
+ }
+ }
+ _calcAvg = enable;
+}
+
+WebRtc_Word16
+CircularBuffer::ArithMean(double& mean)
+{
+ assert(_buffLen > 0);
+
+ if(_buffIsFull)
+ {
+
+ mean = _sum / (double)_buffLen;
+ return 0;
+ }
+ else
+ {
+ if(_idx > 0)
+ {
+ mean = _sum / (double)_idx;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+
+ }
+}
+
+WebRtc_Word16
+CircularBuffer::Variance(double& var)
+{
+ assert(_buffLen > 0);
+
+ if(_buffIsFull)
+ {
+ var = _sumSqr / (double)_buffLen;
+ return 0;
+ }
+ else
+ {
+ if(_idx > 0)
+ {
+ var = _sumSqr / (double)_idx;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+ }
+}
+
+
+
+bool
+FixedPayloadTypeCodec(const char* payloadName)
+{
+ char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = {
+ "PCMU",
+ "PCMA",
+ "GSM",
+ "G723",
+ "DVI4",
+ "LPC",
+ "PCMA",
+ "G722",
+ "QCELP",
+ "CN",
+ "MPA",
+ "G728",
+ "G729"
+ };
+
+ for(int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++)
+ {
+ if(!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n]))
+ {
+ return true;
+ }
+ }
+ return false;
+}
+
+DTMFDetector::DTMFDetector()
+{
+ for(WebRtc_Word16 n = 0; n < 1000; n++)
+ {
+ _toneCntr[n] = 0;
+ }
+}
+
+DTMFDetector::~DTMFDetector()
+{
+}
+
+WebRtc_Word32 DTMFDetector::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool /* toneEnded */)
+{
+ fprintf(stdout, "%d-",digitDtmf);
+ _toneCntr[digitDtmf]++;
+ return 0;
+}
+
+void DTMFDetector::PrintDetectedDigits()
+{
+ for(WebRtc_Word16 n = 0; n < 1000; n++)
+ {
+ if(_toneCntr[n] > 0)
+ {
+ fprintf(stdout, "%d %u msec, \n", n, _toneCntr[n]*10);
+ }
+ }
+ fprintf(stdout, "\n");
+ return;
+}
+
+void
+VADCallback::Reset()
+{
+ for(int n = 0; n < 6; n++)
+ {
+ _numFrameTypes[n] = 0;
+ }
+}
+
+VADCallback::VADCallback()
+{
+ for(int n = 0; n < 6; n++)
+ {
+ _numFrameTypes[n] = 0;
+ }
+}
+
+void
+VADCallback::PrintFrameTypes()
+{
+ fprintf(stdout, "No encoding.................. %d\n", _numFrameTypes[0]);
+ fprintf(stdout, "Active normal encoded........ %d\n", _numFrameTypes[1]);
+ fprintf(stdout, "Passive normal encoded....... %d\n", _numFrameTypes[2]);
+ fprintf(stdout, "Passive DTX wideband......... %d\n", _numFrameTypes[3]);
+ fprintf(stdout, "Passive DTX narrowband....... %d\n", _numFrameTypes[4]);
+ fprintf(stdout, "Passive DTX super-wideband... %d\n", _numFrameTypes[5]);
+}
+
+WebRtc_Word32
+VADCallback::InFrameType(
+ WebRtc_Word16 frameType)
+{
+ _numFrameTypes[frameType]++;
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/utility.h b/trunk/src/modules/audio_coding/main/test/utility.h
new file mode 100644
index 0000000..887c735
--- /dev/null
+++ b/trunk/src/modules/audio_coding/main/test/utility.h
@@ -0,0 +1,195 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef ACM_TEST_UTILITY_H
+#define ACM_TEST_UTILITY_H
+
+#include "audio_coding_module.h"
+#include "gtest/gtest.h"
+
+namespace webrtc {
+
+//-----------------------------
+#define CHECK_ERROR(f) \
+ do { \
+ EXPECT_GE(f, 0) << "Error Calling API"; \
+ }while(0)
+
+//-----------------------------
+#define CHECK_PROTECTED(f) \
+ do { \
+ if(f >= 0) { \
+ ADD_FAILURE() << "Error Calling API"; \
+ } \
+ else { \
+ printf("An expected error is caught.\n"); \
+ } \
+ }while(0)
+
+//----------------------------
+#define CHECK_ERROR_MT(f) \
+ do { \
+ if(f < 0) { \
+ fprintf(stderr, "Error Calling API in file %s at line %d \n", \
+ __FILE__, __LINE__); \
+ } \
+ }while(0)
+
+//----------------------------
+#define CHECK_PROTECTED_MT(f) \
+ do { \
+ if(f >= 0) { \
+ fprintf(stderr, "Error Calling API in file %s at line %d \n", \
+ __FILE__, __LINE__); \
+ } \
+ else { \
+ printf("An expected error is caught.\n"); \
+ } \
+ }while(0)
+
+
+
+#ifdef WIN32
+ /* Exclude rarely-used stuff from Windows headers */
+ //#define WIN32_LEAN_AND_MEAN
+ /* OS-dependent case-insensitive string comparison */
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+ /* OS-dependent case-insensitive string comparison */
+ #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+#define DESTROY_ACM(acm) \
+ do { \
+ if(acm != NULL) { \
+ AudioCodingModule::Destroy(acm); \
+ acm = NULL; \
+ } \
+ } while(0)
+
+
+#define DELETE_POINTER(p) \
+ do { \
+ if(p != NULL) { \
+ delete p; \
+ p = NULL; \
+ } \
+ } while(0)
+
+class ACMTestTimer
+{
+public:
+ ACMTestTimer();
+ ~ACMTestTimer();
+
+ void Reset();
+ void Tick10ms();
+ void Tick1ms();
+ void Tick100ms();
+ void Tick1sec();
+ void CurrentTimeHMS(
+ char* currTime);
+ void CurrentTime(
+ unsigned long& h,
+ unsigned char& m,
+ unsigned char& s,
+ unsigned short& ms);
+
+private:
+ void Adjust();
+
+ unsigned short _msec;
+ unsigned char _sec;
+ unsigned char _min;
+ unsigned long _hour;
+};
+
+
+
+class CircularBuffer
+{
+public:
+ CircularBuffer(WebRtc_UWord32 len);
+ ~CircularBuffer();
+
+ void SetArithMean(
+ bool enable);
+ void SetVariance(
+ bool enable);
+
+ void Update(
+ const double newVal);
+ void IsBufferFull();
+
+ WebRtc_Word16 Variance(double& var);
+ WebRtc_Word16 ArithMean(double& mean);
+
+protected:
+ double* _buff;
+ WebRtc_UWord32 _idx;
+ WebRtc_UWord32 _buffLen;
+
+ bool _buffIsFull;
+ bool _calcAvg;
+ bool _calcVar;
+ double _sum;
+ double _sumSqr;
+};
+
+
+
+
+
+WebRtc_Word16 ChooseCodec(
+ CodecInst& codecInst);
+
+void PrintCodecs();
+
+bool FixedPayloadTypeCodec(const char* payloadName);
+
+
+
+
+class DTMFDetector: public AudioCodingFeedback
+{
+public:
+ DTMFDetector();
+ ~DTMFDetector();
+ // used for inband DTMF detection
+ WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool toneEnded);
+ void PrintDetectedDigits();
+
+private:
+ WebRtc_UWord32 _toneCntr[1000];
+
+};
+
+
+
+
+class VADCallback : public ACMVADCallback
+{
+public:
+ VADCallback();
+ ~VADCallback(){}
+
+ WebRtc_Word32 InFrameType(
+ WebRtc_Word16 frameType);
+
+ void PrintFrameTypes();
+ void Reset();
+
+private:
+ WebRtc_UWord32 _numFrameTypes[6];
+};
+
+} // namespace webrtc
+
+#endif // ACM_TEST_UTILITY_H
diff --git a/trunk/src/modules/audio_coding/neteq/Android.mk b/trunk/src/modules/audio_coding/neteq/Android.mk
new file mode 100644
index 0000000..84267be
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/Android.mk
@@ -0,0 +1,73 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_neteq
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ accelerate.c \
+ automode.c \
+ bgn_update.c \
+ bufstats_decision.c \
+ cng_internal.c \
+ codec_db.c \
+ correlator.c \
+ dsp.c \
+ dsp_helpfunctions.c \
+ dtmf_buffer.c \
+ dtmf_tonegen.c \
+ expand.c \
+ mcu_address_init.c \
+ mcu_dsp_common.c \
+ mcu_reset.c \
+ merge.c \
+ min_distortion.c \
+ mix_voice_unvoice.c \
+ mute_signal.c \
+ normal.c \
+ packet_buffer.c \
+ peak_detection.c \
+ preemptive_expand.c \
+ random_vector.c \
+ recin.c \
+ recout.c \
+ rtcp.c \
+ rtp.c \
+ set_fs.c \
+ signal_mcu.c \
+ split_and_insert.c \
+ unmute_signal.c \
+ webrtc_neteq.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DNETEQ_VOICEENGINE_CODECS'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../codecs/cng/include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/neteq/OWNERS b/trunk/src/modules/audio_coding/neteq/OWNERS
new file mode 100644
index 0000000..1d25542
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/OWNERS
@@ -0,0 +1,2 @@
+henrik.lundin@webrtc.org
+tina.legrand@webrtc.org
diff --git a/trunk/src/modules/audio_coding/neteq/accelerate.c b/trunk/src/modules/audio_coding/neteq/accelerate.c
new file mode 100644
index 0000000..285de4d
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/accelerate.c
@@ -0,0 +1,489 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the Accelerate algorithm that is used to reduce
+ * the delay by removing a part of the audio stream.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+
+#define ACCELERATE_CORR_LEN 50
+#define ACCELERATE_MIN_LAG 10
+#define ACCELERATE_MAX_LAG 60
+#define ACCELERATE_DOWNSAMPLED_LEN (ACCELERATE_CORR_LEN + ACCELERATE_MAX_LAG)
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_downSampSpeech 110 0 109
+ WebRtc_Word32 pw32_corr 2*50 110 209
+ WebRtc_Word16 pw16_corr 50 0 49
+
+ Total: 110+2*50
+ */
+
+#define SCRATCH_PW16_DS_SPEECH 0
+#define SCRATCH_PW32_CORR ACCELERATE_DOWNSAMPLED_LEN
+#define SCRATCH_PW16_CORR 0
+
+/****************************************************************************
+ * WebRtcNetEQ_Accelerate(...)
+ *
+ * This function tries to shorten the audio data by removing one or several
+ * pitch periods. The operation is only carried out if the correlation is
+ * strong or if the signal energy is very low.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to newly decoded speech.
+ * - len : Length of decoded speech.
+ * - BGNonly : If non-zero, Accelerate will only remove the last
+ * DEFAULT_TIME_ADJUST seconds of the input.
+ * No signal matching is done.
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outData : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Number of samples written to outData.
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ const WebRtc_Word16 *pw16_decoded, int len,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly)
+{
+
+#ifdef SCRATCH
+ /* Use scratch memory for internal temporary vectors */
+ WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
+ WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
+ WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
+#else
+ /* Allocate memory for temporary vectors */
+ WebRtc_Word16 pw16_downSampSpeech[ACCELERATE_DOWNSAMPLED_LEN];
+ WebRtc_Word32 pw32_corr[ACCELERATE_CORR_LEN];
+ WebRtc_Word16 pw16_corr[ACCELERATE_CORR_LEN];
+#endif
+ WebRtc_Word16 w16_decodedMax = 0;
+ WebRtc_Word16 w16_tmp;
+ WebRtc_Word16 w16_tmp2;
+ WebRtc_Word32 w32_tmp;
+ WebRtc_Word32 w32_tmp2;
+
+ const WebRtc_Word16 w16_startLag = ACCELERATE_MIN_LAG;
+ const WebRtc_Word16 w16_endLag = ACCELERATE_MAX_LAG;
+ const WebRtc_Word16 w16_corrLen = ACCELERATE_CORR_LEN;
+ const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
+ WebRtc_Word16 *pw16_vectmp;
+ WebRtc_Word16 w16_inc, w16_startfact;
+ WebRtc_Word16 w16_bestIndex, w16_bestVal;
+ WebRtc_Word16 w16_VAD = 1;
+ WebRtc_Word16 fsMult;
+ WebRtc_Word16 fsMult120;
+ WebRtc_Word32 w32_en1, w32_en2, w32_cc;
+ WebRtc_Word16 w16_en1, w16_en2;
+ WebRtc_Word16 w16_en1Scale, w16_en2Scale;
+ WebRtc_Word16 w16_sqrtEn1En2;
+ WebRtc_Word16 w16_bestCorr = 0;
+ int ok;
+
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo *msInfo = inst->msInfo;
+#endif
+
+ fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
+
+ /* Pre-calculate common multiplication with fsMult */
+ fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
+
+ inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
+
+ /* Sanity check for len variable; must be (almost) 30 ms
+ (120*fsMult + max(bestIndex)) */
+ if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult))
+ {
+ /* Length of decoded data too short */
+ inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
+ *pw16_len = len;
+
+ /* simply move all data from decoded to outData */
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /***********************************/
+ /* Special operations for BGN only */
+ /***********************************/
+
+ /* Check if "background noise only" flag is set */
+ if (BGNonly)
+ {
+ /* special operation for BGN only; simply remove a chunk of data */
+ w16_bestIndex = DEFAULT_TIME_ADJUST * WEBRTC_SPL_LSHIFT_W16(fsMult, 3); /* X*fs/1000 */
+
+ /* Sanity check for bestIndex */
+ if (w16_bestIndex > len)
+ { /* not good, do nothing instead */
+ inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
+ *pw16_len = len;
+
+ /* simply move all data from decoded to outData */
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /* set length parameter */
+ *pw16_len = len - w16_bestIndex; /* we remove bestIndex samples */
+
+ /* copy to output */
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, *pw16_len);
+
+ /* set mode */
+ inst->w16_mode = MODE_LOWEN_ACCELERATE;
+
+ /* update statistics */
+ inst->statInst.accelerateLength += w16_bestIndex;
+
+ return 0;
+ } /* end of special code for BGN mode */
+
+#ifdef NETEQ_STEREO
+
+ /* Sanity for msInfo */
+ if (msInfo == NULL)
+ {
+ /* this should not happen here */
+ return MASTER_SLAVE_ERROR;
+ }
+
+ if (msInfo->msMode != NETEQ_SLAVE)
+ {
+ /* Find correlation lag only for non-slave instances */
+
+#endif
+
+ /****************************************************************/
+ /* Find the strongest correlation lag by downsampling to 4 kHz, */
+ /* calculating correlation for downsampled signal and finding */
+ /* the strongest correlation peak. */
+ /****************************************************************/
+
+ /* find maximum absolute value */
+ w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+
+ /* downsample the decoded speech to 4 kHz */
+ ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
+ ACCELERATE_DOWNSAMPLED_LEN, 1 /* compensate delay*/);
+ if (ok != 0)
+ {
+ /* error */
+ inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
+ *pw16_len = len;
+ /* simply move all data from decoded to outData */
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /*
+ * Set scaling factor for cross correlation to protect against overflow
+ * (log2(50) => 6)
+ */
+ w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */
+ WebRtcNetEQ_CrossCorr(
+ pw32_corr, &pw16_downSampSpeech[w16_endLag],
+ &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
+ (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
+
+ /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+ w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
+ w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp);
+
+#ifdef NETEQ_STEREO
+ } /* end if (msInfo->msMode != NETEQ_SLAVE) */
+
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+ /* Find the strongest correlation peak by using the parabolic fit method */
+ WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
+ &w16_bestIndex, &w16_bestVal);
+ /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
+
+ /* Compensate bestIndex for displaced starting position */
+ w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
+ /* 20*fsMult <= bestIndex <= 119*fsMult */
+
+ msInfo->bestIndex = w16_bestIndex;
+ }
+ else if (msInfo->msMode == NETEQ_SLAVE)
+ {
+ if (msInfo->extraInfo == ACC_FAIL)
+ {
+ /* Master has signaled an unsuccessful accelerate */
+ w16_bestIndex = 0;
+ }
+ else
+ {
+ /* Get best index from master */
+ w16_bestIndex = msInfo->bestIndex;
+ }
+ }
+ else
+ {
+ /* Invalid mode */
+ return MASTER_SLAVE_ERROR;
+ }
+
+#else /* NETEQ_STEREO */
+
+ /* Find the strongest correlation peak by using the parabolic fit method */
+ WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
+ &w16_bestIndex, &w16_bestVal);
+ /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
+
+ /* Compensate bestIndex for displaced starting position */
+ w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
+ /* 20*fsMult <= bestIndex <= 119*fsMult */
+
+#endif /* NETEQ_STEREO */
+
+#ifdef NETEQ_STEREO
+
+ if (msInfo->msMode != NETEQ_SLAVE)
+ {
+ /* Calculate correlation only for non-slave instances */
+
+#endif /* NETEQ_STEREO */
+
+ /*****************************************************/
+ /* Calculate correlation bestCorr for the found lag. */
+ /* Also do a simple VAD decision. */
+ /*****************************************************/
+
+ /*
+ * Calculate scaling to ensure that bestIndex samples can be square-summed
+ * without overflowing
+ */
+ w16_tmp = (31
+ - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)));
+ w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex));
+ w16_tmp -= 31;
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ /* vec1 starts at 15 ms minus one pitch period */
+ pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
+ /* vec2 start at 15 ms */
+ pw16_vec2 = &pw16_decoded[fsMult120];
+
+ /* Calculate energies for vec1 and vec2 */
+ w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
+ (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
+ w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
+ (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
+
+ /* Calculate cross-correlation at the found lag */
+ w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
+ w16_bestIndex, w16_tmp);
+
+ /* Check VAD constraint
+ ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
+ w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */
+ if (inst->BGNInst.w16_initialized == 1)
+ {
+ w32_tmp2 = inst->BGNInst.w32_energy;
+ }
+ else
+ {
+ /* if BGN parameters have not been estimated, use a fixed threshold */
+ w32_tmp2 = 75000;
+ }
+ w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
+ w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
+ w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
+ w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
+ w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
+
+ /* Scale w32_tmp properly before comparing with w32_tmp2 */
+ /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */
+ if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
+ {
+ /* Cannot scale only w32_tmp, must scale w32_temp2 too */
+ WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
+ w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
+ WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
+ }
+ else
+ {
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp,
+ WEBRTC_SPL_LSHIFT_W32(w16_tmp,1));
+ }
+
+ if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
+ {
+ /* The signal seems to be passive speech */
+ w16_VAD = 0;
+ w16_bestCorr = 0; /* Correlation does not matter */
+ }
+ else
+ {
+ /* The signal is active speech */
+ w16_VAD = 1;
+
+ /* Calculate correlation (cc/sqrt(en1*en2)) */
+
+ /* Start with calculating scale values */
+ w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
+ w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
+ w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
+ w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
+
+ /* Make sure total scaling is even (to simplify scale factor after sqrt) */
+ if ((w16_en1Scale + w16_en2Scale) & 1)
+ {
+ w16_en1Scale += 1;
+ }
+
+ /* Convert energies to WebRtc_Word16 */
+ w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+ w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+
+ /* Calculate energy product */
+ w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
+
+ /* Calculate square-root of energy product */
+ w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
+
+ /* Calculate cc/sqrt(en1*en2) in Q14 */
+ w16_tmp = 14 - WEBRTC_SPL_RSHIFT_W16(w16_en1Scale+w16_en2Scale, 1);
+ w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
+ w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
+ w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
+ w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
+ }
+
+#ifdef NETEQ_STEREO
+
+ } /* end if (msInfo->msMode != NETEQ_SLAVE) */
+
+#endif /* NETEQ_STEREO */
+
+ /************************************************/
+ /* Check accelerate criteria and remove samples */
+ /************************************************/
+
+ /* Check for strong correlation (>0.9) or passive speech */
+#ifdef NETEQ_STEREO
+ if ((((w16_bestCorr > 14746) || (w16_VAD == 0)) && (msInfo->msMode != NETEQ_SLAVE))
+ || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->extraInfo != ACC_FAIL)))
+#else
+ if ((w16_bestCorr > 14746) || (w16_VAD == 0))
+#endif
+ {
+ /* Do accelerate operation by overlap add */
+
+ /*
+ * Calculate cross-fading slope so that the fading factor goes from
+ * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
+ */
+ w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
+ (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
+
+ /* Initiate fading factor */
+ w16_startfact = 16384 - w16_inc;
+
+ /* vec1 starts at 15 ms minus one pitch period */
+ pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
+ /* vec2 start at 15 ms */
+ pw16_vec2 = &pw16_decoded[fsMult120];
+
+ /* Copy unmodified part [0 to 15 ms minus 1 pitch period] */
+ w16_tmp = (fsMult120 - w16_bestIndex);
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_tmp);
+
+ /* Generate interpolated part of length bestIndex (1 pitch period) */
+ pw16_vectmp = pw16_outData + w16_tmp; /* start of interpolation output */
+ /* Reuse mixing function from Expand */
+ WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec1,
+ (WebRtc_Word16*) pw16_vec2, &w16_startfact, w16_inc, w16_bestIndex);
+
+ /* Move the last part (also unmodified) */
+ /* Take from decoded at 15 ms + 1 pitch period */
+ pw16_vec2 = &pw16_decoded[fsMult120 + w16_bestIndex];
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[fsMult120], pw16_vec2,
+ (WebRtc_Word16) (len - fsMult120 - w16_bestIndex));
+
+ /* Set the mode flag */
+ if (w16_VAD)
+ {
+ inst->w16_mode = MODE_SUCCESS_ACCELERATE;
+ }
+ else
+ {
+ inst->w16_mode = MODE_LOWEN_ACCELERATE;
+ }
+
+ /* Calculate resulting length = original length - pitch period */
+ *pw16_len = len - w16_bestIndex;
+
+ /* Update in-call statistics */
+ inst->statInst.accelerateLength += w16_bestIndex;
+
+ return 0;
+ }
+ else
+ {
+ /* Accelerate not allowed */
+
+#ifdef NETEQ_STEREO
+ /* Signal to slave(s) that this was unsuccessful */
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ msInfo->extraInfo = ACC_FAIL;
+ }
+#endif
+
+ /* Set mode flag to unsuccessful accelerate */
+ inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
+
+ /* Length is unmodified */
+ *pw16_len = len;
+
+ /* Simply move all data from decoded to outData */
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return 0;
+ }
+}
+
+#undef SCRATCH_PW16_DS_SPEECH
+#undef SCRATCH_PW32_CORR
+#undef SCRATCH_PW16_CORR
diff --git a/trunk/src/modules/audio_coding/neteq/automode.c b/trunk/src/modules/audio_coding/neteq/automode.c
new file mode 100644
index 0000000..da127e0
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/automode.c
@@ -0,0 +1,736 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the implementation of automatic buffer level optimization.
+ */
+
+#include "automode.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+
+#include "neteq_defines.h"
+
+#ifdef NETEQ_DELAY_LOGGING
+/* special code for offline delay logging */
+#include <stdio.h>
+#include "delay_logging.h"
+
+extern FILE *delay_fid2; /* file pointer to delay log file */
+#endif /* NETEQ_DELAY_LOGGING */
+
+
+int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
+ WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
+ WebRtc_Word32 fsHz, int mdCodec, int streamingMode)
+{
+ WebRtc_UWord32 timeIat; /* inter-arrival time */
+ int i;
+ WebRtc_Word32 tempsum = 0; /* temp summation */
+ WebRtc_Word32 tempvar; /* temporary variable */
+ int retval = 0; /* return value */
+ WebRtc_Word16 packetLenSamp; /* packet speech length in samples */
+
+ /****************/
+ /* Sanity check */
+ /****************/
+
+ if (maxBufLen <= 1 || fsHz <= 0)
+ {
+ /* maxBufLen must be at least 2 and fsHz must both be strictly positive */
+ return -1;
+ }
+
+ /****************************/
+ /* Update packet statistics */
+ /****************************/
+
+ /* Try calculating packet length from current and previous timestamps */
+ if ((timeStamp <= inst->lastTimeStamp) || (seqNumber <= inst->lastSeqNo))
+ {
+ /* Wrong timestamp or sequence order; revert to backup plan */
+ packetLenSamp = inst->packetSpeechLenSamp; /* use stored value */
+ }
+ else if (timeStamp > inst->lastTimeStamp)
+ {
+ /* calculate timestamps per packet */
+ packetLenSamp = (WebRtc_Word16) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp,
+ seqNumber - inst->lastSeqNo);
+ }
+
+ /* Check that the packet size is positive; if not, the statistics cannot be updated. */
+ if (packetLenSamp > 0)
+ { /* packet size ok */
+
+ /* calculate inter-arrival time in integer packets (rounding down) */
+ timeIat = WebRtcSpl_DivW32W16(inst->packetIatCountSamp, packetLenSamp);
+
+ /* Special operations for streaming mode */
+ if (streamingMode != 0)
+ {
+ /*
+ * Calculate IAT in Q8, including fractions of a packet (i.e., more accurate
+ * than timeIat).
+ */
+ WebRtc_Word16 timeIatQ8 = (WebRtc_Word16) WebRtcSpl_DivW32W16(
+ WEBRTC_SPL_LSHIFT_W32(inst->packetIatCountSamp, 8), packetLenSamp);
+
+ /*
+ * Calculate cumulative sum iat with sequence number compensation (ideal arrival
+ * times makes this sum zero).
+ */
+ inst->cSumIatQ8 += (timeIatQ8
+ - WEBRTC_SPL_LSHIFT_W32(seqNumber - inst->lastSeqNo, 8));
+
+ /* subtract drift term */
+ inst->cSumIatQ8 -= CSUM_IAT_DRIFT;
+
+ /* ensure not negative */
+ inst->cSumIatQ8 = WEBRTC_SPL_MAX(inst->cSumIatQ8, 0);
+
+ /* remember max */
+ if (inst->cSumIatQ8 > inst->maxCSumIatQ8)
+ {
+ inst->maxCSumIatQ8 = inst->cSumIatQ8;
+ inst->maxCSumUpdateTimer = 0;
+ }
+
+ /* too long since the last maximum was observed; decrease max value */
+ if (inst->maxCSumUpdateTimer > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
+ MAX_STREAMING_PEAK_PERIOD))
+ {
+ inst->maxCSumIatQ8 -= 4; /* remove 1000*4/256 = 15.6 ms/s */
+ }
+ } /* end of streaming mode */
+
+ /* check for discontinuous packet sequence and re-ordering */
+ if (seqNumber > inst->lastSeqNo + 1)
+ {
+ /* Compensate for gap in the sequence numbers.
+ * Reduce IAT with expected extra time due to lost packets, but ensure that
+ * the IAT is not negative.
+ */
+ timeIat -= WEBRTC_SPL_MIN(timeIat,
+ (WebRtc_UWord32) (seqNumber - inst->lastSeqNo - 1));
+ }
+ else if (seqNumber < inst->lastSeqNo)
+ {
+ /* compensate for re-ordering */
+ timeIat += (WebRtc_UWord32) (inst->lastSeqNo + 1 - seqNumber);
+ }
+
+ /* saturate IAT at maximum value */
+ timeIat = WEBRTC_SPL_MIN( timeIat, MAX_IAT );
+
+ /* update iatProb = forgetting_factor * iatProb for all elements */
+ for (i = 0; i <= MAX_IAT; i++)
+ {
+ WebRtc_Word32 tempHi, tempLo; /* Temporary variables */
+
+ /*
+ * Multiply iatProbFact (Q15) with iatProb (Q30) and right-shift 15 steps
+ * to come back to Q30. The operation is done in two steps:
+ */
+
+ /*
+ * 1) Multiply the high 16 bits (15 bits + sign) of iatProb. Shift iatProb
+ * 16 steps right to get the high 16 bits in a WebRtc_Word16 prior to
+ * multiplication, and left-shift with 1 afterwards to come back to
+ * Q30 = (Q15 * (Q30>>16)) << 1.
+ */
+ tempHi = WEBRTC_SPL_MUL_16_16(inst->iatProbFact,
+ (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16));
+ tempHi = WEBRTC_SPL_LSHIFT_W32(tempHi, 1); /* left-shift 1 step */
+
+ /*
+ * 2) Isolate and multiply the low 16 bits of iatProb. Right-shift 15 steps
+ * afterwards to come back to Q30 = (Q15 * Q30) >> 15.
+ */
+ tempLo = inst->iatProb[i] & 0x0000FFFF; /* sift out the 16 low bits */
+ tempLo = WEBRTC_SPL_MUL_16_U16(inst->iatProbFact,
+ (WebRtc_UWord16) tempLo);
+ tempLo = WEBRTC_SPL_RSHIFT_W32(tempLo, 15);
+
+ /* Finally, add the high and low parts */
+ inst->iatProb[i] = tempHi + tempLo;
+
+ /* Sum all vector elements while we are at it... */
+ tempsum += inst->iatProb[i];
+ }
+
+ /*
+ * Increase the probability for the currently observed inter-arrival time
+ * with 1 - iatProbFact. The factor is in Q15, iatProb in Q30;
+ * hence, left-shift 15 steps to obtain result in Q30.
+ */
+ inst->iatProb[timeIat] += (32768 - inst->iatProbFact) << 15;
+
+ tempsum += (32768 - inst->iatProbFact) << 15; /* add to vector sum */
+
+ /*
+ * Update iatProbFact (changes only during the first seconds after reset)
+ * The factor converges to IAT_PROB_FACT.
+ */
+ inst->iatProbFact += (IAT_PROB_FACT - inst->iatProbFact + 3) >> 2;
+
+ /* iatProb should sum up to 1 (in Q30). */
+ tempsum -= 1 << 30; /* should be zero */
+
+ /* Check if it does, correct if it doesn't. */
+ if (tempsum > 0)
+ {
+ /* tempsum too large => decrease a few values in the beginning */
+ i = 0;
+ while (i <= MAX_IAT && tempsum > 0)
+ {
+ /* Remove iatProb[i] / 16 from iatProb, but not more than tempsum */
+ tempvar = WEBRTC_SPL_MIN(tempsum, inst->iatProb[i] >> 4);
+ inst->iatProb[i++] -= tempvar;
+ tempsum -= tempvar;
+ }
+ }
+ else if (tempsum < 0)
+ {
+ /* tempsum too small => increase a few values in the beginning */
+ i = 0;
+ while (i <= MAX_IAT && tempsum < 0)
+ {
+ /* Add iatProb[i] / 16 to iatProb, but not more than tempsum */
+ tempvar = WEBRTC_SPL_MIN(-tempsum, inst->iatProb[i] >> 4);
+ inst->iatProb[i++] += tempvar;
+ tempsum += tempvar;
+ }
+ }
+
+ /* Calculate optimal buffer level based on updated statistics */
+ tempvar = (WebRtc_Word32) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat,
+ streamingMode);
+ if (tempvar > 0)
+ {
+ inst->optBufLevel = (WebRtc_UWord16) tempvar;
+
+ if (streamingMode != 0)
+ {
+ inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel,
+ inst->maxCSumIatQ8);
+ }
+
+ /*********/
+ /* Limit */
+ /*********/
+
+ /* Subtract extra delay from maxBufLen */
+ if (inst->extraDelayMs > 0 && inst->packetSpeechLenSamp > 0)
+ {
+ maxBufLen -= inst->extraDelayMs / inst->packetSpeechLenSamp * fsHz / 1000;
+ maxBufLen = WEBRTC_SPL_MAX(maxBufLen, 1); // sanity: at least one packet
+ }
+
+ maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */
+
+ /* Enforce upper limit; 75% of maxBufLen */
+ inst->optBufLevel = (WebRtc_UWord16) WEBRTC_SPL_MIN( inst->optBufLevel,
+ (maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */
+ }
+ else
+ {
+ retval = (int) tempvar;
+ }
+
+ } /* end if */
+
+ /*******************************/
+ /* Update post-call statistics */
+ /*******************************/
+
+ /* Calculate inter-arrival time in ms = packetIatCountSamp / (fsHz / 1000) */
+ timeIat = WEBRTC_SPL_UDIV(
+ WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (WebRtc_Word16) 1000),
+ (WebRtc_UWord32) fsHz);
+
+ /* Increase counter corresponding to current inter-arrival time */
+ if (timeIat > 2000)
+ {
+ inst->countIAT2000ms++;
+ }
+ else if (timeIat > 1000)
+ {
+ inst->countIAT1000ms++;
+ }
+ else if (timeIat > 500)
+ {
+ inst->countIAT500ms++;
+ }
+
+ if (timeIat > inst->longestIATms)
+ {
+ /* update maximum value */
+ inst->longestIATms = timeIat;
+ }
+
+ /***********************************/
+ /* Prepare for next packet arrival */
+ /***********************************/
+
+ inst->packetIatCountSamp = 0; /* reset inter-arrival time counter */
+
+ inst->lastSeqNo = seqNumber; /* remember current sequence number */
+
+ inst->lastTimeStamp = timeStamp; /* remember current timestamp */
+
+ return retval;
+}
+
+
+WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
+ int mdCodec, WebRtc_UWord32 timeIatPkts,
+ int streamingMode)
+{
+
+ WebRtc_Word32 sum1 = 1 << 30; /* assign to 1 in Q30 */
+ WebRtc_Word16 B;
+ WebRtc_UWord16 Bopt;
+ int i;
+ WebRtc_Word32 betaInv; /* optimization parameter */
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ int temp_var;
+#endif
+
+ /****************/
+ /* Sanity check */
+ /****************/
+
+ if (fsHz <= 0)
+ {
+ /* fsHz must be strictly positive */
+ return -1;
+ }
+
+ /***********************************************/
+ /* Get betaInv parameter based on playout mode */
+ /***********************************************/
+
+ if (streamingMode)
+ {
+ /* streaming (listen-only) mode */
+ betaInv = AUTOMODE_STREAMING_BETA_INV_Q30;
+ }
+ else
+ {
+ /* normal mode */
+ betaInv = AUTOMODE_BETA_INV_Q30;
+ }
+
+ /*******************************************************************/
+ /* Calculate optimal buffer level without considering jitter peaks */
+ /*******************************************************************/
+
+ /*
+ * Find the B for which the probability of observing an inter-arrival time larger
+ * than or equal to B is less than or equal to betaInv.
+ */
+ B = 0; /* start from the beginning of iatProb */
+ sum1 -= inst->iatProb[B]; /* ensure that optimal level is not less than 1 */
+
+ do
+ {
+ /*
+ * Subtract the probabilities one by one until the sum is no longer greater
+ * than betaInv.
+ */
+ sum1 -= inst->iatProb[++B];
+ }
+ while ((sum1 > betaInv) && (B < MAX_IAT));
+
+ Bopt = B; /* This is our primary value for the optimal buffer level Bopt */
+
+ if (mdCodec)
+ {
+ /*
+ * Use alternative cost function when multiple description codec is in use.
+ * Do not have to re-calculate all points, just back off a few steps from
+ * previous value of B.
+ */
+ WebRtc_Word32 sum2 = sum1; /* copy sum1 */
+
+ while ((sum2 <= betaInv + inst->iatProb[Bopt]) && (Bopt > 0))
+ {
+ /* Go backwards in the sum until the modified cost function solution is found */
+ sum2 += inst->iatProb[Bopt--];
+ }
+
+ Bopt++; /* This is the optimal level when using an MD codec */
+
+ /* Now, Bopt and B can have different values. */
+ }
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF;
+ fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
+ temp_var = (int) (Bopt * inst->packetSpeechLenSamp);
+#endif
+
+ /******************************************************************/
+ /* Make levelFiltFact adaptive: Larger B <=> larger levelFiltFact */
+ /******************************************************************/
+
+ switch (B)
+ {
+ case 0:
+ case 1:
+ {
+ inst->levelFiltFact = 251;
+ break;
+ }
+ case 2:
+ case 3:
+ {
+ inst->levelFiltFact = 252;
+ break;
+ }
+ case 4:
+ case 5:
+ case 6:
+ case 7:
+ {
+ inst->levelFiltFact = 253;
+ break;
+ }
+ default: /* B > 7 */
+ {
+ inst->levelFiltFact = 254;
+ break;
+ }
+ }
+
+ /************************/
+ /* Peak mode operations */
+ /************************/
+
+ /* Compare current IAT with peak threshold
+ *
+ * If IAT > optimal level + threshold (+1 for MD codecs)
+ * or if IAT > 2 * optimal level (note: optimal level is in Q8):
+ */
+ if (timeIatPkts > (WebRtc_UWord32) (Bopt + inst->peakThresholdPkt + (mdCodec != 0))
+ || timeIatPkts > (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_U16(Bopt, 1))
+ {
+ /* A peak is observed */
+
+ if (inst->peakIndex == -1)
+ {
+ /* this is the first peak; prepare for next peak */
+ inst->peakIndex = 0;
+ /* set the mode-disable counter */
+ inst->peakModeDisabled = WEBRTC_SPL_LSHIFT_W16(1, NUM_PEAKS_REQUIRED-2);
+ }
+ else if (inst->peakIatCountSamp
+ <=
+ (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD))
+ {
+ /* This is not the first peak and the period time is valid */
+
+ /* store time elapsed since last peak */
+ inst->peakPeriodSamp[inst->peakIndex] = inst->peakIatCountSamp;
+
+ /* saturate height to 16 bits */
+ inst->peakHeightPkt[inst->peakIndex]
+ =
+ (WebRtc_Word16) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX);
+
+ /* increment peakIndex and wrap/modulo */
+ inst->peakIndex = (inst->peakIndex + 1) & PEAK_INDEX_MASK;
+
+ /* process peak vectors */
+ inst->curPeakHeight = 0;
+ inst->curPeakPeriod = 0;
+
+ for (i = 0; i < NUM_PEAKS; i++)
+ {
+ /* Find maximum of peak heights and peak periods */
+ inst->curPeakHeight
+ = WEBRTC_SPL_MAX(inst->curPeakHeight, inst->peakHeightPkt[i]);
+ inst->curPeakPeriod
+ = WEBRTC_SPL_MAX(inst->curPeakPeriod, inst->peakPeriodSamp[i]);
+
+ }
+
+ inst->peakModeDisabled >>= 1; /* decrease mode-disable "counter" */
+
+ }
+ else if (inst->peakIatCountSamp > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
+ WEBRTC_SPL_LSHIFT_W16(MAX_PEAK_PERIOD, 1)))
+ {
+ /*
+ * More than 2 * MAX_PEAK_PERIOD has elapsed since last peak;
+ * too long time => reset peak statistics
+ */
+ inst->curPeakHeight = 0;
+ inst->curPeakPeriod = 0;
+ for (i = 0; i < NUM_PEAKS; i++)
+ {
+ inst->peakHeightPkt[i] = 0;
+ inst->peakPeriodSamp[i] = 0;
+ }
+
+ inst->peakIndex = -1; /* Next peak is first peak */
+ inst->peakIatCountSamp = 0;
+ }
+
+ inst->peakIatCountSamp = 0; /* Reset peak interval timer */
+ } /* end if peak is observed */
+
+ /* Evaluate peak mode conditions */
+
+ /*
+ * If not disabled (enough peaks have been observed) and
+ * time since last peak is less than two peak periods.
+ */
+ inst->peakFound = 0;
+ if ((!inst->peakModeDisabled) && (inst->peakIatCountSamp
+ <= WEBRTC_SPL_LSHIFT_W32(inst->curPeakPeriod , 1)))
+ {
+ /* Engage peak mode */
+ inst->peakFound = 1;
+ /* Set optimal buffer level to curPeakHeight (if it's not already larger) */
+ Bopt = WEBRTC_SPL_MAX(Bopt, inst->curPeakHeight);
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ temp_var = (int) -(Bopt * inst->packetSpeechLenSamp);
+#endif
+ }
+
+ /* Scale Bopt to Q8 */
+ Bopt = WEBRTC_SPL_LSHIFT_U16(Bopt,8);
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
+#endif
+
+ /* Sanity check: Bopt must be strictly positive */
+ if (Bopt <= 0)
+ {
+ Bopt = WEBRTC_SPL_LSHIFT_W16(1, 8); /* 1 in Q8 */
+ }
+
+ return Bopt; /* return value in Q8 */
+}
+
+
+int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
+ int sampPerCall, WebRtc_Word16 fsMult)
+{
+
+ WebRtc_Word16 curSizeFrames;
+
+ /****************/
+ /* Sanity check */
+ /****************/
+
+ if (sampPerCall <= 0 || fsMult <= 0)
+ {
+ /* sampPerCall and fsMult must both be strictly positive */
+ return -1;
+ }
+
+ /* Check if packet size has been detected */
+ if (inst->packetSpeechLenSamp > 0)
+ {
+ /*
+ * Current buffer level in packet lengths
+ * = (curSizeMs8 * fsMult) / packetSpeechLenSamp
+ */
+ curSizeFrames = (WebRtc_Word16) WebRtcSpl_DivW32W16(
+ WEBRTC_SPL_MUL_32_16(curSizeMs8, fsMult), inst->packetSpeechLenSamp);
+ }
+ else
+ {
+ curSizeFrames = 0;
+ }
+
+ /* Filter buffer level */
+ if (inst->levelFiltFact > 0) /* check that filter factor is set */
+ {
+ /* Filter:
+ * buffLevelFilt = levelFiltFact * buffLevelFilt
+ * + (1-levelFiltFact) * curSizeFrames
+ *
+ * levelFiltFact is in Q8
+ */
+ inst->buffLevelFilt = (WebRtc_UWord16) (WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL_16_U16(inst->levelFiltFact, inst->buffLevelFilt), 8)
+ + WEBRTC_SPL_MUL_16_16(256 - inst->levelFiltFact, curSizeFrames));
+ }
+
+ /* Account for time-scale operations (accelerate and pre-emptive expand) */
+ if (inst->prevTimeScale)
+ {
+ /*
+ * Time-scaling has been performed since last filter update.
+ * Subtract the sampleMemory from buffLevelFilt after converting sampleMemory
+ * from samples to packets in Q8. Make sure that the filtered value is
+ * non-negative.
+ */
+ inst->buffLevelFilt = (WebRtc_UWord16) WEBRTC_SPL_MAX( inst->buffLevelFilt -
+ WebRtcSpl_DivW32W16(
+ WEBRTC_SPL_LSHIFT_W32(inst->sampleMemory, 8), /* sampleMemory in Q8 */
+ inst->packetSpeechLenSamp ), /* divide by packetSpeechLenSamp */
+ 0);
+
+ /*
+ * Reset flag and set timescaleHoldOff timer to prevent further time-scaling
+ * for some time.
+ */
+ inst->prevTimeScale = 0;
+ inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT;
+ }
+
+ /* Update time counters and HoldOff timer */
+ inst->packetIatCountSamp += sampPerCall; /* packet inter-arrival time */
+ inst->peakIatCountSamp += sampPerCall; /* peak inter-arrival time */
+ inst->timescaleHoldOff >>= 1; /* time-scaling limiter */
+ inst->maxCSumUpdateTimer += sampPerCall; /* cumulative-sum timer */
+
+ return 0;
+
+}
+
+
+int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
+ WebRtc_Word32 fsHz)
+{
+
+ /* Sanity check for newLenSamp and fsHz */
+ if (newLenSamp <= 0 || fsHz <= 0)
+ {
+ return -1;
+ }
+
+ inst->packetSpeechLenSamp = newLenSamp; /* Store packet size in instance */
+
+ /* Make NetEQ wait for first regular packet before starting the timer */
+ inst->lastPackCNGorDTMF = 1;
+
+ inst->packetIatCountSamp = 0; /* Reset packet time counter */
+
+ /*
+ * Calculate peak threshold from packet size. The threshold is defined as
+ * the (fractional) number of packets that corresponds to PEAK_HEIGHT
+ * (in Q8 seconds). That is, threshold = PEAK_HEIGHT/256 * fsHz / packLen.
+ */
+ inst->peakThresholdPkt = (WebRtc_UWord16) WebRtcSpl_DivW32W16ResW16(
+ WEBRTC_SPL_MUL_16_16_RSFT(PEAK_HEIGHT,
+ (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp);
+
+ return 0;
+}
+
+
+int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets)
+{
+
+ int i;
+ WebRtc_UWord16 tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */
+
+ /* Sanity check for maxBufLenPackets */
+ if (maxBufLenPackets <= 1)
+ {
+ /* Invalid value; set to 10 instead (arbitary small number) */
+ maxBufLenPackets = 10;
+ }
+
+ /* Reset filtered buffer level */
+ inst->buffLevelFilt = 0;
+
+ /* Reset packet size to unknown */
+ inst->packetSpeechLenSamp = 0;
+
+ /*
+ * Flag that last packet was special payload, so that automode will treat the next speech
+ * payload as the first payload received.
+ */
+ inst->lastPackCNGorDTMF = 1;
+
+ /* Reset peak detection parameters */
+ inst->peakModeDisabled = 1; /* disable peak mode */
+ inst->peakIatCountSamp = 0;
+ inst->peakIndex = -1; /* indicates that no peak is registered */
+ inst->curPeakHeight = 0;
+ inst->curPeakPeriod = 0;
+ for (i = 0; i < NUM_PEAKS; i++)
+ {
+ inst->peakHeightPkt[i] = 0;
+ inst->peakPeriodSamp[i] = 0;
+ }
+
+ /*
+ * Set the iatProb PDF vector to an exponentially decaying distribution
+ * iatProb[i] = 0.5^(i+1), i = 0, 1, 2, ...
+ * iatProb is in Q30.
+ */
+ for (i = 0; i <= MAX_IAT; i++)
+ {
+ /* iatProb[i] = 0.5^(i+1) = iatProb[i-1] / 2 */
+ tempprob = WEBRTC_SPL_RSHIFT_U16(tempprob, 1);
+ /* store in PDF vector */
+ inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) tempprob, 16);
+ }
+
+ /*
+ * Calculate the optimal buffer level corresponding to the initial PDF.
+ * No need to call WebRtcNetEQ_CalcOptimalBufLvl() since we have just hard-coded
+ * all the variables that the buffer level depends on => we know the result
+ */
+ inst->optBufLevel = WEBRTC_SPL_MIN(4,
+ (maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */
+ inst->levelFiltFact = 253;
+
+ /*
+ * Reset the iat update forgetting factor to 0 to make the impact of the first
+ * incoming packets greater.
+ */
+ inst->iatProbFact = 0;
+
+ /* Reset packet inter-arrival time counter */
+ inst->packetIatCountSamp = 0;
+
+ /* Clear time-scaling related variables */
+ inst->prevTimeScale = 0;
+ inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; /* don't allow time-scaling immediately */
+
+ inst->cSumIatQ8 = 0;
+ inst->maxCSumIatQ8 = 0;
+
+ return 0;
+}
+
+int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst) {
+ int i;
+ int32_t sum_q24 = 0;
+ assert(inst);
+ for (i = 0; i <= MAX_IAT; ++i) {
+ /* Shift 6 to fit worst case: 2^30 * 64. */
+ sum_q24 += (inst->iatProb[i] >> 6) * i;
+ }
+ /* Subtract the nominal inter-arrival time 1 = 2^24 in Q24. */
+ sum_q24 -= (1 << 24);
+ /*
+ * Multiply with 1000000 / 2^24 = 15625 / 2^18 to get in parts-per-million.
+ * Shift 7 to Q17 first, then multiply with 15625 and shift another 11.
+ */
+ return ((sum_q24 >> 7) * 15625) >> 11;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/automode.h b/trunk/src/modules/audio_coding/neteq/automode.h
new file mode 100644
index 0000000..dbd09cf
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/automode.h
@@ -0,0 +1,264 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the functionality for automatic buffer level optimization.
+ */
+
+#ifndef AUTOMODE_H
+#define AUTOMODE_H
+
+#include "typedefs.h"
+
+/*************/
+/* Constants */
+/*************/
+
+/* The beta parameter defines the trade-off between delay and underrun probability. */
+/* It is defined through its inverse in Q30 */
+#define AUTOMODE_BETA_INV_Q30 53687091 /* 1/20 in Q30 */
+#define AUTOMODE_STREAMING_BETA_INV_Q30 536871 /* 1/2000 in Q30 */
+
+/* Forgetting factor for the inter-arrival time statistics */
+#define IAT_PROB_FACT 32745 /* 0.9993 in Q15 */
+
+/* Maximum inter-arrival time to register (in "packet-times") */
+#define MAX_IAT 64
+#define PEAK_HEIGHT 20 /* 0.08s in Q8 */
+
+/* The value (1<<5) sets maximum accelerate "speed" to about 100 ms/s */
+#define AUTOMODE_TIMESCALE_LIMIT (1<<5)
+
+/* Peak mode related parameters */
+/* Number of peaks in peak vector; must be a power of 2 */
+#define NUM_PEAKS 8
+
+/* Must be NUM_PEAKS-1 */
+#define PEAK_INDEX_MASK 0x0007
+
+/* Longest accepted peak distance */
+#define MAX_PEAK_PERIOD 10
+#define MAX_STREAMING_PEAK_PERIOD 600 /* 10 minutes */
+
+/* Number of peaks required before peak mode can be engaged */
+#define NUM_PEAKS_REQUIRED 3
+
+/* Drift term for cumulative sum */
+#define CSUM_IAT_DRIFT 2
+
+/*******************/
+/* Automode struct */
+/*******************/
+
+/* The automode struct is a sub-struct of the
+ bufstats-struct (BufstatsInst_t). */
+
+typedef struct
+{
+
+ /* Filtered current buffer level */
+ WebRtc_UWord16 levelFiltFact; /* filter forgetting factor in Q8 */
+ WebRtc_UWord16 buffLevelFilt; /* filtered buffer level in Q8 */
+
+ /* Inter-arrival time (iat) statistics */
+ WebRtc_Word32 iatProb[MAX_IAT + 1]; /* iat probabilities in Q30 */
+ WebRtc_Word16 iatProbFact; /* iat forgetting factor in Q15 */
+ WebRtc_UWord32 packetIatCountSamp; /* time (in timestamps) elapsed since last
+ packet arrival, based on RecOut calls */
+ WebRtc_UWord16 optBufLevel; /* current optimal buffer level in Q8 */
+
+ /* Packet related information */
+ WebRtc_Word16 packetSpeechLenSamp; /* speech samples per incoming packet */
+ WebRtc_Word16 lastPackCNGorDTMF; /* indicates that the last received packet
+ contained special information */
+ WebRtc_UWord16 lastSeqNo; /* sequence number for last packet received */
+ WebRtc_UWord32 lastTimeStamp; /* timestamp for the last packet received */
+ WebRtc_Word32 sampleMemory; /* memory position for keeping track of how many
+ samples we cut during expand */
+ WebRtc_Word16 prevTimeScale; /* indicates that the last mode was an accelerate
+ or pre-emptive expand operation */
+ WebRtc_UWord32 timescaleHoldOff; /* counter that is shifted one step right each
+ RecOut call; time-scaling allowed when it has
+ reached 0 */
+ WebRtc_Word16 extraDelayMs; /* extra delay for sync with video */
+
+ /* Peak-detection */
+ /* vector with the latest peak periods (peak spacing in samples) */
+ WebRtc_UWord32 peakPeriodSamp[NUM_PEAKS];
+ /* vector with the latest peak heights (in packets) */
+ WebRtc_Word16 peakHeightPkt[NUM_PEAKS];
+ WebRtc_Word16 peakIndex; /* index for the vectors peakPeriodSamp and peakHeightPkt;
+ -1 if still waiting for first peak */
+ WebRtc_UWord16 peakThresholdPkt; /* definition of peak (in packets);
+ calculated from PEAK_HEIGHT */
+ WebRtc_UWord32 peakIatCountSamp; /* samples elapsed since last peak was observed */
+ WebRtc_UWord32 curPeakPeriod; /* current maximum of peakPeriodSamp vector */
+ WebRtc_Word16 curPeakHeight; /* derived from peakHeightPkt vector;
+ used as optimal buffer level in peak mode */
+ WebRtc_Word16 peakModeDisabled; /* ==0 if peak mode can be engaged; >0 if not */
+ uint16_t peakFound; /* 1 if peaks are detected and extra delay is applied;
+ * 0 otherwise. */
+
+ /* Post-call statistics */
+ WebRtc_UWord32 countIAT500ms; /* number of times we got small network outage */
+ WebRtc_UWord32 countIAT1000ms; /* number of times we got medium network outage */
+ WebRtc_UWord32 countIAT2000ms; /* number of times we got large network outage */
+ WebRtc_UWord32 longestIATms; /* mSec duration of longest network outage */
+
+ WebRtc_Word16 cSumIatQ8; /* cumulative sum of inter-arrival times */
+ WebRtc_Word16 maxCSumIatQ8; /* max cumulative sum IAT */
+ WebRtc_UWord32 maxCSumUpdateTimer;/* time elapsed since maximum was observed */
+
+} AutomodeInst_t;
+
+/*************/
+/* Functions */
+/*************/
+
+/****************************************************************************
+ * WebRtcNetEQ_UpdateIatStatistics(...)
+ *
+ * Update the packet inter-arrival time statistics when a new packet arrives.
+ * This function should be called for every arriving packet, with some
+ * exceptions when using DTX/VAD and DTMF. A new optimal buffer level is
+ * calculated after the update.
+ *
+ * Input:
+ * - inst : Automode instance
+ * - maxBufLen : Maximum number of packets the buffer can hold
+ * - seqNumber : RTP sequence number of incoming packet
+ * - timeStamp : RTP timestamp of incoming packet
+ * - fsHz : Sample rate in Hz
+ * - mdCodec : Non-zero if the current codec is a multiple-
+ * description codec
+ * - streamingMode : A non-zero value will increase jitter robustness (and delay)
+ *
+ * Output:
+ * - inst : Updated automode instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
+ WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
+ WebRtc_Word32 fsHz, int mdCodec, int streamingMode);
+
+/****************************************************************************
+ * WebRtcNetEQ_CalcOptimalBufLvl(...)
+ *
+ * Calculate the optimal buffer level based on packet inter-arrival time
+ * statistics.
+ *
+ * Input:
+ * - inst : Automode instance
+ * - fsHz : Sample rate in Hz
+ * - mdCodec : Non-zero if the current codec is a multiple-
+ * description codec
+ * - timeIatPkts : Currently observed inter-arrival time in packets
+ * - streamingMode : A non-zero value will increase jitter robustness (and delay)
+ *
+ * Output:
+ * - inst : Updated automode instance
+ *
+ * Return value : >0 - Optimal buffer level
+ * <0 - Error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
+ int mdCodec, WebRtc_UWord32 timeIatPkts,
+ int streamingMode);
+
+/****************************************************************************
+ * WebRtcNetEQ_BufferLevelFilter(...)
+ *
+ * Update filtered buffer level. The function must be called once for each
+ * RecOut call, since the timing of automode hinges on counters that are
+ * updated by this function.
+ *
+ * Input:
+ * - curSizeMs8 : Total length of unused speech data in packet buffer
+ * and sync buffer, in ms * 8
+ * - inst : Automode instance
+ * - sampPerCall : Number of samples per RecOut call
+ * - fsMult : Sample rate in Hz divided by 8000
+ *
+ * Output:
+ * - inst : Updated automode instance
+ *
+ * Return value : 0 - Ok
+ * : <0 - Error
+ */
+
+int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
+ int sampPerCall, WebRtc_Word16 fsMult);
+
+/****************************************************************************
+ * WebRtcNetEQ_SetPacketSpeechLen(...)
+ *
+ * Provide the number of speech samples extracted from a packet to the
+ * automode instance. Several of the calculations within automode depend
+ * on knowing the packet size.
+ *
+ *
+ * Input:
+ * - inst : Automode instance
+ * - newLenSamp : Number of samples per RecOut call
+ * - fsHz : Sample rate in Hz
+ *
+ * Output:
+ * - inst : Updated automode instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
+ WebRtc_Word32 fsHz);
+
+/****************************************************************************
+ * WebRtcNetEQ_ResetAutomode(...)
+ *
+ * Reset the automode instance.
+ *
+ *
+ * Input:
+ * - inst : Automode instance
+ * - maxBufLenPackets : Maximum number of packets that the packet
+ * buffer can hold (>1)
+ *
+ * Output:
+ * - inst : Updated automode instance
+ *
+ * Return value : 0 - Ok
+ */
+
+int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets);
+
+/****************************************************************************
+ * WebRtcNetEQ_AverageIAT(...)
+ *
+ * Calculate the average inter-arrival time based on current statistics.
+ * The average is expressed in parts per million relative the nominal. That is,
+ * if the average inter-arrival time is equal to the nominal frame time,
+ * the return value is zero. A positive value corresponds to packet spacing
+ * being too large, while a negative value means that the packets arrive with
+ * less spacing than expected.
+ *
+ *
+ * Input:
+ * - inst : Automode instance.
+ *
+ * Return value : Average relative inter-arrival time in samples.
+ */
+
+int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst);
+
+#endif /* AUTOMODE_H */
diff --git a/trunk/src/modules/audio_coding/neteq/bgn_update.c b/trunk/src/modules/audio_coding/neteq/bgn_update.c
new file mode 100644
index 0000000..05956c2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/bgn_update.c
@@ -0,0 +1,247 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the function for updating the background noise estimate.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+
+/* Scratch usage:
+ Designed for BGN_LPC_ORDER <= 10
+
+ Type Name size startpos endpos
+ WebRtc_Word32 pw32_autoCorr 22 0 21 (Length (BGN_LPC_ORDER + 1)*2)
+ WebRtc_Word16 pw16_tempVec 10 22 31 (Length BGN_LPC_ORDER)
+ WebRtc_Word16 pw16_rc 10 32 41 (Length BGN_LPC_ORDER)
+ WebRtc_Word16 pw16_outVec 74 0 73 (Length BGN_LPC_ORDER + 64)
+
+ Total: 74
+ */
+
+#if (BGN_LPC_ORDER > 10) && (defined SCRATCH)
+#error BGN_LPC_ORDER is too large for current scratch memory allocation
+#endif
+
+#define SCRATCH_PW32_AUTO_CORR 0
+#define SCRATCH_PW16_TEMP_VEC 22
+#define SCRATCH_PW16_RC 32
+#define SCRATCH_PW16_OUT_VEC 0
+
+#define NETEQFIX_BGNFRAQINCQ16 229 /* 0.0035 in Q16 */
+
+/****************************************************************************
+ * WebRtcNetEQ_BGNUpdate(...)
+ *
+ * This function updates the background noise parameter estimates.
+ *
+ * Input:
+ * - inst : NetEQ instance, where the speech history is stored.
+ * - scratchPtr : Pointer to scratch vector.
+ *
+ * Output:
+ * - inst : Updated information about the BGN characteristics.
+ *
+ * Return value : No return value
+ */
+
+void WebRtcNetEQ_BGNUpdate(
+#ifdef SCRATCH
+ DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
+#else
+ DSPInst_t *inst
+#endif
+)
+{
+ const WebRtc_Word16 w16_vecLen = 256;
+ BGNInst_t *BGN_Inst = &(inst->BGNInst);
+#ifdef SCRATCH
+ WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_AUTO_CORR);
+ WebRtc_Word16 *pw16_tempVec = pw16_scratchPtr + SCRATCH_PW16_TEMP_VEC;
+ WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
+ WebRtc_Word16 *pw16_outVec = pw16_scratchPtr + SCRATCH_PW16_OUT_VEC;
+#else
+ WebRtc_Word32 pw32_autoCorr[BGN_LPC_ORDER + 1];
+ WebRtc_Word16 pw16_tempVec[BGN_LPC_ORDER];
+ WebRtc_Word16 pw16_outVec[BGN_LPC_ORDER + 64];
+ WebRtc_Word16 pw16_rc[BGN_LPC_ORDER];
+#endif
+ WebRtc_Word16 pw16_A[BGN_LPC_ORDER + 1];
+ WebRtc_Word32 w32_tmp;
+ WebRtc_Word16 *pw16_vec;
+ WebRtc_Word16 w16_maxSample;
+ WebRtc_Word16 w16_tmp, w16_tmp2;
+ WebRtc_Word16 w16_enSampleShift;
+ WebRtc_Word32 w32_en, w32_enBGN;
+ WebRtc_Word32 w32_enUpdateThreashold;
+ WebRtc_Word16 stability;
+
+ pw16_vec = inst->pw16_speechHistory + inst->w16_speechHistoryLen - w16_vecLen;
+
+#ifdef NETEQ_VAD
+ if( !inst->VADInst.VADEnabled /* we are not using post-decode VAD */
+ || inst->VADInst.VADDecision == 0 )
+ { /* ... or, post-decode VAD says passive speaker */
+#endif /* NETEQ_VAD */
+
+ /*Insert zeros to guarantee that boundary values do not distort autocorrelation */
+ WEBRTC_SPL_MEMCPY_W16(pw16_tempVec, pw16_vec - BGN_LPC_ORDER, BGN_LPC_ORDER);
+ WebRtcSpl_MemSetW16(pw16_vec - BGN_LPC_ORDER, 0, BGN_LPC_ORDER);
+
+ w16_maxSample = WebRtcSpl_MaxAbsValueW16(pw16_vec, w16_vecLen);
+ w16_tmp = 8 /* log2(w16_veclen) = 8 */
+ - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_maxSample, w16_maxSample));
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ WebRtcNetEQ_CrossCorr(pw32_autoCorr, pw16_vec, pw16_vec, w16_vecLen, BGN_LPC_ORDER + 1,
+ w16_tmp, -1);
+
+ /* Copy back data */
+ WEBRTC_SPL_MEMCPY_W16(pw16_vec - BGN_LPC_ORDER, pw16_tempVec, BGN_LPC_ORDER);
+
+ w16_enSampleShift = 8 - w16_tmp; /* Number of shifts to get energy/sample */
+ /* pw32_autoCorr[0]>>w16_enSampleShift */
+ w32_en = WEBRTC_SPL_RSHIFT_W32(pw32_autoCorr[0], w16_enSampleShift);
+ if ((w32_en < BGN_Inst->w32_energyUpdate
+#ifdef NETEQ_VAD
+ /* post-decode VAD disabled and w32_en sufficiently low */
+ && !inst->VADInst.VADEnabled)
+ /* ... or, post-decode VAD says passive speaker */
+ || (inst->VADInst.VADEnabled && inst->VADInst.VADDecision == 0)
+#else
+ ) /* just close the extra parenthesis */
+#endif /* NETEQ_VAD */
+ )
+ {
+ /* Generate LPC coefficients */
+ if (pw32_autoCorr[0] > 0)
+ {
+ /* regardless of whether the filter is actually updated or not,
+ update energy threshold levels, since we have in fact observed
+ a low energy signal */
+ if (w32_en < BGN_Inst->w32_energyUpdate)
+ {
+ /* Never get under 1.0 in average sample energy */
+ BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1);
+ BGN_Inst->w32_energyUpdateLow = 0;
+ }
+
+ stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, pw16_A, pw16_rc, BGN_LPC_ORDER);
+ /* Only update BGN if filter is stable */
+ if (stability != 1)
+ {
+ return;
+ }
+ }
+ else
+ {
+ /* Do not update */
+ return;
+ }
+ /* Generate the CNG gain factor by looking at the energy of the residual */
+ WebRtcSpl_FilterMAFastQ12(pw16_vec + w16_vecLen - 64, pw16_outVec, pw16_A,
+ BGN_LPC_ORDER + 1, 64);
+ w32_enBGN = WebRtcNetEQ_DotW16W16(pw16_outVec, pw16_outVec, 64, 0);
+ /* Dot product should never overflow since it is BGN and residual! */
+
+ /*
+ * Check spectral flatness
+ * Comparing the residual variance with the input signal variance tells
+ * if the spectrum is flat or not.
+ * (20*w32_enBGN) >= (w32_en<<6)
+ * Also ensure that the energy is non-zero.
+ */
+ if ((WEBRTC_SPL_MUL_32_16(w32_enBGN, 20) >= WEBRTC_SPL_LSHIFT_W32(w32_en, 6))
+ && (w32_en > 0))
+ {
+ /* spectrum is flat enough; save filter parameters */
+
+ WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filter, pw16_A, BGN_LPC_ORDER+1);
+ WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filterState,
+ pw16_vec + w16_vecLen - BGN_LPC_ORDER, BGN_LPC_ORDER);
+
+ /* Save energy level */
+ BGN_Inst->w32_energy = WEBRTC_SPL_MAX(w32_en, 1);
+
+ /* Update energy threshold levels */
+ /* Never get under 1.0 in average sample energy */
+ BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1);
+ BGN_Inst->w32_energyUpdateLow = 0;
+
+ /* Normalize w32_enBGN to 29 or 30 bits before sqrt */
+ w16_tmp2 = WebRtcSpl_NormW32(w32_enBGN) - 1;
+ if (w16_tmp2 & 0x1)
+ {
+ w16_tmp2 -= 1; /* Even number of shifts required */
+ }
+ w32_enBGN = WEBRTC_SPL_SHIFT_W32(w32_enBGN, w16_tmp2);
+
+ /* Calculate scale and shift factor */
+ BGN_Inst->w16_scale = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_enBGN);
+ BGN_Inst->w16_scaleShift = 13 + ((6 + w16_tmp2) >> 1); /* RANDN table is in Q13, */
+ /* 6=log2(64) */
+
+ BGN_Inst->w16_initialized = 1;
+ }
+
+ }
+ else
+ {
+ /*
+ * Will only happen if post-decode VAD is disabled and w32_en is not low enough.
+ * Increase the threshold for update so that it increases by a factor 4 in four
+ * seconds.
+ * energy = energy * 1.0035
+ */
+ w32_tmp = WEBRTC_SPL_MUL_16_16_RSFT(NETEQFIX_BGNFRAQINCQ16,
+ BGN_Inst->w32_energyUpdateLow, 16);
+ w32_tmp += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
+ (WebRtc_Word16)(BGN_Inst->w32_energyUpdate & 0xFF));
+ w32_tmp += (WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
+ (WebRtc_Word16)((BGN_Inst->w32_energyUpdate>>8) & 0xFF)) << 8);
+ BGN_Inst->w32_energyUpdateLow += w32_tmp;
+
+ BGN_Inst->w32_energyUpdate += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
+ (WebRtc_Word16)(BGN_Inst->w32_energyUpdate>>16));
+ BGN_Inst->w32_energyUpdate += BGN_Inst->w32_energyUpdateLow >> 16;
+ BGN_Inst->w32_energyUpdateLow = (BGN_Inst->w32_energyUpdateLow & 0x0FFFF);
+
+ /* Update maximum energy */
+ /* Decrease by a factor 1/1024 each time */
+ BGN_Inst->w32_energyMax = BGN_Inst->w32_energyMax - (BGN_Inst->w32_energyMax >> 10);
+ if (w32_en > BGN_Inst->w32_energyMax)
+ {
+ BGN_Inst->w32_energyMax = w32_en;
+ }
+
+ /* Set update level to at the minimum 60.21dB lower then the maximum energy */
+ w32_enUpdateThreashold = (BGN_Inst->w32_energyMax + 524288) >> 20;
+ if (w32_enUpdateThreashold > BGN_Inst->w32_energyUpdate)
+ {
+ BGN_Inst->w32_energyUpdate = w32_enUpdateThreashold;
+ }
+ }
+
+#ifdef NETEQ_VAD
+} /* closing initial if-statement */
+#endif /* NETEQ_VAD */
+
+ return;
+}
+
+#undef SCRATCH_PW32_AUTO_CORR
+#undef SCRATCH_PW16_TEMP_VEC
+#undef SCRATCH_PW16_RC
+#undef SCRATCH_PW16_OUT_VEC
+
diff --git a/trunk/src/modules/audio_coding/neteq/buffer_stats.h b/trunk/src/modules/audio_coding/neteq/buffer_stats.h
new file mode 100644
index 0000000..9820519
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/buffer_stats.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Calculates and stores the packet buffer statistics.
+ */
+
+#ifndef BUFFER_STATS_H
+#define BUFFER_STATS_H
+
+#include "automode.h"
+#include "webrtc_neteq.h" /* to define enum WebRtcNetEQPlayoutMode */
+
+/* NetEQ related decisions */
+#define BUFSTATS_DO_NORMAL 0
+#define BUFSTATS_DO_ACCELERATE 1
+#define BUFSTATS_DO_MERGE 2
+#define BUFSTATS_DO_EXPAND 3
+#define BUFSTAT_REINIT 4
+#define BUFSTATS_DO_RFC3389CNG_PACKET 5
+#define BUFSTATS_DO_RFC3389CNG_NOPACKET 6
+#define BUFSTATS_DO_INTERNAL_CNG_NOPACKET 7
+#define BUFSTATS_DO_PREEMPTIVE_EXPAND 8
+#define BUFSTAT_REINIT_DECODER 9
+#define BUFSTATS_DO_DTMF_ONLY 10
+/* Decisions related to when NetEQ is switched off (or in FAX mode) */
+#define BUFSTATS_DO_ALTERNATIVE_PLC 11
+#define BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS 12
+#define BUFSTATS_DO_AUDIO_REPETITION 13
+#define BUFSTATS_DO_AUDIO_REPETITION_INC_TS 14
+
+/* Reinit decoder states after this number of expands (upon arrival of new packet) */
+#define REINIT_AFTER_EXPANDS 100
+
+/* Wait no longer than this number of RecOut calls before using an "early" packet */
+#define MAX_WAIT_FOR_PACKET 10
+
+/* CNG modes */
+#define CNG_OFF 0
+#define CNG_RFC3389_ON 1
+#define CNG_INTERNAL_ON 2
+
+typedef struct
+{
+
+ /* store statistical data here */
+ WebRtc_Word16 w16_cngOn; /* remember if CNG is interrupted by other event (e.g. DTMF) */
+ WebRtc_Word16 w16_noExpand;
+ WebRtc_Word32 uw32_CNGplayedTS;
+
+ /* VQmon data */
+ WebRtc_UWord16 avgDelayMsQ8;
+ WebRtc_Word16 maxDelayMs;
+
+ AutomodeInst_t Automode_inst;
+
+} BufstatsInst_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_BufstatsDecision()
+ *
+ * Gives a decision about what action that is currently desired
+ *
+ *
+ * Input:
+ * inst: The bufstat instance
+ * cur_size: Current buffer size in ms in Q3 domain
+ * targetTS: The desired timestamp to start playout from
+ * availableTS: The closest future value available in buffer
+ * noPacket 1 if no packet is available, makes availableTS undefined
+ * prevPlayMode mode of last NetEq playout
+ * timestampsPerCall number of timestamp for 10ms
+ *
+ * Output:
+ * Returns: A decision, as defined above (see top of file)
+ *
+ */
+
+WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
+ WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
+ WebRtc_UWord32 availableTS, int noPacket,
+ int cngPacket, int prevPlayMode,
+ enum WebRtcNetEQPlayoutMode playoutMode,
+ int timestampsPerCall, int NoOfExpandCalls,
+ WebRtc_Word16 fs_mult,
+ WebRtc_Word16 lastModeBGNonly, int playDtmf);
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/bufstats_decision.c b/trunk/src/modules/audio_coding/neteq/bufstats_decision.c
new file mode 100644
index 0000000..3d37e17
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/bufstats_decision.c
@@ -0,0 +1,426 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the function where the main decision logic for buffer level
+ * adaptation happens.
+ */
+
+#include "buffer_stats.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+
+#include "automode.h"
+#include "neteq_defines.h"
+#include "neteq_error_codes.h"
+#include "webrtc_neteq.h"
+
+#define NETEQ_BUFSTAT_20MS_Q7 2560 /* = 20 ms in Q7 */
+
+WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
+ WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
+ WebRtc_UWord32 availableTS, int noPacket,
+ int cngPacket, int prevPlayMode,
+ enum WebRtcNetEQPlayoutMode playoutMode,
+ int timestampsPerCall, int NoOfExpandCalls,
+ WebRtc_Word16 fs_mult,
+ WebRtc_Word16 lastModeBGNonly, int playDtmf)
+{
+
+ int currentDelayMs;
+ WebRtc_Word32 currSizeSamples = cur_size;
+ WebRtc_Word16 extraDelayPacketsQ8 = 0;
+
+ /* Avoid overflow if the buffer size should be really large (cur_size is limited 256ms) */
+ WebRtc_Word32 curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4);
+ WebRtc_UWord16 level_limit_hi, level_limit_lo;
+
+ inst->Automode_inst.prevTimeScale &= (prevPlayMode == MODE_SUCCESS_ACCELERATE
+ || prevPlayMode == MODE_LOWEN_ACCELERATE || prevPlayMode == MODE_SUCCESS_PREEMPTIVE
+ || prevPlayMode == MODE_LOWEN_PREEMPTIVE);
+
+ if ((prevPlayMode != MODE_RFC3389CNG) && (prevPlayMode != MODE_CODEC_INTERNAL_CNG))
+ {
+ /*
+ * Do not update buffer history if currently playing CNG
+ * since it will bias the filtered buffer level.
+ */
+ WebRtcNetEQ_BufferLevelFilter(cur_size, &(inst->Automode_inst), timestampsPerCall,
+ fs_mult);
+ }
+ else
+ {
+ /* only update time counters */
+ inst->Automode_inst.packetIatCountSamp += timestampsPerCall; /* packet inter-arrival time */
+ inst->Automode_inst.peakIatCountSamp += timestampsPerCall; /* peak inter-arrival time */
+ inst->Automode_inst.timescaleHoldOff >>= 1; /* time-scaling limiter */
+ }
+ cur_size = WEBRTC_SPL_MIN(curr_sizeQ7, WEBRTC_SPL_WORD16_MAX);
+
+ /* Calculate VQmon related variables */
+ /* avgDelay = avgDelay*(511/512) + currentDelay*(1/512) (sample ms delay in Q8) */
+ inst->avgDelayMsQ8 = (WebRtc_Word16) (WEBRTC_SPL_MUL_16_16_RSFT(inst->avgDelayMsQ8,511,9)
+ + (cur_size >> 9));
+
+ /* Update maximum delay if needed */
+ currentDelayMs = (curr_sizeQ7 >> 7);
+ if (currentDelayMs > inst->maxDelayMs)
+ {
+ inst->maxDelayMs = currentDelayMs;
+ }
+
+ /* NetEQ is on with normal or steaming mode */
+ if (playoutMode == kPlayoutOn || playoutMode == kPlayoutStreaming)
+ {
+ /* Guard for errors, so that it should not get stuck in error mode */
+ if (prevPlayMode == MODE_ERROR)
+ {
+ if (noPacket)
+ {
+ return BUFSTATS_DO_EXPAND;
+ }
+ else
+ {
+ return BUFSTAT_REINIT;
+ }
+ }
+
+ if (prevPlayMode != MODE_EXPAND && prevPlayMode != MODE_FADE_TO_BGN)
+ {
+ inst->w16_noExpand = 1;
+ }
+ else
+ {
+ inst->w16_noExpand = 0;
+ }
+
+ if (cngPacket)
+ {
+ /* signed difference between wanted and available TS */
+ WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
+ int32_t optimal_level_samp = (inst->Automode_inst.optBufLevel *
+ inst->Automode_inst.packetSpeechLenSamp) >> 8;
+ int32_t excess_waiting_time_samp = -diffTS - optimal_level_samp;
+
+ if (excess_waiting_time_samp > optimal_level_samp / 2)
+ {
+ /* The waiting time for this packet will be longer than 1.5
+ * times the wanted buffer delay. Advance the clock to cut
+ * waiting time down to the optimal.
+ */
+ inst->uw32_CNGplayedTS += excess_waiting_time_samp;
+ diffTS += excess_waiting_time_samp;
+ }
+
+ if ((diffTS) < 0 && (prevPlayMode == MODE_RFC3389CNG))
+ {
+ /* Not time to play this packet yet. Wait another round before using this
+ * packet. Keep on playing CNG from previous CNG parameters. */
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+
+ /* otherwise, go for the CNG packet now */
+ return BUFSTATS_DO_RFC3389CNG_PACKET;
+ }
+
+ /*Check for expand/cng */
+ if (noPacket)
+ {
+ if (inst->w16_cngOn == CNG_RFC3389_ON)
+ {
+ /* keep on playing CNG */
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ else if (inst->w16_cngOn == CNG_INTERNAL_ON)
+ {
+ /* keep on playing internal CNG */
+ return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
+ }
+ else if (playDtmf == 1)
+ {
+ /* we have not audio data, but can play DTMF */
+ return BUFSTATS_DO_DTMF_ONLY;
+ }
+ else
+ {
+ /* nothing to play => do Expand */
+ return BUFSTATS_DO_EXPAND;
+ }
+ }
+
+ /*
+ * If the expand period was very long, reset NetEQ since it is likely that the
+ * sender was restarted.
+ */
+ if (NoOfExpandCalls > REINIT_AFTER_EXPANDS) return BUFSTAT_REINIT_DECODER;
+
+ /* Calculate extra delay in Q8 packets */
+ if (inst->Automode_inst.extraDelayMs > 0 && inst->Automode_inst.packetSpeechLenSamp
+ > 0)
+ {
+ extraDelayPacketsQ8 = WebRtcSpl_DivW32W16ResW16(
+ (WEBRTC_SPL_MUL(inst->Automode_inst.extraDelayMs, 8 * fs_mult) << 8),
+ inst->Automode_inst.packetSpeechLenSamp);
+ /* (extra delay in samples in Q8) */
+ }
+
+ /* Check if needed packet is available */
+ if (targetTS == availableTS)
+ {
+
+ /* If last mode was not expand, and there is no DTMF to play */
+ if (inst->w16_noExpand == 1 && playDtmf == 0)
+ {
+ /* If so check for accelerate */
+
+ level_limit_lo = ((inst->Automode_inst.optBufLevel) >> 1) /* 50 % */
+ + ((inst->Automode_inst.optBufLevel) >> 2); /* ... + 25% = 75% */
+
+ /* set upper limit to optBufLevel, but make sure that window is at least 20ms */
+ level_limit_hi = WEBRTC_SPL_MAX(inst->Automode_inst.optBufLevel,
+ level_limit_lo +
+ WebRtcSpl_DivW32W16ResW16((WEBRTC_SPL_MUL(20*8, fs_mult) << 8),
+ inst->Automode_inst.packetSpeechLenSamp));
+
+ /* if extra delay is non-zero, add it */
+ if (extraDelayPacketsQ8 > 0)
+ {
+ level_limit_hi += extraDelayPacketsQ8;
+ level_limit_lo += extraDelayPacketsQ8;
+ }
+
+ if (((inst->Automode_inst.buffLevelFilt >= level_limit_hi) &&
+ (inst->Automode_inst.timescaleHoldOff == 0)) ||
+ (inst->Automode_inst.buffLevelFilt >= level_limit_hi << 2))
+ {
+ /*
+ * Buffer level higher than limit and time-scaling allowed,
+ * OR buffer level _really_ high.
+ */
+ return BUFSTATS_DO_ACCELERATE;
+ }
+ else if ((inst->Automode_inst.buffLevelFilt < level_limit_lo)
+ && (inst->Automode_inst.timescaleHoldOff == 0))
+ {
+ return BUFSTATS_DO_PREEMPTIVE_EXPAND;
+ }
+ }
+ return BUFSTATS_DO_NORMAL;
+ }
+
+ /* Check for Merge */
+ else if (availableTS > targetTS)
+ {
+
+ /* Check that we do not play a packet "too early" */
+ if ((prevPlayMode == MODE_EXPAND)
+ && (availableTS - targetTS
+ < (WebRtc_UWord32) WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
+ (WebRtc_Word16)REINIT_AFTER_EXPANDS))
+ && (NoOfExpandCalls < MAX_WAIT_FOR_PACKET)
+ && (availableTS
+ > targetTS
+ + WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
+ (WebRtc_Word16)NoOfExpandCalls))
+ && (inst->Automode_inst.buffLevelFilt <= inst->Automode_inst.optBufLevel
+ + extraDelayPacketsQ8))
+ {
+ if (playDtmf == 1)
+ {
+ /* we still have DTMF to play, so do not perform expand */
+ return BUFSTATS_DO_DTMF_ONLY;
+ }
+ else
+ {
+ /* nothing to play */
+ return BUFSTATS_DO_EXPAND;
+ }
+ }
+
+ /* If previous was CNG period or BGNonly then no merge is needed */
+ if ((prevPlayMode == MODE_RFC3389CNG) || (prevPlayMode == MODE_CODEC_INTERNAL_CNG)
+ || lastModeBGNonly)
+ {
+ /*
+ * Keep the same delay as before the CNG (or maximum 70 ms in buffer as safety
+ * precaution), but make sure that the number of samples in buffer is no
+ * higher than 4 times the optimal level.
+ */
+ WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
+ if (diffTS >= 0
+ || (WEBRTC_SPL_MUL_16_16_RSFT( inst->Automode_inst.optBufLevel
+ + extraDelayPacketsQ8,
+ inst->Automode_inst.packetSpeechLenSamp, 6) < currSizeSamples))
+ {
+ /* it is time to play this new packet */
+ return BUFSTATS_DO_NORMAL;
+ }
+ else
+ {
+ /* it is too early to play this new packet => keep on playing CNG */
+ if (prevPlayMode == MODE_RFC3389CNG)
+ {
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ else if (prevPlayMode == MODE_CODEC_INTERNAL_CNG)
+ {
+ return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
+ }
+ else if (playDtmf == 1)
+ {
+ /* we have not audio data, but can play DTMF */
+ return BUFSTATS_DO_DTMF_ONLY;
+ }
+ else /* lastModeBGNonly */
+ {
+ /* signal expand, but this will result in BGN again */
+ return BUFSTATS_DO_EXPAND;
+ }
+ }
+ }
+
+ /* Do not merge unless we have done a Expand before (for complexity reasons) */
+ if ((inst->w16_noExpand == 0) || ((frameSize < timestampsPerCall) && (cur_size
+ > NETEQ_BUFSTAT_20MS_Q7)))
+ {
+ return BUFSTATS_DO_MERGE;
+ }
+ else if (playDtmf == 1)
+ {
+ /* play DTMF instead of expand */
+ return BUFSTATS_DO_DTMF_ONLY;
+ }
+ else
+ {
+ return BUFSTATS_DO_EXPAND;
+ }
+ }
+ }
+ else
+ { /* kPlayoutOff or kPlayoutFax */
+ if (cngPacket)
+ {
+ if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
+ {
+ /* time to play this packet now */
+ return BUFSTATS_DO_RFC3389CNG_PACKET;
+ }
+ else
+ {
+ /* wait before playing this packet */
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ }
+ if (noPacket)
+ {
+ /*
+ * No packet =>
+ * 1. If in CNG mode play as usual
+ * 2. Otherwise use other method to generate data and hold TS value
+ */
+ if (inst->w16_cngOn == CNG_RFC3389_ON)
+ {
+ /* keep on playing CNG */
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ else if (inst->w16_cngOn == CNG_INTERNAL_ON)
+ {
+ /* keep on playing internal CNG */
+ return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
+ }
+ else
+ {
+ /* nothing to play => invent some data to play out */
+ if (playoutMode == kPlayoutOff)
+ {
+ return BUFSTATS_DO_ALTERNATIVE_PLC;
+ }
+ else if (playoutMode == kPlayoutFax)
+ {
+ return BUFSTATS_DO_AUDIO_REPETITION;
+ }
+ else
+ {
+ /* UNDEFINED, should not get here... */
+ assert(0);
+ return BUFSTAT_REINIT;
+ }
+ }
+ }
+ else if (targetTS == availableTS)
+ {
+ return BUFSTATS_DO_NORMAL;
+ }
+ else
+ {
+ if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
+ {
+ return BUFSTATS_DO_NORMAL;
+ }
+ else if (playoutMode == kPlayoutOff)
+ {
+ /*
+ * If currently playing CNG, continue with that. Don't increase TS
+ * since uw32_CNGplayedTS will be increased.
+ */
+ if (inst->w16_cngOn == CNG_RFC3389_ON)
+ {
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ else if (inst->w16_cngOn == CNG_INTERNAL_ON)
+ {
+ return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
+ }
+ else
+ {
+ /*
+ * Otherwise, do PLC and increase TS while waiting for the time to
+ * play this packet.
+ */
+ return BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS;
+ }
+ }
+ else if (playoutMode == kPlayoutFax)
+ {
+ /*
+ * If currently playing CNG, continue with that don't increase TS since
+ * uw32_CNGplayedTS will be increased.
+ */
+ if (inst->w16_cngOn == CNG_RFC3389_ON)
+ {
+ return BUFSTATS_DO_RFC3389CNG_NOPACKET;
+ }
+ else if (inst->w16_cngOn == CNG_INTERNAL_ON)
+ {
+ return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
+ }
+ else
+ {
+ /*
+ * Otherwise, do audio repetition and increase TS while waiting for the
+ * time to play this packet.
+ */
+ return BUFSTATS_DO_AUDIO_REPETITION_INC_TS;
+ }
+ }
+ else
+ {
+ /* UNDEFINED, should not get here... */
+ assert(0);
+ return BUFSTAT_REINIT;
+ }
+ }
+ }
+ /* We should not get here (but sometimes we do anyway...) */
+ return BUFSTAT_REINIT;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/cng_internal.c b/trunk/src/modules/audio_coding/neteq/cng_internal.c
new file mode 100644
index 0000000..f3a10dc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/cng_internal.c
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the function for obtaining comfort noise from noise parameters
+ * according to IETF RFC 3389.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+#include "webrtc_cng.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+
+/****************************************************************************
+ * WebRtcNetEQ_Cng(...)
+ *
+ * This function produces CNG according to RFC 3389.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - len : Number of samples to produce (max 640 or
+ * 640 - fsHz*5/8000 for first-time CNG, governed by
+ * the definition of WEBRTC_CNG_MAX_OUTSIZE_ORDER in
+ * webrtc_cng.h)
+ *
+ * Output:
+ * - pw16_outData : Output CNG
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+#ifdef NETEQ_CNG_CODEC
+/* Must compile NetEQ with CNG support to enable this function */
+
+int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len)
+{
+ WebRtc_Word16 w16_winMute = 0; /* mixing factor for overlap data */
+ WebRtc_Word16 w16_winUnMute = 0; /* mixing factor for comfort noise */
+ WebRtc_Word16 w16_winMuteInc = 0; /* mixing factor increment (negative) */
+ WebRtc_Word16 w16_winUnMuteInc = 0; /* mixing factor increment */
+ int i;
+
+ /*
+ * Check if last RecOut call was other than RFC3389,
+ * that is, this call is the first of a CNG period.
+ */
+ if (inst->w16_mode != MODE_RFC3389CNG)
+ {
+ /* Reset generation and overlap slightly with old data */
+
+ /* Generate len samples + overlap */
+ if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData,
+ (WebRtc_Word16) (len + inst->ExpandInst.w16_overlap), 1) < 0)
+ {
+ /* error returned */
+ return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
+ }
+
+ /* Set windowing parameters depending on sample rate */
+ if (inst->fs == 8000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC;
+#ifdef NETEQ_WIDEBAND
+ }
+ else if (inst->fs == 16000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ }
+ else if (inst->fs == 32000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ }
+ else if (inst->fs == 48000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC;
+#endif
+ }
+ else
+ {
+ /* Unsupported sample rate (should not be possible) */
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /* Do overlap add between new vector and overlap */
+ for (i = 0; i < inst->ExpandInst.w16_overlap; i++)
+ {
+ /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */
+ inst->ExpandInst.pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL_16_16(
+ inst->ExpandInst.pw16_overlapVec[i], w16_winMute) +
+ WEBRTC_SPL_MUL_16_16(pw16_outData[i], w16_winUnMute)
+ + 16384, 15); /* shift with proper rounding */
+
+ w16_winMute += w16_winMuteInc; /* decrease mute factor (inc<0) */
+ w16_winUnMute += w16_winUnMuteInc; /* increase unmute factor (inc>0) */
+
+ }
+
+ /*
+ * Shift the contents of the outData buffer by overlap samples, since we
+ * already used these first samples in the overlapVec above
+ */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_outData+inst->ExpandInst.w16_overlap, len);
+
+ }
+ else
+ {
+ /* This is a subsequent CNG call; no special overlap needed */
+
+ /* Generate len samples */
+ if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, (WebRtc_Word16) len, 0) < 0)
+ {
+ /* error returned */
+ return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
+ }
+ }
+
+ return 0;
+
+}
+
+#endif /* NETEQ_CNG_CODEC */
+
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db.c b/trunk/src/modules/audio_coding/neteq/codec_db.c
new file mode 100644
index 0000000..cb63aea
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/codec_db.c
@@ -0,0 +1,743 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of the codec database.
+ */
+
+#include "codec_db.h"
+
+#include <string.h> /* to define NULL */
+
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+/*
+ * Resets the codec database.
+ */
+
+int WebRtcNetEQ_DbReset(CodecDbInst_t *inst)
+{
+ int i;
+
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) inst, 0,
+ sizeof(CodecDbInst_t) / sizeof(WebRtc_Word16));
+
+ for (i = 0; i < NUM_TOTAL_CODECS; i++)
+ {
+ inst->position[i] = -1;
+ }
+
+ for (i = 0; i < NUM_CODECS; i++)
+ {
+ inst->payloadType[i] = -1;
+ }
+
+ for (i = 0; i < NUM_CNG_CODECS; i++)
+ {
+ inst->CNGpayloadType[i] = -1;
+ }
+
+ return 0;
+}
+
+/*
+ * Adds a new codec to the database.
+ */
+
+int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
+ WebRtc_Word16 payloadType, FuncDecode funcDecode,
+ FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
+ FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
+ FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
+ FuncUpdBWEst funcUpdBWEst, FuncGetErrorCode funcGetErrorCode,
+ void* codec_state, WebRtc_UWord16 codec_fs)
+{
+
+ int temp;
+ int insertCNGcodec = 0, overwriteCNGcodec = 0, CNGpos = -1;
+
+#ifndef NETEQ_RED_CODEC
+ if (codec == kDecoderRED)
+ {
+ return CODEC_DB_UNSUPPORTED_CODEC;
+ }
+#endif
+ if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec
+ >= (int) kDecoderReservedEnd))
+ {
+ return CODEC_DB_UNSUPPORTED_CODEC;
+ }
+
+ if ((codec_fs != 8000)
+#ifdef NETEQ_WIDEBAND
+ &&(codec_fs!=16000)
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ &&(codec_fs!=32000)
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ &&(codec_fs!=48000)
+#endif
+ )
+ {
+ return CODEC_DB_UNSUPPORTED_FS;
+ }
+
+ /* Ensure that the codec type is supported */
+ switch (codec)
+ {
+#ifdef NETEQ_PCM16B_CODEC
+ case kDecoderPCM16B :
+#endif
+#ifdef NETEQ_G711_CODEC
+ case kDecoderPCMu :
+ case kDecoderPCMa :
+#endif
+#ifdef NETEQ_ILBC_CODEC
+ case kDecoderILBC :
+#endif
+#ifdef NETEQ_ISAC_CODEC
+ case kDecoderISAC :
+#endif
+#ifdef NETEQ_ISAC_SWB_CODEC
+ case kDecoderISACswb :
+#endif
+#ifdef NETEQ_G722_CODEC
+ case kDecoderG722 :
+#endif
+#ifdef NETEQ_WIDEBAND
+ case kDecoderPCM16Bwb :
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ case kDecoderPCM16Bswb32kHz :
+#endif
+#ifdef NETEQ_CNG_CODEC
+ case kDecoderCNG :
+#endif
+#ifdef NETEQ_ATEVENT_DECODE
+ case kDecoderAVT :
+#endif
+#ifdef NETEQ_RED_CODEC
+ case kDecoderRED :
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ case kDecoderPCM16Bswb48kHz :
+#endif
+#ifdef NETEQ_ARBITRARY_CODEC
+ case kDecoderArbitrary:
+#endif
+#ifdef NETEQ_G729_CODEC
+ case kDecoderG729:
+#endif
+#ifdef NETEQ_G729_1_CODEC
+ case kDecoderG729_1 :
+#endif
+#ifdef NETEQ_G726_CODEC
+ case kDecoderG726_16 :
+ case kDecoderG726_24 :
+ case kDecoderG726_32 :
+ case kDecoderG726_40 :
+#endif
+#ifdef NETEQ_G722_1_CODEC
+ case kDecoderG722_1_16 :
+ case kDecoderG722_1_24 :
+ case kDecoderG722_1_32 :
+#endif
+#ifdef NETEQ_G722_1C_CODEC
+ case kDecoderG722_1C_24 :
+ case kDecoderG722_1C_32 :
+ case kDecoderG722_1C_48 :
+#endif
+#ifdef NETEQ_SPEEX_CODEC
+ case kDecoderSPEEX_8 :
+ case kDecoderSPEEX_16 :
+#endif
+#ifdef NETEQ_CELT_CODEC
+ case kDecoderCELT_32 :
+#endif
+#ifdef NETEQ_GSMFR_CODEC
+ case kDecoderGSMFR :
+#endif
+#ifdef NETEQ_AMR_CODEC
+ case kDecoderAMR :
+#endif
+#ifdef NETEQ_AMRWB_CODEC
+ case kDecoderAMRWB :
+#endif
+ {
+ /* If we end up here, the inserted codec is supported => Do nothing */
+ break;
+ }
+ default:
+ {
+ /* If we get to this point, the inserted codec is not supported */
+ return CODEC_DB_UNSUPPORTED_CODEC;
+ }
+ }
+
+ /* Check to see if payload type is taken */
+ if (WebRtcNetEQ_DbGetCodec(inst, payloadType) > 0)
+ {
+ return CODEC_DB_PAYLOAD_TAKEN;
+ }
+
+ /* Special case for CNG codecs */
+ if (codec == kDecoderCNG)
+ {
+ /* check if this is first CNG codec to be registered */
+ if (WebRtcNetEQ_DbGetPayload(inst, codec) == CODEC_DB_NOT_EXIST2)
+ {
+ /* no other CNG codec found */
+ insertCNGcodec = 1;
+ }
+
+ /* find the appropriate insert position in CNG payload vector */
+ switch (codec_fs)
+ {
+#ifdef NETEQ_WIDEBAND
+ case 16000:
+ CNGpos = 1;
+ break;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ case 32000:
+ CNGpos = 2;
+ break;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ case 48000:
+ CNGpos = 3;
+ break;
+#endif
+ default: /* 8000 Hz case */
+ CNGpos = 0;
+ /*
+ * The 8 kHz CNG payload type is the one associated with the regular codec DB
+ * should override any other setting.
+ * Overwrite if this isn't the first CNG
+ */
+ overwriteCNGcodec = !insertCNGcodec;
+ break;
+ }
+
+ /* insert CNG payload type */
+ inst->CNGpayloadType[CNGpos] = payloadType;
+
+ }
+
+ if ((codec != kDecoderCNG) || (insertCNGcodec == 1) || (overwriteCNGcodec == 1))
+ {
+ /* Check if we have reached the maximum numbers of simultaneous codecs */
+ if (inst->nrOfCodecs == NUM_CODECS) return CODEC_DB_FULL;
+
+ /* Check that codec has not already been initialized to DB =>
+ remove it and reinitialize according to new spec */
+ if ((inst->position[codec] != -1) && (overwriteCNGcodec != 1))
+ { /* if registering multiple CNG codecs, don't remove, just overwrite */
+ WebRtcNetEQ_DbRemove(inst, codec);
+ }
+
+ if (overwriteCNGcodec == 1)
+ {
+ temp = inst->position[codec];
+ }
+ else
+ {
+ temp = inst->nrOfCodecs; /* Store this codecs position */
+ inst->position[codec] = temp;
+ inst->nrOfCodecs++;
+ }
+
+ inst->payloadType[temp] = payloadType;
+
+ /* Copy to database */
+ inst->codec_state[temp] = codec_state;
+ inst->funcDecode[temp] = funcDecode;
+ inst->funcDecodeRCU[temp] = funcDecodeRCU;
+ inst->funcAddLatePkt[temp] = funcAddLatePkt;
+ inst->funcDecodeInit[temp] = funcDecodeInit;
+ inst->funcDecodePLC[temp] = funcDecodePLC;
+ inst->funcGetMDinfo[temp] = funcGetMDinfo;
+ inst->funcGetPitch[temp] = funcGetPitch;
+ inst->funcUpdBWEst[temp] = funcUpdBWEst;
+ inst->funcGetErrorCode[temp] = funcGetErrorCode;
+ inst->codec_fs[temp] = codec_fs;
+
+ }
+
+ return 0;
+}
+
+/*
+ * Removes a codec from the database.
+ */
+
+int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec)
+{
+ int i;
+ int pos = -1;
+
+#ifndef NETEQ_RED_CODEC
+ if (codec == kDecoderRED)
+ {
+ return CODEC_DB_UNSUPPORTED_CODEC;
+ }
+#endif
+ if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec
+ >= (int) kDecoderReservedEnd))
+ {
+ return CODEC_DB_UNSUPPORTED_CODEC;
+ }
+
+ pos = inst->position[codec];
+ if (pos == -1)
+ {
+ return CODEC_DB_NOT_EXIST4;
+ }
+ else
+ {
+ /* Remove this codec */
+ inst->position[codec] = -1;
+ for (i = pos; i < (inst->nrOfCodecs - 1); i++)
+ {
+ inst->payloadType[i] = inst->payloadType[i + 1];
+ inst->codec_state[i] = inst->codec_state[i + 1];
+ inst->funcDecode[i] = inst->funcDecode[i + 1];
+ inst->funcDecodeRCU[i] = inst->funcDecodeRCU[i + 1];
+ inst->funcAddLatePkt[i] = inst->funcAddLatePkt[i + 1];
+ inst->funcDecodeInit[i] = inst->funcDecodeInit[i + 1];
+ inst->funcDecodePLC[i] = inst->funcDecodePLC[i + 1];
+ inst->funcGetMDinfo[i] = inst->funcGetMDinfo[i + 1];
+ inst->funcGetPitch[i] = inst->funcGetPitch[i + 1];
+ inst->funcUpdBWEst[i] = inst->funcUpdBWEst[i + 1];
+ inst->funcGetErrorCode[i] = inst->funcGetErrorCode[i + 1];
+ inst->codec_fs[i] = inst->codec_fs[i + 1];
+ }
+ inst->payloadType[i] = -1;
+ inst->codec_state[i] = NULL;
+ inst->funcDecode[i] = NULL;
+ inst->funcDecodeRCU[i] = NULL;
+ inst->funcAddLatePkt[i] = NULL;
+ inst->funcDecodeInit[i] = NULL;
+ inst->funcDecodePLC[i] = NULL;
+ inst->funcGetMDinfo[i] = NULL;
+ inst->funcGetPitch[i] = NULL;
+ inst->funcUpdBWEst[i] = NULL;
+ inst->funcGetErrorCode[i] = NULL;
+ inst->codec_fs[i] = 0;
+ /* Move down all the codecs above this one */
+ for (i = 0; i < NUM_TOTAL_CODECS; i++)
+ {
+ if (inst->position[i] >= pos)
+ {
+ inst->position[i] = inst->position[i] - 1;
+ }
+ }
+ inst->nrOfCodecs--;
+
+ if (codec == kDecoderCNG)
+ {
+ /* also remove all registered CNG payload types */
+ for (i = 0; i < NUM_CNG_CODECS; i++)
+ {
+ inst->CNGpayloadType[i] = -1;
+ }
+ }
+ }
+ return 0;
+}
+
+/*
+ * Get the decoder function pointers for a codec.
+ */
+
+int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
+ CodecFuncInst_t *ptr_inst)
+{
+
+ int pos = inst->position[codec];
+ if ((codec <= kDecoderReservedStart) || (codec >= kDecoderReservedEnd) || (codec
+ > NUM_TOTAL_CODECS))
+ {
+ /* ERROR */
+ pos = -1;
+ }
+ if (pos >= 0)
+ {
+ ptr_inst->codec_state = inst->codec_state[pos];
+ ptr_inst->funcAddLatePkt = inst->funcAddLatePkt[pos];
+ ptr_inst->funcDecode = inst->funcDecode[pos];
+ ptr_inst->funcDecodeRCU = inst->funcDecodeRCU[pos];
+ ptr_inst->funcDecodeInit = inst->funcDecodeInit[pos];
+ ptr_inst->funcDecodePLC = inst->funcDecodePLC[pos];
+ ptr_inst->funcGetMDinfo = inst->funcGetMDinfo[pos];
+ ptr_inst->funcUpdBWEst = inst->funcUpdBWEst[pos];
+ ptr_inst->funcGetErrorCode = inst->funcGetErrorCode[pos];
+ ptr_inst->codec_fs = inst->codec_fs[pos];
+ return 0;
+ }
+ else
+ {
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) ptr_inst, 0,
+ sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
+ return CODEC_DB_NOT_EXIST1;
+ }
+}
+
+/*
+ * Returns payload number given a codec identifier.
+ */
+
+int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID)
+{
+ if (inst->position[codecID] == -1)
+ return CODEC_DB_NOT_EXIST2;
+ else
+ return (inst->payloadType[inst->position[codecID]]);
+
+}
+
+/*
+ * Returns codec identifier given a payload number.
+ * Returns -1 if the payload type does not exist.
+ */
+
+int WebRtcNetEQ_DbGetCodec(CodecDbInst_t *inst, int payloadType)
+{
+ int i, pos;
+
+ for (i = 0; i < NUM_TOTAL_CODECS; i++)
+ {
+ pos = inst->position[i];
+ if (pos != -1)
+ {
+ if (inst->payloadType[pos] == payloadType) return i;
+ }
+ }
+
+ /* did not find payload type */
+ /* check if it's a CNG codec */
+ if (WebRtcNetEQ_DbIsCNGPayload(inst, payloadType))
+ {
+ return kDecoderCNG;
+ }
+
+ /* found no match */
+ return CODEC_DB_NOT_EXIST3;
+}
+
+/*
+ * Extracts the Payload Split information of the codec with the specified payloadType.
+ */
+
+int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID,
+ int codedsize)
+{
+
+ switch (codecID)
+ {
+#ifdef NETEQ_ISAC_CODEC
+ case kDecoderISAC:
+#endif
+#ifdef NETEQ_ISAC_SWB_CODEC
+ case kDecoderISACswb:
+#endif
+#ifdef NETEQ_ARBITRARY_CODEC
+ case kDecoderArbitrary:
+#endif
+#ifdef NETEQ_AMR_CODEC
+ case kDecoderAMR:
+#endif
+#ifdef NETEQ_AMRWB_CODEC
+ case kDecoderAMRWB:
+#endif
+#ifdef NETEQ_G726_CODEC
+ /* Treat G726 as non-splittable to simplify the implementation */
+ case kDecoderG726_16:
+ case kDecoderG726_24:
+ case kDecoderG726_32:
+ case kDecoderG726_40:
+#endif
+#ifdef NETEQ_SPEEX_CODEC
+ case kDecoderSPEEX_8:
+ case kDecoderSPEEX_16:
+#endif
+#ifdef NETEQ_CELT_CODEC
+ case kDecoderCELT_32 :
+#endif
+#ifdef NETEQ_G729_1_CODEC
+ case kDecoderG729_1:
+#endif
+ {
+ /* These codecs' payloads are not splittable */
+ inst->deltaBytes = NO_SPLIT;
+ return 0;
+ }
+
+ /*
+ * Sample based coders are a special case.
+ * In this case, deltaTime signals the number of bytes per timestamp unit times 2
+ * in log2 domain.
+ */
+#if (defined NETEQ_G711_CODEC)
+ case kDecoderPCMu:
+ case kDecoderPCMa:
+ {
+ inst->deltaBytes = -12;
+ inst->deltaTime = 1;
+ return 0;
+ }
+#endif
+#if (defined NETEQ_G722_CODEC)
+ case kDecoderG722:
+ {
+ inst->deltaBytes = -14;
+ inst->deltaTime = 0;
+ return 0;
+ }
+#endif
+#if (defined NETEQ_PCM16B_CODEC)
+ case kDecoderPCM16B:
+ {
+ inst->deltaBytes = -12;
+ inst->deltaTime = 2;
+ return 0;
+ }
+#endif
+#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_WIDEBAND))
+ case kDecoderPCM16Bwb:
+ {
+ inst->deltaBytes = -14;
+ inst->deltaTime = 2;
+ return 0;
+ }
+#endif
+#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_32KHZ_WIDEBAND))
+ case kDecoderPCM16Bswb32kHz:
+ {
+ inst->deltaBytes = -18;
+ inst->deltaTime = 2;
+ return 0;
+ }
+#endif
+#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_48KHZ_WIDEBAND))
+ case kDecoderPCM16Bswb48kHz:
+ {
+ inst->deltaBytes = -22;
+ inst->deltaTime = 2;
+ return 0;
+ }
+#endif
+
+ /* Splittable payloads */
+#ifdef NETEQ_G722_1_CODEC
+ case kDecoderG722_1_16:
+ {
+ inst->deltaBytes = 40;
+ inst->deltaTime = 320;
+ return 0;
+ }
+ case kDecoderG722_1_24:
+ {
+ inst->deltaBytes = 60;
+ inst->deltaTime = 320;
+ return 0;
+ }
+ case kDecoderG722_1_32:
+ {
+ inst->deltaBytes = 80;
+ inst->deltaTime = 320;
+ return 0;
+ }
+#endif
+#ifdef NETEQ_G722_1C_CODEC
+ case kDecoderG722_1C_24:
+ {
+ inst->deltaBytes = 60;
+ inst->deltaTime = 640;
+ return 0;
+ }
+ case kDecoderG722_1C_32:
+ {
+ inst->deltaBytes = 80;
+ inst->deltaTime = 640;
+ return 0;
+ }
+ case kDecoderG722_1C_48:
+ {
+ inst->deltaBytes = 120;
+ inst->deltaTime = 640;
+ return 0;
+ }
+#endif
+#ifdef NETEQ_G729_CODEC
+ case kDecoderG729:
+ {
+ inst->deltaBytes = 10;
+ inst->deltaTime = 80;
+ return 0;
+ }
+#endif
+#ifdef NETEQ_ILBC_CODEC
+ case kDecoderILBC:
+ {
+ /* Check for splitting of iLBC packets.
+ * If payload size is a multiple of 50 bytes it should be split into 30ms frames.
+ * If payload size is a multiple of 38 bytes it should be split into 20ms frames.
+ * Least common multiplier between 38 and 50 is 950, so the payload size must be less than
+ * 950 bytes in order to resolve the frames unambiguously.
+ * Currently max 12 frames in one bundle.
+ */
+ switch (codedsize)
+ {
+ case 50:
+ case 100:
+ case 150:
+ case 200:
+ case 250:
+ case 300:
+ case 350:
+ case 400:
+ case 450:
+ case 500:
+ case 550:
+ case 600:
+ {
+ inst->deltaBytes = 50;
+ inst->deltaTime = 240;
+ break;
+ }
+ case 38:
+ case 76:
+ case 114:
+ case 152:
+ case 190:
+ case 228:
+ case 266:
+ case 304:
+ case 342:
+ case 380:
+ case 418:
+ case 456:
+ {
+ inst->deltaBytes = 38;
+ inst->deltaTime = 160;
+ break;
+ }
+ default:
+ {
+ return AMBIGUOUS_ILBC_FRAME_SIZE; /* Something not supported... */
+ }
+ }
+ return 0;
+ }
+#endif
+#ifdef NETEQ_GSMFR_CODEC
+ case kDecoderGSMFR:
+ {
+ inst->deltaBytes = 33;
+ inst->deltaTime = 160;
+ return 0;
+ }
+#endif
+ default:
+ { /*Unknown codec */
+ inst->deltaBytes = NO_SPLIT;
+ return CODEC_DB_UNKNOWN_CODEC;
+ }
+ } /* end of switch */
+}
+
+/*
+ * Returns 1 if codec is multiple description, 0 otherwise.
+ * NOTE: This function is a stub, since there currently are no MD codecs.
+ */
+int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID)
+{
+ if (0) /* Add test for MD codecs here */
+ return 1;
+ else
+ return 0;
+}
+
+/*
+ * Returns 1 if payload type is registered as a CNG codec, 0 otherwise
+ */
+int WebRtcNetEQ_DbIsCNGPayload(CodecDbInst_t *inst, int payloadType)
+{
+#ifdef NETEQ_CNG_CODEC
+ int i;
+
+ for(i=0; i<NUM_CNG_CODECS; i++)
+ {
+ if( (inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType) )
+ {
+ return 1;
+ }
+ }
+#endif
+
+ return 0;
+
+}
+
+/*
+ * Return the sample rate for the codec with the given payload type, 0 if error
+ */
+WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType)
+{
+ int i;
+ CodecFuncInst_t codecInst;
+
+ /* Sanity */
+ if (inst == NULL)
+ {
+ /* return 0 Hz */
+ return 0;
+ }
+
+ /* Check among CNG payloads */
+ for (i = 0; i < NUM_CNG_CODECS; i++)
+ {
+ if ((inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType))
+ {
+ switch (i)
+ {
+ case 1:
+ return 16000;
+ case 2:
+ return 32000;
+ case 3:
+ return 48000;
+ default:
+ return 8000;
+ }
+ }
+ }
+
+ /* Not a CNG payload, check the other payloads */
+ i = WebRtcNetEQ_DbGetCodec(inst, payloadType);
+ if (i >= 0)
+ {
+ if (WebRtcNetEQ_DbGetPtrs(inst, (enum WebRtcNetEQDecoder) i, &codecInst) != 0)
+ {
+ /* Unexpected error, return 0 Hz */
+ return 0;
+ }
+ return codecInst.codec_fs;
+ }
+
+ /* If we end up here, we got an error, return 0 Hz */
+ return 0;
+
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db.h b/trunk/src/modules/audio_coding/neteq/codec_db.h
new file mode 100644
index 0000000..7f42980
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/codec_db.h
@@ -0,0 +1,126 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Interface for the codec database.
+ */
+
+#ifndef CODEC_DB_H
+#define CODEC_DB_H
+
+#include "typedefs.h"
+
+#include "webrtc_neteq.h"
+#include "codec_db_defines.h"
+#include "neteq_defines.h"
+
+#if defined(NETEQ_48KHZ_WIDEBAND)
+ #define NUM_CNG_CODECS 4
+#elif defined(NETEQ_32KHZ_WIDEBAND)
+ #define NUM_CNG_CODECS 3
+#elif defined(NETEQ_WIDEBAND)
+ #define NUM_CNG_CODECS 2
+#else
+ #define NUM_CNG_CODECS 1
+#endif
+
+typedef struct
+{
+
+ WebRtc_Word16 position[NUM_TOTAL_CODECS];
+ WebRtc_Word16 nrOfCodecs;
+
+ WebRtc_Word16 payloadType[NUM_CODECS];
+ FuncDecode funcDecode[NUM_CODECS];
+ FuncDecode funcDecodeRCU[NUM_CODECS];
+ FuncDecodePLC funcDecodePLC[NUM_CODECS];
+ FuncDecodeInit funcDecodeInit[NUM_CODECS];
+ FuncAddLatePkt funcAddLatePkt[NUM_CODECS];
+ FuncGetMDinfo funcGetMDinfo[NUM_CODECS];
+ FuncGetPitchInfo funcGetPitch[NUM_CODECS];
+ FuncUpdBWEst funcUpdBWEst[NUM_CODECS];
+ FuncGetErrorCode funcGetErrorCode[NUM_CODECS];
+ void * codec_state[NUM_CODECS];
+ WebRtc_UWord16 codec_fs[NUM_CODECS];
+ WebRtc_Word16 CNGpayloadType[NUM_CNG_CODECS];
+
+} CodecDbInst_t;
+
+#define NO_SPLIT -1 /* codec payload cannot be split */
+
+typedef struct
+{
+ WebRtc_Word16 deltaBytes;
+ WebRtc_Word16 deltaTime;
+} SplitInfo_t;
+
+/*
+ * Resets the codec database.
+ */
+int WebRtcNetEQ_DbReset(CodecDbInst_t *inst);
+
+/*
+ * Adds a new codec to the database.
+ */
+int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
+ WebRtc_Word16 payloadType, FuncDecode funcDecode,
+ FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
+ FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
+ FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
+ FuncUpdBWEst funcUpdBWEst, FuncGetErrorCode funcGetErrorCode,
+ void* codec_state, WebRtc_UWord16 codec_fs);
+
+/*
+ * Removes a codec from the database.
+ */
+int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec);
+
+/*
+ * Get the decoder function pointers for a codec.
+ */
+int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder,
+ CodecFuncInst_t *ptr_inst);
+
+/*
+ * Returns payload number given a codec identifier.
+ */
+
+int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID);
+
+/*
+ * Returns codec identifier given a payload number.
+ */
+
+int WebRtcNetEQ_DbGetCodec(CodecDbInst_t *inst, int payloadType);
+
+/*
+ * Extracts the Payload Split information of the codec with the specified payloadType.
+ */
+
+int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID,
+ int codedsize);
+
+/*
+ * Returns 1 if codec is multiple description type, 0 otherwise.
+ */
+int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID);
+
+/*
+ * Returns 1 if payload type is registered as a CNG codec, 0 otherwise.
+ */
+int WebRtcNetEQ_DbIsCNGPayload(CodecDbInst_t *inst, int payloadType);
+
+/*
+ * Return the sample rate for the codec with the given payload type, 0 if error.
+ */
+WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType);
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db_defines.h b/trunk/src/modules/audio_coding/neteq/codec_db_defines.h
new file mode 100644
index 0000000..9b78b86
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/codec_db_defines.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Some definitions related to the codec database.
+ */
+
+#ifndef CODEC_DB_DEFINES_H
+#define CODEC_DB_DEFINES_H
+
+#include "typedefs.h"
+
+#define NUM_CODECS 47 /* probably too large with the limited set of supported codecs*/
+#define NUM_TOTAL_CODECS kDecoderReservedEnd
+
+/*
+ * Pointer to decoder function.
+ */
+typedef WebRtc_Word16 (*FuncDecode)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len,
+ WebRtc_Word16* decoded, WebRtc_Word16* speechType);
+
+/*
+ * Pointer to PLC function.
+ */
+typedef WebRtc_Word16 (*FuncDecodePLC)(void* state, WebRtc_Word16* decodec,
+ WebRtc_Word16 frames);
+
+/*
+ * Pointer to decoder init function.
+ */
+typedef WebRtc_Word16 (*FuncDecodeInit)(void* state);
+
+/*
+ * Pointer to add late packet function.
+ */
+typedef WebRtc_Word16
+ (*FuncAddLatePkt)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len);
+
+/*
+ * Pointer to get MD infofunction.
+ */
+typedef WebRtc_Word16 (*FuncGetMDinfo)(void* state);
+
+/*
+ * Pointer to pitch info function.
+ * Return 0 for unvoiced, -1 if pitch not availiable.
+ */
+typedef WebRtc_Word16 (*FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
+ WebRtc_Word16* length);
+
+/*
+ * Pointer to the update bandwidth estimate function
+ */
+typedef WebRtc_Word16 (*FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts);
+
+/*
+ * Pointer to error code function
+ */
+typedef WebRtc_Word16 (*FuncGetErrorCode)(void* state);
+
+typedef struct CodecFuncInst_t_
+{
+
+ FuncDecode funcDecode;
+ FuncDecode funcDecodeRCU;
+ FuncDecodePLC funcDecodePLC;
+ FuncDecodeInit funcDecodeInit;
+ FuncAddLatePkt funcAddLatePkt;
+ FuncGetMDinfo funcGetMDinfo;
+ FuncUpdBWEst funcUpdBWEst; /* Currently in use for the ISAC family (without LC) only*/
+ FuncGetErrorCode funcGetErrorCode;
+ void * codec_state;
+ WebRtc_UWord16 codec_fs;
+ WebRtc_UWord32 timeStamp;
+
+} CodecFuncInst_t;
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/neteq/correlator.c b/trunk/src/modules/audio_coding/neteq/correlator.c
new file mode 100644
index 0000000..97c41da
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/correlator.c
@@ -0,0 +1,132 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_corrVec 62 0 61
+ WebRtc_Word16 pw16_data_ds 124 0 123
+ WebRtc_Word32 pw32_corr 2*54 124 231
+
+ Total: 232
+ */
+
+#define SCRATCH_pw16_corrVec 0
+#define SCRATCH_pw16_data_ds 0
+#define SCRATCH_pw32_corr 124
+
+#define NETEQ_CORRELATOR_DSVECLEN 124 /* 124 = 60 + 10 + 54 */
+
+WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_data,
+ WebRtc_Word16 w16_dataLen,
+ WebRtc_Word16 *pw16_corrOut,
+ WebRtc_Word16 *pw16_corrScale)
+{
+ WebRtc_Word16 w16_corrLen = 60;
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_data_ds = pw16_scratchPtr + SCRATCH_pw16_corrVec;
+ WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
+ /* WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;*/
+#else
+ WebRtc_Word16 pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN];
+ WebRtc_Word32 pw32_corr[54];
+ /* WebRtc_Word16 pw16_corrVec[4+54+4];*/
+#endif
+ /* WebRtc_Word16 *pw16_corr=&pw16_corrVec[4];*/
+ WebRtc_Word16 w16_maxVal;
+ WebRtc_Word32 w32_maxVal;
+ WebRtc_Word16 w16_normVal;
+ WebRtc_Word16 w16_normVal2;
+ /* WebRtc_Word16 w16_corrUpsLen;*/
+ WebRtc_Word16 *pw16_B = NULL;
+ WebRtc_Word16 w16_Blen = 0;
+ WebRtc_Word16 w16_factor = 0;
+
+ /* Set constants depending on frequency used */
+ if (inst->fs == 8000)
+ {
+ w16_Blen = 3;
+ w16_factor = 2;
+ pw16_B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
+#ifdef NETEQ_WIDEBAND
+ }
+ else if (inst->fs==16000)
+ {
+ w16_Blen = 5;
+ w16_factor = 4;
+ pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ }
+ else if (inst->fs==32000)
+ {
+ w16_Blen = 7;
+ w16_factor = 8;
+ pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ }
+ else /* if inst->fs==48000 */
+ {
+ w16_Blen = 7;
+ w16_factor = 12;
+ pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl;
+#endif
+ }
+
+ /* Downsample data in order to work on a 4 kHz sampled signal */
+ WebRtcSpl_DownsampleFast(
+ pw16_data + w16_dataLen - (NETEQ_CORRELATOR_DSVECLEN * w16_factor),
+ (WebRtc_Word16) (NETEQ_CORRELATOR_DSVECLEN * w16_factor), pw16_data_ds,
+ NETEQ_CORRELATOR_DSVECLEN, pw16_B, w16_Blen, w16_factor, (WebRtc_Word16) 0);
+
+ /* Normalize downsampled vector to using entire 16 bit */
+ w16_maxVal = WebRtcSpl_MaxAbsValueW16(pw16_data_ds, 124);
+ w16_normVal = 16 - WebRtcSpl_NormW32((WebRtc_Word32) w16_maxVal);
+ WebRtcSpl_VectorBitShiftW16(pw16_data_ds, NETEQ_CORRELATOR_DSVECLEN, pw16_data_ds,
+ w16_normVal);
+
+ /* Correlate from lag 10 to lag 60 (20..120 in NB and 40..240 in WB) */
+
+ WebRtcNetEQ_CrossCorr(
+ pw32_corr, &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen],
+ &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen - 10], 60, 54,
+ 6 /*maxValue... shifts*/, -1);
+
+ /*
+ * Move data from w32 to w16 vector.
+ * Normalize downsampled vector to using all 14 bits
+ */
+ w32_maxVal = WebRtcSpl_MaxAbsValueW32(pw32_corr, 54);
+ w16_normVal2 = 18 - WebRtcSpl_NormW32(w32_maxVal);
+ w16_normVal2 = WEBRTC_SPL_MAX(w16_normVal2, 0);
+
+ WebRtcSpl_VectorBitShiftW32ToW16(pw16_corrOut, 54, pw32_corr, w16_normVal2);
+
+ /* Total scale factor (right shifts) of correlation value */
+ *pw16_corrScale = 2 * w16_normVal + 6 + w16_normVal2;
+
+ return (50 + 1);
+}
+
+#undef SCRATCH_pw16_corrVec
+#undef SCRATCH_pw16_data_ds
+#undef SCRATCH_pw32_corr
+
diff --git a/trunk/src/modules/audio_coding/neteq/delay_logging.h b/trunk/src/modules/audio_coding/neteq/delay_logging.h
new file mode 100644
index 0000000..04b1c40
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/delay_logging.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Contains definitions for the delay logging functionality. Only used for debugging and
+ * tracing purposes.
+ */
+
+#ifndef DELAY_LOGGING_H
+#define DELAY_LOGGING_H
+
+#define NETEQ_DELAY_LOGGING_VERSION_STRING "2.0"
+
+#define NETEQ_DELAY_LOGGING_SIGNAL_RECIN 1
+#define NETEQ_DELAY_LOGGING_SIGNAL_FLUSH 2
+#define NETEQ_DELAY_LOGGING_SIGNAL_CLOCK 3
+#define NETEQ_DELAY_LOGGING_SIGNAL_EOF 4
+#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE 5
+#define NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS 6
+#define NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO 7
+#define NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO 8
+#define NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO 9
+#define NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO 10
+#define NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF 11
+#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC 12
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/dsp.c b/trunk/src/modules/audio_coding/neteq/dsp.c
new file mode 100644
index 0000000..96f07d2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dsp.c
@@ -0,0 +1,522 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains some DSP initialization functions and
+ * constant table definitions.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+/* Filter coefficients used when downsampling from the indicated
+ sample rates (8, 16, 32, 48 kHz) to 4 kHz.
+ Coefficients are in Q12. */
+
+/* {0.3, 0.4, 0.3} */
+const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[] = { 1229, 1638, 1229 };
+
+#ifdef NETEQ_WIDEBAND
+/* {0.15, 0.2, 0.3, 0.2, 0.15} */
+const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[] =
+{ 614, 819, 1229, 819, 614};
+#endif
+
+#ifdef NETEQ_32KHZ_WIDEBAND
+/* {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425} */
+const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[] =
+{ 584, 512, 625, 667, 625, 512, 584};
+#endif
+
+#ifdef NETEQ_48KHZ_WIDEBAND
+/* {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487} */
+const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[] =
+{ 1019, 390, 427, 440, 427, 390, 1019};
+#endif
+
+/* Constants used in expand function WebRtcNetEQ_Expand */
+
+/* Q12: -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 */
+const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[4] = { -5179, 19931, -16422, 5776 };
+
+/* Tabulated divisions to save complexity */
+/* 1049/{0, .., 6} */
+const WebRtc_Word16 WebRtcNetEQ_k1049div[7] = { 0, 1049, 524, 349, 262, 209, 174 };
+
+/* 2097/{0, .., 6} */
+const WebRtc_Word16 WebRtcNetEQ_k2097div[7] = { 0, 2097, 1048, 699, 524, 419, 349 };
+
+/* 5243/{0, .., 6} */
+const WebRtc_Word16 WebRtcNetEQ_k5243div[7] = { 0, 5243, 2621, 1747, 1310, 1048, 873 };
+
+#ifdef WEBRTC_NETEQ_40BITACC_TEST
+/*
+ * Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP
+ * implementation where the main (spl and NetEQ) functions have been
+ * 40-bit optimized. For testing purposes.
+ */
+
+/****************************************************************************
+ * WebRtcNetEQ_40BitAccCrossCorr(...)
+ *
+ * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1
+ * is fixed and seq2 slides as the pointer is increased with step
+ *
+ * Input:
+ * - seq1 : First sequence (fixed throughout the correlation)
+ * - seq2 : Second sequence (slided step_seq2 for each
+ * new correlation)
+ * - dimSeq : Number of samples to use in the cross correlation.
+ * Should be no larger than 1024 to avoid overflow.
+ * - dimCrossCorr : Number of CrossCorrelations to calculate (start
+ * position for seq2 is updated for each new one)
+ * - rShift : Number of right shifts to use
+ * - step_seq2 : How many (positive or negative) steps the seq2
+ * pointer should be updated for each new cross
+ * correlation value
+ *
+ * Output:
+ * - crossCorr : The cross correlation in Q-rShift
+ */
+
+void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr,
+ WebRtc_Word16 *seq1,
+ WebRtc_Word16 *seq2,
+ WebRtc_Word16 dimSeq,
+ WebRtc_Word16 dimCrossCorr,
+ WebRtc_Word16 rShift,
+ WebRtc_Word16 step_seq2)
+{
+ int i, j;
+ WebRtc_Word16 *seq1Ptr, *seq2Ptr;
+ WebRtc_Word64 acc;
+
+ for (i = 0; i < dimCrossCorr; i++)
+ {
+ /* Set the pointer to the static vector, set the pointer to
+ the sliding vector and initialize crossCorr */
+ seq1Ptr = seq1;
+ seq2Ptr = seq2 + (step_seq2 * i);
+ acc = 0;
+
+ /* Perform the cross correlation */
+ for (j = 0; j < dimSeq; j++)
+ {
+ acc += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+ seq1Ptr++;
+ seq2Ptr++;
+ }
+
+ (*crossCorr) = (WebRtc_Word32) (acc >> rShift);
+ crossCorr++;
+ }
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_40BitAccDotW16W16(...)
+ *
+ * Calculates the dot product between two vectors (WebRtc_Word16)
+ *
+ * Input:
+ * - vector1 : Vector 1
+ * - vector2 : Vector 2
+ * - len : Number of samples in vector
+ * Should be no larger than 1024 to avoid overflow.
+ * - scaling : The number of left shifts required to avoid overflow
+ * in the dot product
+ * Return value : The dot product
+ */
+
+WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1,
+ WebRtc_Word16 *vector2,
+ int len,
+ int scaling)
+{
+ WebRtc_Word32 sum;
+ int i;
+ WebRtc_Word64 acc;
+
+ acc = 0;
+ for (i = 0; i < len; i++)
+ {
+ acc += WEBRTC_SPL_MUL_16_16(*vector1++, *vector2++);
+ }
+
+ sum = (WebRtc_Word32) (acc >> scaling);
+
+ return(sum);
+}
+
+#endif /* WEBRTC_NETEQ_40BITACC_TEST */
+
+/****************************************************************************
+ * WebRtcNetEQ_DSPInit(...)
+ *
+ * Initializes DSP side of NetEQ.
+ *
+ * Input:
+ * - inst : NetEq DSP instance
+ * - fs : Initial sample rate (may change when decoding data)
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ * : non-zero - error
+ */
+
+int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs)
+{
+
+ int res = 0;
+ WebRtc_Word16 fs_mult;
+
+ /* Pointers and values to save before clearing the instance */
+#ifdef NETEQ_CNG_CODEC
+ void *savedPtr1 = inst->CNG_Codec_inst;
+#endif
+ void *savedPtr2 = inst->pw16_readAddress;
+ void *savedPtr3 = inst->pw16_writeAddress;
+ void *savedPtr4 = inst->main_inst;
+#ifdef NETEQ_VAD
+ void *savedVADptr = inst->VADInst.VADState;
+ VADInitFunction savedVADinit = inst->VADInst.initFunction;
+ VADSetmodeFunction savedVADsetmode = inst->VADInst.setmodeFunction;
+ VADFunction savedVADfunc = inst->VADInst.VADFunction;
+ WebRtc_Word16 savedVADEnabled = inst->VADInst.VADEnabled;
+ WebRtc_Word16 savedVADMode = inst->VADInst.VADMode;
+#endif /* NETEQ_VAD */
+ DSPStats_t saveStats;
+ WebRtc_Word16 saveMsPerCall = inst->millisecondsPerCall;
+ enum BGNMode saveBgnMode = inst->BGNInst.bgnMode;
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo saveMSinfo;
+#endif
+
+ /* copy contents of statInst to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveStats, &(inst->statInst),
+ sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
+
+#ifdef NETEQ_STEREO
+ /* copy contents of msInfo to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveMSinfo, &(inst->msInfo),
+ sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
+#endif
+
+ /* check that the sample rate is valid */
+ if ((fs != 8000)
+#ifdef NETEQ_WIDEBAND
+ &&(fs!=16000)
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ &&(fs!=32000)
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ &&(fs!=48000)
+#endif
+ )
+ {
+ /* invalid rate */
+ return (CODEC_DB_UNSUPPORTED_FS);
+ }
+
+ /* calcualte fs/8000 */
+ fs_mult = WebRtcSpl_DivW32W16ResW16(fs, 8000);
+
+ /* Set everything to zero since most variables should be zero at start */
+ WebRtcSpl_MemSetW16((WebRtc_Word16 *) inst, 0, sizeof(DSPInst_t) / sizeof(WebRtc_Word16));
+
+ /* Restore saved pointers */
+#ifdef NETEQ_CNG_CODEC
+ inst->CNG_Codec_inst = (CNG_dec_inst *)savedPtr1;
+#endif
+ inst->pw16_readAddress = (WebRtc_Word16 *) savedPtr2;
+ inst->pw16_writeAddress = (WebRtc_Word16 *) savedPtr3;
+ inst->main_inst = savedPtr4;
+#ifdef NETEQ_VAD
+ inst->VADInst.VADState = savedVADptr;
+ inst->VADInst.initFunction = savedVADinit;
+ inst->VADInst.setmodeFunction = savedVADsetmode;
+ inst->VADInst.VADFunction = savedVADfunc;
+ inst->VADInst.VADEnabled = savedVADEnabled;
+ inst->VADInst.VADMode = savedVADMode;
+#endif /* NETEQ_VAD */
+
+ /* Initialize main part */
+ inst->fs = fs;
+ inst->millisecondsPerCall = saveMsPerCall;
+ inst->timestampsPerCall = inst->millisecondsPerCall * 8 * fs_mult;
+ inst->ExpandInst.w16_overlap = 5 * fs_mult;
+ inst->endPosition = 565 * fs_mult;
+ inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap;
+ inst->w16_seedInc = 1;
+ inst->uw16_seed = 777;
+ inst->w16_muteFactor = 16384; /* 1.0 in Q14 */
+ inst->w16_frameLen = 3 * inst->timestampsPerCall; /* Dummy initialize to 30ms */
+
+ inst->w16_speechHistoryLen = 256 * fs_mult;
+ inst->pw16_speechHistory = &inst->speechBuffer[inst->endPosition
+ - inst->w16_speechHistoryLen];
+ inst->ExpandInst.pw16_overlapVec = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen
+ - inst->ExpandInst.w16_overlap]);
+
+ /* Reusage of memory in speechBuffer inside Expand */
+ inst->ExpandInst.pw16_expVecs[0] = &inst->speechBuffer[0];
+ inst->ExpandInst.pw16_expVecs[1] = &inst->speechBuffer[126 * fs_mult];
+ inst->ExpandInst.pw16_arState = &inst->speechBuffer[2 * 126 * fs_mult];
+ inst->ExpandInst.pw16_arFilter = &inst->speechBuffer[2 * 126 * fs_mult
+ + UNVOICED_LPC_ORDER];
+ /* Ends at 2*126*fs_mult+UNVOICED_LPC_ORDER+(UNVOICED_LPC_ORDER+1) */
+
+ inst->ExpandInst.w16_expandMuteFactor = 16384; /* 1.0 in Q14 */
+
+ /* Initialize BGN part */
+ inst->BGNInst.pw16_filter[0] = 4096;
+ inst->BGNInst.w16_scale = 20000;
+ inst->BGNInst.w16_scaleShift = 24;
+ inst->BGNInst.w32_energyUpdate = 500000;
+ inst->BGNInst.w32_energyUpdateLow = 0;
+ inst->BGNInst.w32_energy = 2500;
+ inst->BGNInst.w16_initialized = 0;
+ inst->BGNInst.bgnMode = saveBgnMode;
+
+ /* Recreate statistics counters */WEBRTC_SPL_MEMCPY_W16(&(inst->statInst), &saveStats,
+ sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
+
+#ifdef NETEQ_STEREO
+ /* Recreate MSinfo */WEBRTC_SPL_MEMCPY_W16(&(inst->msInfo), &saveMSinfo,
+ sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
+#endif
+
+#ifdef NETEQ_CNG_CODEC
+ if (inst->CNG_Codec_inst!=NULL)
+ {
+ /* initialize comfort noise generator */
+ res |= WebRtcCng_InitDec(inst->CNG_Codec_inst);
+ }
+#endif
+
+#ifdef NETEQ_VAD
+ /* initialize PostDecode VAD instance
+ (don't bother checking for NULL instance, this is done inside init function) */
+ res |= WebRtcNetEQ_InitVAD(&inst->VADInst, fs);
+#endif /* NETEQ_VAD */
+
+ return (res);
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_AddressInit(...)
+ *
+ * Initializes the shared-memory communication on the DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - data2McuAddress : Pointer to memory where DSP writes / MCU reads
+ * - data2DspAddress : Pointer to memory where MCU writes / DSP reads
+ * - mainInst : NetEQ main instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress,
+ const void *data2DspAddress, const void *mainInst)
+{
+
+ /* set shared-memory addresses in the DSP instance */
+ inst->pw16_readAddress = (WebRtc_Word16 *) data2DspAddress;
+ inst->pw16_writeAddress = (WebRtc_Word16 *) data2McuAddress;
+
+ /* set pointer to main NetEQ instance */
+ inst->main_inst = (void *) mainInst;
+
+ /* set output frame size to 10 ms = 80 samples in narrowband */
+ inst->millisecondsPerCall = 10;
+ inst->timestampsPerCall = 80;
+
+ return (0);
+
+}
+
+/****************************************************************************
+ * NETEQDSP_clearInCallStats(...)
+ *
+ * Reset in-call statistics variables on DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst)
+{
+
+ /* Reset statistics counters */
+ inst->statInst.accelerateLength = 0;
+ inst->statInst.expandLength = 0;
+ inst->statInst.preemptiveLength = 0;
+
+ return (0);
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_ClearPostCallStats(...)
+ *
+ * Reset post-call statistics variables on DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst)
+{
+
+ /* Reset statistics counters */
+ inst->statInst.expandedVoiceSamples = 0;
+ inst->statInst.expandedNoiseSamples = 0;
+
+ return (0);
+}
+
+#ifdef NETEQ_VAD
+
+/****************************************************************************
+ * WebRtcNetEQ_InitVAD(...)
+ *
+ * Initializes post-decode VAD instance.
+ *
+ * Input:
+ * - VADinst : PostDecodeVAD instance
+ * - fs : Initial sample rate
+ *
+ * Output:
+ * - VADinst : Updated instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs)
+{
+
+ int res = 0;
+
+ /* initially, disable the post-decode VAD */
+ VADInst->VADEnabled = 0;
+
+ if (VADInst->VADState != NULL /* if VAD state is provided */
+ && VADInst->initFunction != NULL /* and all function ... */
+ && VADInst->setmodeFunction != NULL /* ... pointers ... */
+ && VADInst->VADFunction != NULL) /* ... are defined */
+ {
+ res = VADInst->initFunction( VADInst->VADState ); /* call VAD init function */
+ res |= WebRtcNetEQ_SetVADModeInternal( VADInst, VADInst->VADMode );
+
+ if (res!=0)
+ {
+ /* something is wrong; play it safe and set the VADstate to NULL */
+ VADInst->VADState = NULL;
+ }
+ else if (fs<=16000)
+ {
+ /* enable VAD if NB or WB (VAD cannot handle SWB) */
+ VADInst->VADEnabled = 1;
+ }
+ }
+
+ /* reset SID/CNG interval counter */
+ VADInst->SIDintervalCounter = 0;
+
+ /* initialize with active-speaker decision */
+ VADInst->VADDecision = 1;
+
+ return(res);
+
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADModeInternal(...)
+ *
+ * Set the VAD mode in the VAD struct, and communicate it to the VAD instance
+ * if it exists.
+ *
+ * Input:
+ * - VADinst : PostDecodeVAD instance
+ * - mode : Mode number passed on to the VAD function
+ *
+ * Output:
+ * - VADinst : Updated instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode)
+{
+
+ int res = 0;
+
+ VADInst->VADMode = mode;
+
+ if (VADInst->VADState != NULL)
+ {
+ /* call setmode function */
+ res = VADInst->setmodeFunction(VADInst->VADState, mode);
+ }
+
+ return(res);
+
+}
+
+#endif /* NETEQ_VAD */
+
+/****************************************************************************
+ * WebRtcNetEQ_FlushSpeechBuffer(...)
+ *
+ * Flush the speech buffer.
+ *
+ * Input:
+ * - inst : NetEq DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ * : non-zero - error
+ */
+
+int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst)
+{
+ WebRtc_Word16 fs_mult;
+
+ /* calcualte fs/8000 */
+ fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
+
+ /* clear buffer */
+ WebRtcSpl_MemSetW16(inst->speechBuffer, 0, SPEECH_BUF_SIZE);
+ inst->endPosition = 565 * fs_mult;
+ inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap;
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/dsp.h b/trunk/src/modules/audio_coding/neteq/dsp.h
new file mode 100644
index 0000000..94213e4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dsp.h
@@ -0,0 +1,788 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains some DSP initialization functions,
+ * constant table definitions and other parameters.
+ * Also contains definitions of all DSP-side data structures.
+ */
+
+
+#ifndef DSP_H
+#define DSP_H
+
+#include "typedefs.h"
+
+#include "webrtc_cng.h"
+
+#include "codec_db_defines.h"
+#include "neteq_defines.h"
+#include "neteq_statistics.h"
+
+#ifdef NETEQ_ATEVENT_DECODE
+#include "dtmf_tonegen.h"
+#endif
+
+
+
+/*****************************/
+/* Pre-processor definitions */
+/*****************************/
+
+/* FSMULT is the sample rate divided by 8000 */
+#if defined(NETEQ_48KHZ_WIDEBAND)
+ #define FSMULT 6
+#elif defined(NETEQ_32KHZ_WIDEBAND)
+ #define FSMULT 4
+#elif defined(NETEQ_WIDEBAND)
+ #define FSMULT 2
+#else
+ #define FSMULT 1
+#endif
+
+/* Size of the speech buffer (or synchronization buffer). */
+/* 60 ms decoding + 10 ms syncbuff + 0.625ms lookahead */
+#define SPEECH_BUF_SIZE (565 * FSMULT)
+
+/* Misc definitions */
+#define BGN_LPC_ORDER (4 + FSMULT) /* 5, 6, 8, or 10 */
+#define UNVOICED_LPC_ORDER 6
+#define RANDVEC_NO_OF_SAMPLES 256
+
+/* Number of milliseconds to remove/add during accelerate/pre-emptive expand
+ under BGNonly operation */
+#define DEFAULT_TIME_ADJUST 8
+
+/* Number of RecOut calls without CNG/SID before re-enabling post-decode VAD */
+#define POST_DECODE_VAD_AUTO_ENABLE 3000
+
+/* 8kHz windowing in Q15 (over 5 samples) */
+#define NETEQ_OVERLAP_WINMUTE_8KHZ_START 27307
+#define NETEQ_OVERLAP_WINMUTE_8KHZ_INC -5461
+#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_START 5461
+#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC 5461
+/* 16kHz windowing in Q15 (over 10 samples) */
+#define NETEQ_OVERLAP_WINMUTE_16KHZ_START 29789
+#define NETEQ_OVERLAP_WINMUTE_16KHZ_INC -2979
+#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_START 2979
+#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC 2979
+/* 32kHz windowing in Q15 (over 20 samples) */
+#define NETEQ_OVERLAP_WINMUTE_32KHZ_START 31208
+#define NETEQ_OVERLAP_WINMUTE_32KHZ_INC -1560
+#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_START 1560
+#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC 1560
+/* 48kHz windowing in Q15 (over 30 samples) */
+#define NETEQ_OVERLAP_WINMUTE_48KHZ_START 31711
+#define NETEQ_OVERLAP_WINMUTE_48KHZ_INC -1057
+#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_START 1057
+#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC 1057
+
+/* Fade BGN towards zero after this many Expand calls */
+#define FADE_BGN_TIME 200
+
+
+/*******************/
+/* Constant tables */
+/*******************/
+
+extern const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[];
+extern const WebRtc_Word16 WebRtcNetEQ_k1049div[];
+extern const WebRtc_Word16 WebRtcNetEQ_k2097div[];
+extern const WebRtc_Word16 WebRtcNetEQ_k5243div[];
+
+
+
+/************/
+/* Typedefs */
+/************/
+
+enum BGNMode
+{
+ BGN_ON, /* default "normal" behavior with eternal noise */
+ BGN_FADE, /* noise fades to zero after some time */
+ BGN_OFF /* background noise is always zero */
+};
+
+#ifdef NETEQ_STEREO
+enum MasterSlaveMode
+{
+ NETEQ_MONO, /* stand-alone instance */
+ NETEQ_MASTER, /* master instance in a spatial/stereo configuration */
+ NETEQ_SLAVE /* slave instance in a spatial/stereo configuration */
+};
+
+enum MasterSlaveExtraInfo
+{
+ NO_INFO, /* no info to convey */
+ ACC_FAIL, /* signal that accelerate failed */
+ PE_EXP_FAIL, /* signal that pre-emptive expand failed */
+ DTMF_OVERDUB, /* signal that DTMF overdub is generated */
+ DTMF_ONLY /* signal that DTMF only is played */
+};
+#endif
+
+/****************************/
+/* DSP-side data structures */
+/****************************/
+
+/* Background noise (BGN) instance for storing BGN parameters
+ (sub-instance of NETEQDSP_inst) */
+typedef struct BGNInst_t_
+{
+
+ WebRtc_Word32 w32_energy;
+ WebRtc_Word32 w32_energyMax;
+ WebRtc_Word32 w32_energyUpdate;
+ WebRtc_Word32 w32_energyUpdateLow;
+ WebRtc_Word16 pw16_filterState[BGN_LPC_ORDER];
+ WebRtc_Word16 pw16_filter[BGN_LPC_ORDER + 1];
+ WebRtc_Word16 w16_mutefactor;
+ WebRtc_Word16 w16_scale;
+ WebRtc_Word16 w16_scaleShift;
+ WebRtc_Word16 w16_initialized;
+ enum BGNMode bgnMode;
+
+} BGNInst_t;
+
+/* Expansion instance (sub-instance of NETEQDSP_inst) */
+typedef struct ExpandInst_t_
+{
+
+ WebRtc_Word16 w16_overlap; /* Constant, 5 for NB and 10 for WB */
+ WebRtc_Word16 w16_consecExp; /* Number of consecutive expand calls */
+ WebRtc_Word16 *pw16_arFilter; /* length [UNVOICED_LPC_ORDER+1] */
+ WebRtc_Word16 *pw16_arState; /* length [UNVOICED_LPC_ORDER] */
+ WebRtc_Word16 w16_arGain;
+ WebRtc_Word16 w16_arGainScale;
+ WebRtc_Word16 w16_vFraction; /* Q14 */
+ WebRtc_Word16 w16_currentVFraction; /* Q14 */
+ WebRtc_Word16 *pw16_expVecs[2];
+ WebRtc_Word16 w16_lags[3];
+ WebRtc_Word16 w16_maxLag;
+ WebRtc_Word16 *pw16_overlapVec; /* last samples of speech history */
+ WebRtc_Word16 w16_lagsDirection;
+ WebRtc_Word16 w16_lagsPosition;
+ WebRtc_Word16 w16_expandMuteFactor; /* Q14 */
+ WebRtc_Word16 w16_stopMuting;
+ WebRtc_Word16 w16_onset;
+ WebRtc_Word16 w16_muteSlope; /* Q20 */
+
+} ExpandInst_t;
+
+#ifdef NETEQ_VAD
+
+/*
+ * VAD function pointer types, replicating the typedefs in webrtc_neteq_internal.h.
+ * These function pointers match the definitions of WebRtc VAD functions WebRtcVad_Init,
+ * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h.
+ */
+typedef int (*VADInitFunction)(void *VAD_inst);
+typedef int (*VADSetmodeFunction)(void *VAD_inst, int mode);
+typedef WebRtc_Word16 (*VADFunction)(void *VAD_inst, WebRtc_Word16 fs, WebRtc_Word16 *frame,
+ WebRtc_Word16 frameLen);
+
+/* Post-decode VAD instance (sub-instance of NETEQDSP_inst) */
+typedef struct PostDecodeVAD_t_
+{
+
+ void *VADState; /* pointer to a VAD instance */
+
+ WebRtc_Word16 VADEnabled; /* 1 if enabled, 0 if disabled */
+ int VADMode; /* mode parameter to pass to the VAD function */
+ WebRtc_Word16 VADDecision; /* 1 for active, 0 for passive */
+ WebRtc_Word16 SIDintervalCounter; /* reset when decoding CNG/SID frame,
+ increment for each recout call */
+
+ /* Function pointers */
+ VADInitFunction initFunction; /* VAD init function */
+ VADSetmodeFunction setmodeFunction; /* VAD setmode function */
+ VADFunction VADFunction; /* VAD function */
+
+} PostDecodeVAD_t;
+
+#endif /* NETEQ_VAD */
+
+#ifdef NETEQ_STEREO
+#define MAX_MS_DECODES 10
+
+typedef struct
+{
+ /* Stand-alone, master, or slave */
+ enum MasterSlaveMode msMode;
+
+ enum MasterSlaveExtraInfo extraInfo;
+
+ WebRtc_UWord16 instruction;
+ WebRtc_Word16 distLag;
+ WebRtc_Word16 corrLag;
+ WebRtc_Word16 bestIndex;
+
+ WebRtc_UWord32 endTimestamp;
+ WebRtc_UWord16 samplesLeftWithOverlap;
+
+} MasterSlaveInfo;
+#endif
+
+
+/* "Main" NetEQ DSP instance */
+typedef struct DSPInst_t_
+{
+
+ /* MCU/DSP Communication layer */
+ WebRtc_Word16 *pw16_readAddress;
+ WebRtc_Word16 *pw16_writeAddress;
+ void *main_inst;
+
+ /* Output frame size in ms and samples */
+ WebRtc_Word16 millisecondsPerCall;
+ WebRtc_Word16 timestampsPerCall;
+
+ /*
+ * Example of speech buffer
+ *
+ * -----------------------------------------------------------
+ * | History T-60 to T | Future |
+ * -----------------------------------------------------------
+ * ^ ^
+ * | |
+ * curPosition endPosition
+ *
+ * History is gradually shifted out to the left when inserting
+ * new data at the end.
+ */
+
+ WebRtc_Word16 speechBuffer[SPEECH_BUF_SIZE]; /* History/future speech buffer */
+ int curPosition; /* Next sample to play */
+ int endPosition; /* Position that ends future data */
+ WebRtc_UWord32 endTimestamp; /* Timestamp value at end of future data */
+ WebRtc_UWord32 videoSyncTimestamp; /* (Estimated) timestamp of the last
+ played sample (usually same as
+ endTimestamp-(endPosition-curPosition)
+ except during Expand and CNG) */
+ WebRtc_UWord16 fs; /* sample rate in Hz */
+ WebRtc_Word16 w16_frameLen; /* decoder frame length in samples */
+ WebRtc_Word16 w16_mode; /* operation used during last RecOut call */
+ WebRtc_Word16 w16_muteFactor; /* speech mute factor in Q14 */
+ WebRtc_Word16 *pw16_speechHistory; /* beginning of speech history during Expand */
+ WebRtc_Word16 w16_speechHistoryLen; /* 256 for NB and 512 for WB */
+
+ /* random noise seed parameters */
+ WebRtc_Word16 w16_seedInc;
+ WebRtc_UWord32 uw16_seed;
+
+ /* VQmon related variable */
+ WebRtc_Word16 w16_concealedTS;
+
+ /*****************/
+ /* Sub-instances */
+ /*****************/
+
+ /* Decoder data */
+ CodecFuncInst_t codec_ptr_inst;
+
+#ifdef NETEQ_CNG_CODEC
+ /* CNG "decoder" instance */
+ CNG_dec_inst *CNG_Codec_inst;
+#endif /* NETEQ_CNG_CODEC */
+
+#ifdef NETEQ_ATEVENT_DECODE
+ /* DTMF generator instance */
+ dtmf_tone_inst_t DTMFInst;
+#endif /* NETEQ_CNG_CODEC */
+
+#ifdef NETEQ_VAD
+ /* Post-decode VAD instance */
+ PostDecodeVAD_t VADInst;
+#endif /* NETEQ_VAD */
+
+ /* Expand instance (defined above) */
+ ExpandInst_t ExpandInst;
+
+ /* Background noise instance (defined above) */
+ BGNInst_t BGNInst;
+
+ /* Internal statistics instance */
+ DSPStats_t statInst;
+
+#ifdef NETEQ_STEREO
+ /* Pointer to Master/Slave info */
+ MasterSlaveInfo *msInfo;
+#endif
+
+} DSPInst_t;
+
+
+/*************************/
+/* Function declarations */
+/*************************/
+
+/****************************************************************************
+ * WebRtcNetEQ_DSPInit(...)
+ *
+ * Initializes DSP side of NetEQ.
+ *
+ * Input:
+ * - inst : NetEq DSP instance
+ * - fs : Initial sample rate (may change when decoding data)
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ * : non-zero - error
+ */
+
+int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs);
+
+/****************************************************************************
+ * WebRtcNetEQ_AddressInit(...)
+ *
+ * Initializes the shared-memory communication on the DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - data2McuAddress : Pointer to memory where DSP writes / MCU reads
+ * - data2DspAddress : Pointer to memory where MCU writes / DSP reads
+ * - mainInst : NetEQ main instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress,
+ const void *data2DspAddress, const void *mainInst);
+
+/****************************************************************************
+ * WebRtcNetEQ_ClearInCallStats(...)
+ *
+ * Reset in-call statistics variables on DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_ClearPostCallStats(...)
+ *
+ * Reset post-call statistics variables on DSP side.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ */
+
+int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_RecOutInternal(...)
+ *
+ * This function asks NetEQ for more speech/audio data.
+ *
+ * Input:
+ * - inst : NetEQ instance, i.e. the user that requests more
+ * speech/audio data.
+ * - outdata : Pointer to a memory space where the output data
+ * should be stored.
+ * - BGNonly : If non-zero, RecOut will only produce background
+ * noise. It will still draw packets from the packet
+ * buffer, but they will never be decoded.
+ *
+ * Output:
+ * - inst : Updated user information
+ * - len : Number of samples that were outputted from NetEq
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly);
+
+/****************************************************************************
+ * WebRtcNetEQ_Normal(...)
+ *
+ * This function has the possibility to modify data that is played out in Normal
+ * mode, for example adjust the gain of the signal. The length of the signal
+ * can not be changed.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector
+ * - decoded : Pointer to vector of new data from decoder
+ * - len : Number of input samples
+ *
+ * Output:
+ * - inst : Updated user information
+ * - pw16_len : Pointer to varibale where the number of samples
+ * produced will be written
+ *
+ * Return value : >=0 - Number of samples written to outData
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_Normal(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
+
+/****************************************************************************
+ * WebRtcNetEQ_Expand(...)
+ *
+ * This function produces one "chunk" of expansion data (PLC audio). The
+ * lenght of the produced audio depends on the speech history.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector
+ * - BGNonly : If non-zero, Expand will only produce background
+ * noise.
+ * - pw16_len : Desired number of samples (only for BGN mode).
+ *
+ * Output:
+ * - inst : Updated user information
+ * - outdata : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Number of samples that were outputted from NetEq
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_Expand(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly);
+
+/****************************************************************************
+ * WebRtcNetEQ_GenerateBGN(...)
+ *
+ * This function generates and writes len samples of background noise to the
+ * output vector. The Expand function will be called repeteadly until the
+ * correct number of samples is produced.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector
+ * - len : Desired length of produced BGN.
+ *
+ *
+ * Output:
+ * - pw16_outData : Pointer to a memory space where the output data
+ * should be stored
+ *
+ * Return value : >=0 - Number of noise samples produced and written
+ * to output
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 len);
+
+/****************************************************************************
+ * WebRtcNetEQ_PreEmptiveExpand(...)
+ *
+ * This function tries to extend the audio data by repeating one or several
+ * pitch periods. The operation is only carried out if the correlation is
+ * strong or if the signal energy is very low. The algorithm is the
+ * reciprocal of the Accelerate algorithm.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to newly decoded speech.
+ * - len : Length of decoded speech.
+ * - oldDataLen : Length of the part of decoded that has already been played out.
+ * - BGNonly : If non-zero, Pre-emptive Expand will only copy
+ * the first DEFAULT_TIME_ADJUST seconds of the
+ * input and append to the end. No signal matching is
+ * done.
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outData : Pointer to a memory space where the output data
+ * should be stored. The vector must be at least
+ * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE)
+ * elements long.
+ * - pw16_len : Number of samples written to outData.
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly);
+
+/****************************************************************************
+ * WebRtcNetEQ_Accelerate(...)
+ *
+ * This function tries to shorten the audio data by removing one or several
+ * pitch periods. The operation is only carried out if the correlation is
+ * strong or if the signal energy is very low.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to newly decoded speech.
+ * - len : Length of decoded speech.
+ * - BGNonly : If non-zero, Accelerate will only remove the last
+ * DEFAULT_TIME_ADJUST seconds of the intput.
+ * No signal matching is done.
+ *
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outData : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Number of samples written to outData.
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ const WebRtc_Word16 *pw16_decoded, int len,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly);
+
+/****************************************************************************
+ * WebRtcNetEQ_Merge(...)
+ *
+ * This function is used to merge new data from the decoder to the exisiting
+ * stream in the synchronization buffer. The merge operation is typically
+ * done after a packet loss, where the end of the expanded data does not
+ * fit naturally with the new decoded data.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to new decoded speech.
+ * - len : Number of samples in pw16_decoded.
+ *
+ *
+ * Output:
+ * - inst : Updated user information
+ * - outData : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Number of samples written to pw16_outData
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_Merge(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len);
+
+/****************************************************************************
+ * WebRtcNetEQ_Cng(...)
+ *
+ * This function produces CNG according to RFC 3389
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - len : Number of samples to produce
+ *
+ * Output:
+ * - pw16_outData : Output CNG
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+#ifdef NETEQ_CNG_CODEC
+/* Must compile NetEQ with CNG support to enable this function */
+
+int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len);
+
+#endif /* NETEQ_CNG_CODEC */
+
+/****************************************************************************
+ * WebRtcNetEQ_BGNUpdate(...)
+ *
+ * This function updates the background noise parameter estimates.
+ *
+ * Input:
+ * - inst : NetEQ instance, where the speech history is stored.
+ * - scratchPtr : Pointer to scratch vector.
+ *
+ * Output:
+ * - inst : Updated information about the BGN characteristics.
+ *
+ * Return value : No return value
+ */
+
+void WebRtcNetEQ_BGNUpdate(
+#ifdef SCRATCH
+ DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
+#else
+ DSPInst_t *inst
+#endif
+ );
+
+#ifdef NETEQ_VAD
+/* Functions used by post-decode VAD */
+
+/****************************************************************************
+ * WebRtcNetEQ_InitVAD(...)
+ *
+ * Initializes post-decode VAD instance.
+ *
+ * Input:
+ * - VADinst : PostDecodeVAD instance
+ * - fs : Initial sample rate
+ *
+ * Output:
+ * - VADinst : Updated instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs);
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADModeInternal(...)
+ *
+ * Set the VAD mode in the VAD struct, and communicate it to the VAD instance
+ * if it exists.
+ *
+ * Input:
+ * - VADinst : PostDecodeVAD instance
+ * - mode : Mode number passed on to the VAD function
+ *
+ * Output:
+ * - VADinst : Updated instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode);
+
+#endif /* NETEQ_VAD */
+
+/****************************************************************************
+ * WebRtcNetEQ_FlushSpeechBuffer(...)
+ *
+ * Flush the speech buffer.
+ *
+ * Input:
+ * - inst : NetEq DSP instance
+ *
+ * Output:
+ * - inst : Updated instance
+ *
+ * Return value : 0 - ok
+ * : non-zero - error
+ */
+
+int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst);
+
+#ifndef WEBRTC_NETEQ_40BITACC_TEST
+
+#include "signal_processing_library.h"
+/* Map to regular SPL functions */
+#define WebRtcNetEQ_CrossCorr WebRtcSpl_CrossCorrelation
+#define WebRtcNetEQ_DotW16W16 WebRtcSpl_DotProductWithScale
+
+#else /* WEBRTC_NETEQ_40BITACC_TEST defined */
+/* Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP
+ implementation where the main (splib and NetEQ) functions have been
+ 40-bit optimized. */
+
+/* Map to special 40-bit optimized functions, defined below */
+#define WebRtcNetEQ_CrossCorr WebRtcNetEQ_40BitAccCrossCorr
+#define WebRtcNetEQ_DotW16W16 WebRtcNetEQ_40BitAccDotW16W16
+
+/****************************************************************************
+ * WebRtcNetEQ_40BitAccCrossCorr(...)
+ *
+ * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1
+ * is fixed and seq2 slides as the pointer is increased with step
+ *
+ * Input:
+ * - seq1 : First sequence (fixed throughout the correlation)
+ * - seq2 : Second sequence (slided step_seq2 for each
+ * new correlation)
+ * - dimSeq : Number of samples to use in the cross correlation.
+ * Should be no larger than 1024 to avoid overflow.
+ * - dimCrossCorr : Number of CrossCorrelations to calculate (start
+ * position for seq2 is updated for each new one)
+ * - rShift : Number of right shifts to use
+ * - step_seq2 : How many (positive or negative) steps the seq2
+ * pointer should be updated for each new cross
+ * correlation value
+ *
+ * Output:
+ * - crossCorr : The cross correlation in Q-rShift
+ */
+
+void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr, WebRtc_Word16 *seq1,
+ WebRtc_Word16 *seq2, WebRtc_Word16 dimSeq,
+ WebRtc_Word16 dimCrossCorr, WebRtc_Word16 rShift,
+ WebRtc_Word16 step_seq2);
+
+/****************************************************************************
+ * WebRtcNetEQ_40BitAccDotW16W16(...)
+ *
+ * Calculates the dot product between two vectors (WebRtc_Word16)
+ *
+ * Input:
+ * - vector1 : Vector 1
+ * - vector2 : Vector 2
+ * - len : Number of samples in vector
+ * Should be no larger than 1024 to avoid overflow.
+ * - scaling : The number of right shifts (after multiplication)
+ * required to avoid overflow in the dot product.
+ * Return value : The dot product
+ */
+
+WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
+ int len, int scaling);
+
+#endif /* WEBRTC_NETEQ_40BITACC_TEST */
+
+#endif /* DSP_H */
diff --git a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c
new file mode 100644
index 0000000..6e9a283
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains some help functions that did not fit elsewhere.
+ */
+
+#include "dsp_helpfunctions.h"
+
+
+WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz)
+{
+ switch (fsHz)
+ {
+ case 8000:
+ {
+ return 1;
+ }
+ case 16000:
+ {
+ return 2;
+ }
+ case 32000:
+ {
+ return 4;
+ }
+ case 48000:
+ {
+ return 6;
+ }
+ default:
+ {
+ return 1;
+ }
+ }
+}
+
+
+int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
+ WebRtc_Word16 *out, int outLen, int compensateDelay)
+{
+ WebRtc_Word16 *B; /* filter coefficients */
+ WebRtc_Word16 Blen; /* number of coefficients */
+ WebRtc_Word16 filterDelay; /* phase delay in samples */
+ WebRtc_Word16 factor; /* conversion rate (inFsHz/8000) */
+ int ok;
+
+ /* Set constants depending on frequency used */
+ /* NOTE: The phase delay values are wrong compared to the true phase delay
+ of the filters. However, the error is preserved (through the +1 term)
+ for consistency. */
+ switch (inFsHz)
+ {
+ case 8000:
+ {
+ Blen = 3;
+ factor = 2;
+ B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
+ filterDelay = 1 + 1;
+ break;
+ }
+#ifdef NETEQ_WIDEBAND
+ case 16000:
+ {
+ Blen = 5;
+ factor = 4;
+ B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample16kHzTbl;
+ filterDelay = 2 + 1;
+ break;
+ }
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ case 32000:
+ {
+ Blen = 7;
+ factor = 8;
+ B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample32kHzTbl;
+ filterDelay = 3 + 1;
+ break;
+ }
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ case 48000:
+ {
+ Blen = 7;
+ factor = 12;
+ B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample48kHzTbl;
+ filterDelay = 3 + 1;
+ break;
+ }
+#endif
+ default:
+ {
+ /* unsupported or wrong sample rate */
+ return -1;
+ }
+ }
+
+ if (!compensateDelay)
+ {
+ /* disregard delay compensation */
+ filterDelay = 0;
+ }
+
+ ok = WebRtcSpl_DownsampleFast((WebRtc_Word16*) &in[Blen - 1],
+ (WebRtc_Word16) (inLen - (Blen - 1)), /* number of input samples */
+ out, (WebRtc_Word16) outLen, /* number of output samples to produce */
+ B, Blen, factor, filterDelay); /* filter parameters */
+
+ return ok; /* return value is -1 if input signal is too short */
+
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h
new file mode 100644
index 0000000..f728c09
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h
@@ -0,0 +1,220 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Various help functions used by the DSP functions.
+ */
+
+#ifndef DSP_HELPFUNCTIONS_H
+#define DSP_HELPFUNCTIONS_H
+
+#include "typedefs.h"
+
+#include "dsp.h"
+
+/****************************************************************************
+ * WebRtcNetEQ_Correlator(...)
+ *
+ * Calculate signal correlation.
+ *
+ * Input:
+ * - inst : DSP instance
+ * - data : Speech history to do expand from (older history in data[-4..-1])
+ * - dataLen : Length of data
+ *
+ * Output:
+ * - corrOut : CC of downsampled signal
+ * - corrScale : Scale factor for correlation (-Qdomain)
+ *
+ * Return value : Length of correlated data
+ */
+
+WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
+ WebRtc_Word16 *pw16_corrOut,
+ WebRtc_Word16 *pw16_corrScale);
+
+/****************************************************************************
+ * WebRtcNetEQ_PeakDetection(...)
+ *
+ * Peak detection with parabolic fit.
+ *
+ * Input:
+ * - data : Data sequence for peak detection
+ * - dataLen : Length of data
+ * - nmbPeaks : Number of peaks to detect
+ * - fs_mult : Sample rate multiplier
+ *
+ * Output:
+ * - corrIndex : Index of the peak
+ * - winner : Value of the peak
+ *
+ * Return value : 0 for ok
+ */
+
+WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
+ WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
+ WebRtc_Word16 *pw16_corrIndex,
+ WebRtc_Word16 *pw16_winners);
+
+/****************************************************************************
+ * WebRtcNetEQ_PrblFit(...)
+ *
+ * Three-point parbola fit.
+ *
+ * Input:
+ * - 3pts : Three input samples
+ * - fs_mult : Sample rate multiplier
+ *
+ * Output:
+ * - Ind : Index of the peak
+ * - outVal : Value of the peak
+ *
+ * Return value : 0 for ok
+ */
+
+WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
+ WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult);
+
+/****************************************************************************
+ * WebRtcNetEQ_MinDistortion(...)
+ *
+ * Find the lag that results in minimum distortion.
+ *
+ * Input:
+ * - data : Start of speech to perform distortion on, second vector is assumed
+ * to be data[-Lag]
+ * - minLag : Start lag
+ * - maxLag : End lag
+ * - len : Length to correlate
+ *
+ * Output:
+ * - dist : Distorion value
+ *
+ * Return value : Lag for minimum distortion
+ */
+
+WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
+ WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
+ WebRtc_Word16 len, WebRtc_Word32 *pw16_dist);
+
+/****************************************************************************
+ * WebRtcNetEQ_RandomVec(...)
+ *
+ * Generate random vector.
+ *
+ * Input:
+ * - seed : Current seed (input/output)
+ * - len : Number of samples to generate
+ * - incVal : Jump step
+ *
+ * Output:
+ * - randVec : Generated random vector
+ */
+
+void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
+ WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval);
+
+/****************************************************************************
+ * WebRtcNetEQ_MixVoiceUnvoice(...)
+ *
+ * Mix voiced and unvoiced signal.
+ *
+ * Input:
+ * - voicedVec : Voiced input signal
+ * - unvoicedVec : Unvoiced input signal
+ * - current_vfraction : Current mixing factor
+ * - vfraction_change : Mixing factor change per sample
+ * - N : Number of samples
+ *
+ * Output:
+ * - outData : Mixed signal
+ */
+
+void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
+ WebRtc_Word16 *pw16_unvoicedVec,
+ WebRtc_Word16 *w16_current_vfraction,
+ WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N);
+
+/****************************************************************************
+ * WebRtcNetEQ_UnmuteSignal(...)
+ *
+ * Gradually reduce attenuation.
+ *
+ * Input:
+ * - inVec : Input signal
+ * - startMuteFact : Starting attenuation
+ * - unmuteFact : Factor to "unmute" with (Q20)
+ * - N : Number of samples
+ *
+ * Output:
+ * - outVec : Output signal
+ */
+
+void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
+ WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
+ WebRtc_Word16 N);
+
+/****************************************************************************
+ * WebRtcNetEQ_MuteSignal(...)
+ *
+ * Gradually increase attenuation.
+ *
+ * Input:
+ * - inout : Input/output signal
+ * - muteSlope : Slope of muting
+ * - N : Number of samples
+ */
+
+void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
+ WebRtc_Word16 N);
+
+/****************************************************************************
+ * WebRtcNetEQ_CalcFsMult(...)
+ *
+ * Calculate the sample rate divided by 8000.
+ *
+ * Input:
+ * - fsHz : Sample rate in Hz in {8000, 16000, 32000, 48000}.
+ *
+ * Return value : fsHz/8000 for the valid values, 1 for other inputs
+ */
+
+WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz);
+
+/****************************************************************************
+ * WebRtcNetEQ_DownSampleTo4kHz(...)
+ *
+ * Lowpass filter and downsample a signal to 4 kHz sample rate.
+ *
+ * Input:
+ * - in : Input signal samples.
+ * - inLen : Number of input samples.
+ * - inFsHz : Input sample rate in Hz.
+ * - outLen : Desired number of samples in decimated signal.
+ * - compensateDelay : If non-zero, compensate for the phase delay of
+ * of the anti-alias filter.
+ *
+ * Output:
+ * - out : Output signal samples.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ *
+ */
+
+int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
+ WebRtc_Word16 *out, int outLen, int compensateDelay);
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c
new file mode 100644
index 0000000..f00f9c9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c
@@ -0,0 +1,232 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of packet buffer for DTMF messages.
+ */
+
+#include "dtmf_buffer.h"
+
+#include "typedefs.h" /* to define endianness */
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+
+#ifdef NETEQ_ATEVENT_DECODE
+
+WebRtc_Word16 WebRtcNetEQ_DtmfRemoveEvent(dtmf_inst_t *DTMFdec_inst)
+{
+
+ int i;
+ for (i = 0; i < 3; i++)
+ {
+ DTMFdec_inst->EventQueue[i] = DTMFdec_inst->EventQueue[i + 1];
+ DTMFdec_inst->EventQueueVolume[i] = DTMFdec_inst->EventQueueVolume[i + 1];
+ DTMFdec_inst->EventQueueEnded[i] = DTMFdec_inst->EventQueueEnded[i + 1];
+ DTMFdec_inst->EventQueueStartTime[i] = DTMFdec_inst->EventQueueStartTime[i + 1];
+ DTMFdec_inst->EventQueueEndTime[i] = DTMFdec_inst->EventQueueEndTime[i + 1];
+ }
+ DTMFdec_inst->EventBufferSize--;
+ DTMFdec_inst->EventQueue[3] = -1;
+ DTMFdec_inst->EventQueueVolume[3] = 0;
+ DTMFdec_inst->EventQueueEnded[3] = 0;
+ DTMFdec_inst->EventQueueStartTime[3] = 0;
+ DTMFdec_inst->EventQueueEndTime[3] = 0;
+
+ return 0;
+}
+
+WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
+ WebRtc_Word16 MaxPLCtime)
+{
+ int i;
+ if (((fs != 8000) && (fs != 16000) && (fs != 32000) && (fs != 48000)) || (MaxPLCtime < 0))
+ {
+ return DTMF_DEC_PARAMETER_ERROR;
+ }
+ if (fs == 8000)
+ DTMFdec_inst->framelen = 80;
+ else if (fs == 16000)
+ DTMFdec_inst->framelen = 160;
+ else if (fs == 32000)
+ DTMFdec_inst->framelen = 320;
+ else
+ /* fs == 48000 */
+ DTMFdec_inst->framelen = 480;
+
+ DTMFdec_inst->MaxPLCtime = MaxPLCtime;
+ DTMFdec_inst->CurrentPLCtime = 0;
+ DTMFdec_inst->EventBufferSize = 0;
+ for (i = 0; i < 4; i++)
+ {
+ DTMFdec_inst->EventQueue[i] = -1;
+ DTMFdec_inst->EventQueueVolume[i] = 0;
+ DTMFdec_inst->EventQueueEnded[i] = 0;
+ DTMFdec_inst->EventQueueStartTime[i] = 0;
+ DTMFdec_inst->EventQueueEndTime[i] = 0;
+ }
+ return 0;
+}
+
+WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
+ const WebRtc_Word16 *encoded, WebRtc_Word16 len,
+ WebRtc_UWord32 timeStamp)
+{
+
+ int i;
+ WebRtc_Word16 value;
+ const WebRtc_Word16 *EventStart;
+ WebRtc_Word16 endEvent;
+ WebRtc_Word16 Volume;
+ WebRtc_Word16 Duration;
+ WebRtc_Word16 position = -1;
+
+ /* Extract event */
+ if (len == 4)
+ {
+ EventStart = encoded;
+#ifdef WEBRTC_BIG_ENDIAN
+ value=((*EventStart)>>8);
+ endEvent=((*EventStart)&0x80)>>7;
+ Volume=((*EventStart)&0x3F);
+ Duration=EventStart[1];
+#else
+ value = ((*EventStart) & 0xFF);
+ endEvent = ((*EventStart) & 0x8000) >> 15;
+ Volume = ((*EventStart) & 0x3F00) >> 8;
+ Duration = (((((WebRtc_UWord16) EventStart[1]) >> 8) & 0xFF)
+ | (((WebRtc_UWord16) (EventStart[1] & 0xFF)) << 8));
+#endif
+ /* Only events between 0-15 are supported (DTMF tones) */
+ if ((value < 0) || (value > 15))
+ {
+ return 0;
+ }
+
+ /* Discard all DTMF tones with really low volume (<-36dbm0) */
+ if (Volume > 36)
+ {
+ return 0;
+ }
+
+ /*Are there any unended events of the same type? */
+ for (i = 0; i < DTMFdec_inst->EventBufferSize; i++)
+ {
+ /* Going through the whole queue even when we have found a match will
+ ensure that we add to the latest applicable event */
+ if ((DTMFdec_inst->EventQueue[i] == value) && (!DTMFdec_inst->EventQueueEnded[i]
+ || endEvent)) position = i;
+ }
+ if (position > -1)
+ {
+ DTMFdec_inst->EventQueueVolume[position] = Volume;
+ if ((timeStamp + Duration) > DTMFdec_inst->EventQueueEndTime[position]) DTMFdec_inst->EventQueueEndTime[position]
+ = DTMFdec_inst->EventQueueStartTime[position] + Duration;
+ if (endEvent) DTMFdec_inst->EventQueueEnded[position] = 1;
+ }
+ else
+ {
+ if (DTMFdec_inst->EventBufferSize == MAX_DTMF_QUEUE_SIZE)
+ { /* Buffer full */
+ /* Remove one event */
+ DTMFdec_inst->EventBufferSize--;
+ }
+ /* Store data in the instance on a new position*/
+ DTMFdec_inst->EventQueue[DTMFdec_inst->EventBufferSize] = value;
+ DTMFdec_inst->EventQueueVolume[DTMFdec_inst->EventBufferSize] = Volume;
+ DTMFdec_inst->EventQueueEnded[DTMFdec_inst->EventBufferSize] = endEvent;
+ DTMFdec_inst->EventQueueStartTime[DTMFdec_inst->EventBufferSize] = timeStamp;
+ DTMFdec_inst->EventQueueEndTime[DTMFdec_inst->EventBufferSize] = timeStamp
+ + Duration;
+ DTMFdec_inst->EventBufferSize++;
+ }
+ return 0;
+ }
+ return DTMF_INSERT_ERROR;
+}
+
+WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
+ WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp)
+{
+
+ if (DTMFdec_inst->EventBufferSize < 1) return 0; /* No events to play */
+
+ /* We have events, is it time to play them? */
+ if (currTimeStamp < DTMFdec_inst->EventQueueStartTime[0])
+ {
+ /*No, just return zero */
+ return 0;
+ }
+
+ /* Continue on the event that is currently ongoing */
+ *event = DTMFdec_inst->EventQueue[0];
+ *volume = DTMFdec_inst->EventQueueVolume[0];
+
+ if (DTMFdec_inst->EventQueueEndTime[0] >= (currTimeStamp + DTMFdec_inst->framelen))
+ {
+
+ /* Still at least framLen to play */
+
+ DTMFdec_inst->CurrentPLCtime = 0;
+ if ((DTMFdec_inst->EventQueueEndTime[0] == (currTimeStamp + DTMFdec_inst->framelen))
+ && (DTMFdec_inst->EventQueueEnded[0]))
+ { /* We are done */
+ /*Remove the event from Queue*/
+ WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
+ }
+ return DTMFdec_inst->framelen;
+
+ }
+ else
+ {
+ if ((DTMFdec_inst->EventQueueEnded[0]) || (DTMFdec_inst->EventQueue[1] > -1))
+ {
+ /*
+ * Less than frameLen to play and end of event or already received next event.
+ * Give our a whole frame size of audio to simplify things.
+ */
+
+ /*Remove the event from Queue*/
+ WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
+ DTMFdec_inst->CurrentPLCtime = 0;
+
+ return DTMFdec_inst->framelen;
+
+ }
+ else
+ {
+ /* Less than frameLen to play and not end of event. */
+ DTMFdec_inst->CurrentPLCtime = (WebRtc_Word16) (currTimeStamp
+ - DTMFdec_inst->EventQueueEndTime[0]);
+
+ if ((DTMFdec_inst->CurrentPLCtime > DTMFdec_inst->MaxPLCtime)
+ || (DTMFdec_inst->CurrentPLCtime < -DTMFdec_inst->MaxPLCtime))
+ {
+ /*Remove the event from queue*/
+ WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
+ DTMFdec_inst->CurrentPLCtime = 0;
+ }
+
+ /* If we have a new event that it's time to play */
+ if ((DTMFdec_inst->EventQueue[1] > -1) && (DTMFdec_inst->EventQueueStartTime[1]
+ >= (currTimeStamp + DTMFdec_inst->framelen)))
+ {
+ /*Remove the event from queue*/
+ WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
+ DTMFdec_inst->CurrentPLCtime = 0;
+ }
+
+ return DTMFdec_inst->framelen;
+ }
+ }
+}
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h
new file mode 100644
index 0000000..e185411
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h
@@ -0,0 +1,101 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Packet buffer for DTMF messages.
+ */
+
+#ifndef DTMF_BUFFER_H
+#define DTMF_BUFFER_H
+
+#include "typedefs.h"
+
+#include "neteq_defines.h"
+
+/* Include this code only if ATEVENT (DTMF) is defined in */
+#ifdef NETEQ_ATEVENT_DECODE
+
+#define MAX_DTMF_QUEUE_SIZE 4
+
+typedef struct dtmf_inst_t_
+{
+ WebRtc_Word16 MaxPLCtime;
+ WebRtc_Word16 CurrentPLCtime;
+ WebRtc_Word16 EventQueue[MAX_DTMF_QUEUE_SIZE];
+ WebRtc_Word16 EventQueueVolume[MAX_DTMF_QUEUE_SIZE];
+ WebRtc_Word16 EventQueueEnded[MAX_DTMF_QUEUE_SIZE];
+ WebRtc_UWord32 EventQueueStartTime[MAX_DTMF_QUEUE_SIZE];
+ WebRtc_UWord32 EventQueueEndTime[MAX_DTMF_QUEUE_SIZE];
+ WebRtc_Word16 EventBufferSize;
+ WebRtc_Word16 framelen;
+} dtmf_inst_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_DtmfDecoderInit(...)
+ *
+ * This function initializes a DTMF instance.
+ *
+ * Input:
+ * - DTMF_decinst_t : DTMF instance
+ * - fs : The sample rate used for the DTMF
+ * - MaxPLCtime : Maximum length for a PLC before zeros should be inserted
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
+ WebRtc_Word16 MaxPLCtime);
+
+/****************************************************************************
+ * WebRtcNetEQ_DtmfInsertEvent(...)
+ *
+ * This function decodes a packet with DTMF frames.
+ *
+ * Input:
+ * - DTMFdec_inst : DTMF instance
+ * - encoded : Encoded DTMF frame(s)
+ * - len : Bytes in encoded vector
+ *
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
+ const WebRtc_Word16 *encoded, WebRtc_Word16 len,
+ WebRtc_UWord32 timeStamp);
+
+/****************************************************************************
+ * WebRtcNetEQ_DtmfDecode(...)
+ *
+ * This function decodes a packet with DTMF frame(s). Output will be the
+ * event that should be played for next 10 ms.
+ *
+ * Input:
+ * - DTMFdec_inst : DTMF instance
+ * - currTimeStamp : The current playout timestamp
+ *
+ * Output:
+ * - event : Event number to be played
+ * - volume : Event volume to be played
+ *
+ * Return value : >0 - There is a event to be played
+ * 0 - No event to be played
+ * -1 - Error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
+ WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp);
+
+#endif /* NETEQ_ATEVENT_DECODE */
+
+#endif /* DTMF_BUFFER_H */
+
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c
new file mode 100644
index 0000000..a52f9bc
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c
@@ -0,0 +1,371 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the DTMF tone generator and its parameters.
+ *
+ * A sinusoid is generated using the recursive oscillator model
+ *
+ * y[n] = sin(w*n + phi) = 2*cos(w) * y[n-1] - y[n-2]
+ * = a * y[n-1] - y[n-2]
+ *
+ * initialized with
+ * y[-2] = 0
+ * y[-1] = sin(w)
+ *
+ * A DTMF signal is a combination of two sinusoids, depending
+ * on which event is sent (i.e, which key is pressed). The following
+ * table maps each key (event codes in parentheses) into two tones:
+ *
+ * 1209 Hz 1336 Hz 1477 Hz 1633 Hz
+ * 697 Hz 1 (ev. 1) 2 (ev. 2) 3 (ev. 3) A (ev. 12)
+ * 770 Hz 4 (ev. 4) 5 (ev. 5) 6 (ev. 6) B (ev. 13)
+ * 852 Hz 7 (ev. 7) 8 (ev. 8) 9 (ev. 9) C (ev. 14)
+ * 941 Hz * (ev. 10) 0 (ev. 0) # (ev. 11) D (ev. 15)
+ *
+ * The two tones are added to form the DTMF signal.
+ *
+ */
+
+#include "dtmf_tonegen.h"
+
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+#ifdef NETEQ_ATEVENT_DECODE
+/* Must compile NetEQ with DTMF support to enable the functionality */
+
+/*******************/
+/* Constant tables */
+/*******************/
+
+/*
+ * All tables corresponding to the oscillator model are organized so that
+ * the coefficients for a specific frequency is found in the same position
+ * in every table. The positions for the tones follow this layout:
+ *
+ * dummyVector[8] =
+ * {
+ * 697 Hz, 770 Hz, 852 Hz, 941 Hz,
+ * 1209 Hz, 1336 Hz, 1477 Hz, 1633 Hz
+ * };
+ */
+
+/*
+ * Tables for the constant a = 2*cos(w) = 2*cos(2*pi*f/fs)
+ * in the oscillator model, for 8, 16, 32 and 48 kHz sample rate.
+ * Table values in Q14.
+ */
+
+const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl8Khz[8] =
+{
+ 27980, 26956, 25701, 24219,
+ 19073, 16325, 13085, 9315
+};
+
+#ifdef NETEQ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl16Khz[8]=
+{
+ 31548, 31281, 30951, 30556,
+ 29144, 28361, 27409, 26258
+};
+#endif
+
+#ifdef NETEQ_32KHZ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl32Khz[8]=
+{
+ 32462, 32394, 32311, 32210,
+ 31849, 31647, 31400, 31098
+};
+#endif
+
+#ifdef NETEQ_48KHZ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl48Khz[8]=
+{
+ 32632, 32602, 32564, 32520,
+ 32359, 32268, 32157, 32022
+};
+#endif
+
+/*
+ * Initialization values y[-1] = sin(w) = sin(2*pi*f/fs), for 8, 16, 32 and 48 kHz sample rate.
+ * Table values in Q14.
+ */
+
+const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab8Khz[8] =
+{
+ 8528, 9315, 10163, 11036,
+ 13323, 14206,15021, 15708
+};
+
+#ifdef NETEQ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab16Khz[8]=
+{
+ 4429, 4879, 5380, 5918,
+ 7490, 8207, 8979, 9801
+};
+#endif
+
+#ifdef NETEQ_32KHZ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab32Khz[8]=
+{
+ 2235, 2468, 2728, 3010,
+ 3853, 4249, 4685, 5164
+};
+#endif
+
+#ifdef NETEQ_48KHZ_WIDEBAND
+const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab48Khz[8]=
+{
+ 1493, 1649, 1823, 2013,
+ 2582, 2851, 3148, 3476
+};
+#endif
+
+/* Volume in dBm0 from 0 to -63, where 0 is the first table entry.
+ Everything below -36 is discarded, wherefore the table stops at -36.
+ Table entries are in Q14.
+ */
+
+const WebRtc_Word16 WebRtcNetEQ_dtfm_dBm0[37] = { 16141, 14386, 12821, 11427, 10184, 9077, 8090,
+ 7210, 6426, 5727, 5104, 4549, 4054, 3614,
+ 3221, 2870, 2558, 2280, 2032, 1811, 1614,
+ 1439, 1282, 1143, 1018, 908, 809, 721, 643,
+ 573, 510, 455, 405, 361, 322, 287, 256 };
+
+/****************************************************************************
+ * WebRtcNetEQ_DTMFGenerate(...)
+ *
+ * Generate 10 ms DTMF signal according to input parameters.
+ *
+ * Input:
+ * - DTMFdecInst : DTMF instance
+ * - value : DTMF event number (0-15)
+ * - volume : Volume of generated signal (0-36)
+ * Volume is given in negative dBm0, i.e., volume == 0
+ * means 0 dBm0 while volume == 36 mean -36 dBm0.
+ * - sampFreq : Sample rate in Hz
+ *
+ * Output:
+ * - signal : Pointer to vector where DTMF signal is stored;
+ * Vector must be at least sampFreq/100 samples long.
+ * - DTMFdecInst : Updated DTMF instance
+ *
+ * Return value : >0 - Number of samples written to signal
+ * : <0 - error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, WebRtc_Word16 value,
+ WebRtc_Word16 volume, WebRtc_Word16 *signal,
+ WebRtc_UWord16 sampFreq, WebRtc_Word16 extFrameLen)
+{
+ const WebRtc_Word16 *aTbl; /* pointer to a-coefficient table */
+ const WebRtc_Word16 *yInitTable; /* pointer to initialization value table */
+ WebRtc_Word16 a1 = 0; /* a-coefficient for first tone (low tone) */
+ WebRtc_Word16 a2 = 0; /* a-coefficient for second tone (high tone) */
+ int i;
+ int frameLen; /* number of samples to generate */
+ int lowIndex;
+ int highIndex;
+ WebRtc_Word32 tempVal;
+ WebRtc_Word16 tempValLow;
+ WebRtc_Word16 tempValHigh;
+
+ /* Sanity check for volume */
+ if ((volume < 0) || (volume > 36))
+ {
+ return DTMF_DEC_PARAMETER_ERROR;
+ }
+
+ /* Sanity check for extFrameLen */
+ if (extFrameLen < -1)
+ {
+ return DTMF_DEC_PARAMETER_ERROR;
+ }
+
+ /* Select oscillator coefficient tables based on sample rate */
+ if (sampFreq == 8000)
+ {
+ aTbl = WebRtcNetEQ_dtfm_aTbl8Khz;
+ yInitTable = WebRtcNetEQ_dtfm_yInitTab8Khz;
+ frameLen = 80;
+#ifdef NETEQ_WIDEBAND
+ }
+ else if (sampFreq == 16000)
+ {
+ aTbl = WebRtcNetEQ_dtfm_aTbl16Khz;
+ yInitTable = WebRtcNetEQ_dtfm_yInitTab16Khz;
+ frameLen = 160;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ }
+ else if (sampFreq == 32000)
+ {
+ aTbl = WebRtcNetEQ_dtfm_aTbl32Khz;
+ yInitTable = WebRtcNetEQ_dtfm_yInitTab32Khz;
+ frameLen = 320;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ }
+ else if (sampFreq == 48000)
+ {
+ aTbl = WebRtcNetEQ_dtfm_aTbl48Khz;
+ yInitTable = WebRtcNetEQ_dtfm_yInitTab48Khz;
+ frameLen = 480;
+#endif
+ }
+ else
+ {
+ /* unsupported sample rate */
+ return DTMF_GEN_UNKNOWN_SAMP_FREQ;
+ }
+
+ if (extFrameLen >= 0)
+ {
+ frameLen = extFrameLen;
+ }
+
+ /* select low frequency based on event value */
+ switch (value)
+ {
+ case 1:
+ case 2:
+ case 3:
+ case 12: /* first row on keypad */
+ {
+ lowIndex = 0; /* low frequency: 697 Hz */
+ break;
+ }
+ case 4:
+ case 5:
+ case 6:
+ case 13: /* second row on keypad */
+ {
+ lowIndex = 1; /* low frequency: 770 Hz */
+ break;
+ }
+ case 7:
+ case 8:
+ case 9:
+ case 14: /* third row on keypad */
+ {
+ lowIndex = 2; /* low frequency: 852 Hz */
+ break;
+ }
+ case 0:
+ case 10:
+ case 11:
+ case 15: /* fourth row on keypad */
+ {
+ lowIndex = 3; /* low frequency: 941 Hz */
+ break;
+ }
+ default:
+ {
+ return DTMF_DEC_PARAMETER_ERROR;
+ }
+ } /* end switch */
+
+ /* select high frequency based on event value */
+ switch (value)
+ {
+ case 1:
+ case 4:
+ case 7:
+ case 10: /* first column on keypad */
+ {
+ highIndex = 4; /* high frequency: 1209 Hz */
+ break;
+ }
+ case 2:
+ case 5:
+ case 8:
+ case 0: /* second column on keypad */
+ {
+ highIndex = 5;/* high frequency: 1336 Hz */
+ break;
+ }
+ case 3:
+ case 6:
+ case 9:
+ case 11: /* third column on keypad */
+ {
+ highIndex = 6;/* high frequency: 1477 Hz */
+ break;
+ }
+ case 12:
+ case 13:
+ case 14:
+ case 15: /* fourth column on keypad (special) */
+ {
+ highIndex = 7;/* high frequency: 1633 Hz */
+ break;
+ }
+ default:
+ {
+ return DTMF_DEC_PARAMETER_ERROR;
+ }
+ } /* end switch */
+
+ /* select coefficients based on results from switches above */
+ a1 = aTbl[lowIndex]; /* coefficient for first (low) tone */
+ a2 = aTbl[highIndex]; /* coefficient for second (high) tone */
+
+ if (DTMFdecInst->reinit)
+ {
+ /* set initial values for the recursive model */
+ DTMFdecInst->oldOutputLow[0] = yInitTable[lowIndex];
+ DTMFdecInst->oldOutputLow[1] = 0;
+ DTMFdecInst->oldOutputHigh[0] = yInitTable[highIndex];
+ DTMFdecInst->oldOutputHigh[1] = 0;
+
+ /* reset reinit flag */
+ DTMFdecInst->reinit = 0;
+ }
+
+ /* generate signal sample by sample */
+ for (i = 0; i < frameLen; i++)
+ {
+
+ /* Use rescursion formula y[n] = a*y[n-1] - y[n-2] */
+ tempValLow
+ = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a1, DTMFdecInst->oldOutputLow[1])
+ + 8192) >> 14) - DTMFdecInst->oldOutputLow[0]);
+ tempValHigh
+ = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a2, DTMFdecInst->oldOutputHigh[1])
+ + 8192) >> 14) - DTMFdecInst->oldOutputHigh[0]);
+
+ /* Update recursion memory */
+ DTMFdecInst->oldOutputLow[0] = DTMFdecInst->oldOutputLow[1];
+ DTMFdecInst->oldOutputLow[1] = tempValLow;
+ DTMFdecInst->oldOutputHigh[0] = DTMFdecInst->oldOutputHigh[1];
+ DTMFdecInst->oldOutputHigh[1] = tempValHigh;
+
+ /* scale high tone with 32768 (15 left shifts)
+ and low tone with 23171 (3dB lower than high tone) */
+ tempVal = WEBRTC_SPL_MUL_16_16(DTMF_AMP_LOW, tempValLow)
+ + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tempValHigh, 15);
+
+ /* Norm the signal to Q14 (with proper rounding) */
+ tempVal = (tempVal + 16384) >> 15;
+
+ /* Scale the signal to correct dbM0 value */
+ signal[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(tempVal, WebRtcNetEQ_dtfm_dBm0[volume])
+ + 8192), 14); /* volume value is in Q14; use proper rounding */
+ }
+
+ return frameLen;
+
+}
+
+#endif /* NETEQ_ATEVENT_DECODE */
+
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h
new file mode 100644
index 0000000..add6eb1
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the DTMF tone generator function.
+ */
+
+#ifndef DTMF_TONEGEN_H
+#define DTMF_TONEGEN_H
+
+#include "typedefs.h"
+
+#include "neteq_defines.h"
+
+#ifdef NETEQ_ATEVENT_DECODE
+/* Must compile NetEQ with DTMF support to enable the functionality */
+
+#define DTMF_AMP_LOW 23171 /* 3 dB lower than the high frequency */
+
+/* The DTMF generator struct (part of DSP main struct DSPInst_t) */
+typedef struct dtmf_tone_inst_t_
+{
+
+ WebRtc_Word16 reinit; /* non-zero if the oscillator model should
+ be reinitialized for next event */
+ WebRtc_Word16 oldOutputLow[2]; /* oscillator recursion history (low tone) */
+ WebRtc_Word16 oldOutputHigh[2]; /* oscillator recursion history (high tone) */
+
+ int lastDtmfSample; /* index to the first non-DTMF sample in the
+ speech history, if non-negative */
+}dtmf_tone_inst_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_DTMFGenerate(...)
+ *
+ * Generate 10 ms DTMF signal according to input parameters.
+ *
+ * Input:
+ * - DTMFdecInst : DTMF instance
+ * - value : DTMF event number (0-15)
+ * - volume : Volume of generated signal (0-36)
+ * Volume is given in negative dBm0, i.e., volume == 0
+ * means 0 dBm0 while volume == 36 mean -36 dBm0.
+ * - sampFreq : Sample rate in Hz
+ *
+ * Output:
+ * - signal : Pointer to vector where DTMF signal is stored;
+ * Vector must be at least sampFreq/100 samples long.
+ * - DTMFdecInst : Updated DTMF instance
+ *
+ * Return value : >0 - Number of samples written to signal
+ * : <0 - Error
+ */
+
+WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst,
+ WebRtc_Word16 value,
+ WebRtc_Word16 volume,
+ WebRtc_Word16 *signal,
+ WebRtc_UWord16 sampFreq,
+ WebRtc_Word16 frameLen
+);
+
+#endif /* NETEQ_ATEVENT_DECODE */
+
+#endif /* DTMF_TONEGEN_H */
+
diff --git a/trunk/src/modules/audio_coding/neteq/expand.c b/trunk/src/modules/audio_coding/neteq/expand.c
new file mode 100644
index 0000000..3db7a2a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/expand.c
@@ -0,0 +1,1216 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is the function to expand from the speech history, to produce concealment data or
+ * increasing delay.
+ */
+
+#include "dsp.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+
+#define CHECK_NO_OF_CORRMAX 3
+#define DISTLEN 20
+#define LPCANALASYSLEN 160
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ (First part of first expand)
+ WebRtc_Word16 pw16_bestCorrIndex 3 0 2
+ WebRtc_Word16 pw16_bestCorr 3 3 5
+ WebRtc_Word16 pw16_bestDistIndex 3 6 8
+ WebRtc_Word16 pw16_bestDist 3 9 11
+ WebRtc_Word16 pw16_corrVec 102*fs/8000 12 11+102*fs/8000
+ func WebRtcNetEQ_Correlator 232 12+102*fs/8000 243+102*fs/8000
+
+ (Second part of first expand)
+ WebRtc_Word32 pw32_corr2 99*fs/8000+1 0 99*fs/8000
+ WebRtc_Word32 pw32_autoCorr 2*7 0 13
+ WebRtc_Word16 pw16_rc 6 14 19
+
+ Signal combination:
+ WebRtc_Word16 pw16_randVec 30+120*fs/8000 0 29+120*fs/8000
+ WebRtc_Word16 pw16_scaledRandVec 125*fs/8000 30+120*fs/8000 29+245*fs/8000
+ WebRtc_Word16 pw16_unvoicedVecSpace 10+125*fs/8000 30+245*fs/8000 39+370*fs/8000
+
+ Total: 40+370*fs/8000 (size depends on UNVOICED_LPC_ORDER and BGN_LPC_ORDER)
+ */
+
+#if ((BGN_LPC_ORDER > 10) || (UNVOICED_LPC_ORDER > 10)) && (defined SCRATCH)
+#error BGN_LPC_ORDER and/or BGN_LPC_ORDER are too large for current scratch memory allocation
+#endif
+
+#define SCRATCH_PW16_BEST_CORR_INDEX 0
+#define SCRATCH_PW16_BEST_CORR 3
+#define SCRATCH_PW16_BEST_DIST_INDEX 6
+#define SCRATCH_PW16_BEST_DIST 9
+#define SCRATCH_PW16_CORR_VEC 12
+#define SCRATCH_PW16_CORR2 0
+#define SCRATCH_PW32_AUTO_CORR 0
+#define SCRATCH_PW16_RC 14
+#define SCRATCH_PW16_RAND_VEC 0
+
+#if (defined(NETEQ_48KHZ_WIDEBAND))
+#define SCRATCH_NETEQDSP_CORRELATOR 624
+#define SCRATCH_PW16_SCALED_RAND_VEC 750
+#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1500
+#elif (defined(NETEQ_32KHZ_WIDEBAND))
+#define SCRATCH_NETEQDSP_CORRELATOR 420
+#define SCRATCH_PW16_SCALED_RAND_VEC 510
+#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1010
+#elif (defined(NETEQ_WIDEBAND))
+#define SCRATCH_NETEQDSP_CORRELATOR 216
+#define SCRATCH_PW16_SCALED_RAND_VEC 270
+#define SCRATCH_PW16_UNVOICED_VEC_SPACE 520
+#else /* NB */
+#define SCRATCH_NETEQDSP_CORRELATOR 114
+#define SCRATCH_PW16_SCALED_RAND_VEC 150
+#define SCRATCH_PW16_UNVOICED_VEC_SPACE 275
+#endif
+
+/****************************************************************************
+ * WebRtcNetEQ_Expand(...)
+ *
+ * This function produces one "chunk" of expansion data (PLC audio). The
+ * length of the produced audio depends on the speech history.
+ *
+ * Input:
+ * - inst : DSP instance
+ * - scratchPtr : Pointer to scratch vector
+ * - outdata : Pointer to a memory space where the output data
+ * should be stored
+ * - BGNonly : If non-zero, "expand" will only produce background noise.
+ * - pw16_len : Desired number of samples (only for BGN mode).
+ *
+ * Output:
+ * - inst : Updated instance
+ * - pw16_len : Number of samples that were output from NetEq
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_Expand(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly)
+{
+
+ WebRtc_Word16 fs_mult;
+ ExpandInst_t *ExpandState = &(inst->ExpandInst);
+ BGNInst_t *BGNState = &(inst->BGNInst);
+ int i;
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_randVec = pw16_scratchPtr + SCRATCH_PW16_RAND_VEC;
+ WebRtc_Word16 *pw16_scaledRandVec = pw16_scratchPtr + SCRATCH_PW16_SCALED_RAND_VEC;
+ WebRtc_Word16 *pw16_unvoicedVecSpace = pw16_scratchPtr + SCRATCH_PW16_UNVOICED_VEC_SPACE;
+#else
+ WebRtc_Word16 pw16_randVec[FSMULT * 120 + 30]; /* 150 for NB and 270 for WB */
+ WebRtc_Word16 pw16_scaledRandVec[FSMULT * 125]; /* 125 for NB and 250 for WB */
+ WebRtc_Word16 pw16_unvoicedVecSpace[BGN_LPC_ORDER + FSMULT * 125];
+#endif
+ /* 125 for NB and 250 for WB etc. Reuse pw16_outData[] for this vector */
+ WebRtc_Word16 *pw16_voicedVecStorage = pw16_outData;
+ WebRtc_Word16 *pw16_voicedVec = &pw16_voicedVecStorage[ExpandState->w16_overlap];
+ WebRtc_Word16 *pw16_unvoicedVec = pw16_unvoicedVecSpace + UNVOICED_LPC_ORDER;
+ WebRtc_Word16 *pw16_cngVec = pw16_unvoicedVecSpace + BGN_LPC_ORDER;
+ WebRtc_Word16 w16_expVecsLen, w16_lag = 0, w16_expVecPos;
+ WebRtc_Word16 w16_randLen;
+ WebRtc_Word16 w16_vfractionChange; /* in Q14 */
+ WebRtc_Word16 w16_winMute = 0, w16_winMuteInc = 0, w16_winUnMute = 0, w16_winUnMuteInc = 0;
+ WebRtc_Word32 w32_tmp;
+ WebRtc_Word16 w16_tmp, w16_tmp2;
+ WebRtc_Word16 stability;
+ enum BGNMode bgnMode = inst->BGNInst.bgnMode;
+
+ /* Pre-calculate common multiplications with fs_mult */
+ WebRtc_Word16 fsMult4;
+ WebRtc_Word16 fsMult20;
+ WebRtc_Word16 fsMult120;
+ WebRtc_Word16 fsMultDistLen;
+ WebRtc_Word16 fsMultLPCAnalasysLen;
+
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo *msInfo = inst->msInfo;
+#endif
+
+ /* fs is WebRtc_UWord16 (to hold fs=48000) */
+ fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs); /* calculate fs/8000 */
+
+ /* Pre-calculate common multiplications with fs_mult */
+ fsMult4 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 4);
+ fsMult20 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 20);
+ fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 120);
+ fsMultDistLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, DISTLEN);
+ fsMultLPCAnalasysLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, LPCANALASYSLEN);
+
+ /*
+ * Perform all the initial setup if it's the first expansion.
+ * If background noise (BGN) only, this setup is not needed.
+ */
+ if (ExpandState->w16_consecExp == 0 && !BGNonly)
+ {
+ /* Setup more variables */
+#ifdef SCRATCH
+ WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr
+ + SCRATCH_PW32_AUTO_CORR);
+ WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
+ WebRtc_Word16 *pw16_bestCorrIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR_INDEX;
+ WebRtc_Word16 *pw16_bestCorr = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR;
+ WebRtc_Word16 *pw16_bestDistIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST_INDEX;
+ WebRtc_Word16 *pw16_bestDist = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST;
+ WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_PW16_CORR_VEC;
+ WebRtc_Word32 *pw32_corr2 = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW16_CORR2);
+#else
+ WebRtc_Word32 pw32_autoCorr[UNVOICED_LPC_ORDER+1];
+ WebRtc_Word16 pw16_rc[UNVOICED_LPC_ORDER];
+ WebRtc_Word16 pw16_corrVec[FSMULT*102]; /* 102 for NB */
+ WebRtc_Word16 pw16_bestCorrIndex[CHECK_NO_OF_CORRMAX];
+ WebRtc_Word16 pw16_bestCorr[CHECK_NO_OF_CORRMAX];
+ WebRtc_Word16 pw16_bestDistIndex[CHECK_NO_OF_CORRMAX];
+ WebRtc_Word16 pw16_bestDist[CHECK_NO_OF_CORRMAX];
+ WebRtc_Word32 pw32_corr2[(99*FSMULT)+1];
+#endif
+ WebRtc_Word32 pw32_bestDist[CHECK_NO_OF_CORRMAX];
+ WebRtc_Word16 w16_ind = 0;
+ WebRtc_Word16 w16_corrVecLen;
+ WebRtc_Word16 w16_corrScale;
+ WebRtc_Word16 w16_distScale;
+ WebRtc_Word16 w16_indMin, w16_indMax;
+ WebRtc_Word16 w16_len;
+ WebRtc_Word32 w32_en1, w32_en2, w32_cc;
+ WebRtc_Word16 w16_en1Scale, w16_en2Scale;
+ WebRtc_Word16 w16_en1, w16_en2;
+ WebRtc_Word32 w32_en1_mul_en2;
+ WebRtc_Word16 w16_sqrt_en1en2;
+ WebRtc_Word16 w16_ccShiftL;
+ WebRtc_Word16 w16_bestcorr; /* Correlation in Q14 */
+ WebRtc_Word16 *pw16_vec1, *pw16_vec2;
+ WebRtc_Word16 w16_factor;
+ WebRtc_Word16 w16_DistLag, w16_CorrLag, w16_diffLag;
+ WebRtc_Word16 w16_energyLen;
+ WebRtc_Word16 w16_slope;
+ WebRtc_Word16 w16_startInd;
+ WebRtc_Word16 w16_noOfcorr2;
+ WebRtc_Word16 w16_scale;
+
+ /* Initialize some variables */
+ ExpandState->w16_lagsDirection = 1;
+ ExpandState->w16_lagsPosition = -1;
+ ExpandState->w16_expandMuteFactor = 16384; /* Start from 1.0 (Q14) */
+ BGNState->w16_mutefactor = 0; /* Start with 0 gain for BGN (value in Q14) */
+ inst->w16_seedInc = 1;
+
+#ifdef NETEQ_STEREO
+ /* Sanity for msInfo */
+ if (msInfo == NULL)
+ {
+ /* this should not happen here */
+ return MASTER_SLAVE_ERROR;
+ }
+
+ /*
+ * Do not calculate correlations for slave instance(s)
+ * unless lag info from master is corrupt
+ */
+ if ((msInfo->msMode != NETEQ_SLAVE)
+ || ((msInfo->distLag <= 0) || (msInfo->corrLag <= 0)))
+ {
+#endif
+ /* Calculate correlation vector in downsampled domain (4 kHz sample rate) */
+ w16_corrVecLen = WebRtcNetEQ_Correlator(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQDSP_CORRELATOR,
+#endif
+ inst->pw16_speechHistory, inst->w16_speechHistoryLen, pw16_corrVec,
+ &w16_corrScale);
+
+ /* Find peaks in correlation vector using parabolic fit method */
+ WebRtcNetEQ_PeakDetection(pw16_corrVec, w16_corrVecLen, CHECK_NO_OF_CORRMAX, fs_mult,
+ pw16_bestCorrIndex, pw16_bestCorr);
+
+ /*
+ * Adjust peak locations; cross-correlation lags start at 2.5 ms
+ * (20*fs_mult samples)
+ */
+ pw16_bestCorrIndex[0] += fsMult20;
+ pw16_bestCorrIndex[1] += fsMult20;
+ pw16_bestCorrIndex[2] += fsMult20;
+
+ /* Calculate distortion around the 3 (CHECK_NO_OF_CORRMAX) best lags */
+ w16_distScale = 0;
+ for (i = 0; i < CHECK_NO_OF_CORRMAX; i++)
+ {
+ w16_tmp = fsMult20;
+ w16_tmp2 = pw16_bestCorrIndex[i] - fsMult4;
+ w16_indMin = WEBRTC_SPL_MAX(w16_tmp, w16_tmp2);
+ w16_tmp = fsMult120 - 1;
+ w16_tmp2 = pw16_bestCorrIndex[i] + fsMult4;
+ w16_indMax = WEBRTC_SPL_MIN(w16_tmp, w16_tmp2);
+
+ pw16_bestDistIndex[i] = WebRtcNetEQ_MinDistortion(
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultDistLen]),
+ w16_indMin, w16_indMax, fsMultDistLen, &pw32_bestDist[i]);
+
+ w16_distScale
+ = WEBRTC_SPL_MAX(16 - WebRtcSpl_NormW32(pw32_bestDist[i]), w16_distScale);
+
+ }
+
+ /* Shift the distortion values to fit in WebRtc_Word16 */
+ WebRtcSpl_VectorBitShiftW32ToW16(pw16_bestDist, CHECK_NO_OF_CORRMAX, pw32_bestDist,
+ w16_distScale);
+
+ /*
+ * Find index of maximum criteria, where crit[i] = bestCorr[i])/(bestDist[i])
+ * Do this by a cross multiplication.
+ */
+
+ w32_en1 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0],pw16_bestDist[1]);
+ w32_en2 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[0]);
+ if (w32_en1 >= w32_en2)
+ {
+ /* 0 wins over 1 */
+ w32_en1
+ = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0], pw16_bestDist[2]);
+ w32_en2
+ = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2], pw16_bestDist[0]);
+ if (w32_en1 >= w32_en2)
+ {
+ /* 0 wins over 2 */
+ w16_ind = 0;
+ }
+ else
+ {
+ /* 2 wins over 0 */
+ w16_ind = 2;
+ }
+ }
+ else
+ {
+ /* 1 wins over 0 */
+ w32_en1
+ = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[2]);
+ w32_en2
+ = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2],pw16_bestDist[1]);
+ if ((WebRtc_Word32) w32_en1 >= (WebRtc_Word32) w32_en2)
+ {
+ /* 1 wins over 2 */
+ w16_ind = 1;
+ }
+ else
+ {
+ /* 2 wins over 1 */
+ w16_ind = 2;
+ }
+ }
+
+#ifdef NETEQ_STEREO
+ }
+
+ /* Store DistLag and CorrLag of the position with highest criteria */
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)
+ || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->distLag <= 0 || msInfo->corrLag
+ <= 0)))
+ {
+ /* lags not provided externally */
+ w16_DistLag = pw16_bestDistIndex[w16_ind];
+ w16_CorrLag = pw16_bestCorrIndex[w16_ind];
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ msInfo->distLag = w16_DistLag;
+ msInfo->corrLag = w16_CorrLag;
+ }
+ }
+ else if (msInfo->msMode == NETEQ_SLAVE)
+ {
+ /* lags provided externally (from master) */
+ w16_DistLag = msInfo->distLag;
+ w16_CorrLag = msInfo->corrLag;
+
+ /* sanity for lag values */
+ if ((w16_DistLag <= 0) || (w16_CorrLag <= 0))
+ {
+ return MASTER_SLAVE_ERROR;
+ }
+ }
+ else
+ {
+ /* Invalid mode */
+ return MASTER_SLAVE_ERROR;
+ }
+#else /* not NETEQ_STEREO */
+ w16_DistLag = pw16_bestDistIndex[w16_ind];
+ w16_CorrLag = pw16_bestCorrIndex[w16_ind];
+#endif
+
+ ExpandState->w16_maxLag = WEBRTC_SPL_MAX(w16_DistLag, w16_CorrLag);
+
+ /* Calculate the exact best correlation (in the range within CorrLag-DistLag) */
+ w16_len = w16_DistLag + 10;
+ w16_len = WEBRTC_SPL_MIN(w16_len, fsMult120);
+ w16_len = WEBRTC_SPL_MAX(w16_len, 60 * fs_mult);
+
+ w16_startInd = WEBRTC_SPL_MIN(w16_DistLag, w16_CorrLag);
+ w16_noOfcorr2 = WEBRTC_SPL_ABS_W16((w16_DistLag-w16_CorrLag)) + 1;
+ /* w16_noOfcorr2 maximum value is 99*fs_mult + 1 */
+
+ /* Calculate suitable scaling */
+ w16_tmp
+ = WebRtcSpl_MaxAbsValueW16(
+ &inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd
+ - w16_noOfcorr2],
+ (WebRtc_Word16) (w16_len + w16_startInd + w16_noOfcorr2 - 1));
+ w16_corrScale = ((31 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_tmp, w16_tmp)))
+ + (31 - WebRtcSpl_NormW32(w16_len))) - 31;
+ w16_corrScale = WEBRTC_SPL_MAX(0, w16_corrScale);
+
+ /*
+ * Perform the correlation, store in pw32_corr2
+ */
+
+ WebRtcNetEQ_CrossCorr(pw32_corr2,
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]),
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd]),
+ w16_len, w16_noOfcorr2, w16_corrScale, -1);
+
+ /* Find maximizing index */
+ w16_ind = WebRtcSpl_MaxIndexW32(pw32_corr2, w16_noOfcorr2);
+ w32_cc = pw32_corr2[w16_ind]; /* this is maximum correlation */
+ w16_ind = w16_ind + w16_startInd; /* correct index for start offset */
+
+ /* Calculate energies */
+ w32_en1 = WebRtcNetEQ_DotW16W16(
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]),
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]), w16_len,
+ w16_corrScale);
+ w32_en2 = WebRtcNetEQ_DotW16W16(
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]),
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]),
+ w16_len, w16_corrScale);
+
+ /* Calculate the correlation value w16_bestcorr */
+ if ((w32_en1 > 0) && (w32_en2 > 0))
+ {
+ w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
+ w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
+ w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
+ w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
+ /* Make sure total scaling is even (to simplify scale factor after sqrt) */
+ if ((w16_en1Scale + w16_en2Scale) & 1)
+ {
+ /* if sum is odd */
+ w16_en1Scale += 1;
+ }
+ w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+ w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+ w32_en1_mul_en2 = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
+ w16_sqrt_en1en2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_en1_mul_en2);
+
+ /* Calculate cc/sqrt(en1*en2) in Q14 */
+ w16_ccShiftL = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
+ w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_ccShiftL);
+ w16_bestcorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrt_en1en2);
+ w16_bestcorr = WEBRTC_SPL_MIN(16384, w16_bestcorr); /* set maximum to 1.0 */
+
+ }
+ else
+ {
+ /* if either en1 or en2 is zero */
+ w16_bestcorr = 0;
+ }
+
+ /*
+ * Extract the two vectors, pw16_expVecs[0][] and pw16_expVecs[1][],
+ * from the SpeechHistory[]
+ */
+ w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap;
+ pw16_vec1 = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_expVecsLen]);
+ pw16_vec2 = pw16_vec1 - w16_DistLag;
+ /* Normalize the second vector to the same energy as the first */
+ w32_en1 = WebRtcNetEQ_DotW16W16(pw16_vec1, pw16_vec1, w16_expVecsLen, w16_corrScale);
+ w32_en2 = WebRtcNetEQ_DotW16W16(pw16_vec2, pw16_vec2, w16_expVecsLen, w16_corrScale);
+
+ /*
+ * Confirm that energy factor sqrt(w32_en1/w32_en2) is within difference 0.5 - 2.0
+ * w32_en1/w32_en2 within 0.25 - 4
+ */
+ if (((w32_en1 >> 2) < w32_en2) && ((w32_en1) > (w32_en2 >> 2)))
+ {
+
+ /* Energy constraint fulfilled => use both vectors and scale them accordingly */
+ w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
+ w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
+ w16_en1Scale = w16_en2Scale - 13;
+
+ /* calculate w32_en1/w32_en2 in Q13 */
+ w32_en1_mul_en2 = WebRtcSpl_DivW32W16(
+ WEBRTC_SPL_SHIFT_W32(w32_en1, -w16_en1Scale),
+ (WebRtc_Word16) (WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale)));
+
+ /* calculate factor in Q13 (sqrt of en1/en2 in Q26) */
+ w16_factor = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
+ WEBRTC_SPL_LSHIFT_W32(w32_en1_mul_en2, 13));
+
+ /* Copy the two vectors and give them the same energy */
+
+ WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen);
+ WebRtcSpl_AffineTransformVector(ExpandState->pw16_expVecs[1], pw16_vec2,
+ w16_factor, 4096, 13, w16_expVecsLen);
+
+ }
+ else
+ {
+ /* Energy change constraint not fulfilled => only use last vector */
+
+ WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen);
+ WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[1], ExpandState->pw16_expVecs[0],
+ w16_expVecsLen);
+
+ /* Set the w16_factor since it is used by muting slope */
+ if (((w32_en1 >> 2) < w32_en2) || (w32_en2 == 0))
+ {
+ w16_factor = 4096; /* 0.5 in Q13*/
+ }
+ else
+ {
+ w16_factor = 16384; /* 2.0 in Q13*/
+ }
+ }
+
+ /* Set the 3 lag values */
+ w16_diffLag = w16_DistLag - w16_CorrLag;
+ if (w16_diffLag == 0)
+ {
+ /* DistLag and CorrLag are equal */
+ ExpandState->w16_lags[0] = w16_DistLag;
+ ExpandState->w16_lags[1] = w16_DistLag;
+ ExpandState->w16_lags[2] = w16_DistLag;
+ }
+ else
+ {
+ /* DistLag and CorrLag are not equal; use different combinations of the two */
+ ExpandState->w16_lags[0] = w16_DistLag; /* DistLag only */
+ ExpandState->w16_lags[1] = ((w16_DistLag + w16_CorrLag) >> 1); /* 50/50 */
+ /* Third lag, move one half-step towards CorrLag (in both cases) */
+ if (w16_diffLag > 0)
+ {
+ ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag - 1) >> 1;
+ }
+ else
+ {
+ ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag + 1) >> 1;
+ }
+ }
+
+ /*************************************************
+ * Calculate the LPC and the gain of the filters *
+ *************************************************/
+
+ /* Calculate scale value needed for autocorrelation */
+ w16_tmp = WebRtcSpl_MaxAbsValueW16(
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultLPCAnalasysLen]),
+ fsMultLPCAnalasysLen);
+
+ w16_tmp = 16 - WebRtcSpl_NormW32(w16_tmp);
+ w16_tmp = WEBRTC_SPL_MIN(w16_tmp,0);
+ w16_tmp = (w16_tmp << 1) + 7;
+ w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0);
+
+ /* set w16_ind to simplify the following expressions */
+ w16_ind = inst->w16_speechHistoryLen - fsMultLPCAnalasysLen - UNVOICED_LPC_ORDER;
+
+ /* store first UNVOICED_LPC_ORDER samples in pw16_rc */
+
+ WEBRTC_SPL_MEMCPY_W16(pw16_rc, &inst->pw16_speechHistory[w16_ind], UNVOICED_LPC_ORDER);
+
+ /* set first samples to zero */
+ WebRtcSpl_MemSetW16(&inst->pw16_speechHistory[w16_ind], 0, UNVOICED_LPC_ORDER);
+
+ /* Calculate UNVOICED_LPC_ORDER+1 lags of the ACF */
+
+ WebRtcNetEQ_CrossCorr(
+ pw32_autoCorr, &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]),
+ &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]), fsMultLPCAnalasysLen,
+ UNVOICED_LPC_ORDER + 1, w16_tmp, -1);
+
+ /* Recover the stored samples from pw16_rc */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[w16_ind], pw16_rc, UNVOICED_LPC_ORDER);
+
+ if (pw32_autoCorr[0] > 0)
+ { /* check that variance is positive */
+
+ /* estimate AR filter parameters using Levinson-Durbin algorithm
+ (UNVOICED_LPC_ORDER+1 filter coefficients) */
+ stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, ExpandState->pw16_arFilter,
+ pw16_rc, UNVOICED_LPC_ORDER);
+
+ /* Only update BGN if filter is stable */
+ if (stability != 1)
+ {
+ /* Set first coefficient to 4096 (1.0 in Q12)*/
+ ExpandState->pw16_arFilter[0] = 4096;
+ /* Set remaining UNVOICED_LPC_ORDER coefficients to zero */
+ WebRtcSpl_MemSetW16(ExpandState->pw16_arFilter + 1, 0, UNVOICED_LPC_ORDER);
+ }
+
+ }
+
+ if (w16_DistLag < 40)
+ {
+ w16_energyLen = 2 * w16_DistLag;
+ }
+ else
+ {
+ w16_energyLen = w16_DistLag;
+ }
+ w16_randLen = w16_energyLen + 30; /* Startup part */
+
+ /* Extract a noise segment */
+ if (w16_randLen <= RANDVEC_NO_OF_SAMPLES)
+ {
+ WEBRTC_SPL_MEMCPY_W16(pw16_randVec,
+ (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl, w16_randLen);
+ }
+ else
+ { /* only applies to SWB where length could be larger than 256 */
+#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */
+ WEBRTC_SPL_MEMCPY_W16(pw16_randVec, (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl,
+ RANDVEC_NO_OF_SAMPLES);
+ inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
+ assert(w16_randLen <= FSMULT * 120 + 30);
+ WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
+ (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
+#else
+ assert(0);
+#endif
+ }
+
+ /* Set up state vector and calculate scale factor for unvoiced filtering */
+
+ WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState,
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - UNVOICED_LPC_ORDER]),
+ UNVOICED_LPC_ORDER);
+ WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER,
+ &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128 - UNVOICED_LPC_ORDER]),
+ UNVOICED_LPC_ORDER);
+ WebRtcSpl_FilterMAFastQ12(&inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128],
+ pw16_unvoicedVec, ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, 128);
+ if (WebRtcSpl_MaxAbsValueW16(pw16_unvoicedVec, 128) > 4000)
+ {
+ w16_scale = 4;
+ }
+ else
+ {
+ w16_scale = 0;
+ }
+ w32_tmp = WebRtcNetEQ_DotW16W16(pw16_unvoicedVec, pw16_unvoicedVec, 128, w16_scale);
+
+ /* Normalize w32_tmp to 28 or 29 bits to preserve sqrt() accuracy */
+ w16_tmp = WebRtcSpl_NormW32(w32_tmp) - 3;
+ w16_tmp += ((w16_tmp & 0x1) ^ 0x1); /* Make sure we do an odd number of shifts since we
+ from earlier have 7 shifts from dividing with 128.*/
+ w32_tmp = WEBRTC_SPL_SHIFT_W32(w32_tmp, w16_tmp);
+ w32_tmp = WebRtcSpl_SqrtFloor(w32_tmp);
+ ExpandState->w16_arGainScale = 13 + ((w16_tmp + 7 - w16_scale) >> 1);
+ ExpandState->w16_arGain = (WebRtc_Word16) w32_tmp;
+
+ /********************************************************************
+ * Calculate vfraction from bestcorr *
+ * if (bestcorr>0.480665) *
+ * vfraction = ((bestcorr-0.4)/(1-0.4)).^2 *
+ * else vfraction = 0 *
+ * *
+ * approximation (coefficients in Q12): *
+ * if (x>0.480665) (y(x)<0.3) *
+ * y(x) = -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 *
+ * else y(x) = 0; *
+ ********************************************************************/
+
+ if (w16_bestcorr > 7875)
+ {
+ /* if x>0.480665 */
+ WebRtc_Word16 w16_x1, w16_x2, w16_x3;
+ w16_x1 = w16_bestcorr;
+ w32_tmp = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) w16_x1, w16_x1);
+ w16_x2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
+ w32_tmp = WEBRTC_SPL_MUL_16_16(w16_x1, w16_x2);
+ w16_x3 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
+ w32_tmp
+ = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) WebRtcNetEQ_kMixFractionFuncTbl[0], 14);
+ w32_tmp
+ += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[1], w16_x1);
+ w32_tmp
+ += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[2], w16_x2);
+ w32_tmp
+ += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[3], w16_x3);
+ ExpandState->w16_vFraction = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 12);
+ ExpandState->w16_vFraction = WEBRTC_SPL_MIN(ExpandState->w16_vFraction, 16384);
+ ExpandState->w16_vFraction = WEBRTC_SPL_MAX(ExpandState->w16_vFraction, 0);
+ }
+ else
+ {
+ ExpandState->w16_vFraction = 0;
+ }
+
+ /***********************************************************************
+ * Calculate muting slope, reuse value from earlier scaling of ExpVecs *
+ ***********************************************************************/
+ w16_slope = w16_factor;
+
+ if (w16_slope > 12288)
+ {
+ /* w16_slope > 1.5 ? */
+ /* Calculate (1-(1/slope))/w16_DistLag = (slope-1)/(w16_DistLag*slope) */
+ w32_tmp = w16_slope - 8192;
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 12); /* Value in Q25 (13+12=25) */
+ w16_tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(w16_DistLag,
+ w16_slope, 8); /* Value in Q5 (13-8=5) */
+ w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+ w16_tmp); /* Res in Q20 (25-5=20) */
+
+ if (w16_slope > 14746)
+ { /* w16_slope > 1.8 ? */
+ ExpandState->w16_muteSlope = (w16_tmp + 1) >> 1;
+ }
+ else
+ {
+ ExpandState->w16_muteSlope = (w16_tmp + 4) >> 3;
+ }
+ ExpandState->w16_onset = 1;
+ }
+ else if (ExpandState->w16_vFraction > 13107)
+ {
+ /* w16_vFraction > 0.8 ? */
+ if (w16_slope > 8028)
+ {
+ /* w16_vFraction > 0.98 ? */
+ ExpandState->w16_muteSlope = 0;
+ }
+ else
+ {
+ /* Calculate (1-slope)/w16_DistLag */
+ w32_tmp = 8192 - w16_slope;
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
+ ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+ w16_DistLag); /* Res in Q20 (20-0=20) */
+ }
+ ExpandState->w16_onset = 0;
+ }
+ else
+ {
+ /*
+ * Use the minimum of 0.005 (0.9 on 50 samples in NB and the slope)
+ * and ((1-slope)/w16_DistLag)
+ */
+ w32_tmp = 8192 - w16_slope;
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
+ w32_tmp = WEBRTC_SPL_MAX(w32_tmp, 0);
+ ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
+ w16_DistLag); /* Res in Q20 (20-0=20) */
+ w16_tmp = WebRtcNetEQ_k5243div[fs_mult]; /* 0.005/fs_mult = 5243/fs_mult */
+ ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(w16_tmp, ExpandState->w16_muteSlope);
+ ExpandState->w16_onset = 0;
+ }
+ }
+ else
+ {
+ /* This is not the first Expansion, parameters are already estimated. */
+
+ /* Extract a noise segment */
+ if (BGNonly) /* If we should produce nothing but background noise */
+ {
+ if (*pw16_len > 0)
+ {
+ /*
+ * Set length to input parameter length, but not more than length
+ * of pw16_randVec
+ */
+ w16_lag = WEBRTC_SPL_MIN(*pw16_len, FSMULT * 120 + 30);
+ }
+ else
+ {
+ /* set length to 15 ms */
+ w16_lag = fsMult120;
+ }
+ w16_randLen = w16_lag;
+ }
+ else
+ {
+ w16_randLen = ExpandState->w16_maxLag;
+ }
+
+ if (w16_randLen <= RANDVEC_NO_OF_SAMPLES)
+ {
+ inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
+ WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, w16_randLen,
+ inst->w16_seedInc);
+ }
+ else
+ { /* only applies to SWB where length could be larger than 256 */
+#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */
+ inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
+ WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, RANDVEC_NO_OF_SAMPLES,
+ inst->w16_seedInc);
+ inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
+ assert(w16_randLen <= FSMULT * 120 + 30);
+ WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
+ (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
+#else
+ assert(0);
+#endif
+ }
+ } /* end if(first expand or BGNonly) ... else ... */
+
+ if (!BGNonly) /* Voiced and unvoiced parts not used if generating BGN only */
+ {
+
+ /*************************************************
+ * Generate signal *
+ *************************************************/
+
+ /*
+ * Voiced part
+ */
+
+ /* Linearly mute the use_vfraction value from 1 to vfraction */
+ if (ExpandState->w16_consecExp == 0)
+ {
+ ExpandState->w16_currentVFraction = 16384; /* 1.0 in Q14 */
+ }
+
+ ExpandState->w16_lagsPosition = ExpandState->w16_lagsPosition
+ + ExpandState->w16_lagsDirection;
+
+ /* Change direction if needed */
+ if (ExpandState->w16_lagsPosition == 0)
+ {
+ ExpandState->w16_lagsDirection = 1;
+ }
+ if (ExpandState->w16_lagsPosition == 2)
+ {
+ ExpandState->w16_lagsDirection = -1;
+ }
+
+ /* Generate a weighted vector with the selected lag */
+ w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap;
+ w16_lag = ExpandState->w16_lags[ExpandState->w16_lagsPosition];
+ /* Copy lag+overlap data */
+ w16_expVecPos = w16_expVecsLen - w16_lag - ExpandState->w16_overlap;
+ w16_tmp = w16_lag + ExpandState->w16_overlap;
+ if (ExpandState->w16_lagsPosition == 0)
+ {
+ WEBRTC_SPL_MEMCPY_W16(pw16_voicedVecStorage,
+ &(ExpandState->pw16_expVecs[0][w16_expVecPos]), w16_tmp);
+ }
+ else if (ExpandState->w16_lagsPosition == 1)
+ {
+ WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 3,
+ &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 2, pw16_voicedVecStorage,
+ w16_tmp);
+
+ }
+ else if (ExpandState->w16_lagsPosition == 2)
+ {
+ WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 1,
+ &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 1, pw16_voicedVecStorage,
+ w16_tmp);
+ }
+
+ if (inst->fs == 8000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC;
+#ifdef NETEQ_WIDEBAND
+ }
+ else if (inst->fs == 16000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC;
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ }
+ else if (inst->fs == 32000)
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC;
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ }
+ else /* if (inst->fs==48000) */
+ {
+ /* Windowing in Q15 */
+ w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START;
+ w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC;
+ w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START;
+ w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC;
+#endif
+ }
+
+ /* Smooth the expanded if it has not been muted to or vfraction is larger than 0.5 */
+ if ((ExpandState->w16_expandMuteFactor > 819) && (ExpandState->w16_currentVFraction
+ > 8192))
+ {
+ for (i = 0; i < ExpandState->w16_overlap; i++)
+ {
+ /* Do overlap add between new vector and overlap */
+ ExpandState->pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL_16_16(ExpandState->pw16_overlapVec[i], w16_winMute) +
+ WEBRTC_SPL_MUL_16_16(
+ WEBRTC_SPL_MUL_16_16_RSFT(ExpandState->w16_expandMuteFactor,
+ pw16_voicedVecStorage[i], 14), w16_winUnMute) + 16384, 15);
+ w16_winMute += w16_winMuteInc;
+ w16_winUnMute += w16_winUnMuteInc;
+ }
+ }
+ else if (ExpandState->w16_expandMuteFactor == 0
+#ifdef NETEQ_STEREO
+ && msInfo->msMode == NETEQ_MONO /* only if mono mode is selected */
+#endif
+ )
+ {
+ /* if ExpandState->w16_expandMuteFactor = 0 => all is CNG component
+ set the output length to 15ms (for best CNG production) */
+ w16_tmp = fsMult120;
+ ExpandState->w16_maxLag = w16_tmp;
+ ExpandState->w16_lags[0] = w16_tmp;
+ ExpandState->w16_lags[1] = w16_tmp;
+ ExpandState->w16_lags[2] = w16_tmp;
+ }
+
+ /*
+ * Unvoiced part
+ */
+
+ WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER,
+ ExpandState->pw16_arState,
+ UNVOICED_LPC_ORDER);
+ if (ExpandState->w16_arGainScale > 0)
+ {
+ w32_tmp = ((WebRtc_Word32) 1) << (ExpandState->w16_arGainScale - 1);
+ }
+ else
+ {
+ w32_tmp = 0;
+ }
+
+ /* Note that shift value can be >16 which complicates things for some DSPs */
+ WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec,
+ ExpandState->w16_arGain, w32_tmp, ExpandState->w16_arGainScale, w16_lag);
+
+ WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_unvoicedVec,
+ ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, w16_lag);
+
+ WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState,
+ &(pw16_unvoicedVec[w16_lag - UNVOICED_LPC_ORDER]),
+ UNVOICED_LPC_ORDER);
+
+ /*
+ * Voiced + Unvoiced
+ */
+
+ /* For lag =
+ <=31*fs_mult => go from 1 to 0 in about 8 ms
+ (>=31..<=63)*fs_mult => go from 1 to 0 in about 16 ms
+ >=64*fs_mult => go from 1 to 0 in about 32 ms
+ */
+ w16_tmp = (31 - WebRtcSpl_NormW32(ExpandState->w16_maxLag)) - 5; /* getbits(w16_maxLag) -5 */
+ w16_vfractionChange = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(256, w16_tmp);
+ if (ExpandState->w16_stopMuting == 1)
+ {
+ w16_vfractionChange = 0;
+ }
+
+ /* Create combined signal (unmuted) by shifting in more and more of unvoiced part */
+ w16_tmp = 8 - w16_tmp; /* getbits(w16_vfractionChange) */
+ w16_tmp = (ExpandState->w16_currentVFraction - ExpandState->w16_vFraction) >> w16_tmp;
+ w16_tmp = WEBRTC_SPL_MIN(w16_tmp, w16_lag);
+ WebRtcNetEQ_MixVoiceUnvoice(pw16_outData, pw16_voicedVec, pw16_unvoicedVec,
+ &ExpandState->w16_currentVFraction, w16_vfractionChange, w16_tmp);
+
+ if (w16_tmp < w16_lag)
+ {
+ if (w16_vfractionChange != 0)
+ {
+ ExpandState->w16_currentVFraction = ExpandState->w16_vFraction;
+ }
+ w16_tmp2 = 16384 - ExpandState->w16_currentVFraction;
+ WebRtcSpl_ScaleAndAddVectorsWithRound(pw16_voicedVec + w16_tmp,
+ ExpandState->w16_currentVFraction, pw16_unvoicedVec + w16_tmp, w16_tmp2, 14,
+ pw16_outData + w16_tmp, (WebRtc_Word16) (w16_lag - w16_tmp));
+ }
+
+ /* Select muting factor */
+ if (ExpandState->w16_consecExp == 3)
+ {
+ /* 0.95 on 50 samples in NB (0.0010/fs_mult in Q20) */
+ ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope,
+ WebRtcNetEQ_k1049div[fs_mult]);
+ }
+ if (ExpandState->w16_consecExp == 7)
+ {
+ /* 0.90 on 50 samples in NB (0.0020/fs_mult in Q20) */
+ ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope,
+ WebRtcNetEQ_k2097div[fs_mult]);
+ }
+
+ /* Mute segment according to slope value */
+ if ((ExpandState->w16_consecExp != 0) || (ExpandState->w16_onset != 1))
+ {
+ /* Mute to the previous level, then continue with the muting */
+ WebRtcSpl_AffineTransformVector(pw16_outData, pw16_outData,
+ ExpandState->w16_expandMuteFactor, 8192, 14, w16_lag);
+
+ if ((ExpandState->w16_stopMuting != 1))
+ {
+ WebRtcNetEQ_MuteSignal(pw16_outData, ExpandState->w16_muteSlope, w16_lag);
+
+ w16_tmp = 16384 - (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_lag,
+ ExpandState->w16_muteSlope) + 8192) >> 6); /* 20-14 = 6 */
+ w16_tmp = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp,
+ ExpandState->w16_expandMuteFactor) + 8192) >> 14);
+
+ /* Guard against getting stuck with very small (but sometimes audible) gain */
+ if ((ExpandState->w16_consecExp > 3) && (w16_tmp
+ >= ExpandState->w16_expandMuteFactor))
+ {
+ ExpandState->w16_expandMuteFactor = 0;
+ }
+ else
+ {
+ ExpandState->w16_expandMuteFactor = w16_tmp;
+ }
+ }
+ }
+
+ } /* end if(!BGNonly) */
+
+ /*
+ * BGN
+ */
+
+ if (BGNState->w16_initialized == 1)
+ {
+ /* BGN parameters are initialized; use them */
+
+ WEBRTC_SPL_MEMCPY_W16(pw16_cngVec - BGN_LPC_ORDER,
+ BGNState->pw16_filterState,
+ BGN_LPC_ORDER);
+
+ if (BGNState->w16_scaleShift > 1)
+ {
+ w32_tmp = ((WebRtc_Word32) 1) << (BGNState->w16_scaleShift - 1);
+ }
+ else
+ {
+ w32_tmp = 0;
+ }
+
+ /* Scale random vector to correct energy level */
+ /* Note that shift value can be >16 which complicates things for some DSPs */
+ WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec,
+ BGNState->w16_scale, w32_tmp, BGNState->w16_scaleShift, w16_lag);
+
+ WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_cngVec, BGNState->pw16_filter,
+ BGN_LPC_ORDER + 1, w16_lag);
+
+ WEBRTC_SPL_MEMCPY_W16(BGNState->pw16_filterState,
+ &(pw16_cngVec[w16_lag-BGN_LPC_ORDER]),
+ BGN_LPC_ORDER);
+
+ /* Unmute the insertion of background noise */
+
+ if (bgnMode == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME
+ && BGNState->w16_mutefactor > 0)
+ {
+ /* fade BGN to zero */
+ /* calculate muting slope, approx 2^18/fsHz */
+ WebRtc_Word16 muteFactor;
+ if (fs_mult == 1)
+ {
+ muteFactor = -32;
+ }
+ else if (fs_mult == 2)
+ {
+ muteFactor = -16;
+ }
+ else if (fs_mult == 4)
+ {
+ muteFactor = -8;
+ }
+ else
+ {
+ muteFactor = -5;
+ }
+ /* use UnmuteSignal function with negative slope */
+ WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */
+ pw16_cngVec, muteFactor, /* In Q20 */
+ w16_lag);
+ }
+ else if (BGNState->w16_mutefactor < 16384 && !BGNonly)
+ {
+ /* if (w16_mutefactor < 1) and not BGN only (since then we use no muting) */
+
+ /*
+ * If BGN_OFF, or if BNG_FADE has started fading,
+ * mutefactor should not be increased.
+ */
+ if (ExpandState->w16_stopMuting != 1 && bgnMode != BGN_OFF && !(bgnMode
+ == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME))
+ {
+ WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */
+ pw16_cngVec, ExpandState->w16_muteSlope, /* In Q20 */
+ w16_lag);
+ }
+ else
+ {
+ /* BGN_ON and stop muting, or
+ * BGN_OFF (mute factor is always 0), or
+ * BGN_FADE has reached 0 */
+ WebRtcSpl_AffineTransformVector(pw16_cngVec, pw16_cngVec,
+ BGNState->w16_mutefactor, 8192, 14, w16_lag);
+ }
+ }
+ }
+ else
+ {
+ /* BGN parameters have not been initialized; use zero noise */
+ WebRtcSpl_MemSetW16(pw16_cngVec, 0, w16_lag);
+ }
+
+ if (BGNonly)
+ {
+ /* Copy BGN to outdata */
+ for (i = 0; i < w16_lag; i++)
+ {
+ pw16_outData[i] = pw16_cngVec[i];
+ }
+ }
+ else
+ {
+ /* Add CNG vector to the Voiced + Unvoiced vectors */
+ for (i = 0; i < w16_lag; i++)
+ {
+ pw16_outData[i] = pw16_outData[i] + pw16_cngVec[i];
+ }
+
+ /* increase call number */
+ ExpandState->w16_consecExp = ExpandState->w16_consecExp + 1;
+ if (ExpandState->w16_consecExp < 0) /* Guard against overflow */
+ ExpandState->w16_consecExp = FADE_BGN_TIME; /* "Arbitrary" large num of expands */
+ }
+
+ inst->w16_mode = MODE_EXPAND;
+ *pw16_len = w16_lag;
+
+ /* Update in-call and post-call statistics */
+ if (ExpandState->w16_stopMuting != 1 || BGNonly)
+ {
+ /*
+ * Only do this if StopMuting != 1 or if explicitly BGNonly, otherwise Expand is
+ * called from Merge or Normal and special measures must be taken.
+ */
+ inst->statInst.expandLength += (WebRtc_UWord32) *pw16_len;
+ if (ExpandState->w16_expandMuteFactor == 0 || BGNonly)
+ {
+ /* Only noise expansion */
+ inst->statInst.expandedNoiseSamples += *pw16_len;
+ }
+ else
+ {
+ /* Voice expand (note: not necessarily _voiced_) */
+ inst->statInst.expandedVoiceSamples += *pw16_len;
+ }
+ }
+
+ return 0;
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_GenerateBGN(...)
+ *
+ * This function generates and writes len samples of background noise to the
+ * output vector. The Expand function will be called repeatedly until the
+ * correct number of samples is produced.
+ *
+ * Input:
+ * - inst : NetEq instance, i.e. the user that requests more
+ * speech/audio data
+ * - scratchPtr : Pointer to scratch vector
+ * - len : Desired length of produced BGN.
+ *
+ *
+ * Output:
+ * - pw16_outData : Pointer to a memory space where the output data
+ * should be stored
+ *
+ * Return value : >=0 - Number of noise samples produced and written
+ * to output
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 len)
+{
+
+ WebRtc_Word16 pos = 0;
+ WebRtc_Word16 tempLen = len;
+
+ while (tempLen > 0)
+ {
+ /* while we still need more noise samples, call Expand to obtain background noise */
+ WebRtcNetEQ_Expand(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr,
+#endif
+ &pw16_outData[pos], &tempLen, 1 /*BGNonly*/);
+
+ pos += tempLen; /* we got this many samples */
+ tempLen = len - pos; /* this is the number of samples we still need */
+ }
+
+ return pos;
+}
+
+#undef SCRATCH_PW16_BEST_CORR_INDEX
+#undef SCRATCH_PW16_BEST_CORR
+#undef SCRATCH_PW16_BEST_DIST_INDEX
+#undef SCRATCH_PW16_BEST_DIST
+#undef SCRATCH_PW16_CORR_VEC
+#undef SCRATCH_PW16_CORR2
+#undef SCRATCH_PW32_AUTO_CORR
+#undef SCRATCH_PW16_RC
+#undef SCRATCH_PW16_RAND_VEC
+#undef SCRATCH_NETEQDSP_CORRELATOR
+#undef SCRATCH_PW16_SCALED_RAND_VEC
+#undef SCRATCH_PW16_UNVOICED_VEC_SPACE
+
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
new file mode 100644
index 0000000..909131b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is the main API for NetEQ. Helper macros are located in webrtc_neteq_help_macros.h,
+ * while some internal API functions are found in webrtc_neteq_internal.h.
+ */
+
+#include "typedefs.h"
+
+#ifndef WEBRTC_NETEQ_H
+#define WEBRTC_NETEQ_H
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/**********************************************************
+ * Definitions
+ */
+
+enum WebRtcNetEQDecoder
+{
+ kDecoderReservedStart,
+ kDecoderPCMu,
+ kDecoderPCMa,
+ kDecoderILBC,
+ kDecoderISAC,
+ kDecoderISACswb,
+ kDecoderPCM16B,
+ kDecoderPCM16Bwb,
+ kDecoderPCM16Bswb32kHz,
+ kDecoderPCM16Bswb48kHz,
+ kDecoderG722,
+ kDecoderRED,
+ kDecoderAVT,
+ kDecoderCNG,
+ kDecoderArbitrary,
+ kDecoderG729,
+ kDecoderG729_1,
+ kDecoderG726_16,
+ kDecoderG726_24,
+ kDecoderG726_32,
+ kDecoderG726_40,
+ kDecoderG722_1_16,
+ kDecoderG722_1_24,
+ kDecoderG722_1_32,
+ kDecoderG722_1C_24,
+ kDecoderG722_1C_32,
+ kDecoderG722_1C_48,
+ kDecoderSPEEX_8,
+ kDecoderSPEEX_16,
+ kDecoderCELT_32,
+ kDecoderGSMFR,
+ kDecoderAMR,
+ kDecoderAMRWB,
+ kDecoderReservedEnd
+};
+
+enum WebRtcNetEQNetworkType
+{
+ kUDPNormal,
+ kUDPVideoSync,
+ kTCPNormal,
+ kTCPLargeJitter,
+ kTCPXLargeJitter
+};
+
+enum WebRtcNetEQOutputType
+{
+ kOutputNormal,
+ kOutputPLC,
+ kOutputCNG,
+ kOutputPLCtoCNG,
+ kOutputVADPassive
+};
+
+enum WebRtcNetEQPlayoutMode
+{
+ kPlayoutOn, kPlayoutOff, kPlayoutFax, kPlayoutStreaming
+};
+
+/* Available modes for background noise (inserted after long expands) */
+enum WebRtcNetEQBGNMode
+{
+ kBGNOn, /* default "normal" behavior with eternal noise */
+ kBGNFade, /* noise fades to zero after some time */
+ kBGNOff
+/* background noise is always zero */
+};
+
+/*************************************************
+ * Definitions of decoder calls and the default
+ * API function calls for each codec
+ */
+
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecode)(void* state, WebRtc_Word16* encoded,
+ WebRtc_Word16 len, WebRtc_Word16* decoded,
+ WebRtc_Word16* speechType);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodePLC)(void* state, WebRtc_Word16* decoded,
+ WebRtc_Word16 frames);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodeInit)(void* state);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncAddLatePkt)(void* state, WebRtc_Word16* encoded,
+ WebRtc_Word16 len);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetMDinfo)(void* state);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
+ WebRtc_Word16* length);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
+ WebRtc_Word32 packet_size,
+ WebRtc_UWord16 rtp_seq_number,
+ WebRtc_UWord32 send_ts,
+ WebRtc_UWord32 arr_ts);
+typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetErrorCode)(void* state);
+
+/**********************************************************
+ * Structures
+ */
+
+typedef struct
+{
+ enum WebRtcNetEQDecoder codec;
+ WebRtc_Word16 payloadType;
+ WebRtcNetEQ_FuncDecode funcDecode;
+ WebRtcNetEQ_FuncDecode funcDecodeRCU;
+ WebRtcNetEQ_FuncDecodePLC funcDecodePLC;
+ WebRtcNetEQ_FuncDecodeInit funcDecodeInit;
+ WebRtcNetEQ_FuncAddLatePkt funcAddLatePkt;
+ WebRtcNetEQ_FuncGetMDinfo funcGetMDinfo;
+ WebRtcNetEQ_FuncGetPitchInfo funcGetPitch;
+ WebRtcNetEQ_FuncUpdBWEst funcUpdBWEst;
+ WebRtcNetEQ_FuncGetErrorCode funcGetErrorCode;
+ void* codec_state;
+ WebRtc_UWord16 codec_fs;
+} WebRtcNetEQ_CodecDef;
+
+typedef struct
+{
+ WebRtc_UWord16 fraction_lost;
+ WebRtc_UWord32 cum_lost;
+ WebRtc_UWord32 ext_max;
+ WebRtc_UWord32 jitter;
+} WebRtcNetEQ_RTCPStat;
+
+/**********************************************************
+ * NETEQ Functions
+ */
+
+/* Info functions */
+
+#define WEBRTC_NETEQ_MAX_ERROR_NAME 40
+int WebRtcNetEQ_GetVersion(char *version);
+int WebRtcNetEQ_GetErrorCode(void *inst);
+int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen);
+
+/* Instance memory assign functions */
+
+int WebRtcNetEQ_AssignSize(int *sizeinbytes);
+int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr);
+int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec,
+ int noOfCodecs, enum WebRtcNetEQNetworkType nwType,
+ int *MaxNoOfPackets, int *sizeinbytes);
+int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr,
+ int sizeinbytes);
+
+/* Init functions */
+
+int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs);
+int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon);
+int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs);
+int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode);
+int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode);
+int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode);
+
+/* Codec Database functions */
+
+int WebRtcNetEQ_CodecDbReset(void *inst);
+int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst);
+int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec);
+int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
+ WebRtc_Word16 *MaxEntries);
+int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
+ enum WebRtcNetEQDecoder *codec);
+
+/* Real-time functions */
+
+int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
+ WebRtc_UWord32 uw32_timeRec);
+int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
+int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
+int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
+int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp);
+int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType);
+
+/* VQmon related functions */
+int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
+ WebRtc_UWord16 *concealedVoiceDurationMs,
+ WebRtc_UWord8 *concealedVoiceFlags);
+int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
+ WebRtc_UWord8 *adaptationRate);
+int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
+ WebRtc_UWord16 *maxDelayMs);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
new file mode 100644
index 0000000..214bd10
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
@@ -0,0 +1,387 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains some helper macros that can be used when loading the
+ * NetEQ codec database.
+ */
+
+#ifndef WEBRTC_NETEQ_HELP_MACROS_H
+#define WEBRTC_NETEQ_HELP_MACROS_H
+
+#ifndef NULL
+#define NULL 0
+#endif
+
+/**********************************************************
+ * Help macros for NetEQ initialization
+ */
+
+#define SET_CODEC_PAR(inst,decoder,pt,state,fs) \
+ inst.codec=decoder; \
+ inst.payloadType=pt; \
+ inst.codec_state=state; \
+ inst.codec_fs=fs;
+
+#define SET_PCMU_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeU; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_PCMA_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeA; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_ILBC_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIlbcfix_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcIlbcfix_NetEqPlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIlbcfix_Decoderinit30Ms; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_ISAC_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \
+ inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \
+ inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
+
+#define SET_ISACfix_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsacfix_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsacfix_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsacfix_UpdateBwEstimate; \
+ inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsacfix_GetErrorCode;
+
+#define SET_ISACSWB_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \
+ inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \
+ inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
+
+#define SET_G729_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG729_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG729_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG729_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G729_1_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7291_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7291_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcG7291_DecodeBwe; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_PCM16B_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_PCM16B_WB_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_PCM16B_SWB32_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+
+#define SET_PCM16B_SWB48_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG722_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG722_DecoderInit;\
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1_16_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc16; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit16; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1_24_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode24; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc24; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit24; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1_32_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode32; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc32; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit32; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1C_24_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode24; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc24; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit24; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1C_32_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode32; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc32; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit32; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G722_1C_48_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode48; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc48; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit48; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_AMR_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmr_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmr_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmr_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_AMRWB_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmrWb_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmrWb_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmrWb_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_GSMFR_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcGSMFR_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcGSMFR_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcGSMFR_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G726_16_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode16; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit16; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G726_24_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode24; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit24; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G726_32_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode32; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit32; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_G726_40_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode40; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit40; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_SPEEX_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcSpeex_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcSpeex_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcSpeex_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_CELT_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_CELTSLAVE_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_DecodeSlave; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_RED_FUNCTIONS(inst) \
+ inst.funcDecode=NULL; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_AVT_FUNCTIONS(inst) \
+ inst.funcDecode=NULL; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#define SET_CNG_FUNCTIONS(inst) \
+ inst.funcDecode=NULL; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=NULL; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
+#endif /* WEBRTC_NETEQ_HELP_MACROS_H */
+
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
new file mode 100644
index 0000000..70ed044
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
@@ -0,0 +1,236 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the internal API functions.
+ */
+
+#include "typedefs.h"
+
+#ifndef WEBRTC_NETEQ_INTERNAL_H
+#define WEBRTC_NETEQ_INTERNAL_H
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+typedef struct
+{
+ WebRtc_UWord8 payloadType;
+ WebRtc_UWord16 sequenceNumber;
+ WebRtc_UWord32 timeStamp;
+ WebRtc_UWord32 SSRC;
+ WebRtc_UWord8 markerBit;
+} WebRtcNetEQ_RTPInfo;
+
+/****************************************************************************
+ * WebRtcNetEQ_RecInRTPStruct(...)
+ *
+ * Alternative RecIn function, used when the RTP data has already been
+ * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo).
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - rtpInfo : Pointer to RTP info
+ * - payloadPtr : Pointer to the RTP payload (first byte after header)
+ * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr
+ * - timeRec : Receive time (in timestamps of the used codec)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
+ const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
+ WebRtc_UWord32 timeRec);
+
+/****************************************************************************
+ * WebRtcNetEQ_GetMasterSlaveInfoSize(...)
+ *
+ * Get size in bytes for master/slave struct msInfo used in
+ * WebRtcNetEQ_RecOutMasterSlave.
+ *
+ * Return value : Struct size in bytes
+ *
+ */
+
+int WebRtcNetEQ_GetMasterSlaveInfoSize();
+
+/****************************************************************************
+ * WebRtcNetEQ_RecOutMasterSlave(...)
+ *
+ * RecOut function for running several NetEQ instances in master/slave mode.
+ * One master can be used to control several slaves.
+ * The MasterSlaveInfo struct must be allocated outside NetEQ.
+ * Use function WebRtcNetEQ_GetMasterSlaveInfoSize to get the size needed.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - isMaster : Non-zero indicates that this is the master channel
+ * - msInfo : (slave only) Information from master
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ * - pw16_outData : Pointer to vector where output should be written
+ * - pw16_len : Pointer to variable where output length is returned
+ * - msInfo : (master only) Information to slave(s)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len, void *msInfo,
+ WebRtc_Word16 isMaster);
+
+typedef struct
+{
+ uint16_t currentBufferSize; /* Current jitter buffer size in ms. */
+ uint16_t preferredBufferSize; /* Preferred buffer size in ms. */
+ uint16_t jitterPeaksFound; /* 1 if adding extra delay due to peaky
+ * jitter; 0 otherwise. */
+ uint16_t currentPacketLossRate; /* Loss rate (network + late) (Q14). */
+ uint16_t currentDiscardRate; /* Late loss rate (Q14). */
+ uint16_t currentExpandRate; /* Fraction (of original stream) of
+ * synthesized speech inserted through
+ * expansion (in Q14). */
+ uint16_t currentPreemptiveRate; /* Fraction of data inserted through
+ * pre-emptive expansion (in Q14). */
+ uint16_t currentAccelerateRate; /* Fraction of data removed through
+ * acceleration (in Q14). */
+ int32_t clockDriftPPM; /* Average clock-drift in parts-per-
+ * million (positive or negative). */
+} WebRtcNetEQ_NetworkStatistics;
+
+/*
+ * Get the "in-call" statistics from NetEQ.
+ * The statistics are reset after the query.
+ */
+int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats);
+
+/*
+ * Get the raw waiting times for decoded frames. The function writes the last
+ * recorded waiting times (from frame arrival to frame decoding) to the memory
+ * pointed to by waitingTimeMs. The number of elements written is in the return
+ * value. No more than maxLength elements are written. Statistics are reset on
+ * each query.
+ */
+int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst,
+ int max_length,
+ int* waiting_times_ms);
+
+/***********************************************/
+/* Functions for post-decode VAD functionality */
+/***********************************************/
+
+/* NetEQ must be compiled with the flag NETEQ_VAD enabled for these functions to work. */
+
+/*
+ * VAD function pointer types
+ *
+ * These function pointers match the definitions of webrtc VAD functions WebRtcVad_Init,
+ * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h.
+ */
+typedef int (*WebRtcNetEQ_VADInitFunction)(void *VAD_inst);
+typedef int (*WebRtcNetEQ_VADSetmodeFunction)(void *VAD_inst, int mode);
+typedef WebRtc_Word16 (*WebRtcNetEQ_VADFunction)(void *VAD_inst, WebRtc_Word16 fs,
+ WebRtc_Word16 *frame, WebRtc_Word16 frameLen);
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADInstance(...)
+ *
+ * Provide a pointer to an allocated VAD instance. If function is never
+ * called or it is called with NULL pointer as VAD_inst, the post-decode
+ * VAD functionality is disabled. Also provide pointers to init, setmode
+ * and VAD functions. These are typically pointers to WebRtcVad_Init,
+ * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the
+ * interface file webrtc_vad.h.
+ *
+ * Input:
+ * - NetEQ_inst : NetEQ instance
+ * - VADinst : VAD instance
+ * - initFunction : Pointer to VAD init function
+ * - setmodeFunction : Pointer to VAD setmode function
+ * - VADfunction : Pointer to VAD function
+ *
+ * Output:
+ * - NetEQ_inst : Updated NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst,
+ WebRtcNetEQ_VADInitFunction initFunction,
+ WebRtcNetEQ_VADSetmodeFunction setmodeFunction,
+ WebRtcNetEQ_VADFunction VADFunction);
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADMode(...)
+ *
+ * Pass an aggressiveness mode parameter to the post-decode VAD instance.
+ * If this function is never called, mode 0 (quality mode) is used as default.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - mode : mode parameter (same range as WebRtc VAD mode)
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADMode(void *NetEQ_inst, int mode);
+
+/****************************************************************************
+ * WebRtcNetEQ_RecOutNoDecode(...)
+ *
+ * Special RecOut that does not do any decoding.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ * - pw16_outData : Pointer to vector where output should be written
+ * - pw16_len : Pointer to variable where output length is returned
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len);
+
+/****************************************************************************
+ * WebRtcNetEQ_FlushBuffers(...)
+ *
+ * Flush packet and speech buffers. Does not reset codec database or
+ * jitter statistics.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_FlushBuffers(void *inst);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu.h b/trunk/src/modules/audio_coding/neteq/mcu.h
new file mode 100644
index 0000000..499684a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mcu.h
@@ -0,0 +1,284 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * MCU struct and functions related to the MCU side operations.
+ */
+
+#ifndef MCU_H
+#define MCU_H
+
+#include "typedefs.h"
+
+#include "codec_db.h"
+#include "rtcp.h"
+#include "packet_buffer.h"
+#include "buffer_stats.h"
+#include "neteq_statistics.h"
+
+#ifdef NETEQ_ATEVENT_DECODE
+#include "dtmf_buffer.h"
+#endif
+
+#define MAX_ONE_DESC 5 /* cannot do more than this many consecutive one-descriptor decodings */
+#define MAX_LOSS_REPORT_PERIOD 60 /* number of seconds between auto-reset */
+
+enum TsScaling
+{
+ kTSnoScaling = 0,
+ kTSscalingTwo,
+ kTSscalingTwoThirds,
+ kTSscalingFourThirds
+};
+
+enum { kLenWaitingTimes = 100 };
+
+typedef struct
+{
+
+ WebRtc_Word16 current_Codec;
+ WebRtc_Word16 current_Payload;
+ WebRtc_UWord32 timeStamp; /* Next timestamp that should be played */
+ WebRtc_Word16 millisecondsPerCall;
+ WebRtc_UWord16 timestampsPerCall; /* Output chunk size */
+ WebRtc_UWord16 fs;
+ WebRtc_UWord32 ssrc; /* Current ssrc */
+ WebRtc_Word16 new_codec;
+ WebRtc_Word16 first_packet;
+
+ /* MCU/DSP Communication layer */
+ WebRtc_Word16 *pw16_readAddress;
+ WebRtc_Word16 *pw16_writeAddress;
+ void *main_inst;
+
+ CodecDbInst_t codec_DB_inst; /* Information about all the codecs, i.e. which
+ functions to use and which codpoints that
+ have been assigned */
+ SplitInfo_t PayloadSplit_inst; /* Information about how the current codec
+ payload should be splitted */
+ WebRtcNetEQ_RTCP_t RTCP_inst; /* RTCP statistics */
+ PacketBuf_t PacketBuffer_inst; /* The packet buffer */
+ BufstatsInst_t BufferStat_inst; /* Statistics that are used to make decision
+ for what the DSP should perform */
+#ifdef NETEQ_ATEVENT_DECODE
+ dtmf_inst_t DTMF_inst;
+#endif
+ int NoOfExpandCalls;
+ WebRtc_Word16 AVT_PlayoutOn;
+ enum WebRtcNetEQPlayoutMode NetEqPlayoutMode;
+
+ WebRtc_Word16 one_desc; /* Number of times running on one desc */
+
+ WebRtc_UWord32 lostTS; /* Number of timestamps lost */
+ WebRtc_UWord32 lastReportTS; /* Timestamp elapsed since last report was given */
+
+ int waiting_times[kLenWaitingTimes]; /* Waiting time statistics storage. */
+ int len_waiting_times;
+ int next_waiting_time_index;
+
+ WebRtc_UWord32 externalTS;
+ WebRtc_UWord32 internalTS;
+ WebRtc_Word16 TSscalingInitialized;
+ enum TsScaling scalingFactor;
+
+#ifdef NETEQ_STEREO
+ int usingStereo;
+#endif
+
+} MCUInst_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_McuReset(...)
+ *
+ * Reset the MCU instance.
+ *
+ * Input:
+ * - inst : MCU instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_McuReset(MCUInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_ResetMcuInCallStats(...)
+ *
+ * Reset MCU-side statistics variables for the in-call statistics.
+ *
+ * Input:
+ * - inst : MCU instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_ResetWaitingTimeStats(...)
+ *
+ * Reset waiting-time statistics.
+ *
+ * Input:
+ * - inst : MCU instance.
+ *
+ * Return value : n/a
+ */
+void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_LogWaitingTime(...)
+ *
+ * Log waiting-time to the statistics.
+ *
+ * Input:
+ * - inst : MCU instance.
+ * - waiting_time : Waiting time in "RecOut calls" (i.e., 1 call = 10 ms).
+ *
+ * Return value : n/a
+ */
+void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time);
+
+/****************************************************************************
+ * WebRtcNetEQ_ResetMcuJitterStat(...)
+ *
+ * Reset MCU-side statistics variables for the post-call statistics.
+ *
+ * Input:
+ * - inst : MCU instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_McuAddressInit(...)
+ *
+ * Initializes MCU with read address and write address.
+ *
+ * Input:
+ * - inst : MCU instance
+ * - Data2McuAddress : Pointer to MCU address
+ * - Data2DspAddress : Pointer to DSP address
+ * - main_inst : Pointer to NetEQ main instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress,
+ void * Data2DspAddress, void *main_inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_McuSetFs(...)
+ *
+ * Initializes MCU with read address and write address.
+ *
+ * Input:
+ * - inst : MCU instance
+ * - fs_hz : Sample rate in Hz -- 8000, 16000, 32000, (48000)
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs_hz);
+
+/****************************************************************************
+ * WebRtcNetEQ_SignalMcu(...)
+ *
+ * Signal the MCU that data is available and ask for a RecOut decision.
+ *
+ * Input:
+ * - inst : MCU instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+int WebRtcNetEQ_SignalMcu(MCUInst_t *inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_RecInInternal(...)
+ *
+ * This function inserts a packet into the jitter buffer.
+ *
+ * Input:
+ * - MCU_inst : MCU instance
+ * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct
+ * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacket,
+ WebRtc_UWord32 uw32_timeRec);
+
+/****************************************************************************
+ * WebRtcNetEQ_RecInInternal(...)
+ *
+ * Split the packet according to split_inst and inserts the parts into
+ * Buffer_inst.
+ *
+ * Input:
+ * - MCU_inst : MCU instance
+ * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct
+ * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
+ SplitInfo_t *split_inst, WebRtc_Word16 *flushed);
+
+/****************************************************************************
+ * WebRtcNetEQ_GetTimestampScaling(...)
+ *
+ * Update information about timestamp scaling for a payload type
+ * in MCU_inst->scalingFactor.
+ *
+ * Input:
+ * - MCU_inst : MCU instance
+ * - rtpPayloadType : RTP payload number
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType);
+
+/****************************************************************************
+ * WebRtcNetEQ_ScaleTimestampExternalToInternal(...)
+ *
+ * Convert from external to internal timestamp using current scaling info.
+ *
+ * Input:
+ * - MCU_inst : MCU instance
+ * - externalTS : External timestamp
+ *
+ * Return value : Internal timestamp
+ */
+
+WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
+ WebRtc_UWord32 externalTS);
+
+/****************************************************************************
+ * WebRtcNetEQ_ScaleTimestampInternalToExternal(...)
+ *
+ * Convert from external to internal timestamp using current scaling info.
+ *
+ * Input:
+ * - MCU_inst : MCU instance
+ * - externalTS : Internal timestamp
+ *
+ * Return value : External timestamp
+ */
+
+WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
+ WebRtc_UWord32 internalTS);
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_address_init.c b/trunk/src/modules/audio_coding/neteq/mcu_address_init.c
new file mode 100644
index 0000000..0306a85
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mcu_address_init.c
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "mcu.h"
+
+#include <string.h> /* to define NULL */
+
+/*
+ * Initializes MCU with read address and write address
+ */
+int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress,
+ void * Data2DspAddress, void *main_inst)
+{
+
+ inst->pw16_readAddress = (WebRtc_Word16*) Data2McuAddress;
+ inst->pw16_writeAddress = (WebRtc_Word16*) Data2DspAddress;
+ inst->main_inst = main_inst;
+
+ inst->millisecondsPerCall = 10;
+
+ /* Do expansions in the beginning */
+ if (inst->pw16_writeAddress != NULL) inst->pw16_writeAddress[0] = DSP_INSTR_EXPAND;
+
+ return (0);
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c
new file mode 100644
index 0000000..13025d4
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Communication between MCU and DSP sides.
+ */
+
+#include "mcu_dsp_common.h"
+
+#include <string.h>
+
+/* Initialize instances with read and write address */
+int WebRtcNetEQ_DSPinit(MainInst_t *inst)
+{
+ int res = 0;
+
+ res |= WebRtcNetEQ_AddressInit(&inst->DSPinst, NULL, NULL, inst);
+ res |= WebRtcNetEQ_McuAddressInit(&inst->MCUinst, NULL, NULL, inst);
+
+ return res;
+
+}
+
+/* The DSP side will call this function to interrupt the MCU side */
+int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem)
+{
+ inst->MCUinst.pw16_readAddress = pw16_shared_mem;
+ inst->MCUinst.pw16_writeAddress = pw16_shared_mem;
+ return WebRtcNetEQ_SignalMcu(&inst->MCUinst);
+}
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h
new file mode 100644
index 0000000..e3f4213
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * The main NetEQ instance, which is where the DSP and MCU sides join.
+ */
+
+#ifndef MCU_DSP_COMMON_H
+#define MCU_DSP_COMMON_H
+
+#include "typedefs.h"
+
+#include "dsp.h"
+#include "mcu.h"
+
+/* Define size of shared memory area. */
+#if defined(NETEQ_48KHZ_WIDEBAND)
+ #define SHARED_MEM_SIZE (6*640)
+#elif defined(NETEQ_32KHZ_WIDEBAND)
+ #define SHARED_MEM_SIZE (4*640)
+#elif defined(NETEQ_WIDEBAND)
+ #define SHARED_MEM_SIZE (2*640)
+#else
+ #define SHARED_MEM_SIZE 640
+#endif
+
+/* Struct to hold the NetEQ instance */
+typedef struct
+{
+ DSPInst_t DSPinst; /* DSP part of the NetEQ instance */
+ MCUInst_t MCUinst; /* MCU part of the NetEQ instance */
+ WebRtc_Word16 ErrorCode; /* Store last error code */
+#ifdef NETEQ_STEREO
+ WebRtc_Word16 masterSlave; /* 0 = not set, 1 = master, 2 = slave */
+#endif /* NETEQ_STEREO */
+} MainInst_t;
+
+/* Struct used for communication between DSP and MCU sides of NetEQ */
+typedef struct
+{
+ WebRtc_UWord32 playedOutTS; /* Timestamp position at end of DSP data */
+ WebRtc_UWord16 samplesLeft; /* Number of samples stored */
+ WebRtc_Word16 MD; /* Multiple description codec information */
+ WebRtc_Word16 lastMode; /* Latest mode of NetEQ playout */
+ WebRtc_Word16 frameLen; /* Frame length of previously decoded packet */
+} DSP2MCU_info_t;
+
+/* Initialize instances with read and write address */
+int WebRtcNetEQ_DSPinit(MainInst_t *inst);
+
+/* The DSP side will call this function to interrupt the MCU side */
+int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem);
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_reset.c b/trunk/src/modules/audio_coding/neteq/mcu_reset.c
new file mode 100644
index 0000000..3aae4ce
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mcu_reset.c
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Reset MCU side data.
+ */
+
+#include "mcu.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "automode.h"
+
+int WebRtcNetEQ_McuReset(MCUInst_t *inst)
+{
+
+#ifdef NETEQ_ATEVENT_DECODE
+ int ok;
+#endif
+
+ /* MCU/DSP Communication layer */
+ inst->pw16_readAddress = NULL;
+ inst->pw16_writeAddress = NULL;
+ inst->main_inst = NULL;
+ inst->one_desc = 0;
+ inst->BufferStat_inst.Automode_inst.extraDelayMs = 0;
+ inst->NetEqPlayoutMode = kPlayoutOn;
+
+ WebRtcNetEQ_DbReset(&inst->codec_DB_inst);
+ memset(&inst->PayloadSplit_inst, 0, sizeof(SplitInfo_t));
+
+ /* Clear the Packet buffer and the pointer to memory storage */
+ WebRtcNetEQ_PacketBufferFlush(&inst->PacketBuffer_inst);
+ inst->PacketBuffer_inst.memorySizeW16 = 0;
+ inst->PacketBuffer_inst.maxInsertPositions = 0;
+
+ /* Clear the decision and delay history */
+ memset(&inst->BufferStat_inst, 0, sizeof(BufstatsInst_t));
+#ifdef NETEQ_ATEVENT_DECODE
+ ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560);
+ if (ok != 0)
+ {
+ return ok;
+ }
+#endif
+ inst->NoOfExpandCalls = 0;
+ inst->current_Codec = -1;
+ inst->current_Payload = -1;
+
+ inst->millisecondsPerCall = 10;
+ inst->timestampsPerCall = inst->millisecondsPerCall * 8;
+ inst->fs = 8000;
+ inst->first_packet = 1;
+
+ WebRtcNetEQ_ResetMcuInCallStats(inst);
+
+ WebRtcNetEQ_ResetWaitingTimeStats(inst);
+
+ WebRtcNetEQ_ResetMcuJitterStat(inst);
+
+ WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
+ inst->PacketBuffer_inst.maxInsertPositions);
+
+ return 0;
+}
+
+/*
+ * Reset MCU-side statistics variables for the in-call statistics.
+ */
+
+int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst)
+{
+ inst->lostTS = 0;
+ inst->lastReportTS = 0;
+ inst->PacketBuffer_inst.discardedPackets = 0;
+
+ return 0;
+}
+
+/*
+ * Reset waiting-time statistics.
+ */
+
+void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst) {
+ memset(inst->waiting_times, 0,
+ kLenWaitingTimes * sizeof(inst->waiting_times[0]));
+ inst->len_waiting_times = 0;
+ inst->next_waiting_time_index = 0;
+}
+
+/*
+ * Store waiting-time in the statistics.
+ */
+
+void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time) {
+ assert(inst->next_waiting_time_index < kLenWaitingTimes);
+ inst->waiting_times[inst->next_waiting_time_index] = waiting_time;
+ inst->next_waiting_time_index++;
+ if (inst->next_waiting_time_index >= kLenWaitingTimes) {
+ inst->next_waiting_time_index = 0;
+ }
+ if (inst->len_waiting_times < kLenWaitingTimes) {
+ inst->len_waiting_times++;
+ }
+}
+
+/*
+ * Reset all MCU-side statistics variables for the post-call statistics.
+ */
+
+int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst)
+{
+ inst->BufferStat_inst.Automode_inst.countIAT500ms = 0;
+ inst->BufferStat_inst.Automode_inst.countIAT1000ms = 0;
+ inst->BufferStat_inst.Automode_inst.countIAT2000ms = 0;
+ inst->BufferStat_inst.Automode_inst.longestIATms = 0;
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/merge.c b/trunk/src/modules/audio_coding/neteq/merge.c
new file mode 100644
index 0000000..5f020a9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/merge.c
@@ -0,0 +1,564 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is the function to merge a new packet with expanded data after a packet loss.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+
+/****************************************************************************
+ * WebRtcNetEQ_Merge(...)
+ *
+ * This function...
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to new decoded speech.
+ * - len : Number of samples in pw16_decoded.
+ *
+ *
+ * Output:
+ * - inst : Updated user information
+ * - outData : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Number of samples written to pw16_outData
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_expanded 210*fs/8000 0 209*fs/8000
+ WebRtc_Word16 pw16_expandedLB 100 210*fs/8000 99+210*fs/8000
+ WebRtc_Word16 pw16_decodedLB 40 100+210*fs/8000 139+210*fs/8000
+ WebRtc_Word32 pw32_corr 2*60 140+210*fs/8000 260+210*fs/8000
+ WebRtc_Word16 pw16_corrVec 68 210*fs/8000 67+210*fs/8000
+
+ [gap in scratch vector]
+
+ func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/8000 39+496*fs/8000
+
+ Total: 40+496*fs/8000
+ */
+
+#define SCRATCH_pw16_expanded 0
+#if (defined(NETEQ_48KHZ_WIDEBAND))
+#define SCRATCH_pw16_expandedLB 1260
+#define SCRATCH_pw16_decodedLB 1360
+#define SCRATCH_pw32_corr 1400
+#define SCRATCH_pw16_corrVec 1260
+#define SCRATCH_NETEQ_EXPAND 756
+#elif (defined(NETEQ_32KHZ_WIDEBAND))
+#define SCRATCH_pw16_expandedLB 840
+#define SCRATCH_pw16_decodedLB 940
+#define SCRATCH_pw32_corr 980
+#define SCRATCH_pw16_corrVec 840
+#define SCRATCH_NETEQ_EXPAND 504
+#elif (defined(NETEQ_WIDEBAND))
+#define SCRATCH_pw16_expandedLB 420
+#define SCRATCH_pw16_decodedLB 520
+#define SCRATCH_pw32_corr 560
+#define SCRATCH_pw16_corrVec 420
+#define SCRATCH_NETEQ_EXPAND 252
+#else /* NB */
+#define SCRATCH_pw16_expandedLB 210
+#define SCRATCH_pw16_decodedLB 310
+#define SCRATCH_pw32_corr 350
+#define SCRATCH_pw16_corrVec 210
+#define SCRATCH_NETEQ_EXPAND 126
+#endif
+
+int WebRtcNetEQ_Merge(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len)
+{
+
+ WebRtc_Word16 fs_mult;
+ WebRtc_Word16 fs_shift;
+ WebRtc_Word32 w32_En_new_frame, w32_En_old_frame;
+ WebRtc_Word16 w16_expmax, w16_newmax;
+ WebRtc_Word16 w16_tmp, w16_tmp2;
+ WebRtc_Word32 w32_tmp;
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_pw16_expanded;
+ WebRtc_Word16 *pw16_expandedLB = pw16_scratchPtr + SCRATCH_pw16_expandedLB;
+ WebRtc_Word16 *pw16_decodedLB = pw16_scratchPtr + SCRATCH_pw16_decodedLB;
+ WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
+ WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;
+#else
+ WebRtc_Word16 pw16_expanded[(125+80+5)*FSMULT];
+ WebRtc_Word16 pw16_expandedLB[100];
+ WebRtc_Word16 pw16_decodedLB[40];
+ WebRtc_Word32 pw32_corr[60];
+ WebRtc_Word16 pw16_corrVec[4+60+4];
+#endif
+ WebRtc_Word16 *pw16_corr = &pw16_corrVec[4];
+ WebRtc_Word16 w16_stopPos = 0, w16_bestIndex, w16_interpLen;
+ WebRtc_Word16 w16_bestVal; /* bestVal is dummy */
+ WebRtc_Word16 w16_startfact, w16_inc;
+ WebRtc_Word16 w16_expandedLen;
+ WebRtc_Word16 w16_startPos;
+ WebRtc_Word16 w16_expLen, w16_newLen = 0;
+ WebRtc_Word16 *pw16_decodedOut;
+ WebRtc_Word16 w16_muted;
+
+ int w16_decodedLen = len;
+
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo *msInfo = inst->msInfo;
+#endif
+
+ fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
+ fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */
+
+ /*************************************
+ * Generate data to merge with
+ *************************************/
+ /*
+ * Check how much data that is left since earlier
+ * (at least there should be the overlap)...
+ */
+ w16_startPos = inst->endPosition - inst->curPosition;
+ /* Get one extra expansion to merge and overlap with */
+ inst->ExpandInst.w16_stopMuting = 1;
+ inst->ExpandInst.w16_lagsDirection = 1; /* make sure we get the "optimal" lag */
+ inst->ExpandInst.w16_lagsPosition = -1; /* out of the 3 possible ones */
+ w16_expandedLen = 0; /* Does not fill any function currently */
+
+ if (w16_startPos >= 210 * FSMULT)
+ {
+ /*
+ * The number of samples available in the sync buffer is more than what fits in
+ * pw16_expanded.Keep the first 210*FSMULT samples, but shift them towards the end of
+ * the buffer. This is ok, since all of the buffer will be expand data anyway, so as
+ * long as the beginning is left untouched, we're fine.
+ */
+
+ w16_tmp = w16_startPos - 210 * FSMULT; /* length difference */
+
+ WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[inst->curPosition+w16_tmp] ,
+ &inst->speechBuffer[inst->curPosition], 210*FSMULT);
+
+ inst->curPosition += w16_tmp; /* move start position of sync buffer accordingly */
+ w16_startPos = 210 * FSMULT; /* this is the truncated length */
+ }
+
+ WebRtcNetEQ_Expand(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_expanded, /* let Expand write to beginning of pw16_expanded to avoid overflow */
+ &w16_newLen, 0);
+
+ /*
+ * Now shift the data in pw16_expanded to where it belongs.
+ * Truncate all that ends up outside the vector.
+ */
+
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_expanded[w16_startPos], pw16_expanded,
+ WEBRTC_SPL_MIN(w16_newLen,
+ WEBRTC_SPL_MAX(210*FSMULT - w16_startPos, 0) ) );
+
+ inst->ExpandInst.w16_stopMuting = 0;
+
+ /* Copy what is left since earlier into the expanded vector */
+
+ WEBRTC_SPL_MEMCPY_W16(pw16_expanded, &inst->speechBuffer[inst->curPosition], w16_startPos);
+
+ /*
+ * Do "ugly" copy and paste from the expanded in order to generate more data
+ * to correlate (but not interpolate) with.
+ */
+ w16_expandedLen = (120 + 80 + 2) * fs_mult;
+ w16_expLen = w16_startPos + w16_newLen;
+
+ if (w16_expLen < w16_expandedLen)
+ {
+ while ((w16_expLen + w16_newLen) < w16_expandedLen)
+ {
+ WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos],
+ w16_newLen);
+ w16_expLen += w16_newLen;
+ }
+
+ /* Copy last part (fraction of a whole expansion) */
+
+ WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos],
+ (w16_expandedLen-w16_expLen));
+ }
+ w16_expLen = w16_expandedLen;
+
+ /* Adjust muting factor (main muting factor times expand muting factor) */
+ inst->w16_muteFactor
+ = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
+ inst->ExpandInst.w16_expandMuteFactor, 14);
+
+ /* Adjust muting factor if new vector is more or less of the BGN energy */
+ len = WEBRTC_SPL_MIN(64*fs_mult, w16_decodedLen);
+ w16_expmax = WebRtcSpl_MaxAbsValueW16(pw16_expanded, (WebRtc_Word16) len);
+ w16_newmax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+
+ /* Calculate energy of old data */
+ w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_expmax, w16_expmax));
+ w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0);
+ w32_En_old_frame = WebRtcNetEQ_DotW16W16(pw16_expanded, pw16_expanded, len, w16_tmp);
+
+ /* Calculate energy of new data */
+ w16_tmp2 = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_newmax, w16_newmax));
+ w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2,0);
+ w32_En_new_frame = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, len, w16_tmp2);
+
+ /* Align to same Q-domain */
+ if (w16_tmp2 > w16_tmp)
+ {
+ w32_En_old_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_old_frame, (w16_tmp2-w16_tmp));
+ }
+ else
+ {
+ w32_En_new_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_new_frame, (w16_tmp-w16_tmp2));
+ }
+
+ /* Calculate muting factor to use for new frame */
+ if (w32_En_new_frame > w32_En_old_frame)
+ {
+ /* Normalize w32_En_new_frame to 14 bits */
+ w16_tmp = WebRtcSpl_NormW32(w32_En_new_frame) - 17;
+ w32_En_new_frame = WEBRTC_SPL_SHIFT_W32(w32_En_new_frame, w16_tmp);
+
+ /*
+ * Put w32_En_old_frame in a domain 14 higher, so that
+ * w32_En_old_frame/w32_En_new_frame is in Q14
+ */
+ w16_tmp = w16_tmp + 14;
+ w32_En_old_frame = WEBRTC_SPL_SHIFT_W32(w32_En_old_frame, w16_tmp);
+ w16_tmp
+ = WebRtcSpl_DivW32W16ResW16(w32_En_old_frame, (WebRtc_Word16) w32_En_new_frame);
+ /* Calculate sqrt(w32_En_old_frame/w32_En_new_frame) in Q14 */
+ w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,14));
+ }
+ else
+ {
+ w16_muted = 16384; /* Set = 1.0 when old frame has higher energy than new */
+ }
+
+ /* Set the raise the continued muting factor w16_muted if w16_muteFactor is lower */
+ if (w16_muted > inst->w16_muteFactor)
+ {
+ inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384);
+ }
+
+#ifdef NETEQ_STEREO
+
+ /* Sanity for msInfo */
+ if (msInfo == NULL)
+ {
+ /* this should not happen here */
+ return MASTER_SLAVE_ERROR;
+ }
+
+ /* do not downsample and calculate correlations for slave instance(s) */
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+#endif
+
+ /*********************************************
+ * Downsample to 4kHz and find best overlap
+ *********************************************/
+
+ /* Downsample to 4 kHz */
+ if (inst->fs == 8000)
+ {
+ WebRtcSpl_DownsampleFast(&pw16_expanded[2], (WebRtc_Word16) (w16_expandedLen - 2),
+ pw16_expandedLB, (WebRtc_Word16) (100),
+ (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl, (WebRtc_Word16) 3,
+ (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+ if (w16_decodedLen <= 80)
+ {
+ /* Not quite long enough, so we have to cheat a bit... */
+ WebRtc_Word16 temp_len = w16_decodedLen - 2;
+ w16_tmp = temp_len / 2;
+ WebRtcSpl_DownsampleFast(&pw16_decoded[2], temp_len,
+ pw16_decodedLB, w16_tmp,
+ (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
+ (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+ WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40 - w16_tmp));
+ }
+ else
+ {
+ WebRtcSpl_DownsampleFast(&pw16_decoded[2],
+ (WebRtc_Word16) (w16_decodedLen - 2), pw16_decodedLB,
+ (WebRtc_Word16) (40), (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
+ (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
+ }
+#ifdef NETEQ_WIDEBAND
+ }
+ else if (inst->fs==16000)
+ {
+ WebRtcSpl_DownsampleFast(
+ &pw16_expanded[4], (WebRtc_Word16)(w16_expandedLen-4),
+ pw16_expandedLB, (WebRtc_Word16)(100),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
+ (WebRtc_Word16)4, (WebRtc_Word16)0);
+ if (w16_decodedLen<=160)
+ {
+ /* Not quite long enough, so we have to cheat a bit... */
+ WebRtc_Word16 temp_len = w16_decodedLen - 4;
+ w16_tmp = temp_len / 4;
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[4], temp_len,
+ pw16_decodedLB, w16_tmp,
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
+ (WebRtc_Word16)4, (WebRtc_Word16)0);
+ WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
+ }
+ else
+ {
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[4], (WebRtc_Word16)(w16_decodedLen-4),
+ pw16_decodedLB, (WebRtc_Word16)(40),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
+ (WebRtc_Word16)4, (WebRtc_Word16)0);
+ }
+#endif
+#ifdef NETEQ_32KHZ_WIDEBAND
+ }
+ else if (inst->fs==32000)
+ {
+ /*
+ * TODO(hlundin) Why is the offset into pw16_expanded 6?
+ */
+ WebRtcSpl_DownsampleFast(
+ &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
+ pw16_expandedLB, (WebRtc_Word16)(100),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)8, (WebRtc_Word16)0);
+ if (w16_decodedLen<=320)
+ {
+ /* Not quite long enough, so we have to cheat a bit... */
+ WebRtc_Word16 temp_len = w16_decodedLen - 6;
+ w16_tmp = temp_len / 8;
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[6], temp_len,
+ pw16_decodedLB, w16_tmp,
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)8, (WebRtc_Word16)0);
+ WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
+ }
+ else
+ {
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
+ pw16_decodedLB, (WebRtc_Word16)(40),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)8, (WebRtc_Word16)0);
+ }
+#endif
+#ifdef NETEQ_48KHZ_WIDEBAND
+ }
+ else /* if (inst->fs==48000) */
+ {
+ /*
+ * TODO(hlundin) Why is the offset into pw16_expanded 6?
+ */
+ WebRtcSpl_DownsampleFast(
+ &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
+ pw16_expandedLB, (WebRtc_Word16)(100),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)12, (WebRtc_Word16)0);
+ if (w16_decodedLen<=320)
+ {
+ /* Not quite long enough, so we have to cheat a bit... */
+ /*
+ * TODO(hlundin): Is this correct? Downsampling is a factor 12
+ * but w16_tmp = temp_len / 8.
+ * (Was w16_tmp = ((w16_decodedLen-6)>>3) before re-write.)
+ */
+ WebRtc_Word16 temp_len = w16_decodedLen - 6;
+ w16_tmp = temp_len / 8;
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[6], temp_len,
+ pw16_decodedLB, w16_tmp,
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)12, (WebRtc_Word16)0);
+ WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
+ }
+ else
+ {
+ WebRtcSpl_DownsampleFast(
+ &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
+ pw16_decodedLB, (WebRtc_Word16)(40),
+ (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
+ (WebRtc_Word16)12, (WebRtc_Word16)0);
+ }
+#endif
+ }
+
+ /* Calculate correlation without any normalization (40 samples) */
+ w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) inst->ExpandInst.w16_maxLag,
+ (WebRtc_Word16) (fs_mult * 2)) + 1;
+ w16_stopPos = WEBRTC_SPL_MIN(60, w16_tmp);
+ w32_tmp = WEBRTC_SPL_MUL_16_16(w16_expmax, w16_newmax);
+ if (w32_tmp > 26843546)
+ {
+ w16_tmp = 3;
+ }
+ else
+ {
+ w16_tmp = 0;
+ }
+
+ WebRtcNetEQ_CrossCorr(pw32_corr, pw16_decodedLB, pw16_expandedLB, 40,
+ (WebRtc_Word16) w16_stopPos, w16_tmp, 1);
+
+ /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+ WebRtcSpl_MemSetW16(pw16_corrVec, 0, (4 + 60 + 4));
+ w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_stopPos);
+ w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_stopPos, pw32_corr, w16_tmp);
+
+ /* Calculate allowed starting point for peak finding.
+ The peak location bestIndex must fulfill two criteria:
+ (1) w16_bestIndex+w16_decodedLen < inst->timestampsPerCall+inst->ExpandInst.w16_overlap
+ (2) w16_bestIndex+w16_decodedLen < w16_startPos */
+ w16_tmp = WEBRTC_SPL_MAX(0, WEBRTC_SPL_MAX(w16_startPos,
+ inst->timestampsPerCall+inst->ExpandInst.w16_overlap) - w16_decodedLen);
+ /* Downscale starting index to 4kHz domain */
+ w16_tmp2 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) w16_tmp,
+ (WebRtc_Word16) (fs_mult << 1));
+
+#ifdef NETEQ_STEREO
+ } /* end if (msInfo->msMode != NETEQ_SLAVE) */
+
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+ /* This is master or mono instance; find peak */
+ WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex,
+ &w16_bestVal);
+ w16_bestIndex += w16_tmp; /* compensate for modified starting index */
+ msInfo->bestIndex = w16_bestIndex;
+ }
+ else if (msInfo->msMode == NETEQ_SLAVE)
+ {
+ /* Get peak location from master instance */
+ w16_bestIndex = msInfo->bestIndex;
+ }
+ else
+ {
+ /* Invalid mode */
+ return MASTER_SLAVE_ERROR;
+ }
+
+#else /* NETEQ_STEREO */
+
+ /* Find peak */
+ WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex,
+ &w16_bestVal);
+ w16_bestIndex += w16_tmp; /* compensate for modified starting index */
+
+#endif /* NETEQ_STEREO */
+
+ /*
+ * Ensure that underrun does not occur for 10ms case => we have to get at least
+ * 10ms + overlap . (This should never happen thanks to the above modification of
+ * peak-finding starting point.)
+ * */
+ while ((w16_bestIndex + w16_decodedLen) < (inst->timestampsPerCall
+ + inst->ExpandInst.w16_overlap) || w16_bestIndex + w16_decodedLen < w16_startPos)
+ {
+ w16_bestIndex += w16_newLen; /* Jump one lag ahead */
+ }
+ pw16_decodedOut = pw16_outData + w16_bestIndex;
+
+ /* Mute the new decoded data if needed (and unmute it linearly) */
+ w16_interpLen = WEBRTC_SPL_MIN(60*fs_mult,
+ w16_expandedLen-w16_bestIndex); /* this is the overlapping part of pw16_expanded */
+ w16_interpLen = WEBRTC_SPL_MIN(w16_interpLen, w16_decodedLen);
+ w16_inc = WebRtcSpl_DivW32W16ResW16(4194,
+ fs_mult); /* in Q20, 0.004 for NB and 0.002 for WB */
+ if (inst->w16_muteFactor < 16384)
+ {
+ WebRtcNetEQ_UnmuteSignal(pw16_decoded, &inst->w16_muteFactor, pw16_decoded, w16_inc,
+ (WebRtc_Word16) w16_interpLen);
+ WebRtcNetEQ_UnmuteSignal(&pw16_decoded[w16_interpLen], &inst->w16_muteFactor,
+ &pw16_decodedOut[w16_interpLen], w16_inc,
+ (WebRtc_Word16) (w16_decodedLen - w16_interpLen));
+ }
+ else
+ {
+ /* No muting needed */
+
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_decodedOut[w16_interpLen], &pw16_decoded[w16_interpLen],
+ (w16_decodedLen-w16_interpLen));
+ }
+
+ /* Do overlap and interpolate linearly */
+ w16_inc = WebRtcSpl_DivW32W16ResW16(16384, (WebRtc_Word16) (w16_interpLen + 1)); /* Q14 */
+ w16_startfact = (16384 - w16_inc);
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_expanded, w16_bestIndex);
+ WebRtcNetEQ_MixVoiceUnvoice(pw16_decodedOut, &pw16_expanded[w16_bestIndex], pw16_decoded,
+ &w16_startfact, w16_inc, w16_interpLen);
+
+ inst->w16_mode = MODE_MERGE;
+ inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
+
+ /* New added length (w16_startPos samples were borrowed) */
+ *pw16_len = w16_bestIndex + w16_decodedLen - w16_startPos;
+
+ /* Update VQmon parameter */
+ inst->w16_concealedTS += (*pw16_len - w16_decodedLen);
+ inst->w16_concealedTS = WEBRTC_SPL_MAX(0, inst->w16_concealedTS);
+
+ /* Update in-call and post-call statistics */
+ if (inst->ExpandInst.w16_expandMuteFactor == 0)
+ {
+ /* expansion generates noise only */
+ inst->statInst.expandedNoiseSamples += (*pw16_len - w16_decodedLen);
+ }
+ else
+ {
+ /* expansion generates more than only noise */
+ inst->statInst.expandedVoiceSamples += (*pw16_len - w16_decodedLen);
+ }
+ inst->statInst.expandLength += (*pw16_len - w16_decodedLen);
+
+
+ /* Copy back the first part of the data to the speechHistory */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition], pw16_outData, w16_startPos);
+
+
+ /* Move data to within outData */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, &pw16_outData[w16_startPos], (*pw16_len));
+
+ return 0;
+}
+
+#undef SCRATCH_pw16_expanded
+#undef SCRATCH_pw16_expandedLB
+#undef SCRATCH_pw16_decodedLB
+#undef SCRATCH_pw32_corr
+#undef SCRATCH_pw16_corrVec
+#undef SCRATCH_NETEQ_EXPAND
diff --git a/trunk/src/modules/audio_coding/neteq/min_distortion.c b/trunk/src/modules/audio_coding/neteq/min_distortion.c
new file mode 100644
index 0000000..4c9ee1c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/min_distortion.c
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Calculate best overlap fit according to distortion measure.
+ */
+
+#include "dsp_helpfunctions.h"
+
+#include "signal_processing_library.h"
+
+WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
+ WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
+ WebRtc_Word16 len, WebRtc_Word32 *pw16_dist)
+{
+ int i, j;
+ const WebRtc_Word16 *pw16_data1;
+ const WebRtc_Word16 *pw16_data2;
+ WebRtc_Word32 w32_diff;
+ WebRtc_Word32 w32_sumdiff;
+ WebRtc_Word16 bestIndex = -1;
+ WebRtc_Word32 minDist = WEBRTC_SPL_WORD32_MAX;
+
+ for (i = w16_minLag; i <= w16_maxLag; i++)
+ {
+ w32_sumdiff = 0;
+ pw16_data1 = pw16_data;
+ pw16_data2 = pw16_data - i;
+
+ for (j = 0; j < len; j++)
+ {
+ w32_diff = pw16_data1[j] - pw16_data2[j];
+ w32_sumdiff += WEBRTC_SPL_ABS_W32(w32_diff);
+ }
+
+ /* Compare with previous minimum */
+ if (w32_sumdiff < minDist)
+ {
+ minDist = w32_sumdiff;
+ bestIndex = i;
+ }
+ }
+
+ *pw16_dist = minDist;
+
+ return bestIndex;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c b/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c
new file mode 100644
index 0000000..9895630
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This function mixes a voiced signal with an unvoiced signal and
+ * updates the weight on a sample by sample basis.
+ */
+
+#include "dsp_helpfunctions.h"
+
+#include "signal_processing_library.h"
+
+void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
+ WebRtc_Word16 *pw16_unvoicedVec,
+ WebRtc_Word16 *w16_current_vfraction,
+ WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N)
+{
+ int i;
+ WebRtc_Word16 w16_tmp2;
+ WebRtc_Word16 vfraction = *w16_current_vfraction;
+
+ w16_tmp2 = 16384 - vfraction;
+ for (i = 0; i < N; i++)
+ {
+ pw16_outData[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL_16_16(vfraction, pw16_voicedVec[i]) +
+ WEBRTC_SPL_MUL_16_16(w16_tmp2, pw16_unvoicedVec[i]) + 8192,
+ 14);
+ vfraction -= w16_vfraction_change;
+ w16_tmp2 += w16_vfraction_change;
+ }
+ *w16_current_vfraction = vfraction;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/mute_signal.c b/trunk/src/modules/audio_coding/neteq/mute_signal.c
new file mode 100644
index 0000000..ee899cf
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/mute_signal.c
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This function mutes a signal linearly on a sample by sample basis.
+ */
+
+#include "dsp_helpfunctions.h"
+
+#include "signal_processing_library.h"
+
+void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
+ WebRtc_Word16 N)
+{
+ int i;
+ WebRtc_Word32 w32_tmp = 1048608; /* (16384<<6 + 32) */
+
+ for (i = 0; i < N; i++)
+ {
+ pw16_inout[i]
+ = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16((WebRtc_Word16)(w32_tmp>>6), pw16_inout[i])
+ + 8192) >> 14);
+ w32_tmp -= muteSlope;
+ }
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/neteq.gypi b/trunk/src/modules/audio_coding/neteq/neteq.gypi
new file mode 100644
index 0000000..c72efdd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/neteq.gypi
@@ -0,0 +1,289 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'NetEq',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'CNG',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'defines': [
+ 'NETEQ_VOICEENGINE_CODECS', # TODO: Should create a Chrome define which
+ 'SCRATCH', # specifies a subset of codecs to support.
+ ],
+ 'include_dirs': [
+ 'interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'interface',
+ ],
+ },
+ 'sources': [
+ 'interface/webrtc_neteq.h',
+ 'interface/webrtc_neteq_help_macros.h',
+ 'interface/webrtc_neteq_internal.h',
+ 'accelerate.c',
+ 'automode.c',
+ 'automode.h',
+ 'bgn_update.c',
+ 'buffer_stats.h',
+ 'bufstats_decision.c',
+ 'cng_internal.c',
+ 'codec_db.c',
+ 'codec_db.h',
+ 'codec_db_defines.h',
+ 'correlator.c',
+ 'delay_logging.h',
+ 'dsp.c',
+ 'dsp.h',
+ 'dsp_helpfunctions.c',
+ 'dsp_helpfunctions.h',
+ 'dtmf_buffer.c',
+ 'dtmf_buffer.h',
+ 'dtmf_tonegen.c',
+ 'dtmf_tonegen.h',
+ 'expand.c',
+ 'mcu.h',
+ 'mcu_address_init.c',
+ 'mcu_dsp_common.c',
+ 'mcu_dsp_common.h',
+ 'mcu_reset.c',
+ 'merge.c',
+ 'min_distortion.c',
+ 'mix_voice_unvoice.c',
+ 'mute_signal.c',
+ 'neteq_defines.h',
+ 'neteq_error_codes.h',
+ 'neteq_statistics.h',
+ 'normal.c',
+ 'packet_buffer.c',
+ 'packet_buffer.h',
+ 'peak_detection.c',
+ 'preemptive_expand.c',
+ 'random_vector.c',
+ 'recin.c',
+ 'recout.c',
+ 'rtcp.c',
+ 'rtcp.h',
+ 'rtp.c',
+ 'rtp.h',
+ 'set_fs.c',
+ 'signal_mcu.c',
+ 'split_and_insert.c',
+ 'unmute_signal.c',
+ 'webrtc_neteq.c',
+ ],
+ },
+ ], # targets
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'neteq_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEq',
+ 'NetEqTestTools',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'webrtc_neteq_unittest.cc',
+ ],
+ }, # neteq_unittests
+ {
+ 'target_name': 'NetEqRTPplay',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEq', # NetEQ library defined above
+ 'NetEqTestTools',# Test helpers
+ 'G711',
+ 'G722',
+ 'PCM16B',
+ 'iLBC',
+ 'iSAC',
+ 'CNG',
+ ],
+ 'defines': [
+ # TODO: Make codec selection conditional on definitions in target NetEq
+ 'CODEC_ILBC',
+ 'CODEC_PCM16B',
+ 'CODEC_G711',
+ 'CODEC_G722',
+ 'CODEC_ISAC',
+ 'CODEC_PCM16B_WB',
+ 'CODEC_ISAC_SWB',
+ 'CODEC_PCM16B_32KHZ',
+ 'CODEC_CNGCODEC8',
+ 'CODEC_CNGCODEC16',
+ 'CODEC_CNGCODEC32',
+ 'CODEC_ATEVENT_DECODE',
+ 'CODEC_RED',
+ ],
+ 'include_dirs': [
+ '.',
+ 'test',
+ ],
+ 'sources': [
+ 'test/NetEqRTPplay.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPencode',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEqTestTools',# Test helpers
+ 'G711',
+ 'G722',
+ 'PCM16B',
+ 'iLBC',
+ 'iSAC',
+ 'CNG',
+ '<(webrtc_root)/common_audio/common_audio.gyp:vad',
+ ],
+ 'defines': [
+ # TODO: Make codec selection conditional on definitions in target NetEq
+ 'CODEC_ILBC',
+ 'CODEC_PCM16B',
+ 'CODEC_G711',
+ 'CODEC_G722',
+ 'CODEC_ISAC',
+ 'CODEC_PCM16B_WB',
+ 'CODEC_ISAC_SWB',
+ 'CODEC_PCM16B_32KHZ',
+ 'CODEC_CNGCODEC8',
+ 'CODEC_CNGCODEC16',
+ 'CODEC_CNGCODEC32',
+ 'CODEC_ATEVENT_DECODE',
+ 'CODEC_RED',
+ ],
+ 'include_dirs': [
+ 'interface',
+ 'test',
+ ],
+ 'sources': [
+ 'test/RTPencode.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPjitter',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'test/RTPjitter.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPanalyze',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEqTestTools',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'test/RTPanalyze.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPchange',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEqTestTools',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'test/RTPchange.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPtimeshift',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEqTestTools',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'test/RTPtimeshift.cc',
+ ],
+ },
+ {
+ 'target_name': 'RTPcat',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEqTestTools',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'test/RTPcat.cc',
+ ],
+ },
+ {
+ 'target_name': 'NetEqTestTools',
+ # Collection of useful functions used in other tests
+ 'type': '<(library)',
+ 'dependencies': [
+ 'G711',
+ 'G722',
+ 'PCM16B',
+ 'iLBC',
+ 'iSAC',
+ 'CNG',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'interface',
+ 'test',
+ ],
+ },
+ 'defines': [
+ # TODO: Make codec selection conditional on definitions in target NetEq
+ 'CODEC_ILBC',
+ 'CODEC_PCM16B',
+ 'CODEC_G711',
+ 'CODEC_G722',
+ 'CODEC_ISAC',
+ 'CODEC_PCM16B_WB',
+ 'CODEC_ISAC_SWB',
+ 'CODEC_PCM16B_32KHZ',
+ 'CODEC_CNGCODEC8',
+ 'CODEC_CNGCODEC16',
+ 'CODEC_CNGCODEC32',
+ 'CODEC_ATEVENT_DECODE',
+ 'CODEC_RED',
+ ],
+ 'include_dirs': [
+ 'interface',
+ 'test',
+ ],
+ 'sources': [
+ 'test/NETEQTEST_NetEQClass.cc',
+ 'test/NETEQTEST_RTPpacket.cc',
+ 'test/NETEQTEST_CodecClass.cc',
+ 'test/NETEQTEST_NetEQClass.h',
+ 'test/NETEQTEST_RTPpacket.h',
+ 'test/NETEQTEST_CodecClass.h',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_defines.h b/trunk/src/modules/audio_coding/neteq/neteq_defines.h
new file mode 100644
index 0000000..318e6bb
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/neteq_defines.h
@@ -0,0 +1,356 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*****************************************************************************************
+ *
+ * Compilation flags in NetEQ:
+ *
+ *****************************************************************************************
+ *
+ ***** Platform flags ******
+ *
+ * SCRATCH Run NetEQ with "Scratch memory" to save some stack memory.
+ * Definition can be used on all platforms
+ *
+ ***** Summary flags ******
+ *
+ * NETEQ_ALL_SPECIAL_CODECS Add support for special codecs (CN/RED/DTMF)
+ *
+ * NETEQ_ALL_NB_CODECS Add support for all NB codecs (except CN/RED/DTMF)
+ *
+ * NETEQ_ALL_WB_CODECS Add support for all WB codecs (except CN/RED/DTMF)
+ *
+ * NETEQ_VOICEENGINE_CODECS Support for all NB, WB and SWB32 codecs and CN, RED and DTMF
+ *
+ * NETEQ_ALL_CODECS Support for all NB, WB, SWB 32kHz and SWB 48kHz as well as
+ * CN, RED and DTMF
+ *
+ ***** Sampling frequency ******
+ * (Note: usually not needed when Summary flags are used)
+ *
+ * NETEQ_WIDEBAND Wideband enabled
+ *
+ * NETEQ_32KHZ_WIDEBAND Super wideband @ 32kHz enabled
+ *
+ * NETEQ_48KHZ_WIDEBAND Super wideband @ 48kHz enabled
+ *
+ ***** Special Codec ******
+ * (Note: not needed if NETEQ_ALL_CODECS is used)
+ *
+ * NETEQ_RED_CODEC With this flag you enable NetEQ to understand redundancy in
+ * the RTP. NetEQ will use the redundancy if it's the same
+ * codec
+ *
+ * NETEQ_CNG_CODEC Enable DTX with the CN payload
+ *
+ * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone
+ *
+ ***** Speech Codecs *****
+ * (Note: Not needed if Summary flags are used)
+ *
+ * NETEQ_G711_CODEC Enable G.711 u- and A-law
+ *
+ * NETEQ_PCM16B_CODEC Enable uncompressed 16-bit
+ *
+ * NETEQ_ILBC_CODEC Enable iLBC
+ *
+ * NETEQ_ISAC_CODEC Enable iSAC
+ *
+ * NETEQ_ISAC_SWB_CODEC Enable iSAC-SWB
+ *
+ * NETEQ_G722_CODEC Enable G.722
+ *
+ * NETEQ_G729_CODEC Enable G.729
+ *
+ * NETEQ_G729_1_CODEC Enable G.729.1
+ *
+ * NETEQ_G726_CODEC Enable G.726
+ *
+ * NETEQ_G722_1_CODEC Enable G722.1
+ *
+ * NETEQ_G722_1C_CODEC Enable G722.1 Annex C
+ *
+ * NETEQ_SPEEX_CODEC Enable Speex (at 8 and 16 kHz sample rate)
+ *
+ * NETEQ_CELT_CODEC Enable Celt (at 32 kHz sample rate)
+ *
+ * NETEQ_GSMFR_CODEC Enable GSM-FR
+ *
+ * NETEQ_AMR_CODEC Enable AMR (narrowband)
+ *
+ * NETEQ_AMRWB_CODEC Enable AMR-WB
+ *
+ * NETEQ_CNG_CODEC Enable DTX with the CNG payload
+ *
+ * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone
+ *
+ ***** Test flags ******
+ *
+ * WEBRTC_NETEQ_40BITACC_TEST Run NetEQ with simulated 40-bit accumulator to run
+ * bit-exact to a DSP implementation where the main (splib
+ * and NetEQ) functions have been 40-bit optimized
+ *
+ *****************************************************************************************
+ */
+
+#if !defined NETEQ_DEFINES_H
+#define NETEQ_DEFINES_H
+
+/* Data block structure for MCU to DSP communication:
+ *
+ *
+ * First 3 16-bit words are pre-header that contains instructions and timestamp update
+ * Fourth 16-bit word is length of data block 1
+ * Rest is payload data
+ *
+ * 0 48 64 80
+ * -------------...----------------------------------------------------------------------
+ * | PreHeader ... | Length 1 | Payload data 1 ...... | Lenght 2| Data block 2.... | ...
+ * -------------...----------------------------------------------------------------------
+ *
+ *
+ * Preheader:
+ * 4 MSB can be either of:
+ */
+
+#define DSP_INSTR_NORMAL 0x1000
+/* Payload data will contain the encoded frames */
+
+#define DSP_INSTR_MERGE 0x2000
+/* Payload data block 1 will contain the encoded frame */
+/* Info block will contain the number of missing samples */
+
+#define DSP_INSTR_EXPAND 0x3000
+/* Payload data will be empty */
+
+#define DSP_INSTR_ACCELERATE 0x4000
+/* Payload data will contain the encoded frame */
+
+#define DSP_INSTR_DO_RFC3389CNG 0x5000
+/* Payload data will contain the SID frame if there is one*/
+
+#define DSP_INSTR_DTMF_GENERATE 0x6000
+/* Payload data will be one WebRtc_Word16 with the current DTMF value and one
+ * WebRtc_Word16 with the current volume value
+ */
+#define DSP_INSTR_NORMAL_ONE_DESC 0x7000
+/* No encoded frames */
+
+#define DSP_INSTR_DO_CODEC_INTERNAL_CNG 0x8000
+/* Codec has a built-in VAD/DTX scheme (use the above for "no transmission") */
+
+#define DSP_INSTR_PREEMPTIVE_EXPAND 0x9000
+/* Payload data will contain the encoded frames, if any */
+
+#define DSP_INSTR_DO_ALTERNATIVE_PLC 0xB000
+/* NetEQ switched off and packet missing... */
+
+#define DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS 0xC000
+/* NetEQ switched off and packet missing... */
+
+#define DSP_INSTR_DO_AUDIO_REPETITION 0xD000
+/* NetEQ switched off and packet missing... */
+
+#define DSP_INSTR_DO_AUDIO_REPETITION_INC_TS 0xE000
+/* NetEQ switched off and packet missing... */
+
+#define DSP_INSTR_FADE_TO_BGN 0xF000
+/* Exception handling: fade out to BGN (expand) */
+
+/*
+ * Next 4 bits signal additional data that needs to be transmitted
+ */
+
+#define DSP_CODEC_NO_CHANGE 0x0100
+#define DSP_CODEC_NEW_CODEC 0x0200
+#define DSP_CODEC_ADD_LATE_PKT 0x0300
+#define DSP_CODEC_RESET 0x0400
+#define DSP_DTMF_PAYLOAD 0x0010
+
+/*
+ * The most significant bit of the payload-length
+ * is used to flag whether the associated payload
+ * is redundant payload. This currently useful only for
+ * iSAC, where redundant payloads have to be treated
+ * differently. Every time the length is read it must be
+ * masked by DSP_CODEC_MASK_RED_FLAG to ignore the flag.
+ * Use DSP_CODEC_RED_FLAG to set or retrieve the flag.
+ */
+#define DSP_CODEC_MASK_RED_FLAG 0x7FFF
+#define DSP_CODEC_RED_FLAG 0x8000
+
+/*
+ * The first block of payload data consist of decode function pointers,
+ * and then the speech blocks.
+ *
+ */
+
+
+/*
+ * The playout modes that NetEq produced (i.e. gives more info about if the
+ * Accelerate was successful or not)
+ */
+
+#define MODE_NORMAL 0x0000
+#define MODE_EXPAND 0x0001
+#define MODE_MERGE 0x0002
+#define MODE_SUCCESS_ACCELERATE 0x0003
+#define MODE_UNSUCCESS_ACCELERATE 0x0004
+#define MODE_RFC3389CNG 0x0005
+#define MODE_LOWEN_ACCELERATE 0x0006
+#define MODE_DTMF 0x0007
+#define MODE_ONE_DESCRIPTOR 0x0008
+#define MODE_CODEC_INTERNAL_CNG 0x0009
+#define MODE_SUCCESS_PREEMPTIVE 0x000A
+#define MODE_UNSUCCESS_PREEMPTIVE 0x000B
+#define MODE_LOWEN_PREEMPTIVE 0x000C
+#define MODE_FADE_TO_BGN 0x000D
+
+#define MODE_ERROR 0x0010
+
+#define MODE_AWAITING_CODEC_PTR 0x0100
+
+#define MODE_BGN_ONLY 0x0200
+
+#define MODE_MASTER_DTMF_SIGNAL 0x0400
+
+#define MODE_USING_STEREO 0x0800
+
+
+
+/***********************/
+/* Group codec defines */
+/***********************/
+
+#if (defined(NETEQ_ALL_SPECIAL_CODECS))
+ #define NETEQ_CNG_CODEC
+ #define NETEQ_ATEVENT_DECODE
+ #define NETEQ_RED_CODEC
+ #define NETEQ_VAD
+ #define NETEQ_ARBITRARY_CODEC
+#endif
+
+#if (defined(NETEQ_ALL_NB_CODECS)) /* Except RED, DTMF and CNG */
+ #define NETEQ_PCM16B_CODEC
+ #define NETEQ_G711_CODEC
+ #define NETEQ_ILBC_CODEC
+ #define NETEQ_G729_CODEC
+ #define NETEQ_G726_CODEC
+ #define NETEQ_GSMFR_CODEC
+ #define NETEQ_AMR_CODEC
+#endif
+
+#if (defined(NETEQ_ALL_WB_CODECS)) /* Except RED, DTMF and CNG */
+ #define NETEQ_ISAC_CODEC
+ #define NETEQ_G722_CODEC
+ #define NETEQ_G722_1_CODEC
+ #define NETEQ_G729_1_CODEC
+ #define NETEQ_SPEEX_CODEC
+ #define NETEQ_AMRWB_CODEC
+ #define NETEQ_WIDEBAND
+#endif
+
+#if (defined(NETEQ_ALL_WB32_CODECS)) /* AAC, RED, DTMF and CNG */
+ #define NETEQ_ISAC_SWB_CODEC
+ #define NETEQ_32KHZ_WIDEBAND
+ #define NETEQ_G722_1C_CODEC
+ #define NETEQ_CELT_CODEC
+#endif
+
+#if (defined(NETEQ_VOICEENGINE_CODECS))
+ /* Special codecs */
+ #define NETEQ_CNG_CODEC
+ #define NETEQ_ATEVENT_DECODE
+ #define NETEQ_RED_CODEC
+ #define NETEQ_VAD
+ #define NETEQ_ARBITRARY_CODEC
+
+ /* Narrowband codecs */
+ #define NETEQ_PCM16B_CODEC
+ #define NETEQ_G711_CODEC
+ #define NETEQ_ILBC_CODEC
+ #define NETEQ_AMR_CODEC
+ #define NETEQ_G729_CODEC
+ #define NETEQ_GSMFR_CODEC
+
+ /* Wideband codecs */
+ #define NETEQ_WIDEBAND
+ #define NETEQ_ISAC_CODEC
+ #define NETEQ_G722_CODEC
+ #define NETEQ_G722_1_CODEC
+ #define NETEQ_G729_1_CODEC
+ #define NETEQ_AMRWB_CODEC
+ #define NETEQ_SPEEX_CODEC
+
+ /* Super wideband 32kHz codecs */
+ #define NETEQ_ISAC_SWB_CODEC
+ #define NETEQ_32KHZ_WIDEBAND
+ #define NETEQ_G722_1C_CODEC
+ #define NETEQ_CELT_CODEC
+
+#endif
+
+#if (defined(NETEQ_ALL_CODECS))
+ /* Special codecs */
+ #define NETEQ_CNG_CODEC
+ #define NETEQ_ATEVENT_DECODE
+ #define NETEQ_RED_CODEC
+ #define NETEQ_VAD
+ #define NETEQ_ARBITRARY_CODEC
+
+ /* Narrowband codecs */
+ #define NETEQ_PCM16B_CODEC
+ #define NETEQ_G711_CODEC
+ #define NETEQ_ILBC_CODEC
+ #define NETEQ_G729_CODEC
+ #define NETEQ_G726_CODEC
+ #define NETEQ_GSMFR_CODEC
+ #define NETEQ_AMR_CODEC
+
+ /* Wideband codecs */
+ #define NETEQ_WIDEBAND
+ #define NETEQ_ISAC_CODEC
+ #define NETEQ_G722_CODEC
+ #define NETEQ_G722_1_CODEC
+ #define NETEQ_G729_1_CODEC
+ #define NETEQ_SPEEX_CODEC
+ #define NETEQ_AMRWB_CODEC
+
+ /* Super wideband 32kHz codecs */
+ #define NETEQ_ISAC_SWB_CODEC
+ #define NETEQ_32KHZ_WIDEBAND
+ #define NETEQ_G722_1C_CODEC
+ #define NETEQ_CELT_CODEC
+
+ /* Super wideband 48kHz codecs */
+ #define NETEQ_48KHZ_WIDEBAND
+#endif
+
+/* Max output size from decoding one frame */
+#if defined(NETEQ_48KHZ_WIDEBAND)
+ #define NETEQ_MAX_FRAME_SIZE 2880 /* 60 ms super wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 3600 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
+#elif defined(NETEQ_32KHZ_WIDEBAND)
+ #define NETEQ_MAX_FRAME_SIZE 1920 /* 60 ms super wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 2400 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
+#elif defined(NETEQ_WIDEBAND)
+ #define NETEQ_MAX_FRAME_SIZE 960 /* 60 ms wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 1200 /* 60+15 ms wideband (60 ms decoded + 10 ms for merge overlap) */
+#else
+ #define NETEQ_MAX_FRAME_SIZE 480 /* 60 ms narrowband */
+ #define NETEQ_MAX_OUTPUT_SIZE 600 /* 60+15 ms narrowband (60 ms decoded + 10 ms for merge overlap) */
+#endif
+
+
+/* Enable stereo */
+#define NETEQ_STEREO
+
+#endif /* #if !defined NETEQ_DEFINES_H */
+
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h b/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h
new file mode 100644
index 0000000..1ce4680
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Definition of error codes.
+ *
+ * NOTE: When modifying the error codes,
+ * also modify the function WebRtcNetEQ_GetErrorCode!
+ */
+
+#if !defined NETEQ_ERROR_CODES_H
+#define NETEQ_ERROR_CODES_H
+
+/* Misc Error */
+#define NETEQ_OTHER_ERROR -1000
+
+/* Misc Recout Errors */
+#define FAULTY_INSTRUCTION -1001
+#define FAULTY_NETWORK_TYPE -1002
+#define FAULTY_DELAYVALUE -1003
+#define FAULTY_PLAYOUTMODE -1004
+#define CORRUPT_INSTANCE -1005
+#define ILLEGAL_MASTER_SLAVE_SWITCH -1006
+#define MASTER_SLAVE_ERROR -1007
+
+/* Misc Recout problems */
+#define UNKNOWN_BUFSTAT_DECISION -2001
+#define RECOUT_ERROR_DECODING -2002
+#define RECOUT_ERROR_SAMPLEUNDERRUN -2003
+#define RECOUT_ERROR_DECODED_TOO_MUCH -2004
+
+/* Misc RecIn problems */
+#define RECIN_CNG_ERROR -3001
+#define RECIN_UNKNOWNPAYLOAD -3002
+#define RECIN_BUFFERINSERT_ERROR -3003
+
+/* PBUFFER/BUFSTAT ERRORS */
+#define PBUFFER_INIT_ERROR -4001
+#define PBUFFER_INSERT_ERROR1 -4002
+#define PBUFFER_INSERT_ERROR2 -4003
+#define PBUFFER_INSERT_ERROR3 -4004
+#define PBUFFER_INSERT_ERROR4 -4005
+#define PBUFFER_INSERT_ERROR5 -4006
+#define UNKNOWN_G723_HEADER -4007
+#define PBUFFER_NONEXISTING_PACKET -4008
+#define PBUFFER_NOT_INITIALIZED -4009
+#define AMBIGUOUS_ILBC_FRAME_SIZE -4010
+
+/* CODEC DATABASE ERRORS */
+#define CODEC_DB_FULL -5001
+#define CODEC_DB_NOT_EXIST1 -5002
+#define CODEC_DB_NOT_EXIST2 -5003
+#define CODEC_DB_NOT_EXIST3 -5004
+#define CODEC_DB_NOT_EXIST4 -5005
+#define CODEC_DB_UNKNOWN_CODEC -5006
+#define CODEC_DB_PAYLOAD_TAKEN -5007
+#define CODEC_DB_UNSUPPORTED_CODEC -5008
+#define CODEC_DB_UNSUPPORTED_FS -5009
+
+/* DTMF ERRORS */
+#define DTMF_DEC_PARAMETER_ERROR -6001
+#define DTMF_INSERT_ERROR -6002
+#define DTMF_GEN_UNKNOWN_SAMP_FREQ -6003
+#define DTMF_NOT_SUPPORTED -6004
+
+/* RTP/PACKET ERRORS */
+#define RED_SPLIT_ERROR1 -7001
+#define RED_SPLIT_ERROR2 -7002
+#define RTP_TOO_SHORT_PACKET -7003
+#define RTP_CORRUPT_PACKET -7004
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_statistics.h b/trunk/src/modules/audio_coding/neteq/neteq_statistics.h
new file mode 100644
index 0000000..d07f330
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/neteq_statistics.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Definitions of statistics data structures for MCU and DSP sides.
+ */
+
+#include "typedefs.h"
+
+#ifndef NETEQ_STATISTICS_H
+#define NETEQ_STATISTICS_H
+
+/*
+ * Statistics struct on DSP side
+ */
+typedef struct
+{
+
+ /* variables for in-call statistics; queried through WebRtcNetEQ_GetNetworkStatistics */
+ WebRtc_UWord32 expandLength; /* number of samples produced through expand */
+ WebRtc_UWord32 preemptiveLength; /* number of samples produced through pre-emptive
+ expand */
+ WebRtc_UWord32 accelerateLength; /* number of samples removed through accelerate */
+
+ /* variables for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics */
+ WebRtc_UWord32 expandedVoiceSamples; /* number of voice samples produced through expand */
+ WebRtc_UWord32 expandedNoiseSamples; /* number of noise (background) samples produced
+ through expand */
+
+} DSPStats_t;
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/neteq/normal.c b/trunk/src/modules/audio_coding/neteq/normal.c
new file mode 100644
index 0000000..b33940a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/normal.c
@@ -0,0 +1,279 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the function for handling "normal" speech operation.
+ */
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_expanded 125*fs/8000 0 125*fs/8000-1
+
+ func WebRtcNetEQ_Expand 40+370*fs/8000 125*fs/8000 39+495*fs/8000
+
+ Total: 40+495*fs/8000
+ */
+
+#define SCRATCH_PW16_EXPANDED 0
+#if (defined(NETEQ_48KHZ_WIDEBAND))
+#define SCRATCH_NETEQ_EXPAND 756
+#elif (defined(NETEQ_32KHZ_WIDEBAND))
+#define SCRATCH_NETEQ_EXPAND 504
+#elif (defined(NETEQ_WIDEBAND))
+#define SCRATCH_NETEQ_EXPAND 252
+#else /* NB */
+#define SCRATCH_NETEQ_EXPAND 126
+#endif
+
+/****************************************************************************
+ * WebRtcNetEQ_Normal(...)
+ *
+ * This function has the possibility to modify data that is played out in Normal
+ * mode, for example adjust the gain of the signal. The length of the signal
+ * can not be changed.
+ *
+ * Input:
+ * - inst : NetEq instance, i.e. the user that requests more
+ * speech/audio data
+ * - scratchPtr : Pointer to scratch vector
+ * - decoded : Pointer to vector of new data from decoder
+ * (Vector contents may be altered by the function)
+ * - len : Number of input samples
+ *
+ * Output:
+ * - inst : Updated user information
+ * - outData : Pointer to a memory space where the output data
+ * should be stored
+ * - pw16_len : Pointer to variable where the number of samples
+ * produced will be written
+ *
+ * Return value : >=0 - Number of samples written to outData
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_Normal(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
+{
+
+ int i;
+ WebRtc_Word16 fs_mult;
+ WebRtc_Word16 fs_shift;
+ WebRtc_Word32 w32_En_speech;
+ WebRtc_Word16 enLen;
+ WebRtc_Word16 w16_muted;
+ WebRtc_Word16 w16_inc, w16_frac;
+ WebRtc_Word16 w16_tmp;
+ WebRtc_Word32 w32_tmp;
+
+ /* Sanity check */
+ if (len < 0)
+ {
+ /* Cannot have negative length of input vector */
+ return (-1);
+ }
+
+ if (len == 0)
+ {
+ /* Still got some data to play => continue with the same mode */
+ *pw16_len = len;
+ return (len);
+ }
+
+ fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
+ fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */
+
+ /*
+ * Check if last RecOut call resulted in an Expand or a FadeToBGN. If so, we have to take
+ * care of some cross-fading and unmuting.
+ */
+ if (inst->w16_mode == MODE_EXPAND || inst->w16_mode == MODE_FADE_TO_BGN)
+ {
+
+ /* Define memory where temporary result from Expand algorithm can be stored. */
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_PW16_EXPANDED;
+#else
+ WebRtc_Word16 pw16_expanded[FSMULT * 125];
+#endif
+ WebRtc_Word16 expandedLen = 0;
+ WebRtc_Word16 w16_decodedMax;
+
+ /* Find largest value in new data */
+ w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+
+ /* Generate interpolation data using Expand */
+ /* First, set Expand parameters to appropriate values. */
+ inst->ExpandInst.w16_lagsPosition = 0;
+ inst->ExpandInst.w16_lagsDirection = 0;
+ inst->ExpandInst.w16_stopMuting = 1; /* Do not mute signal any more */
+
+ /* Call Expand */
+ WebRtcNetEQ_Expand(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_expanded, &expandedLen, (WebRtc_Word16) (inst->w16_mode == MODE_FADE_TO_BGN));
+
+ inst->ExpandInst.w16_stopMuting = 0; /* Restore value */
+ inst->ExpandInst.w16_consecExp = 0; /* Last was not Expand any more */
+
+ /* Adjust muting factor (main muting factor times expand muting factor) */
+ if (inst->w16_mode == MODE_FADE_TO_BGN)
+ {
+ /* If last mode was FadeToBGN, the mute factor should be zero. */
+ inst->w16_muteFactor = 0;
+ }
+ else
+ {
+ /* w16_muteFactor * w16_expandMuteFactor */
+ inst->w16_muteFactor
+ = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
+ inst->ExpandInst.w16_expandMuteFactor, 14);
+ }
+
+ /* Adjust muting factor if needed (to BGN level) */
+ enLen = WEBRTC_SPL_MIN(fs_mult<<6, len); /* min( fs_mult * 64, len ) */
+ w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(
+ WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
+ w16_tmp = WEBRTC_SPL_MAX(w16_tmp, 0);
+ w32_En_speech = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, enLen, w16_tmp);
+ w32_En_speech = WebRtcSpl_DivW32W16(w32_En_speech, (WebRtc_Word16) (enLen >> w16_tmp));
+
+ if ((w32_En_speech != 0) && (w32_En_speech > inst->BGNInst.w32_energy))
+ {
+ /* Normalize new frame energy to 15 bits */
+ w16_tmp = WebRtcSpl_NormW32(w32_En_speech) - 16;
+ /* we want inst->BGNInst.energy/En_speech in Q14 */
+ w32_tmp = WEBRTC_SPL_SHIFT_W32(inst->BGNInst.w32_energy, (w16_tmp+14));
+ w16_tmp = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w32_En_speech, w16_tmp);
+ w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp, w16_tmp);
+ w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
+ WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) w16_tmp,
+ 14)); /* w16_muted in Q14 (sqrt(Q28)) */
+ }
+ else
+ {
+ w16_muted = 16384; /* 1.0 in Q14 */
+ }
+ if (w16_muted > inst->w16_muteFactor)
+ {
+ inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384);
+ }
+
+ /* If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14) */
+ w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult);
+ for (i = 0; i < len; i++)
+ {
+ /* scale with mute factor */
+ w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
+ /* shift 14 with proper rounding */
+ pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
+ /* increase mute_factor towards 16384 */
+ inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
+ }
+
+ /*
+ * Interpolate the expanded data into the new vector
+ * (NB/WB/SWB32/SWB40 8/16/32/32 samples)
+ */
+ fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */
+ w16_inc = 4 >> fs_shift;
+ w16_frac = w16_inc;
+ for (i = 0; i < 8 * fs_mult; i++)
+ {
+ pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
+ WEBRTC_SPL_MUL_16_16((32 - w16_frac), pw16_expanded[i]) + 8),
+ 5);
+ w16_frac += w16_inc;
+ }
+
+#ifdef NETEQ_CNG_CODEC
+ }
+ else if (inst->w16_mode==MODE_RFC3389CNG)
+ { /* previous was RFC 3389 CNG...*/
+ WebRtc_Word16 pw16_CngInterp[32];
+ /* Reset mute factor and start up fresh */
+ inst->w16_muteFactor = 16384;
+ if (inst->CNG_Codec_inst != NULL)
+ {
+ /* Generate long enough for 32kHz */
+ if(WebRtcCng_Generate(inst->CNG_Codec_inst,pw16_CngInterp, 32, 0)<0)
+ {
+ /* error returned; set return vector to all zeros */
+ WebRtcSpl_MemSetW16(pw16_CngInterp, 0, 32);
+ }
+ }
+ else
+ {
+ /*
+ * If no CNG instance is defined, just copy from the decoded data.
+ * (This will result in interpolating the decoded with itself.)
+ */
+ WEBRTC_SPL_MEMCPY_W16(pw16_CngInterp, pw16_decoded, fs_mult * 8);
+ }
+ /*
+ * Interpolate the CNG into the new vector
+ * (NB/WB/SWB32kHz/SWB48kHz 8/16/32/32 samples)
+ */
+ fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */
+ w16_inc = 4>>fs_shift;
+ w16_frac = w16_inc;
+ for (i = 0; i < 8 * fs_mult; i++)
+ {
+ pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
+ WEBRTC_SPL_MUL_16_16((32-w16_frac), pw16_CngInterp[i]) + 8),
+ 5);
+ w16_frac += w16_inc;
+ }
+#endif
+
+ }
+ else if (inst->w16_muteFactor < 16384)
+ {
+ /*
+ * Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are still
+ * ramping up from previous muting.
+ * If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14)
+ */
+ w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult);
+ for (i = 0; i < len; i++)
+ {
+ /* scale with mute factor */
+ w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
+ /* shift 14 with proper rounding */
+ pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
+ /* increase mute_factor towards 16384 */
+ inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
+ }
+ }
+
+ /* Copy data to other buffer */WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len);
+
+ inst->w16_mode = MODE_NORMAL;
+ *pw16_len = len;
+ return (len);
+
+}
+
+#undef SCRATCH_PW16_EXPANDED
+#undef SCRATCH_NETEQ_EXPAND
+
diff --git a/trunk/src/modules/audio_coding/neteq/packet_buffer.c b/trunk/src/modules/audio_coding/neteq/packet_buffer.c
new file mode 100644
index 0000000..8b9073c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/packet_buffer.c
@@ -0,0 +1,735 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of the actual packet buffer data structure.
+ */
+
+#include "packet_buffer.h"
+
+#include <string.h> /* to define NULL */
+
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+#ifdef NETEQ_DELAY_LOGGING
+/* special code for offline delay logging */
+#include "delay_logging.h"
+#include <stdio.h>
+
+extern FILE *delay_fid2; /* file pointer to delay log file */
+extern WebRtc_UWord32 tot_received_packets;
+#endif /* NETEQ_DELAY_LOGGING */
+
+
+int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
+ WebRtc_Word16 *pw16_memory, int memorySize)
+{
+ int i;
+ int pos = 0;
+
+ /* Sanity check */
+ if ((memorySize < PBUFFER_MIN_MEMORY_SIZE) || (pw16_memory == NULL)
+ || (maxNoOfPackets < 2) || (maxNoOfPackets > 600))
+ {
+ /* Invalid parameters */
+ return (PBUFFER_INIT_ERROR);
+ }
+
+ /* Clear the buffer instance */
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) bufferInst, 0,
+ sizeof(PacketBuf_t) / sizeof(WebRtc_Word16));
+
+ /* Clear the buffer memory */
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) pw16_memory, 0, memorySize);
+
+ /* Set maximum number of packets */
+ bufferInst->maxInsertPositions = maxNoOfPackets;
+
+ /* Initialize array pointers */
+ /* After each pointer has been set, the index pos is advanced to point immediately
+ * after the the recently allocated vector. Note that one step for the pos index
+ * corresponds to a WebRtc_Word16.
+ */
+
+ bufferInst->timeStamp = (WebRtc_UWord32*) &pw16_memory[pos];
+ pos += maxNoOfPackets << 1; /* advance maxNoOfPackets * WebRtc_UWord32 */
+
+ bufferInst->payloadLocation = (WebRtc_Word16**) &pw16_memory[pos];
+ pos += maxNoOfPackets * (sizeof(WebRtc_Word16*) / sizeof(WebRtc_Word16)); /* advance */
+
+ bufferInst->seqNumber = (WebRtc_UWord16*) &pw16_memory[pos];
+ pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_UWord16 */
+
+ bufferInst->payloadType = &pw16_memory[pos];
+ pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+
+ bufferInst->payloadLengthBytes = &pw16_memory[pos];
+ pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+
+ bufferInst->rcuPlCntr = &pw16_memory[pos];
+ pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
+
+ bufferInst->waitingTime = (int*) (&pw16_memory[pos]);
+ /* Advance maxNoOfPackets * sizeof(waitingTime element). */
+ pos += maxNoOfPackets *
+ sizeof(*bufferInst->waitingTime) / sizeof(*pw16_memory);
+
+ /* The payload memory starts after the slot arrays */
+ bufferInst->startPayloadMemory = &pw16_memory[pos];
+ bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
+ bufferInst->memorySizeW16 = (memorySize - pos); /* Remaining memory */
+
+ /* Initialize each payload slot as empty with infinite delay */
+ for (i = 0; i < bufferInst->maxInsertPositions; i++)
+ {
+ bufferInst->payloadType[i] = -1;
+ }
+
+ /* Reset buffer parameters */
+ bufferInst->numPacketsInBuffer = 0;
+ bufferInst->packSizeSamples = 0;
+ bufferInst->insertPosition = 0;
+
+ /* Reset buffer statistics */
+ bufferInst->discardedPackets = 0;
+
+ return (0);
+}
+
+
+int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst)
+{
+ int i;
+
+ /* Sanity check */
+ if (bufferInst->startPayloadMemory == NULL)
+ {
+ /* Packet buffer has not been initialized */
+ /* Don't do the flushing operation, since we do not
+ know the state of the struct variables */
+ return (0);
+ }
+
+ /* Set all payload lengths to zero */
+ WebRtcSpl_MemSetW16(bufferInst->payloadLengthBytes, 0, bufferInst->maxInsertPositions);
+
+ /* Reset buffer variables */
+ bufferInst->numPacketsInBuffer = 0;
+ bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
+ bufferInst->insertPosition = 0;
+
+ /* Clear all slots, starting with the last one */
+ for (i = (bufferInst->maxInsertPositions - 1); i >= 0; i--)
+ {
+ bufferInst->payloadType[i] = -1;
+ bufferInst->timeStamp[i] = 0;
+ bufferInst->seqNumber[i] = 0;
+ }
+
+ return (0);
+}
+
+
+int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
+ WebRtc_Word16 *flushed)
+{
+ int nextPos;
+ int i;
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ int temp_var;
+#endif /* NETEQ_DELAY_LOGGING */
+
+ /* Initialize to "no flush" */
+ *flushed = 0;
+
+ /* Sanity check */
+ if (bufferInst->startPayloadMemory == NULL)
+ {
+ /* packet buffer has not been initialized */
+ return (-1);
+ }
+
+ /* Sanity check for payload length
+ (payloadLen in bytes and memory size in WebRtc_Word16) */
+ if ((RTPpacket->payloadLen > (bufferInst->memorySizeW16 << 1)) || (RTPpacket->payloadLen
+ <= 0))
+ {
+ /* faulty or too long payload length */
+ return (-1);
+ }
+
+ /* Find a position in the buffer for this packet */
+ if (bufferInst->numPacketsInBuffer != 0)
+ {
+ /* Get the next slot */
+ bufferInst->insertPosition++;
+ if (bufferInst->insertPosition >= bufferInst->maxInsertPositions)
+ {
+ /* "Wrap around" and start from the beginning */
+ bufferInst->insertPosition = 0;
+ }
+
+ /* Check if there is enough space for the new packet */
+ if (bufferInst->currentMemoryPos + ((RTPpacket->payloadLen + 1) >> 1)
+ >= &bufferInst->startPayloadMemory[bufferInst->memorySizeW16])
+ {
+ WebRtc_Word16 *tempMemAddress;
+
+ /*
+ * Payload does not fit at the end of the memory, put it in the beginning
+ * instead
+ */
+ bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
+
+ /*
+ * Now, we must search for the next non-empty payload,
+ * finding the one with the lowest start address for the payload
+ */
+ tempMemAddress = &bufferInst->startPayloadMemory[bufferInst->memorySizeW16];
+ nextPos = -1;
+
+ /* Loop through all slots again */
+ for (i = 0; i < bufferInst->maxInsertPositions; i++)
+ {
+ /* Look for the non-empty slot with the lowest
+ payload location address */
+ if (bufferInst->payloadLengthBytes[i] != 0 && bufferInst->payloadLocation[i]
+ < tempMemAddress)
+ {
+ tempMemAddress = bufferInst->payloadLocation[i];
+ nextPos = i;
+ }
+ }
+
+ /* Check that we did find a previous payload */
+ if (nextPos == -1)
+ {
+ /* The buffer is corrupt => flush and return error */
+ WebRtcNetEQ_PacketBufferFlush(bufferInst);
+ *flushed = 1;
+ return (-1);
+ }
+ }
+ else
+ {
+ /* Payload fits at the end of memory. */
+
+ /* Find the next non-empty slot. */
+ nextPos = bufferInst->insertPosition + 1;
+
+ /* Increase nextPos until a non-empty slot is found or end of array is encountered*/
+ while ((bufferInst->payloadLengthBytes[nextPos] == 0) && (nextPos
+ < bufferInst->maxInsertPositions))
+ {
+ nextPos++;
+ }
+
+ if (nextPos == bufferInst->maxInsertPositions)
+ {
+ /*
+ * Reached the end of the array, so there must be a packet in the first
+ * position instead
+ */
+ nextPos = 0;
+
+ /* Increase nextPos until a non-empty slot is found */
+ while (bufferInst->payloadLengthBytes[nextPos] == 0)
+ {
+ nextPos++;
+ }
+ }
+ } /* end if-else */
+
+ /*
+ * Check if the new payload will extend into a payload later in memory.
+ * If so, the buffer is full.
+ */
+ if ((bufferInst->currentMemoryPos <= bufferInst->payloadLocation[nextPos])
+ && ((&bufferInst->currentMemoryPos[(RTPpacket->payloadLen + 1) >> 1])
+ > bufferInst->payloadLocation[nextPos]))
+ {
+ /* Buffer is full, so the buffer must be flushed */
+ WebRtcNetEQ_PacketBufferFlush(bufferInst);
+ *flushed = 1;
+ }
+
+ if (bufferInst->payloadLengthBytes[bufferInst->insertPosition] != 0)
+ {
+ /* All positions are already taken and entire buffer should be flushed */
+ WebRtcNetEQ_PacketBufferFlush(bufferInst);
+ *flushed = 1;
+ }
+
+ }
+ else
+ {
+ /* Buffer is empty, just insert the packet at the beginning */
+ bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
+ bufferInst->insertPosition = 0;
+ }
+
+ /* Insert packet in the found position */
+ if (RTPpacket->starts_byte1 == 0)
+ {
+ /* Payload is 16-bit aligned => just copy it */
+
+ WEBRTC_SPL_MEMCPY_W16(bufferInst->currentMemoryPos,
+ RTPpacket->payload, (RTPpacket->payloadLen + 1) >> 1);
+ }
+ else
+ {
+ /* Payload is not 16-bit aligned => align it during copy operation */
+ for (i = 0; i < RTPpacket->payloadLen; i++)
+ {
+ /* copy the (i+1)-th byte to the i-th byte */
+
+ WEBRTC_SPL_SET_BYTE(bufferInst->currentMemoryPos,
+ (WEBRTC_SPL_GET_BYTE(RTPpacket->payload, (i + 1))), i);
+ }
+ }
+
+ /* Copy the packet information */
+ bufferInst->payloadLocation[bufferInst->insertPosition] = bufferInst->currentMemoryPos;
+ bufferInst->payloadLengthBytes[bufferInst->insertPosition] = RTPpacket->payloadLen;
+ bufferInst->payloadType[bufferInst->insertPosition] = RTPpacket->payloadType;
+ bufferInst->seqNumber[bufferInst->insertPosition] = RTPpacket->seqNumber;
+ bufferInst->timeStamp[bufferInst->insertPosition] = RTPpacket->timeStamp;
+ bufferInst->rcuPlCntr[bufferInst->insertPosition] = RTPpacket->rcuPlCntr;
+ bufferInst->rcuPlCntr[bufferInst->insertPosition] = 0;
+ bufferInst->waitingTime[bufferInst->insertPosition] = 0;
+ /* Update buffer parameters */
+ bufferInst->numPacketsInBuffer++;
+ bufferInst->currentMemoryPos += (RTPpacket->payloadLen + 1) >> 1;
+
+#ifdef NETEQ_DELAY_LOGGING
+ /* special code for offline delay logging */
+ if (*flushed)
+ {
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_FLUSH;
+ fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
+ }
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_RECIN;
+ fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
+ fwrite( &RTPpacket->timeStamp, sizeof(WebRtc_UWord32), 1, delay_fid2 );
+ fwrite( &RTPpacket->seqNumber, sizeof(WebRtc_UWord16), 1, delay_fid2 );
+ fwrite( &RTPpacket->payloadType, sizeof(int), 1, delay_fid2 );
+ fwrite( &RTPpacket->payloadLen, sizeof(WebRtc_Word16), 1, delay_fid2 );
+ tot_received_packets++;
+#endif /* NETEQ_DELAY_LOGGING */
+
+ return (0);
+}
+
+
+int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket,
+ int bufferPosition, int *waitingTime)
+{
+
+ /* Sanity check */
+ if (bufferInst->startPayloadMemory == NULL)
+ {
+ /* packet buffer has not been initialized */
+ return (PBUFFER_NOT_INITIALIZED);
+ }
+
+ if (bufferPosition < 0 || bufferPosition >= bufferInst->maxInsertPositions)
+ {
+ /* buffer position is outside valid range */
+ return (NETEQ_OTHER_ERROR);
+ }
+
+ /* Check that there is a valid payload in the specified position */
+ if (bufferInst->payloadLengthBytes[bufferPosition] <= 0)
+ {
+ /* The position does not contain a valid payload */
+ RTPpacket->payloadLen = 0; /* Set zero length */
+ return (PBUFFER_NONEXISTING_PACKET); /* Return error */
+ }
+
+ /* Payload exists => extract payload data */
+
+ /* Copy the actual data payload to RTP packet struct */
+
+ WEBRTC_SPL_MEMCPY_W16((WebRtc_Word16*) RTPpacket->payload,
+ bufferInst->payloadLocation[bufferPosition],
+ (bufferInst->payloadLengthBytes[bufferPosition] + 1) >> 1); /*length in WebRtc_Word16*/
+
+ /* Copy payload parameters */
+ RTPpacket->payloadLen = bufferInst->payloadLengthBytes[bufferPosition];
+ RTPpacket->payloadType = bufferInst->payloadType[bufferPosition];
+ RTPpacket->seqNumber = bufferInst->seqNumber[bufferPosition];
+ RTPpacket->timeStamp = bufferInst->timeStamp[bufferPosition];
+ RTPpacket->rcuPlCntr = bufferInst->rcuPlCntr[bufferPosition];
+ *waitingTime = bufferInst->waitingTime[bufferPosition];
+ RTPpacket->starts_byte1 = 0; /* payload is 16-bit aligned */
+
+ /* Clear the position in the packet buffer */
+ bufferInst->payloadType[bufferPosition] = -1;
+ bufferInst->payloadLengthBytes[bufferPosition] = 0;
+ bufferInst->seqNumber[bufferPosition] = 0;
+ bufferInst->timeStamp[bufferPosition] = 0;
+ bufferInst->waitingTime[bufferPosition] = 0;
+ bufferInst->payloadLocation[bufferPosition] = bufferInst->startPayloadMemory;
+
+ /* Reduce packet counter with one */
+ bufferInst->numPacketsInBuffer--;
+
+ return (0);
+}
+
+int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst,
+ uint32_t current_time_stamp,
+ uint32_t* time_stamp,
+ int* buffer_position,
+ int erase_old_packets,
+ int16_t* payload_type) {
+ int32_t time_stamp_diff = WEBRTC_SPL_WORD32_MAX; /* Smallest diff found. */
+ int32_t new_diff;
+ int i;
+ int16_t rcu_payload_cntr;
+
+ if (buffer_inst->startPayloadMemory == NULL) {
+ /* Packet buffer has not been initialized. */
+ return PBUFFER_NOT_INITIALIZED;
+ }
+
+ /* Initialize all return values. */
+ *time_stamp = 0;
+ *payload_type = -1; /* Indicates that no packet was found. */
+ *buffer_position = -1; /* Indicates that no packet was found. */
+ rcu_payload_cntr = WEBRTC_SPL_WORD16_MAX; /* Indicates no packet found. */
+
+ /* Check if buffer is empty. */
+ if (buffer_inst->numPacketsInBuffer <= 0) {
+ return 0;
+ }
+
+ /* Loop through all slots in buffer. */
+ if (erase_old_packets) { /* If old payloads should be discarded. */
+ for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
+ /* Calculate difference between this slot and current_time_stamp. */
+ new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp);
+
+ /* Check if payload should be discarded. */
+ if ((new_diff < 0) /* Payload is too old */
+ && (new_diff > -30000) /* Account for TS wrap-around. */
+ && (buffer_inst->payloadLengthBytes[i] > 0)) { /* Payload exists. */
+ /* Throw away old packet. */
+
+ /* Clear the position in the buffer. */
+ buffer_inst->payloadType[i] = -1;
+ buffer_inst->payloadLengthBytes[i] = 0;
+
+ /* Reduce packet counter by one. */
+ buffer_inst->numPacketsInBuffer--;
+ /* Increase discard counter for in-call statistics. */
+ buffer_inst->discardedPackets++;
+ } else if (((new_diff < time_stamp_diff)
+ || ((new_diff == time_stamp_diff)
+ && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr)))
+ && (buffer_inst->payloadLengthBytes[i] > 0)) {
+ /* New diff is smaller than previous diffs or we have a candidate with a
+ * time stamp as previous candidate but better RCU-counter;
+ * and the payload exists.
+ */
+ /* Save this position as the best candidate. */
+ *buffer_position = i;
+ time_stamp_diff = new_diff;
+ *payload_type = buffer_inst->payloadType[i];
+ rcu_payload_cntr = buffer_inst->rcuPlCntr[i];
+ }
+ }
+ } else {
+ for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
+ /* Calculate difference between this slot and current_time_stamp. */
+ new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp);
+
+ /* Check if this is the oldest packet. */
+ if (((new_diff < time_stamp_diff)
+ || ((new_diff == time_stamp_diff)
+ && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr)))
+ && (buffer_inst->payloadLengthBytes[i] > 0)) {
+ /* New diff is smaller than previous diffs or we have a candidate with a
+ * time_stamp as previous candidate but better RCU-counter;
+ * and the payload exists.
+ */
+ /* Save this position as the best candidate. */
+ *buffer_position = i;
+ time_stamp_diff = new_diff;
+ *payload_type = buffer_inst->payloadType[i];
+ rcu_payload_cntr = buffer_inst->rcuPlCntr[i];
+ }
+ }
+ }
+
+ /* Check that we did find a real position. */
+ if (*buffer_position >= 0) {
+ /* Get the time_stamp for the best position. */
+ *time_stamp = buffer_inst->timeStamp[*buffer_position];
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t *bufferInst)
+{
+ int i, count;
+ WebRtc_Word32 sizeSamples;
+
+ count = 0;
+
+ /* Loop through all slots in the buffer */
+ for (i = 0; i < bufferInst->maxInsertPositions; i++)
+ {
+ /* Only count the packets with non-zero size */
+ if (bufferInst->payloadLengthBytes[i] != 0)
+ {
+ count++;
+ }
+ }
+
+ /*
+ * Calculate buffer size as number of packets times packet size
+ * (packet size is that of the latest decoded packet)
+ */
+ sizeSamples = WEBRTC_SPL_MUL_16_16(bufferInst->packSizeSamples, count);
+
+ /* Sanity check; size cannot be negative */
+ if (sizeSamples < 0)
+ {
+ sizeSamples = 0;
+ }
+
+ return sizeSamples;
+}
+
+void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst) {
+ int i;
+ /* Loop through all slots in the buffer. */
+ for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
+ /* Only increment waiting time for the packets with non-zero size. */
+ if (buffer_inst->payloadLengthBytes[i] != 0) {
+ buffer_inst->waitingTime[i]++;
+ }
+ }
+}
+
+int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
+ int noOfCodecs, int *maxBytes, int *maxSlots)
+{
+ int i;
+ int ok = 0;
+ WebRtc_Word16 w16_tmp;
+ WebRtc_Word16 codecBytes;
+ WebRtc_Word16 codecBuffers;
+
+ /* Initialize return variables to zero */
+ *maxBytes = 0;
+ *maxSlots = 0;
+
+ /* Loop through all codecs supplied to function */
+ for (i = 0; i < noOfCodecs; i++)
+ {
+ /* Find current codec and set parameters accordingly */
+
+ if (codecID[i] == kDecoderPCMu)
+ {
+ codecBytes = 1680; /* Up to 210ms @ 64kbps */
+ codecBuffers = 30; /* Down to 5ms frames */
+ }
+ else if (codecID[i] == kDecoderPCMa)
+ {
+ codecBytes = 1680; /* Up to 210ms @ 64kbps */
+ codecBuffers = 30; /* Down to 5ms frames */
+ }
+ else if (codecID[i] == kDecoderILBC)
+ {
+ codecBytes = 380; /* 200ms @ 15.2kbps (20ms frames) */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderISAC)
+ {
+ codecBytes = 960; /* 240ms @ 32kbps (60ms frames) */
+ codecBuffers = 8;
+ }
+ else if (codecID[i] == kDecoderISACswb)
+ {
+ codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */
+ codecBuffers = 8;
+ }
+ else if (codecID[i] == kDecoderPCM16B)
+ {
+ codecBytes = 3360; /* 210ms */
+ codecBuffers = 15;
+ }
+ else if (codecID[i] == kDecoderPCM16Bwb)
+ {
+ codecBytes = 6720; /* 210ms */
+ codecBuffers = 15;
+ }
+ else if (codecID[i] == kDecoderPCM16Bswb32kHz)
+ {
+ codecBytes = 13440; /* 210ms */
+ codecBuffers = 15;
+ }
+ else if (codecID[i] == kDecoderPCM16Bswb48kHz)
+ {
+ codecBytes = 20160; /* 210ms */
+ codecBuffers = 15;
+ }
+ else if (codecID[i] == kDecoderG722)
+ {
+ codecBytes = 1680; /* 210ms @ 64kbps */
+ codecBuffers = 15;
+ }
+ else if (codecID[i] == kDecoderRED)
+ {
+ codecBytes = 0; /* Should not be max... */
+ codecBuffers = 0;
+ }
+ else if (codecID[i] == kDecoderAVT)
+ {
+ codecBytes = 0; /* Should not be max... */
+ codecBuffers = 0;
+ }
+ else if (codecID[i] == kDecoderCNG)
+ {
+ codecBytes = 0; /* Should not be max... */
+ codecBuffers = 0;
+ }
+ else if (codecID[i] == kDecoderG729)
+ {
+ codecBytes = 210; /* 210ms @ 8kbps */
+ codecBuffers = 20; /* max 200ms supported for 10ms frames */
+ }
+ else if (codecID[i] == kDecoderG729_1)
+ {
+ codecBytes = 840; /* 210ms @ 32kbps */
+ codecBuffers = 10; /* max 200ms supported for 20ms frames */
+ }
+ else if (codecID[i] == kDecoderG726_16)
+ {
+ codecBytes = 400; /* 200ms @ 16kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG726_24)
+ {
+ codecBytes = 600; /* 200ms @ 24kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG726_32)
+ {
+ codecBytes = 800; /* 200ms @ 32kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG726_40)
+ {
+ codecBytes = 1000; /* 200ms @ 40kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1_16)
+ {
+ codecBytes = 420; /* 210ms @ 16kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1_24)
+ {
+ codecBytes = 630; /* 210ms @ 24kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1_32)
+ {
+ codecBytes = 840; /* 210ms @ 32kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1C_24)
+ {
+ codecBytes = 630; /* 210ms @ 24kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1C_32)
+ {
+ codecBytes = 840; /* 210ms @ 32kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderG722_1C_48)
+ {
+ codecBytes = 1260; /* 210ms @ 48kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderSPEEX_8)
+ {
+ codecBytes = 1250; /* 210ms @ 50kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderSPEEX_16)
+ {
+ codecBytes = 1250; /* 210ms @ 50kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderCELT_32)
+ {
+ codecBytes = 1250; /* 210ms @ 50kbps */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderGSMFR)
+ {
+ codecBytes = 340; /* 200ms */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderAMR)
+ {
+ codecBytes = 384; /* 240ms @ 12.2kbps+headers (60ms frames) */
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderAMRWB)
+ {
+ codecBytes = 744;
+ codecBuffers = 10;
+ }
+ else if (codecID[i] == kDecoderArbitrary)
+ {
+ codecBytes = 6720; /* Assume worst case uncompressed WB 210ms */
+ codecBuffers = 15;
+ }
+ else
+ {
+ /*Unknow codec */
+ codecBytes = 0;
+ codecBuffers = 0;
+ ok = CODEC_DB_UNKNOWN_CODEC;
+ }
+
+ /* Update max variables */
+ *maxBytes = WEBRTC_SPL_MAX((*maxBytes), codecBytes);
+ *maxSlots = WEBRTC_SPL_MAX((*maxSlots), codecBuffers);
+
+ } /* end of for loop */
+
+ /*
+ * Add size needed by the additional pointers for each slot inside struct,
+ * as indicated on each line below.
+ */
+ w16_tmp = (sizeof(WebRtc_UWord32) /* timeStamp */
+ + sizeof(WebRtc_Word16*) /* payloadLocation */
+ + sizeof(WebRtc_UWord16) /* seqNumber */
+ + sizeof(WebRtc_Word16) /* payloadType */
+ + sizeof(WebRtc_Word16) /* payloadLengthBytes */
+ + sizeof(WebRtc_Word16) /* rcuPlCntr */
+ + sizeof(int)); /* waitingTime */
+ /* Add the extra size per slot to the memory count */
+ *maxBytes += w16_tmp * (*maxSlots);
+
+ return ok;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/packet_buffer.h b/trunk/src/modules/audio_coding/neteq/packet_buffer.h
new file mode 100644
index 0000000..662f8af
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/packet_buffer.h
@@ -0,0 +1,220 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Interface for the actual packet buffer data structure.
+ */
+
+#ifndef PACKET_BUFFER_H
+#define PACKET_BUFFER_H
+
+#include "typedefs.h"
+
+#include "webrtc_neteq.h"
+#include "rtp.h"
+
+/* Define minimum allowed buffer memory, in 16-bit words */
+#define PBUFFER_MIN_MEMORY_SIZE 150
+
+/****************************/
+/* The packet buffer struct */
+/****************************/
+
+typedef struct
+{
+
+ /* Variables common to the entire buffer */
+ WebRtc_UWord16 packSizeSamples; /* packet size in samples of last decoded packet */
+ WebRtc_Word16 *startPayloadMemory; /* pointer to the payload memory */
+ int memorySizeW16; /* the size (in WebRtc_Word16) of the payload memory */
+ WebRtc_Word16 *currentMemoryPos; /* The memory position to insert next payload */
+ int numPacketsInBuffer; /* The number of packets in the buffer */
+ int insertPosition; /* The position to insert next packet */
+ int maxInsertPositions; /* Maximum number of packets allowed */
+
+ /* Arrays with one entry per packet slot */
+ /* NOTE: If these are changed, the changes must be accounted for at the end of
+ the function WebRtcNetEQ_GetDefaultCodecSettings(). */
+ WebRtc_UWord32 *timeStamp; /* Timestamp in slot n */
+ WebRtc_Word16 **payloadLocation; /* Memory location of payload in slot n */
+ WebRtc_UWord16 *seqNumber; /* Sequence number in slot n */
+ WebRtc_Word16 *payloadType; /* Payload type of packet in slot n */
+ WebRtc_Word16 *payloadLengthBytes; /* Payload length of packet in slot n */
+ WebRtc_Word16 *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload
+ 2 for redundant payload */
+ int *waitingTime;
+
+
+ /* Statistics counter */
+ WebRtc_UWord16 discardedPackets; /* Number of discarded packets */
+
+} PacketBuf_t;
+
+/*************************/
+/* Function declarations */
+/*************************/
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferInit(...)
+ *
+ * This function initializes the packet buffer.
+ *
+ * Input:
+ * - bufferInst : Buffer instance to be initialized
+ * - noOfPackets : Maximum number of packets that buffer should hold
+ * - memory : Pointer to the storage memory for the payloads
+ * - memorySize : The size of the payload memory (in WebRtc_Word16)
+ *
+ * Output:
+ * - bufferInst : Updated buffer instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
+ WebRtc_Word16 *pw16_memory, int memorySize);
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferFlush(...)
+ *
+ * This function flushes all the packets in the buffer.
+ *
+ * Input:
+ * - bufferInst : Buffer instance to be flushed
+ *
+ * Output:
+ * - bufferInst : Flushed buffer instance
+ *
+ * Return value : 0 - Ok
+ */
+
+int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst);
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferInsert(...)
+ *
+ * This function inserts an RTP packet into the packet buffer.
+ *
+ * Input:
+ * - bufferInst : Buffer instance
+ * - RTPpacket : An RTP packet struct (with payload, sequence
+ * number, etc.)
+ *
+ * Output:
+ * - bufferInst : Updated buffer instance
+ * - flushed : 1 if buffer was flushed, 0 otherwise
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
+ WebRtc_Word16 *flushed);
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferExtract(...)
+ *
+ * This function extracts a payload from the buffer.
+ *
+ * Input:
+ * - bufferInst : Buffer instance
+ * - bufferPosition: Position of the packet that should be extracted
+ *
+ * Output:
+ * - RTPpacket : An RTP packet struct (with payload, sequence
+ * number, etc)
+ * - bufferInst : Updated buffer instance
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket,
+ int bufferPosition, int *waitingTime);
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferFindLowestTimestamp(...)
+ *
+ * This function finds the next packet with the lowest timestamp.
+ *
+ * Input:
+ * - buffer_inst : Buffer instance.
+ * - current_time_stamp : The timestamp to compare packet timestamps with.
+ * - erase_old_packets : If non-zero, erase packets older than currentTS.
+ *
+ * Output:
+ * - time_stamp : Lowest timestamp that was found.
+ * - buffer_position : Position of this packet (-1 if there are no
+ * packets in the buffer).
+ * - payload_type : Payload type of the found payload.
+ *
+ * Return value : 0 - Ok;
+ * < 0 - Error.
+ */
+
+int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst,
+ uint32_t current_time_stamp,
+ uint32_t* time_stamp,
+ int* buffer_position,
+ int erase_old_packets,
+ int16_t* payload_type);
+
+/****************************************************************************
+ * WebRtcNetEQ_PacketBufferGetSize(...)
+ *
+ * Calculate and return an estimate of the total data length (in samples)
+ * currently in the buffer. The estimate is calculated as the number of
+ * packets currently in the buffer (which does not have any remaining waiting
+ * time), multiplied with the number of samples obtained from the last
+ * decoded packet.
+ *
+ * Input:
+ * - bufferInst : Buffer instance
+ *
+ * Return value : The buffer size in samples
+ */
+
+WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t *bufferInst);
+
+/****************************************************************************
+ * WebRtcNetEQ_IncrementWaitingTimes(...)
+ *
+ * Increment the waiting time for all packets in the buffer by one.
+ *
+ * Input:
+ * - bufferInst : Buffer instance
+ *
+ * Return value : n/a
+ */
+
+void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst);
+
+/****************************************************************************
+ * WebRtcNetEQ_GetDefaultCodecSettings(...)
+ *
+ * Calculates a recommended buffer size for a specific set of codecs.
+ *
+ * Input:
+ * - codecID : An array of codec types that will be used
+ * - noOfCodecs : Number of codecs in array codecID
+ *
+ * Output:
+ * - maxBytes : Recommended buffer memory size in bytes
+ * - maxSlots : Recommended number of slots in buffer
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
+ int noOfCodecs, int *maxBytes, int *maxSlots);
+
+#endif /* PACKET_BUFFER_H */
diff --git a/trunk/src/modules/audio_coding/neteq/peak_detection.c b/trunk/src/modules/audio_coding/neteq/peak_detection.c
new file mode 100644
index 0000000..678c7f9
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/peak_detection.c
@@ -0,0 +1,232 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of the peak detection used for finding correlation peaks.
+ */
+
+#include "dsp_helpfunctions.h"
+
+#include "signal_processing_library.h"
+
+/* Table of constants used in parabolic fit function WebRtcNetEQ_PrblFit */
+const WebRtc_Word16 WebRtcNetEQ_kPrblCf[17][3] = { { 120, 32, 64 }, { 140, 44, 75 },
+ { 150, 50, 80 }, { 160, 57, 85 },
+ { 180, 72, 96 }, { 200, 89, 107 },
+ { 210, 98, 112 }, { 220, 108, 117 },
+ { 240, 128, 128 }, { 260, 150, 139 },
+ { 270, 162, 144 }, { 280, 174, 149 },
+ { 300, 200, 160 }, { 320, 228, 171 },
+ { 330, 242, 176 }, { 340, 257, 181 },
+ { 360, 288, 192 } };
+
+WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
+ WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
+ WebRtc_Word16 *pw16_winIndex,
+ WebRtc_Word16 *pw16_winValue)
+{
+ /* Local variables */
+ int i;
+ WebRtc_Word16 w16_tmp;
+ WebRtc_Word16 w16_tmp2;
+ WebRtc_Word16 indMin = 0;
+ WebRtc_Word16 indMax = 0;
+
+ /* Peak detection */
+
+ for (i = 0; i <= (w16_nmbPeaks - 1); i++)
+ {
+ if (w16_nmbPeaks == 1)
+ {
+ /*
+ * Single peak
+ * The parabola fit assumes that an extra point is available; worst case it gets
+ * a zero on the high end of the signal.
+ */
+ w16_dataLen++;
+ }
+
+ pw16_winIndex[i] = WebRtcSpl_MaxIndexW16(pw16_data, (WebRtc_Word16) (w16_dataLen - 1));
+
+ if (i != w16_nmbPeaks - 1)
+ {
+ w16_tmp = pw16_winIndex[i] - 2; /* *fs_mult; */
+ indMin = WEBRTC_SPL_MAX(0, w16_tmp);
+ w16_tmp = pw16_winIndex[i] + 2; /* *fs_mult; */
+ w16_tmp2 = w16_dataLen - 1;
+ indMax = WEBRTC_SPL_MIN(w16_tmp2, w16_tmp);
+ }
+
+ if ((pw16_winIndex[i] != 0) && (pw16_winIndex[i] != (w16_dataLen - 2)))
+ {
+ /* Parabola fit*/
+ WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]), &(pw16_winIndex[i]),
+ &(pw16_winValue[i]), fs_mult);
+ }
+ else
+ {
+ if (pw16_winIndex[i] == (w16_dataLen - 2))
+ {
+ if (pw16_data[pw16_winIndex[i]] > pw16_data[pw16_winIndex[i] + 1])
+ {
+ WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]),
+ &(pw16_winIndex[i]), &(pw16_winValue[i]), fs_mult);
+ }
+ else if (pw16_data[pw16_winIndex[i]] <= pw16_data[pw16_winIndex[i] + 1])
+ {
+ pw16_winValue[i] = (pw16_data[pw16_winIndex[i]]
+ + pw16_data[pw16_winIndex[i] + 1]) >> 1; /* lin approx */
+ pw16_winIndex[i] = (pw16_winIndex[i] * 2 + 1) * fs_mult;
+ }
+ }
+ else
+ {
+ pw16_winValue[i] = pw16_data[pw16_winIndex[i]];
+ pw16_winIndex[i] = pw16_winIndex[i] * 2 * fs_mult;
+ }
+ }
+
+ if (i != w16_nmbPeaks - 1)
+ {
+ WebRtcSpl_MemSetW16(&(pw16_data[indMin]), 0, (indMax - indMin + 1));
+ /* for (j=indMin; j<=indMax; j++) pw16_data[j] = 0; */
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
+ WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult)
+{
+ /* Variables */
+ WebRtc_Word32 Num, Den;
+ WebRtc_Word32 temp;
+ WebRtc_Word16 flag, stp, strt, lmt;
+ WebRtc_UWord16 PFind[13];
+
+ if (fs_mult == 1)
+ {
+ PFind[0] = 0;
+ PFind[1] = 8;
+ PFind[2] = 16;
+ }
+ else if (fs_mult == 2)
+ {
+ PFind[0] = 0;
+ PFind[1] = 4;
+ PFind[2] = 8;
+ PFind[3] = 12;
+ PFind[4] = 16;
+ }
+ else if (fs_mult == 4)
+ {
+ PFind[0] = 0;
+ PFind[1] = 2;
+ PFind[2] = 4;
+ PFind[3] = 6;
+ PFind[4] = 8;
+ PFind[5] = 10;
+ PFind[6] = 12;
+ PFind[7] = 14;
+ PFind[8] = 16;
+ }
+ else
+ {
+ PFind[0] = 0;
+ PFind[1] = 1;
+ PFind[2] = 3;
+ PFind[3] = 4;
+ PFind[4] = 5;
+ PFind[5] = 7;
+ PFind[6] = 8;
+ PFind[7] = 9;
+ PFind[8] = 11;
+ PFind[9] = 12;
+ PFind[10] = 13;
+ PFind[11] = 15;
+ PFind[12] = 16;
+ }
+
+ /* Num = -3*pw16_3pts[0] + 4*pw16_3pts[1] - pw16_3pts[2]; */
+ /* Den = pw16_3pts[0] - 2*pw16_3pts[1] + pw16_3pts[2]; */
+ Num = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],-3) + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],4)
+ - pw16_3pts[2];
+
+ Den = pw16_3pts[0] + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],-2) + pw16_3pts[2];
+
+ temp = (WebRtc_Word32) WEBRTC_SPL_MUL(Num, (WebRtc_Word32)120); /* need 32_16 really */
+ flag = 1;
+ stp = WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0] - WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0];
+ strt = (WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0]
+ + WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0]) >> 1;
+
+ if (temp < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)strt))
+ {
+ lmt = strt - stp;
+ while (flag)
+ {
+ if ((flag == fs_mult) || (temp
+ > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
+ {
+ *pw16_outVal
+ = (WebRtc_Word16)
+ (((WebRtc_Word32) ((WebRtc_Word32) WEBRTC_SPL_MUL(Den,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][1])
+ + (WebRtc_Word32) WEBRTC_SPL_MUL(Num,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][2])
+ + WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256))) >> 8);
+ *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) - flag;
+ flag = 0;
+ }
+ else
+ {
+ flag++;
+ lmt -= stp;
+ }
+ }
+ }
+ else if (temp > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)(strt+stp)))
+ {
+ lmt = strt + (stp << 1);
+ while (flag)
+ {
+ if ((flag == fs_mult) || (temp
+ < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
+ {
+ WebRtc_Word32 temp_term_1, temp_term_2, temp_term_3;
+
+ temp_term_1 = WEBRTC_SPL_MUL(Den,
+ (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][1]);
+ temp_term_2 = WEBRTC_SPL_MUL(Num,
+ (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][2]);
+ temp_term_3 = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256);
+
+ *pw16_outVal
+ = (WebRtc_Word16) ((temp_term_1 + temp_term_2 + temp_term_3) >> 8);
+
+ *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) + flag;
+ flag = 0;
+ }
+ else
+ {
+ flag++;
+ lmt += stp;
+ }
+ }
+
+ }
+ else
+ {
+ *pw16_outVal = pw16_3pts[1];
+ *pw16_Ind = (*pw16_Ind) * 2 * fs_mult;
+ }
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/preemptive_expand.c b/trunk/src/modules/audio_coding/neteq/preemptive_expand.c
new file mode 100644
index 0000000..167bc3a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/preemptive_expand.c
@@ -0,0 +1,524 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the Pre-emptive Expand algorithm that is used to increase
+ * the delay by repeating a part of the audio stream.
+ */
+
+#include "dsp.h"
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+
+#define PREEMPTIVE_CORR_LEN 50
+#define PREEMPTIVE_MIN_LAG 10
+#define PREEMPTIVE_MAX_LAG 60
+#define PREEMPTIVE_DOWNSAMPLED_LEN (PREEMPTIVE_CORR_LEN + PREEMPTIVE_MAX_LAG)
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_downSampSpeech 110 0 109
+ WebRtc_Word32 pw32_corr 2*50 110 209
+ WebRtc_Word16 pw16_corr 50 0 49
+
+ Total: 110+2*50
+ */
+
+#define SCRATCH_PW16_DS_SPEECH 0
+#define SCRATCH_PW32_CORR PREEMPTIVE_DOWNSAMPLED_LEN
+#define SCRATCH_PW16_CORR 0
+
+/****************************************************************************
+ * WebRtcNetEQ_PreEmptiveExpand(...)
+ *
+ * This function tries to extend the audio data by repeating one or several
+ * pitch periods. The operation is only carried out if the correlation is
+ * strong or if the signal energy is very low. The algorithm is the
+ * reciprocal of the Accelerate algorithm.
+ *
+ * Input:
+ * - inst : NetEQ DSP instance
+ * - scratchPtr : Pointer to scratch vector.
+ * - decoded : Pointer to newly decoded speech.
+ * - len : Length of decoded speech.
+ * - oldDataLen : Length of the part of decoded that has already been played out.
+ * - BGNonly : If non-zero, Pre-emptive Expand will only copy
+ * the first DEFAULT_TIME_ADJUST seconds of the
+ * input and append to the end. No signal matching is
+ * done.
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outData : Pointer to a memory space where the output data
+ * should be stored. The vector must be at least
+ * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE)
+ * elements long.
+ * - pw16_len : Number of samples written to outData.
+ *
+ * Return value : 0 - Ok
+ * <0 - Error
+ */
+
+int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
+#ifdef SCRATCH
+ WebRtc_Word16 *pw16_scratchPtr,
+#endif
+ const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
+ WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
+ WebRtc_Word16 BGNonly)
+{
+
+#ifdef SCRATCH
+ /* Use scratch memory for internal temporary vectors */
+ WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
+ WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
+ WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
+#else
+ /* Allocate memory for temporary vectors */
+ WebRtc_Word16 pw16_downSampSpeech[PREEMPTIVE_DOWNSAMPLED_LEN];
+ WebRtc_Word32 pw32_corr[PREEMPTIVE_CORR_LEN];
+ WebRtc_Word16 pw16_corr[PREEMPTIVE_CORR_LEN];
+#endif
+ WebRtc_Word16 w16_decodedMax = 0;
+ WebRtc_Word16 w16_tmp = 0;
+ WebRtc_Word16 w16_tmp2;
+ WebRtc_Word32 w32_tmp;
+ WebRtc_Word32 w32_tmp2;
+
+ const WebRtc_Word16 w16_startLag = PREEMPTIVE_MIN_LAG;
+ const WebRtc_Word16 w16_endLag = PREEMPTIVE_MAX_LAG;
+ const WebRtc_Word16 w16_corrLen = PREEMPTIVE_CORR_LEN;
+ const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
+ WebRtc_Word16 *pw16_vectmp;
+ WebRtc_Word16 w16_inc, w16_startfact;
+ WebRtc_Word16 w16_bestIndex, w16_bestVal;
+ WebRtc_Word16 w16_VAD = 1;
+ WebRtc_Word16 fsMult;
+ WebRtc_Word16 fsMult120;
+ WebRtc_Word32 w32_en1, w32_en2, w32_cc;
+ WebRtc_Word16 w16_en1, w16_en2;
+ WebRtc_Word16 w16_en1Scale, w16_en2Scale;
+ WebRtc_Word16 w16_sqrtEn1En2;
+ WebRtc_Word16 w16_bestCorr = 0;
+ int ok;
+
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo *msInfo = inst->msInfo;
+#endif
+
+ fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
+
+ /* Pre-calculate common multiplication with fsMult */
+ fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
+
+ inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
+
+ /*
+ * Sanity check for len variable; must be (almost) 30 ms (120*fsMult + max(bestIndex)).
+ * Also, the new part must be at least .625 ms (w16_overlap).
+ */
+ if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult) || oldDataLen >= len
+ - inst->ExpandInst.w16_overlap)
+ {
+ /* Length of decoded data too short */
+ inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
+ *pw16_len = len;
+
+
+ /* simply move all data from decoded to outData */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /***********************************/
+ /* Special operations for BGN only */
+ /***********************************/
+
+ /* Check if "background noise only" flag is set */
+ if (BGNonly)
+ {
+ /* special operation for BGN only; simply insert a chunk of data */
+ w16_bestIndex = DEFAULT_TIME_ADJUST * (fsMult << 3); /* X*fs/1000 */
+
+ /* Sanity check for bestIndex */
+ if (w16_bestIndex > len)
+ { /* not good, do nothing instead */
+ inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
+ *pw16_len = len;
+
+
+ /* simply move all data from decoded to outData */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /* set length parameter */
+ *pw16_len = len + w16_bestIndex;
+
+
+ /* copy to output */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len);
+ WEBRTC_SPL_MEMCPY_W16(&pw16_outData[len], pw16_decoded, w16_bestIndex);
+
+ /* set mode */
+ inst->w16_mode = MODE_LOWEN_PREEMPTIVE;
+
+ /* update statistics */
+ inst->statInst.preemptiveLength += w16_bestIndex;
+
+ return 0;
+ } /* end of special code for BGN mode */
+
+#ifdef NETEQ_STEREO
+
+ /* Sanity for msInfo */
+ if (msInfo == NULL)
+ {
+ /* this should not happen here */
+ return MASTER_SLAVE_ERROR;
+ }
+
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+ /* Find correlation lag only for non-slave instances */
+
+#endif
+
+ /****************************************************************/
+ /* Find the strongest correlation lag by downsampling to 4 kHz, */
+ /* calculating correlation for downsampled signal and finding */
+ /* the strongest correlation peak. */
+ /****************************************************************/
+
+ /* find maximum absolute value */
+ w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
+
+ /* downsample the decoded speech to 4 kHz */
+ ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
+ PREEMPTIVE_DOWNSAMPLED_LEN, 1 /* compensate delay*/);
+ if (ok != 0)
+ {
+ /* error */
+ inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
+ *pw16_len = len;
+
+
+ /* simply move all data from decoded to outData */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return NETEQ_OTHER_ERROR;
+ }
+
+ /*
+ * Set scaling factor for cross correlation to protect against
+ * overflow (log2(50) => 6)
+ */
+ w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */WebRtcNetEQ_CrossCorr(
+ pw32_corr, &pw16_downSampSpeech[w16_endLag],
+ &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
+ (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
+
+ /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
+ w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
+ w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp);
+
+ /* Find limits for peak finding, in order to avoid overful NetEQ algorithm buffer. */
+ /* Calculate difference between MAX_OUTPUT_SIZE and len in 4 kHz domain. */
+ w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) (NETEQ_MAX_OUTPUT_SIZE - len),
+ (WebRtc_Word16) (fsMult << 1)) - w16_startLag;
+ w16_tmp = WEBRTC_SPL_MIN(w16_corrLen, w16_tmp); /* no more than corrLen = 50 */
+
+#ifdef NETEQ_STEREO
+ } /* end if (msInfo->msMode != NETEQ_SLAVE) */
+
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+ /* Find the strongest correlation peak by using the parabolic fit method */
+ WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal);
+ /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */
+
+ /* Compensate bestIndex for displaced starting position */
+ w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
+ /* 20*fsMult <= bestIndex <= 119*fsMult */
+
+ msInfo->bestIndex = w16_bestIndex;
+ }
+ else if (msInfo->msMode == NETEQ_SLAVE)
+ {
+ if (msInfo->extraInfo == PE_EXP_FAIL)
+ {
+ /* Master has signaled an unsuccessful preemptive expand */
+ w16_bestIndex = 0;
+ }
+ else
+ {
+ /* Get best index from master */
+ w16_bestIndex = msInfo->bestIndex;
+ }
+ }
+ else
+ {
+ /* Invalid mode */
+ return (MASTER_SLAVE_ERROR);
+ }
+
+#else /* NETEQ_STEREO */
+
+ /* Find the strongest correlation peak by using the parabolic fit method */
+ WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal);
+ /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */
+
+ /* Compensate bestIndex for displaced starting position */
+ w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
+ /* 20*fsMult <= bestIndex <= 119*fsMult */
+
+#endif /* NETEQ_STEREO */
+
+#ifdef NETEQ_STEREO
+
+ if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
+ {
+ /* Calculate correlation only for non-slave instances */
+
+#endif /* NETEQ_STEREO */
+
+ /*****************************************************/
+ /* Calculate correlation bestCorr for the found lag. */
+ /* Also do a simple VAD decision. */
+ /*****************************************************/
+
+ /*
+ * Calculate scaling to ensure that bestIndex samples can be square-summed
+ * without overflowing
+ */
+ w16_tmp = (31
+ - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)));
+ w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex));
+ w16_tmp -= 31;
+ w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
+
+ /* vec1 starts at 15 ms minus one pitch period */
+ pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
+ /* vec2 start at 15 ms */
+ pw16_vec2 = &pw16_decoded[fsMult120];
+
+ /* Calculate energies for vec1 and vec2 */
+ w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
+ (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
+ w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
+ (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
+
+ /* Calculate cross-correlation at the found lag */
+ w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
+ w16_bestIndex, w16_tmp);
+
+ /* Check VAD constraint
+ ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
+ w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */
+ if (inst->BGNInst.w16_initialized == 1)
+ {
+ w32_tmp2 = inst->BGNInst.w32_energy;
+ }
+ else
+ {
+ /* if BGN parameters have not been estimated, use a fixed threshold */
+ w32_tmp2 = 75000;
+ }
+ w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
+ w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
+ w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
+ w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
+ w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
+
+ /* Scale w32_tmp properly before comparing with w32_tmp2 */
+ /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */
+ if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
+ {
+ /* Cannot scale only w32_tmp, must scale w32_temp2 too */
+ WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
+ w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
+ WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
+ }
+ else
+ {
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp,
+ WEBRTC_SPL_LSHIFT_W32(w16_tmp,1));
+ }
+
+ if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
+ {
+ /* The signal seems to be passive speech */
+ w16_VAD = 0;
+ w16_bestCorr = 0; /* Correlation does not matter */
+
+ /* For low energy expansion, the new data can be less than 15 ms,
+ but we must ensure that bestIndex is not larger than the new data. */
+ w16_bestIndex = WEBRTC_SPL_MIN( w16_bestIndex, len - oldDataLen );
+ }
+ else
+ {
+ /* The signal is active speech */
+ w16_VAD = 1;
+
+ /* Calculate correlation (cc/sqrt(en1*en2)) */
+
+ /* Start with calculating scale values */
+ w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
+ w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
+ w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
+ w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
+
+ /* Make sure total scaling is even (to simplify scale factor after sqrt) */
+ if ((w16_en1Scale + w16_en2Scale) & 1)
+ {
+ w16_en1Scale += 1;
+ }
+
+ /* Convert energies to WebRtc_Word16 */
+ w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
+ w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
+
+ /* Calculate energy product */
+ w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
+
+ /* Calculate square-root of energy product */
+ w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
+
+ /* Calculate cc/sqrt(en1*en2) in Q14 */
+ w16_tmp = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
+ w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
+ w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
+ w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
+ w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
+ }
+
+#ifdef NETEQ_STEREO
+
+ } /* end if (msInfo->msMode != NETEQ_SLAVE) */
+
+#endif /* NETEQ_STEREO */
+
+ /*******************************************************/
+ /* Check preemptive expand criteria and insert samples */
+ /*******************************************************/
+
+ /* Check for strong correlation (>0.9) and at least 15 ms new data,
+ or passive speech */
+#ifdef NETEQ_STEREO
+ if (((((w16_bestCorr > 14746) && (oldDataLen <= fsMult120)) || (w16_VAD == 0))
+ && (msInfo->msMode != NETEQ_SLAVE)) || ((msInfo->msMode == NETEQ_SLAVE)
+ && (msInfo->extraInfo != PE_EXP_FAIL)))
+#else
+ if (((w16_bestCorr > 14746) && (oldDataLen <= fsMult120))
+ || (w16_VAD == 0))
+#endif
+ {
+ /* Do expand operation by overlap add */
+
+ /* Set length of the first part, not to be modified */
+ WebRtc_Word16 w16_startIndex = WEBRTC_SPL_MAX(oldDataLen, fsMult120);
+
+ /*
+ * Calculate cross-fading slope so that the fading factor goes from
+ * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
+ */
+ w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
+ (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
+
+ /* Initiate fading factor */
+ w16_startfact = 16384 - w16_inc;
+
+ /* vec1 starts at 15 ms minus one pitch period */
+ pw16_vec1 = &pw16_decoded[w16_startIndex - w16_bestIndex];
+ /* vec2 start at 15 ms */
+ pw16_vec2 = &pw16_decoded[w16_startIndex];
+
+
+ /* Copy unmodified part [0 to 15 ms] */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_startIndex);
+
+ /* Generate interpolated part of length bestIndex (1 pitch period) */
+ pw16_vectmp = pw16_outData + w16_startIndex;
+ /* Reuse mixing function from Expand */
+ WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec2,
+ (WebRtc_Word16*) pw16_vec1, &w16_startfact, w16_inc, w16_bestIndex);
+
+ /* Move the last part (also unmodified) */
+ /* Take from decoded at 15 ms */
+ pw16_vec2 = &pw16_decoded[w16_startIndex];
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[w16_startIndex + w16_bestIndex], pw16_vec2,
+ (WebRtc_Word16) (len - w16_startIndex));
+
+ /* Set the mode flag */
+ if (w16_VAD)
+ {
+ inst->w16_mode = MODE_SUCCESS_PREEMPTIVE;
+ }
+ else
+ {
+ inst->w16_mode = MODE_LOWEN_PREEMPTIVE;
+ }
+
+ /* Calculate resulting length = original length + pitch period */
+ *pw16_len = len + w16_bestIndex;
+
+ /* Update in-call statistics */
+ inst->statInst.preemptiveLength += w16_bestIndex;
+
+ return 0;
+ }
+ else
+ {
+ /* Preemptive Expand not allowed */
+
+#ifdef NETEQ_STEREO
+ /* Signal to slave(s) that this was unsuccessful */
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ msInfo->extraInfo = PE_EXP_FAIL;
+ }
+#endif
+
+ /* Set mode flag to unsuccessful preemptive expand */
+ inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
+
+ /* Length is unmodified */
+ *pw16_len = len;
+
+
+ /* Simply move all data from decoded to outData */
+
+ WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
+
+ return 0;
+ }
+}
+
+#undef SCRATCH_PW16_DS_SPEECH
+#undef SCRATCH_PW32_CORR
+#undef SCRATCH_PW16_CORR
diff --git a/trunk/src/modules/audio_coding/neteq/random_vector.c b/trunk/src/modules/audio_coding/neteq/random_vector.c
new file mode 100644
index 0000000..217bacd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/random_vector.c
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This function generates a pseudo-random vector.
+ */
+
+#include "dsp_helpfunctions.h"
+
+/*
+ * Values are normalized so that
+ * sqrt(dot(pw16_NETEQFIX_RANDN_TBL,pw16_NETEQFIX_RANDN_TBL)/256)=2^13
+ */
+const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[RANDVEC_NO_OF_SAMPLES] =
+{
+ 2680, 5532, 441, 5520, 16170, -5146, -1024, -8733, 3115, 9598, -10380, -4959, -1280, -21716, 7133, -1522,
+ 13458, -3902, 2789, -675, 3441, 5016, -13599, -4003, -2739, 3922, -7209, 13352, -11617, -7241, 12905, -2314,
+ 5426, 10121, -9702, 11207, -13542, 1373, 816, -5934, -12504, 4798, 1811, 4112, -613, 201, -10367, -2960,
+ -2419, 3442, 4299, -6116, -6092, 1552, -1650, -480, -1237, 18720, -11858, -8303, -8212, 865, -2890, -16968,
+ 12052, -5845, -5912, 9777, -5665, -6294, 5426, -4737, -6335, 1652, 761, 3832, 641, -8552, -9084, -5753,
+ 8146, 12156, -4915, 15086, -1231, -1869, 11749, -9319, -6403, 11407, 6232, -1683, 24340, -11166, 4017, -10448,
+ 3153, -2936, 6212, 2891, -866, -404, -4807, -2324, -1917, -2388, -6470, -3895, -10300, 5323, -5403, 2205,
+ 4640, 7022, -21186, -6244, -882, -10031, -3395, -12885, 7155, -5339, 5079, -2645, -9515, 6622, 14651, 15852,
+ 359, 122, 8246, -3502, -6696, -3679, -13535, -1409, -704, -7403, -4007, 1798, 279, -420, -12796, -14219,
+ 1141, 3359, 11434, 7049, -6684, -7473, 14283, -4115, -9123, -8969, 4152, 4117, 13792, 5742, 16168, 8661,
+ -1609, -6095, 1881, 14380, -5588, 6758, -6425, -22969, -7269, 7031, 1119, -1611, -5850, -11281, 3559, -8952,
+ -10146, -4667, -16251, -1538, 2062, -1012, -13073, 227, -3142, -5265, 20, 5770, -7559, 4740, -4819, 992,
+ -8208, -7130, -4652, 6725, 7369, -1036, 13144, -1588, -5304, -2344, -449, -5705, -8894, 5205, -17904, -11188,
+ -1022, 4852, 10101, -5255, -4200, -752, 7941, -1543, 5959, 14719, 13346, 17045, -15605, -1678, -1600, -9230,
+ 68, 23348, 1172, 7750, 11212, -18227, 9956, 4161, 883, 3947, 4341, 1014, -4889, -2603, 1246, -5630,
+ -3596, -870, -1298, 2784, -3317, -6612, -20541, 4166, 4181, -8625, 3562, 12890, 4761, 3205, -12259, -8579
+};
+
+
+void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
+ WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval)
+{
+ int i;
+ WebRtc_Word16 w16_pos;
+ for (i = 0; i < w16_len; i++)
+ {
+ *w32_seed = (*w32_seed) + w16_incval;
+ w16_pos = (WebRtc_Word16) ((*w32_seed) & (RANDVEC_NO_OF_SAMPLES - 1));
+ pw16_randVec[i] = WebRtcNetEQ_kRandnTbl[w16_pos];
+ }
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/recin.c b/trunk/src/modules/audio_coding/neteq/recin.c
new file mode 100644
index 0000000..608eb6e
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/recin.c
@@ -0,0 +1,472 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of the RecIn function, which is the main function for inserting RTP
+ * packets into NetEQ.
+ */
+
+#include "mcu.h"
+
+#include <string.h>
+
+#include "signal_processing_library.h"
+
+#include "automode.h"
+#include "dtmf_buffer.h"
+#include "neteq_defines.h"
+#include "neteq_error_codes.h"
+
+
+int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
+ WebRtc_UWord32 uw32_timeRec)
+{
+ RTPPacket_t RTPpacket[2];
+ int i_k;
+ int i_ok = 0, i_No_Of_Payloads = 1;
+ WebRtc_Word16 flushed = 0;
+ WebRtc_Word16 codecPos;
+ int curr_Codec;
+ WebRtc_Word16 isREDPayload = 0;
+ WebRtc_Word32 temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer;
+#ifdef NETEQ_RED_CODEC
+ RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */
+ RTPpacketPtr[0] = &RTPpacket[0];
+ RTPpacketPtr[1] = &RTPpacket[1];
+#endif
+
+ /*
+ * Copy from input RTP packet to local copy
+ * (mainly to enable multiple payloads using RED)
+ */
+
+ WEBRTC_SPL_MEMCPY_W8(&RTPpacket[0], RTPpacketInput, sizeof(RTPPacket_t));
+
+ /* Reinitialize NetEq if it's needed (changed SSRC or first call) */
+
+ if ((RTPpacket[0].ssrc != MCU_inst->ssrc) || (MCU_inst->first_packet == 1))
+ {
+ WebRtcNetEQ_RTCPInit(&MCU_inst->RTCP_inst, RTPpacket[0].seqNumber);
+ MCU_inst->first_packet = 0;
+
+ /* Flush the buffer */
+ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
+
+ /* Store new SSRC */
+ MCU_inst->ssrc = RTPpacket[0].ssrc;
+
+ /* Update codecs */
+ MCU_inst->timeStamp = RTPpacket[0].timeStamp;
+ MCU_inst->current_Payload = RTPpacket[0].payloadType;
+
+ /*Set MCU to update codec on next SignalMCU call */
+ MCU_inst->new_codec = 1;
+
+ /* Reset timestamp scaling */
+ MCU_inst->TSscalingInitialized = 0;
+
+ }
+
+ /* Call RTCP statistics */
+ i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber,
+ RTPpacket[0].timeStamp, uw32_timeRec);
+
+ /* If Redundancy is supported and this is the redundancy payload, separate the payloads */
+#ifdef NETEQ_RED_CODEC
+ if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
+ kDecoderRED))
+ {
+
+ /* Split the payload into a main and a redundancy payloads */
+ i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads);
+ if (i_ok < 0)
+ {
+ /* error returned */
+ return i_ok;
+ }
+
+ /*
+ * Only accept a few redundancies of the same type as the main data,
+ * AVT events and CNG.
+ */
+ if ((i_No_Of_Payloads > 1) && (RTPpacket[0].payloadType != RTPpacket[1].payloadType)
+ && (RTPpacket[0].payloadType != WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
+ kDecoderAVT)) && (RTPpacket[1].payloadType != WebRtcNetEQ_DbGetPayload(
+ &MCU_inst->codec_DB_inst, kDecoderAVT)) && (!WebRtcNetEQ_DbIsCNGPayload(
+ &MCU_inst->codec_DB_inst, RTPpacket[0].payloadType))
+ && (!WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[1].payloadType)))
+ {
+ i_No_Of_Payloads = 1;
+ }
+ isREDPayload = 1;
+ }
+#endif
+
+ /* loop over the number of payloads */
+ for (i_k = 0; i_k < i_No_Of_Payloads; i_k++)
+ {
+
+ if (isREDPayload == 1)
+ {
+ RTPpacket[i_k].rcuPlCntr = i_k;
+ }
+ else
+ {
+ RTPpacket[i_k].rcuPlCntr = 0;
+ }
+
+ /* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */
+ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
+ kDecoderILBC))
+ {
+ i_ok = WebRtcNetEQ_DbGetSplitInfo(
+ &MCU_inst->PayloadSplit_inst,
+ (enum WebRtcNetEQDecoder) WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
+ RTPpacket[i_k].payloadType), RTPpacket[i_k].payloadLen);
+ if (i_ok < 0)
+ {
+ /* error returned */
+ return i_ok;
+ }
+ }
+
+ /* Get information about timestamp scaling for this payload type */
+ i_ok = WebRtcNetEQ_GetTimestampScaling(MCU_inst, RTPpacket[i_k].payloadType);
+ if (i_ok < 0)
+ {
+ /* error returned */
+ return i_ok;
+ }
+
+ if (MCU_inst->TSscalingInitialized == 0 && MCU_inst->scalingFactor != kTSnoScaling)
+ {
+ /* Must initialize scaling with current timestamps */
+ MCU_inst->externalTS = RTPpacket[i_k].timeStamp;
+ MCU_inst->internalTS = RTPpacket[i_k].timeStamp;
+ MCU_inst->TSscalingInitialized = 1;
+ }
+
+ /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */
+ if (MCU_inst->TSscalingInitialized == 1)
+ {
+ WebRtc_UWord32 newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst,
+ RTPpacket[i_k].timeStamp);
+
+ /* save the incoming timestamp for next time */
+ MCU_inst->externalTS = RTPpacket[i_k].timeStamp;
+
+ /* add the scaled difference to last scaled timestamp and save ... */
+ MCU_inst->internalTS = newTS;
+
+ RTPpacket[i_k].timeStamp = newTS;
+ }
+
+ /* Is this a DTMF packet?*/
+ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
+ kDecoderAVT))
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ if (MCU_inst->AVT_PlayoutOn)
+ {
+ i_ok = WebRtcNetEQ_DtmfInsertEvent(&MCU_inst->DTMF_inst,
+ RTPpacket[i_k].payload, RTPpacket[i_k].payloadLen,
+ RTPpacket[i_k].timeStamp);
+ if (i_ok != 0)
+ {
+ return i_ok;
+ }
+ }
+#endif
+#ifdef NETEQ_STEREO
+ if (MCU_inst->usingStereo == 0)
+ {
+ /* do not set this for DTMF packets when using stereo mode */
+ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
+ }
+#else
+ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
+#endif
+ }
+ else if (WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst,
+ RTPpacket[i_k].payloadType))
+ {
+ /* Is this a CNG packet? how should we handle this?*/
+#ifdef NETEQ_CNG_CODEC
+ /* Get CNG sample rate */
+ WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
+ RTPpacket[i_k].payloadType);
+ if ((fsCng != MCU_inst->fs) && (fsCng > 8000))
+ {
+ /*
+ * We have received CNG with a different sample rate from what we are using
+ * now (must be > 8000, since we may use only one CNG type (default) for all
+ * frequencies). Flush buffer and signal new codec.
+ */
+ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
+ MCU_inst->new_codec = 1;
+ MCU_inst->current_Codec = -1;
+ }
+ i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst,
+ &RTPpacket[i_k], &flushed);
+ if (i_ok < 0)
+ {
+ return RECIN_CNG_ERROR;
+ }
+ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
+#else /* NETEQ_CNG_CODEC not defined */
+ return RECIN_UNKNOWNPAYLOAD;
+#endif /* NETEQ_CNG_CODEC */
+ }
+ else
+ {
+ /* Reinitialize the splitting if the payload and/or the payload length has changed */
+ curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
+ RTPpacket[i_k].payloadType);
+ if (curr_Codec != MCU_inst->current_Codec)
+ {
+ if (curr_Codec < 0)
+ {
+ return RECIN_UNKNOWNPAYLOAD;
+ }
+ MCU_inst->current_Codec = curr_Codec;
+ MCU_inst->current_Payload = RTPpacket[i_k].payloadType;
+ i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst,
+ (enum WebRtcNetEQDecoder) MCU_inst->current_Codec,
+ RTPpacket[i_k].payloadLen);
+ if (i_ok < 0)
+ { /* error returned */
+ return i_ok;
+ }
+ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
+ MCU_inst->new_codec = 1;
+ }
+
+ /* Parse the payload and insert it into the buffer */
+ i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k],
+ &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed);
+ if (i_ok < 0)
+ {
+ return i_ok;
+ }
+ if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF != 0)
+ {
+ /* first normal packet after CNG or DTMF */
+ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = -1;
+ }
+ }
+ /* Reset DSP timestamp etc. if packet buffer flushed */
+ if (flushed)
+ {
+ MCU_inst->new_codec = 1;
+ }
+ }
+
+ /*
+ * Update Bandwidth Estimate
+ * Only send the main payload to BWE
+ */
+ if ((curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
+ RTPpacket[0].payloadType)) >= 0)
+ {
+ codecPos = MCU_inst->codec_DB_inst.position[curr_Codec];
+ if (MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos] != NULL) /* codec has BWE function */
+ {
+ if (RTPpacket[0].starts_byte1) /* check for shifted byte alignment */
+ {
+ /* re-align to 16-bit alignment */
+ for (i_k = 0; i_k < RTPpacket[0].payloadLen; i_k++)
+ {
+ WEBRTC_SPL_SET_BYTE(RTPpacket[0].payload,
+ WEBRTC_SPL_GET_BYTE(RTPpacket[0].payload, i_k+1),
+ i_k);
+ }
+ RTPpacket[0].starts_byte1 = 0;
+ }
+
+ MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos](
+ MCU_inst->codec_DB_inst.codec_state[codecPos],
+ (G_CONST WebRtc_UWord16 *) RTPpacket[0].payload,
+ (WebRtc_Word32) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber,
+ (WebRtc_UWord32) RTPpacket[0].timeStamp, (WebRtc_UWord32) uw32_timeRec);
+ }
+ }
+
+ if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == 0)
+ {
+ /* Calculate the total speech length carried in each packet */
+ temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer - temp_bufsize;
+ temp_bufsize *= MCU_inst->PacketBuffer_inst.packSizeSamples;
+
+ if ((temp_bufsize > 0) && (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF
+ == 0) && (temp_bufsize
+ != MCU_inst->BufferStat_inst.Automode_inst.packetSpeechLenSamp))
+ {
+ /* Change the auto-mode parameters if packet length has changed */
+ WebRtcNetEQ_SetPacketSpeechLen(&(MCU_inst->BufferStat_inst.Automode_inst),
+ (WebRtc_Word16) temp_bufsize, MCU_inst->fs);
+ }
+
+ /* update statistics */
+ if ((WebRtc_Word32) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0
+ && !MCU_inst->new_codec)
+ {
+ /*
+ * Only update statistics if incoming packet is not older than last played out
+ * packet, and if new codec flag is not set.
+ */
+ WebRtcNetEQ_UpdateIatStatistics(&MCU_inst->BufferStat_inst.Automode_inst,
+ MCU_inst->PacketBuffer_inst.maxInsertPositions, RTPpacket[0].seqNumber,
+ RTPpacket[0].timeStamp, MCU_inst->fs,
+ WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) MCU_inst->current_Codec),
+ (MCU_inst->NetEqPlayoutMode == kPlayoutStreaming));
+ }
+ }
+ else if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == -1)
+ {
+ /*
+ * This is first "normal" packet after CNG or DTMF.
+ * Reset packet time counter and measure time until next packet,
+ * but don't update statistics.
+ */
+ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 0;
+ MCU_inst->BufferStat_inst.Automode_inst.packetIatCountSamp = 0;
+ }
+ return 0;
+
+}
+
+int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType)
+{
+ enum WebRtcNetEQDecoder codec;
+ int codecNumber;
+
+ codecNumber = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, rtpPayloadType);
+ if (codecNumber < 0)
+ {
+ /* error */
+ return codecNumber;
+ }
+
+ /* cast to enumerator */
+ codec = (enum WebRtcNetEQDecoder) codecNumber;
+
+ /*
+ * The factor obtained below is the number with which the RTP timestamp must be
+ * multiplied to get the true sample count.
+ */
+ switch (codec)
+ {
+ case kDecoderG722:
+ {
+ /* Use timestamp scaling with factor 2 (two output samples per RTP timestamp) */
+ MCU_inst->scalingFactor = kTSscalingTwo;
+ break;
+ }
+ case kDecoderAVT:
+ case kDecoderCNG:
+ {
+ /* do not change the timestamp scaling settings */
+ break;
+ }
+ default:
+ {
+ /* do not use timestamp scaling */
+ MCU_inst->scalingFactor = kTSnoScaling;
+ break;
+ }
+ }
+ return 0;
+}
+
+WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
+ WebRtc_UWord32 externalTS)
+{
+ WebRtc_Word32 timestampDiff;
+ WebRtc_UWord32 internalTS;
+
+ /* difference between this and last incoming timestamp */
+ timestampDiff = externalTS - MCU_inst->externalTS;
+
+ switch (MCU_inst->scalingFactor)
+ {
+ case kTSscalingTwo:
+ {
+ /* multiply with 2 */
+ timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1);
+ break;
+ }
+ case kTSscalingTwoThirds:
+ {
+ /* multiply with 2/3 */
+ timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1);
+ timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3);
+ break;
+ }
+ case kTSscalingFourThirds:
+ {
+ /* multiply with 4/3 */
+ timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 2);
+ timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3);
+ break;
+ }
+ default:
+ {
+ /* no scaling */
+ }
+ }
+
+ /* add the scaled difference to last scaled timestamp and save ... */
+ internalTS = MCU_inst->internalTS + timestampDiff;
+
+ return internalTS;
+}
+
+WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
+ WebRtc_UWord32 internalTS)
+{
+ WebRtc_Word32 timestampDiff;
+ WebRtc_UWord32 externalTS;
+
+ /* difference between this and last incoming timestamp */
+ timestampDiff = (WebRtc_Word32) internalTS - MCU_inst->internalTS;
+
+ switch (MCU_inst->scalingFactor)
+ {
+ case kTSscalingTwo:
+ {
+ /* divide by 2 */
+ timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1);
+ break;
+ }
+ case kTSscalingTwoThirds:
+ {
+ /* multiply with 3/2 */
+ timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3);
+ timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1);
+ break;
+ }
+ case kTSscalingFourThirds:
+ {
+ /* multiply with 3/4 */
+ timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3);
+ timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 2);
+ break;
+ }
+ default:
+ {
+ /* no scaling */
+ }
+ }
+
+ /* add the scaled difference to last scaled timestamp and save ... */
+ externalTS = MCU_inst->externalTS + timestampDiff;
+
+ return externalTS;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/recout.c b/trunk/src/modules/audio_coding/neteq/recout.c
new file mode 100644
index 0000000..dab5540
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/recout.c
@@ -0,0 +1,1429 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of RecOut function, which is the main function for the audio output
+ * process. This function must be called (through the NetEQ API) once every 10 ms.
+ */
+
+#include "dsp.h"
+
+#include <assert.h>
+#include <string.h> /* to define NULL */
+
+#include "signal_processing_library.h"
+
+#include "dsp_helpfunctions.h"
+#include "neteq_error_codes.h"
+#include "neteq_defines.h"
+#include "mcu_dsp_common.h"
+
+/* Audio types */
+#define TYPE_SPEECH 1
+#define TYPE_CNG 2
+
+#ifdef NETEQ_DELAY_LOGGING
+#include "delay_logging.h"
+#include <stdio.h>
+#pragma message("*******************************************************************")
+#pragma message("You have specified to use NETEQ_DELAY_LOGGING in the NetEQ library.")
+#pragma message("Make sure that your test application supports this.")
+#pragma message("*******************************************************************")
+#endif
+
+/* Scratch usage:
+
+ Type Name size startpos endpos
+ WebRtc_Word16 pw16_NetEqAlgorithm_buffer 600*fs/8000 0 600*fs/8000-1
+ struct dspInfo 6 600*fs/8000 605*fs/8000
+
+ func WebRtcNetEQ_Normal 40+495*fs/8000 0 39+495*fs/8000
+ func WebRtcNetEQ_Merge 40+496*fs/8000 0 39+496*fs/8000
+ func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/800 39+496*fs/8000
+ func WebRtcNetEQ_Accelerate 210 240*fs/8000 209+240*fs/8000
+ func WebRtcNetEQ_BGNUpdate 69 480*fs/8000 68+480*fs/8000
+
+ Total: 605*fs/8000
+ */
+
+#define SCRATCH_ALGORITHM_BUFFER 0
+#define SCRATCH_NETEQ_NORMAL 0
+#define SCRATCH_NETEQ_MERGE 0
+
+#if (defined(NETEQ_48KHZ_WIDEBAND))
+#define SCRATCH_DSP_INFO 3600
+#define SCRATCH_NETEQ_ACCELERATE 1440
+#define SCRATCH_NETEQ_BGN_UPDATE 2880
+#define SCRATCH_NETEQ_EXPAND 756
+#elif (defined(NETEQ_32KHZ_WIDEBAND))
+#define SCRATCH_DSP_INFO 2400
+#define SCRATCH_NETEQ_ACCELERATE 960
+#define SCRATCH_NETEQ_BGN_UPDATE 1920
+#define SCRATCH_NETEQ_EXPAND 504
+#elif (defined(NETEQ_WIDEBAND))
+#define SCRATCH_DSP_INFO 1200
+#define SCRATCH_NETEQ_ACCELERATE 480
+#define SCRATCH_NETEQ_BGN_UPDATE 960
+#define SCRATCH_NETEQ_EXPAND 252
+#else /* NB */
+#define SCRATCH_DSP_INFO 600
+#define SCRATCH_NETEQ_ACCELERATE 240
+#define SCRATCH_NETEQ_BGN_UPDATE 480
+#define SCRATCH_NETEQ_EXPAND 126
+#endif
+
+#if (defined(NETEQ_48KHZ_WIDEBAND))
+#define SIZE_SCRATCH_BUFFER 3636
+#elif (defined(NETEQ_32KHZ_WIDEBAND))
+#define SIZE_SCRATCH_BUFFER 2424
+#elif (defined(NETEQ_WIDEBAND))
+#define SIZE_SCRATCH_BUFFER 1212
+#else /* NB */
+#define SIZE_SCRATCH_BUFFER 606
+#endif
+
+#ifdef NETEQ_DELAY_LOGGING
+extern FILE *delay_fid2; /* file pointer to delay log file */
+extern WebRtc_UWord32 tot_received_packets;
+#endif
+
+
+int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len, WebRtc_Word16 BGNonly)
+{
+
+ WebRtc_Word16 blockLen, payloadLen, len = 0, pos;
+ WebRtc_Word16 w16_tmp1, w16_tmp2, w16_tmp3, DataEnough;
+ WebRtc_Word16 *blockPtr;
+ WebRtc_Word16 MD = 0;
+
+ WebRtc_Word16 speechType = TYPE_SPEECH;
+ WebRtc_UWord16 instr;
+ WebRtc_UWord16 uw16_tmp;
+#ifdef SCRATCH
+ char pw8_ScratchBuffer[((SIZE_SCRATCH_BUFFER + 1) * 2)];
+ WebRtc_Word16 *pw16_scratchPtr = (WebRtc_Word16*) pw8_ScratchBuffer;
+ WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
+ WebRtc_Word16 *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr
+ + SCRATCH_ALGORITHM_BUFFER;
+ DSP2MCU_info_t *dspInfo = (DSP2MCU_info_t*) (pw16_scratchPtr + SCRATCH_DSP_INFO);
+#else
+ WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
+ WebRtc_Word16 pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE];
+ DSP2MCU_info_t dspInfoStruct;
+ DSP2MCU_info_t *dspInfo = &dspInfoStruct;
+#endif
+ WebRtc_Word16 fs_mult;
+ int borrowedSamples;
+ int oldBorrowedSamples;
+ int return_value = 0;
+ WebRtc_Word16 lastModeBGNonly = (inst->w16_mode & MODE_BGN_ONLY) != 0; /* check BGN flag */
+ void *mainInstBackup = inst->main_inst;
+
+#ifdef NETEQ_DELAY_LOGGING
+ int temp_var;
+#endif
+ WebRtc_Word16 dtmfValue = -1;
+ WebRtc_Word16 dtmfVolume = -1;
+ int playDtmf = 0;
+#ifdef NETEQ_ATEVENT_DECODE
+ int dtmfSwitch = 0;
+#endif
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo *msInfo = inst->msInfo;
+#endif
+ WebRtc_Word16 *sharedMem = pw16_NetEqAlgorithm_buffer; /* Reuse memory SHARED_MEM_SIZE size */
+ inst->pw16_readAddress = sharedMem;
+ inst->pw16_writeAddress = sharedMem;
+
+ /* Get information about if there is one descriptor left */
+ if (inst->codec_ptr_inst.funcGetMDinfo != NULL)
+ {
+ MD = inst->codec_ptr_inst.funcGetMDinfo(inst->codec_ptr_inst.codec_state);
+ if (MD > 0)
+ MD = 1;
+ else
+ MD = 0;
+ }
+
+#ifdef NETEQ_STEREO
+ if ((msInfo->msMode == NETEQ_SLAVE) && (inst->codec_ptr_inst.funcDecode != NULL))
+ {
+ /*
+ * Valid function pointers indicate that we have decoded something,
+ * and that the timestamp information is correct.
+ */
+
+ /* Get the information from master to correct synchronization */
+ WebRtc_UWord32 currentMasterTimestamp;
+ WebRtc_UWord32 currentSlaveTimestamp;
+
+ currentMasterTimestamp = msInfo->endTimestamp - msInfo->samplesLeftWithOverlap;
+ currentSlaveTimestamp = inst->endTimestamp - (inst->endPosition - inst->curPosition);
+
+ if (currentSlaveTimestamp < currentMasterTimestamp)
+ {
+ /* brute-force discard a number of samples to catch up */
+ inst->curPosition += currentMasterTimestamp - currentSlaveTimestamp;
+
+ /* make sure we have at least "overlap" samples left */
+ inst->curPosition = WEBRTC_SPL_MIN(inst->curPosition,
+ inst->endPosition - inst->ExpandInst.w16_overlap);
+ }
+ else if (currentSlaveTimestamp > currentMasterTimestamp)
+ {
+ /* back off current position to slow down */
+ inst->curPosition -= currentSlaveTimestamp - currentMasterTimestamp;
+
+ /* make sure we do not end up outside the speech history */
+ inst->curPosition = WEBRTC_SPL_MAX(inst->curPosition, 0);
+ }
+ }
+#endif
+
+ /* Write status data to shared memory */
+ dspInfo->playedOutTS = inst->endTimestamp;
+ dspInfo->samplesLeft = inst->endPosition - inst->curPosition
+ - inst->ExpandInst.w16_overlap;
+ dspInfo->MD = MD;
+ dspInfo->lastMode = inst->w16_mode;
+ dspInfo->frameLen = inst->w16_frameLen;
+
+ /* Force update of codec if codec function is NULL */
+ if (inst->codec_ptr_inst.funcDecode == NULL)
+ {
+ dspInfo->lastMode |= MODE_AWAITING_CODEC_PTR;
+ }
+
+#ifdef NETEQ_STEREO
+ if (msInfo->msMode == NETEQ_SLAVE && (msInfo->extraInfo == DTMF_OVERDUB
+ || msInfo->extraInfo == DTMF_ONLY))
+ {
+ /* Signal that the master instance generated DTMF tones */
+ dspInfo->lastMode |= MODE_MASTER_DTMF_SIGNAL;
+ }
+
+ if (msInfo->msMode != NETEQ_MONO)
+ {
+ /* We are using stereo mode; signal this to MCU side */
+ dspInfo->lastMode |= MODE_USING_STEREO;
+ }
+#endif
+
+ WEBRTC_SPL_MEMCPY_W8(inst->pw16_writeAddress,dspInfo,sizeof(DSP2MCU_info_t));
+
+ /* Signal MCU with "interrupt" call to main inst*/
+#ifdef NETEQ_STEREO
+ assert(msInfo != NULL);
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ /* clear info to slave */
+ WebRtcSpl_MemSetW16((WebRtc_Word16 *) msInfo, 0,
+ sizeof(MasterSlaveInfo) / sizeof(WebRtc_Word16));
+ /* re-set mode */
+ msInfo->msMode = NETEQ_MASTER;
+
+ /* Store some information to slave */
+ msInfo->endTimestamp = inst->endTimestamp;
+ msInfo->samplesLeftWithOverlap = inst->endPosition - inst->curPosition;
+ }
+#endif
+
+ /*
+ * This call will trigger the MCU side to make a decision based on buffer contents and
+ * decision history. Instructions, encoded data and function pointers will be written
+ * to the shared memory.
+ */
+ return_value = WebRtcNetEQ_DSP2MCUinterrupt((MainInst_t *) inst->main_inst, sharedMem);
+
+ /* Read MCU data and instructions */
+ instr = (WebRtc_UWord16) (inst->pw16_readAddress[0] & 0xf000);
+
+#ifdef NETEQ_STEREO
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ msInfo->instruction = instr;
+ }
+ else if (msInfo->msMode == NETEQ_SLAVE)
+ {
+ /* Nothing to do */
+ }
+#endif
+
+ /* check for error returned from MCU side, if so, return error */
+ if (return_value < 0)
+ {
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return return_value;
+ }
+
+ blockPtr = &((inst->pw16_readAddress)[3]);
+
+ /* Check for DTMF payload flag */
+ if ((inst->pw16_readAddress[0] & DSP_DTMF_PAYLOAD) != 0)
+ {
+ playDtmf = 1;
+ dtmfValue = blockPtr[1];
+ dtmfVolume = blockPtr[2];
+ blockPtr += 3;
+
+#ifdef NETEQ_STEREO
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ /* signal to slave that master is using DTMF */
+ msInfo->extraInfo = DTMF_OVERDUB;
+ }
+#endif
+ }
+
+ blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of WebRtc_Word16 */
+ payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
+ blockPtr++;
+
+ /* Do we have to change our decoder? */
+ if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_NEW_CODEC)
+ {
+ WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1);
+ if (inst->codec_ptr_inst.codec_fs != 0)
+ {
+ return_value = WebRtcNetEQ_DSPInit(inst, inst->codec_ptr_inst.codec_fs);
+ if (return_value != 0)
+ { /* error returned */
+ instr = DSP_INSTR_FADE_TO_BGN; /* emergency instruction */
+ }
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ fwrite(&inst->fs, sizeof(WebRtc_UWord16), 1, delay_fid2);
+#endif
+ }
+
+ /* Copy it again since the init destroys this part */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1);
+ inst->endTimestamp = inst->codec_ptr_inst.timeStamp;
+ inst->videoSyncTimestamp = inst->codec_ptr_inst.timeStamp;
+ blockPtr += blockLen;
+ blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1;
+ payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
+ blockPtr++;
+ if (inst->codec_ptr_inst.funcDecodeInit != NULL)
+ {
+ inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
+ }
+
+#ifdef NETEQ_CNG_CODEC
+
+ /* Also update the CNG state as this might be uninitialized */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->CNG_Codec_inst,blockPtr,(payloadLen+1)>>1);
+ blockPtr += blockLen;
+ blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1;
+ payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
+ blockPtr++;
+ if (inst->CNG_Codec_inst != NULL)
+ {
+ WebRtcCng_InitDec(inst->CNG_Codec_inst);
+ }
+#endif
+ }
+ else if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_RESET)
+ {
+ /* Reset the current codec (but not DSP struct) */
+ if (inst->codec_ptr_inst.funcDecodeInit != NULL)
+ {
+ inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
+ }
+
+#ifdef NETEQ_CNG_CODEC
+ /* And reset CNG */
+ if (inst->CNG_Codec_inst != NULL)
+ {
+ WebRtcCng_InitDec(inst->CNG_Codec_inst);
+ }
+#endif /*NETEQ_CNG_CODEC*/
+ }
+
+ fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs);
+
+ /* Add late packet? */
+ if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_ADD_LATE_PKT)
+ {
+ if (inst->codec_ptr_inst.funcAddLatePkt != NULL)
+ {
+ /* Only do this if the codec has support for Add Late Pkt */
+ inst->codec_ptr_inst.funcAddLatePkt(inst->codec_ptr_inst.codec_state, blockPtr,
+ payloadLen);
+ }
+ blockPtr += blockLen;
+ blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */
+ payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
+ blockPtr++;
+ }
+
+ /* Do we have to decode data? */
+ if ((instr == DSP_INSTR_NORMAL) || (instr == DSP_INSTR_ACCELERATE) || (instr
+ == DSP_INSTR_MERGE) || (instr == DSP_INSTR_PREEMPTIVE_EXPAND))
+ {
+ /* Do we need to update codec-internal PLC state? */
+ if ((instr == DSP_INSTR_MERGE) && (inst->codec_ptr_inst.funcDecodePLC != NULL))
+ {
+ len = 0;
+ len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
+ &pw16_decoded_buffer[len], 1);
+ }
+ len = 0;
+
+ /* Do decoding */
+ while ((blockLen > 0) && (len < (240 * fs_mult))) /* Guard somewhat against overflow */
+ {
+ if (inst->codec_ptr_inst.funcDecode != NULL)
+ {
+ WebRtc_Word16 dec_Len;
+ if (!BGNonly)
+ {
+ /* Do decoding as normal
+ *
+ * blockPtr is pointing to payload, at this point,
+ * the most significant bit of *(blockPtr - 1) is a flag if set to 1
+ * indicates that the following payload is the redundant payload.
+ */
+ if (((*(blockPtr - 1) & DSP_CODEC_RED_FLAG) != 0)
+ && (inst->codec_ptr_inst.funcDecodeRCU != NULL))
+ {
+ dec_Len = inst->codec_ptr_inst.funcDecodeRCU(
+ inst->codec_ptr_inst.codec_state, blockPtr, payloadLen,
+ &pw16_decoded_buffer[len], &speechType);
+ }
+ else
+ {
+ dec_Len = inst->codec_ptr_inst.funcDecode(
+ inst->codec_ptr_inst.codec_state, blockPtr, payloadLen,
+ &pw16_decoded_buffer[len], &speechType);
+ }
+ }
+ else
+ {
+ /*
+ * Background noise mode: don't decode, just produce the same length BGN.
+ * Don't call Expand for BGN here, since Expand uses the memory where the
+ * bitstreams are stored (sharemem).
+ */
+ dec_Len = inst->w16_frameLen;
+ }
+
+ if (dec_Len > 0)
+ {
+ len += dec_Len;
+ /* Update frameLen */
+ inst->w16_frameLen = dec_Len;
+ }
+ else if (dec_Len < 0)
+ {
+ /* Error */
+ len = -1;
+ break;
+ }
+ /*
+ * Sanity check (although we might still write outside memory when this
+ * happens...)
+ */
+ if (len > NETEQ_MAX_FRAME_SIZE)
+ {
+ WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall);
+ *pw16_len = inst->timestampsPerCall;
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return RECOUT_ERROR_DECODED_TOO_MUCH;
+ }
+
+ /* Verify that instance was not corrupted by decoder */
+ if (mainInstBackup != inst->main_inst)
+ {
+ /* Instance is corrupt */
+ return CORRUPT_INSTANCE;
+ }
+
+ }
+ blockPtr += blockLen;
+ blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */
+ payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
+ blockPtr++;
+ }
+
+ if (len < 0)
+ {
+ len = 0;
+ inst->endTimestamp += inst->w16_frameLen; /* advance one frame */
+ if (inst->codec_ptr_inst.funcGetErrorCode != NULL)
+ {
+ return_value = -inst->codec_ptr_inst.funcGetErrorCode(
+ inst->codec_ptr_inst.codec_state);
+ }
+ else
+ {
+ return_value = RECOUT_ERROR_DECODING;
+ }
+ instr = DSP_INSTR_FADE_TO_BGN;
+ }
+ if (speechType != TYPE_CNG)
+ {
+ /*
+ * Don't increment timestamp if codec returned CNG speech type
+ * since in this case, the MCU side will increment the CNGplayedTS counter.
+ */
+ inst->endTimestamp += len;
+ }
+ }
+ else if (instr == DSP_INSTR_NORMAL_ONE_DESC)
+ {
+ if (inst->codec_ptr_inst.funcDecode != NULL)
+ {
+ len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state, NULL, 0,
+ pw16_decoded_buffer, &speechType);
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32), 1, delay_fid2);
+ fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
+ tot_received_packets++;
+#endif
+ }
+ if (speechType != TYPE_CNG)
+ {
+ /*
+ * Don't increment timestamp if codec returned CNG speech type
+ * since in this case, the MCU side will increment the CNGplayedTS counter.
+ */
+ inst->endTimestamp += len;
+ }
+
+ /* Verify that instance was not corrupted by decoder */
+ if (mainInstBackup != inst->main_inst)
+ {
+ /* Instance is corrupt */
+ return CORRUPT_INSTANCE;
+ }
+
+ if (len <= 0)
+ {
+ len = 0;
+ if (inst->codec_ptr_inst.funcGetErrorCode != NULL)
+ {
+ return_value = -inst->codec_ptr_inst.funcGetErrorCode(
+ inst->codec_ptr_inst.codec_state);
+ }
+ else
+ {
+ return_value = RECOUT_ERROR_DECODING;
+ }
+ if ((inst->codec_ptr_inst.funcDecodeInit != NULL)
+ && (inst->codec_ptr_inst.codec_state != NULL))
+ {
+ /* Reinitialize codec state as something is obviously wrong */
+ inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
+ }
+ inst->endTimestamp += inst->w16_frameLen; /* advance one frame */
+ instr = DSP_INSTR_FADE_TO_BGN;
+ }
+ }
+
+ if (len == 0 && lastModeBGNonly) /* no new data */
+ {
+ BGNonly = 1; /* force BGN this time too */
+ }
+
+#ifdef NETEQ_VAD
+ if ((speechType == TYPE_CNG) /* decoder responded with codec-internal CNG */
+ || ((instr == DSP_INSTR_DO_RFC3389CNG) && (blockLen > 0)) /* ... or, SID frame */
+ || (inst->fs > 16000)) /* ... or, if not NB or WB */
+ {
+ /* disable post-decode VAD upon first sign of send-side DTX/VAD active, or if SWB */
+ inst->VADInst.VADEnabled = 0;
+ inst->VADInst.VADDecision = 1; /* set to always active, just to be on the safe side */
+ inst->VADInst.SIDintervalCounter = 0; /* reset SID interval counter */
+ }
+ else if (!inst->VADInst.VADEnabled) /* VAD disabled and no SID/CNG data observed this time */
+ {
+ inst->VADInst.SIDintervalCounter++; /* increase counter */
+ }
+
+ /* check for re-enabling the VAD */
+ if (inst->VADInst.SIDintervalCounter >= POST_DECODE_VAD_AUTO_ENABLE)
+ {
+ /*
+ * It's been a while since the last CNG/SID frame was observed => re-enable VAD.
+ * (Do not care to look for a VAD instance, since this is done inside the init
+ * function)
+ */
+ WebRtcNetEQ_InitVAD(&inst->VADInst, inst->fs);
+ }
+
+ if (len > 0 /* if we decoded any data */
+ && inst->VADInst.VADEnabled /* and VAD enabled */
+ && inst->fs <= 16000) /* can only do VAD for NB and WB */
+ {
+ int VADframeSize; /* VAD frame size in ms */
+ int VADSamplePtr = 0;
+
+ inst->VADInst.VADDecision = 0;
+
+ if (inst->VADInst.VADFunction != NULL) /* make sure that VAD function is provided */
+ {
+ /* divide the data into groups, as large as possible */
+ for (VADframeSize = 30; VADframeSize >= 10; VADframeSize -= 10)
+ {
+ /* loop through 30, 20, 10 */
+
+ while (inst->VADInst.VADDecision == 0
+ && len - VADSamplePtr >= VADframeSize * fs_mult * 8)
+ {
+ /*
+ * Only continue until first active speech found, and as long as there is
+ * one VADframeSize left.
+ */
+
+ /* call VAD with new decoded data */
+ inst->VADInst.VADDecision |= inst->VADInst.VADFunction(
+ inst->VADInst.VADState, (WebRtc_Word16) inst->fs,
+ (WebRtc_Word16 *) &pw16_decoded_buffer[VADSamplePtr],
+ (WebRtc_Word16) (VADframeSize * fs_mult * 8));
+
+ VADSamplePtr += VADframeSize * fs_mult * 8; /* increment sample counter */
+ }
+ }
+ }
+ else
+ { /* VAD function is NULL */
+ inst->VADInst.VADDecision = 1; /* set decision to active */
+ inst->VADInst.VADEnabled = 0; /* disable VAD since we have no VAD function */
+ }
+
+ }
+#endif /* NETEQ_VAD */
+
+ /* Adjust timestamp if needed */
+ uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[1];
+ inst->endTimestamp += (((WebRtc_UWord32) uw16_tmp) << 16);
+ uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[2];
+ inst->endTimestamp += uw16_tmp;
+
+ if (BGNonly && len > 0)
+ {
+ /*
+ * If BGN mode, we did not produce any data at decoding.
+ * Do it now instead.
+ */
+
+ WebRtcNetEQ_GenerateBGN(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_decoded_buffer, len);
+ }
+
+ /* Switch on the instruction received from the MCU side. */
+ switch (instr)
+ {
+ case DSP_INSTR_NORMAL:
+
+ /* Allow for signal processing to apply gain-back etc */
+ WebRtcNetEQ_Normal(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
+#endif
+ pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
+
+ /* If last packet was decoded as a inband CNG set mode to CNG instead */
+ if ((speechType == TYPE_CNG) || ((inst->w16_mode == MODE_CODEC_INTERNAL_CNG)
+ && (len == 0)))
+ {
+ inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
+ }
+
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ break;
+ case DSP_INSTR_NORMAL_ONE_DESC:
+
+ /* Allow for signal processing to apply gain-back etc */
+ WebRtcNetEQ_Normal(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
+#endif
+ pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ inst->w16_mode = MODE_ONE_DESCRIPTOR;
+ break;
+ case DSP_INSTR_MERGE:
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = -len;
+#endif
+ /* Call Merge with history*/
+ return_value = WebRtcNetEQ_Merge(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_MERGE,
+#endif
+ pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
+
+ if (return_value < 0)
+ {
+ /* error */
+ return return_value;
+ }
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var += len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ /* If last packet was decoded as a inband CNG set mode to CNG instead */
+ if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ break;
+
+ case DSP_INSTR_EXPAND:
+ len = 0;
+ pos = 0;
+ while ((inst->endPosition - inst->curPosition - inst->ExpandInst.w16_overlap + pos)
+ < (inst->timestampsPerCall))
+ {
+ return_value = WebRtcNetEQ_Expand(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_NetEqAlgorithm_buffer, &len, BGNonly);
+ if (return_value < 0)
+ {
+ /* error */
+ return return_value;
+ }
+
+ /*
+ * Update buffer, but only end part (otherwise expand state is destroyed
+ * since it reuses speechBuffer[] memory
+ */
+
+ WEBRTC_SPL_MEMMOVE_W16(inst->pw16_speechHistory,
+ inst->pw16_speechHistory + len,
+ (inst->w16_speechHistoryLen-len));
+ WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[inst->w16_speechHistoryLen-len],
+ pw16_NetEqAlgorithm_buffer, len);
+
+ inst->curPosition -= len;
+
+ /* Update variables for VQmon */
+ inst->w16_concealedTS += len;
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ len = 0; /* already written the data, so do not write it again further down. */
+ }
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ break;
+
+ case DSP_INSTR_ACCELERATE:
+ if (len < 3 * 80 * fs_mult)
+ {
+ /* We need to move data from the speechBuffer[] in order to get 30 ms */
+ borrowedSamples = 3 * 80 * fs_mult - len;
+
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples],
+ pw16_decoded_buffer, len);
+ WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer,
+ &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
+ borrowedSamples);
+
+ return_value = WebRtcNetEQ_Accelerate(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
+#endif
+ pw16_decoded_buffer, 3 * inst->timestampsPerCall,
+ pw16_NetEqAlgorithm_buffer, &len, BGNonly);
+
+ if (return_value < 0)
+ {
+ /* error */
+ return return_value;
+ }
+
+ /* Copy back samples to the buffer */
+ if (len < borrowedSamples)
+ {
+ /*
+ * This destroys the beginning of the buffer, but will not cause any
+ * problems
+ */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples],
+ pw16_NetEqAlgorithm_buffer, len);
+ WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[borrowedSamples-len],
+ inst->speechBuffer,
+ (inst->endPosition-(borrowedSamples-len)));
+
+ inst->curPosition += (borrowedSamples - len);
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = 3 * inst->timestampsPerCall - len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ len = 0;
+ }
+ else
+ {
+ WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples],
+ pw16_NetEqAlgorithm_buffer, borrowedSamples);
+ WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer,
+ &pw16_NetEqAlgorithm_buffer[borrowedSamples],
+ (len-borrowedSamples));
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = 3 * inst->timestampsPerCall - len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ len = len - borrowedSamples;
+ }
+
+ }
+ else
+ {
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = len;
+#endif
+ return_value = WebRtcNetEQ_Accelerate(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
+#endif
+ pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len, BGNonly);
+
+ if (return_value < 0)
+ {
+ /* error */
+ return return_value;
+ }
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var -= len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ }
+ /* If last packet was decoded as a inband CNG set mode to CNG instead */
+ if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ break;
+
+ case DSP_INSTR_DO_RFC3389CNG:
+#ifdef NETEQ_CNG_CODEC
+ if (blockLen > 0)
+ {
+ if (WebRtcCng_UpdateSid(inst->CNG_Codec_inst, (WebRtc_UWord8*) blockPtr,
+ payloadLen) < 0)
+ {
+ /* error returned from CNG function */
+ return_value = -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
+ len = inst->timestampsPerCall;
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
+ break;
+ }
+ }
+
+ if (BGNonly)
+ {
+ /* Get data from BGN function instead of CNG */
+ len = WebRtcNetEQ_GenerateBGN(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_NetEqAlgorithm_buffer, inst->timestampsPerCall);
+ if (len != inst->timestampsPerCall)
+ {
+ /* this is not good, treat this as an error */
+ return_value = -1;
+ }
+ }
+ else
+ {
+ return_value = WebRtcNetEQ_Cng(inst, pw16_NetEqAlgorithm_buffer,
+ inst->timestampsPerCall);
+ }
+ len = inst->timestampsPerCall;
+ inst->ExpandInst.w16_consecExp = 0;
+ inst->w16_mode = MODE_RFC3389CNG;
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+
+ if (return_value < 0)
+ {
+ /* error returned */
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
+ }
+
+ break;
+#else
+ return FAULTY_INSTRUCTION;
+#endif
+ case DSP_INSTR_DO_CODEC_INTERNAL_CNG:
+ /*
+ * This represents the case when there is no transmission and the decoder should
+ * do internal CNG.
+ */
+ len = 0;
+ if (inst->codec_ptr_inst.funcDecode != NULL && !BGNonly)
+ {
+ len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state,
+ blockPtr, 0, pw16_decoded_buffer, &speechType);
+ }
+ else
+ {
+ /* get BGN data */
+ len = WebRtcNetEQ_GenerateBGN(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_decoded_buffer, inst->timestampsPerCall);
+ }
+ WebRtcNetEQ_Normal(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
+#endif
+ pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
+ inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
+ inst->ExpandInst.w16_consecExp = 0;
+ break;
+
+ case DSP_INSTR_DTMF_GENERATE:
+#ifdef NETEQ_ATEVENT_DECODE
+ dtmfSwitch = 0;
+ if ((inst->w16_mode != MODE_DTMF) && (inst->DTMFInst.reinit == 0))
+ {
+ /* Special case; see below.
+ * We must catch this before calling DTMFGenerate,
+ * since reinit is set to 0 in that call.
+ */
+ dtmfSwitch = 1;
+ }
+
+ len = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
+ pw16_NetEqAlgorithm_buffer, inst->fs, -1);
+ if (len < 0)
+ {
+ /* error occurred */
+ return_value = len;
+ len = inst->timestampsPerCall;
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
+ }
+
+ if (dtmfSwitch == 1)
+ {
+ /*
+ * This is the special case where the previous operation was DTMF overdub.
+ * but the current instruction is "regular" DTMF. We must make sure that the
+ * DTMF does not have any discontinuities. The first DTMF sample that we
+ * generate now must be played out immediately, wherefore it must be copied to
+ * the speech buffer.
+ */
+
+ /*
+ * Generate extra DTMF data to fill the space between
+ * curPosition and endPosition
+ */
+ WebRtc_Word16 tempLen;
+
+ tempLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
+ &pw16_NetEqAlgorithm_buffer[len], inst->fs,
+ inst->endPosition - inst->curPosition);
+ if (tempLen < 0)
+ {
+ /* error occurred */
+ return_value = tempLen;
+ len = inst->endPosition - inst->curPosition;
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0,
+ inst->endPosition - inst->curPosition);
+ }
+
+ /* Add to total length */
+ len += tempLen;
+
+ /* Overwrite the "future" part of the speech buffer with the new DTMF data */
+
+ WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition],
+ pw16_NetEqAlgorithm_buffer,
+ inst->endPosition - inst->curPosition);
+
+ /* Shuffle the remaining data to the beginning of algorithm buffer */
+ len -= (inst->endPosition - inst->curPosition);
+ WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer,
+ &pw16_NetEqAlgorithm_buffer[inst->endPosition - inst->curPosition],
+ len);
+ }
+
+ inst->endTimestamp += inst->timestampsPerCall;
+ inst->DTMFInst.reinit = 0;
+ inst->ExpandInst.w16_consecExp = 0;
+ inst->w16_mode = MODE_DTMF;
+ BGNonly = 0; /* override BGN only and let DTMF through */
+
+ playDtmf = 0; /* set to zero because the DTMF is already in the Algorithm buffer */
+ /*
+ * If playDtmf is 1, an extra DTMF vector will be generated and overdubbed
+ * on the output.
+ */
+
+#ifdef NETEQ_STEREO
+ if (msInfo->msMode == NETEQ_MASTER)
+ {
+ /* signal to slave that master is using DTMF only */
+ msInfo->extraInfo = DTMF_ONLY;
+ }
+#endif
+
+ break;
+#else
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return FAULTY_INSTRUCTION;
+#endif
+
+ case DSP_INSTR_DO_ALTERNATIVE_PLC:
+ if (inst->codec_ptr_inst.funcDecodePLC != 0)
+ {
+ len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
+ pw16_NetEqAlgorithm_buffer, 1);
+ }
+ else
+ {
+ len = inst->timestampsPerCall;
+ /* ZeroStuffing... */
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
+ }
+ inst->ExpandInst.w16_consecExp = 0;
+ break;
+ case DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS:
+ if (inst->codec_ptr_inst.funcDecodePLC != 0)
+ {
+ len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
+ pw16_NetEqAlgorithm_buffer, 1);
+ }
+ else
+ {
+ len = inst->timestampsPerCall;
+ /* ZeroStuffing... */
+ WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
+ }
+ inst->ExpandInst.w16_consecExp = 0;
+ inst->endTimestamp += len;
+ break;
+ case DSP_INSTR_DO_AUDIO_REPETITION:
+ len = inst->timestampsPerCall;
+ /* copy->paste... */
+ WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer,
+ &inst->speechBuffer[inst->endPosition-len], len);
+ inst->ExpandInst.w16_consecExp = 0;
+ break;
+ case DSP_INSTR_DO_AUDIO_REPETITION_INC_TS:
+ len = inst->timestampsPerCall;
+ /* copy->paste... */
+ WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer,
+ &inst->speechBuffer[inst->endPosition-len], len);
+ inst->ExpandInst.w16_consecExp = 0;
+ inst->endTimestamp += len;
+ break;
+
+ case DSP_INSTR_PREEMPTIVE_EXPAND:
+ if (len < 3 * inst->timestampsPerCall)
+ {
+ /* borrow samples from sync buffer if necessary */
+ borrowedSamples = 3 * inst->timestampsPerCall - len; /* borrow this many samples */
+ /* calculate how many of these are already played out */
+ oldBorrowedSamples = WEBRTC_SPL_MAX(0,
+ borrowedSamples - (inst->endPosition - inst->curPosition));
+ WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples],
+ pw16_decoded_buffer, len);
+ WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer,
+ &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
+ borrowedSamples);
+ }
+ else
+ {
+ borrowedSamples = 0;
+ oldBorrowedSamples = 0;
+ }
+
+#ifdef NETEQ_DELAY_LOGGING
+ w16_tmp1 = len;
+#endif
+ /* do the expand */
+ return_value = WebRtcNetEQ_PreEmptiveExpand(inst,
+#ifdef SCRATCH
+ /* use same scratch memory as Accelerate */
+ pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
+#endif
+ pw16_decoded_buffer, len + borrowedSamples, oldBorrowedSamples,
+ pw16_NetEqAlgorithm_buffer, &len, BGNonly);
+
+ if (return_value < 0)
+ {
+ /* error */
+ return return_value;
+ }
+
+ if (borrowedSamples > 0)
+ {
+ /* return borrowed samples */
+
+ /* Copy back to last part of speechBuffer from beginning of output buffer */
+ WEBRTC_SPL_MEMCPY_W16( &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
+ pw16_NetEqAlgorithm_buffer,
+ borrowedSamples);
+
+ len -= borrowedSamples; /* remove the borrowed samples from new total length */
+
+ /* Move to beginning of output buffer from end of output buffer */
+ WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer,
+ &pw16_NetEqAlgorithm_buffer[borrowedSamples],
+ len);
+ }
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = len - w16_tmp1; /* number of samples added */
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+ /* If last packet was decoded as inband CNG, set mode to CNG instead */
+ if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
+#ifdef NETEQ_ATEVENT_DECODE
+ if (playDtmf == 0)
+ {
+ inst->DTMFInst.reinit = 1;
+ }
+#endif
+ break;
+
+ case DSP_INSTR_FADE_TO_BGN:
+ {
+ int tempReturnValue;
+ /* do not overwrite return_value, since it likely contains an error code */
+
+ /* calculate interpolation length */
+ w16_tmp3 = WEBRTC_SPL_MIN(inst->endPosition - inst->curPosition,
+ inst->timestampsPerCall);
+ /* check that it will fit in pw16_NetEqAlgorithm_buffer */
+ if (w16_tmp3 + inst->w16_frameLen > NETEQ_MAX_OUTPUT_SIZE)
+ {
+ w16_tmp3 = NETEQ_MAX_OUTPUT_SIZE - inst->w16_frameLen;
+ }
+
+ /* call Expand */
+ len = inst->timestampsPerCall + inst->ExpandInst.w16_overlap;
+ pos = 0;
+
+ tempReturnValue = WebRtcNetEQ_Expand(inst,
+#ifdef SCRATCH
+ pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
+#endif
+ pw16_NetEqAlgorithm_buffer, &len, 1);
+
+ if (tempReturnValue < 0)
+ {
+ /* error */
+ /* this error value will override return_value */
+ return tempReturnValue;
+ }
+
+ pos += len; /* got len samples from expand */
+
+ /* copy to fill the demand */
+ while (pos + len <= inst->w16_frameLen + w16_tmp3)
+ {
+ WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos],
+ pw16_NetEqAlgorithm_buffer, len);
+ pos += len;
+ }
+
+ /* fill with fraction of the expand vector if needed */
+ if (pos < inst->w16_frameLen + w16_tmp3)
+ {
+ WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos], pw16_NetEqAlgorithm_buffer,
+ inst->w16_frameLen + w16_tmp3 - pos);
+ }
+
+ len = inst->w16_frameLen + w16_tmp3; /* truncate any surplus samples since we don't want these */
+
+ /*
+ * Mix with contents in sync buffer. Find largest power of two that is less than
+ * interpolate length divide 16384 with this number; result is in w16_tmp2.
+ */
+ w16_tmp1 = 2;
+ w16_tmp2 = 16384;
+ while (w16_tmp1 <= w16_tmp3)
+ {
+ w16_tmp2 >>= 1; /* divide with 2 */
+ w16_tmp1 <<= 1; /* increase with a factor of 2 */
+ }
+
+ w16_tmp1 = 0;
+ pos = 0;
+ while (w16_tmp1 < 16384)
+ {
+ inst->speechBuffer[inst->curPosition + pos]
+ =
+ (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
+ WEBRTC_SPL_MUL_16_16( inst->speechBuffer[inst->endPosition - w16_tmp3 + pos],
+ 16384-w16_tmp1 ) +
+ WEBRTC_SPL_MUL_16_16( pw16_NetEqAlgorithm_buffer[pos], w16_tmp1 ),
+ 14 );
+ w16_tmp1 += w16_tmp2;
+ pos++;
+ }
+
+ /* overwrite remainder of speech buffer */
+
+ WEBRTC_SPL_MEMCPY_W16( &inst->speechBuffer[inst->endPosition - w16_tmp3 + pos],
+ &pw16_NetEqAlgorithm_buffer[pos], w16_tmp3 - pos);
+
+ len -= w16_tmp3;
+ /* shift algorithm buffer */
+
+ WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer,
+ &pw16_NetEqAlgorithm_buffer[w16_tmp3],
+ len );
+
+ /* Update variables for VQmon */
+ inst->w16_concealedTS += len;
+
+ inst->w16_mode = MODE_FADE_TO_BGN;
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+ temp_var = len;
+ fwrite(&temp_var, sizeof(int), 1, delay_fid2);
+#endif
+
+ break;
+ }
+
+ default:
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return FAULTY_INSTRUCTION;
+ } /* end of grand switch */
+
+ /* Copy data directly to output buffer */
+
+ w16_tmp2 = 0;
+ if ((inst->endPosition + len - inst->curPosition - inst->ExpandInst.w16_overlap)
+ >= inst->timestampsPerCall)
+ {
+ w16_tmp2 = inst->endPosition - inst->curPosition;
+ w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2, 0); /* Additional error protection, just in case */
+ w16_tmp1 = WEBRTC_SPL_MIN(w16_tmp2, inst->timestampsPerCall);
+ w16_tmp2 = inst->timestampsPerCall - w16_tmp1;
+ WEBRTC_SPL_MEMCPY_W16(pw16_outData, &inst->speechBuffer[inst->curPosition], w16_tmp1);
+ WEBRTC_SPL_MEMCPY_W16(&pw16_outData[w16_tmp1], pw16_NetEqAlgorithm_buffer, w16_tmp2);
+ DataEnough = 1;
+ }
+ else
+ {
+ DataEnough = 0;
+ }
+
+ if (playDtmf != 0)
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ WebRtc_Word16 outDataIndex = 0;
+ WebRtc_Word16 overdubLen = -1; /* default len */
+ WebRtc_Word16 dtmfLen;
+
+ /*
+ * Overdub the output with DTMF. Note that this is not executed if the
+ * DSP_INSTR_DTMF_GENERATE operation is performed above.
+ */
+ if (inst->DTMFInst.lastDtmfSample - inst->curPosition > 0)
+ {
+ /* special operation for transition from "DTMF only" to "DTMF overdub" */
+ outDataIndex
+ = WEBRTC_SPL_MIN(inst->DTMFInst.lastDtmfSample - inst->curPosition,
+ inst->timestampsPerCall);
+ overdubLen = inst->timestampsPerCall - outDataIndex;
+ }
+
+ dtmfLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
+ &pw16_outData[outDataIndex], inst->fs, overdubLen);
+ if (dtmfLen < 0)
+ {
+ /* error occurred */
+ return_value = dtmfLen;
+ }
+ inst->DTMFInst.reinit = 0;
+#else
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return FAULTY_INSTRUCTION;
+#endif
+ }
+
+ /*
+ * Shuffle speech buffer to allow more data. Move data from pw16_NetEqAlgorithm_buffer
+ * to speechBuffer.
+ */
+ if (instr != DSP_INSTR_EXPAND)
+ {
+ w16_tmp1 = WEBRTC_SPL_MIN(inst->endPosition, len);
+ WEBRTC_SPL_MEMMOVE_W16(inst->speechBuffer, inst->speechBuffer + w16_tmp1,
+ (inst->endPosition-w16_tmp1));
+ WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-w16_tmp1],
+ &pw16_NetEqAlgorithm_buffer[len-w16_tmp1], w16_tmp1);
+#ifdef NETEQ_ATEVENT_DECODE
+ /* Update index to end of DTMF data in speech buffer */
+ if (instr == DSP_INSTR_DTMF_GENERATE)
+ {
+ /* We have written DTMF data to the end of speech buffer */
+ inst->DTMFInst.lastDtmfSample = inst->endPosition;
+ }
+ else if (inst->DTMFInst.lastDtmfSample > 0)
+ {
+ /* The end of DTMF data in speech buffer has been shuffled */
+ inst->DTMFInst.lastDtmfSample -= w16_tmp1;
+ }
+#endif
+ /*
+ * Update the BGN history if last operation was not expand (nor Merge, Accelerate
+ * or Pre-emptive expand, to save complexity).
+ */
+ if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_MERGE)
+ && (inst->w16_mode != MODE_SUCCESS_ACCELERATE) && (inst->w16_mode
+ != MODE_LOWEN_ACCELERATE) && (inst->w16_mode != MODE_SUCCESS_PREEMPTIVE)
+ && (inst->w16_mode != MODE_LOWEN_PREEMPTIVE) && (inst->w16_mode
+ != MODE_FADE_TO_BGN) && (inst->w16_mode != MODE_DTMF) && (!BGNonly))
+ {
+ WebRtcNetEQ_BGNUpdate(inst
+#ifdef SCRATCH
+ , pw16_scratchPtr + SCRATCH_NETEQ_BGN_UPDATE
+#endif
+ );
+ }
+ }
+ else /* instr == DSP_INSTR_EXPAND */
+ {
+ /* Nothing should be done since data is already copied to output. */
+ }
+
+ inst->curPosition -= len;
+
+ /*
+ * Extra protection in case something should go totally wrong in terms of sizes...
+ * If everything is ok this should NEVER happen.
+ */
+ if (inst->curPosition < -inst->timestampsPerCall)
+ {
+ inst->curPosition = -inst->timestampsPerCall;
+ }
+
+ if ((instr != DSP_INSTR_EXPAND) && (instr != DSP_INSTR_MERGE) && (instr
+ != DSP_INSTR_FADE_TO_BGN))
+ {
+ /* Reset concealed TS parameter if it does not seem to have been flushed */
+ if (inst->w16_concealedTS > inst->timestampsPerCall)
+ {
+ inst->w16_concealedTS = 0;
+ }
+ }
+
+ /*
+ * Double-check that we actually have 10 ms to play. If we haven't, there has been a
+ * serious error.The decoder might have returned way too few samples
+ */
+ if (!DataEnough)
+ {
+ /* This should not happen. Set outdata to zeros, and return error. */
+ WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall);
+ *pw16_len = inst->timestampsPerCall;
+ inst->w16_mode = MODE_ERROR;
+ dspInfo->lastMode = MODE_ERROR;
+ return RECOUT_ERROR_SAMPLEUNDERRUN;
+ }
+
+ /*
+ * Update Videosync timestamp (this special timestamp is needed since the endTimestamp
+ * stops during CNG and Expand periods.
+ */
+ if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_RFC3389CNG))
+ {
+ WebRtc_UWord32 uw32_tmpTS;
+ uw32_tmpTS = inst->endTimestamp - (inst->endPosition - inst->curPosition);
+ if ((WebRtc_Word32) (uw32_tmpTS - inst->videoSyncTimestamp) > 0)
+ {
+ inst->videoSyncTimestamp = uw32_tmpTS;
+ }
+ }
+ else
+ {
+ inst->videoSyncTimestamp += inst->timestampsPerCall;
+ }
+
+ /* After this, regardless of what has happened, deliver 10 ms of future data */
+ inst->curPosition += inst->timestampsPerCall;
+ *pw16_len = inst->timestampsPerCall;
+
+ /* Remember if BGNonly was used */
+ if (BGNonly)
+ {
+ inst->w16_mode |= MODE_BGN_ONLY;
+ }
+
+ return return_value;
+}
+
+#undef SCRATCH_ALGORITHM_BUFFER
+#undef SCRATCH_NETEQ_NORMAL
+#undef SCRATCH_NETEQ_MERGE
+#undef SCRATCH_NETEQ_BGN_UPDATE
+#undef SCRATCH_NETEQ_EXPAND
+#undef SCRATCH_DSP_INFO
+#undef SCRATCH_NETEQ_ACCELERATE
+#undef SIZE_SCRATCH_BUFFER
diff --git a/trunk/src/modules/audio_coding/neteq/rtcp.c b/trunk/src/modules/audio_coding/neteq/rtcp.c
new file mode 100644
index 0000000..35f73da
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/rtcp.c
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of RTCP statistics reporting.
+ */
+
+#include "rtcp.h"
+
+#include <string.h>
+
+#include "signal_processing_library.h"
+
+int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo)
+{
+ /*
+ * Initialize everything to zero and then set the start values for the RTP packet stream.
+ */
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) RTCP_inst, 0,
+ sizeof(WebRtcNetEQ_RTCP_t) / sizeof(WebRtc_Word16));
+ RTCP_inst->base_seq = uw16_seqNo;
+ RTCP_inst->max_seq = uw16_seqNo;
+ return 0;
+}
+
+int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
+ WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime)
+{
+ WebRtc_Word16 w16_SeqDiff;
+ WebRtc_Word32 w32_TimeDiff;
+ WebRtc_Word32 w32_JitterDiff;
+
+ /*
+ * Update number of received packets, and largest packet number received.
+ */
+ RTCP_inst->received++;
+ w16_SeqDiff = uw16_seqNo - RTCP_inst->max_seq;
+ if (w16_SeqDiff >= 0)
+ {
+ if (uw16_seqNo < RTCP_inst->max_seq)
+ {
+ /* Wrap around detected */
+ RTCP_inst->cycles++;
+ }
+ RTCP_inst->max_seq = uw16_seqNo;
+ }
+
+ /* Calculate Jitter, and update previous timestamps */
+ /* Note that the value in RTCP_inst->jitter is in Q4. */
+ if (RTCP_inst->received > 1)
+ {
+ w32_TimeDiff = (uw32_recTime - (uw32_timeStamp - RTCP_inst->transit));
+ w32_TimeDiff = WEBRTC_SPL_ABS_W32(w32_TimeDiff);
+ w32_JitterDiff = WEBRTC_SPL_LSHIFT_W16(w32_TimeDiff, 4) - RTCP_inst->jitter;
+ RTCP_inst->jitter = RTCP_inst->jitter + WEBRTC_SPL_RSHIFT_W32((w32_JitterDiff + 8), 4);
+ }
+ RTCP_inst->transit = (uw32_timeStamp - uw32_recTime);
+ return 0;
+}
+
+int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
+ WebRtc_UWord16 *puw16_fraction_lost,
+ WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
+ WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset)
+{
+ WebRtc_UWord32 uw32_exp_nr, uw32_exp_interval, uw32_rec_interval;
+ WebRtc_Word32 w32_lost;
+
+ /* Extended highest sequence number received */
+ *puw32_ext_max
+ = (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)RTCP_inst->cycles, 16)
+ + RTCP_inst->max_seq;
+
+ /*
+ * Calculate expected number of packets and compare it to the number of packets that
+ * were actually received => the cumulative number of packets lost can be extracted.
+ */
+ uw32_exp_nr = *puw32_ext_max - RTCP_inst->base_seq + 1;
+ if (RTCP_inst->received == 0)
+ {
+ /* no packets received, assume none lost */
+ *puw32_cum_lost = 0;
+ }
+ else if (uw32_exp_nr > RTCP_inst->received)
+ {
+ *puw32_cum_lost = uw32_exp_nr - RTCP_inst->received;
+ if (*puw32_cum_lost > (WebRtc_UWord32) 0xFFFFFF)
+ {
+ *puw32_cum_lost = 0xFFFFFF;
+ }
+ }
+ else
+ {
+ *puw32_cum_lost = 0;
+ }
+
+ /* Fraction lost (Since last report) */
+ uw32_exp_interval = uw32_exp_nr - RTCP_inst->exp_prior;
+ if (!doNotReset)
+ {
+ RTCP_inst->exp_prior = uw32_exp_nr;
+ }
+ uw32_rec_interval = RTCP_inst->received - RTCP_inst->rec_prior;
+ if (!doNotReset)
+ {
+ RTCP_inst->rec_prior = RTCP_inst->received;
+ }
+ w32_lost = (WebRtc_Word32) (uw32_exp_interval - uw32_rec_interval);
+ if (uw32_exp_interval == 0 || w32_lost <= 0 || RTCP_inst->received == 0)
+ {
+ *puw16_fraction_lost = 0;
+ }
+ else
+ {
+ *puw16_fraction_lost = (WebRtc_UWord16) (WEBRTC_SPL_LSHIFT_W32(w32_lost, 8)
+ / uw32_exp_interval);
+ }
+ if (*puw16_fraction_lost > 0xFF)
+ {
+ *puw16_fraction_lost = 0xFF;
+ }
+
+ /* Inter-arrival jitter */
+ *puw32_jitter = (RTCP_inst->jitter) >> 4; /* scaling from Q4 */
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/rtcp.h b/trunk/src/modules/audio_coding/neteq/rtcp.h
new file mode 100644
index 0000000..009e019
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/rtcp.h
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * RTCP statistics reporting.
+ */
+
+#ifndef RTCP_H
+#define RTCP_H
+
+#include "typedefs.h"
+
+typedef struct
+{
+ WebRtc_UWord16 cycles; /* The number of wrap-arounds for the sequence number */
+ WebRtc_UWord16 max_seq; /* The maximum sequence number received
+ (starts from 0 again after wrap around) */
+ WebRtc_UWord16 base_seq; /* The sequence number of the first packet that arrived */
+ WebRtc_UWord32 received; /* The number of packets that has been received */
+ WebRtc_UWord32 rec_prior; /* Number of packets received when last report was generated */
+ WebRtc_UWord32 exp_prior; /* Number of packets that should have been received if no
+ packets were lost. Stored value from last report. */
+ WebRtc_UWord32 jitter; /* Jitter statistics at this instance (calculated according to RFC) */
+ WebRtc_Word32 transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */
+} WebRtcNetEQ_RTCP_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_RTCPInit(...)
+ *
+ * This function calculates the parameters that are needed for the RTCP
+ * report.
+ *
+ * Input:
+ * - RTCP_inst : RTCP instance, that contains information about the
+ * packets that have been received etc.
+ * - seqNo : Packet number of the first received frame.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo);
+
+/****************************************************************************
+ * WebRtcNetEQ_RTCPUpdate(...)
+ *
+ * This function calculates the parameters that are needed for the RTCP
+ * report.
+ *
+ * Input:
+ * - RTCP_inst : RTCP instance, that contains information about the
+ * packets that have been received etc.
+ * - seqNo : Packet number of the first received frame.
+ * - timeStamp : Time stamp from the RTP header.
+ * - recTime : Time (in RTP timestamps) when this packet was received.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
+ WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime);
+
+/****************************************************************************
+ * WebRtcNetEQ_RTCPGetStats(...)
+ *
+ * This function calculates the parameters that are needed for the RTCP
+ * report.
+ *
+ * Input:
+ * - RTCP_inst : RTCP instance, that contains information about the
+ * packets that have been received etc.
+ * - doNotReset : If non-zero, the fraction lost statistics will not
+ * be reset.
+ *
+ * Output:
+ * - RTCP_inst : Updated RTCP information (some statistics are
+ * reset when generating this report)
+ * - fraction_lost : Number of lost RTP packets divided by the number of
+ * expected packets, since the last RTCP Report.
+ * - cum_lost : Cumulative number of lost packets during this
+ * session.
+ * - ext_max : Extended highest sequence number received.
+ * - jitter : Inter-arrival jitter.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
+ WebRtc_UWord16 *puw16_fraction_lost,
+ WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
+ WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset);
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/rtp.c b/trunk/src/modules/audio_coding/neteq/rtp.c
new file mode 100644
index 0000000..bd4f9a2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/rtp.c
@@ -0,0 +1,240 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * RTP related functions.
+ */
+
+#include "rtp.h"
+
+#include "typedefs.h" /* to define endianness */
+
+#include "neteq_error_codes.h"
+
+int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
+ RTPPacket_t* RTPheader)
+{
+ int i_P, i_X, i_CC, i_startPosition;
+ int i_IPver;
+ int i_extlength = -1; /* Default value is there is no extension */
+ int i_padlength = 0; /* Default value if there is no padding */
+
+ if (i_DatagramLen < 12)
+ {
+ return RTP_TOO_SHORT_PACKET;
+ }
+
+#ifdef WEBRTC_BIG_ENDIAN
+ i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */
+ i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */
+ i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */
+ i_CC = ((WebRtc_UWord16) (pw16_Datagram[0] >> 8) & 0xF); /* Get the CC number */
+ RTPheader->payloadType = pw16_Datagram[0] & 0x7F; /* Get the coder type */
+ RTPheader->seqNumber = pw16_Datagram[1]; /* Get the sequence number */
+ RTPheader->timeStamp = ((((WebRtc_UWord32) ((WebRtc_UWord16) pw16_Datagram[2])) << 16)
+ | (WebRtc_UWord16) (pw16_Datagram[3])); /* Get timestamp */
+ RTPheader->ssrc = (((WebRtc_UWord32) pw16_Datagram[4]) << 16)
+ + (((WebRtc_UWord32) pw16_Datagram[5])); /* Get the SSRC */
+
+ if (i_X == 1)
+ {
+ /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
+ i_extlength = pw16_Datagram[7 + 2 * i_CC];
+ }
+ if (i_P == 1)
+ {
+ /* Padding exists. Find out how many bytes the padding consists of. */
+ if (i_DatagramLen & 0x1)
+ {
+ /* odd number of bytes => last byte in higher byte */
+ i_padlength = (((WebRtc_UWord16) pw16_Datagram[i_DatagramLen >> 1]) >> 8);
+ }
+ else
+ {
+ /* even number of bytes => last byte in lower byte */
+ i_padlength = ((pw16_Datagram[(i_DatagramLen >> 1) - 1]) & 0xFF);
+ }
+ }
+#else /* WEBRTC_LITTLE_ENDIAN */
+ i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */
+ i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */
+ i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */
+ i_CC = (WebRtc_UWord16) (pw16_Datagram[0] & 0xF); /* Get the CC number */
+ RTPheader->payloadType = (pw16_Datagram[0] >> 8) & 0x7F; /* Get the coder type */
+ RTPheader->seqNumber = (((((WebRtc_UWord16) pw16_Datagram[1]) >> 8) & 0xFF)
+ | (((WebRtc_UWord16) (pw16_Datagram[1] & 0xFF)) << 8)); /* Get the packet number */
+ RTPheader->timeStamp = ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF) << 24)
+ | ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF00) << 8)
+ | ((((WebRtc_UWord16) pw16_Datagram[3]) >> 8) & 0xFF)
+ | ((((WebRtc_UWord16) pw16_Datagram[3]) & 0xFF) << 8); /* Get timestamp */
+ RTPheader->ssrc = ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF) << 24)
+ | ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF00) << 8)
+ | ((((WebRtc_UWord16) pw16_Datagram[5]) >> 8) & 0xFF)
+ | ((((WebRtc_UWord16) pw16_Datagram[5]) & 0xFF) << 8); /* Get the SSRC */
+
+ if (i_X == 1)
+ {
+ /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
+ i_extlength = (((((WebRtc_UWord16) pw16_Datagram[7 + 2 * i_CC]) >> 8) & 0xFF)
+ | (((WebRtc_UWord16) (pw16_Datagram[7 + 2 * i_CC] & 0xFF)) << 8));
+ }
+ if (i_P == 1)
+ {
+ /* Padding exists. Find out how many bytes the padding consists of. */
+ if (i_DatagramLen & 0x1)
+ {
+ /* odd number of bytes => last byte in higher byte */
+ i_padlength = (pw16_Datagram[i_DatagramLen >> 1] & 0xFF);
+ }
+ else
+ {
+ /* even number of bytes => last byte in lower byte */
+ i_padlength = (((WebRtc_UWord16) pw16_Datagram[(i_DatagramLen >> 1) - 1]) >> 8);
+ }
+ }
+#endif
+
+ i_startPosition = 12 + 4 * (i_extlength + 1) + 4 * i_CC;
+ RTPheader->payload = &pw16_Datagram[i_startPosition >> 1];
+ RTPheader->payloadLen = i_DatagramLen - i_startPosition - i_padlength;
+ RTPheader->starts_byte1 = 0;
+
+ if ((i_IPver != 2) || (RTPheader->payloadLen <= 0) || (RTPheader->payloadLen >= 16000)
+ || (i_startPosition < 12) || (i_startPosition > i_DatagramLen))
+ {
+ return RTP_CORRUPT_PACKET;
+ }
+
+ return 0;
+}
+
+#ifdef NETEQ_RED_CODEC
+
+int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
+ int *i_No_Of_Payloads)
+{
+ const WebRtc_Word16 *pw16_data = RTPheader[0]->payload; /* Pointer to the data */
+ WebRtc_UWord16 uw16_offsetTimeStamp = 65535, uw16_secondPayload = 65535;
+ int i_blockLength, i_k;
+ int i_discardedBlockLength = 0;
+ int singlePayload = 0;
+
+#ifdef WEBRTC_BIG_ENDIAN
+ if ((pw16_data[0] & 0x8000) == 0)
+ {
+ /* Only one payload in this packet*/
+ singlePayload = 1;
+ /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
+ i_blockLength = -4;
+ RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
+ }
+ else
+ {
+ /* Discard all but the two last payloads. */
+ while (((pw16_data[2] & 0x8000) == 1)&&
+ (pw16_data<((RTPheader[0]->payload)+((RTPheader[0]->payloadLen+1)>>1))))
+ {
+ i_discardedBlockLength += (4+(((WebRtc_UWord16)pw16_data[1]) & 0x3FF));
+ pw16_data+=2;
+ }
+ if (pw16_data>=(RTPheader[0]->payload+((RTPheader[0]->payloadLen+1)>>1)))
+ {
+ return RED_SPLIT_ERROR2; /* Error, we are outside the packet */
+ }
+ singlePayload = 0; /* the packet contains more than one payload */
+ uw16_secondPayload = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
+ RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[2]) & 0x7F00) >> 8);
+ uw16_offsetTimeStamp = ((((WebRtc_UWord16)pw16_data[0]) & 0xFF) << 6) +
+ ((((WebRtc_UWord16)pw16_data[1]) & 0xFC00) >> 10);
+ i_blockLength = (((WebRtc_UWord16)pw16_data[1]) & 0x3FF);
+ }
+#else /* WEBRTC_LITTLE_ENDIAN */
+ if ((pw16_data[0] & 0x80) == 0)
+ {
+ /* Only one payload in this packet */
+ singlePayload = 1;
+ /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
+ i_blockLength = -4;
+ RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
+ }
+ else
+ {
+ /* Discard all but the two last payloads. */
+ while (((pw16_data[2] & 0x80) == 1) && (pw16_data < ((RTPheader[0]->payload)
+ + ((RTPheader[0]->payloadLen + 1) >> 1))))
+ {
+ i_discardedBlockLength += (4 + ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
+ + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8));
+ pw16_data += 2;
+ }
+ if (pw16_data >= (RTPheader[0]->payload + ((RTPheader[0]->payloadLen + 1) >> 1)))
+ {
+ return RED_SPLIT_ERROR2; /* Error, we are outside the packet */;
+ }
+ singlePayload = 0; /* the packet contains more than one payload */
+ uw16_secondPayload = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
+ RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[2]) & 0x7F);
+ uw16_offsetTimeStamp = ((((WebRtc_UWord16) pw16_data[0]) & 0xFF00) >> 2)
+ + ((((WebRtc_UWord16) pw16_data[1]) & 0xFC) >> 2);
+ i_blockLength = ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
+ + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8);
+ }
+#endif
+
+ if (i_MaximumPayloads < 2 || singlePayload == 1)
+ {
+ /* Reject the redundancy; or no redundant payload present. */
+ for (i_k = 1; i_k < i_MaximumPayloads; i_k++)
+ {
+ RTPheader[i_k]->payloadType = -1;
+ RTPheader[i_k]->payloadLen = 0;
+ }
+
+ /* update the pointer for the main data */
+ pw16_data = &pw16_data[(5 + i_blockLength) >> 1];
+ RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1;
+ RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5)
+ - i_discardedBlockLength;
+ RTPheader[0]->payload = pw16_data;
+
+ *i_No_Of_Payloads = 1;
+
+ }
+ else
+ {
+ /* Redundancy accepted, put the redundancy in second RTPheader. */
+ RTPheader[1]->payloadType = uw16_secondPayload;
+ RTPheader[1]->payload = &pw16_data[5 >> 1];
+ RTPheader[1]->starts_byte1 = 5 & 0x1;
+ RTPheader[1]->seqNumber = RTPheader[0]->seqNumber;
+ RTPheader[1]->timeStamp = RTPheader[0]->timeStamp - uw16_offsetTimeStamp;
+ RTPheader[1]->ssrc = RTPheader[0]->ssrc;
+ RTPheader[1]->payloadLen = i_blockLength;
+
+ /* Modify first RTP packet, so that it contains the main data. */
+ RTPheader[0]->payload = &pw16_data[(5 + i_blockLength) >> 1];
+ RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1;
+ RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5)
+ - i_discardedBlockLength;
+
+ /* Clear the following payloads. */
+ for (i_k = 2; i_k < i_MaximumPayloads; i_k++)
+ {
+ RTPheader[i_k]->payloadType = -1;
+ RTPheader[i_k]->payloadLen = 0;
+ }
+
+ *i_No_Of_Payloads = 2;
+ }
+ return 0;
+}
+
+#endif
+
diff --git a/trunk/src/modules/audio_coding/neteq/rtp.h b/trunk/src/modules/audio_coding/neteq/rtp.h
new file mode 100644
index 0000000..8490d62
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/rtp.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * RTP data struct and related functions.
+ */
+
+#ifndef RTP_H
+#define RTP_H
+
+#include "typedefs.h"
+
+#include "codec_db.h"
+
+typedef struct
+{
+ WebRtc_UWord16 seqNumber;
+ WebRtc_UWord32 timeStamp;
+ WebRtc_UWord32 ssrc;
+ int payloadType;
+ const WebRtc_Word16 *payload;
+ WebRtc_Word16 payloadLen;
+ WebRtc_Word16 starts_byte1;
+ WebRtc_Word16 rcuPlCntr;
+} RTPPacket_t;
+
+/****************************************************************************
+ * WebRtcNetEQ_RTPPayloadInfo(...)
+ *
+ * Converts a datagram into an RTP header struct.
+ *
+ * Input:
+ * - Datagram : UDP datagram from the network
+ * - DatagramLen : Length in bytes of the datagram
+ *
+ * Output:
+ * - RTPheader : Structure with the datagram info
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
+ RTPPacket_t* RTPheader);
+
+/****************************************************************************
+ * WebRtcNetEQ_RedundancySplit(...)
+ *
+ * Splits a Redundancy RTP struct into two RTP structs. User has to check
+ * that it's really the redundancy payload. No such check is done inside this
+ * function.
+ *
+ * Input:
+ * - RTPheader : First header holds the whole RTP packet (with the redundancy payload)
+ * - MaximumPayloads:
+ * The maximum number of RTP payloads that should be
+ * extracted (1+maximum_no_of_Redundancies).
+ *
+ * Output:
+ * - RTPheader : First header holds the main RTP data, while 2..N
+ * holds the redundancy data.
+ * - No_Of
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
+ int *i_No_Of_Payloads);
+
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/set_fs.c b/trunk/src/modules/audio_coding/neteq/set_fs.c
new file mode 100644
index 0000000..b2ad5ca
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/set_fs.c
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Function were the sample rate is set.
+ */
+
+#include "mcu.h"
+
+#include "dtmf_buffer.h"
+#include "neteq_error_codes.h"
+
+int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs)
+{
+ WebRtc_Word16 ok = 0;
+
+ switch (fs)
+ {
+ case 8000:
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560);
+#endif
+ inst->timestampsPerCall = inst->millisecondsPerCall * 8;
+ break;
+ }
+
+#ifdef NETEQ_WIDEBAND
+ case 16000:
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 16000, 1120);
+#endif
+ inst->timestampsPerCall = inst->millisecondsPerCall * 16;
+ break;
+ }
+#endif
+
+#ifdef NETEQ_32KHZ_WIDEBAND
+ case 32000:
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 32000, 2240);
+#endif
+ inst->timestampsPerCall = inst->millisecondsPerCall * 32;
+ break;
+ }
+#endif
+
+#ifdef NETEQ_48KHZ_WIDEBAND
+ case 48000:
+ {
+#ifdef NETEQ_ATEVENT_DECODE
+ ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 48000, 3360);
+#endif
+ inst->timestampsPerCall = inst->millisecondsPerCall * 48;
+ break;
+ }
+#endif
+
+ default:
+ {
+ /* Not supported yet */
+ return CODEC_DB_UNSUPPORTED_FS;
+ }
+ } /* end switch */
+
+ inst->fs = fs;
+
+ return ok;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/signal_mcu.c b/trunk/src/modules/audio_coding/neteq/signal_mcu.c
new file mode 100644
index 0000000..0180f38
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/signal_mcu.c
@@ -0,0 +1,766 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Signal the MCU that data is available and ask for a RecOut decision.
+ */
+
+#include "mcu.h"
+
+#include <string.h>
+
+#include "signal_processing_library.h"
+
+#include "automode.h"
+#include "dtmf_buffer.h"
+#include "mcu_dsp_common.h"
+#include "neteq_error_codes.h"
+
+#ifdef NETEQ_DELAY_LOGGING
+#include "delay_logging.h"
+#include <stdio.h>
+
+extern FILE *delay_fid2; /* file pointer to delay log file */
+#endif
+
+
+/*
+ * Signals the MCU that DSP status data is available.
+ */
+int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
+{
+
+ int i_bufferpos, i_res;
+ WebRtc_UWord16 uw16_instr;
+ DSP2MCU_info_t dspInfo;
+ WebRtc_Word16 *blockPtr, blockLen;
+ WebRtc_UWord32 uw32_availableTS;
+ RTPPacket_t temp_pkt;
+ WebRtc_Word32 w32_bufsize, w32_tmp;
+ WebRtc_Word16 payloadType = -1;
+ WebRtc_Word16 wantedNoOfTimeStamps;
+ WebRtc_Word32 totalTS;
+ WebRtc_Word16 oldPT, latePacketExist = 0;
+ WebRtc_UWord32 oldTS, prevTS, uw32_tmp;
+ WebRtc_UWord16 prevSeqNo;
+ WebRtc_Word16 nextSeqNoAvail;
+ WebRtc_Word16 fs_mult, w16_tmp;
+ WebRtc_Word16 lastModeBGNonly = 0;
+#ifdef NETEQ_DELAY_LOGGING
+ int temp_var;
+#endif
+ int playDtmf = 0;
+
+ fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
+
+ /* Increment counter since last statistics report */
+ inst->lastReportTS += inst->timestampsPerCall;
+
+ /* Increment waiting time for all packets. */
+ WebRtcNetEQ_IncrementWaitingTimes(&inst->PacketBuffer_inst);
+
+ /* Read info from DSP so we now current status */
+
+ WEBRTC_SPL_MEMCPY_W8(&dspInfo,inst->pw16_readAddress,sizeof(DSP2MCU_info_t));
+
+ /* Set blockPtr to first payload block */
+ blockPtr = &inst->pw16_writeAddress[3];
+
+ /* Clear instruction word and number of lost samples (2*WebRtc_Word16) */
+ inst->pw16_writeAddress[0] = 0;
+ inst->pw16_writeAddress[1] = 0;
+ inst->pw16_writeAddress[2] = 0;
+
+ if ((dspInfo.lastMode & MODE_AWAITING_CODEC_PTR) != 0)
+ {
+ /*
+ * Make sure state is adjusted so that a codec update is
+ * performed when first packet arrives.
+ */
+ if (inst->new_codec != 1)
+ {
+ inst->current_Codec = -1;
+ }
+ dspInfo.lastMode = (dspInfo.lastMode ^ MODE_AWAITING_CODEC_PTR);
+ }
+
+#ifdef NETEQ_STEREO
+ if ((dspInfo.lastMode & MODE_MASTER_DTMF_SIGNAL) != 0)
+ {
+ playDtmf = 1; /* force DTMF decision */
+ dspInfo.lastMode = (dspInfo.lastMode ^ MODE_MASTER_DTMF_SIGNAL);
+ }
+
+ if ((dspInfo.lastMode & MODE_USING_STEREO) != 0)
+ {
+ if (inst->usingStereo == 0)
+ {
+ /* stereo mode changed; reset automode instance to re-synchronize statistics */
+ WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
+ inst->PacketBuffer_inst.maxInsertPositions);
+ }
+ inst->usingStereo = 1;
+ dspInfo.lastMode = (dspInfo.lastMode ^ MODE_USING_STEREO);
+ }
+ else
+ {
+ inst->usingStereo = 0;
+ }
+#endif
+
+ /* detect if BGN_ONLY flag is set in lastMode */
+ if ((dspInfo.lastMode & MODE_BGN_ONLY) != 0)
+ {
+ lastModeBGNonly = 1; /* remember flag */
+ dspInfo.lastMode ^= MODE_BGN_ONLY; /* clear the flag */
+ }
+
+ if ((dspInfo.lastMode == MODE_RFC3389CNG) || (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG)
+ || (dspInfo.lastMode == MODE_EXPAND))
+ {
+ /*
+ * If last mode was CNG (or Expand, since this could be covering up for a lost CNG
+ * packet), increase the CNGplayedTS counter.
+ */
+ inst->BufferStat_inst.uw32_CNGplayedTS += inst->timestampsPerCall;
+
+ if (dspInfo.lastMode == MODE_RFC3389CNG)
+ {
+ /* remember that RFC3389CNG is on (needed if CNG is interrupted by DTMF) */
+ inst->BufferStat_inst.w16_cngOn = CNG_RFC3389_ON;
+ }
+ else if (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG)
+ {
+ /* remember that internal CNG is on (needed if CNG is interrupted by DTMF) */
+ inst->BufferStat_inst.w16_cngOn = CNG_INTERNAL_ON;
+ }
+
+ }
+
+ /* Update packet size from previously decoded packet */
+ if (dspInfo.frameLen > 0)
+ {
+ inst->PacketBuffer_inst.packSizeSamples = dspInfo.frameLen;
+ }
+
+ /* Look for late packet (unless codec has changed) */
+ if (inst->new_codec != 1)
+ {
+ if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec))
+ {
+ WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
+ inst->timeStamp, &uw32_availableTS, &i_bufferpos, 1, &payloadType);
+ if ((inst->new_codec != 1) && (inst->timeStamp == uw32_availableTS)
+ && (inst->timeStamp < dspInfo.playedOutTS) && (i_bufferpos != -1)
+ && (WebRtcNetEQ_DbGetPayload(&(inst->codec_DB_inst),
+ (enum WebRtcNetEQDecoder) inst->current_Codec) == payloadType))
+ {
+ int waitingTime;
+ temp_pkt.payload = blockPtr + 1;
+ i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt,
+ i_bufferpos, &waitingTime);
+ if (i_res < 0)
+ { /* error returned */
+ return i_res;
+ }
+ WebRtcNetEQ_StoreWaitingTime(inst, waitingTime);
+ *blockPtr = temp_pkt.payloadLen;
+ /* set the flag if this is a redundant payload */
+ if (temp_pkt.rcuPlCntr > 0)
+ {
+ *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG);
+ }
+ blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1;
+
+ /*
+ * Close the data with a zero size block, in case we will not write any
+ * more data.
+ */
+ *blockPtr = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff)
+ | DSP_CODEC_ADD_LATE_PKT;
+ latePacketExist = 1;
+ }
+ }
+ }
+
+ i_res = WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
+ dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0),
+ &payloadType);
+ if (i_res < 0)
+ { /* error returned */
+ return i_res;
+ }
+
+ if (inst->BufferStat_inst.w16_cngOn == CNG_RFC3389_ON)
+ {
+ /*
+ * Because of timestamp peculiarities, we have to "manually" disallow using a CNG
+ * packet with the same timestamp as the one that was last played. This can happen
+ * when using redundancy and will cause the timing to shift.
+ */
+ while (i_bufferpos != -1 && WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst,
+ payloadType) && dspInfo.playedOutTS >= uw32_availableTS)
+ {
+
+ /* Don't use this packet, discard it */
+ inst->PacketBuffer_inst.payloadType[i_bufferpos] = -1;
+ inst->PacketBuffer_inst.payloadLengthBytes[i_bufferpos] = 0;
+ inst->PacketBuffer_inst.numPacketsInBuffer--;
+
+ /* Check buffer again */
+ WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
+ dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0),
+ &payloadType);
+ }
+ }
+
+ /* Check packet buffer */
+ w32_bufsize = WebRtcNetEQ_PacketBufferGetSize(&inst->PacketBuffer_inst);
+
+ if (dspInfo.lastMode == MODE_SUCCESS_ACCELERATE || dspInfo.lastMode
+ == MODE_LOWEN_ACCELERATE || dspInfo.lastMode == MODE_SUCCESS_PREEMPTIVE
+ || dspInfo.lastMode == MODE_LOWEN_PREEMPTIVE)
+ {
+ /* Subtract (dspInfo.samplesLeft + inst->timestampsPerCall) from sampleMemory */
+ inst->BufferStat_inst.Automode_inst.sampleMemory -= dspInfo.samplesLeft
+ + inst->timestampsPerCall;
+ }
+
+ /* calculate total current buffer size (in ms*8), including sync buffer */
+ w32_bufsize = WebRtcSpl_DivW32W16((w32_bufsize + dspInfo.samplesLeft), fs_mult);
+
+#ifdef NETEQ_ATEVENT_DECODE
+ /* DTMF data will affect the decision */
+ if (WebRtcNetEQ_DtmfDecode(&inst->DTMF_inst, blockPtr + 1, blockPtr + 2,
+ dspInfo.playedOutTS + inst->BufferStat_inst.uw32_CNGplayedTS) > 0)
+ {
+ playDtmf = 1;
+
+ /* Flag DTMF payload */
+ inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] | DSP_DTMF_PAYLOAD;
+
+ /* Block Length in bytes */
+ blockPtr[0] = 4;
+ /* Advance to next payload position */
+ blockPtr += 3;
+ }
+#endif
+
+ /* Update statistics and make decision */
+ uw16_instr = WebRtcNetEQ_BufstatsDecision(&inst->BufferStat_inst,
+ inst->PacketBuffer_inst.packSizeSamples, w32_bufsize, dspInfo.playedOutTS,
+ uw32_availableTS, i_bufferpos == -1,
+ WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType), dspInfo.lastMode,
+ inst->NetEqPlayoutMode, inst->timestampsPerCall, inst->NoOfExpandCalls, fs_mult,
+ lastModeBGNonly, playDtmf);
+
+ /* Check if time to reset loss counter */
+ if (inst->lastReportTS > WEBRTC_SPL_UMUL(inst->fs, MAX_LOSS_REPORT_PERIOD))
+ {
+ /* reset loss counter */
+ WebRtcNetEQ_ResetMcuInCallStats(inst);
+ }
+
+ /* Check sync buffer size */
+ if ((dspInfo.samplesLeft >= inst->timestampsPerCall) && (uw16_instr
+ != BUFSTATS_DO_ACCELERATE) && (uw16_instr != BUFSTATS_DO_MERGE) && (uw16_instr
+ != BUFSTATS_DO_PREEMPTIVE_EXPAND))
+ {
+ *blockPtr = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_NORMAL;
+ return 0;
+ }
+
+ if (uw16_instr == BUFSTATS_DO_EXPAND)
+ {
+ inst->NoOfExpandCalls++;
+ }
+ else
+ {
+ /* reset counter */
+ inst->NoOfExpandCalls = 0;
+ }
+
+ /* New codec or big change in packet number? */
+ if (((inst->new_codec) || (uw16_instr == BUFSTAT_REINIT)) && (uw16_instr
+ != BUFSTATS_DO_EXPAND))
+ {
+ CodecFuncInst_t cinst;
+
+ /* Clear other instructions */
+ blockPtr = &inst->pw16_writeAddress[3];
+ /* Clear instruction word */
+ inst->pw16_writeAddress[0] = 0;
+
+ inst->timeStamp = uw32_availableTS;
+ dspInfo.playedOutTS = uw32_availableTS;
+ if (inst->current_Codec != -1)
+ {
+ i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst,
+ (enum WebRtcNetEQDecoder) inst->current_Codec, &cinst);
+ if (i_res < 0)
+ { /* error returned */
+ return i_res;
+ }
+ }
+ else
+ {
+ /* The main codec has not been initialized yet (first packets are DTMF or CNG). */
+ if (WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType))
+ {
+ /* The currently extracted packet is CNG; get CNG fs */
+ WebRtc_UWord16 tempFs;
+
+ tempFs = WebRtcNetEQ_DbGetSampleRate(&inst->codec_DB_inst, payloadType);
+ if (tempFs > 0)
+ {
+ inst->fs = tempFs;
+ }
+ }
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) &cinst, 0,
+ sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
+ cinst.codec_fs = inst->fs;
+ }
+ cinst.timeStamp = inst->timeStamp;
+ blockLen = (sizeof(CodecFuncInst_t)) >> (sizeof(WebRtc_Word16) - 1); /* in Word16 */
+ *blockPtr = blockLen * 2;
+ blockPtr++;
+ WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst,sizeof(CodecFuncInst_t));
+ blockPtr += blockLen;
+ inst->new_codec = 0;
+
+ /* Reinitialize the MCU fs */
+ i_res = WebRtcNetEQ_McuSetFs(inst, cinst.codec_fs);
+ if (i_res < 0)
+ { /* error returned */
+ return i_res;
+ }
+
+ /* Set the packet size by guessing */
+ inst->PacketBuffer_inst.packSizeSamples = inst->timestampsPerCall * 3;
+
+ WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
+ inst->PacketBuffer_inst.maxInsertPositions);
+
+#ifdef NETEQ_CNG_CODEC
+ /* Also insert CNG state as this might be needed by DSP */
+ i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst, kDecoderCNG, &cinst);
+ if ((i_res < 0) && (i_res != CODEC_DB_NOT_EXIST1))
+ {
+ /* other error returned */
+ /* (CODEC_DB_NOT_EXIST1 simply indicates that CNG is not used */
+ return i_res;
+ }
+ else
+ {
+ /* CNG exists */
+ blockLen = (sizeof(cinst.codec_state)) >> (sizeof(WebRtc_Word16) - 1);
+ *blockPtr = blockLen * 2;
+ blockPtr++;
+ WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst.codec_state,sizeof(cinst.codec_state));
+ blockPtr += blockLen;
+ }
+#endif
+
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff)
+ | DSP_CODEC_NEW_CODEC;
+
+ if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET)
+ {
+ /*
+ * Change decision to CNG packet, since we do have a CNG packet, but it was
+ * considered too early to use. Now, use it anyway.
+ */
+ uw16_instr = BUFSTATS_DO_RFC3389CNG_PACKET;
+ }
+ else if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET)
+ {
+ uw16_instr = BUFSTATS_DO_NORMAL;
+ }
+
+ /* reset loss counter */
+ WebRtcNetEQ_ResetMcuInCallStats(inst);
+ }
+
+ /* Should we just reset the decoder? */
+ if (uw16_instr == BUFSTAT_REINIT_DECODER)
+ {
+ /* Change decision to normal and flag decoder reset */
+ uw16_instr = BUFSTATS_DO_NORMAL;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff) | DSP_CODEC_RESET;
+ }
+
+ /* Expand requires no new packet */
+ if (uw16_instr == BUFSTATS_DO_EXPAND)
+ {
+
+ inst->timeStamp = dspInfo.playedOutTS;
+
+ /* Have we got one descriptor left? */
+ if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec)
+ && (dspInfo.MD || latePacketExist))
+ {
+
+ if (dspInfo.lastMode != MODE_ONE_DESCRIPTOR)
+ {
+ /* this is the first "consecutive" one-descriptor decoding; reset counter */
+ inst->one_desc = 0;
+ }
+ if (inst->one_desc < MAX_ONE_DESC)
+ {
+ /* use that one descriptor */
+ inst->one_desc++; /* increase counter */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_NORMAL_ONE_DESC;
+
+ /* decrease counter since we did no Expand */
+ inst->NoOfExpandCalls = WEBRTC_SPL_MAX(inst->NoOfExpandCalls - 1, 0);
+ return 0;
+ }
+ else
+ {
+ /* too many consecutive one-descriptor decodings; do expand instead */
+ inst->one_desc = 0; /* reset counter */
+ }
+
+ }
+
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_EXPAND;
+ return 0;
+ }
+
+ /* Merge is not needed if we still have a descriptor */
+ if ((uw16_instr == BUFSTATS_DO_MERGE) && (dspInfo.MD != 0))
+ {
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_NORMAL_ONE_DESC;
+ *blockPtr = 0;
+ return 0;
+ }
+
+ /* Do CNG without trying to extract any packets from buffer */
+ if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET)
+ {
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_RFC3389CNG;
+ *blockPtr = 0;
+ return 0;
+ }
+
+ /* Do built-in CNG without extracting any new packets from buffer */
+ if (uw16_instr == BUFSTATS_DO_INTERNAL_CNG_NOPACKET)
+ {
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_CODEC_INTERNAL_CNG;
+ *blockPtr = 0;
+ return 0;
+ }
+
+ /* Do DTMF without extracting any new packets from buffer */
+ if (uw16_instr == BUFSTATS_DO_DTMF_ONLY)
+ {
+ WebRtc_UWord32 timeStampJump = 0;
+
+ /* Update timestamp */
+ if ((inst->BufferStat_inst.uw32_CNGplayedTS > 0) && (dspInfo.lastMode != MODE_DTMF))
+ {
+ /* Jump in timestamps if needed */
+ timeStampJump = inst->BufferStat_inst.uw32_CNGplayedTS;
+ inst->pw16_writeAddress[1] = (WebRtc_UWord16) (timeStampJump >> 16);
+ inst->pw16_writeAddress[2] = (WebRtc_UWord16) (timeStampJump & 0xFFFF);
+ }
+
+ inst->timeStamp = dspInfo.playedOutTS + timeStampJump;
+
+ inst->BufferStat_inst.uw32_CNGplayedTS = 0;
+ inst->NoOfExpandCalls = 0;
+
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DTMF_GENERATE;
+ *blockPtr = 0;
+ return 0;
+ }
+
+ if (uw16_instr == BUFSTATS_DO_ACCELERATE)
+ {
+ /* In order to do a Accelerate we need at least 30 ms of data */
+ if (dspInfo.samplesLeft >= (3 * 80 * fs_mult))
+ {
+ /* Already have enough data, so we do not need to extract any more */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_ACCELERATE;
+ *blockPtr = 0;
+ inst->BufferStat_inst.Automode_inst.sampleMemory
+ = (WebRtc_Word32) dspInfo.samplesLeft;
+ inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
+ return 0;
+ }
+ else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult))
+ && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
+ {
+ /* Avoid decoding more data as it might overflow playout buffer */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_NORMAL;
+ *blockPtr = 0;
+ return 0;
+ }
+ else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult))
+ && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
+ {
+ /* For >= 30ms allow Accelerate with a decoding to avoid overflow in playout buffer */
+ wantedNoOfTimeStamps = inst->timestampsPerCall;
+ }
+ else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult))
+ {
+ /* We need to decode another 10 ms in order to do an Accelerate */
+ wantedNoOfTimeStamps = inst->timestampsPerCall;
+ }
+ else
+ {
+ /*
+ * Build up decoded data by decoding at least 20 ms of data.
+ * Do not perform Accelerate yet, but wait until we only need to do one decoding.
+ */
+ wantedNoOfTimeStamps = 2 * inst->timestampsPerCall;
+ uw16_instr = BUFSTATS_DO_NORMAL;
+ }
+ }
+ else if (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND)
+ {
+ /* In order to do a Preemptive Expand we need at least 30 ms of data */
+ if (dspInfo.samplesLeft >= (3 * 80 * fs_mult))
+ {
+ /* Already have enough data, so we do not need to extract any more */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_PREEMPTIVE_EXPAND;
+ *blockPtr = 0;
+ inst->BufferStat_inst.Automode_inst.sampleMemory
+ = (WebRtc_Word32) dspInfo.samplesLeft;
+ inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
+ return 0;
+ }
+ else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult))
+ && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
+ {
+ /*
+ * Avoid decoding more data as it might overflow playout buffer;
+ * still try Preemptive Expand though.
+ */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_PREEMPTIVE_EXPAND;
+ *blockPtr = 0;
+ inst->BufferStat_inst.Automode_inst.sampleMemory
+ = (WebRtc_Word32) dspInfo.samplesLeft;
+ inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
+ return 0;
+ }
+ else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult))
+ && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
+ {
+ /*
+ * For >= 30ms allow Preemptive Expand with a decoding to avoid overflow in
+ * playout buffer
+ */
+ wantedNoOfTimeStamps = inst->timestampsPerCall;
+ }
+ else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult))
+ {
+ /* We need to decode another 10 ms in order to do an Preemptive Expand */
+ wantedNoOfTimeStamps = inst->timestampsPerCall;
+ }
+ else
+ {
+ /*
+ * Build up decoded data by decoding at least 20 ms of data,
+ * Still try to perform Preemptive Expand.
+ */
+ wantedNoOfTimeStamps = 2 * inst->timestampsPerCall;
+ }
+ }
+ else
+ {
+ wantedNoOfTimeStamps = inst->timestampsPerCall;
+ }
+
+ /* Otherwise get data from buffer, try to get at least 10ms */
+ totalTS = 0;
+ oldTS = uw32_availableTS;
+ if ((i_bufferpos > -1) && (uw16_instr != BUFSTATS_DO_ALTERNATIVE_PLC) && (uw16_instr
+ != BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS) && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION)
+ && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION_INC_TS))
+ {
+ uw32_tmp = (uw32_availableTS - dspInfo.playedOutTS);
+ inst->pw16_writeAddress[1] = (WebRtc_UWord16) (uw32_tmp >> 16);
+ inst->pw16_writeAddress[2] = (WebRtc_UWord16) (uw32_tmp & 0xFFFF);
+ if (inst->BufferStat_inst.w16_cngOn == CNG_OFF)
+ {
+ /*
+ * Adjustment of TS only corresponds to an actual packet loss
+ * if comfort noise is not played. If comfort noise was just played,
+ * this adjustment of TS is only done to get back in sync with the
+ * stream TS; no loss to report.
+ */
+ inst->lostTS += uw32_tmp;
+ }
+
+ if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET)
+ {
+ /* We are about to decode and use a non-CNG packet => CNG period is ended */
+ inst->BufferStat_inst.w16_cngOn = CNG_OFF;
+ }
+
+ /*
+ * Reset CNG timestamp as a new packet will be delivered.
+ * (Also if CNG packet, since playedOutTS is updated.)
+ */
+ inst->BufferStat_inst.uw32_CNGplayedTS = 0;
+
+ prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos];
+ prevTS = inst->PacketBuffer_inst.timeStamp[i_bufferpos];
+ oldPT = inst->PacketBuffer_inst.payloadType[i_bufferpos];
+
+ /* clear flag bits */
+ inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] & 0xFF3F;
+ do
+ {
+ int waitingTime;
+ inst->timeStamp = uw32_availableTS;
+ /* Write directly to shared memory */
+ temp_pkt.payload = blockPtr + 1;
+ i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt,
+ i_bufferpos, &waitingTime);
+
+ if (i_res < 0)
+ {
+ /* error returned */
+ return i_res;
+ }
+ WebRtcNetEQ_StoreWaitingTime(inst, waitingTime);
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE;
+ fwrite(&temp_var,sizeof(int),1,delay_fid2);
+ fwrite(&temp_pkt.timeStamp,sizeof(WebRtc_UWord32),1,delay_fid2);
+ fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
+#endif
+
+ *blockPtr = temp_pkt.payloadLen;
+ /* set the flag if this is a redundant payload */
+ if (temp_pkt.rcuPlCntr > 0)
+ {
+ *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG);
+ }
+ blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1;
+
+ if (i_bufferpos > -1)
+ {
+ /*
+ * Store number of TS extracted (last extracted is assumed to be of
+ * packSizeSamples).
+ */
+ totalTS = uw32_availableTS - oldTS + inst->PacketBuffer_inst.packSizeSamples;
+ }
+ /* Check what next packet is available */
+ WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
+ inst->timeStamp, &uw32_availableTS, &i_bufferpos, 0, &payloadType);
+
+ nextSeqNoAvail = 0;
+ if ((i_bufferpos > -1) && (oldPT
+ == inst->PacketBuffer_inst.payloadType[i_bufferpos]))
+ {
+ w16_tmp = inst->PacketBuffer_inst.seqNumber[i_bufferpos] - prevSeqNo;
+ w32_tmp = inst->PacketBuffer_inst.timeStamp[i_bufferpos] - prevTS;
+ if ((w16_tmp == 1) || /* Next packet */
+ ((w16_tmp == 0) && (w32_tmp == inst->PacketBuffer_inst.packSizeSamples)))
+ { /* or packet split into frames */
+ nextSeqNoAvail = 1;
+ }
+ prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos];
+ }
+
+ }
+ while ((totalTS < wantedNoOfTimeStamps) && (nextSeqNoAvail == 1));
+ }
+
+ if ((uw16_instr == BUFSTATS_DO_ACCELERATE)
+ || (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND))
+ {
+ /* Check that we have enough data (30ms) to do the Accelearate */
+ if ((totalTS + dspInfo.samplesLeft) < WEBRTC_SPL_MUL(3,inst->timestampsPerCall)
+ && (uw16_instr == BUFSTATS_DO_ACCELERATE))
+ {
+ /* Not enough, do normal operation instead */
+ uw16_instr = BUFSTATS_DO_NORMAL;
+ }
+ else
+ {
+ inst->BufferStat_inst.Automode_inst.sampleMemory
+ = (WebRtc_Word32) dspInfo.samplesLeft + totalTS;
+ inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
+ }
+ }
+
+ /* Close the data with a zero size block */
+ *blockPtr = 0;
+
+ /* Write data to DSP */
+ switch (uw16_instr)
+ {
+ case BUFSTATS_DO_NORMAL:
+ /* Normal with decoding included */
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_NORMAL;
+ break;
+ case BUFSTATS_DO_ACCELERATE:
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_ACCELERATE;
+ break;
+ case BUFSTATS_DO_MERGE:
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_MERGE;
+ break;
+ case BUFSTATS_DO_RFC3389CNG_PACKET:
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_RFC3389CNG;
+ break;
+ case BUFSTATS_DO_ALTERNATIVE_PLC:
+ inst->pw16_writeAddress[1] = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_ALTERNATIVE_PLC;
+ break;
+ case BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS:
+ inst->pw16_writeAddress[1] = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS;
+ break;
+ case BUFSTATS_DO_AUDIO_REPETITION:
+ inst->pw16_writeAddress[1] = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_AUDIO_REPETITION;
+ break;
+ case BUFSTATS_DO_AUDIO_REPETITION_INC_TS:
+ inst->pw16_writeAddress[1] = 0;
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_DO_AUDIO_REPETITION_INC_TS;
+ break;
+ case BUFSTATS_DO_PREEMPTIVE_EXPAND:
+ inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
+ | DSP_INSTR_PREEMPTIVE_EXPAND;
+ break;
+ default:
+ return UNKNOWN_BUFSTAT_DECISION;
+ }
+
+ inst->timeStamp = dspInfo.playedOutTS;
+ return 0;
+
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/split_and_insert.c b/trunk/src/modules/audio_coding/neteq/split_and_insert.c
new file mode 100644
index 0000000..03c1569
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/split_and_insert.c
@@ -0,0 +1,141 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Split an RTP payload (if possible and suitable) and insert into packet buffer.
+ */
+
+#include "mcu.h"
+
+#include <string.h>
+
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+
+int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
+ SplitInfo_t *split_inst, WebRtc_Word16 *flushed)
+{
+
+ int i_ok;
+ int len;
+ int i;
+ RTPPacket_t temp_packet;
+ WebRtc_Word16 localFlushed = 0;
+ const WebRtc_Word16 *pw16_startPayload;
+ *flushed = 0;
+
+ len = packet->payloadLen;
+
+ /* Copy to temp packet that can be modified. */
+
+ WEBRTC_SPL_MEMCPY_W8(&temp_packet,packet,sizeof(RTPPacket_t));
+
+ if (split_inst->deltaBytes == NO_SPLIT)
+ {
+ /* Not splittable codec */
+ i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, packet, &localFlushed);
+ *flushed |= localFlushed;
+ if (i_ok < 0)
+ {
+ return PBUFFER_INSERT_ERROR5;
+ }
+ }
+ else if (split_inst->deltaBytes < -10)
+ {
+ /* G711, PCM16B or G722, use "soft splitting" */
+ int split_size = packet->payloadLen;
+ int mult = WEBRTC_SPL_ABS_W32(split_inst->deltaBytes) - 10;
+
+ /* Find "chunk size" >= 20 ms and < 40 ms
+ * split_inst->deltaTime in this case contains the number of bytes per
+ * timestamp unit times 2
+ */
+ while (split_size >= ((80 << split_inst->deltaTime) * mult))
+ {
+ split_size >>= 1;
+ }
+
+ /* Make the size an even value. */
+ if (split_size > 1)
+ {
+ split_size >>= 1;
+ split_size *= 2;
+ }
+
+ temp_packet.payloadLen = split_size;
+ pw16_startPayload = temp_packet.payload;
+ i = 0;
+ while (len >= (2 * split_size))
+ {
+ /* insert every chunk */
+ i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
+ *flushed |= localFlushed;
+ temp_packet.timeStamp += ((2 * split_size) >> split_inst->deltaTime);
+ i++;
+ temp_packet.payload = &(pw16_startPayload[(i * split_size) >> 1]);
+ temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_size & 0x1);
+
+ len -= split_size;
+ if (i_ok < 0)
+ {
+ return PBUFFER_INSERT_ERROR1;
+ }
+ }
+
+ /* Insert the rest */
+ temp_packet.payloadLen = len;
+ i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
+ *flushed |= localFlushed;
+ if (i_ok < 0)
+ {
+ return PBUFFER_INSERT_ERROR2;
+ }
+ }
+ else
+ {
+ /* Frame based codec, use hard splitting. */
+ i = 0;
+ pw16_startPayload = temp_packet.payload;
+ while (len >= split_inst->deltaBytes)
+ {
+
+ temp_packet.payloadLen = split_inst->deltaBytes;
+ i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
+ *flushed |= localFlushed;
+ i++;
+ temp_packet.payload = &(pw16_startPayload[(i * split_inst->deltaBytes) >> 1]);
+ temp_packet.timeStamp += split_inst->deltaTime;
+ temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_inst->deltaBytes
+ & 0x1);
+
+ if (i_ok < 0)
+ {
+ return PBUFFER_INSERT_ERROR3;
+ }
+ len -= split_inst->deltaBytes;
+
+ }
+ if (len > 0)
+ {
+ /* Must be a either an error or a SID frame at the end of the packet. */
+ temp_packet.payloadLen = len;
+ i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
+ *flushed |= localFlushed;
+ if (i_ok < 0)
+ {
+ return PBUFFER_INSERT_ERROR4;
+ }
+ }
+ }
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
new file mode 100644
index 0000000..64bf508
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
@@ -0,0 +1,657 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "NETEQTEST_CodecClass.h"
+
+#include <stdlib.h> // exit
+
+#include "webrtc_neteq_help_macros.h"
+
+NETEQTEST_Decoder::NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt)
+:
+_decoder(NULL),
+_decoderType(type),
+_pt(pt),
+_fs(fs),
+_name(name)
+{
+}
+
+int NETEQTEST_Decoder::loadToNetEQ(NETEQTEST_NetEQClass & neteq, WebRtcNetEQ_CodecDef & codecInst)
+{
+ SET_CODEC_PAR(codecInst, _decoderType, _pt, _decoder, _fs);
+ int err = neteq.loadCodec(codecInst);
+
+ if (err)
+ {
+ printf("Error loading codec %s into NetEQ database\n", _name.c_str());
+ }
+
+ return(err);
+}
+
+
+// iSAC
+#ifdef CODEC_ISAC
+#include "isac.h"
+
+decoder_iSAC::decoder_iSAC(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderISAC, 16000, "iSAC", pt)
+{
+ WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
+ WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacWideband);
+}
+
+
+decoder_iSAC::~decoder_iSAC()
+{
+ if (_decoder)
+ {
+ WebRtcIsac_Free((ISACStruct *) _decoder);
+ _decoder = NULL;
+ }
+}
+
+
+int decoder_iSAC::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_ISAC_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+
+}
+#endif
+
+#ifdef CODEC_ISAC_SWB
+decoder_iSACSWB::decoder_iSACSWB(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderISACswb, 32000, "iSAC swb", pt)
+{
+ WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
+ WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacSuperWideband);
+}
+
+decoder_iSACSWB::~decoder_iSACSWB()
+{
+ if (_decoder)
+ {
+ WebRtcIsac_Free((ISACStruct *) _decoder);
+ _decoder = NULL;
+ }
+}
+
+int decoder_iSACSWB::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_ISACSWB_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+
+}
+#endif
+
+// PCM u/A
+#ifdef CODEC_G711
+#include "g711_interface.h"
+
+decoder_PCMU::decoder_PCMU(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderPCMu, 8000, "G.711-u", pt)
+{
+ // no state to crate or init
+}
+
+int decoder_PCMU::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCMU_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+
+}
+
+decoder_PCMA::decoder_PCMA(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderPCMa, 8000, "G.711-A", pt)
+{
+ // no state to crate or init
+}
+
+int decoder_PCMA::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCMA_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+// Linear PCM16b
+#if (defined(CODEC_PCM16B) || defined(CODEC_PCM16B_WB) || \
+ defined(CODEC_PCM16B_32KHZ) || defined(CODEC_PCM16B_48KHZ))
+#include "pcm16b.h"
+#endif
+
+#ifdef CODEC_PCM16B
+int decoder_PCM16B_NB::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCM16B_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_PCM16B_WB
+int decoder_PCM16B_WB::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCM16B_WB_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_PCM16B_32KHZ
+int decoder_PCM16B_SWB32::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCM16B_SWB32_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_PCM16B_48KHZ
+int decoder_PCM16B_SWB48::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_PCM16B_SWB48_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_ILBC
+#include "ilbc.h"
+decoder_ILBC::decoder_ILBC(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderILBC, 8000, "iLBC", pt)
+{
+ WebRtc_Word16 err = WebRtcIlbcfix_DecoderCreate((iLBC_decinst_t **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_ILBC::~decoder_ILBC()
+{
+ WebRtcIlbcfix_DecoderFree((iLBC_decinst_t *) _decoder);
+}
+
+int decoder_ILBC::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_ILBC_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G729
+#include "G729Interface.h"
+decoder_G729::decoder_G729(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG729, 8000, "G.729", pt)
+{
+ WebRtc_Word16 err = WebRtcG729_CreateDec((G729_decinst_t **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G729::~decoder_G729()
+{
+ WebRtcG729_FreeDec((G729_decinst_t *) _decoder);
+}
+
+int decoder_G729::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G729_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G729_1
+#include "G729_1Interface.h"
+decoder_G729_1::decoder_G729_1(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG729_1, 16000, "G.729.1", pt)
+{
+ WebRtc_Word16 err = WebRtcG7291_Create((G729_1_inst_t **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G729_1::~decoder_G729_1()
+{
+ WebRtcG7291_Free((G729_1_inst_t *) _decoder);
+}
+
+int decoder_G729_1::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G729_1_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722
+#include "g722_interface.h"
+decoder_G722::decoder_G722(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722, 16000, "G.722", pt)
+{
+ WebRtc_Word16 err = WebRtcG722_CreateDecoder((G722DecInst **) &_decoder);
+ if (err)
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G722::~decoder_G722()
+{
+ WebRtcG722_FreeDecoder((G722DecInst *) _decoder);
+}
+
+int decoder_G722::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#if (defined(CODEC_G722_1_16) || defined(CODEC_G722_1_24) || \
+ defined(CODEC_G722_1_32) || defined(CODEC_G722_1C_24) || \
+ defined(CODEC_G722_1C_32) || defined(CODEC_G722_1C_48))
+#include "G722_1Interface.h"
+#endif
+
+#ifdef CODEC_G722_1_16
+decoder_G722_1_16::decoder_G722_1_16(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1_16, 16000, "G.722.1 (16 kbps)", pt)
+{
+ if (WebRtcG7221_CreateDec16((G722_1_16_decinst_t **) &_decoder))
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G722_1_16::~decoder_G722_1_16()
+{
+ WebRtcG7221_FreeDec16((G722_1_16_decinst_t *) _decoder);
+}
+
+int decoder_G722_1_16::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1_16_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722_1_24
+decoder_G722_1_24::decoder_G722_1_24(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1_24, 16000, "G.722.1 (24 kbps)", pt)
+{
+ if (WebRtcG7221_CreateDec24((G722_1_24_decinst_t **) &_decoder))
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G722_1_24::~decoder_G722_1_24()
+{
+ WebRtcG7221_FreeDec24((G722_1_24_decinst_t *) _decoder);
+}
+
+int decoder_G722_1_24::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1_24_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722_1_32
+decoder_G722_1_32::decoder_G722_1_32(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1_32, 16000, "G.722.1 (32 kbps)", pt)
+{
+ if (WebRtcG7221_CreateDec32((G722_1_32_decinst_t **) &_decoder))
+ {
+ exit(EXIT_FAILURE);
+ }
+}
+
+decoder_G722_1_32::~decoder_G722_1_32()
+{
+ WebRtcG7221_FreeDec32((G722_1_32_decinst_t *) _decoder);
+}
+
+int decoder_G722_1_32::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1_32_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722_1C_24
+decoder_G722_1C_24::decoder_G722_1C_24(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1C_24, 32000, "G.722.1C (24 kbps)", pt)
+{
+ if (WebRtcG7221C_CreateDec24((G722_1C_24_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+}
+
+decoder_G722_1C_24::~decoder_G722_1C_24()
+{
+ WebRtcG7221C_FreeDec24((G722_1C_24_decinst_t *) _decoder);
+}
+
+int decoder_G722_1C_24::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1C_24_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722_1C_32
+decoder_G722_1C_32::decoder_G722_1C_32(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1C_32, 32000, "G.722.1C (32 kbps)", pt)
+{
+ if (WebRtcG7221C_CreateDec32((G722_1C_32_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+}
+
+decoder_G722_1C_32::~decoder_G722_1C_32()
+{
+ WebRtcG7221C_FreeDec32((G722_1C_32_decinst_t *) _decoder);
+}
+
+int decoder_G722_1C_32::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1C_32_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_G722_1C_48
+decoder_G722_1C_48::decoder_G722_1C_48(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderG722_1C_48, 32000, "G.722.1C (48 kbps)", pt)
+{
+ if (WebRtcG7221C_CreateDec48((G722_1C_48_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+}
+
+decoder_G722_1C_48::~decoder_G722_1C_48()
+{
+ WebRtcG7221C_FreeDec48((G722_1C_48_decinst_t *) _decoder);
+}
+
+int decoder_G722_1C_48::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_G722_1C_48_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_AMR
+#include "AMRInterface.h"
+#include "AMRCreation.h"
+decoder_AMR::decoder_AMR(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderAMR, 8000, "AMR", pt)
+{
+ if (WebRtcAmr_CreateDec((AMR_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+
+ WebRtcAmr_DecodeBitmode((AMR_decinst_t *) _decoder, AMRBandwidthEfficient);
+}
+
+decoder_AMR::~decoder_AMR()
+{
+ WebRtcAmr_FreeDec((AMR_decinst_t *) _decoder);
+}
+
+int decoder_AMR::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_AMR_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_AMRWB
+#include "AMRWBInterface.h"
+#include "AMRWBCreation.h"
+decoder_AMRWB::decoder_AMRWB(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderAMRWB, 16000, "AMR wb", pt)
+{
+ if (WebRtcAmrWb_CreateDec((AMRWB_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+
+ WebRtcAmrWb_DecodeBitmode((AMRWB_decinst_t *) _decoder, AMRBandwidthEfficient);
+}
+
+decoder_AMRWB::~decoder_AMRWB()
+{
+ WebRtcAmrWb_FreeDec((AMRWB_decinst_t *) _decoder);
+}
+
+int decoder_AMRWB::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_AMRWB_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_GSMFR
+#include "GSMFRInterface.h"
+#include "GSMFRCreation.h"
+decoder_GSMFR::decoder_GSMFR(WebRtc_UWord8 pt)
+:
+NETEQTEST_Decoder(kDecoderGSMFR, 8000, "GSM-FR", pt)
+{
+ if (WebRtcGSMFR_CreateDec((GSMFR_decinst_t **) &_decoder))
+ exit(EXIT_FAILURE);
+}
+
+decoder_GSMFR::~decoder_GSMFR()
+{
+ WebRtcGSMFR_FreeDec((GSMFR_decinst_t *) _decoder);
+}
+
+int decoder_GSMFR::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_GSMFR_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#if (defined(CODEC_SPEEX_8) || defined (CODEC_SPEEX_16))
+#include "SpeexInterface.h"
+decoder_SPEEX::decoder_SPEEX(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+:
+NETEQTEST_Decoder(fs == 8000 ? kDecoderSPEEX_8 : kDecoderSPEEX_16,
+ fs, "SPEEX " + fs/1000, pt)
+{
+ if (fs != 8000 && fs != 16000)
+ throw std::exception("Wrong sample rate for SPEEX");
+
+ if (WebRtcSpeex_CreateDec((SPEEX_decinst_t **) &_decoder, fs, 1))
+ exit(EXIT_FAILURE);
+}
+
+decoder_SPEEX::~decoder_SPEEX()
+{
+ WebRtcSpeex_FreeDec((SPEEX_decinst_t *) _decoder);
+}
+
+int decoder_SPEEX::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_SPEEX_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_CELT_32
+#include "celt_interface.h"
+decoder_CELT::decoder_CELT(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+:
+NETEQTEST_Decoder(kDecoderCELT_32, fs, "CELT", pt)
+{
+ if (WebRtcCelt_CreateDec((CELT_decinst_t **) &_decoder, 1))
+ exit(EXIT_FAILURE);
+}
+
+decoder_CELT::~decoder_CELT()
+{
+ WebRtcCelt_FreeDec((CELT_decinst_t *) _decoder);
+}
+
+int decoder_CELT::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_CELT_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_RED
+int decoder_RED::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_RED_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#ifdef CODEC_ATEVENT_DECODE
+int decoder_AVT::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_AVT_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+#include "webrtc_cng.h"
+decoder_CNG::decoder_CNG(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+:
+NETEQTEST_Decoder(kDecoderCNG, fs, "CNG " + fs/1000, pt)
+{
+ if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000)
+ exit(EXIT_FAILURE);
+
+ if (WebRtcCng_CreateDec((CNG_dec_inst **) &_decoder))
+ exit(EXIT_FAILURE);
+}
+
+decoder_CNG::~decoder_CNG()
+{
+ WebRtcCng_FreeDec((CNG_dec_inst *) _decoder);
+}
+
+int decoder_CNG::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+ WebRtcNetEQ_CodecDef codecInst;
+
+ SET_CNG_FUNCTIONS(codecInst);
+
+ return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
new file mode 100644
index 0000000..ff49049
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
@@ -0,0 +1,301 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef NETEQTEST_CODECCLASS_H
+#define NETEQTEST_CODECCLASS_H
+
+#include <string>
+#include <string.h>
+
+#include "typedefs.h"
+#include "webrtc_neteq.h"
+#include "NETEQTEST_NetEQClass.h"
+
+class NETEQTEST_Decoder
+{
+public:
+ NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt = 0);
+ virtual ~NETEQTEST_Decoder() {};
+
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
+
+ int getName(char * name, int maxLen) const { strncpy( name, _name.c_str(), maxLen ); return 0;};
+
+ void setPT(WebRtc_UWord8 pt) { _pt = pt; };
+ WebRtc_UWord16 getFs() const { return (_fs); };
+ enum WebRtcNetEQDecoder getType() const { return (_decoderType); };
+ WebRtc_UWord8 getPT() const { return (_pt); };
+
+protected:
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq, WebRtcNetEQ_CodecDef & codecInst);
+
+ void * _decoder;
+ enum WebRtcNetEQDecoder _decoderType;
+ WebRtc_UWord8 _pt;
+ WebRtc_UWord16 _fs;
+ std::string _name;
+
+private:
+};
+
+
+class decoder_iSAC : public NETEQTEST_Decoder
+{
+public:
+ decoder_iSAC(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_iSAC();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_iSACSWB : public NETEQTEST_Decoder
+{
+public:
+ decoder_iSACSWB(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_iSACSWB();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_PCMU : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCMU(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_PCMU() {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_PCMA : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCMA(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_PCMA() {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_PCM16B_NB : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCM16B_NB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16B, 8000, "PCM16 nb", pt) {};
+ virtual ~decoder_PCM16B_NB() {};
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+class decoder_PCM16B_WB : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCM16B_WB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bwb, 16000, "PCM16 wb", pt) {};
+ virtual ~decoder_PCM16B_WB() {};
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+class decoder_PCM16B_SWB32 : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCM16B_SWB32(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb32kHz, 32000, "PCM16 swb32", pt) {};
+ virtual ~decoder_PCM16B_SWB32() {};
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+class decoder_PCM16B_SWB48 : public NETEQTEST_Decoder
+{
+public:
+ decoder_PCM16B_SWB48(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb48kHz, 48000, "PCM16 swb48", pt) {};
+ virtual ~decoder_PCM16B_SWB48() {};
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_ILBC : public NETEQTEST_Decoder
+{
+public:
+ decoder_ILBC(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_ILBC();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_G729 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G729(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G729();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G729_1 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G729_1(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G729_1();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_G722 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_G722_1_16 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1_16(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1_16();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G722_1_24 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1_24(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1_24();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G722_1_32 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1_32(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1_32();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_G722_1C_24 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1C_24(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1C_24();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G722_1C_32 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1C_32(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1C_32();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G722_1C_48 : public NETEQTEST_Decoder
+{
+public:
+ decoder_G722_1C_48(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G722_1C_48();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_AMR : public NETEQTEST_Decoder
+{
+public:
+ decoder_AMR(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_AMR();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_AMRWB : public NETEQTEST_Decoder
+{
+public:
+ decoder_AMRWB(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_AMRWB();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_GSMFR : public NETEQTEST_Decoder
+{
+public:
+ decoder_GSMFR(WebRtc_UWord8 pt = 0);
+ virtual ~decoder_GSMFR();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G726 : public NETEQTEST_Decoder
+{
+public:
+ //virtual decoder_G726(WebRtc_UWord8 pt = 0) = 0;
+ decoder_G726(enum WebRtcNetEQDecoder type, const char * name, WebRtc_UWord8 pt = 0);
+ virtual ~decoder_G726();
+ virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
+};
+
+class decoder_G726_16 : public decoder_G726
+{
+public:
+ decoder_G726_16(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_16, "G.726 (16 kbps)", pt) {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G726_24 : public decoder_G726
+{
+public:
+ decoder_G726_24(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_24, "G.726 (24 kbps)", pt) {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G726_32 : public decoder_G726
+{
+public:
+ decoder_G726_32(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_32, "G.726 (32 kbps)", pt) {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_G726_40 : public decoder_G726
+{
+public:
+ decoder_G726_40(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_40, "G.726 (40 kbps)", pt) {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_SPEEX : public NETEQTEST_Decoder
+{
+public:
+ decoder_SPEEX(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
+ virtual ~decoder_SPEEX();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_CELT : public NETEQTEST_Decoder
+{
+public:
+ decoder_CELT(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 32000);
+ virtual ~decoder_CELT();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_RED : public NETEQTEST_Decoder
+{
+public:
+ decoder_RED(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderRED, 8000, "RED", pt) {};
+ virtual ~decoder_RED() {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+class decoder_AVT : public NETEQTEST_Decoder
+{
+public:
+ decoder_AVT(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderAVT, 8000, "AVT", pt) {};
+ virtual ~decoder_AVT() {};
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+
+class decoder_CNG : public NETEQTEST_Decoder
+{
+public:
+ decoder_CNG(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
+ virtual ~decoder_CNG();
+ int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
+#endif //NETEQTEST_CODECCLASS_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
new file mode 100644
index 0000000..2e60658
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
@@ -0,0 +1,384 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory.h>
+
+#include "NETEQTEST_NetEQClass.h"
+
+
+NETEQTEST_NetEQClass::NETEQTEST_NetEQClass()
+ :
+ _inst(NULL),
+ _instMem(NULL),
+ _bufferMem(NULL),
+ _preparseRTP(false),
+ _fsmult(1),
+ _isMaster(true)
+{
+#ifdef WINDOWS_TIMING
+ _totTimeRecIn.QuadPart = 0;
+ _totTimeRecOut.QuadPart = 0;
+#endif
+}
+
+NETEQTEST_NetEQClass::NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
+ WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
+ :
+ _inst(NULL),
+ _instMem(NULL),
+ _bufferMem(NULL),
+ _preparseRTP(false),
+ _fsmult(1),
+ _isMaster(true)
+{
+#ifdef WINDOWS_TIMING
+ _totTimeRecIn.QuadPart = 0;
+ _totTimeRecOut.QuadPart = 0;
+#endif
+
+ if (assign() == 0)
+ {
+ if (init(fs) == 0)
+ {
+ assignBuffer(usedCodec, noOfCodecs, nwType);
+ }
+ }
+}
+
+
+NETEQTEST_NetEQClass::~NETEQTEST_NetEQClass()
+{
+ if (_instMem)
+ {
+ delete [] _instMem;
+ _instMem = NULL;
+ }
+
+ if (_bufferMem)
+ {
+ delete [] _bufferMem;
+ _bufferMem = NULL;
+ }
+
+ _inst = NULL;
+}
+
+int NETEQTEST_NetEQClass::assign()
+{
+ int memSize;
+
+ WebRtcNetEQ_AssignSize(&memSize);
+
+ if (_instMem)
+ {
+ delete [] _instMem;
+ _instMem = NULL;
+ }
+
+ _instMem = new WebRtc_Word8[memSize];
+
+ int ret = WebRtcNetEQ_Assign(&_inst, _instMem);
+
+ if (ret)
+ {
+ printError();
+ }
+
+ return (ret);
+}
+
+
+int NETEQTEST_NetEQClass::init(WebRtc_UWord16 fs)
+{
+ int ret;
+
+ if (!_inst)
+ {
+ // not assigned
+ ret = assign();
+
+ if (ret != 0)
+ {
+ printError();
+ return (ret);
+ }
+ }
+
+ ret = WebRtcNetEQ_Init(_inst, fs);
+
+ if (ret != 0)
+ {
+ printError();
+ }
+
+ return (ret);
+
+}
+
+
+int NETEQTEST_NetEQClass::assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType)
+{
+ int numPackets, memSize, ret;
+
+ if (!_inst)
+ {
+ // not assigned
+ ret = assign();
+
+ if (ret != 0)
+ {
+ printError();
+ return (ret);
+ }
+
+ ret = init();
+
+ if (ret != 0)
+ {
+ printError();
+ return (ret);
+ }
+ }
+
+ ret = WebRtcNetEQ_GetRecommendedBufferSize(_inst, usedCodec, noOfCodecs, nwType, &numPackets, &memSize);
+
+ if (ret != 0)
+ {
+ printError();
+ return (ret);
+ }
+
+ if (_bufferMem)
+ {
+ delete [] _bufferMem;
+ _bufferMem = NULL;
+ }
+
+ _bufferMem = new WebRtc_Word8[memSize];
+
+ memset(_bufferMem, -1, memSize);
+
+ ret = WebRtcNetEQ_AssignBuffer(_inst, numPackets, _bufferMem, memSize);
+
+ if (ret != 0)
+ {
+ printError();
+ }
+
+ return (ret);
+}
+
+int NETEQTEST_NetEQClass::loadCodec(WebRtcNetEQ_CodecDef &codecInst)
+{
+ int err = WebRtcNetEQ_CodecDbAdd(_inst, &codecInst);
+
+ if (err)
+ {
+ printError();
+ }
+
+ return (err);
+}
+
+void NETEQTEST_NetEQClass::printError()
+{
+ if (_inst)
+ {
+ int errorCode = WebRtcNetEQ_GetErrorCode(_inst);
+
+ if (errorCode)
+ {
+ char errorName[WEBRTC_NETEQ_MAX_ERROR_NAME];
+
+ WebRtcNetEQ_GetErrorName(errorCode, errorName, WEBRTC_NETEQ_MAX_ERROR_NAME);
+
+ printf("Error %i: %s\n", errorCode, errorName);
+ }
+ }
+}
+
+void NETEQTEST_NetEQClass::printError(NETEQTEST_RTPpacket &rtp)
+{
+ // print regular error info
+ printError();
+
+ // print extra info from packet
+ printf("\tRTP: TS=%u, SN=%u, PT=%u, M=%i, len=%i\n",
+ rtp.timeStamp(), rtp.sequenceNumber(), rtp.payloadType(),
+ rtp.markerBit(), rtp.payloadLen());
+
+}
+
+int NETEQTEST_NetEQClass::recIn(NETEQTEST_RTPpacket &rtp)
+{
+
+ int err;
+#ifdef WINDOWS_TIMING
+ LARGE_INTEGER countA, countB;
+#endif
+
+ if (_preparseRTP)
+ {
+ WebRtcNetEQ_RTPInfo rtpInfo;
+ // parse RTP header
+ rtp.parseHeader(rtpInfo);
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countA); // get start count for processor
+#endif
+
+ err = WebRtcNetEQ_RecInRTPStruct(_inst, &rtpInfo, rtp.payload(), rtp.payloadLen(), rtp.time() * _fsmult * 8);
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countB); // get stop count for processor
+ _totTimeRecIn.QuadPart += (countB.QuadPart - countA.QuadPart);
+#endif
+
+ }
+ else
+ {
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countA); // get start count for processor
+#endif
+
+ err = WebRtcNetEQ_RecIn(_inst, (WebRtc_Word16 *) rtp.datagram(), rtp.dataLen(), rtp.time() * _fsmult * 8);
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countB); // get stop count for processor
+ _totTimeRecIn.QuadPart += (countB.QuadPart - countA.QuadPart);
+#endif
+
+ }
+
+ if (err)
+ {
+ printError(rtp);
+ }
+
+ return (err);
+
+}
+
+
+WebRtc_Word16 NETEQTEST_NetEQClass::recOut(WebRtc_Word16 *outData, void *msInfo, enum WebRtcNetEQOutputType *outputType)
+{
+ int err;
+ WebRtc_Word16 outLen = 0;
+#ifdef WINDOWS_TIMING
+ LARGE_INTEGER countA, countB;
+#endif
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countA); // get start count for processor
+#endif
+
+ if (!msInfo)
+ {
+ // no msInfo given, do mono mode
+ err = WebRtcNetEQ_RecOut(_inst, outData, &outLen);
+ }
+ else
+ {
+ // master/slave mode
+ err = WebRtcNetEQ_RecOutMasterSlave(_inst, outData, &outLen, msInfo, static_cast<WebRtc_Word16>(_isMaster));
+ }
+
+#ifdef WINDOWS_TIMING
+ QueryPerformanceCounter(&countB); // get stop count for processor
+ _totTimeRecOut.QuadPart += (countB.QuadPart - countA.QuadPart);
+#endif
+
+ if (err)
+ {
+ printError();
+ }
+ else
+ {
+ int newfsmult = static_cast<int>(outLen / 80);
+
+ if (newfsmult != _fsmult)
+ {
+#ifdef NETEQTEST_PRINT_WARNINGS
+ printf("Warning: output sample rate changed\n");
+#endif // NETEQTEST_PRINT_WARNINGS
+ _fsmult = newfsmult;
+ }
+ }
+
+ if (outputType != NULL)
+ {
+ err = WebRtcNetEQ_GetSpeechOutputType(_inst, outputType);
+
+ if (err)
+ {
+ printError();
+ }
+ }
+
+ return (outLen);
+}
+
+
+WebRtc_UWord32 NETEQTEST_NetEQClass::getSpeechTimeStamp()
+{
+
+ WebRtc_UWord32 ts = 0;
+ int err;
+
+ err = WebRtcNetEQ_GetSpeechTimeStamp(_inst, &ts);
+
+ if (err)
+ {
+ printError();
+ ts = 0;
+ }
+
+ return (ts);
+
+}
+
+WebRtcNetEQOutputType NETEQTEST_NetEQClass::getOutputType() {
+ WebRtcNetEQOutputType type;
+
+ int err = WebRtcNetEQ_GetSpeechOutputType(_inst, &type);
+ if (err)
+ {
+ printError();
+ type = kOutputNormal;
+ }
+ return (type);
+}
+
+//NETEQTEST_NetEQVector::NETEQTEST_NetEQVector(int numChannels)
+//:
+//channels(numChannels, new NETEQTEST_NetEQClass())
+//{
+// //for (int i = 0; i < numChannels; i++)
+// //{
+// // channels.push_back(new NETEQTEST_NetEQClass());
+// //}
+//}
+//
+//NETEQTEST_NetEQVector::NETEQTEST_NetEQVector(int numChannels, enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
+// WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
+// :
+//channels(numChannels, new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, fs, nwType))
+//{
+// //for (int i = 0; i < numChannels; i++)
+// //{
+// // channels.push_back(new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, fs, nwType));
+// //}
+//}
+//
+//NETEQTEST_NetEQVector::~NETEQTEST_NetEQVector()
+//{
+//}
+
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
new file mode 100644
index 0000000..c425b58
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef NETEQTEST_NETEQCLASS_H
+#define NETEQTEST_NETEQCLASS_H
+
+#include <stdio.h>
+#include <vector>
+
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_internal.h"
+
+#include "NETEQTEST_RTPpacket.h"
+
+#ifdef WIN32
+#define WINDOWS_TIMING // complexity measurement only implemented for windows
+//TODO(hlundin):Add complexity testing for Linux.
+#include <windows.h>
+#endif
+
+class NETEQTEST_NetEQClass
+{
+public:
+ NETEQTEST_NetEQClass();
+ NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
+ WebRtc_UWord16 fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
+ ~NETEQTEST_NetEQClass();
+
+ int assign();
+ int init(WebRtc_UWord16 fs = 8000);
+ int assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
+ int loadCodec(WebRtcNetEQ_CodecDef & codecInst);
+ int recIn(NETEQTEST_RTPpacket & rtp);
+ WebRtc_Word16 recOut(WebRtc_Word16 *outData, void *msInfo = NULL, enum WebRtcNetEQOutputType *outputType = NULL);
+ WebRtc_UWord32 getSpeechTimeStamp();
+ WebRtcNetEQOutputType getOutputType();
+
+ void * instance() { return (_inst); };
+ void usePreparseRTP( bool useIt = true ) { _preparseRTP = useIt; };
+ bool usingPreparseRTP() { return (_preparseRTP); };
+ void setMaster( bool isMaster = true ) { _isMaster = isMaster; };
+ void setSlave() { _isMaster = false; };
+ bool isMaster() { return (_isMaster); };
+ bool isSlave() { return (!_isMaster); };
+
+#ifdef WINDOWS_TIMING
+ double getRecInTime() { return (static_cast<double>( _totTimeRecIn.QuadPart )); };
+ double getRecOutTime() { return (static_cast<double>( _totTimeRecOut.QuadPart )); };
+#else
+ double getRecInTime() { return (0.0); };
+ double getRecOutTime() { return (0.0); };
+
+#endif
+
+ void printError();
+ void printError(NETEQTEST_RTPpacket & rtp);
+
+private:
+ void * _inst;
+ WebRtc_Word8 * _instMem;
+ WebRtc_Word8 * _bufferMem;
+ bool _preparseRTP;
+ int _fsmult;
+ bool _isMaster;
+#ifdef WINDOWS_TIMING
+ LARGE_INTEGER _totTimeRecIn;
+ LARGE_INTEGER _totTimeRecOut;
+#endif
+};
+
+
+
+//class NETEQTEST_NetEQVector
+//{
+//public:
+// NETEQTEST_NetEQVector(int numChannels);
+// NETEQTEST_NetEQVector(int numChannels, enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
+// WebRtc_UWord16 fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
+// ~NETEQTEST_NetEQVector();
+//
+//private:
+// std::vector<NETEQTEST_NetEQClass *> channels;
+//};
+
+#endif //NETEQTEST_NETEQCLASS_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
new file mode 100644
index 0000000..0412f06
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
@@ -0,0 +1,875 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "NETEQTEST_RTPpacket.h"
+
+#include <assert.h>
+#include <string.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <netinet/in.h> // for htons, htonl, etc
+#endif
+
+#include <cstdlib>
+
+#include "gtest/gtest.h"
+
+#define HDR_SIZE 8 // rtpplay packet header size in bytes
+
+
+NETEQTEST_RTPpacket::NETEQTEST_RTPpacket()
+:
+_datagram(NULL),
+_payloadPtr(NULL),
+_memSize(0),
+_datagramLen(-1),
+_payloadLen(0),
+_rtpParsed(false),
+_receiveTime(0),
+_lost(false)
+{
+ memset(&_rtpInfo, 0, sizeof(_rtpInfo));
+ _blockList.clear();
+}
+
+NETEQTEST_RTPpacket::NETEQTEST_RTPpacket(const NETEQTEST_RTPpacket& copyFromMe)
+{
+
+ memcpy(this, ©FromMe, sizeof(NETEQTEST_RTPpacket));
+
+ _datagram = NULL;
+ _payloadPtr = NULL;
+
+ if(copyFromMe._datagram)
+ {
+ _datagram = new WebRtc_UWord8[_memSize];
+
+ if(_datagram)
+ {
+ memcpy(_datagram, copyFromMe._datagram, _memSize);
+ }
+ }
+
+ if(copyFromMe._payloadPtr)
+ {
+ _payloadPtr = _datagram + (copyFromMe._payloadPtr - copyFromMe._datagram);
+ }
+
+ _blockList = copyFromMe._blockList;
+
+}
+
+
+NETEQTEST_RTPpacket & NETEQTEST_RTPpacket::operator = (const NETEQTEST_RTPpacket & other)
+{
+ if (this != &other) // protect against invalid self-assignment
+ {
+
+ // deallocate datagram memory if allocated
+ if(_datagram)
+ {
+ delete [] _datagram;
+ }
+
+ // do shallow copy
+ memcpy(this, &other, sizeof(NETEQTEST_RTPpacket));
+
+ // reset pointers
+ _datagram = NULL;
+ _payloadPtr = NULL;
+
+ if(other._datagram)
+ {
+ _datagram = new WebRtc_UWord8[other._memSize];
+ _memSize = other._memSize;
+
+ if(_datagram)
+ {
+ memcpy(_datagram, other._datagram, _memSize);
+ }
+ }
+
+ if(other._payloadPtr)
+ {
+ _payloadPtr = _datagram + (other._payloadPtr - other._datagram);
+ }
+
+ // copy the blocking list (map)
+ _blockList = other._blockList;
+
+ }
+
+ // by convention, always return *this
+ return *this;
+}
+
+
+
+NETEQTEST_RTPpacket::~NETEQTEST_RTPpacket()
+{
+ if(_datagram)
+ {
+ delete [] _datagram;
+ }
+}
+
+
+void NETEQTEST_RTPpacket::reset()
+{
+ if(_datagram) {
+ delete [] _datagram;
+ }
+ _datagram = NULL;
+ _memSize = 0;
+ _datagramLen = -1;
+ _payloadLen = 0;
+ _payloadPtr = NULL;
+ _receiveTime = 0;
+ memset(&_rtpInfo, 0, sizeof(_rtpInfo));
+ _rtpParsed = false;
+
+}
+
+int NETEQTEST_RTPpacket::skipFileHeader(FILE *fp)
+{
+ if (!fp) {
+ return -1;
+ }
+
+ const int kFirstLineLength = 40;
+ char firstline[kFirstLineLength];
+ if (fgets(firstline, kFirstLineLength, fp) == NULL) {
+ return -1;
+ }
+ if (strncmp(firstline, "#!rtpplay", 9) == 0) {
+ if (strncmp(firstline, "#!rtpplay1.0", 12) != 0) {
+ return -1;
+ }
+ }
+ else if (strncmp(firstline, "#!RTPencode", 11) == 0) {
+ if (strncmp(firstline, "#!RTPencode1.0", 14) != 0) {
+ return -1;
+ }
+ }
+ else
+ {
+ return -1;
+ }
+
+ const int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+ if (fseek(fp, kRtpDumpHeaderSize, SEEK_CUR) != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+int NETEQTEST_RTPpacket::readFromFile(FILE *fp)
+{
+ if(!fp)
+ {
+ return(-1);
+ }
+
+ WebRtc_UWord16 length, plen;
+ WebRtc_UWord32 offset;
+
+ if (fread(&length,2,1,fp)==0)
+ {
+ reset();
+ return(-2);
+ }
+ length = ntohs(length);
+
+ if (fread(&plen,2,1,fp)==0)
+ {
+ reset();
+ return(-1);
+ }
+ int packetLen = ntohs(plen);
+
+ if (fread(&offset,4,1,fp)==0)
+ {
+ reset();
+ return(-1);
+ }
+ WebRtc_UWord32 receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
+
+ // Use length here because a plen of 0 specifies rtcp
+ length = (WebRtc_UWord16) (length - HDR_SIZE);
+
+ // check buffer size
+ if (_datagram && _memSize < length)
+ {
+ reset();
+ }
+
+ if (!_datagram)
+ {
+ _datagram = new WebRtc_UWord8[length];
+ _memSize = length;
+ }
+
+ if (fread((unsigned short *) _datagram,1,length,fp) != length)
+ {
+ reset();
+ return(-1);
+ }
+
+ _datagramLen = length;
+ _receiveTime = receiveTime;
+
+ if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
+ {
+ // discard this payload
+ return(readFromFile(fp));
+ }
+
+ return(packetLen);
+
+}
+
+
+int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, size_t length)
+{
+ if (!fp)
+ {
+ return -1;
+ }
+
+ // check buffer size
+ if (_datagram && _memSize < static_cast<int>(length))
+ {
+ reset();
+ }
+
+ if (!_datagram)
+ {
+ _datagram = new WebRtc_UWord8[length];
+ _memSize = length;
+ }
+
+ if (fread(_datagram, 1, length, fp) != length)
+ {
+ reset();
+ return -1;
+ }
+
+ _datagramLen = length;
+ _receiveTime = 0;
+
+ if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
+ {
+ // discard this payload
+ return readFromFile(fp);
+ }
+
+ return length;
+
+}
+
+
+int NETEQTEST_RTPpacket::writeToFile(FILE *fp)
+{
+ if (!fp)
+ {
+ return -1;
+ }
+
+ WebRtc_UWord16 length, plen;
+ WebRtc_UWord32 offset;
+
+ // length including RTPplay header
+ length = htons(_datagramLen + HDR_SIZE);
+ if (fwrite(&length, 2, 1, fp) != 1)
+ {
+ return -1;
+ }
+
+ // payload length
+ plen = htons(_datagramLen);
+ if (fwrite(&plen, 2, 1, fp) != 1)
+ {
+ return -1;
+ }
+
+ // offset (=receive time)
+ offset = htonl(_receiveTime);
+ if (fwrite(&offset, 4, 1, fp) != 1)
+ {
+ return -1;
+ }
+
+
+ // write packet data
+ if (fwrite(_datagram, 1, _datagramLen, fp) !=
+ static_cast<size_t>(_datagramLen))
+ {
+ return -1;
+ }
+
+ return _datagramLen + HDR_SIZE; // total number of bytes written
+
+}
+
+
+void NETEQTEST_RTPpacket::blockPT(WebRtc_UWord8 pt)
+{
+ _blockList[pt] = true;
+}
+
+
+void NETEQTEST_RTPpacket::parseHeader()
+{
+ if (_rtpParsed)
+ {
+ // nothing to do
+ return;
+ }
+
+ if (_datagramLen < 12)
+ {
+ // corrupt packet?
+ return;
+ }
+
+ _payloadLen = parseRTPheader(_datagram, _datagramLen, &_rtpInfo, &_payloadPtr);
+
+ _rtpParsed = true;
+
+ return;
+
+}
+
+void NETEQTEST_RTPpacket::parseHeader(WebRtcNetEQ_RTPInfo & rtpInfo)
+{
+ if (!_rtpParsed)
+ {
+ // parse the header
+ parseHeader();
+ }
+
+ memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcNetEQ_RTPInfo));
+}
+
+WebRtcNetEQ_RTPInfo const * NETEQTEST_RTPpacket::RTPinfo() const
+{
+ if (_rtpParsed)
+ {
+ return &_rtpInfo;
+ }
+ else
+ {
+ return NULL;
+ }
+}
+
+WebRtc_UWord8 * NETEQTEST_RTPpacket::datagram() const
+{
+ if (_datagramLen > 0)
+ {
+ return _datagram;
+ }
+ else
+ {
+ return NULL;
+ }
+}
+
+WebRtc_UWord8 * NETEQTEST_RTPpacket::payload() const
+{
+ if (_payloadLen > 0)
+ {
+ return _payloadPtr;
+ }
+ else
+ {
+ return NULL;
+ }
+}
+
+WebRtc_Word16 NETEQTEST_RTPpacket::payloadLen() const
+{
+ return _payloadLen;
+}
+
+WebRtc_Word16 NETEQTEST_RTPpacket::dataLen() const
+{
+ return _datagramLen;
+}
+
+bool NETEQTEST_RTPpacket::isParsed() const
+{
+ return _rtpParsed;
+}
+
+bool NETEQTEST_RTPpacket::isLost() const
+{
+ return _lost;
+}
+
+WebRtc_UWord8 NETEQTEST_RTPpacket::payloadType() const
+{
+ WebRtcNetEQ_RTPInfo tempRTPinfo;
+
+ if(_datagram)
+ {
+ parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
+ }
+ else
+ {
+ return 0;
+ }
+
+ return tempRTPinfo.payloadType;
+}
+
+WebRtc_UWord16 NETEQTEST_RTPpacket::sequenceNumber() const
+{
+ WebRtcNetEQ_RTPInfo tempRTPinfo;
+
+ if(_datagram)
+ {
+ parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
+ }
+ else
+ {
+ return 0;
+ }
+
+ return tempRTPinfo.sequenceNumber;
+}
+
+WebRtc_UWord32 NETEQTEST_RTPpacket::timeStamp() const
+{
+ WebRtcNetEQ_RTPInfo tempRTPinfo;
+
+ if(_datagram)
+ {
+ parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
+ }
+ else
+ {
+ return 0;
+ }
+
+ return tempRTPinfo.timeStamp;
+}
+
+WebRtc_UWord32 NETEQTEST_RTPpacket::SSRC() const
+{
+ WebRtcNetEQ_RTPInfo tempRTPinfo;
+
+ if(_datagram)
+ {
+ parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
+ }
+ else
+ {
+ return 0;
+ }
+
+ return tempRTPinfo.SSRC;
+}
+
+WebRtc_UWord8 NETEQTEST_RTPpacket::markerBit() const
+{
+ WebRtcNetEQ_RTPInfo tempRTPinfo;
+
+ if(_datagram)
+ {
+ parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
+ }
+ else
+ {
+ return 0;
+ }
+
+ return tempRTPinfo.markerBit;
+}
+
+
+
+int NETEQTEST_RTPpacket::setPayloadType(WebRtc_UWord8 pt)
+{
+
+ if (_datagramLen < 12)
+ {
+ return -1;
+ }
+
+ if (!_rtpParsed)
+ {
+ _rtpInfo.payloadType = pt;
+ }
+
+ _datagram[1]=(unsigned char)(pt & 0xFF);
+
+ return 0;
+
+}
+
+int NETEQTEST_RTPpacket::setSequenceNumber(WebRtc_UWord16 sn)
+{
+
+ if (_datagramLen < 12)
+ {
+ return -1;
+ }
+
+ if (!_rtpParsed)
+ {
+ _rtpInfo.sequenceNumber = sn;
+ }
+
+ _datagram[2]=(unsigned char)((sn>>8)&0xFF);
+ _datagram[3]=(unsigned char)((sn)&0xFF);
+
+ return 0;
+
+}
+
+int NETEQTEST_RTPpacket::setTimeStamp(WebRtc_UWord32 ts)
+{
+
+ if (_datagramLen < 12)
+ {
+ return -1;
+ }
+
+ if (!_rtpParsed)
+ {
+ _rtpInfo.timeStamp = ts;
+ }
+
+ _datagram[4]=(unsigned char)((ts>>24)&0xFF);
+ _datagram[5]=(unsigned char)((ts>>16)&0xFF);
+ _datagram[6]=(unsigned char)((ts>>8)&0xFF);
+ _datagram[7]=(unsigned char)(ts & 0xFF);
+
+ return 0;
+
+}
+
+int NETEQTEST_RTPpacket::setSSRC(WebRtc_UWord32 ssrc)
+{
+
+ if (_datagramLen < 12)
+ {
+ return -1;
+ }
+
+ if (!_rtpParsed)
+ {
+ _rtpInfo.SSRC = ssrc;
+ }
+
+ _datagram[8]=(unsigned char)((ssrc>>24)&0xFF);
+ _datagram[9]=(unsigned char)((ssrc>>16)&0xFF);
+ _datagram[10]=(unsigned char)((ssrc>>8)&0xFF);
+ _datagram[11]=(unsigned char)(ssrc & 0xFF);
+
+ return 0;
+
+}
+
+int NETEQTEST_RTPpacket::setMarkerBit(WebRtc_UWord8 mb)
+{
+
+ if (_datagramLen < 12)
+ {
+ return -1;
+ }
+
+ if (_rtpParsed)
+ {
+ _rtpInfo.markerBit = mb;
+ }
+
+ if (mb)
+ {
+ _datagram[0] |= 0x01;
+ }
+ else
+ {
+ _datagram[0] &= 0xFE;
+ }
+
+ return 0;
+
+}
+
+int NETEQTEST_RTPpacket::setRTPheader(const WebRtcNetEQ_RTPInfo *RTPinfo)
+{
+ if (_datagramLen < 12)
+ {
+ // this packet is not ok
+ return -1;
+ }
+
+ makeRTPheader(_datagram,
+ RTPinfo->payloadType,
+ RTPinfo->sequenceNumber,
+ RTPinfo->timeStamp,
+ RTPinfo->SSRC,
+ RTPinfo->markerBit);
+
+ return 0;
+}
+
+
+int NETEQTEST_RTPpacket::splitStereo(NETEQTEST_RTPpacket& slaveRtp, enum stereoModes mode)
+{
+ // if mono, do nothing
+ if (mode == stereoModeMono)
+ {
+ return 0;
+ }
+
+ // check that the RTP header info is parsed
+ parseHeader();
+
+ // start by copying the main rtp packet
+ slaveRtp = *this;
+
+ if(_payloadLen == 0)
+ {
+ // do no more
+ return 0;
+ }
+
+ if(_payloadLen%2 != 0)
+ {
+ // length must be a factor of 2
+ return -1;
+ }
+
+ switch(mode)
+ {
+ case stereoModeSample1:
+ {
+ // sample based codec with 1-byte samples
+ splitStereoSample(slaveRtp, 1 /* 1 byte/sample */);
+ break;
+ }
+ case stereoModeSample2:
+ {
+ // sample based codec with 2-byte samples
+ splitStereoSample(slaveRtp, 2 /* 2 bytes/sample */);
+ break;
+ }
+ case stereoModeFrame:
+ {
+ // frame based codec
+ splitStereoFrame(slaveRtp);
+ break;
+ }
+ case stereoModeMono:
+ {
+ assert(false);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+
+void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const
+{
+ rtp_data[0]=(unsigned char)0x80;
+ if (markerBit)
+ {
+ rtp_data[0] |= 0x01;
+ }
+ else
+ {
+ rtp_data[0] &= 0xFE;
+ }
+ rtp_data[1]=(unsigned char)(payloadType & 0xFF);
+ rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
+ rtp_data[3]=(unsigned char)((seqNo)&0xFF);
+ rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
+ rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
+
+ rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
+ rtp_data[7]=(unsigned char)(timestamp & 0xFF);
+
+ rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
+ rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
+
+ rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
+ rtp_data[11]=(unsigned char)(ssrc & 0xFF);
+}
+
+
+WebRtc_UWord16 NETEQTEST_RTPpacket::parseRTPheader(const WebRtc_UWord8 *datagram, int datagramLen, WebRtcNetEQ_RTPInfo *RTPinfo, WebRtc_UWord8 **payloadPtr) const
+{
+ WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) datagram;
+ int i_P, i_X, i_CC, i_extlength=-1, i_padlength=0, i_startPosition;
+
+ i_P=(((WebRtc_UWord16)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
+ i_X=(((WebRtc_UWord16)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
+ i_CC=(WebRtc_UWord16)(rtp_data[0] & 0xF); /* Get the CC number */
+ RTPinfo->markerBit = (WebRtc_UWord8) ((rtp_data[0] >> 15) & 0x01); /* Get the marker bit */
+ RTPinfo->payloadType = (WebRtc_UWord8) ((rtp_data[0] >> 8) & 0x7F); /* Get the coder type */
+ RTPinfo->sequenceNumber = ((( ((WebRtc_UWord16)rtp_data[1]) >> 8) & 0xFF) |
+ ( ((WebRtc_UWord16)(rtp_data[1] & 0xFF)) << 8)); /* Get the packet number */
+ RTPinfo->timeStamp = ((((WebRtc_UWord16)rtp_data[2]) & 0xFF) << 24) |
+ ((((WebRtc_UWord16)rtp_data[2]) & 0xFF00) << 8) |
+ ((((WebRtc_UWord16)rtp_data[3]) >> 8) & 0xFF) |
+ ((((WebRtc_UWord16)rtp_data[3]) & 0xFF) << 8); /* Get timestamp */
+ RTPinfo->SSRC=((((WebRtc_UWord16)rtp_data[4]) & 0xFF) << 24) |
+ ((((WebRtc_UWord16)rtp_data[4]) & 0xFF00) << 8) |
+ ((((WebRtc_UWord16)rtp_data[5]) >> 8) & 0xFF) |
+ ((((WebRtc_UWord16)rtp_data[5]) & 0xFF) << 8); /* Get the SSRC */
+
+ if (i_X==1) {
+ /* Extention header exists. Find out how many WebRtc_Word32 it consists of */
+ i_extlength=((( ((WebRtc_UWord16)rtp_data[7+2*i_CC]) >> 8) & 0xFF) |
+ ( ((WebRtc_UWord16)(rtp_data[7+2*i_CC]&0xFF)) << 8));
+ }
+ if (i_P==1) {
+ /* Padding exists. Find out how many bytes the padding consists of */
+ if (datagramLen & 0x1) {
+ /* odd number of bytes => last byte in higher byte */
+ i_padlength=(rtp_data[datagramLen>>1] & 0xFF);
+ } else {
+ /* even number of bytes => last byte in lower byte */
+ i_padlength=(((WebRtc_UWord16)rtp_data[(datagramLen>>1)-1]) >> 8);
+ }
+ }
+
+ i_startPosition=12+4*(i_extlength+1)+4*i_CC;
+
+ if (payloadPtr) {
+ *payloadPtr = (WebRtc_UWord8*) &rtp_data[i_startPosition>>1];
+ }
+
+ return (WebRtc_UWord16) (datagramLen-i_startPosition-i_padlength);
+}
+
+//void NETEQTEST_RTPpacket::splitStereoSample(WebRtc_UWord8 *data, WebRtc_UWord16 *lenBytes, WebRtc_UWord8 *slaveData, WebRtc_UWord16 *slaveLenBytes, int stride)
+void NETEQTEST_RTPpacket::splitStereoSample(NETEQTEST_RTPpacket& slaveRtp, int stride)
+{
+ if(!_payloadPtr || !slaveRtp._payloadPtr
+ || _payloadLen <= 0 || slaveRtp._memSize < _memSize)
+ {
+ return;
+ }
+
+ WebRtc_UWord8 *readDataPtr = _payloadPtr;
+ WebRtc_UWord8 *writeDataPtr = _payloadPtr;
+ WebRtc_UWord8 *slaveData = slaveRtp._payloadPtr;
+
+ while (readDataPtr - _payloadPtr < _payloadLen)
+ {
+ // master data
+ for (int ix = 0; ix < stride; ix++) {
+ *writeDataPtr = *readDataPtr;
+ writeDataPtr++;
+ readDataPtr++;
+ }
+
+ // slave data
+ for (int ix = 0; ix < stride; ix++) {
+ *slaveData = *readDataPtr;
+ slaveData++;
+ readDataPtr++;
+ }
+ }
+
+ _payloadLen /= 2;
+ slaveRtp._payloadLen = _payloadLen;
+}
+
+
+//void NETEQTEST_RTPpacket::splitStereoFrame(WebRtc_UWord8 *data, WebRtc_UWord16 *lenBytes, WebRtc_UWord8 *slaveData, WebRtc_UWord16 *slaveLenBytes)
+void NETEQTEST_RTPpacket::splitStereoFrame(NETEQTEST_RTPpacket& slaveRtp)
+{
+ if(!_payloadPtr || !slaveRtp._payloadPtr
+ || _payloadLen <= 0 || slaveRtp._memSize < _memSize)
+ {
+ return;
+ }
+
+ memmove(slaveRtp._payloadPtr, _payloadPtr + _payloadLen/2, _payloadLen/2);
+
+ _payloadLen /= 2;
+ slaveRtp._payloadLen = _payloadLen;
+}
+
+// Get the RTP header for the RED payload indicated by argument index.
+// The first RED payload is index = 0.
+int NETEQTEST_RTPpacket::extractRED(int index, WebRtcNetEQ_RTPInfo& red)
+{
+//
+// 0 1 2 3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |1| block PT | timestamp offset | block length |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |1| ... |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |0| block PT |
+// +-+-+-+-+-+-+-+-+
+//
+
+ parseHeader();
+
+ WebRtc_UWord8* ptr = payload();
+ WebRtc_UWord8* payloadEndPtr = ptr + payloadLen();
+ int num_encodings = 0;
+ int total_len = 0;
+
+ while ((ptr < payloadEndPtr) && (*ptr & 0x80))
+ {
+ int len = ((ptr[2] & 0x03) << 8) + ptr[3];
+ if (num_encodings == index)
+ {
+ // Header found.
+ red.payloadType = ptr[0] & 0x7F;
+ WebRtc_UWord32 offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
+ red.sequenceNumber = sequenceNumber();
+ red.timeStamp = timeStamp() - offset;
+ red.markerBit = markerBit();
+ red.SSRC = SSRC();
+ return len;
+ }
+ ++num_encodings;
+ total_len += len;
+ ptr += 4;
+ }
+ if ((ptr < payloadEndPtr) && (num_encodings == index))
+ {
+ // Last header.
+ red.payloadType = ptr[0] & 0x7F;
+ red.sequenceNumber = sequenceNumber();
+ red.timeStamp = timeStamp();
+ red.markerBit = markerBit();
+ red.SSRC = SSRC();
+ ++ptr;
+ return payloadLen() - (ptr - payload()) - total_len;
+ }
+ return -1;
+}
+
+// Randomize the payload, not the RTP header.
+void NETEQTEST_RTPpacket::scramblePayload(void)
+{
+ parseHeader();
+
+ for (int i = 0; i < _payloadLen; ++i)
+ {
+ _payloadPtr[i] = static_cast<WebRtc_UWord8>(std::rand());
+ }
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
new file mode 100644
index 0000000..0478568
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef NETEQTEST_RTPPACKET_H
+#define NETEQTEST_RTPPACKET_H
+
+#include <map>
+#include <stdio.h>
+#include "typedefs.h"
+#include "webrtc_neteq_internal.h"
+
+enum stereoModes {
+ stereoModeMono,
+ stereoModeSample1,
+ stereoModeSample2,
+ stereoModeFrame
+};
+
+class NETEQTEST_RTPpacket
+{
+public:
+ NETEQTEST_RTPpacket();
+ NETEQTEST_RTPpacket(const NETEQTEST_RTPpacket& copyFromMe);
+ NETEQTEST_RTPpacket & operator = (const NETEQTEST_RTPpacket & other);
+ bool operator !() const { return (dataLen() < 0); };
+ ~NETEQTEST_RTPpacket();
+ void reset();
+ static int skipFileHeader(FILE *fp);
+ int readFromFile(FILE *fp);
+ int readFixedFromFile(FILE *fp, size_t len);
+ int writeToFile(FILE *fp);
+ void blockPT(WebRtc_UWord8 pt);
+ //WebRtc_Word16 payloadType();
+ void parseHeader();
+ void parseHeader(WebRtcNetEQ_RTPInfo & rtpInfo);
+ WebRtcNetEQ_RTPInfo const * RTPinfo() const;
+ WebRtc_UWord8 * datagram() const;
+ WebRtc_UWord8 * payload() const;
+ WebRtc_Word16 payloadLen() const;
+ WebRtc_Word16 dataLen() const;
+ bool isParsed() const;
+ bool isLost() const;
+ WebRtc_UWord32 time() const { return _receiveTime; };
+
+ WebRtc_UWord8 payloadType() const;
+ WebRtc_UWord16 sequenceNumber() const;
+ WebRtc_UWord32 timeStamp() const;
+ WebRtc_UWord32 SSRC() const;
+ WebRtc_UWord8 markerBit() const;
+
+ int setPayloadType(WebRtc_UWord8 pt);
+ int setSequenceNumber(WebRtc_UWord16 sn);
+ int setTimeStamp(WebRtc_UWord32 ts);
+ int setSSRC(WebRtc_UWord32 ssrc);
+ int setMarkerBit(WebRtc_UWord8 mb);
+ void setTime(WebRtc_UWord32 receiveTime) { _receiveTime = receiveTime; };
+
+ int setRTPheader(const WebRtcNetEQ_RTPInfo *RTPinfo);
+
+ int splitStereo(NETEQTEST_RTPpacket& slaveRtp, enum stereoModes mode);
+
+ int extractRED(int index, WebRtcNetEQ_RTPInfo& red);
+
+ void scramblePayload(void);
+
+ WebRtc_UWord8 * _datagram;
+ WebRtc_UWord8 * _payloadPtr;
+ int _memSize;
+ WebRtc_Word16 _datagramLen;
+ WebRtc_Word16 _payloadLen;
+ WebRtcNetEQ_RTPInfo _rtpInfo;
+ bool _rtpParsed;
+ WebRtc_UWord32 _receiveTime;
+ bool _lost;
+ std::map<WebRtc_UWord8, bool> _blockList;
+
+private:
+ void makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const;
+ WebRtc_UWord16 parseRTPheader(const WebRtc_UWord8 *datagram, int datagramLen, WebRtcNetEQ_RTPInfo *RTPinfo, WebRtc_UWord8 **payloadPtr = NULL) const;
+ void splitStereoSample(NETEQTEST_RTPpacket& slaveRtp, int stride);
+ void splitStereoFrame(NETEQTEST_RTPpacket& slaveRtp);
+};
+
+#endif //NETEQTEST_RTPPACKET_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc b/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
new file mode 100644
index 0000000..d3f7eb5
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
@@ -0,0 +1,1736 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//TODO(hlundin): Reformat file to meet style guide.
+
+/* header includes */
+#include "typedefs.h"
+#include "stdio.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_internal.h"
+#include "webrtc_neteq_help_macros.h"
+#include "neteq_error_codes.h" // for the API test
+
+#include "NETEQTEST_RTPpacket.h"
+#include "NETEQTEST_NetEQClass.h"
+#include "NETEQTEST_CodecClass.h"
+
+#include <string.h>
+#include <stdlib.h>
+#include <time.h>
+#include <map>
+#include <vector>
+
+#ifdef WIN32
+#include <cassert>
+#include <windows.h>
+#endif
+
+#ifdef WEBRTC_LINUX
+#include <netinet/in.h>
+#include <libgen.h>
+#include <cassert>
+#endif
+
+//#include "vld.h"
+
+//#define NETEQ_DELAY_LOGGING
+//#define DUMMY_SLAVE_CHANNEL
+
+#ifdef NETEQ_DELAY_LOGGING
+#include "delay_logging.h"
+#define DUMMY_SLAVE_CHANNEL // do not use a slave channel, only generate zeros instead
+#endif
+
+
+/************************/
+/* Define payload types */
+/************************/
+
+// Payload types are defined in the textfile ptypes.txt, and can be changed after compilation.
+
+
+
+/*********************/
+/* Misc. definitions */
+/*********************/
+
+#define TIME_STEP 1
+#define FIRSTLINELEN 40
+#define MAX_NETEQ_BUFFERSIZE 170000 //100000
+#define CHECK_ZERO(a) {int errCode = a; char tempErrName[WEBRTC_NETEQ_MAX_ERROR_NAME]; if((errCode)!=0){errCode = WebRtcNetEQ_GetErrorCode(inst); WebRtcNetEQ_GetErrorName(errCode, tempErrName, WEBRTC_NETEQ_MAX_ERROR_NAME); printf("\n %s \n line: %d \n error at %s\n Error Code = %d\n",__FILE__,__LINE__,#a, errCode); exit(0);}}
+#define CHECK_NOT_NULL(a) if((a)==NULL){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
+//#define PLAY_CLEAN // ignore arrival times and let the packets arrive according to RTP timestamps
+#define HDR_SIZE 8 // rtpplay packet header size in bytes
+//#define JUNK_DATA // scramble the payloads to test error resilience
+//#define ZERO_TS_START
+
+#ifdef JUNK_DATA
+ #define SEED_FILE "randseed.txt"
+#endif
+
+#ifdef WIN32
+#define MY_MAX_DRIVE _MAX_DRIVE
+#define MY_MAX_PATH _MAX_PATH
+#define MY_MAX_FNAME _MAX_FNAME
+#define MY_MAX_EXT _MAX_EXT
+
+#elif defined(WEBRTC_LINUX)
+#include <linux/limits.h>
+#define MY_MAX_PATH PATH_MAX
+
+#elif defined(WEBRTC_MAC)
+#include <sys/syslimits.h>
+#define MY_MAX_PATH PATH_MAX
+#endif // WEBRTC_MAC
+
+/************/
+/* Typedefs */
+/************/
+
+typedef struct {
+ enum WebRtcNetEQDecoder codec;
+ enum stereoModes stereo;
+ NETEQTEST_Decoder * decoder[2];
+ int fs;
+} decoderStruct;
+
+
+/*************************/
+/* Function declarations */
+/*************************/
+
+void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen);
+int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime);
+void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d);
+bool splitStereo(NETEQTEST_RTPpacket& rtp, NETEQTEST_RTPpacket& rtpSlave,
+ const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
+ const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
+ bool *isStereo);
+void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders);
+int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec);
+void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber);
+void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders);
+int doAPItest();
+bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode);
+
+
+
+/********************/
+/* Global variables */
+/********************/
+
+WebRtc_Word16 NetEqPacketBuffer[MAX_NETEQ_BUFFERSIZE>>1];
+WebRtc_Word16 NetEqPacketBufferSlave[MAX_NETEQ_BUFFERSIZE>>1];
+
+#ifdef NETEQ_DELAY_LOGGING
+extern "C" {
+ FILE *delay_fid2; /* file pointer */
+ WebRtc_UWord32 tot_received_packets=0;
+}
+#endif
+
+#ifdef DEF_BUILD_DATE
+extern char BUILD_DATE;
+#endif
+
+WebRtc_UWord32 writtenSamples = 0;
+WebRtc_UWord32 simClock=0;
+
+int main(int argc, char* argv[])
+{
+ std::vector<NETEQTEST_NetEQClass *> NetEQvector;
+ NETEQTEST_RTPpacket rtp;
+ char version[20];
+
+ NETEQTEST_RTPpacket slaveRtp;
+ //bool switchMS = false;
+ //bool duplicatePayload = false;
+ enum WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd-1];
+ int noOfCodecs;
+ int ok;
+ WebRtc_Word16 out_data[640*2];
+ WebRtc_Word16 outLen, writeLen;
+ int fs = 8000;
+ WebRtcNetEQ_RTCPStat RTCPstat;
+#ifdef WIN32
+ char outdrive[MY_MAX_DRIVE];
+ char outpath[MY_MAX_PATH];
+ char outfile[MY_MAX_FNAME];
+ char outext[MY_MAX_EXT];
+#endif
+ char outfilename[MY_MAX_PATH];
+#ifdef NETEQ_DELAY_LOGGING
+ float clock_float;
+ int temp_var;
+#endif
+#ifdef JUNK_DATA
+ FILE *seedfile;
+#endif
+ FILE *recoutTimes = NULL;
+ FILE *extraDelays = NULL;
+ WebRtcNetEQPlayoutMode streamingMode = kPlayoutOn;
+ bool preParseRTP = false;
+ bool rtpOnly = false;
+ int packetLen = 0;
+ int packetCount = 0;
+ std::map<WebRtc_UWord8, decoderStruct> decoders;
+
+ /* get the version string */
+ WebRtcNetEQ_GetVersion(version);
+ printf("\n\nNetEq version: %s\n", version);
+#ifdef DEF_BUILD_DATE
+ printf("Build time: %s\n", __BUILD_DATE);
+#endif
+
+ /* check number of parameters */
+ if ((argc < 3)
+#ifdef WIN32 // implicit output file name possible for windows
+ && (argc < 2)
+#endif
+ ) {
+ /* print help text and exit */
+ printf("Test program for NetEQ.\n");
+ printf("The program reads an RTP stream from file and inserts it into NetEQ.\n");
+ printf("The format of the RTP stream file should be the same as for rtpplay,\n");
+ printf("and can be obtained e.g., from Ethereal by using\n");
+ printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> Save As\n\n");
+ printf("Usage:\n\n");
+#ifdef WIN32
+ printf("%s RTPfile [outfile] [-options]\n", argv[0]);
+#else
+ printf("%s RTPfile outfile [-options]\n", argv[0]);
+#endif
+ printf("where:\n");
+
+ printf("RTPfile : RTP stream input file\n\n");
+
+ printf("outfile : PCM speech output file\n");
+ printf(" Output file name is derived from RTP file name if omitted\n\n");
+
+ printf("-options are optional switches:\n");
+ printf("\t-recout datfile : supply recout times\n");
+ printf("\t-extradelay datfile : supply extra delay settings and timing\n");
+ printf("\t-streaming : engage streaming mode\n");
+ printf("\t-fax : engage fax mode\n");
+ printf("\t-preparsertp : use RecIn with pre-parsed RTP\n");
+ printf("\t-rtponly packLenBytes : input file consists of constant size RTP packets without RTPplay headers\n");
+ //printf("\t-switchms : switch from mono to stereo (copy channel) after 10 seconds\n");
+ //printf("\t-duplicate : use two instances with identical input (2-channel mono)\n");
+
+ return(0);
+ }
+
+ if (strcmp(argv[1], "-apitest")==0) {
+ // do API test and then return
+ ok=doAPItest();
+
+ if (ok==0)
+ printf("API test successful!\n");
+ else
+ printf("API test failed!\n");
+
+ return(ok);
+ }
+
+ FILE* in_file=fopen(argv[1],"rb");
+ CHECK_NOT_NULL(in_file);
+ printf("Input file: %s\n",argv[1]);
+
+ int argIx = 2; // index of next argument from command line
+
+ if ( argc >= 3 && argv[2][0] != '-' ) { // output name given on command line
+ strcpy(outfilename, argv[2]);
+ argIx++;
+ } else { // derive output name from input name
+#ifdef WIN32
+ _splitpath(argv[1],outdrive,outpath,outfile,outext);
+ _makepath(outfilename,outdrive,outpath,outfile,"pcm");
+#else
+ fprintf(stderr,"Output file name must be specified.\n");
+ return(-1);
+#endif
+ }
+ FILE* out_file=fopen(outfilename,"wb");
+ if (out_file==NULL) {
+ fprintf(stderr,"Could not open file %s for writing\n", outfilename);
+ return(-1);
+ }
+ printf("Output file: %s\n",outfilename);
+
+ // Parse for more arguments, all beginning with '-'
+
+ while( argIx < argc ) {
+ if (argv[argIx][0] != '-') {
+ fprintf(stderr,"Unknown input argument %s\n", argv[argIx]);
+ return(-1);
+ }
+
+ if( strcmp(argv[argIx], "-recout") == 0 ) {
+ argIx++;
+ recoutTimes = fopen(argv[argIx], "rb");
+ CHECK_NOT_NULL(recoutTimes);
+ argIx++;
+ }
+ else if( strcmp(argv[argIx], "-extradelay") == 0 ) {
+ argIx++;
+ extraDelays = fopen(argv[argIx], "rb");
+ CHECK_NOT_NULL(extraDelays);
+ argIx++;
+ }
+ else if( strcmp(argv[argIx], "-streaming") == 0 ) {
+ argIx++;
+ streamingMode = kPlayoutStreaming;
+ }
+ else if( strcmp(argv[argIx], "-fax") == 0 ) {
+ argIx++;
+ streamingMode = kPlayoutFax;
+ }
+ else if( strcmp(argv[argIx], "-preparsertp") == 0 ) {
+ argIx++;
+ preParseRTP = true;
+ }
+ else if( strcmp(argv[argIx], "-rtponly") == 0 ) {
+ argIx++;
+ rtpOnly = true;
+ packetLen = atoi(argv[argIx]);
+ argIx++;
+ if (packetLen <= 0)
+ {
+ printf("Wrong packet size used with argument -rtponly.\n");
+ exit(1);
+ }
+ }
+ //else if( strcmp(argv[argIx], "-switchms") == 0 ) {
+ // argIx++;
+ // switchMS = true;
+ //}
+ //else if( strcmp(argv[argIx], "-duplicate") == 0 ) {
+ // argIx++;
+ // duplicatePayload = true;
+ //}
+ else {
+ fprintf(stderr,"Unknown input argument %s\n", argv[argIx]);
+ return(-1);
+ }
+ }
+
+
+
+#ifdef NETEQ_DELAY_LOGGING
+ char delayfile[MY_MAX_PATH];
+#ifdef WIN32
+ _splitpath(outfilename,outdrive,outpath,outfile,outext);
+ _makepath(delayfile,outdrive,outpath,outfile,"d");
+#else
+ sprintf(delayfile, "%s.d", outfilename);
+#endif
+ delay_fid2 = fopen(delayfile,"wb");
+ fprintf(delay_fid2, "#!NetEQ_Delay_Logging%s\n", NETEQ_DELAY_LOGGING_VERSION_STRING);
+#endif
+
+ char ptypesfile[MY_MAX_PATH];
+#ifdef WIN32
+ _splitpath(argv[0],outdrive,outpath,outfile,outext);
+ _makepath(ptypesfile,outdrive,outpath,"ptypes","txt");
+#else
+ // TODO(hlundin): Include path to ptypes, as for WIN32 above.
+ strcpy(ptypesfile, "ptypes.txt");
+#endif
+ FILE *ptypeFile = fopen(ptypesfile,"rt");
+ if (!ptypeFile) {
+ // Check if we can find the file at the usual place in the trunk.
+ if (strstr(argv[0], "out/Debug/")) {
+ int path_len = strstr(argv[0], "out/Debug/") - argv[0];
+ strncpy(ptypesfile, argv[0], path_len);
+ ptypesfile[path_len] = '\0';
+ strcat(ptypesfile,
+ "src/modules/audio_coding/NetEQ/main/test/ptypes.txt");
+ ptypeFile = fopen(ptypesfile,"rt");
+ }
+ }
+ CHECK_NOT_NULL(ptypeFile);
+ printf("Ptypes file: %s\n\n", ptypesfile);
+
+ parsePtypeFile(ptypeFile, &decoders);
+ fclose(ptypeFile);
+
+ noOfCodecs = populateUsedCodec(&decoders, usedCodec);
+
+
+ /* read RTP file header */
+ if (!rtpOnly)
+ {
+ if (NETEQTEST_RTPpacket::skipFileHeader(in_file) != 0)
+ {
+ fprintf(stderr, "Wrong format in RTP file.\n");
+ return -1;
+ }
+ }
+
+ /* check payload type for first speech packet */
+ long tempFilePos = ftell(in_file);
+ enum stereoModes stereoMode = stereoModeMono;
+
+ if (!rtpOnly)
+ {
+ while (rtp.readFromFile(in_file) >= 0)
+ {
+ if (decoders.count(rtp.payloadType()) > 0
+ && decoders[rtp.payloadType()].codec != kDecoderRED
+ && decoders[rtp.payloadType()].codec != kDecoderAVT
+ && decoders[rtp.payloadType()].codec != kDecoderCNG )
+ {
+ stereoMode = decoders[rtp.payloadType()].stereo;
+ fs = decoders[rtp.payloadType()].fs;
+ break;
+ }
+ }
+ }
+ else
+ {
+ while (rtp.readFixedFromFile(in_file, packetLen) >= 0)
+ {
+ if (decoders.count(rtp.payloadType()) > 0
+ && decoders[rtp.payloadType()].codec != kDecoderRED
+ && decoders[rtp.payloadType()].codec != kDecoderAVT
+ && decoders[rtp.payloadType()].codec != kDecoderCNG )
+ {
+ stereoMode = decoders[rtp.payloadType()].stereo;
+ fs = decoders[rtp.payloadType()].fs;
+ break;
+ }
+ }
+ }
+
+ fseek(in_file, tempFilePos, SEEK_SET /* from beginning */);
+
+
+ /* block some payload types */
+ //rtp.blockPT(72);
+ //rtp.blockPT(23);
+
+ /* read first packet */
+ if (!rtpOnly)
+ {
+ rtp.readFromFile(in_file);
+ }
+ else
+ {
+ rtp.readFixedFromFile(in_file, packetLen);
+ rtp.setTime((1000 * rtp.timeStamp()) / fs);
+ }
+ if (!rtp)
+ {
+ printf("\nWarning: RTP file is empty\n\n");
+ }
+
+
+ /* Initialize NetEQ instances */
+ int numInst = 1;
+ if (stereoMode > stereoModeMono)
+ {
+ numInst = 2;
+ }
+
+ for (int i = 0; i < numInst; i++)
+ {
+ // create memory, allocate, initialize, and allocate packet buffer memory
+ NetEQvector.push_back (new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, static_cast<WebRtc_UWord16>(fs), kTCPLargeJitter));
+
+ createAndInsertDecoders (NetEQvector[i], &decoders, i /* channel */);
+
+ WebRtcNetEQ_SetAVTPlayout(NetEQvector[i]->instance(),1); // enable DTMF playout
+
+ WebRtcNetEQ_SetPlayoutMode(NetEQvector[i]->instance(), streamingMode);
+
+ NetEQvector[i]->usePreparseRTP(preParseRTP);
+
+ if (numInst > 1)
+ {
+ // we are using master/slave mode
+ if (i == 0)
+ {
+ // first instance is master
+ NetEQvector[i]->isMaster();
+ }
+ else
+ {
+ // all other are slaves
+ NetEQvector[i]->isSlave();
+ }
+ }
+ }
+
+
+#ifdef ZERO_TS_START
+ WebRtc_UWord32 firstTS = rtp.timeStamp();
+ rtp.setTimeStamp(0);
+#else
+ WebRtc_UWord32 firstTS = 0;
+#endif
+
+ // check stereo mode
+ if (stereoMode > stereoModeMono)
+ {
+ if(rtp.splitStereo(slaveRtp, stereoMode))
+ {
+ printf("Error in splitStereo\n");
+ }
+ }
+
+#ifdef PLAY_CLEAN
+ WebRtc_UWord32 prevTS = rtp.timeStamp();
+ WebRtc_UWord32 currTS, prev_time;
+#endif
+
+#ifdef JUNK_DATA
+ unsigned int random_seed = (unsigned int) /*1196764538; */time(NULL);
+ srand(random_seed);
+
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "%u\n", random_seed);
+ fclose(seedfile);
+ }
+#endif
+
+ WebRtc_UWord32 nextRecoutTime;
+ int lastRecout = getNextRecoutTime(recoutTimes, &nextRecoutTime); // does nothing if recoutTimes == NULL
+
+ if (recoutTimes)
+ simClock = (rtp.time() < nextRecoutTime ? rtp.time(): nextRecoutTime);
+ else
+ simClock = rtp.time(); // start immediately with first packet
+
+ WebRtc_UWord32 start_clock = simClock;
+
+ WebRtc_UWord32 nextExtraDelayTime;
+ int extraDelay = -1;
+ getNextExtraDelay(extraDelays, &nextExtraDelayTime, &extraDelay);
+
+ void *msInfo;
+ msInfo = malloc(WebRtcNetEQ_GetMasterSlaveInfoSize());
+ if(msInfo == NULL)
+ return(-1);
+
+ while(rtp.dataLen() >= 0 || (recoutTimes && !lastRecout)) {
+// printf("simClock = %Lu\n", simClock);
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CLOCK;
+ clock_float = (float) simClock;
+ fwrite(&temp_var,sizeof(int),1,delay_fid2);
+ fwrite(&clock_float, sizeof(float),1,delay_fid2);
+#endif
+ /* time to set extra delay */
+ if (extraDelay > -1 && simClock >= nextExtraDelayTime) {
+ // set extra delay for all instances
+ for (int i = 0; i < numInst; i++)
+ {
+ WebRtcNetEQ_SetExtraDelay(NetEQvector[i]->instance(), extraDelay);
+ }
+ getNextExtraDelay(extraDelays, &nextExtraDelayTime, &extraDelay);
+ }
+
+ /* check if time to receive */
+ while (simClock >= rtp.time() && rtp.dataLen() >= 0)
+ {
+ if (rtp.dataLen() > 0)
+ {
+
+ // insert main packet
+ NetEQvector[0]->recIn(rtp);
+
+ if (stereoMode > stereoModeMono
+ && slaveRtp.dataLen() > 0)
+ {
+ // insert slave packet
+ NetEQvector[1]->recIn(slaveRtp);
+ }
+
+ }
+
+ /* get next packet */
+#ifdef PLAY_CLEAN
+ prev_time = rtp.time();
+#endif
+ if (!rtpOnly)
+ {
+ rtp.readFromFile(in_file);
+ }
+ else
+ {
+ rtp.readFixedFromFile(in_file, packetLen);
+ rtp.setTime((1000 * rtp.timeStamp()) / fs);
+ }
+
+ if (rtp.dataLen() >= 0)
+ {
+ rtp.setTimeStamp(rtp.timeStamp() - firstTS);
+ }
+
+ packetCount++;
+
+ if (changeStereoMode(rtp, decoders, &stereoMode))
+ {
+ printf("Warning: stereo mode changed\n");
+ }
+
+ if (stereoMode > stereoModeMono)
+ {
+ if(rtp.splitStereo(slaveRtp, stereoMode))
+ {
+ printf("Error in splitStereo\n");
+ }
+ }
+
+#ifdef PLAY_CLEAN
+ currTS = rtp.timeStamp();
+ rtp.setTime(prev_time + (currTS-prevTS)/(fs/1000));
+ prevTS = currTS;
+#endif
+ }
+
+ /* check if time to RecOut */
+ if ( (!recoutTimes && (simClock%10)==0) // recout times not given from file
+ || ( recoutTimes && (simClock >= nextRecoutTime) ) ) // recout times given from file
+ {
+ if (stereoMode > stereoModeMono)
+ {
+ // stereo
+ WebRtc_Word16 tempLen;
+ tempLen = NetEQvector[0]->recOut( out_data, msInfo ); // master
+ outLen = NetEQvector[1]->recOut( &out_data[tempLen], msInfo ); // slave
+
+ assert(tempLen == outLen);
+
+ writeLen = outLen * 2;
+ stereoInterleave(out_data, writeLen);
+ }
+ else
+ {
+ // mono
+ outLen = NetEQvector[0]->recOut( out_data );
+ writeLen = outLen;
+ }
+
+ // write to file
+ fwrite(out_data,writeLen,2,out_file);
+ writtenSamples += writeLen;
+
+
+ lastRecout = getNextRecoutTime(recoutTimes, &nextRecoutTime); // does nothing if recoutTimes == NULL
+
+ /* ask for statistics */
+ WebRtcNetEQ_NetworkStatistics inCallStats;
+ WebRtcNetEQ_GetNetworkStatistics(NetEQvector[0]->instance(), &inCallStats);
+
+ }
+
+ /* increase time */
+ simClock+=TIME_STEP;
+ }
+
+ fclose(in_file);
+ fclose(out_file);
+
+#ifdef NETEQ_DELAY_LOGGING
+ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EOF;
+ fwrite(&temp_var,sizeof(int),1,delay_fid2);
+ fwrite(&tot_received_packets,sizeof(WebRtc_UWord32),1,delay_fid2);
+ fprintf(delay_fid2,"End of file\n");
+ fclose(delay_fid2);
+#endif
+
+ WebRtcNetEQ_GetRTCPStats(NetEQvector[0]->instance(), &RTCPstat);
+ printf("RTCP statistics:\n");
+ printf(" cum_lost : %d\n", (int) RTCPstat.cum_lost);
+ printf(" ext_max : %d\n", (int) RTCPstat.ext_max);
+ printf(" fraction_lost : %d (%f%%)\n", RTCPstat.fraction_lost, (float)(100.0*RTCPstat.fraction_lost/256.0));
+ printf(" jitter : %d\n", (int) RTCPstat.jitter);
+
+ printf("\n Call duration ms : %u\n", simClock-start_clock);
+
+ printf("\nComplexity estimates (including sub-components):\n");
+ printf(" RecIn complexity : %.2f MCPS\n", NetEQvector[0]->getRecInTime() / ((float) 1000*(simClock-start_clock)));
+ printf(" RecOut complexity : %.2f MCPS\n", NetEQvector[0]->getRecOutTime() / ((float) 1000*(simClock-start_clock)));
+
+ free_coders(decoders);
+ //free_coders(0 /* first channel */);
+ // if (stereoMode > stereoModeMono) {
+ // free_coders(1 /* second channel */);
+ // }
+ free(msInfo);
+
+ for (std::vector<NETEQTEST_NetEQClass *>::iterator it = NetEQvector.begin();
+ it < NetEQvector.end(); delete *it++);
+
+ printf("\nSimulation done!\n");
+
+#ifdef JUNK_DATA
+ if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+ fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+ }
+ else {
+ fprintf(seedfile, "ok\n\n");
+ fclose(seedfile);
+ }
+#endif
+
+
+ // Log complexity to file
+/* FILE *statfile;
+ statfile = fopen("complexity.txt","at");
+ fprintf(statfile,"%.4f, %.4f\n", (float) totTime_RecIn.QuadPart / ((float) 1000*(simClock-start_clock)), (float) totTime_RecOut.QuadPart / ((float) 1000*(simClock-start_clock)));
+ fclose(statfile);*/
+
+ return(0);
+
+}
+
+
+
+
+
+/****************/
+/* Subfunctions */
+/****************/
+
+bool splitStereo(NETEQTEST_RTPpacket& rtp, NETEQTEST_RTPpacket& rtpSlave,
+ const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
+ const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
+ bool *isStereo)
+{
+
+ // init
+ //bool isStereo = false;
+ enum stereoModes tempStereoMode = stereoModeMono;
+ bool isCng = false;
+
+ // check payload length
+ if (rtp.dataLen() <= 0) {
+ //*isStereo = false; // don't change
+ return(*isStereo);
+ }
+
+ // check payload type
+ WebRtc_Word16 ptype = rtp.payloadType();
+
+ // is this a cng payload?
+ for (int k = 0; k < noOfCngCodecs; k++) {
+ if (ptype == cngPtype[k]) {
+ // do not change stereo state
+ isCng = true;
+ tempStereoMode = stereoModeFrame;
+ }
+ }
+
+ if (!isCng)
+ {
+ *isStereo = false;
+
+ // is this payload type a stereo codec? which type?
+ for (int k = 0; k < noOfStereoCodecs; k++) {
+ if (ptype == stereoPtype[k]) {
+ tempStereoMode = stereoMode[k];
+ *isStereo = true;
+ break; // exit for loop
+ }
+ }
+ }
+
+ if (*isStereo)
+ {
+ // split the payload if stereo
+
+ if(rtp.splitStereo(rtpSlave, tempStereoMode))
+ {
+ printf("Error in splitStereo\n");
+ }
+
+ }
+
+ return(*isStereo);
+
+}
+
+void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen)
+{
+ int k;
+
+ for(k = totalLen/2; k < totalLen; k++) {
+ WebRtc_Word16 temp = data[k];
+ memmove(&data[2*k - totalLen + 2], &data[2*k - totalLen + 1], (totalLen - k -1) * sizeof(WebRtc_Word16));
+ data[2*k - totalLen + 1] = temp;
+ }
+}
+
+
+int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime) {
+
+ float tempTime;
+
+ if (!fp) {
+ return -1;
+ }
+
+ if (fread(&tempTime, sizeof(float), 1, fp) != 0) {
+ // not end of file
+ *nextTime = (WebRtc_UWord32) tempTime;
+ return 0;
+ }
+
+ *nextTime = 0;
+ fclose(fp);
+
+ return 1;
+}
+
+void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d) {
+
+ float temp[2];
+
+ if(!fp) {
+ *d = -1;
+ return;
+ }
+
+ if (fread(&temp, sizeof(float), 2, fp) != 0) {
+ // not end of file
+ *t = (WebRtc_UWord32) temp[0];
+ *d = (int) temp[1];
+ return;
+ }
+
+ *d = -1;
+ fclose(fp);
+
+ return;
+}
+
+
+void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders)
+{
+ int n, pt;
+ char codec[100];
+ decoderStruct tempDecoder;
+
+ // read first line
+ n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
+
+ while (n==2)
+ {
+ memset(&tempDecoder, 0, sizeof(decoderStruct));
+ tempDecoder.stereo = stereoModeMono;
+
+ if( pt >= 0 // < 0 disables this codec
+ && isalpha(codec[0]) ) // and is a letter
+ {
+
+ /* check for stereo */
+ int L = strlen(codec);
+ bool isStereo = false;
+
+ if (codec[L-1] == '*') {
+ // stereo codec
+ isStereo = true;
+
+ // remove '*'
+ codec[L-1] = '\0';
+ }
+
+#ifdef CODEC_G711
+ if(strcmp(codec, "pcmu") == 0) {
+ tempDecoder.codec = kDecoderPCMu;
+ tempDecoder.fs = 8000;
+ }
+ else if(strcmp(codec, "pcma") == 0) {
+ tempDecoder.codec = kDecoderPCMa;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_IPCMU
+ else if(strcmp(codec, "eg711u") == 0) {
+ tempDecoder.codec = kDecoderEG711u;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_IPCMA
+ else if(strcmp(codec, "eg711a") == 0) {
+ tempDecoder.codec = kDecoderEG711a;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_ILBC
+ else if(strcmp(codec, "ilbc") == 0) {
+ tempDecoder.codec = kDecoderILBC;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_ISAC
+ else if(strcmp(codec, "isac") == 0) {
+ tempDecoder.codec = kDecoderISAC;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_ISACLC
+ else if(strcmp(codec, "isaclc") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_ISACLC;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_ISAC_SWB
+ else if(strcmp(codec, "isacswb") == 0) {
+ tempDecoder.codec = kDecoderISACswb;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_IPCMWB
+ else if(strcmp(codec, "ipcmwb") == 0) {
+ tempDecoder.codec = kDecoderIPCMwb;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_G722
+ else if(strcmp(codec, "g722") == 0) {
+ tempDecoder.codec = kDecoderG722;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_G722_1_16
+ else if(strcmp(codec, "g722_1_16") == 0) {
+ tempDecoder.codec = kDecoderG722_1_16;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_G722_1_24
+ else if(strcmp(codec, "g722_1_24") == 0) {
+ tempDecoder.codec = kDecoderG722_1_24;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_G722_1_32
+ else if(strcmp(codec, "g722_1_32") == 0) {
+ tempDecoder.codec = kDecoderG722_1_32;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_G722_1C_24
+ else if(strcmp(codec, "g722_1c_24") == 0) {
+ tempDecoder.codec = kDecoderG722_1C_24;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_G722_1C_32
+ else if(strcmp(codec, "g722_1c_32") == 0) {
+ tempDecoder.codec = kDecoderG722_1C_32;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_G722_1C_48
+ else if(strcmp(codec, "g722_1c_48") == 0) {
+ tempDecoder.codec = kDecoderG722_1C_48;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_G723
+ else if(strcmp(codec, "g723") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_G723;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_G726
+ else if(strcmp(codec, "g726_16") == 0) {
+ tempDecoder.codec = kDecoderG726_16;
+ tempDecoder.fs = 8000;
+ }
+ else if(strcmp(codec, "g726_24") == 0) {
+ tempDecoder.codec = kDecoderG726_24;
+ tempDecoder.fs = 8000;
+ }
+ else if(strcmp(codec, "g726_32") == 0) {
+ tempDecoder.codec = kDecoderG726_32;
+ tempDecoder.fs = 8000;
+ }
+ else if(strcmp(codec, "g726_40") == 0) {
+ tempDecoder.codec = kDecoderG726_40;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_G729
+ else if(strcmp(codec, "g729") == 0) {
+ tempDecoder.codec = kDecoderG729;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_G729D
+ else if(strcmp(codec, "g729d") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_G729D;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_G729_1
+ else if(strcmp(codec, "g729_1") == 0) {
+ tempDecoder.codec = kDecoderG729_1;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_GSMFR
+ else if(strcmp(codec, "gsmfr") == 0) {
+ tempDecoder.codec = kDecoderGSMFR;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_GSMEFR
+ else if(strcmp(codec, "gsmefr") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_GSMEFR;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_AMR
+ else if(strcmp(codec, "amr") == 0) {
+ tempDecoder.codec = kDecoderAMR;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_AMRWB
+ else if(strcmp(codec, "amrwb") == 0) {
+ tempDecoder.codec = kDecoderAMRWB;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_DVI4
+ else if(strcmp(codec, "dvi4") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_DVI4;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_SPEEX_8
+ else if(strcmp(codec, "speex8") == 0) {
+ tempDecoder.codec = kDecoderSPEEX_8;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_SPEEX_16
+ else if(strcmp(codec, "speex16") == 0) {
+ tempDecoder.codec = kDecoderSPEEX_16;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_CELT_32
+ else if(strcmp(codec, "celt32") == 0) {
+ tempDecoder.codec = kDecoderCELT_32;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_SILK_NB
+ else if(strcmp(codec, "silk8") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_SILK_8;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_SILK_WB
+ else if(strcmp(codec, "silk12") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_SILK_12;
+ tempDecoder.fs = 16000;
+ }
+ else if(strcmp(codec, "silk16") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_SILK_16;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_SILK_SWB
+ else if(strcmp(codec, "silk24") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_SILK_24;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_MELPE
+ else if(strcmp(codec, "melpe") == 0) {
+ tempDecoder.codec = NETEQ_CODEC_MELPE;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_PCM16B
+ else if(strcmp(codec, "pcm16b") == 0) {
+ tempDecoder.codec = kDecoderPCM16B;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_PCM16B_WB
+ else if(strcmp(codec, "pcm16b_wb") == 0) {
+ tempDecoder.codec = kDecoderPCM16Bwb;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ else if(strcmp(codec, "pcm16b_swb32khz") == 0) {
+ tempDecoder.codec = kDecoderPCM16Bswb32kHz;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ else if(strcmp(codec, "pcm16b_swb48khz") == 0) {
+ tempDecoder.codec = kDecoderPCM16Bswb48kHz;
+ tempDecoder.fs = 48000;
+ }
+#endif
+#ifdef CODEC_CNGCODEC8
+ else if(strcmp(codec, "cn") == 0) {
+ tempDecoder.codec = kDecoderCNG;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_CNGCODEC16
+ else if(strcmp(codec, "cn_wb") == 0) {
+ tempDecoder.codec = kDecoderCNG;
+ tempDecoder.fs = 16000;
+ }
+#endif
+#ifdef CODEC_CNGCODEC32
+ else if(strcmp(codec, "cn_swb32") == 0) {
+ tempDecoder.codec = kDecoderCNG;
+ tempDecoder.fs = 32000;
+ }
+#endif
+#ifdef CODEC_CNGCODEC48
+ else if(strcmp(codec, "cn_swb48") == 0) {
+ tempDecoder.codec = kDecoderCNG;
+ tempDecoder.fs = 48000;
+ }
+#endif
+#ifdef CODEC_ATEVENT_DECODE
+ else if(strcmp(codec, "avt") == 0) {
+ tempDecoder.codec = kDecoderAVT;
+ tempDecoder.fs = 8000;
+ }
+#endif
+#ifdef CODEC_RED
+ else if(strcmp(codec, "red") == 0) {
+ tempDecoder.codec = kDecoderRED;
+ tempDecoder.fs = 8000;
+ }
+#endif
+ else if(isalpha(codec[0])) {
+ printf("Unsupported codec %s\n", codec);
+ // read next line and continue while loop
+ n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
+ continue;
+ }
+ else {
+ // name is not recognized, and does not start with a letter
+ // hence, it is commented out
+ // read next line and continue while loop
+ n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
+ continue;
+ }
+
+ // handle stereo
+ if (tempDecoder.codec == kDecoderCNG)
+ {
+ // always set stereo mode for CNG, even if it is not marked at stereo
+ tempDecoder.stereo = stereoModeFrame;
+ }
+ else if(isStereo)
+ {
+ switch(tempDecoder.codec) {
+ // sample based codecs
+ case kDecoderPCMu:
+ case kDecoderPCMa:
+ case kDecoderG722:
+ {
+ // 1 octet per sample
+ tempDecoder.stereo = stereoModeSample1;
+ break;
+ }
+ case kDecoderPCM16B:
+ case kDecoderPCM16Bwb:
+ case kDecoderPCM16Bswb32kHz:
+ case kDecoderPCM16Bswb48kHz:
+ {
+ // 2 octets per sample
+ tempDecoder.stereo = stereoModeSample2;
+ break;
+ }
+
+ // fixed-rate frame codecs
+// case kDecoderG729:
+// case NETEQ_CODEC_G729D:
+// case NETEQ_CODEC_G729E:
+// case kDecoderG722_1_16:
+// case kDecoderG722_1_24:
+// case kDecoderG722_1_32:
+// case kDecoderG722_1C_24:
+// case kDecoderG722_1C_32:
+// case kDecoderG722_1C_48:
+// case NETEQ_CODEC_MELPE:
+// {
+// tempDecoder.stereo = stereoModeFrame;
+// break;
+// }
+ default:
+ {
+ printf("Cannot use codec %s as stereo codec\n", codec);
+ exit(0);
+ }
+ }
+ }
+
+ if (pt > 127)
+ {
+ printf("Payload type must be less than 128\n");
+ exit(0);
+ }
+
+ // insert into codecs map
+ (*decoders)[static_cast<WebRtc_UWord8>(pt)] = tempDecoder;
+
+ }
+
+ n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
+ } // end while
+
+}
+
+
+bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode)
+{
+ if (decoders.count(rtp.payloadType()) > 0
+ && decoders[rtp.payloadType()].codec != kDecoderRED
+ && decoders[rtp.payloadType()].codec != kDecoderAVT
+ && decoders[rtp.payloadType()].codec != kDecoderCNG )
+ {
+ if (decoders[rtp.payloadType()].stereo != *stereoMode)
+ {
+ *stereoMode = decoders[rtp.payloadType()].stereo;
+ return true; // stereo mode did change
+ }
+ }
+
+ return false; // stereo mode did not change
+}
+
+
+int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec)
+{
+ int numCodecs = 0;
+
+ std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+
+ it = decoders->begin();
+
+ for (int i = 0; i < static_cast<int>(decoders->size()); i++, it++)
+ {
+ usedCodec[numCodecs] = (*it).second.codec;
+ numCodecs++;
+ }
+
+ return numCodecs;
+}
+
+
+void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber)
+{
+ std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+
+ for (it = decoders->begin(); it != decoders->end(); it++)
+ {
+ if (channelNumber == 0 ||
+ ((*it).second.stereo > stereoModeMono ))
+ {
+ // create decoder instance
+ WebRtc_UWord8 pt = static_cast<WebRtc_UWord8>( (*it).first );
+ NETEQTEST_Decoder **dec = &((*it).second.decoder[channelNumber]);
+ enum WebRtcNetEQDecoder type = (*it).second.codec;
+
+ switch (type)
+ {
+#ifdef CODEC_G711
+ case kDecoderPCMu:
+ *dec = new decoder_PCMU( pt );
+ break;
+ case kDecoderPCMa:
+ *dec = new decoder_PCMA( pt );
+ break;
+#endif
+#ifdef CODEC_IPCMU
+ case kDecoderEG711u:
+ *dec = new decoder_IPCMU( pt );
+ break;
+#endif
+#ifdef CODEC_IPCMA
+ case kDecoderEG711a:
+ *dec = new decoder_IPCMA( pt );
+ break;
+#endif
+#ifdef CODEC_IPCMWB
+ case kDecoderIPCMwb:
+ *dec = new decoder_IPCMWB( pt );
+ break;
+#endif
+#ifdef CODEC_ILBC
+ case kDecoderILBC:
+ *dec = new decoder_ILBC( pt );
+ break;
+#endif
+#ifdef CODEC_ISAC
+ case kDecoderISAC:
+ *dec = new decoder_iSAC( pt );
+ break;
+#endif
+#ifdef CODEC_ISAC_SWB
+ case kDecoderISACswb:
+ *dec = new decoder_iSACSWB( pt );
+ break;
+#endif
+#ifdef CODEC_G729
+ case kDecoderG729:
+ *dec = new decoder_G729( pt );
+ break;
+ case NETEQ_CODEC_G729D:
+ printf("Error: G729D not supported\n");
+ break;
+#endif
+#ifdef CODEC_G729E
+ case NETEQ_CODEC_G729E:
+ *dec = new decoder_G729E( pt );
+ break;
+#endif
+#ifdef CODEC_G729_1
+ case kDecoderG729_1:
+ *dec = new decoder_G729_1( pt );
+ break;
+#endif
+#ifdef CODEC_G723
+ case NETEQ_CODEC_G723:
+ *dec = new decoder_G723( pt );
+ break;
+#endif
+#ifdef CODEC_PCM16B
+ case kDecoderPCM16B:
+ *dec = new decoder_PCM16B_NB( pt );
+ break;
+#endif
+#ifdef CODEC_PCM16B_WB
+ case kDecoderPCM16Bwb:
+ *dec = new decoder_PCM16B_WB( pt );
+ break;
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ case kDecoderPCM16Bswb32kHz:
+ *dec = new decoder_PCM16B_SWB32( pt );
+ break;
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ case kDecoderPCM16Bswb48kHz:
+ *dec = new decoder_PCM16B_SWB48( pt );
+ break;
+#endif
+#ifdef CODEC_DVI4
+ case NETEQ_CODEC_DVI4:
+ *dec = new decoder_DVI4( pt );
+ break;
+#endif
+#ifdef CODEC_G722
+ case kDecoderG722:
+ *dec = new decoder_G722( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1_16
+ case kDecoderG722_1_16:
+ *dec = new decoder_G722_1_16( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1_24
+ case kDecoderG722_1_24:
+ *dec = new decoder_G722_1_24( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1_32
+ case kDecoderG722_1_32:
+ *dec = new decoder_G722_1_32( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1C_24
+ case kDecoderG722_1C_24:
+ *dec = new decoder_G722_1C_24( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1C_32
+ case kDecoderG722_1C_32:
+ *dec = new decoder_G722_1C_32( pt );
+ break;
+#endif
+#ifdef CODEC_G722_1C_48
+ case kDecoderG722_1C_48:
+ *dec = new decoder_G722_1C_48( pt );
+ break;
+#endif
+#ifdef CODEC_AMR
+ case kDecoderAMR:
+ *dec = new decoder_AMR( pt );
+ break;
+#endif
+#ifdef CODEC_AMRWB
+ case kDecoderAMRWB:
+ *dec = new decoder_AMRWB( pt );
+ break;
+#endif
+#ifdef CODEC_GSMFR
+ case kDecoderGSMFR:
+ *dec = new decoder_GSMFR( pt );
+ break;
+#endif
+#ifdef CODEC_GSMEFR
+ case NETEQ_CODEC_GSMEFR:
+ *dec = new decoder_GSMEFR( pt );
+ break;
+#endif
+#ifdef CODEC_G726
+ case kDecoderG726_16:
+ *dec = new decoder_G726_16( pt );
+ break;
+ case kDecoderG726_24:
+ *dec = new decoder_G726_24( pt );
+ break;
+ case kDecoderG726_32:
+ *dec = new decoder_G726_32( pt );
+ break;
+ case kDecoderG726_40:
+ *dec = new decoder_G726_40( pt );
+ break;
+#endif
+#ifdef CODEC_MELPE
+ case NETEQ_CODEC_MELPE:
+#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
+ *dec = new decoder_MELPE( pt );
+#endif
+ break;
+#endif
+#ifdef CODEC_SPEEX_8
+ case kDecoderSPEEX_8:
+ *dec = new decoder_SPEEX( pt, 8000 );
+ break;
+#endif
+#ifdef CODEC_SPEEX_16
+ case kDecoderSPEEX_16:
+ *dec = new decoder_SPEEX( pt, 16000 );
+ break;
+#endif
+#ifdef CODEC_CELT_32
+ case kDecoderCELT_32:
+ *dec = new decoder_CELT( pt, 32000 );
+ break;
+#endif
+#ifdef CODEC_RED
+ case kDecoderRED:
+ *dec = new decoder_RED( pt );
+ break;
+#endif
+#ifdef CODEC_ATEVENT_DECODE
+ case kDecoderAVT:
+ *dec = new decoder_AVT( pt );
+ break;
+#endif
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ case kDecoderCNG:
+ *dec = new decoder_CNG( pt, static_cast<WebRtc_UWord16>((*it).second.fs) );
+ break;
+#endif
+#ifdef CODEC_ISACLC
+ case NETEQ_CODEC_ISACLC:
+ *dec = new decoder_iSACLC( pt );
+ break;
+#endif
+#ifdef CODEC_SILK_NB
+ case NETEQ_CODEC_SILK_8:
+#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
+ *dec = new decoder_SILK8( pt );
+#endif
+ break;
+#endif
+#ifdef CODEC_SILK_WB
+ case NETEQ_CODEC_SILK_12:
+#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
+ *dec = new decoder_SILK12( pt );
+#endif
+ break;
+#endif
+#ifdef CODEC_SILK_WB
+ case NETEQ_CODEC_SILK_16:
+#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
+ *dec = new decoder_SILK16( pt );
+#endif
+ break;
+#endif
+#ifdef CODEC_SILK_SWB
+ case NETEQ_CODEC_SILK_24:
+#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
+ *dec = new decoder_SILK24( pt );
+#endif
+ break;
+#endif
+
+ default:
+ printf("Unknown codec type encountered in createAndInsertDecoders\n");
+ exit(0);
+ }
+
+ // insert into codec DB
+ if (*dec)
+ {
+ (*dec)->loadToNetEQ(*neteq);
+ }
+ }
+ }
+
+}
+
+
+void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders)
+{
+ std::map<WebRtc_UWord8, decoderStruct>::iterator it;
+
+ for (it = decoders.begin(); it != decoders.end(); it++)
+ {
+ if ((*it).second.decoder[0])
+ {
+ delete (*it).second.decoder[0];
+ }
+
+ if ((*it).second.decoder[1])
+ {
+ delete (*it).second.decoder[1];
+ }
+ }
+}
+
+
+
+#include "pcm16b.h"
+#include "g711_interface.h"
+#include "isac.h"
+
+int doAPItest() {
+
+ char version[20];
+ void *inst;
+ enum WebRtcNetEQDecoder usedCodec;
+ int NetEqBufferMaxPackets, BufferSizeInBytes;
+ WebRtcNetEQ_CodecDef codecInst;
+ WebRtcNetEQ_RTCPStat RTCPstat;
+ WebRtc_UWord32 timestamp;
+ int memorySize;
+ int ok;
+
+ printf("API-test:\n");
+
+ /* get the version string */
+ WebRtcNetEQ_GetVersion(version);
+ printf("NetEq version: %s\n\n", version);
+
+ /* test that API functions return -1 if instance is NULL */
+#define CHECK_MINUS_ONE(x) {int errCode = x; if((errCode)!=-1){printf("\n API test failed at line %d: %s. Function did not return -1 as expected\n",__LINE__,#x); return(-1);}}
+//#define RESET_ERROR(x) ((MainInst_t*) x)->ErrorCode = 0;
+ inst = NULL;
+
+ CHECK_MINUS_ONE(WebRtcNetEQ_GetErrorCode(inst))
+ CHECK_MINUS_ONE(WebRtcNetEQ_Assign(&inst, NULL))
+// printf("WARNING: Test of WebRtcNetEQ_Assign() is disabled due to a bug.\n");
+ usedCodec=kDecoderPCMu;
+ CHECK_MINUS_ONE(WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter, &NetEqBufferMaxPackets, &BufferSizeInBytes))
+ CHECK_MINUS_ONE(WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, BufferSizeInBytes))
+
+ CHECK_MINUS_ONE(WebRtcNetEQ_Init(inst, 8000))
+ CHECK_MINUS_ONE(WebRtcNetEQ_SetAVTPlayout(inst, 0))
+ CHECK_MINUS_ONE(WebRtcNetEQ_SetExtraDelay(inst, 17))
+ CHECK_MINUS_ONE(WebRtcNetEQ_SetPlayoutMode(inst, kPlayoutOn))
+
+ CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbReset(inst))
+ CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
+ CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbRemove(inst, usedCodec))
+ WebRtc_Word16 temp1, temp2;
+ CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetSizeInfo(inst, &temp1, &temp2))
+ CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetCodecInfo(inst, 0, &usedCodec))
+
+ CHECK_MINUS_ONE(WebRtcNetEQ_RecIn(inst, &temp1, 17, 4711))
+ CHECK_MINUS_ONE(WebRtcNetEQ_RecOut(inst, &temp1, &temp2))
+ CHECK_MINUS_ONE(WebRtcNetEQ_GetRTCPStats(inst, &RTCPstat)); // error here!!!
+ CHECK_MINUS_ONE(WebRtcNetEQ_GetSpeechTimeStamp(inst, ×tamp))
+ WebRtcNetEQOutputType temptype;
+ CHECK_MINUS_ONE(WebRtcNetEQ_GetSpeechOutputType(inst, &temptype))
+
+ WebRtc_UWord8 tempFlags;
+ WebRtc_UWord16 utemp1, utemp2;
+ CHECK_MINUS_ONE(WebRtcNetEQ_VQmonRecOutStatistics(inst, &utemp1, &utemp2, &tempFlags))
+ CHECK_MINUS_ONE(WebRtcNetEQ_VQmonGetRxStatistics(inst, &utemp1, &utemp2))
+
+ WebRtcNetEQ_AssignSize(&memorySize);
+ CHECK_ZERO(WebRtcNetEQ_Assign(&inst, malloc(memorySize)))
+
+ /* init with wrong sample frequency */
+ CHECK_MINUS_ONE(WebRtcNetEQ_Init(inst, 17))
+
+ /* init with correct fs */
+ CHECK_ZERO(WebRtcNetEQ_Init(inst, 8000))
+
+ /* GetRecommendedBufferSize with wrong codec */
+ usedCodec=kDecoderReservedStart;
+ ok = WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter , &NetEqBufferMaxPackets, &BufferSizeInBytes);
+ if((ok!=-1) || ((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNKNOWN_CODEC))){
+ printf("WebRtcNetEQ_GetRecommendedBufferSize() did not return proper error code for wrong codec.\n");
+ printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));
+ }
+ //RESET_ERROR(inst)
+
+ /* GetRecommendedBufferSize with wrong network type */
+ usedCodec = kDecoderPCMu;
+ ok=WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, (enum WebRtcNetEQNetworkType) 4711 , &NetEqBufferMaxPackets, &BufferSizeInBytes);
+ if ((ok!=-1) || ((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_NETWORK_TYPE))) {
+ printf("WebRtcNetEQ_GetRecommendedBufferSize() did not return proper error code for wrong network type.\n");
+ printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));
+ //RESET_ERROR(inst)
+ }
+ CHECK_ZERO(WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter , &NetEqBufferMaxPackets, &BufferSizeInBytes))
+
+ /* try to do RecIn before assigning the packet buffer */
+/* makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, 17,4711, 1235412312);
+ makeDTMFpayload(&rtp_data[12], 1, 1, 10, 100);
+ ok = WebRtcNetEQ_RecIn(inst, (short *) rtp_data, 12+4, 4711);
+ printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));*/
+
+ /* check all limits of WebRtcNetEQ_AssignBuffer */
+ ok=WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, 149<<1);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
+ printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong sizeinbytes\n");
+ }
+ ok=WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NULL, BufferSizeInBytes);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
+ printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for NULL memory pointer\n");
+ }
+ ok=WebRtcNetEQ_AssignBuffer(inst, 1, NetEqPacketBuffer, BufferSizeInBytes);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
+ printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong MaxNoOfPackets\n");
+ }
+ ok=WebRtcNetEQ_AssignBuffer(inst, 601, NetEqPacketBuffer, BufferSizeInBytes);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
+ printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong MaxNoOfPackets\n");
+ }
+
+ /* do correct assignbuffer */
+ CHECK_ZERO(WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, BufferSizeInBytes))
+
+ ok=WebRtcNetEQ_SetExtraDelay(inst, -1);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_DELAYVALUE))) {
+ printf("WebRtcNetEQ_SetExtraDelay() did not return proper error code for too small delay\n");
+ }
+ ok=WebRtcNetEQ_SetExtraDelay(inst, 1001);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_DELAYVALUE))) {
+ printf("WebRtcNetEQ_SetExtraDelay() did not return proper error code for too large delay\n");
+ }
+
+ ok=WebRtcNetEQ_SetPlayoutMode(inst,(enum WebRtcNetEQPlayoutMode) 4711);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_PLAYOUTMODE))) {
+ printf("WebRtcNetEQ_SetPlayoutMode() did not return proper error code for wrong mode\n");
+ }
+
+ /* number of codecs should return zero before adding any codecs */
+ WebRtcNetEQ_CodecDbGetSizeInfo(inst, &temp1, &temp2);
+ if(temp1!=0)
+ printf("WebRtcNetEQ_CodecDbGetSizeInfo() return non-zero number of codecs in DB before adding any codecs\n");
+
+ /* get info from empty database */
+ ok=WebRtcNetEQ_CodecDbGetCodecInfo(inst, 17, &usedCodec);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_NOT_EXIST1))) {
+ printf("WebRtcNetEQ_CodecDbGetCodecInfo() did not return proper error code for out-of-range entry number\n");
+ }
+
+ /* remove codec from empty database */
+ ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderPCMa);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_NOT_EXIST4))) {
+ printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that has not been added\n");
+ }
+
+ /* add codec with unsupported fs */
+#ifdef CODEC_PCM16B
+#ifndef NETEQ_48KHZ_WIDEBAND
+ SET_CODEC_PAR(codecInst,kDecoderPCM16Bswb48kHz,77,NULL,48000);
+ SET_PCM16B_SWB48_FUNCTIONS(codecInst);
+ ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_FS))) {
+ printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding codec with unsupported sample freq\n");
+ }
+#else
+ printf("Could not test adding codec with unsupported sample frequency since NetEQ is compiled with 48kHz support.\n");
+#endif
+#else
+ printf("Could not test adding codec with unsupported sample frequency since NetEQ is compiled without PCM16B support.\n");
+#endif
+
+ /* add two codecs with identical payload types */
+ SET_CODEC_PAR(codecInst,kDecoderPCMa,17,NULL,8000);
+ SET_PCMA_FUNCTIONS(codecInst);
+ CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
+
+ SET_CODEC_PAR(codecInst,kDecoderPCMu,17,NULL,8000);
+ SET_PCMU_FUNCTIONS(codecInst);
+ ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
+ printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding two codecs with identical payload types\n");
+ }
+
+ /* try adding several payload types for CNG codecs */
+ SET_CODEC_PAR(codecInst,kDecoderCNG,105,NULL,16000);
+ SET_CNG_FUNCTIONS(codecInst);
+ CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
+ SET_CODEC_PAR(codecInst,kDecoderCNG,13,NULL,8000);
+ SET_CNG_FUNCTIONS(codecInst);
+ CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
+
+ /* try adding a speech codec over a CNG codec */
+ SET_CODEC_PAR(codecInst,kDecoderISAC,105,NULL,16000); /* same as WB CNG above */
+ SET_ISAC_FUNCTIONS(codecInst);
+ ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
+ printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding a speech codec over a CNG codec\n");
+ }
+
+ /* try adding a CNG codec over a speech codec */
+ SET_CODEC_PAR(codecInst,kDecoderCNG,17,NULL,32000); /* same as PCMU above */
+ SET_CNG_FUNCTIONS(codecInst);
+ ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
+ printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding a speech codec over a CNG codec\n");
+ }
+
+
+ /* remove codec out of range */
+ ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderReservedStart);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_CODEC))) {
+ printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that is out of range\n");
+ }
+ ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderReservedEnd);
+ if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_CODEC))) {
+ printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that is out of range\n");
+ }
+
+ /*SET_CODEC_PAR(codecInst,kDecoderEG711a,NETEQ_CODEC_EG711A_PT,NetEqiPCMAState,8000);
+ SET_IPCMA_FUNCTIONS(codecInst);
+ CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
+*/
+ free(inst);
+
+ return(0);
+
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h b/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h
new file mode 100644
index 0000000..f6cc3da
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* PayloadTypes.h */
+/* Used by NetEqRTPplay application */
+
+/* RTP defined codepoints */
+#define NETEQ_CODEC_PCMU_PT 0
+#define NETEQ_CODEC_GSMFR_PT 3
+#define NETEQ_CODEC_G723_PT 4
+#define NETEQ_CODEC_DVI4_PT 125 // 8 kHz version
+//#define NETEQ_CODEC_DVI4_16_PT 6 // 16 kHz version
+#define NETEQ_CODEC_PCMA_PT 8
+#define NETEQ_CODEC_G722_PT 9
+#define NETEQ_CODEC_CN_PT 13
+//#define NETEQ_CODEC_G728_PT 15
+//#define NETEQ_CODEC_DVI4_11_PT 16 // 11.025 kHz version
+//#define NETEQ_CODEC_DVI4_22_PT 17 // 22.050 kHz version
+#define NETEQ_CODEC_G729_PT 18
+
+/* Dynamic RTP codepoints as defined in VoiceEngine (file VEAPI.cpp) */
+#define NETEQ_CODEC_IPCMWB_PT 97
+#define NETEQ_CODEC_SPEEX8_PT 98
+#define NETEQ_CODEC_SPEEX16_PT 99
+#define NETEQ_CODEC_EG711U_PT 100
+#define NETEQ_CODEC_EG711A_PT 101
+#define NETEQ_CODEC_ILBC_PT 102
+#define NETEQ_CODEC_ISAC_PT 103
+#define NETEQ_CODEC_ISACLC_PT 119
+#define NETEQ_CODEC_ISACSWB_PT 104
+#define NETEQ_CODEC_AVT_PT 106
+#define NETEQ_CODEC_G722_1_16_PT 108
+#define NETEQ_CODEC_G722_1_24_PT 109
+#define NETEQ_CODEC_G722_1_32_PT 110
+#define NETEQ_CODEC_SC3_PT 111
+#define NETEQ_CODEC_AMR_PT 112
+#define NETEQ_CODEC_GSMEFR_PT 113
+//#define NETEQ_CODEC_ILBCRCU_PT 114
+#define NETEQ_CODEC_G726_16_PT 115
+#define NETEQ_CODEC_G726_24_PT 116
+#define NETEQ_CODEC_G726_32_PT 121
+#define NETEQ_CODEC_RED_PT 117
+#define NETEQ_CODEC_G726_40_PT 118
+//#define NETEQ_CODEC_ENERGY_PT 120
+#define NETEQ_CODEC_CN_WB_PT 105
+#define NETEQ_CODEC_CN_SWB_PT 126
+#define NETEQ_CODEC_G729_1_PT 107
+#define NETEQ_CODEC_G729D_PT 123
+#define NETEQ_CODEC_MELPE_PT 124
+#define NETEQ_CODEC_CELT32_PT 114
+
+/* Extra dynamic codepoints */
+#define NETEQ_CODEC_AMRWB_PT 120
+#define NETEQ_CODEC_PCM16B_PT 93
+#define NETEQ_CODEC_PCM16B_WB_PT 94
+#define NETEQ_CODEC_PCM16B_SWB32KHZ_PT 95
+#define NETEQ_CODEC_PCM16B_SWB48KHZ_PT 96
+#define NETEQ_CODEC_MPEG4AAC_PT 122
+
+
+/* Not default in VoiceEngine */
+#define NETEQ_CODEC_G722_1C_24_PT 84
+#define NETEQ_CODEC_G722_1C_32_PT 85
+#define NETEQ_CODEC_G722_1C_48_PT 86
+
+#define NETEQ_CODEC_SILK_8_PT 80
+#define NETEQ_CODEC_SILK_12_PT 81
+#define NETEQ_CODEC_SILK_16_PT 82
+#define NETEQ_CODEC_SILK_24_PT 83
+
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc b/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc
new file mode 100644
index 0000000..12617dd
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <stdio.h>
+#include <vector>
+
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+
+enum {
+ kRedPayloadType = 127
+};
+
+int main(int argc, char* argv[]) {
+ FILE* in_file = fopen(argv[1], "rb");
+ if (!in_file) {
+ printf("Cannot open input file %s\n", argv[1]);
+ return -1;
+ }
+ printf("Input file: %s\n", argv[1]);
+
+ FILE* out_file = fopen(argv[2], "wt");
+ if (!out_file) {
+ printf("Cannot open output file %s\n", argv[2]);
+ return -1;
+ }
+ printf("Output file: %s\n\n", argv[2]);
+
+ // Print file header.
+ fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M\n");
+
+ // Read file header.
+ NETEQTEST_RTPpacket::skipFileHeader(in_file);
+ NETEQTEST_RTPpacket packet;
+
+ while (packet.readFromFile(in_file) >= 0) {
+ // Write packet data to file.
+ fprintf(out_file, "%5u %10u %10u %5i %5i %2i\n",
+ packet.sequenceNumber(), packet.timeStamp(), packet.time(),
+ packet.dataLen(), packet.payloadType(), packet.markerBit());
+ if (packet.payloadType() == kRedPayloadType) {
+ WebRtcNetEQ_RTPInfo red_header;
+ int len;
+ int red_index = 0;
+ while ((len = packet.extractRED(red_index++, red_header)) >= 0) {
+ fprintf(out_file, "* %5u %10u %10u %5i %5i\n",
+ red_header.sequenceNumber, red_header.timeStamp,
+ packet.time(), len, red_header.payloadType);
+ }
+ assert(red_index > 1); // We must get at least one payload.
+ }
+ }
+
+ fclose(in_file);
+ fclose(out_file);
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc b/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc
new file mode 100644
index 0000000..001b00b
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "gtest/gtest.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+
+#define FIRSTLINELEN 40
+
+int main(int argc, char* argv[]) {
+ if (argc < 3) {
+ printf("Usage: RTPcat in1.rtp int2.rtp [...] out.rtp\n");
+ exit(1);
+ }
+
+ FILE* in_file = fopen(argv[1], "rb");
+ if (!in_file) {
+ printf("Cannot open input file %s\n", argv[1]);
+ return -1;
+ }
+
+ FILE* out_file = fopen(argv[argc - 1], "wb"); // Last parameter is out file.
+ if (!out_file) {
+ printf("Cannot open output file %s\n", argv[argc - 1]);
+ return -1;
+ }
+ printf("Output RTP file: %s\n\n", argv[argc - 1]);
+
+ // Read file header and write directly to output file.
+ char firstline[FIRSTLINELEN];
+ const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+ EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
+ EXPECT_GT(fputs(firstline, out_file), 0);
+ EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
+ in_file));
+ EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
+ out_file));
+
+ // Close input file and re-open it later (easier to write the loop below).
+ fclose(in_file);
+
+ for (int i = 1; i < argc - 1; i++) {
+ in_file = fopen(argv[i], "rb");
+ if (!in_file) {
+ printf("Cannot open input file %s\n", argv[i]);
+ return -1;
+ }
+ printf("Input RTP file: %s\n", argv[i]);
+
+ NETEQTEST_RTPpacket::skipFileHeader(in_file);
+ NETEQTEST_RTPpacket packet;
+ int pack_len = packet.readFromFile(in_file);
+ if (pack_len < 0) {
+ exit(1);
+ }
+ while (pack_len >= 0) {
+ packet.writeToFile(out_file);
+ pack_len = packet.readFromFile(in_file);
+ }
+ fclose(in_file);
+ }
+ fclose(out_file);
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc b/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc
new file mode 100644
index 0000000..ecbd81c
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "gtest/gtest.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+
+#define FIRSTLINELEN 40
+
+static bool pktCmp(NETEQTEST_RTPpacket *a, NETEQTEST_RTPpacket *b) {
+ return (a->time() < b->time());
+}
+
+int main(int argc, char* argv[]) {
+ FILE* in_file = fopen(argv[1], "rb");
+ if (!in_file) {
+ printf("Cannot open input file %s\n", argv[1]);
+ return -1;
+ }
+ printf("Input RTP file: %s\n", argv[1]);
+
+ FILE* stat_file = fopen(argv[2], "rt");
+ if (!stat_file) {
+ printf("Cannot open timing file %s\n", argv[2]);
+ return -1;
+ }
+ printf("Timing file: %s\n", argv[2]);
+
+ FILE* out_file = fopen(argv[3], "wb");
+ if (!out_file) {
+ printf("Cannot open output file %s\n", argv[3]);
+ return -1;
+ }
+ printf("Output RTP file: %s\n\n", argv[3]);
+
+ // Read all statistics and insert into map.
+ // Read first line.
+ char temp_str[100];
+ if (fgets(temp_str, 100, stat_file) == NULL) {
+ printf("Failed to read timing file %s\n", argv[2]);
+ return -1;
+ }
+ // Define map.
+ std::map<std::pair<uint16_t, uint32_t>, uint32_t> packet_stats;
+ uint16_t seq_no;
+ uint32_t ts;
+ uint32_t send_time;
+
+ while (fscanf(stat_file,
+ "%hu %u %u %*i %*i\n", &seq_no, &ts, &send_time) == 3) {
+ std::pair<uint16_t, uint32_t>
+ temp_pair = std::pair<uint16_t, uint32_t>(seq_no, ts);
+
+ packet_stats[temp_pair] = send_time;
+ }
+
+ fclose(stat_file);
+
+ // Read file header and write directly to output file.
+ char first_line[FIRSTLINELEN];
+ if (fgets(first_line, FIRSTLINELEN, in_file) == NULL) {
+ printf("Failed to read first line of input file %s\n", argv[1]);
+ return -1;
+ }
+ fputs(first_line, out_file);
+ // start_sec + start_usec + source + port + padding
+ const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+ if (fread(first_line, 1, kRtpDumpHeaderSize, in_file)
+ != kRtpDumpHeaderSize) {
+ printf("Failed to read RTP dump header from input file %s\n", argv[1]);
+ return -1;
+ }
+ if (fwrite(first_line, 1, kRtpDumpHeaderSize, out_file)
+ != kRtpDumpHeaderSize) {
+ printf("Failed to write RTP dump header to output file %s\n", argv[3]);
+ return -1;
+ }
+
+ std::vector<NETEQTEST_RTPpacket *> packet_vec;
+
+ while (1) {
+ // Insert in vector.
+ NETEQTEST_RTPpacket *new_packet = new NETEQTEST_RTPpacket();
+ if (new_packet->readFromFile(in_file) < 0) {
+ // End of file.
+ break;
+ }
+
+ // Look for new send time in statistics vector.
+ std::pair<uint16_t, uint32_t> temp_pair =
+ std::pair<uint16_t, uint32_t>(new_packet->sequenceNumber(),
+ new_packet->timeStamp());
+
+ uint32_t new_send_time = packet_stats[temp_pair];
+ new_packet->setTime(new_send_time); // Set new send time.
+ packet_vec.push_back(new_packet); // Insert in vector.
+ }
+
+ // Sort the vector according to send times.
+ std::sort(packet_vec.begin(), packet_vec.end(), pktCmp);
+
+ std::vector<NETEQTEST_RTPpacket *>::iterator it;
+ for (it = packet_vec.begin(); it != packet_vec.end(); it++) {
+ // Write to out file.
+ if ((*it)->writeToFile(out_file) < 0) {
+ printf("Error writing to file\n");
+ return -1;
+ }
+ // Delete packet.
+ delete *it;
+ }
+
+ fclose(in_file);
+ fclose(out_file);
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc b/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc
new file mode 100644
index 0000000..f328794
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc
@@ -0,0 +1,2031 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//TODO(hlundin): Reformat file to meet style guide.
+
+/* header includes */
+#include "typedefs.h"
+#include "stdio.h"
+#include "webrtc_neteq.h" // needed for enum WebRtcNetEQDecoder
+#include <string.h>
+#include <stdlib.h>
+#include <cassert>
+
+#ifdef WIN32
+#include <winsock2.h>
+#endif
+#ifdef WEBRTC_LINUX
+#include <netinet/in.h>
+#endif
+
+
+/************************/
+/* Define payload types */
+/************************/
+
+#include "PayloadTypes.h"
+
+
+
+/*********************/
+/* Misc. definitions */
+/*********************/
+
+#define STOPSENDTIME 3000
+#define RESTARTSENDTIME 0 //162500
+#define FIRSTLINELEN 40
+#define CHECK_NOT_NULL(a) if((a)==0){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
+
+//#define MULTIPLE_SAME_TIMESTAMP
+#define REPEAT_PACKET_DISTANCE 17
+#define REPEAT_PACKET_COUNT 1 // number of extra packets to send
+
+//#define INSERT_OLD_PACKETS
+#define OLD_PACKET 5 // how many seconds too old should the packet be?
+
+//#define TIMESTAMP_WRAPAROUND
+
+//#define RANDOM_DATA
+//#define RANDOM_PAYLOAD_DATA
+#define RANDOM_SEED 10
+
+//#define INSERT_DTMF_PACKETS
+//#define NO_DTMF_OVERDUB
+#define DTMF_PACKET_INTERVAL 2000
+#define DTMF_DURATION 500
+
+#define STEREO_MODE_FRAME 0
+#define STEREO_MODE_SAMPLE_1 1 //1 octet per sample
+#define STEREO_MODE_SAMPLE_2 2 //2 octets per sample
+
+/*************************/
+/* Function declarations */
+/*************************/
+
+void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed);
+int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
+void defineCodecs(enum WebRtcNetEQDecoder *usedCodec, int *noOfCodecs );
+int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels);
+int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc);
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
+ int seqNo, WebRtc_UWord32 ssrc);
+int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
+void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples);
+void stereoInterleave(unsigned char* data, int dataLen, int stride);
+
+/*********************/
+/* Codec definitions */
+/*********************/
+
+#include "webrtc_vad.h"
+
+#if ((defined CODEC_PCM16B)||(defined NETEQ_ARBITRARY_CODEC))
+ #include "pcm16b.h"
+#endif
+#ifdef CODEC_G711
+ #include "g711_interface.h"
+#endif
+#ifdef CODEC_G729
+ #include "G729Interface.h"
+#endif
+#ifdef CODEC_G729_1
+ #include "G729_1Interface.h"
+#endif
+#ifdef CODEC_AMR
+ #include "AMRInterface.h"
+ #include "AMRCreation.h"
+#endif
+#ifdef CODEC_AMRWB
+ #include "AMRWBInterface.h"
+ #include "AMRWBCreation.h"
+#endif
+#ifdef CODEC_ILBC
+ #include "ilbc.h"
+#endif
+#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
+ #include "isac.h"
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ #include "isacfix.h"
+ #ifdef CODEC_ISAC
+ #error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
+ #endif
+#endif
+#ifdef CODEC_G722
+ #include "g722_interface.h"
+#endif
+#ifdef CODEC_G722_1_24
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1_32
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1_16
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_24
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_32
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_48
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G726
+ #include "G726Creation.h"
+ #include "G726Interface.h"
+#endif
+#ifdef CODEC_GSMFR
+ #include "GSMFRInterface.h"
+ #include "GSMFRCreation.h"
+#endif
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ #include "webrtc_cng.h"
+#endif
+#if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16))
+ #include "SpeexInterface.h"
+#endif
+#ifdef CODEC_CELT_32
+#include "celt_interface.h"
+#endif
+
+
+/***********************************/
+/* Global codec instance variables */
+/***********************************/
+
+WebRtcVadInst *VAD_inst[2];
+
+#ifdef CODEC_G722
+ G722EncInst *g722EncState[2];
+#endif
+
+#ifdef CODEC_G722_1_24
+ G722_1_24_encinst_t *G722_1_24enc_inst[2];
+#endif
+#ifdef CODEC_G722_1_32
+ G722_1_32_encinst_t *G722_1_32enc_inst[2];
+#endif
+#ifdef CODEC_G722_1_16
+ G722_1_16_encinst_t *G722_1_16enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_24
+ G722_1C_24_encinst_t *G722_1C_24enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_32
+ G722_1C_32_encinst_t *G722_1C_32enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_48
+ G722_1C_48_encinst_t *G722_1C_48enc_inst[2];
+#endif
+#ifdef CODEC_G726
+ G726_encinst_t *G726enc_inst[2];
+#endif
+#ifdef CODEC_G729
+ G729_encinst_t *G729enc_inst[2];
+#endif
+#ifdef CODEC_G729_1
+ G729_1_inst_t *G729_1_inst[2];
+#endif
+#ifdef CODEC_AMR
+ AMR_encinst_t *AMRenc_inst[2];
+ WebRtc_Word16 AMR_bitrate;
+#endif
+#ifdef CODEC_AMRWB
+ AMRWB_encinst_t *AMRWBenc_inst[2];
+ WebRtc_Word16 AMRWB_bitrate;
+#endif
+#ifdef CODEC_ILBC
+ iLBC_encinst_t *iLBCenc_inst[2];
+#endif
+#ifdef CODEC_ISAC
+ ISACStruct *ISAC_inst[2];
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ ISACFIX_MainStruct *ISAC_inst[2];
+#endif
+#ifdef CODEC_ISAC_SWB
+ ISACStruct *ISACSWB_inst[2];
+#endif
+#ifdef CODEC_GSMFR
+ GSMFR_encinst_t *GSMFRenc_inst[2];
+#endif
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ CNG_enc_inst *CNGenc_inst[2];
+#endif
+#ifdef CODEC_SPEEX_8
+ SPEEX_encinst_t *SPEEX8enc_inst[2];
+#endif
+#ifdef CODEC_SPEEX_16
+ SPEEX_encinst_t *SPEEX16enc_inst[2];
+#endif
+#ifdef CODEC_CELT_32
+ CELT_encinst_t *CELT32enc_inst[2];
+#endif
+#ifdef CODEC_G711
+ void *G711state[2]={NULL, NULL};
+#endif
+
+
+int main(int argc, char* argv[])
+{
+ int packet_size, fs;
+ enum WebRtcNetEQDecoder usedCodec;
+ int payloadType;
+ int bitrate = 0;
+ int useVAD, vad;
+ int useRed=0;
+ int len, enc_len;
+ WebRtc_Word16 org_data[4000];
+ unsigned char rtp_data[8000];
+ WebRtc_Word16 seqNo=0xFFF;
+ WebRtc_UWord32 ssrc=1235412312;
+ WebRtc_UWord32 timestamp=0xAC1245;
+ WebRtc_UWord16 length, plen;
+ WebRtc_UWord32 offset;
+ double sendtime = 0;
+ int red_PT[2] = {0};
+ WebRtc_UWord32 red_TS[2] = {0};
+ WebRtc_UWord16 red_len[2] = {0};
+ int RTPheaderLen=12;
+ unsigned char red_data[8000];
+#ifdef INSERT_OLD_PACKETS
+ WebRtc_UWord16 old_length, old_plen;
+ int old_enc_len;
+ int first_old_packet=1;
+ unsigned char old_rtp_data[8000];
+ int packet_age=0;
+#endif
+#ifdef INSERT_DTMF_PACKETS
+ int NTone = 1;
+ int DTMFfirst = 1;
+ WebRtc_UWord32 DTMFtimestamp;
+ bool dtmfSent = false;
+#endif
+ bool usingStereo = false;
+ int stereoMode = 0;
+ int numChannels = 1;
+
+ /* check number of parameters */
+ if ((argc != 6) && (argc != 7)) {
+ /* print help text and exit */
+ printf("Application to encode speech into an RTP stream.\n");
+ printf("The program reads a PCM file and encodes is using the specified codec.\n");
+ printf("The coded speech is packetized in RTP packest and written to the output file.\n");
+ printf("The format of the RTP stream file is simlilar to that of rtpplay,\n");
+ printf("but with the receive time euqal to 0 for all packets.\n");
+ printf("Usage:\n\n");
+ printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
+ printf("where:\n");
+
+ printf("PCMfile : PCM speech input file\n\n");
+
+ printf("RTPfile : RTP stream output file\n\n");
+
+ printf("frameLen : 80...960... Number of samples per packet (limit depends on codec)\n\n");
+
+ printf("codecName\n");
+#ifdef CODEC_PCM16B
+ printf(" : pcm16b 16 bit PCM (8kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_WB
+ printf(" : pcm16b_wb 16 bit PCM (16kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n");
+#endif
+#ifdef CODEC_G711
+ printf(" : pcma g711 A-law (8kHz)\n");
+#endif
+#ifdef CODEC_G711
+ printf(" : pcmu g711 u-law (8kHz)\n");
+#endif
+#ifdef CODEC_G729
+ printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n");
+#endif
+#ifdef CODEC_G729_1
+ printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 kbps)\n");
+#endif
+#ifdef CODEC_G722_1_16
+ printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with 16kbps)\n");
+#endif
+#ifdef CODEC_G722_1_24
+ printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps version)\n");
+#endif
+#ifdef CODEC_G722_1_32
+ printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_24
+ printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_32
+ printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_48
+ printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n");
+#endif
+
+#ifdef CODEC_G726
+ printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
+ printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
+ printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
+ printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
+#endif
+#ifdef CODEC_AMR
+ printf(" : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n");
+ printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n");
+#endif
+#ifdef CODEC_AMRWB
+ printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP codec (16kHz)\n");
+ printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
+#endif
+#ifdef CODEC_ILBC
+ printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
+#endif
+#ifdef CODEC_ISAC
+ printf(" : isac iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n");
+#endif
+#ifdef CODEC_ISAC_SWB
+ printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
+#endif
+#ifdef CODEC_GSMFR
+ printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
+#endif
+#ifdef CODEC_G722
+ printf(" : g722 g722 coder (16kHz) (the 64kbps version)\n");
+#endif
+#ifdef CODEC_SPEEX_8
+ printf(" : speex8 speex coder (8 kHz)\n");
+#endif
+#ifdef CODEC_SPEEX_16
+ printf(" : speex16 speex coder (16 kHz)\n");
+#endif
+#ifdef CODEC_CELT_32
+ printf(" : celt32 celt coder (32 kHz)\n");
+#endif
+#ifdef CODEC_RED
+#ifdef CODEC_G711
+ printf(" : red_pcm Redundancy RTP packet with 2*G711A frames\n");
+#endif
+#ifdef CODEC_ISAC
+ printf(" : red_isac Redundancy RTP packet with 2*iSAC frames\n");
+#endif
+#endif
+ printf("\n");
+
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ printf("useVAD : 0 Voice Activity Detection is switched off\n");
+ printf(" : 1 Voice Activity Detection is switched on\n\n");
+#else
+ printf("useVAD : 0 Voice Activity Detection switched off (on not supported)\n\n");
+#endif
+ printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n");
+
+ return(0);
+ }
+
+ FILE* in_file=fopen(argv[1],"rb");
+ CHECK_NOT_NULL(in_file);
+ printf("Input file: %s\n",argv[1]);
+ FILE* out_file=fopen(argv[2],"wb");
+ CHECK_NOT_NULL(out_file);
+ printf("Output file: %s\n\n",argv[2]);
+ packet_size=atoi(argv[3]);
+ CHECK_NOT_NULL(packet_size);
+ printf("Packet size: %i\n",packet_size);
+
+ // check for stereo
+ if(argv[4][strlen(argv[4])-1] == '*') {
+ // use stereo
+ usingStereo = true;
+ numChannels = 2;
+ argv[4][strlen(argv[4])-1] = '\0';
+ }
+
+ NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed);
+
+ if(useRed) {
+ RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */
+ }
+
+ useVAD=atoi(argv[5]);
+#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ if (useVAD!=0) {
+ printf("Error: this simulation does not support VAD/DTX/CNG\n");
+ }
+#endif
+
+ // check stereo type
+ if(usingStereo)
+ {
+ switch(usedCodec)
+ {
+ // sample based codecs
+ case kDecoderPCMu:
+ case kDecoderPCMa:
+ case kDecoderG722:
+ {
+ // 1 octet per sample
+ stereoMode = STEREO_MODE_SAMPLE_1;
+ break;
+ }
+ case kDecoderPCM16B:
+ case kDecoderPCM16Bwb:
+ case kDecoderPCM16Bswb32kHz:
+ case kDecoderPCM16Bswb48kHz:
+ {
+ // 2 octets per sample
+ stereoMode = STEREO_MODE_SAMPLE_2;
+ break;
+ }
+
+ // fixed-rate frame codecs (with internal VAD)
+ case kDecoderG729:
+ {
+ if(useVAD) {
+ printf("Cannot use codec-internal VAD and stereo\n");
+ exit(0);
+ }
+ // break intentionally omitted
+ }
+ case kDecoderG722_1_16:
+ case kDecoderG722_1_24:
+ case kDecoderG722_1_32:
+ case kDecoderG722_1C_24:
+ case kDecoderG722_1C_32:
+ case kDecoderG722_1C_48:
+ {
+ stereoMode = STEREO_MODE_FRAME;
+ break;
+ }
+ default:
+ {
+ printf("Cannot use codec %s as stereo codec\n", argv[4]);
+ exit(0);
+ }
+ }
+ }
+
+ if ((usedCodec == kDecoderISAC) || (usedCodec == kDecoderISACswb))
+ {
+ if (argc != 7)
+ {
+ if (usedCodec == kDecoderISAC)
+ {
+ bitrate = 32000;
+ printf(
+ "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
+ }
+ else // (usedCodec==kDecoderISACswb)
+ {
+ bitrate = 56000;
+ printf(
+ "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n");
+ }
+ }
+ else
+ {
+ bitrate = atoi(argv[6]);
+ if (usedCodec == kDecoderISAC)
+ {
+ if ((bitrate < 10000) || (bitrate > 32000))
+ {
+ printf(
+ "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n",
+ bitrate);
+ exit(0);
+ }
+ printf("Running iSAC at bitrate of %i bps\n", bitrate);
+ }
+ else // (usedCodec==kDecoderISACswb)
+ {
+ if ((bitrate < 32000) || (bitrate > 56000))
+ {
+ printf(
+ "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
+ bitrate);
+ exit(0);
+ }
+ }
+ }
+ }
+ else
+ {
+ if (argc == 7)
+ {
+ printf(
+ "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n");
+ exit(0);
+ }
+ }
+
+ if(useRed) {
+ printf("Redundancy engaged. ");
+ }
+ printf("Used codec: %i\n",usedCodec);
+ printf("Payload type: %i\n",payloadType);
+
+ NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels);
+
+ /* write file header */
+ //fprintf(out_file, "#!RTPencode%s\n", "1.0");
+ fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
+ WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
+ fwrite(&dummy_variable, 4, 1, out_file);
+ fwrite(&dummy_variable, 4, 1, out_file);
+ fwrite(&dummy_variable, 4, 1, out_file);
+ fwrite(&dummy_variable, 2, 1, out_file);
+ fwrite(&dummy_variable, 2, 1, out_file);
+
+#ifdef TIMESTAMP_WRAPAROUND
+ timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
+#endif
+#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
+ srand(RANDOM_SEED);
+#endif
+
+ /* if redundancy is used, the first redundant payload is zero length */
+ red_len[0] = 0;
+
+ /* read first frame */
+ len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
+
+ /* de-interleave if stereo */
+ if ( usingStereo )
+ {
+ stereoDeInterleave(org_data, len * numChannels);
+ }
+
+ while (len==packet_size) {
+
+#ifdef INSERT_DTMF_PACKETS
+ dtmfSent = false;
+
+ if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
+ if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
+ // tone has not ended
+ if (DTMFfirst==1) {
+ DTMFtimestamp = timestamp; // save this timestamp
+ DTMFfirst=0;
+ }
+ makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
+ enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
+ }
+ else {
+ // tone has ended
+ makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
+ enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000));
+ NTone++;
+ DTMFfirst=1;
+ }
+
+ /* write RTP packet to file */
+ length = htons(12 + enc_len + 8);
+ plen = htons(12 + enc_len);
+ offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
+ offset = htonl(offset);
+ fwrite(&length, 2, 1, out_file);
+ fwrite(&plen, 2, 1, out_file);
+ fwrite(&offset, 4, 1, out_file);
+ fwrite(rtp_data, 12 + enc_len, 1, out_file);
+
+ dtmfSent = true;
+ }
+#endif
+
+#ifdef NO_DTMF_OVERDUB
+ /* If DTMF is sent, we should not send any speech packets during the same time */
+ if (dtmfSent) {
+ enc_len = 0;
+ }
+ else {
+#endif
+ /* encode frame */
+ enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels);
+ if (enc_len==-1) {
+ printf("Error encoding frame\n");
+ exit(0);
+ }
+
+ if ( usingStereo &&
+ stereoMode != STEREO_MODE_FRAME &&
+ vad == 1 )
+ {
+ // interleave the encoded payload for sample-based codecs (not for CNG)
+ stereoInterleave(&rtp_data[12], enc_len, stereoMode);
+ }
+#ifdef NO_DTMF_OVERDUB
+ }
+#endif
+
+ if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
+ if(useRed) {
+ if(red_len[0] > 0) {
+ memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len);
+ memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
+
+ red_len[1] = enc_len;
+ red_TS[1] = timestamp;
+ if(vad)
+ red_PT[1] = payloadType;
+ else
+ red_PT[1] = NETEQ_CODEC_CN_PT;
+
+ makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+
+
+ enc_len += red_len[0] + RTPheaderLen - 12;
+ }
+ else { // do not use redundancy payload for this packet, i.e., only last payload
+ memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len);
+ //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
+
+ red_len[1] = enc_len;
+ red_TS[1] = timestamp;
+ if(vad)
+ red_PT[1] = payloadType;
+ else
+ red_PT[1] = NETEQ_CODEC_CN_PT;
+
+ makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+
+
+ enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used)
+ }
+ }
+ else {
+
+ /* make RTP header */
+ if (vad) // regular speech data
+ makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
+ else // CNG data
+ makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
+
+ }
+#ifdef MULTIPLE_SAME_TIMESTAMP
+ int mult_pack=0;
+ do {
+#endif //MULTIPLE_SAME_TIMESTAMP
+ /* write RTP packet to file */
+ length = htons(12 + enc_len + 8);
+ plen = htons(12 + enc_len);
+ offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
+ offset = htonl(offset);
+ fwrite(&length, 2, 1, out_file);
+ fwrite(&plen, 2, 1, out_file);
+ fwrite(&offset, 4, 1, out_file);
+#ifdef RANDOM_DATA
+ for (int k=0; k<12+enc_len; k++) {
+ rtp_data[k] = rand() + rand();
+ }
+#endif
+#ifdef RANDOM_PAYLOAD_DATA
+ for (int k=12; k<12+enc_len; k++) {
+ rtp_data[k] = rand() + rand();
+ }
+#endif
+ fwrite(rtp_data, 12 + enc_len, 1, out_file);
+#ifdef MULTIPLE_SAME_TIMESTAMP
+ } while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
+#endif //MULTIPLE_SAME_TIMESTAMP
+
+#ifdef INSERT_OLD_PACKETS
+ if (packet_age >= OLD_PACKET*fs) {
+ if (!first_old_packet) {
+ // send the old packet
+ fwrite(&old_length, 2, 1, out_file);
+ fwrite(&old_plen, 2, 1, out_file);
+ fwrite(&offset, 4, 1, out_file);
+ fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file);
+ }
+ // store current packet as old
+ old_length=length;
+ old_plen=plen;
+ memcpy(old_rtp_data,rtp_data,12+enc_len);
+ old_enc_len=enc_len;
+ first_old_packet=0;
+ packet_age=0;
+
+ }
+ packet_age += packet_size;
+#endif
+
+ if(useRed) {
+ /* move data to redundancy store */
+#ifdef CODEC_ISAC
+ if(usedCodec==kDecoderISAC)
+ {
+ assert(!usingStereo); // Cannot handle stereo yet
+ red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (WebRtc_Word16*)red_data);
+ }
+ else
+ {
+#endif
+ memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len);
+ red_len[0]=red_len[1];
+#ifdef CODEC_ISAC
+ }
+#endif
+ red_TS[0]=red_TS[1];
+ red_PT[0]=red_PT[1];
+ }
+
+ }
+
+ /* read next frame */
+ len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
+ /* de-interleave if stereo */
+ if ( usingStereo )
+ {
+ stereoDeInterleave(org_data, len * numChannels);
+ }
+
+ if (payloadType==NETEQ_CODEC_G722_PT)
+ timestamp+=len>>1;
+ else
+ timestamp+=len;
+
+ sendtime += (double) len/(fs/1000);
+ }
+
+ NetEQTest_free_coders(usedCodec, numChannels);
+ fclose(in_file);
+ fclose(out_file);
+ printf("Done!\n");
+
+ return(0);
+}
+
+
+
+
+/****************/
+/* Subfunctions */
+/****************/
+
+void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed) {
+
+ *bitrate = 0; /* Default bitrate setting */
+ *useRed = 0; /* Default no redundancy */
+
+ if(!strcmp(name,"pcmu")){
+ *codec=kDecoderPCMu;
+ *PT=NETEQ_CODEC_PCMU_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcma")){
+ *codec=kDecoderPCMa;
+ *PT=NETEQ_CODEC_PCMA_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcm16b")){
+ *codec=kDecoderPCM16B;
+ *PT=NETEQ_CODEC_PCM16B_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcm16b_wb")){
+ *codec=kDecoderPCM16Bwb;
+ *PT=NETEQ_CODEC_PCM16B_WB_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"pcm16b_swb32")){
+ *codec=kDecoderPCM16Bswb32kHz;
+ *PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
+ *fs=32000;
+ }
+ else if(!strcmp(name,"pcm16b_swb48")){
+ *codec=kDecoderPCM16Bswb48kHz;
+ *PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
+ *fs=48000;
+ }
+ else if(!strcmp(name,"g722")){
+ *codec=kDecoderG722;
+ *PT=NETEQ_CODEC_G722_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"g722.1_16")){
+ *codec=kDecoderG722_1_16;
+ *PT=NETEQ_CODEC_G722_1_16_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"g722.1_24")){
+ *codec=kDecoderG722_1_24;
+ *PT=NETEQ_CODEC_G722_1_24_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"g722.1_32")){
+ *codec=kDecoderG722_1_32;
+ *PT=NETEQ_CODEC_G722_1_32_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"g722.1C_24")){
+ *codec=kDecoderG722_1C_24;
+ *PT=NETEQ_CODEC_G722_1C_24_PT;
+ *fs=32000;
+ }
+ else if(!strcmp(name,"g722.1C_32")){
+ *codec=kDecoderG722_1C_32;
+ *PT=NETEQ_CODEC_G722_1C_32_PT;
+ *fs=32000;
+ }
+ else if(!strcmp(name,"g722.1C_48")){
+ *codec=kDecoderG722_1C_48;
+ *PT=NETEQ_CODEC_G722_1C_48_PT;
+ *fs=32000;
+ }
+ else if(!strcmp(name,"g726_16")){
+ *fs=8000;
+ *codec=kDecoderG726_16;
+ *PT=NETEQ_CODEC_G726_16_PT;
+ *bitrate=16;
+ }
+ else if(!strcmp(name,"g726_24")){
+ *fs=8000;
+ *codec=kDecoderG726_24;
+ *PT=NETEQ_CODEC_G726_24_PT;
+ *bitrate=24;
+ }
+ else if(!strcmp(name,"g726_32")){
+ *fs=8000;
+ *codec=kDecoderG726_32;
+ *PT=NETEQ_CODEC_G726_32_PT;
+ *bitrate=32;
+ }
+ else if(!strcmp(name,"g726_40")){
+ *fs=8000;
+ *codec=kDecoderG726_40;
+ *PT=NETEQ_CODEC_G726_40_PT;
+ *bitrate=40;
+ }
+ else if((!strcmp(name,"amr4.75k"))||(!strcmp(name,"amr5.15k"))||(!strcmp(name,"amr5.9k"))||
+ (!strcmp(name,"amr6.7k"))||(!strcmp(name,"amr7.4k"))||(!strcmp(name,"amr7.95k"))||
+ (!strcmp(name,"amr10.2k"))||(!strcmp(name,"amr12.2k"))) {
+ *fs=8000;
+ if (!strcmp(name,"amr4.75k"))
+ *bitrate = 0;
+ if (!strcmp(name,"amr5.15k"))
+ *bitrate = 1;
+ if (!strcmp(name,"amr5.9k"))
+ *bitrate = 2;
+ if (!strcmp(name,"amr6.7k"))
+ *bitrate = 3;
+ if (!strcmp(name,"amr7.4k"))
+ *bitrate = 4;
+ if (!strcmp(name,"amr7.95k"))
+ *bitrate = 5;
+ if (!strcmp(name,"amr10.2k"))
+ *bitrate = 6;
+ if (!strcmp(name,"amr12.2k"))
+ *bitrate = 7;
+ *codec=kDecoderAMR;
+ *PT=NETEQ_CODEC_AMR_PT;
+ }
+ else if((!strcmp(name,"amrwb7k"))||(!strcmp(name,"amrwb9k"))||(!strcmp(name,"amrwb12k"))||
+ (!strcmp(name,"amrwb14k"))||(!strcmp(name,"amrwb16k"))||(!strcmp(name,"amrwb18k"))||
+ (!strcmp(name,"amrwb20k"))||(!strcmp(name,"amrwb23k"))||(!strcmp(name,"amrwb24k"))) {
+ *fs=16000;
+ if (!strcmp(name,"amrwb7k"))
+ *bitrate = 7000;
+ if (!strcmp(name,"amrwb9k"))
+ *bitrate = 9000;
+ if (!strcmp(name,"amrwb12k"))
+ *bitrate = 12000;
+ if (!strcmp(name,"amrwb14k"))
+ *bitrate = 14000;
+ if (!strcmp(name,"amrwb16k"))
+ *bitrate = 16000;
+ if (!strcmp(name,"amrwb18k"))
+ *bitrate = 18000;
+ if (!strcmp(name,"amrwb20k"))
+ *bitrate = 20000;
+ if (!strcmp(name,"amrwb23k"))
+ *bitrate = 23000;
+ if (!strcmp(name,"amrwb24k"))
+ *bitrate = 24000;
+ *codec=kDecoderAMRWB;
+ *PT=NETEQ_CODEC_AMRWB_PT;
+ }
+ else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){
+ *fs=8000;
+ *codec=kDecoderILBC;
+ *PT=NETEQ_CODEC_ILBC_PT;
+ }
+ else if(!strcmp(name,"isac")){
+ *fs=16000;
+ *codec=kDecoderISAC;
+ *PT=NETEQ_CODEC_ISAC_PT;
+ }
+ else if(!strcmp(name,"isacswb")){
+ *fs=32000;
+ *codec=kDecoderISACswb;
+ *PT=NETEQ_CODEC_ISACSWB_PT;
+ }
+ else if(!strcmp(name,"g729")){
+ *fs=8000;
+ *codec=kDecoderG729;
+ *PT=NETEQ_CODEC_G729_PT;
+ }
+ else if(!strcmp(name,"g729.1")){
+ *fs=16000;
+ *codec=kDecoderG729_1;
+ *PT=NETEQ_CODEC_G729_1_PT;
+ }
+ else if(!strcmp(name,"gsmfr")){
+ *fs=8000;
+ *codec=kDecoderGSMFR;
+ *PT=NETEQ_CODEC_GSMFR_PT;
+ }
+ else if(!strcmp(name,"speex8")){
+ *fs=8000;
+ *codec=kDecoderSPEEX_8;
+ *PT=NETEQ_CODEC_SPEEX8_PT;
+ }
+ else if(!strcmp(name,"speex16")){
+ *fs=16000;
+ *codec=kDecoderSPEEX_16;
+ *PT=NETEQ_CODEC_SPEEX16_PT;
+ }
+ else if(!strcmp(name,"celt32")){
+ *fs=32000;
+ *codec=kDecoderCELT_32;
+ *PT=NETEQ_CODEC_CELT32_PT;
+ }
+ else if(!strcmp(name,"red_pcm")){
+ *codec=kDecoderPCMa;
+ *PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
+ *fs=8000;
+ *useRed = 1;
+ } else if(!strcmp(name,"red_isac")){
+ *codec=kDecoderISAC;
+ *PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
+ *fs=16000;
+ *useRed = 1;
+ } else {
+ printf("Error: Not a supported codec (%s)\n", name);
+ exit(0);
+ }
+
+}
+
+
+
+
+int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){
+
+ int ok=0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ ok=WebRtcVad_Create(&VAD_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for VAD instance\n");
+ exit(0);
+ }
+ ok=WebRtcVad_Init(VAD_inst[k]);
+ if (ok==-1) {
+ printf("Error: Initialization of VAD struct failed\n");
+ exit(0);
+ }
+
+
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for CNG encoding instance\n");
+ exit(0);
+ }
+ if(sampfreq <= 16000) {
+ ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5);
+ if (ok==-1) {
+ printf("Error: Initialization of CNG struct failed. Error code %d\n",
+ WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
+ exit(0);
+ }
+ }
+#endif
+
+ switch (coder) {
+ case kDecoderReservedStart : // dummy codec
+#ifdef CODEC_PCM16B
+ case kDecoderPCM16B :
+#endif
+#ifdef CODEC_PCM16B_WB
+ case kDecoderPCM16Bwb :
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ case kDecoderPCM16Bswb32kHz :
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ case kDecoderPCM16Bswb48kHz :
+#endif
+#ifdef CODEC_G711
+ case kDecoderPCMu :
+ case kDecoderPCMa :
+#endif
+ // do nothing
+ break;
+#ifdef CODEC_G729
+ case kDecoderG729:
+ if (sampfreq==8000) {
+ if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) {
+ ok=WebRtcG729_CreateEnc(&G729enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G729 encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG729_EncoderInit(G729enc_inst[k], vad);
+ if ((vad==1)&&(enc_frameSize!=80)) {
+ printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3));
+ }
+ } else {
+ printf("\nError - g729 is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G729_1
+ case kDecoderG729_1:
+ if (sampfreq==16000) {
+ if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)
+ ) {
+ ok=WebRtcG7291_Create(&G729_1_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.729.1 codec instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
+ exit(0);
+ }
+ if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) {
+ /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
+ printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n");
+ exit(0);
+ }
+ WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/);
+ } else {
+ printf("\nError - G.729.1 input is always 16 kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_SPEEX_8
+ case kDecoderSPEEX_8 :
+ if (sampfreq==8000) {
+ if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Speex encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+ exit(0);
+ }
+ if ((vad==1)&&(enc_frameSize!=160)) {
+ printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3));
+ vad=0;
+ }
+ ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n");
+ }
+ break;
+#endif
+#ifdef CODEC_SPEEX_16
+ case kDecoderSPEEX_16 :
+ if (sampfreq==16000) {
+ if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) {
+ ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Speex encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+ exit(0);
+ }
+ if ((vad==1)&&(enc_frameSize!=320)) {
+ printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4));
+ vad=0;
+ }
+ ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n");
+ }
+ break;
+#endif
+#ifdef CODEC_CELT_32
+ case kDecoderCELT_32 :
+ if (sampfreq==32000) {
+ if (enc_frameSize==320) {
+ ok=WebRtcCelt_CreateEnc(&CELT32enc_inst[k], 1 /*mono*/);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Celt encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Celt only supports 10 ms!!\n\n");
+ exit(0);
+ }
+ ok=WebRtcCelt_EncoderInit(CELT32enc_inst[k], 1 /*mono*/, 48000 /*bitrate*/);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Celt32 called with sample frequency other than 32 kHz.\n\n");
+ }
+ break;
+#endif
+
+#ifdef CODEC_G722_1_16
+ case kDecoderG722_1_16 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1_24
+ case kDecoderG722_1_24 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1_32
+ case kDecoderG722_1_32 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_24
+ case kDecoderG722_1C_24 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_32
+ case kDecoderG722_1C_32 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_48
+ case kDecoderG722_1C_48 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722
+ case kDecoderG722 :
+ if (sampfreq==16000) {
+ if (enc_frameSize%2==0) {
+ } else {
+ printf("\nError - g722 frames must have an even number of enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcG722_CreateEncoder(&g722EncState[k]);
+ WebRtcG722_EncoderInit(g722EncState[k]);
+ } else {
+ printf("\nError - g722 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_AMR
+ case kDecoderAMR :
+ if (sampfreq==8000) {
+ ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for AMR encoding instance\n");
+ exit(0);
+ }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
+ WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
+ AMR_bitrate = bitrate;
+ } else {
+ printf("\nError - AMR is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_AMRWB
+ case kDecoderAMRWB :
+ if (sampfreq==16000) {
+ ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for AMRWB encoding instance\n");
+ exit(0);
+ }
+ if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) {
+ printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
+ exit(0);
+ }
+ WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
+ if (bitrate==7000) {
+ AMRWB_bitrate = AMRWB_MODE_7k;
+ } else if (bitrate==9000) {
+ AMRWB_bitrate = AMRWB_MODE_9k;
+ } else if (bitrate==12000) {
+ AMRWB_bitrate = AMRWB_MODE_12k;
+ } else if (bitrate==14000) {
+ AMRWB_bitrate = AMRWB_MODE_14k;
+ } else if (bitrate==16000) {
+ AMRWB_bitrate = AMRWB_MODE_16k;
+ } else if (bitrate==18000) {
+ AMRWB_bitrate = AMRWB_MODE_18k;
+ } else if (bitrate==20000) {
+ AMRWB_bitrate = AMRWB_MODE_20k;
+ } else if (bitrate==23000) {
+ AMRWB_bitrate = AMRWB_MODE_23k;
+ } else if (bitrate==24000) {
+ AMRWB_bitrate = AMRWB_MODE_24k;
+ }
+ WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
+
+ } else {
+ printf("\nError - AMRwb is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ILBC
+ case kDecoderILBC :
+ if (sampfreq==8000) {
+ ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iLBC encoding instance\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==320)) {
+ /* 20 ms version */
+ WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
+ } else {
+ /* 30 ms version */
+ WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
+ }
+ } else {
+ printf("\nError - iLBC is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ISAC
+ case kDecoderISAC:
+ if (sampfreq==16000) {
+ ok=WebRtcIsac_Create(&ISAC_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC instance\n");
+ exit(0);
+ }if ((enc_frameSize==480)||(enc_frameSize==960)) {
+ } else {
+ printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+ exit(0);
+ }
+ WebRtcIsac_EncoderInit(ISAC_inst[k],1);
+ if ((bitrate<10000)||(bitrate>32000)) {
+ printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
+ } else {
+ printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ case kDecoderISAC:
+ if (sampfreq==16000) {
+ ok=WebRtcIsacfix_Create(&ISAC_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC instance\n");
+ exit(0);
+ }if ((enc_frameSize==480)||(enc_frameSize==960)) {
+ } else {
+ printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+ exit(0);
+ }
+ WebRtcIsacfix_EncoderInit(ISAC_inst[k],1);
+ if ((bitrate<10000)||(bitrate>32000)) {
+ printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
+ } else {
+ printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ISAC_SWB
+ case kDecoderISACswb:
+ if (sampfreq==32000) {
+ ok=WebRtcIsac_Create(&ISACSWB_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
+ exit(0);
+ }if (enc_frameSize==960) {
+ } else {
+ printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
+ exit(0);
+ }
+ ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], kIsacSuperWideband);
+ if (ok!=0) {
+ printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
+ exit(0);
+ }
+ WebRtcIsac_EncoderInit(ISACSWB_inst[k],1);
+ if ((bitrate<32000)||(bitrate>56000)) {
+ printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5);
+ } else {
+ printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_GSMFR
+ case kDecoderGSMFR:
+ if (sampfreq==8000) {
+ ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for GSM FR encoding instance\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
+ } else {
+ printf("\nError - GSM FR is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+ default :
+ printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
+ exit(0);
+ break;
+ }
+
+ if (ok != 0) {
+ return(ok);
+ }
+ } // end for
+
+ return(0);
+}
+
+
+
+
+int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels) {
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ WebRtcVad_Free(VAD_inst[k]);
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ WebRtcCng_FreeEnc(CNGenc_inst[k]);
+#endif
+
+ switch (coder)
+ {
+ case kDecoderReservedStart : // dummy codec
+#ifdef CODEC_PCM16B
+ case kDecoderPCM16B :
+#endif
+#ifdef CODEC_PCM16B_WB
+ case kDecoderPCM16Bwb :
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ case kDecoderPCM16Bswb32kHz :
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ case kDecoderPCM16Bswb48kHz :
+#endif
+#ifdef CODEC_G711
+ case kDecoderPCMu :
+ case kDecoderPCMa :
+#endif
+ // do nothing
+ break;
+#ifdef CODEC_G729
+ case kDecoderG729:
+ WebRtcG729_FreeEnc(G729enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G729_1
+ case kDecoderG729_1:
+ WebRtcG7291_Free(G729_1_inst[k]);
+ break;
+#endif
+#ifdef CODEC_SPEEX_8
+ case kDecoderSPEEX_8 :
+ WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_SPEEX_16
+ case kDecoderSPEEX_16 :
+ WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_CELT_32
+ case kDecoderCELT_32 :
+ WebRtcCelt_FreeEnc(CELT32enc_inst[k]);
+ break;
+#endif
+
+#ifdef CODEC_G722_1_16
+ case kDecoderG722_1_16 :
+ WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1_24
+ case kDecoderG722_1_24 :
+ WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1_32
+ case kDecoderG722_1_32 :
+ WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_24
+ case kDecoderG722_1C_24 :
+ WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_32
+ case kDecoderG722_1C_32 :
+ WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_48
+ case kDecoderG722_1C_48 :
+ WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722
+ case kDecoderG722 :
+ WebRtcG722_FreeEncoder(g722EncState[k]);
+ break;
+#endif
+#ifdef CODEC_AMR
+ case kDecoderAMR :
+ WebRtcAmr_FreeEnc(AMRenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_AMRWB
+ case kDecoderAMRWB :
+ WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ILBC
+ case kDecoderILBC :
+ WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ISAC
+ case kDecoderISAC:
+ WebRtcIsac_Free(ISAC_inst[k]);
+ break;
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ case kDecoderISAC:
+ WebRtcIsacfix_Free(ISAC_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ISAC_SWB
+ case kDecoderISACswb:
+ WebRtcIsac_Free(ISACSWB_inst[k]);
+ break;
+#endif
+#ifdef CODEC_GSMFR
+ case kDecoderGSMFR:
+ WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
+ break;
+#endif
+ default :
+ printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
+ exit(0);
+ break;
+ }
+ }
+
+ return(0);
+}
+
+
+
+
+
+
+int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate ,
+ int * vad, int useVAD, int bitrate, int numChannels){
+
+ short cdlen = 0;
+ WebRtc_Word16 *tempdata;
+ static int first_cng=1;
+ WebRtc_Word16 tempLen;
+
+ *vad =1;
+
+ // check VAD first
+ if(useVAD&&
+ (coder!=kDecoderG729)&&(coder!=kDecoderAMR)&&
+ (coder!=kDecoderSPEEX_8)&&(coder!=kDecoderSPEEX_16))
+ {
+ *vad = 0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ tempLen = frameLen;
+ tempdata = &indata[k*frameLen];
+ int localVad=0;
+ /* Partition the signal and test each chunk for VAD.
+ All chunks must be VAD=0 to produce a total VAD=0. */
+ while (tempLen >= 10*sampleRate/1000) {
+ if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000);
+ tempdata += 30*sampleRate/1000;
+ tempLen -= 30*sampleRate/1000;
+ }
+ else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000);
+ tempdata += 20*sampleRate/1000;
+ tempLen -= 20*sampleRate/1000;
+ }
+ else { // use 10ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000);
+ tempdata += 10*sampleRate/1000;
+ tempLen -= 10*sampleRate/1000;
+ }
+ }
+
+ // aggregate all VAD decisions over all channels
+ *vad |= localVad;
+ }
+
+ if(!*vad){
+ // all channels are silent
+ cdlen = 0;
+ for (int k = 0; k < numChannels; k++)
+ {
+ WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */,
+ encoded,&tempLen,first_cng);
+ encoded += tempLen;
+ cdlen += tempLen;
+ }
+ *vad=0;
+ first_cng=0;
+ return(cdlen);
+ }
+ }
+
+
+ // loop over all channels
+ int totalLen = 0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ /* Encode with the selected coder type */
+ if (coder==kDecoderPCMu) { /*g711 u-law */
+#ifdef CODEC_G711
+ cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+#endif
+ }
+ else if (coder==kDecoderPCMa) { /*g711 A-law */
+#ifdef CODEC_G711
+ cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+ }
+#endif
+#ifdef CODEC_PCM16B
+ else if ((coder==kDecoderPCM16B)||(coder==kDecoderPCM16Bwb)||
+ (coder==kDecoderPCM16Bswb32kHz)||(coder==kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
+ cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (WebRtc_Word16*) encoded);
+ }
+#endif
+#ifdef CODEC_G722
+ else if (coder==kDecoderG722) { /*g722 */
+ cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ cdlen=frameLen>>1;
+ }
+#endif
+#ifdef CODEC_G722_1_16
+ else if (coder==kDecoderG722_1_16) { /* g722.1 16kbit/s mode */
+ cdlen=WebRtcG7221_Encode16((G722_1_16_encinst_t*)G722_1_16enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G722_1_24
+ else if (coder==kDecoderG722_1_24) { /* g722.1 24kbit/s mode*/
+ cdlen=WebRtcG7221_Encode24((G722_1_24_encinst_t*)G722_1_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G722_1_32
+ else if (coder==kDecoderG722_1_32) { /* g722.1 32kbit/s mode */
+ cdlen=WebRtcG7221_Encode32((G722_1_32_encinst_t*)G722_1_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G722_1C_24
+ else if (coder==kDecoderG722_1C_24) { /* g722.1 32 kHz 24kbit/s mode*/
+ cdlen=WebRtcG7221C_Encode24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G722_1C_32
+ else if (coder==kDecoderG722_1C_32) { /* g722.1 32 kHz 32kbit/s mode */
+ cdlen=WebRtcG7221C_Encode32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G722_1C_48
+ else if (coder==kDecoderG722_1C_48) { /* g722.1 32 kHz 48kbit/s mode */
+ cdlen=WebRtcG7221C_Encode48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_G729
+ else if (coder==kDecoderG729) { /*g729 */
+ WebRtc_Word16 dataPos=0;
+ WebRtc_Word16 len=0;
+ cdlen = 0;
+ for (dataPos=0;dataPos<frameLen;dataPos+=80) {
+ len=WebRtcG729_Encode(G729enc_inst[k], &indata[dataPos], 80, (WebRtc_Word16*)(&encoded[cdlen]));
+ cdlen += len;
+ }
+ }
+#endif
+#ifdef CODEC_G729_1
+ else if (coder==kDecoderG729_1) { /*g729.1 */
+ WebRtc_Word16 dataPos=0;
+ WebRtc_Word16 len=0;
+ cdlen = 0;
+ for (dataPos=0;dataPos<frameLen;dataPos+=160) {
+ len=WebRtcG7291_Encode(G729_1_inst[k], &indata[dataPos], (WebRtc_Word16*)(&encoded[cdlen]), bitrate, frameLen/320 /* num 20ms frames*/);
+ cdlen += len;
+ }
+ }
+#endif
+#ifdef CODEC_AMR
+ else if (coder==kDecoderAMR) { /*AMR */
+ cdlen=WebRtcAmr_Encode(AMRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMR_bitrate);
+ }
+#endif
+#ifdef CODEC_AMRWB
+ else if (coder==kDecoderAMRWB) { /*AMR-wb */
+ cdlen=WebRtcAmrWb_Encode(AMRWBenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMRWB_bitrate);
+ }
+#endif
+#ifdef CODEC_ILBC
+ else if (coder==kDecoderILBC) { /*iLBC */
+ cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(WebRtc_Word16*)encoded);
+ }
+#endif
+#if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
+ else if (coder==kDecoderISAC) { /*iSAC */
+ int noOfCalls=0;
+ cdlen=0;
+ while (cdlen<=0) {
+#ifdef CODEC_ISAC /* floating point */
+ cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+#else /* fixed point */
+ cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+#endif
+ noOfCalls++;
+ }
+ }
+#endif
+#ifdef CODEC_ISAC_SWB
+ else if (coder==kDecoderISACswb) { /* iSAC SWB */
+ int noOfCalls=0;
+ cdlen=0;
+ while (cdlen<=0) {
+ cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(WebRtc_Word16*)encoded);
+ noOfCalls++;
+ }
+ }
+#endif
+#ifdef CODEC_GSMFR
+ else if (coder==kDecoderGSMFR) { /* GSM FR */
+ cdlen=WebRtcGSMFR_Encode(GSMFRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_SPEEX_8
+ else if (coder==kDecoderSPEEX_8) { /* Speex */
+ int encodedLen = 0;
+ int retVal = 1;
+ while (retVal == 1 && encodedLen < frameLen) {
+ retVal = WebRtcSpeex_Encode(SPEEX8enc_inst[k], &indata[encodedLen], 15000);
+ encodedLen += 20*8; /* 20 ms */
+ }
+ if( (retVal == 0 && encodedLen != frameLen) || retVal < 0) {
+ printf("Error encoding speex frame!\n");
+ exit(0);
+ }
+ cdlen=WebRtcSpeex_GetBitstream(SPEEX8enc_inst[k], (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_SPEEX_16
+ else if (coder==kDecoderSPEEX_16) { /* Speex */
+ int encodedLen = 0;
+ int retVal = 1;
+ while (retVal == 1 && encodedLen < frameLen) {
+ retVal = WebRtcSpeex_Encode(SPEEX16enc_inst[k], &indata[encodedLen], 15000);
+ encodedLen += 20*16; /* 20 ms */
+ }
+ if( (retVal == 0 && encodedLen != frameLen) || retVal < 0) {
+ printf("Error encoding speex frame!\n");
+ exit(0);
+ }
+ cdlen=WebRtcSpeex_GetBitstream(SPEEX16enc_inst[k], (WebRtc_Word16*)encoded);
+ }
+#endif
+#ifdef CODEC_CELT_32
+ else if (coder==kDecoderCELT_32) { /* Celt */
+ int encodedLen = 0;
+ cdlen = 0;
+ while (cdlen <= 0) {
+ cdlen = WebRtcCelt_Encode(CELT32enc_inst[k], &indata[encodedLen], encoded);
+ encodedLen += 10*32; /* 10 ms */
+ }
+ if( (encodedLen != frameLen) || cdlen < 0) {
+ printf("Error encoding Celt frame!\n");
+ exit(0);
+ }
+ }
+#endif
+
+ indata += frameLen;
+ encoded += cdlen;
+ totalLen += cdlen;
+
+ } // end for
+
+ first_cng=1;
+ return(totalLen);
+}
+
+
+
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc){
+
+ rtp_data[0]=(unsigned char)0x80;
+ rtp_data[1]=(unsigned char)(payloadType & 0xFF);
+ rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
+ rtp_data[3]=(unsigned char)((seqNo)&0xFF);
+ rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
+ rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
+
+ rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
+ rtp_data[7]=(unsigned char)(timestamp & 0xFF);
+
+ rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
+ rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
+
+ rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
+ rtp_data[11]=(unsigned char)(ssrc & 0xFF);
+}
+
+
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
+ int seqNo, WebRtc_UWord32 ssrc)
+{
+
+ int i;
+ unsigned char *rtpPointer;
+ WebRtc_UWord16 offset;
+
+ /* first create "standard" RTP header */
+ makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
+
+ rtpPointer = &rtp_data[12];
+
+ /* add one sub-header for each redundant payload (not the primary) */
+ for(i=0; i<numPayloads-1; i++) { /* |0 1 2 3 4 5 6 7| */
+ if(blockLen[i] > 0) {
+ offset = (WebRtc_UWord16) (timestamp[numPayloads-1] - timestamp[i]);
+
+ rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F| block PT | */
+ rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF); /* | timestamp- | */
+ rtpPointer[2] = (unsigned char) ( ((offset & 0x3F)<<2) |
+ ( (blockLen[i]>>8) & 0x03 ) ); /* | -offset |bl-| */
+ rtpPointer[3] = (unsigned char) ( blockLen[i] & 0xFF ); /* | -ock length | */
+
+ rtpPointer += 4;
+ }
+ }
+
+ /* last sub-header */
+ rtpPointer[0]= (unsigned char) (0x00 | (0x7F&payloadType[numPayloads-1]));/* |F| block PT | */
+ rtpPointer += 1;
+
+ return(rtpPointer - rtp_data); /* length of header in bytes */
+}
+
+
+
+int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) {
+ unsigned char E,R,V;
+ R=0;
+ V=(unsigned char)Volume;
+ if (End==0) {
+ E = 0x00;
+ } else {
+ E = 0x80;
+ }
+ payload_data[0]=(unsigned char)Event;
+ payload_data[1]=(unsigned char)(E|R|V);
+ //Duration equals 8 times time_ms, default is 8000 Hz.
+ payload_data[2]=(unsigned char)((Duration>>8)&0xFF);
+ payload_data[3]=(unsigned char)(Duration&0xFF);
+ return(4);
+}
+
+void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples)
+{
+
+ WebRtc_Word16 *tempVec;
+ WebRtc_Word16 *readPtr, *writeL, *writeR;
+
+ if (numSamples <= 0)
+ return;
+
+ tempVec = (WebRtc_Word16 *) malloc(sizeof(WebRtc_Word16) * numSamples);
+ if (tempVec == NULL) {
+ printf("Error allocating memory\n");
+ exit(0);
+ }
+
+ memcpy(tempVec, audioSamples, numSamples*sizeof(WebRtc_Word16));
+
+ writeL = audioSamples;
+ writeR = &audioSamples[numSamples/2];
+ readPtr = tempVec;
+
+ for (int k = 0; k < numSamples; k += 2)
+ {
+ *writeL = *readPtr;
+ readPtr++;
+ *writeR = *readPtr;
+ readPtr++;
+ writeL++;
+ writeR++;
+ }
+
+ free(tempVec);
+
+}
+
+
+void stereoInterleave(unsigned char* data, int dataLen, int stride)
+{
+
+ unsigned char *ptrL, *ptrR;
+ unsigned char temp[10];
+
+ if (stride > 10)
+ {
+ exit(0);
+ }
+
+ if (dataLen%1 != 0)
+ {
+ // must be even number of samples
+ printf("Error: cannot interleave odd sample number\n");
+ exit(0);
+ }
+
+ ptrL = data + stride;
+ ptrR = &data[dataLen/2];
+
+ while (ptrL < ptrR) {
+ // copy from right pointer to temp
+ memcpy(temp, ptrR, stride);
+
+ // shift data between pointers
+ memmove(ptrL + stride, ptrL, ptrR - ptrL);
+
+ // copy from temp to left pointer
+ memcpy(ptrL, temp, stride);
+
+ // advance pointers
+ ptrL += stride*2;
+ ptrR += stride;
+ }
+
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc b/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc
new file mode 100644
index 0000000..795111f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc
@@ -0,0 +1,196 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//TODO(hlundin): Reformat file to meet style guide.
+
+/* header includes */
+#include "typedefs.h"
+#include <stdio.h>
+#include <stdlib.h>
+#ifdef WIN32
+#include <winsock2.h>
+#include <io.h>
+#endif
+#ifdef WEBRTC_LINUX
+#include <netinet/in.h>
+#endif
+#include <search.h>
+#include <float.h>
+
+#include "gtest/gtest.h"
+
+/*********************/
+/* Misc. definitions */
+/*********************/
+
+#define FIRSTLINELEN 40
+#define CHECK_ZERO(a) {int errCode = a; if((errCode)!=0){fprintf(stderr,"\n %s \n line: %d \n error at %s\n Error Code = %d\n",__FILE__,__LINE__,#a, WebRtcNetEQ_GetErrorCode(inst)); exit(0);}}
+#define CHECK_NOT_NULL(a) if((a)==NULL){fprintf(stderr,"\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
+
+struct arr_time {
+ float time;
+ WebRtc_UWord32 ix;
+};
+
+int filelen(FILE *fid)
+{
+ fpos_t cur_pos;
+ int len;
+
+ if (!fid || fgetpos(fid, &cur_pos)) {
+ return(-1);
+ }
+
+ fseek(fid, 0, SEEK_END);
+ len = ftell(fid);
+
+ fsetpos(fid, &cur_pos);
+
+ return (len);
+}
+
+int compare_arr_time(const void *x, const void *y);
+
+int main(int argc, char* argv[])
+{
+ unsigned int dat_len, rtp_len, Npack, k;
+ arr_time *time_vec;
+ char firstline[FIRSTLINELEN];
+ unsigned char *rtp_vec = NULL, **packet_ptr, *temp_packet;
+ const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+ WebRtc_UWord16 len;
+ WebRtc_UWord32 *offset;
+
+/* check number of parameters */
+ if (argc != 4) {
+ /* print help text and exit */
+ printf("Apply jitter on RTP stream.\n");
+ printf("The program reads an RTP stream and packet timing from two files.\n");
+ printf("The RTP stream is modified to have the same jitter as described in the timing files.\n");
+ printf("The format of the RTP stream file should be the same as for rtpplay,\n");
+ printf("and can be obtained e.g., from Ethereal by using\n");
+ printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> Save As\n\n");
+ printf("Usage:\n\n");
+ printf("%s RTP_infile dat_file RTP_outfile\n", argv[0]);
+ printf("where:\n");
+
+ printf("RTP_infile : RTP stream input file\n\n");
+
+ printf("dat_file : file with packet arrival times in ms\n\n");
+
+ printf("RTP_outfile : RTP stream output file\n\n");
+
+ return(0);
+ }
+
+ FILE* in_file=fopen(argv[1],"rb");
+ CHECK_NOT_NULL(in_file);
+ printf("Input file: %s\n",argv[1]);
+ FILE* dat_file=fopen(argv[2],"rb");
+ CHECK_NOT_NULL(dat_file);
+ printf("Dat-file: %s\n",argv[2]);
+ FILE* out_file=fopen(argv[3],"wb");
+ CHECK_NOT_NULL(out_file);
+ printf("Output file: %s\n\n",argv[3]);
+
+ time_vec = (arr_time *) malloc(sizeof(arr_time)*(filelen(dat_file)/sizeof(float)) + 1000); // add 1000 bytes to avoid (rare) strange error
+ if (time_vec==NULL) {
+ fprintf(stderr, "Error: could not allocate memory for reading dat file\n");
+ goto closing;
+ }
+
+ dat_len=0;
+ while(fread(&(time_vec[dat_len].time),sizeof(float),1,dat_file)>0) {
+ time_vec[dat_len].ix=dat_len;
+ dat_len++;
+ }
+
+ qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
+
+
+ rtp_vec = (unsigned char *) malloc(sizeof(unsigned char)*filelen(in_file));
+ if (rtp_vec==NULL) {
+ fprintf(stderr,"Error: could not allocate memory for reading rtp file\n");
+ goto closing;
+ }
+
+ // read file header and write directly to output file
+ EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
+ EXPECT_GT(fputs(firstline, out_file), 0);
+ EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
+ in_file));
+ EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
+ out_file));
+
+ // read all RTP packets into vector
+ rtp_len=0;
+ Npack=0;
+ len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
+ while(len==2) {
+ len = ntohs(*((WebRtc_UWord16 *)(rtp_vec + rtp_len)));
+ rtp_len += 2;
+ if(fread(&rtp_vec[rtp_len], sizeof(unsigned char), len-2, in_file)!=(unsigned) (len-2)) {
+ fprintf(stderr,"Error: currupt packet length\n");
+ goto closing;
+ }
+ rtp_len += len-2;
+ Npack++;
+ len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
+ }
+
+ packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
+
+ packet_ptr[0]=rtp_vec;
+ k=1;
+ while(k<Npack) {
+ len = ntohs(*((WebRtc_UWord16 *) packet_ptr[k-1]));
+ packet_ptr[k]=packet_ptr[k-1]+len;
+ k++;
+ }
+
+ for(k=0; k<dat_len && k<Npack; k++) {
+ if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){
+ temp_packet = packet_ptr[time_vec[k].ix];
+ offset = (WebRtc_UWord32 *) (temp_packet+4);
+ if ( time_vec[k].time >= 0 ) {
+ *offset = htonl((WebRtc_UWord32) time_vec[k].time);
+ }
+ else {
+ *offset = htonl((WebRtc_UWord32) 0);
+ fprintf(stderr, "Warning: negative receive time in dat file transformed to 0.\n");
+ }
+
+ // write packet to file
+ fwrite(temp_packet, sizeof(unsigned char), ntohs(*((WebRtc_UWord16*) temp_packet)), out_file);
+ }
+ }
+
+
+closing:
+ free(time_vec);
+ free(rtp_vec);
+ fclose(in_file);
+ fclose(dat_file);
+ fclose(out_file);
+
+ return(0);
+}
+
+
+
+int compare_arr_time(const void *xp, const void *yp) {
+
+ if(((arr_time *)xp)->time == ((arr_time *)yp)->time)
+ return(0);
+ else if(((arr_time *)xp)->time > ((arr_time *)yp)->time)
+ return(1);
+
+ return(-1);
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc b/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc
new file mode 100644
index 0000000..dc7ff9f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc
@@ -0,0 +1,97 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <stdio.h>
+#include <vector>
+
+#include "NETEQTEST_RTPpacket.h"
+#include "gtest/gtest.h"
+
+/*********************/
+/* Misc. definitions */
+/*********************/
+
+#define FIRSTLINELEN 40
+
+
+int main(int argc, char* argv[])
+{
+ if(argc < 4 || argc > 6)
+ {
+ printf("Usage: RTPtimeshift in.rtp out.rtp newStartTS [newStartSN [newStartArrTime]]\n");
+ exit(1);
+ }
+
+ FILE *inFile=fopen(argv[1],"rb");
+ if (!inFile)
+ {
+ printf("Cannot open input file %s\n", argv[1]);
+ return(-1);
+ }
+ printf("Input RTP file: %s\n",argv[1]);
+
+ FILE *outFile=fopen(argv[2],"wb");
+ if (!outFile)
+ {
+ printf("Cannot open output file %s\n", argv[2]);
+ return(-1);
+ }
+ printf("Output RTP file: %s\n\n",argv[2]);
+
+ // read file header and write directly to output file
+ const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+ char firstline[FIRSTLINELEN];
+ EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
+ EXPECT_GT(fputs(firstline, outFile), 0);
+ EXPECT_EQ(kRtpDumpHeaderSize,
+ fread(firstline, 1, kRtpDumpHeaderSize, inFile));
+ EXPECT_EQ(kRtpDumpHeaderSize,
+ fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
+ NETEQTEST_RTPpacket packet;
+ int packLen = packet.readFromFile(inFile);
+ if (packLen < 0)
+ {
+ exit(1);
+ }
+
+ // get new start TS and start SeqNo from arguments
+ WebRtc_UWord32 TSdiff = atoi(argv[3]) - packet.timeStamp();
+ WebRtc_UWord16 SNdiff = 0;
+ WebRtc_UWord32 ATdiff = 0;
+ if (argc > 4)
+ {
+ if (argv[4] >= 0)
+ SNdiff = atoi(argv[4]) - packet.sequenceNumber();
+ if (argc > 5)
+ {
+ if (argv[5] >= 0)
+ ATdiff = atoi(argv[5]) - packet.time();
+ }
+ }
+
+ while (packLen >= 0)
+ {
+
+ packet.setTimeStamp(packet.timeStamp() + TSdiff);
+ packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
+ packet.setTime(packet.time() + ATdiff);
+
+ packet.writeToFile(outFile);
+
+ packLen = packet.readFromFile(inFile);
+
+ }
+
+ fclose(inFile);
+ fclose(outFile);
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m b/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
new file mode 100644
index 0000000..77b394f
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
@@ -0,0 +1,191 @@
+function outStruct = parse_delay_file(file)
+
+fid = fopen(file, 'rb');
+if fid == -1
+ error('Cannot open file %s', file);
+end
+
+textline = fgetl(fid);
+if ~strncmp(textline, '#!NetEQ_Delay_Logging', 21)
+ error('Wrong file format');
+end
+
+ver = sscanf(textline, '#!NetEQ_Delay_Logging%d.%d');
+if ~all(ver == [2; 0])
+ error('Wrong version of delay logging function')
+end
+
+
+start_pos = ftell(fid);
+fseek(fid, -12, 'eof');
+textline = fgetl(fid);
+if ~strncmp(textline, 'End of file', 21)
+ error('File ending is not correct. Seems like the simulation ended abnormally.');
+end
+
+fseek(fid,-12-4, 'eof');
+Npackets = fread(fid, 1, 'int32');
+fseek(fid, start_pos, 'bof');
+
+rtpts = zeros(Npackets, 1);
+seqno = zeros(Npackets, 1);
+pt = zeros(Npackets, 1);
+plen = zeros(Npackets, 1);
+recin_t = nan*ones(Npackets, 1);
+decode_t = nan*ones(Npackets, 1);
+playout_delay = zeros(Npackets, 1);
+optbuf = zeros(Npackets, 1);
+
+fs_ix = 1;
+clock = 0;
+ts_ix = 1;
+ended = 0;
+late_packets = 0;
+fs_now = 8000;
+last_decode_k = 0;
+tot_expand = 0;
+tot_accelerate = 0;
+tot_preemptive = 0;
+
+while not(ended)
+ signal = fread(fid, 1, '*int32');
+
+ switch signal
+ case 3 % NETEQ_DELAY_LOGGING_SIGNAL_CLOCK
+ clock = fread(fid, 1, '*float32');
+
+ % keep on reading batches of M until the signal is no longer "3"
+ % read int32 + float32 in one go
+ % this is to save execution time
+ temp = [3; 0];
+ M = 120;
+ while all(temp(1,:) == 3)
+ fp = ftell(fid);
+ temp = fread(fid, [2 M], '*int32');
+ end
+
+ % back up to last clock event
+ fseek(fid, fp - ftell(fid) + ...
+ (find(temp(1,:) ~= 3, 1 ) - 2) * 2 * 4 + 4, 'cof');
+ % read the last clock value
+ clock = fread(fid, 1, '*float32');
+
+ case 1 % NETEQ_DELAY_LOGGING_SIGNAL_RECIN
+ temp_ts = fread(fid, 1, 'uint32');
+
+ if late_packets > 0
+ temp_ix = ts_ix - 1;
+ while (temp_ix >= 1) && (rtpts(temp_ix) ~= temp_ts)
+ % TODO(hlundin): use matlab vector search instead?
+ temp_ix = temp_ix - 1;
+ end
+
+ if temp_ix >= 1
+ % the ts was found in the vector
+ late_packets = late_packets - 1;
+ else
+ temp_ix = ts_ix;
+ ts_ix = ts_ix + 1;
+ end
+ else
+ temp_ix = ts_ix;
+ ts_ix = ts_ix + 1;
+ end
+
+ rtpts(temp_ix) = temp_ts;
+ seqno(temp_ix) = fread(fid, 1, 'uint16');
+ pt(temp_ix) = fread(fid, 1, 'int32');
+ plen(temp_ix) = fread(fid, 1, 'int16');
+ recin_t(temp_ix) = clock;
+
+ case 2 % NETEQ_DELAY_LOGGING_SIGNAL_FLUSH
+ % do nothing
+
+ case 4 % NETEQ_DELAY_LOGGING_SIGNAL_EOF
+ ended = 1;
+
+ case 5 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE
+ last_decode_ts = fread(fid, 1, 'uint32');
+ temp_delay = fread(fid, 1, 'uint16');
+
+ k = find(rtpts(1:(ts_ix - 1))==last_decode_ts,1,'last');
+ if ~isempty(k)
+ decode_t(k) = clock;
+ playout_delay(k) = temp_delay + ...
+ 5 * fs_now / 8000; % add overlap length
+ last_decode_k = k;
+ end
+
+ case 6 % NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS
+ fsvec(fs_ix) = fread(fid, 1, 'uint16');
+ fschange_ts(fs_ix) = last_decode_ts;
+ fs_now = fsvec(fs_ix);
+ fs_ix = fs_ix + 1;
+
+ case 7 % NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO
+ playout_delay(last_decode_k) = playout_delay(last_decode_k) ...
+ + fread(fid, 1, 'int32');
+
+ case 8 % NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO
+ temp = fread(fid, 1, 'int32');
+ if last_decode_k ~= 0
+ tot_expand = tot_expand + temp / (fs_now / 1000);
+ end
+
+ case 9 % NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO
+ temp = fread(fid, 1, 'int32');
+ if last_decode_k ~= 0
+ tot_accelerate = tot_accelerate + temp / (fs_now / 1000);
+ end
+
+ case 10 % NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO
+ temp = fread(fid, 1, 'int32');
+ if last_decode_k ~= 0
+ tot_preemptive = tot_preemptive + temp / (fs_now / 1000);
+ end
+
+ case 11 % NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF
+ optbuf(last_decode_k) = fread(fid, 1, 'int32');
+
+ case 12 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC
+ last_decode_ts = fread(fid, 1, 'uint32');
+ k = ts_ix - 1;
+
+ while (k >= 1) && (rtpts(k) ~= last_decode_ts)
+ % TODO(hlundin): use matlab vector search instead?
+ k = k - 1;
+ end
+
+ if k < 1
+ % packet not received yet
+ k = ts_ix;
+ rtpts(ts_ix) = last_decode_ts;
+ late_packets = late_packets + 1;
+ end
+
+ decode_t(k) = clock;
+ playout_delay(k) = fread(fid, 1, 'uint16') + ...
+ 5 * fs_now / 8000; % add overlap length
+ last_decode_k = k;
+
+ end
+
+end
+
+
+fclose(fid);
+
+outStruct = struct(...
+ 'ts', rtpts, ...
+ 'sn', seqno, ...
+ 'pt', pt,...
+ 'plen', plen,...
+ 'arrival', recin_t,...
+ 'decode', decode_t,...
+ 'fs', fsvec(:),...
+ 'fschange_ts', fschange_ts(:),...
+ 'playout_delay', playout_delay,...
+ 'tot_expand', tot_expand,...
+ 'tot_accelerate', tot_accelerate,...
+ 'tot_preemptive', tot_preemptive,...
+ 'optbuf', optbuf);
diff --git a/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m b/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
new file mode 100644
index 0000000..bc1c85a
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
@@ -0,0 +1,187 @@
+function [delay_struct, delayvalues] = plot_neteq_delay(delayfile, varargin)
+
+% InfoStruct = plot_neteq_delay(delayfile)
+% InfoStruct = plot_neteq_delay(delayfile, 'skipdelay', skip_seconds)
+%
+% Henrik Lundin, 2006-11-17
+% Henrik Lundin, 2011-05-17
+%
+
+try
+ s = parse_delay_file(delayfile);
+catch
+ error(lasterr);
+end
+
+delayskip=0;
+noplot=0;
+arg_ptr=1;
+delaypoints=[];
+
+s.sn=unwrap_seqno(s.sn);
+
+while arg_ptr+1 <= nargin
+ switch lower(varargin{arg_ptr})
+ case {'skipdelay', 'delayskip'}
+ % skip a number of seconds in the beginning when calculating delays
+ delayskip = varargin{arg_ptr+1};
+ arg_ptr = arg_ptr + 2;
+ case 'noplot'
+ noplot=1;
+ arg_ptr = arg_ptr + 1;
+ case {'get_delay', 'getdelay'}
+ % return a vector of delay values for the points in the given vector
+ delaypoints = varargin{arg_ptr+1};
+ arg_ptr = arg_ptr + 2;
+ otherwise
+ warning('Unknown switch %s\n', varargin{arg_ptr});
+ arg_ptr = arg_ptr + 1;
+ end
+end
+
+% find lost frames that were covered by one-descriptor decoding
+one_desc_ix=find(isnan(s.arrival));
+for k=1:length(one_desc_ix)
+ ix=find(s.ts==max(s.ts(s.ts(one_desc_ix(k))>s.ts)));
+ s.sn(one_desc_ix(k))=s.sn(ix)+1;
+ s.pt(one_desc_ix(k))=s.pt(ix);
+ s.arrival(one_desc_ix(k))=s.arrival(ix)+s.decode(one_desc_ix(k))-s.decode(ix);
+end
+
+% remove duplicate received frames that were never decoded (RED codec)
+if length(unique(s.ts(isfinite(s.ts)))) < length(s.ts(isfinite(s.ts)))
+ ix=find(isfinite(s.decode));
+ s.sn=s.sn(ix);
+ s.ts=s.ts(ix);
+ s.arrival=s.arrival(ix);
+ s.playout_delay=s.playout_delay(ix);
+ s.pt=s.pt(ix);
+ s.optbuf=s.optbuf(ix);
+ plen=plen(ix);
+ s.decode=s.decode(ix);
+end
+
+% find non-unique sequence numbers
+[~,un_ix]=unique(s.sn);
+nonun_ix=setdiff(1:length(s.sn),un_ix);
+if ~isempty(nonun_ix)
+ warning('RTP sequence numbers are in error');
+end
+
+% sort vectors
+[s.sn,sort_ix]=sort(s.sn);
+s.ts=s.ts(sort_ix);
+s.arrival=s.arrival(sort_ix);
+s.decode=s.decode(sort_ix);
+s.playout_delay=s.playout_delay(sort_ix);
+s.pt=s.pt(sort_ix);
+
+send_t=s.ts-s.ts(1);
+if length(s.fs)<1
+ warning('No info about sample rate found in file. Using default 8000.');
+ s.fs(1)=8000;
+ s.fschange_ts(1)=min(s.ts);
+elseif s.fschange_ts(1)>min(s.ts)
+ s.fschange_ts(1)=min(s.ts);
+end
+
+end_ix=length(send_t);
+for k=length(s.fs):-1:1
+ start_ix=find(s.ts==s.fschange_ts(k));
+ send_t(start_ix:end_ix)=send_t(start_ix:end_ix)/s.fs(k)*1000;
+ s.playout_delay(start_ix:end_ix)=s.playout_delay(start_ix:end_ix)/s.fs(k)*1000;
+ s.optbuf(start_ix:end_ix)=s.optbuf(start_ix:end_ix)/s.fs(k)*1000;
+ end_ix=start_ix-1;
+end
+
+tot_time=max(send_t)-min(send_t);
+
+seq_ix=s.sn-min(s.sn)+1;
+send_t=send_t+max(min(s.arrival-send_t),0);
+
+plot_send_t=nan*ones(max(seq_ix),1);
+plot_send_t(seq_ix)=send_t;
+plot_nw_delay=nan*ones(max(seq_ix),1);
+plot_nw_delay(seq_ix)=s.arrival-send_t;
+
+cng_ix=find(s.pt~=13); % find those packets that are not CNG/SID
+
+if noplot==0
+ h=plot(plot_send_t/1000,plot_nw_delay);
+ set(h,'color',0.75*[1 1 1]);
+ hold on
+ if any(s.optbuf~=0)
+ peak_ix=find(s.optbuf(cng_ix)<0); % peak mode is labeled with negative values
+ no_peak_ix=find(s.optbuf(cng_ix)>0); %setdiff(1:length(cng_ix),peak_ix);
+ h1=plot(send_t(cng_ix(peak_ix))/1000,...
+ s.arrival(cng_ix(peak_ix))+abs(s.optbuf(cng_ix(peak_ix)))-send_t(cng_ix(peak_ix)),...
+ 'r.');
+ h2=plot(send_t(cng_ix(no_peak_ix))/1000,...
+ s.arrival(cng_ix(no_peak_ix))+abs(s.optbuf(cng_ix(no_peak_ix)))-send_t(cng_ix(no_peak_ix)),...
+ 'g.');
+ set([h1, h2],'markersize',1)
+ end
+ %h=plot(send_t(seq_ix)/1000,s.decode+s.playout_delay-send_t(seq_ix));
+ h=plot(send_t(cng_ix)/1000,s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix));
+ set(h,'linew',1.5);
+ hold off
+ ax1=axis;
+ axis tight
+ ax2=axis;
+ axis([ax2(1:3) ax1(4)])
+end
+
+
+% calculate delays and other parameters
+
+delayskip_ix = find(send_t-send_t(1)>=delayskip*1000, 1 );
+
+use_ix = intersect(cng_ix,... % use those that are not CNG/SID frames...
+ intersect(find(isfinite(s.decode)),... % ... that did arrive ...
+ (delayskip_ix:length(s.decode))')); % ... and are sent after delayskip seconds
+
+mean_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-send_t(use_ix));
+neteq_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-s.arrival(use_ix));
+
+Npack=max(s.sn(delayskip_ix:end))-min(s.sn(delayskip_ix:end))+1;
+nw_lossrate=(Npack-length(s.sn(delayskip_ix:end)))/Npack;
+neteq_lossrate=(length(s.sn(delayskip_ix:end))-length(use_ix))/Npack;
+
+delay_struct=struct('mean_delay',mean_delay,'neteq_delay',neteq_delay,...
+ 'nw_lossrate',nw_lossrate,'neteq_lossrate',neteq_lossrate,...
+ 'tot_expand',round(s.tot_expand),'tot_accelerate',round(s.tot_accelerate),...
+ 'tot_preemptive',round(s.tot_preemptive),'tot_time',tot_time,...
+ 'filename',delayfile,'units','ms','fs',unique(s.fs));
+
+if not(isempty(delaypoints))
+ delayvalues=interp1(send_t(cng_ix),...
+ s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix),...
+ delaypoints,'nearest',NaN);
+else
+ delayvalues=[];
+end
+
+
+
+% SUBFUNCTIONS %
+
+function y=unwrap_seqno(x)
+
+jumps=find(abs((diff(x)-1))>65000);
+
+while ~isempty(jumps)
+ n=jumps(1);
+ if x(n+1)-x(n) < 0
+ % negative jump
+ x(n+1:end)=x(n+1:end)+65536;
+ else
+ % positive jump
+ x(n+1:end)=x(n+1:end)-65536;
+ end
+
+ jumps=find(abs((diff(x(n+1:end))-1))>65000);
+end
+
+y=x;
+
+return;
diff --git a/trunk/src/modules/audio_coding/neteq/test/ptypes.txt b/trunk/src/modules/audio_coding/neteq/test/ptypes.txt
new file mode 100644
index 0000000..c3d4e25
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/test/ptypes.txt
@@ -0,0 +1,20 @@
+pcmu 0
+pcma 8
+cn 13
+//ipcmwb 97
+//eg711u 100
+//eg711a 101
+ilbc 102
+isac 103
+isacswb 104
+avt 106
+red 117
+cn_wb 98
+cn_swb32 99
+pcm16b 93
+pcm16b_wb 94
+pcm16b_swb32khz 95
+//pcm16b_swb48khz 96
+//mpeg4aac 122
+g722 9
+celt32 114
diff --git a/trunk/src/modules/audio_coding/neteq/unmute_signal.c b/trunk/src/modules/audio_coding/neteq/unmute_signal.c
new file mode 100644
index 0000000..ee9daa8
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/unmute_signal.c
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This function "unmutes" a vector on a sample by sample basis.
+ */
+
+#include "dsp_helpfunctions.h"
+
+#include "signal_processing_library.h"
+
+
+void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
+ WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
+ WebRtc_Word16 N)
+{
+ int i;
+ WebRtc_UWord16 w16_tmp;
+ WebRtc_Word32 w32_tmp;
+
+ w16_tmp = (WebRtc_UWord16) *startMuteFact;
+ w32_tmp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,6) + 32;
+ for (i = 0; i < N; i++)
+ {
+ pw16_outVec[i]
+ = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp, pw16_inVec[i]) + 8192) >> 14);
+ w32_tmp += unmuteFact;
+ w32_tmp = WEBRTC_SPL_MAX(0, w32_tmp);
+ w16_tmp = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 6); /* 20 - 14 = 6 */
+ w16_tmp = WEBRTC_SPL_MIN(16384, w16_tmp);
+ }
+ *startMuteFact = (WebRtc_Word16) w16_tmp;
+}
+
diff --git a/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c b/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c
new file mode 100644
index 0000000..5e99fd8
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c
@@ -0,0 +1,1654 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Implementation of main NetEQ API.
+ */
+
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_internal.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+#include "neteq_error_codes.h"
+#include "mcu_dsp_common.h"
+#include "rtcp.h"
+
+#define RETURN_ON_ERROR( macroExpr, macroInstPtr ) { \
+ if ((macroExpr) != 0) { \
+ if ((macroExpr) == -1) { \
+ (macroInstPtr)->ErrorCode = - (NETEQ_OTHER_ERROR); \
+ } else { \
+ (macroInstPtr)->ErrorCode = -((WebRtc_Word16) (macroExpr)); \
+ } \
+ return(-1); \
+ } }
+
+int WebRtcNetEQ_strncpy(char *strDest, int numberOfElements,
+ const char *strSource, int count)
+{
+ /* check vector lengths */
+ if (count > numberOfElements)
+ {
+ strDest[0] = '\0';
+ return (-1);
+ }
+ else
+ {
+ strncpy(strDest, strSource, count);
+ return (0);
+ }
+}
+
+/**********************************************************
+ * NETEQ Functions
+ */
+
+/*****************************************
+ * Info functions
+ */
+
+int WebRtcNetEQ_GetVersion(char *version)
+{
+ char versionString[] = "3.3.0\0 ";
+ char endChar[] = " ";
+ int i = 0;
+ while ((versionString[i] != endChar[0]) && (i <= 20))
+ {
+ version[i] = versionString[i]; /* To avoid using strcpy */
+ i++;
+ }
+ return (0);
+}
+
+int WebRtcNetEQ_GetErrorCode(void *inst)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ return (NetEqMainInst->ErrorCode);
+}
+
+int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen)
+{
+ if ((errorName == NULL) || (maxStrLen <= 0))
+ {
+ return (-1);
+ }
+
+ if (errorCode < 0)
+ {
+ errorCode = -errorCode; // absolute value
+ }
+
+ switch (errorCode)
+ {
+ case 1: // could be -1
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "OTHER_ERROR", maxStrLen);
+ break;
+ }
+ case 1001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_INSTRUCTION", maxStrLen);
+ break;
+ }
+ case 1002:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_NETWORK_TYPE", maxStrLen);
+ break;
+ }
+ case 1003:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_DELAYVALUE", maxStrLen);
+ break;
+ }
+ case 1004:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_PLAYOUTMODE", maxStrLen);
+ break;
+ }
+ case 1005:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CORRUPT_INSTANCE", maxStrLen);
+ break;
+ }
+ case 1006:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "ILLEGAL_MASTER_SLAVE_SWITCH", maxStrLen);
+ break;
+ }
+ case 1007:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "MASTER_SLAVE_ERROR", maxStrLen);
+ break;
+ }
+ case 2001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_BUFSTAT_DECISION", maxStrLen);
+ break;
+ }
+ case 2002:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODING", maxStrLen);
+ break;
+ }
+ case 2003:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_SAMPLEUNDERRUN", maxStrLen);
+ break;
+ }
+ case 2004:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODED_TOO_MUCH",
+ maxStrLen);
+ break;
+ }
+ case 3001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_CNG_ERROR", maxStrLen);
+ break;
+ }
+ case 3002:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_UNKNOWNPAYLOAD", maxStrLen);
+ break;
+ }
+ case 3003:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_BUFFERINSERT_ERROR", maxStrLen);
+ break;
+ }
+ case 4001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INIT_ERROR", maxStrLen);
+ break;
+ }
+ case 4002:
+ case 4003:
+ case 4004:
+ case 4005:
+ case 4006:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INSERT_ERROR1", maxStrLen);
+ break;
+ }
+ case 4007:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_G723_HEADER", maxStrLen);
+ break;
+ }
+ case 4008:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NONEXISTING_PACKET", maxStrLen);
+ break;
+ }
+ case 4009:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NOT_INITIALIZED", maxStrLen);
+ break;
+ }
+ case 4010:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "AMBIGUOUS_ILBC_FRAME_SIZE", maxStrLen);
+ break;
+ }
+ case 5001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_FULL", maxStrLen);
+ break;
+ }
+ case 5002:
+ case 5003:
+ case 5004:
+ case 5005:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_NOT_EXIST", maxStrLen);
+ break;
+ }
+ case 5006:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNKNOWN_CODEC", maxStrLen);
+ break;
+ }
+ case 5007:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_PAYLOAD_TAKEN", maxStrLen);
+ break;
+ }
+ case 5008:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_CODEC", maxStrLen);
+ break;
+ }
+ case 5009:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_FS", maxStrLen);
+ break;
+ }
+ case 6001:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_DEC_PARAMETER_ERROR", maxStrLen);
+ break;
+ }
+ case 6002:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_INSERT_ERROR", maxStrLen);
+ break;
+ }
+ case 6003:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_GEN_UNKNOWN_SAMP_FREQ", maxStrLen);
+ break;
+ }
+ case 6004:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_NOT_SUPPORTED", maxStrLen);
+ break;
+ }
+ case 7001:
+ case 7002:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RED_SPLIT_ERROR", maxStrLen);
+ break;
+ }
+ case 7003:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_TOO_SHORT_PACKET", maxStrLen);
+ break;
+ }
+ case 7004:
+ {
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_CORRUPT_PACKET", maxStrLen);
+ break;
+ }
+ default:
+ {
+ /* check for decoder error ranges */
+ if (errorCode >= 6010 && errorCode <= 6810)
+ {
+ /* iSAC error code */
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "iSAC ERROR", maxStrLen);
+ break;
+ }
+
+ WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_ERROR", maxStrLen);
+ return (-1);
+ }
+ }
+
+ return (0);
+}
+
+/* Assign functions (create not allowed in order to avoid malloc in lib) */
+int WebRtcNetEQ_AssignSize(int *sizeinbytes)
+{
+ *sizeinbytes = (sizeof(MainInst_t) * 2) / sizeof(WebRtc_Word16);
+ return (0);
+}
+
+int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) NETEQ_inst_Addr;
+ *inst = NETEQ_inst_Addr;
+ if (*inst == NULL) return (-1);
+ /* Clear memory */
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) NetEqMainInst, 0,
+ (sizeof(MainInst_t) / sizeof(WebRtc_Word16)));
+ ok = WebRtcNetEQ_McuReset(&NetEqMainInst->MCUinst);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (0);
+}
+
+int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec,
+ int noOfCodecs, enum WebRtcNetEQNetworkType nwType,
+ int *MaxNoOfPackets, int *sizeinbytes)
+{
+ int ok = 0;
+ int multiplier;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ *MaxNoOfPackets = 0;
+ *sizeinbytes = 0;
+ ok = WebRtcNetEQ_GetDefaultCodecSettings(codec, noOfCodecs, sizeinbytes, MaxNoOfPackets);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ if (nwType == kUDPNormal)
+ {
+ multiplier = 1;
+ }
+ else if (nwType == kUDPVideoSync)
+ {
+ multiplier = 4;
+ }
+ else if (nwType == kTCPNormal)
+ {
+ multiplier = 4;
+ }
+ else if (nwType == kTCPLargeJitter)
+ {
+ multiplier = 8;
+ }
+ else if (nwType == kTCPXLargeJitter)
+ {
+ multiplier = 20;
+ }
+ else
+ {
+ NetEqMainInst->ErrorCode = -FAULTY_NETWORK_TYPE;
+ return (-1);
+ }
+ *MaxNoOfPackets = (*MaxNoOfPackets) * multiplier;
+ *sizeinbytes = (*sizeinbytes) * multiplier;
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr,
+ int sizeinbytes)
+{
+ int ok;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ ok = WebRtcNetEQ_PacketBufferInit(&NetEqMainInst->MCUinst.PacketBuffer_inst,
+ MaxNoOfPackets, (WebRtc_Word16*) NETEQ_Buffer_Addr, (sizeinbytes >> 1));
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+/************************************************
+ * Init functions
+ */
+
+/****************************************************************************
+ * WebRtcNetEQ_Init(...)
+ *
+ * Initialize NetEQ.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - fs : Initial sample rate in Hz (may change with payload)
+ *
+ * Output:
+ * - inst : Initialized NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs)
+{
+ int ok = 0;
+
+ /* Typecast inst to internal instance format */
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+#ifdef NETEQ_VAD
+ /* Start out with no PostDecode VAD instance */
+ NetEqMainInst->DSPinst.VADInst.VADState = NULL;
+ /* Also set all VAD function pointers to NULL */
+ NetEqMainInst->DSPinst.VADInst.initFunction = NULL;
+ NetEqMainInst->DSPinst.VADInst.setmodeFunction = NULL;
+ NetEqMainInst->DSPinst.VADInst.VADFunction = NULL;
+#endif /* NETEQ_VAD */
+
+ ok = WebRtcNetEQ_DSPinit(NetEqMainInst); /* Init addresses between MCU and DSP */
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+
+ ok = WebRtcNetEQ_DSPInit(&NetEqMainInst->DSPinst, fs); /* Init dsp side */
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+ /* set BGN mode to default, since it is not cleared by DSP init function */
+ NetEqMainInst->DSPinst.BGNInst.bgnMode = BGN_ON;
+
+ /* init statistics functions and counters */
+ ok = WebRtcNetEQ_ClearInCallStats(&NetEqMainInst->DSPinst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+ ok = WebRtcNetEQ_ClearPostCallStats(&NetEqMainInst->DSPinst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+ ok = WebRtcNetEQ_ResetMcuJitterStat(&NetEqMainInst->MCUinst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+
+ /* flush packet buffer */
+ ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+
+ /* set some variables to initial values */
+ NetEqMainInst->MCUinst.current_Codec = -1;
+ NetEqMainInst->MCUinst.current_Payload = -1;
+ NetEqMainInst->MCUinst.first_packet = 1;
+ NetEqMainInst->MCUinst.one_desc = 0;
+ NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = 0;
+ NetEqMainInst->MCUinst.NoOfExpandCalls = 0;
+ NetEqMainInst->MCUinst.fs = fs;
+
+#ifdef NETEQ_ATEVENT_DECODE
+ /* init DTMF decoder */
+ ok = WebRtcNetEQ_DtmfDecoderInit(&(NetEqMainInst->MCUinst.DTMF_inst),fs,560);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+#endif
+
+ /* init RTCP statistics */
+ WebRtcNetEQ_RTCPInit(&(NetEqMainInst->MCUinst.RTCP_inst), 0);
+
+ /* set BufferStat struct to zero */
+ WebRtcSpl_MemSetW16((WebRtc_Word16*) &(NetEqMainInst->MCUinst.BufferStat_inst), 0,
+ sizeof(BufstatsInst_t) / sizeof(WebRtc_Word16));
+
+ /* reset automode */
+ WebRtcNetEQ_ResetAutomode(&(NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst),
+ NetEqMainInst->MCUinst.PacketBuffer_inst.maxInsertPositions);
+
+ NetEqMainInst->ErrorCode = 0;
+
+#ifdef NETEQ_STEREO
+ /* set master/slave info to undecided */
+ NetEqMainInst->masterSlave = 0;
+#endif
+
+ return (ok);
+}
+
+int WebRtcNetEQ_FlushBuffers(void *inst)
+{
+ int ok = 0;
+
+ /* Typecast inst to internal instance format */
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+ /* Flush packet buffer */
+ ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+
+ /* Set MCU to wait for new codec */
+ NetEqMainInst->MCUinst.first_packet = 1;
+
+ /* Flush speech buffer */
+ ok = WebRtcNetEQ_FlushSpeechBuffer(&NetEqMainInst->DSPinst);
+ RETURN_ON_ERROR(ok, NetEqMainInst);
+
+ return 0;
+}
+
+int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+#ifdef NETEQ_ATEVENT_DECODE
+ NetEqMainInst->MCUinst.AVT_PlayoutOn = PlayoutAVTon;
+ return(0);
+#else
+ if (PlayoutAVTon != 0)
+ {
+ NetEqMainInst->ErrorCode = -DTMF_NOT_SUPPORTED;
+ return (-1);
+ }
+ else
+ {
+ return (0);
+ }
+#endif
+}
+
+int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ if ((DelayInMs < 0) || (DelayInMs > 1000))
+ {
+ NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE;
+ return (-1);
+ }
+ NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = DelayInMs;
+ return (0);
+}
+
+int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ if ((playoutMode != kPlayoutOn) && (playoutMode != kPlayoutOff) && (playoutMode
+ != kPlayoutFax) && (playoutMode != kPlayoutStreaming))
+ {
+ NetEqMainInst->ErrorCode = -FAULTY_PLAYOUTMODE;
+ return (-1);
+ }
+ else
+ {
+ NetEqMainInst->MCUinst.NetEqPlayoutMode = playoutMode;
+ return (0);
+ }
+}
+
+int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode)
+{
+
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ /* Instance sanity */
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+ NetEqMainInst->DSPinst.BGNInst.bgnMode = (enum BGNMode) bgnMode;
+
+ return (0);
+}
+
+int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode)
+{
+
+ const MainInst_t *NetEqMainInst = (const MainInst_t*) inst;
+
+ /* Instance sanity */
+ if (NetEqMainInst == NULL) return (-1);
+
+ *bgnMode = (enum WebRtcNetEQBGNMode) NetEqMainInst->DSPinst.BGNInst.bgnMode;
+
+ return (0);
+}
+
+/************************************************
+ * CodecDB functions
+ */
+
+int WebRtcNetEQ_CodecDbReset(void *inst)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ ok = WebRtcNetEQ_DbReset(&NetEqMainInst->MCUinst.codec_DB_inst);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+
+ /* set function pointers to NULL to prevent RecOut from using the codec */
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL;
+
+ return (0);
+}
+
+int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
+ WebRtc_Word16 *MaxEntries)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ *MaxEntries = NUM_CODECS;
+ *UsedEntries = NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs;
+ return (0);
+}
+
+int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
+ enum WebRtcNetEQDecoder *codec)
+{
+ int i;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ *codec = (enum WebRtcNetEQDecoder) 0;
+ if ((Entry >= 0) && (Entry < NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs))
+ {
+ for (i = 0; i < NUM_TOTAL_CODECS; i++)
+ {
+ if (NetEqMainInst->MCUinst.codec_DB_inst.position[i] == Entry)
+ {
+ *codec = (enum WebRtcNetEQDecoder) i;
+ }
+ }
+ }
+ else
+ {
+ NetEqMainInst->ErrorCode = -(CODEC_DB_NOT_EXIST1);
+ return (-1);
+ }
+ return (0);
+}
+
+int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ ok = WebRtcNetEQ_DbAdd(&NetEqMainInst->MCUinst.codec_DB_inst, codecInst->codec,
+ codecInst->payloadType, codecInst->funcDecode, codecInst->funcDecodeRCU,
+ codecInst->funcDecodePLC, codecInst->funcDecodeInit, codecInst->funcAddLatePkt,
+ codecInst->funcGetMDinfo, codecInst->funcGetPitch, codecInst->funcUpdBWEst,
+ codecInst->funcGetErrorCode, codecInst->codec_state, codecInst->codec_fs);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* check if currently used codec is being removed */
+ if (NetEqMainInst->MCUinst.current_Codec == (WebRtc_Word16) codec)
+ {
+ /* set function pointers to NULL to prevent RecOut from using the codec */
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL;
+ NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL;
+ }
+
+ ok = WebRtcNetEQ_DbRemove(&NetEqMainInst->MCUinst.codec_DB_inst, codec);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+/*********************************
+ * Real-time functions
+ */
+
+int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
+ WebRtc_UWord32 uw32_timeRec)
+{
+ int ok = 0;
+ RTPPacket_t RTPpacket;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+ /* Parse RTP header */
+ ok = WebRtcNetEQ_RTPPayloadInfo(p_w16datagramstart, w16_RTPlen, &RTPpacket);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+
+ ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_RecInRTPStruct(...)
+ *
+ * Alternative RecIn function, used when the RTP data has already been
+ * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo).
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - rtpInfo : Pointer to RTP info
+ * - payloadPtr : Pointer to the RTP payload (first byte after header)
+ * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr
+ * - timeRec : Receive time (in timestamps of the used codec)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
+ const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
+ WebRtc_UWord32 uw32_timeRec)
+{
+ int ok = 0;
+ RTPPacket_t RTPpacket;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+ /* Load NetEQ's RTP struct from Module RTP struct */
+ RTPpacket.payloadType = rtpInfo->payloadType;
+ RTPpacket.seqNumber = rtpInfo->sequenceNumber;
+ RTPpacket.timeStamp = rtpInfo->timeStamp;
+ RTPpacket.ssrc = rtpInfo->SSRC;
+ RTPpacket.payload = (const WebRtc_Word16*) payloadPtr;
+ RTPpacket.payloadLen = payloadLenBytes;
+ RTPpacket.starts_byte1 = 0;
+
+ ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo msInfo;
+ msInfo.msMode = NETEQ_MONO;
+#endif
+
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+#ifdef NETEQ_STEREO
+ NetEqMainInst->DSPinst.msInfo = &msInfo;
+#endif
+
+ ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
+ pw16_len, 0 /* not BGN only */);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_RecOutMasterSlave(...)
+ *
+ * RecOut function for running several NetEQ instances in master/slave mode.
+ * One master can be used to control several slaves.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - isMaster : Non-zero indicates that this is the master channel
+ * - msInfo : (slave only) Information from master
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ * - pw16_outData : Pointer to vector where output should be written
+ * - pw16_len : Pointer to variable where output length is returned
+ * - msInfo : (master only) Information to slave(s)
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len, void *msInfo,
+ WebRtc_Word16 isMaster)
+{
+#ifndef NETEQ_STEREO
+ /* Stereo not supported */
+ return(-1);
+#else
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+ if (msInfo == NULL)
+ {
+ /* msInfo not provided */
+ NetEqMainInst->ErrorCode = NETEQ_OTHER_ERROR;
+ return (-1);
+ }
+
+ /* translate from external to internal Master/Slave information */
+ NetEqMainInst->DSPinst.msInfo = (MasterSlaveInfo *) msInfo;
+
+ /* check that we have not done a master/slave switch without first re-initializing */
+ if ((NetEqMainInst->masterSlave == 1 && !isMaster) || /* switch from master to slave */
+ (NetEqMainInst->masterSlave == 2 && isMaster)) /* switch from slave to master */
+ {
+ NetEqMainInst->ErrorCode = ILLEGAL_MASTER_SLAVE_SWITCH;
+ return (-1);
+ }
+
+ if (!isMaster)
+ {
+ /* this is the slave */
+ NetEqMainInst->masterSlave = 2;
+ NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_SLAVE;
+ }
+ else
+ {
+ NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_MASTER;
+ }
+
+ ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
+ pw16_len, 0 /* not BGN only */);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+
+ if (isMaster)
+ {
+ /* this is the master */
+ NetEqMainInst->masterSlave = 1;
+ }
+
+ return (ok);
+#endif
+}
+
+int WebRtcNetEQ_GetMasterSlaveInfoSize()
+{
+#ifdef NETEQ_STEREO
+ return (sizeof(MasterSlaveInfo));
+#else
+ return(-1);
+#endif
+}
+
+/* Special RecOut that does not do any decoding. */
+int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
+ WebRtc_Word16 *pw16_len)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+#ifdef NETEQ_STEREO
+ MasterSlaveInfo msInfo;
+#endif
+
+ if (NetEqMainInst == NULL) return (-1);
+
+ /* Check for corrupt/cleared instance */
+ if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
+ {
+ /* Instance is corrupt */
+ NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
+ return (-1);
+ }
+
+#ifdef NETEQ_STEREO
+ /* keep same mode as before */
+ switch (NetEqMainInst->masterSlave)
+ {
+ case 1:
+ {
+ msInfo.msMode = NETEQ_MASTER;
+ break;
+ }
+ case 2:
+ {
+ msInfo.msMode = NETEQ_SLAVE;
+ break;
+ }
+ default:
+ {
+ msInfo.msMode = NETEQ_MONO;
+ break;
+ }
+ }
+
+ NetEqMainInst->DSPinst.msInfo = &msInfo;
+#endif
+
+ ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
+ pw16_len, 1 /* BGN only */);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst,
+ &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max,
+ &RTCP_inst->jitter, 0);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst)
+{
+ int ok = 0;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst,
+ &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max,
+ &RTCP_inst->jitter, 1);
+ if (ok != 0)
+ {
+ NetEqMainInst->ErrorCode = -ok;
+ return (-1);
+ }
+ return (ok);
+}
+
+int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+
+ if (NetEqMainInst->MCUinst.TSscalingInitialized)
+ {
+ *timestamp = WebRtcNetEQ_ScaleTimestampInternalToExternal(&NetEqMainInst->MCUinst,
+ NetEqMainInst->DSPinst.videoSyncTimestamp);
+ }
+ else
+ {
+ *timestamp = NetEqMainInst->DSPinst.videoSyncTimestamp;
+ }
+
+ return (0);
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_GetSpeechOutputType(...)
+ *
+ * Get the output type for the audio provided by the latest call to
+ * WebRtcNetEQ_RecOut().
+ *
+ * kOutputNormal = normal audio (possibly processed)
+ * kOutputPLC = loss concealment through stretching audio
+ * kOutputCNG = comfort noise (codec-internal or RFC3389)
+ * kOutputPLCtoCNG = background noise only due to long expand or error
+ * kOutputVADPassive = PostDecode VAD signalling passive speaker
+ *
+ * Input:
+ * - inst : NetEQ instance
+ *
+ * Output:
+ * - outputType : Output type from enum list WebRtcNetEQOutputType
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType)
+{
+ /* Typecast to internal instance type */
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+ if ((NetEqMainInst->DSPinst.w16_mode & MODE_BGN_ONLY) != 0)
+ {
+ /* If last mode was background noise only */
+ *outputType = kOutputPLCtoCNG;
+
+ }
+ else if ((NetEqMainInst->DSPinst.w16_mode == MODE_CODEC_INTERNAL_CNG)
+ || (NetEqMainInst->DSPinst.w16_mode == MODE_RFC3389CNG))
+ {
+ /* If CN or internal CNG */
+ *outputType = kOutputCNG;
+
+#ifdef NETEQ_VAD
+ }
+ else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
+ {
+ /* post-decode VAD says passive speaker */
+ *outputType = kOutputVADPassive;
+#endif /* NETEQ_VAD */
+
+ }
+ else if ((NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
+ && (NetEqMainInst->DSPinst.ExpandInst.w16_expandMuteFactor == 0))
+ {
+ /* Expand mode has faded down to background noise only (very long expand) */
+ *outputType = kOutputPLCtoCNG;
+
+ }
+ else if (NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
+ {
+ /* PLC mode */
+ *outputType = kOutputPLC;
+
+ }
+ else
+ {
+ /* Normal speech output type (can still be manipulated, e.g., accelerated) */
+ *outputType = kOutputNormal;
+ }
+
+ return (0);
+}
+
+/**********************************
+ * Functions related to VQmon
+ */
+
+#define WEBRTC_NETEQ_CONCEALMENTFLAG_LOST 0x01
+#define WEBRTC_NETEQ_CONCEALMENTFLAG_DISCARDED 0x02
+#define WEBRTC_NETEQ_CONCEALMENTFLAG_SUPRESS 0x04
+#define WEBRTC_NETEQ_CONCEALMENTFLAG_CNGACTIVE 0x80
+
+int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
+ WebRtc_UWord16 *concealedVoiceDurationMs,
+ WebRtc_UWord8 *concealedVoiceFlags)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ WebRtc_Word16 fs_mult;
+ WebRtc_Word16 ms_lost;
+ if (NetEqMainInst == NULL) return (-1);
+ fs_mult = WebRtcSpl_DivW32W16ResW16(NetEqMainInst->MCUinst.fs, 8000);
+
+ ms_lost = WebRtcSpl_DivW32W16ResW16(
+ (WebRtc_Word32) NetEqMainInst->DSPinst.w16_concealedTS, (WebRtc_Word16) (8 * fs_mult));
+ if (ms_lost > NetEqMainInst->DSPinst.millisecondsPerCall) ms_lost
+ = NetEqMainInst->DSPinst.millisecondsPerCall;
+
+ *validVoiceDurationMs = NetEqMainInst->DSPinst.millisecondsPerCall - ms_lost;
+ *concealedVoiceDurationMs = ms_lost;
+ if (ms_lost > 0)
+ {
+ *concealedVoiceFlags = WEBRTC_NETEQ_CONCEALMENTFLAG_LOST;
+ }
+ else
+ {
+ *concealedVoiceFlags = 0;
+ }
+ NetEqMainInst->DSPinst.w16_concealedTS -= ms_lost * (8 * fs_mult);
+
+ return (0);
+}
+
+int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
+ WebRtc_UWord8 *adaptationRate)
+{
+ /* Dummy check the inst, just to avoid compiler warnings. */
+ if (inst == NULL)
+ {
+ /* Do nothing. */
+ }
+
+ /* Hardcoded variables that are used for VQmon as jitter buffer parameters */
+ *absMaxDelayMs = 240;
+ *adaptationRate = 1;
+ return (0);
+}
+
+int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
+ WebRtc_UWord16 *maxDelayMs)
+{
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL) return (-1);
+ *avgDelayMs = (WebRtc_UWord16) (NetEqMainInst->MCUinst.BufferStat_inst.avgDelayMsQ8 >> 8);
+ *maxDelayMs = (WebRtc_UWord16) NetEqMainInst->MCUinst.BufferStat_inst.maxDelayMs;
+ return (0);
+}
+
+/*************************************
+ * Statistics functions
+ */
+
+/* Get the "in-call" statistics from NetEQ.
+ * The statistics are reset after the query. */
+int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats)
+
+{
+
+ WebRtc_UWord16 tempU16;
+ WebRtc_UWord32 tempU32, tempU32_2;
+ int numShift;
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+
+ /* Instance sanity */
+ if (NetEqMainInst == NULL) return (-1);
+
+ /*******************/
+ /* Get buffer size */
+ /*******************/
+
+ if (NetEqMainInst->MCUinst.fs != 0)
+ {
+ WebRtc_Word32 temp32;
+ /* Query packet buffer for number of samples. */
+ temp32 = WebRtcNetEQ_PacketBufferGetSize(
+ &NetEqMainInst->MCUinst.PacketBuffer_inst);
+
+ /* Divide by sample rate.
+ * Calculate temp32 * 1000 / fs to get result in ms. */
+ stats->currentBufferSize = (WebRtc_UWord16)
+ WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
+
+ /* Add number of samples yet to play in sync buffer. */
+ temp32 = (WebRtc_Word32) (NetEqMainInst->DSPinst.endPosition -
+ NetEqMainInst->DSPinst.curPosition);
+ stats->currentBufferSize += (WebRtc_UWord16)
+ WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
+ }
+ else
+ {
+ /* Sample rate not initialized. */
+ stats->currentBufferSize = 0;
+ }
+
+ /***************************/
+ /* Get optimal buffer size */
+ /***************************/
+
+ if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
+ {
+ /* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
+ stats->preferredBufferSize
+ = (WebRtc_UWord16) WEBRTC_SPL_MUL_16_16(
+ (WebRtc_Word16) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */
+ WebRtcSpl_DivW32W16ResW16(
+ (WebRtc_Word32) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */
+ WebRtcSpl_DivW32W16ResW16( (WebRtc_Word32) NetEqMainInst->MCUinst.fs, (WebRtc_Word16) 1000 ) /* samples per ms */
+ ) );
+
+ /* add extra delay */
+ if (NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs > 0)
+ {
+ stats->preferredBufferSize
+ += NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs;
+ }
+ }
+ else
+ {
+ /* sample rate not initialized */
+ stats->preferredBufferSize = 0;
+ }
+
+ /***********************************/
+ /* Check if jitter peaks are found */
+ /***********************************/
+
+ stats->jitterPeaksFound =
+ NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.peakFound;
+
+ /***********************/
+ /* Calculate loss rate */
+ /***********************/
+
+ /* timestamps elapsed since last report */
+ tempU32 = NetEqMainInst->MCUinst.lastReportTS;
+
+ if (NetEqMainInst->MCUinst.lostTS == 0)
+ {
+ /* no losses */
+ stats->currentPacketLossRate = 0;
+ }
+ else if (NetEqMainInst->MCUinst.lostTS < tempU32)
+ {
+ /* calculate shifts; we want the result in Q14 */
+ numShift = WebRtcSpl_NormU32(NetEqMainInst->MCUinst.lostTS); /* numerator shift for normalize */
+
+ if (numShift < 14)
+ {
+ /* cannot shift numerator 14 steps; shift denominator too */
+ tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
+ }
+ else
+ {
+ /* shift no more than 14 steps */
+ numShift = 14;
+ }
+
+ if (tempU32 == 0)
+ {
+ /* check for zero denominator; result should be zero in this case */
+ stats->currentPacketLossRate = 0;
+ }
+ else
+ {
+ /* check that denominator fits in signed 16-bit */
+ while (tempU32 > WEBRTC_SPL_WORD16_MAX)
+ {
+ tempU32 >>= 1; /* right-shift 1 step */
+ numShift--; /* compensate in numerator */
+ }
+ tempU16 = (WebRtc_UWord16) tempU32;
+
+ /* do the shift of numerator */
+ tempU32
+ = WEBRTC_SPL_SHIFT_W32( (WebRtc_UWord32) NetEqMainInst->MCUinst.lostTS, numShift);
+
+ stats->currentPacketLossRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+ tempU16);
+ }
+ }
+ else
+ {
+ /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
+ /* set loss rate = 1 */
+ stats->currentPacketLossRate = 1 << 14; /* 1 in Q14 */
+ }
+
+ /**************************/
+ /* Calculate discard rate */
+ /**************************/
+
+ /* timestamps elapsed since last report */
+ tempU32 = NetEqMainInst->MCUinst.lastReportTS;
+
+ /* number of discarded samples */
+ tempU32_2
+ = WEBRTC_SPL_MUL_16_U16( (WebRtc_Word16) NetEqMainInst->MCUinst.PacketBuffer_inst.packSizeSamples,
+ NetEqMainInst->MCUinst.PacketBuffer_inst.discardedPackets);
+
+ if (tempU32_2 == 0)
+ {
+ /* no discarded samples */
+ stats->currentDiscardRate = 0;
+ }
+ else if (tempU32_2 < tempU32)
+ {
+ /* calculate shifts; we want the result in Q14 */
+ numShift = WebRtcSpl_NormU32(tempU32_2); /* numerator shift for normalize */
+
+ if (numShift < 14)
+ {
+ /* cannot shift numerator 14 steps; shift denominator too */
+ tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
+ }
+ else
+ {
+ /* shift no more than 14 steps */
+ numShift = 14;
+ }
+
+ if (tempU32 == 0)
+ {
+ /* check for zero denominator; result should be zero in this case */
+ stats->currentDiscardRate = 0;
+ }
+ else
+ {
+ /* check that denominator fits in signed 16-bit */
+ while (tempU32 > WEBRTC_SPL_WORD16_MAX)
+ {
+ tempU32 >>= 1; /* right-shift 1 step */
+ numShift--; /* compensate in numerator */
+ }
+ tempU16 = (WebRtc_UWord16) tempU32;
+
+ /* do the shift of numerator */
+ tempU32 = WEBRTC_SPL_SHIFT_W32( tempU32_2, numShift);
+
+ stats->currentDiscardRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
+ }
+ }
+ else
+ {
+ /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
+ /* set loss rate = 1 */
+ stats->currentDiscardRate = 1 << 14; /* 1 in Q14 */
+ }
+
+ /*************************************************************/
+ /* Calculate Accelerate, Expand and Pre-emptive Expand rates */
+ /*************************************************************/
+
+ /* timestamps elapsed since last report */
+ tempU32 = NetEqMainInst->MCUinst.lastReportTS;
+
+ if (NetEqMainInst->DSPinst.statInst.accelerateLength == 0)
+ {
+ /* no accelerate */
+ stats->currentAccelerateRate = 0;
+ }
+ else if (NetEqMainInst->DSPinst.statInst.accelerateLength < tempU32)
+ {
+ /* calculate shifts; we want the result in Q14 */
+ numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.accelerateLength); /* numerator shift for normalize */
+
+ if (numShift < 14)
+ {
+ /* cannot shift numerator 14 steps; shift denominator too */
+ tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
+ }
+ else
+ {
+ /* shift no more than 14 steps */
+ numShift = 14;
+ }
+
+ if (tempU32 == 0)
+ {
+ /* check for zero denominator; result should be zero in this case */
+ stats->currentAccelerateRate = 0;
+ }
+ else
+ {
+ /* check that denominator fits in signed 16-bit */
+ while (tempU32 > WEBRTC_SPL_WORD16_MAX)
+ {
+ tempU32 >>= 1; /* right-shift 1 step */
+ numShift--; /* compensate in numerator */
+ }
+ tempU16 = (WebRtc_UWord16) tempU32;
+
+ /* do the shift of numerator */
+ tempU32
+ = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.accelerateLength, numShift);
+
+ stats->currentAccelerateRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+ tempU16);
+ }
+ }
+ else
+ {
+ /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
+ /* set loss rate = 1 */
+ stats->currentAccelerateRate = 1 << 14; /* 1 in Q14 */
+ }
+
+ /* timestamps elapsed since last report */
+ tempU32 = NetEqMainInst->MCUinst.lastReportTS;
+
+ if (NetEqMainInst->DSPinst.statInst.expandLength == 0)
+ {
+ /* no expand */
+ stats->currentExpandRate = 0;
+ }
+ else if (NetEqMainInst->DSPinst.statInst.expandLength < tempU32)
+ {
+ /* calculate shifts; we want the result in Q14 */
+ numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.expandLength); /* numerator shift for normalize */
+
+ if (numShift < 14)
+ {
+ /* cannot shift numerator 14 steps; shift denominator too */
+ tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
+ }
+ else
+ {
+ /* shift no more than 14 steps */
+ numShift = 14;
+ }
+
+ if (tempU32 == 0)
+ {
+ /* check for zero denominator; result should be zero in this case */
+ stats->currentExpandRate = 0;
+ }
+ else
+ {
+ /* check that denominator fits in signed 16-bit */
+ while (tempU32 > WEBRTC_SPL_WORD16_MAX)
+ {
+ tempU32 >>= 1; /* right-shift 1 step */
+ numShift--; /* compensate in numerator */
+ }
+ tempU16 = (WebRtc_UWord16) tempU32;
+
+ /* do the shift of numerator */
+ tempU32
+ = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.expandLength, numShift);
+
+ stats->currentExpandRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
+ }
+ }
+ else
+ {
+ /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
+ /* set loss rate = 1 */
+ stats->currentExpandRate = 1 << 14; /* 1 in Q14 */
+ }
+
+ /* timestamps elapsed since last report */
+ tempU32 = NetEqMainInst->MCUinst.lastReportTS;
+
+ if (NetEqMainInst->DSPinst.statInst.preemptiveLength == 0)
+ {
+ /* no pre-emptive expand */
+ stats->currentPreemptiveRate = 0;
+ }
+ else if (NetEqMainInst->DSPinst.statInst.preemptiveLength < tempU32)
+ {
+ /* calculate shifts; we want the result in Q14 */
+ numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.preemptiveLength); /* numerator shift for normalize */
+
+ if (numShift < 14)
+ {
+ /* cannot shift numerator 14 steps; shift denominator too */
+ tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
+ }
+ else
+ {
+ /* shift no more than 14 steps */
+ numShift = 14;
+ }
+
+ if (tempU32 == 0)
+ {
+ /* check for zero denominator; result should be zero in this case */
+ stats->currentPreemptiveRate = 0;
+ }
+ else
+ {
+ /* check that denominator fits in signed 16-bit */
+ while (tempU32 > WEBRTC_SPL_WORD16_MAX)
+ {
+ tempU32 >>= 1; /* right-shift 1 step */
+ numShift--; /* compensate in numerator */
+ }
+ tempU16 = (WebRtc_UWord16) tempU32;
+
+ /* do the shift of numerator */
+ tempU32
+ = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.preemptiveLength, numShift);
+
+ stats->currentPreemptiveRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
+ tempU16);
+ }
+ }
+ else
+ {
+ /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
+ /* set loss rate = 1 */
+ stats->currentPreemptiveRate = 1 << 14; /* 1 in Q14 */
+ }
+
+ stats->clockDriftPPM = WebRtcNetEQ_AverageIAT(
+ &NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst);
+
+ /* reset counters */
+ WebRtcNetEQ_ResetMcuInCallStats(&(NetEqMainInst->MCUinst));
+ WebRtcNetEQ_ClearInCallStats(&(NetEqMainInst->DSPinst));
+
+ return (0);
+}
+
+int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst,
+ int max_length,
+ int* waiting_times_ms) {
+ int i = 0;
+ MainInst_t *main_inst = (MainInst_t*) inst;
+ if (main_inst == NULL) return -1;
+
+ while ((i < max_length) && (i < main_inst->MCUinst.len_waiting_times)) {
+ waiting_times_ms[i] = main_inst->MCUinst.waiting_times[i] *
+ main_inst->DSPinst.millisecondsPerCall;
+ ++i;
+ }
+ assert(i <= kLenWaitingTimes);
+ WebRtcNetEQ_ResetWaitingTimeStats(&main_inst->MCUinst);
+ return i;
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADInstance(...)
+ *
+ * Provide a pointer to an allocated VAD instance. If function is never
+ * called or it is called with NULL pointer as VAD_inst, the post-decode
+ * VAD functionality is disabled. Also provide pointers to init, setmode
+ * and VAD functions. These are typically pointers to WebRtcVad_Init,
+ * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the
+ * interface file webrtc_vad.h.
+ *
+ * Input:
+ * - NetEQ_inst : NetEQ instance
+ * - VADinst : VAD instance
+ * - initFunction : Pointer to VAD init function
+ * - setmodeFunction : Pointer to VAD setmode function
+ * - VADfunction : Pointer to VAD function
+ *
+ * Output:
+ * - NetEQ_inst : Updated NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst,
+ WebRtcNetEQ_VADInitFunction initFunction,
+ WebRtcNetEQ_VADSetmodeFunction setmodeFunction,
+ WebRtcNetEQ_VADFunction VADFunction)
+{
+
+ /* Typecast to internal instance type */
+ MainInst_t *NetEqMainInst = (MainInst_t*) NetEQ_inst;
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+#ifdef NETEQ_VAD
+
+ /* Store pointer in PostDecode VAD struct */
+ NetEqMainInst->DSPinst.VADInst.VADState = VAD_inst;
+
+ /* Store function pointers */
+ NetEqMainInst->DSPinst.VADInst.initFunction = initFunction;
+ NetEqMainInst->DSPinst.VADInst.setmodeFunction = setmodeFunction;
+ NetEqMainInst->DSPinst.VADInst.VADFunction = VADFunction;
+
+ /* Call init function and return the result (ok or fail) */
+ return(WebRtcNetEQ_InitVAD(&NetEqMainInst->DSPinst.VADInst, NetEqMainInst->DSPinst.fs));
+
+#else /* NETEQ_VAD not defined */
+ return (-1);
+#endif /* NETEQ_VAD */
+
+}
+
+/****************************************************************************
+ * WebRtcNetEQ_SetVADMode(...)
+ *
+ * Pass an aggressiveness mode parameter to the post-decode VAD instance.
+ * If this function is never called, mode 0 (quality mode) is used as default.
+ *
+ * Input:
+ * - inst : NetEQ instance
+ * - mode : mode parameter (same range as WebRtc VAD mode)
+ *
+ * Output:
+ * - inst : Updated NetEQ instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcNetEQ_SetVADMode(void *inst, int mode)
+{
+
+ /* Typecast to internal instance type */
+ MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+ if (NetEqMainInst == NULL)
+ {
+ return (-1);
+ }
+
+#ifdef NETEQ_VAD
+
+ /* Set mode and return result */
+ return(WebRtcNetEQ_SetVADModeInternal(&NetEqMainInst->DSPinst.VADInst, mode));
+
+#else /* NETEQ_VAD not defined */
+ return (-1);
+#endif /* NETEQ_VAD */
+
+}
diff --git a/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc b/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
new file mode 100644
index 0000000..fbb9cc2
--- /dev/null
+++ b/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
@@ -0,0 +1,605 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes unit tests for NetEQ.
+ */
+
+#include <stdlib.h>
+#include <string.h> // memset
+
+#include <string>
+#include <vector>
+
+#include "gtest/gtest.h"
+
+#include "modules/audio_coding/neteq/interface/webrtc_neteq.h"
+#include "modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
+#include "modules/audio_coding/neteq/interface/webrtc_neteq_internal.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+#include "testsupport/fileutils.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+class RefFiles {
+ public:
+ RefFiles(const std::string& input_file, const std::string& output_file);
+ ~RefFiles();
+ template<class T> void ProcessReference(const T& test_results);
+ template<typename T, size_t n> void ProcessReference(
+ const T (&test_results)[n],
+ size_t length);
+ template<typename T, size_t n> void WriteToFile(
+ const T (&test_results)[n],
+ size_t length);
+ template<typename T, size_t n> void ReadFromFileAndCompare(
+ const T (&test_results)[n],
+ size_t length);
+ void WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats);
+ void ReadFromFileAndCompare(const WebRtcNetEQ_NetworkStatistics& stats);
+ void WriteToFile(const WebRtcNetEQ_RTCPStat& stats);
+ void ReadFromFileAndCompare(const WebRtcNetEQ_RTCPStat& stats);
+
+ FILE* input_fp_;
+ FILE* output_fp_;
+};
+
+RefFiles::RefFiles(const std::string &input_file,
+ const std::string &output_file)
+ : input_fp_(NULL),
+ output_fp_(NULL) {
+ if (!input_file.empty()) {
+ input_fp_ = fopen(input_file.c_str(), "rb");
+ EXPECT_TRUE(input_fp_ != NULL);
+ }
+ if (!output_file.empty()) {
+ output_fp_ = fopen(output_file.c_str(), "wb");
+ EXPECT_TRUE(output_fp_ != NULL);
+ }
+}
+
+RefFiles::~RefFiles() {
+ if (input_fp_) {
+ EXPECT_EQ(EOF, fgetc(input_fp_)); // Make sure that we reached the end.
+ fclose(input_fp_);
+ }
+ if (output_fp_) fclose(output_fp_);
+}
+
+template<class T>
+void RefFiles::ProcessReference(const T& test_results) {
+ WriteToFile(test_results);
+ ReadFromFileAndCompare(test_results);
+}
+
+template<typename T, size_t n>
+void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) {
+ WriteToFile(test_results, length);
+ ReadFromFileAndCompare(test_results, length);
+}
+
+template<typename T, size_t n>
+void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) {
+ if (output_fp_) {
+ ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
+ }
+}
+
+template<typename T, size_t n>
+void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
+ size_t length) {
+ if (input_fp_) {
+ // Read from ref file.
+ T* ref = new T[length];
+ ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
+ // Compare
+ EXPECT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
+ delete [] ref;
+ }
+}
+
+void RefFiles::WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats) {
+ if (output_fp_) {
+ ASSERT_EQ(1u, fwrite(&stats, sizeof(WebRtcNetEQ_NetworkStatistics), 1,
+ output_fp_));
+ }
+}
+
+void RefFiles::ReadFromFileAndCompare(
+ const WebRtcNetEQ_NetworkStatistics& stats) {
+ if (input_fp_) {
+ // Read from ref file.
+ size_t stat_size = sizeof(WebRtcNetEQ_NetworkStatistics);
+ WebRtcNetEQ_NetworkStatistics ref_stats;
+ ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
+ // Compare
+ EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size));
+ }
+}
+
+void RefFiles::WriteToFile(const WebRtcNetEQ_RTCPStat& stats) {
+ if (output_fp_) {
+ ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
+ output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.cum_lost), sizeof(stats.cum_lost), 1,
+ output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.ext_max), sizeof(stats.ext_max), 1,
+ output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
+ output_fp_));
+ }
+}
+
+void RefFiles::ReadFromFileAndCompare(
+ const WebRtcNetEQ_RTCPStat& stats) {
+ if (input_fp_) {
+ // Read from ref file.
+ WebRtcNetEQ_RTCPStat ref_stats;
+ ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
+ sizeof(ref_stats.fraction_lost), 1, input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.cum_lost), sizeof(ref_stats.cum_lost), 1,
+ input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.ext_max), sizeof(ref_stats.ext_max), 1,
+ input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
+ input_fp_));
+ // Compare
+ EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
+ EXPECT_EQ(ref_stats.cum_lost, stats.cum_lost);
+ EXPECT_EQ(ref_stats.ext_max, stats.ext_max);
+ EXPECT_EQ(ref_stats.jitter, stats.jitter);
+ }
+}
+
+class NetEqDecodingTest : public ::testing::Test {
+ protected:
+ // NetEQ must be polled for data once every 10 ms. Thus, neither of the
+ // constants below can be changed.
+ static const int kTimeStepMs = 10;
+ static const int kBlockSize8kHz = kTimeStepMs * 8;
+ static const int kBlockSize16kHz = kTimeStepMs * 16;
+ static const int kBlockSize32kHz = kTimeStepMs * 32;
+ static const int kMaxBlockSize = kBlockSize32kHz;
+
+ NetEqDecodingTest();
+ virtual void SetUp();
+ virtual void TearDown();
+ void SelectDecoders(WebRtcNetEQDecoder* used_codec);
+ void LoadDecoders();
+ void OpenInputFile(const std::string &rtp_file);
+ void Process(NETEQTEST_RTPpacket* rtp_ptr, int16_t* out_len);
+ void DecodeAndCompare(const std::string &rtp_file,
+ const std::string &ref_file);
+ void DecodeAndCheckStats(const std::string &rtp_file,
+ const std::string &stat_ref_file,
+ const std::string &rtcp_ref_file);
+ static void PopulateRtpInfo(int frame_index,
+ int timestamp,
+ WebRtcNetEQ_RTPInfo* rtp_info);
+ static void PopulateCng(int frame_index,
+ int timestamp,
+ WebRtcNetEQ_RTPInfo* rtp_info,
+ uint8_t* payload,
+ int* payload_len);
+
+ NETEQTEST_NetEQClass* neteq_inst_;
+ std::vector<NETEQTEST_Decoder*> dec_;
+ FILE* rtp_fp_;
+ unsigned int sim_clock_;
+ int16_t out_data_[kMaxBlockSize];
+};
+
+NetEqDecodingTest::NetEqDecodingTest()
+ : neteq_inst_(NULL),
+ rtp_fp_(NULL),
+ sim_clock_(0) {
+ memset(out_data_, 0, sizeof(out_data_));
+}
+
+void NetEqDecodingTest::SetUp() {
+ WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1];
+
+ SelectDecoders(usedCodec);
+ neteq_inst_ = new NETEQTEST_NetEQClass(usedCodec, dec_.size(), 8000,
+ kTCPLargeJitter);
+ ASSERT_TRUE(neteq_inst_);
+ LoadDecoders();
+}
+
+void NetEqDecodingTest::TearDown() {
+ if (neteq_inst_)
+ delete neteq_inst_;
+ for (size_t i = 0; i < dec_.size(); ++i) {
+ if (dec_[i])
+ delete dec_[i];
+ }
+ if (rtp_fp_)
+ fclose(rtp_fp_);
+}
+
+void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) {
+ *used_codec++ = kDecoderPCMu;
+ dec_.push_back(new decoder_PCMU(0));
+ *used_codec++ = kDecoderPCMa;
+ dec_.push_back(new decoder_PCMA(8));
+ *used_codec++ = kDecoderILBC;
+ dec_.push_back(new decoder_ILBC(102));
+ *used_codec++ = kDecoderISAC;
+ dec_.push_back(new decoder_iSAC(103));
+ *used_codec++ = kDecoderISACswb;
+ dec_.push_back(new decoder_iSACSWB(104));
+ *used_codec++ = kDecoderPCM16B;
+ dec_.push_back(new decoder_PCM16B_NB(93));
+ *used_codec++ = kDecoderPCM16Bwb;
+ dec_.push_back(new decoder_PCM16B_WB(94));
+ *used_codec++ = kDecoderPCM16Bswb32kHz;
+ dec_.push_back(new decoder_PCM16B_SWB32(95));
+ *used_codec++ = kDecoderCNG;
+ dec_.push_back(new decoder_CNG(13, 8000));
+ *used_codec++ = kDecoderCNG;
+ dec_.push_back(new decoder_CNG(98, 16000));
+}
+
+void NetEqDecodingTest::LoadDecoders() {
+ for (size_t i = 0; i < dec_.size(); ++i) {
+ ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_));
+ }
+}
+
+void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
+ rtp_fp_ = fopen(rtp_file.c_str(), "rb");
+ ASSERT_TRUE(rtp_fp_ != NULL);
+ ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_));
+}
+
+void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int16_t* out_len) {
+ // Check if time to receive.
+ while ((sim_clock_ >= rtp->time()) &&
+ (rtp->dataLen() >= 0)) {
+ if (rtp->dataLen() > 0) {
+ ASSERT_EQ(0, neteq_inst_->recIn(*rtp));
+ }
+ // Get next packet.
+ ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
+ }
+
+ // RecOut
+ *out_len = neteq_inst_->recOut(out_data_);
+ ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
+ (*out_len == kBlockSize16kHz) ||
+ (*out_len == kBlockSize32kHz));
+
+ // Increase time.
+ sim_clock_ += kTimeStepMs;
+}
+
+void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
+ const std::string &ref_file) {
+ OpenInputFile(rtp_file);
+
+ std::string ref_out_file = "";
+ if (ref_file.empty()) {
+ ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm";
+ }
+ RefFiles ref_files(ref_file, ref_out_file);
+
+ NETEQTEST_RTPpacket rtp;
+ ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
+ while (rtp.dataLen() >= 0) {
+ int16_t out_len;
+ Process(&rtp, &out_len);
+ ref_files.ProcessReference(out_data_, out_len);
+ }
+}
+
+void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file,
+ const std::string &stat_ref_file,
+ const std::string &rtcp_ref_file) {
+ OpenInputFile(rtp_file);
+ std::string stat_out_file = "";
+ if (stat_ref_file.empty()) {
+ stat_out_file = webrtc::test::OutputPath() +
+ "neteq_network_stats.dat";
+ }
+ RefFiles network_stat_files(stat_ref_file, stat_out_file);
+
+ std::string rtcp_out_file = "";
+ if (rtcp_ref_file.empty()) {
+ rtcp_out_file = webrtc::test::OutputPath() +
+ "neteq_rtcp_stats.dat";
+ }
+ RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
+
+ NETEQTEST_RTPpacket rtp;
+ ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
+ while (rtp.dataLen() >= 0) {
+ int16_t out_len;
+ Process(&rtp, &out_len);
+
+ // Query the network statistics API once per second
+ if (sim_clock_ % 1000 == 0) {
+ // Process NetworkStatistics.
+ WebRtcNetEQ_NetworkStatistics network_stats;
+ ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
+ &network_stats));
+ network_stat_files.ProcessReference(network_stats);
+
+ // Process RTCPstat.
+ WebRtcNetEQ_RTCPStat rtcp_stats;
+ ASSERT_EQ(0, WebRtcNetEQ_GetRTCPStats(neteq_inst_->instance(),
+ &rtcp_stats));
+ rtcp_stat_files.ProcessReference(rtcp_stats);
+ }
+ }
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+ int timestamp,
+ WebRtcNetEQ_RTPInfo* rtp_info) {
+ rtp_info->sequenceNumber = frame_index;
+ rtp_info->timeStamp = timestamp;
+ rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
+ rtp_info->payloadType = 94; // PCM16b WB codec.
+ rtp_info->markerBit = 0;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+ int timestamp,
+ WebRtcNetEQ_RTPInfo* rtp_info,
+ uint8_t* payload,
+ int* payload_len) {
+ rtp_info->sequenceNumber = frame_index;
+ rtp_info->timeStamp = timestamp;
+ rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
+ rtp_info->payloadType = 98; // WB CNG.
+ rtp_info->markerBit = 0;
+ payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
+ *payload_len = 1; // Only noise level, no spectral parameters.
+}
+
+TEST_F(NetEqDecodingTest, TestBitExactness) {
+ const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
+ "resources/neteq_universal.rtp";
+ const std::string kInputRefFile =
+ webrtc::test::ResourcePath("neteq_universal_ref", "pcm");
+ DecodeAndCompare(kInputRtpFile, kInputRefFile);
+}
+
+TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
+ const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
+ "resources/neteq_universal.rtp";
+ const std::string kNetworkStatRefFile =
+ webrtc::test::ResourcePath("neteq_network_stats", "dat");
+ const std::string kRtcpStatRefFile =
+ webrtc::test::ResourcePath("neteq_rtcp_stats", "dat");
+ DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
+}
+
+TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
+ // Use fax mode to avoid time-scaling. This is to simplify the testing of
+ // packet waiting times in the packet buffer.
+ ASSERT_EQ(0,
+ WebRtcNetEQ_SetPlayoutMode(neteq_inst_->instance(), kPlayoutFax));
+ // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
+ int num_frames = 30;
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ for (int i = 0; i < num_frames; ++i) {
+ uint16_t payload[kSamples] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ rtp_info.sequenceNumber = i;
+ rtp_info.timeStamp = i * kSamples;
+ rtp_info.SSRC = 0x1234; // Just an arbitrary SSRC.
+ rtp_info.payloadType = 94; // PCM16b WB codec.
+ rtp_info.markerBit = 0;
+ ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
+ reinterpret_cast<uint8_t*>(payload),
+ kPayloadBytes, 0));
+ }
+ // Pull out all data.
+ for (int i = 0; i < num_frames; ++i) {
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+ const int kVecLen = 110; // More than kLenWaitingTimes in mcu.h.
+ int waiting_times[kVecLen];
+ int len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
+ kVecLen, waiting_times);
+ EXPECT_EQ(num_frames, len);
+ // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
+ // spacing (per definition), we expect the delay to increase with 10 ms for
+ // each packet.
+ for (int i = 0; i < len; ++i) {
+ EXPECT_EQ((i + 1) * 10, waiting_times[i]);
+ }
+
+ // Check statistics again and make sure it's been reset.
+ EXPECT_EQ(0, WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
+ kVecLen, waiting_times));
+
+ // Process > 100 frames, and make sure that that we get statistics
+ // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
+ num_frames = 110;
+ for (int i = 0; i < num_frames; ++i) {
+ uint16_t payload[kSamples] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ rtp_info.sequenceNumber = i;
+ rtp_info.timeStamp = i * kSamples;
+ rtp_info.SSRC = 0x1235; // Just an arbitrary SSRC.
+ rtp_info.payloadType = 94; // PCM16b WB codec.
+ rtp_info.markerBit = 0;
+ ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
+ reinterpret_cast<uint8_t*>(payload),
+ kPayloadBytes, 0));
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+
+ len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
+ kVecLen, waiting_times);
+ EXPECT_EQ(100, len);
+}
+
+TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
+ const int kNumFrames = 3000; // Needed for convergence.
+ int frame_index = 0;
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ while (frame_index < kNumFrames) {
+ // Insert one packet each time, except every 10th time where we insert two
+ // packets at once. This will create a negative clock-drift of approx. 10%.
+ int num_packets = (frame_index % 10 == 0 ? 2 : 1);
+ for (int n = 0; n < num_packets; ++n) {
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
+ ASSERT_EQ(0,
+ WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
+ &rtp_info,
+ payload,
+ kPayloadBytes, 0));
+ ++frame_index;
+ }
+
+ // Pull out data once.
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+
+ WebRtcNetEQ_NetworkStatistics network_stats;
+ ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
+ &network_stats));
+ EXPECT_EQ(-106911, network_stats.clockDriftPPM);
+}
+
+TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
+ const int kNumFrames = 5000; // Needed for convergence.
+ int frame_index = 0;
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ for (int i = 0; i < kNumFrames; ++i) {
+ // Insert one packet each time, except every 10th time where we don't insert
+ // any packet. This will create a positive clock-drift of approx. 11%.
+ int num_packets = (i % 10 == 9 ? 0 : 1);
+ for (int n = 0; n < num_packets; ++n) {
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
+ ASSERT_EQ(0,
+ WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
+ &rtp_info,
+ payload,
+ kPayloadBytes, 0));
+ ++frame_index;
+ }
+
+ // Pull out data once.
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+
+ WebRtcNetEQ_NetworkStatistics network_stats;
+ ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
+ &network_stats));
+ EXPECT_EQ(108352, network_stats.clockDriftPPM);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
+ uint16_t seq_no = 0;
+ uint32_t timestamp = 0;
+ const int kFrameSizeMs = 30;
+ const int kSamples = kFrameSizeMs * 16;
+ const int kPayloadBytes = kSamples * 2;
+ // Apply a clock drift of -25 ms / s (sender faster than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+ double next_input_time_ms = 0.0;
+ double t_ms;
+
+ // Insert speech for 5 seconds.
+ const int kSpeechDurationMs = 5000;
+ for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+ // Each turn in this for loop is 10 ms.
+ while (next_input_time_ms <= t_ms) {
+ // Insert one 30 ms speech frame.
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0,
+ WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
+ &rtp_info,
+ payload,
+ kPayloadBytes, 0));
+ ++seq_no;
+ timestamp += kSamples;
+ next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
+ }
+ // Pull out data once.
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+
+ EXPECT_EQ(kOutputNormal, neteq_inst_->getOutputType());
+ int32_t delay_before = timestamp - neteq_inst_->getSpeechTimeStamp();
+
+ // Insert CNG for 1 minute (= 60000 ms).
+ const int kCngPeriodMs = 100;
+ const int kCngPeriodSamples = kCngPeriodMs * 16; // Period in 16 kHz samples.
+ const int kCngDurationMs = 60000;
+ for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
+ // Each turn in this for loop is 10 ms.
+ while (next_input_time_ms <= t_ms) {
+ // Insert one CNG frame each 100 ms.
+ uint8_t payload[kPayloadBytes];
+ int payload_len;
+ WebRtcNetEQ_RTPInfo rtp_info;
+ PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+ ASSERT_EQ(0,
+ WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
+ &rtp_info,
+ payload,
+ payload_len, 0));
+ ++seq_no;
+ timestamp += kCngPeriodSamples;
+ next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor;
+ }
+ // Pull out data once.
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ }
+
+ EXPECT_EQ(kOutputCNG, neteq_inst_->getOutputType());
+
+ // Insert speech again until output type is speech.
+ while (neteq_inst_->getOutputType() != kOutputNormal) {
+ // Each turn in this for loop is 10 ms.
+ while (next_input_time_ms <= t_ms) {
+ // Insert one 30 ms speech frame.
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcNetEQ_RTPInfo rtp_info;
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0,
+ WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
+ &rtp_info,
+ payload,
+ kPayloadBytes, 0));
+ ++seq_no;
+ timestamp += kSamples;
+ next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
+ }
+ // Pull out data once.
+ ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
+ // Increase clock.
+ t_ms += 10;
+ }
+
+ int32_t delay_after = timestamp - neteq_inst_->getSpeechTimeStamp();
+ // Compare delay before and after, and make sure it differs less than 20 ms.
+ EXPECT_LE(delay_after, delay_before + 20 * 16);
+ EXPECT_GE(delay_after, delay_before - 20 * 16);
+}
+
+} // namespace
diff --git a/trunk/src/modules/audio_conference_mixer/OWNERS b/trunk/src/modules/audio_conference_mixer/OWNERS
new file mode 100644
index 0000000..7dc791e
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/OWNERS
@@ -0,0 +1,3 @@
+henrike@webrtc.org
+pwestin@webrtc.org
+andrew@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
new file mode 100644
index 0000000..4ece1bf
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+
+#include "audio_conference_mixer_defines.h"
+#include "module.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+class AudioMixerOutputReceiver;
+class AudioMixerStatusReceiver;
+class MixerParticipant;
+class Trace;
+
+class AudioConferenceMixer : public Module
+{
+public:
+ enum {kMaximumAmountOfMixedParticipants = 3};
+ enum Frequency
+ {
+ kNbInHz = 8000,
+ kWbInHz = 16000,
+ kSwbInHz = 32000,
+ kLowestPossible = -1,
+ kDefaultFrequency = kWbInHz
+ };
+
+ // Factory method. Constructor disabled.
+ static AudioConferenceMixer* Create(int id);
+ virtual ~AudioConferenceMixer() {}
+
+ // Module functions
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+ virtual WebRtc_Word32 TimeUntilNextProcess() = 0 ;
+ virtual WebRtc_Word32 Process() = 0;
+
+ // Register/unregister a callback class for receiving the mixed audio.
+ virtual WebRtc_Word32 RegisterMixedStreamCallback(
+ AudioMixerOutputReceiver& receiver) = 0;
+ virtual WebRtc_Word32 UnRegisterMixedStreamCallback() = 0;
+
+ // Register/unregister a callback class for receiving status information.
+ virtual WebRtc_Word32 RegisterMixerStatusCallback(
+ AudioMixerStatusReceiver& mixerStatusCallback,
+ const WebRtc_UWord32 amountOf10MsBetweenCallbacks) = 0;
+ virtual WebRtc_Word32 UnRegisterMixerStatusCallback() = 0;
+
+ // Add/remove participants as candidates for mixing.
+ virtual WebRtc_Word32 SetMixabilityStatus(
+ MixerParticipant& participant,
+ const bool mixable) = 0;
+ // mixable is set to true if a participant is a candidate for mixing.
+ virtual WebRtc_Word32 MixabilityStatus(
+ MixerParticipant& participant,
+ bool& mixable) = 0;
+
+ // Inform the mixer that the participant should always be mixed and not
+ // count toward the number of mixed participants. Note that a participant
+ // must have been added to the mixer (by calling SetMixabilityStatus())
+ // before this function can be successfully called.
+ virtual WebRtc_Word32 SetAnonymousMixabilityStatus(
+ MixerParticipant& participant, const bool mixable) = 0;
+ // mixable is set to true if the participant is mixed anonymously.
+ virtual WebRtc_Word32 AnonymousMixabilityStatus(
+ MixerParticipant& participant, bool& mixable) = 0;
+
+ // Set the minimum sampling frequency at which to mix. The mixing algorithm
+ // may still choose to mix at a higher samling frequency to avoid
+ // downsampling of audio contributing to the mixed audio.
+ virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq) = 0;
+
+protected:
+ AudioConferenceMixer() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
new file mode 100644
index 0000000..718470d
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+
+#include "map_wrapper.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class MixHistory;
+
+// A callback class that all mixer participants must inherit from/implement.
+class MixerParticipant
+{
+public:
+ // The implementation of this function should update audioFrame with new
+ // audio every time it's called.
+ //
+ // If it returns -1, the frame will not be added to the mix.
+ virtual WebRtc_Word32 GetAudioFrame(const WebRtc_Word32 id,
+ AudioFrame& audioFrame) = 0;
+
+ // mixed will be set to true if the participant was mixed this mix iteration
+ WebRtc_Word32 IsMixed(bool& mixed) const;
+
+ // This function specifies the sampling frequency needed for the AudioFrame
+ // for future GetAudioFrame(..) calls.
+ virtual WebRtc_Word32 NeededFrequency(const WebRtc_Word32 id) = 0;
+
+ MixHistory* _mixHistory;
+protected:
+ MixerParticipant();
+ virtual ~MixerParticipant();
+};
+
+// Container struct for participant statistics.
+struct ParticipantStatistics
+{
+ WebRtc_Word32 participant;
+ WebRtc_Word32 level;
+};
+
+class AudioMixerStatusReceiver
+{
+public:
+ // Callback function that provides an array of ParticipantStatistics for the
+ // participants that were mixed last mix iteration.
+ virtual void MixedParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size) = 0;
+ // Callback function that provides an array of the ParticipantStatistics for
+ // the participants that had a positiv VAD last mix iteration.
+ virtual void VADPositiveParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size) = 0;
+ // Callback function that provides the audio level of the mixed audio frame
+ // from the last mix iteration.
+ virtual void MixedAudioLevel(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord32 level) = 0;
+protected:
+ AudioMixerStatusReceiver() {}
+ virtual ~AudioMixerStatusReceiver() {}
+};
+
+class AudioMixerOutputReceiver
+{
+public:
+ // This callback function provides the mixed audio for this mix iteration.
+ // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
+ // size according to the size parameter.
+ virtual void NewMixedAudio(const WebRtc_Word32 id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const WebRtc_UWord32 size) = 0;
+protected:
+ AudioMixerOutputReceiver() {}
+ virtual ~AudioMixerOutputReceiver() {}
+};
+
+class AudioRelayReceiver
+{
+public:
+ // This callback function provides the mix decision for this mix iteration.
+ // mixerList is a list of elements of the type
+ // [int,MixerParticipant*]
+ virtual void NewAudioToRelay(const WebRtc_Word32 id,
+ const MapWrapper& mixerList) = 0;
+protected:
+ AudioRelayReceiver() {}
+ virtual ~AudioRelayReceiver() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/Android.mk b/trunk/src/modules/audio_conference_mixer/source/Android.mk
new file mode 100644
index 0000000..a90ff1d
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/Android.mk
@@ -0,0 +1,44 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_audio_conference_mixer
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ audio_frame_manipulator.cc \
+ level_indicator.cc \
+ audio_conference_mixer_impl.cc \
+ time_scheduler.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../audio_processing/include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi
new file mode 100644
index 0000000..fd116c6
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi
@@ -0,0 +1,70 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audio_conference_mixer',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'audio_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/audio_conference_mixer.h',
+ '../interface/audio_conference_mixer_defines.h',
+ 'audio_frame_manipulator.cc',
+ 'audio_frame_manipulator.h',
+ 'level_indicator.cc',
+ 'level_indicator.h',
+ 'memory_pool.h',
+ 'memory_pool_posix.h',
+ 'memory_pool_win.h',
+ 'audio_conference_mixer_impl.cc',
+ 'audio_conference_mixer_impl.h',
+ 'time_scheduler.cc',
+ 'time_scheduler.h',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'audio_conference_mixer_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_conference_mixer',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'sources': [
+ 'audio_conference_mixer_unittest.cc',
+ ],
+ }, # audio_conference_mixer_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
new file mode 100644
index 0000000..8da0ecd
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -0,0 +1,1187 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_conference_mixer_defines.h"
+#include "audio_conference_mixer_impl.h"
+#include "audio_frame_manipulator.h"
+#include "audio_processing.h"
+#include "critical_section_wrapper.h"
+#include "map_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+namespace {
+void SetParticipantStatistics(ParticipantStatistics* stats,
+ const AudioFrame& frame)
+{
+ stats->participant = frame._id;
+ stats->level = frame._volume;
+}
+} // namespace
+
+MixerParticipant::MixerParticipant()
+ : _mixHistory(new MixHistory())
+{
+}
+
+MixerParticipant::~MixerParticipant()
+{
+ delete _mixHistory;
+}
+
+WebRtc_Word32 MixerParticipant::IsMixed(bool& mixed) const
+{
+ return _mixHistory->IsMixed(mixed);
+}
+
+MixHistory::MixHistory()
+ : _isMixed(0)
+{
+}
+
+MixHistory::~MixHistory()
+{
+}
+
+WebRtc_Word32 MixHistory::IsMixed(bool& mixed) const
+{
+ mixed = (_isMixed.Value() == 1);
+ return 0;
+}
+
+WebRtc_Word32 MixHistory::WasMixed(bool& wasMixed) const
+{
+ // Was mixed is the same as is mixed depending on perspective. This function
+ // is for the perspective of AudioConferenceMixerImpl.
+ return IsMixed(wasMixed);
+}
+
+WebRtc_Word32 MixHistory::SetIsMixed(const bool mixed)
+{
+ _isMixed = mixed ? 1 : 0;
+ return 0;
+}
+
+void MixHistory::ResetMixedStatus()
+{
+ _isMixed = 0;
+}
+
+AudioConferenceMixer* AudioConferenceMixer::Create(int id)
+{
+ AudioConferenceMixerImpl* mixer = new AudioConferenceMixerImpl(id);
+ if(!mixer->Init())
+ {
+ delete mixer;
+ return NULL;
+ }
+ return mixer;
+}
+
+AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id)
+ : _scratchParticipantsToMixAmount(0),
+ _scratchMixedParticipants(),
+ _scratchVadPositiveParticipantsAmount(0),
+ _scratchVadPositiveParticipants(),
+ _crit(NULL),
+ _cbCrit(NULL),
+ _id(id),
+ _minimumMixingFreq(kLowestPossible),
+ _mixReceiver(NULL),
+ _mixerStatusCallback(NULL),
+ _amountOf10MsBetweenCallbacks(1),
+ _amountOf10MsUntilNextCallback(0),
+ _mixerStatusCb(false),
+ _outputFrequency(kDefaultFrequency),
+ _sampleSize(0),
+ _audioFramePool(NULL),
+ _participantList(),
+ _additionalParticipantList(),
+ _numMixedParticipants(0),
+ _timeStamp(0),
+ _timeScheduler(kProcessPeriodicityInMs),
+ _mixedAudioLevel(),
+ _processCalls(0),
+ _limiter(NULL)
+{}
+
+bool AudioConferenceMixerImpl::Init()
+{
+ _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
+ if (_crit.get() == NULL)
+ return false;
+
+ _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
+ if(_cbCrit.get() == NULL)
+ return false;
+
+ _limiter.reset(AudioProcessing::Create(_id));
+ if(_limiter.get() == NULL)
+ return false;
+
+ MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool,
+ DEFAULT_AUDIO_FRAME_POOLSIZE);
+ if(_audioFramePool == NULL)
+ return false;
+
+ if(SetOutputFrequency(kDefaultFrequency) == -1)
+ return false;
+
+ // Assume mono.
+ if (!SetNumLimiterChannels(1))
+ return false;
+
+ if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
+ _limiter->kNoError)
+ return false;
+
+ // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
+ // divide-by-2 but -7 is used instead to give a bit of headroom since the
+ // AGC is not a hard limiter.
+ if(_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
+ return false;
+
+ if(_limiter->gain_control()->set_compression_gain_db(0)
+ != _limiter->kNoError)
+ return false;
+
+ if(_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
+ return false;
+
+ if(_limiter->gain_control()->Enable(true) != _limiter->kNoError)
+ return false;
+
+ return true;
+}
+
+AudioConferenceMixerImpl::~AudioConferenceMixerImpl()
+{
+ MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
+ assert(_audioFramePool == NULL);
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+// Process should be called every kProcessPeriodicityInMs ms
+WebRtc_Word32 AudioConferenceMixerImpl::TimeUntilNextProcess()
+{
+ WebRtc_Word32 timeUntilNextProcess = 0;
+ CriticalSectionScoped cs(_crit.get());
+ if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "failed in TimeToNextUpdate() call");
+ // Sanity check
+ assert(false);
+ return -1;
+ }
+ return timeUntilNextProcess;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::Process()
+{
+ WebRtc_UWord32 remainingParticipantsAllowedToMix =
+ kMaximumAmountOfMixedParticipants;
+ {
+ CriticalSectionScoped cs(_crit.get());
+ assert(_processCalls == 0);
+ _processCalls++;
+
+ // Let the scheduler know that we are running one iteration.
+ _timeScheduler.UpdateScheduler();
+ }
+
+ ListWrapper mixList;
+ ListWrapper rampOutList;
+ ListWrapper additionalFramesList;
+ MapWrapper mixedParticipantsMap;
+ {
+ CriticalSectionScoped cs(_cbCrit.get());
+
+ WebRtc_Word32 lowFreq = GetLowestMixingFrequency();
+ // SILK can run in 12 kHz and 24 kHz. These frequencies are not
+ // supported so use the closest higher frequency to not lose any
+ // information.
+ // TODO(henrike): this is probably more appropriate to do in
+ // GetLowestMixingFrequency().
+ if (lowFreq == 12000)
+ {
+ lowFreq = 16000;
+ } else if (lowFreq == 24000) {
+ lowFreq = 32000;
+ }
+ if(lowFreq <= 0)
+ {
+ CriticalSectionScoped cs(_crit.get());
+ _processCalls--;
+ return 0;
+ } else {
+ switch(lowFreq)
+ {
+ case 8000:
+ if(OutputFrequency() != kNbInHz)
+ {
+ SetOutputFrequency(kNbInHz);
+ }
+ break;
+ case 16000:
+ if(OutputFrequency() != kWbInHz)
+ {
+ SetOutputFrequency(kWbInHz);
+ }
+ break;
+ case 32000:
+ if(OutputFrequency() != kSwbInHz)
+ {
+ SetOutputFrequency(kSwbInHz);
+ }
+ break;
+ default:
+ assert(false);
+
+ CriticalSectionScoped cs(_crit.get());
+ _processCalls--;
+ return -1;
+ }
+ }
+
+ UpdateToMix(mixList, rampOutList, mixedParticipantsMap,
+ remainingParticipantsAllowedToMix);
+
+ GetAdditionalAudio(additionalFramesList);
+ UpdateMixedStatus(mixedParticipantsMap);
+ _scratchParticipantsToMixAmount = mixedParticipantsMap.Size();
+ }
+
+ // Clear mixedParticipantsMap to avoid memory leak warning.
+ // Please note that the mixedParticipantsMap doesn't own any dynamically
+ // allocated memory.
+ while(mixedParticipantsMap.Erase(mixedParticipantsMap.First()) == 0) {}
+
+ // Get an AudioFrame for mixing from the memory pool.
+ AudioFrame* mixedAudio = NULL;
+ if(_audioFramePool->PopMemory(mixedAudio) == -1)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
+ "failed PopMemory() call");
+ assert(false);
+ return -1;
+ }
+
+ bool timeForMixerCallback = false;
+ int retval = 0;
+ WebRtc_Word32 audioLevel = 0;
+ {
+ const ListItem* firstItem = mixList.First();
+ // Assume mono.
+ WebRtc_UWord8 numberOfChannels = 1;
+ if(firstItem != NULL)
+ {
+ // Use the same number of channels as the first frame to be mixed.
+ numberOfChannels = static_cast<const AudioFrame*>(
+ firstItem->GetItem())->_audioChannel;
+ }
+ // TODO(henrike): it might be better to decide the number of channels
+ // with an API instead of dynamically.
+
+ CriticalSectionScoped cs(_crit.get());
+ if (!SetNumLimiterChannels(numberOfChannels))
+ retval = -1;
+
+ mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadPassive, numberOfChannels);
+
+ _timeStamp += _sampleSize;
+
+ MixFromList(*mixedAudio, mixList);
+ MixAnonomouslyFromList(*mixedAudio, additionalFramesList);
+ MixAnonomouslyFromList(*mixedAudio, rampOutList);
+
+ if(mixedAudio->_payloadDataLengthInSamples == 0)
+ {
+ // Nothing was mixed, set the audio samples to silence.
+ memset(mixedAudio->_payloadData, 0, _sampleSize);
+ mixedAudio->_payloadDataLengthInSamples = _sampleSize;
+ }
+ else
+ {
+ // Only call the limiter if we have something to mix.
+ if(!LimitMixedAudio(*mixedAudio))
+ retval = -1;
+ }
+
+ _mixedAudioLevel.ComputeLevel(mixedAudio->_payloadData,_sampleSize);
+ audioLevel = _mixedAudioLevel.GetLevel();
+
+ if(_mixerStatusCb)
+ {
+ _scratchVadPositiveParticipantsAmount = 0;
+ UpdateVADPositiveParticipants(mixList);
+ if(_amountOf10MsUntilNextCallback-- == 0)
+ {
+ _amountOf10MsUntilNextCallback = _amountOf10MsBetweenCallbacks;
+ timeForMixerCallback = true;
+ }
+ }
+ }
+
+ {
+ CriticalSectionScoped cs(_cbCrit.get());
+ if(_mixReceiver != NULL)
+ {
+ const AudioFrame** dummy = NULL;
+ _mixReceiver->NewMixedAudio(
+ _id,
+ *mixedAudio,
+ dummy,
+ 0);
+ }
+
+ if((_mixerStatusCallback != NULL) &&
+ timeForMixerCallback)
+ {
+ _mixerStatusCallback->MixedParticipants(
+ _id,
+ _scratchMixedParticipants,
+ _scratchParticipantsToMixAmount);
+
+ _mixerStatusCallback->VADPositiveParticipants(
+ _id,
+ _scratchVadPositiveParticipants,
+ _scratchVadPositiveParticipantsAmount);
+ _mixerStatusCallback->MixedAudioLevel(_id,audioLevel);
+ }
+ }
+
+ // Reclaim all outstanding memory.
+ _audioFramePool->PushMemory(mixedAudio);
+ ClearAudioFrameList(mixList);
+ ClearAudioFrameList(rampOutList);
+ ClearAudioFrameList(additionalFramesList);
+ {
+ CriticalSectionScoped cs(_crit.get());
+ _processCalls--;
+ }
+ return retval;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixedStreamCallback(
+ AudioMixerOutputReceiver& mixReceiver)
+{
+ CriticalSectionScoped cs(_cbCrit.get());
+ if(_mixReceiver != NULL)
+ {
+ return -1;
+ }
+ _mixReceiver = &mixReceiver;
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback()
+{
+ CriticalSectionScoped cs(_cbCrit.get());
+ if(_mixReceiver == NULL)
+ {
+ return -1;
+ }
+ _mixReceiver = NULL;
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::SetOutputFrequency(
+ const Frequency frequency)
+{
+ CriticalSectionScoped cs(_crit.get());
+ const int error = _limiter->set_sample_rate_hz(frequency);
+ if(error != _limiter->kNoError)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "Error from AudioProcessing: %d", error);
+ return -1;
+ }
+
+ _outputFrequency = frequency;
+ _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;
+
+ return 0;
+}
+
+AudioConferenceMixer::Frequency
+AudioConferenceMixerImpl::OutputFrequency() const
+{
+ CriticalSectionScoped cs(_crit.get());
+ return _outputFrequency;
+}
+
+bool AudioConferenceMixerImpl::SetNumLimiterChannels(int numChannels)
+{
+ if(_limiter->num_input_channels() != numChannels)
+ {
+ const int error = _limiter->set_num_channels(numChannels,
+ numChannels);
+ if(error != _limiter->kNoError)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "Error from AudioProcessing: %d", error);
+ assert(false);
+ return false;
+ }
+ }
+
+ return true;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback(
+ AudioMixerStatusReceiver& mixerStatusCallback,
+ const WebRtc_UWord32 amountOf10MsBetweenCallbacks)
+{
+ if(amountOf10MsBetweenCallbacks == 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceAudioMixerServer,
+ _id,
+ "amountOf10MsBetweenCallbacks(%d) needs to be larger than 0");
+ return -1;
+ }
+ {
+ CriticalSectionScoped cs(_cbCrit.get());
+ if(_mixerStatusCallback != NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "Mixer status callback already registered");
+ return -1;
+ }
+ _mixerStatusCallback = &mixerStatusCallback;
+ }
+ {
+ CriticalSectionScoped cs(_crit.get());
+ _amountOf10MsBetweenCallbacks = amountOf10MsBetweenCallbacks;
+ _amountOf10MsUntilNextCallback = 0;
+ _mixerStatusCb = true;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback()
+{
+ {
+ CriticalSectionScoped cs(_crit.get());
+ if(!_mixerStatusCb)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "Mixer status callback not registered");
+ return -1;
+ }
+ _mixerStatusCb = false;
+ }
+ {
+ CriticalSectionScoped cs(_cbCrit.get());
+ _mixerStatusCallback = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus(
+ MixerParticipant& participant,
+ const bool mixable)
+{
+ if (!mixable)
+ {
+ // Anonymous participants are in a separate list. Make sure that the
+ // participant is in the _participantList if it is being mixed.
+ SetAnonymousMixabilityStatus(participant, false);
+ }
+ WebRtc_UWord32 numMixedParticipants;
+ {
+ CriticalSectionScoped cs(_cbCrit.get());
+ const bool isMixed =
+ IsParticipantInList(participant,_participantList);
+ // API must be called with a new state.
+ if(!(mixable ^ isMixed))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "Mixable is aready %s",
+ isMixed ? "ON" : "off");
+ return -1;
+ }
+ bool success = false;
+ if(mixable)
+ {
+ success = AddParticipantToList(participant,_participantList);
+ }
+ else
+ {
+ success = RemoveParticipantFromList(participant,_participantList);
+ }
+ if(!success)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "failed to %s participant",
+ mixable ? "add" : "remove");
+ assert(false);
+ return -1;
+ }
+
+ int numMixedNonAnonymous = _participantList.GetSize();
+ if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants)
+ {
+ numMixedNonAnonymous = kMaximumAmountOfMixedParticipants;
+ }
+ numMixedParticipants = numMixedNonAnonymous +
+ _additionalParticipantList.GetSize();
+ }
+ // A MixerParticipant was added or removed. Make sure the scratch
+ // buffer is updated if necessary.
+ // Note: The scratch buffer may only be updated in Process().
+ CriticalSectionScoped cs(_crit.get());
+ _numMixedParticipants = numMixedParticipants;
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::MixabilityStatus(
+ MixerParticipant& participant,
+ bool& mixable)
+{
+ CriticalSectionScoped cs(_cbCrit.get());
+ mixable = IsParticipantInList(participant, _participantList);
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
+ MixerParticipant& participant, const bool anonymous)
+{
+ CriticalSectionScoped cs(_cbCrit.get());
+ if(IsParticipantInList(participant, _additionalParticipantList))
+ {
+ if(anonymous)
+ {
+ return 0;
+ }
+ if(!RemoveParticipantFromList(participant, _additionalParticipantList))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "unable to remove participant from anonymous list");
+ assert(false);
+ return -1;
+ }
+ return AddParticipantToList(participant, _participantList) ? 0 : -1;
+ }
+ if(!anonymous)
+ {
+ return 0;
+ }
+ const bool mixable = RemoveParticipantFromList(participant,
+ _participantList);
+ if(!mixable)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceAudioMixerServer,
+ _id,
+ "participant must be registered before turning it into anonymous");
+ // Setting anonymous status is only possible if MixerParticipant is
+ // already registered.
+ return -1;
+ }
+ return AddParticipantToList(participant, _additionalParticipantList) ?
+ 0 : -1;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::AnonymousMixabilityStatus(
+ MixerParticipant& participant, bool& mixable)
+{
+ CriticalSectionScoped cs(_cbCrit.get());
+ mixable = IsParticipantInList(participant,
+ _additionalParticipantList);
+ return 0;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::SetMinimumMixingFrequency(
+ Frequency freq)
+{
+ // Make sure that only allowed sampling frequencies are used. Use closest
+ // higher sampling frequency to avoid losing information.
+ if (static_cast<int>(freq) == 12000)
+ {
+ freq = kWbInHz;
+ } else if (static_cast<int>(freq) == 24000) {
+ freq = kSwbInHz;
+ }
+
+ if((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) ||
+ (freq == kLowestPossible))
+ {
+ _minimumMixingFreq=freq;
+ return 0;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "SetMinimumMixingFrequency incorrect frequency: %i",freq);
+ assert(false);
+ return -1;
+ }
+}
+
+// Check all AudioFrames that are to be mixed. The highest sampling frequency
+// found is the lowest that can be used without losing information.
+WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequency()
+{
+ const int participantListFrequency =
+ GetLowestMixingFrequencyFromList(_participantList);
+ const int anonymousListFrequency =
+ GetLowestMixingFrequencyFromList(_additionalParticipantList);
+ const int highestFreq =
+ (participantListFrequency > anonymousListFrequency) ?
+ participantListFrequency : anonymousListFrequency;
+ // Check if the user specified a lowest mixing frequency.
+ if(_minimumMixingFreq != kLowestPossible)
+ {
+ if(_minimumMixingFreq > highestFreq)
+ {
+ return _minimumMixingFreq;
+ }
+ }
+ return highestFreq;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList(
+ ListWrapper& mixList)
+{
+ WebRtc_Word32 highestFreq = 8000;
+ ListItem* item = mixList.First();
+ while(item)
+ {
+ MixerParticipant* participant =
+ static_cast<MixerParticipant*>(item->GetItem());
+ const WebRtc_Word32 neededFrequency = participant->NeededFrequency(_id);
+ if(neededFrequency > highestFreq)
+ {
+ highestFreq = neededFrequency;
+ }
+ item = mixList.Next(item);
+ }
+ return highestFreq;
+}
+
+void AudioConferenceMixerImpl::UpdateToMix(
+ ListWrapper& mixList,
+ ListWrapper& rampOutList,
+ MapWrapper& mixParticipantList,
+ WebRtc_UWord32& maxAudioFrameCounter)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)",
+ maxAudioFrameCounter);
+ const WebRtc_UWord32 mixListStartSize = mixList.GetSize();
+ ListWrapper activeList; // Elements are AudioFrames
+ // Struct needed by the passive lists to keep track of which AudioFrame
+ // belongs to which MixerParticipant.
+ struct ParticipantFramePair
+ {
+ MixerParticipant* participant;
+ AudioFrame* audioFrame;
+ };
+ ListWrapper passiveWasNotMixedList; // Elements are MixerParticipant
+ ListWrapper passiveWasMixedList; // Elements are MixerParticipant
+ ListItem* item = _participantList.First();
+ while(item)
+ {
+ // Stop keeping track of passive participants if there are already
+ // enough participants available (they wont be mixed anyway).
+ bool mustAddToPassiveList = (maxAudioFrameCounter >
+ (activeList.GetSize() +
+ passiveWasMixedList.GetSize() +
+ passiveWasNotMixedList.GetSize()));
+
+ MixerParticipant* participant = static_cast<MixerParticipant*>(
+ item->GetItem());
+ bool wasMixed = false;
+ participant->_mixHistory->WasMixed(wasMixed);
+ AudioFrame* audioFrame = NULL;
+ if(_audioFramePool->PopMemory(audioFrame) == -1)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
+ "failed PopMemory() call");
+ assert(false);
+ return;
+ }
+ audioFrame->_frequencyInHz = _outputFrequency;
+
+ if(participant->GetAudioFrame(_id,*audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "failed to GetAudioFrame() from participant");
+ _audioFramePool->PushMemory(audioFrame);
+ item = _participantList.Next(item);
+ continue;
+ }
+ // TODO(henrike): this assert triggers in some test cases where SRTP is
+ // used which prevents NetEQ from making a VAD. Temporarily disable this
+ // assert until the problem is fixed on a higher level.
+ // assert(audioFrame->_vadActivity != AudioFrame::kVadUnknown);
+ if (audioFrame->_vadActivity == AudioFrame::kVadUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "invalid VAD state from participant");
+ }
+
+ if(audioFrame->_vadActivity == AudioFrame::kVadActive)
+ {
+ if(!wasMixed)
+ {
+ RampIn(*audioFrame);
+ }
+
+ if(activeList.GetSize() >= maxAudioFrameCounter)
+ {
+ // There are already more active participants than should be
+ // mixed. Only keep the ones with the highest energy.
+ ListItem* replaceItem = NULL;
+ CalculateEnergy(*audioFrame);
+ WebRtc_UWord32 lowestEnergy = audioFrame->_energy;
+
+ ListItem* activeItem = activeList.First();
+ while(activeItem)
+ {
+ AudioFrame* replaceFrame = static_cast<AudioFrame*>(
+ activeItem->GetItem());
+ CalculateEnergy(*replaceFrame);
+ if(replaceFrame->_energy < lowestEnergy)
+ {
+ replaceItem = activeItem;
+ lowestEnergy = replaceFrame->_energy;
+ }
+ activeItem = activeList.Next(activeItem);
+ }
+ if(replaceItem != NULL)
+ {
+ AudioFrame* replaceFrame = static_cast<AudioFrame*>(
+ replaceItem->GetItem());
+
+ bool replaceWasMixed = false;
+ MapItem* replaceParticipant = mixParticipantList.Find(
+ replaceFrame->_id);
+ // When a frame is pushed to |activeList| it is also pushed
+ // to mixParticipantList with the frame's id. This means
+ // that the Find call above should never fail.
+ if(replaceParticipant == NULL)
+ {
+ assert(false);
+ } else {
+ static_cast<MixerParticipant*>(
+ replaceParticipant->GetItem())->_mixHistory->
+ WasMixed(replaceWasMixed);
+
+ mixParticipantList.Erase(replaceFrame->_id);
+ activeList.Erase(replaceItem);
+
+ activeList.PushFront(static_cast<void*>(audioFrame));
+ mixParticipantList.Insert(
+ audioFrame->_id,
+ static_cast<void*>(participant));
+ assert(mixParticipantList.Size() <=
+ kMaximumAmountOfMixedParticipants);
+
+ if(replaceWasMixed)
+ {
+ RampOut(*replaceFrame);
+ rampOutList.PushBack(
+ static_cast<void*>(replaceFrame));
+ assert(rampOutList.GetSize() <=
+ kMaximumAmountOfMixedParticipants);
+ } else {
+ _audioFramePool->PushMemory(replaceFrame);
+ }
+ }
+ } else {
+ if(wasMixed)
+ {
+ RampOut(*audioFrame);
+ rampOutList.PushBack(static_cast<void*>(audioFrame));
+ assert(rampOutList.GetSize() <=
+ kMaximumAmountOfMixedParticipants);
+ } else {
+ _audioFramePool->PushMemory(audioFrame);
+ }
+ }
+ } else {
+ activeList.PushFront(static_cast<void*>(audioFrame));
+ mixParticipantList.Insert(audioFrame->_id,
+ static_cast<void*>(participant));
+ assert(mixParticipantList.Size() <=
+ kMaximumAmountOfMixedParticipants);
+ }
+ } else {
+ if(wasMixed)
+ {
+ ParticipantFramePair* pair = new ParticipantFramePair;
+ pair->audioFrame = audioFrame;
+ pair->participant = participant;
+ passiveWasMixedList.PushBack(static_cast<void*>(pair));
+ } else if(mustAddToPassiveList) {
+ RampIn(*audioFrame);
+ ParticipantFramePair* pair = new ParticipantFramePair;
+ pair->audioFrame = audioFrame;
+ pair->participant = participant;
+ passiveWasNotMixedList.PushBack(static_cast<void*>(pair));
+ } else {
+ _audioFramePool->PushMemory(audioFrame);
+ }
+ }
+ item = _participantList.Next(item);
+ }
+ assert(activeList.GetSize() <= maxAudioFrameCounter);
+ // At this point it is known which participants should be mixed. Transfer
+ // this information to this functions output parameters.
+ while(!activeList.Empty())
+ {
+ ListItem* mixItem = activeList.First();
+ mixList.PushBack(mixItem->GetItem());
+ activeList.Erase(mixItem);
+ }
+ // Always mix a constant number of AudioFrames. If there aren't enough
+ // active participants mix passive ones. Starting with those that was mixed
+ // last iteration.
+ while(!passiveWasMixedList.Empty())
+ {
+ ListItem* mixItem = passiveWasMixedList.First();
+ ParticipantFramePair* pair = static_cast<ParticipantFramePair*>(
+ mixItem->GetItem());
+ if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize)
+ {
+ mixList.PushBack(pair->audioFrame);
+ mixParticipantList.Insert(pair->audioFrame->_id,
+ static_cast<void*>(pair->participant));
+ assert(mixParticipantList.Size() <=
+ kMaximumAmountOfMixedParticipants);
+ }
+ else
+ {
+ _audioFramePool->PushMemory(pair->audioFrame);
+ }
+ delete pair;
+ passiveWasMixedList.Erase(mixItem);
+ }
+ // And finally the ones that have not been mixed for a while.
+ while(!passiveWasNotMixedList.Empty())
+ {
+ ListItem* mixItem = passiveWasNotMixedList.First();
+ ParticipantFramePair* pair = static_cast<ParticipantFramePair*>(
+ mixItem->GetItem());
+ if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize)
+ {
+ mixList.PushBack(pair->audioFrame);
+ mixParticipantList.Insert(pair->audioFrame->_id,
+ static_cast<void*>(pair->participant));
+ assert(mixParticipantList.Size() <=
+ kMaximumAmountOfMixedParticipants);
+ }
+ else
+ {
+ _audioFramePool->PushMemory(pair->audioFrame);
+ }
+ delete pair;
+ passiveWasNotMixedList.Erase(mixItem);
+ }
+ assert(maxAudioFrameCounter + mixListStartSize >= mixList.GetSize());
+ maxAudioFrameCounter += mixListStartSize - mixList.GetSize();
+}
+
+void AudioConferenceMixerImpl::GetAdditionalAudio(
+ ListWrapper& additionalFramesList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "GetAdditionalAudio(additionalFramesList)");
+ ListItem* item = _additionalParticipantList.First();
+ while(item)
+ {
+ // The GetAudioFrame() callback may remove the current item. Store the
+ // next item just in case that happens.
+ ListItem* nextItem = _additionalParticipantList.Next(item);
+
+ MixerParticipant* participant = static_cast<MixerParticipant*>(
+ item->GetItem());
+ AudioFrame* audioFrame = NULL;
+ if(_audioFramePool->PopMemory(audioFrame) == -1)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
+ "failed PopMemory() call");
+ assert(false);
+ return;
+ }
+ audioFrame->_frequencyInHz = _outputFrequency;
+ if(participant->GetAudioFrame(_id, *audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ "failed to GetAudioFrame() from participant");
+ _audioFramePool->PushMemory(audioFrame);
+ item = nextItem;
+ continue;
+ }
+ if(audioFrame->_payloadDataLengthInSamples == 0)
+ {
+ // Empty frame. Don't use it.
+ _audioFramePool->PushMemory(audioFrame);
+ item = nextItem;
+ continue;
+ }
+ additionalFramesList.PushBack(static_cast<void*>(audioFrame));
+ item = nextItem;
+ }
+}
+
+void AudioConferenceMixerImpl::UpdateMixedStatus(
+ MapWrapper& mixedParticipantsMap)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "UpdateMixedStatus(mixedParticipantsMap)");
+ assert(mixedParticipantsMap.Size() <= kMaximumAmountOfMixedParticipants);
+
+ // Loop through all participants. If they are in the mix map they
+ // were mixed.
+ ListItem* participantItem = _participantList.First();
+ while(participantItem != NULL)
+ {
+ bool isMixed = false;
+ MixerParticipant* participant =
+ static_cast<MixerParticipant*>(participantItem->GetItem());
+
+ MapItem* mixedItem = mixedParticipantsMap.First();
+ while(mixedItem)
+ {
+ if(participant == mixedItem->GetItem())
+ {
+ isMixed = true;
+ break;
+ }
+ mixedItem = mixedParticipantsMap.Next(mixedItem);
+ }
+ participant->_mixHistory->SetIsMixed(isMixed);
+ participantItem = _participantList.Next(participantItem);
+ }
+}
+
+void AudioConferenceMixerImpl::ClearAudioFrameList(ListWrapper& audioFrameList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "ClearAudioFrameList(audioFrameList)");
+ ListItem* item = audioFrameList.First();
+ while(item)
+ {
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+ _audioFramePool->PushMemory(audioFrame);
+ audioFrameList.Erase(item);
+ item = audioFrameList.First();
+ }
+}
+
+void AudioConferenceMixerImpl::UpdateVADPositiveParticipants(
+ ListWrapper& mixList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "UpdateVADPositiveParticipants(mixList)");
+
+ ListItem* item = mixList.First();
+ while(item != NULL)
+ {
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+ CalculateEnergy(*audioFrame);
+ if(audioFrame->_vadActivity == AudioFrame::kVadActive)
+ {
+ _scratchVadPositiveParticipants[
+ _scratchVadPositiveParticipantsAmount].participant =
+ audioFrame->_id;
+ _scratchVadPositiveParticipants[
+ _scratchVadPositiveParticipantsAmount].level =
+ audioFrame->_volume;
+ _scratchVadPositiveParticipantsAmount++;
+ }
+ item = mixList.Next(item);
+ }
+}
+
+bool AudioConferenceMixerImpl::IsParticipantInList(
+ MixerParticipant& participant,
+ ListWrapper& participantList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "IsParticipantInList(participant,participantList)");
+ ListItem* item = participantList.First();
+ while(item != NULL)
+ {
+ MixerParticipant* rhsParticipant =
+ static_cast<MixerParticipant*>(item->GetItem());
+ if(&participant == rhsParticipant)
+ {
+ return true;
+ }
+ item = participantList.Next(item);
+ }
+ return false;
+}
+
+bool AudioConferenceMixerImpl::AddParticipantToList(
+ MixerParticipant& participant,
+ ListWrapper& participantList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "AddParticipantToList(participant, participantList)");
+ if(participantList.PushBack(static_cast<void*>(&participant)) == -1)
+ {
+ return false;
+ }
+ // Make sure that the mixed status is correct for new MixerParticipant.
+ participant._mixHistory->ResetMixedStatus();
+ return true;
+}
+
+bool AudioConferenceMixerImpl::RemoveParticipantFromList(
+ MixerParticipant& participant,
+ ListWrapper& participantList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "RemoveParticipantFromList(participant, participantList)");
+ ListItem* item = participantList.First();
+ while(item)
+ {
+ if(item->GetItem() == &participant)
+ {
+ participantList.Erase(item);
+ // Participant is no longer mixed, reset to default.
+ participant._mixHistory->ResetMixedStatus();
+ return true;
+ }
+ item = participantList.Next(item);
+ }
+ return false;
+}
+
+WebRtc_Word32 AudioConferenceMixerImpl::MixFromList(
+ AudioFrame& mixedAudio,
+ const ListWrapper& audioFrameList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "MixFromList(mixedAudio, audioFrameList)");
+ WebRtc_UWord32 position = 0;
+ ListItem* item = audioFrameList.First();
+ if(item == NULL)
+ {
+ return 0;
+ }
+
+ if(_numMixedParticipants == 1)
+ {
+ // No mixing required here; skip the saturation protection.
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+ mixedAudio = *audioFrame;
+ SetParticipantStatistics(&_scratchMixedParticipants[position],
+ *audioFrame);
+ return 0;
+ }
+
+ while(item != NULL)
+ {
+ if(position >= kMaximumAmountOfMixedParticipants)
+ {
+ WEBRTC_TRACE(
+ kTraceMemory,
+ kTraceAudioMixerServer,
+ _id,
+ "Trying to mix more than max amount of mixed participants:%d!",
+ kMaximumAmountOfMixedParticipants);
+ // Assert and avoid crash
+ assert(false);
+ position = 0;
+ }
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+
+ // Divide by two to avoid saturation in the mixing.
+ *audioFrame >>= 1;
+ mixedAudio += *audioFrame;
+
+ SetParticipantStatistics(&_scratchMixedParticipants[position],
+ *audioFrame);
+
+ position++;
+ item = audioFrameList.Next(item);
+ }
+
+ return 0;
+}
+
+// TODO(andrew): consolidate this function with MixFromList.
+WebRtc_Word32 AudioConferenceMixerImpl::MixAnonomouslyFromList(
+ AudioFrame& mixedAudio,
+ const ListWrapper& audioFrameList)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ "MixAnonomouslyFromList(mixedAudio, audioFrameList)");
+ ListItem* item = audioFrameList.First();
+ if(item == NULL)
+ return 0;
+
+ if(_numMixedParticipants == 1)
+ {
+ // No mixing required here; skip the saturation protection.
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+ mixedAudio = *audioFrame;
+ return 0;
+ }
+
+ while(item != NULL)
+ {
+ AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
+ // Divide by two to avoid saturation in the mixing.
+ *audioFrame >>= 1;
+ mixedAudio += *audioFrame;
+ item = audioFrameList.Next(item);
+ }
+ return 0;
+}
+
+bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio)
+{
+ if(_numMixedParticipants == 1)
+ {
+ return true;
+ }
+
+ // Smoothly limit the mixed frame.
+ const int error = _limiter->ProcessStream(&mixedAudio);
+
+ // And now we can safely restore the level. This procedure results in
+ // some loss of resolution, deemed acceptable.
+ //
+ // It's possible to apply the gain in the AGC (with a target level of 0 dbFS
+ // and compression gain of 6 dB). However, in the transition frame when this
+ // is enabled (moving from one to two participants) it has the potential to
+ // create discontinuities in the mixed frame.
+ //
+ // Instead we double the frame (with addition since left-shifting a
+ // negative value is undefined).
+ mixedAudio += mixedAudio;
+
+ if(error != _limiter->kNoError)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ "Error from AudioProcessing: %d", error);
+ assert(false);
+ return false;
+ }
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
new file mode 100644
index 0000000..efa5d68
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -0,0 +1,209 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
+
+#include "atomic32_wrapper.h"
+#include "audio_conference_mixer.h"
+#include "engine_configurations.h"
+#include "level_indicator.h"
+#include "list_wrapper.h"
+#include "memory_pool.h"
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "time_scheduler.h"
+
+namespace webrtc {
+class AudioProcessing;
+class CriticalSectionWrapper;
+
+// Cheshire cat implementation of MixerParticipant's non virtual functions.
+class MixHistory
+{
+public:
+ MixHistory();
+ ~MixHistory();
+
+ // MixerParticipant function
+ WebRtc_Word32 IsMixed(bool& mixed) const;
+
+ // Sets wasMixed to true if the participant was mixed previous mix
+ // iteration.
+ WebRtc_Word32 WasMixed(bool& wasMixed) const;
+
+ // Updates the mixed status.
+ WebRtc_Word32 SetIsMixed(const bool mixed);
+
+ void ResetMixedStatus();
+private:
+ Atomic32Wrapper _isMixed; // 0 = false, 1 = true
+};
+
+class AudioConferenceMixerImpl : public AudioConferenceMixer
+{
+public:
+ // AudioProcessing only accepts 10 ms frames.
+ enum {kProcessPeriodicityInMs = 10};
+
+ AudioConferenceMixerImpl(int id);
+ ~AudioConferenceMixerImpl();
+
+ // Must be called after ctor.
+ bool Init();
+
+ // Module functions
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ // AudioConferenceMixer functions
+ virtual WebRtc_Word32 RegisterMixedStreamCallback(
+ AudioMixerOutputReceiver& mixReceiver);
+ virtual WebRtc_Word32 UnRegisterMixedStreamCallback();
+ virtual WebRtc_Word32 RegisterMixerStatusCallback(
+ AudioMixerStatusReceiver& mixerStatusCallback,
+ const WebRtc_UWord32 amountOf10MsBetweenCallbacks);
+ virtual WebRtc_Word32 UnRegisterMixerStatusCallback();
+ virtual WebRtc_Word32 SetMixabilityStatus(MixerParticipant& participant,
+ const bool mixable);
+ virtual WebRtc_Word32 MixabilityStatus(MixerParticipant& participant,
+ bool& mixable);
+ virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq);
+ virtual WebRtc_Word32 SetAnonymousMixabilityStatus(
+ MixerParticipant& participant, const bool mixable);
+ virtual WebRtc_Word32 AnonymousMixabilityStatus(
+ MixerParticipant& participant, bool& mixable);
+private:
+ enum{DEFAULT_AUDIO_FRAME_POOLSIZE = 50};
+
+ // Set/get mix frequency
+ WebRtc_Word32 SetOutputFrequency(const Frequency frequency);
+ Frequency OutputFrequency() const;
+
+ // Must be called whenever an audio frame indicates the number of channels
+ // has changed.
+ bool SetNumLimiterChannels(int numChannels);
+
+ // Fills mixList with the AudioFrames pointers that should be used when
+ // mixing. Fills mixParticipantList with ParticipantStatistics for the
+ // participants who's AudioFrames are inside mixList.
+ // maxAudioFrameCounter both input and output specifies how many more
+ // AudioFrames that are allowed to be mixed.
+ // rampOutList contain AudioFrames corresponding to an audio stream that
+ // used to be mixed but shouldn't be mixed any longer. These AudioFrames
+ // should be ramped out over this AudioFrame to avoid audio discontinuities.
+ void UpdateToMix(ListWrapper& mixList, ListWrapper& rampOutList,
+ MapWrapper& mixParticipantList,
+ WebRtc_UWord32& maxAudioFrameCounter);
+
+ // Return the lowest mixing frequency that can be used without having to
+ // downsample any audio.
+ WebRtc_Word32 GetLowestMixingFrequency();
+ WebRtc_Word32 GetLowestMixingFrequencyFromList(ListWrapper& mixList);
+
+ // Return the AudioFrames that should be mixed anonymously.
+ void GetAdditionalAudio(ListWrapper& additionalFramesList);
+
+ // Update the MixHistory of all MixerParticipants. mixedParticipantsList
+ // should contain a map of MixerParticipants that have been mixed.
+ void UpdateMixedStatus(MapWrapper& mixedParticipantsList);
+
+ // Clears audioFrameList and reclaims all memory associated with it.
+ void ClearAudioFrameList(ListWrapper& audioFrameList);
+
+ // Update the list of MixerParticipants who have a positive VAD. mixList
+ // should be a list of AudioFrames
+ void UpdateVADPositiveParticipants(
+ ListWrapper& mixList);
+
+ // This function returns true if it finds the MixerParticipant in the
+ // specified list of MixerParticipants.
+ bool IsParticipantInList(
+ MixerParticipant& participant,
+ ListWrapper& participantList);
+
+ // Add/remove the MixerParticipant to the specified
+ // MixerParticipant list.
+ bool AddParticipantToList(
+ MixerParticipant& participant,
+ ListWrapper& participantList);
+ bool RemoveParticipantFromList(
+ MixerParticipant& removeParticipant,
+ ListWrapper& participantList);
+
+ // Mix the AudioFrames stored in audioFrameList into mixedAudio.
+ WebRtc_Word32 MixFromList(
+ AudioFrame& mixedAudio,
+ const ListWrapper& audioFrameList);
+ // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
+ // record will be kept of this mix (e.g. the corresponding MixerParticipants
+ // will not be marked as IsMixed()
+ WebRtc_Word32 MixAnonomouslyFromList(AudioFrame& mixedAudio,
+ const ListWrapper& audioFrameList);
+
+ bool LimitMixedAudio(AudioFrame& mixedAudio);
+
+ // Scratch memory
+ // Note that the scratch memory may only be touched in the scope of
+ // Process().
+ WebRtc_UWord32 _scratchParticipantsToMixAmount;
+ ParticipantStatistics _scratchMixedParticipants[
+ kMaximumAmountOfMixedParticipants];
+ WebRtc_UWord32 _scratchVadPositiveParticipantsAmount;
+ ParticipantStatistics _scratchVadPositiveParticipants[
+ kMaximumAmountOfMixedParticipants];
+
+ scoped_ptr<CriticalSectionWrapper> _crit;
+ scoped_ptr<CriticalSectionWrapper> _cbCrit;
+
+ WebRtc_Word32 _id;
+
+ Frequency _minimumMixingFreq;
+
+ // Mix result callback
+ AudioMixerOutputReceiver* _mixReceiver;
+
+ AudioMixerStatusReceiver* _mixerStatusCallback;
+ WebRtc_UWord32 _amountOf10MsBetweenCallbacks;
+ WebRtc_UWord32 _amountOf10MsUntilNextCallback;
+ bool _mixerStatusCb;
+
+ // The current sample frequency and sample size when mixing.
+ Frequency _outputFrequency;
+ WebRtc_UWord16 _sampleSize;
+
+ // Memory pool to avoid allocating/deallocating AudioFrames
+ MemoryPool<AudioFrame>* _audioFramePool;
+
+ // List of all participants. Note all lists are disjunct
+ ListWrapper _participantList; // May be mixed.
+ ListWrapper _additionalParticipantList; // Always mixed, anonomously.
+
+ WebRtc_UWord32 _numMixedParticipants;
+
+ WebRtc_UWord32 _timeStamp;
+
+ // Metronome class.
+ TimeScheduler _timeScheduler;
+
+ // Smooth level indicator.
+ LevelIndicator _mixedAudioLevel;
+
+ // Counter keeping track of concurrent calls to process.
+ // Note: should never be higher than 1 or lower than 0.
+ WebRtc_Word16 _processCalls;
+
+ // Used for inhibiting saturation in mixing.
+ scoped_ptr<AudioProcessing> _limiter;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc
new file mode 100644
index 0000000..f895fbd
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "audio_conference_mixer.h"
+#include "gtest/gtest.h"
+
+TEST(AudioConferenceMixerTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
new file mode 100644
index 0000000..8716454
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_frame_manipulator.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+
+namespace {
+// Linear ramping over 80 samples.
+// TODO(hellner): ramp using fix point?
+const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
+ 0.0506f, 0.0633f, 0.0759f, 0.0886f,
+ 0.1013f, 0.1139f, 0.1266f, 0.1392f,
+ 0.1519f, 0.1646f, 0.1772f, 0.1899f,
+ 0.2025f, 0.2152f, 0.2278f, 0.2405f,
+ 0.2532f, 0.2658f, 0.2785f, 0.2911f,
+ 0.3038f, 0.3165f, 0.3291f, 0.3418f,
+ 0.3544f, 0.3671f, 0.3797f, 0.3924f,
+ 0.4051f, 0.4177f, 0.4304f, 0.4430f,
+ 0.4557f, 0.4684f, 0.4810f, 0.4937f,
+ 0.5063f, 0.5190f, 0.5316f, 0.5443f,
+ 0.5570f, 0.5696f, 0.5823f, 0.5949f,
+ 0.6076f, 0.6203f, 0.6329f, 0.6456f,
+ 0.6582f, 0.6709f, 0.6835f, 0.6962f,
+ 0.7089f, 0.7215f, 0.7342f, 0.7468f,
+ 0.7595f, 0.7722f, 0.7848f, 0.7975f,
+ 0.8101f, 0.8228f, 0.8354f, 0.8481f,
+ 0.8608f, 0.8734f, 0.8861f, 0.8987f,
+ 0.9114f, 0.9241f, 0.9367f, 0.9494f,
+ 0.9620f, 0.9747f, 0.9873f, 1.0000f};
+const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
+} // namespace
+
+namespace webrtc {
+void CalculateEnergy(AudioFrame& audioFrame)
+{
+ if(audioFrame._energy != 0xffffffff)
+ {
+ return;
+ }
+ audioFrame._energy = 0;
+ for(int position = 0; position < audioFrame._payloadDataLengthInSamples;
+ position++)
+ {
+ // TODO(andrew): this can easily overflow.
+ audioFrame._energy += audioFrame._payloadData[position] *
+ audioFrame._payloadData[position];
+ }
+}
+
+void RampIn(AudioFrame& audioFrame)
+{
+ assert(rampSize <= audioFrame._payloadDataLengthInSamples);
+ for(int i = 0; i < rampSize; i++)
+ {
+ audioFrame._payloadData[i] = static_cast<WebRtc_Word16>
+ (rampArray[i] * audioFrame._payloadData[i]);
+ }
+}
+
+void RampOut(AudioFrame& audioFrame)
+{
+ assert(rampSize <= audioFrame._payloadDataLengthInSamples);
+ for(int i = 0; i < rampSize; i++)
+ {
+ const int rampPos = rampSize - 1 - i;
+ audioFrame._payloadData[i] = static_cast<WebRtc_Word16>
+ (rampArray[rampPos] * audioFrame._payloadData[i]);
+ }
+ memset(&audioFrame._payloadData[rampSize], 0,
+ (audioFrame._payloadDataLengthInSamples - rampSize) *
+ sizeof(audioFrame._payloadData[0]));
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h
new file mode 100644
index 0000000..fdf5d33
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
+
+namespace webrtc {
+class AudioFrame;
+
+// Updates the audioFrame's energy (based on its samples).
+void CalculateEnergy(AudioFrame& audioFrame);
+
+// Apply linear step function that ramps in/out the audio samples in audioFrame
+void RampIn(AudioFrame& audioFrame);
+void RampOut(AudioFrame& audioFrame);
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc b/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc
new file mode 100644
index 0000000..799a47d
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "level_indicator.h"
+
+namespace webrtc {
+// Array for adding smothing to level changes (ad-hoc).
+const WebRtc_UWord32 perm[] =
+ {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
+
+LevelIndicator::LevelIndicator()
+ : _max(0),
+ _count(0),
+ _currentLevel(0)
+{
+}
+
+LevelIndicator::~LevelIndicator()
+{
+}
+
+// Level is based on the highest absolute value for all samples.
+void LevelIndicator::ComputeLevel(const WebRtc_Word16* speech,
+ const WebRtc_UWord16 nrOfSamples)
+{
+ WebRtc_Word32 min = 0;
+ for(WebRtc_UWord32 i = 0; i < nrOfSamples; i++)
+ {
+ if(_max < speech[i])
+ {
+ _max = speech[i];
+ }
+ if(min > speech[i])
+ {
+ min = speech[i];
+ }
+ }
+
+ // Absolute max value.
+ if(-min > _max)
+ {
+ _max = -min;
+ }
+
+ if(_count == TICKS_BEFORE_CALCULATION)
+ {
+ // Highest sample value maps directly to a level.
+ WebRtc_Word32 position = _max / 1000;
+ if ((position == 0) &&
+ (_max > 250))
+ {
+ position = 1;
+ }
+ _currentLevel = perm[position];
+ // The max value is decayed and stored so that it can be reused to slow
+ // down decreases in level.
+ _max = _max >> 1;
+ _count = 0;
+ } else {
+ _count++;
+ }
+}
+
+WebRtc_Word32 LevelIndicator::GetLevel()
+{
+ return _currentLevel;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/level_indicator.h b/trunk/src/modules/audio_conference_mixer/source/level_indicator.h
new file mode 100644
index 0000000..bdcdf8e
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/level_indicator.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class LevelIndicator
+{
+public:
+ enum{TICKS_BEFORE_CALCULATION = 10};
+
+ LevelIndicator();
+ ~LevelIndicator();
+
+ // Updates the level.
+ void ComputeLevel(const WebRtc_Word16* speech,
+ const WebRtc_UWord16 nrOfSamples);
+
+ WebRtc_Word32 GetLevel();
+private:
+ WebRtc_Word32 _max;
+ WebRtc_UWord32 _count;
+ WebRtc_UWord32 _currentLevel;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool.h
new file mode 100644
index 0000000..caf5d93
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/memory_pool.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
+
+#include <assert.h>
+
+#include "typedefs.h"
+
+#if _WIN32
+#include "memory_pool_win.h"
+#else
+#include "memory_pool_posix.h"
+#endif
+
+namespace webrtc {
+
+template<class MemoryType>
+class MemoryPool
+{
+public:
+ // Factory method, constructor disabled.
+ static WebRtc_Word32 CreateMemoryPool(MemoryPool*& memoryPool,
+ WebRtc_UWord32 initialPoolSize);
+
+ // Try to delete the memory pool. Fail with return value -1 if there is
+ // outstanding memory.
+ static WebRtc_Word32 DeleteMemoryPool(
+ MemoryPool*& memoryPool);
+
+ // Get/return unused memory.
+ WebRtc_Word32 PopMemory(MemoryType*& memory);
+ WebRtc_Word32 PushMemory(MemoryType*& memory);
+private:
+ MemoryPool(WebRtc_Word32 initialPoolSize);
+ ~MemoryPool();
+
+ MemoryPoolImpl<MemoryType>* _ptrImpl;
+};
+
+template<class MemoryType>
+MemoryPool<MemoryType>::MemoryPool(WebRtc_Word32 initialPoolSize)
+{
+ _ptrImpl = new MemoryPoolImpl<MemoryType>(initialPoolSize);
+}
+
+template<class MemoryType>
+MemoryPool<MemoryType>::~MemoryPool()
+{
+ delete _ptrImpl;
+}
+
+template<class MemoryType> WebRtc_Word32
+MemoryPool<MemoryType>::CreateMemoryPool(MemoryPool*& memoryPool,
+ WebRtc_UWord32 initialPoolSize)
+{
+ memoryPool = new MemoryPool(initialPoolSize);
+ if(memoryPool == NULL)
+ {
+ return -1;
+ }
+ if(memoryPool->_ptrImpl == NULL)
+ {
+ delete memoryPool;
+ memoryPool = NULL;
+ return -1;
+ }
+ if(!memoryPool->_ptrImpl->Initialize())
+ {
+ delete memoryPool;
+ memoryPool = NULL;
+ return -1;
+ }
+ return 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPool<MemoryType>::DeleteMemoryPool(MemoryPool*& memoryPool)
+{
+ if(memoryPool == NULL)
+ {
+ return -1;
+ }
+ if(memoryPool->_ptrImpl == NULL)
+ {
+ return -1;
+ }
+ if(memoryPool->_ptrImpl->Terminate() == -1)
+ {
+ return -1;
+ }
+ delete memoryPool;
+ memoryPool = NULL;
+ return 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPool<MemoryType>::PopMemory(MemoryType*& memory)
+{
+ return _ptrImpl->PopMemory(memory);
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPool<MemoryType>::PushMemory(MemoryType*& memory)
+{
+ if(memory == NULL)
+ {
+ return -1;
+ }
+ return _ptrImpl->PushMemory(memory);
+}
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h
new file mode 100644
index 0000000..45f800b
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h
@@ -0,0 +1,168 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
+
+#include <assert.h>
+
+#include "critical_section_wrapper.h"
+#include "list_wrapper.h"
+#include "typedefs.h"
+
+namespace webrtc {
+template<class MemoryType>
+class MemoryPoolImpl
+{
+public:
+ // MemoryPool functions.
+ WebRtc_Word32 PopMemory(MemoryType*& memory);
+ WebRtc_Word32 PushMemory(MemoryType*& memory);
+
+ MemoryPoolImpl(WebRtc_Word32 initialPoolSize);
+ ~MemoryPoolImpl();
+
+ // Atomic functions
+ WebRtc_Word32 Terminate();
+ bool Initialize();
+private:
+ // Non-atomic function.
+ WebRtc_Word32 CreateMemory(WebRtc_UWord32 amountToCreate);
+
+ CriticalSectionWrapper* _crit;
+
+ bool _terminate;
+
+ ListWrapper _memoryPool;
+
+ WebRtc_UWord32 _initialPoolSize;
+ WebRtc_UWord32 _createdMemory;
+ WebRtc_UWord32 _outstandingMemory;
+};
+
+template<class MemoryType>
+MemoryPoolImpl<MemoryType>::MemoryPoolImpl(WebRtc_Word32 initialPoolSize)
+ : _crit(CriticalSectionWrapper::CreateCriticalSection()),
+ _terminate(false),
+ _memoryPool(),
+ _initialPoolSize(initialPoolSize),
+ _createdMemory(0),
+ _outstandingMemory(0)
+{
+}
+
+template<class MemoryType>
+MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
+{
+ // Trigger assert if there is outstanding memory.
+ assert(_createdMemory == 0);
+ assert(_outstandingMemory == 0);
+ delete _crit;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
+{
+ CriticalSectionScoped cs(_crit);
+ if(_terminate)
+ {
+ memory = NULL;
+ return -1;
+ }
+ ListItem* item = _memoryPool.First();
+ if(item == NULL)
+ {
+ // _memoryPool empty create new memory.
+ CreateMemory(_initialPoolSize);
+ item = _memoryPool.First();
+ if(item == NULL)
+ {
+ memory = NULL;
+ return -1;
+ }
+ }
+ memory = static_cast<MemoryType*>(item->GetItem());
+ _memoryPool.Erase(item);
+ _outstandingMemory++;
+ return 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
+{
+ if(memory == NULL)
+ {
+ return -1;
+ }
+ CriticalSectionScoped cs(_crit);
+ _outstandingMemory--;
+ if(_memoryPool.GetSize() > (_initialPoolSize << 1))
+ {
+ // Reclaim memory if less than half of the pool is unused.
+ _createdMemory--;
+ delete memory;
+ memory = NULL;
+ return 0;
+ }
+ _memoryPool.PushBack(static_cast<void*>(memory));
+ memory = NULL;
+ return 0;
+}
+
+template<class MemoryType>
+bool MemoryPoolImpl<MemoryType>::Initialize()
+{
+ CriticalSectionScoped cs(_crit);
+ return CreateMemory(_initialPoolSize) == 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::Terminate()
+{
+ CriticalSectionScoped cs(_crit);
+ assert(_createdMemory == _outstandingMemory + _memoryPool.GetSize());
+
+ _terminate = true;
+ // Reclaim all memory.
+ while(_createdMemory > 0)
+ {
+ ListItem* item = _memoryPool.First();
+ if(item == NULL)
+ {
+ // There is memory that hasn't been returned yet.
+ return -1;
+ }
+ MemoryType* memory = static_cast<MemoryType*>(item->GetItem());
+ delete memory;
+ _memoryPool.Erase(item);
+ _createdMemory--;
+ }
+ return 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::CreateMemory(
+ WebRtc_UWord32 amountToCreate)
+{
+ for(WebRtc_UWord32 i = 0; i < amountToCreate; i++)
+ {
+ MemoryType* memory = new MemoryType();
+ if(memory == NULL)
+ {
+ return -1;
+ }
+ _memoryPool.PushBack(static_cast<void*>(memory));
+ _createdMemory++;
+ }
+ return 0;
+}
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h
new file mode 100644
index 0000000..1275ca1
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h
@@ -0,0 +1,199 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
+
+#include <assert.h>
+#include <windows.h>
+
+#include "aligned_malloc.h"
+#include "atomic32_wrapper.h"
+#include "typedefs.h"
+
+namespace webrtc {
+template<class MemoryType> struct MemoryPoolItem;
+
+template<class MemoryType>
+struct MemoryPoolItemPayload
+{
+ MemoryPoolItemPayload()
+ : memoryType(),
+ base(NULL)
+ {
+ }
+ MemoryType memoryType;
+ MemoryPoolItem<MemoryType>* base;
+};
+
+template<class MemoryType>
+struct MemoryPoolItem
+{
+ // Atomic single linked list entry header.
+ SLIST_ENTRY itemEntry;
+ // Atomic single linked list payload.
+ MemoryPoolItemPayload<MemoryType>* payload;
+};
+
+template<class MemoryType>
+class MemoryPoolImpl
+{
+public:
+ // MemoryPool functions.
+ WebRtc_Word32 PopMemory(MemoryType*& memory);
+ WebRtc_Word32 PushMemory(MemoryType*& memory);
+
+ MemoryPoolImpl(WebRtc_Word32 /*initialPoolSize*/);
+ ~MemoryPoolImpl();
+
+ // Atomic functions.
+ WebRtc_Word32 Terminate();
+ bool Initialize();
+private:
+ // Non-atomic function.
+ MemoryPoolItem<MemoryType>* CreateMemory();
+
+ // Windows implementation of single linked atomic list, documented here:
+ // http://msdn.microsoft.com/en-us/library/ms686962(VS.85).aspx
+
+ // Atomic single linked list head.
+ PSLIST_HEADER _pListHead;
+
+ Atomic32Wrapper _createdMemory;
+ Atomic32Wrapper _outstandingMemory;
+};
+
+template<class MemoryType>
+MemoryPoolImpl<MemoryType>::MemoryPoolImpl(
+ WebRtc_Word32 /*initialPoolSize*/)
+ : _pListHead(NULL),
+ _createdMemory(0),
+ _outstandingMemory(0)
+{
+}
+
+template<class MemoryType>
+MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
+{
+ Terminate();
+ if(_pListHead != NULL)
+ {
+ AlignedFree(reinterpret_cast<void*>(_pListHead));
+ _pListHead = NULL;
+ }
+ // Trigger assert if there is outstanding memory.
+ assert(_createdMemory.Value() == 0);
+ assert(_outstandingMemory.Value() == 0);
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
+{
+ PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
+ if(pListEntry == NULL)
+ {
+ MemoryPoolItem<MemoryType>* item = CreateMemory();
+ if(item == NULL)
+ {
+ return -1;
+ }
+ pListEntry = &(item->itemEntry);
+ }
+ ++_outstandingMemory;
+ memory = &((MemoryPoolItem<MemoryType>*)pListEntry)->payload->memoryType;
+ return 0;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
+{
+ if(memory == NULL)
+ {
+ return -1;
+ }
+
+ MemoryPoolItem<MemoryType>* item =
+ ((MemoryPoolItemPayload<MemoryType>*)memory)->base;
+
+ const WebRtc_Word32 usedItems = --_outstandingMemory;
+ const WebRtc_Word32 totalItems = _createdMemory.Value();
+ const WebRtc_Word32 freeItems = totalItems - usedItems;
+ if(freeItems < 0)
+ {
+ assert(false);
+ delete item->payload;
+ AlignedFree(item);
+ return -1;
+ }
+ if(freeItems >= totalItems>>1)
+ {
+ delete item->payload;
+ AlignedFree(item);
+ --_createdMemory;
+ return 0;
+ }
+ InterlockedPushEntrySList(_pListHead,&(item->itemEntry));
+ return 0;
+}
+
+template<class MemoryType>
+bool MemoryPoolImpl<MemoryType>::Initialize()
+{
+ _pListHead = (PSLIST_HEADER)AlignedMalloc(sizeof(SLIST_HEADER),
+ MEMORY_ALLOCATION_ALIGNMENT);
+ if(_pListHead == NULL)
+ {
+ return false;
+ }
+ InitializeSListHead(_pListHead);
+ return true;
+}
+
+template<class MemoryType>
+WebRtc_Word32 MemoryPoolImpl<MemoryType>::Terminate()
+{
+ WebRtc_Word32 itemsFreed = 0;
+ PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
+ while(pListEntry != NULL)
+ {
+ MemoryPoolItem<MemoryType>* item = ((MemoryPoolItem<MemoryType>*)pListEntry);
+ delete item->payload;
+ AlignedFree(item);
+ --_createdMemory;
+ itemsFreed++;
+ pListEntry = InterlockedPopEntrySList(_pListHead);
+ }
+ return itemsFreed;
+}
+
+template<class MemoryType>
+MemoryPoolItem<MemoryType>* MemoryPoolImpl<MemoryType>::CreateMemory()
+{
+ MemoryPoolItem<MemoryType>* returnValue = (MemoryPoolItem<MemoryType>*)
+ AlignedMalloc(sizeof(MemoryPoolItem<MemoryType>),
+ MEMORY_ALLOCATION_ALIGNMENT);
+ if(returnValue == NULL)
+ {
+ return NULL;
+ }
+
+ returnValue->payload = new MemoryPoolItemPayload<MemoryType>();
+ if(returnValue->payload == NULL)
+ {
+ delete returnValue;
+ return NULL;
+ }
+ returnValue->payload->base = returnValue;
+ ++_createdMemory;
+ return returnValue;
+}
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc
new file mode 100644
index 0000000..183005e
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "critical_section_wrapper.h"
+#include "time_scheduler.h"
+
+namespace webrtc {
+TimeScheduler::TimeScheduler(const WebRtc_UWord32 periodicityInMs)
+ : _crit(CriticalSectionWrapper::CreateCriticalSection()),
+ _isStarted(false),
+ _lastPeriodMark(),
+ _periodicityInMs(periodicityInMs),
+ _periodicityInTicks(TickTime::MillisecondsToTicks(periodicityInMs)),
+ _missedPeriods(0)
+ {
+ }
+
+TimeScheduler::~TimeScheduler()
+{
+ delete _crit;
+}
+
+WebRtc_Word32 TimeScheduler::UpdateScheduler()
+{
+ CriticalSectionScoped cs(_crit);
+ if(!_isStarted)
+ {
+ _isStarted = true;
+ _lastPeriodMark = TickTime::Now();
+ return 0;
+ }
+ // Don't perform any calculations until the debt of pending periods have
+ // been worked off.
+ if(_missedPeriods > 0)
+ {
+ _missedPeriods--;
+ return 0;
+ }
+
+ // Calculate the time that has past since previous call to this function.
+ TickTime tickNow = TickTime::Now();
+ TickInterval amassedTicks = tickNow - _lastPeriodMark;
+ WebRtc_Word64 amassedMs = amassedTicks.Milliseconds();
+
+ // Calculate the number of periods the time that has passed correspond to.
+ WebRtc_Word32 periodsToClaim = (WebRtc_Word32)amassedMs /
+ ((WebRtc_Word32)_periodicityInMs);
+
+ // One period will be worked off by this call. Make sure that the number of
+ // pending periods don't end up being negative (e.g. if this function is
+ // called to often).
+ if(periodsToClaim < 1)
+ {
+ periodsToClaim = 1;
+ }
+
+ // Update the last period mark without introducing any drifting.
+ // Note that if this fuunction is called to often _lastPeriodMark can
+ // refer to a time in the future which in turn will yield TimeToNextUpdate
+ // that is greater than the periodicity
+ for(WebRtc_Word32 i = 0; i < periodsToClaim; i++)
+ {
+ _lastPeriodMark += _periodicityInTicks;
+ }
+
+ // Update the total amount of missed periods note that we have processed
+ // one period hence the - 1
+ _missedPeriods += periodsToClaim - 1;
+ return 0;
+}
+
+WebRtc_Word32 TimeScheduler::TimeToNextUpdate(
+ WebRtc_Word32& updateTimeInMS) const
+{
+ CriticalSectionScoped cs(_crit);
+ // Missed periods means that the next UpdateScheduler() should happen
+ // immediately.
+ if(_missedPeriods > 0)
+ {
+ updateTimeInMS = 0;
+ return 0;
+ }
+
+ // Calculate the time (in ms) that has past since last call to
+ // UpdateScheduler()
+ TickTime tickNow = TickTime::Now();
+ TickInterval ticksSinceLastUpdate = tickNow - _lastPeriodMark;
+ const WebRtc_Word32 millisecondsSinceLastUpdate =
+ (WebRtc_Word32) ticksSinceLastUpdate.Milliseconds();
+
+ updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate;
+ updateTimeInMS = (updateTimeInMS < 0) ? 0 : updateTimeInMS;
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h
new file mode 100644
index 0000000..e2674d9
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// The TimeScheduler class keeps track of periodic events. It is non-drifting
+// and keeps track of any missed periods so that it is possible to catch up.
+// (compare to a metronome)
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
+
+#include "tick_util.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class TimeScheduler
+{
+public:
+ TimeScheduler(const WebRtc_UWord32 periodicityInMs);
+ ~TimeScheduler();
+
+ // Signal that a periodic event has been triggered.
+ WebRtc_Word32 UpdateScheduler();
+
+ // Set updateTimeInMs to the amount of time until UpdateScheduler() should
+ // be called. This time will never be negative.
+ WebRtc_Word32 TimeToNextUpdate(WebRtc_Word32& updateTimeInMS) const;
+
+private:
+ CriticalSectionWrapper* _crit;
+
+ bool _isStarted;
+ TickTime _lastPeriodMark;
+
+ WebRtc_UWord32 _periodicityInMs;
+ WebRtc_Word64 _periodicityInTicks;
+ WebRtc_UWord32 _missedPeriods;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
diff --git a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc
new file mode 100644
index 0000000..e34a3cb
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc
@@ -0,0 +1,1098 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <iostream>
+#include <time.h>
+
+#include "functionTest.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "webrtc_vad.h"
+
+#if (defined(WEBRTC_LINUX) || defined(WEBRTC_MAC))
+ #include <sys/stat.h>
+ #define MY_PERMISSION_MASK S_IRWXU | S_IRWXG | S_IRWXO
+ #define MKDIR(directory) mkdir(directory,MY_PERMISSION_MASK)
+#else // defined(WINDOWS)
+ #include <direct.h>
+ #define MKDIR(directory) mkdir(directory)
+#endif
+
+int main(int /*argc*/, char* /*argv[]*/)
+{
+ // Initialize random number generator
+ //unsigned int seed = 1220716312; // just a seed that can be used
+ unsigned int seed = (unsigned)time( NULL );
+ srand(seed);
+ std::cout << "Starting function test. Seed = " << seed << std::endl;
+ std::cout << "Press enter to continue" << std::endl;
+ getchar();
+ MixerWrapper* testInstance1 = MixerWrapper::CreateMixerWrapper();
+ MixerWrapper* testInstance2 = MixerWrapper::CreateMixerWrapper();
+ if((testInstance1 == NULL) ||
+ (testInstance2 == NULL))
+ {
+ assert(false);
+ return 0;
+ }
+
+ char versionString[256] = "";
+ WebRtc_UWord32 remainingBufferInBytes = 256;
+ WebRtc_UWord32 position = 0;
+ AudioConferenceMixer::GetVersion(versionString,remainingBufferInBytes,position);
+
+ int read = 1;
+ while(read != 0)
+ {
+ std::cout << versionString << std::endl;
+ std::cout << "--------Menu-----------" << std::endl;
+ std::cout << std::endl;
+ std::cout << "0. Quit" << std::endl;
+ std::cout << "2. StartMixing" << std::endl;
+ std::cout << "3. StopMixing" << std::endl;
+ std::cout << "4. Create participant(s)" << std::endl;
+ std::cout << "5. Delete participant(s)" << std::endl;
+ std::cout << "6. List participants " << std::endl;
+ std::cout << "7. Print mix status " << std::endl;
+ std::cout << "8. Run identical scenario:" << std::endl;
+ std::cout << " a. 1 VIP, 3 regular, amount of mixed = 3" << std::endl;
+ std::cout << " b. 1 anonymous, 3 regular, amount of mixed = 2" << std::endl;
+ scanf("%i",&read);
+ getchar();
+ MixerParticipant::ParticipantType participantType;
+ int option = 0;
+ WebRtc_UWord32 id = 0;
+ ListItem* item = NULL;
+ ListWrapper participants;
+ if(read == 0)
+ {
+ // donothing
+ }
+ else if(read == 1)
+ {
+ }
+ else if(read == 2)
+ {
+ testInstance1->StartMixing();
+ }
+ else if(read == 3)
+ {
+ testInstance1->StopMixing();
+ }
+ else if(read == 4)
+ {
+ while(true)
+ {
+ std::cout << "VIP(music) = " << MixerParticipant::VIP << std::endl;
+ std::cout << "Regular(speech) = " << MixerParticipant::REGULAR << std::endl;
+ std::cout << "Anonymous(music) = " << MixerParticipant::MIXED_ANONYMOUS << std::endl;
+ std::cout << "Select type of participant: ";
+ scanf("%i",&option);
+ if(option == MixerParticipant::VIP ||
+ option == MixerParticipant::REGULAR ||
+ option == MixerParticipant::MIXED_ANONYMOUS)
+ {
+ break;
+ }
+ }
+ participantType = (MixerParticipant::ParticipantType)option;
+ testInstance1->CreateParticipant(participantType);
+ }
+ else if(read == 5)
+ {
+ std::cout << "Select participant to delete: ";
+ scanf("%i",&option);
+ id = option;
+ testInstance1->DeleteParticipant(id);
+ break;
+ }
+ else if(read == 6)
+ {
+ testInstance1->GetParticipantList(participants);
+ item = participants.First();
+ std::cout << "The following participants have been created: " << std::endl;
+ while(item)
+ {
+ WebRtc_UWord32 id = item->GetUnsignedItem();
+ std::cout << id;
+ item = participants.Next(item);
+ if(item != NULL)
+ {
+ std::cout << ", ";
+ }
+ else
+ {
+ std::cout << std::endl;
+ }
+ }
+ }
+ else if(read == 7)
+ {
+ std::cout << "-------------Mixer Status-------------" << std::endl;
+ testInstance1->PrintStatus();
+ testInstance2->PrintStatus();
+ std::cout << "Press enter to continue";
+ getchar();
+ std::cout << std::endl;
+ std::cout << std::endl;
+ }
+ else if(read == 8)
+ {
+ const WebRtc_Word32 amountOfParticipants = 4;
+ MixerParticipant::ParticipantType instance1Participants[] =
+ {MixerParticipant::VIP,
+ MixerParticipant::REGULAR,
+ MixerParticipant::REGULAR,
+ MixerParticipant::REGULAR};
+ MixerParticipant::ParticipantType instance2Participants[] =
+ {MixerParticipant::MIXED_ANONYMOUS,
+ MixerParticipant::REGULAR,
+ MixerParticipant::REGULAR,
+ MixerParticipant::REGULAR};
+ for(WebRtc_Word32 i = 0; i < amountOfParticipants; i++)
+ {
+ WebRtc_Word32 startPosition = 0;
+ GenerateRandomPosition(startPosition);
+ testInstance1->CreateParticipant(instance1Participants[i],startPosition);
+ testInstance2->CreateParticipant(instance2Participants[i],startPosition);
+ }
+ bool success = true;
+ success = testInstance1->StartMixing();
+ assert(success);
+ success = testInstance2->StartMixing(2);
+ assert(success);
+ }
+ }
+
+ std::cout << "Press enter to stop" << std::endl;
+ getchar();
+ delete testInstance1;
+ delete testInstance2;
+ return 0;
+}
+
+FileWriter::FileWriter()
+ :
+ _file(NULL)
+{
+}
+
+FileWriter::~FileWriter()
+{
+ if(_file)
+ {
+ fclose(_file);
+ }
+}
+
+bool
+FileWriter::SetFileName(
+ const char* fileName)
+{
+ if(_file)
+ {
+ fclose(_file);
+ }
+ _file = fopen(fileName,"wb");
+ return _file != NULL;
+}
+
+bool
+FileWriter::WriteToFile(
+ const AudioFrame& audioFrame)
+{
+ WebRtc_Word32 written = (WebRtc_Word32)fwrite(audioFrame._payloadData,sizeof(WebRtc_Word16),audioFrame._payloadDataLengthInSamples,_file);
+ // Do not flush buffers since that will add (a lot of) delay
+ return written == audioFrame._payloadDataLengthInSamples;
+}
+
+FileReader::FileReader()
+ :
+ _frequency(kDefaultFrequency),
+ _sampleSize((_frequency*kProcessPeriodicityInMs)/1000),
+ _timeStamp(0),
+ _file(NULL),
+ _vadInstr(NULL),
+ _automaticVad(false),
+ _vad(false)
+{
+ if(WebRtcVad_Create(&_vadInstr) == 0)
+ {
+ if(WebRtcVad_Init(_vadInstr) != 0)
+ {
+ assert(false);
+ WebRtcVad_Free(_vadInstr);
+ _vadInstr = NULL;
+ }
+ }
+ else
+ {
+ assert(false);
+ }
+}
+
+FileReader::~FileReader()
+{
+ if(_file)
+ {
+ fclose(_file);
+ }
+ if(_vadInstr)
+ {
+ WebRtcVad_Free(_vadInstr);
+ }
+}
+
+bool
+FileReader::SetFileName(
+ const char* fileName)
+{
+ if(_file)
+ {
+ fclose(_file);
+ }
+ _file = fopen(fileName,"rb");
+ return _file != NULL;
+}
+
+bool
+FileReader::ReadFromFile(
+ AudioFrame& audioFrame)
+{
+
+ WebRtc_Word16 buffer[AudioFrame::kMaxAudioFrameSizeSamples];
+ LoopedFileRead(buffer,AudioFrame::kMaxAudioFrameSizeSamples,_sampleSize,_file);
+
+ bool vad = false;
+ GetVAD(buffer,_sampleSize,vad);
+ AudioFrame::VADActivity activity = vad ? AudioFrame::kVadActive :
+ AudioFrame::kVadPassive;
+
+ _volumeCalculator.ComputeLevel(buffer,_sampleSize);
+ const WebRtc_Word32 level = _volumeCalculator.GetLevel();
+ return audioFrame.UpdateFrame( -1,
+ _timeStamp,
+ buffer,
+ _sampleSize,
+ _frequency,
+ AudioFrame::kNormalSpeech,
+ activity,
+ 0,
+ level) == 0;
+
+}
+
+bool
+FileReader::FastForwardFile(
+ const WebRtc_Word32 samples)
+{
+ WebRtc_Word16* tempBuffer = new WebRtc_Word16[samples];
+ bool success = LoopedFileRead(tempBuffer,samples,samples,_file);
+ delete[] tempBuffer;
+ return success;
+}
+
+bool
+FileReader::EnableAutomaticVAD(
+ bool enable,
+ int mode)
+{
+ if(!_automaticVad &&
+ enable)
+ {
+ if(WebRtcVad_Init(_vadInstr) == -1)
+ {
+ return false;
+ }
+ }
+ WebRtcVad_set_mode(_vadInstr,mode);
+ _automaticVad = enable;
+ return true;
+}
+
+bool
+FileReader::SetVAD(
+ bool vad)
+{
+ if(_automaticVad)
+ {
+ return false;
+ }
+ _vad = vad;
+ return true;
+}
+
+bool
+FileReader::GetVAD(
+ WebRtc_Word16* buffer,
+ WebRtc_UWord8 bufferLengthInSamples,
+ bool& vad)
+{
+ if(_automaticVad)
+ {
+ WebRtc_Word16 result = WebRtcVad_Process(_vadInstr,_frequency,buffer,bufferLengthInSamples);
+ if(result == -1)
+ {
+ assert(false);
+ return false;
+ }
+ _vad = vad = (result == 1);
+ }
+ vad = _vad;
+ return true;
+}
+
+MixerParticipant*
+MixerParticipant::CreateParticipant(
+ const WebRtc_UWord32 id,
+ ParticipantType participantType,
+ const WebRtc_Word32 startPosition,
+ char* outputPath)
+{
+ if(participantType == RANDOM)
+ {
+ participantType = (ParticipantType)(rand() % 3);
+ }
+ MixerParticipant* participant = new MixerParticipant(id,participantType);
+ // Randomize the start position so we only need one input file
+ // assume file is smaller than 1 minute wideband = 60 * 16000
+ // Always start at a multiple of 10ms wideband
+ if(!participant->InitializeFileReader(startPosition) ||
+ !participant->InitializeFileWriter(outputPath))
+ {
+ delete participant;
+ return NULL;
+ }
+ return participant;
+}
+
+MixerParticipant::MixerParticipant(
+ const WebRtc_UWord32 id,
+ ParticipantType participantType)
+ :
+ _id(id),
+ _participantType(participantType),
+ _fileReader(),
+ _fileWriter()
+{
+}
+
+MixerParticipant::~MixerParticipant()
+{
+}
+
+WebRtc_Word32
+MixerParticipant::GetAudioFrame(
+ const WebRtc_Word32 /*id*/,
+ AudioFrame& audioFrame)
+{
+ if(!_fileReader.ReadFromFile(audioFrame))
+ {
+ return -1;
+ }
+ audioFrame._id = _id;
+ return 0;
+}
+
+WebRtc_Word32
+MixerParticipant::MixedAudioFrame(
+ const AudioFrame& audioFrame)
+{
+ return _fileWriter.WriteToFile(audioFrame);
+}
+
+WebRtc_Word32
+MixerParticipant::GetParticipantType(
+ ParticipantType& participantType)
+{
+ participantType = _participantType;
+ return 0;
+}
+
+bool
+MixerParticipant::InitializeFileReader(
+ const WebRtc_Word32 startPositionInSamples)
+{
+ char fileName[128] = "";
+ if(_participantType == REGULAR)
+ {
+ sprintf(fileName,"convFile.pcm");
+ }
+ else
+ {
+ sprintf(fileName,"musicFile.pcm");
+ }
+ if(!_fileReader.SetFileName(fileName))
+ {
+ return false;
+ }
+ if(!_fileReader.EnableAutomaticVAD(true,2))
+ {
+ assert(false);
+ }
+ return _fileReader.FastForwardFile(startPositionInSamples);
+}
+
+bool
+MixerParticipant::InitializeFileWriter(
+ char* outputPath)
+{
+ const WebRtc_Word32 stringsize = 128;
+ char fileName[stringsize] = "";
+ strncpy(fileName,outputPath,stringsize);
+ fileName[stringsize-1] = '\0';
+
+ char tempName[stringsize];
+ tempName[0] = '\0';
+ sprintf(tempName,"outputFile%d.pcm",(int)_id);
+ strncat(fileName,tempName,(stringsize - strlen(fileName)));
+ fileName[stringsize-1] = '\0';
+
+ return _fileWriter.SetFileName(fileName);
+}
+
+StatusReceiver::StatusReceiver(
+ const WebRtc_Word32 id)
+ :
+ _id(id),
+ _mixedParticipants(NULL),
+ _mixedParticipantsAmount(0),
+ _mixedParticipantsSize(0),
+ _vadPositiveParticipants(NULL),
+ _vadPositiveParticipantsAmount(0),
+ _vadPositiveParticipantsSize(0),
+ _mixedAudioLevel(0)
+{
+}
+
+StatusReceiver::~StatusReceiver()
+{
+ delete[] _mixedParticipants;
+ delete[] _vadPositiveParticipants;
+}
+
+void
+StatusReceiver::MixedParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size)
+{
+ if(id != _id)
+ {
+ assert(false);
+ }
+ if(_mixedParticipantsSize < size)
+ {
+ delete[] _mixedParticipants;
+ _mixedParticipantsSize = size;
+ _mixedParticipants = new ParticipantStatistics[size];
+ }
+ _mixedParticipantsAmount = size;
+ memcpy(_mixedParticipants,participantStatistics,sizeof(ParticipantStatistics)*size);
+}
+
+void
+StatusReceiver::VADPositiveParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size)
+{
+ if(id != _id)
+ {
+ assert(false);
+ }
+
+ if(_vadPositiveParticipantsSize < size)
+ {
+ delete[] _vadPositiveParticipants;
+ _vadPositiveParticipantsSize = size;
+ _vadPositiveParticipants = new ParticipantStatistics[size];
+ }
+ _vadPositiveParticipantsAmount = size;
+ memcpy(_vadPositiveParticipants,participantStatistics,sizeof(ParticipantStatistics)*size);
+}
+
+void
+StatusReceiver::MixedAudioLevel(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord32 level)
+{
+ if(id != _id)
+ {
+ assert(false);
+ }
+ _mixedAudioLevel = level;
+}
+
+void
+StatusReceiver::PrintMixedParticipants()
+{
+ std::cout << "Mixed participants" << std::endl;
+ if(_mixedParticipantsAmount == 0)
+ {
+ std::cout << "N/A" << std::endl;
+ }
+ for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++)
+ {
+ std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl;
+ }
+}
+
+void
+StatusReceiver::PrintVadPositiveParticipants()
+{
+ std::cout << "VAD positive participants" << std::endl;
+ if(_mixedParticipantsAmount == 0)
+ {
+ std::cout << "N/A" << std::endl;
+ }
+ for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++)
+ {
+ std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl;
+ }
+}
+
+void
+StatusReceiver::PrintMixedAudioLevel()
+{
+ std::cout << "Mixed audio level = " << _mixedAudioLevel << std::endl;
+}
+
+WebRtc_Word32 MixerWrapper::_mixerWrapperIdCounter = 0;
+
+MixerWrapper::MixerWrapper()
+ :
+ _processThread(NULL),
+ _threadId(0),
+ _firstProcessCall(true),
+ _previousTime(),
+ _periodicityInTicks(TickTime::MillisecondsToTicks(FileReader::kProcessPeriodicityInMs)),
+ _synchronizationEvent(EventWrapper::Create()),
+ _freeItemIds(),
+ _itemIdCounter(0),
+ _mixerParticipants(),
+ _mixerWrappererId(_mixerWrapperIdCounter++),
+ _instanceOutputPath(),
+ _trace(NULL),
+ _statusReceiver(_mixerWrappererId),
+ _generalAudioWriter()
+{
+ sprintf(_instanceOutputPath,"instance%d/",(int)_mixerWrappererId);
+ MKDIR(_instanceOutputPath);
+ _mixer = AudioConferenceMixer::CreateAudioConferenceMixer(
+ _mixerWrappererId);
+ if(_mixer != NULL)
+ {
+ bool success = true;
+
+ success = _mixer->RegisterMixedStreamCallback(*this) == 0;
+ assert(success);
+ success = _mixer->RegisterMixedStreamCallback(*this) == -1;
+ assert(success);
+ success = _mixer->UnRegisterMixedStreamCallback() == 0;
+ assert(success);
+ success = _mixer->UnRegisterMixedStreamCallback() == -1;
+ assert(success);
+ success = _mixer->RegisterMixedStreamCallback(*this) == 0;
+ assert(success);
+
+ success = _mixer->RegisterMixerStatusCallback(_statusReceiver,2) == 0;
+ assert(success);
+ success = _mixer->RegisterMixerStatusCallback(_statusReceiver,1) == -1;
+ assert(success);
+ success = _mixer->UnRegisterMixerStatusCallback() == 0;
+ assert(success);
+ success = _mixer->UnRegisterMixerStatusCallback() == -1;
+ assert(success);
+ success = _mixer->RegisterMixerStatusCallback(_statusReceiver,1) == 0;
+ assert(success);
+ }
+ else
+ {
+ assert(false);
+ std::cout << "Failed to create mixer instance";
+ }
+}
+
+MixerWrapper*
+MixerWrapper::CreateMixerWrapper()
+{
+ MixerWrapper* mixerWrapper = new MixerWrapper();
+ if(!mixerWrapper->InitializeFileWriter())
+ {
+ delete mixerWrapper;
+ return NULL;
+ }
+ return mixerWrapper;
+}
+
+MixerWrapper::~MixerWrapper()
+{
+ StopMixing();
+ ClearAllItemIds();
+ _synchronizationEvent->StopTimer();
+ delete _synchronizationEvent;
+ delete _mixer;
+}
+
+bool
+MixerWrapper::CreateParticipant(
+ MixerParticipant::ParticipantType participantType)
+{
+ WebRtc_Word32 startPosition = 0;
+ GenerateRandomPosition(startPosition);
+ return CreateParticipant(participantType,startPosition);
+}
+
+bool
+MixerWrapper::CreateParticipant(
+ MixerParticipant::ParticipantType participantType,
+ const WebRtc_Word32 startPosition)
+{
+ WebRtc_UWord32 id;
+ if(!GetFreeItemIds(id))
+ {
+ return false;
+ }
+
+ MixerParticipant* participant = MixerParticipant::CreateParticipant(id,participantType,startPosition,_instanceOutputPath);
+ if(!participant)
+ {
+ return false;
+ }
+ if(_mixerParticipants.Insert(id,static_cast<void*>(participant)) != 0)
+ {
+ delete participant;
+ return false;
+ }
+ if(!StartMixingParticipant(id))
+ {
+ DeleteParticipant(id);
+ return false;
+ }
+ return true;
+}
+
+bool
+MixerWrapper::DeleteParticipant(
+ const WebRtc_UWord32 id)
+{
+ bool success = StopMixingParticipant(id);
+ if(!success)
+ {
+ assert(false);
+ return false;
+ }
+ MapItem* item = _mixerParticipants.Find(id);
+ if(item == NULL)
+ {
+ return false;
+ }
+ MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
+ delete participant;
+ _mixerParticipants.Erase(item);
+ AddFreeItemIds(id);
+ return true;
+}
+
+bool
+MixerWrapper::StartMixing(
+ const WebRtc_UWord32 mixedParticipants)
+{
+ if(_processThread)
+ {
+ return false;
+ }
+ if(_mixer->SetAmountOfMixedParticipants(mixedParticipants) != 0)
+ {
+ assert(false);
+ }
+ WebRtc_UWord32 mixedParticipantsTest = 0;
+ _mixer->AmountOfMixedParticipants(mixedParticipantsTest);
+ assert(mixedParticipantsTest == mixedParticipants);
+
+ if(!_synchronizationEvent->StartTimer(true,10))
+ {
+ assert(false);
+ return false;
+ }
+ _processThread = ThreadWrapper::CreateThread(Process, this, kLowPriority);
+ if(!_processThread->Start(_threadId))
+ {
+ delete _processThread;
+ _processThread = NULL;
+ assert(false);
+ return false;
+ }
+
+ return true;
+}
+
+bool
+MixerWrapper::StopMixing()
+{
+ while(_processThread &&
+ !_processThread->Stop())
+ {}
+ _synchronizationEvent->StopTimer();
+
+ delete _processThread;
+ _processThread = NULL;
+ return true;
+}
+
+void
+MixerWrapper::NewMixedAudio(
+ const WebRtc_Word32 id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const WebRtc_UWord32 size)
+{
+ if(id < 0)
+ {
+ assert(false);
+ }
+ // Store the general audio
+ _generalAudioWriter.WriteToFile(generalAudioFrame);
+
+ // Send the unique audio frames to its corresponding participants
+ ListWrapper uniqueAudioFrameList;
+ for(WebRtc_UWord32 i = 0; i < size; i++)
+ {
+ WebRtc_UWord32 id = (uniqueAudioFrames[i])->_id;
+ MapItem* resultItem = _mixerParticipants.Find(id);
+ if(resultItem == NULL)
+ {
+ assert(false);
+ continue;
+ }
+ MixerParticipant* participant = static_cast<MixerParticipant*>(resultItem->GetItem());
+ participant->MixedAudioFrame(*(uniqueAudioFrames[i]));
+ uniqueAudioFrameList.PushBack(resultItem->GetItem());
+ }
+
+ // Send the general audio frames to the remaining participants
+ MapItem* item = _mixerParticipants.First();
+ while(item)
+ {
+ bool isUnique = false;
+ ListItem* compareItem = uniqueAudioFrameList.First();
+ while(compareItem)
+ {
+ if(compareItem->GetItem() == item->GetItem())
+ {
+ isUnique = true;
+ break;
+ }
+ compareItem = uniqueAudioFrameList.Next(compareItem);
+ }
+ if(!isUnique)
+ {
+ MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
+ participant->MixedAudioFrame(generalAudioFrame);
+ }
+ item = _mixerParticipants.Next(item);
+ }
+}
+
+bool
+MixerWrapper::GetParticipantList(
+ ListWrapper& participants)
+{
+ MapItem* item = _mixerParticipants.First();
+ while(item)
+ {
+ participants.PushBack(item->GetId());
+ item = _mixerParticipants.Next(item);
+ }
+ return true;
+}
+
+void
+MixerWrapper::PrintStatus()
+{
+ std::cout << "instance " << _mixerWrappererId << std::endl;
+ std::cout << std::endl;
+ _statusReceiver.PrintMixedParticipants();
+ std::cout << std::endl;
+ _statusReceiver.PrintVadPositiveParticipants();
+ std::cout << std::endl;
+ _statusReceiver.PrintMixedAudioLevel();
+ std::cout << "---------------------------------------" << std::endl;
+}
+
+bool
+MixerWrapper::InitializeFileWriter()
+{
+ const WebRtc_Word32 stringsize = 128;
+ char fileName[stringsize] = "";
+ strncpy(fileName,_instanceOutputPath,stringsize);
+ fileName[stringsize-1] = '\0';
+
+ strncat(fileName,"generalOutputFile.pcm",(stringsize - strlen(fileName)));
+ fileName[stringsize-1] = '\0';
+ return _generalAudioWriter.SetFileName(fileName);
+}
+
+bool
+MixerWrapper::Process(
+ void* instance)
+{
+ MixerWrapper* mixerWrapper = static_cast<MixerWrapper*>(instance);
+ return mixerWrapper->Process();
+}
+
+bool
+MixerWrapper::Process()
+{
+ switch(_synchronizationEvent->Wait(1000))
+ {
+ case kEventSignaled:
+ // Normal operation, ~10 ms has passed
+ break;
+ case kEventError:
+ // Error occured end the thread and throw an assertion
+ assert(false);
+ return false;
+ case kEventTimeout:
+ // One second has passed without a timeout something is wrong
+ // end the thread and throw an assertion
+ assert(false);
+ return false;
+ }
+ WebRtc_Word32 processOfset = 0;
+ const TickTime currentTime = TickTime::Now();
+ if(_firstProcessCall)
+ {
+ _previousTime = TickTime::Now();
+ _firstProcessCall = false;
+ }
+ else
+ {
+ TickInterval deltaTime = (currentTime - _previousTime);
+ _previousTime += _periodicityInTicks;
+ processOfset = (WebRtc_Word32) deltaTime.Milliseconds();
+ processOfset -= FileReader::kProcessPeriodicityInMs;
+ }
+
+ _mixer->Process();
+ WebRtc_Word32 timeUntilNextProcess = _mixer->TimeUntilNextProcess();
+ if(processOfset > FileReader::kProcessPeriodicityInMs)
+ {
+ std::cout << "Performance Warning: Process running " << processOfset << " too slow" << std::endl;
+ _previousTime = currentTime;
+ if(timeUntilNextProcess > 0)
+ {
+ std::cout << "Performance Warning: test performance and module performance missmatch" << std::endl;
+ }
+ }
+ else if(processOfset < -FileReader::kProcessPeriodicityInMs)
+ {
+ std::cout << "Performance Warning: Process running " << -processOfset << " too fast" << std::endl;
+ _previousTime = currentTime;
+ if(timeUntilNextProcess < FileReader::kProcessPeriodicityInMs)
+ {
+ std::cout << "Performance Warning: test performance and module performance missmatch" << std::endl;
+ }
+ }
+ return true;
+}
+
+
+bool
+MixerWrapper::StartMixingParticipant(
+ const WebRtc_UWord32 id)
+{
+ MapItem* item = _mixerParticipants.Find(id);
+ if(item == NULL)
+ {
+ return false;
+ }
+ MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
+ MixerParticipant::ParticipantType participantType = MixerParticipant::REGULAR;
+ participant->GetParticipantType(participantType);
+ if(participantType == MixerParticipant::MIXED_ANONYMOUS)
+ {
+ bool anonymouslyMixed = false;
+ bool success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == 0;
+ assert(success);
+ success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
+ assert(success);
+ assert(anonymouslyMixed);
+ success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == -1;
+ assert(success);
+ success = _mixer->SetAnonymousMixabilityStatus(*participant,false) == 0;
+ assert(success);
+ success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
+ assert(success);
+ assert(!anonymouslyMixed);
+ success = _mixer->SetAnonymousMixabilityStatus(*participant,false) == -1;
+ assert(success);
+ success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == 0;
+ assert(success);
+ success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
+ assert(success);
+ assert(anonymouslyMixed);
+ return success;
+ }
+ WebRtc_UWord32 previousAmountOfMixableParticipants = 0;
+ bool success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0;
+ assert(success);
+
+ success = _mixer->SetMixabilityStatus(*participant,true) == 0;
+ assert(success);
+ success = _mixer->SetMixabilityStatus(*participant,true) == -1;
+ assert(success);
+ success = _mixer->SetMixabilityStatus(*participant,false) == 0;
+ assert(success);
+ success = _mixer->SetMixabilityStatus(*participant,false) == -1;
+ assert(success);
+ success = _mixer->SetMixabilityStatus(*participant,true) == 0;
+ assert(success);
+ if(!success)
+ {
+ return false;
+ }
+
+ WebRtc_UWord32 currentAmountOfMixableParticipants = 0;
+ success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0;
+ assert(currentAmountOfMixableParticipants == previousAmountOfMixableParticipants + 1);
+
+ bool mixable = true;
+ success = _mixer->MixabilityStatus(*participant,mixable) == 0;
+ assert(success);
+ assert(mixable);
+ if(participantType == MixerParticipant::REGULAR)
+ {
+ return true;
+ }
+ bool IsVIP = false;
+ success = _mixer->SetVIPStatus(*participant,true) == 0;
+ assert(success);
+ success = _mixer->VIPStatus(*participant,IsVIP) == 0;
+ assert(success);
+ assert(IsVIP);
+ success = _mixer->SetVIPStatus(*participant,true) == -1;
+ assert(success);
+ success = _mixer->SetVIPStatus(*participant,false) == 0;
+ assert(success);
+ success = _mixer->VIPStatus(*participant,IsVIP) == 0;
+ assert(success);
+ assert(!IsVIP);
+ success = _mixer->SetVIPStatus(*participant,false) == -1;
+ assert(success);
+ success = _mixer->SetVIPStatus(*participant,true) == 0;
+ assert(success);
+ success = _mixer->VIPStatus(*participant,IsVIP) == 0;
+ assert(success);
+ assert(IsVIP);
+ assert(success);
+ return success;
+}
+
+bool
+MixerWrapper::StopMixingParticipant(
+ const WebRtc_UWord32 id)
+{
+ MapItem* item = _mixerParticipants.Find(id);
+ if(item == NULL)
+ {
+ return false;
+ }
+ MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
+ bool success = false;
+ WebRtc_UWord32 previousAmountOfMixableParticipants = 0;
+ success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0;
+ assert(success);
+ success = _mixer->SetMixabilityStatus(*participant,false) == 0;
+ assert(success);
+ WebRtc_UWord32 currentAmountOfMixableParticipants = 0;
+ success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0;
+ assert(success);
+ assert(success ? currentAmountOfMixableParticipants == previousAmountOfMixableParticipants -1 :
+ currentAmountOfMixableParticipants == previousAmountOfMixableParticipants);
+ return success;
+}
+
+bool
+MixerWrapper::GetFreeItemIds(
+ WebRtc_UWord32& itemId)
+{
+ if(!_freeItemIds.Empty())
+ {
+ ListItem* item = _freeItemIds.First();
+ WebRtc_UWord32* id = static_cast<WebRtc_UWord32*>(item->GetItem());
+ itemId = *id;
+ delete id;
+ return true;
+ }
+ if(_itemIdCounter == (WebRtc_UWord32) -1)
+ {
+ return false;
+ }
+ itemId = _itemIdCounter++;
+ return true;
+}
+
+void
+MixerWrapper::AddFreeItemIds(
+ const WebRtc_UWord32 itemId)
+{
+ WebRtc_UWord32* id = new WebRtc_UWord32;
+ *id = itemId;
+ _freeItemIds.PushBack(static_cast<void*>(id));
+}
+
+void
+MixerWrapper::ClearAllItemIds()
+{
+ ListItem* item = _freeItemIds.First();
+ while(item != NULL)
+ {
+ WebRtc_UWord32* id = static_cast<WebRtc_UWord32*>(item->GetItem());
+ delete id;
+ _freeItemIds.Erase(item);
+ item = _freeItemIds.First();
+ }
+}
+
+bool
+LoopedFileRead(
+ WebRtc_Word16* buffer,
+ WebRtc_UWord32 bufferSizeInSamples,
+ WebRtc_UWord32 samplesToRead,
+ FILE* file)
+{
+ if(bufferSizeInSamples < samplesToRead)
+ {
+ return false;
+ }
+ WebRtc_UWord32 gottenSamples = (WebRtc_UWord32)fread(buffer,sizeof(WebRtc_Word16),samplesToRead,file);
+ if(gottenSamples != samplesToRead)
+ {
+ WebRtc_UWord32 missingSamples = samplesToRead - gottenSamples;
+ fseek(file,0,0);
+ gottenSamples += (WebRtc_UWord32)fread(&buffer[gottenSamples],sizeof(WebRtc_Word16),missingSamples,file);
+ }
+ if(gottenSamples != samplesToRead)
+ {
+ return false;
+ }
+ return true;
+}
+
+void
+GenerateRandomPosition(
+ WebRtc_Word32& startPosition)
+{
+ startPosition = (rand() % (60*16000/160)) * 160;
+}
diff --git a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h
new file mode 100644
index 0000000..f25e5f1
--- /dev/null
+++ b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h
@@ -0,0 +1,276 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
+
+#include "module_common_types.h"
+#include "level_indicator.h"
+#include "list_wrapper.h"
+#include "map_wrapper.h"
+#include "audio_conference_mixer.h"
+#include "audio_conference_mixer_defines.h"
+#include "tick_util.h"
+
+namespace webrtc {
+class EventWrapper;
+class ThreadWrapper;
+class Trace;
+}
+struct WebRtcVadInst;
+
+class FileWriter
+{
+public:
+ FileWriter();
+ ~FileWriter();
+
+ bool SetFileName(
+ const char* fileName);
+
+ bool WriteToFile(
+ const AudioFrame& audioFrame);
+private:
+ FILE* _file;
+};
+
+class FileReader
+{
+public:
+ enum {kProcessPeriodicityInMs = 10};
+ enum Frequency
+ {
+ kNbInHz = 8000,
+ kWbInHz = 16000,
+ kDefaultFrequency = kWbInHz
+ };
+
+ FileReader();
+ ~FileReader();
+
+ bool SetFileName(
+ const char* fileName);
+
+ bool ReadFromFile(
+ AudioFrame& audioFrame);
+
+ bool FastForwardFile(
+ const WebRtc_Word32 samples);
+
+ bool EnableAutomaticVAD(
+ bool enable,
+ int mode);
+
+ bool SetVAD(
+ bool vad);
+private:
+ bool GetVAD(
+ WebRtc_Word16* buffer,
+ WebRtc_UWord8 bufferLengthInSamples,
+ bool& vad);
+
+ Frequency _frequency;
+ WebRtc_UWord8 _sampleSize;
+
+ WebRtc_UWord32 _timeStamp;
+
+ FILE* _file;
+
+ WebRtcVadInst* _vadInstr;
+ bool _automaticVad;
+ bool _vad;
+
+ LevelIndicator _volumeCalculator;
+};
+
+class MixerParticipant : public MixerParticipant
+{
+public:
+ enum ParticipantType
+ {
+ VIP = 0,
+ REGULAR = 1,
+ MIXED_ANONYMOUS = 2,
+ RANDOM = 3
+ };
+
+ static MixerParticipant* CreateParticipant(
+ const WebRtc_UWord32 id,
+ ParticipantType participantType,
+ const WebRtc_Word32 startPosition,
+ char* outputPath);
+ ~MixerParticipant();
+
+ WebRtc_Word32 GetAudioFrame(
+ const WebRtc_Word32 id,
+ AudioFrame& audioFrame);
+
+ WebRtc_Word32 MixedAudioFrame(
+ const AudioFrame& audioFrame);
+
+ WebRtc_Word32 GetParticipantType(
+ ParticipantType& participantType);
+private:
+ MixerParticipant(
+ const WebRtc_UWord32 id,
+ ParticipantType participantType);
+
+ bool InitializeFileReader(
+ const WebRtc_Word32 startPositionInSamples);
+
+ bool InitializeFileWriter(
+ char* outputPath);
+
+ WebRtc_UWord32 _id;
+ ParticipantType _participantType;
+
+ FileReader _fileReader;
+ FileWriter _fileWriter;
+};
+
+class StatusReceiver : public AudioMixerStatusReceiver
+{
+public:
+ StatusReceiver(
+ const WebRtc_Word32 id);
+ ~StatusReceiver();
+
+ void MixedParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size);
+
+ void VADPositiveParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size);
+
+ void MixedAudioLevel(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord32 level);
+
+ void PrintMixedParticipants();
+
+ void PrintVadPositiveParticipants();
+
+ void PrintMixedAudioLevel();
+private:
+ WebRtc_Word32 _id;
+
+ ParticipantStatistics* _mixedParticipants;
+ WebRtc_UWord32 _mixedParticipantsAmount;
+ WebRtc_UWord32 _mixedParticipantsSize;
+
+ ParticipantStatistics* _vadPositiveParticipants;
+ WebRtc_UWord32 _vadPositiveParticipantsAmount;
+ WebRtc_UWord32 _vadPositiveParticipantsSize;
+
+ WebRtc_UWord32 _mixedAudioLevel;
+};
+
+class MixerWrapper : public AudioMixerOutputReceiver
+{
+public:
+ static MixerWrapper* CreateMixerWrapper();
+ ~MixerWrapper();
+
+ bool SetMixFrequency(
+ const AudioConferenceMixer::Frequency frequency);
+
+ bool CreateParticipant(
+ MixerParticipant::ParticipantType participantType);
+
+ bool CreateParticipant(
+ MixerParticipant::ParticipantType participantType,
+ const WebRtc_Word32 startPosition);
+
+ bool DeleteParticipant(
+ const WebRtc_UWord32 id);
+
+ bool StartMixing(
+ const WebRtc_UWord32 mixedParticipants = AudioConferenceMixer::kDefaultAmountOfMixedParticipants);
+
+ bool StopMixing();
+
+ void NewMixedAudio(
+ const WebRtc_Word32 id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const WebRtc_UWord32 size);
+
+ bool GetParticipantList(
+ ListWrapper& participants);
+
+ void PrintStatus();
+private:
+ MixerWrapper();
+
+ bool InitializeFileWriter();
+
+ static bool Process(
+ void* instance);
+
+ bool Process();
+
+ bool StartMixingParticipant(
+ const WebRtc_UWord32 id);
+
+ bool StopMixingParticipant(
+ const WebRtc_UWord32 id);
+
+ bool GetFreeItemIds(
+ WebRtc_UWord32& itemId);
+
+ void AddFreeItemIds(
+ const WebRtc_UWord32 itemId);
+
+ void ClearAllItemIds();
+
+ webrtc::ThreadWrapper* _processThread;
+ unsigned int _threadId;
+
+ // Performance hooks
+ enum{WARNING_COUNTER = 100};
+
+ bool _firstProcessCall;
+ TickTime _previousTime; // Tick time of previous process
+ const WebRtc_Word64 _periodicityInTicks; // Periodicity
+
+ webrtc::EventWrapper* _synchronizationEvent;
+
+ ListWrapper _freeItemIds;
+ WebRtc_UWord32 _itemIdCounter;
+
+ MapWrapper _mixerParticipants;
+
+ static WebRtc_Word32 _mixerWrapperIdCounter;
+ WebRtc_Word32 _mixerWrappererId;
+ char _instanceOutputPath[128];
+
+ webrtc::Trace* _trace;
+ AudioConferenceMixer* _mixer;
+
+ StatusReceiver _statusReceiver;
+
+ FileWriter _generalAudioWriter;
+};
+
+bool
+LoopedFileRead(
+ WebRtc_Word16* buffer,
+ WebRtc_UWord32 bufferSizeInSamples,
+ WebRtc_UWord32 samplesToRead,
+ FILE* file);
+
+void
+GenerateRandomPosition(
+ WebRtc_Word32& startPosition);
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
diff --git a/trunk/src/modules/audio_device/OWNERS b/trunk/src/modules/audio_device/OWNERS
new file mode 100644
index 0000000..a07ced3
--- /dev/null
+++ b/trunk/src/modules/audio_device/OWNERS
@@ -0,0 +1,4 @@
+henrikg@webrtc.org
+henrika@webrtc.org
+niklas.enbom@webrtc.org
+xians@webrtc.org
diff --git a/trunk/src/modules/audio_device/main/interface/audio_device.h b/trunk/src/modules/audio_device/main/interface/audio_device.h
new file mode 100644
index 0000000..1149055
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/interface/audio_device.h
@@ -0,0 +1,207 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
+
+#include "module.h"
+#include "audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceModule : public RefCountedModule {
+ public:
+ enum ErrorCode {
+ kAdmErrNone = 0,
+ kAdmErrArgument = 1
+ };
+
+ enum AudioLayer {
+ kPlatformDefaultAudio = 0,
+ kWindowsWaveAudio = 1,
+ kWindowsCoreAudio = 2,
+ kLinuxAlsaAudio = 3,
+ kLinuxPulseAudio = 4,
+ kDummyAudio = 5
+ };
+
+ enum WindowsDeviceType {
+ kDefaultCommunicationDevice = -1,
+ kDefaultDevice = -2
+ };
+
+ enum BufferType {
+ kFixedBufferSize = 0,
+ kAdaptiveBufferSize = 1
+ };
+
+ enum ChannelType {
+ kChannelLeft = 0,
+ kChannelRight = 1,
+ kChannelBoth = 2
+ };
+
+ public:
+ // Retrieve the currently utilized audio layer
+ virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
+
+ // Error handling
+ virtual ErrorCode LastError() const = 0;
+ virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) = 0;
+
+ // Full-duplex transportation of PCM audio
+ virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
+
+ // Main initialization and termination
+ virtual int32_t Init() = 0;
+ virtual int32_t Terminate() = 0;
+ virtual bool Initialized() const = 0;
+
+ // Device enumeration
+ virtual int16_t PlayoutDevices() = 0;
+ virtual int16_t RecordingDevices() = 0;
+ virtual int32_t PlayoutDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) = 0;
+ virtual int32_t RecordingDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) = 0;
+
+ // Device selection
+ virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+ virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
+ virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+ virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
+
+ // Audio transport initialization
+ virtual int32_t PlayoutIsAvailable(bool* available) = 0;
+ virtual int32_t InitPlayout() = 0;
+ virtual bool PlayoutIsInitialized() const = 0;
+ virtual int32_t RecordingIsAvailable(bool* available) = 0;
+ virtual int32_t InitRecording() = 0;
+ virtual bool RecordingIsInitialized() const = 0;
+
+ // Audio transport control
+ virtual int32_t StartPlayout() = 0;
+ virtual int32_t StopPlayout() = 0;
+ virtual bool Playing() const = 0;
+ virtual int32_t StartRecording() = 0;
+ virtual int32_t StopRecording() = 0;
+ virtual bool Recording() const = 0;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual int32_t SetAGC(bool enable) = 0;
+ virtual bool AGC() const = 0;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
+ uint16_t volumeRight) = 0;
+ virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
+ uint16_t* volumeRight) const = 0;
+
+ // Audio mixer initialization
+ virtual int32_t SpeakerIsAvailable(bool* available) = 0;
+ virtual int32_t InitSpeaker() = 0;
+ virtual bool SpeakerIsInitialized() const = 0;
+ virtual int32_t MicrophoneIsAvailable(bool* available) = 0;
+ virtual int32_t InitMicrophone() = 0;
+ virtual bool MicrophoneIsInitialized() const = 0;
+
+ // Speaker volume controls
+ virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
+ virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+ virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
+ virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
+ virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
+ virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const = 0;
+
+ // Microphone volume controls
+ virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
+ virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+ virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
+ virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
+ virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
+ virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const = 0;
+
+ // Speaker mute control
+ virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
+ virtual int32_t SetSpeakerMute(bool enable) = 0;
+ virtual int32_t SpeakerMute(bool* enabled) const = 0;
+
+ // Microphone mute control
+ virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
+ virtual int32_t SetMicrophoneMute(bool enable) = 0;
+ virtual int32_t MicrophoneMute(bool* enabled) const = 0;
+
+ // Microphone boost control
+ virtual int32_t MicrophoneBoostIsAvailable(bool* available) = 0;
+ virtual int32_t SetMicrophoneBoost(bool enable) = 0;
+ virtual int32_t MicrophoneBoost(bool* enabled) const = 0;
+
+ // Stereo support
+ virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
+ virtual int32_t SetStereoPlayout(bool enable) = 0;
+ virtual int32_t StereoPlayout(bool* enabled) const = 0;
+ virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
+ virtual int32_t SetStereoRecording(bool enable) = 0;
+ virtual int32_t StereoRecording(bool* enabled) const = 0;
+ virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
+ virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
+
+ // Delay information and control
+ virtual int32_t SetPlayoutBuffer(const BufferType type,
+ uint16_t sizeMS = 0) = 0;
+ virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const = 0;
+ virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
+ virtual int32_t RecordingDelay(uint16_t* delayMS) const = 0;
+
+ // CPU load
+ virtual int32_t CPULoad(uint16_t* load) const = 0;
+
+ // Recording of raw PCM data
+ virtual int32_t StartRawOutputFileRecording(
+ const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
+ virtual int32_t StopRawOutputFileRecording() = 0;
+ virtual int32_t StartRawInputFileRecording(
+ const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
+ virtual int32_t StopRawInputFileRecording() = 0;
+
+ // Native sample rate controls (samples/sec)
+ virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;
+ virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0;
+ virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0;
+ virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
+
+ // Mobile device specific functions
+ virtual int32_t ResetAudioDevice() = 0;
+ virtual int32_t SetLoudspeakerStatus(bool enable) = 0;
+ virtual int32_t GetLoudspeakerStatus(bool* enabled) const = 0;
+
+ // *Experimental - not recommended for use.*
+ // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
+ //
+ // Must be called before InitRecording(). When enabled:
+ // 1. StartPlayout() must be called before StartRecording().
+ // 2. StopRecording() should be called before StopPlayout().
+ // The reverse order may cause garbage audio to be rendered or the
+ // capture side to halt until StopRecording() is called.
+ virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
+ virtual bool BuiltInAECIsEnabled() const { return false; }
+
+ protected:
+ virtual ~AudioDeviceModule() {};
+};
+
+AudioDeviceModule* CreateAudioDeviceModule(
+ WebRtc_Word32 id, AudioDeviceModule::AudioLayer audioLayer);
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
diff --git a/trunk/src/modules/audio_device/main/interface/audio_device_defines.h b/trunk/src/modules/audio_device/main/interface/audio_device_defines.h
new file mode 100644
index 0000000..67db65d
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/interface/audio_device_defines.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+// AudioDeviceObserver
+// ----------------------------------------------------------------------------
+
+class AudioDeviceObserver
+{
+public:
+ enum ErrorCode
+ {
+ kRecordingError = 0,
+ kPlayoutError = 1
+ };
+ enum WarningCode
+ {
+ kRecordingWarning = 0,
+ kPlayoutWarning = 1
+ };
+
+ virtual void OnErrorIsReported(const ErrorCode error) = 0;
+ virtual void OnWarningIsReported(const WarningCode warning) = 0;
+
+protected:
+ virtual ~AudioDeviceObserver() {}
+};
+
+// ----------------------------------------------------------------------------
+// AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport
+{
+public:
+ virtual int32_t RecordedDataIsAvailable(const char* audioSamples,
+ const uint32_t nSamples,
+ const uint8_t nBytesPerSample,
+ const uint8_t nChannels,
+ const uint32_t samplesPerSec,
+ const uint32_t totalDelayMS,
+ const int32_t clockDrift,
+ const uint32_t currentMicLevel,
+ uint32_t& newMicLevel) = 0;
+
+ virtual int32_t NeedMorePlayData(const uint32_t nSamples,
+ const uint8_t nBytesPerSample,
+ const uint8_t nChannels,
+ const uint32_t samplesPerSec,
+ char* audioSamples,
+ uint32_t& nSamplesOut) = 0;
+
+protected:
+ virtual ~AudioTransport() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
diff --git a/trunk/src/modules/audio_device/main/source/Android.mk b/trunk/src/modules/audio_device/main/source/Android.mk
new file mode 100644
index 0000000..affa5e1
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/Android.mk
@@ -0,0 +1,57 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_audio_device
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ audio_device_buffer.cc \
+ audio_device_generic.cc \
+ audio_device_utility.cc \
+ audio_device_impl.cc \
+ android/audio_device_android_opensles.cc \
+ android/audio_device_utility_android.cc \
+ dummy/audio_device_utility_dummy.cc \
+ dummy/audio_device_dummy.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_ANDROID_OPENSLES'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/android \
+ $(LOCAL_PATH)/dummy \
+ $(LOCAL_PATH)/linux \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../../common_audio/resampler/include \
+ $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface \
+ system/media/wilhelm/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport \
+ libOpenSLES
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc
new file mode 100644
index 0000000..c275171
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc
@@ -0,0 +1,2920 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Android audio device implementation (JNI/AudioTrack/AudioRecord usage)
+ */
+
+// TODO(xians): Break out attach and detach current thread to JVM to
+// separate functions.
+
+#include <stdlib.h>
+#include "audio_device_utility.h"
+#include "audio_device_android_jni.h"
+#include "audio_device_config.h"
+
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+
+// Android logging, uncomment to print trace to logcat instead of
+// trace file/callback
+//#include <android/log.h>
+//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, \
+// "WebRTC AD jni", __VA_ARGS__)
+
+namespace webrtc
+{
+
+JavaVM* globalJvm = NULL;
+JNIEnv* globalJNIEnv = NULL;
+jobject globalSndContext = NULL;
+jclass globalScClass = NULL;
+
+// ----------------------------------------------------------------------------
+// SetAndroidAudioDeviceObjects
+//
+// Global function for setting Java pointers and creating Java
+// objects that are global to all instances of VoiceEngine used
+// by the same Java application.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 SetAndroidAudioDeviceObjects(void* javaVM, void* env,
+ void* context)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1, "%s",
+ __FUNCTION__);
+
+ globalJvm = (JavaVM*) javaVM;
+ globalSndContext = (jobject) context;
+
+ if (env)
+ {
+ globalJNIEnv = (JNIEnv *) env;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1,
+ "%s: will find class", __FUNCTION__);
+
+ // get java class type (note path to class packet)
+ jclass
+ javaScClassLocal =
+ globalJNIEnv->FindClass(
+ "org/webrtc/voiceengine/AudioDeviceAndroid");
+ if (!javaScClassLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "%s: could not find java class", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1,
+ "%s: will create global reference", __FUNCTION__);
+
+ // create a global reference to the class (to tell JNI that we are
+ // referencing it after this function has returned)
+ globalScClass
+ = reinterpret_cast<jclass> (globalJNIEnv->NewGlobalRef(
+ javaScClassLocal));
+ if (!globalScClass)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "%s: could not create reference", __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local class ref, we only use the global ref
+ globalJNIEnv->DeleteLocalRef(javaScClassLocal);
+ }
+ else // User is resetting the env variable
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
+ "%s: env is NULL, assuming deinit", __FUNCTION__);
+
+ if (!globalJNIEnv)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
+ "%s: saved env already NULL", __FUNCTION__);
+ return 0;
+ }
+
+ globalJNIEnv->DeleteGlobalRef(globalScClass);
+ globalJNIEnv = (JNIEnv *) NULL;
+ }
+
+ return 0;
+}
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceAndroidJni - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceAndroidJni::AudioDeviceAndroidJni(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _timeEventRec(*EventWrapper::Create()),
+ _timeEventPlay(*EventWrapper::Create()),
+ _recStartStopEvent(*EventWrapper::Create()),
+ _playStartStopEvent(*EventWrapper::Create()),
+ _ptrThreadPlay(NULL),
+ _ptrThreadRec(NULL),
+ _recThreadID(0),
+ _playThreadID(0),
+ _playThreadIsInitialized(false),
+ _recThreadIsInitialized(false),
+ _shutdownPlayThread(false),
+ _shutdownRecThread(false),
+ // _recBuffer[2*REC_BUF_SIZE_IN_SAMPLES]
+ _recordingDeviceIsSpecified(false),
+ _playoutDeviceIsSpecified(false), _initialized(false),
+ _recording(false), _playing(false), _recIsInitialized(false),
+ _playIsInitialized(false), _micIsInitialized(false),
+ _speakerIsInitialized(false), _startRec(false), _stopRec(false),
+ _startPlay(false), _stopPlay(false), _playWarning(0),
+ _playError(0), _recWarning(0), _recError(0), _delayPlayout(0),
+ _delayRecording(0),
+ _AGC(false),
+ _samplingFreqIn(0),
+ _samplingFreqOut(0),
+ _maxSpeakerVolume(0),
+ _loudSpeakerOn(false),
+ _recAudioSource(1), // 1 is AudioSource.MIC which is our default
+ _javaVM(NULL), _javaContext(NULL), _jniEnvPlay(NULL),
+ _jniEnvRec(NULL), _javaScClass(0), _javaScObj(0),
+ _javaPlayBuffer(0), _javaRecBuffer(0), _javaDirectPlayBuffer(NULL),
+ _javaDirectRecBuffer(NULL), _javaMidPlayAudio(0),
+ _javaMidRecAudio(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+
+ memset(_recBuffer, 0, sizeof(_recBuffer));
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceAndroidJni - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceAndroidJni::~AudioDeviceAndroidJni()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ delete &_recStartStopEvent;
+ delete &_playStartStopEvent;
+ delete &_timeEventRec;
+ delete &_timeEventPlay;
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AttachAudioBuffer
+// ----------------------------------------------------------------------------
+
+void AudioDeviceAndroidJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // inform the AudioBuffer about default settings for this implementation
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+ _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const
+{
+
+ audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Init
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ // Init Java member variables
+ // and set up JNI interface to
+ // AudioDeviceAndroid java class
+ if (InitJavaResources() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Failed to init Java resources", __FUNCTION__);
+ return -1;
+ }
+
+ // Check the sample rate to be used for playback and recording
+ // and the max playout volume
+ if (InitSampleRate() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Failed to init samplerate", __FUNCTION__);
+ return -1;
+ }
+
+ // RECORDING
+ const char* threadName = "webrtc_jni_audio_capture_thread";
+ _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
+ kRealtimePriority, threadName);
+ if (_ptrThreadRec == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the rec audio thread");
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThreadRec->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the rec audio thread");
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ return -1;
+ }
+ _recThreadID = threadID;
+
+ // PLAYOUT
+ threadName = "webrtc_jni_audio_render_thread";
+ _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
+ kRealtimePriority, threadName);
+ if (_ptrThreadPlay == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the play audio thread");
+ return -1;
+ }
+
+ threadID = 0;
+ if (!_ptrThreadPlay->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the play audio thread");
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ return -1;
+ }
+ _playThreadID = threadID;
+
+ _initialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Terminate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::Terminate()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ // RECORDING
+ StopRecording();
+ _shutdownRecThread = true;
+ _timeEventRec.Set(); // Release rec thread from waiting state
+ if (_ptrThreadRec)
+ {
+ // First, the thread must detach itself from Java VM
+ _critSect.Leave();
+ if (kEventSignaled != _recStartStopEvent.Wait(5000))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceAudioDevice,
+ _id,
+ "%s: Recording thread shutdown timed out, cannot "
+ "terminate thread",
+ __FUNCTION__);
+ // If we close thread anyway, the app will crash
+ return -1;
+ }
+ _recStartStopEvent.Reset();
+ _critSect.Enter();
+
+ // Close down rec thread
+ ThreadWrapper* tmpThread = _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ _critSect.Leave();
+ tmpThread->SetNotAlive();
+ // Release again, we might have returned to waiting state
+ _timeEventRec.Set();
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ _jniEnvRec = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the rec audio thread");
+ }
+ _critSect.Enter();
+
+ _recThreadIsInitialized = false;
+ }
+ _micIsInitialized = false;
+ _recordingDeviceIsSpecified = false;
+
+ // PLAYOUT
+ StopPlayout();
+ _shutdownPlayThread = true;
+ _timeEventPlay.Set(); // Release rec thread from waiting state
+ if (_ptrThreadPlay)
+ {
+ // First, the thread must detach itself from Java VM
+ _critSect.Leave();
+ if (kEventSignaled != _playStartStopEvent.Wait(5000))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceAudioDevice,
+ _id,
+ "%s: Playout thread shutdown timed out, cannot "
+ "terminate thread",
+ __FUNCTION__);
+ // If we close thread anyway, the app will crash
+ return -1;
+ }
+ _playStartStopEvent.Reset();
+ _critSect.Enter();
+
+ // Close down play thread
+ ThreadWrapper* tmpThread = _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ _critSect.Leave();
+ tmpThread->SetNotAlive();
+ _timeEventPlay.Set();
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ _jniEnvPlay = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the play audio thread");
+ }
+ _critSect.Enter();
+
+ _playThreadIsInitialized = false;
+ }
+ _speakerIsInitialized = false;
+ _playoutDeviceIsSpecified = false;
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // Make method IDs and buffer pointers unusable
+ _javaMidPlayAudio = 0;
+ _javaMidRecAudio = 0;
+ _javaDirectPlayBuffer = NULL;
+ _javaDirectRecBuffer = NULL;
+
+ // Delete the references to the java buffers, this allows the
+ // garbage collector to delete them
+ env->DeleteGlobalRef(_javaPlayBuffer);
+ _javaPlayBuffer = 0;
+ env->DeleteGlobalRef(_javaRecBuffer);
+ _javaRecBuffer = 0;
+
+ // Delete the references to the java object and class, this allows the
+ // garbage collector to delete them
+ env->DeleteGlobalRef(_javaScObj);
+ _javaScObj = 0;
+ _javaScClass = 0;
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ _initialized = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::Initialized() const
+{
+
+ return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerIsAvailable(bool& available)
+{
+
+ // We always assume it's available
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Playout already started");
+ return -1;
+ }
+
+ if (!_playoutDeviceIsSpecified)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout device is not specified");
+ return -1;
+ }
+
+ // Nothing needs to be done here, we use a flag to have consistent
+ // behavior with other platforms
+ _speakerIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneIsAvailable(bool& available)
+{
+
+ // We always assume it's available
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Recording already started");
+ return -1;
+ }
+
+ if (!_recordingDeviceIsSpecified)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording device is not specified");
+ return -1;
+ }
+
+ // Nothing needs to be done here, we use a flag to have consistent
+ // behavior with other platforms
+ _micIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::SpeakerIsInitialized() const
+{
+
+ return _speakerIsInitialized;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::MicrophoneIsInitialized() const
+{
+
+ return _micIsInitialized;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ available = true; // We assume we are always be able to set/get volume
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+ if (!_javaContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Context is not set");
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID setPlayoutVolumeID = env->GetMethodID(_javaScClass,
+ "SetPlayoutVolume", "(I)I");
+
+ // call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, setPlayoutVolumeID,
+ static_cast<int> (volume));
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "SetPlayoutVolume failed (%d)", res);
+ return -1;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+ if (!_javaContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Context is not set");
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID getPlayoutVolumeID = env->GetMethodID(_javaScClass,
+ "GetPlayoutVolume", "()I");
+
+ // call java sc object method
+ jint level = env->CallIntMethod(_javaScObj, getPlayoutVolumeID);
+ if (level < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetPlayoutVolume failed (%d)", level);
+ return -1;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ volume = static_cast<WebRtc_UWord32> (level);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetWaveOutVolume(
+ WebRtc_UWord16 /*volumeLeft*/,
+ WebRtc_UWord16 /*volumeRight*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::WaveOutVolume(
+ WebRtc_UWord16& /*volumeLeft*/,
+ WebRtc_UWord16& /*volumeRight*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MaxSpeakerVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ maxVolume = _maxSpeakerVolume;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MinSpeakerVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ minVolume = 0;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ stepSize = 1;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerMuteIsAvailable(bool& available)
+{
+
+ available = false; // Speaker mute not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetSpeakerMute(bool /*enable*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SpeakerMute(bool& /*enabled*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ available = false; // Mic mute not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneMute(bool /*enable*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneMute(bool& /*enabled*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ available = false; // Mic boost not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneBoost(bool enable)
+{
+
+ if (!_micIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Microphone not initialized");
+ return -1;
+ }
+
+ if (enable)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneBoost(bool& enabled) const
+{
+
+ if (!_micIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Microphone not initialized");
+ return -1;
+ }
+
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StereoRecordingIsAvailable(bool& available)
+{
+
+ available = false; // Stereo recording not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoRecording
+//
+// Specifies the number of input channels.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetStereoRecording(bool enable)
+{
+
+ if (enable)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StereoRecording(bool& enabled) const
+{
+
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StereoPlayoutIsAvailable(bool& available)
+{
+
+ available = false; // Stereo playout not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetStereoPlayout(bool enable)
+{
+
+ if (enable)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StereoPlayout(bool& enabled) const
+{
+
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetAGC
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetAGC(bool enable)
+{
+
+ _AGC = enable;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// AGC
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::AGC() const
+{
+
+ return _AGC;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolumeIsAvailable(
+ bool& available)
+{
+
+ available = false; // Mic volume not supported on Android
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneVolume(
+ WebRtc_UWord32 /*volume*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolume(
+ WebRtc_UWord32& /*volume*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MaxMicrophoneVolume(
+ WebRtc_UWord32& /*maxVolume*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MinMicrophoneVolume(
+ WebRtc_UWord32& /*minVolume*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& /*stepSize*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceAndroidJni::PlayoutDevices()
+{
+
+ // There is one device only
+ return 1;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout already initialized");
+ return -1;
+ }
+
+ if (0 != index)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Do nothing but set a flag, this is to have consistent behavior
+ // with other platforms
+ _playoutDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize])
+{
+
+ if (0 != index)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Return empty string
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::RecordingDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize])
+{
+
+ if (0 != index)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Return empty string
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceAndroidJni::RecordingDevices()
+{
+
+ // There is one device only
+ return 1;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording already initialized");
+ return -1;
+ }
+
+ // Recording device index will be used for specifying recording
+ // audio source, allow any value
+ _recAudioSource = index;
+ _recordingDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::PlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return res;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::RecordingIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return res;
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Not initialized");
+ return -1;
+ }
+
+ if (_playing)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Playout already started");
+ return -1;
+ }
+
+ if (!_playoutDeviceIsSpecified)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout device is not specified");
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout already initialized");
+ return 0;
+ }
+
+ // Initialize the speaker
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitSpeaker() failed");
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "attaching");
+
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID initPlaybackID = env->GetMethodID(_javaScClass, "InitPlayback",
+ "(I)I");
+
+ int samplingFreq = 44100;
+ if (_samplingFreqOut != 44)
+ {
+ samplingFreq = _samplingFreqOut * 1000;
+ }
+
+ int retVal = -1;
+
+ // Call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, initPlaybackID, samplingFreq);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "InitPlayback failed (%d)", res);
+ }
+ else
+ {
+ // Set the audio device buffer sampling rate
+ _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreqOut * 1000);
+ _playIsInitialized = true;
+ retVal = 0;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "detaching");
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return retVal;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Not initialized");
+ return -1;
+ }
+
+ if (_recording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Recording already started");
+ return -1;
+ }
+
+ if (!_recordingDeviceIsSpecified)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording device is not specified");
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording already initialized");
+ return 0;
+ }
+
+ // Initialize the microphone
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitMicrophone() failed");
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID initRecordingID = env->GetMethodID(_javaScClass, "InitRecording",
+ "(II)I");
+
+ int samplingFreq = 44100;
+ if (_samplingFreqIn != 44)
+ {
+ samplingFreq = _samplingFreqIn * 1000;
+ }
+
+ int retVal = -1;
+
+ // call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, initRecordingID, _recAudioSource,
+ samplingFreq);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "InitRecording failed (%d)", res);
+ }
+ else
+ {
+ // Set the audio device buffer sampling rate
+ _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreqIn * 1000);
+
+ // the init rec function returns a fixed delay
+ _delayRecording = res / _samplingFreqIn;
+
+ _recIsInitialized = true;
+ retVal = 0;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return retVal;
+}
+
+// ----------------------------------------------------------------------------
+// StartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StartRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording not initialized");
+ return -1;
+ }
+
+ if (_recording)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording already started");
+ return 0;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID startRecordingID = env->GetMethodID(_javaScClass,
+ "StartRecording", "()I");
+
+ // Call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, startRecordingID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "StartRecording failed (%d)", res);
+ return -1;
+ }
+
+ _recWarning = 0;
+ _recError = 0;
+
+ // Signal to recording thread that we want to start
+ _startRec = true;
+ _timeEventRec.Set(); // Release thread from waiting state
+ _critSect.Leave();
+ // Wait for thread to init
+ if (kEventSignaled != _recStartStopEvent.Wait(5000))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Timeout or error starting");
+ }
+ _recStartStopEvent.Reset();
+ _critSect.Enter();
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StopRecording()
+
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording is not initialized");
+ return 0;
+ }
+
+ // make sure we don't start recording (it's asynchronous),
+ // assuming that we are under lock
+ _startRec = false;
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID stopRecordingID = env->GetMethodID(_javaScClass, "StopRecording",
+ "()I");
+
+ // Call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, stopRecordingID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "StopRecording failed (%d)", res);
+ }
+
+ _recIsInitialized = false;
+ _recording = false;
+ _recWarning = 0;
+ _recError = 0;
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::RecordingIsInitialized() const
+{
+
+ return _recIsInitialized;
+}
+
+// ----------------------------------------------------------------------------
+// Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::Recording() const
+{
+
+ return _recording;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::PlayoutIsInitialized() const
+{
+
+ return _playIsInitialized;
+}
+
+// ----------------------------------------------------------------------------
+// StartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StartPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout not initialized");
+ return -1;
+ }
+
+ if (_playing)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout already started");
+ return 0;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID startPlaybackID = env->GetMethodID(_javaScClass, "StartPlayback",
+ "()I");
+
+ // Call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, startPlaybackID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "StartPlayback failed (%d)", res);
+ return -1;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+
+ // Signal to playout thread that we want to start
+ _startPlay = true;
+ _timeEventPlay.Set(); // Release thread from waiting state
+ _critSect.Leave();
+ // Wait for thread to init
+ if (kEventSignaled != _playStartStopEvent.Wait(5000))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Timeout or error starting");
+ }
+ _playStartStopEvent.Reset();
+ _critSect.Enter();
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::StopPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout is not initialized");
+ return 0;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID stopPlaybackID = env->GetMethodID(_javaScClass, "StopPlayback",
+ "()I");
+
+ // Call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, stopPlaybackID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "StopPlayback failed (%d)", res);
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+ _playWarning = 0;
+ _playError = 0;
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDelay
+//
+// Remaining amount of data still in the playout buffer.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = _delayPlayout;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDelay
+//
+// Remaining amount of data still in the recording buffer.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::RecordingDelay(
+ WebRtc_UWord16& delayMS) const
+{
+ delayMS = _delayRecording;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::Playing() const
+{
+
+ return _playing;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType /*type*/,
+ WebRtc_UWord16 /*sizeMS*/)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const
+{
+
+ type = AudioDeviceModule::kAdaptiveBufferSize;
+ sizeMS = _delayPlayout; // Set to current playout delay
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// CPULoad
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::CPULoad(WebRtc_UWord16& /*load*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::PlayoutWarning() const
+{
+ return (_playWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::PlayoutError() const
+{
+ return (_playError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::RecordingWarning() const
+{
+ return (_recWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::RecordingError() const
+{
+ return (_recError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceAndroidJni::ClearPlayoutWarning()
+{
+ _playWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceAndroidJni::ClearPlayoutError()
+{
+ _playError = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceAndroidJni::ClearRecordingWarning()
+{
+ _recWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceAndroidJni::ClearRecordingError()
+{
+ _recError = 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingSampleRate(
+ const WebRtc_UWord32 samplesPerSec)
+{
+
+ if (samplesPerSec > 48000 || samplesPerSec < 8000)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Invalid sample rate");
+ return -1;
+ }
+
+ // set the recording sample rate to use
+ if (samplesPerSec == 44100)
+ {
+ _samplingFreqIn = 44;
+ }
+ else
+ {
+ _samplingFreqIn = samplesPerSec / 1000;
+ }
+
+ // Update the AudioDeviceBuffer
+ _ptrAudioBuffer->SetRecordingSampleRate(samplesPerSec);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutSampleRate(
+ const WebRtc_UWord32 samplesPerSec)
+{
+
+ if (samplesPerSec > 48000 || samplesPerSec < 8000)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Invalid sample rate");
+ return -1;
+ }
+
+ // set the playout sample rate to use
+ if (samplesPerSec == 44100)
+ {
+ _samplingFreqOut = 44;
+ }
+ else
+ {
+ _samplingFreqOut = samplesPerSec / 1000;
+ }
+
+ // Update the AudioDeviceBuffer
+ _ptrAudioBuffer->SetPlayoutSampleRate(samplesPerSec);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetLoudspeakerStatus
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::SetLoudspeakerStatus(bool enable)
+{
+
+ if (!_javaContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ " Context is not set");
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ " Could not attach thread to JVM (%d, %p)", res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the method ID
+ jmethodID setPlayoutSpeakerID = env->GetMethodID(_javaScClass,
+ "SetPlayoutSpeaker",
+ "(Z)I");
+
+ // call java sc object method
+ jint res = env->CallIntMethod(_javaScObj, setPlayoutSpeakerID, enable);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ " SetPlayoutSpeaker failed (%d)", res);
+ return -1;
+ }
+
+ _loudSpeakerOn = enable;
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
+ " Could not detach thread from JVM");
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// GetLoudspeakerStatus
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::GetLoudspeakerStatus(bool& enabled) const
+{
+
+ enabled = _loudSpeakerOn;
+
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+
+// ----------------------------------------------------------------------------
+// InitJavaResources
+//
+// Initializes needed Java resources like the JNI interface to
+// AudioDeviceAndroid.java
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitJavaResources()
+{
+ // todo: Check if we already have created the java object
+ _javaVM = globalJvm;
+ _javaContext = globalSndContext;
+ _javaScClass = globalScClass;
+
+ // use the jvm that has been set
+ if (!_javaVM)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Not a valid Java VM pointer", __FUNCTION__);
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "get method id");
+
+ // get the method ID for the void(void) constructor
+ jmethodID cid = env->GetMethodID(_javaScClass, "<init>", "()V");
+ if (cid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get constructor ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "construct object", __FUNCTION__);
+
+ // construct the object
+ jobject javaScObjLocal = env->NewObject(_javaScClass, cid);
+ if (!javaScObjLocal)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "%s: could not create Java sc object", __FUNCTION__);
+ return -1;
+ }
+
+ // create a reference to the object (to tell JNI that we are referencing it
+ // after this function has returned)
+ _javaScObj = env->NewGlobalRef(javaScObjLocal);
+ if (!_javaScObj)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not create Java sc object reference",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local object ref, we only use the global ref
+ env->DeleteLocalRef(javaScObjLocal);
+
+ //////////////////////
+ // AUDIO MANAGEMENT
+
+ // This is not mandatory functionality
+ if (_javaContext)
+ {
+ // Get Context field ID
+ jfieldID fidContext = env->GetFieldID(_javaScClass, "_context",
+ "Landroid/content/Context;");
+ if (!fidContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get Context fid", __FUNCTION__);
+ return -1;
+ }
+
+ // Set the Java application Context so we can use AudioManager
+ // Get Context object and check it
+ jobject javaContext = (jobject) _javaContext;
+ env->SetObjectField(_javaScObj, fidContext, javaContext);
+ javaContext = env->GetObjectField(_javaScObj, fidContext);
+ if (!javaContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not set Context", __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local object ref
+ env->DeleteLocalRef(javaContext);
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceAudioDevice,
+ _id,
+ "%s: did not set Context - some functionality is not "
+ "supported",
+ __FUNCTION__);
+ }
+
+ /////////////
+ // PLAYOUT
+
+ // Get play buffer field ID
+ jfieldID fidPlayBuffer = env->GetFieldID(_javaScClass, "_playBuffer",
+ "Ljava/nio/ByteBuffer;");
+ if (!fidPlayBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get play buffer fid", __FUNCTION__);
+ return -1;
+ }
+
+ // Get play buffer object
+ jobject javaPlayBufferLocal =
+ env->GetObjectField(_javaScObj, fidPlayBuffer);
+ if (!javaPlayBufferLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get play buffer", __FUNCTION__);
+ return -1;
+ }
+
+ // Create a global reference to the object (to tell JNI that we are
+ // referencing it after this function has returned)
+ // NOTE: we are referencing it only through the direct buffer (see below)
+ _javaPlayBuffer = env->NewGlobalRef(javaPlayBufferLocal);
+ if (!_javaPlayBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get play buffer reference", __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local object ref, we only use the global ref
+ env->DeleteLocalRef(javaPlayBufferLocal);
+
+ // Get direct buffer
+ _javaDirectPlayBuffer = env->GetDirectBufferAddress(_javaPlayBuffer);
+ if (!_javaDirectPlayBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get direct play buffer", __FUNCTION__);
+ return -1;
+ }
+
+ // Get the play audio method ID
+ _javaMidPlayAudio = env->GetMethodID(_javaScClass, "PlayAudio", "(I)I");
+ if (!_javaMidPlayAudio)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get play audio mid", __FUNCTION__);
+ return -1;
+ }
+
+ //////////////
+ // RECORDING
+
+ // Get rec buffer field ID
+ jfieldID fidRecBuffer = env->GetFieldID(_javaScClass, "_recBuffer",
+ "Ljava/nio/ByteBuffer;");
+ if (!fidRecBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get rec buffer fid", __FUNCTION__);
+ return -1;
+ }
+
+ // Get rec buffer object
+ jobject javaRecBufferLocal = env->GetObjectField(_javaScObj, fidRecBuffer);
+ if (!javaRecBufferLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get rec buffer", __FUNCTION__);
+ return -1;
+ }
+
+ // Create a global reference to the object (to tell JNI that we are
+ // referencing it after this function has returned)
+ // NOTE: we are referencing it only through the direct buffer (see below)
+ _javaRecBuffer = env->NewGlobalRef(javaRecBufferLocal);
+ if (!_javaRecBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get rec buffer reference", __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local object ref, we only use the global ref
+ env->DeleteLocalRef(javaRecBufferLocal);
+
+ // Get direct buffer
+ _javaDirectRecBuffer = env->GetDirectBufferAddress(_javaRecBuffer);
+ if (!_javaDirectRecBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get direct rec buffer", __FUNCTION__);
+ return -1;
+ }
+
+ // Get the rec audio method ID
+ _javaMidRecAudio = env->GetMethodID(_javaScClass, "RecordAudio", "(I)I");
+ if (!_javaMidRecAudio)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: could not get rec audio mid", __FUNCTION__);
+ return -1;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitSampleRate
+//
+// checks supported sample rates for playback
+// and recording and initializes the rates to be used
+// Also stores the max playout volume returned from InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidJni::InitSampleRate()
+{
+ int samplingFreq = 44100;
+ jint res = 0;
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ if (_samplingFreqIn > 0)
+ {
+ // read the configured sampling rate
+ samplingFreq = 44100;
+ if (_samplingFreqIn != 44)
+ {
+ samplingFreq = _samplingFreqIn * 1000;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " Trying configured recording sampling rate %d",
+ samplingFreq);
+ }
+
+ // get the method ID
+ jmethodID initRecordingID = env->GetMethodID(_javaScClass, "InitRecording",
+ "(II)I");
+
+ bool keepTrying = true;
+ while (keepTrying)
+ {
+ // call java sc object method
+ res = env->CallIntMethod(_javaScObj, initRecordingID, _recAudioSource,
+ samplingFreq);
+ if (res < 0)
+ {
+ switch (samplingFreq)
+ {
+ case 44100:
+ samplingFreq = 16000;
+ break;
+ case 16000:
+ samplingFreq = 8000;
+ break;
+ default: // error
+ WEBRTC_TRACE(kTraceError,
+ kTraceAudioDevice, _id,
+ "%s: InitRecording failed (%d)", __FUNCTION__,
+ res);
+ return -1;
+ }
+ }
+ else
+ {
+ keepTrying = false;
+ }
+ }
+
+ // set the recording sample rate to use
+ if (samplingFreq == 44100)
+ {
+ _samplingFreqIn = 44;
+ }
+ else
+ {
+ _samplingFreqIn = samplingFreq / 1000;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ "Recording sample rate set to (%d)", _samplingFreqIn);
+
+ // get the method ID
+ jmethodID stopRecordingID = env->GetMethodID(_javaScClass, "StopRecording",
+ "()I");
+
+ // Call java sc object method
+ res = env->CallIntMethod(_javaScObj, stopRecordingID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "StopRecording failed (%d)", res);
+ }
+
+ // get the method ID
+ jmethodID initPlaybackID = env->GetMethodID(_javaScClass, "InitPlayback",
+ "(I)I");
+
+ if (_samplingFreqOut > 0)
+ {
+ // read the configured sampling rate
+ samplingFreq = 44100;
+ if (_samplingFreqOut != 44)
+ {
+ samplingFreq = _samplingFreqOut * 1000;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " Trying configured playback sampling rate %d",
+ samplingFreq);
+ }
+ else
+ {
+ // set the preferred sampling frequency
+ if (samplingFreq == 8000)
+ {
+ // try 16000
+ samplingFreq = 16000;
+ }
+ // else use same as recording
+ }
+
+ keepTrying = true;
+ while (keepTrying)
+ {
+ // call java sc object method
+ res = env->CallIntMethod(_javaScObj, initPlaybackID, samplingFreq);
+ if (res < 0)
+ {
+ switch (samplingFreq)
+ {
+ case 44100:
+ samplingFreq = 16000;
+ break;
+ case 16000:
+ samplingFreq = 8000;
+ break;
+ default: // error
+ WEBRTC_TRACE(kTraceError,
+ kTraceAudioDevice, _id,
+ "InitPlayback failed (%d)", res);
+ return -1;
+ }
+ }
+ else
+ {
+ keepTrying = false;
+ }
+ }
+
+ // Store max playout volume
+ _maxSpeakerVolume = static_cast<WebRtc_UWord32> (res);
+ if (_maxSpeakerVolume < 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Did not get valid max speaker volume value (%d)",
+ _maxSpeakerVolume);
+ }
+
+ // set the playback sample rate to use
+ if (samplingFreq == 44100)
+ {
+ _samplingFreqOut = 44;
+ }
+ else
+ {
+ _samplingFreqOut = samplingFreq / 1000;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ "Playback sample rate set to (%d)", _samplingFreqOut);
+
+ // get the method ID
+ jmethodID stopPlaybackID = env->GetMethodID(_javaScClass, "StopPlayback",
+ "()I");
+
+ // Call java sc object method
+ res = env->CallIntMethod(_javaScObj, stopPlaybackID);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "StopPlayback failed (%d)", res);
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ return 0;
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// PlayThreadFunc
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::PlayThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceAndroidJni*> (pThis)->PlayThreadProcess());
+}
+
+// ----------------------------------------------------------------------------
+// RecThreadFunc
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::RecThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceAndroidJni*> (pThis)->RecThreadProcess());
+}
+
+// ----------------------------------------------------------------------------
+// PlayThreadProcess
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::PlayThreadProcess()
+{
+ if (!_playThreadIsInitialized)
+ {
+ // Do once when thread is started
+
+ // Attach this thread to JVM and get the JNI env for this thread
+ jint res = _javaVM->AttachCurrentThread(&_jniEnvPlay, NULL);
+ if ((res < 0) || !_jniEnvPlay)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id,
+ "Could not attach playout thread to JVM (%d, %p)",
+ res, _jniEnvPlay);
+ return false; // Close down thread
+ }
+
+ _playThreadIsInitialized = true;
+ }
+
+ if (!_playing)
+ {
+ switch (_timeEventPlay.Wait(1000))
+ {
+ case kEventSignaled:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, "Playout thread event signal");
+ _timeEventPlay.Reset();
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, "Playout thread event error");
+ return true;
+ case kEventTimeout:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, "Playout thread event timeout");
+ return true;
+ }
+ }
+
+ Lock();
+
+ if (_startPlay)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "_startPlay true, performing initial actions");
+ _startPlay = false;
+ _playing = true;
+ _playWarning = 0;
+ _playError = 0;
+ _playStartStopEvent.Set();
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Sent signal");
+ }
+
+ if (_playing)
+ {
+ WebRtc_Word8 playBuffer[2 * 480]; // Max 10 ms @ 48 kHz / 16 bit
+ WebRtc_UWord32 samplesToPlay = _samplingFreqOut * 10;
+
+ // ask for new PCM data to be played out using the AudioDeviceBuffer
+ // ensure that this callback is executed without taking the
+ // audio-thread lock
+ UnLock();
+ WebRtc_UWord32 nSamples =
+ _ptrAudioBuffer->RequestPlayoutData(samplesToPlay);
+ Lock();
+
+ // Check again since play may have stopped during unlocked period
+ if (!_playing)
+ {
+ UnLock();
+ return true;
+ }
+
+ nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ if (nSamples != samplesToPlay)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " invalid number of output samples(%d)", nSamples);
+ _playWarning = 1;
+ }
+
+ // Copy data to our direct buffer (held by java sc object)
+ // todo: Give _javaDirectPlayBuffer directly to VoE?
+ memcpy(_javaDirectPlayBuffer, playBuffer, nSamples * 2);
+
+ UnLock();
+
+ // Call java sc object method to process data in direct buffer
+ // Will block until data has been put in OS playout buffer
+ // (see java sc class)
+ jint res = _jniEnvPlay->CallIntMethod(_javaScObj, _javaMidPlayAudio,
+ 2 * nSamples);
+ if (res < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "PlayAudio failed (%d)", res);
+ _playWarning = 1;
+ }
+ else if (res > 0)
+ {
+ // we are not recording and have got a delay value from playback
+ _delayPlayout = res / _samplingFreqOut;
+ }
+ // If 0 is returned we are recording and then play delay is updated
+ // in RecordProcess
+
+ Lock();
+
+ } // _playing
+
+ if (_shutdownPlayThread)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Detaching thread from Java VM");
+
+ // Detach thread from Java VM
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, "Could not detach playout thread from JVM");
+ _shutdownPlayThread = false;
+ // If we say OK (i.e. set event) and close thread anyway,
+ // app will crash
+ }
+ else
+ {
+ _jniEnvPlay = NULL;
+ _shutdownPlayThread = false;
+ _playStartStopEvent.Set(); // Signal to Terminate() that we are done
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Sent signal");
+ }
+ }
+
+ UnLock();
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// RecThreadProcess
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceAndroidJni::RecThreadProcess()
+{
+ if (!_recThreadIsInitialized)
+ {
+ // Do once when thread is started
+
+ // Attach this thread to JVM
+ jint res = _javaVM->AttachCurrentThread(&_jniEnvRec, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !_jniEnvRec)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, "Could not attach rec thread to JVM (%d, %p)",
+ res, _jniEnvRec);
+ return false; // Close down thread
+ }
+
+ _recThreadIsInitialized = true;
+ }
+
+ // just sleep if rec has not started
+ if (!_recording)
+ {
+ switch (_timeEventRec.Wait(1000))
+ {
+ case kEventSignaled:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, "Recording thread event signal");
+ _timeEventRec.Reset();
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, "Recording thread event error");
+ return true;
+ case kEventTimeout:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, "Recording thread event timeout");
+ return true;
+ }
+ }
+
+ Lock();
+
+ if (_startRec)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "_startRec true, performing initial actions");
+ _startRec = false;
+ _recording = true;
+ _recWarning = 0;
+ _recError = 0;
+ _recStartStopEvent.Set();
+ }
+
+ if (_recording)
+ {
+ WebRtc_UWord32 samplesToRec = _samplingFreqIn * 10;
+
+ // Call java sc object method to record data to direct buffer
+ // Will block until data has been recorded (see java sc class),
+ // therefore we must release the lock
+ UnLock();
+ jint playDelayInSamples = _jniEnvRec->CallIntMethod(_javaScObj,
+ _javaMidRecAudio,
+ 2 * samplesToRec);
+ if (playDelayInSamples < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "RecordAudio failed");
+ _recWarning = 1;
+ }
+ else
+ {
+ _delayPlayout = playDelayInSamples / _samplingFreqOut;
+ }
+ Lock();
+
+ // Check again since recording may have stopped during Java call
+ if (_recording)
+ {
+// WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+// "total delay is %d", msPlayDelay + _delayRecording);
+
+ // Copy data to our direct buffer (held by java sc object)
+ // todo: Give _javaDirectRecBuffer directly to VoE?
+ // todo: Check count <= 480 ?
+ memcpy(_recBuffer, _javaDirectRecBuffer, 2 * samplesToRec);
+
+ // store the recorded buffer (no action will be taken if the
+ // #recorded samples is not a full buffer)
+ _ptrAudioBuffer->SetRecordedBuffer(_recBuffer, samplesToRec);
+
+ // store vqe delay values
+ _ptrAudioBuffer->SetVQEData(_delayPlayout, _delayRecording, 0);
+
+ // deliver recorded samples at specified sample rate, mic level
+ // etc. to the observer using callback
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ Lock();
+ }
+
+ } // _recording
+
+ if (_shutdownRecThread)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Detaching rec thread from Java VM");
+
+ // Detach thread from Java VM
+ if (_javaVM->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, "Could not detach recording thread from JVM");
+ _shutdownRecThread = false;
+ // If we say OK (i.e. set event) and close thread anyway,
+ // app will crash
+ }
+ else
+ {
+ _jniEnvRec = NULL;
+ _shutdownRecThread = false;
+ _recStartStopEvent.Set(); // Signal to Terminate() that we are done
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Sent signal rec");
+ }
+ }
+
+ UnLock();
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h
new file mode 100644
index 0000000..e127e26
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h
@@ -0,0 +1,268 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Android audio device interface (JNI/AudioTrack/AudioRecord usage)
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
+
+#include "audio_device_generic.h"
+#include "critical_section_wrapper.h"
+
+#include <jni.h> // For accessing AudioDeviceAndroid java class
+
+namespace webrtc
+{
+class EventWrapper;
+
+const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz
+const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz
+
+const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
+const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout
+
+const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
+
+
+WebRtc_Word32 SetAndroidAudioDeviceObjects(void* javaVM, void* env,
+ void* context);
+
+class ThreadWrapper;
+
+class AudioDeviceAndroidJni: public AudioDeviceGeneric
+{
+public:
+ AudioDeviceAndroidJni(const WebRtc_Word32 id);
+ ~AudioDeviceAndroidJni();
+
+ virtual WebRtc_Word32 ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]);
+
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const;
+
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize)
+ const;
+
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ virtual WebRtc_Word32 SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(
+ AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ virtual WebRtc_Word32 SetRecordingSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+ virtual WebRtc_Word32 SetPlayoutSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+
+ virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
+ virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
+
+private:
+ // Lock
+ void Lock()
+ {
+ _critSect.Enter();
+ };
+ void UnLock()
+ {
+ _critSect.Leave();
+ };
+
+ // Init
+ WebRtc_Word32 InitJavaResources();
+ WebRtc_Word32 InitSampleRate();
+
+ // Threads
+ static bool RecThreadFunc(void*);
+ static bool PlayThreadFunc(void*);
+ bool RecThreadProcess();
+ bool PlayThreadProcess();
+
+ // Misc
+ AudioDeviceBuffer* _ptrAudioBuffer;
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+
+ // Events
+ EventWrapper& _timeEventRec;
+ EventWrapper& _timeEventPlay;
+ EventWrapper& _recStartStopEvent;
+ EventWrapper& _playStartStopEvent;
+
+ // Threads
+ ThreadWrapper* _ptrThreadPlay;
+ ThreadWrapper* _ptrThreadRec;
+ WebRtc_UWord32 _recThreadID;
+ WebRtc_UWord32 _playThreadID;
+ bool _playThreadIsInitialized;
+ bool _recThreadIsInitialized;
+ bool _shutdownPlayThread;
+ bool _shutdownRecThread;
+
+ // Rec buffer
+ WebRtc_Word8 _recBuffer[2 * REC_BUF_SIZE_IN_SAMPLES];
+
+ // States
+ bool _recordingDeviceIsSpecified;
+ bool _playoutDeviceIsSpecified;
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _micIsInitialized;
+ bool _speakerIsInitialized;
+
+ // Signal flags to threads
+ bool _startRec;
+ bool _stopRec;
+ bool _startPlay;
+ bool _stopPlay;
+
+ // Warnings and errors
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ // Delay
+ WebRtc_UWord16 _delayPlayout;
+ WebRtc_UWord16 _delayRecording;
+
+ // AGC state
+ bool _AGC;
+
+ // Stored device properties
+ WebRtc_UWord16 _samplingFreqIn; // Sampling frequency for Mic
+ WebRtc_UWord16 _samplingFreqOut; // Sampling frequency for Speaker
+ WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value
+ bool _loudSpeakerOn;
+ // Stores the desired audio source to use, set in SetRecordingDevice
+ int _recAudioSource;
+
+ // JNI and Java
+ JavaVM* _javaVM; // denotes a Java VM
+ jobject _javaContext; // the application context
+
+ JNIEnv* _jniEnvPlay; // The JNI env for playout thread
+ JNIEnv* _jniEnvRec; // The JNI env for recording thread
+
+ jclass _javaScClass; // AudioDeviceAndroid class
+ jobject _javaScObj; // AudioDeviceAndroid object
+
+ // The play buffer field in AudioDeviceAndroid object (global ref)
+ jobject _javaPlayBuffer;
+ // The rec buffer field in AudioDeviceAndroid object (global ref)
+ jobject _javaRecBuffer;
+ void* _javaDirectPlayBuffer; // Direct buffer pointer to play buffer
+ void* _javaDirectRecBuffer; // Direct buffer pointer to rec buffer
+ jmethodID _javaMidPlayAudio; // Method ID of play in AudioDeviceAndroid
+ jmethodID _javaMidRecAudio; // Method ID of rec in AudioDeviceAndroid
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc
new file mode 100644
index 0000000..da88a0e
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc
@@ -0,0 +1,2164 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <time.h>
+#include <sys/time.h>
+
+#include "audio_device_utility.h"
+#include "audio_device_android_opensles.h"
+#include "audio_device_config.h"
+
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+
+#ifdef WEBRTC_ANDROID_DEBUG
+#include <android/log.h>
+#define WEBRTC_TRACE(a,b,c,...) __android_log_print( \
+ ANDROID_LOG_DEBUG, "WebRTC ADM OpenSLES", __VA_ARGS__)
+#endif
+
+namespace webrtc {
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceAndroidOpenSLES - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceAndroidOpenSLES::AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _slEngineObject(NULL),
+ _slPlayer(NULL),
+ _slEngine(NULL),
+ _slPlayerPlay(NULL),
+ _slOutputMixObject(NULL),
+ _slSpeakerVolume(NULL),
+ _slRecorder(NULL),
+ _slRecorderRecord(NULL),
+ _slAudioIODeviceCapabilities(NULL),
+ _slRecorderSimpleBufferQueue(NULL),
+ _slMicVolume(NULL),
+ _micDeviceId(0),
+ _recQueueSeq(0),
+ _timeEventRec(*EventWrapper::Create()),
+ _ptrThreadRec(NULL),
+ _recThreadID(0),
+ _playQueueSeq(0),
+ _recCurrentSeq(0),
+ _recBufferTotalSize(0),
+ _recordingDeviceIsSpecified(false),
+ _playoutDeviceIsSpecified(false),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _micIsInitialized(false),
+ _speakerIsInitialized(false),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _playoutDelay(0),
+ _recordingDelay(0),
+ _AGC(false),
+ _adbSampleRate(0),
+ _samplingRateIn(SL_SAMPLINGRATE_16),
+ _samplingRateOut(SL_SAMPLINGRATE_16),
+ _maxSpeakerVolume(0),
+ _minSpeakerVolume(0),
+ _loudSpeakerOn(false) {
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created",
+ __FUNCTION__);
+ memset(_playQueueBuffer, 0, sizeof(_playQueueBuffer));
+ memset(_recQueueBuffer, 0, sizeof(_recQueueBuffer));
+ memset(_recBuffer, 0, sizeof(_recBuffer));
+ memset(_recLength, 0, sizeof(_recLength));
+ memset(_recSeqNumber, 0, sizeof(_recSeqNumber));
+}
+
+AudioDeviceAndroidOpenSLES::~AudioDeviceAndroidOpenSLES() {
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
+ __FUNCTION__);
+
+ Terminate();
+
+ delete &_timeEventRec;
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+void AudioDeviceAndroidOpenSLES::AttachAudioBuffer(
+ AudioDeviceBuffer* audioBuffer) {
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // inform the AudioBuffer about default settings for this implementation
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+ _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const {
+
+ audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::Init() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized) {
+ return 0;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ SLEngineOption EngineOption[] = { (SLuint32) SL_ENGINEOPTION_THREADSAFE,
+ (SLuint32) SL_BOOLEAN_TRUE };
+ WebRtc_Word32 res = slCreateEngine(&_slEngineObject, 1, EngineOption, 0,
+ NULL, NULL);
+ //WebRtc_Word32 res = slCreateEngine( &_slEngineObject, 0, NULL, 0, NULL,
+ // NULL);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create SL Engine Object");
+ return -1;
+ }
+ /* Realizing the SL Engine in synchronous mode. */
+ if ((*_slEngineObject)->Realize(_slEngineObject, SL_BOOLEAN_FALSE)
+ != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to Realize SL Engine");
+ return -1;
+ }
+
+ if ((*_slEngineObject)->GetInterface(_slEngineObject, SL_IID_ENGINE,
+ (void*) &_slEngine)
+ != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get SL Engine interface");
+ return -1;
+ }
+
+ // Check the sample rate to be used for playback and recording
+ if (InitSampleRate() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "%s: Failed to init samplerate", __FUNCTION__);
+ return -1;
+ }
+
+ // Set the audio device buffer sampling rate, we assume we get the same
+ // for play and record
+ if (_ptrAudioBuffer->SetRecordingSampleRate(_adbSampleRate) < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not set audio device buffer recording "
+ "sampling rate (%d)", _adbSampleRate);
+ }
+ if (_ptrAudioBuffer->SetPlayoutSampleRate(_adbSampleRate) < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not set audio device buffer playout sampling "
+ "rate (%d)", _adbSampleRate);
+ }
+
+ _initialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::Terminate() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized) {
+ return 0;
+ }
+
+ // RECORDING
+ StopRecording();
+
+ _micIsInitialized = false;
+ _recordingDeviceIsSpecified = false;
+
+ // PLAYOUT
+ StopPlayout();
+
+ if (_slEngineObject != NULL) {
+ (*_slEngineObject)->Destroy(_slEngineObject);
+ _slEngineObject = NULL;
+ _slEngine = NULL;
+ }
+
+ _initialized = false;
+
+ return 0;
+}
+
+bool AudioDeviceAndroidOpenSLES::Initialized() const {
+
+ return (_initialized);
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerIsAvailable(bool& available) {
+
+ // We always assume it's available
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitSpeaker() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Playout already started");
+ return -1;
+ }
+
+ if (!_playoutDeviceIsSpecified) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout device is not specified");
+ return -1;
+ }
+
+ // Nothing needs to be done here, we use a flag to have consistent
+ // behavior with other platforms
+ _speakerIsInitialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneIsAvailable(bool& available) {
+
+ // We always assume it's available
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitMicrophone() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Recording already started");
+ return -1;
+ }
+
+ if (!_recordingDeviceIsSpecified) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording device is not specified");
+ return -1;
+ }
+
+ // Nothing needs to be done here, we use a flag to have consistent
+ // behavior with other platforms
+ _micIsInitialized = true;
+
+ return 0;
+}
+
+bool AudioDeviceAndroidOpenSLES::SpeakerIsInitialized() const {
+
+ return _speakerIsInitialized;
+}
+
+bool AudioDeviceAndroidOpenSLES::MicrophoneIsInitialized() const {
+
+ return _micIsInitialized;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolumeIsAvailable(
+ bool& available) {
+
+ available = true; // We assume we are always be able to set/get volume
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetSpeakerVolume(
+ WebRtc_UWord32 volume) {
+
+ if (!_speakerIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ if (_slEngineObject == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "SetSpeakerVolume, SL Engine object doesnt exist");
+ return -1;
+ }
+
+ if (_slEngine == NULL) {
+ // Get the SL Engine Interface which is implicit
+ if ((*_slEngineObject)->GetInterface(_slEngineObject, SL_IID_ENGINE,
+ (void*) &_slEngine)
+ != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to GetInterface SL Engine Interface");
+ return -1;
+ }
+ }
+ /* if (_slOutputMixObject == NULL && _slEngine != NULL)
+ {
+ // Set arrays required[] and iidArray[] for VOLUME interface
+ const SLInterfaceID ids[1] = {SL_IID_VOLUME};
+ const SLboolean req[1] = {SL_BOOLEAN_TRUE};
+ // Create Output Mix object to be used by player
+ if ((*_slEngine)->CreateOutputMix(_slEngine, &_slOutputMixObject, 1, ids,
+ req) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Output Mix object");
+ return -1;
+ }
+
+ // Realizing the Output Mix object in synchronous mode.
+ if ((*_slOutputMixObject)->Realize(_slOutputMixObject,
+ SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to realize the output mix object");
+ return -1;
+ }
+ }
+ if (_slSpeakerVolume == NULL && _slOutputMixObject != NULL)
+ {
+ if ((*_slOutputMixObject)->GetInterface(_slOutputMixObject,
+ SL_IID_VOLUME, (void*)&_slSpeakerVolume) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get interface for Speaker Volume");
+ return -1;
+ }
+ }
+
+ WebRtc_Word32 vol(0);
+ vol = ((volume*(_maxSpeakerVolume-_minSpeakerVolume) + (int)(255/2))
+ / (255)) + _minSpeakerVolume;
+ if (_slSpeakerVolume != NULL)
+ {
+ if ((*_slSpeakerVolume)->SetVolumeLevel(_slSpeakerVolume, vol)
+ != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to set speaker volume");
+ return -1;
+ }
+ }
+ */
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolume(
+ WebRtc_UWord32& volume) const {
+
+ /* if (!_speakerIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ if (_slEngineObject == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "SetSpeakerVolume, SL Engine object doesnt exist");
+ return -1;
+ }
+
+ if (_slEngine == NULL && _slEngineObject != NULL)
+ {
+ // Get the SL Engine Interface which is implicit
+ if ((*_slEngineObject)->GetInterface(_slEngineObject,
+ SL_IID_ENGINE, (void*)&_slEngine) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get SL Engine interface");
+ return -1;
+ }
+ }
+ if (_slOutputMixObject == NULL && _slEngine != NULL)
+ {
+ // Set arrays required[] and iidArray[] for VOLUME interface
+ const SLInterfaceID ids[1] = {SL_IID_VOLUME};
+ const SLboolean req[1] = {SL_BOOLEAN_TRUE};
+ // Create Output Mix object to be used by player
+ if ((*_slEngine)->CreateOutputMix(_slEngine,
+ &_slOutputMixObject, 1, ids, req) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Output Mixer object");
+ return -1;
+ }
+ // Realizing the Output Mix object in synchronous mode.
+ if ((*_slOutputMixObject)->Realize(_slOutputMixObject,
+ SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to realize Output Mix object");
+ return -1;
+ }
+ }
+ if (_slSpeakerVolume == NULL && _slOutputMixObject != NULL)
+ {
+ if ((*_slOutputMixObject)->GetInterface(_slOutputMixObject,
+ SL_IID_VOLUME, (void*)&_slSpeakerVolume) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get speaker volume interface");
+ return -1;
+ }
+ }
+
+ SLmillibel vol(0);
+ if (_slSpeakerVolume != NULL)
+ {
+ if ((*_slSpeakerVolume)->GetVolumeLevel(_slSpeakerVolume, &vol)
+ != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get speaker volume");
+ return -1;
+ }
+ }
+ // volume has to be mapped from millibel to [0, 255]
+ // volume = (WebRtc_UWord32) (((vol - _minSpeakerVolume) * 255 +
+ * (int)((_maxSpeakerVolume - _minSpeakerVolume)/2)) /
+ * (_maxSpeakerVolume - _minSpeakerVolume));
+ */
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetWaveOutVolume(
+ WebRtc_UWord16 /*volumeLeft*/,
+ WebRtc_UWord16 /*volumeRight*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::WaveOutVolume(
+ WebRtc_UWord16& /*volumeLeft*/,
+ WebRtc_UWord16& /*volumeRight*/) const {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MaxSpeakerVolume(
+ WebRtc_UWord32& maxVolume) const {
+
+ if (!_speakerIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+
+ maxVolume = _maxSpeakerVolume;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MinSpeakerVolume(
+ WebRtc_UWord32& minVolume) const {
+
+ if (!_speakerIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;//
+ }
+
+ minVolume = _minSpeakerVolume;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const {
+
+ if (!_speakerIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Speaker not initialized");
+ return -1;
+ }
+ stepSize = 1;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerMuteIsAvailable(
+ bool& available) {
+
+ available = false; // Speaker mute not supported on Android
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetSpeakerMute(bool /*enable*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerMute(bool& /*enabled*/) const {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneMuteIsAvailable(
+ bool& available) {
+
+ available = false; // Mic mute not supported on Android
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneMute(bool /*enable*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneMute(
+ bool& /*enabled*/) const {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneBoostIsAvailable(
+ bool& available) {
+
+ available = false; // Mic boost not supported on Android
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneBoost(bool enable) {
+
+ if (!_micIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Microphone not initialized");
+ return -1;
+ }
+
+ if (enable) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneBoost(bool& enabled) const {
+
+ if (!_micIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Microphone not initialized");
+ return -1;
+ }
+
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoRecordingIsAvailable(
+ bool& available) {
+
+ available = false; // Stereo recording not supported on Android
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetStereoRecording(bool enable) {
+
+ if (enable) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoRecording(bool& enabled) const {
+
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoPlayoutIsAvailable(
+ bool& available) {
+
+ available = false; // Stereo playout not supported on Android
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetStereoPlayout(bool enable) {
+
+ if (enable) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Enabling not available");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoPlayout(bool& enabled) const {
+
+ enabled = false;
+
+ return 0;
+}
+
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetAGC(bool enable) {
+
+ _AGC = enable;
+
+ return 0;
+}
+
+bool AudioDeviceAndroidOpenSLES::AGC() const {
+
+ return _AGC;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolumeIsAvailable(
+ bool& available) {
+
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneVolume(
+ WebRtc_UWord32 volume) {
+
+ if (_slEngineObject == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "SetMicrophoneVolume, SL Engine Object doesnt exist");
+ return -1;
+ }
+
+ /* Get the optional DEVICE VOLUME interface from the engine */
+ if (_slMicVolume == NULL) {
+ // Get the optional DEVICE VOLUME interface from the engine
+ if ((*_slEngineObject)->GetInterface(_slEngineObject,
+ SL_IID_DEVICEVOLUME,
+ (void*) &_slMicVolume)
+ != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Output Mix object");
+ }
+ }
+
+ if (_slMicVolume != NULL) {
+ WebRtc_Word32 vol(0);
+ vol = ((volume * (_maxSpeakerVolume - _minSpeakerVolume) + (int) (255
+ / 2)) / (255)) + _minSpeakerVolume;
+ if ((*_slMicVolume)->SetVolume(_slMicVolume, _micDeviceId, vol)
+ != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Output Mix object");
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolume(
+ WebRtc_UWord32& /*volume*/) const {
+
+ /* if (_slEngineObject == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "MicrophoneVolume, SL Engine Object doesnt exist");
+ return -1;
+ }
+
+ // Get the optional DEVICE VOLUME interface from the engine
+ if (_slMicVolume == NULL)
+ {
+ // Get the optional DEVICE VOLUME interface from the engine
+ if ((*_slEngineObject)->GetInterface(_slEngineObject,
+ SL_IID_DEVICEVOLUME, (void*)&_slMicVolume) != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Microphone Volume interface");
+ }
+ }
+
+ SLint32 vol(0);
+ if (_slMicVolume != NULL)
+ {
+ if ((*_slMicVolume)->GetVolume(_slMicVolume, _micDeviceId, &vol)
+ != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Microphone Volume");
+ }
+ }
+ // volume has to be mapped from millibel to [0, 255]
+ // volume = (WebRtc_UWord32) (((vol - _minSpeakerVolume) * 255 +
+ * (int)((_maxSpeakerVolume - _minSpeakerVolume)/2)) /
+ * (_maxSpeakerVolume - _minSpeakerVolume));
+ */
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MaxMicrophoneVolume(
+ WebRtc_UWord32& /*maxVolume*/) const {
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MinMicrophoneVolume(
+ WebRtc_UWord32& minVolume) const {
+
+ minVolume = 0;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const {
+
+ stepSize = 1;
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceAndroidOpenSLES::PlayoutDevices() {
+
+ return 1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutDevice(WebRtc_UWord16 index) {
+
+ if (_playIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout already initialized");
+ return -1;
+ }
+
+ if (0 != index) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Do nothing but set a flag, this is to have consistent behaviour
+ // with other platforms
+ _playoutDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
+
+ if (0 != index) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Return empty string
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingDeviceName(
+ WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
+
+ if (0 != index) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Return empty string
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceAndroidOpenSLES::RecordingDevices() {
+
+ return 1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetRecordingDevice(
+ WebRtc_UWord16 index) {
+
+ if (_recIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording already initialized");
+ return -1;
+ }
+
+ if (0 != index) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Device index is out of range [0,0]");
+ return -1;
+ }
+
+ // Do nothing but set a flag, this is to have consistent behaviour with
+ // other platforms
+ _recordingDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutIsAvailable(bool& available) {
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1) {
+ available = true;
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingIsAvailable(bool& available) {
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1) {
+ available = true;
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitPlayout() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Not initialized");
+ return -1;
+ }
+
+ if (_playing) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Playout already started");
+ return -1;
+ }
+
+ if (!_playoutDeviceIsSpecified) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout device is not specified");
+ return -1;
+ }
+
+ if (_playIsInitialized) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout already initialized");
+ return 0;
+ }
+
+ // Initialize the speaker
+ if (InitSpeaker() == -1) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitSpeaker() failed");
+ }
+
+ if (_slEngineObject == NULL || _slEngine == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " SLObject or Engiine is NULL");
+ return -1;
+ }
+
+ WebRtc_Word32 res(-1);
+ SLDataFormat_PCM pcm;
+ SLDataSource audioSource;
+ SLDataLocator_AndroidSimpleBufferQueue simpleBufferQueue;
+ SLDataSink audioSink;
+ SLDataLocator_OutputMix locator_outputmix;
+
+ // Create Output Mix object to be used by player
+ SLInterfaceID ids[N_MAX_INTERFACES];
+ SLboolean req[N_MAX_INTERFACES];
+ for (unsigned int i = 0; i < N_MAX_INTERFACES; i++) {
+ ids[i] = SL_IID_NULL;
+ req[i] = SL_BOOLEAN_FALSE;
+ }
+ ids[0] = SL_IID_ENVIRONMENTALREVERB;
+ res = (*_slEngine)->CreateOutputMix(_slEngine, &_slOutputMixObject, 1, ids,
+ req);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get SL Output Mix object");
+ return -1;
+ }
+ // Realizing the Output Mix object in synchronous mode.
+ res = (*_slOutputMixObject)->Realize(_slOutputMixObject, SL_BOOLEAN_FALSE);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to realize SL Output Mix object");
+ return -1;
+ }
+ // Get the speaker mixer
+ /* res = (*_slOutputMixObject)->GetInterface(_slOutputMixObject,
+ * SL_IID_VOLUME, (void*)&_slSpeakerVolume);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Speaker Mixer");
+ return -1;
+ }
+ */
+ // The code below can be moved to startplayout instead
+ /* Setup the data source structure for the buffer queue */
+ simpleBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+ /* Two buffers in our buffer queue, to have low latency*/
+ simpleBufferQueue.numBuffers = N_PLAY_QUEUE_BUFFERS;
+ // TODO(xians), figure out if we should support stereo playout for android
+ /* Setup the format of the content in the buffer queue */
+ pcm.formatType = SL_DATAFORMAT_PCM;
+ pcm.numChannels = 1;
+ // _samplingRateOut is initilized in InitSampleRate()
+ pcm.samplesPerSec = SL_SAMPLINGRATE_16;
+ pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+ pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+ pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+ pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+ audioSource.pFormat = (void *) &pcm;
+ audioSource.pLocator = (void *) &simpleBufferQueue;
+ /* Setup the data sink structure */
+ locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+ locator_outputmix.outputMix = _slOutputMixObject;
+ audioSink.pLocator = (void *) &locator_outputmix;
+ audioSink.pFormat = NULL;
+
+ // Set arrays required[] and iidArray[] for SEEK interface
+ // (PlayItf is implicit)
+ ids[0] = SL_IID_BUFFERQUEUE;
+ ids[1] = SL_IID_EFFECTSEND;
+ req[0] = SL_BOOLEAN_TRUE;
+ req[1] = SL_BOOLEAN_TRUE;
+ // Create the music player
+ res = (*_slEngine)->CreateAudioPlayer(_slEngine, &_slPlayer, &audioSource,
+ &audioSink, 2, ids, req);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Audio Player");
+ return -1;
+ }
+
+ // Realizing the player in synchronous mode.
+ res = (*_slPlayer)->Realize(_slPlayer, SL_BOOLEAN_FALSE);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to realize the player");
+ return -1;
+ }
+ // Get seek and play interfaces
+ res = (*_slPlayer)->GetInterface(_slPlayer, SL_IID_PLAY,
+ (void*) &_slPlayerPlay);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Player interface");
+ return -1;
+ }
+ res = (*_slPlayer)->GetInterface(_slPlayer, SL_IID_BUFFERQUEUE,
+ (void*) &_slPlayerSimpleBufferQueue);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Player Simple Buffer Queue interface");
+ return -1;
+ }
+
+ // Setup to receive buffer queue event callbacks
+ res = (*_slPlayerSimpleBufferQueue)->RegisterCallback(
+ _slPlayerSimpleBufferQueue,
+ PlayerSimpleBufferQueueCallback,
+ this);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to register Player Callback");
+ return -1;
+ }
+ _playIsInitialized = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitRecording() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Not initialized");
+ return -1;
+ }
+
+ if (_recording) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Recording already started");
+ return -1;
+ }
+
+ if (!_recordingDeviceIsSpecified) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording device is not specified");
+ return -1;
+ }
+
+ if (_recIsInitialized) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording already initialized");
+ return 0;
+ }
+
+ // Initialize the microphone
+ if (InitMicrophone() == -1) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitMicrophone() failed");
+ }
+
+ if (_slEngineObject == NULL || _slEngine == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording object is NULL");
+ return -1;
+ }
+ WebRtc_Word32 res(-1);
+ WebRtc_Word32 numInputs(0);
+ WebRtc_UWord32 inputDeviceIDs[N_MAX_INPUT_DEVICES];
+ SLAudioInputDescriptor audioInputDescriptor;
+ SLDataSource audioSource;
+ SLDataLocator_IODevice micLocator;
+ SLDataSink audioSink;
+ SLDataFormat_PCM pcm;
+ SLDataLocator_AndroidSimpleBufferQueue simpleBufferQueue;
+ bool micAvailable(false);
+ WebRtc_UWord32 micDeviceID(0);
+
+ /* // Get the Audio IO DEVICE CAPABILITIES interface, which is also implicit
+ res = (*_slEngineObject)->GetInterface(_slEngineObject,
+ SL_IID_AUDIOIODEVICECAPABILITIES, (void*)&_slAudioIODeviceCapabilities);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Audio IO device Capacilities interface");
+ return -1;
+ }
+ numInputs = N_MAX_OUTPUT_DEVICES;
+
+ res = (*_slAudioIODeviceCapabilities)->GetAvailableAudioInputs(
+ _slAudioIODeviceCapabilities, &numInputs, inputDeviceIDs);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get the number of Input Devices");
+ return -1;
+ }
+
+ // Search for either earpiece microphone or headset microphone input
+ // device - with a preference for the latter
+ for (int i=0;i<numInputs; i++)
+ {
+ res = (*_slAudioIODeviceCapabilities)->QueryAudioInputCapabilities(
+ _slAudioIODeviceCapabilities, inputDeviceIDs[i], &audioInputDescriptor);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to query info for the Input Devices");
+ return -1;
+ }
+ if((audioInputDescriptor.deviceConnection ==
+ SL_DEVCONNECTION_ATTACHED_WIRED)
+ && (audioInputDescriptor.deviceScope == SL_DEVSCOPE_USER)
+ && (audioInputDescriptor.deviceLocation == SL_DEVLOCATION_HEADSET))
+ {
+ micDeviceID = inputDeviceIDs[i];
+ micAvailable = true;
+ break;
+ }
+ else if((audioInputDescriptor.deviceConnection ==
+ SL_DEVCONNECTION_INTEGRATED)
+ && (audioInputDescriptor.deviceScope == SL_DEVSCOPE_USER)
+ && (audioInputDescriptor.deviceLocation == SL_DEVLOCATION_HANDSET))
+ {
+ micDeviceID = inputDeviceIDs[i];
+ micAvailable = true;
+ break;
+ }
+ }
+ // If neither of the preferred input audio devices is available,
+ // no point in continuing
+ if (!micAvailable)
+ {
+ return -1;
+ }
+ */
+
+ // Get the optional DEVICE VOLUME interface from the engine,
+ // should this be done somewhere else
+ /* res = (*_slEngineObject)->GetInterface(_slEngineObject,
+ * SL_IID_DEVICEVOLUME, (void*)&_slMicVolume);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Microphone Volume interface");
+ return -1;
+ }
+ */
+ // Setup the data source structure
+ micLocator.locatorType = SL_DATALOCATOR_IODEVICE;
+ micLocator.deviceType = SL_IODEVICE_AUDIOINPUT;
+ micLocator.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; //micDeviceID;
+ micLocator.device = NULL;
+ audioSource.pLocator = (void *) &micLocator;
+ audioSource.pFormat = NULL;
+
+ /* Setup the data source structure for the buffer queue */
+ simpleBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+ simpleBufferQueue.numBuffers = N_REC_QUEUE_BUFFERS;
+ /* Setup the format of the content in the buffer queue */
+ pcm.formatType = SL_DATAFORMAT_PCM;
+ pcm.numChannels = 1;
+ // _samplingRateIn is initialized in initSampleRate()
+ pcm.samplesPerSec = SL_SAMPLINGRATE_16;
+ pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+ pcm.containerSize = 16;
+ pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+ pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+ audioSink.pFormat = (void *) &pcm;
+ audioSink.pLocator = (void *) &simpleBufferQueue;
+
+ // Create audio recorder
+ const SLInterfaceID id[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+ const SLboolean req[1] = { SL_BOOLEAN_TRUE };
+ res = (*_slEngine)->CreateAudioRecorder(_slEngine, &_slRecorder,
+ &audioSource, &audioSink, 1, id,
+ req);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create Recorder");
+ return -1;
+ }
+
+ // Realizing the recorder in synchronous mode.
+ res = (*_slRecorder)->Realize(_slRecorder, SL_BOOLEAN_FALSE);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to realize Recorder");
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " get Recorder interface");
+ // Get the RECORD interface - it is an implicit interface
+ res = (*_slRecorder)->GetInterface(_slRecorder, SL_IID_RECORD,
+ (void*) &_slRecorderRecord);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Recorder interface");
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " get Recorder Simple Buffer Queue 2");
+ // Get the simpleBufferQueue interface
+ res = (*_slRecorder)->GetInterface(_slRecorder,
+ SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+ (void*) &_slRecorderSimpleBufferQueue);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Recorder Simple Buffer Queue");
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " register Recorder Callback 2");
+ // Setup to receive buffer queue event callbacks
+ res = (*_slRecorderSimpleBufferQueue)->RegisterCallback(
+ _slRecorderSimpleBufferQueue,
+ RecorderSimpleBufferQueueCallback,
+ this);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to register Recorder Callback");
+ return -1;
+ }
+
+ _recIsInitialized = true;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StartRecording() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording not initialized");
+ return -1;
+ }
+
+ if (_recording) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording already started");
+ return 0;
+ }
+
+ if (_slRecorderRecord == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " RecordITF is NULL");
+ return -1;
+ }
+
+ if (_slRecorderSimpleBufferQueue == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recorder Simple Buffer Queue is NULL");
+ return -1;
+ }
+
+ // Reset recording buffer
+ memset(_recQueueBuffer, 0, sizeof(_recQueueBuffer)); // empty the queue
+ _recQueueSeq = 0;
+
+ const char* threadName = "webrtc_opensles_audio_capture_thread";
+ _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
+ kRealtimePriority, threadName);
+ if (_ptrThreadRec == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the rec audio thread");
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThreadRec->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the rec audio thread");
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ return -1;
+ }
+ _recThreadID = threadID;
+ _recThreadIsInitialized = true;
+ memset(_recBuffer, 0, sizeof(_recBuffer));
+ memset(_recLength, 0, sizeof(_recLength));
+ memset(_recSeqNumber, 0, sizeof(_recSeqNumber));
+ _recCurrentSeq = 0;
+ _recBufferTotalSize = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ // Enqueue N_REC_QUEUE_BUFFERS -1 zero buffers to get the ball rolling
+ // find out how it behaves when the sample rate is 44100
+ WebRtc_Word32 res(-1);
+ WebRtc_UWord32 nSample10ms = _adbSampleRate / 100;
+ for (int i = 0; i < (N_REC_QUEUE_BUFFERS - 1); i++) {
+ // We assign 10ms buffer to each queue, size given in bytes.
+ res = (*_slRecorderSimpleBufferQueue)->Enqueue(
+ _slRecorderSimpleBufferQueue,
+ (void*) _recQueueBuffer[_recQueueSeq],
+ 2 * nSample10ms);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to Enqueue Empty Buffer to recorder");
+ return -1;
+ }
+ _recQueueSeq++;
+ }
+ // Record the audio
+ res = (*_slRecorderRecord)->SetRecordState(_slRecorderRecord,
+ SL_RECORDSTATE_RECORDING);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to start recording");
+ return -1;
+ }
+ _recording = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StopRecording() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recording is not initialized");
+ return 0;
+ }
+
+ // Stop the recording thread
+ if (_ptrThreadRec != NULL)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Stopping capture thread");
+ bool res = _ptrThreadRec->Stop();
+ if (!res) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Failed to stop Capture thread ");
+ } else {
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ _recThreadIsInitialized = false;
+ }
+ }
+
+ if ((_slRecorderRecord != NULL) && (_slRecorder != NULL)) {
+ // Record the audio
+ WebRtc_Word32 res = (*_slRecorderRecord)->SetRecordState(
+ _slRecorderRecord,
+ SL_RECORDSTATE_STOPPED);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to stop recording");
+ return -1;
+ }
+ res = (*_slRecorderSimpleBufferQueue)->Clear(
+ _slRecorderSimpleBufferQueue);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to clear recorder buffer queue");
+ return -1;
+ }
+
+ // Destroy the recorder object
+ (*_slRecorder)->Destroy(_slRecorder);
+ _slRecorder = NULL;
+ _slRecorderRecord = NULL;
+ _slRecorderRecord = NULL;
+ }
+
+ _recIsInitialized = false;
+ _recording = false;
+ _recWarning = 0;
+ _recError = 0;
+ _recQueueSeq = 0;
+ return 0;
+}
+
+bool AudioDeviceAndroidOpenSLES::RecordingIsInitialized() const {
+
+ return _recIsInitialized;
+}
+
+
+bool AudioDeviceAndroidOpenSLES::Recording() const {
+
+ return _recording;
+}
+
+bool AudioDeviceAndroidOpenSLES::PlayoutIsInitialized() const {
+
+ return _playIsInitialized;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StartPlayout() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playout not initialized");
+ return -1;
+ }
+
+ if (_playing) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout already started");
+ return 0;
+ }
+
+ if (_slPlayerPlay == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " PlayItf is NULL");
+ return -1;
+ }
+ if (_slPlayerSimpleBufferQueue == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PlayerSimpleBufferQueue is NULL");
+ return -1;
+ }
+
+ _recQueueSeq = 0;
+
+ WebRtc_Word32 res(-1);
+ /* Enqueue a set of zero buffers to get the ball rolling */
+ WebRtc_UWord32 nSample10ms = _adbSampleRate / 100;
+ WebRtc_Word8 playBuffer[2 * nSample10ms];
+ WebRtc_UWord32 noSamplesOut(0);
+ /* res = (*_slPlayerSimpleBufferQueue)->Clear(_slPlayerSimpleBufferQueue);
+ if (res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " player simpler buffer queue clean failed");
+ //return ; dong return
+ }
+ */// for (int i = 0; i<(N_PLAY_QUEUE_BUFFERS -1); i++)
+ {
+ noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(nSample10ms);
+ //Lock();
+ // Get data from Audio Device Buffer
+ noSamplesOut = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ // Insert what we have in data buffer
+ memcpy(_playQueueBuffer[_playQueueSeq], playBuffer, 2 * noSamplesOut);
+ //UnLock();
+
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ // "_playQueueSeq (%u) noSamplesOut (%d)", _playQueueSeq,
+ //noSamplesOut);
+ // write the buffer data we got from VoE into the device
+ res = (*_slPlayerSimpleBufferQueue)->Enqueue(
+ _slPlayerSimpleBufferQueue,
+ (void*) _playQueueBuffer[_playQueueSeq],
+ 2 * noSamplesOut);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " player simpler buffer queue Enqueue failed, %d",
+ noSamplesOut);
+ //return ; dong return
+ }
+ _playQueueSeq = (_playQueueSeq + 1) % N_PLAY_QUEUE_BUFFERS;
+ }
+
+ // Play the PCM samples using a buffer queue
+ res = (*_slPlayerPlay)->SetPlayState(_slPlayerPlay, SL_PLAYSTATE_PLAYING);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to start playout");
+ return -1;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _playing = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::StopPlayout() {
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout is not initialized");
+ return 0;
+ }
+
+ if ((_slPlayerPlay != NULL) && (_slOutputMixObject == NULL) && (_slPlayer
+ == NULL)) {
+ // Make sure player is stopped
+ WebRtc_Word32 res =
+ (*_slPlayerPlay)->SetPlayState(_slPlayerPlay,
+ SL_PLAYSTATE_STOPPED);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to stop playout");
+ return -1;
+ }
+ res = (*_slPlayerSimpleBufferQueue)->Clear(_slPlayerSimpleBufferQueue);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to clear recorder buffer queue");
+ return -1;
+ }
+
+ // Destroy the player
+ (*_slPlayer)->Destroy(_slPlayer);
+ // Destroy Output Mix object
+ (*_slOutputMixObject)->Destroy(_slOutputMixObject);
+ _slPlayer = NULL;
+ _slPlayerPlay = NULL;
+ _slPlayerSimpleBufferQueue = NULL;
+ _slOutputMixObject = NULL;
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+ _playWarning = 0;
+ _playError = 0;
+ _playQueueSeq = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutDelay(WebRtc_UWord16& delayMS) const {
+ delayMS = _playoutDelay;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingDelay(WebRtc_UWord16& delayMS) const {
+ delayMS = _recordingDelay;
+
+ return 0;
+}
+
+bool AudioDeviceAndroidOpenSLES::Playing() const {
+
+ return _playing;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType /*type*/,
+ WebRtc_UWord16 /*sizeMS*/) {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const {
+
+ type = AudioDeviceModule::kAdaptiveBufferSize;
+ sizeMS = _playoutDelay; // Set to current playout delay
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::CPULoad(WebRtc_UWord16& /*load*/) const {
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+bool AudioDeviceAndroidOpenSLES::PlayoutWarning() const {
+ return (_playWarning > 0);
+}
+
+bool AudioDeviceAndroidOpenSLES::PlayoutError() const {
+ return (_playError > 0);
+}
+
+bool AudioDeviceAndroidOpenSLES::RecordingWarning() const {
+ return (_recWarning > 0);
+}
+
+bool AudioDeviceAndroidOpenSLES::RecordingError() const {
+ return (_recError > 0);
+}
+
+void AudioDeviceAndroidOpenSLES::ClearPlayoutWarning() {
+ _playWarning = 0;
+}
+
+void AudioDeviceAndroidOpenSLES::ClearPlayoutError() {
+ _playError = 0;
+}
+
+void AudioDeviceAndroidOpenSLES::ClearRecordingWarning() {
+ _recWarning = 0;
+}
+
+void AudioDeviceAndroidOpenSLES::ClearRecordingError() {
+ _recError = 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetLoudspeakerStatus(bool enable) {
+
+ // if (!_javaContext)
+ // {
+ // WEBRTC_TRACE(kTraceError, kTraceUtility, -1, " Context is not set");
+ // return -1;
+ // }
+
+ // get the JNI env for this thread
+ // JNIEnv *env;
+ // bool isAttached = false;
+
+ // if (_javaVM->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
+ // {
+ // try to attach the thread and get the env
+ // Attach this thread to JVMslPlayoutCallback
+ // jint res = _javaVM->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ // if ((res < 0) || !env)
+ // {
+ // WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ // " Could not attach thread to JVM (%d, %p)", res, env);
+ // return -1;
+ // }
+ // isAttached = true;
+ // }
+
+ // get the method ID
+ // jmethodID setPlayoutSpeakerID = env->GetMethodID(_javaScClass,
+ // "SetPlayoutSpeaker", "(Z)I");
+
+ // call java sc object method
+ // jint res = env->CallIntMethod(_javaScObj, setPlayoutSpeakerID, enable);
+ // if (res < 0)
+ // {
+ // WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ // " SetPlayoutSpeaker failed (%d)", res);
+ // return -1;
+ // }
+
+ _loudSpeakerOn = enable;
+
+ // Detach this thread if it was attached
+ // if (isAttached)
+ // {
+ // if (_javaVM->DetachCurrentThread() < 0)
+ // {
+ // WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
+ // " Could not detach thread from JVM");
+ // }
+ // }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::GetLoudspeakerStatus(
+ bool& enabled) const {
+
+ enabled = _loudSpeakerOn;
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+void AudioDeviceAndroidOpenSLES::PlayerSimpleBufferQueueCallback(
+ SLAndroidSimpleBufferQueueItf queueItf,
+ void *pContext) {
+ AudioDeviceAndroidOpenSLES* ptrThis =
+ static_cast<AudioDeviceAndroidOpenSLES*> (pContext);
+ ptrThis->PlayerSimpleBufferQueueCallbackHandler(queueItf);
+}
+
+void AudioDeviceAndroidOpenSLES::PlayerSimpleBufferQueueCallbackHandler(
+ SLAndroidSimpleBufferQueueItf queueItf) {
+ WebRtc_Word32 res;
+ //Lock();
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ //"_playQueueSeq (%u)", _playQueueSeq);
+ if (_playing && (_playQueueSeq < N_PLAY_QUEUE_BUFFERS)) {
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ //_id, "playout callback ");
+ unsigned int noSamp10ms = _adbSampleRate / 100;
+ // Max 10 ms @ samplerate kHz / 16 bit
+ WebRtc_Word8 playBuffer[2 * noSamp10ms];
+ int noSamplesOut = 0;
+
+ // Assumption for implementation
+ // assert(PLAYBUFSIZESAMPLES == noSamp10ms);
+
+ // TODO(xians), update the playout delay
+ //UnLock();
+
+ noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(noSamp10ms);
+ //Lock();
+ // Get data from Audio Device Buffer
+ noSamplesOut = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ // Cast OK since only equality comparison
+ if (noSamp10ms != (unsigned int) noSamplesOut) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "noSamp10ms (%u) != noSamplesOut (%d)", noSamp10ms,
+ noSamplesOut);
+
+ if (_playWarning > 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Pending play warning exists");
+ }
+ _playWarning = 1;
+ }
+ // Insert what we have in data buffer
+ memcpy(_playQueueBuffer[_playQueueSeq], playBuffer, 2 * noSamplesOut);
+ //UnLock();
+
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ //"_playQueueSeq (%u) noSamplesOut (%d)", _playQueueSeq, noSamplesOut);
+ // write the buffer data we got from VoE into the device
+ res
+ = (*_slPlayerSimpleBufferQueue)->Enqueue(
+ _slPlayerSimpleBufferQueue,
+ _playQueueBuffer[_playQueueSeq],
+ 2 * noSamplesOut);
+ if (res != SL_RESULT_SUCCESS) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " player simpler buffer queue Enqueue failed, %d",
+ noSamplesOut);
+ return;
+ }
+ // update the playout delay
+ UpdatePlayoutDelay(noSamplesOut);
+ // update the play buffer sequency
+ _playQueueSeq = (_playQueueSeq + 1) % N_PLAY_QUEUE_BUFFERS;
+ }
+}
+
+void AudioDeviceAndroidOpenSLES::RecorderSimpleBufferQueueCallback(
+ SLAndroidSimpleBufferQueueItf queueItf,
+ void *pContext) {
+ AudioDeviceAndroidOpenSLES* ptrThis =
+ static_cast<AudioDeviceAndroidOpenSLES*> (pContext);
+ ptrThis->RecorderSimpleBufferQueueCallbackHandler(queueItf);
+}
+
+void AudioDeviceAndroidOpenSLES::RecorderSimpleBufferQueueCallbackHandler(
+ SLAndroidSimpleBufferQueueItf queueItf) {
+ WebRtc_Word32 res;
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ //" RecorderSimpleBufferQueueCallbackHandler");
+ if (_recording) {
+ // Insert all data in temp buffer into recording buffers
+ // There is zero or one buffer partially full at any given time,
+ // all others are full or empty
+ // Full means filled with noSamp10ms samples.
+
+ const unsigned int noSamp10ms = _adbSampleRate / 100;
+ // WebRtc_UWord16 queuePos = 0;
+ // WebRtc_UWord16 checkQueuePos = 0;
+ unsigned int dataPos = 0;
+ WebRtc_UWord16 bufPos = 0;
+ WebRtc_Word16 insertPos = -1;
+ unsigned int nCopy = 0; // Number of samples to copy
+ // WebRtc_Word32 isData = 0;
+
+ while (dataPos < noSamp10ms)//REC_BUF_SIZE_IN_SAMPLES) //noSamp10ms)
+
+ {
+ // Loop over all recording buffers or until we find the partially
+ // full buffer
+ // First choice is to insert into partially full buffer,
+ // second choice is to insert into empty buffer
+ bufPos = 0;
+ insertPos = -1;
+ nCopy = 0;
+ while (bufPos < N_REC_BUFFERS)
+ {
+ if ((_recLength[bufPos] > 0) && (_recLength[bufPos]
+ < noSamp10ms))
+ {
+ // Found the partially full buffer
+ insertPos = static_cast<WebRtc_Word16> (bufPos);
+ bufPos = N_REC_BUFFERS; // Don't need to search more
+ }
+ else if ((-1 == insertPos) && (0 == _recLength[bufPos]))
+ {
+ // Found an empty buffer
+ insertPos = static_cast<WebRtc_Word16> (bufPos);
+ }
+ ++bufPos;
+ }
+
+ if (insertPos > -1)
+ {
+ // We found a non-full buffer, copy data from the buffer queue
+ // o recBuffer
+ unsigned int dataToCopy = noSamp10ms - dataPos;
+ unsigned int currentRecLen = _recLength[insertPos];
+ unsigned int roomInBuffer = noSamp10ms - currentRecLen;
+ nCopy = (dataToCopy < roomInBuffer ? dataToCopy : roomInBuffer);
+ memcpy(&_recBuffer[insertPos][currentRecLen],
+ &_recQueueBuffer[_recQueueSeq][dataPos],
+ nCopy * sizeof(short));
+ if (0 == currentRecLen)
+ {
+ _recSeqNumber[insertPos] = _recCurrentSeq;
+ ++_recCurrentSeq;
+ }
+ _recBufferTotalSize += nCopy;
+ // Has to be done last to avoid interrupt problems
+ // between threads
+ _recLength[insertPos] += nCopy;
+ dataPos += nCopy;
+ }
+ else
+ {
+ // Didn't find a non-full buffer
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, " Could not insert into recording buffer");
+ if (_recWarning > 0)
+ {
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceAudioDevice, _id,
+ " Pending rec warning exists");
+ }
+ _recWarning = 1;
+ dataPos = noSamp10ms; // Don't try to insert more
+ }
+ }
+
+ // clean the queue buffer
+ // Start with empty buffer
+ memset(_recQueueBuffer[_recQueueSeq], 0, 2 * REC_BUF_SIZE_IN_SAMPLES);
+ // write the empty buffer to the queue
+ res = (*_slRecorderSimpleBufferQueue)->Enqueue(
+ _slRecorderSimpleBufferQueue,
+ (void*) _recQueueBuffer[_recQueueSeq],
+ 2 * noSamp10ms);
+ if (res != SL_RESULT_SUCCESS) {
+ return;
+ }
+ // update the rec queue seq
+ _recQueueSeq = (_recQueueSeq + 1) % N_REC_QUEUE_BUFFERS;
+ // wake up the recording thread
+ _timeEventRec.Set();
+ }
+}
+
+void AudioDeviceAndroidOpenSLES::CheckErr(SLresult res) {
+ if (res != SL_RESULT_SUCCESS) {
+ // Debug printing to be placed here
+ exit(-1);
+ }
+}
+
+void AudioDeviceAndroidOpenSLES::UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed) {
+ // currently just do some simple calculation, should we setup a timer for
+ // the callback to have a more accurate delay
+ // Android CCD asks for 10ms as the maximum warm output latency, so we
+ // simply add (nPlayQueueBuffer -1 + 0.5)*10ms
+ // This playout delay should be seldom changed
+ _playoutDelay = (N_PLAY_QUEUE_BUFFERS - 0.5) * 10 + N_PLAY_QUEUE_BUFFERS
+ * nSamplePlayed / (_adbSampleRate / 1000);
+}
+
+void AudioDeviceAndroidOpenSLES::UpdateRecordingDelay() {
+ // // Android CCD asks for 10ms as the maximum warm input latency,
+ // so we simply add 10ms
+ _recordingDelay = 10;
+ const WebRtc_UWord32 noSamp10ms = _adbSampleRate / 100;
+ // if (_recBufferTotalSize > noSamp10ms)
+ // {
+ _recordingDelay += (N_REC_QUEUE_BUFFERS * noSamp10ms) / (_adbSampleRate
+ / 1000);
+ // }
+}
+
+WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitSampleRate() {
+
+ WebRtc_Word32 res(-1);
+ WebRtc_Word32 numOutputs(0);
+ WebRtc_UWord32 headsetDeviceID(0);
+ WebRtc_UWord32 earpieceDeviceID(0);
+ bool headsetAvailable(false);
+ bool earpieceAvailable(false);
+ bool foundSampleRate(false);
+ WebRtc_UWord32 outputDeviceIDs[N_MAX_OUTPUT_DEVICES];
+ SLAudioOutputDescriptor audioOutputDescriptor;
+
+ if (_slEngineObject == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " SL Object is NULL");
+ return -1;
+ }
+
+ /* // Get the Audio IO DEVICE CAPABILITIES interface
+ res = (*_slEngineObject)->GetInterface(_slEngineObject,
+ SL_IID_AUDIOIODEVICECAPABILITIES, (void*)&_slAudioIODeviceCapabilities);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get Device Capabilities interface");
+ return -1;
+ }
+ numOutputs = N_MAX_OUTPUT_DEVICES;
+ res = (*_slAudioIODeviceCapabilities)->GetAvailableAudioOutputs(
+ _slAudioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to get number of Output Devices");
+ return -1;
+ }
+
+ // Search for headset output and phone handsfree speaker device,
+ // we prefer headset to earpiece
+ for (int i=0;i<numOutputs; i++)
+ {
+ res = (*_slAudioIODeviceCapabilities)->QueryAudioOutputCapabilities(
+ _slAudioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor);
+ if ( res != SL_RESULT_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to query Output Devices info");
+ return -1;
+ }
+ if((audioOutputDescriptor.deviceConnection ==
+ SL_DEVCONNECTION_ATTACHED_WIRED)&&
+ (audioOutputDescriptor.deviceScope == SL_DEVSCOPE_USER)&&
+ (audioOutputDescriptor.deviceLocation == SL_DEVLOCATION_HEADSET))
+ {
+ headsetDeviceID = outputDeviceIDs[i];
+ headsetAvailable = true;
+ break;
+ }
+ else if((audioOutputDescriptor.deviceConnection ==
+ SL_DEVCONNECTION_INTEGRATED)&&
+ (audioOutputDescriptor.deviceScope == SL_DEVSCOPE_ENVIRONMENT)&&
+ (audioOutputDescriptor.deviceLocation == SL_DEVLOCATION_HANDSET))
+ {
+ earpieceDeviceID = outputDeviceIDs[i];
+ earpieceAvailable = true;
+ break;
+ }
+ }
+ // Neither headset output nor phone handsfree speaker is available
+ if (headsetAvailable == false && earpieceAvailable == false)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " No playout device");
+ return -1;
+ }
+
+ _sampleRateInMilliHz = SL_SAMPLINGRATE_16;
+ if (audioOutputDescriptor.isFreqRangeContinuous == SL_BOOLEAN_FALSE)
+ {
+ while (!foundSampleRate)
+ {
+ for (int i=0; i<audioOutputDescriptor.numOfSamplingRatesSupported; i++)
+ {
+ if (audioOutputDescriptor.samplingRatesSupported[i]
+ == _sampleRateInMilliHz) // supported sampling rate in milliHertz
+ {
+ switch (_sampleRateInMilliHz)
+ {
+ case SL_SAMPLINGRATE_44_1:
+ _adbSampleRate = 44000;
+ break;
+ case SL_SAMPLINGRATE_16:
+ _adbSampleRate = 16000;
+ break;
+ case SL_SAMPLINGRATE_8:
+ _adbSampleRate = 8000;
+ break;
+ default: // error
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " SampleRate(%d) is not supported", _sampleRateInMilliHz);
+ return -1;
+ } // switch
+ foundSampleRate = true;
+ break;
+ } //if(audioOutputDescriptor.samplingRatesSupported[i] == _sampleRate
+ } //for (int i=0; i<audioOutputDescriptor.numOfSamplingRatesSupported; i++)
+ switch (_sampleRateInMilliHz)
+ {
+ case SL_SAMPLINGRATE_16:
+ _sampleRateInMilliHz = SL_SAMPLINGRATE_44_1;
+ break;
+ case SL_SAMPLINGRATE_44_1:
+ _sampleRateInMilliHz = SL_SAMPLINGRATE_8;
+ break;
+ default:
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " SampleRate is not supported");
+ return -1;
+ } // switch
+ } // while (!foundSampleRate)
+ }
+ else //audioOutputDescriptor.isFreqRangeContinuous == SL_BOOLEAN_TRUE
+ {
+ // minSampleRate < SL_SAMPLINGRATE_16 < maxSampleRate
+ if ((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_16) &&
+ (SL_SAMPLINGRATE_16 < audioOutputDescriptor.maxSampleRate))
+ {
+ _adbSampleRate = 16000;
+ } // minSampleRate < SL_SAMPLINGRATE_44_1 < maxSampleRate
+ else if((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_44_1) &&
+ (SL_SAMPLINGRATE_44_1 < audioOutputDescriptor.maxSampleRate))
+ {
+ _adbSampleRate = 44000;
+ } // minSampleRate < SL_SAMPLINGRATE_8 < maxSampleRate
+ else if ((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_8) &&
+ (SL_SAMPLINGRATE_8 < audioOutputDescriptor.maxSampleRate))
+ {
+ _adbSampleRate = 8000;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " SampleRate is not supported");
+ return -1;
+ }
+ } // else
+ */
+ _samplingRateIn = SL_SAMPLINGRATE_16;
+ _samplingRateOut = SL_SAMPLINGRATE_16;
+ _adbSampleRate = 16000;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " sample rate set to (%d)", _adbSampleRate);
+ return 0;
+
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+bool AudioDeviceAndroidOpenSLES::RecThreadFunc(void* pThis) {
+ return (static_cast<AudioDeviceAndroidOpenSLES*> (pThis)->RecThreadProcess());
+}
+
+bool AudioDeviceAndroidOpenSLES::RecThreadProcess() {
+
+ // Lock();
+ // Wait for 100ms for the signal from device callback
+ // In case no callback comes in 100ms, we check the buffer anyway
+ _timeEventRec.Wait(100);
+
+ int bufPos = 0;
+ unsigned int lowestSeq = 0;
+ int lowestSeqBufPos = 0;
+ bool foundBuf = true;
+ const unsigned int noSamp10ms = _adbSampleRate / 100;
+
+ while (foundBuf)
+ {
+ // Check if we have any buffer with data to insert into the
+ // Audio Device Buffer,
+ // and find the one with the lowest seq number
+ foundBuf = false;
+
+ for (bufPos = 0; bufPos < N_REC_BUFFERS; ++bufPos)
+ {
+ if (noSamp10ms == _recLength[bufPos])
+ {
+ if (!foundBuf) {
+ lowestSeq = _recSeqNumber[bufPos];
+ lowestSeqBufPos = bufPos;
+ foundBuf = true;
+ } else if (_recSeqNumber[bufPos] < lowestSeq)
+ {
+ lowestSeq = _recSeqNumber[bufPos];
+ lowestSeqBufPos = bufPos;
+ }
+ }
+ } // for
+
+ // Insert data into the Audio Device Buffer if found any
+ if (foundBuf)
+ {
+ UpdateRecordingDelay();
+ // Set the recorded buffer
+ _ptrAudioBuffer->SetRecordedBuffer(_recBuffer[lowestSeqBufPos],
+ noSamp10ms);
+
+ // Don't need to set the current mic level in ADB since we only
+ // support digital AGC,
+ // and besides we cannot get or set the iPhone mic level anyway.
+
+ // Set VQE info, use clockdrift == 0
+ _ptrAudioBuffer->SetVQEData(_playoutDelay, _recordingDelay, 0);
+
+ // Deliver recorded samples at specified sample rate, mic level
+ // etc. to the observer using callback
+ //UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ //Lock();
+
+ // Make buffer available
+ _recSeqNumber[lowestSeqBufPos] = 0;
+ _recBufferTotalSize -= _recLength[lowestSeqBufPos];
+ // Must be done last to avoid interrupt problems between threads
+ _recLength[lowestSeqBufPos] = 0;
+ }
+
+ } // while (foundBuf)
+ //UnLock();
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h
new file mode 100644
index 0000000..974ae90
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h
@@ -0,0 +1,310 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
+
+#include "audio_device_generic.h"
+#include "critical_section_wrapper.h"
+
+#include <jni.h> // For accessing AudioDeviceAndroid.java
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+namespace webrtc
+{
+class EventWrapper;
+
+const WebRtc_UWord32 N_MAX_INTERFACES = 3;
+const WebRtc_UWord32 N_MAX_OUTPUT_DEVICES = 6;
+const WebRtc_UWord32 N_MAX_INPUT_DEVICES = 3;
+
+const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 16000;//44000; // Default fs
+const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000;//44000; // Default fs
+
+const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
+const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout
+
+const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
+const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = 480;
+
+// Number of the buffers in playout queue
+const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 2;
+// Number of buffers in recording queue
+const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 2;
+// Number of 10 ms recording blocks in rec buffer
+const WebRtc_UWord16 N_REC_BUFFERS = 20;
+
+class ThreadWrapper;
+
+class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric
+{
+public:
+ AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id);
+ ~AudioDeviceAndroidOpenSLES();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32
+ ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32
+ PlayoutDeviceName(WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32
+ RecordingDeviceName(WebRtc_UWord16 index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32
+ SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32
+ SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ SLPlayItf playItf;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32
+ MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32
+ SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+ // Error and warning information
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+ // Attach audio buffer
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ // Speaker audio routing
+ virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
+ virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
+
+private:
+ // Lock
+ void Lock()
+ {
+ _critSect.Enter();
+ };
+ void UnLock()
+ {
+ _critSect.Leave();
+ };
+
+ static void PlayerSimpleBufferQueueCallback(
+ SLAndroidSimpleBufferQueueItf queueItf,
+ void *pContext);
+ void PlayerSimpleBufferQueueCallbackHandler(
+ SLAndroidSimpleBufferQueueItf queueItf);
+ static void RecorderSimpleBufferQueueCallback(
+ SLAndroidSimpleBufferQueueItf queueItf,
+ void *pContext);
+ void RecorderSimpleBufferQueueCallbackHandler(
+ SLAndroidSimpleBufferQueueItf queueItf);
+ void CheckErr(SLresult res);
+
+ // Delay updates
+ void UpdateRecordingDelay();
+ void UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed);
+
+ // Init
+ WebRtc_Word32 InitSampleRate();
+
+ // Threads
+ static bool RecThreadFunc(void*);
+ static bool PlayThreadFunc(void*);
+ bool RecThreadProcess();
+ bool PlayThreadProcess();
+
+ // Misc
+ AudioDeviceBuffer* _ptrAudioBuffer;
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+
+ // audio unit
+ SLObjectItf _slEngineObject;
+
+ // playout device
+ SLObjectItf _slPlayer;
+ SLEngineItf _slEngine;
+ SLPlayItf _slPlayerPlay;
+ SLAndroidSimpleBufferQueueItf _slPlayerSimpleBufferQueue;
+ SLObjectItf _slOutputMixObject;
+ SLVolumeItf _slSpeakerVolume;
+
+ // recording device
+ SLObjectItf _slRecorder;
+ SLRecordItf _slRecorderRecord;
+ SLAudioIODeviceCapabilitiesItf _slAudioIODeviceCapabilities;
+ SLAndroidSimpleBufferQueueItf _slRecorderSimpleBufferQueue;
+ SLDeviceVolumeItf _slMicVolume;
+
+ WebRtc_UWord32 _micDeviceId;
+
+ // Events
+ EventWrapper& _timeEventRec;
+ // Threads
+ ThreadWrapper* _ptrThreadRec;
+ WebRtc_UWord32 _recThreadID;
+ // TODO(xians), remove the following flag
+ bool _recThreadIsInitialized;
+
+ // Playout buffer
+ WebRtc_Word8 _playQueueBuffer[N_PLAY_QUEUE_BUFFERS][2
+ * PLAY_BUF_SIZE_IN_SAMPLES];
+ WebRtc_UWord32 _playQueueSeq;
+ // Recording buffer
+ WebRtc_Word8 _recQueueBuffer[N_REC_QUEUE_BUFFERS][2
+ * REC_BUF_SIZE_IN_SAMPLES];
+ WebRtc_UWord32 _recQueueSeq;
+ WebRtc_Word8 _recBuffer[N_REC_BUFFERS][2*REC_BUF_SIZE_IN_SAMPLES];
+ WebRtc_UWord32 _recLength[N_REC_BUFFERS];
+ WebRtc_UWord32 _recSeqNumber[N_REC_BUFFERS];
+ WebRtc_UWord32 _recCurrentSeq;
+ // Current total size all data in buffers, used for delay estimate
+ WebRtc_UWord32 _recBufferTotalSize;
+
+ // States
+ bool _recordingDeviceIsSpecified;
+ bool _playoutDeviceIsSpecified;
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _micIsInitialized;
+ bool _speakerIsInitialized;
+
+ // Warnings and errors
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ // Delay
+ WebRtc_UWord16 _playoutDelay;
+ WebRtc_UWord16 _recordingDelay;
+
+ // AGC state
+ bool _AGC;
+
+ // The sampling rate to use with Audio Device Buffer
+ WebRtc_UWord32 _adbSampleRate;
+ // Stored device properties
+ WebRtc_UWord32 _samplingRateIn; // Sampling frequency for Mic
+ WebRtc_UWord32 _samplingRateOut; // Sampling frequency for Speaker
+ WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value
+ WebRtc_UWord32 _minSpeakerVolume; // The minimum speaker volume value
+ bool _loudSpeakerOn;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc
new file mode 100644
index 0000000..84b4c60
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Android audio device utility implementation
+ */
+
+#include "audio_device_utility_android.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+AudioDeviceUtilityAndroid::AudioDeviceUtilityAndroid(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
+ _lastError(AudioDeviceModule::kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+}
+
+AudioDeviceUtilityAndroid::~AudioDeviceUtilityAndroid()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+ }
+
+ delete &_critSect;
+}
+
+WebRtc_Word32 AudioDeviceUtilityAndroid::Init()
+{
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " OS info: %s", "Android");
+
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h
new file mode 100644
index 0000000..81f685a
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Android audio device utility interface
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
+
+#include "audio_device_utility.h"
+#include "audio_device.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class AudioDeviceUtilityAndroid: public AudioDeviceUtility
+{
+public:
+ AudioDeviceUtilityAndroid(const WebRtc_Word32 id);
+ ~AudioDeviceUtilityAndroid();
+
+ virtual WebRtc_Word32 Init();
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ AudioDeviceModule::ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
diff --git a/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java b/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
new file mode 100644
index 0000000..b56085b
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
@@ -0,0 +1,509 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Android audio device test app
+ */
+
+package org.webrtc.voiceengine;
+
+import java.nio.ByteBuffer;
+import java.util.concurrent.locks.ReentrantLock;
+
+import android.content.Context;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.util.Log;
+
+
+class AudioDeviceAndroid {
+ private AudioTrack _audioTrack = null;
+ private AudioRecord _audioRecord = null;
+
+ private Context _context;
+ private AudioManager _audioManager;
+
+ private ByteBuffer _playBuffer;
+ private ByteBuffer _recBuffer;
+ private byte[] _tempBufPlay;
+ private byte[] _tempBufRec;
+
+ private final ReentrantLock _playLock = new ReentrantLock();
+ private final ReentrantLock _recLock = new ReentrantLock();
+
+ private boolean _doPlayInit = true;
+ private boolean _doRecInit = true;
+ private boolean _isRecording = false;
+ private boolean _isPlaying = false;
+
+ private int _bufferedRecSamples = 0;
+ private int _bufferedPlaySamples = 0;
+ private int _playPosition = 0;
+
+ AudioDeviceAndroid() {
+ try {
+ _playBuffer = ByteBuffer.allocateDirect(2 * 480); // Max 10 ms @ 48
+ // kHz
+ _recBuffer = ByteBuffer.allocateDirect(2 * 480); // Max 10 ms @ 48
+ // kHz
+ } catch (Exception e) {
+ DoLog(e.getMessage());
+ }
+
+ _tempBufPlay = new byte[2 * 480];
+ _tempBufRec = new byte[2 * 480];
+ }
+
+ @SuppressWarnings("unused")
+ private int InitRecording(int audioSource, int sampleRate) {
+ // get the minimum buffer size that can be used
+ int minRecBufSize =
+ AudioRecord.getMinBufferSize(sampleRate,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+
+ // DoLog("min rec buf size is " + minRecBufSize);
+
+ // double size to be more safe
+ int recBufSize = minRecBufSize * 2;
+ _bufferedRecSamples = (5 * sampleRate) / 200;
+ // DoLog("rough rec delay set to " + _bufferedRecSamples);
+
+ // release the object
+ if (_audioRecord != null) {
+ _audioRecord.release();
+ _audioRecord = null;
+ }
+
+ try {
+ _audioRecord = new AudioRecord(
+ audioSource,
+ sampleRate,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ recBufSize);
+
+ } catch (Exception e) {
+ DoLog(e.getMessage());
+ return -1;
+ }
+
+ // check that the audioRecord is ready to be used
+ if (_audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
+ // DoLog("rec not initialized " + sampleRate);
+ return -1;
+ }
+
+ // DoLog("rec sample rate set to " + sampleRate);
+
+ return _bufferedRecSamples;
+ }
+
+ @SuppressWarnings("unused")
+ private int StartRecording() {
+ if (_isPlaying == false) {
+ SetAudioMode(true);
+ }
+
+ // start recording
+ try {
+ _audioRecord.startRecording();
+
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ return -1;
+ }
+
+ _isRecording = true;
+ return 0;
+ }
+
+ @SuppressWarnings("unused")
+ private int InitPlayback(int sampleRate) {
+ // get the minimum buffer size that can be used
+ int minPlayBufSize =
+ AudioTrack.getMinBufferSize(sampleRate,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+
+ // DoLog("min play buf size is " + minPlayBufSize);
+
+ int playBufSize = minPlayBufSize;
+ if (playBufSize < 6000) {
+ playBufSize *= 2;
+ }
+ _bufferedPlaySamples = 0;
+ // DoLog("play buf size is " + playBufSize);
+
+ // release the object
+ if (_audioTrack != null) {
+ _audioTrack.release();
+ _audioTrack = null;
+ }
+
+ try {
+ _audioTrack = new AudioTrack(
+ AudioManager.STREAM_VOICE_CALL,
+ sampleRate,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ playBufSize, AudioTrack.MODE_STREAM);
+ } catch (Exception e) {
+ DoLog(e.getMessage());
+ return -1;
+ }
+
+ // check that the audioRecord is ready to be used
+ if (_audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
+ // DoLog("play not initialized " + sampleRate);
+ return -1;
+ }
+
+ // DoLog("play sample rate set to " + sampleRate);
+
+ if (_audioManager == null && _context != null) {
+ _audioManager = (AudioManager)
+ _context.getSystemService(Context.AUDIO_SERVICE);
+ }
+
+ // Return max playout volume
+ if (_audioManager == null) {
+ // Don't know the max volume but still init is OK for playout,
+ // so we should not return error.
+ return 0;
+ }
+ return _audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
+ }
+
+ @SuppressWarnings("unused")
+ private int StartPlayback() {
+ if (_isRecording == false) {
+ SetAudioMode(true);
+ }
+
+ // start playout
+ try {
+ _audioTrack.play();
+
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ return -1;
+ }
+
+ _isPlaying = true;
+ return 0;
+ }
+
+ @SuppressWarnings("unused")
+ private int StopRecording() {
+ _recLock.lock();
+ try {
+ // only stop if we are recording
+ if (_audioRecord.getRecordingState() ==
+ AudioRecord.RECORDSTATE_RECORDING) {
+ // stop recording
+ try {
+ _audioRecord.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ return -1;
+ }
+ }
+
+ // release the object
+ _audioRecord.release();
+ _audioRecord = null;
+
+ } finally {
+ // Ensure we always unlock, both for success, exception or error
+ // return.
+ _doRecInit = true;
+ _recLock.unlock();
+ }
+
+ if (_isPlaying == false) {
+ SetAudioMode(false);
+ }
+
+ _isRecording = false;
+ return 0;
+ }
+
+ @SuppressWarnings("unused")
+ private int StopPlayback() {
+ _playLock.lock();
+ try {
+ // only stop if we are playing
+ if (_audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
+ // stop playout
+ try {
+ _audioTrack.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ return -1;
+ }
+
+ // flush the buffers
+ _audioTrack.flush();
+ }
+
+ // release the object
+ _audioTrack.release();
+ _audioTrack = null;
+
+ } finally {
+ // Ensure we always unlock, both for success, exception or error
+ // return.
+ _doPlayInit = true;
+ _playLock.unlock();
+ }
+
+ if (_isRecording == false) {
+ SetAudioMode(false);
+ }
+
+ _isPlaying = false;
+ return 0;
+ }
+
+ @SuppressWarnings("unused")
+ private int PlayAudio(int lengthInBytes) {
+
+ int bufferedSamples = 0;
+
+ _playLock.lock();
+ try {
+ if (_audioTrack == null) {
+ return -2; // We have probably closed down while waiting for
+ // play lock
+ }
+
+ // Set priority, only do once
+ if (_doPlayInit == true) {
+ try {
+ android.os.Process.setThreadPriority(
+ android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+ } catch (Exception e) {
+ DoLog("Set play thread priority failed: " + e.getMessage());
+ }
+ _doPlayInit = false;
+ }
+
+ int written = 0;
+ _playBuffer.get(_tempBufPlay);
+ written = _audioTrack.write(_tempBufPlay, 0, lengthInBytes);
+ _playBuffer.rewind(); // Reset the position to start of buffer
+
+ // DoLog("Wrote data to sndCard");
+
+ // increase by number of written samples
+ _bufferedPlaySamples += (written >> 1);
+
+ // decrease by number of played samples
+ int pos = _audioTrack.getPlaybackHeadPosition();
+ if (pos < _playPosition) { // wrap or reset by driver
+ _playPosition = 0; // reset
+ }
+ _bufferedPlaySamples -= (pos - _playPosition);
+ _playPosition = pos;
+
+ if (!_isRecording) {
+ bufferedSamples = _bufferedPlaySamples;
+ }
+
+ if (written != lengthInBytes) {
+ // DoLog("Could not write all data to sc (written = " + written
+ // + ", length = " + lengthInBytes + ")");
+ return -1;
+ }
+
+ } finally {
+ // Ensure we always unlock, both for success, exception or error
+ // return.
+ _playLock.unlock();
+ }
+
+ return bufferedSamples;
+ }
+
+ @SuppressWarnings("unused")
+ private int RecordAudio(int lengthInBytes) {
+ _recLock.lock();
+
+ try {
+ if (_audioRecord == null) {
+ return -2; // We have probably closed down while waiting for rec
+ // lock
+ }
+
+ // Set priority, only do once
+ if (_doRecInit == true) {
+ try {
+ android.os.Process.setThreadPriority(
+ android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+ } catch (Exception e) {
+ DoLog("Set rec thread priority failed: " + e.getMessage());
+ }
+ _doRecInit = false;
+ }
+
+ int readBytes = 0;
+ _recBuffer.rewind(); // Reset the position to start of buffer
+ readBytes = _audioRecord.read(_tempBufRec, 0, lengthInBytes);
+ // DoLog("read " + readBytes + "from SC");
+ _recBuffer.put(_tempBufRec);
+
+ if (readBytes != lengthInBytes) {
+ // DoLog("Could not read all data from sc (read = " + readBytes
+ // + ", length = " + lengthInBytes + ")");
+ return -1;
+ }
+
+ } catch (Exception e) {
+ DoLogErr("RecordAudio try failed: " + e.getMessage());
+
+ } finally {
+ // Ensure we always unlock, both for success, exception or error
+ // return.
+ _recLock.unlock();
+ }
+
+ return (_bufferedPlaySamples);
+ }
+
+ @SuppressWarnings("unused")
+ private int SetPlayoutSpeaker(boolean loudspeakerOn) {
+ // create audio manager if needed
+ if (_audioManager == null && _context != null) {
+ _audioManager = (AudioManager)
+ _context.getSystemService(Context.AUDIO_SERVICE);
+ }
+
+ if (_audioManager == null) {
+ DoLogErr("Could not change audio routing - no audio manager");
+ return -1;
+ }
+
+ int apiLevel = Integer.parseInt(android.os.Build.VERSION.SDK);
+
+ if ((3 == apiLevel) || (4 == apiLevel)) {
+ // 1.5 and 1.6 devices
+ if (loudspeakerOn) {
+ // route audio to back speaker
+ _audioManager.setMode(AudioManager.MODE_NORMAL);
+ } else {
+ // route audio to earpiece
+ _audioManager.setMode(AudioManager.MODE_IN_CALL);
+ }
+ } else {
+ // 2.x devices
+ if ((android.os.Build.BRAND.equals("Samsung") ||
+ android.os.Build.BRAND.equals("samsung")) &&
+ ((5 == apiLevel) || (6 == apiLevel) ||
+ (7 == apiLevel))) {
+ // Samsung 2.0, 2.0.1 and 2.1 devices
+ if (loudspeakerOn) {
+ // route audio to back speaker
+ _audioManager.setMode(AudioManager.MODE_IN_CALL);
+ _audioManager.setSpeakerphoneOn(loudspeakerOn);
+ } else {
+ // route audio to earpiece
+ _audioManager.setSpeakerphoneOn(loudspeakerOn);
+ _audioManager.setMode(AudioManager.MODE_NORMAL);
+ }
+ } else {
+ // Non-Samsung and Samsung 2.2 and up devices
+ _audioManager.setSpeakerphoneOn(loudspeakerOn);
+ }
+ }
+
+ return 0;
+ }
+
+ @SuppressWarnings("unused")
+ private int SetPlayoutVolume(int level) {
+
+ // create audio manager if needed
+ if (_audioManager == null && _context != null) {
+ _audioManager = (AudioManager)
+ _context.getSystemService(Context.AUDIO_SERVICE);
+ }
+
+ int retVal = -1;
+
+ if (_audioManager != null) {
+ _audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
+ level, 0);
+ retVal = 0;
+ }
+
+ return retVal;
+ }
+
+ @SuppressWarnings("unused")
+ private int GetPlayoutVolume() {
+
+ // create audio manager if needed
+ if (_audioManager == null && _context != null) {
+ _audioManager = (AudioManager)
+ _context.getSystemService(Context.AUDIO_SERVICE);
+ }
+
+ int level = -1;
+
+ if (_audioManager != null) {
+ level = _audioManager.getStreamVolume(
+ AudioManager.STREAM_VOICE_CALL);
+ }
+
+ return level;
+ }
+
+ private void SetAudioMode(boolean startCall) {
+ int apiLevel = Integer.parseInt(android.os.Build.VERSION.SDK);
+
+ if (_audioManager == null && _context != null) {
+ _audioManager = (AudioManager)
+ _context.getSystemService(Context.AUDIO_SERVICE);
+ }
+
+ if (_audioManager == null) {
+ DoLogErr("Could not set audio mode - no audio manager");
+ return;
+ }
+
+ // ***IMPORTANT*** When the API level for honeycomb (H) has been
+ // decided,
+ // the condition should be changed to include API level 8 to H-1.
+ if ((android.os.Build.BRAND.equals("Samsung") || android.os.Build.BRAND
+ .equals("samsung")) && (8 == apiLevel)) {
+ // Set Samsung specific VoIP mode for 2.2 devices
+ int mode =
+ (startCall ? 4 /* VoIP mode */
+ : AudioManager.MODE_NORMAL);
+ _audioManager.setMode(mode);
+ if (_audioManager.getMode() != mode) {
+ DoLogErr("Could not set audio mode for Samsung device");
+ }
+ }
+ }
+
+ final String logTag = "WebRTC AD java";
+
+ private void DoLog(String msg) {
+ Log.d(logTag, msg);
+ }
+
+ private void DoLogErr(String msg) {
+ Log.e(logTag, msg);
+ }
+}
diff --git a/trunk/src/modules/audio_device/main/source/audio_device.gypi b/trunk/src/modules/audio_device/main/source/audio_device.gypi
new file mode 100644
index 0000000..2f3f9b1
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device.gypi
@@ -0,0 +1,210 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audio_device',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '.',
+ '../../../interface',
+ '../interface',
+ 'dummy', # dummy audio device
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../../../../',
+ '../../../interface',
+ '../interface',
+ ],
+ },
+ # TODO(xians): Rename files to e.g. *_linux.{ext}, remove sources in conditions section
+ 'sources': [
+ '../interface/audio_device.h',
+ '../interface/audio_device_defines.h',
+ 'audio_device_buffer.cc',
+ 'audio_device_buffer.h',
+ 'audio_device_generic.cc',
+ 'audio_device_generic.h',
+ 'audio_device_utility.cc',
+ 'audio_device_utility.h',
+ 'audio_device_impl.cc',
+ 'audio_device_impl.h',
+ 'audio_device_config.h',
+ 'dummy/audio_device_dummy.cc',
+ 'dummy/audio_device_dummy.h',
+ 'dummy/audio_device_utility_dummy.cc',
+ 'dummy/audio_device_utility_dummy.h',
+ ],
+ 'conditions': [
+ ['OS=="linux"', {
+ 'include_dirs': [
+ 'linux',
+ ],
+ }], # OS==linux
+ ['OS=="mac"', {
+ 'include_dirs': [
+ 'mac',
+ ],
+ }], # OS==mac
+ ['OS=="win"', {
+ 'include_dirs': [
+ 'win',
+ '../../../../../..',
+ ],
+ }],
+ ['OS=="android"', {
+ 'include_dirs': [
+ 'android',
+ ],
+ }], # OS==android
+ ['include_internal_audio_device==0', {
+ 'defines': [
+ 'WEBRTC_DUMMY_AUDIO_BUILD',
+ ],
+ }],
+ ['include_internal_audio_device==1', {
+ 'sources': [
+ 'linux/alsasymboltable_linux.cc',
+ 'linux/alsasymboltable_linux.h',
+ 'linux/audio_device_alsa_linux.cc',
+ 'linux/audio_device_alsa_linux.h',
+ 'linux/audio_device_utility_linux.cc',
+ 'linux/audio_device_utility_linux.h',
+ 'linux/audio_mixer_manager_alsa_linux.cc',
+ 'linux/audio_mixer_manager_alsa_linux.h',
+ 'linux/latebindingsymboltable_linux.cc',
+ 'linux/latebindingsymboltable_linux.h',
+ 'mac/audio_device_mac.cc',
+ 'mac/audio_device_mac.h',
+ 'mac/audio_device_utility_mac.cc',
+ 'mac/audio_device_utility_mac.h',
+ 'mac/audio_mixer_manager_mac.cc',
+ 'mac/audio_mixer_manager_mac.h',
+ 'mac/portaudio/pa_memorybarrier.h',
+ 'mac/portaudio/pa_ringbuffer.c',
+ 'mac/portaudio/pa_ringbuffer.h',
+ 'win/audio_device_core_win.cc',
+ 'win/audio_device_core_win.h',
+ 'win/audio_device_wave_win.cc',
+ 'win/audio_device_wave_win.h',
+ 'win/audio_device_utility_win.cc',
+ 'win/audio_device_utility_win.h',
+ 'win/audio_mixer_manager_win.cc',
+ 'win/audio_mixer_manager_win.h',
+ 'android/audio_device_android_opensles.cc',
+ 'android/audio_device_android_opensles.h',
+ 'android/audio_device_utility_android.cc',
+ 'android/audio_device_utility_android.h',
+ ],
+ 'conditions': [
+ ['OS=="linux"', {
+ 'defines': [
+ 'LINUX_ALSA',
+ ],
+ 'link_settings': {
+ 'libraries': [
+ '-ldl',
+ '-lasound',
+ ],
+ },
+ 'conditions': [
+ ['include_pulse_audio==1', {
+ 'defines': [
+ 'LINUX_PULSE',
+ ],
+ 'sources': [
+ 'linux/audio_device_pulse_linux.cc',
+ 'linux/audio_device_pulse_linux.h',
+ 'linux/audio_mixer_manager_pulse_linux.cc',
+ 'linux/audio_mixer_manager_pulse_linux.h',
+ 'linux/pulseaudiosymboltable_linux.cc',
+ 'linux/pulseaudiosymboltable_linux.h',
+ ],
+ 'link_settings': {
+ 'libraries': [
+ '-lpulse',
+ ],
+ },
+ }],
+ ],
+ }],
+ ['OS=="mac"', {
+ 'link_settings': {
+ 'libraries': [
+ '$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
+ '$(SDKROOT)/System/Library/Frameworks/CoreAudio.framework',
+ ],
+ },
+ }],
+ ['OS=="win"', {
+ 'link_settings': {
+ 'libraries': [
+ # Required for the built-in WASAPI AEC.
+ '-ldmoguids.lib',
+ '-lwmcodecdspuuid.lib',
+ '-lamstrmid.lib',
+ '-lmsdmo.lib',
+ ],
+ },
+ }],
+ ], # conditions
+ }], # include_internal_audio_device==1
+ ], # conditions
+ },
+ ],
+ # Exclude the test targets when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'audio_device_test_api',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_device',
+ 'webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'sources': [
+ '../test/audio_device_test_api.cc',
+ '../test/audio_device_test_defines.h',
+ ],
+ },
+ {
+ 'target_name': 'audio_device_test_func',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_device',
+ 'webrtc_utility',
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ ],
+ 'sources': [
+ '../test/audio_device_test_func.cc',
+ '../test/audio_device_test_defines.h',
+ '../test/func_test_manager.cc',
+ '../test/func_test_manager.h',
+ ],
+ },
+ ],
+ }],
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc b/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc
new file mode 100644
index 0000000..6edec6a
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc
@@ -0,0 +1,652 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "critical_section_wrapper.h"
+#include "audio_device_buffer.h"
+#include "audio_device_utility.h"
+#include "audio_device_config.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include <cassert>
+
+#include "signal_processing_library.h"
+
+namespace webrtc {
+
+// ----------------------------------------------------------------------------
+// ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceBuffer::AudioDeviceBuffer() :
+ _id(-1),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrCbAudioTransport(NULL),
+ _recSampleRate(0),
+ _playSampleRate(0),
+ _recChannels(0),
+ _playChannels(0),
+ _recChannel(AudioDeviceModule::kChannelBoth),
+ _recBytesPerSample(0),
+ _playBytesPerSample(0),
+ _recSamples(0),
+ _recSize(0),
+ _playSamples(0),
+ _playSize(0),
+ _recFile(*FileWrapper::Create()),
+ _playFile(*FileWrapper::Create()),
+ _currentMicLevel(0),
+ _newMicLevel(0),
+ _playDelayMS(0),
+ _recDelayMS(0),
+ _clockDrift(0),
+ _measureDelay(false), // should always be 'false' (EXPERIMENTAL)
+ _pulseList(),
+ _lastPulseTime(AudioDeviceUtility::GetTimeInMS())
+{
+ // valid ID will be set later by SetId, use -1 for now
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
+ memset(_recBuffer, 0, kMaxBufferSizeBytes);
+ memset(_playBuffer, 0, kMaxBufferSizeBytes);
+}
+
+// ----------------------------------------------------------------------------
+// dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceBuffer::~AudioDeviceBuffer()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ _recFile.Flush();
+ _recFile.CloseFile();
+ delete &_recFile;
+
+ _playFile.Flush();
+ _playFile.CloseFile();
+ delete &_playFile;
+
+ _EmptyList();
+ }
+
+ delete &_critSect;
+ delete &_critSectCb;
+}
+
+// ----------------------------------------------------------------------------
+// SetId
+// ----------------------------------------------------------------------------
+
+void AudioDeviceBuffer::SetId(WebRtc_UWord32 id)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id);
+ _id = id;
+}
+
+// ----------------------------------------------------------------------------
+// RegisterAudioCallback
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback)
+{
+
+ CriticalSectionScoped lock(_critSectCb);
+ _ptrCbAudioTransport = audioCallback;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::InitPlayout()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_measureDelay)
+ {
+ _EmptyList();
+ _lastPulseTime = AudioDeviceUtility::GetTimeInMS();
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::InitRecording()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_measureDelay)
+ {
+ _EmptyList();
+ _lastPulseTime = AudioDeviceUtility::GetTimeInMS();
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetRecordingSampleRate(WebRtc_UWord32 fsHz)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz);
+
+ CriticalSectionScoped lock(_critSect);
+ _recSampleRate = fsHz;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetPlayoutSampleRate(WebRtc_UWord32 fsHz)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz);
+
+ CriticalSectionScoped lock(_critSect);
+ _playSampleRate = fsHz;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::RecordingSampleRate() const
+{
+ return _recSampleRate;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::PlayoutSampleRate() const
+{
+ return _playSampleRate;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingChannels
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannels(WebRtc_UWord8 channels)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels);
+
+ CriticalSectionScoped lock(_critSect);
+ _recChannels = channels;
+ _recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutChannels
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetPlayoutChannels(WebRtc_UWord8 channels)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels);
+
+ CriticalSectionScoped lock(_critSect);
+ _playChannels = channels;
+ // 16 bits per sample in mono, 32 bits in stereo
+ _playBytesPerSample = 2*channels;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingChannel
+//
+// Select which channel to use while recording.
+// This API requires that stereo is enabled.
+//
+// Note that, the nChannel parameter in RecordedDataIsAvailable will be
+// set to 2 even for kChannelLeft and kChannelRight. However, nBytesPerSample
+// will be 2 instead of 4 four these cases.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannel(const AudioDeviceModule::ChannelType channel)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recChannels == 1)
+ {
+ return -1;
+ }
+
+ if (channel == AudioDeviceModule::kChannelBoth)
+ {
+ // two bytes per channel
+ _recBytesPerSample = 4;
+ }
+ else
+ {
+ // only utilize one out of two possible channels (left or right)
+ _recBytesPerSample = 2;
+ }
+ _recChannel = channel;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingChannel
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const
+{
+ channel = _recChannel;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingChannels
+// ----------------------------------------------------------------------------
+
+WebRtc_UWord8 AudioDeviceBuffer::RecordingChannels() const
+{
+ return _recChannels;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutChannels
+// ----------------------------------------------------------------------------
+
+WebRtc_UWord8 AudioDeviceBuffer::PlayoutChannels() const
+{
+ return _playChannels;
+}
+
+// ----------------------------------------------------------------------------
+// SetCurrentMicLevel
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetCurrentMicLevel(WebRtc_UWord32 level)
+{
+ _currentMicLevel = level;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// NewMicLevel
+// ----------------------------------------------------------------------------
+
+WebRtc_UWord32 AudioDeviceBuffer::NewMicLevel() const
+{
+ return _newMicLevel;
+}
+
+// ----------------------------------------------------------------------------
+// SetVQEData
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift)
+{
+ if ((playDelayMS + recDelayMS) > 300)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "too long delay (play:%i rec:%i)", playDelayMS, recDelayMS, clockDrift);
+ }
+
+ _playDelayMS = playDelayMS;
+ _recDelayMS = recDelayMS;
+ _clockDrift = clockDrift;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StartInputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize])
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _recFile.Flush();
+ _recFile.CloseFile();
+
+ return (_recFile.OpenFile(fileName, false, false, false));
+}
+
+// ----------------------------------------------------------------------------
+// StopInputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::StopInputFileRecording()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _recFile.Flush();
+ _recFile.CloseFile();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StartOutputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize])
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _playFile.Flush();
+ _playFile.CloseFile();
+
+ return (_playFile.OpenFile(fileName, false, false, false));
+}
+
+// ----------------------------------------------------------------------------
+// StopOutputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::StopOutputFileRecording()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _playFile.Flush();
+ _playFile.CloseFile();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordedBuffer
+//
+// Store recorded audio buffer in local memory ready for the actual
+// "delivery" using a callback.
+//
+// This method can also parse out left or right channel from a stereo
+// input signal, i.e., emulate mono.
+//
+// Examples:
+//
+// 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes
+// 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recBytesPerSample == 0)
+ {
+ assert(false);
+ return -1;
+ }
+
+ _recSamples = nSamples;
+ _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
+ if (_recSize > kMaxBufferSizeBytes)
+ {
+ assert(false);
+ return -1;
+ }
+
+ if (nSamples != _recSamples)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples);
+ return -1;
+ }
+
+ if (_recChannel == AudioDeviceModule::kChannelBoth)
+ {
+ // (default) copy the complete input buffer to the local buffer
+ memcpy(&_recBuffer[0], audioBuffer, _recSize);
+ }
+ else
+ {
+ WebRtc_Word16* ptr16In = (WebRtc_Word16*)audioBuffer;
+ WebRtc_Word16* ptr16Out = (WebRtc_Word16*)&_recBuffer[0];
+
+ if (AudioDeviceModule::kChannelRight == _recChannel)
+ {
+ ptr16In++;
+ }
+
+ // exctract left or right channel from input buffer to the local buffer
+ for (WebRtc_UWord32 i = 0; i < _recSamples; i++)
+ {
+ *ptr16Out = *ptr16In;
+ ptr16Out++;
+ ptr16In++;
+ ptr16In++;
+ }
+ }
+
+ if (_recFile.Open())
+ {
+ // write to binary file in mono or stereo (interleaved)
+ _recFile.Write(&_recBuffer[0], _recSize);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// DeliverRecordedData
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::DeliverRecordedData()
+{
+ CriticalSectionScoped lock(_critSectCb);
+
+ // Ensure that user has initialized all essential members
+ if ((_recSampleRate == 0) ||
+ (_recSamples == 0) ||
+ (_recBytesPerSample == 0) ||
+ (_recChannels == 0))
+ {
+ assert(false);
+ return -1;
+ }
+
+ if (_ptrCbAudioTransport == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to deliver recorded data (AudioTransport does not exist)");
+ return 0;
+ }
+
+ WebRtc_Word32 res(0);
+ WebRtc_UWord32 newMicLevel(0);
+ WebRtc_UWord32 totalDelayMS = _playDelayMS +_recDelayMS;
+
+ if (_measureDelay)
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ memset(&_recBuffer[0], 0, _recSize);
+ WebRtc_UWord32 time = AudioDeviceUtility::GetTimeInMS();
+ if (time - _lastPulseTime > 500)
+ {
+ _pulseList.PushBack(time);
+ _lastPulseTime = time;
+
+ WebRtc_Word16* ptr16 = (WebRtc_Word16*)&_recBuffer[0];
+ *ptr16 = 30000;
+ }
+ }
+
+ res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0],
+ _recSamples,
+ _recBytesPerSample,
+ _recChannels,
+ _recSampleRate,
+ totalDelayMS,
+ _clockDrift,
+ _currentMicLevel,
+ newMicLevel);
+ if (res != -1)
+ {
+ _newMicLevel = newMicLevel;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RequestPlayoutData
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::RequestPlayoutData(WebRtc_UWord32 nSamples)
+{
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ // Ensure that user has initialized all essential members
+ if ((_playBytesPerSample == 0) ||
+ (_playChannels == 0) ||
+ (_playSampleRate == 0))
+ {
+ assert(false);
+ return -1;
+ }
+
+ _playSamples = nSamples;
+ _playSize = _playBytesPerSample * nSamples; // {2,4}*nSamples
+ if (_playSize > kMaxBufferSizeBytes)
+ {
+ assert(false);
+ return -1;
+ }
+
+ if (nSamples != _playSamples)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of samples to be played out (%d)", nSamples);
+ return -1;
+ }
+ }
+
+ WebRtc_UWord32 nSamplesOut(0);
+
+ CriticalSectionScoped lock(_critSectCb);
+
+ if (_ptrCbAudioTransport == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to feed data to playout (AudioTransport does not exist)");
+ return 0;
+ }
+
+ if (_ptrCbAudioTransport)
+ {
+ WebRtc_UWord32 res(0);
+
+ res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples,
+ _playBytesPerSample,
+ _playChannels,
+ _playSampleRate,
+ &_playBuffer[0],
+ nSamplesOut);
+ if (res != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed");
+ }
+
+ // --- Experimental delay-measurement implementation
+ // *** not be used in released code ***
+
+ if (_measureDelay)
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ WebRtc_Word16 maxAbs = WebRtcSpl_MaxAbsValueW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels);
+ if (maxAbs > 1000)
+ {
+ WebRtc_UWord32 nowTime = AudioDeviceUtility::GetTimeInMS();
+
+ if (!_pulseList.Empty())
+ {
+ ListItem* item = _pulseList.First();
+ if (item)
+ {
+ WebRtc_Word16 maxIndex = WebRtcSpl_MaxAbsIndexW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels);
+ WebRtc_UWord32 pulseTime = item->GetUnsignedItem();
+ WebRtc_UWord32 diff = nowTime - pulseTime + (10*maxIndex)/(nSamplesOut*_playChannels);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "diff time in playout delay (%d)", diff);
+ }
+ _pulseList.PopFront();
+ }
+ }
+ }
+ }
+
+ return nSamplesOut;
+}
+
+// ----------------------------------------------------------------------------
+// GetPlayoutData
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceBuffer::GetPlayoutData(WebRtc_Word8* audioBuffer)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playSize > kMaxBufferSizeBytes)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds "
+ "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize);
+ assert(false);
+ return -1;
+ }
+
+ memcpy(audioBuffer, &_playBuffer[0], _playSize);
+
+ if (_playFile.Open())
+ {
+ // write to binary file in mono or stereo (interleaved)
+ _playFile.Write(&_playBuffer[0], _playSize);
+ }
+
+ return _playSamples;
+}
+
+// ----------------------------------------------------------------------------
+// _EmptyList
+// ----------------------------------------------------------------------------
+
+void AudioDeviceBuffer::_EmptyList()
+{
+ while (!_pulseList.Empty())
+ {
+ ListItem* item = _pulseList.First();
+ if (item)
+ {
+ // WebRtc_UWord32 ts = item->GetUnsignedItem();
+ }
+ _pulseList.PopFront();
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_buffer.h b/trunk/src/modules/audio_device/main/source/audio_device_buffer.h
new file mode 100644
index 0000000..e7db275
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_buffer.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
+
+#include "typedefs.h"
+#include "../../../../common_audio/resampler/include/resampler.h"
+#include "file_wrapper.h"
+#include "audio_device.h"
+#include "list_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+const WebRtc_UWord32 kPulsePeriodMs = 1000;
+const WebRtc_UWord32 kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
+
+class AudioDeviceObserver;
+class MediaFile;
+
+class AudioDeviceBuffer
+{
+public:
+ void SetId(WebRtc_UWord32 id);
+ WebRtc_Word32 RegisterAudioCallback(AudioTransport* audioCallback);
+
+ WebRtc_Word32 InitPlayout();
+ WebRtc_Word32 InitRecording();
+
+ WebRtc_Word32 SetRecordingSampleRate(WebRtc_UWord32 fsHz);
+ WebRtc_Word32 SetPlayoutSampleRate(WebRtc_UWord32 fsHz);
+ WebRtc_Word32 RecordingSampleRate() const;
+ WebRtc_Word32 PlayoutSampleRate() const;
+
+ WebRtc_Word32 SetRecordingChannels(WebRtc_UWord8 channels);
+ WebRtc_Word32 SetPlayoutChannels(WebRtc_UWord8 channels);
+ WebRtc_UWord8 RecordingChannels() const;
+ WebRtc_UWord8 PlayoutChannels() const;
+ WebRtc_Word32 SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
+ WebRtc_Word32 RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
+
+ WebRtc_Word32 SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples);
+ WebRtc_Word32 SetCurrentMicLevel(WebRtc_UWord32 level);
+ WebRtc_Word32 SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift);
+ WebRtc_Word32 DeliverRecordedData();
+ WebRtc_UWord32 NewMicLevel() const;
+
+ WebRtc_Word32 RequestPlayoutData(WebRtc_UWord32 nSamples);
+ WebRtc_Word32 GetPlayoutData(WebRtc_Word8* audioBuffer);
+
+ WebRtc_Word32 StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
+ WebRtc_Word32 StopInputFileRecording();
+ WebRtc_Word32 StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
+ WebRtc_Word32 StopOutputFileRecording();
+
+ AudioDeviceBuffer();
+ ~AudioDeviceBuffer();
+
+private:
+ void _EmptyList();
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _critSect;
+ CriticalSectionWrapper& _critSectCb;
+
+ AudioTransport* _ptrCbAudioTransport;
+
+ WebRtc_UWord32 _recSampleRate;
+ WebRtc_UWord32 _playSampleRate;
+
+ WebRtc_UWord8 _recChannels;
+ WebRtc_UWord8 _playChannels;
+
+ // selected recording channel (left/right/both)
+ AudioDeviceModule::ChannelType _recChannel;
+
+ // 2 or 4 depending on mono or stereo
+ WebRtc_UWord8 _recBytesPerSample;
+ WebRtc_UWord8 _playBytesPerSample;
+
+ // 10ms in stereo @ 96kHz
+ WebRtc_Word8 _recBuffer[kMaxBufferSizeBytes];
+
+ // one sample <=> 2 or 4 bytes
+ WebRtc_UWord32 _recSamples;
+ WebRtc_UWord32 _recSize; // in bytes
+
+ // 10ms in stereo @ 96kHz
+ WebRtc_Word8 _playBuffer[kMaxBufferSizeBytes];
+
+ // one sample <=> 2 or 4 bytes
+ WebRtc_UWord32 _playSamples;
+ WebRtc_UWord32 _playSize; // in bytes
+
+ FileWrapper& _recFile;
+ FileWrapper& _playFile;
+
+ WebRtc_UWord32 _currentMicLevel;
+ WebRtc_UWord32 _newMicLevel;
+
+ WebRtc_UWord32 _playDelayMS;
+ WebRtc_UWord32 _recDelayMS;
+
+ WebRtc_Word32 _clockDrift;
+
+ bool _measureDelay;
+ ListWrapper _pulseList;
+ WebRtc_UWord32 _lastPulseTime;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_config.h b/trunk/src/modules/audio_device/main/source/audio_device_config.h
new file mode 100644
index 0000000..23b9d55
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_config.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
+
+// Enumerators
+//
+enum { kAdmMaxIdleTimeProcess = 1000 };
+enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
+
+// Platform specifics
+//
+#if defined(_WIN32)
+#if (_MSC_VER >= 1400)
+// Windows Core Audio is the default audio layer in Windows.
+// Only supported for VS 2005 and higher.
+#define WEBRTC_WINDOWS_CORE_AUDIO_BUILD
+#endif
+#endif
+
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#include <windows.h>
+#include <tchar.h>
+#include <strsafe.h>
+#define DEBUG_PRINT(...) \
+{ \
+ TCHAR msg[256]; \
+ StringCchPrintf(msg, 256, __VA_ARGS__); \
+ OutputDebugString(msg); \
+}
+#else
+#define DEBUG_PRINT(exp) ((void)0)
+#endif
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
+
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_generic.cc b/trunk/src/modules/audio_device/main/source/audio_device_generic.cc
new file mode 100644
index 0000000..7093d80
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_generic.cc
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_generic.h"
+#include "trace.h"
+
+namespace webrtc {
+
+WebRtc_Word32 AudioDeviceGeneric::SetRecordingSampleRate(
+ const WebRtc_UWord32 samplesPerSec)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Set recording sample rate not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceGeneric::SetPlayoutSampleRate(
+ const WebRtc_UWord32 samplesPerSec)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Set playout sample rate not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceGeneric::SetLoudspeakerStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Set loudspeaker status not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceGeneric::GetLoudspeakerStatus(bool& enable) const
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Get loudspeaker status not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceGeneric::ResetAudioDevice()
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Reset audio device not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceGeneric::SoundDeviceControl(unsigned int par1,
+ unsigned int par2, unsigned int par3, unsigned int par4)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Sound device control not supported on this platform");
+ return -1;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Windows AEC not supported on this platform");
+ return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInAECIsEnabled() const
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Windows AEC not supported on this platform");
+ return false;
+}
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_generic.h b/trunk/src/modules/audio_device/main/source/audio_device_generic.h
new file mode 100644
index 0000000..13d5f83
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_generic.h
@@ -0,0 +1,183 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
+
+#include "audio_device.h"
+#include "audio_device_buffer.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric
+{
+ public:
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const = 0;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init() = 0;
+ virtual WebRtc_Word32 Terminate() = 0;
+ virtual bool Initialized() const = 0;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices() = 0;
+ virtual WebRtc_Word16 RecordingDevices() = 0;
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]) = 0;
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]) = 0;
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index) = 0;
+ virtual WebRtc_Word32 SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device) = 0;
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index) = 0;
+ virtual WebRtc_Word32 SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device) = 0;
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 InitPlayout() = 0;
+ virtual bool PlayoutIsInitialized() const = 0;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 InitRecording() = 0;
+ virtual bool RecordingIsInitialized() const = 0;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout() = 0;
+ virtual WebRtc_Word32 StopPlayout() = 0;
+ virtual bool Playing() const = 0;
+ virtual WebRtc_Word32 StartRecording() = 0;
+ virtual WebRtc_Word32 StopRecording() = 0;
+ virtual bool Recording() const = 0;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable) = 0;
+ virtual bool AGC() const = 0;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight) = 0;
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const = 0;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 InitSpeaker() = 0;
+ virtual bool SpeakerIsInitialized() const = 0;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 InitMicrophone() = 0;
+ virtual bool MicrophoneIsInitialized() const = 0;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume) = 0;
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const = 0;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const = 0;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const = 0;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const = 0;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume) = 0;
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const = 0;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(
+ WebRtc_UWord32& maxVolume) const = 0;
+ virtual WebRtc_Word32 MinMicrophoneVolume(
+ WebRtc_UWord32& minVolume) const = 0;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const = 0;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable) = 0;
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const = 0;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable) = 0;
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const = 0;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable) = 0;
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const = 0;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable) = 0;
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const = 0;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available) = 0;
+ virtual WebRtc_Word32 SetStereoRecording(bool enable) = 0;
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const = 0;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS = 0) = 0;
+ virtual WebRtc_Word32 PlayoutBuffer(
+ AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const = 0;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const = 0;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const = 0;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const = 0;
+
+ // Native sample rate controls (samples/sec)
+ virtual WebRtc_Word32 SetRecordingSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+ virtual WebRtc_Word32 SetPlayoutSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+
+ // Speaker audio routing (for mobile devices)
+ virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
+ virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
+
+ // Reset Audio Device (for mobile devices)
+ virtual WebRtc_Word32 ResetAudioDevice();
+
+ // Sound Audio Device control (for WinCE only)
+ virtual WebRtc_Word32 SoundDeviceControl(unsigned int par1 = 0,
+ unsigned int par2 = 0,
+ unsigned int par3 = 0,
+ unsigned int par4 = 0);
+
+ // Windows Core Audio only.
+ virtual int32_t EnableBuiltInAEC(bool enable);
+ virtual bool BuiltInAECIsEnabled() const;
+
+public:
+ virtual bool PlayoutWarning() const = 0;
+ virtual bool PlayoutError() const = 0;
+ virtual bool RecordingWarning() const = 0;
+ virtual bool RecordingError() const = 0;
+ virtual void ClearPlayoutWarning() = 0;
+ virtual void ClearPlayoutError() = 0;
+ virtual void ClearRecordingWarning() = 0;
+ virtual void ClearRecordingError() = 0;
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+
+ virtual ~AudioDeviceGeneric() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
+
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_impl.cc b/trunk/src/modules/audio_device/main/source/audio_device_impl.cc
new file mode 100644
index 0000000..1c4c445
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_impl.cc
@@ -0,0 +1,2062 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_impl.h"
+#include "audio_device_config.h"
+#include "system_wrappers/interface/ref_count.h"
+
+#include <assert.h>
+#include <string.h>
+
+#if defined(_WIN32)
+ #include "audio_device_utility_win.h"
+ #include "audio_device_wave_win.h"
+ #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ #include "audio_device_core_win.h"
+ #endif
+#elif defined(WEBRTC_ANDROID_OPENSLES)
+ #include <stdlib.h>
+ #include "audio_device_utility_android.h"
+ #include "audio_device_android_opensles.h"
+#elif defined(WEBRTC_ANDROID)
+ #include <stdlib.h>
+ #include "audio_device_utility_android.h"
+ #include "audio_device_android_jni.h"
+#elif defined(WEBRTC_LINUX)
+ #include "audio_device_utility_linux.h"
+ #if defined(LINUX_ALSA)
+ #include "audio_device_alsa_linux.h"
+ #endif
+ #if defined(LINUX_PULSE)
+ #include "audio_device_pulse_linux.h"
+ #endif
+#elif defined(MAC_IPHONE)
+ #include "audio_device_utility_iphone.h"
+ #include "audio_device_iphone.h"
+#elif (defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC))
+ #include "audio_device_utility_mac.h"
+ #include "audio_device_mac.h"
+#endif
+#include "audio_device_dummy.h"
+#include "audio_device_utility_dummy.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#define CHECK_INITIALIZED() \
+{ \
+ if (!_initialized) { \
+ return -1; \
+ }; \
+}
+
+#define CHECK_INITIALIZED_BOOL() \
+{ \
+ if (!_initialized) { \
+ return false; \
+ }; \
+}
+
+namespace webrtc
+{
+
+AudioDeviceModule* CreateAudioDeviceModule(
+ WebRtc_Word32 id, AudioDeviceModule::AudioLayer audioLayer) {
+ return AudioDeviceModuleImpl::Create(id, audioLayer);
+}
+
+
+// ============================================================================
+// Static methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceModule::Create()
+// ----------------------------------------------------------------------------
+
+AudioDeviceModule* AudioDeviceModuleImpl::Create(const WebRtc_Word32 id,
+ const AudioLayer audioLayer)
+{
+
+ // Create the generic ref counted (platform independent) implementation.
+ RefCountImpl<AudioDeviceModuleImpl>* audioDevice =
+ new RefCountImpl<AudioDeviceModuleImpl>(id, audioLayer);
+
+ // Ensure that the current platform is supported.
+ if (audioDevice->CheckPlatform() == -1)
+ {
+ delete audioDevice;
+ return NULL;
+ }
+
+ // Create the platform-dependent implementation.
+ if (audioDevice->CreatePlatformSpecificObjects() == -1)
+ {
+ delete audioDevice;
+ return NULL;
+ }
+
+ // Ensure that the generic audio buffer can communicate with the
+ // platform-specific parts.
+ if (audioDevice->AttachAudioBuffer() == -1)
+ {
+ delete audioDevice;
+ return NULL;
+ }
+
+ return audioDevice;
+}
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceModuleImpl - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceModuleImpl::AudioDeviceModuleImpl(const WebRtc_Word32 id, const AudioLayer audioLayer) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
+ _critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrCbAudioDeviceObserver(NULL),
+ _ptrAudioDeviceUtility(NULL),
+ _ptrAudioDevice(NULL),
+ _id(id),
+ _platformAudioLayer(audioLayer),
+ _lastProcessTime(AudioDeviceUtility::GetTimeInMS()),
+ _platformType(kPlatformNotSupported),
+ _initialized(false),
+ _lastError(kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+}
+
+// ----------------------------------------------------------------------------
+// CheckPlatform
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::CheckPlatform()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ // Ensure that the current platform is supported
+ //
+ PlatformType platform(kPlatformNotSupported);
+
+#if defined(_WIN32)
+ platform = kPlatformWin32;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is WIN32");
+#elif defined(WEBRTC_ANDROID)
+ platform = kPlatformAndroid;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is ANDROID");
+#elif defined(WEBRTC_LINUX)
+ platform = kPlatformLinux;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is LINUX");
+#elif (defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC))
+ platform = kPlatformMac;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is MAC");
+#endif
+
+ if (platform == kPlatformNotSupported)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "current platform is not supported => this module will self destruct!");
+ return -1;
+ }
+
+ // Store valid output results
+ //
+ _platformType = platform;
+
+ return 0;
+}
+
+
+// ----------------------------------------------------------------------------
+// CreatePlatformSpecificObjects
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ AudioDeviceGeneric* ptrAudioDevice(NULL);
+ AudioDeviceUtility* ptrAudioDeviceUtility(NULL);
+
+#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+ ptrAudioDevice = new AudioDeviceDummy(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
+
+ if (ptrAudioDevice != NULL)
+ {
+ ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
+ }
+#else
+ const AudioLayer audioLayer(PlatformAudioLayer());
+
+ // Create the *Windows* implementation of the Audio Device
+ //
+#if defined(_WIN32)
+ if ((audioLayer == kWindowsWaveAudio)
+#if !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ // Wave audio is default if Core audio is not supported in this build
+ || (audioLayer == kPlatformDefaultAudio)
+#endif
+ )
+ {
+ // create *Windows Wave Audio* implementation
+ ptrAudioDevice = new AudioDeviceWindowsWave(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Windows Wave APIs will be utilized");
+ }
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ if ((audioLayer == kWindowsCoreAudio) ||
+ (audioLayer == kPlatformDefaultAudio)
+ )
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Windows Core Audio APIs...");
+
+ if (AudioDeviceWindowsCore::CoreAudioIsSupported())
+ {
+ // create *Windows Core Audio* implementation
+ ptrAudioDevice = new AudioDeviceWindowsCore(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Windows Core Audio APIs will be utilized");
+ }
+ else
+ {
+ // create *Windows Wave Audio* implementation
+ ptrAudioDevice = new AudioDeviceWindowsWave(Id());
+ if (ptrAudioDevice != NULL)
+ {
+ // Core Audio was not supported => revert to Windows Wave instead
+ _platformAudioLayer = kWindowsWaveAudio; // modify the state set at construction
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Windows Core Audio is *not* supported => Wave APIs will be utilized instead");
+ }
+ }
+ }
+#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Windows implementation of the Device Utility.
+ // This class is independent of the selected audio layer
+ // for Windows.
+ //
+ ptrAudioDeviceUtility = new AudioDeviceUtilityWindows(Id());
+ }
+#endif // #if defined(_WIN32)
+
+ // Create the *Android OpenSLES* implementation of the Audio Device
+ //
+#if defined(WEBRTC_ANDROID_OPENSLES)
+ if (audioLayer == kPlatformDefaultAudio)
+ {
+ // Create *Android OpenELSE Audio* implementation
+ ptrAudioDevice = new AudioDeviceAndroidOpenSLES(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Android OpenSLES Audio APIs will be utilized");
+ }
+
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Android implementation of the Device Utility.
+ ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
+ }
+ // END #if defined(WEBRTC_ANDROID_OPENSLES)
+
+ // Create the *Android Java* implementation of the Audio Device
+ //
+#elif defined(WEBRTC_ANDROID)
+ if (audioLayer == kPlatformDefaultAudio)
+ {
+ // Create *Android JNI Audio* implementation
+ ptrAudioDevice = new AudioDeviceAndroidJni(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Android JNI Audio APIs will be utilized");
+ }
+
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Android implementation of the Device Utility.
+ ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
+ }
+ // END #if defined(WEBRTC_ANDROID)
+
+ // Create the *Linux* implementation of the Audio Device
+ //
+#elif defined(WEBRTC_LINUX)
+ if ((audioLayer == kLinuxPulseAudio) || (audioLayer == kPlatformDefaultAudio))
+ {
+#if defined(LINUX_PULSE)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Linux PulseAudio APIs...");
+
+ if (AudioDeviceLinuxPulse::PulseAudioIsSupported())
+ {
+ // create *Linux PulseAudio* implementation
+ ptrAudioDevice = new AudioDeviceLinuxPulse(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Linux PulseAudio APIs will be utilized");
+ }
+ else
+ {
+#endif
+#if defined(LINUX_ALSA)
+ // create *Linux ALSA Audio* implementation
+ ptrAudioDevice = new AudioDeviceLinuxALSA(Id());
+ if (ptrAudioDevice != NULL)
+ {
+ // Pulse Audio was not supported => revert to ALSA instead
+ _platformAudioLayer = kLinuxAlsaAudio; // modify the state set at construction
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Linux PulseAudio is *not* supported => ALSA APIs will be utilized instead");
+ }
+#endif
+#if defined(LINUX_PULSE)
+ }
+#endif
+ }
+ else if (audioLayer == kLinuxAlsaAudio)
+ {
+#if defined(LINUX_ALSA)
+ // create *Linux ALSA Audio* implementation
+ ptrAudioDevice = new AudioDeviceLinuxALSA(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Linux ALSA APIs will be utilized");
+#endif
+ }
+
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Linux implementation of the Device Utility.
+ // This class is independent of the selected audio layer
+ // for Linux.
+ //
+ ptrAudioDeviceUtility = new AudioDeviceUtilityLinux(Id());
+ }
+#endif // #if defined(WEBRTC_LINUX)
+
+ // Create the *iPhone* implementation of the Audio Device
+ //
+#if defined(MAC_IPHONE)
+ if (audioLayer == kPlatformDefaultAudio)
+ {
+ // Create *iPhone Audio* implementation
+ ptrAudioDevice = new AudioDeviceIPhone(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "iPhone Audio APIs will be utilized");
+ }
+
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Mac implementation of the Device Utility.
+ ptrAudioDeviceUtility = new AudioDeviceUtilityIPhone(Id());
+ }
+ // END #if defined(MAC_IPHONE)
+
+ // Create the *Mac* implementation of the Audio Device
+ //
+#elif defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
+ if (audioLayer == kPlatformDefaultAudio)
+ {
+ // Create *Mac Audio* implementation
+ ptrAudioDevice = new AudioDeviceMac(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Mac OS X Audio APIs will be utilized");
+ }
+
+ if (ptrAudioDevice != NULL)
+ {
+ // Create the Mac implementation of the Device Utility.
+ ptrAudioDeviceUtility = new AudioDeviceUtilityMac(Id());
+ }
+#endif // #if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
+
+ // Create the *Dummy* implementation of the Audio Device
+ // Available for all platforms
+ //
+ if (audioLayer == kDummyAudio)
+ {
+ // Create *Dummy Audio* implementation
+ assert(!ptrAudioDevice);
+ ptrAudioDevice = new AudioDeviceDummy(Id());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
+
+ if (ptrAudioDevice != NULL)
+ {
+ ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
+ }
+ }
+#endif // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+
+ if (ptrAudioDevice == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "unable to create the platform specific audio device implementation");
+ return -1;
+ }
+
+ if (ptrAudioDeviceUtility == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "unable to create the platform specific audio device utility");
+ return -1;
+ }
+
+ // Store valid output pointers
+ //
+ _ptrAudioDevice = ptrAudioDevice;
+ _ptrAudioDeviceUtility = ptrAudioDeviceUtility;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// AttachAudioBuffer
+//
+// Install "bridge" between the platform implemetation and the generic
+// implementation. The "child" shall set the native sampling rate and the
+// number of channels in this function call.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::AttachAudioBuffer()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ _audioDeviceBuffer.SetId(_id);
+ _ptrAudioDevice->AttachAudioBuffer(&_audioDeviceBuffer);
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// ~AudioDeviceModuleImpl - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceModuleImpl::~AudioDeviceModuleImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+
+ if (_ptrAudioDevice)
+ {
+ delete _ptrAudioDevice;
+ _ptrAudioDevice = NULL;
+ }
+
+ if (_ptrAudioDeviceUtility)
+ {
+ delete _ptrAudioDeviceUtility;
+ _ptrAudioDeviceUtility = NULL;
+ }
+
+ delete &_critSect;
+ delete &_critSectEventCb;
+ delete &_critSectAudioCb;
+}
+
+// ============================================================================
+// Module
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Module::ChangeUniqueId
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Module::TimeUntilNextProcess
+//
+// Returns the number of milliseconds until the module want a worker thread
+// to call Process().
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::TimeUntilNextProcess()
+{
+ WebRtc_UWord32 now = AudioDeviceUtility::GetTimeInMS();
+ WebRtc_Word32 deltaProcess = kAdmMaxIdleTimeProcess - (now - _lastProcessTime);
+ return (deltaProcess);
+}
+
+// ----------------------------------------------------------------------------
+// Module::Process
+//
+// Check for posted error and warning reports. Generate callbacks if
+// new reports exists.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::Process()
+{
+
+ _lastProcessTime = AudioDeviceUtility::GetTimeInMS();
+
+ // kPlayoutWarning
+ if (_ptrAudioDevice->PlayoutWarning())
+ {
+ CriticalSectionScoped lock(_critSectEventCb);
+ if (_ptrCbAudioDeviceObserver)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "=> OnWarningIsReported(kPlayoutWarning)");
+ _ptrCbAudioDeviceObserver->OnWarningIsReported(AudioDeviceObserver::kPlayoutWarning);
+ }
+ _ptrAudioDevice->ClearPlayoutWarning();
+ }
+
+ // kPlayoutError
+ if (_ptrAudioDevice->PlayoutError())
+ {
+ CriticalSectionScoped lock(_critSectEventCb);
+ if (_ptrCbAudioDeviceObserver)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "=> OnErrorIsReported(kPlayoutError)");
+ _ptrCbAudioDeviceObserver->OnErrorIsReported(AudioDeviceObserver::kPlayoutError);
+ }
+ _ptrAudioDevice->ClearPlayoutError();
+ }
+
+ // kRecordingWarning
+ if (_ptrAudioDevice->RecordingWarning())
+ {
+ CriticalSectionScoped lock(_critSectEventCb);
+ if (_ptrCbAudioDeviceObserver)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "=> OnWarningIsReported(kRecordingWarning)");
+ _ptrCbAudioDeviceObserver->OnWarningIsReported(AudioDeviceObserver::kRecordingWarning);
+ }
+ _ptrAudioDevice->ClearRecordingWarning();
+ }
+
+ // kRecordingError
+ if (_ptrAudioDevice->RecordingError())
+ {
+ CriticalSectionScoped lock(_critSectEventCb);
+ if (_ptrCbAudioDeviceObserver)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "=> OnErrorIsReported(kRecordingError)");
+ _ptrCbAudioDeviceObserver->OnErrorIsReported(AudioDeviceObserver::kRecordingError);
+ }
+ _ptrAudioDevice->ClearRecordingError();
+ }
+
+ return 0;
+}
+
+// ============================================================================
+// Public API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const
+{
+
+ AudioLayer activeAudio;
+
+ if (_ptrAudioDevice->ActiveAudioLayer(activeAudio) == -1)
+ {
+ return -1;
+ }
+
+ *audioLayer = activeAudio;
+
+ if (*audioLayer == AudioDeviceModule::kWindowsWaveAudio)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsWaveAudio");
+ }
+ else if (*audioLayer == AudioDeviceModule::kWindowsCoreAudio)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsCoreAudio");
+ }
+ else if (*audioLayer == AudioDeviceModule::kLinuxAlsaAudio)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kLinuxAlsaAudio");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: NOT_SUPPORTED");
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// LastError
+// ----------------------------------------------------------------------------
+
+AudioDeviceModule::ErrorCode AudioDeviceModuleImpl::LastError() const
+{
+ return _lastError;
+}
+
+// ----------------------------------------------------------------------------
+// Init
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::Init()
+{
+
+ if (_initialized)
+ return 0;
+
+ if (!_ptrAudioDeviceUtility)
+ return -1;
+
+ if (!_ptrAudioDevice)
+ return -1;
+
+ _ptrAudioDeviceUtility->Init();
+
+ if (_ptrAudioDevice->Init() == -1)
+ {
+ return -1;
+ }
+
+ _initialized = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Terminate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::Terminate()
+{
+
+ if (!_initialized)
+ return 0;
+
+ if (_ptrAudioDevice->Terminate() == -1)
+ {
+ return -1;
+ }
+
+ _initialized = false;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::Initialized() const
+{
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", _initialized);
+ return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->SpeakerIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// InitSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::InitSpeaker()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->InitSpeaker());
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->MicrophoneIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// InitMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::InitMicrophone()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->InitMicrophone());
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->SpeakerVolumeIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetSpeakerVolume(volume));
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolume(WebRtc_UWord32* volume) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 level(0);
+
+ if (_ptrAudioDevice->SpeakerVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ *volume = level;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: volume=%u", *volume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetWaveOutVolume(volumeLeft, volumeRight));
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::WaveOutVolume(WebRtc_UWord16* volumeLeft, WebRtc_UWord16* volumeRight) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 volLeft(0);
+ WebRtc_UWord16 volRight(0);
+
+ if (_ptrAudioDevice->WaveOutVolume(volLeft, volRight) == -1)
+ {
+ return -1;
+ }
+
+ *volumeLeft = volLeft;
+ *volumeRight = volRight;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "outputs: volumeLeft=%u, volumeRight=%u",
+ *volumeLeft, *volumeRight);
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::SpeakerIsInitialized() const
+{
+ CHECK_INITIALIZED_BOOL();
+
+ bool isInitialized = _ptrAudioDevice->SpeakerIsInitialized();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", isInitialized);
+ return (isInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const
+{
+ CHECK_INITIALIZED_BOOL();
+
+ bool isInitialized = _ptrAudioDevice->MicrophoneIsInitialized();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", isInitialized);
+ return (isInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MaxSpeakerVolume(WebRtc_UWord32* maxVolume) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_ptrAudioDevice->MaxSpeakerVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ *maxVolume = maxVol;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: maxVolume=%d", *maxVolume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MinSpeakerVolume(WebRtc_UWord32* minVolume) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_ptrAudioDevice->MinSpeakerVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ *minVolume = minVol;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: minVolume=%u", *minVolume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolumeStepSize(WebRtc_UWord16* stepSize) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 delta(0);
+
+ if (_ptrAudioDevice->SpeakerVolumeStepSize(delta) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the speaker-volume step size");
+ return -1;
+ }
+
+ *stepSize = delta;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: stepSize=%u", *stepSize);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->SpeakerMuteIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetSpeakerMute(bool enable)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetSpeakerMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ bool muted(false);
+
+ if (_ptrAudioDevice->SpeakerMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ *enabled = muted;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->MicrophoneMuteIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneMute(bool enable)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetMicrophoneMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ bool muted(false);
+
+ if (_ptrAudioDevice->MicrophoneMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ *enabled = muted;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneBoostIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->MicrophoneBoostIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneBoost(bool enable)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetMicrophoneBoost(enable));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneBoost(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ bool onOff(false);
+
+ if (_ptrAudioDevice->MicrophoneBoost(onOff) == -1)
+ {
+ return -1;
+ }
+
+ *enabled = onOff;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->MicrophoneVolumeIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetMicrophoneVolume(volume));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolume(WebRtc_UWord32* volume) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 level(0);
+
+ if (_ptrAudioDevice->MicrophoneVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ *volume = level;
+
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: volume=%u", *volume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StereoRecordingIsAvailable(bool* available) const
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->StereoRecordingIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetStereoRecording(bool enable)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->RecordingIsInitialized())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "recording in stereo is not supported");
+ return -1;
+ }
+
+ if (_ptrAudioDevice->SetStereoRecording(enable) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to enable stereo recording");
+ return -1;
+ }
+
+ WebRtc_Word8 nChannels(1);
+ if (enable)
+ {
+ nChannels = 2;
+ }
+ _audioDeviceBuffer.SetRecordingChannels(nChannels);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StereoRecording(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ bool stereo(false);
+
+ if (_ptrAudioDevice->StereoRecording(stereo) == -1)
+ {
+ return -1;
+ }
+
+ *enabled = stereo;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingChannel
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingChannel(const ChannelType channel)
+{
+ if (channel == kChannelBoth)
+ {
+ }
+ else if (channel == kChannelLeft)
+ {
+ }
+ else
+ {
+ }
+ CHECK_INITIALIZED();
+
+ bool stereo(false);
+
+ if (_ptrAudioDevice->StereoRecording(stereo) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "recording in stereo is not supported");
+ return -1;
+ }
+
+ return (_audioDeviceBuffer.SetRecordingChannel(channel));
+}
+
+// ----------------------------------------------------------------------------
+// RecordingChannel
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RecordingChannel(ChannelType* channel) const
+{
+ CHECK_INITIALIZED();
+
+ ChannelType chType;
+
+ if (_audioDeviceBuffer.RecordingChannel(chType) == -1)
+ {
+ return -1;
+ }
+
+ *channel = chType;
+
+ if (*channel == kChannelBoth)
+ {
+ }
+ else if (*channel == kChannelLeft)
+ {
+ }
+ else
+ {
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->StereoPlayoutIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetStereoPlayout(bool enable)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->PlayoutIsInitialized())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "unable to set stereo mode while playing side is initialized");
+ return -1;
+ }
+
+ if (_ptrAudioDevice->SetStereoPlayout(enable))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "stereo playout is not supported");
+ return -1;
+ }
+
+ WebRtc_Word8 nChannels(1);
+ if (enable)
+ {
+ nChannels = 2;
+ }
+ _audioDeviceBuffer.SetPlayoutChannels(nChannels);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ bool stereo(false);
+
+ if (_ptrAudioDevice->StereoPlayout(stereo) == -1)
+ {
+ return -1;
+ }
+
+ *enabled = stereo;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetAGC
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetAGC(bool enable)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetAGC(enable));
+}
+
+// ----------------------------------------------------------------------------
+// AGC
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::AGC() const
+{
+ CHECK_INITIALIZED_BOOL();
+ return (_ptrAudioDevice->AGC());
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->PlayoutIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RecordingIsAvailable(bool* available)
+{
+ CHECK_INITIALIZED();
+
+ bool isAvailable(0);
+
+ if (_ptrAudioDevice->RecordingIsAvailable(isAvailable) == -1)
+ {
+ return -1;
+ }
+
+ *available = isAvailable;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MaxMicrophoneVolume(WebRtc_UWord32* maxVolume) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_ptrAudioDevice->MaxMicrophoneVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ *maxVolume = maxVol;
+
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: maxVolume=%d", *maxVolume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MinMicrophoneVolume(WebRtc_UWord32* minVolume) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_ptrAudioDevice->MinMicrophoneVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ *minVolume = minVol;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: minVolume=%u", *minVolume);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolumeStepSize(WebRtc_UWord16* stepSize) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 delta(0);
+
+ if (_ptrAudioDevice->MicrophoneVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ *stepSize = delta;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: stepSize=%u", *stepSize);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceModuleImpl::PlayoutDevices()
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 nPlayoutDevices = _ptrAudioDevice->PlayoutDevices();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: #playout devices=%d", nPlayoutDevices);
+ return ((WebRtc_Word16)(nPlayoutDevices));
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetPlayoutDevice(index));
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device)
+{
+ if (device == kDefaultDevice)
+ {
+ }
+ else
+ {
+ }
+ CHECK_INITIALIZED();
+
+ return (_ptrAudioDevice->SetPlayoutDevice(device));
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+ CHECK_INITIALIZED();
+
+ if (name == NULL)
+ {
+ _lastError = kAdmErrArgument;
+ return -1;
+ }
+
+ if (_ptrAudioDevice->PlayoutDeviceName(index, name, guid) == -1)
+ {
+ return -1;
+ }
+
+ if (name != NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: name=%s", name);
+ }
+ if (guid != NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: guid=%s", guid);
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+ CHECK_INITIALIZED();
+
+ if (name == NULL)
+ {
+ _lastError = kAdmErrArgument;
+ return -1;
+ }
+
+ if (_ptrAudioDevice->RecordingDeviceName(index, name, guid) == -1)
+ {
+ return -1;
+ }
+
+ if (name != NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: name=%s", name);
+ }
+ if (guid != NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: guid=%s", guid);
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceModuleImpl::RecordingDevices()
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 nRecordingDevices = _ptrAudioDevice->RecordingDevices();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: #recording devices=%d", nRecordingDevices);
+ return ((WebRtc_Word16)nRecordingDevices);
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingDevice(WebRtc_UWord16 index)
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->SetRecordingDevice(index));
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device)
+{
+ if (device == kDefaultDevice)
+ {
+ }
+ else
+ {
+ }
+ CHECK_INITIALIZED();
+
+ return (_ptrAudioDevice->SetRecordingDevice(device));
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::InitPlayout()
+{
+ CHECK_INITIALIZED();
+ _audioDeviceBuffer.InitPlayout();
+ return (_ptrAudioDevice->InitPlayout());
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::InitRecording()
+{
+ CHECK_INITIALIZED();
+ _audioDeviceBuffer.InitRecording();
+ return (_ptrAudioDevice->InitRecording());
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::PlayoutIsInitialized() const
+{
+ CHECK_INITIALIZED_BOOL();
+ return (_ptrAudioDevice->PlayoutIsInitialized());
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::RecordingIsInitialized() const
+{
+ CHECK_INITIALIZED_BOOL();
+ return (_ptrAudioDevice->RecordingIsInitialized());
+}
+
+// ----------------------------------------------------------------------------
+// StartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StartPlayout()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->StartPlayout());
+}
+
+// ----------------------------------------------------------------------------
+// StopPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StopPlayout()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->StopPlayout());
+}
+
+// ----------------------------------------------------------------------------
+// Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::Playing() const
+{
+ CHECK_INITIALIZED_BOOL();
+ return (_ptrAudioDevice->Playing());
+}
+
+// ----------------------------------------------------------------------------
+// StartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StartRecording()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->StartRecording());
+}
+// ----------------------------------------------------------------------------
+// StopRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StopRecording()
+{
+ CHECK_INITIALIZED();
+ return (_ptrAudioDevice->StopRecording());
+}
+
+// ----------------------------------------------------------------------------
+// Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceModuleImpl::Recording() const
+{
+ CHECK_INITIALIZED_BOOL();
+ return (_ptrAudioDevice->Recording());
+}
+
+// ----------------------------------------------------------------------------
+// RegisterEventObserver
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RegisterEventObserver(AudioDeviceObserver* eventCallback)
+{
+
+ CriticalSectionScoped lock(_critSectEventCb);
+ _ptrCbAudioDeviceObserver = eventCallback;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RegisterAudioCallback
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RegisterAudioCallback(AudioTransport* audioCallback)
+{
+
+ CriticalSectionScoped lock(_critSectAudioCb);
+ _audioDeviceBuffer.RegisterAudioCallback(audioCallback);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StartRawInputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StartRawInputFileRecording(const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize])
+{
+ CHECK_INITIALIZED();
+
+ if (NULL == pcmFileNameUTF8)
+ {
+ return -1;
+ }
+
+ return (_audioDeviceBuffer.StartInputFileRecording(pcmFileNameUTF8));
+}
+
+// ----------------------------------------------------------------------------
+// StopRawInputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StopRawInputFileRecording()
+{
+ CHECK_INITIALIZED();
+
+ return (_audioDeviceBuffer.StopInputFileRecording());
+}
+
+// ----------------------------------------------------------------------------
+// StartRawOutputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StartRawOutputFileRecording(const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize])
+{
+ CHECK_INITIALIZED();
+
+ if (NULL == pcmFileNameUTF8)
+ {
+ return -1;
+ }
+
+ return (_audioDeviceBuffer.StartOutputFileRecording(pcmFileNameUTF8));
+}
+
+// ----------------------------------------------------------------------------
+// StopRawOutputFileRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::StopRawOutputFileRecording()
+{
+ CHECK_INITIALIZED();
+
+ return (_audioDeviceBuffer.StopOutputFileRecording());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutBuffer(const BufferType type, WebRtc_UWord16 sizeMS)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->PlayoutIsInitialized())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "unable to modify the playout buffer while playing side is initialized");
+ return -1;
+ }
+
+ WebRtc_Word32 ret(0);
+
+ if (kFixedBufferSize == type)
+ {
+ if (sizeMS < kAdmMinPlayoutBufferSizeMs || sizeMS > kAdmMaxPlayoutBufferSizeMs)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "size parameter is out of range");
+ return -1;
+ }
+ }
+
+ if ((ret = _ptrAudioDevice->SetPlayoutBuffer(type, sizeMS)) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to set the playout buffer (error: %d)", LastError());
+ }
+
+ return ret;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::PlayoutBuffer(BufferType* type, WebRtc_UWord16* sizeMS) const
+{
+ CHECK_INITIALIZED();
+
+ BufferType bufType;
+ WebRtc_UWord16 size(0);
+
+ if (_ptrAudioDevice->PlayoutBuffer(bufType, size) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the buffer type and size");
+ return -1;
+ }
+
+ *type = bufType;
+ *sizeMS = size;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: type=%u, sizeMS=%u", *type, *sizeMS);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::PlayoutDelay(WebRtc_UWord16* delayMS) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 delay(0);
+
+ if (_ptrAudioDevice->PlayoutDelay(delay) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the playout delay");
+ return -1;
+ }
+
+ *delayMS = delay;
+
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: delayMS=%u", *delayMS);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RecordingDelay(WebRtc_UWord16* delayMS) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 delay(0);
+
+ if (_ptrAudioDevice->RecordingDelay(delay) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the recording delay");
+ return -1;
+ }
+
+ *delayMS = delay;
+
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: delayMS=%u", *delayMS);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// CPULoad
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::CPULoad(WebRtc_UWord16* load) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_UWord16 cpuLoad(0);
+
+ if (_ptrAudioDevice->CPULoad(cpuLoad) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the CPU load");
+ return -1;
+ }
+
+ *load = cpuLoad;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: load=%u", *load);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingSampleRate(const WebRtc_UWord32 samplesPerSec)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->SetRecordingSampleRate(samplesPerSec) != 0)
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::RecordingSampleRate(WebRtc_UWord32* samplesPerSec) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_Word32 sampleRate = _audioDeviceBuffer.RecordingSampleRate();
+
+ if (sampleRate == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the sample rate");
+ return -1;
+ }
+
+ *samplesPerSec = sampleRate;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutSampleRate(const WebRtc_UWord32 samplesPerSec)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->SetPlayoutSampleRate(samplesPerSec) != 0)
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::PlayoutSampleRate(WebRtc_UWord32* samplesPerSec) const
+{
+ CHECK_INITIALIZED();
+
+ WebRtc_Word32 sampleRate = _audioDeviceBuffer.PlayoutSampleRate();
+
+ if (sampleRate == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the sample rate");
+ return -1;
+ }
+
+ *samplesPerSec = sampleRate;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec);
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// ResetAudioDevice
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::ResetAudioDevice()
+{
+ CHECK_INITIALIZED();
+
+
+ if (_ptrAudioDevice->ResetAudioDevice() == -1)
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SetLoudspeakerStatus
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::SetLoudspeakerStatus(bool enable)
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->SetLoudspeakerStatus(enable) != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// GetLoudspeakerStatus
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceModuleImpl::GetLoudspeakerStatus(bool* enabled) const
+{
+ CHECK_INITIALIZED();
+
+ if (_ptrAudioDevice->GetLoudspeakerStatus(*enabled) != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable)
+{
+ CHECK_INITIALIZED();
+
+ return _ptrAudioDevice->EnableBuiltInAEC(enable);
+}
+
+bool AudioDeviceModuleImpl::BuiltInAECIsEnabled() const
+{
+ CHECK_INITIALIZED_BOOL();
+
+ return _ptrAudioDevice->BuiltInAECIsEnabled();
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Platform
+// ----------------------------------------------------------------------------
+
+AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const
+{
+ return _platformType;
+}
+
+// ----------------------------------------------------------------------------
+// PlatformAudioLayer
+// ----------------------------------------------------------------------------
+
+AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer() const
+{
+
+ switch (_platformAudioLayer)
+ {
+ case kPlatformDefaultAudio:
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kPlatformDefaultAudio");
+ break;
+ case kWindowsWaveAudio:
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsWaveAudio");
+ break;
+ case kWindowsCoreAudio:
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsCoreAudio");
+ break;
+ case kLinuxAlsaAudio:
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kLinuxAlsaAudio");
+ break;
+ case kDummyAudio:
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kDummyAudio");
+ break;
+ default:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "output: INVALID");
+ break;
+ }
+
+ return _platformAudioLayer;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_impl.h b/trunk/src/modules/audio_device/main/source/audio_device_impl.h
new file mode 100644
index 0000000..5edec0d
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_impl.h
@@ -0,0 +1,235 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
+
+#include "audio_device.h"
+#include "audio_device_buffer.h"
+
+namespace webrtc
+{
+
+class AudioDeviceGeneric;
+class AudioDeviceUtility;
+class CriticalSectionWrapper;
+
+class AudioDeviceModuleImpl : public AudioDeviceModule
+{
+public:
+ enum PlatformType
+ {
+ kPlatformNotSupported = 0,
+ kPlatformWin32 = 1,
+ kPlatformWinCe = 2,
+ kPlatformLinux = 3,
+ kPlatformMac = 4,
+ kPlatformAndroid = 5
+ };
+
+ WebRtc_Word32 CheckPlatform();
+ WebRtc_Word32 CreatePlatformSpecificObjects();
+ WebRtc_Word32 AttachAudioBuffer();
+
+ AudioDeviceModuleImpl(const WebRtc_Word32 id, const AudioLayer audioLayer);
+ virtual ~AudioDeviceModuleImpl();
+
+public: // RefCountedModule
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+public:
+ // Factory methods (resource allocation/deallocation)
+ static AudioDeviceModule* Create(
+ const WebRtc_Word32 id,
+ const AudioLayer audioLayer = kPlatformDefaultAudio);
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(AudioLayer* audioLayer) const;
+
+ // Error handling
+ virtual ErrorCode LastError() const;
+ virtual WebRtc_Word32 RegisterEventObserver(
+ AudioDeviceObserver* eventCallback);
+
+ // Full-duplex transportation of PCM audio
+ virtual WebRtc_Word32 RegisterAudioCallback(
+ AudioTransport* audioCallback);
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool* available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool* available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16* volumeLeft,
+ WebRtc_UWord16* volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool* available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool* available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool* available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32* volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32* maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32* minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(
+ WebRtc_UWord16* stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool* available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32* volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(
+ WebRtc_UWord32* maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(
+ WebRtc_UWord32* minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(
+ WebRtc_UWord16* stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool* available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool* enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool* available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool* enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool* available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool* enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool* available) const;
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool* enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool* available) const;
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool* enabled) const;
+ virtual WebRtc_Word32 SetRecordingChannel(const ChannelType channel);
+ virtual WebRtc_Word32 RecordingChannel(ChannelType* channel) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(const BufferType type,
+ WebRtc_UWord16 sizeMS = 0);
+ virtual WebRtc_Word32 PlayoutBuffer(BufferType* type,
+ WebRtc_UWord16* sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16* delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16* delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16* load) const;
+
+ // Recording of raw PCM data
+ virtual WebRtc_Word32 StartRawOutputFileRecording(
+ const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]);
+ virtual WebRtc_Word32 StopRawOutputFileRecording();
+ virtual WebRtc_Word32 StartRawInputFileRecording(
+ const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]);
+ virtual WebRtc_Word32 StopRawInputFileRecording();
+
+ // Native sample rate controls (samples/sec)
+ virtual WebRtc_Word32 SetRecordingSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+ virtual WebRtc_Word32 RecordingSampleRate(
+ WebRtc_UWord32* samplesPerSec) const;
+ virtual WebRtc_Word32 SetPlayoutSampleRate(
+ const WebRtc_UWord32 samplesPerSec);
+ virtual WebRtc_Word32 PlayoutSampleRate(
+ WebRtc_UWord32* samplesPerSec) const;
+
+ // Mobile device specific functions
+ virtual WebRtc_Word32 ResetAudioDevice();
+ virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
+ virtual WebRtc_Word32 GetLoudspeakerStatus(bool* enabled) const;
+
+ virtual int32_t EnableBuiltInAEC(bool enable);
+ virtual bool BuiltInAECIsEnabled() const;
+
+public:
+ WebRtc_Word32 Id() {return _id;}
+
+private:
+ PlatformType Platform() const;
+ AudioLayer PlatformAudioLayer() const;
+
+private:
+ CriticalSectionWrapper& _critSect;
+ CriticalSectionWrapper& _critSectEventCb;
+ CriticalSectionWrapper& _critSectAudioCb;
+
+ AudioDeviceObserver* _ptrCbAudioDeviceObserver;
+
+ AudioDeviceUtility* _ptrAudioDeviceUtility;
+ AudioDeviceGeneric* _ptrAudioDevice;
+
+ AudioDeviceBuffer _audioDeviceBuffer;
+
+ WebRtc_Word32 _id;
+ AudioLayer _platformAudioLayer;
+ WebRtc_UWord32 _lastProcessTime;
+ PlatformType _platformType;
+ bool _initialized;
+ mutable ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_utility.cc b/trunk/src/modules/audio_device/main/source/audio_device_utility.cc
new file mode 100644
index 0000000..c256f9e
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_utility.cc
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "audio_device_utility.h"
+
+#if defined(_WIN32)
+
+// ============================================================================
+// Windows
+// ============================================================================
+
+#include <windows.h>
+#include <conio.h>
+#include <ctype.h>
+#include <stdio.h>
+#include <mmsystem.h>
+
+namespace webrtc
+{
+
+void AudioDeviceUtility::Sleep(WebRtc_UWord32 milliseconds)
+{
+ return ::Sleep(milliseconds);
+}
+
+void AudioDeviceUtility::WaitForKey()
+{
+ _getch();
+}
+
+WebRtc_UWord32 AudioDeviceUtility::GetTimeInMS()
+{
+ return timeGetTime();
+}
+
+bool AudioDeviceUtility::StringCompare(const WebRtc_Word8* str1 , const WebRtc_Word8* str2, const WebRtc_UWord32 length)
+{
+ return ((_strnicmp(str1, str2, length) == 0) ? true : false);
+}
+
+} // namespace webrtc
+
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+
+// ============================================================================
+// Linux & Mac
+// ============================================================================
+
+#include <sys/time.h> // gettimeofday
+#include <time.h> // nanosleep, gettimeofday
+#include <string.h> // strncasecmp
+#include <stdio.h> // getchar
+#include <termios.h> // tcgetattr
+
+#include <unistd.h>
+
+namespace webrtc
+{
+
+void AudioDeviceUtility::WaitForKey()
+{
+
+ struct termios oldt, newt;
+
+ tcgetattr( STDIN_FILENO, &oldt );
+
+ // we don't want getchar to echo!
+
+ newt = oldt;
+ newt.c_lflag &= ~( ICANON | ECHO );
+ tcsetattr( STDIN_FILENO, TCSANOW, &newt );
+
+ // catch any newline that's hanging around...
+
+ // you'll have to hit enter twice if you
+
+ // choose enter out of all available keys
+
+ if (getchar() == '\n')
+ {
+ getchar();
+ }
+
+ tcsetattr( STDIN_FILENO, TCSANOW, &oldt );
+}
+
+WebRtc_UWord32 AudioDeviceUtility::GetTimeInMS()
+{
+ struct timeval tv;
+ struct timezone tz;
+ WebRtc_UWord32 val;
+
+ gettimeofday(&tv, &tz);
+ val = (WebRtc_UWord32)(tv.tv_sec*1000 + tv.tv_usec/1000);
+ return val;
+}
+
+void AudioDeviceUtility::Sleep(WebRtc_UWord32 milliseconds)
+{
+ timespec t;
+ t.tv_sec = milliseconds/1000;
+ t.tv_nsec = (milliseconds-(milliseconds/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+}
+
+bool AudioDeviceUtility::StringCompare(const WebRtc_Word8* str1 , const WebRtc_Word8* str2, const WebRtc_UWord32 length)
+{
+ return (strncasecmp(str1, str2, length) == 0)?true: false;
+}
+
+} // namespace webrtc
+
+#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+
+
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_utility.h b/trunk/src/modules/audio_device/main/source/audio_device_utility.h
new file mode 100644
index 0000000..b967950
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/audio_device_utility.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class AudioDeviceUtility
+{
+public:
+ static WebRtc_UWord32 GetTimeInMS();
+ static void Sleep(WebRtc_UWord32 milliseconds);
+ static void WaitForKey();
+ static bool StringCompare(const WebRtc_Word8* str1,
+ const WebRtc_Word8* str2,
+ const WebRtc_UWord32 length);
+ virtual WebRtc_Word32 Init() = 0;
+
+ virtual ~AudioDeviceUtility() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
+
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc
new file mode 100644
index 0000000..2c8d5c7
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc
@@ -0,0 +1,1336 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_dummy.h"
+
+#include <string.h>
+
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+
+// Enable to record playout data
+//#define RECORD_PLAYOUT 1
+
+namespace webrtc {
+
+const WebRtc_UWord32 REC_TIMER_PERIOD_MS = 10;
+const WebRtc_UWord32 PLAY_TIMER_PERIOD_MS = 10;
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceDummy() - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceDummy::AudioDeviceDummy(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _timeEventRec(*EventWrapper::Create()),
+ _timeEventPlay(*EventWrapper::Create()),
+ _recStartEvent(*EventWrapper::Create()),
+ _playStartEvent(*EventWrapper::Create()),
+ _ptrThreadRec(NULL),
+ _ptrThreadPlay(NULL),
+ _recThreadID(0),
+ _playThreadID(0),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _speakerIsInitialized(false),
+ _microphoneIsInitialized(false),
+ _playDataFile(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+
+ memset(_recBuffer, 0, sizeof(_recBuffer));
+ WebRtc_Word16* tmp = (WebRtc_Word16*)_recBuffer;
+
+ // Saw tooth -16000 to 16000, 100 Hz @ fs = 16 kHz
+// for(int i=0; i<160; ++i)
+// {
+// tmp[i] = i*200-16000;
+// }
+
+ // Rough sinus 2 kHz @ fs = 16 kHz
+ for(int i=0; i<20; ++i)
+ {
+ tmp[i*8] = 0;
+ tmp[i*8+1] = -5000;
+ tmp[i*8+2] = -16000;
+ tmp[i*8+3] = -5000;
+ tmp[i*8+4] = 0;
+ tmp[i*8+5] = 5000;
+ tmp[i*8+6] = 16000;
+ tmp[i*8+7] = 5000;
+ }
+
+#ifdef RECORD_PLAYOUT
+ _playDataFile = fopen("webrtc_VoiceEngine_playout.pcm", "wb");
+ if (!_playDataFile)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not open file for writing playout data");
+ }
+#endif
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceDummy() - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceDummy::~AudioDeviceDummy()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ _ptrAudioBuffer = NULL;
+
+ delete &_recStartEvent;
+ delete &_playStartEvent;
+ delete &_timeEventRec;
+ delete &_timeEventPlay;
+ delete &_critSect;
+
+ if (_playDataFile)
+ {
+ fclose(_playDataFile);
+ }
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AttachAudioBuffer
+// ----------------------------------------------------------------------------
+
+void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // Inform the AudioBuffer about default settings for this implementation.
+ _ptrAudioBuffer->SetRecordingSampleRate(16000);
+ _ptrAudioBuffer->SetPlayoutSampleRate(16000);
+ _ptrAudioBuffer->SetRecordingChannels(1);
+ _ptrAudioBuffer->SetPlayoutChannels(1);
+}
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kDummyAudio;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Init
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ const bool periodic(true);
+ unsigned int threadID(0);
+ char threadName[64] = {0};
+
+ // RECORDING
+ strncpy(threadName, "webrtc_audio_module_rec_thread", 63);
+ _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this, kRealtimePriority, threadName);
+ if (_ptrThreadRec == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to create the rec audio thread");
+ return -1;
+ }
+
+ if (!_ptrThreadRec->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the rec audio thread");
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ return -1;
+ }
+ _recThreadID = threadID;
+
+ if (!_timeEventRec.StartTimer(periodic, REC_TIMER_PERIOD_MS))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the rec timer event");
+ if (_ptrThreadRec->Stop())
+ {
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " unable to stop the activated rec thread");
+ }
+ return -1;
+ }
+
+ // PLAYOUT
+ strncpy(threadName, "webrtc_audio_module_play_thread", 63);
+ _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this, kRealtimePriority, threadName);
+ if (_ptrThreadPlay == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to create the play audio thread");
+ return -1;
+ }
+
+ threadID = 0;
+ if (!_ptrThreadPlay->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the play audio thread");
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ return -1;
+ }
+ _playThreadID = threadID;
+
+ if (!_timeEventPlay.StartTimer(periodic, PLAY_TIMER_PERIOD_MS))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the play timer event");
+ if (_ptrThreadPlay->Stop())
+ {
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " unable to stop the activated play thread");
+ }
+ return -1;
+ }
+
+ _initialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Terminate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::Terminate()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ // RECORDING
+ if (_ptrThreadRec)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+ _timeEventRec.Set();
+
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " failed to close down the rec audio thread");
+ }
+
+ _critSect.Enter();
+ }
+
+ _timeEventRec.StopTimer();
+
+ // PLAYOUT
+ if (_ptrThreadPlay)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+ _timeEventPlay.Set();
+
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " failed to close down the play audio thread");
+ }
+
+ _critSect.Enter();
+ }
+
+ _timeEventPlay.StopTimer();
+
+ _initialized = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::Initialized() const
+{
+ return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ _speakerIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ _microphoneIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::SpeakerIsInitialized() const
+{
+
+ return (_speakerIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::MicrophoneIsInitialized() const
+{
+
+ return (_microphoneIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetSpeakerMute(bool enable)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SpeakerMute(bool& enabled) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetMicrophoneMute(bool enable)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneMute(bool& enabled) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ available = false;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetMicrophoneBoost(bool enable)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneBoost(bool& enabled) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StereoRecordingIsAvailable(bool& available)
+{
+
+ available = false;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetStereoRecording(bool enable)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (enable)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StereoRecording(bool& enabled) const
+{
+
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetStereoPlayout(bool enable)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (enable)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StereoPlayout(bool& enabled) const
+{
+
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetAGC
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetAGC(bool enable)
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// AGC
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::AGC() const
+{
+ // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ return false;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ available = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceDummy::SetMicrophoneVolume(volume=%u)", volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+ // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceDummy::PlayoutDevices()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ return 1;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (index != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ if (index != 0)
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ if (index != 0)
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceDummy::RecordingDevices()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ return 1;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (index != 0 )
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::PlayoutIsAvailable(bool& available)
+{
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::RecordingIsAvailable(bool& available)
+{
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " InitSpeaker() failed");
+ }
+
+ _playIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " InitMicrophone() failed");
+ }
+
+ _recIsInitialized = true;
+
+ return 0;
+
+}
+
+// ----------------------------------------------------------------------------
+// StartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StartRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ _recording = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StopRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ _recIsInitialized = false;
+ _recording = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::Recording() const
+{
+ return (_recording);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::PlayoutIsInitialized() const
+{
+
+ return (_playIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// StartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StartPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ _playing = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::StopPlayout()
+{
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+ delayMS = 0;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+ delayMS = 0;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::Playing() const
+{
+ return (_playing);
+}
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Just ignore
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+
+ type = AudioDeviceModule::kAdaptiveBufferSize;
+ sizeMS = 0;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// CPULoad
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceDummy::CPULoad(WebRtc_UWord16& load) const
+{
+
+ load = 0;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::PlayoutWarning() const
+{
+ return false;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::PlayoutError() const
+{
+ return false;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::RecordingWarning() const
+{
+ return false;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::RecordingError() const
+{
+ return false;
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceDummy::ClearPlayoutWarning()
+{
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceDummy::ClearPlayoutError()
+{
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceDummy::ClearRecordingWarning()
+{
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceDummy::ClearRecordingError()
+{
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// PlayThreadFunc
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::PlayThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceDummy*>(pThis)->PlayThreadProcess());
+}
+
+// ----------------------------------------------------------------------------
+// RecThreadFunc
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::RecThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceDummy*>(pThis)->RecThreadProcess());
+}
+
+// ----------------------------------------------------------------------------
+// PlayThreadProcess
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::PlayThreadProcess()
+{
+ switch (_timeEventPlay.Wait(1000))
+ {
+ case kEventSignaled:
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "EventWrapper::Wait() failed => restarting timer");
+ _timeEventPlay.StopTimer();
+ _timeEventPlay.StartTimer(true, PLAY_TIMER_PERIOD_MS);
+ return true;
+ case kEventTimeout:
+ return true;
+ }
+
+ Lock();
+
+ if(_playing)
+ {
+ WebRtc_Word8 playBuffer[2*160];
+
+ UnLock();
+ WebRtc_Word32 nSamples = (WebRtc_Word32)_ptrAudioBuffer->RequestPlayoutData(160);
+ Lock();
+
+ if (!_playing)
+ {
+ UnLock();
+ return true;
+ }
+
+ nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ if (nSamples != 160)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " invalid number of output samples(%d)", nSamples);
+ }
+
+ if (_playDataFile)
+ {
+ int wr = fwrite(playBuffer, 2, 160, _playDataFile);
+ if (wr != 160)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Could not write playout data to file (%d) ferror = %d",
+ wr, ferror(_playDataFile));
+ }
+ }
+ }
+
+ UnLock();
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// RecThreadProcess
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceDummy::RecThreadProcess()
+{
+ switch (_timeEventRec.Wait(1000))
+ {
+ case kEventSignaled:
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "EventWrapper::Wait() failed => restarting timer");
+ _timeEventRec.StopTimer();
+ _timeEventRec.StartTimer(true, REC_TIMER_PERIOD_MS);
+ return true;
+ case kEventTimeout:
+ return true;
+ }
+
+ Lock();
+
+ if (_recording)
+ {
+ // store the recorded buffer
+ _ptrAudioBuffer->SetRecordedBuffer(_recBuffer, 160);
+
+ // store vqe delay values
+ _ptrAudioBuffer->SetVQEData(0, 0, 0);
+
+ // deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ }
+ else
+ {
+ UnLock();
+ }
+
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h
new file mode 100644
index 0000000..5a979d2
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h
@@ -0,0 +1,178 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
+
+#include <stdio.h>
+
+#include "audio_device_generic.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+class EventWrapper;
+class ThreadWrapper;
+
+class AudioDeviceDummy : public AudioDeviceGeneric
+{
+public:
+ AudioDeviceDummy(const WebRtc_Word32 id);
+ ~AudioDeviceDummy();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private:
+ void Lock() { _critSect.Enter(); };
+ void UnLock() { _critSect.Leave(); };
+
+ static bool RecThreadFunc(void*);
+ static bool PlayThreadFunc(void*);
+ bool RecThreadProcess();
+ bool PlayThreadProcess();
+
+ AudioDeviceBuffer* _ptrAudioBuffer;
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+
+ EventWrapper& _timeEventRec;
+ EventWrapper& _timeEventPlay;
+ EventWrapper& _recStartEvent;
+ EventWrapper& _playStartEvent;
+
+ ThreadWrapper* _ptrThreadRec;
+ ThreadWrapper* _ptrThreadPlay;
+ WebRtc_UWord32 _recThreadID;
+ WebRtc_UWord32 _playThreadID;
+
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _speakerIsInitialized;
+ bool _microphoneIsInitialized;
+
+ WebRtc_Word8 _recBuffer[2*160];
+
+ FILE* _playDataFile;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc
new file mode 100644
index 0000000..44f2c25
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility_dummy.h"
+#include "audio_device_config.h" // DEBUG_PRINT()
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+AudioDeviceUtilityDummy::AudioDeviceUtilityDummy(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _lastError(AudioDeviceModule::kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+}
+
+AudioDeviceUtilityDummy::~AudioDeviceUtilityDummy()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ // free stuff here...
+ }
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+
+WebRtc_Word32 AudioDeviceUtilityDummy::Init()
+{
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " OS info: %s", "Dummy");
+
+ return 0;
+}
+
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h
new file mode 100644
index 0000000..601c448
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
+
+#include "audio_device_utility.h"
+#include "audio_device.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class AudioDeviceUtilityDummy: public AudioDeviceUtility
+{
+public:
+ AudioDeviceUtilityDummy(const WebRtc_Word32 id);
+ ~AudioDeviceUtilityDummy();
+
+ virtual WebRtc_Word32 Init();
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ AudioDeviceModule::ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc
new file mode 100644
index 0000000..1b1707c
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc
@@ -0,0 +1,39 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "alsasymboltable_linux.h"
+
+namespace webrtc_adm_linux_alsa {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
+#define X(sym) \
+ LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
+
+} // namespace webrtc_adm_linux_alsa
diff --git a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h
new file mode 100644
index 0000000..d25bbd7
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h
@@ -0,0 +1,147 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
+
+#include "latebindingsymboltable_linux.h"
+
+namespace webrtc_adm_linux_alsa {
+
+// The ALSA symbols we need, as an X-Macro list.
+// This list must contain precisely every libasound function that is used in
+// alsasoundsystem.cc.
+#define ALSA_SYMBOLS_LIST \
+ X(snd_device_name_free_hint) \
+ X(snd_device_name_get_hint) \
+ X(snd_device_name_hint) \
+ X(snd_pcm_avail_update) \
+ X(snd_pcm_close) \
+ X(snd_pcm_delay) \
+ X(snd_pcm_drop) \
+ X(snd_pcm_open) \
+ X(snd_pcm_prepare) \
+ X(snd_pcm_readi) \
+ X(snd_pcm_recover) \
+ X(snd_pcm_resume) \
+ X(snd_pcm_reset) \
+ X(snd_pcm_state) \
+ X(snd_pcm_set_params) \
+ X(snd_pcm_get_params) \
+ X(snd_pcm_start) \
+ X(snd_pcm_stream) \
+ X(snd_pcm_frames_to_bytes) \
+ X(snd_pcm_bytes_to_frames) \
+ X(snd_pcm_wait) \
+ X(snd_pcm_writei) \
+ X(snd_pcm_info_get_class) \
+ X(snd_pcm_info_get_subdevices_avail) \
+ X(snd_pcm_info_get_subdevice_name) \
+ X(snd_pcm_info_set_subdevice) \
+ X(snd_pcm_info_get_id) \
+ X(snd_pcm_info_set_device) \
+ X(snd_pcm_info_set_stream) \
+ X(snd_pcm_info_get_name) \
+ X(snd_pcm_info_get_subdevices_count) \
+ X(snd_pcm_info_sizeof) \
+ X(snd_pcm_hw_params) \
+ X(snd_pcm_hw_params_malloc) \
+ X(snd_pcm_hw_params_free) \
+ X(snd_pcm_hw_params_any) \
+ X(snd_pcm_hw_params_set_access) \
+ X(snd_pcm_hw_params_set_format) \
+ X(snd_pcm_hw_params_set_channels) \
+ X(snd_pcm_hw_params_set_rate_near) \
+ X(snd_pcm_hw_params_set_buffer_size_near) \
+ X(snd_card_next) \
+ X(snd_card_get_name) \
+ X(snd_config_update) \
+ X(snd_config_copy) \
+ X(snd_config_get_id) \
+ X(snd_ctl_open) \
+ X(snd_ctl_close) \
+ X(snd_ctl_card_info) \
+ X(snd_ctl_card_info_sizeof) \
+ X(snd_ctl_card_info_get_id) \
+ X(snd_ctl_card_info_get_name) \
+ X(snd_ctl_pcm_next_device) \
+ X(snd_ctl_pcm_info) \
+ X(snd_mixer_load) \
+ X(snd_mixer_free) \
+ X(snd_mixer_detach) \
+ X(snd_mixer_close) \
+ X(snd_mixer_open) \
+ X(snd_mixer_attach) \
+ X(snd_mixer_first_elem) \
+ X(snd_mixer_elem_next) \
+ X(snd_mixer_selem_get_name) \
+ X(snd_mixer_selem_is_active) \
+ X(snd_mixer_selem_register) \
+ X(snd_mixer_selem_set_playback_volume_all) \
+ X(snd_mixer_selem_get_playback_volume) \
+ X(snd_mixer_selem_has_playback_volume) \
+ X(snd_mixer_selem_get_playback_volume_range) \
+ X(snd_mixer_selem_has_playback_switch) \
+ X(snd_mixer_selem_get_playback_switch) \
+ X(snd_mixer_selem_set_playback_switch_all) \
+ X(snd_mixer_selem_has_capture_switch) \
+ X(snd_mixer_selem_get_capture_switch) \
+ X(snd_mixer_selem_set_capture_switch_all) \
+ X(snd_mixer_selem_has_capture_volume) \
+ X(snd_mixer_selem_set_capture_volume_all) \
+ X(snd_mixer_selem_get_capture_volume) \
+ X(snd_mixer_selem_get_capture_volume_range) \
+ X(snd_dlopen) \
+ X(snd_dlclose) \
+ X(snd_config) \
+ X(snd_config_search) \
+ X(snd_config_get_string) \
+ X(snd_config_search_definition) \
+ X(snd_config_get_type) \
+ X(snd_config_delete) \
+ X(snd_config_iterator_entry) \
+ X(snd_config_iterator_first) \
+ X(snd_config_iterator_next) \
+ X(snd_config_iterator_end) \
+ X(snd_config_delete_compound_members) \
+ X(snd_config_get_integer) \
+ X(snd_config_get_bool) \
+ X(snd_dlsym) \
+ X(snd_strerror) \
+ X(snd_lib_error) \
+ X(snd_lib_error_set_handler)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
+#define X(sym) \
+ LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
+
+} // namespace webrtc_adm_linux_alsa
+
+#endif // WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc
new file mode 100644
index 0000000..f82d502
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc
@@ -0,0 +1,2320 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "audio_device_utility.h"
+#include "audio_device_alsa_linux.h"
+#include "audio_device_config.h"
+
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+
+
+webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym) \
+ LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
+
+// Redefine these here to be able to do late-binding
+#undef snd_ctl_card_info_alloca
+#define snd_ctl_card_info_alloca(ptr) \
+ do { *ptr = (snd_ctl_card_info_t *) \
+ __builtin_alloca (LATE(snd_ctl_card_info_sizeof)()); \
+ memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); } while (0)
+
+#undef snd_pcm_info_alloca
+#define snd_pcm_info_alloca(pInfo) \
+ do { *pInfo = (snd_pcm_info_t *) \
+ __builtin_alloca (LATE(snd_pcm_info_sizeof)()); \
+ memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)()); } while (0)
+
+// snd_lib_error_handler_t
+void WebrtcAlsaErrorHandler(const char *file,
+ int line,
+ const char *function,
+ int err,
+ const char *fmt,...){};
+
+namespace webrtc
+{
+static const unsigned int ALSA_PLAYOUT_FREQ = 48000;
+static const unsigned int ALSA_PLAYOUT_CH = 2;
+static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us
+static const unsigned int ALSA_CAPTURE_FREQ = 48000;
+static const unsigned int ALSA_CAPTURE_CH = 2;
+static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us
+static const unsigned int ALSA_PLAYOUT_WAIT_TIMEOUT = 5; // in ms
+static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
+
+#define FUNC_GET_NUM_OF_DEVICE 0
+#define FUNC_GET_DEVICE_NAME 1
+#define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2
+
+AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrThreadRec(NULL),
+ _ptrThreadPlay(NULL),
+ _recThreadID(0),
+ _playThreadID(0),
+ _id(id),
+ _mixerManager(id),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _inputDeviceIsSpecified(false),
+ _outputDeviceIsSpecified(false),
+ _handleRecord(NULL),
+ _handlePlayout(NULL),
+ _recordingBuffersizeInFrame(0),
+ _recordingPeriodSizeInFrame(0),
+ _playoutBufferSizeInFrame(0),
+ _playoutPeriodSizeInFrame(0),
+ _recordingBufferSizeIn10MS(0),
+ _playoutBufferSizeIn10MS(0),
+ _recordingFramesIn10MS(0),
+ _playoutFramesIn10MS(0),
+ _recordingFreq(ALSA_CAPTURE_FREQ),
+ _playoutFreq(ALSA_PLAYOUT_FREQ),
+ _recChannels(ALSA_CAPTURE_CH),
+ _playChannels(ALSA_PLAYOUT_CH),
+ _recordingBuffer(NULL),
+ _playoutBuffer(NULL),
+ _recordingFramesLeft(0),
+ _playoutFramesLeft(0),
+ _playbackBufferSize(0),
+ _playBufType(AudioDeviceModule::kFixedBufferSize),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _AGC(false),
+ _recordingDelay(0),
+ _playoutDelay(0),
+ _writeErrors(0),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _playBufDelay(80),
+ _playBufDelayFixed(80)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceLinuxALSA - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ // Clean up the recording buffer and playout buffer.
+ if (_recordingBuffer)
+ {
+ delete [] _recordingBuffer;
+ _recordingBuffer = NULL;
+ }
+ if (_playoutBuffer)
+ {
+ delete [] _playoutBuffer;
+ _playoutBuffer = NULL;
+ }
+ delete &_critSect;
+}
+
+void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // Inform the AudioBuffer about default settings for this implementation.
+ // Set all values to zero here since the actual settings will be done by
+ // InitPlayout and InitRecording later.
+ _ptrAudioBuffer->SetRecordingSampleRate(0);
+ _ptrAudioBuffer->SetPlayoutSampleRate(0);
+ _ptrAudioBuffer->SetRecordingChannels(0);
+ _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Load libasound
+ if (!AlsaSymbolTable.Load())
+ {
+ // Alsa is not installed on
+ // this system
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to load symbol table");
+ return -1;
+ }
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ _initialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::Terminate()
+{
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ _mixerManager.Close();
+
+ // RECORDING
+ if (_ptrThreadRec)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the rec audio thread");
+ }
+
+ _critSect.Enter();
+ }
+
+ // PLAYOUT
+ if (_ptrThreadPlay)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the play audio thread");
+ }
+
+ _critSect.Enter();
+ }
+
+ _initialized = false;
+ _outputDeviceIsSpecified = false;
+ _inputDeviceIsSpecified = false;
+
+ return 0;
+}
+
+bool AudioDeviceLinuxALSA::Initialized() const
+{
+ return (_initialized);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a valid speaker
+ // exists
+ available = true;
+
+ // Close the initialized output mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ char devName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+ return _mixerManager.OpenSpeaker(devName);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a valid
+ // microphone exists
+ available = true;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ char devName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+ return _mixerManager.OpenMicrophone(devName);
+}
+
+bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const
+{
+ return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const
+{
+ return (_mixerManager.MicrophoneIsInitialized());
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a volume control
+ // exists
+ available = true;
+
+ // Close the initialized output mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.SpeakerVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ volume = level;
+
+ return 0;
+}
+
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::WaveOutVolume(
+ WebRtc_UWord16& /*volumeLeft*/,
+ WebRtc_UWord16& /*volumeRight*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MaxSpeakerVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MinSpeakerVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinSpeakerVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected speaker has a mute control
+ _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerMute(bool enable)
+{
+ return (_mixerManager.SetSpeakerMute(enable));
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.SpeakerMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a mute control
+ //
+ _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable)
+{
+ return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.MicrophoneMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Enumerate all avaliable microphone and make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a boost control
+ _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneBoost(bool enable)
+{
+
+ return (_mixerManager.SetMicrophoneBoost(enable));
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoost(bool& enabled) const
+{
+
+ bool onOff(0);
+
+ if (_mixerManager.MicrophoneBoost(onOff) == -1)
+ {
+ return -1;
+ }
+
+ enabled = onOff;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ // If we already have initialized in stereo it's obviously available
+ if (_recIsInitialized && (2 == _recChannels))
+ {
+ available = true;
+ return 0;
+ }
+
+ // Save rec states and the number of rec channels
+ bool recIsInitialized = _recIsInitialized;
+ bool recording = _recording;
+ int recChannels = _recChannels;
+
+ available = false;
+
+ // Stop/uninitialize recording if initialized (and possibly started)
+ if (_recIsInitialized)
+ {
+ StopRecording();
+ }
+
+ // Try init in stereo;
+ _recChannels = 2;
+ if (InitRecording() == 0)
+ {
+ available = true;
+ }
+
+ // Stop/uninitialize recording
+ StopRecording();
+
+ // Recover previous states
+ _recChannels = recChannels;
+ if (recIsInitialized)
+ {
+ InitRecording();
+ }
+ if (recording)
+ {
+ StartRecording();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoRecording(bool enable)
+{
+
+ if (enable)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const
+{
+
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ // If we already have initialized in stereo it's obviously available
+ if (_playIsInitialized && (2 == _playChannels))
+ {
+ available = true;
+ return 0;
+ }
+
+ // Save rec states and the number of rec channels
+ bool playIsInitialized = _playIsInitialized;
+ bool playing = _playing;
+ int playChannels = _playChannels;
+
+ available = false;
+
+ // Stop/uninitialize recording if initialized (and possibly started)
+ if (_playIsInitialized)
+ {
+ StopPlayout();
+ }
+
+ // Try init in stereo;
+ _playChannels = 2;
+ if (InitPlayout() == 0)
+ {
+ available = true;
+ }
+
+ // Stop/uninitialize recording
+ StopPlayout();
+
+ // Recover previous states
+ _playChannels = playChannels;
+ if (playIsInitialized)
+ {
+ InitPlayout();
+ }
+ if (playing)
+ {
+ StartPlayout();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoPlayout(bool enable)
+{
+
+ if (enable)
+ _playChannels = 2;
+ else
+ _playChannels = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const
+{
+
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetAGC(bool enable)
+{
+
+ _AGC = enable;
+
+ return 0;
+}
+
+bool AudioDeviceLinuxALSA::AGC() const
+{
+
+ return _AGC;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a volume control
+ // exists
+ available = true;
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetMicrophoneVolume(volume));
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.MicrophoneVolume(level) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to retrive current microphone level");
+ return -1;
+ }
+
+ volume = level;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MaxMicrophoneVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MinMicrophoneVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceLinuxALSA::PlayoutDevices()
+{
+
+ return (WebRtc_Word16)GetDevicesInfo(0, true);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 nDevices = GetDevicesInfo(0, true);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable audio output devices is %u", nDevices);
+
+ if (index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ _outputDeviceIndex = index;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices(PlayoutDevices());
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices(RecordingDevices());
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
+}
+
+WebRtc_Word16 AudioDeviceLinuxALSA::RecordingDevices()
+{
+
+ return (WebRtc_Word16)GetDevicesInfo(0, false);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 nDevices = GetDevicesInfo(0, false);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable audio input devices is %u", nDevices);
+
+ if (index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ _inputDeviceIndex = index;
+ _inputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side with mono
+ // Assumes that user set num channels after calling this function
+ _playChannels = 1;
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+ else
+ {
+ // It may be possible to play out in stereo
+ res = StereoPlayoutIsAvailable(available);
+ if (available)
+ {
+ // Then set channels to 2 so InitPlayout doesn't fail
+ _playChannels = 2;
+ }
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the recording side with mono
+ // Assumes that user set num channels after calling this function
+ _recChannels = 1;
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+ else
+ {
+ // It may be possible to record in stereo
+ res = StereoRecordingIsAvailable(available);
+ if (available)
+ {
+ // Then set channels to 2 so InitPlayout doesn't fail
+ _recChannels = 2;
+ }
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::InitPlayout()
+{
+
+ int errVal = 0;
+
+ CriticalSectionScoped lock(_critSect);
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (!_outputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitSpeaker() failed");
+ }
+
+ // Start by closing any existing wave-output devices
+ //
+ if (_handlePlayout != NULL)
+ {
+ LATE(snd_pcm_close)(_handlePlayout);
+ _handlePlayout = NULL;
+ _playIsInitialized = false;
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error closing current playout sound device, error:"
+ " %s", LATE(snd_strerror)(errVal));
+ }
+ }
+
+ // Open PCM device for playout
+ char deviceName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
+ kAdmMaxDeviceNameSize);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " InitPlayout open (%s)", deviceName);
+
+ errVal = LATE(snd_pcm_open)
+ (&_handlePlayout,
+ deviceName,
+ SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK);
+
+ if (errVal == -EBUSY) // Device busy - try some more!
+ {
+ for (int i=0; i < 5; i++)
+ {
+ sleep(1);
+ errVal = LATE(snd_pcm_open)
+ (&_handlePlayout,
+ deviceName,
+ SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK);
+ if (errVal == 0)
+ {
+ break;
+ }
+ }
+ }
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " unable to open playback device: %s (%d)",
+ LATE(snd_strerror)(errVal),
+ errVal);
+ _handlePlayout = NULL;
+ return -1;
+ }
+
+ _playoutFramesIn10MS = _playoutFreq/100;
+ if ((errVal = LATE(snd_pcm_set_params)( _handlePlayout,
+#if defined(WEBRTC_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE,
+#else
+ SND_PCM_FORMAT_S16_LE, //format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, //access
+ _playChannels, //channels
+ _playoutFreq, //rate
+ 1, //soft_resample
+ ALSA_PLAYOUT_LATENCY //40*1000 //latency required overall latency in us
+ )) < 0)
+ { /* 0.5sec */
+ _playoutFramesIn10MS = 0;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " unable to set playback device: %s (%d)",
+ LATE(snd_strerror)(errVal),
+ errVal);
+ ErrorRecovery(errVal, _handlePlayout);
+ errVal = LATE(snd_pcm_close)(_handlePlayout);
+ _handlePlayout = NULL;
+ return -1;
+ }
+
+ errVal = LATE(snd_pcm_get_params)(_handlePlayout,
+ &_playoutBufferSizeInFrame, &_playoutPeriodSizeInFrame);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_pcm_get_params %s",
+ LATE(snd_strerror)(errVal),
+ errVal);
+ _playoutBufferSizeInFrame = 0;
+ _playoutPeriodSizeInFrame = 0;
+ }
+ else {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " playout snd_pcm_get_params "
+ "buffer_size:%d period_size :%d",
+ _playoutBufferSizeInFrame, _playoutPeriodSizeInFrame);
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
+ _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
+ }
+
+ // Set play buffer size
+ _playoutBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
+ _handlePlayout, _playoutFramesIn10MS);
+
+ // Init varaibles used for play
+ _playWarning = 0;
+ _playError = 0;
+
+ if (_handlePlayout != NULL)
+ {
+ _playIsInitialized = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::InitRecording()
+{
+
+ int errVal = 0;
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitMicrophone() failed");
+ }
+
+ // Start by closing any existing pcm-input devices
+ //
+ if (_handleRecord != NULL)
+ {
+ int errVal = LATE(snd_pcm_close)(_handleRecord);
+ _handleRecord = NULL;
+ _recIsInitialized = false;
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error closing current recording sound device,"
+ " error: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ }
+
+ // Open PCM device for recording
+ // The corresponding settings for playout are made after the record settings
+ char deviceName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
+ kAdmMaxDeviceNameSize);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "InitRecording open (%s)", deviceName);
+ errVal = LATE(snd_pcm_open)
+ (&_handleRecord,
+ deviceName,
+ SND_PCM_STREAM_CAPTURE,
+ SND_PCM_NONBLOCK);
+
+ // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+ if (errVal == -EBUSY) // Device busy - try some more!
+ {
+ for (int i=0; i < 5; i++)
+ {
+ sleep(1);
+ errVal = LATE(snd_pcm_open)
+ (&_handleRecord,
+ deviceName,
+ SND_PCM_STREAM_CAPTURE,
+ SND_PCM_NONBLOCK);
+ if (errVal == 0)
+ {
+ break;
+ }
+ }
+ }
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " unable to open record device: %s",
+ LATE(snd_strerror)(errVal));
+ _handleRecord = NULL;
+ return -1;
+ }
+
+ _recordingFramesIn10MS = _recordingFreq/100;
+ if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE, //format
+#else
+ SND_PCM_FORMAT_S16_LE, //format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, //access
+ _recChannels, //channels
+ _recordingFreq, //rate
+ 1, //soft_resample
+ ALSA_CAPTURE_LATENCY //latency in us
+ )) < 0)
+ {
+ // Fall back to another mode then.
+ if (_recChannels == 1)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+
+ if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE, //format
+#else
+ SND_PCM_FORMAT_S16_LE, //format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, //access
+ _recChannels, //channels
+ _recordingFreq, //rate
+ 1, //soft_resample
+ ALSA_CAPTURE_LATENCY //latency in us
+ )) < 0)
+ {
+ _recordingFramesIn10MS = 0;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " unable to set record settings: %s (%d)",
+ LATE(snd_strerror)(errVal), errVal);
+ ErrorRecovery(errVal, _handleRecord);
+ errVal = LATE(snd_pcm_close)(_handleRecord);
+ _handleRecord = NULL;
+ return -1;
+ }
+ }
+
+ errVal = LATE(snd_pcm_get_params)(_handleRecord,
+ &_recordingBuffersizeInFrame, &_recordingPeriodSizeInFrame);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_pcm_get_params %s",
+ LATE(snd_strerror)(errVal), errVal);
+ _recordingBuffersizeInFrame = 0;
+ _recordingPeriodSizeInFrame = 0;
+ }
+ else {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " capture snd_pcm_get_params "
+ "buffer_size:%d period_size:%d",
+ _recordingBuffersizeInFrame, _recordingPeriodSizeInFrame);
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
+ _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+ }
+
+ // Set rec buffer size and create buffer
+ _recordingBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
+ _handleRecord, _recordingFramesIn10MS);
+
+ if (_handleRecord != NULL)
+ {
+ // Mark recording side as initialized
+ _recIsInitialized = true;
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StartRecording()
+{
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ _recording = true;
+
+ int errVal = 0;
+ _recordingFramesLeft = _recordingFramesIn10MS;
+
+ // Make sure we only create the buffer once.
+ if (!_recordingBuffer)
+ _recordingBuffer = new WebRtc_Word8[_recordingBufferSizeIn10MS];
+ if (!_recordingBuffer)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to alloc recording buffer");
+ _recording = false;
+ return -1;
+ }
+ // RECORDING
+ const char* threadName = "webrtc_audio_module_capture_thread";
+ _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc,
+ this,
+ kRealtimePriority,
+ threadName);
+ if (_ptrThreadRec == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the rec audio thread");
+ _recording = false;
+ delete [] _recordingBuffer;
+ _recordingBuffer = NULL;
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThreadRec->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the rec audio thread");
+ _recording = false;
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ delete [] _recordingBuffer;
+ _recordingBuffer = NULL;
+ return -1;
+ }
+ _recThreadID = threadID;
+
+ errVal = LATE(snd_pcm_prepare)(_handleRecord);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " capture snd_pcm_prepare failed (%s)\n",
+ LATE(snd_strerror)(errVal));
+ // just log error
+ // if snd_pcm_open fails will return -1
+ }
+
+ errVal = LATE(snd_pcm_start)(_handleRecord);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " capture snd_pcm_start err: %s",
+ LATE(snd_strerror)(errVal));
+ errVal = LATE(snd_pcm_start)(_handleRecord);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " capture snd_pcm_start 2nd try err: %s",
+ LATE(snd_strerror)(errVal));
+ StopRecording();
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StopRecording()
+{
+
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_handleRecord == NULL)
+ {
+ return -1;
+ }
+
+ // Make sure we don't start recording (it's asynchronous).
+ _recIsInitialized = false;
+ _recording = false;
+ }
+
+ if (_ptrThreadRec && !_ptrThreadRec->Stop())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to stop the rec audio thread");
+ return -1;
+ }
+ else {
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+ _recordingFramesLeft = 0;
+ if (_recordingBuffer)
+ {
+ delete [] _recordingBuffer;
+ _recordingBuffer = NULL;
+ }
+
+ // Stop and close pcm recording device.
+ int errVal = LATE(snd_pcm_drop)(_handleRecord);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error stop recording: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ errVal = LATE(snd_pcm_close)(_handleRecord);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error closing record sound device, error: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ // Check if we have muted and unmute if so.
+ bool muteEnabled = false;
+ MicrophoneMute(muteEnabled);
+ if (muteEnabled)
+ {
+ SetMicrophoneMute(false);
+ }
+
+ // set the pcm input handle to NULL
+ _handleRecord = NULL;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " handle_capture is now set to NULL");
+
+ return 0;
+}
+
+bool AudioDeviceLinuxALSA::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxALSA::Recording() const
+{
+ return (_recording);
+}
+
+bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const
+{
+ return (_playIsInitialized);
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StartPlayout()
+{
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ _playing = true;
+
+ _playoutFramesLeft = 0;
+ if (!_playoutBuffer)
+ _playoutBuffer = new WebRtc_Word8[_playoutBufferSizeIn10MS];
+ if (!_playoutBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to alloc playout buf");
+ _playing = false;
+ return -1;
+ }
+
+ // PLAYOUT
+ const char* threadName = "webrtc_audio_module_play_thread";
+ _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc,
+ this,
+ kRealtimePriority,
+ threadName);
+ if (_ptrThreadPlay == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the play audio thread");
+ _playing = false;
+ delete [] _playoutBuffer;
+ _playoutBuffer = NULL;
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThreadPlay->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the play audio thread");
+ _playing = false;
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ delete [] _playoutBuffer;
+ _playoutBuffer = NULL;
+ return -1;
+ }
+ _playThreadID = threadID;
+
+ int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " playout snd_pcm_prepare failed (%s)\n",
+ LATE(snd_strerror)(errVal));
+ // just log error
+ // if snd_pcm_open fails will return -1
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::StopPlayout()
+{
+
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_handlePlayout == NULL)
+ {
+ return -1;
+ }
+
+ _playing = false;
+ }
+
+ // stop playout thread first
+ if (_ptrThreadPlay && !_ptrThreadPlay->Stop())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to stop the play audio thread");
+ return -1;
+ }
+ else {
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ _playoutFramesLeft = 0;
+ delete [] _playoutBuffer;
+ _playoutBuffer = NULL;
+
+ // stop and close pcm playout device
+ int errVal = LATE(snd_pcm_drop)(_handlePlayout);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error stop playing: %s",
+ LATE(snd_strerror)(errVal));
+ }
+
+ errVal = LATE(snd_pcm_close)(_handlePlayout);
+ if (errVal < 0)
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error closing playout sound device, error: %s",
+ LATE(snd_strerror)(errVal));
+
+ // set the pcm input handle to NULL
+ _playIsInitialized = false;
+ _handlePlayout = NULL;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " handle_playout is now set to NULL");
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = (WebRtc_UWord16)_playoutDelay * 1000 / _playoutFreq;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ // Adding 10ms adjusted value to the record delay due to 10ms buffering.
+ delayMS = (WebRtc_UWord16)(10 + _recordingDelay * 1000 / _recordingFreq);
+ return 0;
+}
+
+bool AudioDeviceLinuxALSA::Playing() const
+{
+ return (_playing);
+}
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS)
+{
+ _playBufType = type;
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ _playBufDelayFixed = sizeMS;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const
+{
+ type = _playBufType;
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ sizeMS = _playBufDelayFixed;
+ }
+ else
+ {
+ sizeMS = _playBufDelay;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::CPULoad(WebRtc_UWord16& load) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+bool AudioDeviceLinuxALSA::PlayoutWarning() const
+{
+ return (_playWarning > 0);
+}
+
+bool AudioDeviceLinuxALSA::PlayoutError() const
+{
+ return (_playError > 0);
+}
+
+bool AudioDeviceLinuxALSA::RecordingWarning() const
+{
+ return (_recWarning > 0);
+}
+
+bool AudioDeviceLinuxALSA::RecordingError() const
+{
+ return (_recError > 0);
+}
+
+void AudioDeviceLinuxALSA::ClearPlayoutWarning()
+{
+ _playWarning = 0;
+}
+
+void AudioDeviceLinuxALSA::ClearPlayoutError()
+{
+ _playError = 0;
+}
+
+void AudioDeviceLinuxALSA::ClearRecordingWarning()
+{
+ _recWarning = 0;
+}
+
+void AudioDeviceLinuxALSA::ClearRecordingError()
+{
+ _recError = 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+WebRtc_Word32 AudioDeviceLinuxALSA::GetDevicesInfo(
+ const WebRtc_Word32 function,
+ const bool playback,
+ const WebRtc_Word32 enumDeviceNo,
+ char* enumDeviceName,
+ const WebRtc_Word32 ednLen) const
+{
+
+ // Device enumeration based on libjingle implementation
+ // by Tristan Schmelcher at Google Inc.
+
+ const char *type = playback ? "Output" : "Input";
+ // dmix and dsnoop are only for playback and capture, respectively, but ALSA
+ // stupidly includes them in both lists.
+ const char *ignorePrefix = playback ? "dsnoop:" : "dmix:" ;
+ // (ALSA lists many more "devices" of questionable interest, but we show them
+ // just in case the weird devices may actually be desirable for some
+ // users/systems.)
+
+ int err;
+ int enumCount(0);
+ bool keepSearching(true);
+
+ void **hints;
+ err = LATE(snd_device_name_hint)(-1, // All cards
+ "pcm", // Only PCM devices
+ &hints);
+ if (err != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetDevicesInfo - device name hint error: %s",
+ LATE(snd_strerror)(err));
+ return -1;
+ }
+
+ enumCount++; // default is 0
+ if (function == FUNC_GET_DEVICE_NAME && enumDeviceNo == 0)
+ {
+ strcpy(enumDeviceName, "default");
+ return 0;
+ }
+ if (function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM && enumDeviceNo == 0)
+ {
+ strcpy(enumDeviceName, "default");
+ return 0;
+ }
+
+ for (void **list = hints; *list != NULL; ++list)
+ {
+ char *actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
+ if (actualType)
+ { // NULL means it's both.
+ bool wrongType = (strcmp(actualType, type) != 0);
+ free(actualType);
+ if (wrongType)
+ {
+ // Wrong type of device (i.e., input vs. output).
+ continue;
+ }
+ }
+
+ char *name = LATE(snd_device_name_get_hint)(*list, "NAME");
+ if (!name)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Device has no name");
+ // Skip it.
+ continue;
+ }
+
+ // Now check if we actually want to show this device.
+ if (strcmp(name, "default") != 0 &&
+ strcmp(name, "null") != 0 &&
+ strcmp(name, "pulse") != 0 &&
+ strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0)
+ {
+ // Yes, we do.
+ char *desc = LATE(snd_device_name_get_hint)(*list, "DESC");
+ if (!desc)
+ {
+ // Virtual devices don't necessarily have descriptions.
+ // Use their names instead
+ desc = name;
+ }
+
+ if (FUNC_GET_NUM_OF_DEVICE == function)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Enum device %d - %s", enumCount, name);
+
+ }
+ if ((FUNC_GET_DEVICE_NAME == function) &&
+ (enumDeviceNo == enumCount))
+ {
+
+ // We have found the enum device, copy the name to buffer
+ strncpy(enumDeviceName, desc, ednLen);
+ enumDeviceName[ednLen-1] = '\0';
+ keepSearching = false;
+ // replace '\n' with '-'
+ char * pret = strchr(enumDeviceName, '\n'/*0xa*/); //LF
+ if (pret)
+ *pret = '-';
+ }
+ if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
+ (enumDeviceNo == enumCount))
+ {
+ // We have found the enum device, copy the name to buffer
+ strncpy(enumDeviceName, name, ednLen);
+ enumDeviceName[ednLen-1] = '\0';
+ keepSearching = false;
+ }
+ if (keepSearching)
+ {
+ ++enumCount;
+ }
+
+ if (desc != name)
+ {
+ free(desc);
+ }
+ }
+
+ free(name);
+
+ if (!keepSearching)
+ {
+ break;
+ }
+ }
+
+ err = LATE(snd_device_name_free_hint)(hints);
+ if (err != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetDevicesInfo - device name free hint error: %s",
+ LATE(snd_strerror)(err));
+ // Continue and return true anyways, since we did get the whole list.
+ }
+
+ if (FUNC_GET_NUM_OF_DEVICE == function)
+ {
+ if (enumCount == 1) // only default?
+ enumCount = 0;
+ return enumCount; // Normal return point for function 0
+ }
+
+ if (keepSearching)
+ {
+ // If we get here for function 1 and 2, we didn't find the specified
+ // enum device
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetDevicesInfo - Could not find device name or numbers");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const
+{
+ if (_handleRecord == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " input state has been modified during unlocked period");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const
+{
+ if (_handlePlayout == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " output state has been modified during unlocked period");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxALSA::ErrorRecovery(WebRtc_Word32 error,
+ snd_pcm_t* deviceHandle)
+{
+ int st = LATE(snd_pcm_state)(deviceHandle);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Trying to recover from error: %s (%d) (state %d)",
+ (LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) ?
+ "capture" : "playout", LATE(snd_strerror)(error), error, st);
+
+ // It is recommended to use snd_pcm_recover for all errors. If that function
+ // cannot handle the error, the input error code will be returned, otherwise
+ // 0 is returned. From snd_pcm_recover API doc: "This functions handles
+ // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
+ // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
+ // trying to prepare given stream for next I/O."
+
+ /** Open */
+ // SND_PCM_STATE_OPEN = 0,
+ /** Setup installed */
+ // SND_PCM_STATE_SETUP,
+ /** Ready to start */
+ // SND_PCM_STATE_PREPARED,
+ /** Running */
+ // SND_PCM_STATE_RUNNING,
+ /** Stopped: underrun (playback) or overrun (capture) detected */
+ // SND_PCM_STATE_XRUN,= 4
+ /** Draining: running (playback) or stopped (capture) */
+ // SND_PCM_STATE_DRAINING,
+ /** Paused */
+ // SND_PCM_STATE_PAUSED,
+ /** Hardware is suspended */
+ // SND_PCM_STATE_SUSPENDED,
+ // ** Hardware is disconnected */
+ // SND_PCM_STATE_DISCONNECTED,
+ // SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
+
+ // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
+ // in Sthlm lab.
+
+ int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
+ if (0 == res)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Recovery - snd_pcm_recover OK");
+
+ if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
+ _recording &&
+ LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE)
+ {
+ // For capture streams we also have to repeat the explicit start()
+ // to get data flowing again.
+ int err = LATE(snd_pcm_start)(deviceHandle);
+ if (err != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recovery - snd_pcm_start error: %u", err);
+ return -1;
+ }
+ }
+
+ if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
+ _playing &&
+ LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK)
+ {
+ // For capture streams we also have to repeat the explicit start() to get
+ // data flowing again.
+ int err = LATE(snd_pcm_start)(deviceHandle);
+ if (err != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recovery - snd_pcm_start error: %s",
+ LATE(snd_strerror)(err));
+ return -1;
+ }
+ }
+
+ return -EPIPE == error ? 1 : 0;
+ }
+ else {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Terriable, it shouldn't happen");
+ }
+
+ return res;
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+bool AudioDeviceLinuxALSA::PlayThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceLinuxALSA*>(pThis)->PlayThreadProcess());
+}
+
+bool AudioDeviceLinuxALSA::RecThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceLinuxALSA*>(pThis)->RecThreadProcess());
+}
+
+bool AudioDeviceLinuxALSA::PlayThreadProcess()
+{
+ if(!_playing)
+ return false;
+
+ int err;
+ snd_pcm_sframes_t frames;
+ snd_pcm_sframes_t avail_frames;
+
+ Lock();
+ //return a positive number of frames ready otherwise a negative error code
+ avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
+ if (avail_frames < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "playout snd_pcm_avail_update error: %s",
+ LATE(snd_strerror)(avail_frames));
+ ErrorRecovery(avail_frames, _handlePlayout);
+ UnLock();
+ return true;
+ }
+ else if (avail_frames == 0)
+ {
+ UnLock();
+
+ //maximum tixe in milliseconds to wait, a negative value means infinity
+ err = LATE(snd_pcm_wait)(_handlePlayout, 2);
+ if (err == 0)
+ { //timeout occured
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+ "playout snd_pcm_wait timeout");
+ }
+
+ return true;
+ }
+
+ if (_playoutFramesLeft <= 0)
+ {
+ UnLock();
+ _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+ Lock();
+
+ _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+ assert(_playoutFramesLeft == _playoutFramesIn10MS);
+ }
+
+ if (static_cast<WebRtc_UWord32>(avail_frames) > _playoutFramesLeft)
+ avail_frames = _playoutFramesLeft;
+
+ int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout,
+ _playoutFramesLeft);
+ frames = LATE(snd_pcm_writei)(
+ _handlePlayout,
+ &_playoutBuffer[_playoutBufferSizeIn10MS - size],
+ avail_frames);
+
+ if (frames < 0)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+ "playout snd_pcm_avail_update error: %s",
+ LATE(snd_strerror)(frames));
+ _playoutFramesLeft = 0;
+ ErrorRecovery(frames, _handlePlayout);
+ UnLock();
+ return true;
+ }
+ else {
+ assert(frames==avail_frames);
+ _playoutFramesLeft -= frames;
+ }
+
+ UnLock();
+ return true;
+}
+
+bool AudioDeviceLinuxALSA::RecThreadProcess()
+{
+ if (!_recording)
+ return false;
+
+ int err;
+ snd_pcm_sframes_t frames;
+ snd_pcm_sframes_t avail_frames;
+ WebRtc_Word8 buffer[_recordingBufferSizeIn10MS];
+
+ Lock();
+
+ //return a positive number of frames ready otherwise a negative error code
+ avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
+ if (avail_frames < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "capture snd_pcm_avail_update error: %s",
+ LATE(snd_strerror)(avail_frames));
+ ErrorRecovery(avail_frames, _handleRecord);
+ UnLock();
+ return true;
+ }
+ else if (avail_frames == 0)
+ { // no frame is available now
+ UnLock();
+
+ //maximum time in milliseconds to wait, a negative value means infinity
+ err = LATE(snd_pcm_wait)(_handleRecord,
+ ALSA_CAPTURE_WAIT_TIMEOUT);
+ if (err == 0) //timeout occured
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+ "caputre snd_pcm_wait timeout");
+
+ return true;
+ }
+
+ if (static_cast<WebRtc_UWord32>(avail_frames) > _recordingFramesLeft)
+ avail_frames = _recordingFramesLeft;
+
+ frames = LATE(snd_pcm_readi)(_handleRecord,
+ buffer, avail_frames); // frames to be written
+ if (frames < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "caputre snd_pcm_readi error: %s",
+ LATE(snd_strerror)(frames));
+ ErrorRecovery(frames, _handleRecord);
+ UnLock();
+ return true;
+ }
+ else if (frames > 0)
+ {
+ assert(frames == avail_frames);
+
+ int left_size = LATE(snd_pcm_frames_to_bytes)(_handleRecord,
+ _recordingFramesLeft);
+ int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
+
+ memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size],
+ buffer, size);
+ _recordingFramesLeft -= frames;
+
+ if (!_recordingFramesLeft)
+ { // buf is full
+ _recordingFramesLeft = _recordingFramesIn10MS;
+
+ // store the recorded buffer (no action will be taken if the
+ // #recorded samples is not a full buffer)
+ _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+ _recordingFramesIn10MS);
+
+ WebRtc_UWord32 currentMicLevel = 0;
+ WebRtc_UWord32 newMicLevel = 0;
+
+ if (AGC())
+ {
+ // store current mic level in the audio buffer if AGC is enabled
+ if (MicrophoneVolume(currentMicLevel) == 0)
+ {
+ if (currentMicLevel == 0xffffffff)
+ currentMicLevel = 100;
+ // this call does not affect the actual microphone volume
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ }
+
+ // calculate delay
+ _playoutDelay = 0;
+ _recordingDelay = 0;
+ if (_handlePlayout)
+ {
+ err = LATE(snd_pcm_delay)(_handlePlayout,
+ &_playoutDelay); // returned delay in frames
+ if (err < 0)
+ {
+ // TODO(xians): Shall we call ErrorRecovery() here?
+ _playoutDelay = 0;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "playout snd_pcm_delay: %s",
+ LATE(snd_strerror)(err));
+ }
+ }
+
+ err = LATE(snd_pcm_delay)(_handleRecord,
+ &_recordingDelay); // returned delay in frames
+ if (err < 0)
+ {
+ // TODO(xians): Shall we call ErrorRecovery() here?
+ _recordingDelay = 0;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "caputre snd_pcm_delay: %s",
+ LATE(snd_strerror)(err));
+ }
+
+ // TODO(xians): Shall we add 10ms buffer delay to the record delay?
+ _ptrAudioBuffer->SetVQEData(
+ _playoutDelay * 1000 / _playoutFreq,
+ _recordingDelay * 1000 / _recordingFreq, 0);
+
+ // Deliver recorded samples at specified sample rate, mic level etc.
+ // to the observer using callback.
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ Lock();
+
+ if (AGC())
+ {
+ newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0)
+ {
+ // The VQE will only deliver non-zero microphone levels when a
+ // change is needed. Set this new mic level (received from the
+ // observer as return value in the callback).
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " the required modification of the "
+ "microphone volume failed");
+ }
+ }
+ }
+ }
+
+ UnLock();
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h
new file mode 100644
index 0000000..9a5a032
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
+
+#include "audio_device_generic.h"
+#include "critical_section_wrapper.h"
+#include "audio_mixer_manager_alsa_linux.h"
+
+#include <sys/soundcard.h>
+#include <sys/ioctl.h>
+
+#include <alsa/asoundlib.h>
+
+namespace webrtc
+{
+class EventWrapper;
+class ThreadWrapper;
+
+class AudioDeviceLinuxALSA : public AudioDeviceGeneric
+{
+public:
+ AudioDeviceLinuxALSA(const WebRtc_Word32 id);
+ ~AudioDeviceLinuxALSA();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+public:
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private:
+ WebRtc_Word32 GetDevicesInfo(const WebRtc_Word32 function,
+ const bool playback,
+ const WebRtc_Word32 enumDeviceNo = 0,
+ char* enumDeviceName = NULL,
+ const WebRtc_Word32 ednLen = 0) const;
+ WebRtc_Word32 ErrorRecovery(WebRtc_Word32 error, snd_pcm_t* deviceHandle);
+
+private:
+ void Lock() { _critSect.Enter(); };
+ void UnLock() { _critSect.Leave(); };
+private:
+ inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
+ inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
+
+private:
+ static bool RecThreadFunc(void*);
+ static bool PlayThreadFunc(void*);
+ bool RecThreadProcess();
+ bool PlayThreadProcess();
+
+private:
+ AudioDeviceBuffer* _ptrAudioBuffer;
+
+ CriticalSectionWrapper& _critSect;
+
+ ThreadWrapper* _ptrThreadRec;
+ ThreadWrapper* _ptrThreadPlay;
+ WebRtc_UWord32 _recThreadID;
+ WebRtc_UWord32 _playThreadID;
+
+ WebRtc_Word32 _id;
+
+ AudioMixerManagerLinuxALSA _mixerManager;
+
+ WebRtc_UWord16 _inputDeviceIndex;
+ WebRtc_UWord16 _outputDeviceIndex;
+ bool _inputDeviceIsSpecified;
+ bool _outputDeviceIsSpecified;
+
+ snd_pcm_t* _handleRecord;
+ snd_pcm_t* _handlePlayout;
+
+ snd_pcm_uframes_t _recordingBuffersizeInFrame;
+ snd_pcm_uframes_t _recordingPeriodSizeInFrame;
+ snd_pcm_uframes_t _playoutBufferSizeInFrame;
+ snd_pcm_uframes_t _playoutPeriodSizeInFrame;
+
+ ssize_t _recordingBufferSizeIn10MS;
+ ssize_t _playoutBufferSizeIn10MS;
+ WebRtc_UWord32 _recordingFramesIn10MS;
+ WebRtc_UWord32 _playoutFramesIn10MS;
+
+ WebRtc_UWord32 _recordingFreq;
+ WebRtc_UWord32 _playoutFreq;
+ WebRtc_UWord8 _recChannels;
+ WebRtc_UWord8 _playChannels;
+
+ WebRtc_Word8* _recordingBuffer; // in byte
+ WebRtc_Word8* _playoutBuffer; // in byte
+ WebRtc_UWord32 _recordingFramesLeft;
+ WebRtc_UWord32 _playoutFramesLeft;
+
+ WebRtc_UWord32 _playbackBufferSize;
+
+ AudioDeviceModule::BufferType _playBufType;
+
+private:
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _AGC;
+
+ snd_pcm_sframes_t _recordingDelay;
+ snd_pcm_sframes_t _playoutDelay;
+
+ WebRtc_Word32 _writeErrors;
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ WebRtc_UWord16 _playBufDelay; // playback delay
+ WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
+};
+
+}
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
new file mode 100644
index 0000000..16e1d33
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
@@ -0,0 +1,3150 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "audio_device_utility.h"
+#include "audio_device_pulse_linux.h"
+#include "audio_device_config.h"
+
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+
+webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym) \
+ LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, sym)
+
+namespace webrtc
+{
+
+// ============================================================================
+// Static Methods
+// ============================================================================
+
+bool AudioDeviceLinuxPulse::PulseAudioIsSupported()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s",
+ __FUNCTION__);
+
+ bool pulseAudioIsSupported(true);
+
+ // Check that we can initialize
+ AudioDeviceLinuxPulse* admPulse = new AudioDeviceLinuxPulse(-1);
+ if (admPulse->InitPulseAudio() == -1)
+ {
+ pulseAudioIsSupported = false;
+ }
+ admPulse->TerminatePulseAudio();
+ delete admPulse;
+
+ if (pulseAudioIsSupported)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
+ "*** Linux Pulse Audio is supported ***");
+ } else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
+ "*** Linux Pulse Audio is NOT supported => will revert to the ALSA API ***");
+ }
+
+ return (pulseAudioIsSupported);
+}
+
+AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _timeEventRec(*EventWrapper::Create()),
+ _timeEventPlay(*EventWrapper::Create()),
+ _recStartEvent(*EventWrapper::Create()),
+ _playStartEvent(*EventWrapper::Create()),
+ _ptrThreadPlay(NULL),
+ _ptrThreadRec(NULL),
+ _recThreadID(0),
+ _playThreadID(0),
+ _id(id),
+ _mixerManager(id),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _inputDeviceIsSpecified(false),
+ _outputDeviceIsSpecified(false),
+ _samplingFreq(0),
+ _recChannels(1),
+ _playChannels(1),
+ _playBufType(AudioDeviceModule::kFixedBufferSize),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _startRec(false),
+ _stopRec(false),
+ _startPlay(false),
+ _stopPlay(false),
+ _AGC(false),
+ _playBufDelayFixed(20),
+ _sndCardPlayDelay(0),
+ _sndCardRecDelay(0),
+ _writeErrors(0),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _deviceIndex(-1),
+ _numPlayDevices(0),
+ _numRecDevices(0),
+ _playDeviceName(NULL),
+ _recDeviceName(NULL),
+ _playDisplayDeviceName(NULL),
+ _recDisplayDeviceName(NULL),
+ _playBuffer(NULL),
+ _playbackBufferSize(0),
+ _playbackBufferUnused(0),
+ _tempBufferSpace(0),
+ _recBuffer(NULL),
+ _recordBufferSize(0),
+ _recordBufferUsed(0),
+ _tempSampleData(NULL),
+ _tempSampleDataSize(0),
+ _configuredLatencyPlay(0),
+ _configuredLatencyRec(0),
+ _paDeviceIndex(-1),
+ _paStateChanged(false),
+ _paMainloop(NULL),
+ _paMainloopApi(NULL),
+ _paContext(NULL),
+ _recStream(NULL),
+ _playStream(NULL),
+ _recStreamFlags(0),
+ _playStreamFlags(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+
+ memset(_paServerVersion, 0, sizeof(_paServerVersion));
+ memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
+ memset(&_recBufferAttr, 0, sizeof(_recBufferAttr));
+}
+
+AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ if (_recBuffer)
+ {
+ delete [] _recBuffer;
+ _recBuffer = NULL;
+ }
+ if (_playBuffer)
+ {
+ delete [] _playBuffer;
+ _playBuffer = NULL;
+ }
+ if (_playDeviceName)
+ {
+ delete [] _playDeviceName;
+ _playDeviceName = NULL;
+ }
+ if (_recDeviceName)
+ {
+ delete [] _recDeviceName;
+ _recDeviceName = NULL;
+ }
+
+ delete &_recStartEvent;
+ delete &_playStartEvent;
+ delete &_timeEventRec;
+ delete &_timeEventPlay;
+ delete &_critSect;
+}
+
+void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // Inform the AudioBuffer about default settings for this implementation.
+ // Set all values to zero here since the actual settings will be done by
+ // InitPlayout and InitRecording later.
+ _ptrAudioBuffer->SetRecordingSampleRate(0);
+ _ptrAudioBuffer->SetPlayoutSampleRate(0);
+ _ptrAudioBuffer->SetRecordingChannels(0);
+ _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceLinuxPulse::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kLinuxPulseAudio;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ // Initialize PulseAudio
+ if (InitPulseAudio() < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to initialize PulseAudio");
+
+ if (TerminatePulseAudio() < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to terminate PulseAudio");
+ }
+
+ return -1;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ // RECORDING
+ const char* threadName = "webrtc_audio_module_rec_thread";
+ _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
+ kRealtimePriority, threadName);
+ if (_ptrThreadRec == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the rec audio thread");
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThreadRec->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the rec audio thread");
+
+ delete _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ return -1;
+ }
+ _recThreadID = threadID;
+
+ // PLAYOUT
+ threadName = "webrtc_audio_module_play_thread";
+ _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
+ kRealtimePriority, threadName);
+ if (_ptrThreadPlay == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to create the play audio thread");
+ return -1;
+ }
+
+ threadID = 0;
+ if (!_ptrThreadPlay->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " failed to start the play audio thread");
+
+ delete _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ return -1;
+ }
+ _playThreadID = threadID;
+
+ _initialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::Terminate()
+{
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ Lock();
+
+ _mixerManager.Close();
+
+ // RECORDING
+ if (_ptrThreadRec)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadRec;
+ _ptrThreadRec = NULL;
+ UnLock();
+
+ tmpThread->SetNotAlive();
+ _timeEventRec.Set();
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the rec audio thread");
+ }
+ // Lock again since we need to protect _ptrThreadPlay.
+ Lock();
+ }
+
+ // PLAYOUT
+ if (_ptrThreadPlay)
+ {
+ ThreadWrapper* tmpThread = _ptrThreadPlay;
+ _ptrThreadPlay = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+ _timeEventPlay.Set();
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to close down the play audio thread");
+ }
+ } else {
+ UnLock();
+ }
+
+ // Terminate PulseAudio
+ if (TerminatePulseAudio() < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to terminate PulseAudio");
+ return -1;
+ }
+
+ _initialized = false;
+ _outputDeviceIsSpecified = false;
+ _inputDeviceIsSpecified = false;
+
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::Initialized() const
+{
+ return (_initialized);
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a valid speaker exists
+ //
+ available = true;
+
+ // Close the initialized output mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (!_outputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ // check if default device
+ if (_outputDeviceIndex == 0)
+ {
+ WebRtc_UWord16 deviceIndex = 0;
+ GetDefaultDeviceInfo(false, NULL, deviceIndex);
+ _paDeviceIndex = deviceIndex;
+ } else
+ {
+ // get the PA device index from
+ // the callback
+ _deviceIndex = _outputDeviceIndex;
+
+ // get playout devices
+ PlayoutDevices();
+ }
+
+ // the callback has now set the _paDeviceIndex to
+ // the PulseAudio index of the device
+ if (_mixerManager.OpenSpeaker(_paDeviceIndex) == -1)
+ {
+ return -1;
+ }
+
+ // clear _deviceIndex
+ _deviceIndex = -1;
+ _paDeviceIndex = -1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a valid microphone
+ // exists
+ available = true;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ // Check if default device
+ if (_inputDeviceIndex == 0)
+ {
+ WebRtc_UWord16 deviceIndex = 0;
+ GetDefaultDeviceInfo(true, NULL, deviceIndex);
+ _paDeviceIndex = deviceIndex;
+ } else
+ {
+ // Get the PA device index from
+ // the callback
+ _deviceIndex = _inputDeviceIndex;
+
+ // get recording devices
+ RecordingDevices();
+ }
+
+ // The callback has now set the _paDeviceIndex to
+ // the PulseAudio index of the device
+ if (_mixerManager.OpenMicrophone(_paDeviceIndex) == -1)
+ {
+ return -1;
+ }
+
+ // Clear _deviceIndex
+ _deviceIndex = -1;
+ _paDeviceIndex = -1;
+
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const
+{
+ return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const
+{
+ return (_mixerManager.MicrophoneIsInitialized());
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a volume control exists
+ available = true;
+
+ // Close the initialized output mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.SpeakerVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ volume = level;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetWaveOutVolume(
+ WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::WaveOutVolume(
+ WebRtc_UWord16& /*volumeLeft*/,
+ WebRtc_UWord16& /*volumeRight*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MaxSpeakerVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MinSpeakerVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinSpeakerVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected speaker has a mute control
+ _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetSpeakerMute(bool enable)
+{
+
+ return (_mixerManager.SetSpeakerMute(enable));
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const
+{
+
+ bool muted(0);
+ if (_mixerManager.SpeakerMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a mute control
+ //
+ _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable)
+{
+
+ return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const
+{
+
+ bool muted(0);
+ if (_mixerManager.MicrophoneMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Enumerate all avaliable microphone and make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a boost control
+ _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneBoost(bool enable)
+{
+
+ return (_mixerManager.SetMicrophoneBoost(enable));
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneBoost(bool& enabled) const
+{
+
+ bool onOff(0);
+
+ if (_mixerManager.MicrophoneBoost(onOff) == -1)
+ {
+ return -1;
+ }
+
+ enabled = onOff;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available)
+{
+
+ if (_recChannels == 2 && _recording) {
+ available = true;
+ return 0;
+ }
+
+ available = false;
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+ int error = 0;
+
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // Cannot open the specified device
+ available = false;
+ return 0;
+ }
+
+#ifndef WEBRTC_PA_GTALK
+ // Check if the selected microphone can record stereo.
+ bool isAvailable(false);
+ error = _mixerManager.StereoRecordingIsAvailable(isAvailable);
+ if (!error)
+ available = isAvailable;
+#endif
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return error;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetStereoRecording(bool enable)
+{
+
+#ifndef WEBRTC_PA_GTALK
+ if (enable)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+#endif
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const
+{
+
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available)
+{
+
+ if (_playChannels == 2 && _playing) {
+ available = true;
+ return 0;
+ }
+
+ available = false;
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+ int error = 0;
+
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // Cannot open the specified device.
+ return -1;
+ }
+
+#ifndef WEBRTC_PA_GTALK
+ // Check if the selected speaker can play stereo.
+ bool isAvailable(false);
+ error = _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+ if (!error)
+ available = isAvailable;
+#endif
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return error;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetStereoPlayout(bool enable)
+{
+
+#ifndef WEBRTC_PA_GTALK
+ if (enable)
+ _playChannels = 2;
+ else
+ _playChannels = 1;
+#endif
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
+{
+
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetAGC(bool enable)
+{
+
+ _AGC = enable;
+
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::AGC() const
+{
+
+ return _AGC;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(
+ bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a volume control
+ // exists
+ available = true;
+
+ // Close the initialized input mixer
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolume(
+ WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.MicrophoneVolume(level) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to retrive current microphone level");
+ return -1;
+ }
+
+ volume = level;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MaxMicrophoneVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MinMicrophoneVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceLinuxPulse::PlayoutDevices()
+{
+
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+ _numPlayDevices = 1; // init to 1 to account for "default"
+
+ // get the whole list of devices and update _numPlayDevices
+ paOperation = LATE(pa_context_get_sink_info_list)(_paContext,
+ PaSinkInfoCallback,
+ this);
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ return _numPlayDevices;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ const WebRtc_UWord16 nDevices = PlayoutDevices();
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable output devices is %u", nDevices);
+
+ if (index > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices - 1));
+ return -1;
+ }
+
+ _outputDeviceIndex = index;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices = PlayoutDevices();
+
+ if ((index > (nDevices - 1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ // Check if default device
+ if (index == 0)
+ {
+ WebRtc_UWord16 deviceIndex = 0;
+ return GetDefaultDeviceInfo(false, name, deviceIndex);
+ }
+
+ // Tell the callback that we want
+ // The name for this device
+ _playDisplayDeviceName = name;
+ _deviceIndex = index;
+
+ // get playout devices
+ PlayoutDevices();
+
+ // clear device name and index
+ _playDisplayDeviceName = NULL;
+ _deviceIndex = -1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices(RecordingDevices());
+
+ if ((index > (nDevices - 1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ // Check if default device
+ if (index == 0)
+ {
+ WebRtc_UWord16 deviceIndex = 0;
+ return GetDefaultDeviceInfo(true, name, deviceIndex);
+ }
+
+ // Tell the callback that we want
+ // the name for this device
+ _recDisplayDeviceName = name;
+ _deviceIndex = index;
+
+ // Get recording devices
+ RecordingDevices();
+
+ // Clear device name and index
+ _recDisplayDeviceName = NULL;
+ _deviceIndex = -1;
+
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceLinuxPulse::RecordingDevices()
+{
+
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+ _numRecDevices = 1; // Init to 1 to account for "default"
+
+ // Get the whole list of devices and update _numRecDevices
+ paOperation = LATE(pa_context_get_source_info_list)(_paContext,
+ PaSourceInfoCallback,
+ this);
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ return _numRecDevices;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ const WebRtc_UWord16 nDevices(RecordingDevices());
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable input devices is %u", nDevices);
+
+ if (index > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices - 1));
+ return -1;
+ }
+
+ _inputDeviceIndex = index;
+ _inputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (!_outputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitSpeaker() failed");
+ }
+
+ // Set sampling rate to use
+ WebRtc_UWord32 samplingRate = _samplingFreq * 1000;
+ if (samplingRate == 44000)
+ {
+ samplingRate = 44100;
+ }
+
+ // Set the play sample specification
+ pa_sample_spec playSampleSpec;
+ playSampleSpec.channels = _playChannels;
+ playSampleSpec.format = PA_SAMPLE_S16LE;
+ playSampleSpec.rate = samplingRate;
+
+ // Create a new play stream
+ _playStream = LATE(pa_stream_new)(_paContext, "playStream",
+ &playSampleSpec, NULL);
+
+ if (!_playStream)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create play stream, err=%d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ // Provide the playStream to the mixer
+ _mixerManager.SetPlayStream(_playStream);
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreq * 1000);
+ _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " stream state %d\n", LATE(pa_stream_get_state)(_playStream));
+
+ // Set stream flags
+ _playStreamFlags = (pa_stream_flags_t) (PA_STREAM_AUTO_TIMING_UPDATE
+ | PA_STREAM_INTERPOLATE_TIMING);
+
+ if (_configuredLatencyPlay != WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
+ {
+ // If configuring a specific latency then we want to specify
+ // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+ // automatically to reach that target latency. However, that flag doesn't
+ // exist in Ubuntu 8.04 and many people still use that, so we have to check
+ // the protocol version of libpulse.
+ if (LATE(pa_context_get_protocol_version)(_paContext)
+ >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
+ {
+ _playStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+ }
+
+ const pa_sample_spec *spec =
+ LATE(pa_stream_get_sample_spec)(_playStream);
+ if (!spec)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " pa_stream_get_sample_spec()");
+ return -1;
+ }
+
+ size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+ WebRtc_UWord32 latency = bytesPerSec
+ * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+
+ // Set the play buffer attributes
+ _playBufferAttr.maxlength = latency; // num bytes stored in the buffer
+ _playBufferAttr.tlength = latency; // target fill level of play buffer
+ // minimum free num bytes before server request more data
+ _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+ _playBufferAttr.prebuf = _playBufferAttr.tlength
+ - _playBufferAttr.minreq; // prebuffer tlength before starting playout
+
+ _configuredLatencyPlay = latency;
+ }
+
+ // num samples in bytes * num channels
+ _playbackBufferSize = _samplingFreq * 10 * 2 * _playChannels;
+ _playbackBufferUnused = _playbackBufferSize;
+ _playBuffer = new WebRtc_Word8[_playbackBufferSize];
+
+ // Enable underflow callback
+ LATE(pa_stream_set_underflow_callback)(_playStream,
+ PaStreamUnderflowCallback, this);
+
+ // Set the state callback function for the stream
+ LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
+
+ // Mark playout side as initialized
+ _playIsInitialized = true;
+ _sndCardPlayDelay = 0;
+ _sndCardRecDelay = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitMicrophone() failed");
+ }
+
+ // Set sampling rate to use
+ WebRtc_UWord32 samplingRate = _samplingFreq * 1000;
+ if (samplingRate == 44000)
+ {
+ samplingRate = 44100;
+ }
+
+ // Set the rec sample specification
+ pa_sample_spec recSampleSpec;
+ recSampleSpec.channels = _recChannels;
+ recSampleSpec.format = PA_SAMPLE_S16LE;
+ recSampleSpec.rate = samplingRate;
+
+ // Create a new rec stream
+ _recStream = LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec,
+ NULL);
+ if (!_recStream)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create rec stream, err=%d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ // Provide the recStream to the mixer
+ _mixerManager.SetRecStream(_recStream);
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreq * 1000);
+ _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
+ }
+
+ if (_configuredLatencyRec != WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
+ {
+ _recStreamFlags = (pa_stream_flags_t) (PA_STREAM_AUTO_TIMING_UPDATE
+ | PA_STREAM_INTERPOLATE_TIMING);
+
+ // If configuring a specific latency then we want to specify
+ // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+ // automatically to reach that target latency. However, that flag doesn't
+ // exist in Ubuntu 8.04 and many people still use that, so we have to check
+ // the protocol version of libpulse.
+ if (LATE(pa_context_get_protocol_version)(_paContext)
+ >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
+ {
+ _recStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+ }
+
+ const pa_sample_spec *spec =
+ LATE(pa_stream_get_sample_spec)(_recStream);
+ if (!spec)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " pa_stream_get_sample_spec(rec)");
+ return -1;
+ }
+
+ size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+ WebRtc_UWord32 latency = bytesPerSec
+ * WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+
+ // Set the rec buffer attributes
+ // Note: fragsize specifies a maximum transfer size, not a minimum, so
+ // it is not possible to force a high latency setting, only a low one.
+ _recBufferAttr.fragsize = latency; // size of fragment
+ _recBufferAttr.maxlength = latency + bytesPerSec
+ * WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+
+ _configuredLatencyRec = latency;
+ }
+
+ _recordBufferSize = _samplingFreq * 10 * 2 * _recChannels;
+ _recordBufferUsed = 0;
+ _recBuffer = new WebRtc_Word8[_recordBufferSize];
+
+ // Enable overflow callback
+ LATE(pa_stream_set_overflow_callback)(_recStream, PaStreamOverflowCallback,
+ this);
+
+ // Set the state callback function for the stream
+ LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this);
+
+ // Mark recording side as initialized
+ _recIsInitialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StartRecording()
+{
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ // set state to ensure that the recording starts from the audio thread
+ _startRec = true;
+
+ // the audio thread will signal when recording has started
+ _timeEventRec.Set();
+ if (kEventTimeout == _recStartEvent.Wait(10000))
+ {
+ _startRec = false;
+ StopRecording();
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to activate recording");
+ return -1;
+ }
+
+ if (_recording)
+ {
+ // the recording state is set by the audio thread after recording has started
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to activate recording");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StopRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_recStream == NULL)
+ {
+ return -1;
+ }
+
+ _recIsInitialized = false;
+ _recording = false;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " stopping recording");
+
+ // Stop Recording
+ PaLock();
+
+ DisableReadCallback();
+ LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL);
+
+ // Unset this here so that we don't get a TERMINATED callback
+ LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL);
+
+ if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED)
+ {
+ // Disconnect the stream
+ if (LATE(pa_stream_disconnect)(_recStream) != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to disconnect rec stream, err=%d\n",
+ LATE(pa_context_errno)(_paContext));
+ PaUnLock();
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " disconnected recording");
+ }
+
+ LATE(pa_stream_unref)(_recStream);
+ _recStream = NULL;
+
+ PaUnLock();
+
+ // Provide the recStream to the mixer
+ _mixerManager.SetRecStream(_recStream);
+
+ if (_recBuffer)
+ {
+ delete [] _recBuffer;
+ _recBuffer = NULL;
+ }
+
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxPulse::Recording() const
+{
+ return (_recording);
+}
+
+bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const
+{
+ return (_playIsInitialized);
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StartPlayout()
+{
+
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ // set state to ensure that playout starts from the audio thread
+ _startPlay = true;
+
+ // the audio thread will signal when playout has started
+ _timeEventPlay.Set();
+ if (kEventTimeout == _playStartEvent.Wait(10000))
+ {
+ _startPlay = false;
+ StopPlayout();
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to activate playout");
+ return -1;
+ }
+
+ if (_playing)
+ {
+ // the playing state is set by the audio thread after playout has started
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to activate playing");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::StopPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_playStream == NULL)
+ {
+ return -1;
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+ _sndCardPlayDelay = 0;
+ _sndCardRecDelay = 0;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " stopping playback");
+
+ // Stop Playout
+ PaLock();
+
+ DisableWriteCallback();
+ LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL);
+
+ // Unset this here so that we don't get a TERMINATED callback
+ LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL);
+
+ if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED)
+ {
+ // Disconnect the stream
+ if (LATE(pa_stream_disconnect)(_playStream) != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to disconnect play stream, err=%d",
+ LATE(pa_context_errno)(_paContext));
+ PaUnLock();
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " disconnected playback");
+ }
+
+ LATE(pa_stream_unref)(_playStream);
+ _playStream = NULL;
+
+ PaUnLock();
+
+ // Provide the playStream to the mixer
+ _mixerManager.SetPlayStream(_playStream);
+
+ if (_playBuffer)
+ {
+ delete [] _playBuffer;
+ _playBuffer = NULL;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = (WebRtc_UWord16) _sndCardPlayDelay;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = (WebRtc_UWord16) _sndCardRecDelay;
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::Playing() const
+{
+ return (_playing);
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS)
+{
+
+ if (type != AudioDeviceModule::kFixedBufferSize)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Adaptive buffer size not supported on this platform");
+ return -1;
+ }
+
+ _playBufType = type;
+ _playBufDelayFixed = sizeMS;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const
+{
+
+ type = _playBufType;
+ sizeMS = _playBufDelayFixed;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::CPULoad(WebRtc_UWord16& /*load*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+bool AudioDeviceLinuxPulse::PlayoutWarning() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return (_playWarning > 0);
+}
+
+bool AudioDeviceLinuxPulse::PlayoutError() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return (_playError > 0);
+}
+
+bool AudioDeviceLinuxPulse::RecordingWarning() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return (_recWarning > 0);
+}
+
+bool AudioDeviceLinuxPulse::RecordingError() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return (_recError > 0);
+}
+
+void AudioDeviceLinuxPulse::ClearPlayoutWarning()
+{
+ CriticalSectionScoped lock(_critSect);
+ _playWarning = 0;
+}
+
+void AudioDeviceLinuxPulse::ClearPlayoutError()
+{
+ CriticalSectionScoped lock(_critSect);
+ _playError = 0;
+}
+
+void AudioDeviceLinuxPulse::ClearRecordingWarning()
+{
+ CriticalSectionScoped lock(_critSect);
+ _recWarning = 0;
+}
+
+void AudioDeviceLinuxPulse::ClearRecordingError()
+{
+ CriticalSectionScoped lock(_critSect);
+ _recError = 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context *c, void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaContextStateCallbackHandler(
+ c);
+}
+
+// ----------------------------------------------------------------------------
+// PaSinkInfoCallback
+// ----------------------------------------------------------------------------
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
+ const pa_sink_info *i, int eol,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaSinkInfoCallbackHandler(
+ i, eol);
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
+ const pa_source_info *i,
+ int eol, void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaSourceInfoCallbackHandler(
+ i, eol);
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context */*c*/,
+ const pa_server_info *i,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaServerInfoCallbackHandler(i);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream *p, void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamStateCallbackHandler(p);
+}
+
+void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context *c)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " context state cb");
+
+ pa_context_state_t state = LATE(pa_context_get_state)(c);
+ switch (state)
+ {
+ case PA_CONTEXT_UNCONNECTED:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " unconnected");
+ break;
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " no state");
+ break;
+ case PA_CONTEXT_FAILED:
+ case PA_CONTEXT_TERMINATED:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " failed");
+ _paStateChanged = true;
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ break;
+ case PA_CONTEXT_READY:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " ready");
+ _paStateChanged = true;
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ break;
+ }
+}
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallbackHandler(const pa_sink_info *i,
+ int eol)
+{
+ if (eol)
+ {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
+
+ if (_numPlayDevices == _deviceIndex)
+ {
+ // Convert the device index to the one of the sink
+ _paDeviceIndex = i->index;
+
+ if (_playDeviceName)
+ {
+ // Copy the sink name
+ strncpy(_playDeviceName, i->name, kAdmMaxDeviceNameSize);
+ _playDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+ if (_playDisplayDeviceName)
+ {
+ // Copy the sink display name
+ strncpy(_playDisplayDeviceName, i->description,
+ kAdmMaxDeviceNameSize);
+ _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+ }
+
+ _numPlayDevices++;
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(
+ const pa_source_info *i,
+ int eol)
+{
+ if (eol)
+ {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
+
+ // We don't want to list output devices
+ if (i->monitor_of_sink == PA_INVALID_INDEX)
+ {
+ if (_numRecDevices == _deviceIndex)
+ {
+ // Convert the device index to the one of the source
+ _paDeviceIndex = i->index;
+
+ if (_recDeviceName)
+ {
+ // copy the source name
+ strncpy(_recDeviceName, i->name, kAdmMaxDeviceNameSize);
+ _recDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+ if (_recDisplayDeviceName)
+ {
+ // Copy the source display name
+ strncpy(_recDisplayDeviceName, i->description,
+ kAdmMaxDeviceNameSize);
+ _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+ }
+
+ _numRecDevices++;
+ }
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(const pa_server_info *i)
+{
+ // Use PA native sampling rate
+ WebRtc_UWord32 paSampleRate = i->sample_spec.rate;
+ if (paSampleRate == 44100)
+ {
+#ifdef WEBRTC_PA_GTALK
+ paSampleRate = 48000;
+#else
+ paSampleRate = 44000;
+#endif
+ }
+
+ _samplingFreq = paSampleRate / 1000;
+
+ // Copy the PA server version
+ if (_paServerVersion)
+ {
+ strncpy(_paServerVersion, i->server_version, 31);
+ _paServerVersion[31] = '\0';
+ }
+
+ if (_recDisplayDeviceName)
+ {
+ // Copy the source name
+ strncpy(_recDisplayDeviceName, i->default_source_name,
+ kAdmMaxDeviceNameSize);
+ _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+
+ if (_playDisplayDeviceName)
+ {
+ // Copy the sink name
+ strncpy(_playDisplayDeviceName, i->default_sink_name,
+ kAdmMaxDeviceNameSize);
+ _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+ }
+
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream *p)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " stream state cb");
+
+ pa_stream_state_t state = LATE(pa_stream_get_state)(p);
+ switch (state)
+ {
+ case PA_STREAM_UNCONNECTED:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " unconnected");
+ break;
+ case PA_STREAM_CREATING:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " creating");
+ break;
+ case PA_STREAM_FAILED:
+ case PA_STREAM_TERMINATED:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " failed");
+ break;
+ case PA_STREAM_READY:
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " ready");
+ break;
+ }
+
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::CheckPulseAudioVersion()
+{
+ /*WebRtc_Word32 index = 0;
+ WebRtc_Word32 partIndex = 0;
+ WebRtc_Word32 partNum = 1;
+ WebRtc_Word32 minVersion[3] = {0, 9, 15};
+ bool versionOk = false;
+ char str[8] = {0};*/
+
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+
+ // get the server info and update deviceName
+ paOperation = LATE(pa_context_get_server_info)(_paContext,
+ PaServerInfoCallback, this);
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
+ " checking PulseAudio version: %s", _paServerVersion);
+
+ /* Saved because it may turn out that we need to check the version in the future
+ while (true)
+ {
+ if (_paServerVersion[index] == '.')
+ {
+ index++;
+ str[partIndex] = '\0';
+ partIndex = 0;
+
+ if(partNum == 2)
+ {
+ if (atoi(str) < minVersion[1])
+ {
+ break;
+ }
+ partNum = 3;
+ }
+ else
+ {
+ if (atoi(str) > minVersion[0])
+ {
+ versionOk = true;
+ break;
+ }
+ partNum = 2;
+ }
+ }
+ else if (_paServerVersion[index] == '\0' || _paServerVersion[index] == '-')
+ {
+ str[partIndex] = '\0';
+ if (atoi(str) >= minVersion[2])
+ {
+ versionOk = true;
+ }
+ break;
+ }
+
+ str[partIndex] = _paServerVersion[index];
+ index++;
+ partIndex++;
+ }
+
+ if (!versionOk)
+ {
+ return -1;
+ }
+ */
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitSamplingFrequency()
+{
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+
+ // Get the server info and update _samplingFreq
+ paOperation = LATE(pa_context_get_server_info)(_paContext,
+ PaServerInfoCallback, this);
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
+ WebRtc_Word8* name,
+ WebRtc_UWord16& index)
+{
+ WebRtc_Word8 tmpName[kAdmMaxDeviceNameSize] = {0};
+ // subtract length of "default: "
+ WebRtc_UWord16 nameLen = kAdmMaxDeviceNameSize - 9;
+ WebRtc_Word8* pName = NULL;
+
+ if (name)
+ {
+ // Add "default: "
+ strcpy(name, "default: ");
+ pName = &name[9];
+ }
+
+ // Tell the callback that we want
+ // the name for this device
+ if (recDevice)
+ {
+ _recDisplayDeviceName = tmpName;
+ } else
+ {
+ _playDisplayDeviceName = tmpName;
+ }
+
+ // Set members
+ _paDeviceIndex = -1;
+ _deviceIndex = 0;
+ _numPlayDevices = 0;
+ _numRecDevices = 0;
+
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+
+ // Get the server info and update deviceName
+ paOperation = LATE(pa_context_get_server_info)(_paContext,
+ PaServerInfoCallback, this);
+
+ WaitForOperationCompletion(paOperation);
+
+ // Get the device index
+ if (recDevice)
+ {
+ paOperation
+ = LATE(pa_context_get_source_info_by_name)(_paContext,
+ (char *) tmpName,
+ PaSourceInfoCallback,
+ this);
+ } else
+ {
+ paOperation
+ = LATE(pa_context_get_sink_info_by_name)(_paContext,
+ (char *) tmpName,
+ PaSinkInfoCallback, this);
+ }
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ // Set the index
+ index = _paDeviceIndex;
+
+ if (name)
+ {
+ // Copy to name string
+ strncpy(pName, tmpName, nameLen);
+ }
+
+ // Clear members
+ _playDisplayDeviceName = NULL;
+ _recDisplayDeviceName = NULL;
+ _paDeviceIndex = -1;
+ _deviceIndex = -1;
+ _numPlayDevices = 0;
+ _numRecDevices = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::InitPulseAudio()
+{
+ int retVal = 0;
+
+ // Load libpulse
+ if (!PaSymbolTable.Load())
+ {
+ // Most likely the Pulse library and sound server are not installed on
+ // this system
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to load symbol table");
+ return -1;
+ }
+
+ // Create a mainloop API and connection to the default server
+ // the mainloop is the internal asynchronous API event loop
+ if (_paMainloop) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PA mainloop has already existed");
+ return -1;
+ }
+ _paMainloop = LATE(pa_threaded_mainloop_new)();
+ if (!_paMainloop)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " could not create mainloop");
+ return -1;
+ }
+
+ // Start the threaded main loop
+ retVal = LATE(pa_threaded_mainloop_start)(_paMainloop);
+ if (retVal != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to start main loop, error=%d", retVal);
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " mainloop running!");
+
+ PaLock();
+
+ _paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop);
+ if (!_paMainloopApi)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " could not create mainloop API");
+ PaUnLock();
+ return -1;
+ }
+
+ // Create a new PulseAudio context
+ if (_paContext){
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PA context has already existed");
+ PaUnLock();
+ return -1;
+ }
+ _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
+
+ if (!_paContext)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " could not create context");
+ PaUnLock();
+ return -1;
+ }
+
+ // Set state callback function
+ LATE(pa_context_set_state_callback)(_paContext, PaContextStateCallback,
+ this);
+
+ // Connect the context to a server (default)
+ _paStateChanged = false;
+ retVal = LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN,
+ NULL);
+
+ if (retVal != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to connect context, error=%d", retVal);
+ PaUnLock();
+ return -1;
+ }
+
+ // Wait for state change
+ while (!_paStateChanged)
+ {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
+
+ // Now check to see what final state we reached.
+ pa_context_state_t state = LATE(pa_context_get_state)(_paContext);
+
+ if (state != PA_CONTEXT_READY)
+ {
+ if (state == PA_CONTEXT_FAILED)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to connect to PulseAudio sound server");
+ } else if (state == PA_CONTEXT_TERMINATED)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PulseAudio connection terminated early");
+ } else
+ {
+ // Shouldn't happen, because we only signal on one of those three
+ // states
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " unknown problem connecting to PulseAudio");
+ }
+ PaUnLock();
+ return -1;
+ }
+
+ PaUnLock();
+
+ // Give the objects to the mixer manager
+ _mixerManager.SetPulseAudioObjects(_paMainloop, _paContext);
+
+ // Check the version
+ if (CheckPulseAudioVersion() < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PulseAudio version %s not supported", _paServerVersion);
+ return -1;
+ }
+
+ // Initialize sampling frequency
+ if (InitSamplingFrequency() < 0 || _samplingFreq == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to initialize sampling frequency, set to %d",
+ _samplingFreq);
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::TerminatePulseAudio()
+{
+ // Do nothing if the instance doesn't exist
+ // likely PaSymbolTable.Load() fails
+ if (!_paMainloop) {
+ return 0;
+ }
+
+ PaLock();
+
+ // Disconnect the context
+ if (_paContext)
+ {
+ LATE(pa_context_disconnect)(_paContext);
+ }
+
+ // Unreference the context
+ if (_paContext)
+ {
+ LATE(pa_context_unref)(_paContext);
+ }
+
+ PaUnLock();
+ _paContext = NULL;
+
+ // Stop the threaded main loop
+ if (_paMainloop)
+ {
+ LATE(pa_threaded_mainloop_stop)(_paMainloop);
+ }
+
+ // Free the mainloop
+ if (_paMainloop)
+ {
+ LATE(pa_threaded_mainloop_free)(_paMainloop);
+ }
+
+ _paMainloop = NULL;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " PulseAudio terminated");
+
+ return 0;
+}
+
+void AudioDeviceLinuxPulse::PaLock()
+{
+ LATE(pa_threaded_mainloop_lock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::PaUnLock()
+{
+ LATE(pa_threaded_mainloop_unlock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::WaitForOperationCompletion(
+ pa_operation* paOperation) const
+{
+ if (!paOperation)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "paOperation NULL in WaitForOperationCompletion");
+ return;
+ }
+
+ while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING)
+ {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
+
+ LATE(pa_operation_unref)(paOperation);
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::EnableWriteCallback()
+{
+ if (LATE(pa_stream_get_state)(_playStream) == PA_STREAM_READY)
+ {
+ // May already have available space. Must check.
+ _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream);
+ if (_tempBufferSpace > 0)
+ {
+ // Yup, there is already space available, so if we register a write
+ // callback then it will not receive any event. So dispatch one ourself
+ // instead
+ _timeEventPlay.Set();
+ return;
+ }
+ }
+
+ LATE(pa_stream_set_write_callback)(_playStream, &PaStreamWriteCallback,
+ this);
+}
+
+void AudioDeviceLinuxPulse::DisableWriteCallback()
+{
+ LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream */*unused*/,
+ size_t buffer_space,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamWriteCallbackHandler(
+ buffer_space);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace)
+{
+ _tempBufferSpace = bufferSpace;
+
+ // Since we write the data asynchronously on a different thread, we have
+ // to temporarily disable the write callback or else Pulse will call it
+ // continuously until we write the data. We re-enable it below.
+ DisableWriteCallback();
+ _timeEventPlay.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream */*unused*/,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamUnderflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
+{
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Playout underflow");
+
+ if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
+ {
+ // We didn't configure a pa_buffer_attr before, so switching to one now
+ // would be questionable.
+ return;
+ }
+
+ // Otherwise reconfigure the stream with a higher target latency.
+
+ const pa_sample_spec *spec = LATE(pa_stream_get_sample_spec)(_playStream);
+ if (!spec)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " pa_stream_get_sample_spec()");
+ return;
+ }
+
+ size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+ WebRtc_UWord32 newLatency = _configuredLatencyPlay + bytesPerSec
+ * WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+
+ // Set the play buffer attributes
+ _playBufferAttr.maxlength = newLatency;
+ _playBufferAttr.tlength = newLatency;
+ _playBufferAttr.minreq = newLatency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+ _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
+
+ pa_operation *op = LATE(pa_stream_set_buffer_attr)(_playStream,
+ &_playBufferAttr, NULL,
+ NULL);
+ if (!op)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " pa_stream_set_buffer_attr()");
+ return;
+ }
+
+ // Don't need to wait for this to complete.
+ LATE(pa_operation_unref)(op);
+
+ // Save the new latency in case we underflow again.
+ _configuredLatencyPlay = newLatency;
+}
+
+void AudioDeviceLinuxPulse::EnableReadCallback()
+{
+ LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this);
+}
+
+void AudioDeviceLinuxPulse::DisableReadCallback()
+{
+ LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream */*unused1*/,
+ size_t /*unused2*/,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamReadCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler()
+{
+ // We get the data pointer and size now in order to save one Lock/Unlock
+ // in the worker thread
+ if (LATE(pa_stream_peek)(_recStream, &_tempSampleData, &_tempSampleDataSize)
+ != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Can't read data!");
+ return;
+ }
+
+ // Since we consume the data asynchronously on a different thread, we have
+ // to temporarily disable the read callback or else Pulse will call it
+ // continuously until we consume the data. We re-enable it below
+ DisableReadCallback();
+ _timeEventRec.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream */*unused*/,
+ void *pThis)
+{
+ static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamOverflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler()
+{
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Recording overflow");
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::LatencyUsecs(pa_stream *stream)
+{
+ if (!WEBRTC_PA_REPORT_LATENCY)
+ {
+ return 0;
+ }
+
+ if (!stream)
+ {
+ return 0;
+ }
+
+ pa_usec_t latency;
+ int negative;
+ if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Can't query latency");
+ // We'd rather continue playout/capture with an incorrect delay than stop
+ // it altogether, so return a valid value.
+ return 0;
+ }
+
+ if (negative)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " warning: pa_stream_get_latency reported negative delay");
+
+ // The delay can be negative for monitoring streams if the captured
+ // samples haven't been played yet. In such a case, "latency" contains the
+ // magnitude, so we must negate it to get the real value.
+ WebRtc_Word32 tmpLatency = (WebRtc_Word32) -latency;
+ if (tmpLatency < 0)
+ {
+ // Make sure that we don't use a negative delay
+ tmpLatency = 0;
+ }
+
+ return tmpLatency;
+ } else
+ {
+ return (WebRtc_Word32) latency;
+ }
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
+ size_t bufferSize)
+{
+ size_t size = bufferSize;
+ WebRtc_UWord32 numRecSamples = _recordBufferSize / (2 * _recChannels);
+
+ // Account for the peeked data and the used data
+ WebRtc_UWord32 recDelay = (WebRtc_UWord32) ((LatencyUsecs(_recStream)
+ / 1000) + 10 * ((size + _recordBufferUsed) / _recordBufferSize));
+
+ _sndCardRecDelay = recDelay;
+
+ if (_playStream)
+ {
+ // Get the playout delay
+ _sndCardPlayDelay = (WebRtc_UWord32) (LatencyUsecs(_playStream) / 1000);
+ }
+
+ if (_recordBufferUsed > 0)
+ {
+ // Have to copy to the buffer until it is full
+ size_t copy = _recordBufferSize - _recordBufferUsed;
+ if (size < copy)
+ {
+ copy = size;
+ }
+
+ memcpy(&_recBuffer[_recordBufferUsed], bufferData, copy);
+ _recordBufferUsed += copy;
+ bufferData = static_cast<const char *> (bufferData) + copy;
+ size -= copy;
+
+ if (_recordBufferUsed != _recordBufferSize)
+ {
+ // Not enough data yet to pass to VoE
+ return 0;
+ }
+
+ // Provide data to VoiceEngine
+ if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1)
+ {
+ // We have stopped recording
+ return -1;
+ }
+
+ _recordBufferUsed = 0;
+ }
+
+ // Now process full 10ms sample sets directly from the input
+ while (size >= _recordBufferSize)
+ {
+ // Provide data to VoiceEngine
+ if (ProcessRecordedData(
+ static_cast<WebRtc_Word8 *> (const_cast<void *> (bufferData)),
+ numRecSamples, recDelay) == -1)
+ {
+ // We have stopped recording
+ return -1;
+ }
+
+ bufferData = static_cast<const char *> (bufferData) + _recordBufferSize;
+ size -= _recordBufferSize;
+
+ // We have consumed 10ms of data
+ recDelay -= 10;
+ }
+
+ // Now save any leftovers for later.
+ if (size > 0)
+ {
+ memcpy(_recBuffer, bufferData, size);
+ _recordBufferUsed = size;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceLinuxPulse::ProcessRecordedData(
+ WebRtc_Word8 *bufferData,
+ WebRtc_UWord32 bufferSizeInSamples,
+ WebRtc_UWord32 recDelay)
+{
+ WebRtc_UWord32 currentMicLevel(0);
+ WebRtc_UWord32 newMicLevel(0);
+
+ _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
+
+ if (AGC())
+ {
+ // Store current mic level in the audio buffer if AGC is enabled
+ if (MicrophoneVolume(currentMicLevel) == 0)
+ {
+ // This call does not affect the actual microphone volume
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ }
+
+ // Set vqe data
+ const WebRtc_UWord32 clockDrift(0);
+ _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay, clockDrift);
+
+ // Deliver recorded samples at specified sample rate,
+ // mic level etc. to the observer using callback
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ Lock();
+
+ // We have been unlocked - check the flag again
+ if (!_recording)
+ {
+ return -1;
+ }
+
+ if (AGC())
+ {
+ newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0)
+ {
+ // The VQE will only deliver non-zero microphone levels when a
+ // change is needed.
+ // Set this new mic level (received from the observer as return
+ // value in the callback).
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+ " AGC change of volume: old=%u => new=%u",
+ currentMicLevel, newMicLevel);
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id,
+ " the required modification of the microphone "
+ "volume failed");
+ }
+ }
+ }
+
+ return 0;
+}
+
+bool AudioDeviceLinuxPulse::PlayThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceLinuxPulse*> (pThis)->PlayThreadProcess());
+}
+
+bool AudioDeviceLinuxPulse::RecThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceLinuxPulse*> (pThis)->RecThreadProcess());
+}
+
+bool AudioDeviceLinuxPulse::PlayThreadProcess()
+{
+ switch (_timeEventPlay.Wait(1000))
+ {
+ case kEventSignaled:
+ _timeEventPlay.Reset();
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "EventWrapper::Wait() failed");
+ return true;
+ case kEventTimeout:
+ return true;
+ }
+
+ Lock();
+
+ if (_startPlay)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "_startPlay true, performing initial actions");
+
+ _startPlay = false;
+ _playDeviceName = NULL;
+
+ // Set if not default device
+ if (_outputDeviceIndex > 0)
+ {
+ // Get the playout device name
+ _playDeviceName = new WebRtc_Word8[kAdmMaxDeviceNameSize];
+ _deviceIndex = _outputDeviceIndex;
+ PlayoutDevices();
+ }
+
+ // Start muted only supported on 0.9.11 and up
+ if (LATE(pa_context_get_protocol_version)(_paContext)
+ >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
+ {
+ // Get the currently saved speaker mute status
+ // and set the initial mute status accordingly
+ bool enabled(false);
+ _mixerManager.SpeakerMute(enabled);
+ if (enabled)
+ {
+ _playStreamFlags |= PA_STREAM_START_MUTED;
+ }
+ }
+
+ // Get the currently saved speaker volume
+ WebRtc_UWord32 volume = 0;
+ _mixerManager.SpeakerVolume(volume);
+
+ PaLock();
+
+ // Set the same volume for all channels
+ pa_cvolume cVolumes;
+ const pa_sample_spec *spec =
+ LATE(pa_stream_get_sample_spec)(_playStream);
+ LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+
+ // Connect the stream to a sink
+ if (LATE(pa_stream_connect_playback)(
+ _playStream,
+ _playDeviceName,
+ &_playBufferAttr,
+ (pa_stream_flags_t) _playStreamFlags,
+ &cVolumes, NULL) != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to connect play stream, err=%d",
+ LATE(pa_context_errno)(_paContext));
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " play stream connected");
+
+ // Wait for state change
+ while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY)
+ {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " play stream ready");
+
+ // We can now handle write callbacks
+ EnableWriteCallback();
+
+ PaUnLock();
+
+ // Clear device name
+ if (_playDeviceName)
+ {
+ delete [] _playDeviceName;
+ _playDeviceName = NULL;
+ }
+
+ _playing = true;
+ _playStartEvent.Set();
+
+ UnLock();
+ return true;
+ }
+
+ if (_playing)
+ {
+ if (!_recording)
+ {
+ // Update the playout delay
+ _sndCardPlayDelay = (WebRtc_UWord32) (LatencyUsecs(_playStream)
+ / 1000);
+ }
+
+ if (_playbackBufferUnused < _playbackBufferSize)
+ {
+
+ size_t write = _playbackBufferSize - _playbackBufferUnused;
+ if (_tempBufferSpace < write)
+ {
+ write = _tempBufferSpace;
+ }
+
+ PaLock();
+ if (LATE(pa_stream_write)(
+ _playStream,
+ (void *) &_playBuffer[_playbackBufferUnused],
+ write, NULL, (int64_t) 0,
+ PA_SEEK_RELATIVE) != PA_OK)
+ {
+ _writeErrors++;
+ if (_writeErrors > 10)
+ {
+ if (_playError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceUtility, _id,
+ " pending playout error exists");
+ }
+ _playError = 1; // Triggers callback from module process thread
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceUtility,
+ _id,
+ " kPlayoutError message posted: "
+ "_writeErrors=%u, error=%d",
+ _writeErrors,
+ LATE(pa_context_errno)(_paContext));
+ _writeErrors = 0;
+ }
+ }
+ PaUnLock();
+
+ _playbackBufferUnused += write;
+ _tempBufferSpace -= write;
+ }
+
+ WebRtc_UWord32 numPlaySamples = _playbackBufferSize / (2
+ * _playChannels);
+ if (_tempBufferSpace > 0) // Might have been reduced to zero by the above
+ {
+ // Ask for new PCM data to be played out using the AudioDeviceBuffer
+ // ensure that this callback is executed without taking the
+ // audio-thread lock
+ UnLock();
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " requesting data");
+ WebRtc_UWord32 nSamples =
+ _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
+ Lock();
+
+ // We have been unlocked - check the flag again
+ if (!_playing)
+ {
+ UnLock();
+ return true;
+ }
+
+ nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
+ if (nSamples != numPlaySamples)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
+ _id, " invalid number of output samples(%d)",
+ nSamples);
+ }
+
+ size_t write = _playbackBufferSize;
+ if (_tempBufferSpace < write)
+ {
+ write = _tempBufferSpace;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " will write");
+ PaLock();
+ if (LATE(pa_stream_write)(_playStream, (void *) &_playBuffer[0],
+ write, NULL, (int64_t) 0,
+ PA_SEEK_RELATIVE) != PA_OK)
+ {
+ _writeErrors++;
+ if (_writeErrors > 10)
+ {
+ if (_playError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceUtility, _id,
+ " pending playout error exists");
+ }
+ _playError = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceUtility,
+ _id,
+ " kPlayoutError message posted: "
+ "_writeErrors=%u, error=%d",
+ _writeErrors,
+ LATE(pa_context_errno)(_paContext));
+ _writeErrors = 0;
+ }
+ }
+ PaUnLock();
+
+ _playbackBufferUnused = write;
+ }
+
+ _tempBufferSpace = 0;
+ PaLock();
+ EnableWriteCallback();
+ PaUnLock();
+
+ } // _playing
+
+ UnLock();
+ return true;
+}
+
+bool AudioDeviceLinuxPulse::RecThreadProcess()
+{
+ switch (_timeEventRec.Wait(1000))
+ {
+ case kEventSignaled:
+ _timeEventRec.Reset();
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "EventWrapper::Wait() failed");
+ return true;
+ case kEventTimeout:
+ return true;
+ }
+
+ Lock();
+
+ if (_startRec)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "_startRec true, performing initial actions");
+
+ _recDeviceName = NULL;
+
+ // Set if not default device
+ if (_inputDeviceIndex > 0)
+ {
+ // Get the recording device name
+ _recDeviceName = new WebRtc_Word8[kAdmMaxDeviceNameSize];
+ _deviceIndex = _inputDeviceIndex;
+ RecordingDevices();
+ }
+
+ PaLock();
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " connecting stream");
+
+ // Connect the stream to a source
+ if (LATE(pa_stream_connect_record)(_recStream, _recDeviceName,
+ &_recBufferAttr,
+ (pa_stream_flags_t) _recStreamFlags)
+ != PA_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to connect rec stream, err=%d",
+ LATE(pa_context_errno)(_paContext));
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " connected");
+
+ // Wait for state change
+ while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY)
+ {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " done");
+
+ // We can now handle read callbacks
+ EnableReadCallback();
+
+ PaUnLock();
+
+ // Clear device name
+ if (_recDeviceName)
+ {
+ delete [] _recDeviceName;
+ _recDeviceName = NULL;
+ }
+
+ _startRec = false;
+ _recording = true;
+ _recStartEvent.Set();
+
+ UnLock();
+ return true;
+ }
+
+ if (_recording)
+ {
+ // Read data and provide it to VoiceEngine
+ if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1)
+ {
+ UnLock();
+ return true;
+ }
+
+ _tempSampleData = NULL;
+ _tempSampleDataSize = 0;
+
+ PaLock();
+ while (true)
+ {
+ // Ack the last thing we read
+ if (LATE(pa_stream_drop)(_recStream) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, " failed to drop, err=%d\n",
+ LATE(pa_context_errno)(_paContext));
+ }
+
+ if (LATE(pa_stream_readable_size)(_recStream) <= 0)
+ {
+ // Then that was all the data
+ break;
+ }
+
+ // Else more data.
+ const void *sampleData;
+ size_t sampleDataSize;
+
+ if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize)
+ != 0)
+ {
+ _recError = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
+ _id, " RECORD_ERROR message posted, error = %d",
+ LATE(pa_context_errno)(_paContext));
+ break;
+ }
+
+ _sndCardRecDelay = (WebRtc_UWord32) (LatencyUsecs(_recStream)
+ / 1000);
+
+ // Drop lock for sigslot dispatch, which could take a while.
+ PaUnLock();
+ // Read data and provide it to VoiceEngine
+ if (ReadRecordedData(sampleData, sampleDataSize) == -1)
+ {
+ UnLock();
+ return true;
+ }
+ PaLock();
+
+ // Return to top of loop for the ack and the check for more data.
+ }
+
+ EnableReadCallback();
+ PaUnLock();
+
+ } // _recording
+
+ UnLock();
+ return true;
+}
+
+}
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h
new file mode 100644
index 0000000..693a32b
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h
@@ -0,0 +1,385 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
+
+#include "audio_device_generic.h"
+#include "audio_mixer_manager_pulse_linux.h"
+#include "critical_section_wrapper.h"
+
+#include <pulse/pulseaudio.h>
+
+// Set this define to make the code behave like in GTalk/libjingle
+//#define WEBRTC_PA_GTALK
+
+// We define this flag if it's missing from our headers, because we want to be
+// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
+// if run against a recent version of the library.
+#ifndef PA_STREAM_ADJUST_LATENCY
+#define PA_STREAM_ADJUST_LATENCY 0x2000U
+#endif
+#ifndef PA_STREAM_START_MUTED
+#define PA_STREAM_START_MUTED 0x1000U
+#endif
+
+// Set this constant to 0 to disable latency reading
+const WebRtc_UWord32 WEBRTC_PA_REPORT_LATENCY = 1;
+
+// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
+
+// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
+const WebRtc_UWord32 WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
+
+// Some timing constants for optimal operation. See
+// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
+// for a good explanation of some of the factors that go into this.
+
+// Playback.
+
+// For playback, there is a round-trip delay to fill the server-side playback
+// buffer, so setting too low of a latency is a buffer underflow risk. We will
+// automatically increase the latency if a buffer underflow does occur, but we
+// also enforce a sane minimum at start-up time. Anything lower would be
+// virtually guaranteed to underflow at least once, so there's no point in
+// allowing lower latencies.
+const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
+
+// Every time a playback stream underflows, we will reconfigure it with target
+// latency that is greater by this amount.
+const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
+
+// We also need to configure a suitable request size. Too small and we'd burn
+// CPU from the overhead of transfering small amounts of data at once. Too large
+// and the amount of data remaining in the buffer right before refilling it
+// would be a buffer underflow risk. We set it to half of the buffer size.
+const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
+
+// Capture.
+
+// For capture, low latency is not a buffer overflow risk, but it makes us burn
+// CPU from the overhead of transfering small amounts of data at once, so we set
+// a recommended value that we use for the kLowLatency constant (but if the user
+// explicitly requests something lower then we will honour it).
+// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
+const WebRtc_UWord32 WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
+
+// There is a round-trip delay to ack the data to the server, so the
+// server-side buffer needs extra space to prevent buffer overflow. 20ms is
+// sufficient, but there is no penalty to making it bigger, so we make it huge.
+// (750ms is libpulse's default value for the _total_ buffer size in the
+// kNoLatencyRequirements case.)
+const WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
+
+const WebRtc_UWord32 WEBRTC_PA_MSECS_PER_SEC = 1000;
+
+// Init _configuredLatencyRec/Play to this value to disable latency requirements
+const WebRtc_Word32 WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
+
+// Set this const to 1 to account for peeked and used data in latency calculation
+const WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
+
+namespace webrtc
+{
+class EventWrapper;
+class ThreadWrapper;
+
+class AudioDeviceLinuxPulse: public AudioDeviceGeneric
+{
+public:
+ AudioDeviceLinuxPulse(const WebRtc_Word32 id);
+ ~AudioDeviceLinuxPulse();
+
+ static bool PulseAudioIsSupported();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32
+ ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32
+ SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+public:
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private:
+ void Lock()
+ {
+ _critSect.Enter();
+ }
+ ;
+ void UnLock()
+ {
+ _critSect.Leave();
+ }
+ ;
+ void WaitForOperationCompletion(pa_operation* paOperation) const;
+ void WaitForSuccess(pa_operation* paOperation) const;
+
+private:
+ static void PaContextStateCallback(pa_context *c, void *pThis);
+ static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
+ int eol, void *pThis);
+ static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
+ int eol, void *pThis);
+ static void PaServerInfoCallback(pa_context *c, const pa_server_info *i,
+ void *pThis);
+ static void PaStreamStateCallback(pa_stream *p, void *pThis);
+ void PaContextStateCallbackHandler(pa_context *c);
+ void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
+ void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
+ void PaServerInfoCallbackHandler(const pa_server_info *i);
+ void PaStreamStateCallbackHandler(pa_stream *p);
+
+ void EnableWriteCallback();
+ void DisableWriteCallback();
+ static void PaStreamWriteCallback(pa_stream *unused, size_t buffer_space,
+ void *pThis);
+ void PaStreamWriteCallbackHandler(size_t buffer_space);
+ static void PaStreamUnderflowCallback(pa_stream *unused, void *pThis);
+ void PaStreamUnderflowCallbackHandler();
+ void EnableReadCallback();
+ void DisableReadCallback();
+ static void PaStreamReadCallback(pa_stream *unused1, size_t unused2,
+ void *pThis);
+ void PaStreamReadCallbackHandler();
+ static void PaStreamOverflowCallback(pa_stream *unused, void *pThis);
+ void PaStreamOverflowCallbackHandler();
+ WebRtc_Word32 LatencyUsecs(pa_stream *stream);
+ WebRtc_Word32 ReadRecordedData(const void* bufferData, size_t bufferSize);
+ WebRtc_Word32 ProcessRecordedData(WebRtc_Word8 *bufferData,
+ WebRtc_UWord32 bufferSizeInSamples,
+ WebRtc_UWord32 recDelay);
+
+ WebRtc_Word32 CheckPulseAudioVersion();
+ WebRtc_Word32 InitSamplingFrequency();
+ WebRtc_Word32 GetDefaultDeviceInfo(bool recDevice, WebRtc_Word8* name,
+ WebRtc_UWord16& index);
+ WebRtc_Word32 InitPulseAudio();
+ WebRtc_Word32 TerminatePulseAudio();
+
+ void PaLock();
+ void PaUnLock();
+
+ static bool RecThreadFunc(void*);
+ static bool PlayThreadFunc(void*);
+ bool RecThreadProcess();
+ bool PlayThreadProcess();
+
+private:
+ AudioDeviceBuffer* _ptrAudioBuffer;
+
+ CriticalSectionWrapper& _critSect;
+ EventWrapper& _timeEventRec;
+ EventWrapper& _timeEventPlay;
+ EventWrapper& _recStartEvent;
+ EventWrapper& _playStartEvent;
+
+ ThreadWrapper* _ptrThreadPlay;
+ ThreadWrapper* _ptrThreadRec;
+ WebRtc_UWord32 _recThreadID;
+ WebRtc_UWord32 _playThreadID;
+ WebRtc_Word32 _id;
+
+ AudioMixerManagerLinuxPulse _mixerManager;
+
+ WebRtc_UWord16 _inputDeviceIndex;
+ WebRtc_UWord16 _outputDeviceIndex;
+ bool _inputDeviceIsSpecified;
+ bool _outputDeviceIsSpecified;
+
+ WebRtc_UWord32 _samplingFreq;
+ WebRtc_UWord8 _recChannels;
+ WebRtc_UWord8 _playChannels;
+
+ AudioDeviceModule::BufferType _playBufType;
+
+private:
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _startRec;
+ bool _stopRec;
+ bool _startPlay;
+ bool _stopPlay;
+ bool _AGC;
+
+private:
+ WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
+
+ WebRtc_UWord32 _sndCardPlayDelay;
+ WebRtc_UWord32 _sndCardRecDelay;
+
+ WebRtc_Word32 _writeErrors;
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ WebRtc_UWord16 _deviceIndex;
+ WebRtc_Word16 _numPlayDevices;
+ WebRtc_Word16 _numRecDevices;
+ WebRtc_Word8* _playDeviceName;
+ WebRtc_Word8* _recDeviceName;
+ WebRtc_Word8* _playDisplayDeviceName;
+ WebRtc_Word8* _recDisplayDeviceName;
+ WebRtc_Word8 _paServerVersion[32];
+
+ WebRtc_Word8* _playBuffer;
+ size_t _playbackBufferSize;
+ size_t _playbackBufferUnused;
+ size_t _tempBufferSpace;
+ WebRtc_Word8* _recBuffer;
+ size_t _recordBufferSize;
+ size_t _recordBufferUsed;
+ const void* _tempSampleData;
+ size_t _tempSampleDataSize;
+ WebRtc_Word32 _configuredLatencyPlay;
+ WebRtc_Word32 _configuredLatencyRec;
+
+ // PulseAudio
+ WebRtc_UWord16 _paDeviceIndex;
+ bool _paStateChanged;
+
+ pa_threaded_mainloop* _paMainloop;
+ pa_mainloop_api* _paMainloopApi;
+ pa_context* _paContext;
+
+ pa_stream* _recStream;
+ pa_stream* _playStream;
+ WebRtc_UWord32 _recStreamFlags;
+ WebRtc_UWord32 _playStreamFlags;
+ pa_buffer_attr _playBufferAttr;
+ pa_buffer_attr _recBufferAttr;
+};
+
+}
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc
new file mode 100644
index 0000000..0b9dd54
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility_linux.h"
+#include "audio_device_config.h" // DEBUG_PRINT()
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+AudioDeviceUtilityLinux::AudioDeviceUtilityLinux(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
+ _lastError(AudioDeviceModule::kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+}
+
+AudioDeviceUtilityLinux::~AudioDeviceUtilityLinux()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ // free stuff here...
+ }
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+
+WebRtc_Word32 AudioDeviceUtilityLinux::Init()
+{
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " OS info: %s", "Linux");
+
+ return 0;
+}
+
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h
new file mode 100644
index 0000000..8df7acc
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
+
+#include "audio_device_utility.h"
+#include "audio_device.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class AudioDeviceUtilityLinux: public AudioDeviceUtility
+{
+public:
+ AudioDeviceUtilityLinux(const WebRtc_Word32 id);
+ ~AudioDeviceUtilityLinux();
+
+ virtual WebRtc_Word32 Init();
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ AudioDeviceModule::ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc
new file mode 100644
index 0000000..306ad5d
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc
@@ -0,0 +1,1317 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "audio_mixer_manager_alsa_linux.h"
+#include "trace.h"
+
+extern webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libalsa, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym) \
+ LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
+
+namespace webrtc
+{
+
+AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _outputMixerHandle(NULL),
+ _inputMixerHandle(NULL),
+ _outputMixerElement(NULL),
+ _inputMixerElement(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s constructed", __FUNCTION__);
+
+ memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+ memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+}
+
+AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destructed", __FUNCTION__);
+
+ Close();
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// PUBLIC METHODS
+// ============================================================================
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::Close()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ CloseSpeaker();
+ CloseMicrophone();
+
+ return 0;
+
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::CloseSpeaker()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ int errVal = 0;
+
+ if (_outputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing playout mixer");
+ LATE(snd_mixer_free)(_outputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error freeing playout mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error detachinging playout mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error snd_mixer_close(handleMixer) errVal=%d",
+ errVal);
+ }
+ _outputMixerHandle = NULL;
+ _outputMixerElement = NULL;
+ }
+ memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::CloseMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ int errVal = 0;
+
+ if (_inputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer");
+
+ LATE(snd_mixer_free)(_inputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error freeing record mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer 2");
+
+ errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error detachinging record mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer 3");
+
+ errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error snd_mixer_close(handleMixer) errVal=%d",
+ errVal);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer 4");
+ _inputMixerHandle = NULL;
+ _inputMixerElement = NULL;
+ }
+ memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::OpenSpeaker(name=%s)", deviceName);
+
+ CriticalSectionScoped lock(_critSect);
+
+ int errVal = 0;
+
+ // Close any existing output mixer handle
+ //
+ if (_outputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing playout mixer");
+
+ LATE(snd_mixer_free)(_outputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error freeing playout mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error detachinging playout mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error snd_mixer_close(handleMixer) errVal=%d",
+ errVal);
+ }
+ }
+ _outputMixerHandle = NULL;
+ _outputMixerElement = NULL;
+
+ errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "snd_mixer_open(&_outputMixerHandle, 0) - error");
+ return -1;
+ }
+
+ char controlName[kAdmMaxDeviceNameSize] = { 0 };
+ GetControlName(controlName, deviceName);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " snd_mixer_attach(_outputMixerHandle, %s)", controlName);
+
+ errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_attach(_outputMixerHandle, %s) error: %s",
+ controlName, LATE(snd_strerror)(errVal));
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+ strcpy(_outputMixerStr, controlName);
+
+ errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_selem_register(_outputMixerHandle,"
+ " NULL, NULL), error: %s",
+ LATE(snd_strerror)(errVal));
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+
+ // Load and find the proper mixer element
+ if (LoadSpeakerMixerElement() < 0)
+ {
+ return -1;
+ }
+
+ if (_outputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " the output mixer device is now open (0x%x)",
+ _outputMixerHandle);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::OpenMicrophone(char *deviceName)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::OpenMicrophone(name=%s)",
+ deviceName);
+
+ CriticalSectionScoped lock(_critSect);
+
+ int errVal = 0;
+
+ // Close any existing input mixer handle
+ //
+ if (_inputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer");
+
+ LATE(snd_mixer_free)(_inputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error freeing record mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer");
+
+ errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error detachinging record mixer: %s",
+ LATE(snd_strerror)(errVal));
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer");
+
+ errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error snd_mixer_close(handleMixer) errVal=%d",
+ errVal);
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Closing record mixer");
+ }
+ _inputMixerHandle = NULL;
+ _inputMixerElement = NULL;
+
+ errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_open(&_inputMixerHandle, 0) - error");
+ return -1;
+ }
+
+ char controlName[kAdmMaxDeviceNameSize] = { 0 };
+ GetControlName(controlName, deviceName);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " snd_mixer_attach(_inputMixerHandle, %s)", controlName);
+
+ errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_attach(_inputMixerHandle, %s) error: %s",
+ controlName, LATE(snd_strerror)(errVal));
+
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+ strcpy(_inputMixerStr, controlName);
+
+ errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_selem_register(_inputMixerHandle,"
+ " NULL, NULL), error: %s",
+ LATE(snd_strerror)(errVal));
+
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+ // Load and find the proper mixer element
+ if (LoadMicMixerElement() < 0)
+ {
+ return -1;
+ }
+
+ if (_inputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " the input mixer device is now open (0x%x)",
+ _inputMixerHandle);
+ }
+
+ return 0;
+}
+
+bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ return (_outputMixerHandle != NULL);
+}
+
+bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ return (_inputMixerHandle != NULL);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SetSpeakerVolume(
+ WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume=%u)",
+ volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ int errVal =
+ LATE(snd_mixer_selem_set_playback_volume_all)(_outputMixerElement,
+ volume);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error changing master volume: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ return (0);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolume(
+ WebRtc_UWord32& volume) const
+{
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int vol(0);
+
+ int
+ errVal = LATE(snd_mixer_selem_get_playback_volume)(
+ _outputMixerElement,
+ (snd_mixer_selem_channel_id_t) 0,
+ &vol);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Error getting outputvolume: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxALSA::SpeakerVolume() => vol=%i",
+ vol);
+
+ volume = static_cast<WebRtc_UWord32> (vol);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avilable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal =
+ LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
+ &minVol, &maxVol);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout hardware volume range, min: %d, max: %d",
+ minVol, maxVol);
+
+ if (maxVol <= minVol)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting get_playback_volume_range: %s",
+ LATE(snd_strerror)(errVal));
+ }
+
+ maxVolume = static_cast<WebRtc_UWord32> (maxVol);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MinSpeakerVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal =
+ LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
+ &minVol, &maxVol);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout hardware volume range, min: %d, max: %d",
+ minVol, maxVol);
+
+ if (maxVol <= minVol)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting get_playback_volume_range: %s",
+ LATE(snd_strerror)(errVal));
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (minVol);
+
+ return 0;
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ // SetMaxSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMaxSpeakerVolume(
+ WebRtc_UWord32 maxVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Error getting playback volume range: %s", snd_strerror(errVal));
+ }
+
+ maxVol = maxVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout hardware volume range, min: %d, max: %d", minVol, maxVol);
+ if (errVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting playback volume range: %s", snd_strerror(errVal));
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ // SetMinSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMinSpeakerVolume(
+ WebRtc_UWord32 minVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Error getting playback volume range: %s", snd_strerror(errVal));
+ }
+
+ minVol = minVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Playout hardware volume range, min: %d, max: %d", minVol, maxVol);
+ if (errVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting playback volume range: %s", snd_strerror(errVal));
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer exists");
+ return -1;
+ }
+
+ // The step size is always 1 for ALSA
+ stepSize = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(
+ bool& available)
+{
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(
+ bool& available)
+{
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ // Ensure that the selected speaker destination has a valid mute control.
+ bool available(false);
+ SpeakerMuteIsAvailable(available);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to mute the speaker");
+ return -1;
+ }
+
+ // Note value = 0 (off) means muted
+ int errVal =
+ LATE(snd_mixer_selem_set_playback_switch_all)(_outputMixerElement,
+ !enable);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting playback switch: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ return (0);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const
+{
+
+ if (_outputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected speaker destination has a valid mute control.
+ bool available =
+ LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to mute the speaker");
+ return -1;
+ }
+
+ int value(false);
+
+ // Retrieve one boolean control value for a specified mute-control
+ //
+ int
+ errVal = LATE(snd_mixer_selem_get_playback_switch)(
+ _outputMixerElement,
+ (snd_mixer_selem_channel_id_t) 0,
+ &value);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting playback switch: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ // Note value = 0 (off) means muted
+ enabled = (bool) !value;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(
+ bool& available)
+{
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ // Ensure that the selected microphone destination has a valid mute control.
+ bool available(false);
+ MicrophoneMuteIsAvailable(available);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to mute the microphone");
+ return -1;
+ }
+
+ // Note value = 0 (off) means muted
+ int errVal =
+ LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement,
+ !enable);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting capture switch: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ return (0);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const
+{
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected microphone destination has a valid mute control.
+ bool available =
+ LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to mute the microphone");
+ return -1;
+ }
+
+ int value(false);
+
+ // Retrieve one boolean control value for a specified mute-control
+ //
+ int
+ errVal = LATE(snd_mixer_selem_get_capture_switch)(
+ _inputMixerElement,
+ (snd_mixer_selem_channel_id_t) 0,
+ &value);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting capture switch: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ // Note value = 0 (off) means muted
+ enabled = (bool) !value;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneBoostIsAvailable(
+ bool& available)
+{
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Microphone boost cannot be enabled through ALSA Simple Mixer Interface
+ available = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneBoost(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::SetMicrophoneBoost(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected microphone destination has a valid mute control.
+ bool available(false);
+ MicrophoneMuteIsAvailable(available);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to enable microphone boost");
+ return -1;
+ }
+
+ // It is assumed that the call above fails!
+
+ return (0);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneBoost(bool& enabled) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Microphone boost cannot be enabled on this platform!
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
+ bool& available)
+{
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneVolume(
+ WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=%u)",
+ volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ int
+ errVal =
+ LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement,
+ volume);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error changing microphone volume: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+
+ return (0);
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ // SetMaxMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMaxMicrophoneVolume(
+ WebRtc_UWord32 maxVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(_inputMixerElement,
+ &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Error getting capture volume range: %s", snd_strerror(errVal));
+ }
+
+ maxVol = (long int)maxVolume;
+ printf("min %d max %d", minVol, maxVol);
+ errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol, maxVol);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Capture hardware volume range, min: %d, max: %d", minVol, maxVol);
+ if (errVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting capture volume range: %s", snd_strerror(errVal));
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ // SetMinMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMinMicrophoneVolume(
+ WebRtc_UWord32 minVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable output mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(
+ _inputMixerElement, &minVol, &maxVol);
+ if (maxVol <= minVol)
+ {
+ //maxVol = 255;
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Error getting capture volume range: %s", snd_strerror(errVal));
+ }
+
+ printf("min %d max %d", minVol, maxVol);
+ minVol = (long int)minVolume;
+ errVal = snd_mixer_selem_set_capture_volume_range(
+ _inputMixerElement, minVol, maxVol);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Capture hardware volume range, min: %d, max: %d", minVol, maxVol);
+ if (errVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error setting capture volume range: %s", snd_strerror(errVal));
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolume(
+ WebRtc_UWord32& volume) const
+{
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ long int vol(0);
+
+ int
+ errVal =
+ LATE(snd_mixer_selem_get_capture_volume)(
+ _inputMixerElement,
+ (snd_mixer_selem_channel_id_t) 0,
+ &vol);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Error getting inputvolume: %s",
+ LATE(snd_strerror)(errVal));
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol=%i",
+ vol);
+
+ volume = static_cast<WebRtc_UWord32> (vol);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
+ WebRtc_UWord32& maxVolume) const
+{
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ // check if we have mic volume at all
+ if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " No microphone volume available");
+ return -1;
+ }
+
+ int errVal =
+ LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
+ &minVol, &maxVol);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Microphone hardware volume range, min: %d, max: %d",
+ minVol, maxVol);
+ if (maxVol <= minVol)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting microphone volume range: %s",
+ LATE(snd_strerror)(errVal));
+ }
+
+ maxVolume = static_cast<WebRtc_UWord32> (maxVol);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
+ WebRtc_UWord32& minVolume) const
+{
+
+ if (_inputMixerElement == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer element exists");
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal =
+ LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
+ &minVol, &maxVol);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Microphone hardware volume range, min: %d, max: %d",
+ minVol, maxVol);
+ if (maxVol <= minVol)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error getting microphone volume range: %s",
+ LATE(snd_strerror)(errVal));
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (minVol);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " no avaliable input mixer exists");
+ return -1;
+ }
+
+ // The step size is always 1 for ALSA
+ stepSize = 1;
+
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::LoadMicMixerElement() const
+{
+ int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "snd_mixer_load(_inputMixerHandle), error: %s",
+ LATE(snd_strerror)(errVal));
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+
+ snd_mixer_elem_t *elem = NULL;
+ snd_mixer_elem_t *micElem = NULL;
+ unsigned mixerIdx = 0;
+ const char *selemName = NULL;
+
+ // Find and store handles to the right mixer elements
+ for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem; elem
+ = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
+ {
+ if (LATE(snd_mixer_selem_is_active)(elem))
+ {
+ selemName = LATE(snd_mixer_selem_get_name)(elem);
+ if (strcmp(selemName, "Capture") == 0) // "Capture", "Mic"
+ {
+ _inputMixerElement = elem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, " Capture element set");
+ } else if (strcmp(selemName, "Mic") == 0)
+ {
+ micElem = elem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, " Mic element found");
+ }
+ }
+
+ if (_inputMixerElement)
+ {
+ // Use the first Capture element that is found
+ // The second one may not work
+ break;
+ }
+ }
+
+ if (_inputMixerElement == NULL)
+ {
+ // We didn't find a Capture handle, use Mic.
+ if (micElem != NULL)
+ {
+ _inputMixerElement = micElem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Using Mic as capture volume.");
+ } else
+ {
+ _inputMixerElement = NULL;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Could not find capture volume on the mixer.");
+
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const
+{
+ int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
+ if (errVal < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " snd_mixer_load(_outputMixerHandle), error: %s",
+ LATE(snd_strerror)(errVal));
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+
+ snd_mixer_elem_t *elem = NULL;
+ snd_mixer_elem_t *masterElem = NULL;
+ snd_mixer_elem_t *speakerElem = NULL;
+ unsigned mixerIdx = 0;
+ const char *selemName = NULL;
+
+ // Find and store handles to the right mixer elements
+ for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem; elem
+ = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
+ {
+ if (LATE(snd_mixer_selem_is_active)(elem))
+ {
+ selemName = LATE(snd_mixer_selem_get_name)(elem);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "snd_mixer_selem_get_name %d: %s =%x", mixerIdx,
+ selemName, elem);
+
+ // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
+ if (strcmp(selemName, "PCM") == 0)
+ {
+ _outputMixerElement = elem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, " PCM element set");
+ } else if (strcmp(selemName, "Master") == 0)
+ {
+ masterElem = elem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, " Master element found");
+ } else if (strcmp(selemName, "Speaker") == 0)
+ {
+ speakerElem = elem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, " Speaker element found");
+ }
+ }
+
+ if (_outputMixerElement)
+ {
+ // We have found the element we want
+ break;
+ }
+ }
+
+ // If we didn't find a PCM Handle, use Master or Speaker
+ if (_outputMixerElement == NULL)
+ {
+ if (masterElem != NULL)
+ {
+ _outputMixerElement = masterElem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Using Master as output volume.");
+ } else if (speakerElem != NULL)
+ {
+ _outputMixerElement = speakerElem;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Using Speaker as output volume.");
+ } else
+ {
+ _outputMixerElement = NULL;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Could not find output volume in the mixer.");
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
+ char* deviceName) const
+{
+ // Example
+ // deviceName: "front:CARD=Intel,DEV=0"
+ // controlName: "hw:CARD=Intel"
+ char* pos1 = strchr(deviceName, ':');
+ char* pos2 = strchr(deviceName, ',');
+ if (!pos2)
+ {
+ // Can also be default:CARD=Intel
+ pos2 = &deviceName[strlen(deviceName)];
+ }
+ if (pos1 && pos2)
+ {
+ strcpy(controlName, "hw");
+ int nChar = (int) (pos2 - pos1);
+ strncpy(&controlName[2], pos1, nChar);
+ controlName[2 + nChar] = '\0';
+ } else
+ {
+ strcpy(controlName, deviceName);
+ }
+
+}
+
+}
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h
new file mode 100644
index 0000000..94ea982
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
+
+#include "typedefs.h"
+#include "audio_device.h"
+#include "critical_section_wrapper.h"
+#include "alsasymboltable_linux.h"
+
+#include <alsa/asoundlib.h>
+
+namespace webrtc
+{
+
+class AudioMixerManagerLinuxALSA
+{
+public:
+ WebRtc_Word32 OpenSpeaker(char* deviceName);
+ WebRtc_Word32 OpenMicrophone(char* deviceName);
+ WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetSpeakerMute(bool enable);
+ WebRtc_Word32 SpeakerMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneMute(bool enable);
+ WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+ WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 Close();
+ WebRtc_Word32 CloseSpeaker();
+ WebRtc_Word32 CloseMicrophone();
+ bool SpeakerIsInitialized() const;
+ bool MicrophoneIsInitialized() const;
+
+public:
+ AudioMixerManagerLinuxALSA(const WebRtc_Word32 id);
+ ~AudioMixerManagerLinuxALSA();
+
+private:
+ WebRtc_Word32 LoadMicMixerElement() const;
+ WebRtc_Word32 LoadSpeakerMixerElement() const;
+ void GetControlName(char *controlName, char* deviceName) const;
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ mutable snd_mixer_t* _outputMixerHandle;
+ char _outputMixerStr[kAdmMaxDeviceNameSize];
+ mutable snd_mixer_t* _inputMixerHandle;
+ char _inputMixerStr[kAdmMaxDeviceNameSize];
+ mutable snd_mixer_elem_t* _outputMixerElement;
+ mutable snd_mixer_elem_t* _inputMixerElement;
+};
+
+}
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc
new file mode 100644
index 0000000..01f5172
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc
@@ -0,0 +1,1271 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "audio_mixer_manager_pulse_linux.h"
+#include "trace.h"
+
+extern webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym) \
+ LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, sym)
+
+namespace webrtc
+{
+
+enum { kMaxRetryOnFailure = 2 };
+
+AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _paOutputDeviceIndex(-1),
+ _paInputDeviceIndex(-1),
+ _paPlayStream(NULL),
+ _paRecStream(NULL),
+ _paMainloop(NULL),
+ _paContext(NULL),
+ _paVolume(0),
+ _paMute(0),
+ _paVolSteps(0),
+ _paSpeakerMute(false),
+ _paSpeakerVolume(0),
+ _paChannels(0),
+ _paObjectsSet(false),
+ _callbackValues(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s constructed", __FUNCTION__);
+}
+
+AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destructed", __FUNCTION__);
+
+ Close();
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// PUBLIC METHODS
+// ============================================================================
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
+ pa_threaded_mainloop* mainloop,
+ pa_context* context)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!mainloop || !context)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " could not set PulseAudio objects for mixer");
+ return -1;
+ }
+
+ _paMainloop = mainloop;
+ _paContext = context;
+ _paObjectsSet = true;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " the PulseAudio objects for the mixer has been set");
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::Close()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ CloseSpeaker();
+ CloseMicrophone();
+
+ _paMainloop = NULL;
+ _paContext = NULL;
+ _paObjectsSet = false;
+
+ return 0;
+
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::CloseSpeaker()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Reset the index to -1
+ _paOutputDeviceIndex = -1;
+ _paPlayStream = NULL;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::CloseMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Reset the index to -1
+ _paInputDeviceIndex = -1;
+ _paRecStream = NULL;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)");
+
+ CriticalSectionScoped lock(_critSect);
+ _paPlayStream = playStream;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetRecStream(recStream)");
+
+ CriticalSectionScoped lock(_critSect);
+ _paRecStream = recStream;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::OpenSpeaker(
+ WebRtc_UWord16 deviceIndex)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=%d)",
+ deviceIndex);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // No point in opening the speaker
+ // if PA objects have not been set
+ if (!_paObjectsSet)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PulseAudio objects has not been set");
+ return -1;
+ }
+
+ // Set the index for the PulseAudio
+ // output device to control
+ _paOutputDeviceIndex = deviceIndex;
+
+ // Init the speaker volume to the normal volume
+ _paSpeakerVolume = PA_VOLUME_NORM;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " the output mixer device is now open");
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::OpenMicrophone(
+ WebRtc_UWord16 deviceIndex)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex=%d)",
+ deviceIndex);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // No point in opening the microphone
+ // if PA objects have not been set
+ if (!_paObjectsSet)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " PulseAudio objects have not been set");
+ return -1;
+ }
+
+ // Set the index for the PulseAudio
+ // input device to control
+ _paInputDeviceIndex = deviceIndex;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " the input mixer device is now open");
+
+ return 0;
+}
+
+bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ return (_paOutputDeviceIndex != -1);
+}
+
+bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ return (_paInputDeviceIndex != -1);
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetSpeakerVolume(
+ WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume=%u)",
+ volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ bool setFailed(false);
+
+ if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ // We can only really set the volume if we have a connected stream
+ PaLock();
+
+ // Get the number of channels from the sample specification
+ const pa_sample_spec *spec =
+ LATE(pa_stream_get_sample_spec)(_paPlayStream);
+ if (!spec)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " could not get sample specification");
+ PaUnLock();
+ return -1;
+ }
+
+ // Set the same volume for all channels
+ pa_cvolume cVolumes;
+ LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+
+ pa_operation* paOperation = NULL;
+ paOperation = LATE(pa_context_set_sink_input_volume)(
+ _paContext,
+ LATE(pa_stream_get_index)(_paPlayStream),
+ &cVolumes,
+ PaSetVolumeCallback, NULL);
+ if (!paOperation)
+ {
+ setFailed = true;
+ }
+
+ // Don't need to wait for the completion
+ LATE(pa_operation_unref)(paOperation);
+
+ PaUnLock();
+ } else
+ {
+ // We have not created a stream or it's not connected to the sink
+ // Save the volume to be set at connection
+ _paSpeakerVolume = volume;
+ }
+
+ if (setFailed)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " could not set speaker volume, error%d",
+ LATE(pa_context_errno)(_paContext));
+
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ // We can only get the volume if we have a connected stream
+ if (!GetSinkInputInfo())
+ return -1;
+
+ volume = static_cast<WebRtc_UWord32> (_paVolume);
+ ResetCallbackVariables();
+ } else
+ {
+ volume = _paSpeakerVolume;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::SpeakerVolume() => vol=%i",
+ volume);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ // PA_VOLUME_NORM corresponds to 100% (0db)
+ // but PA allows up to 150 db amplification
+ maxVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_NORM);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_MUTED);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ // The sink input (stream) will always have step size = 1
+ // There are PA_VOLUME_NORM+1 steps
+ stepSize = 1;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize() => "
+ "size=%i, stepSize");
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
+{
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available)
+{
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ bool setFailed(false);
+
+ if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ // We can only really mute if we have a connected stream
+ PaLock();
+
+ pa_operation* paOperation = NULL;
+ paOperation = LATE(pa_context_set_sink_input_mute)(
+ _paContext,
+ LATE(pa_stream_get_index)(_paPlayStream),
+ (int) enable,
+ PaSetVolumeCallback,
+ NULL);
+ if (!paOperation)
+ {
+ setFailed = true;
+ }
+
+ // Don't need to wait for the completion
+ LATE(pa_operation_unref)(paOperation);
+
+ PaUnLock();
+ } else
+ {
+ // We have not created a stream or it's not connected to the sink
+ // Save the mute status to be set at connection
+ _paSpeakerMute = enable;
+ }
+
+ if (setFailed)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " could not mute speaker, error%d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const
+{
+
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ // We can only get the mute status if we have a connected stream
+ if (!GetSinkInputInfo())
+ return -1;
+
+ enabled = static_cast<bool> (_paMute);
+ ResetCallbackVariables();
+ } else
+ {
+ enabled = _paSpeakerMute;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::SpeakerMute() => "
+ "enabled=%i, enabled");
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available)
+{
+ if (_paOutputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " output device index has not been set");
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t) _paOutputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
+ }
+
+ PaUnLock();
+
+ if (!GetSinkInfoByIndex(deviceIndex))
+ return -1;
+
+ available = static_cast<bool> (_paChannels == 2);
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
+{
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ // Get info for this source
+ // We want to know if the actual device can record in stereo
+ paOperation = LATE(pa_context_get_source_info_by_index)(
+ _paContext, deviceIndex,
+ PaSourceInfoCallback,
+ (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+ PaUnLock();
+
+ if (!_callbackValues)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Error getting number of input channels: %d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ available = static_cast<bool> (_paChannels == 2);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+ " => available=%i, available");
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
+ bool& available)
+{
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ bool setFailed(false);
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ // Set mute switch for the source
+ paOperation = LATE(pa_context_set_source_mute_by_index)(
+ _paContext, deviceIndex,
+ enable,
+ PaSetVolumeCallback, NULL);
+
+ if (!paOperation)
+ {
+ setFailed = true;
+ }
+
+ // Don't need to wait for this to complete.
+ LATE(pa_operation_unref)(paOperation);
+
+ PaUnLock();
+
+ // Reset variables altered by callback
+ ResetCallbackVariables();
+
+ if (setFailed)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " could not mute microphone, error%d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ PaUnLock();
+
+ if (!GetSourceInfoByIndex(deviceIndex))
+ return -1;
+
+ enabled = static_cast<bool> (_paMute);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::MicrophoneMute() =>"
+ " enabled=%i, enabled");
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
+{
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Always unavailable in Pulse Audio
+ // Could make it possible to use PA_VOLUME_MAX
+ // but that gives bad audio with some sound cards
+ available = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::SetMicrophoneBoost(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetMicrophoneBoost(enable=%u)",
+ enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Ensure that the selected microphone destination has a valid boost control
+ bool available(false);
+ MicrophoneBoostIsAvailable(available);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to enable microphone boost");
+ return -1;
+ }
+
+ // It is assumed that the call above fails!
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneBoost(bool& enabled) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Microphone boost cannot be enabled on this platform!
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
+ bool& available)
+{
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=%u)",
+ volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // Unlike output streams, input streams have no concept of a stream volume,
+ // only a device volume. So we have to change the volume of the device
+ // itself.
+
+ // The device may have a different number of channels than the stream and
+ // their mapping may be different, so we don't want to use the channel count
+ // from our sample spec. We could use PA_CHANNELS_MAX to cover our bases,
+ // and the server allows that even if the device's channel count is lower,
+ // but some buggy PA clients don't like that (the pavucontrol on Hardy dies
+ // in an assert if the channel count is different). So instead we look up
+ // the actual number of channels that the device has.
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ bool setFailed(false);
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ // Get the number of channels for this source
+ paOperation
+ = LATE(pa_context_get_source_info_by_index)(_paContext, deviceIndex,
+ PaSourceInfoCallback,
+ (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+
+ if (!_callbackValues)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Error getting input channels: %d",
+ LATE(pa_context_errno)(_paContext));
+ PaUnLock();
+ return -1;
+ }
+
+ WebRtc_UWord8 channels = _paChannels;
+ ResetCallbackVariables();
+
+ pa_cvolume cVolumes;
+ LATE(pa_cvolume_set)(&cVolumes, channels, volume);
+
+ // Set the volume for the source
+ paOperation
+ = LATE(pa_context_set_source_volume_by_index)(_paContext, deviceIndex,
+ &cVolumes,
+ PaSetVolumeCallback, NULL);
+
+ if (!paOperation)
+ {
+ setFailed = true;
+ }
+
+ // Don't need to wait for this to complete.
+ LATE(pa_operation_unref)(paOperation);
+
+ PaUnLock();
+
+ // Reset variables altered by callback
+ ResetCallbackVariables();
+
+ if (setFailed)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " could not set microphone volume, error%d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ PaUnLock();
+
+ if (!GetSourceInfoByIndex(deviceIndex))
+ return -1;
+
+ volume = static_cast<WebRtc_UWord32> (_paVolume);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=%i, volume");
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ // PA_VOLUME_NORM corresponds to 100% (0db)
+ // PA allows up to 150 db amplification (PA_VOLUME_MAX)
+ // but that doesn't work well for all sound cards
+ maxVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_NORM);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerLinuxPulse::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_MUTED);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
+ WebRtc_UWord16& stepSize) const
+{
+
+ if (_paInputDeviceIndex == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " input device index has not been set");
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+
+ PaLock();
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+ != PA_STREAM_UNCONNECTED))
+ {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ // Get info for this source
+ paOperation
+ = LATE(pa_context_get_source_info_by_index)(_paContext, deviceIndex,
+ PaSourceInfoCallback,
+ (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+
+ PaUnLock();
+
+ if (!_callbackValues)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Error getting step size: %d",
+ LATE(pa_context_errno)(_paContext));
+ return -1;
+ }
+
+ stepSize = static_cast<WebRtc_UWord16> ((PA_VOLUME_NORM + 1) / _paVolSteps);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize()"
+ " => size=%i, stepSize");
+
+ // Reset members modified by callback
+ ResetCallbackVariables();
+
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
+ const pa_sink_info *i,
+ int eol, void *pThis)
+{
+ static_cast<AudioMixerManagerLinuxPulse*> (pThis)-> PaSinkInfoCallbackHandler(
+ i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+ pa_context */*c*/,
+ const pa_sink_input_info *i,
+ int eol, void *pThis)
+{
+ static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
+ PaSinkInputInfoCallbackHandler(i, eol);
+}
+
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
+ const pa_source_info *i,
+ int eol, void *pThis)
+{
+ static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
+ PaSourceInfoCallbackHandler(i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context * c,
+ int success, void */*pThis*/)
+{
+ if (!success)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ " failed to set volume");
+ }
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
+ const pa_sink_info *i,
+ int eol)
+{
+ if (eol)
+ {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
+
+ _callbackValues = true;
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j)
+ {
+ if (paVolume < i->volume.values[j])
+ {
+ paVolume = i->volume.values[j];
+ }
+ }
+ _paVolume = paVolume; // get the max volume for any channel
+ _paMute = i->mute; // get mute status
+
+ // supported since PA 0.9.15
+ //_paVolSteps = i->n_volume_steps; // get the number of volume steps
+ // default value is PA_VOLUME_NORM+1
+ _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
+ const pa_sink_input_info *i,
+ int eol)
+{
+ if (eol)
+ {
+ // Signal that we are done.
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
+
+ _callbackValues = true;
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j)
+ {
+ if (paVolume < i->volume.values[j])
+ {
+ paVolume = i->volume.values[j];
+ }
+ }
+ _paVolume = paVolume; // Get the max volume for any channel
+ _paMute = i->mute; // Get mute status
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
+ const pa_source_info *i,
+ int eol)
+{
+ if (eol)
+ {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
+
+ _callbackValues = true;
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j)
+ {
+ if (paVolume < i->volume.values[j])
+ {
+ paVolume = i->volume.values[j];
+ }
+ }
+ _paVolume = paVolume; // Get the max volume for any channel
+ _paMute = i->mute; // Get mute status
+
+ // supported since PA 0.9.15
+ //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
+ // default value is PA_VOLUME_NORM+1
+ _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::ResetCallbackVariables() const
+{
+ _paVolume = 0;
+ _paMute = 0;
+ _paVolSteps = 0;
+ _paChannels = 0;
+ _callbackValues = false;
+}
+
+void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
+ pa_operation* paOperation) const
+{
+ while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING)
+ {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
+
+ LATE(pa_operation_unref)(paOperation);
+}
+
+void AudioMixerManagerLinuxPulse::PaLock() const
+{
+ LATE(pa_threaded_mainloop_lock)(_paMainloop);
+}
+
+void AudioMixerManagerLinuxPulse::PaUnLock() const
+{
+ LATE(pa_threaded_mainloop_unlock)(_paMainloop);
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ PaLock();
+ for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
+ retries ++) {
+ // Get info for this stream (sink input).
+ paOperation = LATE(pa_context_get_sink_input_info)(
+ _paContext,
+ LATE(pa_stream_get_index)(_paPlayStream),
+ PaSinkInputInfoCallback,
+ (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+ }
+ PaUnLock();
+
+ if (!_callbackValues) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetSinkInputInfo failed to get volume info : %d",
+ LATE(pa_context_errno)(_paContext));
+ return false;
+ }
+
+ return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(
+ int device_index) const {
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ PaLock();
+ for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
+ retries ++) {
+ paOperation = LATE(pa_context_get_sink_info_by_index)(_paContext,
+ device_index, PaSinkInfoCallback, (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+ }
+ PaUnLock();
+
+ if (!_callbackValues) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetSinkInfoByIndex failed to get volume info: %d",
+ LATE(pa_context_errno)(_paContext));
+ return false;
+ }
+
+ return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(
+ int device_index) const {
+ pa_operation* paOperation = NULL;
+ ResetCallbackVariables();
+
+ PaLock();
+ for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
+ retries ++) {
+ paOperation = LATE(pa_context_get_source_info_by_index)(
+ _paContext, device_index, PaSourceInfoCallback, (void*) this);
+
+ WaitForOperationCompletion(paOperation);
+ }
+
+ PaUnLock();
+
+ if (!_callbackValues) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "GetSourceInfoByIndex error: %d",
+ LATE(pa_context_errno)(_paContext));
+ return false;
+ }
+
+ return true;
+}
+
+}
+
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h
new file mode 100644
index 0000000..22d6da5
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H
+
+#include "typedefs.h"
+#include "audio_device.h"
+#include "critical_section_wrapper.h"
+#include "pulseaudiosymboltable_linux.h"
+
+#include <stdint.h>
+#include <pulse/pulseaudio.h>
+
+#ifndef UINT32_MAX
+#define UINT32_MAX ((uint32_t)-1)
+#endif
+
+namespace webrtc
+{
+
+class AudioMixerManagerLinuxPulse
+{
+public:
+ WebRtc_Word32 SetPlayStream(pa_stream* playStream);
+ WebRtc_Word32 SetRecStream(pa_stream* recStream);
+ WebRtc_Word32 OpenSpeaker(WebRtc_UWord16 deviceIndex);
+ WebRtc_Word32 OpenMicrophone(WebRtc_UWord16 deviceIndex);
+ WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetSpeakerMute(bool enable);
+ WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ WebRtc_Word32 SpeakerMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneMute(bool enable);
+ WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+ WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
+ pa_context* context);
+ WebRtc_Word32 Close();
+ WebRtc_Word32 CloseSpeaker();
+ WebRtc_Word32 CloseMicrophone();
+ bool SpeakerIsInitialized() const;
+ bool MicrophoneIsInitialized() const;
+
+public:
+ AudioMixerManagerLinuxPulse(const WebRtc_Word32 id);
+ ~AudioMixerManagerLinuxPulse();
+
+private:
+ static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
+ int eol, void *pThis);
+ static void PaSinkInputInfoCallback(pa_context *c,
+ const pa_sink_input_info *i, int eol,
+ void *pThis);
+ static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
+ int eol, void *pThis);
+ static void
+ PaSetVolumeCallback(pa_context * /*c*/, int success, void */*pThis*/);
+ void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
+ void PaSinkInputInfoCallbackHandler(const pa_sink_input_info *i, int eol);
+ void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
+
+ void ResetCallbackVariables() const;
+ void WaitForOperationCompletion(pa_operation* paOperation) const;
+ void PaLock() const;
+ void PaUnLock() const;
+
+ bool GetSinkInputInfo() const;
+ bool GetSinkInfoByIndex(int device_index)const ;
+ bool GetSourceInfoByIndex(int device_index) const;
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ WebRtc_Word16 _paOutputDeviceIndex;
+ WebRtc_Word16 _paInputDeviceIndex;
+
+ pa_stream* _paPlayStream;
+ pa_stream* _paRecStream;
+
+ pa_threaded_mainloop* _paMainloop;
+ pa_context* _paContext;
+
+ mutable WebRtc_UWord32 _paVolume;
+ mutable WebRtc_UWord32 _paMute;
+ mutable WebRtc_UWord32 _paVolSteps;
+ bool _paSpeakerMute;
+ mutable WebRtc_UWord32 _paSpeakerVolume;
+ mutable WebRtc_UWord8 _paChannels;
+ bool _paObjectsSet;
+ mutable bool _callbackValues;
+
+ WebRtc_UWord8 _micVolChannels;
+ WebRtc_UWord8 _spkVolChannels;
+};
+
+}
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc
new file mode 100644
index 0000000..8f3c7c8
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc
@@ -0,0 +1,116 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "latebindingsymboltable_linux.h"
+
+#ifdef WEBRTC_LINUX
+#include <dlfcn.h>
+#endif
+
+// TODO(grunell): Either put inside webrtc namespace or use webrtc:: instead.
+using namespace webrtc;
+
+namespace webrtc_adm_linux {
+
+inline static const char *GetDllError() {
+#ifdef WEBRTC_LINUX
+ char *err = dlerror();
+ if (err) {
+ return err;
+ } else {
+ return "No error";
+ }
+#else
+#error Not implemented
+#endif
+}
+
+DllHandle InternalLoadDll(const char dll_name[]) {
+#ifdef WEBRTC_LINUX
+ DllHandle handle = dlopen(dll_name, RTLD_NOW);
+#else
+#error Not implemented
+#endif
+ if (handle == kInvalidDllHandle) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
+ "Can't load %s : %d", dll_name, GetDllError());
+ }
+ return handle;
+}
+
+void InternalUnloadDll(DllHandle handle) {
+#ifdef WEBRTC_LINUX
+ if (dlclose(handle) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "%d", GetDllError());
+ }
+#else
+#error Not implemented
+#endif
+}
+
+static bool LoadSymbol(DllHandle handle,
+ const char *symbol_name,
+ void **symbol) {
+#ifdef WEBRTC_LINUX
+ *symbol = dlsym(handle, symbol_name);
+ char *err = dlerror();
+ if (err) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Error loading symbol %s : %d", symbol_name, err);
+ return false;
+ } else if (!*symbol) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "Symbol %s is NULL", symbol_name);
+ return false;
+ }
+ return true;
+#else
+#error Not implemented
+#endif
+}
+
+// This routine MUST assign SOME value for every symbol, even if that value is
+// NULL, or else some symbols may be left with uninitialized data that the
+// caller may later interpret as a valid address.
+bool InternalLoadSymbols(DllHandle handle,
+ int num_symbols,
+ const char *const symbol_names[],
+ void *symbols[]) {
+#ifdef WEBRTC_LINUX
+ // Clear any old errors.
+ dlerror();
+#endif
+ for (int i = 0; i < num_symbols; ++i) {
+ if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
+ return false;
+ }
+ }
+ return true;
+}
+
+} // namespace webrtc_adm_linux
diff --git a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h
new file mode 100644
index 0000000..91d25aa
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h
@@ -0,0 +1,195 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H
+
+#include <assert.h>
+#include <stddef.h> // for NULL
+#include <string.h>
+
+#include "constructor_magic.h"
+#include "trace.h"
+
+// This file provides macros for creating "symbol table" classes to simplify the
+// dynamic loading of symbols from DLLs. Currently the implementation only
+// supports Linux and pure C symbols.
+// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
+
+namespace webrtc_adm_linux {
+
+#ifdef WEBRTC_LINUX
+typedef void *DllHandle;
+
+const DllHandle kInvalidDllHandle = NULL;
+#else
+#error Not implemented
+#endif
+
+// These are helpers for use only by the class below.
+DllHandle InternalLoadDll(const char dll_name[]);
+
+void InternalUnloadDll(DllHandle handle);
+
+bool InternalLoadSymbols(DllHandle handle,
+ int num_symbols,
+ const char *const symbol_names[],
+ void *symbols[]);
+
+template <int SYMBOL_TABLE_SIZE,
+ const char kDllName[],
+ const char *const kSymbolNames[]>
+class LateBindingSymbolTable {
+ public:
+ LateBindingSymbolTable()
+ : handle_(kInvalidDllHandle),
+ undefined_symbols_(false) {
+ memset(symbols_, 0, sizeof(symbols_));
+ }
+
+ ~LateBindingSymbolTable() {
+ Unload();
+ }
+
+ static int NumSymbols() {
+ return SYMBOL_TABLE_SIZE;
+ }
+
+ // We do not use this, but we offer it for theoretical convenience.
+ static const char *GetSymbolName(int index) {
+ assert(index < NumSymbols());
+ return kSymbolNames[index];
+ }
+
+ bool IsLoaded() const {
+ return handle_ != kInvalidDllHandle;
+ }
+
+ // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
+ // table loaded successfully.
+ bool Load() {
+ if (IsLoaded()) {
+ return true;
+ }
+ if (undefined_symbols_) {
+ // We do not attempt to load again because repeated attempts are not
+ // likely to succeed and DLL loading is costly.
+ //WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ // "We know there are undefined symbols");
+ return false;
+ }
+ handle_ = InternalLoadDll(kDllName);
+ if (!IsLoaded()) {
+ return false;
+ }
+ if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
+ undefined_symbols_ = true;
+ Unload();
+ return false;
+ }
+ return true;
+ }
+
+ void Unload() {
+ if (!IsLoaded()) {
+ return;
+ }
+ InternalUnloadDll(handle_);
+ handle_ = kInvalidDllHandle;
+ memset(symbols_, 0, sizeof(symbols_));
+ }
+
+ // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
+ // instead of this.
+ void *GetSymbol(int index) const {
+ assert(IsLoaded());
+ assert(index < NumSymbols());
+ return symbols_[index];
+ }
+
+ private:
+ DllHandle handle_;
+ bool undefined_symbols_;
+ void *symbols_[SYMBOL_TABLE_SIZE];
+
+ DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
+};
+
+// This macro must be invoked in a header to declare a symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) \
+enum {
+
+// This macro must be invoked in the header declaration once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This macro defines an enum with names built from the symbols, which
+// essentially creates a hash table in the compiler from symbol names to their
+// indices in the symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
+ ClassName##_SYMBOL_TABLE_INDEX_##sym,
+
+// This macro completes the header declaration.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName) \
+ ClassName##_SYMBOL_TABLE_SIZE \
+}; \
+\
+extern const char ClassName##_kDllName[]; \
+extern const char *const \
+ ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
+\
+typedef ::webrtc_adm_linux::LateBindingSymbolTable<ClassName##_SYMBOL_TABLE_SIZE, \
+ ClassName##_kDllName, \
+ ClassName##_kSymbolNames> \
+ ClassName;
+
+// This macro must be invoked in a .cc file to define a previously-declared
+// symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
+const char ClassName##_kDllName[] = dllName; \
+const char *const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
+
+// This macro must be invoked in the .cc definition once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This would have to use the mangled name if we were to ever support C++
+// symbols.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) \
+ #sym,
+
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
+};
+
+// Index of a given symbol in the given symbol table class.
+#define LATESYM_INDEXOF(ClassName, sym) \
+ (ClassName##_SYMBOL_TABLE_INDEX_##sym)
+
+// Returns a reference to the given late-binded symbol, with the correct type.
+#define LATESYM_GET(ClassName, inst, sym) \
+ (*reinterpret_cast<typeof(&sym)>( \
+ (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
+
+} // namespace webrtc_adm_linux
+
+#endif // WEBRTC_ADM_LATEBINDINGSYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc
new file mode 100644
index 0000000..ae663f7
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc
@@ -0,0 +1,39 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "pulseaudiosymboltable_linux.h"
+
+namespace webrtc_adm_linux_pulse {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
+#define X(sym) \
+ LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
+
+} // namespace webrtc_adm_linux_pulse
diff --git a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h
new file mode 100644
index 0000000..049509b
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h
@@ -0,0 +1,104 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
+
+#include "latebindingsymboltable_linux.h"
+
+namespace webrtc_adm_linux_pulse {
+
+// The PulseAudio symbols we need, as an X-Macro list.
+// This list must contain precisely every libpulse function that is used in
+// the ADM LINUX PULSE Device and Mixer classes
+#define PULSE_AUDIO_SYMBOLS_LIST \
+ X(pa_bytes_per_second) \
+ X(pa_context_connect) \
+ X(pa_context_disconnect) \
+ X(pa_context_errno) \
+ X(pa_context_get_protocol_version) \
+ X(pa_context_get_server_info) \
+ X(pa_context_get_sink_info_list) \
+ X(pa_context_get_sink_info_by_index) \
+ X(pa_context_get_sink_info_by_name) \
+ X(pa_context_get_sink_input_info) \
+ X(pa_context_get_source_info_by_index) \
+ X(pa_context_get_source_info_by_name) \
+ X(pa_context_get_source_info_list) \
+ X(pa_context_get_state) \
+ X(pa_context_new) \
+ X(pa_context_set_sink_input_volume) \
+ X(pa_context_set_sink_input_mute) \
+ X(pa_context_set_source_volume_by_index) \
+ X(pa_context_set_source_mute_by_index) \
+ X(pa_context_set_state_callback) \
+ X(pa_context_unref) \
+ X(pa_cvolume_set) \
+ X(pa_operation_get_state) \
+ X(pa_operation_unref) \
+ X(pa_stream_connect_playback) \
+ X(pa_stream_connect_record) \
+ X(pa_stream_disconnect) \
+ X(pa_stream_drop) \
+ X(pa_stream_get_device_index) \
+ X(pa_stream_get_index) \
+ X(pa_stream_get_latency) \
+ X(pa_stream_get_sample_spec) \
+ X(pa_stream_get_state) \
+ X(pa_stream_new) \
+ X(pa_stream_peek) \
+ X(pa_stream_readable_size) \
+ X(pa_stream_set_buffer_attr) \
+ X(pa_stream_set_overflow_callback) \
+ X(pa_stream_set_read_callback) \
+ X(pa_stream_set_state_callback) \
+ X(pa_stream_set_underflow_callback) \
+ X(pa_stream_set_write_callback) \
+ X(pa_stream_unref) \
+ X(pa_stream_writable_size) \
+ X(pa_stream_write) \
+ X(pa_strerror) \
+ X(pa_threaded_mainloop_free) \
+ X(pa_threaded_mainloop_get_api) \
+ X(pa_threaded_mainloop_lock) \
+ X(pa_threaded_mainloop_new) \
+ X(pa_threaded_mainloop_signal) \
+ X(pa_threaded_mainloop_start) \
+ X(pa_threaded_mainloop_stop) \
+ X(pa_threaded_mainloop_unlock) \
+ X(pa_threaded_mainloop_wait)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
+#define X(sym) \
+ LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
+
+} // namespace webrtc_adm_linux_pulse
+
+#endif // WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc
new file mode 100644
index 0000000..e927ae7
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc
@@ -0,0 +1,3241 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility.h"
+#include "audio_device_mac.h"
+#include "audio_device_config.h"
+
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+
+#include <cassert>
+
+#include <sys/sysctl.h> // sysctlbyname()
+#include <mach/mach.h> // mach_task_self()
+#include <libkern/OSAtomic.h> // OSAtomicCompareAndSwap()
+#include "portaudio/pa_ringbuffer.h"
+
+namespace webrtc
+{
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceError, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ return -1; \
+ } \
+ } while(0)
+
+#define WEBRTC_CA_LOG_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceError, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ } \
+ } while(0)
+
+#define WEBRTC_CA_LOG_WARN(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceWarning, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ } \
+ } while(0)
+
+enum
+{
+ MaxNumberDevices = 64
+};
+
+void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
+{
+ while (1)
+ {
+ int32_t oldValue = *theValue;
+ if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
+ == true)
+ {
+ return;
+ }
+ }
+}
+
+int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
+{
+ while (1)
+ {
+ WebRtc_Word32 value = *theValue;
+ if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
+ {
+ return value;
+ }
+ }
+}
+
+// CoreAudio errors are best interpreted as four character strings.
+void AudioDeviceMac::logCAMsg(const TraceLevel level,
+ const TraceModule module,
+ const WebRtc_Word32 id, const char *msg,
+ const char *err)
+{
+ assert(msg != NULL);
+ assert(err != NULL);
+
+#ifdef WEBRTC_BIG_ENDIAN
+ WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+#else
+ // We need to flip the characters in this case.
+ WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
+ + 2, err + 1, err);
+#endif
+}
+
+AudioDeviceMac::AudioDeviceMac(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _stopEventRec(*EventWrapper::Create()),
+ _stopEvent(*EventWrapper::Create()),
+ _captureWorkerThread(NULL),
+ _renderWorkerThread(NULL),
+ _captureWorkerThreadId(0),
+ _renderWorkerThreadId(0),
+ _id(id),
+ _mixerManager(id),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _inputDeviceID(kAudioObjectUnknown),
+ _outputDeviceID(kAudioObjectUnknown),
+ _inputDeviceIsSpecified(false),
+ _outputDeviceIsSpecified(false),
+ _recChannels(N_REC_CHANNELS),
+ _playChannels(N_PLAY_CHANNELS),
+ _captureBufData(NULL),
+ _renderBufData(NULL),
+ _playBufType(AudioDeviceModule::kFixedBufferSize),
+ _initialized(false),
+ _isShutDown(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _startRec(false),
+ _stopRec(false),
+ _stopPlay(false),
+ _AGC(false),
+ _renderDeviceIsAlive(1),
+ _captureDeviceIsAlive(1),
+ _twoDevices(true),
+ _doStop(false),
+ _doStopRec(false),
+ _macBookPro(false),
+ _macBookProPanRight(false),
+ _captureLatencyUs(0),
+ _renderLatencyUs(0),
+ _captureDelayUs(0),
+ _renderDelayUs(0),
+ _renderDelayOffsetSamples(0),
+ _playBufDelayFixed(20),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _paCaptureBuffer(NULL),
+ _paRenderBuffer(NULL),
+ _captureBufSizeSamples(0),
+ _renderBufSizeSamples(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+
+ assert(&_stopEvent != NULL);
+ assert(&_stopEventRec != NULL);
+
+ memset(_renderConvertData, 0, sizeof(_renderConvertData));
+ memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+ memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+ memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+ memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+}
+
+
+AudioDeviceMac::~AudioDeviceMac()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+
+ if (!_isShutDown)
+ {
+ Terminate();
+ }
+
+ if (_captureWorkerThread)
+ {
+ delete _captureWorkerThread;
+ _captureWorkerThread = NULL;
+ }
+
+ if (_renderWorkerThread)
+ {
+ delete _renderWorkerThread;
+ _renderWorkerThread = NULL;
+ }
+
+ if (_paRenderBuffer)
+ {
+ delete _paRenderBuffer;
+ _paRenderBuffer = NULL;
+ }
+
+ if (_paCaptureBuffer)
+ {
+ delete _paCaptureBuffer;
+ _paCaptureBuffer = NULL;
+ }
+
+ if (_renderBufData)
+ {
+ delete[] _renderBufData;
+ _renderBufData = NULL;
+ }
+
+ if (_captureBufData)
+ {
+ delete[] _captureBufData;
+ _captureBufData = NULL;
+ }
+
+ kern_return_t kernErr = KERN_SUCCESS;
+ kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_destroy() error: %d", kernErr);
+ }
+
+ kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_destroy() error: %d", kernErr);
+ }
+
+ delete &_stopEvent;
+ delete &_stopEventRec;
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // inform the AudioBuffer about default settings for this implementation
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+ _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+WebRtc_Word32 AudioDeviceMac::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ OSStatus err = noErr;
+
+ _isShutDown = false;
+
+ // PortAudio ring buffers require an elementCount which is a power of two.
+ if (_renderBufData == NULL)
+ {
+ UInt32 powerOfTwo = 1;
+ while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
+ {
+ powerOfTwo <<= 1;
+ }
+ _renderBufSizeSamples = powerOfTwo;
+ _renderBufData = new SInt16[_renderBufSizeSamples];
+ }
+
+ if (_paRenderBuffer == NULL)
+ {
+ _paRenderBuffer = new PaUtilRingBuffer;
+ ring_buffer_size_t bufSize = -1;
+ bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
+ _renderBufSizeSamples,
+ _renderBufData);
+ if (bufSize == -1)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, " PaUtil_InitializeRingBuffer() error");
+ return -1;
+ }
+ }
+
+ if (_captureBufData == NULL)
+ {
+ UInt32 powerOfTwo = 1;
+ while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
+ {
+ powerOfTwo <<= 1;
+ }
+ _captureBufSizeSamples = powerOfTwo;
+ _captureBufData = new Float32[_captureBufSizeSamples];
+ }
+
+ if (_paCaptureBuffer == NULL)
+ {
+ _paCaptureBuffer = new PaUtilRingBuffer;
+ ring_buffer_size_t bufSize = -1;
+ bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
+ sizeof(Float32),
+ _captureBufSizeSamples,
+ _captureBufData);
+ if (bufSize == -1)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, " PaUtil_InitializeRingBuffer() error");
+ return -1;
+ }
+ }
+
+ if (_renderWorkerThread == NULL)
+ {
+ _renderWorkerThread
+ = ThreadWrapper::CreateThread(RunRender, this, kRealtimePriority,
+ "RenderWorkerThread");
+ if (_renderWorkerThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, " Render CreateThread() error");
+ return -1;
+ }
+ }
+
+ if (_captureWorkerThread == NULL)
+ {
+ _captureWorkerThread
+ = ThreadWrapper::CreateThread(RunCapture, this, kRealtimePriority,
+ "CaptureWorkerThread");
+ if (_captureWorkerThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
+ _id, " Capture CreateThread() error");
+ return -1;
+ }
+ }
+
+ kern_return_t kernErr = KERN_SUCCESS;
+ kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
+ SYNC_POLICY_FIFO, 0);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " semaphore_create() error: %d", kernErr);
+ return -1;
+ }
+
+ kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
+ SYNC_POLICY_FIFO, 0);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ " semaphore_create() error: %d", kernErr);
+ return -1;
+ }
+
+ // Setting RunLoop to NULL here instructs HAL to manage its own thread for
+ // notifications. This was the default behaviour on OS X 10.5 and earlier, but now
+ // must be explicitly specified. HAL would otherwise try to use the main thread to
+ // issue notifications.
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioHardwarePropertyRunLoop,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ CFRunLoopRef runLoop = NULL;
+ UInt32 size = sizeof(CFRunLoopRef);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
+ &propertyAddress, 0, NULL, size, &runLoop));
+
+ // Listen for any device changes.
+ propertyAddress.mSelector = kAudioHardwarePropertyDevices;
+ WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
+ &propertyAddress, &objectListenerProc, this));
+
+ // Determine if this is a MacBook Pro
+ _macBookPro = false;
+ _macBookProPanRight = false;
+ char buf[128];
+ size_t length = sizeof(buf);
+ memset(buf, 0, length);
+
+ int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
+ if (intErr != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error in sysctlbyname(): %d", err);
+ } else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Hardware model: %s", buf);
+ if (strncmp(buf, "MacBookPro", 10) == 0)
+ {
+ _macBookPro = true;
+ }
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ _initialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::Terminate()
+{
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ if (_recording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording must be stopped");
+ return -1;
+ }
+
+ if (_playing)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Playback must be stopped");
+ return -1;
+ }
+
+ _critSect.Enter();
+
+ _mixerManager.Close();
+
+ OSStatus err = noErr;
+ int retVal = 0;
+
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
+ &propertyAddress, &objectListenerProc, this));
+
+ err = AudioHardwareUnload();
+ if (err != noErr)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Error in AudioHardwareUnload()", (const char*) &err);
+ retVal = -1;
+ }
+
+ _critSect.Leave();
+
+ _isShutDown = true;
+ _initialized = false;
+ _outputDeviceIsSpecified = false;
+ _inputDeviceIsSpecified = false;
+
+ return retVal;
+}
+
+bool AudioDeviceMac::Initialized() const
+{
+ return (_initialized);
+}
+
+WebRtc_Word32 AudioDeviceMac::SpeakerIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a valid speaker exists
+ //
+ available = true;
+
+ // Close the initialized output mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
+ {
+ return -1;
+ }
+
+ if (_inputDeviceID == _outputDeviceID)
+ {
+ _twoDevices = false;
+ } else
+ {
+ _twoDevices = true;
+ }
+
+ if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a valid microphone exists
+ //
+ available = true;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+
+WebRtc_Word32 AudioDeviceMac::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
+ {
+ return -1;
+ }
+
+ if (_inputDeviceID == _outputDeviceID)
+ {
+ _twoDevices = false;
+ } else
+ {
+ _twoDevices = true;
+ }
+
+ if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+bool AudioDeviceMac::SpeakerIsInitialized() const
+{
+ return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceMac::MicrophoneIsInitialized() const
+{
+ return (_mixerManager.MicrophoneIsInitialized());
+}
+
+WebRtc_Word32 AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a volume control exists
+ //
+ available = true;
+
+ // Close the initialized output mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+WebRtc_Word32 AudioDeviceMac::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.SpeakerVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ volume = level;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight)
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32
+AudioDeviceMac::WaveOutVolume(WebRtc_UWord16& /*volumeLeft*/,
+ WebRtc_UWord16& /*volumeRight*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceMac::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinSpeakerVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+ return 0;
+}
+
+WebRtc_Word32
+AudioDeviceMac::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected speaker has a mute control
+ //
+ _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetSpeakerMute(bool enable)
+{
+ return (_mixerManager.SetSpeakerMute(enable));
+}
+
+WebRtc_Word32 AudioDeviceMac::SpeakerMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.SpeakerMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a mute control
+ //
+ _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetMicrophoneMute(bool enable)
+{
+ return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.MicrophoneMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Enumerate all avaliable microphone and make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a boost control
+ //
+ _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetMicrophoneBoost(bool enable)
+{
+
+ return (_mixerManager.SetMicrophoneBoost(enable));
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneBoost(bool& enabled) const
+{
+
+ bool onOff(0);
+
+ if (_mixerManager.MicrophoneBoost(onOff) == -1)
+ {
+ return -1;
+ }
+
+ enabled = onOff;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // Cannot open the specified device
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone can record stereo
+ //
+ _mixerManager.StereoRecordingIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetStereoRecording(bool enable)
+{
+
+ if (enable)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StereoRecording(bool& enabled) const
+{
+
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ if (!wasInitialized && InitSpeaker() == -1)
+ {
+ // Cannot open the specified device
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone can record stereo
+ //
+ _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetStereoPlayout(bool enable)
+{
+
+ if (enable)
+ _playChannels = 2;
+ else
+ _playChannels = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StereoPlayout(bool& enabled) const
+{
+
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetAGC(bool enable)
+{
+
+ _AGC = enable;
+
+ return 0;
+}
+
+bool AudioDeviceMac::AGC() const
+{
+
+ return _AGC;
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
+{
+
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control.
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a volume control
+ // exists
+ //
+ available = true;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized)
+ {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+WebRtc_Word32 AudioDeviceMac::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.MicrophoneVolume(level) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " failed to retrive current microphone level");
+ return -1;
+ }
+
+ volume = level;
+ return 0;
+}
+
+WebRtc_Word32
+AudioDeviceMac::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+ return 0;
+}
+
+WebRtc_Word32
+AudioDeviceMac::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+ return 0;
+}
+
+WebRtc_Word32
+AudioDeviceMac::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+ return 0;
+}
+
+WebRtc_Word16 AudioDeviceMac::PlayoutDevices()
+{
+
+ AudioDeviceID playDevices[MaxNumberDevices];
+ return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
+ MaxNumberDevices);
+}
+
+WebRtc_Word32 AudioDeviceMac::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ AudioDeviceID playDevices[MaxNumberDevices];
+ WebRtc_UWord32 nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
+ playDevices, MaxNumberDevices);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable waveform-audio output devices is %u",
+ nDevices);
+
+ if (index > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices - 1));
+ return -1;
+ }
+
+ _outputDeviceIndex = index;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceMac::PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices(PlayoutDevices());
+
+ if ((index > (nDevices - 1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
+}
+
+WebRtc_Word32 AudioDeviceMac::RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ const WebRtc_UWord16 nDevices(RecordingDevices());
+
+ if ((index > (nDevices - 1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
+}
+
+WebRtc_Word16 AudioDeviceMac::RecordingDevices()
+{
+
+ AudioDeviceID recDevices[MaxNumberDevices];
+ return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
+ MaxNumberDevices);
+}
+
+WebRtc_Word32 AudioDeviceMac::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ AudioDeviceID recDevices[MaxNumberDevices];
+ WebRtc_UWord32 nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
+ recDevices, MaxNumberDevices);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " number of availiable waveform-audio input devices is %u",
+ nDevices);
+
+ if (index > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " device index is out of range [0,%u]", (nDevices - 1));
+ return -1;
+ }
+
+ _inputDeviceIndex = index;
+ _inputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+
+WebRtc_Word32
+AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "WindowsDeviceType not supported");
+ return -1;
+}
+
+WebRtc_Word32 AudioDeviceMac::PlayoutIsAvailable(bool& available)
+{
+
+ available = true;
+
+ // Try to initialize the playout side
+ if (InitPlayout() == -1)
+ {
+ available = false;
+ }
+
+ // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
+ // We must actually start playout here in order to have the IOProc
+ // deleted by calling StopPlayout().
+ if (StartPlayout() == -1)
+ {
+ available = false;
+ }
+
+ // Cancel effect of initialization
+ if (StopPlayout() == -1)
+ {
+ available = false;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::RecordingIsAvailable(bool& available)
+{
+
+ available = true;
+
+ // Try to initialize the recording side
+ if (InitRecording() == -1)
+ {
+ available = false;
+ }
+
+ // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
+ // We must actually start recording here in order to have the IOProc
+ // deleted by calling StopRecording().
+ if (StartRecording() == -1)
+ {
+ available = false;
+ }
+
+ // Cancel effect of initialization
+ if (StopRecording() == -1)
+ {
+ available = false;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (!_outputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitSpeaker() failed");
+ }
+
+ if (!MicrophoneIsInitialized())
+ {
+ // Make this call to check if we are using
+ // one or two devices (_twoDevices)
+ bool available = false;
+ if (MicrophoneIsAvailable(available) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " MicrophoneIsAvailable() failed");
+ }
+ }
+
+ PaUtil_FlushRingBuffer(_paRenderBuffer);
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ _renderDelayOffsetSamples = 0;
+ _renderDelayUs = 0;
+ _renderLatencyUs = 0;
+ _renderDeviceIsAlive = 1;
+ _doStop = false;
+
+ // The internal microphone of a MacBook Pro is located under the left speaker
+ // grille. When the internal speakers are in use, we want to fully stereo
+ // pan to the right.
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyDataSource,
+ kAudioDevicePropertyScopeOutput, 0 };
+ if (_macBookPro)
+ {
+ _macBookProPanRight = false;
+ Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ UInt32 dataSource = 0;
+ size = sizeof(dataSource);
+ WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &dataSource));
+
+ if (dataSource == 'ispk')
+ {
+ _macBookProPanRight = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id,
+ "MacBook Pro using internal speakers; stereo"
+ " panning right");
+ } else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
+ _id, "MacBook Pro not using internal speakers");
+ }
+
+ // Add a listener to determine if the status changes.
+ WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+ }
+ }
+
+ // Get current stream description
+ propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+ memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
+ size = sizeof(_outStreamFormat);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &_outStreamFormat));
+
+ if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Unacceptable output stream format -> mFormatID",
+ (const char *) &_outStreamFormat.mFormatID);
+ return -1;
+ }
+
+ if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Too many channels on device -> mChannelsPerFrame = %d",
+ _outStreamFormat.mChannelsPerFrame);
+ return -1;
+ }
+
+ if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Non-interleaved audio data is not supported.",
+ "AudioHardware streams should not have this format.");
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Ouput stream format:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mSampleRate = %f, mChannelsPerFrame = %u",
+ _outStreamFormat.mSampleRate,
+ _outStreamFormat.mChannelsPerFrame);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mBytesPerPacket = %u, mFramesPerPacket = %u",
+ _outStreamFormat.mBytesPerPacket,
+ _outStreamFormat.mFramesPerPacket);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mBytesPerFrame = %u, mBitsPerChannel = %u",
+ _outStreamFormat.mBytesPerFrame,
+ _outStreamFormat.mBitsPerChannel);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mFormatFlags = %u, mChannelsPerFrame = %u",
+ _outStreamFormat.mFormatFlags,
+ _outStreamFormat.mChannelsPerFrame);
+ logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+ (const char *) &_outStreamFormat.mFormatID);
+
+ // Our preferred format to work with
+ _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
+ if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
+ {
+ _outDesiredFormat.mChannelsPerFrame = 2;
+ } else
+ {
+ // Disable stereo playout when we only have one channel on the device.
+ _outDesiredFormat.mChannelsPerFrame = 1;
+ _playChannels = 1;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Stereo playout unavailable on this device");
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
+ }
+
+ _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT
+ * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+
+ _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame
+ * sizeof(SInt16);
+ _outDesiredFormat.mFramesPerPacket = 1; // In uncompressed audio,
+ // a packet is one frame.
+ _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame
+ * sizeof(SInt16);
+ _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+ _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
+ | kLinearPCMFormatFlagIsPacked;
+#ifdef WEBRTC_BIG_ENDIAN
+ _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+ _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &_outStreamFormat,
+ &_renderConverter));
+
+ // First try to set buffer size to desired value (_playBufDelayFixed)
+ UInt32 bufByteCount = (UInt32)((_outStreamFormat.mSampleRate / 1000.0)
+ * _playBufDelayFixed * _outStreamFormat.mChannelsPerFrame
+ * sizeof(Float32));
+ if (_outStreamFormat.mFramesPerPacket != 0)
+ {
+ if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
+ {
+ bufByteCount = ((UInt32)(bufByteCount
+ / _outStreamFormat.mFramesPerPacket) + 1)
+ * _outStreamFormat.mFramesPerPacket;
+ }
+ }
+
+ // Ensure the buffer size is within the acceptable range provided by the device.
+ propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+ AudioValueRange range;
+ size = sizeof(range);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &range));
+ if (range.mMinimum > bufByteCount)
+ {
+ bufByteCount = range.mMinimum;
+ } else if (range.mMaximum < bufByteCount)
+ {
+ bufByteCount = range.mMaximum;
+ }
+
+ propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+ size = sizeof(bufByteCount);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, size, &bufByteCount));
+
+ // Get render device latency
+ propertyAddress.mSelector = kAudioDevicePropertyLatency;
+ UInt32 latency = 0;
+ size = sizeof(UInt32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &latency));
+ _renderLatencyUs = (WebRtc_UWord32) ((1.0e6 * latency)
+ / _outStreamFormat.mSampleRate);
+
+ // Get render stream latency
+ propertyAddress.mSelector = kAudioDevicePropertyStreams;
+ AudioStreamID stream = 0;
+ size = sizeof(AudioStreamID);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &stream));
+ propertyAddress.mSelector = kAudioStreamPropertyLatency;
+ size = sizeof(UInt32);
+ latency = 0;
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &latency));
+ _renderLatencyUs += (WebRtc_UWord32) ((1.0e6 * latency)
+ / _outStreamFormat.mSampleRate);
+
+ // Listen for format changes
+ propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ // Listen for processor overloads
+ propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+ WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ if (_twoDevices || !_recIsInitialized)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
+ deviceIOProc, this, &_deviceIOProcID));
+ }
+
+ // Mark playout side as initialized
+ _playIsInitialized = true;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " initial playout status: _renderDelayOffsetSamples=%d,"
+ " _renderDelayUs=%d, _renderLatencyUs=%d",
+ _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " InitMicrophone() failed");
+ }
+
+ if (!SpeakerIsInitialized())
+ {
+ // Make this call to check if we are using
+ // one or two devices (_twoDevices)
+ bool available = false;
+ if (SpeakerIsAvailable(available) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " SpeakerIsAvailable() failed");
+ }
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+
+ PaUtil_FlushRingBuffer(_paCaptureBuffer);
+
+ _captureDelayUs = 0;
+ _captureLatencyUs = 0;
+ _captureDeviceIsAlive = 1;
+ _doStopRec = false;
+
+ // Get current stream description
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyStreamFormat,
+ kAudioDevicePropertyScopeInput, 0 };
+ memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
+ size = sizeof(_inStreamFormat);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &_inStreamFormat));
+
+ if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Unacceptable input stream format -> mFormatID",
+ (const char *) &_inStreamFormat.mFormatID);
+ return -1;
+ }
+
+ if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ ", Too many channels on device (mChannelsPerFrame = %d)",
+ _inStreamFormat.mChannelsPerFrame);
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Input stream format:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " mSampleRate = %f, mChannelsPerFrame = %u",
+ _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " mBytesPerPacket = %u, mFramesPerPacket = %u",
+ _inStreamFormat.mBytesPerPacket,
+ _inStreamFormat.mFramesPerPacket);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " mBytesPerFrame = %u, mBitsPerChannel = %u",
+ _inStreamFormat.mBytesPerFrame,
+ _inStreamFormat.mBitsPerChannel);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " mFormatFlags = %u, mChannelsPerFrame = %u",
+ _inStreamFormat.mFormatFlags,
+ _inStreamFormat.mChannelsPerFrame);
+ logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+ (const char *) &_inStreamFormat.mFormatID);
+
+ // Our preferred format to work with
+ if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
+ {
+ _inDesiredFormat.mChannelsPerFrame = 2;
+ } else
+ {
+ // Disable stereo recording when we only have one channel on the device.
+ _inDesiredFormat.mChannelsPerFrame = 1;
+ _recChannels = 1;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Stereo recording unavailable on this device");
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
+ }
+
+ _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
+ _inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
+ * sizeof(SInt16);
+ _inDesiredFormat.mFramesPerPacket = 1;
+ _inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
+ * sizeof(SInt16);
+ _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+ _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
+ | kLinearPCMFormatFlagIsPacked;
+#ifdef WEBRTC_BIG_ENDIAN
+ _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+ _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
+ &_captureConverter));
+
+ // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
+ // TODO(xians): investigate this block.
+ UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
+ * 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
+ * sizeof(Float32));
+ if (_inStreamFormat.mFramesPerPacket != 0)
+ {
+ if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
+ {
+ bufByteCount = ((UInt32)(bufByteCount
+ / _inStreamFormat.mFramesPerPacket) + 1)
+ * _inStreamFormat.mFramesPerPacket;
+ }
+ }
+
+ // Ensure the buffer size is within the acceptable range provided by the device.
+ propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+ AudioValueRange range;
+ size = sizeof(range);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &range));
+ if (range.mMinimum > bufByteCount)
+ {
+ bufByteCount = range.mMinimum;
+ } else if (range.mMaximum < bufByteCount)
+ {
+ bufByteCount = range.mMaximum;
+ }
+
+ propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+ size = sizeof(bufByteCount);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, size, &bufByteCount));
+
+ // Get capture device latency
+ propertyAddress.mSelector = kAudioDevicePropertyLatency;
+ UInt32 latency = 0;
+ size = sizeof(UInt32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &latency));
+ _captureLatencyUs = (UInt32)((1.0e6 * latency)
+ / _inStreamFormat.mSampleRate);
+
+ // Get capture stream latency
+ propertyAddress.mSelector = kAudioDevicePropertyStreams;
+ AudioStreamID stream = 0;
+ size = sizeof(AudioStreamID);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &stream));
+ propertyAddress.mSelector = kAudioStreamPropertyLatency;
+ size = sizeof(UInt32);
+ latency = 0;
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &latency));
+ _captureLatencyUs += (UInt32)((1.0e6 * latency)
+ / _inStreamFormat.mSampleRate);
+
+ // Listen for format changes
+ // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
+ propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ // Listen for processor overloads
+ propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+ WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ if (_twoDevices)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
+ inDeviceIOProc, this, &_inDeviceIOProcID));
+ } else if (!_playIsInitialized)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
+ deviceIOProc, this, &_deviceIOProcID));
+ }
+
+ // Mark recording side as initialized
+ _recIsInitialized = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StartRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ if (!_initialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Recording worker thread has not been started");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+
+ unsigned int threadID(0);
+ if (_captureWorkerThread != NULL)
+ {
+ _captureWorkerThread->Start(threadID);
+ }
+ _captureWorkerThreadId = threadID;
+
+ if (_twoDevices)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
+ } else if (!_playing)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
+ }
+
+ _recording = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StopRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ OSStatus err = noErr;
+
+ // Stop device
+ int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
+ if (_twoDevices)
+ {
+ if (_recording && captureDeviceIsAlive == 1)
+ {
+ _recording = false;
+ _doStopRec = true; // Signal to io proc to stop audio device
+ _critSect.Leave(); // Cannot be under lock, risk of deadlock
+ if (kEventTimeout == _stopEventRec.Wait(2000))
+ {
+ CriticalSectionScoped critScoped(_critSect);
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Timed out stopping the capture IOProc. "
+ "We may have failed to detect a device removal.");
+
+ WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
+ _inDeviceIOProcID));
+ WEBRTC_CA_LOG_WARN(
+ AudioDeviceDestroyIOProcID(_inputDeviceID,
+ _inDeviceIOProcID));
+ }
+ _critSect.Enter();
+ _doStopRec = false;
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " Recording stopped");
+ }
+ }
+ else
+ {
+ // We signal a stop for a shared device even when rendering has
+ // not yet ended. This is to ensure the IOProc will return early as
+ // intended (by checking |_recording|) before accessing
+ // resources we free below (e.g. the capture converter).
+ //
+ // In the case of a shared devcie, the IOProc will verify
+ // rendering has ended before stopping itself.
+ if (_recording && captureDeviceIsAlive == 1)
+ {
+ _recording = false;
+ _doStop = true; // Signal to io proc to stop audio device
+ _critSect.Leave(); // Cannot be under lock, risk of deadlock
+ if (kEventTimeout == _stopEvent.Wait(2000))
+ {
+ CriticalSectionScoped critScoped(_critSect);
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Timed out stopping the shared IOProc. "
+ "We may have failed to detect a device removal.");
+
+ // We assume rendering on a shared device has stopped as well if
+ // the IOProc times out.
+ WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
+ _deviceIOProcID));
+ WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
+ _deviceIOProcID));
+ }
+ _critSect.Enter();
+ _doStop = false;
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ " Recording stopped (shared)");
+ }
+ }
+
+ // Setting this signal will allow the worker thread to be stopped.
+ AtomicSet32(&_captureDeviceIsAlive, 0);
+ _critSect.Leave();
+ if (_captureWorkerThread != NULL)
+ {
+ if (!_captureWorkerThread->Stop())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Timed out waiting for the render worker thread to "
+ "stop.");
+ }
+ }
+ _critSect.Enter();
+
+ WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
+
+ // Remove listeners.
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyStreamFormat,
+ kAudioDevicePropertyScopeInput, 0 };
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ _recIsInitialized = false;
+ _recording = false;
+
+ return 0;
+}
+
+bool AudioDeviceMac::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+bool AudioDeviceMac::Recording() const
+{
+ return (_recording);
+}
+
+bool AudioDeviceMac::PlayoutIsInitialized() const
+{
+ return (_playIsInitialized);
+}
+
+WebRtc_Word32 AudioDeviceMac::StartPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ OSStatus err = noErr;
+
+ unsigned int threadID(0);
+ if (_renderWorkerThread != NULL)
+ {
+ _renderWorkerThread->Start(threadID);
+ }
+ _renderWorkerThreadId = threadID;
+
+ if (_twoDevices || !_recording)
+ {
+ WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
+ }
+ _playing = true;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::StopPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ OSStatus err = noErr;
+
+ int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
+ if (_playing && renderDeviceIsAlive == 1)
+ {
+ // We signal a stop for a shared device even when capturing has not
+ // yet ended. This is to ensure the IOProc will return early as
+ // intended (by checking |_playing|) before accessing resources we
+ // free below (e.g. the render converter).
+ //
+ // In the case of a shared device, the IOProc will verify capturing
+ // has ended before stopping itself.
+ _playing = false;
+ _doStop = true; // Signal to io proc to stop audio device
+ _critSect.Leave(); // Cannot be under lock, risk of deadlock
+ if (kEventTimeout == _stopEvent.Wait(2000))
+ {
+ CriticalSectionScoped critScoped(_critSect);
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Timed out stopping the render IOProc. "
+ "We may have failed to detect a device removal.");
+
+ // We assume capturing on a shared device has stopped as well if the
+ // IOProc times out.
+ WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
+ _deviceIOProcID));
+ WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
+ _deviceIOProcID));
+ }
+ _critSect.Enter();
+ _doStop = false;
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Playout stopped");
+ }
+
+ // Setting this signal will allow the worker thread to be stopped.
+ AtomicSet32(&_renderDeviceIsAlive, 0);
+ _critSect.Leave();
+ if (_renderWorkerThread != NULL)
+ {
+ if (!_renderWorkerThread->Stop())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Timed out waiting for the render worker thread to "
+ "stop.");
+ }
+ }
+ _critSect.Enter();
+
+ WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
+
+ // Remove listeners.
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
+ 0 };
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+
+ if (_macBookPro)
+ {
+ Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ propertyAddress.mSelector = kAudioDevicePropertyDataSource;
+ WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
+ &propertyAddress, &objectListenerProc, this));
+ }
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+ delayMS = static_cast<WebRtc_UWord16> (1e-3 * (renderDelayUs
+ + _renderLatencyUs) + 0.5);
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+ delayMS = static_cast<WebRtc_UWord16> (1e-3 * (captureDelayUs
+ + _captureLatencyUs) + 0.5);
+ return 0;
+}
+
+bool AudioDeviceMac::Playing() const
+{
+ return (_playing);
+}
+
+WebRtc_Word32 AudioDeviceMac::SetPlayoutBuffer(
+ const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS)
+{
+
+ if (type != AudioDeviceModule::kFixedBufferSize)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Adaptive buffer size not supported on this platform");
+ return -1;
+ }
+
+ _playBufType = type;
+ _playBufDelayFixed = sizeMS;
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::PlayoutBuffer(
+ AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const
+{
+
+ type = _playBufType;
+ sizeMS = _playBufDelayFixed;
+
+ return 0;
+}
+
+// Not implemented for Mac.
+WebRtc_Word32 AudioDeviceMac::CPULoad(WebRtc_UWord16& /*load*/) const
+{
+
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " API call not supported on this platform");
+
+ return -1;
+}
+
+bool AudioDeviceMac::PlayoutWarning() const
+{
+ return (_playWarning > 0);
+}
+
+bool AudioDeviceMac::PlayoutError() const
+{
+ return (_playError > 0);
+}
+
+bool AudioDeviceMac::RecordingWarning() const
+{
+ return (_recWarning > 0);
+}
+
+bool AudioDeviceMac::RecordingError() const
+{
+ return (_recError > 0);
+}
+
+void AudioDeviceMac::ClearPlayoutWarning()
+{
+ _playWarning = 0;
+}
+
+void AudioDeviceMac::ClearPlayoutError()
+{
+ _playError = 0;
+}
+
+void AudioDeviceMac::ClearRecordingWarning()
+{
+ _recWarning = 0;
+}
+
+void AudioDeviceMac::ClearRecordingError()
+{
+ _recError = 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+WebRtc_Word32
+AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
+ AudioDeviceID scopedDeviceIds[],
+ const WebRtc_UWord32 deviceListLength)
+{
+ OSStatus err = noErr;
+
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ UInt32 size = 0;
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
+ &propertyAddress, 0, NULL, &size));
+ if (size == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "No devices");
+ return 0;
+ }
+
+ AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
+ UInt32 numberDevices = size / sizeof(AudioDeviceID);
+ AudioBufferList* bufferList = NULL;
+ UInt32 numberScopedDevices = 0;
+
+ // First check if there is a default device and list it
+ UInt32 hardwareProperty = 0;
+ if (scope == kAudioDevicePropertyScopeOutput)
+ {
+ hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+ } else
+ {
+ hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+ }
+
+ AudioObjectPropertyAddress
+ propertyAddressDefault = { hardwareProperty,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ AudioDeviceID usedID;
+ UInt32 uintSize = sizeof(UInt32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &propertyAddressDefault, 0, NULL, &uintSize, &usedID));
+ if (usedID != kAudioDeviceUnknown)
+ {
+ scopedDeviceIds[numberScopedDevices] = usedID;
+ numberScopedDevices++;
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "GetNumberDevices(): Default device unknown");
+ }
+
+ // Then list the rest of the devices
+ bool listOK = true;
+
+ WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &propertyAddress, 0, NULL, &size, deviceIds));
+ if (err != noErr)
+ {
+ listOK = false;
+ } else
+ {
+ propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+ propertyAddress.mScope = scope;
+ propertyAddress.mElement = 0;
+ for (UInt32 i = 0; i < numberDevices; i++)
+ {
+ // Check for input channels
+ WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
+ &propertyAddress, 0, NULL, &size));
+ if (err == kAudioHardwareBadDeviceError)
+ {
+ // This device doesn't actually exist; continue iterating.
+ continue;
+ } else if (err != noErr)
+ {
+ listOK = false;
+ break;
+ }
+
+ bufferList = (AudioBufferList*) malloc(size);
+ WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
+ &propertyAddress, 0, NULL, &size, bufferList));
+ if (err != noErr)
+ {
+ listOK = false;
+ break;
+ }
+
+ if (bufferList->mNumberBuffers > 0)
+ {
+ if (numberScopedDevices >= deviceListLength)
+ {
+ WEBRTC_TRACE(kTraceError,
+ kTraceAudioDevice, _id,
+ "Device list is not long enough");
+ listOK = false;
+ break;
+ }
+
+ scopedDeviceIds[numberScopedDevices] = deviceIds[i];
+ numberScopedDevices++;
+ }
+
+ free(bufferList);
+ bufferList = NULL;
+ } // for
+ }
+
+ if (!listOK)
+ {
+ if (deviceIds)
+ {
+ free(deviceIds);
+ deviceIds = NULL;
+ }
+
+ if (bufferList)
+ {
+ free(bufferList);
+ bufferList = NULL;
+ }
+
+ return -1;
+ }
+
+ // Happy ending
+ if (deviceIds)
+ {
+ free(deviceIds);
+ deviceIds = NULL;
+ }
+
+ return numberScopedDevices;
+}
+
+WebRtc_Word32
+AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
+ const WebRtc_UWord16 index,
+ char* name)
+{
+ OSStatus err = noErr;
+ UInt32 len = kAdmMaxDeviceNameSize;
+ AudioDeviceID deviceIds[MaxNumberDevices];
+
+ int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
+ if (numberDevices < 0)
+ {
+ return -1;
+ } else if (numberDevices == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "No devices");
+ return -1;
+ }
+
+ // If the number is below the number of devices, assume it's "WEBRTC ID"
+ // otherwise assume it's a CoreAudio ID
+ AudioDeviceID usedID;
+
+ // Check if there is a default device
+ bool isDefaultDevice = false;
+ if (index == 0)
+ {
+ UInt32 hardwareProperty = 0;
+ if (scope == kAudioDevicePropertyScopeOutput)
+ {
+ hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+ } else
+ {
+ hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+ }
+ AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ UInt32 size = sizeof(UInt32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &propertyAddress, 0, NULL, &size, &usedID));
+ if (usedID == kAudioDeviceUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "GetDeviceName(): Default device unknown");
+ } else
+ {
+ isDefaultDevice = true;
+ }
+ }
+
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyDeviceName, scope, 0 };
+
+ if (isDefaultDevice)
+ {
+ char devName[len];
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
+ &propertyAddress, 0, NULL, &len, devName));
+
+ sprintf(name, "default (%s)", devName);
+ } else
+ {
+ if (index < numberDevices)
+ {
+ usedID = deviceIds[index];
+ } else
+ {
+ usedID = index;
+ }
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
+ &propertyAddress, 0, NULL, &len, name));
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::InitDevice(const WebRtc_UWord16 userDeviceIndex,
+ AudioDeviceID& deviceId,
+ const bool isInput)
+{
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ AudioObjectPropertyScope deviceScope;
+ AudioObjectPropertySelector defaultDeviceSelector;
+ AudioDeviceID deviceIds[MaxNumberDevices];
+
+ if (isInput)
+ {
+ deviceScope = kAudioDevicePropertyScopeInput;
+ defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
+ } else
+ {
+ deviceScope = kAudioDevicePropertyScopeOutput;
+ defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+ }
+
+ AudioObjectPropertyAddress
+ propertyAddress = { defaultDeviceSelector,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ // Get the actual device IDs
+ int numberDevices = GetNumberDevices(deviceScope, deviceIds,
+ MaxNumberDevices);
+ if (numberDevices < 0)
+ {
+ return -1;
+ } else if (numberDevices == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "InitDevice(): No devices");
+ return -1;
+ }
+
+ bool isDefaultDevice = false;
+ deviceId = kAudioDeviceUnknown;
+ if (userDeviceIndex == 0)
+ {
+ // Try to use default system device
+ size = sizeof(AudioDeviceID);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+ &propertyAddress, 0, NULL, &size, &deviceId));
+ if (deviceId == kAudioDeviceUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " No default device exists");
+ } else
+ {
+ isDefaultDevice = true;
+ }
+ }
+
+ if (!isDefaultDevice)
+ {
+ deviceId = deviceIds[userDeviceIndex];
+ }
+
+ // Obtain device name and manufacturer for logging.
+ // Also use this as a test to ensure a user-set device ID is valid.
+ char devName[128];
+ char devManf[128];
+ memset(devName, 0, sizeof(devName));
+ memset(devManf, 0, sizeof(devManf));
+
+ propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
+ propertyAddress.mScope = deviceScope;
+ propertyAddress.mElement = 0;
+ size = sizeof(devName);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
+ &propertyAddress, 0, NULL, &size, devName));
+
+ propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
+ size = sizeof(devManf);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
+ &propertyAddress, 0, NULL, &size, devManf));
+
+ if (isInput)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Input device: %s %s", devManf, devName);
+ } else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Output device: %s %s", devManf, devName);
+ }
+
+ return 0;
+}
+
+OSStatus AudioDeviceMac::objectListenerProc(
+ AudioObjectID objectId,
+ UInt32 numberAddresses,
+ const AudioObjectPropertyAddress addresses[],
+ void* clientData)
+{
+ AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
+ assert(ptrThis != NULL);
+
+ ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
+
+ // AudioObjectPropertyListenerProc functions are supposed to return 0
+ return 0;
+}
+
+OSStatus AudioDeviceMac::implObjectListenerProc(
+ const AudioObjectID objectId,
+ const UInt32 numberAddresses,
+ const AudioObjectPropertyAddress addresses[])
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "AudioDeviceMac::implObjectListenerProc()");
+
+ for (UInt32 i = 0; i < numberAddresses; i++)
+ {
+ if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
+ {
+ HandleDeviceChange();
+ } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
+ {
+ HandleStreamFormatChange(objectId, addresses[i]);
+ } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
+ {
+ HandleDataSourceChange(objectId, addresses[i]);
+ } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
+ {
+ HandleProcessorOverload(addresses[i]);
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::HandleDeviceChange()
+{
+ OSStatus err = noErr;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "kAudioHardwarePropertyDevices");
+
+ // A device has changed. Check if our registered devices have been removed.
+ // Ensure the devices have been initialized, meaning the IDs are valid.
+ if (MicrophoneIsInitialized())
+ {
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyDeviceIsAlive,
+ kAudioDevicePropertyScopeInput, 0 };
+ UInt32 deviceIsAlive = 1;
+ UInt32 size = sizeof(UInt32);
+ err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
+ NULL, &size, &deviceIsAlive);
+
+ if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "Capture device is not alive (probably removed)");
+ AtomicSet32(&_captureDeviceIsAlive, 0);
+ _mixerManager.CloseMicrophone();
+ if (_recError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, " pending recording error exists");
+ }
+ _recError = 1; // triggers callback from module process thread
+ } else if (err != noErr)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Error in AudioDeviceGetProperty()", (const char*) &err);
+ return -1;
+ }
+ }
+
+ if (SpeakerIsInitialized())
+ {
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyDeviceIsAlive,
+ kAudioDevicePropertyScopeOutput, 0 };
+ UInt32 deviceIsAlive = 1;
+ UInt32 size = sizeof(UInt32);
+ err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
+ NULL, &size, &deviceIsAlive);
+
+ if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "Render device is not alive (probably removed)");
+ AtomicSet32(&_renderDeviceIsAlive, 0);
+ _mixerManager.CloseSpeaker();
+ if (_playError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
+ _id, " pending playout error exists");
+ }
+ _playError = 1; // triggers callback from module process thread
+ } else if (err != noErr)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Error in AudioDeviceGetProperty()", (const char*) &err);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::HandleStreamFormatChange(
+ const AudioObjectID objectId,
+ const AudioObjectPropertyAddress propertyAddress)
+{
+ OSStatus err = noErr;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Stream format changed");
+
+ if (objectId != _inputDeviceID && objectId != _outputDeviceID)
+ {
+ return 0;
+ }
+
+ // Get the new device format
+ AudioStreamBasicDescription streamFormat;
+ UInt32 size = sizeof(streamFormat);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
+ &propertyAddress, 0, NULL, &size, &streamFormat));
+
+ if (streamFormat.mFormatID != kAudioFormatLinearPCM)
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Unacceptable input stream format -> mFormatID",
+ (const char *) &streamFormat.mFormatID);
+ return -1;
+ }
+
+ if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Too many channels on device (mChannelsPerFrame = %d)",
+ streamFormat.mChannelsPerFrame);
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Stream format:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mSampleRate = %f, mChannelsPerFrame = %u",
+ streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mBytesPerPacket = %u, mFramesPerPacket = %u",
+ streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mBytesPerFrame = %u, mBitsPerChannel = %u",
+ streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "mFormatFlags = %u, mChannelsPerFrame = %u",
+ streamFormat.mFormatFlags, streamFormat.mChannelsPerFrame);
+ logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+ (const char *) &streamFormat.mFormatID);
+
+ if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
+ {
+ memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
+
+ if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
+ {
+ _inDesiredFormat.mChannelsPerFrame = 2;
+ } else
+ {
+ // Disable stereo recording when we only have one channel on the device.
+ _inDesiredFormat.mChannelsPerFrame = 1;
+ _recChannels = 1;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Stereo recording unavailable on this device");
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
+ }
+
+ // Recreate the converter with the new format
+ // TODO(xians): make this thread safe
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
+ &_captureConverter));
+ } else
+ {
+ memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
+
+ if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
+ {
+ _outDesiredFormat.mChannelsPerFrame = 2;
+ } else
+ {
+ // Disable stereo playout when we only have one channel on the device.
+ _outDesiredFormat.mChannelsPerFrame = 1;
+ _playChannels = 1;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Stereo playout unavailable on this device");
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ // Update audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
+ }
+
+ _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT
+ * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
+ * _outDesiredFormat.mChannelsPerFrame;
+
+ // Recreate the converter with the new format
+ // TODO(xians): make this thread safe
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_renderConverter));
+
+ WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &streamFormat,
+ &_renderConverter));
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioDeviceMac::HandleDataSourceChange(
+ const AudioObjectID objectId,
+ const AudioObjectPropertyAddress propertyAddress)
+{
+ OSStatus err = noErr;
+
+ if (_macBookPro && propertyAddress.mScope
+ == kAudioDevicePropertyScopeOutput)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+ "Data source changed");
+
+ _macBookProPanRight = false;
+ UInt32 dataSource = 0;
+ UInt32 size = sizeof(UInt32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
+ &propertyAddress, 0, NULL, &size, &dataSource));
+ if (dataSource == 'ispk')
+ {
+ _macBookProPanRight = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "MacBook Pro using internal speakers; stereo panning right");
+ } else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "MacBook Pro not using internal speakers");
+ }
+ }
+
+ return 0;
+}
+WebRtc_Word32 AudioDeviceMac::HandleProcessorOverload(
+ const AudioObjectPropertyAddress propertyAddress)
+{
+ // TODO(xians): we probably want to notify the user in some way of the
+ // overload. However, the Windows interpretations of these errors seem to
+ // be more severe than what ProcessorOverload is thrown for.
+ //
+ // We don't log the notification, as it's sent from the HAL's IO thread. We
+ // don't want to slow it down even further.
+ if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
+ {
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
+ // overload");
+ //_callback->ProblemIsReported(
+ // SndCardStreamObserver::ERecordingProblem);
+ } else
+ {
+ //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ // "Render processor overload");
+ //_callback->ProblemIsReported(
+ // SndCardStreamObserver::EPlaybackProblem);
+ }
+
+ return 0;
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+ const AudioBufferList* inputData,
+ const AudioTimeStamp* inputTime,
+ AudioBufferList* outputData,
+ const AudioTimeStamp* outputTime,
+ void *clientData)
+{
+ AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
+ assert(ptrThis != NULL);
+
+ ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
+
+ // AudioDeviceIOProc functions are supposed to return 0
+ return 0;
+}
+
+OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
+ UInt32 *numberDataPackets,
+ AudioBufferList *data,
+ AudioStreamPacketDescription **,
+ void *userData)
+{
+ AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
+ assert(ptrThis != NULL);
+
+ return ptrThis->implOutConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+ const AudioBufferList* inputData,
+ const AudioTimeStamp* inputTime,
+ AudioBufferList*,
+ const AudioTimeStamp*, void* clientData)
+{
+ AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
+ assert(ptrThis != NULL);
+
+ ptrThis->implInDeviceIOProc(inputData, inputTime);
+
+ // AudioDeviceIOProc functions are supposed to return 0
+ return 0;
+}
+
+OSStatus AudioDeviceMac::inConverterProc(
+ AudioConverterRef,
+ UInt32 *numberDataPackets,
+ AudioBufferList *data,
+ AudioStreamPacketDescription ** /*dataPacketDescription*/,
+ void *userData)
+{
+ AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
+ assert(ptrThis != NULL);
+
+ return ptrThis->implInConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime,
+ AudioBufferList *outputData,
+ const AudioTimeStamp *outputTime)
+{
+ OSStatus err = noErr;
+ UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
+ UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+ if (!_twoDevices && _recording)
+ {
+ implInDeviceIOProc(inputData, inputTime);
+ }
+
+ // Check if we should close down audio device
+ // Double-checked locking optimization to remove locking overhead
+ if (_doStop)
+ {
+ _critSect.Enter();
+ if (_doStop)
+ {
+ if (_twoDevices || (!_recording && !_playing))
+ {
+ // In the case of a shared device, the single driving ioProc
+ // is stopped here
+ WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
+ _deviceIOProcID));
+ WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
+ _deviceIOProcID));
+ if (err == noErr)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, " Playout or shared device stopped");
+ }
+ }
+
+ _doStop = false;
+ _stopEvent.Set();
+ _critSect.Leave();
+ return 0;
+ }
+ _critSect.Leave();
+ }
+
+ if (!_playing)
+ {
+ // This can be the case when a shared device is capturing but not
+ // rendering. We allow the checks above before returning to avoid a
+ // timeout when capturing is stopped.
+ return 0;
+ }
+
+ assert(_outStreamFormat.mBytesPerFrame != 0);
+ UInt32 size = outputData->mBuffers->mDataByteSize
+ / _outStreamFormat.mBytesPerFrame;
+
+ // TODO(xians): signal an error somehow?
+ err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
+ this, &size, outputData, NULL);
+ if (err != noErr)
+ {
+ if (err == 1)
+ {
+ // This is our own error.
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Error in AudioConverterFillComplexBuffer()");
+ return 1;
+ } else
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Error in AudioConverterFillComplexBuffer()",
+ (const char *) &err);
+ return 1;
+ }
+ }
+
+ ring_buffer_size_t bufSizeSamples =
+ PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
+
+ int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
+ + 0.5);
+ renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
+ / _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
+ + 0.5);
+
+ AtomicSet32(&_renderDelayUs, renderDelayUs);
+
+ return 0;
+}
+
+OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
+ AudioBufferList *data)
+{
+ assert(data->mNumberBuffers == 1);
+ ring_buffer_size_t numSamples = *numberDataPackets
+ * _outDesiredFormat.mChannelsPerFrame;
+
+ data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
+ // Always give the converter as much as it wants, zero padding as required.
+ data->mBuffers->mDataByteSize = *numberDataPackets
+ * _outDesiredFormat.mBytesPerPacket;
+ data->mBuffers->mData = _renderConvertData;
+ memset(_renderConvertData, 0, sizeof(_renderConvertData));
+
+ PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
+
+ kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_signal_all() error: %d", kernErr);
+ return 1;
+ }
+
+ return 0;
+}
+
+OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime)
+{
+ OSStatus err = noErr;
+ UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
+ UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+ // Check if we should close down audio device
+ // Double-checked locking optimization to remove locking overhead
+ if (_doStopRec)
+ {
+ _critSect.Enter();
+ if (_doStopRec)
+ {
+ // This will be signalled only when a shared device is not in use.
+ WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+ WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
+ _inDeviceIOProcID));
+ if (err == noErr)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
+ _id, " Recording device stopped");
+ }
+
+ _doStopRec = false;
+ _stopEventRec.Set();
+ _critSect.Leave();
+ return 0;
+ }
+ _critSect.Leave();
+ }
+
+ if (!_recording)
+ {
+ // Allow above checks to avoid a timeout on stopping capture.
+ return 0;
+ }
+
+ ring_buffer_size_t bufSizeSamples =
+ PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
+
+ int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
+ + 0.5);
+ captureDelayUs
+ += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
+ / _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
+ + 0.5);
+
+ AtomicSet32(&_captureDelayUs, captureDelayUs);
+
+ assert(inputData->mNumberBuffers == 1);
+ ring_buffer_size_t numSamples = inputData->mBuffers->mDataByteSize
+ * _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
+ PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
+ numSamples);
+
+ kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
+ if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_signal_all() error: %d", kernErr);
+ }
+
+ return err;
+}
+
+OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
+ AudioBufferList *data)
+{
+ assert(data->mNumberBuffers == 1);
+ ring_buffer_size_t numSamples = *numberDataPackets
+ * _inStreamFormat.mChannelsPerFrame;
+
+ while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
+ {
+ mach_timespec_t timeout;
+ timeout.tv_sec = 0;
+ timeout.tv_nsec = TIMER_PERIOD_MS;
+
+ kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
+ if (kernErr == KERN_OPERATION_TIMED_OUT)
+ {
+ int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
+ if (signal == 0)
+ {
+ // The capture device is no longer alive; stop the worker thread.
+ *numberDataPackets = 0;
+ return 1;
+ }
+ } else if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_wait() error: %d", kernErr);
+ }
+ }
+
+ // Pass the read pointer directly to the converter to avoid a memcpy.
+ void* dummyPtr;
+ ring_buffer_size_t dummySize;
+ PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
+ &data->mBuffers->mData, &numSamples,
+ &dummyPtr, &dummySize);
+ PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+
+ data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
+ *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
+ data->mBuffers->mDataByteSize = *numberDataPackets
+ * _inStreamFormat.mBytesPerPacket;
+
+ return 0;
+}
+
+bool AudioDeviceMac::RunRender(void* ptrThis)
+{
+ return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
+}
+
+bool AudioDeviceMac::RenderWorkerThread()
+{
+ ring_buffer_size_t numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
+ * _outDesiredFormat.mChannelsPerFrame;
+ while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
+ - _renderDelayOffsetSamples < numSamples)
+ {
+ mach_timespec_t timeout;
+ timeout.tv_sec = 0;
+ timeout.tv_nsec = TIMER_PERIOD_MS;
+
+ kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
+ if (kernErr == KERN_OPERATION_TIMED_OUT)
+ {
+ int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
+ if (signal == 0)
+ {
+ // The render device is no longer alive; stop the worker thread.
+ return false;
+ }
+ } else if (kernErr != KERN_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " semaphore_timedwait() error: %d", kernErr);
+ }
+ }
+
+ WebRtc_Word8 playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
+
+ if (!_ptrAudioBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " capture AudioBuffer is invalid");
+ return false;
+ }
+
+ // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+ WebRtc_UWord32 nSamples =
+ _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+
+ nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " invalid number of output samples(%d)", nSamples);
+ }
+
+ WebRtc_UWord32 nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+
+ SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
+ if (_macBookProPanRight && (_playChannels == 2))
+ {
+ // Mix entirely into the right channel and zero the left channel.
+ SInt32 sampleInt32 = 0;
+ for (WebRtc_UWord32 sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
+ += 2)
+ {
+ sampleInt32 = pPlayBuffer[sampleIdx];
+ sampleInt32 += pPlayBuffer[sampleIdx + 1];
+ sampleInt32 /= 2;
+
+ if (sampleInt32 > 32767)
+ {
+ sampleInt32 = 32767;
+ } else if (sampleInt32 < -32768)
+ {
+ sampleInt32 = -32768;
+ }
+
+ pPlayBuffer[sampleIdx] = 0;
+ pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
+ }
+ }
+
+ PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
+
+ return true;
+}
+
+bool AudioDeviceMac::RunCapture(void* ptrThis)
+{
+ return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
+}
+
+bool AudioDeviceMac::CaptureWorkerThread()
+{
+ OSStatus err = noErr;
+ UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
+ * _inDesiredFormat.mChannelsPerFrame;
+ SInt16 recordBuffer[noRecSamples];
+ UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
+
+ AudioBufferList engineBuffer;
+ engineBuffer.mNumberBuffers = 1; // Interleaved channels.
+ engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
+ engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
+ * noRecSamples;
+ engineBuffer.mBuffers->mData = recordBuffer;
+
+ err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
+ this, &size, &engineBuffer, NULL);
+ if (err != noErr)
+ {
+ if (err == 1)
+ {
+ // This is our own error.
+ return false;
+ } else
+ {
+ logCAMsg(kTraceError, kTraceAudioDevice, _id,
+ "Error in AudioConverterFillComplexBuffer()",
+ (const char *) &err);
+ return false;
+ }
+ }
+
+ // TODO(xians): what if the returned size is incorrect?
+ if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
+ {
+ WebRtc_UWord32 currentMicLevel(0);
+ WebRtc_UWord32 newMicLevel(0);
+ WebRtc_Word32 msecOnPlaySide;
+ WebRtc_Word32 msecOnRecordSide;
+
+ int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+ int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+
+ msecOnPlaySide = static_cast<WebRtc_Word32> (1e-3 * (renderDelayUs
+ + _renderLatencyUs) + 0.5);
+ msecOnRecordSide = static_cast<WebRtc_Word32> (1e-3 * (captureDelayUs
+ + _captureLatencyUs) + 0.5);
+
+ if (!_ptrAudioBuffer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " capture AudioBuffer is invalid");
+ return false;
+ }
+
+ // store the recorded buffer (no action will be taken if the
+ // #recorded samples is not a full buffer)
+ _ptrAudioBuffer->SetRecordedBuffer((WebRtc_Word8*) &recordBuffer,
+ (WebRtc_UWord32) size);
+
+ if (AGC())
+ {
+ // store current mic level in the audio buffer if AGC is enabled
+ if (MicrophoneVolume(currentMicLevel) == 0)
+ {
+ // this call does not affect the actual microphone volume
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ }
+
+ _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
+
+ // deliver recorded samples at specified sample rate, mic level etc.
+ // to the observer using callback
+ _ptrAudioBuffer->DeliverRecordedData();
+
+ if (AGC())
+ {
+ newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0)
+ {
+ // The VQE will only deliver non-zero microphone levels when
+ // a change is needed.
+ // Set this new mic level (received from the observer as return
+ // value in the callback).
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
+ _id, " AGC change of volume: old=%u => new=%u",
+ currentMicLevel, newMicLevel);
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " the required modification of the microphone "
+ "volume failed");
+ }
+ }
+ }
+ }
+
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h
new file mode 100644
index 0000000..9dc8917
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h
@@ -0,0 +1,399 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
+
+#include "audio_device_generic.h"
+#include "critical_section_wrapper.h"
+#include "audio_mixer_manager_mac.h"
+
+#include <CoreAudio/CoreAudio.h>
+#include <AudioToolbox/AudioConverter.h>
+#include <mach/semaphore.h>
+
+struct PaUtilRingBuffer;
+
+namespace webrtc
+{
+class EventWrapper;
+class ThreadWrapper;
+
+const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
+const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
+
+const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
+const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
+const WebRtc_UWord32 N_DEVICE_CHANNELS = 8;
+
+const WebRtc_UWord32 ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC
+ / 100);
+const WebRtc_UWord32 ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC
+ / 100);
+
+enum
+{
+ N_BLOCKS_IO = 2
+};
+enum
+{
+ N_BUFFERS_IN = 10
+};
+enum
+{
+ N_BUFFERS_OUT = 3
+}; // Must be at least N_BLOCKS_IO
+
+const WebRtc_UWord32 TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
+
+const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES
+ * N_DEVICE_CHANNELS * N_BUFFERS_IN);
+const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES =
+ (ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT);
+
+class AudioDeviceMac: public AudioDeviceGeneric
+{
+public:
+ AudioDeviceMac(const WebRtc_Word32 id);
+ ~AudioDeviceMac();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32
+ ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(
+ WebRtc_UWord16 index,
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize],
+ WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
+ WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
+ WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32
+ MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32
+ SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+ WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
+ WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+public:
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private:
+ void Lock()
+ {
+ _critSect.Enter();
+ }
+ ;
+ void UnLock()
+ {
+ _critSect.Leave();
+ }
+ ;
+ WebRtc_Word32 Id()
+ {
+ return _id;
+ }
+
+ static void AtomicSet32(int32_t* theValue, int32_t newValue);
+ static int32_t AtomicGet32(int32_t* theValue);
+
+ static void logCAMsg(const TraceLevel level,
+ const TraceModule module,
+ const WebRtc_Word32 id, const char *msg,
+ const char *err);
+
+ WebRtc_Word32 GetNumberDevices(const AudioObjectPropertyScope scope,
+ AudioDeviceID scopedDeviceIds[],
+ const WebRtc_UWord32 deviceListLength);
+
+ WebRtc_Word32 GetDeviceName(const AudioObjectPropertyScope scope,
+ const WebRtc_UWord16 index, char* name);
+
+ WebRtc_Word32 InitDevice(WebRtc_UWord16 userDeviceIndex,
+ AudioDeviceID& deviceId, bool isInput);
+
+ static OSStatus
+ objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
+ const AudioObjectPropertyAddress addresses[],
+ void* clientData);
+
+ OSStatus
+ implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
+ const AudioObjectPropertyAddress addresses[]);
+
+ WebRtc_Word32 HandleDeviceChange();
+
+ WebRtc_Word32
+ HandleStreamFormatChange(AudioObjectID objectId,
+ AudioObjectPropertyAddress propertyAddress);
+
+ WebRtc_Word32
+ HandleDataSourceChange(AudioObjectID objectId,
+ AudioObjectPropertyAddress propertyAddress);
+
+ WebRtc_Word32
+ HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
+
+private:
+ static OSStatus deviceIOProc(AudioDeviceID device,
+ const AudioTimeStamp *now,
+ const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime,
+ AudioBufferList *outputData,
+ const AudioTimeStamp* outputTime,
+ void *clientData);
+
+ static OSStatus
+ outConverterProc(AudioConverterRef audioConverter,
+ UInt32 *numberDataPackets, AudioBufferList *data,
+ AudioStreamPacketDescription **dataPacketDescription,
+ void *userData);
+
+ static OSStatus inDeviceIOProc(AudioDeviceID device,
+ const AudioTimeStamp *now,
+ const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime,
+ AudioBufferList *outputData,
+ const AudioTimeStamp *outputTime,
+ void *clientData);
+
+ static OSStatus
+ inConverterProc(AudioConverterRef audioConverter,
+ UInt32 *numberDataPackets, AudioBufferList *data,
+ AudioStreamPacketDescription **dataPacketDescription,
+ void *inUserData);
+
+ OSStatus implDeviceIOProc(const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime,
+ AudioBufferList *outputData,
+ const AudioTimeStamp *outputTime);
+
+ OSStatus implOutConverterProc(UInt32 *numberDataPackets,
+ AudioBufferList *data);
+
+ OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
+ const AudioTimeStamp *inputTime);
+
+ OSStatus implInConverterProc(UInt32 *numberDataPackets,
+ AudioBufferList *data);
+
+ static bool RunCapture(void*);
+ static bool RunRender(void*);
+ bool CaptureWorkerThread();
+ bool RenderWorkerThread();
+
+private:
+ AudioDeviceBuffer* _ptrAudioBuffer;
+
+ CriticalSectionWrapper& _critSect;
+
+ EventWrapper& _stopEventRec;
+ EventWrapper& _stopEvent;
+
+ ThreadWrapper* _captureWorkerThread;
+ ThreadWrapper* _renderWorkerThread;
+ WebRtc_UWord32 _captureWorkerThreadId;
+ WebRtc_UWord32 _renderWorkerThreadId;
+
+ WebRtc_Word32 _id;
+
+ AudioMixerManagerMac _mixerManager;
+
+ WebRtc_UWord16 _inputDeviceIndex;
+ WebRtc_UWord16 _outputDeviceIndex;
+ AudioDeviceID _inputDeviceID;
+ AudioDeviceID _outputDeviceID;
+#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
+ AudioDeviceIOProcID _inDeviceIOProcID;
+ AudioDeviceIOProcID _deviceIOProcID;
+#endif
+ bool _inputDeviceIsSpecified;
+ bool _outputDeviceIsSpecified;
+
+ WebRtc_UWord8 _recChannels;
+ WebRtc_UWord8 _playChannels;
+
+ Float32* _captureBufData;
+ SInt16* _renderBufData;
+
+ SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
+
+ AudioDeviceModule::BufferType _playBufType;
+
+private:
+ bool _initialized;
+ bool _isShutDown;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _startRec;
+ bool _stopRec;
+ bool _stopPlay;
+ bool _AGC;
+
+ // Atomically set varaibles
+ int32_t _renderDeviceIsAlive;
+ int32_t _captureDeviceIsAlive;
+
+ bool _twoDevices;
+ bool _doStop; // For play if not shared device or play+rec if shared device
+ bool _doStopRec; // For rec if not shared device
+ bool _macBookPro;
+ bool _macBookProPanRight;
+ bool _stereoRender;
+ bool _stereoRenderRequested;
+
+ AudioConverterRef _captureConverter;
+ AudioConverterRef _renderConverter;
+
+ AudioStreamBasicDescription _outStreamFormat;
+ AudioStreamBasicDescription _outDesiredFormat;
+ AudioStreamBasicDescription _inStreamFormat;
+ AudioStreamBasicDescription _inDesiredFormat;
+
+ WebRtc_UWord32 _captureLatencyUs;
+ WebRtc_UWord32 _renderLatencyUs;
+
+ // Atomically set variables
+ mutable int32_t _captureDelayUs;
+ mutable int32_t _renderDelayUs;
+
+ WebRtc_Word32 _renderDelayOffsetSamples;
+
+private:
+ WebRtc_UWord16 _playBufDelay; // playback delay
+ WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
+
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ PaUtilRingBuffer* _paCaptureBuffer;
+ PaUtilRingBuffer* _paRenderBuffer;
+
+ semaphore_t _renderSemaphore;
+ semaphore_t _captureSemaphore;
+
+ WebRtc_UWord32 _captureBufSizeSamples;
+ WebRtc_UWord32 _renderBufSizeSamples;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
new file mode 100644
index 0000000..94bdf26
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility_mac.h"
+#include "audio_device_config.h" // DEBUG_PRINT()
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+AudioDeviceUtilityMac::AudioDeviceUtilityMac(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _lastError(AudioDeviceModule::kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
+ "%s created", __FUNCTION__);
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceUtilityMac() - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceUtilityMac::~AudioDeviceUtilityMac()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ // free stuff here...
+ }
+
+ delete &_critSect;
+}
+
+WebRtc_Word32 AudioDeviceUtilityMac::Init()
+{
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
+ " OS info: %s", "OS X");
+
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
new file mode 100644
index 0000000..ccb3d99
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
+
+#include "audio_device_utility.h"
+#include "audio_device.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class AudioDeviceUtilityMac: public AudioDeviceUtility
+{
+public:
+ AudioDeviceUtilityMac(const WebRtc_Word32 id);
+ ~AudioDeviceUtilityMac();
+
+ virtual WebRtc_Word32 Init();
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ AudioDeviceModule::ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc
new file mode 100644
index 0000000..40850d4
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc
@@ -0,0 +1,1167 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_mixer_manager_mac.h"
+#include "trace.h"
+
+#include <unistd.h> // getpid()
+
+namespace webrtc {
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceError, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ return -1; \
+ } \
+ } while(0)
+
+#define WEBRTC_CA_LOG_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceError, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ } \
+ } while(0)
+
+#define WEBRTC_CA_LOG_WARN(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(kTraceWarning, kTraceAudioDevice, _id, \
+ "Error in " #expr, (const char *)&err); \
+ } \
+ } while(0)
+
+AudioMixerManagerMac::AudioMixerManagerMac(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _inputDeviceID(kAudioObjectUnknown),
+ _outputDeviceID(kAudioObjectUnknown),
+ _noInputChannels(0),
+ _noOutputChannels(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s constructed", __FUNCTION__);
+}
+
+AudioMixerManagerMac::~AudioMixerManagerMac()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
+ "%s destructed", __FUNCTION__);
+
+ Close();
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// PUBLIC METHODS
+// ============================================================================
+
+WebRtc_Word32 AudioMixerManagerMac::Close()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ CloseSpeaker();
+ CloseMicrophone();
+
+ return 0;
+
+}
+
+WebRtc_Word32 AudioMixerManagerMac::CloseSpeaker()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _outputDeviceID = kAudioObjectUnknown;
+ _noOutputChannels = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::CloseMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ _inputDeviceID = kAudioObjectUnknown;
+ _noInputChannels = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
+
+ CriticalSectionScoped lock(_critSect);
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ pid_t hogPid = -1;
+
+ _outputDeviceID = deviceID;
+
+ // Check which process, if any, has hogged the device.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
+ kAudioDevicePropertyScopeOutput, 0 };
+
+ size = sizeof(hogPid);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &hogPid));
+
+ if (hogPid == -1)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " No process has hogged the input device");
+ }
+ // getpid() is apparently "always successful"
+ else if (hogPid == getpid())
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Our process has hogged the input device");
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Another process (pid = %d) has hogged the input device",
+ static_cast<int> (hogPid));
+
+ return -1;
+ }
+
+ // get number of channels from stream format
+ propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+
+ // Get the stream format, to be able to read the number of channels.
+ AudioStreamBasicDescription streamFormat;
+ size = sizeof(AudioStreamBasicDescription);
+ memset(&streamFormat, 0, size);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &streamFormat));
+
+ _noOutputChannels = streamFormat.mChannelsPerFrame;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
+
+ CriticalSectionScoped lock(_critSect);
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ pid_t hogPid = -1;
+
+ _inputDeviceID = deviceID;
+
+ // Check which process, if any, has hogged the device.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
+ kAudioDevicePropertyScopeInput, 0 };
+ size = sizeof(hogPid);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &hogPid));
+ if (hogPid == -1)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " No process has hogged the input device");
+ }
+ // getpid() is apparently "always successful"
+ else if (hogPid == getpid())
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " Our process has hogged the input device");
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Another process (pid = %d) has hogged the input device",
+ static_cast<int> (hogPid));
+
+ return -1;
+ }
+
+ // get number of channels from stream format
+ propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+
+ // Get the stream format, to be able to read the number of channels.
+ AudioStreamBasicDescription streamFormat;
+ size = sizeof(AudioStreamBasicDescription);
+ memset(&streamFormat, 0, size);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &streamFormat));
+
+ _noInputChannels = streamFormat.mChannelsPerFrame;
+
+ return 0;
+}
+
+bool AudioMixerManagerMac::SpeakerIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ return (_outputDeviceID != kAudioObjectUnknown);
+}
+
+bool AudioMixerManagerMac::MicrophoneIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
+ __FUNCTION__);
+
+ return (_inputDeviceID != kAudioObjectUnknown);
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ bool success = false;
+
+ // volume range is 0.0 - 1.0, convert from 0 -255
+ const Float32 vol = (Float32)(volume / 255.0);
+
+ assert(vol <= 1.0 && vol >= 0.0);
+
+ // Does the capture device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
+ 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(vol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, size, &vol));
+
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(vol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, size, &vol));
+ }
+ success = true;
+ }
+
+ if (!success)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to set a volume on any output channel");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ unsigned int channels = 0;
+ Float32 channelVol = 0;
+ Float32 vol = 0;
+
+ // Does the device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
+ 0 };
+ Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(vol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &vol));
+
+ // vol 0.0 to 1.0 -> convert to 0 - 255
+ volume = static_cast<WebRtc_UWord32> (vol * 255 + 0.5);
+ } else
+ {
+ // Otherwise get the average volume across channels.
+ vol = 0;
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ channelVol = 0;
+ propertyAddress.mElement = i;
+ hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(channelVol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &channelVol));
+
+ vol += channelVol;
+ channels++;
+ }
+ }
+
+ if (channels == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to get a volume on any channel");
+ return -1;
+ }
+
+ assert(channels > 0);
+ // vol 0.0 to 1.0 -> convert to 0 - 255
+ volume = static_cast<WebRtc_UWord32> (255 * vol / channels + 0.5);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 255
+ maxVolume = 255;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 255
+ minVolume = 0;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 255
+ stepSize = 1;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available)
+{
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+
+ // Does the capture device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = {
+ kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
+ 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ available = true;
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err != noErr || !isSettable)
+ {
+ available = false;
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Volume cannot be set for output channel %d, err=%d",
+ i, err);
+ return -1;
+ }
+ }
+
+ available = true;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available)
+{
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+
+ // Does the capture device have a master mute control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeOutput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ available = true;
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err != noErr || !isSettable)
+ {
+ available = false;
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Mute cannot be set for output channel %d, err=%d",
+ i, err);
+ return -1;
+ }
+ }
+
+ available = true;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SetSpeakerMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ UInt32 mute = enable ? 1 : 0;
+ bool success = false;
+
+ // Does the render device have a master mute control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeOutput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(mute);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, size, &mute));
+
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(mute);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, size, &mute));
+ }
+ success = true;
+ }
+
+ if (!success)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to set mute on any input channel");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SpeakerMute(bool& enabled) const
+{
+
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ unsigned int channels = 0;
+ UInt32 channelMuted = 0;
+ UInt32 muted = 0;
+
+ // Does the device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeOutput, 0 };
+ Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(muted);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &muted));
+
+ // 1 means muted
+ enabled = static_cast<bool> (muted);
+ } else
+ {
+ // Otherwise check if all channels are muted.
+ for (UInt32 i = 1; i <= _noOutputChannels; i++)
+ {
+ muted = 0;
+ propertyAddress.mElement = i;
+ hasProperty = AudioObjectHasProperty(_outputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(channelMuted);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
+ &propertyAddress, 0, NULL, &size, &channelMuted));
+
+ muted = (muted && channelMuted);
+ channels++;
+ }
+ }
+
+ if (channels == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to get mute for any channel");
+ return -1;
+ }
+
+ assert(channels > 0);
+ // 1 means muted
+ enabled = static_cast<bool> (muted);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available)
+{
+ if (_outputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ available = (_noOutputChannels == 2);
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available)
+{
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ available = (_noInputChannels == 2);
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available)
+{
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+
+ // Does the capture device have a master mute control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ available = true;
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err != noErr || !isSettable)
+ {
+ available = false;
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Mute cannot be set for output channel %d, err=%d",
+ i, err);
+ return -1;
+ }
+ }
+
+ available = true;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ UInt32 mute = enable ? 1 : 0;
+ bool success = false;
+
+ // Does the capture device have a master mute control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(mute);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, size, &mute));
+
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(mute);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, size, &mute));
+ }
+ success = true;
+ }
+
+ if (!success)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to set mute on any input channel");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::MicrophoneMute(bool& enabled) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ unsigned int channels = 0;
+ UInt32 channelMuted = 0;
+ UInt32 muted = 0;
+
+ // Does the device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(muted);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &muted));
+
+ // 1 means muted
+ enabled = static_cast<bool> (muted);
+ } else
+ {
+ // Otherwise check if all channels are muted.
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ muted = 0;
+ propertyAddress.mElement = i;
+ hasProperty = AudioObjectHasProperty(_inputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(channelMuted);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &channelMuted));
+
+ muted = (muted && channelMuted);
+ channels++;
+ }
+ }
+
+ if (channels == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to get mute for any channel");
+ return -1;
+ }
+
+ assert(channels > 0);
+ // 1 means muted
+ enabled = static_cast<bool> (muted);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
+ enabled);
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available)
+{
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ available = false; // No AudioObjectPropertySelector value for Mic Boost
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneBoost(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // Ensure that the selected microphone has a valid boost control.
+ bool available(false);
+ MicrophoneBoostIsAvailable(available);
+ if (!available)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " it is not possible to enable microphone boost");
+ return -1;
+ }
+
+ // It is assumed that the call above fails!
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // Microphone boost cannot be enabled on this platform!
+ enabled = false;
+
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available)
+{
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+
+ // Does the capture device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyVolumeScalar,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ available = true;
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err != noErr || !isSettable)
+ {
+ available = false;
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Volume cannot be set for input channel %d, err=%d",
+ i, err);
+ return -1;
+ }
+ }
+
+ available = true;
+ return 0;
+}
+
+WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ bool success = false;
+
+ // volume range is 0.0 - 1.0, convert from 0 - 255
+ const Float32 vol = (Float32)(volume / 255.0);
+
+ assert(vol <= 1.0 && vol >= 0.0);
+
+ // Does the capture device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyVolumeScalar,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(vol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, size, &vol));
+
+ return 0;
+ }
+
+ // Otherwise try to set each channel.
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ propertyAddress.mElement = i;
+ isSettable = false;
+ err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+ &isSettable);
+ if (err == noErr && isSettable)
+ {
+ size = sizeof(vol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, size, &vol));
+ }
+ success = true;
+ }
+
+ if (!success)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to set a level on any input channel");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ OSStatus err = noErr;
+ UInt32 size = 0;
+ unsigned int channels = 0;
+ Float32 channelVol = 0;
+ Float32 volFloat32 = 0;
+
+ // Does the device have a master volume control?
+ // If so, use it exclusively.
+ AudioObjectPropertyAddress
+ propertyAddress = { kAudioDevicePropertyVolumeScalar,
+ kAudioDevicePropertyScopeInput, 0 };
+ Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(volFloat32);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &volFloat32));
+
+ // vol 0.0 to 1.0 -> convert to 0 - 255
+ volume = static_cast<WebRtc_UWord32> (volFloat32 * 255 + 0.5);
+ } else
+ {
+ // Otherwise get the average volume across channels.
+ volFloat32 = 0;
+ for (UInt32 i = 1; i <= _noInputChannels; i++)
+ {
+ channelVol = 0;
+ propertyAddress.mElement = i;
+ hasProperty = AudioObjectHasProperty(_inputDeviceID,
+ &propertyAddress);
+ if (hasProperty)
+ {
+ size = sizeof(channelVol);
+ WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
+ &propertyAddress, 0, NULL, &size, &channelVol));
+
+ volFloat32 += channelVol;
+ channels++;
+ }
+ }
+
+ if (channels == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " Unable to get a level on any channel");
+ return -1;
+ }
+
+ assert(channels > 0);
+ // vol 0.0 to 1.0 -> convert to 0 - 255
+ volume = static_cast<WebRtc_UWord32>
+ (255 * volFloat32 / channels + 0.5);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
+ volume);
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 255
+ maxVolume = 255;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 10
+ minVolume = 0;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioMixerManagerMac::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (_inputDeviceID == kAudioObjectUnknown)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " device ID has not been set");
+ return -1;
+ }
+
+ // volume range is 0.0 to 1.0
+ // we convert that to 0 - 10
+ stepSize = 1;
+
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+// CoreAudio errors are best interpreted as four character strings.
+void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
+ const TraceModule module,
+ const WebRtc_Word32 id, const char *msg,
+ const char *err)
+{
+ assert(msg != NULL);
+ assert(err != NULL);
+
+#ifdef WEBRTC_BIG_ENDIAN
+ WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+#else
+ // We need to flip the characters in this case.
+ WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
+ + 2, err + 1, err);
+#endif
+}
+
+} // namespace webrtc
+// EOF
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h
new file mode 100644
index 0000000..7209f91
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
+
+#include "typedefs.h"
+#include "audio_device.h"
+#include "critical_section_wrapper.h"
+
+#include <CoreAudio/CoreAudio.h>
+
+namespace webrtc {
+
+class AudioMixerManagerMac
+{
+public:
+ WebRtc_Word32 OpenSpeaker(AudioDeviceID deviceID);
+ WebRtc_Word32 OpenMicrophone(AudioDeviceID deviceID);
+ WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetSpeakerMute(bool enable);
+ WebRtc_Word32 SpeakerMute(bool& enabled) const;
+ WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneMute(bool enable);
+ WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+ WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 Close();
+ WebRtc_Word32 CloseSpeaker();
+ WebRtc_Word32 CloseMicrophone();
+ bool SpeakerIsInitialized() const;
+ bool MicrophoneIsInitialized() const;
+
+public:
+ AudioMixerManagerMac(const WebRtc_Word32 id);
+ ~AudioMixerManagerMac();
+
+private:
+ static void logCAMsg(const TraceLevel level,
+ const TraceModule module,
+ const WebRtc_Word32 id, const char *msg,
+ const char *err);
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+
+ AudioDeviceID _inputDeviceID;
+ AudioDeviceID _outputDeviceID;
+
+ WebRtc_UWord16 _noInputChannels;
+ WebRtc_UWord16 _noOutputChannels;
+
+};
+
+} //namespace webrtc
+
+#endif // AUDIO_MIXER_MAC_H
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h
new file mode 100644
index 0000000..f689622
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h
@@ -0,0 +1,127 @@
+/*
+ * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
+ * Portable Audio I/O Library
+ * Memory barrier utilities
+ *
+ * Author: Bjorn Roche, XO Audio, LLC
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/**
+ @file pa_memorybarrier.h
+ @ingroup common_src
+*/
+
+/****************
+ * Some memory barrier primitives based on the system.
+ * right now only OS X, FreeBSD, and Linux are supported. In addition to providing
+ * memory barriers, these functions should ensure that data cached in registers
+ * is written out to cache where it can be snooped by other CPUs. (ie, the volatile
+ * keyword should not be required)
+ *
+ * the primitives that must be defined are:
+ *
+ * PaUtil_FullMemoryBarrier()
+ * PaUtil_ReadMemoryBarrier()
+ * PaUtil_WriteMemoryBarrier()
+ *
+ ****************/
+
+#if defined(__APPLE__)
+# include <libkern/OSAtomic.h>
+ /* Here are the memory barrier functions. Mac OS X only provides
+ full memory barriers, so the three types of barriers are the same,
+ however, these barriers are superior to compiler-based ones. */
+# define PaUtil_FullMemoryBarrier() OSMemoryBarrier()
+# define PaUtil_ReadMemoryBarrier() OSMemoryBarrier()
+# define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
+#elif defined(__GNUC__)
+ /* GCC >= 4.1 has built-in intrinsics. We'll use those */
+# if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
+# define PaUtil_FullMemoryBarrier() __sync_synchronize()
+# define PaUtil_ReadMemoryBarrier() __sync_synchronize()
+# define PaUtil_WriteMemoryBarrier() __sync_synchronize()
+ /* as a fallback, GCC understands volatile asm and "memory" to mean it
+ * should not reorder memory read/writes */
+ /* Note that it is not clear that any compiler actually defines __PPC__,
+ * it can probably removed safely. */
+# elif defined( __ppc__ ) || defined( __powerpc__) || defined( __PPC__ )
+# define PaUtil_FullMemoryBarrier() asm volatile("sync":::"memory")
+# define PaUtil_ReadMemoryBarrier() asm volatile("sync":::"memory")
+# define PaUtil_WriteMemoryBarrier() asm volatile("sync":::"memory")
+# elif defined( __i386__ ) || defined( __i486__ ) || defined( __i586__ ) || \
+ defined( __i686__ ) || defined( __x86_64__ )
+# define PaUtil_FullMemoryBarrier() asm volatile("mfence":::"memory")
+# define PaUtil_ReadMemoryBarrier() asm volatile("lfence":::"memory")
+# define PaUtil_WriteMemoryBarrier() asm volatile("sfence":::"memory")
+# else
+# ifdef ALLOW_SMP_DANGERS
+# warning Memory barriers not defined on this system or system unknown
+# warning For SMP safety, you should fix this.
+# define PaUtil_FullMemoryBarrier()
+# define PaUtil_ReadMemoryBarrier()
+# define PaUtil_WriteMemoryBarrier()
+# else
+# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
+# endif
+# endif
+#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
+# include <intrin.h>
+# pragma intrinsic(_ReadWriteBarrier)
+# pragma intrinsic(_ReadBarrier)
+# pragma intrinsic(_WriteBarrier)
+# define PaUtil_FullMemoryBarrier() _ReadWriteBarrier()
+# define PaUtil_ReadMemoryBarrier() _ReadBarrier()
+# define PaUtil_WriteMemoryBarrier() _WriteBarrier()
+#elif defined(_WIN32_WCE)
+# define PaUtil_FullMemoryBarrier()
+# define PaUtil_ReadMemoryBarrier()
+# define PaUtil_WriteMemoryBarrier()
+#elif defined(_MSC_VER) || defined(__BORLANDC__)
+# define PaUtil_FullMemoryBarrier() _asm { lock add [esp], 0 }
+# define PaUtil_ReadMemoryBarrier() _asm { lock add [esp], 0 }
+# define PaUtil_WriteMemoryBarrier() _asm { lock add [esp], 0 }
+#else
+# ifdef ALLOW_SMP_DANGERS
+# warning Memory barriers not defined on this system or system unknown
+# warning For SMP safety, you should fix this.
+# define PaUtil_FullMemoryBarrier()
+# define PaUtil_ReadMemoryBarrier()
+# define PaUtil_WriteMemoryBarrier()
+# else
+# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
+# endif
+#endif
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c
new file mode 100644
index 0000000..310d719
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c
@@ -0,0 +1,227 @@
+/*
+ * $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
+ * Portable Audio I/O Library
+ * Ring Buffer utility.
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on Mac OS X by Bjorn Roche
+ * modified for SMP safety on Linux by Leland Lucius
+ * also, allowed for const where possible
+ * modified for multiple-byte-sized data elements by Sven Fischer
+ *
+ * Note that this is safe only for a single-thread reader and a
+ * single-thread writer.
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup common_src
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include "pa_ringbuffer.h"
+#include <string.h>
+#include "pa_memorybarrier.h"
+
+/***************************************************************************
+ * Initialize FIFO.
+ * elementCount must be power of 2, returns -1 if not.
+ */
+ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr )
+{
+ if( ((elementCount-1) & elementCount) != 0) return -1; /* Not Power of two. */
+ rbuf->bufferSize = elementCount;
+ rbuf->buffer = (char *)dataPtr;
+ PaUtil_FlushRingBuffer( rbuf );
+ rbuf->bigMask = (elementCount*2)-1;
+ rbuf->smallMask = (elementCount)-1;
+ rbuf->elementSizeBytes = elementSizeBytes;
+ return 0;
+}
+
+/***************************************************************************
+** Return number of elements available for reading. */
+ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf )
+{
+ PaUtil_ReadMemoryBarrier();
+ return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
+}
+/***************************************************************************
+** Return number of elements available for writing. */
+ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf )
+{
+ /* Since we are calling PaUtil_GetRingBufferReadAvailable, we don't need an aditional MB */
+ return ( rbuf->bufferSize - PaUtil_GetRingBufferReadAvailable(rbuf));
+}
+
+/***************************************************************************
+** Clear buffer. Should only be called when buffer is NOT being read. */
+void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf )
+{
+ rbuf->writeIndex = rbuf->readIndex = 0;
+}
+
+/***************************************************************************
+** Get address of region(s) to which we can write data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+ void **dataPtr1, ring_buffer_size_t *sizePtr1,
+ void **dataPtr2, ring_buffer_size_t *sizePtr2 )
+{
+ ring_buffer_size_t index;
+ ring_buffer_size_t available = PaUtil_GetRingBufferWriteAvailable( rbuf );
+ if( elementCount > available ) elementCount = available;
+ /* Check to see if write is not contiguous. */
+ index = rbuf->writeIndex & rbuf->smallMask;
+ if( (index + elementCount) > rbuf->bufferSize )
+ {
+ /* Write data in two blocks that wrap the buffer. */
+ ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
+ *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+ *sizePtr1 = firstHalf;
+ *dataPtr2 = &rbuf->buffer[0];
+ *sizePtr2 = elementCount - firstHalf;
+ }
+ else
+ {
+ *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+ *sizePtr1 = elementCount;
+ *dataPtr2 = NULL;
+ *sizePtr2 = 0;
+ }
+ return elementCount;
+}
+
+
+/***************************************************************************
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
+{
+ /* we need to ensure that previous writes are seen before we update the write index */
+ PaUtil_WriteMemoryBarrier();
+ return rbuf->writeIndex = (rbuf->writeIndex + elementCount) & rbuf->bigMask;
+}
+
+/***************************************************************************
+** Get address of region(s) from which we can read data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+ void **dataPtr1, ring_buffer_size_t *sizePtr1,
+ void **dataPtr2, ring_buffer_size_t *sizePtr2 )
+{
+ ring_buffer_size_t index;
+ ring_buffer_size_t available = PaUtil_GetRingBufferReadAvailable( rbuf );
+ if( elementCount > available ) elementCount = available;
+ /* Check to see if read is not contiguous. */
+ index = rbuf->readIndex & rbuf->smallMask;
+ if( (index + elementCount) > rbuf->bufferSize )
+ {
+ /* Write data in two blocks that wrap the buffer. */
+ ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
+ *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+ *sizePtr1 = firstHalf;
+ *dataPtr2 = &rbuf->buffer[0];
+ *sizePtr2 = elementCount - firstHalf;
+ }
+ else
+ {
+ *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+ *sizePtr1 = elementCount;
+ *dataPtr2 = NULL;
+ *sizePtr2 = 0;
+ }
+ return elementCount;
+}
+/***************************************************************************
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
+{
+ /* we need to ensure that previous writes are always seen before updating the index. */
+ PaUtil_WriteMemoryBarrier();
+ return rbuf->readIndex = (rbuf->readIndex + elementCount) & rbuf->bigMask;
+}
+
+/***************************************************************************
+** Return elements written. */
+ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount )
+{
+ ring_buffer_size_t size1, size2, numWritten;
+ void *data1, *data2;
+ numWritten = PaUtil_GetRingBufferWriteRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
+ if( size2 > 0 )
+ {
+
+ memcpy( data1, data, size1*rbuf->elementSizeBytes );
+ data = ((char *)data) + size1*rbuf->elementSizeBytes;
+ memcpy( data2, data, size2*rbuf->elementSizeBytes );
+ }
+ else
+ {
+ memcpy( data1, data, size1*rbuf->elementSizeBytes );
+ }
+ PaUtil_AdvanceRingBufferWriteIndex( rbuf, numWritten );
+ return numWritten;
+}
+
+/***************************************************************************
+** Return elements read. */
+ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount )
+{
+ ring_buffer_size_t size1, size2, numRead;
+ void *data1, *data2;
+ numRead = PaUtil_GetRingBufferReadRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
+ if( size2 > 0 )
+ {
+ memcpy( data, data1, size1*rbuf->elementSizeBytes );
+ data = ((char *)data) + size1*rbuf->elementSizeBytes;
+ memcpy( data, data2, size2*rbuf->elementSizeBytes );
+ }
+ else
+ {
+ memcpy( data, data1, size1*rbuf->elementSizeBytes );
+ }
+ PaUtil_AdvanceRingBufferReadIndex( rbuf, numRead );
+ return numRead;
+}
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h
new file mode 100644
index 0000000..393f6f8
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h
@@ -0,0 +1,233 @@
+#ifndef WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
+#define WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
+/*
+ * $Id: pa_ringbuffer.h 1421 2009-11-18 16:09:05Z bjornroche $
+ * Portable Audio I/O Library
+ * Ring Buffer utility.
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on OS X by Bjorn Roche.
+ * also allowed for const where possible.
+ * modified for multiple-byte-sized data elements by Sven Fischer
+ *
+ * Note that this is safe only for a single-thread reader
+ * and a single-thread writer.
+ *
+ * This program is distributed with the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/** @file
+ @ingroup common_src
+ @brief Single-reader single-writer lock-free ring buffer
+
+ PaUtilRingBuffer is a ring buffer used to transport samples between
+ different execution contexts (threads, OS callbacks, interrupt handlers)
+ without requiring the use of any locks. This only works when there is
+ a single reader and a single writer (ie. one thread or callback writes
+ to the ring buffer, another thread or callback reads from it).
+
+ The PaUtilRingBuffer structure manages a ring buffer containing N
+ elements, where N must be a power of two. An element may be any size
+ (specified in bytes).
+
+ The memory area used to store the buffer elements must be allocated by
+ the client prior to calling PaUtil_InitializeRingBuffer() and must outlive
+ the use of the ring buffer.
+*/
+
+#if defined(__APPLE__)
+#include <sys/types.h>
+typedef int32_t ring_buffer_size_t;
+#elif defined( __GNUC__ )
+typedef long ring_buffer_size_t;
+#elif (_MSC_VER >= 1400)
+typedef long ring_buffer_size_t;
+#elif defined(_MSC_VER) || defined(__BORLANDC__)
+typedef long ring_buffer_size_t;
+#else
+typedef long ring_buffer_size_t;
+#endif
+
+
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+typedef struct PaUtilRingBuffer
+{
+ ring_buffer_size_t bufferSize; /**< Number of elements in FIFO. Power of 2. Set by PaUtil_InitRingBuffer. */
+ ring_buffer_size_t writeIndex; /**< Index of next writable element. Set by PaUtil_AdvanceRingBufferWriteIndex. */
+ ring_buffer_size_t readIndex; /**< Index of next readable element. Set by PaUtil_AdvanceRingBufferReadIndex. */
+ ring_buffer_size_t bigMask; /**< Used for wrapping indices with extra bit to distinguish full/empty. */
+ ring_buffer_size_t smallMask; /**< Used for fitting indices to buffer. */
+ ring_buffer_size_t elementSizeBytes; /**< Number of bytes per element. */
+ char *buffer; /**< Pointer to the buffer containing the actual data. */
+}PaUtilRingBuffer;
+
+/** Initialize Ring Buffer.
+
+ @param rbuf The ring buffer.
+
+ @param elementSizeBytes The size of a single data element in bytes.
+
+ @param elementCount The number of elements in the buffer (must be power of 2).
+
+ @param dataPtr A pointer to a previously allocated area where the data
+ will be maintained. It must be elementCount*elementSizeBytes long.
+
+ @return -1 if elementCount is not a power of 2, otherwise 0.
+*/
+ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr );
+
+/** Clear buffer. Should only be called when buffer is NOT being read.
+
+ @param rbuf The ring buffer.
+*/
+void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf );
+
+/** Retrieve the number of elements available in the ring buffer for writing.
+
+ @param rbuf The ring buffer.
+
+ @return The number of elements available for writing.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf );
+
+/** Retrieve the number of elements available in the ring buffer for reading.
+
+ @param rbuf The ring buffer.
+
+ @return The number of elements available for reading.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf );
+
+/** Write data to the ring buffer.
+
+ @param rbuf The ring buffer.
+
+ @param data The address of new data to write to the buffer.
+
+ @param elementCount The number of elements to be written.
+
+ @return The number of elements written.
+*/
+ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount );
+
+/** Read data from the ring buffer.
+
+ @param rbuf The ring buffer.
+
+ @param data The address where the data should be stored.
+
+ @param elementCount The number of elements to be read.
+
+ @return The number of elements read.
+*/
+ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount );
+
+/** Get address of region(s) to which we can write data.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements desired.
+
+ @param dataPtr1 The address where the first (or only) region pointer will be
+ stored.
+
+ @param sizePtr1 The address where the first (or only) region length will be
+ stored.
+
+ @param dataPtr2 The address where the second region pointer will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @param sizePtr2 The address where the second region length will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @return The room available to be written or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+ void **dataPtr1, ring_buffer_size_t *sizePtr1,
+ void **dataPtr2, ring_buffer_size_t *sizePtr2 );
+
+/** Advance the write index to the next location to be written.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements to advance.
+
+ @return The new position.
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
+
+/** Get address of region(s) from which we can write data.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements desired.
+
+ @param dataPtr1 The address where the first (or only) region pointer will be
+ stored.
+
+ @param sizePtr1 The address where the first (or only) region length will be
+ stored.
+
+ @param dataPtr2 The address where the second region pointer will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @param sizePtr2 The address where the second region length will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @return The number of elements available for reading.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+ void **dataPtr1, ring_buffer_size_t *sizePtr1,
+ void **dataPtr2, ring_buffer_size_t *sizePtr2 );
+
+/** Advance the read index to the next location to be read.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements to advance.
+
+ @return The new position.
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif /* MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_PORTAUDIO_PA_RINGBUFFER_H_ */
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc
new file mode 100644
index 0000000..d0418d2
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc
@@ -0,0 +1,5173 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma warning(disable: 4995) // name was marked as #pragma deprecated
+
+#if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
+// Reports the major and minor versions of the compiler.
+// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version 13 and a 1.0 point release.
+// The Visual C++ 2005 compiler version is 1400.
+// Type cl /? at the command line to see the major and minor versions of your compiler along with the build number.
+#pragma message(">> INFO: Windows Core Audio is not supported in VS 2003")
+#endif
+
+#include "audio_device_config.h"
+
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+#pragma message(">> INFO: WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined")
+#else
+#pragma message(">> INFO: WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined")
+#endif
+
+#ifdef WEBRTC_WINDOWS_CORE_AUDIO_BUILD
+
+#include "audio_device_core_win.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include <windows.h>
+#include <comdef.h>
+#include <dmo.h>
+#include "Functiondiscoverykeys_devpkey.h"
+#include <mmsystem.h>
+#include <strsafe.h>
+#include <uuids.h>
+
+#include "audio_device_utility.h"
+#include "trace.h"
+
+// Macro that calls a COM method returning HRESULT value.
+#define EXIT_ON_ERROR(hres) do { if (FAILED(hres)) goto Exit; } while(0)
+
+// Macro that releases a COM object if not NULL.
+#define SAFE_RELEASE(p) do { if ((p)) { (p)->Release(); (p) = NULL; } } while(0)
+
+#define ROUND(x) ((x) >=0 ? (int)((x) + 0.5) : (int)((x) - 0.5))
+
+// REFERENCE_TIME time units per millisecond
+#define REFTIMES_PER_MILLISEC 10000
+
+typedef struct tagTHREADNAME_INFO
+{
+ DWORD dwType; // must be 0x1000
+ LPCSTR szName; // pointer to name (in user addr space)
+ DWORD dwThreadID; // thread ID (-1=caller thread)
+ DWORD dwFlags; // reserved for future use, must be zero
+} THREADNAME_INFO;
+
+namespace webrtc {
+namespace {
+
+enum { COM_THREADING_MODEL = COINIT_MULTITHREADED };
+
+enum
+{
+ kAecCaptureStreamIndex = 0,
+ kAecRenderStreamIndex = 1
+};
+
+// An implementation of IMediaBuffer, as required for
+// IMediaObject::ProcessOutput(). After consuming data provided by
+// ProcessOutput(), call SetLength() to update the buffer availability.
+//
+// Example implementation:
+// http://msdn.microsoft.com/en-us/library/dd376684(v=vs.85).aspx
+class MediaBufferImpl : public IMediaBuffer
+{
+public:
+ explicit MediaBufferImpl(DWORD maxLength)
+ : _data(new BYTE[maxLength]),
+ _length(0),
+ _maxLength(maxLength),
+ _refCount(0)
+ {}
+
+ // IMediaBuffer methods.
+ STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength))
+ {
+ if (!ppBuffer || !pcbLength)
+ {
+ return E_POINTER;
+ }
+
+ *ppBuffer = _data;
+ *pcbLength = _length;
+
+ return S_OK;
+ }
+
+ STDMETHOD(GetMaxLength(DWORD* pcbMaxLength))
+ {
+ if (!pcbMaxLength)
+ {
+ return E_POINTER;
+ }
+
+ *pcbMaxLength = _maxLength;
+ return S_OK;
+ }
+
+ STDMETHOD(SetLength(DWORD cbLength))
+ {
+ if (cbLength > _maxLength)
+ {
+ return E_INVALIDARG;
+ }
+
+ _length = cbLength;
+ return S_OK;
+ }
+
+ // IUnknown methods.
+ STDMETHOD_(ULONG, AddRef())
+ {
+ return InterlockedIncrement(&_refCount);
+ }
+
+ STDMETHOD(QueryInterface(REFIID riid, void** ppv))
+ {
+ if (!ppv)
+ {
+ return E_POINTER;
+ }
+ else if (riid != IID_IMediaBuffer && riid != IID_IUnknown)
+ {
+ return E_NOINTERFACE;
+ }
+
+ *ppv = static_cast<IMediaBuffer*>(this);
+ AddRef();
+ return S_OK;
+ }
+
+ STDMETHOD_(ULONG, Release())
+ {
+ LONG refCount = InterlockedDecrement(&_refCount);
+ if (refCount == 0)
+ {
+ delete this;
+ }
+
+ return refCount;
+ }
+
+private:
+ ~MediaBufferImpl()
+ {
+ delete [] _data;
+ }
+
+ BYTE* _data;
+ DWORD _length;
+ const DWORD _maxLength;
+ LONG _refCount;
+};
+} // namespace
+
+// ============================================================================
+// Static Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// CoreAudioIsSupported
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::CoreAudioIsSupported()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s", __FUNCTION__);
+
+ bool MMDeviceIsAvailable(false);
+ bool coreAudioIsSupported(false);
+
+ HRESULT hr(S_OK);
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR errorText[MAXERRORLENGTH];
+
+ // 1) Check if Windows version is Vista SP1 or later.
+ //
+ // CoreAudio is only available on Vista SP1 and later.
+ //
+ OSVERSIONINFOEX osvi;
+ DWORDLONG dwlConditionMask = 0;
+ int op = VER_LESS_EQUAL;
+
+ // Initialize the OSVERSIONINFOEX structure.
+ ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
+ osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+ osvi.dwMajorVersion = 6;
+ osvi.dwMinorVersion = 0;
+ osvi.wServicePackMajor = 0;
+ osvi.wServicePackMinor = 0;
+ osvi.wProductType = VER_NT_WORKSTATION;
+
+ // Initialize the condition mask.
+ VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
+
+ DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
+ VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
+ VER_PRODUCT_TYPE;
+
+ // Perform the test.
+ BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask,
+ dwlConditionMask);
+ if (isVistaRTMorXP != 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
+ "*** Windows Core Audio is only supported on Vista SP1 or later "
+ "=> will revert to the Wave API ***");
+ return false;
+ }
+
+ // 2) Initializes the COM library for use by the calling thread.
+
+ // The COM init wrapper sets the thread's concurrency model to MTA,
+ // and creates a new apartment for the thread if one is required. The
+ // wrapper also ensures that each call to CoInitializeEx is balanced
+ // by a corresponding call to CoUninitialize.
+ //
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ // Things will work even if an STA thread is calling this method but we
+ // want to ensure that MTA is used and therefore return false here.
+ return false;
+ }
+
+ // 3) Check if the MMDevice API is available.
+ //
+ // The Windows Multimedia Device (MMDevice) API enables audio clients to
+ // discover audio endpoint devices, determine their capabilities, and create
+ // driver instances for those devices.
+ // Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
+ // The MMDevice API consists of several interfaces. The first of these is the
+ // IMMDeviceEnumerator interface. To access the interfaces in the MMDevice API,
+ // a client obtains a reference to the IMMDeviceEnumerator interface of a
+ // device-enumerator object by calling the CoCreateInstance function.
+ //
+ // Through the IMMDeviceEnumerator interface, the client can obtain references
+ // to the other interfaces in the MMDevice API. The MMDevice API implements
+ // the following interfaces:
+ //
+ // IMMDevice Represents an audio device.
+ // IMMDeviceCollection Represents a collection of audio devices.
+ // IMMDeviceEnumerator Provides methods for enumerating audio devices.
+ // IMMEndpoint Represents an audio endpoint device.
+ //
+ IMMDeviceEnumerator* pIMMD(NULL);
+ const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
+ const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
+
+ hr = CoCreateInstance(
+ CLSID_MMDeviceEnumerator, // GUID value of MMDeviceEnumerator coclass
+ NULL,
+ CLSCTX_ALL,
+ IID_IMMDeviceEnumerator, // GUID value of the IMMDeviceEnumerator interface
+ (void**)&pIMMD );
+
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
+ "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to create the required COM object", hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
+ "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) failed (hr=0x%x)", hr);
+
+ const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
+ FORMAT_MESSAGE_IGNORE_INSERTS;
+ const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+
+ // Gets the system's human readable message string for this HRESULT.
+ // All error message in English by default.
+ DWORD messageLength = ::FormatMessageW(dwFlags,
+ 0,
+ hr,
+ dwLangID,
+ errorText,
+ MAXERRORLENGTH,
+ NULL);
+
+ assert(messageLength <= MAXERRORLENGTH);
+
+ // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+ for (; messageLength && ::isspace(errorText[messageLength - 1]);
+ --messageLength)
+ {
+ errorText[messageLength - 1] = '\0';
+ }
+
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ StringCchCat(buf, MAXERRORLENGTH, errorText);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%S", buf);
+ }
+ else
+ {
+ MMDeviceIsAvailable = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
+ "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) succeeded", hr);
+ SAFE_RELEASE(pIMMD);
+ }
+
+ // 4) Verify that we can create and initialize our Core Audio class.
+ //
+ // Also, perform a limited "API test" to ensure that Core Audio is supported for all devices.
+ //
+ if (MMDeviceIsAvailable)
+ {
+ coreAudioIsSupported = false;
+
+ AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore(-1);
+ if (p == NULL)
+ {
+ return false;
+ }
+
+ int ok(0);
+ int temp_ok(0);
+ bool available(false);
+
+ ok |= p->Init();
+
+ WebRtc_Word16 numDevsRec = p->RecordingDevices();
+ for (WebRtc_UWord16 i = 0; i < numDevsRec; i++)
+ {
+ ok |= p->SetRecordingDevice(i);
+ temp_ok = p->RecordingIsAvailable(available);
+ ok |= temp_ok;
+ ok |= (available == false);
+ if (available)
+ {
+ ok |= p->InitMicrophone();
+ }
+ if (ok)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
+ "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Recording for device id=%i", i);
+ }
+ }
+
+ WebRtc_Word16 numDevsPlay = p->PlayoutDevices();
+ for (WebRtc_UWord16 i = 0; i < numDevsPlay; i++)
+ {
+ ok |= p->SetPlayoutDevice(i);
+ temp_ok = p->PlayoutIsAvailable(available);
+ ok |= temp_ok;
+ ok |= (available == false);
+ if (available)
+ {
+ ok |= p->InitSpeaker();
+ }
+ if (ok)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1 ,
+ "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Playout for device id=%i", i);
+ }
+ }
+
+ ok |= p->Terminate();
+
+ if (ok == 0)
+ {
+ coreAudioIsSupported = true;
+ }
+
+ delete p;
+ }
+
+ if (coreAudioIsSupported)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is supported ***");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is NOT supported => will revert to the Wave API ***");
+ }
+
+ return (coreAudioIsSupported);
+}
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceWindowsCore() - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsCore::AudioDeviceWindowsCore(const WebRtc_Word32 id) :
+ _comInit(ScopedCOMInitializer::kMTA),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _volumeMutex(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _ptrAudioBuffer(NULL),
+ _ptrEnumerator(NULL),
+ _ptrRenderCollection(NULL),
+ _ptrCaptureCollection(NULL),
+ _ptrDeviceOut(NULL),
+ _ptrDeviceIn(NULL),
+ _ptrClientOut(NULL),
+ _ptrClientIn(NULL),
+ _ptrRenderClient(NULL),
+ _ptrCaptureClient(NULL),
+ _ptrCaptureVolume(NULL),
+ _ptrRenderSimpleVolume(NULL),
+ _dmo(NULL),
+ _mediaBuffer(NULL),
+ _builtInAecEnabled(false),
+ _playAudioFrameSize(0),
+ _playSampleRate(0),
+ _playBlockSize(0),
+ _playChannels(2),
+ _sndCardPlayDelay(0),
+ _sndCardRecDelay(0),
+ _sampleDriftAt48kHz(0),
+ _driftAccumulator(0),
+ _writtenSamples(0),
+ _readSamples(0),
+ _playAcc(0),
+ _recAudioFrameSize(0),
+ _recSampleRate(0),
+ _recBlockSize(0),
+ _recChannels(2),
+ _avrtLibrary(NULL),
+ _winSupportAvrt(false),
+ _hRenderSamplesReadyEvent(NULL),
+ _hPlayThread(NULL),
+ _hCaptureSamplesReadyEvent(NULL),
+ _hRecThread(NULL),
+ _hShutdownRenderEvent(NULL),
+ _hShutdownCaptureEvent(NULL),
+ _hRenderStartedEvent(NULL),
+ _hCaptureStartedEvent(NULL),
+ _hGetCaptureVolumeThread(NULL),
+ _hSetCaptureVolumeThread(NULL),
+ _hSetCaptureVolumeEvent(NULL),
+ _hMmTask(NULL),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _speakerIsInitialized(false),
+ _microphoneIsInitialized(false),
+ _AGC(false),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
+ _playBufDelay(80),
+ _playBufDelayFixed(80),
+ _usingInputDeviceIndex(false),
+ _usingOutputDeviceIndex(false),
+ _inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
+ _outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _newMicLevel(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+ assert(_comInit.succeeded());
+
+ // Try to load the Avrt DLL
+ if (!_avrtLibrary)
+ {
+ // Get handle to the Avrt DLL module.
+ _avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
+ if (_avrtLibrary)
+ {
+ // Handle is valid (should only happen if OS larger than vista & win7).
+ // Try to get the function addresses.
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() The Avrt DLL module is now loaded");
+
+ _PAvRevertMmThreadCharacteristics = (PAvRevertMmThreadCharacteristics)GetProcAddress(_avrtLibrary, "AvRevertMmThreadCharacteristics");
+ _PAvSetMmThreadCharacteristicsA = (PAvSetMmThreadCharacteristicsA)GetProcAddress(_avrtLibrary, "AvSetMmThreadCharacteristicsA");
+ _PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(_avrtLibrary, "AvSetMmThreadPriority");
+
+ if ( _PAvRevertMmThreadCharacteristics &&
+ _PAvSetMmThreadCharacteristicsA &&
+ _PAvSetMmThreadPriority)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvRevertMmThreadCharacteristics() is OK");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadCharacteristicsA() is OK");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadPriority() is OK");
+ _winSupportAvrt = true;
+ }
+ }
+ }
+
+ // Create our samples ready events - we want auto reset events that start in the not-signaled state.
+ // The state of an auto-reset event object remains signaled until a single waiting thread is released,
+ // at which time the system automatically sets the state to nonsignaled. If no threads are waiting,
+ // the event object's state remains signaled.
+ // (Except for _hShutdownCaptureEvent, which is used to shutdown multiple threads).
+ _hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
+ _hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+
+ _perfCounterFreq.QuadPart = 1;
+ _perfCounterFactor = 0.0;
+ _avgCPULoad = 0.0;
+
+ // list of number of channels to use on recording side
+ _recChannelsPrioList[0] = 2; // stereo is prio 1
+ _recChannelsPrioList[1] = 1; // mono is prio 2
+
+ // list of number of channels to use on playout side
+ _playChannelsPrioList[0] = 2; // stereo is prio 1
+ _playChannelsPrioList[1] = 1; // mono is prio 2
+
+ HRESULT hr;
+
+ // We know that this API will work since it has already been verified in
+ // CoreAudioIsSupported, hence no need to check for errors here as well.
+
+ // Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
+ // TODO(henrika): we should probably move this allocation to Init() instead
+ // and deallocate in Terminate() to make the implementation more symmetric.
+ CoCreateInstance(
+ __uuidof(MMDeviceEnumerator),
+ NULL,
+ CLSCTX_ALL,
+ __uuidof(IMMDeviceEnumerator),
+ reinterpret_cast<void**>(&_ptrEnumerator));
+ assert(NULL != _ptrEnumerator);
+
+ // DMO initialization for built-in WASAPI AEC.
+ {
+ IMediaObject* ptrDMO = NULL;
+ hr = CoCreateInstance(CLSID_CWMAudioAEC,
+ NULL,
+ CLSCTX_INPROC_SERVER,
+ IID_IMediaObject,
+ reinterpret_cast<void**>(&ptrDMO));
+ if (FAILED(hr) || ptrDMO == NULL)
+ {
+ // Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
+ // feature is prevented from being enabled.
+ _builtInAecEnabled = false;
+ _TraceCOMError(hr);
+ }
+ _dmo = ptrDMO;
+ SAFE_RELEASE(ptrDMO);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceWindowsCore() - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ // The IMMDeviceEnumerator is created during construction. Must release
+ // it here and not in Terminate() since we don't recreate it in Init().
+ SAFE_RELEASE(_ptrEnumerator);
+
+ _ptrAudioBuffer = NULL;
+
+ if (NULL != _hRenderSamplesReadyEvent)
+ {
+ CloseHandle(_hRenderSamplesReadyEvent);
+ _hRenderSamplesReadyEvent = NULL;
+ }
+
+ if (NULL != _hCaptureSamplesReadyEvent)
+ {
+ CloseHandle(_hCaptureSamplesReadyEvent);
+ _hCaptureSamplesReadyEvent = NULL;
+ }
+
+ if (NULL != _hRenderStartedEvent)
+ {
+ CloseHandle(_hRenderStartedEvent);
+ _hRenderStartedEvent = NULL;
+ }
+
+ if (NULL != _hCaptureStartedEvent)
+ {
+ CloseHandle(_hCaptureStartedEvent);
+ _hCaptureStartedEvent = NULL;
+ }
+
+ if (NULL != _hShutdownRenderEvent)
+ {
+ CloseHandle(_hShutdownRenderEvent);
+ _hShutdownRenderEvent = NULL;
+ }
+
+ if (NULL != _hShutdownCaptureEvent)
+ {
+ CloseHandle(_hShutdownCaptureEvent);
+ _hShutdownCaptureEvent = NULL;
+ }
+
+ if (NULL != _hSetCaptureVolumeEvent)
+ {
+ CloseHandle(_hSetCaptureVolumeEvent);
+ _hSetCaptureVolumeEvent = NULL;
+ }
+
+ if (_avrtLibrary)
+ {
+ BOOL freeOK = FreeLibrary(_avrtLibrary);
+ if (!freeOK)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() failed to free the loaded Avrt DLL module correctly");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() the Avrt DLL module is now unloaded");
+ }
+ }
+
+ delete &_critSect;
+ delete &_volumeMutex;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AttachAudioBuffer
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // Inform the AudioBuffer about default settings for this implementation.
+ // Set all values to zero here since the actual settings will be done by
+ // InitPlayout and InitRecording later.
+ _ptrAudioBuffer->SetRecordingSampleRate(0);
+ _ptrAudioBuffer->SetPlayoutSampleRate(0);
+ _ptrAudioBuffer->SetRecordingChannels(0);
+ _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kWindowsCoreAudio;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Init
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ // Enumerate all audio rendering and capturing endpoint devices.
+ // Note that, some of these will not be able to select by the user.
+ // The complete collection is for internal use only.
+ //
+ _EnumerateEndpointDevicesAll(eRender);
+ _EnumerateEndpointDevicesAll(eCapture);
+
+ _initialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Terminate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::Terminate()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_initialized) {
+ return 0;
+ }
+
+ _initialized = false;
+ _speakerIsInitialized = false;
+ _microphoneIsInitialized = false;
+ _playing = false;
+ _recording = false;
+
+ SAFE_RELEASE(_ptrRenderCollection);
+ SAFE_RELEASE(_ptrCaptureCollection);
+ SAFE_RELEASE(_ptrDeviceOut);
+ SAFE_RELEASE(_ptrDeviceIn);
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrRenderClient);
+ SAFE_RELEASE(_ptrCaptureClient);
+ SAFE_RELEASE(_ptrCaptureVolume);
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Initialized() const
+{
+ return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ if (_usingOutputDeviceIndex)
+ {
+ WebRtc_Word16 nDevices = PlayoutDevices();
+ if (_outputDeviceIndex > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
+ return -1;
+ }
+ }
+
+ WebRtc_Word32 ret(0);
+
+ SAFE_RELEASE(_ptrDeviceOut);
+ if (_usingOutputDeviceIndex)
+ {
+ // Refresh the selected rendering endpoint device using current index
+ ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
+ }
+ else
+ {
+ ERole role;
+ (_outputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
+ // Refresh the selected rendering endpoint device using role
+ ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
+ }
+
+ if (ret != 0 || (_ptrDeviceOut == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the rendering enpoint device");
+ SAFE_RELEASE(_ptrDeviceOut);
+ return -1;
+ }
+
+ IAudioSessionManager* pManager = NULL;
+ ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager),
+ CLSCTX_ALL,
+ NULL,
+ (void**)&pManager);
+ if (ret != 0 || pManager == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to initialize the render manager");
+ SAFE_RELEASE(pManager);
+ return -1;
+ }
+
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+ ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
+ if (ret != 0 || _ptrRenderSimpleVolume == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to initialize the render simple volume");
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+ return -1;
+ }
+ SAFE_RELEASE(pManager);
+
+ _speakerIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ available = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ if (_usingInputDeviceIndex)
+ {
+ WebRtc_Word16 nDevices = RecordingDevices();
+ if (_inputDeviceIndex > (nDevices - 1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
+ return -1;
+ }
+ }
+
+ WebRtc_Word32 ret(0);
+
+ SAFE_RELEASE(_ptrDeviceIn);
+ if (_usingInputDeviceIndex)
+ {
+ // Refresh the selected capture endpoint device using current index
+ ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
+ }
+ else
+ {
+ ERole role;
+ (_inputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
+ // Refresh the selected capture endpoint device using role
+ ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
+ }
+
+ if (ret != 0 || (_ptrDeviceIn == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the capturing enpoint device");
+ SAFE_RELEASE(_ptrDeviceIn);
+ return -1;
+ }
+
+ ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume),
+ CLSCTX_ALL,
+ NULL,
+ reinterpret_cast<void **>(&_ptrCaptureVolume));
+ if (ret != 0 || _ptrCaptureVolume == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to initialize the capture volume");
+ SAFE_RELEASE(_ptrCaptureVolume);
+ return -1;
+ }
+
+ _microphoneIsInitialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::SpeakerIsInitialized() const
+{
+
+ return (_speakerIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
+{
+
+ return (_microphoneIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioSessionManager* pManager = NULL;
+ ISimpleAudioVolume* pVolume = NULL;
+
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL, (void**)&pManager);
+ EXIT_ON_ERROR(hr);
+
+ hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
+ EXIT_ON_ERROR(hr);
+
+ float volume(0.0f);
+ hr = pVolume->GetMasterVolume(&volume);
+ if (FAILED(hr))
+ {
+ available = false;
+ }
+ available = true;
+
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(pVolume);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+ }
+
+ if (volume < (WebRtc_UWord32)MIN_CORE_SPEAKER_VOLUME ||
+ volume > (WebRtc_UWord32)MAX_CORE_SPEAKER_VOLUME)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+
+ // scale input volume to valid range (0.0 to 1.0)
+ const float fLevel = (float)volume/MAX_CORE_SPEAKER_VOLUME;
+ _volumeMutex.Enter();
+ hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel,NULL);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+ }
+
+ HRESULT hr = S_OK;
+ float fLevel(0.0f);
+
+ _volumeMutex.Enter();
+ hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ // scale input volume range [0.0,1.0] to valid output range
+ volume = static_cast<WebRtc_UWord32> (fLevel*MAX_CORE_SPEAKER_VOLUME);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
+{
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
+{
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+//
+// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
+// silence and 1.0 indicates full volume (no attenuation).
+// We add our (webrtc-internal) own max level to match the Wave API and
+// how it is used today in VoE.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ maxVolume = static_cast<WebRtc_UWord32> (MAX_CORE_SPEAKER_VOLUME);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (MIN_CORE_SPEAKER_VOLUME);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ stepSize = CORE_SPEAKER_VOLUME_STEP_SIZE;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Query the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume),
+ CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ if (FAILED(hr))
+ available = false;
+ else
+ available = true;
+
+ SAFE_RELEASE(pVolume);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Set the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ const BOOL mute(enable);
+ hr = pVolume->SetMute(mute, NULL);
+ EXIT_ON_ERROR(hr);
+
+ SAFE_RELEASE(pVolume);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const
+{
+
+ if (!_speakerIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Query the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ EXIT_ON_ERROR(hr);
+
+ enabled = (mute == TRUE) ? true : false;
+
+ SAFE_RELEASE(pVolume);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Query the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ if (FAILED(hr))
+ available = false;
+ else
+ available = true;
+
+ SAFE_RELEASE(pVolume);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneMute(bool enable)
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Set the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ const BOOL mute(enable);
+ hr = pVolume->SetMute(mute, NULL);
+ EXIT_ON_ERROR(hr);
+
+ SAFE_RELEASE(pVolume);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ // Query the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ EXIT_ON_ERROR(hr);
+
+ enabled = (mute == TRUE) ? true : false;
+
+ SAFE_RELEASE(pVolume);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ available = false;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneBoost(bool enable)
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneBoost(bool& enabled) const
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
+{
+
+ available = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetStereoRecording(bool enable)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (enable)
+ {
+ _recChannelsPrioList[0] = 2; // try stereo first
+ _recChannelsPrioList[1] = 1;
+ _recChannels = 2;
+ }
+ else
+ {
+ _recChannelsPrioList[0] = 1; // try mono first
+ _recChannelsPrioList[1] = 2;
+ _recChannels = 1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StereoRecording(bool& enabled) const
+{
+
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
+{
+
+ available = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (enable)
+ {
+ _playChannelsPrioList[0] = 2; // try stereo first
+ _playChannelsPrioList[1] = 1;
+ _playChannels = 2;
+ }
+ else
+ {
+ _playChannelsPrioList[0] = 1; // try mono first
+ _playChannelsPrioList[1] = 2;
+ _playChannels = 1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
+{
+
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetAGC
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetAGC(bool enable)
+{
+ CriticalSectionScoped lock(_critSect);
+ _AGC = enable;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// AGC
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::AGC() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return _AGC;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
+
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
+
+ float volume(0.0f);
+ hr = pVolume->GetMasterVolumeLevelScalar(&volume);
+ if (FAILED(hr))
+ {
+ available = false;
+ }
+ available = true;
+
+ SAFE_RELEASE(pVolume);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
+
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+ }
+
+ if (volume < static_cast<WebRtc_UWord32>(MIN_CORE_MICROPHONE_VOLUME) ||
+ volume > static_cast<WebRtc_UWord32>(MAX_CORE_MICROPHONE_VOLUME))
+ {
+ return -1;
+ }
+
+ HRESULT hr = S_OK;
+ // scale input volume to valid range (0.0 to 1.0)
+ const float fLevel = static_cast<float>(volume)/MAX_CORE_MICROPHONE_VOLUME;
+ _volumeMutex.Enter();
+ _ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+ }
+
+ HRESULT hr = S_OK;
+ float fLevel(0.0f);
+ volume = 0;
+ _volumeMutex.Enter();
+ hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ // scale input volume range [0.0,1.0] to valid output range
+ volume = static_cast<WebRtc_UWord32> (fLevel*MAX_CORE_MICROPHONE_VOLUME);
+
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+//
+// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
+// silence and 1.0 indicates full volume (no attenuation).
+// We add our (webrtc-internal) own max level to match the Wave API and
+// how it is used today in VoE.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ maxVolume = static_cast<WebRtc_UWord32> (MAX_CORE_MICROPHONE_VOLUME);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ minVolume = static_cast<WebRtc_UWord32> (MIN_CORE_MICROPHONE_VOLUME);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (!_microphoneIsInitialized)
+ {
+ return -1;
+ }
+
+ stepSize = CORE_MICROPHONE_VOLUME_STEP_SIZE;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceWindowsCore::PlayoutDevices()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_RefreshDeviceList(eRender) != -1)
+ {
+ return (_DeviceListCount(eRender));
+ }
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ // Get current number of available rendering endpoint devices and refresh the rendering collection.
+ UINT nDevices = PlayoutDevices();
+
+ if (index < 0 || index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ HRESULT hr(S_OK);
+
+ assert(_ptrRenderCollection != NULL);
+
+ // Select an endpoint rendering device given the specified index
+ SAFE_RELEASE(_ptrDeviceOut);
+ hr = _ptrRenderCollection->Item(
+ index,
+ &_ptrDeviceOut);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(_ptrDeviceOut);
+ return -1;
+ }
+
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ }
+
+ _usingOutputDeviceIndex = true;
+ _outputDeviceIndex = index;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ ERole role(eCommunications);
+
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ role = eConsole;
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ role = eCommunications;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Refresh the list of rendering endpoint devices
+ _RefreshDeviceList(eRender);
+
+ HRESULT hr(S_OK);
+
+ assert(_ptrEnumerator != NULL);
+
+ // Select an endpoint rendering device given the specified role
+ SAFE_RELEASE(_ptrDeviceOut);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(
+ eRender,
+ role,
+ &_ptrDeviceOut);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(_ptrDeviceOut);
+ return -1;
+ }
+
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ }
+
+ _usingOutputDeviceIndex = false;
+ _outputDevice = device;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ bool defaultCommunicationDevice(false);
+ const WebRtc_Word16 nDevices(PlayoutDevices()); // also updates the list of devices
+
+ // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
+ if (index == (WebRtc_UWord16)(-1))
+ {
+ defaultCommunicationDevice = true;
+ index = 0;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
+ }
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ HRESULT hr(S_OK);
+ WebRtc_Word32 ret(-1);
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (defaultCommunicationDevice)
+ {
+ ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName, bufferLen);
+ }
+ else
+ {
+ ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
+ }
+
+ if (ret == 0)
+ {
+ // Convert the endpoint device's friendly-name to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ }
+ }
+
+ // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
+ if (defaultCommunicationDevice)
+ {
+ ret = _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
+ }
+ else
+ {
+ ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
+ }
+
+ if (guid != NULL && ret == 0)
+ {
+ // Convert the endpoint device's ID string to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ }
+ }
+
+ return ret;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ bool defaultCommunicationDevice(false);
+ const WebRtc_Word16 nDevices(RecordingDevices()); // also updates the list of devices
+
+ // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
+ if (index == (WebRtc_UWord16)(-1))
+ {
+ defaultCommunicationDevice = true;
+ index = 0;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
+ }
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ HRESULT hr(S_OK);
+ WebRtc_Word32 ret(-1);
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (defaultCommunicationDevice)
+ {
+ ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName, bufferLen);
+ }
+ else
+ {
+ ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
+ }
+
+ if (ret == 0)
+ {
+ // Convert the endpoint device's friendly-name to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ }
+ }
+
+ // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
+ if (defaultCommunicationDevice)
+ {
+ ret = _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
+ }
+ else
+ {
+ ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
+ }
+
+ if (guid != NULL && ret == 0)
+ {
+ // Convert the endpoint device's ID string to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ }
+ }
+
+ return ret;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceWindowsCore::RecordingDevices()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_RefreshDeviceList(eCapture) != -1)
+ {
+ return (_DeviceListCount(eCapture));
+ }
+
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ // Get current number of available capture endpoint devices and refresh the capture collection.
+ UINT nDevices = RecordingDevices();
+
+ if (index < 0 || index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ HRESULT hr(S_OK);
+
+ assert(_ptrCaptureCollection != NULL);
+
+ // Select an endpoint capture device given the specified index
+ SAFE_RELEASE(_ptrDeviceIn);
+ hr = _ptrCaptureCollection->Item(
+ index,
+ &_ptrDeviceIn);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(_ptrDeviceIn);
+ return -1;
+ }
+
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ }
+
+ _usingInputDeviceIndex = true;
+ _inputDeviceIndex = index;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ ERole role(eCommunications);
+
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ role = eConsole;
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ role = eCommunications;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Refresh the list of capture endpoint devices
+ _RefreshDeviceList(eCapture);
+
+ HRESULT hr(S_OK);
+
+ assert(_ptrEnumerator != NULL);
+
+ // Select an endpoint capture device given the specified role
+ SAFE_RELEASE(_ptrDeviceIn);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(
+ eCapture,
+ role,
+ &_ptrDeviceIn);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(_ptrDeviceIn);
+ return -1;
+ }
+
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ }
+
+ _usingInputDeviceIndex = false;
+ _inputDevice = device;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the recording side
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
+ }
+
+ // Ensure that the updated rendering endpoint device is valid
+ if (_ptrDeviceOut == NULL)
+ {
+ return -1;
+ }
+
+ if (_builtInAecEnabled && _recIsInitialized)
+ {
+ // Ensure the correct render device is configured in case
+ // InitRecording() was called before InitPlayout().
+ if (SetDMOProperties() == -1)
+ {
+ return -1;
+ }
+ }
+
+ HRESULT hr = S_OK;
+ WAVEFORMATEX* pWfxOut = NULL;
+ WAVEFORMATEX Wfx;
+ WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+ // Create COM object with IAudioClient interface.
+ SAFE_RELEASE(_ptrClientOut);
+ hr = _ptrDeviceOut->Activate(
+ __uuidof(IAudioClient),
+ CLSCTX_ALL,
+ NULL,
+ (void**)&_ptrClientOut);
+ EXIT_ON_ERROR(hr);
+
+ // Retrieve the stream format that the audio engine uses for its internal
+ // processing (mixing) of shared-mode streams.
+ hr = _ptrClientOut->GetMixFormat(&pWfxOut);
+ if (SUCCEEDED(hr))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current rendering mix format:");
+ // format type
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxOut->wFormatTag, pWfxOut->wFormatTag);
+ // number of channels (i.e. mono, stereo...)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxOut->nChannels);
+ // sample rate
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxOut->nSamplesPerSec);
+ // for buffer estimation
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxOut->nAvgBytesPerSec);
+ // block size of data
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxOut->nBlockAlign);
+ // number of bits per sample of mono data
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxOut->wBitsPerSample);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxOut->cbSize);
+ }
+
+ // Set wave format
+ Wfx.wFormatTag = WAVE_FORMAT_PCM;
+ Wfx.wBitsPerSample = 16;
+ Wfx.cbSize = 0;
+
+ const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
+ hr = S_FALSE;
+
+ // Iterate over frequencies and channels, in order of priority
+ for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
+ {
+ for (int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++)
+ {
+ Wfx.nChannels = _playChannelsPrioList[chan];
+ Wfx.nSamplesPerSec = freqs[freq];
+ Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
+ Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
+ // If the method succeeds and the audio endpoint device supports the specified stream format,
+ // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
+ // it returns S_FALSE.
+ hr = _ptrClientOut->IsFormatSupported(
+ AUDCLNT_SHAREMODE_SHARED,
+ &Wfx,
+ &pWfxClosestMatch);
+ if (hr == S_OK)
+ {
+ break;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels=%d, nSamplesPerSec=%d is not supported",
+ Wfx.nChannels, Wfx.nSamplesPerSec);
+ }
+ }
+ if (hr == S_OK)
+ break;
+ }
+
+ // TODO(andrew): what happens in the event of failure in the above loop?
+ // Is _ptrClientOut->Initialize expected to fail?
+ // Same in InitRecording().
+ if (hr == S_OK)
+ {
+ _playAudioFrameSize = Wfx.nBlockAlign;
+ _playBlockSize = Wfx.nSamplesPerSec/100;
+ _playSampleRate = Wfx.nSamplesPerSec;
+ _devicePlaySampleRate = Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
+ _devicePlayBlockSize = Wfx.nSamplesPerSec/100;
+ if (_playBlockSize == 441)
+ {
+ _playSampleRate = 44000; // we are actually running at 44000 Hz and *not* 44100 Hz
+ _playBlockSize = 440; // adjust to size we can handle
+ }
+ _playChannels = Wfx.nChannels;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this rendering format:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.wFormatTag, Wfx.wFormatTag);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.nChannels);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.nSamplesPerSec);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.nAvgBytesPerSec);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.nBlockAlign);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.wBitsPerSample);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playAudioFrameSize: %d", _playAudioFrameSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playBlockSize : %d", _playBlockSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playChannels : %d", _playChannels);
+ }
+
+ _Get44kHzDrift();
+
+ // Create a rendering stream.
+ //
+ // ****************************************************************************
+ // For a shared-mode stream that uses event-driven buffering, the caller must
+ // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
+ // determines how large a buffer to allocate based on the scheduling period
+ // of the audio engine. Although the client's buffer processing thread is
+ // event driven, the basic buffer management process, as described previously,
+ // is unaltered.
+ // Each time the thread awakens, it should call IAudioClient::GetCurrentPadding
+ // to determine how much data to write to a rendering buffer or read from a capture
+ // buffer. In contrast to the two buffers that the Initialize method allocates
+ // for an exclusive-mode stream that uses event-driven buffering, a shared-mode
+ // stream requires a single buffer.
+ // ****************************************************************************
+ //
+ REFERENCE_TIME hnsBufferDuration = 0; // ask for minimum buffer size (default)
+ if (_devicePlaySampleRate == 44100)
+ {
+ // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
+ // There seems to be a larger risk of underruns for 44.1 compared
+ // with the default rate (48kHz). When using default, we set the requested
+ // buffer duration to 0, which sets the buffer to the minimum size
+ // required by the engine thread. The actual buffer size can then be
+ // read by GetBufferSize() and it is 20ms on most machines.
+ hnsBufferDuration = 30*10000;
+ }
+ hr = _ptrClientOut->Initialize(
+ AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK, // processing of the audio buffer by the client will be event driven
+ hnsBufferDuration, // requested buffer capacity as a time value (in 100-nanosecond units)
+ 0, // periodicity
+ &Wfx, // selected wave format
+ NULL); // session GUID
+
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
+ if (pWfxClosestMatch != NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "closest mix format: #channels=%d, samples/sec=%d, bits/sample=%d",
+ pWfxClosestMatch->nChannels, pWfxClosestMatch->nSamplesPerSec, pWfxClosestMatch->wBitsPerSample);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "no format suggested");
+ }
+ }
+ EXIT_ON_ERROR(hr);
+
+ if (_ptrAudioBuffer)
+ {
+ // Update the audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
+ _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8)_playChannels);
+ }
+ else
+ {
+ // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
+ // has been created, hence the AudioDeviceBuffer does not exist.
+ // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
+ }
+
+ // Get the actual size of the shared (endpoint buffer).
+ // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+ UINT bufferFrameCount(0);
+ hr = _ptrClientOut->GetBufferSize(
+ &bufferFrameCount);
+ if (SUCCEEDED(hr))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
+ bufferFrameCount, bufferFrameCount*_playAudioFrameSize);
+ }
+
+ // Set the event handle that the system signals when an audio buffer is ready
+ // to be processed by the client.
+ hr = _ptrClientOut->SetEventHandle(
+ _hRenderSamplesReadyEvent);
+ EXIT_ON_ERROR(hr);
+
+ // Get an IAudioRenderClient interface.
+ SAFE_RELEASE(_ptrRenderClient);
+ hr = _ptrClientOut->GetService(
+ __uuidof(IAudioRenderClient),
+ (void**)&_ptrRenderClient);
+ EXIT_ON_ERROR(hr);
+
+ // Mark playout side as initialized
+ _playIsInitialized = true;
+
+ CoTaskMemFree(pWfxOut);
+ CoTaskMemFree(pWfxClosestMatch);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render side is now initialized");
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ CoTaskMemFree(pWfxOut);
+ CoTaskMemFree(pWfxClosestMatch);
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+ return -1;
+}
+
+// Capture initialization when the built-in AEC DirectX Media Object (DMO) is
+// used. Called from InitRecording(), most of which is skipped over. The DMO
+// handles device initialization itself.
+// Reference: http://msdn.microsoft.com/en-us/library/ff819492(v=vs.85).aspx
+WebRtc_Word32 AudioDeviceWindowsCore::InitRecordingDMO()
+{
+ assert(_builtInAecEnabled);
+ assert(_dmo != NULL);
+
+ if (SetDMOProperties() == -1)
+ {
+ return -1;
+ }
+
+ DMO_MEDIA_TYPE mt = {0};
+ HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
+ if (FAILED(hr))
+ {
+ MoFreeMediaType(&mt);
+ _TraceCOMError(hr);
+ return -1;
+ }
+ mt.majortype = MEDIATYPE_Audio;
+ mt.subtype = MEDIASUBTYPE_PCM;
+ mt.formattype = FORMAT_WaveFormatEx;
+
+ // Supported formats
+ // nChannels: 1 (in AEC-only mode)
+ // nSamplesPerSec: 8000, 11025, 16000, 22050
+ // wBitsPerSample: 16
+ WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
+ ptrWav->wFormatTag = WAVE_FORMAT_PCM;
+ ptrWav->nChannels = 1;
+ // 16000 is the highest we can support with our resampler.
+ ptrWav->nSamplesPerSec = 16000;
+ ptrWav->nAvgBytesPerSec = 32000;
+ ptrWav->nBlockAlign = 2;
+ ptrWav->wBitsPerSample = 16;
+ ptrWav->cbSize = 0;
+
+ // Set the VoE format equal to the AEC output format.
+ _recAudioFrameSize = ptrWav->nBlockAlign;
+ _recSampleRate = ptrWav->nSamplesPerSec;
+ _recBlockSize = ptrWav->nSamplesPerSec / 100;
+ _recChannels = ptrWav->nChannels;
+
+ // Set the DMO output format parameters.
+ hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
+ MoFreeMediaType(&mt);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ if (_ptrAudioBuffer)
+ {
+ _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+ _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+ }
+ else
+ {
+ // Refer to InitRecording() for comments.
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "AudioDeviceBuffer must be attached before streaming can start");
+ }
+
+ _mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
+
+ // Optional, but if called, must be after media types are set.
+ hr = _dmo->AllocateStreamingResources();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ _recIsInitialized = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Capture side is now initialized");
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ if (QueryPerformanceFrequency(&_perfCounterFreq) == 0)
+ {
+ return -1;
+ }
+ _perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
+
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
+ }
+
+ // Ensure that the updated capturing endpoint device is valid
+ if (_ptrDeviceIn == NULL)
+ {
+ return -1;
+ }
+
+ if (_builtInAecEnabled)
+ {
+ // The DMO will configure the capture device.
+ return InitRecordingDMO();
+ }
+
+ HRESULT hr = S_OK;
+ WAVEFORMATEX* pWfxIn = NULL;
+ WAVEFORMATEX Wfx;
+ WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+ // Create COM object with IAudioClient interface.
+ SAFE_RELEASE(_ptrClientIn);
+ hr = _ptrDeviceIn->Activate(
+ __uuidof(IAudioClient),
+ CLSCTX_ALL,
+ NULL,
+ (void**)&_ptrClientIn);
+ EXIT_ON_ERROR(hr);
+
+ // Retrieve the stream format that the audio engine uses for its internal
+ // processing (mixing) of shared-mode streams.
+ hr = _ptrClientIn->GetMixFormat(&pWfxIn);
+ if (SUCCEEDED(hr))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current capturing mix format:");
+ // format type
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxIn->wFormatTag, pWfxIn->wFormatTag);
+ // number of channels (i.e. mono, stereo...)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxIn->nChannels);
+ // sample rate
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxIn->nSamplesPerSec);
+ // for buffer estimation
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxIn->nAvgBytesPerSec);
+ // block size of data
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxIn->nBlockAlign);
+ // number of bits per sample of mono data
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxIn->wBitsPerSample);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxIn->cbSize);
+ }
+
+ // Set wave format
+ Wfx.wFormatTag = WAVE_FORMAT_PCM;
+ Wfx.wBitsPerSample = 16;
+ Wfx.cbSize = 0;
+
+ const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
+ hr = S_FALSE;
+
+ // Iterate over frequencies and channels, in order of priority
+ for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
+ {
+ for (int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++)
+ {
+ Wfx.nChannels = _recChannelsPrioList[chan];
+ Wfx.nSamplesPerSec = freqs[freq];
+ Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
+ Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
+ // If the method succeeds and the audio endpoint device supports the specified stream format,
+ // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
+ // it returns S_FALSE.
+ hr = _ptrClientIn->IsFormatSupported(
+ AUDCLNT_SHAREMODE_SHARED,
+ &Wfx,
+ &pWfxClosestMatch);
+ if (hr == S_OK)
+ {
+ break;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels=%d, nSamplesPerSec=%d is not supported",
+ Wfx.nChannels, Wfx.nSamplesPerSec);
+ }
+ }
+ if (hr == S_OK)
+ break;
+ }
+
+ if (hr == S_OK)
+ {
+ _recAudioFrameSize = Wfx.nBlockAlign;
+ _recSampleRate = Wfx.nSamplesPerSec;
+ _recBlockSize = Wfx.nSamplesPerSec/100;
+ _recChannels = Wfx.nChannels;
+ if (_recBlockSize == 441)
+ {
+ _recSampleRate = 44000; // we are actually using 44000 Hz and *not* 44100 Hz
+ _recBlockSize = 440; // adjust to size we can handle
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this capturing format:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.wFormatTag, Wfx.wFormatTag);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.nChannels);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.nSamplesPerSec);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.nAvgBytesPerSec);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.nBlockAlign);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.wBitsPerSample);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recAudioFrameSize: %d", _recAudioFrameSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recBlockSize : %d", _recBlockSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recChannels : %d", _recChannels);
+ }
+
+ _Get44kHzDrift();
+
+ // Create a capturing stream.
+ hr = _ptrClientIn->Initialize(
+ AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK | // processing of the audio buffer by the client will be event driven
+ AUDCLNT_STREAMFLAGS_NOPERSIST, // volume and mute settings for an audio session will not persist across system restarts
+ 0, // required for event-driven shared mode
+ 0, // periodicity
+ &Wfx, // selected wave format
+ NULL); // session GUID
+
+
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
+ if (pWfxClosestMatch != NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "closest mix format: #channels=%d, samples/sec=%d, bits/sample=%d",
+ pWfxClosestMatch->nChannels, pWfxClosestMatch->nSamplesPerSec, pWfxClosestMatch->wBitsPerSample);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "no format suggested");
+ }
+ }
+ EXIT_ON_ERROR(hr);
+
+ if (_ptrAudioBuffer)
+ {
+ // Update the audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+ _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8)_recChannels);
+ }
+ else
+ {
+ // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
+ // has been created, hence the AudioDeviceBuffer does not exist.
+ // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
+ }
+
+ // Get the actual size of the shared (endpoint buffer).
+ // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+ UINT bufferFrameCount(0);
+ hr = _ptrClientIn->GetBufferSize(
+ &bufferFrameCount);
+ if (SUCCEEDED(hr))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
+ bufferFrameCount, bufferFrameCount*_recAudioFrameSize);
+ }
+
+ // Set the event handle that the system signals when an audio buffer is ready
+ // to be processed by the client.
+ hr = _ptrClientIn->SetEventHandle(
+ _hCaptureSamplesReadyEvent);
+ EXIT_ON_ERROR(hr);
+
+ // Get an IAudioCaptureClient interface.
+ SAFE_RELEASE(_ptrCaptureClient);
+ hr = _ptrClientIn->GetService(
+ __uuidof(IAudioCaptureClient),
+ (void**)&_ptrCaptureClient);
+ EXIT_ON_ERROR(hr);
+
+ // Mark capture side as initialized
+ _recIsInitialized = true;
+
+ CoTaskMemFree(pWfxIn);
+ CoTaskMemFree(pWfxClosestMatch);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "capture side is now initialized");
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ CoTaskMemFree(pWfxIn);
+ CoTaskMemFree(pWfxClosestMatch);
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// StartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StartRecording()
+{
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_hRecThread != NULL)
+ {
+ return 0;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ HRESULT hr = S_OK;
+ {
+ CriticalSectionScoped critScoped(_critSect);
+
+ // Create thread which will drive the capturing
+ LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
+ if (_builtInAecEnabled)
+ {
+ // Redirect to the DMO polling method.
+ lpStartAddress = WSAPICaptureThreadPollDMO;
+
+ if (!_playing)
+ {
+ // The DMO won't provide us captured output data unless we
+ // give it render data to process.
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Playout must be started before recording when using the "
+ "built-in AEC");
+ return -1;
+ }
+ }
+
+ assert(_hRecThread == NULL);
+ _hRecThread = CreateThread(NULL,
+ 0,
+ lpStartAddress,
+ this,
+ 0,
+ NULL);
+ if (_hRecThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to create the recording thread");
+ return -1;
+ }
+
+ // Set thread priority to highest possible
+ SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
+
+ assert(_hGetCaptureVolumeThread == NULL);
+ _hGetCaptureVolumeThread = CreateThread(NULL,
+ 0,
+ GetCaptureVolumeThread,
+ this,
+ 0,
+ NULL);
+ if (_hGetCaptureVolumeThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create the volume getter thread");
+ return -1;
+ }
+
+ assert(_hSetCaptureVolumeThread == NULL);
+ _hSetCaptureVolumeThread = CreateThread(NULL,
+ 0,
+ SetCaptureVolumeThread,
+ this,
+ 0,
+ NULL);
+ if (_hSetCaptureVolumeThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create the volume setter thread");
+ return -1;
+ }
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "capturing did not start up properly");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "capture audio stream has now started...");
+
+ _avgCPULoad = 0.0f;
+ _playAcc = 0;
+ _recording = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StopRecording()
+{
+ WebRtc_Word32 err = 0;
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ _Lock();
+
+ if (_hRecThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "no capturing stream is active => close down WASAPI only");
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
+ _recIsInitialized = false;
+ _recording = false;
+ _UnLock();
+ return 0;
+ }
+
+ // Stop the driving thread...
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "closing down the webrtc_core_audio_capture_thread...");
+ // Manual-reset event; it will remain signalled to stop all capture threads.
+ SetEvent(_hShutdownCaptureEvent);
+
+ _UnLock();
+ DWORD ret = WaitForSingleObject(_hRecThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to close down webrtc_core_audio_capture_thread");
+ err = -1;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "webrtc_core_audio_capture_thread is now closed");
+ }
+
+ ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ // the thread did not stop as it should
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to close down volume getter thread");
+ err = -1;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " volume getter thread is now closed");
+ }
+
+ ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ // the thread did not stop as it should
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to close down volume setter thread");
+ err = -1;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " volume setter thread is now closed");
+ }
+ _Lock();
+
+ ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
+ // Ensure that the thread has released these interfaces properly.
+ assert(err == -1 || _ptrClientIn == NULL);
+ assert(err == -1 || _ptrCaptureClient == NULL);
+
+ _recIsInitialized = false;
+ _recording = false;
+
+ // These will create thread leaks in the result of an error,
+ // but we can at least resume the call.
+ CloseHandle(_hRecThread);
+ _hRecThread = NULL;
+
+ CloseHandle(_hGetCaptureVolumeThread);
+ _hGetCaptureVolumeThread = NULL;
+
+ CloseHandle(_hSetCaptureVolumeThread);
+ _hSetCaptureVolumeThread = NULL;
+
+ if (_builtInAecEnabled)
+ {
+ assert(_dmo != NULL);
+ // This is necessary. Otherwise the DMO can generate garbage render
+ // audio even after rendering has stopped.
+ HRESULT hr = _dmo->FreeStreamingResources();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ err = -1;
+ }
+ }
+
+ _UnLock();
+
+ // Reset the recording delay value.
+ _sndCardRecDelay = 0;
+
+ return err;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Recording() const
+{
+ return (_recording);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::PlayoutIsInitialized() const
+{
+
+ return (_playIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// StartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StartPlayout()
+{
+
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_hPlayThread != NULL)
+ {
+ return 0;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ HRESULT hr = S_OK;
+ {
+ CriticalSectionScoped critScoped(_critSect);
+
+ // Create thread which will drive the rendering.
+ assert(_hPlayThread == NULL);
+ _hPlayThread = CreateThread(
+ NULL,
+ 0,
+ WSAPIRenderThread,
+ this,
+ 0,
+ NULL);
+ if (_hPlayThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to create the playout thread");
+ return -1;
+ }
+
+ // Set thread priority to highest possible.
+ SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "rendering did not start up properly");
+ return -1;
+ }
+
+ _playing = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "rendering audio stream has now started...");
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::StopPlayout()
+{
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ {
+ CriticalSectionScoped critScoped(_critSect) ;
+
+ if (_hPlayThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "no rendering stream is active => close down WASAPI only");
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+ _playIsInitialized = false;
+ _playing = false;
+ return 0;
+ }
+
+ // stop the driving thread...
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "closing down the webrtc_core_audio_render_thread...");
+ SetEvent(_hShutdownRenderEvent);
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ // the thread did not stop as it should
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to close down webrtc_core_audio_render_thread");
+ CloseHandle(_hPlayThread);
+ _hPlayThread = NULL;
+ _playIsInitialized = false;
+ _playing = false;
+ return -1;
+ }
+
+ {
+ CriticalSectionScoped critScoped(_critSect);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "webrtc_core_audio_render_thread is now closed");
+
+ // to reset this event manually at each time we finish with it,
+ // in case that the render thread has exited before StopPlayout(),
+ // this event might be caught by the new render thread within same VoE instance.
+ ResetEvent(_hShutdownRenderEvent);
+
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+
+ _playIsInitialized = false;
+ _playing = false;
+
+ CloseHandle(_hPlayThread);
+ _hPlayThread = NULL;
+
+ if (_builtInAecEnabled && _recording)
+ {
+ // The DMO won't provide us captured output data unless we
+ // give it render data to process.
+ //
+ // We still permit the playout to shutdown, and trace a warning.
+ // Otherwise, VoE can get into a state which will never permit
+ // playout to stop properly.
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "Recording should be stopped before playout when using the "
+ "built-in AEC");
+ }
+ } // critScoped
+
+ // Reset the playout delay value.
+ _sndCardPlayDelay = 0;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ delayMS = static_cast<WebRtc_UWord16>(_sndCardRecDelay.Value());
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Playing() const
+{
+ return (_playing);
+}
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _playBufType = type;
+
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ _playBufDelayFixed = sizeMS;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
+{
+ {
+ CriticalSectionScoped lock(_critSect);
+ type = _playBufType;
+ }
+
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ CriticalSectionScoped lock(_critSect);
+ sizeMS = _playBufDelayFixed;
+ }
+ else
+ {
+ // Use same value as for PlayoutDelay
+ sizeMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// CPULoad
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::CPULoad(WebRtc_UWord16& load) const
+{
+
+ load = static_cast<WebRtc_UWord16> (100*_avgCPULoad);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::PlayoutWarning() const
+{
+ return ( _playWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::PlayoutError() const
+{
+ return ( _playError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::RecordingWarning() const
+{
+ return ( _recWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::RecordingError() const
+{
+ return ( _recError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::ClearPlayoutWarning()
+{
+ _playWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::ClearPlayoutError()
+{
+ _playError = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::ClearRecordingWarning()
+{
+ _recWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::ClearRecordingError()
+{
+ _recError = 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// [static] WSAPIRenderThread
+// ----------------------------------------------------------------------------
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context)
+{
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
+ DoRenderThread();
+}
+
+// ----------------------------------------------------------------------------
+// [static] WSAPICaptureThread
+// ----------------------------------------------------------------------------
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context)
+{
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
+ DoCaptureThread();
+}
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context)
+{
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
+ DoCaptureThreadPollDMO();
+}
+
+DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context)
+{
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
+ DoGetCaptureVolumeThread();
+}
+
+DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context)
+{
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
+ DoSetCaptureVolumeThread();
+}
+
+DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread()
+{
+ HANDLE waitObject = _hShutdownCaptureEvent;
+
+ while (1)
+ {
+ if (AGC())
+ {
+ WebRtc_UWord32 currentMicLevel = 0;
+ if (MicrophoneVolume(currentMicLevel) == 0)
+ {
+ // This doesn't set the system volume, just stores it.
+ _Lock();
+ if (_ptrAudioBuffer)
+ {
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ _UnLock();
+ }
+ }
+
+ DWORD waitResult = WaitForSingleObject(waitObject,
+ GET_MIC_VOLUME_INTERVAL_MS);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ return 0;
+ case WAIT_TIMEOUT: // timeout notification
+ break;
+ default: // unexpected error
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unknown wait termination on get volume thread");
+ return -1;
+ }
+ }
+}
+
+DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
+{
+ HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
+
+ while (1)
+ {
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ return 0;
+ case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
+ break;
+ default: // unexpected error
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unknown wait termination on set volume thread");
+ return -1;
+ }
+
+ _Lock();
+ WebRtc_UWord32 newMicLevel = _newMicLevel;
+ _UnLock();
+
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " the required modification of the microphone volume failed");
+ }
+ }
+}
+
+// ----------------------------------------------------------------------------
+// DoRenderThread
+// ----------------------------------------------------------------------------
+
+DWORD AudioDeviceWindowsCore::DoRenderThread()
+{
+
+ bool keepPlaying = true;
+ HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
+ HRESULT hr = S_OK;
+ HANDLE hMmTask = NULL;
+
+ LARGE_INTEGER t1;
+ LARGE_INTEGER t2;
+ WebRtc_Word32 time(0);
+
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to initialize COM in render thread");
+ return -1;
+ }
+
+ _SetThreadName(-1, "webrtc_core_audio_render_thread");
+
+ // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority.
+ //
+ if (_winSupportAvrt)
+ {
+ DWORD taskIndex(0);
+ hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+ if (hMmTask)
+ {
+ if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to boost play-thread using MMCSS");
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render thread is now registered with MMCSS (taskIndex=%d)", taskIndex);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to enable MMCSS on render thread (err=%d)", GetLastError());
+ _TraceCOMError(GetLastError());
+ }
+ }
+
+ _Lock();
+
+ // Get size of rendering buffer (length is expressed as the number of audio frames the buffer can hold).
+ // This value is fixed during the rendering session.
+ //
+ UINT32 bufferLength = 0;
+ hr = _ptrClientOut->GetBufferSize(&bufferLength);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] size of buffer : %u", bufferLength);
+
+ // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
+ //
+ REFERENCE_TIME latency;
+ _ptrClientOut->GetStreamLatency(&latency);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] max stream latency : %u (%3.2f ms)",
+ (DWORD)latency, (double)(latency/10000.0));
+
+ // Get the length of the periodic interval separating successive processing passes by
+ // the audio engine on the data in the endpoint buffer.
+ //
+ // The period between processing passes by the audio engine is fixed for a particular
+ // audio endpoint device and represents the smallest processing quantum for the audio engine.
+ // This period plus the stream latency between the buffer and endpoint device represents
+ // the minimum possible latency that an audio application can achieve.
+ // Typical value: 100000 <=> 0.01 sec = 10ms.
+ //
+ REFERENCE_TIME devPeriod = 0;
+ REFERENCE_TIME devPeriodMin = 0;
+ _ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] device period : %u (%3.2f ms)",
+ (DWORD)devPeriod, (double)(devPeriod/10000.0));
+
+ // The Event Driven renderer will be woken up every defaultDevicePeriod hundred-nano-seconds.
+ // Convert that time into a number of frames.
+ //
+ double devicePeriodInSeconds = devPeriod / (10000.0*1000.0);
+ UINT32 devicePeriodInFrames = static_cast<UINT32>(_playSampleRate * devicePeriodInSeconds + 0.5);
+
+ // Derive inital rendering delay.
+ // Example: 10*(960/480) + 15 = 20 + 15 = 35ms
+ //
+ int playout_delay = 10 * (bufferLength / _playBlockSize) +
+ (int)((latency + devPeriod) / 10000);
+ _sndCardPlayDelay = playout_delay;
+ _writtenSamples = 0;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "[REND] initial delay : %u", playout_delay);
+
+ double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
+
+ // Before starting the stream, fill the rendering buffer with silence.
+ //
+ BYTE *pData = NULL;
+ hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
+ EXIT_ON_ERROR(hr);
+
+ hr = _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
+ EXIT_ON_ERROR(hr);
+
+ _writtenSamples += bufferLength;
+
+ IAudioClock* clock = NULL;
+ hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
+ if (FAILED(hr)) {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "failed to get IAudioClock interface from the IAudioClient");
+ }
+
+ // Start up the rendering audio stream.
+ hr = _ptrClientOut->Start();
+ EXIT_ON_ERROR(hr);
+
+ _UnLock();
+
+ // Set event which will ensure that the calling thread modifies the playing state to true.
+ //
+ SetEvent(_hRenderStartedEvent);
+
+ // >> ------------------ THREAD LOOP ------------------
+
+ while (keepPlaying)
+ {
+ // Wait for a render notification event or a shutdown event
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0 + 0: // _hShutdownRenderEvent
+ keepPlaying = false;
+ break;
+ case WAIT_OBJECT_0 + 1: // _hRenderSamplesReadyEvent
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ _ptrClientOut->Stop();
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "render event timed out after 0.5 seconds");
+ goto Exit;
+ default: // unexpected error
+ _ptrClientOut->Stop();
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on render side");
+ goto Exit;
+ }
+
+ while (keepPlaying)
+ {
+ _Lock();
+
+ // Get the number of frames of padding (queued up to play) in the endpoint buffer.
+ UINT32 padding = 0;
+ hr = _ptrClientOut->GetCurrentPadding(&padding);
+ EXIT_ON_ERROR(hr);
+
+ // Derive the amount of available space in the output buffer
+ WebRtc_UWord32 framesAvailable = bufferLength - padding;
+ // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "#avaliable audio frames = %u", framesAvailable);
+
+ // Do we have 10 ms available in the render buffer?
+ if (framesAvailable < _playBlockSize)
+ {
+ // Not enough space in render buffer to store next render packet.
+ _UnLock();
+ break;
+ }
+
+ // Write n*10ms buffers to the render buffer
+ const WebRtc_UWord32 n10msBuffers = (framesAvailable / _playBlockSize);
+ for (WebRtc_UWord32 n = 0; n < n10msBuffers; n++)
+ {
+ // Get pointer (i.e., grab the buffer) to next space in the shared render buffer.
+ hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
+ EXIT_ON_ERROR(hr);
+
+ QueryPerformanceCounter(&t1); // measure time: START
+
+ if (_ptrAudioBuffer)
+ {
+ // Request data to be played out (#bytes = _playBlockSize*_audioFrameSize)
+ _UnLock();
+ WebRtc_Word32 nSamples =
+ _ptrAudioBuffer->RequestPlayoutData(_playBlockSize);
+ _Lock();
+
+ if (nSamples == -1)
+ {
+ _UnLock();
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ "failed to read data from render client");
+ goto Exit;
+ }
+
+ // Sanity check to ensure that essential states are not modified during the unlocked period
+ if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
+ {
+ _UnLock();
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
+ goto Exit;
+ }
+ if (nSamples != _playBlockSize)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
+ }
+
+ // Get the actual (stored) data
+ nSamples = _ptrAudioBuffer->GetPlayoutData((WebRtc_Word8*)pData);
+ }
+
+ QueryPerformanceCounter(&t2); // measure time: STOP
+ time = (int)(t2.QuadPart-t1.QuadPart);
+ _playAcc += time;
+
+ DWORD dwFlags(0);
+ hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, dwFlags);
+ // See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
+ // for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
+ EXIT_ON_ERROR(hr);
+
+ _writtenSamples += _playBlockSize;
+ }
+
+ // Check the current delay on the playout side.
+ if (clock) {
+ UINT64 pos = 0;
+ UINT64 freq = 1;
+ clock->GetPosition(&pos, NULL);
+ clock->GetFrequency(&freq);
+ playout_delay = ROUND((double(_writtenSamples) /
+ _devicePlaySampleRate - double(pos) / freq) * 1000.0);
+ _sndCardPlayDelay = playout_delay;
+ }
+
+ _UnLock();
+ }
+ }
+
+ // ------------------ THREAD LOOP ------------------ <<
+
+ Sleep(static_cast<DWORD>(endpointBufferSizeMS+0.5));
+ hr = _ptrClientOut->Stop();
+
+Exit:
+ SAFE_RELEASE(clock);
+
+ if (FAILED(hr))
+ {
+ _UnLock();
+ _ptrClientOut->Stop();
+ _TraceCOMError(hr);
+ }
+
+ if (_winSupportAvrt)
+ {
+ if (NULL != hMmTask)
+ {
+ _PAvRevertMmThreadCharacteristics(hMmTask);
+ }
+ }
+
+ if (keepPlaying)
+ {
+ hr = _ptrClientOut->Stop();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ }
+ hr = _ptrClientOut->Reset();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ }
+
+ // Trigger callback from module process thread
+ _playError = 1;
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: rendering thread has ended pre-maturely");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Rendering thread is now terminated properly");
+ }
+
+ return (DWORD)hr;
+}
+
+DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
+{
+ HRESULT hr = S_OK;
+ _hMmTask = NULL;
+
+ _SetThreadName(-1, "webrtc_core_audio_capture_thread");
+
+ // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
+ // priority.
+ if (_winSupportAvrt)
+ {
+ DWORD taskIndex(0);
+ _hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+ if (_hMmTask)
+ {
+ if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "failed to boost rec-thread using MMCSS");
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "capture thread is now registered with MMCSS (taskIndex=%d)",
+ taskIndex);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "failed to enable MMCSS on capture thread (err=%d)",
+ GetLastError());
+ _TraceCOMError(GetLastError());
+ }
+ }
+
+ return S_OK;
+}
+
+void AudioDeviceWindowsCore::RevertCaptureThreadPriority()
+{
+ if (_winSupportAvrt)
+ {
+ if (NULL != _hMmTask)
+ {
+ _PAvRevertMmThreadCharacteristics(_hMmTask);
+ }
+ }
+
+ _hMmTask = NULL;
+}
+
+DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
+{
+ assert(_mediaBuffer != NULL);
+ bool keepRecording = true;
+
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to initialize COM in polling DMO thread");
+ return -1;
+ }
+
+ HRESULT hr = InitCaptureThreadPriority();
+ if (FAILED(hr))
+ {
+ return hr;
+ }
+
+ // Set event which will ensure that the calling thread modifies the
+ // recording state to true.
+ SetEvent(_hCaptureStartedEvent);
+
+ // >> ---------------------------- THREAD LOOP ----------------------------
+ while (keepRecording)
+ {
+ // Poll the DMO every 5 ms.
+ // (The same interval used in the Wave implementation.)
+ DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ keepRecording = false;
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ break;
+ default: // unexpected error
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "Unknown wait termination on capture side");
+ hr = -1; // To signal an error callback.
+ keepRecording = false;
+ break;
+ }
+
+ while (keepRecording)
+ {
+ CriticalSectionScoped critScoped(_critSect);
+
+ DWORD dwStatus = 0;
+ {
+ DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
+ dmoBuffer.pBuffer = _mediaBuffer;
+ dmoBuffer.pBuffer->AddRef();
+
+ // Poll the DMO for AEC processed capture data. The DMO will
+ // copy available data to |dmoBuffer|, and should only return
+ // 10 ms frames. The value of |dwStatus| should be ignored.
+ hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
+ SAFE_RELEASE(dmoBuffer.pBuffer);
+ dwStatus = dmoBuffer.dwStatus;
+ }
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ ULONG bytesProduced = 0;
+ BYTE* data;
+ // Get a pointer to the data buffer. This should be valid until
+ // the next call to ProcessOutput.
+ hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ // TODO(andrew): handle AGC.
+
+ if (bytesProduced > 0)
+ {
+ const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
+ // TODO(andrew): verify that this is always satisfied. It might
+ // be that ProcessOutput will try to return more than 10 ms if
+ // we fail to call it frequently enough.
+ assert(kSamplesProduced == _recBlockSize);
+ assert(sizeof(BYTE) == sizeof(WebRtc_Word8));
+ _ptrAudioBuffer->SetRecordedBuffer(
+ reinterpret_cast<WebRtc_Word8*>(data),
+ kSamplesProduced);
+ _ptrAudioBuffer->SetVQEData(0, 0, 0);
+
+ _UnLock(); // Release lock while making the callback.
+ _ptrAudioBuffer->DeliverRecordedData();
+ _Lock();
+ }
+
+ // Reset length to indicate buffer availability.
+ hr = _mediaBuffer->SetLength(0);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE))
+ {
+ // The DMO cannot currently produce more data. This is the
+ // normal case; otherwise it means the DMO had more than 10 ms
+ // of data available and ProcessOutput should be called again.
+ break;
+ }
+ }
+ }
+ // ---------------------------- THREAD LOOP ---------------------------- <<
+
+ RevertCaptureThreadPriority();
+
+ if (FAILED(hr))
+ {
+ // Trigger callback from module process thread
+ _recError = 1;
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
+ "kRecordingError message posted: capturing thread has ended "
+ "prematurely");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Capturing thread is now terminated properly");
+ }
+
+ return hr;
+}
+
+
+// ----------------------------------------------------------------------------
+// DoCaptureThread
+// ----------------------------------------------------------------------------
+
+DWORD AudioDeviceWindowsCore::DoCaptureThread()
+{
+
+ bool keepRecording = true;
+ HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
+ HRESULT hr = S_OK;
+ HANDLE hMmTask = NULL;
+
+ LARGE_INTEGER t1;
+ LARGE_INTEGER t2;
+ WebRtc_Word32 time(0);
+
+ BYTE* syncBuffer = NULL;
+ UINT32 syncBufIndex = 0;
+
+ WebRtc_UWord32 newMicLevel(0);
+ WebRtc_UWord32 currentMicLevel(0);
+
+ _readSamples = 0;
+
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "failed to initialize COM in capture thread");
+ return -1;
+ }
+
+ hr = InitCaptureThreadPriority();
+ if (FAILED(hr))
+ {
+ return hr;
+ }
+
+ _Lock();
+
+ // Get size of capturing buffer (length is expressed as the number of audio frames the buffer can hold).
+ // This value is fixed during the capturing session.
+ //
+ UINT32 bufferLength = 0;
+ hr = _ptrClientIn->GetBufferSize(&bufferLength);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of buffer : %u", bufferLength);
+
+ // Allocate memory for sync buffer.
+ // It is used for compensation between native 44.1 and internal 44.0 and
+ // for cases when the capture buffer is larger than 10ms.
+ //
+ const UINT32 syncBufferSize = 2*(bufferLength * _recAudioFrameSize);
+ syncBuffer = new BYTE[syncBufferSize];
+ if (syncBuffer == NULL)
+ {
+ return E_POINTER;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of sync buffer : %u [bytes]", syncBufferSize);
+
+ // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
+ //
+ REFERENCE_TIME latency;
+ _ptrClientIn->GetStreamLatency(&latency);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] max stream latency : %u (%3.2f ms)",
+ (DWORD)latency, (double)(latency / 10000.0));
+
+ // Get the length of the periodic interval separating successive processing passes by
+ // the audio engine on the data in the endpoint buffer.
+ //
+ REFERENCE_TIME devPeriod = 0;
+ REFERENCE_TIME devPeriodMin = 0;
+ _ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] device period : %u (%3.2f ms)",
+ (DWORD)devPeriod, (double)(devPeriod / 10000.0));
+
+ double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] extraDelayMS : %3.2f", extraDelayMS);
+
+ double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_recBlockSize);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
+
+ // Start up the capturing stream.
+ //
+ hr = _ptrClientIn->Start();
+ EXIT_ON_ERROR(hr);
+
+ _UnLock();
+
+ // Set event which will ensure that the calling thread modifies the recording state to true.
+ //
+ SetEvent(_hCaptureStartedEvent);
+
+ // >> ---------------------------- THREAD LOOP ----------------------------
+
+ while (keepRecording)
+ {
+ // Wait for a capture notification event or a shutdown event
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0 + 0: // _hShutdownCaptureEvent
+ keepRecording = false;
+ break;
+ case WAIT_OBJECT_0 + 1: // _hCaptureSamplesReadyEvent
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ _ptrClientIn->Stop();
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "capture event timed out after 0.5 seconds");
+ goto Exit;
+ default: // unexpected error
+ _ptrClientIn->Stop();
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on capture side");
+ goto Exit;
+ }
+
+ while (keepRecording)
+ {
+ BYTE *pData = 0;
+ UINT32 framesAvailable = 0;
+ DWORD flags = 0;
+ UINT64 recTime = 0;
+ UINT64 recPos = 0;
+
+ _Lock();
+
+ // Find out how much capture data is available
+ //
+ hr = _ptrCaptureClient->GetBuffer(&pData, // packet which is ready to be read by used
+ &framesAvailable, // #frames in the captured packet (can be zero)
+ &flags, // support flags (check)
+ &recPos, // device position of first audio frame in data packet
+ &recTime); // value of performance counter at the time of recording the first audio frame
+
+ if (SUCCEEDED(hr))
+ {
+ if (AUDCLNT_S_BUFFER_EMPTY == hr)
+ {
+ // Buffer was empty => start waiting for a new capture notification event
+ _UnLock();
+ break;
+ }
+
+ if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
+ {
+ // Treat all of the data in the packet as silence and ignore the actual data values.
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "AUDCLNT_BUFFERFLAGS_SILENT");
+ pData = NULL;
+ }
+
+ assert(framesAvailable != 0);
+
+ if (pData)
+ {
+ CopyMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], pData, framesAvailable*_recAudioFrameSize);
+ }
+ else
+ {
+ ZeroMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], framesAvailable*_recAudioFrameSize);
+ }
+ assert(syncBufferSize >= (syncBufIndex*_recAudioFrameSize)+framesAvailable*_recAudioFrameSize);
+
+ // Release the capture buffer
+ //
+ hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
+ EXIT_ON_ERROR(hr);
+
+ _readSamples += framesAvailable;
+ syncBufIndex += framesAvailable;
+
+ QueryPerformanceCounter(&t1);
+
+ // Get the current recording and playout delay.
+ WebRtc_UWord32 sndCardRecDelay = (WebRtc_UWord32)
+ (((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime)
+ / 10000) + (10*syncBufIndex) / _recBlockSize - 10);
+ WebRtc_UWord32 sndCardPlayDelay =
+ static_cast<WebRtc_UWord32>(_sndCardPlayDelay.Value());
+
+ _sndCardRecDelay = sndCardRecDelay;
+
+ while (syncBufIndex >= _recBlockSize)
+ {
+ if (_ptrAudioBuffer)
+ {
+ _ptrAudioBuffer->SetRecordedBuffer((const WebRtc_Word8*)syncBuffer, _recBlockSize);
+
+ _driftAccumulator += _sampleDriftAt48kHz;
+ const WebRtc_Word32 clockDrift =
+ static_cast<WebRtc_Word32>(_driftAccumulator);
+ _driftAccumulator -= clockDrift;
+
+ _ptrAudioBuffer->SetVQEData(sndCardPlayDelay,
+ sndCardRecDelay,
+ clockDrift);
+
+ QueryPerformanceCounter(&t1); // measure time: START
+
+ _UnLock(); // release lock while making the callback
+ _ptrAudioBuffer->DeliverRecordedData();
+ _Lock(); // restore the lock
+
+ QueryPerformanceCounter(&t2); // measure time: STOP
+
+ // Measure "average CPU load".
+ // Basically what we do here is to measure how many percent of our 10ms period
+ // is used for encoding and decoding. This value shuld be used as a warning indicator
+ // only and not seen as an absolute value. Running at ~100% will lead to bad QoS.
+ time = (int)(t2.QuadPart - t1.QuadPart);
+ _avgCPULoad = (float)(_avgCPULoad*.99 + (time + _playAcc) / (double)(_perfCounterFreq.QuadPart));
+ _playAcc = 0;
+
+ // Sanity check to ensure that essential states are not modified during the unlocked period
+ if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
+ {
+ _UnLock();
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
+ goto Exit;
+ }
+ }
+
+ // store remaining data which was not able to deliver as 10ms segment
+ MoveMemory(&syncBuffer[0], &syncBuffer[_recBlockSize*_recAudioFrameSize], (syncBufIndex-_recBlockSize)*_recAudioFrameSize);
+ syncBufIndex -= _recBlockSize;
+ sndCardRecDelay -= 10;
+ }
+
+ if (_AGC)
+ {
+ WebRtc_UWord32 newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0)
+ {
+ // The VQE will only deliver non-zero microphone levels when a change is needed.
+ // Set this new mic level (received from the observer as return value in the callback).
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AGC change of volume: new=%u", newMicLevel);
+ // We store this outside of the audio buffer to avoid
+ // having it overwritten by the getter thread.
+ _newMicLevel = newMicLevel;
+ SetEvent(_hSetCaptureVolumeEvent);
+ }
+ }
+ }
+ else
+ {
+ // If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the audio samples
+ // must wait for the next processing pass. The client might benefit from keeping a count
+ // of the failed GetBuffer calls. If GetBuffer returns this error repeatedly, the client
+ // can start a new processing loop after shutting down the current client by calling
+ // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio client.
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "IAudioCaptureClient::GetBuffer returned AUDCLNT_E_BUFFER_ERROR, hr = 0x%08X", hr);
+ goto Exit;
+ }
+
+ _UnLock();
+ }
+ }
+
+ // ---------------------------- THREAD LOOP ---------------------------- <<
+
+ hr = _ptrClientIn->Stop();
+
+Exit:
+ if (FAILED(hr))
+ {
+ _UnLock();
+ _ptrClientIn->Stop();
+ _TraceCOMError(hr);
+ }
+
+ RevertCaptureThreadPriority();
+
+ if (keepRecording)
+ {
+ if (_ptrClientIn != NULL)
+ {
+ hr = _ptrClientIn->Stop();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ }
+ hr = _ptrClientIn->Reset();
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ }
+ }
+
+ // Trigger callback from module process thread
+ _recError = 1;
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: capturing thread has ended pre-maturely");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Capturing thread is now terminated properly");
+ }
+
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
+
+ if (syncBuffer)
+ {
+ delete [] syncBuffer;
+ }
+
+ return (DWORD)hr;
+}
+
+int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable)
+{
+
+ if (_recIsInitialized)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Attempt to set Windows AEC with recording already initialized");
+ return -1;
+ }
+
+ if (_dmo == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Built-in AEC DMO was not initialized properly at create time");
+ return -1;
+ }
+
+ _builtInAecEnabled = enable;
+ return 0;
+}
+
+bool AudioDeviceWindowsCore::BuiltInAECIsEnabled() const
+{
+ return _builtInAecEnabled;
+}
+
+int AudioDeviceWindowsCore::SetDMOProperties()
+{
+ HRESULT hr = S_OK;
+ assert(_dmo != NULL);
+
+ scoped_refptr<IPropertyStore> ps;
+ {
+ IPropertyStore* ptrPS = NULL;
+ hr = _dmo->QueryInterface(IID_IPropertyStore,
+ reinterpret_cast<void**>(&ptrPS));
+ if (FAILED(hr) || ptrPS == NULL)
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ ps = ptrPS;
+ SAFE_RELEASE(ptrPS);
+ }
+
+ // Set the AEC system mode.
+ // SINGLE_CHANNEL_AEC - AEC processing only.
+ if (SetVtI4Property(ps,
+ MFPKEY_WMAAECMA_SYSTEM_MODE,
+ SINGLE_CHANNEL_AEC))
+ {
+ return -1;
+ }
+
+ // Set the AEC source mode.
+ // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
+ if (SetBoolProperty(ps,
+ MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
+ VARIANT_TRUE) == -1)
+ {
+ return -1;
+ }
+
+ // Enable the feature mode.
+ // This lets us override all the default processing settings below.
+ if (SetBoolProperty(ps,
+ MFPKEY_WMAAECMA_FEATURE_MODE,
+ VARIANT_TRUE) == -1)
+ {
+ return -1;
+ }
+
+ // Disable analog AGC (default enabled).
+ if (SetBoolProperty(ps,
+ MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
+ VARIANT_FALSE) == -1)
+ {
+ return -1;
+ }
+
+ // Disable noise suppression (default enabled).
+ // 0 - Disabled, 1 - Enabled
+ if (SetVtI4Property(ps,
+ MFPKEY_WMAAECMA_FEATR_NS,
+ 0) == -1)
+ {
+ return -1;
+ }
+
+ // Relevant parameters to leave at default settings:
+ // MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
+ // MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
+ // MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
+ // TODO(andrew): investigate decresing the length to 128 ms.
+ // MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
+ // 0 is automatic; defaults to 160 samples (or 10 ms frames at the
+ // selected 16 kHz) as long as mic array processing is disabled.
+ // MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
+ // MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
+
+ // Set the devices selected by VoE. If using a default device, we need to
+ // search for the device index.
+ int inDevIndex = _inputDeviceIndex;
+ int outDevIndex = _outputDeviceIndex;
+ if (!_usingInputDeviceIndex)
+ {
+ ERole role = eCommunications;
+ if (_inputDevice == AudioDeviceModule::kDefaultDevice)
+ {
+ role = eConsole;
+ }
+
+ if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1)
+ {
+ return -1;
+ }
+ }
+
+ if (!_usingOutputDeviceIndex)
+ {
+ ERole role = eCommunications;
+ if (_outputDevice == AudioDeviceModule::kDefaultDevice)
+ {
+ role = eConsole;
+ }
+
+ if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1)
+ {
+ return -1;
+ }
+ }
+
+ DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
+ static_cast<uint32_t>(0x0000ffff & inDevIndex);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Capture device index: %d, render device index: %d",
+ inDevIndex, outDevIndex);
+ if (SetVtI4Property(ps,
+ MFPKEY_WMAAECMA_DEVICE_INDEXES,
+ devIndex) == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+int AudioDeviceWindowsCore::SetBoolProperty(IPropertyStore* ptrPS,
+ REFPROPERTYKEY key,
+ VARIANT_BOOL value)
+{
+ PROPVARIANT pv;
+ PropVariantInit(&pv);
+ pv.vt = VT_BOOL;
+ pv.boolVal = value;
+ HRESULT hr = ptrPS->SetValue(key, pv);
+ PropVariantClear(&pv);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ return 0;
+}
+
+int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
+ REFPROPERTYKEY key,
+ LONG value)
+{
+ PROPVARIANT pv;
+ PropVariantInit(&pv);
+ pv.vt = VT_I4;
+ pv.lVal = value;
+ HRESULT hr = ptrPS->SetValue(key, pv);
+ PropVariantClear(&pv);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _RefreshDeviceList
+//
+// Creates a new list of endpoint rendering or capture devices after
+// deleting any previously created (and possibly out-of-date) list of
+// such devices.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ IMMDeviceCollection *pCollection = NULL;
+
+ assert(dir == eRender || dir == eCapture);
+ assert(_ptrEnumerator != NULL);
+
+ // Create a fresh list of devices using the specified direction
+ hr = _ptrEnumerator->EnumAudioEndpoints(
+ dir,
+ DEVICE_STATE_ACTIVE,
+ &pCollection);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
+
+ if (dir == eRender)
+ {
+ SAFE_RELEASE(_ptrRenderCollection);
+ _ptrRenderCollection = pCollection;
+ }
+ else
+ {
+ SAFE_RELEASE(_ptrCaptureCollection);
+ _ptrCaptureCollection = pCollection;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _DeviceListCount
+//
+// Gets a count of the endpoint rendering or capture devices in the
+// current list of such devices.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ UINT count = 0;
+
+ assert(eRender == dir || eCapture == dir);
+
+ if (eRender == dir && NULL != _ptrRenderCollection)
+ {
+ hr = _ptrRenderCollection->GetCount(&count);
+ }
+ else if (NULL != _ptrCaptureCollection)
+ {
+ hr = _ptrCaptureCollection->GetCount(&count);
+ }
+
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ return static_cast<WebRtc_Word16> (count);
+}
+
+// ----------------------------------------------------------------------------
+// _GetListDeviceName
+//
+// Gets the friendly name of an endpoint rendering or capture device
+// from the current list of such devices. The caller uses an index
+// into the list to identify the device.
+//
+// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
+// in _RefreshDeviceList().
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ IMMDevice *pDevice = NULL;
+
+ assert(dir == eRender || dir == eCapture);
+
+ if (eRender == dir && NULL != _ptrRenderCollection)
+ {
+ hr = _ptrRenderCollection->Item(index, &pDevice);
+ }
+ else if (NULL != _ptrCaptureCollection)
+ {
+ hr = _ptrCaptureCollection->Item(index, &pDevice);
+ }
+
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pDevice);
+ return -1;
+ }
+
+ WebRtc_Word32 res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
+}
+
+// ----------------------------------------------------------------------------
+// _GetDefaultDeviceName
+//
+// Gets the friendly name of an endpoint rendering or capture device
+// given a specified device role.
+//
+// Uses: _ptrEnumerator
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ IMMDevice *pDevice = NULL;
+
+ assert(dir == eRender || dir == eCapture);
+ assert(role == eConsole || role == eCommunications);
+ assert(_ptrEnumerator != NULL);
+
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(
+ dir,
+ role,
+ &pDevice);
+
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pDevice);
+ return -1;
+ }
+
+ WebRtc_Word32 res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
+}
+
+// ----------------------------------------------------------------------------
+// _GetListDeviceID
+//
+// Gets the unique ID string of an endpoint rendering or capture device
+// from the current list of such devices. The caller uses an index
+// into the list to identify the device.
+//
+// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
+// in _RefreshDeviceList().
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ IMMDevice *pDevice = NULL;
+
+ assert(dir == eRender || dir == eCapture);
+
+ if (eRender == dir && NULL != _ptrRenderCollection)
+ {
+ hr = _ptrRenderCollection->Item(index, &pDevice);
+ }
+ else if (NULL != _ptrCaptureCollection)
+ {
+ hr = _ptrCaptureCollection->Item(index, &pDevice);
+ }
+
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pDevice);
+ return -1;
+ }
+
+ WebRtc_Word32 res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
+}
+
+// ----------------------------------------------------------------------------
+// _GetDefaultDeviceID
+//
+// Gets the uniqe device ID of an endpoint rendering or capture device
+// given a specified device role.
+//
+// Uses: _ptrEnumerator
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ IMMDevice *pDevice = NULL;
+
+ assert(dir == eRender || dir == eCapture);
+ assert(role == eConsole || role == eCommunications);
+ assert(_ptrEnumerator != NULL);
+
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(
+ dir,
+ role,
+ &pDevice);
+
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pDevice);
+ return -1;
+ }
+
+ WebRtc_Word32 res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
+}
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
+ ERole role,
+ int* index)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr = S_OK;
+ WCHAR szDefaultDeviceID[MAX_PATH] = {0};
+ WCHAR szDeviceID[MAX_PATH] = {0};
+
+ const size_t kDeviceIDLength = sizeof(szDeviceID)/sizeof(szDeviceID[0]);
+ assert(kDeviceIDLength ==
+ sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
+
+ if (_GetDefaultDeviceID(dir,
+ role,
+ szDefaultDeviceID,
+ kDeviceIDLength) == -1)
+ {
+ return -1;
+ }
+
+ IMMDeviceCollection* collection = _ptrCaptureCollection;
+ if (dir == eRender)
+ {
+ collection = _ptrRenderCollection;
+ }
+
+ if (!collection)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Device collection not valid");
+ return -1;
+ }
+
+ UINT count = 0;
+ hr = collection->GetCount(&count);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ *index = -1;
+ for (UINT i = 0; i < count; i++)
+ {
+ memset(szDeviceID, 0, sizeof(szDeviceID));
+ scoped_refptr<IMMDevice> device;
+ {
+ IMMDevice* ptrDevice = NULL;
+ hr = collection->Item(i, &ptrDevice);
+ if (FAILED(hr) || ptrDevice == NULL)
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ device = ptrDevice;
+ SAFE_RELEASE(ptrDevice);
+ }
+
+ if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1)
+ {
+ return -1;
+ }
+
+ if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0)
+ {
+ // Found a match.
+ *index = i;
+ break;
+ }
+
+ }
+
+ if (*index == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Unable to find collection index for default device");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _GetDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
+ LPWSTR pszBuffer,
+ int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ static const WCHAR szDefault[] = L"<Device not available>";
+
+ HRESULT hr = E_FAIL;
+ IPropertyStore *pProps = NULL;
+ PROPVARIANT varName;
+
+ assert(pszBuffer != NULL);
+ assert(bufferLen > 0);
+
+ if (pDevice != NULL)
+ {
+ hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "IMMDevice::OpenPropertyStore failed, hr = 0x%08X", hr);
+ }
+ }
+
+ // Initialize container for property value.
+ PropVariantInit(&varName);
+
+ if (SUCCEEDED(hr))
+ {
+ // Get the endpoint device's friendly-name property.
+ hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "IPropertyStore::GetValue failed, hr = 0x%08X", hr);
+ }
+ }
+
+ if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt))
+ {
+ hr = E_FAIL;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "IPropertyStore::GetValue returned no value, hr = 0x%08X", hr);
+ }
+
+ if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt))
+ {
+ // The returned value is not a wide null terminated string.
+ hr = E_UNEXPECTED;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "IPropertyStore::GetValue returned unexpected type, hr = 0x%08X", hr);
+ }
+
+ if (SUCCEEDED(hr) && (varName.pwszVal != NULL))
+ {
+ // Copy the valid device name to the provided ouput buffer.
+ wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
+ }
+ else
+ {
+ // Failed to find the device name.
+ wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+ }
+
+ PropVariantClear(&varName);
+ SAFE_RELEASE(pProps);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _GetDeviceID
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ static const WCHAR szDefault[] = L"<Device not available>";
+
+ HRESULT hr = E_FAIL;
+ LPWSTR pwszID = NULL;
+
+ assert(pszBuffer != NULL);
+ assert(bufferLen > 0);
+
+ if (pDevice != NULL)
+ {
+ hr = pDevice->GetId(&pwszID);
+ }
+
+ if (hr == S_OK)
+ {
+ // Found the device ID.
+ wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
+ }
+ else
+ {
+ // Failed to find the device ID.
+ wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+ }
+
+ CoTaskMemFree(pwszID);
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _GetDefaultDevice
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ HRESULT hr(S_OK);
+
+ assert(_ptrEnumerator != NULL);
+
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(
+ dir,
+ role,
+ ppDevice);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _GetListDevice
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice)
+{
+ HRESULT hr(S_OK);
+
+ assert(_ptrEnumerator != NULL);
+
+ IMMDeviceCollection *pCollection = NULL;
+
+ hr = _ptrEnumerator->EnumAudioEndpoints(
+ dir,
+ DEVICE_STATE_ACTIVE, // only active endpoints are OK
+ &pCollection);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
+
+ hr = pCollection->Item(
+ index,
+ ppDevice);
+ if (FAILED(hr))
+ {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// _EnumerateEndpointDevicesAll
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ assert(_ptrEnumerator != NULL);
+
+ HRESULT hr = S_OK;
+ IMMDeviceCollection *pCollection = NULL;
+
+ // Generate a collection of audio endpoint devices in the system.
+ // Get states for *all* endpoint devices.
+ // Output: IMMDeviceCollection interface.
+ hr = _ptrEnumerator->EnumAudioEndpoints(
+ dataFlow, // data-flow direction (input parameter)
+ DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_NOTPRESENT | DEVICE_STATE_UNPLUGGED,
+ &pCollection); // release interface when done
+
+ EXIT_ON_ERROR(hr);
+
+ // use the IMMDeviceCollection interface...
+
+ UINT count;
+ IMMDevice *pEndpoint = NULL;
+ IPropertyStore *pProps = NULL;
+ IAudioEndpointVolume* pEndpointVolume = NULL;
+ LPWSTR pwszID = NULL;
+
+ // Retrieve a count of the devices in the device collection.
+ hr = pCollection->GetCount(&count);
+ EXIT_ON_ERROR(hr);
+ if (dataFlow == eRender)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#rendering endpoint devices (counting all): %u", count);
+ else if (dataFlow == eCapture)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#capturing endpoint devices (counting all): %u", count);
+
+ if (count == 0)
+ {
+ return 0;
+ }
+
+ // Each loop prints the name of an endpoint device.
+ for (ULONG i = 0; i < count; i++)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Endpoint %d:", i);
+
+ // Get pointer to endpoint number i.
+ // Output: IMMDevice interface.
+ hr = pCollection->Item(
+ i,
+ &pEndpoint);
+ EXIT_ON_ERROR(hr);
+
+ // use the IMMDevice interface of the specified endpoint device...
+
+ // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
+ hr = pEndpoint->GetId(&pwszID);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "ID string : %S", pwszID);
+
+ // Retrieve an interface to the device's property store.
+ // Output: IPropertyStore interface.
+ hr = pEndpoint->OpenPropertyStore(
+ STGM_READ,
+ &pProps);
+ EXIT_ON_ERROR(hr);
+
+ // use the IPropertyStore interface...
+
+ PROPVARIANT varName;
+ // Initialize container for property value.
+ PropVariantInit(&varName);
+
+ // Get the endpoint's friendly-name property.
+ // Example: "Speakers (Realtek High Definition Audio)"
+ hr = pProps->GetValue(
+ PKEY_Device_FriendlyName,
+ &varName);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", varName.pwszVal);
+
+ // Get the endpoint's current device state
+ DWORD dwState;
+ hr = pEndpoint->GetState(&dwState);
+ EXIT_ON_ERROR(hr);
+ if (dwState & DEVICE_STATE_ACTIVE)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : *ACTIVE*", dwState);
+ if (dwState & DEVICE_STATE_DISABLED)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : DISABLED", dwState);
+ if (dwState & DEVICE_STATE_NOTPRESENT)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : NOTPRESENT", dwState);
+ if (dwState & DEVICE_STATE_UNPLUGGED)
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : UNPLUGGED", dwState);
+
+ // Check the hardware volume capabilities.
+ DWORD dwHwSupportMask = 0;
+ hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL,
+ NULL, (void**)&pEndpointVolume);
+ EXIT_ON_ERROR(hr);
+ hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
+ EXIT_ON_ERROR(hr);
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
+ // The audio endpoint device supports a hardware volume control
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_VOLUME", dwHwSupportMask);
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
+ // The audio endpoint device supports a hardware mute control
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_MUTE", dwHwSupportMask);
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
+ // The audio endpoint device supports a hardware peak meter
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_METER", dwHwSupportMask);
+
+ // Check the channel count (#channels in the audio stream that enters or leaves the audio endpoint device)
+ UINT nChannelCount(0);
+ hr = pEndpointVolume->GetChannelCount(
+ &nChannelCount);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#channels : %u", nChannelCount);
+
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
+ {
+ // Get the volume range.
+ float fLevelMinDB(0.0);
+ float fLevelMaxDB(0.0);
+ float fVolumeIncrementDB(0.0);
+ hr = pEndpointVolume->GetVolumeRange(
+ &fLevelMinDB,
+ &fLevelMaxDB,
+ &fVolumeIncrementDB);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume range : %4.2f (min), %4.2f (max), %4.2f (inc) [dB]",
+ fLevelMinDB, fLevelMaxDB, fVolumeIncrementDB);
+
+ // The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is divided
+ // into n uniform intervals of size vinc = fVolumeIncrementDB, where
+ // n = (vmax ?vmin) / vinc.
+ // The values vmin, vmax, and vinc are measured in decibels. The client can set
+ // the volume level to one of n + 1 discrete values in the range from vmin to vmax.
+ int n = (int)((fLevelMaxDB-fLevelMinDB)/fVolumeIncrementDB);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#intervals : %d", n);
+
+ // Get information about the current step in the volume range.
+ // This method represents the volume level of the audio stream that enters or leaves
+ // the audio endpoint device as an index or "step" in a range of discrete volume levels.
+ // Output value nStepCount is the number of steps in the range. Output value nStep
+ // is the step index of the current volume level. If the number of steps is n = nStepCount,
+ // then step index nStep can assume values from 0 (minimum volume) to n ?1 (maximum volume).
+ UINT nStep(0);
+ UINT nStepCount(0);
+ hr = pEndpointVolume->GetVolumeStepInfo(
+ &nStep,
+ &nStepCount);
+ EXIT_ON_ERROR(hr);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume steps : %d (nStep), %d (nStepCount)", nStep, nStepCount);
+ }
+
+ CoTaskMemFree(pwszID);
+ pwszID = NULL;
+ PropVariantClear(&varName);
+ SAFE_RELEASE(pProps);
+ SAFE_RELEASE(pEndpoint);
+ SAFE_RELEASE(pEndpointVolume);
+ }
+ SAFE_RELEASE(pCollection);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ CoTaskMemFree(pwszID);
+ pwszID = NULL;
+ SAFE_RELEASE(pCollection);
+ SAFE_RELEASE(pEndpoint);
+ SAFE_RELEASE(pEndpointVolume);
+ SAFE_RELEASE(pProps);
+ return -1;
+}
+
+// ----------------------------------------------------------------------------
+// _TraceCOMError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
+{
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR errorText[MAXERRORLENGTH];
+
+ const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
+ FORMAT_MESSAGE_IGNORE_INSERTS;
+ const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+
+ // Gets the system's human readable message string for this HRESULT.
+ // All error message in English by default.
+ DWORD messageLength = ::FormatMessageW(dwFlags,
+ 0,
+ hr,
+ dwLangID,
+ errorText,
+ MAXERRORLENGTH,
+ NULL);
+
+ assert(messageLength <= MAXERRORLENGTH);
+
+ // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+ for (; messageLength && ::isspace(errorText[messageLength - 1]);
+ --messageLength)
+ {
+ errorText[messageLength - 1] = '\0';
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ "Core Audio method failed (hr=0x%x)", hr);
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ StringCchCat(buf, MAXERRORLENGTH, errorText);
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// _SetThreadName
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::_SetThreadName(DWORD dwThreadID, LPCSTR szThreadName)
+{
+ // See http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx for details on the code
+ // in this function. Name of article is "Setting a Thread Name (Unmanaged)".
+
+ THREADNAME_INFO info;
+ info.dwType = 0x1000;
+ info.szName = szThreadName;
+ info.dwThreadID = dwThreadID;
+ info.dwFlags = 0;
+
+ __try
+ {
+ RaiseException( 0x406D1388, 0, sizeof(info)/sizeof(DWORD), (ULONG_PTR *)&info );
+ }
+ __except (EXCEPTION_CONTINUE_EXECUTION)
+ {
+ }
+}
+
+// ----------------------------------------------------------------------------
+// _Get44kHzDrift
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::_Get44kHzDrift()
+{
+ // We aren't able to resample at 44.1 kHz. Instead we run at 44 kHz and push/pull
+ // from the engine faster to compensate. If only one direction is set to 44.1 kHz
+ // the result is indistinguishable from clock drift to the AEC. We can compensate
+ // internally if we inform the AEC about the drift.
+ _sampleDriftAt48kHz = 0;
+ _driftAccumulator = 0;
+
+ if (_playSampleRate == 44000 && _recSampleRate != 44000)
+ {
+ _sampleDriftAt48kHz = 480.0f/440;
+ }
+ else if(_playSampleRate != 44000 && _recSampleRate == 44000)
+ {
+ _sampleDriftAt48kHz = -480.0f/441;
+ }
+}
+
+// ----------------------------------------------------------------------------
+// WideToUTF8
+// ----------------------------------------------------------------------------
+
+char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const {
+#ifdef UNICODE
+ const size_t kStrLen = sizeof(_str);
+ memset(_str, 0, kStrLen);
+ // Get required size (in bytes) to be able to complete the conversion.
+ int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
+ if (required_size <= kStrLen)
+ {
+ // Process the entire input string, including the terminating null char.
+ if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
+ memset(_str, 0, kStrLen);
+ }
+ return _str;
+#else
+ return const_cast<char*>(src);
+#endif
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_WINDOWS_CORE_AUDIO_BUILD
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h
new file mode 100644
index 0000000..da37a8c
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h
@@ -0,0 +1,381 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
+
+#if (_MSC_VER >= 1400) // only include for VS 2005 and higher
+
+#include "audio_device_generic.h"
+
+#pragma once
+#include <wmcodecdsp.h> // CLSID_CWMAudioAEC
+ // (must be before audioclient.h)
+#include <Audioclient.h> // WASAPI
+#include <Audiopolicy.h>
+#include <avrt.h> // Avrt
+#include <endpointvolume.h>
+#include <mediaobj.h> // IMediaObject
+#include <Mmdeviceapi.h> // MMDevice
+
+#include "atomic32_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "scoped_refptr.h"
+
+// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
+#pragma comment( lib, "avrt.lib" )
+// AVRT function pointers
+typedef BOOL (WINAPI *PAvRevertMmThreadCharacteristics)(HANDLE);
+typedef HANDLE (WINAPI *PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD);
+typedef BOOL (WINAPI *PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY);
+
+namespace webrtc {
+
+const float MAX_CORE_SPEAKER_VOLUME = 255.0f;
+const float MIN_CORE_SPEAKER_VOLUME = 0.0f;
+const float MAX_CORE_MICROPHONE_VOLUME = 255.0f;
+const float MIN_CORE_MICROPHONE_VOLUME = 0.0f;
+const WebRtc_UWord16 CORE_SPEAKER_VOLUME_STEP_SIZE = 1;
+const WebRtc_UWord16 CORE_MICROPHONE_VOLUME_STEP_SIZE = 1;
+
+// Utility class which initializes COM in the constructor (STA or MTA),
+// and uninitializes COM in the destructor.
+class ScopedCOMInitializer {
+ public:
+ // Enum value provided to initialize the thread as an MTA instead of STA.
+ enum SelectMTA { kMTA };
+
+ // Constructor for STA initialization.
+ ScopedCOMInitializer() {
+ Initialize(COINIT_APARTMENTTHREADED);
+ }
+
+ // Constructor for MTA initialization.
+ explicit ScopedCOMInitializer(SelectMTA mta) {
+ Initialize(COINIT_MULTITHREADED);
+ }
+
+ ScopedCOMInitializer::~ScopedCOMInitializer() {
+ if (SUCCEEDED(hr_))
+ CoUninitialize();
+ }
+
+ bool succeeded() const { return SUCCEEDED(hr_); }
+
+ private:
+ void Initialize(COINIT init) {
+ hr_ = CoInitializeEx(NULL, init);
+ }
+
+ HRESULT hr_;
+
+ ScopedCOMInitializer(const ScopedCOMInitializer&);
+ void operator=(const ScopedCOMInitializer&);
+};
+
+
+class AudioDeviceWindowsCore : public AudioDeviceGeneric
+{
+public:
+ AudioDeviceWindowsCore(const WebRtc_Word32 id);
+ ~AudioDeviceWindowsCore();
+
+ static bool CoreAudioIsSupported();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+ virtual int32_t EnableBuiltInAEC(bool enable);
+ virtual bool BuiltInAECIsEnabled() const;
+
+public:
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private: // avrt function pointers
+ PAvRevertMmThreadCharacteristics _PAvRevertMmThreadCharacteristics;
+ PAvSetMmThreadCharacteristicsA _PAvSetMmThreadCharacteristicsA;
+ PAvSetMmThreadPriority _PAvSetMmThreadPriority;
+ HMODULE _avrtLibrary;
+ bool _winSupportAvrt;
+
+private: // thread functions
+ DWORD InitCaptureThreadPriority();
+ void RevertCaptureThreadPriority();
+ static DWORD WINAPI WSAPICaptureThread(LPVOID context);
+ DWORD DoCaptureThread();
+
+ static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context);
+ DWORD DoCaptureThreadPollDMO();
+
+ static DWORD WINAPI WSAPIRenderThread(LPVOID context);
+ DWORD DoRenderThread();
+
+ static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
+ DWORD DoGetCaptureVolumeThread();
+
+ static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
+ DWORD DoSetCaptureVolumeThread();
+
+ void _SetThreadName(DWORD dwThreadID, LPCSTR szThreadName);
+ void _Lock() { _critSect.Enter(); };
+ void _UnLock() { _critSect.Leave(); };
+
+private:
+ WebRtc_Word32 Id() {return _id;}
+
+private:
+ int SetDMOProperties();
+
+ int SetBoolProperty(IPropertyStore* ptrPS,
+ REFPROPERTYKEY key,
+ VARIANT_BOOL value);
+
+ int SetVtI4Property(IPropertyStore* ptrPS,
+ REFPROPERTYKEY key,
+ LONG value);
+
+ WebRtc_Word32 _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const;
+ void _TraceCOMError(HRESULT hr) const;
+
+ WebRtc_Word32 _RefreshDeviceList(EDataFlow dir);
+ WebRtc_Word16 _DeviceListCount(EDataFlow dir);
+ WebRtc_Word32 _GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
+ WebRtc_Word32 _GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
+ WebRtc_Word32 _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
+ WebRtc_Word32 _GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
+ WebRtc_Word32 _GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
+ WebRtc_Word32 _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index);
+ WebRtc_Word32 _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
+ WebRtc_Word32 _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice);
+ WebRtc_Word32 _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice);
+
+ void _Get44kHzDrift();
+
+ // Converts from wide-char to UTF-8 if UNICODE is defined.
+ // Does nothing if UNICODE is undefined.
+ char* WideToUTF8(const TCHAR* src) const;
+
+ WebRtc_Word32 InitRecordingDMO();
+
+private:
+ ScopedCOMInitializer _comInit;
+ AudioDeviceBuffer* _ptrAudioBuffer;
+ CriticalSectionWrapper& _critSect;
+ CriticalSectionWrapper& _volumeMutex;
+ WebRtc_Word32 _id;
+
+private: // MMDevice
+ IMMDeviceEnumerator* _ptrEnumerator;
+ IMMDeviceCollection* _ptrRenderCollection;
+ IMMDeviceCollection* _ptrCaptureCollection;
+ IMMDevice* _ptrDeviceOut;
+ IMMDevice* _ptrDeviceIn;
+
+private: // WASAPI
+ IAudioClient* _ptrClientOut;
+ IAudioClient* _ptrClientIn;
+ IAudioRenderClient* _ptrRenderClient;
+ IAudioCaptureClient* _ptrCaptureClient;
+ IAudioEndpointVolume* _ptrCaptureVolume;
+ ISimpleAudioVolume* _ptrRenderSimpleVolume;
+
+ // DirectX Media Object (DMO) for the built-in AEC.
+ scoped_refptr<IMediaObject> _dmo;
+ scoped_refptr<IMediaBuffer> _mediaBuffer;
+ bool _builtInAecEnabled;
+
+ HANDLE _hRenderSamplesReadyEvent;
+ HANDLE _hPlayThread;
+ HANDLE _hRenderStartedEvent;
+ HANDLE _hShutdownRenderEvent;
+
+ HANDLE _hCaptureSamplesReadyEvent;
+ HANDLE _hRecThread;
+ HANDLE _hCaptureStartedEvent;
+ HANDLE _hShutdownCaptureEvent;
+
+ HANDLE _hGetCaptureVolumeThread;
+ HANDLE _hSetCaptureVolumeThread;
+ HANDLE _hSetCaptureVolumeEvent;
+
+ HANDLE _hMmTask;
+
+ UINT _playAudioFrameSize;
+ WebRtc_UWord32 _playSampleRate;
+ WebRtc_UWord32 _devicePlaySampleRate;
+ WebRtc_UWord32 _playBlockSize;
+ WebRtc_UWord32 _devicePlayBlockSize;
+ WebRtc_UWord32 _playChannels;
+ Atomic32Wrapper _sndCardPlayDelay;
+ UINT64 _writtenSamples;
+ LONGLONG _playAcc;
+
+ UINT _recAudioFrameSize;
+ WebRtc_UWord32 _recSampleRate;
+ WebRtc_UWord32 _recBlockSize;
+ WebRtc_UWord32 _recChannels;
+ UINT64 _readSamples;
+ Atomic32Wrapper _sndCardRecDelay;
+
+ float _sampleDriftAt48kHz;
+ float _driftAccumulator;
+
+ WebRtc_UWord16 _recChannelsPrioList[2];
+ WebRtc_UWord16 _playChannelsPrioList[2];
+
+ LARGE_INTEGER _perfCounterFreq;
+ double _perfCounterFactor;
+ float _avgCPULoad;
+
+private:
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _speakerIsInitialized;
+ bool _microphoneIsInitialized;
+
+ bool _usingInputDeviceIndex;
+ bool _usingOutputDeviceIndex;
+ AudioDeviceModule::WindowsDeviceType _inputDevice;
+ AudioDeviceModule::WindowsDeviceType _outputDevice;
+ WebRtc_UWord16 _inputDeviceIndex;
+ WebRtc_UWord16 _outputDeviceIndex;
+
+ bool _AGC;
+
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ AudioDeviceModule::BufferType _playBufType;
+ WebRtc_UWord16 _playBufDelay;
+ WebRtc_UWord16 _playBufDelayFixed;
+
+ WebRtc_UWord16 _newMicLevel;
+
+ mutable char _str[512];
+};
+
+#endif // #if (_MSC_VER >= 1400)
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
+
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc
new file mode 100644
index 0000000..5a29972
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc
@@ -0,0 +1,232 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility_win.h"
+#include "audio_device_config.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#include <windows.h>
+#include <tchar.h>
+#include <strsafe.h>
+
+#define STRING_MAX_SIZE 256
+
+typedef void (WINAPI *PGNSI)(LPSYSTEM_INFO);
+typedef BOOL (WINAPI *PGPI)(DWORD, DWORD, DWORD, DWORD, PDWORD);
+
+namespace webrtc
+{
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceUtilityWindows() - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceUtilityWindows::AudioDeviceUtilityWindows(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _lastError(AudioDeviceModule::kAdmErrNone)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceUtilityWindows() - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceUtilityWindows::~AudioDeviceUtilityWindows()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+ {
+ CriticalSectionScoped lock(_critSect);
+
+ // free stuff here...
+ }
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Init()
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceUtilityWindows::Init()
+{
+
+ TCHAR szOS[STRING_MAX_SIZE];
+
+ if (GetOSDisplayString(szOS))
+ {
+#ifdef _UNICODE
+ char os[STRING_MAX_SIZE];
+ if (WideCharToMultiByte(CP_UTF8, 0, szOS, -1, os, STRING_MAX_SIZE, NULL, NULL) == 0)
+ {
+ DWORD err = GetLastError();
+ strncpy(os, "Could not get OS info", STRING_MAX_SIZE);
+ }
+ // DEBUG_PRINTP("OS info: %s\n", os);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", os);
+#else
+ // DEBUG_PRINTP("OS info: %s\n", szOS);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", szOS);
+#endif
+ }
+
+ return 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
+{
+ OSVERSIONINFOEX osvi;
+ SYSTEM_INFO si;
+ PGNSI pGNSI;
+ BOOL bOsVersionInfoEx;
+
+ ZeroMemory(&si, sizeof(SYSTEM_INFO));
+ ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
+
+ osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+
+ // Retrieve information about the current operating system
+ //
+ if (!(bOsVersionInfoEx = GetVersionEx((OSVERSIONINFO *) &osvi)))
+ return FALSE;
+
+ // Parse our OS version string
+ //
+ if (VER_PLATFORM_WIN32_NT == osvi.dwPlatformId && osvi.dwMajorVersion > 4)
+ {
+ StringCchCopy(pszOS, STRING_MAX_SIZE, TEXT("Microsoft "));
+
+ // Test for the specific product
+ //
+ // Operating system Version number
+ // --------------------------------------
+ // Windows 7 6.1
+ // Windows Server 2008 R2 6.1
+ // Windows Server 2008 6.0
+ // Windows Vista 6.0
+ // - - - - - - - - - - - - - - - - - - -
+ // Windows Server 2003 R2 5.2
+ // Windows Server 2003 5.2
+ // Windows XP 5.1
+ // Windows 2000 5.0
+ //
+ // see http://msdn.microsoft.com/en-us/library/ms724832(VS.85).aspx for details
+ //
+ if (osvi.dwMajorVersion == 6)
+ {
+ if (osvi.dwMinorVersion == 0)
+ {
+ // Windows Vista or Server 2008
+ if (osvi.wProductType == VER_NT_WORKSTATION)
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Vista "));
+ else
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 " ));
+ }
+
+ if (osvi.dwMinorVersion == 1)
+ {
+ // Windows 7 or Server 2008 R2
+ if (osvi.wProductType == VER_NT_WORKSTATION)
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 7 "));
+ else
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 R2 " ));
+ }
+ }
+
+ if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 2)
+ {
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2003"));
+ }
+
+ if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 1)
+ {
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows XP "));
+ if (osvi.wSuiteMask & VER_SUITE_PERSONAL)
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Home Edition" ));
+ else
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
+ }
+
+ if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 0)
+ {
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 2000 "));
+
+ if (osvi.wProductType == VER_NT_WORKSTATION )
+ {
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
+ }
+ else
+ {
+ if (osvi.wSuiteMask & VER_SUITE_DATACENTER)
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Datacenter Server" ));
+ else if (osvi.wSuiteMask & VER_SUITE_ENTERPRISE)
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Advanced Server" ));
+ else StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Server" ));
+ }
+ }
+
+ // Include service pack (if any)
+ //
+ if (_tcslen(osvi.szCSDVersion) > 0)
+ {
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(" "));
+ StringCchCat(pszOS, STRING_MAX_SIZE, osvi.szCSDVersion);
+ }
+
+ TCHAR buf[80];
+
+ // Include build number
+ //
+ StringCchPrintf( buf, 80, TEXT(" (build %d)"), osvi.dwBuildNumber);
+ StringCchCat(pszOS, STRING_MAX_SIZE, buf);
+
+ // Call GetNativeSystemInfo if supported or GetSystemInfo otherwise
+ //
+ pGNSI = (PGNSI) GetProcAddress(GetModuleHandle(TEXT("kernel32.dll")), "GetNativeSystemInfo");
+ if (NULL != pGNSI)
+ pGNSI(&si);
+ else
+ GetSystemInfo(&si);
+
+ // Add 64-bit or 32-bit for OS versions "later than" Vista
+ //
+ if (osvi.dwMajorVersion >= 6)
+ {
+ if ((si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) ||
+ (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_IA64))
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( ", 64-bit" ));
+ else if (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_INTEL )
+ StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(", 32-bit"));
+ }
+
+ return TRUE;
+ }
+ else
+ {
+ return FALSE;
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h
new file mode 100644
index 0000000..77b4c22
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
+
+#include "audio_device_utility.h"
+#include "audio_device.h"
+#include <windows.h>
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class AudioDeviceUtilityWindows : public AudioDeviceUtility
+{
+public:
+ AudioDeviceUtilityWindows(const WebRtc_Word32 id);
+ ~AudioDeviceUtilityWindows();
+
+ virtual WebRtc_Word32 Init();
+
+private:
+ BOOL GetOSDisplayString(LPTSTR pszOS);
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ AudioDeviceModule::ErrorCode _lastError;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc
new file mode 100644
index 0000000..509732c
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc
@@ -0,0 +1,3818 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_utility.h"
+#include "audio_device_wave_win.h"
+#include "audio_device_config.h"
+
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+
+#include <windows.h>
+#include <objbase.h> // CoTaskMemAlloc, CoTaskMemFree
+#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
+#include <cassert>
+
+// Avoids the need of Windows 7 SDK
+#ifndef WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE
+#define WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE 0x0010
+#endif
+
+// Supported in Windows Vista and Windows 7.
+// http://msdn.microsoft.com/en-us/library/dd370819(v=VS.85).aspx
+// Taken from Mmddk.h.
+#define DRV_RESERVED 0x0800
+#define DRV_QUERYFUNCTIONINSTANCEID (DRV_RESERVED + 17)
+#define DRV_QUERYFUNCTIONINSTANCEIDSIZE (DRV_RESERVED + 18)
+
+#define POW2(A) (2 << ((A) - 1))
+
+namespace webrtc {
+
+// ============================================================================
+// Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AudioDeviceWindowsWave - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsWave::AudioDeviceWindowsWave(const WebRtc_Word32 id) :
+ _ptrAudioBuffer(NULL),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _timeEvent(*EventWrapper::Create()),
+ _recStartEvent(*EventWrapper::Create()),
+ _playStartEvent(*EventWrapper::Create()),
+ _hGetCaptureVolumeThread(NULL),
+ _hShutdownGetVolumeEvent(NULL),
+ _hSetCaptureVolumeThread(NULL),
+ _hShutdownSetVolumeEvent(NULL),
+ _hSetCaptureVolumeEvent(NULL),
+ _ptrThread(NULL),
+ _threadID(0),
+ _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _mixerManager(id),
+ _usingInputDeviceIndex(false),
+ _usingOutputDeviceIndex(false),
+ _inputDevice(AudioDeviceModule::kDefaultDevice),
+ _outputDevice(AudioDeviceModule::kDefaultDevice),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _inputDeviceIsSpecified(false),
+ _outputDeviceIsSpecified(false),
+ _initialized(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _recording(false),
+ _playing(false),
+ _startRec(false),
+ _stopRec(false),
+ _startPlay(false),
+ _stopPlay(false),
+ _AGC(false),
+ _hWaveIn(NULL),
+ _hWaveOut(NULL),
+ _recChannels(N_REC_CHANNELS),
+ _playChannels(N_PLAY_CHANNELS),
+ _recBufCount(0),
+ _recPutBackDelay(0),
+ _recDelayCount(0),
+ _playBufCount(0),
+ _prevPlayTime(0),
+ _prevRecTime(0),
+ _prevTimerCheckTime(0),
+ _timesdwBytes(0),
+ _timerFaults(0),
+ _timerRestartAttempts(0),
+ _no_of_msecleft_warnings(0),
+ _MAX_minBuffer(65),
+ _useHeader(0),
+ _dTcheckPlayBufDelay(10),
+ _playBufDelay(80),
+ _playBufDelayFixed(80),
+ _minPlayBufDelay(20),
+ _avgCPULoad(0),
+ _sndCardPlayDelay(0),
+ _sndCardRecDelay(0),
+ _plSampOld(0),
+ _rcSampOld(0),
+ _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
+ _recordedBytes(0),
+ _playWarning(0),
+ _playError(0),
+ _recWarning(0),
+ _recError(0),
+ _newMicLevel(0),
+ _minMicVolume(0),
+ _maxMicVolume(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+
+ // Initialize value, set to 0 if it fails
+ if (!QueryPerformanceFrequency(&_perfFreq))
+ {
+ _perfFreq.QuadPart = 0;
+ }
+
+ _hShutdownGetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hShutdownSetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+ _hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+}
+
+// ----------------------------------------------------------------------------
+// AudioDeviceWindowsWave - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsWave::~AudioDeviceWindowsWave()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+
+ Terminate();
+
+ delete &_recStartEvent;
+ delete &_playStartEvent;
+ delete &_timeEvent;
+ delete &_critSect;
+ delete &_critSectCb;
+
+ if (NULL != _hShutdownGetVolumeEvent)
+ {
+ CloseHandle(_hShutdownGetVolumeEvent);
+ _hShutdownGetVolumeEvent = NULL;
+ }
+
+ if (NULL != _hShutdownSetVolumeEvent)
+ {
+ CloseHandle(_hShutdownSetVolumeEvent);
+ _hShutdownSetVolumeEvent = NULL;
+ }
+
+ if (NULL != _hSetCaptureVolumeEvent)
+ {
+ CloseHandle(_hSetCaptureVolumeEvent);
+ _hSetCaptureVolumeEvent = NULL;
+ }
+}
+
+// ============================================================================
+// API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// AttachAudioBuffer
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ _ptrAudioBuffer = audioBuffer;
+
+ // inform the AudioBuffer about default settings for this implementation
+ _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+ _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+ _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+// ----------------------------------------------------------------------------
+// ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
+{
+ audioLayer = AudioDeviceModule::kWindowsWaveAudio;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Init
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::Init()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_initialized)
+ {
+ return 0;
+ }
+
+ const WebRtc_UWord32 nowTime(AudioDeviceUtility::GetTimeInMS());
+
+ _recordedBytes = 0;
+ _prevRecByteCheckTime = nowTime;
+ _prevRecTime = nowTime;
+ _prevPlayTime = nowTime;
+ _prevTimerCheckTime = nowTime;
+
+ _playWarning = 0;
+ _playError = 0;
+ _recWarning = 0;
+ _recError = 0;
+
+ _mixerManager.EnumerateAll();
+
+ if (_ptrThread)
+ {
+ // thread is already created and active
+ return 0;
+ }
+
+ const char* threadName = "webrtc_audio_module_thread";
+ _ptrThread = ThreadWrapper::CreateThread(ThreadFunc,
+ this,
+ kRealtimePriority,
+ threadName);
+ if (_ptrThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ "failed to create the audio thread");
+ return -1;
+ }
+
+ unsigned int threadID(0);
+ if (!_ptrThread->Start(threadID))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ "failed to start the audio thread");
+ delete _ptrThread;
+ _ptrThread = NULL;
+ return -1;
+ }
+ _threadID = threadID;
+
+ const bool periodic(true);
+ if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+ "failed to start the timer event");
+ if (_ptrThread->Stop())
+ {
+ delete _ptrThread;
+ _ptrThread = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "unable to stop the activated thread");
+ }
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
+
+ _hGetCaptureVolumeThread = CreateThread(NULL,
+ 0,
+ GetCaptureVolumeThread,
+ this,
+ 0,
+ NULL);
+ if (_hGetCaptureVolumeThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create the volume getter thread");
+ return -1;
+ }
+
+ SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
+
+ _hSetCaptureVolumeThread = CreateThread(NULL,
+ 0,
+ SetCaptureVolumeThread,
+ this,
+ 0,
+ NULL);
+ if (_hSetCaptureVolumeThread == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to create the volume setter thread");
+ return -1;
+ }
+
+ SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
+
+ _initialized = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Terminate
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::Terminate()
+{
+
+ if (!_initialized)
+ {
+ return 0;
+ }
+
+ _critSect.Enter();
+
+ _mixerManager.Close();
+
+ if (_ptrThread)
+ {
+ ThreadWrapper* tmpThread = _ptrThread;
+ _ptrThread = NULL;
+ _critSect.Leave();
+
+ tmpThread->SetNotAlive();
+ _timeEvent.Set();
+
+ if (tmpThread->Stop())
+ {
+ delete tmpThread;
+ }
+ else
+ {
+ _critSect.Leave();
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ "failed to close down the audio thread");
+ return -1;
+ }
+ }
+ else
+ {
+ _critSect.Leave();
+ }
+
+ _critSect.Enter();
+ SetEvent(_hShutdownGetVolumeEvent);
+ _critSect.Leave();
+ WebRtc_Word32 ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ // the thread did not stop as it should
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to close down volume getter thread");
+ CloseHandle(_hGetCaptureVolumeThread);
+ _hGetCaptureVolumeThread = NULL;
+ return -1;
+ }
+ _critSect.Enter();
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " volume getter thread is now closed");
+
+ SetEvent(_hShutdownSetVolumeEvent);
+ _critSect.Leave();
+ ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0)
+ {
+ // the thread did not stop as it should
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " failed to close down volume setter thread");
+ CloseHandle(_hSetCaptureVolumeThread);
+ _hSetCaptureVolumeThread = NULL;
+ return -1;
+ }
+ _critSect.Enter();
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ " volume setter thread is now closed");
+
+ CloseHandle(_hGetCaptureVolumeThread);
+ _hGetCaptureVolumeThread = NULL;
+
+ CloseHandle(_hSetCaptureVolumeThread);
+ _hSetCaptureVolumeThread = NULL;
+
+ _critSect.Leave();
+
+ _timeEvent.StopTimer();
+
+ _initialized = false;
+ _outputDeviceIsSpecified = false;
+ _inputDeviceIsSpecified = false;
+
+ return 0;
+}
+
+
+DWORD WINAPI AudioDeviceWindowsWave::GetCaptureVolumeThread(LPVOID context)
+{
+ return(((AudioDeviceWindowsWave*)context)->DoGetCaptureVolumeThread());
+}
+
+DWORD WINAPI AudioDeviceWindowsWave::SetCaptureVolumeThread(LPVOID context)
+{
+ return(((AudioDeviceWindowsWave*)context)->DoSetCaptureVolumeThread());
+}
+
+DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
+{
+ HANDLE waitObject = _hShutdownGetVolumeEvent;
+
+ while (1)
+ {
+ DWORD waitResult = WaitForSingleObject(waitObject,
+ GET_MIC_VOLUME_INTERVAL_MS);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0: // _hShutdownGetVolumeEvent
+ return 0;
+ case WAIT_TIMEOUT: // timeout notification
+ break;
+ default: // unexpected error
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unknown wait termination on get volume thread");
+ return -1;
+ }
+
+ if (AGC())
+ {
+ WebRtc_UWord32 currentMicLevel = 0;
+ if (MicrophoneVolume(currentMicLevel) == 0)
+ {
+ // This doesn't set the system volume, just stores it.
+ _critSect.Enter();
+ if (_ptrAudioBuffer)
+ {
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ _critSect.Leave();
+ }
+ }
+ }
+}
+
+DWORD AudioDeviceWindowsWave::DoSetCaptureVolumeThread()
+{
+ HANDLE waitArray[2] = {_hShutdownSetVolumeEvent, _hSetCaptureVolumeEvent};
+
+ while (1)
+ {
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
+ switch (waitResult)
+ {
+ case WAIT_OBJECT_0: // _hShutdownSetVolumeEvent
+ return 0;
+ case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
+ break;
+ default: // unexpected error
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unknown wait termination on set volume thread");
+ return -1;
+ }
+
+ _critSect.Enter();
+ WebRtc_UWord32 newMicLevel = _newMicLevel;
+ _critSect.Leave();
+
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " the required modification of the microphone volume failed");
+ }
+ }
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::Initialized() const
+{
+ return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerIsAvailable(bool& available)
+{
+
+ // Enumerate all avaliable speakers and make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (InitSpeaker() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a valid speaker exists
+ //
+ available = true;
+
+ // Close the initialized output mixer
+ //
+ _mixerManager.CloseSpeaker();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::InitSpeaker()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (_mixerManager.EnumerateSpeakers() == -1)
+ {
+ // failed to locate any valid/controllable speaker
+ return -1;
+ }
+
+ if (IsUsingOutputDeviceIndex())
+ {
+ if (_mixerManager.OpenSpeaker(OutputDeviceIndex()) == -1)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ if (_mixerManager.OpenSpeaker(OutputDevice()) == -1)
+ {
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneIsAvailable(bool& available)
+{
+
+ // Enumerate all avaliable microphones and make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (InitMicrophone() == -1)
+ {
+ available = false;
+ return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a valid microphone exists
+ //
+ available = true;
+
+ // Close the initialized input mixer
+ //
+ _mixerManager.CloseMicrophone();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::InitMicrophone()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (_mixerManager.EnumerateMicrophones() == -1)
+ {
+ // failed to locate any valid/controllable microphone
+ return -1;
+ }
+
+ if (IsUsingInputDeviceIndex())
+ {
+ if (_mixerManager.OpenMicrophone(InputDeviceIndex()) == -1)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ if (_mixerManager.OpenMicrophone(InputDevice()) == -1)
+ {
+ return -1;
+ }
+ }
+
+ WebRtc_UWord32 maxVol = 0;
+ if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unable to retrieve max microphone volume");
+ }
+ _maxMicVolume = maxVol;
+
+ WebRtc_UWord32 minVol = 0;
+ if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+ " unable to retrieve min microphone volume");
+ }
+ _minMicVolume = minVol;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::SpeakerIsInitialized() const
+{
+ return (_mixerManager.SpeakerIsInitialized());
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::MicrophoneIsInitialized() const
+{
+ return (_mixerManager.MicrophoneIsInitialized());
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+
+ // Enumerate all avaliable speakers and make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (InitSpeaker() == -1)
+ {
+ // failed to find a valid speaker
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected speaker has a volume control
+ //
+ _mixerManager.SpeakerVolumeIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ //
+ _mixerManager.CloseSpeaker();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+
+ return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.SpeakerVolume(level) == -1)
+ {
+ return -1;
+ }
+
+ volume = level;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetWaveOutVolume
+//
+// The low-order word contains the left-channel volume setting, and the
+// high-order word contains the right-channel setting.
+// A value of 0xFFFF represents full volume, and a value of 0x0000 is silence.
+//
+// If a device does not support both left and right volume control,
+// the low-order word of dwVolume specifies the volume level,
+// and the high-order word is ignored.
+//
+// Most devices do not support the full 16 bits of volume-level control
+// and will not use the least-significant bits of the requested volume setting.
+// For example, if a device supports 4 bits of volume control, the values
+// 0x4000, 0x4FFF, and 0x43BE will all be truncated to 0x4000.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
+{
+
+ MMRESULT res(0);
+ WAVEOUTCAPS caps;
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_hWaveOut == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
+ }
+
+ // To determine whether the device supports volume control on both
+ // the left and right channels, use the WAVECAPS_LRVOLUME flag.
+ //
+ res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+ if (!(caps.dwSupport & WAVECAPS_VOLUME))
+ {
+ // this device does not support volume control using the waveOutSetVolume API
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
+ return -1;
+ }
+ if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
+ {
+ // high-order word (right channel) is ignored
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
+ }
+
+ DWORD dwVolume(0x00000000);
+ dwVolume = (DWORD)(((volumeRight & 0xFFFF) << 16) | (volumeLeft & 0xFFFF));
+
+ res = waveOutSetVolume(_hWaveOut, dwVolume);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutSetVolume() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// WaveOutVolume
+//
+// The low-order word of this location contains the left-channel volume setting,
+// and the high-order word contains the right-channel setting.
+// A value of 0xFFFF (65535) represents full volume, and a value of 0x0000
+// is silence.
+//
+// If a device does not support both left and right volume control,
+// the low-order word of the specified location contains the mono volume level.
+//
+// The full 16-bit setting(s) set with the waveOutSetVolume function is returned,
+// regardless of whether the device supports the full 16 bits of volume-level
+// control.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
+{
+
+ MMRESULT res(0);
+ WAVEOUTCAPS caps;
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_hWaveOut == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
+ }
+
+ // To determine whether the device supports volume control on both
+ // the left and right channels, use the WAVECAPS_LRVOLUME flag.
+ //
+ res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+ if (!(caps.dwSupport & WAVECAPS_VOLUME))
+ {
+ // this device does not support volume control using the waveOutSetVolume API
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
+ return -1;
+ }
+ if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
+ {
+ // high-order word (right channel) is ignored
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
+ }
+
+ DWORD dwVolume(0x00000000);
+
+ res = waveOutGetVolume(_hWaveOut, &dwVolume);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutGetVolume() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ return -1;
+ }
+
+ WORD wVolumeLeft = LOWORD(dwVolume);
+ WORD wVolumeRight = HIWORD(dwVolume);
+
+ volumeLeft = static_cast<WebRtc_UWord16> (wVolumeLeft);
+ volumeRight = static_cast<WebRtc_UWord16> (wVolumeRight);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ WebRtc_UWord32 maxVol(0);
+
+ if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
+ {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ WebRtc_UWord32 minVol(0);
+
+ if (_mixerManager.MinSpeakerVolume(minVol) == -1)
+ {
+ return -1;
+ }
+
+ minVolume = minVol;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+
+ // Enumerate all avaliable speakers and make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (InitSpeaker() == -1)
+ {
+ // If we end up here it means that the selected speaker has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected speaker has a mute control
+ //
+ _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ //
+ _mixerManager.CloseSpeaker();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerMute(bool enable)
+{
+ return (_mixerManager.SetSpeakerMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.SpeakerMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMuteIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+
+ // Enumerate all avaliable microphones and make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a mute control
+ //
+ _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ _mixerManager.CloseMicrophone();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneMute(bool enable)
+{
+ return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMute(bool& enabled) const
+{
+
+ bool muted(0);
+
+ if (_mixerManager.MicrophoneMute(muted) == -1)
+ {
+ return -1;
+ }
+
+ enabled = muted;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoostIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+
+ // Enumerate all avaliable microphones and make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (InitMicrophone() == -1)
+ {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no boost control
+ // already at this stage.
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a boost control
+ //
+ _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ _mixerManager.CloseMicrophone();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneBoost(bool enable)
+{
+
+ return (_mixerManager.SetMicrophoneBoost(enable));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoost(bool& enabled) const
+{
+
+ bool onOff(0);
+
+ if (_mixerManager.MicrophoneBoost(onOff) == -1)
+ {
+ return -1;
+ }
+
+ enabled = onOff;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StereoRecordingIsAvailable(bool& available)
+{
+ available = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetStereoRecording(bool enable)
+{
+
+ if (enable)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StereoRecording(bool& enabled) const
+{
+
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayoutIsAvailable(bool& available)
+{
+ available = true;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetStereoPlayout
+//
+// Specifies the number of output channels.
+//
+// NOTE - the setting will only have an effect after InitPlayout has
+// been called.
+//
+// 16-bit mono:
+//
+// Each sample is 2 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
+// For each sample, the first byte is the low-order byte of channel 0 and the
+// second byte is the high-order byte of channel 0.
+//
+// 16-bit stereo:
+//
+// Each sample is 4 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
+// For each sample, the first byte is the low-order byte of channel 0 (left channel);
+// the second byte is the high-order byte of channel 0; the third byte is the
+// low-order byte of channel 1 (right channel); and the fourth byte is the
+// high-order byte of channel 1.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetStereoPlayout(bool enable)
+{
+
+ if (enable)
+ _playChannels = 2;
+ else
+ _playChannels = 1;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StereoPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayout(bool& enabled) const
+{
+
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetAGC
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetAGC(bool enable)
+{
+
+ _AGC = enable;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// AGC
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::AGC() const
+{
+ return _AGC;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeIsAvailable(bool& available)
+{
+
+ bool isAvailable(false);
+
+ // Enumerate all avaliable microphones and make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ //
+ if (InitMicrophone() == -1)
+ {
+ // Failed to find valid microphone
+ available = false;
+ return 0;
+ }
+
+ // Check if the selected microphone has a volume control
+ //
+ _mixerManager.MicrophoneVolumeIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ _mixerManager.CloseMicrophone();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+ WebRtc_UWord32 level(0);
+
+ if (_mixerManager.MicrophoneVolume(level) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to retrive current microphone level");
+ return -1;
+ }
+
+ volume = level;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+ // _maxMicVolume can be zero in AudioMixerManager::MaxMicrophoneVolume():
+ // (1) API GetLineControl() returns failure at querying the max Mic level.
+ // (2) API GetLineControl() returns maxVolume as zero in rare cases.
+ // Both cases show we don't have access to the mixer controls.
+ // We return -1 here to indicate that.
+ if (_maxMicVolume == 0)
+ {
+ return -1;
+ }
+
+ maxVolume = _maxMicVolume;;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+ minVolume = _minMicVolume;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ WebRtc_UWord16 delta(0);
+
+ if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
+ {
+ return -1;
+ }
+
+ stepSize = delta;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceWindowsWave::PlayoutDevices()
+{
+
+ return (waveOutGetNumDevs());
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(WebRtc_UWord16 index)
+{
+
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ UINT nDevices = waveOutGetNumDevs();
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio output devices is %u", nDevices);
+
+ if (index < 0 || index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ _usingOutputDeviceIndex = true;
+ _outputDeviceIndex = index;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ }
+
+ _usingOutputDeviceIndex = false;
+ _outputDevice = device;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ WebRtc_UWord16 nDevices(PlayoutDevices());
+
+ // Special fix for the case when the user asks for the name of the default device.
+ //
+ if (index == (WebRtc_UWord16)(-1))
+ {
+ index = 0;
+ }
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ WAVEOUTCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
+ MMRESULT res;
+
+ res = waveOutGetDevCapsW(index, &caps, sizeof(WAVEOUTCAPSW));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCapsW() failed (err=%d)", res);
+ return -1;
+ }
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
+ }
+
+ if (guid == NULL)
+ {
+ return 0;
+ }
+
+ // It is possible to get the unique endpoint ID string using the Wave API.
+ // However, it is only supported on Windows Vista and Windows 7.
+
+ size_t cbEndpointId(0);
+
+ // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
+ // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
+ res = waveOutMessage((HWAVEOUT)IntToPtr(index),
+ DRV_QUERYFUNCTIONINSTANCEIDSIZE,
+ (DWORD_PTR)&cbEndpointId, NULL);
+ if (res != MMSYSERR_NOERROR)
+ {
+ // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
+ TraceWaveOutError(res);
+ // Best we can do is to copy the friendly name and use it as guid
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
+ }
+ return 0;
+ }
+
+ // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
+
+ WCHAR *pstrEndpointId = NULL;
+ pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
+
+ // Get the endpoint ID string for this waveOut device.
+ res = waveOutMessage((HWAVEOUT)IntToPtr(index),
+ DRV_QUERYFUNCTIONINSTANCEID,
+ (DWORD_PTR)pstrEndpointId,
+ cbEndpointId);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
+ TraceWaveOutError(res);
+ // Best we can do is to copy the friendly name and use it as guid
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
+ }
+ CoTaskMemFree(pstrEndpointId);
+ return 0;
+ }
+
+ if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
+ }
+ CoTaskMemFree(pstrEndpointId);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
+{
+
+ WebRtc_UWord16 nDevices(RecordingDevices());
+
+ // Special fix for the case when the user asks for the name of the default device.
+ //
+ if (index == (WebRtc_UWord16)(-1))
+ {
+ index = 0;
+ }
+
+ if ((index > (nDevices-1)) || (name == NULL))
+ {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL)
+ {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ WAVEINCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
+ MMRESULT res;
+
+ res = waveInGetDevCapsW(index, &caps, sizeof(WAVEINCAPSW));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCapsW() failed (err=%d)", res);
+ return -1;
+ }
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
+ }
+
+ if (guid == NULL)
+ {
+ return 0;
+ }
+
+ // It is possible to get the unique endpoint ID string using the Wave API.
+ // However, it is only supported on Windows Vista and Windows 7.
+
+ size_t cbEndpointId(0);
+
+ // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
+ // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
+ res = waveInMessage((HWAVEIN)IntToPtr(index),
+ DRV_QUERYFUNCTIONINSTANCEIDSIZE,
+ (DWORD_PTR)&cbEndpointId, NULL);
+ if (res != MMSYSERR_NOERROR)
+ {
+ // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
+ TraceWaveInError(res);
+ // Best we can do is to copy the friendly name and use it as guid
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
+ }
+ return 0;
+ }
+
+ // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
+
+ WCHAR *pstrEndpointId = NULL;
+ pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
+
+ // Get the endpoint ID string for this waveOut device.
+ res = waveInMessage((HWAVEIN)IntToPtr(index),
+ DRV_QUERYFUNCTIONINSTANCEID,
+ (DWORD_PTR)pstrEndpointId,
+ cbEndpointId);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
+ TraceWaveInError(res);
+ // Best we can do is to copy the friendly name and use it as guid
+ if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
+ }
+ CoTaskMemFree(pstrEndpointId);
+ return 0;
+ }
+
+ if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
+ }
+ CoTaskMemFree(pstrEndpointId);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word16 AudioDeviceWindowsWave::RecordingDevices()
+{
+
+ return (waveInGetNumDevs());
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(WebRtc_UWord16 index)
+{
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ UINT nDevices = waveInGetNumDevs();
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio input devices is %u", nDevices);
+
+ if (index < 0 || index > (nDevices-1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ return -1;
+ }
+
+ _usingInputDeviceIndex = true;
+ _inputDeviceIndex = index;
+ _inputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ }
+
+ if (_recIsInitialized)
+ {
+ return -1;
+ }
+
+ _usingInputDeviceIndex = false;
+ _inputDevice = device;
+ _inputDeviceIsSpecified = true;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PlayoutIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the playout side
+ WebRtc_Word32 res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::RecordingIsAvailable(bool& available)
+{
+
+ available = false;
+
+ // Try to initialize the recording side
+ WebRtc_Word32 res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1)
+ {
+ available = true;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::InitPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_playing)
+ {
+ return -1;
+ }
+
+ if (!_outputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ return 0;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
+ }
+
+ // Enumerate all availiable output devices
+ EnumeratePlayoutDevices();
+
+ // Start by closing any existing wave-output devices
+ //
+ MMRESULT res(MMSYSERR_ERROR);
+
+ if (_hWaveOut != NULL)
+ {
+ res = waveOutClose(_hWaveOut);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+ }
+
+ // Set the output wave format
+ //
+ WAVEFORMATEX waveFormat;
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = _playChannels; // mono <=> 1, stereo <=> 2
+ waveFormat.nSamplesPerSec = N_PLAY_SAMPLES_PER_SEC;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0;
+
+ // Open the given waveform-audio output device for playout
+ //
+ HWAVEOUT hWaveOut(NULL);
+
+ if (IsUsingOutputDeviceIndex())
+ {
+ // verify settings first
+ res = waveOutOpen(NULL, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // open the given waveform-audio output device for recording
+ res = waveOutOpen(&hWaveOut, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening output device corresponding to device ID %u", _outputDeviceIndex);
+ }
+ }
+ else
+ {
+ if (_outputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ // check if it is possible to open the default communication device (supported on Windows 7)
+ res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // if so, open the default communication device for real
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
+ }
+ else
+ {
+ // use default device since default communication device was not avaliable
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
+ }
+ }
+ else if (_outputDevice == AudioDeviceModule::kDefaultDevice)
+ {
+ // open default device since it has been requested
+ res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
+ }
+ }
+ }
+
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ return -1;
+ }
+
+ // Log information about the aquired output device
+ //
+ WAVEOUTCAPS caps;
+
+ res = waveOutGetDevCaps((UINT_PTR)hWaveOut, &caps, sizeof(WAVEOUTCAPS));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+
+ UINT deviceID(0);
+ res = waveOutGetID(hWaveOut, &deviceID);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetID() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
+
+ // Store valid handle for the open waveform-audio output device
+ _hWaveOut = hWaveOut;
+
+ // Store the input wave header as well
+ _waveFormatOut = waveFormat;
+
+ // Prepare wave-out headers
+ //
+ const WebRtc_UWord8 bytesPerSample = 2*_playChannels;
+
+ for (int n = 0; n < N_BUFFERS_OUT; n++)
+ {
+ // set up the output wave header
+ _waveHeaderOut[n].lpData = _playBuffer[n];
+ _waveHeaderOut[n].dwBufferLength = bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES;
+ _waveHeaderOut[n].dwFlags = 0;
+ _waveHeaderOut[n].dwLoops = 0;
+
+ memset(_playBuffer[n], 0, bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES);
+
+ // The waveOutPrepareHeader function prepares a waveform-audio data block for playback.
+ // The lpData, dwBufferLength, and dwFlags members of the WAVEHDR structure must be set
+ // before calling this function.
+ //
+ res = waveOutPrepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (err=%d)", n, res);
+ TraceWaveOutError(res);
+ }
+
+ // perform extra check to ensure that the header is prepared
+ if (_waveHeaderOut[n].dwFlags != WHDR_PREPARED)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (dwFlags != WHDR_PREPARED)", n);
+ }
+ }
+
+ // Mark playout side as initialized
+ _playIsInitialized = true;
+
+ _dTcheckPlayBufDelay = 10; // check playback buffer delay every 10 ms
+ _playBufCount = 0; // index of active output wave header (<=> output buffer index)
+ _playBufDelay = 80; // buffer delay/size is initialized to 80 ms and slowly decreased until er < 25
+ _minPlayBufDelay = 25; // minimum playout buffer delay
+ _MAX_minBuffer = 65; // adaptive minimum playout buffer delay cannot be larger than this value
+ _intro = 1; // Used to make sure that adaption starts after (2000-1700)/100 seconds
+ _waitCounter = 1700; // Counter for start of adaption of playback buffer
+ _erZeroCounter = 0; // Log how many times er = 0 in consequtive calls to RecTimeProc
+ _useHeader = 0; // Counts number of "useHeader" detections. Stops at 2.
+
+ _writtenSamples = 0;
+ _writtenSamplesOld = 0;
+ _playedSamplesOld = 0;
+ _sndCardPlayDelay = 0;
+ _sndCardRecDelay = 0;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id,"initial playout status: _playBufDelay=%d, _minPlayBufDelay=%d",
+ _playBufDelay, _minPlayBufDelay);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// InitRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::InitRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_recording)
+ {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified)
+ {
+ return -1;
+ }
+
+ if (_recIsInitialized)
+ {
+ return 0;
+ }
+
+ _avgCPULoad = 0;
+ _playAcc = 0;
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
+ }
+
+ // Enumerate all availiable input devices
+ EnumerateRecordingDevices();
+
+ // Start by closing any existing wave-input devices
+ //
+ MMRESULT res(MMSYSERR_ERROR);
+
+ if (_hWaveIn != NULL)
+ {
+ res = waveInClose(_hWaveIn);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+ }
+
+ // Set the input wave format
+ //
+ WAVEFORMATEX waveFormat;
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = _recChannels; // mono <=> 1, stereo <=> 2
+ waveFormat.nSamplesPerSec = N_REC_SAMPLES_PER_SEC;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0;
+
+ // Open the given waveform-audio input device for recording
+ //
+ HWAVEIN hWaveIn(NULL);
+
+ if (IsUsingInputDeviceIndex())
+ {
+ // verify settings first
+ res = waveInOpen(NULL, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // open the given waveform-audio input device for recording
+ res = waveInOpen(&hWaveIn, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening input device corresponding to device ID %u", _inputDeviceIndex);
+ }
+ }
+ else
+ {
+ if (_inputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ // check if it is possible to open the default communication device (supported on Windows 7)
+ res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // if so, open the default communication device for real
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
+ }
+ else
+ {
+ // use default device since default communication device was not avaliable
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
+ }
+ }
+ else if (_inputDevice == AudioDeviceModule::kDefaultDevice)
+ {
+ // open default device since it has been requested
+ res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
+ }
+ }
+ }
+
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
+ TraceWaveInError(res);
+ return -1;
+ }
+
+ // Log information about the aquired input device
+ //
+ WAVEINCAPS caps;
+
+ res = waveInGetDevCaps((UINT_PTR)hWaveIn, &caps, sizeof(WAVEINCAPS));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ UINT deviceID(0);
+ res = waveInGetID(hWaveIn, &deviceID);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetID() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
+
+ // Store valid handle for the open waveform-audio input device
+ _hWaveIn = hWaveIn;
+
+ // Store the input wave header as well
+ _waveFormatIn = waveFormat;
+
+ // Mark recording side as initialized
+ _recIsInitialized = true;
+
+ _recBufCount = 0; // index of active input wave header (<=> input buffer index)
+ _recDelayCount = 0; // ensures that input buffers are returned with certain delay
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StartRecording()
+{
+
+ if (!_recIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_recording)
+ {
+ return 0;
+ }
+
+ // set state to ensure that the recording starts from the audio thread
+ _startRec = true;
+
+ // the audio thread will signal when recording has stopped
+ if (kEventTimeout == _recStartEvent.Wait(10000))
+ {
+ _startRec = false;
+ StopRecording();
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
+ return -1;
+ }
+
+ if (_recording)
+ {
+ // the recording state is set by the audio thread after recording has started
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StopRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_recIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_hWaveIn == NULL)
+ {
+ return -1;
+ }
+
+ bool wasRecording = _recording;
+ _recIsInitialized = false;
+ _recording = false;
+
+ MMRESULT res;
+
+ // Stop waveform-adio input. If there are any buffers in the queue, the
+ // current buffer will be marked as done (the dwBytesRecorded member in
+ // the header will contain the length of data), but any empty buffers in
+ // the queue will remain there.
+ //
+ res = waveInStop(_hWaveIn);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStop() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ // Stop input on the given waveform-audio input device and resets the current
+ // position to zero. All pending buffers are marked as done and returned to
+ // the application.
+ //
+ res = waveInReset(_hWaveIn);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInReset() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ // Clean up the preparation performed by the waveInPrepareHeader function.
+ // Only unprepare header if recording was ever started (and headers are prepared).
+ //
+ if (wasRecording)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInUnprepareHeader() will be performed");
+ for (int n = 0; n < N_BUFFERS_IN; n++)
+ {
+ res = waveInUnprepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+ }
+ }
+
+ // Close the given waveform-audio input device.
+ //
+ res = waveInClose(_hWaveIn);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ // Set the wave input handle to NULL
+ //
+ _hWaveIn = NULL;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveIn is now set to NULL");
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::RecordingIsInitialized() const
+{
+ return (_recIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::Recording() const
+{
+ return (_recording);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::PlayoutIsInitialized() const
+{
+ return (_playIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+// StartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StartPlayout()
+{
+
+ if (!_playIsInitialized)
+ {
+ return -1;
+ }
+
+ if (_playing)
+ {
+ return 0;
+ }
+
+ // set state to ensure that playout starts from the audio thread
+ _startPlay = true;
+
+ // the audio thread will signal when recording has started
+ if (kEventTimeout == _playStartEvent.Wait(10000))
+ {
+ _startPlay = false;
+ StopPlayout();
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playout");
+ return -1;
+ }
+
+ if (_playing)
+ {
+ // the playing state is set by the audio thread after playout has started
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playing");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::StopPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playIsInitialized)
+ {
+ return 0;
+ }
+
+ if (_hWaveOut == NULL)
+ {
+ return -1;
+ }
+
+ _playIsInitialized = false;
+ _playing = false;
+ _sndCardPlayDelay = 0;
+ _sndCardRecDelay = 0;
+
+ MMRESULT res;
+
+ // The waveOutReset function stops playback on the given waveform-audio
+ // output device and resets the current position to zero. All pending
+ // playback buffers are marked as done (WHDR_DONE) and returned to the application.
+ // After this function returns, the application can send new playback buffers
+ // to the device by calling waveOutWrite, or close the device by calling waveOutClose.
+ //
+ res = waveOutReset(_hWaveOut);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutReset() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+
+ // The waveOutUnprepareHeader function cleans up the preparation performed
+ // by the waveOutPrepareHeader function. This function must be called after
+ // the device driver is finished with a data block.
+ // You must call this function before freeing the buffer.
+ //
+ for (int n = 0; n < N_BUFFERS_OUT; n++)
+ {
+ res = waveOutUnprepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutUnprepareHeader() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+ }
+
+ // The waveOutClose function closes the given waveform-audio output device.
+ // The close operation fails if the device is still playing a waveform-audio
+ // buffer that was previously sent by calling waveOutWrite. Before calling
+ // waveOutClose, the application must wait for all buffers to finish playing
+ // or call the waveOutReset function to terminate playback.
+ //
+ res = waveOutClose(_hWaveOut);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+
+ _hWaveOut = NULL;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveOut is now set to NULL");
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDelay(WebRtc_UWord16& delayMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+ delayMS = (WebRtc_UWord16)_sndCardPlayDelay;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// RecordingDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::RecordingDelay(WebRtc_UWord16& delayMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+ delayMS = (WebRtc_UWord16)_sndCardRecDelay;
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::Playing() const
+{
+ return (_playing);
+}
+// ----------------------------------------------------------------------------
+// SetPlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
+{
+ CriticalSectionScoped lock(_critSect);
+ _playBufType = type;
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ _playBufDelayFixed = sizeMS;
+ }
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutBuffer
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
+{
+ CriticalSectionScoped lock(_critSect);
+ type = _playBufType;
+ if (type == AudioDeviceModule::kFixedBufferSize)
+ {
+ sizeMS = _playBufDelayFixed;
+ }
+ else
+ {
+ sizeMS = _playBufDelay;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// CPULoad
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::CPULoad(WebRtc_UWord16& load) const
+{
+
+ load = static_cast<WebRtc_UWord16>(100*_avgCPULoad);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::PlayoutWarning() const
+{
+ return ( _playWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// PlayoutError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::PlayoutError() const
+{
+ return ( _playError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingWarning
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::RecordingWarning() const
+{
+ return ( _recWarning > 0);
+}
+
+// ----------------------------------------------------------------------------
+// RecordingError
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::RecordingError() const
+{
+ return ( _recError > 0);
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::ClearPlayoutWarning()
+{
+ _playWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearPlayoutError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::ClearPlayoutError()
+{
+ _playError = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingWarning
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::ClearRecordingWarning()
+{
+ _recWarning = 0;
+}
+
+// ----------------------------------------------------------------------------
+// ClearRecordingError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::ClearRecordingError()
+{
+ _recError = 0;
+}
+
+// ============================================================================
+// Private Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// InputSanityCheckAfterUnlockedPeriod
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::InputSanityCheckAfterUnlockedPeriod() const
+{
+ if (_hWaveIn == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
+ return -1;
+ }
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// OutputSanityCheckAfterUnlockedPeriod
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::OutputSanityCheckAfterUnlockedPeriod() const
+{
+ if (_hWaveOut == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
+ return -1;
+ }
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// EnumeratePlayoutDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::EnumeratePlayoutDevices()
+{
+
+ WebRtc_UWord16 nDevices(PlayoutDevices());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#output devices: %u", nDevices);
+
+ WAVEOUTCAPS caps;
+ MMRESULT res;
+
+ for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
+ {
+ res = waveOutGetDevCaps(deviceID, &caps, sizeof(WAVEOUTCAPS));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
+ if (caps.dwFormats & WAVE_FORMAT_48S16)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
+ }
+ if (caps.dwFormats & WAVE_FORMAT_48M16)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
+ TraceSupportFlags(caps.dwSupport);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// EnumerateRecordingDevices
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::EnumerateRecordingDevices()
+{
+
+ WebRtc_UWord16 nDevices(RecordingDevices());
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#input devices: %u", nDevices);
+
+ WAVEINCAPS caps;
+ MMRESULT res;
+
+ for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
+ {
+ res = waveInGetDevCaps(deviceID, &caps, sizeof(WAVEINCAPS));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
+ if (caps.dwFormats & WAVE_FORMAT_48S16)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
+ }
+ if (caps.dwFormats & WAVE_FORMAT_48M16)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// TraceSupportFlags
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::TraceSupportFlags(DWORD dwSupport) const
+{
+ TCHAR buf[256];
+
+ StringCchPrintf(buf, 128, TEXT("support flags : 0x%x "), dwSupport);
+
+ if (dwSupport & WAVECAPS_PITCH)
+ {
+ // supports pitch control
+ StringCchCat(buf, 256, TEXT("(PITCH)"));
+ }
+ if (dwSupport & WAVECAPS_PLAYBACKRATE)
+ {
+ // supports playback rate control
+ StringCchCat(buf, 256, TEXT("(PLAYBACKRATE)"));
+ }
+ if (dwSupport & WAVECAPS_VOLUME)
+ {
+ // supports volume control
+ StringCchCat(buf, 256, TEXT("(VOLUME)"));
+ }
+ if (dwSupport & WAVECAPS_LRVOLUME)
+ {
+ // supports separate left and right volume control
+ StringCchCat(buf, 256, TEXT("(LRVOLUME)"));
+ }
+ if (dwSupport & WAVECAPS_SYNC)
+ {
+ // the driver is synchronous and will block while playing a buffer
+ StringCchCat(buf, 256, TEXT("(SYNC)"));
+ }
+ if (dwSupport & WAVECAPS_SAMPLEACCURATE)
+ {
+ // returns sample-accurate position information
+ StringCchCat(buf, 256, TEXT("(SAMPLEACCURATE)"));
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
+}
+
+// ----------------------------------------------------------------------------
+// TraceWaveInError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::TraceWaveInError(MMRESULT error) const
+{
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR msg[MAXERRORLENGTH];
+
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ waveInGetErrorText(error, msg, MAXERRORLENGTH);
+ StringCchCat(buf, MAXERRORLENGTH, msg);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
+}
+
+// ----------------------------------------------------------------------------
+// TraceWaveOutError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsWave::TraceWaveOutError(MMRESULT error) const
+{
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR msg[MAXERRORLENGTH];
+
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ waveOutGetErrorText(error, msg, MAXERRORLENGTH);
+ StringCchCat(buf, MAXERRORLENGTH, msg);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
+}
+
+// ----------------------------------------------------------------------------
+// PrepareStartPlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartPlayout()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_hWaveOut == NULL)
+ {
+ return -1;
+ }
+
+ // A total of 30ms of data is immediately placed in the SC buffer
+ //
+ WebRtc_Word8 zeroVec[4*PLAY_BUF_SIZE_IN_SAMPLES]; // max allocation
+ memset(zeroVec, 0, 4*PLAY_BUF_SIZE_IN_SAMPLES);
+
+ {
+ Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
+ Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
+ Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
+ }
+
+ _playAcc = 0;
+ _playWarning = 0;
+ _playError = 0;
+ _dc_diff_mean = 0;
+ _dc_y_prev = 0;
+ _dc_penalty_counter = 20;
+ _dc_prevtime = 0;
+ _dc_prevplay = 0;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// PrepareStartRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartRecording()
+{
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_hWaveIn == NULL)
+ {
+ return -1;
+ }
+
+ _playAcc = 0;
+ _recordedBytes = 0;
+ _recPutBackDelay = REC_PUT_BACK_DELAY;
+
+ MMRESULT res;
+ MMTIME mmtime;
+ mmtime.wType = TIME_SAMPLES;
+
+ res = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition(TIME_SAMPLES) failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ _read_samples = mmtime.u.sample;
+ _read_samples_old = _read_samples;
+ _rec_samples_old = mmtime.u.sample;
+ _wrapCounter = 0;
+
+ for (int n = 0; n < N_BUFFERS_IN; n++)
+ {
+ const WebRtc_UWord8 nBytesPerSample = 2*_recChannels;
+
+ // set up the input wave header
+ _waveHeaderIn[n].lpData = _recBuffer[n];
+ _waveHeaderIn[n].dwBufferLength = nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
+ _waveHeaderIn[n].dwFlags = 0;
+ _waveHeaderIn[n].dwBytesRecorded = 0;
+ _waveHeaderIn[n].dwUser = 0;
+
+ memset(_recBuffer[n], 0, nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES);
+
+ // prepare a buffer for waveform-audio input
+ res = waveInPrepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", n, res);
+ TraceWaveInError(res);
+ }
+
+ // send an input buffer to the given waveform-audio input device
+ res = waveInAddBuffer(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", n, res);
+ TraceWaveInError(res);
+ }
+ }
+
+ // start input on the given waveform-audio input device
+ res = waveInStart(_hWaveIn);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStart() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// GetPlayoutBufferDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples)
+{
+ int i;
+ int ms_Header;
+ long playedDifference;
+ int msecInPlayoutBuffer(0); // #milliseconds of audio in the playout buffer
+
+ const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_PLAY_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
+
+ MMRESULT res;
+ MMTIME mmtime;
+
+ if (!_playing)
+ {
+ playedSamples = 0;
+ return (0);
+ }
+
+ // Retrieve the current playback position.
+ //
+ mmtime.wType = TIME_SAMPLES; // number of waveform-audio samples
+ res = waveOutGetPosition(_hWaveOut, &mmtime, sizeof(mmtime));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetPosition() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+
+ writtenSamples = _writtenSamples; // #samples written to the playout buffer
+ playedSamples = mmtime.u.sample; // current playout position in the playout buffer
+
+ // derive remaining amount (in ms) of data in the playout buffer
+ msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
+ // DEBUG_PRINTP("msecInPlayoutBuffer=%u\n", msecInPlayoutBuffer);
+
+ playedDifference = (long) (_playedSamplesOld - playedSamples);
+
+ if (playedDifference > 64000)
+ {
+ // If the sound cards number-of-played-out-samples variable wraps around before
+ // written_sampels wraps around this needs to be adjusted. This can happen on
+ // sound cards that uses less than 32 bits to keep track of number of played out
+ // sampels. To avoid being fooled by sound cards that sometimes produces false
+ // output we compare old value minus the new value with a large value. This is
+ // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
+ // would trigger the wrap-around function if we didn't compare with a large value.
+ // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
+
+ i = 31;
+ while((_playedSamplesOld <= (unsigned long)POW2(i)) && (i > 14)) {
+ i--;
+ }
+
+ if((i < 31) && (i > 14)) {
+ // Avoid adjusting when there is 32-bit wrap-around since that is
+ // something neccessary.
+ //
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "msecleft() => wrap around occured: %d bits used by sound card)", (i+1));
+
+ _writtenSamples = _writtenSamples - POW2(i + 1);
+ writtenSamples = _writtenSamples;
+ msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
+ }
+ }
+ else if ((_writtenSamplesOld > POW2(31)) && (writtenSamples < 96000))
+ {
+ // Wrap around as expected after having used all 32 bits. (But we still
+ // test if the wrap around happened earlier which it should not)
+
+ i = 31;
+ while (_writtenSamplesOld <= (unsigned long)POW2(i)) {
+ i--;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured after having used all 32 bits)");
+
+ _writtenSamplesOld = writtenSamples;
+ _playedSamplesOld = playedSamples;
+ msecInPlayoutBuffer = (int)((writtenSamples + POW2(i + 1) - playedSamples)/nSamplesPerMs);
+
+ }
+ else if ((writtenSamples < 96000) && (playedSamples > POW2(31)))
+ {
+ // Wrap around has, as expected, happened for written_sampels before
+ // playedSampels so we have to adjust for this until also playedSampels
+ // has had wrap around.
+
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured: correction of output is done)");
+
+ _writtenSamplesOld = writtenSamples;
+ _playedSamplesOld = playedSamples;
+ msecInPlayoutBuffer = (int)((writtenSamples + POW2(32) - playedSamples)/nSamplesPerMs);
+ }
+
+ _writtenSamplesOld = writtenSamples;
+ _playedSamplesOld = playedSamples;
+
+
+ // We use the following formaula to track that playout works as it should
+ // y=playedSamples/48 - timeGetTime();
+ // y represent the clock drift between system clock and sound card clock - should be fairly stable
+ // When the exponential mean value of diff(y) goes away from zero something is wrong
+ // The exponential formula will accept 1% clock drift but not more
+ // The driver error means that we will play to little audio and have a high negative clock drift
+ // We kick in our alternative method when the clock drift reaches 20%
+
+ int diff,y;
+ int unsigned time =0;
+
+ // If we have other problems that causes playout glitches
+ // we don't want to switch playout method.
+ // Check if playout buffer is extremely low, or if we haven't been able to
+ // exectue our code in more than 40 ms
+
+ time = timeGetTime();
+
+ if ((msecInPlayoutBuffer < 20) || (time - _dc_prevtime > 40))
+ {
+ _dc_penalty_counter = 100;
+ }
+
+ if ((playedSamples != 0))
+ {
+ y = playedSamples/48 - time;
+ if ((_dc_y_prev != 0) && (_dc_penalty_counter == 0))
+ {
+ diff = y - _dc_y_prev;
+ _dc_diff_mean = (990*_dc_diff_mean)/1000 + 10*diff;
+ }
+ _dc_y_prev = y;
+ }
+
+ if (_dc_penalty_counter)
+ {
+ _dc_penalty_counter--;
+ }
+
+ if (_dc_diff_mean < -200)
+ {
+ // Always reset the filter
+ _dc_diff_mean = 0;
+
+ // Problem is detected. Switch delay method and set min buffer to 80.
+ // Reset the filter and keep monitoring the filter output.
+ // If issue is detected a second time, increase min buffer to 100.
+ // If that does not help, we must modify this scheme further.
+
+ _useHeader++;
+ if (_useHeader == 1)
+ {
+ _minPlayBufDelay = 80;
+ _playWarning = 1; // only warn first time
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #1: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
+ }
+ else if (_useHeader == 2)
+ {
+ _minPlayBufDelay = 100; // add some more safety
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #2: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
+ }
+ else
+ {
+ // This state should not be entered... (HA)
+ WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "further actions are required!");
+ }
+ if (_playWarning == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout warning exists");
+ }
+ _playWarning = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kPlayoutWarning message posted: switching to alternative playout delay method");
+ }
+ _dc_prevtime = time;
+ _dc_prevplay = playedSamples;
+
+ // Try a very rough method of looking at how many buffers are still playing
+ ms_Header = 0;
+ for (i = 0; i < N_BUFFERS_OUT; i++) {
+ if ((_waveHeaderOut[i].dwFlags & WHDR_INQUEUE)!=0) {
+ ms_Header += 10;
+ }
+ }
+
+ if ((ms_Header-50) > msecInPlayoutBuffer) {
+ // Test for cases when GetPosition appears to be screwed up (currently just log....)
+ TCHAR infoStr[300];
+ if (_no_of_msecleft_warnings%20==0)
+ {
+ StringCchPrintf(infoStr, 300, TEXT("writtenSamples=%i, playedSamples=%i, msecInPlayoutBuffer=%i, ms_Header=%i"), writtenSamples, playedSamples, msecInPlayoutBuffer, ms_Header);
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)infoStr);
+ }
+ _no_of_msecleft_warnings++;
+ }
+
+ // If this is true we have had a problem with the playout
+ if (_useHeader > 0)
+ {
+ return (ms_Header);
+ }
+
+
+ if (ms_Header < msecInPlayoutBuffer)
+ {
+ if (_no_of_msecleft_warnings % 100 == 0)
+ {
+ TCHAR str[300];
+ StringCchPrintf(str, 300, TEXT("_no_of_msecleft_warnings=%i, msecInPlayoutBuffer=%i ms_Header=%i (minBuffer=%i buffersize=%i writtenSamples=%i playedSamples=%i)"),
+ _no_of_msecleft_warnings, msecInPlayoutBuffer, ms_Header, _minPlayBufDelay, _playBufDelay, writtenSamples, playedSamples);
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)str);
+ }
+ _no_of_msecleft_warnings++;
+ ms_Header -= 6; // Round off as we only have 10ms resolution + Header info is usually slightly delayed compared to GetPosition
+
+ if (ms_Header < 0)
+ ms_Header = 0;
+
+ return (ms_Header);
+ }
+ else
+ {
+ return (msecInPlayoutBuffer);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// GetRecordingBufferDelay
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples)
+{
+ long recDifference;
+ MMTIME mmtime;
+ MMRESULT mmr;
+
+ const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_REC_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
+
+ // Retrieve the current input position of the given waveform-audio input device
+ //
+ mmtime.wType = TIME_SAMPLES;
+ mmr = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
+ if (MMSYSERR_NOERROR != mmr)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition() failed (err=%d)", mmr);
+ TraceWaveInError(mmr);
+ }
+
+ readSamples = _read_samples; // updated for each full fram in RecProc()
+ recSamples = mmtime.u.sample; // remaining time in input queue (recorded but not read yet)
+
+ recDifference = (long) (_rec_samples_old - recSamples);
+
+ if( recDifference > 64000) {
+ WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 1 (recDifference =%d)", recDifference);
+ // If the sound cards number-of-recorded-samples variable wraps around before
+ // read_sampels wraps around this needs to be adjusted. This can happen on
+ // sound cards that uses less than 32 bits to keep track of number of played out
+ // sampels. To avoid being fooled by sound cards that sometimes produces false
+ // output we compare old value minus the new value with a large value. This is
+ // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
+ // would trigger the wrap-around function if we didn't compare with a large value.
+ // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
+ //
+ int i = 31;
+ while((_rec_samples_old <= (unsigned long)POW2(i)) && (i > 14))
+ i--;
+
+ if((i < 31) && (i > 14)) {
+ // Avoid adjusting when there is 32-bit wrap-around since that is
+ // somethying neccessary.
+ //
+ _read_samples = _read_samples - POW2(i + 1);
+ readSamples = _read_samples;
+ _wrapCounter++;
+ } else {
+ WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"AEC (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
+ }
+ }
+
+ if((_wrapCounter>200)){
+ // Do nothing, handled later
+ }
+ else if((_rec_samples_old > POW2(31)) && (recSamples < 96000)) {
+ WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 2 (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
+ // Wrap around as expected after having used all 32 bits.
+ _read_samples_old = readSamples;
+ _rec_samples_old = recSamples;
+ _wrapCounter++;
+ return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
+
+
+ } else if((recSamples < 96000) && (readSamples > POW2(31))) {
+ WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 3 (readSamples %d recSamples %d)",readSamples, recSamples);
+ // Wrap around has, as expected, happened for rec_sampels before
+ // readSampels so we have to adjust for this until also readSampels
+ // has had wrap around.
+ _read_samples_old = readSamples;
+ _rec_samples_old = recSamples;
+ _wrapCounter++;
+ return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
+ }
+
+ _read_samples_old = _read_samples;
+ _rec_samples_old = recSamples;
+ int res=(((int)_rec_samples_old - (int)_read_samples_old)/nSamplesPerMs);
+
+ if((res > 2000)||(res < 0)||(_wrapCounter>200)){
+ // Reset everything
+ WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"msec_read error (res %d wrapCounter %d)",res, _wrapCounter);
+ MMTIME mmtime;
+ mmtime.wType = TIME_SAMPLES;
+
+ mmr=waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
+ if (mmr != MMSYSERR_NOERROR) {
+ WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "waveInGetPosition failed (mmr=%d)", mmr);
+ }
+ _read_samples=mmtime.u.sample;
+ _read_samples_old=_read_samples;
+ _rec_samples_old=mmtime.u.sample;
+
+ // Guess a decent value
+ res = 20;
+ }
+
+ _wrapCounter = 0;
+ return res;
+}
+
+// ============================================================================
+// Thread Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// ThreadFunc
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::ThreadFunc(void* pThis)
+{
+ return (static_cast<AudioDeviceWindowsWave*>(pThis)->ThreadProcess());
+}
+
+// ----------------------------------------------------------------------------
+// ThreadProcess
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsWave::ThreadProcess()
+{
+ WebRtc_UWord32 time(0);
+ WebRtc_UWord32 playDiff(0);
+ WebRtc_UWord32 recDiff(0);
+
+ LONGLONG playTime(0);
+ LONGLONG recTime(0);
+
+ switch (_timeEvent.Wait(1000))
+ {
+ case kEventSignaled:
+ break;
+ case kEventError:
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "EventWrapper::Wait() failed => restarting timer");
+ _timeEvent.StopTimer();
+ _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
+ return true;
+ case kEventTimeout:
+ return true;
+ }
+
+ time = AudioDeviceUtility::GetTimeInMS();
+
+ if (_startPlay)
+ {
+ if (PrepareStartPlayout() == 0)
+ {
+ _prevTimerCheckTime = time;
+ _prevPlayTime = time;
+ _startPlay = false;
+ _playing = true;
+ _playStartEvent.Set();
+ }
+ }
+
+ if (_startRec)
+ {
+ if (PrepareStartRecording() == 0)
+ {
+ _prevTimerCheckTime = time;
+ _prevRecTime = time;
+ _prevRecByteCheckTime = time;
+ _startRec = false;
+ _recording = true;
+ _recStartEvent.Set();
+ }
+ }
+
+ if (_playing)
+ {
+ playDiff = time - _prevPlayTime;
+ }
+
+ if (_recording)
+ {
+ recDiff = time - _prevRecTime;
+ }
+
+ if (_playing || _recording)
+ {
+ RestartTimerIfNeeded(time);
+ }
+
+ if (_playing &&
+ (playDiff > (WebRtc_UWord32)(_dTcheckPlayBufDelay - 1)) ||
+ (playDiff < 0))
+ {
+ Lock();
+ if (_playing)
+ {
+ if (PlayProc(playTime) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
+ }
+ _prevPlayTime = time;
+ if (playTime != 0)
+ _playAcc += playTime;
+ }
+ UnLock();
+ }
+
+ if (_playing && (playDiff > 12))
+ {
+ // It has been a long time since we were able to play out, try to
+ // compensate by calling PlayProc again.
+ //
+ Lock();
+ if (_playing)
+ {
+ if (PlayProc(playTime))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
+ }
+ _prevPlayTime = time;
+ if (playTime != 0)
+ _playAcc += playTime;
+ }
+ UnLock();
+ }
+
+ if (_recording &&
+ (recDiff > REC_CHECK_TIME_PERIOD_MS) ||
+ (recDiff < 0))
+ {
+ Lock();
+ if (_recording)
+ {
+ WebRtc_Word32 nRecordedBytes(0);
+ WebRtc_UWord16 maxIter(10);
+
+ // Deliver all availiable recorded buffers and update the CPU load measurement.
+ // We use a while loop here to compensate for the fact that the multi-media timer
+ // can sometimed enter a "bad state" after hibernation where the resolution is
+ // reduced from ~1ms to ~10-15 ms.
+ //
+ while ((nRecordedBytes = RecProc(recTime)) > 0)
+ {
+ maxIter--;
+ _recordedBytes += nRecordedBytes;
+ if (recTime && _perfFreq.QuadPart)
+ {
+ // Measure the average CPU load:
+ // This is a simplified expression where an exponential filter is used:
+ // _avgCPULoad = 0.99 * _avgCPULoad + 0.01 * newCPU,
+ // newCPU = (recTime+playAcc)/f is time in seconds
+ // newCPU / 0.01 is the fraction of a 10 ms period
+ // The two 0.01 cancels each other.
+ // NOTE - assumes 10ms audio buffers.
+ //
+ _avgCPULoad = (float)(_avgCPULoad*.99 + (recTime+_playAcc)/(double)(_perfFreq.QuadPart));
+ _playAcc = 0;
+ }
+ if (maxIter == 0)
+ {
+ // If we get this message ofte, our compensation scheme is not sufficient.
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "failed to compensate for reduced MM-timer resolution");
+ }
+ }
+
+ if (nRecordedBytes == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "RecProc() failed");
+ }
+
+ _prevRecTime = time;
+
+ // Monitor the recording process and generate error/warning callbacks if needed
+ MonitorRecording(time);
+ }
+ UnLock();
+ }
+
+ if (!_recording)
+ {
+ _prevRecByteCheckTime = time;
+ _avgCPULoad = 0;
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// RecProc
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
+{
+ MMRESULT res;
+ WebRtc_UWord32 bufCount(0);
+ WebRtc_UWord32 nBytesRecorded(0);
+
+ consumedTime = 0;
+
+ // count modulo N_BUFFERS_IN (0,1,2,...,(N_BUFFERS_IN-1),0,1,2,..)
+ if (_recBufCount == N_BUFFERS_IN)
+ {
+ _recBufCount = 0;
+ }
+
+ bufCount = _recBufCount;
+
+ // take mono/stereo mode into account when deriving size of a full buffer
+ const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
+ const WebRtc_UWord32 fullBufferSizeInBytes = bytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
+
+ // read number of recorded bytes for the given input-buffer
+ nBytesRecorded = _waveHeaderIn[bufCount].dwBytesRecorded;
+
+ if (nBytesRecorded == fullBufferSizeInBytes ||
+ (nBytesRecorded > 0))
+ {
+ WebRtc_Word32 msecOnPlaySide;
+ WebRtc_Word32 msecOnRecordSide;
+ WebRtc_UWord32 writtenSamples;
+ WebRtc_UWord32 playedSamples;
+ WebRtc_UWord32 readSamples, recSamples;
+ bool send = true;
+
+ WebRtc_UWord32 nSamplesRecorded = (nBytesRecorded/bytesPerSample); // divide by 2 or 4 depending on mono or stereo
+
+ if (nBytesRecorded == fullBufferSizeInBytes)
+ {
+ _timesdwBytes = 0;
+ }
+ else
+ {
+ // Test if it is stuck on this buffer
+ _timesdwBytes++;
+ if (_timesdwBytes < 5)
+ {
+ // keep trying
+ return (0);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id,"nBytesRecorded=%d => don't use", nBytesRecorded);
+ _timesdwBytes = 0;
+ send = false;
+ }
+ }
+
+ // store the recorded buffer (no action will be taken if the #recorded samples is not a full buffer)
+ _ptrAudioBuffer->SetRecordedBuffer(_waveHeaderIn[bufCount].lpData, nSamplesRecorded);
+
+ // update #samples read
+ _read_samples += nSamplesRecorded;
+
+ // Check how large the playout and recording buffers are on the sound card.
+ // This info is needed by the AEC.
+ //
+ msecOnPlaySide = GetPlayoutBufferDelay(writtenSamples, playedSamples);
+ msecOnRecordSide = GetRecordingBufferDelay(readSamples, recSamples);
+
+ // If we use the alternative playout delay method, skip the clock drift compensation
+ // since it will be an unreliable estimate and might degrade AEC performance.
+ WebRtc_Word32 drift = (_useHeader > 0) ? 0 : GetClockDrift(playedSamples, recSamples);
+
+ _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, drift);
+
+ // Store the play and rec delay values for video synchronization
+ _sndCardPlayDelay = msecOnPlaySide;
+ _sndCardRecDelay = msecOnRecordSide;
+
+ LARGE_INTEGER t1,t2;
+
+ if (send)
+ {
+ QueryPerformanceCounter(&t1);
+
+ // deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ Lock();
+
+ QueryPerformanceCounter(&t2);
+
+ if (InputSanityCheckAfterUnlockedPeriod() == -1)
+ {
+ // assert(false);
+ return -1;
+ }
+ }
+
+ if (_AGC)
+ {
+ WebRtc_UWord32 newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0)
+ {
+ // The VQE will only deliver non-zero microphone levels when a change is needed.
+ WEBRTC_TRACE(kTraceStream, kTraceUtility, _id,"AGC change of volume: => new=%u", newMicLevel);
+
+ // We store this outside of the audio buffer to avoid
+ // having it overwritten by the getter thread.
+ _newMicLevel = newMicLevel;
+ SetEvent(_hSetCaptureVolumeEvent);
+ }
+ }
+
+ // return utilized buffer to queue after specified delay (default is 4)
+ if (_recDelayCount > (_recPutBackDelay-1))
+ {
+ // deley buffer counter to compensate for "put-back-delay"
+ bufCount = (bufCount + N_BUFFERS_IN - _recPutBackDelay) % N_BUFFERS_IN;
+
+ // reset counter so we can make new detection
+ _waveHeaderIn[bufCount].dwBytesRecorded = 0;
+
+ // return the utilized wave-header after certain delay (given by _recPutBackDelay)
+ res = waveInUnprepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader(%d) failed (err=%d)", bufCount, res);
+ TraceWaveInError(res);
+ }
+
+ // ensure that the utilized header can be used again
+ res = waveInPrepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", bufCount, res);
+ TraceWaveInError(res);
+ return -1;
+ }
+
+ // add the utilized buffer to the queue again
+ res = waveInAddBuffer(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", bufCount, res);
+ TraceWaveInError(res);
+ if (_recPutBackDelay < 50)
+ {
+ _recPutBackDelay++;
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "_recPutBackDelay increased to %d", _recPutBackDelay);
+ }
+ else
+ {
+ if (_recError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
+ }
+ _recError = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: _recPutBackDelay=%u", _recPutBackDelay);
+ }
+ }
+ } // if (_recDelayCount > (_recPutBackDelay-1))
+
+ if (_recDelayCount < (_recPutBackDelay+1))
+ {
+ _recDelayCount++;
+ }
+
+ // increase main buffer count since one complete buffer has now been delivered
+ _recBufCount++;
+
+ if (send) {
+ // Calculate processing time
+ consumedTime = (int)(t2.QuadPart-t1.QuadPart);
+ // handle wraps, time should not be higher than a second
+ if ((consumedTime > _perfFreq.QuadPart) || (consumedTime < 0))
+ consumedTime = 0;
+ }
+
+ } // if ((nBytesRecorded == fullBufferSizeInBytes))
+
+ return nBytesRecorded;
+}
+
+// ----------------------------------------------------------------------------
+// PlayProc
+// ----------------------------------------------------------------------------
+
+int AudioDeviceWindowsWave::PlayProc(LONGLONG& consumedTime)
+{
+ WebRtc_Word32 remTimeMS(0);
+ WebRtc_Word8 playBuffer[4*PLAY_BUF_SIZE_IN_SAMPLES];
+ WebRtc_UWord32 writtenSamples(0);
+ WebRtc_UWord32 playedSamples(0);
+
+ LARGE_INTEGER t1;
+ LARGE_INTEGER t2;
+
+ consumedTime = 0;
+ _waitCounter++;
+
+ // Get number of ms of sound that remains in the sound card buffer for playback.
+ //
+ remTimeMS = GetPlayoutBufferDelay(writtenSamples, playedSamples);
+
+ // The threshold can be adaptive or fixed. The adaptive scheme is updated
+ // also for fixed mode but the updated threshold is not utilized.
+ //
+ const WebRtc_UWord16 thresholdMS =
+ (_playBufType == AudioDeviceModule::kAdaptiveBufferSize) ? _playBufDelay : _playBufDelayFixed;
+
+ if (remTimeMS < thresholdMS + 9)
+ {
+ _dTcheckPlayBufDelay = 5;
+
+ if (remTimeMS == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id, "playout buffer is empty => we must adapt...");
+ if (_waitCounter > 30)
+ {
+ _erZeroCounter++;
+ if (_erZeroCounter == 2)
+ {
+ _playBufDelay += 15;
+ _minPlayBufDelay += 20;
+ _waitCounter = 50;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0,erZero=2): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
+ }
+ else if (_erZeroCounter == 3)
+ {
+ _erZeroCounter = 0;
+ _playBufDelay += 30;
+ _minPlayBufDelay += 25;
+ _waitCounter = 0;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=3): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
+ }
+ else
+ {
+ _minPlayBufDelay += 10;
+ _playBufDelay += 15;
+ _waitCounter = 50;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=1): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
+ }
+ }
+ }
+ else if (remTimeMS < _minPlayBufDelay)
+ {
+ // If there is less than 25 ms of audio in the play out buffer
+ // increase the buffersize limit value. _waitCounter prevents
+ // _playBufDelay to be increased every time this function is called.
+
+ if (_waitCounter > 30)
+ {
+ _playBufDelay += 10;
+ if (_intro == 0)
+ _waitCounter = 0;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is increased: playBufDelay=%u", _playBufDelay);
+ }
+ }
+ else if (remTimeMS < thresholdMS - 9)
+ {
+ _erZeroCounter = 0;
+ }
+ else
+ {
+ _erZeroCounter = 0;
+ _dTcheckPlayBufDelay = 10;
+ }
+
+ QueryPerformanceCounter(&t1); // measure time: START
+
+ // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+ // Ensure that this callback is executed without taking the audio-thread lock.
+ //
+ UnLock();
+ WebRtc_UWord32 nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES);
+ Lock();
+
+ if (OutputSanityCheckAfterUnlockedPeriod() == -1)
+ {
+ // assert(false);
+ return -1;
+ }
+
+ nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+ if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
+ }
+
+ QueryPerformanceCounter(&t2); // measure time: STOP
+ consumedTime = (int)(t2.QuadPart - t1.QuadPart);
+
+ Write(playBuffer, PLAY_BUF_SIZE_IN_SAMPLES);
+
+ } // if (er < thresholdMS + 9)
+ else if (thresholdMS + 9 < remTimeMS )
+ {
+ _erZeroCounter = 0;
+ _dTcheckPlayBufDelay = 2; // check buffer more often
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Need to check playout buffer more often (dT=%u, remTimeMS=%u)", _dTcheckPlayBufDelay, remTimeMS);
+ }
+
+ // If the buffersize has been stable for 20 seconds try to decrease the buffer size
+ if (_waitCounter > 2000)
+ {
+ _intro = 0;
+ _playBufDelay--;
+ _waitCounter = 1990;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is decreased: playBufDelay=%u", _playBufDelay);
+ }
+
+ // Limit the minimum sound card (playback) delay to adaptive minimum delay
+ if (_playBufDelay < _minPlayBufDelay)
+ {
+ _playBufDelay = _minPlayBufDelay;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %u", _minPlayBufDelay);
+ }
+
+ // Limit the maximum sound card (playback) delay to 150 ms
+ if (_playBufDelay > 150)
+ {
+ _playBufDelay = 150;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %d", _playBufDelay);
+ }
+
+ // Upper limit of the minimum sound card (playback) delay to 65 ms.
+ // Deactivated during "useHeader mode" (_useHeader > 0).
+ if (_minPlayBufDelay > _MAX_minBuffer &&
+ (_useHeader == 0))
+ {
+ _minPlayBufDelay = _MAX_minBuffer;
+ WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Minimum playout threshold is limited to %d", _MAX_minBuffer);
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// Write
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples)
+{
+ if (_hWaveOut == NULL)
+ {
+ return -1;
+ }
+
+ if (_playIsInitialized)
+ {
+ MMRESULT res;
+
+ const WebRtc_UWord16 bufCount(_playBufCount);
+
+ // Place data in the memory associated with _waveHeaderOut[bufCount]
+ //
+ const WebRtc_Word16 nBytes = (2*_playChannels)*nSamples;
+ memcpy(&_playBuffer[bufCount][0], &data[0], nBytes);
+
+ // Send a data block to the given waveform-audio output device.
+ //
+ // When the buffer is finished, the WHDR_DONE bit is set in the dwFlags
+ // member of the WAVEHDR structure. The buffer must be prepared with the
+ // waveOutPrepareHeader function before it is passed to waveOutWrite.
+ // Unless the device is paused by calling the waveOutPause function,
+ // playback begins when the first data block is sent to the device.
+ //
+ res = waveOutWrite(_hWaveOut, &_waveHeaderOut[bufCount], sizeof(_waveHeaderOut[bufCount]));
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutWrite(%d) failed (err=%d)", bufCount, res);
+ TraceWaveOutError(res);
+
+ _writeErrors++;
+ if (_writeErrors > 10)
+ {
+ if (_playError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout error exists");
+ }
+ _playError = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: _writeErrors=%u", _writeErrors);
+ }
+
+ return -1;
+ }
+
+ _playBufCount = (_playBufCount+1) % N_BUFFERS_OUT; // increase buffer counter modulo size of total buffer
+ _writtenSamples += nSamples; // each sample is 2 or 4 bytes
+ _writeErrors = 0;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// GetClockDrift
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp)
+{
+ int drift = 0;
+ unsigned int plSampDiff = 0, rcSampDiff = 0;
+
+ if (plSamp >= _plSampOld)
+ {
+ plSampDiff = plSamp - _plSampOld;
+ }
+ else
+ {
+ // Wrap
+ int i = 31;
+ while(_plSampOld <= (unsigned int)POW2(i))
+ {
+ i--;
+ }
+
+ // Add the amount remaining prior to wrapping
+ plSampDiff = plSamp + POW2(i + 1) - _plSampOld;
+ }
+
+ if (rcSamp >= _rcSampOld)
+ {
+ rcSampDiff = rcSamp - _rcSampOld;
+ }
+ else
+ { // Wrap
+ int i = 31;
+ while(_rcSampOld <= (unsigned int)POW2(i))
+ {
+ i--;
+ }
+
+ rcSampDiff = rcSamp + POW2(i + 1) - _rcSampOld;
+ }
+
+ drift = plSampDiff - rcSampDiff;
+
+ _plSampOld = plSamp;
+ _rcSampOld = rcSamp;
+
+ return drift;
+}
+
+// ----------------------------------------------------------------------------
+// MonitorRecording
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::MonitorRecording(const WebRtc_UWord32 time)
+{
+ const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
+ const WebRtc_UWord32 nRecordedSamples = _recordedBytes/bytesPerSample;
+
+ if (nRecordedSamples > 5*N_REC_SAMPLES_PER_SEC)
+ {
+ // 5 seconds of audio has been recorded...
+ if ((time - _prevRecByteCheckTime) > 5700)
+ {
+ // ...and it was more than 5.7 seconds since we last did this check <=>
+ // we have not been able to record 5 seconds of audio in 5.7 seconds,
+ // hence a problem should be reported.
+ // This problem can be related to USB overload.
+ //
+ if (_recWarning == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording warning exists");
+ }
+ _recWarning = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kRecordingWarning message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
+ }
+
+ _recordedBytes = 0; // restart "check again when 5 seconds are recorded"
+ _prevRecByteCheckTime = time; // reset timer to measure time for recording of 5 seconds
+ }
+
+ if ((time - _prevRecByteCheckTime) > 8000)
+ {
+ // It has been more than 8 seconds since we able to confirm that 5 seconds of
+ // audio was recorded, hence we have not been able to record 5 seconds in
+ // 8 seconds => the complete recording process is most likely dead.
+ //
+ if (_recError == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
+ }
+ _recError = 1; // triggers callback from module process thread
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
+
+ _prevRecByteCheckTime = time;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MonitorRecording
+//
+// Restart timer if needed (they seem to be messed up after a hibernate).
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioDeviceWindowsWave::RestartTimerIfNeeded(const WebRtc_UWord32 time)
+{
+ const WebRtc_UWord32 diffMS = time - _prevTimerCheckTime;
+ _prevTimerCheckTime = time;
+
+ if (diffMS > 7)
+ {
+ // one timer-issue detected...
+ _timerFaults++;
+ if (_timerFaults > 5 && _timerRestartAttempts < 2)
+ {
+ // Reinitialize timer event if event fails to execute at least every 5ms.
+ // On some machines it helps and the timer starts working as it should again;
+ // however, not all machines (we have seen issues on e.g. IBM T60).
+ // Therefore, the scheme below ensures that we do max 2 attempts to restart the timer.
+ // For the cases where restart does not do the trick, we compensate for the reduced
+ // resolution on both the recording and playout sides.
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, " timer issue detected => timer is restarted");
+ _timeEvent.StopTimer();
+ _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
+ // make sure timer gets time to start up and we don't kill/start timer serveral times over and over again
+ _timerFaults = -20;
+ _timerRestartAttempts++;
+ }
+ }
+ else
+ {
+ // restart timer-check scheme since we are OK
+ _timerFaults = 0;
+ _timerRestartAttempts = 0;
+ }
+
+ return 0;
+}
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h
new file mode 100644
index 0000000..76607f6
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h
@@ -0,0 +1,333 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
+
+#include "audio_device_generic.h"
+#include "audio_mixer_manager_win.h"
+
+#pragma comment( lib, "winmm.lib" )
+
+namespace webrtc {
+class EventWrapper;
+class ThreadWrapper;
+
+const WebRtc_UWord32 TIMER_PERIOD_MS = 2;
+const WebRtc_UWord32 REC_CHECK_TIME_PERIOD_MS = 4;
+const WebRtc_UWord16 REC_PUT_BACK_DELAY = 4;
+
+const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
+const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
+
+const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
+const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
+
+// NOTE - CPU load will not be correct for other sizes than 10ms
+const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC/100);
+const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC/100);
+
+enum { N_BUFFERS_IN = 200 };
+enum { N_BUFFERS_OUT = 200 };
+
+class AudioDeviceWindowsWave : public AudioDeviceGeneric
+{
+public:
+ AudioDeviceWindowsWave(const WebRtc_Word32 id);
+ ~AudioDeviceWindowsWave();
+
+ // Retrieve the currently utilized audio layer
+ virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+
+ // Main initializaton and termination
+ virtual WebRtc_Word32 Init();
+ virtual WebRtc_Word32 Terminate();
+ virtual bool Initialized() const;
+
+ // Device enumeration
+ virtual WebRtc_Word16 PlayoutDevices();
+ virtual WebRtc_Word16 RecordingDevices();
+ virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+ virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
+
+ // Device selection
+ virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+ virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
+ virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
+
+ // Audio transport initialization
+ virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitPlayout();
+ virtual bool PlayoutIsInitialized() const;
+ virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitRecording();
+ virtual bool RecordingIsInitialized() const;
+
+ // Audio transport control
+ virtual WebRtc_Word32 StartPlayout();
+ virtual WebRtc_Word32 StopPlayout();
+ virtual bool Playing() const;
+ virtual WebRtc_Word32 StartRecording();
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool Recording() const;
+
+ // Microphone Automatic Gain Control (AGC)
+ virtual WebRtc_Word32 SetAGC(bool enable);
+ virtual bool AGC() const;
+
+ // Volume control based on the Windows Wave API (Windows only)
+ virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
+ virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
+
+ // Audio mixer initialization
+ virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitSpeaker();
+ virtual bool SpeakerIsInitialized() const;
+ virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
+ virtual WebRtc_Word32 InitMicrophone();
+ virtual bool MicrophoneIsInitialized() const;
+
+ // Speaker volume controls
+ virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Microphone volume controls
+ virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+
+ // Speaker mute control
+ virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetSpeakerMute(bool enable);
+ virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
+
+ // Microphone mute control
+ virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
+ virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+
+ // Microphone boost control
+ virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+
+ // Stereo support
+ virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoPlayout(bool enable);
+ virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
+ virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
+ virtual WebRtc_Word32 SetStereoRecording(bool enable);
+ virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
+
+ // Delay information and control
+ virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
+ virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
+ virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
+ virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
+
+ // CPU load
+ virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
+
+public:
+ virtual bool PlayoutWarning() const;
+ virtual bool PlayoutError() const;
+ virtual bool RecordingWarning() const;
+ virtual bool RecordingError() const;
+ virtual void ClearPlayoutWarning();
+ virtual void ClearPlayoutError();
+ virtual void ClearRecordingWarning();
+ virtual void ClearRecordingError();
+
+public:
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+private:
+ void Lock() { _critSect.Enter(); };
+ void UnLock() { _critSect.Leave(); };
+ WebRtc_Word32 Id() {return _id;}
+ bool IsUsingOutputDeviceIndex() const {return _usingOutputDeviceIndex;}
+ AudioDeviceModule::WindowsDeviceType OutputDevice() const {return _outputDevice;}
+ WebRtc_UWord16 OutputDeviceIndex() const {return _outputDeviceIndex;}
+ bool IsUsingInputDeviceIndex() const {return _usingInputDeviceIndex;}
+ AudioDeviceModule::WindowsDeviceType InputDevice() const {return _inputDevice;}
+ WebRtc_UWord16 InputDeviceIndex() const {return _inputDeviceIndex;}
+
+private:
+ inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
+ inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
+
+private:
+ WebRtc_Word32 EnumeratePlayoutDevices();
+ WebRtc_Word32 EnumerateRecordingDevices();
+ void TraceSupportFlags(DWORD dwSupport) const;
+ void TraceWaveInError(MMRESULT error) const;
+ void TraceWaveOutError(MMRESULT error) const;
+ WebRtc_Word32 PrepareStartRecording();
+ WebRtc_Word32 PrepareStartPlayout();
+
+ WebRtc_Word32 RecProc(LONGLONG& consumedTime);
+ int PlayProc(LONGLONG& consumedTime);
+
+ WebRtc_Word32 GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples);
+ WebRtc_Word32 GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples);
+ WebRtc_Word32 Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples);
+ WebRtc_Word32 GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp);
+ WebRtc_Word32 MonitorRecording(const WebRtc_UWord32 time);
+ WebRtc_Word32 RestartTimerIfNeeded(const WebRtc_UWord32 time);
+
+private:
+ static bool ThreadFunc(void*);
+ bool ThreadProcess();
+
+ static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
+ DWORD DoGetCaptureVolumeThread();
+
+ static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
+ DWORD DoSetCaptureVolumeThread();
+
+private:
+ AudioDeviceBuffer* _ptrAudioBuffer;
+
+ CriticalSectionWrapper& _critSect;
+ EventWrapper& _timeEvent;
+ EventWrapper& _recStartEvent;
+ EventWrapper& _playStartEvent;
+
+ HANDLE _hGetCaptureVolumeThread;
+ HANDLE _hShutdownGetVolumeEvent;
+ HANDLE _hSetCaptureVolumeThread;
+ HANDLE _hShutdownSetVolumeEvent;
+ HANDLE _hSetCaptureVolumeEvent;
+
+ ThreadWrapper* _ptrThread;
+ WebRtc_UWord32 _threadID;
+
+ CriticalSectionWrapper& _critSectCb;
+
+ WebRtc_Word32 _id;
+
+ AudioMixerManager _mixerManager;
+
+ bool _usingInputDeviceIndex;
+ bool _usingOutputDeviceIndex;
+ AudioDeviceModule::WindowsDeviceType _inputDevice;
+ AudioDeviceModule::WindowsDeviceType _outputDevice;
+ WebRtc_UWord16 _inputDeviceIndex;
+ WebRtc_UWord16 _outputDeviceIndex;
+ bool _inputDeviceIsSpecified;
+ bool _outputDeviceIsSpecified;
+
+ WAVEFORMATEX _waveFormatIn;
+ WAVEFORMATEX _waveFormatOut;
+
+ HWAVEIN _hWaveIn;
+ HWAVEOUT _hWaveOut;
+
+ WAVEHDR _waveHeaderIn[N_BUFFERS_IN];
+ WAVEHDR _waveHeaderOut[N_BUFFERS_OUT];
+
+ WebRtc_UWord8 _recChannels;
+ WebRtc_UWord8 _playChannels;
+ WebRtc_UWord16 _recBufCount;
+ WebRtc_UWord16 _recDelayCount;
+ WebRtc_UWord16 _recPutBackDelay;
+
+ WebRtc_Word8 _recBuffer[N_BUFFERS_IN][4*REC_BUF_SIZE_IN_SAMPLES];
+ WebRtc_Word8 _playBuffer[N_BUFFERS_OUT][4*PLAY_BUF_SIZE_IN_SAMPLES];
+
+ AudioDeviceModule::BufferType _playBufType;
+
+private:
+ bool _initialized;
+ bool _recording;
+ bool _playing;
+ bool _recIsInitialized;
+ bool _playIsInitialized;
+ bool _startRec;
+ bool _stopRec;
+ bool _startPlay;
+ bool _stopPlay;
+ bool _AGC;
+
+private:
+ WebRtc_UWord32 _prevPlayTime;
+ WebRtc_UWord32 _prevRecTime;
+ WebRtc_UWord32 _prevTimerCheckTime;
+
+ WebRtc_UWord16 _playBufCount; // playout buffer index
+ WebRtc_UWord16 _dTcheckPlayBufDelay; // dT for check of play buffer, {2,5,10} [ms]
+ WebRtc_UWord16 _playBufDelay; // playback delay
+ WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
+ WebRtc_UWord16 _minPlayBufDelay; // minimum playback delay
+ WebRtc_UWord16 _MAX_minBuffer; // level of (adaptive) min threshold must be < _MAX_minBuffer
+
+ WebRtc_Word32 _erZeroCounter; // counts "buffer-is-empty" events
+ WebRtc_Word32 _intro;
+ WebRtc_Word32 _waitCounter;
+
+ WebRtc_UWord32 _writtenSamples;
+ WebRtc_UWord32 _writtenSamplesOld;
+ WebRtc_UWord32 _playedSamplesOld;
+
+ WebRtc_UWord32 _sndCardPlayDelay;
+ WebRtc_UWord32 _sndCardRecDelay;
+
+ WebRtc_UWord32 _plSampOld;
+ WebRtc_UWord32 _rcSampOld;
+
+ WebRtc_UWord32 _read_samples;
+ WebRtc_UWord32 _read_samples_old;
+ WebRtc_UWord32 _rec_samples_old;
+
+ // State that detects driver problems:
+ WebRtc_Word32 _dc_diff_mean;
+ WebRtc_Word32 _dc_y_prev;
+ WebRtc_Word32 _dc_penalty_counter;
+ WebRtc_Word32 _dc_prevtime;
+ WebRtc_UWord32 _dc_prevplay;
+
+ WebRtc_UWord32 _recordedBytes; // accumulated #recorded bytes (reset periodically)
+ WebRtc_UWord32 _prevRecByteCheckTime; // time when we last checked the recording process
+
+ // CPU load measurements
+ LARGE_INTEGER _perfFreq;
+ LONGLONG _playAcc; // accumulated time for playout callback
+ float _avgCPULoad; // average total (rec+play) CPU load
+
+ WebRtc_Word32 _wrapCounter;
+
+ WebRtc_Word32 _useHeader;
+ WebRtc_Word16 _timesdwBytes;
+ WebRtc_Word32 _no_of_msecleft_warnings;
+ WebRtc_Word32 _writeErrors;
+ WebRtc_Word32 _timerFaults;
+ WebRtc_Word32 _timerRestartAttempts;
+
+ WebRtc_UWord16 _playWarning;
+ WebRtc_UWord16 _playError;
+ WebRtc_UWord16 _recWarning;
+ WebRtc_UWord16 _recError;
+
+ WebRtc_UWord32 _newMicLevel;
+ WebRtc_UWord32 _minMicVolume;
+ WebRtc_UWord32 _maxMicVolume;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc
new file mode 100644
index 0000000..c3cf72e
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc
@@ -0,0 +1,2722 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_mixer_manager_win.h"
+#include "trace.h"
+
+#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
+#include <cassert> // assert()
+
+#ifdef _WIN32
+// removes warning: "reinterpret_cast: conversion from 'UINT' to 'HMIXEROBJ'
+// of greater size"
+#pragma warning(disable:4312)
+#endif
+
+// Avoids the need of Windows 7 SDK
+#ifndef WAVE_MAPPED_kDefaultCommunicationDevice
+#define WAVE_MAPPED_kDefaultCommunicationDevice 0x0010
+#endif
+
+namespace webrtc {
+
+// ============================================================================
+// CONSTRUCTION/DESTRUCTION
+// ============================================================================
+
+AudioMixerManager::AudioMixerManager(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _inputMixerHandle(NULL),
+ _outputMixerHandle(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s constructed", __FUNCTION__);
+ ClearSpeakerState();
+ ClearMicrophoneState();
+}
+
+AudioMixerManager::~AudioMixerManager()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destructed", __FUNCTION__);
+
+ Close();
+
+ delete &_critSect;
+}
+
+// ============================================================================
+// PUBLIC METHODS
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Close
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::Close()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerHandle != NULL)
+ {
+ mixerClose(_outputMixerHandle);
+ _outputMixerHandle = NULL;
+ }
+ if (_inputMixerHandle != NULL)
+ {
+ mixerClose(_inputMixerHandle);
+ _inputMixerHandle = NULL;
+ }
+ return 0;
+
+}
+
+// ----------------------------------------------------------------------------
+// CloseSpeaker
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::CloseSpeaker()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerHandle == NULL)
+ {
+ return -1;
+ }
+
+ ClearSpeakerState(_outputMixerID);
+
+ mixerClose(_outputMixerHandle);
+ _outputMixerHandle = NULL;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// CloseMicrophone
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::CloseMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ return -1;
+ }
+
+ ClearMicrophoneState(_inputMixerID);
+
+ mixerClose(_inputMixerHandle);
+ _inputMixerHandle = NULL;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// EnumerateAll
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::EnumerateAll()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ UINT nDevices = mixerGetNumDevs();
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
+
+ MIXERCAPS caps;
+ MIXERLINE destLine;
+ MIXERLINE sourceLine;
+ MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
+
+ UINT mixId(0);
+ UINT destId(0);
+ UINT sourceId(0);
+
+ for (mixId = 0; mixId < nDevices; mixId++)
+ {
+ if (!GetCapabilities(mixId, caps, true))
+ continue;
+
+ for (destId = 0; destId < caps.cDestinations; destId++)
+ {
+ GetDestinationLineInfo(mixId, destId, destLine, true);
+ GetAllLineControls(mixId, destLine, controlArray, true);
+
+ for (sourceId = 0; sourceId < destLine.cConnections; sourceId++)
+ {
+ GetSourceLineInfo(mixId, destId, sourceId, sourceLine, true);
+ GetAllLineControls(mixId, sourceLine, controlArray, true);
+ }
+ }
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// EnumerateSpeakers
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::EnumerateSpeakers()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ UINT nDevices = mixerGetNumDevs();
+ if (nDevices > MAX_NUMBER_MIXER_DEVICES)
+ {
+ assert(false);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
+
+ MIXERCAPS caps;
+ MIXERLINE destLine;
+ MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
+
+ UINT mixId(0);
+ UINT destId(0);
+
+ ClearSpeakerState();
+
+ // scan all avaliable mixer devices
+ for (mixId = 0; mixId < nDevices; mixId++)
+ {
+ // get capabilities for the specified mixer ID
+ GetCapabilities(mixId, caps);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
+ // scan all avaliable destinations for this mixer
+ for (destId = 0; destId < caps.cDestinations; destId++)
+ {
+ GetDestinationLineInfo(mixId, destId, destLine);
+ if ((destLine.cControls == 0) || // no controls or
+ (destLine.cConnections == 0) || // no source lines or
+ (destLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) || // disconnected or
+ !(destLine.fdwLine & MIXERLINE_LINEF_ACTIVE)) // inactive
+ {
+ // don't store this line ID since it will not be possible to control
+ continue;
+ }
+ if ((destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_SPEAKERS) ||
+ (destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_HEADPHONES))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found valid speaker/headphone (name: %s, ID: %u)", WideToUTF8(destLine.szName), destLine.dwLineID);
+ _speakerState[mixId].dwLineID = destLine.dwLineID;
+ _speakerState[mixId].speakerIsValid = true;
+ // retrieve all controls for the speaker component
+ GetAllLineControls(mixId, destLine, controlArray);
+ for (UINT c = 0; c < destLine.cControls; c++)
+ {
+ if (controlArray[c].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
+ {
+ _speakerState[mixId].dwVolumeControlID = controlArray[c].dwControlID;
+ _speakerState[mixId].volumeControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[c].szName), controlArray[c].dwControlID);
+ }
+ else if (controlArray[c].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
+ {
+ _speakerState[mixId].dwMuteControlID = controlArray[c].dwControlID;
+ _speakerState[mixId].muteControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[c].szName), controlArray[c].dwControlID);
+ }
+ }
+ break;
+ }
+ }
+ if (!SpeakerIsValid(mixId))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to find a valid speaker destination line", mixId);
+ }
+ }
+
+ if (ValidSpeakers() == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to locate any valid speaker line");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// EnumerateMicrophones
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::EnumerateMicrophones()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ UINT nDevices = mixerGetNumDevs();
+ if (nDevices > MAX_NUMBER_MIXER_DEVICES)
+ {
+ assert(false);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
+
+ MIXERCAPS caps;
+ MIXERLINE destLine;
+ MIXERLINE sourceLine;
+ MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
+
+ UINT mixId(0);
+ UINT destId(0);
+
+ ClearMicrophoneState();
+
+ // scan all avaliable mixer devices
+ for (mixId = 0; mixId < nDevices; mixId++)
+ {
+ // get capabilities for the specified mixer ID
+ GetCapabilities(mixId, caps);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
+ // scan all avaliable destinations for this mixer
+ for (destId = 0; destId < caps.cDestinations; destId++)
+ {
+ GetDestinationLineInfo(mixId, destId, destLine);
+
+ if ((destLine.cConnections == 0) || // no source lines or
+ (destLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) || // disconnected or
+ !(destLine.fdwLine & MIXERLINE_LINEF_ACTIVE)) // inactive
+ {
+ // Don't store this line ID since there are no sources connected to this destination.
+ // Compare with the speaker side where we also exclude lines with no controls.
+ continue;
+ }
+
+ if (destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_WAVEIN)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found valid Wave In destination (name: %s, ID: %u)", WideToUTF8(destLine.szName), destLine.dwLineID);
+ _microphoneState[mixId].dwLineID = destLine.dwLineID;
+ _microphoneState[mixId].microphoneIsValid = true;
+
+ // retrieve all controls for the identified wave-in destination
+ if (!GetAllLineControls(mixId, destLine, controlArray))
+ {
+ // This destination has no controls. We must try to control
+ // one of its sources instead.
+ // This is a rare state but has been found for some
+ // Logitech USB headsets.
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "this destination has no controls => must control source");
+ for (DWORD sourceId = 0; sourceId < destLine.cConnections; sourceId++)
+ {
+ GetSourceLineInfo(mixId, destId, sourceId, sourceLine, false);
+ if (sourceLine.dwComponentType ==
+ MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "found microphone source ( name: %s, ID: %u)",
+ WideToUTF8(sourceLine.szName), sourceId);
+ GetAllLineControls(mixId, sourceLine, controlArray, false);
+ // scan the controls for this source and search for volume,
+ // mute and on/off (<=> boost) controls
+ for (UINT sc = 0; sc < sourceLine.cControls; sc++)
+ {
+ if (controlArray[sc].dwControlType ==
+ MIXERCONTROL_CONTROLTYPE_VOLUME)
+ {
+ // store this volume control
+ _microphoneState[mixId].dwVolumeControlID =
+ controlArray[sc].dwControlID;
+ _microphoneState[mixId].volumeControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "found volume control (name: %s, ID: %u)",
+ WideToUTF8(controlArray[sc].szName),
+ controlArray[sc].dwControlID);
+ }
+ else if (controlArray[sc].dwControlType ==
+ MIXERCONTROL_CONTROLTYPE_MUTE)
+ {
+ // store this mute control
+ _microphoneState[mixId].dwMuteControlID =
+ controlArray[sc].dwControlID;
+ _microphoneState[mixId].muteControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "found mute control (name: %s, ID: %u)",
+ WideToUTF8(controlArray[sc].szName),
+ controlArray[sc].dwControlID);
+ }
+ else if (controlArray[sc].dwControlType ==
+ MIXERCONTROL_CONTROLTYPE_ONOFF ||
+ controlArray[sc].dwControlType ==
+ MIXERCONTROL_CONTROLTYPE_LOUDNESS)
+ {
+ // store this on/off control (most likely a Boost control)
+ _microphoneState[mixId].dwOnOffControlID =
+ controlArray[sc].dwControlID;
+ _microphoneState[mixId].onOffControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "found on/off control (name: %s, ID: %u)",
+ WideToUTF8(controlArray[sc].szName),
+ controlArray[sc].dwControlID);
+ }
+ }
+ }
+ }
+
+ break;
+ }
+
+ // It seems like there are three different configurations we can find in this state:
+ //
+ // (1) The Wave-in destination contains one MUX control only
+ // (2) The Wave-in destination contains one or more controls where one is a volume control
+ // (3) On Vista and Win 7, it seems like case 2 above is extended.
+ // It is common that a Wave-in destination has two master controls (volume and mute),
+ // AND a microphone source as well with its own volume and mute controls with unique
+ // identifiers. Initial tests have shown that it is sufficient to modify the master
+ // controls only. The source controls will "follow" the master settings, hence the
+ // source controls seem to be redundant.
+ //
+ // For case 1, we should locate the selected source and its controls. The MUX setting will
+ // give us the selected source. NOTE - the selecion might not be a microphone.
+ //
+ // For case 2, the volume control works as a master level control and we should use that one.
+ //
+ // For case 3, we use the master controls only and assume that the source control will "follow".
+ //
+ // Examples of case 1: - SigmaTel Audio (built-in)
+ // - add more..
+ //
+ // Examples of case 2: - Plantronics USB Headset
+ // - Eutectics IPP 200 USB phone
+ // - add more...
+ //
+ // Examples of case 3: - Realtek High Definition on Vista (TL)
+ // - add more...
+
+ if ((destLine.cControls == 1) &&
+ (controlArray[0].dwControlType == MIXERCONTROL_CONTROLTYPE_MUX))
+ {
+ // Case 1: MUX control detected => locate the selected source and its volume control
+ // Note that, the selecion might not be a microphone. A warning is given for
+ // this case only, i.e., it is OK to control a selected Line In source as long
+ // as it is connected to the wave-in destination.
+
+ UINT selection(0);
+ const DWORD nItemsInMux(controlArray[0].cMultipleItems);
+
+ // decide which source line that is selected in the mux
+ if (GetSelectedMuxSource(mixId, controlArray[0].dwControlID, nItemsInMux, selection))
+ {
+ // selection now contains the index of the selected source =>
+ // read the line information for this source
+ // if conditions listed below
+ // condition 1: invalid source
+ // condition 2: no controls
+ // condition 3: disconnected
+ // condition 4: inactive
+ if (!GetSourceLineInfo(mixId, destId, selection, sourceLine) ||
+ (sourceLine.cControls == 0) ||
+ (sourceLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) ||
+ !(sourceLine.fdwLine & MIXERLINE_LINEF_ACTIVE))
+ {
+ continue;
+ }
+
+ if (sourceLine.dwComponentType != MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE)
+ {
+ // add more details about the selected source (not a microphone)
+ TraceComponentType(sourceLine.dwComponentType);
+ // send a warning just to inform about the fact that a non-microphone source will be controlled
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "the selected (to be controlled) source is not a microphone type");
+ }
+
+ // retrieve all controls for the selected source
+ GetAllLineControls(mixId, sourceLine, controlArray);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "MUX selection is %u [0,%u]", selection, nItemsInMux-1);
+
+ // scan the controls for this source and search for volume, mute and on/off (<=> boost) controls
+ for (UINT sc = 0; sc < sourceLine.cControls; sc++)
+ {
+ if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
+ {
+ // store this volume control
+ _microphoneState[mixId].dwVolumeControlID = controlArray[sc].dwControlID;
+ _microphoneState[mixId].volumeControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
+ }
+ else if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
+ {
+ // store this mute control
+ _microphoneState[mixId].dwMuteControlID = controlArray[sc].dwControlID;
+ _microphoneState[mixId].muteControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
+ }
+ else if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_ONOFF ||
+ controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_LOUDNESS)
+ {
+ // store this on/off control (most likely a Boost control)
+ _microphoneState[mixId].dwOnOffControlID = controlArray[sc].dwControlID;
+ _microphoneState[mixId].onOffControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found on/off control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
+ }
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to detect which source to control");
+ }
+
+ }
+ else if (destLine.cConnections == 1)
+ {
+ // Case 2 or Case 3:
+
+ GetSourceLineInfo(mixId, destId, 0, sourceLine);
+ if ((sourceLine.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE) &&
+ (sourceLine.cControls > 0))
+ {
+ // Case 3: same as Case 2 below but we have also detected a Microphone source
+ // with its own controls. So far, I have not been able to find any device
+ // where it is required to modify these controls. Until I have found such
+ // a device, this case will be handled as a Case 2 (see below).
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "microphone source controls will not be controlled");
+ }
+ else if ((sourceLine.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE) &&
+ (sourceLine.cControls == 0))
+ {
+ // default state on non Vista/Win 7 machines
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "microphone source has no controls => use master controls instead");
+ }
+ else
+ {
+ // add more details about the selected source (not a microphone)
+ TraceComponentType(sourceLine.dwComponentType);
+ // send a warning just to inform about the fact that a non-microphone source will be controlled
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "the connected (to be controlled) source is not a microphone type");
+ }
+
+ // Case 2 : one source only and no MUX control detected =>
+ // locate the master volume control (and mute + boost controls if possible)
+
+ // scan the controls for this wave-in destination and search for volume, mute and on/off (<=> boost) controls
+ for (UINT dc = 0; dc < destLine.cControls; dc++)
+ {
+ if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
+ {
+ // store this volume control
+ _microphoneState[mixId].dwVolumeControlID = controlArray[dc].dwControlID;
+ _microphoneState[mixId].volumeControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
+ }
+ else if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
+ {
+ // store this mute control
+ _microphoneState[mixId].dwMuteControlID = controlArray[dc].dwControlID;
+ _microphoneState[mixId].muteControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
+ }
+ else if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_ONOFF ||
+ controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_LOUDNESS ||
+ controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_BOOLEAN)
+ {
+ // store this on/off control
+ _microphoneState[mixId].dwOnOffControlID = controlArray[dc].dwControlID;
+ _microphoneState[mixId].onOffControlIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found on/off control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
+ }
+ }
+ }
+ else
+ {
+ // We are in a state where more than one source is connected to the wave-in destination.
+ // I am bailing out here for now until I understand this case better.
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to locate valid microphone controls for this mixer");
+ }
+ break;
+ }
+ } // for (destId = 0; destId < caps.cDestinations; destId++)
+
+ if (!MicrophoneIsValid(mixId))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to find a valid microphone destination line", mixId);
+ }
+ } // for (mixId = 0; mixId < nDevices; mixId++)
+
+ if (ValidMicrophones() == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to locate any valid microphone line");
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// OpenSpeaker I(II)
+//
+// Verifies that the mixer contains a valid speaker destination line.
+// Avoids opening the mixer if valid control has not been found.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::OpenSpeaker(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(kDefaultDevice)");
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(kDefaultCommunicationDevice)");
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Close any existing output mixer handle
+ //
+ if (_outputMixerHandle != NULL)
+ {
+ mixerClose(_outputMixerHandle);
+ _outputMixerHandle = NULL;
+ }
+
+ MMRESULT res;
+ WAVEFORMATEX waveFormat;
+ HWAVEOUT hWaveOut(NULL);
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
+ waveFormat.nChannels = 2;
+ waveFormat.nSamplesPerSec = 48000;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0;
+
+ // We need a waveform-audio output handle for the currently selected output device.
+ // This handle will then give us the corresponding mixer identifier. Once the mixer
+ // ID is known, it is possible to open the output mixer.
+ //
+ if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ // check if it is possible to open the default communication device (supported on Windows 7)
+ res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL |
+ WAVE_MAPPED_kDefaultCommunicationDevice | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // if so, open the default communication device for real
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_kDefaultCommunicationDevice);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
+ }
+ else
+ {
+ // use default device since default communication device was not avaliable
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "unable to open default communication device => using default instead");
+ }
+ }
+ else if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ // open default device since it has been requested
+ res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
+ }
+
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
+ TraceWaveOutError(res);
+ }
+
+ UINT mixerId(0);
+ HMIXER hMixer(NULL);
+
+ // Retrieve the device identifier for a mixer device associated with the
+ // aquired waveform-audio output handle.
+ //
+ res = mixerGetID((HMIXEROBJ)hWaveOut, &mixerId, MIXER_OBJECTF_HWAVEOUT);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEOUT) failed (err=%d)", res);
+ // identification failed => use default mixer identifier (=0)
+ mixerId = 0;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified output device <=> mixer ID %u", mixerId);
+
+ // The waveform-audio output handle is no longer needed.
+ //
+ waveOutClose(hWaveOut);
+
+ // Verify that the mixer contains a valid speaker destination line.
+ // Avoid opening the mixer if valid control has not been found.
+ //
+ if (!SpeakerIsValid(mixerId))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the speaker volume for this mixer device");
+ return -1;
+ }
+
+ // Open the specified mixer device and ensure that the device will not
+ // be removed until the application closes the handle.
+ //
+ res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
+ }
+
+ // Store the output mixer handle and active mixer identifier
+ //
+ _outputMixerHandle = hMixer;
+ _outputMixerID = mixerId;
+
+ if (_outputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the output mixer device is now open (0x%x)", _outputMixerHandle);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// OpenSpeaker II(II)
+//
+// Verifies that the mixer contains a valid speaker destination line.
+// Avoids opening the mixer if valid control has not been found.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::OpenSpeaker(WebRtc_UWord16 index)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(index=%d)", index);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Close any existing output mixer handle
+ //
+ if (_outputMixerHandle != NULL)
+ {
+ mixerClose(_outputMixerHandle);
+ _outputMixerHandle = NULL;
+ }
+
+ MMRESULT res;
+ WAVEFORMATEX waveFormat;
+ HWAVEOUT hWaveOut(NULL);
+
+ const UINT deviceID(index); // use index parameter as device identifier
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
+ waveFormat.nChannels = 2;
+ waveFormat.nSamplesPerSec = 48000;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0;
+
+ // We need a waveform-audio output handle for the currently selected output device.
+ // This handle will then give us the corresponding mixer identifier. Once the mixer
+ // ID is known, it is possible to open the output mixer.
+ //
+ res = waveOutOpen(&hWaveOut, deviceID, &waveFormat, 0, 0, CALLBACK_NULL);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutOpen(deviceID=%u) failed (err=%d)", index, res);
+ TraceWaveOutError(res);
+ }
+
+ UINT mixerId(0);
+ HMIXER hMixer(NULL);
+
+ // Retrieve the device identifier for a mixer device associated with the
+ // aquired waveform-audio output handle.
+ //
+ res = mixerGetID((HMIXEROBJ)hWaveOut, &mixerId, MIXER_OBJECTF_HWAVEOUT);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEOUT) failed (err=%d)", res);
+ // identification failed => use default mixer identifier (=0)
+ mixerId = 0;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified output device <=> mixer ID %u", mixerId);
+
+ // The waveform-audio output handle is no longer needed.
+ //
+ waveOutClose(hWaveOut);
+
+ // Verify that the mixer contains a valid speaker destination line.
+ // Avoid opening the mixer if valid control has not been found.
+ //
+ if (!SpeakerIsValid(mixerId))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the speaker volume for this mixer device");
+ return -1;
+ }
+
+ // Open the specified mixer device and ensure that the device will not
+ // be removed until the application closes the handle.
+ //
+ res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
+ }
+
+ // Store the output mixer handle and active mixer identifier
+ //
+ _outputMixerHandle = hMixer;
+ _outputMixerID = mixerId;
+
+ if (_outputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the output mixer device is now open (0x%x)", _outputMixerHandle);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// OpenMicrophone I(II)
+//
+// Verifies that the mixer contains a valid wave-in destination line.
+// Avoids opening the mixer if valid control has not been found.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::OpenMicrophone(AudioDeviceModule::WindowsDeviceType device)
+{
+ if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(kDefaultDevice)");
+ }
+ else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(kDefaultCommunicationDevice)");
+ }
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Close any existing output mixer handle
+ //
+ if (_inputMixerHandle != NULL)
+ {
+ mixerClose(_inputMixerHandle);
+ _inputMixerHandle = NULL;
+ }
+
+ MMRESULT res;
+ WAVEFORMATEX waveFormat;
+ HWAVEIN hWaveIn(NULL);
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
+ waveFormat.nChannels = 1;
+ waveFormat.nSamplesPerSec = 48000;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0 ;
+
+ // We need a waveform-audio input handle for the currently selected input device.
+ // This handle will then give us the corresponding mixer identifier. Once the mixer
+ // ID is known, it is possible to open the input mixer.
+ //
+ if (device == AudioDeviceModule::kDefaultCommunicationDevice)
+ {
+ // check if it is possible to open the default communication device (supported on Windows 7)
+ res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL |
+ WAVE_MAPPED_kDefaultCommunicationDevice | WAVE_FORMAT_QUERY);
+ if (MMSYSERR_NOERROR == res)
+ {
+ // if so, open the default communication device for real
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_kDefaultCommunicationDevice);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
+ }
+ else
+ {
+ // use default device since default communication device was not avaliable
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "unable to open default communication device => using default instead");
+ }
+ }
+ else if (device == AudioDeviceModule::kDefaultDevice)
+ {
+ // open default device since it has been requested
+ res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
+ }
+
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
+ TraceWaveInError(res);
+ }
+
+ UINT mixerId(0);
+ HMIXER hMixer(NULL);
+
+ // Retrieve the device identifier for a mixer device associated with the
+ // aquired waveform-audio input handle.
+ //
+ res = mixerGetID((HMIXEROBJ)hWaveIn, &mixerId, MIXER_OBJECTF_HWAVEIN);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEIN) failed (err=%d)", res);
+ // identification failed => use default mixer identifier (=0)
+ mixerId = 0;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified input device <=> mixer ID %u", mixerId);
+
+ // The waveform-audio input handle is no longer needed.
+ //
+ waveInClose(hWaveIn);
+
+ // Verify that the mixer contains a valid wave-in destination line and a volume control.
+ // Avoid opening the mixer if valid control has not been found.
+ //
+ if (!MicrophoneIsValid(mixerId))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the microphone volume for this mixer device");
+ return -1;
+ }
+
+ // Open the specified mixer device and ensure that the device will not
+ // be removed until the application closes the handle.
+ //
+ res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
+ }
+
+ // Store the input mixer handle and active mixer identifier
+ //
+ _inputMixerHandle = hMixer;
+ _inputMixerID = mixerId;
+
+ if (_inputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the input mixer device is now open (0x%x)", _inputMixerHandle);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// OpenMicrophone II(II)
+//
+// Verifies that the mixer contains a valid wave-in destination line.
+// Avoids opening the mixer if valid control has not been found.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::OpenMicrophone(WebRtc_UWord16 index)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(index=%d)", index);
+
+ CriticalSectionScoped lock(_critSect);
+
+ // Close any existing input mixer handle
+ //
+ if (_inputMixerHandle != NULL)
+ {
+ mixerClose(_inputMixerHandle);
+ _inputMixerHandle = NULL;
+ }
+
+ MMRESULT res;
+ WAVEFORMATEX waveFormat;
+ HWAVEIN hWaveIn(NULL);
+
+ const UINT deviceID(index); // use index parameter as device identifier
+
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
+ waveFormat.nChannels = 1;
+ waveFormat.nSamplesPerSec = 48000;
+ waveFormat.wBitsPerSample = 16;
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ waveFormat.cbSize = 0;
+
+ // We need a waveform-audio input handle for the currently selected input device.
+ // This handle will then give us the corresponding mixer identifier. Once the mixer
+ // ID is known, it is possible to open the input mixer.
+ //
+ res = waveInOpen(&hWaveIn, deviceID, &waveFormat, 0, 0, CALLBACK_NULL);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInOpen(deviceID=%u) failed (err=%d)", index, res);
+ TraceWaveInError(res);
+ }
+
+ UINT mixerId(0);
+ HMIXER hMixer(NULL);
+
+ // Retrieve the device identifier for a mixer device associated with the
+ // aquired waveform-audio input handle.
+ //
+ res = mixerGetID((HMIXEROBJ)hWaveIn, &mixerId, MIXER_OBJECTF_HWAVEIN);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEIN) failed (err=%d)", res);
+ // identification failed => use default mixer identifier (=0)
+ mixerId = 0;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified input device <=> mixer ID %u", mixerId);
+
+ // The waveform-audio input handle is no longer needed.
+ //
+ waveInClose(hWaveIn);
+
+ // Verify that the mixer contains a valid wave-in destination line.
+ // Avoid opening the mixer if valid control has not been found.
+ //
+ if (!MicrophoneIsValid(mixerId))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the microphone volume for this mixer device");
+ return -1;
+ }
+
+ // Open the specified mixer device and ensure that the device will not
+ // be removed until the application closes the handle.
+ //
+ res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
+ if (MMSYSERR_NOERROR != res)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
+ }
+
+ // Store the input mixer handle and active mixer identifier
+ //
+ _inputMixerHandle = hMixer;
+ _inputMixerID = mixerId;
+
+ if (_inputMixerHandle != NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the input mixer device is now open (0x%x)", _inputMixerHandle);
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::SpeakerIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ return (_outputMixerHandle != NULL);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::MicrophoneIsInitialized() const
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ return (_inputMixerHandle != NULL);
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SetSpeakerVolume(WebRtc_UWord32 volume)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetSpeakerVolume(volume=%u)", volume);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_outputMixerID);
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
+ DWORD dwValue(volume);
+
+ // Set one unsigned control value for a specified volume-control identifier
+ //
+ if (!SetUnsignedControlValue(mixerID, dwControlID, dwValue))
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolume
+//
+// Note that (MIXERCONTROL_CONTROLTYPE_VOLUME & MIXERCONTROL_CT_UNITS_MASK)
+// always equals MIXERCONTROL_CT_UNITS_UNSIGNED;
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SpeakerVolume(WebRtc_UWord32& volume) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_outputMixerID);
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
+ DWORD dwValue(0);
+
+ // Retrieve one unsigned control value for a specified volume-control identifier
+ //
+ if (!GetUnsignedControlValue(mixerID, dwControlID, dwValue))
+ {
+ return -1;
+ }
+
+ volume = dwValue;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MaxSpeakerVolume
+//
+// Note that (MIXERCONTROL_CONTROLTYPE_VOLUME & MIXERCONTROL_CT_UNITS_MASK)
+// always equals MIXERCONTROL_CT_UNITS_UNSIGNED
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_outputMixerID);
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, dwControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ maxVolume = mixerControl.Bounds.dwMaximum;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_outputMixerID);
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, dwControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ minVolume = mixerControl.Bounds.dwMinimum;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_outputMixerID);
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, _speakerState[mixerID].dwVolumeControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ stepSize = static_cast<WebRtc_UWord16> (mixerControl.Metrics.cSteps);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SpeakerVolumeIsAvailable(bool& available)
+{
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ available = _speakerState[_outputMixerID].volumeControlIsValid;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SpeakerMuteIsAvailable(bool& available)
+{
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ available = _speakerState[_outputMixerID].muteControlIsValid;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetSpeakerMute
+//
+// This mute function works a master mute for the output speaker.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SetSpeakerMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetSpeakerMute(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected speaker destination has a valid mute control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the output mixer handle exists.
+ //
+ if (!_speakerState[_outputMixerID].muteControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this speaker line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwMuteControlID);
+
+ // Set one boolean control value for the specified mute-control
+ //
+ if (!SetBooleanControlValue(_outputMixerID, dwControlID, enable))
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SpeakerMute(bool& enabled) const
+{
+
+ if (_outputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected speaker destination has a valid mute control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the output mixer handle exists.
+ //
+ if (!_speakerState[_outputMixerID].muteControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this speaker line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_speakerState[_outputMixerID].dwMuteControlID);
+ bool value(false);
+
+ // Retrieve one boolean control value for a specified mute-control identifier
+ //
+ if (!GetBooleanControlValue(_outputMixerID, dwControlID, value))
+ {
+ return -1;
+ }
+
+ enabled = value;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneMuteIsAvailable(bool& available)
+{
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ available = _microphoneState[_inputMixerID].muteControlIsValid;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneMute
+//
+// This mute function works a master mute for the input microphone.
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SetMicrophoneMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetMicrophoneMute(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected wave-in destinationhas a valid mute control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the input mixer handle exists.
+ //
+ if (!_microphoneState[_inputMixerID].muteControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this microphone line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwMuteControlID);
+
+ // Set one boolean control value for the specified mute-control
+ //
+ if (!SetBooleanControlValue(_inputMixerID, dwControlID, enable))
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneMute
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneMute(bool& enabled) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected wave-in destinationhas a valid mute control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the input mixer handle exists.
+ //
+ if (!_microphoneState[_inputMixerID].muteControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this microphone line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwMuteControlID);
+ bool value(false);
+
+ // Retrieve one boolean control value for a specified mute-control identifier
+ //
+ if (!GetBooleanControlValue(_inputMixerID, dwControlID, value))
+ {
+ return -1;
+ }
+
+ enabled = value;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoostIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneBoostIsAvailable(bool& available)
+{
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ available = _microphoneState[_inputMixerID].onOffControlIsValid;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SetMicrophoneBoost(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetMicrophoneBoost(enable=%u)", enable);
+
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected wave-in destination has a valid boost (on/off) control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the input mixer handle exists.
+ //
+ if (!_microphoneState[_inputMixerID].onOffControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no boost control exists for this wave-in line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwOnOffControlID);
+
+ // Set one boolean control value for the specified boost (on/off) control
+ //
+ if (!SetBooleanControlValue(_inputMixerID, dwControlID, enable))
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneBoost
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneBoost(bool& enabled) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ // Ensure that the selected wave-in destination has a valid boost (on/off) control.
+ // If so, its identifier was stored during the enumeration phase which must
+ // have taken place since the input mixer handle exists.
+ //
+ if (!_microphoneState[_inputMixerID].onOffControlIsValid)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no boost control exists for this wave-in line");
+ return -1;
+ }
+
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwOnOffControlID);
+ bool value(false);
+
+ // Retrieve one boolean control value for a specified boost-control identifier
+ //
+ if (!GetBooleanControlValue(_inputMixerID, dwControlID, value))
+ {
+ return -1;
+ }
+
+ enabled = value;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneVolumeIsAvailable(bool& available)
+{
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ available = _microphoneState[_inputMixerID].volumeControlIsValid;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::SetMicrophoneVolume(WebRtc_UWord32 volume)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_inputMixerID);
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
+ DWORD dwValue(volume);
+
+ // Set one unsigned control value for a specified volume-control identifier
+ //
+ if (!SetUnsignedControlValue(mixerID, dwControlID, dwValue))
+ {
+ return -1;
+ }
+
+ return (0);
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneVolume(WebRtc_UWord32& volume) const
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_inputMixerID);
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
+ DWORD dwValue(0);
+
+ // Retrieve one unsigned control value for a specified volume-control identifier
+ //
+ if (!GetUnsignedControlValue(mixerID, dwControlID, dwValue))
+ {
+ return -1;
+ }
+
+ volume = dwValue;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MaxMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_inputMixerID);
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, dwControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ maxVolume = mixerControl.Bounds.dwMaximum;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_inputMixerID);
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, dwControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ minVolume = mixerControl.Bounds.dwMinimum;
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneVolumeStepSize
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioMixerManager::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
+{
+
+ if (_inputMixerHandle == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
+ return -1;
+ }
+
+ const UINT mixerID(_inputMixerID);
+ const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
+ MIXERCONTROL mixerControl;
+
+ // Retrieve one control line for a specified volume-control identifier
+ //
+ if (!GetLineControl(mixerID, dwControlID, mixerControl))
+ {
+ return -1;
+ }
+
+ stepSize = static_cast<WebRtc_UWord16> (mixerControl.Metrics.cSteps);
+
+ return 0;
+}
+
+// ============================================================================
+// PRIVATE METHODS
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+// Devices
+//
+// A given audio card has one Mixer device associated with it. All of the
+// various components on that card are controlled through that card's one
+// Mixer device.
+// ----------------------------------------------------------------------------
+
+UINT AudioMixerManager::Devices() const
+{
+ UINT nDevs = mixerGetNumDevs();
+ return nDevs;
+}
+
+// ----------------------------------------------------------------------------
+// DestinationLines
+//
+// # destination lines given mixer ID.
+// ----------------------------------------------------------------------------
+
+UINT AudioMixerManager::DestinationLines(UINT mixId) const
+{
+ MIXERCAPS caps;
+ if (!GetCapabilities(mixId, caps))
+ {
+ return 0;
+ }
+ return (caps.cDestinations);
+}
+// ----------------------------------------------------------------------------
+// DestinationLines
+//
+// # source lines given mixer ID and destination ID.
+// ----------------------------------------------------------------------------
+
+UINT AudioMixerManager::SourceLines(UINT mixId, DWORD destId) const
+{
+ MIXERLINE dline;
+ if (!GetDestinationLineInfo(mixId, destId, dline))
+ {
+ return 0;
+ }
+ return (dline.cConnections);
+}
+
+// ----------------------------------------------------------------------------
+// GetCapabilities
+//
+// Queries a specified mixer device to determine its capabilities.
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetCapabilities(UINT mixId, MIXERCAPS& caps, bool trace) const
+{
+ MMRESULT res;
+ MIXERCAPS mcaps;
+
+ res = mixerGetDevCaps(mixId, &mcaps, sizeof(MIXERCAPS));
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetDevCaps() failed (err=%d)", res);
+ return false;
+ }
+
+ memcpy(&caps, &mcaps, sizeof(MIXERCAPS));
+
+ if (trace)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Mixer ID %u:", mixId);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u", caps.wPid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u", caps.vDriverVersion);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", WideToUTF8(caps.szPname));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "misc. support bits : %u", caps.fdwSupport);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "count of destinations: %u (+)", caps.cDestinations);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
+ }
+
+ if (caps.cDestinations == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "invalid number of mixer destinations");
+ return false;
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetDestinationLineInfo
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetDestinationLineInfo(UINT mixId, DWORD destId, MIXERLINE& line, bool trace) const
+{
+ MMRESULT res;
+ MIXERLINE mline;
+
+ mline.cbStruct = sizeof(MIXERLINE);
+ mline.dwDestination = destId; // max destination index is cDestinations-1
+ mline.dwSource = 0; // not set for MIXER_GETLINEINFOF_DESTINATION
+
+ // Retrieve information about the specified destination line of a mixer device.
+ // Note that we use the mixer ID here and not a handle to an opened mixer.
+ // It is not required to open the mixer for enumeration purposes only.
+ //
+ res = mixerGetLineInfo(reinterpret_cast<HMIXEROBJ>(mixId), &mline, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_DESTINATION);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineInfo(MIXER_GETLINEINFOF_DESTINATION) failed (err=%d)", res);
+ return false;
+ }
+
+ memcpy(&line, &mline, sizeof(MIXERLINE));
+
+ if (trace)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "> Destination Line ID %u:", destId);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "destination line index : %u", mline.dwDestination);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwLineID : %lu (unique)", mline.dwLineID);
+ TraceStatusAndSupportFlags(mline.fdwLine);
+ TraceComponentType(mline.dwComponentType);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "count of channels : %u", mline.cChannels);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# audio source lines : %u (+)", mline.cConnections); // valid only for destinations
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# controls : %u (*)", mline.cControls); // can be zero
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "short name : %s", WideToUTF8(mline.szShortName));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(mline.szName));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
+ TraceTargetType(mline.Target.dwType);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "target device ID : %lu", mline.Target.dwDeviceID);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", mline.Target.wMid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u", mline.Target.wPid);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "driver version : %u", mline.Target.vDriverVersion);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", WideToUTF8(mline.Target.szPname));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "---------------------------------------------------------------");
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetSourceLineInfo
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetSourceLineInfo(UINT mixId, DWORD destId, DWORD srcId, MIXERLINE& line, bool trace) const
+{
+ MMRESULT res;
+ MIXERLINE mline;
+
+ mline.cbStruct = sizeof(MIXERLINE);
+ mline.dwDestination = destId; // we want the source info for this destination
+ mline.dwSource = srcId; // source index (enumerate over these)
+
+ // Retrieve information about the specified source line of a mixer device.
+ // Note that we use the mixer ID here and not a handle to an opened mixer.
+ // It is not required to open the mixer for enumeration purposes only.
+ //
+ res = mixerGetLineInfo(reinterpret_cast<HMIXEROBJ>(mixId), &mline, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_SOURCE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineInfo(MIXER_GETLINEINFOF_SOURCE) failed (err=%d)", res);
+ return false;
+ }
+
+ memcpy(&line, &mline, sizeof(MIXERLINE));
+
+ if (trace)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " >> Source Line ID %u:", srcId);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "destination line index : %u", mline.dwDestination);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwSource : %u", mline.dwSource);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwLineID : %lu (unique)", mline.dwLineID);
+ TraceStatusAndSupportFlags(mline.fdwLine);
+ TraceComponentType(mline.dwComponentType);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# controls : %u (*)", mline.cControls);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(mline.szName));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
+ TraceTargetType(mline.Target.dwType);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "---------------------------------------------------------------");
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetAllLineControls
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetAllLineControls(UINT mixId, const MIXERLINE& line, MIXERCONTROL* controlArray, bool trace) const
+{
+ // Ensure that we don't try to aquire information if there are no controls for this line
+ //
+ if (line.cControls == 0)
+ return false;
+
+ MMRESULT res;
+ MIXERLINECONTROLS mlineControls; // contains information about the controls of an audio line
+
+ mlineControls.dwLineID = line.dwLineID; // unique audio line identifier
+ mlineControls.cControls = line.cControls; // number of controls associated with the line
+ mlineControls.pamxctrl = controlArray; // points to the first MIXERCONTROL structure to be filled
+ mlineControls.cbStruct = sizeof(MIXERLINECONTROLS);
+ mlineControls.cbmxctrl = sizeof(MIXERCONTROL);
+
+ // Get information on ALL controls associated with the specified audio line
+ //
+ res = mixerGetLineControls(reinterpret_cast<HMIXEROBJ>(mixId), &mlineControls, MIXER_OBJECTF_MIXER | MIXER_GETLINECONTROLSF_ALL);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineControls(MIXER_GETLINECONTROLSF_ALL) failed (err=%d)", res);
+ return false;
+ }
+
+ if (trace)
+ {
+ for (UINT c = 0; c < line.cControls; c++)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " >> Control ID %u:", c);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwControlID : %u (unique)", controlArray[c].dwControlID);
+ TraceControlType(controlArray[c].dwControlType);
+ TraceControlStatusAndSupportFlags(controlArray[c].fdwControl);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cMultipleItems : %u", controlArray[c].cMultipleItems);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "short name : %s", WideToUTF8(controlArray[c].szShortName));
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(controlArray[c].szName));
+ if ((controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_SIGNED)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "min signed value : %d", controlArray[c].Bounds.lMinimum);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "max signed value : %d", controlArray[c].Bounds.lMaximum);
+ }
+ else if ((controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_UNSIGNED ||
+ (controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_BOOLEAN)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "min unsigned value : %u", controlArray[c].Bounds.dwMinimum);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "max unsigned value : %u", controlArray[c].Bounds.dwMaximum);
+ }
+ if (controlArray[c].dwControlType != MIXERCONTROL_CONTROLTYPE_CUSTOM)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cSteps : %u", controlArray[c].Metrics.cSteps);
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "...............................................................");
+ GetControlDetails(mixId, controlArray[c], true);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "...............................................................");
+
+ }
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetLineControls
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetLineControl(UINT mixId, DWORD dwControlID, MIXERCONTROL& control) const
+{
+ MMRESULT res;
+ MIXERLINECONTROLS mlineControl;
+
+ mlineControl.dwControlID = dwControlID;
+ mlineControl.cControls = 1;
+ mlineControl.pamxctrl = &control;
+ mlineControl.cbStruct = sizeof(MIXERLINECONTROLS);
+ mlineControl.cbmxctrl = sizeof(MIXERCONTROL);
+
+ // Get information on one controls associated with the specified conrol identifier
+ //
+ res = mixerGetLineControls(reinterpret_cast<HMIXEROBJ>(mixId), &mlineControl, MIXER_OBJECTF_MIXER | MIXER_GETLINECONTROLSF_ONEBYID);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineControls(MIXER_GETLINECONTROLSF_ONEBYID) failed (err=%d)", res);
+ return false;
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetControlDetails
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetControlDetails(UINT mixId, MIXERCONTROL& controlArray, bool trace) const
+{
+ assert(controlArray.cMultipleItems <= MAX_NUMBER_OF_MULTIPLE_ITEMS);
+
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+
+ MIXERCONTROLDETAILS_UNSIGNED valueUnsigned[MAX_NUMBER_OF_MULTIPLE_ITEMS];
+ MIXERCONTROLDETAILS_SIGNED valueSigned[MAX_NUMBER_OF_MULTIPLE_ITEMS];
+ MIXERCONTROLDETAILS_BOOLEAN valueBoolean[MAX_NUMBER_OF_MULTIPLE_ITEMS];
+
+ enum ControlType
+ {
+ CT_UNITS_UNSIGNED,
+ CT_UNITS_SIGNED,
+ CT_UNITS_BOOLEAN
+ };
+
+ ControlType ctype(CT_UNITS_UNSIGNED);
+
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.dwControlID = controlArray.dwControlID; // control identifier
+ controlDetails.cChannels = 1; // we need to set values as if they were uniform
+ controlDetails.cMultipleItems = controlArray.cMultipleItems; // only nonzero for CONTROLF_MULTIPLE controls
+ // can e.g. happen for CONTROLTYPE_MUX
+ if (controlDetails.cMultipleItems > MAX_NUMBER_OF_MULTIPLE_ITEMS)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "cMultipleItems > %d", MAX_NUMBER_OF_MULTIPLE_ITEMS);
+ controlDetails.cMultipleItems = MAX_NUMBER_OF_MULTIPLE_ITEMS;
+ }
+
+ if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_SIGNED)
+ {
+ ctype = CT_UNITS_SIGNED;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_SIGNED);
+ controlDetails.paDetails = &valueSigned[0];
+ }
+ else if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_UNSIGNED)
+ {
+ ctype = CT_UNITS_UNSIGNED;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
+ controlDetails.paDetails = &valueUnsigned[0];
+ }
+ else if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_BOOLEAN)
+ {
+ ctype = CT_UNITS_BOOLEAN;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
+ controlDetails.paDetails = &valueBoolean[0];
+ }
+
+ // Retrieve a control's value
+ //
+ res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ if (trace)
+ {
+ UINT nItems(1);
+ nItems = (controlDetails.cMultipleItems > 0 ? controlDetails.cMultipleItems : 1);
+ for (UINT i = 0; i < nItems; i++)
+ {
+ if (ctype == CT_UNITS_SIGNED)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "signed value : %d", valueSigned[i].lValue);
+ }
+ else if (ctype == CT_UNITS_UNSIGNED)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unsigned value : %u", valueUnsigned[i].dwValue);
+ }
+ else if (ctype == CT_UNITS_BOOLEAN)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "boolean value : %u", valueBoolean[i].fValue);
+ }
+ }
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetUnsignedControlValue
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD& dwValue) const
+{
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+ MIXERCONTROLDETAILS_UNSIGNED valueUnsigned;
+
+ controlDetails.dwControlID = dwControlID;
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.cChannels = 1;
+ controlDetails.cMultipleItems = 0;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
+ controlDetails.paDetails = &valueUnsigned;
+
+ // Retrieve the unsigned value
+ //
+ res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ // Deliver the retrieved value
+ //
+ dwValue = valueUnsigned.dwValue;
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// SetUnsignedControlValue
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::SetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD dwValue) const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioMixerManager::SetUnsignedControlValue(mixId=%u, dwControlID=%d, dwValue=%d)", mixId, dwControlID, dwValue);
+
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+ MIXERCONTROLDETAILS_UNSIGNED valueUnsigned;
+
+ controlDetails.dwControlID = dwControlID;
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.cChannels = 1;
+ controlDetails.cMultipleItems = 0;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
+ controlDetails.paDetails = &valueUnsigned;
+
+ valueUnsigned.dwValue = dwValue;
+
+ // Set the unsigned value
+ //
+ res = mixerSetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerSetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// SetBooleanControlValue
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::SetBooleanControlValue(UINT mixId, DWORD dwControlID, bool value) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetBooleanControlValue(mixId=%u, dwControlID=%d, value=%d)", mixId, dwControlID, value);
+
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+ MIXERCONTROLDETAILS_BOOLEAN valueBoolean;
+
+ controlDetails.dwControlID = dwControlID;
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.cChannels = 1;
+ controlDetails.cMultipleItems = 0;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
+ controlDetails.paDetails = &valueBoolean;
+
+ if (value == true)
+ valueBoolean.fValue = TRUE;
+ else
+ valueBoolean.fValue = FALSE;
+
+ // Set the boolean value
+ //
+ res = mixerSetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerSetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetBooleanControlValue
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetBooleanControlValue(UINT mixId, DWORD dwControlID, bool& value) const
+{
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+ MIXERCONTROLDETAILS_BOOLEAN valueBoolean;
+
+ controlDetails.dwControlID = dwControlID;
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.cChannels = 1;
+ controlDetails.cMultipleItems = 0;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
+ controlDetails.paDetails = &valueBoolean;
+
+ // Retrieve the boolean value
+ //
+ res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ // Deliver the retrieved value
+ //
+ if (valueBoolean.fValue == 0)
+ value = false;
+ else
+ value = true;
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// GetSelectedMuxSource
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::GetSelectedMuxSource(UINT mixId, DWORD dwControlID, DWORD cMultipleItems, UINT& index) const
+{
+ assert(cMultipleItems <= MAX_NUMBER_OF_MULTIPLE_ITEMS);
+
+ MMRESULT res;
+ MIXERCONTROLDETAILS controlDetails;
+ MIXERCONTROLDETAILS_BOOLEAN valueBoolean[MAX_NUMBER_OF_MULTIPLE_ITEMS];
+ memset(&valueBoolean, 0, sizeof(valueBoolean));
+
+ controlDetails.dwControlID = dwControlID;
+ controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
+ controlDetails.cChannels = 1;
+ controlDetails.cMultipleItems = cMultipleItems;
+ controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
+ controlDetails.paDetails = &valueBoolean;
+
+ // Retrieve the boolean values
+ //
+ res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
+ if (res != MMSYSERR_NOERROR)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
+ return false;
+ }
+
+ // Map the current MUX setting to an index corresponding to a source index.
+ // e.g. with cMultipleItems = 3,
+ // valueBoolean[] = {1,0,0} => index = 2
+ // valueBoolean[] = {0,1,0} => index = 1
+ // valueBoolean[] = {0,0,1} => index = 0
+ //
+ // If there is no "1" in the array, we assume index should be 0.
+ index = 0;
+ for (DWORD i = 0; i < cMultipleItems; i++)
+ {
+ if (valueBoolean[i].fValue > 0)
+ {
+ index = (cMultipleItems - 1) - i;
+ break;
+ }
+ }
+
+ return true;
+}
+
+// ----------------------------------------------------------------------------
+// TraceStatusAndSupportFlags
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceStatusAndSupportFlags(DWORD fdwLine) const
+{
+ TCHAR buf[128];
+
+ StringCchPrintf(buf, 128, TEXT("status & support flags : 0x%x "), fdwLine);
+
+ switch (fdwLine)
+ {
+ case MIXERLINE_LINEF_ACTIVE:
+ StringCchCat(buf, 128, TEXT("(ACTIVE DESTINATION)"));
+ break;
+ case MIXERLINE_LINEF_DISCONNECTED:
+ StringCchCat(buf, 128, TEXT("(DISCONNECTED)"));
+ break;
+ case MIXERLINE_LINEF_SOURCE:
+ StringCchCat(buf, 128, TEXT("(INACTIVE SOURCE)"));
+ break;
+ case MIXERLINE_LINEF_SOURCE | MIXERLINE_LINEF_ACTIVE:
+ StringCchCat(buf, 128, TEXT("(ACTIVE SOURCE)"));
+ break;
+ default:
+ StringCchCat(buf, 128, TEXT("(INVALID)"));
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// TraceComponentType
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceComponentType(DWORD dwComponentType) const
+{
+ TCHAR buf[128];
+
+ StringCchPrintf(buf, 128, TEXT("component type : 0x%x "), dwComponentType);
+
+ switch (dwComponentType)
+ {
+ // Destination
+ case MIXERLINE_COMPONENTTYPE_DST_UNDEFINED:
+ StringCchCat(buf, 128, TEXT("(DST_UNDEFINED)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_DIGITAL:
+ StringCchCat(buf, 128, TEXT("(DST_DIGITAL)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_LINE:
+ StringCchCat(buf, 128, TEXT("(DST_LINE)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_MONITOR:
+ StringCchCat(buf, 128, TEXT("(DST_MONITOR)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_SPEAKERS:
+ StringCchCat(buf, 128, TEXT("(DST_SPEAKERS)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_HEADPHONES:
+ StringCchCat(buf, 128, TEXT("(DST_HEADPHONES)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_TELEPHONE:
+ StringCchCat(buf, 128, TEXT("(DST_TELEPHONE)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_WAVEIN:
+ StringCchCat(buf, 128, TEXT("(DST_WAVEIN)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_DST_VOICEIN:
+ StringCchCat(buf, 128, TEXT("(DST_VOICEIN)"));
+ break;
+ // Source
+ case MIXERLINE_COMPONENTTYPE_SRC_UNDEFINED:
+ StringCchCat(buf, 128, TEXT("(SRC_UNDEFINED)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_DIGITAL:
+ StringCchCat(buf, 128, TEXT("(SRC_DIGITAL)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_LINE:
+ StringCchCat(buf, 128, TEXT("(SRC_LINE)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE:
+ StringCchCat(buf, 128, TEXT("(SRC_MICROPHONE)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_SYNTHESIZER:
+ StringCchCat(buf, 128, TEXT("(SRC_SYNTHESIZER)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_COMPACTDISC:
+ StringCchCat(buf, 128, TEXT("(SRC_COMPACTDISC)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_TELEPHONE:
+ StringCchCat(buf, 128, TEXT("(SRC_TELEPHONE)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_PCSPEAKER:
+ StringCchCat(buf, 128, TEXT("(SRC_PCSPEAKER)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_WAVEOUT:
+ StringCchCat(buf, 128, TEXT("(SRC_WAVEOUT)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_AUXILIARY:
+ StringCchCat(buf, 128, TEXT("(SRC_AUXILIARY)"));
+ break;
+ case MIXERLINE_COMPONENTTYPE_SRC_ANALOG:
+ StringCchCat(buf, 128, TEXT("(SRC_ANALOG)"));
+ break;
+ default:
+ StringCchCat(buf, 128, TEXT("(INVALID)"));
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// TraceTargetType
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceTargetType(DWORD dwType) const
+{
+ TCHAR buf[128];
+
+ StringCchPrintf(buf, 128, TEXT("media device type : 0x%x "), dwType);
+
+ switch (dwType)
+ {
+ case MIXERLINE_TARGETTYPE_UNDEFINED:
+ StringCchCat(buf, 128, TEXT("(UNDEFINED)"));
+ break;
+ case MIXERLINE_TARGETTYPE_WAVEOUT:
+ StringCchCat(buf, 128, TEXT("(WAVEOUT)"));
+ break;
+ case MIXERLINE_TARGETTYPE_WAVEIN:
+ StringCchCat(buf, 128, TEXT("(WAVEIN)"));
+ break;
+ case MIXERLINE_TARGETTYPE_MIDIOUT:
+ StringCchCat(buf, 128, TEXT("(MIDIOUT)"));
+ break;
+ case MIXERLINE_TARGETTYPE_MIDIIN:
+ StringCchCat(buf, 128, TEXT("(MIDIIN)"));
+ break;
+ default:
+ StringCchCat(buf, 128, TEXT("(INVALID)"));
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// TraceControlType
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceControlType(DWORD dwControlType) const
+{
+ TCHAR buf[128];
+
+ // Class type classification
+ //
+ StringCchPrintf(buf, 128, TEXT("class type : 0x%x "), dwControlType);
+
+ switch (dwControlType & MIXERCONTROL_CT_CLASS_MASK)
+ {
+ case MIXERCONTROL_CT_CLASS_CUSTOM:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_CUSTOM)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_METER:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_METER)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_SWITCH:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_SWITCH)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_NUMBER:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_NUMBER)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_SLIDER:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_SLIDER)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_FADER:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_FADER)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_TIME:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_TIME)"));
+ break;
+ case MIXERCONTROL_CT_CLASS_LIST:
+ StringCchCat(buf, 128, TEXT("(CT_CLASS_LIST)"));
+ break;
+ default:
+ StringCchCat(buf, 128, TEXT("(INVALID)"));
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+
+ // Control type (for each class)
+ //
+ StringCchPrintf(buf, 128, TEXT("control type : 0x%x "), dwControlType);
+
+ switch (dwControlType)
+ {
+ case MIXERCONTROL_CONTROLTYPE_CUSTOM:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_CUSTOM)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_BOOLEANMETER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BOOLEANMETER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_SIGNEDMETER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SIGNEDMETER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_PEAKMETER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PEAKMETER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_UNSIGNEDMETER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_UNSIGNEDMETER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_BOOLEAN:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BOOLEAN)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_ONOFF:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_ONOFF)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MUTE:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MUTE)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MONO:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MONO)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_LOUDNESS:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_LOUDNESS)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_STEREOENH:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_STEREOENH)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_BASS_BOOST:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BASS_BOOST)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_BUTTON:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BUTTON)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_DECIBELS:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_DECIBELS)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_SIGNED:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SIGNED)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_UNSIGNED:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_UNSIGNED)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_PERCENT:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PERCENT)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_SLIDER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SLIDER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_PAN:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PAN)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_QSOUNDPAN:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_QSOUNDPAN)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_FADER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_FADER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_VOLUME:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_VOLUME)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_BASS:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BASS)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_TREBLE:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_TREBLE)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_EQUALIZER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_EQUALIZER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_SINGLESELECT:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SINGLESELECT)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MUX:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MUX)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MULTIPLESELECT:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MULTIPLESELECT)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MIXER:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MIXER)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MICROTIME:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MICROTIME)"));
+ break;
+ case MIXERCONTROL_CONTROLTYPE_MILLITIME:
+ StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MILLITIME)"));
+ break;
+ default:
+ StringCchCat(buf, 128, TEXT("(INVALID)"));
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// TraceControlStatusAndSupportFlags
+//
+// fdwControl
+//
+// Status and support flags for the audio line control. The following values
+// are defined:
+//
+// MIXERCONTROL_CONTROLF_DISABLED
+//
+// The control is disabled, perhaps due to other settings for the mixer hardware,
+// and cannot be used. An application can read current settings from a
+// disabled control, but it cannot apply settings.
+//
+// MIXERCONTROL_CONTROLF_MULTIPLE
+//
+// The control has two or more settings per channel. An equalizer, for example,
+// requires this flag because each frequency band can be set to a different value.
+// An equalizer that affects both channels of a stereo line in a uniform fashion
+// will also specify the MIXERCONTROL_CONTROLF_UNIFORM flag.
+//
+// MIXERCONTROL_CONTROLF_UNIFORM
+//
+// The control acts on all channels of a multichannel line in a uniform fashion.
+// For example, a control that mutes both channels of a stereo line would set
+// this flag. Most MIXERCONTROL_CONTROLTYPE_MUX and
+// MIXERCONTROL_CONTROLTYPE_MIXER controls also specify the
+// MIXERCONTROL_CONTROLF_UNIFORM flag.
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceControlStatusAndSupportFlags(DWORD fdwControl) const
+{
+ TCHAR buf[128];
+
+ StringCchPrintf(buf, 128, TEXT("control support flags : 0x%x "), fdwControl);
+
+ if (fdwControl & MIXERCONTROL_CONTROLF_DISABLED)
+ {
+ // The control is disabled, perhaps due to other settings for the mixer hardware,
+ // and cannot be used. An application can read current settings from a disabled
+ // control, but it cannot apply settings.
+ StringCchCat(buf, 128, TEXT("(CONTROLF_DISABLED)"));
+ }
+
+ if (fdwControl & MIXERCONTROL_CONTROLF_MULTIPLE)
+ {
+ // The control has two or more settings per channel. An equalizer, for example,
+ // requires this flag because each frequency band can be set to a different
+ // value. An equalizer that affects both channels of a stereo line in a
+ // uniform fashion will also specify the MIXERCONTROL_CONTROLF_UNIFORM flag.
+ StringCchCat(buf, 128, TEXT("(CONTROLF_MULTIPLE)"));
+ }
+
+ if (fdwControl & MIXERCONTROL_CONTROLF_UNIFORM)
+ {
+ // The control acts on all channels of a multichannel line in a uniform
+ // fashion. For example, a control that mutes both channels of a stereo
+ // line would set this flag. Most MIXERCONTROL_CONTROLTYPE_MUX and
+ // MIXERCONTROL_CONTROLTYPE_MIXER controls also specify the
+ // MIXERCONTROL_CONTROLF_UNIFORM flag.
+ StringCchCat(buf, 128, TEXT("(CONTROLF_UNIFORM)"));
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// ClearSpeakerState I (II)
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::ClearSpeakerState(UINT idx)
+{
+ _speakerState[idx].dwLineID = 0L;
+ _speakerState[idx].dwVolumeControlID = 0L;
+ _speakerState[idx].dwMuteControlID = 0L;
+ _speakerState[idx].speakerIsValid = false;
+ _speakerState[idx].muteControlIsValid = false;
+ _speakerState[idx].volumeControlIsValid = false;
+}
+
+// ----------------------------------------------------------------------------
+// ClearSpeakerState II (II)
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::ClearSpeakerState()
+{
+ for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
+ {
+ ClearSpeakerState(i);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// SpeakerIsValid
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::SpeakerIsValid(UINT idx) const
+{
+ return (_speakerState[idx].speakerIsValid);
+}
+
+// ----------------------------------------------------------------------------
+// ValidSpeakers
+//
+// Counts number of valid speaker destinations for all mixer devices.
+// ----------------------------------------------------------------------------
+
+UINT AudioMixerManager::ValidSpeakers() const
+{
+ UINT nSpeakers(0);
+ for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
+ {
+ if (SpeakerIsValid(i))
+ nSpeakers++;
+ }
+ return nSpeakers;
+}
+
+// ----------------------------------------------------------------------------
+// ClearMicrophoneState I (II)
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::ClearMicrophoneState(UINT idx)
+{
+ _microphoneState[idx].dwLineID = 0L;
+ _microphoneState[idx].dwVolumeControlID = 0L;
+ _microphoneState[idx].dwMuteControlID = 0L;
+ _microphoneState[idx].dwOnOffControlID = 0L;
+ _microphoneState[idx].microphoneIsValid = false;
+ _microphoneState[idx].muteControlIsValid = false;
+ _microphoneState[idx].volumeControlIsValid = false;
+ _microphoneState[idx].onOffControlIsValid = false;
+}
+
+// ----------------------------------------------------------------------------
+// ClearMicrophoneState II (II)
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::ClearMicrophoneState()
+{
+ for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
+ {
+ ClearMicrophoneState(i);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// MicrophoneIsValid
+// ----------------------------------------------------------------------------
+
+bool AudioMixerManager::MicrophoneIsValid(UINT idx) const
+{
+ return (_microphoneState[idx].microphoneIsValid);
+
+}
+
+// ----------------------------------------------------------------------------
+// ValidMicrophones
+//
+// Counts number of valid speaker destinations for all mixer devices.
+// To be valid, a speaker destination line must exist.
+// ----------------------------------------------------------------------------
+
+UINT AudioMixerManager::ValidMicrophones() const
+{
+ UINT nMicrophones(0);
+ for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
+ {
+ if (MicrophoneIsValid(i))
+ nMicrophones++;
+ }
+ return nMicrophones;
+}
+
+// ----------------------------------------------------------------------------
+// TraceWaveInError
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceWaveInError(MMRESULT error) const
+{
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR msg[MAXERRORLENGTH];
+
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ waveInGetErrorText(error, msg, MAXERRORLENGTH);
+ StringCchCat(buf, MAXERRORLENGTH, msg);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// TraceWaveOutError
+// ----------------------------------------------------------------------------
+
+void AudioMixerManager::TraceWaveOutError(MMRESULT error) const
+{
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR msg[MAXERRORLENGTH];
+
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ waveOutGetErrorText(error, msg, MAXERRORLENGTH);
+ StringCchCat(buf, MAXERRORLENGTH, msg);
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+}
+
+// ----------------------------------------------------------------------------
+// WideToUTF8
+// ----------------------------------------------------------------------------
+
+char* AudioMixerManager::WideToUTF8(const TCHAR* src) const {
+#ifdef UNICODE
+ const size_t kStrLen = sizeof(_str);
+ memset(_str, 0, kStrLen);
+ // Get required size (in bytes) to be able to complete the conversion.
+ int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
+ if (required_size <= kStrLen)
+ {
+ // Process the entire input string, including the terminating null char.
+ if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
+ memset(_str, 0, kStrLen);
+ }
+ return _str;
+#else
+ return const_cast<char*>(src);
+#endif
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h
new file mode 100644
index 0000000..da9de47
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_WIN_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_WIN_H
+
+#include "typedefs.h"
+#include "audio_device.h"
+#include "critical_section_wrapper.h"
+#include <Windows.h>
+#include <mmsystem.h>
+
+namespace webrtc {
+
+class AudioMixerManager
+{
+public:
+ enum { MAX_NUMBER_MIXER_DEVICES = 40 };
+ enum { MAX_NUMBER_OF_LINE_CONTROLS = 20 };
+ enum { MAX_NUMBER_OF_MULTIPLE_ITEMS = 20 };
+ struct SpeakerLineInfo
+ {
+ DWORD dwLineID;
+ bool speakerIsValid;
+ DWORD dwVolumeControlID;
+ bool volumeControlIsValid;
+ DWORD dwMuteControlID;
+ bool muteControlIsValid;
+ };
+ struct MicrophoneLineInfo
+ {
+ DWORD dwLineID;
+ bool microphoneIsValid;
+ DWORD dwVolumeControlID;
+ bool volumeControlIsValid;
+ DWORD dwMuteControlID;
+ bool muteControlIsValid;
+ DWORD dwOnOffControlID;
+ bool onOffControlIsValid;
+ };
+public:
+ WebRtc_Word32 EnumerateAll();
+ WebRtc_Word32 EnumerateSpeakers();
+ WebRtc_Word32 EnumerateMicrophones();
+ WebRtc_Word32 OpenSpeaker(AudioDeviceModule::WindowsDeviceType device);
+ WebRtc_Word32 OpenSpeaker(WebRtc_UWord16 index);
+ WebRtc_Word32 OpenMicrophone(AudioDeviceModule::WindowsDeviceType device);
+ WebRtc_Word32 OpenMicrophone(WebRtc_UWord16 index);
+ WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetSpeakerMute(bool enable);
+ WebRtc_Word32 SpeakerMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneMute(bool enable);
+ WebRtc_Word32 MicrophoneMute(bool& enabled) const;
+ WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneBoost(bool enable);
+ WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
+ WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
+ WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
+ WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
+ WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
+ WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
+ WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
+ WebRtc_Word32 Close();
+ WebRtc_Word32 CloseSpeaker();
+ WebRtc_Word32 CloseMicrophone();
+ bool SpeakerIsInitialized() const;
+ bool MicrophoneIsInitialized() const;
+ UINT Devices() const;
+
+private:
+ UINT DestinationLines(UINT mixId) const;
+ UINT SourceLines(UINT mixId, DWORD destId) const;
+ bool GetCapabilities(UINT mixId, MIXERCAPS& caps, bool trace = false) const;
+ bool GetDestinationLineInfo(UINT mixId, DWORD destId, MIXERLINE& line, bool trace = false) const;
+ bool GetSourceLineInfo(UINT mixId, DWORD destId, DWORD srcId, MIXERLINE& line, bool trace = false) const;
+
+ bool GetAllLineControls(UINT mixId, const MIXERLINE& line, MIXERCONTROL* controlArray, bool trace = false) const;
+ bool GetLineControl(UINT mixId, DWORD dwControlID, MIXERCONTROL& control) const;
+ bool GetControlDetails(UINT mixId, MIXERCONTROL& controlArray, bool trace = false) const;
+ bool GetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD& dwValue) const;
+ bool SetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD dwValue) const;
+ bool SetBooleanControlValue(UINT mixId, DWORD dwControlID, bool value) const;
+ bool GetBooleanControlValue(UINT mixId, DWORD dwControlID, bool& value) const;
+ bool GetSelectedMuxSource(UINT mixId, DWORD dwControlID, DWORD cMultipleItems, UINT& index) const;
+
+private:
+ void ClearSpeakerState();
+ void ClearSpeakerState(UINT idx);
+ void ClearMicrophoneState();
+ void ClearMicrophoneState(UINT idx);
+ bool SpeakerIsValid(UINT idx) const;
+ UINT ValidSpeakers() const;
+ bool MicrophoneIsValid(UINT idx) const;
+ UINT ValidMicrophones() const;
+
+ void TraceStatusAndSupportFlags(DWORD fdwLine) const;
+ void TraceTargetType(DWORD dwType) const;
+ void TraceComponentType(DWORD dwComponentType) const;
+ void TraceControlType(DWORD dwControlType) const;
+ void TraceControlStatusAndSupportFlags(DWORD fdwControl) const;
+ void TraceWaveInError(MMRESULT error) const;
+ void TraceWaveOutError(MMRESULT error) const;
+ // Converts from wide-char to UTF-8 if UNICODE is defined.
+ // Does nothing if UNICODE is undefined.
+ char* WideToUTF8(const TCHAR* src) const;
+
+public:
+ AudioMixerManager(const WebRtc_Word32 id);
+ ~AudioMixerManager();
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ HMIXER _outputMixerHandle;
+ UINT _outputMixerID;
+ HMIXER _inputMixerHandle;
+ UINT _inputMixerID;
+ SpeakerLineInfo _speakerState[MAX_NUMBER_MIXER_DEVICES];
+ MicrophoneLineInfo _microphoneState[MAX_NUMBER_MIXER_DEVICES];
+ mutable char _str[MAXERRORLENGTH];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_H
diff --git a/trunk/src/modules/audio_device/main/test/README.txt b/trunk/src/modules/audio_device/main/test/README.txt
new file mode 100644
index 0000000..7435ac5
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/README.txt
@@ -0,0 +1,23 @@
+INSTRUCTIONS:
+
+- Start with test #3 (Device enumeration) to get an overview of the available
+ audio devices.
+- Next, proceed with test #4 (Device selection) to get more details about
+ the supported functions for each audio device.
+- Verify two-way audio in test #5.
+ Repeat this test for different selections of playout and recording devices.
+- More detailed tests (volume, mute etc.) can also be performed using #6-#11.
+
+NOTE:
+
+- Some tests requires that the user opens up the audio mixer dialog and
+ verifies that a certain action (e.g. Mute ON/OFF) is executed correctly.
+- Files can be recorded during some tests to enable off-line analysis.
+- Full support of 'Default Communication' devices requires Windows 7.
+- If a test consists of several sub tests, press any key to start a new sub test.
+
+KNOWN ISSUES:
+
+- Microphone Boost control is not supported on Windows Vista or Windows 7.
+- Speaker and microphone volume controls will not work as intended on Windows
+ Vista if a 'Default Communication' device is selected in any direction.
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath
new file mode 100644
index 0000000..6e9239f
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath
@@ -0,0 +1,7 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+ <classpathentry kind="src" path="src"/>
+ <classpathentry kind="src" path="gen"/>
+ <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+ <classpathentry kind="output" path="bin"/>
+</classpath>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml
new file mode 100644
index 0000000..d8117f5
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml
@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ android:versionCode="1"
+ android:versionName="1.0" package="org.webrtc.voiceengine.test">
+ <application android:icon="@drawable/icon"
+ android:label="@string/app_name"
+ android:debuggable="true">
+ <activity android:label="@string/app_name"
+ android:name="AudioDeviceAndroidTest">
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+ </application>
+
+ <uses-sdk android:minSdkVersion="3"></uses-sdk>
+ <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS">
+ </uses-permission>
+ <uses-permission android:name="android.permission.RECORD_AUDIO">
+ </uses-permission>
+</manifest>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties
new file mode 100644
index 0000000..19ddebd
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target.
+target=android-3
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java
new file mode 100644
index 0000000..a295780
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java
@@ -0,0 +1,26 @@
+/* AUTO-GENERATED FILE. DO NOT MODIFY.
+ *
+ * This class was automatically generated by the
+ * aapt tool from the resource data it found. It
+ * should not be modified by hand.
+ */
+
+package org.webrtc.voiceengine.test;
+
+public final class R {
+ public static final class attr {
+ }
+ public static final class drawable {
+ public static final int icon=0x7f020000;
+ }
+ public static final class id {
+ public static final int Button01=0x7f050000;
+ }
+ public static final class layout {
+ public static final int main=0x7f030000;
+ }
+ public static final class string {
+ public static final int app_name=0x7f040000;
+ public static final int run_button=0x7f040001;
+ }
+}
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc
new file mode 100644
index 0000000..f46c6b2
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc
@@ -0,0 +1,108 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h> // memset
+#include <android/log.h>
+
+#include "org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h"
+
+#include "../../../../interface/audio_device.h"
+
+#define LOG_TAG "WebRTC ADM Native"
+
+void api_test();
+void func_test(int);
+
+typedef struct
+{
+ // Other
+ JavaVM* jvm;
+} AdmData;
+
+static AdmData admData;
+
+jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
+{
+ if (!vm)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, LOG_TAG,
+ "JNI_OnLoad did not receive a valid VM pointer");
+ return -1;
+ }
+
+ // Get JNI
+ JNIEnv* env;
+ if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
+ JNI_VERSION_1_4))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, LOG_TAG,
+ "JNI_OnLoad could not get JNI env");
+ return -1;
+ }
+
+ // Get class to register the native functions with
+ // jclass regClass =
+ // env->FindClass("org/webrtc/voiceengine/test/AudioDeviceAndroidTest");
+ // if (!regClass) {
+ // return -1; // Exception thrown
+ // }
+
+ // Register native functions
+ // JNINativeMethod methods[1];
+ // methods[0].name = NULL;
+ // methods[0].signature = NULL;
+ // methods[0].fnPtr = NULL;
+ // if (JNI_OK != env->RegisterNatives(regClass, methods, 1))
+ // {
+ // return -1;
+ // }
+
+ // Init VoiceEngine data
+ memset(&admData, 0, sizeof(admData));
+
+ // Store the JVM
+ admData.jvm = vm;
+
+ return JNI_VERSION_1_4;
+}
+
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_NativeInit(JNIEnv * env,
+ jclass)
+{
+ // Look up and cache any interesting class, field and method IDs for
+ // any used java class here
+
+ return true;
+}
+
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_RunTest(JNIEnv *env,
+ jobject context,
+ jint test)
+{
+ // Set instance independent Java objects
+ webrtc::AudioDeviceModule::SetAndroidObjects(admData.jvm, env, context);
+
+ // Start test
+ if (0 == test)
+ {
+ api_test();
+ }
+ else
+ {
+ func_test(test);
+ }
+
+ // Clear instance independent Java objects
+ webrtc::AudioDeviceModule::SetAndroidObjects(NULL, NULL, NULL);
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h
new file mode 100644
index 0000000..5cbc56f
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h
@@ -0,0 +1,29 @@
+/* DO NOT EDIT THIS FILE - it is machine generated */
+#include <jni.h>
+/* Header for class org_webrtc_voiceengine_test_AudioDeviceAndroidTest */
+
+#ifndef _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest
+#define _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest
+#ifdef __cplusplus
+extern "C" {
+#endif
+/*
+ * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest
+ * Method: NativeInit
+ * Signature: ()Z
+ */
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_NativeInit
+ (JNIEnv *, jclass);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest
+ * Method: RunTest
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_RunTest
+ (JNIEnv *, jobject, jint);
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png
new file mode 100644
index 0000000..a07c69f
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png
Binary files differ
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml
new file mode 100644
index 0000000..6161f1d
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml
@@ -0,0 +1,12 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+
+ <Button android:text="@string/run_button"
+ android:id="@+id/Button01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+</LinearLayout>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml
new file mode 100644
index 0000000..bbb6f51
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml
@@ -0,0 +1,6 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+
+ <string name="app_name">WebRTC Audio Device Android Test</string>
+<string name="run_button">Run Test</string>
+</resources>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java
new file mode 100644
index 0000000..4863168
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java
@@ -0,0 +1 @@
+../../../../../../../source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java
new file mode 100644
index 0000000..b87af46
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java
@@ -0,0 +1,69 @@
+package org.webrtc.voiceengine.test;
+
+import android.app.Activity;
+import android.media.AudioManager;
+import android.os.Bundle;
+import android.util.Log;
+import android.view.View;
+import android.widget.Button;
+
+public class AudioDeviceAndroidTest extends Activity {
+ private Thread _testThread;
+
+ /** Called when the activity is first created. */
+ @Override
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ setContentView(R.layout.main);
+
+ final Button buttonStart = (Button) findViewById(R.id.Button01);
+ // buttonStart.setWidth(200);
+ // button.layout(50, 50, 100, 40);
+ buttonStart.setOnClickListener(new View.OnClickListener() {
+ public void onClick(View v) {
+ _testThread = new Thread(_testProc);
+ _testThread.start();
+ }
+ });
+
+ // Suggest to use the voice call audio stream for hardware volume
+ // controls
+ setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
+
+ DoLog("Started WebRTC Android ADM Test");
+ }
+
+ private Runnable _testProc = new Runnable() {
+ public void run() {
+ // TODO(xians), choose test from GUI
+ // Select test here, 0 for API test, 1-> for Func tests
+ RunTest(5);
+ }
+ };
+
+ private void DoLog(String msg) {
+ Log.d("*WebRTC ADM*", msg);
+ }
+
+ // //////////////// Native function prototypes ////////////////////
+
+ // Init wrapper
+ private native static boolean NativeInit();
+
+ // Function used to call test
+ private native int RunTest(int testType);
+
+ // Load native library
+ static {
+ Log.d("*WebRTC ADM*", "Loading audio_device_android_test...");
+ System.loadLibrary("audio_device_android_test");
+
+ Log.d("*WebRTC ADM*", "Calling native init...");
+ if (!NativeInit()) {
+ Log.e("*WebRTC ADM*", "Native init failed");
+ throw new RuntimeException("Native init failed");
+ } else {
+ Log.d("*WebRTC ADM*", "Native init successful");
+ }
+ }
+}
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc b/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc
new file mode 100644
index 0000000..22ac91a
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc
@@ -0,0 +1,2243 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include "audio_device_test_defines.h"
+
+#include "../source/audio_device_config.h"
+#include "../source/audio_device_impl.h"
+#include "../source/audio_device_utility.h"
+
+// Helper functions
+#if defined(ANDROID)
+char filenameStr[2][256] =
+{ 0}; // Allow two buffers for those API calls taking two filenames
+int currentStr = 0;
+
+char* GetFilename(char* filename)
+{
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", filename);
+ return filenameStr[currentStr];
+}
+const char* GetFilename(const char* filename)
+{
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", filename);
+ return filenameStr[currentStr];
+}
+int GetResource(char* resource, char* dest, int destLen)
+{
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
+ strncpy(dest, filenameStr[currentStr], destLen-1);
+ return 0;
+}
+char* GetResource(char* resource)
+{
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
+ return filenameStr[currentStr];
+}
+const char* GetResource(const char* resource)
+{
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
+ return filenameStr[currentStr];
+}
+#elif !defined(MAC_IPHONE)
+char* GetFilename(char* filename)
+{
+ return filename;
+}
+const char* GetFilename(const char* filename)
+{
+ return filename;
+}
+char* GetResource(char* resource)
+{
+ return resource;
+}
+const char* GetResource(const char* resource)
+{
+ return resource;
+}
+#endif
+
+using namespace webrtc;
+
+// ----------------------------------------------------------------------------
+// AudioEventObserverAPI
+// ----------------------------------------------------------------------------
+
+class AudioEventObserverAPI: public AudioDeviceObserver
+{
+public:
+ AudioEventObserverAPI(AudioDeviceModule* audioDevice) :
+ _error(kRecordingError),
+ _warning(kRecordingWarning),
+ _audioDevice(audioDevice)
+ {
+ }
+ ;
+ ~AudioEventObserverAPI()
+ {
+ }
+ ;
+ virtual void OnErrorIsReported(const ErrorCode error)
+ {
+ TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n\n", error);
+ _error = error;
+ // TEST(_audioDevice->StopRecording() == 0);
+ // TEST(_audioDevice->StopPlayout() == 0);
+ }
+ ;
+ virtual void OnWarningIsReported(const WarningCode warning)
+ {
+ TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n\n", warning);
+ _warning = warning;
+ TEST(_audioDevice->StopRecording() == 0);
+ TEST(_audioDevice->StopPlayout() == 0);
+ }
+ ;
+public:
+ ErrorCode _error;
+ WarningCode _warning;
+private:
+ AudioDeviceModule* _audioDevice;
+};
+
+class AudioTransportAPI: public AudioTransport
+{
+public:
+ AudioTransportAPI(AudioDeviceModule* audioDevice) :
+ _audioDevice(audioDevice), _recCount(0), _playCount(0)
+ {
+ }
+ ;
+
+ ~AudioTransportAPI()
+ {
+ }
+ ;
+
+ virtual WebRtc_Word32 RecordedDataIsAvailable(
+ const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 sampleRate,
+ const WebRtc_UWord32 totalDelay,
+ const WebRtc_Word32 clockSkew,
+ const WebRtc_UWord32 currentMicLevel,
+ WebRtc_UWord32& newMicLevel)
+ {
+ _recCount++;
+ if (_recCount % 100 == 0)
+ {
+ if (nChannels == 1)
+ {
+ // mono
+ TEST_LOG("-");
+ } else if ((nChannels == 2) && (nBytesPerSample == 2))
+ {
+ // stereo but only using one channel
+ TEST_LOG("-|");
+ } else
+ {
+ // stereo
+ TEST_LOG("--");
+ }
+ }
+
+ return 0;
+ }
+
+ virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 sampleRate,
+ WebRtc_Word8* audioSamples,
+ WebRtc_UWord32& nSamplesOut)
+ {
+ _playCount++;
+ if (_playCount % 100 == 0)
+ {
+ if (nChannels == 1)
+ {
+ TEST_LOG("+");
+ } else
+ {
+ TEST_LOG("++");
+ }
+ }
+
+ nSamplesOut = 480;
+
+ return 0;
+ }
+ ;
+private:
+ AudioDeviceModule* _audioDevice;
+ WebRtc_UWord32 _recCount;
+ WebRtc_UWord32 _playCount;
+};
+
+int api_test();
+
+
+#if !defined(MAC_IPHONE) && !defined(ANDROID)
+int api_test();
+
+int main(int /*argc*/, char* /*argv*/[])
+{
+ api_test();
+}
+#endif
+
+int api_test()
+{
+ int i(0);
+
+ TEST_LOG("========================================\n");
+ TEST_LOG("API Test of the WebRtcAudioDevice Module\n");
+ TEST_LOG("========================================\n\n");
+
+ ProcessThread* processThread = ProcessThread::CreateProcessThread();
+ processThread->Start();
+
+ // =======================================================
+ // AudioDeviceModuleImpl::Create
+ //
+ // Windows:
+ // if (WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ // user can select between default (Core) or Wave
+ // else
+ // user can select between default (Wave) or Wave
+ // =======================================================
+
+ const WebRtc_Word32 myId = 444;
+ AudioDeviceModule* audioDevice(NULL);
+
+#if defined(_WIN32)
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined!\n\n");
+ // create default implementation (=Core Audio) instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+ audioDevice->AddRef();
+ TEST(audioDevice->Release() == 0);
+ // create non-default (=Wave Audio) instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
+ audioDevice->AddRef();
+ TEST(audioDevice->Release() == 0);
+ // explicitly specify usage of Core Audio (same as default)
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsCoreAudio)) != NULL);
+#else
+ TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined!\n");
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
+ // create default implementation (=Wave Audio) instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+ audioDevice->AddRef();
+ TEST(audioDevice->Release() == 0);
+ // explicitly specify usage of Wave Audio (same as default)
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
+#endif
+#endif
+
+#if defined(ANDROID)
+ // Fails tests
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
+ // Create default implementation instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+#elif defined(WEBRTC_LINUX)
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
+ // create default implementation instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+ audioDevice->AddRef();
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Release() == 0);
+ // explicitly specify usage of Pulse Audio (same as default)
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxPulseAudio)) != NULL);
+#endif
+
+#if defined(WEBRTC_MAC)
+ // Fails tests
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
+ // Create default implementation instance
+ TEST((audioDevice = AudioDeviceModuleImpl::Create(
+ myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+#endif
+
+ if (audioDevice == NULL)
+ {
+#ifdef _WIN32
+ goto Exit;
+#else
+ TEST_LOG("Failed creating audio device object! \n");
+ return 0;
+#endif
+ }
+
+ // The ADM is reference counted.
+ audioDevice->AddRef();
+
+ processThread->RegisterModule(audioDevice);
+
+ AudioDeviceModule::AudioLayer audioLayer =
+ AudioDeviceModule::kPlatformDefaultAudio;
+ TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
+ if (audioLayer == AudioDeviceModule::kLinuxAlsaAudio) {
+ TEST_LOG("API Test is not available on ALSA. \n");
+ processThread->DeRegisterModule(audioDevice);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Release() == 0);
+ return 0;
+ }
+
+ TEST_LOG("Testing...\n\n");
+
+ // =====================
+ // RegisterEventObserver
+ // =====================
+
+ AudioEventObserverAPI* eventObserver =
+ new AudioEventObserverAPI(audioDevice);
+
+ TEST(audioDevice->RegisterEventObserver(NULL) == 0);
+ TEST(audioDevice->RegisterEventObserver(eventObserver) == 0);
+ TEST(audioDevice->RegisterEventObserver(NULL) == 0);
+
+ // =====================
+ // RegisterAudioCallback
+ // =====================
+
+ AudioTransportAPI* audioTransport = new AudioTransportAPI(audioDevice);
+
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ // ====
+ // Init
+ // ====
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ // =========
+ // Terminate
+ // =========
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ // ------------------------------------------------------------------------
+ // Ensure that we keep audio device initialized for all API tests:
+ //
+ TEST(audioDevice->Init() == 0);
+ // ------------------------------------------------------------------------
+
+ // goto SHORTCUT;
+
+ WebRtc_Word16 nDevices(0);
+
+ // ==============
+ // PlayoutDevices
+ // ==============
+
+ TEST((nDevices = audioDevice->PlayoutDevices()) > 0);
+ TEST((nDevices = audioDevice->PlayoutDevices()) > 0);
+
+ // ================
+ // RecordingDevices
+ // ================
+
+ TEST((nDevices = audioDevice->RecordingDevices()) > 0);
+ TEST((nDevices = audioDevice->RecordingDevices()) > 0);
+
+ // =================
+ // PlayoutDeviceName
+ // =================
+
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize];
+ WebRtc_Word8 guid[kAdmMaxGuidSize];
+
+ nDevices = audioDevice->PlayoutDevices();
+
+ // fail tests
+ TEST(audioDevice->PlayoutDeviceName(-2, name, guid) == -1);
+ TEST(audioDevice->PlayoutDeviceName(nDevices, name, guid) == -1);
+ TEST(audioDevice->PlayoutDeviceName(0, NULL, guid) == -1);
+
+ // bulk tests
+ TEST(audioDevice->PlayoutDeviceName(0, name, NULL) == 0);
+#ifdef _WIN32
+ TEST(audioDevice->PlayoutDeviceName(-1, name, NULL) == 0); // shall be mapped to 0
+#else
+ TEST(audioDevice->PlayoutDeviceName(-1, name, NULL) == -1);
+#endif
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->PlayoutDeviceName(i, name, guid) == 0);
+ TEST(audioDevice->PlayoutDeviceName(i, name, NULL) == 0);
+ }
+
+ // ===================
+ // RecordingDeviceName
+ // ===================
+
+ nDevices = audioDevice->RecordingDevices();
+
+ // fail tests
+ TEST(audioDevice->RecordingDeviceName(-2, name, guid) == -1);
+ TEST(audioDevice->RecordingDeviceName(nDevices, name, guid) == -1);
+ TEST(audioDevice->RecordingDeviceName(0, NULL, guid) == -1);
+
+ // bulk tests
+ TEST(audioDevice->RecordingDeviceName(0, name, NULL) == 0);
+#ifdef _WIN32
+ TEST(audioDevice->RecordingDeviceName(-1, name, NULL) == 0); // shall me mapped to 0
+#else
+ TEST(audioDevice->RecordingDeviceName(-1, name, NULL) == -1);
+#endif
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->RecordingDeviceName(i, name, guid) == 0);
+ TEST(audioDevice->RecordingDeviceName(i, name, NULL) == 0);
+ }
+
+ // ================
+ // SetPlayoutDevice
+ // ================
+
+ nDevices = audioDevice->PlayoutDevices();
+
+ // fail tests
+ TEST(audioDevice->SetPlayoutDevice(-1) == -1);
+ TEST(audioDevice->SetPlayoutDevice(nDevices) == -1);
+
+ // bulk tests
+#ifdef _WIN32
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
+#else
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == -1);
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == -1);
+#endif
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ }
+
+ // ==================
+ // SetRecordingDevice
+ // ==================
+
+ nDevices = audioDevice->RecordingDevices();
+
+ // fail tests
+ TEST(audioDevice->SetRecordingDevice(-1) == -1);
+ TEST(audioDevice->SetRecordingDevice(nDevices) == -1);
+
+ // bulk tests
+#ifdef _WIN32
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
+#else
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == -1);
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultDevice) == -1);
+#endif
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ }
+
+ // ==================
+ // PlayoutIsAvailable
+ // ==================
+
+ bool available(false);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+#ifdef _WIN32
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false); // availability check should not initialize
+
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+#endif
+
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ }
+
+ // ====================
+ // RecordingIsAvailable
+ // ====================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+#ifdef _WIN32
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+#endif
+
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ }
+
+ // ===========
+ // InitPlayout
+ // ===========
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial state
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+
+ // ensure that device must be set before we can initialize
+ TEST(audioDevice->InitPlayout() == -1);
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+
+ // bulk tests
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == -1);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ }
+
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ // Sleep is needed for e.g. iPhone since we after stopping then starting may
+ // have a hangover time of a couple of ms before initialized.
+ AudioDeviceUtility::Sleep(50);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ }
+
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ }
+ }
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+
+ // =============
+ // InitRecording
+ // =============
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial state
+ TEST(audioDevice->RecordingIsInitialized() == false);
+
+ // ensure that device must be set before we can initialize
+ TEST(audioDevice->InitRecording() == -1);
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == true);
+
+ // bulk tests
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == true);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == -1);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ }
+
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ AudioDeviceUtility::Sleep(50);
+ TEST(audioDevice->RecordingIsInitialized() == true);
+ }
+
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == true);
+ }
+ }
+ }
+
+ TEST(audioDevice->StopRecording() == 0);
+
+ // ============
+ // StartPlayout
+ // StopPlayout
+ // ============
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ TEST(audioDevice->StartPlayout() == -1);
+ TEST(audioDevice->StopPlayout() == 0);
+
+#ifdef _WIN32
+ // kDefaultCommunicationDevice
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->Playing() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ }
+#endif
+
+ // repeat test but for kDefaultDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->Playing() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->Playing() == false);
+ }
+
+ // repeat test for all devices
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->Playing() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->Playing() == false);
+ }
+ }
+
+ // ==============
+ // StartRecording
+ // StopRecording
+ // ==============
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+
+ TEST(audioDevice->StartRecording() == -1);
+ TEST(audioDevice->StopRecording() == 0);
+
+#ifdef _WIN32
+ // kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ }
+#endif
+
+ // repeat test but for kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->Recording() == false);
+ }
+
+ // repeat test for all devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->Recording() == false);
+ }
+ }
+
+ WebRtc_UWord32 vol(0);
+
+#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+
+ // ================
+ // SetWaveOutVolume
+ // GetWaveOutVolume
+ // ================
+
+ // NOTE 1: Windows Wave only!
+ // NOTE 2: It seems like the waveOutSetVolume API returns
+ // MMSYSERR_NOTSUPPORTED on some Vista machines!
+
+ const WebRtc_UWord16 maxVol(0xFFFF);
+ WebRtc_UWord16 volL, volR;
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // make dummy test to see if this API is supported
+ WebRtc_Word32 works = audioDevice->SetWaveOutVolume(vol, vol);
+ WARNING(works == 0);
+
+ if (works == 0)
+ {
+ // set volume without open playout device
+ for (vol = 0; vol <= maxVol; vol += (maxVol/5))
+ {
+ TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
+ TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
+ TEST((volL==vol) && (volR==vol));
+ }
+
+ // repeat test but this time with an open (default) output device
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ for (vol = 0; vol <= maxVol; vol += (maxVol/5))
+ {
+ TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
+ TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
+ TEST((volL==vol) && (volR==vol));
+ }
+
+ // as above but while playout is active
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->Playing() == true);
+ for (vol = 0; vol <= maxVol; vol += (maxVol/5))
+ {
+ TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
+ TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
+ TEST((volL==vol) && (volR==vol));
+ }
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->Playing() == false);
+
+#endif // defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ // ==================
+ // SpeakerIsAvailable
+ // ==================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+ }
+
+ // ===========
+ // InitSpeaker
+ // ===========
+
+ // NOTE: we call Terminate followed by Init to ensure that any existing output mixer
+ // handle is set to NULL. The mixer handle is closed and reopend again for each call to
+ // SetPlayoutDevice.
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // kDefaultCommunicationDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ }
+
+ // fail tests
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->InitSpeaker() == -1);
+ TEST(audioDevice->StopPlayout() == 0);
+ }
+
+ // kDefaultDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ }
+
+ // repeat test for all devices
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ }
+ }
+
+ // =====================
+ // MicrophoneIsAvailable
+ // =====================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+ }
+
+ // ==============
+ // InitMicrophone
+ // ==============
+
+ // NOTE: we call Terminate followed by Init to ensure that any existing input mixer
+ // handle is set to NULL. The mixer handle is closed and reopend again for each call to
+ // SetRecordingDevice.
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ }
+
+ // fail tests
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->InitMicrophone() == -1);
+ TEST(audioDevice->StopRecording() == 0);
+ }
+
+ // kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ }
+
+ // repeat test for all devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ }
+ }
+
+ // ========================
+ // SpeakerVolumeIsAvailable
+ // ========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+ }
+
+ // ================
+ // SetSpeakerVolume
+ // SpeakerVolume
+ // MaxSpeakerVolume
+ // MinSpeakerVolume
+ // ================
+
+ WebRtc_UWord32 volume(0);
+ WebRtc_UWord32 maxVolume(0);
+ WebRtc_UWord32 minVolume(0);
+ WebRtc_UWord16 stepSize(0);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // fail tests
+ TEST(audioDevice->SetSpeakerVolume(0) == -1); // speaker must be initialized first
+ TEST(audioDevice->SpeakerVolume(&volume) == -1);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == -1);
+ TEST(audioDevice->MinSpeakerVolume(&minVolume) == -1);
+ TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == -1);
+
+#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ // test for warning (can e.g. happen on Vista with Wave API)
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SetSpeakerVolume(19001) == 0);
+ TEST(audioDevice->SpeakerVolume(&volume) == 0);
+ WARNING(volume == 19001);
+ }
+#endif
+
+#ifdef _WIN32
+ // use kDefaultCommunicationDevice and modify/retrieve the volume
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
+ TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
+ for (vol = minVolume; vol < (int)maxVolume; vol += 20*stepSize)
+ {
+ TEST(audioDevice->SetSpeakerVolume(vol) == 0);
+ TEST(audioDevice->SpeakerVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+#endif
+
+ // use kDefaultDevice and modify/retrieve the volume
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
+ TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
+ WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
+ step = (step < stepSize ? stepSize : step);
+ for (vol = minVolume; vol <= maxVolume; vol += step)
+ {
+ TEST(audioDevice->SetSpeakerVolume(vol) == 0);
+ TEST(audioDevice->SpeakerVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+
+ // use all (indexed) devices and modify/retrieve the volume
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
+ TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
+ WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
+ step = (step < stepSize ? stepSize : step);
+ for (vol = minVolume; vol <= maxVolume; vol += step)
+ {
+ TEST(audioDevice->SetSpeakerVolume(vol) == 0);
+ TEST(audioDevice->SpeakerVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+ }
+
+ // restore reasonable level
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetSpeakerVolume(maxVolume < 10 ?
+ maxVolume/3 : maxVolume/10) == 0);
+ }
+
+ // ======
+ // SetAGC
+ // AGC
+ // ======
+
+ // NOTE: The AGC API only enables/disables the AGC. To ensure that it will
+ // have an effect, use it in combination with MicrophoneVolumeIsAvailable.
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+ TEST(audioDevice->AGC() == false);
+
+ // set/get tests
+ TEST(audioDevice->SetAGC(true) == 0);
+ TEST(audioDevice->AGC() == true);
+ TEST(audioDevice->SetAGC(false) == 0);
+ TEST(audioDevice->AGC() == false);
+
+ // ===========================
+ // MicrophoneVolumeIsAvailable
+ // ===========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+ }
+
+ // ===================
+ // SetMicrophoneVolume
+ // MicrophoneVolume
+ // MaxMicrophoneVolume
+ // MinMicrophoneVolume
+ // ===================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // fail tests
+ TEST(audioDevice->SetMicrophoneVolume(0) == -1); // must be initialized first
+ TEST(audioDevice->MicrophoneVolume(&volume) == -1);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == -1);
+ TEST(audioDevice->MinMicrophoneVolume(&minVolume) == -1);
+ TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == -1);
+
+#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ // test for warning (can e.g. happen on Vista with Wave API)
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneVolume(19001) == 0);
+ TEST(audioDevice->MicrophoneVolume(&volume) == 0);
+ WARNING(volume == 19001);
+ }
+#endif
+
+#ifdef _WIN32
+ // initialize kDefaultCommunicationDevice and modify/retrieve the volume
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
+ TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
+ for (vol = minVolume; vol < (int)maxVolume; vol += 10*stepSize)
+ {
+ TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
+ TEST(audioDevice->MicrophoneVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+#endif
+
+ // reinitialize kDefaultDevice and modify/retrieve the volume
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
+ TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
+ for (vol = minVolume; vol < maxVolume; vol += 10 * stepSize)
+ {
+ TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
+ TEST(audioDevice->MicrophoneVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+
+ // use all (indexed) devices and modify/retrieve the volume
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
+ TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
+ for (vol = minVolume; vol < maxVolume; vol += 20 * stepSize)
+ {
+ TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
+ TEST(audioDevice->MicrophoneVolume(&volume) == 0);
+ TEST((volume == vol) || (volume == vol-1));
+ }
+ }
+ }
+
+ // restore reasonable level
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetMicrophoneVolume(maxVolume/10) == 0);
+ }
+
+ // ======================
+ // SpeakerMuteIsAvailable
+ // ======================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->PlayoutDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+ }
+
+ // =========================
+ // MicrophoneMuteIsAvailable
+ // =========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ #endif
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+ }
+
+ // ==========================
+ // MicrophoneBoostIsAvailable
+ // ==========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+#ifdef _WIN32
+ // check the kDefaultCommunicationDevice
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ // check for availability should not lead to initialization
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+#endif
+
+ // check the kDefaultDevice
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // check all availiable devices
+ nDevices = audioDevice->RecordingDevices();
+ for (i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+ }
+
+ // ==============
+ // SetSpeakerMute
+ // SpeakerMute
+ // ==============
+
+ bool enabled(false);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->SpeakerIsInitialized() == false);
+
+ // fail tests
+ TEST(audioDevice->SetSpeakerMute(true) == -1); // requires initialization
+ TEST(audioDevice->SpeakerMute(&enabled) == -1);
+
+#ifdef _WIN32
+ // initialize kDefaultCommunicationDevice and modify/retrieve the mute state
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SetSpeakerMute(true) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetSpeakerMute(false) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+#endif
+
+ // reinitialize kDefaultDevice and modify/retrieve the mute state
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SetSpeakerMute(true) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetSpeakerMute(false) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // reinitialize the default device (0) and modify/retrieve the mute state
+ TEST(audioDevice->SetPlayoutDevice(0) == 0);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SetSpeakerMute(true) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetSpeakerMute(false) == 0);
+ TEST(audioDevice->SpeakerMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // ==================
+ // SetMicrophoneMute
+ // MicrophoneMute
+ // ==================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // fail tests
+ TEST(audioDevice->SetMicrophoneMute(true) == -1); // requires initialization
+ TEST(audioDevice->MicrophoneMute(&enabled) == -1);
+
+#ifdef _WIN32
+ // initialize kDefaultCommunicationDevice and modify/retrieve the mute
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneMute(true) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneMute(false) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+#endif
+
+ // reinitialize kDefaultDevice and modify/retrieve the mute
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneMute(true) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneMute(false) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // reinitialize the default device (0) and modify/retrieve the Mute
+ TEST(audioDevice->SetRecordingDevice(0) == 0);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneMute(true) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneMute(false) == 0);
+ TEST(audioDevice->MicrophoneMute(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // ==================
+ // SetMicrophoneBoost
+ // MicrophoneBoost
+ // ==================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+ TEST(audioDevice->MicrophoneIsInitialized() == false);
+
+ // fail tests
+ TEST(audioDevice->SetMicrophoneBoost(true) == -1); // requires initialization
+ TEST(audioDevice->MicrophoneBoost(&enabled) == -1);
+
+#ifdef _WIN32
+ // initialize kDefaultCommunicationDevice and modify/retrieve the boost
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneBoost(true) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneBoost(false) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == false);
+ }
+#endif
+
+ // reinitialize kDefaultDevice and modify/retrieve the boost
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneBoost(true) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneBoost(false) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // reinitialize the default device (0) and modify/retrieve the boost
+ TEST(audioDevice->SetRecordingDevice(0) == 0);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->SetMicrophoneBoost(true) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetMicrophoneBoost(false) == 0);
+ TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // ================
+ // SetStereoPlayout
+ // StereoPlayout
+ // ================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // fail tests
+ TEST(audioDevice->InitPlayout() == -1);
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ // must be performed before initialization
+ TEST(audioDevice->SetStereoPlayout(true) == -1);
+
+ // ensure that we can set the stereo mode for playout
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+
+ // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoPlayout(false) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == false);
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ }
+
+ // initialize kDefaultDevice and modify/retrieve stereo support
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoPlayout(false) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == false);
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ }
+
+ // initialize default device (0) and modify/retrieve stereo support
+ TEST(audioDevice->SetPlayoutDevice(0) == 0);
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoPlayout(false) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == false);
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ TEST(enabled == true);
+ }
+
+ // ==================
+ // SetStereoRecording
+ // StereoRecording
+ // ==================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // fail tests
+ TEST(audioDevice->InitRecording() == -1);
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == true);
+ // must be performed before initialization
+ TEST(audioDevice->SetStereoRecording(true) == -1);
+
+ // ensures that we can set the stereo mode for recording
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+
+ // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoRecording(false) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // initialize kDefaultDevice and modify/retrieve stereo support
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoRecording(false) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // initialize default device (0) and modify/retrieve stereo support
+ TEST(audioDevice->SetRecordingDevice(0) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == true);
+ TEST(audioDevice->SetStereoRecording(false) == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ TEST(enabled == false);
+ }
+
+ // ===================
+ // SetRecordingChannel
+ // RecordingChannel
+ // ==================
+
+ // the user in Win Core Audio
+
+ AudioDeviceModule::ChannelType channelType(AudioDeviceModule::kChannelBoth);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // fail tests
+ TEST(audioDevice->SetStereoRecording(false) == 0);
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == -1);
+
+ // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == 0);
+ TEST(audioDevice->RecordingChannel(&channelType) == 0);
+ TEST(channelType == AudioDeviceModule::kChannelBoth);
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
+ TEST(audioDevice->RecordingChannel(&channelType) == 0);
+ TEST(channelType == AudioDeviceModule::kChannelLeft);
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
+ TEST(audioDevice->RecordingChannel(&channelType) == 0);
+ TEST(channelType == AudioDeviceModule::kChannelRight);
+ }
+
+ // ================
+ // SetPlayoutBuffer
+ // PlayoutBuffer
+ // ================
+
+ AudioDeviceModule::BufferType bufferType;
+ WebRtc_UWord16 sizeMS(0);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+#if defined(_WIN32) || defined(ANDROID) || defined(MAC_IPHONE)
+ TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
+#else
+ TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
+#endif
+
+ // fail tests
+ TEST(audioDevice->InitPlayout() == -1); // must set device first
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutIsInitialized() == true);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
+ 100) == -1);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ kAdmMinPlayoutBufferSizeMs-1) == -1);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ kAdmMaxPlayoutBufferSizeMs+1) == -1);
+
+ // bulk tests (all should be successful)
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+#ifdef _WIN32
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
+ 0) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+ TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
+ 10000) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+ TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
+#endif
+#if defined(ANDROID) || defined(MAC_IPHONE)
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ kAdmMinPlayoutBufferSizeMs) == -1);
+#else
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ kAdmMinPlayoutBufferSizeMs) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+ TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
+ TEST(sizeMS == kAdmMinPlayoutBufferSizeMs);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ kAdmMaxPlayoutBufferSizeMs) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+ TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
+ TEST(sizeMS == kAdmMaxPlayoutBufferSizeMs);
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
+ 100) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+ TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
+ TEST(sizeMS == 100);
+#endif
+
+#ifdef _WIN32
+ // restore default
+ TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
+ 0) == 0);
+ TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
+#endif
+
+ // ============
+ // PlayoutDelay
+ // ============
+
+ // NOTE: this API is better tested in a functional test
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // bulk tests
+ TEST(audioDevice->PlayoutDelay(&sizeMS) == 0);
+ TEST(audioDevice->PlayoutDelay(&sizeMS) == 0);
+
+ // ==============
+ // RecordingDelay
+ // ==============
+
+ // NOTE: this API is better tested in a functional test
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+
+ // bulk tests
+ TEST(audioDevice->RecordingDelay(&sizeMS) == 0);
+ TEST(audioDevice->RecordingDelay(&sizeMS) == 0);
+
+ // =======
+ // CPULoad
+ // =======
+
+ // NOTE: this API is better tested in a functional test
+
+ WebRtc_UWord16 load(0);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+
+ // bulk tests
+#ifdef _WIN32
+ TEST(audioDevice->CPULoad(&load) == 0);
+ TEST(load == 0);
+#else
+ TEST(audioDevice->CPULoad(&load) == -1);
+#endif
+
+ // ===========================
+ // StartRawOutputFileRecording
+ // StopRawOutputFileRecording
+ // ===========================
+
+ // NOTE: this API is better tested in a functional test
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // fail tests
+ TEST(audioDevice->StartRawOutputFileRecording(NULL) == -1);
+
+ // bulk tests
+ TEST(audioDevice->StartRawOutputFileRecording(
+ GetFilename("raw_output_not_playing.pcm")) == 0);
+ TEST(audioDevice->StopRawOutputFileRecording() == 0);
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->StartRawOutputFileRecording(
+ GetFilename("raw_output_playing.pcm")) == 0);
+ AudioDeviceUtility::Sleep(100);
+ TEST(audioDevice->StopRawOutputFileRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->StartRawOutputFileRecording(
+ GetFilename("raw_output_not_playing.pcm")) == 0);
+ TEST(audioDevice->StopRawOutputFileRecording() == 0);
+
+ // results after this test:
+ //
+ // - size of raw_output_not_playing.pcm shall be 0
+ // - size of raw_output_playing.pcm shall be > 0
+
+ // ==========================
+ // StartRawInputFileRecording
+ // StopRawInputFileRecording
+ // ==========================
+
+ // NOTE: this API is better tested in a functional test
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ // fail tests
+ TEST(audioDevice->StartRawInputFileRecording(NULL) == -1);
+
+ // bulk tests
+ TEST(audioDevice->StartRawInputFileRecording(
+ GetFilename("raw_input_not_recording.pcm")) == 0);
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->StartRawInputFileRecording(
+ GetFilename("raw_input_recording.pcm")) == 0);
+ AudioDeviceUtility::Sleep(100);
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StartRawInputFileRecording(
+ GetFilename("raw_input_not_recording.pcm")) == 0);
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+
+ // results after this test:
+ //
+ // - size of raw_input_not_recording.pcm shall be 0
+ // - size of raw_input_not_recording.pcm shall be > 0
+
+ // ===================
+ // RecordingSampleRate
+ // ===================
+
+ WebRtc_UWord32 sampleRate(0);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+
+ // bulk tests
+ TEST(audioDevice->RecordingSampleRate(&sampleRate) == 0);
+#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ TEST(sampleRate == 48000);
+#elif defined(ANDROID)
+ TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
+ TEST((sampleRate == 44000) || (sampleRate == 16000));
+#elif defined(MAC_IPHONE)
+ TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
+ TEST((sampleRate == 44000) || (sampleRate == 16000) || (sampleRate == 8000));
+#endif
+
+ // @TODO(xians) - add tests for all platforms here...
+
+ // =================
+ // PlayoutSampleRate
+ // =================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+
+ // bulk tests
+ TEST(audioDevice->PlayoutSampleRate(&sampleRate) == 0);
+#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+ TEST(sampleRate == 48000);
+#elif defined(ANDROID)
+ TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
+ TEST((sampleRate == 44000) || (sampleRate == 16000));
+#elif defined(MAC_IPHONE)
+ TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
+ TEST((sampleRate == 44000) || (sampleRate == 16000) || (sampleRate == 8000));
+#endif
+
+ // ==========================
+ // ResetAudioDevice
+ // ==========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+ TEST(audioDevice->RecordingIsInitialized() == false);
+ TEST(audioDevice->Recording() == false);
+
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+ TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
+
+#if defined(MAC_IPHONE)
+ // Not playing or recording, should just return 0
+ TEST(audioDevice->ResetAudioDevice() == 0);
+
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ for (int l=0; l<20; ++l)
+ {
+ TEST_LOG("Resetting sound device several time with pause %d ms\n", l);
+ TEST(audioDevice->ResetAudioDevice() == 0);
+ AudioDeviceUtility::Sleep(l);
+ }
+#else
+ // Fail tests
+ TEST(audioDevice->ResetAudioDevice() == -1);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->ResetAudioDevice() == -1);
+#endif
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+
+ // ==========================
+ // SetPlayoutSpeaker
+ // ==========================
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Init() == 0);
+
+ // check initial states
+ TEST(audioDevice->Initialized() == true);
+ TEST(audioDevice->PlayoutIsInitialized() == false);
+ TEST(audioDevice->Playing() == false);
+
+ TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
+
+ bool loudspeakerOn(false);
+#if defined(MAC_IPHONE)
+ // Not playing or recording, should just return a success
+ TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == true);
+ TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == false);
+
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == true);
+ TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == false);
+
+#else
+ // Fail tests
+ TEST(audioDevice->SetLoudspeakerStatus(true) == -1);
+ TEST(audioDevice->SetLoudspeakerStatus(false) == -1);
+ TEST(audioDevice->SetLoudspeakerStatus(true) == -1);
+ TEST(audioDevice->SetLoudspeakerStatus(false) == -1);
+
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ TEST(audioDevice->GetLoudspeakerStatus(&loudspeakerOn) == -1);
+#endif
+ TEST(audioDevice->StopPlayout() == 0);
+
+#ifdef _WIN32
+ Exit:
+#endif
+
+ // ------------------------------------------------------------------------
+ // Terminate the module when all tests are done:
+ //
+ TEST(audioDevice->Terminate() == 0);
+ // ------------------------------------------------------------------------
+
+ // ===================================================
+ // AudioDeviceModuleImpl::Destroy
+ // ===================================================
+
+
+ // release the ProcessThread object
+ if (processThread)
+ {
+ processThread->DeRegisterModule(audioDevice);
+ processThread->Stop();
+ ProcessThread::DestroyProcessThread(processThread);
+ }
+
+ // delete the event observer
+ if (eventObserver)
+ {
+ delete eventObserver;
+ eventObserver = NULL;
+ }
+
+ // delete the audio transport
+ if (audioTransport)
+ {
+ delete audioTransport;
+ audioTransport = NULL;
+ }
+
+ // release the AudioDeviceModule object
+ if (audioDevice) {
+ TEST(audioDevice->Release() == 0);
+ }
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h b/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h
new file mode 100644
index 0000000..5c5f4e2
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h
@@ -0,0 +1,116 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
+
+#include "common_types.h"
+#include "audio_device.h"
+#include "process_thread.h"
+#include "trace.h"
+
+#ifdef _WIN32
+#define MACRO_DEFAULT_DEVICE AudioDeviceModule::kDefaultDevice
+#define MACRO_DEFAULT_COMMUNICATION_DEVICE AudioDeviceModule::kDefaultCommunicationDevice
+#else
+#define MACRO_DEFAULT_DEVICE 0
+#define MACRO_DEFAULT_COMMUNICATION_DEVICE 0
+#endif
+
+#ifdef ANDROID
+#include <android/log.h>
+#define LOG_TAG "WebRtc ADM TEST"
+#define TEST_LOG(...) __android_log_print(ANDROID_LOG_DEBUG, LOG_TAG, __VA_ARGS__)
+#define TEST_LOG_ERROR(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#else
+#define TEST_LOG printf
+#define TEST_LOG_ERROR(...) fprintf(stderr, __VA_ARGS__)
+#endif
+
+static int testCount = 0;
+static int errorCount = 0;
+static int warningCount = 0;
+
+#define RESET_TEST \
+ do { \
+ testCount = 0; \
+ errorCount = 0; \
+ warningCount = 0; \
+ } while(0) \
+
+#define PRINT_ERR_MSG(msg) \
+ do { \
+ TEST_LOG_ERROR("Error at line %i of %s\n%s", \
+ __LINE__, __FILE__, msg); \
+ } while(0)
+
+#define WARNING(expr) \
+ do { \
+ testCount++; \
+ if (!(expr)) { \
+ TEST_LOG_ERROR("WARNING #%d: at line %i\n\n", \
+ warningCount+1, __LINE__); \
+ warningCount++; \
+ } \
+ } while(0)
+
+
+#define TEST(expr) \
+ do { \
+ testCount++; \
+ if (!(expr)) { \
+ PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
+ errorCount++; \
+ } \
+ } while(0)
+
+#define TEST_ERR(expr, err) \
+ do { \
+ testCount++; \
+ if (!(expr)) { \
+ PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
+ errorCount++; \
+ } \
+ if (audioDevice->LastError() != err) { \
+ PRINT_ERR_MSG("Assertion failed: " #err "\n\n"); \
+ errorCount++; \
+ } \
+ } while(0)
+
+
+#define PRINT_TEST_RESULTS \
+ do { \
+ TEST_LOG("\n>> %i tests completed <<\n", testCount); \
+ if (errorCount > 0) { \
+ TEST_LOG(">> %i FAILED! <<\n\n", errorCount); \
+ } \
+ else if (warningCount > 0) \
+ { \
+ TEST_LOG(">> ALL PASSED (with %d warnings) <<\n\n", \
+ warningCount); \
+ } \
+ else \
+ { \
+ TEST_LOG(">> ALL PASSED <<\n\n"); \
+ } \
+ } while(0)
+
+// Helper functions
+// For iPhone, they are defined in iPhone specific test code.
+// For Android, they are defined in API test only (since both
+// API and Func tests are built into the same lib).
+// For other, they are defined in both API test and Func test.
+char* GetFilename(char* filename);
+const char* GetFilename(const char* filename);
+char* GetResource(char* resource);
+const char* GetResource(const char* resource);
+
+#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
+
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc b/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc
new file mode 100644
index 0000000..87a70aa
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include "audio_device_test_defines.h"
+#include "func_test_manager.h"
+
+#ifndef __GNUC__
+// Disable warning message 4996 ('scanf': This function or variable may be unsafe)
+#pragma warning( disable : 4996 )
+#endif
+
+using namespace webrtc;
+
+int func_test(int);
+
+// ----------------------------------------------------------------------------
+// main()
+// ----------------------------------------------------------------------------
+
+#if !defined(MAC_IPHONE) && !defined(ANDROID)
+int main(int /*argc*/, char* /*argv*/[])
+{
+ func_test(0);
+}
+#endif
+
+// ----------------------------------------------------------------------------
+// func_test()
+// ----------------------------------------------------------------------------
+
+int func_test(int sel)
+{
+ TEST_LOG("=========================================\n");
+ TEST_LOG("Func Test of the WebRtcAudioDevice Module\n");
+ TEST_LOG("=========================================\n\n");
+
+ // Initialize the counters here to get rid of "unused variables" warnings.
+ testCount = 0;
+ errorCount = 0;
+ warningCount = 0;
+
+ FuncTestManager funcMgr;
+
+ funcMgr.Init();
+
+ bool quit(false);
+
+ while (!quit)
+ {
+ TEST_LOG("---------------------------------------\n");
+ TEST_LOG("Select type of test\n\n");
+ TEST_LOG(" (0) Quit\n");
+ TEST_LOG(" (1) All\n");
+ TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
+ TEST_LOG(" (2) Audio-layer selection\n");
+ TEST_LOG(" (3) Device enumeration\n");
+ TEST_LOG(" (4) Device selection\n");
+ TEST_LOG(" (5) Audio transport\n");
+ TEST_LOG(" (6) Speaker volume\n");
+ TEST_LOG(" (7) Microphone volume\n");
+ TEST_LOG(" (8) Speaker mute\n");
+ TEST_LOG(" (9) Microphone mute\n");
+ TEST_LOG(" (10) Microphone boost\n");
+ TEST_LOG(" (11) Microphone AGC\n");
+ TEST_LOG(" (12) Loopback measurements\n");
+ TEST_LOG(" (13) Device removal\n");
+ TEST_LOG(" (14) Advanced mobile device API\n");
+ TEST_LOG(" (66) XTEST\n");
+ TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
+ TEST_LOG("\n: ");
+
+ int selection(0);
+ enum TestType testType(TTInvalid);
+
+SHOW_MENU:
+
+ if (sel > 0)
+ {
+ selection = sel;
+ }
+ else
+ {
+ if (scanf("%d", &selection) < 0) {
+ perror("Failed to get selection.");
+ }
+ }
+
+ switch (selection)
+ {
+ case 0:
+ quit = true;
+ break;
+ case 1:
+ testType = TTAll;
+ break;
+ case 2:
+ testType = TTAudioLayerSelection;
+ break;
+ case 3:
+ testType = TTDeviceEnumeration;
+ break;
+ case 4:
+ testType = TTDeviceSelection;
+ break;
+ case 5:
+ testType = TTAudioTransport;
+ break;
+ case 6:
+ testType = TTSpeakerVolume;
+ break;
+ case 7:
+ testType = TTMicrophoneVolume;
+ break;
+ case 8:
+ testType = TTSpeakerMute;
+ break;
+ case 9:
+ testType = TTMicrophoneMute;
+ break;
+ case 10:
+ testType = TTMicrophoneBoost;
+ break;
+ case 11:
+ testType = TTMicrophoneAGC;
+ break;
+ case 12:
+ testType = TTLoopback;
+ break;
+ case 13:
+ testType = TTDeviceRemoval;
+ break;
+ case 14:
+ testType = TTMobileAPI;
+ break;
+ case 66:
+ testType = TTTest;
+ break;
+ default:
+ testType = TTInvalid;
+ TEST_LOG(": ");
+ goto SHOW_MENU;
+ break;
+ }
+
+ funcMgr.DoTest(testType);
+
+ if (sel > 0)
+ {
+ quit = true;
+ }
+ }
+
+ funcMgr.Close();
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_device/main/test/func_test_manager.cc b/trunk/src/modules/audio_device/main/test/func_test_manager.cc
new file mode 100644
index 0000000..8bc7646
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/func_test_manager.cc
@@ -0,0 +1,2745 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <ctype.h>
+#include <cassert>
+#include <string.h>
+
+#include "func_test_manager.h"
+#include "testsupport/fileutils.h"
+
+#include "../source/audio_device_config.h"
+#include "../source/audio_device_impl.h"
+
+#ifndef __GNUC__
+// Disable warning message ('sprintf': name was marked as #pragma deprecated)
+#pragma warning( disable : 4995 )
+// Disable warning message 4996 ('scanf': This function or variable may be unsafe)
+#pragma warning( disable : 4996 )
+#endif
+
+const char* RecordedMicrophoneFile = "recorded_microphone_mono_48.pcm";
+const char* RecordedMicrophoneVolumeFile =
+"recorded_microphone_volume_mono_48.pcm";
+const char* RecordedMicrophoneMuteFile = "recorded_microphone_mute_mono_48.pcm";
+const char* RecordedMicrophoneBoostFile =
+"recorded_microphone_boost_mono_48.pcm";
+const char* RecordedMicrophoneAGCFile = "recorded_microphone_AGC_mono_48.pcm";
+const char* RecordedSpeakerFile = "recorded_speaker_48.pcm";
+
+struct AudioPacket
+{
+ WebRtc_UWord8 dataBuffer[4 * 960];
+ WebRtc_UWord16 nSamples;
+ WebRtc_UWord16 nBytesPerSample;
+ WebRtc_UWord8 nChannels;
+ WebRtc_UWord32 samplesPerSec;
+};
+
+// Helper functions
+#if !defined(MAC_IPHONE) && !defined(ANDROID)
+char* GetFilename(char* filename)
+{
+ return filename;
+}
+const char* GetFilename(const char* filename)
+{
+ return filename;
+}
+char* GetResource(char* resource)
+{
+ return resource;
+}
+const char* GetResource(const char* resource)
+{
+ return resource;
+}
+#endif
+
+namespace webrtc
+{
+
+AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) :
+ _audioDevice(audioDevice)
+{
+}
+;
+
+AudioEventObserver::~AudioEventObserver()
+{
+}
+;
+
+void AudioEventObserver::OnErrorIsReported(const ErrorCode error)
+{
+ TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n \n", error);
+ _error = error;
+ // TEST(_audioDevice->StopRecording() == 0);
+ // TEST(_audioDevice->StopPlayout() == 0);
+}
+;
+
+
+void AudioEventObserver::OnWarningIsReported(const WarningCode warning)
+{
+ TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n \n", warning);
+ _warning = warning;
+ //TEST(_audioDevice->StopRecording() == 0);
+ //TEST(_audioDevice->StopPlayout() == 0);
+}
+;
+
+AudioTransportImpl::AudioTransportImpl(AudioDeviceModule* audioDevice) :
+ _audioDevice(audioDevice),
+ _playFromFile(false),
+ _fullDuplex(false),
+ _speakerVolume(false),
+ _speakerMute(false),
+ _microphoneVolume(false),
+ _microphoneMute(false),
+ _microphoneBoost(false),
+ _microphoneAGC(false),
+ _loopBackMeasurements(false),
+ _playFile(*FileWrapper::Create()),
+ _recCount(0),
+ _playCount(0),
+ _audioList()
+{
+ _resampler.Reset(48000, 48000, kResamplerSynchronousStereo);
+}
+;
+
+AudioTransportImpl::~AudioTransportImpl()
+{
+ _playFile.Flush();
+ _playFile.CloseFile();
+ delete &_playFile;
+
+ while (!_audioList.Empty())
+ {
+ ListItem* item = _audioList.First();
+ if (item)
+ {
+ AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
+ if (packet)
+ {
+ delete packet;
+ }
+ }
+ _audioList.PopFront();
+ }
+}
+;
+
+// ----------------------------------------------------------------------------
+// AudioTransportImpl::SetFilePlayout
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32 AudioTransportImpl::SetFilePlayout(bool enable,
+ const WebRtc_Word8* fileName)
+{
+ _playFromFile = enable;
+ if (enable)
+ {
+ return (_playFile.OpenFile(fileName, true, true, false));
+ } else
+ {
+ _playFile.Flush();
+ return (_playFile.CloseFile());
+ }
+}
+;
+
+void AudioTransportImpl::SetFullDuplex(bool enable)
+{
+ _fullDuplex = enable;
+
+ while (!_audioList.Empty())
+ {
+ ListItem* item = _audioList.First();
+ if (item)
+ {
+ AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
+ if (packet)
+ {
+ delete packet;
+ }
+ }
+ _audioList.PopFront();
+ }
+}
+;
+
+WebRtc_Word32 AudioTransportImpl::RecordedDataIsAvailable(
+ const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord32 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord32 currentMicLevel,
+ WebRtc_UWord32& newMicLevel)
+{
+ if (_fullDuplex && _audioList.GetSize() < 15)
+ {
+ AudioPacket* packet = new AudioPacket();
+ memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
+ packet->nSamples = (WebRtc_UWord16) nSamples;
+ packet->nBytesPerSample = nBytesPerSample;
+ packet->nChannels = nChannels;
+ packet->samplesPerSec = samplesPerSec;
+ _audioList.PushBack(packet);
+ }
+
+ _recCount++;
+ if (_recCount % 100 == 0)
+ {
+ bool addMarker(true);
+
+ if (_loopBackMeasurements)
+ {
+ addMarker = false;
+ }
+
+ if (_microphoneVolume)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ WebRtc_UWord32 minVolume(0);
+ WebRtc_UWord32 volume(0);
+ WebRtc_UWord16 stepSize(0);
+ TEST(_audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(_audioDevice->MinMicrophoneVolume(&minVolume) == 0);
+ TEST(_audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
+ TEST(_audioDevice->MicrophoneVolume(&volume) == 0);
+ if (volume == 0)
+ {
+ TEST_LOG("[0]");
+ addMarker = false;
+ }
+ int stepScale = (int) ((maxVolume - minVolume) / (stepSize * 10));
+ volume += (stepScale * stepSize);
+ if (volume > maxVolume)
+ {
+ TEST_LOG("[MAX]");
+ volume = 0;
+ addMarker = false;
+ }
+ TEST(_audioDevice->SetMicrophoneVolume(volume) == 0);
+ }
+
+ if (_microphoneAGC)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ WebRtc_UWord32 minVolume(0);
+ WebRtc_UWord16 stepSize(0);
+ TEST(_audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(_audioDevice->MinMicrophoneVolume(&minVolume) == 0);
+ TEST(_audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
+ // emulate real AGC (min->max->min->max etc.)
+ if (currentMicLevel <= 1)
+ {
+ TEST_LOG("[MIN]");
+ addMarker = false;
+ }
+ int stepScale = (int) ((maxVolume - minVolume) / (stepSize * 10));
+ newMicLevel = currentMicLevel + (stepScale * stepSize);
+ if (newMicLevel > maxVolume)
+ {
+ TEST_LOG("[MAX]");
+ newMicLevel = 1; // set lowest (non-zero) AGC level
+ addMarker = false;
+ }
+ }
+
+ if (_microphoneMute && (_recCount % 500 == 0))
+ {
+ bool muted(false);
+ TEST(_audioDevice->MicrophoneMute(&muted) == 0);
+ muted = !muted;
+ TEST(_audioDevice->SetMicrophoneMute(muted) == 0);
+ if (muted)
+ {
+ TEST_LOG("[MUTE ON]");
+ addMarker = false;
+ } else
+ {
+ TEST_LOG("[MUTE OFF]");
+ addMarker = false;
+ }
+ }
+
+ if (_microphoneBoost && (_recCount % 500 == 0))
+ {
+ bool boosted(false);
+ TEST(_audioDevice->MicrophoneBoost(&boosted) == 0);
+ boosted = !boosted;
+ TEST(_audioDevice->SetMicrophoneBoost(boosted) == 0);
+ if (boosted)
+ {
+ TEST_LOG("[BOOST ON]");
+ addMarker = false;
+ } else
+ {
+ TEST_LOG("[BOOST OFF]");
+ addMarker = false;
+ }
+ }
+
+ if ((nChannels == 1) && addMarker)
+ {
+ // mono
+ TEST_LOG("-");
+ } else if ((nChannels == 2) && (nBytesPerSample == 2) && addMarker)
+ {
+ AudioDeviceModule::ChannelType
+ chType(AudioDeviceModule::kChannelLeft);
+ TEST(_audioDevice->RecordingChannel(&chType) == 0);
+ if (chType == AudioDeviceModule::kChannelLeft)
+ TEST_LOG("-|");
+ else
+ TEST_LOG("|-");
+ } else if (addMarker)
+ {
+ // stereo
+ TEST_LOG("--");
+ }
+
+ if (nChannels == 2 && nBytesPerSample == 2)
+ {
+ // TEST_LOG("=> emulated mono (one channel exctracted from stereo input)\n");
+ }
+ }
+
+ return 0;
+}
+
+
+WebRtc_Word32 AudioTransportImpl::NeedMorePlayData(
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ WebRtc_Word8* audioSamples,
+ WebRtc_UWord32& nSamplesOut)
+{
+ if (_fullDuplex)
+ {
+ if (_audioList.Empty())
+ {
+ // use zero stuffing when not enough data
+ memset(audioSamples, 0, nBytesPerSample * nSamples);
+ } else
+ {
+ ListItem* item = _audioList.First();
+ AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
+ if (packet)
+ {
+ int ret(0);
+ int lenOut(0);
+ WebRtc_Word16 tmpBuf_96kHz[80 * 12];
+ WebRtc_Word16* ptr16In = NULL;
+ WebRtc_Word16* ptr16Out = NULL;
+
+ const WebRtc_UWord16 nSamplesIn = packet->nSamples;
+ const WebRtc_UWord8 nChannelsIn = packet->nChannels;
+ const WebRtc_UWord32 samplesPerSecIn = packet->samplesPerSec;
+ const WebRtc_UWord16 nBytesPerSampleIn =
+ packet->nBytesPerSample;
+
+ WebRtc_Word32 fsInHz(samplesPerSecIn);
+ WebRtc_Word32 fsOutHz(samplesPerSec);
+
+ if (fsInHz == 44100)
+ fsInHz = 44000;
+
+ if (fsOutHz == 44100)
+ fsOutHz = 44000;
+
+ if (nChannelsIn == 2 && nBytesPerSampleIn == 4)
+ {
+ // input is stereo => we will resample in stereo
+ ret = _resampler.ResetIfNeeded(fsInHz, fsOutHz,
+ kResamplerSynchronousStereo);
+ if (ret == 0)
+ {
+ if (nChannels == 2)
+ {
+ _resampler.Push(
+ (const WebRtc_Word16*) packet->dataBuffer,
+ 2 * nSamplesIn,
+ (WebRtc_Word16*) audioSamples, 2
+ * nSamples, lenOut);
+ } else
+ {
+ _resampler.Push(
+ (const WebRtc_Word16*) packet->dataBuffer,
+ 2 * nSamplesIn, tmpBuf_96kHz, 2
+ * nSamples, lenOut);
+
+ ptr16In = &tmpBuf_96kHz[0];
+ ptr16Out = (WebRtc_Word16*) audioSamples;
+
+ // do stereo -> mono
+ for (unsigned int i = 0; i < nSamples; i++)
+ {
+ *ptr16Out = *ptr16In; // use left channel
+ ptr16Out++;
+ ptr16In++;
+ ptr16In++;
+ }
+ }
+ assert(2*nSamples == (WebRtc_UWord32)lenOut);
+ } else
+ {
+ if (_playCount % 100 == 0)
+ TEST_LOG(
+ "ERROR: unable to resample from %d to %d\n",
+ samplesPerSecIn, samplesPerSec);
+ }
+ } else
+ {
+ // input is mono (can be "reduced from stereo" as well) =>
+ // we will resample in mono
+ ret = _resampler.ResetIfNeeded(fsInHz, fsOutHz,
+ kResamplerSynchronous);
+ if (ret == 0)
+ {
+ if (nChannels == 1)
+ {
+ _resampler.Push(
+ (const WebRtc_Word16*) packet->dataBuffer,
+ nSamplesIn,
+ (WebRtc_Word16*) audioSamples,
+ nSamples, lenOut);
+ } else
+ {
+ _resampler.Push(
+ (const WebRtc_Word16*) packet->dataBuffer,
+ nSamplesIn, tmpBuf_96kHz, nSamples,
+ lenOut);
+
+ ptr16In = &tmpBuf_96kHz[0];
+ ptr16Out = (WebRtc_Word16*) audioSamples;
+
+ // do mono -> stereo
+ for (unsigned int i = 0; i < nSamples; i++)
+ {
+ *ptr16Out = *ptr16In; // left
+ ptr16Out++;
+ *ptr16Out = *ptr16In; // right (same as left sample)
+ ptr16Out++;
+ ptr16In++;
+ }
+ }
+ assert(nSamples == (WebRtc_UWord32)lenOut);
+ } else
+ {
+ if (_playCount % 100 == 0)
+ TEST_LOG("ERROR: unable to resample from %d to %d\n",
+ samplesPerSecIn, samplesPerSec);
+ }
+ }
+ nSamplesOut = nSamples;
+ delete packet;
+ }
+ _audioList.PopFront();
+ }
+ } // if (_fullDuplex)
+
+ if (_playFromFile && _playFile.Open())
+ {
+ WebRtc_Word16 fileBuf[480];
+
+ // read mono-file
+ WebRtc_Word32 len = _playFile.Read((WebRtc_Word8*) fileBuf, 2
+ * nSamples);
+ if (len != 2 * (WebRtc_Word32) nSamples)
+ {
+ _playFile.Rewind();
+ _playFile.Read((WebRtc_Word8*) fileBuf, 2 * nSamples);
+ }
+
+ // convert to stero if required
+ if (nChannels == 1)
+ {
+ memcpy(audioSamples, fileBuf, 2 * nSamples);
+ } else
+ {
+ // mono sample from file is duplicated and sent to left and right
+ // channels
+ WebRtc_Word16* audio16 = (WebRtc_Word16*) audioSamples;
+ for (unsigned int i = 0; i < nSamples; i++)
+ {
+ (*audio16) = fileBuf[i]; // left
+ audio16++;
+ (*audio16) = fileBuf[i]; // right
+ audio16++;
+ }
+ }
+ } // if (_playFromFile && _playFile.Open())
+
+ _playCount++;
+
+ if (_playCount % 100 == 0)
+ {
+ bool addMarker(true);
+
+ if (_speakerVolume)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ WebRtc_UWord32 minVolume(0);
+ WebRtc_UWord32 volume(0);
+ WebRtc_UWord16 stepSize(0);
+ TEST(_audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(_audioDevice->MinSpeakerVolume(&minVolume) == 0);
+ TEST(_audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
+ TEST(_audioDevice->SpeakerVolume(&volume) == 0);
+ if (volume == 0)
+ {
+ TEST_LOG("[0]");
+ addMarker = false;
+ }
+ WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
+ step = (step < stepSize ? stepSize : step);
+ volume += step;
+ if (volume > maxVolume)
+ {
+ TEST_LOG("[MAX]");
+ volume = 0;
+ addMarker = false;
+ }
+ TEST(_audioDevice->SetSpeakerVolume(volume) == 0);
+ }
+
+ if (_speakerMute && (_playCount % 500 == 0))
+ {
+ bool muted(false);
+ TEST(_audioDevice->SpeakerMute(&muted) == 0);
+ muted = !muted;
+ TEST(_audioDevice->SetSpeakerMute(muted) == 0);
+ if (muted)
+ {
+ TEST_LOG("[MUTE ON]");
+ addMarker = false;
+ } else
+ {
+ TEST_LOG("[MUTE OFF]");
+ addMarker = false;
+ }
+ }
+
+ if (_loopBackMeasurements)
+ {
+ WebRtc_UWord16 recDelayMS(0);
+ WebRtc_UWord16 playDelayMS(0);
+ WebRtc_UWord32 nItemsInList(0);
+
+ nItemsInList = _audioList.GetSize();
+ TEST(_audioDevice->RecordingDelay(&recDelayMS) == 0);
+ TEST(_audioDevice->PlayoutDelay(&playDelayMS) == 0);
+ TEST_LOG("Delay (rec+play)+buf: %3u (%3u+%3u)+%3u [ms]\n",
+ recDelayMS + playDelayMS + 10 * (nItemsInList + 1),
+ recDelayMS, playDelayMS, 10 * (nItemsInList + 1));
+
+ addMarker = false;
+ }
+
+ if ((nChannels == 1) && addMarker)
+ {
+ TEST_LOG("+");
+ } else if ((nChannels == 2) && addMarker)
+ {
+ TEST_LOG("++");
+ }
+ } // if (_playCount % 100 == 0)
+
+ nSamplesOut = nSamples;
+
+ return 0;
+}
+;
+
+FuncTestManager::FuncTestManager() :
+ _resourcePath(webrtc::test::ProjectRootPath() +
+ "test/data/audio_device/"),
+ _processThread(NULL),
+ _audioDevice(NULL),
+ _audioEventObserver(NULL),
+ _audioTransport(NULL)
+{
+ assert(!_resourcePath.empty());
+ _playoutFile48 = _resourcePath + "audio_short48.pcm";
+ _playoutFile44 = _resourcePath + "audio_short44.pcm";
+ _playoutFile16 = _resourcePath + "audio_short16.pcm";
+ _playoutFile8 = _resourcePath + "audio_short8.pcm";
+}
+
+FuncTestManager::~FuncTestManager()
+{
+}
+
+WebRtc_Word32 FuncTestManager::Init()
+{
+ TEST((_processThread = ProcessThread::CreateProcessThread()) != NULL);
+ if (_processThread == NULL)
+ {
+ return -1;
+ }
+ _processThread->Start();
+
+ // create the Audio Device module
+ TEST((_audioDevice = AudioDeviceModuleImpl::Create(
+ 555, ADM_AUDIO_LAYER)) != NULL);
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+ TEST(_audioDevice->AddRef() == 1);
+
+ // register the Audio Device module
+ _processThread->RegisterModule(_audioDevice);
+
+ // register event observer
+ _audioEventObserver = new AudioEventObserver(_audioDevice);
+ TEST(_audioDevice->RegisterEventObserver(_audioEventObserver) == 0);
+
+ // register audio transport
+ _audioTransport = new AudioTransportImpl(_audioDevice);
+ TEST(_audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::Close()
+{
+ TEST(_audioDevice->RegisterEventObserver(NULL) == 0);
+ TEST(_audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(_audioDevice->Terminate() == 0);
+
+ // release the ProcessThread object
+ if (_processThread)
+ {
+ _processThread->DeRegisterModule(_audioDevice);
+ _processThread->Stop();
+ ProcessThread::DestroyProcessThread(_processThread);
+ }
+
+ // delete the audio observer
+ if (_audioEventObserver)
+ {
+ delete _audioEventObserver;
+ _audioEventObserver = NULL;
+ }
+
+ // delete the audio transport
+ if (_audioTransport)
+ {
+ delete _audioTransport;
+ _audioTransport = NULL;
+ }
+
+ // release the AudioDeviceModule object
+ if (_audioDevice)
+ {
+ TEST(_audioDevice->Release() == 0);
+ _audioDevice = NULL;
+ }
+
+ // return the ThreadWrapper (singleton)
+ Trace::ReturnTrace();
+
+ // PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::DoTest(const TestType testType)
+{
+ switch (testType)
+ {
+ case TTAll:
+ TestAudioLayerSelection();
+ TestDeviceEnumeration();
+ TestDeviceSelection();
+ TestAudioTransport();
+ TestSpeakerVolume();
+ TestMicrophoneVolume();
+ TestLoopback();
+ case TTAudioLayerSelection:
+ TestAudioLayerSelection();
+ break;
+ case TTDeviceEnumeration:
+ TestDeviceEnumeration();
+ break;
+ case TTDeviceSelection:
+ TestDeviceSelection();
+ break;
+ case TTAudioTransport:
+ TestAudioTransport();
+ break;
+ case TTSpeakerVolume:
+ TestSpeakerVolume();
+ break;
+ case TTMicrophoneVolume:
+ TestMicrophoneVolume();
+ break;
+ case TTSpeakerMute:
+ TestSpeakerMute();
+ break;
+ case TTMicrophoneMute:
+ TestMicrophoneMute();
+ break;
+ case TTMicrophoneBoost:
+ TestMicrophoneBoost();
+ break;
+ case TTMicrophoneAGC:
+ TestMicrophoneAGC();
+ break;
+ case TTLoopback:
+ TestLoopback();
+ break;
+ case TTDeviceRemoval:
+ TestDeviceRemoval();
+ break;
+ case TTMobileAPI:
+ TestAdvancedMBAPI();
+ case TTTest:
+ TestExtra();
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+;
+
+WebRtc_Word32 FuncTestManager::TestAudioLayerSelection()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Audio Layer test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ AudioDeviceModule::AudioLayer audioLayer;
+ TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
+
+ if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
+ {
+ TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio\n \n");
+ } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
+ {
+ TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio\n \n");
+ } else if (audioLayer == AudioDeviceModule::kLinuxAlsaAudio)
+ {
+ TEST_LOG("\nActiveAudioLayer: kLinuxAlsaAudio\n \n");
+ } else if (audioLayer == AudioDeviceModule::kLinuxPulseAudio)
+ {
+ TEST_LOG("\nActiveAudioLayer: kLinuxPulseAudio\n \n");
+ } else
+ {
+ TEST_LOG("\nActiveAudioLayer: INVALID\n \n");
+ }
+
+ char ch;
+ bool tryWinWave(false);
+ bool tryWinCore(false);
+
+ if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
+ {
+ TEST_LOG("Would you like to try kWindowsCoreAudio instead "
+ "[requires Win Vista or Win 7] (Y/N)?\n: ");
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ tryWinCore = true;
+ }
+ } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
+ {
+ TEST_LOG("Would you like to try kWindowsWaveAudio instead (Y/N)?\n: ");
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ tryWinWave = true;
+ }
+ }
+
+ if (tryWinWave || tryWinCore)
+ {
+ // =======================================
+ // First, close down what we have started
+
+ // terminate
+ TEST(_audioDevice->RegisterEventObserver(NULL) == 0);
+ TEST(_audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(_audioDevice->Terminate() == 0);
+
+ // release the ProcessThread object
+ if (_processThread)
+ {
+ _processThread->DeRegisterModule(_audioDevice);
+ _processThread->Stop();
+ ProcessThread::DestroyProcessThread(_processThread);
+ }
+
+ // delete the audio observer
+ if (_audioEventObserver)
+ {
+ delete _audioEventObserver;
+ _audioEventObserver = NULL;
+ }
+
+ // delete the audio transport
+ if (_audioTransport)
+ {
+ delete _audioTransport;
+ _audioTransport = NULL;
+ }
+
+ // release the AudioDeviceModule object
+ if (_audioDevice)
+ {
+ TEST(_audioDevice->Release() == 0);
+ _audioDevice = NULL;
+ }
+
+ // ==================================================
+ // Next, try to make fresh start with new audio layer
+
+ TEST((_processThread = ProcessThread::CreateProcessThread()) != NULL);
+ if (_processThread == NULL)
+ {
+ return -1;
+ }
+ _processThread->Start();
+
+ // create the Audio Device module based on selected audio layer
+ if (tryWinWave)
+ {
+ _audioDevice = AudioDeviceModuleImpl::Create(
+ 555,
+ AudioDeviceModule::kWindowsWaveAudio);
+ } else if (tryWinCore)
+ {
+ _audioDevice = AudioDeviceModuleImpl::Create(
+ 555,
+ AudioDeviceModule::kWindowsCoreAudio);
+ }
+
+ if (_audioDevice == NULL)
+ {
+ TEST_LOG("\nERROR: Switch of audio layer failed!\n");
+ // restore default audio layer instead
+ TEST((_audioDevice = AudioDeviceModuleImpl::Create(
+ 555, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
+ }
+
+ if (_audioDevice == NULL)
+ {
+ TEST_LOG("\nERROR: Failed to revert back to default audio layer!\n");
+ return -1;
+ }
+
+ TEST(_audioDevice->AddRef() == 1);
+
+ // register the Audio Device module
+ _processThread->RegisterModule(_audioDevice);
+
+ // register event observer
+ _audioEventObserver = new AudioEventObserver(_audioDevice);
+ TEST(_audioDevice->RegisterEventObserver(_audioEventObserver) == 0);
+
+ // register audio transport
+ _audioTransport = new AudioTransportImpl(_audioDevice);
+ TEST(_audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ TEST(_audioDevice->ActiveAudioLayer(&audioLayer) == 0);
+
+ if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
+ {
+ if (tryWinCore)
+ TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio <=> "
+ "switch was *not* possible\n \n");
+ else
+ TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio <=> "
+ "switch was possible\n \n");
+ } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
+ {
+ if (tryWinWave)
+ TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio <=> "
+ "switch was *not* possible\n \n");
+ else
+ TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio <=> "
+ "switch was possible\n \n");
+ }
+ } // if (tryWinWave || tryWinCore)
+
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestDeviceEnumeration()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Device Enumeration test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize];
+ WebRtc_Word8 guid[kAdmMaxGuidSize];
+
+ const WebRtc_Word16 nPlayoutDevices(audioDevice->PlayoutDevices());
+ TEST(nPlayoutDevices >= 0);
+ TEST_LOG("\nPlayoutDevices: %u\n \n", nPlayoutDevices);
+ for (int n = 0; n < nPlayoutDevices; n++)
+ {
+ TEST(audioDevice->PlayoutDeviceName(n, name, guid) == 0);
+ TEST_LOG(
+ "PlayoutDeviceName(%d) : name=%s \n \
+ guid=%s\n",
+ n, name, guid);
+ }
+
+#ifdef _WIN32
+ // default (-1)
+ TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == 0);
+ TEST_LOG("PlayoutDeviceName(%d): default name=%s \n \
+ default guid=%s\n", -1, name, guid);
+#else
+ // should fail
+ TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == -1);
+#endif
+
+ const WebRtc_Word16 nRecordingDevices(audioDevice->RecordingDevices());
+ TEST(nRecordingDevices >= 0);
+ TEST_LOG("\nRecordingDevices: %u\n \n", nRecordingDevices);
+ for (int n = 0; n < nRecordingDevices; n++)
+ {
+ TEST(audioDevice->RecordingDeviceName(n, name, guid) == 0);
+ TEST_LOG(
+ "RecordingDeviceName(%d) : name=%s \n \
+ guid=%s\n",
+ n, name, guid);
+ }
+
+#ifdef _WIN32
+ // default (-1)
+ TEST(audioDevice->RecordingDeviceName(-1, name, guid) == 0);
+ TEST_LOG("RecordingDeviceName(%d): default name=%s \n \
+ default guid=%s\n", -1, name, guid);
+#else
+ // should fail
+ TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == -1);
+#endif
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestDeviceSelection()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Device Selection test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+#define PRINT_HEADING(a, b) \
+ { \
+ TEST_LOG("Set" #a "Device(" #b ") => \n"); \
+ } \
+
+#define PRINT_HEADING_IDX(a, b,c ) \
+ { \
+ TEST_LOG("Set" #a "Device(%d) (%s) => \n", b, c); \
+ } \
+
+#define PRINT_STR(a, b) \
+ { \
+ char str[128]; \
+ (b == true) ? (sprintf(str, " %-17s: available\n", #a)) : (sprintf(str, " %-17s: NA\n", #a)); \
+ TEST_LOG("%s", str); \
+ } \
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ bool available(false);
+ WebRtc_Word16 nDevices(-1);
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize];
+ WebRtc_Word8 guid[kAdmMaxGuidSize];
+
+ // =======
+ // Playout
+
+ nDevices = audioDevice->PlayoutDevices();
+ TEST(nDevices >= 0);
+
+ TEST_LOG("\n");
+#ifdef _WIN32
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ PRINT_HEADING(Playout, kDefaultCommunicationDevice);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Playout, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Playout, available);
+ }
+ else
+ {
+ PRINT_STR(Stereo Playout, false);
+ }
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ PRINT_STR(Speaker, available);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Volume, available);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Mute, available);
+
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ PRINT_HEADING(Playout, kDefaultDevice);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Playout, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Playout, available);
+ }
+ else
+ {
+ PRINT_STR(Stereo Playout, false);
+ }
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ PRINT_STR(Speaker, available);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Volume, available);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Mute, available);
+#else
+ TEST(audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == -1);
+ TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == -1);
+#endif
+
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetPlayoutDevice(i) == 0);
+ TEST(audioDevice->PlayoutDeviceName(i, name, guid) == 0);
+ PRINT_HEADING_IDX(Playout, i, name);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Playout, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Playout, available);
+ } else
+ {
+ PRINT_STR(Stereo Playout, false);
+ }
+ TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
+ PRINT_STR(Speaker, available);
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Volume, available);
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ PRINT_STR(Speaker Mute, available);
+ }
+
+ // =========
+ // Recording
+
+ nDevices = audioDevice->RecordingDevices();
+ TEST(nDevices >= 0);
+
+ TEST_LOG("\n");
+#ifdef _WIN32
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == 0);
+ PRINT_HEADING(Recording, kDefaultCommunicationDevice);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ PRINT_STR(Recording, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Recording, available);
+ }
+ else
+ {
+ // special fix to ensure that we don't log 'available' when recording is not OK
+ PRINT_STR(Stereo Recording, false);
+ }
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ PRINT_STR(Microphone, available);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Volume, available);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Mute, available);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Boost, available);
+
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
+ PRINT_HEADING(Recording, kDefaultDevice);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ PRINT_STR(Recording, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Recording, available);
+ }
+ else
+ {
+ // special fix to ensure that we don't log 'available' when recording is not OK
+ PRINT_STR(Stereo Recording, false);
+ }
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ PRINT_STR(Microphone, available);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Volume, available);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Mute, available);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Boost, available);
+#else
+ TEST(audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice) == -1);
+ TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == -1);
+#endif
+
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(audioDevice->SetRecordingDevice(i) == 0);
+ TEST(audioDevice->RecordingDeviceName(i, name, guid) == 0);
+ PRINT_HEADING_IDX(Recording, i, name);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ PRINT_STR(Recording, available);
+ if (available)
+ {
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ PRINT_STR(Stereo Recording, available);
+ } else
+ {
+ // special fix to ensure that we don't log 'available' when recording
+ // is not OK
+ PRINT_STR(Stereo Recording, false);
+ }
+ TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
+ PRINT_STR(Microphone, available);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Volume, available);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Mute, available);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ PRINT_STR(Microphone Boost, available);
+ }
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestAudioTransport()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Audio Transport test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ bool recIsAvailable(false);
+ bool playIsAvailable(false);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
+ if (!recIsAvailable)
+ {
+ TEST_LOG(
+ "\nWARNING: Recording is not available for the selected device!\n \n");
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
+ if (recIsAvailable && playIsAvailable)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else if (!playIsAvailable)
+ {
+ TEST_LOG(
+ "\nWARNING: Playout is not available for the selected device!\n \n");
+ }
+
+ bool available(false);
+ WebRtc_UWord32 samplesPerSec(0);
+
+ if (playIsAvailable)
+ {
+ // =========================================
+ // Start by playing out an existing PCM file
+
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetSpeakerVolume(maxVolume/2) == 0);
+ }
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
+ if (samplesPerSec == 48000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile48.c_str()));
+ } else if (samplesPerSec == 44100 || samplesPerSec == 44000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile44.c_str()));
+ } else if (samplesPerSec == 16000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile16.c_str()));
+ } else if (samplesPerSec == 8000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile8.c_str()));
+ } else {
+ TEST_LOG("\nERROR: Sample rate (%u) is not supported!\n \n",
+ samplesPerSec);
+ return -1;
+ }
+ TEST(audioDevice->StartPlayout() == 0);
+
+ if (audioDevice->Playing())
+ {
+ TEST_LOG("\n> Listen to the file being played (fs=%d) out "
+ "and verify that the audio quality is OK.\n"
+ "> Press any key to stop playing...\n \n",
+ samplesPerSec);
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFilePlayout(false);
+ }
+
+ bool enabled(false);
+ if (recIsAvailable)
+ {
+ // ====================================
+ // Next, record from microphone to file
+
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
+ }
+
+ TEST(audioDevice->StartRawInputFileRecording(
+ GetFilename(RecordedMicrophoneFile)) == 0);
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ // ensure file recording in mono
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
+ }
+ TEST(audioDevice->StartRecording() == 0);
+ AudioDeviceUtility::Sleep(100);
+
+ TEST(audioDevice->Recording() == true);
+ if (audioDevice->Recording())
+ {
+ TEST_LOG("\n \n> The microphone input signal is now being recorded "
+ "to a PCM file.\n"
+ "> Speak into the microphone to ensure that your voice is"
+ " recorded.\n> Press any key to stop recording...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == 0);
+ }
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ }
+
+ if (recIsAvailable && playIsAvailable)
+ {
+ // ==========================
+ // Play out the recorded file
+
+ _audioTransport->SetFilePlayout(true,
+ GetFilename(RecordedMicrophoneFile));
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ AudioDeviceUtility::Sleep(100);
+ }
+
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Playing())
+ {
+ TEST_LOG("\n \n> Listen to the recorded file and verify that the "
+ "audio quality is OK.\n"
+ "> Press any key to stop listening...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFilePlayout(false);
+ }
+
+ if (recIsAvailable && playIsAvailable)
+ {
+ // ==============================
+ // Finally, make full duplex test
+
+ WebRtc_UWord32 playSamplesPerSec(0);
+ WebRtc_UWord32 recSamplesPerSecRec(0);
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ _audioTransport->SetFullDuplex(true);
+
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
+ }
+
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
+ TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
+ if (playSamplesPerSec != recSamplesPerSecRec)
+ {
+ TEST_LOG("\nERROR: sample rates does not match (fs_play=%u, fs_rec=%u)",
+ playSamplesPerSec, recSamplesPerSecRec);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ _audioTransport->SetFullDuplex(false);
+ return -1;
+ }
+
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ AudioDeviceUtility::Sleep(100);
+
+ if (audioDevice->Playing() && audioDevice->Recording())
+ {
+ TEST_LOG("\n \n> Full duplex audio (fs=%u) is now active.\n"
+ "> Speak into the microphone and verify that your voice is "
+ "played out in loopback.\n> Press any key to stop...\n \n",
+ playSamplesPerSec);
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFullDuplex(false);
+ }
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestSpeakerVolume()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Speaker Volume test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ WebRtc_UWord32 startVolume(0);
+ WebRtc_UWord32 samplesPerSec(0);
+
+ TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetSpeakerVolume(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Volume control is not available for the selected "
+ "device!\n \n");
+ return -1;
+ }
+
+ // store initial volume setting
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SpeakerVolume(&startVolume) == 0);
+
+ // start at volume 0
+ TEST(audioDevice->SetSpeakerVolume(0) == 0);
+
+ // ======================================
+ // Start playing out an existing PCM file
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
+ if (48000 == samplesPerSec) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile48.c_str()));
+ } else if (44100 == samplesPerSec || samplesPerSec == 44000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile44.c_str()));
+ } else if (samplesPerSec == 16000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile16.c_str()));
+ } else if (samplesPerSec == 8000) {
+ _audioTransport->SetFilePlayout(
+ true, GetResource(_playoutFile8.c_str()));
+ } else {
+ TEST_LOG("\nERROR: Sample rate (%d) is not supported!\n \n",
+ samplesPerSec);
+ return -1;
+ }
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Playing())
+ {
+ TEST_LOG("\n> Listen to the file being played out and verify that the "
+ "selected speaker volume is varied between [~0] and [~MAX].\n"
+ "> The file shall be played out with an increasing volume level "
+ "correlated to the speaker volume.\n"
+ "> Press any key to stop playing...\n \n");
+ PAUSE(10000);
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetSpeakerVolume(false);
+ _audioTransport->SetFilePlayout(false);
+
+ // restore volume setting
+ TEST(audioDevice->SetSpeakerVolume(startVolume) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestSpeakerMute()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Speaker Mute test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ bool startMute(false);
+ WebRtc_UWord32 samplesPerSec(0);
+
+ TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetSpeakerMute(true);
+ } else
+ {
+ TEST_LOG(
+ "\nERROR: Mute control is not available for the selected"
+ " device!\n \n");
+ return -1;
+ }
+
+ // store initial mute setting
+ TEST(audioDevice->InitSpeaker() == 0);
+ TEST(audioDevice->SpeakerMute(&startMute) == 0);
+
+ // start with no mute
+ TEST(audioDevice->SetSpeakerMute(false) == 0);
+
+ // ======================================
+ // Start playing out an existing PCM file
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
+ if (48000 == samplesPerSec)
+ _audioTransport->SetFilePlayout(true, _playoutFile48.c_str());
+ else if (44100 == samplesPerSec || 44000 == samplesPerSec)
+ _audioTransport->SetFilePlayout(true, _playoutFile44.c_str());
+ else
+ {
+ TEST_LOG("\nERROR: Sample rate (%d) is not supported!\n \n",
+ samplesPerSec);
+ return -1;
+ }
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Playing())
+ {
+ TEST_LOG("\n> Listen to the file being played out and verify that the"
+ " selected speaker mute control is toggled between [MUTE ON] and"
+ " [MUTE OFF].\n> You should only hear the file during the"
+ " 'MUTE OFF' periods.\n"
+ "> Press any key to stop playing...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetSpeakerMute(false);
+ _audioTransport->SetFilePlayout(false);
+
+ // restore mute setting
+ TEST(audioDevice->SetSpeakerMute(startMute) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestMicrophoneVolume()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Microphone Volume test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetMicrophoneVolume(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Volume control is not available for the selected "
+ "device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected "
+ "device!\n \n");
+ return -1;
+ }
+
+ TEST_LOG("\nEnable recording of microphone input to file (%s) during this"
+ " test (Y/N)?\n: ",
+ RecordedMicrophoneVolumeFile);
+ char ch;
+ bool fileRecording(false);
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ fileRecording = true;
+ }
+
+ WebRtc_UWord32 startVolume(0);
+ bool enabled(false);
+
+ // store initial volume setting
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MicrophoneVolume(&startVolume) == 0);
+
+ // start at volume 0
+ TEST(audioDevice->SetMicrophoneVolume(0) == 0);
+
+ // ======================================================================
+ // Start recording from the microphone while the mic volume is changed
+ // continuously.
+ // Also, start playing out the input to enable real-time verification.
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneVolumeFile) == 0);
+ }
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ // ensures a mono file
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
+ }
+ TEST(audioDevice->StartRecording() == 0);
+ }
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the selected "
+ "microphone volume is varied between [~0] and [~MAX].\n"
+ "> You should hear your own voice with an increasing volume level"
+ " correlated to the microphone volume.\n"
+ "> After a finalized test (and if file recording was enabled) "
+ "verify the recorded result off line.\n"
+ "> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ }
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+
+ _audioTransport->SetMicrophoneVolume(false);
+ _audioTransport->SetFullDuplex(false);
+
+ // restore volume setting
+ TEST(audioDevice->SetMicrophoneVolume(startVolume) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestMicrophoneMute()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Microphone Mute test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetMicrophoneMute(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Mute control is not available for the selected"
+ " device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected "
+ "device!\n \n");
+ return -1;
+ }
+
+ TEST_LOG("\nEnable recording of microphone input to file (%s) during this "
+ "test (Y/N)?\n: ",
+ RecordedMicrophoneMuteFile);
+ char ch;
+ bool fileRecording(false);
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ fileRecording = true;
+ }
+
+ bool startMute(false);
+ bool enabled(false);
+
+ // store initial volume setting
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MicrophoneMute(&startMute) == 0);
+
+ // start at no mute
+ TEST(audioDevice->SetMicrophoneMute(false) == 0);
+
+ // ==================================================================
+ // Start recording from the microphone while the mic mute is toggled
+ // continuously.
+ // Also, start playing out the input to enable real-time verification.
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneMuteFile) == 0);
+ }
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ // ensure file recording in mono
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
+ }
+ TEST(audioDevice->StartRecording() == 0);
+ }
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the selected "
+ "microphone mute control is toggled between [MUTE ON] and [MUTE OFF]."
+ "\n> You should only hear your own voice in loopback during the"
+ " 'MUTE OFF' periods.\n> After a finalized test (and if file "
+ "recording was enabled) verify the recorded result off line.\n"
+ "> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ }
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetMicrophoneMute(false);
+ _audioTransport->SetFullDuplex(false);
+
+ // restore volume setting
+ TEST(audioDevice->SetMicrophoneMute(startMute) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestMicrophoneBoost()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Microphone Boost test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetMicrophoneBoost(true);
+ } else
+ {
+ TEST_LOG(
+ "\nERROR: Boost control is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ TEST_LOG("\nEnable recording of microphone input to file (%s) during this "
+ "test (Y/N)?\n: ",
+ RecordedMicrophoneBoostFile);
+ char ch;
+ bool fileRecording(false);
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ fileRecording = true;
+ }
+
+ bool startBoost(false);
+ bool enabled(false);
+
+ // store initial volume setting
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MicrophoneBoost(&startBoost) == 0);
+
+ // start at no boost
+ TEST(audioDevice->SetMicrophoneBoost(false) == 0);
+
+ // ==================================================================
+ // Start recording from the microphone while the mic boost is toggled
+ // continuously.
+ // Also, start playing out the input to enable real-time verification.
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneBoostFile) == 0);
+ }
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ // ensure file recording in mono
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
+ }
+ TEST(audioDevice->StartRecording() == 0);
+ }
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the selected "
+ "microphone boost control is toggled between [BOOST ON] and [BOOST OFF].\n"
+ "> You should hear your own voice with an increased volume level "
+ "during the 'BOOST ON' periods.\n \n"
+ "> After a finalized test (and if file recording was enabled) verify"
+ " the recorded result off line.\n"
+ "> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ }
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetMicrophoneBoost(false);
+ _audioTransport->SetFullDuplex(false);
+
+ // restore boost setting
+ TEST(audioDevice->SetMicrophoneBoost(startBoost) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestMicrophoneAGC()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Microphone AGC test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetMicrophoneAGC(true);
+ } else
+ {
+ TEST_LOG("\nERROR: It is not possible to control the microphone volume"
+ " for the selected device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ TEST_LOG("\nEnable recording of microphone input to file (%s) during "
+ "this test (Y/N)?\n: ",
+ RecordedMicrophoneAGCFile);
+ char ch;
+ bool fileRecording(false);
+ TEST(scanf(" %c", &ch) > 0);
+ ch = toupper(ch);
+ if (ch == 'Y')
+ {
+ fileRecording = true;
+ }
+
+ WebRtc_UWord32 startVolume(0);
+ bool enabled(false);
+
+ // store initial volume setting
+ TEST(audioDevice->InitMicrophone() == 0);
+ TEST(audioDevice->MicrophoneVolume(&startVolume) == 0);
+
+ // ====================================================================
+ // Start recording from the microphone while the mic volume is changed
+ // continuously
+ // by the emulated AGC (implemented by our audio transport).
+ // Also, start playing out the input to enable real-time verification.
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneAGCFile) == 0);
+ }
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ TEST(audioDevice->RecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetAGC(true) == 0);
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ if (enabled)
+ {
+ // ensures a mono file
+ TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
+ }
+ TEST(audioDevice->StartRecording() == 0);
+ }
+ TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+ }
+
+ TEST(audioDevice->AGC() == true);
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->Playing() == true);
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the volume of"
+ " the selected microphone is varied between [~0] and [~MAX].\n"
+ "> You should hear your own voice with an increasing volume level"
+ " correlated to an emulated AGC setting.\n"
+ "> After a finalized test (and if file recording was enabled) verify"
+ " the recorded result off line.\n"
+ "> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ if (fileRecording)
+ {
+ TEST(audioDevice->StopRawInputFileRecording() == 0);
+ }
+ TEST(audioDevice->SetAGC(false) == 0);
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+
+ _audioTransport->SetMicrophoneAGC(false);
+ _audioTransport->SetFullDuplex(false);
+
+ // restore volume setting
+ TEST(audioDevice->SetMicrophoneVolume(startVolume) == 0);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestLoopback()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Loopback measurement test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ bool recIsAvailable(false);
+ bool playIsAvailable(false);
+ WebRtc_UWord8 nPlayChannels(0);
+ WebRtc_UWord8 nRecChannels(0);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
+ if (!recIsAvailable)
+ {
+ TEST_LOG("\nERROR: Recording is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
+ if (recIsAvailable && playIsAvailable)
+ {
+ _audioTransport->SetFullDuplex(true);
+ _audioTransport->SetLoopbackMeasurements(true);
+ } else if (!playIsAvailable)
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ bool enabled(false);
+ bool available(false);
+
+ if (recIsAvailable && playIsAvailable)
+ {
+ WebRtc_UWord32 playSamplesPerSec(0);
+ WebRtc_UWord32 recSamplesPerSecRec(0);
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ _audioTransport->SetFullDuplex(true);
+
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ }
+
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ }
+
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
+ }
+
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
+ TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ enabled ? nPlayChannels = 2 : nPlayChannels = 1;
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ enabled ? nRecChannels = 2 : nRecChannels = 1;
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+
+ if (audioDevice->Playing() && audioDevice->Recording())
+ {
+ TEST_LOG("\n \n> Loopback audio is now active.\n"
+ "> Rec : fs=%u, #channels=%u.\n"
+ "> Play: fs=%u, #channels=%u.\n"
+ "> Speak into the microphone and verify that your voice is"
+ " played out in loopback.\n"
+ "> Press any key to stop...\n \n",
+ recSamplesPerSecRec, nRecChannels, playSamplesPerSec,
+ nPlayChannels);
+ PAUSE(30000);
+ }
+
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFullDuplex(false);
+ _audioTransport->SetLoopbackMeasurements(false);
+ }
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestDeviceRemoval()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Device removal test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ bool recIsAvailable(false);
+ bool playIsAvailable(false);
+ WebRtc_UWord8 nPlayChannels(0);
+ WebRtc_UWord8 nRecChannels(0);
+ WebRtc_UWord8 loopCount(0);
+
+ while (loopCount < 2)
+ {
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
+ if (!recIsAvailable)
+ {
+ TEST_LOG("\nERROR: Recording is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+
+ TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
+ if (recIsAvailable && playIsAvailable)
+ {
+ _audioTransport->SetFullDuplex(true);
+ } else if (!playIsAvailable)
+ {
+ TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
+ return -1;
+ }
+
+ bool available(false);
+ bool enabled(false);
+
+ if (recIsAvailable && playIsAvailable)
+ {
+ WebRtc_UWord32 playSamplesPerSec(0);
+ WebRtc_UWord32 recSamplesPerSecRec(0);
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+
+ _audioTransport->SetFullDuplex(true);
+
+ TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoRecording(true) == 0);
+ }
+
+ TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
+ if (available)
+ {
+ TEST(audioDevice->SetStereoPlayout(true) == 0);
+ }
+
+ TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
+ if (available)
+ {
+ WebRtc_UWord32 maxVolume(0);
+ TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
+ TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
+ }
+
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
+ TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
+ TEST(audioDevice->StereoPlayout(&enabled) == 0);
+ enabled ? nPlayChannels = 2 : nPlayChannels = 1;
+ TEST(audioDevice->StereoRecording(&enabled) == 0);
+ enabled ? nRecChannels = 2 : nRecChannels = 1;
+ TEST(audioDevice->StartRecording() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+
+ AudioDeviceModule::AudioLayer audioLayer;
+ TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
+
+ if (audioLayer == AudioDeviceModule::kLinuxPulseAudio)
+ {
+ TEST_LOG("\n \n> PulseAudio loopback audio is now active.\n"
+ "> Rec : fs=%u, #channels=%u.\n"
+ "> Play: fs=%u, #channels=%u.\n"
+ "> Speak into the microphone and verify that your voice is"
+ " played out in loopback.\n"
+ "> Unplug the device and make sure that your voice is played"
+ " out in loop back on the built-in soundcard.\n"
+ "> Then press any key...\n",
+ recSamplesPerSecRec, nRecChannels, playSamplesPerSec,
+ nPlayChannels);
+
+ PAUSE(DEFAULT_PAUSE_TIME);
+ } else if (audioDevice->Playing() && audioDevice->Recording())
+ {
+ if (loopCount < 1)
+ {
+ TEST_LOG("\n \n> Loopback audio is now active.\n"
+ "> Rec : fs=%u, #channels=%u.\n"
+ "> Play: fs=%u, #channels=%u.\n"
+ "> Speak into the microphone and verify that your voice"
+ " is played out in loopback.\n"
+ "> Unplug the device and wait for the error message...\n",
+ recSamplesPerSecRec, nRecChannels,
+ playSamplesPerSec, nPlayChannels);
+
+ _audioEventObserver->_error
+ = (AudioDeviceObserver::ErrorCode) (-1);
+ while (_audioEventObserver->_error
+ == (AudioDeviceObserver::ErrorCode) (-1))
+ {
+ SLEEP(500);
+ }
+ } else
+ {
+ TEST_LOG("\n \n> Loopback audio is now active.\n"
+ "> Rec : fs=%u, #channels=%u.\n"
+ "> Play: fs=%u, #channels=%u.\n"
+ "> Speak into the microphone and verify that your voice"
+ " is played out in loopback.\n"
+ "> Press any key to stop...\n",
+ recSamplesPerSecRec, nRecChannels,
+ playSamplesPerSec, nPlayChannels);
+
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+ }
+
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFullDuplex(false);
+
+ if (loopCount < 1)
+ {
+ TEST_LOG("\n \n> Stopped!\n");
+ TEST_LOG("> Now reinsert device if you want to enumerate it.\n");
+ TEST_LOG("> Press any key when done.\n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ loopCount++;
+ }
+ } // loopCount
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::TestExtra()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Extra test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ TEST(audioDevice->Terminate() == 0);
+ TEST(audioDevice->Initialized() == false);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+WebRtc_Word32 FuncTestManager::SelectRecordingDevice()
+{
+ WebRtc_Word16 nDevices = _audioDevice->RecordingDevices();
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize];
+ WebRtc_Word8 guid[kAdmMaxGuidSize];
+ WebRtc_Word32 ret(-1);
+
+#ifdef _WIN32
+ TEST_LOG("\nSelect Recording Device\n \n");
+ TEST_LOG(" (%d) Default\n", 0);
+ TEST_LOG(" (%d) Default Communication [Win 7]\n", 1);
+ TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(_audioDevice->RecordingDeviceName(i, name, guid) == 0);
+ TEST_LOG(" (%d) Device %d (%s)\n", i+10, i, name);
+ }
+ TEST_LOG("\n: ");
+
+ int sel(0);
+
+ scanf("%u", &sel);
+
+ if (sel == 0)
+ {
+ TEST((ret = _audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice)) == 0);
+ }
+ else if (sel == 1)
+ {
+ TEST((ret = _audioDevice->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice)) == 0);
+ }
+ else if (sel < (nDevices+10))
+ {
+ TEST((ret = _audioDevice->SetRecordingDevice(sel-10)) == 0);
+ }
+ else
+ {
+ return -1;
+ }
+#else
+ TEST_LOG("\nSelect Recording Device\n \n");
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(_audioDevice->RecordingDeviceName(i, name, guid) == 0);
+ TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
+ }
+ TEST_LOG("\n: ");
+ int sel(0);
+ TEST(scanf("%u", &sel) > 0);
+ if (sel < (nDevices))
+ {
+ TEST((ret = _audioDevice->SetRecordingDevice(sel)) == 0);
+ } else
+ {
+ return -1;
+ }
+#endif
+
+ return ret;
+}
+
+WebRtc_Word32 FuncTestManager::SelectPlayoutDevice()
+{
+ WebRtc_Word16 nDevices = _audioDevice->PlayoutDevices();
+ WebRtc_Word8 name[kAdmMaxDeviceNameSize];
+ WebRtc_Word8 guid[kAdmMaxGuidSize];
+
+#ifdef _WIN32
+ TEST_LOG("\nSelect Playout Device\n \n");
+ TEST_LOG(" (%d) Default\n", 0);
+ TEST_LOG(" (%d) Default Communication [Win 7]\n", 1);
+ TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(_audioDevice->PlayoutDeviceName(i, name, guid) == 0);
+ TEST_LOG(" (%d) Device %d (%s)\n", i+10, i, name);
+ }
+ TEST_LOG("\n: ");
+
+ int sel(0);
+
+ scanf("%u", &sel);
+
+ WebRtc_Word32 ret(0);
+
+ if (sel == 0)
+ {
+ TEST((ret = _audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultDevice)) == 0);
+ }
+ else if (sel == 1)
+ {
+ TEST((ret = _audioDevice->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice)) == 0);
+ }
+ else if (sel < (nDevices+10))
+ {
+ TEST((ret = _audioDevice->SetPlayoutDevice(sel-10)) == 0);
+ }
+ else
+ {
+ return -1;
+ }
+#else
+ TEST_LOG("\nSelect Playout Device\n \n");
+ for (int i = 0; i < nDevices; i++)
+ {
+ TEST(_audioDevice->PlayoutDeviceName(i, name, guid) == 0);
+ TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
+ }
+ TEST_LOG("\n: ");
+ int sel(0);
+ TEST(scanf("%u", &sel) > 0);
+ WebRtc_Word32 ret(0);
+ if (sel < (nDevices))
+ {
+ TEST((ret = _audioDevice->SetPlayoutDevice(sel)) == 0);
+ } else
+ {
+ return -1;
+ }
+#endif
+
+ return ret;
+}
+
+WebRtc_Word32 FuncTestManager::TestAdvancedMBAPI()
+{
+ TEST_LOG("\n=======================================\n");
+ TEST_LOG(" Advanced mobile device API test:\n");
+ TEST_LOG("=======================================\n");
+
+ if (_audioDevice == NULL)
+ {
+ return -1;
+ }
+
+ RESET_TEST;
+
+ AudioDeviceModule* audioDevice = _audioDevice;
+
+ TEST(audioDevice->Init() == 0);
+ TEST(audioDevice->Initialized() == true);
+
+ if (SelectRecordingDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+ if (SelectPlayoutDevice() == -1)
+ {
+ TEST_LOG("\nERROR: Device selection failed!\n \n");
+ return -1;
+ }
+ _audioTransport->SetFullDuplex(true);
+ _audioTransport->SetLoopbackMeasurements(true);
+
+ TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
+ // Start recording
+ TEST(audioDevice->InitRecording() == 0);
+ TEST(audioDevice->StartRecording() == 0);
+ // Start playout
+ TEST(audioDevice->InitPlayout() == 0);
+ TEST(audioDevice->StartPlayout() == 0);
+
+ TEST(audioDevice->Recording() == true);
+ TEST(audioDevice->Playing() == true);
+
+#if defined(_WIN32_WCE) || defined(MAC_IPHONE)
+ TEST_LOG("\nResetAudioDevice\n \n");
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n\
+> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+ for (int p=0; p<=60; p+=20)
+ {
+ TEST_LOG("Resetting sound device several time with pause %d ms\n", p);
+ for (int l=0; l<20; ++l)
+ {
+ TEST(audioDevice->ResetAudioDevice() == 0);
+ AudioDeviceUtility::Sleep(p);
+ }
+ TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n");
+ AudioDeviceUtility::Sleep(2000);
+ }
+#endif
+
+#if defined(MAC_IPHONE)
+ bool loudspeakerOn(false);
+ TEST_LOG("\nSet playout spaker\n \n");
+ if (audioDevice->Recording() && audioDevice->Playing())
+ {
+ TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n\
+> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ }
+
+ TEST_LOG("Set to use speaker\n");
+ TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
+ TEST_LOG("\n> Speak into the microphone and verify that the audio is"
+ " from the loudspeaker.\n\
+> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == true);
+
+ TEST_LOG("Set to not use speaker\n");
+ TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
+ TEST_LOG("\n> Speak into the microphone and verify that the audio is not"
+ " from the loudspeaker.\n\
+> Press any key to stop...\n \n");
+ PAUSE(DEFAULT_PAUSE_TIME);
+ TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
+ TEST(loudspeakerOn == false);
+#endif
+
+ TEST(audioDevice->StopRecording() == 0);
+ TEST(audioDevice->StopPlayout() == 0);
+ TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
+
+ _audioTransport->SetFullDuplex(false);
+
+ TEST_LOG("\n");
+ PRINT_TEST_RESULTS;
+
+ return 0;
+}
+
+} // namespace webrtc
+
+// EOF
diff --git a/trunk/src/modules/audio_device/main/test/func_test_manager.h b/trunk/src/modules/audio_device/main/test/func_test_manager.h
new file mode 100644
index 0000000..5efae50
--- /dev/null
+++ b/trunk/src/modules/audio_device/main/test/func_test_manager.h
@@ -0,0 +1,228 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
+#define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
+
+#include "../source/audio_device_utility.h"
+
+#include <string>
+
+#include "typedefs.h"
+#include "audio_device.h"
+#include "audio_device_test_defines.h"
+#include "file_wrapper.h"
+#include "list_wrapper.h"
+#include "resampler.h"
+
+#if defined(MAC_IPHONE) || defined(ANDROID)
+#define USE_SLEEP_AS_PAUSE
+#else
+//#define USE_SLEEP_AS_PAUSE
+#endif
+
+// Sets the default pause time if using sleep as pause
+#define DEFAULT_PAUSE_TIME 5000
+
+#if defined(USE_SLEEP_AS_PAUSE)
+#define PAUSE(a) AudioDeviceUtility::Sleep(a);
+#else
+#define PAUSE(a) AudioDeviceUtility::WaitForKey();
+#endif
+
+#define SLEEP(a) AudioDeviceUtility::Sleep(a);
+
+#define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
+//#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio
+
+enum TestType
+{
+ TTInvalid = -1,
+ TTAll = 0,
+ TTAudioLayerSelection = 1,
+ TTDeviceEnumeration = 2,
+ TTDeviceSelection = 3,
+ TTAudioTransport = 4,
+ TTSpeakerVolume = 5,
+ TTMicrophoneVolume = 6,
+ TTSpeakerMute = 7,
+ TTMicrophoneMute = 8,
+ TTMicrophoneBoost = 9,
+ TTMicrophoneAGC = 10,
+ TTLoopback = 11,
+ TTDeviceRemoval = 13,
+ TTMobileAPI = 14,
+ TTTest = 66,
+};
+
+class ProcessThread;
+
+namespace webrtc
+{
+
+class AudioDeviceModule;
+class AudioEventObserver;
+class AudioTransport;
+
+// ----------------------------------------------------------------------------
+// AudioEventObserver
+// ----------------------------------------------------------------------------
+
+class AudioEventObserver: public AudioDeviceObserver
+{
+public:
+ virtual void OnErrorIsReported(const ErrorCode error);
+ virtual void OnWarningIsReported(const WarningCode warning);
+ AudioEventObserver(AudioDeviceModule* audioDevice);
+ ~AudioEventObserver();
+public:
+ ErrorCode _error;
+ WarningCode _warning;
+private:
+ AudioDeviceModule* _audioDevice;
+};
+
+// ----------------------------------------------------------------------------
+// AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransportImpl: public AudioTransport
+{
+public:
+ virtual WebRtc_Word32
+ RecordedDataIsAvailable(const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord32 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord32 currentMicLevel,
+ WebRtc_UWord32& newMicLevel);
+
+ virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ WebRtc_Word8* audioSamples,
+ WebRtc_UWord32& nSamplesOut);
+
+ AudioTransportImpl(AudioDeviceModule* audioDevice);
+ ~AudioTransportImpl();
+
+public:
+ WebRtc_Word32 SetFilePlayout(bool enable, const WebRtc_Word8* fileName =
+ NULL);
+ void SetFullDuplex(bool enable);
+ void SetSpeakerVolume(bool enable)
+ {
+ _speakerVolume = enable;
+ }
+ ;
+ void SetSpeakerMute(bool enable)
+ {
+ _speakerMute = enable;
+ }
+ ;
+ void SetMicrophoneMute(bool enable)
+ {
+ _microphoneMute = enable;
+ }
+ ;
+ void SetMicrophoneVolume(bool enable)
+ {
+ _microphoneVolume = enable;
+ }
+ ;
+ void SetMicrophoneBoost(bool enable)
+ {
+ _microphoneBoost = enable;
+ }
+ ;
+ void SetLoopbackMeasurements(bool enable)
+ {
+ _loopBackMeasurements = enable;
+ }
+ ;
+ void SetMicrophoneAGC(bool enable)
+ {
+ _microphoneAGC = enable;
+ }
+ ;
+
+private:
+ AudioDeviceModule* _audioDevice;
+
+ bool _playFromFile;
+ bool _fullDuplex;
+ bool _speakerVolume;
+ bool _speakerMute;
+ bool _microphoneVolume;
+ bool _microphoneMute;
+ bool _microphoneBoost;
+ bool _microphoneAGC;
+ bool _loopBackMeasurements;
+
+ FileWrapper& _playFile;
+
+ WebRtc_UWord32 _recCount;
+ WebRtc_UWord32 _playCount;
+
+ ListWrapper _audioList;
+
+ Resampler _resampler;
+};
+
+// ----------------------------------------------------------------------------
+// FuncTestManager
+// ----------------------------------------------------------------------------
+
+class FuncTestManager
+{
+public:
+ FuncTestManager();
+ ~FuncTestManager();
+ WebRtc_Word32 Init();
+ WebRtc_Word32 Close();
+ WebRtc_Word32 DoTest(const TestType testType);
+private:
+ WebRtc_Word32 TestAudioLayerSelection();
+ WebRtc_Word32 TestDeviceEnumeration();
+ WebRtc_Word32 TestDeviceSelection();
+ WebRtc_Word32 TestAudioTransport();
+ WebRtc_Word32 TestSpeakerVolume();
+ WebRtc_Word32 TestMicrophoneVolume();
+ WebRtc_Word32 TestSpeakerMute();
+ WebRtc_Word32 TestMicrophoneMute();
+ WebRtc_Word32 TestMicrophoneBoost();
+ WebRtc_Word32 TestLoopback();
+ WebRtc_Word32 TestDeviceRemoval();
+ WebRtc_Word32 TestExtra();
+ WebRtc_Word32 TestMicrophoneAGC();
+ WebRtc_Word32 SelectPlayoutDevice();
+ WebRtc_Word32 SelectRecordingDevice();
+ WebRtc_Word32 TestAdvancedMBAPI();
+private:
+ // Paths to where the resource files to be used for this test are located.
+ std::string _resourcePath;
+ std::string _playoutFile48;
+ std::string _playoutFile44;
+ std::string _playoutFile16;
+ std::string _playoutFile8;
+
+ ProcessThread* _processThread;
+ AudioDeviceModule* _audioDevice;
+ AudioEventObserver* _audioEventObserver;
+ AudioTransportImpl* _audioTransport;
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
diff --git a/trunk/src/modules/audio_processing/Android.mk b/trunk/src/modules/audio_processing/Android.mk
new file mode 100644
index 0000000..2ab5bb6
--- /dev/null
+++ b/trunk/src/modules/audio_processing/Android.mk
@@ -0,0 +1,143 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_apm
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ $(call all-proto-files-under, .) \
+ audio_buffer.cc \
+ audio_processing_impl.cc \
+ echo_cancellation_impl.cc \
+ echo_control_mobile_impl.cc \
+ gain_control_impl.cc \
+ high_pass_filter_impl.cc \
+ level_estimator_impl.cc \
+ noise_suppression_impl.cc \
+ splitting_filter.cc \
+ processing_component.cc \
+ voice_detection_impl.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_NS_FIXED'
+# floating point
+# -DWEBRTC_NS_FLOAT'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/aec/include \
+ $(LOCAL_PATH)/aecm/include \
+ $(LOCAL_PATH)/agc/include \
+ $(LOCAL_PATH)/ns/include \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../common_audio/vad/include \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ external/protobuf/src
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+# apm process test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ $(call all-proto-files-under, .) \
+ test/process_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ external/gtest/include
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libwebrtc_audio_preprocessing
+
+LOCAL_MODULE:= webrtc_apm_process_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
+
+# apm unit test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ $(call all-proto-files-under, test) \
+ test/unit_test.cc \
+ ../../../test/testsupport/fileutils.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_APM_UNIT_TEST_FIXED_PROFILE'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../../test \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ $(LOCAL_PATH)/../../common_audio/signal_processing/include \
+ external/gtest/include \
+ external/protobuf/src
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libwebrtc_audio_preprocessing
+
+LOCAL_MODULE:= webrtc_apm_unit_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
diff --git a/trunk/src/modules/audio_processing/OWNERS b/trunk/src/modules/audio_processing/OWNERS
new file mode 100644
index 0000000..5a25634
--- /dev/null
+++ b/trunk/src/modules/audio_processing/OWNERS
@@ -0,0 +1,2 @@
+andrew@webrtc.org
+bjornv@webrtc.org
diff --git a/trunk/src/modules/audio_processing/aec/Android.mk b/trunk/src/modules/audio_processing/aec/Android.mk
new file mode 100644
index 0000000..26679d8
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/Android.mk
@@ -0,0 +1,48 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aec
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ echo_cancellation.c \
+ aec_resampler.c \
+ aec_core.c \
+ aec_rdft.c \
+
+ifeq ($(TARGET_ARCH),x86)
+LOCAL_SRC_FILES += \
+ aec_core_sse2.c \
+ aec_rdft_sse2.c
+endif
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/aec/aec.gypi b/trunk/src/modules/audio_processing/aec/aec.gypi
new file mode 100644
index 0000000..2e34c5b
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec.gypi
@@ -0,0 +1,74 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'aec',
+ 'type': '<(library)',
+ 'variables': {
+ # Outputs some low-level debug files.
+ 'aec_debug_dump%': 0,
+ },
+ 'dependencies': [
+ 'apm_util',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/echo_cancellation.h',
+ 'echo_cancellation.c',
+ 'aec_core.h',
+ 'aec_core.c',
+ 'aec_rdft.h',
+ 'aec_rdft.c',
+ 'aec_resampler.h',
+ 'aec_resampler.c',
+ ],
+ 'conditions': [
+ ['target_arch=="ia32" or target_arch=="x64"', {
+ 'dependencies': [ 'aec_sse2', ],
+ }],
+ ['aec_debug_dump==1', {
+ 'defines': [ 'WEBRTC_AEC_DEBUG_DUMP', ],
+ }],
+ ],
+ },
+ ],
+ 'conditions': [
+ ['target_arch=="ia32" or target_arch=="x64"', {
+ 'targets': [
+ {
+ 'target_name': 'aec_sse2',
+ 'type': '<(library)',
+ 'sources': [
+ 'aec_core_sse2.c',
+ 'aec_rdft_sse2.c',
+ ],
+ 'conditions': [
+ ['os_posix==1 and OS!="mac"', {
+ 'cflags': [ '-msse2', ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ 'OTHER_CFLAGS': [ '-msse2', ],
+ },
+ }],
+ ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/aec/aec_core.c b/trunk/src/modules/audio_processing/aec/aec_core.c
new file mode 100644
index 0000000..1637e6f
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_core.c
@@ -0,0 +1,1527 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * The core AEC algorithm, which is presented with time-aligned signals.
+ */
+
+#include "aec_core.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stddef.h> // size_t
+#include <stdlib.h>
+#include <string.h>
+
+#include "aec_rdft.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "delay_estimator_wrapper.h"
+#include "ring_buffer.h"
+#include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "typedefs.h"
+
+// Buffer size (samples)
+static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
+
+// Noise suppression
+static const int converged = 250;
+
+// Metrics
+static const int subCountLen = 4;
+static const int countLen = 50;
+
+// Quantities to control H band scaling for SWB input
+static const int flagHbandCn = 1; // flag for adding comfort noise in H band
+static const float cnScaleHband = (float)0.4; // scale for comfort noise in H band
+// Initial bin for averaging nlp gain in low band
+static const int freqAvgIc = PART_LEN / 2;
+
+// Matlab code to produce table:
+// win = sqrt(hanning(63)); win = [0 ; win(1:32)];
+// fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
+static const float sqrtHanning[65] = {
+ 0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
+ 0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
+ 0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
+ 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+ 0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
+ 0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
+ 0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
+ 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+ 0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
+ 0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
+ 0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
+ 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+ 0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
+ 0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
+ 0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
+ 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+ 0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
+ 0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
+ 0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
+ 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+ 0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
+ 0.99969881869620f, 1.00000000000000f
+};
+
+// Matlab code to produce table:
+// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
+const float WebRtcAec_weightCurve[65] = {
+ 0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f,
+ 0.1845f, 0.1926f, 0.2000f, 0.2069f, 0.2134f, 0.2195f,
+ 0.2254f, 0.2309f, 0.2363f, 0.2414f, 0.2464f, 0.2512f,
+ 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
+ 0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f,
+ 0.3035f, 0.3070f, 0.3104f, 0.3138f, 0.3171f, 0.3204f,
+ 0.3236f, 0.3268f, 0.3299f, 0.3330f, 0.3360f, 0.3390f,
+ 0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
+ 0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f,
+ 0.3752f, 0.3777f, 0.3803f, 0.3828f, 0.3854f, 0.3878f,
+ 0.3903f, 0.3928f, 0.3952f, 0.3976f, 0.4000f
+};
+
+// Matlab code to produce table:
+// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
+const float WebRtcAec_overDriveCurve[65] = {
+ 1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f,
+ 1.3062f, 1.3307f, 1.3536f, 1.3750f, 1.3953f, 1.4146f,
+ 1.4330f, 1.4507f, 1.4677f, 1.4841f, 1.5000f, 1.5154f,
+ 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
+ 1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f,
+ 1.6847f, 1.6960f, 1.7071f, 1.7181f, 1.7289f, 1.7395f,
+ 1.7500f, 1.7603f, 1.7706f, 1.7806f, 1.7906f, 1.8004f,
+ 1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
+ 1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f,
+ 1.9186f, 1.9270f, 1.9354f, 1.9437f, 1.9520f, 1.9601f,
+ 1.9682f, 1.9763f, 1.9843f, 1.9922f, 2.0000f
+};
+
+// "Private" function prototypes.
+static void ProcessBlock(aec_t* aec);
+
+static void NonLinearProcessing(aec_t *aec, short *output, short *outputH);
+
+static void GetHighbandGain(const float *lambda, float *nlpGainHband);
+
+// Comfort_noise also computes noise for H band returned in comfortNoiseHband
+static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
+ complex_t *comfortNoiseHband,
+ const float *noisePow, const float *lambda);
+
+static void WebRtcAec_InitLevel(power_level_t *level);
+static void WebRtcAec_InitStats(stats_t *stats);
+static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]);
+static void UpdateMetrics(aec_t *aec);
+// Convert from time domain to frequency domain. Note that |time_data| are
+// overwritten.
+static void TimeToFrequency(float time_data[PART_LEN2],
+ float freq_data[2][PART_LEN1],
+ int window);
+
+__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
+{
+ return aRe * bRe - aIm * bIm;
+}
+
+__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
+{
+ return aRe * bIm + aIm * bRe;
+}
+
+static int CmpFloat(const void *a, const void *b)
+{
+ const float *da = (const float *)a;
+ const float *db = (const float *)b;
+
+ return (*da > *db) - (*da < *db);
+}
+
+int WebRtcAec_CreateAec(aec_t **aecInst)
+{
+ aec_t *aec = malloc(sizeof(aec_t));
+ *aecInst = aec;
+ if (aec == NULL) {
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aec->nearFrBuf,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aec->outFrBuf,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aec->nearFrBufH,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aec->outFrBufH,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+
+ // Create far-end buffers.
+ if (WebRtc_CreateBuffer(&aec->far_buf, kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+ if (WebRtc_CreateBuffer(&aec->far_buf_windowed, kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_CreateBuffer(&aec->far_time_buf, kBufSizePartitions,
+ sizeof(int16_t) * PART_LEN) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+#endif
+ if (WebRtc_CreateDelayEstimator(&aec->delay_estimator,
+ PART_LEN1,
+ kMaxDelayBlocks,
+ kLookaheadBlocks) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+int WebRtcAec_FreeAec(aec_t *aec)
+{
+ if (aec == NULL) {
+ return -1;
+ }
+
+ WebRtc_FreeBuffer(aec->nearFrBuf);
+ WebRtc_FreeBuffer(aec->outFrBuf);
+
+ WebRtc_FreeBuffer(aec->nearFrBufH);
+ WebRtc_FreeBuffer(aec->outFrBufH);
+
+ WebRtc_FreeBuffer(aec->far_buf);
+ WebRtc_FreeBuffer(aec->far_buf_windowed);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_FreeBuffer(aec->far_time_buf);
+#endif
+ WebRtc_FreeDelayEstimator(aec->delay_estimator);
+
+ free(aec);
+ return 0;
+}
+
+static void FilterFar(aec_t *aec, float yf[2][PART_LEN1])
+{
+ int i;
+ for (i = 0; i < NR_PART; i++) {
+ int j;
+ int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+ int pos = i * PART_LEN1;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART*(PART_LEN1);
+ }
+
+ for (j = 0; j < PART_LEN1; j++) {
+ yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
+ yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
+ }
+ }
+}
+
+static void ScaleErrorSignal(aec_t *aec, float ef[2][PART_LEN1])
+{
+ int i;
+ float absEf;
+ for (i = 0; i < (PART_LEN1); i++) {
+ ef[0][i] /= (aec->xPow[i] + 1e-10f);
+ ef[1][i] /= (aec->xPow[i] + 1e-10f);
+ absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+
+ if (absEf > aec->errThresh) {
+ absEf = aec->errThresh / (absEf + 1e-10f);
+ ef[0][i] *= absEf;
+ ef[1][i] *= absEf;
+ }
+
+ // Stepsize factor
+ ef[0][i] *= aec->mu;
+ ef[1][i] *= aec->mu;
+ }
+}
+
+// Time-unconstrined filter adaptation.
+// TODO(andrew): consider for a low-complexity mode.
+//static void FilterAdaptationUnconstrained(aec_t *aec, float *fft,
+// float ef[2][PART_LEN1]) {
+// int i, j;
+// for (i = 0; i < NR_PART; i++) {
+// int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+// int pos;
+// // Check for wrap
+// if (i + aec->xfBufBlockPos >= NR_PART) {
+// xPos -= NR_PART * PART_LEN1;
+// }
+//
+// pos = i * PART_LEN1;
+//
+// for (j = 0; j < PART_LEN1; j++) {
+// aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
+// aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
+// }
+// }
+//}
+
+static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
+ int i, j;
+ for (i = 0; i < NR_PART; i++) {
+ int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+ int pos;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART * PART_LEN1;
+ }
+
+ pos = i * PART_LEN1;
+
+ for (j = 0; j < PART_LEN; j++) {
+
+ fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
+ -aec->xfBuf[1][xPos + j],
+ ef[0][j], ef[1][j]);
+ fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
+ -aec->xfBuf[1][xPos + j],
+ ef[0][j], ef[1][j]);
+ }
+ fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
+ -aec->xfBuf[1][xPos + PART_LEN],
+ ef[0][PART_LEN], ef[1][PART_LEN]);
+
+ aec_rdft_inverse_128(fft);
+ memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
+
+ // fft scaling
+ {
+ float scale = 2.0f / PART_LEN2;
+ for (j = 0; j < PART_LEN; j++) {
+ fft[j] *= scale;
+ }
+ }
+ aec_rdft_forward_128(fft);
+
+ aec->wfBuf[0][pos] += fft[0];
+ aec->wfBuf[0][pos + PART_LEN] += fft[1];
+
+ for (j = 1; j < PART_LEN; j++) {
+ aec->wfBuf[0][pos + j] += fft[2 * j];
+ aec->wfBuf[1][pos + j] += fft[2 * j + 1];
+ }
+ }
+}
+
+static void OverdriveAndSuppress(aec_t *aec, float hNl[PART_LEN1],
+ const float hNlFb,
+ float efw[2][PART_LEN1]) {
+ int i;
+ for (i = 0; i < PART_LEN1; i++) {
+ // Weight subbands
+ if (hNl[i] > hNlFb) {
+ hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
+ (1 - WebRtcAec_weightCurve[i]) * hNl[i];
+ }
+ hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+
+ // Suppress error signal
+ efw[0][i] *= hNl[i];
+ efw[1][i] *= hNl[i];
+
+ // Ooura fft returns incorrect sign on imaginary component. It matters here
+ // because we are making an additive change with comfort noise.
+ efw[1][i] *= -1;
+ }
+}
+
+WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
+WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
+WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
+WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
+
+int WebRtcAec_InitAec(aec_t *aec, int sampFreq)
+{
+ int i;
+
+ aec->sampFreq = sampFreq;
+
+ if (sampFreq == 8000) {
+ aec->mu = 0.6f;
+ aec->errThresh = 2e-6f;
+ }
+ else {
+ aec->mu = 0.5f;
+ aec->errThresh = 1.5e-6f;
+ }
+
+ if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
+ return -1;
+ }
+
+ if (WebRtc_InitBuffer(aec->outFrBuf) == -1) {
+ return -1;
+ }
+
+ if (WebRtc_InitBuffer(aec->nearFrBufH) == -1) {
+ return -1;
+ }
+
+ if (WebRtc_InitBuffer(aec->outFrBufH) == -1) {
+ return -1;
+ }
+
+ // Initialize far-end buffers.
+ if (WebRtc_InitBuffer(aec->far_buf) == -1) {
+ return -1;
+ }
+ if (WebRtc_InitBuffer(aec->far_buf_windowed) == -1) {
+ return -1;
+ }
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_InitBuffer(aec->far_time_buf) == -1) {
+ return -1;
+ }
+#endif
+ aec->system_delay = 0;
+
+ if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
+ return -1;
+ }
+ aec->delay_logging_enabled = 0;
+ memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
+
+ // Default target suppression level
+ aec->targetSupp = -11.5;
+ aec->minOverDrive = 2.0;
+
+ // Sampling frequency multiplier
+ // SWB is processed as 160 frame size
+ if (aec->sampFreq == 32000) {
+ aec->mult = (short)aec->sampFreq / 16000;
+ }
+ else {
+ aec->mult = (short)aec->sampFreq / 8000;
+ }
+
+ aec->farBufWritePos = 0;
+ aec->farBufReadPos = 0;
+
+ aec->inSamples = 0;
+ aec->outSamples = 0;
+ aec->knownDelay = 0;
+
+ // Initialize buffers
+ memset(aec->dBuf, 0, sizeof(aec->dBuf));
+ memset(aec->eBuf, 0, sizeof(aec->eBuf));
+ // For H band
+ memset(aec->dBufH, 0, sizeof(aec->dBufH));
+
+ memset(aec->xPow, 0, sizeof(aec->xPow));
+ memset(aec->dPow, 0, sizeof(aec->dPow));
+ memset(aec->dInitMinPow, 0, sizeof(aec->dInitMinPow));
+ aec->noisePow = aec->dInitMinPow;
+ aec->noiseEstCtr = 0;
+
+ // Initial comfort noise power
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->dMinPow[i] = 1.0e6f;
+ }
+
+ // Holds the last block written to
+ aec->xfBufBlockPos = 0;
+ // TODO: Investigate need for these initializations. Deleting them doesn't
+ // change the output at all and yields 0.4% overall speedup.
+ memset(aec->xfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+ memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+ memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
+ memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
+ memset(aec->xfwBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+ memset(aec->se, 0, sizeof(float) * PART_LEN1);
+
+ // To prevent numerical instability in the first block.
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sd[i] = 1;
+ }
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sx[i] = 1;
+ }
+
+ memset(aec->hNs, 0, sizeof(aec->hNs));
+ memset(aec->outBuf, 0, sizeof(float) * PART_LEN);
+
+ aec->hNlFbMin = 1;
+ aec->hNlFbLocalMin = 1;
+ aec->hNlXdAvgMin = 1;
+ aec->hNlNewMin = 0;
+ aec->hNlMinCtr = 0;
+ aec->overDrive = 2;
+ aec->overDriveSm = 2;
+ aec->delayIdx = 0;
+ aec->stNearState = 0;
+ aec->echoState = 0;
+ aec->divergeState = 0;
+
+ aec->seed = 777;
+ aec->delayEstCtr = 0;
+
+ // Metrics disabled by default
+ aec->metricsMode = 0;
+ WebRtcAec_InitMetrics(aec);
+
+ // Assembly optimization
+ WebRtcAec_FilterFar = FilterFar;
+ WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
+ WebRtcAec_FilterAdaptation = FilterAdaptation;
+ WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+ if (WebRtc_GetCPUInfo(kSSE2)) {
+ WebRtcAec_InitAec_SSE2();
+ }
+#endif
+
+ aec_rdft_init();
+
+ return 0;
+}
+
+void WebRtcAec_InitMetrics(aec_t *aec)
+{
+ aec->stateCounter = 0;
+ WebRtcAec_InitLevel(&aec->farlevel);
+ WebRtcAec_InitLevel(&aec->nearlevel);
+ WebRtcAec_InitLevel(&aec->linoutlevel);
+ WebRtcAec_InitLevel(&aec->nlpoutlevel);
+
+ WebRtcAec_InitStats(&aec->erl);
+ WebRtcAec_InitStats(&aec->erle);
+ WebRtcAec_InitStats(&aec->aNlp);
+ WebRtcAec_InitStats(&aec->rerl);
+}
+
+
+void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend) {
+ float fft[PART_LEN2];
+ float xf[2][PART_LEN1];
+
+ // Check if the buffer is full, and in that case flush the oldest data.
+ if (WebRtc_available_write(aec->far_buf) < 1) {
+ WebRtc_MoveReadPtr(aec->far_buf, 1);
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, 1);
+ aec->system_delay -= PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, 1);
+#endif
+ }
+ // Convert far-end partition to the frequency domain without windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 0);
+ WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
+
+ // Convert far-end partition to the frequency domain with windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 1);
+ WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
+}
+
+void WebRtcAec_ProcessFrame(aec_t *aec,
+ const short *nearend,
+ const short *nearendH,
+ int knownDelay)
+{
+ // For each frame the process is as follows:
+ // 1) If the system_delay indicates on being too small for processing a
+ // frame we stuff the buffer with enough data for 10 ms.
+ // 2) Adjust the buffer to the system delay, by moving the read pointer.
+ // 3) If we can't move read pointer due to buffer size limitations we
+ // flush/stuff the buffer.
+ // 4) Process as many partitions as possible.
+ // 5) Update the |system_delay| with respect to a full frame of FRAME_LEN
+ // samples. Even though we will have data left to process (we work with
+ // partitions) we consider updating a whole frame, since that's the
+ // amount of data we input and output in audio_processing.
+
+ // TODO(bjornv): Investigate how we should round the delay difference; right
+ // now we know that incoming |knownDelay| is underestimated when it's less
+ // than |aec->knownDelay|. We therefore, round (-32) in that direction. In
+ // the other direction, we don't have this situation, but might flush one
+ // partition too little. This can cause non-causality, which should be
+ // investigated. Maybe, allow for a non-symmetric rounding, like -16.
+ int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
+ int moved_elements = 0;
+
+ // TODO(bjornv): Change the near-end buffer handling to be the same as for
+ // far-end, that is, with a near_pre_buf.
+ // Buffer the near-end frame.
+ WebRtc_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
+ // For H band
+ if (aec->sampFreq == 32000) {
+ WebRtc_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
+ }
+
+ // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
+ // have enough far-end data for that by stuffing the buffer if the
+ // |system_delay| indicates others.
+ if (aec->system_delay < FRAME_LEN) {
+ // We don't have enough data so we rewind 10 ms.
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, -(aec->mult + 1));
+ aec->system_delay -= WebRtc_MoveReadPtr(aec->far_buf, -(aec->mult + 1)) *
+ PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, -(aec->mult + 1));
+#endif
+ }
+
+ // 2) Compensate for a possible change in the system delay.
+
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
+ moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
+ aec->knownDelay -= moved_elements * PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+#endif
+
+ // 4) Process as many blocks as possible.
+ while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
+ ProcessBlock(aec);
+ }
+
+ // 5) Update system delay with respect to the entire frame.
+ aec->system_delay -= FRAME_LEN;
+}
+
+static void ProcessBlock(aec_t* aec) {
+ int i;
+ float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
+ float scale;
+
+ float fft[PART_LEN2];
+ float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
+ float df[2][PART_LEN1];
+ float far_spectrum = 0.0f;
+ float near_spectrum = 0.0f;
+ float abs_far_spectrum[PART_LEN1];
+ float abs_near_spectrum[PART_LEN1];
+
+ const float gPow[2] = {0.9f, 0.1f};
+
+ // Noise estimate constants.
+ const int noiseInitBlocks = 500 * aec->mult;
+ const float step = 0.1f;
+ const float ramp = 1.0002f;
+ const float gInitNoise[2] = {0.999f, 0.001f};
+
+ int16_t nearend[PART_LEN];
+ int16_t* nearend_ptr = NULL;
+ int16_t output[PART_LEN];
+ int16_t outputH[PART_LEN];
+
+ float* xf_ptr = NULL;
+
+ memset(dH, 0, sizeof(dH));
+ if (aec->sampFreq == 32000) {
+ // Get the upper band first so we can reuse |nearend|.
+ WebRtc_ReadBuffer(aec->nearFrBufH,
+ (void**) &nearend_ptr,
+ nearend,
+ PART_LEN);
+ for (i = 0; i < PART_LEN; i++) {
+ dH[i] = (float) (nearend_ptr[i]);
+ }
+ memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
+ }
+ WebRtc_ReadBuffer(aec->nearFrBuf, (void**) &nearend_ptr, nearend, PART_LEN);
+
+ // ---------- Ooura fft ----------
+ // Concatenate old and new nearend blocks.
+ for (i = 0; i < PART_LEN; i++) {
+ d[i] = (float) (nearend_ptr[i]);
+ }
+ memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t farend[PART_LEN];
+ int16_t* farend_ptr = NULL;
+ WebRtc_ReadBuffer(aec->far_time_buf, (void**) &farend_ptr, farend, 1);
+ fwrite(farend_ptr, sizeof(int16_t), PART_LEN, aec->farFile);
+ fwrite(nearend_ptr, sizeof(int16_t), PART_LEN, aec->nearFile);
+ }
+#endif
+
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf) > 0);
+ WebRtc_ReadBuffer(aec->far_buf, (void**) &xf_ptr, &xf[0][0], 1);
+
+ // Near fft
+ memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, df, 0);
+
+ // Power smoothing
+ for (i = 0; i < PART_LEN1; i++) {
+ far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
+ (xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
+ aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART * far_spectrum;
+ // Calculate absolute spectra
+ abs_far_spectrum[i] = sqrtf(far_spectrum);
+
+ near_spectrum = df[0][i] * df[0][i] + df[1][i] * df[1][i];
+ aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] * near_spectrum;
+ // Calculate absolute spectra
+ abs_near_spectrum[i] = sqrtf(near_spectrum);
+ }
+
+ // Estimate noise power. Wait until dPow is more stable.
+ if (aec->noiseEstCtr > 50) {
+ for (i = 0; i < PART_LEN1; i++) {
+ if (aec->dPow[i] < aec->dMinPow[i]) {
+ aec->dMinPow[i] = (aec->dPow[i] + step * (aec->dMinPow[i] -
+ aec->dPow[i])) * ramp;
+ }
+ else {
+ aec->dMinPow[i] *= ramp;
+ }
+ }
+ }
+
+ // Smooth increasing noise power from zero at the start,
+ // to avoid a sudden burst of comfort noise.
+ if (aec->noiseEstCtr < noiseInitBlocks) {
+ aec->noiseEstCtr++;
+ for (i = 0; i < PART_LEN1; i++) {
+ if (aec->dMinPow[i] > aec->dInitMinPow[i]) {
+ aec->dInitMinPow[i] = gInitNoise[0] * aec->dInitMinPow[i] +
+ gInitNoise[1] * aec->dMinPow[i];
+ }
+ else {
+ aec->dInitMinPow[i] = aec->dMinPow[i];
+ }
+ }
+ aec->noisePow = aec->dInitMinPow;
+ }
+ else {
+ aec->noisePow = aec->dMinPow;
+ }
+
+ // Block wise delay estimation used for logging
+ if (aec->delay_logging_enabled) {
+ int delay_estimate = 0;
+ // Estimate the delay
+ delay_estimate = WebRtc_DelayEstimatorProcessFloat(aec->delay_estimator,
+ abs_far_spectrum,
+ abs_near_spectrum,
+ PART_LEN1);
+ if (delay_estimate >= 0) {
+ // Update delay estimate buffer.
+ aec->delay_histogram[delay_estimate]++;
+ }
+ }
+
+ // Update the xfBuf block position.
+ aec->xfBufBlockPos--;
+ if (aec->xfBufBlockPos == -1) {
+ aec->xfBufBlockPos = NR_PART - 1;
+ }
+
+ // Buffer xf
+ memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1, xf_ptr,
+ sizeof(float) * PART_LEN1);
+ memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, &xf_ptr[PART_LEN1],
+ sizeof(float) * PART_LEN1);
+
+ memset(yf[0], 0, sizeof(float) * (PART_LEN1 * 2));
+
+ // Filter far
+ WebRtcAec_FilterFar(aec, yf);
+
+ // Inverse fft to obtain echo estimate and error.
+ fft[0] = yf[0][0];
+ fft[1] = yf[0][PART_LEN];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2 * i] = yf[0][i];
+ fft[2 * i + 1] = yf[1][i];
+ }
+ aec_rdft_inverse_128(fft);
+
+ scale = 2.0f / PART_LEN2;
+ for (i = 0; i < PART_LEN; i++) {
+ y[i] = fft[PART_LEN + i] * scale; // fft scaling
+ }
+
+ for (i = 0; i < PART_LEN; i++) {
+ e[i] = d[i] - y[i];
+ }
+
+ // Error fft
+ memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
+ memset(fft, 0, sizeof(float) * PART_LEN);
+ memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
+ // TODO(bjornv): Change to use TimeToFrequency().
+ aec_rdft_forward_128(fft);
+
+ ef[1][0] = 0;
+ ef[1][PART_LEN] = 0;
+ ef[0][0] = fft[0];
+ ef[0][PART_LEN] = fft[1];
+ for (i = 1; i < PART_LEN; i++) {
+ ef[0][i] = fft[2 * i];
+ ef[1][i] = fft[2 * i + 1];
+ }
+
+ if (aec->metricsMode == 1) {
+ // Note that the first PART_LEN samples in fft (before transformation) are
+ // zero. Hence, the scaling by two in UpdateLevel() should not be
+ // performed. That scaling is taken care of in UpdateMetrics() instead.
+ UpdateLevel(&aec->linoutlevel, ef);
+ }
+
+ // Scale error signal inversely with far power.
+ WebRtcAec_ScaleErrorSignal(aec, ef);
+ WebRtcAec_FilterAdaptation(aec, fft, ef);
+ NonLinearProcessing(aec, output, outputH);
+
+ if (aec->metricsMode == 1) {
+ // Update power levels and echo metrics
+ UpdateLevel(&aec->farlevel, (float (*)[PART_LEN1]) xf_ptr);
+ UpdateLevel(&aec->nearlevel, df);
+ UpdateMetrics(aec);
+ }
+
+ // Store the output block.
+ WebRtc_WriteBuffer(aec->outFrBuf, output, PART_LEN);
+ // For H band
+ if (aec->sampFreq == 32000) {
+ WebRtc_WriteBuffer(aec->outFrBufH, outputH, PART_LEN);
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t eInt16[PART_LEN];
+ for (i = 0; i < PART_LEN; i++) {
+ eInt16[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
+ WEBRTC_SPL_WORD16_MIN);
+ }
+
+ fwrite(eInt16, sizeof(int16_t), PART_LEN, aec->outLinearFile);
+ fwrite(output, sizeof(int16_t), PART_LEN, aec->outFile);
+ }
+#endif
+}
+
+static void NonLinearProcessing(aec_t *aec, short *output, short *outputH)
+{
+ float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1];
+ complex_t comfortNoiseHband[PART_LEN1];
+ float fft[PART_LEN2];
+ float scale, dtmp;
+ float nlpGainHband;
+ int i, j, pos;
+
+ // Coherence and non-linear filter
+ float cohde[PART_LEN1], cohxd[PART_LEN1];
+ float hNlDeAvg, hNlXdAvg;
+ float hNl[PART_LEN1];
+ float hNlPref[PREF_BAND_SIZE];
+ float hNlFb = 0, hNlFbLow = 0;
+ const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
+ const int prefBandSize = PREF_BAND_SIZE / aec->mult;
+ const int minPrefBand = 4 / aec->mult;
+
+ // Near and error power sums
+ float sdSum = 0, seSum = 0;
+
+ // Power estimate smoothing coefficients
+ const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
+ const float *ptrGCoh = gCoh[aec->mult - 1];
+
+ // Filter energy
+ float wfEnMax = 0, wfEn = 0;
+ const int delayEstInterval = 10 * aec->mult;
+
+ float* xfw_ptr = NULL;
+
+ aec->delayEstCtr++;
+ if (aec->delayEstCtr == delayEstInterval) {
+ aec->delayEstCtr = 0;
+ }
+
+ // initialize comfort noise for H band
+ memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
+ nlpGainHband = (float)0.0;
+ dtmp = (float)0.0;
+
+ // Measure energy in each filter partition to determine delay.
+ // TODO: Spread by computing one partition per block?
+ if (aec->delayEstCtr == 0) {
+ wfEnMax = 0;
+ aec->delayIdx = 0;
+ for (i = 0; i < NR_PART; i++) {
+ pos = i * PART_LEN1;
+ wfEn = 0;
+ for (j = 0; j < PART_LEN1; j++) {
+ wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
+ aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
+ }
+
+ if (wfEn > wfEnMax) {
+ wfEnMax = wfEn;
+ aec->delayIdx = i;
+ }
+ }
+ }
+
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
+ // NLP
+ WebRtc_ReadBuffer(aec->far_buf_windowed, (void**) &xfw_ptr, &xfw[0][0], 1);
+
+ // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
+ // |xfwBuf|.
+ // Buffer far.
+ memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
+
+ // Use delayed far.
+ memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1, sizeof(xfw));
+
+ // Windowed near fft
+ for (i = 0; i < PART_LEN; i++) {
+ fft[i] = aec->dBuf[i] * sqrtHanning[i];
+ fft[PART_LEN + i] = aec->dBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
+ }
+ aec_rdft_forward_128(fft);
+
+ dfw[1][0] = 0;
+ dfw[1][PART_LEN] = 0;
+ dfw[0][0] = fft[0];
+ dfw[0][PART_LEN] = fft[1];
+ for (i = 1; i < PART_LEN; i++) {
+ dfw[0][i] = fft[2 * i];
+ dfw[1][i] = fft[2 * i + 1];
+ }
+
+ // Windowed error fft
+ for (i = 0; i < PART_LEN; i++) {
+ fft[i] = aec->eBuf[i] * sqrtHanning[i];
+ fft[PART_LEN + i] = aec->eBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
+ }
+ aec_rdft_forward_128(fft);
+ efw[1][0] = 0;
+ efw[1][PART_LEN] = 0;
+ efw[0][0] = fft[0];
+ efw[0][PART_LEN] = fft[1];
+ for (i = 1; i < PART_LEN; i++) {
+ efw[0][i] = fft[2 * i];
+ efw[1][i] = fft[2 * i + 1];
+ }
+
+ // Smoothed PSD
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sd[i] = ptrGCoh[0] * aec->sd[i] + ptrGCoh[1] *
+ (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
+ aec->se[i] = ptrGCoh[0] * aec->se[i] + ptrGCoh[1] *
+ (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
+ // We threshold here to protect against the ill-effects of a zero farend.
+ // The threshold is not arbitrarily chosen, but balances protection and
+ // adverse interaction with the algorithm's tuning.
+ // TODO: investigate further why this is so sensitive.
+ aec->sx[i] = ptrGCoh[0] * aec->sx[i] + ptrGCoh[1] *
+ WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i], 15);
+
+ aec->sde[i][0] = ptrGCoh[0] * aec->sde[i][0] + ptrGCoh[1] *
+ (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
+ aec->sde[i][1] = ptrGCoh[0] * aec->sde[i][1] + ptrGCoh[1] *
+ (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
+
+ aec->sxd[i][0] = ptrGCoh[0] * aec->sxd[i][0] + ptrGCoh[1] *
+ (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
+ aec->sxd[i][1] = ptrGCoh[0] * aec->sxd[i][1] + ptrGCoh[1] *
+ (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
+
+ sdSum += aec->sd[i];
+ seSum += aec->se[i];
+ }
+
+ // Divergent filter safeguard.
+ if (aec->divergeState == 0) {
+ if (seSum > sdSum) {
+ aec->divergeState = 1;
+ }
+ }
+ else {
+ if (seSum * 1.05f < sdSum) {
+ aec->divergeState = 0;
+ }
+ }
+
+ if (aec->divergeState == 1) {
+ memcpy(efw, dfw, sizeof(efw));
+ }
+
+ // Reset if error is significantly larger than nearend (13 dB).
+ if (seSum > (19.95f * sdSum)) {
+ memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+ }
+
+ // Subband coherence
+ for (i = 0; i < PART_LEN1; i++) {
+ cohde[i] = (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
+ (aec->sd[i] * aec->se[i] + 1e-10f);
+ cohxd[i] = (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
+ (aec->sx[i] * aec->sd[i] + 1e-10f);
+ }
+
+ hNlXdAvg = 0;
+ for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+ hNlXdAvg += cohxd[i];
+ }
+ hNlXdAvg /= prefBandSize;
+ hNlXdAvg = 1 - hNlXdAvg;
+
+ hNlDeAvg = 0;
+ for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+ hNlDeAvg += cohde[i];
+ }
+ hNlDeAvg /= prefBandSize;
+
+ if (hNlXdAvg < 0.75f && hNlXdAvg < aec->hNlXdAvgMin) {
+ aec->hNlXdAvgMin = hNlXdAvg;
+ }
+
+ if (hNlDeAvg > 0.98f && hNlXdAvg > 0.9f) {
+ aec->stNearState = 1;
+ }
+ else if (hNlDeAvg < 0.95f || hNlXdAvg < 0.8f) {
+ aec->stNearState = 0;
+ }
+
+ if (aec->hNlXdAvgMin == 1) {
+ aec->echoState = 0;
+ aec->overDrive = aec->minOverDrive;
+
+ if (aec->stNearState == 1) {
+ memcpy(hNl, cohde, sizeof(hNl));
+ hNlFb = hNlDeAvg;
+ hNlFbLow = hNlDeAvg;
+ }
+ else {
+ for (i = 0; i < PART_LEN1; i++) {
+ hNl[i] = 1 - cohxd[i];
+ }
+ hNlFb = hNlXdAvg;
+ hNlFbLow = hNlXdAvg;
+ }
+ }
+ else {
+
+ if (aec->stNearState == 1) {
+ aec->echoState = 0;
+ memcpy(hNl, cohde, sizeof(hNl));
+ hNlFb = hNlDeAvg;
+ hNlFbLow = hNlDeAvg;
+ }
+ else {
+ aec->echoState = 1;
+ for (i = 0; i < PART_LEN1; i++) {
+ hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
+ }
+
+ // Select an order statistic from the preferred bands.
+ // TODO: Using quicksort now, but a selection algorithm may be preferred.
+ memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
+ qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
+ hNlFb = hNlPref[(int)floor(prefBandQuant * (prefBandSize - 1))];
+ hNlFbLow = hNlPref[(int)floor(prefBandQuantLow * (prefBandSize - 1))];
+ }
+ }
+
+ // Track the local filter minimum to determine suppression overdrive.
+ if (hNlFbLow < 0.6f && hNlFbLow < aec->hNlFbLocalMin) {
+ aec->hNlFbLocalMin = hNlFbLow;
+ aec->hNlFbMin = hNlFbLow;
+ aec->hNlNewMin = 1;
+ aec->hNlMinCtr = 0;
+ }
+ aec->hNlFbLocalMin = WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1);
+ aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1);
+
+ if (aec->hNlNewMin == 1) {
+ aec->hNlMinCtr++;
+ }
+ if (aec->hNlMinCtr == 2) {
+ aec->hNlNewMin = 0;
+ aec->hNlMinCtr = 0;
+ aec->overDrive = WEBRTC_SPL_MAX(aec->targetSupp /
+ ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f), aec->minOverDrive);
+ }
+
+ // Smooth the overdrive.
+ if (aec->overDrive < aec->overDriveSm) {
+ aec->overDriveSm = 0.99f * aec->overDriveSm + 0.01f * aec->overDrive;
+ }
+ else {
+ aec->overDriveSm = 0.9f * aec->overDriveSm + 0.1f * aec->overDrive;
+ }
+
+ WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
+
+ // Add comfort noise.
+ ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
+
+ // TODO(bjornv): Investigate how to take the windowing below into account if
+ // needed.
+ if (aec->metricsMode == 1) {
+ // Note that we have a scaling by two in the time domain |eBuf|.
+ // In addition the time domain signal is windowed before transformation,
+ // losing half the energy on the average. We take care of the first
+ // scaling only in UpdateMetrics().
+ UpdateLevel(&aec->nlpoutlevel, efw);
+ }
+ // Inverse error fft.
+ fft[0] = efw[0][0];
+ fft[1] = efw[0][PART_LEN];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2*i] = efw[0][i];
+ // Sign change required by Ooura fft.
+ fft[2*i + 1] = -efw[1][i];
+ }
+ aec_rdft_inverse_128(fft);
+
+ // Overlap and add to obtain output.
+ scale = 2.0f / PART_LEN2;
+ for (i = 0; i < PART_LEN; i++) {
+ fft[i] *= scale; // fft scaling
+ fft[i] = fft[i]*sqrtHanning[i] + aec->outBuf[i];
+
+ // Saturation protection
+ output[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fft[i],
+ WEBRTC_SPL_WORD16_MIN);
+
+ fft[PART_LEN + i] *= scale; // fft scaling
+ aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i];
+ }
+
+ // For H band
+ if (aec->sampFreq == 32000) {
+
+ // H band gain
+ // average nlp over low band: average over second half of freq spectrum
+ // (4->8khz)
+ GetHighbandGain(hNl, &nlpGainHband);
+
+ // Inverse comfort_noise
+ if (flagHbandCn == 1) {
+ fft[0] = comfortNoiseHband[0][0];
+ fft[1] = comfortNoiseHband[PART_LEN][0];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2*i] = comfortNoiseHband[i][0];
+ fft[2*i + 1] = comfortNoiseHband[i][1];
+ }
+ aec_rdft_inverse_128(fft);
+ scale = 2.0f / PART_LEN2;
+ }
+
+ // compute gain factor
+ for (i = 0; i < PART_LEN; i++) {
+ dtmp = (float)aec->dBufH[i];
+ dtmp = (float)dtmp * nlpGainHband; // for variable gain
+
+ // add some comfort noise where Hband is attenuated
+ if (flagHbandCn == 1) {
+ fft[i] *= scale; // fft scaling
+ dtmp += cnScaleHband * fft[i];
+ }
+
+ // Saturation protection
+ outputH[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, dtmp,
+ WEBRTC_SPL_WORD16_MIN);
+ }
+ }
+
+ // Copy the current block to the old position.
+ memcpy(aec->dBuf, aec->dBuf + PART_LEN, sizeof(float) * PART_LEN);
+ memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
+
+ // Copy the current block to the old position for H band
+ if (aec->sampFreq == 32000) {
+ memcpy(aec->dBufH, aec->dBufH + PART_LEN, sizeof(float) * PART_LEN);
+ }
+
+ memmove(aec->xfwBuf + PART_LEN1, aec->xfwBuf, sizeof(aec->xfwBuf) -
+ sizeof(complex_t) * PART_LEN1);
+}
+
+static void GetHighbandGain(const float *lambda, float *nlpGainHband)
+{
+ int i;
+
+ nlpGainHband[0] = (float)0.0;
+ for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
+ nlpGainHband[0] += lambda[i];
+ }
+ nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
+}
+
+static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
+ complex_t *comfortNoiseHband, const float *noisePow, const float *lambda)
+{
+ int i, num;
+ float rand[PART_LEN];
+ float noise, noiseAvg, tmp, tmpAvg;
+ WebRtc_Word16 randW16[PART_LEN];
+ complex_t u[PART_LEN1];
+
+ const float pi2 = 6.28318530717959f;
+
+ // Generate a uniform random array on [0 1]
+ WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
+ for (i = 0; i < PART_LEN; i++) {
+ rand[i] = ((float)randW16[i]) / 32768;
+ }
+
+ // Reject LF noise
+ u[0][0] = 0;
+ u[0][1] = 0;
+ for (i = 1; i < PART_LEN1; i++) {
+ tmp = pi2 * rand[i - 1];
+
+ noise = sqrtf(noisePow[i]);
+ u[i][0] = noise * (float)cos(tmp);
+ u[i][1] = -noise * (float)sin(tmp);
+ }
+ u[PART_LEN][1] = 0;
+
+ for (i = 0; i < PART_LEN1; i++) {
+ // This is the proper weighting to match the background noise power
+ tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
+ //tmp = 1 - lambda[i];
+ efw[0][i] += tmp * u[i][0];
+ efw[1][i] += tmp * u[i][1];
+ }
+
+ // For H band comfort noise
+ // TODO: don't compute noise and "tmp" twice. Use the previous results.
+ noiseAvg = 0.0;
+ tmpAvg = 0.0;
+ num = 0;
+ if (aec->sampFreq == 32000 && flagHbandCn == 1) {
+
+ // average noise scale
+ // average over second half of freq spectrum (i.e., 4->8khz)
+ // TODO: we shouldn't need num. We know how many elements we're summing.
+ for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
+ num++;
+ noiseAvg += sqrtf(noisePow[i]);
+ }
+ noiseAvg /= (float)num;
+
+ // average nlp scale
+ // average over second half of freq spectrum (i.e., 4->8khz)
+ // TODO: we shouldn't need num. We know how many elements we're summing.
+ num = 0;
+ for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
+ num++;
+ tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
+ }
+ tmpAvg /= (float)num;
+
+ // Use average noise for H band
+ // TODO: we should probably have a new random vector here.
+ // Reject LF noise
+ u[0][0] = 0;
+ u[0][1] = 0;
+ for (i = 1; i < PART_LEN1; i++) {
+ tmp = pi2 * rand[i - 1];
+
+ // Use average noise for H band
+ u[i][0] = noiseAvg * (float)cos(tmp);
+ u[i][1] = -noiseAvg * (float)sin(tmp);
+ }
+ u[PART_LEN][1] = 0;
+
+ for (i = 0; i < PART_LEN1; i++) {
+ // Use average NLP weight for H band
+ comfortNoiseHband[i][0] = tmpAvg * u[i][0];
+ comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+ }
+ }
+}
+
+static void WebRtcAec_InitLevel(power_level_t *level)
+{
+ const float bigFloat = 1E17f;
+
+ level->averagelevel = 0;
+ level->framelevel = 0;
+ level->minlevel = bigFloat;
+ level->frsum = 0;
+ level->sfrsum = 0;
+ level->frcounter = 0;
+ level->sfrcounter = 0;
+}
+
+static void WebRtcAec_InitStats(stats_t *stats)
+{
+ stats->instant = offsetLevel;
+ stats->average = offsetLevel;
+ stats->max = offsetLevel;
+ stats->min = offsetLevel * (-1);
+ stats->sum = 0;
+ stats->hisum = 0;
+ stats->himean = offsetLevel;
+ stats->counter = 0;
+ stats->hicounter = 0;
+}
+
+static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]) {
+ // Do the energy calculation in the frequency domain. The FFT is performed on
+ // a segment of PART_LEN2 samples due to overlap, but we only want the energy
+ // of half that data (the last PART_LEN samples). Parseval's relation states
+ // that the energy is preserved according to
+ //
+ // \sum_{n=0}^{N-1} |x(n)|^2 = 1/N * \sum_{n=0}^{N-1} |X(n)|^2
+ // = ENERGY,
+ //
+ // where N = PART_LEN2. Since we are only interested in calculating the energy
+ // for the last PART_LEN samples we approximate by calculating ENERGY and
+ // divide by 2,
+ //
+ // \sum_{n=N/2}^{N-1} |x(n)|^2 ~= ENERGY / 2
+ //
+ // Since we deal with real valued time domain signals we only store frequency
+ // bins [0, PART_LEN], which is what |in| consists of. To calculate ENERGY we
+ // need to add the contribution from the missing part in
+ // [PART_LEN+1, PART_LEN2-1]. These values are, up to a phase shift, identical
+ // with the values in [1, PART_LEN-1], hence multiply those values by 2. This
+ // is the values in the for loop below, but multiplication by 2 and division
+ // by 2 cancel.
+
+ // TODO(bjornv): Investigate reusing energy calculations performed at other
+ // places in the code.
+ int k = 1;
+ // Imaginary parts are zero at end points and left out of the calculation.
+ float energy = (in[0][0] * in[0][0]) / 2;
+ energy += (in[0][PART_LEN] * in[0][PART_LEN]) / 2;
+
+ for (k = 1; k < PART_LEN; k++) {
+ energy += (in[0][k] * in[0][k] + in[1][k] * in[1][k]);
+ }
+ energy /= PART_LEN2;
+
+ level->sfrsum += energy;
+ level->sfrcounter++;
+
+ if (level->sfrcounter > subCountLen) {
+ level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
+ level->sfrsum = 0;
+ level->sfrcounter = 0;
+ if (level->framelevel > 0) {
+ if (level->framelevel < level->minlevel) {
+ level->minlevel = level->framelevel; // New minimum.
+ } else {
+ level->minlevel *= (1 + 0.001f); // Small increase.
+ }
+ }
+ level->frcounter++;
+ level->frsum += level->framelevel;
+ if (level->frcounter > countLen) {
+ level->averagelevel = level->frsum / countLen;
+ level->frsum = 0;
+ level->frcounter = 0;
+ }
+ }
+}
+
+static void UpdateMetrics(aec_t *aec)
+{
+ float dtmp, dtmp2;
+
+ const float actThresholdNoisy = 8.0f;
+ const float actThresholdClean = 40.0f;
+ const float safety = 0.99995f;
+ const float noisyPower = 300000.0f;
+
+ float actThreshold;
+ float echo, suppressedEcho;
+
+ if (aec->echoState) { // Check if echo is likely present
+ aec->stateCounter++;
+ }
+
+ if (aec->farlevel.frcounter == 0) {
+
+ if (aec->farlevel.minlevel < noisyPower) {
+ actThreshold = actThresholdClean;
+ }
+ else {
+ actThreshold = actThresholdNoisy;
+ }
+
+ if ((aec->stateCounter > (0.5f * countLen * subCountLen))
+ && (aec->farlevel.sfrcounter == 0)
+
+ // Estimate in active far-end segments only
+ && (aec->farlevel.averagelevel > (actThreshold * aec->farlevel.minlevel))
+ ) {
+
+ // Subtract noise power
+ echo = aec->nearlevel.averagelevel - safety * aec->nearlevel.minlevel;
+
+ // ERL
+ dtmp = 10 * (float)log10(aec->farlevel.averagelevel /
+ aec->nearlevel.averagelevel + 1e-10f);
+ dtmp2 = 10 * (float)log10(aec->farlevel.averagelevel / echo + 1e-10f);
+
+ aec->erl.instant = dtmp;
+ if (dtmp > aec->erl.max) {
+ aec->erl.max = dtmp;
+ }
+
+ if (dtmp < aec->erl.min) {
+ aec->erl.min = dtmp;
+ }
+
+ aec->erl.counter++;
+ aec->erl.sum += dtmp;
+ aec->erl.average = aec->erl.sum / aec->erl.counter;
+
+ // Upper mean
+ if (dtmp > aec->erl.average) {
+ aec->erl.hicounter++;
+ aec->erl.hisum += dtmp;
+ aec->erl.himean = aec->erl.hisum / aec->erl.hicounter;
+ }
+
+ // A_NLP
+ dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
+ (2 * aec->linoutlevel.averagelevel) + 1e-10f);
+
+ // subtract noise power
+ suppressedEcho = 2 * (aec->linoutlevel.averagelevel -
+ safety * aec->linoutlevel.minlevel);
+
+ dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
+
+ aec->aNlp.instant = dtmp2;
+ if (dtmp > aec->aNlp.max) {
+ aec->aNlp.max = dtmp;
+ }
+
+ if (dtmp < aec->aNlp.min) {
+ aec->aNlp.min = dtmp;
+ }
+
+ aec->aNlp.counter++;
+ aec->aNlp.sum += dtmp;
+ aec->aNlp.average = aec->aNlp.sum / aec->aNlp.counter;
+
+ // Upper mean
+ if (dtmp > aec->aNlp.average) {
+ aec->aNlp.hicounter++;
+ aec->aNlp.hisum += dtmp;
+ aec->aNlp.himean = aec->aNlp.hisum / aec->aNlp.hicounter;
+ }
+
+ // ERLE
+
+ // subtract noise power
+ suppressedEcho = 2 * (aec->nlpoutlevel.averagelevel -
+ safety * aec->nlpoutlevel.minlevel);
+
+ dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
+ (2 * aec->nlpoutlevel.averagelevel) + 1e-10f);
+ dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
+
+ dtmp = dtmp2;
+ aec->erle.instant = dtmp;
+ if (dtmp > aec->erle.max) {
+ aec->erle.max = dtmp;
+ }
+
+ if (dtmp < aec->erle.min) {
+ aec->erle.min = dtmp;
+ }
+
+ aec->erle.counter++;
+ aec->erle.sum += dtmp;
+ aec->erle.average = aec->erle.sum / aec->erle.counter;
+
+ // Upper mean
+ if (dtmp > aec->erle.average) {
+ aec->erle.hicounter++;
+ aec->erle.hisum += dtmp;
+ aec->erle.himean = aec->erle.hisum / aec->erle.hicounter;
+ }
+ }
+
+ aec->stateCounter = 0;
+ }
+}
+
+static void TimeToFrequency(float time_data[PART_LEN2],
+ float freq_data[2][PART_LEN1],
+ int window) {
+ int i = 0;
+
+ // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
+ if (window) {
+ for (i = 0; i < PART_LEN; i++) {
+ time_data[i] *= sqrtHanning[i];
+ time_data[PART_LEN + i] *= sqrtHanning[PART_LEN - i];
+ }
+ }
+
+ aec_rdft_forward_128(time_data);
+ // Reorder.
+ freq_data[1][0] = 0;
+ freq_data[1][PART_LEN] = 0;
+ freq_data[0][0] = time_data[0];
+ freq_data[0][PART_LEN] = time_data[1];
+ for (i = 1; i < PART_LEN; i++) {
+ freq_data[0][i] = time_data[2 * i];
+ freq_data[1][i] = time_data[2 * i + 1];
+ }
+}
diff --git a/trunk/src/modules/audio_processing/aec/aec_core.h b/trunk/src/modules/audio_processing/aec/aec_core.h
new file mode 100644
index 0000000..d326a68
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_core.h
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Specifies the interface for the AEC core.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#endif
+
+#include "typedefs.h"
+
+#define FRAME_LEN 80
+#define PART_LEN 64 // Length of partition
+#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
+#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
+#define NR_PART 12 // Number of partitions in filter.
+#define PREF_BAND_SIZE 24
+
+// Delay estimator constants, used for logging.
+enum { kMaxDelayBlocks = 60 };
+enum { kLookaheadBlocks = 15 };
+enum { kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks };
+
+typedef float complex_t[2];
+// For performance reasons, some arrays of complex numbers are replaced by twice
+// as long arrays of float, all the real parts followed by all the imaginary
+// ones (complex_t[SIZE] -> float[2][SIZE]). This allows SIMD optimizations and
+// is better than two arrays (one for the real parts and one for the imaginary
+// parts) as this other way would require two pointers instead of one and cause
+// extra register spilling. This also allows the offsets to be calculated at
+// compile time.
+
+// Metrics
+enum {offsetLevel = -100};
+
+typedef struct {
+ float sfrsum;
+ int sfrcounter;
+ float framelevel;
+ float frsum;
+ int frcounter;
+ float minlevel;
+ float averagelevel;
+} power_level_t;
+
+typedef struct {
+ float instant;
+ float average;
+ float min;
+ float max;
+ float sum;
+ float hisum;
+ float himean;
+ int counter;
+ int hicounter;
+} stats_t;
+
+typedef struct {
+ int farBufWritePos, farBufReadPos;
+
+ int knownDelay;
+ int inSamples, outSamples;
+ int delayEstCtr;
+
+ void *nearFrBuf, *outFrBuf;
+
+ void *nearFrBufH;
+ void *outFrBufH;
+
+ float dBuf[PART_LEN2]; // nearend
+ float eBuf[PART_LEN2]; // error
+
+ float dBufH[PART_LEN2]; // nearend
+
+ float xPow[PART_LEN1];
+ float dPow[PART_LEN1];
+ float dMinPow[PART_LEN1];
+ float dInitMinPow[PART_LEN1];
+ float *noisePow;
+
+ float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
+ float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
+ complex_t sde[PART_LEN1]; // cross-psd of nearend and error
+ complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
+ complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
+
+ float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
+ float hNs[PART_LEN1];
+ float hNlFbMin, hNlFbLocalMin;
+ float hNlXdAvgMin;
+ int hNlNewMin, hNlMinCtr;
+ float overDrive, overDriveSm;
+ float targetSupp, minOverDrive;
+ float outBuf[PART_LEN];
+ int delayIdx;
+
+ short stNearState, echoState;
+ short divergeState;
+
+ int xfBufBlockPos;
+
+ void* far_buf;
+ void* far_buf_windowed;
+ int system_delay; // Current system delay buffered in AEC.
+
+ int mult; // sampling frequency multiple
+ int sampFreq;
+ WebRtc_UWord32 seed;
+
+ float mu; // stepsize
+ float errThresh; // error threshold
+
+ int noiseEstCtr;
+
+ power_level_t farlevel;
+ power_level_t nearlevel;
+ power_level_t linoutlevel;
+ power_level_t nlpoutlevel;
+
+ int metricsMode;
+ int stateCounter;
+ stats_t erl;
+ stats_t erle;
+ stats_t aNlp;
+ stats_t rerl;
+
+ // Quantities to control H band scaling for SWB input
+ int freq_avg_ic; //initial bin for averaging nlp gain
+ int flag_Hband_cn; //for comfort noise
+ float cn_scale_Hband; //scale for comfort noise in H band
+
+ int delay_histogram[kHistorySizeBlocks];
+ int delay_logging_enabled;
+ void* delay_estimator;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ void* far_time_buf;
+ FILE *farFile;
+ FILE *nearFile;
+ FILE *outFile;
+ FILE *outLinearFile;
+#endif
+} aec_t;
+
+typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
+extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
+typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
+extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
+typedef void (*WebRtcAec_FilterAdaptation_t)
+ (aec_t *aec, float *fft, float ef[2][PART_LEN1]);
+extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
+typedef void (*WebRtcAec_OverdriveAndSuppress_t)
+ (aec_t *aec, float hNl[PART_LEN1], const float hNlFb, float efw[2][PART_LEN1]);
+extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
+
+int WebRtcAec_CreateAec(aec_t **aec);
+int WebRtcAec_FreeAec(aec_t *aec);
+int WebRtcAec_InitAec(aec_t *aec, int sampFreq);
+void WebRtcAec_InitAec_SSE2(void);
+
+void WebRtcAec_InitMetrics(aec_t *aec);
+void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend);
+void WebRtcAec_ProcessFrame(aec_t* aec,
+ const short *nearend,
+ const short *nearendH,
+ int knownDelay);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
diff --git a/trunk/src/modules/audio_processing/aec/aec_core_sse2.c b/trunk/src/modules/audio_processing/aec/aec_core_sse2.c
new file mode 100644
index 0000000..74a1c48
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_core_sse2.c
@@ -0,0 +1,415 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * The core AEC algorithm, SSE2 version of speed-critical functions.
+ */
+
+#include "aec_core.h"
+
+#include <emmintrin.h>
+#include <math.h>
+#include <string.h> // memset
+
+#include "aec_rdft.h"
+
+__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
+{
+ return aRe * bRe - aIm * bIm;
+}
+
+__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
+{
+ return aRe * bIm + aIm * bRe;
+}
+
+static void FilterFarSSE2(aec_t *aec, float yf[2][PART_LEN1])
+{
+ int i;
+ for (i = 0; i < NR_PART; i++) {
+ int j;
+ int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+ int pos = i * PART_LEN1;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART*(PART_LEN1);
+ }
+
+ // vectorized code (four at once)
+ for (j = 0; j + 3 < PART_LEN1; j += 4) {
+ const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
+ const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
+ const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
+ const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
+ const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
+ const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
+ const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
+ const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
+ const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
+ const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
+ const __m128 e = _mm_sub_ps(a, b);
+ const __m128 f = _mm_add_ps(c, d);
+ const __m128 g = _mm_add_ps(yf_re, e);
+ const __m128 h = _mm_add_ps(yf_im, f);
+ _mm_storeu_ps(&yf[0][j], g);
+ _mm_storeu_ps(&yf[1][j], h);
+ }
+ // scalar code for the remaining items.
+ for (; j < PART_LEN1; j++) {
+ yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
+ yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
+ }
+ }
+}
+
+static void ScaleErrorSignalSSE2(aec_t *aec, float ef[2][PART_LEN1])
+{
+ const __m128 k1e_10f = _mm_set1_ps(1e-10f);
+ const __m128 kThresh = _mm_set1_ps(aec->errThresh);
+ const __m128 kMu = _mm_set1_ps(aec->mu);
+
+ int i;
+ // vectorized code (four at once)
+ for (i = 0; i + 3 < PART_LEN1; i += 4) {
+ const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
+ const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
+ const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
+
+ const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
+ __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
+ __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
+ const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
+ const __m128 ef_im2 = _mm_mul_ps(ef_im, ef_im);
+ const __m128 ef_sum2 = _mm_add_ps(ef_re2, ef_im2);
+ const __m128 absEf = _mm_sqrt_ps(ef_sum2);
+ const __m128 bigger = _mm_cmpgt_ps(absEf, kThresh);
+ __m128 absEfPlus = _mm_add_ps(absEf, k1e_10f);
+ const __m128 absEfInv = _mm_div_ps(kThresh, absEfPlus);
+ __m128 ef_re_if = _mm_mul_ps(ef_re, absEfInv);
+ __m128 ef_im_if = _mm_mul_ps(ef_im, absEfInv);
+ ef_re_if = _mm_and_ps(bigger, ef_re_if);
+ ef_im_if = _mm_and_ps(bigger, ef_im_if);
+ ef_re = _mm_andnot_ps(bigger, ef_re);
+ ef_im = _mm_andnot_ps(bigger, ef_im);
+ ef_re = _mm_or_ps(ef_re, ef_re_if);
+ ef_im = _mm_or_ps(ef_im, ef_im_if);
+ ef_re = _mm_mul_ps(ef_re, kMu);
+ ef_im = _mm_mul_ps(ef_im, kMu);
+
+ _mm_storeu_ps(&ef[0][i], ef_re);
+ _mm_storeu_ps(&ef[1][i], ef_im);
+ }
+ // scalar code for the remaining items.
+ for (; i < (PART_LEN1); i++) {
+ float absEf;
+ ef[0][i] /= (aec->xPow[i] + 1e-10f);
+ ef[1][i] /= (aec->xPow[i] + 1e-10f);
+ absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+
+ if (absEf > aec->errThresh) {
+ absEf = aec->errThresh / (absEf + 1e-10f);
+ ef[0][i] *= absEf;
+ ef[1][i] *= absEf;
+ }
+
+ // Stepsize factor
+ ef[0][i] *= aec->mu;
+ ef[1][i] *= aec->mu;
+ }
+}
+
+static void FilterAdaptationSSE2(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
+ int i, j;
+ for (i = 0; i < NR_PART; i++) {
+ int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+ int pos = i * PART_LEN1;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART * PART_LEN1;
+ }
+
+ // Process the whole array...
+ for (j = 0; j < PART_LEN; j+= 4) {
+ // Load xfBuf and ef.
+ const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
+ const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
+ const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
+ const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
+ // Calculate the product of conjugate(xfBuf) by ef.
+ // re(conjugate(a) * b) = aRe * bRe + aIm * bIm
+ // im(conjugate(a) * b)= aRe * bIm - aIm * bRe
+ const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
+ const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
+ const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
+ const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
+ const __m128 e = _mm_add_ps(a, b);
+ const __m128 f = _mm_sub_ps(c, d);
+ // Interleave real and imaginary parts.
+ const __m128 g = _mm_unpacklo_ps(e, f);
+ const __m128 h = _mm_unpackhi_ps(e, f);
+ // Store
+ _mm_storeu_ps(&fft[2*j + 0], g);
+ _mm_storeu_ps(&fft[2*j + 4], h);
+ }
+ // ... and fixup the first imaginary entry.
+ fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
+ -aec->xfBuf[1][xPos + PART_LEN],
+ ef[0][PART_LEN], ef[1][PART_LEN]);
+
+ aec_rdft_inverse_128(fft);
+ memset(fft + PART_LEN, 0, sizeof(float)*PART_LEN);
+
+ // fft scaling
+ {
+ float scale = 2.0f / PART_LEN2;
+ const __m128 scale_ps = _mm_load_ps1(&scale);
+ for (j = 0; j < PART_LEN; j+=4) {
+ const __m128 fft_ps = _mm_loadu_ps(&fft[j]);
+ const __m128 fft_scale = _mm_mul_ps(fft_ps, scale_ps);
+ _mm_storeu_ps(&fft[j], fft_scale);
+ }
+ }
+ aec_rdft_forward_128(fft);
+
+ {
+ float wt1 = aec->wfBuf[1][pos];
+ aec->wfBuf[0][pos + PART_LEN] += fft[1];
+ for (j = 0; j < PART_LEN; j+= 4) {
+ __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
+ __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
+ const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
+ const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
+ const __m128 fft_re = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(2, 0, 2 ,0));
+ const __m128 fft_im = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3 ,1));
+ wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
+ wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
+ _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
+ _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
+ }
+ aec->wfBuf[1][pos] = wt1;
+ }
+ }
+}
+
+static __m128 mm_pow_ps(__m128 a, __m128 b)
+{
+ // a^b = exp2(b * log2(a))
+ // exp2(x) and log2(x) are calculated using polynomial approximations.
+ __m128 log2_a, b_log2_a, a_exp_b;
+
+ // Calculate log2(x), x = a.
+ {
+ // To calculate log2(x), we decompose x like this:
+ // x = y * 2^n
+ // n is an integer
+ // y is in the [1.0, 2.0) range
+ //
+ // log2(x) = log2(y) + n
+ // n can be evaluated by playing with float representation.
+ // log2(y) in a small range can be approximated, this code uses an order
+ // five polynomial approximation. The coefficients have been
+ // estimated with the Remez algorithm and the resulting
+ // polynomial has a maximum relative error of 0.00086%.
+
+ // Compute n.
+ // This is done by masking the exponent, shifting it into the top bit of
+ // the mantissa, putting eight into the biased exponent (to shift/
+ // compensate the fact that the exponent has been shifted in the top/
+ // fractional part and finally getting rid of the implicit leading one
+ // from the mantissa by substracting it out.
+ static const ALIGN16_BEG int float_exponent_mask[4] ALIGN16_END =
+ {0x7F800000, 0x7F800000, 0x7F800000, 0x7F800000};
+ static const ALIGN16_BEG int eight_biased_exponent[4] ALIGN16_END =
+ {0x43800000, 0x43800000, 0x43800000, 0x43800000};
+ static const ALIGN16_BEG int implicit_leading_one[4] ALIGN16_END =
+ {0x43BF8000, 0x43BF8000, 0x43BF8000, 0x43BF8000};
+ static const int shift_exponent_into_top_mantissa = 8;
+ const __m128 two_n = _mm_and_ps(a, *((__m128 *)float_exponent_mask));
+ const __m128 n_1 = _mm_castsi128_ps(_mm_srli_epi32(_mm_castps_si128(two_n),
+ shift_exponent_into_top_mantissa));
+ const __m128 n_0 = _mm_or_ps(n_1, *((__m128 *)eight_biased_exponent));
+ const __m128 n = _mm_sub_ps(n_0, *((__m128 *)implicit_leading_one));
+
+ // Compute y.
+ static const ALIGN16_BEG int mantissa_mask[4] ALIGN16_END =
+ {0x007FFFFF, 0x007FFFFF, 0x007FFFFF, 0x007FFFFF};
+ static const ALIGN16_BEG int zero_biased_exponent_is_one[4] ALIGN16_END =
+ {0x3F800000, 0x3F800000, 0x3F800000, 0x3F800000};
+ const __m128 mantissa = _mm_and_ps(a, *((__m128 *)mantissa_mask));
+ const __m128 y = _mm_or_ps(
+ mantissa, *((__m128 *)zero_biased_exponent_is_one));
+
+ // Approximate log2(y) ~= (y - 1) * pol5(y).
+ // pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0
+ static const ALIGN16_BEG float ALIGN16_END C5[4] =
+ {-3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f};
+ static const ALIGN16_BEG float ALIGN16_END C4[4] =
+ {3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f};
+ static const ALIGN16_BEG float ALIGN16_END C3[4] =
+ {-1.2315303f, -1.2315303f, -1.2315303f, -1.2315303f};
+ static const ALIGN16_BEG float ALIGN16_END C2[4] =
+ {2.5988452f, 2.5988452f, 2.5988452f, 2.5988452f};
+ static const ALIGN16_BEG float ALIGN16_END C1[4] =
+ {-3.3241990f, -3.3241990f, -3.3241990f, -3.3241990f};
+ static const ALIGN16_BEG float ALIGN16_END C0[4] =
+ {3.1157899f, 3.1157899f, 3.1157899f, 3.1157899f};
+ const __m128 pol5_y_0 = _mm_mul_ps(y, *((__m128 *)C5));
+ const __m128 pol5_y_1 = _mm_add_ps(pol5_y_0, *((__m128 *)C4));
+ const __m128 pol5_y_2 = _mm_mul_ps(pol5_y_1, y);
+ const __m128 pol5_y_3 = _mm_add_ps(pol5_y_2, *((__m128 *)C3));
+ const __m128 pol5_y_4 = _mm_mul_ps(pol5_y_3, y);
+ const __m128 pol5_y_5 = _mm_add_ps(pol5_y_4, *((__m128 *)C2));
+ const __m128 pol5_y_6 = _mm_mul_ps(pol5_y_5, y);
+ const __m128 pol5_y_7 = _mm_add_ps(pol5_y_6, *((__m128 *)C1));
+ const __m128 pol5_y_8 = _mm_mul_ps(pol5_y_7, y);
+ const __m128 pol5_y = _mm_add_ps(pol5_y_8, *((__m128 *)C0));
+ const __m128 y_minus_one = _mm_sub_ps(
+ y, *((__m128 *)zero_biased_exponent_is_one));
+ const __m128 log2_y = _mm_mul_ps(y_minus_one , pol5_y);
+
+ // Combine parts.
+ log2_a = _mm_add_ps(n, log2_y);
+ }
+
+ // b * log2(a)
+ b_log2_a = _mm_mul_ps(b, log2_a);
+
+ // Calculate exp2(x), x = b * log2(a).
+ {
+ // To calculate 2^x, we decompose x like this:
+ // x = n + y
+ // n is an integer, the value of x - 0.5 rounded down, therefore
+ // y is in the [0.5, 1.5) range
+ //
+ // 2^x = 2^n * 2^y
+ // 2^n can be evaluated by playing with float representation.
+ // 2^y in a small range can be approximated, this code uses an order two
+ // polynomial approximation. The coefficients have been estimated
+ // with the Remez algorithm and the resulting polynomial has a
+ // maximum relative error of 0.17%.
+
+ // To avoid over/underflow, we reduce the range of input to ]-127, 129].
+ static const ALIGN16_BEG float max_input[4] ALIGN16_END =
+ {129.f, 129.f, 129.f, 129.f};
+ static const ALIGN16_BEG float min_input[4] ALIGN16_END =
+ {-126.99999f, -126.99999f, -126.99999f, -126.99999f};
+ const __m128 x_min = _mm_min_ps(b_log2_a, *((__m128 *)max_input));
+ const __m128 x_max = _mm_max_ps(x_min, *((__m128 *)min_input));
+ // Compute n.
+ static const ALIGN16_BEG float half[4] ALIGN16_END =
+ {0.5f, 0.5f, 0.5f, 0.5f};
+ const __m128 x_minus_half = _mm_sub_ps(x_max, *((__m128 *)half));
+ const __m128i x_minus_half_floor = _mm_cvtps_epi32(x_minus_half);
+ // Compute 2^n.
+ static const ALIGN16_BEG int float_exponent_bias[4] ALIGN16_END =
+ {127, 127, 127, 127};
+ static const int float_exponent_shift = 23;
+ const __m128i two_n_exponent = _mm_add_epi32(
+ x_minus_half_floor, *((__m128i *)float_exponent_bias));
+ const __m128 two_n = _mm_castsi128_ps(_mm_slli_epi32(
+ two_n_exponent, float_exponent_shift));
+ // Compute y.
+ const __m128 y = _mm_sub_ps(x_max, _mm_cvtepi32_ps(x_minus_half_floor));
+ // Approximate 2^y ~= C2 * y^2 + C1 * y + C0.
+ static const ALIGN16_BEG float C2[4] ALIGN16_END =
+ {3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f};
+ static const ALIGN16_BEG float C1[4] ALIGN16_END =
+ {6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f};
+ static const ALIGN16_BEG float C0[4] ALIGN16_END =
+ {1.0017247f, 1.0017247f, 1.0017247f, 1.0017247f};
+ const __m128 exp2_y_0 = _mm_mul_ps(y, *((__m128 *)C2));
+ const __m128 exp2_y_1 = _mm_add_ps(exp2_y_0, *((__m128 *)C1));
+ const __m128 exp2_y_2 = _mm_mul_ps(exp2_y_1, y);
+ const __m128 exp2_y = _mm_add_ps(exp2_y_2, *((__m128 *)C0));
+
+ // Combine parts.
+ a_exp_b = _mm_mul_ps(exp2_y, two_n);
+ }
+ return a_exp_b;
+}
+
+extern const float WebRtcAec_weightCurve[65];
+extern const float WebRtcAec_overDriveCurve[65];
+
+static void OverdriveAndSuppressSSE2(aec_t *aec, float hNl[PART_LEN1],
+ const float hNlFb,
+ float efw[2][PART_LEN1]) {
+ int i;
+ const __m128 vec_hNlFb = _mm_set1_ps(hNlFb);
+ const __m128 vec_one = _mm_set1_ps(1.0f);
+ const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
+ const __m128 vec_overDriveSm = _mm_set1_ps(aec->overDriveSm);
+ // vectorized code (four at once)
+ for (i = 0; i + 3 < PART_LEN1; i+=4) {
+ // Weight subbands
+ __m128 vec_hNl = _mm_loadu_ps(&hNl[i]);
+ const __m128 vec_weightCurve = _mm_loadu_ps(&WebRtcAec_weightCurve[i]);
+ const __m128 bigger = _mm_cmpgt_ps(vec_hNl, vec_hNlFb);
+ const __m128 vec_weightCurve_hNlFb = _mm_mul_ps(
+ vec_weightCurve, vec_hNlFb);
+ const __m128 vec_one_weightCurve = _mm_sub_ps(vec_one, vec_weightCurve);
+ const __m128 vec_one_weightCurve_hNl = _mm_mul_ps(
+ vec_one_weightCurve, vec_hNl);
+ const __m128 vec_if0 = _mm_andnot_ps(bigger, vec_hNl);
+ const __m128 vec_if1 = _mm_and_ps(
+ bigger, _mm_add_ps(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl));
+ vec_hNl = _mm_or_ps(vec_if0, vec_if1);
+
+ {
+ const __m128 vec_overDriveCurve = _mm_loadu_ps(
+ &WebRtcAec_overDriveCurve[i]);
+ const __m128 vec_overDriveSm_overDriveCurve = _mm_mul_ps(
+ vec_overDriveSm, vec_overDriveCurve);
+ vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
+ _mm_storeu_ps(&hNl[i], vec_hNl);
+ }
+
+ // Suppress error signal
+ {
+ __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
+ __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
+ vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
+ vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
+
+ // Ooura fft returns incorrect sign on imaginary component. It matters
+ // here because we are making an additive change with comfort noise.
+ vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
+ _mm_storeu_ps(&efw[0][i], vec_efw_re);
+ _mm_storeu_ps(&efw[1][i], vec_efw_im);
+ }
+ }
+ // scalar code for the remaining items.
+ for (; i < PART_LEN1; i++) {
+ // Weight subbands
+ if (hNl[i] > hNlFb) {
+ hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
+ (1 - WebRtcAec_weightCurve[i]) * hNl[i];
+ }
+ hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+
+ // Suppress error signal
+ efw[0][i] *= hNl[i];
+ efw[1][i] *= hNl[i];
+
+ // Ooura fft returns incorrect sign on imaginary component. It matters
+ // here because we are making an additive change with comfort noise.
+ efw[1][i] *= -1;
+ }
+}
+
+void WebRtcAec_InitAec_SSE2(void) {
+ WebRtcAec_FilterFar = FilterFarSSE2;
+ WebRtcAec_ScaleErrorSignal = ScaleErrorSignalSSE2;
+ WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
+ WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
+}
+
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft.c b/trunk/src/modules/audio_processing/aec/aec_rdft.c
new file mode 100644
index 0000000..19908d8
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_rdft.c
@@ -0,0 +1,587 @@
+/*
+ * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
+ * Copyright Takuya OOURA, 1996-2001
+ *
+ * You may use, copy, modify and distribute this code for any purpose (include
+ * commercial use) and without fee. Please refer to this package when you modify
+ * this code.
+ *
+ * Changes by the WebRTC authors:
+ * - Trivial type modifications.
+ * - Minimal code subset to do rdft of length 128.
+ * - Optimizations because of known length.
+ *
+ * All changes are covered by the WebRTC license and IP grant:
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aec_rdft.h"
+
+#include <math.h>
+
+#include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "typedefs.h"
+
+// constants shared by all paths (C, SSE2).
+float rdft_w[64];
+// constants used by the C path.
+float rdft_wk3ri_first[32];
+float rdft_wk3ri_second[32];
+// constants used by SSE2 but initialized in C path.
+ALIGN16_BEG float ALIGN16_END rdft_wk1r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk2r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk3r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk1i[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk2i[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk3i[32];
+ALIGN16_BEG float ALIGN16_END cftmdl_wk1r[4];
+
+static int ip[16];
+
+static void bitrv2_32or128(int n, int *ip, float *a) {
+ // n is 32 or 128
+ int j, j1, k, k1, m, m2;
+ float xr, xi, yr, yi;
+
+ ip[0] = 0;
+ {
+ int l = n;
+ m = 1;
+ while ((m << 3) < l) {
+ l >>= 1;
+ for (j = 0; j < m; j++) {
+ ip[m + j] = ip[j] + l;
+ }
+ m <<= 1;
+ }
+ }
+ m2 = 2 * m;
+ for (k = 0; k < m; k++) {
+ for (j = 0; j < k; j++) {
+ j1 = 2 * j + ip[k];
+ k1 = 2 * k + ip[j];
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 -= m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ j1 = 2 * k + m2 + ip[k];
+ k1 = j1 + m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+}
+
+static void makewt_32(void) {
+ const int nw = 32;
+ int j, nwh;
+ float delta, x, y;
+
+ ip[0] = nw;
+ ip[1] = 1;
+ nwh = nw >> 1;
+ delta = atanf(1.0f) / nwh;
+ rdft_w[0] = 1;
+ rdft_w[1] = 0;
+ rdft_w[nwh] = cosf(delta * nwh);
+ rdft_w[nwh + 1] = rdft_w[nwh];
+ for (j = 2; j < nwh; j += 2) {
+ x = cosf(delta * j);
+ y = sinf(delta * j);
+ rdft_w[j] = x;
+ rdft_w[j + 1] = y;
+ rdft_w[nw - j] = y;
+ rdft_w[nw - j + 1] = x;
+ }
+ bitrv2_32or128(nw, ip + 2, rdft_w);
+
+ // pre-calculate constants used by cft1st_128 and cftmdl_128...
+ cftmdl_wk1r[0] = rdft_w[2];
+ cftmdl_wk1r[1] = rdft_w[2];
+ cftmdl_wk1r[2] = rdft_w[2];
+ cftmdl_wk1r[3] = -rdft_w[2];
+ {
+ int k1;
+
+ for (k1 = 0, j = 0; j < 128; j += 16, k1 += 2) {
+ const int k2 = 2 * k1;
+ const float wk2r = rdft_w[k1 + 0];
+ const float wk2i = rdft_w[k1 + 1];
+ float wk1r, wk1i;
+ // ... scalar version.
+ wk1r = rdft_w[k2 + 0];
+ wk1i = rdft_w[k2 + 1];
+ rdft_wk3ri_first[k1 + 0] = wk1r - 2 * wk2i * wk1i;
+ rdft_wk3ri_first[k1 + 1] = 2 * wk2i * wk1r - wk1i;
+ wk1r = rdft_w[k2 + 2];
+ wk1i = rdft_w[k2 + 3];
+ rdft_wk3ri_second[k1 + 0] = wk1r - 2 * wk2r * wk1i;
+ rdft_wk3ri_second[k1 + 1] = 2 * wk2r * wk1r - wk1i;
+ // ... vector version.
+ rdft_wk1r[k2 + 0] = rdft_w[k2 + 0];
+ rdft_wk1r[k2 + 1] = rdft_w[k2 + 0];
+ rdft_wk1r[k2 + 2] = rdft_w[k2 + 2];
+ rdft_wk1r[k2 + 3] = rdft_w[k2 + 2];
+ rdft_wk2r[k2 + 0] = rdft_w[k1 + 0];
+ rdft_wk2r[k2 + 1] = rdft_w[k1 + 0];
+ rdft_wk2r[k2 + 2] = -rdft_w[k1 + 1];
+ rdft_wk2r[k2 + 3] = -rdft_w[k1 + 1];
+ rdft_wk3r[k2 + 0] = rdft_wk3ri_first[k1 + 0];
+ rdft_wk3r[k2 + 1] = rdft_wk3ri_first[k1 + 0];
+ rdft_wk3r[k2 + 2] = rdft_wk3ri_second[k1 + 0];
+ rdft_wk3r[k2 + 3] = rdft_wk3ri_second[k1 + 0];
+ rdft_wk1i[k2 + 0] = -rdft_w[k2 + 1];
+ rdft_wk1i[k2 + 1] = rdft_w[k2 + 1];
+ rdft_wk1i[k2 + 2] = -rdft_w[k2 + 3];
+ rdft_wk1i[k2 + 3] = rdft_w[k2 + 3];
+ rdft_wk2i[k2 + 0] = -rdft_w[k1 + 1];
+ rdft_wk2i[k2 + 1] = rdft_w[k1 + 1];
+ rdft_wk2i[k2 + 2] = -rdft_w[k1 + 0];
+ rdft_wk2i[k2 + 3] = rdft_w[k1 + 0];
+ rdft_wk3i[k2 + 0] = -rdft_wk3ri_first[k1 + 1];
+ rdft_wk3i[k2 + 1] = rdft_wk3ri_first[k1 + 1];
+ rdft_wk3i[k2 + 2] = -rdft_wk3ri_second[k1 + 1];
+ rdft_wk3i[k2 + 3] = rdft_wk3ri_second[k1 + 1];
+ }
+ }
+}
+
+static void makect_32(void) {
+ float *c = rdft_w + 32;
+ const int nc = 32;
+ int j, nch;
+ float delta;
+
+ ip[1] = nc;
+ nch = nc >> 1;
+ delta = atanf(1.0f) / nch;
+ c[0] = cosf(delta * nch);
+ c[nch] = 0.5f * c[0];
+ for (j = 1; j < nch; j++) {
+ c[j] = 0.5f * cosf(delta * j);
+ c[nc - j] = 0.5f * sinf(delta * j);
+ }
+}
+
+static void cft1st_128_C(float *a) {
+ const int n = 128;
+ int j, k1, k2;
+ float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ x0r = a[0] + a[2];
+ x0i = a[1] + a[3];
+ x1r = a[0] - a[2];
+ x1i = a[1] - a[3];
+ x2r = a[4] + a[6];
+ x2i = a[5] + a[7];
+ x3r = a[4] - a[6];
+ x3i = a[5] - a[7];
+ a[0] = x0r + x2r;
+ a[1] = x0i + x2i;
+ a[4] = x0r - x2r;
+ a[5] = x0i - x2i;
+ a[2] = x1r - x3i;
+ a[3] = x1i + x3r;
+ a[6] = x1r + x3i;
+ a[7] = x1i - x3r;
+ wk1r = rdft_w[2];
+ x0r = a[8] + a[10];
+ x0i = a[9] + a[11];
+ x1r = a[8] - a[10];
+ x1i = a[9] - a[11];
+ x2r = a[12] + a[14];
+ x2i = a[13] + a[15];
+ x3r = a[12] - a[14];
+ x3i = a[13] - a[15];
+ a[8] = x0r + x2r;
+ a[9] = x0i + x2i;
+ a[12] = x2i - x0i;
+ a[13] = x0r - x2r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[10] = wk1r * (x0r - x0i);
+ a[11] = wk1r * (x0r + x0i);
+ x0r = x3i + x1r;
+ x0i = x3r - x1i;
+ a[14] = wk1r * (x0i - x0r);
+ a[15] = wk1r * (x0i + x0r);
+ k1 = 0;
+ for (j = 16; j < n; j += 16) {
+ k1 += 2;
+ k2 = 2 * k1;
+ wk2r = rdft_w[k1 + 0];
+ wk2i = rdft_w[k1 + 1];
+ wk1r = rdft_w[k2 + 0];
+ wk1i = rdft_w[k2 + 1];
+ wk3r = rdft_wk3ri_first[k1 + 0];
+ wk3i = rdft_wk3ri_first[k1 + 1];
+ x0r = a[j + 0] + a[j + 2];
+ x0i = a[j + 1] + a[j + 3];
+ x1r = a[j + 0] - a[j + 2];
+ x1i = a[j + 1] - a[j + 3];
+ x2r = a[j + 4] + a[j + 6];
+ x2i = a[j + 5] + a[j + 7];
+ x3r = a[j + 4] - a[j + 6];
+ x3i = a[j + 5] - a[j + 7];
+ a[j + 0] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j + 4] = wk2r * x0r - wk2i * x0i;
+ a[j + 5] = wk2r * x0i + wk2i * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j + 2] = wk1r * x0r - wk1i * x0i;
+ a[j + 3] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j + 6] = wk3r * x0r - wk3i * x0i;
+ a[j + 7] = wk3r * x0i + wk3i * x0r;
+ wk1r = rdft_w[k2 + 2];
+ wk1i = rdft_w[k2 + 3];
+ wk3r = rdft_wk3ri_second[k1 + 0];
+ wk3i = rdft_wk3ri_second[k1 + 1];
+ x0r = a[j + 8] + a[j + 10];
+ x0i = a[j + 9] + a[j + 11];
+ x1r = a[j + 8] - a[j + 10];
+ x1i = a[j + 9] - a[j + 11];
+ x2r = a[j + 12] + a[j + 14];
+ x2i = a[j + 13] + a[j + 15];
+ x3r = a[j + 12] - a[j + 14];
+ x3i = a[j + 13] - a[j + 15];
+ a[j + 8] = x0r + x2r;
+ a[j + 9] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j + 12] = -wk2i * x0r - wk2r * x0i;
+ a[j + 13] = -wk2i * x0i + wk2r * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j + 10] = wk1r * x0r - wk1i * x0i;
+ a[j + 11] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j + 14] = wk3r * x0r - wk3i * x0i;
+ a[j + 15] = wk3r * x0i + wk3i * x0r;
+ }
+}
+
+static void cftmdl_128_C(float *a) {
+ const int l = 8;
+ const int n = 128;
+ const int m = 32;
+ int j0, j1, j2, j3, k, k1, k2, m2;
+ float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ for (j0 = 0; j0 < l; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2 + 0] - a[j3 + 0];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ a[j2 + 0] = x0r - x2r;
+ a[j2 + 1] = x0i - x2i;
+ a[j1 + 0] = x1r - x3i;
+ a[j1 + 1] = x1i + x3r;
+ a[j3 + 0] = x1r + x3i;
+ a[j3 + 1] = x1i - x3r;
+ }
+ wk1r = rdft_w[2];
+ for (j0 = m; j0 < l + m; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2 + 0] - a[j3 + 0];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ a[j2 + 0] = x2i - x0i;
+ a[j2 + 1] = x0r - x2r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1 + 0] = wk1r * (x0r - x0i);
+ a[j1 + 1] = wk1r * (x0r + x0i);
+ x0r = x3i + x1r;
+ x0i = x3r - x1i;
+ a[j3 + 0] = wk1r * (x0i - x0r);
+ a[j3 + 1] = wk1r * (x0i + x0r);
+ }
+ k1 = 0;
+ m2 = 2 * m;
+ for (k = m2; k < n; k += m2) {
+ k1 += 2;
+ k2 = 2 * k1;
+ wk2r = rdft_w[k1 + 0];
+ wk2i = rdft_w[k1 + 1];
+ wk1r = rdft_w[k2 + 0];
+ wk1i = rdft_w[k2 + 1];
+ wk3r = rdft_wk3ri_first[k1 + 0];
+ wk3i = rdft_wk3ri_first[k1 + 1];
+ for (j0 = k; j0 < l + k; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2 + 0] - a[j3 + 0];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j2 + 0] = wk2r * x0r - wk2i * x0i;
+ a[j2 + 1] = wk2r * x0i + wk2i * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1 + 0] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 1] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j3 + 0] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 1] = wk3r * x0i + wk3i * x0r;
+ }
+ wk1r = rdft_w[k2 + 2];
+ wk1i = rdft_w[k2 + 3];
+ wk3r = rdft_wk3ri_second[k1 + 0];
+ wk3i = rdft_wk3ri_second[k1 + 1];
+ for (j0 = k + m; j0 < l + (k + m); j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2 + 0] - a[j3 + 0];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j2 + 0] = -wk2i * x0r - wk2r * x0i;
+ a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1 + 0] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 1] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j3 + 0] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 1] = wk3r * x0i + wk3i * x0r;
+ }
+ }
+}
+
+static void cftfsub_128(float *a) {
+ int j, j1, j2, j3, l;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ cft1st_128(a);
+ cftmdl_128(a);
+ l = 32;
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ a[j2] = x0r - x2r;
+ a[j2 + 1] = x0i - x2i;
+ a[j1] = x1r - x3i;
+ a[j1 + 1] = x1i + x3r;
+ a[j3] = x1r + x3i;
+ a[j3 + 1] = x1i - x3r;
+ }
+}
+
+static void cftbsub_128(float *a) {
+ int j, j1, j2, j3, l;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ cft1st_128(a);
+ cftmdl_128(a);
+ l = 32;
+
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = -a[j + 1] - a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = -a[j + 1] + a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i - x2i;
+ a[j2] = x0r - x2r;
+ a[j2 + 1] = x0i + x2i;
+ a[j1] = x1r - x3i;
+ a[j1 + 1] = x1i - x3r;
+ a[j3] = x1r + x3i;
+ a[j3 + 1] = x1i + x3r;
+ }
+}
+
+static void rftfsub_128_C(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr - wki * xi;
+ yi = wkr * xi + wki * xr;
+ a[j2 + 0] -= yr;
+ a[j2 + 1] -= yi;
+ a[k2 + 0] += yr;
+ a[k2 + 1] -= yi;
+ }
+}
+
+static void rftbsub_128_C(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ a[1] = -a[1];
+ for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr + wki * xi;
+ yi = wkr * xi - wki * xr;
+ a[j2 + 0] = a[j2 + 0] - yr;
+ a[j2 + 1] = yi - a[j2 + 1];
+ a[k2 + 0] = yr + a[k2 + 0];
+ a[k2 + 1] = yi - a[k2 + 1];
+ }
+ a[65] = -a[65];
+}
+
+void aec_rdft_forward_128(float *a) {
+ const int n = 128;
+ float xi;
+
+ bitrv2_32or128(n, ip + 2, a);
+ cftfsub_128(a);
+ rftfsub_128(a);
+ xi = a[0] - a[1];
+ a[0] += a[1];
+ a[1] = xi;
+}
+
+void aec_rdft_inverse_128(float *a) {
+ const int n = 128;
+
+ a[1] = 0.5f * (a[0] - a[1]);
+ a[0] -= a[1];
+ rftbsub_128(a);
+ bitrv2_32or128(n, ip + 2, a);
+ cftbsub_128(a);
+}
+
+// code path selection
+rft_sub_128_t cft1st_128;
+rft_sub_128_t cftmdl_128;
+rft_sub_128_t rftfsub_128;
+rft_sub_128_t rftbsub_128;
+
+void aec_rdft_init(void) {
+ cft1st_128 = cft1st_128_C;
+ cftmdl_128 = cftmdl_128_C;
+ rftfsub_128 = rftfsub_128_C;
+ rftbsub_128 = rftbsub_128_C;
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+ if (WebRtc_GetCPUInfo(kSSE2)) {
+ aec_rdft_init_sse2();
+ }
+#endif
+ // init library constants.
+ makewt_32();
+ makect_32();
+}
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft.h b/trunk/src/modules/audio_processing/aec/aec_rdft.h
new file mode 100644
index 0000000..91bedc9
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_rdft.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+
+// These intrinsics were unavailable before VS 2008.
+// TODO(andrew): move to a common file.
+#if defined(_MSC_VER) && _MSC_VER < 1500
+#include <emmintrin.h>
+static __inline __m128 _mm_castsi128_ps(__m128i a) { return *(__m128*)&a; }
+static __inline __m128i _mm_castps_si128(__m128 a) { return *(__m128i*)&a; }
+#endif
+
+#ifdef _MSC_VER /* visual c++ */
+# define ALIGN16_BEG __declspec(align(16))
+# define ALIGN16_END
+#else /* gcc or icc */
+# define ALIGN16_BEG
+# define ALIGN16_END __attribute__((aligned(16)))
+#endif
+
+// constants shared by all paths (C, SSE2).
+extern float rdft_w[64];
+// constants used by the C path.
+extern float rdft_wk3ri_first[32];
+extern float rdft_wk3ri_second[32];
+// constants used by SSE2 but initialized in C path.
+extern float rdft_wk1r[32];
+extern float rdft_wk2r[32];
+extern float rdft_wk3r[32];
+extern float rdft_wk1i[32];
+extern float rdft_wk2i[32];
+extern float rdft_wk3i[32];
+extern float cftmdl_wk1r[4];
+
+// code path selection function pointers
+typedef void (*rft_sub_128_t)(float *a);
+extern rft_sub_128_t rftfsub_128;
+extern rft_sub_128_t rftbsub_128;
+extern rft_sub_128_t cft1st_128;
+extern rft_sub_128_t cftmdl_128;
+
+// entry points
+void aec_rdft_init(void);
+void aec_rdft_init_sse2(void);
+void aec_rdft_forward_128(float *a);
+void aec_rdft_inverse_128(float *a);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c b/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c
new file mode 100644
index 0000000..eeb3152
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c
@@ -0,0 +1,427 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aec_rdft.h"
+
+#include <emmintrin.h>
+
+static const ALIGN16_BEG float ALIGN16_END k_swap_sign[4] =
+ {-1.f, 1.f, -1.f, 1.f};
+
+static void cft1st_128_SSE2(float *a) {
+ const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
+ int j, k2;
+
+ for (k2 = 0, j = 0; j < 128; j += 16, k2 += 4) {
+ __m128 a00v = _mm_loadu_ps(&a[j + 0]);
+ __m128 a04v = _mm_loadu_ps(&a[j + 4]);
+ __m128 a08v = _mm_loadu_ps(&a[j + 8]);
+ __m128 a12v = _mm_loadu_ps(&a[j + 12]);
+ __m128 a01v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 a23v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(3, 2, 3 ,2));
+ __m128 a45v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 a67v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(3, 2, 3 ,2));
+
+ const __m128 wk1rv = _mm_load_ps(&rdft_wk1r[k2]);
+ const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2]);
+ const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2]);
+ const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2]);
+ const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2]);
+ const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2]);
+ __m128 x0v = _mm_add_ps(a01v, a23v);
+ const __m128 x1v = _mm_sub_ps(a01v, a23v);
+ const __m128 x2v = _mm_add_ps(a45v, a67v);
+ const __m128 x3v = _mm_sub_ps(a45v, a67v);
+ __m128 x0w;
+ a01v = _mm_add_ps(x0v, x2v);
+ x0v = _mm_sub_ps(x0v, x2v);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ {
+ const __m128 a45_0v = _mm_mul_ps(wk2rv, x0v);
+ const __m128 a45_1v = _mm_mul_ps(wk2iv, x0w);
+ a45v = _mm_add_ps(a45_0v, a45_1v);
+ }
+ {
+ __m128 a23_0v, a23_1v;
+ const __m128 x3w = _mm_shuffle_ps(x3v, x3v, _MM_SHUFFLE(2, 3, 0 ,1));
+ const __m128 x3s = _mm_mul_ps(mm_swap_sign, x3w);
+ x0v = _mm_add_ps(x1v, x3s);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ a23_0v = _mm_mul_ps(wk1rv, x0v);
+ a23_1v = _mm_mul_ps(wk1iv, x0w);
+ a23v = _mm_add_ps(a23_0v, a23_1v);
+
+ x0v = _mm_sub_ps(x1v, x3s);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ }
+ {
+ const __m128 a67_0v = _mm_mul_ps(wk3rv, x0v);
+ const __m128 a67_1v = _mm_mul_ps(wk3iv, x0w);
+ a67v = _mm_add_ps(a67_0v, a67_1v);
+ }
+
+ a00v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(1, 0, 1 ,0));
+ a04v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(1, 0, 1 ,0));
+ a08v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(3, 2, 3 ,2));
+ a12v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(3, 2, 3 ,2));
+ _mm_storeu_ps(&a[j + 0], a00v);
+ _mm_storeu_ps(&a[j + 4], a04v);
+ _mm_storeu_ps(&a[j + 8], a08v);
+ _mm_storeu_ps(&a[j + 12], a12v);
+ }
+}
+
+static void cftmdl_128_SSE2(float *a) {
+ const int l = 8;
+ const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
+ int j0;
+
+ __m128 wk1rv = _mm_load_ps(cftmdl_wk1r);
+ for (j0 = 0; j0 < l; j0 += 2) {
+ const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
+ const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
+ const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
+ const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
+ const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
+ _mm_castsi128_ps(a_32),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
+ _mm_castsi128_ps(a_40),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
+ const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
+
+ const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
+ const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
+ const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
+ const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
+ const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
+ _mm_castsi128_ps(a_48),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
+ _mm_castsi128_ps(a_56),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
+ const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
+
+ const __m128 xx0 = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+
+ const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
+ _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
+ _MM_SHUFFLE(2, 3, 0, 1)));
+ const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
+ const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+ const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+
+ const __m128 yy0 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
+ _MM_SHUFFLE(2, 2, 2 ,2));
+ const __m128 yy1 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
+ _MM_SHUFFLE(3, 3, 3 ,3));
+ const __m128 yy2 = _mm_mul_ps(mm_swap_sign, yy1);
+ const __m128 yy3 = _mm_add_ps(yy0, yy2);
+ const __m128 yy4 = _mm_mul_ps(wk1rv, yy3);
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx0));
+ _mm_storel_epi64((__m128i*)&a[j0 + 32],
+ _mm_shuffle_epi32(_mm_castps_si128(xx0),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx1));
+ _mm_storel_epi64((__m128i*)&a[j0 + 48],
+ _mm_shuffle_epi32(_mm_castps_si128(xx1),
+ _MM_SHUFFLE(2, 3, 2, 3)));
+ a[j0 + 48] = -a[j0 + 48];
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(x1_x3_add));
+ _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(x1_x3_sub));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 40], _mm_castps_si128(yy4));
+ _mm_storel_epi64((__m128i*)&a[j0 + 56],
+ _mm_shuffle_epi32(_mm_castps_si128(yy4),
+ _MM_SHUFFLE(2, 3, 2, 3)));
+ }
+
+ {
+ int k = 64;
+ int k1 = 2;
+ int k2 = 2 * k1;
+ const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2+0]);
+ const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2+0]);
+ const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2+0]);
+ const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2+0]);
+ const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2+0]);
+ wk1rv = _mm_load_ps(&rdft_wk1r[k2+0]);
+ for (j0 = k; j0 < l + k; j0 += 2) {
+ const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
+ const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
+ const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
+ const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
+ const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
+ _mm_castsi128_ps(a_32),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
+ _mm_castsi128_ps(a_40),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
+ const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
+
+ const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
+ const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
+ const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
+ const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
+ const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
+ _mm_castsi128_ps(a_48),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
+ _mm_castsi128_ps(a_56),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
+ const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
+
+ const __m128 xx = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx2 = _mm_mul_ps(xx1 , wk2rv);
+ const __m128 xx3 = _mm_mul_ps(wk2iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(xx1),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx4 = _mm_add_ps(xx2, xx3);
+
+ const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
+ _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
+ _MM_SHUFFLE(2, 3, 0, 1)));
+ const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
+ const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+ const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+
+ const __m128 xx10 = _mm_mul_ps(x1_x3_add, wk1rv);
+ const __m128 xx11 = _mm_mul_ps(wk1iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_add),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx12 = _mm_add_ps(xx10, xx11);
+
+ const __m128 xx20 = _mm_mul_ps(x1_x3_sub, wk3rv);
+ const __m128 xx21 = _mm_mul_ps(wk3iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_sub),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx22 = _mm_add_ps(xx20, xx21);
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx));
+ _mm_storel_epi64((__m128i*)&a[j0 + 32],
+ _mm_shuffle_epi32(_mm_castps_si128(xx),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx4));
+ _mm_storel_epi64((__m128i*)&a[j0 + 48],
+ _mm_shuffle_epi32(_mm_castps_si128(xx4),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(xx12));
+ _mm_storel_epi64((__m128i*)&a[j0 + 40],
+ _mm_shuffle_epi32(_mm_castps_si128(xx12),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(xx22));
+ _mm_storel_epi64((__m128i*)&a[j0 + 56],
+ _mm_shuffle_epi32(_mm_castps_si128(xx22),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+ }
+ }
+}
+
+static void rftfsub_128_SSE2(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ static const ALIGN16_BEG float ALIGN16_END k_half[4] =
+ {0.5f, 0.5f, 0.5f, 0.5f};
+ const __m128 mm_half = _mm_load_ps(k_half);
+
+ // Vectorized code (four at once).
+ // Note: commented number are indexes for the first iteration of the loop.
+ for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
+ // Load 'wk'.
+ const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
+ const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
+ const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
+ const __m128 wkr_ =
+ _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
+ const __m128 wki_ = c_j1; // 1, 2, 3, 4,
+ // Load and shuffle 'a'.
+ const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
+ const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
+ const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
+ const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
+ const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
+ const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
+ const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
+ const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
+ // Calculate 'x'.
+ const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
+ // 2-126, 4-124, 6-122, 8-120,
+ const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
+ // 3-127, 5-125, 7-123, 9-121,
+ // Calculate product into 'y'.
+ // yr = wkr * xr - wki * xi;
+ // yi = wkr * xi + wki * xr;
+ const __m128 a_ = _mm_mul_ps(wkr_, xr_);
+ const __m128 b_ = _mm_mul_ps(wki_, xi_);
+ const __m128 c_ = _mm_mul_ps(wkr_, xi_);
+ const __m128 d_ = _mm_mul_ps(wki_, xr_);
+ const __m128 yr_ = _mm_sub_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
+ const __m128 yi_ = _mm_add_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
+ // Update 'a'.
+ // a[j2 + 0] -= yr;
+ // a[j2 + 1] -= yi;
+ // a[k2 + 0] += yr;
+ // a[k2 + 1] -= yi;
+ const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
+ const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_); // 3, 5, 7, 9,
+ const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
+ const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_); // 127, 125, 123, 121,
+ // Shuffle in right order and store.
+ const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
+ // 2, 3, 4, 5,
+ const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
+ // 6, 7, 8, 9,
+ const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
+ // 122, 123, 120, 121,
+ const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
+ // 126, 127, 124, 125,
+ const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
+ const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
+ _mm_storeu_ps(&a[0 + j2], a_j2_0n);
+ _mm_storeu_ps(&a[4 + j2], a_j2_4n);
+ _mm_storeu_ps(&a[122 - j2], a_k2_0n);
+ _mm_storeu_ps(&a[126 - j2], a_k2_4n);
+ }
+ // Scalar code for the remaining items.
+ for (; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr - wki * xi;
+ yi = wkr * xi + wki * xr;
+ a[j2 + 0] -= yr;
+ a[j2 + 1] -= yi;
+ a[k2 + 0] += yr;
+ a[k2 + 1] -= yi;
+ }
+}
+
+static void rftbsub_128_SSE2(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ static const ALIGN16_BEG float ALIGN16_END k_half[4] =
+ {0.5f, 0.5f, 0.5f, 0.5f};
+ const __m128 mm_half = _mm_load_ps(k_half);
+
+ a[1] = -a[1];
+ // Vectorized code (four at once).
+ // Note: commented number are indexes for the first iteration of the loop.
+ for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
+ // Load 'wk'.
+ const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
+ const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
+ const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
+ const __m128 wkr_ =
+ _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
+ const __m128 wki_ = c_j1; // 1, 2, 3, 4,
+ // Load and shuffle 'a'.
+ const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
+ const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
+ const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
+ const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
+ const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
+ const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
+ const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
+ const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
+ // Calculate 'x'.
+ const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
+ // 2-126, 4-124, 6-122, 8-120,
+ const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
+ // 3-127, 5-125, 7-123, 9-121,
+ // Calculate product into 'y'.
+ // yr = wkr * xr + wki * xi;
+ // yi = wkr * xi - wki * xr;
+ const __m128 a_ = _mm_mul_ps(wkr_, xr_);
+ const __m128 b_ = _mm_mul_ps(wki_, xi_);
+ const __m128 c_ = _mm_mul_ps(wkr_, xi_);
+ const __m128 d_ = _mm_mul_ps(wki_, xr_);
+ const __m128 yr_ = _mm_add_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
+ const __m128 yi_ = _mm_sub_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
+ // Update 'a'.
+ // a[j2 + 0] = a[j2 + 0] - yr;
+ // a[j2 + 1] = yi - a[j2 + 1];
+ // a[k2 + 0] = yr + a[k2 + 0];
+ // a[k2 + 1] = yi - a[k2 + 1];
+ const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
+ const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1); // 3, 5, 7, 9,
+ const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
+ const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1); // 127, 125, 123, 121,
+ // Shuffle in right order and store.
+ const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
+ // 2, 3, 4, 5,
+ const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
+ // 6, 7, 8, 9,
+ const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
+ // 122, 123, 120, 121,
+ const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
+ // 126, 127, 124, 125,
+ const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
+ const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
+ _mm_storeu_ps(&a[0 + j2], a_j2_0n);
+ _mm_storeu_ps(&a[4 + j2], a_j2_4n);
+ _mm_storeu_ps(&a[122 - j2], a_k2_0n);
+ _mm_storeu_ps(&a[126 - j2], a_k2_4n);
+ }
+ // Scalar code for the remaining items.
+ for (; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr + wki * xi;
+ yi = wkr * xi - wki * xr;
+ a[j2 + 0] = a[j2 + 0] - yr;
+ a[j2 + 1] = yi - a[j2 + 1];
+ a[k2 + 0] = yr + a[k2 + 0];
+ a[k2 + 1] = yi - a[k2 + 1];
+ }
+ a[65] = -a[65];
+}
+
+void aec_rdft_init_sse2(void) {
+ cft1st_128 = cft1st_128_SSE2;
+ cftmdl_128 = cftmdl_128_SSE2;
+ rftfsub_128 = rftfsub_128_SSE2;
+ rftbsub_128 = rftbsub_128_SSE2;
+}
+
diff --git a/trunk/src/modules/audio_processing/aec/aec_resampler.c b/trunk/src/modules/audio_processing/aec/aec_resampler.c
new file mode 100644
index 0000000..ea980cd
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_resampler.c
@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for clock
+ * skew by resampling the farend signal.
+ */
+
+#include "aec_resampler.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+#include "aec_core.h"
+
+enum { kEstimateLengthFrames = 400 };
+
+typedef struct {
+ short buffer[kResamplerBufferSize];
+ float position;
+
+ int deviceSampleRateHz;
+ int skewData[kEstimateLengthFrames];
+ int skewDataIndex;
+ float skewEstimate;
+} resampler_t;
+
+static int EstimateSkew(const int* rawSkew,
+ int size,
+ int absLimit,
+ float *skewEst);
+
+int WebRtcAec_CreateResampler(void **resampInst)
+{
+ resampler_t *obj = malloc(sizeof(resampler_t));
+ *resampInst = obj;
+ if (obj == NULL) {
+ return -1;
+ }
+
+ return 0;
+}
+
+int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz)
+{
+ resampler_t *obj = (resampler_t*) resampInst;
+ memset(obj->buffer, 0, sizeof(obj->buffer));
+ obj->position = 0.0;
+
+ obj->deviceSampleRateHz = deviceSampleRateHz;
+ memset(obj->skewData, 0, sizeof(obj->skewData));
+ obj->skewDataIndex = 0;
+ obj->skewEstimate = 0.0;
+
+ return 0;
+}
+
+int WebRtcAec_FreeResampler(void *resampInst)
+{
+ resampler_t *obj = (resampler_t*) resampInst;
+ free(obj);
+
+ return 0;
+}
+
+int WebRtcAec_ResampleLinear(void *resampInst,
+ const short *inspeech,
+ int size,
+ float skew,
+ short *outspeech)
+{
+ resampler_t *obj = (resampler_t*) resampInst;
+
+ short *y;
+ float be, tnew, interp;
+ int tn, outsize, mm;
+
+ if (size < 0 || size > 2 * FRAME_LEN) {
+ return -1;
+ }
+
+ // Add new frame data in lookahead
+ memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
+ inspeech,
+ size * sizeof(short));
+
+ // Sample rate ratio
+ be = 1 + skew;
+
+ // Loop over input frame
+ mm = 0;
+ y = &obj->buffer[FRAME_LEN]; // Point at current frame
+
+ tnew = be * mm + obj->position;
+ tn = (int) tnew;
+
+ while (tn < size) {
+
+ // Interpolation
+ interp = y[tn] + (tnew - tn) * (y[tn+1] - y[tn]);
+
+ if (interp > 32767) {
+ interp = 32767;
+ }
+ else if (interp < -32768) {
+ interp = -32768;
+ }
+
+ outspeech[mm] = (short) interp;
+ mm++;
+
+ tnew = be * mm + obj->position;
+ tn = (int) tnew;
+ }
+
+ outsize = mm;
+ obj->position += outsize * be - size;
+
+ // Shift buffer
+ memmove(obj->buffer,
+ &obj->buffer[size],
+ (kResamplerBufferSize - size) * sizeof(short));
+
+ return outsize;
+}
+
+int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst)
+{
+ resampler_t *obj = (resampler_t*)resampInst;
+ int err = 0;
+
+ if (obj->skewDataIndex < kEstimateLengthFrames) {
+ obj->skewData[obj->skewDataIndex] = rawSkew;
+ obj->skewDataIndex++;
+ }
+ else if (obj->skewDataIndex == kEstimateLengthFrames) {
+ err = EstimateSkew(obj->skewData,
+ kEstimateLengthFrames,
+ obj->deviceSampleRateHz,
+ skewEst);
+ obj->skewEstimate = *skewEst;
+ obj->skewDataIndex++;
+ }
+ else {
+ *skewEst = obj->skewEstimate;
+ }
+
+ return err;
+}
+
+int EstimateSkew(const int* rawSkew,
+ int size,
+ int deviceSampleRateHz,
+ float *skewEst)
+{
+ const int absLimitOuter = (int)(0.04f * deviceSampleRateHz);
+ const int absLimitInner = (int)(0.0025f * deviceSampleRateHz);
+ int i = 0;
+ int n = 0;
+ float rawAvg = 0;
+ float err = 0;
+ float rawAbsDev = 0;
+ int upperLimit = 0;
+ int lowerLimit = 0;
+ float cumSum = 0;
+ float x = 0;
+ float x2 = 0;
+ float y = 0;
+ float xy = 0;
+ float xAvg = 0;
+ float denom = 0;
+ float skew = 0;
+
+ *skewEst = 0; // Set in case of error below.
+ for (i = 0; i < size; i++) {
+ if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
+ n++;
+ rawAvg += rawSkew[i];
+ }
+ }
+
+ if (n == 0) {
+ return -1;
+ }
+ assert(n > 0);
+ rawAvg /= n;
+
+ for (i = 0; i < size; i++) {
+ if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
+ err = rawSkew[i] - rawAvg;
+ rawAbsDev += err >= 0 ? err : -err;
+ }
+ }
+ assert(n > 0);
+ rawAbsDev /= n;
+ upperLimit = (int)(rawAvg + 5 * rawAbsDev + 1); // +1 for ceiling.
+ lowerLimit = (int)(rawAvg - 5 * rawAbsDev - 1); // -1 for floor.
+
+ n = 0;
+ for (i = 0; i < size; i++) {
+ if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
+ (rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
+ n++;
+ cumSum += rawSkew[i];
+ x += n;
+ x2 += n*n;
+ y += cumSum;
+ xy += n * cumSum;
+ }
+ }
+
+ if (n == 0) {
+ return -1;
+ }
+ assert(n > 0);
+ xAvg = x / n;
+ denom = x2 - xAvg*x;
+
+ if (denom != 0) {
+ skew = (xy - xAvg*y) / denom;
+ }
+
+ *skewEst = skew;
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/aec/aec_resampler.h b/trunk/src/modules/audio_processing/aec/aec_resampler.h
new file mode 100644
index 0000000..ab4cc6e
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/aec_resampler.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+
+#include "aec_core.h"
+
+enum { kResamplingDelay = 1 };
+enum { kResamplerBufferSize = FRAME_LEN * 4 };
+
+// Unless otherwise specified, functions return 0 on success and -1 on error
+int WebRtcAec_CreateResampler(void **resampInst);
+int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz);
+int WebRtcAec_FreeResampler(void *resampInst);
+
+// Estimates skew from raw measurement.
+int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
+
+// Resamples input using linear interpolation.
+// Returns size of resampled array.
+int WebRtcAec_ResampleLinear(void *resampInst,
+ const short *inspeech,
+ int size,
+ float skew,
+ short *outspeech);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
diff --git a/trunk/src/modules/audio_processing/aec/echo_cancellation.c b/trunk/src/modules/audio_processing/aec/echo_cancellation.c
new file mode 100644
index 0000000..bde9c87
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/echo_cancellation.c
@@ -0,0 +1,924 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Contains the API functions for the AEC.
+ */
+#include "echo_cancellation.h"
+
+#include <math.h>
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#endif
+#include <stdlib.h>
+#include <string.h>
+
+#include "aec_core.h"
+#include "aec_resampler.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "ring_buffer.h"
+#include "typedefs.h"
+
+// Maximum length of resampled signal. Must be an integer multiple of frames
+// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
+// The factor of 2 handles wb, and the + 1 is as a safety margin
+// TODO(bjornv): Replace with kResamplerBufferSize
+#define MAX_RESAMP_LEN (5 * FRAME_LEN)
+
+static const int kMaxBufSizeStart = 62; // In partitions
+static const int sampMsNb = 8; // samples per ms in nb
+// Target suppression levels for nlp modes
+// log{0.001, 0.00001, 0.00000001}
+static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
+static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
+static const int initCheck = 42;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+static int instance_count = 0;
+#endif
+
+typedef struct {
+ int delayCtr;
+ int sampFreq;
+ int splitSampFreq;
+ int scSampFreq;
+ float sampFactor; // scSampRate / sampFreq
+ short nlpMode;
+ short autoOnOff;
+ short activity;
+ short skewMode;
+ int bufSizeStart;
+ //short bufResetCtr; // counts number of noncausal frames
+ int knownDelay;
+
+ short initFlag; // indicates if AEC has been initialized
+
+ // Variables used for averaging far end buffer size
+ short counter;
+ int sum;
+ short firstVal;
+ short checkBufSizeCtr;
+
+ // Variables used for delay shifts
+ short msInSndCardBuf;
+ short filtDelay; // Filtered delay estimate.
+ int timeForDelayChange;
+ int ECstartup;
+ int checkBuffSize;
+ short lastDelayDiff;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ void* far_pre_buf_s16; // Time domain far-end pre-buffer in int16_t.
+ FILE *bufFile;
+ FILE *delayFile;
+ FILE *skewFile;
+#endif
+
+ // Structures
+ void *resampler;
+
+ int skewFrCtr;
+ int resample; // if the skew is small enough we don't resample
+ int highSkewCtr;
+ float skew;
+
+ void* far_pre_buf; // Time domain far-end pre-buffer.
+
+ int lastError;
+
+ aec_t *aec;
+} aecpc_t;
+
+// Estimates delay to set the position of the far-end buffer read pointer
+// (controlled by knownDelay)
+static int EstBufDelay(aecpc_t *aecInst);
+
+WebRtc_Word32 WebRtcAec_Create(void **aecInst)
+{
+ aecpc_t *aecpc;
+ if (aecInst == NULL) {
+ return -1;
+ }
+
+ aecpc = malloc(sizeof(aecpc_t));
+ *aecInst = aecpc;
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+
+ if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+ // Create far-end pre-buffer. The buffer size has to be large enough for
+ // largest possible drift compensation (kResamplerBufferSize) + "almost" an
+ // FFT buffer (PART_LEN2 - 1).
+ if (WebRtc_CreateBuffer(&aecpc->far_pre_buf,
+ PART_LEN2 + kResamplerBufferSize,
+ sizeof(float)) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+
+ aecpc->initFlag = 0;
+ aecpc->lastError = 0;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_CreateBuffer(&aecpc->far_pre_buf_s16,
+ PART_LEN2 + kResamplerBufferSize,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+ {
+ char filename[64];
+ sprintf(filename, "aec_far%d.pcm", instance_count);
+ aecpc->aec->farFile = fopen(filename, "wb");
+ sprintf(filename, "aec_near%d.pcm", instance_count);
+ aecpc->aec->nearFile = fopen(filename, "wb");
+ sprintf(filename, "aec_out%d.pcm", instance_count);
+ aecpc->aec->outFile = fopen(filename, "wb");
+ sprintf(filename, "aec_out_linear%d.pcm", instance_count);
+ aecpc->aec->outLinearFile = fopen(filename, "wb");
+ sprintf(filename, "aec_buf%d.dat", instance_count);
+ aecpc->bufFile = fopen(filename, "wb");
+ sprintf(filename, "aec_skew%d.dat", instance_count);
+ aecpc->skewFile = fopen(filename, "wb");
+ sprintf(filename, "aec_delay%d.dat", instance_count);
+ aecpc->delayFile = fopen(filename, "wb");
+ instance_count++;
+ }
+#endif
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_Free(void *aecInst)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ WebRtc_FreeBuffer(aecpc->far_pre_buf);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_FreeBuffer(aecpc->far_pre_buf_s16);
+ fclose(aecpc->aec->farFile);
+ fclose(aecpc->aec->nearFile);
+ fclose(aecpc->aec->outFile);
+ fclose(aecpc->aec->outLinearFile);
+ fclose(aecpc->bufFile);
+ fclose(aecpc->skewFile);
+ fclose(aecpc->delayFile);
+#endif
+
+ WebRtcAec_FreeAec(aecpc->aec);
+ WebRtcAec_FreeResampler(aecpc->resampler);
+ free(aecpc);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
+{
+ aecpc_t *aecpc = aecInst;
+ AecConfig aecConfig;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->sampFreq = sampFreq;
+
+ if (scSampFreq < 1 || scSampFreq > 96000) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->scSampFreq = scSampFreq;
+
+ // Initialize echo canceller core
+ if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ if (WebRtc_InitBuffer(aecpc->far_pre_buf) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap.
+
+ aecpc->initFlag = initCheck; // indicates that initialization has been done
+
+ if (aecpc->sampFreq == 32000) {
+ aecpc->splitSampFreq = 16000;
+ }
+ else {
+ aecpc->splitSampFreq = sampFreq;
+ }
+
+ aecpc->skewFrCtr = 0;
+ aecpc->activity = 0;
+
+ aecpc->delayCtr = 0;
+
+ aecpc->sum = 0;
+ aecpc->counter = 0;
+ aecpc->checkBuffSize = 1;
+ aecpc->firstVal = 0;
+
+ aecpc->ECstartup = 1;
+ aecpc->bufSizeStart = 0;
+ aecpc->checkBufSizeCtr = 0;
+ aecpc->filtDelay = 0;
+ aecpc->timeForDelayChange = 0;
+ aecpc->knownDelay = 0;
+ aecpc->lastDelayDiff = 0;
+
+ aecpc->skew = 0;
+ aecpc->resample = kAecFalse;
+ aecpc->highSkewCtr = 0;
+ aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
+
+ // Default settings.
+ aecConfig.nlpMode = kAecNlpModerate;
+ aecConfig.skewMode = kAecFalse;
+ aecConfig.metricsMode = kAecFalse;
+ aecConfig.delay_logging = kAecFalse;
+
+ if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_InitBuffer(aecpc->far_pre_buf_s16) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN); // Start overlap.
+#endif
+
+ return 0;
+}
+
+// only buffer L band for farend
+WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
+ WebRtc_Word16 nrOfSamples)
+{
+ aecpc_t *aecpc = aecInst;
+ WebRtc_Word32 retVal = 0;
+ int newNrOfSamples = (int) nrOfSamples;
+ short newFarend[MAX_RESAMP_LEN];
+ const int16_t* farend_ptr = farend;
+ float tmp_farend[MAX_RESAMP_LEN];
+ const float* farend_float = tmp_farend;
+ float skew;
+ int i = 0;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (farend == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // number of samples == 160 for SWB input
+ if (nrOfSamples != 80 && nrOfSamples != 160) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ skew = aecpc->skew;
+
+ if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
+ // Resample and get a new number of samples
+ newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
+ farend,
+ nrOfSamples,
+ skew,
+ newFarend);
+ farend_ptr = (const int16_t*) newFarend;
+ }
+
+ aecpc->aec->system_delay += newNrOfSamples;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_WriteBuffer(aecpc->far_pre_buf_s16, farend_ptr,
+ (size_t) newNrOfSamples);
+#endif
+ // Cast to float and write the time-domain data to |far_pre_buf|.
+ for (i = 0; i < newNrOfSamples; i++) {
+ tmp_farend[i] = (float) farend_ptr[i];
+ }
+ WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_float,
+ (size_t) newNrOfSamples);
+
+ // Transform to frequency domain if we have enough data.
+ while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
+ // We have enough data to pass to the FFT, hence read PART_LEN2 samples.
+ WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**) &farend_float, tmp_farend,
+ PART_LEN2);
+
+ WebRtcAec_BufferFarendPartition(aecpc->aec, farend_float);
+
+ // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_ReadBuffer(aecpc->far_pre_buf_s16, (void**) &farend_ptr, newFarend,
+ PART_LEN2);
+ WebRtc_WriteBuffer(aecpc->aec->far_time_buf, &farend_ptr[PART_LEN], 1);
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN);
+#endif
+ }
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
+ const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
+ WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
+{
+ aecpc_t *aecpc = aecInst;
+ WebRtc_Word32 retVal = 0;
+ short i;
+ short nBlocks10ms;
+ short nFrames;
+ // Limit resampling to doubling/halving of signal
+ const float minSkewEst = -0.5f;
+ const float maxSkewEst = 1.0f;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (nearend == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (out == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // number of samples == 160 for SWB input
+ if (nrOfSamples != 80 && nrOfSamples != 160) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ // Check for valid pointers based on sampling rate
+ if (aecpc->sampFreq == 32000 && nearendH == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (msInSndCardBuf < 0) {
+ msInSndCardBuf = 0;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ }
+ else if (msInSndCardBuf > 500) {
+ msInSndCardBuf = 500;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ }
+ // TODO(andrew): we need to investigate if this +10 is really wanted.
+ msInSndCardBuf += 10;
+ aecpc->msInSndCardBuf = msInSndCardBuf;
+
+ if (aecpc->skewMode == kAecTrue) {
+ if (aecpc->skewFrCtr < 25) {
+ aecpc->skewFrCtr++;
+ }
+ else {
+ retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
+ if (retVal == -1) {
+ aecpc->skew = 0;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ }
+
+ aecpc->skew /= aecpc->sampFactor*nrOfSamples;
+
+ if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
+ aecpc->resample = kAecFalse;
+ }
+ else {
+ aecpc->resample = kAecTrue;
+ }
+
+ if (aecpc->skew < minSkewEst) {
+ aecpc->skew = minSkewEst;
+ }
+ else if (aecpc->skew > maxSkewEst) {
+ aecpc->skew = maxSkewEst;
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
+#endif
+ }
+ }
+
+ nFrames = nrOfSamples / FRAME_LEN;
+ nBlocks10ms = nFrames / aecpc->aec->mult;
+
+ if (aecpc->ECstartup) {
+ if (nearend != out) {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, nearend, sizeof(short) * nrOfSamples);
+ }
+
+ // The AEC is in the start up mode
+ // AEC is disabled until the system delay is OK
+
+ // Mechanism to ensure that the system delay is reasonably stable.
+ if (aecpc->checkBuffSize) {
+ aecpc->checkBufSizeCtr++;
+ // Before we fill up the far-end buffer we require the system delay
+ // to be stable (+/-8 ms) compared to the first value. This
+ // comparison is made during the following 6 consecutive 10 ms
+ // blocks. If it seems to be stable then we start to fill up the
+ // far-end buffer.
+ if (aecpc->counter == 0) {
+ aecpc->firstVal = aecpc->msInSndCardBuf;
+ aecpc->sum = 0;
+ }
+
+ if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
+ WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
+ aecpc->sum += aecpc->msInSndCardBuf;
+ aecpc->counter++;
+ }
+ else {
+ aecpc->counter = 0;
+ }
+
+ if (aecpc->counter * nBlocks10ms >= 6) {
+ // The far-end buffer size is determined in partitions of
+ // PART_LEN samples. Use 75% of the average value of the system
+ // delay as buffer size to start with.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
+ aecpc->aec->mult * 8) / (4 * aecpc->counter * PART_LEN),
+ kMaxBufSizeStart);
+ // Buffer size has now been determined.
+ aecpc->checkBuffSize = 0;
+ }
+
+ if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
+ // For really bad systems, don't disable the echo canceller for
+ // more than 0.5 sec.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
+ aecpc->aec->mult * 3) / 40, kMaxBufSizeStart);
+ aecpc->checkBuffSize = 0;
+ }
+ }
+
+ // If |checkBuffSize| changed in the if-statement above.
+ if (!aecpc->checkBuffSize) {
+ // The system delay is now reasonably stable (or has been unstable
+ // for too long). When the far-end buffer is filled with
+ // approximately the same amount of data as reported by the system
+ // we end the startup phase.
+ int overhead_elements = aecpc->aec->system_delay / PART_LEN -
+ aecpc->bufSizeStart;
+ if (overhead_elements == 0) {
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ } else if (overhead_elements > 0) {
+ WebRtc_MoveReadPtr(aecpc->aec->far_buf_windowed,
+ overhead_elements);
+ WebRtc_MoveReadPtr(aecpc->aec->far_buf, overhead_elements);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aecpc->aec->far_time_buf, overhead_elements);
+#endif
+ // TODO(bjornv): Do we need a check on how much we actually
+ // moved the read pointer? It should always be possible to move
+ // the pointer |overhead_elements| since we have only added data
+ // to the buffer and no delay compensation nor AEC processing
+ // has been done.
+ aecpc->aec->system_delay -= overhead_elements * PART_LEN;
+
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ }
+ }
+ } else {
+ // AEC is enabled.
+
+ int out_elements = 0;
+
+ EstBufDelay(aecpc);
+
+ // Note that 1 frame is supported for NB and 2 frames for WB.
+ for (i = 0; i < nFrames; i++) {
+ int16_t* out_ptr = NULL;
+ int16_t out_tmp[FRAME_LEN];
+
+ // Call the AEC.
+ WebRtcAec_ProcessFrame(aecpc->aec,
+ &nearend[FRAME_LEN * i],
+ &nearendH[FRAME_LEN * i],
+ aecpc->knownDelay);
+ // TODO(bjornv): Re-structure such that we don't have to pass
+ // |aecpc->knownDelay| as input. Change name to something like
+ // |system_buffer_diff|.
+
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ out_elements = (int) WebRtc_available_read(aecpc->aec->outFrBuf);
+ if (out_elements < FRAME_LEN) {
+ WebRtc_MoveReadPtr(aecpc->aec->outFrBuf,
+ out_elements - FRAME_LEN);
+ if (aecpc->sampFreq == 32000) {
+ WebRtc_MoveReadPtr(aecpc->aec->outFrBufH,
+ out_elements - FRAME_LEN);
+ }
+ }
+
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aecpc->aec->outFrBuf, (void**) &out_ptr,
+ out_tmp, FRAME_LEN);
+ memcpy(&out[FRAME_LEN * i], out_ptr, sizeof(int16_t) * FRAME_LEN);
+ // For H band
+ if (aecpc->sampFreq == 32000) {
+ WebRtc_ReadBuffer(aecpc->aec->outFrBufH, (void**) &out_ptr,
+ out_tmp, FRAME_LEN);
+ memcpy(&outH[FRAME_LEN * i], out_ptr,
+ sizeof(int16_t) * FRAME_LEN);
+ }
+ }
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t far_buf_size_ms = (int16_t) (aecpc->aec->system_delay /
+ (sampMsNb * aecpc->aec->mult));
+ fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
+ fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
+ }
+#endif
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->skewMode = config.skewMode;
+
+ if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
+ kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->nlpMode = config.nlpMode;
+ aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
+ aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
+
+ if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->aec->metricsMode = config.metricsMode;
+ if (aecpc->aec->metricsMode == kAecTrue) {
+ WebRtcAec_InitMetrics(aecpc->aec);
+ }
+
+ if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->aec->delay_logging_enabled = config.delay_logging;
+ if (aecpc->aec->delay_logging_enabled == kAecTrue) {
+ memset(aecpc->aec->delay_histogram, 0, sizeof(aecpc->aec->delay_histogram));
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (config == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ config->nlpMode = aecpc->nlpMode;
+ config->skewMode = aecpc->skewMode;
+ config->metricsMode = aecpc->aec->metricsMode;
+ config->delay_logging = aecpc->aec->delay_logging_enabled;
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (status == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ *status = aecpc->aec->echoState;
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
+{
+ const float upweight = 0.7f;
+ float dtmp;
+ short stmp;
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (metrics == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // ERL
+ metrics->erl.instant = (short) aecpc->aec->erl.instant;
+
+ if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
+ metrics->erl.average = (short) dtmp;
+ }
+ else {
+ metrics->erl.average = offsetLevel;
+ }
+
+ metrics->erl.max = (short) aecpc->aec->erl.max;
+
+ if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
+ metrics->erl.min = (short) aecpc->aec->erl.min;
+ }
+ else {
+ metrics->erl.min = offsetLevel;
+ }
+
+ // ERLE
+ metrics->erle.instant = (short) aecpc->aec->erle.instant;
+
+ if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
+ metrics->erle.average = (short) dtmp;
+ }
+ else {
+ metrics->erle.average = offsetLevel;
+ }
+
+ metrics->erle.max = (short) aecpc->aec->erle.max;
+
+ if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
+ metrics->erle.min = (short) aecpc->aec->erle.min;
+ } else {
+ metrics->erle.min = offsetLevel;
+ }
+
+ // RERL
+ if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
+ stmp = metrics->erl.average + metrics->erle.average;
+ }
+ else {
+ stmp = offsetLevel;
+ }
+ metrics->rerl.average = stmp;
+
+ // No other statistics needed, but returned for completeness
+ metrics->rerl.instant = stmp;
+ metrics->rerl.max = stmp;
+ metrics->rerl.min = stmp;
+
+ // A_NLP
+ metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
+
+ if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
+ metrics->aNlp.average = (short) dtmp;
+ }
+ else {
+ metrics->aNlp.average = offsetLevel;
+ }
+
+ metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
+
+ if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
+ metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
+ }
+ else {
+ metrics->aNlp.min = offsetLevel;
+ }
+
+ return 0;
+}
+
+int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
+ aecpc_t* self = handle;
+ int i = 0;
+ int delay_values = 0;
+ int num_delay_values = 0;
+ int my_median = 0;
+ const int kMsPerBlock = (PART_LEN * 1000) / self->splitSampFreq;
+ float l1_norm = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (median == NULL) {
+ self->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (std == NULL) {
+ self->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (self->initFlag != initCheck) {
+ self->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+ if (self->aec->delay_logging_enabled == 0) {
+ // Logging disabled
+ self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR;
+ return -1;
+ }
+
+ // Get number of delay values since last update
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ num_delay_values += self->aec->delay_histogram[i];
+ }
+ if (num_delay_values == 0) {
+ // We have no new delay value data. Even though -1 is a valid estimate, it
+ // will practically never be used since multiples of |kMsPerBlock| will
+ // always be returned.
+ *median = -1;
+ *std = -1;
+ return 0;
+ }
+
+ delay_values = num_delay_values >> 1; // Start value for median count down
+ // Get median of delay values since last update
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ delay_values -= self->aec->delay_histogram[i];
+ if (delay_values < 0) {
+ my_median = i;
+ break;
+ }
+ }
+ // Account for lookahead.
+ *median = (my_median - kLookaheadBlocks) * kMsPerBlock;
+
+ // Calculate the L1 norm, with median value as central moment
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ l1_norm += (float) (fabs(i - my_median) * self->aec->delay_histogram[i]);
+ }
+ *std = (int) (l1_norm / (float) num_delay_values + 0.5f) * kMsPerBlock;
+
+ // Reset histogram
+ memset(self->aec->delay_histogram, 0, sizeof(self->aec->delay_histogram));
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ return aecpc->lastError;
+}
+
+static int EstBufDelay(aecpc_t* aecpc) {
+ int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
+ int current_delay = nSampSndCard - aecpc->aec->system_delay;
+ int delay_difference = 0;
+
+ // Before we proceed with the delay estimate filtering we:
+ // 1) Compensate for the frame that will be read.
+ // 2) Compensate for drift resampling.
+
+ // 1) Compensating for the frame(s) that will be read/processed.
+ current_delay += FRAME_LEN * aecpc->aec->mult;
+
+ // 2) Account for resampling frame delay.
+ if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
+ current_delay -= kResamplingDelay;
+ }
+
+ aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
+ 0.2 * current_delay));
+
+ delay_difference = aecpc->filtDelay - aecpc->knownDelay;
+ if (delay_difference > 224) {
+ if (aecpc->lastDelayDiff < 96) {
+ aecpc->timeForDelayChange = 0;
+ } else {
+ aecpc->timeForDelayChange++;
+ }
+ } else if (delay_difference < 96 && aecpc->knownDelay > 0) {
+ if (aecpc->lastDelayDiff > 224) {
+ aecpc->timeForDelayChange = 0;
+ } else {
+ aecpc->timeForDelayChange++;
+ }
+ } else {
+ aecpc->timeForDelayChange = 0;
+ }
+ aecpc->lastDelayDiff = delay_difference;
+
+ if (aecpc->timeForDelayChange > 25) {
+ aecpc->knownDelay = WEBRTC_SPL_MAX((int) aecpc->filtDelay - 160, 0);
+ }
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h b/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h
new file mode 100644
index 0000000..a266e84
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h
@@ -0,0 +1,262 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+
+#include "typedefs.h"
+
+// Errors
+#define AEC_UNSPECIFIED_ERROR 12000
+#define AEC_UNSUPPORTED_FUNCTION_ERROR 12001
+#define AEC_UNINITIALIZED_ERROR 12002
+#define AEC_NULL_POINTER_ERROR 12003
+#define AEC_BAD_PARAMETER_ERROR 12004
+
+// Warnings
+#define AEC_BAD_PARAMETER_WARNING 12050
+
+enum {
+ kAecNlpConservative = 0,
+ kAecNlpModerate,
+ kAecNlpAggressive
+};
+
+enum {
+ kAecFalse = 0,
+ kAecTrue
+};
+
+typedef struct {
+ WebRtc_Word16 nlpMode; // default kAecNlpModerate
+ WebRtc_Word16 skewMode; // default kAecFalse
+ WebRtc_Word16 metricsMode; // default kAecFalse
+ int delay_logging; // default kAecFalse
+ //float realSkew;
+} AecConfig;
+
+typedef struct {
+ WebRtc_Word16 instant;
+ WebRtc_Word16 average;
+ WebRtc_Word16 max;
+ WebRtc_Word16 min;
+} AecLevel;
+
+typedef struct {
+ AecLevel rerl;
+ AecLevel erl;
+ AecLevel erle;
+ AecLevel aNlp;
+} AecMetrics;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Allocates the memory needed by the AEC. The memory needs to be initialized
+ * separately using the WebRtcAec_Init() function.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void **aecInst Pointer to the AEC instance to be created
+ * and initialized
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_Create(void **aecInst);
+
+/*
+ * This function releases the memory allocated by WebRtcAec_Create().
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_Free(void *aecInst);
+
+/*
+ * Initializes an AEC instance.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ * WebRtc_Word32 sampFreq Sampling frequency of data
+ * WebRtc_Word32 scSampFreq Soundcard sampling frequency
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_Init(void *aecInst,
+ WebRtc_Word32 sampFreq,
+ WebRtc_Word32 scSampFreq);
+
+/*
+ * Inserts an 80 or 160 sample block of data into the farend buffer.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ * WebRtc_Word16 *farend In buffer containing one frame of
+ * farend signal for L band
+ * WebRtc_Word16 nrOfSamples Number of samples in farend buffer
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
+ const WebRtc_Word16 *farend,
+ WebRtc_Word16 nrOfSamples);
+
+/*
+ * Runs the echo canceller on an 80 or 160 sample blocks of data.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ * WebRtc_Word16 *nearend In buffer containing one frame of
+ * nearend+echo signal for L band
+ * WebRtc_Word16 *nearendH In buffer containing one frame of
+ * nearend+echo signal for H band
+ * WebRtc_Word16 nrOfSamples Number of samples in nearend buffer
+ * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
+ * system buffers
+ * WebRtc_Word16 skew Difference between number of samples played
+ * and recorded at the soundcard (for clock skew
+ * compensation)
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16 *out Out buffer, one frame of processed nearend
+ * for L band
+ * WebRtc_Word16 *outH Out buffer, one frame of processed nearend
+ * for H band
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_Process(void *aecInst,
+ const WebRtc_Word16 *nearend,
+ const WebRtc_Word16 *nearendH,
+ WebRtc_Word16 *out,
+ WebRtc_Word16 *outH,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_Word16 msInSndCardBuf,
+ WebRtc_Word32 skew);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ * AecConfig config Config instance that contains all
+ * properties to be set
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
+
+/*
+ * Gets the on-the-fly paramters.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * AecConfig *config Pointer to the config instance that
+ * all properties will be written to
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
+
+/*
+ * Gets the current echo status of the nearend signal.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16 *status 0: Almost certainly nearend single-talk
+ * 1: Might not be neared single-talk
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
+
+/*
+ * Gets the current echo metrics for the session.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * AecMetrics *metrics Struct which will be filled out with the
+ * current echo metrics.
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
+
+/*
+ * Gets the current delay metrics for the session.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* handle Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * int* median Delay median value.
+ * int* std Delay standard deviation.
+ *
+ * int return 0: OK
+ * -1: error
+ */
+int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std);
+
+/*
+ * Gets the last error code.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecInst Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 11000-11100: error code
+ */
+WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
+
+#ifdef __cplusplus
+}
+#endif
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
diff --git a/trunk/src/modules/audio_processing/aecm/Android.mk b/trunk/src/modules/audio_processing/aecm/Android.mk
new file mode 100644
index 0000000..2d64b85
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/Android.mk
@@ -0,0 +1,78 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+#############################
+# Build the non-neon library.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aecm
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ echo_control_mobile.c \
+ aecm_core.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#########################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aecm_neon
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := aecm_core_neon.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ -mfpu=neon \
+ -mfloat-abi=softfp \
+ -flax-vector-conversions
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/trunk/src/modules/audio_processing/aecm/aecm.gypi b/trunk/src/modules/audio_processing/aecm/aecm.gypi
new file mode 100644
index 0000000..3e4a113
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/aecm.gypi
@@ -0,0 +1,35 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'aecm',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ 'apm_util'
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/echo_control_mobile.h',
+ 'echo_control_mobile.c',
+ 'aecm_core.c',
+ 'aecm_core.h',
+ ],
+ },
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core.c b/trunk/src/modules/audio_processing/aecm/aecm_core.c
new file mode 100644
index 0000000..4a51d56
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/aecm_core.c
@@ -0,0 +1,2131 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aecm_core.h"
+
+#include <assert.h>
+#include <stdlib.h>
+
+#include "cpu_features_wrapper.h"
+#include "delay_estimator_wrapper.h"
+#include "echo_control_mobile.h"
+#include "ring_buffer.h"
+#include "typedefs.h"
+
+#ifdef ARM_WINM_LOG
+#include <stdio.h>
+#include <windows.h>
+#endif
+
+#ifdef AEC_DEBUG
+FILE *dfile;
+FILE *testfile;
+#endif
+
+#ifdef _MSC_VER // visual c++
+#define ALIGN8_BEG __declspec(align(8))
+#define ALIGN8_END
+#else // gcc or icc
+#define ALIGN8_BEG
+#define ALIGN8_END __attribute__((aligned(8)))
+#endif
+
+#ifdef AECM_SHORT
+
+// Square root of Hanning window in Q14
+const WebRtc_Word16 WebRtcAecm_kSqrtHanning[] =
+{
+ 0, 804, 1606, 2404, 3196, 3981, 4756, 5520,
+ 6270, 7005, 7723, 8423, 9102, 9760, 10394, 11003,
+ 11585, 12140, 12665, 13160, 13623, 14053, 14449, 14811,
+ 15137, 15426, 15679, 15893, 16069, 16207, 16305, 16364,
+ 16384
+};
+
+#else
+
+// Square root of Hanning window in Q14
+const ALIGN8_BEG WebRtc_Word16 WebRtcAecm_kSqrtHanning[] ALIGN8_END =
+{
+ 0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
+ 3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019, 8364,
+ 8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514, 11795, 12068, 12335,
+ 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189, 14384, 14571, 14749, 14918,
+ 15079, 15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034, 16111, 16179, 16237,
+ 16286, 16325, 16354, 16373, 16384
+};
+
+#endif
+
+//Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha1 = 32584;
+//Q15 beta = 0.12967166976970 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta1 = 4249;
+//Q15 alpha = 0.94234827210087 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha2 = 30879;
+//Q15 beta = 0.33787806009150 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta2 = 11072;
+//Q15 alpha = 0.82247698684306 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha3 = 26951;
+//Q15 beta = 0.57762063060713 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta3 = 18927;
+
+// Initialization table for echo channel in 8 kHz
+static const WebRtc_Word16 kChannelStored8kHz[PART_LEN1] = {
+ 2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418,
+ 1451, 1506, 1562, 1644, 1726, 1804, 1882, 1918,
+ 1953, 1982, 2010, 2025, 2040, 2034, 2027, 2021,
+ 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683,
+ 1635, 1604, 1572, 1545, 1517, 1481, 1444, 1405,
+ 1367, 1331, 1294, 1270, 1245, 1239, 1233, 1247,
+ 1260, 1282, 1303, 1338, 1373, 1407, 1441, 1470,
+ 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649,
+ 1676
+};
+
+// Initialization table for echo channel in 16 kHz
+static const WebRtc_Word16 kChannelStored16kHz[PART_LEN1] = {
+ 2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882,
+ 1953, 2010, 2040, 2027, 2014, 1980, 1869, 1732,
+ 1635, 1572, 1517, 1444, 1367, 1294, 1245, 1233,
+ 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621,
+ 1676, 1741, 1802, 1861, 1921, 1983, 2040, 2102,
+ 2170, 2265, 2375, 2515, 2651, 2781, 2922, 3075,
+ 3253, 3471, 3738, 3976, 4151, 4258, 4308, 4288,
+ 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484,
+ 3153
+};
+
+static const WebRtc_Word16 kCosTable[] = {
+ 8192, 8190, 8187, 8180, 8172, 8160, 8147, 8130, 8112,
+ 8091, 8067, 8041, 8012, 7982, 7948, 7912, 7874, 7834,
+ 7791, 7745, 7697, 7647, 7595, 7540, 7483, 7424, 7362,
+ 7299, 7233, 7164, 7094, 7021, 6947, 6870, 6791, 6710,
+ 6627, 6542, 6455, 6366, 6275, 6182, 6087, 5991, 5892,
+ 5792, 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
+ 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971, 3845,
+ 3719, 3591, 3462, 3331, 3200, 3068, 2935, 2801, 2667,
+ 2531, 2395, 2258, 2120, 1981, 1842, 1703, 1563, 1422,
+ 1281, 1140, 998, 856, 713, 571, 428, 285, 142,
+ 0, -142, -285, -428, -571, -713, -856, -998, -1140,
+ -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
+ -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
+ -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
+ -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
+ -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
+ -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
+ -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
+ -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
+ -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
+ -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
+ -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
+ -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
+ -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
+ -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
+ -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
+ -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
+ -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
+ -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
+ -1281, -1140, -998, -856, -713, -571, -428, -285, -142,
+ 0, 142, 285, 428, 571, 713, 856, 998, 1140,
+ 1281, 1422, 1563, 1703, 1842, 1981, 2120, 2258, 2395,
+ 2531, 2667, 2801, 2935, 3068, 3200, 3331, 3462, 3591,
+ 3719, 3845, 3971, 4095, 4219, 4341, 4461, 4580, 4698,
+ 4815, 4930, 5043, 5155, 5265, 5374, 5481, 5586, 5690,
+ 5792, 5892, 5991, 6087, 6182, 6275, 6366, 6455, 6542,
+ 6627, 6710, 6791, 6870, 6947, 7021, 7094, 7164, 7233,
+ 7299, 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
+ 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041, 8067,
+ 8091, 8112, 8130, 8147, 8160, 8172, 8180, 8187, 8190
+};
+
+static const WebRtc_Word16 kSinTable[] = {
+ 0, 142, 285, 428, 571, 713, 856, 998,
+ 1140, 1281, 1422, 1563, 1703, 1842, 1981, 2120,
+ 2258, 2395, 2531, 2667, 2801, 2935, 3068, 3200,
+ 3331, 3462, 3591, 3719, 3845, 3971, 4095, 4219,
+ 4341, 4461, 4580, 4698, 4815, 4930, 5043, 5155,
+ 5265, 5374, 5481, 5586, 5690, 5792, 5892, 5991,
+ 6087, 6182, 6275, 6366, 6455, 6542, 6627, 6710,
+ 6791, 6870, 6947, 7021, 7094, 7164, 7233, 7299,
+ 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
+ 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041,
+ 8067, 8091, 8112, 8130, 8147, 8160, 8172, 8180,
+ 8187, 8190, 8191, 8190, 8187, 8180, 8172, 8160,
+ 8147, 8130, 8112, 8091, 8067, 8041, 8012, 7982,
+ 7948, 7912, 7874, 7834, 7791, 7745, 7697, 7647,
+ 7595, 7540, 7483, 7424, 7362, 7299, 7233, 7164,
+ 7094, 7021, 6947, 6870, 6791, 6710, 6627, 6542,
+ 6455, 6366, 6275, 6182, 6087, 5991, 5892, 5792,
+ 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
+ 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971,
+ 3845, 3719, 3591, 3462, 3331, 3200, 3068, 2935,
+ 2801, 2667, 2531, 2395, 2258, 2120, 1981, 1842,
+ 1703, 1563, 1422, 1281, 1140, 998, 856, 713,
+ 571, 428, 285, 142, 0, -142, -285, -428,
+ -571, -713, -856, -998, -1140, -1281, -1422, -1563,
+ -1703, -1842, -1981, -2120, -2258, -2395, -2531, -2667,
+ -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719,
+ -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
+ -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586,
+ -5690, -5792, -5892, -5991, -6087, -6182, -6275, -6366,
+ -6455, -6542, -6627, -6710, -6791, -6870, -6947, -7021,
+ -7094, -7164, -7233, -7299, -7362, -7424, -7483, -7540,
+ -7595, -7647, -7697, -7745, -7791, -7834, -7874, -7912,
+ -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130,
+ -8147, -8160, -8172, -8180, -8187, -8190, -8191, -8190,
+ -8187, -8180, -8172, -8160, -8147, -8130, -8112, -8091,
+ -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
+ -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424,
+ -7362, -7299, -7233, -7164, -7094, -7021, -6947, -6870,
+ -6791, -6710, -6627, -6542, -6455, -6366, -6275, -6182,
+ -6087, -5991, -5892, -5792, -5690, -5586, -5481, -5374,
+ -5265, -5155, -5043, -4930, -4815, -4698, -4580, -4461,
+ -4341, -4219, -4096, -3971, -3845, -3719, -3591, -3462,
+ -3331, -3200, -3068, -2935, -2801, -2667, -2531, -2395,
+ -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
+ -1140, -998, -856, -713, -571, -428, -285, -142
+};
+
+static const WebRtc_Word16 kNoiseEstQDomain = 15;
+static const WebRtc_Word16 kNoiseEstIncCount = 5;
+
+static void ComfortNoise(AecmCore_t* aecm,
+ const WebRtc_UWord16* dfa,
+ complex16_t* out,
+ const WebRtc_Word16* lambda);
+
+static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm);
+
+// Moves the pointer to the next entry and inserts |far_spectrum| and
+// corresponding Q-domain in its buffer.
+//
+// Inputs:
+// - self : Pointer to the delay estimation instance
+// - far_spectrum : Pointer to the far end spectrum
+// - far_q : Q-domain of far end spectrum
+//
+static void UpdateFarHistory(AecmCore_t* self,
+ uint16_t* far_spectrum,
+ int far_q) {
+ // Get new buffer position
+ self->far_history_pos++;
+ if (self->far_history_pos >= MAX_DELAY) {
+ self->far_history_pos = 0;
+ }
+ // Update Q-domain buffer
+ self->far_q_domains[self->far_history_pos] = far_q;
+ // Update far end spectrum buffer
+ memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]),
+ far_spectrum,
+ sizeof(uint16_t) * PART_LEN1);
+}
+
+// Returns a pointer to the far end spectrum aligned to current near end
+// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
+// called before AlignedFarend(...). Otherwise, you get the pointer to the
+// previous frame. The memory is only valid until the next call of
+// WebRtc_DelayEstimatorProcessFix(...).
+//
+// Inputs:
+// - self : Pointer to the AECM instance.
+// - delay : Current delay estimate.
+//
+// Output:
+// - far_q : The Q-domain of the aligned far end spectrum
+//
+// Return value:
+// - far_spectrum : Pointer to the aligned far end spectrum
+// NULL - Error
+//
+static const uint16_t* AlignedFarend(AecmCore_t* self, int* far_q, int delay) {
+ int buffer_position = 0;
+ assert(self != NULL);
+ buffer_position = self->far_history_pos - delay;
+
+ // Check buffer position
+ if (buffer_position < 0) {
+ buffer_position += MAX_DELAY;
+ }
+ // Get Q-domain
+ *far_q = self->far_q_domains[buffer_position];
+ // Return far end spectrum
+ return &(self->far_history[buffer_position * PART_LEN1]);
+}
+
+#ifdef ARM_WINM_LOG
+HANDLE logFile = NULL;
+#endif
+
+// Declare function pointers.
+CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+WindowAndFFT WebRtcAecm_WindowAndFFT;
+InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
+
+int WebRtcAecm_CreateCore(AecmCore_t **aecmInst)
+{
+ AecmCore_t *aecm = malloc(sizeof(AecmCore_t));
+ *aecmInst = aecm;
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateDelayEstimator(&aecm->delay_estimator,
+ PART_LEN1,
+ MAX_DELAY,
+ 0) == -1) {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ // Init some aecm pointers. 16 and 32 byte alignment is only necessary
+ // for Neon code currently.
+ aecm->xBuf = (WebRtc_Word16*) (((uintptr_t)aecm->xBuf_buf + 31) & ~ 31);
+ aecm->dBufClean = (WebRtc_Word16*) (((uintptr_t)aecm->dBufClean_buf + 31) & ~ 31);
+ aecm->dBufNoisy = (WebRtc_Word16*) (((uintptr_t)aecm->dBufNoisy_buf + 31) & ~ 31);
+ aecm->outBuf = (WebRtc_Word16*) (((uintptr_t)aecm->outBuf_buf + 15) & ~ 15);
+ aecm->channelStored = (WebRtc_Word16*) (((uintptr_t)
+ aecm->channelStored_buf + 15) & ~ 15);
+ aecm->channelAdapt16 = (WebRtc_Word16*) (((uintptr_t)
+ aecm->channelAdapt16_buf + 15) & ~ 15);
+ aecm->channelAdapt32 = (WebRtc_Word32*) (((uintptr_t)
+ aecm->channelAdapt32_buf + 31) & ~ 31);
+
+ return 0;
+}
+
+void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path)
+{
+ int i = 0;
+
+ // Reset the stored channel
+ memcpy(aecm->channelStored, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
+ // Reset the adapted channels
+ memcpy(aecm->channelAdapt16, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)(aecm->channelAdapt16[i]), 16);
+ }
+
+ // Reset channel storing variables
+ aecm->mseAdaptOld = 1000;
+ aecm->mseStoredOld = 1000;
+ aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
+ aecm->mseChannelCount = 0;
+}
+
+static void WindowAndFFTC(WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling)
+{
+ int i, j;
+
+ memset(fft, 0, sizeof(WebRtc_Word16) * PART_LEN4);
+ // FFT of signal
+ for (i = 0, j = 0; i < PART_LEN; i++, j += 2)
+ {
+ // Window time domain signal and insert into real part of
+ // transformation array |fft|
+ fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (time_signal[i] << time_signal_scaling),
+ WebRtcAecm_kSqrtHanning[i],
+ 14);
+ fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (time_signal[i + PART_LEN] << time_signal_scaling),
+ WebRtcAecm_kSqrtHanning[PART_LEN - i],
+ 14);
+ // Inserting zeros in imaginary parts not necessary since we
+ // initialized the array with all zeros
+ }
+
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the first PART_LEN2 samples
+ for (i = 0, j = 0; j < PART_LEN2; i += 1, j += 2)
+ {
+ freq_signal[i].real = fft[j];
+
+ // The imaginary part has to switch sign
+ freq_signal[i].imag = - fft[j+1];
+ }
+}
+
+static void InverseFFTAndWindowC(AecmCore_t* aecm,
+ WebRtc_Word16* fft,
+ complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean)
+{
+ int i, j, outCFFT;
+ WebRtc_Word32 tmp32no1;
+
+ // Synthesis
+ for (i = 1; i < PART_LEN; i++)
+ {
+ j = WEBRTC_SPL_LSHIFT_W32(i, 1);
+ fft[j] = efw[i].real;
+
+ // mirrored data, even
+ fft[PART_LEN4 - j] = efw[i].real;
+ fft[j + 1] = -efw[i].imag;
+
+ //mirrored data, odd
+ fft[PART_LEN4 - (j - 1)] = efw[i].imag;
+ }
+ fft[0] = efw[0].real;
+ fft[1] = -efw[0].imag;
+
+ fft[PART_LEN2] = efw[PART_LEN].real;
+ fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
+
+ // inverse FFT, result should be scaled with outCFFT
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
+
+ //take only the real values and scale with outCFFT
+ for (i = 0; i < PART_LEN2; i++)
+ {
+ j = WEBRTC_SPL_LSHIFT_W32(i, 1);
+ fft[i] = fft[j];
+ }
+
+ for (i = 0; i < PART_LEN; i++)
+ {
+ fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ fft[i],
+ WebRtcAecm_kSqrtHanning[i],
+ 14);
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
+ outCFFT - aecm->dfaCleanQDomain);
+ fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ tmp32no1 + aecm->outBuf[i],
+ WEBRTC_SPL_WORD16_MIN);
+ output[i] = fft[i];
+
+ tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
+ fft[PART_LEN + i],
+ WebRtcAecm_kSqrtHanning[PART_LEN - i],
+ 14);
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
+ outCFFT - aecm->dfaCleanQDomain);
+ aecm->outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX,
+ tmp32no1,
+ WEBRTC_SPL_WORD16_MIN);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+#endif
+
+ // Copy the current block to the old position (aecm->outBuf is shifted elsewhere)
+ memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ if (nearendClean != NULL)
+ {
+ memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ }
+}
+
+static void CalcLinearEnergiesC(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored)
+{
+ int i;
+
+ // Get energy for the delayed far end signal and estimated
+ // echo using both stored and adapted channels.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+ (*far_energy) += (WebRtc_UWord32)(far_spectrum[i]);
+ (*echo_energy_adapt) += WEBRTC_SPL_UMUL_16_16(aecm->channelAdapt16[i],
+ far_spectrum[i]);
+ (*echo_energy_stored) += (WebRtc_UWord32)echo_est[i];
+ }
+}
+
+static void StoreAdaptiveChannelC(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est)
+{
+ int i;
+
+ // During startup we store the channel every block.
+ memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
+ // Recalculate echo estimate
+ for (i = 0; i < PART_LEN; i += 4)
+ {
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+ echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
+ far_spectrum[i + 1]);
+ echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
+ far_spectrum[i + 2]);
+ echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
+ far_spectrum[i + 3]);
+ }
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+}
+
+static void ResetAdaptiveChannelC(AecmCore_t* aecm)
+{
+ int i;
+
+ // The stored channel has a significantly lower MSE than the adaptive one for
+ // two consecutive calculations. Reset the adaptive channel.
+ memcpy(aecm->channelAdapt16, aecm->channelStored,
+ sizeof(WebRtc_Word16) * PART_LEN1);
+ // Restore the W32 channel
+ for (i = 0; i < PART_LEN; i += 4)
+ {
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i], 16);
+ aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 1], 16);
+ aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 2], 16);
+ aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 3], 16);
+ }
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
+}
+
+// WebRtcAecm_InitCore(...)
+//
+// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
+// Input:
+// - aecm : Pointer to the Echo Suppression instance
+// - samplingFreq : Sampling Frequency
+//
+// Output:
+// - aecm : Initialized instance
+//
+// Return value : 0 - Ok
+// -1 - Error
+//
+int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq)
+{
+ int i = 0;
+ WebRtc_Word32 tmp32 = PART_LEN1 * PART_LEN1;
+ WebRtc_Word16 tmp16 = PART_LEN1;
+
+ if (samplingFreq != 8000 && samplingFreq != 16000)
+ {
+ samplingFreq = 8000;
+ return -1;
+ }
+ // sanity check of sampling frequency
+ aecm->mult = (WebRtc_Word16)samplingFreq / 8000;
+
+ aecm->farBufWritePos = 0;
+ aecm->farBufReadPos = 0;
+ aecm->knownDelay = 0;
+ aecm->lastKnownDelay = 0;
+
+ WebRtc_InitBuffer(aecm->farFrameBuf);
+ WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
+ WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
+ WebRtc_InitBuffer(aecm->outFrameBuf);
+
+ memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
+ memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
+ memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
+ memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
+
+ aecm->seed = 666;
+ aecm->totCount = 0;
+
+ if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
+ return -1;
+ }
+ // Set far end histories to zero
+ memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
+ memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
+ aecm->far_history_pos = MAX_DELAY;
+
+ aecm->nlpFlag = 1;
+ aecm->fixedDelay = -1;
+
+ aecm->dfaCleanQDomain = 0;
+ aecm->dfaCleanQDomainOld = 0;
+ aecm->dfaNoisyQDomain = 0;
+ aecm->dfaNoisyQDomainOld = 0;
+
+ memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
+ aecm->farLogEnergy = 0;
+ memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
+ memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
+
+ // Initialize the echo channels with a stored shape.
+ if (samplingFreq == 8000)
+ {
+ WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
+ }
+ else
+ {
+ WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
+ }
+
+ memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
+ memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
+ aecm->noiseEstCtr = 0;
+
+ aecm->cngMode = AecmTrue;
+
+ memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
+ memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
+ // Shape the initial noise level to an approximate pink noise.
+ for (i = 0; i < (PART_LEN1 >> 1) - 1; i++)
+ {
+ aecm->noiseEst[i] = (tmp32 << 8);
+ tmp16--;
+ tmp32 -= (WebRtc_Word32)((tmp16 << 1) + 1);
+ }
+ for (; i < PART_LEN1; i++)
+ {
+ aecm->noiseEst[i] = (tmp32 << 8);
+ }
+
+ aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
+ aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
+ aecm->farEnergyMaxMin = 0;
+ aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
+ // beginning.
+ aecm->farEnergyMSE = 0;
+ aecm->currentVADValue = 0;
+ aecm->vadUpdateCount = 0;
+ aecm->firstVAD = 1;
+
+ aecm->startupState = 0;
+ aecm->supGain = SUPGAIN_DEFAULT;
+ aecm->supGainOld = SUPGAIN_DEFAULT;
+
+ aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
+ aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
+ aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
+ aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
+
+ assert(PART_LEN % 16 == 0);
+
+ // Initialize function pointers.
+ WebRtcAecm_WindowAndFFT = WindowAndFFTC;
+ WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowC;
+ WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
+ WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
+ WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
+
+#ifdef WEBRTC_DETECT_ARM_NEON
+ uint64_t features = WebRtc_GetCPUFeaturesARM();
+ if ((features & kCPUFeatureNEON) != 0)
+ {
+ WebRtcAecm_InitNeon();
+ }
+#elif defined(WEBRTC_ARCH_ARM_NEON)
+ WebRtcAecm_InitNeon();
+#endif
+
+ return 0;
+}
+
+// TODO(bjornv): This function is currently not used. Add support for these
+// parameters from a higher level
+int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag)
+{
+ aecm->nlpFlag = nlpFlag;
+ aecm->fixedDelay = delay;
+
+ return 0;
+}
+
+int WebRtcAecm_FreeCore(AecmCore_t *aecm)
+{
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ WebRtc_FreeBuffer(aecm->farFrameBuf);
+ WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
+ WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
+ WebRtc_FreeBuffer(aecm->outFrameBuf);
+
+ WebRtc_FreeDelayEstimator(aecm->delay_estimator);
+ free(aecm);
+
+ return 0;
+}
+
+int WebRtcAecm_ProcessFrame(AecmCore_t * aecm,
+ const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * out)
+{
+ WebRtc_Word16 outBlock_buf[PART_LEN + 8]; // Align buffer to 8-byte boundary.
+ WebRtc_Word16* outBlock = (WebRtc_Word16*) (((uintptr_t) outBlock_buf + 15) & ~ 15);
+
+ WebRtc_Word16 farFrame[FRAME_LEN];
+ const int16_t* out_ptr = NULL;
+ int size = 0;
+
+ // Buffer the current frame.
+ // Fetch an older one corresponding to the delay.
+ WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
+ WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
+
+ // Buffer the synchronized far and near frames,
+ // to pass the smaller blocks individually.
+ WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
+ WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
+ if (nearendClean != NULL)
+ {
+ WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
+ }
+
+ // Process as many blocks as possible.
+ while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN)
+ {
+ int16_t far_block[PART_LEN];
+ const int16_t* far_block_ptr = NULL;
+ int16_t near_noisy_block[PART_LEN];
+ const int16_t* near_noisy_block_ptr = NULL;
+
+ WebRtc_ReadBuffer(aecm->farFrameBuf, (void**) &far_block_ptr, far_block,
+ PART_LEN);
+ WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf,
+ (void**) &near_noisy_block_ptr,
+ near_noisy_block,
+ PART_LEN);
+ if (nearendClean != NULL)
+ {
+ int16_t near_clean_block[PART_LEN];
+ const int16_t* near_clean_block_ptr = NULL;
+
+ WebRtc_ReadBuffer(aecm->nearCleanFrameBuf,
+ (void**) &near_clean_block_ptr,
+ near_clean_block,
+ PART_LEN);
+ if (WebRtcAecm_ProcessBlock(aecm,
+ far_block_ptr,
+ near_noisy_block_ptr,
+ near_clean_block_ptr,
+ outBlock) == -1)
+ {
+ return -1;
+ }
+ } else
+ {
+ if (WebRtcAecm_ProcessBlock(aecm,
+ far_block_ptr,
+ near_noisy_block_ptr,
+ NULL,
+ outBlock) == -1)
+ {
+ return -1;
+ }
+ }
+
+ WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
+ }
+
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ size = (int) WebRtc_available_read(aecm->outFrameBuf);
+ if (size < FRAME_LEN)
+ {
+ WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
+ }
+
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aecm->outFrameBuf, (void**) &out_ptr, out, FRAME_LEN);
+ if (out_ptr != out) {
+ // ReadBuffer() hasn't copied to |out| in this case.
+ memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
+ }
+
+ return 0;
+}
+
+// WebRtcAecm_AsymFilt(...)
+//
+// Performs asymmetric filtering.
+//
+// Inputs:
+// - filtOld : Previous filtered value.
+// - inVal : New input value.
+// - stepSizePos : Step size when we have a positive contribution.
+// - stepSizeNeg : Step size when we have a negative contribution.
+//
+// Output:
+//
+// Return: - Filtered value.
+//
+WebRtc_Word16 WebRtcAecm_AsymFilt(const WebRtc_Word16 filtOld, const WebRtc_Word16 inVal,
+ const WebRtc_Word16 stepSizePos,
+ const WebRtc_Word16 stepSizeNeg)
+{
+ WebRtc_Word16 retVal;
+
+ if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
+ {
+ return inVal;
+ }
+ retVal = filtOld;
+ if (filtOld > inVal)
+ {
+ retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
+ } else
+ {
+ retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
+ }
+
+ return retVal;
+}
+
+// WebRtcAecm_CalcEnergies(...)
+//
+// This function calculates the log of energies for nearend, farend and estimated
+// echoes. There is also an update of energy decision levels, i.e. internal VAD.
+//
+//
+// @param aecm [i/o] Handle of the AECM instance.
+// @param far_spectrum [in] Pointer to farend spectrum.
+// @param far_q [in] Q-domain of farend spectrum.
+// @param nearEner [in] Near end energy for current block in
+// Q(aecm->dfaQDomain).
+// @param echoEst [out] Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
+ const WebRtc_UWord16* far_spectrum,
+ const WebRtc_Word16 far_q,
+ const WebRtc_UWord32 nearEner,
+ WebRtc_Word32 * echoEst)
+{
+ // Local variables
+ WebRtc_UWord32 tmpAdapt = 0;
+ WebRtc_UWord32 tmpStored = 0;
+ WebRtc_UWord32 tmpFar = 0;
+
+ int i;
+
+ WebRtc_Word16 zeros, frac;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word16 increase_max_shifts = 4;
+ WebRtc_Word16 decrease_max_shifts = 11;
+ WebRtc_Word16 increase_min_shifts = 11;
+ WebRtc_Word16 decrease_min_shifts = 3;
+ WebRtc_Word16 kLogLowValue = WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
+
+ // Get log of near end energy and store in buffer
+
+ // Shift buffer
+ memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+
+ // Logarithm of integrated magnitude spectrum (nearEner)
+ tmp16 = kLogLowValue;
+ if (nearEner)
+ {
+ zeros = WebRtcSpl_NormU32(nearEner);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
+ (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
+ 23);
+ // log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
+ }
+ aecm->nearLogEnergy[0] = tmp16;
+ // END: Get log of near end energy
+
+ WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt, &tmpStored);
+
+ // Shift buffers
+ memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+ memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+
+ // Logarithm of delayed far end energy
+ tmp16 = kLogLowValue;
+ if (tmpFar)
+ {
+ zeros = WebRtcSpl_NormU32(tmpFar);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
+ & 0x7FFFFFFF), 23);
+ // log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8);
+ }
+ aecm->farLogEnergy = tmp16;
+
+ // Logarithm of estimated echo energy through adapted channel
+ tmp16 = kLogLowValue;
+ if (tmpAdapt)
+ {
+ zeros = WebRtcSpl_NormU32(tmpAdapt);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
+ & 0x7FFFFFFF), 23);
+ //log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
+ }
+ aecm->echoAdaptLogEnergy[0] = tmp16;
+
+ // Logarithm of estimated echo energy through stored channel
+ tmp16 = kLogLowValue;
+ if (tmpStored)
+ {
+ zeros = WebRtcSpl_NormU32(tmpStored);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
+ & 0x7FFFFFFF), 23);
+ //log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
+ }
+ aecm->echoStoredLogEnergy[0] = tmp16;
+
+ // Update farend energy levels (min, max, vad, mse)
+ if (aecm->farLogEnergy > FAR_ENERGY_MIN)
+ {
+ if (aecm->startupState == 0)
+ {
+ increase_max_shifts = 2;
+ decrease_min_shifts = 2;
+ increase_min_shifts = 8;
+ }
+
+ aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
+ increase_min_shifts, decrease_min_shifts);
+ aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
+ increase_max_shifts, decrease_max_shifts);
+ aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
+
+ // Dynamic VAD region size
+ tmp16 = 2560 - aecm->farEnergyMin;
+ if (tmp16 > 0)
+ {
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9);
+ } else
+ {
+ tmp16 = 0;
+ }
+ tmp16 += FAR_ENERGY_VAD_REGION;
+
+ if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
+ {
+ // In startup phase or VAD update halted
+ aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
+ } else
+ {
+ if (aecm->farEnergyVAD > aecm->farLogEnergy)
+ {
+ aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy +
+ tmp16 -
+ aecm->farEnergyVAD,
+ 6);
+ aecm->vadUpdateCount = 0;
+ } else
+ {
+ aecm->vadUpdateCount++;
+ }
+ }
+ // Put MSE threshold higher than VAD
+ aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
+ }
+
+ // Update VAD variables
+ if (aecm->farLogEnergy > aecm->farEnergyVAD)
+ {
+ if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
+ {
+ // We are in startup or have significant dynamics in input speech level
+ aecm->currentVADValue = 1;
+ }
+ } else
+ {
+ aecm->currentVADValue = 0;
+ }
+ if ((aecm->currentVADValue) && (aecm->firstVAD))
+ {
+ aecm->firstVAD = 0;
+ if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
+ {
+ // The estimated echo has higher energy than the near end signal.
+ // This means that the initialization was too aggressive. Scale
+ // down by a factor 8
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ aecm->channelAdapt16[i] >>= 3;
+ }
+ // Compensate the adapted echo energy level accordingly.
+ aecm->echoAdaptLogEnergy[0] -= (3 << 8);
+ aecm->firstVAD = 1;
+ }
+ }
+}
+
+// WebRtcAecm_CalcStepSize(...)
+//
+// This function calculates the step size used in channel estimation
+//
+//
+// @param aecm [in] Handle of the AECM instance.
+// @param mu [out] (Return value) Stepsize in log2(), i.e. number of shifts.
+//
+//
+WebRtc_Word16 WebRtcAecm_CalcStepSize(AecmCore_t * const aecm)
+{
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word16 mu = MU_MAX;
+
+ // Here we calculate the step size mu used in the
+ // following NLMS based Channel estimation algorithm
+ if (!aecm->currentVADValue)
+ {
+ // Far end energy level too low, no channel update
+ mu = 0;
+ } else if (aecm->startupState > 0)
+ {
+ if (aecm->farEnergyMin >= aecm->farEnergyMax)
+ {
+ mu = MU_MIN;
+ } else
+ {
+ tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
+ tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, MU_DIFF);
+ tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
+ mu = MU_MIN - 1 - (WebRtc_Word16)(tmp32);
+ // The -1 is an alternative to rounding. This way we get a larger
+ // stepsize, so we in some sense compensate for truncation in NLMS
+ }
+ if (mu < MU_MAX)
+ {
+ mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
+ }
+ }
+
+ return mu;
+}
+
+// WebRtcAecm_UpdateChannel(...)
+//
+// This function performs channel estimation. NLMS and decision on channel storage.
+//
+//
+// @param aecm [i/o] Handle of the AECM instance.
+// @param far_spectrum [in] Absolute value of the farend signal in Q(far_q)
+// @param far_q [in] Q-domain of the farend signal
+// @param dfa [in] Absolute value of the nearend signal (Q[aecm->dfaQDomain])
+// @param mu [in] NLMS step size.
+// @param echoEst [i/o] Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_UpdateChannel(AecmCore_t * aecm,
+ const WebRtc_UWord16* far_spectrum,
+ const WebRtc_Word16 far_q,
+ const WebRtc_UWord16 * const dfa,
+ const WebRtc_Word16 mu,
+ WebRtc_Word32 * echoEst)
+{
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 mseStored;
+ WebRtc_Word32 mseAdapt;
+
+ int i;
+
+ WebRtc_Word16 zerosFar, zerosNum, zerosCh, zerosDfa;
+ WebRtc_Word16 shiftChFar, shiftNum, shift2ResChan;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 xfaQ, dfaQ;
+
+ // This is the channel estimation algorithm. It is base on NLMS but has a variable step
+ // length, which was calculated above.
+ if (mu)
+ {
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Determine norm of channel and farend to make sure we don't get overflow in
+ // multiplication
+ zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
+ zerosFar = WebRtcSpl_NormU32((WebRtc_UWord32)far_spectrum[i]);
+ if (zerosCh + zerosFar > 31)
+ {
+ // Multiplication is safe
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
+ far_spectrum[i]);
+ shiftChFar = 0;
+ } else
+ {
+ // We need to shift down before multiplication
+ shiftChFar = 32 - zerosCh - zerosFar;
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], shiftChFar),
+ far_spectrum[i]);
+ }
+ // Determine Q-domain of numerator
+ zerosNum = WebRtcSpl_NormU32(tmpU32no1);
+ if (dfa[i])
+ {
+ zerosDfa = WebRtcSpl_NormU32((WebRtc_UWord32)dfa[i]);
+ } else
+ {
+ zerosDfa = 32;
+ }
+ tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain -
+ RESOLUTION_CHANNEL32 - far_q + shiftChFar;
+ if (zerosNum > tmp16no1 + 1)
+ {
+ xfaQ = tmp16no1;
+ dfaQ = zerosDfa - 2;
+ } else
+ {
+ xfaQ = zerosNum - 2;
+ dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
+ shiftChFar + xfaQ;
+ }
+ // Add in the same Q-domain
+ tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
+ tmpU32no2 = WEBRTC_SPL_SHIFT_W32((WebRtc_UWord32)dfa[i], dfaQ);
+ tmp32no1 = (WebRtc_Word32)tmpU32no2 - (WebRtc_Word32)tmpU32no1;
+ zerosNum = WebRtcSpl_NormW32(tmp32no1);
+ if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q)))
+ {
+ //
+ // Update is needed
+ //
+ // This is what we would like to compute
+ //
+ // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
+ // tmp32norm = (i + 1)
+ // aecm->channelAdapt[i] += (2^mu) * tmp32no1
+ // / (tmp32norm * far_spectrum[i])
+ //
+
+ // Make sure we don't get overflow in multiplication.
+ if (zerosNum + zerosFar > 31)
+ {
+ if (tmp32no1 > 0)
+ {
+ tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmp32no1,
+ far_spectrum[i]);
+ } else
+ {
+ tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
+ far_spectrum[i]);
+ }
+ shiftNum = 0;
+ } else
+ {
+ shiftNum = 32 - (zerosNum + zerosFar);
+ if (tmp32no1 > 0)
+ {
+ tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(tmp32no1, shiftNum),
+ far_spectrum[i]);
+ } else
+ {
+ tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(-tmp32no1, shiftNum),
+ far_spectrum[i]);
+ }
+ }
+ // Normalize with respect to frequency bin
+ tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
+ // Make sure we are in the right Q-domain
+ shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
+ if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
+ {
+ tmp32no2 = WEBRTC_SPL_WORD32_MAX;
+ } else
+ {
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
+ }
+ aecm->channelAdapt32[i] = WEBRTC_SPL_ADD_SAT_W32(aecm->channelAdapt32[i],
+ tmp32no2);
+ if (aecm->channelAdapt32[i] < 0)
+ {
+ // We can never have negative channel gain
+ aecm->channelAdapt32[i] = 0;
+ }
+ aecm->channelAdapt16[i]
+ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], 16);
+ }
+ }
+ }
+ // END: Adaptive channel update
+
+ // Determine if we should store or restore the channel
+ if ((aecm->startupState == 0) & (aecm->currentVADValue))
+ {
+ // During startup we store the channel every block,
+ // and we recalculate echo estimate
+ WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+ } else
+ {
+ if (aecm->farLogEnergy < aecm->farEnergyMSE)
+ {
+ aecm->mseChannelCount = 0;
+ } else
+ {
+ aecm->mseChannelCount++;
+ }
+ // Enough data for validation. Store channel if we can.
+ if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
+ {
+ // We have enough data.
+ // Calculate MSE of "Adapt" and "Stored" versions.
+ // It is actually not MSE, but average absolute error.
+ mseStored = 0;
+ mseAdapt = 0;
+ for (i = 0; i < MIN_MSE_COUNT; i++)
+ {
+ tmp32no1 = ((WebRtc_Word32)aecm->echoStoredLogEnergy[i]
+ - (WebRtc_Word32)aecm->nearLogEnergy[i]);
+ tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+ mseStored += tmp32no2;
+
+ tmp32no1 = ((WebRtc_Word32)aecm->echoAdaptLogEnergy[i]
+ - (WebRtc_Word32)aecm->nearLogEnergy[i]);
+ tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+ mseAdapt += tmp32no2;
+ }
+ if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
+ & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
+ * aecm->mseAdaptOld)))
+ {
+ // The stored channel has a significantly lower MSE than the adaptive one for
+ // two consecutive calculations. Reset the adaptive channel.
+ WebRtcAecm_ResetAdaptiveChannel(aecm);
+ } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
+ < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
+ {
+ // The adaptive channel has a significantly lower MSE than the stored one.
+ // The MSE for the adaptive channel has also been low for two consecutive
+ // calculations. Store the adaptive channel.
+ WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+
+ // Update threshold
+ if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
+ {
+ aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
+ } else
+ {
+ aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt
+ - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8);
+ }
+
+ }
+
+ // Reset counter
+ aecm->mseChannelCount = 0;
+
+ // Store the MSE values.
+ aecm->mseStoredOld = mseStored;
+ aecm->mseAdaptOld = mseAdapt;
+ }
+ }
+ // END: Determine if we should store or reset channel estimate.
+}
+
+// CalcSuppressionGain(...)
+//
+// This function calculates the suppression gain that is used in the Wiener filter.
+//
+//
+// @param aecm [i/n] Handle of the AECM instance.
+// @param supGain [out] (Return value) Suppression gain with which to scale the noise
+// level (Q14).
+//
+//
+static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm)
+{
+ WebRtc_Word32 tmp32no1;
+
+ WebRtc_Word16 supGain = SUPGAIN_DEFAULT;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 dE = 0;
+
+ // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
+ // end energy and echo estimation error.
+ // Adjust for the far end signal level. A low signal level indicates no far end signal,
+ // hence we set the suppression gain to 0
+ if (!aecm->currentVADValue)
+ {
+ supGain = 0;
+ } else
+ {
+ // Adjust for possible double talk. If we have large variations in estimation error we
+ // likely have double talk (or poor channel).
+ tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
+ dE = WEBRTC_SPL_ABS_W16(tmp16no1);
+
+ if (dE < ENERGY_DEV_TOL)
+ {
+ // Likely no double talk. The better estimation, the more we can suppress signal.
+ // Update counters
+ if (dE < SUPGAIN_EPC_DT)
+ {
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffAB, dE);
+ tmp32no1 += (SUPGAIN_EPC_DT >> 1);
+ tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
+ supGain = aecm->supGainErrParamA - tmp16no1;
+ } else
+ {
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffBD,
+ (ENERGY_DEV_TOL - dE));
+ tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
+ tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
+ - SUPGAIN_EPC_DT));
+ supGain = aecm->supGainErrParamD + tmp16no1;
+ }
+ } else
+ {
+ // Likely in double talk. Use default value
+ supGain = aecm->supGainErrParamD;
+ }
+ }
+
+ if (supGain > aecm->supGainOld)
+ {
+ tmp16no1 = supGain;
+ } else
+ {
+ tmp16no1 = aecm->supGainOld;
+ }
+ aecm->supGainOld = supGain;
+ if (tmp16no1 < aecm->supGain)
+ {
+ aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
+ } else
+ {
+ aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
+ }
+
+ // END: Update suppression gain
+
+ return aecm->supGain;
+}
+
+// Transforms a time domain signal into the frequency domain, outputting the
+// complex valued signal, absolute value and sum of absolute values.
+//
+// time_signal [in] Pointer to time domain signal
+// freq_signal_real [out] Pointer to real part of frequency domain array
+// freq_signal_imag [out] Pointer to imaginary part of frequency domain
+// array
+// freq_signal_abs [out] Pointer to absolute value of frequency domain
+// array
+// freq_signal_sum_abs [out] Pointer to the sum of all absolute values in
+// the frequency domain array
+// return value The Q-domain of current frequency values
+//
+static int TimeToFrequencyDomain(const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ WebRtc_UWord16* freq_signal_abs,
+ WebRtc_UWord32* freq_signal_sum_abs)
+{
+ int i = 0;
+ int time_signal_scaling = 0;
+
+ WebRtc_Word32 tmp32no1;
+ WebRtc_Word32 tmp32no2;
+
+ // In fft_buf, +16 for 32-byte alignment.
+ WebRtc_Word16 fft_buf[PART_LEN4 + 16];
+ WebRtc_Word16 *fft = (WebRtc_Word16 *) (((uintptr_t) fft_buf + 31) & ~31);
+
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 tmp16no2;
+#ifdef AECM_WITH_ABS_APPROX
+ WebRtc_Word16 max_value = 0;
+ WebRtc_Word16 min_value = 0;
+ WebRtc_UWord16 alpha = 0;
+ WebRtc_UWord16 beta = 0;
+#endif
+
+#ifdef AECM_DYNAMIC_Q
+ tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
+ time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
+#endif
+
+ WebRtcAecm_WindowAndFFT(fft, time_signal, freq_signal, time_signal_scaling);
+
+ // Extract imaginary and real part, calculate the magnitude for all frequency bins
+ freq_signal[0].imag = 0;
+ freq_signal[PART_LEN].imag = 0;
+ freq_signal[PART_LEN].real = fft[PART_LEN2];
+ freq_signal_abs[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[0].real);
+ freq_signal_abs[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[PART_LEN].real);
+ (*freq_signal_sum_abs) = (WebRtc_UWord32)(freq_signal_abs[0]) +
+ (WebRtc_UWord32)(freq_signal_abs[PART_LEN]);
+
+ for (i = 1; i < PART_LEN; i++)
+ {
+ if (freq_signal[i].real == 0)
+ {
+ freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[i].imag);
+ }
+ else if (freq_signal[i].imag == 0)
+ {
+ freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[i].real);
+ }
+ else
+ {
+ // Approximation for magnitude of complex fft output
+ // magn = sqrt(real^2 + imag^2)
+ // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
+ //
+ // The parameters alpha and beta are stored in Q15
+
+#ifdef AECM_WITH_ABS_APPROX
+ tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+ tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+
+ if(tmp16no1 > tmp16no2)
+ {
+ max_value = tmp16no1;
+ min_value = tmp16no2;
+ } else
+ {
+ max_value = tmp16no2;
+ min_value = tmp16no1;
+ }
+
+ // Magnitude in Q(-6)
+ if ((max_value >> 2) > min_value)
+ {
+ alpha = kAlpha1;
+ beta = kBeta1;
+ } else if ((max_value >> 1) > min_value)
+ {
+ alpha = kAlpha2;
+ beta = kBeta2;
+ } else
+ {
+ alpha = kAlpha3;
+ beta = kBeta3;
+ }
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(max_value,
+ alpha,
+ 15);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(min_value,
+ beta,
+ 15);
+ freq_signal_abs[i] = (WebRtc_UWord16)tmp16no1 +
+ (WebRtc_UWord16)tmp16no2;
+#else
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm __volatile(
+ "smulbb %[tmp32no1], %[real], %[real]\n\t"
+ "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
+ :[tmp32no1]"=r"(tmp32no1),
+ [tmp32no2]"=r"(tmp32no2)
+ :[real]"r"(freq_signal[i].real),
+ [imag]"r"(freq_signal[i].imag)
+ );
+#else
+ tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+ tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
+ tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
+#endif // WEBRTC_ARCH_ARM_V7A
+ tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
+
+ freq_signal_abs[i] = (WebRtc_UWord16)tmp32no1;
+#endif // AECM_WITH_ABS_APPROX
+ }
+ (*freq_signal_sum_abs) += (WebRtc_UWord32)freq_signal_abs[i];
+ }
+
+ return time_signal_scaling;
+}
+
+int WebRtcAecm_ProcessBlock(AecmCore_t * aecm,
+ const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * output)
+{
+ int i;
+
+ WebRtc_UWord32 xfaSum;
+ WebRtc_UWord32 dfaNoisySum;
+ WebRtc_UWord32 dfaCleanSum;
+ WebRtc_UWord32 echoEst32Gained;
+ WebRtc_UWord32 tmpU32;
+
+ WebRtc_Word32 tmp32no1;
+
+ WebRtc_UWord16 xfa[PART_LEN1];
+ WebRtc_UWord16 dfaNoisy[PART_LEN1];
+ WebRtc_UWord16 dfaClean[PART_LEN1];
+ WebRtc_UWord16* ptrDfaClean = dfaClean;
+ const WebRtc_UWord16* far_spectrum_ptr = NULL;
+
+ // 32 byte aligned buffers (with +8 or +16).
+ // TODO (kma): define fft with complex16_t.
+ WebRtc_Word16 fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
+ WebRtc_Word32 echoEst32_buf[PART_LEN1 + 8];
+ WebRtc_Word32 dfw_buf[PART_LEN1 + 8];
+ WebRtc_Word32 efw_buf[PART_LEN1 + 8];
+
+ WebRtc_Word16* fft = (WebRtc_Word16*) (((uintptr_t) fft_buf + 31) & ~ 31);
+ WebRtc_Word32* echoEst32 = (WebRtc_Word32*) (((uintptr_t) echoEst32_buf + 31) & ~ 31);
+ complex16_t* dfw = (complex16_t*) (((uintptr_t) dfw_buf + 31) & ~ 31);
+ complex16_t* efw = (complex16_t*) (((uintptr_t) efw_buf + 31) & ~ 31);
+
+ WebRtc_Word16 hnl[PART_LEN1];
+ WebRtc_Word16 numPosCoef = 0;
+ WebRtc_Word16 nlpGain = ONE_Q14;
+ int delay;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 tmp16no2;
+ WebRtc_Word16 mu;
+ WebRtc_Word16 supGain;
+ WebRtc_Word16 zeros32, zeros16;
+ WebRtc_Word16 zerosDBufNoisy, zerosDBufClean, zerosXBuf;
+ int far_q;
+ WebRtc_Word16 resolutionDiff, qDomainDiff;
+
+ const int kMinPrefBand = 4;
+ const int kMaxPrefBand = 24;
+ WebRtc_Word32 avgHnl32 = 0;
+
+#ifdef ARM_WINM_LOG_
+ DWORD temp;
+ static int flag0 = 0;
+ __int64 freq, start, end, diff__;
+ unsigned int milliseconds;
+#endif
+
+ // Determine startup state. There are three states:
+ // (0) the first CONV_LEN blocks
+ // (1) another CONV_LEN blocks
+ // (2) the rest
+
+ if (aecm->startupState < 2)
+ {
+ aecm->startupState = (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
+ }
+ // END: Determine startup state
+
+ // Buffer near and far end signals
+ memcpy(aecm->xBuf + PART_LEN, farend, sizeof(WebRtc_Word16) * PART_LEN);
+ memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(WebRtc_Word16) * PART_LEN);
+ if (nearendClean != NULL)
+ {
+ memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(WebRtc_Word16) * PART_LEN);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick start
+ QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Transform far end signal from time domain to frequency domain.
+ far_q = TimeToFrequencyDomain(aecm->xBuf,
+ dfw,
+ xfa,
+ &xfaSum);
+
+ // Transform noisy near end signal from time domain to frequency domain.
+ zerosDBufNoisy = TimeToFrequencyDomain(aecm->dBufNoisy,
+ dfw,
+ dfaNoisy,
+ &dfaNoisySum);
+ aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
+ aecm->dfaNoisyQDomain = (WebRtc_Word16)zerosDBufNoisy;
+
+
+ if (nearendClean == NULL)
+ {
+ ptrDfaClean = dfaNoisy;
+ aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
+ aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
+ dfaCleanSum = dfaNoisySum;
+ } else
+ {
+ // Transform clean near end signal from time domain to frequency domain.
+ zerosDBufClean = TimeToFrequencyDomain(aecm->dBufClean,
+ dfw,
+ dfaClean,
+ &dfaCleanSum);
+ aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
+ aecm->dfaCleanQDomain = (WebRtc_Word16)zerosDBufClean;
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Get the delay
+ // Save far-end history and estimate delay
+ UpdateFarHistory(aecm, xfa, far_q);
+ delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator,
+ xfa,
+ dfaNoisy,
+ PART_LEN1,
+ far_q,
+ zerosDBufNoisy);
+ if (delay == -1)
+ {
+ return -1;
+ }
+ else if (delay == -2)
+ {
+ // If the delay is unknown, we assume zero.
+ // NOTE: this will have to be adjusted if we ever add lookahead.
+ delay = 0;
+ }
+
+ if (aecm->fixedDelay >= 0)
+ {
+ // Use fixed delay
+ delay = aecm->fixedDelay;
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+ // Get aligned far end spectrum
+ far_spectrum_ptr = AlignedFarend(aecm, &far_q, delay);
+ zerosXBuf = (WebRtc_Word16) far_q;
+ if (far_spectrum_ptr == NULL)
+ {
+ return -1;
+ }
+
+ // Calculate log(energy) and update energy threshold levels
+ WebRtcAecm_CalcEnergies(aecm,
+ far_spectrum_ptr,
+ zerosXBuf,
+ dfaNoisySum,
+ echoEst32);
+
+ // Calculate stepsize
+ mu = WebRtcAecm_CalcStepSize(aecm);
+
+ // Update counters
+ aecm->totCount++;
+
+ // This is the channel estimation algorithm.
+ // It is base on NLMS but has a variable step length, which was calculated above.
+ WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu, echoEst32);
+ supGain = CalcSuppressionGain(aecm);
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Calculate Wiener filter hnl[]
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Far end signal through channel estimate in Q8
+ // How much can we shift right to preserve resolution
+ tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
+ aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
+
+ zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
+ zeros16 = WebRtcSpl_NormW16(supGain) + 1;
+ if (zeros32 + zeros16 > 16)
+ {
+ // Multiplication is safe
+ // Result in Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
+ (WebRtc_UWord16)supGain);
+ resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+ resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+ } else
+ {
+ tmp16no1 = 17 - zeros32 - zeros16;
+ resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+ resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+ if (zeros32 > tmp16no1)
+ {
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
+ (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W16(supGain,
+ tmp16no1)); // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+ } else
+ {
+ // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
+ (WebRtc_UWord32)WEBRTC_SPL_RSHIFT_W32(aecm->echoFilt[i], tmp16no1),
+ (WebRtc_UWord16)supGain);
+ }
+ }
+
+ zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
+ if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld))
+ & (aecm->nearFilt[i]))
+ {
+ tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16);
+ qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld;
+ } else
+ {
+ tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i],
+ aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld);
+ qDomainDiff = 0;
+ }
+ tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff);
+ tmp32no1 = (WebRtc_Word32)(tmp16no2 - tmp16no1);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 4);
+ tmp16no2 += tmp16no1;
+ zeros16 = WebRtcSpl_NormW16(tmp16no2);
+ if ((tmp16no2) & (-qDomainDiff > zeros16))
+ {
+ aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
+ } else
+ {
+ aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff);
+ }
+
+ // Wiener filter coefficients, resulting hnl in Q14
+ if (echoEst32Gained == 0)
+ {
+ hnl[i] = ONE_Q14;
+ } else if (aecm->nearFilt[i] == 0)
+ {
+ hnl[i] = 0;
+ } else
+ {
+ // Multiply the suppression gain
+ // Rounding
+ echoEst32Gained += (WebRtc_UWord32)(aecm->nearFilt[i] >> 1);
+ tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained, (WebRtc_UWord16)aecm->nearFilt[i]);
+
+ // Current resolution is
+ // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN - max(0, 17 - zeros16 - zeros32))
+ // Make sure we are in Q14
+ tmp32no1 = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
+ if (tmp32no1 > ONE_Q14)
+ {
+ hnl[i] = 0;
+ } else if (tmp32no1 < 0)
+ {
+ hnl[i] = ONE_Q14;
+ } else
+ {
+ // 1-echoEst/dfa
+ hnl[i] = ONE_Q14 - (WebRtc_Word16)tmp32no1;
+ if (hnl[i] < 0)
+ {
+ hnl[i] = 0;
+ }
+ }
+ }
+ if (hnl[i])
+ {
+ numPosCoef++;
+ }
+ }
+ // Only in wideband. Prevent the gain in upper band from being larger than
+ // in lower band.
+ if (aecm->mult == 2)
+ {
+ // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
+ // speech distortion in double-talk.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], hnl[i], 14);
+ }
+
+ for (i = kMinPrefBand; i <= kMaxPrefBand; i++)
+ {
+ avgHnl32 += (WebRtc_Word32)hnl[i];
+ }
+ assert(kMaxPrefBand - kMinPrefBand + 1 > 0);
+ avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
+
+ for (i = kMaxPrefBand; i < PART_LEN1; i++)
+ {
+ if (hnl[i] > (WebRtc_Word16)avgHnl32)
+ {
+ hnl[i] = (WebRtc_Word16)avgHnl32;
+ }
+ }
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Calculate NLP gain, result is in Q14
+ if (aecm->nlpFlag)
+ {
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Truncate values close to zero and one.
+ if (hnl[i] > NLP_COMP_HIGH)
+ {
+ hnl[i] = ONE_Q14;
+ } else if (hnl[i] < NLP_COMP_LOW)
+ {
+ hnl[i] = 0;
+ }
+
+ // Remove outliers
+ if (numPosCoef < 3)
+ {
+ nlpGain = 0;
+ } else
+ {
+ nlpGain = ONE_Q14;
+ }
+
+ // NLP
+ if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
+ {
+ hnl[i] = ONE_Q14;
+ } else
+ {
+ hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
+ }
+
+ // multiply with Wiener coefficients
+ efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
+ hnl[i], 14));
+ efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
+ hnl[i], 14));
+ }
+ }
+ else
+ {
+ // multiply with Wiener coefficients
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
+ hnl[i], 14));
+ efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
+ hnl[i], 14));
+ }
+ }
+
+ if (aecm->cngMode == AecmTrue)
+ {
+ ComfortNoise(aecm, ptrDfaClean, efw, hnl);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ WebRtcAecm_InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
+
+ return 0;
+}
+
+
+// Generate comfort noise and add to output signal.
+//
+// \param[in] aecm Handle of the AECM instance.
+// \param[in] dfa Absolute value of the nearend signal (Q[aecm->dfaQDomain]).
+// \param[in,out] outReal Real part of the output signal (Q[aecm->dfaQDomain]).
+// \param[in,out] outImag Imaginary part of the output signal (Q[aecm->dfaQDomain]).
+// \param[in] lambda Suppression gain with which to scale the noise level (Q14).
+//
+static void ComfortNoise(AecmCore_t* aecm,
+ const WebRtc_UWord16* dfa,
+ complex16_t* out,
+ const WebRtc_Word16* lambda)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word32 tmp32;
+
+ WebRtc_Word16 randW16[PART_LEN];
+ WebRtc_Word16 uReal[PART_LEN1];
+ WebRtc_Word16 uImag[PART_LEN1];
+ WebRtc_Word32 outLShift32;
+ WebRtc_Word16 noiseRShift16[PART_LEN1];
+
+ WebRtc_Word16 shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
+ WebRtc_Word16 minTrackShift;
+
+ assert(shiftFromNearToNoise >= 0);
+ assert(shiftFromNearToNoise < 16);
+
+ if (aecm->noiseEstCtr < 100)
+ {
+ // Track the minimum more quickly initially.
+ aecm->noiseEstCtr++;
+ minTrackShift = 6;
+ } else
+ {
+ minTrackShift = 9;
+ }
+
+ // Estimate noise power.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+
+ // Shift to the noise domain.
+ tmp32 = (WebRtc_Word32)dfa[i];
+ outLShift32 = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+
+ if (outLShift32 < aecm->noiseEst[i])
+ {
+ // Reset "too low" counter
+ aecm->noiseEstTooLowCtr[i] = 0;
+ // Track the minimum.
+ if (aecm->noiseEst[i] < (1 << minTrackShift))
+ {
+ // For small values, decrease noiseEst[i] every
+ // |kNoiseEstIncCount| block. The regular approach below can not
+ // go further down due to truncation.
+ aecm->noiseEstTooHighCtr[i]++;
+ if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount)
+ {
+ aecm->noiseEst[i]--;
+ aecm->noiseEstTooHighCtr[i] = 0; // Reset the counter
+ }
+ }
+ else
+ {
+ aecm->noiseEst[i] -= ((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
+ }
+ } else
+ {
+ // Reset "too high" counter
+ aecm->noiseEstTooHighCtr[i] = 0;
+ // Ramp slowly upwards until we hit the minimum again.
+ if ((aecm->noiseEst[i] >> 19) > 0)
+ {
+ // Avoid overflow.
+ // Multiplication with 2049 will cause wrap around. Scale
+ // down first and then multiply
+ aecm->noiseEst[i] >>= 11;
+ aecm->noiseEst[i] *= 2049;
+ }
+ else if ((aecm->noiseEst[i] >> 11) > 0)
+ {
+ // Large enough for relative increase
+ aecm->noiseEst[i] *= 2049;
+ aecm->noiseEst[i] >>= 11;
+ }
+ else
+ {
+ // Make incremental increases based on size every
+ // |kNoiseEstIncCount| block
+ aecm->noiseEstTooLowCtr[i]++;
+ if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount)
+ {
+ aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
+ aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
+ }
+ }
+ }
+ }
+
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(aecm->noiseEst[i], shiftFromNearToNoise);
+ if (tmp32 > 32767)
+ {
+ tmp32 = 32767;
+ aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+ }
+ noiseRShift16[i] = (WebRtc_Word16)tmp32;
+
+ tmp16 = ONE_Q14 - lambda[i];
+ noiseRShift16[i]
+ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, noiseRShift16[i], 14);
+ }
+
+ // Generate a uniform random array on [0 2^15-1].
+ WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
+
+ // Generate noise according to estimated energy.
+ uReal[0] = 0; // Reject LF noise.
+ uImag[0] = 0;
+ for (i = 1; i < PART_LEN1; i++)
+ {
+ // Get a random index for the cos and sin tables over [0 359].
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
+
+ // Tables are in Q13.
+ uReal[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i],
+ kCosTable[tmp16], 13);
+ uImag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i],
+ kSinTable[tmp16], 13);
+ }
+ uImag[PART_LEN] = 0;
+
+#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ }
+#else
+ for (i = 0; i < PART_LEN1 -1; )
+ {
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ i++;
+
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ i++;
+ }
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+#endif
+}
+
+void WebRtcAecm_BufferFarFrame(AecmCore_t* const aecm,
+ const WebRtc_Word16* const farend,
+ const int farLen)
+{
+ int writeLen = farLen, writePos = 0;
+
+ // Check if the write position must be wrapped
+ while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
+ {
+ // Write to remaining buffer space before wrapping
+ writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
+ memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+ sizeof(WebRtc_Word16) * writeLen);
+ aecm->farBufWritePos = 0;
+ writePos = writeLen;
+ writeLen = farLen - writeLen;
+ }
+
+ memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+ sizeof(WebRtc_Word16) * writeLen);
+ aecm->farBufWritePos += writeLen;
+}
+
+void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
+ const int farLen, const int knownDelay)
+{
+ int readLen = farLen;
+ int readPos = 0;
+ int delayChange = knownDelay - aecm->lastKnownDelay;
+
+ aecm->farBufReadPos -= delayChange;
+
+ // Check if delay forces a read position wrap
+ while (aecm->farBufReadPos < 0)
+ {
+ aecm->farBufReadPos += FAR_BUF_LEN;
+ }
+ while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
+ {
+ aecm->farBufReadPos -= FAR_BUF_LEN;
+ }
+
+ aecm->lastKnownDelay = knownDelay;
+
+ // Check if read position must be wrapped
+ while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
+ {
+
+ // Read from remaining buffer space before wrapping
+ readLen = FAR_BUF_LEN - aecm->farBufReadPos;
+ memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+ sizeof(WebRtc_Word16) * readLen);
+ aecm->farBufReadPos = 0;
+ readPos = readLen;
+ readLen = farLen - readLen;
+ }
+ memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+ sizeof(WebRtc_Word16) * readLen);
+ aecm->farBufReadPos += readLen;
+}
+
+
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core.h b/trunk/src/modules/audio_processing/aecm/aecm_core.h
new file mode 100644
index 0000000..0ec62ec
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/aecm_core.h
@@ -0,0 +1,375 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs echo control (suppression) with fft routines in fixed-point
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
+
+#define AECM_DYNAMIC_Q // turn on/off dynamic Q-domain
+//#define AECM_WITH_ABS_APPROX
+//#define AECM_SHORT // for 32 sample partition length (otherwise 64)
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+// Algorithm parameters
+
+#define FRAME_LEN 80 // Total frame length, 10 ms
+#ifdef AECM_SHORT
+
+#define PART_LEN 32 // Length of partition
+#define PART_LEN_SHIFT 6 // Length of (PART_LEN * 2) in base 2
+
+#else
+
+#define PART_LEN 64 // Length of partition
+#define PART_LEN_SHIFT 7 // Length of (PART_LEN * 2) in base 2
+
+#endif
+
+#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
+#define PART_LEN2 (PART_LEN << 1) // Length of partition * 2
+#define PART_LEN4 (PART_LEN << 2) // Length of partition * 4
+#define FAR_BUF_LEN PART_LEN4 // Length of buffers
+#define MAX_DELAY 100
+
+// Counter parameters
+#ifdef AECM_SHORT
+
+#define CONV_LEN 1024 // Convergence length used at startup
+#else
+
+#define CONV_LEN 512 // Convergence length used at startup
+#endif
+
+#define CONV_LEN2 (CONV_LEN << 1) // Convergence length * 2 used at startup
+// Energy parameters
+#define MAX_BUF_LEN 64 // History length of energy signals
+
+#define FAR_ENERGY_MIN 1025 // Lowest Far energy level: At least 2 in energy
+#define FAR_ENERGY_DIFF 929 // Allowed difference between max and min
+
+#define ENERGY_DEV_OFFSET 0 // The energy error offset in Q8
+#define ENERGY_DEV_TOL 400 // The energy estimation tolerance in Q8
+#define FAR_ENERGY_VAD_REGION 230 // Far VAD tolerance region
+// Stepsize parameters
+#define MU_MIN 10 // Min stepsize 2^-MU_MIN (far end energy dependent)
+#define MU_MAX 1 // Max stepsize 2^-MU_MAX (far end energy dependent)
+#define MU_DIFF 9 // MU_MIN - MU_MAX
+// Channel parameters
+#define MIN_MSE_COUNT 20 // Min number of consecutive blocks with enough far end
+ // energy to compare channel estimates
+#define MIN_MSE_DIFF 29 // The ratio between adapted and stored channel to
+ // accept a new storage (0.8 in Q-MSE_RESOLUTION)
+#define MSE_RESOLUTION 5 // MSE parameter resolution
+#define RESOLUTION_CHANNEL16 12 // W16 Channel in Q-RESOLUTION_CHANNEL16
+#define RESOLUTION_CHANNEL32 28 // W32 Channel in Q-RESOLUTION_CHANNEL
+#define CHANNEL_VAD 16 // Minimum energy in frequency band to update channel
+// Suppression gain parameters: SUPGAIN_ parameters in Q-(RESOLUTION_SUPGAIN)
+#define RESOLUTION_SUPGAIN 8 // Channel in Q-(RESOLUTION_SUPGAIN)
+#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN) // Default suppression gain
+#define SUPGAIN_ERROR_PARAM_A 3072 // Estimation error parameter (Maximum gain) (8 in Q8)
+#define SUPGAIN_ERROR_PARAM_B 1536 // Estimation error parameter (Gain before going down)
+#define SUPGAIN_ERROR_PARAM_D SUPGAIN_DEFAULT // Estimation error parameter
+ // (Should be the same as Default) (1 in Q8)
+#define SUPGAIN_EPC_DT 200 // = SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL
+// Defines for "check delay estimation"
+#define CORR_WIDTH 31 // Number of samples to correlate over.
+#define CORR_MAX 16 // Maximum correlation offset
+#define CORR_MAX_BUF 63
+#define CORR_DEV 4
+#define CORR_MAX_LEVEL 20
+#define CORR_MAX_LOW 4
+#define CORR_BUF_LEN (CORR_MAX << 1) + 1
+// Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN
+
+#define ONE_Q14 (1 << 14)
+
+// NLP defines
+#define NLP_COMP_LOW 3277 // 0.2 in Q14
+#define NLP_COMP_HIGH ONE_Q14 // 1 in Q14
+
+extern const WebRtc_Word16 WebRtcAecm_kSqrtHanning[];
+
+typedef struct {
+ WebRtc_Word16 real;
+ WebRtc_Word16 imag;
+} complex16_t;
+
+typedef struct
+{
+ int farBufWritePos;
+ int farBufReadPos;
+ int knownDelay;
+ int lastKnownDelay;
+ int firstVAD; // Parameter to control poorly initialized channels
+
+ void *farFrameBuf;
+ void *nearNoisyFrameBuf;
+ void *nearCleanFrameBuf;
+ void *outFrameBuf;
+
+ WebRtc_Word16 farBuf[FAR_BUF_LEN];
+
+ WebRtc_Word16 mult;
+ WebRtc_UWord32 seed;
+
+ // Delay estimation variables
+ void* delay_estimator;
+ WebRtc_UWord16 currentDelay;
+ // Far end history variables
+ // TODO(bjornv): Replace |far_history| with ring_buffer.
+ uint16_t far_history[PART_LEN1 * MAX_DELAY];
+ int far_history_pos;
+ int far_q_domains[MAX_DELAY];
+
+ WebRtc_Word16 nlpFlag;
+ WebRtc_Word16 fixedDelay;
+
+ WebRtc_UWord32 totCount;
+
+ WebRtc_Word16 dfaCleanQDomain;
+ WebRtc_Word16 dfaCleanQDomainOld;
+ WebRtc_Word16 dfaNoisyQDomain;
+ WebRtc_Word16 dfaNoisyQDomainOld;
+
+ WebRtc_Word16 nearLogEnergy[MAX_BUF_LEN];
+ WebRtc_Word16 farLogEnergy;
+ WebRtc_Word16 echoAdaptLogEnergy[MAX_BUF_LEN];
+ WebRtc_Word16 echoStoredLogEnergy[MAX_BUF_LEN];
+
+ // The extra 16 or 32 bytes in the following buffers are for alignment based Neon code.
+ // It's designed this way since the current GCC compiler can't align a buffer in 16 or 32
+ // byte boundaries properly.
+ WebRtc_Word16 channelStored_buf[PART_LEN1 + 8];
+ WebRtc_Word16 channelAdapt16_buf[PART_LEN1 + 8];
+ WebRtc_Word32 channelAdapt32_buf[PART_LEN1 + 8];
+ WebRtc_Word16 xBuf_buf[PART_LEN2 + 16]; // farend
+ WebRtc_Word16 dBufClean_buf[PART_LEN2 + 16]; // nearend
+ WebRtc_Word16 dBufNoisy_buf[PART_LEN2 + 16]; // nearend
+ WebRtc_Word16 outBuf_buf[PART_LEN + 8];
+
+ // Pointers to the above buffers
+ WebRtc_Word16 *channelStored;
+ WebRtc_Word16 *channelAdapt16;
+ WebRtc_Word32 *channelAdapt32;
+ WebRtc_Word16 *xBuf;
+ WebRtc_Word16 *dBufClean;
+ WebRtc_Word16 *dBufNoisy;
+ WebRtc_Word16 *outBuf;
+
+ WebRtc_Word32 echoFilt[PART_LEN1];
+ WebRtc_Word16 nearFilt[PART_LEN1];
+ WebRtc_Word32 noiseEst[PART_LEN1];
+ int noiseEstTooLowCtr[PART_LEN1];
+ int noiseEstTooHighCtr[PART_LEN1];
+ WebRtc_Word16 noiseEstCtr;
+ WebRtc_Word16 cngMode;
+
+ WebRtc_Word32 mseAdaptOld;
+ WebRtc_Word32 mseStoredOld;
+ WebRtc_Word32 mseThreshold;
+
+ WebRtc_Word16 farEnergyMin;
+ WebRtc_Word16 farEnergyMax;
+ WebRtc_Word16 farEnergyMaxMin;
+ WebRtc_Word16 farEnergyVAD;
+ WebRtc_Word16 farEnergyMSE;
+ int currentVADValue;
+ WebRtc_Word16 vadUpdateCount;
+
+ WebRtc_Word16 startupState;
+ WebRtc_Word16 mseChannelCount;
+ WebRtc_Word16 supGain;
+ WebRtc_Word16 supGainOld;
+
+ WebRtc_Word16 supGainErrParamA;
+ WebRtc_Word16 supGainErrParamD;
+ WebRtc_Word16 supGainErrParamDiffAB;
+ WebRtc_Word16 supGainErrParamDiffBD;
+
+#ifdef AEC_DEBUG
+ FILE *farFile;
+ FILE *nearFile;
+ FILE *outFile;
+#endif
+} AecmCore_t;
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_CreateCore(...)
+//
+// Allocates the memory needed by the AECM. The memory needs to be
+// initialized separately using the WebRtcAecm_InitCore() function.
+//
+// Input:
+// - aecm : Instance that should be created
+//
+// Output:
+// - aecm : Created instance
+//
+// Return value : 0 - Ok
+// -1 - Error
+//
+int WebRtcAecm_CreateCore(AecmCore_t **aecm);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_InitCore(...)
+//
+// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
+// Input:
+// - aecm : Pointer to the AECM instance
+// - samplingFreq : Sampling Frequency
+//
+// Output:
+// - aecm : Initialized instance
+//
+// Return value : 0 - Ok
+// -1 - Error
+//
+int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_FreeCore(...)
+//
+// This function releases the memory allocated by WebRtcAecm_CreateCore()
+// Input:
+// - aecm : Pointer to the AECM instance
+//
+// Return value : 0 - Ok
+// -1 - Error
+// 11001-11016: Error
+//
+int WebRtcAecm_FreeCore(AecmCore_t *aecm);
+
+int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_InitEchoPathCore(...)
+//
+// This function resets the echo channel adaptation with the specified channel.
+// Input:
+// - aecm : Pointer to the AECM instance
+// - echo_path : Pointer to the data that should initialize the echo path
+//
+// Output:
+// - aecm : Initialized instance
+//
+void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_ProcessFrame(...)
+//
+// This function processes frames and sends blocks to WebRtcAecm_ProcessBlock(...)
+//
+// Inputs:
+// - aecm : Pointer to the AECM instance
+// - farend : In buffer containing one frame of echo signal
+// - nearendNoisy : In buffer containing one frame of nearend+echo signal without NS
+// - nearendClean : In buffer containing one frame of nearend+echo signal with NS
+//
+// Output:
+// - out : Out buffer, one frame of nearend signal :
+//
+//
+int WebRtcAecm_ProcessFrame(AecmCore_t * aecm, const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * out);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_ProcessBlock(...)
+//
+// This function is called for every block within one frame
+// This function is called by WebRtcAecm_ProcessFrame(...)
+//
+// Inputs:
+// - aecm : Pointer to the AECM instance
+// - farend : In buffer containing one block of echo signal
+// - nearendNoisy : In buffer containing one frame of nearend+echo signal without NS
+// - nearendClean : In buffer containing one frame of nearend+echo signal with NS
+//
+// Output:
+// - out : Out buffer, one block of nearend signal :
+//
+//
+int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * noisyClean,
+ WebRtc_Word16 * out);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_BufferFarFrame()
+//
+// Inserts a frame of data into farend buffer.
+//
+// Inputs:
+// - aecm : Pointer to the AECM instance
+// - farend : In buffer containing one frame of farend signal
+// - farLen : Length of frame
+//
+void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
+ const int farLen);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_FetchFarFrame()
+//
+// Read the farend buffer to account for known delay
+//
+// Inputs:
+// - aecm : Pointer to the AECM instance
+// - farend : In buffer containing one frame of farend signal
+// - farLen : Length of frame
+// - knownDelay : known delay
+//
+void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
+ const int farLen, const int knownDelay);
+
+///////////////////////////////////////////////////////////////////////////////
+// Some function pointers, for internal functions shared by ARM NEON and
+// generic C code.
+//
+typedef void (*CalcLinearEnergies)(
+ AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echoEst,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored);
+extern CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+
+typedef void (*StoreAdaptiveChannel)(
+ AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est);
+extern StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+
+typedef void (*ResetAdaptiveChannel)(AecmCore_t* aecm);
+extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+
+typedef void (*WindowAndFFT)(
+ WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling);
+extern WindowAndFFT WebRtcAecm_WindowAndFFT;
+
+typedef void (*InverseFFTAndWindow)(
+ AecmCore_t* aecm,
+ WebRtc_Word16* fft, complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean);
+extern InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
+
+// Initialization of the above function pointers for ARM Neon.
+void WebRtcAecm_InitNeon(void);
+
+
+#endif
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c b/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c
new file mode 100644
index 0000000..ab448b4
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c
@@ -0,0 +1,303 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aecm_core.h"
+
+#include <arm_neon.h>
+#include <assert.h>
+
+
+// Square root of Hanning window in Q14.
+static const WebRtc_Word16 kSqrtHanningReversed[] __attribute__((aligned(8))) = {
+ 16384, 16373, 16354, 16325,
+ 16286, 16237, 16179, 16111,
+ 16034, 15947, 15851, 15746,
+ 15631, 15506, 15373, 15231,
+ 15079, 14918, 14749, 14571,
+ 14384, 14189, 13985, 13773,
+ 13553, 13325, 13089, 12845,
+ 12594, 12335, 12068, 11795,
+ 11514, 11227, 10933, 10633,
+ 10326, 10013, 9695, 9370,
+ 9040, 8705, 8364, 8019,
+ 7668, 7313, 6954, 6591,
+ 6224, 5853, 5478, 5101,
+ 4720, 4337, 3951, 3562,
+ 3172, 2780, 2386, 1990,
+ 1594, 1196, 798, 399
+};
+
+static void WindowAndFFTNeon(WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling) {
+ int i, j;
+
+ int16x4_t tmp16x4_scaling = vdup_n_s16(time_signal_scaling);
+ __asm__("vmov.i16 d21, #0" ::: "d21");
+
+ for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int32x4_t tmp32x4_0;
+
+ /* Window near end */
+ // fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((time_signal[i]
+ // << time_signal_scaling), WebRtcAecm_kSqrtHanning[i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i]));
+ tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
+
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
+ tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
+
+ __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
+ __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[j]) : "q10");
+
+ // fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ // (time_signal[PART_LEN + i] << time_signal_scaling),
+ // WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i + PART_LEN]));
+ tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
+
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
+ tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
+
+ __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
+ __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[PART_LEN2 + j]) : "q10");
+ }
+
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the first PART_LEN2 samples, and switch the sign of the imaginary part.
+ for (i = 0, j = 0; j < PART_LEN2; i += 8, j += 16) {
+ __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
+ __asm__("vneg.s16 d22, d22" : : : "q10");
+ __asm__("vneg.s16 d23, d23" : : : "q11");
+ __asm__("vst2.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&freq_signal[i].real): "q10", "q11");
+ }
+}
+
+static void InverseFFTAndWindowNeon(AecmCore_t* aecm,
+ WebRtc_Word16* fft,
+ complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean) {
+ int i, j, outCFFT;
+ WebRtc_Word32 tmp32no1;
+
+ // Synthesis
+ for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
+ // We overwrite two more elements in fft[], but it's ok.
+ __asm__("vld2.16 {d20, d21}, [%0, :128]" : : "r"(&(efw[i].real)) : "q10");
+ __asm__("vmov q11, q10" : : : "q10", "q11");
+
+ __asm__("vneg.s16 d23, d23" : : : "q11");
+ __asm__("vst2.16 {d22, d23}, [%0, :128]" : : "r"(&fft[j]): "q11");
+
+ __asm__("vrev64.16 q10, q10" : : : "q10");
+ __asm__("vst2.16 {d20, d21}, [%0]" : : "r"(&fft[PART_LEN4 - j - 6]): "q10");
+ }
+
+ fft[PART_LEN2] = efw[PART_LEN].real;
+ fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
+
+ // Inverse FFT, result should be scaled with outCFFT.
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the real values and scale with outCFFT.
+ for (i = 0, j = 0; i < PART_LEN2; i += 8, j += 16) {
+ __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
+ __asm__("vst1.16 {d20, d21}, [%0, :128]" : : "r"(&fft[i]): "q10");
+ }
+
+ int32x4_t tmp32x4_2;
+ __asm__("vdup.32 %q0, %1" : "=w"(tmp32x4_2) : "r"((WebRtc_Word32)
+ (outCFFT - aecm->dfaCleanQDomain)));
+ for (i = 0; i < PART_LEN; i += 4) {
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int32x4_t tmp32x4_0;
+ int32x4_t tmp32x4_1;
+
+ // fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // fft[i], WebRtcAecm_kSqrtHanning[i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[i]));
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
+ __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
+ __asm__("vrshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
+
+ // tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
+ // outCFFT - aecm->dfaCleanQDomain);
+ __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
+
+ // fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ // tmp32no1 + outBuf[i], WEBRTC_SPL_WORD16_MIN);
+ // output[i] = fft[i];
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&aecm->outBuf[i]));
+ __asm__("vmovl.s16 %q0, %P1" : "=w"(tmp32x4_1) : "w"(tmp16x4_0));
+ __asm__("vadd.i32 %q0, %q1" : : "w"(tmp32x4_0), "w"(tmp32x4_1));
+ __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&fft[i]));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&output[i]));
+
+ // tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
+ // fft[PART_LEN + i], WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[PART_LEN + i]));
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
+ __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
+ __asm__("vshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
+
+ // tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
+ __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
+ // outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
+ // WEBRTC_SPL_WORD16_MAX, tmp32no1, WEBRTC_SPL_WORD16_MIN);
+ __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&aecm->outBuf[i]));
+ }
+
+ // Copy the current block to the old position (outBuf is shifted elsewhere).
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->xBuf[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&aecm->xBuf[i]): "q10");
+ }
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufNoisy[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufNoisy[i]): "q10");
+ }
+ if (nearendClean != NULL) {
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufClean[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufClean[i]): "q10");
+ }
+ }
+}
+
+static void CalcLinearEnergiesNeon(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored) {
+ int i;
+
+ register WebRtc_UWord32 far_energy_r;
+ register WebRtc_UWord32 echo_energy_stored_r;
+ register WebRtc_UWord32 echo_energy_adapt_r;
+ uint32x4_t tmp32x4_0;
+
+ __asm__("vmov.i32 q14, #0" : : : "q14"); // far_energy
+ __asm__("vmov.i32 q8, #0" : : : "q8"); // echo_energy_stored
+ __asm__("vmov.i32 q9, #0" : : : "q9"); // echo_energy_adapt
+
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // far_energy += (WebRtc_UWord32)(far_spectrum[i]);
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
+ __asm__("vaddw.u16 q14, q14, d26" : : : "q14", "q13");
+ __asm__("vaddw.u16 q14, q14, d27" : : : "q14", "q13");
+
+ // Get estimated echo energies for adaptive channel and stored channel.
+ // echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vst1.32 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&echo_est[i]):
+ "q10", "q11");
+
+ // echo_energy_stored += (WebRtc_UWord32)echoEst[i];
+ __asm__("vadd.u32 q8, q10" : : : "q10", "q8");
+ __asm__("vadd.u32 q8, q11" : : : "q11", "q8");
+
+ // echo_energy_adapt += WEBRTC_SPL_UMUL_16_16(
+ // aecm->channelAdapt16[i], far_spectrum[i]);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vadd.u32 q9, q10" : : : "q9", "q15");
+ __asm__("vadd.u32 q9, q11" : : : "q9", "q11");
+ }
+
+ __asm__("vadd.u32 d28, d29" : : : "q14");
+ __asm__("vpadd.u32 d28, d28" : : : "q14");
+ __asm__("vmov.32 %0, d28[0]" : "=r"(far_energy_r): : "q14");
+
+ __asm__("vadd.u32 d18, d19" : : : "q9");
+ __asm__("vpadd.u32 d18, d18" : : : "q9");
+ __asm__("vmov.32 %0, d18[0]" : "=r"(echo_energy_adapt_r): : "q9");
+
+ __asm__("vadd.u32 d16, d17" : : : "q8");
+ __asm__("vpadd.u32 d16, d16" : : : "q8");
+ __asm__("vmov.32 %0, d16[0]" : "=r"(echo_energy_stored_r): : "q8");
+
+ // Get estimated echo energies for adaptive channel and stored channel.
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ *echo_energy_stored = echo_energy_stored_r + (WebRtc_UWord32)echo_est[i];
+ *far_energy = far_energy_r + (WebRtc_UWord32)(far_spectrum[i]);
+ *echo_energy_adapt = echo_energy_adapt_r + WEBRTC_SPL_UMUL_16_16(
+ aecm->channelAdapt16[i], far_spectrum[i]);
+}
+
+static void StoreAdaptiveChannelNeon(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est) {
+ int i;
+
+ // During startup we store the channel every block.
+ // Recalculate echo estimate.
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // aecm->channelStored[i] = acem->channelAdapt16[i];
+ // echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vst1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&echo_est[i]) : "q10", "q11");
+ }
+ aecm->channelStored[i] = aecm->channelAdapt16[i];
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+}
+
+static void ResetAdaptiveChannelNeon(AecmCore_t* aecm) {
+ int i;
+
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // aecm->channelAdapt16[i] = aecm->channelStored[i];
+ // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)
+ // aecm->channelStored[i], 16);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : :
+ "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vst1.16 {d24, d25}, [%0, :128]" : :
+ "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vshll.s16 q10, d24, #16" : : : "q12", "q13", "q10");
+ __asm__("vshll.s16 q11, d25, #16" : : : "q12", "q13", "q11");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->channelAdapt32[i]): "q10", "q11");
+ }
+ aecm->channelAdapt16[i] = aecm->channelStored[i];
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i], 16);
+}
+
+void WebRtcAecm_InitNeon(void) {
+ WebRtcAecm_WindowAndFFT = WindowAndFFTNeon;
+ WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowNeon;
+ WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesNeon;
+ WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelNeon;
+ WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelNeon;
+}
diff --git a/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c b/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c
new file mode 100644
index 0000000..566ae00
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c
@@ -0,0 +1,792 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+//#include <string.h>
+
+#include "echo_control_mobile.h"
+#include "aecm_core.h"
+#include "ring_buffer.h"
+#ifdef AEC_DEBUG
+#include <stdio.h>
+#endif
+#ifdef MAC_IPHONE_PRINT
+#include <time.h>
+#include <stdio.h>
+#elif defined ARM_WINM_LOG
+#include "windows.h"
+extern HANDLE logFile;
+#endif
+
+#define BUF_SIZE_FRAMES 50 // buffer size (frames)
+// Maximum length of resampled signal. Must be an integer multiple of frames
+// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
+// The factor of 2 handles wb, and the + 1 is as a safety margin
+#define MAX_RESAMP_LEN (5 * FRAME_LEN)
+
+static const size_t kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
+static const int kSampMsNb = 8; // samples per ms in nb
+// Target suppression levels for nlp modes
+// log{0.001, 0.00001, 0.00000001}
+static const int kInitCheck = 42;
+
+typedef struct
+{
+ int sampFreq;
+ int scSampFreq;
+ short bufSizeStart;
+ int knownDelay;
+
+ // Stores the last frame added to the farend buffer
+ short farendOld[2][FRAME_LEN];
+ short initFlag; // indicates if AEC has been initialized
+
+ // Variables used for averaging far end buffer size
+ short counter;
+ short sum;
+ short firstVal;
+ short checkBufSizeCtr;
+
+ // Variables used for delay shifts
+ short msInSndCardBuf;
+ short filtDelay;
+ int timeForDelayChange;
+ int ECstartup;
+ int checkBuffSize;
+ int delayChange;
+ short lastDelayDiff;
+
+ WebRtc_Word16 echoMode;
+
+#ifdef AEC_DEBUG
+ FILE *bufFile;
+ FILE *delayFile;
+ FILE *preCompFile;
+ FILE *postCompFile;
+#endif // AEC_DEBUG
+ // Structures
+ void *farendBuf;
+
+ int lastError;
+
+ AecmCore_t *aecmCore;
+} aecmob_t;
+
+// Estimates delay to set the position of the farend buffer read pointer
+// (controlled by knownDelay)
+static int WebRtcAecm_EstBufDelay(aecmob_t *aecmInst, short msInSndCardBuf);
+
+// Stuffs the farend buffer if the estimated delay is too large
+static int WebRtcAecm_DelayComp(aecmob_t *aecmInst);
+
+WebRtc_Word32 WebRtcAecm_Create(void **aecmInst)
+{
+ aecmob_t *aecm;
+ if (aecmInst == NULL)
+ {
+ return -1;
+ }
+
+ aecm = malloc(sizeof(aecmob_t));
+ *aecmInst = aecm;
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (WebRtcAecm_CreateCore(&aecm->aecmCore) == -1)
+ {
+ WebRtcAecm_Free(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->farendBuf, kBufSizeSamp,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_Free(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ aecm->initFlag = 0;
+ aecm->lastError = 0;
+
+#ifdef AEC_DEBUG
+ aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
+ aecm->aecmCore->nearFile = fopen("aecNear.pcm","wb");
+ aecm->aecmCore->outFile = fopen("aecOut.pcm","wb");
+ //aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
+
+ aecm->bufFile = fopen("aecBuf.dat", "wb");
+ aecm->delayFile = fopen("aecDelay.dat", "wb");
+ aecm->preCompFile = fopen("preComp.pcm", "wb");
+ aecm->postCompFile = fopen("postComp.pcm", "wb");
+#endif // AEC_DEBUG
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_Free(void *aecmInst)
+{
+ aecmob_t *aecm = aecmInst;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+#ifdef AEC_DEBUG
+ fclose(aecm->aecmCore->farFile);
+ fclose(aecm->aecmCore->nearFile);
+ fclose(aecm->aecmCore->outFile);
+ //fclose(aecm->aecmCore->outLpFile);
+
+ fclose(aecm->bufFile);
+ fclose(aecm->delayFile);
+ fclose(aecm->preCompFile);
+ fclose(aecm->postCompFile);
+#endif // AEC_DEBUG
+ WebRtcAecm_FreeCore(aecm->aecmCore);
+ WebRtc_FreeBuffer(aecm->farendBuf);
+ free(aecm);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq)
+{
+ aecmob_t *aecm = aecmInst;
+ AecmConfig aecConfig;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (sampFreq != 8000 && sampFreq != 16000)
+ {
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecm->sampFreq = sampFreq;
+
+ // Initialize AECM core
+ if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
+ {
+ aecm->lastError = AECM_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ // Initialize farend buffer
+ if (WebRtc_InitBuffer(aecm->farendBuf) == -1)
+ {
+ aecm->lastError = AECM_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ aecm->initFlag = kInitCheck; // indicates that initialization has been done
+
+ aecm->delayChange = 1;
+
+ aecm->sum = 0;
+ aecm->counter = 0;
+ aecm->checkBuffSize = 1;
+ aecm->firstVal = 0;
+
+ aecm->ECstartup = 1;
+ aecm->bufSizeStart = 0;
+ aecm->checkBufSizeCtr = 0;
+ aecm->filtDelay = 0;
+ aecm->timeForDelayChange = 0;
+ aecm->knownDelay = 0;
+ aecm->lastDelayDiff = 0;
+
+ memset(&aecm->farendOld[0][0], 0, 160);
+
+ // Default settings.
+ aecConfig.cngMode = AecmTrue;
+ aecConfig.echoMode = 3;
+
+ if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
+ {
+ aecm->lastError = AECM_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_BufferFarend(void *aecmInst, const WebRtc_Word16 *farend,
+ WebRtc_Word16 nrOfSamples)
+{
+ aecmob_t *aecm = aecmInst;
+ WebRtc_Word32 retVal = 0;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (farend == NULL)
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (nrOfSamples != 80 && nrOfSamples != 160)
+ {
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ // TODO: Is this really a good idea?
+ if (!aecm->ECstartup)
+ {
+ WebRtcAecm_DelayComp(aecm);
+ }
+
+ WebRtc_WriteBuffer(aecm->farendBuf, farend, (size_t) nrOfSamples);
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAecm_Process(void *aecmInst, const WebRtc_Word16 *nearendNoisy,
+ const WebRtc_Word16 *nearendClean, WebRtc_Word16 *out,
+ WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf)
+{
+ aecmob_t *aecm = aecmInst;
+ WebRtc_Word32 retVal = 0;
+ short i;
+ short nmbrOfFilledBuffers;
+ short nBlocks10ms;
+ short nFrames;
+#ifdef AEC_DEBUG
+ short msInAECBuf;
+#endif
+
+#ifdef ARM_WINM_LOG
+ __int64 freq, start, end, diff;
+ unsigned int milliseconds;
+ DWORD temp;
+#elif defined MAC_IPHONE_PRINT
+ // double endtime = 0, starttime = 0;
+ struct timeval starttime;
+ struct timeval endtime;
+ static long int timeused = 0;
+ static int timecount = 0;
+#endif
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (nearendNoisy == NULL)
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (out == NULL)
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (nrOfSamples != 80 && nrOfSamples != 160)
+ {
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ if (msInSndCardBuf < 0)
+ {
+ msInSndCardBuf = 0;
+ aecm->lastError = AECM_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ } else if (msInSndCardBuf > 500)
+ {
+ msInSndCardBuf = 500;
+ aecm->lastError = AECM_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ }
+ msInSndCardBuf += 10;
+ aecm->msInSndCardBuf = msInSndCardBuf;
+
+ nFrames = nrOfSamples / FRAME_LEN;
+ nBlocks10ms = nFrames / aecm->aecmCore->mult;
+
+ if (aecm->ECstartup)
+ {
+ if (nearendClean == NULL)
+ {
+ if (out != nearendNoisy)
+ {
+ memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
+ }
+ } else if (out != nearendClean)
+ {
+ memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
+ }
+
+ nmbrOfFilledBuffers =
+ (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
+ // The AECM is in the start up mode
+ // AECM is disabled until the soundcard buffer and farend buffers are OK
+
+ // Mechanism to ensure that the soundcard buffer is reasonably stable.
+ if (aecm->checkBuffSize)
+ {
+ aecm->checkBufSizeCtr++;
+ // Before we fill up the far end buffer we require the amount of data on the
+ // sound card to be stable (+/-8 ms) compared to the first value. This
+ // comparison is made during the following 4 consecutive frames. If it seems
+ // to be stable then we start to fill up the far end buffer.
+
+ if (aecm->counter == 0)
+ {
+ aecm->firstVal = aecm->msInSndCardBuf;
+ aecm->sum = 0;
+ }
+
+ if (abs(aecm->firstVal - aecm->msInSndCardBuf)
+ < WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb))
+ {
+ aecm->sum += aecm->msInSndCardBuf;
+ aecm->counter++;
+ } else
+ {
+ aecm->counter = 0;
+ }
+
+ if (aecm->counter * nBlocks10ms >= 6)
+ {
+ // The farend buffer size is determined in blocks of 80 samples
+ // Use 75% of the average value of the soundcard buffer
+ aecm->bufSizeStart
+ = WEBRTC_SPL_MIN((3 * aecm->sum
+ * aecm->aecmCore->mult) / (aecm->counter * 40), BUF_SIZE_FRAMES);
+ // buffersize has now been determined
+ aecm->checkBuffSize = 0;
+ }
+
+ if (aecm->checkBufSizeCtr * nBlocks10ms > 50)
+ {
+ // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
+ aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf
+ * aecm->aecmCore->mult) / 40, BUF_SIZE_FRAMES);
+ aecm->checkBuffSize = 0;
+ }
+ }
+
+ // if checkBuffSize changed in the if-statement above
+ if (!aecm->checkBuffSize)
+ {
+ // soundcard buffer is now reasonably stable
+ // When the far end buffer is filled with approximately the same amount of
+ // data as the amount on the sound card we end the start up phase and start
+ // to cancel echoes.
+
+ if (nmbrOfFilledBuffers == aecm->bufSizeStart)
+ {
+ aecm->ECstartup = 0; // Enable the AECM
+ } else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
+ {
+ WebRtc_MoveReadPtr(aecm->farendBuf,
+ (int) WebRtc_available_read(aecm->farendBuf)
+ - (int) aecm->bufSizeStart * FRAME_LEN);
+ aecm->ECstartup = 0;
+ }
+ }
+
+ } else
+ {
+ // AECM is enabled
+
+ // Note only 1 block supported for nb and 2 blocks for wb
+ for (i = 0; i < nFrames; i++)
+ {
+ int16_t farend[FRAME_LEN];
+ const int16_t* farend_ptr = NULL;
+
+ nmbrOfFilledBuffers =
+ (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
+
+ // Check that there is data in the far end buffer
+ if (nmbrOfFilledBuffers > 0)
+ {
+ // Get the next 80 samples from the farend buffer
+ WebRtc_ReadBuffer(aecm->farendBuf, (void**) &farend_ptr, farend,
+ FRAME_LEN);
+
+ // Always store the last frame for use when we run out of data
+ memcpy(&(aecm->farendOld[i][0]), farend_ptr,
+ FRAME_LEN * sizeof(short));
+ } else
+ {
+ // We have no data so we use the last played frame
+ memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
+ farend_ptr = farend;
+ }
+
+ // Call buffer delay estimator when all data is extracted,
+ // i,e. i = 0 for NB and i = 1 for WB
+ if ((i == 0 && aecm->sampFreq == 8000) || (i == 1 && aecm->sampFreq == 16000))
+ {
+ WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
+ }
+
+#ifdef ARM_WINM_LOG
+ // measure tick start
+ QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#elif defined MAC_IPHONE_PRINT
+ // starttime = clock()/(double)CLOCKS_PER_SEC;
+ gettimeofday(&starttime, NULL);
+#endif
+ // Call the AECM
+ /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
+ &out[FRAME_LEN * i], aecm->knownDelay);*/
+ if (nearendClean == NULL)
+ {
+ if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
+ farend_ptr,
+ &nearendNoisy[FRAME_LEN * i],
+ NULL,
+ &out[FRAME_LEN * i]) == -1)
+ {
+ return -1;
+ }
+ } else
+ {
+ if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
+ farend_ptr,
+ &nearendNoisy[FRAME_LEN * i],
+ &nearendClean[FRAME_LEN * i],
+ &out[FRAME_LEN * i]) == -1)
+ {
+ return -1;
+ }
+ }
+
+#ifdef ARM_WINM_LOG
+
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+
+ if(end > start)
+ {
+ diff = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ }
+#elif defined MAC_IPHONE_PRINT
+ // endtime = clock()/(double)CLOCKS_PER_SEC;
+ // printf("%f\n", endtime - starttime);
+
+ gettimeofday(&endtime, NULL);
+
+ if( endtime.tv_usec > starttime.tv_usec)
+ {
+ timeused += endtime.tv_usec - starttime.tv_usec;
+ } else
+ {
+ timeused += endtime.tv_usec + 1000000 - starttime.tv_usec;
+ }
+
+ if(++timecount == 1000)
+ {
+ timecount = 0;
+ printf("AEC: %ld\n", timeused);
+ timeused = 0;
+ }
+#endif
+
+ }
+ }
+
+#ifdef AEC_DEBUG
+ msInAECBuf = (short) WebRtc_available_read(aecm->farendBuf) /
+ (kSampMsNb * aecm->aecmCore->mult);
+ fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
+ fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
+#endif
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
+{
+ aecmob_t *aecm = aecmInst;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
+ {
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecm->aecmCore->cngMode = config.cngMode;
+
+ if (config.echoMode < 0 || config.echoMode > 4)
+ {
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecm->echoMode = config.echoMode;
+
+ if (aecm->echoMode == 0)
+ {
+ aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
+ aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
+ aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
+ aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
+ aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 3)
+ - (SUPGAIN_ERROR_PARAM_B >> 3);
+ aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 3)
+ - (SUPGAIN_ERROR_PARAM_D >> 3);
+ } else if (aecm->echoMode == 1)
+ {
+ aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
+ aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
+ aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
+ aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
+ aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 2)
+ - (SUPGAIN_ERROR_PARAM_B >> 2);
+ aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 2)
+ - (SUPGAIN_ERROR_PARAM_D >> 2);
+ } else if (aecm->echoMode == 2)
+ {
+ aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
+ aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
+ aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
+ aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
+ aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 1)
+ - (SUPGAIN_ERROR_PARAM_B >> 1);
+ aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 1)
+ - (SUPGAIN_ERROR_PARAM_D >> 1);
+ } else if (aecm->echoMode == 3)
+ {
+ aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
+ aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
+ aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
+ aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
+ aecm->aecmCore->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
+ aecm->aecmCore->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
+ } else if (aecm->echoMode == 4)
+ {
+ aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
+ aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
+ aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
+ aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
+ aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A << 1)
+ - (SUPGAIN_ERROR_PARAM_B << 1);
+ aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B << 1)
+ - (SUPGAIN_ERROR_PARAM_D << 1);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst, AecmConfig *config)
+{
+ aecmob_t *aecm = aecmInst;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (config == NULL)
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ config->cngMode = aecm->aecmCore->cngMode;
+ config->echoMode = aecm->echoMode;
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
+ const void* echo_path,
+ size_t size_bytes)
+{
+ aecmob_t *aecm = aecmInst;
+ const WebRtc_Word16* echo_path_ptr = echo_path;
+
+ if ((aecm == NULL) || (echo_path == NULL))
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (size_bytes != WebRtcAecm_echo_path_size_bytes())
+ {
+ // Input channel size does not match the size of AECM
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
+ void* echo_path,
+ size_t size_bytes)
+{
+ aecmob_t *aecm = aecmInst;
+ WebRtc_Word16* echo_path_ptr = echo_path;
+
+ if ((aecm == NULL) || (echo_path == NULL))
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (size_bytes != WebRtcAecm_echo_path_size_bytes())
+ {
+ // Input channel size does not match the size of AECM
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
+ return 0;
+}
+
+size_t WebRtcAecm_echo_path_size_bytes()
+{
+ return (PART_LEN1 * sizeof(WebRtc_Word16));
+}
+
+WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst)
+{
+ aecmob_t *aecm = aecmInst;
+
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ return aecm->lastError;
+}
+
+static int WebRtcAecm_EstBufDelay(aecmob_t *aecm, short msInSndCardBuf)
+{
+ short delayNew, nSampSndCard;
+ short nSampFar = (short) WebRtc_available_read(aecm->farendBuf);
+ short diff;
+
+ nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
+
+ delayNew = nSampSndCard - nSampFar;
+
+ if (delayNew < FRAME_LEN)
+ {
+ WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
+ delayNew += FRAME_LEN;
+ }
+
+ aecm->filtDelay = WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
+
+ diff = aecm->filtDelay - aecm->knownDelay;
+ if (diff > 224)
+ {
+ if (aecm->lastDelayDiff < 96)
+ {
+ aecm->timeForDelayChange = 0;
+ } else
+ {
+ aecm->timeForDelayChange++;
+ }
+ } else if (diff < 96 && aecm->knownDelay > 0)
+ {
+ if (aecm->lastDelayDiff > 224)
+ {
+ aecm->timeForDelayChange = 0;
+ } else
+ {
+ aecm->timeForDelayChange++;
+ }
+ } else
+ {
+ aecm->timeForDelayChange = 0;
+ }
+ aecm->lastDelayDiff = diff;
+
+ if (aecm->timeForDelayChange > 25)
+ {
+ aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
+ }
+ return 0;
+}
+
+static int WebRtcAecm_DelayComp(aecmob_t *aecm)
+{
+ int nSampFar = (int) WebRtc_available_read(aecm->farendBuf);
+ int nSampSndCard, delayNew, nSampAdd;
+ const int maxStuffSamp = 10 * FRAME_LEN;
+
+ nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
+ delayNew = nSampSndCard - nSampFar;
+
+ if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult)
+ {
+ // The difference of the buffer sizes is larger than the maximum
+ // allowed known delay. Compensate by stuffing the buffer.
+ nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
+ FRAME_LEN));
+ nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
+
+ WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
+ aecm->delayChange = 1; // the delay needs to be updated
+ }
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h b/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h
new file mode 100644
index 0000000..da0ad86
--- /dev/null
+++ b/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h
@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+
+#include "typedefs.h"
+
+enum {
+ AecmFalse = 0,
+ AecmTrue
+};
+
+// Errors
+#define AECM_UNSPECIFIED_ERROR 12000
+#define AECM_UNSUPPORTED_FUNCTION_ERROR 12001
+#define AECM_UNINITIALIZED_ERROR 12002
+#define AECM_NULL_POINTER_ERROR 12003
+#define AECM_BAD_PARAMETER_ERROR 12004
+
+// Warnings
+#define AECM_BAD_PARAMETER_WARNING 12100
+
+typedef struct {
+ WebRtc_Word16 cngMode; // AECM_FALSE, AECM_TRUE (default)
+ WebRtc_Word16 echoMode; // 0, 1, 2, 3 (default), 4
+} AecmConfig;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Allocates the memory needed by the AECM. The memory needs to be
+ * initialized separately using the WebRtcAecm_Init() function.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void **aecmInst Pointer to the AECM instance to be
+ * created and initialized
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_Create(void **aecmInst);
+
+/*
+ * This function releases the memory allocated by WebRtcAecm_Create()
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_Free(void *aecmInst);
+
+/*
+ * Initializes an AECM instance.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ * WebRtc_Word32 sampFreq Sampling frequency of data
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_Init(void* aecmInst,
+ WebRtc_Word32 sampFreq);
+
+/*
+ * Inserts an 80 or 160 sample block of data into the farend buffer.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ * WebRtc_Word16 *farend In buffer containing one frame of
+ * farend signal
+ * WebRtc_Word16 nrOfSamples Number of samples in farend buffer
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_BufferFarend(void* aecmInst,
+ const WebRtc_Word16* farend,
+ WebRtc_Word16 nrOfSamples);
+
+/*
+ * Runs the AECM on an 80 or 160 sample blocks of data.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ * WebRtc_Word16 *nearendNoisy In buffer containing one frame of
+ * reference nearend+echo signal. If
+ * noise reduction is active, provide
+ * the noisy signal here.
+ * WebRtc_Word16 *nearendClean In buffer containing one frame of
+ * nearend+echo signal. If noise
+ * reduction is active, provide the
+ * clean signal here. Otherwise pass a
+ * NULL pointer.
+ * WebRtc_Word16 nrOfSamples Number of samples in nearend buffer
+ * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
+ * system buffers
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16 *out Out buffer, one frame of processed nearend
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_Process(void* aecmInst,
+ const WebRtc_Word16* nearendNoisy,
+ const WebRtc_Word16* nearendClean,
+ WebRtc_Word16* out,
+ WebRtc_Word16 nrOfSamples,
+ WebRtc_Word16 msInSndCardBuf);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ * AecmConfig config Config instance that contains all
+ * properties to be set
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_set_config(void* aecmInst,
+ AecmConfig config);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * AecmConfig *config Pointer to the config instance that
+ * all properties will be written to
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst,
+ AecmConfig *config);
+
+/*
+ * This function enables the user to set the echo path on-the-fly.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* aecmInst Pointer to the AECM instance
+ * void* echo_path Pointer to the echo path to be set
+ * size_t size_bytes Size in bytes of the echo path
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
+ const void* echo_path,
+ size_t size_bytes);
+
+/*
+ * This function enables the user to get the currently used echo path
+ * on-the-fly
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* aecmInst Pointer to the AECM instance
+ * void* echo_path Pointer to echo path
+ * size_t size_bytes Size in bytes of the echo path
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
+ void* echo_path,
+ size_t size_bytes);
+
+/*
+ * This function enables the user to get the echo path size in bytes
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * size_t return : size in bytes
+ */
+size_t WebRtcAecm_echo_path_size_bytes();
+
+/*
+ * Gets the last error code.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void *aecmInst Pointer to the AECM instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 11000-11100: error code
+ */
+WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst);
+
+#ifdef __cplusplus
+}
+#endif
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
diff --git a/trunk/src/modules/audio_processing/agc/Android.mk b/trunk/src/modules/audio_processing/agc/Android.mk
new file mode 100644
index 0000000..5eceb14
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_agc
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ analog_agc.c \
+ digital_agc.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/agc/agc.gypi b/trunk/src/modules/audio_processing/agc/agc.gypi
new file mode 100644
index 0000000..f96f237
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/agc.gypi
@@ -0,0 +1,34 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'agc',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/gain_control.h',
+ 'analog_agc.c',
+ 'analog_agc.h',
+ 'digital_agc.c',
+ 'digital_agc.h',
+ ],
+ },
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/agc/analog_agc.c b/trunk/src/modules/audio_processing/agc/analog_agc.c
new file mode 100644
index 0000000..558a6cb
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/analog_agc.c
@@ -0,0 +1,1694 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* analog_agc.c
+ *
+ * Using a feedback system, determines an appropriate analog volume level
+ * given an input signal and current volume level. Targets a conservative
+ * signal level and is intended for use with a digital AGC to apply
+ * additional gain.
+ *
+ */
+
+#include <assert.h>
+#include <stdlib.h>
+#ifdef AGC_DEBUG //test log
+#include <stdio.h>
+#endif
+#include "analog_agc.h"
+
+/* The slope of in Q13*/
+static const WebRtc_Word16 kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
+
+/* The offset in Q14 */
+static const WebRtc_Word16 kOffset1[8] = {25395, 23911, 22206, 20737, 19612, 18805, 17951,
+ 17367};
+
+/* The slope of in Q13*/
+static const WebRtc_Word16 kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
+
+/* The offset in Q14 */
+static const WebRtc_Word16 kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670,
+ 17286};
+
+static const WebRtc_Word16 kMuteGuardTimeMs = 8000;
+static const WebRtc_Word16 kInitCheck = 42;
+
+/* Default settings if config is not used */
+#define AGC_DEFAULT_TARGET_LEVEL 3
+#define AGC_DEFAULT_COMP_GAIN 9
+/* This is the target level for the analog part in ENV scale. To convert to RMS scale you
+ * have to add OFFSET_ENV_TO_RMS.
+ */
+#define ANALOG_TARGET_LEVEL 11
+#define ANALOG_TARGET_LEVEL_2 5 // ANALOG_TARGET_LEVEL / 2
+/* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
+ * varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future replace it with
+ * a table.
+ */
+#define OFFSET_ENV_TO_RMS 9
+/* The reference input level at which the digital part gives an output of targetLevelDbfs
+ * (desired level) if we have no compression gain. This level should be set high enough not
+ * to compress the peaks due to the dynamics.
+ */
+#define DIGITAL_REF_AT_0_COMP_GAIN 4
+/* Speed of reference level decrease.
+ */
+#define DIFF_REF_TO_ANALOG 5
+
+#ifdef MIC_LEVEL_FEEDBACK
+#define NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET 7
+#endif
+/* Size of analog gain table */
+#define GAIN_TBL_LEN 32
+/* Matlab code:
+ * fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
+ */
+/* Q12 */
+static const WebRtc_UWord16 kGainTableAnalog[GAIN_TBL_LEN] = {4096, 4251, 4412, 4579, 4752,
+ 4932, 5118, 5312, 5513, 5722, 5938, 6163, 6396, 6638, 6889, 7150, 7420, 7701, 7992,
+ 8295, 8609, 8934, 9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};
+
+/* Gain/Suppression tables for virtual Mic (in Q10) */
+static const WebRtc_UWord16 kGainTableVirtualMic[128] = {1052, 1081, 1110, 1141, 1172, 1204,
+ 1237, 1271, 1305, 1341, 1378, 1416, 1454, 1494, 1535, 1577, 1620, 1664, 1710, 1757,
+ 1805, 1854, 1905, 1957, 2010, 2065, 2122, 2180, 2239, 2301, 2364, 2428, 2495, 2563,
+ 2633, 2705, 2779, 2855, 2933, 3013, 3096, 3180, 3267, 3357, 3449, 3543, 3640, 3739,
+ 3842, 3947, 4055, 4166, 4280, 4397, 4517, 4640, 4767, 4898, 5032, 5169, 5311, 5456,
+ 5605, 5758, 5916, 6078, 6244, 6415, 6590, 6770, 6956, 7146, 7341, 7542, 7748, 7960,
+ 8178, 8402, 8631, 8867, 9110, 9359, 9615, 9878, 10148, 10426, 10711, 11004, 11305,
+ 11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807, 15212, 15628,
+ 16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923, 20468, 21028, 21603,
+ 22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808, 27541, 28295, 29069, 29864,
+ 30681, 31520, 32382};
+static const WebRtc_UWord16 kSuppressionTableVirtualMic[128] = {1024, 1006, 988, 970, 952,
+ 935, 918, 902, 886, 870, 854, 839, 824, 809, 794, 780, 766, 752, 739, 726, 713, 700,
+ 687, 675, 663, 651, 639, 628, 616, 605, 594, 584, 573, 563, 553, 543, 533, 524, 514,
+ 505, 496, 487, 478, 470, 461, 453, 445, 437, 429, 421, 414, 406, 399, 392, 385, 378,
+ 371, 364, 358, 351, 345, 339, 333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278,
+ 273, 268, 263, 258, 254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
+ 200, 197, 193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155, 153, 150,
+ 147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118, 116, 114, 112, 110,
+ 108, 106, 104, 102};
+
+/* Table for target energy levels. Values in Q(-7)
+ * Matlab code
+ * targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n', round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
+
+static const WebRtc_Word32 kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67264106,
+ 53429779, 42440782, 33711911, 26778323, 21270778, 16895980, 13420954, 10660642,
+ 8468049, 6726411, 5342978, 4244078, 3371191, 2677832, 2127078, 1689598, 1342095,
+ 1066064, 846805, 672641, 534298, 424408, 337119, 267783, 212708, 168960, 134210,
+ 106606, 84680, 67264, 53430, 42441, 33712, 26778, 21271, 16896, 13421, 10661, 8468,
+ 6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
+ 213, 169, 134, 107, 85, 67};
+
+int WebRtcAgc_AddMic(void *state, WebRtc_Word16 *in_mic, WebRtc_Word16 *in_mic_H,
+ WebRtc_Word16 samples)
+{
+ WebRtc_Word32 nrg, max_nrg, sample, tmp32;
+ WebRtc_Word32 *ptr;
+ WebRtc_UWord16 targetGainIdx, gain;
+ WebRtc_Word16 i, n, L, M, subFrames, tmp16, tmp_speech[16];
+ Agc_t *stt;
+ stt = (Agc_t *)state;
+
+ //default/initial values corresponding to 10ms for wb and swb
+ M = 10;
+ L = 16;
+ subFrames = 160;
+
+ if (stt->fs == 8000)
+ {
+ if (samples == 80)
+ {
+ subFrames = 80;
+ M = 10;
+ L = 8;
+ } else if (samples == 160)
+ {
+ subFrames = 80;
+ M = 20;
+ L = 8;
+ } else
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_mic, frame %d: Invalid number of samples\n\n",
+ (stt->fcount + 1));
+#endif
+ return -1;
+ }
+ } else if (stt->fs == 16000)
+ {
+ if (samples == 160)
+ {
+ subFrames = 160;
+ M = 10;
+ L = 16;
+ } else if (samples == 320)
+ {
+ subFrames = 160;
+ M = 20;
+ L = 16;
+ } else
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_mic, frame %d: Invalid number of samples\n\n",
+ (stt->fcount + 1));
+#endif
+ return -1;
+ }
+ } else if (stt->fs == 32000)
+ {
+ /* SWB is processed as 160 sample for L and H bands */
+ if (samples == 160)
+ {
+ subFrames = 160;
+ M = 10;
+ L = 16;
+ } else
+ {
+#ifdef AGC_DEBUG
+ fprintf(stt->fpt,
+ "AGC->add_mic, frame %d: Invalid sample rate\n\n",
+ (stt->fcount + 1));
+#endif
+ return -1;
+ }
+ }
+
+ /* Check for valid pointers based on sampling rate */
+ if ((stt->fs == 32000) && (in_mic_H == NULL))
+ {
+ return -1;
+ }
+ /* Check for valid pointer for low band */
+ if (in_mic == NULL)
+ {
+ return -1;
+ }
+
+ /* apply slowly varying digital gain */
+ if (stt->micVol > stt->maxAnalog)
+ {
+ /* |maxLevel| is strictly >= |micVol|, so this condition should be
+ * satisfied here, ensuring there is no divide-by-zero. */
+ assert(stt->maxLevel > stt->maxAnalog);
+
+ /* Q1 */
+ tmp16 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog);
+ tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
+ tmp16 = (WebRtc_Word16)(stt->maxLevel - stt->maxAnalog);
+ targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16);
+ assert(targetGainIdx < GAIN_TBL_LEN);
+
+ /* Increment through the table towards the target gain.
+ * If micVol drops below maxAnalog, we allow the gain
+ * to be dropped immediately. */
+ if (stt->gainTableIdx < targetGainIdx)
+ {
+ stt->gainTableIdx++;
+ } else if (stt->gainTableIdx > targetGainIdx)
+ {
+ stt->gainTableIdx--;
+ }
+
+ /* Q12 */
+ gain = kGainTableAnalog[stt->gainTableIdx];
+
+ for (i = 0; i < samples; i++)
+ {
+ // For lower band
+ tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic[i], gain);
+ sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
+ if (sample > 32767)
+ {
+ in_mic[i] = 32767;
+ } else if (sample < -32768)
+ {
+ in_mic[i] = -32768;
+ } else
+ {
+ in_mic[i] = (WebRtc_Word16)sample;
+ }
+
+ // For higher band
+ if (stt->fs == 32000)
+ {
+ tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic_H[i], gain);
+ sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
+ if (sample > 32767)
+ {
+ in_mic_H[i] = 32767;
+ } else if (sample < -32768)
+ {
+ in_mic_H[i] = -32768;
+ } else
+ {
+ in_mic_H[i] = (WebRtc_Word16)sample;
+ }
+ }
+ }
+ } else
+ {
+ stt->gainTableIdx = 0;
+ }
+
+ /* compute envelope */
+ if ((M == 10) && (stt->inQueue > 0))
+ {
+ ptr = stt->env[1];
+ } else
+ {
+ ptr = stt->env[0];
+ }
+
+ for (i = 0; i < M; i++)
+ {
+ /* iterate over samples */
+ max_nrg = 0;
+ for (n = 0; n < L; n++)
+ {
+ nrg = WEBRTC_SPL_MUL_16_16(in_mic[i * L + n], in_mic[i * L + n]);
+ if (nrg > max_nrg)
+ {
+ max_nrg = nrg;
+ }
+ }
+ ptr[i] = max_nrg;
+ }
+
+ /* compute energy */
+ if ((M == 10) && (stt->inQueue > 0))
+ {
+ ptr = stt->Rxx16w32_array[1];
+ } else
+ {
+ ptr = stt->Rxx16w32_array[0];
+ }
+
+ for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++)
+ {
+ if (stt->fs == 16000)
+ {
+ WebRtcSpl_DownsampleBy2(&in_mic[i * 32], 32, tmp_speech, stt->filterState);
+ } else
+ {
+ memcpy(tmp_speech, &in_mic[i * 16], 16 * sizeof(short));
+ }
+ /* Compute energy in blocks of 16 samples */
+ ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
+ }
+
+ /* update queue information */
+ if ((stt->inQueue == 0) && (M == 10))
+ {
+ stt->inQueue = 1;
+ } else
+ {
+ stt->inQueue = 2;
+ }
+
+ /* call VAD (use low band only) */
+ for (i = 0; i < samples; i += subFrames)
+ {
+ WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames);
+ }
+
+ return 0;
+}
+
+int WebRtcAgc_AddFarend(void *state, const WebRtc_Word16 *in_far, WebRtc_Word16 samples)
+{
+ WebRtc_Word32 errHandle = 0;
+ WebRtc_Word16 i, subFrames;
+ Agc_t *stt;
+ stt = (Agc_t *)state;
+
+ if (stt == NULL)
+ {
+ return -1;
+ }
+
+ if (stt->fs == 8000)
+ {
+ if ((samples != 80) && (samples != 160))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
+ stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 80;
+ } else if (stt->fs == 16000)
+ {
+ if ((samples != 160) && (samples != 320))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
+ stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 160;
+ } else if (stt->fs == 32000)
+ {
+ if ((samples != 160) && (samples != 320))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
+ stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 160;
+ } else
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->add_far_end, frame %d: Invalid sample rate\n\n",
+ stt->fcount + 1);
+#endif
+ return -1;
+ }
+
+ for (i = 0; i < samples; i += subFrames)
+ {
+ errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames);
+ }
+
+ return errHandle;
+}
+
+int WebRtcAgc_VirtualMic(void *agcInst, WebRtc_Word16 *in_near, WebRtc_Word16 *in_near_H,
+ WebRtc_Word16 samples, WebRtc_Word32 micLevelIn,
+ WebRtc_Word32 *micLevelOut)
+{
+ WebRtc_Word32 tmpFlt, micLevelTmp, gainIdx;
+ WebRtc_UWord16 gain;
+ WebRtc_Word16 ii;
+ Agc_t *stt;
+
+ WebRtc_UWord32 nrg;
+ WebRtc_Word16 sampleCntr;
+ WebRtc_UWord32 frameNrg = 0;
+ WebRtc_UWord32 frameNrgLimit = 5500;
+ WebRtc_Word16 numZeroCrossing = 0;
+ const WebRtc_Word16 kZeroCrossingLowLim = 15;
+ const WebRtc_Word16 kZeroCrossingHighLim = 20;
+
+ stt = (Agc_t *)agcInst;
+
+ /*
+ * Before applying gain decide if this is a low-level signal.
+ * The idea is that digital AGC will not adapt to low-level
+ * signals.
+ */
+ if (stt->fs != 8000)
+ {
+ frameNrgLimit = frameNrgLimit << 1;
+ }
+
+ frameNrg = WEBRTC_SPL_MUL_16_16(in_near[0], in_near[0]);
+ for (sampleCntr = 1; sampleCntr < samples; sampleCntr++)
+ {
+
+ // increment frame energy if it is less than the limit
+ // the correct value of the energy is not important
+ if (frameNrg < frameNrgLimit)
+ {
+ nrg = WEBRTC_SPL_MUL_16_16(in_near[sampleCntr], in_near[sampleCntr]);
+ frameNrg += nrg;
+ }
+
+ // Count the zero crossings
+ numZeroCrossing += ((in_near[sampleCntr] ^ in_near[sampleCntr - 1]) < 0);
+ }
+
+ if ((frameNrg < 500) || (numZeroCrossing <= 5))
+ {
+ stt->lowLevelSignal = 1;
+ } else if (numZeroCrossing <= kZeroCrossingLowLim)
+ {
+ stt->lowLevelSignal = 0;
+ } else if (frameNrg <= frameNrgLimit)
+ {
+ stt->lowLevelSignal = 1;
+ } else if (numZeroCrossing >= kZeroCrossingHighLim)
+ {
+ stt->lowLevelSignal = 1;
+ } else
+ {
+ stt->lowLevelSignal = 0;
+ }
+
+ micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
+ /* Set desired level */
+ gainIdx = stt->micVol;
+ if (stt->micVol > stt->maxAnalog)
+ {
+ gainIdx = stt->maxAnalog;
+ }
+ if (micLevelTmp != stt->micRef)
+ {
+ /* Something has happened with the physical level, restart. */
+ stt->micRef = micLevelTmp;
+ stt->micVol = 127;
+ *micLevelOut = 127;
+ stt->micGainIdx = 127;
+ gainIdx = 127;
+ }
+ /* Pre-process the signal to emulate the microphone level. */
+ /* Take one step at a time in the gain table. */
+ if (gainIdx > 127)
+ {
+ gain = kGainTableVirtualMic[gainIdx - 128];
+ } else
+ {
+ gain = kSuppressionTableVirtualMic[127 - gainIdx];
+ }
+ for (ii = 0; ii < samples; ii++)
+ {
+ tmpFlt = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_U16(in_near[ii], gain), 10);
+ if (tmpFlt > 32767)
+ {
+ tmpFlt = 32767;
+ gainIdx--;
+ if (gainIdx >= 127)
+ {
+ gain = kGainTableVirtualMic[gainIdx - 127];
+ } else
+ {
+ gain = kSuppressionTableVirtualMic[127 - gainIdx];
+ }
+ }
+ if (tmpFlt < -32768)
+ {
+ tmpFlt = -32768;
+ gainIdx--;
+ if (gainIdx >= 127)
+ {
+ gain = kGainTableVirtualMic[gainIdx - 127];
+ } else
+ {
+ gain = kSuppressionTableVirtualMic[127 - gainIdx];
+ }
+ }
+ in_near[ii] = (WebRtc_Word16)tmpFlt;
+ if (stt->fs == 32000)
+ {
+ tmpFlt = WEBRTC_SPL_MUL_16_U16(in_near_H[ii], gain);
+ tmpFlt = WEBRTC_SPL_RSHIFT_W32(tmpFlt, 10);
+ if (tmpFlt > 32767)
+ {
+ tmpFlt = 32767;
+ }
+ if (tmpFlt < -32768)
+ {
+ tmpFlt = -32768;
+ }
+ in_near_H[ii] = (WebRtc_Word16)tmpFlt;
+ }
+ }
+ /* Set the level we (finally) used */
+ stt->micGainIdx = gainIdx;
+// *micLevelOut = stt->micGainIdx;
+ *micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale);
+ /* Add to Mic as if it was the output from a true microphone */
+ if (WebRtcAgc_AddMic(agcInst, in_near, in_near_H, samples) != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+void WebRtcAgc_UpdateAgcThresholds(Agc_t *stt)
+{
+
+ WebRtc_Word16 tmp16;
+#ifdef MIC_LEVEL_FEEDBACK
+ int zeros;
+
+ if (stt->micLvlSat)
+ {
+ /* Lower the analog target level since we have reached its maximum */
+ zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
+ stt->targetIdxOffset = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2);
+ }
+#endif
+
+ /* Set analog target level in envelope dBOv scale */
+ tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
+ tmp16 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32)tmp16, ANALOG_TARGET_LEVEL);
+ stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
+ if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN)
+ {
+ stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
+ }
+ if (stt->agcMode == kAgcModeFixedDigital)
+ {
+ /* Adjust for different parameter interpretation in FixedDigital mode */
+ stt->analogTarget = stt->compressionGaindB;
+ }
+#ifdef MIC_LEVEL_FEEDBACK
+ stt->analogTarget += stt->targetIdxOffset;
+#endif
+ /* Since the offset between RMS and ENV is not constant, we should make this into a
+ * table, but for now, we'll stick with a constant, tuned for the chosen analog
+ * target level.
+ */
+ stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
+#ifdef MIC_LEVEL_FEEDBACK
+ stt->targetIdx += stt->targetIdxOffset;
+#endif
+ /* Analog adaptation limits */
+ /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
+ stt->analogTargetLevel = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
+ stt->startUpperLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 1];/* -19 dBov */
+ stt->startLowerLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 1];/* -21 dBov */
+ stt->upperPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 2];/* -18 dBov */
+ stt->lowerPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 2];/* -22 dBov */
+ stt->upperSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 5];/* -15 dBov */
+ stt->lowerSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 5];/* -25 dBov */
+ stt->upperLimit = stt->startUpperLimit;
+ stt->lowerLimit = stt->startLowerLimit;
+}
+
+void WebRtcAgc_SaturationCtrl(Agc_t *stt, WebRtc_UWord8 *saturated, WebRtc_Word32 *env)
+{
+ WebRtc_Word16 i, tmpW16;
+
+ /* Check if the signal is saturated */
+ for (i = 0; i < 10; i++)
+ {
+ tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(env[i], 20);
+ if (tmpW16 > 875)
+ {
+ stt->envSum += tmpW16;
+ }
+ }
+
+ if (stt->envSum > 25000)
+ {
+ *saturated = 1;
+ stt->envSum = 0;
+ }
+
+ /* stt->envSum *= 0.99; */
+ stt->envSum = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum,
+ (WebRtc_Word16)32440, 15);
+}
+
+void WebRtcAgc_ZeroCtrl(Agc_t *stt, WebRtc_Word32 *inMicLevel, WebRtc_Word32 *env)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word32 tmp32 = 0;
+ WebRtc_Word32 midVal;
+
+ /* Is the input signal zero? */
+ for (i = 0; i < 10; i++)
+ {
+ tmp32 += env[i];
+ }
+
+ /* Each block is allowed to have a few non-zero
+ * samples.
+ */
+ if (tmp32 < 500)
+ {
+ stt->msZero += 10;
+ } else
+ {
+ stt->msZero = 0;
+ }
+
+ if (stt->muteGuardMs > 0)
+ {
+ stt->muteGuardMs -= 10;
+ }
+
+ if (stt->msZero > 500)
+ {
+ stt->msZero = 0;
+
+ /* Increase microphone level only if it's less than 50% */
+ midVal = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog + stt->minLevel + 1, 1);
+ if (*inMicLevel < midVal)
+ {
+ /* *inMicLevel *= 1.1; */
+ tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel);
+ *inMicLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
+ /* Reduces risk of a muted mic repeatedly triggering excessive levels due
+ * to zero signal detection. */
+ *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
+ stt->micVol = *inMicLevel;
+ }
+
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
+ stt->fcount, stt->micVol);
+#endif
+
+ stt->activeSpeech = 0;
+ stt->Rxx16_LPw32Max = 0;
+
+ /* The AGC has a tendency (due to problems with the VAD parameters), to
+ * vastly increase the volume after a muting event. This timer prevents
+ * upwards adaptation for a short period. */
+ stt->muteGuardMs = kMuteGuardTimeMs;
+ }
+}
+
+void WebRtcAgc_SpeakerInactiveCtrl(Agc_t *stt)
+{
+ /* Check if the near end speaker is inactive.
+ * If that is the case the VAD threshold is
+ * increased since the VAD speech model gets
+ * more sensitive to any sound after a long
+ * silence.
+ */
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word16 vadThresh;
+
+ if (stt->vadMic.stdLongTerm < 2500)
+ {
+ stt->vadThreshold = 1500;
+ } else
+ {
+ vadThresh = kNormalVadThreshold;
+ if (stt->vadMic.stdLongTerm < 4500)
+ {
+ /* Scale between min and max threshold */
+ vadThresh += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1);
+ }
+
+ /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
+ tmp32 = (WebRtc_Word32)vadThresh;
+ tmp32 += WEBRTC_SPL_MUL_16_16((WebRtc_Word16)31, stt->vadThreshold);
+ stt->vadThreshold = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 5);
+ }
+}
+
+void WebRtcAgc_ExpCurve(WebRtc_Word16 volume, WebRtc_Word16 *index)
+{
+ // volume in Q14
+ // index in [0-7]
+ /* 8 different curves */
+ if (volume > 5243)
+ {
+ if (volume > 7864)
+ {
+ if (volume > 12124)
+ {
+ *index = 7;
+ } else
+ {
+ *index = 6;
+ }
+ } else
+ {
+ if (volume > 6554)
+ {
+ *index = 5;
+ } else
+ {
+ *index = 4;
+ }
+ }
+ } else
+ {
+ if (volume > 2621)
+ {
+ if (volume > 3932)
+ {
+ *index = 3;
+ } else
+ {
+ *index = 2;
+ }
+ } else
+ {
+ if (volume > 1311)
+ {
+ *index = 1;
+ } else
+ {
+ *index = 0;
+ }
+ }
+ }
+}
+
+WebRtc_Word32 WebRtcAgc_ProcessAnalog(void *state, WebRtc_Word32 inMicLevel,
+ WebRtc_Word32 *outMicLevel,
+ WebRtc_Word16 vadLogRatio,
+ WebRtc_Word16 echo, WebRtc_UWord8 *saturationWarning)
+{
+ WebRtc_UWord32 tmpU32;
+ WebRtc_Word32 Rxx16w32, tmp32;
+ WebRtc_Word32 inMicLevelTmp, lastMicVol;
+ WebRtc_Word16 i;
+ WebRtc_UWord8 saturated = 0;
+ Agc_t *stt;
+
+ stt = (Agc_t *)state;
+ inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
+
+ if (inMicLevelTmp > stt->maxAnalog)
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
+#endif
+ return -1;
+ } else if (inMicLevelTmp < stt->minLevel)
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
+#endif
+ return -1;
+ }
+
+ if (stt->firstCall == 0)
+ {
+ WebRtc_Word32 tmpVol;
+ stt->firstCall = 1;
+ tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
+ tmpVol = (stt->minLevel + tmp32);
+
+ /* If the mic level is very low at start, increase it! */
+ if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog))
+ {
+ inMicLevelTmp = tmpVol;
+ }
+ stt->micVol = inMicLevelTmp;
+ }
+
+ /* Set the mic level to the previous output value if there is digital input gain */
+ if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog))
+ {
+ inMicLevelTmp = stt->micVol;
+ }
+
+ /* If the mic level was manually changed to a very low value raise it! */
+ if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput))
+ {
+ tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
+ inMicLevelTmp = (stt->minLevel + tmp32);
+ stt->micVol = inMicLevelTmp;
+#ifdef MIC_LEVEL_FEEDBACK
+ //stt->numBlocksMicLvlSat = 0;
+#endif
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
+ stt->fcount);
+#endif
+ }
+
+ if (inMicLevelTmp != stt->micVol)
+ {
+ // Incoming level mismatch; update our level.
+ // This could be the case if the volume is changed manually, or if the
+ // sound device has a low volume resolution.
+ stt->micVol = inMicLevelTmp;
+ }
+
+ if (inMicLevelTmp > stt->maxLevel)
+ {
+ // Always allow the user to raise the volume above the maxLevel.
+ stt->maxLevel = inMicLevelTmp;
+ }
+
+ // Store last value here, after we've taken care of manual updates etc.
+ lastMicVol = stt->micVol;
+
+ /* Checks if the signal is saturated. Also a check if individual samples
+ * are larger than 12000 is done. If they are the counter for increasing
+ * the volume level is set to -100ms
+ */
+ WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);
+
+ /* The AGC is always allowed to lower the level if the signal is saturated */
+ if (saturated == 1)
+ {
+ /* Lower the recording level
+ * Rxx160_LP is adjusted down because it is so slow it could
+ * cause the AGC to make wrong decisions. */
+ /* stt->Rxx160_LPw32 *= 0.875; */
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7);
+
+ stt->zeroCtrlMax = stt->micVol;
+
+ /* stt->micVol *= 0.903; */
+ tmp32 = inMicLevelTmp - stt->minLevel;
+ tmpU32 = WEBRTC_SPL_UMUL(29591, (WebRtc_UWord32)(tmp32));
+ stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
+ if (stt->micVol > lastMicVol - 2)
+ {
+ stt->micVol = lastMicVol - 2;
+ }
+ inMicLevelTmp = stt->micVol;
+
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
+ stt->fcount, stt->micVol);
+#endif
+
+ if (stt->micVol < stt->minOutput)
+ {
+ *saturationWarning = 1;
+ }
+
+ /* Reset counter for decrease of volume level to avoid
+ * decreasing too much. The saturation control can still
+ * lower the level if needed. */
+ stt->msTooHigh = -100;
+
+ /* Enable the control mechanism to ensure that our measure,
+ * Rxx160_LP, is in the correct range. This must be done since
+ * the measure is very slow. */
+ stt->activeSpeech = 0;
+ stt->Rxx16_LPw32Max = 0;
+
+ /* Reset to initial values */
+ stt->msecSpeechInnerChange = kMsecSpeechInner;
+ stt->msecSpeechOuterChange = kMsecSpeechOuter;
+ stt->changeToSlowMode = 0;
+
+ stt->muteGuardMs = 0;
+
+ stt->upperLimit = stt->startUpperLimit;
+ stt->lowerLimit = stt->startLowerLimit;
+#ifdef MIC_LEVEL_FEEDBACK
+ //stt->numBlocksMicLvlSat = 0;
+#endif
+ }
+
+ /* Check if the input speech is zero. If so the mic volume
+ * is increased. On some computers the input is zero up as high
+ * level as 17% */
+ WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);
+
+ /* Check if the near end speaker is inactive.
+ * If that is the case the VAD threshold is
+ * increased since the VAD speech model gets
+ * more sensitive to any sound after a long
+ * silence.
+ */
+ WebRtcAgc_SpeakerInactiveCtrl(stt);
+
+ for (i = 0; i < 5; i++)
+ {
+ /* Computed on blocks of 16 samples */
+
+ Rxx16w32 = stt->Rxx16w32_array[0][i];
+
+ /* Rxx160w32 in Q(-7) */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos], 3);
+ stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
+ stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
+
+ /* Circular buffer */
+ stt->Rxx16pos++;
+ if (stt->Rxx16pos == RXX_BUFFER_LEN)
+ {
+ stt->Rxx16pos = 0;
+ }
+
+ /* Rxx16_LPw32 in Q(-4) */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_LPw32, kAlphaShortTerm);
+ stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;
+
+ if (vadLogRatio > stt->vadThreshold)
+ {
+ /* Speech detected! */
+
+ /* Check if Rxx160_LP is in the correct range. If
+ * it is too high/low then we set it to the maximum of
+ * Rxx16_LPw32 during the first 200ms of speech.
+ */
+ if (stt->activeSpeech < 250)
+ {
+ stt->activeSpeech += 2;
+
+ if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max)
+ {
+ stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
+ }
+ } else if (stt->activeSpeech == 250)
+ {
+ stt->activeSpeech += 2;
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx16_LPw32Max, 3);
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN);
+ }
+
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160w32 - stt->Rxx160_LPw32, kAlphaLongTerm);
+ stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;
+
+ if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit)
+ {
+ stt->msTooHigh += 2;
+ stt->msTooLow = 0;
+ stt->changeToSlowMode = 0;
+
+ if (stt->msTooHigh > stt->msecSpeechOuterChange)
+ {
+ stt->msTooHigh = 0;
+
+ /* Lower the recording level */
+ /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
+
+ /* Reduce the max gain to avoid excessive oscillation
+ * (but never drop below the maximum analog level).
+ * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
+ */
+ tmp32 = (15 * stt->maxLevel) + stt->micVol;
+ stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
+ stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
+
+ stt->zeroCtrlMax = stt->micVol;
+
+ /* 0.95 in Q15 */
+ tmp32 = inMicLevelTmp - stt->minLevel;
+ tmpU32 = WEBRTC_SPL_UMUL(31130, (WebRtc_UWord32)(tmp32));
+ stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
+ if (stt->micVol > lastMicVol - 1)
+ {
+ stt->micVol = lastMicVol - 1;
+ }
+ inMicLevelTmp = stt->micVol;
+
+ /* Enable the control mechanism to ensure that our measure,
+ * Rxx160_LP, is in the correct range.
+ */
+ stt->activeSpeech = 0;
+ stt->Rxx16_LPw32Max = 0;
+#ifdef MIC_LEVEL_FEEDBACK
+ //stt->numBlocksMicLvlSat = 0;
+#endif
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
+ stt->fcount, stt->micVol, stt->maxLevel);
+#endif
+ }
+ } else if (stt->Rxx160_LPw32 > stt->upperLimit)
+ {
+ stt->msTooHigh += 2;
+ stt->msTooLow = 0;
+ stt->changeToSlowMode = 0;
+
+ if (stt->msTooHigh > stt->msecSpeechInnerChange)
+ {
+ /* Lower the recording level */
+ stt->msTooHigh = 0;
+ /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
+
+ /* Reduce the max gain to avoid excessive oscillation
+ * (but never drop below the maximum analog level).
+ * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
+ */
+ tmp32 = (15 * stt->maxLevel) + stt->micVol;
+ stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
+ stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
+
+ stt->zeroCtrlMax = stt->micVol;
+
+ /* 0.965 in Q15 */
+ tmp32 = inMicLevelTmp - stt->minLevel;
+ tmpU32 = WEBRTC_SPL_UMUL(31621, (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
+ stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
+ if (stt->micVol > lastMicVol - 1)
+ {
+ stt->micVol = lastMicVol - 1;
+ }
+ inMicLevelTmp = stt->micVol;
+
+#ifdef MIC_LEVEL_FEEDBACK
+ //stt->numBlocksMicLvlSat = 0;
+#endif
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
+ stt->fcount, stt->micVol, stt->maxLevel);
+#endif
+ }
+ } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
+ {
+ stt->msTooHigh = 0;
+ stt->changeToSlowMode = 0;
+ stt->msTooLow += 2;
+
+ if (stt->msTooLow > stt->msecSpeechOuterChange)
+ {
+ /* Raise the recording level */
+ WebRtc_Word16 index, weightFIX;
+ WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
+
+ stt->msTooLow = 0;
+
+ /* Normalize the volume level */
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
+ if (stt->maxInit != stt->minLevel)
+ {
+ volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
+ (stt->maxInit - stt->minLevel));
+ }
+
+ /* Find correct curve */
+ WebRtcAgc_ExpCurve(volNormFIX, &index);
+
+ /* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05 */
+ weightFIX = kOffset1[index]
+ - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope1[index],
+ volNormFIX, 13);
+
+ /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
+
+ tmp32 = inMicLevelTmp - stt->minLevel;
+ tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
+ stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
+ if (stt->micVol < lastMicVol + 2)
+ {
+ stt->micVol = lastMicVol + 2;
+ }
+
+ inMicLevelTmp = stt->micVol;
+
+#ifdef MIC_LEVEL_FEEDBACK
+ /* Count ms in level saturation */
+ //if (stt->micVol > stt->maxAnalog) {
+ if (stt->micVol > 150)
+ {
+ /* mic level is saturated */
+ stt->numBlocksMicLvlSat++;
+ fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
+ }
+#endif
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
+ stt->fcount, stt->micVol);
+#endif
+ }
+ } else if (stt->Rxx160_LPw32 < stt->lowerLimit)
+ {
+ stt->msTooHigh = 0;
+ stt->changeToSlowMode = 0;
+ stt->msTooLow += 2;
+
+ if (stt->msTooLow > stt->msecSpeechInnerChange)
+ {
+ /* Raise the recording level */
+ WebRtc_Word16 index, weightFIX;
+ WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
+
+ stt->msTooLow = 0;
+
+ /* Normalize the volume level */
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
+ if (stt->maxInit != stt->minLevel)
+ {
+ volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
+ (stt->maxInit - stt->minLevel));
+ }
+
+ /* Find correct curve */
+ WebRtcAgc_ExpCurve(volNormFIX, &index);
+
+ /* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1 */
+ weightFIX = kOffset2[index]
+ - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope2[index],
+ volNormFIX, 13);
+
+ /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
+ stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
+
+ tmp32 = inMicLevelTmp - stt->minLevel;
+ tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
+ stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
+ if (stt->micVol < lastMicVol + 1)
+ {
+ stt->micVol = lastMicVol + 1;
+ }
+
+ inMicLevelTmp = stt->micVol;
+
+#ifdef MIC_LEVEL_FEEDBACK
+ /* Count ms in level saturation */
+ //if (stt->micVol > stt->maxAnalog) {
+ if (stt->micVol > 150)
+ {
+ /* mic level is saturated */
+ stt->numBlocksMicLvlSat++;
+ fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
+ }
+#endif
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
+ stt->fcount, stt->micVol);
+#endif
+
+ }
+ } else
+ {
+ /* The signal is inside the desired range which is:
+ * lowerLimit < Rxx160_LP/640 < upperLimit
+ */
+ if (stt->changeToSlowMode > 4000)
+ {
+ stt->msecSpeechInnerChange = 1000;
+ stt->msecSpeechOuterChange = 500;
+ stt->upperLimit = stt->upperPrimaryLimit;
+ stt->lowerLimit = stt->lowerPrimaryLimit;
+ } else
+ {
+ stt->changeToSlowMode += 2; // in milliseconds
+ }
+ stt->msTooLow = 0;
+ stt->msTooHigh = 0;
+
+ stt->micVol = inMicLevelTmp;
+
+ }
+#ifdef MIC_LEVEL_FEEDBACK
+ if (stt->numBlocksMicLvlSat > NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET)
+ {
+ stt->micLvlSat = 1;
+ fprintf(stderr, "target before = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
+ WebRtcAgc_UpdateAgcThresholds(stt);
+ WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]),
+ stt->compressionGaindB, stt->targetLevelDbfs, stt->limiterEnable,
+ stt->analogTarget);
+ stt->numBlocksMicLvlSat = 0;
+ stt->micLvlSat = 0;
+ fprintf(stderr, "target offset = %d\n", stt->targetIdxOffset);
+ fprintf(stderr, "target after = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
+ }
+#endif
+ }
+ }
+
+ /* Ensure gain is not increased in presence of echo or after a mute event
+ * (but allow the zeroCtrl() increase on the frame of a mute detection).
+ */
+ if (echo == 1 || (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs))
+ {
+ if (stt->micVol > lastMicVol)
+ {
+ stt->micVol = lastMicVol;
+ }
+ }
+
+ /* limit the gain */
+ if (stt->micVol > stt->maxLevel)
+ {
+ stt->micVol = stt->maxLevel;
+ } else if (stt->micVol < stt->minOutput)
+ {
+ stt->micVol = stt->minOutput;
+ }
+
+ *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->micVol, stt->scale);
+ if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale))
+ {
+ *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale);
+ }
+
+ return 0;
+}
+
+int WebRtcAgc_Process(void *agcInst, const WebRtc_Word16 *in_near,
+ const WebRtc_Word16 *in_near_H, WebRtc_Word16 samples,
+ WebRtc_Word16 *out, WebRtc_Word16 *out_H, WebRtc_Word32 inMicLevel,
+ WebRtc_Word32 *outMicLevel, WebRtc_Word16 echo,
+ WebRtc_UWord8 *saturationWarning)
+{
+ Agc_t *stt;
+ WebRtc_Word32 inMicLevelTmp;
+ WebRtc_Word16 subFrames, i;
+ WebRtc_UWord8 satWarningTmp = 0;
+
+ stt = (Agc_t *)agcInst;
+
+ //
+ if (stt == NULL)
+ {
+ return -1;
+ }
+ //
+
+
+ if (stt->fs == 8000)
+ {
+ if ((samples != 80) && (samples != 160))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 80;
+ } else if (stt->fs == 16000)
+ {
+ if ((samples != 160) && (samples != 320))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 160;
+ } else if (stt->fs == 32000)
+ {
+ if ((samples != 160) && (samples != 320))
+ {
+#ifdef AGC_DEBUG //test log
+ fprintf(stt->fpt,
+ "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+ subFrames = 160;
+ } else
+ {
+#ifdef AGC_DEBUG// test log
+ fprintf(stt->fpt,
+ "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+
+ /* Check for valid pointers based on sampling rate */
+ if (stt->fs == 32000 && in_near_H == NULL)
+ {
+ return -1;
+ }
+ /* Check for valid pointers for low band */
+ if (in_near == NULL)
+ {
+ return -1;
+ }
+
+ *saturationWarning = 0;
+ //TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS
+ *outMicLevel = inMicLevel;
+ inMicLevelTmp = inMicLevel;
+
+ // TODO(andrew): clearly we don't need input and output pointers...
+ // Change the interface to take a shared input/output.
+ if (in_near != out)
+ {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, in_near, samples * sizeof(WebRtc_Word16));
+ }
+ if (stt->fs == 32000)
+ {
+ if (in_near_H != out_H)
+ {
+ memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16));
+ }
+ }
+
+#ifdef AGC_DEBUG//test log
+ stt->fcount++;
+#endif
+
+ for (i = 0; i < samples; i += subFrames)
+ {
+ if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
+ stt->fs, stt->lowLevelSignal) == -1)
+ {
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+ if ((stt->agcMode < kAgcModeFixedDigital) && ((stt->lowLevelSignal == 0)
+ || (stt->agcMode != kAgcModeAdaptiveDigital)))
+ {
+ if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevelTmp, outMicLevel,
+ stt->vadMic.logRatio, echo, saturationWarning) == -1)
+ {
+ return -1;
+ }
+ }
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
+#endif
+
+ /* update queue */
+ if (stt->inQueue > 1)
+ {
+ memcpy(stt->env[0], stt->env[1], 10 * sizeof(WebRtc_Word32));
+ memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(WebRtc_Word32));
+ }
+
+ if (stt->inQueue > 0)
+ {
+ stt->inQueue--;
+ }
+
+ /* If 20ms frames are used the input mic level must be updated so that
+ * the analog AGC does not think that there has been a manual volume
+ * change. */
+ inMicLevelTmp = *outMicLevel;
+
+ /* Store a positive saturation warning. */
+ if (*saturationWarning == 1)
+ {
+ satWarningTmp = 1;
+ }
+ }
+
+ /* Trigger the saturation warning if displayed by any of the frames. */
+ *saturationWarning = satWarningTmp;
+
+ return 0;
+}
+
+int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig)
+{
+ Agc_t *stt;
+ stt = (Agc_t *)agcInst;
+
+ if (stt == NULL)
+ {
+ return -1;
+ }
+
+ if (stt->initFlag != kInitCheck)
+ {
+ stt->lastError = AGC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (agcConfig.limiterEnable != kAgcFalse && agcConfig.limiterEnable != kAgcTrue)
+ {
+ stt->lastError = AGC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ stt->limiterEnable = agcConfig.limiterEnable;
+ stt->compressionGaindB = agcConfig.compressionGaindB;
+ if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31))
+ {
+ stt->lastError = AGC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ stt->targetLevelDbfs = agcConfig.targetLevelDbfs;
+
+ if (stt->agcMode == kAgcModeFixedDigital)
+ {
+ /* Adjust for different parameter interpretation in FixedDigital mode */
+ stt->compressionGaindB += agcConfig.targetLevelDbfs;
+ }
+
+ /* Update threshold levels for analog adaptation */
+ WebRtcAgc_UpdateAgcThresholds(stt);
+
+ /* Recalculate gain table */
+ if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
+ stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
+ {
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
+#endif
+ return -1;
+ }
+ /* Store the config in a WebRtcAgc_config_t */
+ stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
+ stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
+ stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;
+
+ return 0;
+}
+
+int WebRtcAgc_get_config(void *agcInst, WebRtcAgc_config_t *config)
+{
+ Agc_t *stt;
+ stt = (Agc_t *)agcInst;
+
+ if (stt == NULL)
+ {
+ return -1;
+ }
+
+ if (config == NULL)
+ {
+ stt->lastError = AGC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (stt->initFlag != kInitCheck)
+ {
+ stt->lastError = AGC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ config->limiterEnable = stt->usedConfig.limiterEnable;
+ config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
+ config->compressionGaindB = stt->usedConfig.compressionGaindB;
+
+ return 0;
+}
+
+int WebRtcAgc_Create(void **agcInst)
+{
+ Agc_t *stt;
+ if (agcInst == NULL)
+ {
+ return -1;
+ }
+ stt = (Agc_t *)malloc(sizeof(Agc_t));
+
+ *agcInst = stt;
+ if (stt == NULL)
+ {
+ return -1;
+ }
+
+#ifdef AGC_DEBUG
+ stt->fpt = fopen("./agc_test_log.txt", "wt");
+ stt->agcLog = fopen("./agc_debug_log.txt", "wt");
+ stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
+#endif
+
+ stt->initFlag = 0;
+ stt->lastError = 0;
+
+ return 0;
+}
+
+int WebRtcAgc_Free(void *state)
+{
+ Agc_t *stt;
+
+ stt = (Agc_t *)state;
+#ifdef AGC_DEBUG
+ fclose(stt->fpt);
+ fclose(stt->agcLog);
+ fclose(stt->digitalAgc.logFile);
+#endif
+ free(stt);
+
+ return 0;
+}
+
+/* minLevel - Minimum volume level
+ * maxLevel - Maximum volume level
+ */
+int WebRtcAgc_Init(void *agcInst, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel,
+ WebRtc_Word16 agcMode, WebRtc_UWord32 fs)
+{
+ WebRtc_Word32 max_add, tmp32;
+ WebRtc_Word16 i;
+ int tmpNorm;
+ Agc_t *stt;
+
+ /* typecast state pointer */
+ stt = (Agc_t *)agcInst;
+
+ if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0)
+ {
+ stt->lastError = AGC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ /* Analog AGC variables */
+ stt->envSum = 0;
+
+ /* mode = 0 - Only saturation protection
+ * 1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
+ * 2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
+ * 3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
+ */
+#ifdef AGC_DEBUG//test log
+ stt->fcount = 0;
+ fprintf(stt->fpt, "AGC->Init\n");
+#endif
+ if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
+ {
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
+#endif
+ return -1;
+ }
+ stt->agcMode = agcMode;
+ stt->fs = fs;
+
+ /* initialize input VAD */
+ WebRtcAgc_InitVad(&stt->vadMic);
+
+ /* If the volume range is smaller than 0-256 then
+ * the levels are shifted up to Q8-domain */
+ tmpNorm = WebRtcSpl_NormU32((WebRtc_UWord32)maxLevel);
+ stt->scale = tmpNorm - 23;
+ if (stt->scale < 0)
+ {
+ stt->scale = 0;
+ }
+ // TODO(bjornv): Investigate if we really need to scale up a small range now when we have
+ // a guard against zero-increments. For now, we do not support scale up (scale = 0).
+ stt->scale = 0;
+ maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale);
+ minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale);
+
+ /* Make minLevel and maxLevel static in AdaptiveDigital */
+ if (stt->agcMode == kAgcModeAdaptiveDigital)
+ {
+ minLevel = 0;
+ maxLevel = 255;
+ stt->scale = 0;
+ }
+ /* The maximum supplemental volume range is based on a vague idea
+ * of how much lower the gain will be than the real analog gain. */
+ max_add = WEBRTC_SPL_RSHIFT_W32(maxLevel - minLevel, 2);
+
+ /* Minimum/maximum volume level that can be set */
+ stt->minLevel = minLevel;
+ stt->maxAnalog = maxLevel;
+ stt->maxLevel = maxLevel + max_add;
+ stt->maxInit = stt->maxLevel;
+
+ stt->zeroCtrlMax = stt->maxAnalog;
+
+ /* Initialize micVol parameter */
+ stt->micVol = stt->maxAnalog;
+ if (stt->agcMode == kAgcModeAdaptiveDigital)
+ {
+ stt->micVol = 127; /* Mid-point of mic level */
+ }
+ stt->micRef = stt->micVol;
+ stt->micGainIdx = 127;
+#ifdef MIC_LEVEL_FEEDBACK
+ stt->numBlocksMicLvlSat = 0;
+ stt->micLvlSat = 0;
+#endif
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt,
+ "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
+ stt->minLevel, stt->maxAnalog, stt->maxLevel);
+#endif
+
+ /* Minimum output volume is 4% higher than the available lowest volume level */
+ tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)10, 8);
+ stt->minOutput = (stt->minLevel + tmp32);
+
+ stt->msTooLow = 0;
+ stt->msTooHigh = 0;
+ stt->changeToSlowMode = 0;
+ stt->firstCall = 0;
+ stt->msZero = 0;
+ stt->muteGuardMs = 0;
+ stt->gainTableIdx = 0;
+
+ stt->msecSpeechInnerChange = kMsecSpeechInner;
+ stt->msecSpeechOuterChange = kMsecSpeechOuter;
+
+ stt->activeSpeech = 0;
+ stt->Rxx16_LPw32Max = 0;
+
+ stt->vadThreshold = kNormalVadThreshold;
+ stt->inActive = 0;
+
+ for (i = 0; i < RXX_BUFFER_LEN; i++)
+ {
+ stt->Rxx16_vectorw32[i] = (WebRtc_Word32)1000; /* -54dBm0 */
+ }
+ stt->Rxx160w32 = 125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */
+
+ stt->Rxx16pos = 0;
+ stt->Rxx16_LPw32 = (WebRtc_Word32)16284; /* Q(-4) */
+
+ for (i = 0; i < 5; i++)
+ {
+ stt->Rxx16w32_array[0][i] = 0;
+ }
+ for (i = 0; i < 20; i++)
+ {
+ stt->env[0][i] = 0;
+ }
+ stt->inQueue = 0;
+
+#ifdef MIC_LEVEL_FEEDBACK
+ stt->targetIdxOffset = 0;
+#endif
+
+ WebRtcSpl_MemSetW32(stt->filterState, 0, 8);
+
+ stt->initFlag = kInitCheck;
+ // Default config settings.
+ stt->defaultConfig.limiterEnable = kAgcTrue;
+ stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
+ stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;
+
+ if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1)
+ {
+ stt->lastError = AGC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+ stt->Rxx160_LPw32 = stt->analogTargetLevel; // Initialize rms value
+
+ stt->lowLevelSignal = 0;
+
+ /* Only positive values are allowed that are not too large */
+ if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
+ {
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
+#endif
+ return -1;
+ } else
+ {
+#ifdef AGC_DEBUG//test log
+ fprintf(stt->fpt, "\n");
+#endif
+ return 0;
+ }
+}
diff --git a/trunk/src/modules/audio_processing/agc/analog_agc.h b/trunk/src/modules/audio_processing/agc/analog_agc.h
new file mode 100644
index 0000000..b32ac65
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/analog_agc.h
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
+
+#include "typedefs.h"
+#include "gain_control.h"
+#include "digital_agc.h"
+
+//#define AGC_DEBUG
+//#define MIC_LEVEL_FEEDBACK
+#ifdef AGC_DEBUG
+#include <stdio.h>
+#endif
+
+/* Analog Automatic Gain Control variables:
+ * Constant declarations (inner limits inside which no changes are done)
+ * In the beginning the range is narrower to widen as soon as the measure
+ * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
+ * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
+ * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
+ * The limits are created by running the AGC with a file having the desired
+ * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
+ * by out=10*log10(in/260537279.7); Set the target level to the average level
+ * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
+ * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
+ */
+#define RXX_BUFFER_LEN 10
+
+static const WebRtc_Word16 kMsecSpeechInner = 520;
+static const WebRtc_Word16 kMsecSpeechOuter = 340;
+
+static const WebRtc_Word16 kNormalVadThreshold = 400;
+
+static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
+static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
+
+typedef struct
+{
+ // Configurable parameters/variables
+ WebRtc_UWord32 fs; // Sampling frequency
+ WebRtc_Word16 compressionGaindB; // Fixed gain level in dB
+ WebRtc_Word16 targetLevelDbfs; // Target level in -dBfs of envelope (default -3)
+ WebRtc_Word16 agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig)
+ WebRtc_UWord8 limiterEnable; // Enabling limiter (on/off (default off))
+ WebRtcAgc_config_t defaultConfig;
+ WebRtcAgc_config_t usedConfig;
+
+ // General variables
+ WebRtc_Word16 initFlag;
+ WebRtc_Word16 lastError;
+
+ // Target level parameters
+ // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
+ WebRtc_Word32 analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs
+ WebRtc_Word32 startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs
+ WebRtc_Word32 startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs
+ WebRtc_Word32 upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs
+ WebRtc_Word32 lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs
+ WebRtc_Word32 upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs
+ WebRtc_Word32 lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs
+ WebRtc_UWord16 targetIdx; // Table index for corresponding target level
+#ifdef MIC_LEVEL_FEEDBACK
+ WebRtc_UWord16 targetIdxOffset; // Table index offset for level compensation
+#endif
+ WebRtc_Word16 analogTarget; // Digital reference level in ENV scale
+
+ // Analog AGC specific variables
+ WebRtc_Word32 filterState[8]; // For downsampling wb to nb
+ WebRtc_Word32 upperLimit; // Upper limit for mic energy
+ WebRtc_Word32 lowerLimit; // Lower limit for mic energy
+ WebRtc_Word32 Rxx160w32; // Average energy for one frame
+ WebRtc_Word32 Rxx16_LPw32; // Low pass filtered subframe energies
+ WebRtc_Word32 Rxx160_LPw32; // Low pass filtered frame energies
+ WebRtc_Word32 Rxx16_LPw32Max; // Keeps track of largest energy subframe
+ WebRtc_Word32 Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
+ WebRtc_Word32 Rxx16w32_array[2][5];// Energy values of microphone signal
+ WebRtc_Word32 env[2][10]; // Envelope values of subframes
+
+ WebRtc_Word16 Rxx16pos; // Current position in the Rxx16_vectorw32
+ WebRtc_Word16 envSum; // Filtered scaled envelope in subframes
+ WebRtc_Word16 vadThreshold; // Threshold for VAD decision
+ WebRtc_Word16 inActive; // Inactive time in milliseconds
+ WebRtc_Word16 msTooLow; // Milliseconds of speech at a too low level
+ WebRtc_Word16 msTooHigh; // Milliseconds of speech at a too high level
+ WebRtc_Word16 changeToSlowMode; // Change to slow mode after some time at target
+ WebRtc_Word16 firstCall; // First call to the process-function
+ WebRtc_Word16 msZero; // Milliseconds of zero input
+ WebRtc_Word16 msecSpeechOuterChange;// Min ms of speech between volume changes
+ WebRtc_Word16 msecSpeechInnerChange;// Min ms of speech between volume changes
+ WebRtc_Word16 activeSpeech; // Milliseconds of active speech
+ WebRtc_Word16 muteGuardMs; // Counter to prevent mute action
+ WebRtc_Word16 inQueue; // 10 ms batch indicator
+
+ // Microphone level variables
+ WebRtc_Word32 micRef; // Remember ref. mic level for virtual mic
+ WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table
+ WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly
+ WebRtc_Word32 micVol; // Remember volume between frames
+ WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain
+ WebRtc_Word32 maxAnalog; // Maximum possible analog volume level
+ WebRtc_Word32 maxInit; // Initial value of "max"
+ WebRtc_Word32 minLevel; // Minimum possible volume level
+ WebRtc_Word32 minOutput; // Minimum output volume level
+ WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input
+
+ WebRtc_Word16 scale; // Scale factor for internal volume levels
+#ifdef MIC_LEVEL_FEEDBACK
+ WebRtc_Word16 numBlocksMicLvlSat;
+ WebRtc_UWord8 micLvlSat;
+#endif
+ // Structs for VAD and digital_agc
+ AgcVad_t vadMic;
+ DigitalAgc_t digitalAgc;
+
+#ifdef AGC_DEBUG
+ FILE* fpt;
+ FILE* agcLog;
+ WebRtc_Word32 fcount;
+#endif
+
+ WebRtc_Word16 lowLevelSignal;
+} Agc_t;
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/trunk/src/modules/audio_processing/agc/digital_agc.c b/trunk/src/modules/audio_processing/agc/digital_agc.c
new file mode 100644
index 0000000..3b4b39b
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/digital_agc.c
@@ -0,0 +1,798 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* digital_agc.c
+ *
+ */
+
+#include "digital_agc.h"
+
+#include <assert.h>
+#include <string.h>
+#ifdef AGC_DEBUG
+#include <stdio.h>
+#endif
+
+#include "gain_control.h"
+
+// To generate the gaintable, copy&paste the following lines to a Matlab window:
+// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
+// zeros = 0:31; lvl = 2.^(1-zeros);
+// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
+// B = MaxGain - MinGain;
+// gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
+// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
+// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines):
+// in = 10*log10(lvl); out = 20*log10(gains/65536);
+// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
+// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
+// zoom on;
+
+// Generator table for y=log2(1+e^x) in Q8.
+enum { kGenFuncTableSize = 128 };
+static const WebRtc_UWord16 kGenFuncTable[kGenFuncTableSize] = {
+ 256, 485, 786, 1126, 1484, 1849, 2217, 2586,
+ 2955, 3324, 3693, 4063, 4432, 4801, 5171, 5540,
+ 5909, 6279, 6648, 7017, 7387, 7756, 8125, 8495,
+ 8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449,
+ 11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404,
+ 14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359,
+ 17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313,
+ 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268,
+ 23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222,
+ 26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177,
+ 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
+ 32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086,
+ 35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041,
+ 38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996,
+ 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950,
+ 44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905
+};
+
+static const WebRtc_Word16 kAvgDecayTime = 250; // frames; < 3000
+
+WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
+ WebRtc_Word16 digCompGaindB, // Q0
+ WebRtc_Word16 targetLevelDbfs,// Q0
+ WebRtc_UWord8 limiterEnable,
+ WebRtc_Word16 analogTarget) // Q0
+{
+ // This function generates the compressor gain table used in the fixed digital part.
+ WebRtc_UWord32 tmpU32no1, tmpU32no2, absInLevel, logApprox;
+ WebRtc_Word32 inLevel, limiterLvl;
+ WebRtc_Word32 tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
+ const WebRtc_UWord16 kLog10 = 54426; // log2(10) in Q14
+ const WebRtc_UWord16 kLog10_2 = 49321; // 10*log10(2) in Q14
+ const WebRtc_UWord16 kLogE_1 = 23637; // log2(e) in Q14
+ WebRtc_UWord16 constMaxGain;
+ WebRtc_UWord16 tmpU16, intPart, fracPart;
+ const WebRtc_Word16 kCompRatio = 3;
+ const WebRtc_Word16 kSoftLimiterLeft = 1;
+ WebRtc_Word16 limiterOffset = 0; // Limiter offset
+ WebRtc_Word16 limiterIdx, limiterLvlX;
+ WebRtc_Word16 constLinApprox, zeroGainLvl, maxGain, diffGain;
+ WebRtc_Word16 i, tmp16, tmp16no1;
+ int zeros, zerosScale;
+
+ // Constants
+// kLogE_1 = 23637; // log2(e) in Q14
+// kLog10 = 54426; // log2(10) in Q14
+// kLog10_2 = 49321; // 10*log10(2) in Q14
+
+ // Calculate maximum digital gain and zero gain level
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB - analogTarget, kCompRatio - 1);
+ tmp16no1 = analogTarget - targetLevelDbfs;
+ tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
+ maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(maxGain, kCompRatio);
+ zeroGainLvl = digCompGaindB;
+ zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1),
+ kCompRatio - 1);
+ if ((digCompGaindB <= analogTarget) && (limiterEnable))
+ {
+ zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft);
+ limiterOffset = 0;
+ }
+
+ // Calculate the difference between maximum gain and gain at 0dB0v:
+ // diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio
+ // = (compRatio-1)*digCompGaindB/compRatio
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
+ diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
+ if (diffGain < 0 || diffGain >= kGenFuncTableSize)
+ {
+ assert(0);
+ return -1;
+ }
+
+ // Calculate the limiter level and index:
+ // limiterLvlX = analogTarget - limiterOffset
+ // limiterLvl = targetLevelDbfs + limiterOffset/compRatio
+ limiterLvlX = analogTarget - limiterOffset;
+ limiterIdx = 2
+ + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13),
+ WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1));
+ tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
+ limiterLvl = targetLevelDbfs + tmp16no1;
+
+ // Calculate (through table lookup):
+ // constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
+ constMaxGain = kGenFuncTable[diffGain]; // in Q8
+
+ // Calculate a parameter used to approximate the fractional part of 2^x with a
+ // piecewise linear function in Q14:
+ // constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
+ constLinApprox = 22817; // in Q14
+
+ // Calculate a denominator used in the exponential part to convert from dB to linear scale:
+ // den = 20*constMaxGain (in Q8)
+ den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8
+
+ for (i = 0; i < 32; i++)
+ {
+ // Calculate scaled input level (compressor):
+ // inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
+ tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
+ inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
+
+ // Calculate diffGain-inLevel, to map using the genFuncTable
+ inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14
+
+ // Make calculations on abs(inLevel) and compensate for the sign afterwards.
+ absInLevel = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(inLevel); // Q14
+
+ // LUT with interpolation
+ intPart = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
+ fracPart = (WebRtc_UWord16)(absInLevel & 0x00003FFF); // extract the fractional part
+ tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
+ tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
+ tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)kGenFuncTable[intPart], 14); // Q22
+ logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
+ // Compensate for negative exponent using the relation:
+ // log2(1 + 2^-x) = log2(1 + 2^x) - x
+ if (inLevel < 0)
+ {
+ zeros = WebRtcSpl_NormU32(absInLevel);
+ zerosScale = 0;
+ if (zeros < 15)
+ {
+ // Not enough space for multiplication
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
+ if (zeros < 9)
+ {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
+ zerosScale = 9 - zeros;
+ } else
+ {
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
+ }
+ } else
+ {
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
+ }
+ logApprox = 0;
+ if (tmpU32no2 < tmpU32no1)
+ {
+ logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
+ }
+ }
+ numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
+ numFIX -= WEBRTC_SPL_MUL_32_16((WebRtc_Word32)logApprox, diffGain); // Q14
+
+ // Calculate ratio
+ // Shift |numFIX| as much as possible.
+ // Ensure we avoid wrap-around in |den| as well.
+ if (numFIX > (den >> 8)) // |den| is Q8.
+ {
+ zeros = WebRtcSpl_NormW32(numFIX);
+ } else
+ {
+ zeros = WebRtcSpl_NormW32(den) + 8;
+ }
+ numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
+
+ // Shift den so we end up in Qy1
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
+ if (numFIX < 0)
+ {
+ numFIX -= WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
+ } else
+ {
+ numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
+ }
+ y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
+ if (limiterEnable && (i < limiterIdx))
+ {
+ tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
+ tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
+ y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
+ }
+ if (y32 > 39000)
+ {
+ tmp32 = WEBRTC_SPL_MUL(y32 >> 1, kLog10) + 4096; // in Q27
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 13); // in Q14
+ } else
+ {
+ tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
+ }
+ tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
+
+ // Calculate power
+ if (tmp32 > 0)
+ {
+ intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 14);
+ fracPart = (WebRtc_UWord16)(tmp32 & 0x00003FFF); // in Q14
+ if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
+ {
+ tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
+ tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
+ tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
+ tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
+ } else
+ {
+ tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
+ tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
+ }
+ fracPart = (WebRtc_UWord16)tmp32no2;
+ gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
+ + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
+ } else
+ {
+ gainTable[i] = 0;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *stt, WebRtc_Word16 agcMode)
+{
+
+ if (agcMode == kAgcModeFixedDigital)
+ {
+ // start at minimum to find correct gain faster
+ stt->capacitorSlow = 0;
+ } else
+ {
+ // start out with 0 dB gain
+ stt->capacitorSlow = 134217728; // (WebRtc_Word32)(0.125f * 32768.0f * 32768.0f);
+ }
+ stt->capacitorFast = 0;
+ stt->gain = 65536;
+ stt->gatePrevious = 0;
+ stt->agcMode = agcMode;
+#ifdef AGC_DEBUG
+ stt->frameCounter = 0;
+#endif
+
+ // initialize VADs
+ WebRtcAgc_InitVad(&stt->vadNearend);
+ WebRtcAgc_InitVad(&stt->vadFarend);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_far,
+ WebRtc_Word16 nrSamples)
+{
+ // Check for valid pointer
+ if (&stt->vadFarend == NULL)
+ {
+ return -1;
+ }
+
+ // VAD for far end
+ WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_near,
+ const WebRtc_Word16 *in_near_H, WebRtc_Word16 *out,
+ WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
+ WebRtc_Word16 lowlevelSignal)
+{
+ // array for gains (one value per ms, incl start & end)
+ WebRtc_Word32 gains[11];
+
+ WebRtc_Word32 out_tmp, tmp32;
+ WebRtc_Word32 env[10];
+ WebRtc_Word32 nrg, max_nrg;
+ WebRtc_Word32 cur_level;
+ WebRtc_Word32 gain32, delta;
+ WebRtc_Word16 logratio;
+ WebRtc_Word16 lower_thr, upper_thr;
+ WebRtc_Word16 zeros, zeros_fast, frac;
+ WebRtc_Word16 decay;
+ WebRtc_Word16 gate, gain_adj;
+ WebRtc_Word16 k, n;
+ WebRtc_Word16 L, L2; // samples/subframe
+
+ // determine number of samples per ms
+ if (FS == 8000)
+ {
+ L = 8;
+ L2 = 3;
+ } else if (FS == 16000)
+ {
+ L = 16;
+ L2 = 4;
+ } else if (FS == 32000)
+ {
+ L = 16;
+ L2 = 4;
+ } else
+ {
+ return -1;
+ }
+
+ // TODO(andrew): again, we don't need input and output pointers...
+ if (in_near != out)
+ {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
+ }
+ if (FS == 32000)
+ {
+ if (in_near_H != out_H)
+ {
+ memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
+ }
+ }
+ // VAD for near end
+ logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
+
+ // Account for far end VAD
+ if (stt->vadFarend.counter > 10)
+ {
+ tmp32 = WEBRTC_SPL_MUL_16_16(3, logratio);
+ logratio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 - stt->vadFarend.logRatio, 2);
+ }
+
+ // Determine decay factor depending on VAD
+ // upper_thr = 1.0f;
+ // lower_thr = 0.25f;
+ upper_thr = 1024; // Q10
+ lower_thr = 0; // Q10
+ if (logratio > upper_thr)
+ {
+ // decay = -2^17 / DecayTime; -> -65
+ decay = -65;
+ } else if (logratio < lower_thr)
+ {
+ decay = 0;
+ } else
+ {
+ // decay = (WebRtc_Word16)(((lower_thr - logratio)
+ // * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
+ // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65
+ tmp32 = WEBRTC_SPL_MUL_16_16((lower_thr - logratio), 65);
+ decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
+ }
+
+ // adjust decay factor for long silence (detected as low standard deviation)
+ // This is only done in the adaptive modes
+ if (stt->agcMode != kAgcModeFixedDigital)
+ {
+ if (stt->vadNearend.stdLongTerm < 4000)
+ {
+ decay = 0;
+ } else if (stt->vadNearend.stdLongTerm < 8096)
+ {
+ // decay = (WebRtc_Word16)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12);
+ tmp32 = WEBRTC_SPL_MUL_16_16((stt->vadNearend.stdLongTerm - 4000), decay);
+ decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
+ }
+
+ if (lowlevelSignal != 0)
+ {
+ decay = 0;
+ }
+ }
+#ifdef AGC_DEBUG
+ stt->frameCounter++;
+ fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
+#endif
+ // Find max amplitude per sub frame
+ // iterate over sub frames
+ for (k = 0; k < 10; k++)
+ {
+ // iterate over samples
+ max_nrg = 0;
+ for (n = 0; n < L; n++)
+ {
+ nrg = WEBRTC_SPL_MUL_16_16(out[k * L + n], out[k * L + n]);
+ if (nrg > max_nrg)
+ {
+ max_nrg = nrg;
+ }
+ }
+ env[k] = max_nrg;
+ }
+
+ // Calculate gain per sub frame
+ gains[0] = stt->gain;
+ for (k = 0; k < 10; k++)
+ {
+ // Fast envelope follower
+ // decay time = -131000 / -1000 = 131 (ms)
+ stt->capacitorFast = AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast);
+ if (env[k] > stt->capacitorFast)
+ {
+ stt->capacitorFast = env[k];
+ }
+ // Slow envelope follower
+ if (env[k] > stt->capacitorSlow)
+ {
+ // increase capacitorSlow
+ stt->capacitorSlow
+ = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), stt->capacitorSlow);
+ } else
+ {
+ // decrease capacitorSlow
+ stt->capacitorSlow
+ = AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow);
+ }
+
+ // use maximum of both capacitors as current level
+ if (stt->capacitorFast > stt->capacitorSlow)
+ {
+ cur_level = stt->capacitorFast;
+ } else
+ {
+ cur_level = stt->capacitorSlow;
+ }
+ // Translate signal level into gain, using a piecewise linear approximation
+ // find number of leading zeros
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)cur_level);
+ if (cur_level == 0)
+ {
+ zeros = 31;
+ }
+ tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
+ tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
+ gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
+#ifdef AGC_DEBUG
+ if (k == 0)
+ {
+ fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
+ }
+#endif
+ }
+
+ // Gate processing (lower gain during absence of speech)
+ zeros = WEBRTC_SPL_LSHIFT_W16(zeros, 9) - WEBRTC_SPL_RSHIFT_W16(frac, 3);
+ // find number of leading zeros
+ zeros_fast = WebRtcSpl_NormU32((WebRtc_UWord32)stt->capacitorFast);
+ if (stt->capacitorFast == 0)
+ {
+ zeros_fast = 31;
+ }
+ tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
+ zeros_fast = WEBRTC_SPL_LSHIFT_W16(zeros_fast, 9);
+ zeros_fast -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
+
+ gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
+
+ if (gate < 0)
+ {
+ stt->gatePrevious = 0;
+ } else
+ {
+ tmp32 = WEBRTC_SPL_MUL_16_16(stt->gatePrevious, 7);
+ gate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)gate + tmp32, 3);
+ stt->gatePrevious = gate;
+ }
+ // gate < 0 -> no gate
+ // gate > 2500 -> max gate
+ if (gate > 0)
+ {
+ if (gate < 2500)
+ {
+ gain_adj = WEBRTC_SPL_RSHIFT_W16(2500 - gate, 5);
+ } else
+ {
+ gain_adj = 0;
+ }
+ for (k = 0; k < 10; k++)
+ {
+ if ((gains[k + 1] - stt->gainTable[0]) > 8388608)
+ {
+ // To prevent wraparound
+ tmp32 = WEBRTC_SPL_RSHIFT_W32((gains[k+1] - stt->gainTable[0]), 8);
+ tmp32 = WEBRTC_SPL_MUL(tmp32, (178 + gain_adj));
+ } else
+ {
+ tmp32 = WEBRTC_SPL_MUL((gains[k+1] - stt->gainTable[0]), (178 + gain_adj));
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 8);
+ }
+ gains[k + 1] = stt->gainTable[0] + tmp32;
+ }
+ }
+
+ // Limit gain to avoid overload distortion
+ for (k = 0; k < 10; k++)
+ {
+ // To prevent wrap around
+ zeros = 10;
+ if (gains[k + 1] > 47453132)
+ {
+ zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
+ }
+ gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
+ gain32 = WEBRTC_SPL_MUL(gain32, gain32);
+ // check for overflow
+ while (AGC_MUL32(WEBRTC_SPL_RSHIFT_W32(env[k], 12) + 1, gain32)
+ > WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)32767, 2 * (1 - zeros + 10)))
+ {
+ // multiply by 253/256 ==> -0.1 dB
+ if (gains[k + 1] > 8388607)
+ {
+ // Prevent wrap around
+ gains[k + 1] = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(gains[k+1], 8), 253);
+ } else
+ {
+ gains[k + 1] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(gains[k+1], 253), 8);
+ }
+ gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
+ gain32 = WEBRTC_SPL_MUL(gain32, gain32);
+ }
+ }
+ // gain reductions should be done 1 ms earlier than gain increases
+ for (k = 1; k < 10; k++)
+ {
+ if (gains[k] > gains[k + 1])
+ {
+ gains[k] = gains[k + 1];
+ }
+ }
+ // save start gain for next frame
+ stt->gain = gains[10];
+
+ // Apply gain
+ // handle first sub frame separately
+ delta = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
+ gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
+ // iterate over samples
+ for (n = 0; n < L; n++)
+ {
+ // For lower band
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
+ out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ if (out_tmp > 4095)
+ {
+ out[n] = (WebRtc_Word16)32767;
+ } else if (out_tmp < -4096)
+ {
+ out[n] = (WebRtc_Word16)-32768;
+ } else
+ {
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32, 4));
+ out[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ }
+ // For higher band
+ if (FS == 32000)
+ {
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
+ WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
+ out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ if (out_tmp > 4095)
+ {
+ out_H[n] = (WebRtc_Word16)32767;
+ } else if (out_tmp < -4096)
+ {
+ out_H[n] = (WebRtc_Word16)-32768;
+ } else
+ {
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
+ WEBRTC_SPL_RSHIFT_W32(gain32, 4));
+ out_H[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ }
+ }
+ //
+
+ gain32 += delta;
+ }
+ // iterate over subframes
+ for (k = 1; k < 10; k++)
+ {
+ delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
+ gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
+ // iterate over samples
+ for (n = 0; n < L; n++)
+ {
+ // For lower band
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[k * L + n],
+ WEBRTC_SPL_RSHIFT_W32(gain32, 4));
+ out[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ // For higher band
+ if (FS == 32000)
+ {
+ tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[k * L + n],
+ WEBRTC_SPL_RSHIFT_W32(gain32, 4));
+ out_H[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
+ }
+ gain32 += delta;
+ }
+ }
+
+ return 0;
+}
+
+void WebRtcAgc_InitVad(AgcVad_t *state)
+{
+ WebRtc_Word16 k;
+
+ state->HPstate = 0; // state of high pass filter
+ state->logRatio = 0; // log( P(active) / P(inactive) )
+ // average input level (Q10)
+ state->meanLongTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
+
+ // variance of input level (Q8)
+ state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
+
+ state->stdLongTerm = 0; // standard deviation of input level in dB
+ // short-term average input level (Q10)
+ state->meanShortTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
+
+ // short-term variance of input level (Q8)
+ state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
+
+ state->stdShortTerm = 0; // short-term standard deviation of input level in dB
+ state->counter = 3; // counts updates
+ for (k = 0; k < 8; k++)
+ {
+ // downsampling filter
+ state->downState[k] = 0;
+ }
+}
+
+WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
+ const WebRtc_Word16 *in, // (i) Speech signal
+ WebRtc_Word16 nrSamples) // (i) number of samples
+{
+ WebRtc_Word32 out, nrg, tmp32, tmp32b;
+ WebRtc_UWord16 tmpU16;
+ WebRtc_Word16 k, subfr, tmp16;
+ WebRtc_Word16 buf1[8];
+ WebRtc_Word16 buf2[4];
+ WebRtc_Word16 HPstate;
+ WebRtc_Word16 zeros, dB;
+
+ // process in 10 sub frames of 1 ms (to save on memory)
+ nrg = 0;
+ HPstate = state->HPstate;
+ for (subfr = 0; subfr < 10; subfr++)
+ {
+ // downsample to 4 kHz
+ if (nrSamples == 160)
+ {
+ for (k = 0; k < 8; k++)
+ {
+ tmp32 = (WebRtc_Word32)in[2 * k] + (WebRtc_Word32)in[2 * k + 1];
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
+ buf1[k] = (WebRtc_Word16)tmp32;
+ }
+ in += 16;
+
+ WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState);
+ } else
+ {
+ WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState);
+ in += 8;
+ }
+
+ // high pass filter and compute energy
+ for (k = 0; k < 4; k++)
+ {
+ out = buf2[k] + HPstate;
+ tmp32 = WEBRTC_SPL_MUL(600, out);
+ HPstate = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmp32, 10) - buf2[k]);
+ tmp32 = WEBRTC_SPL_MUL(out, out);
+ nrg += WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
+ }
+ }
+ state->HPstate = HPstate;
+
+ // find number of leading zeros
+ if (!(0xFFFF0000 & nrg))
+ {
+ zeros = 16;
+ } else
+ {
+ zeros = 0;
+ }
+ if (!(0xFF000000 & (nrg << zeros)))
+ {
+ zeros += 8;
+ }
+ if (!(0xF0000000 & (nrg << zeros)))
+ {
+ zeros += 4;
+ }
+ if (!(0xC0000000 & (nrg << zeros)))
+ {
+ zeros += 2;
+ }
+ if (!(0x80000000 & (nrg << zeros)))
+ {
+ zeros += 1;
+ }
+
+ // energy level (range {-32..30}) (Q10)
+ dB = WEBRTC_SPL_LSHIFT_W16(15 - zeros, 11);
+
+ // Update statistics
+
+ if (state->counter < kAvgDecayTime)
+ {
+ // decay time = AvgDecTime * 10 ms
+ state->counter++;
+ }
+
+ // update short-term estimate of mean energy level (Q10)
+ tmp32 = (WEBRTC_SPL_MUL_16_16(state->meanShortTerm, 15) + (WebRtc_Word32)dB);
+ state->meanShortTerm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
+
+ // update short-term estimate of variance in energy level (Q8)
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
+ tmp32 += WEBRTC_SPL_MUL(state->varianceShortTerm, 15);
+ state->varianceShortTerm = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
+
+ // update short-term estimate of standard deviation in energy level (Q10)
+ tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
+ state->stdShortTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
+
+ // update long-term estimate of mean energy level (Q10)
+ tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->counter) + (WebRtc_Word32)dB;
+ state->meanLongTerm = WebRtcSpl_DivW32W16ResW16(tmp32,
+ WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
+
+ // update long-term estimate of variance in energy level (Q8)
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
+ tmp32 += WEBRTC_SPL_MUL(state->varianceLongTerm, state->counter);
+ state->varianceLongTerm = WebRtcSpl_DivW32W16(tmp32,
+ WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
+
+ // update long-term estimate of standard deviation in energy level (Q10)
+ tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
+ tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
+ state->stdLongTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
+
+ // update voice activity measure (Q10)
+ tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
+ tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm));
+ tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
+ tmpU16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)13, 12);
+ tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
+ tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10);
+
+ state->logRatio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
+
+ // limit
+ if (state->logRatio > 2048)
+ {
+ state->logRatio = 2048;
+ }
+ if (state->logRatio < -2048)
+ {
+ state->logRatio = -2048;
+ }
+
+ return state->logRatio; // Q10
+}
diff --git a/trunk/src/modules/audio_processing/agc/digital_agc.h b/trunk/src/modules/audio_processing/agc/digital_agc.h
new file mode 100644
index 0000000..240b220
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/digital_agc.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
+
+#ifdef AGC_DEBUG
+#include <stdio.h>
+#endif
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+// the 32 most significant bits of A(19) * B(26) >> 13
+#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
+// C + the 32 most significant bits of A * B
+#define AGC_SCALEDIFF32(A, B, C) ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
+
+typedef struct
+{
+ WebRtc_Word32 downState[8];
+ WebRtc_Word16 HPstate;
+ WebRtc_Word16 counter;
+ WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
+ WebRtc_Word16 meanLongTerm; // Q10
+ WebRtc_Word32 varianceLongTerm; // Q8
+ WebRtc_Word16 stdLongTerm; // Q10
+ WebRtc_Word16 meanShortTerm; // Q10
+ WebRtc_Word32 varianceShortTerm; // Q8
+ WebRtc_Word16 stdShortTerm; // Q10
+} AgcVad_t; // total = 54 bytes
+
+typedef struct
+{
+ WebRtc_Word32 capacitorSlow;
+ WebRtc_Word32 capacitorFast;
+ WebRtc_Word32 gain;
+ WebRtc_Word32 gainTable[32];
+ WebRtc_Word16 gatePrevious;
+ WebRtc_Word16 agcMode;
+ AgcVad_t vadNearend;
+ AgcVad_t vadFarend;
+#ifdef AGC_DEBUG
+ FILE* logFile;
+ int frameCounter;
+#endif
+} DigitalAgc_t;
+
+WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
+
+WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
+ const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
+ WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
+ WebRtc_Word16 lowLevelSignal);
+
+WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
+ WebRtc_Word16 nrSamples);
+
+void WebRtcAgc_InitVad(AgcVad_t *vadInst);
+
+WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
+ const WebRtc_Word16 *in, // (i) Speech signal
+ WebRtc_Word16 nrSamples); // (i) number of samples
+
+WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
+ WebRtc_Word16 compressionGaindB, // Q0 (in dB)
+ WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
+ WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/trunk/src/modules/audio_processing/agc/include/gain_control.h b/trunk/src/modules/audio_processing/agc/include/gain_control.h
new file mode 100644
index 0000000..8af5c71
--- /dev/null
+++ b/trunk/src/modules/audio_processing/agc/include/gain_control.h
@@ -0,0 +1,259 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
+
+#include "typedefs.h"
+
+// Errors
+#define AGC_UNSPECIFIED_ERROR 18000
+#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
+#define AGC_UNINITIALIZED_ERROR 18002
+#define AGC_NULL_POINTER_ERROR 18003
+#define AGC_BAD_PARAMETER_ERROR 18004
+
+// Warnings
+#define AGC_BAD_PARAMETER_WARNING 18050
+
+enum
+{
+ kAgcModeUnchanged,
+ kAgcModeAdaptiveAnalog,
+ kAgcModeAdaptiveDigital,
+ kAgcModeFixedDigital
+};
+
+enum
+{
+ kAgcFalse = 0,
+ kAgcTrue
+};
+
+typedef struct
+{
+ WebRtc_Word16 targetLevelDbfs; // default 3 (-3 dBOv)
+ WebRtc_Word16 compressionGaindB; // default 9 dB
+ WebRtc_UWord8 limiterEnable; // default kAgcTrue (on)
+} WebRtcAgc_config_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*
+ * This function processes a 10/20ms frame of far-end speech to determine
+ * if there is active speech. Far-end speech length can be either 10ms or
+ * 20ms. The length of the input speech vector must be given in samples
+ * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
+ *
+ * Input:
+ * - agcInst : AGC instance.
+ * - inFar : Far-end input speech vector (10 or 20ms)
+ * - samples : Number of samples in input vector
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_AddFarend(void* agcInst,
+ const WebRtc_Word16* inFar,
+ WebRtc_Word16 samples);
+
+/*
+ * This function processes a 10/20ms frame of microphone speech to determine
+ * if there is active speech. Microphone speech length can be either 10ms or
+ * 20ms. The length of the input speech vector must be given in samples
+ * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
+ * input levels, the input signal is increased in level by multiplying and
+ * overwriting the samples in inMic[].
+ *
+ * This function should be called before any further processing of the
+ * near-end microphone signal.
+ *
+ * Input:
+ * - agcInst : AGC instance.
+ * - inMic : Microphone input speech vector (10 or 20 ms) for
+ * L band
+ * - inMic_H : Microphone input speech vector (10 or 20 ms) for
+ * H band
+ * - samples : Number of samples in input vector
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_AddMic(void* agcInst,
+ WebRtc_Word16* inMic,
+ WebRtc_Word16* inMic_H,
+ WebRtc_Word16 samples);
+
+/*
+ * This function replaces the analog microphone with a virtual one.
+ * It is a digital gain applied to the input signal and is used in the
+ * agcAdaptiveDigital mode where no microphone level is adjustable.
+ * Microphone speech length can be either 10ms or 20ms. The length of the
+ * input speech vector must be given in samples (80/160 when FS=8000, and
+ * 160/320 when FS=16000 or FS=32000).
+ *
+ * Input:
+ * - agcInst : AGC instance.
+ * - inMic : Microphone input speech vector for (10 or 20 ms)
+ * L band
+ * - inMic_H : Microphone input speech vector for (10 or 20 ms)
+ * H band
+ * - samples : Number of samples in input vector
+ * - micLevelIn : Input level of microphone (static)
+ *
+ * Output:
+ * - inMic : Microphone output after processing (L band)
+ * - inMic_H : Microphone output after processing (H band)
+ * - micLevelOut : Adjusted microphone level after processing
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_VirtualMic(void* agcInst,
+ WebRtc_Word16* inMic,
+ WebRtc_Word16* inMic_H,
+ WebRtc_Word16 samples,
+ WebRtc_Word32 micLevelIn,
+ WebRtc_Word32* micLevelOut);
+
+/*
+ * This function processes a 10/20ms frame and adjusts (normalizes) the gain
+ * both analog and digitally. The gain adjustments are done only during
+ * active periods of speech. The input speech length can be either 10ms or
+ * 20ms and the output is of the same length. The length of the speech
+ * vectors must be given in samples (80/160 when FS=8000, and 160/320 when
+ * FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
+ * not adjust upward in the presence of echo.
+ *
+ * This function should be called after processing the near-end microphone
+ * signal, in any case after any echo cancellation.
+ *
+ * Input:
+ * - agcInst : AGC instance
+ * - inNear : Near-end input speech vector (10 or 20 ms) for
+ * L band
+ * - inNear_H : Near-end input speech vector (10 or 20 ms) for
+ * H band
+ * - samples : Number of samples in input/output vector
+ * - inMicLevel : Current microphone volume level
+ * - echo : Set to 0 if the signal passed to add_mic is
+ * almost certainly free of echo; otherwise set
+ * to 1. If you have no information regarding echo
+ * set to 0.
+ *
+ * Output:
+ * - outMicLevel : Adjusted microphone volume level
+ * - out : Gain-adjusted near-end speech vector (L band)
+ * : May be the same vector as the input.
+ * - out_H : Gain-adjusted near-end speech vector (H band)
+ * - saturationWarning : A returned value of 1 indicates a saturation event
+ * has occurred and the volume cannot be further
+ * reduced. Otherwise will be set to 0.
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_Process(void* agcInst,
+ const WebRtc_Word16* inNear,
+ const WebRtc_Word16* inNear_H,
+ WebRtc_Word16 samples,
+ WebRtc_Word16* out,
+ WebRtc_Word16* out_H,
+ WebRtc_Word32 inMicLevel,
+ WebRtc_Word32* outMicLevel,
+ WebRtc_Word16 echo,
+ WebRtc_UWord8* saturationWarning);
+
+/*
+ * This function sets the config parameters (targetLevelDbfs,
+ * compressionGaindB and limiterEnable).
+ *
+ * Input:
+ * - agcInst : AGC instance
+ * - config : config struct
+ *
+ * Output:
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);
+
+/*
+ * This function returns the config parameters (targetLevelDbfs,
+ * compressionGaindB and limiterEnable).
+ *
+ * Input:
+ * - agcInst : AGC instance
+ *
+ * Output:
+ * - config : config struct
+ *
+ * Return value:
+ * : 0 - Normal operation.
+ * : -1 - Error
+ */
+int WebRtcAgc_get_config(void* agcInst, WebRtcAgc_config_t* config);
+
+/*
+ * This function creates an AGC instance, which will contain the state
+ * information for one (duplex) channel.
+ *
+ * Return value : AGC instance if successful
+ * : 0 (i.e., a NULL pointer) if unsuccessful
+ */
+int WebRtcAgc_Create(void **agcInst);
+
+/*
+ * This function frees the AGC instance created at the beginning.
+ *
+ * Input:
+ * - agcInst : AGC instance.
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcAgc_Free(void *agcInst);
+
+/*
+ * This function initializes an AGC instance.
+ *
+ * Input:
+ * - agcInst : AGC instance.
+ * - minLevel : Minimum possible mic level
+ * - maxLevel : Maximum possible mic level
+ * - agcMode : 0 - Unchanged
+ * : 1 - Adaptive Analog Automatic Gain Control -3dBOv
+ * : 2 - Adaptive Digital Automatic Gain Control -3dBOv
+ * : 3 - Fixed Digital Gain 0dB
+ * - fs : Sampling frequency
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcAgc_Init(void *agcInst,
+ WebRtc_Word32 minLevel,
+ WebRtc_Word32 maxLevel,
+ WebRtc_Word16 agcMode,
+ WebRtc_UWord32 fs);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
diff --git a/trunk/src/modules/audio_processing/apm_tests.gypi b/trunk/src/modules/audio_processing/apm_tests.gypi
new file mode 100644
index 0000000..c00a6d2
--- /dev/null
+++ b/trunk/src/modules/audio_processing/apm_tests.gypi
@@ -0,0 +1,75 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audioproc_unittest',
+ 'type': 'executable',
+ 'conditions': [
+ ['prefer_fixed_point==1', {
+ 'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
+ }, {
+ 'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
+ }],
+ ['enable_protobuf==1', {
+ 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
+ }],
+ ],
+ 'dependencies': [
+ 'audio_processing',
+ 'audioproc_unittest_proto',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [ 'test/unit_test.cc', ],
+ },
+ {
+ 'target_name': 'audioproc_unittest_proto',
+ 'type': 'static_library',
+ 'sources': [ 'test/unittest.proto', ],
+ 'variables': {
+ 'proto_in_dir': 'test',
+ # Workaround to protect against gyp's pathname relativization when this
+ # file is included by modules.gyp.
+ 'proto_out_protected': 'webrtc/audio_processing',
+ 'proto_out_dir': '<(proto_out_protected)',
+ },
+ 'includes': [ '../../build/protoc.gypi', ],
+ },
+ ],
+ 'conditions': [
+ ['enable_protobuf==1', {
+ 'targets': [
+ {
+ 'target_name': 'audioproc',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_processing',
+ 'audioproc_debug_proto',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [ 'test/process_test.cc', ],
+ },
+ {
+ 'target_name': 'unpack_aecdump',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audioproc_debug_proto',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
+ ],
+ 'sources': [ 'test/unpack.cc', ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/audio_buffer.cc b/trunk/src/modules/audio_processing/audio_buffer.cc
new file mode 100644
index 0000000..a7fb04d
--- /dev/null
+++ b/trunk/src/modules/audio_processing/audio_buffer.cc
@@ -0,0 +1,306 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_buffer.h"
+
+#include "signal_processing_library.h"
+
+namespace webrtc {
+namespace {
+
+enum {
+ kSamplesPer8kHzChannel = 80,
+ kSamplesPer16kHzChannel = 160,
+ kSamplesPer32kHzChannel = 320
+};
+
+void StereoToMono(const int16_t* left, const int16_t* right,
+ int16_t* out, int samples_per_channel) {
+ assert(left != NULL && right != NULL && out != NULL);
+ for (int i = 0; i < samples_per_channel; i++) {
+ int32_t data32 = (static_cast<int32_t>(left[i]) +
+ static_cast<int32_t>(right[i])) >> 1;
+
+ out[i] = WebRtcSpl_SatW32ToW16(data32);
+ }
+}
+} // namespace
+
+struct AudioChannel {
+ AudioChannel() {
+ memset(data, 0, sizeof(data));
+ }
+
+ int16_t data[kSamplesPer32kHzChannel];
+};
+
+struct SplitAudioChannel {
+ SplitAudioChannel() {
+ memset(low_pass_data, 0, sizeof(low_pass_data));
+ memset(high_pass_data, 0, sizeof(high_pass_data));
+ memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
+ memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
+ memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
+ memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
+ }
+
+ int16_t low_pass_data[kSamplesPer16kHzChannel];
+ int16_t high_pass_data[kSamplesPer16kHzChannel];
+
+ WebRtc_Word32 analysis_filter_state1[6];
+ WebRtc_Word32 analysis_filter_state2[6];
+ WebRtc_Word32 synthesis_filter_state1[6];
+ WebRtc_Word32 synthesis_filter_state2[6];
+};
+
+// TODO(andrew): check range of input parameters?
+AudioBuffer::AudioBuffer(int max_num_channels,
+ int samples_per_channel)
+ : max_num_channels_(max_num_channels),
+ num_channels_(0),
+ num_mixed_channels_(0),
+ num_mixed_low_pass_channels_(0),
+ data_was_mixed_(false),
+ samples_per_channel_(samples_per_channel),
+ samples_per_split_channel_(samples_per_channel),
+ reference_copied_(false),
+ activity_(AudioFrame::kVadUnknown),
+ is_muted_(false),
+ data_(NULL),
+ channels_(NULL),
+ split_channels_(NULL),
+ mixed_channels_(NULL),
+ mixed_low_pass_channels_(NULL),
+ low_pass_reference_channels_(NULL) {
+ if (max_num_channels_ > 1) {
+ channels_.reset(new AudioChannel[max_num_channels_]);
+ mixed_channels_.reset(new AudioChannel[max_num_channels_]);
+ mixed_low_pass_channels_.reset(new AudioChannel[max_num_channels_]);
+ }
+ low_pass_reference_channels_.reset(new AudioChannel[max_num_channels_]);
+
+ if (samples_per_channel_ == kSamplesPer32kHzChannel) {
+ split_channels_.reset(new SplitAudioChannel[max_num_channels_]);
+ samples_per_split_channel_ = kSamplesPer16kHzChannel;
+ }
+}
+
+AudioBuffer::~AudioBuffer() {}
+
+int16_t* AudioBuffer::data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (data_ != NULL) {
+ return data_;
+ }
+
+ return channels_[channel].data;
+}
+
+int16_t* AudioBuffer::low_pass_split_data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (split_channels_.get() == NULL) {
+ return data(channel);
+ }
+
+ return split_channels_[channel].low_pass_data;
+}
+
+int16_t* AudioBuffer::high_pass_split_data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (split_channels_.get() == NULL) {
+ return NULL;
+ }
+
+ return split_channels_[channel].high_pass_data;
+}
+
+int16_t* AudioBuffer::mixed_data(int channel) const {
+ assert(channel >= 0 && channel < num_mixed_channels_);
+
+ return mixed_channels_[channel].data;
+}
+
+int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
+ assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
+
+ return mixed_low_pass_channels_[channel].data;
+}
+
+int16_t* AudioBuffer::low_pass_reference(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (!reference_copied_) {
+ return NULL;
+ }
+
+ return low_pass_reference_channels_[channel].data;
+}
+
+WebRtc_Word32* AudioBuffer::analysis_filter_state1(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].analysis_filter_state1;
+}
+
+WebRtc_Word32* AudioBuffer::analysis_filter_state2(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].analysis_filter_state2;
+}
+
+WebRtc_Word32* AudioBuffer::synthesis_filter_state1(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].synthesis_filter_state1;
+}
+
+WebRtc_Word32* AudioBuffer::synthesis_filter_state2(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].synthesis_filter_state2;
+}
+
+void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
+ activity_ = activity;
+}
+
+AudioFrame::VADActivity AudioBuffer::activity() const {
+ return activity_;
+}
+
+bool AudioBuffer::is_muted() const {
+ return is_muted_;
+}
+
+int AudioBuffer::num_channels() const {
+ return num_channels_;
+}
+
+int AudioBuffer::samples_per_channel() const {
+ return samples_per_channel_;
+}
+
+int AudioBuffer::samples_per_split_channel() const {
+ return samples_per_split_channel_;
+}
+
+// TODO(andrew): Do deinterleaving and mixing in one step?
+void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
+ assert(frame->_audioChannel <= max_num_channels_);
+ assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
+
+ num_channels_ = frame->_audioChannel;
+ data_was_mixed_ = false;
+ num_mixed_channels_ = 0;
+ num_mixed_low_pass_channels_ = 0;
+ reference_copied_ = false;
+ activity_ = frame->_vadActivity;
+ is_muted_ = false;
+ if (frame->_energy == 0) {
+ is_muted_ = true;
+ }
+
+ if (num_channels_ == 1) {
+ // We can get away with a pointer assignment in this case.
+ data_ = frame->_payloadData;
+ return;
+ }
+
+ int16_t* interleaved = frame->_payloadData;
+ for (int i = 0; i < num_channels_; i++) {
+ int16_t* deinterleaved = channels_[i].data;
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel_; j++) {
+ deinterleaved[j] = interleaved[interleaved_idx];
+ interleaved_idx += num_channels_;
+ }
+ }
+}
+
+void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
+ assert(frame->_audioChannel == num_channels_);
+ assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
+ frame->_vadActivity = activity_;
+
+ if (!data_changed) {
+ return;
+ }
+
+ if (num_channels_ == 1) {
+ if (data_was_mixed_) {
+ memcpy(frame->_payloadData,
+ channels_[0].data,
+ sizeof(int16_t) * samples_per_channel_);
+ } else {
+ // These should point to the same buffer in this case.
+ assert(data_ == frame->_payloadData);
+ }
+
+ return;
+ }
+
+ int16_t* interleaved = frame->_payloadData;
+ for (int i = 0; i < num_channels_; i++) {
+ int16_t* deinterleaved = channels_[i].data;
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel_; j++) {
+ interleaved[interleaved_idx] = deinterleaved[j];
+ interleaved_idx += num_channels_;
+ }
+ }
+}
+
+// TODO(andrew): would be good to support the no-mix case with pointer
+// assignment.
+// TODO(andrew): handle mixing to multiple channels?
+void AudioBuffer::Mix(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(channels_[0].data,
+ channels_[1].data,
+ channels_[0].data,
+ samples_per_channel_);
+
+ num_channels_ = num_mixed_channels;
+ data_was_mixed_ = true;
+}
+
+void AudioBuffer::CopyAndMix(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(channels_[0].data,
+ channels_[1].data,
+ mixed_channels_[0].data,
+ samples_per_channel_);
+
+ num_mixed_channels_ = num_mixed_channels;
+}
+
+void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(low_pass_split_data(0),
+ low_pass_split_data(1),
+ mixed_low_pass_channels_[0].data,
+ samples_per_split_channel_);
+
+ num_mixed_low_pass_channels_ = num_mixed_channels;
+}
+
+void AudioBuffer::CopyLowPassToReference() {
+ reference_copied_ = true;
+ for (int i = 0; i < num_channels_; i++) {
+ memcpy(low_pass_reference_channels_[i].data,
+ low_pass_split_data(i),
+ sizeof(int16_t) * samples_per_split_channel_);
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/audio_buffer.h b/trunk/src/modules/audio_processing/audio_buffer.h
new file mode 100644
index 0000000..87d6972
--- /dev/null
+++ b/trunk/src/modules/audio_processing/audio_buffer.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
+
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+struct AudioChannel;
+struct SplitAudioChannel;
+
+class AudioBuffer {
+ public:
+ AudioBuffer(int max_num_channels, int samples_per_channel);
+ virtual ~AudioBuffer();
+
+ int num_channels() const;
+ int samples_per_channel() const;
+ int samples_per_split_channel() const;
+
+ int16_t* data(int channel) const;
+ int16_t* low_pass_split_data(int channel) const;
+ int16_t* high_pass_split_data(int channel) const;
+ int16_t* mixed_data(int channel) const;
+ int16_t* mixed_low_pass_data(int channel) const;
+ int16_t* low_pass_reference(int channel) const;
+
+ int32_t* analysis_filter_state1(int channel) const;
+ int32_t* analysis_filter_state2(int channel) const;
+ int32_t* synthesis_filter_state1(int channel) const;
+ int32_t* synthesis_filter_state2(int channel) const;
+
+ void set_activity(AudioFrame::VADActivity activity);
+ AudioFrame::VADActivity activity() const;
+
+ bool is_muted() const;
+
+ void DeinterleaveFrom(AudioFrame* audioFrame);
+ void InterleaveTo(AudioFrame* audioFrame) const;
+ // If |data_changed| is false, only the non-audio data members will be copied
+ // to |frame|.
+ void InterleaveTo(AudioFrame* frame, bool data_changed) const;
+ void Mix(int num_mixed_channels);
+ void CopyAndMix(int num_mixed_channels);
+ void CopyAndMixLowPass(int num_mixed_channels);
+ void CopyLowPassToReference();
+
+ private:
+ const int max_num_channels_;
+ int num_channels_;
+ int num_mixed_channels_;
+ int num_mixed_low_pass_channels_;
+ // Whether the original data was replaced with mixed data.
+ bool data_was_mixed_;
+ const int samples_per_channel_;
+ int samples_per_split_channel_;
+ bool reference_copied_;
+ AudioFrame::VADActivity activity_;
+ bool is_muted_;
+
+ int16_t* data_;
+ scoped_array<AudioChannel> channels_;
+ scoped_array<SplitAudioChannel> split_channels_;
+ scoped_array<AudioChannel> mixed_channels_;
+ // TODO(andrew): improve this, we don't need the full 32 kHz space here.
+ scoped_array<AudioChannel> mixed_low_pass_channels_;
+ scoped_array<AudioChannel> low_pass_reference_channels_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
diff --git a/trunk/src/modules/audio_processing/audio_processing.gypi b/trunk/src/modules/audio_processing/audio_processing.gypi
new file mode 100644
index 0000000..00078c4
--- /dev/null
+++ b/trunk/src/modules/audio_processing/audio_processing.gypi
@@ -0,0 +1,91 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audio_processing',
+ 'type': '<(library)',
+ 'conditions': [
+ ['prefer_fixed_point==1', {
+ 'dependencies': [ 'ns_fix' ],
+ 'defines': [ 'WEBRTC_NS_FIXED' ],
+ }, {
+ 'dependencies': [ 'ns' ],
+ 'defines': [ 'WEBRTC_NS_FLOAT' ],
+ }],
+ ['enable_protobuf==1', {
+ 'dependencies': [ 'audioproc_debug_proto' ],
+ 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
+ }],
+ ],
+ 'dependencies': [
+ 'aec',
+ 'aecm',
+ 'agc',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/common_audio/common_audio.gyp:vad',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ 'include',
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ '../interface',
+ ],
+ },
+ 'sources': [
+ 'include/audio_processing.h',
+ 'audio_buffer.cc',
+ 'audio_buffer.h',
+ 'audio_processing_impl.cc',
+ 'audio_processing_impl.h',
+ 'echo_cancellation_impl.cc',
+ 'echo_cancellation_impl.h',
+ 'echo_control_mobile_impl.cc',
+ 'echo_control_mobile_impl.h',
+ 'gain_control_impl.cc',
+ 'gain_control_impl.h',
+ 'high_pass_filter_impl.cc',
+ 'high_pass_filter_impl.h',
+ 'level_estimator_impl.cc',
+ 'level_estimator_impl.h',
+ 'noise_suppression_impl.cc',
+ 'noise_suppression_impl.h',
+ 'splitting_filter.cc',
+ 'splitting_filter.h',
+ 'processing_component.cc',
+ 'processing_component.h',
+ 'voice_detection_impl.cc',
+ 'voice_detection_impl.h',
+ ],
+ },
+ ],
+ 'conditions': [
+ ['enable_protobuf==1', {
+ 'targets': [
+ {
+ 'target_name': 'audioproc_debug_proto',
+ 'type': 'static_library',
+ 'sources': [ 'debug.proto', ],
+ 'variables': {
+ 'proto_in_dir': '.',
+ # Workaround to protect against gyp's pathname relativization when
+ # this file is included by modules.gyp.
+ 'proto_out_protected': 'webrtc/audio_processing',
+ 'proto_out_dir': '<(proto_out_protected)',
+ },
+ 'includes': [ '../../build/protoc.gypi', ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/audio_processing_impl.cc b/trunk/src/modules/audio_processing/audio_processing_impl.cc
new file mode 100644
index 0000000..35d266d
--- /dev/null
+++ b/trunk/src/modules/audio_processing/audio_processing_impl.cc
@@ -0,0 +1,643 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_processing_impl.h"
+
+#include <assert.h>
+
+#include "audio_buffer.h"
+#include "critical_section_wrapper.h"
+#include "echo_cancellation_impl.h"
+#include "echo_control_mobile_impl.h"
+#include "file_wrapper.h"
+#include "high_pass_filter_impl.h"
+#include "gain_control_impl.h"
+#include "level_estimator_impl.h"
+#include "module_common_types.h"
+#include "noise_suppression_impl.h"
+#include "processing_component.h"
+#include "splitting_filter.h"
+#include "voice_detection_impl.h"
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+namespace webrtc {
+AudioProcessing* AudioProcessing::Create(int id) {
+
+ AudioProcessingImpl* apm = new AudioProcessingImpl(id);
+ if (apm->Initialize() != kNoError) {
+ delete apm;
+ apm = NULL;
+ }
+
+ return apm;
+}
+
+void AudioProcessing::Destroy(AudioProcessing* apm) {
+ delete static_cast<AudioProcessingImpl*>(apm);
+}
+
+AudioProcessingImpl::AudioProcessingImpl(int id)
+ : id_(id),
+ echo_cancellation_(NULL),
+ echo_control_mobile_(NULL),
+ gain_control_(NULL),
+ high_pass_filter_(NULL),
+ level_estimator_(NULL),
+ noise_suppression_(NULL),
+ voice_detection_(NULL),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ render_audio_(NULL),
+ capture_audio_(NULL),
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ debug_file_(FileWrapper::Create()),
+ event_msg_(new audioproc::Event()),
+#endif
+ sample_rate_hz_(kSampleRate16kHz),
+ split_sample_rate_hz_(kSampleRate16kHz),
+ samples_per_channel_(sample_rate_hz_ / 100),
+ stream_delay_ms_(0),
+ was_stream_delay_set_(false),
+ num_reverse_channels_(1),
+ num_input_channels_(1),
+ num_output_channels_(1) {
+
+ echo_cancellation_ = new EchoCancellationImpl(this);
+ component_list_.push_back(echo_cancellation_);
+
+ echo_control_mobile_ = new EchoControlMobileImpl(this);
+ component_list_.push_back(echo_control_mobile_);
+
+ gain_control_ = new GainControlImpl(this);
+ component_list_.push_back(gain_control_);
+
+ high_pass_filter_ = new HighPassFilterImpl(this);
+ component_list_.push_back(high_pass_filter_);
+
+ level_estimator_ = new LevelEstimatorImpl(this);
+ component_list_.push_back(level_estimator_);
+
+ noise_suppression_ = new NoiseSuppressionImpl(this);
+ component_list_.push_back(noise_suppression_);
+
+ voice_detection_ = new VoiceDetectionImpl(this);
+ component_list_.push_back(voice_detection_);
+}
+
+AudioProcessingImpl::~AudioProcessingImpl() {
+ while (!component_list_.empty()) {
+ ProcessingComponent* component = component_list_.front();
+ component->Destroy();
+ delete component;
+ component_list_.pop_front();
+ }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ debug_file_->CloseFile();
+ }
+#endif
+
+ delete crit_;
+ crit_ = NULL;
+
+ if (render_audio_) {
+ delete render_audio_;
+ render_audio_ = NULL;
+ }
+
+ if (capture_audio_) {
+ delete capture_audio_;
+ capture_audio_ = NULL;
+ }
+}
+
+CriticalSectionWrapper* AudioProcessingImpl::crit() const {
+ return crit_;
+}
+
+int AudioProcessingImpl::split_sample_rate_hz() const {
+ return split_sample_rate_hz_;
+}
+
+int AudioProcessingImpl::Initialize() {
+ CriticalSectionScoped crit_scoped(crit_);
+ return InitializeLocked();
+}
+
+int AudioProcessingImpl::InitializeLocked() {
+ if (render_audio_ != NULL) {
+ delete render_audio_;
+ render_audio_ = NULL;
+ }
+
+ if (capture_audio_ != NULL) {
+ delete capture_audio_;
+ capture_audio_ = NULL;
+ }
+
+ render_audio_ = new AudioBuffer(num_reverse_channels_,
+ samples_per_channel_);
+ capture_audio_ = new AudioBuffer(num_input_channels_,
+ samples_per_channel_);
+
+ was_stream_delay_set_ = false;
+
+ // Initialize all components.
+ std::list<ProcessingComponent*>::iterator it;
+ for (it = component_list_.begin(); it != component_list_.end(); it++) {
+ int err = (*it)->Initialize();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+#endif
+
+ return kNoError;
+}
+
+int AudioProcessingImpl::set_sample_rate_hz(int rate) {
+ CriticalSectionScoped crit_scoped(crit_);
+ if (rate != kSampleRate8kHz &&
+ rate != kSampleRate16kHz &&
+ rate != kSampleRate32kHz) {
+ return kBadParameterError;
+ }
+
+ sample_rate_hz_ = rate;
+ samples_per_channel_ = rate / 100;
+
+ if (sample_rate_hz_ == kSampleRate32kHz) {
+ split_sample_rate_hz_ = kSampleRate16kHz;
+ } else {
+ split_sample_rate_hz_ = sample_rate_hz_;
+ }
+
+ return InitializeLocked();
+}
+
+int AudioProcessingImpl::sample_rate_hz() const {
+ return sample_rate_hz_;
+}
+
+int AudioProcessingImpl::set_num_reverse_channels(int channels) {
+ CriticalSectionScoped crit_scoped(crit_);
+ // Only stereo supported currently.
+ if (channels > 2 || channels < 1) {
+ return kBadParameterError;
+ }
+
+ num_reverse_channels_ = channels;
+
+ return InitializeLocked();
+}
+
+int AudioProcessingImpl::num_reverse_channels() const {
+ return num_reverse_channels_;
+}
+
+int AudioProcessingImpl::set_num_channels(
+ int input_channels,
+ int output_channels) {
+ CriticalSectionScoped crit_scoped(crit_);
+ if (output_channels > input_channels) {
+ return kBadParameterError;
+ }
+
+ // Only stereo supported currently.
+ if (input_channels > 2 || input_channels < 1) {
+ return kBadParameterError;
+ }
+
+ if (output_channels > 2 || output_channels < 1) {
+ return kBadParameterError;
+ }
+
+ num_input_channels_ = input_channels;
+ num_output_channels_ = output_channels;
+
+ return InitializeLocked();
+}
+
+int AudioProcessingImpl::num_input_channels() const {
+ return num_input_channels_;
+}
+
+int AudioProcessingImpl::num_output_channels() const {
+ return num_output_channels_;
+}
+
+int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
+ CriticalSectionScoped crit_scoped(crit_);
+ int err = kNoError;
+
+ if (frame == NULL) {
+ return kNullPointerError;
+ }
+
+ if (frame->_frequencyInHz != sample_rate_hz_) {
+ return kBadSampleRateError;
+ }
+
+ if (frame->_audioChannel != num_input_channels_) {
+ return kBadNumberChannelsError;
+ }
+
+ if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
+ return kBadDataLengthError;
+ }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ event_msg_->set_type(audioproc::Event::STREAM);
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_input_data(frame->_payloadData, data_size);
+ msg->set_delay(stream_delay_ms_);
+ msg->set_drift(echo_cancellation_->stream_drift_samples());
+ msg->set_level(gain_control_->stream_analog_level());
+ }
+#endif
+
+ capture_audio_->DeinterleaveFrom(frame);
+
+ // TODO(ajm): experiment with mixing and AEC placement.
+ if (num_output_channels_ < num_input_channels_) {
+ capture_audio_->Mix(num_output_channels_);
+ frame->_audioChannel = num_output_channels_;
+ }
+
+ bool data_changed = stream_data_changed();
+ if (analysis_needed(data_changed)) {
+ for (int i = 0; i < num_output_channels_; i++) {
+ // Split into a low and high band.
+ SplittingFilterAnalysis(capture_audio_->data(i),
+ capture_audio_->low_pass_split_data(i),
+ capture_audio_->high_pass_split_data(i),
+ capture_audio_->analysis_filter_state1(i),
+ capture_audio_->analysis_filter_state2(i));
+ }
+ }
+
+ err = high_pass_filter_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = gain_control_->AnalyzeCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = echo_cancellation_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ if (echo_control_mobile_->is_enabled() &&
+ noise_suppression_->is_enabled()) {
+ capture_audio_->CopyLowPassToReference();
+ }
+
+ err = noise_suppression_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = voice_detection_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = gain_control_->ProcessCaptureAudio(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ if (synthesis_needed(data_changed)) {
+ for (int i = 0; i < num_output_channels_; i++) {
+ // Recombine low and high bands.
+ SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
+ capture_audio_->high_pass_split_data(i),
+ capture_audio_->data(i),
+ capture_audio_->synthesis_filter_state1(i),
+ capture_audio_->synthesis_filter_state2(i));
+ }
+ }
+
+ // The level estimator operates on the recombined data.
+ err = level_estimator_->ProcessStream(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ capture_audio_->InterleaveTo(frame, data_changed);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_output_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+#endif
+
+ was_stream_delay_set_ = false;
+ return kNoError;
+}
+
+int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
+ CriticalSectionScoped crit_scoped(crit_);
+ int err = kNoError;
+
+ if (frame == NULL) {
+ return kNullPointerError;
+ }
+
+ if (frame->_frequencyInHz != sample_rate_hz_) {
+ return kBadSampleRateError;
+ }
+
+ if (frame->_audioChannel != num_reverse_channels_) {
+ return kBadNumberChannelsError;
+ }
+
+ if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
+ return kBadDataLengthError;
+ }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
+ audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+#endif
+
+ render_audio_->DeinterleaveFrom(frame);
+
+ // TODO(ajm): turn the splitting filter into a component?
+ if (sample_rate_hz_ == kSampleRate32kHz) {
+ for (int i = 0; i < num_reverse_channels_; i++) {
+ // Split into low and high band.
+ SplittingFilterAnalysis(render_audio_->data(i),
+ render_audio_->low_pass_split_data(i),
+ render_audio_->high_pass_split_data(i),
+ render_audio_->analysis_filter_state1(i),
+ render_audio_->analysis_filter_state2(i));
+ }
+ }
+
+ // TODO(ajm): warnings possible from components?
+ err = echo_cancellation_->ProcessRenderAudio(render_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = echo_control_mobile_->ProcessRenderAudio(render_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ err = gain_control_->ProcessRenderAudio(render_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+
+ return err; // TODO(ajm): this is for returning warnings; necessary?
+}
+
+int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+ was_stream_delay_set_ = true;
+ if (delay < 0) {
+ return kBadParameterError;
+ }
+
+ // TODO(ajm): the max is rather arbitrarily chosen; investigate.
+ if (delay > 500) {
+ stream_delay_ms_ = 500;
+ return kBadStreamParameterWarning;
+ }
+
+ stream_delay_ms_ = delay;
+ return kNoError;
+}
+
+int AudioProcessingImpl::stream_delay_ms() const {
+ return stream_delay_ms_;
+}
+
+bool AudioProcessingImpl::was_stream_delay_set() const {
+ return was_stream_delay_set_;
+}
+
+int AudioProcessingImpl::StartDebugRecording(
+ const char filename[AudioProcessing::kMaxFilenameSize]) {
+ CriticalSectionScoped crit_scoped(crit_);
+ assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
+
+ if (filename == NULL) {
+ return kNullPointerError;
+ }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // Stop any ongoing recording.
+ if (debug_file_->Open()) {
+ if (debug_file_->CloseFile() == -1) {
+ return kFileError;
+ }
+ }
+
+ if (debug_file_->OpenFile(filename, false) == -1) {
+ debug_file_->CloseFile();
+ return kFileError;
+ }
+
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
+ }
+ return kNoError;
+#else
+ return kUnsupportedFunctionError;
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+int AudioProcessingImpl::StopDebugRecording() {
+ CriticalSectionScoped crit_scoped(crit_);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // We just return if recording hasn't started.
+ if (debug_file_->Open()) {
+ if (debug_file_->CloseFile() == -1) {
+ return kFileError;
+ }
+ }
+ return kNoError;
+#else
+ return kUnsupportedFunctionError;
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
+ return echo_cancellation_;
+}
+
+EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
+ return echo_control_mobile_;
+}
+
+GainControl* AudioProcessingImpl::gain_control() const {
+ return gain_control_;
+}
+
+HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
+ return high_pass_filter_;
+}
+
+LevelEstimator* AudioProcessingImpl::level_estimator() const {
+ return level_estimator_;
+}
+
+NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
+ return noise_suppression_;
+}
+
+VoiceDetection* AudioProcessingImpl::voice_detection() const {
+ return voice_detection_;
+}
+
+WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
+ CriticalSectionScoped crit_scoped(crit_);
+ id_ = id;
+
+ return kNoError;
+}
+
+bool AudioProcessingImpl::stream_data_changed() const {
+ int enabled_count = 0;
+ std::list<ProcessingComponent*>::const_iterator it;
+ for (it = component_list_.begin(); it != component_list_.end(); it++) {
+ if ((*it)->is_component_enabled()) {
+ enabled_count++;
+ }
+ }
+
+ // Data is unchanged if no components are enabled, or if only level_estimator_
+ // or voice_detection_ is enabled.
+ if (enabled_count == 0) {
+ return false;
+ } else if (enabled_count == 1) {
+ if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
+ return false;
+ }
+ } else if (enabled_count == 2) {
+ if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool AudioProcessingImpl::synthesis_needed(bool stream_data_changed) const {
+ return (stream_data_changed && sample_rate_hz_ == kSampleRate32kHz);
+}
+
+bool AudioProcessingImpl::analysis_needed(bool stream_data_changed) const {
+ if (!stream_data_changed && !voice_detection_->is_enabled()) {
+ // Only level_estimator_ is enabled.
+ return false;
+ } else if (sample_rate_hz_ == kSampleRate32kHz) {
+ // Something besides level_estimator_ is enabled, and we have super-wb.
+ return true;
+ }
+ return false;
+}
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+int AudioProcessingImpl::WriteMessageToDebugFile() {
+ int32_t size = event_msg_->ByteSize();
+ if (size <= 0) {
+ return kUnspecifiedError;
+ }
+#if defined(WEBRTC_BIG_ENDIAN)
+ // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
+ // pretty safe in assuming little-endian.
+#endif
+
+ if (!event_msg_->SerializeToString(&event_str_)) {
+ return kUnspecifiedError;
+ }
+
+ // Write message preceded by its size.
+ if (!debug_file_->Write(&size, sizeof(int32_t))) {
+ return kFileError;
+ }
+ if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
+ return kFileError;
+ }
+
+ event_msg_->Clear();
+
+ return 0;
+}
+
+int AudioProcessingImpl::WriteInitMessage() {
+ event_msg_->set_type(audioproc::Event::INIT);
+ audioproc::Init* msg = event_msg_->mutable_init();
+ msg->set_sample_rate(sample_rate_hz_);
+ msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz());
+ msg->set_num_input_channels(num_input_channels_);
+ msg->set_num_output_channels(num_output_channels_);
+ msg->set_num_reverse_channels(num_reverse_channels_);
+
+ int err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+
+ return kNoError;
+}
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/audio_processing_impl.h b/trunk/src/modules/audio_processing/audio_processing_impl.h
new file mode 100644
index 0000000..c1ab476
--- /dev/null
+++ b/trunk/src/modules/audio_processing/audio_processing_impl.h
@@ -0,0 +1,126 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
+
+#include "audio_processing.h"
+
+#include <list>
+#include <string>
+
+#include "scoped_ptr.h"
+
+namespace webrtc {
+class AudioBuffer;
+class CriticalSectionWrapper;
+class EchoCancellationImpl;
+class EchoControlMobileImpl;
+class FileWrapper;
+class GainControlImpl;
+class HighPassFilterImpl;
+class LevelEstimatorImpl;
+class NoiseSuppressionImpl;
+class ProcessingComponent;
+class VoiceDetectionImpl;
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+namespace audioproc {
+
+class Event;
+
+} // namespace audioproc
+#endif
+
+class AudioProcessingImpl : public AudioProcessing {
+ public:
+ enum {
+ kSampleRate8kHz = 8000,
+ kSampleRate16kHz = 16000,
+ kSampleRate32kHz = 32000
+ };
+
+ explicit AudioProcessingImpl(int id);
+ virtual ~AudioProcessingImpl();
+
+ CriticalSectionWrapper* crit() const;
+
+ int split_sample_rate_hz() const;
+ bool was_stream_delay_set() const;
+
+ // AudioProcessing methods.
+ virtual int Initialize();
+ virtual int InitializeLocked();
+ virtual int set_sample_rate_hz(int rate);
+ virtual int sample_rate_hz() const;
+ virtual int set_num_channels(int input_channels, int output_channels);
+ virtual int num_input_channels() const;
+ virtual int num_output_channels() const;
+ virtual int set_num_reverse_channels(int channels);
+ virtual int num_reverse_channels() const;
+ virtual int ProcessStream(AudioFrame* frame);
+ virtual int AnalyzeReverseStream(AudioFrame* frame);
+ virtual int set_stream_delay_ms(int delay);
+ virtual int stream_delay_ms() const;
+ virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
+ virtual int StopDebugRecording();
+ virtual EchoCancellation* echo_cancellation() const;
+ virtual EchoControlMobile* echo_control_mobile() const;
+ virtual GainControl* gain_control() const;
+ virtual HighPassFilter* high_pass_filter() const;
+ virtual LevelEstimator* level_estimator() const;
+ virtual NoiseSuppression* noise_suppression() const;
+ virtual VoiceDetection* voice_detection() const;
+
+ // Module methods.
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ private:
+ bool stream_data_changed() const;
+ bool synthesis_needed(bool stream_data_changed) const;
+ bool analysis_needed(bool stream_data_changed) const;
+
+ int id_;
+
+ EchoCancellationImpl* echo_cancellation_;
+ EchoControlMobileImpl* echo_control_mobile_;
+ GainControlImpl* gain_control_;
+ HighPassFilterImpl* high_pass_filter_;
+ LevelEstimatorImpl* level_estimator_;
+ NoiseSuppressionImpl* noise_suppression_;
+ VoiceDetectionImpl* voice_detection_;
+
+ std::list<ProcessingComponent*> component_list_;
+ CriticalSectionWrapper* crit_;
+ AudioBuffer* render_audio_;
+ AudioBuffer* capture_audio_;
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // TODO(andrew): make this more graceful. Ideally we would split this stuff
+ // out into a separate class with an "enabled" and "disabled" implementation.
+ int WriteMessageToDebugFile();
+ int WriteInitMessage();
+ scoped_ptr<FileWrapper> debug_file_;
+ scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
+ std::string event_str_; // Memory for protobuf serialization.
+#endif
+
+ int sample_rate_hz_;
+ int split_sample_rate_hz_;
+ int samples_per_channel_;
+ int stream_delay_ms_;
+ bool was_stream_delay_set_;
+
+ int num_reverse_channels_;
+ int num_input_channels_;
+ int num_output_channels_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/debug.proto b/trunk/src/modules/audio_processing/debug.proto
new file mode 100644
index 0000000..4b3a163
--- /dev/null
+++ b/trunk/src/modules/audio_processing/debug.proto
@@ -0,0 +1,37 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+message Init {
+ optional int32 sample_rate = 1;
+ optional int32 device_sample_rate = 2;
+ optional int32 num_input_channels = 3;
+ optional int32 num_output_channels = 4;
+ optional int32 num_reverse_channels = 5;
+}
+
+message ReverseStream {
+ optional bytes data = 1;
+}
+
+message Stream {
+ optional bytes input_data = 1;
+ optional bytes output_data = 2;
+ optional int32 delay = 3;
+ optional sint32 drift = 4;
+ optional int32 level = 5;
+}
+
+message Event {
+ enum Type {
+ INIT = 0;
+ REVERSE_STREAM = 1;
+ STREAM = 2;
+ }
+
+ required Type type = 1;
+
+ optional Init init = 2;
+ optional ReverseStream reverse_stream = 3;
+ optional Stream stream = 4;
+}
diff --git a/trunk/src/modules/audio_processing/echo_cancellation_impl.cc b/trunk/src/modules/audio_processing/echo_cancellation_impl.cc
new file mode 100644
index 0000000..d4c5523
--- /dev/null
+++ b/trunk/src/modules/audio_processing/echo_cancellation_impl.cc
@@ -0,0 +1,371 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "echo_cancellation_impl.h"
+
+#include <cassert>
+#include <string.h>
+
+#include "critical_section_wrapper.h"
+#include "echo_cancellation.h"
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+
+typedef void Handle;
+
+namespace {
+WebRtc_Word16 MapSetting(EchoCancellation::SuppressionLevel level) {
+ switch (level) {
+ case EchoCancellation::kLowSuppression:
+ return kAecNlpConservative;
+ case EchoCancellation::kModerateSuppression:
+ return kAecNlpModerate;
+ case EchoCancellation::kHighSuppression:
+ return kAecNlpAggressive;
+ }
+ assert(false);
+ return -1;
+}
+
+AudioProcessing::Error MapError(int err) {
+ switch (err) {
+ case AEC_UNSUPPORTED_FUNCTION_ERROR:
+ return AudioProcessing::kUnsupportedFunctionError;
+ case AEC_BAD_PARAMETER_ERROR:
+ return AudioProcessing::kBadParameterError;
+ case AEC_BAD_PARAMETER_WARNING:
+ return AudioProcessing::kBadStreamParameterWarning;
+ default:
+ // AEC_UNSPECIFIED_ERROR
+ // AEC_UNINITIALIZED_ERROR
+ // AEC_NULL_POINTER_ERROR
+ return AudioProcessing::kUnspecifiedError;
+ }
+}
+} // namespace
+
+EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm),
+ drift_compensation_enabled_(false),
+ metrics_enabled_(false),
+ suppression_level_(kModerateSuppression),
+ device_sample_rate_hz_(48000),
+ stream_drift_samples_(0),
+ was_stream_drift_set_(false),
+ stream_has_echo_(false),
+ delay_logging_enabled_(false) {}
+
+EchoCancellationImpl::~EchoCancellationImpl() {}
+
+int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == apm_->num_reverse_channels());
+
+ int err = apm_->kNoError;
+
+ // The ordering convention must be followed to pass to the correct AEC.
+ size_t handle_index = 0;
+ for (int i = 0; i < apm_->num_output_channels(); i++) {
+ for (int j = 0; j < audio->num_channels(); j++) {
+ Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+ err = WebRtcAec_BufferFarend(
+ my_handle,
+ audio->low_pass_split_data(j),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle); // TODO(ajm): warning possible?
+ }
+
+ handle_index++;
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ if (!apm_->was_stream_delay_set()) {
+ return apm_->kStreamParameterNotSetError;
+ }
+
+ if (drift_compensation_enabled_ && !was_stream_drift_set_) {
+ return apm_->kStreamParameterNotSetError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == apm_->num_output_channels());
+
+ int err = apm_->kNoError;
+
+ // The ordering convention must be followed to pass to the correct AEC.
+ size_t handle_index = 0;
+ stream_has_echo_ = false;
+ for (int i = 0; i < audio->num_channels(); i++) {
+ for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ Handle* my_handle = handle(handle_index);
+ err = WebRtcAec_Process(
+ my_handle,
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
+ apm_->stream_delay_ms(),
+ stream_drift_samples_);
+
+ if (err != apm_->kNoError) {
+ err = GetHandleError(my_handle);
+ // TODO(ajm): Figure out how to return warnings properly.
+ if (err != apm_->kBadStreamParameterWarning) {
+ return err;
+ }
+ }
+
+ WebRtc_Word16 status = 0;
+ err = WebRtcAec_get_echo_status(my_handle, &status);
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ if (status == 1) {
+ stream_has_echo_ = true;
+ }
+
+ handle_index++;
+ }
+ }
+
+ was_stream_drift_set_ = false;
+ return apm_->kNoError;
+}
+
+int EchoCancellationImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ // Ensure AEC and AECM are not both enabled.
+ if (enable && apm_->echo_control_mobile()->is_enabled()) {
+ return apm_->kBadParameterError;
+ }
+
+ return EnableComponent(enable);
+}
+
+bool EchoCancellationImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (MapSetting(level) == -1) {
+ return apm_->kBadParameterError;
+ }
+
+ suppression_level_ = level;
+ return Configure();
+}
+
+EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
+ const {
+ return suppression_level_;
+}
+
+int EchoCancellationImpl::enable_drift_compensation(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ drift_compensation_enabled_ = enable;
+ return Configure();
+}
+
+bool EchoCancellationImpl::is_drift_compensation_enabled() const {
+ return drift_compensation_enabled_;
+}
+
+int EchoCancellationImpl::set_device_sample_rate_hz(int rate) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (rate < 8000 || rate > 96000) {
+ return apm_->kBadParameterError;
+ }
+
+ device_sample_rate_hz_ = rate;
+ return Initialize();
+}
+
+int EchoCancellationImpl::device_sample_rate_hz() const {
+ return device_sample_rate_hz_;
+}
+
+int EchoCancellationImpl::set_stream_drift_samples(int drift) {
+ was_stream_drift_set_ = true;
+ stream_drift_samples_ = drift;
+ return apm_->kNoError;
+}
+
+int EchoCancellationImpl::stream_drift_samples() const {
+ return stream_drift_samples_;
+}
+
+int EchoCancellationImpl::enable_metrics(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ metrics_enabled_ = enable;
+ return Configure();
+}
+
+bool EchoCancellationImpl::are_metrics_enabled() const {
+ return metrics_enabled_;
+}
+
+// TODO(ajm): we currently just use the metrics from the first AEC. Think more
+// aboue the best way to extend this to multi-channel.
+int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (metrics == NULL) {
+ return apm_->kNullPointerError;
+ }
+
+ if (!is_component_enabled() || !metrics_enabled_) {
+ return apm_->kNotEnabledError;
+ }
+
+ AecMetrics my_metrics;
+ memset(&my_metrics, 0, sizeof(my_metrics));
+ memset(metrics, 0, sizeof(Metrics));
+
+ Handle* my_handle = static_cast<Handle*>(handle(0));
+ int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
+ metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
+ metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
+ metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
+
+ metrics->echo_return_loss.instant = my_metrics.erl.instant;
+ metrics->echo_return_loss.average = my_metrics.erl.average;
+ metrics->echo_return_loss.maximum = my_metrics.erl.max;
+ metrics->echo_return_loss.minimum = my_metrics.erl.min;
+
+ metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
+ metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
+ metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
+ metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
+
+ metrics->a_nlp.instant = my_metrics.aNlp.instant;
+ metrics->a_nlp.average = my_metrics.aNlp.average;
+ metrics->a_nlp.maximum = my_metrics.aNlp.max;
+ metrics->a_nlp.minimum = my_metrics.aNlp.min;
+
+ return apm_->kNoError;
+}
+
+bool EchoCancellationImpl::stream_has_echo() const {
+ return stream_has_echo_;
+}
+
+int EchoCancellationImpl::enable_delay_logging(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ delay_logging_enabled_ = enable;
+ return Configure();
+}
+
+bool EchoCancellationImpl::is_delay_logging_enabled() const {
+ return delay_logging_enabled_;
+}
+
+// TODO(bjornv): How should we handle the multi-channel case?
+int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (median == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (std == NULL) {
+ return apm_->kNullPointerError;
+ }
+
+ if (!is_component_enabled() || !delay_logging_enabled_) {
+ return apm_->kNotEnabledError;
+ }
+
+ Handle* my_handle = static_cast<Handle*>(handle(0));
+ if (WebRtcAec_GetDelayMetrics(my_handle, median, std) !=
+ apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoCancellationImpl::Initialize() {
+ int err = ProcessingComponent::Initialize();
+ if (err != apm_->kNoError || !is_component_enabled()) {
+ return err;
+ }
+
+ was_stream_drift_set_ = false;
+
+ return apm_->kNoError;
+}
+
+void* EchoCancellationImpl::CreateHandle() const {
+ Handle* handle = NULL;
+ if (WebRtcAec_Create(&handle) != apm_->kNoError) {
+ handle = NULL;
+ } else {
+ assert(handle != NULL);
+ }
+
+ return handle;
+}
+
+int EchoCancellationImpl::DestroyHandle(void* handle) const {
+ assert(handle != NULL);
+ return WebRtcAec_Free(static_cast<Handle*>(handle));
+}
+
+int EchoCancellationImpl::InitializeHandle(void* handle) const {
+ assert(handle != NULL);
+ return WebRtcAec_Init(static_cast<Handle*>(handle),
+ apm_->sample_rate_hz(),
+ device_sample_rate_hz_);
+}
+
+int EchoCancellationImpl::ConfigureHandle(void* handle) const {
+ assert(handle != NULL);
+ AecConfig config;
+ config.metricsMode = metrics_enabled_;
+ config.nlpMode = MapSetting(suppression_level_);
+ config.skewMode = drift_compensation_enabled_;
+ config.delay_logging = delay_logging_enabled_;
+
+ return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
+}
+
+int EchoCancellationImpl::num_handles_required() const {
+ return apm_->num_output_channels() *
+ apm_->num_reverse_channels();
+}
+
+int EchoCancellationImpl::GetHandleError(void* handle) const {
+ assert(handle != NULL);
+ return MapError(WebRtcAec_get_error_code(static_cast<Handle*>(handle)));
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/echo_cancellation_impl.h b/trunk/src/modules/audio_processing/echo_cancellation_impl.h
new file mode 100644
index 0000000..3c2198c
--- /dev/null
+++ b/trunk/src/modules/audio_processing/echo_cancellation_impl.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class EchoCancellationImpl : public EchoCancellation,
+ public ProcessingComponent {
+ public:
+ explicit EchoCancellationImpl(const AudioProcessingImpl* apm);
+ virtual ~EchoCancellationImpl();
+
+ int ProcessRenderAudio(const AudioBuffer* audio);
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // EchoCancellation implementation.
+ virtual bool is_enabled() const;
+ virtual int device_sample_rate_hz() const;
+ virtual int stream_drift_samples() const;
+
+ // ProcessingComponent implementation.
+ virtual int Initialize();
+
+ private:
+ // EchoCancellation implementation.
+ virtual int Enable(bool enable);
+ virtual int enable_drift_compensation(bool enable);
+ virtual bool is_drift_compensation_enabled() const;
+ virtual int set_device_sample_rate_hz(int rate);
+ virtual int set_stream_drift_samples(int drift);
+ virtual int set_suppression_level(SuppressionLevel level);
+ virtual SuppressionLevel suppression_level() const;
+ virtual int enable_metrics(bool enable);
+ virtual bool are_metrics_enabled() const;
+ virtual bool stream_has_echo() const;
+ virtual int GetMetrics(Metrics* metrics);
+ virtual int enable_delay_logging(bool enable);
+ virtual bool is_delay_logging_enabled() const;
+ virtual int GetDelayMetrics(int* median, int* std);
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+ bool drift_compensation_enabled_;
+ bool metrics_enabled_;
+ SuppressionLevel suppression_level_;
+ int device_sample_rate_hz_;
+ int stream_drift_samples_;
+ bool was_stream_drift_set_;
+ bool stream_has_echo_;
+ bool delay_logging_enabled_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc b/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc
new file mode 100644
index 0000000..9427789
--- /dev/null
+++ b/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc
@@ -0,0 +1,300 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "echo_control_mobile_impl.h"
+
+#include <cassert>
+#include <cstring>
+
+#include "critical_section_wrapper.h"
+#include "echo_control_mobile.h"
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+
+typedef void Handle;
+
+namespace {
+WebRtc_Word16 MapSetting(EchoControlMobile::RoutingMode mode) {
+ switch (mode) {
+ case EchoControlMobile::kQuietEarpieceOrHeadset:
+ return 0;
+ case EchoControlMobile::kEarpiece:
+ return 1;
+ case EchoControlMobile::kLoudEarpiece:
+ return 2;
+ case EchoControlMobile::kSpeakerphone:
+ return 3;
+ case EchoControlMobile::kLoudSpeakerphone:
+ return 4;
+ }
+ assert(false);
+ return -1;
+}
+
+AudioProcessing::Error MapError(int err) {
+ switch (err) {
+ case AECM_UNSUPPORTED_FUNCTION_ERROR:
+ return AudioProcessing::kUnsupportedFunctionError;
+ case AECM_NULL_POINTER_ERROR:
+ return AudioProcessing::kNullPointerError;
+ case AECM_BAD_PARAMETER_ERROR:
+ return AudioProcessing::kBadParameterError;
+ case AECM_BAD_PARAMETER_WARNING:
+ return AudioProcessing::kBadStreamParameterWarning;
+ default:
+ // AECM_UNSPECIFIED_ERROR
+ // AECM_UNINITIALIZED_ERROR
+ return AudioProcessing::kUnspecifiedError;
+ }
+}
+} // namespace
+
+size_t EchoControlMobile::echo_path_size_bytes() {
+ return WebRtcAecm_echo_path_size_bytes();
+}
+
+EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm),
+ routing_mode_(kSpeakerphone),
+ comfort_noise_enabled_(true),
+ external_echo_path_(NULL) {}
+
+EchoControlMobileImpl::~EchoControlMobileImpl() {
+ if (external_echo_path_ != NULL) {
+ delete [] external_echo_path_;
+ external_echo_path_ = NULL;
+ }
+}
+
+int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == apm_->num_reverse_channels());
+
+ int err = apm_->kNoError;
+
+ // The ordering convention must be followed to pass to the correct AECM.
+ size_t handle_index = 0;
+ for (int i = 0; i < apm_->num_output_channels(); i++) {
+ for (int j = 0; j < audio->num_channels(); j++) {
+ Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+ err = WebRtcAecm_BufferFarend(
+ my_handle,
+ audio->low_pass_split_data(j),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle); // TODO(ajm): warning possible?
+ }
+
+ handle_index++;
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ if (!apm_->was_stream_delay_set()) {
+ return apm_->kStreamParameterNotSetError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == apm_->num_output_channels());
+
+ int err = apm_->kNoError;
+
+ // The ordering convention must be followed to pass to the correct AECM.
+ size_t handle_index = 0;
+ for (int i = 0; i < audio->num_channels(); i++) {
+ // TODO(ajm): improve how this works, possibly inside AECM.
+ // This is kind of hacked up.
+ WebRtc_Word16* noisy = audio->low_pass_reference(i);
+ WebRtc_Word16* clean = audio->low_pass_split_data(i);
+ if (noisy == NULL) {
+ noisy = clean;
+ clean = NULL;
+ }
+ for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+ Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+ err = WebRtcAecm_Process(
+ my_handle,
+ noisy,
+ clean,
+ audio->low_pass_split_data(i),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
+ apm_->stream_delay_ms());
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle); // TODO(ajm): warning possible?
+ }
+
+ handle_index++;
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoControlMobileImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ // Ensure AEC and AECM are not both enabled.
+ if (enable && apm_->echo_cancellation()->is_enabled()) {
+ return apm_->kBadParameterError;
+ }
+
+ return EnableComponent(enable);
+}
+
+bool EchoControlMobileImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (MapSetting(mode) == -1) {
+ return apm_->kBadParameterError;
+ }
+
+ routing_mode_ = mode;
+ return Configure();
+}
+
+EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
+ const {
+ return routing_mode_;
+}
+
+int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ comfort_noise_enabled_ = enable;
+ return Configure();
+}
+
+bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
+ return comfort_noise_enabled_;
+}
+
+int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
+ size_t size_bytes) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (echo_path == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (size_bytes != echo_path_size_bytes()) {
+ // Size mismatch
+ return apm_->kBadParameterError;
+ }
+
+ if (external_echo_path_ == NULL) {
+ external_echo_path_ = new unsigned char[size_bytes];
+ }
+ memcpy(external_echo_path_, echo_path, size_bytes);
+
+ return Initialize();
+}
+
+int EchoControlMobileImpl::GetEchoPath(void* echo_path,
+ size_t size_bytes) const {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (echo_path == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (size_bytes != echo_path_size_bytes()) {
+ // Size mismatch
+ return apm_->kBadParameterError;
+ }
+ if (!is_component_enabled()) {
+ return apm_->kNotEnabledError;
+ }
+
+ // Get the echo path from the first channel
+ Handle* my_handle = static_cast<Handle*>(handle(0));
+ if (WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes) != 0) {
+ return GetHandleError(my_handle);
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoControlMobileImpl::Initialize() {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ if (apm_->sample_rate_hz() == apm_->kSampleRate32kHz) {
+ // AECM doesn't support super-wideband.
+ return apm_->kBadSampleRateError;
+ }
+
+ return ProcessingComponent::Initialize();
+}
+
+void* EchoControlMobileImpl::CreateHandle() const {
+ Handle* handle = NULL;
+ if (WebRtcAecm_Create(&handle) != apm_->kNoError) {
+ handle = NULL;
+ } else {
+ assert(handle != NULL);
+ }
+
+ return handle;
+}
+
+int EchoControlMobileImpl::DestroyHandle(void* handle) const {
+ return WebRtcAecm_Free(static_cast<Handle*>(handle));
+}
+
+int EchoControlMobileImpl::InitializeHandle(void* handle) const {
+ assert(handle != NULL);
+ Handle* my_handle = static_cast<Handle*>(handle);
+ if (WebRtcAecm_Init(my_handle, apm_->sample_rate_hz()) != 0) {
+ return GetHandleError(my_handle);
+ }
+ if (external_echo_path_ != NULL) {
+ if (WebRtcAecm_InitEchoPath(my_handle,
+ external_echo_path_,
+ echo_path_size_bytes()) != 0) {
+ return GetHandleError(my_handle);
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
+ AecmConfig config;
+ config.cngMode = comfort_noise_enabled_;
+ config.echoMode = MapSetting(routing_mode_);
+
+ return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
+}
+
+int EchoControlMobileImpl::num_handles_required() const {
+ return apm_->num_output_channels() *
+ apm_->num_reverse_channels();
+}
+
+int EchoControlMobileImpl::GetHandleError(void* handle) const {
+ assert(handle != NULL);
+ return MapError(WebRtcAecm_get_error_code(static_cast<Handle*>(handle)));
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/echo_control_mobile_impl.h b/trunk/src/modules/audio_processing/echo_control_mobile_impl.h
new file mode 100644
index 0000000..6d9e369
--- /dev/null
+++ b/trunk/src/modules/audio_processing/echo_control_mobile_impl.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class EchoControlMobileImpl : public EchoControlMobile,
+ public ProcessingComponent {
+ public:
+ explicit EchoControlMobileImpl(const AudioProcessingImpl* apm);
+ virtual ~EchoControlMobileImpl();
+
+ int ProcessRenderAudio(const AudioBuffer* audio);
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // EchoControlMobile implementation.
+ virtual bool is_enabled() const;
+
+ // ProcessingComponent implementation.
+ virtual int Initialize();
+
+ private:
+ // EchoControlMobile implementation.
+ virtual int Enable(bool enable);
+ virtual int set_routing_mode(RoutingMode mode);
+ virtual RoutingMode routing_mode() const;
+ virtual int enable_comfort_noise(bool enable);
+ virtual bool is_comfort_noise_enabled() const;
+ virtual int SetEchoPath(const void* echo_path, size_t size_bytes);
+ virtual int GetEchoPath(void* echo_path, size_t size_bytes) const;
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+ RoutingMode routing_mode_;
+ bool comfort_noise_enabled_;
+ unsigned char* external_echo_path_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/gain_control_impl.cc b/trunk/src/modules/audio_processing/gain_control_impl.cc
new file mode 100644
index 0000000..a518ab5
--- /dev/null
+++ b/trunk/src/modules/audio_processing/gain_control_impl.cc
@@ -0,0 +1,367 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gain_control_impl.h"
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#include "gain_control.h"
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+
+typedef void Handle;
+
+namespace {
+WebRtc_Word16 MapSetting(GainControl::Mode mode) {
+ switch (mode) {
+ case GainControl::kAdaptiveAnalog:
+ return kAgcModeAdaptiveAnalog;
+ case GainControl::kAdaptiveDigital:
+ return kAgcModeAdaptiveDigital;
+ case GainControl::kFixedDigital:
+ return kAgcModeFixedDigital;
+ }
+ assert(false);
+ return -1;
+}
+} // namespace
+
+GainControlImpl::GainControlImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm),
+ mode_(kAdaptiveAnalog),
+ minimum_capture_level_(0),
+ maximum_capture_level_(255),
+ limiter_enabled_(true),
+ target_level_dbfs_(3),
+ compression_gain_db_(9),
+ analog_capture_level_(0),
+ was_analog_level_set_(false),
+ stream_is_saturated_(false) {}
+
+GainControlImpl::~GainControlImpl() {}
+
+int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+
+ WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
+ if (audio->num_channels() > 1) {
+ audio->CopyAndMixLowPass(1);
+ mixed_data = audio->mixed_low_pass_data(0);
+ }
+
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+ int err = WebRtcAgc_AddFarend(
+ my_handle,
+ mixed_data,
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == num_handles());
+
+ int err = apm_->kNoError;
+
+ if (mode_ == kAdaptiveAnalog) {
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+ err = WebRtcAgc_AddMic(
+ my_handle,
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+ }
+ } else if (mode_ == kAdaptiveDigital) {
+
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+ WebRtc_Word32 capture_level_out = 0;
+
+ err = WebRtcAgc_VirtualMic(
+ my_handle,
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
+ //capture_levels_[i],
+ analog_capture_level_,
+ &capture_level_out);
+
+ capture_levels_[i] = capture_level_out;
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
+ return apm_->kStreamParameterNotSetError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == num_handles());
+
+ stream_is_saturated_ = false;
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+ WebRtc_Word32 capture_level_out = 0;
+ WebRtc_UWord8 saturation_warning = 0;
+
+ int err = WebRtcAgc_Process(
+ my_handle,
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ capture_levels_[i],
+ &capture_level_out,
+ apm_->echo_cancellation()->stream_has_echo(),
+ &saturation_warning);
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ capture_levels_[i] = capture_level_out;
+ if (saturation_warning == 1) {
+ stream_is_saturated_ = true;
+ }
+ }
+
+ if (mode_ == kAdaptiveAnalog) {
+ // Take the analog level to be the average across the handles.
+ analog_capture_level_ = 0;
+ for (int i = 0; i < num_handles(); i++) {
+ analog_capture_level_ += capture_levels_[i];
+ }
+
+ analog_capture_level_ /= num_handles();
+ }
+
+ was_analog_level_set_ = false;
+ return apm_->kNoError;
+}
+
+// TODO(ajm): ensure this is called under kAdaptiveAnalog.
+int GainControlImpl::set_stream_analog_level(int level) {
+ was_analog_level_set_ = true;
+ if (level < minimum_capture_level_ || level > maximum_capture_level_) {
+ return apm_->kBadParameterError;
+ }
+
+ if (mode_ == kAdaptiveAnalog) {
+ if (level != analog_capture_level_) {
+ // The analog level has been changed; update our internal levels.
+ capture_levels_.assign(num_handles(), level);
+ }
+ }
+ analog_capture_level_ = level;
+
+ return apm_->kNoError;
+}
+
+int GainControlImpl::stream_analog_level() {
+ // TODO(ajm): enable this assertion?
+ //assert(mode_ == kAdaptiveAnalog);
+
+ return analog_capture_level_;
+}
+
+int GainControlImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool GainControlImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int GainControlImpl::set_mode(Mode mode) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (MapSetting(mode) == -1) {
+ return apm_->kBadParameterError;
+ }
+
+ mode_ = mode;
+ return Initialize();
+}
+
+GainControl::Mode GainControlImpl::mode() const {
+ return mode_;
+}
+
+int GainControlImpl::set_analog_level_limits(int minimum,
+ int maximum) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (minimum < 0) {
+ return apm_->kBadParameterError;
+ }
+
+ if (maximum > 65535) {
+ return apm_->kBadParameterError;
+ }
+
+ if (maximum < minimum) {
+ return apm_->kBadParameterError;
+ }
+
+ minimum_capture_level_ = minimum;
+ maximum_capture_level_ = maximum;
+
+ return Initialize();
+}
+
+int GainControlImpl::analog_level_minimum() const {
+ return minimum_capture_level_;
+}
+
+int GainControlImpl::analog_level_maximum() const {
+ return maximum_capture_level_;
+}
+
+bool GainControlImpl::stream_is_saturated() const {
+ return stream_is_saturated_;
+}
+
+int GainControlImpl::set_target_level_dbfs(int level) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (level > 31 || level < 0) {
+ return apm_->kBadParameterError;
+ }
+
+ target_level_dbfs_ = level;
+ return Configure();
+}
+
+int GainControlImpl::target_level_dbfs() const {
+ return target_level_dbfs_;
+}
+
+int GainControlImpl::set_compression_gain_db(int gain) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (gain < 0 || gain > 90) {
+ return apm_->kBadParameterError;
+ }
+
+ compression_gain_db_ = gain;
+ return Configure();
+}
+
+int GainControlImpl::compression_gain_db() const {
+ return compression_gain_db_;
+}
+
+int GainControlImpl::enable_limiter(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ limiter_enabled_ = enable;
+ return Configure();
+}
+
+bool GainControlImpl::is_limiter_enabled() const {
+ return limiter_enabled_;
+}
+
+int GainControlImpl::Initialize() {
+ int err = ProcessingComponent::Initialize();
+ if (err != apm_->kNoError || !is_component_enabled()) {
+ return err;
+ }
+
+ analog_capture_level_ =
+ (maximum_capture_level_ - minimum_capture_level_) >> 1;
+ capture_levels_.assign(num_handles(), analog_capture_level_);
+ was_analog_level_set_ = false;
+
+ return apm_->kNoError;
+}
+
+void* GainControlImpl::CreateHandle() const {
+ Handle* handle = NULL;
+ if (WebRtcAgc_Create(&handle) != apm_->kNoError) {
+ handle = NULL;
+ } else {
+ assert(handle != NULL);
+ }
+
+ return handle;
+}
+
+int GainControlImpl::DestroyHandle(void* handle) const {
+ return WebRtcAgc_Free(static_cast<Handle*>(handle));
+}
+
+int GainControlImpl::InitializeHandle(void* handle) const {
+ return WebRtcAgc_Init(static_cast<Handle*>(handle),
+ minimum_capture_level_,
+ maximum_capture_level_,
+ MapSetting(mode_),
+ apm_->sample_rate_hz());
+}
+
+int GainControlImpl::ConfigureHandle(void* handle) const {
+ WebRtcAgc_config_t config;
+ // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
+ // change the interface.
+ //assert(target_level_dbfs_ <= 0);
+ //config.targetLevelDbfs = static_cast<WebRtc_Word16>(-target_level_dbfs_);
+ config.targetLevelDbfs = static_cast<WebRtc_Word16>(target_level_dbfs_);
+ config.compressionGaindB =
+ static_cast<WebRtc_Word16>(compression_gain_db_);
+ config.limiterEnable = limiter_enabled_;
+
+ return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
+}
+
+int GainControlImpl::num_handles_required() const {
+ return apm_->num_output_channels();
+}
+
+int GainControlImpl::GetHandleError(void* handle) const {
+ // The AGC has no get_error() function.
+ // (Despite listing errors in its interface...)
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/gain_control_impl.h b/trunk/src/modules/audio_processing/gain_control_impl.h
new file mode 100644
index 0000000..5915eeb
--- /dev/null
+++ b/trunk/src/modules/audio_processing/gain_control_impl.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+
+#include <vector>
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class GainControlImpl : public GainControl,
+ public ProcessingComponent {
+ public:
+ explicit GainControlImpl(const AudioProcessingImpl* apm);
+ virtual ~GainControlImpl();
+
+ int ProcessRenderAudio(AudioBuffer* audio);
+ int AnalyzeCaptureAudio(AudioBuffer* audio);
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // ProcessingComponent implementation.
+ virtual int Initialize();
+
+ // GainControl implementation.
+ virtual bool is_enabled() const;
+ virtual int stream_analog_level();
+
+ private:
+ // GainControl implementation.
+ virtual int Enable(bool enable);
+ virtual int set_stream_analog_level(int level);
+ virtual int set_mode(Mode mode);
+ virtual Mode mode() const;
+ virtual int set_target_level_dbfs(int level);
+ virtual int target_level_dbfs() const;
+ virtual int set_compression_gain_db(int gain);
+ virtual int compression_gain_db() const;
+ virtual int enable_limiter(bool enable);
+ virtual bool is_limiter_enabled() const;
+ virtual int set_analog_level_limits(int minimum, int maximum);
+ virtual int analog_level_minimum() const;
+ virtual int analog_level_maximum() const;
+ virtual bool stream_is_saturated() const;
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+ Mode mode_;
+ int minimum_capture_level_;
+ int maximum_capture_level_;
+ bool limiter_enabled_;
+ int target_level_dbfs_;
+ int compression_gain_db_;
+ std::vector<int> capture_levels_;
+ int analog_capture_level_;
+ bool was_analog_level_set_;
+ bool stream_is_saturated_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/high_pass_filter_impl.cc b/trunk/src/modules/audio_processing/high_pass_filter_impl.cc
new file mode 100644
index 0000000..b20fed8
--- /dev/null
+++ b/trunk/src/modules/audio_processing/high_pass_filter_impl.cc
@@ -0,0 +1,173 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "high_pass_filter_impl.h"
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+namespace {
+const WebRtc_Word16 kFilterCoefficients8kHz[5] =
+ {3798, -7596, 3798, 7807, -3733};
+
+const WebRtc_Word16 kFilterCoefficients[5] =
+ {4012, -8024, 4012, 8002, -3913};
+
+struct FilterState {
+ WebRtc_Word16 y[4];
+ WebRtc_Word16 x[2];
+ const WebRtc_Word16* ba;
+};
+
+int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
+ assert(hpf != NULL);
+
+ if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) {
+ hpf->ba = kFilterCoefficients8kHz;
+ } else {
+ hpf->ba = kFilterCoefficients;
+ }
+
+ WebRtcSpl_MemSetW16(hpf->x, 0, 2);
+ WebRtcSpl_MemSetW16(hpf->y, 0, 4);
+
+ return AudioProcessing::kNoError;
+}
+
+int Filter(FilterState* hpf, WebRtc_Word16* data, int length) {
+ assert(hpf != NULL);
+
+ WebRtc_Word32 tmp_int32 = 0;
+ WebRtc_Word16* y = hpf->y;
+ WebRtc_Word16* x = hpf->x;
+ const WebRtc_Word16* ba = hpf->ba;
+
+ for (int i = 0; i < length; i++) {
+ // y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
+ // + -a[1] * y[i-1] + -a[2] * y[i-2];
+
+ tmp_int32 =
+ WEBRTC_SPL_MUL_16_16(y[1], ba[3]); // -a[1] * y[i-1] (low part)
+ tmp_int32 +=
+ WEBRTC_SPL_MUL_16_16(y[3], ba[4]); // -a[2] * y[i-2] (low part)
+ tmp_int32 = (tmp_int32 >> 15);
+ tmp_int32 +=
+ WEBRTC_SPL_MUL_16_16(y[0], ba[3]); // -a[1] * y[i-1] (high part)
+ tmp_int32 +=
+ WEBRTC_SPL_MUL_16_16(y[2], ba[4]); // -a[2] * y[i-2] (high part)
+ tmp_int32 = (tmp_int32 << 1);
+
+ tmp_int32 += WEBRTC_SPL_MUL_16_16(data[i], ba[0]); // b[0]*x[0]
+ tmp_int32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); // b[1]*x[i-1]
+ tmp_int32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); // b[2]*x[i-2]
+
+ // Update state (input part)
+ x[1] = x[0];
+ x[0] = data[i];
+
+ // Update state (filtered part)
+ y[2] = y[0];
+ y[3] = y[1];
+ y[0] = static_cast<WebRtc_Word16>(tmp_int32 >> 13);
+ y[1] = static_cast<WebRtc_Word16>((tmp_int32 -
+ WEBRTC_SPL_LSHIFT_W32(static_cast<WebRtc_Word32>(y[0]), 13)) << 2);
+
+ // Rounding in Q12, i.e. add 2^11
+ tmp_int32 += 2048;
+
+ // Saturate (to 2^27) so that the HP filtered signal does not overflow
+ tmp_int32 = WEBRTC_SPL_SAT(static_cast<WebRtc_Word32>(134217727),
+ tmp_int32,
+ static_cast<WebRtc_Word32>(-134217728));
+
+ // Convert back to Q0 and use rounding
+ data[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_int32, 12);
+
+ }
+
+ return AudioProcessing::kNoError;
+}
+} // namespace
+
+typedef FilterState Handle;
+
+HighPassFilterImpl::HighPassFilterImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm) {}
+
+HighPassFilterImpl::~HighPassFilterImpl() {}
+
+int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ int err = apm_->kNoError;
+
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ assert(audio->samples_per_split_channel() <= 160);
+
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+ err = Filter(my_handle,
+ audio->low_pass_split_data(i),
+ audio->samples_per_split_channel());
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int HighPassFilterImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool HighPassFilterImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+void* HighPassFilterImpl::CreateHandle() const {
+ return new FilterState;
+}
+
+int HighPassFilterImpl::DestroyHandle(void* handle) const {
+ delete static_cast<Handle*>(handle);
+ return apm_->kNoError;
+}
+
+int HighPassFilterImpl::InitializeHandle(void* handle) const {
+ return InitializeFilter(static_cast<Handle*>(handle),
+ apm_->sample_rate_hz());
+}
+
+int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const {
+ return apm_->kNoError; // Not configurable.
+}
+
+int HighPassFilterImpl::num_handles_required() const {
+ return apm_->num_output_channels();
+}
+
+int HighPassFilterImpl::GetHandleError(void* handle) const {
+ // The component has no detailed errors.
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/high_pass_filter_impl.h b/trunk/src/modules/audio_processing/high_pass_filter_impl.h
new file mode 100644
index 0000000..94a9c89
--- /dev/null
+++ b/trunk/src/modules/audio_processing/high_pass_filter_impl.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class HighPassFilterImpl : public HighPassFilter,
+ public ProcessingComponent {
+ public:
+ explicit HighPassFilterImpl(const AudioProcessingImpl* apm);
+ virtual ~HighPassFilterImpl();
+
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // HighPassFilter implementation.
+ virtual bool is_enabled() const;
+
+ private:
+ // HighPassFilter implementation.
+ virtual int Enable(bool enable);
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/include/audio_processing.h b/trunk/src/modules/audio_processing/include/audio_processing.h
new file mode 100644
index 0000000..aefb824
--- /dev/null
+++ b/trunk/src/modules/audio_processing/include/audio_processing.h
@@ -0,0 +1,595 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
+
+#include <stddef.h> // size_t
+
+#include "typedefs.h"
+#include "module.h"
+
+namespace webrtc {
+
+class AudioFrame;
+class EchoCancellation;
+class EchoControlMobile;
+class GainControl;
+class HighPassFilter;
+class LevelEstimator;
+class NoiseSuppression;
+class VoiceDetection;
+
+// The Audio Processing Module (APM) provides a collection of voice processing
+// components designed for real-time communications software.
+//
+// APM operates on two audio streams on a frame-by-frame basis. Frames of the
+// primary stream, on which all processing is applied, are passed to
+// |ProcessStream()|. Frames of the reverse direction stream, which are used for
+// analysis by some components, are passed to |AnalyzeReverseStream()|. On the
+// client-side, this will typically be the near-end (capture) and far-end
+// (render) streams, respectively. APM should be placed in the signal chain as
+// close to the audio hardware abstraction layer (HAL) as possible.
+//
+// On the server-side, the reverse stream will normally not be used, with
+// processing occurring on each incoming stream.
+//
+// Component interfaces follow a similar pattern and are accessed through
+// corresponding getters in APM. All components are disabled at create-time,
+// with default settings that are recommended for most situations. New settings
+// can be applied without enabling a component. Enabling a component triggers
+// memory allocation and initialization to allow it to start processing the
+// streams.
+//
+// Thread safety is provided with the following assumptions to reduce locking
+// overhead:
+// 1. The stream getters and setters are called from the same thread as
+// ProcessStream(). More precisely, stream functions are never called
+// concurrently with ProcessStream().
+// 2. Parameter getters are never called concurrently with the corresponding
+// setter.
+//
+// APM accepts only 16-bit linear PCM audio data in frames of 10 ms. Multiple
+// channels should be interleaved.
+//
+// Usage example, omitting error checking:
+// AudioProcessing* apm = AudioProcessing::Create(0);
+// apm->set_sample_rate_hz(32000); // Super-wideband processing.
+//
+// // Mono capture and stereo render.
+// apm->set_num_channels(1, 1);
+// apm->set_num_reverse_channels(2);
+//
+// apm->high_pass_filter()->Enable(true);
+//
+// apm->echo_cancellation()->enable_drift_compensation(false);
+// apm->echo_cancellation()->Enable(true);
+//
+// apm->noise_reduction()->set_level(kHighSuppression);
+// apm->noise_reduction()->Enable(true);
+//
+// apm->gain_control()->set_analog_level_limits(0, 255);
+// apm->gain_control()->set_mode(kAdaptiveAnalog);
+// apm->gain_control()->Enable(true);
+//
+// apm->voice_detection()->Enable(true);
+//
+// // Start a voice call...
+//
+// // ... Render frame arrives bound for the audio HAL ...
+// apm->AnalyzeReverseStream(render_frame);
+//
+// // ... Capture frame arrives from the audio HAL ...
+// // Call required set_stream_ functions.
+// apm->set_stream_delay_ms(delay_ms);
+// apm->gain_control()->set_stream_analog_level(analog_level);
+//
+// apm->ProcessStream(capture_frame);
+//
+// // Call required stream_ functions.
+// analog_level = apm->gain_control()->stream_analog_level();
+// has_voice = apm->stream_has_voice();
+//
+// // Repeate render and capture processing for the duration of the call...
+// // Start a new call...
+// apm->Initialize();
+//
+// // Close the application...
+// AudioProcessing::Destroy(apm);
+// apm = NULL;
+//
+class AudioProcessing : public Module {
+ public:
+ // Creates a APM instance, with identifier |id|. Use one instance for every
+ // primary audio stream requiring processing. On the client-side, this would
+ // typically be one instance for the near-end stream, and additional instances
+ // for each far-end stream which requires processing. On the server-side,
+ // this would typically be one instance for every incoming stream.
+ static AudioProcessing* Create(int id);
+ virtual ~AudioProcessing() {};
+
+ // TODO(andrew): remove this method. We now allow users to delete instances
+ // directly, useful for scoped_ptr.
+ // Destroys a |apm| instance.
+ static void Destroy(AudioProcessing* apm);
+
+ // Initializes internal states, while retaining all user settings. This
+ // should be called before beginning to process a new audio stream. However,
+ // it is not necessary to call before processing the first stream after
+ // creation.
+ virtual int Initialize() = 0;
+
+ // Sets the sample |rate| in Hz for both the primary and reverse audio
+ // streams. 8000, 16000 or 32000 Hz are permitted.
+ virtual int set_sample_rate_hz(int rate) = 0;
+ virtual int sample_rate_hz() const = 0;
+
+ // Sets the number of channels for the primary audio stream. Input frames must
+ // contain a number of channels given by |input_channels|, while output frames
+ // will be returned with number of channels given by |output_channels|.
+ virtual int set_num_channels(int input_channels, int output_channels) = 0;
+ virtual int num_input_channels() const = 0;
+ virtual int num_output_channels() const = 0;
+
+ // Sets the number of channels for the reverse audio stream. Input frames must
+ // contain a number of channels given by |channels|.
+ virtual int set_num_reverse_channels(int channels) = 0;
+ virtual int num_reverse_channels() const = 0;
+
+ // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
+ // this is the near-end (or captured) audio.
+ //
+ // If needed for enabled functionality, any function with the set_stream_ tag
+ // must be called prior to processing the current frame. Any getter function
+ // with the stream_ tag which is needed should be called after processing.
+ //
+ // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
+ // members of |frame| must be valid, and correspond to settings supplied
+ // to APM.
+ virtual int ProcessStream(AudioFrame* frame) = 0;
+
+ // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame
+ // will not be modified. On the client-side, this is the far-end (or to be
+ // rendered) audio.
+ //
+ // It is only necessary to provide this if echo processing is enabled, as the
+ // reverse stream forms the echo reference signal. It is recommended, but not
+ // necessary, to provide if gain control is enabled. On the server-side this
+ // typically will not be used. If you're not sure what to pass in here,
+ // chances are you don't need to use it.
+ //
+ // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
+ // members of |frame| must be valid.
+ //
+ // TODO(ajm): add const to input; requires an implementation fix.
+ virtual int AnalyzeReverseStream(AudioFrame* frame) = 0;
+
+ // This must be called if and only if echo processing is enabled.
+ //
+ // Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end
+ // frame and ProcessStream() receiving a near-end frame containing the
+ // corresponding echo. On the client-side this can be expressed as
+ // delay = (t_render - t_analyze) + (t_process - t_capture)
+ // where,
+ // - t_analyze is the time a frame is passed to AnalyzeReverseStream() and
+ // t_render is the time the first sample of the same frame is rendered by
+ // the audio hardware.
+ // - t_capture is the time the first sample of a frame is captured by the
+ // audio hardware and t_pull is the time the same frame is passed to
+ // ProcessStream().
+ virtual int set_stream_delay_ms(int delay) = 0;
+ virtual int stream_delay_ms() const = 0;
+
+ // Starts recording debugging information to a file specified by |filename|,
+ // a NULL-terminated string. If there is an ongoing recording, the old file
+ // will be closed, and recording will continue in the newly specified file.
+ // An already existing file will be overwritten without warning.
+ static const size_t kMaxFilenameSize = 1024;
+ virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
+
+ // Stops recording debugging information, and closes the file. Recording
+ // cannot be resumed in the same file (without overwriting it).
+ virtual int StopDebugRecording() = 0;
+
+ // These provide access to the component interfaces and should never return
+ // NULL. The pointers will be valid for the lifetime of the APM instance.
+ // The memory for these objects is entirely managed internally.
+ virtual EchoCancellation* echo_cancellation() const = 0;
+ virtual EchoControlMobile* echo_control_mobile() const = 0;
+ virtual GainControl* gain_control() const = 0;
+ virtual HighPassFilter* high_pass_filter() const = 0;
+ virtual LevelEstimator* level_estimator() const = 0;
+ virtual NoiseSuppression* noise_suppression() const = 0;
+ virtual VoiceDetection* voice_detection() const = 0;
+
+ struct Statistic {
+ int instant; // Instantaneous value.
+ int average; // Long-term average.
+ int maximum; // Long-term maximum.
+ int minimum; // Long-term minimum.
+ };
+
+ enum Error {
+ // Fatal errors.
+ kNoError = 0,
+ kUnspecifiedError = -1,
+ kCreationFailedError = -2,
+ kUnsupportedComponentError = -3,
+ kUnsupportedFunctionError = -4,
+ kNullPointerError = -5,
+ kBadParameterError = -6,
+ kBadSampleRateError = -7,
+ kBadDataLengthError = -8,
+ kBadNumberChannelsError = -9,
+ kFileError = -10,
+ kStreamParameterNotSetError = -11,
+ kNotEnabledError = -12,
+
+ // Warnings are non-fatal.
+ // This results when a set_stream_ parameter is out of range. Processing
+ // will continue, but the parameter may have been truncated.
+ kBadStreamParameterWarning = -13
+ };
+
+ // Inherited from Module.
+ virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; };
+ virtual WebRtc_Word32 Process() { return -1; };
+};
+
+// The acoustic echo cancellation (AEC) component provides better performance
+// than AECM but also requires more processing power and is dependent on delay
+// stability and reporting accuracy. As such it is well-suited and recommended
+// for PC and IP phone applications.
+//
+// Not recommended to be enabled on the server-side.
+class EchoCancellation {
+ public:
+ // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
+ // Enabling one will disable the other.
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // Differences in clock speed on the primary and reverse streams can impact
+ // the AEC performance. On the client-side, this could be seen when different
+ // render and capture devices are used, particularly with webcams.
+ //
+ // This enables a compensation mechanism, and requires that
+ // |set_device_sample_rate_hz()| and |set_stream_drift_samples()| be called.
+ virtual int enable_drift_compensation(bool enable) = 0;
+ virtual bool is_drift_compensation_enabled() const = 0;
+
+ // Provides the sampling rate of the audio devices. It is assumed the render
+ // and capture devices use the same nominal sample rate. Required if and only
+ // if drift compensation is enabled.
+ virtual int set_device_sample_rate_hz(int rate) = 0;
+ virtual int device_sample_rate_hz() const = 0;
+
+ // Sets the difference between the number of samples rendered and captured by
+ // the audio devices since the last call to |ProcessStream()|. Must be called
+ // if and only if drift compensation is enabled, prior to |ProcessStream()|.
+ virtual int set_stream_drift_samples(int drift) = 0;
+ virtual int stream_drift_samples() const = 0;
+
+ enum SuppressionLevel {
+ kLowSuppression,
+ kModerateSuppression,
+ kHighSuppression
+ };
+
+ // Sets the aggressiveness of the suppressor. A higher level trades off
+ // double-talk performance for increased echo suppression.
+ virtual int set_suppression_level(SuppressionLevel level) = 0;
+ virtual SuppressionLevel suppression_level() const = 0;
+
+ // Returns false if the current frame almost certainly contains no echo
+ // and true if it _might_ contain echo.
+ virtual bool stream_has_echo() const = 0;
+
+ // Enables the computation of various echo metrics. These are obtained
+ // through |GetMetrics()|.
+ virtual int enable_metrics(bool enable) = 0;
+ virtual bool are_metrics_enabled() const = 0;
+
+ // Each statistic is reported in dB.
+ // P_far: Far-end (render) signal power.
+ // P_echo: Near-end (capture) echo signal power.
+ // P_out: Signal power at the output of the AEC.
+ // P_a: Internal signal power at the point before the AEC's non-linear
+ // processor.
+ struct Metrics {
+ // RERL = ERL + ERLE
+ AudioProcessing::Statistic residual_echo_return_loss;
+
+ // ERL = 10log_10(P_far / P_echo)
+ AudioProcessing::Statistic echo_return_loss;
+
+ // ERLE = 10log_10(P_echo / P_out)
+ AudioProcessing::Statistic echo_return_loss_enhancement;
+
+ // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
+ AudioProcessing::Statistic a_nlp;
+ };
+
+ // TODO(ajm): discuss the metrics update period.
+ virtual int GetMetrics(Metrics* metrics) = 0;
+
+ // Enables computation and logging of delay values. Statistics are obtained
+ // through |GetDelayMetrics()|.
+ virtual int enable_delay_logging(bool enable) = 0;
+ virtual bool is_delay_logging_enabled() const = 0;
+
+ // The delay metrics consists of the delay |median| and the delay standard
+ // deviation |std|. The values are averaged over the time period since the
+ // last call to |GetDelayMetrics()|.
+ virtual int GetDelayMetrics(int* median, int* std) = 0;
+
+ protected:
+ virtual ~EchoCancellation() {};
+};
+
+// The acoustic echo control for mobile (AECM) component is a low complexity
+// robust option intended for use on mobile devices.
+//
+// Not recommended to be enabled on the server-side.
+class EchoControlMobile {
+ public:
+ // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
+ // Enabling one will disable the other.
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // Recommended settings for particular audio routes. In general, the louder
+ // the echo is expected to be, the higher this value should be set. The
+ // preferred setting may vary from device to device.
+ enum RoutingMode {
+ kQuietEarpieceOrHeadset,
+ kEarpiece,
+ kLoudEarpiece,
+ kSpeakerphone,
+ kLoudSpeakerphone
+ };
+
+ // Sets echo control appropriate for the audio routing |mode| on the device.
+ // It can and should be updated during a call if the audio routing changes.
+ virtual int set_routing_mode(RoutingMode mode) = 0;
+ virtual RoutingMode routing_mode() const = 0;
+
+ // Comfort noise replaces suppressed background noise to maintain a
+ // consistent signal level.
+ virtual int enable_comfort_noise(bool enable) = 0;
+ virtual bool is_comfort_noise_enabled() const = 0;
+
+ // A typical use case is to initialize the component with an echo path from a
+ // previous call. The echo path is retrieved using |GetEchoPath()|, typically
+ // at the end of a call. The data can then be stored for later use as an
+ // initializer before the next call, using |SetEchoPath()|.
+ //
+ // Controlling the echo path this way requires the data |size_bytes| to match
+ // the internal echo path size. This size can be acquired using
+ // |echo_path_size_bytes()|. |SetEchoPath()| causes an entire reset, worth
+ // noting if it is to be called during an ongoing call.
+ //
+ // It is possible that version incompatibilities may result in a stored echo
+ // path of the incorrect size. In this case, the stored path should be
+ // discarded.
+ virtual int SetEchoPath(const void* echo_path, size_t size_bytes) = 0;
+ virtual int GetEchoPath(void* echo_path, size_t size_bytes) const = 0;
+
+ // The returned path size is guaranteed not to change for the lifetime of
+ // the application.
+ static size_t echo_path_size_bytes();
+
+ protected:
+ virtual ~EchoControlMobile() {};
+};
+
+// The automatic gain control (AGC) component brings the signal to an
+// appropriate range. This is done by applying a digital gain directly and, in
+// the analog mode, prescribing an analog gain to be applied at the audio HAL.
+//
+// Recommended to be enabled on the client-side.
+class GainControl {
+ public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // When an analog mode is set, this must be called prior to |ProcessStream()|
+ // to pass the current analog level from the audio HAL. Must be within the
+ // range provided to |set_analog_level_limits()|.
+ virtual int set_stream_analog_level(int level) = 0;
+
+ // When an analog mode is set, this should be called after |ProcessStream()|
+ // to obtain the recommended new analog level for the audio HAL. It is the
+ // users responsibility to apply this level.
+ virtual int stream_analog_level() = 0;
+
+ enum Mode {
+ // Adaptive mode intended for use if an analog volume control is available
+ // on the capture device. It will require the user to provide coupling
+ // between the OS mixer controls and AGC through the |stream_analog_level()|
+ // functions.
+ //
+ // It consists of an analog gain prescription for the audio device and a
+ // digital compression stage.
+ kAdaptiveAnalog,
+
+ // Adaptive mode intended for situations in which an analog volume control
+ // is unavailable. It operates in a similar fashion to the adaptive analog
+ // mode, but with scaling instead applied in the digital domain. As with
+ // the analog mode, it additionally uses a digital compression stage.
+ kAdaptiveDigital,
+
+ // Fixed mode which enables only the digital compression stage also used by
+ // the two adaptive modes.
+ //
+ // It is distinguished from the adaptive modes by considering only a
+ // short time-window of the input signal. It applies a fixed gain through
+ // most of the input level range, and compresses (gradually reduces gain
+ // with increasing level) the input signal at higher levels. This mode is
+ // preferred on embedded devices where the capture signal level is
+ // predictable, so that a known gain can be applied.
+ kFixedDigital
+ };
+
+ virtual int set_mode(Mode mode) = 0;
+ virtual Mode mode() const = 0;
+
+ // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
+ // from digital full-scale). The convention is to use positive values. For
+ // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
+ // level 3 dB below full-scale. Limited to [0, 31].
+ //
+ // TODO(ajm): use a negative value here instead, if/when VoE will similarly
+ // update its interface.
+ virtual int set_target_level_dbfs(int level) = 0;
+ virtual int target_level_dbfs() const = 0;
+
+ // Sets the maximum |gain| the digital compression stage may apply, in dB. A
+ // higher number corresponds to greater compression, while a value of 0 will
+ // leave the signal uncompressed. Limited to [0, 90].
+ virtual int set_compression_gain_db(int gain) = 0;
+ virtual int compression_gain_db() const = 0;
+
+ // When enabled, the compression stage will hard limit the signal to the
+ // target level. Otherwise, the signal will be compressed but not limited
+ // above the target level.
+ virtual int enable_limiter(bool enable) = 0;
+ virtual bool is_limiter_enabled() const = 0;
+
+ // Sets the |minimum| and |maximum| analog levels of the audio capture device.
+ // Must be set if and only if an analog mode is used. Limited to [0, 65535].
+ virtual int set_analog_level_limits(int minimum,
+ int maximum) = 0;
+ virtual int analog_level_minimum() const = 0;
+ virtual int analog_level_maximum() const = 0;
+
+ // Returns true if the AGC has detected a saturation event (period where the
+ // signal reaches digital full-scale) in the current frame and the analog
+ // level cannot be reduced.
+ //
+ // This could be used as an indicator to reduce or disable analog mic gain at
+ // the audio HAL.
+ virtual bool stream_is_saturated() const = 0;
+
+ protected:
+ virtual ~GainControl() {};
+};
+
+// A filtering component which removes DC offset and low-frequency noise.
+// Recommended to be enabled on the client-side.
+class HighPassFilter {
+ public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ protected:
+ virtual ~HighPassFilter() {};
+};
+
+// An estimation component used to retrieve level metrics.
+class LevelEstimator {
+ public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // Returns the root mean square (RMS) level in dBFs (decibels from digital
+ // full-scale), or alternately dBov. It is computed over all primary stream
+ // frames since the last call to RMS(). The returned value is positive but
+ // should be interpreted as negative. It is constrained to [0, 127].
+ //
+ // The computation follows:
+ // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-05
+ // with the intent that it can provide the RTP audio level indication.
+ //
+ // Frames passed to ProcessStream() with an |_energy| of zero are considered
+ // to have been muted. The RMS of the frame will be interpreted as -127.
+ virtual int RMS() = 0;
+
+ protected:
+ virtual ~LevelEstimator() {};
+};
+
+// The noise suppression (NS) component attempts to remove noise while
+// retaining speech. Recommended to be enabled on the client-side.
+//
+// Recommended to be enabled on the client-side.
+class NoiseSuppression {
+ public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // Determines the aggressiveness of the suppression. Increasing the level
+ // will reduce the noise level at the expense of a higher speech distortion.
+ enum Level {
+ kLow,
+ kModerate,
+ kHigh,
+ kVeryHigh
+ };
+
+ virtual int set_level(Level level) = 0;
+ virtual Level level() const = 0;
+
+ protected:
+ virtual ~NoiseSuppression() {};
+};
+
+// The voice activity detection (VAD) component analyzes the stream to
+// determine if voice is present. A facility is also provided to pass in an
+// external VAD decision.
+//
+// In addition to |stream_has_voice()| the VAD decision is provided through the
+// |AudioFrame| passed to |ProcessStream()|. The |_vadActivity| member will be
+// modified to reflect the current decision.
+class VoiceDetection {
+ public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
+ // Returns true if voice is detected in the current frame. Should be called
+ // after |ProcessStream()|.
+ virtual bool stream_has_voice() const = 0;
+
+ // Some of the APM functionality requires a VAD decision. In the case that
+ // a decision is externally available for the current frame, it can be passed
+ // in here, before |ProcessStream()| is called.
+ //
+ // VoiceDetection does _not_ need to be enabled to use this. If it happens to
+ // be enabled, detection will be skipped for any frame in which an external
+ // VAD decision is provided.
+ virtual int set_stream_has_voice(bool has_voice) = 0;
+
+ // Specifies the likelihood that a frame will be declared to contain voice.
+ // A higher value makes it more likely that speech will not be clipped, at
+ // the expense of more noise being detected as voice.
+ enum Likelihood {
+ kVeryLowLikelihood,
+ kLowLikelihood,
+ kModerateLikelihood,
+ kHighLikelihood
+ };
+
+ virtual int set_likelihood(Likelihood likelihood) = 0;
+ virtual Likelihood likelihood() const = 0;
+
+ // Sets the |size| of the frames in ms on which the VAD will operate. Larger
+ // frames will improve detection accuracy, but reduce the frequency of
+ // updates.
+ //
+ // This does not impact the size of frames passed to |ProcessStream()|.
+ virtual int set_frame_size_ms(int size) = 0;
+ virtual int frame_size_ms() const = 0;
+
+ protected:
+ virtual ~VoiceDetection() {};
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
diff --git a/trunk/src/modules/audio_processing/level_estimator_impl.cc b/trunk/src/modules/audio_processing/level_estimator_impl.cc
new file mode 100644
index 0000000..42cac99
--- /dev/null
+++ b/trunk/src/modules/audio_processing/level_estimator_impl.cc
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "level_estimator_impl.h"
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+namespace {
+
+const double kMaxSquaredLevel = 32768.0 * 32768.0;
+
+class Level {
+ public:
+ static const int kMinLevel = 127;
+
+ Level()
+ : sum_square_(0.0),
+ sample_count_(0) {}
+ ~Level() {}
+
+ void Init() {
+ sum_square_ = 0.0;
+ sample_count_ = 0;
+ }
+
+ void Process(int16_t* data, int length) {
+ assert(data != NULL);
+ assert(length > 0);
+ sum_square_ += SumSquare(data, length);
+ sample_count_ += length;
+ }
+
+ void ProcessMuted(int length) {
+ assert(length > 0);
+ sample_count_ += length;
+ }
+
+ int RMS() {
+ if (sample_count_ == 0 || sum_square_ == 0.0) {
+ Init();
+ return kMinLevel;
+ }
+
+ // Normalize by the max level.
+ double rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
+ // 20log_10(x^0.5) = 10log_10(x)
+ rms = 10 * log10(rms);
+ if (rms > 0)
+ rms = 0;
+ else if (rms < -kMinLevel)
+ rms = -kMinLevel;
+
+ rms = -rms;
+ Init();
+ return static_cast<int>(rms + 0.5);
+ }
+
+ private:
+ static double SumSquare(int16_t* data, int length) {
+ double sum_square = 0.0;
+ for (int i = 0; i < length; ++i) {
+ double data_d = static_cast<double>(data[i]);
+ sum_square += data_d * data_d;
+ }
+ return sum_square;
+ }
+
+ double sum_square_;
+ int sample_count_;
+};
+} // namespace
+
+LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm) {}
+
+LevelEstimatorImpl::~LevelEstimatorImpl() {}
+
+int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ Level* level = static_cast<Level*>(handle(0));
+ if (audio->is_muted()) {
+ level->ProcessMuted(audio->samples_per_channel());
+ return apm_->kNoError;
+ }
+
+ int16_t* mixed_data = audio->data(0);
+ if (audio->num_channels() > 1) {
+ audio->CopyAndMix(1);
+ mixed_data = audio->mixed_data(0);
+ }
+
+ level->Process(mixed_data, audio->samples_per_channel());
+
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool LevelEstimatorImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int LevelEstimatorImpl::RMS() {
+ if (!is_component_enabled()) {
+ return apm_->kNotEnabledError;
+ }
+
+ Level* level = static_cast<Level*>(handle(0));
+ return level->RMS();
+}
+
+void* LevelEstimatorImpl::CreateHandle() const {
+ return new Level;
+}
+
+int LevelEstimatorImpl::DestroyHandle(void* handle) const {
+ assert(handle != NULL);
+ Level* level = static_cast<Level*>(handle);
+ delete level;
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::InitializeHandle(void* handle) const {
+ assert(handle != NULL);
+ Level* level = static_cast<Level*>(handle);
+ level->Init();
+
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::num_handles_required() const {
+ return 1;
+}
+
+int LevelEstimatorImpl::GetHandleError(void* handle) const {
+ // The component has no detailed errors.
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/level_estimator_impl.h b/trunk/src/modules/audio_processing/level_estimator_impl.h
new file mode 100644
index 0000000..1a06343
--- /dev/null
+++ b/trunk/src/modules/audio_processing/level_estimator_impl.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class LevelEstimatorImpl : public LevelEstimator,
+ public ProcessingComponent {
+ public:
+ explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
+ virtual ~LevelEstimatorImpl();
+
+ int ProcessStream(AudioBuffer* audio);
+
+ // LevelEstimator implementation.
+ virtual bool is_enabled() const;
+
+ private:
+ // LevelEstimator implementation.
+ virtual int Enable(bool enable);
+ virtual int RMS();
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/noise_suppression_impl.cc b/trunk/src/modules/audio_processing/noise_suppression_impl.cc
new file mode 100644
index 0000000..c44d3fe
--- /dev/null
+++ b/trunk/src/modules/audio_processing/noise_suppression_impl.cc
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "noise_suppression_impl.h"
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#if defined(WEBRTC_NS_FLOAT)
+#include "noise_suppression.h"
+#elif defined(WEBRTC_NS_FIXED)
+#include "noise_suppression_x.h"
+#endif
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_NS_FLOAT)
+typedef NsHandle Handle;
+#elif defined(WEBRTC_NS_FIXED)
+typedef NsxHandle Handle;
+#endif
+
+namespace {
+int MapSetting(NoiseSuppression::Level level) {
+ switch (level) {
+ case NoiseSuppression::kLow:
+ return 0;
+ case NoiseSuppression::kModerate:
+ return 1;
+ case NoiseSuppression::kHigh:
+ return 2;
+ case NoiseSuppression::kVeryHigh:
+ return 3;
+ }
+ assert(false);
+ return -1;
+}
+} // namespace
+
+NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm),
+ level_(kModerate) {}
+
+NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
+
+int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ int err = apm_->kNoError;
+
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+ assert(audio->samples_per_split_channel() <= 160);
+ assert(audio->num_channels() == num_handles());
+
+ for (int i = 0; i < num_handles(); i++) {
+ Handle* my_handle = static_cast<Handle*>(handle(i));
+#if defined(WEBRTC_NS_FLOAT)
+ err = WebRtcNs_Process(static_cast<Handle*>(handle(i)),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i));
+#elif defined(WEBRTC_NS_FIXED)
+ err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i),
+ audio->low_pass_split_data(i),
+ audio->high_pass_split_data(i));
+#endif
+
+ if (err != apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+ }
+
+ return apm_->kNoError;
+}
+
+int NoiseSuppressionImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool NoiseSuppressionImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int NoiseSuppressionImpl::set_level(Level level) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (MapSetting(level) == -1) {
+ return apm_->kBadParameterError;
+ }
+
+ level_ = level;
+ return Configure();
+}
+
+NoiseSuppression::Level NoiseSuppressionImpl::level() const {
+ return level_;
+}
+
+void* NoiseSuppressionImpl::CreateHandle() const {
+ Handle* handle = NULL;
+#if defined(WEBRTC_NS_FLOAT)
+ if (WebRtcNs_Create(&handle) != apm_->kNoError)
+#elif defined(WEBRTC_NS_FIXED)
+ if (WebRtcNsx_Create(&handle) != apm_->kNoError)
+#endif
+ {
+ handle = NULL;
+ } else {
+ assert(handle != NULL);
+ }
+
+ return handle;
+}
+
+int NoiseSuppressionImpl::DestroyHandle(void* handle) const {
+#if defined(WEBRTC_NS_FLOAT)
+ return WebRtcNs_Free(static_cast<Handle*>(handle));
+#elif defined(WEBRTC_NS_FIXED)
+ return WebRtcNsx_Free(static_cast<Handle*>(handle));
+#endif
+}
+
+int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
+#if defined(WEBRTC_NS_FLOAT)
+ return WebRtcNs_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
+#elif defined(WEBRTC_NS_FIXED)
+ return WebRtcNsx_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
+#endif
+}
+
+int NoiseSuppressionImpl::ConfigureHandle(void* handle) const {
+#if defined(WEBRTC_NS_FLOAT)
+ return WebRtcNs_set_policy(static_cast<Handle*>(handle),
+ MapSetting(level_));
+#elif defined(WEBRTC_NS_FIXED)
+ return WebRtcNsx_set_policy(static_cast<Handle*>(handle),
+ MapSetting(level_));
+#endif
+}
+
+int NoiseSuppressionImpl::num_handles_required() const {
+ return apm_->num_output_channels();
+}
+
+int NoiseSuppressionImpl::GetHandleError(void* handle) const {
+ // The NS has no get_error() function.
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
+
diff --git a/trunk/src/modules/audio_processing/noise_suppression_impl.h b/trunk/src/modules/audio_processing/noise_suppression_impl.h
new file mode 100644
index 0000000..7b65b70
--- /dev/null
+++ b/trunk/src/modules/audio_processing/noise_suppression_impl.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class NoiseSuppressionImpl : public NoiseSuppression,
+ public ProcessingComponent {
+ public:
+ explicit NoiseSuppressionImpl(const AudioProcessingImpl* apm);
+ virtual ~NoiseSuppressionImpl();
+
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // NoiseSuppression implementation.
+ virtual bool is_enabled() const;
+
+ private:
+ // NoiseSuppression implementation.
+ virtual int Enable(bool enable);
+ virtual int set_level(Level level);
+ virtual Level level() const;
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+ Level level_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/ns/Android.mk b/trunk/src/modules/audio_processing/ns/Android.mk
new file mode 100644
index 0000000..5e442f6
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/Android.mk
@@ -0,0 +1,79 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+#############################
+# Build the non-neon library.
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_ns
+LOCAL_MODULE_TAGS := optional
+LOCAL_GENERATED_SOURCES :=
+LOCAL_SRC_FILES := \
+ noise_suppression_x.c \
+ nsx_core.c
+
+# Files for floating point.
+# noise_suppression.c ns_core.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#############################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_ns_neon
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := nsx_core_neon.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ -mfpu=neon \
+ -mfloat-abi=softfp \
+ -flax-vector-conversions
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/trunk/src/modules/audio_processing/ns/defines.h b/trunk/src/modules/audio_processing/ns/defines.h
new file mode 100644
index 0000000..d253967
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/defines.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
+
+//#define PROCESS_FLOW_0 // Use the traditional method.
+//#define PROCESS_FLOW_1 // Use traditional with DD estimate of prior SNR.
+#define PROCESS_FLOW_2 // Use the new method of speech/noise classification.
+
+#define BLOCKL_MAX 160 // max processing block length: 160
+#define ANAL_BLOCKL_MAX 256 // max analysis block length: 256
+#define HALF_ANAL_BLOCKL 129 // half max analysis block length + 1
+
+#define QUANTILE (float)0.25
+
+#define SIMULT 3
+#define END_STARTUP_LONG 200
+#define END_STARTUP_SHORT 50
+#define FACTOR (float)40.0
+#define WIDTH (float)0.01
+
+#define SMOOTH (float)0.75 // filter smoothing
+// Length of fft work arrays.
+#define IP_LENGTH (ANAL_BLOCKL_MAX >> 1) // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
+#define W_LENGTH (ANAL_BLOCKL_MAX >> 1)
+
+//PARAMETERS FOR NEW METHOD
+#define DD_PR_SNR (float)0.98 // DD update of prior SNR
+#define LRT_TAVG (float)0.50 // tavg parameter for LRT (previously 0.90)
+#define SPECT_FL_TAVG (float)0.30 // tavg parameter for spectral flatness measure
+#define SPECT_DIFF_TAVG (float)0.30 // tavg parameter for spectral difference measure
+#define PRIOR_UPDATE (float)0.10 // update parameter of prior model
+#define NOISE_UPDATE (float)0.90 // update parameter for noise
+#define SPEECH_UPDATE (float)0.99 // update parameter when likely speech
+#define WIDTH_PR_MAP (float)4.0 // width parameter in sigmoid map for prior model
+#define LRT_FEATURE_THR (float)0.5 // default threshold for LRT feature
+#define SF_FEATURE_THR (float)0.5 // default threshold for Spectral Flatness feature
+#define SD_FEATURE_THR (float)0.5 // default threshold for Spectral Difference feature
+#define PROB_RANGE (float)0.20 // probability threshold for noise state in
+ // speech/noise likelihood
+#define HIST_PAR_EST 1000 // histogram size for estimation of parameters
+#define GAMMA_PAUSE (float)0.05 // update for conservative noise estimate
+//
+#define B_LIM (float)0.5 // threshold in final energy gain factor calculation
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
diff --git a/trunk/src/modules/audio_processing/ns/include/noise_suppression.h b/trunk/src/modules/audio_processing/ns/include/noise_suppression.h
new file mode 100644
index 0000000..1f498c1
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/include/noise_suppression.h
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+
+#include "typedefs.h"
+
+typedef struct NsHandleT NsHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ * - NS_inst : Pointer to noise reduction instance that should be
+ * created
+ *
+ * Output:
+ * - NS_inst : Pointer to created noise reduction instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_Create(NsHandle** NS_inst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Reduction
+ * instance.
+ *
+ * Input:
+ * - NS_inst : Pointer to NS instance that should be freed
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_Free(NsHandle* NS_inst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ * - NS_inst : Instance that should be initialized
+ * - fs : sampling frequency
+ *
+ * Output:
+ * - NS_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - NS_inst : Instance that should be initialized
+ * - mode : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ * - NS_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_set_policy(NsHandle* NS_inst, int mode);
+
+
+/*
+ * This functions does Noise Suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ * - NS_inst : NS Instance. Needs to be initiated before call.
+ * - spframe : Pointer to speech frame buffer for L band
+ * - spframe_H : Pointer to speech frame buffer for H band
+ * - fs : sampling frequency
+ *
+ * Output:
+ * - NS_inst : Updated NS instance
+ * - outframe : Pointer to output frame for L band
+ * - outframe_H : Pointer to output frame for H band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+int WebRtcNs_Process(NsHandle* NS_inst,
+ short* spframe,
+ short* spframe_H,
+ short* outframe,
+ short* outframe_H);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
diff --git a/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h b/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h
new file mode 100644
index 0000000..b6eef90
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+
+#include "typedefs.h"
+
+typedef struct NsxHandleT NsxHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ * - nsxInst : Pointer to noise reduction instance that should be
+ * created
+ *
+ * Output:
+ * - nsxInst : Pointer to created noise reduction instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_Create(NsxHandle** nsxInst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Suppression
+ * instance.
+ *
+ * Input:
+ * - nsxInst : Pointer to NS instance that should be freed
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_Free(NsxHandle* nsxInst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ * - nsxInst : Instance that should be initialized
+ * - fs : sampling frequency
+ *
+ * Output:
+ * - nsxInst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - nsxInst : Instance that should be initialized
+ * - mode : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ * - nsxInst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode);
+
+/*
+ * This functions does noise suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ * - nsxInst : NSx instance. Needs to be initiated before call.
+ * - speechFrame : Pointer to speech frame buffer for L band
+ * - speechFrameHB : Pointer to speech frame buffer for H band
+ * - fs : sampling frequency
+ *
+ * Output:
+ * - nsxInst : Updated NSx instance
+ * - outFrame : Pointer to output frame for L band
+ * - outFrameHB : Pointer to output frame for H band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+int WebRtcNsx_Process(NsxHandle* nsxInst,
+ short* speechFrame,
+ short* speechFrameHB,
+ short* outFrame,
+ short* outFrameHB);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
diff --git a/trunk/src/modules/audio_processing/ns/noise_suppression.c b/trunk/src/modules/audio_processing/ns/noise_suppression.c
new file mode 100644
index 0000000..a1e5ae8
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/noise_suppression.c
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "noise_suppression.h"
+#include "ns_core.h"
+#include "defines.h"
+
+int WebRtcNs_Create(NsHandle** NS_inst) {
+ *NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
+ if (*NS_inst != NULL) {
+ (*(NSinst_t**)NS_inst)->initFlag = 0;
+ return 0;
+ } else {
+ return -1;
+ }
+
+}
+
+int WebRtcNs_Free(NsHandle* NS_inst) {
+ free(NS_inst);
+ return 0;
+}
+
+
+int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs) {
+ return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
+}
+
+int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
+ return WebRtcNs_set_policy_core((NSinst_t*) NS_inst, mode);
+}
+
+
+int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H,
+ short* outframe, short* outframe_H) {
+ return WebRtcNs_ProcessCore(
+ (NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
+}
diff --git a/trunk/src/modules/audio_processing/ns/noise_suppression_x.c b/trunk/src/modules/audio_processing/ns/noise_suppression_x.c
new file mode 100644
index 0000000..6d27d0e
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/noise_suppression_x.c
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "noise_suppression_x.h"
+#include "nsx_core.h"
+#include "nsx_defines.h"
+
+int WebRtcNsx_Create(NsxHandle** nsxInst) {
+ *nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
+ if (*nsxInst != NULL) {
+ (*(NsxInst_t**)nsxInst)->initFlag = 0;
+ return 0;
+ } else {
+ return -1;
+ }
+
+}
+
+int WebRtcNsx_Free(NsxHandle* nsxInst) {
+ free(nsxInst);
+ return 0;
+}
+
+int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs) {
+ return WebRtcNsx_InitCore((NsxInst_t*)nsxInst, fs);
+}
+
+int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode) {
+ return WebRtcNsx_set_policy_core((NsxInst_t*)nsxInst, mode);
+}
+
+int WebRtcNsx_Process(NsxHandle* nsxInst, short* speechFrame,
+ short* speechFrameHB, short* outFrame,
+ short* outFrameHB) {
+ return WebRtcNsx_ProcessCore(
+ (NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame, outFrameHB);
+}
+
diff --git a/trunk/src/modules/audio_processing/ns/ns.gypi b/trunk/src/modules/audio_processing/ns/ns.gypi
new file mode 100644
index 0000000..147f7ea
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/ns.gypi
@@ -0,0 +1,59 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'ns',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ 'apm_util'
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/noise_suppression.h',
+ 'noise_suppression.c',
+ 'windows_private.h',
+ 'defines.h',
+ 'ns_core.c',
+ 'ns_core.h',
+ ],
+ },
+ {
+ 'target_name': 'ns_fix',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/noise_suppression_x.h',
+ 'noise_suppression_x.c',
+ 'nsx_defines.h',
+ 'nsx_core.c',
+ 'nsx_core.h',
+ ],
+ },
+ ],
+}
diff --git a/trunk/src/modules/audio_processing/ns/ns_core.c b/trunk/src/modules/audio_processing/ns/ns_core.c
new file mode 100644
index 0000000..e5aa4ff
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/ns_core.c
@@ -0,0 +1,1303 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+#include <math.h>
+//#include <stdio.h>
+#include <stdlib.h>
+#include "noise_suppression.h"
+#include "ns_core.h"
+#include "windows_private.h"
+#include "fft4g.h"
+#include "signal_processing_library.h"
+
+// Set Feature Extraction Parameters
+void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) {
+ //bin size of histogram
+ inst->featureExtractionParams.binSizeLrt = (float)0.1;
+ inst->featureExtractionParams.binSizeSpecFlat = (float)0.05;
+ inst->featureExtractionParams.binSizeSpecDiff = (float)0.1;
+
+ //range of histogram over which lrt threshold is computed
+ inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0;
+
+ //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
+ // thresholds for prior model
+ inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff
+ inst->featureExtractionParams.factor2ModelPars = (float)0.9; //for spectral_flatness:
+ // used when noise is flatter than speech
+
+ //peak limit for spectral flatness (varies between 0 and 1)
+ inst->featureExtractionParams.thresPosSpecFlat = (float)0.6;
+
+ //limit on spacing of two highest peaks in histogram: spacing determined by bin size
+ inst->featureExtractionParams.limitPeakSpacingSpecFlat =
+ 2 * inst->featureExtractionParams.binSizeSpecFlat;
+ inst->featureExtractionParams.limitPeakSpacingSpecDiff =
+ 2 * inst->featureExtractionParams.binSizeSpecDiff;
+
+ //limit on relevance of second peak:
+ inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5;
+ inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5;
+
+ // fluctuation limit of lrt feature
+ inst->featureExtractionParams.thresFluctLrt = (float)0.05;
+
+ //limit on the max and min values for the feature thresholds
+ inst->featureExtractionParams.maxLrt = (float)1.0;
+ inst->featureExtractionParams.minLrt = (float)0.20;
+
+ inst->featureExtractionParams.maxSpecFlat = (float)0.95;
+ inst->featureExtractionParams.minSpecFlat = (float)0.10;
+
+ inst->featureExtractionParams.maxSpecDiff = (float)1.0;
+ inst->featureExtractionParams.minSpecDiff = (float)0.16;
+
+ //criteria of weight of histogram peak to accept/reject feature
+ inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3
+ * (inst->modelUpdatePars[1])); //for spectral flatness
+ inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3
+ * (inst->modelUpdatePars[1])); //for spectral difference
+}
+
+// Initialize state
+int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs) {
+ int i;
+ //We only support 10ms frames
+
+ //check for valid pointer
+ if (inst == NULL) {
+ return -1;
+ }
+
+ // Initialization of struct
+ if (fs == 8000 || fs == 16000 || fs == 32000) {
+ inst->fs = fs;
+ } else {
+ return -1;
+ }
+ inst->windShift = 0;
+ if (fs == 8000) {
+ // We only support 10ms frames
+ inst->blockLen = 80;
+ inst->blockLen10ms = 80;
+ inst->anaLen = 128;
+ inst->window = kBlocks80w128;
+ inst->outLen = 0;
+ } else if (fs == 16000) {
+ // We only support 10ms frames
+ inst->blockLen = 160;
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->window = kBlocks160w256;
+ inst->outLen = 0;
+ } else if (fs == 32000) {
+ // We only support 10ms frames
+ inst->blockLen = 160;
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->window = kBlocks160w256;
+ inst->outLen = 0;
+ }
+ inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
+
+ // Initialize fft work arrays.
+ inst->ip[0] = 0; // Setting this triggers initialization.
+ memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ WebRtc_rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft);
+
+ memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+
+ //for HB processing
+ memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+
+ //for quantile noise estimation
+ memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
+ inst->lquantile[i] = (float)8.0;
+ inst->density[i] = (float)0.3;
+ }
+
+ for (i = 0; i < SIMULT; i++) {
+ inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
+ }
+
+ inst->updates = 0;
+
+ // Wiener filter initialization
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->smooth[i] = (float)1.0;
+ }
+
+ // Set the aggressiveness: default
+ inst->aggrMode = 0;
+
+ //initialize variables for new method
+ inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->magnPrev[i] = (float)0.0; //previous mag spectrum
+ inst->noisePrev[i] = (float)0.0; //previous noise-spectrum
+ inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold)
+ inst->magnAvgPause[i] = (float)0.0; //conservative noise spectrum estimate
+ inst->speechProbHB[i] = (float)0.0; //for estimation of HB in second pass
+ inst->initMagnEst[i] = (float)0.0; //initial average mag spectrum
+ }
+
+ //feature quantities
+ inst->featureData[0] = SF_FEATURE_THR; //spectral flatness (start on threshold)
+ inst->featureData[1] = (float)0.0; //spectral entropy: not used in this version
+ inst->featureData[2] = (float)0.0; //spectral variance: not used in this version
+ inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold)
+ inst->featureData[4] = SF_FEATURE_THR; //spectral template diff (start on threshold)
+ inst->featureData[5] = (float)0.0; //normalization for spectral-diff
+ inst->featureData[6] = (float)0.0; //window time-average of input magnitude spectrum
+
+ //histogram quantities: used to estimate/update thresholds for features
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ inst->histLrt[i] = 0;
+ inst->histSpecFlat[i] = 0;
+ inst->histSpecDiff[i] = 0;
+ }
+
+ inst->blockInd = -1; //frame counter
+ inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature
+ inst->priorModelPars[1] = (float)0.5; //threshold for spectral flatness:
+ // determined on-line
+ inst->priorModelPars[2] = (float)1.0; //sgn_map par for spectral measure:
+ // 1 for flatness measure
+ inst->priorModelPars[3] = (float)0.5; //threshold for template-difference feature:
+ // determined on-line
+ inst->priorModelPars[4] = (float)1.0; //default weighting parameter for lrt feature
+ inst->priorModelPars[5] = (float)0.0; //default weighting parameter for
+ // spectral flatness feature
+ inst->priorModelPars[6] = (float)0.0; //default weighting parameter for
+ // spectral difference feature
+
+ inst->modelUpdatePars[0] = 2; //update flag for parameters:
+ // 0 no update, 1=update once, 2=update every window
+ inst->modelUpdatePars[1] = 500; //window for update
+ inst->modelUpdatePars[2] = 0; //counter for update of conservative noise spectrum
+ //counter if the feature thresholds are updated during the sequence
+ inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
+
+ inst->signalEnergy = 0.0;
+ inst->sumMagn = 0.0;
+ inst->whiteNoiseLevel = 0.0;
+ inst->pinkNoiseNumerator = 0.0;
+ inst->pinkNoiseExp = 0.0;
+
+ WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration
+
+ //default mode
+ WebRtcNs_set_policy_core(inst, 0);
+
+
+ memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
+
+ inst->initFlag = 1;
+ return 0;
+}
+
+int WebRtcNs_set_policy_core(NSinst_t* inst, int mode) {
+ // allow for modes:0,1,2,3
+ if (mode < 0 || mode > 3) {
+ return (-1);
+ }
+
+ inst->aggrMode = mode;
+ if (mode == 0) {
+ inst->overdrive = (float)1.0;
+ inst->denoiseBound = (float)0.5;
+ inst->gainmap = 0;
+ } else if (mode == 1) {
+ //inst->overdrive = (float)1.25;
+ inst->overdrive = (float)1.0;
+ inst->denoiseBound = (float)0.25;
+ inst->gainmap = 1;
+ } else if (mode == 2) {
+ //inst->overdrive = (float)1.25;
+ inst->overdrive = (float)1.1;
+ inst->denoiseBound = (float)0.125;
+ inst->gainmap = 1;
+ } else if (mode == 3) {
+ //inst->overdrive = (float)1.30;
+ inst->overdrive = (float)1.25;
+ inst->denoiseBound = (float)0.09;
+ inst->gainmap = 1;
+ }
+ return 0;
+}
+
+// Estimate noise
+void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) {
+ int i, s, offset;
+ float lmagn[HALF_ANAL_BLOCKL], delta;
+
+ if (inst->updates < END_STARTUP_LONG) {
+ inst->updates++;
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ lmagn[i] = (float)log(magn[i]);
+ }
+
+ // loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // newquantest(...)
+ for (i = 0; i < inst->magnLen; i++) {
+ // compute delta
+ if (inst->density[offset + i] > 1.0) {
+ delta = FACTOR * (float)1.0 / inst->density[offset + i];
+ } else {
+ delta = FACTOR;
+ }
+
+ // update log quantile estimate
+ if (lmagn[i] > inst->lquantile[offset + i]) {
+ inst->lquantile[offset + i] += QUANTILE * delta
+ / (float)(inst->counter[s] + 1);
+ } else {
+ inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta
+ / (float)(inst->counter[s] + 1);
+ }
+
+ // update density estimate
+ if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH) {
+ inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
+ + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s] + 1);
+ }
+ } // end loop over magnitude spectrum
+
+ if (inst->counter[s] >= END_STARTUP_LONG) {
+ inst->counter[s] = 0;
+ if (inst->updates >= END_STARTUP_LONG) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
+ }
+ }
+ }
+
+ inst->counter[s]++;
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->updates < END_STARTUP_LONG) {
+ // Use the last "s" to get noise during startup that differ from zero.
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
+ }
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = inst->quantile[i];
+ }
+}
+
+// Extract thresholds for feature parameters
+// histograms are computed over some window_size (given by inst->modelUpdatePars[1])
+// thresholds and weights are extracted every window
+// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
+// threshold and weights are returned in: inst->priorModelPars
+void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) {
+ int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
+ int maxPeak1, maxPeak2;
+ int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff;
+
+ float binMid, featureSum;
+ float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
+ float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
+
+ //3 features: lrt, flatness, difference
+ //lrt_feature = inst->featureData[3];
+ //flat_feature = inst->featureData[0];
+ //diff_feature = inst->featureData[4];
+
+ //update histograms
+ if (flag == 0) {
+ // LRT
+ if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt)
+ && (inst->featureData[3] >= 0.0)) {
+ i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt);
+ inst->histLrt[i]++;
+ }
+ // Spectral flatness
+ if ((inst->featureData[0] < HIST_PAR_EST
+ * inst->featureExtractionParams.binSizeSpecFlat)
+ && (inst->featureData[0] >= 0.0)) {
+ i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat);
+ inst->histSpecFlat[i]++;
+ }
+ // Spectral difference
+ if ((inst->featureData[4] < HIST_PAR_EST
+ * inst->featureExtractionParams.binSizeSpecDiff)
+ && (inst->featureData[4] >= 0.0)) {
+ i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff);
+ inst->histSpecDiff[i]++;
+ }
+ }
+
+ // extract parameters for speech/noise probability
+ if (flag == 1) {
+ //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt
+ avgHistLrt = 0.0;
+ avgHistLrtCompl = 0.0;
+ avgSquareHistLrt = 0.0;
+ numHistLrt = 0;
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt;
+ if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt) {
+ avgHistLrt += inst->histLrt[i] * binMid;
+ numHistLrt += inst->histLrt[i];
+ }
+ avgSquareHistLrt += inst->histLrt[i] * binMid * binMid;
+ avgHistLrtCompl += inst->histLrt[i] * binMid;
+ }
+ if (numHistLrt > 0) {
+ avgHistLrt = avgHistLrt / ((float)numHistLrt);
+ }
+ avgHistLrtCompl = avgHistLrtCompl / ((float)inst->modelUpdatePars[1]);
+ avgSquareHistLrt = avgSquareHistLrt / ((float)inst->modelUpdatePars[1]);
+ fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
+ // get threshold for lrt feature:
+ if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
+ //very low fluct, so likely noise
+ inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
+ } else {
+ inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars
+ * avgHistLrt;
+ // check if value is within min/max range
+ if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt) {
+ inst->priorModelPars[0] = inst->featureExtractionParams.minLrt;
+ }
+ if (inst->priorModelPars[0] > inst->featureExtractionParams.maxLrt) {
+ inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
+ }
+ }
+ // done with lrt feature
+
+ //
+ // for spectral flatness and spectral difference: compute the main peaks of histogram
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecFlat = 0.0;
+ posPeak2SpecFlat = 0.0;
+ weightPeak1SpecFlat = 0;
+ weightPeak2SpecFlat = 0;
+
+ // peaks for flatness
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat;
+ if (inst->histSpecFlat[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecFlat = weightPeak1SpecFlat;
+ posPeak2SpecFlat = posPeak1SpecFlat;
+
+ maxPeak1 = inst->histSpecFlat[i];
+ weightPeak1SpecFlat = inst->histSpecFlat[i];
+ posPeak1SpecFlat = binMid;
+ } else if (inst->histSpecFlat[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecFlat[i];
+ weightPeak2SpecFlat = inst->histSpecFlat[i];
+ posPeak2SpecFlat = binMid;
+ }
+ }
+
+ //compute two peaks for spectral difference
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecDiff = 0.0;
+ posPeak2SpecDiff = 0.0;
+ weightPeak1SpecDiff = 0;
+ weightPeak2SpecDiff = 0;
+ // peaks for spectral difference
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff;
+ if (inst->histSpecDiff[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecDiff = weightPeak1SpecDiff;
+ posPeak2SpecDiff = posPeak1SpecDiff;
+
+ maxPeak1 = inst->histSpecDiff[i];
+ weightPeak1SpecDiff = inst->histSpecDiff[i];
+ posPeak1SpecDiff = binMid;
+ } else if (inst->histSpecDiff[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecDiff[i];
+ weightPeak2SpecDiff = inst->histSpecDiff[i];
+ posPeak2SpecDiff = binMid;
+ }
+ }
+
+ // for spectrum flatness feature
+ useFeatureSpecFlat = 1;
+ // merge the two peaks if they are close
+ if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat)
+ < inst->featureExtractionParams.limitPeakSpacingSpecFlat)
+ && (weightPeak2SpecFlat
+ > inst->featureExtractionParams.limitPeakWeightsSpecFlat
+ * weightPeak1SpecFlat)) {
+ weightPeak1SpecFlat += weightPeak2SpecFlat;
+ posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat);
+ }
+ //reject if weight of peaks is not large enough, or peak value too small
+ if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat
+ || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat) {
+ useFeatureSpecFlat = 0;
+ }
+ // if selected, get the threshold
+ if (useFeatureSpecFlat == 1) {
+ // compute the threshold
+ inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars
+ * posPeak1SpecFlat;
+ //check if value is within min/max range
+ if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat) {
+ inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat;
+ }
+ if (inst->priorModelPars[1] > inst->featureExtractionParams.maxSpecFlat) {
+ inst->priorModelPars[1] = inst->featureExtractionParams.maxSpecFlat;
+ }
+ }
+ // done with flatness feature
+
+ // for template feature
+ useFeatureSpecDiff = 1;
+ // merge the two peaks if they are close
+ if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff)
+ < inst->featureExtractionParams.limitPeakSpacingSpecDiff)
+ && (weightPeak2SpecDiff
+ > inst->featureExtractionParams.limitPeakWeightsSpecDiff
+ * weightPeak1SpecDiff)) {
+ weightPeak1SpecDiff += weightPeak2SpecDiff;
+ posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff);
+ }
+ // get the threshold value
+ inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars
+ * posPeak1SpecDiff;
+ //reject if weight of peaks is not large enough
+ if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff) {
+ useFeatureSpecDiff = 0;
+ }
+ //check if value is within min/max range
+ if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff) {
+ inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff;
+ }
+ if (inst->priorModelPars[3] > inst->featureExtractionParams.maxSpecDiff) {
+ inst->priorModelPars[3] = inst->featureExtractionParams.maxSpecDiff;
+ }
+ // done with spectral difference feature
+
+ // don't use template feature if fluctuation of lrt feature is very low:
+ // most likely just noise state
+ if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
+ useFeatureSpecDiff = 0;
+ }
+
+ // select the weights between the features
+ // inst->priorModelPars[4] is weight for lrt: always selected
+ // inst->priorModelPars[5] is weight for spectral flatness
+ // inst->priorModelPars[6] is weight for spectral difference
+ featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
+ inst->priorModelPars[4] = (float)1.0 / featureSum;
+ inst->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
+ inst->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
+
+ // set hists to zero for next update
+ if (inst->modelUpdatePars[0] >= 1) {
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ inst->histLrt[i] = 0;
+ inst->histSpecFlat[i] = 0;
+ inst->histSpecDiff[i] = 0;
+ }
+ }
+ } // end of flag == 1
+}
+
+// Compute spectral flatness on input spectrum
+// magnIn is the magnitude spectrum
+// spectral flatness is returned in inst->featureData[0]
+void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) {
+ int i;
+ int shiftLP = 1; //option to remove first bin(s) from spectral measures
+ float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
+
+ // comute spectral measures
+ // for flatness
+ avgSpectralFlatnessNum = 0.0;
+ avgSpectralFlatnessDen = inst->sumMagn;
+ for (i = 0; i < shiftLP; i++) {
+ avgSpectralFlatnessDen -= magnIn[i];
+ }
+ // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
+ for (i = shiftLP; i < inst->magnLen; i++) {
+ if (magnIn[i] > 0.0) {
+ avgSpectralFlatnessNum += (float)log(magnIn[i]);
+ } else {
+ inst->featureData[0] -= SPECT_FL_TAVG * inst->featureData[0];
+ return;
+ }
+ }
+ //normalize
+ avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen;
+ avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen;
+
+ //ratio and inverse log: check for case of log(0)
+ spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
+
+ //time-avg update of spectral flatness feature
+ inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]);
+ // done with flatness feature
+}
+
+// Compute the difference measure between input spectrum and a template/learned noise spectrum
+// magnIn is the input spectrum
+// the reference/template spectrum is inst->magnAvgPause[i]
+// returns (normalized) spectral difference in inst->featureData[4]
+void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) {
+ // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
+ int i;
+ float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
+
+ avgPause = 0.0;
+ avgMagn = inst->sumMagn;
+ // compute average quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ //conservative smooth noise spectrum from pause frames
+ avgPause += inst->magnAvgPause[i];
+ }
+ avgPause = avgPause / ((float)inst->magnLen);
+ avgMagn = avgMagn / ((float)inst->magnLen);
+
+ covMagnPause = 0.0;
+ varPause = 0.0;
+ varMagn = 0.0;
+ // compute variance and covariance quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause);
+ varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause);
+ varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
+ }
+ covMagnPause = covMagnPause / ((float)inst->magnLen);
+ varPause = varPause / ((float)inst->magnLen);
+ varMagn = varMagn / ((float)inst->magnLen);
+ // update of average magnitude spectrum
+ inst->featureData[6] += inst->signalEnergy;
+
+ avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001);
+ // normalize and compute time-avg update of difference feature
+ avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001));
+ inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]);
+}
+
+// Compute speech/noise probability
+// speech/noise probability is returned in: probSpeechFinal
+//magn is the input magnitude spectrum
+//noise is the noise spectrum
+//snrLocPrior is the prior snr for each freq.
+//snr loc_post is the post snr for each freq.
+void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snrLocPrior,
+ float* snrLocPost) {
+ int i, sgnMap;
+ float invLrt, gainPrior, indPrior;
+ float logLrtTimeAvgKsum, besselTmp;
+ float indicator0, indicator1, indicator2;
+ float tmpFloat1, tmpFloat2;
+ float weightIndPrior0, weightIndPrior1, weightIndPrior2;
+ float threshPrior0, threshPrior1, threshPrior2;
+ float widthPrior, widthPrior0, widthPrior1, widthPrior2;
+
+ widthPrior0 = WIDTH_PR_MAP;
+ widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region:
+ // lower range, so increase width in tanh map
+ widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure
+
+ //threshold parameters for features
+ threshPrior0 = inst->priorModelPars[0];
+ threshPrior1 = inst->priorModelPars[1];
+ threshPrior2 = inst->priorModelPars[3];
+
+ //sign for flatness feature
+ sgnMap = (int)(inst->priorModelPars[2]);
+
+ //weight parameters for features
+ weightIndPrior0 = inst->priorModelPars[4];
+ weightIndPrior1 = inst->priorModelPars[5];
+ weightIndPrior2 = inst->priorModelPars[6];
+
+ // compute feature based on average LR factor
+ // this is the average over all frequencies of the smooth log lrt
+ logLrtTimeAvgKsum = 0.0;
+ for (i = 0; i < inst->magnLen; i++) {
+ tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i];
+ tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001);
+ besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2;
+ inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1)
+ - inst->logLrtTimeAvg[i]);
+ logLrtTimeAvgKsum += inst->logLrtTimeAvg[i];
+ }
+ logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen);
+ inst->featureData[3] = logLrtTimeAvgKsum;
+ // done with computation of LR factor
+
+ //
+ //compute the indicator functions
+ //
+
+ // average lrt feature
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (logLrtTimeAvgKsum < threshPrior0) {
+ widthPrior = widthPrior1;
+ }
+ // compute indicator function: sigmoid map
+ indicator0 = (float)0.5 * ((float)tanh(widthPrior *
+ (logLrtTimeAvgKsum - threshPrior0)) + (float)1.0);
+
+ //spectral flatness feature
+ tmpFloat1 = inst->featureData[0];
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (sgnMap == 1 && (tmpFloat1 > threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ if (sgnMap == -1 && (tmpFloat1 < threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ // compute indicator function: sigmoid map
+ indicator1 = (float)0.5 * ((float)tanh((float)sgnMap *
+ widthPrior * (threshPrior1 - tmpFloat1)) + (float)1.0);
+
+ //for template spectrum-difference
+ tmpFloat1 = inst->featureData[4];
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (tmpFloat1 < threshPrior2) {
+ widthPrior = widthPrior2;
+ }
+ // compute indicator function: sigmoid map
+ indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2))
+ + (float)1.0);
+
+ //combine the indicator function with the feature weights
+ indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2
+ * indicator2;
+ // done with computing indicator function
+
+ //compute the prior probability
+ inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb);
+ // make sure probabilities are within range: keep floor to 0.01
+ if (inst->priorSpeechProb > 1.0) {
+ inst->priorSpeechProb = (float)1.0;
+ }
+ if (inst->priorSpeechProb < 0.01) {
+ inst->priorSpeechProb = (float)0.01;
+ }
+
+ //final speech probability: combine prior model with LR factor:
+ gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001);
+ for (i = 0; i < inst->magnLen; i++) {
+ invLrt = (float)exp(-inst->logLrtTimeAvg[i]);
+ invLrt = (float)gainPrior * invLrt;
+ probSpeechFinal[i] = (float)1.0 / ((float)1.0 + invLrt);
+ }
+}
+
+int WebRtcNs_ProcessCore(NSinst_t* inst,
+ short* speechFrame,
+ short* speechFrameHB,
+ short* outFrame,
+ short* outFrameHB) {
+ // main routine for noise reduction
+
+ int flagHB = 0;
+ int i;
+ const int kStartBand = 5; // Skip first frequency bins during estimation.
+ int updateParsFlag;
+
+ float energy1, energy2, gain, factor, factor1, factor2;
+ float signalEnergy, sumMagn;
+ float snrPrior, currentEstimateStsa;
+ float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
+ float gammaNoiseTmp, gammaNoiseOld;
+ float noiseUpdateTmp, fTmp, dTmp;
+ float fin[BLOCKL_MAX], fout[BLOCKL_MAX];
+ float winData[ANAL_BLOCKL_MAX];
+ float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
+ float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
+ float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
+ float probSpeechFinal[HALF_ANAL_BLOCKL], previousEstimateStsa[HALF_ANAL_BLOCKL];
+ float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
+ // Variables during startup
+ float sum_log_i = 0.0;
+ float sum_log_i_square = 0.0;
+ float sum_log_magn = 0.0;
+ float sum_log_i_log_magn = 0.0;
+ float parametric_noise = 0.0;
+ float parametric_exp = 0.0;
+ float parametric_num = 0.0;
+
+ // SWB variables
+ int deltaBweHB = 1;
+ int deltaGainHB = 1;
+ float decayBweHB = 1.0;
+ float gainMapParHB = 1.0;
+ float gainTimeDomainHB = 1.0;
+ float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
+
+ // Check that initiation has been done
+ if (inst->initFlag != 1) {
+ return (-1);
+ }
+ // Check for valid pointers based on sampling rate
+ if (inst->fs == 32000) {
+ if (speechFrameHB == NULL) {
+ return -1;
+ }
+ flagHB = 1;
+ // range for averaging low band quantities for H band gain
+ deltaBweHB = (int)inst->magnLen / 4;
+ deltaGainHB = deltaBweHB;
+ }
+ //
+ updateParsFlag = inst->modelUpdatePars[0];
+ //
+
+ //for LB do all processing
+ // convert to float
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fin[i] = (float)speechFrame[i];
+ }
+ // update analysis buffer for L band
+ memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
+ sizeof(float) * (inst->anaLen - inst->blockLen10ms));
+ memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
+ sizeof(float) * inst->blockLen10ms);
+
+ if (flagHB == 1) {
+ // convert to float
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fin[i] = (float)speechFrameHB[i];
+ }
+ // update analysis buffer for H band
+ memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
+ sizeof(float) * (inst->anaLen - inst->blockLen10ms));
+ memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
+ sizeof(float) * inst->blockLen10ms);
+ }
+
+ // check if processing needed
+ if (inst->outLen == 0) {
+ // windowing
+ energy1 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ winData[i] = inst->window[i] * inst->dataBuf[i];
+ energy1 += winData[i] * winData[i];
+ }
+ if (energy1 == 0.0) {
+ // synthesize the special case of zero input
+ // we want to avoid updating statistics in this case:
+ // Updating feature statistics when we have zeros only will cause thresholds to
+ // move towards zero signal situations. This in turn has the effect that once the
+ // signal is "turned on" (non-zero values) everything will be treated as speech
+ // and there is no noise suppression effect. Depending on the duration of the
+ // inactive signal it takes a considerable amount of time for the system to learn
+ // what is noise and what is speech.
+
+ // read out fully processed segment
+ for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
+ fout[i - inst->windShift] = inst->syntBuf[i];
+ }
+ // update synthesis buffer
+ memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
+ sizeof(float) * inst->blockLen);
+
+ // out buffer
+ inst->outLen = inst->blockLen - inst->blockLen10ms;
+ if (inst->blockLen > inst->blockLen10ms) {
+ for (i = 0; i < inst->outLen; i++) {
+ inst->outBuf[i] = fout[i + inst->blockLen10ms];
+ }
+ }
+ // convert to short
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = fout[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrame[i] = (short)dTmp;
+ }
+
+ // for time-domain gain of HB
+ if (flagHB == 1) {
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = inst->dataBufHB[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrameHB[i] = (short)dTmp;
+ }
+ } // end of H band gain computation
+ //
+ return 0;
+ }
+
+ //
+ inst->blockInd++; // Update the block index only when we process a block.
+ // FFT
+ WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
+
+ imag[0] = 0;
+ real[0] = winData[0];
+ magn[0] = (float)(fabs(real[0]) + 1.0f);
+ imag[inst->magnLen - 1] = 0;
+ real[inst->magnLen - 1] = winData[1];
+ magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
+ signalEnergy = (float)(real[0] * real[0]) +
+ (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]);
+ sumMagn = magn[0] + magn[inst->magnLen - 1];
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[0] += magn[0];
+ inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
+ tmpFloat2 = log((float)(inst->magnLen - 1));
+ sum_log_i = tmpFloat2;
+ sum_log_i_square = tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[inst->magnLen - 1]);
+ sum_log_magn = tmpFloat1;
+ sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
+ }
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ real[i] = winData[2 * i];
+ imag[i] = winData[2 * i + 1];
+ // magnitude spectrum
+ fTmp = real[i] * real[i];
+ fTmp += imag[i] * imag[i];
+ signalEnergy += fTmp;
+ magn[i] = ((float)sqrt(fTmp)) + 1.0f;
+ sumMagn += magn[i];
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[i] += magn[i];
+ if (i >= kStartBand) {
+ tmpFloat2 = log((float)i);
+ sum_log_i += tmpFloat2;
+ sum_log_i_square += tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[i]);
+ sum_log_magn += tmpFloat1;
+ sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
+ }
+ }
+ }
+ signalEnergy = signalEnergy / ((float)inst->magnLen);
+ inst->signalEnergy = signalEnergy;
+ inst->sumMagn = sumMagn;
+
+ //compute spectral flatness on input spectrum
+ WebRtcNs_ComputeSpectralFlatness(inst, magn);
+ // quantile noise estimate
+ WebRtcNs_NoiseEstimation(inst, magn, noise);
+ //compute simplified noise model during startup
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ // Estimate White noise
+ inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
+ // Estimate Pink noise parameters
+ tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
+ tmpFloat1 -= (sum_log_i * sum_log_i);
+ tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the estimated spectrum to be positive
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ inst->pinkNoiseNumerator += tmpFloat3;
+ tmpFloat2 = (sum_log_i * sum_log_magn);
+ tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the pink noise power to be in the interval [0, 1];
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ if (tmpFloat3 > 1.0f) {
+ tmpFloat3 = 1.0f;
+ }
+ inst->pinkNoiseExp += tmpFloat3;
+
+ // Calculate frequency independent parts of parametric noise estimate.
+ if (inst->pinkNoiseExp == 0.0f) {
+ // Use white noise estimate
+ parametric_noise = inst->whiteNoiseLevel;
+ } else {
+ // Use pink noise estimate
+ parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
+ parametric_num *= (float)(inst->blockInd + 1);
+ parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
+ parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp);
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Estimate the background noise using the white and pink noise parameters
+ if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand)) {
+ // Use pink noise estimate
+ parametric_noise = parametric_num / pow((float)i, parametric_exp);
+ }
+ theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise);
+ theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
+ // Weight quantile noise with modeled noise
+ noise[i] *= (inst->blockInd);
+ tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd);
+ noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
+ noise[i] /= END_STARTUP_SHORT;
+ }
+ }
+ //compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure
+ if (inst->blockInd < END_STARTUP_LONG) {
+ inst->featureData[5] *= inst->blockInd;
+ inst->featureData[5] += signalEnergy;
+ inst->featureData[5] /= (inst->blockInd + 1);
+ }
+
+#ifdef PROCESS_FLOW_0
+ if (inst->blockInd > END_STARTUP_LONG) {
+ //option: average the quantile noise: for check with AEC2
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (float)0.6 * inst->noisePrev[i] + (float)0.4 * noise[i];
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Wiener with over sub-substraction:
+ theFilter[i] = (magn[i] - inst->overdrive * noise[i]) / (magn[i] + (float)0.0001);
+ }
+ }
+#else
+ //start processing at frames == converged+1
+ //
+ // STEP 1: compute prior and post snr based on quantile noise est
+ //
+
+ // compute DD estimate of prior SNR: needed for new method
+ for (i = 0; i < inst->magnLen; i++) {
+ // post snr
+ snrLocPost[i] = (float)0.0;
+ if (magn[i] > noise[i]) {
+ snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
+ }
+ // previous post snr
+ // previous estimate: based on previous frame with gain filter
+ previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001)
+ * (inst->smooth[i]);
+ // DD estimate is sum of two terms: current estimate and previous estimate
+ // directed decision update of snrPrior
+ snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
+ * snrLocPost[i];
+ // post and prior snr needed for step 2
+ } // end of loop over freqs
+#ifdef PROCESS_FLOW_1
+ for (i = 0; i < inst->magnLen; i++) {
+ // gain filter
+ tmpFloat1 = inst->overdrive + snrLocPrior[i];
+ tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
+ theFilter[i] = (float)tmpFloat2;
+ } // end of loop over freqs
+#endif
+ // done with step 1: dd computation of prior and post snr
+
+ //
+ //STEP 2: compute speech/noise likelihood
+ //
+#ifdef PROCESS_FLOW_2
+ // compute difference of input spectrum with learned/estimated noise spectrum
+ WebRtcNs_ComputeSpectralDifference(inst, magn);
+ // compute histograms for parameter decisions (thresholds and weights for features)
+ // parameters are extracted once every window time (=inst->modelUpdatePars[1])
+ if (updateParsFlag >= 1) {
+ // counter update
+ inst->modelUpdatePars[3]--;
+ // update histogram
+ if (inst->modelUpdatePars[3] > 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 0);
+ }
+ // compute model parameters
+ if (inst->modelUpdatePars[3] == 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 1);
+ inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
+ // if wish to update only once, set flag to zero
+ if (updateParsFlag == 1) {
+ inst->modelUpdatePars[0] = 0;
+ } else {
+ // update every window:
+ // get normalization for spectral difference for next window estimate
+ inst->featureData[6] = inst->featureData[6]
+ / ((float)inst->modelUpdatePars[1]);
+ inst->featureData[5] = (float)0.5 * (inst->featureData[6]
+ + inst->featureData[5]);
+ inst->featureData[6] = (float)0.0;
+ }
+ }
+ }
+ // compute speech/noise probability
+ WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost);
+ // time-avg parameter for noise update
+ gammaNoiseTmp = NOISE_UPDATE;
+ for (i = 0; i < inst->magnLen; i++) {
+ probSpeech = probSpeechFinal[i];
+ probNonSpeech = (float)1.0 - probSpeech;
+ // temporary noise update:
+ // use it for speech frames if update value is less than previous
+ noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
+ * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
+ //
+ // time-constant based on speech/noise state
+ gammaNoiseOld = gammaNoiseTmp;
+ gammaNoiseTmp = NOISE_UPDATE;
+ // increase gamma (i.e., less noise update) for frame likely to be speech
+ if (probSpeech > PROB_RANGE) {
+ gammaNoiseTmp = SPEECH_UPDATE;
+ }
+ // conservative noise update
+ if (probSpeech < PROB_RANGE) {
+ inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
+ }
+ // noise update
+ if (gammaNoiseTmp == gammaNoiseOld) {
+ noise[i] = noiseUpdateTmp;
+ } else {
+ noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
+ * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
+ // allow for noise update downwards:
+ // if noise update decreases the noise, it is safe, so allow it to happen
+ if (noiseUpdateTmp < noise[i]) {
+ noise[i] = noiseUpdateTmp;
+ }
+ }
+ } // end of freq loop
+ // done with step 2: noise update
+
+ //
+ // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
+ //
+ for (i = 0; i < inst->magnLen; i++) {
+ // post and prior snr
+ currentEstimateStsa = (float)0.0;
+ if (magn[i] > noise[i]) {
+ currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
+ }
+ // DD estimate is sume of two terms: current estimate and previous estimate
+ // directed decision update of snrPrior
+ snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
+ * currentEstimateStsa;
+ // gain filter
+ tmpFloat1 = inst->overdrive + snrPrior;
+ tmpFloat2 = (float)snrPrior / tmpFloat1;
+ theFilter[i] = (float)tmpFloat2;
+ } // end of loop over freqs
+ // done with step3
+#endif
+#endif
+
+ for (i = 0; i < inst->magnLen; i++) {
+ // flooring bottom
+ if (theFilter[i] < inst->denoiseBound) {
+ theFilter[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilter[i] > (float)1.0) {
+ theFilter[i] = 1.0;
+ }
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ // flooring bottom
+ if (theFilterTmp[i] < inst->denoiseBound) {
+ theFilterTmp[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilterTmp[i] > (float)1.0) {
+ theFilterTmp[i] = 1.0;
+ }
+ // Weight the two suppression filters
+ theFilter[i] *= (inst->blockInd);
+ theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
+ theFilter[i] += theFilterTmp[i];
+ theFilter[i] /= (END_STARTUP_SHORT);
+ }
+ // smoothing
+#ifdef PROCESS_FLOW_0
+ inst->smooth[i] *= SMOOTH; // value set to 0.7 in define.h file
+ inst->smooth[i] += ((float)1.0 - SMOOTH) * theFilter[i];
+#else
+ inst->smooth[i] = theFilter[i];
+#endif
+ real[i] *= inst->smooth[i];
+ imag[i] *= inst->smooth[i];
+ }
+ // keep track of noise and magn spectrum for next frame
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->noisePrev[i] = noise[i];
+ inst->magnPrev[i] = magn[i];
+ }
+ // back to time domain
+ winData[0] = real[0];
+ winData[1] = real[inst->magnLen - 1];
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ winData[2 * i] = real[i];
+ winData[2 * i + 1] = imag[i];
+ }
+ WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
+
+ for (i = 0; i < inst->anaLen; i++) {
+ real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
+ }
+
+ //scale factor: only do it after END_STARTUP_LONG time
+ factor = (float)1.0;
+ if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) {
+ factor1 = (float)1.0;
+ factor2 = (float)1.0;
+
+ energy2 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ energy2 += (float)real[i] * (float)real[i];
+ }
+ gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
+
+#ifdef PROCESS_FLOW_2
+ // scaling for new version
+ if (gain > B_LIM) {
+ factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
+ if (gain * factor1 > (float)1.0) {
+ factor1 = (float)1.0 / gain;
+ }
+ }
+ if (gain < B_LIM) {
+ //don't reduce scale too much for pause regions:
+ // attenuation here should be controlled by flooring
+ if (gain <= inst->denoiseBound) {
+ gain = inst->denoiseBound;
+ }
+ factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
+ }
+ //combine both scales with speech/noise prob:
+ // note prior (priorSpeechProb) is not frequency dependent
+ factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb)
+ * factor2;
+#else
+ if (gain > B_LIM) {
+ factor = (float)1.0 + (float)1.3 * (gain - B_LIM);
+ } else {
+ factor = (float)1.0 + (float)2.0 * (gain - B_LIM);
+ }
+ if (gain * factor > (float)1.0) {
+ factor = (float)1.0 / gain;
+ }
+#endif
+ } // out of inst->gainmap==1
+
+ // synthesis
+ for (i = 0; i < inst->anaLen; i++) {
+ inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
+ }
+ // read out fully processed segment
+ for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
+ fout[i - inst->windShift] = inst->syntBuf[i];
+ }
+ // update synthesis buffer
+ memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
+ sizeof(float) * inst->blockLen);
+
+ // out buffer
+ inst->outLen = inst->blockLen - inst->blockLen10ms;
+ if (inst->blockLen > inst->blockLen10ms) {
+ for (i = 0; i < inst->outLen; i++) {
+ inst->outBuf[i] = fout[i + inst->blockLen10ms];
+ }
+ }
+ } // end of if out.len==0
+ else {
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fout[i] = inst->outBuf[i];
+ }
+ memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms,
+ sizeof(float) * (inst->outLen - inst->blockLen10ms));
+ memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0,
+ sizeof(float) * inst->blockLen10ms);
+ inst->outLen -= inst->blockLen10ms;
+ }
+
+ // convert to short
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = fout[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrame[i] = (short)dTmp;
+ }
+
+ // for time-domain gain of HB
+ if (flagHB == 1) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->speechProbHB[i] = probSpeechFinal[i];
+ }
+ // average speech prob from low band
+ // avg over second half (i.e., 4->8kHz) of freq. spectrum
+ avgProbSpeechHB = 0.0;
+ for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) {
+ avgProbSpeechHB += inst->speechProbHB[i];
+ }
+ avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
+ // average filter gain from low band
+ // average over second half (i.e., 4->8kHz) of freq. spectrum
+ avgFilterGainHB = 0.0;
+ for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++) {
+ avgFilterGainHB += inst->smooth[i];
+ }
+ avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
+ avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
+ // gain based on speech prob:
+ gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
+ //combine gain with low band gain
+ gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
+ if (avgProbSpeechHB >= (float)0.5) {
+ gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
+ }
+ gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
+ //make sure gain is within flooring range
+ // flooring bottom
+ if (gainTimeDomainHB < inst->denoiseBound) {
+ gainTimeDomainHB = inst->denoiseBound;
+ }
+ // flooring top
+ if (gainTimeDomainHB > (float)1.0) {
+ gainTimeDomainHB = 1.0;
+ }
+ //apply gain
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = gainTimeDomainHB * inst->dataBufHB[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrameHB[i] = (short)dTmp;
+ }
+ } // end of H band gain computation
+ //
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/ns/ns_core.h b/trunk/src/modules/audio_processing/ns/ns_core.h
new file mode 100644
index 0000000..2f4c34f
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/ns_core.h
@@ -0,0 +1,179 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
+
+#include "defines.h"
+
+typedef struct NSParaExtract_t_ {
+
+ //bin size of histogram
+ float binSizeLrt;
+ float binSizeSpecFlat;
+ float binSizeSpecDiff;
+ //range of histogram over which lrt threshold is computed
+ float rangeAvgHistLrt;
+ //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
+ //thresholds for prior model
+ float factor1ModelPars; //for lrt and spectral difference
+ float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech
+ //peak limit for spectral flatness (varies between 0 and 1)
+ float thresPosSpecFlat;
+ //limit on spacing of two highest peaks in histogram: spacing determined by bin size
+ float limitPeakSpacingSpecFlat;
+ float limitPeakSpacingSpecDiff;
+ //limit on relevance of second peak:
+ float limitPeakWeightsSpecFlat;
+ float limitPeakWeightsSpecDiff;
+ //limit on fluctuation of lrt feature
+ float thresFluctLrt;
+ //limit on the max and min values for the feature thresholds
+ float maxLrt;
+ float minLrt;
+ float maxSpecFlat;
+ float minSpecFlat;
+ float maxSpecDiff;
+ float minSpecDiff;
+ //criteria of weight of histogram peak to accept/reject feature
+ int thresWeightSpecFlat;
+ int thresWeightSpecDiff;
+
+} NSParaExtract_t;
+
+typedef struct NSinst_t_ {
+
+ WebRtc_UWord32 fs;
+ int blockLen;
+ int blockLen10ms;
+ int windShift;
+ int outLen;
+ int anaLen;
+ int magnLen;
+ int aggrMode;
+ const float* window;
+ float dataBuf[ANAL_BLOCKL_MAX];
+ float syntBuf[ANAL_BLOCKL_MAX];
+ float outBuf[3 * BLOCKL_MAX];
+
+ int initFlag;
+ // parameters for quantile noise estimation
+ float density[SIMULT* HALF_ANAL_BLOCKL];
+ float lquantile[SIMULT* HALF_ANAL_BLOCKL];
+ float quantile[HALF_ANAL_BLOCKL];
+ int counter[SIMULT];
+ int updates;
+ // parameters for Wiener filter
+ float smooth[HALF_ANAL_BLOCKL];
+ float overdrive;
+ float denoiseBound;
+ int gainmap;
+ // fft work arrays.
+ int ip[IP_LENGTH];
+ float wfft[W_LENGTH];
+
+ // parameters for new method: some not needed, will reduce/cleanup later
+ WebRtc_Word32 blockInd; //frame index counter
+ int modelUpdatePars[4]; //parameters for updating or estimating
+ // thresholds/weights for prior model
+ float priorModelPars[7]; //parameters for prior model
+ float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
+ float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame
+ float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing
+ float priorSpeechProb; //prior speech/noise probability
+ float featureData[7]; //data for features
+ float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate
+ float signalEnergy; //energy of magn
+ float sumMagn; //sum of magn
+ float whiteNoiseLevel; //initial noise estimate
+ float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate
+ float pinkNoiseNumerator; //pink noise parameter: numerator
+ float pinkNoiseExp; //pink noise parameter: power of freq
+ NSParaExtract_t featureExtractionParams; //parameters for feature extraction
+ //histograms for parameter estimation
+ int histLrt[HIST_PAR_EST];
+ int histSpecFlat[HIST_PAR_EST];
+ int histSpecDiff[HIST_PAR_EST];
+ //quantities for high band estimate
+ float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT
+ float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB
+
+} NSinst_t;
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcNs_InitCore(...)
+ *
+ * This function initializes a noise suppression instance
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - fs : Sampling frequency
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs);
+
+/****************************************************************************
+ * WebRtcNs_set_policy_core(...)
+ *
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
+ *
+ * Output:
+ * - NS_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_set_policy_core(NSinst_t* inst, int mode);
+
+/****************************************************************************
+ * WebRtcNs_ProcessCore
+ *
+ * Do noise suppression.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - inFrameLow : Input speech frame for lower band
+ * - inFrameHigh : Input speech frame for higher band
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outFrameLow : Output speech frame for lower band
+ * - outFrameHigh : Output speech frame for higher band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+
+
+int WebRtcNs_ProcessCore(NSinst_t* inst,
+ short* inFrameLow,
+ short* inFrameHigh,
+ short* outFrameLow,
+ short* outFrameHigh);
+
+
+#ifdef __cplusplus
+}
+#endif
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core.c b/trunk/src/modules/audio_processing/ns/nsx_core.c
new file mode 100644
index 0000000..51bde0c
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/nsx_core.c
@@ -0,0 +1,2444 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "noise_suppression_x.h"
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+
+#include "cpu_features_wrapper.h"
+#include "nsx_core.h"
+
+// Skip first frequency bins during estimation. (0 <= value < 64)
+static const int kStartBand = 5;
+
+// Constants to compensate for shifting signal log(2^shifts).
+const WebRtc_Word16 WebRtcNsx_kLogTable[9] = {
+ 0, 177, 355, 532, 710, 887, 1065, 1242, 1420
+};
+
+const WebRtc_Word16 WebRtcNsx_kCounterDiv[201] = {
+ 32767, 16384, 10923, 8192, 6554, 5461, 4681,
+ 4096, 3641, 3277, 2979, 2731, 2521, 2341, 2185, 2048, 1928, 1820, 1725, 1638, 1560,
+ 1489, 1425, 1365, 1311, 1260, 1214, 1170, 1130, 1092, 1057, 1024, 993, 964, 936, 910,
+ 886, 862, 840, 819, 799, 780, 762, 745, 728, 712, 697, 683, 669, 655, 643, 630, 618,
+ 607, 596, 585, 575, 565, 555, 546, 537, 529, 520, 512, 504, 496, 489, 482, 475, 468,
+ 462, 455, 449, 443, 437, 431, 426, 420, 415, 410, 405, 400, 395, 390, 386, 381, 377,
+ 372, 368, 364, 360, 356, 352, 349, 345, 341, 338, 334, 331, 328, 324, 321, 318, 315,
+ 312, 309, 306, 303, 301, 298, 295, 293, 290, 287, 285, 282, 280, 278, 275, 273, 271,
+ 269, 266, 264, 262, 260, 258, 256, 254, 252, 250, 248, 246, 245, 243, 241, 239, 237,
+ 236, 234, 232, 231, 229, 228, 226, 224, 223, 221, 220, 218, 217, 216, 214, 213, 211,
+ 210, 209, 207, 206, 205, 204, 202, 201, 200, 199, 197, 196, 195, 194, 193, 192, 191,
+ 189, 188, 187, 186, 185, 184, 183, 182, 181, 180, 179, 178, 177, 176, 175, 174, 173,
+ 172, 172, 171, 170, 169, 168, 167, 166, 165, 165, 164, 163
+};
+
+const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256] = {
+ 0, 1, 3, 4, 6, 7, 9, 10, 11, 13, 14, 16, 17, 18, 20, 21,
+ 22, 24, 25, 26, 28, 29, 30, 32, 33, 34, 36, 37, 38, 40, 41, 42,
+ 44, 45, 46, 47, 49, 50, 51, 52, 54, 55, 56, 57, 59, 60, 61, 62,
+ 63, 65, 66, 67, 68, 69, 71, 72, 73, 74, 75, 77, 78, 79, 80, 81,
+ 82, 84, 85, 86, 87, 88, 89, 90, 92, 93, 94, 95, 96, 97, 98, 99,
+ 100, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 116, 117,
+ 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133,
+ 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
+ 150, 151, 152, 153, 154, 155, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
+ 165, 166, 167, 168, 169, 169, 170, 171, 172, 173, 174, 175, 176, 177, 178, 178,
+ 179, 180, 181, 182, 183, 184, 185, 185, 186, 187, 188, 189, 190, 191, 192, 192,
+ 193, 194, 195, 196, 197, 198, 198, 199, 200, 201, 202, 203, 203, 204, 205, 206,
+ 207, 208, 208, 209, 210, 211, 212, 212, 213, 214, 215, 216, 216, 217, 218, 219,
+ 220, 220, 221, 222, 223, 224, 224, 225, 226, 227, 228, 228, 229, 230, 231, 231,
+ 232, 233, 234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 244,
+ 244, 245, 246, 247, 247, 248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
+};
+
+static const WebRtc_Word16 kPowTableFrac[1024] = {
+ 0, 1, 1, 2, 3, 3, 4, 5,
+ 6, 6, 7, 8, 8, 9, 10, 10,
+ 11, 12, 13, 13, 14, 15, 15, 16,
+ 17, 17, 18, 19, 20, 20, 21, 22,
+ 22, 23, 24, 25, 25, 26, 27, 27,
+ 28, 29, 30, 30, 31, 32, 32, 33,
+ 34, 35, 35, 36, 37, 37, 38, 39,
+ 40, 40, 41, 42, 42, 43, 44, 45,
+ 45, 46, 47, 48, 48, 49, 50, 50,
+ 51, 52, 53, 53, 54, 55, 56, 56,
+ 57, 58, 58, 59, 60, 61, 61, 62,
+ 63, 64, 64, 65, 66, 67, 67, 68,
+ 69, 69, 70, 71, 72, 72, 73, 74,
+ 75, 75, 76, 77, 78, 78, 79, 80,
+ 81, 81, 82, 83, 84, 84, 85, 86,
+ 87, 87, 88, 89, 90, 90, 91, 92,
+ 93, 93, 94, 95, 96, 96, 97, 98,
+ 99, 100, 100, 101, 102, 103, 103, 104,
+ 105, 106, 106, 107, 108, 109, 109, 110,
+ 111, 112, 113, 113, 114, 115, 116, 116,
+ 117, 118, 119, 119, 120, 121, 122, 123,
+ 123, 124, 125, 126, 126, 127, 128, 129,
+ 130, 130, 131, 132, 133, 133, 134, 135,
+ 136, 137, 137, 138, 139, 140, 141, 141,
+ 142, 143, 144, 144, 145, 146, 147, 148,
+ 148, 149, 150, 151, 152, 152, 153, 154,
+ 155, 156, 156, 157, 158, 159, 160, 160,
+ 161, 162, 163, 164, 164, 165, 166, 167,
+ 168, 168, 169, 170, 171, 172, 173, 173,
+ 174, 175, 176, 177, 177, 178, 179, 180,
+ 181, 181, 182, 183, 184, 185, 186, 186,
+ 187, 188, 189, 190, 190, 191, 192, 193,
+ 194, 195, 195, 196, 197, 198, 199, 200,
+ 200, 201, 202, 203, 204, 205, 205, 206,
+ 207, 208, 209, 210, 210, 211, 212, 213,
+ 214, 215, 215, 216, 217, 218, 219, 220,
+ 220, 221, 222, 223, 224, 225, 225, 226,
+ 227, 228, 229, 230, 231, 231, 232, 233,
+ 234, 235, 236, 237, 237, 238, 239, 240,
+ 241, 242, 243, 243, 244, 245, 246, 247,
+ 248, 249, 249, 250, 251, 252, 253, 254,
+ 255, 255, 256, 257, 258, 259, 260, 261,
+ 262, 262, 263, 264, 265, 266, 267, 268,
+ 268, 269, 270, 271, 272, 273, 274, 275,
+ 276, 276, 277, 278, 279, 280, 281, 282,
+ 283, 283, 284, 285, 286, 287, 288, 289,
+ 290, 291, 291, 292, 293, 294, 295, 296,
+ 297, 298, 299, 299, 300, 301, 302, 303,
+ 304, 305, 306, 307, 308, 308, 309, 310,
+ 311, 312, 313, 314, 315, 316, 317, 318,
+ 318, 319, 320, 321, 322, 323, 324, 325,
+ 326, 327, 328, 328, 329, 330, 331, 332,
+ 333, 334, 335, 336, 337, 338, 339, 339,
+ 340, 341, 342, 343, 344, 345, 346, 347,
+ 348, 349, 350, 351, 352, 352, 353, 354,
+ 355, 356, 357, 358, 359, 360, 361, 362,
+ 363, 364, 365, 366, 367, 367, 368, 369,
+ 370, 371, 372, 373, 374, 375, 376, 377,
+ 378, 379, 380, 381, 382, 383, 384, 385,
+ 385, 386, 387, 388, 389, 390, 391, 392,
+ 393, 394, 395, 396, 397, 398, 399, 400,
+ 401, 402, 403, 404, 405, 406, 407, 408,
+ 409, 410, 410, 411, 412, 413, 414, 415,
+ 416, 417, 418, 419, 420, 421, 422, 423,
+ 424, 425, 426, 427, 428, 429, 430, 431,
+ 432, 433, 434, 435, 436, 437, 438, 439,
+ 440, 441, 442, 443, 444, 445, 446, 447,
+ 448, 449, 450, 451, 452, 453, 454, 455,
+ 456, 457, 458, 459, 460, 461, 462, 463,
+ 464, 465, 466, 467, 468, 469, 470, 471,
+ 472, 473, 474, 475, 476, 477, 478, 479,
+ 480, 481, 482, 483, 484, 485, 486, 487,
+ 488, 489, 490, 491, 492, 493, 494, 495,
+ 496, 498, 499, 500, 501, 502, 503, 504,
+ 505, 506, 507, 508, 509, 510, 511, 512,
+ 513, 514, 515, 516, 517, 518, 519, 520,
+ 521, 522, 523, 525, 526, 527, 528, 529,
+ 530, 531, 532, 533, 534, 535, 536, 537,
+ 538, 539, 540, 541, 542, 544, 545, 546,
+ 547, 548, 549, 550, 551, 552, 553, 554,
+ 555, 556, 557, 558, 560, 561, 562, 563,
+ 564, 565, 566, 567, 568, 569, 570, 571,
+ 572, 574, 575, 576, 577, 578, 579, 580,
+ 581, 582, 583, 584, 585, 587, 588, 589,
+ 590, 591, 592, 593, 594, 595, 596, 597,
+ 599, 600, 601, 602, 603, 604, 605, 606,
+ 607, 608, 610, 611, 612, 613, 614, 615,
+ 616, 617, 618, 620, 621, 622, 623, 624,
+ 625, 626, 627, 628, 630, 631, 632, 633,
+ 634, 635, 636, 637, 639, 640, 641, 642,
+ 643, 644, 645, 646, 648, 649, 650, 651,
+ 652, 653, 654, 656, 657, 658, 659, 660,
+ 661, 662, 664, 665, 666, 667, 668, 669,
+ 670, 672, 673, 674, 675, 676, 677, 678,
+ 680, 681, 682, 683, 684, 685, 687, 688,
+ 689, 690, 691, 692, 693, 695, 696, 697,
+ 698, 699, 700, 702, 703, 704, 705, 706,
+ 708, 709, 710, 711, 712, 713, 715, 716,
+ 717, 718, 719, 720, 722, 723, 724, 725,
+ 726, 728, 729, 730, 731, 732, 733, 735,
+ 736, 737, 738, 739, 741, 742, 743, 744,
+ 745, 747, 748, 749, 750, 751, 753, 754,
+ 755, 756, 757, 759, 760, 761, 762, 763,
+ 765, 766, 767, 768, 770, 771, 772, 773,
+ 774, 776, 777, 778, 779, 780, 782, 783,
+ 784, 785, 787, 788, 789, 790, 792, 793,
+ 794, 795, 796, 798, 799, 800, 801, 803,
+ 804, 805, 806, 808, 809, 810, 811, 813,
+ 814, 815, 816, 818, 819, 820, 821, 823,
+ 824, 825, 826, 828, 829, 830, 831, 833,
+ 834, 835, 836, 838, 839, 840, 841, 843,
+ 844, 845, 846, 848, 849, 850, 851, 853,
+ 854, 855, 857, 858, 859, 860, 862, 863,
+ 864, 866, 867, 868, 869, 871, 872, 873,
+ 874, 876, 877, 878, 880, 881, 882, 883,
+ 885, 886, 887, 889, 890, 891, 893, 894,
+ 895, 896, 898, 899, 900, 902, 903, 904,
+ 906, 907, 908, 909, 911, 912, 913, 915,
+ 916, 917, 919, 920, 921, 923, 924, 925,
+ 927, 928, 929, 931, 932, 933, 935, 936,
+ 937, 938, 940, 941, 942, 944, 945, 946,
+ 948, 949, 950, 952, 953, 955, 956, 957,
+ 959, 960, 961, 963, 964, 965, 967, 968,
+ 969, 971, 972, 973, 975, 976, 977, 979,
+ 980, 981, 983, 984, 986, 987, 988, 990,
+ 991, 992, 994, 995, 996, 998, 999, 1001,
+ 1002, 1003, 1005, 1006, 1007, 1009, 1010, 1012,
+ 1013, 1014, 1016, 1017, 1018, 1020, 1021, 1023
+};
+
+static const WebRtc_Word16 kIndicatorTable[17] = {
+ 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718,
+ 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187
+};
+
+// hybrib Hanning & flat window
+static const WebRtc_Word16 kBlocks80w128x[128] = {
+ 0, 536, 1072, 1606, 2139, 2669, 3196, 3720, 4240, 4756, 5266,
+ 5771, 6270, 6762, 7246, 7723, 8192, 8652, 9102, 9543, 9974, 10394,
+ 10803, 11200, 11585, 11958, 12318, 12665, 12998, 13318, 13623, 13913, 14189,
+ 14449, 14694, 14924, 15137, 15334, 15515, 15679, 15826, 15956, 16069, 16165,
+ 16244, 16305, 16349, 16375, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16375, 16349, 16305, 16244, 16165, 16069, 15956,
+ 15826, 15679, 15515, 15334, 15137, 14924, 14694, 14449, 14189, 13913, 13623,
+ 13318, 12998, 12665, 12318, 11958, 11585, 11200, 10803, 10394, 9974, 9543,
+ 9102, 8652, 8192, 7723, 7246, 6762, 6270, 5771, 5266, 4756, 4240,
+ 3720, 3196, 2669, 2139, 1606, 1072, 536
+};
+
+// hybrib Hanning & flat window
+static const WebRtc_Word16 kBlocks160w256x[256] = {
+ 0, 268, 536, 804, 1072, 1339, 1606, 1872,
+ 2139, 2404, 2669, 2933, 3196, 3459, 3720, 3981,
+ 4240, 4499, 4756, 5012, 5266, 5520, 5771, 6021,
+ 6270, 6517, 6762, 7005, 7246, 7486, 7723, 7959,
+ 8192, 8423, 8652, 8878, 9102, 9324, 9543, 9760,
+ 9974, 10185, 10394, 10600, 10803, 11003, 11200, 11394,
+ 11585, 11773, 11958, 12140, 12318, 12493, 12665, 12833,
+ 12998, 13160, 13318, 13472, 13623, 13770, 13913, 14053,
+ 14189, 14321, 14449, 14574, 14694, 14811, 14924, 15032,
+ 15137, 15237, 15334, 15426, 15515, 15599, 15679, 15754,
+ 15826, 15893, 15956, 16015, 16069, 16119, 16165, 16207,
+ 16244, 16277, 16305, 16329, 16349, 16364, 16375, 16382,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16382, 16375, 16364, 16349, 16329, 16305, 16277,
+ 16244, 16207, 16165, 16119, 16069, 16015, 15956, 15893,
+ 15826, 15754, 15679, 15599, 15515, 15426, 15334, 15237,
+ 15137, 15032, 14924, 14811, 14694, 14574, 14449, 14321,
+ 14189, 14053, 13913, 13770, 13623, 13472, 13318, 13160,
+ 12998, 12833, 12665, 12493, 12318, 12140, 11958, 11773,
+ 11585, 11394, 11200, 11003, 10803, 10600, 10394, 10185,
+ 9974, 9760, 9543, 9324, 9102, 8878, 8652, 8423,
+ 8192, 7959, 7723, 7486, 7246, 7005, 6762, 6517,
+ 6270, 6021, 5771, 5520, 5266, 5012, 4756, 4499,
+ 4240, 3981, 3720, 3459, 3196, 2933, 2669, 2404,
+ 2139, 1872, 1606, 1339, 1072, 804, 536, 268
+};
+
+// Gain factor1 table: Input value in Q8 and output value in Q13
+// original floating point code
+// if (gain > blim) {
+// factor1 = 1.0 + 1.3 * (gain - blim);
+// if (gain * factor1 > 1.0) {
+// factor1 = 1.0 / gain;
+// }
+// } else {
+// factor1 = 1.0;
+// }
+static const WebRtc_Word16 kFactor1Table[257] = {
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8233, 8274, 8315, 8355, 8396, 8436, 8475, 8515, 8554, 8592, 8631, 8669,
+ 8707, 8745, 8783, 8820, 8857, 8894, 8931, 8967, 9003, 9039, 9075, 9111, 9146, 9181,
+ 9216, 9251, 9286, 9320, 9354, 9388, 9422, 9456, 9489, 9523, 9556, 9589, 9622, 9655,
+ 9687, 9719, 9752, 9784, 9816, 9848, 9879, 9911, 9942, 9973, 10004, 10035, 10066,
+ 10097, 10128, 10158, 10188, 10218, 10249, 10279, 10308, 10338, 10368, 10397, 10426,
+ 10456, 10485, 10514, 10543, 10572, 10600, 10629, 10657, 10686, 10714, 10742, 10770,
+ 10798, 10826, 10854, 10882, 10847, 10810, 10774, 10737, 10701, 10666, 10631, 10596,
+ 10562, 10527, 10494, 10460, 10427, 10394, 10362, 10329, 10297, 10266, 10235, 10203,
+ 10173, 10142, 10112, 10082, 10052, 10023, 9994, 9965, 9936, 9908, 9879, 9851, 9824,
+ 9796, 9769, 9742, 9715, 9689, 9662, 9636, 9610, 9584, 9559, 9534, 9508, 9484, 9459,
+ 9434, 9410, 9386, 9362, 9338, 9314, 9291, 9268, 9245, 9222, 9199, 9176, 9154, 9132,
+ 9110, 9088, 9066, 9044, 9023, 9002, 8980, 8959, 8939, 8918, 8897, 8877, 8857, 8836,
+ 8816, 8796, 8777, 8757, 8738, 8718, 8699, 8680, 8661, 8642, 8623, 8605, 8586, 8568,
+ 8550, 8532, 8514, 8496, 8478, 8460, 8443, 8425, 8408, 8391, 8373, 8356, 8339, 8323,
+ 8306, 8289, 8273, 8256, 8240, 8224, 8208, 8192
+};
+
+// For Factor2 tables
+// original floating point code
+// if (gain > blim) {
+// factor2 = 1.0;
+// } else {
+// factor2 = 1.0 - 0.3 * (blim - gain);
+// if (gain <= inst->denoiseBound) {
+// factor2 = 1.0 - 0.3 * (blim - inst->denoiseBound);
+// }
+// }
+//
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness1[257] = {
+ 7577, 7577, 7577, 7577, 7577, 7577,
+ 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness2[257] = {
+ 7270, 7270, 7270, 7270, 7270, 7306,
+ 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness3[257] = {
+ 7184, 7184, 7184, 7229, 7270, 7306,
+ 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// sum of log2(i) from table index to inst->anaLen2 in Q5
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kSumLogIndex[66] = {
+ 0, 22917, 22917, 22885, 22834, 22770, 22696, 22613,
+ 22524, 22428, 22326, 22220, 22109, 21994, 21876, 21754,
+ 21629, 21501, 21370, 21237, 21101, 20963, 20822, 20679,
+ 20535, 20388, 20239, 20089, 19937, 19783, 19628, 19470,
+ 19312, 19152, 18991, 18828, 18664, 18498, 18331, 18164,
+ 17994, 17824, 17653, 17480, 17306, 17132, 16956, 16779,
+ 16602, 16423, 16243, 16063, 15881, 15699, 15515, 15331,
+ 15146, 14960, 14774, 14586, 14398, 14209, 14019, 13829,
+ 13637, 13445
+};
+
+// sum of log2(i)^2 from table index to inst->anaLen2 in Q2
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kSumSquareLogIndex[66] = {
+ 0, 16959, 16959, 16955, 16945, 16929, 16908, 16881,
+ 16850, 16814, 16773, 16729, 16681, 16630, 16575, 16517,
+ 16456, 16392, 16325, 16256, 16184, 16109, 16032, 15952,
+ 15870, 15786, 15700, 15612, 15521, 15429, 15334, 15238,
+ 15140, 15040, 14938, 14834, 14729, 14622, 14514, 14404,
+ 14292, 14179, 14064, 13947, 13830, 13710, 13590, 13468,
+ 13344, 13220, 13094, 12966, 12837, 12707, 12576, 12444,
+ 12310, 12175, 12039, 11902, 11763, 11624, 11483, 11341,
+ 11198, 11054
+};
+
+// log2(table index) in Q12
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kLogIndex[129] = {
+ 0, 0, 4096, 6492, 8192, 9511, 10588, 11499,
+ 12288, 12984, 13607, 14170, 14684, 15157, 15595, 16003,
+ 16384, 16742, 17080, 17400, 17703, 17991, 18266, 18529,
+ 18780, 19021, 19253, 19476, 19691, 19898, 20099, 20292,
+ 20480, 20662, 20838, 21010, 21176, 21338, 21496, 21649,
+ 21799, 21945, 22087, 22226, 22362, 22495, 22625, 22752,
+ 22876, 22998, 23117, 23234, 23349, 23462, 23572, 23680,
+ 23787, 23892, 23994, 24095, 24195, 24292, 24388, 24483,
+ 24576, 24668, 24758, 24847, 24934, 25021, 25106, 25189,
+ 25272, 25354, 25434, 25513, 25592, 25669, 25745, 25820,
+ 25895, 25968, 26041, 26112, 26183, 26253, 26322, 26390,
+ 26458, 26525, 26591, 26656, 26721, 26784, 26848, 26910,
+ 26972, 27033, 27094, 27154, 27213, 27272, 27330, 27388,
+ 27445, 27502, 27558, 27613, 27668, 27722, 27776, 27830,
+ 27883, 27935, 27988, 28039, 28090, 28141, 28191, 28241,
+ 28291, 28340, 28388, 28437, 28484, 28532, 28579, 28626,
+ 28672
+};
+
+// determinant of estimation matrix in Q0 corresponding to the log2 tables above
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kDeterminantEstMatrix[66] = {
+ 0, 29814, 25574, 22640, 20351, 18469, 16873, 15491,
+ 14277, 13199, 12233, 11362, 10571, 9851, 9192, 8587,
+ 8030, 7515, 7038, 6596, 6186, 5804, 5448, 5115,
+ 4805, 4514, 4242, 3988, 3749, 3524, 3314, 3116,
+ 2930, 2755, 2590, 2435, 2289, 2152, 2022, 1900,
+ 1785, 1677, 1575, 1478, 1388, 1302, 1221, 1145,
+ 1073, 1005, 942, 881, 825, 771, 721, 674,
+ 629, 587, 547, 510, 475, 442, 411, 382,
+ 355, 330
+};
+
+// Declare function pointers.
+NoiseEstimation WebRtcNsx_NoiseEstimation;
+PrepareSpectrum WebRtcNsx_PrepareSpectrum;
+SynthesisUpdate WebRtcNsx_SynthesisUpdate;
+AnalysisUpdate WebRtcNsx_AnalysisUpdate;
+Denormalize WebRtcNsx_Denormalize;
+CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
+
+// Update the noise estimation information.
+static void UpdateNoiseEstimate(NsxInst_t* inst, int offset) {
+ WebRtc_Word32 tmp32no1 = 0;
+ WebRtc_Word32 tmp32no2 = 0;
+ WebRtc_Word16 tmp16 = 0;
+ const WebRtc_Word16 kExp2Const = 11819; // Q13
+
+ int i = 0;
+
+ tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
+ inst->magnLen);
+ // Guarantee a Q-domain as high as possible and still fit in int16
+ inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ kExp2Const, tmp16, 21);
+ for (i = 0; i < inst->magnLen; i++) {
+ // inst->quantile[i]=exp(inst->lquantile[offset+i]);
+ // in Q21
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ inst->noiseEstLogQuantile[offset + i]);
+ tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+ tmp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ tmp16 -= 21;// shift 21 to get result in Q0
+ tmp16 += (WebRtc_Word16) inst->qNoise; //shift to get result in Q(qNoise)
+ if (tmp16 < 0) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ }
+ inst->noiseEstQuantile[i] = WebRtcSpl_SatW32ToW16(tmp32no1);
+ }
+}
+
+// Noise Estimation
+static void NoiseEstimationC(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise) {
+ WebRtc_Word16 lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
+ WebRtc_Word16 countProd, delta, zeros, frac;
+ WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
+ const int16_t log2_const = 22713; // Q15
+ const int16_t width_factor = 21845;
+
+ int i, s, offset;
+
+ tabind = inst->stages - inst->normData;
+ assert(tabind < 9);
+ assert(tabind > -9);
+ if (tabind < 0) {
+ logval = -WebRtcNsx_kLogTable[-tabind];
+ } else {
+ logval = WebRtcNsx_kLogTable[tabind];
+ }
+
+ // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
+ // magn is in Q(-stages), and the real lmagn values are:
+ // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
+ // lmagn in Q8
+ for (i = 0; i < inst->magnLen; i++) {
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magn[i] << zeros)
+ & 0x7FFFFFFF) >> 23);
+ // log2(magn(i))
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ // log2(magn(i))*log(2)
+ lmagn[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
+ // + log(2^stages)
+ lmagn[i] += logval;
+ } else {
+ lmagn[i] = logval;//0;
+ }
+ }
+
+ // loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // Get counter values from state
+ counter = inst->noiseEstCounter[s];
+ assert(counter < 201);
+ countDiv = WebRtcNsx_kCounterDiv[counter];
+ countProd = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(counter, countDiv);
+
+ // quant_est(...)
+ for (i = 0; i < inst->magnLen; i++) {
+ // compute delta
+ if (inst->noiseEstDensity[offset + i] > 512) {
+ // Get the value for delta by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
+ delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ } else {
+ delta = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ delta = FACTOR_Q7_STARTUP;
+ }
+ }
+
+ // update log quantile estimate
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
+ // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
+ // CounterDiv=1/(inst->counter[s]+1) in Q15
+ tmp16 += 2;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
+ inst->noiseEstLogQuantile[offset + i] += tmp16no1;
+ } else {
+ tmp16 += 1;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
+ // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
+ inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
+ if (inst->noiseEstLogQuantile[offset + i] < logval) {
+ // This is the smallest fixed point representation we can
+ // have, hence we limit the output.
+ inst->noiseEstLogQuantile[offset + i] = logval;
+ }
+ }
+
+ // update density estimate
+ if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
+ < WIDTH_Q8) {
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->noiseEstDensity[offset + i], countProd, 15);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ width_factor, countDiv, 15);
+ inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
+ }
+ } // end loop over magnitude spectrum
+
+ if (counter >= END_STARTUP_LONG) {
+ inst->noiseEstCounter[s] = 0;
+ if (inst->blockIndex >= END_STARTUP_LONG) {
+ UpdateNoiseEstimate(inst, offset);
+ }
+ }
+ inst->noiseEstCounter[s]++;
+
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ UpdateNoiseEstimate(inst, offset);
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (WebRtc_UWord32)(inst->noiseEstQuantile[i]); // Q(qNoise)
+ }
+ (*q_noise) = (WebRtc_Word16)inst->qNoise;
+}
+
+// Filter the data in the frequency domain, and create spectrum.
+static void PrepareSpectrumC(NsxInst_t* inst, int16_t* freq_buf) {
+ int i = 0, j = 0;
+ int16_t tmp16 = 0;
+
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
+ (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ inst->imag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
+ (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ }
+
+ freq_buf[0] = inst->real[0];
+ freq_buf[1] = -inst->imag[0];
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ tmp16 = (inst->anaLen << 1) - j;
+ freq_buf[j] = inst->real[i];
+ freq_buf[j + 1] = -inst->imag[i];
+ freq_buf[tmp16] = inst->real[i];
+ freq_buf[tmp16 + 1] = inst->imag[i];
+ }
+ freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+}
+
+// Denormalize the input buffer.
+static __inline void DenormalizeC(NsxInst_t* inst, int16_t* in, int factor) {
+ int i = 0, j = 0;
+ int32_t tmp32 = 0;
+ for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
+ tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)in[j],
+ factor - inst->normData);
+ inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ }
+}
+
+// For the noise supression process, synthesis, read out fully processed
+// segment, and update synthesis buffer.
+static void SynthesisUpdateC(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor) {
+ int i = 0;
+ int16_t tmp16a = 0;
+ int16_t tmp16b = 0;
+ int32_t tmp32 = 0;
+
+ // synthesis
+ for (i = 0; i < inst->anaLen; i++) {
+ tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13); // Q0
+ // Down shift with rounding
+ tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(inst->synthesisBuffer[i],
+ tmp16b); // Q0
+ }
+
+ // read out fully processed segment
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ out_frame[i] = inst->synthesisBuffer[i]; // Q0
+ }
+
+ // update synthesis buffer
+ WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ inst->synthesisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
+ + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
+}
+
+// Update analysis buffer for lower band, and window data before FFT.
+static void AnalysisUpdateC(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech) {
+ int i = 0;
+
+ // For lower band update analysis buffer.
+ WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
+ inst->analysisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
+ + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+
+ // Window data before FFT.
+ for (i = 0; i < inst->anaLen; i++) {
+ out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ }
+}
+
+// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
+// zeros, and normalize it.
+static __inline void CreateComplexBufferC(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out) {
+ int i = 0, j = 0;
+ for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
+ out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ out[j + 1] = 0; // Insert zeros in imaginary part
+ }
+}
+
+void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
+ WebRtc_Word16 pink_noise_exp_avg,
+ WebRtc_Word32 pink_noise_num_avg,
+ int freq_index,
+ WebRtc_UWord32* noise_estimate,
+ WebRtc_UWord32* noise_estimate_avg) {
+ WebRtc_Word32 tmp32no1 = 0;
+ WebRtc_Word32 tmp32no2 = 0;
+
+ WebRtc_Word16 int_part = 0;
+ WebRtc_Word16 frac_part = 0;
+
+ // Use pink noise estimate
+ // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
+ assert(freq_index >= 0);
+ assert(freq_index < 129);
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(pink_noise_exp_avg, kLogIndex[freq_index]); // Q26
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 15); // Q11
+ tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
+
+ // Calculate output: 2^tmp32no1
+ // Output in Q(minNorm-stages)
+ tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11);
+ if (tmp32no1 > 0) {
+ int_part = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
+ frac_part = (WebRtc_Word16)(tmp32no1 & 0x000007ff); // Q11
+ // Piecewise linear approximation of 'b' in
+ // 2^(int_part+frac_part) = 2^int_part * (1 + b)
+ // 'b' is given in Q11 and below stored in frac_part.
+ if (WEBRTC_SPL_RSHIFT_W16(frac_part, 10)) {
+ // Upper fractional part
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(2048 - frac_part, 1244); // Q21
+ tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
+ } else {
+ // Lower fractional part
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(frac_part, 804), 10);
+ }
+ // Shift fractional part to Q(minNorm-stages)
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
+ *noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (WebRtc_UWord32)tmp32no2;
+ // Scale up to initMagnEst, which is not block averaged
+ *noise_estimate = (*noise_estimate_avg) * (WebRtc_UWord32)(inst->blockIndex + 1);
+ }
+}
+
+// Initialize state
+WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs) {
+ int i;
+
+ //check for valid pointer
+ if (inst == NULL) {
+ return -1;
+ }
+ //
+
+ // Initialization of struct
+ if (fs == 8000 || fs == 16000 || fs == 32000) {
+ inst->fs = fs;
+ } else {
+ return -1;
+ }
+
+ if (fs == 8000) {
+ inst->blockLen10ms = 80;
+ inst->anaLen = 128;
+ inst->stages = 7;
+ inst->window = kBlocks80w128x;
+ inst->thresholdLogLrt = 131072; //default threshold for LRT feature
+ inst->maxLrt = 0x0040000;
+ inst->minLrt = 52429;
+ } else if (fs == 16000) {
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->stages = 8;
+ inst->window = kBlocks160w256x;
+ inst->thresholdLogLrt = 212644; //default threshold for LRT feature
+ inst->maxLrt = 0x0080000;
+ inst->minLrt = 104858;
+ } else if (fs == 32000) {
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->stages = 8;
+ inst->window = kBlocks160w256x;
+ inst->thresholdLogLrt = 212644; //default threshold for LRT feature
+ inst->maxLrt = 0x0080000;
+ inst->minLrt = 104858;
+ }
+ inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
+ inst->magnLen = inst->anaLen2 + 1;
+
+ WebRtcSpl_ZerosArrayW16(inst->analysisBuffer, ANAL_BLOCKL_MAX);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer, ANAL_BLOCKL_MAX);
+
+ // for HB processing
+ WebRtcSpl_ZerosArrayW16(inst->dataBufHBFX, ANAL_BLOCKL_MAX);
+ // for quantile noise estimation
+ WebRtcSpl_ZerosArrayW16(inst->noiseEstQuantile, HALF_ANAL_BLOCKL);
+ for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
+ inst->noiseEstLogQuantile[i] = 2048; // Q8
+ inst->noiseEstDensity[i] = 153; // Q9
+ }
+ for (i = 0; i < SIMULT; i++) {
+ inst->noiseEstCounter[i] = (WebRtc_Word16)(END_STARTUP_LONG * (i + 1)) / SIMULT;
+ }
+
+ // Initialize suppression filter with ones
+ WebRtcSpl_MemSetW16((WebRtc_Word16*)inst->noiseSupFilter, 16384, HALF_ANAL_BLOCKL);
+
+ // Set the aggressiveness: default
+ inst->aggrMode = 0;
+
+ //initialize variables for new method
+ inst->priorNonSpeechProb = 8192; // Q14(0.5) prior probability for speech/noise
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->prevMagnU16[i] = 0;
+ inst->prevNoiseU32[i] = 0; //previous noise-spectrum
+ inst->logLrtTimeAvgW32[i] = 0; //smooth LR ratio
+ inst->avgMagnPause[i] = 0; //conservative noise spectrum estimate
+ inst->initMagnEst[i] = 0; //initial average magnitude spectrum
+ }
+
+ //feature quantities
+ inst->thresholdSpecDiff = 50; //threshold for difference feature: determined on-line
+ inst->thresholdSpecFlat = 20480; //threshold for flatness: determined on-line
+ inst->featureLogLrt = inst->thresholdLogLrt; //average LRT factor (= threshold)
+ inst->featureSpecFlat = inst->thresholdSpecFlat; //spectral flatness (= threshold)
+ inst->featureSpecDiff = inst->thresholdSpecDiff; //spectral difference (= threshold)
+ inst->weightLogLrt = 6; //default weighting par for LRT feature
+ inst->weightSpecFlat = 0; //default weighting par for spectral flatness feature
+ inst->weightSpecDiff = 0; //default weighting par for spectral difference feature
+
+ inst->curAvgMagnEnergy = 0; //window time-average of input magnitude spectrum
+ inst->timeAvgMagnEnergy = 0; //normalization for spectral difference
+ inst->timeAvgMagnEnergyTmp = 0; //normalization for spectral difference
+
+ //histogram quantities: used to estimate/update thresholds for features
+ WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
+
+ inst->blockIndex = -1; //frame counter
+
+ //inst->modelUpdate = 500; //window for update
+ inst->modelUpdate = (1 << STAT_UPDATES); //window for update
+ inst->cntThresUpdate = 0; //counter feature thresholds updates
+
+ inst->sumMagn = 0;
+ inst->magnEnergy = 0;
+ inst->prevQMagn = 0;
+ inst->qNoise = 0;
+ inst->prevQNoise = 0;
+
+ inst->energyIn = 0;
+ inst->scaleEnergyIn = 0;
+
+ inst->whiteNoiseLevel = 0;
+ inst->pinkNoiseNumerator = 0;
+ inst->pinkNoiseExp = 0;
+ inst->minNorm = 15; // Start with full scale
+ inst->zeroInputSignal = 0;
+
+ //default mode
+ WebRtcNsx_set_policy_core(inst, 0);
+
+#ifdef NS_FILEDEBUG
+ inst->infile = fopen("indebug.pcm", "wb");
+ inst->outfile = fopen("outdebug.pcm", "wb");
+ inst->file1 = fopen("file1.pcm", "wb");
+ inst->file2 = fopen("file2.pcm", "wb");
+ inst->file3 = fopen("file3.pcm", "wb");
+ inst->file4 = fopen("file4.pcm", "wb");
+ inst->file5 = fopen("file5.pcm", "wb");
+#endif
+
+ // Initialize function pointers.
+ WebRtcNsx_NoiseEstimation = NoiseEstimationC;
+ WebRtcNsx_PrepareSpectrum = PrepareSpectrumC;
+ WebRtcNsx_SynthesisUpdate = SynthesisUpdateC;
+ WebRtcNsx_AnalysisUpdate = AnalysisUpdateC;
+ WebRtcNsx_Denormalize = DenormalizeC;
+ WebRtcNsx_CreateComplexBuffer = CreateComplexBufferC;
+
+#ifdef WEBRTC_DETECT_ARM_NEON
+ uint64_t features = WebRtc_GetCPUFeaturesARM();
+ if ((features & kCPUFeatureNEON) != 0)
+ {
+ WebRtcNsx_InitNeon();
+ }
+#elif defined(WEBRTC_ARCH_ARM_NEON)
+ WebRtcNsx_InitNeon();
+#endif
+
+ inst->initFlag = 1;
+
+ return 0;
+}
+
+int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode) {
+ // allow for modes:0,1,2,3
+ if (mode < 0 || mode > 3) {
+ return -1;
+ }
+
+ inst->aggrMode = mode;
+ if (mode == 0) {
+ inst->overdrive = 256; // Q8(1.0)
+ inst->denoiseBound = 8192; // Q14(0.5)
+ inst->gainMap = 0; // No gain compensation
+ } else if (mode == 1) {
+ inst->overdrive = 256; // Q8(1.0)
+ inst->denoiseBound = 4096; // Q14(0.25)
+ inst->factor2Table = kFactor2Aggressiveness1;
+ inst->gainMap = 1;
+ } else if (mode == 2) {
+ inst->overdrive = 282; // ~= Q8(1.1)
+ inst->denoiseBound = 2048; // Q14(0.125)
+ inst->factor2Table = kFactor2Aggressiveness2;
+ inst->gainMap = 1;
+ } else if (mode == 3) {
+ inst->overdrive = 320; // Q8(1.25)
+ inst->denoiseBound = 1475; // ~= Q14(0.09)
+ inst->factor2Table = kFactor2Aggressiveness3;
+ inst->gainMap = 1;
+ }
+ return 0;
+}
+
+// Extract thresholds for feature parameters
+// histograms are computed over some window_size (given by window_pars)
+// thresholds and weights are extracted every window
+// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
+// threshold and weights are returned in: inst->priorModelPars
+void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
+ WebRtc_UWord32 tmpU32;
+ WebRtc_UWord32 histIndex;
+ WebRtc_UWord32 posPeak1SpecFlatFX, posPeak2SpecFlatFX;
+ WebRtc_UWord32 posPeak1SpecDiffFX, posPeak2SpecDiffFX;
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word32 fluctLrtFX, thresFluctLrtFX;
+ WebRtc_Word32 avgHistLrtFX, avgSquareHistLrtFX, avgHistLrtComplFX;
+
+ WebRtc_Word16 j;
+ WebRtc_Word16 numHistLrt;
+
+ int i;
+ int useFeatureSpecFlat, useFeatureSpecDiff, featureSum;
+ int maxPeak1, maxPeak2;
+ int weightPeak1SpecFlat, weightPeak2SpecFlat;
+ int weightPeak1SpecDiff, weightPeak2SpecDiff;
+
+ //update histograms
+ if (!flag) {
+ // LRT
+ // Type casting to UWord32 is safe since negative values will not be wrapped to larger
+ // values than HIST_PAR_EST
+ histIndex = (WebRtc_UWord32)(inst->featureLogLrt);
+ if (histIndex < HIST_PAR_EST) {
+ inst->histLrt[histIndex]++;
+ }
+ // Spectral flatness
+ // (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
+ histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
+ if (histIndex < HIST_PAR_EST) {
+ inst->histSpecFlat[histIndex]++;
+ }
+ // Spectral difference
+ histIndex = HIST_PAR_EST;
+ if (inst->timeAvgMagnEnergy > 0) {
+ // Guard against division by zero
+ // If timeAvgMagnEnergy == 0 we have no normalizing statistics and
+ // therefore can't update the histogram
+ histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
+ inst->timeAvgMagnEnergy);
+ }
+ if (histIndex < HIST_PAR_EST) {
+ inst->histSpecDiff[histIndex]++;
+ }
+ }
+
+ // extract parameters for speech/noise probability
+ if (flag) {
+ useFeatureSpecDiff = 1;
+ //for LRT feature:
+ // compute the average over inst->featureExtractionParams.rangeAvgHistLrt
+ avgHistLrtFX = 0;
+ avgSquareHistLrtFX = 0;
+ numHistLrt = 0;
+ for (i = 0; i < BIN_SIZE_LRT; i++) {
+ j = (2 * i + 1);
+ tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
+ avgHistLrtFX += tmp32;
+ numHistLrt += inst->histLrt[i];
+ avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
+ }
+ avgHistLrtComplFX = avgHistLrtFX;
+ for (; i < HIST_PAR_EST; i++) {
+ j = (2 * i + 1);
+ tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
+ avgHistLrtComplFX += tmp32;
+ avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
+ }
+ fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
+ fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
+ thresFluctLrtFX = THRES_FLUCT_LRT * numHistLrt;
+ // get threshold for LRT feature:
+ tmpU32 = (FACTOR_1_LRT_DIFF * (WebRtc_UWord32)avgHistLrtFX);
+ if ((fluctLrtFX < thresFluctLrtFX) || (numHistLrt == 0) ||
+ (tmpU32 > (WebRtc_UWord32)(100 * numHistLrt))) {
+ //very low fluctuation, so likely noise
+ inst->thresholdLogLrt = inst->maxLrt;
+ } else {
+ tmp32 = (WebRtc_Word32)((tmpU32 << (9 + inst->stages)) / numHistLrt /
+ 25);
+ // check if value is within min/max range
+ inst->thresholdLogLrt = WEBRTC_SPL_SAT(inst->maxLrt,
+ tmp32,
+ inst->minLrt);
+ }
+ if (fluctLrtFX < thresFluctLrtFX) {
+ // Do not use difference feature if fluctuation of LRT feature is very low:
+ // most likely just noise state
+ useFeatureSpecDiff = 0;
+ }
+
+ // for spectral flatness and spectral difference: compute the main peaks of histogram
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecFlatFX = 0;
+ posPeak2SpecFlatFX = 0;
+ weightPeak1SpecFlat = 0;
+ weightPeak2SpecFlat = 0;
+
+ // peaks for flatness
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ if (inst->histSpecFlat[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecFlat = weightPeak1SpecFlat;
+ posPeak2SpecFlatFX = posPeak1SpecFlatFX;
+
+ maxPeak1 = inst->histSpecFlat[i];
+ weightPeak1SpecFlat = inst->histSpecFlat[i];
+ posPeak1SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
+ } else if (inst->histSpecFlat[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecFlat[i];
+ weightPeak2SpecFlat = inst->histSpecFlat[i];
+ posPeak2SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
+ }
+ }
+
+ // for spectral flatness feature
+ useFeatureSpecFlat = 1;
+ // merge the two peaks if they are close
+ if ((posPeak1SpecFlatFX - posPeak2SpecFlatFX < LIM_PEAK_SPACE_FLAT_DIFF)
+ && (weightPeak2SpecFlat * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecFlat)) {
+ weightPeak1SpecFlat += weightPeak2SpecFlat;
+ posPeak1SpecFlatFX = (posPeak1SpecFlatFX + posPeak2SpecFlatFX) >> 1;
+ }
+ //reject if weight of peaks is not large enough, or peak value too small
+ if (weightPeak1SpecFlat < THRES_WEIGHT_FLAT_DIFF || posPeak1SpecFlatFX
+ < THRES_PEAK_FLAT) {
+ useFeatureSpecFlat = 0;
+ } else { // if selected, get the threshold
+ // compute the threshold and check if value is within min/max range
+ inst->thresholdSpecFlat = WEBRTC_SPL_SAT(MAX_FLAT_Q10, FACTOR_2_FLAT_Q10
+ * posPeak1SpecFlatFX, MIN_FLAT_Q10); //Q10
+ }
+ // done with flatness feature
+
+ if (useFeatureSpecDiff) {
+ //compute two peaks for spectral difference
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecDiffFX = 0;
+ posPeak2SpecDiffFX = 0;
+ weightPeak1SpecDiff = 0;
+ weightPeak2SpecDiff = 0;
+ // peaks for spectral difference
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ if (inst->histSpecDiff[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecDiff = weightPeak1SpecDiff;
+ posPeak2SpecDiffFX = posPeak1SpecDiffFX;
+
+ maxPeak1 = inst->histSpecDiff[i];
+ weightPeak1SpecDiff = inst->histSpecDiff[i];
+ posPeak1SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
+ } else if (inst->histSpecDiff[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecDiff[i];
+ weightPeak2SpecDiff = inst->histSpecDiff[i];
+ posPeak2SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
+ }
+ }
+
+ // merge the two peaks if they are close
+ if ((posPeak1SpecDiffFX - posPeak2SpecDiffFX < LIM_PEAK_SPACE_FLAT_DIFF)
+ && (weightPeak2SpecDiff * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecDiff)) {
+ weightPeak1SpecDiff += weightPeak2SpecDiff;
+ posPeak1SpecDiffFX = (posPeak1SpecDiffFX + posPeak2SpecDiffFX) >> 1;
+ }
+ // get the threshold value and check if value is within min/max range
+ inst->thresholdSpecDiff = WEBRTC_SPL_SAT(MAX_DIFF, FACTOR_1_LRT_DIFF
+ * posPeak1SpecDiffFX, MIN_DIFF); //5x bigger
+ //reject if weight of peaks is not large enough
+ if (weightPeak1SpecDiff < THRES_WEIGHT_FLAT_DIFF) {
+ useFeatureSpecDiff = 0;
+ }
+ // done with spectral difference feature
+ }
+
+ // select the weights between the features
+ // inst->priorModelPars[4] is weight for LRT: always selected
+ featureSum = 6 / (1 + useFeatureSpecFlat + useFeatureSpecDiff);
+ inst->weightLogLrt = featureSum;
+ inst->weightSpecFlat = useFeatureSpecFlat * featureSum;
+ inst->weightSpecDiff = useFeatureSpecDiff * featureSum;
+
+ // set histograms to zero for next update
+ WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
+ } // end of flag == 1
+}
+
+
+// Compute spectral flatness on input spectrum
+// magn is the magnitude spectrum
+// spectral flatness is returned in inst->featureSpecFlat
+void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, WebRtc_UWord16* magn) {
+ WebRtc_UWord32 tmpU32;
+ WebRtc_UWord32 avgSpectralFlatnessNum, avgSpectralFlatnessDen;
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word32 currentSpectralFlatness, logCurSpectralFlatness;
+
+ WebRtc_Word16 zeros, frac, intPart;
+
+ int i;
+
+ // for flatness
+ avgSpectralFlatnessNum = 0;
+ avgSpectralFlatnessDen = inst->sumMagn - (WebRtc_UWord32)magn[0]; // Q(normData-stages)
+
+ // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
+ // flatness = exp( sum(log(magn[i]))/N - log(sum(magn[i])/N) )
+ // = exp( sum(log(magn[i]))/N ) * N / sum(magn[i])
+ // = 2^( sum(log2(magn[i]))/N - (log2(sum(magn[i])) - log2(N)) ) [This is used]
+ for (i = 1; i < inst->magnLen; i++) {
+ // First bin is excluded from spectrum measures. Number of bins is now a power of 2
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
+ frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_UWord32)(magn[i]) << zeros)
+ & 0x7FFFFFFF) >> 23);
+ // log2(magn(i))
+ assert(frac < 256);
+ tmpU32 = (WebRtc_UWord32)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]); // Q8
+ avgSpectralFlatnessNum += tmpU32; // Q8
+ } else {
+ //if at least one frequency component is zero, treat separately
+ tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
+ inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
+ return;
+ }
+ }
+ //ratio and inverse log: check for case of log(0)
+ zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
+ frac = (WebRtc_Word16)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
+ // log2(avgSpectralFlatnessDen)
+ assert(frac < 256);
+ tmp32 = (WebRtc_Word32)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]); // Q8
+ logCurSpectralFlatness = (WebRtc_Word32)avgSpectralFlatnessNum;
+ logCurSpectralFlatness += ((WebRtc_Word32)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
+ logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
+ logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
+ tmp32 = (WebRtc_Word32)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
+ & 0x0001FFFF)); //Q17
+ intPart = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
+ intPart += 7; // Shift 7 to get the output in Q10 (from Q17 = -17+10)
+ if (intPart > 0) {
+ currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
+ } else {
+ currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
+ }
+
+ //time average update of spectral flatness feature
+ tmp32 = currentSpectralFlatness - (WebRtc_Word32)inst->featureSpecFlat; // Q10
+ tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
+ inst->featureSpecFlat = (WebRtc_UWord32)((WebRtc_Word32)inst->featureSpecFlat
+ + WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
+ // done with flatness feature
+}
+
+
+// Compute the difference measure between input spectrum and a template/learned noise spectrum
+// magn_tmp is the input spectrum
+// the reference/template spectrum is inst->magn_avg_pause[i]
+// returns (normalized) spectral difference in inst->featureSpecDiff
+void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, WebRtc_UWord16* magnIn) {
+ // This is to be calculated:
+ // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+ WebRtc_UWord32 varMagnUFX, varPauseUFX, avgDiffNormMagnUFX;
+
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 avgPauseFX, avgMagnFX, covMagnPauseFX;
+ WebRtc_Word32 maxPause, minPause;
+
+ WebRtc_Word16 tmp16no1;
+
+ int i, norm32, nShifts;
+
+ avgPauseFX = 0;
+ maxPause = 0;
+ minPause = inst->avgMagnPause[0]; // Q(prevQMagn)
+ // compute average quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ // Compute mean of magn_pause
+ avgPauseFX += inst->avgMagnPause[i]; // in Q(prevQMagn)
+ maxPause = WEBRTC_SPL_MAX(maxPause, inst->avgMagnPause[i]);
+ minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]);
+ }
+ // normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
+ avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
+ avgMagnFX = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
+ // Largest possible deviation in magnPause for (co)var calculations
+ tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
+ // Get number of shifts to make sure we don't get wrap around in varPause
+ nShifts = WEBRTC_SPL_MAX(0, 10 + inst->stages - WebRtcSpl_NormW32(tmp32no1));
+
+ varMagnUFX = 0;
+ varPauseUFX = 0;
+ covMagnPauseFX = 0;
+ for (i = 0; i < inst->magnLen; i++) {
+ // Compute var and cov of magn and magn_pause
+ tmp16no1 = (WebRtc_Word16)((WebRtc_Word32)magnIn[i] - avgMagnFX);
+ tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
+ varMagnUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
+ covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
+ varPauseUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
+ }
+ //update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
+ inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
+ + inst->stages - 1);
+
+ avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
+ if ((varPauseUFX) && (covMagnPauseFX)) {
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
+ norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
+ if (norm32 > 0) {
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
+ } else {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
+ }
+ tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
+
+ nShifts += norm32;
+ nShifts <<= 1;
+ if (nShifts < 0) {
+ varPauseUFX >>= (-nShifts); // Q(2*(qMagn+norm32+minPause))
+ nShifts = 0;
+ }
+ if (varPauseUFX > 0) {
+ // Q(2*(qMagn+norm32-16+minPause))
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX);
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
+
+ // Q(2*qMagn)
+ avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1);
+ } else {
+ avgDiffNormMagnUFX = 0;
+ }
+ }
+ //normalize and compute time average update of difference feature
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
+ if (inst->featureSpecDiff > tmpU32no1) {
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
+ SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
+ inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
+ SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
+ inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
+ }
+}
+
+// Compute speech/noise probability
+// speech/noise probability is returned in: probSpeechFinal
+//snrLocPrior is the prior SNR for each frequency (in Q11)
+//snrLocPost is the post SNR for each frequency (in Q11)
+void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, WebRtc_UWord16* nonSpeechProbFinal,
+ WebRtc_UWord32* priorLocSnr, WebRtc_UWord32* postLocSnr) {
+ WebRtc_UWord32 zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
+
+ WebRtc_Word32 invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
+ WebRtc_Word32 frac32, logTmp;
+ WebRtc_Word32 logLrtTimeAvgKsumFX;
+
+ WebRtc_Word16 indPriorFX16;
+ WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
+
+ int i, normTmp, normTmp2, nShifts;
+
+ // compute feature based on average LR factor
+ // this is the average over all frequencies of the smooth log LRT
+ logLrtTimeAvgKsumFX = 0;
+ for (i = 0; i < inst->magnLen; i++) {
+ besselTmpFX32 = (WebRtc_Word32)postLocSnr[i]; // Q11
+ normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
+ num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp)
+ if (normTmp > 10) {
+ den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
+ } else {
+ den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
+ }
+ if (den > 0) {
+ besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
+ } else {
+ besselTmpFX32 -= num; // Q11
+ }
+
+ // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]);
+ // Here, LRT_TAVG = 0.5
+ zeros = WebRtcSpl_NormU32(priorLocSnr[i]);
+ frac32 = (WebRtc_Word32)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19);
+ tmp32 = WEBRTC_SPL_MUL(frac32, frac32);
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19);
+ tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)frac32, 5412, 12);
+ frac32 = tmp32 + 37;
+ // tmp32 = log2(priorLocSnr[i])
+ tmp32 = (WebRtc_Word32)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
+ logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12
+ inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
+
+ logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12
+ }
+ inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2
+ // done with computation of LR factor
+
+ //
+ //compute the indicator functions
+ //
+
+ // average LRT feature
+ // FLOAT code
+ // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0);
+ tmpIndFX = 16384; // Q14(1.0)
+ tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12
+ nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5;
+ //use larger width in tanh map for pause regions
+ if (tmp32no1 < 0) {
+ tmpIndFX = 0;
+ tmp32no1 = -tmp32no1;
+ //widthPrior = widthPrior * 2.0;
+ nShifts++;
+ }
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
+ // compute indicator function: sigmoid map
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14);
+ if ((tableIndex < 16) && (tableIndex >= 0)) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmp32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
+ if (tmpIndFX == 0) {
+ tmpIndFX = 8192 - tmp16no2; // Q14
+ } else {
+ tmpIndFX = 8192 + tmp16no2; // Q14
+ }
+ }
+ indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14
+
+ //spectral flatness feature
+ if (inst->weightSpecFlat) {
+ tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10
+ tmpIndFX = 16384; // Q14(1.0)
+ //use larger width in tanh map for pause regions
+ tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10
+ nShifts = 4;
+ if (inst->thresholdSpecFlat < tmpU32no1) {
+ tmpIndFX = 0;
+ tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat;
+ //widthPrior = widthPrior * 2.0;
+ nShifts++;
+ }
+ tmp32no1 = (WebRtc_Word32)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
+ nShifts), 25); //Q14
+ tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14
+ // compute indicator function: sigmoid map
+ // FLOAT code
+ // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0);
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
+ if (tableIndex < 16) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
+ if (tmpIndFX) {
+ tmpIndFX = 8192 + tmp16no2; // Q14
+ } else {
+ tmpIndFX = 8192 - tmp16no2; // Q14
+ }
+ }
+ indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14
+ }
+
+ //for template spectral-difference
+ if (inst->weightSpecDiff) {
+ tmpU32no1 = 0;
+ if (inst->featureSpecDiff) {
+ normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
+ WebRtcSpl_NormU32(inst->featureSpecDiff));
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages
+ - normTmp);
+ if (tmpU32no2 > 0) {
+ // Q(20 - inst->stages)
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2);
+ } else {
+ tmpU32no1 = (WebRtc_UWord32)(0x7fffffff);
+ }
+ }
+ tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25);
+ tmpU32no2 = tmpU32no1 - tmpU32no3;
+ nShifts = 1;
+ tmpIndFX = 16384; // Q14(1.0)
+ //use larger width in tanh map for pause regions
+ if (tmpU32no2 & 0x80000000) {
+ tmpIndFX = 0;
+ tmpU32no2 = tmpU32no3 - tmpU32no1;
+ //widthPrior = widthPrior * 2.0;
+ nShifts--;
+ }
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
+ // compute indicator function: sigmoid map
+ /* FLOAT code
+ indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
+ */
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
+ if (tableIndex < 16) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ tmp16no1, frac, 14);
+ if (tmpIndFX) {
+ tmpIndFX = 8192 + tmp16no2;
+ } else {
+ tmpIndFX = 8192 - tmp16no2;
+ }
+ }
+ indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14
+ }
+
+ //combine the indicator function with the feature weights
+ // FLOAT code
+ // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2);
+ indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14
+ // done with computing indicator function
+
+ //compute the prior probability
+ // FLOAT code
+ // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb);
+ tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14
+ inst->priorNonSpeechProb += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ PRIOR_UPDATE_Q14, tmp16, 14); // Q14
+
+ //final speech probability: combine prior model with LR factor:
+
+ memset(nonSpeechProbFinal, 0, sizeof(WebRtc_UWord16) * inst->magnLen);
+
+ if (inst->priorNonSpeechProb > 0) {
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT code
+ // invLrt = exp(inst->logLrtTimeAvg[i]);
+ // invLrt = inst->priorSpeechProb * invLrt;
+ // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt);
+ // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt;
+ // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt);
+ if (inst->logLrtTimeAvgW32[i] < 65300) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637),
+ 14); // Q12
+ intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12);
+ if (intPart < -8) {
+ intPart = -8;
+ }
+ frac = (WebRtc_Word16)(tmp32no1 & 0x00000fff); // Q12
+
+ // Quadratic approximation of 2^frac
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
+ tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
+ invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
+ + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
+
+ normTmp = WebRtcSpl_NormW32(invLrtFX);
+ normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
+ if (normTmp + normTmp2 >= 7) {
+ if (normTmp + normTmp2 < 15) {
+ invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp);
+ // Q(normTmp+normTmp2-7)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb));
+ // Q(normTmp+normTmp2+7)
+ invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14
+ } else {
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22
+ invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
+ }
+
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22
+
+ nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1,
+ (WebRtc_Word32)inst->priorNonSpeechProb + invLrtFX); // Q8
+ }
+ }
+ }
+ }
+}
+
+// Transform input (speechFrame) to frequency domain magnitude (magnU16)
+void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, WebRtc_UWord16* magnU16) {
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+
+ WebRtc_Word32 tmp_1_w32 = 0;
+ WebRtc_Word32 tmp_2_w32 = 0;
+ WebRtc_Word32 sum_log_magn = 0;
+ WebRtc_Word32 sum_log_i_log_magn = 0;
+
+ WebRtc_UWord16 sum_log_magn_u16 = 0;
+ WebRtc_UWord16 tmp_u16 = 0;
+
+ WebRtc_Word16 sum_log_i = 0;
+ WebRtc_Word16 sum_log_i_square = 0;
+ WebRtc_Word16 frac = 0;
+ WebRtc_Word16 log2 = 0;
+ WebRtc_Word16 matrix_determinant = 0;
+ WebRtc_Word16 winData[ANAL_BLOCKL_MAX], maxWinData;
+ WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
+
+ int i, j;
+ int zeros;
+ int net_norm = 0;
+ int right_shifts_in_magnU16 = 0;
+ int right_shifts_in_initMagnEst = 0;
+
+ // Update analysis buffer for lower band, and window data before FFT.
+ WebRtcNsx_AnalysisUpdate(inst, winData, speechFrame);
+
+ // Get input energy
+ inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
+
+ // Reset zero input flag
+ inst->zeroInputSignal = 0;
+ // Acquire norm for winData
+ maxWinData = WebRtcSpl_MaxAbsValueW16(winData, inst->anaLen);
+ inst->normData = WebRtcSpl_NormW16(maxWinData);
+ if (maxWinData == 0) {
+ // Treat zero input separately.
+ inst->zeroInputSignal = 1;
+ return;
+ }
+
+ // Determine the net normalization in the frequency domain
+ net_norm = inst->stages - inst->normData;
+ // Track lowest normalization factor and use it to prevent wrap around in shifting
+ right_shifts_in_magnU16 = inst->normData - inst->minNorm;
+ right_shifts_in_initMagnEst = WEBRTC_SPL_MAX(-right_shifts_in_magnU16, 0);
+ inst->minNorm -= right_shifts_in_initMagnEst;
+ right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0);
+
+ // create realImag as winData interleaved with zeros (= imag. part), normalize it
+ WebRtcNsx_CreateComplexBuffer(inst, winData, realImag);
+
+ // bit-reverse position of elements in array and FFT the array
+ WebRtcSpl_ComplexBitReverse(realImag, inst->stages); // Q(normData-stages)
+ WebRtcSpl_ComplexFFT(realImag, inst->stages, 1);
+
+ inst->imag[0] = 0; // Q(normData-stages)
+ inst->imag[inst->anaLen2] = 0;
+ inst->real[0] = realImag[0]; // Q(normData-stages)
+ inst->real[inst->anaLen2] = realImag[inst->anaLen];
+ // Q(2*(normData-stages))
+ inst->magnEnergy = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[0], inst->real[0]);
+ inst->magnEnergy += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[inst->anaLen2],
+ inst->real[inst->anaLen2]);
+ magnU16[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[0]); // Q(normData-stages)
+ magnU16[inst->anaLen2] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[inst->anaLen2]);
+ inst->sumMagn = (WebRtc_UWord32)magnU16[0]; // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[inst->anaLen2];
+
+ if (inst->blockIndex >= END_STARTUP_SHORT) {
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ inst->real[i] = realImag[j];
+ inst->imag[i] = -realImag[j + 1];
+ // magnitude spectrum
+ // energy in Q(2*(normData-stages))
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
+ tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
+ inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
+
+ magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
+ }
+ } else {
+ //
+ // Gather information during startup for noise parameter estimation
+ //
+
+ // Switch initMagnEst to Q(minNorm-stages)
+ inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
+ right_shifts_in_initMagnEst);
+ inst->initMagnEst[inst->anaLen2] =
+ WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
+ right_shifts_in_initMagnEst); // Q(minNorm-stages)
+
+ // Shift magnU16 to same domain as initMagnEst
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[0],
+ right_shifts_in_magnU16); // Q(minNorm-stages)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[inst->anaLen2],
+ right_shifts_in_magnU16); // Q(minNorm-stages)
+
+ // Update initMagnEst
+ inst->initMagnEst[0] += tmpU32no1; // Q(minNorm-stages)
+ inst->initMagnEst[inst->anaLen2] += tmpU32no2; // Q(minNorm-stages)
+
+ log2 = 0;
+ if (magnU16[inst->anaLen2]) {
+ // Calculate log2(magnU16[inst->anaLen2])
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[inst->anaLen2]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[inst->anaLen2] << zeros) &
+ 0x7FFFFFFF) >> 23); // Q8
+ // log2(magnU16(i)) in Q8
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]);
+ }
+
+ sum_log_magn = (WebRtc_Word32)log2; // Q8
+ // sum_log_i_log_magn in Q17
+ sum_log_i_log_magn = (WEBRTC_SPL_MUL_16_16(kLogIndex[inst->anaLen2], log2) >> 3);
+
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ inst->real[i] = realImag[j];
+ inst->imag[i] = -realImag[j + 1];
+ // magnitude spectrum
+ // energy in Q(2*(normData-stages))
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
+ tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
+ inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
+
+ magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
+
+ // Switch initMagnEst to Q(minNorm-stages)
+ inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
+ right_shifts_in_initMagnEst);
+
+ // Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[i],
+ right_shifts_in_magnU16);
+ // Update initMagnEst
+ inst->initMagnEst[i] += tmpU32no1; // Q(minNorm-stages)
+
+ if (i >= kStartBand) {
+ // For pink noise estimation. Collect data neglecting lower frequency band
+ log2 = 0;
+ if (magnU16[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[i]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[i] << zeros) &
+ 0x7FFFFFFF) >> 23);
+ // log2(magnU16(i)) in Q8
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ }
+ sum_log_magn += (WebRtc_Word32)log2; // Q8
+ // sum_log_i_log_magn in Q17
+ sum_log_i_log_magn += (WEBRTC_SPL_MUL_16_16(kLogIndex[i], log2) >> 3);
+ }
+ }
+
+ //
+ //compute simplified noise model during startup
+ //
+
+ // Estimate White noise
+
+ // Switch whiteNoiseLevel to Q(minNorm-stages)
+ inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
+ right_shifts_in_initMagnEst);
+
+ // Update the average magnitude spectrum, used as noise estimate.
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
+
+ // Replacing division above with 'stages' shifts
+ // Shift to same Q-domain as whiteNoiseLevel
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
+ // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
+ assert(END_STARTUP_SHORT < 128);
+ inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
+
+ // Estimate Pink noise parameters
+ // Denominator used in both parameter estimates.
+ // The value is only dependent on the size of the frequency band (kStartBand)
+ // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
+ assert(kStartBand < 66);
+ matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
+ sum_log_i = kSumLogIndex[kStartBand]; // Q5
+ sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
+ if (inst->fs == 8000) {
+ // Adjust values to shorter blocks in narrow band.
+ tmp_1_w32 = (WebRtc_Word32)matrix_determinant;
+ tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
+ tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
+ tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4);
+ tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)
+ (inst->magnLen - kStartBand), kSumSquareLogIndex[65], 2);
+ matrix_determinant = (WebRtc_Word16)tmp_1_w32;
+ sum_log_i -= kSumLogIndex[65]; // Q5
+ sum_log_i_square -= kSumSquareLogIndex[65]; // Q2
+ }
+
+ // Necessary number of shifts to fit sum_log_magn in a word16
+ zeros = 16 - WebRtcSpl_NormW32(sum_log_magn);
+ if (zeros < 0) {
+ zeros = 0;
+ }
+ tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
+ sum_log_magn_u16 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
+
+ // Calculate and update pinkNoiseNumerator. Result in Q11.
+ tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((WebRtc_UWord32)sum_log_i_log_magn, 12); // Q5
+
+ // Shift the largest value of sum_log_i and tmp32no3 before multiplication
+ tmp_u16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)sum_log_i, 1); // Q6
+ if ((WebRtc_UWord32)sum_log_i > tmpU32no1) {
+ tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros);
+ } else {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
+ }
+ tmp_2_w32 -= (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
+ matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
+ tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
+ tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11
+ if (tmp_2_w32 < 0) {
+ tmp_2_w32 = 0;
+ }
+ inst->pinkNoiseNumerator += tmp_2_w32; // Q11
+
+ // Calculate and update pinkNoiseExp. Result in Q14.
+ tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i, sum_log_magn_u16); // Q(14-zeros)
+ tmp_1_w32 = WEBRTC_SPL_RSHIFT_W32(sum_log_i_log_magn, 3 + zeros);
+ tmp_1_w32 = WEBRTC_SPL_MUL((WebRtc_Word32)(inst->magnLen - kStartBand),
+ tmp_1_w32);
+ tmp_2_w32 -= tmp_1_w32; // Q(14-zeros)
+ if (tmp_2_w32 > 0) {
+ // If the exponential parameter is negative force it to zero, which means a
+ // flat spectrum.
+ tmp_1_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q14
+ inst->pinkNoiseExp += WEBRTC_SPL_SAT(16384, tmp_1_w32, 0); // Q14
+ }
+ }
+}
+
+void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) {
+ WebRtc_Word32 energyOut;
+
+ WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
+ WebRtc_Word16 tmp16no1, tmp16no2;
+ WebRtc_Word16 energyRatio;
+ WebRtc_Word16 gainFactor, gainFactor1, gainFactor2;
+
+ int i;
+ int outCIFFT;
+ int scaleEnergyOut = 0;
+
+ if (inst->zeroInputSignal) {
+ // synthesize the special case of zero input
+ // read out fully processed segment
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrame[i] = inst->synthesisBuffer[i]; // Q0
+ }
+ // update synthesis buffer
+ WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ inst->synthesisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
+ inst->blockLen10ms);
+ return;
+ }
+
+ // Filter the data in the frequency domain, and create spectrum.
+ WebRtcNsx_PrepareSpectrum(inst, realImag);
+
+ // bit-reverse position of elements in array and IFFT it
+ WebRtcSpl_ComplexBitReverse(realImag, inst->stages);
+ outCIFFT = WebRtcSpl_ComplexIFFT(realImag, inst->stages, 1);
+
+ // Denormalize.
+ WebRtcNsx_Denormalize(inst, realImag, outCIFFT);
+
+ //scale factor: only do it after END_STARTUP_LONG time
+ gainFactor = 8192; // 8192 = Q13(1.0)
+ if (inst->gainMap == 1 &&
+ inst->blockIndex > END_STARTUP_LONG &&
+ inst->energyIn > 0) {
+ energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
+ if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000)) {
+ energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
+ - inst->scaleEnergyIn);
+ } else {
+ inst->energyIn = WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 8 + scaleEnergyOut
+ - inst->scaleEnergyIn); // Q(-8-scaleEnergyOut)
+ }
+
+ assert(inst->energyIn > 0);
+ energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut
+ + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
+ // Limit the ratio to [0, 1] in Q8, i.e., [0, 256]
+ energyRatio = WEBRTC_SPL_SAT(256, energyRatio, 0);
+
+ // all done in lookup tables now
+ assert(energyRatio < 257);
+ gainFactor1 = kFactor1Table[energyRatio]; // Q8
+ gainFactor2 = inst->factor2Table[energyRatio]; // Q8
+
+ //combine both scales with speech/noise prob: note prior (priorSpeechProb) is not frequency dependent
+
+ // factor = inst->priorSpeechProb*factor1 + (1.0-inst->priorSpeechProb)*factor2; // original code
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(16384 - inst->priorNonSpeechProb,
+ gainFactor1, 14); // Q13 16384 = Q14(1.0)
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
+ gainFactor2, 14); // Q13;
+ gainFactor = tmp16no1 + tmp16no2; // Q13
+ } // out of flag_gain_map==1
+
+ // Synthesis, read out fully processed segment, and update synthesis buffer.
+ WebRtcNsx_SynthesisUpdate(inst, outFrame, gainFactor);
+}
+
+int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFrameHB,
+ short* outFrame, short* outFrameHB) {
+ // main routine for noise suppression
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2, tmpU32no3;
+ WebRtc_UWord32 satMax, maxNoiseU32;
+ WebRtc_UWord32 tmpMagnU32, tmpNoiseU32;
+ WebRtc_UWord32 nearMagnEst;
+ WebRtc_UWord32 noiseUpdateU32;
+ WebRtc_UWord32 noiseU32[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 postLocSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 priorLocSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 prevNearSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 curNearSnr;
+ WebRtc_UWord32 priorSnr;
+ WebRtc_UWord32 noise_estimate = 0;
+ WebRtc_UWord32 noise_estimate_avg = 0;
+ WebRtc_UWord32 numerator = 0;
+
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 pink_noise_num_avg = 0;
+
+ WebRtc_UWord16 tmpU16no1;
+ WebRtc_UWord16 magnU16[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 prevNoiseU16[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 nonSpeechProbFinal[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 gammaNoise, prevGammaNoise;
+ WebRtc_UWord16 noiseSupFilterTmp[HALF_ANAL_BLOCKL];
+
+ WebRtc_Word16 qMagn, qNoise;
+ WebRtc_Word16 avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
+ WebRtc_Word16 pink_noise_exp_avg = 0;
+
+ int i;
+ int nShifts, postShifts;
+ int norm32no1, norm32no2;
+ int flag, sign;
+ int q_domain_to_use = 0;
+
+ // Code for ARMv7-Neon platform assumes the following:
+ assert(inst->anaLen % 16 == 0);
+ assert(inst->anaLen2 % 8 == 0);
+ assert(inst->blockLen10ms % 16 == 0);
+ assert(inst->magnLen == inst->anaLen2 + 1);
+
+#ifdef NS_FILEDEBUG
+ fwrite(spframe, sizeof(short), inst->blockLen10ms, inst->infile);
+#endif
+
+ // Check that initialization has been done
+ if (inst->initFlag != 1) {
+ return -1;
+ }
+ // Check for valid pointers based on sampling rate
+ if ((inst->fs == 32000) && (speechFrameHB == NULL)) {
+ return -1;
+ }
+
+ // Store speechFrame and transform to frequency domain
+ WebRtcNsx_DataAnalysis(inst, speechFrame, magnU16);
+
+ if (inst->zeroInputSignal) {
+ WebRtcNsx_DataSynthesis(inst, outFrame);
+
+ if (inst->fs == 32000) {
+ // update analysis buffer for H band
+ // append new data to buffer FX
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms,
+ speechFrameHB, inst->blockLen10ms);
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
+ }
+ } // end of H band gain computation
+ return 0;
+ }
+
+ // Update block index when we have something to process
+ inst->blockIndex++;
+ //
+
+ // Norm of magn
+ qMagn = inst->normData - inst->stages;
+
+ // Compute spectral flatness on input spectrum
+ WebRtcNsx_ComputeSpectralFlatness(inst, magnU16);
+
+ // quantile noise estimate
+ WebRtcNsx_NoiseEstimation(inst, magnU16, noiseU32, &qNoise);
+
+ //noise estimate from previous frame
+ for (i = 0; i < inst->magnLen; i++) {
+ prevNoiseU16[i] = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
+ }
+
+ if (inst->blockIndex < END_STARTUP_SHORT) {
+ // Noise Q-domain to be used later; see description at end of section.
+ q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages);
+
+ // Calculate frequency independent parts in parametric noise estimate and calculate
+ // the estimate for the lower frequency band (same values for all frequency bins)
+ if (inst->pinkNoiseExp) {
+ pink_noise_exp_avg = (WebRtc_Word16)WebRtcSpl_DivW32W16(inst->pinkNoiseExp,
+ (WebRtc_Word16)(inst->blockIndex + 1)); // Q14
+ pink_noise_num_avg = WebRtcSpl_DivW32W16(inst->pinkNoiseNumerator,
+ (WebRtc_Word16)(inst->blockIndex + 1)); // Q11
+ WebRtcNsx_CalcParametricNoiseEstimate(inst,
+ pink_noise_exp_avg,
+ pink_noise_num_avg,
+ kStartBand,
+ &noise_estimate,
+ &noise_estimate_avg);
+ } else {
+ // Use white noise estimate if we have poor pink noise parameter estimates
+ noise_estimate = inst->whiteNoiseLevel; // Q(minNorm-stages)
+ noise_estimate_avg = noise_estimate / (inst->blockIndex + 1); // Q(minNorm-stages)
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Estimate the background noise using the pink noise parameters if permitted
+ if ((inst->pinkNoiseExp) && (i >= kStartBand)) {
+ // Reset noise_estimate
+ noise_estimate = 0;
+ noise_estimate_avg = 0;
+ // Calculate the parametric noise estimate for current frequency bin
+ WebRtcNsx_CalcParametricNoiseEstimate(inst,
+ pink_noise_exp_avg,
+ pink_noise_num_avg,
+ i,
+ &noise_estimate,
+ &noise_estimate_avg);
+ }
+ // Calculate parametric Wiener filter
+ noiseSupFilterTmp[i] = inst->denoiseBound;
+ if (inst->initMagnEst[i]) {
+ // numerator = (initMagnEst - noise_estimate * overdrive)
+ // Result in Q(8+minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
+ numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8);
+ if (numerator > tmpU32no1) {
+ // Suppression filter coefficient larger than zero, so calculate.
+ numerator -= tmpU32no1;
+
+ // Determine number of left shifts in numerator for best accuracy after
+ // division
+ nShifts = WebRtcSpl_NormU32(numerator);
+ nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
+
+ // Shift numerator to Q(nShifts+8+minNorm-stages)
+ numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts);
+
+ // Shift denominator to Q(nShifts-6+minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
+ if (tmpU32no1 == 0) {
+ // This is only possible if numerator = 0, in which case
+ // we don't need any division.
+ tmpU32no1 = 1;
+ }
+ tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
+ noiseSupFilterTmp[i] = (WebRtc_UWord16)WEBRTC_SPL_SAT(16384, tmpU32no2,
+ (WebRtc_UWord32)(inst->denoiseBound)); // Q14
+ }
+ }
+ // Weight quantile noise 'noiseU32' with modeled noise 'noise_estimate_avg'
+ // 'noiseU32 is in Q(qNoise) and 'noise_estimate' in Q(minNorm-stages)
+ // To guarantee that we do not get wrap around when shifting to the same domain
+ // we use the lowest one. Furthermore, we need to save 6 bits for the weighting.
+ // 'noise_estimate_avg' can handle this operation by construction, but 'noiseU32'
+ // may not.
+
+ // Shift 'noiseU32' to 'q_domain_to_use'
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
+ // Shift 'noise_estimate_avg' to 'q_domain_to_use'
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
+ - q_domain_to_use);
+ // Make a simple check to see if we have enough room for weighting 'tmpU32no1'
+ // without wrap around
+ nShifts = 0;
+ if (tmpU32no1 & 0xfc000000) {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
+ nShifts = 6;
+ }
+ tmpU32no1 *= inst->blockIndex;
+ tmpU32no2 *= (END_STARTUP_SHORT - inst->blockIndex);
+ // Add them together and divide by startup length
+ noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
+ // Shift back if necessary
+ noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
+ }
+ // Update new Q-domain for 'noiseU32'
+ qNoise = q_domain_to_use;
+ }
+ // compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // substituting division with shift ending up in Q(-2*stages)
+ inst->timeAvgMagnEnergyTmp
+ += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
+ 2 * inst->normData + inst->stages - 1);
+ inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
+ inst->blockIndex + 1);
+ }
+
+ //start processing at frames == converged+1
+ // STEP 1: compute prior and post SNR based on quantile noise estimates
+
+ // compute direct decision (DD) estimate of prior SNR: needed for new method
+ satMax = (WebRtc_UWord32)1048575;// Largest possible value without getting overflow despite shifting 12 steps
+ postShifts = 6 + qMagn - qNoise;
+ nShifts = 5 - inst->prevQMagn + inst->prevQNoise;
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT:
+ // post SNR
+ // postLocSnr[i] = 0.0;
+ // if (magn[i] > noise[i])
+ // {
+ // postLocSnr[i] = magn[i] / (noise[i] + 0.0001);
+ // }
+ // // previous post SNR
+ // // previous estimate: based on previous frame with gain filter (smooth is previous filter)
+ //
+ // prevNearSnr[i] = inst->prevMagnU16[i] / (inst->noisePrev[i] + 0.0001) * (inst->smooth[i]);
+ //
+ // // DD estimate is sum of two terms: current estimate and previous estimate
+ // // directed decision update of priorSnr (or we actually store [2*priorSnr+1])
+ //
+ // priorLocSnr[i] = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * (postLocSnr[i] - 1.0);
+
+ // calculate post SNR: output in Q11
+ postLocSnr[i] = 2048; // 1.0 in Q11
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], 6); // Q(6+qMagn)
+ if (postShifts < 0) {
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
+ }
+ if (tmpU32no1 > tmpU32no2) {
+ // Current magnitude larger than noise
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ } else {
+ postLocSnr[i] = satMax;
+ }
+ }
+
+ // calculate prevNearSnr[i] and save for later instead of recalculating it later
+ nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
+
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ } else {
+ tmpU32no1 = satMax; // Q11
+ }
+ prevNearSnr[i] = tmpU32no1; // Q11
+
+ //directed decision update of priorSnr
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
+ priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
+ // priorLocSnr = 1 + 2*priorSnr
+ priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
+ } // end of loop over frequencies
+ // done with step 1: DD computation of prior and post SNR
+
+ // STEP 2: compute speech/noise likelihood
+
+ //compute difference of input spectrum with learned/estimated noise spectrum
+ WebRtcNsx_ComputeSpectralDifference(inst, magnU16);
+ //compute histograms for determination of parameters (thresholds and weights for features)
+ //parameters are extracted once every window time (=inst->modelUpdate)
+ //counter update
+ inst->cntThresUpdate++;
+ flag = (int)(inst->cntThresUpdate == inst->modelUpdate);
+ //update histogram
+ WebRtcNsx_FeatureParameterExtraction(inst, flag);
+ //compute model parameters
+ if (flag) {
+ inst->cntThresUpdate = 0; // Reset counter
+ //update every window:
+ // get normalization for spectral difference for next window estimate
+
+ // Shift to Q(-2*stages)
+ inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
+
+ tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
+ // Update featureSpecDiff
+ if ((tmpU32no1 != inst->timeAvgMagnEnergy) && (inst->featureSpecDiff) &&
+ (inst->timeAvgMagnEnergy > 0)) {
+ norm32no1 = 0;
+ tmpU32no3 = tmpU32no1;
+ while (0xFFFF0000 & tmpU32no3) {
+ tmpU32no3 >>= 1;
+ norm32no1++;
+ }
+ tmpU32no2 = inst->featureSpecDiff;
+ while (0xFFFF0000 & tmpU32no2) {
+ tmpU32no2 >>= 1;
+ norm32no1++;
+ }
+ tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
+ tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
+ if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1) {
+ inst->featureSpecDiff = 0x007FFFFF;
+ } else {
+ inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
+ WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
+ }
+ }
+
+ inst->timeAvgMagnEnergy = tmpU32no1; // Q(-2*stages)
+ inst->curAvgMagnEnergy = 0;
+ }
+
+ //compute speech/noise probability
+ WebRtcNsx_SpeechNoiseProb(inst, nonSpeechProbFinal, priorLocSnr, postLocSnr);
+
+ //time-avg parameter for noise update
+ gammaNoise = NOISE_UPDATE_Q8; // Q8
+
+ maxNoiseU32 = 0;
+ postShifts = inst->prevQNoise - qMagn;
+ nShifts = inst->prevQMagn - qMagn;
+ for (i = 0; i < inst->magnLen; i++) {
+ // temporary noise update: use it for speech frames if update value is less than previous
+ // the formula has been rewritten into:
+ // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
+
+ if (postShifts < 0) {
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
+ }
+ if (prevNoiseU16[i] > tmpU32no2) {
+ sign = -1;
+ tmpU32no1 = prevNoiseU16[i] - tmpU32no2;
+ } else {
+ sign = 1;
+ tmpU32no1 = tmpU32no2 - prevNoiseU16[i];
+ }
+ noiseUpdateU32 = inst->prevNoiseU32[i]; // Q(prevQNoise+11)
+ tmpU32no3 = 0;
+ if ((tmpU32no1) && (nonSpeechProbFinal[i])) {
+ // This value will be used later, if gammaNoise changes
+ tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
+ if (0x7c000000 & tmpU32no3) {
+ // Shifting required before multiplication
+ tmpU32no2
+ = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
+ } else {
+ // We can do shifting after multiplication
+ tmpU32no2
+ = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
+ }
+ if (sign > 0) {
+ noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
+ } else {
+ // This operation is safe. We can never get wrap around, since worst
+ // case scenario means magnU16 = 0
+ noiseUpdateU32 -= tmpU32no2; // Q(prevQNoise+11)
+ }
+ }
+
+ //increase gamma (i.e., less noise update) for frame likely to be speech
+ prevGammaNoise = gammaNoise;
+ gammaNoise = NOISE_UPDATE_Q8;
+ //time-constant based on speech/noise state
+ //increase gamma (i.e., less noise update) for frames likely to be speech
+ if (nonSpeechProbFinal[i] < ONE_MINUS_PROB_RANGE_Q8) {
+ gammaNoise = GAMMA_NOISE_TRANS_AND_SPEECH_Q8;
+ }
+
+ if (prevGammaNoise != gammaNoise) {
+ // new noise update
+ // this line is the same as above, only that the result is stored in a different variable and the gammaNoise
+ // has changed
+ //
+ // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
+
+ if (0x7c000000 & tmpU32no3) {
+ // Shifting required before multiplication
+ tmpU32no2
+ = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
+ } else {
+ // We can do shifting after multiplication
+ tmpU32no2
+ = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
+ }
+ if (sign > 0) {
+ tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
+ } else {
+ tmpU32no1 = inst->prevNoiseU32[i] - tmpU32no2; // Q(prevQNoise+11)
+ }
+ if (noiseUpdateU32 > tmpU32no1) {
+ noiseUpdateU32 = tmpU32no1; // Q(prevQNoise+11)
+ }
+ }
+ noiseU32[i] = noiseUpdateU32; // Q(prevQNoise+11)
+ if (noiseUpdateU32 > maxNoiseU32) {
+ maxNoiseU32 = noiseUpdateU32;
+ }
+
+ // conservative noise update
+ // // original FLOAT code
+ // if (prob_speech < PROB_RANGE) {
+ // inst->avgMagnPause[i] = inst->avgMagnPause[i] + (1.0 - gamma_pause)*(magn[i] - inst->avgMagnPause[i]);
+ // }
+
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(inst->avgMagnPause[i], -nShifts);
+ if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8) {
+ if (nShifts < 0) {
+ tmp32no1 = (WebRtc_Word32)magnU16[i] - tmp32no2; // Q(qMagn)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts)
+ - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
+ }
+ tmp32no2 += tmp32no1; // Q(qMagn)
+ }
+ inst->avgMagnPause[i] = tmp32no2;
+ } // end of frequency loop
+
+ norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
+ qNoise = inst->prevQNoise + norm32no1 - 5;
+ // done with step 2: noise update
+
+ // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
+ nShifts = inst->prevQNoise + 11 - qMagn;
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT code
+ // // post and prior SNR
+ // curNearSnr = 0.0;
+ // if (magn[i] > noise[i])
+ // {
+ // curNearSnr = magn[i] / (noise[i] + 0.0001) - 1.0;
+ // }
+ // // DD estimate is sum of two terms: current estimate and previous estimate
+ // // directed decision update of snrPrior
+ // snrPrior = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * curNearSnr;
+ // // gain filter
+ // tmpFloat1 = inst->overdrive + snrPrior;
+ // tmpFloat2 = snrPrior / tmpFloat1;
+ // theFilter[i] = tmpFloat2;
+
+ // calculate curNearSnr again, this is necessary because a new noise estimate has been made since then. for the original
+ curNearSnr = 0; // Q11
+ if (nShifts < 0) {
+ // This case is equivalent with magn < noise which implies curNearSnr = 0;
+ tmpMagnU32 = (WebRtc_UWord32)magnU16[i]; // Q(qMagn)
+ tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn)
+ } else if (nShifts > 17) {
+ tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17)
+ tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
+ } else {
+ tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], nShifts); // Q(qNoise_prev+11)
+ tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
+ }
+ if (tmpMagnU32 > tmpNoiseU32) {
+ tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
+ norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ }
+ curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ }
+
+ //directed decision update of priorSnr
+ // FLOAT
+ // priorSnr = DD_PR_SNR * prevNearSnr + (1.0-DD_PR_SNR) * curNearSnr;
+
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(curNearSnr, ONE_MINUS_DD_PR_SNR_Q11); // Q22
+ priorSnr = tmpU32no1 + tmpU32no2; // Q22
+
+ //gain filter
+ tmpU32no1 = (WebRtc_UWord32)(inst->overdrive)
+ + WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
+ assert(inst->overdrive > 0);
+ tmpU16no1 = (WebRtc_UWord16)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
+ inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
+
+ // Weight in the parametric Wiener filter during startup
+ if (inst->blockIndex < END_STARTUP_SHORT) {
+ // Weight the two suppression filters
+ tmpU32no1 = WEBRTC_SPL_UMUL_16_16(inst->noiseSupFilter[i],
+ (WebRtc_UWord16)inst->blockIndex);
+ tmpU32no2 = WEBRTC_SPL_UMUL_16_16(noiseSupFilterTmp[i],
+ (WebRtc_UWord16)(END_STARTUP_SHORT
+ - inst->blockIndex));
+ tmpU32no1 += tmpU32no2;
+ inst->noiseSupFilter[i] = (WebRtc_UWord16)WebRtcSpl_DivU32U16(tmpU32no1,
+ END_STARTUP_SHORT);
+ }
+ } // end of loop over frequencies
+ //done with step3
+
+ // save noise and magnitude spectrum for next frame
+ inst->prevQNoise = qNoise;
+ inst->prevQMagn = qMagn;
+ if (norm32no1 > 5) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11)
+ inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
+ }
+ } else {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
+ inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
+ }
+ }
+
+ WebRtcNsx_DataSynthesis(inst, outFrame);
+#ifdef NS_FILEDEBUG
+ fwrite(outframe, sizeof(short), inst->blockLen10ms, inst->outfile);
+#endif
+
+ //for H band:
+ // only update data buffer, then apply time-domain gain is applied derived from L band
+ if (inst->fs == 32000) {
+ // update analysis buffer for H band
+ // append new data to buffer FX
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms, inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms, speechFrameHB, inst->blockLen10ms);
+ // range for averaging low band quantities for H band gain
+
+ gainTimeDomainHB = 16384; // 16384 = Q14(1.0)
+ //average speech prob from low band
+ //average filter gain from low band
+ //avg over second half (i.e., 4->8kHz) of freq. spectrum
+ tmpU32no1 = 0; // Q12
+ tmpU16no1 = 0; // Q8
+ for (i = inst->anaLen2 - (inst->anaLen2 >> 2); i < inst->anaLen2; i++) {
+ tmpU16no1 += nonSpeechProbFinal[i]; // Q8
+ tmpU32no1 += (WebRtc_UWord32)(inst->noiseSupFilter[i]); // Q14
+ }
+ avgProbSpeechHB = (WebRtc_Word16)(4096
+ - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12
+ avgFilterGainHB = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
+ tmpU32no1, inst->stages - 3); // Q14
+
+ // // original FLOAT code
+ // // gain based on speech probability:
+ // avg_prob_speech_tt=(float)2.0*avg_prob_speech-(float)1.0;
+ // gain_mod=(float)0.5*((float)1.0+(float)tanh(avg_prob_speech_tt)); // between 0 and 1
+
+ // gain based on speech probability:
+ // original expression: "0.5 * (1 + tanh(2x-1))"
+ // avgProbSpeechHB has been anyway saturated to a value between 0 and 1 so the other cases don't have to be dealt with
+ // avgProbSpeechHB and gainModHB are in Q12, 3607 = Q12(0.880615234375) which is a zero point of
+ // |0.5 * (1 + tanh(2x-1)) - x| - |0.5 * (1 + tanh(2x-1)) - 0.880615234375| meaning that from that point the error of approximating
+ // the expression with f(x) = x would be greater than the error of approximating the expression with f(x) = 0.880615234375
+ // error: "|0.5 * (1 + tanh(2x-1)) - x| from x=0 to 0.880615234375" -> http://www.wolframalpha.com/input/?i=|0.5+*+(1+%2B+tanh(2x-1))+-+x|+from+x%3D0+to+0.880615234375
+ // and: "|0.5 * (1 + tanh(2x-1)) - 0.880615234375| from x=0.880615234375 to 1" -> http://www.wolframalpha.com/input/?i=+|0.5+*+(1+%2B+tanh(2x-1))+-+0.880615234375|+from+x%3D0.880615234375+to+1
+ gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607);
+
+ // // original FLOAT code
+ // //combine gain with low band gain
+ // if (avg_prob_speech < (float)0.5) {
+ // gain_time_domain_HB=(float)0.5*gain_mod+(float)0.5*avg_filter_gain;
+ // }
+ // else {
+ // gain_time_domain_HB=(float)0.25*gain_mod+(float)0.75*avg_filter_gain;
+ // }
+
+
+ //combine gain with low band gain
+ if (avgProbSpeechHB < 2048) {
+ // 2048 = Q12(0.5)
+ // the next two lines in float are "gain_time_domain = 0.5 * gain_mod + 0.5 * avg_filter_gain"; Q2(0.5) = 2 equals one left shift
+ gainTimeDomainHB = (gainModHB << 1) + (avgFilterGainHB >> 1); // Q14
+ } else {
+ // "gain_time_domain = 0.25 * gain_mod + 0.75 * agv_filter_gain;"
+ gainTimeDomainHB = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, avgFilterGainHB, 2); // 3 = Q2(0.75); Q14
+ gainTimeDomainHB += gainModHB; // Q14
+ }
+ //make sure gain is within flooring range
+ gainTimeDomainHB
+ = WEBRTC_SPL_SAT(16384, gainTimeDomainHB, (WebRtc_Word16)(inst->denoiseBound)); // 16384 = Q14(1.0)
+
+
+ //apply gain
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrameHB[i]
+ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
+ }
+ } // end of H band gain computation
+
+ return 0;
+}
+
+
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core.h b/trunk/src/modules/audio_processing/ns/nsx_core.h
new file mode 100644
index 0000000..0a0faf9
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/nsx_core.h
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+#include "nsx_defines.h"
+
+#ifdef NS_FILEDEBUG
+#include <stdio.h>
+#endif
+
+typedef struct NsxInst_t_ {
+ WebRtc_UWord32 fs;
+
+ const WebRtc_Word16* window;
+ WebRtc_Word16 analysisBuffer[ANAL_BLOCKL_MAX];
+ WebRtc_Word16 synthesisBuffer[ANAL_BLOCKL_MAX];
+ WebRtc_UWord16 noiseSupFilter[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 overdrive; /* Q8 */
+ WebRtc_UWord16 denoiseBound; /* Q14 */
+ const WebRtc_Word16* factor2Table;
+ WebRtc_Word16 noiseEstLogQuantile[SIMULT* HALF_ANAL_BLOCKL];
+ WebRtc_Word16 noiseEstDensity[SIMULT* HALF_ANAL_BLOCKL];
+ WebRtc_Word16 noiseEstCounter[SIMULT];
+ WebRtc_Word16 noiseEstQuantile[HALF_ANAL_BLOCKL];
+
+ WebRtc_Word16 anaLen;
+ int anaLen2;
+ int magnLen;
+ int aggrMode;
+ int stages;
+ int initFlag;
+ int gainMap;
+
+ WebRtc_Word32 maxLrt;
+ WebRtc_Word32 minLrt;
+ WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
+ WebRtc_Word32 featureLogLrt;
+ WebRtc_Word32 thresholdLogLrt;
+ WebRtc_Word16 weightLogLrt;
+
+ WebRtc_UWord32 featureSpecDiff;
+ WebRtc_UWord32 thresholdSpecDiff;
+ WebRtc_Word16 weightSpecDiff;
+
+ WebRtc_UWord32 featureSpecFlat;
+ WebRtc_UWord32 thresholdSpecFlat;
+ WebRtc_Word16 weightSpecFlat;
+
+ WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
+ WebRtc_UWord32 magnEnergy;
+ WebRtc_UWord32 sumMagn;
+ WebRtc_UWord32 curAvgMagnEnergy;
+ WebRtc_UWord32 timeAvgMagnEnergy;
+ WebRtc_UWord32 timeAvgMagnEnergyTmp;
+
+ WebRtc_UWord32 whiteNoiseLevel; //initial noise estimate
+ WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
+ WebRtc_Word32 pinkNoiseNumerator; //pink noise parameter: numerator
+ WebRtc_Word32 pinkNoiseExp; //pink noise parameter: power of freq
+ int minNorm; //smallest normalization factor
+ int zeroInputSignal; //zero input signal flag
+
+ WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
+ WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
+ WebRtc_Word16 priorNonSpeechProb; //prior speech/noise probability // Q14
+
+ int blockIndex; //frame index counter
+ int modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
+ int cntThresUpdate;
+
+ //histograms for parameter estimation
+ WebRtc_Word16 histLrt[HIST_PAR_EST];
+ WebRtc_Word16 histSpecFlat[HIST_PAR_EST];
+ WebRtc_Word16 histSpecDiff[HIST_PAR_EST];
+
+ //quantities for high band estimate
+ WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
+
+ int qNoise;
+ int prevQNoise;
+ int prevQMagn;
+ int blockLen10ms;
+
+ WebRtc_Word16 real[ANAL_BLOCKL_MAX];
+ WebRtc_Word16 imag[ANAL_BLOCKL_MAX];
+ WebRtc_Word32 energyIn;
+ int scaleEnergyIn;
+ int normData;
+
+} NsxInst_t;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/****************************************************************************
+ * WebRtcNsx_InitCore(...)
+ *
+ * This function initializes a noise suppression instance
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - fs : Sampling frequency
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs);
+
+/****************************************************************************
+ * WebRtcNsx_set_policy_core(...)
+ *
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode);
+
+/****************************************************************************
+ * WebRtcNsx_ProcessCore
+ *
+ * Do noise suppression.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - inFrameLow : Input speech frame for lower band
+ * - inFrameHigh : Input speech frame for higher band
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outFrameLow : Output speech frame for lower band
+ * - outFrameHigh : Output speech frame for higher band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+int WebRtcNsx_ProcessCore(NsxInst_t* inst,
+ short* inFrameLow,
+ short* inFrameHigh,
+ short* outFrameLow,
+ short* outFrameHigh);
+
+/****************************************************************************
+ * Some function pointers, for internal functions shared by ARM NEON and
+ * generic C code.
+ */
+// Noise Estimation.
+typedef void (*NoiseEstimation)(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise);
+extern NoiseEstimation WebRtcNsx_NoiseEstimation;
+
+// Filter the data in the frequency domain, and create spectrum.
+typedef void (*PrepareSpectrum)(NsxInst_t* inst,
+ int16_t* freq_buff);
+extern PrepareSpectrum WebRtcNsx_PrepareSpectrum;
+
+// For the noise supression process, synthesis, read out fully processed
+// segment, and update synthesis buffer.
+typedef void (*SynthesisUpdate)(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor);
+extern SynthesisUpdate WebRtcNsx_SynthesisUpdate;
+
+// Update analysis buffer for lower band, and window data before FFT.
+typedef void (*AnalysisUpdate)(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech);
+extern AnalysisUpdate WebRtcNsx_AnalysisUpdate;
+
+// Denormalize the input buffer.
+typedef void (*Denormalize)(NsxInst_t* inst,
+ int16_t* in,
+ int factor);
+extern Denormalize WebRtcNsx_Denormalize;
+
+// Create a complex number buffer, as the intput interleaved with zeros,
+// and normalize it.
+typedef void (*CreateComplexBuffer)(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out);
+extern CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
+
+/****************************************************************************
+ * Initialization of the above function pointers for ARM Neon.
+ */
+void WebRtcNsx_InitNeon(void);
+
+extern const WebRtc_Word16 WebRtcNsx_kLogTable[9];
+extern const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256];
+extern const WebRtc_Word16 WebRtcNsx_kCounterDiv[201];
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core_neon.c b/trunk/src/modules/audio_processing/ns/nsx_core_neon.c
new file mode 100644
index 0000000..2f85abd
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/nsx_core_neon.c
@@ -0,0 +1,734 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "nsx_core.h"
+
+#include <arm_neon.h>
+#include <assert.h>
+
+// Update the noise estimation information.
+static void UpdateNoiseEstimateNeon(NsxInst_t* inst, int offset) {
+ int i = 0;
+ const int16_t kExp2Const = 11819; // Q13
+ int16_t* ptr_noiseEstLogQuantile = NULL;
+ int16_t* ptr_noiseEstQuantile = NULL;
+ int16x4_t kExp2Const16x4 = vdup_n_s16(kExp2Const);
+ int32x4_t twentyOne32x4 = vdupq_n_s32(21);
+ int32x4_t constA32x4 = vdupq_n_s32(0x1fffff);
+ int32x4_t constB32x4 = vdupq_n_s32(0x200000);
+
+ int16_t tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
+ inst->magnLen);
+
+ // Guarantee a Q-domain as high as possible and still fit in int16
+ inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2Const,
+ tmp16,
+ 21);
+
+ int32x4_t qNoise32x4 = vdupq_n_s32(inst->qNoise);
+
+ for (ptr_noiseEstLogQuantile = &inst->noiseEstLogQuantile[offset],
+ ptr_noiseEstQuantile = &inst->noiseEstQuantile[0];
+ ptr_noiseEstQuantile < &inst->noiseEstQuantile[inst->magnLen - 3];
+ ptr_noiseEstQuantile += 4, ptr_noiseEstLogQuantile += 4) {
+
+ // tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ // inst->noiseEstLogQuantile[offset + i]);
+ int16x4_t v16x4 = vld1_s16(ptr_noiseEstLogQuantile);
+ int32x4_t v32x4B = vmull_s16(v16x4, kExp2Const16x4);
+
+ // tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+ int32x4_t v32x4A = vandq_s32(v32x4B, constA32x4);
+ v32x4A = vorrq_s32(v32x4A, constB32x4);
+
+ // tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ v32x4B = vshrq_n_s32(v32x4B, 21);
+
+ // tmp16 -= 21;// shift 21 to get result in Q0
+ v32x4B = vsubq_s32(v32x4B, twentyOne32x4);
+
+ // tmp16 += (int16_t) inst->qNoise;
+ // shift to get result in Q(qNoise)
+ v32x4B = vaddq_s32(v32x4B, qNoise32x4);
+
+ // if (tmp16 < 0) {
+ // tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ // } else {
+ // tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ // }
+ v32x4B = vshlq_s32(v32x4A, v32x4B);
+
+ // tmp16 = WebRtcSpl_SatW32ToW16(tmp32no1);
+ v16x4 = vqmovn_s32(v32x4B);
+
+ //inst->noiseEstQuantile[i] = tmp16;
+ vst1_s16(ptr_noiseEstQuantile, v16x4);
+ }
+
+ // Last iteration:
+
+ // inst->quantile[i]=exp(inst->lquantile[offset+i]);
+ // in Q21
+ int32_t tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ *ptr_noiseEstLogQuantile);
+ int32_t tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+
+ tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ tmp16 -= 21;// shift 21 to get result in Q0
+ tmp16 += (int16_t) inst->qNoise; //shift to get result in Q(qNoise)
+ if (tmp16 < 0) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ }
+ *ptr_noiseEstQuantile = WebRtcSpl_SatW32ToW16(tmp32no1);
+}
+
+// Noise Estimation
+static void NoiseEstimationNeon(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise) {
+ int16_t lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
+ int16_t countProd, delta, zeros, frac;
+ int16_t log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
+ const int16_t log2_const = 22713;
+ const int16_t width_factor = 21845;
+
+ int i, s, offset;
+
+ tabind = inst->stages - inst->normData;
+ assert(tabind < 9);
+ assert(tabind > -9);
+ if (tabind < 0) {
+ logval = -WebRtcNsx_kLogTable[-tabind];
+ } else {
+ logval = WebRtcNsx_kLogTable[tabind];
+ }
+
+ int16x8_t logval_16x8 = vdupq_n_s16(logval);
+
+ // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
+ // magn is in Q(-stages), and the real lmagn values are:
+ // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
+ // lmagn in Q8
+ for (i = 0; i < inst->magnLen; i++) {
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((uint32_t)magn[i]);
+ frac = (int16_t)((((uint32_t)magn[i] << zeros)
+ & 0x7FFFFFFF) >> 23);
+ assert(frac < 256);
+ // log2(magn(i))
+ log2 = (int16_t)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ // log2(magn(i))*log(2)
+ lmagn[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
+ // + log(2^stages)
+ lmagn[i] += logval;
+ } else {
+ lmagn[i] = logval;
+ }
+ }
+
+ int16x4_t Q3_16x4 = vdup_n_s16(3);
+ int16x8_t WIDTHQ8_16x8 = vdupq_n_s16(WIDTH_Q8);
+ int16x8_t WIDTHFACTOR_16x8 = vdupq_n_s16(width_factor);
+
+ int16_t factor = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG)
+ factor = FACTOR_Q7_STARTUP;
+
+ // Loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // Get counter values from state
+ counter = inst->noiseEstCounter[s];
+ assert(counter < 201);
+ countDiv = WebRtcNsx_kCounterDiv[counter];
+ countProd = (int16_t)WEBRTC_SPL_MUL_16_16(counter, countDiv);
+
+ // quant_est(...)
+ int16_t deltaBuff[8];
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int16x4_t countDiv_16x4 = vdup_n_s16(countDiv);
+ int16x8_t countProd_16x8 = vdupq_n_s16(countProd);
+ int16x8_t tmp16x8_0 = vdupq_n_s16(countDiv);
+ int16x8_t prod16x8 = vqrdmulhq_s16(WIDTHFACTOR_16x8, tmp16x8_0);
+ int16x8_t tmp16x8_1;
+ int16x8_t tmp16x8_2;
+ int16x8_t tmp16x8_3;
+ int16x8_t tmp16x8_4;
+ int16x8_t tmp16x8_5;
+ int32x4_t tmp32x4;
+
+ for (i = 0; i < inst->magnLen - 7; i += 8) {
+ // Compute delta.
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ tmp16x8_0 = vdupq_n_s16(factor);
+ vst1q_s16(deltaBuff, tmp16x8_0);
+
+ int j;
+ for (j = 0; j < 8; j++) {
+ if (inst->noiseEstDensity[offset + i + j] > 512) {
+ // Get values for deltaBuff by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i + j]);
+ deltaBuff[j] = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ }
+ }
+
+ // Update log quantile estimate
+
+ // tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[0]), countDiv_16x4);
+ tmp16x4_1 = vshrn_n_s32(tmp32x4, 14);
+ tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[4]), countDiv_16x4);
+ tmp16x4_0 = vshrn_n_s32(tmp32x4, 14);
+ tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // Keep for several lines.
+
+ // prepare for the "if" branch
+ // tmp16 += 2;
+ // tmp16_1 = (Word16)(tmp16>>2);
+ tmp16x8_1 = vrshrq_n_s16(tmp16x8_0, 2);
+
+ // inst->noiseEstLogQuantile[offset+i] + tmp16_1;
+ tmp16x8_2 = vld1q_s16(&inst->noiseEstLogQuantile[offset + i]); // Keep
+ tmp16x8_1 = vaddq_s16(tmp16x8_2, tmp16x8_1); // Keep for several lines
+
+ // Prepare for the "else" branch
+ // tmp16 += 1;
+ // tmp16_1 = (Word16)(tmp16>>1);
+ tmp16x8_0 = vrshrq_n_s16(tmp16x8_0, 1);
+
+ // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
+ tmp32x4 = vmull_s16(vget_low_s16(tmp16x8_0), Q3_16x4);
+ tmp16x4_1 = vshrn_n_s32(tmp32x4, 1);
+
+ // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
+ tmp32x4 = vmull_s16(vget_high_s16(tmp16x8_0), Q3_16x4);
+ tmp16x4_0 = vshrn_n_s32(tmp32x4, 1);
+
+ // inst->noiseEstLogQuantile[offset + i] - tmp16_2;
+ tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // keep
+ tmp16x8_0 = vsubq_s16(tmp16x8_2, tmp16x8_0);
+
+ // logval is the smallest fixed point representation we can have. Values
+ // below that will correspond to values in the interval [0, 1], which
+ // can't possibly occur.
+ tmp16x8_0 = vmaxq_s16(tmp16x8_0, logval_16x8);
+
+ // Do the if-else branches:
+ tmp16x8_3 = vld1q_s16(&lmagn[i]); // keep for several lines
+ tmp16x8_5 = vsubq_s16(tmp16x8_3, tmp16x8_2);
+ __asm__("vcgt.s16 %q0, %q1, #0"::"w"(tmp16x8_4), "w"(tmp16x8_5));
+ __asm__("vbit %q0, %q1, %q2"::
+ "w"(tmp16x8_2), "w"(tmp16x8_1), "w"(tmp16x8_4));
+ __asm__("vbif %q0, %q1, %q2"::
+ "w"(tmp16x8_2), "w"(tmp16x8_0), "w"(tmp16x8_4));
+ vst1q_s16(&inst->noiseEstLogQuantile[offset + i], tmp16x8_2);
+
+ // Update density estimate
+ // tmp16_1 + tmp16_2
+ tmp16x8_1 = vld1q_s16(&inst->noiseEstDensity[offset + i]);
+ tmp16x8_0 = vqrdmulhq_s16(tmp16x8_1, countProd_16x8);
+ tmp16x8_0 = vaddq_s16(tmp16x8_0, prod16x8);
+
+ // lmagn[i] - inst->noiseEstLogQuantile[offset + i]
+ tmp16x8_3 = vsubq_s16(tmp16x8_3, tmp16x8_2);
+ tmp16x8_3 = vabsq_s16(tmp16x8_3);
+ tmp16x8_4 = vcgtq_s16(WIDTHQ8_16x8, tmp16x8_3);
+ __asm__("vbit %q0, %q1, %q2"::
+ "w"(tmp16x8_1), "w"(tmp16x8_0), "w"(tmp16x8_4));
+ vst1q_s16(&inst->noiseEstDensity[offset + i], tmp16x8_1);
+ } // End loop over magnitude spectrum
+
+ // Last iteration over magnitude spectrum:
+ // compute delta
+ if (inst->noiseEstDensity[offset + i] > 512) {
+ // Get values for deltaBuff by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
+ delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ } else {
+ delta = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ delta = FACTOR_Q7_STARTUP;
+ }
+ }
+ // update log quantile estimate
+ tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
+ // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
+ // CounterDiv=1/(inst->counter[s]+1) in Q15
+ tmp16 += 2;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
+ inst->noiseEstLogQuantile[offset + i] += tmp16no1;
+ } else {
+ tmp16 += 1;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
+ // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
+ tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
+ inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
+ if (inst->noiseEstLogQuantile[offset + i] < logval) {
+ // logval is the smallest fixed point representation we can have.
+ // Values below that will correspond to values in the interval
+ // [0, 1], which can't possibly occur.
+ inst->noiseEstLogQuantile[offset + i] = logval;
+ }
+ }
+
+ // update density estimate
+ if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
+ < WIDTH_Q8) {
+ tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->noiseEstDensity[offset + i], countProd, 15);
+ tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ width_factor, countDiv, 15);
+ inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
+ }
+
+
+ if (counter >= END_STARTUP_LONG) {
+ inst->noiseEstCounter[s] = 0;
+ if (inst->blockIndex >= END_STARTUP_LONG) {
+ UpdateNoiseEstimateNeon(inst, offset);
+ }
+ }
+ inst->noiseEstCounter[s]++;
+
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ UpdateNoiseEstimateNeon(inst, offset);
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (uint32_t)(inst->noiseEstQuantile[i]); // Q(qNoise)
+ }
+ (*q_noise) = (int16_t)inst->qNoise;
+}
+
+// Filter the data in the frequency domain, and create spectrum.
+static void PrepareSpectrumNeon(NsxInst_t* inst, int16_t* freq_buf) {
+
+ // (1) Filtering.
+
+ // Fixed point C code for the next block is as follows:
+ // for (i = 0; i < inst->magnLen; i++) {
+ // inst->real[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
+ // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ // inst->imag[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
+ // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ // }
+
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_imag = &inst->imag[0];
+ uint16_t* ptr_noiseSupFilter = &inst->noiseSupFilter[0];
+
+ // Filter the rest in the frequency domain.
+ for (; ptr_real < &inst->real[inst->magnLen - 1];) {
+ // Loop unrolled once. Both pointers are incremented by 4 twice.
+ __asm__ __volatile__(
+ "vld1.16 d20, [%[ptr_real]]\n\t"
+ "vld1.16 d22, [%[ptr_imag]]\n\t"
+ "vld1.16 d23, [%[ptr_noiseSupFilter]]!\n\t"
+ "vmull.s16 q10, d20, d23\n\t"
+ "vmull.s16 q11, d22, d23\n\t"
+ "vshrn.s32 d20, q10, #14\n\t"
+ "vshrn.s32 d22, q11, #14\n\t"
+ "vst1.16 d20, [%[ptr_real]]!\n\t"
+ "vst1.16 d22, [%[ptr_imag]]!\n\t"
+
+ "vld1.16 d18, [%[ptr_real]]\n\t"
+ "vld1.16 d24, [%[ptr_imag]]\n\t"
+ "vld1.16 d25, [%[ptr_noiseSupFilter]]!\n\t"
+ "vmull.s16 q9, d18, d25\n\t"
+ "vmull.s16 q12, d24, d25\n\t"
+ "vshrn.s32 d18, q9, #14\n\t"
+ "vshrn.s32 d24, q12, #14\n\t"
+ "vst1.16 d18, [%[ptr_real]]!\n\t"
+ "vst1.16 d24, [%[ptr_imag]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_imag]"+r"(ptr_imag),
+ [ptr_real]"+r"(ptr_real),
+ [ptr_noiseSupFilter]"+r"(ptr_noiseSupFilter)
+ :
+ :"d18", "d19", "d20", "d21", "d22", "d23", "d24", "d25",
+ "q9", "q10", "q11", "q12"
+ );
+ }
+
+ // Filter the last pair of elements in the frequency domain.
+ *ptr_real = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_real,
+ (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
+ *ptr_imag = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_imag,
+ (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
+
+ // (2) Create spectrum.
+
+ // Fixed point C code for the rest of the function is as follows:
+ // freq_buf[0] = inst->real[0];
+ // freq_buf[1] = -inst->imag[0];
+ // for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ // tmp16 = (inst->anaLen << 1) - j;
+ // freq_buf[j] = inst->real[i];
+ // freq_buf[j + 1] = -inst->imag[i];
+ // freq_buf[tmp16] = inst->real[i];
+ // freq_buf[tmp16 + 1] = inst->imag[i];
+ // }
+ // freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ // freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+
+ freq_buf[0] = inst->real[0];
+ freq_buf[1] = -inst->imag[0];
+
+ int offset = -16;
+ int16_t* ptr_realImag1 = &freq_buf[2];
+ int16_t* ptr_realImag2 = ptr_realImag2 = &freq_buf[(inst->anaLen << 1) - 8];
+ ptr_real = &inst->real[1];
+ ptr_imag = &inst->imag[1];
+ for (; ptr_real < &inst->real[inst->anaLen2 - 11];) {
+ // Loop unrolled once. All pointers are incremented twice.
+ __asm__ __volatile__(
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_imag]]!\n\t"
+ // Negate and interleave:
+ "vmov.s16 d20, d22\n\t"
+ "vneg.s16 d21, d23\n\t"
+ "vzip.16 d20, d21\n\t"
+ // Write 8 elements to &freq_buf[j]
+ "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
+ // Interleave and reverse elements:
+ "vzip.16 d22, d23\n\t"
+ "vrev64.32 d18, d23\n\t"
+ "vrev64.32 d19, d22\n\t"
+ // Write 8 elements to &freq_buf[tmp16]
+ "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
+
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_imag]]!\n\t"
+ // Negate and interleave:
+ "vmov.s16 d20, d22\n\t"
+ "vneg.s16 d21, d23\n\t"
+ "vzip.16 d20, d21\n\t"
+ // Write 8 elements to &freq_buf[j]
+ "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
+ // Interleave and reverse elements:
+ "vzip.16 d22, d23\n\t"
+ "vrev64.32 d18, d23\n\t"
+ "vrev64.32 d19, d22\n\t"
+ // Write 8 elements to &freq_buf[tmp16]
+ "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
+
+ // Specify constraints.
+ :[ptr_imag]"+r"(ptr_imag),
+ [ptr_real]"+r"(ptr_real),
+ [ptr_realImag1]"+r"(ptr_realImag1),
+ [ptr_realImag2]"+r"(ptr_realImag2)
+ :[offset]"r"(offset)
+ :"d18", "d19", "d20", "d21", "d22", "d23"
+ );
+ }
+ for (ptr_realImag2 += 6;
+ ptr_real <= &inst->real[inst->anaLen2];
+ ptr_real += 1, ptr_imag += 1, ptr_realImag1 += 2, ptr_realImag2 -= 2) {
+ *ptr_realImag1 = *ptr_real;
+ *(ptr_realImag1 + 1) = -(*ptr_imag);
+ *ptr_realImag2 = *ptr_real;
+ *(ptr_realImag2 + 1) = *ptr_imag;
+ }
+
+ freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+}
+
+// Denormalize the input buffer.
+static __inline void DenormalizeNeon(NsxInst_t* inst, int16_t* in, int factor) {
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_in = &in[0];
+
+ __asm__ __volatile__("vdup.32 q10, %0" ::
+ "r"((int32_t)(factor - inst->normData)) : "q10");
+ for (; ptr_real < &inst->real[inst->anaLen];) {
+
+ // Loop unrolled once. Both pointers are incremented.
+ __asm__ __volatile__(
+ // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
+ // factor - inst->normData);
+ "vld2.16 {d24, d25}, [%[ptr_in]]!\n\t"
+ "vmovl.s16 q12, d24\n\t"
+ "vshl.s32 q12, q10\n\t"
+ // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqmovn.s32 d24, q12\n\t"
+ "vst1.16 d24, [%[ptr_real]]!\n\t"
+
+ // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
+ // factor - inst->normData);
+ "vld2.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vmovl.s16 q11, d22\n\t"
+ "vshl.s32 q11, q10\n\t"
+ // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqmovn.s32 d22, q11\n\t"
+ "vst1.16 d22, [%[ptr_real]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_in]"+r"(ptr_in),
+ [ptr_real]"+r"(ptr_real)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+}
+
+// For the noise supress process, synthesis, read out fully processed segment,
+// and update synthesis buffer.
+static void SynthesisUpdateNeon(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor) {
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_syn = &inst->synthesisBuffer[0];
+ int16_t* ptr_window = &inst->window[0];
+
+ // synthesis
+ __asm__ __volatile__("vdup.16 d24, %0" : : "r"(gain_factor) : "d24");
+ // Loop unrolled once. All pointers are incremented in the assembly code.
+ for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
+ __asm__ __volatile__(
+ // Load variables.
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_window]]!\n\t"
+ "vld1.16 d25, [%[ptr_syn]]\n\t"
+ // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ "vmull.s16 q11, d22, d23\n\t"
+ "vrshrn.i32 d22, q11, #14\n\t"
+ // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
+ "vmull.s16 q11, d24, d22\n\t"
+ // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqrshrn.s32 d22, q11, #13\n\t"
+ // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
+ // inst->synthesisBuffer[i], tmp16b); // Q0
+ "vqadd.s16 d25, d22\n\t"
+ "vst1.16 d25, [%[ptr_syn]]!\n\t"
+
+ // Load variables.
+ "vld1.16 d26, [%[ptr_real]]!\n\t"
+ "vld1.16 d27, [%[ptr_window]]!\n\t"
+ "vld1.16 d28, [%[ptr_syn]]\n\t"
+ // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ "vmull.s16 q13, d26, d27\n\t"
+ "vrshrn.i32 d26, q13, #14\n\t"
+ // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
+ "vmull.s16 q13, d24, d26\n\t"
+ // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqrshrn.s32 d26, q13, #13\n\t"
+ // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
+ // inst->synthesisBuffer[i], tmp16b); // Q0
+ "vqadd.s16 d28, d26\n\t"
+ "vst1.16 d28, [%[ptr_syn]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_real]"+r"(ptr_real),
+ [ptr_window]"+r"(ptr_window),
+ [ptr_syn]"+r"(ptr_syn)
+ :
+ :"d22", "d23", "d24", "d25", "d26", "d27", "d28", "q11", "q12", "q13"
+ );
+ }
+
+ int16_t* ptr_out = &out_frame[0];
+ ptr_syn = &inst->synthesisBuffer[0];
+ // read out fully processed segment
+ for (; ptr_syn < &inst->synthesisBuffer[inst->blockLen10ms];) {
+ // Loop unrolled once. Both pointers are incremented in the assembly code.
+ __asm__ __volatile__(
+ // out_frame[i] = inst->synthesisBuffer[i]; // Q0
+ "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
+ "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
+ :[ptr_syn]"+r"(ptr_syn),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+
+ // Update synthesis buffer.
+ // C code:
+ // WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ // inst->synthesisBuffer + inst->blockLen10ms,
+ // inst->anaLen - inst->blockLen10ms);
+ ptr_out = &inst->synthesisBuffer[0],
+ ptr_syn = &inst->synthesisBuffer[inst->blockLen10ms];
+ for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
+ // Loop unrolled once. Both pointers are incremented in the assembly code.
+ __asm__ __volatile__(
+ "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
+ "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
+ :[ptr_syn]"+r"(ptr_syn),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+
+ // C code:
+ // WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
+ // + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
+ __asm__ __volatile__("vdup.16 q10, %0" : : "r"(0) : "q10");
+ for (; ptr_out < &inst->synthesisBuffer[inst->anaLen];) {
+ // Loop unrolled once. Pointer is incremented in the assembly code.
+ __asm__ __volatile__(
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ :[ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21"
+ );
+ }
+}
+
+// Update analysis buffer for lower band, and window data before FFT.
+static void AnalysisUpdateNeon(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech) {
+
+ int16_t* ptr_ana = &inst->analysisBuffer[inst->blockLen10ms];
+ int16_t* ptr_out = &inst->analysisBuffer[0];
+
+ // For lower band update analysis buffer.
+ // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
+ // inst->analysisBuffer + inst->blockLen10ms,
+ // inst->anaLen - inst->blockLen10ms);
+ for (; ptr_out < &inst->analysisBuffer[inst->anaLen - inst->blockLen10ms];) {
+ // Loop unrolled once, so both pointers are incremented by 8 twice.
+ __asm__ __volatile__(
+ "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23"
+ );
+ }
+
+ // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
+ // + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+ for (ptr_ana = new_speech; ptr_out < &inst->analysisBuffer[inst->anaLen];) {
+ // Loop unrolled once, so both pointers are incremented by 8 twice.
+ __asm__ __volatile__(
+ "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23"
+ );
+ }
+
+ // Window data before FFT
+ int16_t* ptr_window = &inst->window[0];
+ ptr_out = &out[0];
+ ptr_ana = &inst->analysisBuffer[0];
+ for (; ptr_out < &out[inst->anaLen];) {
+
+ // Loop unrolled once, so all pointers are incremented by 4 twice.
+ __asm__ __volatile__(
+ "vld1.16 d20, [%[ptr_ana]]!\n\t"
+ "vld1.16 d21, [%[ptr_window]]!\n\t"
+ // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ "vmull.s16 q10, d20, d21\n\t"
+ "vrshrn.i32 d20, q10, #14\n\t"
+ "vst1.16 d20, [%[ptr_out]]!\n\t"
+
+ "vld1.16 d22, [%[ptr_ana]]!\n\t"
+ "vld1.16 d23, [%[ptr_window]]!\n\t"
+ // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ "vmull.s16 q11, d22, d23\n\t"
+ "vrshrn.i32 d22, q11, #14\n\t"
+ "vst1.16 d22, [%[ptr_out]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_window]"+r"(ptr_window),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23", "q10", "q11"
+ );
+ }
+}
+
+// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
+// zeros, and normalize it.
+static __inline void CreateComplexBufferNeon(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out) {
+ int16_t* ptr_out = &out[0];
+ int16_t* ptr_in = &in[0];
+
+ __asm__ __volatile__("vdup.16 d25, %0" : : "r"(0) : "d25");
+ __asm__ __volatile__("vdup.16 q10, %0" : : "r"(inst->normData) : "q10");
+ for (; ptr_in < &in[inst->anaLen];) {
+
+ // Loop unrolled once, so ptr_in is incremented by 8 twice,
+ // and ptr_out is incremented by 8 four times.
+ __asm__ __volatile__(
+ // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vshl.s16 q11, q10\n\t"
+ "vmov d24, d23\n\t"
+
+ // out[j + 1] = 0; // Insert zeros in imaginary part
+ "vmov d23, d25\n\t"
+ "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
+
+ // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vshl.s16 q11, q10\n\t"
+ "vmov d24, d23\n\t"
+
+ // out[j + 1] = 0; // Insert zeros in imaginary part
+ "vmov d23, d25\n\t"
+ "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_in]"+r"(ptr_in),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25", "q10", "q11"
+ );
+ }
+}
+
+void WebRtcNsx_InitNeon(void) {
+ WebRtcNsx_NoiseEstimation = NoiseEstimationNeon;
+ WebRtcNsx_PrepareSpectrum = PrepareSpectrumNeon;
+ WebRtcNsx_SynthesisUpdate = SynthesisUpdateNeon;
+ WebRtcNsx_AnalysisUpdate = AnalysisUpdateNeon;
+ WebRtcNsx_Denormalize = DenormalizeNeon;
+ WebRtcNsx_CreateComplexBuffer = CreateComplexBufferNeon;
+}
diff --git a/trunk/src/modules/audio_processing/ns/nsx_defines.h b/trunk/src/modules/audio_processing/ns/nsx_defines.h
new file mode 100644
index 0000000..cd1e3bf
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/nsx_defines.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
+
+#define ANAL_BLOCKL_MAX 256 // max analysis block length
+#define HALF_ANAL_BLOCKL 129 // half max analysis block length + 1
+#define SIMULT 3
+#define END_STARTUP_LONG 200
+#define END_STARTUP_SHORT 50
+#define FACTOR_Q16 (WebRtc_Word32)2621440 // 40 in Q16
+#define FACTOR_Q7 (WebRtc_Word16)5120 // 40 in Q7
+#define FACTOR_Q7_STARTUP (WebRtc_Word16)1024 // 8 in Q7
+#define WIDTH_Q8 3 // 0.01 in Q8 (or 25 )
+//PARAMETERS FOR NEW METHOD
+#define DD_PR_SNR_Q11 2007 // ~= Q11(0.98) DD update of prior SNR
+#define ONE_MINUS_DD_PR_SNR_Q11 41 // DD update of prior SNR
+#define SPECT_FLAT_TAVG_Q14 4915 // (0.30) tavg parameter for spectral flatness measure
+#define SPECT_DIFF_TAVG_Q8 77 // (0.30) tavg parameter for spectral flatness measure
+#define PRIOR_UPDATE_Q14 1638 // Q14(0.1) update parameter of prior model
+#define NOISE_UPDATE_Q8 26 // 26 ~= Q8(0.1) update parameter for noise
+// probability threshold for noise state in speech/noise likelihood
+#define ONE_MINUS_PROB_RANGE_Q8 205 // 205 ~= Q8(0.8)
+#define HIST_PAR_EST 1000 // histogram size for estimation of parameters
+//FEATURE EXTRACTION CONFIG
+//bin size of histogram
+#define BIN_SIZE_LRT 10
+//scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
+// thresholds for prior model
+#define FACTOR_1_LRT_DIFF 6 //for LRT and spectral difference (5 times bigger)
+//for spectral_flatness: used when noise is flatter than speech (10 times bigger)
+#define FACTOR_2_FLAT_Q10 922
+//peak limit for spectral flatness (varies between 0 and 1)
+#define THRES_PEAK_FLAT 24 // * 2 * BIN_SIZE_FLAT_FX
+//limit on spacing of two highest peaks in histogram: spacing determined by bin size
+#define LIM_PEAK_SPACE_FLAT_DIFF 4 // * 2 * BIN_SIZE_DIFF_FX
+//limit on relevance of second peak:
+#define LIM_PEAK_WEIGHT_FLAT_DIFF 2
+#define THRES_FLUCT_LRT 10240 //=20 * inst->modelUpdate; fluctuation limit of LRT feat.
+//limit on the max and min values for the feature thresholds
+#define MAX_FLAT_Q10 38912 // * 2 * BIN_SIZE_FLAT_FX
+#define MIN_FLAT_Q10 4096 // * 2 * BIN_SIZE_FLAT_FX
+#define MAX_DIFF 100 // * 2 * BIN_SIZE_DIFF_FX
+#define MIN_DIFF 16 // * 2 * BIN_SIZE_DIFF_FX
+//criteria of weight of histogram peak to accept/reject feature
+#define THRES_WEIGHT_FLAT_DIFF 154//(int)(0.3*(inst->modelUpdate)) for flatness and difference
+//
+#define STAT_UPDATES 9 // Update every 512 = 1 << 9 block
+#define ONE_MINUS_GAMMA_PAUSE_Q8 13 // ~= Q8(0.05) update for conservative noise estimate
+#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 3 // ~= Q8(0.01) update for transition and noise region
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
diff --git a/trunk/src/modules/audio_processing/ns/windows_private.h b/trunk/src/modules/audio_processing/ns/windows_private.h
new file mode 100644
index 0000000..44c2e84
--- /dev/null
+++ b/trunk/src/modules/audio_processing/ns/windows_private.h
@@ -0,0 +1,574 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
+
+// Hanning window for 4ms 16kHz
+static const float kHanning64w128[128] = {
+ 0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
+ 0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
+ 0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
+ 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+ 0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
+ 0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
+ 0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
+ 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+ 0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
+ 0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
+ 0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
+ 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+ 0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
+ 0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
+ 0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
+ 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+ 0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
+ 0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
+ 0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
+ 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+ 0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
+ 0.99969881869620f, 1.00000000000000f,
+ 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
+ 0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
+ 0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
+ 0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
+ 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
+ 0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
+ 0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
+ 0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
+ 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
+ 0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
+ 0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
+ 0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
+ 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
+ 0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
+ 0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
+ 0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
+ 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
+ 0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
+ 0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
+ 0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
+ 0.07356456359967f, 0.04906767432742f, 0.02454122852291f
+};
+
+
+
+// hybrib Hanning & flat window
+static const float kBlocks80w128[128] = {
+ (float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
+ (float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
+ (float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
+ (float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
+ (float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
+ (float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
+ (float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
+ (float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
+ (float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
+ (float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
+ (float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
+ (float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
+ (float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
+ (float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
+ (float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
+ (float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
+ (float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
+ (float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
+ (float)0.09801714, (float)0.06540313, (float)0.03271908
+};
+
+// hybrib Hanning & flat window
+static const float kBlocks160w256[256] = {
+ (float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
+ (float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
+ (float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
+ (float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
+ (float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
+ (float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
+ (float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
+ (float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
+ (float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
+ (float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
+ (float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
+ (float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
+ (float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
+ (float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
+ (float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
+ (float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
+ (float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
+ (float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
+ (float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
+ (float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
+ (float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
+ (float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
+ (float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
+ (float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
+ (float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
+ (float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
+ (float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
+ (float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
+ (float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
+ (float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
+ (float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
+ (float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
+ (float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
+ (float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
+ (float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
+ (float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
+ (float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
+ (float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
+ (float)0.01636173
+};
+
+// hybrib Hanning & flat window: for 20ms
+static const float kBlocks320w512[512] = {
+ (float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
+ (float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
+ (float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
+ (float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
+ (float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
+ (float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
+ (float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
+ (float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
+ (float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
+ (float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
+ (float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
+ (float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
+ (float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
+ (float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
+ (float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
+ (float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
+ (float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
+ (float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
+ (float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
+ (float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
+ (float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
+ (float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
+ (float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
+ (float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
+ (float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
+ (float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
+ (float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
+ (float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
+ (float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
+ (float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
+ (float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
+ (float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
+ (float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
+ (float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
+ (float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
+ (float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
+ (float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
+ (float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
+ (float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
+ (float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
+ (float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
+ (float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
+ (float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
+ (float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
+ (float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
+ (float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
+ (float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
+ (float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
+ (float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
+ (float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
+ (float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
+ (float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
+ (float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
+ (float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
+ (float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
+ (float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
+ (float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
+ (float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
+ (float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
+ (float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
+ (float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
+ (float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
+ (float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
+ (float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
+ (float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
+ (float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
+ (float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
+ (float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
+ (float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
+ (float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
+ (float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
+ (float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
+ (float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
+ (float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
+ (float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
+ (float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
+ (float)0.01636173, (float)0.00818114
+};
+
+
+// Hanning window: for 15ms at 16kHz with symmetric zeros
+static const float kBlocks240w512[512] = {
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
+ (float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
+ (float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
+ (float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
+ (float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
+ (float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
+ (float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
+ (float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
+ (float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
+ (float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
+ (float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
+ (float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
+ (float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
+ (float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
+ (float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
+ (float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
+ (float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
+ (float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
+ (float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
+ (float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
+ (float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
+ (float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
+ (float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
+ (float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
+ (float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
+ (float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
+ (float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
+ (float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
+ (float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
+ (float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
+ (float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
+ (float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
+ (float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
+ (float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
+ (float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
+ (float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
+ (float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
+ (float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
+ (float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
+ (float)0.95501995, (float)0.95694035, (float)0.95881975, (float)0.96065807, (float)0.96245527,
+ (float)0.96421117, (float)0.96592581, (float)0.96759909, (float)0.96923089, (float)0.97082120,
+ (float)0.97236991, (float)0.97387701, (float)0.97534233, (float)0.97676587, (float)0.97814763,
+ (float)0.97948742, (float)0.98078531, (float)0.98204112, (float)0.98325491, (float)0.98442656,
+ (float)0.98555607, (float)0.98664331, (float)0.98768836, (float)0.98869103, (float)0.98965138,
+ (float)0.99056935, (float)0.99144489, (float)0.99227792, (float)0.99306846, (float)0.99381649,
+ (float)0.99452192, (float)0.99518472, (float)0.99580491, (float)0.99638247, (float)0.99691731,
+ (float)0.99740952, (float)0.99785894, (float)0.99826562, (float)0.99862951, (float)0.99895066,
+ (float)0.99922901, (float)0.99946457, (float)0.99965733, (float)0.99980724, (float)0.99991435,
+ (float)0.99997860, (float)1.00000000, (float)0.99997860, (float)0.99991435, (float)0.99980724,
+ (float)0.99965733, (float)0.99946457, (float)0.99922901, (float)0.99895066, (float)0.99862951,
+ (float)0.99826562, (float)0.99785894, (float)0.99740946, (float)0.99691731, (float)0.99638247,
+ (float)0.99580491, (float)0.99518472, (float)0.99452192, (float)0.99381644, (float)0.99306846,
+ (float)0.99227792, (float)0.99144489, (float)0.99056935, (float)0.98965138, (float)0.98869103,
+ (float)0.98768836, (float)0.98664331, (float)0.98555607, (float)0.98442656, (float)0.98325491,
+ (float)0.98204112, (float)0.98078525, (float)0.97948742, (float)0.97814757, (float)0.97676587,
+ (float)0.97534227, (float)0.97387695, (float)0.97236991, (float)0.97082120, (float)0.96923089,
+ (float)0.96759909, (float)0.96592581, (float)0.96421117, (float)0.96245521, (float)0.96065807,
+ (float)0.95881969, (float)0.95694029, (float)0.95501995, (float)0.95305860, (float)0.95105648,
+ (float)0.94901365, (float)0.94693011, (float)0.94480604, (float)0.94264150, (float)0.94043654,
+ (float)0.93819129, (float)0.93590593, (float)0.93358046, (float)0.93121493, (float)0.92880952,
+ (float)0.92636436, (float)0.92387950, (float)0.92135507, (float)0.91879123, (float)0.91618794,
+ (float)0.91354543, (float)0.91086382, (float)0.90814310, (float)0.90538365, (float)0.90258527,
+ (float)0.89974827, (float)0.89687276, (float)0.89395875, (float)0.89100647, (float)0.88801610,
+ (float)0.88498759, (float)0.88192123, (float)0.87881714, (float)0.87567532, (float)0.87249595,
+ (float)0.86927933, (float)0.86602539, (float)0.86273432, (float)0.85940641, (float)0.85604161,
+ (float)0.85264009, (float)0.84920216, (float)0.84572780, (float)0.84221715, (float)0.83867055,
+ (float)0.83508795, (float)0.83146954, (float)0.82781565, (float)0.82412612, (float)0.82040137,
+ (float)0.81664157, (float)0.81284660, (float)0.80901700, (float)0.80515265, (float)0.80125374,
+ (float)0.79732066, (float)0.79335332, (float)0.78935200, (float)0.78531694, (float)0.78124815,
+ (float)0.77714586, (float)0.77301049, (float)0.76884180, (float)0.76464021, (float)0.76040596,
+ (float)0.75613904, (float)0.75183970, (float)0.74750835, (float)0.74314481, (float)0.73874938,
+ (float)0.73432249, (float)0.72986400, (float)0.72537428, (float)0.72085363, (float)0.71630186,
+ (float)0.71171951, (float)0.70710677, (float)0.70246363, (float)0.69779032, (float)0.69308734,
+ (float)0.68835449, (float)0.68359220, (float)0.67880070, (float)0.67398006, (float)0.66913044,
+ (float)0.66425240, (float)0.65934575, (float)0.65441096, (float)0.64944804, (float)0.64445722,
+ (float)0.63943905, (float)0.63439327, (float)0.62932026, (float)0.62422055, (float)0.61909389,
+ (float)0.61394072, (float)0.60876143, (float)0.60355592, (float)0.59832448, (float)0.59306765,
+ (float)0.58778518, (float)0.58247757, (float)0.57714522, (float)0.57178789, (float)0.56640613,
+ (float)0.56100023, (float)0.55557019, (float)0.55011630, (float)0.54463905, (float)0.53913826,
+ (float)0.53361434, (float)0.52806783, (float)0.52249849, (float)0.51690674, (float)0.51129305,
+ (float)0.50565726, (float)0.50000006, (float)0.49432117, (float)0.48862115, (float)0.48290038,
+ (float)0.47715873, (float)0.47139663, (float)0.46561456, (float)0.45981231, (float)0.45399037,
+ (float)0.44814920, (float)0.44228864, (float)0.43640912, (float)0.43051112, (float)0.42459446,
+ (float)0.41865960, (float)0.41270703, (float)0.40673658, (float)0.40074870, (float)0.39474386,
+ (float)0.38872188, (float)0.38268328, (float)0.37662849, (float)0.37055734, (float)0.36447033,
+ (float)0.35836792, (float)0.35224995, (float)0.34611690, (float)0.33996922, (float)0.33380675,
+ (float)0.32763001, (float)0.32143945, (float)0.31523487, (float)0.30901679, (float)0.30278572,
+ (float)0.29654145, (float)0.29028472, (float)0.28401530, (float)0.27773371, (float)0.27144048,
+ (float)0.26513538, (float)0.25881892, (float)0.25249159, (float)0.24615324, (float)0.23980433,
+ (float)0.23344538, (float)0.22707619, (float)0.22069728, (float)0.21430916, (float)0.20791161,
+ (float)0.20150517, (float)0.19509031, (float)0.18866688, (float)0.18223536, (float)0.17579627,
+ (float)0.16934940, (float)0.16289529, (float)0.15643445, (float)0.14996666, (float)0.14349243,
+ (float)0.13701232, (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
+ (float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220, (float)0.07845904,
+ (float)0.07193252, (float)0.06540315, (float)0.05887074, (float)0.05233581, (float)0.04579888,
+ (float)0.03925974, (float)0.03271893, (float)0.02617695, (float)0.01963361, (float)0.01308943,
+ (float)0.00654493, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000
+};
+
+
+// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
+static const float kBlocks480w1024[1024] = {
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
+ (float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
+ (float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
+ (float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
+ (float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
+ (float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
+ (float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
+ (float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
+ (float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
+ (float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
+ (float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
+ (float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
+ (float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
+ (float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
+ (float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
+ (float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
+ (float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
+ (float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
+ (float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
+ (float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
+ (float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
+ (float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
+ (float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
+ (float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
+ (float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
+ (float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
+ (float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
+ (float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
+ (float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
+ (float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
+ (float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
+ (float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
+ (float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
+ (float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
+ (float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
+ (float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
+ (float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
+ (float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
+ (float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
+ (float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
+ (float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
+ (float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
+ (float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
+ (float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
+ (float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
+ (float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
+ (float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
+ (float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
+ (float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
+ (float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
+ (float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
+ (float)0.73653996, (float)0.73874950, (float)0.74095118, (float)0.74314487, (float)0.74533057,
+ (float)0.74750835, (float)0.74967808, (float)0.75183982, (float)0.75399351, (float)0.75613910,
+ (float)0.75827658, (float)0.76040596, (float)0.76252723, (float)0.76464027, (float)0.76674515,
+ (float)0.76884186, (float)0.77093029, (float)0.77301043, (float)0.77508241, (float)0.77714598,
+ (float)0.77920127, (float)0.78124821, (float)0.78328675, (float)0.78531694, (float)0.78733873,
+ (float)0.78935206, (float)0.79135692, (float)0.79335338, (float)0.79534125, (float)0.79732066,
+ (float)0.79929149, (float)0.80125386, (float)0.80320752, (float)0.80515265, (float)0.80708915,
+ (float)0.80901700, (float)0.81093621, (float)0.81284672, (float)0.81474853, (float)0.81664157,
+ (float)0.81852591, (float)0.82040149, (float)0.82226825, (float)0.82412618, (float)0.82597536,
+ (float)0.82781565, (float)0.82964706, (float)0.83146966, (float)0.83328325, (float)0.83508795,
+ (float)0.83688378, (float)0.83867061, (float)0.84044838, (float)0.84221727, (float)0.84397703,
+ (float)0.84572780, (float)0.84746957, (float)0.84920216, (float)0.85092574, (float)0.85264021,
+ (float)0.85434544, (float)0.85604161, (float)0.85772866, (float)0.85940641, (float)0.86107504,
+ (float)0.86273444, (float)0.86438453, (float)0.86602545, (float)0.86765707, (float)0.86927933,
+ (float)0.87089235, (float)0.87249607, (float)0.87409031, (float)0.87567532, (float)0.87725097,
+ (float)0.87881714, (float)0.88037390, (float)0.88192129, (float)0.88345921, (float)0.88498765,
+ (float)0.88650668, (float)0.88801610, (float)0.88951612, (float)0.89100653, (float)0.89248741,
+ (float)0.89395881, (float)0.89542055, (float)0.89687276, (float)0.89831537, (float)0.89974827,
+ (float)0.90117162, (float)0.90258533, (float)0.90398932, (float)0.90538365, (float)0.90676826,
+ (float)0.90814316, (float)0.90950841, (float)0.91086388, (float)0.91220951, (float)0.91354549,
+ (float)0.91487163, (float)0.91618794, (float)0.91749454, (float)0.91879123, (float)0.92007810,
+ (float)0.92135513, (float)0.92262226, (float)0.92387950, (float)0.92512691, (float)0.92636442,
+ (float)0.92759192, (float)0.92880958, (float)0.93001723, (float)0.93121493, (float)0.93240267,
+ (float)0.93358046, (float)0.93474817, (float)0.93590593, (float)0.93705362, (float)0.93819135,
+ (float)0.93931901, (float)0.94043654, (float)0.94154406, (float)0.94264150, (float)0.94372880,
+ (float)0.94480604, (float)0.94587320, (float)0.94693011, (float)0.94797695, (float)0.94901365,
+ (float)0.95004016, (float)0.95105654, (float)0.95206273, (float)0.95305866, (float)0.95404440,
+ (float)0.95501995, (float)0.95598525, (float)0.95694035, (float)0.95788521, (float)0.95881975,
+ (float)0.95974404, (float)0.96065807, (float)0.96156180, (float)0.96245527, (float)0.96333838,
+ (float)0.96421117, (float)0.96507370, (float)0.96592581, (float)0.96676767, (float)0.96759909,
+ (float)0.96842021, (float)0.96923089, (float)0.97003126, (float)0.97082120, (float)0.97160077,
+ (float)0.97236991, (float)0.97312868, (float)0.97387701, (float)0.97461486, (float)0.97534233,
+ (float)0.97605932, (float)0.97676587, (float)0.97746199, (float)0.97814763, (float)0.97882277,
+ (float)0.97948742, (float)0.98014158, (float)0.98078531, (float)0.98141843, (float)0.98204112,
+ (float)0.98265332, (float)0.98325491, (float)0.98384601, (float)0.98442656, (float)0.98499662,
+ (float)0.98555607, (float)0.98610497, (float)0.98664331, (float)0.98717111, (float)0.98768836,
+ (float)0.98819500, (float)0.98869103, (float)0.98917651, (float)0.98965138, (float)0.99011570,
+ (float)0.99056935, (float)0.99101239, (float)0.99144489, (float)0.99186671, (float)0.99227792,
+ (float)0.99267852, (float)0.99306846, (float)0.99344778, (float)0.99381649, (float)0.99417448,
+ (float)0.99452192, (float)0.99485862, (float)0.99518472, (float)0.99550015, (float)0.99580491,
+ (float)0.99609905, (float)0.99638247, (float)0.99665523, (float)0.99691731, (float)0.99716878,
+ (float)0.99740952, (float)0.99763954, (float)0.99785894, (float)0.99806762, (float)0.99826562,
+ (float)0.99845290, (float)0.99862951, (float)0.99879545, (float)0.99895066, (float)0.99909520,
+ (float)0.99922901, (float)0.99935216, (float)0.99946457, (float)0.99956632, (float)0.99965733,
+ (float)0.99973762, (float)0.99980724, (float)0.99986613, (float)0.99991435, (float)0.99995178,
+ (float)0.99997860, (float)0.99999464, (float)1.00000000, (float)0.99999464, (float)0.99997860,
+ (float)0.99995178, (float)0.99991435, (float)0.99986613, (float)0.99980724, (float)0.99973762,
+ (float)0.99965733, (float)0.99956632, (float)0.99946457, (float)0.99935216, (float)0.99922901,
+ (float)0.99909520, (float)0.99895066, (float)0.99879545, (float)0.99862951, (float)0.99845290,
+ (float)0.99826562, (float)0.99806762, (float)0.99785894, (float)0.99763954, (float)0.99740946,
+ (float)0.99716872, (float)0.99691731, (float)0.99665523, (float)0.99638247, (float)0.99609905,
+ (float)0.99580491, (float)0.99550015, (float)0.99518472, (float)0.99485862, (float)0.99452192,
+ (float)0.99417448, (float)0.99381644, (float)0.99344778, (float)0.99306846, (float)0.99267852,
+ (float)0.99227792, (float)0.99186671, (float)0.99144489, (float)0.99101239, (float)0.99056935,
+ (float)0.99011564, (float)0.98965138, (float)0.98917651, (float)0.98869103, (float)0.98819494,
+ (float)0.98768836, (float)0.98717111, (float)0.98664331, (float)0.98610497, (float)0.98555607,
+ (float)0.98499656, (float)0.98442656, (float)0.98384601, (float)0.98325491, (float)0.98265326,
+ (float)0.98204112, (float)0.98141843, (float)0.98078525, (float)0.98014158, (float)0.97948742,
+ (float)0.97882277, (float)0.97814757, (float)0.97746193, (float)0.97676587, (float)0.97605932,
+ (float)0.97534227, (float)0.97461486, (float)0.97387695, (float)0.97312862, (float)0.97236991,
+ (float)0.97160077, (float)0.97082120, (float)0.97003126, (float)0.96923089, (float)0.96842015,
+ (float)0.96759909, (float)0.96676761, (float)0.96592581, (float)0.96507365, (float)0.96421117,
+ (float)0.96333838, (float)0.96245521, (float)0.96156180, (float)0.96065807, (float)0.95974404,
+ (float)0.95881969, (float)0.95788515, (float)0.95694029, (float)0.95598525, (float)0.95501995,
+ (float)0.95404440, (float)0.95305860, (float)0.95206267, (float)0.95105648, (float)0.95004016,
+ (float)0.94901365, (float)0.94797695, (float)0.94693011, (float)0.94587314, (float)0.94480604,
+ (float)0.94372880, (float)0.94264150, (float)0.94154406, (float)0.94043654, (float)0.93931895,
+ (float)0.93819129, (float)0.93705362, (float)0.93590593, (float)0.93474817, (float)0.93358046,
+ (float)0.93240267, (float)0.93121493, (float)0.93001723, (float)0.92880952, (float)0.92759192,
+ (float)0.92636436, (float)0.92512691, (float)0.92387950, (float)0.92262226, (float)0.92135507,
+ (float)0.92007804, (float)0.91879123, (float)0.91749448, (float)0.91618794, (float)0.91487157,
+ (float)0.91354543, (float)0.91220951, (float)0.91086382, (float)0.90950835, (float)0.90814310,
+ (float)0.90676820, (float)0.90538365, (float)0.90398932, (float)0.90258527, (float)0.90117157,
+ (float)0.89974827, (float)0.89831525, (float)0.89687276, (float)0.89542055, (float)0.89395875,
+ (float)0.89248741, (float)0.89100647, (float)0.88951600, (float)0.88801610, (float)0.88650662,
+ (float)0.88498759, (float)0.88345915, (float)0.88192123, (float)0.88037384, (float)0.87881714,
+ (float)0.87725091, (float)0.87567532, (float)0.87409031, (float)0.87249595, (float)0.87089223,
+ (float)0.86927933, (float)0.86765701, (float)0.86602539, (float)0.86438447, (float)0.86273432,
+ (float)0.86107504, (float)0.85940641, (float)0.85772860, (float)0.85604161, (float)0.85434544,
+ (float)0.85264009, (float)0.85092574, (float)0.84920216, (float)0.84746951, (float)0.84572780,
+ (float)0.84397697, (float)0.84221715, (float)0.84044844, (float)0.83867055, (float)0.83688372,
+ (float)0.83508795, (float)0.83328319, (float)0.83146954, (float)0.82964706, (float)0.82781565,
+ (float)0.82597530, (float)0.82412612, (float)0.82226813, (float)0.82040137, (float)0.81852591,
+ (float)0.81664157, (float)0.81474847, (float)0.81284660, (float)0.81093609, (float)0.80901700,
+ (float)0.80708915, (float)0.80515265, (float)0.80320752, (float)0.80125374, (float)0.79929143,
+ (float)0.79732066, (float)0.79534125, (float)0.79335332, (float)0.79135686, (float)0.78935200,
+ (float)0.78733861, (float)0.78531694, (float)0.78328675, (float)0.78124815, (float)0.77920121,
+ (float)0.77714586, (float)0.77508223, (float)0.77301049, (float)0.77093029, (float)0.76884180,
+ (float)0.76674509, (float)0.76464021, (float)0.76252711, (float)0.76040596, (float)0.75827658,
+ (float)0.75613904, (float)0.75399339, (float)0.75183970, (float)0.74967796, (float)0.74750835,
+ (float)0.74533057, (float)0.74314481, (float)0.74095106, (float)0.73874938, (float)0.73653996,
+ (float)0.73432249, (float)0.73209721, (float)0.72986400, (float)0.72762305, (float)0.72537428,
+ (float)0.72311789, (float)0.72085363, (float)0.71858162, (float)0.71630186, (float)0.71401453,
+ (float)0.71171951, (float)0.70941705, (float)0.70710677, (float)0.70478898, (float)0.70246363,
+ (float)0.70013070, (float)0.69779032, (float)0.69544268, (float)0.69308734, (float)0.69072461,
+ (float)0.68835449, (float)0.68597704, (float)0.68359220, (float)0.68120021, (float)0.67880070,
+ (float)0.67639399, (float)0.67398006, (float)0.67155886, (float)0.66913044, (float)0.66669512,
+ (float)0.66425240, (float)0.66180259, (float)0.65934575, (float)0.65688181, (float)0.65441096,
+ (float)0.65193301, (float)0.64944804, (float)0.64695609, (float)0.64445722, (float)0.64195150,
+ (float)0.63943905, (float)0.63691956, (float)0.63439327, (float)0.63186014, (float)0.62932026,
+ (float)0.62677372, (float)0.62422055, (float)0.62166059, (float)0.61909389, (float)0.61652064,
+ (float)0.61394072, (float)0.61135429, (float)0.60876143, (float)0.60616189, (float)0.60355592,
+ (float)0.60094339, (float)0.59832448, (float)0.59569913, (float)0.59306765, (float)0.59042960,
+ (float)0.58778518, (float)0.58513451, (float)0.58247757, (float)0.57981461, (float)0.57714522,
+ (float)0.57446963, (float)0.57178789, (float)0.56910002, (float)0.56640613, (float)0.56370628,
+ (float)0.56100023, (float)0.55828822, (float)0.55557019, (float)0.55284619, (float)0.55011630,
+ (float)0.54738069, (float)0.54463905, (float)0.54189152, (float)0.53913826, (float)0.53637916,
+ (float)0.53361434, (float)0.53084403, (float)0.52806783, (float)0.52528596, (float)0.52249849,
+ (float)0.51970541, (float)0.51690674, (float)0.51410276, (float)0.51129305, (float)0.50847787,
+ (float)0.50565726, (float)0.50283122, (float)0.50000006, (float)0.49716327, (float)0.49432117,
+ (float)0.49147379, (float)0.48862115, (float)0.48576325, (float)0.48290038, (float)0.48003212,
+ (float)0.47715873, (float)0.47428021, (float)0.47139663, (float)0.46850798, (float)0.46561456,
+ (float)0.46271589, (float)0.45981231, (float)0.45690379, (float)0.45399037, (float)0.45107210,
+ (float)0.44814920, (float)0.44522130, (float)0.44228864, (float)0.43935123, (float)0.43640912,
+ (float)0.43346232, (float)0.43051112, (float)0.42755505, (float)0.42459446, (float)0.42162928,
+ (float)0.41865960, (float)0.41568545, (float)0.41270703, (float)0.40972400, (float)0.40673658,
+ (float)0.40374479, (float)0.40074870, (float)0.39774850, (float)0.39474386, (float)0.39173496,
+ (float)0.38872188, (float)0.38570464, (float)0.38268328, (float)0.37965804, (float)0.37662849,
+ (float)0.37359491, (float)0.37055734, (float)0.36751580, (float)0.36447033, (float)0.36142117,
+ (float)0.35836792, (float)0.35531086, (float)0.35224995, (float)0.34918529, (float)0.34611690,
+ (float)0.34304500, (float)0.33996922, (float)0.33688980, (float)0.33380675, (float)0.33072016,
+ (float)0.32763001, (float)0.32453656, (float)0.32143945, (float)0.31833887, (float)0.31523487,
+ (float)0.31212750, (float)0.30901679, (float)0.30590302, (float)0.30278572, (float)0.29966521,
+ (float)0.29654145, (float)0.29341453, (float)0.29028472, (float)0.28715155, (float)0.28401530,
+ (float)0.28087601, (float)0.27773371, (float)0.27458847, (float)0.27144048, (float)0.26828936,
+ (float)0.26513538, (float)0.26197854, (float)0.25881892, (float)0.25565651, (float)0.25249159,
+ (float)0.24932374, (float)0.24615324, (float)0.24298008, (float)0.23980433, (float)0.23662600,
+ (float)0.23344538, (float)0.23026201, (float)0.22707619, (float)0.22388794, (float)0.22069728,
+ (float)0.21750426, (float)0.21430916, (float)0.21111152, (float)0.20791161, (float)0.20470949,
+ (float)0.20150517, (float)0.19829892, (float)0.19509031, (float)0.19187963, (float)0.18866688,
+ (float)0.18545210, (float)0.18223536, (float)0.17901689, (float)0.17579627, (float)0.17257376,
+ (float)0.16934940, (float)0.16612324, (float)0.16289529, (float)0.15966584, (float)0.15643445,
+ (float)0.15320137, (float)0.14996666, (float)0.14673033, (float)0.14349243, (float)0.14025325,
+ (float)0.13701232, (float)0.13376991, (float)0.13052608, (float)0.12728085, (float)0.12403426,
+ (float)0.12078657, (float)0.11753736, (float)0.11428688, (float)0.11103519, (float)0.10778230,
+ (float)0.10452849, (float)0.10127334, (float)0.09801710, (float)0.09475980, (float)0.09150149,
+ (float)0.08824220, (float)0.08498220, (float)0.08172106, (float)0.07845904, (float)0.07519618,
+ (float)0.07193252, (float)0.06866808, (float)0.06540315, (float)0.06213728, (float)0.05887074,
+ (float)0.05560357, (float)0.05233581, (float)0.04906749, (float)0.04579888, (float)0.04252954,
+ (float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798, (float)0.02617695,
+ (float)0.02290541, (float)0.01963361, (float)0.01636161, (float)0.01308943, (float)0.00981712,
+ (float)0.00654493, (float)0.00327244, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000
+};
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
diff --git a/trunk/src/modules/audio_processing/processing_component.cc b/trunk/src/modules/audio_processing/processing_component.cc
new file mode 100644
index 0000000..9ac1257
--- /dev/null
+++ b/trunk/src/modules/audio_processing/processing_component.cc
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "processing_component.h"
+
+#include <cassert>
+
+#include "audio_processing_impl.h"
+
+namespace webrtc {
+
+ProcessingComponent::ProcessingComponent(const AudioProcessingImpl* apm)
+ : apm_(apm),
+ initialized_(false),
+ enabled_(false),
+ num_handles_(0) {}
+
+ProcessingComponent::~ProcessingComponent() {
+ assert(initialized_ == false);
+}
+
+int ProcessingComponent::Destroy() {
+ while (!handles_.empty()) {
+ DestroyHandle(handles_.back());
+ handles_.pop_back();
+ }
+ initialized_ = false;
+
+ return apm_->kNoError;
+}
+
+int ProcessingComponent::EnableComponent(bool enable) {
+ if (enable && !enabled_) {
+ enabled_ = enable; // Must be set before Initialize() is called.
+
+ int err = Initialize();
+ if (err != apm_->kNoError) {
+ enabled_ = false;
+ return err;
+ }
+ } else {
+ enabled_ = enable;
+ }
+
+ return apm_->kNoError;
+}
+
+bool ProcessingComponent::is_component_enabled() const {
+ return enabled_;
+}
+
+void* ProcessingComponent::handle(int index) const {
+ assert(index < num_handles_);
+ return handles_[index];
+}
+
+int ProcessingComponent::num_handles() const {
+ return num_handles_;
+}
+
+int ProcessingComponent::Initialize() {
+ if (!enabled_) {
+ return apm_->kNoError;
+ }
+
+ num_handles_ = num_handles_required();
+ if (num_handles_ > static_cast<int>(handles_.size())) {
+ handles_.resize(num_handles_, NULL);
+ }
+
+ assert(static_cast<int>(handles_.size()) >= num_handles_);
+ for (int i = 0; i < num_handles_; i++) {
+ if (handles_[i] == NULL) {
+ handles_[i] = CreateHandle();
+ if (handles_[i] == NULL) {
+ return apm_->kCreationFailedError;
+ }
+ }
+
+ int err = InitializeHandle(handles_[i]);
+ if (err != apm_->kNoError) {
+ return GetHandleError(handles_[i]);
+ }
+ }
+
+ initialized_ = true;
+ return Configure();
+}
+
+int ProcessingComponent::Configure() {
+ if (!initialized_) {
+ return apm_->kNoError;
+ }
+
+ assert(static_cast<int>(handles_.size()) >= num_handles_);
+ for (int i = 0; i < num_handles_; i++) {
+ int err = ConfigureHandle(handles_[i]);
+ if (err != apm_->kNoError) {
+ return GetHandleError(handles_[i]);
+ }
+ }
+
+ return apm_->kNoError;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/processing_component.h b/trunk/src/modules/audio_processing/processing_component.h
new file mode 100644
index 0000000..b3457b5
--- /dev/null
+++ b/trunk/src/modules/audio_processing/processing_component.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
+
+#include <vector>
+
+#include "audio_processing.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+
+class ProcessingComponent {
+ public:
+ explicit ProcessingComponent(const AudioProcessingImpl* apm);
+ virtual ~ProcessingComponent();
+
+ virtual int Initialize();
+ virtual int Destroy();
+
+ bool is_component_enabled() const;
+
+ protected:
+ virtual int Configure();
+ int EnableComponent(bool enable);
+ void* handle(int index) const;
+ int num_handles() const;
+
+ private:
+ virtual void* CreateHandle() const = 0;
+ virtual int InitializeHandle(void* handle) const = 0;
+ virtual int ConfigureHandle(void* handle) const = 0;
+ virtual int DestroyHandle(void* handle) const = 0;
+ virtual int num_handles_required() const = 0;
+ virtual int GetHandleError(void* handle) const = 0;
+
+ const AudioProcessingImpl* apm_;
+ std::vector<void*> handles_;
+ bool initialized_;
+ bool enabled_;
+ int num_handles_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H__
diff --git a/trunk/src/modules/audio_processing/splitting_filter.cc b/trunk/src/modules/audio_processing/splitting_filter.cc
new file mode 100644
index 0000000..1526141
--- /dev/null
+++ b/trunk/src/modules/audio_processing/splitting_filter.cc
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "splitting_filter.h"
+#include "signal_processing_library.h"
+
+namespace webrtc {
+
+void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
+ WebRtc_Word16* low_band,
+ WebRtc_Word16* high_band,
+ WebRtc_Word32* filter_state1,
+ WebRtc_Word32* filter_state2)
+{
+ WebRtcSpl_AnalysisQMF(in_data, low_band, high_band, filter_state1, filter_state2);
+}
+
+void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
+ const WebRtc_Word16* high_band,
+ WebRtc_Word16* out_data,
+ WebRtc_Word32* filt_state1,
+ WebRtc_Word32* filt_state2)
+{
+ WebRtcSpl_SynthesisQMF(low_band, high_band, out_data, filt_state1, filt_state2);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/splitting_filter.h b/trunk/src/modules/audio_processing/splitting_filter.h
new file mode 100644
index 0000000..661bfb2
--- /dev/null
+++ b/trunk/src/modules/audio_processing/splitting_filter.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+namespace webrtc {
+/*
+ * SplittingFilterbank_analysisQMF(...)
+ *
+ * Splits a super-wb signal into two subbands: 0-8 kHz and 8-16 kHz.
+ *
+ * Input:
+ * - in_data : super-wb audio signal
+ *
+ * Input & Output:
+ * - filt_state1: Filter state for first all-pass filter
+ * - filt_state2: Filter state for second all-pass filter
+ *
+ * Output:
+ * - low_band : The signal from the 0-4 kHz band
+ * - high_band : The signal from the 4-8 kHz band
+ */
+void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
+ WebRtc_Word16* low_band,
+ WebRtc_Word16* high_band,
+ WebRtc_Word32* filt_state1,
+ WebRtc_Word32* filt_state2);
+
+/*
+ * SplittingFilterbank_synthesisQMF(...)
+ *
+ * Combines the two subbands (0-8 and 8-16 kHz) into a super-wb signal.
+ *
+ * Input:
+ * - low_band : The signal with the 0-8 kHz band
+ * - high_band : The signal with the 8-16 kHz band
+ *
+ * Input & Output:
+ * - filt_state1: Filter state for first all-pass filter
+ * - filt_state2: Filter state for second all-pass filter
+ *
+ * Output:
+ * - out_data : super-wb speech signal
+ */
+void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
+ const WebRtc_Word16* high_band,
+ WebRtc_Word16* out_data,
+ WebRtc_Word32* filt_state1,
+ WebRtc_Word32* filt_state2);
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml b/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
new file mode 100644
index 0000000..c6063b3
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- BEGIN_INCLUDE(manifest) -->
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ package="com.example.native_activity"
+ android:versionCode="1"
+ android:versionName="1.0">
+
+ <!-- This is the platform API where NativeActivity was introduced. -->
+ <uses-sdk android:minSdkVersion="8" />
+
+ <!-- This .apk has no Java code itself, so set hasCode to false. -->
+ <application android:label="@string/app_name" android:hasCode="false" android:debuggable="true">
+
+ <!-- Our activity is the built-in NativeActivity framework class.
+ This will take care of integrating with our NDK code. -->
+ <activity android:name="android.app.NativeActivity"
+ android:label="@string/app_name"
+ android:configChanges="orientation|keyboardHidden">
+ <!-- Tell NativeActivity the name of or .so -->
+ <meta-data android:name="android.app.lib_name"
+ android:value="apmtest-activity" />
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+ </application>
+
+</manifest>
+<!-- END_INCLUDE(manifest) -->
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/default.properties b/trunk/src/modules/audio_processing/test/android/apmtest/default.properties
new file mode 100644
index 0000000..9a2c9f6
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target.
+target=android-9
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk
new file mode 100644
index 0000000..eaf3c9d
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk
@@ -0,0 +1,26 @@
+# Copyright (C) 2010 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+#
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := apmtest-activity
+LOCAL_SRC_FILES := main.c
+LOCAL_LDLIBS := -llog -landroid -lEGL -lGLESv1_CM
+LOCAL_STATIC_LIBRARIES := android_native_app_glue
+
+include $(BUILD_SHARED_LIBRARY)
+
+$(call import-module,android/native_app_glue)
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk
new file mode 100644
index 0000000..22d188e
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk
@@ -0,0 +1 @@
+APP_PLATFORM := android-9
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c b/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c
new file mode 100644
index 0000000..2e19635
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+//BEGIN_INCLUDE(all)
+#include <jni.h>
+#include <errno.h>
+
+#include <EGL/egl.h>
+#include <GLES/gl.h>
+
+#include <android/sensor.h>
+#include <android/log.h>
+#include <android_native_app_glue.h>
+
+#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "native-activity", __VA_ARGS__))
+#define LOGW(...) ((void)__android_log_print(ANDROID_LOG_WARN, "native-activity", __VA_ARGS__))
+
+/**
+ * Our saved state data.
+ */
+struct saved_state {
+ float angle;
+ int32_t x;
+ int32_t y;
+};
+
+/**
+ * Shared state for our app.
+ */
+struct engine {
+ struct android_app* app;
+
+ ASensorManager* sensorManager;
+ const ASensor* accelerometerSensor;
+ ASensorEventQueue* sensorEventQueue;
+
+ int animating;
+ EGLDisplay display;
+ EGLSurface surface;
+ EGLContext context;
+ int32_t width;
+ int32_t height;
+ struct saved_state state;
+};
+
+/**
+ * Initialize an EGL context for the current display.
+ */
+static int engine_init_display(struct engine* engine) {
+ // initialize OpenGL ES and EGL
+
+ /*
+ * Here specify the attributes of the desired configuration.
+ * Below, we select an EGLConfig with at least 8 bits per color
+ * component compatible with on-screen windows
+ */
+ const EGLint attribs[] = {
+ EGL_SURFACE_TYPE, EGL_WINDOW_BIT,
+ EGL_BLUE_SIZE, 8,
+ EGL_GREEN_SIZE, 8,
+ EGL_RED_SIZE, 8,
+ EGL_NONE
+ };
+ EGLint w, h, dummy, format;
+ EGLint numConfigs;
+ EGLConfig config;
+ EGLSurface surface;
+ EGLContext context;
+
+ EGLDisplay display = eglGetDisplay(EGL_DEFAULT_DISPLAY);
+
+ eglInitialize(display, 0, 0);
+
+ /* Here, the application chooses the configuration it desires. In this
+ * sample, we have a very simplified selection process, where we pick
+ * the first EGLConfig that matches our criteria */
+ eglChooseConfig(display, attribs, &config, 1, &numConfigs);
+
+ /* EGL_NATIVE_VISUAL_ID is an attribute of the EGLConfig that is
+ * guaranteed to be accepted by ANativeWindow_setBuffersGeometry().
+ * As soon as we picked a EGLConfig, we can safely reconfigure the
+ * ANativeWindow buffers to match, using EGL_NATIVE_VISUAL_ID. */
+ eglGetConfigAttrib(display, config, EGL_NATIVE_VISUAL_ID, &format);
+
+ ANativeWindow_setBuffersGeometry(engine->app->window, 0, 0, format);
+
+ surface = eglCreateWindowSurface(display, config, engine->app->window, NULL);
+ context = eglCreateContext(display, config, NULL, NULL);
+
+ if (eglMakeCurrent(display, surface, surface, context) == EGL_FALSE) {
+ LOGW("Unable to eglMakeCurrent");
+ return -1;
+ }
+
+ eglQuerySurface(display, surface, EGL_WIDTH, &w);
+ eglQuerySurface(display, surface, EGL_HEIGHT, &h);
+
+ engine->display = display;
+ engine->context = context;
+ engine->surface = surface;
+ engine->width = w;
+ engine->height = h;
+ engine->state.angle = 0;
+
+ // Initialize GL state.
+ glHint(GL_PERSPECTIVE_CORRECTION_HINT, GL_FASTEST);
+ glEnable(GL_CULL_FACE);
+ glShadeModel(GL_SMOOTH);
+ glDisable(GL_DEPTH_TEST);
+
+ return 0;
+}
+
+/**
+ * Just the current frame in the display.
+ */
+static void engine_draw_frame(struct engine* engine) {
+ if (engine->display == NULL) {
+ // No display.
+ return;
+ }
+
+ // Just fill the screen with a color.
+ glClearColor(((float)engine->state.x)/engine->width, engine->state.angle,
+ ((float)engine->state.y)/engine->height, 1);
+ glClear(GL_COLOR_BUFFER_BIT);
+
+ eglSwapBuffers(engine->display, engine->surface);
+}
+
+/**
+ * Tear down the EGL context currently associated with the display.
+ */
+static void engine_term_display(struct engine* engine) {
+ if (engine->display != EGL_NO_DISPLAY) {
+ eglMakeCurrent(engine->display, EGL_NO_SURFACE, EGL_NO_SURFACE, EGL_NO_CONTEXT);
+ if (engine->context != EGL_NO_CONTEXT) {
+ eglDestroyContext(engine->display, engine->context);
+ }
+ if (engine->surface != EGL_NO_SURFACE) {
+ eglDestroySurface(engine->display, engine->surface);
+ }
+ eglTerminate(engine->display);
+ }
+ engine->animating = 0;
+ engine->display = EGL_NO_DISPLAY;
+ engine->context = EGL_NO_CONTEXT;
+ engine->surface = EGL_NO_SURFACE;
+}
+
+/**
+ * Process the next input event.
+ */
+static int32_t engine_handle_input(struct android_app* app, AInputEvent* event) {
+ struct engine* engine = (struct engine*)app->userData;
+ if (AInputEvent_getType(event) == AINPUT_EVENT_TYPE_MOTION) {
+ engine->animating = 1;
+ engine->state.x = AMotionEvent_getX(event, 0);
+ engine->state.y = AMotionEvent_getY(event, 0);
+ return 1;
+ }
+ return 0;
+}
+
+/**
+ * Process the next main command.
+ */
+static void engine_handle_cmd(struct android_app* app, int32_t cmd) {
+ struct engine* engine = (struct engine*)app->userData;
+ switch (cmd) {
+ case APP_CMD_SAVE_STATE:
+ // The system has asked us to save our current state. Do so.
+ engine->app->savedState = malloc(sizeof(struct saved_state));
+ *((struct saved_state*)engine->app->savedState) = engine->state;
+ engine->app->savedStateSize = sizeof(struct saved_state);
+ break;
+ case APP_CMD_INIT_WINDOW:
+ // The window is being shown, get it ready.
+ if (engine->app->window != NULL) {
+ engine_init_display(engine);
+ engine_draw_frame(engine);
+ }
+ break;
+ case APP_CMD_TERM_WINDOW:
+ // The window is being hidden or closed, clean it up.
+ engine_term_display(engine);
+ break;
+ case APP_CMD_GAINED_FOCUS:
+ // When our app gains focus, we start monitoring the accelerometer.
+ if (engine->accelerometerSensor != NULL) {
+ ASensorEventQueue_enableSensor(engine->sensorEventQueue,
+ engine->accelerometerSensor);
+ // We'd like to get 60 events per second (in us).
+ ASensorEventQueue_setEventRate(engine->sensorEventQueue,
+ engine->accelerometerSensor, (1000L/60)*1000);
+ }
+ break;
+ case APP_CMD_LOST_FOCUS:
+ // When our app loses focus, we stop monitoring the accelerometer.
+ // This is to avoid consuming battery while not being used.
+ if (engine->accelerometerSensor != NULL) {
+ ASensorEventQueue_disableSensor(engine->sensorEventQueue,
+ engine->accelerometerSensor);
+ }
+ // Also stop animating.
+ engine->animating = 0;
+ engine_draw_frame(engine);
+ break;
+ }
+}
+
+/**
+ * This is the main entry point of a native application that is using
+ * android_native_app_glue. It runs in its own thread, with its own
+ * event loop for receiving input events and doing other things.
+ */
+void android_main(struct android_app* state) {
+ struct engine engine;
+
+ // Make sure glue isn't stripped.
+ app_dummy();
+
+ memset(&engine, 0, sizeof(engine));
+ state->userData = &engine;
+ state->onAppCmd = engine_handle_cmd;
+ state->onInputEvent = engine_handle_input;
+ engine.app = state;
+
+ // Prepare to monitor accelerometer
+ engine.sensorManager = ASensorManager_getInstance();
+ engine.accelerometerSensor = ASensorManager_getDefaultSensor(engine.sensorManager,
+ ASENSOR_TYPE_ACCELEROMETER);
+ engine.sensorEventQueue = ASensorManager_createEventQueue(engine.sensorManager,
+ state->looper, LOOPER_ID_USER, NULL, NULL);
+
+ if (state->savedState != NULL) {
+ // We are starting with a previous saved state; restore from it.
+ engine.state = *(struct saved_state*)state->savedState;
+ }
+
+ // loop waiting for stuff to do.
+
+ while (1) {
+ // Read all pending events.
+ int ident;
+ int events;
+ struct android_poll_source* source;
+
+ // If not animating, we will block forever waiting for events.
+ // If animating, we loop until all events are read, then continue
+ // to draw the next frame of animation.
+ while ((ident=ALooper_pollAll(engine.animating ? 0 : -1, NULL, &events,
+ (void**)&source)) >= 0) {
+
+ // Process this event.
+ if (source != NULL) {
+ source->process(state, source);
+ }
+
+ // If a sensor has data, process it now.
+ if (ident == LOOPER_ID_USER) {
+ if (engine.accelerometerSensor != NULL) {
+ ASensorEvent event;
+ while (ASensorEventQueue_getEvents(engine.sensorEventQueue,
+ &event, 1) > 0) {
+ LOGI("accelerometer: x=%f y=%f z=%f",
+ event.acceleration.x, event.acceleration.y,
+ event.acceleration.z);
+ }
+ }
+ }
+
+ // Check if we are exiting.
+ if (state->destroyRequested != 0) {
+ engine_term_display(&engine);
+ return;
+ }
+ }
+
+ if (engine.animating) {
+ // Done with events; draw next animation frame.
+ engine.state.angle += .01f;
+ if (engine.state.angle > 1) {
+ engine.state.angle = 0;
+ }
+
+ // Drawing is throttled to the screen update rate, so there
+ // is no need to do timing here.
+ engine_draw_frame(&engine);
+ }
+ }
+}
+//END_INCLUDE(all)
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml b/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
new file mode 100644
index 0000000..d0bd0f3
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
@@ -0,0 +1,4 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+ <string name="app_name">apmtest</string>
+</resources>
diff --git a/trunk/src/modules/audio_processing/test/apmtest.m b/trunk/src/modules/audio_processing/test/apmtest.m
new file mode 100644
index 0000000..3172cd1
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/apmtest.m
@@ -0,0 +1,355 @@
+function apmtest(task, testname, filepath, casenumber, legacy)
+%APMTEST is a tool to process APM file sets and easily display the output.
+% APMTEST(TASK, TESTNAME, CASENUMBER) performs one of several TASKs:
+% 'test' Processes the files to produce test output.
+% 'list' Prints a list of cases in the test set, preceded by their
+% CASENUMBERs.
+% 'show' Uses spclab to show the test case specified by the
+% CASENUMBER parameter.
+%
+% using a set of test files determined by TESTNAME:
+% 'all' All tests.
+% 'apm' The standard APM test set (default).
+% 'apmm' The mobile APM test set.
+% 'aec' The AEC test set.
+% 'aecm' The AECM test set.
+% 'agc' The AGC test set.
+% 'ns' The NS test set.
+% 'vad' The VAD test set.
+%
+% FILEPATH specifies the path to the test data files.
+%
+% CASENUMBER can be used to select a single test case. Omit CASENUMBER,
+% or set to zero, to use all test cases.
+%
+
+if nargin < 5 || isempty(legacy)
+ % Set to true to run old VQE recordings.
+ legacy = false;
+end
+
+if nargin < 4 || isempty(casenumber)
+ casenumber = 0;
+end
+
+if nargin < 3 || isempty(filepath)
+ filepath = 'data/';
+end
+
+if nargin < 2 || isempty(testname)
+ testname = 'all';
+end
+
+if nargin < 1 || isempty(task)
+ task = 'test';
+end
+
+if ~strcmp(task, 'test') && ~strcmp(task, 'list') && ~strcmp(task, 'show')
+ error(['TASK ' task ' is not recognized']);
+end
+
+if casenumber == 0 && strcmp(task, 'show')
+ error(['CASENUMBER must be specified for TASK ' task]);
+end
+
+inpath = [filepath 'input/'];
+outpath = [filepath 'output/'];
+refpath = [filepath 'reference/'];
+
+if strcmp(testname, 'all')
+ tests = {'apm','apmm','aec','aecm','agc','ns','vad'};
+else
+ tests = {testname};
+end
+
+if legacy
+ progname = './test';
+else
+ progname = './process_test';
+end
+
+global farFile;
+global nearFile;
+global eventFile;
+global delayFile;
+global driftFile;
+
+if legacy
+ farFile = 'vqeFar.pcm';
+ nearFile = 'vqeNear.pcm';
+ eventFile = 'vqeEvent.dat';
+ delayFile = 'vqeBuf.dat';
+ driftFile = 'vqeDrift.dat';
+else
+ farFile = 'apm_far.pcm';
+ nearFile = 'apm_near.pcm';
+ eventFile = 'apm_event.dat';
+ delayFile = 'apm_delay.dat';
+ driftFile = 'apm_drift.dat';
+end
+
+simulateMode = false;
+nErr = 0;
+nCases = 0;
+for i=1:length(tests)
+ simulateMode = false;
+
+ if strcmp(tests{i}, 'apm')
+ testdir = ['apm/'];
+ outfile = ['out'];
+ if legacy
+ opt = ['-ec 1 -agc 2 -nc 2 -vad 3'];
+ else
+ opt = ['--no_progress -hpf' ...
+ ' -aec --drift_compensation -agc --fixed_digital' ...
+ ' -ns --ns_moderate -vad'];
+ end
+
+ elseif strcmp(tests{i}, 'apm-swb')
+ simulateMode = true;
+ testdir = ['apm-swb/'];
+ outfile = ['out'];
+ if legacy
+ opt = ['-fs 32000 -ec 1 -agc 2 -nc 2'];
+ else
+ opt = ['--no_progress -fs 32000 -hpf' ...
+ ' -aec --drift_compensation -agc --adaptive_digital' ...
+ ' -ns --ns_moderate -vad'];
+ end
+ elseif strcmp(tests{i}, 'apmm')
+ testdir = ['apmm/'];
+ outfile = ['out'];
+ opt = ['-aec --drift_compensation -agc --fixed_digital -hpf -ns ' ...
+ '--ns_moderate'];
+
+ else
+ error(['TESTNAME ' tests{i} ' is not recognized']);
+ end
+
+ inpathtest = [inpath testdir];
+ outpathtest = [outpath testdir];
+ refpathtest = [refpath testdir];
+
+ if ~exist(inpathtest,'dir')
+ error(['Input directory ' inpathtest ' does not exist']);
+ end
+
+ if ~exist(refpathtest,'dir')
+ warning(['Reference directory ' refpathtest ' does not exist']);
+ end
+
+ [status, errMsg] = mkdir(outpathtest);
+ if (status == 0)
+ error(errMsg);
+ end
+
+ [nErr, nCases] = recurseDir(inpathtest, outpathtest, refpathtest, outfile, ...
+ progname, opt, simulateMode, nErr, nCases, task, casenumber, legacy);
+
+ if strcmp(task, 'test') || strcmp(task, 'show')
+ system(['rm ' farFile]);
+ system(['rm ' nearFile]);
+ if simulateMode == false
+ system(['rm ' eventFile]);
+ system(['rm ' delayFile]);
+ system(['rm ' driftFile]);
+ end
+ end
+end
+
+if ~strcmp(task, 'list')
+ if nErr == 0
+ fprintf(1, '\nAll files are bit-exact to reference\n', nErr);
+ else
+ fprintf(1, '\n%d files are NOT bit-exact to reference\n', nErr);
+ end
+end
+
+
+function [nErrOut, nCases] = recurseDir(inpath, outpath, refpath, ...
+ outfile, progname, opt, simulateMode, nErr, nCases, task, casenumber, ...
+ legacy)
+
+global farFile;
+global nearFile;
+global eventFile;
+global delayFile;
+global driftFile;
+
+dirs = dir(inpath);
+nDirs = 0;
+nErrOut = nErr;
+for i=3:length(dirs) % skip . and ..
+ nDirs = nDirs + dirs(i).isdir;
+end
+
+
+if nDirs == 0
+ nCases = nCases + 1;
+
+ if casenumber == nCases || casenumber == 0
+
+ if strcmp(task, 'list')
+ fprintf([num2str(nCases) '. ' outfile '\n'])
+ else
+ vadoutfile = ['vad_' outfile '.dat'];
+ outfile = [outfile '.pcm'];
+
+ % Check for VAD test
+ vadTest = 0;
+ if ~isempty(findstr(opt, '-vad'))
+ vadTest = 1;
+ if legacy
+ opt = [opt ' ' outpath vadoutfile];
+ else
+ opt = [opt ' --vad_out_file ' outpath vadoutfile];
+ end
+ end
+
+ if exist([inpath 'vqeFar.pcm'])
+ system(['ln -s -f ' inpath 'vqeFar.pcm ' farFile]);
+ elseif exist([inpath 'apm_far.pcm'])
+ system(['ln -s -f ' inpath 'apm_far.pcm ' farFile]);
+ end
+
+ if exist([inpath 'vqeNear.pcm'])
+ system(['ln -s -f ' inpath 'vqeNear.pcm ' nearFile]);
+ elseif exist([inpath 'apm_near.pcm'])
+ system(['ln -s -f ' inpath 'apm_near.pcm ' nearFile]);
+ end
+
+ if exist([inpath 'vqeEvent.dat'])
+ system(['ln -s -f ' inpath 'vqeEvent.dat ' eventFile]);
+ elseif exist([inpath 'apm_event.dat'])
+ system(['ln -s -f ' inpath 'apm_event.dat ' eventFile]);
+ end
+
+ if exist([inpath 'vqeBuf.dat'])
+ system(['ln -s -f ' inpath 'vqeBuf.dat ' delayFile]);
+ elseif exist([inpath 'apm_delay.dat'])
+ system(['ln -s -f ' inpath 'apm_delay.dat ' delayFile]);
+ end
+
+ if exist([inpath 'vqeSkew.dat'])
+ system(['ln -s -f ' inpath 'vqeSkew.dat ' driftFile]);
+ elseif exist([inpath 'vqeDrift.dat'])
+ system(['ln -s -f ' inpath 'vqeDrift.dat ' driftFile]);
+ elseif exist([inpath 'apm_drift.dat'])
+ system(['ln -s -f ' inpath 'apm_drift.dat ' driftFile]);
+ end
+
+ if simulateMode == false
+ command = [progname ' -o ' outpath outfile ' ' opt];
+ else
+ if legacy
+ inputCmd = [' -in ' nearFile];
+ else
+ inputCmd = [' -i ' nearFile];
+ end
+
+ if exist([farFile])
+ if legacy
+ inputCmd = [' -if ' farFile inputCmd];
+ else
+ inputCmd = [' -ir ' farFile inputCmd];
+ end
+ end
+ command = [progname inputCmd ' -o ' outpath outfile ' ' opt];
+ end
+ % This prevents MATLAB from using its own C libraries.
+ shellcmd = ['bash -c "unset LD_LIBRARY_PATH;'];
+ fprintf([command '\n']);
+ [status, result] = system([shellcmd command '"']);
+ fprintf(result);
+
+ fprintf(['Reference file: ' refpath outfile '\n']);
+
+ if vadTest == 1
+ equal_to_ref = are_files_equal([outpath vadoutfile], ...
+ [refpath vadoutfile], ...
+ 'int8');
+ if ~equal_to_ref
+ nErr = nErr + 1;
+ end
+ end
+
+ [equal_to_ref, diffvector] = are_files_equal([outpath outfile], ...
+ [refpath outfile], ...
+ 'int16');
+ if ~equal_to_ref
+ nErr = nErr + 1;
+ end
+
+ if strcmp(task, 'show')
+ % Assume the last init gives the sample rate of interest.
+ str_idx = strfind(result, 'Sample rate:');
+ fs = str2num(result(str_idx(end) + 13:str_idx(end) + 17));
+ fprintf('Using %d Hz\n', fs);
+
+ if exist([farFile])
+ spclab(fs, farFile, nearFile, [refpath outfile], ...
+ [outpath outfile], diffvector);
+ %spclab(fs, diffvector);
+ else
+ spclab(fs, nearFile, [refpath outfile], [outpath outfile], ...
+ diffvector);
+ %spclab(fs, diffvector);
+ end
+ end
+ end
+ end
+else
+
+ for i=3:length(dirs)
+ if dirs(i).isdir
+ [nErr, nCases] = recurseDir([inpath dirs(i).name '/'], outpath, ...
+ refpath,[outfile '_' dirs(i).name], progname, opt, ...
+ simulateMode, nErr, nCases, task, casenumber, legacy);
+ end
+ end
+end
+nErrOut = nErr;
+
+function [are_equal, diffvector] = ...
+ are_files_equal(newfile, reffile, precision, diffvector)
+
+are_equal = false;
+diffvector = 0;
+if ~exist(newfile,'file')
+ warning(['Output file ' newfile ' does not exist']);
+ return
+end
+
+if ~exist(reffile,'file')
+ warning(['Reference file ' reffile ' does not exist']);
+ return
+end
+
+fid = fopen(newfile,'rb');
+new = fread(fid,inf,precision);
+fclose(fid);
+
+fid = fopen(reffile,'rb');
+ref = fread(fid,inf,precision);
+fclose(fid);
+
+if length(new) ~= length(ref)
+ warning('Reference is not the same length as output');
+ minlength = min(length(new), length(ref));
+ new = new(1:minlength);
+ ref = ref(1:minlength);
+end
+diffvector = new - ref;
+
+if isequal(new, ref)
+ fprintf([newfile ' is bit-exact to reference\n']);
+ are_equal = true;
+else
+ if isempty(new)
+ warning([newfile ' is empty']);
+ return
+ end
+ snr = snrseg(new,ref,80);
+ fprintf('\n');
+ are_equal = false;
+end
diff --git a/trunk/src/modules/audio_processing/test/process_test.cc b/trunk/src/modules/audio_processing/test/process_test.cc
new file mode 100644
index 0000000..2023ddb
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/process_test.cc
@@ -0,0 +1,964 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <sys/stat.h>
+#endif
+
+#include "gtest/gtest.h"
+
+#include "audio_processing.h"
+#include "cpu_features_wrapper.h"
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "tick_util.h"
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+
+using webrtc::AudioFrame;
+using webrtc::AudioProcessing;
+using webrtc::EchoCancellation;
+using webrtc::GainControl;
+using webrtc::NoiseSuppression;
+using webrtc::scoped_array;
+using webrtc::TickInterval;
+using webrtc::TickTime;
+
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace {
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file,
+ ::google::protobuf::MessageLite* msg) {
+ // The "wire format" for the size is little-endian.
+ // Assume process_test is running on a little-endian machine.
+ int32_t size = 0;
+ if (fread(&size, sizeof(int32_t), 1, file) != 1) {
+ return false;
+ }
+ if (size <= 0) {
+ return false;
+ }
+ const size_t usize = static_cast<size_t>(size);
+
+ scoped_array<char> array(new char[usize]);
+ if (fread(array.get(), sizeof(char), usize, file) != usize) {
+ return false;
+ }
+
+ msg->Clear();
+ return msg->ParseFromArray(array.get(), usize);
+}
+
+void PrintStat(const AudioProcessing::Statistic& stat) {
+ printf("%d, %d, %d\n", stat.average,
+ stat.maximum,
+ stat.minimum);
+}
+
+void usage() {
+ printf(
+ "Usage: process_test [options] [-pb PROTOBUF_FILE]\n"
+ " [-ir REVERSE_FILE] [-i PRIMARY_FILE] [-o OUT_FILE]\n");
+ printf(
+ "process_test is a test application for AudioProcessing.\n\n"
+ "When a protobuf debug file is available, specify it with -pb.\n"
+ "Alternately, when -ir or -i is used, the specified files will be\n"
+ "processed directly in a simulation mode. Otherwise the full set of\n"
+ "legacy test files is expected to be present in the working directory.\n");
+ printf("\n");
+ printf("Options\n");
+ printf("General configuration (only used for the simulation mode):\n");
+ printf(" -fs SAMPLE_RATE_HZ\n");
+ printf(" -ch CHANNELS_IN CHANNELS_OUT\n");
+ printf(" -rch REVERSE_CHANNELS\n");
+ printf("\n");
+ printf("Component configuration:\n");
+ printf(
+ "All components are disabled by default. Each block below begins with a\n"
+ "flag to enable the component with default settings. The subsequent flags\n"
+ "in the block are used to provide configuration settings.\n");
+ printf("\n -aec Echo cancellation\n");
+ printf(" --drift_compensation\n");
+ printf(" --no_drift_compensation\n");
+ printf(" --no_echo_metrics\n");
+ printf(" --no_delay_logging\n");
+ printf("\n -aecm Echo control mobile\n");
+ printf(" --aecm_echo_path_in_file FILE\n");
+ printf(" --aecm_echo_path_out_file FILE\n");
+ printf("\n -agc Gain control\n");
+ printf(" --analog\n");
+ printf(" --adaptive_digital\n");
+ printf(" --fixed_digital\n");
+ printf(" --target_level LEVEL\n");
+ printf(" --compression_gain GAIN\n");
+ printf(" --limiter\n");
+ printf(" --no_limiter\n");
+ printf("\n -hpf High pass filter\n");
+ printf("\n -ns Noise suppression\n");
+ printf(" --ns_low\n");
+ printf(" --ns_moderate\n");
+ printf(" --ns_high\n");
+ printf(" --ns_very_high\n");
+ printf("\n -vad Voice activity detection\n");
+ printf(" --vad_out_file FILE\n");
+ printf("\n Level metrics (enabled by default)\n");
+ printf(" --no_level_metrics\n");
+ printf("\n");
+ printf("Modifiers:\n");
+ printf(" --noasm Disable SSE optimization.\n");
+ printf(" --delay DELAY Add DELAY ms to input value.\n");
+ printf(" --perf Measure performance.\n");
+ printf(" --quiet Suppress text output.\n");
+ printf(" --no_progress Suppress progress.\n");
+ printf(" --debug_file FILE Dump a debug recording.\n");
+}
+
+// void function for gtest.
+void void_main(int argc, char* argv[]) {
+ if (argc > 1 && strcmp(argv[1], "--help") == 0) {
+ usage();
+ return;
+ }
+
+ if (argc < 2) {
+ printf("Did you mean to run without arguments?\n");
+ printf("Try `process_test --help' for more information.\n\n");
+ }
+
+ AudioProcessing* apm = AudioProcessing::Create(0);
+ ASSERT_TRUE(apm != NULL);
+
+ const char* pb_filename = NULL;
+ const char* far_filename = NULL;
+ const char* near_filename = NULL;
+ const char* out_filename = NULL;
+ const char* vad_out_filename = NULL;
+ const char* aecm_echo_path_in_filename = NULL;
+ const char* aecm_echo_path_out_filename = NULL;
+
+ int32_t sample_rate_hz = 16000;
+ int32_t device_sample_rate_hz = 16000;
+
+ int num_capture_input_channels = 1;
+ int num_capture_output_channels = 1;
+ int num_render_channels = 1;
+
+ int samples_per_channel = sample_rate_hz / 100;
+
+ bool simulating = false;
+ bool perf_testing = false;
+ bool verbose = true;
+ bool progress = true;
+ int extra_delay_ms = 0;
+ //bool interleaved = true;
+
+ ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+ for (int i = 1; i < argc; i++) {
+ if (strcmp(argv[i], "-pb") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify protobuf filename after -pb";
+ pb_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-ir") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -ir";
+ far_filename = argv[i];
+ simulating = true;
+
+ } else if (strcmp(argv[i], "-i") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -i";
+ near_filename = argv[i];
+ simulating = true;
+
+ } else if (strcmp(argv[i], "-o") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -o";
+ out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-fs") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify sample rate after -fs";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
+ samples_per_channel = sample_rate_hz / 100;
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(sample_rate_hz));
+
+ } else if (strcmp(argv[i], "-ch") == 0) {
+ i++;
+ ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
+ i++;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_channels(num_capture_input_channels,
+ num_capture_output_channels));
+
+ } else if (strcmp(argv[i], "-rch") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify number of channels after -rch";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_reverse_channels(num_render_channels));
+
+ } else if (strcmp(argv[i], "-aec") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_metrics(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_delay_logging(true));
+
+ } else if (strcmp(argv[i], "--drift_compensation") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ // TODO(ajm): this is enabled in the VQE test app by default. Investigate
+ // why it can give better performance despite passing zeros.
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_drift_compensation(true));
+ } else if (strcmp(argv[i], "--no_drift_compensation") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_drift_compensation(false));
+
+ } else if (strcmp(argv[i], "--no_echo_metrics") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_metrics(false));
+
+ } else if (strcmp(argv[i], "--no_delay_logging") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_delay_logging(false));
+
+ } else if (strcmp(argv[i], "--no_level_metrics") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
+
+ } else if (strcmp(argv[i], "-aecm") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
+
+ } else if (strcmp(argv[i], "--aecm_echo_path_in_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_in_file";
+ aecm_echo_path_in_filename = argv[i];
+
+ } else if (strcmp(argv[i], "--aecm_echo_path_out_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_out_file";
+ aecm_echo_path_out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-agc") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+
+ } else if (strcmp(argv[i], "--analog") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
+
+ } else if (strcmp(argv[i], "--adaptive_digital") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+
+ } else if (strcmp(argv[i], "--fixed_digital") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kFixedDigital));
+
+ } else if (strcmp(argv[i], "--target_level") == 0) {
+ i++;
+ int level;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &level));
+
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_target_level_dbfs(level));
+
+ } else if (strcmp(argv[i], "--compression_gain") == 0) {
+ i++;
+ int gain;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &gain));
+
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_compression_gain_db(gain));
+
+ } else if (strcmp(argv[i], "--limiter") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->enable_limiter(true));
+
+ } else if (strcmp(argv[i], "--no_limiter") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->enable_limiter(false));
+
+ } else if (strcmp(argv[i], "-hpf") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->high_pass_filter()->Enable(true));
+
+ } else if (strcmp(argv[i], "-ns") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+
+ } else if (strcmp(argv[i], "--ns_low") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kLow));
+
+ } else if (strcmp(argv[i], "--ns_moderate") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kModerate));
+
+ } else if (strcmp(argv[i], "--ns_high") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kHigh));
+
+ } else if (strcmp(argv[i], "--ns_very_high") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kVeryHigh));
+
+ } else if (strcmp(argv[i], "-vad") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+
+ } else if (strcmp(argv[i], "--vad_out_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --vad_out_file";
+ vad_out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "--noasm") == 0) {
+ WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
+ // We need to reinitialize here if components have already been enabled.
+ ASSERT_EQ(apm->kNoError, apm->Initialize());
+
+ } else if (strcmp(argv[i], "--delay") == 0) {
+ i++;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &extra_delay_ms));
+
+ } else if (strcmp(argv[i], "--perf") == 0) {
+ perf_testing = true;
+
+ } else if (strcmp(argv[i], "--quiet") == 0) {
+ verbose = false;
+ progress = false;
+
+ } else if (strcmp(argv[i], "--no_progress") == 0) {
+ progress = false;
+
+ } else if (strcmp(argv[i], "--debug_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --debug_file";
+ ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i]));
+ } else {
+ FAIL() << "Unrecognized argument " << argv[i];
+ }
+ }
+ // If we're reading a protobuf file, ensure a simulation hasn't also
+ // been requested (which makes no sense...)
+ ASSERT_FALSE(pb_filename && simulating);
+
+ if (verbose) {
+ printf("Sample rate: %d Hz\n", sample_rate_hz);
+ printf("Primary channels: %d (in), %d (out)\n",
+ num_capture_input_channels,
+ num_capture_output_channels);
+ printf("Reverse channels: %d \n", num_render_channels);
+ }
+
+ const char far_file_default[] = "apm_far.pcm";
+ const char near_file_default[] = "apm_near.pcm";
+ const char out_file_default[] = "out.pcm";
+ const char event_filename[] = "apm_event.dat";
+ const char delay_filename[] = "apm_delay.dat";
+ const char drift_filename[] = "apm_drift.dat";
+ const char vad_file_default[] = "vad_out.dat";
+
+ if (!simulating) {
+ far_filename = far_file_default;
+ near_filename = near_file_default;
+ }
+
+ if (!out_filename) {
+ out_filename = out_file_default;
+ }
+
+ if (!vad_out_filename) {
+ vad_out_filename = vad_file_default;
+ }
+
+ FILE* pb_file = NULL;
+ FILE* far_file = NULL;
+ FILE* near_file = NULL;
+ FILE* out_file = NULL;
+ FILE* event_file = NULL;
+ FILE* delay_file = NULL;
+ FILE* drift_file = NULL;
+ FILE* vad_out_file = NULL;
+ FILE* aecm_echo_path_in_file = NULL;
+ FILE* aecm_echo_path_out_file = NULL;
+
+ if (pb_filename) {
+ pb_file = fopen(pb_filename, "rb");
+ ASSERT_TRUE(NULL != pb_file) << "Unable to open protobuf file "
+ << pb_filename;
+ } else {
+ if (far_filename) {
+ far_file = fopen(far_filename, "rb");
+ ASSERT_TRUE(NULL != far_file) << "Unable to open far-end audio file "
+ << far_filename;
+ }
+
+ near_file = fopen(near_filename, "rb");
+ ASSERT_TRUE(NULL != near_file) << "Unable to open near-end audio file "
+ << near_filename;
+ if (!simulating) {
+ event_file = fopen(event_filename, "rb");
+ ASSERT_TRUE(NULL != event_file) << "Unable to open event file "
+ << event_filename;
+
+ delay_file = fopen(delay_filename, "rb");
+ ASSERT_TRUE(NULL != delay_file) << "Unable to open buffer file "
+ << delay_filename;
+
+ drift_file = fopen(drift_filename, "rb");
+ ASSERT_TRUE(NULL != drift_file) << "Unable to open drift file "
+ << drift_filename;
+ }
+ }
+
+ out_file = fopen(out_filename, "wb");
+ ASSERT_TRUE(NULL != out_file) << "Unable to open output audio file "
+ << out_filename;
+
+ int near_size_bytes = 0;
+ if (pb_file) {
+ struct stat st;
+ stat(pb_filename, &st);
+ // Crude estimate, but should be good enough.
+ near_size_bytes = st.st_size / 3;
+ } else {
+ struct stat st;
+ stat(near_filename, &st);
+ near_size_bytes = st.st_size;
+ }
+
+ if (apm->voice_detection()->is_enabled()) {
+ vad_out_file = fopen(vad_out_filename, "wb");
+ ASSERT_TRUE(NULL != vad_out_file) << "Unable to open VAD output file "
+ << vad_out_file;
+ }
+
+ if (aecm_echo_path_in_filename != NULL) {
+ aecm_echo_path_in_file = fopen(aecm_echo_path_in_filename, "rb");
+ ASSERT_TRUE(NULL != aecm_echo_path_in_file) << "Unable to open file "
+ << aecm_echo_path_in_filename;
+
+ const size_t path_size =
+ apm->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path(new char[path_size]);
+ ASSERT_EQ(path_size, fread(echo_path.get(),
+ sizeof(char),
+ path_size,
+ aecm_echo_path_in_file));
+ EXPECT_EQ(apm->kNoError,
+ apm->echo_control_mobile()->SetEchoPath(echo_path.get(),
+ path_size));
+ fclose(aecm_echo_path_in_file);
+ aecm_echo_path_in_file = NULL;
+ }
+
+ if (aecm_echo_path_out_filename != NULL) {
+ aecm_echo_path_out_file = fopen(aecm_echo_path_out_filename, "wb");
+ ASSERT_TRUE(NULL != aecm_echo_path_out_file) << "Unable to open file "
+ << aecm_echo_path_out_filename;
+ }
+
+ size_t read_count = 0;
+ int reverse_count = 0;
+ int primary_count = 0;
+ int near_read_bytes = 0;
+ TickInterval acc_ticks;
+
+ AudioFrame far_frame;
+ AudioFrame near_frame;
+
+ int delay_ms = 0;
+ int drift_samples = 0;
+ int capture_level = 127;
+ int8_t stream_has_voice = 0;
+
+ TickTime t0 = TickTime::Now();
+ TickTime t1 = t0;
+ WebRtc_Word64 max_time_us = 0;
+ WebRtc_Word64 max_time_reverse_us = 0;
+ WebRtc_Word64 min_time_us = 1e6;
+ WebRtc_Word64 min_time_reverse_us = 1e6;
+
+ // TODO(ajm): Ideally we would refactor this block into separate functions,
+ // but for now we want to share the variables.
+ if (pb_file) {
+ Event event_msg;
+ while (ReadMessageFromFile(pb_file, &event_msg)) {
+ std::ostringstream trace_stream;
+ trace_stream << "Processed frames: " << reverse_count << " (reverse), "
+ << primary_count << " (primary)";
+ SCOPED_TRACE(trace_stream.str());
+
+ if (event_msg.type() == Event::INIT) {
+ ASSERT_TRUE(event_msg.has_init());
+ const Init msg = event_msg.init();
+
+ ASSERT_TRUE(msg.has_sample_rate());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(msg.sample_rate()));
+
+ ASSERT_TRUE(msg.has_device_sample_rate());
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_device_sample_rate_hz(
+ msg.device_sample_rate()));
+
+ ASSERT_TRUE(msg.has_num_input_channels());
+ ASSERT_TRUE(msg.has_num_output_channels());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_channels(msg.num_input_channels(),
+ msg.num_output_channels()));
+
+ ASSERT_TRUE(msg.has_num_reverse_channels());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_reverse_channels(msg.num_reverse_channels()));
+
+ samples_per_channel = msg.sample_rate() / 100;
+ far_frame._frequencyInHz = msg.sample_rate();
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = msg.num_reverse_channels();
+ near_frame._frequencyInHz = msg.sample_rate();
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (verbose) {
+ printf("Init at frame: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+ printf(" Sample rate: %d Hz\n", msg.sample_rate());
+ printf(" Primary channels: %d (in), %d (out)\n",
+ msg.num_input_channels(),
+ msg.num_output_channels());
+ printf(" Reverse channels: %d \n", msg.num_reverse_channels());
+ }
+
+ } else if (event_msg.type() == Event::REVERSE_STREAM) {
+ ASSERT_TRUE(event_msg.has_reverse_stream());
+ const ReverseStream msg = event_msg.reverse_stream();
+ reverse_count++;
+
+ ASSERT_TRUE(msg.has_data());
+ ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
+ far_frame._audioChannel, msg.data().size());
+ memcpy(far_frame._payloadData, msg.data().data(), msg.data().size());
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->AnalyzeReverseStream(&far_frame));
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_reverse_us) {
+ max_time_reverse_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_reverse_us) {
+ min_time_reverse_us = tick_diff.Microseconds();
+ }
+ }
+
+ } else if (event_msg.type() == Event::STREAM) {
+ ASSERT_TRUE(event_msg.has_stream());
+ const Stream msg = event_msg.stream();
+ primary_count++;
+
+ // ProcessStream could have changed this for the output frame.
+ near_frame._audioChannel = apm->num_input_channels();
+
+ ASSERT_TRUE(msg.has_input_data());
+ ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
+ near_frame._audioChannel, msg.input_data().size());
+ memcpy(near_frame._payloadData,
+ msg.input_data().data(),
+ msg.input_data().size());
+
+ near_read_bytes += msg.input_data().size();
+ if (progress && primary_count % 100 == 0) {
+ printf("%.0f%% complete\r",
+ (near_read_bytes * 100.0) / near_size_bytes);
+ fflush(stdout);
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_stream_analog_level(msg.level()));
+ ASSERT_EQ(apm->kNoError,
+ apm->set_stream_delay_ms(msg.delay() + extra_delay_ms));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_stream_drift_samples(msg.drift()));
+
+ int err = apm->ProcessStream(&near_frame);
+ if (err == apm->kBadStreamParameterWarning) {
+ printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
+ }
+ ASSERT_TRUE(err == apm->kNoError ||
+ err == apm->kBadStreamParameterWarning);
+ ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
+
+ capture_level = apm->gain_control()->stream_analog_level();
+
+ stream_has_voice =
+ static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
+ if (vad_out_file != NULL) {
+ ASSERT_EQ(1u, fwrite(&stream_has_voice,
+ sizeof(stream_has_voice),
+ 1,
+ vad_out_file));
+ }
+
+ if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
+ ASSERT_EQ(msg.level(), capture_level);
+ }
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_us) {
+ max_time_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_us) {
+ min_time_us = tick_diff.Microseconds();
+ }
+ }
+
+ size_t size = samples_per_channel * near_frame._audioChannel;
+ ASSERT_EQ(size, fwrite(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ out_file));
+ }
+ }
+
+ ASSERT_TRUE(feof(pb_file));
+
+ } else {
+ enum Events {
+ kInitializeEvent,
+ kRenderEvent,
+ kCaptureEvent,
+ kResetEventDeprecated
+ };
+ int16_t event = 0;
+ while (simulating || feof(event_file) == 0) {
+ std::ostringstream trace_stream;
+ trace_stream << "Processed frames: " << reverse_count << " (reverse), "
+ << primary_count << " (primary)";
+ SCOPED_TRACE(trace_stream.str());
+
+ if (simulating) {
+ if (far_file == NULL) {
+ event = kCaptureEvent;
+ } else {
+ if (event == kRenderEvent) {
+ event = kCaptureEvent;
+ } else {
+ event = kRenderEvent;
+ }
+ }
+ } else {
+ read_count = fread(&event, sizeof(event), 1, event_file);
+ if (read_count != 1) {
+ break;
+ }
+ }
+
+ far_frame._frequencyInHz = sample_rate_hz;
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = num_render_channels;
+ near_frame._frequencyInHz = sample_rate_hz;
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (event == kInitializeEvent || event == kResetEventDeprecated) {
+ ASSERT_EQ(1u,
+ fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file));
+ samples_per_channel = sample_rate_hz / 100;
+
+ ASSERT_EQ(1u,
+ fread(&device_sample_rate_hz,
+ sizeof(device_sample_rate_hz),
+ 1,
+ event_file));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(sample_rate_hz));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_device_sample_rate_hz(
+ device_sample_rate_hz));
+
+ far_frame._frequencyInHz = sample_rate_hz;
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = num_render_channels;
+ near_frame._frequencyInHz = sample_rate_hz;
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (verbose) {
+ printf("Init at frame: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+ printf(" Sample rate: %d Hz\n", sample_rate_hz);
+ }
+
+ } else if (event == kRenderEvent) {
+ reverse_count++;
+
+ size_t size = samples_per_channel * num_render_channels;
+ read_count = fread(far_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ far_file);
+
+ if (simulating) {
+ if (read_count != size) {
+ // Read an equal amount from the near file to avoid errors due to
+ // not reaching end-of-file.
+ EXPECT_EQ(0, fseek(near_file, read_count * sizeof(int16_t),
+ SEEK_CUR));
+ break; // This is expected.
+ }
+ } else {
+ ASSERT_EQ(size, read_count);
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->AnalyzeReverseStream(&far_frame));
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_reverse_us) {
+ max_time_reverse_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_reverse_us) {
+ min_time_reverse_us = tick_diff.Microseconds();
+ }
+ }
+
+ } else if (event == kCaptureEvent) {
+ primary_count++;
+ near_frame._audioChannel = num_capture_input_channels;
+
+ size_t size = samples_per_channel * num_capture_input_channels;
+ read_count = fread(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ near_file);
+
+ near_read_bytes += read_count * sizeof(int16_t);
+ if (progress && primary_count % 100 == 0) {
+ printf("%.0f%% complete\r",
+ (near_read_bytes * 100.0) / near_size_bytes);
+ fflush(stdout);
+ }
+ if (simulating) {
+ if (read_count != size) {
+ break; // This is expected.
+ }
+
+ delay_ms = 0;
+ drift_samples = 0;
+ } else {
+ ASSERT_EQ(size, read_count);
+
+ // TODO(ajm): sizeof(delay_ms) for current files?
+ ASSERT_EQ(1u,
+ fread(&delay_ms, 2, 1, delay_file));
+ ASSERT_EQ(1u,
+ fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ // TODO(ajm): fake an analog gain while simulating.
+
+ int capture_level_in = capture_level;
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_stream_analog_level(capture_level));
+ ASSERT_EQ(apm->kNoError,
+ apm->set_stream_delay_ms(delay_ms + extra_delay_ms));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
+
+ int err = apm->ProcessStream(&near_frame);
+ if (err == apm->kBadStreamParameterWarning) {
+ printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
+ }
+ ASSERT_TRUE(err == apm->kNoError ||
+ err == apm->kBadStreamParameterWarning);
+ ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
+
+ capture_level = apm->gain_control()->stream_analog_level();
+
+ stream_has_voice =
+ static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
+ if (vad_out_file != NULL) {
+ ASSERT_EQ(1u, fwrite(&stream_has_voice,
+ sizeof(stream_has_voice),
+ 1,
+ vad_out_file));
+ }
+
+ if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
+ ASSERT_EQ(capture_level_in, capture_level);
+ }
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_us) {
+ max_time_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_us) {
+ min_time_us = tick_diff.Microseconds();
+ }
+ }
+
+ size = samples_per_channel * near_frame._audioChannel;
+ ASSERT_EQ(size, fwrite(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ out_file));
+ }
+ else {
+ FAIL() << "Event " << event << " is unrecognized";
+ }
+ }
+ }
+ printf("100%% complete\r");
+
+ if (aecm_echo_path_out_file != NULL) {
+ const size_t path_size =
+ apm->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path(new char[path_size]);
+ apm->echo_control_mobile()->GetEchoPath(echo_path.get(), path_size);
+ ASSERT_EQ(path_size, fwrite(echo_path.get(),
+ sizeof(char),
+ path_size,
+ aecm_echo_path_out_file));
+ fclose(aecm_echo_path_out_file);
+ aecm_echo_path_out_file = NULL;
+ }
+
+ if (verbose) {
+ printf("\nProcessed frames: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+
+ if (apm->level_estimator()->is_enabled()) {
+ printf("\n--Level metrics--\n");
+ printf("RMS: %d dBFS\n", -apm->level_estimator()->RMS());
+ }
+ if (apm->echo_cancellation()->are_metrics_enabled()) {
+ EchoCancellation::Metrics metrics;
+ apm->echo_cancellation()->GetMetrics(&metrics);
+ printf("\n--Echo metrics--\n");
+ printf("(avg, max, min)\n");
+ printf("ERL: ");
+ PrintStat(metrics.echo_return_loss);
+ printf("ERLE: ");
+ PrintStat(metrics.echo_return_loss_enhancement);
+ printf("ANLP: ");
+ PrintStat(metrics.a_nlp);
+ }
+ if (apm->echo_cancellation()->is_delay_logging_enabled()) {
+ int median = 0;
+ int std = 0;
+ apm->echo_cancellation()->GetDelayMetrics(&median, &std);
+ printf("\n--Delay metrics--\n");
+ printf("Median: %3d\n", median);
+ printf("Standard deviation: %3d\n", std);
+ }
+ }
+
+ if (!pb_file) {
+ int8_t temp_int8;
+ if (far_file) {
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, far_file);
+ EXPECT_NE(0, feof(far_file)) << "Far-end file not fully processed";
+ }
+
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, near_file);
+ EXPECT_NE(0, feof(near_file)) << "Near-end file not fully processed";
+
+ if (!simulating) {
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, event_file);
+ EXPECT_NE(0, feof(event_file)) << "Event file not fully processed";
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, delay_file);
+ EXPECT_NE(0, feof(delay_file)) << "Delay file not fully processed";
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, drift_file);
+ EXPECT_NE(0, feof(drift_file)) << "Drift file not fully processed";
+ }
+ }
+
+ if (perf_testing) {
+ if (primary_count > 0) {
+ WebRtc_Word64 exec_time = acc_ticks.Milliseconds();
+ printf("\nTotal time: %.3f s, file time: %.2f s\n",
+ exec_time * 0.001, primary_count * 0.01);
+ printf("Time per frame: %.3f ms (average), %.3f ms (max),"
+ " %.3f ms (min)\n",
+ (exec_time * 1.0) / primary_count,
+ (max_time_us + max_time_reverse_us) / 1000.0,
+ (min_time_us + min_time_reverse_us) / 1000.0);
+ } else {
+ printf("Warning: no capture frames\n");
+ }
+ }
+
+ AudioProcessing::Destroy(apm);
+ apm = NULL;
+}
+} // namespace
+
+int main(int argc, char* argv[])
+{
+ void_main(argc, argv);
+
+ // Optional, but removes memory leak noise from Valgrind.
+ google::protobuf::ShutdownProtobufLibrary();
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/test/unit_test.cc b/trunk/src/modules/audio_processing/test/unit_test.cc
new file mode 100644
index 0000000..4fa5d7c
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/unit_test.cc
@@ -0,0 +1,1322 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "gtest/gtest.h"
+
+#include "audio_processing.h"
+#include "event_wrapper.h"
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "signal_processing_library.h"
+#include "testsupport/fileutils.h"
+#include "thread_wrapper.h"
+#include "trace.h"
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/test/unittest.pb.h"
+#else
+#include "webrtc/audio_processing/unittest.pb.h"
+#endif
+
+#if (defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)) || \
+ (defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && !defined(NDEBUG))
+# define WEBRTC_AUDIOPROC_BIT_EXACT
+#endif
+
+using webrtc::AudioProcessing;
+using webrtc::AudioFrame;
+using webrtc::GainControl;
+using webrtc::NoiseSuppression;
+using webrtc::EchoCancellation;
+using webrtc::EventWrapper;
+using webrtc::scoped_array;
+using webrtc::Trace;
+using webrtc::LevelEstimator;
+using webrtc::EchoCancellation;
+using webrtc::EchoControlMobile;
+using webrtc::VoiceDetection;
+
+namespace {
+// When false, this will compare the output data with the results stored to
+// file. This is the typical case. When the file should be updated, it can
+// be set to true with the command-line switch --write_ref_data.
+bool write_ref_data = false;
+
+class ApmTest : public ::testing::Test {
+ protected:
+ ApmTest();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ static void SetUpTestCase() {
+ Trace::CreateTrace();
+ std::string trace_filename = webrtc::test::OutputPath() +
+ "audioproc_trace.txt";
+ ASSERT_EQ(0, Trace::SetTraceFile(trace_filename.c_str()));
+ }
+
+ static void TearDownTestCase() {
+ Trace::ReturnTrace();
+ }
+
+ void Init(int sample_rate_hz, int num_reverse_channels,
+ int num_input_channels, int num_output_channels,
+ bool open_output_file);
+ std::string ResourceFilePath(std::string name, int sample_rate_hz);
+ std::string OutputFilePath(std::string name,
+ int sample_rate_hz,
+ int num_reverse_channels,
+ int num_input_channels,
+ int num_output_channels);
+
+ const std::string output_path_;
+ const std::string ref_path_;
+ const std::string ref_filename_;
+ webrtc::AudioProcessing* apm_;
+ webrtc::AudioFrame* frame_;
+ webrtc::AudioFrame* revframe_;
+ FILE* far_file_;
+ FILE* near_file_;
+ FILE* out_file_;
+};
+
+ApmTest::ApmTest()
+ : output_path_(webrtc::test::OutputPath()),
+ ref_path_(webrtc::test::ProjectRootPath() +
+ "test/data/audio_processing/"),
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+ ref_filename_(ref_path_ + "output_data_fixed.pb"),
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ ref_filename_(ref_path_ + "output_data_float.pb"),
+#endif
+ apm_(NULL),
+ frame_(NULL),
+ revframe_(NULL),
+ far_file_(NULL),
+ near_file_(NULL),
+ out_file_(NULL) {}
+
+void ApmTest::SetUp() {
+ apm_ = AudioProcessing::Create(0);
+ ASSERT_TRUE(apm_ != NULL);
+
+ frame_ = new AudioFrame();
+ revframe_ = new AudioFrame();
+
+ Init(32000, 2, 2, 2, false);
+}
+
+void ApmTest::TearDown() {
+ if (frame_) {
+ delete frame_;
+ }
+ frame_ = NULL;
+
+ if (revframe_) {
+ delete revframe_;
+ }
+ revframe_ = NULL;
+
+ if (far_file_) {
+ ASSERT_EQ(0, fclose(far_file_));
+ }
+ far_file_ = NULL;
+
+ if (near_file_) {
+ ASSERT_EQ(0, fclose(near_file_));
+ }
+ near_file_ = NULL;
+
+ if (out_file_) {
+ ASSERT_EQ(0, fclose(out_file_));
+ }
+ out_file_ = NULL;
+
+ if (apm_ != NULL) {
+ AudioProcessing::Destroy(apm_);
+ }
+ apm_ = NULL;
+}
+
+std::string ApmTest::ResourceFilePath(std::string name, int sample_rate_hz) {
+ std::ostringstream ss;
+ // Resource files are all stereo.
+ ss << name << sample_rate_hz / 1000 << "_stereo";
+ return webrtc::test::ResourcePath(ss.str(), "pcm");
+}
+
+std::string ApmTest::OutputFilePath(std::string name,
+ int sample_rate_hz,
+ int num_reverse_channels,
+ int num_input_channels,
+ int num_output_channels) {
+ std::ostringstream ss;
+ ss << name << sample_rate_hz / 1000 << "_" << num_reverse_channels << "r" <<
+ num_input_channels << "i" << "_";
+ if (num_output_channels == 1) {
+ ss << "mono";
+ } else if (num_output_channels == 2) {
+ ss << "stereo";
+ } else {
+ assert(false);
+ return "";
+ }
+ ss << ".pcm";
+
+ return output_path_ + ss.str();
+}
+
+
+void ApmTest::Init(int sample_rate_hz, int num_reverse_channels,
+ int num_input_channels, int num_output_channels,
+ bool open_output_file) {
+ ASSERT_EQ(apm_->kNoError, apm_->Initialize());
+
+ // Handles error checking of the parameters as well. No need to repeat it.
+ ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(sample_rate_hz));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(num_input_channels,
+ num_output_channels));
+ ASSERT_EQ(apm_->kNoError,
+ apm_->set_num_reverse_channels(num_reverse_channels));
+
+ // We always use 10 ms frames.
+ const int samples_per_channel = sample_rate_hz / 100;
+ frame_->_payloadDataLengthInSamples = samples_per_channel;
+ frame_->_audioChannel = num_input_channels;
+ frame_->_frequencyInHz = sample_rate_hz;
+ revframe_->_payloadDataLengthInSamples = samples_per_channel;
+ revframe_->_audioChannel = num_reverse_channels;
+ revframe_->_frequencyInHz = sample_rate_hz;
+
+ if (far_file_) {
+ ASSERT_EQ(0, fclose(far_file_));
+ }
+ std::string filename = ResourceFilePath("far", sample_rate_hz);
+ far_file_ = fopen(filename.c_str(), "rb");
+ ASSERT_TRUE(far_file_ != NULL) << "Could not open file " <<
+ filename << "\n";
+
+ if (near_file_) {
+ ASSERT_EQ(0, fclose(near_file_));
+ }
+ filename = ResourceFilePath("near", sample_rate_hz);
+ near_file_ = fopen(filename.c_str(), "rb");
+ ASSERT_TRUE(near_file_ != NULL) << "Could not open file " <<
+ filename << "\n";
+
+ if (open_output_file) {
+ if (out_file_) {
+ ASSERT_EQ(0, fclose(out_file_));
+ }
+ filename = OutputFilePath("out", sample_rate_hz, num_reverse_channels,
+ num_input_channels, num_output_channels);
+ out_file_ = fopen(filename.c_str(), "wb");
+ ASSERT_TRUE(out_file_ != NULL) << "Could not open file " <<
+ filename << "\n";
+ }
+}
+
+void MixStereoToMono(const int16_t* stereo,
+ int16_t* mono,
+ int samples_per_channel) {
+ for (int i = 0; i < samples_per_channel; i++) {
+ int32_t int32 = (static_cast<int32_t>(stereo[i * 2]) +
+ static_cast<int32_t>(stereo[i * 2 + 1])) >> 1;
+ mono[i] = static_cast<int16_t>(int32);
+ }
+}
+
+template <class T>
+T MaxValue(T a, T b) {
+ return a > b ? a : b;
+}
+
+template <class T>
+T AbsValue(T a) {
+ return a > 0 ? a : -a;
+}
+
+void SetFrameTo(AudioFrame* frame, int16_t value) {
+ for (int i = 0; i < frame->_payloadDataLengthInSamples * frame->_audioChannel;
+ ++i) {
+ frame->_payloadData[i] = value;
+ }
+}
+
+int16_t MaxAudioFrame(const AudioFrame& frame) {
+ const int length = frame._payloadDataLengthInSamples * frame._audioChannel;
+ int16_t max = AbsValue(frame._payloadData[0]);
+ for (int i = 1; i < length; i++) {
+ max = MaxValue(max, AbsValue(frame._payloadData[i]));
+ }
+
+ return max;
+}
+
+bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+ if (frame1._payloadDataLengthInSamples !=
+ frame2._payloadDataLengthInSamples) {
+ return false;
+ }
+ if (frame1._audioChannel !=
+ frame2._audioChannel) {
+ return false;
+ }
+ if (memcmp(frame1._payloadData, frame2._payloadData,
+ frame1._payloadDataLengthInSamples * frame1._audioChannel *
+ sizeof(int16_t))) {
+ return false;
+ }
+ return true;
+}
+
+void TestStats(const AudioProcessing::Statistic& test,
+ const webrtc::audioproc::Test::Statistic& reference) {
+ EXPECT_EQ(reference.instant(), test.instant);
+ EXPECT_EQ(reference.average(), test.average);
+ EXPECT_EQ(reference.maximum(), test.maximum);
+ EXPECT_EQ(reference.minimum(), test.minimum);
+}
+
+void WriteStatsMessage(const AudioProcessing::Statistic& output,
+ webrtc::audioproc::Test::Statistic* message) {
+ message->set_instant(output.instant);
+ message->set_average(output.average);
+ message->set_maximum(output.maximum);
+ message->set_minimum(output.minimum);
+}
+
+void WriteMessageLiteToFile(const std::string filename,
+ const ::google::protobuf::MessageLite& message) {
+ FILE* file = fopen(filename.c_str(), "wb");
+ ASSERT_TRUE(file != NULL) << "Could not open " << filename;
+ int size = message.ByteSize();
+ ASSERT_GT(size, 0);
+ unsigned char* array = new unsigned char[size];
+ ASSERT_TRUE(message.SerializeToArray(array, size));
+
+ ASSERT_EQ(1u, fwrite(&size, sizeof(int), 1, file));
+ ASSERT_EQ(static_cast<size_t>(size),
+ fwrite(array, sizeof(unsigned char), size, file));
+
+ delete [] array;
+ fclose(file);
+}
+
+void ReadMessageLiteFromFile(const std::string filename,
+ ::google::protobuf::MessageLite* message) {
+ assert(message != NULL);
+
+ FILE* file = fopen(filename.c_str(), "rb");
+ ASSERT_TRUE(file != NULL) << "Could not open " << filename;
+ int size = 0;
+ ASSERT_EQ(1u, fread(&size, sizeof(int), 1, file));
+ ASSERT_GT(size, 0);
+ unsigned char* array = new unsigned char[size];
+ ASSERT_EQ(static_cast<size_t>(size),
+ fread(array, sizeof(unsigned char), size, file));
+
+ ASSERT_TRUE(message->ParseFromArray(array, size));
+
+ delete [] array;
+ fclose(file);
+}
+
+struct ThreadData {
+ ThreadData(int thread_num_, AudioProcessing* ap_)
+ : thread_num(thread_num_),
+ error(false),
+ ap(ap_) {}
+ int thread_num;
+ bool error;
+ AudioProcessing* ap;
+};
+
+// Don't use GTest here; non-thread-safe on Windows (as of 1.5.0).
+bool DeadlockProc(void* thread_object) {
+ ThreadData* thread_data = static_cast<ThreadData*>(thread_object);
+ AudioProcessing* ap = thread_data->ap;
+ int err = ap->kNoError;
+
+ AudioFrame primary_frame;
+ AudioFrame reverse_frame;
+ primary_frame._payloadDataLengthInSamples = 320;
+ primary_frame._audioChannel = 2;
+ primary_frame._frequencyInHz = 32000;
+ reverse_frame._payloadDataLengthInSamples = 320;
+ reverse_frame._audioChannel = 2;
+ reverse_frame._frequencyInHz = 32000;
+
+ ap->echo_cancellation()->Enable(true);
+ ap->gain_control()->Enable(true);
+ ap->high_pass_filter()->Enable(true);
+ ap->level_estimator()->Enable(true);
+ ap->noise_suppression()->Enable(true);
+ ap->voice_detection()->Enable(true);
+
+ if (thread_data->thread_num % 2 == 0) {
+ err = ap->AnalyzeReverseStream(&reverse_frame);
+ if (err != ap->kNoError) {
+ printf("Error in AnalyzeReverseStream(): %d\n", err);
+ thread_data->error = true;
+ return false;
+ }
+ }
+
+ if (thread_data->thread_num % 2 == 1) {
+ ap->set_stream_delay_ms(0);
+ ap->echo_cancellation()->set_stream_drift_samples(0);
+ ap->gain_control()->set_stream_analog_level(0);
+ err = ap->ProcessStream(&primary_frame);
+ if (err == ap->kStreamParameterNotSetError) {
+ printf("Expected kStreamParameterNotSetError in ProcessStream(): %d\n",
+ err);
+ } else if (err != ap->kNoError) {
+ printf("Error in ProcessStream(): %d\n", err);
+ thread_data->error = true;
+ return false;
+ }
+ ap->gain_control()->stream_analog_level();
+ }
+
+ EventWrapper* event = EventWrapper::Create();
+ event->Wait(1);
+ delete event;
+ event = NULL;
+
+ return true;
+}
+
+/*TEST_F(ApmTest, Deadlock) {
+ const int num_threads = 16;
+ std::vector<ThreadWrapper*> threads(num_threads);
+ std::vector<ThreadData*> thread_data(num_threads);
+
+ ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
+
+ for (int i = 0; i < num_threads; i++) {
+ thread_data[i] = new ThreadData(i, apm_);
+ threads[i] = ThreadWrapper::CreateThread(DeadlockProc,
+ thread_data[i],
+ kNormalPriority,
+ 0);
+ ASSERT_TRUE(threads[i] != NULL);
+ unsigned int thread_id = 0;
+ threads[i]->Start(thread_id);
+ }
+
+ EventWrapper* event = EventWrapper::Create();
+ ASSERT_EQ(kEventTimeout, event->Wait(5000));
+ delete event;
+ event = NULL;
+
+ for (int i = 0; i < num_threads; i++) {
+ // This will return false if the thread has deadlocked.
+ ASSERT_TRUE(threads[i]->Stop());
+ ASSERT_FALSE(thread_data[i]->error);
+ delete threads[i];
+ threads[i] = NULL;
+ delete thread_data[i];
+ thread_data[i] = NULL;
+ }
+}*/
+
+TEST_F(ApmTest, StreamParameters) {
+ // No errors when the components are disabled.
+ EXPECT_EQ(apm_->kNoError,
+ apm_->ProcessStream(frame_));
+
+ // -- Missing AGC level --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError,
+ apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(false));
+
+ // -- Missing delay --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+
+ // -- Missing drift --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // -- No stream parameters --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->AnalyzeReverseStream(revframe_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError,
+ apm_->ProcessStream(frame_));
+
+ // -- All there --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+}
+
+TEST_F(ApmTest, Channels) {
+ // Testing number of invalid channels
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
+ // Testing number of valid channels
+ for (int i = 1; i < 3; i++) {
+ for (int j = 1; j < 3; j++) {
+ if (j > i) {
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
+ } else {
+ EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
+ EXPECT_EQ(j, apm_->num_output_channels());
+ }
+ }
+ EXPECT_EQ(i, apm_->num_input_channels());
+ EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
+ EXPECT_EQ(i, apm_->num_reverse_channels());
+ }
+}
+
+TEST_F(ApmTest, SampleRates) {
+ // Testing invalid sample rates
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
+ // Testing valid sample rates
+ int fs[] = {8000, 16000, 32000};
+ for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
+ EXPECT_EQ(fs[i], apm_->sample_rate_hz());
+ }
+}
+
+
+TEST_F(ApmTest, EchoCancellation) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(4000));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(100000));
+
+ int rate[] = {16000, 44100, 48000};
+ for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
+ EXPECT_EQ(rate[i],
+ apm_->echo_cancellation()->device_sample_rate_hz());
+ }
+
+ EchoCancellation::SuppressionLevel level[] = {
+ EchoCancellation::kLowSuppression,
+ EchoCancellation::kModerateSuppression,
+ EchoCancellation::kHighSuppression,
+ };
+ for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_suppression_level(level[i]));
+ EXPECT_EQ(level[i],
+ apm_->echo_cancellation()->suppression_level());
+ }
+
+ EchoCancellation::Metrics metrics;
+ EXPECT_EQ(apm_->kNotEnabledError,
+ apm_->echo_cancellation()->GetMetrics(&metrics));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
+
+ int median = 0;
+ int std = 0;
+ EXPECT_EQ(apm_->kNotEnabledError,
+ apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_delay_logging_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_delay_logging_enabled());
+
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
+}
+
+TEST_F(ApmTest, EchoControlMobile) {
+ // AECM won't use super-wideband.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
+ // Turn AECM on (and AEC off)
+ Init(16000, 2, 2, 2, false);
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+ EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
+
+ // Toggle routing modes
+ EchoControlMobile::RoutingMode mode[] = {
+ EchoControlMobile::kQuietEarpieceOrHeadset,
+ EchoControlMobile::kEarpiece,
+ EchoControlMobile::kLoudEarpiece,
+ EchoControlMobile::kSpeakerphone,
+ EchoControlMobile::kLoudSpeakerphone,
+ };
+ for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->set_routing_mode(mode[i]));
+ EXPECT_EQ(mode[i],
+ apm_->echo_control_mobile()->routing_mode());
+ }
+ // Turn comfort noise off/on
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->enable_comfort_noise(false));
+ EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->enable_comfort_noise(true));
+ EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
+ // Set and get echo path
+ const size_t echo_path_size =
+ apm_->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path_in(new char[echo_path_size]);
+ scoped_array<char> echo_path_out(new char[echo_path_size]);
+ EXPECT_EQ(apm_->kNullPointerError,
+ apm_->echo_control_mobile()->SetEchoPath(NULL, echo_path_size));
+ EXPECT_EQ(apm_->kNullPointerError,
+ apm_->echo_control_mobile()->GetEchoPath(NULL, echo_path_size));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(), 1));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
+ echo_path_size));
+ for (size_t i = 0; i < echo_path_size; i++) {
+ echo_path_in[i] = echo_path_out[i] + 1;
+ }
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(), 1));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(),
+ echo_path_size));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
+ echo_path_size));
+ for (size_t i = 0; i < echo_path_size; i++) {
+ EXPECT_EQ(echo_path_in[i], echo_path_out[i]);
+ }
+
+ // Process a few frames with NS in the default disabled state. This exercises
+ // a different codepath than with it enabled.
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+
+ // Turn AECM off
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+ EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
+}
+
+TEST_F(ApmTest, GainControl) {
+ // Testing gain modes
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(
+ apm_->gain_control()->mode()));
+
+ GainControl::Mode mode[] = {
+ GainControl::kAdaptiveAnalog,
+ GainControl::kAdaptiveDigital,
+ GainControl::kFixedDigital
+ };
+ for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(mode[i]));
+ EXPECT_EQ(mode[i], apm_->gain_control()->mode());
+ }
+ // Testing invalid target levels
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_target_level_dbfs(-3));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_target_level_dbfs(-40));
+ // Testing valid target levels
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_target_level_dbfs(
+ apm_->gain_control()->target_level_dbfs()));
+
+ int level_dbfs[] = {0, 6, 31};
+ for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
+ EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
+ }
+
+ // Testing invalid compression gains
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_compression_gain_db(-1));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_compression_gain_db(100));
+
+ // Testing valid compression gains
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_compression_gain_db(
+ apm_->gain_control()->compression_gain_db()));
+
+ int gain_db[] = {0, 10, 90};
+ for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_compression_gain_db(gain_db[i]));
+ EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
+ }
+
+ // Testing limiter off/on
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
+ EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
+ EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
+
+ // Testing invalid level limits
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(-1, 512));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(100000, 512));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, -1));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, 100000));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, 255));
+
+ // Testing valid level limits
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(
+ apm_->gain_control()->analog_level_minimum(),
+ apm_->gain_control()->analog_level_maximum()));
+
+ int min_level[] = {0, 255, 1024};
+ for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
+ EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
+ }
+
+ int max_level[] = {0, 1024, 65535};
+ for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
+ EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
+ }
+
+ // TODO(ajm): stream_is_saturated() and stream_analog_level()
+
+ // Turn AGC off
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+ EXPECT_FALSE(apm_->gain_control()->is_enabled());
+}
+
+TEST_F(ApmTest, NoiseSuppression) {
+ // Test valid suppression levels.
+ NoiseSuppression::Level level[] = {
+ NoiseSuppression::kLow,
+ NoiseSuppression::kModerate,
+ NoiseSuppression::kHigh,
+ NoiseSuppression::kVeryHigh
+ };
+ for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->noise_suppression()->set_level(level[i]));
+ EXPECT_EQ(level[i], apm_->noise_suppression()->level());
+ }
+
+ // Turn NS on/off
+ EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
+ EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
+ EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
+}
+
+TEST_F(ApmTest, HighPassFilter) {
+ // Turn HP filter on/off
+ EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
+ EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
+ EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
+}
+
+TEST_F(ApmTest, LevelEstimator) {
+ // Turn level estimator on/off
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_FALSE(apm_->level_estimator()->is_enabled());
+
+ EXPECT_EQ(apm_->kNotEnabledError, apm_->level_estimator()->RMS());
+
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_TRUE(apm_->level_estimator()->is_enabled());
+
+ // Run this test in wideband; in super-wb, the splitting filter distorts the
+ // audio enough to cause deviation from the expectation for small values.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ frame_->_payloadDataLengthInSamples = 160;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 16000;
+
+ // Min value if no frames have been processed.
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+
+ // Min value on zero frames.
+ SetFrameTo(frame_, 0);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+
+ // Try a few RMS values.
+ // (These also test that the value resets after retrieving it.)
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(0, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 30000);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(1, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 10000);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(10, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 10);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(70, apm_->level_estimator()->RMS());
+
+ // Min value if _energy == 0.
+ SetFrameTo(frame_, 10000);
+ uint32_t energy = frame_->_energy; // Save default to restore below.
+ frame_->_energy = 0;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+ frame_->_energy = energy;
+
+ // Verify reset after enable/disable.
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ SetFrameTo(frame_, 1);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(90, apm_->level_estimator()->RMS());
+
+ // Verify reset after initialize.
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ SetFrameTo(frame_, 1);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(90, apm_->level_estimator()->RMS());
+}
+
+TEST_F(ApmTest, VoiceDetection) {
+ // Test external VAD
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_stream_has_voice(true));
+ EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_stream_has_voice(false));
+ EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
+
+ // Test valid likelihoods
+ VoiceDetection::Likelihood likelihood[] = {
+ VoiceDetection::kVeryLowLikelihood,
+ VoiceDetection::kLowLikelihood,
+ VoiceDetection::kModerateLikelihood,
+ VoiceDetection::kHighLikelihood
+ };
+ for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_likelihood(likelihood[i]));
+ EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
+ }
+
+ /* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
+ // Test invalid frame sizes
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->voice_detection()->set_frame_size_ms(12));
+
+ // Test valid frame sizes
+ for (int i = 10; i <= 30; i += 10) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_frame_size_ms(i));
+ EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
+ }
+ */
+
+ // Turn VAD on/off
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_TRUE(apm_->voice_detection()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+ EXPECT_FALSE(apm_->voice_detection()->is_enabled());
+
+ // Test that AudioFrame activity is maintained when VAD is disabled.
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+ AudioFrame::VADActivity activity[] = {
+ AudioFrame::kVadActive,
+ AudioFrame::kVadPassive,
+ AudioFrame::kVadUnknown
+ };
+ for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) {
+ frame_->_vadActivity = activity[i];
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(activity[i], frame_->_vadActivity);
+ }
+
+ // Test that AudioFrame activity is set when VAD is enabled.
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ frame_->_vadActivity = AudioFrame::kVadUnknown;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_NE(AudioFrame::kVadUnknown, frame_->_vadActivity);
+
+ // TODO(bjornv): Add tests for streamed voice; stream_has_voice()
+}
+
+TEST_F(ApmTest, SplittingFilter) {
+ // Verify the filter is not active through undistorted audio when:
+ // 1. No components are enabled...
+ SetFrameTo(frame_, 1000);
+ AudioFrame frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+
+ // 2. Only the level estimator is enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+
+ // 3. Only VAD is enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+
+ // 4. Both VAD and the level estimator are enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+
+ // 5. Not using super-wb.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ frame_->_payloadDataLengthInSamples = 160;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 16000;
+ // Enable AEC, which would require the filter in super-wb. We rely on the
+ // first few frames of data being unaffected by the AEC.
+ // TODO(andrew): This test, and the one below, rely rather tenuously on the
+ // behavior of the AEC. Think of something more robust.
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+
+ // Check the test is valid. We should have distortion from the filter
+ // when AEC is enabled (which won't affect the audio).
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ frame_->_payloadDataLengthInSamples = 320;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 32000;
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
+}
+
+// TODO(andrew): expand test to verify output.
+TEST_F(ApmTest, DebugDump) {
+ const std::string filename = webrtc::test::OutputPath() + "debug.aec";
+ EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(NULL));
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // Stopping without having started should be OK.
+ EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
+ EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+
+ // Verify the file has been written.
+ FILE* fid = fopen(filename.c_str(), "r");
+ ASSERT_TRUE(fid != NULL);
+
+ // Clean it up.
+ ASSERT_EQ(0, fclose(fid));
+ ASSERT_EQ(0, remove(filename.c_str()));
+#else
+ EXPECT_EQ(apm_->kUnsupportedFunctionError,
+ apm_->StartDebugRecording(filename.c_str()));
+ EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
+
+ // Verify the file has NOT been written.
+ ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+// TODO(andrew): Add a test to process a few frames with different combinations
+// of enabled components.
+
+// TODO(andrew): Make this test more robust such that it can be run on multiple
+// platforms. It currently requires bit-exactness.
+#ifdef WEBRTC_AUDIOPROC_BIT_EXACT
+TEST_F(ApmTest, Process) {
+ GOOGLE_PROTOBUF_VERIFY_VERSION;
+ webrtc::audioproc::OutputData ref_data;
+
+ if (!write_ref_data) {
+ ReadMessageLiteFromFile(ref_filename_, &ref_data);
+ } else {
+ // Write the desired tests to the protobuf reference file.
+ const int channels[] = {1, 2};
+ const size_t channels_size = sizeof(channels) / sizeof(*channels);
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+ // AECM doesn't support super-wb.
+ const int sample_rates[] = {8000, 16000};
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ const int sample_rates[] = {8000, 16000, 32000};
+#endif
+ const size_t sample_rates_size = sizeof(sample_rates) / sizeof(*sample_rates);
+ for (size_t i = 0; i < channels_size; i++) {
+ for (size_t j = 0; j < channels_size; j++) {
+ // We can't have more output than input channels.
+ for (size_t k = 0; k <= j; k++) {
+ for (size_t l = 0; l < sample_rates_size; l++) {
+ webrtc::audioproc::Test* test = ref_data.add_test();
+ test->set_num_reverse_channels(channels[i]);
+ test->set_num_input_channels(channels[j]);
+ test->set_num_output_channels(channels[k]);
+ test->set_sample_rate(sample_rates[l]);
+ }
+ }
+ }
+ }
+ }
+
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(true));
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(0, 255));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+#endif
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->high_pass_filter()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->level_estimator()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->noise_suppression()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->Enable(true));
+
+ for (int i = 0; i < ref_data.test_size(); i++) {
+ printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
+
+ webrtc::audioproc::Test* test = ref_data.mutable_test(i);
+ Init(test->sample_rate(), test->num_reverse_channels(),
+ test->num_input_channels(), test->num_output_channels(), true);
+
+ const int samples_per_channel = test->sample_rate() / 100;
+ int frame_count = 0;
+ int has_echo_count = 0;
+ int has_voice_count = 0;
+ int is_saturated_count = 0;
+ int analog_level = 127;
+ int analog_level_average = 0;
+ int max_output_average = 0;
+
+ while (1) {
+ // Read far-end frame, always stereo.
+ size_t frame_size = samples_per_channel * 2;
+ size_t read_count = fread(revframe_->_payloadData,
+ sizeof(int16_t),
+ frame_size,
+ far_file_);
+ if (read_count != frame_size) {
+ // Check that the file really ended.
+ ASSERT_NE(0, feof(far_file_));
+ break; // This is expected.
+ }
+
+ if (revframe_->_audioChannel == 1) {
+ MixStereoToMono(revframe_->_payloadData, revframe_->_payloadData,
+ samples_per_channel);
+ }
+
+ EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
+
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(analog_level));
+
+ // Read near-end frame, always stereo.
+ read_count = fread(frame_->_payloadData,
+ sizeof(int16_t),
+ frame_size,
+ near_file_);
+ if (read_count != frame_size) {
+ // Check that the file really ended.
+ ASSERT_NE(0, feof(near_file_));
+ break; // This is expected.
+ }
+
+ if (frame_->_audioChannel == 1) {
+ MixStereoToMono(frame_->_payloadData, frame_->_payloadData,
+ samples_per_channel);
+ }
+ frame_->_vadActivity = AudioFrame::kVadUnknown;
+
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ // Ensure the frame was downmixed properly.
+ EXPECT_EQ(test->num_output_channels(), frame_->_audioChannel);
+
+ max_output_average += MaxAudioFrame(*frame_);
+
+ if (apm_->echo_cancellation()->stream_has_echo()) {
+ has_echo_count++;
+ }
+
+ analog_level = apm_->gain_control()->stream_analog_level();
+ analog_level_average += analog_level;
+ if (apm_->gain_control()->stream_is_saturated()) {
+ is_saturated_count++;
+ }
+ if (apm_->voice_detection()->stream_has_voice()) {
+ has_voice_count++;
+ EXPECT_EQ(AudioFrame::kVadActive, frame_->_vadActivity);
+ } else {
+ EXPECT_EQ(AudioFrame::kVadPassive, frame_->_vadActivity);
+ }
+
+ frame_size = samples_per_channel * frame_->_audioChannel;
+ size_t write_count = fwrite(frame_->_payloadData,
+ sizeof(int16_t),
+ frame_size,
+ out_file_);
+ ASSERT_EQ(frame_size, write_count);
+
+ // Reset in case of downmixing.
+ frame_->_audioChannel = test->num_input_channels();
+ frame_count++;
+ }
+ max_output_average /= frame_count;
+ analog_level_average /= frame_count;
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ EchoCancellation::Metrics echo_metrics;
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->GetMetrics(&echo_metrics));
+ int median = 0;
+ int std = 0;
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
+
+ int rms_level = apm_->level_estimator()->RMS();
+ EXPECT_LE(0, rms_level);
+ EXPECT_GE(127, rms_level);
+#endif
+
+ if (!write_ref_data) {
+ EXPECT_EQ(test->has_echo_count(), has_echo_count);
+ EXPECT_EQ(test->has_voice_count(), has_voice_count);
+ EXPECT_EQ(test->is_saturated_count(), is_saturated_count);
+
+ EXPECT_EQ(test->analog_level_average(), analog_level_average);
+ EXPECT_EQ(test->max_output_average(), max_output_average);
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ webrtc::audioproc::Test::EchoMetrics reference =
+ test->echo_metrics();
+ TestStats(echo_metrics.residual_echo_return_loss,
+ reference.residual_echo_return_loss());
+ TestStats(echo_metrics.echo_return_loss,
+ reference.echo_return_loss());
+ TestStats(echo_metrics.echo_return_loss_enhancement,
+ reference.echo_return_loss_enhancement());
+ TestStats(echo_metrics.a_nlp,
+ reference.a_nlp());
+
+ webrtc::audioproc::Test::DelayMetrics reference_delay =
+ test->delay_metrics();
+ EXPECT_EQ(reference_delay.median(), median);
+ EXPECT_EQ(reference_delay.std(), std);
+
+ EXPECT_EQ(test->rms_level(), rms_level);
+#endif
+ } else {
+ test->set_has_echo_count(has_echo_count);
+ test->set_has_voice_count(has_voice_count);
+ test->set_is_saturated_count(is_saturated_count);
+
+ test->set_analog_level_average(analog_level_average);
+ test->set_max_output_average(max_output_average);
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ webrtc::audioproc::Test::EchoMetrics* message =
+ test->mutable_echo_metrics();
+ WriteStatsMessage(echo_metrics.residual_echo_return_loss,
+ message->mutable_residual_echo_return_loss());
+ WriteStatsMessage(echo_metrics.echo_return_loss,
+ message->mutable_echo_return_loss());
+ WriteStatsMessage(echo_metrics.echo_return_loss_enhancement,
+ message->mutable_echo_return_loss_enhancement());
+ WriteStatsMessage(echo_metrics.a_nlp,
+ message->mutable_a_nlp());
+
+ webrtc::audioproc::Test::DelayMetrics* message_delay =
+ test->mutable_delay_metrics();
+ message_delay->set_median(median);
+ message_delay->set_std(std);
+
+ test->set_rms_level(rms_level);
+#endif
+ }
+
+ rewind(far_file_);
+ rewind(near_file_);
+ }
+
+ if (write_ref_data) {
+ WriteMessageLiteToFile(ref_filename_, ref_data);
+ }
+}
+#endif // WEBRTC_AUDIOPROC_BIT_EXACT
+
+} // namespace
+
+int main(int argc, char** argv) {
+ ::testing::InitGoogleTest(&argc, argv);
+
+ for (int i = 1; i < argc; i++) {
+ if (strcmp(argv[i], "--write_ref_data") == 0) {
+ write_ref_data = true;
+ }
+ }
+
+ int err = RUN_ALL_TESTS();
+
+ // Optional, but removes memory leak noise from Valgrind.
+ google::protobuf::ShutdownProtobufLibrary();
+ return err;
+}
diff --git a/trunk/src/modules/audio_processing/test/unittest.proto b/trunk/src/modules/audio_processing/test/unittest.proto
new file mode 100644
index 0000000..67ba722
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/unittest.proto
@@ -0,0 +1,52 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+message Test {
+ optional int32 num_reverse_channels = 1;
+ optional int32 num_input_channels = 2;
+ optional int32 num_output_channels = 3;
+ optional int32 sample_rate = 4;
+
+ message Frame {
+ }
+
+ repeated Frame frame = 5;
+
+ optional int32 analog_level_average = 6;
+ optional int32 max_output_average = 7;
+
+ optional int32 has_echo_count = 8;
+ optional int32 has_voice_count = 9;
+ optional int32 is_saturated_count = 10;
+
+ message Statistic {
+ optional int32 instant = 1;
+ optional int32 average = 2;
+ optional int32 maximum = 3;
+ optional int32 minimum = 4;
+ }
+
+ message EchoMetrics {
+ optional Statistic residual_echo_return_loss = 1;
+ optional Statistic echo_return_loss = 2;
+ optional Statistic echo_return_loss_enhancement = 3;
+ optional Statistic a_nlp = 4;
+ }
+
+ optional EchoMetrics echo_metrics = 11;
+
+ message DelayMetrics {
+ optional int32 median = 1;
+ optional int32 std = 2;
+ }
+
+ optional DelayMetrics delay_metrics = 12;
+
+ optional int32 rms_level = 13;
+}
+
+message OutputData {
+ repeated Test test = 1;
+}
+
diff --git a/trunk/src/modules/audio_processing/test/unpack.cc b/trunk/src/modules/audio_processing/test/unpack.cc
new file mode 100644
index 0000000..2337131
--- /dev/null
+++ b/trunk/src/modules/audio_processing/test/unpack.cc
@@ -0,0 +1,216 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Commandline tool to unpack audioproc debug files.
+//
+// The debug files are dumped as protobuf blobs. For analysis, it's necessary
+// to unpack the file into its component parts: audio and other data.
+
+#include <stdio.h>
+
+#include "google/gflags.h"
+#include "scoped_ptr.h"
+#include "typedefs.h"
+#include "webrtc/audio_processing/debug.pb.h"
+
+using webrtc::scoped_array;
+
+using webrtc::audioproc::Event;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+using webrtc::audioproc::Init;
+
+// TODO(andrew): unpack more of the data.
+DEFINE_string(input_file, "input.pcm", "The name of the input stream file.");
+DEFINE_string(output_file, "ref_out.pcm",
+ "The name of the reference output stream file.");
+DEFINE_string(reverse_file, "reverse.pcm",
+ "The name of the reverse input stream file.");
+DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
+DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
+DEFINE_string(level_file, "level.int32", "The name of the level file.");
+DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
+DEFINE_bool(full, false,
+ "Unpack the full set of files (normally not needed).");
+
+// TODO(andrew): move this to a helper class to share with process_test.cc?
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file,
+ ::google::protobuf::MessageLite* msg) {
+ // The "wire format" for the size is little-endian.
+ // Assume process_test is running on a little-endian machine.
+ int32_t size = 0;
+ if (fread(&size, sizeof(int32_t), 1, file) != 1) {
+ return false;
+ }
+ if (size <= 0) {
+ return false;
+ }
+ const size_t usize = static_cast<size_t>(size);
+
+ scoped_array<char> array(new char[usize]);
+ if (fread(array.get(), sizeof(char), usize, file) != usize) {
+ return false;
+ }
+
+ msg->Clear();
+ return msg->ParseFromArray(array.get(), usize);
+}
+
+int main(int argc, char* argv[]) {
+ std::string program_name = argv[0];
+ std::string usage = "Commandline tool to unpack audioproc debug files.\n"
+ "Example usage:\n" + program_name + " debug_dump.pb\n";
+ google::SetUsageMessage(usage);
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ if (argc < 2) {
+ printf("%s", google::ProgramUsage());
+ return 1;
+ }
+
+ FILE* debug_file = fopen(argv[1], "rb");
+ if (debug_file == NULL) {
+ printf("Unable to open %s\n", argv[1]);
+ return 1;
+ }
+ FILE* input_file = fopen(FLAGS_input_file.c_str(), "wb");
+ if (input_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_input_file.c_str());
+ return 1;
+ }
+ FILE* output_file = fopen(FLAGS_output_file.c_str(), "wb");
+ if (output_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_output_file.c_str());
+ return 1;
+ }
+ FILE* reverse_file = fopen(FLAGS_reverse_file.c_str(), "wb");
+ if (reverse_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_reverse_file.c_str());
+ return 1;
+ }
+ FILE* settings_file = fopen(FLAGS_settings_file.c_str(), "wb");
+ if (settings_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_settings_file.c_str());
+ return 1;
+ }
+
+ FILE* delay_file = NULL;
+ FILE* drift_file = NULL;
+ FILE* level_file = NULL;
+ if (FLAGS_full) {
+ delay_file = fopen(FLAGS_delay_file.c_str(), "wb");
+ if (delay_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_delay_file.c_str());
+ return 1;
+ }
+ drift_file = fopen(FLAGS_drift_file.c_str(), "wb");
+ if (drift_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_drift_file.c_str());
+ return 1;
+ }
+ level_file = fopen(FLAGS_level_file.c_str(), "wb");
+ if (level_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_level_file.c_str());
+ return 1;
+ }
+ }
+
+ Event event_msg;
+ int frame_count = 0;
+ while (ReadMessageFromFile(debug_file, &event_msg)) {
+ if (event_msg.type() == Event::REVERSE_STREAM) {
+ if (!event_msg.has_reverse_stream()) {
+ printf("Corrupted input file: ReverseStream missing.\n");
+ return 1;
+ }
+
+ const ReverseStream msg = event_msg.reverse_stream();
+ if (msg.has_data()) {
+ if (fwrite(msg.data().data(), msg.data().size(), 1, reverse_file) !=
+ 1) {
+ printf("Error when writing to %s\n", FLAGS_reverse_file.c_str());
+ return 1;
+ }
+ }
+ } else if (event_msg.type() == Event::STREAM) {
+ frame_count++;
+ if (!event_msg.has_stream()) {
+ printf("Corrupted input file: Stream missing.\n");
+ return 1;
+ }
+
+ const Stream msg = event_msg.stream();
+ if (msg.has_input_data()) {
+ if (fwrite(msg.input_data().data(), msg.input_data().size(), 1,
+ input_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_input_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_output_data()) {
+ if (fwrite(msg.output_data().data(), msg.output_data().size(), 1,
+ output_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_output_file.c_str());
+ return 1;
+ }
+ }
+
+ if (FLAGS_full) {
+ if (msg.has_delay()) {
+ int32_t delay = msg.delay();
+ if (fwrite(&delay, sizeof(int32_t), 1, delay_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_delay_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_drift()) {
+ int32_t drift = msg.drift();
+ if (fwrite(&drift, sizeof(int32_t), 1, drift_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_drift_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_level()) {
+ int32_t level = msg.level();
+ if (fwrite(&level, sizeof(int32_t), 1, level_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_level_file.c_str());
+ return 1;
+ }
+ }
+ }
+ } else if (event_msg.type() == Event::INIT) {
+ if (!event_msg.has_init()) {
+ printf("Corrupted input file: Init missing.\n");
+ return 1;
+ }
+
+ const Init msg = event_msg.init();
+ // These should print out zeros if they're missing.
+ fprintf(settings_file, "Init at frame: %d\n", frame_count);
+ fprintf(settings_file, " Sample rate: %d\n", msg.sample_rate());
+ fprintf(settings_file, " Device sample rate: %d\n",
+ msg.device_sample_rate());
+ fprintf(settings_file, " Input channels: %d\n",
+ msg.num_input_channels());
+ fprintf(settings_file, " Output channels: %d\n",
+ msg.num_output_channels());
+ fprintf(settings_file, " Reverse channels: %d\n",
+ msg.num_reverse_channels());
+
+ fprintf(settings_file, "\n");
+ }
+ }
+
+ return 0;
+}
diff --git a/trunk/src/modules/audio_processing/utility/Android.mk b/trunk/src/modules/audio_processing/utility/Android.mk
new file mode 100644
index 0000000..bd3d039
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/Android.mk
@@ -0,0 +1,43 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_apm_utility
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ fft4g.c \
+ ring_buffer.c \
+ delay_estimator.c \
+ delay_estimator_wrapper.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator.c b/trunk/src/modules/audio_processing/utility/delay_estimator.c
new file mode 100644
index 0000000..24ee74d
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/delay_estimator.c
@@ -0,0 +1,319 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "delay_estimator.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+// Number of right shifts for scaling is linearly depending on number of bits in
+// the far-end binary spectrum.
+static const int kShiftsAtZero = 13; // Right shifts at zero binary spectrum.
+static const int kShiftsLinearSlope = 3;
+
+static const int32_t kProbabilityOffset = 1024; // 2 in Q9.
+static const int32_t kProbabilityLowerLimit = 8704; // 17 in Q9.
+static const int32_t kProbabilityMinSpread = 2816; // 5.5 in Q9.
+
+// Counts and returns number of bits of a 32-bit word.
+static int BitCount(uint32_t u32) {
+ uint32_t tmp = u32 - ((u32 >> 1) & 033333333333) -
+ ((u32 >> 2) & 011111111111);
+ tmp = ((tmp + (tmp >> 3)) & 030707070707);
+ tmp = (tmp + (tmp >> 6));
+ tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
+
+ return ((int) tmp);
+}
+
+// Compares the |binary_vector| with all rows of the |binary_matrix| and counts
+// per row the number of times they have the same value.
+//
+// Inputs:
+// - binary_vector : binary "vector" stored in a long
+// - binary_matrix : binary "matrix" stored as a vector of long
+// - matrix_size : size of binary "matrix"
+//
+// Output:
+// - bit_counts : "Vector" stored as a long, containing for each
+// row the number of times the matrix row and the
+// input vector have the same value
+//
+static void BitCountComparison(uint32_t binary_vector,
+ const uint32_t* binary_matrix,
+ int matrix_size,
+ int32_t* bit_counts) {
+ int n = 0;
+
+ // Compare |binary_vector| with all rows of the |binary_matrix|
+ for (; n < matrix_size; n++) {
+ bit_counts[n] = (int32_t) BitCount(binary_vector ^ binary_matrix[n]);
+ }
+}
+
+int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+
+ if (handle->mean_bit_counts != NULL) {
+ free(handle->mean_bit_counts);
+ handle->mean_bit_counts = NULL;
+ }
+ if (handle->bit_counts != NULL) {
+ free(handle->bit_counts);
+ handle->bit_counts = NULL;
+ }
+ if (handle->binary_far_history != NULL) {
+ free(handle->binary_far_history);
+ handle->binary_far_history = NULL;
+ }
+ if (handle->binary_near_history != NULL) {
+ free(handle->binary_near_history);
+ handle->binary_near_history = NULL;
+ }
+ if (handle->far_bit_counts != NULL) {
+ free(handle->far_bit_counts);
+ handle->far_bit_counts = NULL;
+ }
+
+ free(handle);
+
+ return 0;
+}
+
+int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
+ int max_delay,
+ int lookahead) {
+ BinaryDelayEstimator* self = NULL;
+ int history_size = max_delay + lookahead;
+
+ if (handle == NULL) {
+ return -1;
+ }
+ if (max_delay < 0) {
+ return -1;
+ }
+ if (lookahead < 0) {
+ return -1;
+ }
+ if (history_size < 2) {
+ // Must be this large for buffer shifting.
+ return -1;
+ }
+
+ self = malloc(sizeof(BinaryDelayEstimator));
+ *handle = self;
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->mean_bit_counts = NULL;
+ self->bit_counts = NULL;
+ self->binary_far_history = NULL;
+ self->far_bit_counts = NULL;
+
+ self->history_size = history_size;
+ self->near_history_size = lookahead + 1;
+
+ // Allocate memory for spectrum buffers.
+ self->mean_bit_counts = malloc(history_size * sizeof(int32_t));
+ if (self->mean_bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->bit_counts = malloc(history_size * sizeof(int32_t));
+ if (self->bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ // Allocate memory for history buffers.
+ self->binary_far_history = malloc(history_size * sizeof(uint32_t));
+ if (self->binary_far_history == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->binary_near_history = malloc(self->near_history_size *
+ sizeof(uint32_t));
+ if (self->binary_near_history == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->far_bit_counts = malloc(history_size * sizeof(int));
+ if (self->far_bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle) {
+ int i = 0;
+ assert(handle != NULL);
+
+ memset(handle->bit_counts, 0, sizeof(int32_t) * handle->history_size);
+ memset(handle->binary_far_history, 0,
+ sizeof(uint32_t) * handle->history_size);
+ memset(handle->binary_near_history, 0,
+ sizeof(uint32_t) * handle->near_history_size);
+ memset(handle->far_bit_counts, 0, sizeof(int) * handle->history_size);
+ for (i = 0; i < handle->history_size; ++i) {
+ handle->mean_bit_counts[i] = (20 << 9); // 20 in Q9.
+ }
+ handle->minimum_probability = (32 << 9); // 32 in Q9.
+ handle->last_delay_probability = (32 << 9); // 32 in Q9.
+
+ // Default return value if we're unable to estimate. -1 is used for errors.
+ handle->last_delay = -2;
+
+ return 0;
+}
+
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
+ uint32_t binary_far_spectrum,
+ uint32_t binary_near_spectrum) {
+ int i = 0;
+ int candidate_delay = -1;
+
+ int32_t value_best_candidate = 16384; // 32 in Q9, (max |mean_bit_counts|).
+ int32_t value_worst_candidate = 0;
+
+ assert(handle != NULL);
+ // Shift binary spectrum history and insert current |binary_far_spectrum|.
+ memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
+ (handle->history_size - 1) * sizeof(uint32_t));
+ handle->binary_far_history[0] = binary_far_spectrum;
+
+ // Shift history of far-end binary spectrum bit counts and insert bit count
+ // of current |binary_far_spectrum|.
+ memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
+ (handle->history_size - 1) * sizeof(int));
+ handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
+
+ if (handle->near_history_size > 1) {
+ // If we apply lookahead, shift near-end binary spectrum history. Insert
+ // current |binary_near_spectrum| and pull out the delayed one.
+ memmove(&(handle->binary_near_history[1]),
+ &(handle->binary_near_history[0]),
+ (handle->near_history_size - 1) * sizeof(uint32_t));
+ handle->binary_near_history[0] = binary_near_spectrum;
+ binary_near_spectrum =
+ handle->binary_near_history[handle->near_history_size - 1];
+ }
+
+ // Compare with delayed spectra and store the |bit_counts| for each delay.
+ BitCountComparison(binary_near_spectrum,
+ handle->binary_far_history,
+ handle->history_size,
+ handle->bit_counts);
+
+ // Update |mean_bit_counts|, which is the smoothed version of |bit_counts|.
+ for (i = 0; i < handle->history_size; i++) {
+ // |bit_counts| is constrained to [0, 32], meaning we can smooth with a
+ // factor up to 2^26. We use Q9.
+ int32_t bit_count = (handle->bit_counts[i] << 9); // Q9.
+
+ // Update |mean_bit_counts| only when far-end signal has something to
+ // contribute. If |far_bit_counts| is zero the far-end signal is weak and
+ // we likely have a poor echo condition, hence don't update.
+ if (handle->far_bit_counts[i] > 0) {
+ // Make number of right shifts piecewise linear w.r.t. |far_bit_counts|.
+ int shifts = kShiftsAtZero;
+ shifts -= (kShiftsLinearSlope * handle->far_bit_counts[i]) >> 4;
+ WebRtc_MeanEstimatorFix(bit_count, shifts, &(handle->mean_bit_counts[i]));
+ }
+ }
+
+ // Find |candidate_delay|, |value_best_candidate| and |value_worst_candidate|
+ // of |mean_bit_counts|.
+ for (i = 0; i < handle->history_size; i++) {
+ if (handle->mean_bit_counts[i] < value_best_candidate) {
+ value_best_candidate = handle->mean_bit_counts[i];
+ candidate_delay = i;
+ }
+ if (handle->mean_bit_counts[i] > value_worst_candidate) {
+ value_worst_candidate = handle->mean_bit_counts[i];
+ }
+ }
+
+ // The |value_best_candidate| is a good indicator on the probability of
+ // |candidate_delay| being an accurate delay (a small |value_best_candidate|
+ // means a good binary match). In the following sections we make a decision
+ // whether to update |last_delay| or not.
+ // 1) If the difference bit counts between the best and the worst delay
+ // candidates is too small we consider the situation to be unreliable and
+ // don't update |last_delay|.
+ // 2) If the situation is reliable we update |last_delay| if the value of the
+ // best candidate delay has a value less than
+ // i) an adaptive threshold |minimum_probability|, or
+ // ii) this corresponding value |last_delay_probability|, but updated at
+ // this time instant.
+
+ // Update |minimum_probability|.
+ if ((handle->minimum_probability > kProbabilityLowerLimit) &&
+ (value_worst_candidate - value_best_candidate > kProbabilityMinSpread)) {
+ // The "hard" threshold can't be lower than 17 (in Q9).
+ // The valley in the curve also has to be distinct, i.e., the
+ // difference between |value_worst_candidate| and |value_best_candidate| has
+ // to be large enough.
+ int32_t threshold = value_best_candidate + kProbabilityOffset;
+ if (threshold < kProbabilityLowerLimit) {
+ threshold = kProbabilityLowerLimit;
+ }
+ if (handle->minimum_probability > threshold) {
+ handle->minimum_probability = threshold;
+ }
+ }
+ // Update |last_delay_probability|.
+ // We use a Markov type model, i.e., a slowly increasing level over time.
+ handle->last_delay_probability++;
+ if (value_worst_candidate > value_best_candidate + kProbabilityOffset) {
+ // Reliable delay value for usage.
+ if (value_best_candidate < handle->minimum_probability) {
+ handle->last_delay = candidate_delay;
+ }
+ if (value_best_candidate < handle->last_delay_probability) {
+ handle->last_delay = candidate_delay;
+ // Reset |last_delay_probability|.
+ handle->last_delay_probability = value_best_candidate;
+ }
+ }
+
+ return handle->last_delay;
+}
+
+int WebRtc_binary_last_delay(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+ return handle->last_delay;
+}
+
+int WebRtc_history_size(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+ return handle->history_size;
+}
+
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+ int factor,
+ int32_t* mean_value) {
+ int32_t diff = new_value - *mean_value;
+
+ // mean_new = mean_value + ((new_value - mean_value) >> factor);
+ if (diff < 0) {
+ diff = -((-diff) >> factor);
+ } else {
+ diff = (diff >> factor);
+ }
+ *mean_value += diff;
+}
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator.h b/trunk/src/modules/audio_processing/utility/delay_estimator.h
new file mode 100644
index 0000000..a376dfe
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/delay_estimator.h
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on binary converted spectra.
+// The return value is 0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+
+#include "typedefs.h"
+
+typedef struct {
+ // Pointer to bit counts.
+ int32_t* mean_bit_counts;
+ int* far_bit_counts;
+
+ // Array only used locally in ProcessBinarySpectrum() but whose size is
+ // determined at run-time.
+ int32_t* bit_counts;
+
+ // Binary history variables.
+ uint32_t* binary_far_history;
+ uint32_t* binary_near_history;
+
+ // Delay estimation variables.
+ int32_t minimum_probability;
+ int last_delay_probability;
+
+ // Delay memory.
+ int last_delay;
+
+ // Buffer size.
+ int history_size;
+
+ // Near-end buffer size.
+ int near_history_size;
+} BinaryDelayEstimator;
+
+// Releases the memory allocated by WebRtc_CreateBinaryDelayEstimator(...).
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle);
+
+// Refer to WebRtc_CreateDelayEstimator() in delay_estimator_wrapper.h.
+int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
+ int max_delay,
+ int lookahead);
+
+// Initializes the delay estimation instance created with
+// WebRtc_CreateBinaryDelayEstimator(...).
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Output:
+// - handle : Initialized instance.
+//
+int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle);
+
+// Estimates and returns the delay between the binary far-end and binary near-
+// end spectra. The value will be offset by the lookahead (i.e. the lookahead
+// should be subtracted from the returned value).
+// Inputs:
+// - handle : Pointer to the delay estimation instance.
+// - binary_far_spectrum : Far-end binary spectrum.
+// - binary_near_spectrum : Near-end binary spectrum of the current block.
+//
+// Output:
+// - handle : Updated instance.
+//
+// Return value:
+// - delay : >= 0 - Calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
+ uint32_t binary_far_spectrum,
+ uint32_t binary_near_spectrum);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_ProcessBinarySpectrum(...).
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - delay : >= 0 - Last calculated delay value
+// -1 - Error
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_binary_last_delay(BinaryDelayEstimator* handle);
+
+// Returns the history size used in the far-end buffers to calculate the delay
+// over.
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - history_size : > 0 - Far-end history size.
+// -1 - Error.
+//
+int WebRtc_history_size(BinaryDelayEstimator* handle);
+
+// Updates the |mean_value| recursively with a step size of 2^-|factor|. This
+// function is used internally in the Binary Delay Estimator as well as the
+// Fixed point wrapper.
+//
+// Inputs:
+// - new_value : The new value the mean should be updated with.
+// - factor : The step size, in number of right shifts.
+//
+// Input/Output:
+// - mean_value : Pointer to the mean value.
+//
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+ int factor,
+ int32_t* mean_value);
+
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c
new file mode 100644
index 0000000..438c95f
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c
@@ -0,0 +1,336 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "delay_estimator_wrapper.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "delay_estimator.h"
+
+typedef union {
+ float float_;
+ int32_t int32_;
+} SpectrumType;
+
+typedef struct {
+ // Pointers to mean values of spectrum.
+ SpectrumType* mean_far_spectrum;
+ SpectrumType* mean_near_spectrum;
+ // |mean_*_spectrum| initialization indicator.
+ int far_spectrum_initialized;
+ int near_spectrum_initialized;
+
+ int spectrum_size;
+
+ // Binary spectrum based delay estimator
+ BinaryDelayEstimator* binary_handle;
+} DelayEstimator;
+
+// Only bit |kBandFirst| through bit |kBandLast| are processed and
+// |kBandFirst| - |kBandLast| must be < 32.
+static const int kBandFirst = 12;
+static const int kBandLast = 43;
+
+static __inline uint32_t SetBit(uint32_t in, int pos) {
+ uint32_t mask = (1 << pos);
+ uint32_t out = (in | mask);
+
+ return out;
+}
+
+// Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(),
+// but for float.
+//
+// Inputs:
+// - new_value : New additional value.
+// - scale : Scale for smoothing (should be less than 1.0).
+//
+// Input/Output:
+// - mean_value : Pointer to the mean value for updating.
+//
+static void MeanEstimatorFloat(float new_value,
+ float scale,
+ float* mean_value) {
+ assert(scale < 1.0f);
+ *mean_value += (new_value - *mean_value) * scale;
+}
+
+// Computes the binary spectrum by comparing the input |spectrum| with a
+// |threshold_spectrum|. Float and fixed point versions.
+//
+// Inputs:
+// - spectrum : Spectrum of which the binary spectrum should be
+// calculated.
+// - threshold_spectrum : Threshold spectrum with which the input
+// spectrum is compared.
+// Return:
+// - out : Binary spectrum.
+//
+static uint32_t BinarySpectrumFix(uint16_t* spectrum,
+ SpectrumType* threshold_spectrum,
+ int q_domain,
+ int* threshold_initialized) {
+ int i = kBandFirst;
+ uint32_t out = 0;
+
+ assert(q_domain < 16);
+
+ if (!(*threshold_initialized)) {
+ // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // value. This speeds up the convergence.
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ if (spectrum[i] > 0) {
+ // Convert input spectrum from Q(|q_domain|) to Q15.
+ int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
+ threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
+ *threshold_initialized = 1;
+ }
+ }
+ }
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ // Convert input spectrum from Q(|q_domain|) to Q15.
+ int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
+ // Update the |threshold_spectrum|.
+ WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
+ // Convert |spectrum| at current frequency bin to a binary value.
+ if (spectrum_q15 > threshold_spectrum[i].int32_) {
+ out = SetBit(out, i - kBandFirst);
+ }
+ }
+
+ return out;
+}
+
+static uint32_t BinarySpectrumFloat(float* spectrum,
+ SpectrumType* threshold_spectrum,
+ int* threshold_initialized) {
+ int i = kBandFirst;
+ uint32_t out = 0;
+ const float kScale = 1 / 64.0;
+
+ if (!(*threshold_initialized)) {
+ // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // value. This speeds up the convergence.
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ if (spectrum[i] > 0.0f) {
+ threshold_spectrum[i].float_ = (spectrum[i] / 2);
+ *threshold_initialized = 1;
+ }
+ }
+ }
+
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ // Update the |threshold_spectrum|.
+ MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
+ // Convert |spectrum| at current frequency bin to a binary value.
+ if (spectrum[i] > threshold_spectrum[i].float_) {
+ out = SetBit(out, i - kBandFirst);
+ }
+ }
+
+ return out;
+}
+
+int WebRtc_FreeDelayEstimator(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ if (self->mean_far_spectrum != NULL) {
+ free(self->mean_far_spectrum);
+ self->mean_far_spectrum = NULL;
+ }
+ if (self->mean_near_spectrum != NULL) {
+ free(self->mean_near_spectrum);
+ self->mean_near_spectrum = NULL;
+ }
+
+ WebRtc_FreeBinaryDelayEstimator(self->binary_handle);
+
+ free(self);
+
+ return 0;
+}
+
+int WebRtc_CreateDelayEstimator(void** handle,
+ int spectrum_size,
+ int max_delay,
+ int lookahead) {
+ DelayEstimator* self = NULL;
+
+ // Check if the sub band used in the delay estimation is small enough to fit
+ // the binary spectra in a uint32_t.
+ assert(kBandLast - kBandFirst < 32);
+
+ if (handle == NULL) {
+ return -1;
+ }
+ if (spectrum_size < kBandLast) {
+ return -1;
+ }
+
+ self = malloc(sizeof(DelayEstimator));
+ *handle = self;
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->mean_far_spectrum = NULL;
+ self->mean_near_spectrum = NULL;
+
+ // Create binary delay estimator.
+ if (WebRtc_CreateBinaryDelayEstimator(&self->binary_handle,
+ max_delay,
+ lookahead) != 0) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ // Allocate memory for spectrum buffers.
+ self->mean_far_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
+ if (self->mean_far_spectrum == NULL) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->mean_near_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
+ if (self->mean_near_spectrum == NULL) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+
+ self->spectrum_size = spectrum_size;
+
+ return 0;
+}
+
+int WebRtc_InitDelayEstimator(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ // Initialize binary delay estimator.
+ if (WebRtc_InitBinaryDelayEstimator(self->binary_handle) != 0) {
+ return -1;
+ }
+ // Set averaged far and near end spectra to zero.
+ memset(self->mean_far_spectrum, 0,
+ sizeof(SpectrumType) * self->spectrum_size);
+ memset(self->mean_near_spectrum, 0,
+ sizeof(SpectrumType) * self->spectrum_size);
+ // Reset initialization indicators.
+ self->far_spectrum_initialized = 0;
+ self->near_spectrum_initialized = 0;
+
+ return 0;
+}
+
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+ uint16_t* far_spectrum,
+ uint16_t* near_spectrum,
+ int spectrum_size,
+ int far_q,
+ int near_q) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+ uint32_t binary_far_spectrum = 0;
+ uint32_t binary_near_spectrum = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (far_spectrum == NULL) {
+ // Empty far end spectrum.
+ return -1;
+ }
+ if (near_spectrum == NULL) {
+ // Empty near end spectrum.
+ return -1;
+ }
+ if (spectrum_size != self->spectrum_size) {
+ // Data sizes don't match.
+ return -1;
+ }
+ if (far_q > 15) {
+ // If |far_q| is larger than 15 we cannot guarantee no wrap around.
+ return -1;
+ }
+ if (near_q > 15) {
+ // If |near_q| is larger than 15 we cannot guarantee no wrap around.
+ return -1;
+ }
+
+ // Get binary spectra.
+ binary_far_spectrum = BinarySpectrumFix(far_spectrum,
+ self->mean_far_spectrum,
+ far_q,
+ &(self->far_spectrum_initialized));
+ binary_near_spectrum = BinarySpectrumFix(near_spectrum,
+ self->mean_near_spectrum,
+ near_q,
+ &(self->near_spectrum_initialized));
+
+ return WebRtc_ProcessBinarySpectrum(self->binary_handle,
+ binary_far_spectrum,
+ binary_near_spectrum);
+}
+
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+ float* far_spectrum,
+ float* near_spectrum,
+ int spectrum_size) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+ uint32_t binary_far_spectrum = 0;
+ uint32_t binary_near_spectrum = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (far_spectrum == NULL) {
+ // Empty far end spectrum.
+ return -1;
+ }
+ if (near_spectrum == NULL) {
+ // Empty near end spectrum.
+ return -1;
+ }
+ if (spectrum_size != self->spectrum_size) {
+ // Data sizes don't match.
+ return -1;
+ }
+
+ // Get binary spectra.
+ binary_far_spectrum = BinarySpectrumFloat(far_spectrum,
+ self->mean_far_spectrum,
+ &(self->far_spectrum_initialized));
+ binary_near_spectrum = BinarySpectrumFloat(near_spectrum,
+ self->mean_near_spectrum,
+ &(self->near_spectrum_initialized));
+
+ return WebRtc_ProcessBinarySpectrum(self->binary_handle,
+ binary_far_spectrum,
+ binary_near_spectrum);
+}
+
+int WebRtc_last_delay(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ return WebRtc_binary_last_delay(self->binary_handle);
+}
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h
new file mode 100644
index 0000000..2a47b5d
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on block by block basis.
+// The return value is 0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+
+#include "typedefs.h"
+
+// Releases the memory allocated by WebRtc_CreateDelayEstimator(...)
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+int WebRtc_FreeDelayEstimator(void* handle);
+
+// Allocates the memory needed by the delay estimation. The memory needs to be
+// initialized separately through WebRtc_InitDelayEstimator(...).
+//
+// Inputs:
+// - handle : Instance that should be created.
+// - spectrum_size : Size of the spectrum used both in far-end and
+// near-end. Used to allocate memory for spectrum
+// specific buffers.
+// - max_delay : The maximum delay which can be estimated. Needed to
+// allocate memory for history buffers.
+// - lookahead : Amount of non-causal lookahead to use. This can
+// detect cases in which a near-end signal occurs before
+// the corresponding far-end signal. It will delay the
+// estimate for the current block by an equal amount,
+// and the returned values will be offset by it.
+//
+// A value of zero is the typical no-lookahead case.
+// This also represents the minimum delay which can be
+// estimated.
+//
+// Output:
+// - handle : Created instance.
+//
+int WebRtc_CreateDelayEstimator(void** handle,
+ int spectrum_size,
+ int max_delay,
+ int lookahead);
+
+// Initializes the delay estimation instance created with
+// WebRtc_CreateDelayEstimator(...)
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Output:
+// - handle : Initialized instance.
+//
+int WebRtc_InitDelayEstimator(void* handle);
+
+// Estimates and returns the delay between the far-end and near-end blocks. The
+// value will be offset by the lookahead (i.e. the lookahead should be
+// subtracted from the returned value).
+// Inputs:
+// - handle : Pointer to the delay estimation instance.
+// - far_spectrum : Pointer to the far-end spectrum data.
+// - near_spectrum : Pointer to the near-end spectrum data of the current
+// block.
+// - spectrum_size : The size of the data arrays (same for both far- and
+// near-end).
+// - far_q : The Q-domain of the far-end data.
+// - near_q : The Q-domain of the near-end data.
+//
+// Output:
+// - handle : Updated instance.
+//
+// Return value:
+// - delay : >= 0 - Calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+ uint16_t* far_spectrum,
+ uint16_t* near_spectrum,
+ int spectrum_size,
+ int far_q,
+ int near_q);
+
+// See WebRtc_DelayEstimatorProcessFix() for description.
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+ float* far_spectrum,
+ float* near_spectrum,
+ int spectrum_size);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_DelayEstimatorProcess(...).
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - delay : >= 0 - Last calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_last_delay(void* handle);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
diff --git a/trunk/src/modules/audio_processing/utility/fft4g.c b/trunk/src/modules/audio_processing/utility/fft4g.c
new file mode 100644
index 0000000..cbc4dc3
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/fft4g.c
@@ -0,0 +1,1326 @@
+/*
+ * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
+ * Copyright Takuya OOURA, 1996-2001
+ *
+ * You may use, copy, modify and distribute this code for any purpose (include
+ * commercial use) and without fee. Please refer to this package when you modify
+ * this code.
+ *
+ * Changes:
+ * Trivial type modifications by the WebRTC authors.
+ */
+
+/*
+Fast Fourier/Cosine/Sine Transform
+ dimension :one
+ data length :power of 2
+ decimation :frequency
+ radix :4, 2
+ data :inplace
+ table :use
+functions
+ cdft: Complex Discrete Fourier Transform
+ rdft: Real Discrete Fourier Transform
+ ddct: Discrete Cosine Transform
+ ddst: Discrete Sine Transform
+ dfct: Cosine Transform of RDFT (Real Symmetric DFT)
+ dfst: Sine Transform of RDFT (Real Anti-symmetric DFT)
+function prototypes
+ void cdft(int, int, float *, int *, float *);
+ void rdft(int, int, float *, int *, float *);
+ void ddct(int, int, float *, int *, float *);
+ void ddst(int, int, float *, int *, float *);
+ void dfct(int, float *, float *, int *, float *);
+ void dfst(int, float *, float *, int *, float *);
+
+
+-------- Complex DFT (Discrete Fourier Transform) --------
+ [definition]
+ <case1>
+ X[k] = sum_j=0^n-1 x[j]*exp(2*pi*i*j*k/n), 0<=k<n
+ <case2>
+ X[k] = sum_j=0^n-1 x[j]*exp(-2*pi*i*j*k/n), 0<=k<n
+ (notes: sum_j=0^n-1 is a summation from j=0 to n-1)
+ [usage]
+ <case1>
+ ip[0] = 0; // first time only
+ cdft(2*n, 1, a, ip, w);
+ <case2>
+ ip[0] = 0; // first time only
+ cdft(2*n, -1, a, ip, w);
+ [parameters]
+ 2*n :data length (int)
+ n >= 1, n = power of 2
+ a[0...2*n-1] :input/output data (float *)
+ input data
+ a[2*j] = Re(x[j]),
+ a[2*j+1] = Im(x[j]), 0<=j<n
+ output data
+ a[2*k] = Re(X[k]),
+ a[2*k+1] = Im(X[k]), 0<=k<n
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n/2-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ cdft(2*n, -1, a, ip, w);
+ is
+ cdft(2*n, 1, a, ip, w);
+ for (j = 0; j <= 2 * n - 1; j++) {
+ a[j] *= 1.0 / n;
+ }
+ .
+
+
+-------- Real DFT / Inverse of Real DFT --------
+ [definition]
+ <case1> RDFT
+ R[k] = sum_j=0^n-1 a[j]*cos(2*pi*j*k/n), 0<=k<=n/2
+ I[k] = sum_j=0^n-1 a[j]*sin(2*pi*j*k/n), 0<k<n/2
+ <case2> IRDFT (excluding scale)
+ a[k] = (R[0] + R[n/2]*cos(pi*k))/2 +
+ sum_j=1^n/2-1 R[j]*cos(2*pi*j*k/n) +
+ sum_j=1^n/2-1 I[j]*sin(2*pi*j*k/n), 0<=k<n
+ [usage]
+ <case1>
+ ip[0] = 0; // first time only
+ rdft(n, 1, a, ip, w);
+ <case2>
+ ip[0] = 0; // first time only
+ rdft(n, -1, a, ip, w);
+ [parameters]
+ n :data length (int)
+ n >= 2, n = power of 2
+ a[0...n-1] :input/output data (float *)
+ <case1>
+ output data
+ a[2*k] = R[k], 0<=k<n/2
+ a[2*k+1] = I[k], 0<k<n/2
+ a[1] = R[n/2]
+ <case2>
+ input data
+ a[2*j] = R[j], 0<=j<n/2
+ a[2*j+1] = I[j], 0<j<n/2
+ a[1] = R[n/2]
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n/2)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n/2-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ rdft(n, 1, a, ip, w);
+ is
+ rdft(n, -1, a, ip, w);
+ for (j = 0; j <= n - 1; j++) {
+ a[j] *= 2.0 / n;
+ }
+ .
+
+
+-------- DCT (Discrete Cosine Transform) / Inverse of DCT --------
+ [definition]
+ <case1> IDCT (excluding scale)
+ C[k] = sum_j=0^n-1 a[j]*cos(pi*j*(k+1/2)/n), 0<=k<n
+ <case2> DCT
+ C[k] = sum_j=0^n-1 a[j]*cos(pi*(j+1/2)*k/n), 0<=k<n
+ [usage]
+ <case1>
+ ip[0] = 0; // first time only
+ ddct(n, 1, a, ip, w);
+ <case2>
+ ip[0] = 0; // first time only
+ ddct(n, -1, a, ip, w);
+ [parameters]
+ n :data length (int)
+ n >= 2, n = power of 2
+ a[0...n-1] :input/output data (float *)
+ output data
+ a[k] = C[k], 0<=k<n
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n/2)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n*5/4-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ ddct(n, -1, a, ip, w);
+ is
+ a[0] *= 0.5;
+ ddct(n, 1, a, ip, w);
+ for (j = 0; j <= n - 1; j++) {
+ a[j] *= 2.0 / n;
+ }
+ .
+
+
+-------- DST (Discrete Sine Transform) / Inverse of DST --------
+ [definition]
+ <case1> IDST (excluding scale)
+ S[k] = sum_j=1^n A[j]*sin(pi*j*(k+1/2)/n), 0<=k<n
+ <case2> DST
+ S[k] = sum_j=0^n-1 a[j]*sin(pi*(j+1/2)*k/n), 0<k<=n
+ [usage]
+ <case1>
+ ip[0] = 0; // first time only
+ ddst(n, 1, a, ip, w);
+ <case2>
+ ip[0] = 0; // first time only
+ ddst(n, -1, a, ip, w);
+ [parameters]
+ n :data length (int)
+ n >= 2, n = power of 2
+ a[0...n-1] :input/output data (float *)
+ <case1>
+ input data
+ a[j] = A[j], 0<j<n
+ a[0] = A[n]
+ output data
+ a[k] = S[k], 0<=k<n
+ <case2>
+ output data
+ a[k] = S[k], 0<k<n
+ a[0] = S[n]
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n/2)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n*5/4-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ ddst(n, -1, a, ip, w);
+ is
+ a[0] *= 0.5;
+ ddst(n, 1, a, ip, w);
+ for (j = 0; j <= n - 1; j++) {
+ a[j] *= 2.0 / n;
+ }
+ .
+
+
+-------- Cosine Transform of RDFT (Real Symmetric DFT) --------
+ [definition]
+ C[k] = sum_j=0^n a[j]*cos(pi*j*k/n), 0<=k<=n
+ [usage]
+ ip[0] = 0; // first time only
+ dfct(n, a, t, ip, w);
+ [parameters]
+ n :data length - 1 (int)
+ n >= 2, n = power of 2
+ a[0...n] :input/output data (float *)
+ output data
+ a[k] = C[k], 0<=k<=n
+ t[0...n/2] :work area (float *)
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n/4)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n/4+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n*5/8-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ a[0] *= 0.5;
+ a[n] *= 0.5;
+ dfct(n, a, t, ip, w);
+ is
+ a[0] *= 0.5;
+ a[n] *= 0.5;
+ dfct(n, a, t, ip, w);
+ for (j = 0; j <= n; j++) {
+ a[j] *= 2.0 / n;
+ }
+ .
+
+
+-------- Sine Transform of RDFT (Real Anti-symmetric DFT) --------
+ [definition]
+ S[k] = sum_j=1^n-1 a[j]*sin(pi*j*k/n), 0<k<n
+ [usage]
+ ip[0] = 0; // first time only
+ dfst(n, a, t, ip, w);
+ [parameters]
+ n :data length + 1 (int)
+ n >= 2, n = power of 2
+ a[0...n-1] :input/output data (float *)
+ output data
+ a[k] = S[k], 0<k<n
+ (a[0] is used for work area)
+ t[0...n/2-1] :work area (float *)
+ ip[0...*] :work area for bit reversal (int *)
+ length of ip >= 2+sqrt(n/4)
+ strictly,
+ length of ip >=
+ 2+(1<<(int)(log(n/4+0.5)/log(2))/2).
+ ip[0],ip[1] are pointers of the cos/sin table.
+ w[0...n*5/8-1] :cos/sin table (float *)
+ w[],ip[] are initialized if ip[0] == 0.
+ [remark]
+ Inverse of
+ dfst(n, a, t, ip, w);
+ is
+ dfst(n, a, t, ip, w);
+ for (j = 1; j <= n - 1; j++) {
+ a[j] *= 2.0 / n;
+ }
+ .
+
+
+Appendix :
+ The cos/sin table is recalculated when the larger table required.
+ w[] and ip[] are compatible with all routines.
+*/
+
+static void makewt(int nw, int *ip, float *w);
+static void makect(int nc, int *ip, float *c);
+static void bitrv2(int n, int *ip, float *a);
+static void bitrv2conj(int n, int *ip, float *a);
+static void cftfsub(int n, float *a, float *w);
+static void cftbsub(int n, float *a, float *w);
+static void cft1st(int n, float *a, float *w);
+static void cftmdl(int n, int l, float *a, float *w);
+static void rftfsub(int n, float *a, int nc, float *c);
+static void rftbsub(int n, float *a, int nc, float *c);
+#if 0 // Not used.
+static void dctsub(int n, float *a, int nc, float *c)
+static void dstsub(int n, float *a, int nc, float *c)
+#endif
+
+
+void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w)
+{
+ if (n > (ip[0] << 2)) {
+ makewt(n >> 2, ip, w);
+ }
+ if (n > 4) {
+ if (isgn >= 0) {
+ bitrv2(n, ip + 2, a);
+ cftfsub(n, a, w);
+ } else {
+ bitrv2conj(n, ip + 2, a);
+ cftbsub(n, a, w);
+ }
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+}
+
+
+void WebRtc_rdft(int n, int isgn, float *a, int *ip, float *w)
+{
+ int nw, nc;
+ float xi;
+
+ nw = ip[0];
+ if (n > (nw << 2)) {
+ nw = n >> 2;
+ makewt(nw, ip, w);
+ }
+ nc = ip[1];
+ if (n > (nc << 2)) {
+ nc = n >> 2;
+ makect(nc, ip, w + nw);
+ }
+ if (isgn >= 0) {
+ if (n > 4) {
+ bitrv2(n, ip + 2, a);
+ cftfsub(n, a, w);
+ rftfsub(n, a, nc, w + nw);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ xi = a[0] - a[1];
+ a[0] += a[1];
+ a[1] = xi;
+ } else {
+ a[1] = 0.5f * (a[0] - a[1]);
+ a[0] -= a[1];
+ if (n > 4) {
+ rftbsub(n, a, nc, w + nw);
+ bitrv2(n, ip + 2, a);
+ cftbsub(n, a, w);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ }
+}
+
+#if 0 // Not used.
+static void ddct(int n, int isgn, float *a, int *ip, float *w)
+{
+ int j, nw, nc;
+ float xr;
+
+ nw = ip[0];
+ if (n > (nw << 2)) {
+ nw = n >> 2;
+ makewt(nw, ip, w);
+ }
+ nc = ip[1];
+ if (n > nc) {
+ nc = n;
+ makect(nc, ip, w + nw);
+ }
+ if (isgn < 0) {
+ xr = a[n - 1];
+ for (j = n - 2; j >= 2; j -= 2) {
+ a[j + 1] = a[j] - a[j - 1];
+ a[j] += a[j - 1];
+ }
+ a[1] = a[0] - xr;
+ a[0] += xr;
+ if (n > 4) {
+ rftbsub(n, a, nc, w + nw);
+ bitrv2(n, ip + 2, a);
+ cftbsub(n, a, w);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ }
+ dctsub(n, a, nc, w + nw);
+ if (isgn >= 0) {
+ if (n > 4) {
+ bitrv2(n, ip + 2, a);
+ cftfsub(n, a, w);
+ rftfsub(n, a, nc, w + nw);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ xr = a[0] - a[1];
+ a[0] += a[1];
+ for (j = 2; j < n; j += 2) {
+ a[j - 1] = a[j] - a[j + 1];
+ a[j] += a[j + 1];
+ }
+ a[n - 1] = xr;
+ }
+}
+
+
+static void ddst(int n, int isgn, float *a, int *ip, float *w)
+{
+ int j, nw, nc;
+ float xr;
+
+ nw = ip[0];
+ if (n > (nw << 2)) {
+ nw = n >> 2;
+ makewt(nw, ip, w);
+ }
+ nc = ip[1];
+ if (n > nc) {
+ nc = n;
+ makect(nc, ip, w + nw);
+ }
+ if (isgn < 0) {
+ xr = a[n - 1];
+ for (j = n - 2; j >= 2; j -= 2) {
+ a[j + 1] = -a[j] - a[j - 1];
+ a[j] -= a[j - 1];
+ }
+ a[1] = a[0] + xr;
+ a[0] -= xr;
+ if (n > 4) {
+ rftbsub(n, a, nc, w + nw);
+ bitrv2(n, ip + 2, a);
+ cftbsub(n, a, w);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ }
+ dstsub(n, a, nc, w + nw);
+ if (isgn >= 0) {
+ if (n > 4) {
+ bitrv2(n, ip + 2, a);
+ cftfsub(n, a, w);
+ rftfsub(n, a, nc, w + nw);
+ } else if (n == 4) {
+ cftfsub(n, a, w);
+ }
+ xr = a[0] - a[1];
+ a[0] += a[1];
+ for (j = 2; j < n; j += 2) {
+ a[j - 1] = -a[j] - a[j + 1];
+ a[j] -= a[j + 1];
+ }
+ a[n - 1] = -xr;
+ }
+}
+
+
+static void dfct(int n, float *a, float *t, int *ip, float *w)
+{
+ int j, k, l, m, mh, nw, nc;
+ float xr, xi, yr, yi;
+
+ nw = ip[0];
+ if (n > (nw << 3)) {
+ nw = n >> 3;
+ makewt(nw, ip, w);
+ }
+ nc = ip[1];
+ if (n > (nc << 1)) {
+ nc = n >> 1;
+ makect(nc, ip, w + nw);
+ }
+ m = n >> 1;
+ yi = a[m];
+ xi = a[0] + a[n];
+ a[0] -= a[n];
+ t[0] = xi - yi;
+ t[m] = xi + yi;
+ if (n > 2) {
+ mh = m >> 1;
+ for (j = 1; j < mh; j++) {
+ k = m - j;
+ xr = a[j] - a[n - j];
+ xi = a[j] + a[n - j];
+ yr = a[k] - a[n - k];
+ yi = a[k] + a[n - k];
+ a[j] = xr;
+ a[k] = yr;
+ t[j] = xi - yi;
+ t[k] = xi + yi;
+ }
+ t[mh] = a[mh] + a[n - mh];
+ a[mh] -= a[n - mh];
+ dctsub(m, a, nc, w + nw);
+ if (m > 4) {
+ bitrv2(m, ip + 2, a);
+ cftfsub(m, a, w);
+ rftfsub(m, a, nc, w + nw);
+ } else if (m == 4) {
+ cftfsub(m, a, w);
+ }
+ a[n - 1] = a[0] - a[1];
+ a[1] = a[0] + a[1];
+ for (j = m - 2; j >= 2; j -= 2) {
+ a[2 * j + 1] = a[j] + a[j + 1];
+ a[2 * j - 1] = a[j] - a[j + 1];
+ }
+ l = 2;
+ m = mh;
+ while (m >= 2) {
+ dctsub(m, t, nc, w + nw);
+ if (m > 4) {
+ bitrv2(m, ip + 2, t);
+ cftfsub(m, t, w);
+ rftfsub(m, t, nc, w + nw);
+ } else if (m == 4) {
+ cftfsub(m, t, w);
+ }
+ a[n - l] = t[0] - t[1];
+ a[l] = t[0] + t[1];
+ k = 0;
+ for (j = 2; j < m; j += 2) {
+ k += l << 2;
+ a[k - l] = t[j] - t[j + 1];
+ a[k + l] = t[j] + t[j + 1];
+ }
+ l <<= 1;
+ mh = m >> 1;
+ for (j = 0; j < mh; j++) {
+ k = m - j;
+ t[j] = t[m + k] - t[m + j];
+ t[k] = t[m + k] + t[m + j];
+ }
+ t[mh] = t[m + mh];
+ m = mh;
+ }
+ a[l] = t[0];
+ a[n] = t[2] - t[1];
+ a[0] = t[2] + t[1];
+ } else {
+ a[1] = a[0];
+ a[2] = t[0];
+ a[0] = t[1];
+ }
+}
+
+static void dfst(int n, float *a, float *t, int *ip, float *w)
+{
+ int j, k, l, m, mh, nw, nc;
+ float xr, xi, yr, yi;
+
+ nw = ip[0];
+ if (n > (nw << 3)) {
+ nw = n >> 3;
+ makewt(nw, ip, w);
+ }
+ nc = ip[1];
+ if (n > (nc << 1)) {
+ nc = n >> 1;
+ makect(nc, ip, w + nw);
+ }
+ if (n > 2) {
+ m = n >> 1;
+ mh = m >> 1;
+ for (j = 1; j < mh; j++) {
+ k = m - j;
+ xr = a[j] + a[n - j];
+ xi = a[j] - a[n - j];
+ yr = a[k] + a[n - k];
+ yi = a[k] - a[n - k];
+ a[j] = xr;
+ a[k] = yr;
+ t[j] = xi + yi;
+ t[k] = xi - yi;
+ }
+ t[0] = a[mh] - a[n - mh];
+ a[mh] += a[n - mh];
+ a[0] = a[m];
+ dstsub(m, a, nc, w + nw);
+ if (m > 4) {
+ bitrv2(m, ip + 2, a);
+ cftfsub(m, a, w);
+ rftfsub(m, a, nc, w + nw);
+ } else if (m == 4) {
+ cftfsub(m, a, w);
+ }
+ a[n - 1] = a[1] - a[0];
+ a[1] = a[0] + a[1];
+ for (j = m - 2; j >= 2; j -= 2) {
+ a[2 * j + 1] = a[j] - a[j + 1];
+ a[2 * j - 1] = -a[j] - a[j + 1];
+ }
+ l = 2;
+ m = mh;
+ while (m >= 2) {
+ dstsub(m, t, nc, w + nw);
+ if (m > 4) {
+ bitrv2(m, ip + 2, t);
+ cftfsub(m, t, w);
+ rftfsub(m, t, nc, w + nw);
+ } else if (m == 4) {
+ cftfsub(m, t, w);
+ }
+ a[n - l] = t[1] - t[0];
+ a[l] = t[0] + t[1];
+ k = 0;
+ for (j = 2; j < m; j += 2) {
+ k += l << 2;
+ a[k - l] = -t[j] - t[j + 1];
+ a[k + l] = t[j] - t[j + 1];
+ }
+ l <<= 1;
+ mh = m >> 1;
+ for (j = 1; j < mh; j++) {
+ k = m - j;
+ t[j] = t[m + k] + t[m + j];
+ t[k] = t[m + k] - t[m + j];
+ }
+ t[0] = t[m + mh];
+ m = mh;
+ }
+ a[l] = t[0];
+ }
+ a[0] = 0;
+}
+#endif // Not used.
+
+
+/* -------- initializing routines -------- */
+
+
+#include <math.h>
+
+static void makewt(int nw, int *ip, float *w)
+{
+ int j, nwh;
+ float delta, x, y;
+
+ ip[0] = nw;
+ ip[1] = 1;
+ if (nw > 2) {
+ nwh = nw >> 1;
+ delta = (float)atan(1.0f) / nwh;
+ w[0] = 1;
+ w[1] = 0;
+ w[nwh] = (float)cos(delta * nwh);
+ w[nwh + 1] = w[nwh];
+ if (nwh > 2) {
+ for (j = 2; j < nwh; j += 2) {
+ x = (float)cos(delta * j);
+ y = (float)sin(delta * j);
+ w[j] = x;
+ w[j + 1] = y;
+ w[nw - j] = y;
+ w[nw - j + 1] = x;
+ }
+ bitrv2(nw, ip + 2, w);
+ }
+ }
+}
+
+
+static void makect(int nc, int *ip, float *c)
+{
+ int j, nch;
+ float delta;
+
+ ip[1] = nc;
+ if (nc > 1) {
+ nch = nc >> 1;
+ delta = (float)atan(1.0f) / nch;
+ c[0] = (float)cos(delta * nch);
+ c[nch] = 0.5f * c[0];
+ for (j = 1; j < nch; j++) {
+ c[j] = 0.5f * (float)cos(delta * j);
+ c[nc - j] = 0.5f * (float)sin(delta * j);
+ }
+ }
+}
+
+
+/* -------- child routines -------- */
+
+
+static void bitrv2(int n, int *ip, float *a)
+{
+ int j, j1, k, k1, l, m, m2;
+ float xr, xi, yr, yi;
+
+ ip[0] = 0;
+ l = n;
+ m = 1;
+ while ((m << 3) < l) {
+ l >>= 1;
+ for (j = 0; j < m; j++) {
+ ip[m + j] = ip[j] + l;
+ }
+ m <<= 1;
+ }
+ m2 = 2 * m;
+ if ((m << 3) == l) {
+ for (k = 0; k < m; k++) {
+ for (j = 0; j < k; j++) {
+ j1 = 2 * j + ip[k];
+ k1 = 2 * k + ip[j];
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 -= m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ j1 = 2 * k + m2 + ip[k];
+ k1 = j1 + m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ } else {
+ for (k = 1; k < m; k++) {
+ for (j = 0; j < k; j++) {
+ j1 = 2 * j + ip[k];
+ k1 = 2 * k + ip[j];
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += m2;
+ xr = a[j1];
+ xi = a[j1 + 1];
+ yr = a[k1];
+ yi = a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ }
+ }
+}
+
+
+static void bitrv2conj(int n, int *ip, float *a)
+{
+ int j, j1, k, k1, l, m, m2;
+ float xr, xi, yr, yi;
+
+ ip[0] = 0;
+ l = n;
+ m = 1;
+ while ((m << 3) < l) {
+ l >>= 1;
+ for (j = 0; j < m; j++) {
+ ip[m + j] = ip[j] + l;
+ }
+ m <<= 1;
+ }
+ m2 = 2 * m;
+ if ((m << 3) == l) {
+ for (k = 0; k < m; k++) {
+ for (j = 0; j < k; j++) {
+ j1 = 2 * j + ip[k];
+ k1 = 2 * k + ip[j];
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 -= m2;
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += 2 * m2;
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ k1 = 2 * k + ip[k];
+ a[k1 + 1] = -a[k1 + 1];
+ j1 = k1 + m2;
+ k1 = j1 + m2;
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ k1 += m2;
+ a[k1 + 1] = -a[k1 + 1];
+ }
+ } else {
+ a[1] = -a[1];
+ a[m2 + 1] = -a[m2 + 1];
+ for (k = 1; k < m; k++) {
+ for (j = 0; j < k; j++) {
+ j1 = 2 * j + ip[k];
+ k1 = 2 * k + ip[j];
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ j1 += m2;
+ k1 += m2;
+ xr = a[j1];
+ xi = -a[j1 + 1];
+ yr = a[k1];
+ yi = -a[k1 + 1];
+ a[j1] = yr;
+ a[j1 + 1] = yi;
+ a[k1] = xr;
+ a[k1 + 1] = xi;
+ }
+ k1 = 2 * k + ip[k];
+ a[k1 + 1] = -a[k1 + 1];
+ a[k1 + m2 + 1] = -a[k1 + m2 + 1];
+ }
+ }
+}
+
+
+static void cftfsub(int n, float *a, float *w)
+{
+ int j, j1, j2, j3, l;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ l = 2;
+ if (n > 8) {
+ cft1st(n, a, w);
+ l = 8;
+ while ((l << 2) < n) {
+ cftmdl(n, l, a, w);
+ l <<= 2;
+ }
+ }
+ if ((l << 2) == n) {
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ a[j2] = x0r - x2r;
+ a[j2 + 1] = x0i - x2i;
+ a[j1] = x1r - x3i;
+ a[j1 + 1] = x1i + x3r;
+ a[j3] = x1r + x3i;
+ a[j3 + 1] = x1i - x3r;
+ }
+ } else {
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ x0r = a[j] - a[j1];
+ x0i = a[j + 1] - a[j1 + 1];
+ a[j] += a[j1];
+ a[j + 1] += a[j1 + 1];
+ a[j1] = x0r;
+ a[j1 + 1] = x0i;
+ }
+ }
+}
+
+
+static void cftbsub(int n, float *a, float *w)
+{
+ int j, j1, j2, j3, l;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ l = 2;
+ if (n > 8) {
+ cft1st(n, a, w);
+ l = 8;
+ while ((l << 2) < n) {
+ cftmdl(n, l, a, w);
+ l <<= 2;
+ }
+ }
+ if ((l << 2) == n) {
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = -a[j + 1] - a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = -a[j + 1] + a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i - x2i;
+ a[j2] = x0r - x2r;
+ a[j2 + 1] = x0i + x2i;
+ a[j1] = x1r - x3i;
+ a[j1 + 1] = x1i - x3r;
+ a[j3] = x1r + x3i;
+ a[j3 + 1] = x1i + x3r;
+ }
+ } else {
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ x0r = a[j] - a[j1];
+ x0i = -a[j + 1] + a[j1 + 1];
+ a[j] += a[j1];
+ a[j + 1] = -a[j + 1] - a[j1 + 1];
+ a[j1] = x0r;
+ a[j1 + 1] = x0i;
+ }
+ }
+}
+
+
+static void cft1st(int n, float *a, float *w)
+{
+ int j, k1, k2;
+ float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ x0r = a[0] + a[2];
+ x0i = a[1] + a[3];
+ x1r = a[0] - a[2];
+ x1i = a[1] - a[3];
+ x2r = a[4] + a[6];
+ x2i = a[5] + a[7];
+ x3r = a[4] - a[6];
+ x3i = a[5] - a[7];
+ a[0] = x0r + x2r;
+ a[1] = x0i + x2i;
+ a[4] = x0r - x2r;
+ a[5] = x0i - x2i;
+ a[2] = x1r - x3i;
+ a[3] = x1i + x3r;
+ a[6] = x1r + x3i;
+ a[7] = x1i - x3r;
+ wk1r = w[2];
+ x0r = a[8] + a[10];
+ x0i = a[9] + a[11];
+ x1r = a[8] - a[10];
+ x1i = a[9] - a[11];
+ x2r = a[12] + a[14];
+ x2i = a[13] + a[15];
+ x3r = a[12] - a[14];
+ x3i = a[13] - a[15];
+ a[8] = x0r + x2r;
+ a[9] = x0i + x2i;
+ a[12] = x2i - x0i;
+ a[13] = x0r - x2r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[10] = wk1r * (x0r - x0i);
+ a[11] = wk1r * (x0r + x0i);
+ x0r = x3i + x1r;
+ x0i = x3r - x1i;
+ a[14] = wk1r * (x0i - x0r);
+ a[15] = wk1r * (x0i + x0r);
+ k1 = 0;
+ for (j = 16; j < n; j += 16) {
+ k1 += 2;
+ k2 = 2 * k1;
+ wk2r = w[k1];
+ wk2i = w[k1 + 1];
+ wk1r = w[k2];
+ wk1i = w[k2 + 1];
+ wk3r = wk1r - 2 * wk2i * wk1i;
+ wk3i = 2 * wk2i * wk1r - wk1i;
+ x0r = a[j] + a[j + 2];
+ x0i = a[j + 1] + a[j + 3];
+ x1r = a[j] - a[j + 2];
+ x1i = a[j + 1] - a[j + 3];
+ x2r = a[j + 4] + a[j + 6];
+ x2i = a[j + 5] + a[j + 7];
+ x3r = a[j + 4] - a[j + 6];
+ x3i = a[j + 5] - a[j + 7];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j + 4] = wk2r * x0r - wk2i * x0i;
+ a[j + 5] = wk2r * x0i + wk2i * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j + 2] = wk1r * x0r - wk1i * x0i;
+ a[j + 3] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j + 6] = wk3r * x0r - wk3i * x0i;
+ a[j + 7] = wk3r * x0i + wk3i * x0r;
+ wk1r = w[k2 + 2];
+ wk1i = w[k2 + 3];
+ wk3r = wk1r - 2 * wk2r * wk1i;
+ wk3i = 2 * wk2r * wk1r - wk1i;
+ x0r = a[j + 8] + a[j + 10];
+ x0i = a[j + 9] + a[j + 11];
+ x1r = a[j + 8] - a[j + 10];
+ x1i = a[j + 9] - a[j + 11];
+ x2r = a[j + 12] + a[j + 14];
+ x2i = a[j + 13] + a[j + 15];
+ x3r = a[j + 12] - a[j + 14];
+ x3i = a[j + 13] - a[j + 15];
+ a[j + 8] = x0r + x2r;
+ a[j + 9] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j + 12] = -wk2i * x0r - wk2r * x0i;
+ a[j + 13] = -wk2i * x0i + wk2r * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j + 10] = wk1r * x0r - wk1i * x0i;
+ a[j + 11] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j + 14] = wk3r * x0r - wk3i * x0i;
+ a[j + 15] = wk3r * x0i + wk3i * x0r;
+ }
+}
+
+
+static void cftmdl(int n, int l, float *a, float *w)
+{
+ int j, j1, j2, j3, k, k1, k2, m, m2;
+ float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
+ float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
+
+ m = l << 2;
+ for (j = 0; j < l; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ a[j2] = x0r - x2r;
+ a[j2 + 1] = x0i - x2i;
+ a[j1] = x1r - x3i;
+ a[j1 + 1] = x1i + x3r;
+ a[j3] = x1r + x3i;
+ a[j3 + 1] = x1i - x3r;
+ }
+ wk1r = w[2];
+ for (j = m; j < l + m; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ a[j2] = x2i - x0i;
+ a[j2 + 1] = x0r - x2r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1] = wk1r * (x0r - x0i);
+ a[j1 + 1] = wk1r * (x0r + x0i);
+ x0r = x3i + x1r;
+ x0i = x3r - x1i;
+ a[j3] = wk1r * (x0i - x0r);
+ a[j3 + 1] = wk1r * (x0i + x0r);
+ }
+ k1 = 0;
+ m2 = 2 * m;
+ for (k = m2; k < n; k += m2) {
+ k1 += 2;
+ k2 = 2 * k1;
+ wk2r = w[k1];
+ wk2i = w[k1 + 1];
+ wk1r = w[k2];
+ wk1i = w[k2 + 1];
+ wk3r = wk1r - 2 * wk2i * wk1i;
+ wk3i = 2 * wk2i * wk1r - wk1i;
+ for (j = k; j < l + k; j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j2] = wk2r * x0r - wk2i * x0i;
+ a[j2 + 1] = wk2r * x0i + wk2i * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 1] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j3] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 1] = wk3r * x0i + wk3i * x0r;
+ }
+ wk1r = w[k2 + 2];
+ wk1i = w[k2 + 3];
+ wk3r = wk1r - 2 * wk2r * wk1i;
+ wk3i = 2 * wk2r * wk1r - wk1i;
+ for (j = k + m; j < l + (k + m); j += 2) {
+ j1 = j + l;
+ j2 = j1 + l;
+ j3 = j2 + l;
+ x0r = a[j] + a[j1];
+ x0i = a[j + 1] + a[j1 + 1];
+ x1r = a[j] - a[j1];
+ x1i = a[j + 1] - a[j1 + 1];
+ x2r = a[j2] + a[j3];
+ x2i = a[j2 + 1] + a[j3 + 1];
+ x3r = a[j2] - a[j3];
+ x3i = a[j2 + 1] - a[j3 + 1];
+ a[j] = x0r + x2r;
+ a[j + 1] = x0i + x2i;
+ x0r -= x2r;
+ x0i -= x2i;
+ a[j2] = -wk2i * x0r - wk2r * x0i;
+ a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
+ x0r = x1r - x3i;
+ x0i = x1i + x3r;
+ a[j1] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 1] = wk1r * x0i + wk1i * x0r;
+ x0r = x1r + x3i;
+ x0i = x1i - x3r;
+ a[j3] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 1] = wk3r * x0i + wk3i * x0r;
+ }
+ }
+}
+
+
+static void rftfsub(int n, float *a, int nc, float *c)
+{
+ int j, k, kk, ks, m;
+ float wkr, wki, xr, xi, yr, yi;
+
+ m = n >> 1;
+ ks = 2 * nc / m;
+ kk = 0;
+ for (j = 2; j < m; j += 2) {
+ k = n - j;
+ kk += ks;
+ wkr = 0.5f - c[nc - kk];
+ wki = c[kk];
+ xr = a[j] - a[k];
+ xi = a[j + 1] + a[k + 1];
+ yr = wkr * xr - wki * xi;
+ yi = wkr * xi + wki * xr;
+ a[j] -= yr;
+ a[j + 1] -= yi;
+ a[k] += yr;
+ a[k + 1] -= yi;
+ }
+}
+
+
+static void rftbsub(int n, float *a, int nc, float *c)
+{
+ int j, k, kk, ks, m;
+ float wkr, wki, xr, xi, yr, yi;
+
+ a[1] = -a[1];
+ m = n >> 1;
+ ks = 2 * nc / m;
+ kk = 0;
+ for (j = 2; j < m; j += 2) {
+ k = n - j;
+ kk += ks;
+ wkr = 0.5f - c[nc - kk];
+ wki = c[kk];
+ xr = a[j] - a[k];
+ xi = a[j + 1] + a[k + 1];
+ yr = wkr * xr + wki * xi;
+ yi = wkr * xi - wki * xr;
+ a[j] -= yr;
+ a[j + 1] = yi - a[j + 1];
+ a[k] += yr;
+ a[k + 1] = yi - a[k + 1];
+ }
+ a[m + 1] = -a[m + 1];
+}
+
+#if 0 // Not used.
+static void dctsub(int n, float *a, int nc, float *c)
+{
+ int j, k, kk, ks, m;
+ float wkr, wki, xr;
+
+ m = n >> 1;
+ ks = nc / n;
+ kk = 0;
+ for (j = 1; j < m; j++) {
+ k = n - j;
+ kk += ks;
+ wkr = c[kk] - c[nc - kk];
+ wki = c[kk] + c[nc - kk];
+ xr = wki * a[j] - wkr * a[k];
+ a[j] = wkr * a[j] + wki * a[k];
+ a[k] = xr;
+ }
+ a[m] *= c[0];
+}
+
+
+static void dstsub(int n, float *a, int nc, float *c)
+{
+ int j, k, kk, ks, m;
+ float wkr, wki, xr;
+
+ m = n >> 1;
+ ks = nc / n;
+ kk = 0;
+ for (j = 1; j < m; j++) {
+ k = n - j;
+ kk += ks;
+ wkr = c[kk] - c[nc - kk];
+ wki = c[kk] + c[nc - kk];
+ xr = wki * a[k] - wkr * a[j];
+ a[k] = wkr * a[k] + wki * a[j];
+ a[j] = xr;
+ }
+ a[m] *= c[0];
+}
+#endif // Not used.
diff --git a/trunk/src/modules/audio_processing/utility/fft4g.h b/trunk/src/modules/audio_processing/utility/fft4g.h
new file mode 100644
index 0000000..14a52a1
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/fft4g.h
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
+
+void WebRtc_rdft(int, int, float *, int *, float *);
+void WebRtc_cdft(int, int, float *, int *, float *);
+
+#endif
diff --git a/trunk/src/modules/audio_processing/utility/ring_buffer.c b/trunk/src/modules/audio_processing/utility/ring_buffer.c
new file mode 100644
index 0000000..8b2b436
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/ring_buffer.c
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
+// otherwise specified, functions return 0 on success and -1 on error.
+
+#include "ring_buffer.h"
+
+#include <stddef.h> // size_t
+#include <stdlib.h>
+#include <string.h>
+
+enum Wrap {
+ SAME_WRAP,
+ DIFF_WRAP
+};
+
+typedef struct {
+ size_t read_pos;
+ size_t write_pos;
+ size_t element_count;
+ size_t element_size;
+ enum Wrap rw_wrap;
+ char* data;
+} buf_t;
+
+// Get address of region(s) from which we can read data.
+// If the region is contiguous, |data_ptr_bytes_2| will be zero.
+// If non-contiguous, |data_ptr_bytes_2| will be the size in bytes of the second
+// region. Returns room available to be read or |element_count|, whichever is
+// smaller.
+static size_t GetBufferReadRegions(buf_t* buf,
+ size_t element_count,
+ void** data_ptr_1,
+ size_t* data_ptr_bytes_1,
+ void** data_ptr_2,
+ size_t* data_ptr_bytes_2) {
+
+ const size_t readable_elements = WebRtc_available_read(buf);
+ const size_t read_elements = (readable_elements < element_count ?
+ readable_elements : element_count);
+ const size_t margin = buf->element_count - buf->read_pos;
+
+ // Check to see if read is not contiguous.
+ if (read_elements > margin) {
+ // Write data in two blocks that wrap the buffer.
+ *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
+ *data_ptr_bytes_1 = margin * buf->element_size;
+ *data_ptr_2 = buf->data;
+ *data_ptr_bytes_2 = (read_elements - margin) * buf->element_size;
+ } else {
+ *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
+ *data_ptr_bytes_1 = read_elements * buf->element_size;
+ *data_ptr_2 = NULL;
+ *data_ptr_bytes_2 = 0;
+ }
+
+ return read_elements;
+}
+
+int WebRtc_CreateBuffer(void** handle,
+ size_t element_count,
+ size_t element_size) {
+ buf_t* self = NULL;
+
+ if (handle == NULL) {
+ return -1;
+ }
+
+ self = malloc(sizeof(buf_t));
+ if (self == NULL) {
+ return -1;
+ }
+ *handle = self;
+
+ self->data = malloc(element_count * element_size);
+ if (self->data == NULL) {
+ free(self);
+ self = NULL;
+ return -1;
+ }
+
+ self->element_count = element_count;
+ self->element_size = element_size;
+
+ return 0;
+}
+
+int WebRtc_InitBuffer(void* handle) {
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->read_pos = 0;
+ self->write_pos = 0;
+ self->rw_wrap = SAME_WRAP;
+
+ // Initialize buffer to zeros
+ memset(self->data, 0, self->element_count * self->element_size);
+
+ return 0;
+}
+
+int WebRtc_FreeBuffer(void* handle) {
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ free(self->data);
+ free(self);
+
+ return 0;
+}
+
+size_t WebRtc_ReadBuffer(void* handle,
+ void** data_ptr,
+ void* data,
+ size_t element_count) {
+
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+ if (data == NULL) {
+ return 0;
+ }
+ if (data_ptr == NULL) {
+ return 0;
+ }
+
+ {
+ void* buf_ptr_1 = NULL;
+ void* buf_ptr_2 = NULL;
+ size_t buf_ptr_bytes_1 = 0;
+ size_t buf_ptr_bytes_2 = 0;
+ const size_t read_count = GetBufferReadRegions(self,
+ element_count,
+ &buf_ptr_1,
+ &buf_ptr_bytes_1,
+ &buf_ptr_2,
+ &buf_ptr_bytes_2);
+
+ if (buf_ptr_bytes_2 > 0) {
+ // We have a wrap around when reading the buffer. Copy the buffer data to
+ // |data| and point to it.
+ memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
+ memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
+ *data_ptr = data;
+ } else {
+ *data_ptr = buf_ptr_1;
+ }
+
+ // Update read position
+ WebRtc_MoveReadPtr(handle, (int) read_count);
+
+ return read_count;
+ }
+}
+
+size_t WebRtc_WriteBuffer(void* handle,
+ const void* data,
+ size_t element_count) {
+
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+ if (data == NULL) {
+ return 0;
+ }
+
+ {
+ const size_t free_elements = WebRtc_available_write(handle);
+ const size_t write_elements = (free_elements < element_count ? free_elements
+ : element_count);
+ size_t n = write_elements;
+ const size_t margin = self->element_count - self->write_pos;
+
+ if (write_elements > margin) {
+ // Buffer wrap around when writing.
+ memcpy(self->data + self->write_pos * self->element_size,
+ data, margin * self->element_size);
+ self->write_pos = 0;
+ n -= margin;
+ self->rw_wrap = DIFF_WRAP;
+ }
+ memcpy(self->data + self->write_pos * self->element_size,
+ ((const char*) data) + ((write_elements - n) * self->element_size),
+ n * self->element_size);
+ self->write_pos += n;
+
+ return write_elements;
+ }
+}
+
+int WebRtc_MoveReadPtr(void* handle, int element_count) {
+
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+
+ {
+ // We need to be able to take care of negative changes, hence use "int"
+ // instead of "size_t".
+ const int free_elements = (int) WebRtc_available_write(handle);
+ const int readable_elements = (int) WebRtc_available_read(handle);
+ int read_pos = (int) self->read_pos;
+
+ if (element_count > readable_elements) {
+ element_count = readable_elements;
+ }
+ if (element_count < -free_elements) {
+ element_count = -free_elements;
+ }
+
+ read_pos += element_count;
+ if (read_pos > (int) self->element_count) {
+ // Buffer wrap around. Restart read position and wrap indicator.
+ read_pos -= (int) self->element_count;
+ self->rw_wrap = SAME_WRAP;
+ }
+ if (read_pos < 0) {
+ // Buffer wrap around. Restart read position and wrap indicator.
+ read_pos += (int) self->element_count;
+ self->rw_wrap = DIFF_WRAP;
+ }
+
+ self->read_pos = (size_t) read_pos;
+
+ return element_count;
+ }
+}
+
+size_t WebRtc_available_read(const void* handle) {
+ const buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+
+ if (self->rw_wrap == SAME_WRAP) {
+ return self->write_pos - self->read_pos;
+ } else {
+ return self->element_count - self->read_pos + self->write_pos;
+ }
+}
+
+size_t WebRtc_available_write(const void* handle) {
+ const buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+
+ return self->element_count - WebRtc_available_read(handle);
+}
diff --git a/trunk/src/modules/audio_processing/utility/ring_buffer.h b/trunk/src/modules/audio_processing/utility/ring_buffer.h
new file mode 100644
index 0000000..3c44029
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/ring_buffer.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
+// otherwise specified, functions return 0 on success and -1 on error.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+
+#include <stddef.h> // size_t
+
+int WebRtc_CreateBuffer(void** handle,
+ size_t element_count,
+ size_t element_size);
+int WebRtc_InitBuffer(void* handle);
+int WebRtc_FreeBuffer(void* handle);
+
+// Reads data from the buffer. The |data_ptr| will point to the address where
+// it is located. If all |element_count| data are feasible to read without
+// buffer wrap around |data_ptr| will point to the location in the buffer.
+// Otherwise, the data will be copied to |data| (memory allocation done by the
+// user) and |data_ptr| points to the address of |data|. |data_ptr| is only
+// guaranteed to be valid until the next call to WebRtc_WriteBuffer().
+// Returns number of elements read.
+size_t WebRtc_ReadBuffer(void* handle,
+ void** data_ptr,
+ void* data,
+ size_t element_count);
+
+// Writes |data| to buffer and returns the number of elements written.
+size_t WebRtc_WriteBuffer(void* handle, const void* data, size_t element_count);
+
+// Moves the buffer read position and returns the number of elements moved.
+// Positive |element_count| moves the read position towards the write position,
+// that is, flushing the buffer. Negative |element_count| moves the read
+// position away from the the write position, that is, stuffing the buffer.
+// Returns number of elements moved.
+int WebRtc_MoveReadPtr(void* handle, int element_count);
+
+// Returns number of available elements to read.
+size_t WebRtc_available_read(const void* handle);
+
+// Returns number of available elements for write.
+size_t WebRtc_available_write(const void* handle);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
diff --git a/trunk/src/modules/audio_processing/utility/util.gypi b/trunk/src/modules/audio_processing/utility/util.gypi
new file mode 100644
index 0000000..3c3024a
--- /dev/null
+++ b/trunk/src/modules/audio_processing/utility/util.gypi
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'apm_util',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '.',
+ ],
+ },
+ 'sources': [
+ 'delay_estimator.c',
+ 'delay_estimator.h',
+ 'delay_estimator_wrapper.c',
+ 'delay_estimator_wrapper.h',
+ 'fft4g.c',
+ 'fft4g.h',
+ 'ring_buffer.c',
+ 'ring_buffer.h',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_processing/voice_detection_impl.cc b/trunk/src/modules/audio_processing/voice_detection_impl.cc
new file mode 100644
index 0000000..50b99a0
--- /dev/null
+++ b/trunk/src/modules/audio_processing/voice_detection_impl.cc
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voice_detection_impl.h"
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#include "webrtc_vad.h"
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+
+namespace webrtc {
+
+typedef VadInst Handle;
+
+namespace {
+int MapSetting(VoiceDetection::Likelihood likelihood) {
+ switch (likelihood) {
+ case VoiceDetection::kVeryLowLikelihood:
+ return 3;
+ case VoiceDetection::kLowLikelihood:
+ return 2;
+ case VoiceDetection::kModerateLikelihood:
+ return 1;
+ case VoiceDetection::kHighLikelihood:
+ return 0;
+ }
+ assert(false);
+ return -1;
+}
+} // namespace
+
+VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm),
+ stream_has_voice_(false),
+ using_external_vad_(false),
+ likelihood_(kLowLikelihood),
+ frame_size_ms_(10),
+ frame_size_samples_(0) {}
+
+VoiceDetectionImpl::~VoiceDetectionImpl() {}
+
+int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ if (using_external_vad_) {
+ using_external_vad_ = false;
+ return apm_->kNoError;
+ }
+ assert(audio->samples_per_split_channel() <= 160);
+
+ WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
+ if (audio->num_channels() > 1) {
+ audio->CopyAndMixLowPass(1);
+ mixed_data = audio->mixed_low_pass_data(0);
+ }
+
+ // TODO(ajm): concatenate data in frame buffer here.
+
+ int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
+ apm_->split_sample_rate_hz(),
+ mixed_data,
+ frame_size_samples_);
+ if (vad_ret == 0) {
+ stream_has_voice_ = false;
+ audio->set_activity(AudioFrame::kVadPassive);
+ } else if (vad_ret == 1) {
+ stream_has_voice_ = true;
+ audio->set_activity(AudioFrame::kVadActive);
+ } else {
+ return apm_->kUnspecifiedError;
+ }
+
+ return apm_->kNoError;
+}
+
+int VoiceDetectionImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool VoiceDetectionImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
+ using_external_vad_ = true;
+ stream_has_voice_ = has_voice;
+ return apm_->kNoError;
+}
+
+bool VoiceDetectionImpl::stream_has_voice() const {
+ // TODO(ajm): enable this assertion?
+ //assert(using_external_vad_ || is_component_enabled());
+ return stream_has_voice_;
+}
+
+int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ if (MapSetting(likelihood) == -1) {
+ return apm_->kBadParameterError;
+ }
+
+ likelihood_ = likelihood;
+ return Configure();
+}
+
+VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
+ return likelihood_;
+}
+
+int VoiceDetectionImpl::set_frame_size_ms(int size) {
+ CriticalSectionScoped crit_scoped(apm_->crit());
+ assert(size == 10); // TODO(ajm): remove when supported.
+ if (size != 10 &&
+ size != 20 &&
+ size != 30) {
+ return apm_->kBadParameterError;
+ }
+
+ frame_size_ms_ = size;
+
+ return Initialize();
+}
+
+int VoiceDetectionImpl::frame_size_ms() const {
+ return frame_size_ms_;
+}
+
+int VoiceDetectionImpl::Initialize() {
+ int err = ProcessingComponent::Initialize();
+ if (err != apm_->kNoError || !is_component_enabled()) {
+ return err;
+ }
+
+ using_external_vad_ = false;
+ frame_size_samples_ = frame_size_ms_ * (apm_->split_sample_rate_hz() / 1000);
+ // TODO(ajm): intialize frame buffer here.
+
+ return apm_->kNoError;
+}
+
+void* VoiceDetectionImpl::CreateHandle() const {
+ Handle* handle = NULL;
+ if (WebRtcVad_Create(&handle) != apm_->kNoError) {
+ handle = NULL;
+ } else {
+ assert(handle != NULL);
+ }
+
+ return handle;
+}
+
+int VoiceDetectionImpl::DestroyHandle(void* handle) const {
+ return WebRtcVad_Free(static_cast<Handle*>(handle));
+}
+
+int VoiceDetectionImpl::InitializeHandle(void* handle) const {
+ return WebRtcVad_Init(static_cast<Handle*>(handle));
+}
+
+int VoiceDetectionImpl::ConfigureHandle(void* handle) const {
+ return WebRtcVad_set_mode(static_cast<Handle*>(handle),
+ MapSetting(likelihood_));
+}
+
+int VoiceDetectionImpl::num_handles_required() const {
+ return 1;
+}
+
+int VoiceDetectionImpl::GetHandleError(void* handle) const {
+ // The VAD has no get_error() function.
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/voice_detection_impl.h b/trunk/src/modules/audio_processing/voice_detection_impl.h
new file mode 100644
index 0000000..52d92e0
--- /dev/null
+++ b/trunk/src/modules/audio_processing/voice_detection_impl.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class VoiceDetectionImpl : public VoiceDetection,
+ public ProcessingComponent {
+ public:
+ explicit VoiceDetectionImpl(const AudioProcessingImpl* apm);
+ virtual ~VoiceDetectionImpl();
+
+ int ProcessCaptureAudio(AudioBuffer* audio);
+
+ // VoiceDetection implementation.
+ virtual bool is_enabled() const;
+
+ // ProcessingComponent implementation.
+ virtual int Initialize();
+
+ private:
+ // VoiceDetection implementation.
+ virtual int Enable(bool enable);
+ virtual int set_stream_has_voice(bool has_voice);
+ virtual bool stream_has_voice() const;
+ virtual int set_likelihood(Likelihood likelihood);
+ virtual Likelihood likelihood() const;
+ virtual int set_frame_size_ms(int size);
+ virtual int frame_size_ms() const;
+
+ // ProcessingComponent implementation.
+ virtual void* CreateHandle() const;
+ virtual int InitializeHandle(void* handle) const;
+ virtual int ConfigureHandle(void* handle) const;
+ virtual int DestroyHandle(void* handle) const;
+ virtual int num_handles_required() const;
+ virtual int GetHandleError(void* handle) const;
+
+ const AudioProcessingImpl* apm_;
+ bool stream_has_voice_;
+ bool using_external_vad_;
+ Likelihood likelihood_;
+ int frame_size_ms_;
+ int frame_size_samples_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
diff --git a/trunk/src/modules/interface/module.h b/trunk/src/modules/interface/module.h
new file mode 100644
index 0000000..37e5027
--- /dev/null
+++ b/trunk/src/modules/interface/module.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_INTERFACE_MODULE_H_
+#define MODULES_INTERFACE_MODULE_H_
+
+#include <assert.h>
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+class Module {
+ public:
+ // Change the unique identifier of this object.
+ virtual int32_t ChangeUniqueId(const int32_t id) = 0;
+
+ // Returns the number of milliseconds until the module want a worker
+ // thread to call Process.
+ virtual int32_t TimeUntilNextProcess() = 0;
+
+ // Process any pending tasks such as timeouts.
+ virtual int32_t Process() = 0;
+
+ protected:
+ virtual ~Module() {}
+};
+
+// Reference counted version of the module interface.
+class RefCountedModule : public Module {
+ public:
+ // Increase the reference count by one.
+ // Returns the incremented reference count.
+ // TODO(perkj): Make this pure virtual when Chromium have implemented
+ // reference counting ADM and Video capture module.
+ virtual int32_t AddRef() {
+ assert(false && "Not implemented.");
+ return 1;
+ }
+
+ // Decrease the reference count by one.
+ // Returns the decreased reference count.
+ // Returns 0 if the last reference was just released.
+ // When the reference count reach 0 the object will self-destruct.
+ // TODO(perkj): Make this pure virtual when Chromium have implemented
+ // reference counting ADM and Video capture module.
+ virtual int32_t Release() {
+ assert(false && "Not implemented.");
+ return 1;
+ }
+
+ protected:
+ virtual ~RefCountedModule() {}
+};
+
+} // namespace webrtc
+
+#endif // MODULES_INTERFACE_MODULE_H_
diff --git a/trunk/src/modules/interface/module_common_types.h b/trunk/src/modules/interface/module_common_types.h
new file mode 100644
index 0000000..2e1beaa
--- /dev/null
+++ b/trunk/src/modules/interface/module_common_types.h
@@ -0,0 +1,1047 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULE_COMMON_TYPES_H
+#define MODULE_COMMON_TYPES_H
+
+#include <cstring> // memcpy
+#include <assert.h>
+
+#include "typedefs.h"
+#include "common_types.h"
+
+#ifdef _WIN32
+ #pragma warning(disable:4351) // remove warning "new behavior: elements of array
+ // 'array' will be default initialized"
+#endif
+
+namespace webrtc
+{
+struct RTPHeader
+{
+ bool markerBit;
+ WebRtc_UWord8 payloadType;
+ WebRtc_UWord16 sequenceNumber;
+ WebRtc_UWord32 timestamp;
+ WebRtc_UWord32 ssrc;
+ WebRtc_UWord8 numCSRCs;
+ WebRtc_UWord32 arrOfCSRCs[kRtpCsrcSize];
+ WebRtc_UWord8 paddingLength;
+ WebRtc_UWord16 headerLength;
+};
+
+struct RTPHeaderExtension
+{
+ WebRtc_Word32 transmissionTimeOffset;
+};
+
+struct RTPAudioHeader
+{
+ WebRtc_UWord8 numEnergy; // number of valid entries in arrOfEnergy
+ WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel
+ bool isCNG; // is this CNG
+ WebRtc_UWord8 channel; // number of channels 2 = stereo
+};
+
+enum {kNoPictureId = -1};
+enum {kNoTl0PicIdx = -1};
+enum {kNoTemporalIdx = -1};
+enum {kNoKeyIdx = -1};
+enum {kNoSimulcastIdx = 0};
+
+struct RTPVideoHeaderVP8
+{
+ void InitRTPVideoHeaderVP8()
+ {
+ nonReference = false;
+ pictureId = kNoPictureId;
+ tl0PicIdx = kNoTl0PicIdx;
+ temporalIdx = kNoTemporalIdx;
+ layerSync = false;
+ keyIdx = kNoKeyIdx;
+ partitionId = 0;
+ beginningOfPartition = false;
+ frameWidth = 0;
+ frameHeight = 0;
+ }
+
+ bool nonReference; // Frame is discardable.
+ WebRtc_Word16 pictureId; // Picture ID index, 15 bits;
+ // kNoPictureId if PictureID does not exist.
+ WebRtc_Word16 tl0PicIdx; // TL0PIC_IDX, 8 bits;
+ // kNoTl0PicIdx means no value provided.
+ WebRtc_Word8 temporalIdx; // Temporal layer index, or kNoTemporalIdx.
+ bool layerSync; // This frame is a layer sync frame.
+ // Disabled if temporalIdx == kNoTemporalIdx.
+ int keyIdx; // 5 bits; kNoKeyIdx means not used.
+ int partitionId; // VP8 partition ID
+ bool beginningOfPartition; // True if this packet is the first
+ // in a VP8 partition. Otherwise false
+ int frameWidth; // Exists for key frames.
+ int frameHeight; // Exists for key frames.
+};
+union RTPVideoTypeHeader
+{
+ RTPVideoHeaderVP8 VP8;
+};
+
+enum RTPVideoCodecTypes
+{
+ kRTPVideoGeneric = 0,
+ kRTPVideoVP8 = 8,
+ kRTPVideoNoVideo = 10,
+ kRTPVideoFEC = 11,
+ kRTPVideoI420 = 12
+};
+struct RTPVideoHeader
+{
+ WebRtc_UWord16 width; // size
+ WebRtc_UWord16 height;
+
+ bool isFirstPacket; // first packet in frame
+ WebRtc_UWord8 simulcastIdx; // Index if the simulcast encoder creating
+ // this frame, 0 if not using simulcast.
+ RTPVideoCodecTypes codec;
+ RTPVideoTypeHeader codecHeader;
+};
+union RTPTypeHeader
+{
+ RTPAudioHeader Audio;
+ RTPVideoHeader Video;
+};
+
+struct WebRtcRTPHeader
+{
+ RTPHeader header;
+ FrameType frameType;
+ RTPTypeHeader type;
+ RTPHeaderExtension extension;
+};
+
+class RTPFragmentationHeader
+{
+public:
+ RTPFragmentationHeader() :
+ fragmentationVectorSize(0),
+ fragmentationOffset(NULL),
+ fragmentationLength(NULL),
+ fragmentationTimeDiff(NULL),
+ fragmentationPlType(NULL)
+ {};
+
+ ~RTPFragmentationHeader()
+ {
+ delete [] fragmentationOffset;
+ delete [] fragmentationLength;
+ delete [] fragmentationTimeDiff;
+ delete [] fragmentationPlType;
+ }
+
+ RTPFragmentationHeader& operator=(const RTPFragmentationHeader& header)
+ {
+ if(this == &header)
+ {
+ return *this;
+ }
+
+ if(header.fragmentationVectorSize != fragmentationVectorSize)
+ {
+ // new size of vectors
+
+ // delete old
+ delete [] fragmentationOffset;
+ fragmentationOffset = NULL;
+ delete [] fragmentationLength;
+ fragmentationLength = NULL;
+ delete [] fragmentationTimeDiff;
+ fragmentationTimeDiff = NULL;
+ delete [] fragmentationPlType;
+ fragmentationPlType = NULL;
+
+ if(header.fragmentationVectorSize > 0)
+ {
+ // allocate new
+ if(header.fragmentationOffset)
+ {
+ fragmentationOffset = new WebRtc_UWord32[header.fragmentationVectorSize];
+ }
+ if(header.fragmentationLength)
+ {
+ fragmentationLength = new WebRtc_UWord32[header.fragmentationVectorSize];
+ }
+ if(header.fragmentationTimeDiff)
+ {
+ fragmentationTimeDiff = new WebRtc_UWord16[header.fragmentationVectorSize];
+ }
+ if(header.fragmentationPlType)
+ {
+ fragmentationPlType = new WebRtc_UWord8[header.fragmentationVectorSize];
+ }
+ }
+ // set new size
+ fragmentationVectorSize = header.fragmentationVectorSize;
+ }
+
+ if(header.fragmentationVectorSize > 0)
+ {
+ // copy values
+ if(header.fragmentationOffset)
+ {
+ memcpy(fragmentationOffset, header.fragmentationOffset,
+ header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
+ }
+ if(header.fragmentationLength)
+ {
+ memcpy(fragmentationLength, header.fragmentationLength,
+ header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
+ }
+ if(header.fragmentationTimeDiff)
+ {
+ memcpy(fragmentationTimeDiff, header.fragmentationTimeDiff,
+ header.fragmentationVectorSize * sizeof(WebRtc_UWord16));
+ }
+ if(header.fragmentationPlType)
+ {
+ memcpy(fragmentationPlType, header.fragmentationPlType,
+ header.fragmentationVectorSize * sizeof(WebRtc_UWord8));
+ }
+ }
+ return *this;
+ }
+ void VerifyAndAllocateFragmentationHeader( const WebRtc_UWord16 size)
+ {
+ if( fragmentationVectorSize < size)
+ {
+ WebRtc_UWord16 oldVectorSize = fragmentationVectorSize;
+ {
+ // offset
+ WebRtc_UWord32* oldOffsets = fragmentationOffset;
+ fragmentationOffset = new WebRtc_UWord32[size];
+ memset(fragmentationOffset+oldVectorSize, 0,
+ sizeof(WebRtc_UWord32)*(size-oldVectorSize));
+ // copy old values
+ memcpy(fragmentationOffset,oldOffsets, sizeof(WebRtc_UWord32) * oldVectorSize);
+ delete[] oldOffsets;
+ }
+ // length
+ {
+ WebRtc_UWord32* oldLengths = fragmentationLength;
+ fragmentationLength = new WebRtc_UWord32[size];
+ memset(fragmentationLength+oldVectorSize, 0,
+ sizeof(WebRtc_UWord32) * (size- oldVectorSize));
+ memcpy(fragmentationLength, oldLengths,
+ sizeof(WebRtc_UWord32) * oldVectorSize);
+ delete[] oldLengths;
+ }
+ // time diff
+ {
+ WebRtc_UWord16* oldTimeDiffs = fragmentationTimeDiff;
+ fragmentationTimeDiff = new WebRtc_UWord16[size];
+ memset(fragmentationTimeDiff+oldVectorSize, 0,
+ sizeof(WebRtc_UWord16) * (size- oldVectorSize));
+ memcpy(fragmentationTimeDiff, oldTimeDiffs,
+ sizeof(WebRtc_UWord16) * oldVectorSize);
+ delete[] oldTimeDiffs;
+ }
+ // payload type
+ {
+ WebRtc_UWord8* oldTimePlTypes = fragmentationPlType;
+ fragmentationPlType = new WebRtc_UWord8[size];
+ memset(fragmentationPlType+oldVectorSize, 0,
+ sizeof(WebRtc_UWord8) * (size- oldVectorSize));
+ memcpy(fragmentationPlType, oldTimePlTypes,
+ sizeof(WebRtc_UWord8) * oldVectorSize);
+ delete[] oldTimePlTypes;
+ }
+ fragmentationVectorSize = size;
+ }
+ }
+
+ WebRtc_UWord16 fragmentationVectorSize; // Number of fragmentations
+ WebRtc_UWord32* fragmentationOffset; // Offset of pointer to data for each fragm.
+ WebRtc_UWord32* fragmentationLength; // Data size for each fragmentation
+ WebRtc_UWord16* fragmentationTimeDiff; // Timestamp difference relative "now" for
+ // each fragmentation
+ WebRtc_UWord8* fragmentationPlType; // Payload type of each fragmentation
+};
+
+struct RTCPVoIPMetric
+{
+ // RFC 3611 4.7
+ WebRtc_UWord8 lossRate;
+ WebRtc_UWord8 discardRate;
+ WebRtc_UWord8 burstDensity;
+ WebRtc_UWord8 gapDensity;
+ WebRtc_UWord16 burstDuration;
+ WebRtc_UWord16 gapDuration;
+ WebRtc_UWord16 roundTripDelay;
+ WebRtc_UWord16 endSystemDelay;
+ WebRtc_UWord8 signalLevel;
+ WebRtc_UWord8 noiseLevel;
+ WebRtc_UWord8 RERL;
+ WebRtc_UWord8 Gmin;
+ WebRtc_UWord8 Rfactor;
+ WebRtc_UWord8 extRfactor;
+ WebRtc_UWord8 MOSLQ;
+ WebRtc_UWord8 MOSCQ;
+ WebRtc_UWord8 RXconfig;
+ WebRtc_UWord16 JBnominal;
+ WebRtc_UWord16 JBmax;
+ WebRtc_UWord16 JBabsMax;
+};
+
+// class describing a complete, or parts of an encoded frame.
+class EncodedVideoData
+{
+public:
+ EncodedVideoData() :
+ completeFrame(false),
+ missingFrame(false),
+ payloadData(NULL),
+ payloadSize(0),
+ bufferSize(0)
+ {};
+
+ EncodedVideoData(const EncodedVideoData& data)
+ {
+ payloadType = data.payloadType;
+ timeStamp = data.timeStamp;
+ renderTimeMs = data.renderTimeMs;
+ encodedWidth = data.encodedWidth;
+ encodedHeight = data.encodedHeight;
+ completeFrame = data.completeFrame;
+ missingFrame = data.missingFrame;
+ payloadSize = data.payloadSize;
+ fragmentationHeader = data.fragmentationHeader;
+ frameType = data.frameType;
+ codec = data.codec;
+ if (data.payloadSize > 0)
+ {
+ payloadData = new WebRtc_UWord8[data.payloadSize];
+ memcpy(payloadData, data.payloadData, data.payloadSize);
+ }
+ else
+ {
+ payloadData = NULL;
+ }
+ }
+
+
+ ~EncodedVideoData()
+ {
+ delete [] payloadData;
+ };
+
+ EncodedVideoData& operator=(const EncodedVideoData& data)
+ {
+ if (this == &data)
+ {
+ return *this;
+ }
+ payloadType = data.payloadType;
+ timeStamp = data.timeStamp;
+ renderTimeMs = data.renderTimeMs;
+ encodedWidth = data.encodedWidth;
+ encodedHeight = data.encodedHeight;
+ completeFrame = data.completeFrame;
+ missingFrame = data.missingFrame;
+ payloadSize = data.payloadSize;
+ fragmentationHeader = data.fragmentationHeader;
+ frameType = data.frameType;
+ codec = data.codec;
+ if (data.payloadSize > 0)
+ {
+ delete [] payloadData;
+ payloadData = new WebRtc_UWord8[data.payloadSize];
+ memcpy(payloadData, data.payloadData, data.payloadSize);
+ bufferSize = data.payloadSize;
+ }
+ return *this;
+ };
+ void VerifyAndAllocate( const WebRtc_UWord32 size)
+ {
+ if (bufferSize < size)
+ {
+ WebRtc_UWord8* oldPayload = payloadData;
+ payloadData = new WebRtc_UWord8[size];
+ memcpy(payloadData, oldPayload, sizeof(WebRtc_UWord8) * payloadSize);
+
+ bufferSize = size;
+ delete[] oldPayload;
+ }
+ }
+
+ WebRtc_UWord8 payloadType;
+ WebRtc_UWord32 timeStamp;
+ WebRtc_Word64 renderTimeMs;
+ WebRtc_UWord32 encodedWidth;
+ WebRtc_UWord32 encodedHeight;
+ bool completeFrame;
+ bool missingFrame;
+ WebRtc_UWord8* payloadData;
+ WebRtc_UWord32 payloadSize;
+ WebRtc_UWord32 bufferSize;
+ RTPFragmentationHeader fragmentationHeader;
+ FrameType frameType;
+ VideoCodecType codec;
+};
+
+struct VideoContentMetrics {
+ VideoContentMetrics()
+ : motion_magnitude(0.0f),
+ spatial_pred_err(0.0f),
+ spatial_pred_err_h(0.0f),
+ spatial_pred_err_v(0.0f) {
+ }
+
+ void Reset() {
+ motion_magnitude = 0.0f;
+ spatial_pred_err = 0.0f;
+ spatial_pred_err_h = 0.0f;
+ spatial_pred_err_v = 0.0f;
+ }
+ float motion_magnitude;
+ float spatial_pred_err;
+ float spatial_pred_err_h;
+ float spatial_pred_err_v;
+};
+
+/*************************************************
+ *
+ * VideoFrame class
+ *
+ * The VideoFrame class allows storing and
+ * handling of video frames.
+ *
+ *
+ *************************************************/
+class VideoFrame
+{
+public:
+ VideoFrame();
+ ~VideoFrame();
+ /**
+ * Verifies that current allocated buffer size is larger than or equal to the input size.
+ * If the current buffer size is smaller, a new allocation is made and the old buffer data
+ * is copied to the new buffer.
+ * Buffer size is updated to minimumSize.
+ */
+ WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
+ /**
+ * Update length of data buffer in frame. Function verifies that new length is less or
+ * equal to allocated size.
+ */
+ WebRtc_Word32 SetLength(const WebRtc_UWord32 newLength);
+ /*
+ * Swap buffer and size data
+ */
+ WebRtc_Word32 Swap(WebRtc_UWord8*& newMemory,
+ WebRtc_UWord32& newLength,
+ WebRtc_UWord32& newSize);
+ /*
+ * Swap buffer and size data
+ */
+ WebRtc_Word32 SwapFrame(VideoFrame& videoFrame);
+ /**
+ * Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
+ * is allocated.
+ */
+ WebRtc_Word32 CopyFrame(const VideoFrame& videoFrame);
+ /**
+ * Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
+ * is allocated.
+ */
+ WebRtc_Word32 CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer);
+ /**
+ * Delete VideoFrame and resets members to zero
+ */
+ void Free();
+ /**
+ * Set frame timestamp (90kHz)
+ */
+ void SetTimeStamp(const WebRtc_UWord32 timeStamp) {_timeStamp = timeStamp;}
+ /**
+ * Get pointer to frame buffer
+ */
+ WebRtc_UWord8* Buffer() const {return _buffer;}
+
+ WebRtc_UWord8*& Buffer() {return _buffer;}
+
+ /**
+ * Get allocated buffer size
+ */
+ WebRtc_UWord32 Size() const {return _bufferSize;}
+ /**
+ * Get frame length
+ */
+ WebRtc_UWord32 Length() const {return _bufferLength;}
+ /**
+ * Get frame timestamp (90kHz)
+ */
+ WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
+ /**
+ * Get frame width
+ */
+ WebRtc_UWord32 Width() const {return _width;}
+ /**
+ * Get frame height
+ */
+ WebRtc_UWord32 Height() const {return _height;}
+ /**
+ * Set frame width
+ */
+ void SetWidth(const WebRtc_UWord32 width) {_width = width;}
+ /**
+ * Set frame height
+ */
+ void SetHeight(const WebRtc_UWord32 height) {_height = height;}
+ /**
+ * Set render time in miliseconds
+ */
+ void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
+ /**
+ * Get render time in miliseconds
+ */
+ WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
+
+private:
+ void Set(WebRtc_UWord8* buffer,
+ WebRtc_UWord32 size,
+ WebRtc_UWord32 length,
+ WebRtc_UWord32 timeStamp);
+
+ WebRtc_UWord8* _buffer; // Pointer to frame buffer
+ WebRtc_UWord32 _bufferSize; // Allocated buffer size
+ WebRtc_UWord32 _bufferLength; // Length (in bytes) of buffer
+ WebRtc_UWord32 _timeStamp; // Timestamp of frame (90kHz)
+ WebRtc_UWord32 _width;
+ WebRtc_UWord32 _height;
+ WebRtc_Word64 _renderTimeMs;
+}; // end of VideoFrame class declaration
+
+// inline implementation of VideoFrame class:
+inline
+VideoFrame::VideoFrame():
+ _buffer(0),
+ _bufferSize(0),
+ _bufferLength(0),
+ _timeStamp(0),
+ _width(0),
+ _height(0),
+ _renderTimeMs(0)
+{
+ //
+}
+inline
+VideoFrame::~VideoFrame()
+{
+ if(_buffer)
+ {
+ delete [] _buffer;
+ _buffer = NULL;
+ }
+}
+
+
+inline
+WebRtc_Word32
+VideoFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
+{
+ if (minimumSize < 1)
+ {
+ return -1;
+ }
+ if(minimumSize > _bufferSize)
+ {
+ // create buffer of sufficient size
+ WebRtc_UWord8* newBufferBuffer = new WebRtc_UWord8[minimumSize];
+ if(_buffer)
+ {
+ // copy old data
+ memcpy(newBufferBuffer, _buffer, _bufferSize);
+ delete [] _buffer;
+ }
+ else
+ {
+ memset(newBufferBuffer, 0, minimumSize * sizeof(WebRtc_UWord8));
+ }
+ _buffer = newBufferBuffer;
+ _bufferSize = minimumSize;
+ }
+ return 0;
+}
+
+inline
+WebRtc_Word32
+VideoFrame::SetLength(const WebRtc_UWord32 newLength)
+{
+ if (newLength >_bufferSize )
+ { // can't accomodate new value
+ return -1;
+ }
+ _bufferLength = newLength;
+ return 0;
+}
+
+inline
+WebRtc_Word32
+VideoFrame::SwapFrame(VideoFrame& videoFrame)
+{
+ WebRtc_UWord32 tmpTimeStamp = _timeStamp;
+ WebRtc_UWord32 tmpWidth = _width;
+ WebRtc_UWord32 tmpHeight = _height;
+ WebRtc_Word64 tmpRenderTime = _renderTimeMs;
+
+ _timeStamp = videoFrame._timeStamp;
+ _width = videoFrame._width;
+ _height = videoFrame._height;
+ _renderTimeMs = videoFrame._renderTimeMs;
+
+ videoFrame._timeStamp = tmpTimeStamp;
+ videoFrame._width = tmpWidth;
+ videoFrame._height = tmpHeight;
+ videoFrame._renderTimeMs = tmpRenderTime;
+
+ return Swap(videoFrame._buffer, videoFrame._bufferLength, videoFrame._bufferSize);
+}
+
+inline
+WebRtc_Word32
+VideoFrame::Swap(WebRtc_UWord8*& newMemory, WebRtc_UWord32& newLength, WebRtc_UWord32& newSize)
+{
+ WebRtc_UWord8* tmpBuffer = _buffer;
+ WebRtc_UWord32 tmpLength = _bufferLength;
+ WebRtc_UWord32 tmpSize = _bufferSize;
+ _buffer = newMemory;
+ _bufferLength = newLength;
+ _bufferSize = newSize;
+ newMemory = tmpBuffer;
+ newLength = tmpLength;
+ newSize = tmpSize;
+ return 0;
+}
+
+inline
+WebRtc_Word32
+VideoFrame::CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer)
+{
+ if (length > _bufferSize)
+ {
+ WebRtc_Word32 ret = VerifyAndAllocate(length);
+ if (ret < 0)
+ {
+ return ret;
+ }
+ }
+ memcpy(_buffer, sourceBuffer, length);
+ _bufferLength = length;
+ return 0;
+}
+
+inline
+WebRtc_Word32
+VideoFrame::CopyFrame(const VideoFrame& videoFrame)
+{
+ if(CopyFrame(videoFrame.Length(), videoFrame.Buffer()) != 0)
+ {
+ return -1;
+ }
+ _timeStamp = videoFrame._timeStamp;
+ _width = videoFrame._width;
+ _height = videoFrame._height;
+ _renderTimeMs = videoFrame._renderTimeMs;
+ return 0;
+}
+
+inline
+void
+VideoFrame::Free()
+{
+ _timeStamp = 0;
+ _bufferLength = 0;
+ _bufferSize = 0;
+ _height = 0;
+ _width = 0;
+ _renderTimeMs = 0;
+
+ if(_buffer)
+ {
+ delete [] _buffer;
+ _buffer = NULL;
+ }
+}
+
+
+/*************************************************
+ *
+ * AudioFrame class
+ *
+ * The AudioFrame class holds up to 60 ms wideband
+ * audio. It allows for adding and subtracting frames
+ * while keeping track of the resulting states.
+ *
+ * Note
+ * - The +operator assume that you would never add
+ * exact opposite frames when deciding the resulting
+ * state. To do this use the -operator.
+ *
+ * - _audioChannel of 1 indicated mono, and 2
+ * indicates stereo.
+ *
+ * - _payloadDataLengthInSamples is the number of
+ * samples per channel. Therefore, the total
+ * number of samples in _payloadData is
+ * (_payloadDataLengthInSamples * _audioChannel).
+ *
+ * - Stereo data is stored in interleaved fashion
+ * starting with the left channel.
+ *
+ *************************************************/
+class AudioFrame
+{
+public:
+ enum{kMaxAudioFrameSizeSamples = 3840}; // stereo 32KHz 60ms 2*32*60
+
+ enum VADActivity
+ {
+ kVadActive = 0,
+ kVadPassive = 1,
+ kVadUnknown = 2
+ };
+ enum SpeechType
+ {
+ kNormalSpeech = 0,
+ kPLC = 1,
+ kCNG = 2,
+ kPLCCNG = 3,
+ kUndefined = 4
+ };
+
+ AudioFrame();
+ virtual ~AudioFrame();
+
+ WebRtc_Word32 UpdateFrame(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16* payloadData,
+ const WebRtc_UWord16 payloadDataLengthInSamples,
+ const int frequencyInHz,
+ const SpeechType speechType,
+ const VADActivity vadActivity,
+ const WebRtc_UWord8 audioChannel = 1,
+ const WebRtc_Word32 volume = -1,
+ const WebRtc_Word32 energy = -1);
+
+ AudioFrame& Append(const AudioFrame& rhs);
+
+ void Mute();
+
+ AudioFrame& operator=(const AudioFrame& rhs);
+ AudioFrame& operator>>=(const WebRtc_Word32 rhs);
+ AudioFrame& operator+=(const AudioFrame& rhs);
+ AudioFrame& operator-=(const AudioFrame& rhs);
+
+ WebRtc_Word32 _id;
+ WebRtc_UWord32 _timeStamp;
+
+ // Supporting Stereo, stereo samples are interleaved
+ WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
+ WebRtc_UWord16 _payloadDataLengthInSamples;
+ int _frequencyInHz;
+ WebRtc_UWord8 _audioChannel;
+ SpeechType _speechType;
+ VADActivity _vadActivity;
+
+ WebRtc_UWord32 _energy;
+ WebRtc_Word32 _volume;
+};
+
+inline
+AudioFrame::AudioFrame()
+ :
+ _id(-1),
+ _timeStamp(0),
+ _payloadData(),
+ _payloadDataLengthInSamples(0),
+ _frequencyInHz(0),
+ _audioChannel(1),
+ _speechType(kUndefined),
+ _vadActivity(kVadUnknown),
+ _energy(0xffffffff),
+ _volume(0xffffffff)
+{
+}
+
+inline
+AudioFrame::~AudioFrame()
+{
+}
+
+inline
+WebRtc_Word32
+AudioFrame::UpdateFrame(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_Word16* payloadData,
+ const WebRtc_UWord16 payloadDataLengthInSamples,
+ const int frequencyInHz,
+ const SpeechType speechType,
+ const VADActivity vadActivity,
+ const WebRtc_UWord8 audioChannel,
+ const WebRtc_Word32 volume,
+ const WebRtc_Word32 energy)
+{
+ _id = id;
+ _timeStamp = timeStamp;
+ _frequencyInHz = frequencyInHz;
+ _speechType = speechType;
+ _vadActivity = vadActivity;
+ _volume = volume;
+ _audioChannel = audioChannel;
+ _energy = energy;
+
+ if((payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
+ (audioChannel > 2) || (audioChannel < 1))
+ {
+ _payloadDataLengthInSamples = 0;
+ return -1;
+ }
+ _payloadDataLengthInSamples = payloadDataLengthInSamples;
+ if(payloadData != NULL)
+ {
+ memcpy(_payloadData, payloadData, sizeof(WebRtc_Word16) *
+ payloadDataLengthInSamples * _audioChannel);
+ }
+ else
+ {
+ memset(_payloadData,0,sizeof(WebRtc_Word16) *
+ payloadDataLengthInSamples * _audioChannel);
+ }
+ return 0;
+}
+
+inline
+void
+AudioFrame::Mute()
+{
+ memset(_payloadData, 0, _payloadDataLengthInSamples * sizeof(WebRtc_Word16));
+}
+
+inline
+AudioFrame&
+AudioFrame::operator=(const AudioFrame& rhs)
+{
+ // Sanity Check
+ if((rhs._payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
+ (rhs._audioChannel > 2) ||
+ (rhs._audioChannel < 1))
+ {
+ return *this;
+ }
+ if(this == &rhs)
+ {
+ return *this;
+ }
+ _id = rhs._id;
+ _timeStamp = rhs._timeStamp;
+ _frequencyInHz = rhs._frequencyInHz;
+ _speechType = rhs._speechType;
+ _vadActivity = rhs._vadActivity;
+ _volume = rhs._volume;
+ _audioChannel = rhs._audioChannel;
+ _energy = rhs._energy;
+
+ _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
+ memcpy(_payloadData, rhs._payloadData,
+ sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
+
+ return *this;
+}
+
+inline
+AudioFrame&
+AudioFrame::operator>>=(const WebRtc_Word32 rhs)
+{
+ assert((_audioChannel > 0) && (_audioChannel < 3));
+ if((_audioChannel > 2) ||
+ (_audioChannel < 1))
+ {
+ return *this;
+ }
+ for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
+ {
+ _payloadData[i] = WebRtc_Word16(_payloadData[i] >> rhs);
+ }
+ return *this;
+}
+
+inline
+AudioFrame&
+AudioFrame::Append(const AudioFrame& rhs)
+{
+ // Sanity check
+ assert((_audioChannel > 0) && (_audioChannel < 3));
+ if((_audioChannel > 2) ||
+ (_audioChannel < 1))
+ {
+ return *this;
+ }
+ if(_audioChannel != rhs._audioChannel)
+ {
+ return *this;
+ }
+ if((_vadActivity == kVadActive) ||
+ rhs._vadActivity == kVadActive)
+ {
+ _vadActivity = kVadActive;
+ }
+ else if((_vadActivity == kVadUnknown) ||
+ rhs._vadActivity == kVadUnknown)
+ {
+ _vadActivity = kVadUnknown;
+ }
+ if(_speechType != rhs._speechType)
+ {
+ _speechType = kUndefined;
+ }
+
+ WebRtc_UWord16 offset = _payloadDataLengthInSamples * _audioChannel;
+ for(WebRtc_UWord16 i = 0;
+ i < rhs._payloadDataLengthInSamples * rhs._audioChannel;
+ i++)
+ {
+ _payloadData[offset+i] = rhs._payloadData[i];
+ }
+ _payloadDataLengthInSamples += rhs._payloadDataLengthInSamples;
+ return *this;
+}
+
+// merge vectors
+inline
+AudioFrame&
+AudioFrame::operator+=(const AudioFrame& rhs)
+{
+ // Sanity check
+ assert((_audioChannel > 0) && (_audioChannel < 3));
+ if((_audioChannel > 2) ||
+ (_audioChannel < 1))
+ {
+ return *this;
+ }
+ if(_audioChannel != rhs._audioChannel)
+ {
+ return *this;
+ }
+ bool noPrevData = false;
+ if(_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples)
+ {
+ if(_payloadDataLengthInSamples == 0)
+ {
+ // special case we have no data to start with
+ _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
+ noPrevData = true;
+ } else
+ {
+ return *this;
+ }
+ }
+
+ if((_vadActivity == kVadActive) ||
+ rhs._vadActivity == kVadActive)
+ {
+ _vadActivity = kVadActive;
+ }
+ else if((_vadActivity == kVadUnknown) ||
+ rhs._vadActivity == kVadUnknown)
+ {
+ _vadActivity = kVadUnknown;
+ }
+
+ if(_speechType != rhs._speechType)
+ {
+ _speechType = kUndefined;
+ }
+
+ if(noPrevData)
+ {
+ memcpy(_payloadData, rhs._payloadData,
+ sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
+ } else
+ {
+ // IMPROVEMENT this can be done very fast in assembly
+ for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
+ {
+ WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] +
+ (WebRtc_Word32)rhs._payloadData[i];
+ if(wrapGuard < -32768)
+ {
+ _payloadData[i] = -32768;
+ }else if(wrapGuard > 32767)
+ {
+ _payloadData[i] = 32767;
+ }else
+ {
+ _payloadData[i] = (WebRtc_Word16)wrapGuard;
+ }
+ }
+ }
+ _energy = 0xffffffff;
+ _volume = 0xffffffff;
+ return *this;
+}
+
+inline
+AudioFrame&
+AudioFrame::operator-=(const AudioFrame& rhs)
+{
+ // Sanity check
+ assert((_audioChannel > 0) && (_audioChannel < 3));
+ if((_audioChannel > 2)||
+ (_audioChannel < 1))
+ {
+ return *this;
+ }
+ if((_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples) ||
+ (_audioChannel != rhs._audioChannel))
+ {
+ return *this;
+ }
+ if((_vadActivity != kVadPassive) ||
+ rhs._vadActivity != kVadPassive)
+ {
+ _vadActivity = kVadUnknown;
+ }
+ _speechType = kUndefined;
+
+ for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
+ {
+ WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] -
+ (WebRtc_Word32)rhs._payloadData[i];
+ if(wrapGuard < -32768)
+ {
+ _payloadData[i] = -32768;
+ }
+ else if(wrapGuard > 32767)
+ {
+ _payloadData[i] = 32767;
+ }
+ else
+ {
+ _payloadData[i] = (WebRtc_Word16)wrapGuard;
+ }
+ }
+ _energy = 0xffffffff;
+ _volume = 0xffffffff;
+ return *this;
+}
+
+} // namespace webrtc
+
+#endif // MODULE_COMMON_TYPES_H
diff --git a/trunk/src/modules/media_file/OWNERS b/trunk/src/modules/media_file/OWNERS
new file mode 100644
index 0000000..2cc47e4
--- /dev/null
+++ b/trunk/src/modules/media_file/OWNERS
@@ -0,0 +1,4 @@
+pwestin@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
+niklas.enbom@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/media_file/interface/media_file.h b/trunk/src/modules/media_file/interface/media_file.h
new file mode 100644
index 0000000..bc857ad
--- /dev/null
+++ b/trunk/src/modules/media_file/interface/media_file.h
@@ -0,0 +1,227 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+
+#include "common_types.h"
+#include "typedefs.h"
+#include "module.h"
+#include "module_common_types.h"
+#include "media_file_defines.h"
+
+namespace webrtc {
+class MediaFile : public Module
+{
+public:
+ // Factory method. Constructor disabled. id is the identifier for the
+ // MediaFile instance.
+ static MediaFile* CreateMediaFile(const WebRtc_Word32 id);
+ static void DestroyMediaFile(MediaFile* module);
+
+ // Set the MediaFile instance identifier.
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ // Put 10-60ms of audio data from file into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes is both an input and output
+ // parameter. As input parameter it indicates the size of audioBuffer.
+ // As output parameter it indicates the number of bytes written to
+ // audioBuffer.
+ // Note: This API only play mono audio but can be used on file containing
+ // audio with more channels (in which case the audio will be converted to
+ // mono).
+ virtual WebRtc_Word32 PlayoutAudioData(
+ WebRtc_Word8* audioBuffer,
+ WebRtc_UWord32& dataLengthInBytes) = 0;
+
+ // Put one video frame into videoBuffer. dataLengthInBytes is both an input
+ // and output parameter. As input parameter it indicates the size of
+ // videoBuffer. As output parameter it indicates the number of bytes written
+ // to videoBuffer.
+ virtual WebRtc_Word32 PlayoutAVIVideoData(
+ WebRtc_Word8* videoBuffer,
+ WebRtc_UWord32& dataLengthInBytes) = 0;
+
+ // Put 10-60ms, depending on codec frame size, of audio data from file into
+ // audioBufferLeft and audioBufferRight. The buffers contain the left and
+ // right channel of played out stereo audio.
+ // dataLengthInBytes is both an input and output parameter. As input
+ // parameter it indicates the size of both audioBufferLeft and
+ // audioBufferRight. As output parameter it indicates the number of bytes
+ // written to both audio buffers.
+ // Note: This API can only be successfully called for WAV files with stereo
+ // audio.
+ virtual WebRtc_Word32 PlayoutStereoData(
+ WebRtc_Word8* audioBufferLeft,
+ WebRtc_Word8* audioBufferRight,
+ WebRtc_UWord32& dataLengthInBytes) = 0;
+
+ // Open the file specified by fileName (relative path is allowed) for
+ // reading. FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. If loop is true the file will be played until
+ // StopPlaying() is called. When end of file is reached the file is read
+ // from the start. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo audio is only supported for WAV files.
+ virtual WebRtc_Word32 StartPlayingAudioFile(
+ const WebRtc_Word8* fileName,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const bool loop = false,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0) = 0;
+
+ // Open the file specified by fileName for reading (relative path is
+ // allowed). If loop is true the file will be played until StopPlaying() is
+ // called. When end of file is reached the file is read from the start.
+ // format specifies the type of file fileName refers to. Only video will be
+ // read if videoOnly is true.
+ virtual WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
+ const bool loop,
+ bool videoOnly,
+ const FileFormats format) = 0;
+
+ // Prepare for playing audio from stream.
+ // FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo audio is only supported for WAV files.
+ virtual WebRtc_Word32 StartPlayingAudioStream(
+ InStream& stream,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0) = 0;
+
+ // Stop playing from file or stream.
+ virtual WebRtc_Word32 StopPlaying() = 0;
+
+ // Return true if playing.
+ virtual bool IsPlaying() = 0;
+
+
+ // Set durationMs to the number of ms that has been played from file.
+ virtual WebRtc_Word32 PlayoutPositionMs(
+ WebRtc_UWord32& durationMs) const = 0;
+
+ // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to file. The audio frame size is determined by the codecInst.pacsize
+ // parameter of the last sucessfull StartRecordingAudioFile(..) call.
+ // Note: bufferLength must be exactly one frame.
+ virtual WebRtc_Word32 IncomingAudioData(
+ const WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 bufferLength) = 0;
+
+ // Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
+ // to file.
+ // Note: videoBuffer can contain encoded data. The codec used must be the
+ // same as what was specified by videoCodecInst for the last successfull
+ // StartRecordingVideoFile(..) call. The videoBuffer must contain exactly
+ // one video frame.
+ virtual WebRtc_Word32 IncomingAVIVideoData(
+ const WebRtc_Word8* videoBuffer,
+ const WebRtc_UWord32 bufferLength) = 0;
+
+ // Open/creates file specified by fileName for writing (relative path is
+ // allowed). FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that should be created/opened.
+ // codecInst specifies the encoding of the audio data. maxSizeBytes
+ // specifies the number of bytes allowed to be written to file if it is
+ // greater than zero.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ virtual WebRtc_Word32 StartRecordingAudioFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const WebRtc_UWord32 maxSizeBytes = 0) = 0;
+
+ // Open/create the file specified by fileName for writing audio/video data
+ // (relative path is allowed). format specifies the type of file fileName
+ // should be. codecInst specifies the encoding of the audio data.
+ // videoCodecInst specifies the encoding of the video data. Only video data
+ // will be recorded if videoOnly is true.
+ virtual WebRtc_Word32 StartRecordingVideoFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ bool videoOnly = false) = 0;
+
+ // Prepare for recording audio to stream.
+ // FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that stream should correspond to.
+ // codecInst specifies the encoding of the audio data.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ virtual WebRtc_Word32 StartRecordingAudioStream(
+ OutStream& stream,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0) = 0;
+
+ // Stop recording to file or stream.
+ virtual WebRtc_Word32 StopRecording() = 0;
+
+ // Return true if recording.
+ virtual bool IsRecording() = 0;
+
+ // Set durationMs to the number of ms that has been recorded to file.
+ virtual WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs) = 0;
+
+ // Return true if recording or playing is stereo.
+ virtual bool IsStereo() = 0;
+
+ // Register callback to receive media file related notifications. Disables
+ // callbacks if callback is NULL.
+ virtual WebRtc_Word32 SetModuleFileCallback(FileCallback* callback) = 0;
+
+ // Set durationMs to the size of the file (in ms) specified by fileName.
+ // format specifies the type of file fileName refers to. freqInHz specifies
+ // the sampling frequency of the file.
+ virtual WebRtc_Word32 FileDurationMs(
+ const WebRtc_Word8* fileName,
+ WebRtc_UWord32& durationMs,
+ const FileFormats format,
+ const WebRtc_UWord32 freqInHz = 16000) = 0;
+
+ // Update codecInst according to the current audio codec being used for
+ // reading or writing.
+ virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0;
+
+ // Update videoCodecInst according to the current video codec being used for
+ // reading or writing.
+ virtual WebRtc_Word32 VideoCodecInst(VideoCodec& videoCodecInst) const = 0;
+
+protected:
+ MediaFile() {}
+ virtual ~MediaFile() {}
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
diff --git a/trunk/src/modules/media_file/interface/media_file_defines.h b/trunk/src/modules/media_file/interface/media_file_defines.h
new file mode 100644
index 0000000..38af562
--- /dev/null
+++ b/trunk/src/modules/media_file/interface/media_file_defines.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+
+#include "engine_configurations.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+
+namespace webrtc {
+// Callback class for the MediaFile class.
+class FileCallback
+{
+public:
+ virtual ~FileCallback(){}
+
+ // This function is called by MediaFile when a file has been playing for
+ // durationMs ms. id is the identifier for the MediaFile instance calling
+ // the callback.
+ virtual void PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs) = 0;
+
+ // This function is called by MediaFile when a file has been recording for
+ // durationMs ms. id is the identifier for the MediaFile instance calling
+ // the callback.
+ virtual void RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs) = 0;
+
+ // This function is called by MediaFile when a file has been stopped
+ // playing. id is the identifier for the MediaFile instance calling the
+ // callback.
+ virtual void PlayFileEnded(const WebRtc_Word32 id) = 0;
+
+ // This function is called by MediaFile when a file has been stopped
+ // recording. id is the identifier for the MediaFile instance calling the
+ // callback.
+ virtual void RecordFileEnded(const WebRtc_Word32 id) = 0;
+
+protected:
+ FileCallback() {}
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
diff --git a/trunk/src/modules/media_file/source/Android.mk b/trunk/src/modules/media_file/source/Android.mk
new file mode 100644
index 0000000..2c408c8
--- /dev/null
+++ b/trunk/src/modules/media_file/source/Android.mk
@@ -0,0 +1,46 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_media_file
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ avi_file.cc \
+ media_file_impl.cc \
+ media_file_utility.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../utility/interface \
+ $(LOCAL_PATH)/../../audio_coding/main/interface \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/media_file/source/avi_file.cc b/trunk/src/modules/media_file/source/avi_file.cc
new file mode 100644
index 0000000..74f7908
--- /dev/null
+++ b/trunk/src/modules/media_file/source/avi_file.cc
@@ -0,0 +1,1773 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// TODO(henrike): reassess the error handling in this class. Currently failure
+// is detected by asserts in many places. Also a refactoring of this class would
+// be beneficial.
+
+#include "avi_file.h"
+
+#include <assert.h>
+#include <string.h>
+
+#ifdef _WIN32
+#include <windows.h>
+#endif
+
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "list_wrapper.h"
+#include "trace.h"
+
+// http://msdn2.microsoft.com/en-us/library/ms779636.aspx
+// A chunk has the following form:
+// ckID ckSize ckData
+// where ckID is a FOURCC that identifies the data contained in the
+// chunk, ckData is a 4-byte value giving the size of the data in
+// ckData, and ckData is zero or more bytes of data. The data is always
+// padded to nearest WORD boundary. ckSize gives the size of the valid
+// data in the chunk; it does not include the padding, the size of
+// ckID, or the size of ckSize.
+//http://msdn2.microsoft.com/en-us/library/ms779632.aspx
+//NOTE: Workaround to make MPEG4 files play on WMP. MPEG files
+// place the config parameters efter the BITMAPINFOHEADER and
+// *NOT* in the 'strd'!
+// http://msdn.microsoft.com/en-us/library/dd183375.aspx
+// http://msdn.microsoft.com/en-us/library/dd183376.aspx
+
+namespace webrtc {
+namespace {
+static const WebRtc_UWord32 kAvifHasindex = 0x00000010;
+static const WebRtc_UWord32 kAvifMustuseindex = 0x00000020;
+static const WebRtc_UWord32 kAvifIsinterleaved = 0x00000100;
+static const WebRtc_UWord32 kAvifTrustcktype = 0x00000800;
+static const WebRtc_UWord32 kAvifWascapturefile = 0x00010000;
+
+template <class T>
+T MinValue(T a, T b)
+{
+ return a < b ? a : b;
+}
+} // namespace
+
+AviFile::AVIMAINHEADER::AVIMAINHEADER()
+ : fcc( 0),
+ cb( 0),
+ dwMicroSecPerFrame( 0),
+ dwMaxBytesPerSec( 0),
+ dwPaddingGranularity( 0),
+ dwFlags( 0),
+ dwTotalFrames( 0),
+ dwInitialFrames( 0),
+ dwStreams( 0),
+ dwSuggestedBufferSize(0),
+ dwWidth( 0),
+ dwHeight( 0)
+{
+ dwReserved[0] = 0;
+ dwReserved[1] = 0;
+ dwReserved[2] = 0;
+ dwReserved[3] = 0;
+}
+
+AVISTREAMHEADER::AVISTREAMHEADER()
+ : fcc( 0),
+ cb( 0),
+ fccType( 0),
+ fccHandler( 0),
+ dwFlags( 0),
+ wPriority( 0),
+ wLanguage( 0),
+ dwInitialFrames( 0),
+ dwScale( 0),
+ dwRate( 0),
+ dwStart( 0),
+ dwLength( 0),
+ dwSuggestedBufferSize(0),
+ dwQuality( 0),
+ dwSampleSize( 0)
+{
+ rcFrame.left = 0;
+ rcFrame.top = 0;
+ rcFrame.right = 0;
+ rcFrame.bottom = 0;
+}
+
+BITMAPINFOHEADER::BITMAPINFOHEADER()
+ : biSize( 0),
+ biWidth( 0),
+ biHeight( 0),
+ biPlanes( 0),
+ biBitCount( 0),
+ biCompression( 0),
+ biSizeImage( 0),
+ biXPelsPerMeter(0),
+ biYPelsPerMeter(0),
+ biClrUsed( 0),
+ biClrImportant( 0)
+{
+}
+
+WAVEFORMATEX::WAVEFORMATEX()
+ : wFormatTag( 0),
+ nChannels( 0),
+ nSamplesPerSec( 0),
+ nAvgBytesPerSec(0),
+ nBlockAlign( 0),
+ wBitsPerSample( 0),
+ cbSize( 0)
+{
+}
+
+AviFile::AVIINDEXENTRY::AVIINDEXENTRY(WebRtc_UWord32 inckid,
+ WebRtc_UWord32 indwFlags,
+ WebRtc_UWord32 indwChunkOffset,
+ WebRtc_UWord32 indwChunkLength)
+ : ckid(inckid),
+ dwFlags(indwFlags),
+ dwChunkOffset(indwChunkOffset),
+ dwChunkLength(indwChunkLength)
+{
+}
+
+AviFile::AviFile()
+ : _crit(CriticalSectionWrapper::CreateCriticalSection()),
+ _aviFile(NULL),
+ _aviHeader(),
+ _videoStreamHeader(),
+ _audioStreamHeader(),
+ _videoFormatHeader(),
+ _audioFormatHeader(),
+ _videoConfigParameters(),
+ _videoConfigLength(0),
+ _videoStreamName(),
+ _videoStreamNameLength(0),
+ _audioConfigParameters(),
+ _audioStreamName(),
+ _videoStream(),
+ _audioStream(),
+ _nrStreams(0),
+ _aviLength(0),
+ _dataLength(0),
+ _bytesRead(0),
+ _dataStartByte(0),
+ _framesRead(0),
+ _videoFrames(0),
+ _audioFrames(0),
+ _reading(false),
+ _openedAs(AVI_AUDIO),
+ _loop(false),
+ _writing(false),
+ _bytesWritten(0),
+ _riffSizeMark(0),
+ _moviSizeMark(0),
+ _totNumFramesMark(0),
+ _videoStreamLengthMark(0),
+ _audioStreamLengthMark(0),
+ _moviListOffset(0),
+ _writeAudioStream(false),
+ _writeVideoStream(false),
+ _aviMode(NotSet),
+ _videoCodecConfigParams(NULL),
+ _videoCodecConfigParamsLength(0),
+ _videoStreamDataChunkPrefix(0),
+ _audioStreamDataChunkPrefix(0),
+ _created(false),
+ _indexList(new ListWrapper())
+{
+ ResetComplexMembers();
+}
+
+AviFile::~AviFile()
+{
+ Close();
+
+ delete _indexList;
+ delete[] _videoCodecConfigParams;
+ delete _crit;
+}
+
+WebRtc_Word32 AviFile::Open(AVIStreamType streamType, const char* fileName,
+ bool loop)
+{
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1, "OpenAVIFile(%s)",
+ fileName);
+ _crit->Enter();
+
+ if (_aviMode != NotSet)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ _aviMode = Read;
+
+ if (!fileName)
+ {
+ _crit->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "\tfileName not valid!");
+ return -1;
+ }
+
+#ifdef _WIN32
+ // fopen does not support wide characters on Windows, ergo _wfopen.
+ wchar_t wideFileName[FileWrapper::kMaxFileNameSize];
+ wideFileName[0] = 0;
+ MultiByteToWideChar(CP_UTF8,0,fileName, -1, // convert the whole string
+ wideFileName, FileWrapper::kMaxFileNameSize);
+
+ _aviFile = _wfopen(wideFileName, L"rb");
+#else
+ _aviFile = fopen(fileName, "rb");
+#endif
+
+ if (!_aviFile)
+ {
+ _crit->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Could not open file!");
+ return -1;
+ }
+
+ // ReadRIFF verifies that the file is AVI and figures out the file length.
+ WebRtc_Word32 err = ReadRIFF();
+ if (err)
+ {
+ if (_aviFile)
+ {
+ fclose(_aviFile);
+ _aviFile = NULL;
+ }
+ _crit->Leave();
+ return -1;
+ }
+
+ err = ReadHeaders();
+ if (err)
+ {
+ if (_aviFile)
+ {
+ fclose(_aviFile);
+ _aviFile = NULL;
+ }
+ _crit->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Unsupported or corrupt AVI format");
+ return -1;
+ }
+
+ _dataStartByte = _bytesRead;
+ _reading = true;
+ _openedAs = streamType;
+ _loop = loop;
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::Close()
+{
+ _crit->Enter();
+ switch (_aviMode)
+ {
+ case Read:
+ CloseRead();
+ break;
+ case Write:
+ CloseWrite();
+ break;
+ default:
+ break;
+ }
+
+ if (_videoCodecConfigParams)
+ {
+ delete [] _videoCodecConfigParams;
+ _videoCodecConfigParams = 0;
+ }
+ ResetMembers();
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_UWord32 AviFile::MakeFourCc(WebRtc_UWord8 ch0, WebRtc_UWord8 ch1,
+ WebRtc_UWord8 ch2, WebRtc_UWord8 ch3)
+{
+ return ((WebRtc_UWord32)(WebRtc_UWord8)(ch0) |
+ ((WebRtc_UWord32)(WebRtc_UWord8)(ch1) << 8) |
+ ((WebRtc_UWord32)(WebRtc_UWord8)(ch2) << 16) |
+ ((WebRtc_UWord32)(WebRtc_UWord8)(ch3) << 24 ));
+}
+
+WebRtc_Word32 AviFile::GetVideoStreamInfo(AVISTREAMHEADER& videoStreamHeader,
+ BITMAPINFOHEADER& bitmapInfo,
+ char* codecConfigParameters,
+ WebRtc_Word32& configLength)
+{
+ _crit->Enter();
+ if (!_reading && !_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ memcpy(&videoStreamHeader, &_videoStreamHeader, sizeof(_videoStreamHeader));
+ memcpy(&bitmapInfo, &_videoFormatHeader, sizeof(_videoFormatHeader));
+
+ if (configLength <= _videoConfigLength)
+ {
+ memcpy(codecConfigParameters, _videoConfigParameters,
+ _videoConfigLength);
+ configLength = _videoConfigLength;
+ }
+ else
+ {
+ configLength = 0;
+ }
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::GetDuration(WebRtc_Word32& durationMs)
+{
+ _crit->Enter();
+ if (_videoStreamHeader.dwRate==0 || _videoStreamHeader.dwScale==0)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ durationMs = _videoStreamHeader.dwLength * 1000 /
+ (_videoStreamHeader.dwRate/_videoStreamHeader.dwScale);
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::GetAudioStreamInfo(WAVEFORMATEX& waveHeader)
+{
+ _crit->Enter();
+ if (_aviMode != Read)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (!_reading && !_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ memcpy(&waveHeader, &_audioFormatHeader, sizeof(_audioFormatHeader));
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAudio(const WebRtc_UWord8* data,
+ WebRtc_Word32 length)
+{
+ _crit->Enter();
+ size_t newBytesWritten = _bytesWritten;
+
+ if (_aviMode != Write)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (!_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (!_writeAudioStream)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ // Start of chunk.
+ const WebRtc_UWord32 chunkOffset = ftell(_aviFile) - _moviListOffset;
+ _bytesWritten += PutLE32(_audioStreamDataChunkPrefix);
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t chunkSizeMark = _bytesWritten;
+
+ _bytesWritten += PutBuffer(data, length);
+
+ const long chunkSize = PutLE32LengthFromCurrent(
+ static_cast<long>(chunkSizeMark));
+
+ // Make sure that the chunk is aligned on 2 bytes (= 1 sample).
+ if (chunkSize % 2)
+ {
+ _bytesWritten += PutByte(0);
+ }
+ // End of chunk
+
+ // Save chunk information for use when closing file.
+ AddChunkToIndexList(_audioStreamDataChunkPrefix, 0, // No flags.
+ chunkOffset, chunkSize);
+
+ ++_audioFrames;
+ newBytesWritten = _bytesWritten - newBytesWritten;
+ _crit->Leave();
+ return static_cast<WebRtc_Word32>(newBytesWritten);
+}
+
+WebRtc_Word32 AviFile::WriteVideo(const WebRtc_UWord8* data,
+ WebRtc_Word32 length)
+{
+ _crit->Enter();
+ size_t newBytesWritten = _bytesWritten;
+ if (_aviMode != Write)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (!_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (!_writeVideoStream)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ // Start of chunk.
+ const WebRtc_UWord32 chunkOffset = ftell(_aviFile) - _moviListOffset;
+ _bytesWritten += PutLE32(_videoStreamDataChunkPrefix);
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t chunkSizeMark = _bytesWritten;
+
+ _bytesWritten += PutBuffer(data, length);
+
+ const long chunkSize = PutLE32LengthFromCurrent(
+ static_cast<long>(chunkSizeMark));
+
+ // Make sure that the chunk is aligned on 2 bytes (= 1 sample).
+ if (chunkSize % 2)
+ {
+ //Pad one byte, to WORD align.
+ _bytesWritten += PutByte(0);
+ }
+ //End chunk!
+ AddChunkToIndexList(_videoStreamDataChunkPrefix, 0, // No flags.
+ chunkOffset, static_cast<WebRtc_UWord32>(chunkSize));
+
+ ++_videoFrames;
+ newBytesWritten = _bytesWritten - newBytesWritten;
+ _crit->Leave();
+ return static_cast<WebRtc_Word32>(newBytesWritten);
+}
+
+WebRtc_Word32 AviFile::PrepareDataChunkHeaders()
+{
+ // 00 video stream, 01 audio stream.
+ // db uncompresses video, dc compressed video, wb WAV audio
+ if (_writeVideoStream)
+ {
+ if (strncmp((const char*) &_videoStreamHeader.fccHandler, "I420", 4) ==
+ 0)
+ {
+ _videoStreamDataChunkPrefix = MakeFourCc('0', '0', 'd', 'b');
+ }
+ else
+ {
+ _videoStreamDataChunkPrefix = MakeFourCc('0', '0', 'd', 'c');
+ }
+ _audioStreamDataChunkPrefix = MakeFourCc('0', '1', 'w', 'b');
+ }
+ else
+ {
+ _audioStreamDataChunkPrefix = MakeFourCc('0', '0', 'w', 'b');
+ }
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadMoviSubChunk(WebRtc_UWord8* data,
+ WebRtc_Word32& length,
+ WebRtc_UWord32 tag1,
+ WebRtc_UWord32 tag2)
+{
+ if (!_reading)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
+ "AviFile::ReadMoviSubChunk(): File not open!");
+ length = 0;
+ return -1;
+ }
+
+ WebRtc_UWord32 size;
+ bool isEOFReached = false;
+ // Try to read one data chunk header
+ while (true)
+ {
+ // TODO (hellner): what happens if an empty AVI file is opened with
+ // _loop set to true? Seems like this while-loop would never exit!
+
+ // tag = db uncompresses video, dc compressed video or wb WAV audio.
+ WebRtc_UWord32 tag;
+ _bytesRead += GetLE32(tag);
+ _bytesRead += GetLE32(size);
+
+ const WebRtc_Word32 eof = feof(_aviFile);
+ if (!eof)
+ {
+ if (tag == tag1)
+ {
+ // Supported tag found.
+ break;
+ }
+ else if ((tag == tag2) && (tag2 != 0))
+ {
+ // Supported tag found.
+ break;
+ }
+
+ // Jump to next chunk. The size is in bytes but chunks are aligned
+ // on 2 byte boundaries.
+ const WebRtc_UWord32 seekSize = (size % 2) ? size + 1 : size;
+ const WebRtc_Word32 err = fseek(_aviFile, seekSize, SEEK_CUR);
+
+ if (err)
+ {
+ isEOFReached = true;
+ }
+ }
+ else
+ {
+ isEOFReached = true;
+ }
+
+ if (isEOFReached)
+ {
+ clearerr(_aviFile);
+
+ if (_loop)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
+ "AviFile::ReadMoviSubChunk(): Reached end of AVI\
+ data file, starting from the beginning.");
+
+ fseek(_aviFile, static_cast<long>(_dataStartByte), SEEK_SET);
+
+ _bytesRead = _dataStartByte;
+ _framesRead = 0;
+ isEOFReached = false;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
+ "AviFile::ReadMoviSubChunk(): Reached end of AVI\
+ file!");
+ length = 0;
+ return -1;
+ }
+ }
+ _bytesRead += size;
+ }
+
+ if (static_cast<WebRtc_Word32>(size) > length)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
+ "AviFile::ReadMoviSubChunk(): AVI read buffer too small!");
+
+ // Jump to next chunk. The size is in bytes but chunks are aligned
+ // on 2 byte boundaries.
+ const WebRtc_UWord32 seekSize = (size % 2) ? size + 1 : size;
+ fseek(_aviFile, seekSize, SEEK_CUR);
+ _bytesRead += seekSize;
+ length = 0;
+ return -1;
+ }
+ _bytesRead += GetBuffer(data, size);
+
+ // The size is in bytes but chunks are aligned on 2 byte boundaries.
+ if (size % 2)
+ {
+ WebRtc_UWord8 dummy_byte;
+ _bytesRead += GetByte(dummy_byte);
+ }
+ length = size;
+ ++_framesRead;
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadAudio(WebRtc_UWord8* data, WebRtc_Word32& length)
+{
+ _crit->Enter();
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "AviFile::ReadAudio()");
+
+ if (_aviMode != Read)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (_openedAs != AVI_AUDIO)
+ {
+ length = 0;
+ _crit->Leave();
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "File not open as audio!");
+ return -1;
+ }
+
+ const WebRtc_Word32 ret = ReadMoviSubChunk(
+ data,
+ length,
+ StreamAndTwoCharCodeToTag(_audioStream.streamNumber, "wb"));
+
+ _crit->Leave();
+ return ret;
+}
+
+WebRtc_Word32 AviFile::ReadVideo(WebRtc_UWord8* data, WebRtc_Word32& length)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "AviFile::ReadVideo()");
+
+ _crit->Enter();
+ if (_aviMode != Read)
+ {
+ //Has to be Read!
+ _crit->Leave();
+ return -1;
+ }
+ if (_openedAs != AVI_VIDEO)
+ {
+ length = 0;
+ _crit->Leave();
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "File not open as video!");
+ return -1;
+ }
+
+ const WebRtc_Word32 ret = ReadMoviSubChunk(
+ data,
+ length,
+ StreamAndTwoCharCodeToTag(_videoStream.streamNumber, "dc"),
+ StreamAndTwoCharCodeToTag(_videoStream.streamNumber, "db"));
+ _crit->Leave();
+ return ret;
+}
+
+WebRtc_Word32 AviFile::Create(const char* fileName)
+{
+ _crit->Enter();
+ if (_aviMode != Write)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ if (!_writeVideoStream && !_writeAudioStream)
+ {
+ _crit->Leave();
+ return -1;
+ }
+ if (_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+#ifdef _WIN32
+ // fopen does not support wide characters on Windows, ergo _wfopen.
+ wchar_t wideFileName[FileWrapper::kMaxFileNameSize];
+ wideFileName[0] = 0;
+
+ MultiByteToWideChar(CP_UTF8,0,fileName, -1, // convert the whole string
+ wideFileName, FileWrapper::kMaxFileNameSize);
+
+ _aviFile = _wfopen(wideFileName, L"w+b");
+ if (!_aviFile)
+ {
+ _crit->Leave();
+ return -1;
+ }
+#else
+ _aviFile = fopen(fileName, "w+b");
+ if (!_aviFile)
+ {
+ _crit->Leave();
+ return -1;
+ }
+#endif
+
+ WriteRIFF();
+ WriteHeaders();
+
+ _created = true;
+
+ PrepareDataChunkHeaders();
+ ClearIndexList();
+ WriteMoviStart();
+ _aviMode = Write;
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::CreateVideoStream(
+ const AVISTREAMHEADER& videoStreamHeader,
+ const BITMAPINFOHEADER& bitMapInfoHeader,
+ const WebRtc_UWord8* codecConfigParams,
+ WebRtc_Word32 codecConfigParamsLength)
+{
+ _crit->Enter();
+ if (_aviMode == Read)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ if (_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ _aviMode = Write;
+ _writeVideoStream = true;
+
+ _videoStreamHeader = videoStreamHeader;
+ _videoFormatHeader = bitMapInfoHeader;
+
+ if (codecConfigParams && codecConfigParamsLength > 0)
+ {
+ if (_videoCodecConfigParams)
+ {
+ delete [] _videoCodecConfigParams;
+ _videoCodecConfigParams = 0;
+ }
+
+ _videoCodecConfigParams = new WebRtc_UWord8[codecConfigParamsLength];
+ _videoCodecConfigParamsLength = codecConfigParamsLength;
+
+ memcpy(_videoCodecConfigParams, codecConfigParams,
+ _videoCodecConfigParamsLength);
+ }
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::CreateAudioStream(
+ const AVISTREAMHEADER& audioStreamHeader,
+ const WAVEFORMATEX& waveFormatHeader)
+{
+ _crit->Enter();
+
+ if (_aviMode == Read)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ if (_created)
+ {
+ _crit->Leave();
+ return -1;
+ }
+
+ _aviMode = Write;
+ _writeAudioStream = true;
+ _audioStreamHeader = audioStreamHeader;
+ _audioFormatHeader = waveFormatHeader;
+ _crit->Leave();
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteRIFF()
+{
+ const WebRtc_UWord32 riffTag = MakeFourCc('R', 'I', 'F', 'F');
+ _bytesWritten += PutLE32(riffTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ _riffSizeMark = _bytesWritten;
+
+ const WebRtc_UWord32 aviTag = MakeFourCc('A', 'V', 'I', ' ');
+ _bytesWritten += PutLE32(aviTag);
+
+ return 0;
+}
+
+
+WebRtc_Word32 AviFile::WriteHeaders()
+{
+ // Main AVI header list.
+ const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
+ _bytesWritten += PutLE32(listTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t listhdrlSizeMark = _bytesWritten;
+
+ const WebRtc_UWord32 hdrlTag = MakeFourCc('h', 'd', 'r', 'l');
+ _bytesWritten += PutLE32(hdrlTag);
+
+ WriteAVIMainHeader();
+ WriteAVIStreamHeaders();
+
+ const long hdrlLen = PutLE32LengthFromCurrent(
+ static_cast<long>(listhdrlSizeMark));
+
+ // Junk chunk to align on 2048 boundry (CD-ROM sector boundary).
+ const WebRtc_UWord32 junkTag = MakeFourCc('J', 'U', 'N', 'K');
+ _bytesWritten += PutLE32(junkTag);
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t junkSizeMark = _bytesWritten;
+
+ const WebRtc_UWord32 junkBufferSize =
+ 0x800 // 2048 byte alignment
+ - 12 // RIFF SIZE 'AVI '
+ - 8 // LIST SIZE
+ - hdrlLen //
+ - 8 // JUNK SIZE
+ - 12; // LIST SIZE 'MOVI'
+
+ // TODO (hellner): why not just fseek here?
+ WebRtc_UWord8* junkBuffer = new WebRtc_UWord8[junkBufferSize];
+ memset(junkBuffer, 0, junkBufferSize);
+ _bytesWritten += PutBuffer(junkBuffer, junkBufferSize);
+ delete [] junkBuffer;
+
+ PutLE32LengthFromCurrent(static_cast<long>(junkSizeMark));
+ // End of JUNK chunk.
+ // End of main AVI header list.
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIMainHeader()
+{
+ const WebRtc_UWord32 avihTag = MakeFourCc('a', 'v', 'i', 'h');
+ _bytesWritten += PutLE32(avihTag);
+ _bytesWritten += PutLE32(14 * sizeof(WebRtc_UWord32));
+
+ const WebRtc_UWord32 scale = _videoStreamHeader.dwScale ?
+ _videoStreamHeader.dwScale : 1;
+ const WebRtc_UWord32 microSecPerFrame = 1000000 /
+ (_videoStreamHeader.dwRate / scale);
+ _bytesWritten += PutLE32(microSecPerFrame);
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+
+ WebRtc_UWord32 numStreams = 0;
+ if (_writeVideoStream)
+ {
+ ++numStreams;
+ }
+ if (_writeAudioStream)
+ {
+ ++numStreams;
+ }
+
+ if (numStreams == 1)
+ {
+ _bytesWritten += PutLE32(
+ kAvifTrustcktype
+ | kAvifHasindex
+ | kAvifWascapturefile);
+ }
+ else
+ {
+ _bytesWritten += PutLE32(
+ kAvifTrustcktype
+ | kAvifHasindex
+ | kAvifWascapturefile
+ | kAvifIsinterleaved);
+ }
+
+ _totNumFramesMark = _bytesWritten;
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(numStreams);
+
+ if (_writeVideoStream)
+ {
+ _bytesWritten += PutLE32(
+ _videoStreamHeader.dwSuggestedBufferSize);
+ _bytesWritten += PutLE32(
+ _videoStreamHeader.rcFrame.right-_videoStreamHeader.rcFrame.left);
+ _bytesWritten += PutLE32(
+ _videoStreamHeader.rcFrame.bottom-_videoStreamHeader.rcFrame.top);
+ } else {
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ }
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ _bytesWritten += PutLE32(0);
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIStreamHeaders()
+{
+ if (_writeVideoStream)
+ {
+ WriteAVIVideoStreamHeaders();
+ }
+ if (_writeAudioStream)
+ {
+ WriteAVIAudioStreamHeaders();
+ }
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIVideoStreamHeaders()
+{
+ const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
+ _bytesWritten += PutLE32(listTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t liststrlSizeMark = _bytesWritten;
+
+ const WebRtc_UWord32 hdrlTag = MakeFourCc('s', 't', 'r', 'l');
+ _bytesWritten += PutLE32(hdrlTag);
+
+ WriteAVIVideoStreamHeaderChunks();
+
+ PutLE32LengthFromCurrent(static_cast<long>(liststrlSizeMark));
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIVideoStreamHeaderChunks()
+{
+ // Start of strh
+ const WebRtc_UWord32 strhTag = MakeFourCc('s', 't', 'r', 'h');
+ _bytesWritten += PutLE32(strhTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strhSizeMark = _bytesWritten;
+
+ _bytesWritten += PutLE32(_videoStreamHeader.fccType);
+ _bytesWritten += PutLE32(_videoStreamHeader.fccHandler);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwFlags);
+ _bytesWritten += PutLE16(_videoStreamHeader.wPriority);
+ _bytesWritten += PutLE16(_videoStreamHeader.wLanguage);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwInitialFrames);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwScale);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwRate);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwStart);
+
+ _videoStreamLengthMark = _bytesWritten;
+ _bytesWritten += PutLE32(_videoStreamHeader.dwLength);
+
+ _bytesWritten += PutLE32(_videoStreamHeader.dwSuggestedBufferSize);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwQuality);
+ _bytesWritten += PutLE32(_videoStreamHeader.dwSampleSize);
+ _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.left);
+ _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.top);
+ _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.right);
+ _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.bottom);
+
+ PutLE32LengthFromCurrent(static_cast<long>(strhSizeMark));
+ // End of strh
+
+ // Start of strf
+ const WebRtc_UWord32 strfTag = MakeFourCc('s', 't', 'r', 'f');
+ _bytesWritten += PutLE32(strfTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strfSizeMark = _bytesWritten;
+
+ _bytesWritten += PutLE32(_videoFormatHeader.biSize);
+ _bytesWritten += PutLE32(_videoFormatHeader.biWidth);
+ _bytesWritten += PutLE32(_videoFormatHeader.biHeight);
+ _bytesWritten += PutLE16(_videoFormatHeader.biPlanes);
+ _bytesWritten += PutLE16(_videoFormatHeader.biBitCount);
+ _bytesWritten += PutLE32(_videoFormatHeader.biCompression);
+ _bytesWritten += PutLE32(_videoFormatHeader.biSizeImage);
+ _bytesWritten += PutLE32(_videoFormatHeader.biXPelsPerMeter);
+ _bytesWritten += PutLE32(_videoFormatHeader.biYPelsPerMeter);
+ _bytesWritten += PutLE32(_videoFormatHeader.biClrUsed);
+ _bytesWritten += PutLE32(_videoFormatHeader.biClrImportant);
+
+ const bool isMpegFile = _videoStreamHeader.fccHandler ==
+ AviFile::MakeFourCc('M','4','S','2');
+ if (isMpegFile)
+ {
+ if (_videoCodecConfigParams && _videoCodecConfigParamsLength > 0)
+ {
+ _bytesWritten += PutBuffer(_videoCodecConfigParams,
+ _videoCodecConfigParamsLength);
+ }
+ }
+
+ PutLE32LengthFromCurrent(static_cast<long>(strfSizeMark));
+ // End of strf
+
+ if ( _videoCodecConfigParams
+ && (_videoCodecConfigParamsLength > 0)
+ && !isMpegFile)
+ {
+ // Write strd, unless it's an MPEG file
+ const WebRtc_UWord32 strdTag = MakeFourCc('s', 't', 'r', 'd');
+ _bytesWritten += PutLE32(strdTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strdSizeMark = _bytesWritten;
+
+ _bytesWritten += PutBuffer(_videoCodecConfigParams,
+ _videoCodecConfigParamsLength);
+
+ PutLE32LengthFromCurrent(static_cast<long>(strdSizeMark));
+ // End of strd
+ }
+
+ // Start of strn
+ const WebRtc_UWord32 strnTag = MakeFourCc('s', 't', 'r', 'n');
+ _bytesWritten += PutLE32(strnTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strnSizeMark = _bytesWritten;
+
+ _bytesWritten += PutBufferZ("WebRtc.avi ");
+
+ PutLE32LengthFromCurrent(static_cast<long>(strnSizeMark));
+ // End of strd
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIAudioStreamHeaders()
+{
+ // Start of LIST
+ WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
+ _bytesWritten += PutLE32(listTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t liststrlSizeMark = _bytesWritten;
+
+ WebRtc_UWord32 hdrlTag = MakeFourCc('s', 't', 'r', 'l');
+ _bytesWritten += PutLE32(hdrlTag);
+
+ WriteAVIAudioStreamHeaderChunks();
+
+ PutLE32LengthFromCurrent(static_cast<long>(liststrlSizeMark));
+ //End of LIST
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteAVIAudioStreamHeaderChunks()
+{
+ // Start of strh
+ const WebRtc_UWord32 strhTag = MakeFourCc('s', 't', 'r', 'h');
+ _bytesWritten += PutLE32(strhTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strhSizeMark = _bytesWritten;
+
+ _bytesWritten += PutLE32(_audioStreamHeader.fccType);
+ _bytesWritten += PutLE32(_audioStreamHeader.fccHandler);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwFlags);
+ _bytesWritten += PutLE16(_audioStreamHeader.wPriority);
+ _bytesWritten += PutLE16(_audioStreamHeader.wLanguage);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwInitialFrames);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwScale);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwRate);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwStart);
+
+ _audioStreamLengthMark = _bytesWritten;
+ _bytesWritten += PutLE32(_audioStreamHeader.dwLength);
+
+ _bytesWritten += PutLE32(_audioStreamHeader.dwSuggestedBufferSize);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwQuality);
+ _bytesWritten += PutLE32(_audioStreamHeader.dwSampleSize);
+ _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.left);
+ _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.top);
+ _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.right);
+ _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.bottom);
+
+ PutLE32LengthFromCurrent(static_cast<long>(strhSizeMark));
+ // End of strh
+
+ // Start of strf
+ const WebRtc_UWord32 strfTag = MakeFourCc('s', 't', 'r', 'f');
+ _bytesWritten += PutLE32(strfTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strfSizeMark = _bytesWritten;
+
+ _bytesWritten += PutLE16(_audioFormatHeader.wFormatTag);
+ _bytesWritten += PutLE16(_audioFormatHeader.nChannels);
+ _bytesWritten += PutLE32(_audioFormatHeader.nSamplesPerSec);
+ _bytesWritten += PutLE32(_audioFormatHeader.nAvgBytesPerSec);
+ _bytesWritten += PutLE16(_audioFormatHeader.nBlockAlign);
+ _bytesWritten += PutLE16(_audioFormatHeader.wBitsPerSample);
+ _bytesWritten += PutLE16(_audioFormatHeader.cbSize);
+
+ PutLE32LengthFromCurrent(static_cast<long>(strfSizeMark));
+ // End end of strf.
+
+ // Audio doesn't have strd.
+
+ // Start of strn
+ const WebRtc_UWord32 strnTag = MakeFourCc('s', 't', 'r', 'n');
+ _bytesWritten += PutLE32(strnTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t strnSizeMark = _bytesWritten;
+
+ _bytesWritten += PutBufferZ("WebRtc.avi ");
+
+ PutLE32LengthFromCurrent(static_cast<long>(strnSizeMark));
+ // End of strd.
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::WriteMoviStart()
+{
+ // Create template movi list. Fill out size when known (i.e. when closing
+ // file).
+ const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
+ _bytesWritten += PutLE32(listTag);
+
+ _bytesWritten += PutLE32(0); //Size! Change later!
+ _moviSizeMark = _bytesWritten;
+ _moviListOffset = ftell(_aviFile);
+
+ const WebRtc_UWord32 moviTag = MakeFourCc('m', 'o', 'v', 'i');
+ _bytesWritten += PutLE32(moviTag);
+
+ return 0;
+}
+
+size_t AviFile::PutByte(WebRtc_UWord8 byte)
+{
+ return fwrite(&byte, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord8),
+ _aviFile);
+}
+
+size_t AviFile::PutLE16(WebRtc_UWord16 word)
+{
+ return fwrite(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord16),
+ _aviFile);
+}
+
+size_t AviFile::PutLE32(WebRtc_UWord32 word)
+{
+ return fwrite(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord32),
+ _aviFile);
+}
+
+size_t AviFile::PutBuffer(const WebRtc_UWord8* str, size_t size)
+{
+ return fwrite(str, sizeof(WebRtc_UWord8), size,
+ _aviFile);
+}
+
+size_t AviFile::PutBufferZ(const char* str)
+{
+ // Include NULL charachter, hence the + 1
+ return PutBuffer(reinterpret_cast<const WebRtc_UWord8*>(str),
+ strlen(str) + 1);
+}
+
+long AviFile::PutLE32LengthFromCurrent(long startPos)
+{
+ const long endPos = ftell(_aviFile);
+ if (endPos < 0) {
+ return 0;
+ }
+ bool success = (0 == fseek(_aviFile, startPos - 4, SEEK_SET));
+ if (!success) {
+ assert(false);
+ return 0;
+ }
+ const long len = endPos - startPos;
+ if (endPos > startPos) {
+ PutLE32(len);
+ }
+ else {
+ assert(false);
+ }
+ success = (0 == fseek(_aviFile, endPos, SEEK_SET));
+ assert(success);
+ return len;
+}
+
+void AviFile::PutLE32AtPos(long pos, WebRtc_UWord32 word)
+{
+ const long currPos = ftell(_aviFile);
+ if (currPos < 0) {
+ assert(false);
+ return;
+ }
+ bool success = (0 == fseek(_aviFile, pos, SEEK_SET));
+ if (!success) {
+ assert(false);
+ return;
+ }
+ PutLE32(word);
+ success = (0 == fseek(_aviFile, currPos, SEEK_SET));
+ assert(success);
+}
+
+void AviFile::CloseRead()
+{
+ if (_aviFile)
+ {
+ fclose(_aviFile);
+ _aviFile = NULL;
+ }
+}
+
+void AviFile::CloseWrite()
+{
+ if (_created)
+ {
+ // Update everything that isn't known until the file is closed. The
+ // marks indicate where in the headers this update should be.
+ PutLE32LengthFromCurrent(static_cast<long>(_moviSizeMark));
+
+ PutLE32AtPos(static_cast<long>(_totNumFramesMark), _videoFrames);
+
+ if (_writeVideoStream)
+ {
+ PutLE32AtPos(static_cast<long>(_videoStreamLengthMark),
+ _videoFrames);
+ }
+
+ if (_writeAudioStream)
+ {
+ PutLE32AtPos(static_cast<long>(_audioStreamLengthMark),
+ _audioFrames);
+ }
+
+ WriteIndex();
+ PutLE32LengthFromCurrent(static_cast<long>(_riffSizeMark));
+ ClearIndexList();
+
+ if (_aviFile)
+ {
+ fclose(_aviFile);
+ _aviFile = NULL;
+ }
+ }
+}
+
+void AviFile::ResetMembers()
+{
+ ResetComplexMembers();
+
+ _aviFile = NULL;
+
+ _nrStreams = 0;
+ _aviLength = 0;
+ _dataLength = 0;
+ _bytesRead = 0;
+ _dataStartByte = 0;
+ _framesRead = 0;
+ _videoFrames = 0;
+ _audioFrames = 0;
+
+ _reading = false;
+ _openedAs = AVI_AUDIO;
+ _loop = false;
+ _writing = false;
+
+ _bytesWritten = 0;
+
+ _riffSizeMark = 0;
+ _moviSizeMark = 0;
+ _totNumFramesMark = 0;
+ _videoStreamLengthMark = 0;
+ _audioStreamLengthMark = 0;
+
+ _writeAudioStream = false;
+ _writeVideoStream = false;
+
+ _aviMode = NotSet;
+ _videoCodecConfigParams = 0;
+ _videoCodecConfigParamsLength = 0;
+
+ _videoStreamDataChunkPrefix = 0;
+ _audioStreamDataChunkPrefix = 0;
+
+ _created = false;
+
+ _moviListOffset = 0;
+
+ _videoConfigLength = 0;
+}
+
+void AviFile::ResetComplexMembers()
+{
+ memset(&_aviHeader, 0, sizeof(AVIMAINHEADER));
+ memset(&_videoStreamHeader, 0, sizeof(AVISTREAMHEADER));
+ memset(&_audioStreamHeader, 0, sizeof(AVISTREAMHEADER));
+ memset(&_videoFormatHeader, 0, sizeof(BITMAPINFOHEADER));
+ memset(&_audioFormatHeader, 0, sizeof(WAVEFORMATEX));
+ memset(_videoConfigParameters, 0, CODEC_CONFIG_LENGTH);
+ memset(_videoStreamName, 0, STREAM_NAME_LENGTH);
+ memset(_audioStreamName, 0, STREAM_NAME_LENGTH);
+ memset(&_videoStream, 0, sizeof(AVIStream));
+ memset(&_audioStream, 0, sizeof(AVIStream));
+}
+
+size_t AviFile::GetByte(WebRtc_UWord8& word)
+{
+ return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord8), _aviFile);
+}
+
+size_t AviFile::GetLE16(WebRtc_UWord16& word)
+{
+ return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord16),
+ _aviFile);
+}
+
+size_t AviFile::GetLE32(WebRtc_UWord32& word)
+{
+ return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord32),
+ _aviFile);
+}
+
+size_t AviFile::GetBuffer(WebRtc_UWord8* str, size_t size)
+{
+ return fread(str, sizeof(WebRtc_UWord8), size, _aviFile);
+}
+
+WebRtc_Word32 AviFile::ReadRIFF()
+{
+ WebRtc_UWord32 tag;
+ _bytesRead = GetLE32(tag);
+ if (tag != MakeFourCc('R', 'I', 'F', 'F'))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not a RIFF file!");
+ return -1;
+ }
+
+ WebRtc_UWord32 size;
+ _bytesRead += GetLE32(size);
+ _aviLength = size;
+
+ _bytesRead += GetLE32(tag);
+ if (tag != MakeFourCc('A', 'V', 'I', ' '))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not an AVI file!");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadHeaders()
+{
+ WebRtc_UWord32 tag;
+ _bytesRead += GetLE32(tag);
+ WebRtc_UWord32 size;
+ _bytesRead += GetLE32(size);
+
+ if (tag != MakeFourCc('L', 'I', 'S', 'T'))
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 listTag;
+ _bytesRead += GetLE32(listTag);
+ if (listTag != MakeFourCc('h', 'd', 'r', 'l'))
+ {
+ return -1;
+ }
+
+ WebRtc_Word32 err = ReadAVIMainHeader();
+ if (err)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadAVIMainHeader()
+{
+ _bytesRead += GetLE32(_aviHeader.fcc);
+ _bytesRead += GetLE32(_aviHeader.cb);
+ _bytesRead += GetLE32(_aviHeader.dwMicroSecPerFrame);
+ _bytesRead += GetLE32(_aviHeader.dwMaxBytesPerSec);
+ _bytesRead += GetLE32(_aviHeader.dwPaddingGranularity);
+ _bytesRead += GetLE32(_aviHeader.dwFlags);
+ _bytesRead += GetLE32(_aviHeader.dwTotalFrames);
+ _bytesRead += GetLE32(_aviHeader.dwInitialFrames);
+ _bytesRead += GetLE32(_aviHeader.dwStreams);
+ _bytesRead += GetLE32(_aviHeader.dwSuggestedBufferSize);
+ _bytesRead += GetLE32(_aviHeader.dwWidth);
+ _bytesRead += GetLE32(_aviHeader.dwHeight);
+ _bytesRead += GetLE32(_aviHeader.dwReserved[0]);
+ _bytesRead += GetLE32(_aviHeader.dwReserved[1]);
+ _bytesRead += GetLE32(_aviHeader.dwReserved[2]);
+ _bytesRead += GetLE32(_aviHeader.dwReserved[3]);
+
+ if (_aviHeader.fcc != MakeFourCc('a', 'v', 'i', 'h'))
+ {
+ return -1;
+ }
+
+ if (_aviHeader.dwFlags & kAvifMustuseindex)
+ {
+ return -1;
+ }
+
+ bool readVideoStreamHeader = false;
+ bool readAudioStreamHeader = false;
+ unsigned int streamsRead = 0;
+ while (_aviHeader.dwStreams > streamsRead)
+ {
+ WebRtc_UWord32 strltag;
+ _bytesRead += GetLE32(strltag);
+ WebRtc_UWord32 strlsize;
+ _bytesRead += GetLE32(strlsize);
+ const long endSeekPos = ftell(_aviFile) +
+ static_cast<WebRtc_Word32>(strlsize);
+
+ if (strltag != MakeFourCc('L', 'I', 'S', 'T'))
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 listTag;
+ _bytesRead += GetLE32(listTag);
+ if (listTag != MakeFourCc('s', 't', 'r', 'l'))
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 chunktag;
+ _bytesRead += GetLE32(chunktag);
+ WebRtc_UWord32 chunksize;
+ _bytesRead += GetLE32(chunksize);
+
+ if (chunktag != MakeFourCc('s', 't', 'r', 'h'))
+ {
+ return -1;
+ }
+
+ AVISTREAMHEADER tmpStreamHeader;
+ tmpStreamHeader.fcc = chunktag;
+ tmpStreamHeader.cb = chunksize;
+
+ _bytesRead += GetLE32(tmpStreamHeader.fccType);
+ _bytesRead += GetLE32(tmpStreamHeader.fccHandler);
+ _bytesRead += GetLE32(tmpStreamHeader.dwFlags);
+ _bytesRead += GetLE16(tmpStreamHeader.wPriority);
+ _bytesRead += GetLE16(tmpStreamHeader.wLanguage);
+ _bytesRead += GetLE32(tmpStreamHeader.dwInitialFrames);
+ _bytesRead += GetLE32(tmpStreamHeader.dwScale);
+ _bytesRead += GetLE32(tmpStreamHeader.dwRate);
+ _bytesRead += GetLE32(tmpStreamHeader.dwStart);
+ _bytesRead += GetLE32(tmpStreamHeader.dwLength);
+ _bytesRead += GetLE32(tmpStreamHeader.dwSuggestedBufferSize);
+ _bytesRead += GetLE32(tmpStreamHeader.dwQuality);
+ _bytesRead += GetLE32(tmpStreamHeader.dwSampleSize);
+
+ WebRtc_UWord16 left;
+ _bytesRead += GetLE16(left);
+ tmpStreamHeader.rcFrame.left = left;
+ WebRtc_UWord16 top;
+ _bytesRead += GetLE16(top);
+ tmpStreamHeader.rcFrame.top = top;
+ WebRtc_UWord16 right;
+ _bytesRead += GetLE16(right);
+ tmpStreamHeader.rcFrame.right = right;
+ WebRtc_UWord16 bottom;
+ _bytesRead += GetLE16(bottom);
+ tmpStreamHeader.rcFrame.bottom = bottom;
+
+ if (!readVideoStreamHeader
+ && (tmpStreamHeader.fccType == MakeFourCc('v', 'i', 'd', 's')))
+ {
+ _videoStreamHeader = tmpStreamHeader; //Bitwise copy is OK!
+ const WebRtc_Word32 err = ReadAVIVideoStreamHeader(endSeekPos);
+ if (err)
+ {
+ return -1;
+ }
+ // Make sure there actually is video data in the file...
+ if (_videoStreamHeader.dwLength == 0)
+ {
+ return -1;
+ }
+ readVideoStreamHeader = true;
+ } else if(!readAudioStreamHeader &&
+ (tmpStreamHeader.fccType == MakeFourCc('a', 'u', 'd', 's'))) {
+ _audioStreamHeader = tmpStreamHeader;
+ const WebRtc_Word32 err = ReadAVIAudioStreamHeader(endSeekPos);
+ if (err)
+ {
+ return -1;
+ }
+ readAudioStreamHeader = true;
+ }
+ else
+ {
+ fseek(_aviFile, endSeekPos, SEEK_SET);
+ _bytesRead += endSeekPos;
+ }
+
+ ++streamsRead;
+ }
+
+ if (!readVideoStreamHeader && !readAudioStreamHeader)
+ {
+ return -1;
+ }
+
+ WebRtc_UWord32 tag;
+ _bytesRead += GetLE32(tag);
+ WebRtc_UWord32 size;
+ _bytesRead += GetLE32(size);
+
+ if (tag == MakeFourCc('J', 'U', 'N', 'K'))
+ {
+ fseek(_aviFile, size, SEEK_CUR);
+ _bytesRead += size;
+ _bytesRead += GetLE32(tag);
+ _bytesRead += GetLE32(size);
+ }
+ if (tag != MakeFourCc('L', 'I', 'S', 'T'))
+ {
+ return -1;
+ }
+ WebRtc_UWord32 listTag;
+ _bytesRead += GetLE32(listTag);
+ if (listTag != MakeFourCc('m', 'o', 'v', 'i'))
+ {
+ return -1;
+ }
+ _dataLength = size;
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadAVIVideoStreamHeader(WebRtc_Word32 endpos)
+{
+ WebRtc_UWord32 chunktag;
+ _bytesRead += GetLE32(chunktag);
+ WebRtc_UWord32 chunksize;
+ _bytesRead += GetLE32(chunksize);
+
+ if (chunktag != MakeFourCc('s', 't', 'r', 'f'))
+ {
+ return -1;
+ }
+
+ _bytesRead += GetLE32(_videoFormatHeader.biSize);
+ _bytesRead += GetLE32(_videoFormatHeader.biWidth);
+ _bytesRead += GetLE32(_videoFormatHeader.biHeight);
+ _bytesRead += GetLE16(_videoFormatHeader.biPlanes);
+ _bytesRead += GetLE16(_videoFormatHeader.biBitCount);
+ _bytesRead += GetLE32(_videoFormatHeader.biCompression);
+ _bytesRead += GetLE32(_videoFormatHeader.biSizeImage);
+ _bytesRead += GetLE32(_videoFormatHeader.biXPelsPerMeter);
+ _bytesRead += GetLE32(_videoFormatHeader.biYPelsPerMeter);
+ _bytesRead += GetLE32(_videoFormatHeader.biClrUsed);
+ _bytesRead += GetLE32(_videoFormatHeader.biClrImportant);
+
+ if (chunksize > _videoFormatHeader.biSize)
+ {
+ const WebRtc_UWord32 size = chunksize - _videoFormatHeader.biSize;
+ const WebRtc_UWord32 readSize = MinValue(size, CODEC_CONFIG_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_videoConfigParameters), readSize);
+ _videoConfigLength = readSize;
+ WebRtc_Word32 skipSize = chunksize - _videoFormatHeader.biSize -
+ readSize;
+ if (skipSize > 0)
+ {
+ fseek(_aviFile, skipSize, SEEK_CUR);
+ _bytesRead += skipSize;
+ }
+ }
+
+ while (static_cast<long>(_bytesRead) < endpos)
+ {
+ WebRtc_UWord32 chunktag;
+ _bytesRead += GetLE32(chunktag);
+ WebRtc_UWord32 chunksize;
+ _bytesRead += GetLE32(chunksize);
+
+ if (chunktag == MakeFourCc('s', 't', 'r', 'n'))
+ {
+ const WebRtc_UWord32 size = MinValue(chunksize, STREAM_NAME_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_videoStreamName), size);
+ }
+ else if (chunktag == MakeFourCc('s', 't', 'r', 'd'))
+ {
+ const WebRtc_UWord32 size = MinValue(chunksize,
+ CODEC_CONFIG_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_videoConfigParameters), size);
+ _videoConfigLength = size;
+ }
+ else
+ {
+ fseek(_aviFile, chunksize, SEEK_CUR);
+ _bytesRead += chunksize;
+ }
+
+ if (feof(_aviFile))
+ {
+ return -1;
+ }
+ }
+ _videoStream.streamType = AviFile::AVI_VIDEO;
+ _videoStream.streamNumber = _nrStreams++;
+
+ return 0;
+}
+
+WebRtc_Word32 AviFile::ReadAVIAudioStreamHeader(WebRtc_Word32 endpos)
+{
+ WebRtc_UWord32 chunktag;
+ _bytesRead += GetLE32(chunktag);
+ WebRtc_UWord32 chunksize;
+ _bytesRead += GetLE32(chunksize);
+
+ if (chunktag != MakeFourCc('s', 't', 'r', 'f'))
+ {
+ return -1;
+ }
+
+ const size_t startRead = _bytesRead;
+ _bytesRead += GetLE16(_audioFormatHeader.wFormatTag);
+ _bytesRead += GetLE16(_audioFormatHeader.nChannels);
+ _bytesRead += GetLE32(_audioFormatHeader.nSamplesPerSec);
+ _bytesRead += GetLE32(_audioFormatHeader.nAvgBytesPerSec);
+ _bytesRead += GetLE16(_audioFormatHeader.nBlockAlign);
+ _bytesRead += GetLE16(_audioFormatHeader.wBitsPerSample);
+ _bytesRead += GetLE16(_audioFormatHeader.cbSize);
+
+ const WebRtc_UWord32 diffRead = chunksize - (_bytesRead - startRead);
+ if (diffRead > 0)
+ {
+ const WebRtc_UWord32 size = MinValue(diffRead, CODEC_CONFIG_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_audioConfigParameters), size);
+ }
+
+ while (static_cast<long>(_bytesRead) < endpos)
+ {
+ WebRtc_UWord32 chunktag;
+ _bytesRead += GetLE32(chunktag);
+ WebRtc_UWord32 chunksize;
+ _bytesRead += GetLE32(chunksize);
+
+ if (chunktag == MakeFourCc('s', 't', 'r', 'n'))
+ {
+ const WebRtc_UWord32 size = MinValue(chunksize, STREAM_NAME_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_audioStreamName), size);
+ }
+ else if (chunktag == MakeFourCc('s', 't', 'r', 'd'))
+ {
+ const WebRtc_UWord32 size = MinValue(chunksize,
+ CODEC_CONFIG_LENGTH);
+ _bytesRead += GetBuffer(
+ reinterpret_cast<WebRtc_UWord8*>(_audioConfigParameters), size);
+ }
+ else
+ {
+ fseek(_aviFile, chunksize, SEEK_CUR);
+ _bytesRead += chunksize;
+ }
+
+ if (feof(_aviFile))
+ {
+ return -1;
+ }
+ }
+ _audioStream.streamType = AviFile::AVI_AUDIO;
+ _audioStream.streamNumber = _nrStreams++;
+ return 0;
+}
+
+WebRtc_UWord32 AviFile::StreamAndTwoCharCodeToTag(WebRtc_Word32 streamNum,
+ const char* twoCharCode)
+{
+ WebRtc_UWord8 a = '0';
+ WebRtc_UWord8 b;
+ switch (streamNum)
+ {
+ case 1:
+ b = '1';
+ break;
+ case 2:
+ b = '2';
+ break;
+ default:
+ b = '0';
+ }
+ return MakeFourCc(a, b, twoCharCode[0], twoCharCode[1]);
+}
+
+void AviFile::ClearIndexList()
+{
+ while (!_indexList->Empty())
+ {
+ ListItem* listItem = _indexList->First();
+ if (listItem == 0)
+ {
+ break;
+ }
+
+ AVIINDEXENTRY* item = static_cast<AVIINDEXENTRY*>(listItem->GetItem());
+ if (item != NULL)
+ {
+ delete item;
+ }
+ _indexList->PopFront();
+ }
+}
+
+void AviFile::AddChunkToIndexList(WebRtc_UWord32 inChunkId,
+ WebRtc_UWord32 inFlags,
+ WebRtc_UWord32 inOffset,
+ WebRtc_UWord32 inSize)
+{
+ _indexList->PushBack(new AVIINDEXENTRY(inChunkId, inFlags, inOffset,
+ inSize));
+}
+
+void AviFile::WriteIndex()
+{
+ const WebRtc_UWord32 idxTag = MakeFourCc('i', 'd', 'x', '1');
+ _bytesWritten += PutLE32(idxTag);
+
+ // Size is unknown at this point. Update later.
+ _bytesWritten += PutLE32(0);
+ const size_t idxChunkSize = _bytesWritten;
+
+ for (ListItem* listItem = _indexList->First();
+ listItem != NULL;
+ listItem = _indexList->Next(listItem))
+ {
+ const AVIINDEXENTRY* item =
+ static_cast<AVIINDEXENTRY*>(listItem->GetItem());
+ if (item != NULL)
+ {
+ _bytesWritten += PutLE32(item->ckid);
+ _bytesWritten += PutLE32(item->dwFlags);
+ _bytesWritten += PutLE32(item->dwChunkOffset);
+ _bytesWritten += PutLE32(item->dwChunkLength);
+ }
+ }
+ PutLE32LengthFromCurrent(static_cast<long>(idxChunkSize));
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/avi_file.h b/trunk/src/modules/media_file/source/avi_file.h
new file mode 100644
index 0000000..73a19af
--- /dev/null
+++ b/trunk/src/modules/media_file/source/avi_file.h
@@ -0,0 +1,278 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Class for reading (x)or writing to an AVI file.
+// Note: the class cannot be used for reading and writing at the same time.
+#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
+
+#include <stdio.h>
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class ListWrapper;
+
+struct AVISTREAMHEADER
+{
+ AVISTREAMHEADER();
+ WebRtc_UWord32 fcc;
+ WebRtc_UWord32 cb;
+ WebRtc_UWord32 fccType;
+ WebRtc_UWord32 fccHandler;
+ WebRtc_UWord32 dwFlags;
+ WebRtc_UWord16 wPriority;
+ WebRtc_UWord16 wLanguage;
+ WebRtc_UWord32 dwInitialFrames;
+ WebRtc_UWord32 dwScale;
+ WebRtc_UWord32 dwRate;
+ WebRtc_UWord32 dwStart;
+ WebRtc_UWord32 dwLength;
+ WebRtc_UWord32 dwSuggestedBufferSize;
+ WebRtc_UWord32 dwQuality;
+ WebRtc_UWord32 dwSampleSize;
+ struct
+ {
+ WebRtc_Word16 left;
+ WebRtc_Word16 top;
+ WebRtc_Word16 right;
+ WebRtc_Word16 bottom;
+ } rcFrame;
+};
+
+struct BITMAPINFOHEADER
+{
+ BITMAPINFOHEADER();
+ WebRtc_UWord32 biSize;
+ WebRtc_UWord32 biWidth;
+ WebRtc_UWord32 biHeight;
+ WebRtc_UWord16 biPlanes;
+ WebRtc_UWord16 biBitCount;
+ WebRtc_UWord32 biCompression;
+ WebRtc_UWord32 biSizeImage;
+ WebRtc_UWord32 biXPelsPerMeter;
+ WebRtc_UWord32 biYPelsPerMeter;
+ WebRtc_UWord32 biClrUsed;
+ WebRtc_UWord32 biClrImportant;
+};
+
+struct WAVEFORMATEX
+{
+ WAVEFORMATEX();
+ WebRtc_UWord16 wFormatTag;
+ WebRtc_UWord16 nChannels;
+ WebRtc_UWord32 nSamplesPerSec;
+ WebRtc_UWord32 nAvgBytesPerSec;
+ WebRtc_UWord16 nBlockAlign;
+ WebRtc_UWord16 wBitsPerSample;
+ WebRtc_UWord16 cbSize;
+};
+
+class AviFile
+{
+public:
+ enum AVIStreamType
+ {
+ AVI_AUDIO = 0,
+ AVI_VIDEO = 1
+ };
+
+ // Unsigned, for comparison with must-be-unsigned types.
+ static const unsigned int CODEC_CONFIG_LENGTH = 64;
+ static const unsigned int STREAM_NAME_LENGTH = 32;
+
+ AviFile();
+ ~AviFile();
+
+ WebRtc_Word32 Open(AVIStreamType streamType, const char* fileName,
+ bool loop = false);
+
+ WebRtc_Word32 CreateVideoStream(const AVISTREAMHEADER& videoStreamHeader,
+ const BITMAPINFOHEADER& bitMapInfoHeader,
+ const WebRtc_UWord8* codecConfigParams,
+ WebRtc_Word32 codecConfigParamsLength);
+
+ WebRtc_Word32 CreateAudioStream(const AVISTREAMHEADER& audioStreamHeader,
+ const WAVEFORMATEX& waveFormatHeader);
+ WebRtc_Word32 Create(const WebRtc_Word8* fileName);
+
+ WebRtc_Word32 WriteAudio(const WebRtc_UWord8* data, WebRtc_Word32 length);
+ WebRtc_Word32 WriteVideo(const WebRtc_UWord8* data, WebRtc_Word32 length);
+
+ WebRtc_Word32 GetVideoStreamInfo(AVISTREAMHEADER& videoStreamHeader,
+ BITMAPINFOHEADER& bitmapInfo,
+ WebRtc_Word8* codecConfigParameters,
+ WebRtc_Word32& configLength);
+
+ WebRtc_Word32 GetDuration(WebRtc_Word32& durationMs);
+
+ WebRtc_Word32 GetAudioStreamInfo(WAVEFORMATEX& waveHeader);
+
+ WebRtc_Word32 ReadAudio(WebRtc_UWord8* data, WebRtc_Word32& length);
+ WebRtc_Word32 ReadVideo(WebRtc_UWord8* data, WebRtc_Word32& length);
+
+ WebRtc_Word32 Close();
+
+ static WebRtc_UWord32 MakeFourCc(WebRtc_UWord8 ch0, WebRtc_UWord8 ch1,
+ WebRtc_UWord8 ch2, WebRtc_UWord8 ch3);
+
+private:
+ enum AVIFileMode
+ {
+ NotSet,
+ Read,
+ Write
+ };
+
+ struct AVIINDEXENTRY
+ {
+ AVIINDEXENTRY(WebRtc_UWord32 inckid, WebRtc_UWord32 indwFlags,
+ WebRtc_UWord32 indwChunkOffset,
+ WebRtc_UWord32 indwChunkLength);
+ WebRtc_UWord32 ckid;
+ WebRtc_UWord32 dwFlags;
+ WebRtc_UWord32 dwChunkOffset;
+ WebRtc_UWord32 dwChunkLength;
+ };
+
+ WebRtc_Word32 PrepareDataChunkHeaders();
+
+ WebRtc_Word32 ReadMoviSubChunk(WebRtc_UWord8* data, WebRtc_Word32& length,
+ WebRtc_UWord32 tag1,
+ WebRtc_UWord32 tag2 = 0);
+
+ WebRtc_Word32 WriteRIFF();
+ WebRtc_Word32 WriteHeaders();
+ WebRtc_Word32 WriteAVIMainHeader();
+ WebRtc_Word32 WriteAVIStreamHeaders();
+ WebRtc_Word32 WriteAVIVideoStreamHeaders();
+ WebRtc_Word32 WriteAVIVideoStreamHeaderChunks();
+ WebRtc_Word32 WriteAVIAudioStreamHeaders();
+ WebRtc_Word32 WriteAVIAudioStreamHeaderChunks();
+
+ WebRtc_Word32 WriteMoviStart();
+
+ size_t PutByte(WebRtc_UWord8 byte);
+ size_t PutLE16(WebRtc_UWord16 word);
+ size_t PutLE32(WebRtc_UWord32 word);
+ size_t PutBuffer(const WebRtc_UWord8* str, size_t size);
+ size_t PutBufferZ(const char* str);
+ long PutLE32LengthFromCurrent(long startPos);
+ void PutLE32AtPos(long pos, WebRtc_UWord32 word);
+
+ size_t GetByte(WebRtc_UWord8& word);
+ size_t GetLE16(WebRtc_UWord16& word);
+ size_t GetLE32(WebRtc_UWord32& word);
+ size_t GetBuffer(WebRtc_UWord8* str, size_t size);
+
+ void CloseRead();
+ void CloseWrite();
+
+ void ResetMembers();
+ void ResetComplexMembers();
+
+ WebRtc_Word32 ReadRIFF();
+ WebRtc_Word32 ReadHeaders();
+ WebRtc_Word32 ReadAVIMainHeader();
+ WebRtc_Word32 ReadAVIVideoStreamHeader(WebRtc_Word32 endpos);
+ WebRtc_Word32 ReadAVIAudioStreamHeader(WebRtc_Word32 endpos);
+
+ WebRtc_UWord32 StreamAndTwoCharCodeToTag(WebRtc_Word32 streamNum,
+ const char* twoCharCode);
+
+ void ClearIndexList();
+ void AddChunkToIndexList(WebRtc_UWord32 inChunkId, WebRtc_UWord32 inFlags,
+ WebRtc_UWord32 inOffset, WebRtc_UWord32 inSize);
+
+ void WriteIndex();
+
+private:
+ struct AVIMAINHEADER
+ {
+ AVIMAINHEADER();
+ WebRtc_UWord32 fcc;
+ WebRtc_UWord32 cb;
+ WebRtc_UWord32 dwMicroSecPerFrame;
+ WebRtc_UWord32 dwMaxBytesPerSec;
+ WebRtc_UWord32 dwPaddingGranularity;
+ WebRtc_UWord32 dwFlags;
+ WebRtc_UWord32 dwTotalFrames;
+ WebRtc_UWord32 dwInitialFrames;
+ WebRtc_UWord32 dwStreams;
+ WebRtc_UWord32 dwSuggestedBufferSize;
+ WebRtc_UWord32 dwWidth;
+ WebRtc_UWord32 dwHeight;
+ WebRtc_UWord32 dwReserved[4];
+ };
+
+ struct AVIStream
+ {
+ AVIStreamType streamType;
+ int streamNumber;
+ };
+
+ CriticalSectionWrapper* _crit;
+ FILE* _aviFile;
+ AVIMAINHEADER _aviHeader;
+ AVISTREAMHEADER _videoStreamHeader;
+ AVISTREAMHEADER _audioStreamHeader;
+ BITMAPINFOHEADER _videoFormatHeader;
+ WAVEFORMATEX _audioFormatHeader;
+
+ WebRtc_Word8 _videoConfigParameters[CODEC_CONFIG_LENGTH];
+ WebRtc_Word32 _videoConfigLength;
+ WebRtc_Word8 _videoStreamName[STREAM_NAME_LENGTH];
+ WebRtc_Word32 _videoStreamNameLength;
+ WebRtc_Word8 _audioConfigParameters[CODEC_CONFIG_LENGTH];
+ WebRtc_Word8 _audioStreamName[STREAM_NAME_LENGTH];
+
+ AVIStream _videoStream;
+ AVIStream _audioStream;
+
+ WebRtc_Word32 _nrStreams;
+ WebRtc_Word32 _aviLength;
+ WebRtc_Word32 _dataLength;
+ size_t _bytesRead;
+ size_t _dataStartByte;
+ WebRtc_Word32 _framesRead;
+ WebRtc_Word32 _videoFrames;
+ WebRtc_Word32 _audioFrames;
+
+ bool _reading;
+ AVIStreamType _openedAs;
+ bool _loop;
+ bool _writing;
+
+ size_t _bytesWritten;
+
+ size_t _riffSizeMark;
+ size_t _moviSizeMark;
+ size_t _totNumFramesMark;
+ size_t _videoStreamLengthMark;
+ size_t _audioStreamLengthMark;
+ WebRtc_Word32 _moviListOffset;
+
+ bool _writeAudioStream;
+ bool _writeVideoStream;
+
+ AVIFileMode _aviMode;
+ WebRtc_UWord8* _videoCodecConfigParams;
+ WebRtc_Word32 _videoCodecConfigParamsLength;
+
+ WebRtc_UWord32 _videoStreamDataChunkPrefix;
+ WebRtc_UWord32 _audioStreamDataChunkPrefix;
+ bool _created;
+
+ ListWrapper* _indexList; // Elements are of type AVIINDEXENTRY.
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
diff --git a/trunk/src/modules/media_file/source/media_file.gypi b/trunk/src/modules/media_file/source/media_file.gypi
new file mode 100644
index 0000000..2f9ffe9
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file.gypi
@@ -0,0 +1,67 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'media_file',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'defines': [
+ 'WEBRTC_MODULE_UTILITY_VIDEO', # for compiling support for video recording
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/media_file.h',
+ '../interface/media_file_defines.h',
+ 'avi_file.cc',
+ 'avi_file.h',
+ 'media_file_impl.cc',
+ 'media_file_impl.h',
+ 'media_file_utility.cc',
+ 'media_file_utility.h',
+ ], # source
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'media_file_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'media_file',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'media_file_unittest.cc',
+ ],
+ }, # media_file_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/media_file/source/media_file_impl.cc b/trunk/src/modules/media_file/source/media_file_impl.cc
new file mode 100644
index 0000000..6c42fa1
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file_impl.cc
@@ -0,0 +1,1372 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "media_file_impl.h"
+#include "tick_util.h"
+#include "trace.h"
+
+#if (defined(WIN32) || defined(WINCE))
+ #define STR_CASE_CMP _stricmp
+ #define STR_NCASE_CMP _strnicmp
+#else
+ #define STR_CASE_CMP strcasecmp
+ #define STR_NCASE_CMP strncasecmp
+#endif
+
+namespace webrtc {
+MediaFile* MediaFile::CreateMediaFile(const WebRtc_Word32 id)
+{
+ return new MediaFileImpl(id);
+}
+
+void MediaFile::DestroyMediaFile(MediaFile* module)
+{
+ delete static_cast<MediaFileImpl*>(module);
+}
+
+MediaFileImpl::MediaFileImpl(const WebRtc_Word32 id)
+ : _id(id),
+ _crit(CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCrit(CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrFileUtilityObj(NULL),
+ codec_info_(),
+ _ptrInStream(NULL),
+ _ptrOutStream(NULL),
+ _fileFormat((FileFormats)-1),
+ _recordDurationMs(0),
+ _playoutPositionMs(0),
+ _notificationMs(0),
+ _playingActive(false),
+ _recordingActive(false),
+ _isStereo(false),
+ _openFile(false),
+ _fileName(),
+ _ptrCallback(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, id, "Created");
+
+ codec_info_.plname[0] = '\0';
+ _fileName[0] = '\0';
+}
+
+
+MediaFileImpl::~MediaFileImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id, "~MediaFileImpl()");
+ {
+ CriticalSectionScoped lock(_crit);
+
+ if(_playingActive)
+ {
+ StopPlaying();
+ }
+
+ if(_recordingActive)
+ {
+ StopRecording();
+ }
+
+ delete _ptrFileUtilityObj;
+
+ if(_openFile)
+ {
+ delete _ptrInStream;
+ _ptrInStream = NULL;
+ delete _ptrOutStream;
+ _ptrOutStream = NULL;
+ }
+ }
+
+ delete _crit;
+ delete _callbackCrit;
+}
+
+WebRtc_Word32 MediaFileImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::TimeUntilNextProcess()
+{
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceFile,
+ _id,
+ "TimeUntilNextProcess: This method is not used by MediaFile class.");
+ return -1;
+}
+
+WebRtc_Word32 MediaFileImpl::Process()
+{
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Process: This method is not used by MediaFile class.");
+ return -1;
+}
+
+WebRtc_Word32 MediaFileImpl::PlayoutAVIVideoData(
+ WebRtc_Word8* buffer,
+ WebRtc_UWord32& dataLengthInBytes)
+{
+ return PlayoutData( buffer, dataLengthInBytes, true);
+}
+
+WebRtc_Word32 MediaFileImpl::PlayoutAudioData(WebRtc_Word8* buffer,
+ WebRtc_UWord32& dataLengthInBytes)
+{
+ return PlayoutData( buffer, dataLengthInBytes, false);
+}
+
+WebRtc_Word32 MediaFileImpl::PlayoutData(WebRtc_Word8* buffer,
+ WebRtc_UWord32& dataLengthInBytes,
+ bool video)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "MediaFileImpl::PlayoutData(buffer= 0x%x, bufLen= %ld)",
+ buffer, dataLengthInBytes);
+
+ const WebRtc_UWord32 bufferLengthInBytes = dataLengthInBytes;
+ dataLengthInBytes = 0;
+
+ if(buffer == NULL || bufferLengthInBytes == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Buffer pointer or length is NULL!");
+ return -1;
+ }
+
+ WebRtc_Word32 bytesRead = 0;
+ {
+ CriticalSectionScoped lock(_crit);
+
+ if(!_playingActive)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Not currently playing!");
+ return -1;
+ }
+
+ if(!_ptrFileUtilityObj)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Playing, but no FileUtility object!");
+ StopPlaying();
+ return -1;
+ }
+
+ switch(_fileFormat)
+ {
+ case kFileFormatPcm32kHzFile:
+ case kFileFormatPcm16kHzFile:
+ case kFileFormatPcm8kHzFile:
+ bytesRead = _ptrFileUtilityObj->ReadPCMData(
+ *_ptrInStream,
+ buffer,
+ bufferLengthInBytes);
+ break;
+ case kFileFormatCompressedFile:
+ bytesRead = _ptrFileUtilityObj->ReadCompressedData(
+ *_ptrInStream,
+ buffer,
+ bufferLengthInBytes);
+ break;
+ case kFileFormatWavFile:
+ bytesRead = _ptrFileUtilityObj->ReadWavDataAsMono(
+ *_ptrInStream,
+ buffer,
+ bufferLengthInBytes);
+ break;
+ case kFileFormatPreencodedFile:
+ bytesRead = _ptrFileUtilityObj->ReadPreEncodedData(
+ *_ptrInStream,
+ buffer,
+ bufferLengthInBytes);
+ if(bytesRead > 0)
+ {
+ dataLengthInBytes = bytesRead;
+ return 0;
+ }
+ break;
+ case kFileFormatAviFile:
+ {
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ if(video)
+ {
+ bytesRead = _ptrFileUtilityObj->ReadAviVideoData(
+ buffer,
+ bufferLengthInBytes);
+ }
+ else
+ {
+ bytesRead = _ptrFileUtilityObj->ReadAviAudioData(
+ buffer,
+ bufferLengthInBytes);
+ }
+ break;
+#else
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Invalid file format: %d", kFileFormatAviFile);
+ assert(false);
+ break;
+#endif
+ }
+ }
+
+ if( bytesRead > 0)
+ {
+ dataLengthInBytes =(WebRtc_UWord32) bytesRead;
+ }
+ }
+ HandlePlayCallbacks(bytesRead);
+ return 0;
+}
+
+void MediaFileImpl::HandlePlayCallbacks(WebRtc_Word32 bytesRead)
+{
+ bool playEnded = false;
+ WebRtc_UWord32 callbackNotifyMs = 0;
+
+ if(bytesRead > 0)
+ {
+ // Check if it's time for PlayNotification(..).
+ _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
+ if(_notificationMs)
+ {
+ if(_playoutPositionMs >= _notificationMs)
+ {
+ _notificationMs = 0;
+ callbackNotifyMs = _playoutPositionMs;
+ }
+ }
+ }
+ else
+ {
+ // If no bytes were read assume end of file.
+ StopPlaying();
+ playEnded = true;
+ }
+
+ // Only _callbackCrit may and should be taken when making callbacks.
+ CriticalSectionScoped lock(_callbackCrit);
+ if(_ptrCallback)
+ {
+ if(callbackNotifyMs)
+ {
+ _ptrCallback->PlayNotification(_id, callbackNotifyMs);
+ }
+ if(playEnded)
+ {
+ _ptrCallback->PlayFileEnded(_id);
+ }
+ }
+}
+
+WebRtc_Word32 MediaFileImpl::PlayoutStereoData(
+ WebRtc_Word8* bufferLeft,
+ WebRtc_Word8* bufferRight,
+ WebRtc_UWord32& dataLengthInBytes)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "MediaFileImpl::PlayoutStereoData(Left = 0x%x, Right = 0x%x,\
+ Len= %ld)",
+ bufferLeft,
+ bufferRight,
+ dataLengthInBytes);
+
+ const WebRtc_UWord32 bufferLengthInBytes = dataLengthInBytes;
+ dataLengthInBytes = 0;
+
+ if(bufferLeft == NULL || bufferRight == NULL || bufferLengthInBytes == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "A buffer pointer or the length is NULL!");
+ return -1;
+ }
+
+ bool playEnded = false;
+ WebRtc_UWord32 callbackNotifyMs = 0;
+ {
+ CriticalSectionScoped lock(_crit);
+
+ if(!_playingActive || !_isStereo)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Not currently playing stereo!");
+ return -1;
+ }
+
+ if(!_ptrFileUtilityObj)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "Playing stereo, but the FileUtility objects is NULL!");
+ StopPlaying();
+ return -1;
+ }
+
+ // Stereo playout only supported for WAV files.
+ WebRtc_Word32 bytesRead = 0;
+ switch(_fileFormat)
+ {
+ case kFileFormatWavFile:
+ bytesRead = _ptrFileUtilityObj->ReadWavDataAsStereo(
+ *_ptrInStream,
+ bufferLeft,
+ bufferRight,
+ bufferLengthInBytes);
+ break;
+ default:
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Trying to read non-WAV as stereo audio\
+ (not supported)");
+ break;
+ }
+
+ if(bytesRead > 0)
+ {
+ dataLengthInBytes = bytesRead;
+
+ // Check if it's time for PlayNotification(..).
+ _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
+ if(_notificationMs)
+ {
+ if(_playoutPositionMs >= _notificationMs)
+ {
+ _notificationMs = 0;
+ callbackNotifyMs = _playoutPositionMs;
+ }
+ }
+ }
+ else
+ {
+ // If no bytes were read assume end of file.
+ StopPlaying();
+ playEnded = true;
+ }
+ }
+
+ CriticalSectionScoped lock(_callbackCrit);
+ if(_ptrCallback)
+ {
+ if(callbackNotifyMs)
+ {
+ _ptrCallback->PlayNotification(_id, callbackNotifyMs);
+ }
+ if(playEnded)
+ {
+ _ptrCallback->PlayFileEnded(_id);
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StartPlayingAudioFile(
+ const WebRtc_Word8* fileName,
+ const WebRtc_UWord32 notificationTimeMs,
+ const bool loop,
+ const FileFormats format,
+ const CodecInst* codecInst,
+ const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs)
+{
+ const bool videoOnly = false;
+ return StartPlayingFile(fileName, notificationTimeMs, loop, videoOnly,
+ format, codecInst, startPointMs, stopPointMs);
+}
+
+
+WebRtc_Word32 MediaFileImpl::StartPlayingVideoFile(const WebRtc_Word8* fileName,
+ const bool loop,
+ bool videoOnly,
+ const FileFormats format)
+{
+
+ const WebRtc_UWord32 notificationTimeMs = 0;
+ const WebRtc_UWord32 startPointMs = 0;
+ const WebRtc_UWord32 stopPointMs = 0;
+ return StartPlayingFile(fileName, notificationTimeMs, loop, videoOnly,
+ format, 0, startPointMs, stopPointMs);
+}
+
+WebRtc_Word32 MediaFileImpl::StartPlayingFile(
+ const WebRtc_Word8* fileName,
+ const WebRtc_UWord32 notificationTimeMs,
+ const bool loop,
+ bool videoOnly,
+ const FileFormats format,
+ const CodecInst* codecInst,
+ const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs)
+{
+
+ if(!ValidFileName(fileName))
+ {
+ return -1;
+ }
+ if(!ValidFileFormat(format,codecInst))
+ {
+ return -1;
+ }
+ if(!ValidFilePositions(startPointMs,stopPointMs))
+ {
+ return -1;
+ }
+
+ // Check that the file will play longer than notificationTimeMs ms.
+ if((startPointMs && stopPointMs && !loop) &&
+ (notificationTimeMs > (stopPointMs - startPointMs)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "specified notification time is longer than amount of ms that will\
+ be played");
+ return -1;
+ }
+
+ FileWrapper* inputStream = FileWrapper::Create();
+ if(inputStream == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "Failed to allocate input stream for file %s", fileName);
+ return -1;
+ }
+
+ // TODO (hellner): make all formats support reading from stream.
+ bool useStream = (format != kFileFormatAviFile);
+ if( useStream)
+ {
+ if(inputStream->OpenFile(fileName, true, loop) != 0)
+ {
+ delete inputStream;
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Could not open input file %s", fileName);
+ return -1;
+ }
+ }
+
+ if(StartPlayingStream(*inputStream, fileName, loop, notificationTimeMs,
+ format, codecInst, startPointMs, stopPointMs,
+ videoOnly) == -1)
+ {
+ if( useStream)
+ {
+ inputStream->CloseFile();
+ }
+ delete inputStream;
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_crit);
+ _openFile = true;
+ strncpy(_fileName, fileName, sizeof(_fileName));
+ _fileName[sizeof(_fileName) - 1] = '\0';
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StartPlayingAudioStream(
+ InStream& stream,
+ const WebRtc_UWord32 notificationTimeMs,
+ const FileFormats format,
+ const CodecInst* codecInst,
+ const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs)
+{
+ return StartPlayingStream(stream, 0, false, notificationTimeMs, format,
+ codecInst, startPointMs, stopPointMs);
+}
+
+WebRtc_Word32 MediaFileImpl::StartPlayingStream(
+ InStream& stream,
+ const WebRtc_Word8* filename,
+ bool loop,
+ const WebRtc_UWord32 notificationTimeMs,
+ const FileFormats format,
+ const CodecInst* codecInst,
+ const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs,
+ bool videoOnly)
+{
+ if(!ValidFileFormat(format,codecInst))
+ {
+ return -1;
+ }
+
+ if(!ValidFilePositions(startPointMs,stopPointMs))
+ {
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_crit);
+ if(_playingActive || _recordingActive)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "StartPlaying called, but already playing or recording file %s",
+ (_fileName[0] == '\0') ? "(name not set)" : _fileName);
+ return -1;
+ }
+
+ if(_ptrFileUtilityObj != NULL)
+ {
+ WEBRTC_TRACE(kTraceError,
+ kTraceFile,
+ _id,
+ "StartPlaying called, but FileUtilityObj already exists!");
+ StopPlaying();
+ return -1;
+ }
+
+ _ptrFileUtilityObj = new ModuleFileUtility(_id);
+ if(_ptrFileUtilityObj == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "Failed to create FileUtilityObj!");
+ return -1;
+ }
+
+ switch(format)
+ {
+ case kFileFormatWavFile:
+ {
+ if(_ptrFileUtilityObj->InitWavReading(stream, startPointMs,
+ stopPointMs) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a valid WAV file!");
+ StopPlaying();
+ return -1;
+ }
+ _fileFormat = kFileFormatWavFile;
+ break;
+ }
+ case kFileFormatCompressedFile:
+ {
+ if(_ptrFileUtilityObj->InitCompressedReading(stream, startPointMs,
+ stopPointMs) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a valid Compressed file!");
+ StopPlaying();
+ return -1;
+ }
+ _fileFormat = kFileFormatCompressedFile;
+ break;
+ }
+ case kFileFormatPcm8kHzFile:
+ case kFileFormatPcm16kHzFile:
+ case kFileFormatPcm32kHzFile:
+ {
+ // ValidFileFormat() called in the beginneing of this function
+ // prevents codecInst from being NULL here.
+ assert(codecInst != NULL);
+ if(!ValidFrequency(codecInst->plfreq) ||
+ _ptrFileUtilityObj->InitPCMReading(stream, startPointMs,
+ stopPointMs,
+ codecInst->plfreq) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a valid raw 8 or 16 KHz PCM file!");
+ StopPlaying();
+ return -1;
+ }
+
+ _fileFormat = format;
+ break;
+ }
+ case kFileFormatPreencodedFile:
+ {
+ // ValidFileFormat() called in the beginneing of this function
+ // prevents codecInst from being NULL here.
+ assert(codecInst != NULL);
+ if(_ptrFileUtilityObj->InitPreEncodedReading(stream, *codecInst) ==
+ -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a valid PreEncoded file!");
+ StopPlaying();
+ return -1;
+ }
+
+ _fileFormat = kFileFormatPreencodedFile;
+ break;
+ }
+ case kFileFormatAviFile:
+ {
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ if(_ptrFileUtilityObj->InitAviReading( filename, videoOnly, loop))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a valid AVI file!");
+ StopPlaying();
+
+ return -1;
+ }
+
+ _ptrFileUtilityObj->codec_info(codec_info_);
+
+ _fileFormat = kFileFormatAviFile;
+ break;
+#else
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Invalid file format: %d", kFileFormatAviFile);
+ assert(false);
+ break;
+#endif
+ }
+ }
+ if(_ptrFileUtilityObj->codec_info(codec_info_) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to retrieve codec info!");
+ StopPlaying();
+ return -1;
+ }
+
+ _isStereo = (codec_info_.channels == 2);
+ if(_isStereo && (_fileFormat != kFileFormatWavFile))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Stereo is only allowed for WAV files");
+ StopPlaying();
+ return -1;
+ }
+ _playingActive = true;
+ _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
+ _ptrInStream = &stream;
+ _notificationMs = notificationTimeMs;
+
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StopPlaying()
+{
+
+ CriticalSectionScoped lock(_crit);
+ _isStereo = false;
+ if(_ptrFileUtilityObj)
+ {
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ }
+ if(_ptrInStream)
+ {
+ // If MediaFileImpl opened the InStream it must be reclaimed here.
+ if(_openFile)
+ {
+ delete _ptrInStream;
+ _openFile = false;
+ }
+ _ptrInStream = NULL;
+ }
+
+ codec_info_.pltype = 0;
+ codec_info_.plname[0] = '\0';
+
+ if(!_playingActive)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "playing is not active!");
+ return -1;
+ }
+
+ _playingActive = false;
+ return 0;
+}
+
+bool MediaFileImpl::IsPlaying()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsPlaying()");
+ CriticalSectionScoped lock(_crit);
+ return _playingActive;
+}
+
+WebRtc_Word32 MediaFileImpl::IncomingAudioData(
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 bufferLengthInBytes)
+{
+ return IncomingAudioVideoData( buffer, bufferLengthInBytes, false);
+}
+
+WebRtc_Word32 MediaFileImpl::IncomingAVIVideoData(
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 bufferLengthInBytes)
+{
+ return IncomingAudioVideoData( buffer, bufferLengthInBytes, true);
+}
+
+WebRtc_Word32 MediaFileImpl::IncomingAudioVideoData(
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 bufferLengthInBytes,
+ const bool video)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "MediaFile::IncomingData(buffer= 0x%x, bufLen= %hd",
+ buffer, bufferLengthInBytes);
+
+ if(buffer == NULL || bufferLengthInBytes == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Buffer pointer or length is NULL!");
+ return -1;
+ }
+
+ bool recordingEnded = false;
+ WebRtc_UWord32 callbackNotifyMs = 0;
+ {
+ CriticalSectionScoped lock(_crit);
+
+ if(!_recordingActive)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Not currently recording!");
+ return -1;
+ }
+ if(_ptrOutStream == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Recording is active, but output stream is NULL!");
+ assert(false);
+ return -1;
+ }
+
+ WebRtc_Word32 bytesWritten = 0;
+ WebRtc_UWord32 samplesWritten = codec_info_.pacsize;
+ if(_ptrFileUtilityObj)
+ {
+ switch(_fileFormat)
+ {
+ case kFileFormatPcm8kHzFile:
+ case kFileFormatPcm16kHzFile:
+ case kFileFormatPcm32kHzFile:
+ bytesWritten = _ptrFileUtilityObj->WritePCMData(
+ *_ptrOutStream,
+ buffer,
+ bufferLengthInBytes);
+
+ // Sample size is 2 bytes.
+ if(bytesWritten > 0)
+ {
+ samplesWritten = bytesWritten/sizeof(WebRtc_Word16);
+ }
+ break;
+ case kFileFormatCompressedFile:
+ bytesWritten = _ptrFileUtilityObj->WriteCompressedData(
+ *_ptrOutStream, buffer, bufferLengthInBytes);
+ break;
+ case kFileFormatWavFile:
+ bytesWritten = _ptrFileUtilityObj->WriteWavData(
+ *_ptrOutStream,
+ buffer,
+ bufferLengthInBytes);
+ if(bytesWritten > 0 && STR_NCASE_CMP(codec_info_.plname,
+ "L16", 4) == 0)
+ {
+ // Sample size is 2 bytes.
+ samplesWritten = bytesWritten/sizeof(WebRtc_Word16);
+ }
+ break;
+ case kFileFormatPreencodedFile:
+ bytesWritten = _ptrFileUtilityObj->WritePreEncodedData(
+ *_ptrOutStream, buffer, bufferLengthInBytes);
+ break;
+ case kFileFormatAviFile:
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ if(video)
+ {
+ bytesWritten = _ptrFileUtilityObj->WriteAviVideoData(
+ buffer, bufferLengthInBytes);
+ }else
+ {
+ bytesWritten = _ptrFileUtilityObj->WriteAviAudioData(
+ buffer, bufferLengthInBytes);
+ }
+ break;
+#else
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Invalid file format: %d", kFileFormatAviFile);
+ assert(false);
+ break;
+#endif
+ }
+ } else {
+ // TODO (hellner): quick look at the code makes me think that this
+ // code is never executed. Remove?
+ if(_ptrOutStream)
+ {
+ if(_ptrOutStream->Write(buffer, bufferLengthInBytes))
+ {
+ bytesWritten = bufferLengthInBytes;
+ }
+ }
+ }
+
+ if(!video)
+ {
+ _recordDurationMs += samplesWritten / (codec_info_.plfreq / 1000);
+ }
+
+ // Check if it's time for RecordNotification(..).
+ if(_notificationMs)
+ {
+ if(_recordDurationMs >= _notificationMs)
+ {
+ _notificationMs = 0;
+ callbackNotifyMs = _recordDurationMs;
+ }
+ }
+ if(bytesWritten < (WebRtc_Word32)bufferLengthInBytes)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Failed to write all requested bytes!");
+ StopRecording();
+ recordingEnded = true;
+ }
+ }
+
+ // Only _callbackCrit may and should be taken when making callbacks.
+ CriticalSectionScoped lock(_callbackCrit);
+ if(_ptrCallback)
+ {
+ if(callbackNotifyMs)
+ {
+ _ptrCallback->RecordNotification(_id, callbackNotifyMs);
+ }
+ if(recordingEnded)
+ {
+ _ptrCallback->RecordFileEnded(_id);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StartRecordingAudioFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs,
+ const WebRtc_UWord32 maxSizeBytes)
+{
+ VideoCodec dummyCodecInst;
+ return StartRecordingFile(fileName, format, codecInst, dummyCodecInst,
+ notificationTimeMs, maxSizeBytes);
+}
+
+
+WebRtc_Word32 MediaFileImpl::StartRecordingVideoFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ bool videoOnly)
+{
+ const WebRtc_UWord32 notificationTimeMs = 0;
+ const WebRtc_UWord32 maxSizeBytes = 0;
+
+ return StartRecordingFile(fileName, format, codecInst, videoCodecInst,
+ notificationTimeMs, maxSizeBytes, videoOnly);
+}
+
+WebRtc_Word32 MediaFileImpl::StartRecordingFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ const WebRtc_UWord32 notificationTimeMs,
+ const WebRtc_UWord32 maxSizeBytes,
+ bool videoOnly)
+{
+
+ if(!ValidFileName(fileName))
+ {
+ return -1;
+ }
+ if(!ValidFileFormat(format,&codecInst))
+ {
+ return -1;
+ }
+
+ FileWrapper* outputStream = FileWrapper::Create();
+ if(outputStream == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "Failed to allocate memory for output stream");
+ return -1;
+ }
+
+ // TODO (hellner): make all formats support writing to stream.
+ const bool useStream = ( format != kFileFormatAviFile);
+ if( useStream)
+ {
+ if(outputStream->OpenFile(fileName, false) != 0)
+ {
+ delete outputStream;
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Could not open output file '%s' for writing!",
+ fileName);
+ return -1;
+ }
+ }
+ if(maxSizeBytes)
+ {
+ outputStream->SetMaxFileSize(maxSizeBytes);
+ }
+
+ if(StartRecordingStream(*outputStream, fileName, format, codecInst,
+ videoCodecInst, notificationTimeMs,
+ videoOnly) == -1)
+ {
+ if( useStream)
+ {
+ outputStream->CloseFile();
+ }
+ delete outputStream;
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_crit);
+ _openFile = true;
+ strncpy(_fileName, fileName, sizeof(_fileName));
+ _fileName[sizeof(_fileName) - 1] = '\0';
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StartRecordingAudioStream(
+ OutStream& stream,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs)
+{
+ VideoCodec dummyCodecInst;
+ return StartRecordingStream(stream, 0, format, codecInst, dummyCodecInst,
+ notificationTimeMs);
+}
+
+WebRtc_Word32 MediaFileImpl::StartRecordingStream(
+ OutStream& stream,
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ const WebRtc_UWord32 notificationTimeMs,
+ bool videoOnly)
+{
+
+ // Check codec info
+ if(!ValidFileFormat(format,&codecInst))
+ {
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_crit);
+ if(_recordingActive || _playingActive)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "StartRecording called, but already recording or playing file %s!",
+ _fileName);
+ return -1;
+ }
+
+ if(_ptrFileUtilityObj != NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "StartRecording called, but fileUtilityObj already exists!");
+ StopRecording();
+ return -1;
+ }
+
+ _ptrFileUtilityObj = new ModuleFileUtility(_id);
+ if(_ptrFileUtilityObj == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "Cannot allocate fileUtilityObj!");
+ return -1;
+ }
+
+ CodecInst tmpAudioCodec;
+ memcpy(&tmpAudioCodec, &codecInst, sizeof(CodecInst));
+ switch(format)
+ {
+ case kFileFormatWavFile:
+ {
+ if(_ptrFileUtilityObj->InitWavWriting(stream, codecInst) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to initialize WAV file!");
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+ _fileFormat = kFileFormatWavFile;
+ break;
+ }
+ case kFileFormatCompressedFile:
+ {
+ // Write compression codec name at beginning of file
+ if(_ptrFileUtilityObj->InitCompressedWriting(stream, codecInst) ==
+ -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to initialize Compressed file!");
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+ _fileFormat = kFileFormatCompressedFile;
+ break;
+ }
+ case kFileFormatPcm8kHzFile:
+ case kFileFormatPcm16kHzFile:
+ {
+ if(!ValidFrequency(codecInst.plfreq) ||
+ _ptrFileUtilityObj->InitPCMWriting(stream, codecInst.plfreq) ==
+ -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to initialize 8 or 16KHz PCM file!");
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+ _fileFormat = format;
+ break;
+ }
+ case kFileFormatPreencodedFile:
+ {
+ if(_ptrFileUtilityObj->InitPreEncodedWriting(stream, codecInst) ==
+ -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to initialize Pre-Encoded file!");
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+
+ _fileFormat = kFileFormatPreencodedFile;
+ break;
+ }
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ case kFileFormatAviFile:
+ {
+ if( (_ptrFileUtilityObj->InitAviWriting(
+ fileName,
+ codecInst,
+ videoCodecInst,videoOnly) == -1) ||
+ (_ptrFileUtilityObj->codec_info(tmpAudioCodec) != 0))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Failed to initialize AVI file!");
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+ _fileFormat = kFileFormatAviFile;
+ break;
+ }
+#endif
+ default:
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Invalid file format %d specified!", format);
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ return -1;
+ }
+ }
+ _isStereo = (tmpAudioCodec.channels == 2);
+ if(_isStereo)
+ {
+ if(_fileFormat != kFileFormatWavFile)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "Stereo is only allowed for WAV files");
+ StopRecording();
+ return -1;
+ }
+ if((STR_NCASE_CMP(tmpAudioCodec.plname, "L16", 4) != 0) &&
+ (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMU", 5) != 0) &&
+ (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMA", 5) != 0))
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceFile,
+ _id,
+ "Stereo is only allowed for codec PCMU, PCMA and L16 ");
+ StopRecording();
+ return -1;
+ }
+ }
+ memcpy(&codec_info_, &tmpAudioCodec, sizeof(CodecInst));
+ _recordingActive = true;
+ _ptrOutStream = &stream;
+ _notificationMs = notificationTimeMs;
+ _recordDurationMs = 0;
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::StopRecording()
+{
+
+ CriticalSectionScoped lock(_crit);
+ if(!_recordingActive)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
+ "recording is not active!");
+ return -1;
+ }
+
+ _isStereo = false;
+
+ if(_ptrFileUtilityObj != NULL)
+ {
+ // Both AVI and WAV header has to be updated before closing the stream
+ // because they contain size information.
+ if((_fileFormat == kFileFormatWavFile) &&
+ (_ptrOutStream != NULL))
+ {
+ _ptrFileUtilityObj->UpdateWavHeader(*_ptrOutStream);
+ }
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ else if( _fileFormat == kFileFormatAviFile)
+ {
+ _ptrFileUtilityObj->CloseAviFile( );
+ }
+#endif
+ delete _ptrFileUtilityObj;
+ _ptrFileUtilityObj = NULL;
+ }
+
+ if(_ptrOutStream != NULL)
+ {
+ // If MediaFileImpl opened the OutStream it must be reclaimed here.
+ if(_openFile)
+ {
+ delete _ptrOutStream;
+ _openFile = false;
+ }
+ _ptrOutStream = NULL;
+ }
+
+ _recordingActive = false;
+ codec_info_.pltype = 0;
+ codec_info_.plname[0] = '\0';
+
+ return 0;
+}
+
+bool MediaFileImpl::IsRecording()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsRecording()");
+ CriticalSectionScoped lock(_crit);
+ return _recordingActive;
+}
+
+WebRtc_Word32 MediaFileImpl::RecordDurationMs(WebRtc_UWord32& durationMs)
+{
+
+ CriticalSectionScoped lock(_crit);
+ if(!_recordingActive)
+ {
+ durationMs = 0;
+ return -1;
+ }
+ durationMs = _recordDurationMs;
+ return 0;
+}
+
+bool MediaFileImpl::IsStereo()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsStereo()");
+ CriticalSectionScoped lock(_crit);
+ return _isStereo;
+}
+
+WebRtc_Word32 MediaFileImpl::SetModuleFileCallback(FileCallback* callback)
+{
+
+ CriticalSectionScoped lock(_callbackCrit);
+
+ _ptrCallback = callback;
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::FileDurationMs(const WebRtc_Word8* fileName,
+ WebRtc_UWord32& durationMs,
+ const FileFormats format,
+ const WebRtc_UWord32 freqInHz)
+{
+
+ if(!ValidFileName(fileName))
+ {
+ return -1;
+ }
+ if(!ValidFrequency(freqInHz))
+ {
+ return -1;
+ }
+
+ ModuleFileUtility* utilityObj = new ModuleFileUtility(_id);
+ if(utilityObj == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "failed to allocate utility object!");
+ return -1;
+ }
+
+ const WebRtc_Word32 duration = utilityObj->FileDurationMs(fileName, format,
+ freqInHz);
+ delete utilityObj;
+ if(duration == -1)
+ {
+ durationMs = 0;
+ return -1;
+ }
+
+ durationMs = duration;
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::PlayoutPositionMs(WebRtc_UWord32& positionMs) const
+{
+ CriticalSectionScoped lock(_crit);
+ if(!_playingActive)
+ {
+ positionMs = 0;
+ return -1;
+ }
+ positionMs = _playoutPositionMs;
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::codec_info(CodecInst& codecInst) const
+{
+ CriticalSectionScoped lock(_crit);
+ if(!_playingActive && !_recordingActive)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Neither playout nor recording has been initialized!");
+ return -1;
+ }
+ if (codec_info_.pltype == 0 && codec_info_.plname[0] == '\0')
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "The CodecInst for %s is unknown!",
+ _playingActive ? "Playback" : "Recording");
+ return -1;
+ }
+ memcpy(&codecInst,&codec_info_,sizeof(CodecInst));
+ return 0;
+}
+
+WebRtc_Word32 MediaFileImpl::VideoCodecInst(VideoCodec& codecInst) const
+{
+ CriticalSectionScoped lock(_crit);
+ if(!_playingActive && !_recordingActive)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Neither playout nor recording has been initialized!");
+ return -1;
+ }
+ if( _ptrFileUtilityObj == NULL)
+ {
+ return -1;
+ }
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ VideoCodec videoCodec;
+ if( _ptrFileUtilityObj->VideoCodecInst( videoCodec) != 0)
+ {
+ return -1;
+ }
+ memcpy(&codecInst,&videoCodec,sizeof(VideoCodec));
+ return 0;
+#else
+ return -1;
+#endif
+}
+
+bool MediaFileImpl::ValidFileFormat(const FileFormats format,
+ const CodecInst* codecInst)
+{
+ if(codecInst == NULL)
+ {
+ if(format == kFileFormatPreencodedFile ||
+ format == kFileFormatPcm8kHzFile ||
+ format == kFileFormatPcm16kHzFile ||
+ format == kFileFormatPcm32kHzFile)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "Codec info required for file format specified!");
+ return false;
+ }
+ }
+ return true;
+}
+
+bool MediaFileImpl::ValidFileName(const WebRtc_Word8* fileName)
+{
+ if((fileName == NULL) ||(fileName[0] == '\0'))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1, "FileName not specified!");
+ return false;
+ }
+ return true;
+}
+
+
+bool MediaFileImpl::ValidFilePositions(const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs)
+{
+ if(startPointMs == 0 && stopPointMs == 0) // Default values
+ {
+ return true;
+ }
+ if(stopPointMs &&(startPointMs >= stopPointMs))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "startPointMs must be less than stopPointMs!");
+ return false;
+ }
+ if(stopPointMs &&((stopPointMs - startPointMs) < 20))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "minimum play duration for files is 20 ms!");
+ return false;
+ }
+ return true;
+}
+
+bool MediaFileImpl::ValidFrequency(const WebRtc_UWord32 frequency)
+{
+ if((frequency == 8000) || (frequency == 16000)|| (frequency == 32000))
+ {
+ return true;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "Frequency should be 8000, 16000 or 32000 (Hz)");
+ return false;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/media_file_impl.h b/trunk/src/modules/media_file/source/media_file_impl.h
new file mode 100644
index 0000000..06e20d0
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file_impl.h
@@ -0,0 +1,246 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+
+#include "common_types.h"
+#include "media_file.h"
+#include "media_file_defines.h"
+#include "media_file_utility.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+class MediaFileImpl : public MediaFile
+{
+
+public:
+ MediaFileImpl(const WebRtc_Word32 id);
+ ~MediaFileImpl();
+
+ WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ WebRtc_Word32 Process();
+ WebRtc_Word32 TimeUntilNextProcess();
+
+ // MediaFile functions
+ WebRtc_Word32 PlayoutAudioData(WebRtc_Word8* audioBuffer,
+ WebRtc_UWord32& dataLengthInBytes);
+ WebRtc_Word32 PlayoutAVIVideoData(WebRtc_Word8* videoBuffer,
+ WebRtc_UWord32& dataLengthInBytes);
+ WebRtc_Word32 PlayoutStereoData(WebRtc_Word8* audioBufferLeft,
+ WebRtc_Word8* audioBufferRight,
+ WebRtc_UWord32& dataLengthInBytes);
+ virtual WebRtc_Word32 StartPlayingAudioFile(
+ const WebRtc_Word8* fileName,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const bool loop = false,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0);
+ WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
+ const bool loop,
+ bool videoOnly,
+ const FileFormats format);
+ WebRtc_Word32 StartPlayingAudioStream(
+ InStream& stream,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0);
+ WebRtc_Word32 StopPlaying();
+ bool IsPlaying();
+ WebRtc_Word32 PlayoutPositionMs(WebRtc_UWord32& positionMs) const;
+ WebRtc_Word32 IncomingAudioData(const WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 bufferLength);
+ WebRtc_Word32 IncomingAVIVideoData(const WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 bufferLength);
+ WebRtc_Word32 StartRecordingAudioFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const WebRtc_UWord32 maxSizeBytes = 0);
+ WebRtc_Word32 StartRecordingVideoFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ bool videoOnly = false);
+ WebRtc_Word32 StartRecordingAudioStream(
+ OutStream& stream,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0);
+ WebRtc_Word32 StopRecording();
+ bool IsRecording();
+ WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs);
+ bool IsStereo();
+ WebRtc_Word32 SetModuleFileCallback(FileCallback* callback);
+ WebRtc_Word32 FileDurationMs(
+ const WebRtc_Word8* fileName,
+ WebRtc_UWord32& durationMs,
+ const FileFormats format,
+ const WebRtc_UWord32 freqInHz = 16000);
+ WebRtc_Word32 codec_info(CodecInst& codecInst) const;
+ WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst) const;
+
+private:
+ // Returns true if the combination of format and codecInst is valid.
+ static bool ValidFileFormat(const FileFormats format,
+ const CodecInst* codecInst);
+
+
+ // Returns true if the filename is valid
+ static bool ValidFileName(const WebRtc_Word8* fileName);
+
+ // Returns true if the combination of startPointMs and stopPointMs is valid.
+ static bool ValidFilePositions(const WebRtc_UWord32 startPointMs,
+ const WebRtc_UWord32 stopPointMs);
+
+ // Open the file specified by fileName for reading (relative path is
+ // allowed). FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. If loop is true the file will be played until
+ // StopPlaying() is called. When end of file is reached the file is read
+ // from the start. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. Only video will be read if videoOnly is
+ // true. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ WebRtc_Word32 StartPlayingFile(
+ const WebRtc_Word8* fileName,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const bool loop = false,
+ bool videoOnly = false,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0);
+
+ // Opens the file specified by fileName for reading (relative path is
+ // allowed) if format is kFileFormatAviFile otherwise use stream for
+ // reading. FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. If loop is true the file will be played until
+ // StopPlaying() is called. When end of file is reached the file is read
+ // from the start. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. Only video will be read if videoOnly is
+ // true. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ // TODO (hellner): there is no reason why fileName should be needed here.
+ WebRtc_Word32 StartPlayingStream(
+ InStream& stream,
+ const WebRtc_Word8* fileName,
+ bool loop,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0,
+ bool videoOnly = true);
+
+ // Writes one frame into dataBuffer. dataLengthInBytes is both an input and
+ // output parameter. As input parameter it indicates the size of
+ // audioBuffer. As output parameter it indicates the number of bytes
+ // written to audioBuffer. If video is true the data written is a video
+ // frame otherwise it is an audio frame.
+ WebRtc_Word32 PlayoutData(WebRtc_Word8* dataBuffer,
+ WebRtc_UWord32& dataLengthInBytes, bool video);
+
+ // Write one frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to file. The frame is an audio frame if video is true otherwise it is an
+ // audio frame.
+ WebRtc_Word32 IncomingAudioVideoData(const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 bufferLength,
+ const bool video);
+
+ // Open/creates file specified by fileName for writing (relative path is
+ // allowed) if format is kFileFormatAviFile otherwise use stream for
+ // writing. FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that should be created/opened.
+ // codecInst specifies the encoding of the audio data. videoCodecInst
+ // specifies the encoding of the video data. maxSizeBytes specifies the
+ // number of bytes allowed to be written to file if it is greater than zero.
+ // If format is kFileFormatAviFile and videoOnly is true the AVI file will
+ // only contain video frames.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ WebRtc_Word32 StartRecordingFile(
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const WebRtc_UWord32 maxSizeBytes = 0,
+ bool videoOnly = false);
+
+ // Open/creates file specified by fileName for writing (relative path is
+ // allowed). FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that should be created/opened.
+ // codecInst specifies the encoding of the audio data. videoCodecInst
+ // specifies the encoding of the video data. maxSizeBytes specifies the
+ // number of bytes allowed to be written to file if it is greater than zero.
+ // If format is kFileFormatAviFile and videoOnly is true the AVI file will
+ // only contain video frames.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ // TODO (hellner): there is no reason why fileName should be needed here.
+ WebRtc_Word32 StartRecordingStream(
+ OutStream& stream,
+ const WebRtc_Word8* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ const WebRtc_UWord32 notificationTimeMs = 0,
+ const bool videoOnly = false);
+
+ // Returns true if frequencyInHz is a supported frequency.
+ static bool ValidFrequency(const WebRtc_UWord32 frequencyInHz);
+
+ void HandlePlayCallbacks(WebRtc_Word32 bytesRead);
+
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper* _crit;
+ CriticalSectionWrapper* _callbackCrit;
+
+ ModuleFileUtility* _ptrFileUtilityObj;
+ CodecInst codec_info_;
+
+ InStream* _ptrInStream;
+ OutStream* _ptrOutStream;
+
+ FileFormats _fileFormat;
+ WebRtc_UWord32 _recordDurationMs;
+ WebRtc_UWord32 _playoutPositionMs;
+ WebRtc_UWord32 _notificationMs;
+
+ bool _playingActive;
+ bool _recordingActive;
+ bool _isStereo;
+ bool _openFile;
+
+ WebRtc_Word8 _fileName[512];
+
+ FileCallback* _ptrCallback;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
diff --git a/trunk/src/modules/media_file/source/media_file_unittest.cc b/trunk/src/modules/media_file/source/media_file_unittest.cc
new file mode 100644
index 0000000..86f37f3
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "media_file.h"
+#include "gtest/gtest.h"
+
+TEST(MediaFileTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/media_file/source/media_file_utility.cc b/trunk/src/modules/media_file/source/media_file_utility.cc
new file mode 100644
index 0000000..eeeb209
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file_utility.cc
@@ -0,0 +1,2563 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "file_wrapper.h"
+#include "media_file_utility.h"
+#include "module_common_types.h"
+#include "trace.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ #include "avi_file.h"
+#endif
+
+#if (defined(WIN32) || defined(WINCE))
+ #define STR_CASE_CMP _stricmp
+ #define STR_NCASE_CMP _strnicmp
+#else
+ #define STR_CASE_CMP strcasecmp
+ #define STR_NCASE_CMP strncasecmp
+#endif
+
+namespace {
+enum WaveFormats
+{
+ kWaveFormatPcm = 0x0001,
+ kWaveFormatALaw = 0x0006,
+ kWaveFormatMuLaw = 0x0007
+};
+
+// First 16 bytes the WAVE header. ckID should be "RIFF", wave_ckID should be
+// "WAVE" and ckSize is the chunk size (4 + n)
+struct WAVE_RIFF_header
+{
+ WebRtc_Word8 ckID[4];
+ WebRtc_Word32 ckSize;
+ WebRtc_Word8 wave_ckID[4];
+};
+
+// First 8 byte of the format chunk. fmt_ckID should be "fmt ". fmt_ckSize is
+// the chunk size (16, 18 or 40 byte)
+struct WAVE_CHUNK_header
+{
+ WebRtc_Word8 fmt_ckID[4];
+ WebRtc_Word32 fmt_ckSize;
+};
+} // unnamed namespace
+
+namespace webrtc {
+ModuleFileUtility::ModuleFileUtility(const WebRtc_Word32 id)
+ : _wavFormatObj(),
+ _dataSize(0),
+ _readSizeBytes(0),
+ _id(id),
+ _stopPointInMs(0),
+ _startPointInMs(0),
+ _playoutPositionMs(0),
+ _bytesWritten(0),
+ codec_info_(),
+ _codecId(kCodecNoCodec),
+ _bytesPerSample(0),
+ _readPos(0),
+ _reading(false),
+ _writing(false),
+ _tempData()
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ ,
+ _aviAudioInFile(0),
+ _aviVideoInFile(0),
+ _aviOutFile(0)
+#endif
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "ModuleFileUtility::ModuleFileUtility()");
+ memset(&codec_info_,0,sizeof(CodecInst));
+ codec_info_.pltype = -1;
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ memset(&_videoCodec,0,sizeof(_videoCodec));
+#endif
+}
+
+ModuleFileUtility::~ModuleFileUtility()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "ModuleFileUtility::~ModuleFileUtility()");
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ delete _aviAudioInFile;
+ delete _aviVideoInFile;
+#endif
+}
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+WebRtc_Word32 ModuleFileUtility::InitAviWriting(
+ const WebRtc_Word8* filename,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ const bool videoOnly /*= false*/)
+{
+ _writing = false;
+
+ delete _aviOutFile;
+ _aviOutFile = new AviFile( );
+
+ AVISTREAMHEADER videoStreamHeader;
+ videoStreamHeader.fccType = AviFile::MakeFourCc('v', 'i', 'd', 's');
+
+#ifdef VIDEOCODEC_I420
+ if (strncmp(videoCodecInst.plName, "I420", 7) == 0)
+ {
+ videoStreamHeader.fccHandler = AviFile::MakeFourCc('I','4','2','0');
+ }
+#endif
+#ifdef VIDEOCODEC_VP8
+ if (strncmp(videoCodecInst.plName, "VP8", 7) == 0)
+ {
+ videoStreamHeader.fccHandler = AviFile::MakeFourCc('V','P','8','0');
+ }
+#endif
+ if (videoStreamHeader.fccHandler == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "InitAviWriting() Codec not supported");
+
+ return -1;
+ }
+ videoStreamHeader.dwScale = 1;
+ videoStreamHeader.dwRate = videoCodecInst.maxFramerate;
+ videoStreamHeader.dwSuggestedBufferSize = videoCodecInst.height *
+ (videoCodecInst.width >> 1) * 3;
+ videoStreamHeader.dwQuality = (WebRtc_UWord32)-1;
+ videoStreamHeader.dwSampleSize = 0;
+ videoStreamHeader.rcFrame.top = 0;
+ videoStreamHeader.rcFrame.bottom = videoCodecInst.height;
+ videoStreamHeader.rcFrame.left = 0;
+ videoStreamHeader.rcFrame.right = videoCodecInst.width;
+
+ BITMAPINFOHEADER bitMapInfoHeader;
+ bitMapInfoHeader.biSize = sizeof(BITMAPINFOHEADER);
+ bitMapInfoHeader.biHeight = videoCodecInst.height;
+ bitMapInfoHeader.biWidth = videoCodecInst.width;
+ bitMapInfoHeader.biPlanes = 1;
+ bitMapInfoHeader.biBitCount = 12;
+ bitMapInfoHeader.biClrImportant = 0;
+ bitMapInfoHeader.biClrUsed = 0;
+ bitMapInfoHeader.biCompression = videoStreamHeader.fccHandler;
+ bitMapInfoHeader.biSizeImage = bitMapInfoHeader.biWidth *
+ bitMapInfoHeader.biHeight * bitMapInfoHeader.biBitCount / 8;
+
+ if (_aviOutFile->CreateVideoStream(
+ videoStreamHeader,
+ bitMapInfoHeader,
+ NULL,
+ 0) != 0)
+ {
+ return -1;
+ }
+
+ if(!videoOnly)
+ {
+ AVISTREAMHEADER audioStreamHeader;
+ audioStreamHeader.fccType = AviFile::MakeFourCc('a', 'u', 'd', 's');
+ // fccHandler is the FOURCC of the codec for decoding the stream.
+ // It's an optional parameter that is not used by audio streams.
+ audioStreamHeader.fccHandler = 0;
+ audioStreamHeader.dwScale = 1;
+
+ WAVEFORMATEX waveFormatHeader;
+ waveFormatHeader.cbSize = 0;
+ waveFormatHeader.nChannels = 1;
+
+ if (strncmp(audioCodecInst.plname, "PCMU", 4) == 0)
+ {
+ audioStreamHeader.dwSampleSize = 1;
+ audioStreamHeader.dwRate = 8000;
+ audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
+ audioStreamHeader.dwSuggestedBufferSize = 80;
+
+ waveFormatHeader.nAvgBytesPerSec = 8000;
+ waveFormatHeader.nSamplesPerSec = 8000;
+ waveFormatHeader.wBitsPerSample = 8;
+ waveFormatHeader.nBlockAlign = 1;
+ waveFormatHeader.wFormatTag = kWaveFormatMuLaw;
+
+ } else if (strncmp(audioCodecInst.plname, "PCMA", 4) == 0)
+ {
+ audioStreamHeader.dwSampleSize = 1;
+ audioStreamHeader.dwRate = 8000;
+ audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
+ audioStreamHeader.dwSuggestedBufferSize = 80;
+
+ waveFormatHeader.nAvgBytesPerSec = 8000;
+ waveFormatHeader.nSamplesPerSec = 8000;
+ waveFormatHeader.wBitsPerSample = 8;
+ waveFormatHeader.nBlockAlign = 1;
+ waveFormatHeader.wFormatTag = kWaveFormatALaw;
+
+ } else if (strncmp(audioCodecInst.plname, "L16", 3) == 0)
+ {
+ audioStreamHeader.dwSampleSize = 2;
+ audioStreamHeader.dwRate = audioCodecInst.plfreq;
+ audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
+ audioStreamHeader.dwSuggestedBufferSize =
+ (audioCodecInst.plfreq/100) * 2;
+
+ waveFormatHeader.nAvgBytesPerSec = audioCodecInst.plfreq * 2;
+ waveFormatHeader.nSamplesPerSec = audioCodecInst.plfreq;
+ waveFormatHeader.wBitsPerSample = 16;
+ waveFormatHeader.nBlockAlign = 2;
+ waveFormatHeader.wFormatTag = kWaveFormatPcm;
+ } else
+ {
+ return -1;
+ }
+
+ if(_aviOutFile->CreateAudioStream(
+ audioStreamHeader,
+ waveFormatHeader) != 0)
+ {
+ return -1;
+ }
+
+
+ if( InitWavCodec(waveFormatHeader.nSamplesPerSec,
+ waveFormatHeader.nChannels,
+ waveFormatHeader.wBitsPerSample,
+ waveFormatHeader.wFormatTag) != 0)
+ {
+ return -1;
+ }
+ }
+ _aviOutFile->Create(filename);
+ _writing = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::WriteAviAudioData(
+ const WebRtc_Word8* buffer,
+ WebRtc_UWord32 bufferLengthInBytes)
+{
+ if( _aviOutFile != 0)
+ {
+ return _aviOutFile->WriteAudio(
+ reinterpret_cast<const WebRtc_UWord8*>(buffer),
+ bufferLengthInBytes);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
+ return -1;
+ }
+}
+
+WebRtc_Word32 ModuleFileUtility::WriteAviVideoData(
+ const WebRtc_Word8* buffer,
+ WebRtc_UWord32 bufferLengthInBytes)
+{
+ if( _aviOutFile != 0)
+ {
+ return _aviOutFile->WriteVideo(
+ reinterpret_cast<const WebRtc_UWord8*>(buffer),
+ bufferLengthInBytes);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
+ return -1;
+ }
+}
+
+
+WebRtc_Word32 ModuleFileUtility::CloseAviFile( )
+{
+ if( _reading && _aviAudioInFile)
+ {
+ delete _aviAudioInFile;
+ _aviAudioInFile = 0;
+ }
+
+ if( _reading && _aviVideoInFile)
+ {
+ delete _aviVideoInFile;
+ _aviVideoInFile = 0;
+ }
+
+ if( _writing && _aviOutFile)
+ {
+ delete _aviOutFile;
+ _aviOutFile = 0;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32 ModuleFileUtility::InitAviReading(const WebRtc_Word8* filename,
+ bool videoOnly, bool loop)
+{
+ _reading = false;
+ delete _aviVideoInFile;
+ _aviVideoInFile = new AviFile( );
+
+ if ((_aviVideoInFile != 0) && _aviVideoInFile->Open(AviFile::AVI_VIDEO,
+ filename, loop) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Unable to open AVI file (video)");
+ return -1;
+ }
+
+
+ AVISTREAMHEADER videoInStreamHeader;
+ BITMAPINFOHEADER bitmapInfo;
+ char codecConfigParameters[AviFile::CODEC_CONFIG_LENGTH] = {};
+ WebRtc_Word32 configLength = 0;
+ if( _aviVideoInFile->GetVideoStreamInfo(videoInStreamHeader, bitmapInfo,
+ codecConfigParameters,
+ configLength) != 0)
+ {
+ return -1;
+ }
+ _videoCodec.width = static_cast<WebRtc_UWord16>(
+ videoInStreamHeader.rcFrame.right);
+ _videoCodec.height = static_cast<WebRtc_UWord16>(
+ videoInStreamHeader.rcFrame.bottom);
+ _videoCodec.maxFramerate = static_cast<WebRtc_UWord8>(
+ videoInStreamHeader.dwRate);
+
+ const size_t plnameLen = sizeof(_videoCodec.plName) / sizeof(char);
+ if (bitmapInfo.biCompression == AviFile::MakeFourCc('I','4','2','0'))
+ {
+ strncpy(_videoCodec.plName, "I420", plnameLen);
+ _videoCodec.codecType = kVideoCodecI420;
+ }
+ else if (bitmapInfo.biCompression ==
+ AviFile::MakeFourCc('V', 'P', '8', '0'))
+ {
+ strncpy(_videoCodec.plName, "VP8", plnameLen);
+ _videoCodec.codecType = kVideoCodecVP8;
+ }
+ else
+ {
+ return -1;
+ }
+
+ if(!videoOnly)
+ {
+ delete _aviAudioInFile;
+ _aviAudioInFile = new AviFile();
+
+ if ( (_aviAudioInFile != 0) &&
+ _aviAudioInFile->Open(AviFile::AVI_AUDIO, filename, loop) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Unable to open AVI file (audio)");
+ return -1;
+ }
+
+ WAVEFORMATEX waveHeader;
+ if(_aviAudioInFile->GetAudioStreamInfo(waveHeader) != 0)
+ {
+ return -1;
+ }
+ if(InitWavCodec(waveHeader.nSamplesPerSec, waveHeader.nChannels,
+ waveHeader.wBitsPerSample, waveHeader.wFormatTag) != 0)
+ {
+ return -1;
+ }
+ }
+ _reading = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadAviAudioData(
+ WebRtc_Word8* outBuffer,
+ const WebRtc_UWord32 bufferLengthInBytes)
+{
+ if(_aviAudioInFile == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
+ return -1;
+ }
+
+ WebRtc_Word32 length = bufferLengthInBytes;
+ if(_aviAudioInFile->ReadAudio(
+ reinterpret_cast<WebRtc_UWord8*>(outBuffer),
+ length) != 0)
+ {
+ return -1;
+ }
+ else
+ {
+ return length;
+ }
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadAviVideoData(
+ WebRtc_Word8* outBuffer,
+ const WebRtc_UWord32 bufferLengthInBytes)
+{
+ if(_aviVideoInFile == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
+ return -1;
+ }
+
+ WebRtc_Word32 length = bufferLengthInBytes;
+ if( _aviVideoInFile->ReadVideo(
+ reinterpret_cast<WebRtc_UWord8*>(outBuffer),
+ length) != 0)
+ {
+ return -1;
+ } else {
+ return length;
+ }
+}
+
+WebRtc_Word32 ModuleFileUtility::VideoCodecInst(VideoCodec& codecInst)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::CodecInst(codecInst= 0x%x)", &codecInst);
+
+ if(!_reading)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "CodecInst: not currently reading audio file!");
+ return -1;
+ }
+ memcpy(&codecInst,&_videoCodec,sizeof(VideoCodec));
+ return 0;
+}
+#endif
+
+WebRtc_Word32 ModuleFileUtility::ReadWavHeader(InStream& wav)
+{
+ WAVE_RIFF_header RIFFheaderObj;
+ WAVE_CHUNK_header CHUNKheaderObj;
+ // TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
+ WebRtc_Word8 tmpStr[6] = "FOUR";
+ WebRtc_UWord8 tmpStr2[4];
+ WebRtc_Word32 i, len;
+ bool dataFound = false;
+ bool fmtFound = false;
+ WebRtc_Word8 dummyRead;
+
+
+ _dataSize = 0;
+ len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
+ if(len != sizeof(WAVE_RIFF_header))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a wave file (too short)");
+ return -1;
+ }
+
+ for (i = 0; i < 4; i++)
+ {
+ tmpStr[i] = RIFFheaderObj.ckID[i];
+ }
+ if(strcmp(tmpStr, "RIFF") != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a wave file (does not have RIFF)");
+ return -1;
+ }
+ for (i = 0; i < 4; i++)
+ {
+ tmpStr[i] = RIFFheaderObj.wave_ckID[i];
+ }
+ if(strcmp(tmpStr, "WAVE") != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Not a wave file (does not have WAVE)");
+ return -1;
+ }
+
+ len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
+
+ // WAVE files are stored in little endian byte order. Make sure that the
+ // data can be read on big endian as well.
+ // TODO (hellner): little endian to system byte order should be done in
+ // in a subroutine.
+ memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
+ CHUNKheaderObj.fmt_ckSize =
+ (WebRtc_Word32) ((WebRtc_UWord32) tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8) +
+ (((WebRtc_UWord32)tmpStr2[2])<<16) +
+ (((WebRtc_UWord32)tmpStr2[3])<<24));
+
+ memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
+
+ while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
+ {
+ if(strcmp(tmpStr, "fmt ") == 0)
+ {
+ len = wav.Read(&_wavFormatObj, sizeof(WAVE_FMTINFO_header));
+
+ memcpy(tmpStr2, &_wavFormatObj.formatTag, 2);
+ _wavFormatObj.formatTag =
+ (WaveFormats) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8));
+ memcpy(tmpStr2, &_wavFormatObj.nChannels, 2);
+ _wavFormatObj.nChannels =
+ (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8));
+ memcpy(tmpStr2, &_wavFormatObj.nSamplesPerSec, 4);
+ _wavFormatObj.nSamplesPerSec =
+ (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8) +
+ (((WebRtc_UWord32)tmpStr2[2])<<16) +
+ (((WebRtc_UWord32)tmpStr2[3])<<24));
+ memcpy(tmpStr2, &_wavFormatObj.nAvgBytesPerSec, 4);
+ _wavFormatObj.nAvgBytesPerSec =
+ (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8) +
+ (((WebRtc_UWord32)tmpStr2[2])<<16) +
+ (((WebRtc_UWord32)tmpStr2[3])<<24));
+ memcpy(tmpStr2, &_wavFormatObj.nBlockAlign, 2);
+ _wavFormatObj.nBlockAlign =
+ (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8));
+ memcpy(tmpStr2, &_wavFormatObj.nBitsPerSample, 2);
+ _wavFormatObj.nBitsPerSample =
+ (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8));
+
+ for (i = 0;
+ i < (CHUNKheaderObj.fmt_ckSize -
+ (WebRtc_Word32)sizeof(WAVE_FMTINFO_header));
+ i++)
+ {
+ len = wav.Read(&dummyRead, 1);
+ if(len != 1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "File corrupted, reached EOF (reading fmt)");
+ return -1;
+ }
+ }
+ fmtFound = true;
+ }
+ else if(strcmp(tmpStr, "data") == 0)
+ {
+ _dataSize = CHUNKheaderObj.fmt_ckSize;
+ dataFound = true;
+ break;
+ }
+ else
+ {
+ for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
+ {
+ len = wav.Read(&dummyRead, 1);
+ if(len != 1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "File corrupted, reached EOF (reading other)");
+ return -1;
+ }
+ }
+ }
+
+ len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
+
+ memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
+ CHUNKheaderObj.fmt_ckSize =
+ (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
+ (((WebRtc_UWord32)tmpStr2[1])<<8) +
+ (((WebRtc_UWord32)tmpStr2[2])<<16) +
+ (((WebRtc_UWord32)tmpStr2[3])<<24));
+
+ memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
+ }
+
+ // Either a proper format chunk has been read or a data chunk was come
+ // across.
+ if( (_wavFormatObj.formatTag != kWaveFormatPcm) &&
+ (_wavFormatObj.formatTag != kWaveFormatALaw) &&
+ (_wavFormatObj.formatTag != kWaveFormatMuLaw))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Coding formatTag value=%d not supported!",
+ _wavFormatObj.formatTag);
+ return -1;
+ }
+ if((_wavFormatObj.nChannels < 1) ||
+ (_wavFormatObj.nChannels > 2))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "nChannels value=%d not supported!",
+ _wavFormatObj.nChannels);
+ return -1;
+ }
+
+ if((_wavFormatObj.nBitsPerSample != 8) &&
+ (_wavFormatObj.nBitsPerSample != 16))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "nBitsPerSample value=%d not supported!",
+ _wavFormatObj.nBitsPerSample);
+ return -1;
+ }
+
+ // Calculate the number of bytes that 10 ms of audio data correspond to.
+ if(_wavFormatObj.formatTag == kWaveFormatPcm)
+ {
+ // TODO (hellner): integer division for 22050 and 11025 would yield
+ // the same result as the else statement. Remove those
+ // special cases?
+ if(_wavFormatObj.nSamplesPerSec == 44100)
+ {
+ _readSizeBytes = 440 * _wavFormatObj.nChannels *
+ (_wavFormatObj.nBitsPerSample / 8);
+ } else if(_wavFormatObj.nSamplesPerSec == 22050) {
+ _readSizeBytes = 220 * _wavFormatObj.nChannels *
+ (_wavFormatObj.nBitsPerSample / 8);
+ } else if(_wavFormatObj.nSamplesPerSec == 11025) {
+ _readSizeBytes = 110 * _wavFormatObj.nChannels *
+ (_wavFormatObj.nBitsPerSample / 8);
+ } else {
+ _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
+ _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
+ }
+
+ } else {
+ _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
+ _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitWavCodec(WebRtc_UWord32 samplesPerSec,
+ WebRtc_UWord32 channels,
+ WebRtc_UWord32 bitsPerSample,
+ WebRtc_UWord32 formatTag)
+{
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = samplesPerSec;
+ codec_info_.channels = channels;
+ codec_info_.rate = bitsPerSample * samplesPerSec;
+
+ // Calculate the packet size for 10ms frames
+ switch(formatTag)
+ {
+ case kWaveFormatALaw:
+ strcpy(codec_info_.plname, "PCMA");
+ _codecId = kCodecPcma;
+ codec_info_.pltype = 8;
+ codec_info_.pacsize = codec_info_.plfreq / 100;
+ break;
+ case kWaveFormatMuLaw:
+ strcpy(codec_info_.plname, "PCMU");
+ _codecId = kCodecPcmu;
+ codec_info_.pltype = 0;
+ codec_info_.pacsize = codec_info_.plfreq / 100;
+ break;
+ case kWaveFormatPcm:
+ codec_info_.pacsize = (bitsPerSample * (codec_info_.plfreq / 100)) / 8;
+ if(samplesPerSec == 8000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_8Khz;
+ }
+ else if(samplesPerSec == 16000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_16kHz;
+ }
+ else if(samplesPerSec == 32000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_32Khz;
+ }
+ // Set the packet size for "odd" sampling frequencies so that it
+ // properly corresponds to _readSizeBytes.
+ else if(samplesPerSec == 11025)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_16kHz;
+ codec_info_.pacsize = 110;
+ codec_info_.plfreq = 11000;
+ }
+ else if(samplesPerSec == 22050)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_16kHz;
+ codec_info_.pacsize = 220;
+ codec_info_.plfreq = 22000;
+ }
+ else if(samplesPerSec == 44100)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_16kHz;
+ codec_info_.pacsize = 440;
+ codec_info_.plfreq = 44000;
+ }
+ else if(samplesPerSec == 48000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ _codecId = kCodecL16_16kHz;
+ codec_info_.pacsize = 480;
+ codec_info_.plfreq = 48000;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Unsupported PCM frequency!");
+ return -1;
+ }
+ break;
+ default:
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "unknown WAV format TAG!");
+ return -1;
+ break;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitWavReading(InStream& wav,
+ const WebRtc_UWord32 start,
+ const WebRtc_UWord32 stop)
+{
+
+ _reading = false;
+
+ if(ReadWavHeader(wav) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "failed to read WAV header!");
+ return -1;
+ }
+
+ _playoutPositionMs = 0;
+ _readPos = 0;
+
+ if(start > 0)
+ {
+ WebRtc_UWord8 dummy[WAV_MAX_BUFFER_SIZE];
+ WebRtc_Word32 readLength;
+ if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
+ {
+ while (_playoutPositionMs < start)
+ {
+ readLength = wav.Read(dummy, _readSizeBytes);
+ if(readLength == _readSizeBytes)
+ {
+ _readPos += readLength;
+ _playoutPositionMs += 10;
+ }
+ else // Must have reached EOF before start position!
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "InitWavReading(), EOF before start position");
+ return -1;
+ }
+ }
+ }
+ else
+ {
+ return -1;
+ }
+ }
+ if( InitWavCodec(_wavFormatObj.nSamplesPerSec, _wavFormatObj.nChannels,
+ _wavFormatObj.nBitsPerSample,
+ _wavFormatObj.formatTag) != 0)
+ {
+ return -1;
+ }
+ _bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
+
+
+ _startPointInMs = start;
+ _stopPointInMs = stop;
+ _reading = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadWavDataAsMono(
+ InStream& wav,
+ WebRtc_Word8* outData,
+ const WebRtc_UWord32 bufferSize)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadWavDataAsMono(wav= 0x%x, outData= 0x%d,\
+ bufSize= %ld)",
+ &wav,
+ outData,
+ bufferSize);
+
+ // The number of bytes that should be read from file.
+ const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
+ // The number of bytes that will be written to outData.
+ const WebRtc_UWord32 bytesRequested = (codec_info_.channels == 2) ?
+ totalBytesNeeded >> 1 : totalBytesNeeded;
+ if(bufferSize < bytesRequested)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: output buffer is too short!");
+ return -1;
+ }
+ if(outData == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: output buffer NULL!");
+ return -1;
+ }
+
+ if(!_reading)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: no longer reading file.");
+ return -1;
+ }
+
+ WebRtc_Word32 bytesRead = ReadWavData(
+ wav,
+ (codec_info_.channels == 2) ? _tempData : (WebRtc_UWord8*)outData,
+ totalBytesNeeded);
+ if(bytesRead == 0)
+ {
+ return 0;
+ }
+ if(bytesRead < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: failed to read data from WAV file.");
+ return -1;
+ }
+ // Output data is should be mono.
+ if(codec_info_.channels == 2)
+ {
+ for (WebRtc_UWord32 i = 0; i < bytesRequested / _bytesPerSample; i++)
+ {
+ // Sample value is the average of left and right buffer rounded to
+ // closest integer value. Note samples can be either 1 or 2 byte.
+ if(_bytesPerSample == 1)
+ {
+ _tempData[i] = ((_tempData[2 * i] + _tempData[(2 * i) + 1] +
+ 1) >> 1);
+ }
+ else
+ {
+ WebRtc_Word16* sampleData = (WebRtc_Word16*) _tempData;
+ sampleData[i] = ((sampleData[2 * i] + sampleData[(2 * i) + 1] +
+ 1) >> 1);
+ }
+ }
+ memcpy(outData, _tempData, bytesRequested);
+ }
+ return bytesRequested;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadWavDataAsStereo(
+ InStream& wav,
+ WebRtc_Word8* outDataLeft,
+ WebRtc_Word8* outDataRight,
+ const WebRtc_UWord32 bufferSize)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadWavDataAsStereo(wav= 0x%x, outLeft= 0x%x,\
+ outRight= 0x%x, bufSize= %ld)",
+ &wav,
+ outDataLeft,
+ outDataRight,
+ bufferSize);
+
+ if((outDataLeft == NULL) ||
+ (outDataRight == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: an input buffer is NULL!");
+ return -1;
+ }
+ if(codec_info_.channels != 2)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "ReadWavDataAsStereo: WAV file does not contain stereo data!");
+ return -1;
+ }
+ if(! _reading)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsStereo: no longer reading file.");
+ return -1;
+ }
+
+ // The number of bytes that should be read from file.
+ const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
+ // The number of bytes that will be written to the left and the right
+ // buffers.
+ const WebRtc_UWord32 bytesRequested = totalBytesNeeded >> 1;
+ if(bufferSize < bytesRequested)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavData: Output buffers are too short!");
+ assert(false);
+ return -1;
+ }
+
+ WebRtc_Word32 bytesRead = ReadWavData(wav, _tempData, totalBytesNeeded);
+ if(bytesRead <= 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsStereo: failed to read data from WAV file.");
+ return -1;
+ }
+
+ // Turn interleaved audio to left and right buffer. Note samples can be
+ // either 1 or 2 bytes
+ if(_bytesPerSample == 1)
+ {
+ for (WebRtc_UWord32 i = 0; i < bytesRequested; i++)
+ {
+ outDataLeft[i] = _tempData[2 * i];
+ outDataRight[i] = _tempData[(2 * i) + 1];
+ }
+ }
+ else if(_bytesPerSample == 2)
+ {
+ WebRtc_Word16* sampleData = reinterpret_cast<WebRtc_Word16*>(_tempData);
+ WebRtc_Word16* outLeft = reinterpret_cast<WebRtc_Word16*>(outDataLeft);
+ WebRtc_Word16* outRight = reinterpret_cast<WebRtc_Word16*>(
+ outDataRight);
+
+ // Bytes requested to samples requested.
+ WebRtc_UWord32 sampleCount = bytesRequested >> 1;
+ for (WebRtc_UWord32 i = 0; i < sampleCount; i++)
+ {
+ outLeft[i] = sampleData[2 * i];
+ outRight[i] = sampleData[(2 * i) + 1];
+ }
+ } else {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavStereoData: unsupported sample size %d!",
+ _bytesPerSample);
+ assert(false);
+ return -1;
+ }
+ return bytesRequested;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadWavData(
+ InStream& wav,
+ WebRtc_UWord8* buffer,
+ const WebRtc_UWord32 dataLengthInBytes)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
+ &wav,
+ buffer,
+ dataLengthInBytes);
+
+
+ if(buffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadWavDataAsMono: output buffer NULL!");
+ return -1;
+ }
+
+ // Make sure that a read won't return too few samples.
+ // TODO (hellner): why not read the remaining bytes needed from the start
+ // of the file?
+ if((_dataSize - _readPos) < (WebRtc_Word32)dataLengthInBytes)
+ {
+ // Rewind() being -1 may be due to the file not supposed to be looped.
+ if(wav.Rewind() == -1)
+ {
+ _reading = false;
+ return 0;
+ }
+ if(InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1)
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+
+ WebRtc_Word32 bytesRead = wav.Read(buffer, dataLengthInBytes);
+ if(bytesRead < 0)
+ {
+ _reading = false;
+ return -1;
+ }
+
+ // This should never happen due to earlier sanity checks.
+ // TODO (hellner): change to an assert and fail here since this should
+ // never happen...
+ if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
+ {
+ if((wav.Rewind() == -1) ||
+ (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
+ {
+ _reading = false;
+ return -1;
+ }
+ else
+ {
+ bytesRead = wav.Read(buffer, dataLengthInBytes);
+ if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ }
+
+ _readPos += bytesRead;
+
+ // TODO (hellner): Why is dataLengthInBytes let dictate the number of bytes
+ // to read when exactly 10ms should be read?!
+ _playoutPositionMs += 10;
+ if((_stopPointInMs > 0) &&
+ (_playoutPositionMs >= _stopPointInMs))
+ {
+ if((wav.Rewind() == -1) ||
+ (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
+ {
+ _reading = false;
+ }
+ }
+ return bytesRead;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitWavWriting(OutStream& wav,
+ const CodecInst& codecInst)
+{
+
+ if(set_codec_info(codecInst) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "codecInst identifies unsupported codec!");
+ return -1;
+ }
+ _writing = false;
+ WebRtc_UWord32 channels = (codecInst.channels == 0) ?
+ 1 : codecInst.channels;
+
+ if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
+ {
+ _bytesPerSample = 1;
+ if(WriteWavHeader(wav, 8000, _bytesPerSample, channels,
+ kWaveFormatMuLaw, 0) == -1)
+ {
+ return -1;
+ }
+ }else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
+ {
+ _bytesPerSample = 1;
+ if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWaveFormatALaw,
+ 0) == -1)
+ {
+ return -1;
+ }
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
+ {
+ _bytesPerSample = 2;
+ if(WriteWavHeader(wav, codecInst.plfreq, _bytesPerSample, channels,
+ kWaveFormatPcm, 0) == -1)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "codecInst identifies unsupported codec for WAV file!");
+ return -1;
+ }
+ _writing = true;
+ _bytesWritten = 0;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::WriteWavData(OutStream& out,
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 dataLength)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %d)",
+ &out,
+ buffer,
+ dataLength);
+
+ if(buffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "WriteWavData: input buffer NULL!");
+ return -1;
+ }
+
+ if(!out.Write(buffer, dataLength))
+ {
+ return -1;
+ }
+ _bytesWritten += dataLength;
+ return dataLength;
+}
+
+
+WebRtc_Word32 ModuleFileUtility::WriteWavHeader(
+ OutStream& wav,
+ const WebRtc_UWord32 freq,
+ const WebRtc_UWord32 bytesPerSample,
+ const WebRtc_UWord32 channels,
+ const WebRtc_UWord32 format,
+ const WebRtc_UWord32 lengthInBytes)
+{
+
+ // Frame size in bytes for 10 ms of audio.
+ // TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
+ // be taken into consideration here!
+ WebRtc_Word32 frameSize = (freq / 100) * bytesPerSample * channels;
+
+ // Calculate the number of full frames that the wave file contain.
+ const WebRtc_Word32 dataLengthInBytes = frameSize *
+ (lengthInBytes / frameSize);
+
+ WebRtc_Word8 tmpStr[4];
+ WebRtc_Word8 tmpChar;
+ WebRtc_UWord32 tmpLong;
+
+ memcpy(tmpStr, "RIFF", 4);
+ wav.Write(tmpStr, 4);
+
+ tmpLong = dataLengthInBytes + 36;
+ tmpChar = (WebRtc_Word8)(tmpLong);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 8);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 16);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 24);
+ wav.Write(&tmpChar, 1);
+
+ memcpy(tmpStr, "WAVE", 4);
+ wav.Write(tmpStr, 4);
+
+ memcpy(tmpStr, "fmt ", 4);
+ wav.Write(tmpStr, 4);
+
+ tmpChar = 16;
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+
+ tmpChar = (WebRtc_Word8)(format);
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+
+ tmpChar = (WebRtc_Word8)(channels);
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+
+ tmpLong = freq;
+ tmpChar = (WebRtc_Word8)(tmpLong);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 8);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 16);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 24);
+ wav.Write(&tmpChar, 1);
+
+ // nAverageBytesPerSec = Sample rate * Bytes per sample * Channels
+ tmpLong = bytesPerSample * freq * channels;
+ tmpChar = (WebRtc_Word8)(tmpLong);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 8);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 16);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 24);
+ wav.Write(&tmpChar, 1);
+
+ // nBlockAlign = Bytes per sample * Channels
+ tmpChar = (WebRtc_Word8)(bytesPerSample * channels);
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+
+ tmpChar = (WebRtc_Word8)(bytesPerSample*8);
+ wav.Write(&tmpChar, 1);
+ tmpChar = 0;
+ wav.Write(&tmpChar, 1);
+
+ memcpy(tmpStr, "data", 4);
+ wav.Write(tmpStr, 4);
+
+ tmpLong = dataLengthInBytes;
+ tmpChar = (WebRtc_Word8)(tmpLong);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 8);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 16);
+ wav.Write(&tmpChar, 1);
+ tmpChar = (WebRtc_Word8)(tmpLong >> 24);
+ wav.Write(&tmpChar, 1);
+
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::UpdateWavHeader(OutStream& wav)
+{
+ WebRtc_Word32 res = -1;
+ if(wav.Rewind() == -1)
+ {
+ return -1;
+ }
+ WebRtc_UWord32 channels = (codec_info_.channels == 0) ?
+ 1 : codec_info_.channels;
+
+ if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
+ {
+ res = WriteWavHeader(wav, codec_info_.plfreq, 2, channels,
+ kWaveFormatPcm, _bytesWritten);
+ } else if(STR_CASE_CMP(codec_info_.plname, "PCMU") == 0) {
+ res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatMuLaw,
+ _bytesWritten);
+ } else if(STR_CASE_CMP(codec_info_.plname, "PCMA") == 0) {
+ res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatALaw,
+ _bytesWritten);
+ } else {
+ // Allow calling this API even if not writing to a WAVE file.
+ // TODO (hellner): why?!
+ return 0;
+ }
+ return res;
+}
+
+
+WebRtc_Word32 ModuleFileUtility::InitPreEncodedReading(InStream& in,
+ const CodecInst& cinst)
+{
+
+ WebRtc_UWord8 preEncodedID;
+ in.Read(&preEncodedID, 1);
+
+ MediaFileUtility_CodecType codecType =
+ (MediaFileUtility_CodecType)preEncodedID;
+
+ if(set_codec_info(cinst) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Pre-encoded file send codec mismatch!");
+ return -1;
+ }
+ if(codecType != _codecId)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "Pre-encoded file format codec mismatch!");
+ return -1;
+ }
+ memcpy(&codec_info_,&cinst,sizeof(CodecInst));
+ _reading = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadPreEncodedData(
+ InStream& in,
+ WebRtc_Word8* outData,
+ const WebRtc_UWord32 bufferSize)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x,\
+ bufferSize= %d)",
+ &in,
+ outData,
+ bufferSize);
+
+ if(outData == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
+ }
+
+ WebRtc_UWord32 frameLen;
+ WebRtc_UWord8 buf[64];
+ // Each frame has a two byte header containing the frame length.
+ WebRtc_Word32 res = in.Read(buf, 2);
+ if(res != 2)
+ {
+ if(!in.Rewind())
+ {
+ // The first byte is the codec identifier.
+ in.Read(buf, 1);
+ res = in.Read(buf, 2);
+ }
+ else
+ {
+ return -1;
+ }
+ }
+ frameLen = buf[0] + buf[1] * 256;
+ if(bufferSize < frameLen)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "buffer not large enough to read %d bytes of pre-encoded data!",
+ frameLen);
+ return -1;
+ }
+ return in.Read(outData, frameLen);
+}
+
+WebRtc_Word32 ModuleFileUtility::InitPreEncodedWriting(
+ OutStream& out,
+ const CodecInst& codecInst)
+{
+
+ if(set_codec_info(codecInst) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "CodecInst not recognized!");
+ return -1;
+ }
+ _writing = true;
+ _bytesWritten = 1;
+ out.Write(&_codecId, 1);
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::WritePreEncodedData(
+ OutStream& out,
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 dataLength)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x,\
+ dataLen= %d)",
+ &out,
+ buffer,
+ dataLength);
+
+ if(buffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+ }
+
+ WebRtc_Word32 bytesWritten = 0;
+ // The first two bytes is the size of the frame.
+ WebRtc_Word16 lengthBuf;
+ lengthBuf = (WebRtc_Word16)dataLength;
+ if(!out.Write(&lengthBuf, 2))
+ {
+ return -1;
+ }
+ bytesWritten = 2;
+
+ if(!out.Write(buffer, dataLength))
+ {
+ return -1;
+ }
+ bytesWritten += dataLength;
+ return bytesWritten;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitCompressedReading(
+ InStream& in,
+ const WebRtc_UWord32 start,
+ const WebRtc_UWord32 stop)
+{
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
+ stop= %d)",
+ &in,
+ start,
+ stop);
+
+#if defined(WEBRTC_CODEC_GSMAMR) || defined(WEBRTC_CODEC_GSMAMRWB) || \
+ defined(WEBRTC_CODEC_ILBC)
+ WebRtc_Word16 read_len = 0;
+#endif
+ _codecId = kCodecNoCodec;
+ _playoutPositionMs = 0;
+ _reading = false;
+
+ _startPointInMs = start;
+ _stopPointInMs = stop;
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ WebRtc_Word32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ WebRtc_Word32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
+#endif
+
+ // Read the codec name
+ WebRtc_Word32 cnt = 0;
+ WebRtc_Word8 buf[64];
+ do
+ {
+ in.Read(&buf[cnt++], 1);
+ } while ((buf[cnt-1] != '\n') && (64 > cnt));
+
+ if(cnt==64)
+ {
+ return -1;
+ } else {
+ buf[cnt]=0;
+ }
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ if(!strcmp("#!AMR\n", buf))
+ {
+ strcpy(codec_info_.plname, "amr");
+ codec_info_.pacsize = 160;
+ _codecId = kCodecAmr;
+ codec_info_.pltype = 112;
+ codec_info_.rate = 12200;
+ codec_info_.plfreq = 8000;
+ codec_info_.channels = 1;
+
+ WebRtc_Word16 mode = 0;
+ if(_startPointInMs > 0)
+ {
+ while (_playoutPositionMs <= _startPointInMs)
+ {
+ // First read byte contain the AMR mode.
+ read_len = in.Read(buf, 1);
+ if(read_len != 1)
+ {
+ return -1;
+ }
+
+ mode = (buf[0]>>3)&0xF;
+ if((mode < 0) || (mode > 8))
+ {
+ if(mode != 15)
+ {
+ return -1;
+ }
+ }
+ if(mode != 15)
+ {
+ read_len = in.Read(&buf[1], AMRmode2bytes[mode]);
+ if(read_len != AMRmode2bytes[mode])
+ {
+ return -1;
+ }
+ }
+ _playoutPositionMs += 20;
+ }
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if(!strcmp("#!AMRWB\n", buf))
+ {
+ strcpy(codec_info_.plname, "amr-wb");
+ codec_info_.pacsize = 320;
+ _codecId = kCodecAmrWb;
+ codec_info_.pltype = 120;
+ codec_info_.rate = 20000;
+ codec_info_.plfreq = 16000;
+ codec_info_.channels = 1;
+
+ WebRtc_Word16 mode = 0;
+ if(_startPointInMs > 0)
+ {
+ while (_playoutPositionMs <= _startPointInMs)
+ {
+ // First read byte contain the AMR mode.
+ read_len = in.Read(buf, 1);
+ if(read_len != 1)
+ {
+ return -1;
+ }
+
+ mode = (buf[0]>>3)&0xF;
+ if((mode < 0) || (mode > 9))
+ {
+ if(mode != 15)
+ {
+ return -1;
+ }
+ }
+ if(mode != 15)
+ {
+ read_len = in.Read(&buf[1], AMRWBmode2bytes[mode]);
+ if(read_len != AMRWBmode2bytes[mode])
+ {
+ return -1;
+ }
+ }
+ _playoutPositionMs += 20;
+ }
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if(!strcmp("#!iLBC20\n", buf))
+ {
+ codec_info_.pltype = 102;
+ strcpy(codec_info_.plname, "ilbc");
+ codec_info_.plfreq = 8000;
+ codec_info_.pacsize = 160;
+ codec_info_.channels = 1;
+ codec_info_.rate = 13300;
+ _codecId = kCodecIlbc20Ms;
+
+ if(_startPointInMs > 0)
+ {
+ while (_playoutPositionMs <= _startPointInMs)
+ {
+ read_len = in.Read(buf, 38);
+ if(read_len == 38)
+ {
+ _playoutPositionMs += 20;
+ }
+ else
+ {
+ return -1;
+ }
+ }
+ }
+ }
+
+ if(!strcmp("#!iLBC30\n", buf))
+ {
+ codec_info_.pltype = 102;
+ strcpy(codec_info_.plname, "ilbc");
+ codec_info_.plfreq = 8000;
+ codec_info_.pacsize = 240;
+ codec_info_.channels = 1;
+ codec_info_.rate = 13300;
+ _codecId = kCodecIlbc30Ms;
+
+ if(_startPointInMs > 0)
+ {
+ while (_playoutPositionMs <= _startPointInMs)
+ {
+ read_len = in.Read(buf, 50);
+ if(read_len == 50)
+ {
+ _playoutPositionMs += 20;
+ }
+ else
+ {
+ return -1;
+ }
+ }
+ }
+ }
+#endif
+ if(_codecId == kCodecNoCodec)
+ {
+ return -1;
+ }
+ _reading = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadCompressedData(InStream& in,
+ WebRtc_Word8* outData,
+ WebRtc_UWord32 bufferSize)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x,\
+ bytes=%ld)",
+ &in,
+ outData,
+ bufferSize);
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ WebRtc_UWord32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ WebRtc_UWord32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
+#endif
+ WebRtc_UWord32 bytesRead = 0;
+
+ if(! _reading)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "not currently reading!");
+ return -1;
+ }
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ if(_codecId == kCodecAmr)
+ {
+ WebRtc_Word32 res = in.Read(outData, 1);
+ if(res != 1)
+ {
+ if(!in.Rewind())
+ {
+ InitCompressedReading(in, _startPointInMs, _stopPointInMs);
+ res = in.Read(outData, 1);
+ if(res != 1)
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ else
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ const WebRtc_Word16 mode = (outData[0]>>3)&0xF;
+ if((mode < 0) ||
+ (mode > 8))
+ {
+ if(mode != 15)
+ {
+ return -1;
+ }
+ }
+ if(mode != 15)
+ {
+ if(bufferSize < AMRmode2bytes[mode] + 1)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceFile,
+ _id,
+ "output buffer is too short to read AMR compressed data.");
+ assert(false);
+ return -1;
+ }
+ bytesRead = in.Read(&outData[1], AMRmode2bytes[mode]);
+ if(bytesRead != AMRmode2bytes[mode])
+ {
+ _reading = false;
+ return -1;
+ }
+ // Count the mode byte to bytes read.
+ bytesRead++;
+ }
+ else
+ {
+ bytesRead = 1;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if(_codecId == kCodecAmrWb)
+ {
+ WebRtc_Word32 res = in.Read(outData, 1);
+ if(res != 1)
+ {
+ if(!in.Rewind())
+ {
+ InitCompressedReading(in, _startPointInMs, _stopPointInMs);
+ res = in.Read(outData, 1);
+ if(res != 1)
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ else
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ WebRtc_Word16 mode = (outData[0]>>3)&0xF;
+ if((mode < 0) ||
+ (mode > 8))
+ {
+ if(mode != 15)
+ {
+ return -1;
+ }
+ }
+ if(mode != 15)
+ {
+ if(bufferSize < AMRWBmode2bytes[mode] + 1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "output buffer is too short to read AMRWB\
+ compressed.");
+ assert(false);
+ return -1;
+ }
+ bytesRead = in.Read(&outData[1], AMRWBmode2bytes[mode]);
+ if(bytesRead != AMRWBmode2bytes[mode])
+ {
+ _reading = false;
+ return -1;
+ }
+ bytesRead++;
+ }
+ else
+ {
+ bytesRead = 1;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if((_codecId == kCodecIlbc20Ms) ||
+ (_codecId == kCodecIlbc30Ms))
+ {
+ WebRtc_UWord32 byteSize = 0;
+ if(_codecId == kCodecIlbc30Ms)
+ {
+ byteSize = 50;
+ }
+ if(_codecId == kCodecIlbc20Ms)
+ {
+ byteSize = 38;
+ }
+ if(bufferSize < byteSize)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "output buffer is too short to read ILBC compressed\
+ data.");
+ assert(false);
+ return -1;
+ }
+
+ bytesRead = in.Read(outData, byteSize);
+ if(bytesRead != byteSize)
+ {
+ if(!in.Rewind())
+ {
+ InitCompressedReading(in, _startPointInMs, _stopPointInMs);
+ bytesRead = in.Read(outData, byteSize);
+ if(bytesRead != byteSize)
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ else
+ {
+ _reading = false;
+ return -1;
+ }
+ }
+ }
+#endif
+ if(bytesRead == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadCompressedData() no bytes read, codec not supported");
+ return -1;
+ }
+
+ _playoutPositionMs += 20;
+ if((_stopPointInMs > 0) &&
+ (_playoutPositionMs >= _stopPointInMs))
+ {
+ if(!in.Rewind())
+ {
+ InitCompressedReading(in, _startPointInMs, _stopPointInMs);
+ }
+ else
+ {
+ _reading = false;
+ }
+ }
+ return bytesRead;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitCompressedWriting(
+ OutStream& out,
+ const CodecInst& codecInst)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
+ "ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
+ codecName= %s)",
+ &out, codecInst.plname);
+
+ _writing = false;
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
+ {
+ if(codecInst.pacsize == 160)
+ {
+ memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
+ _codecId = kCodecAmr;
+ out.Write("#!AMR\n",6);
+ _writing = true;
+ return 0;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
+ {
+ if(codecInst.pacsize == 320)
+ {
+ memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
+ _codecId = kCodecAmrWb;
+ out.Write("#!AMRWB\n",8);
+ _writing = true;
+ return 0;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
+ {
+ if(codecInst.pacsize == 160)
+ {
+ _codecId = kCodecIlbc20Ms;
+ out.Write("#!iLBC20\n",9);
+ }
+ else if(codecInst.pacsize == 240)
+ {
+ _codecId = kCodecIlbc30Ms;
+ out.Write("#!iLBC30\n",9);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "codecInst defines unsupported compression codec!");
+ return -1;
+ }
+ memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
+ _writing = true;
+ return 0;
+ }
+#endif
+
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "codecInst defines unsupported compression codec!");
+ return -1;
+}
+
+WebRtc_Word32 ModuleFileUtility::WriteCompressedData(
+ OutStream& out,
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 dataLength)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x,\
+ dataLen= %d)",
+ &out,
+ buffer,
+ dataLength);
+
+ if(buffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+ }
+
+ if(!out.Write(buffer, dataLength))
+ {
+ return -1;
+ }
+ return dataLength;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitPCMReading(InStream& pcm,
+ const WebRtc_UWord32 start,
+ const WebRtc_UWord32 stop,
+ WebRtc_UWord32 freq)
+{
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
+ freq=%d)",
+ &pcm,
+ start,
+ stop,
+ freq);
+
+ WebRtc_Word8 dummy[320];
+ WebRtc_Word32 read_len;
+
+ _playoutPositionMs = 0;
+ _startPointInMs = start;
+ _stopPointInMs = stop;
+ _reading = false;
+
+ if(freq == 8000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 8000;
+ codec_info_.pacsize = 160;
+ codec_info_.channels = 1;
+ codec_info_.rate = 128000;
+ _codecId = kCodecL16_8Khz;
+ }
+ else if(freq == 16000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 16000;
+ codec_info_.pacsize = 320;
+ codec_info_.channels = 1;
+ codec_info_.rate = 256000;
+ _codecId = kCodecL16_16kHz;
+ }
+ else if(freq == 32000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 32000;
+ codec_info_.pacsize = 320;
+ codec_info_.channels = 1;
+ codec_info_.rate = 512000;
+ _codecId = kCodecL16_32Khz;
+ }
+
+ // Readsize for 10ms of audio data (2 bytes per sample).
+ _readSizeBytes = 2 * codec_info_. plfreq / 100;
+ if(_startPointInMs > 0)
+ {
+ while (_playoutPositionMs < _startPointInMs)
+ {
+ read_len = pcm.Read(dummy, _readSizeBytes);
+ if(read_len == _readSizeBytes)
+ {
+ _playoutPositionMs += 10;
+ }
+ else // Must have reached EOF before start position!
+ {
+ return -1;
+ }
+ }
+ }
+ _reading = true;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::ReadPCMData(InStream& pcm,
+ WebRtc_Word8* outData,
+ WebRtc_UWord32 bufferSize)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %d)",
+ &pcm,
+ outData,
+ bufferSize);
+
+ if(outData == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+ }
+
+ // Readsize for 10ms of audio data (2 bytes per sample).
+ WebRtc_UWord32 bytesRequested = 2 * codec_info_.plfreq / 100;
+ if(bufferSize < bytesRequested)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadPCMData: buffer not long enough for a 10ms frame.");
+ assert(false);
+ return -1;
+ }
+
+ WebRtc_UWord32 bytesRead = pcm.Read(outData, bytesRequested);
+ if(bytesRead < bytesRequested)
+ {
+ if(pcm.Rewind() == -1)
+ {
+ _reading = false;
+ }
+ else
+ {
+ if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
+ codec_info_.plfreq) == -1)
+ {
+ _reading = false;
+ }
+ else
+ {
+ WebRtc_Word32 rest = bytesRequested - bytesRead;
+ WebRtc_Word32 len = pcm.Read(&(outData[bytesRead]), rest);
+ if(len == rest)
+ {
+ bytesRead += len;
+ }
+ else
+ {
+ _reading = false;
+ }
+ }
+ if(bytesRead <= 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "ReadPCMData: Failed to rewind audio file.");
+ return -1;
+ }
+ }
+ }
+
+ if(bytesRead <= 0)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ReadPCMData: end of file");
+ return -1;
+ }
+ _playoutPositionMs += 10;
+ if(_stopPointInMs && _playoutPositionMs >= _stopPointInMs)
+ {
+ if(!pcm.Rewind())
+ {
+ if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
+ codec_info_.plfreq) == -1)
+ {
+ _reading = false;
+ }
+ }
+ }
+ return bytesRead;
+}
+
+WebRtc_Word32 ModuleFileUtility::InitPCMWriting(OutStream& out,
+ WebRtc_UWord32 freq)
+{
+
+ if(freq == 8000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 8000;
+ codec_info_.pacsize = 160;
+ codec_info_.channels = 1;
+ codec_info_.rate = 128000;
+
+ _codecId = kCodecL16_8Khz;
+ }
+ else if(freq == 16000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 16000;
+ codec_info_.pacsize = 320;
+ codec_info_.channels = 1;
+ codec_info_.rate = 256000;
+
+ _codecId = kCodecL16_16kHz;
+ }
+ else if(freq == 32000)
+ {
+ strcpy(codec_info_.plname, "L16");
+ codec_info_.pltype = -1;
+ codec_info_.plfreq = 32000;
+ codec_info_.pacsize = 320;
+ codec_info_.channels = 1;
+ codec_info_.rate = 512000;
+
+ _codecId = kCodecL16_32Khz;
+ }
+ if((_codecId != kCodecL16_8Khz) &&
+ (_codecId != kCodecL16_16kHz) &&
+ (_codecId != kCodecL16_32Khz))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "CodecInst is not 8KHz PCM or 16KHz PCM!");
+ return -1;
+ }
+ _writing = true;
+ _bytesWritten = 0;
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::WritePCMData(OutStream& out,
+ const WebRtc_Word8* buffer,
+ const WebRtc_UWord32 dataLength)
+{
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceFile,
+ _id,
+ "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %d)",
+ &out,
+ buffer,
+ dataLength);
+
+ if(buffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
+ }
+
+ if(!out.Write(buffer, dataLength))
+ {
+ return -1;
+ }
+
+ _bytesWritten += dataLength;
+ return dataLength;
+}
+
+WebRtc_Word32 ModuleFileUtility::codec_info(CodecInst& codecInst)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::codec_info(codecInst= 0x%x)", &codecInst);
+
+ if(!_reading && !_writing)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "CodecInst: not currently reading audio file!");
+ return -1;
+ }
+ memcpy(&codecInst,&codec_info_,sizeof(CodecInst));
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::set_codec_info(const CodecInst& codecInst)
+{
+
+ _codecId = kCodecNoCodec;
+ if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
+ {
+ _codecId = kCodecPcmu;
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
+ {
+ _codecId = kCodecPcma;
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
+ {
+ if(codecInst.plfreq == 8000)
+ {
+ _codecId = kCodecL16_8Khz;
+ }
+ else if(codecInst.plfreq == 16000)
+ {
+ _codecId = kCodecL16_16kHz;
+ }
+ else if(codecInst.plfreq == 32000)
+ {
+ _codecId = kCodecL16_32Khz;
+ }
+ }
+#ifdef WEBRTC_CODEC_GSMAMR
+ else if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
+ {
+ _codecId = kCodecAmr;
+ }
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ else if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
+ {
+ _codecId = kCodecAmrWb;
+ }
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ else if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
+ {
+ if(codecInst.pacsize == 160)
+ {
+ _codecId = kCodecIlbc20Ms;
+ }
+ else if(codecInst.pacsize == 240)
+ {
+ _codecId = kCodecIlbc30Ms;
+ }
+ }
+#endif
+#if(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ else if(STR_CASE_CMP(codecInst.plname, "isac") == 0)
+ {
+ if(codecInst.plfreq == 16000)
+ {
+ _codecId = kCodecIsac;
+ }
+ else if(codecInst.plfreq == 32000)
+ {
+ _codecId = kCodecIsacSwb;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_ISACLC
+ else if(STR_CASE_CMP(codecInst.plname, "isaclc") == 0)
+ {
+ _codecId = kCodecIsacLc;
+ }
+#endif
+#ifdef WEBRTC_CODEC_G722
+ else if(STR_CASE_CMP(codecInst.plname, "G722") == 0)
+ {
+ _codecId = kCodecG722;
+ }
+#endif
+ else if(STR_CASE_CMP(codecInst.plname, "G7221") == 0)
+ {
+#ifdef WEBRTC_CODEC_G722_1
+ if(codecInst.plfreq == 16000)
+ {
+ if(codecInst.rate == 16000)
+ {
+ _codecId = kCodecG722_1_16Kbps;
+ }
+ else if(codecInst.rate == 24000)
+ {
+ _codecId = kCodecG722_1_24Kbps;
+ }
+ else if(codecInst.rate == 32000)
+ {
+ _codecId = kCodecG722_1_32Kbps;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ if(codecInst.plfreq == 32000)
+ {
+ if(codecInst.rate == 48000)
+ {
+ _codecId = kCodecG722_1c_48;
+ }
+ else if(codecInst.rate == 32000)
+ {
+ _codecId = kCodecG722_1c_32;
+ }
+ else if(codecInst.rate == 24000)
+ {
+ _codecId = kCodecG722_1c_24;
+ }
+ }
+#endif
+ }
+#ifdef WEBRTC_CODEC_G726
+ else if(STR_CASE_CMP(codecInst.plname, "G726-40") == 0)
+ {
+ _codecId = kCodecG726_40;
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "G726-32") == 0)
+ {
+ _codecId = kCodecG726_24;
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "G726-24") == 0)
+ {
+ _codecId = kCodecG726_32;
+ }
+ else if(STR_CASE_CMP(codecInst.plname, "G726-16") == 0)
+ {
+ _codecId = kCodecG726_16;
+ }
+#endif
+#ifdef WEBRTC_CODEC_G729
+ else if(STR_CASE_CMP(codecInst.plname, "G729") == 0)
+ {
+ _codecId = kCodecG729;
+ }
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ else if(STR_CASE_CMP(codecInst.plname, "G7291") == 0)
+ {
+ _codecId = kCodecG729_1;
+ }
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ else if(STR_CASE_CMP(codecInst.plname, "speex") == 0)
+ {
+ if(codecInst.plfreq == 8000)
+ {
+ _codecId = kCodecSpeex8Khz;
+ }
+ else if(codecInst.plfreq == 16000)
+ {
+ _codecId = kCodecSpeex16Khz;
+ }
+ }
+#endif
+ if(_codecId == kCodecNoCodec)
+ {
+ return -1;
+ }
+ memcpy(&codec_info_, &codecInst, sizeof(CodecInst));
+ return 0;
+}
+
+WebRtc_Word32 ModuleFileUtility::FileDurationMs(const WebRtc_Word8* fileName,
+ const FileFormats fileFormat,
+ const WebRtc_UWord32 freqInHz)
+{
+
+ if(fileName == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id, "filename NULL");
+ return -1;
+ }
+
+ WebRtc_Word32 time_in_ms = -1;
+ struct stat file_size;
+ if(stat(fileName,&file_size) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "failed to retrieve file size with stat!");
+ return -1;
+ }
+ FileWrapper* inStreamObj = FileWrapper::Create();
+ if(inStreamObj == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
+ "failed to create InStream object!");
+ return -1;
+ }
+ if(inStreamObj->OpenFile(fileName, true) == -1)
+ {
+ delete inStreamObj;
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "failed to open file %s!", fileName);
+ return -1;
+ }
+
+ switch (fileFormat)
+ {
+ case kFileFormatWavFile:
+ {
+ if(ReadWavHeader(*inStreamObj) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "failed to read WAV file header!");
+ return -1;
+ }
+ time_in_ms = ((file_size.st_size - 44) /
+ (_wavFormatObj.nAvgBytesPerSec/1000));
+ break;
+ }
+ case kFileFormatPcm16kHzFile:
+ {
+ // 16 samples per ms. 2 bytes per sample.
+ WebRtc_Word32 denominator = 16*2;
+ time_in_ms = (file_size.st_size)/denominator;
+ break;
+ }
+ case kFileFormatPcm8kHzFile:
+ {
+ // 8 samples per ms. 2 bytes per sample.
+ WebRtc_Word32 denominator = 8*2;
+ time_in_ms = (file_size.st_size)/denominator;
+ break;
+ }
+ case kFileFormatCompressedFile:
+ {
+ WebRtc_Word32 cnt = 0;
+ WebRtc_Word32 read_len = 0;
+ WebRtc_Word8 buf[64];
+ do
+ {
+ read_len = inStreamObj->Read(&buf[cnt++], 1);
+ if(read_len != 1)
+ {
+ return -1;
+ }
+ } while ((buf[cnt-1] != '\n') && (64 > cnt));
+
+ if(cnt == 64)
+ {
+ return -1;
+ }
+ else
+ {
+ buf[cnt] = 0;
+ }
+#ifdef WEBRTC_CODEC_GSMAMR
+ if(!strcmp("#!AMR\n", buf))
+ {
+ WebRtc_UWord8 dummy;
+ read_len = inStreamObj->Read(&dummy, 1);
+ if(read_len != 1)
+ {
+ return -1;
+ }
+
+ WebRtc_Word16 AMRMode = (dummy>>3)&0xF;
+
+ // TODO (hellner): use tables instead of hardcoding like this!
+ // Additionally, this calculation does not
+ // take octet alignment into consideration.
+ switch (AMRMode)
+ {
+ // Mode 0: 4.75 kbit/sec -> 95 bits per 20 ms frame.
+ // 20 ms = 95 bits ->
+ // file size in bytes * 8 / 95 is the number of
+ // 20 ms frames in the file ->
+ // time_in_ms = file size * 8 / 95 * 20
+ case 0:
+ time_in_ms = ((file_size.st_size)*160)/95;
+ break;
+ // Mode 1: 5.15 kbit/sec -> 103 bits per 20 ms frame.
+ case 1:
+ time_in_ms = ((file_size.st_size)*160)/103;
+ break;
+ // Mode 2: 5.90 kbit/sec -> 118 bits per 20 ms frame.
+ case 2:
+ time_in_ms = ((file_size.st_size)*160)/118;
+ break;
+ // Mode 3: 6.70 kbit/sec -> 134 bits per 20 ms frame.
+ case 3:
+ time_in_ms = ((file_size.st_size)*160)/134;
+ break;
+ // Mode 4: 7.40 kbit/sec -> 148 bits per 20 ms frame.
+ case 4:
+ time_in_ms = ((file_size.st_size)*160)/148;
+ break;
+ // Mode 5: 7.95 bit/sec -> 159 bits per 20 ms frame.
+ case 5:
+ time_in_ms = ((file_size.st_size)*160)/159;
+ break;
+ // Mode 6: 10.2 bit/sec -> 204 bits per 20 ms frame.
+ case 6:
+ time_in_ms = ((file_size.st_size)*160)/204;
+ break;
+ // Mode 7: 12.2 bit/sec -> 244 bits per 20 ms frame.
+ case 7:
+ time_in_ms = ((file_size.st_size)*160)/244;
+ break;
+ // Mode 8: SID Mode -> 39 bits per 20 ms frame.
+ case 8:
+ time_in_ms = ((file_size.st_size)*160)/39;
+ break;
+ default:
+ break;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if(!strcmp("#!AMRWB\n", buf))
+ {
+ WebRtc_UWord8 dummy;
+ read_len = inStreamObj->Read(&dummy, 1);
+ if(read_len != 1)
+ {
+ return -1;
+ }
+
+ // TODO (hellner): use tables instead of hardcoding like this!
+ WebRtc_Word16 AMRWBMode = (dummy>>3)&0xF;
+ switch(AMRWBMode)
+ {
+ // Mode 0: 6.6 kbit/sec -> 132 bits per 20 ms frame.
+ case 0:
+ time_in_ms = ((file_size.st_size)*160)/132;
+ break;
+ // Mode 1: 8.85 kbit/sec -> 177 bits per 20 ms frame.
+ case 1:
+ time_in_ms = ((file_size.st_size)*160)/177;
+ break;
+ // Mode 2: 12.65 kbit/sec -> 253 bits per 20 ms frame.
+ case 2:
+ time_in_ms = ((file_size.st_size)*160)/253;
+ break;
+ // Mode 3: 14.25 kbit/sec -> 285 bits per 20 ms frame.
+ case 3:
+ time_in_ms = ((file_size.st_size)*160)/285;
+ break;
+ // Mode 4: 15.85 kbit/sec -> 317 bits per 20 ms frame.
+ case 4:
+ time_in_ms = ((file_size.st_size)*160)/317;
+ break;
+ // Mode 5: 18.25 kbit/sec -> 365 bits per 20 ms frame.
+ case 5:
+ time_in_ms = ((file_size.st_size)*160)/365;
+ break;
+ // Mode 6: 19.85 kbit/sec -> 397 bits per 20 ms frame.
+ case 6:
+ time_in_ms = ((file_size.st_size)*160)/397;
+ break;
+ // Mode 7: 23.05 kbit/sec -> 461 bits per 20 ms frame.
+ case 7:
+ time_in_ms = ((file_size.st_size)*160)/461;
+ break;
+ // Mode 8: 23.85 kbit/sec -> 477 bits per 20 ms frame.
+ case 8:
+ time_in_ms = ((file_size.st_size)*160)/477;
+ break;
+ default:
+ delete inStreamObj;
+ return -1;
+ }
+ }
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if(!strcmp("#!iLBC20\n", buf))
+ {
+ // 20 ms is 304 bits
+ time_in_ms = ((file_size.st_size)*160)/304;
+ break;
+ }
+ if(!strcmp("#!iLBC30\n", buf))
+ {
+ // 30 ms takes 400 bits.
+ // file size in bytes * 8 / 400 is the number of
+ // 30 ms frames in the file ->
+ // time_in_ms = file size * 8 / 400 * 30
+ time_in_ms = ((file_size.st_size)*240)/400;
+ break;
+ }
+#endif
+ }
+ case kFileFormatPreencodedFile:
+ {
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "cannot determine duration of Pre-Encoded file!");
+ break;
+ }
+ default:
+ WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+ "unsupported file format %d!", fileFormat);
+ break;
+ }
+ inStreamObj->CloseFile();
+ delete inStreamObj;
+ return time_in_ms;
+}
+
+WebRtc_UWord32 ModuleFileUtility::PlayoutPositionMs()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+ "ModuleFileUtility::PlayoutPosition()");
+
+ if(_reading)
+ {
+ return _playoutPositionMs;
+ }
+ else
+ {
+ return 0;
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/media_file_utility.h b/trunk/src/modules/media_file/source/media_file_utility.h
new file mode 100644
index 0000000..cfe0314
--- /dev/null
+++ b/trunk/src/modules/media_file/source/media_file_utility.h
@@ -0,0 +1,349 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Note: the class cannot be used for reading and writing at the same time.
+#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+
+#include <stdio.h>
+
+#include "common_types.h"
+#include "media_file_defines.h"
+
+namespace webrtc {
+class AviFile;
+class InStream;
+class OutStream;
+
+class ModuleFileUtility
+{
+public:
+
+ ModuleFileUtility(const WebRtc_Word32 id);
+ ~ModuleFileUtility();
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ // Open the file specified by fileName for reading (relative path is
+ // allowed). If loop is true the file will be played until StopPlaying() is
+ // called. When end of file is reached the file is read from the start.
+ // Only video will be read if videoOnly is true.
+ WebRtc_Word32 InitAviReading(const WebRtc_Word8* fileName, bool videoOnly,
+ bool loop);
+
+ // Put 10-60ms of audio data from file into the outBuffer depending on
+ // codec frame size. bufferLengthInBytes indicates the size of outBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ // Note: This API only play mono audio but can be used on file containing
+ // audio with more channels (in which case the audio will be coverted to
+ // mono).
+ WebRtc_Word32 ReadAviAudioData(WebRtc_Word8* outBuffer,
+ const WebRtc_UWord32 bufferLengthInBytes);
+
+ // Put one video frame into outBuffer. bufferLengthInBytes indicates the
+ // size of outBuffer.
+ // The return value is the number of bytes written to videoBuffer.
+ WebRtc_Word32 ReadAviVideoData(WebRtc_Word8* videoBuffer,
+ const WebRtc_UWord32 bufferLengthInBytes);
+
+ // Open/create the file specified by fileName for writing audio/video data
+ // (relative path is allowed). codecInst specifies the encoding of the audio
+ // data. videoCodecInst specifies the encoding of the video data. Only video
+ // data will be recorded if videoOnly is true.
+ WebRtc_Word32 InitAviWriting(const WebRtc_Word8* filename,
+ const CodecInst& codecInst,
+ const VideoCodec& videoCodecInst,
+ const bool videoOnly);
+
+ // Write one audio frame, i.e. the bufferLengthinBytes first bytes of
+ // audioBuffer, to file. The audio frame size is determined by the
+ // codecInst.pacsize parameter of the last sucessfull
+ // InitAviWriting(..) call.
+ // Note: bufferLength must be exactly one frame.
+ WebRtc_Word32 WriteAviAudioData(const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord32 bufferLengthInBytes);
+
+
+ // Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
+ // to file.
+ // Note: videoBuffer can contain encoded data. The codec used must be the
+ // same as what was specified by videoCodecInst for the last successfull
+ // InitAviWriting(..) call. The videoBuffer must contain exactly
+ // one video frame.
+ WebRtc_Word32 WriteAviVideoData(const WebRtc_Word8* videoBuffer,
+ WebRtc_UWord32 bufferLengthInBytes);
+
+ // Stop recording to file or stream.
+ WebRtc_Word32 CloseAviFile();
+
+ WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst);
+#endif // #ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+ // Prepare for playing audio from stream.
+ // startPointMs and stopPointMs, unless zero, specify what part of the file
+ // should be read. From startPointMs ms to stopPointMs ms.
+ WebRtc_Word32 InitWavReading(InStream& stream,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0);
+
+ // Put 10-60ms of audio data from stream into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ // Note: This API only play mono audio but can be used on file containing
+ // audio with more channels (in which case the audio will be converted to
+ // mono).
+ WebRtc_Word32 ReadWavDataAsMono(InStream& stream, WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Put 10-60ms, depending on codec frame size, of audio data from file into
+ // audioBufferLeft and audioBufferRight. The buffers contain the left and
+ // right channel of played out stereo audio.
+ // dataLengthInBytes indicates the size of both audioBufferLeft and
+ // audioBufferRight.
+ // The return value is the number of bytes read for each buffer.
+ // Note: This API can only be successfully called for WAV files with stereo
+ // audio.
+ WebRtc_Word32 ReadWavDataAsStereo(InStream& wav,
+ WebRtc_Word8* audioBufferLeft,
+ WebRtc_Word8* audioBufferRight,
+ const WebRtc_UWord32 bufferLength);
+
+ // Prepare for recording audio to stream.
+ // codecInst specifies the encoding of the audio data.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ WebRtc_Word32 InitWavWriting(OutStream& stream, const CodecInst& codecInst);
+
+ // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to file. The audio frame size is determined by the codecInst.pacsize
+ // parameter of the last sucessfull StartRecordingAudioFile(..) call.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 WriteWavData(OutStream& stream,
+ const WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 bufferLength);
+
+ // Finalizes the WAV header so that it is correct if nothing more will be
+ // written to stream.
+ // Note: this API must be called before closing stream to ensure that the
+ // WAVE header is updated with the file size. Don't call this API
+ // if more samples are to be written to stream.
+ WebRtc_Word32 UpdateWavHeader(OutStream& stream);
+
+ // Prepare for playing audio from stream.
+ // startPointMs and stopPointMs, unless zero, specify what part of the file
+ // should be read. From startPointMs ms to stopPointMs ms.
+ // freqInHz is the PCM sampling frequency.
+ // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
+ WebRtc_Word32 InitPCMReading(InStream& stream,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0,
+ const WebRtc_UWord32 freqInHz = 16000);
+
+ // Put 10-60ms of audio data from stream into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 ReadPCMData(InStream& stream, WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Prepare for recording audio to stream.
+ // freqInHz is the PCM sampling frequency.
+ // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
+ WebRtc_Word32 InitPCMWriting(OutStream& stream,
+ const WebRtc_UWord32 freqInHz = 16000);
+
+ // Write one 10ms audio frame, i.e. the bufferLength first bytes of
+ // audioBuffer, to file. The audio frame size is determined by the freqInHz
+ // parameter of the last sucessfull InitPCMWriting(..) call.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 WritePCMData(OutStream& stream,
+ const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord32 bufferLength);
+
+ // Prepare for playing audio from stream.
+ // startPointMs and stopPointMs, unless zero, specify what part of the file
+ // should be read. From startPointMs ms to stopPointMs ms.
+ WebRtc_Word32 InitCompressedReading(InStream& stream,
+ const WebRtc_UWord32 startPointMs = 0,
+ const WebRtc_UWord32 stopPointMs = 0);
+
+ // Put 10-60ms of audio data from stream into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 ReadCompressedData(InStream& stream,
+ WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Prepare for recording audio to stream.
+ // codecInst specifies the encoding of the audio data.
+ WebRtc_Word32 InitCompressedWriting(OutStream& stream,
+ const CodecInst& codecInst);
+
+ // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to file. The audio frame size is determined by the codecInst.pacsize
+ // parameter of the last sucessfull InitCompressedWriting(..) call.
+ // The return value is the number of bytes written to stream.
+ // Note: bufferLength must be exactly one frame.
+ WebRtc_Word32 WriteCompressedData(OutStream& stream,
+ const WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 bufferLength);
+
+ // Prepare for playing audio from stream.
+ // codecInst specifies the encoding of the audio data.
+ WebRtc_Word32 InitPreEncodedReading(InStream& stream,
+ const CodecInst& codecInst);
+
+ // Put 10-60ms of audio data from stream into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 ReadPreEncodedData(InStream& stream,
+ WebRtc_Word8* audioBuffer,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Prepare for recording audio to stream.
+ // codecInst specifies the encoding of the audio data.
+ WebRtc_Word32 InitPreEncodedWriting(OutStream& stream,
+ const CodecInst& codecInst);
+
+ // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to stream. The audio frame size is determined by the codecInst.pacsize
+ // parameter of the last sucessfull InitPreEncodedWriting(..) call.
+ // The return value is the number of bytes written to stream.
+ // Note: bufferLength must be exactly one frame.
+ WebRtc_Word32 WritePreEncodedData(OutStream& stream,
+ const WebRtc_Word8* inData,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Set durationMs to the size of the file (in ms) specified by fileName.
+ // freqInHz specifies the sampling frequency of the file.
+ WebRtc_Word32 FileDurationMs(const WebRtc_Word8* fileName,
+ const FileFormats fileFormat,
+ const WebRtc_UWord32 freqInHz = 16000);
+
+ // Return the number of ms that have been played so far.
+ WebRtc_UWord32 PlayoutPositionMs();
+
+ // Update codecInst according to the current audio codec being used for
+ // reading or writing.
+ WebRtc_Word32 codec_info(CodecInst& codecInst);
+
+private:
+ // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
+ enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
+
+
+ WebRtc_Word32 InitWavCodec(WebRtc_UWord32 samplesPerSec,
+ WebRtc_UWord32 channels,
+ WebRtc_UWord32 bitsPerSample,
+ WebRtc_UWord32 formatTag);
+
+ // Parse the WAV header in stream.
+ WebRtc_Word32 ReadWavHeader(InStream& stream);
+
+ // Update the WAV header. freqInHz, bytesPerSample, channels, format,
+ // lengthInBytes specify characterists of the audio data.
+ // freqInHz is the sampling frequency. bytesPerSample is the sample size in
+ // bytes. channels is the number of channels, e.g. 1 is mono and 2 is
+ // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
+ // lengthInBytes is the number of bytes the audio samples are using up.
+ WebRtc_Word32 WriteWavHeader(OutStream& stream,
+ const WebRtc_UWord32 freqInHz,
+ const WebRtc_UWord32 bytesPerSample,
+ const WebRtc_UWord32 channels,
+ const WebRtc_UWord32 format,
+ const WebRtc_UWord32 lengthInBytes);
+
+ // Put dataLengthInBytes of audio data from stream into the audioBuffer.
+ // The return value is the number of bytes written to audioBuffer.
+ WebRtc_Word32 ReadWavData(InStream& stream, WebRtc_UWord8* audioBuffer,
+ const WebRtc_UWord32 dataLengthInBytes);
+
+ // Update the current audio codec being used for reading or writing
+ // according to codecInst.
+ WebRtc_Word32 set_codec_info(const CodecInst& codecInst);
+
+ struct WAVE_FMTINFO_header
+ {
+ WebRtc_Word16 formatTag;
+ WebRtc_Word16 nChannels;
+ WebRtc_Word32 nSamplesPerSec;
+ WebRtc_Word32 nAvgBytesPerSec;
+ WebRtc_Word16 nBlockAlign;
+ WebRtc_Word16 nBitsPerSample;
+ };
+ // Identifiers for preencoded files.
+ enum MediaFileUtility_CodecType
+ {
+ kCodecNoCodec = 0,
+ kCodecIsac,
+ kCodecIsacSwb,
+ kCodecIsacLc,
+ kCodecL16_8Khz,
+ kCodecL16_16kHz,
+ kCodecL16_32Khz,
+ kCodecPcmu,
+ kCodecPcma,
+ kCodecIlbc20Ms,
+ kCodecIlbc30Ms,
+ kCodecG722,
+ kCodecG722_1_32Kbps,
+ kCodecG722_1_24Kbps,
+ kCodecG722_1_16Kbps,
+ kCodecG722_1c_48,
+ kCodecG722_1c_32,
+ kCodecG722_1c_24,
+ kCodecAmr,
+ kCodecAmrWb,
+ kCodecG729,
+ kCodecG729_1,
+ kCodecG726_40,
+ kCodecG726_32,
+ kCodecG726_24,
+ kCodecG726_16,
+ kCodecSpeex8Khz,
+ kCodecSpeex16Khz
+ };
+
+ // TODO (hellner): why store multiple formats. Just store either codec_info_
+ // or _wavFormatObj and supply conversion functions.
+ WAVE_FMTINFO_header _wavFormatObj;
+ WebRtc_Word32 _dataSize; // Chunk size if reading a WAV file
+ // Number of bytes to read. I.e. frame size in bytes. May be multiple
+ // chunks if reading WAV.
+ WebRtc_Word32 _readSizeBytes;
+
+ WebRtc_Word32 _id;
+
+ WebRtc_UWord32 _stopPointInMs;
+ WebRtc_UWord32 _startPointInMs;
+ WebRtc_UWord32 _playoutPositionMs;
+ WebRtc_UWord32 _bytesWritten;
+
+ CodecInst codec_info_;
+ MediaFileUtility_CodecType _codecId;
+
+ // The amount of bytes, on average, used for one audio sample.
+ WebRtc_Word32 _bytesPerSample;
+ WebRtc_Word32 _readPos;
+
+ // Only reading or writing can be enabled, not both.
+ bool _reading;
+ bool _writing;
+
+ // Scratch buffer used for turning stereo audio to mono.
+ WebRtc_UWord8 _tempData[WAV_MAX_BUFFER_SIZE];
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ AviFile* _aviAudioInFile;
+ AviFile* _aviVideoInFile;
+ AviFile* _aviOutFile;
+ VideoCodec _videoCodec;
+#endif
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
diff --git a/trunk/src/modules/modules.gyp b/trunk/src/modules/modules.gyp
new file mode 100644
index 0000000..83b5cc6
--- /dev/null
+++ b/trunk/src/modules/modules.gyp
@@ -0,0 +1,60 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ 'audio_coding/codecs/cng/cng.gypi',
+ 'audio_coding/codecs/g711/g711.gypi',
+ 'audio_coding/codecs/g722/g722.gypi',
+ 'audio_coding/codecs/ilbc/ilbc.gypi',
+ 'audio_coding/codecs/iSAC/main/source/isac.gypi',
+ 'audio_coding/codecs/iSAC/fix/source/isacfix.gypi',
+ 'audio_coding/codecs/pcm16b/pcm16b.gypi',
+ 'audio_coding/main/source/audio_coding_module.gypi',
+ 'audio_coding/neteq/neteq.gypi',
+ 'audio_conference_mixer/source/audio_conference_mixer.gypi',
+ 'audio_device/main/source/audio_device.gypi',
+ 'audio_processing/audio_processing.gypi',
+ 'audio_processing/aec/aec.gypi',
+ 'audio_processing/aecm/aecm.gypi',
+ 'audio_processing/agc/agc.gypi',
+ 'audio_processing/ns/ns.gypi',
+ 'audio_processing/utility/util.gypi',
+ 'media_file/source/media_file.gypi',
+ 'udp_transport/source/udp_transport.gypi',
+ 'utility/source/utility.gypi',
+ 'video_coding/codecs/i420/main/source/i420.gypi',
+ 'video_coding/codecs/test_framework/test_framework.gypi',
+ 'video_coding/codecs/vp8/main/source/vp8.gypi',
+ 'video_coding/main/source/video_coding.gypi',
+ 'video_capture/main/source/video_capture.gypi',
+ 'video_processing/main/source/video_processing.gypi',
+ 'video_render/main/source/video_render.gypi',
+ 'rtp_rtcp/source/rtp_rtcp.gypi',
+ ],
+
+ # Test targets, excluded when building with Chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'includes': [
+ 'audio_coding/codecs/iSAC/isac_test.gypi',
+ 'audio_coding/codecs/iSAC/isacfix_test.gypi',
+ 'audio_processing/apm_tests.gypi',
+ 'rtp_rtcp/source/rtp_rtcp_tests.gypi',
+ 'rtp_rtcp/test/test_bwe/test_bwe.gypi',
+ 'rtp_rtcp/test/testFec/test_fec.gypi',
+ 'rtp_rtcp/test/testAPI/test_api.gypi',
+ 'video_coding/main/source/video_coding_test.gypi',
+ 'video_coding/codecs/test/video_codecs_test_framework.gypi',
+ 'video_coding/codecs/tools/video_codecs_tools.gypi',
+ 'video_processing/main/test/vpm_tests.gypi',
+ ], # includes
+ }], # build_with_chromium
+ ], # conditions
+}
diff --git a/trunk/src/modules/rtp_rtcp/OWNERS b/trunk/src/modules/rtp_rtcp/OWNERS
new file mode 100644
index 0000000..c968564
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/OWNERS
@@ -0,0 +1,5 @@
+pwestin@webrtc.org
+stefan@webrtc.org
+henrik.lundin@webrtc.org
+mflodman@webrtc.org
+asapersson@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h
new file mode 100644
index 0000000..6b55202
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -0,0 +1,1094 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+
+#include <vector>
+
+#include "module.h"
+#include "rtp_rtcp_defines.h"
+
+namespace webrtc {
+// forward declaration
+class Transport;
+
+class RtpRtcp : public Module
+{
+public:
+ /*
+ * create a RTP/RTCP module object using the system clock
+ *
+ * id - unique identifier of this RTP/RTCP module object
+ * audio - true for a audio version of the RTP/RTCP module object false will create a video version
+ */
+ static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id,
+ const bool audio);
+
+ /*
+ * create a RTP/RTCP module object
+ *
+ * id - unique identifier of this RTP/RTCP module object
+ * audio - true for a audio version of the RTP/RTCP module object
+ * false will create a video version
+ * clock - the clock to use to read time; must not be NULL
+ */
+ static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock);
+
+ /*
+ * destroy a RTP/RTCP module object
+ *
+ * module - object to destroy
+ */
+ static void DestroyRtpRtcp(RtpRtcp* module);
+
+ /*
+ * Change the unique identifier of this object
+ *
+ * id - new unique identifier of this RTP/RTCP module object
+ */
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ /*
+ * De-muxing functionality for conferencing
+ *
+ * register a module that will act as a default module for this module
+ * used for feedback messages back to the encoder when one encoded stream
+ * is sent to multiple destinations
+ *
+ * module - default module
+ */
+ virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module) = 0;
+
+ /*
+ * unregister the default module
+ * will stop the demuxing feedback
+ */
+ virtual WebRtc_Word32 DeRegisterDefaultModule() = 0;
+
+ /*
+ * returns true if a default module is registered, false otherwise
+ */
+ virtual bool DefaultModuleRegistered() = 0;
+
+ /*
+ * returns number of registered child modules
+ */
+ virtual WebRtc_UWord32 NumberChildModules() = 0;
+
+ /*
+ * Lip-sync between voice-video
+ *
+ * module - audio module
+ *
+ * Note: only allowed on a video module
+ */
+ virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module) = 0;
+
+ /*
+ * Turn off lip-sync between voice-video
+ */
+ virtual WebRtc_Word32 DeRegisterSyncModule() = 0;
+
+ /**************************************************************************
+ *
+ * Receiver functions
+ *
+ ***************************************************************************/
+
+ /*
+ * Initialize receive side
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 InitReceiver() = 0;
+
+ /*
+ * Used by the module to deliver the incoming data to the codec module
+ *
+ * incomingDataCallback - callback object that will receive the incoming data
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback) = 0;
+
+ /*
+ * Used by the module to deliver messages to the codec module/appliation
+ *
+ * incomingMessagesCallback - callback object that will receive the incoming messages
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback) = 0;
+
+ /*
+ * configure a RTP packet timeout value
+ *
+ * RTPtimeoutMS - time in milliseconds after last received RTP packet
+ * RTCPtimeoutMS - time in milliseconds after last received RTCP packet
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
+ const WebRtc_UWord32 RTCPtimeoutMS) = 0;
+
+ /*
+ * Set periodic dead or alive notification
+ *
+ * enable - turn periodic dead or alive notification on/off
+ * sampleTimeSeconds - sample interval in seconds for dead or alive notifications
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(const bool enable,
+ const WebRtc_UWord8 sampleTimeSeconds) = 0;
+
+ /*
+ * Get periodic dead or alive notification status
+ *
+ * enable - periodic dead or alive notification on/off
+ * sampleTimeSeconds - sample interval in seconds for dead or alive notifications
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(bool &enable,
+ WebRtc_UWord8 &sampleTimeSeconds) = 0;
+
+ /*
+ * set voice codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterReceivePayload(
+ const CodecInst& voiceCodec) = 0;
+
+ /*
+ * set video codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterReceivePayload(
+ const VideoCodec& videoCodec) = 0;
+
+ /*
+ * get payload type for a voice codec
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ReceivePayloadType(
+ const CodecInst& voiceCodec,
+ WebRtc_Word8* plType) = 0;
+
+ /*
+ * get payload type for a video codec
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ReceivePayloadType(
+ const VideoCodec& videoCodec,
+ WebRtc_Word8* plType) = 0;
+
+ /*
+ * Remove a registered payload type from list of accepted payloads
+ *
+ * payloadType - payload type of codec
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 DeRegisterReceivePayload(
+ const WebRtc_Word8 payloadType) = 0;
+
+ /*
+ * (De)register RTP header extension type and id.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id) = 0;
+
+ virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type) = 0;
+
+ /*
+ * Get last received remote timestamp
+ */
+ virtual WebRtc_UWord32 RemoteTimestamp() const = 0;
+
+ /*
+ * Get the current estimated remote timestamp
+ *
+ * timestamp - estimated timestamp
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const = 0;
+
+ /*
+ * Get incoming SSRC
+ */
+ virtual WebRtc_UWord32 RemoteSSRC() const = 0;
+
+ /*
+ * Get remote CSRC
+ *
+ * arrOfCSRC - array that will receive the CSRCs
+ *
+ * return -1 on failure else the number of valid entries in the list
+ */
+ virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
+
+ /*
+ * get the currently configured SSRC filter
+ *
+ * allowedSSRC - SSRC that will be allowed through
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const = 0;
+
+ /*
+ * set a SSRC to be used as a filter for incoming RTP streams
+ *
+ * allowedSSRC - SSRC that will be allowed through
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
+ const WebRtc_UWord32 allowedSSRC) = 0;
+
+ /*
+ * Turn on/off receiving RTX (RFC 4588) on a specific SSRC.
+ */
+ virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
+ const WebRtc_UWord32 SSRC) = 0;
+
+ /*
+ * Get status of receiving RTX (RFC 4588) on a specific SSRC.
+ */
+ virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const = 0;
+
+ /*
+ * called by the network module when we receive a packet
+ *
+ * incomingPacket - incoming packet buffer
+ * packetLength - length of incoming buffer
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPacket,
+ const WebRtc_UWord16 packetLength) = 0;
+
+
+ /*
+ * Option when not using the RegisterSyncModule function
+ *
+ * Inform the module about the received audion NTP
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 IncomingAudioNTP(
+ const WebRtc_UWord32 audioReceivedNTPsecs,
+ const WebRtc_UWord32 audioReceivedNTPfrac,
+ const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
+ const WebRtc_UWord32 audioRTCPArrivalTimeFrac) = 0;
+
+ /**************************************************************************
+ *
+ * Sender
+ *
+ ***************************************************************************/
+
+ /*
+ * Initialize send side
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 InitSender() = 0;
+
+ /*
+ * Used by the module to send RTP and RTCP packet to the network module
+ *
+ * outgoingTransport - transport object that will be called when packets are ready to be sent out on the network
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport) = 0;
+
+ /*
+ * set MTU
+ *
+ * size - Max transfer unit in bytes, default is 1500
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size) = 0;
+
+ /*
+ * set transtport overhead
+ * default is IPv4 and UDP with no encryption
+ *
+ * TCP - true for TCP false UDP
+ * IPv6 - true for IP version 6 false for version 4
+ * authenticationOverhead - number of bytes to leave for an authentication header
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
+ const bool IPV6,
+ const WebRtc_UWord8 authenticationOverhead = 0) = 0;
+
+ /*
+ * Get max payload length
+ *
+ * A combination of the configuration MaxTransferUnit and TransportOverhead.
+ * Does not account FEC/ULP/RED overhead if FEC is enabled.
+ * Does not account for RTP headers
+ */
+ virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
+
+ /*
+ * Get max data payload length
+ *
+ * A combination of the configuration MaxTransferUnit, headers and TransportOverhead.
+ * Takes into account FEC/ULP/RED overhead if FEC is enabled.
+ * Takes into account RTP headers
+ */
+ virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
+
+ /*
+ * set RTPKeepaliveStatus
+ *
+ * enable - on/off
+ * unknownPayloadType - payload type to use for RTP keepalive
+ * deltaTransmitTimeMS - delta time between RTP keepalive packets
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetRTPKeepaliveStatus(
+ const bool enable,
+ const WebRtc_Word8 unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS) = 0;
+
+ /*
+ * Get RTPKeepaliveStatus
+ *
+ * enable - on/off
+ * unknownPayloadType - payload type in use for RTP keepalive
+ * deltaTransmitTimeMS - delta time between RTP keepalive packets
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RTPKeepaliveStatus(
+ bool* enable,
+ WebRtc_Word8* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const = 0;
+
+ /*
+ * check if RTPKeepaliveStatus is enabled
+ */
+ virtual bool RTPKeepalive() const = 0;
+
+ /*
+ * set codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterSendPayload(
+ const CodecInst& voiceCodec) = 0;
+
+ /*
+ * set codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterSendPayload(
+ const VideoCodec& videoCodec) = 0;
+
+ /*
+ * Unregister a send payload
+ *
+ * payloadType - payload type of codec
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 DeRegisterSendPayload(
+ const WebRtc_Word8 payloadType) = 0;
+
+ /*
+ * (De)register RTP header extension type and id.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id) = 0;
+
+ virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
+ const RTPExtensionType type) = 0;
+
+ /*
+ * Enable/disable traffic smoothing of sending stream.
+ */
+ virtual void SetTransmissionSmoothingStatus(const bool enable) = 0;
+
+ virtual bool TransmissionSmoothingStatus() const = 0;
+
+ /*
+ * get start timestamp
+ */
+ virtual WebRtc_UWord32 StartTimestamp() const = 0;
+
+ /*
+ * configure start timestamp, default is a random number
+ *
+ * timestamp - start timestamp
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetStartTimestamp(
+ const WebRtc_UWord32 timestamp) = 0;
+
+ /*
+ * Get SequenceNumber
+ */
+ virtual WebRtc_UWord16 SequenceNumber() const = 0;
+
+ /*
+ * Set SequenceNumber, default is a random number
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq) = 0;
+
+ /*
+ * Get SSRC
+ */
+ virtual WebRtc_UWord32 SSRC() const = 0;
+
+ /*
+ * configure SSRC, default is a random number
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc) = 0;
+
+ /*
+ * Get CSRC
+ *
+ * arrOfCSRC - array of CSRCs
+ *
+ * return -1 on failure else number of valid entries in the array
+ */
+ virtual WebRtc_Word32 CSRCs(
+ WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
+
+ /*
+ * Set CSRC
+ *
+ * arrOfCSRC - array of CSRCs
+ * arrLength - number of valid entries in the array
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetCSRCs(
+ const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength) = 0;
+
+ /*
+ * includes CSRCs in RTP header if enabled
+ *
+ * include CSRC - on/off
+ *
+ * default:on
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetCSRCStatus(const bool include) = 0;
+
+ /*
+ * Turn on/off sending RTX (RFC 4588) on a specific SSRC.
+ */
+ virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
+ const bool setSSRC,
+ const WebRtc_UWord32 SSRC) = 0;
+
+
+ /*
+ * Get status of sending RTX (RFC 4588) on a specific SSRC.
+ */
+ virtual WebRtc_Word32 RTXSendStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const = 0;
+
+ /*
+ * sends kRtcpByeCode when going from true to false
+ *
+ * sending - on/off
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSendingStatus(const bool sending) = 0;
+
+ /*
+ * get send status
+ */
+ virtual bool Sending() const = 0;
+
+ /*
+ * Starts/Stops media packets, on by default
+ *
+ * sending - on/off
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending) = 0;
+
+ /*
+ * get send status
+ */
+ virtual bool SendingMedia() const = 0;
+
+ /*
+ * get sent bitrate in Kbit/s
+ */
+ virtual void BitrateSent(WebRtc_UWord32* totalRate,
+ WebRtc_UWord32* videoRate,
+ WebRtc_UWord32* fecRate,
+ WebRtc_UWord32* nackRate) const = 0;
+
+ /*
+ * Get the send-side estimate of the available bandwidth.
+ */
+ virtual int EstimatedSendBandwidth(
+ WebRtc_UWord32* available_bandwidth) const = 0;
+
+ /*
+ * Get the receive-side estimate of the available bandwidth.
+ */
+ virtual int EstimatedReceiveBandwidth(
+ WebRtc_UWord32* available_bandwidth) const = 0;
+
+ /*
+ * Used by the codec module to deliver a video or audio frame for packetization
+ *
+ * frameType - type of frame to send
+ * payloadType - payload type of frame to send
+ * timestamp - timestamp of frame to send
+ * payloadData - payload buffer of frame to send
+ * payloadSize - size of payload buffer to send
+ * fragmentation - fragmentation offset data for fragmented frames such as layers or RED
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SendOutgoingData(
+ const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation = NULL,
+ const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
+
+ /**************************************************************************
+ *
+ * RTCP
+ *
+ ***************************************************************************/
+
+ /*
+ * RegisterIncomingRTCPCallback
+ *
+ * incomingMessagesCallback - callback object that will receive messages from RTCP
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback) = 0;
+
+ /*
+ * Get RTCP status
+ */
+ virtual RTCPMethod RTCP() const = 0;
+
+ /*
+ * configure RTCP status i.e on(compound or non- compound)/off
+ *
+ * method - RTCP method to use
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method) = 0;
+
+ /*
+ * Set RTCP CName (i.e unique identifier)
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
+
+ /*
+ * Get RTCP CName (i.e unique identifier)
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]) = 0;
+
+ /*
+ * Get remote CName
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoteCNAME(
+ const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const = 0;
+
+ /*
+ * Get remote NTP
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoteNTP(
+ WebRtc_UWord32 *ReceivedNTPsecs,
+ WebRtc_UWord32 *ReceivedNTPfrac,
+ WebRtc_UWord32 *RTCPArrivalTimeSecs,
+ WebRtc_UWord32 *RTCPArrivalTimeFrac) const = 0;
+
+ /*
+ * AddMixedCNAME
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 AddMixedCNAME(
+ const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]) = 0;
+
+ /*
+ * RemoveMixedCNAME
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC) = 0;
+
+ /*
+ * Get RoundTripTime
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
+ WebRtc_UWord16* RTT,
+ WebRtc_UWord16* avgRTT,
+ WebRtc_UWord16* minRTT,
+ WebRtc_UWord16* maxRTT) const = 0 ;
+
+ /*
+ * Reset RoundTripTime statistics
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC)= 0 ;
+
+ /*
+ * Force a send of a RTCP packet
+ * normal SR and RR are triggered via the process function
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport) = 0;
+
+ /*
+ * Good state of RTP receiver inform sender
+ */
+ virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID) = 0;
+
+ /*
+ * Send a RTCP Slice Loss Indication (SLI)
+ * 6 least significant bits of pictureID
+ */
+ virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID) = 0;
+
+ /*
+ * Reset RTP statistics
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ResetStatisticsRTP() = 0;
+
+ /*
+ * statistics of our localy created statistics of the received RTP stream
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost, // scale 0 to 255
+ WebRtc_UWord32 *cum_lost, // number of lost packets
+ WebRtc_UWord32 *ext_max, // highest sequence number received
+ WebRtc_UWord32 *jitter,
+ WebRtc_UWord32 *max_jitter = NULL) const = 0;
+
+ /*
+ * Reset RTP data counters for the receiving side
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ResetReceiveDataCountersRTP() = 0;
+
+ /*
+ * Reset RTP data counters for the sending side
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 ResetSendDataCountersRTP() = 0;
+
+ /*
+ * statistics of the amount of data sent and received
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
+ WebRtc_UWord32 *packetsSent,
+ WebRtc_UWord32 *bytesReceived,
+ WebRtc_UWord32 *packetsReceived) const = 0;
+ /*
+ * Get received RTCP sender info
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
+
+ /*
+ * Get received RTCP report block
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoteRTCPStat(
+ std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
+ /*
+ * Set received RTCP report block
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 AddRTCPReportBlock(
+ const WebRtc_UWord32 SSRC,
+ const RTCPReportBlock* receiveBlock) = 0;
+
+ /*
+ * RemoveRTCPReportBlock
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC) = 0;
+
+ /*
+ * (APP) Application specific data
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 length) = 0;
+ /*
+ * (XR) VOIP metric
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) = 0;
+
+ /*
+ * (REMB) Receiver Estimated Max Bitrate
+ */
+ virtual bool REMB() const = 0;
+
+ virtual WebRtc_Word32 SetREMBStatus(const bool enable) = 0;
+
+ virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
+ const WebRtc_UWord8 numberOfSSRC,
+ const WebRtc_UWord32* SSRC) = 0;
+
+ // Used to set maximum bitrate estimate received in a REMB packet.
+ virtual WebRtc_Word32 SetMaximumBitrateEstimate(
+ const WebRtc_UWord32 bitrate) = 0;
+
+ // Registers an observer to call when the estimate of the incoming channel
+ // changes.
+ virtual bool SetRemoteBitrateObserver(
+ RtpRemoteBitrateObserver* observer) = 0;
+
+ /*
+ * (IJ) Extended jitter report.
+ */
+ virtual bool IJ() const = 0;
+
+ virtual WebRtc_Word32 SetIJStatus(const bool enable) = 0;
+
+ /*
+ * (TMMBR) Temporary Max Media Bit Rate
+ */
+ virtual bool TMMBR() const = 0;
+
+ /*
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetTMMBRStatus(const bool enable) = 0;
+
+ /*
+ * local bw estimation changed
+ *
+ * for video called by internal estimator
+ * for audio (iSAC) called by engine, geting the data from the decoder
+ */
+ virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit) = 0;
+
+ /*
+ * (NACK)
+ */
+ virtual NACKMethod NACK() const = 0;
+
+ /*
+ * Turn negative acknowledgement requests on/off
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0;
+
+ /*
+ * TODO(holmer): Propagate this API to VideoEngine.
+ * Returns the currently configured selective retransmission settings.
+ */
+ virtual int SelectiveRetransmissions() const = 0;
+
+ /*
+ * TODO(holmer): Propagate this API to VideoEngine.
+ * Sets the selective retransmission settings, which will decide which
+ * packets will be retransmitted if NACKed. Settings are constructed by
+ * combining the constants in enum RetransmissionMode with bitwise OR.
+ * All packets are retransmitted if kRetransmitAllPackets is set, while no
+ * packets are retransmitted if kRetransmitOff is set.
+ * By default all packets except FEC packets are retransmitted. For VP8
+ * with temporal scalability only base layer packets are retransmitted.
+ *
+ * Returns -1 on failure, otherwise 0.
+ */
+ virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
+
+ /*
+ * Send a Negative acknowledgement packet
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
+ const WebRtc_UWord16 size) = 0;
+
+ /*
+ * Store the sent packets, needed to answer to a Negative acknowledgement requests
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200) = 0;
+
+ /**************************************************************************
+ *
+ * Audio
+ *
+ ***************************************************************************/
+
+ /*
+ * RegisterAudioCallback
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback) = 0;
+
+ /*
+ * set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples) = 0;
+
+ /*
+ * Outband TelephoneEvent(DTMF) detection
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
+ const bool forwardToDecoder,
+ const bool detectEndOfTone = false) = 0;
+
+ /*
+ * Is outband TelephoneEvent(DTMF) turned on/off?
+ */
+ virtual bool TelephoneEvent() const = 0;
+
+ /*
+ * Returns true if received DTMF events are forwarded to the decoder using
+ * the OnPlayTelephoneEvent callback.
+ */
+ virtual bool TelephoneEventForwardToDecoder() const = 0;
+
+ /*
+ * SendTelephoneEventActive
+ *
+ * return true if we currently send a telephone event and 100 ms after an event is sent
+ * used to prevent teh telephone event tone to be recorded by the microphone and send inband
+ * just after the tone has ended
+ */
+ virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const = 0;
+
+ /*
+ * Send a TelephoneEvent tone using RFC 2833 (4733)
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level) = 0;
+
+ /*
+ * Set payload type for Redundant Audio Data RFC 2198
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType) = 0;
+
+ /*
+ * Get payload type for Redundant Audio Data RFC 2198
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const = 0;
+
+ /*
+ * Set status and ID for header-extension-for-audio-level-indication.
+ * See https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+ * for more details.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID) = 0;
+
+ /*
+ * Get status and ID for header-extension-for-audio-level-indication.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const = 0;
+
+ /*
+ * Store the audio level in dBov for header-extension-for-audio-level-indication.
+ * This API shall be called before transmision of an RTP packet to ensure
+ * that the |level| part of the extended RTP header is updated.
+ *
+ * return -1 on failure else 0.
+ */
+ virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov) = 0;
+
+ /**************************************************************************
+ *
+ * Video
+ *
+ ***************************************************************************/
+
+ /*
+ * Register a callback object that will receive callbacks for video related events
+ * such as an incoming key frame request.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback) = 0;
+
+ /*
+ * Set the estimated camera delay in MS
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0;
+
+ /*
+ * Set the start and max send bitrate
+ * used by the bandwidth management
+ *
+ * Not calling this or setting startBitrateKbit to 0 disables the bandwidth management
+ *
+ * minBitrateKbit = 0 equals no min bitrate
+ * maxBitrateKbit = 0 equals no max bitrate
+ *
+ * return -1 on failure else 0
+ */
+ virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
+ const WebRtc_UWord16 minBitrateKbit,
+ const WebRtc_UWord16 maxBitrateKbit) = 0;
+
+ /*
+ * Turn on/off generic FEC
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC) = 0;
+
+ /*
+ * Get generic FEC setting
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) = 0;
+
+
+ /*
+ * Set FEC code rate of key and delta frames
+ * codeRate on a scale of 0 to 255 where 255 is 100% added packets, hence protect up to 50% packet loss
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate) = 0;
+
+
+ /*
+ * Set FEC unequal protection (UEP) across packets,
+ * for key and delta frames.
+ *
+ * If keyUseUepProtection is true UEP is enabled for key frames.
+ * If deltaUseUepProtection is true UEP is enabled for delta frames.
+ *
+ * UEP skews the FEC protection towards being spent more on the
+ * important packets, at the cost of less FEC protection for the
+ * non-important packets.
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection) = 0;
+
+
+ /*
+ * Set method for requestion a new key frame
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method) = 0;
+
+ /*
+ * send a request for a keyframe
+ *
+ * return -1 on failure else 0
+ */
+ virtual WebRtc_Word32 RequestKeyFrame() = 0;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
diff --git a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
new file mode 100644
index 0000000..b6a8a54
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -0,0 +1,282 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+#ifndef NULL
+ #define NULL 0
+#endif
+
+#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500 // we assume ethernet
+#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
+#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds
+
+namespace webrtc{
+enum RTCPMethod
+{
+ kRtcpOff = 0,
+ kRtcpCompound = 1,
+ kRtcpNonCompound = 2
+};
+
+enum RTPAliveType
+{
+ kRtpDead = 0,
+ kRtpNoRtp = 1,
+ kRtpAlive = 2
+};
+
+enum StorageType {
+ kDontStore,
+ kDontRetransmit,
+ kAllowRetransmission
+};
+
+enum RTPExtensionType
+{
+ kRtpExtensionNone,
+ kRtpExtensionTransmissionTimeOffset,
+ kRtpExtensionAudioLevel,
+};
+
+enum RTCPAppSubTypes
+{
+ kAppSubtypeBwe = 0x00
+};
+
+enum RTCPPacketType
+{
+ kRtcpReport = 0x0001,
+ kRtcpSr = 0x0002,
+ kRtcpRr = 0x0004,
+ kRtcpBye = 0x0008,
+ kRtcpPli = 0x0010,
+ kRtcpNack = 0x0020,
+ kRtcpFir = 0x0040,
+ kRtcpTmmbr = 0x0080,
+ kRtcpTmmbn = 0x0100,
+ kRtcpSrReq = 0x0200,
+ kRtcpXrVoipMetric = 0x0400,
+ kRtcpApp = 0x0800,
+ kRtcpSli = 0x4000,
+ kRtcpRpsi = 0x8000,
+ kRtcpRemb = 0x10000,
+ kRtcpTransmissionTimeOffset = 0x20000
+};
+
+enum KeyFrameRequestMethod
+{
+ kKeyFrameReqFirRtp = 1,
+ kKeyFrameReqPliRtcp = 2,
+ kKeyFrameReqFirRtcp = 3
+};
+
+enum RtpRtcpPacketType
+{
+ kPacketRtp = 0,
+ kPacketKeepAlive = 1
+};
+
+enum NACKMethod
+{
+ kNackOff = 0,
+ kNackRtcp = 2
+};
+
+enum RetransmissionMode {
+ kRetransmitOff = 0x0,
+ kRetransmitFECPackets = 0x1,
+ kRetransmitBaseLayer = 0x2,
+ kRetransmitHigherLayers = 0x4,
+ kRetransmitAllPackets = 0xFF
+};
+
+struct RTCPSenderInfo
+{
+ WebRtc_UWord32 NTPseconds;
+ WebRtc_UWord32 NTPfraction;
+ WebRtc_UWord32 RTPtimeStamp;
+ WebRtc_UWord32 sendPacketCount;
+ WebRtc_UWord32 sendOctetCount;
+};
+
+struct RTCPReportBlock
+{
+ // Fields as described by RFC 3550 6.4.2.
+ WebRtc_UWord32 remoteSSRC; // SSRC of sender of this report.
+ WebRtc_UWord32 sourceSSRC; // SSRC of the RTP packet sender.
+ WebRtc_UWord8 fractionLost;
+ WebRtc_UWord32 cumulativeLost; // 24 bits valid
+ WebRtc_UWord32 extendedHighSeqNum;
+ WebRtc_UWord32 jitter;
+ WebRtc_UWord32 lastSR;
+ WebRtc_UWord32 delaySinceLastSR;
+};
+
+class RtpData
+{
+public:
+ virtual WebRtc_Word32 OnReceivedPayloadData(
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader) = 0;
+protected:
+ virtual ~RtpData() {}
+};
+
+class RtcpFeedback
+{
+public:
+ // if audioVideoOffset > 0 video is behind audio
+ virtual void OnLipSyncUpdate(const WebRtc_Word32 /*id*/,
+ const WebRtc_Word32 /*audioVideoOffset*/) {};
+
+ virtual void OnApplicationDataReceived(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord8 /*subType*/,
+ const WebRtc_UWord32 /*name*/,
+ const WebRtc_UWord16 /*length*/,
+ const WebRtc_UWord8* /*data*/) {};
+
+ virtual void OnXRVoIPMetricReceived(
+ const WebRtc_Word32 /*id*/,
+ const RTCPVoIPMetric* /*metric*/,
+ const WebRtc_Word8 /*VoIPmetricBuffer*/[28]) {};
+
+ virtual void OnRTCPPacketTimeout(const WebRtc_Word32 /*id*/) {};
+
+ virtual void OnTMMBRReceived(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord16 /*bwEstimateKbit*/) {};
+
+ virtual void OnSLIReceived(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord8 /*pictureId*/) {};
+
+ virtual void OnRPSIReceived(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord64 /*pictureId*/) {};
+
+ virtual void OnReceiverEstimatedMaxBitrateReceived(
+ const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord32 /*bitRate*/) {};
+
+ virtual void OnSendReportReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord32 senderSSRC) {};
+
+ virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord32 senderSSRC) {};
+
+protected:
+ virtual ~RtcpFeedback() {}
+};
+
+class RtpFeedback
+{
+public:
+ // Receiving payload change or SSRC change. (return success!)
+ /*
+ * channels - number of channels in codec (1 = mono, 2 = stereo)
+ */
+ virtual WebRtc_Word32 OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payloadType,
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) = 0;
+
+ virtual void OnPacketTimeout(const WebRtc_Word32 id) = 0;
+
+ virtual void OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType) = 0;
+
+ virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive) = 0;
+
+ virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC) = 0;
+
+ virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added) = 0;
+
+protected:
+ virtual ~RtpFeedback() {}
+};
+
+class RtpAudioFeedback
+{
+public:
+ virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const bool endOfEvent) = 0;
+
+ virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const WebRtc_UWord16 lengthMs,
+ const WebRtc_UWord8 volume) = 0;
+
+protected:
+ virtual ~RtpAudioFeedback() {}
+};
+
+
+class RtpVideoFeedback
+{
+public:
+ // this function should call codec module to inform it about the request
+ virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+ const FrameType type,
+ const WebRtc_UWord8 streamIdx) = 0;
+
+ virtual void OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 bitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs) = 0;
+
+protected:
+ virtual ~RtpVideoFeedback() {}
+};
+
+// A clock interface that allows reading of absolute and relative
+// timestamps in an RTP/RTCP module.
+class RtpRtcpClock {
+public:
+ virtual ~RtpRtcpClock() {}
+
+ // Return a timestamp in milliseconds relative to some arbitrary
+ // source; the source is fixed for this clock.
+ virtual WebRtc_UWord32 GetTimeInMS() = 0;
+
+ // Retrieve an NTP absolute timestamp.
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) = 0;
+};
+
+// RtpReceiveBitrateUpdate is used to signal changes in bitrate estimates for
+// the incoming stream.
+class RtpRemoteBitrateObserver
+{
+public:
+ // Called when a receive channel has a new bitrate estimate for the incoming
+ // stream.
+ virtual void OnReceiveBitrateChanged(unsigned int ssrc,
+ unsigned int bitrate) = 0;
+
+ // Called when a REMB packet has been received.
+ virtual void OnReceivedRemb(unsigned int bitrate) = 0;
+
+ virtual ~RtpRemoteBitrateObserver() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
diff --git a/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
new file mode 100644
index 0000000..cc71e7b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -0,0 +1,288 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../testing/gmock/include/gmock/gmock.h"
+
+#include "modules/interface/module.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class MockRtpRtcp : public RtpRtcp {
+ public:
+ MOCK_METHOD1(ChangeUniqueId,
+ WebRtc_Word32(const WebRtc_Word32 id));
+ MOCK_METHOD1(RegisterDefaultModule,
+ WebRtc_Word32(RtpRtcp* module));
+ MOCK_METHOD0(DeRegisterDefaultModule,
+ WebRtc_Word32());
+ MOCK_METHOD0(DefaultModuleRegistered,
+ bool());
+ MOCK_METHOD0(NumberChildModules,
+ WebRtc_UWord32());
+ MOCK_METHOD1(RegisterSyncModule,
+ WebRtc_Word32(RtpRtcp* module));
+ MOCK_METHOD0(DeRegisterSyncModule,
+ WebRtc_Word32());
+ MOCK_METHOD0(InitReceiver,
+ WebRtc_Word32());
+ MOCK_METHOD1(RegisterIncomingDataCallback,
+ WebRtc_Word32(RtpData* incomingDataCallback));
+ MOCK_METHOD1(RegisterIncomingRTPCallback,
+ WebRtc_Word32(RtpFeedback* incomingMessagesCallback));
+ MOCK_METHOD2(SetPacketTimeout,
+ WebRtc_Word32(const WebRtc_UWord32 RTPtimeoutMS, const WebRtc_UWord32 RTCPtimeoutMS));
+ MOCK_METHOD2(SetPeriodicDeadOrAliveStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord8 sampleTimeSeconds));
+ MOCK_METHOD2(PeriodicDeadOrAliveStatus,
+ WebRtc_Word32(bool &enable, WebRtc_UWord8 &sampleTimeSeconds));
+ MOCK_METHOD1(RegisterReceivePayload,
+ WebRtc_Word32(const CodecInst& voiceCodec));
+ MOCK_METHOD1(RegisterReceivePayload,
+ WebRtc_Word32(const VideoCodec& videoCodec));
+ MOCK_METHOD2(ReceivePayloadType,
+ WebRtc_Word32(const CodecInst& voiceCodec, WebRtc_Word8* plType));
+ MOCK_METHOD2(ReceivePayloadType,
+ WebRtc_Word32(const VideoCodec& videoCodec, WebRtc_Word8* plType));
+ MOCK_METHOD1(DeRegisterReceivePayload,
+ WebRtc_Word32(const WebRtc_Word8 payloadType));
+ MOCK_METHOD2(RegisterReceiveRtpHeaderExtension,
+ WebRtc_Word32(const RTPExtensionType type, const WebRtc_UWord8 id));
+ MOCK_METHOD1(DeregisterReceiveRtpHeaderExtension,
+ WebRtc_Word32(const RTPExtensionType type));
+ MOCK_CONST_METHOD0(RemoteTimestamp,
+ WebRtc_UWord32());
+ MOCK_CONST_METHOD1(EstimatedRemoteTimeStamp,
+ WebRtc_Word32(WebRtc_UWord32& timestamp));
+ MOCK_CONST_METHOD0(RemoteSSRC,
+ WebRtc_UWord32());
+ MOCK_CONST_METHOD1(RemoteCSRCs,
+ WebRtc_Word32(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]));
+ MOCK_CONST_METHOD1(SSRCFilter,
+ WebRtc_Word32(WebRtc_UWord32& allowedSSRC));
+ MOCK_METHOD2(SetSSRCFilter,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord32 allowedSSRC));
+ MOCK_METHOD2(SetRTXReceiveStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord32 SSRC));
+ MOCK_CONST_METHOD2(RTXReceiveStatus,
+ WebRtc_Word32(bool* enable, WebRtc_UWord32* SSRC));
+ MOCK_METHOD2(IncomingPacket,
+ WebRtc_Word32(const WebRtc_UWord8* incomingPacket, const WebRtc_UWord16 packetLength));
+ MOCK_METHOD4(IncomingAudioNTP,
+ WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs, const WebRtc_UWord32 audioReceivedNTPfrac, const WebRtc_UWord32 audioRTCPArrivalTimeSecs, const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
+ MOCK_METHOD0(InitSender,
+ WebRtc_Word32());
+ MOCK_METHOD1(RegisterSendTransport,
+ WebRtc_Word32(Transport* outgoingTransport));
+ MOCK_METHOD1(SetMaxTransferUnit,
+ WebRtc_Word32(const WebRtc_UWord16 size));
+ MOCK_METHOD3(SetTransportOverhead,
+ WebRtc_Word32(const bool TCP, const bool IPV6, const WebRtc_UWord8 authenticationOverhead));
+ MOCK_CONST_METHOD0(MaxPayloadLength,
+ WebRtc_UWord16());
+ MOCK_CONST_METHOD0(MaxDataPayloadLength,
+ WebRtc_UWord16());
+ MOCK_METHOD3(SetRTPKeepaliveStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_Word8 unknownPayloadType, const WebRtc_UWord16 deltaTransmitTimeMS));
+ MOCK_CONST_METHOD3(RTPKeepaliveStatus,
+ WebRtc_Word32(bool* enable, WebRtc_Word8* unknownPayloadType, WebRtc_UWord16* deltaTransmitTimeMS));
+ MOCK_CONST_METHOD0(RTPKeepalive,
+ bool());
+ MOCK_METHOD1(RegisterSendPayload,
+ WebRtc_Word32(const CodecInst& voiceCodec));
+ MOCK_METHOD1(RegisterSendPayload,
+ WebRtc_Word32(const VideoCodec& videoCodec));
+ MOCK_METHOD1(DeRegisterSendPayload,
+ WebRtc_Word32(const WebRtc_Word8 payloadType));
+ MOCK_METHOD2(RegisterSendRtpHeaderExtension,
+ WebRtc_Word32(const RTPExtensionType type, const WebRtc_UWord8 id));
+ MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
+ WebRtc_Word32(const RTPExtensionType type));
+ MOCK_METHOD1(SetTransmissionSmoothingStatus,
+ void(const bool enable));
+ MOCK_CONST_METHOD0(TransmissionSmoothingStatus,
+ bool());
+ MOCK_CONST_METHOD0(StartTimestamp,
+ WebRtc_UWord32());
+ MOCK_METHOD1(SetStartTimestamp,
+ WebRtc_Word32(const WebRtc_UWord32 timestamp));
+ MOCK_CONST_METHOD0(SequenceNumber,
+ WebRtc_UWord16());
+ MOCK_METHOD1(SetSequenceNumber,
+ WebRtc_Word32(const WebRtc_UWord16 seq));
+ MOCK_CONST_METHOD0(SSRC,
+ WebRtc_UWord32());
+ MOCK_METHOD1(SetSSRC,
+ WebRtc_Word32(const WebRtc_UWord32 ssrc));
+ MOCK_CONST_METHOD1(CSRCs,
+ WebRtc_Word32(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]));
+ MOCK_METHOD2(SetCSRCs,
+ WebRtc_Word32(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], const WebRtc_UWord8 arrLength));
+ MOCK_METHOD1(SetCSRCStatus,
+ WebRtc_Word32(const bool include));
+ MOCK_METHOD3(SetRTXSendStatus,
+ WebRtc_Word32(const bool enable, const bool setSSRC, const WebRtc_UWord32 SSRC));
+ MOCK_CONST_METHOD2(RTXSendStatus,
+ WebRtc_Word32(bool* enable, WebRtc_UWord32* SSRC));
+ MOCK_METHOD1(SetSendingStatus,
+ WebRtc_Word32(const bool sending));
+ MOCK_CONST_METHOD0(Sending,
+ bool());
+ MOCK_METHOD1(SetSendingMediaStatus,
+ WebRtc_Word32(const bool sending));
+ MOCK_CONST_METHOD0(SendingMedia,
+ bool());
+ MOCK_CONST_METHOD4(BitrateSent,
+ void(WebRtc_UWord32* totalRate, WebRtc_UWord32* videoRate, WebRtc_UWord32* fecRate, WebRtc_UWord32* nackRate));
+ MOCK_CONST_METHOD1(EstimatedSendBandwidth,
+ int(WebRtc_UWord32* available_bandwidth));
+ MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
+ int(WebRtc_UWord32* available_bandwidth));
+ MOCK_METHOD7(SendOutgoingData,
+ WebRtc_Word32(const FrameType frameType, const WebRtc_Word8 payloadType, const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpVideoHdr));
+ MOCK_METHOD1(RegisterIncomingRTCPCallback,
+ WebRtc_Word32(RtcpFeedback* incomingMessagesCallback));
+ MOCK_CONST_METHOD0(RTCP,
+ RTCPMethod());
+ MOCK_METHOD1(SetRTCPStatus,
+ WebRtc_Word32(const RTCPMethod method));
+ MOCK_METHOD1(SetCNAME,
+ WebRtc_Word32(const char cName[RTCP_CNAME_SIZE]));
+ MOCK_METHOD1(CNAME,
+ WebRtc_Word32(char cName[RTCP_CNAME_SIZE]));
+ MOCK_CONST_METHOD2(RemoteCNAME,
+ WebRtc_Word32(const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]));
+ MOCK_CONST_METHOD4(RemoteNTP,
+ WebRtc_Word32(WebRtc_UWord32 *ReceivedNTPsecs, WebRtc_UWord32 *ReceivedNTPfrac, WebRtc_UWord32 *RTCPArrivalTimeSecs, WebRtc_UWord32 *RTCPArrivalTimeFrac));
+ MOCK_METHOD2(AddMixedCNAME,
+ WebRtc_Word32(const WebRtc_UWord32 SSRC, const WebRtc_Word8 cName[RTCP_CNAME_SIZE]));
+ MOCK_METHOD1(RemoveMixedCNAME,
+ WebRtc_Word32(const WebRtc_UWord32 SSRC));
+ MOCK_CONST_METHOD5(RTT,
+ WebRtc_Word32(const WebRtc_UWord32 remoteSSRC, WebRtc_UWord16* RTT, WebRtc_UWord16* avgRTT, WebRtc_UWord16* minRTT, WebRtc_UWord16* maxRTT));
+ MOCK_METHOD1(ResetRTT,
+ WebRtc_Word32(const WebRtc_UWord32 remoteSSRC));
+ MOCK_METHOD1(SendRTCP,
+ WebRtc_Word32(WebRtc_UWord32 rtcpPacketType));
+ MOCK_METHOD1(SendRTCPReferencePictureSelection,
+ WebRtc_Word32(const WebRtc_UWord64 pictureID));
+ MOCK_METHOD1(SendRTCPSliceLossIndication,
+ WebRtc_Word32(const WebRtc_UWord8 pictureID));
+ MOCK_METHOD0(ResetStatisticsRTP,
+ WebRtc_Word32());
+ MOCK_CONST_METHOD5(StatisticsRTP,
+ WebRtc_Word32(WebRtc_UWord8 *fraction_lost, WebRtc_UWord32 *cum_lost, WebRtc_UWord32 *ext_max, WebRtc_UWord32 *jitter, WebRtc_UWord32 *max_jitter));
+ MOCK_METHOD0(ResetReceiveDataCountersRTP,
+ WebRtc_Word32());
+ MOCK_METHOD0(ResetSendDataCountersRTP,
+ WebRtc_Word32());
+ MOCK_CONST_METHOD4(DataCountersRTP,
+ WebRtc_Word32(WebRtc_UWord32 *bytesSent, WebRtc_UWord32 *packetsSent, WebRtc_UWord32 *bytesReceived, WebRtc_UWord32 *packetsReceived));
+ MOCK_METHOD1(RemoteRTCPStat,
+ WebRtc_Word32(RTCPSenderInfo* senderInfo));
+ MOCK_CONST_METHOD1(RemoteRTCPStat,
+ WebRtc_Word32(std::vector<RTCPReportBlock>* receiveBlocks));
+ MOCK_METHOD2(AddRTCPReportBlock,
+ WebRtc_Word32(const WebRtc_UWord32 SSRC, const RTCPReportBlock* receiveBlock));
+ MOCK_METHOD1(RemoveRTCPReportBlock,
+ WebRtc_Word32(const WebRtc_UWord32 SSRC));
+ MOCK_METHOD4(SetRTCPApplicationSpecificData,
+ WebRtc_Word32(const WebRtc_UWord8 subType, const WebRtc_UWord32 name, const WebRtc_UWord8* data, const WebRtc_UWord16 length));
+ MOCK_METHOD1(SetRTCPVoIPMetrics,
+ WebRtc_Word32(const RTCPVoIPMetric* VoIPMetric));
+ MOCK_CONST_METHOD0(REMB,
+ bool());
+ MOCK_METHOD1(SetREMBStatus,
+ WebRtc_Word32(const bool enable));
+ MOCK_METHOD3(SetREMBData,
+ WebRtc_Word32(const WebRtc_UWord32 bitrate, const WebRtc_UWord8 numberOfSSRC, const WebRtc_UWord32* SSRC));
+ MOCK_METHOD1(SetMaximumBitrateEstimate,
+ WebRtc_Word32(const WebRtc_UWord32 bitrate));
+ MOCK_METHOD1(SetRemoteBitrateObserver,
+ bool(RtpRemoteBitrateObserver*));
+ MOCK_CONST_METHOD0(IJ,
+ bool());
+ MOCK_METHOD1(SetIJStatus,
+ WebRtc_Word32(const bool));
+ MOCK_CONST_METHOD0(TMMBR,
+ bool());
+ MOCK_METHOD1(SetTMMBRStatus,
+ WebRtc_Word32(const bool enable));
+ MOCK_METHOD1(OnBandwidthEstimateUpdate,
+ void(WebRtc_UWord16 bandWidthKbit));
+ MOCK_CONST_METHOD0(NACK,
+ NACKMethod());
+ MOCK_METHOD1(SetNACKStatus,
+ WebRtc_Word32(const NACKMethod method));
+ MOCK_CONST_METHOD0(SelectiveRetransmissions,
+ int());
+ MOCK_METHOD1(SetSelectiveRetransmissions,
+ int(uint8_t settings));
+ MOCK_METHOD2(SendNACK,
+ WebRtc_Word32(const WebRtc_UWord16* nackList, const WebRtc_UWord16 size));
+ MOCK_METHOD2(SetStorePacketsStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord16 numberToStore));
+ MOCK_METHOD1(RegisterAudioCallback,
+ WebRtc_Word32(RtpAudioFeedback* messagesCallback));
+ MOCK_METHOD1(SetAudioPacketSize,
+ WebRtc_Word32(const WebRtc_UWord16 packetSizeSamples));
+ MOCK_METHOD3(SetTelephoneEventStatus,
+ WebRtc_Word32(const bool enable, const bool forwardToDecoder, const bool detectEndOfTone));
+ MOCK_CONST_METHOD0(TelephoneEvent,
+ bool());
+ MOCK_CONST_METHOD0(TelephoneEventForwardToDecoder,
+ bool());
+ MOCK_CONST_METHOD1(SendTelephoneEventActive,
+ bool(WebRtc_Word8& telephoneEvent));
+ MOCK_METHOD3(SendTelephoneEventOutband,
+ WebRtc_Word32(const WebRtc_UWord8 key, const WebRtc_UWord16 time_ms, const WebRtc_UWord8 level));
+ MOCK_METHOD1(SetSendREDPayloadType,
+ WebRtc_Word32(const WebRtc_Word8 payloadType));
+ MOCK_CONST_METHOD1(SendREDPayloadType,
+ WebRtc_Word32(WebRtc_Word8& payloadType));
+ MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord8 ID));
+ MOCK_CONST_METHOD2(GetRTPAudioLevelIndicationStatus,
+ WebRtc_Word32(bool& enable, WebRtc_UWord8& ID));
+ MOCK_METHOD1(SetAudioLevel,
+ WebRtc_Word32(const WebRtc_UWord8 level_dBov));
+ MOCK_METHOD1(RegisterIncomingVideoCallback,
+ WebRtc_Word32(RtpVideoFeedback* incomingMessagesCallback));
+ MOCK_METHOD1(SetCameraDelay,
+ WebRtc_Word32(const WebRtc_Word32 delayMS));
+ MOCK_METHOD3(SetSendBitrate,
+ void(const WebRtc_UWord32 startBitrate, const WebRtc_UWord16 minBitrateKbit, const WebRtc_UWord16 maxBitrateKbit));
+ MOCK_METHOD3(SetGenericFECStatus,
+ WebRtc_Word32(const bool enable, const WebRtc_UWord8 payloadTypeRED, const WebRtc_UWord8 payloadTypeFEC));
+ MOCK_METHOD3(GenericFECStatus,
+ WebRtc_Word32(bool& enable, WebRtc_UWord8& payloadTypeRED, WebRtc_UWord8& payloadTypeFEC));
+ MOCK_METHOD2(SetFECCodeRate,
+ WebRtc_Word32(const WebRtc_UWord8 keyFrameCodeRate, const WebRtc_UWord8 deltaFrameCodeRate));
+ MOCK_METHOD2(SetFECUepProtection,
+ WebRtc_Word32(const bool keyUseUepProtection, const bool deltaUseUepProtection));
+ MOCK_METHOD1(SetKeyFrameRequestMethod,
+ WebRtc_Word32(const KeyFrameRequestMethod method));
+ MOCK_METHOD0(RequestKeyFrame,
+ WebRtc_Word32());
+
+ MOCK_CONST_METHOD3(Version,
+ int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
+ MOCK_METHOD0(TimeUntilNextProcess,
+ int32_t());
+ MOCK_METHOD0(Process,
+ int32_t());
+
+ // Members.
+ unsigned int remote_ssrc_;
+};
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/Android.mk b/trunk/src/modules/rtp_rtcp/source/Android.mk
new file mode 100644
index 0000000..68295be
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/Android.mk
@@ -0,0 +1,67 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_rtp_rtcp
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_GENERATED_SOURCES :=
+LOCAL_SRC_FILES := \
+ bitrate.cc \
+ rtp_rtcp_impl.cc \
+ rtcp_receiver.cc \
+ rtcp_receiver_help.cc \
+ rtcp_sender.cc \
+ rtcp_utility.cc \
+ rtp_receiver.cc \
+ rtp_sender.cc \
+ rtp_utility.cc \
+ rtp_header_extension.cc \
+ ssrc_database.cc \
+ tmmbr_help.cc \
+ dtmf_queue.cc \
+ rtp_receiver_audio.cc \
+ rtp_sender_audio.cc \
+ bandwidth_management.cc \
+ forward_error_correction.cc \
+ forward_error_correction_internal.cc \
+ overuse_detector.cc \
+ remote_rate_control.cc \
+ rtp_packet_history.cc \
+ receiver_fec.cc \
+ rtp_receiver_video.cc \
+ rtp_sender_video.cc \
+ rtp_format_vp8.cc \
+ transmission_bucket.cc \
+ vp8_partition_aggregator.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/rtp_rtcp/source/Bitrate.h b/trunk/src/modules/rtp_rtcp/source/Bitrate.h
new file mode 100644
index 0000000..ab5637b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/Bitrate.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
+#include "common_types.h" // Transport
+#include <stdio.h>
+#include <list>
+
+namespace webrtc {
+class RtpRtcpClock;
+
+class Bitrate
+{
+public:
+ Bitrate(RtpRtcpClock* clock);
+
+ // initialize members
+ void Init();
+
+ // calculate rates
+ void Process();
+
+ // update with a packet
+ void Update(const WebRtc_Word32 bytes);
+
+ // packet rate last second, updated roughly every 100 ms
+ WebRtc_UWord32 PacketRate() const;
+
+ // bitrate last second, updated roughly every 100 ms
+ WebRtc_UWord32 BitrateLast() const;
+
+ // bitrate last second, updated now
+ WebRtc_UWord32 BitrateNow() const;
+
+protected:
+ RtpRtcpClock& _clock;
+
+private:
+ WebRtc_UWord32 _packetRate;
+ WebRtc_UWord32 _bitrate;
+ WebRtc_UWord8 _bitrateNextIdx;
+ WebRtc_UWord32 _packetRateArray[10];
+ WebRtc_UWord32 _bitrateArray[10];
+ WebRtc_UWord32 _bitrateDiffMS[10];
+ WebRtc_UWord32 _timeLastRateUpdate;
+ WebRtc_UWord32 _bytesCount;
+ WebRtc_UWord32 _packetCount;
+};
+
+struct DataTimeSizeTuple
+{
+ DataTimeSizeTuple(WebRtc_UWord32 sizeBytes, WebRtc_Word64 timeCompleteMs) :
+ _sizeBytes(sizeBytes),
+ _timeCompleteMs(timeCompleteMs) {}
+
+ WebRtc_UWord32 _sizeBytes;
+ WebRtc_Word64 _timeCompleteMs;
+};
+
+class BitRateStats
+{
+public:
+ BitRateStats();
+ ~BitRateStats();
+
+ void Init();
+ void Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs);
+ WebRtc_UWord32 BitRate(WebRtc_Word64 nowMs);
+
+private:
+ void EraseOld(WebRtc_Word64 nowMs);
+
+ std::list<DataTimeSizeTuple*> _dataSamples;
+ WebRtc_UWord32 _accumulatedBytes;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc
new file mode 100644
index 0000000..05b7e2f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc
@@ -0,0 +1,580 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "bitstream_builder.h"
+
+#include <string.h>
+
+namespace webrtc {
+BitstreamBuilder::BitstreamBuilder(WebRtc_UWord8* data, const WebRtc_UWord32 dataSize) :
+ _data(data),
+ _dataSize(dataSize),
+ _byteOffset(0),
+ _bitOffset(0)
+{
+ memset(data, 0, dataSize);
+}
+
+WebRtc_UWord32
+BitstreamBuilder::Length() const
+{
+ return _byteOffset+ (_bitOffset?1:0);
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add1Bit(const WebRtc_UWord8 bit)
+{
+ // sanity
+ if(_bitOffset + 1 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bit);
+ return 0;
+}
+
+void
+BitstreamBuilder::Add1BitWithoutSanity(const WebRtc_UWord8 bit)
+{
+ if(bit & 0x1)
+ {
+ _data[_byteOffset] += (1 << (7-_bitOffset));
+ }
+
+ if(_bitOffset == 7)
+ {
+ // last bit in byte
+ _bitOffset = 0;
+ _byteOffset++;
+ } else
+ {
+ _bitOffset++;
+ }
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add2Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 2 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add3Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 3 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 2);
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add4Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 4 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 3);
+ Add1BitWithoutSanity(bits >> 2);
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add5Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 5 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 4);
+ Add1BitWithoutSanity(bits >> 3);
+ Add1BitWithoutSanity(bits >> 2);
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add6Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 6 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 5);
+ Add1BitWithoutSanity(bits >> 4);
+ Add1BitWithoutSanity(bits >> 3);
+ Add1BitWithoutSanity(bits >> 2);
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add7Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_bitOffset + 7 > 8)
+ {
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ }
+ Add1BitWithoutSanity(bits >> 6);
+ Add1BitWithoutSanity(bits >> 5);
+ Add1BitWithoutSanity(bits >> 4);
+ Add1BitWithoutSanity(bits >> 3);
+ Add1BitWithoutSanity(bits >> 2);
+ Add1BitWithoutSanity(bits >> 1);
+ Add1BitWithoutSanity(bits);
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add8Bits(const WebRtc_UWord8 bits)
+{
+ // sanity
+ if(_dataSize < Length()+1)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ if(_bitOffset == 0)
+ {
+ _data[_byteOffset] = bits;
+ } else
+ {
+ _data[_byteOffset] += (bits >> _bitOffset);
+ _data[_byteOffset+1] += (bits << (8-_bitOffset));
+ }
+ _byteOffset++;
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add16Bits(const WebRtc_UWord16 bits)
+{
+ // sanity
+ if(_dataSize < Length()+2)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ if(_bitOffset == 0)
+ {
+ _data[_byteOffset] = (WebRtc_UWord8)(bits >> 8);
+ _data[_byteOffset+1] = (WebRtc_UWord8)(bits);
+ } else
+ {
+ _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset + 8));
+ _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> _bitOffset);
+ _data[_byteOffset+2] += (WebRtc_UWord8)(bits << (8-_bitOffset));
+ }
+ _byteOffset += 2;
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add24Bits(const WebRtc_UWord32 bits)
+{
+ // sanity
+ if(_dataSize < Length()+3)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ if(_bitOffset == 0)
+ {
+ _data[_byteOffset] = (WebRtc_UWord8)(bits >> 16);
+ _data[_byteOffset+1] = (WebRtc_UWord8)(bits >> 8);
+ _data[_byteOffset+2] = (WebRtc_UWord8)(bits);
+ } else
+ {
+ _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset+16));
+ _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> (_bitOffset+8));
+ _data[_byteOffset+2] += (WebRtc_UWord8)(bits >> (_bitOffset));
+ _data[_byteOffset+3] += (WebRtc_UWord8)(bits << (8-_bitOffset));
+ }
+ _byteOffset += 3;
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::Add32Bits(const WebRtc_UWord32 bits)
+{
+ // sanity
+ if(_dataSize < Length()+4)
+ {
+ // not enough space in buffer
+ return -1;
+ }
+ if(_bitOffset == 0)
+ {
+ _data[_byteOffset] = (WebRtc_UWord8)(bits >> 24);
+ _data[_byteOffset+1] = (WebRtc_UWord8)(bits >> 16);
+ _data[_byteOffset+2] = (WebRtc_UWord8)(bits >> 8);
+ _data[_byteOffset+3] = (WebRtc_UWord8)(bits);
+ } else
+ {
+ _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset+24));
+ _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> (_bitOffset+16));
+ _data[_byteOffset+2] += (WebRtc_UWord8)(bits >> (_bitOffset+8));
+ _data[_byteOffset+3] += (WebRtc_UWord8)(bits >> (_bitOffset));
+ _data[_byteOffset+4] += (WebRtc_UWord8)(bits << (8-_bitOffset));
+ }
+ _byteOffset += 4;
+ return 0;
+}
+
+// Exp-Golomb codes
+/*
+ with "prefix" and "suffix" bits and assignment to codeNum ranges (informative)
+ Bit string form Range of codeNum
+ 1 0
+ 0 1 x0 1..2 2bits-1
+ 0 0 1 x1 x0 3..6 3bits-1
+ 0 0 0 1 x2 x1 x0 7..14 4bits-1
+ 0 0 0 0 1 x3 x2 x1 x0 15..30
+ 0 0 0 0 0 1 x4 x3 x2 x1 x0 31..62
+*/
+WebRtc_Word32
+BitstreamBuilder::AddUE(const WebRtc_UWord32 value)
+{
+ // un-rolled on 8 bit base to avoid too deep if else chain
+ if(value < 0x0000ffff)
+ {
+ if(value < 0x000000ff)
+ {
+ if(value == 0)
+ {
+ if(AddPrefix(0) != 0)
+ {
+ return -1;
+ }
+ } else if(value < 3)
+ {
+ if(AddPrefix(1) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(1, value-1);
+ } else if(value < 7)
+ {
+ if(AddPrefix(2) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(2, value-3);
+ } else if(value < 15)
+ {
+ if(AddPrefix(3) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(3, value-7);
+ } else if(value < 31)
+ {
+ if(AddPrefix(4) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(4, value-15);
+ } else if(value < 63)
+ {
+ if(AddPrefix(5) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(5, value-31);
+ } else if(value < 127)
+ {
+ if(AddPrefix(6) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(6, value-63);
+ } else
+ {
+ if(AddPrefix(7) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(7, value-127);
+ }
+ }else
+ {
+ if(value < 0x000001ff)
+ {
+ if(AddPrefix(8) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(8, value-0x000000ff);
+ } else if(value < 0x000003ff)
+ {
+ if(AddPrefix(9) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(9, value-0x000001ff);
+ } else if(value < 0x000007ff)
+ {
+ if(AddPrefix(10) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(10, value-0x000003ff);
+ } else if(value < 0x00000fff)
+ {
+ if(AddPrefix(11) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(1, value-0x000007ff);
+ } else if(value < 0x00001fff)
+ {
+ if(AddPrefix(12) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(12, value-0x00000fff);
+ } else if(value < 0x00003fff)
+ {
+ if(AddPrefix(13) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(13, value-0x00001fff);
+ } else if(value < 0x00007fff)
+ {
+ if(AddPrefix(14) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(14, value-0x00003fff);
+ } else
+ {
+ if(AddPrefix(15) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(15, value-0x00007fff);
+ }
+ }
+ }else
+ {
+ if(value < 0x00ffffff)
+ {
+ if(value < 0x0001ffff)
+ {
+ if(AddPrefix(16) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(16, value-0x0000ffff);
+ } else if(value < 0x0003ffff)
+ {
+ if(AddPrefix(17) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(17, value-0x0001ffff);
+ } else if(value < 0x0007ffff)
+ {
+ if(AddPrefix(18) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(18, value-0x0003ffff);
+ } else if(value < 0x000fffff)
+ {
+ if(AddPrefix(19) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(19, value-0x0007ffff);
+ } else if(value < 0x001fffff)
+ {
+ if(AddPrefix(20) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(20, value-0x000fffff);
+ } else if(value < 0x003fffff)
+ {
+ if(AddPrefix(21) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(21, value-0x001fffff);
+ } else if(value < 0x007fffff)
+ {
+ if(AddPrefix(22) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(22, value-0x003fffff);
+ } else
+ {
+ if(AddPrefix(23) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(23, value-0x007fffff);
+ }
+ } else
+ {
+ if(value < 0x01ffffff)
+ {
+ if(AddPrefix(24) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(24, value-0x00ffffff);
+ } else if(value < 0x03ffffff)
+ {
+ if(AddPrefix(25) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(25, value-0x01ffffff);
+ } else if(value < 0x07ffffff)
+ {
+ if(AddPrefix(26) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(26, value-0x03ffffff);
+ } else if(value < 0x0fffffff)
+ {
+ if(AddPrefix(27) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(27, value-0x07ffffff);
+ } else if(value < 0x1fffffff)
+ {
+ if(AddPrefix(28) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(28, value-0x0fffffff);
+ } else if(value < 0x3fffffff)
+ {
+ if(AddPrefix(29) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(29, value-0x1fffffff);
+ } else if(value < 0x7fffffff)
+ {
+ if(AddPrefix(30) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(30, value-0x3fffffff);
+ } else if(value < 0xffffffff)
+ {
+ if(AddPrefix(31) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(31, value-0x7ffffff);
+ } else
+ {
+ if(AddPrefix(32) != 0)
+ {
+ return -1;
+ }
+ AddSuffix(32, 0); // exactly 0xffffffff
+ }
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+BitstreamBuilder::AddPrefix(const WebRtc_UWord8 numZeros)
+{
+ // sanity for the sufix too
+ WebRtc_UWord32 numBitsToAdd = numZeros * 2 + 1;
+ if(((_dataSize - _byteOffset) *8 + 8-_bitOffset) < numBitsToAdd)
+ {
+ return -1;
+ }
+
+ // add numZeros
+ for (WebRtc_UWord32 i = 0; i < numZeros; i++)
+ {
+ Add1Bit(0);
+ }
+ Add1Bit(1);
+ return 0;
+}
+
+void
+BitstreamBuilder::AddSuffix(const WebRtc_UWord8 numBits, const WebRtc_UWord32 rest)
+{
+ // most significant bit first
+ for(WebRtc_Word32 i = numBits - 1; i >= 0; i--)
+ {
+ if(( rest >> i) & 0x1)
+ {
+ Add1Bit(1);
+ }else
+ {
+ Add1Bit(0);
+ }
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h
new file mode 100644
index 0000000..c88ef8f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class BitstreamBuilder
+{
+public:
+ BitstreamBuilder(WebRtc_UWord8* data, const WebRtc_UWord32 dataSize);
+
+ WebRtc_UWord32 Length() const;
+
+ WebRtc_Word32 Add1Bit(const WebRtc_UWord8 bit);
+ WebRtc_Word32 Add2Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add3Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add4Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add5Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add6Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add7Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add8Bits(const WebRtc_UWord8 bits);
+ WebRtc_Word32 Add16Bits(const WebRtc_UWord16 bits);
+ WebRtc_Word32 Add24Bits(const WebRtc_UWord32 bits);
+ WebRtc_Word32 Add32Bits(const WebRtc_UWord32 bits);
+
+ // Exp-Golomb codes
+ WebRtc_Word32 AddUE(const WebRtc_UWord32 value);
+
+private:
+ WebRtc_Word32 AddPrefix(const WebRtc_UWord8 numZeros);
+ void AddSuffix(const WebRtc_UWord8 numBits, const WebRtc_UWord32 rest);
+ void Add1BitWithoutSanity(const WebRtc_UWord8 bit);
+
+ WebRtc_UWord8* _data;
+ WebRtc_UWord32 _dataSize;
+
+ WebRtc_UWord32 _byteOffset;
+ WebRtc_UWord8 _bitOffset;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc
new file mode 100644
index 0000000..79ec967
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "bitstream_parser.h"
+
+namespace webrtc {
+BitstreamParser::BitstreamParser(const WebRtc_UWord8* data, const WebRtc_UWord32 dataLength) :
+ _data(data),
+ _dataLength(dataLength),
+ _byteOffset(0),
+ _bitOffset(0)
+{
+}
+ // todo should we have any error codes from this?
+
+WebRtc_UWord8
+BitstreamParser::Get1Bit()
+{
+ WebRtc_UWord8 retVal = 0x1 & (_data[_byteOffset] >> (7-_bitOffset++));
+
+ // prepare next byte
+ if(_bitOffset == 8)
+ {
+ _bitOffset = 0;
+ _byteOffset++;
+ }
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get2Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get3Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 2);
+ retVal += (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get4Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 3);
+ retVal += (Get1Bit() << 2);
+ retVal += (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get5Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 4);
+ retVal += (Get1Bit() << 3);
+ retVal += (Get1Bit() << 2);
+ retVal += (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get6Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 5);
+ retVal += (Get1Bit() << 4);
+ retVal += (Get1Bit() << 3);
+ retVal += (Get1Bit() << 2);
+ retVal += (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get7Bits()
+{
+ WebRtc_UWord8 retVal = (Get1Bit() << 6);
+ retVal += (Get1Bit() << 5);
+ retVal += (Get1Bit() << 4);
+ retVal += (Get1Bit() << 3);
+ retVal += (Get1Bit() << 2);
+ retVal += (Get1Bit() << 1);
+ retVal += Get1Bit();
+ return retVal;
+}
+
+WebRtc_UWord8
+BitstreamParser::Get8Bits()
+{
+ WebRtc_UWord16 retVal;
+
+ if(_bitOffset != 0)
+ {
+ // read 16 bits
+ retVal = (_data[_byteOffset] << 8)+ (_data[_byteOffset+1]) ;
+ retVal = retVal >> (8-_bitOffset);
+ } else
+ {
+ retVal = _data[_byteOffset];
+ }
+ _byteOffset++;
+ return (WebRtc_UWord8)retVal;
+}
+
+WebRtc_UWord16
+BitstreamParser::Get16Bits()
+{
+ WebRtc_UWord32 retVal;
+
+ if(_bitOffset != 0)
+ {
+ // read 24 bits
+ retVal = (_data[_byteOffset] << 16) + (_data[_byteOffset+1] << 8) + (_data[_byteOffset+2]);
+ retVal = retVal >> (8-_bitOffset);
+ }else
+ {
+ // read 16 bits
+ retVal = (_data[_byteOffset] << 8) + (_data[_byteOffset+1]) ;
+ }
+ _byteOffset += 2;
+ return (WebRtc_UWord16)retVal;
+}
+
+WebRtc_UWord32
+BitstreamParser::Get24Bits()
+{
+ WebRtc_UWord32 retVal;
+
+ if(_bitOffset != 0)
+ {
+ // read 32 bits
+ retVal = (_data[_byteOffset] << 24) + (_data[_byteOffset+1] << 16) + (_data[_byteOffset+2] << 8) + (_data[_byteOffset+3]);
+ retVal = retVal >> (8-_bitOffset);
+ }else
+ {
+ // read 24 bits
+ retVal = (_data[_byteOffset] << 16) + (_data[_byteOffset+1] << 8) + (_data[_byteOffset+2]) ;
+ }
+ _byteOffset += 3;
+ return retVal & 0x00ffffff; // we need to clean up the high 8 bits
+}
+
+WebRtc_UWord32
+BitstreamParser::Get32Bits()
+{
+ WebRtc_UWord32 retVal;
+
+ if(_bitOffset != 0)
+ {
+ // read 40 bits
+ WebRtc_UWord64 tempVal = _data[_byteOffset];
+ tempVal <<= 8;
+ tempVal += _data[_byteOffset+1];
+ tempVal <<= 8;
+ tempVal += _data[_byteOffset+2];
+ tempVal <<= 8;
+ tempVal += _data[_byteOffset+3];
+ tempVal <<= 8;
+ tempVal += _data[_byteOffset+4];
+ tempVal >>= (8-_bitOffset);
+
+ retVal = WebRtc_UWord32(tempVal);
+ }else
+ {
+ // read 32 bits
+ retVal = (_data[_byteOffset]<< 24) + (_data[_byteOffset+1] << 16) + (_data[_byteOffset+2] << 8) + (_data[_byteOffset+3]) ;
+ }
+ _byteOffset += 4;
+ return retVal;
+}
+
+// Exp-Golomb codes
+/*
+ with "prefix" and "suffix" bits and assignment to codeNum ranges (informative)
+ Bit string form Range of codeNum
+ 1 0
+ 0 1 x0 1..2
+ 0 0 1 x1 x0 3..6
+ 0 0 0 1 x2 x1 x0 7..14
+ 0 0 0 0 1 x3 x2 x1 x0 15..30
+ 0 0 0 0 0 1 x4 x3 x2 x1 x0 31..62
+*/
+
+WebRtc_UWord32
+BitstreamParser::GetUE()
+{
+ WebRtc_UWord32 retVal = 0;
+ WebRtc_UWord8 numLeadingZeros = 0;
+
+ while (Get1Bit() != 1)
+ {
+ numLeadingZeros++;
+ }
+ // prefix
+ retVal = (1 << numLeadingZeros) - 1;
+
+ // suffix
+ while (numLeadingZeros)
+ {
+ retVal += (Get1Bit() << --numLeadingZeros);
+ }
+ return retVal;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h
new file mode 100644
index 0000000..3d8f9ef
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class BitstreamParser
+{
+public:
+ BitstreamParser(const WebRtc_UWord8* data, const WebRtc_UWord32 dataLength);
+
+ WebRtc_UWord8 Get1Bit();
+ WebRtc_UWord8 Get2Bits();
+ WebRtc_UWord8 Get3Bits();
+ WebRtc_UWord8 Get4Bits();
+ WebRtc_UWord8 Get5Bits();
+ WebRtc_UWord8 Get6Bits();
+ WebRtc_UWord8 Get7Bits();
+ WebRtc_UWord8 Get8Bits();
+ WebRtc_UWord16 Get16Bits();
+ WebRtc_UWord32 Get24Bits();
+ WebRtc_UWord32 Get32Bits();
+
+ // Exp-Golomb codes
+ WebRtc_UWord32 GetUE();
+
+private:
+ const WebRtc_UWord8* _data;
+ const WebRtc_UWord32 _dataLength;
+
+ WebRtc_UWord32 _byteOffset;
+ WebRtc_UWord8 _bitOffset;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc
new file mode 100644
index 0000000..cf6b549
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc
@@ -0,0 +1,818 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+#include "h264_information.h"
+
+//#define DEBUG_SEI_MESSAGE 1
+
+#ifdef DEBUG_SEI_MESSAGE
+ #include "bitstream_parser.h"
+ #include <stdio.h>
+ #include <math.h>
+
+ WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
+ {
+ return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
+ }
+
+#endif
+
+namespace webrtc {
+H264Information::H264Information(const bool SVC)
+ : _SVC(SVC)
+
+{
+}
+
+H264Information::~H264Information()
+{
+
+}
+
+void
+H264Information::Reset()
+{
+ _parsedLength = 0;
+ _remLength = 0;
+ _length = 0;
+ _info.numNALUs = 0;
+ _info.numLayers = 0;
+
+ memset(_info.startCodeSize, 0, sizeof(_info.startCodeSize));
+ memset(_info.payloadSize, 0, sizeof(_info.payloadSize));
+ memset(_info.NRI, 0, sizeof(_info.NRI));
+ memset(_info.type, 0, sizeof(_info.type));
+ memset(_info.accLayerSize, 0, sizeof(_info.accLayerSize));
+
+ for (WebRtc_Word32 i = 0; i < KMaxNumberOfNALUs; i++)
+ {
+ _info.SVCheader[i].idr = 0;
+ _info.SVCheader[i].priorityID = 0;
+ _info.SVCheader[i].interLayerPred = 0;
+ _info.SVCheader[i].dependencyID = 0;
+ _info.SVCheader[i].qualityID = 0;
+ _info.SVCheader[i].temporalID = 0;
+ _info.SVCheader[i].useRefBasePic = 0;
+ _info.SVCheader[i].discardable = 0;
+ _info.SVCheader[i].output = 0;
+
+ _info.PACSI[i].X = 0;
+ _info.PACSI[i].Y = 0;
+// _info.PACSI[i].T = 0;
+ _info.PACSI[i].A = 0;
+ _info.PACSI[i].P = 0;
+ _info.PACSI[i].C = 0;
+ _info.PACSI[i].S = 0;
+ _info.PACSI[i].E = 0;
+ _info.PACSI[i].TL0picIDx = 0;
+ _info.PACSI[i].IDRpicID = 0;
+ _info.PACSI[i].DONC = 0;
+ _info.PACSI[i].numSEINALUs = 0;
+ _info.PACSI[i].NALlength = 5;
+ }
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 GetInfo(const WebRtc_UWord8* ptrEncodedBuffer,
+ * const WebRtc_UWord32 length,
+ * const H264Info*& ptrInfo);
+ *
+ * Gets information from an encoded stream.
+ *
+ * Input:
+ * - ptrEncodedBuffer : Pointer to encoded stream.
+ * - length : Length in bytes of encoded stream.
+ *
+ * Output:
+ * - ptrInfo : Pointer to struct with H.264 info.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::GetInfo(const WebRtc_UWord8* ptrEncodedBuffer,
+ const WebRtc_UWord32 length,
+ const H264Info*& ptrInfo)
+{
+ if (!ptrEncodedBuffer || length < 4)
+ {
+ return -1;
+ }
+
+ if (!HasInfo(length))
+ {
+ if (-1 == FindInfo(ptrEncodedBuffer, length))
+ {
+ Reset();
+ return -1;
+ }
+ }
+ ptrInfo = &_info;
+ return 0;
+}
+
+RtpVideoCodecTypes
+H264Information::Type()
+{
+ if(_SVC)
+ {
+ return RTP_H264_SVCVideo;
+ }
+ return RTP_H264Video;
+}
+
+
+/*******************************************************************************
+ * bool HasInfo(const WebRtc_UWord32 length);
+ *
+ * Checks if information has already been stored for this encoded stream.
+ *
+ * Input:
+ * - length : Length in bytes of encoded stream.
+ *
+ * Return value:
+ * - true (false) : Information has (not) been stored.
+ */
+
+bool
+H264Information::HasInfo(const WebRtc_UWord32 length)
+{
+ if (!_info.numNALUs)
+ {
+ return false;
+ }
+
+ // has info, make sure current length matches info length
+ if (length != _length)
+ {
+ Reset();
+ return false;
+ }
+
+ return true;
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 FindInfo(const WebRtc_UWord8* ptrEncodedBuffer,
+ * const WebRtc_UWord32 length);
+ *
+ * Parses the encoded stream.
+ *
+ * Input:
+ * - ptrEncodedBuffer : Pointer to encoded stream.
+ * - length : Length in bytes of encoded stream.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::FindInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length)
+{
+ _ptrData = ptrEncodedBuffer;
+ _length = length;
+ _parsedLength = 0;
+ _remLength = length;
+
+ do
+ {
+ // Get start code length
+ if (FindNALUStartCodeSize() == -1)
+ {
+ Reset();
+ return -1;
+ }
+
+ // Get NAL unit payload size
+ WebRtc_Word32 foundLast = FindNALU();
+ if (foundLast == -1)
+ {
+ Reset();
+ return -1;
+ }
+
+ // Validate parsed length
+ if (_parsedLength > _length)
+ {
+ Reset();
+ return -1;
+ }
+
+ // Get NRI
+ GetNRI();
+
+ // Get type
+ if (FindNALUType() == -1)
+ {
+ Reset();
+ return -1;
+ }
+
+ // Set layer start end bit
+ SetLayerSEBit(foundLast);
+
+
+ // Last NAL unit found?
+ if (foundLast == 1)
+ {
+ if (_parsedLength != _length)
+ {
+ Reset();
+ return -1;
+ }
+ _info.numNALUs++;
+ return SetLayerLengths();
+ }
+
+ // Next NAL unit
+ _ptrData += (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]);
+ _remLength -= (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]);
+ _info.numNALUs++;
+
+ // Validate memory allocation
+ if (_info.numNALUs >= KMaxNumberOfNALUs)
+ {
+ Reset();
+ return -1;
+ }
+ }
+ while(true);
+
+ return 0;
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 FindNALUStartCodeSize();
+ *
+ * Finds the start code length of the current NAL unit.
+ *
+ * Output:
+ * - _info.startCodeSize[currentNALU] : Start code length in bytes of NAL unit.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::FindNALUStartCodeSize()
+{
+ // NAL unit start code. Ex. {0,0,1} or {0,0,0,1}
+ for (WebRtc_UWord32 i = 2; i < _remLength; i++)
+ {
+ if (_ptrData[i] == 1 && _ptrData[i - 1] == 0 && _ptrData[i - 2] == 0)
+ {
+ _info.startCodeSize[_info.numNALUs] = WebRtc_UWord8(i + 1);
+ return 0;
+ }
+ }
+ return -1;
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 FindNALU();
+ *
+ * Finds the length of the current NAL unit.
+ *
+ * Output:
+ * - _info.payloadSize[currentNALU] : Payload length in bytes of NAL unit
+ * (start code length not included).
+ * - _parsedLength : Current parsed length in bytes.
+ *
+ * Return value:
+ * - 1 : ok. Last NAL unit found.
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::FindNALU()
+{
+ for (WebRtc_UWord32 i = _info.startCodeSize[_info.numNALUs]; i < _remLength - 2; i += 2)
+ {
+ if (_ptrData[i] == 0)
+ {
+ WebRtc_Word32 size = 0;
+ if ((_ptrData[i + 1] == 1 && _ptrData[i - 1] == 0) ||
+ (_ptrData[i + 2] == 1 && _ptrData[i + 1] == 0))
+ {
+ // Found a header
+ // Reduce size by preceding zeroes
+ while (_ptrData[i - 1] == 0)
+ {
+ i--;
+ }
+ size = i;
+ }
+ if (size > 0)
+ {
+ _info.payloadSize[_info.numNALUs] = size - _info.startCodeSize[_info.numNALUs];
+ _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs];
+ return 0;
+ }
+ }
+ }
+ // Last NAL unit
+ _info.payloadSize[_info.numNALUs] = _remLength - _info.startCodeSize[_info.numNALUs];
+ if (_info.payloadSize[_info.numNALUs] > 0)
+ {
+ _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs];
+ return 1;
+ }
+ return -1;
+}
+
+/*******************************************************************************
+ * void GetNRI();
+ *
+ * Finds the NRI of the current NAL unit.
+ *
+ * Output:
+ * - _info.NRI[currentNALU] : NRI of NAL unit.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+void
+H264Information::GetNRI()
+{
+ // NAL unit header (1 byte)
+ // ---------------------------------
+ // | start code |F|NRI| Type |
+ // ---------------------------------
+
+ // NRI (2 bits) - nal_ref_idc. '00' - the NAL unit is not used to reconstruct reference pictures.
+ // >00 - the NAL unit is required to reconstruct reference pictures
+ // in the same layer, or contains a parameter set.
+
+
+ const WebRtc_UWord8 type = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x1f;
+
+ // NALU type of 5, 7 and 8 shoud have NRI to b011
+ if( type == 5 ||
+ type == 7 ||
+ type == 8)
+ {
+ _info.NRI[_info.numNALUs] = 0x60;
+ }else
+ {
+ _info.NRI[_info.numNALUs] = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x60;
+ }
+}
+
+
+/*******************************************************************************
+ * WebRtc_Word32 FindNALUType();
+ *
+ * Finds the type of the current NAL unit.
+ *
+ * Output:
+ * - _info.type[currentNALU] : Type of NAL unit
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::FindNALUType()
+{
+ // NAL unit header (1 byte)
+ // ---------------------------------
+ // | start code |F|NRI| Type |
+ // ---------------------------------
+
+ _info.type[_info.numNALUs] = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x1f;
+
+ if (_info.type[_info.numNALUs] == 0)
+ {
+ return -1;
+ }
+
+ // SVC NAL units, extended header
+ if (ParseSVCNALUHeader() == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 ParseSVCNALUHeader();
+ *
+ * Finds the extended header of the current NAL unit. Included for NAL unit types 14 and 20.
+ *
+ * Output:
+ * - _info.SVCheader[currentNALU] : SVC header of NAL unit.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ */
+WebRtc_Word32
+H264Information::ParseSVCNALUHeader()
+{
+ if (_info.type[_info.numNALUs] == 5)
+ {
+ _info.SVCheader[_info.numNALUs].idr = 1;
+ }
+ if (_info.type[_info.numNALUs] == 6)
+ {
+ WebRtc_UWord32 seiPayloadSize;
+ do
+ {
+ // SEI message
+ seiPayloadSize = 0;
+
+ WebRtc_UWord32 curByte = _info.startCodeSize[_info.numNALUs] + 1;
+ const WebRtc_UWord32 seiStartOffset = curByte;
+
+ WebRtc_UWord32 seiPayloadType = 0;
+ while(_ptrData[curByte] == 0xff)
+ {
+ seiPayloadType += 255;
+ curByte++;
+ }
+ seiPayloadType += _ptrData[curByte++];
+
+ while(_ptrData[curByte] == 0xff)
+ {
+ seiPayloadSize += 255;
+ curByte++;
+ }
+ seiPayloadSize += _ptrData[curByte++];
+
+ if(_info.payloadSize[_info.numNALUs] < _info.startCodeSize[_info.numNALUs] + seiPayloadSize)
+ {
+ // sanity of remaining buffer
+ // return 0 since no one "need" SEI messages
+ assert(false);
+ return 0;
+ }
+
+ if(seiPayloadType == 24)
+ {
+ // we add this to NALU 0 to be signaled in the first PACSI packet
+ _info.PACSI[0].numSEINALUs = 1; // we allways add this to NALU 0 to send it in the first packet
+ if(_info.PACSI[0].seiMessageLength[0] != seiPayloadSize)
+ {
+ _info.PACSI[0].seiMessageLength[0] = seiPayloadSize;
+ delete [] _info.PACSI[0].seiMessageData[0];
+ _info.PACSI[0].seiMessageData[0] = new WebRtc_UWord8[seiPayloadSize];
+ }
+ memcpy(_info.PACSI[0].seiMessageData[0], _ptrData+seiStartOffset, seiPayloadSize);
+
+ _info.PACSI[0].NALlength += seiPayloadSize + 2; // additional 2 is the length
+
+#ifdef DEBUG_SEI_MESSAGE
+ const WebRtc_UWord8 numberOfLayers = 10;
+ WebRtc_UWord16 avgBitrate[numberOfLayers]= {0,0,0,0,0,0,0,0,0,0};
+ WebRtc_UWord16 maxBitrateLayer[numberOfLayers]= {0,0,0,0,0,0,0,0,0,0};
+ WebRtc_UWord16 maxBitrateLayerRepresentation[numberOfLayers] = {0,0,0,0,0,0,0,0,0,0};
+ WebRtc_UWord16 maxBitrareCalcWindow[numberOfLayers] = {0,0,0,0,0,0,0,0,0,0};
+
+ BitstreamParser parserScalabilityInfo(_ptrData+curByte, seiPayloadSize);
+
+ parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+ const WebRtc_UWord8 priority_layer_info_present = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 priority_id_setting_flag = parserScalabilityInfo.Get1Bit();
+
+ WebRtc_UWord32 numberOfLayersMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 j = 0; j<= numberOfLayersMinusOne; j++)
+ {
+ printf("\nLayer ID:%d \n",parserScalabilityInfo.GetUE());
+ printf("Priority ID:%d \n", parserScalabilityInfo.Get6Bits());
+ printf("Discardable:%d \n", parserScalabilityInfo.Get1Bit());
+
+ printf("Dependency ID:%d \n", parserScalabilityInfo.Get3Bits());
+ printf("Quality ID:%d \n", parserScalabilityInfo.Get4Bits());
+ printf("Temporal ID:%d \n", parserScalabilityInfo.Get3Bits());
+
+ const WebRtc_UWord8 sub_pic_layer_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 sub_region_layer_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 iroi_division_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 profile_level_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 bitrate_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 frm_rate_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 frm_size_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 layer_dependency_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 parameter_sets_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 bitstream_restriction_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 exact_inter_layer_pred_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+
+ if(sub_pic_layer_flag || iroi_division_info_present_flag)
+ {
+ parserScalabilityInfo.Get1Bit();
+ }
+ const WebRtc_UWord8 layer_conversion_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 layer_output_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+
+ if(profile_level_info_present_flag)
+ {
+ parserScalabilityInfo.Get24Bits();
+ }
+ if(bitrate_info_present_flag)
+ {
+ // this is what we want
+ avgBitrate[j] = parserScalabilityInfo.Get16Bits();
+ maxBitrateLayer[j] = parserScalabilityInfo.Get16Bits();
+ maxBitrateLayerRepresentation[j] = parserScalabilityInfo.Get16Bits();
+ maxBitrareCalcWindow[j] = parserScalabilityInfo.Get16Bits();
+
+ printf("\tAvg:%d\n", BitRateBPS(avgBitrate[j]));
+ printf("\tmaxBitrate:%d\n", BitRateBPS(maxBitrateLayer[j]));
+ printf("\tmaxBitrate rep:%d\n", BitRateBPS(maxBitrateLayerRepresentation[j]));
+ printf("\tCalcWindow:%d\n", maxBitrareCalcWindow[j]);
+ }
+ if(frm_rate_info_present_flag)
+ {
+ printf("\tFrame rate constant:%d\n", parserScalabilityInfo.Get2Bits()); // 0 = not constant, 1 = constant, 2 = maybe...
+ printf("\tFrame rate avg:%d\n", parserScalabilityInfo.Get16Bits()/256);
+ }
+ if(frm_size_info_present_flag || iroi_division_info_present_flag)
+ {
+ printf("\tFrame Width:%d\n",(parserScalabilityInfo.GetUE()+1)*16);
+ printf("\tFrame Height:%d\n",(parserScalabilityInfo.GetUE()+1)*16);
+ }
+ if(sub_region_layer_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ if(sub_pic_layer_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(iroi_division_info_present_flag)
+ {
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }else
+ {
+ const WebRtc_UWord32 numRoisMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= numRoisMinusOne; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }
+ }
+ }
+ if(layer_dependency_info_present_flag)
+ {
+ const WebRtc_UWord32 numDirectlyDependentLayers = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k < numDirectlyDependentLayers; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ } else
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(parameter_sets_info_present_flag)
+ {
+ const WebRtc_UWord32 numSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= numSeqParameterSetMinusOne; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ const WebRtc_UWord32 numSubsetSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 l = 0; l <= numSubsetSeqParameterSetMinusOne; l++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ const WebRtc_UWord32 numPicParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 m = 0; m <= numPicParameterSetMinusOne; m++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ }else
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(bitstream_restriction_info_present_flag)
+ {
+ parserScalabilityInfo.Get1Bit();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }
+ if(layer_conversion_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <2;k++)
+ {
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.Get24Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ }
+ }
+ if(priority_layer_info_present)
+ {
+ const WebRtc_UWord32 prNumDidMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= prNumDidMinusOne;k++)
+ {
+ parserScalabilityInfo.Get3Bits();
+ const WebRtc_UWord32 prNumMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 l = 0; l <= prNumMinusOne; l++)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.Get24Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ }
+ if(priority_id_setting_flag)
+ {
+ WebRtc_UWord8 priorityIdSettingUri;
+ WebRtc_UWord32 priorityIdSettingUriIdx = 0;
+ do
+ {
+ priorityIdSettingUri = parserScalabilityInfo.Get8Bits();
+ } while (priorityIdSettingUri != 0);
+ }
+#endif
+ } else
+ {
+ // not seiPayloadType 24 ignore
+ }
+ //check if we have more SEI in NALU
+ } while (_info.payloadSize[_info.numNALUs] > _info.startCodeSize[_info.numNALUs] + seiPayloadSize);
+ }
+
+ // Extended NAL unit header (3 bytes).
+ // +---------------+---------------+---------------+
+ // |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |R|I| PRID |N| DID | QID | TID |U|D|O| RR|
+ // +---------------+---------------+---------------+
+
+ // R - Reserved for future extensions (MUST be 1). Receivers SHOULD ignore the value of R.
+ // I - Is layer representation an IDR layer (1) or not (0).
+ // PRID - Priority identifier for the NAL unit.
+ // N - Specifies whether inter-layer prediction may be used for decoding the coded slice (1) or not (0).
+ // DID - Indicates the inter-layer coding dependency level of a layer representation.
+ // QID - Indicates the quality level of an MGS layer representation.
+ // TID - Indicates the temporal level of a layer representation.
+ // U - Use only reference base pictures during the inter prediction process (1) or not (0).
+ // D - Discardable flag.
+ // O - Output_flag. Affects the decoded picture output process as defined in Annex C of [H.264].
+ // RR - Reserved_three_2bits (MUST be '11'). Receivers SHOULD ignore the value of RR.
+
+ if (_info.type[_info.numNALUs] == 14 ||
+ _info.type[_info.numNALUs] == 20)
+ {
+ WebRtc_UWord32 curByte = _info.startCodeSize[_info.numNALUs] + 1;
+
+ if (_remLength < curByte + 3)
+ {
+ return -1;
+ }
+
+ _info.SVCheader[_info.numNALUs].idr = (_ptrData[curByte] >> 6) & 0x01;
+ _info.SVCheader[_info.numNALUs].priorityID = (_ptrData[curByte++] & 0x3F);
+
+ _info.SVCheader[_info.numNALUs].interLayerPred = (_ptrData[curByte] >> 7) & 0x01;
+ _info.SVCheader[_info.numNALUs].dependencyID = (_ptrData[curByte] >> 4) & 0x07;
+ _info.SVCheader[_info.numNALUs].qualityID = (_ptrData[curByte++] & 0x0F);
+
+ _info.SVCheader[_info.numNALUs].temporalID = (_ptrData[curByte] >> 5) & 0x07;
+ _info.SVCheader[_info.numNALUs].useRefBasePic = (_ptrData[curByte] >> 4) & 0x01;
+ _info.SVCheader[_info.numNALUs].discardable = (_ptrData[curByte] >> 3) & 0x01;
+ _info.SVCheader[_info.numNALUs].output = (_ptrData[curByte] >> 2) & 0x01;
+
+ if (_info.type[_info.numNALUs] == 14)
+ {
+ // inform the next NALU
+ memcpy(&(_info.SVCheader[_info.numNALUs+1]), &(_info.SVCheader[_info.numNALUs]), sizeof(_H264_SVC_NALUHeader));
+ }
+ }
+ return 0;
+}
+
+
+/*******************************************************************************
+ * void SetLayerSEBit();
+ *
+ * Sets start and end bits for the current NAL unit.
+ *
+ * Output:
+ * - _info.PACSI[currentNALU].S : First NAL unit in a layer (S = 1).
+ * - _info.PACSI[currentNALU].E : Last NAL unit in a layer (E = 1).
+ *
+ */
+void
+H264Information::SetLayerSEBit(WebRtc_Word32 foundLast)
+{
+ if (_info.numNALUs == 0)
+ {
+ // First NAL unit
+ _info.PACSI[_info.numNALUs].S = 1;
+ }
+
+ if (_info.numNALUs > 0)
+ {
+ if (_info.type[_info.numNALUs] != _info.type[_info.numNALUs-1] &&
+ (_info.type[_info.numNALUs] == 20))
+ {
+ // First layer in scalable extension
+ _info.PACSI[_info.numNALUs].S = 1;
+ _info.PACSI[_info.numNALUs-1].E = 1;
+ }
+
+ if (_info.type[_info.numNALUs] == 20 && _info.type[_info.numNALUs-1] == 20)
+ {
+ if (_info.SVCheader[_info.numNALUs].temporalID != _info.SVCheader[_info.numNALUs-1].temporalID ||
+ _info.SVCheader[_info.numNALUs].dependencyID != _info.SVCheader[_info.numNALUs-1].dependencyID ||
+ _info.SVCheader[_info.numNALUs].qualityID != _info.SVCheader[_info.numNALUs-1].qualityID)
+ {
+ // New layer in scalable extension
+ _info.PACSI[_info.numNALUs].S = 1;
+ _info.PACSI[_info.numNALUs-1].E = 1;
+ }
+ }
+ }
+
+ if (foundLast)
+ {
+ // Last NAL unit
+ _info.PACSI[_info.numNALUs].E = 1;
+ }
+
+}
+
+/*******************************************************************************
+ * WebRtc_Word32 SetLayerLengths();
+ *
+ * Sets the accumulated layer length.
+ *
+ * Output:
+ * - _info.accLayerSize[currentLayer] : Size in bytes of layer: 0 - currentLayer.
+ *
+ * Return value:
+ * - 0 : ok
+ * - (-1) : Error
+ *
+ */
+WebRtc_Word32
+H264Information::SetLayerLengths()
+{
+ for (WebRtc_UWord32 curNALU = 0; curNALU < _info.numNALUs; curNALU++)
+ {
+ _info.accLayerSize[_info.numLayers] += _info.startCodeSize[curNALU] + _info.payloadSize[curNALU];
+
+ if (_info.PACSI[curNALU].E == 1)
+ {
+ _info.numLayers++;
+ if (curNALU == WebRtc_UWord32(_info.numNALUs - 1))
+ {
+ break;
+ }
+ if (_info.numLayers >= KMaxNumberOfLayers)
+ {
+ Reset();
+ return -1;
+ }
+ _info.accLayerSize[_info.numLayers] += _info.accLayerSize[_info.numLayers - 1];
+ }
+ }
+
+ if (_info.numLayers < 1 && _info.numLayers > KMaxNumberOfLayers)
+ {
+ Reset();
+ return -1;
+ }
+
+ if (_info.accLayerSize[_info.numLayers - 1] != WebRtc_Word32(_length))
+ {
+ Reset();
+ return -1;
+ }
+
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h
new file mode 100644
index 0000000..c7f5214
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h
@@ -0,0 +1,170 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
+
+#include "VideoCodecInformation.h"
+#include "typedefs.h"
+
+namespace webrtc {
+enum
+{
+ KMaxNumberOfNALUs = 128,
+ KMaxNumberOfSEINALUs = 2,
+ KMaxNumberOfLayers = 16
+};
+
+struct H264_SVC_NALUHeader
+{
+ H264_SVC_NALUHeader()
+ :
+ r(1),
+ idr(0),
+ priorityID(0),
+ interLayerPred(0),
+ dependencyID(0),
+ qualityID(0),
+ temporalID(0),
+ useRefBasePic(0),
+ discardable(0),
+ output(0),
+ rr(3),
+ length(3)
+ {
+ }
+ const WebRtc_UWord8 r;
+ WebRtc_UWord8 idr;
+ WebRtc_UWord8 priorityID;
+ WebRtc_UWord8 interLayerPred;
+ WebRtc_UWord8 dependencyID;
+ WebRtc_UWord8 qualityID;
+ WebRtc_UWord8 temporalID;
+ WebRtc_UWord8 useRefBasePic;
+ WebRtc_UWord8 discardable;
+ WebRtc_UWord8 output;
+ const WebRtc_UWord8 rr;
+ const WebRtc_UWord8 length;
+};
+
+class H264_PACSI_NALU
+{
+public:
+ H264_PACSI_NALU() :
+ NALlength(5),
+ type(30),
+ X(0),
+ Y(0),
+// T(0),
+ A(0),
+ P(0),
+ C(0),
+ S(0),
+ E(0),
+ TL0picIDx(0),
+ IDRpicID(0),
+ DONC(0),
+ numSEINALUs(0)
+ {
+ memset(seiMessageLength, 0, sizeof(seiMessageLength));
+ memset(seiMessageData, 0, sizeof(seiMessageData));
+ }
+ ~H264_PACSI_NALU()
+ {
+ for(int i = 0; i<KMaxNumberOfSEINALUs; i++)
+ {
+ if(seiMessageData[i])
+ {
+ delete [] seiMessageData[i];
+ }
+ }
+ }
+
+ WebRtc_UWord32 NALlength;
+ const WebRtc_UWord8 type;
+ WebRtc_UWord8 X;
+ WebRtc_UWord8 Y;
+// WebRtc_UWord8 T;
+ WebRtc_UWord8 A;
+ WebRtc_UWord8 P;
+ WebRtc_UWord8 C;
+ WebRtc_UWord8 S;
+ WebRtc_UWord8 E;
+ WebRtc_UWord8 TL0picIDx;
+ WebRtc_UWord16 IDRpicID;
+ WebRtc_UWord16 DONC;
+ WebRtc_UWord32 numSEINALUs;
+ WebRtc_UWord32 seiMessageLength[KMaxNumberOfSEINALUs]; // we allow KMaxNumberOfSEINALUs SEI messages
+ WebRtc_UWord8* seiMessageData[KMaxNumberOfSEINALUs];
+};
+
+struct H264Info
+{
+ H264Info()
+ :
+ numNALUs(0),
+ numLayers(0)
+ {
+ memset(startCodeSize, 0, sizeof(startCodeSize));
+ memset(payloadSize, 0, sizeof(payloadSize));
+ memset(NRI, 0, sizeof(NRI));
+ memset(type, 0, sizeof(type));
+ memset(accLayerSize, 0, sizeof(accLayerSize));
+ }
+ WebRtc_UWord16 numNALUs;
+ WebRtc_UWord8 numLayers;
+ WebRtc_UWord8 startCodeSize[KMaxNumberOfNALUs];
+ WebRtc_UWord32 payloadSize[KMaxNumberOfNALUs];
+ WebRtc_UWord8 NRI[KMaxNumberOfNALUs];
+ WebRtc_UWord8 type[KMaxNumberOfNALUs];
+ H264_SVC_NALUHeader SVCheader[KMaxNumberOfNALUs];
+ H264_PACSI_NALU PACSI[KMaxNumberOfNALUs];
+ WebRtc_Word32 accLayerSize[KMaxNumberOfLayers];
+};
+
+
+class H264Information : public VideoCodecInformation
+{
+public:
+ H264Information(const bool SVC);
+ ~H264Information();
+
+ virtual void Reset();
+
+ virtual RtpVideoCodecTypes Type();
+
+ virtual WebRtc_Word32 GetInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length, const H264Info*& ptrInfo);
+
+
+protected:
+ bool HasInfo(const WebRtc_UWord32 length);
+ WebRtc_Word32 FindInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length);
+
+ void GetNRI();
+ WebRtc_Word32 FindNALU();
+ WebRtc_Word32 FindNALUStartCodeSize();
+ WebRtc_Word32 FindNALUType();
+
+ WebRtc_Word32 ParseSVCNALUHeader();
+
+ void SetLayerSEBit(WebRtc_Word32 foundLast);
+ WebRtc_Word32 SetLayerLengths();
+
+private:
+ const bool _SVC;
+ const WebRtc_UWord8* _ptrData;
+ WebRtc_UWord32 _length;
+ WebRtc_UWord32 _parsedLength;
+ WebRtc_UWord32 _remLength;
+ H264Info _info;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc
new file mode 100644
index 0000000..1f35526
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc
@@ -0,0 +1,1280 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_sender_h264.h"
+
+#include "rtp_utility.h"
+
+namespace webrtc {
+RTPSenderH264::RTPSenderH264(RTPSenderInterface* rtpSender) :
+ // H264
+ _rtpSender(*rtpSender),
+ _h264Mode(H264_SINGLE_NAL_MODE),
+ _h264SendPPS_SPS(true),
+ _h264SVCPayloadType(-1),
+ _h264SVCRelaySequenceNumber(0),
+ _h264SVCRelayTimeStamp(0),
+ _h264SVCRelayLayerComplete(false),
+
+ _useHighestSendLayer(false),
+ _highestDependencyLayerOld(MAX_NUMBER_OF_TEMPORAL_ID-1),
+ _highestDependencyQualityIDOld(MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID-1),
+ _highestDependencyLayer(0),
+ _highestDependencyQualityID(0),
+ _highestTemporalLayer(0)
+{
+}
+
+RTPSenderH264::~RTPSenderH264()
+{
+}
+
+WebRtc_Word32
+RTPSenderH264::Init()
+{
+ _h264SendPPS_SPS = true;
+ _h264Mode = H264_SINGLE_NAL_MODE;
+ return 0;
+}
+
+/*
+ multi-session
+ 3 modes supported
+ NI-T timestamps
+ NI-TC timestamps/CS-DON
+ NI-C CS-DON
+
+ Non-interleaved timestamp based mode (NI-T)
+ Non-interleaved cross-session decoding order number (CS-DON) based mode (NI-C)
+ Non-interleaved combined timestamp and CS-DON mode (NI-TC)
+
+ NOT supported Interleaved CS-DON (I-C) mode.
+
+ NI-T and NI-TC modes both use timestamps to recover the decoding
+ order. In order to be able to do so, it is necessary for the RTP
+ packet stream to contain data for all sampling instances of a given
+ RTP session in all enhancement RTP sessions that depend on the given
+ RTP session. The NI-C and I-C modes do not have this limitation,
+ and use the CS-DON values as a means to explicitly indicate decoding
+ order, either directly coded in PACSI NAL units, or inferred from
+ them using the packetization rules. It is noted that the NI-TC mode
+ offers both alternatives and it is up to the receiver to select
+ which one to use.
+*/
+
+bool
+RTPSenderH264::AddH264SVCNALUHeader(const H264_SVC_NALUHeader& svc,
+ WebRtc_UWord8* databuffer,
+ WebRtc_Word32& curByte) const
+{
+ // +---------------+---------------+---------------+
+ // |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |R|I| PRID |N| DID | QID | TID |U|D|O| RR|
+ // +---------------+---------------+---------------+
+
+ // R - Reserved for future extensions (MUST be 1). Receivers SHOULD ignore the value of R.
+ // I - Is layer representation an IDR layer (1) or not (0).
+ // PRID - Priority identifier for the NAL unit.
+ // N - Specifies whether inter-layer prediction may be used for decoding the coded slice (1) or not (0).
+ // DID - Indicates the WebRtc_Word32er-layer coding dependency level of a layer representation.
+ // QID - Indicates the quality level of an MGS layer representation.
+ // TID - Indicates the temporal level of a layer representation.
+ // U - Use only reference base pictures during the WebRtc_Word32er prediction process (1) or not (0).
+ // D - Discardable flag.
+ // O - Output_flag. Affects the decoded picture output process as defined in Annex C of [H.264].
+ // RR - Reserved_three_2bits (MUST be '11'). Receivers SHOULD ignore the value of RR.
+
+ // Add header data
+ databuffer[curByte++] = (svc.r << 7) + (svc.idr << 6) + (svc.priorityID & 0x3F);
+ databuffer[curByte++] = (svc.interLayerPred << 7) + (svc.dependencyID << 4) + (svc.qualityID & 0x0F);
+ databuffer[curByte++] = (svc.temporalID << 5) + (svc.useRefBasePic << 4) + (svc.discardable << 3) +
+ (svc.output << 2) + (svc.rr & 0x03);
+ return true;
+}
+
+WebRtc_Word32
+RTPSenderH264::AddH264PACSINALU(const bool firstPacketInNALU,
+ const bool lastPacketInNALU,
+ const H264_PACSI_NALU& pacsi,
+ const H264_SVC_NALUHeader& svc,
+ const WebRtc_UWord16 DONC,
+ WebRtc_UWord8* databuffer,
+ WebRtc_Word32& curByte) const
+{
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |F|NRI|Type(30) | SVC NAL unit header |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |X|Y|T|A|P|C|S|E| TL0PICIDX (o.)| IDRPICID (o.) |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | DONC (o.) | NAL unit size 1 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | |
+ // | SEI NAL unit 1 |
+ // | |
+ // | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | | NAL unit size 2 | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+ // | |
+ // | SEI NAL unit 2 |
+ // | +-+-+-+-+-+-+-+-+-+-+
+ // | |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+
+ // If present, MUST be first NAL unit in aggregation packet + there MUST be at least
+ // one additional unit in the same packet! The RTPHeader and payload header are set as if the 2nd NAL unit
+ // (first non-PACSI NAL unit) is encapsulated in the same packet.
+ // contains scalability info common for all remaining NAL units.
+
+ // todo add API to configure this required for multisession
+ const bool addDONC = false;
+
+ if (svc.length == 0 || pacsi.NALlength == 0)
+ {
+ return 0;
+ }
+
+ WebRtc_Word32 startByte = curByte;
+
+ // NAL unit header
+ databuffer[curByte++] = 30; // NRI will be added later
+
+ // Extended SVC header
+ AddH264SVCNALUHeader(svc, databuffer, curByte);
+
+ // Flags
+ databuffer[curByte++] = (pacsi.X << 7) +
+ (pacsi.Y << 6) +
+ (addDONC << 5) +
+ (pacsi.A << 4) +
+ (pacsi.P << 3) +
+ (pacsi.C << 2) +
+ firstPacketInNALU?(pacsi.S << 1):0 +
+ lastPacketInNALU?(pacsi.E):0;
+
+ // Optional fields
+ if (pacsi.Y)
+ {
+ databuffer[curByte++] = pacsi.TL0picIDx;
+ databuffer[curByte++] = (WebRtc_UWord8)(pacsi.IDRpicID >> 8);
+ databuffer[curByte++] = (WebRtc_UWord8)(pacsi.IDRpicID);
+ }
+ // Decoding order number
+ if (addDONC) // pacsi.T
+ {
+ databuffer[curByte++] = (WebRtc_UWord8)(DONC >> 8);
+ databuffer[curByte++] = (WebRtc_UWord8)(DONC);
+ }
+
+ // SEI NALU
+ if(firstPacketInNALU) // IMPROVEMENT duplicate it to make sure it arrives...
+ {
+ // we only set this for NALU 0 to make sure we send it only once per frame
+ for (WebRtc_UWord32 i = 0; i < pacsi.numSEINALUs; i++)
+ {
+ // NALU size
+ databuffer[curByte++] = (WebRtc_UWord8)(pacsi.seiMessageLength[i] >> 8);
+ databuffer[curByte++] = (WebRtc_UWord8)(pacsi.seiMessageLength[i]);
+
+ // NALU data
+ memcpy(databuffer + curByte, pacsi.seiMessageData[i], pacsi.seiMessageLength[i]);
+ curByte += pacsi.seiMessageLength[i];
+ }
+ }
+ return curByte - startByte;
+}
+
+WebRtc_Word32
+RTPSenderH264::SetH264RelaySequenceNumber(const WebRtc_UWord16 seqNum)
+{
+ _h264SVCRelaySequenceNumber = seqNum;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SetH264RelayCompleteLayer(const bool complete)
+{
+ _h264SVCRelayLayerComplete = complete;
+ return 0;
+}
+
+/*
+ 12 Filler data
+
+ The only restriction of filler data NAL units within an
+ access unit is that they shall not precede the first VCL
+ NAL unit with the same access unit.
+*/
+WebRtc_Word32
+RTPSenderH264::SendH264FillerData(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord16 bytesToSend,
+ const WebRtc_UWord32 ssrc)
+{
+ WebRtc_UWord16 fillerLength = bytesToSend - 12 - 1;
+
+ if (fillerLength > WEBRTC_IP_PACKET_SIZE - 12 - 1)
+ {
+ return 0;
+ }
+
+ if (fillerLength == 0)
+ {
+ // do not send an empty packet, will not reach JB
+ fillerLength = 1;
+ }
+
+ // send codec valid data, H.264 has defined data which is binary 1111111
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
+ dataBuffer[1] = rtpHeader->header.payloadType;
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp);
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
+
+ // set filler NALU type
+ dataBuffer[12] = 12; // NRI field = 0, type 12
+
+ // fill with 0xff
+ memset(dataBuffer + 12 + 1, 0xff, fillerLength);
+
+ return _rtpSender.SendToNetwork(dataBuffer,
+ fillerLength,
+ 12 + 1);
+}
+
+WebRtc_Word32
+RTPSenderH264::SendH264FillerData(const WebRtc_UWord32 captureTimestamp,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 bytes
+ )
+{
+
+ const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
+ WebRtc_UWord16 maxLength = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - _rtpSender.RTPHeaderLength();
+
+ WebRtc_Word32 bytesToSend=bytes;
+ WebRtc_UWord16 fillerLength=0;
+
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+
+ while(bytesToSend>0)
+ {
+ fillerLength=maxLength;
+ if(fillerLength<maxLength)
+ {
+ fillerLength = (WebRtc_UWord16) bytesToSend;
+ }
+
+ bytesToSend-=fillerLength;
+
+ if (fillerLength > WEBRTC_IP_PACKET_SIZE - 12 - 1)
+ {
+ return 0;
+ }
+
+ if (fillerLength == 0)
+ {
+ // do not send an empty packet, will not reach JB
+ fillerLength = 1;
+ }
+
+ // send paded data
+ // correct seq num, time stamp and payloadtype
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType, false,captureTimestamp, true, true);
+
+ // set filler NALU type
+ dataBuffer[12] = 12; // NRI field = 0, type 12
+
+ // send codec valid data, H.264 has defined data which is binary 1111111
+ // fill with 0xff
+ memset(dataBuffer + 12 + 1, 0xff, fillerLength-1);
+
+ if( _rtpSender.SendToNetwork(dataBuffer,
+ fillerLength,
+ 12)<0)
+ {
+
+ return -1;;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* incomingRTPPacket,
+ const WebRtc_UWord16 incomingRTPPacketSize,
+ const WebRtc_UWord32 ssrc,
+ const bool higestLayer)
+{
+ if (rtpHeader->header.sequenceNumber != (WebRtc_UWord16)(_h264SVCRelaySequenceNumber + 1))
+ {
+ // not continous, signal loss
+ _rtpSender.IncrementSequenceNumber();
+ }
+ _h264SVCRelaySequenceNumber = rtpHeader->header.sequenceNumber;
+
+
+ if (rtpHeader->header.timestamp != _h264SVCRelayTimeStamp)
+ {
+ // new frame
+ _h264SVCRelayLayerComplete = false;
+ }
+
+ if (rtpHeader->header.timestamp == _h264SVCRelayTimeStamp &&
+ _h264SVCRelayLayerComplete)
+ {
+ // sanity, end of layer already sent
+ // Could happened for fragmented packet with missing PACSI info (PACSI packet reorded and received after packet it belongs to)
+ // fragmented packet has no layer info set (default info 0)
+ return 0;
+ }
+ _h264SVCRelayTimeStamp = rtpHeader->header.timestamp;
+
+ // re-packetize H.264-SVC packets
+ // we keep the timestap unchanged
+ // make a copy and only change the SSRC and seqNum
+
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+ memcpy(dataBuffer, incomingRTPPacket, incomingRTPPacketSize);
+
+ // _sequenceNumber initiated in Init()
+ // _ssrc initiated in constructor
+
+ // re-write payload type
+ if(_h264SVCPayloadType != -1)
+ {
+ dataBuffer[1] &= kRtpMarkerBitMask;
+ dataBuffer[1] += _h264SVCPayloadType;
+ }
+
+ // _sequenceNumber will not work for re-ordering by NACK from original sender
+ // engine responsible for this
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning
+ //ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc);
+
+ // how do we know it's the last relayed packet in a frame?
+ // 1) packets arrive in order, the engine manages that
+ // 2) highest layer that we relay
+ // 3) the end bit is set for the highest layer
+
+ if(higestLayer && rtpHeader->type.Video.codecHeader.H264.relayE)
+ {
+ // set marker bit
+ dataBuffer[1] |= kRtpMarkerBitMask;
+
+ // set relayed layer as complete
+ _h264SVCRelayLayerComplete = true;
+ }
+ return _rtpSender.SendToNetwork(dataBuffer,
+ incomingRTPPacketSize - rtpHeader->header.headerLength,
+ rtpHeader->header.headerLength);
+}
+
+WebRtc_Word32
+RTPSenderH264::SendH264_STAP_A(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ bool& switchToFUA,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength)
+{
+ const WebRtc_Word32 H264_NALU_LENGTH = 2;
+
+ WebRtc_UWord16 h264HeaderLength = 1; // normal header length
+ WebRtc_UWord16 maxPayloadLengthSTAP_A = _rtpSender.MaxPayloadLength() -
+ FECPacketOverhead() - rtpHeaderLength -
+ h264HeaderLength - H264_NALU_LENGTH;
+
+ WebRtc_Word32 dataOffset = rtpHeaderLength + h264HeaderLength;
+ WebRtc_UWord8 NRI = 0;
+ WebRtc_UWord16 payloadBytesInPacket = 0;
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+
+ if (ptrH264Info->payloadSize[idxNALU] > maxPayloadLengthSTAP_A)
+ {
+ // we need to fragment NAL switch to mode FU-A
+ switchToFUA = true;
+ } else
+ {
+ // combine as many NAL units in every IP packet
+ do
+ {
+ if(!_h264SendPPS_SPS)
+ {
+ // don't send NALU of type 7 and 8 SPS and PPS
+ if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
+ {
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ idxNALU++;
+ continue;
+ }
+ }
+ if(ptrH264Info->payloadSize[idxNALU] + payloadBytesInPacket <= maxPayloadLengthSTAP_A)
+ {
+ if(ptrH264Info->NRI[idxNALU] > NRI)
+ {
+ NRI = ptrH264Info->NRI[idxNALU];
+ }
+ // put NAL size into packet
+ dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] >> 8);
+ dataOffset++;
+ dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] & 0xff);
+ dataOffset++;
+ // Put payload in packet
+ memcpy(&dataBuffer[dataOffset], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
+ dataOffset += ptrH264Info->payloadSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ payloadBytesInPacket += (WebRtc_UWord16)(ptrH264Info->payloadSize[idxNALU] + H264_NALU_LENGTH);
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ } else
+ {
+ // we don't fitt the next NALU in this packet
+ break;
+ }
+ idxNALU++;
+ }while(payloadBytesToSend);
+ }
+
+ // sanity
+ // don't send empty packets
+ if (payloadBytesInPacket)
+ {
+ // add RTP header
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType, (payloadBytesToSend==0)?true:false, captureTimeStamp);
+ dataBuffer[rtpHeaderLength] = 24 + NRI; // STAP-A == 24
+ WebRtc_UWord16 payloadLength = payloadBytesInPacket + h264HeaderLength;
+
+ if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength))
+ {
+ return -1;
+ }
+ }
+ return 0;
+} // end STAP-A
+
+// STAP-A for H.264 SVC
+WebRtc_Word32
+RTPSenderH264::SendH264_STAP_A_PACSI(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ bool& switchToFUA,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength,
+ WebRtc_UWord16& decodingOrderNumber)
+{
+ const WebRtc_Word32 H264_NALU_LENGTH = 2;
+
+ WebRtc_UWord16 h264HeaderLength = 1; // normal header length
+ WebRtc_UWord16 maxPayloadLengthSTAP_A = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - rtpHeaderLength - h264HeaderLength - H264_NALU_LENGTH;
+ WebRtc_Word32 dataOffset = rtpHeaderLength + h264HeaderLength;
+ WebRtc_UWord8 NRI = 0;
+ WebRtc_UWord16 payloadBytesInPacket = 0;
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+ bool firstNALUNotIDR = true; //delta
+
+ // Put PACSI NAL unit into packet
+ WebRtc_Word32 lengthPACSI = 0;
+ WebRtc_UWord32 PACSI_NALlength = ptrH264Info->PACSI[idxNALU].NALlength;
+ if (PACSI_NALlength > maxPayloadLengthSTAP_A)
+ {
+ return -1;
+ }
+ dataBuffer[dataOffset++] = (WebRtc_UWord8)(PACSI_NALlength >> 8);
+ dataBuffer[dataOffset++] = (WebRtc_UWord8)(PACSI_NALlength & 0xff);
+
+ // end bit will be updated later, since another NALU in this packet might be the last
+ WebRtc_Word32 lengthPASCINALU = AddH264PACSINALU(true,
+ false,
+ ptrH264Info->PACSI[idxNALU],
+ ptrH264Info->SVCheader[idxNALU],
+ decodingOrderNumber,
+ dataBuffer,
+ dataOffset);
+ if (lengthPASCINALU <= 0)
+ {
+ return -1;
+ }
+ decodingOrderNumber++;
+
+ lengthPACSI = H264_NALU_LENGTH + lengthPASCINALU;
+ maxPayloadLengthSTAP_A -= (WebRtc_UWord16)lengthPACSI;
+ if (ptrH264Info->payloadSize[idxNALU] > maxPayloadLengthSTAP_A)
+ {
+ // we need to fragment NAL switch to mode FU-A
+ switchToFUA = true;
+ return 0;
+ }
+ if(!ptrH264Info->SVCheader[idxNALU].idr)
+ {
+ firstNALUNotIDR = true;
+ }
+
+ WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
+ (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
+ ptrH264Info->SVCheader[idxNALU].temporalID;
+
+ {
+ // combine as many NAL units in every IP packet, with the same priorityID
+ // Improvement we could allow several very small MGS NALU from different layers to be sent in one packet
+
+ do
+ {
+ if(!_h264SendPPS_SPS)
+ {
+ // Don't send NALU of type 7 and 8 SPS and PPS,
+ // they could be signaled outofband
+ if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
+ {
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ idxNALU++;
+ continue;
+ }
+ }
+ // don't send NALU type 6 (SEI message) not allowed when we send it in PACSI
+ if(ptrH264Info->type[idxNALU] == 6)
+ {
+ // SEI NALU Don't send, not allowed when we send it in PACSI
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ idxNALU++;
+ continue;
+ }
+
+ const WebRtc_UWord32 layerNALU = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
+ (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
+ ptrH264Info->SVCheader[idxNALU].temporalID;
+
+ // we need to break on a new layer
+ if( ptrH264Info->payloadSize[idxNALU] + payloadBytesInPacket <= maxPayloadLengthSTAP_A &&
+ layerNALU == layer)
+ {
+ if(ptrH264Info->NRI[idxNALU] > NRI)
+ {
+ NRI = ptrH264Info->NRI[idxNALU];
+ }
+ // put NAL size into packet
+ dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] >> 8);
+ dataOffset++;
+ dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] & 0xff);
+ dataOffset++;
+ // Put payload in packet
+ memcpy(&dataBuffer[dataOffset], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
+ dataOffset += ptrH264Info->payloadSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ payloadBytesInPacket += (WebRtc_UWord16)(ptrH264Info->payloadSize[idxNALU] + H264_NALU_LENGTH);
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ } else
+ {
+ // we don't fitt the next NALU in this packet or,
+ // it's the next layer
+
+ // check if we should send this NALU
+ // based on the layer
+
+ if(_useHighestSendLayer && layerNALU != layer)
+ {
+ // we don't send this NALU due to it's a new layer
+ // check if we should send the next or if this is the last
+ const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[idxNALU].dependencyID << 4) + ptrH264Info->SVCheader[idxNALU].qualityID;
+
+ bool highestLayer;
+ if(SendH264SVCLayer(frameType,
+ ptrH264Info->SVCheader[idxNALU].temporalID,
+ dependencyQualityID,
+ highestLayer) == false)
+ {
+ // will trigger markerbit and stop sending this frame
+ payloadBytesToSend = 0;
+ }
+ }
+ break;
+ }
+ idxNALU++;
+
+ }while(payloadBytesToSend);
+ }
+
+ // sanity, don't send empty packets
+ if (payloadBytesInPacket)
+ {
+ // add RTP header
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType, (payloadBytesToSend==0)?true:false, captureTimeStamp);
+
+ dataBuffer[rtpHeaderLength] = 24 + NRI; // STAP-A == 24
+
+ // NRI for PACSI
+ dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 1] &= 0x1f; // zero out NRI field
+ dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 1] |= NRI;
+
+ if(ptrH264Info->PACSI[idxNALU-1].E)
+ {
+ // update end bit
+ dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 5] |= 0x01;
+ }
+ if(firstNALUNotIDR)
+ {
+ // we have to check if any of the NALU in this packet is an IDR NALU
+ bool setIBit = false;
+ for(int i = 0; i < idxNALU; i++)
+ {
+ if(ptrH264Info->SVCheader[i].idr)
+ {
+ setIBit = true;
+ break;
+ }
+ }
+ if(setIBit)
+ {
+ // update I bit
+ dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 2] |= 0x40;
+ }
+ }
+ const WebRtc_UWord16 payloadLength = payloadBytesInPacket + h264HeaderLength + (WebRtc_UWord16)lengthPACSI;
+ if(-1 == SendVideoPacket(frameType,
+ dataBuffer,
+ payloadLength,
+ rtpHeaderLength,
+ layer==0))
+ {
+ return -1;
+ }
+ }
+ return 0;
+} // end STAP-A
+
+WebRtc_Word32
+RTPSenderH264::SendH264_FU_A(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength,
+ WebRtc_UWord16& decodingOrderNumber,
+ const bool sendSVCPACSI)
+{
+
+ // FUA for the rest of the frame
+ WebRtc_UWord16 maxPayloadLength = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - rtpHeaderLength;
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+ WebRtc_UWord32 payloadBytesRemainingInNALU = ptrH264Info->payloadSize[idxNALU];
+
+ bool isBaseLayer=false;
+
+ if(payloadBytesRemainingInNALU > maxPayloadLength)
+ {
+ // we need to fragment NALU
+ const WebRtc_UWord16 H264_FUA_LENGTH = 2; // FU-a H.264 header is 2 bytes
+
+ if(sendSVCPACSI)
+ {
+ SendH264_SinglePACSI(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ true,
+ false);
+
+ WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
+ (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
+ ptrH264Info->SVCheader[idxNALU].temporalID;
+ isBaseLayer=(layer==0);
+ }
+
+ // First packet
+ _rtpSender.BuildRTPheader(dataBuffer,payloadType, false, captureTimeStamp);
+
+ WebRtc_UWord16 maxPayloadLengthFU_A = maxPayloadLength - H264_FUA_LENGTH ;
+ WebRtc_UWord8 fuaIndc = 28 + ptrH264Info->NRI[idxNALU];
+ dataBuffer[rtpHeaderLength] = fuaIndc; // FU-A indicator
+ dataBuffer[rtpHeaderLength+1] = (WebRtc_UWord8)(ptrH264Info->type[idxNALU] + 0x80)/*start*/; // FU-A header
+
+ memcpy(&dataBuffer[rtpHeaderLength + H264_FUA_LENGTH], &data[ptrH264Info->startCodeSize[idxNALU]+1], maxPayloadLengthFU_A);
+ WebRtc_UWord16 payloadLength = maxPayloadLengthFU_A + H264_FUA_LENGTH;
+ if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength, isBaseLayer))
+ {
+ return -1;
+ }
+
+ //+1 is from the type that is coded into the FU-a header
+ data += maxPayloadLengthFU_A + 1 + ptrH264Info->startCodeSize[idxNALU]; // inc data ptr
+ payloadBytesToSend -= maxPayloadLengthFU_A+1+ptrH264Info->startCodeSize[idxNALU];
+ payloadBytesRemainingInNALU -= maxPayloadLengthFU_A+1;
+
+ // all non first/last packets
+ while(payloadBytesRemainingInNALU > maxPayloadLengthFU_A)
+ {
+ if(sendSVCPACSI)
+ {
+ SendH264_SinglePACSI(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ false,
+ false);
+ }
+
+ // prepare next header
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType, false, captureTimeStamp);
+
+ dataBuffer[rtpHeaderLength] = (WebRtc_UWord8)fuaIndc; // FU-A indicator
+ dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU]; // FU-A header
+
+ memcpy(&dataBuffer[rtpHeaderLength+H264_FUA_LENGTH], data, maxPayloadLengthFU_A);
+ payloadLength = maxPayloadLengthFU_A + H264_FUA_LENGTH;
+
+ if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength,isBaseLayer))
+ {
+ return -1;
+ }
+ data += maxPayloadLengthFU_A; // inc data ptr
+ payloadBytesToSend -= maxPayloadLengthFU_A;
+ payloadBytesRemainingInNALU -= maxPayloadLengthFU_A;
+ dataBuffer[rtpHeaderLength] = fuaIndc; // FU-A indicator
+ dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU]; // FU-A header
+ }
+ if(sendSVCPACSI)
+ {
+ SendH264_SinglePACSI(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ false,
+ true); // last packet in NALU
+
+ if(_useHighestSendLayer && idxNALU+1 < ptrH264Info->numNALUs)
+ {
+ // not last NALU in frame
+ // check if it's the the next layer should not be sent
+
+ // check if we should send the next or if this is the last
+ const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[idxNALU+1].dependencyID << 4) +
+ ptrH264Info->SVCheader[idxNALU+1].qualityID;
+
+ bool highestLayer;
+ if(SendH264SVCLayer(frameType,
+ ptrH264Info->SVCheader[idxNALU+1].temporalID,
+ dependencyQualityID,
+ highestLayer) == false)
+ {
+ // will trigger markerbit and stop sending this frame
+ payloadBytesToSend = payloadBytesRemainingInNALU;
+ }
+ }
+ }
+ // last packet in NALU
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType,(payloadBytesToSend == (WebRtc_Word32)payloadBytesRemainingInNALU)?true:false, captureTimeStamp);
+ dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU] + 0x40/*stop*/; // FU-A header
+
+ memcpy(&dataBuffer[rtpHeaderLength+H264_FUA_LENGTH], data, payloadBytesRemainingInNALU);
+ payloadLength = (WebRtc_UWord16)payloadBytesRemainingInNALU + H264_FUA_LENGTH;
+ payloadBytesToSend -= payloadBytesRemainingInNALU;
+ if(payloadBytesToSend != 0)
+ {
+ data += payloadBytesRemainingInNALU; // inc data ptr
+ }
+ idxNALU++;
+ if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength,isBaseLayer))
+ {
+ return -1;
+ }
+ } else
+ {
+ // send NAL unit in singel mode
+ return SendH264_SingleMode(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ payloadBytesToSend,
+ data,
+ rtpHeaderLength,
+ sendSVCPACSI);
+ }
+ // end FU-a
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SendH264_SingleMode(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength,
+ WebRtc_UWord16& decodingOrderNumber,
+ const bool sendSVCPACSI)
+{
+ // no H.264 header lenght in single mode
+ // we use WEBRTC_IP_PACKET_SIZE instead of the configured MTU since it's better to send fragmented UDP than not to send
+ const WebRtc_UWord16 maxPayloadLength = WEBRTC_IP_PACKET_SIZE - _rtpSender.PacketOverHead() - FECPacketOverhead() - rtpHeaderLength;
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+ bool isBaseLayer=false;
+
+ if(ptrH264Info->payloadSize[idxNALU] > maxPayloadLength)
+ {
+ return -3;
+ }
+ if(!_h264SendPPS_SPS)
+ {
+ // don't send NALU of type 7 and 8 SPS and PPS
+ if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
+ {
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ idxNALU++;
+ return 0;
+ }
+ }
+ if(sendSVCPACSI)
+ {
+ SendH264_SinglePACSI(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ true,
+ true);
+
+ WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
+ (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
+ ptrH264Info->SVCheader[idxNALU].temporalID;
+ isBaseLayer=(layer==0);
+ }
+
+ // Put payload in packet
+ memcpy(&dataBuffer[rtpHeaderLength], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
+
+ WebRtc_UWord16 payloadBytesInPacket = (WebRtc_UWord16)ptrH264Info->payloadSize[idxNALU];
+ payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU]; // left to send
+
+ //
+ _rtpSender.BuildRTPheader(dataBuffer,payloadType,(payloadBytesToSend ==0)?true:false, captureTimeStamp);
+
+ dataBuffer[rtpHeaderLength] &= 0x1f; // zero out NRI field
+ dataBuffer[rtpHeaderLength] |= ptrH264Info->NRI[idxNALU]; // nri
+ if(payloadBytesToSend > 0)
+ {
+ data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
+ }
+ idxNALU++;
+ if(-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket, rtpHeaderLength,isBaseLayer))
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SendH264_SinglePACSI(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ const WebRtc_UWord16 idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const bool firstPacketInNALU,
+ const bool lastPacketInNALU);
+{
+ // Send PACSI in single mode
+ WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
+ WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)_rtpSender.BuildRTPheader(dataBuffer, payloadType,false, captureTimeStamp);
+ WebRtc_Word32 dataOffset = rtpHeaderLength;
+
+ WebRtc_Word32 lengthPASCINALU = AddH264PACSINALU(firstPacketInNALU,
+ lastPacketInNALU,
+ ptrH264Info->PACSI[idxNALU],
+ ptrH264Info->SVCheader[idxNALU],
+ decodingOrderNumber,
+ dataBuffer,
+ dataOffset);
+
+ if (lengthPASCINALU <= 0)
+ {
+ return -1;
+ }
+ decodingOrderNumber++;
+
+ WebRtc_UWord16 payloadBytesInPacket = (WebRtc_UWord16)lengthPASCINALU;
+
+ // Set payload header (first payload byte co-serves as the payload header)
+ dataBuffer[rtpHeaderLength] &= 0x1f; // zero out NRI field
+ dataBuffer[rtpHeaderLength] |= ptrH264Info->NRI[idxNALU]; // nri
+
+ const WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
+ (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
+ ptrH264Info->SVCheader[idxNALU].temporalID;
+
+ if (-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket, rtpHeaderLength,layer==0))
+ {
+ return -1;
+ }
+ return 0;
+}
+
+
+
+
+WebRtc_Word32
+RTPSenderH264::SendH264SVC(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ H264Information& h264Information,
+ WebRtc_UWord16& decodingOrderNumber)
+{
+ WebRtc_Word32 payloadBytesToSend = payloadSize;
+ const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
+
+ const H264Info* ptrH264Info = NULL;
+ if (h264Information.GetInfo(payloadData,payloadSize, ptrH264Info) == -1)
+ {
+ return -1;
+ }
+ if(_useHighestSendLayer)
+ {
+ // we need to check if we should drop the frame
+ // it could be a temporal layer (aka a temporal frame)
+ const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[0].dependencyID << 4) + ptrH264Info->SVCheader[0].qualityID;
+
+ bool dummyHighestLayer;
+ if(SendH264SVCLayer(frameType,
+ ptrH264Info->SVCheader[0].temporalID,
+ dependencyQualityID,
+ dummyHighestLayer) == false)
+ {
+ // skip send this frame
+ return 0;
+ }
+ }
+
+ WebRtc_UWord16 idxNALU = 0;
+ while (payloadBytesToSend > 0)
+ {
+ bool switchToFUA = false;
+ if (SendH264_STAP_A_PACSI(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ switchToFUA,
+ payloadBytesToSend,
+ payloadData,
+ rtpHeaderLength,
+ decodingOrderNumber) != 0)
+ {
+ return -1;
+ }
+ if(switchToFUA)
+ {
+ // FU_A for this NALU
+ if (SendH264_FU_A(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ payloadBytesToSend,
+ payloadData,
+ rtpHeaderLength,
+ true) != 0)
+ {
+ return -1;
+ }
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SetH264PacketizationMode(const H264PacketizationMode mode)
+{
+ _h264Mode = mode;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::SetH264SendModeNALU_PPS_SPS(const bool dontSend)
+{
+ _h264SendPPS_SPS = !dontSend;
+ return 0;
+}
+
+bool
+RTPSenderH264::SendH264SVCLayer(const FrameType frameType,
+ const WebRtc_UWord8 temporalID,
+ const WebRtc_UWord8 dependencyQualityID,
+ bool& higestLayer)
+{
+ WebRtc_UWord8 dependencyID = dependencyQualityID >> 4;
+
+ // keyframe required to switch between dependency layers not quality and temporal
+ if( _highestDependencyLayer != _highestDependencyLayerOld)
+ {
+ // we want to switch dependency layer
+ if(frameType == kVideoFrameKey)
+ {
+ // key frame we can change layer if it's correct layer
+ if(_highestDependencyLayer > _highestDependencyLayerOld)
+ {
+ // we want to switch up
+ // does this packet belong to a new layer?
+
+ if( dependencyID > _highestDependencyLayerOld &&
+ dependencyID <= _highestDependencyLayer)
+ {
+ _highestDependencyLayerOld = dependencyID;
+ _highestDependencyQualityIDOld = _highestDependencyQualityID;
+
+ if( dependencyID == _highestDependencyLayer &&
+ dependencyQualityID == _highestDependencyQualityID)
+ {
+ higestLayer = true;
+ }
+ // relay
+ return true;
+ }
+ }
+ if(_highestDependencyLayer < _highestDependencyLayerOld)
+ {
+ // we want to switch down
+ // does this packet belong to a low layer?
+ if( dependencyID <= _highestDependencyLayer)
+ {
+ _highestDependencyLayerOld = dependencyID;
+ _highestDependencyQualityIDOld = _highestDependencyQualityID;
+ if( dependencyID == _highestDependencyLayer &&
+ dependencyQualityID == _highestDependencyQualityID)
+ {
+ higestLayer = true;
+ }
+ // relay
+ return true;
+ }
+ }
+ } else
+ {
+ // Delta frame and we are waiting to switch dependency layer
+ if(_highestDependencyLayer > _highestDependencyLayerOld)
+ {
+ // we want to switch up to a higher dependency layer
+ // use old setting until we get a key-frame
+
+ // filter based on old dependency
+ // we could have allowed to add a MGS layer lower than the dependency ID
+ // but then we can't know the highest layer relayed we assume that the user
+ // will add one layer at a time
+ if( _highestTemporalLayer < temporalID ||
+ _highestDependencyLayerOld < dependencyID ||
+ _highestDependencyQualityIDOld < dependencyQualityID)
+ {
+ // drop
+ return false;
+ }
+ // highest layer based on old
+ if( dependencyID == _highestDependencyLayerOld &&
+ dependencyQualityID == _highestDependencyQualityIDOld)
+ {
+ higestLayer = true;
+ }
+ } else
+ {
+ // we want to switch down to a lower dependency layer,
+ // use old setting, done bellow
+ // drop all temporal layers while waiting for the key-frame
+ if(temporalID > 0)
+ {
+ // drop
+ return false;
+ }
+ // we can't drop a lower MGS layer since this might depend on it
+ // however we can drop MGS layers larger than dependecyQualityId
+ // with dependency from old and quality 0
+ if( _highestDependencyLayerOld < dependencyID ||
+ (_highestDependencyQualityIDOld & 0xf0) < dependencyQualityID)
+ {
+ // drop
+ return false;
+ }
+ if( dependencyID == _highestDependencyLayerOld &&
+ dependencyQualityID == (_highestDependencyQualityIDOld & 0xf0))
+ {
+ higestLayer = true;
+ }
+ }
+ }
+ } else
+ {
+ // filter based on current state
+ if( _highestTemporalLayer < temporalID ||
+ _highestDependencyLayer < dependencyID ||
+ _highestDependencyQualityID < dependencyQualityID)
+ {
+ // drop
+ return false;
+ }
+ if( dependencyID == _highestDependencyLayer &&
+ dependencyQualityID == _highestDependencyQualityID)
+ {
+ higestLayer = true;
+ }
+ }
+ return true;
+}
+
+WebRtc_Word32
+RTPSenderH264::SetHighestSendLayer(const WebRtc_UWord8 dependencyQualityLayer,
+ const WebRtc_UWord8 temporalLayer)
+{
+ const WebRtc_UWord8 dependencyLayer = (dependencyQualityLayer >> 4);
+
+ if(_highestDependencyLayerOld != _highestDependencyLayer)
+ {
+ // we have not switched to the new dependency yet
+ } else
+ {
+ if(_highestDependencyLayer == dependencyLayer)
+ {
+ // no change of dependency
+ // switch now _highestDependencyQualityIDOld
+ _highestDependencyQualityIDOld = dependencyQualityLayer;
+ }else
+ {
+ // change of dependency, update _highestDependencyQualityIDOld store as old
+ _highestDependencyQualityIDOld = _highestDependencyQualityID;
+ }
+ }
+ _useHighestSendLayer = true;
+ _highestDependencyLayer = dependencyLayer;
+ _highestDependencyQualityID = dependencyQualityLayer;
+ _highestTemporalLayer = temporalLayer;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderH264::HighestSendLayer(WebRtc_UWord8& dependencyQualityLayer,
+ WebRtc_UWord8& temporalLayer)
+{
+ if (!_useHighestSendLayer)
+ {
+ // No information set
+ return -1;
+ }
+ dependencyQualityLayer = _highestDependencyQualityID;
+ temporalLayer = _highestTemporalLayer;
+ return 0;
+}
+/*
+* H.264
+*/
+WebRtc_Word32
+RTPSenderH264::SendH264(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ H264Information& h264Information)
+{
+ WebRtc_Word32 payloadBytesToSend = payloadSize;
+ const WebRtc_UWord8* data = payloadData;
+ bool switchToFUA = false;
+ const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
+
+ const H264Info* ptrH264Info = NULL;
+ if (h264Information.GetInfo(payloadData,payloadSize, ptrH264Info) == -1)
+ {
+ return -1;
+ }
+ WebRtc_UWord16 idxNALU = 0;
+ WebRtc_UWord16 DONCdummy = 0;
+
+ while (payloadBytesToSend > 0)
+ {
+ switch(_h264Mode)
+ {
+ case H264_NON_INTERLEAVED_MODE:
+
+ if(!switchToFUA)
+ {
+ if(SendH264_STAP_A(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ switchToFUA,
+ payloadBytesToSend,
+ data,
+ rtpHeaderLength) != 0)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ // FUA for the rest of the frame
+ if(SendH264_FU_A(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ payloadBytesToSend,
+ data,
+ rtpHeaderLength,
+ DONCdummy) != 0)
+ {
+ return -1;
+ }
+ // try to go back to STAP_A
+ switchToFUA = false;
+ }
+ break;
+ case H264_SINGLE_NAL_MODE:
+ {
+ // modeSingleU
+ if(SendH264_SingleMode(frameType,
+ ptrH264Info,
+ idxNALU,
+ payloadType,
+ captureTimeStamp,
+ payloadBytesToSend,
+ data,
+ rtpHeaderLength,
+ DONCdummy) != 0)
+ {
+ return -1;
+ }
+ break;
+ }
+ case H264_INTERLEAVED_MODE:
+ // not supported
+ assert(false);
+ return -1;
+ }
+ }
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h
new file mode 100644
index 0000000..564b870
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h
@@ -0,0 +1,179 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
+
+#include "typedefs.h"
+#include "ModuleRTPRTCPConfig.h"
+#include "rtp_rtcp_defines.h"
+#include "h264_information.h"
+
+#include "RTPSender.h"
+
+namespace webrtc {
+class RTPSenderH264
+{
+public:
+ WebRtc_Word32 SendH264(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ H264Information& h264Information);
+
+ WebRtc_Word32 SendH264SVC(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ H264Information& h264Information);
+
+ // H.264 AVC
+ WebRtc_Word32 SetH264PacketizationMode(const H264PacketizationMode mode);
+
+ WebRtc_Word32 SetH264SendModeNALU_PPS_SPS(const bool dontSend);
+
+ // H.264 SVC
+ WebRtc_Word32 SetHighestSendLayer(const WebRtc_UWord8 dependencyQualityLayer,
+ const WebRtc_UWord8 temporalLayer);
+
+ WebRtc_Word32 HighestSendLayer(WebRtc_UWord8& dependencyQualityLayer,
+ WebRtc_UWord8& temporalLayer);
+
+protected:
+ RTPSenderH264(RTPSenderInterface* rtpSender);
+ virtual ~RTPSenderH264();
+
+ WebRtc_Word32 Init();
+
+ virtual WebRtc_UWord16 FECPacketOverhead() const = 0;
+ virtual RtpVideoCodecTypes VideoCodecType() const = 0;
+
+ virtual WebRtc_Word32 SendVideoPacket(const FrameType frameType,
+ const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 payloadLength,
+ const WebRtc_UWord16 rtpHeaderLength,
+ bool baseLayerVideoPacket=false) = 0;
+
+
+ bool SendH264SVCLayer(const FrameType frameType,
+ const WebRtc_UWord8 temporalID,
+ const WebRtc_UWord8 dependencyQualityID,
+ bool& higestLayer);
+
+ // H.264 SVC
+ WebRtc_Word32 AddH264PACSINALU(const bool firstPacketInNALU,
+ const bool lastPacketInNALU,
+ const H264_PACSI_NALU& paci,
+ const H264_SVC_NALUHeader& svc,
+ const WebRtc_UWord16 DONC,
+ WebRtc_UWord8* databuffer,
+ WebRtc_Word32& curByte) const;
+
+ WebRtc_Word32 SendH264FillerData(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord16 bytesToSend,
+ const WebRtc_UWord32 ssrc);
+
+ WebRtc_Word32 SendH264FillerData(const WebRtc_UWord32 captureTimestamp,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 bytesToSend);
+
+ WebRtc_Word32 SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* incomingRTPPacket,
+ const WebRtc_UWord16 incomingRTPPacketSize,
+ const WebRtc_UWord32 ssrc,
+ const bool higestLayer);
+
+ WebRtc_Word32 SetH264RelaySequenceNumber(const WebRtc_UWord16 seqNum);
+
+ WebRtc_Word32 SetH264RelayCompleteLayer(const bool complete);
+
+ // H.264
+ H264PacketizationMode _h264Mode;
+ bool _h264SendPPS_SPS;
+
+ // H.264-SVC
+ WebRtc_Word8 _h264SVCPayloadType;
+ WebRtc_UWord16 _h264SVCRelaySequenceNumber;
+ WebRtc_UWord32 _h264SVCRelayTimeStamp;
+ bool _h264SVCRelayLayerComplete;
+
+
+private:
+ // H.264
+ WebRtc_Word32 SendH264_SingleMode(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength,
+ const bool sendSVCPACSI=false);
+
+ WebRtc_Word32 SendH264_FU_A(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength,
+ const bool sendSVCPACSI = false);
+
+ WebRtc_Word32 SendH264_STAP_A(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ bool& switchToFUA,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLength);
+
+ WebRtc_Word32 SendH264_STAP_A_PACSI(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ WebRtc_UWord16 &idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ bool& switchToFUA,
+ WebRtc_Word32 &payloadBytesToSend,
+ const WebRtc_UWord8*& data,
+ const WebRtc_UWord16 rtpHeaderLengh)
+
+ WebRtc_Word32 SendH264_SinglePACSI(const FrameType frameType,
+ const H264Info* ptrH264Info,
+ const WebRtc_UWord16 idxNALU,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const bool firstPacketInNALU,
+ const bool lastPacketInNALU);
+
+ bool AddH264SVCNALUHeader(const H264_SVC_NALUHeader& svc,
+ WebRtc_UWord8* databuffer,
+ WebRtc_Word32& curByte) const;
+
+ RTPSenderInterface& _rtpSender;
+
+ // relay
+ bool _useHighestSendLayer;
+ WebRtc_UWord8 _highestDependencyLayerOld;
+ WebRtc_UWord8 _highestDependencyQualityIDOld;
+ WebRtc_UWord8 _highestDependencyLayer;
+ WebRtc_UWord8 _highestDependencyQualityID;
+ WebRtc_UWord8 _highestTemporalLayer;
+
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc
new file mode 100644
index 0000000..a14a2ef
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc
@@ -0,0 +1,316 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "bandwidth_management.h"
+#include "trace.h"
+#include "rtp_utility.h"
+#include "rtp_rtcp_config.h"
+
+#include <math.h> // sqrt()
+
+namespace webrtc {
+
+BandwidthManagement::BandwidthManagement(const WebRtc_Word32 id) :
+ _id(id),
+ _critsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _lastPacketLossExtendedHighSeqNum(0),
+ _lastReportAllLost(false),
+ _lastLoss(0),
+ _accumulateLostPacketsQ8(0),
+ _accumulateExpectedPackets(0),
+ _bitRate(0),
+ _minBitRateConfigured(0),
+ _maxBitRateConfigured(0),
+ _last_fraction_loss(0),
+ _last_round_trip_time(0),
+ _bwEstimateIncoming(0),
+ _smoothedFractionLostQ4(-1), // indicate uninitialized
+ _sFLFactorQ4(14), // 0.875 in Q4
+ _timeLastIncrease(0)
+{
+}
+
+BandwidthManagement::~BandwidthManagement()
+{
+ delete _critsect;
+}
+
+void
+BandwidthManagement::SetSendBitrate(const WebRtc_UWord32 startBitrate,
+ const WebRtc_UWord16 minBitrateKbit,
+ const WebRtc_UWord16 maxBitrateKbit)
+{
+ CriticalSectionScoped cs(_critsect);
+
+ _bitRate = startBitrate;
+ _minBitRateConfigured = minBitrateKbit*1000;
+ if(maxBitrateKbit == 0)
+ {
+ // no max configured use 1Gbit/s
+ _maxBitRateConfigured = 1000000000;
+ } else
+ {
+ _maxBitRateConfigured = maxBitrateKbit*1000;
+ }
+}
+
+WebRtc_Word32
+BandwidthManagement::MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit)
+{
+ CriticalSectionScoped cs(_critsect);
+
+ if(_maxBitRateConfigured == 0)
+ {
+ return -1;
+ }
+ *maxBitrateKbit = (WebRtc_UWord16)(_maxBitRateConfigured/1000);
+ return 0;
+}
+
+WebRtc_Word32
+BandwidthManagement::UpdateBandwidthEstimate(const WebRtc_UWord16 bandWidthKbit,
+ WebRtc_UWord32* newBitrate,
+ WebRtc_UWord8* fractionLost,
+ WebRtc_UWord16* roundTripTime)
+{
+ *newBitrate = 0;
+ CriticalSectionScoped cs(_critsect);
+
+ _bwEstimateIncoming = bandWidthKbit*1000;
+
+ if(_bitRate == 0)
+ {
+ // BandwidthManagement off
+ return -1;
+ }
+ if (_bwEstimateIncoming > 0 && _bitRate > _bwEstimateIncoming)
+ {
+ _bitRate = _bwEstimateIncoming;
+ } else
+ {
+ return -1;
+ }
+ *newBitrate = _bitRate;
+ *fractionLost = _last_fraction_loss;
+ *roundTripTime = _last_round_trip_time;
+ return 0;
+}
+
+WebRtc_Word32 BandwidthManagement::UpdatePacketLoss(
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
+ WebRtc_UWord32 sentBitrate,
+ const WebRtc_UWord16 rtt,
+ WebRtc_UWord8* loss,
+ WebRtc_UWord32* newBitrate,
+ WebRtc_Word64 nowMS)
+{
+ CriticalSectionScoped cs(_critsect);
+
+ _last_fraction_loss = *loss;
+ _last_round_trip_time = rtt;
+
+ if(_bitRate == 0)
+ {
+ // BandwidthManagement off
+ return -1;
+ }
+
+ // Check sequence number diff and weight loss report
+ if (_lastPacketLossExtendedHighSeqNum > 0 &&
+ (lastReceivedExtendedHighSeqNum >= _lastPacketLossExtendedHighSeqNum))
+ {
+ // This is not the first loss report and the sequence number is
+ // non-decreasing. Calculate sequence number diff.
+ WebRtc_UWord32 seqNumDiff = lastReceivedExtendedHighSeqNum
+ - _lastPacketLossExtendedHighSeqNum;
+
+ // Check if this report and the last was 100% loss, then report
+ // 100% loss even though seqNumDiff is small.
+ // If not, go on with the checks.
+ if (!(_lastReportAllLost && *loss == 255))
+ {
+ _lastReportAllLost = (*loss == 255);
+
+ // Calculate number of lost packets.
+ // loss = 256 * numLostPackets / expectedPackets.
+ const int numLostPacketsQ8 = *loss * seqNumDiff;
+
+ // Accumulate reports.
+ _accumulateLostPacketsQ8 += numLostPacketsQ8;
+ _accumulateExpectedPackets += seqNumDiff;
+
+ // Report loss if the total report is based on sufficiently
+ // many packets.
+ const int limitNumPackets = 10;
+ if (_accumulateExpectedPackets >= limitNumPackets)
+ {
+ *loss = _accumulateLostPacketsQ8 / _accumulateExpectedPackets;
+
+ // Reset accumulators
+ _accumulateLostPacketsQ8 = 0;
+ _accumulateExpectedPackets = 0;
+ }
+ else
+ {
+ // Report zero loss until we have enough data to estimate
+ // the loss rate.
+ *loss = 0;
+ }
+ }
+ }
+ // Keep for next time.
+ _lastLoss = *loss;
+
+ // Remember the sequence number until next time
+ _lastPacketLossExtendedHighSeqNum = lastReceivedExtendedHighSeqNum;
+
+ WebRtc_UWord32 bitRate = ShapeSimple(*loss, rtt, sentBitrate, nowMS);
+ if (bitRate == 0)
+ {
+ // no change
+ return -1;
+ }
+ _bitRate = bitRate;
+ *newBitrate = bitRate;
+ return 0;
+}
+
+WebRtc_Word32 BandwidthManagement::AvailableBandwidth(
+ WebRtc_UWord32* bandwidthKbit) const {
+ CriticalSectionScoped cs(_critsect);
+ if (_bitRate == 0) {
+ return -1;
+ }
+ if (!bandwidthKbit) {
+ return -1;
+ }
+ *bandwidthKbit = _bitRate;
+ return 0;
+}
+
+/* Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
+ * The formula in RFC 3448, Section 3.1, is used.
+ */
+
+// protected
+WebRtc_Word32 BandwidthManagement::CalcTFRCbps(WebRtc_Word16 avgPackSizeBytes,
+ WebRtc_Word32 rttMs,
+ WebRtc_Word32 packetLoss)
+{
+ if (avgPackSizeBytes <= 0 || rttMs <= 0 || packetLoss <= 0)
+ {
+ // input variables out of range; return -1
+ return -1;
+ }
+
+ double R = static_cast<double>(rttMs)/1000; // RTT in seconds
+ int b = 1; // number of packets acknowledged by a single TCP acknowledgement; recommended = 1
+ double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds; recommended = 4*R
+ double p = static_cast<double>(packetLoss)/255; // packet loss rate in [0, 1)
+ double s = static_cast<double>(avgPackSizeBytes);
+
+ // calculate send rate in bytes/second
+ double X = s / (R * sqrt(2 * b * p / 3) + (t_RTO * (3 * sqrt( 3 * b * p / 8) * p * (1 + 32 * p * p))));
+
+ return (static_cast<WebRtc_Word32>(X*8)); // bits/second
+}
+
+/*
+* Simple bandwidth estimation. Depends a lot on bwEstimateIncoming and packetLoss.
+*/
+// protected
+WebRtc_UWord32 BandwidthManagement::ShapeSimple(WebRtc_Word32 packetLoss,
+ WebRtc_Word32 rtt,
+ WebRtc_UWord32 sentBitrate,
+ WebRtc_Word64 nowMS)
+{
+ WebRtc_UWord32 newBitRate = 0;
+ bool reducing = false;
+
+ // Limit the rate increases to once a second.
+ if (packetLoss <= 5)
+ {
+ if ((nowMS - _timeLastIncrease) <
+ kBWEUpdateIntervalMs)
+ {
+ return _bitRate;
+ }
+ _timeLastIncrease = nowMS;
+ }
+
+ if (packetLoss > 5 && packetLoss <= 26)
+ {
+ // 2% - 10%
+ newBitRate = _bitRate;
+ }
+ else if (packetLoss > 26)
+ {
+ // 26/256 ~= 10%
+ // reduce rate: newRate = rate * (1 - 0.5*lossRate)
+ // packetLoss = 256*lossRate
+ newBitRate = static_cast<WebRtc_UWord32>(
+ (sentBitrate * static_cast<double>(512 - packetLoss)) / 512.0);
+ reducing = true;
+ }
+ else
+ {
+ // increase rate by 8%
+ newBitRate = static_cast<WebRtc_UWord32>(_bitRate * 1.08 + 0.5);
+
+ // add 1 kbps extra, just to make sure that we do not get stuck
+ // (gives a little extra increase at low rates, negligible at higher rates)
+ newBitRate += 1000;
+ }
+
+ // Calculate smoothed loss number
+ if (_smoothedFractionLostQ4 < 0)
+ {
+ // startup
+ _smoothedFractionLostQ4 = static_cast<WebRtc_UWord16>(packetLoss);
+ }
+ else
+ {
+ _smoothedFractionLostQ4 = ((_sFLFactorQ4 * _smoothedFractionLostQ4 + 8) >> 4) // Q4*Q4 = Q8; down to Q4 again with proper rounding
+ + (16 - _sFLFactorQ4) * static_cast<WebRtc_UWord16>(packetLoss); // Q4 * Q0 = Q4
+ }
+
+ // Calculate what rate TFRC would apply in this situation
+ //WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, _smoothedFractionLostQ4 >> 4); // scale loss to Q0 (back to [0, 255])
+ WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, packetLoss); // scale loss to Q0 (back to [0, 255])
+
+ if (reducing &&
+ tfrcRate > 0 &&
+ static_cast<WebRtc_UWord32>(tfrcRate) > newBitRate)
+ {
+ // do not reduce further if rate is below TFRC rate
+ newBitRate = _bitRate;
+ }
+
+ if (_bwEstimateIncoming > 0 && newBitRate > _bwEstimateIncoming)
+ {
+ newBitRate = _bwEstimateIncoming;
+ }
+ if (newBitRate > _maxBitRateConfigured)
+ {
+ newBitRate = _maxBitRateConfigured;
+ }
+ if (newBitRate < _minBitRateConfigured)
+ {
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceRtpRtcp,
+ _id,
+ "The configured min bitrate (%u kbps) is greater than the "
+ "estimated available bandwidth (%u kbps).\n",
+ _minBitRateConfigured / 1000, newBitRate / 1000);
+ newBitRate = _minBitRateConfigured;
+ }
+ return newBitRate;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h
new file mode 100644
index 0000000..2516b86
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp_config.h"
+#include "critical_section_wrapper.h"
+
+/*
+* FEC and NACK added bitrate is handled outside class
+*/
+
+namespace webrtc {
+class BandwidthManagement
+{
+public:
+ BandwidthManagement(const WebRtc_Word32 id);
+ ~BandwidthManagement();
+
+ // Call when we receive a RTCP message with TMMBR or REMB
+ WebRtc_Word32 UpdateBandwidthEstimate(const WebRtc_UWord16 bandWidthKbit,
+ WebRtc_UWord32* newBitrate,
+ WebRtc_UWord8* fractionLost,
+ WebRtc_UWord16* roundTripTime);
+
+ // Call when we receive a RTCP message with a ReceiveBlock
+ WebRtc_Word32 UpdatePacketLoss(
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
+ WebRtc_UWord32 sentBitrate,
+ const WebRtc_UWord16 rtt,
+ WebRtc_UWord8* loss,
+ WebRtc_UWord32* newBitrate,
+ WebRtc_Word64 nowMS);
+
+ // If no bandwidth estimate is available or if |bandwidthKbit| is NULL,
+ // -1 is returned.
+ WebRtc_Word32 AvailableBandwidth(WebRtc_UWord32* bandwidthKbit) const;
+
+ void SetSendBitrate(const WebRtc_UWord32 startBitrate,
+ const WebRtc_UWord16 minBitrateKbit,
+ const WebRtc_UWord16 maxBitrateKbit);
+
+ WebRtc_Word32 MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit);
+
+protected:
+ WebRtc_UWord32 ShapeSimple(WebRtc_Word32 packetLoss,
+ WebRtc_Word32 rtt,
+ WebRtc_UWord32 sentBitrate,
+ WebRtc_Word64 nowMS);
+
+ WebRtc_Word32 CalcTFRCbps(WebRtc_Word16 avgPackSizeBytes,
+ WebRtc_Word32 rttMs,
+ WebRtc_Word32 packetLoss);
+
+private:
+ enum { kBWEUpdateIntervalMs = 1000 };
+
+ WebRtc_Word32 _id;
+
+ CriticalSectionWrapper* _critsect;
+
+ // incoming filters
+ WebRtc_UWord32 _lastPacketLossExtendedHighSeqNum;
+ bool _lastReportAllLost;
+ WebRtc_UWord8 _lastLoss;
+ int _accumulateLostPacketsQ8;
+ int _accumulateExpectedPackets;
+
+ // bitrate
+ WebRtc_UWord32 _bitRate;
+ WebRtc_UWord32 _minBitRateConfigured;
+ WebRtc_UWord32 _maxBitRateConfigured;
+
+ WebRtc_UWord8 _last_fraction_loss;
+ WebRtc_UWord16 _last_round_trip_time;
+
+ // bandwidth estimate
+ WebRtc_UWord32 _bwEstimateIncoming;
+ WebRtc_Word16 _smoothedFractionLostQ4;
+ WebRtc_Word16 _sFLFactorQ4; // forgetting factor for _smoothedFractionLostQ4
+ WebRtc_Word64 _timeLastIncrease;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/bitrate.cc b/trunk/src/modules/rtp_rtcp/source/bitrate.cc
new file mode 100644
index 0000000..be0c7dd
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/bitrate.cc
@@ -0,0 +1,203 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "Bitrate.h"
+#include "rtp_utility.h"
+
+#define BITRATE_AVERAGE_WINDOW 2000
+
+namespace webrtc {
+Bitrate::Bitrate(RtpRtcpClock* clock) :
+ _clock(*clock),
+ _packetRate(0),
+ _bitrate(0),
+ _bitrateNextIdx(0),
+ _timeLastRateUpdate(0),
+ _bytesCount(0),
+ _packetCount(0)
+{
+ memset(_packetRateArray, 0, sizeof(_packetRateArray));
+ memset(_bitrateDiffMS, 0, sizeof(_bitrateDiffMS));
+ memset(_bitrateArray, 0, sizeof(_bitrateArray));
+}
+
+void
+Bitrate::Init()
+{
+ _packetRate = 0;
+ _bitrate = 0;
+ _timeLastRateUpdate = 0;
+ _bytesCount = 0;
+ _packetCount = 0;
+ _bitrateNextIdx = 0;
+
+ memset(_packetRateArray, 0, sizeof(_packetRateArray));
+ memset(_bitrateDiffMS, 0, sizeof(_bitrateDiffMS));
+ memset(_bitrateArray, 0, sizeof(_bitrateArray));
+}
+
+void
+Bitrate::Update(const WebRtc_Word32 bytes)
+{
+ _bytesCount += bytes;
+ _packetCount++;
+}
+
+WebRtc_UWord32
+Bitrate::PacketRate() const
+{
+ return _packetRate;
+}
+
+WebRtc_UWord32
+Bitrate::BitrateLast() const
+{
+ return _bitrate;
+}
+
+WebRtc_UWord32
+Bitrate::BitrateNow() const
+{
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+ WebRtc_UWord32 diffMS = now -_timeLastRateUpdate;
+
+ if(diffMS > 10000) // 10 sec
+ {
+ // too high diff ignore
+ return _bitrate; // bits/s
+ }
+ WebRtc_UWord64 bitsSinceLastRateUpdate = 8*_bytesCount*1000;
+
+ // have to consider the time when the measurement was done
+ // ((bits/sec * sec) + (bits)) / sec
+ WebRtc_UWord64 bitrate = (((WebRtc_UWord64)_bitrate * 1000) + bitsSinceLastRateUpdate)/(1000+diffMS);
+ return (WebRtc_UWord32)bitrate;
+}
+
+void
+Bitrate::Process()
+{
+ // triggered by timer
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+ WebRtc_UWord32 diffMS = now -_timeLastRateUpdate;
+
+ if(diffMS > 100)
+ {
+ if(diffMS > 10000) // 10 sec
+ {
+ // too high diff ignore
+ _timeLastRateUpdate = now;
+ _bytesCount = 0;
+ _packetCount = 0;
+ return;
+ }
+ _packetRateArray[_bitrateNextIdx] = (_packetCount*1000)/diffMS;
+ _bitrateArray[_bitrateNextIdx] = 8*((_bytesCount*1000)/diffMS);
+ // will overflow at ~34 Mbit/s
+ _bitrateDiffMS[_bitrateNextIdx] = diffMS;
+ _bitrateNextIdx++;
+ if(_bitrateNextIdx >= 10)
+ {
+ _bitrateNextIdx = 0;
+ }
+
+ WebRtc_UWord32 sumDiffMS = 0;
+ WebRtc_UWord64 sumBitrateMS = 0;
+ WebRtc_UWord32 sumPacketrateMS = 0;
+ for(int i= 0; i <10; i++)
+ {
+ // sum of time
+ sumDiffMS += _bitrateDiffMS[i];
+ sumBitrateMS += _bitrateArray[i] * _bitrateDiffMS[i];
+ sumPacketrateMS += _packetRateArray[i] * _bitrateDiffMS[i];
+ }
+ _timeLastRateUpdate = now;
+ _bytesCount = 0;
+ _packetCount = 0;
+
+ _packetRate = sumPacketrateMS/sumDiffMS;
+ _bitrate = WebRtc_UWord32(sumBitrateMS / sumDiffMS);
+ }
+}
+
+BitRateStats::BitRateStats()
+ :_dataSamples(), _accumulatedBytes(0)
+{
+}
+
+BitRateStats::~BitRateStats()
+{
+ while (_dataSamples.size() > 0)
+ {
+ delete _dataSamples.front();
+ _dataSamples.pop_front();
+ }
+}
+
+void BitRateStats::Init()
+{
+ _accumulatedBytes = 0;
+ while (_dataSamples.size() > 0)
+ {
+ delete _dataSamples.front();
+ _dataSamples.pop_front();
+ }
+}
+
+void BitRateStats::Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs)
+{
+ // Find an empty slot for storing the new sample and at the same time
+ // accumulate the history.
+ _dataSamples.push_back(new DataTimeSizeTuple(packetSizeBytes, nowMs));
+ _accumulatedBytes += packetSizeBytes;
+ EraseOld(nowMs);
+}
+
+void BitRateStats::EraseOld(WebRtc_Word64 nowMs)
+{
+ while (_dataSamples.size() > 0)
+ {
+ if (nowMs - _dataSamples.front()->_timeCompleteMs >
+ BITRATE_AVERAGE_WINDOW)
+ {
+ // Delete old sample
+ _accumulatedBytes -= _dataSamples.front()->_sizeBytes;
+ delete _dataSamples.front();
+ _dataSamples.pop_front();
+ }
+ else
+ {
+ break;
+ }
+ }
+}
+
+WebRtc_UWord32 BitRateStats::BitRate(WebRtc_Word64 nowMs)
+{
+ // Calculate the average bit rate the past BITRATE_AVERAGE_WINDOW ms.
+ // Removes any old samples from the list.
+ EraseOld(nowMs);
+ WebRtc_Word64 timeOldest = nowMs;
+ if (_dataSamples.size() > 0)
+ {
+ timeOldest = _dataSamples.front()->_timeCompleteMs;
+ }
+ // Update average bit rate
+ float denom = static_cast<float>(nowMs - timeOldest);
+ if (nowMs == timeOldest)
+ {
+ // Calculate with a one second window when we haven't
+ // received more than one packet.
+ denom = 1000.0;
+ }
+ return static_cast<WebRtc_UWord32>(_accumulatedBytes * 8.0f * 1000.0f /
+ denom + 0.5f);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/bwe_defines.h b/trunk/src/modules/rtp_rtcp/source/bwe_defines.h
new file mode 100644
index 0000000..8fdc985
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/bwe_defines.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+
+#include "typedefs.h"
+
+#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
+#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
+
+namespace webrtc {
+enum BandwidthUsage
+{
+ kBwNormal,
+ kBwOverusing,
+ kBwUnderUsing
+};
+
+enum RateControlState
+{
+ kRcHold,
+ kRcIncrease,
+ kRcDecrease
+};
+
+enum RateControlRegion
+{
+ kRcNearMax,
+ kRcAboveMax,
+ kRcMaxUnknown
+};
+
+class RateControlInput
+{
+public:
+ RateControlInput(BandwidthUsage bwState,
+ WebRtc_UWord32 incomingBitRate,
+ double noiseVar) :
+ _bwState(bwState), _incomingBitRate(incomingBitRate), _noiseVar(noiseVar)
+ {};
+
+ BandwidthUsage _bwState;
+ WebRtc_UWord32 _incomingBitRate;
+ double _noiseVar;
+};
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc
new file mode 100644
index 0000000..749309b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dtmf_queue.h"
+
+#include <string.h> //memset
+
+namespace webrtc {
+DTMFqueue::DTMFqueue():
+ _DTMFCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _nextEmptyIndex(0)
+{
+ memset(_DTMFKey,0, sizeof(_DTMFKey));
+ memset(_DTMFLen,0, sizeof(_DTMFLen));
+ memset(_DTMFLevel,0, sizeof(_DTMFLevel));
+}
+
+DTMFqueue::~DTMFqueue()
+{
+ delete _DTMFCritsect;
+}
+
+WebRtc_Word32
+DTMFqueue::AddDTMF(WebRtc_UWord8 key, WebRtc_UWord16 len, WebRtc_UWord8 level)
+{
+ CriticalSectionScoped lock(_DTMFCritsect);
+
+ if(_nextEmptyIndex >= DTMF_OUTBAND_MAX)
+ {
+ return -1;
+ }
+ WebRtc_Word32 index = _nextEmptyIndex;
+ _DTMFKey[index] = key;
+ _DTMFLen[index] = len;
+ _DTMFLevel[index] = level;
+ _nextEmptyIndex++;
+ return 0;
+}
+
+WebRtc_Word8
+DTMFqueue::NextDTMF(WebRtc_UWord8* DTMFKey, WebRtc_UWord16* len, WebRtc_UWord8* level)
+{
+ CriticalSectionScoped lock(_DTMFCritsect);
+
+ if(!PendingDTMF())
+ {
+ return -1;
+ }
+ *DTMFKey=_DTMFKey[0];
+ *len=_DTMFLen[0];
+ *level=_DTMFLevel[0];
+
+ memmove(&(_DTMFKey[0]), &(_DTMFKey[1]), _nextEmptyIndex*sizeof(WebRtc_UWord8));
+ memmove(&(_DTMFLen[0]), &(_DTMFLen[1]), _nextEmptyIndex*sizeof(WebRtc_UWord16));
+ memmove(&(_DTMFLevel[0]), &(_DTMFLevel[1]), _nextEmptyIndex*sizeof(WebRtc_UWord8));
+
+ _nextEmptyIndex--;
+ return 0;
+}
+
+bool
+DTMFqueue::PendingDTMF()
+{
+ return(_nextEmptyIndex>0);
+}
+
+void
+DTMFqueue::ResetDTMF()
+{
+ _nextEmptyIndex = 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h
new file mode 100644
index 0000000..8451a21
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp_config.h"
+
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+class DTMFqueue
+{
+public:
+ DTMFqueue();
+ virtual ~DTMFqueue();
+
+ WebRtc_Word32 AddDTMF(WebRtc_UWord8 DTMFKey, WebRtc_UWord16 len, WebRtc_UWord8 level);
+ WebRtc_Word8 NextDTMF(WebRtc_UWord8* DTMFKey, WebRtc_UWord16 * len, WebRtc_UWord8 * level);
+ bool PendingDTMF();
+ void ResetDTMF();
+
+private:
+ CriticalSectionWrapper* _DTMFCritsect;
+ WebRtc_UWord8 _nextEmptyIndex;
+ WebRtc_UWord8 _DTMFKey[DTMF_OUTBAND_MAX];
+ WebRtc_UWord16 _DTMFLen[DTMF_OUTBAND_MAX];
+ WebRtc_UWord8 _DTMFLevel[DTMF_OUTBAND_MAX];
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h b/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h
new file mode 100644
index 0000000..699ee5a
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h
@@ -0,0 +1,25741 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+
+// Ensure the tables have internal linkage, as we only require them in one file.
+namespace
+{
+ const WebRtc_UWord8 mask10_1[2] =
+ {
+ 0xff, 0xc0
+ };
+
+ const WebRtc_UWord8 mask10_10[20] =
+ {
+ 0x4c, 0x00,
+ 0x51, 0x00,
+ 0xa0, 0x40,
+ 0x04, 0xc0,
+ 0x03, 0x80,
+ 0x86, 0x00,
+ 0x29, 0x00,
+ 0x42, 0x40,
+ 0x98, 0x00,
+ 0x30, 0x80
+ };
+
+ const WebRtc_UWord8 mask10_2[4] =
+ {
+ 0xaa, 0x80,
+ 0xd5, 0x40
+ };
+
+ const WebRtc_UWord8 mask10_3[6] =
+ {
+ 0xa4, 0x40,
+ 0xc9, 0x00,
+ 0x52, 0x80
+ };
+
+ const WebRtc_UWord8 mask10_4[8] =
+ {
+ 0xca, 0x00,
+ 0x32, 0x80,
+ 0xa1, 0x40,
+ 0x55, 0x00
+ };
+
+ const WebRtc_UWord8 mask10_5[10] =
+ {
+ 0xca, 0x00,
+ 0x32, 0x80,
+ 0xa1, 0x40,
+ 0x55, 0x00,
+ 0x08, 0xc0
+ };
+
+ const WebRtc_UWord8 mask10_6[12] =
+ {
+ 0x0e, 0x00,
+ 0x33, 0x00,
+ 0x10, 0xc0,
+ 0x45, 0x40,
+ 0x88, 0x80,
+ 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask10_7[14] =
+ {
+ 0x46, 0x00,
+ 0x33, 0x00,
+ 0x80, 0xc0,
+ 0x0c, 0x40,
+ 0x28, 0x80,
+ 0x94, 0x00,
+ 0xc1, 0x00
+ };
+
+ const WebRtc_UWord8 mask10_8[16] =
+ {
+ 0x2c, 0x00,
+ 0x81, 0x80,
+ 0xa0, 0x40,
+ 0x05, 0x40,
+ 0x18, 0x80,
+ 0xc2, 0x00,
+ 0x22, 0x80,
+ 0x50, 0x40
+ };
+
+ const WebRtc_UWord8 mask10_9[18] =
+ {
+ 0x4c, 0x00,
+ 0x23, 0x00,
+ 0x88, 0xc0,
+ 0x21, 0x40,
+ 0x52, 0x80,
+ 0x94, 0x00,
+ 0x26, 0x00,
+ 0x48, 0x40,
+ 0x91, 0x80
+ };
+
+ const WebRtc_UWord8 mask11_1[2] =
+ {
+ 0xff, 0xe0
+ };
+
+ const WebRtc_UWord8 mask11_10[20] =
+ {
+ 0x64, 0x40,
+ 0x51, 0x40,
+ 0xa9, 0x00,
+ 0x04, 0xc0,
+ 0xd0, 0x00,
+ 0x82, 0x40,
+ 0x21, 0x20,
+ 0x0c, 0x20,
+ 0x4a, 0x00,
+ 0x12, 0xa0
+ };
+
+ const WebRtc_UWord8 mask11_11[22] =
+ {
+ 0x46, 0x40,
+ 0x33, 0x20,
+ 0x99, 0x00,
+ 0x05, 0x80,
+ 0x80, 0xa0,
+ 0x84, 0x40,
+ 0x40, 0x60,
+ 0x0a, 0x80,
+ 0x68, 0x00,
+ 0x10, 0x20,
+ 0x30, 0x40
+ };
+
+ const WebRtc_UWord8 mask11_2[4] =
+ {
+ 0xec, 0xc0,
+ 0x9b, 0xa0
+ };
+
+ const WebRtc_UWord8 mask11_3[6] =
+ {
+ 0xca, 0xc0,
+ 0xf1, 0x40,
+ 0xb6, 0x20
+ };
+
+ const WebRtc_UWord8 mask11_4[8] =
+ {
+ 0xc4, 0xc0,
+ 0x31, 0x60,
+ 0x4b, 0x20,
+ 0x2c, 0xa0
+ };
+
+ const WebRtc_UWord8 mask11_5[10] =
+ {
+ 0x86, 0x80,
+ 0x23, 0x20,
+ 0x16, 0x20,
+ 0x4c, 0x20,
+ 0x41, 0xc0
+ };
+
+ const WebRtc_UWord8 mask11_6[12] =
+ {
+ 0x64, 0x40,
+ 0x51, 0x40,
+ 0x0c, 0xa0,
+ 0xa1, 0x20,
+ 0x12, 0xa0,
+ 0x8a, 0x40
+ };
+
+ const WebRtc_UWord8 mask11_7[14] =
+ {
+ 0x46, 0x40,
+ 0x33, 0x20,
+ 0x91, 0x80,
+ 0xa4, 0x20,
+ 0x50, 0xa0,
+ 0x84, 0xc0,
+ 0x09, 0x60
+ };
+
+ const WebRtc_UWord8 mask11_8[16] =
+ {
+ 0x0c, 0x80,
+ 0x80, 0x60,
+ 0xa0, 0x80,
+ 0x05, 0x40,
+ 0x43, 0x00,
+ 0x1a, 0x00,
+ 0x60, 0x20,
+ 0x14, 0x20
+ };
+
+ const WebRtc_UWord8 mask11_9[18] =
+ {
+ 0x46, 0x40,
+ 0x62, 0x60,
+ 0x8c, 0x00,
+ 0x01, 0x60,
+ 0x07, 0x80,
+ 0xa0, 0x80,
+ 0x18, 0xa0,
+ 0x91, 0x00,
+ 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask12_1[2] =
+ {
+ 0xff, 0xf0
+ };
+
+ const WebRtc_UWord8 mask12_10[20] =
+ {
+ 0x51, 0x40,
+ 0x45, 0x10,
+ 0x80, 0xd0,
+ 0x24, 0x20,
+ 0x0a, 0x20,
+ 0x00, 0xe0,
+ 0xb8, 0x00,
+ 0x09, 0x10,
+ 0x56, 0x00,
+ 0xa2, 0x80
+ };
+
+ const WebRtc_UWord8 mask12_11[22] =
+ {
+ 0x53, 0x60,
+ 0x21, 0x30,
+ 0x10, 0x90,
+ 0x00, 0x70,
+ 0x0c, 0x10,
+ 0x40, 0xc0,
+ 0x6a, 0x00,
+ 0x86, 0x00,
+ 0x24, 0x80,
+ 0x89, 0x00,
+ 0xc0, 0x20
+ };
+
+ const WebRtc_UWord8 mask12_12[24] =
+ {
+ 0x10, 0x60,
+ 0x02, 0x30,
+ 0x40, 0x50,
+ 0x21, 0x80,
+ 0x81, 0x10,
+ 0x14, 0x80,
+ 0x98, 0x00,
+ 0x08, 0x90,
+ 0x62, 0x00,
+ 0x24, 0x20,
+ 0x8a, 0x00,
+ 0x84, 0x40
+ };
+
+ const WebRtc_UWord8 mask12_2[4] =
+ {
+ 0xec, 0xc0,
+ 0x93, 0xb0
+ };
+
+ const WebRtc_UWord8 mask12_3[6] =
+ {
+ 0x9b, 0x80,
+ 0x4f, 0x10,
+ 0x3c, 0x60
+ };
+
+ const WebRtc_UWord8 mask12_4[8] =
+ {
+ 0x8b, 0x20,
+ 0x14, 0xb0,
+ 0x22, 0xd0,
+ 0x45, 0x50
+ };
+
+ const WebRtc_UWord8 mask12_5[10] =
+ {
+ 0x53, 0x60,
+ 0x64, 0x20,
+ 0x0c, 0xc0,
+ 0x82, 0xa0,
+ 0x09, 0x30
+ };
+
+ const WebRtc_UWord8 mask12_6[12] =
+ {
+ 0x51, 0x40,
+ 0xc5, 0x10,
+ 0x21, 0x80,
+ 0x12, 0x30,
+ 0x08, 0xe0,
+ 0x2e, 0x00
+ };
+
+ const WebRtc_UWord8 mask12_7[14] =
+ {
+ 0x53, 0x60,
+ 0x21, 0x30,
+ 0x90, 0x90,
+ 0x02, 0x50,
+ 0x06, 0xa0,
+ 0x2c, 0x00,
+ 0x88, 0x60
+ };
+
+ const WebRtc_UWord8 mask12_8[16] =
+ {
+ 0x20, 0x60,
+ 0x80, 0x30,
+ 0x42, 0x40,
+ 0x01, 0x90,
+ 0x14, 0x10,
+ 0x0a, 0x80,
+ 0x38, 0x00,
+ 0xc5, 0x00
+ };
+
+ const WebRtc_UWord8 mask12_9[18] =
+ {
+ 0x53, 0x60,
+ 0xe4, 0x20,
+ 0x24, 0x40,
+ 0xa1, 0x10,
+ 0x18, 0x30,
+ 0x03, 0x90,
+ 0x8a, 0x10,
+ 0x04, 0x90,
+ 0x00, 0xe0
+ };
+
+ const WebRtc_UWord8 mask13_1[2] =
+ {
+ 0xff, 0xf8
+ };
+
+ const WebRtc_UWord8 mask13_10[20] =
+ {
+ 0xd1, 0x00,
+ 0x44, 0x50,
+ 0x10, 0x98,
+ 0xa0, 0x50,
+ 0x4a, 0x08,
+ 0x40, 0x30,
+ 0x80, 0x28,
+ 0x0c, 0x90,
+ 0x05, 0x88,
+ 0x62, 0x20
+ };
+
+ const WebRtc_UWord8 mask13_11[22] =
+ {
+ 0x51, 0x20,
+ 0x22, 0x10,
+ 0x13, 0x40,
+ 0x25, 0x00,
+ 0x18, 0x18,
+ 0x0a, 0x20,
+ 0x88, 0x88,
+ 0x06, 0x80,
+ 0xe0, 0x20,
+ 0x84, 0x40,
+ 0x44, 0x18
+ };
+
+ const WebRtc_UWord8 mask13_12[24] =
+ {
+ 0x28, 0x28,
+ 0x84, 0x50,
+ 0x60, 0x40,
+ 0x05, 0x48,
+ 0x02, 0x98,
+ 0x01, 0x30,
+ 0x48, 0x10,
+ 0x24, 0x80,
+ 0x94, 0x00,
+ 0x8a, 0x00,
+ 0x11, 0x80,
+ 0x52, 0x20
+ };
+
+ const WebRtc_UWord8 mask13_13[26] =
+ {
+ 0x51, 0x20,
+ 0x66, 0x40,
+ 0x05, 0x48,
+ 0x81, 0x20,
+ 0x94, 0x00,
+ 0x30, 0x80,
+ 0x21, 0x10,
+ 0x03, 0xc0,
+ 0xe8, 0x00,
+ 0x0a, 0x10,
+ 0x80, 0x18,
+ 0x04, 0x90,
+ 0x08, 0xa8
+ };
+
+ const WebRtc_UWord8 mask13_2[4] =
+ {
+ 0xec, 0xc0,
+ 0x1b, 0x38
+ };
+
+ const WebRtc_UWord8 mask13_3[6] =
+ {
+ 0x99, 0xb0,
+ 0x46, 0xd8,
+ 0x37, 0x28
+ };
+
+ const WebRtc_UWord8 mask13_4[8] =
+ {
+ 0x49, 0xb0,
+ 0x26, 0xd0,
+ 0x85, 0x68,
+ 0x52, 0x58
+ };
+
+ const WebRtc_UWord8 mask13_5[10] =
+ {
+ 0x51, 0x30,
+ 0x66, 0x40,
+ 0x0c, 0x68,
+ 0xa1, 0xc0,
+ 0x22, 0x98
+ };
+
+ const WebRtc_UWord8 mask13_6[12] =
+ {
+ 0xd1, 0x20,
+ 0x46, 0xd0,
+ 0x15, 0x48,
+ 0x21, 0x70,
+ 0x28, 0xc8,
+ 0xaa, 0x20
+ };
+
+ const WebRtc_UWord8 mask13_7[14] =
+ {
+ 0x59, 0x20,
+ 0x26, 0x50,
+ 0xb1, 0x40,
+ 0x2b, 0x08,
+ 0x14, 0xc8,
+ 0xc8, 0x88,
+ 0x84, 0xb0
+ };
+
+ const WebRtc_UWord8 mask13_8[16] =
+ {
+ 0x80, 0xa8,
+ 0x30, 0x90,
+ 0x16, 0x08,
+ 0x03, 0x30,
+ 0x44, 0x60,
+ 0x08, 0x18,
+ 0xd8, 0x00,
+ 0xa1, 0x40
+ };
+
+ const WebRtc_UWord8 mask13_9[18] =
+ {
+ 0x59, 0x20,
+ 0x66, 0x40,
+ 0x14, 0x40,
+ 0x21, 0x48,
+ 0x02, 0xc8,
+ 0x94, 0x10,
+ 0x80, 0xa8,
+ 0x0a, 0x90,
+ 0x40, 0x18
+ };
+
+ const WebRtc_UWord8 mask14_1[2] =
+ {
+ 0xff, 0xfc
+ };
+
+ const WebRtc_UWord8 mask14_10[20] =
+ {
+ 0xc0, 0xd4,
+ 0x1d, 0x40,
+ 0xd4, 0x08,
+ 0x02, 0x60,
+ 0x04, 0x28,
+ 0x20, 0x98,
+ 0x40, 0x44,
+ 0x08, 0x84,
+ 0x68, 0x00,
+ 0x23, 0x10
+ };
+
+ const WebRtc_UWord8 mask14_11[22] =
+ {
+ 0x62, 0xd0,
+ 0x35, 0x20,
+ 0x14, 0x14,
+ 0xc5, 0x08,
+ 0x22, 0x0c,
+ 0x88, 0xb8,
+ 0x42, 0x54,
+ 0x28, 0xa4,
+ 0x94, 0x20,
+ 0x1b, 0x04,
+ 0x22, 0xc0
+ };
+
+ const WebRtc_UWord8 mask14_12[24] =
+ {
+ 0x81, 0x04,
+ 0x40, 0x68,
+ 0x90, 0x24,
+ 0x28, 0x28,
+ 0x52, 0x10,
+ 0x41, 0x88,
+ 0x09, 0x30,
+ 0x48, 0x44,
+ 0x04, 0x44,
+ 0x0e, 0x80,
+ 0xa5, 0x90,
+ 0x12, 0x0c
+ };
+
+ const WebRtc_UWord8 mask14_13[26] =
+ {
+ 0x62, 0x54,
+ 0x34, 0x60,
+ 0x48, 0x04,
+ 0x00, 0xac,
+ 0x28, 0x08,
+ 0x81, 0x08,
+ 0x23, 0x04,
+ 0x06, 0x80,
+ 0x80, 0x14,
+ 0x30, 0x10,
+ 0x8c, 0x20,
+ 0x54, 0x00,
+ 0x80, 0xc0
+ };
+
+ const WebRtc_UWord8 mask14_14[28] =
+ {
+ 0x40, 0x54,
+ 0x15, 0x40,
+ 0xc0, 0x04,
+ 0x28, 0x10,
+ 0x05, 0x0c,
+ 0x64, 0x80,
+ 0x81, 0x80,
+ 0x10, 0x98,
+ 0x84, 0x20,
+ 0x12, 0x30,
+ 0x62, 0x00,
+ 0x28, 0x60,
+ 0x0e, 0x08,
+ 0x10, 0x84
+ };
+
+ const WebRtc_UWord8 mask14_2[4] =
+ {
+ 0xec, 0xe8,
+ 0x3b, 0x9c
+ };
+
+ const WebRtc_UWord8 mask14_3[6] =
+ {
+ 0xac, 0xd8,
+ 0x55, 0x6c,
+ 0x27, 0xb4
+ };
+
+ const WebRtc_UWord8 mask14_4[8] =
+ {
+ 0x2c, 0xd8,
+ 0x93, 0x68,
+ 0x1a, 0xb4,
+ 0x47, 0x2c
+ };
+
+ const WebRtc_UWord8 mask14_5[10] =
+ {
+ 0x64, 0xd8,
+ 0xa5, 0x68,
+ 0x52, 0xb4,
+ 0x1d, 0xa8,
+ 0x9c, 0x54
+ };
+
+ const WebRtc_UWord8 mask14_6[12] =
+ {
+ 0x4a, 0x54,
+ 0x95, 0x48,
+ 0x14, 0xb4,
+ 0x51, 0xa8,
+ 0x22, 0x6c,
+ 0x88, 0x8c
+ };
+
+ const WebRtc_UWord8 mask14_7[14] =
+ {
+ 0x62, 0x54,
+ 0xb9, 0x20,
+ 0x18, 0xb4,
+ 0x54, 0x98,
+ 0x06, 0x6c,
+ 0x85, 0x54,
+ 0xaa, 0x88
+ };
+
+ const WebRtc_UWord8 mask14_8[16] =
+ {
+ 0xc0, 0x14,
+ 0x41, 0x60,
+ 0x88, 0x30,
+ 0x20, 0xa4,
+ 0x0a, 0x48,
+ 0x04, 0x98,
+ 0x94, 0x40,
+ 0x72, 0x00
+ };
+
+ const WebRtc_UWord8 mask14_9[18] =
+ {
+ 0xa2, 0x54,
+ 0x34, 0x60,
+ 0x4a, 0x24,
+ 0x20, 0xa8,
+ 0x11, 0x84,
+ 0x49, 0x08,
+ 0x86, 0x0c,
+ 0x20, 0xd4,
+ 0x88, 0x48
+ };
+
+ const WebRtc_UWord8 mask15_1[2] =
+ {
+ 0xff, 0xfe
+ };
+
+ const WebRtc_UWord8 mask15_10[20] =
+ {
+ 0xc0, 0xa0,
+ 0x15, 0x56,
+ 0x74, 0x40,
+ 0x00, 0x9c,
+ 0x01, 0x2c,
+ 0x44, 0x92,
+ 0x88, 0x50,
+ 0x20, 0xa4,
+ 0xaa, 0x04,
+ 0x02, 0x62
+ };
+
+ const WebRtc_UWord8 mask15_11[22] =
+ {
+ 0x62, 0x22,
+ 0xf1, 0x10,
+ 0x10, 0x0e,
+ 0x10, 0xb0,
+ 0x24, 0x24,
+ 0x01, 0x12,
+ 0x00, 0xc4,
+ 0x04, 0xa2,
+ 0x02, 0x58,
+ 0x2b, 0x00,
+ 0x98, 0x40
+ };
+
+ const WebRtc_UWord8 mask15_12[24] =
+ {
+ 0x88, 0x90,
+ 0x40, 0x54,
+ 0x82, 0x62,
+ 0x21, 0xa4,
+ 0x10, 0x64,
+ 0x44, 0x0a,
+ 0x10, 0xc8,
+ 0x4d, 0x2a,
+ 0x38, 0x02,
+ 0x17, 0x48,
+ 0x90, 0x84,
+ 0x72, 0x14
+ };
+
+ const WebRtc_UWord8 mask15_13[26] =
+ {
+ 0x62, 0xa2,
+ 0x34, 0x44,
+ 0x40, 0x4a,
+ 0xc4, 0x04,
+ 0x08, 0x60,
+ 0x94, 0x12,
+ 0x88, 0xc0,
+ 0x21, 0x32,
+ 0xc1, 0x40,
+ 0x10, 0x68,
+ 0x06, 0x90,
+ 0x59, 0x00,
+ 0x0a, 0x0c
+ };
+
+ const WebRtc_UWord8 mask15_14[28] =
+ {
+ 0x40, 0x82,
+ 0x15, 0x54,
+ 0x88, 0x12,
+ 0xc0, 0x10,
+ 0x80, 0xa0,
+ 0x01, 0x22,
+ 0x40, 0x2c,
+ 0x22, 0x02,
+ 0x90, 0x04,
+ 0x12, 0x40,
+ 0x5d, 0x00,
+ 0x20, 0x54,
+ 0x86, 0x08,
+ 0x28, 0x88
+ };
+
+ const WebRtc_UWord8 mask15_15[30] =
+ {
+ 0x62, 0x22,
+ 0x31, 0x10,
+ 0x58, 0x00,
+ 0x01, 0x12,
+ 0x88, 0x20,
+ 0x44, 0x02,
+ 0x29, 0x04,
+ 0x82, 0xa0,
+ 0x0a, 0x1a,
+ 0x11, 0xe0,
+ 0x84, 0x04,
+ 0x86, 0x40,
+ 0x00, 0x86,
+ 0x44, 0x48,
+ 0x10, 0x98
+ };
+
+ const WebRtc_UWord8 mask15_2[4] =
+ {
+ 0xec, 0xea,
+ 0xbb, 0x9c
+ };
+
+ const WebRtc_UWord8 mask15_3[6] =
+ {
+ 0xac, 0x92,
+ 0x55, 0x4a,
+ 0x43, 0x36
+ };
+
+ const WebRtc_UWord8 mask15_4[8] =
+ {
+ 0x25, 0xaa,
+ 0x95, 0x54,
+ 0x1a, 0x6a,
+ 0x43, 0xd4
+ };
+
+ const WebRtc_UWord8 mask15_5[10] =
+ {
+ 0x64, 0xa2,
+ 0x25, 0x54,
+ 0x49, 0x68,
+ 0x53, 0x90,
+ 0x8e, 0x30
+ };
+
+ const WebRtc_UWord8 mask15_6[12] =
+ {
+ 0x62, 0x8a,
+ 0x15, 0x54,
+ 0x4c, 0x46,
+ 0x52, 0x94,
+ 0x23, 0x64,
+ 0x8a, 0x58
+ };
+
+ const WebRtc_UWord8 mask15_7[14] =
+ {
+ 0x62, 0xa2,
+ 0xb1, 0x14,
+ 0x18, 0x6a,
+ 0x44, 0xd4,
+ 0x13, 0x64,
+ 0x49, 0x1a,
+ 0x86, 0x8c
+ };
+
+ const WebRtc_UWord8 mask15_8[16] =
+ {
+ 0x90, 0x22,
+ 0x09, 0x50,
+ 0x00, 0x6a,
+ 0x20, 0x34,
+ 0x14, 0x44,
+ 0xc2, 0x10,
+ 0x00, 0xc6,
+ 0x65, 0x80
+ };
+
+ const WebRtc_UWord8 mask15_9[18] =
+ {
+ 0x62, 0x22,
+ 0x24, 0x44,
+ 0xc0, 0x50,
+ 0x03, 0x0c,
+ 0x16, 0x28,
+ 0x89, 0x00,
+ 0x82, 0x90,
+ 0x08, 0xa4,
+ 0x90, 0x48
+ };
+
+ const WebRtc_UWord8 mask16_1[2] =
+ {
+ 0xff, 0xff
+ };
+
+ const WebRtc_UWord8 mask16_10[20] =
+ {
+ 0x45, 0x51,
+ 0x10, 0xa2,
+ 0x01, 0x25,
+ 0x0b, 0x42,
+ 0xd8, 0x20,
+ 0x82, 0x8c,
+ 0x24, 0x4a,
+ 0x38, 0x18,
+ 0x2a, 0x25,
+ 0x84, 0x92
+ };
+
+ const WebRtc_UWord8 mask16_11[22] =
+ {
+ 0x55, 0x55,
+ 0x2a, 0x22,
+ 0x31, 0x11,
+ 0x83, 0x42,
+ 0x06, 0x98,
+ 0x40, 0xe1,
+ 0x2c, 0x44,
+ 0xd8, 0x28,
+ 0x92, 0x81,
+ 0x84, 0x32,
+ 0x68, 0x0c
+ };
+
+ const WebRtc_UWord8 mask16_12[24] =
+ {
+ 0x84, 0x31,
+ 0x18, 0xa2,
+ 0x4e, 0x01,
+ 0x44, 0xc8,
+ 0x0e, 0x90,
+ 0x20, 0xcc,
+ 0x93, 0x40,
+ 0x2d, 0x10,
+ 0x31, 0x44,
+ 0xc0, 0x23,
+ 0x11, 0x25,
+ 0xe8, 0x80
+ };
+
+ const WebRtc_UWord8 mask16_13[26] =
+ {
+ 0x45, 0x15,
+ 0x22, 0x22,
+ 0x96, 0x0c,
+ 0x0c, 0x50,
+ 0x62, 0x04,
+ 0x49, 0x06,
+ 0x11, 0x82,
+ 0x12, 0x38,
+ 0x40, 0x71,
+ 0xa8, 0x8a,
+ 0x08, 0xa1,
+ 0xa0, 0xc0,
+ 0xc5, 0x10
+ };
+
+ const WebRtc_UWord8 mask16_14[28] =
+ {
+ 0x45, 0x51,
+ 0x22, 0x0a,
+ 0x84, 0xd0,
+ 0x0c, 0x8a,
+ 0x18, 0x06,
+ 0x30, 0x03,
+ 0x61, 0x08,
+ 0x40, 0x11,
+ 0x10, 0x2c,
+ 0x09, 0x60,
+ 0x00, 0x94,
+ 0x52, 0x40,
+ 0xa4, 0x24,
+ 0x82, 0x88
+ };
+
+ const WebRtc_UWord8 mask16_15[30] =
+ {
+ 0x55, 0x11,
+ 0x22, 0x22,
+ 0x11, 0x11,
+ 0x80, 0x45,
+ 0x20, 0x1a,
+ 0x08, 0x68,
+ 0x22, 0x84,
+ 0x48, 0x09,
+ 0x07, 0x01,
+ 0x94, 0x20,
+ 0x82, 0x06,
+ 0x60, 0x48,
+ 0x89, 0x80,
+ 0x00, 0x8e,
+ 0x18, 0x22
+ };
+
+ const WebRtc_UWord8 mask16_16[32] =
+ {
+ 0xa4, 0x10,
+ 0x01, 0x2a,
+ 0x06, 0x42,
+ 0x08, 0x68,
+ 0x81, 0x90,
+ 0x00, 0xf0,
+ 0x50, 0x05,
+ 0x20, 0x51,
+ 0x43, 0x08,
+ 0x68, 0x80,
+ 0x80, 0x0b,
+ 0x10, 0x4c,
+ 0x12, 0x30,
+ 0x40, 0x85,
+ 0x0e, 0x04,
+ 0x18, 0x12
+ };
+
+ const WebRtc_UWord8 mask16_2[4] =
+ {
+ 0xae, 0xae,
+ 0x79, 0x79
+ };
+
+ const WebRtc_UWord8 mask16_3[6] =
+ {
+ 0xad, 0x2d,
+ 0x76, 0x36,
+ 0x26, 0xdb
+ };
+
+ const WebRtc_UWord8 mask16_4[8] =
+ {
+ 0x55, 0x55,
+ 0xaa, 0xaa,
+ 0x35, 0x35,
+ 0xca, 0xca
+ };
+
+ const WebRtc_UWord8 mask16_5[10] =
+ {
+ 0x55, 0x55,
+ 0x2a, 0x2a,
+ 0x24, 0x25,
+ 0x84, 0xc8,
+ 0x10, 0xb6
+ };
+
+ const WebRtc_UWord8 mask16_6[12] =
+ {
+ 0x51, 0x51,
+ 0x0a, 0x2a,
+ 0xa2, 0x15,
+ 0x84, 0x4a,
+ 0x30, 0x92,
+ 0x04, 0xac
+ };
+
+ const WebRtc_UWord8 mask16_7[14] =
+ {
+ 0x45, 0x51,
+ 0x22, 0x2a,
+ 0x91, 0x11,
+ 0x2e, 0x08,
+ 0x48, 0x34,
+ 0x90, 0x29,
+ 0x09, 0x86
+ };
+
+ const WebRtc_UWord8 mask16_8[16] =
+ {
+ 0x20, 0x54,
+ 0x18, 0x88,
+ 0x84, 0x07,
+ 0x60, 0x48,
+ 0x12, 0x82,
+ 0x81, 0x41,
+ 0x40, 0x62,
+ 0x16, 0x30
+ };
+
+ const WebRtc_UWord8 mask16_9[18] =
+ {
+ 0x55, 0x51,
+ 0x22, 0x2a,
+ 0x05, 0x85,
+ 0x09, 0x4a,
+ 0x84, 0x32,
+ 0xc0, 0x0d,
+ 0x20, 0xa6,
+ 0x1a, 0x09,
+ 0x44, 0x64
+ };
+
+ const WebRtc_UWord8 mask17_1[6] =
+ {
+ 0xff, 0xff, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_10[60] =
+ {
+ 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
+ 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
+ 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
+ 0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
+ 0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
+ 0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
+ 0xc8, 0x02, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_11[66] =
+ {
+ 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
+ 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
+ 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
+ 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
+ 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_12[72] =
+ {
+ 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
+ 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
+ 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
+ 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
+ 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
+ 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x35, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_13[78] =
+ {
+ 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
+ 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
+ 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
+ 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
+ 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
+ 0x83, 0x34, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_14[84] =
+ {
+ 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
+ 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
+ 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
+ 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
+ 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
+ 0xb0, 0xde, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_15[90] =
+ {
+ 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
+ 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
+ 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
+ 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
+ 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
+ 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_16[96] =
+ {
+ 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
+ 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
+ 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
+ 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
+ 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
+ 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
+ 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x0a, 0x1c, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_17[102] =
+ {
+ 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
+ 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
+ 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
+ 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0x4c, 0x00, 0x00, 0x00, 0x00,
+ 0x8a, 0x66, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x91, 0x00, 0x00, 0x00, 0x00,
+ 0x68, 0x42, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0xa4, 0x00, 0x00, 0x00, 0x00,
+ 0x43, 0x13, 0x00, 0x00, 0x00, 0x00,
+ 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x1c, 0x88, 0x80, 0x00, 0x00, 0x00,
+ 0x3c, 0x09, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_2[12] =
+ {
+ 0xce, 0xce, 0x00, 0x00, 0x00, 0x00,
+ 0xb9, 0x39, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_3[18] =
+ {
+ 0xcd, 0xcc, 0x00, 0x00, 0x00, 0x00,
+ 0x97, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0xb8, 0xd1, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_4[24] =
+ {
+ 0xca, 0xec, 0x00, 0x00, 0x00, 0x00,
+ 0xa9, 0x67, 0x00, 0x00, 0x00, 0x00,
+ 0x3a, 0xb1, 0x80, 0x00, 0x00, 0x00,
+ 0x55, 0x5a, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_5[30] =
+ {
+ 0x55, 0x44, 0x80, 0x00, 0x00, 0x00,
+ 0x2a, 0x66, 0x00, 0x00, 0x00, 0x00,
+ 0x25, 0xa1, 0x80, 0x00, 0x00, 0x00,
+ 0xe2, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x99, 0x98, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_6[36] =
+ {
+ 0xd1, 0x4c, 0x00, 0x00, 0x00, 0x00,
+ 0xa2, 0xc5, 0x00, 0x00, 0x00, 0x00,
+ 0x95, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0xca, 0x0a, 0x80, 0x00, 0x00, 0x00,
+ 0xa4, 0xaa, 0x00, 0x00, 0x00, 0x00,
+ 0x78, 0x15, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_7[42] =
+ {
+ 0x15, 0x44, 0x80, 0x00, 0x00, 0x00,
+ 0x8a, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x85, 0x91, 0x00, 0x00, 0x00, 0x00,
+ 0x32, 0x0a, 0x80, 0x00, 0x00, 0x00,
+ 0x58, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x43, 0xc8, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_8[48] =
+ {
+ 0x64, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0xa2, 0xc2, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x60, 0x80, 0x00, 0x00, 0x00,
+ 0x4a, 0x85, 0x00, 0x00, 0x00, 0x00,
+ 0x38, 0x4c, 0x00, 0x00, 0x00, 0x00,
+ 0x89, 0x29, 0x00, 0x00, 0x00, 0x00,
+ 0x07, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x94, 0xb0, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask17_9[54] =
+ {
+ 0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
+ 0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
+ 0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
+ 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
+ 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
+ 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_1[6] =
+ {
+ 0xff, 0xff, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_10[60] =
+ {
+ 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
+ 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
+ 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
+ 0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
+ 0x2b, 0x15, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
+ 0x25, 0x92, 0xc0, 0x00, 0x00, 0x00,
+ 0xfd, 0x9d, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_11[66] =
+ {
+ 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
+ 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
+ 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
+ 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
+ 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x89, 0x44, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_12[72] =
+ {
+ 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
+ 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
+ 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
+ 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
+ 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
+ 0x5b, 0x0c, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_13[78] =
+ {
+ 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
+ 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
+ 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
+ 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
+ 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_14[84] =
+ {
+ 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
+ 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
+ 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
+ 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
+ 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
+ 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
+ 0x7f, 0x4f, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_15[90] =
+ {
+ 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
+ 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
+ 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
+ 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
+ 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_16[96] =
+ {
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
+ 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
+ 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
+ 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
+ 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
+ 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0xef, 0xf2, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_17[102] =
+ {
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
+ 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
+ 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
+ 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
+ 0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
+ 0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
+ 0x09, 0x04, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_18[108] =
+ {
+ 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
+ 0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
+ 0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
+ 0x09, 0x04, 0x80, 0x00, 0x00, 0x00,
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
+ 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
+ 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
+ 0xd0, 0x03, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_2[12] =
+ {
+ 0xce, 0x67, 0x00, 0x00, 0x00, 0x00,
+ 0x39, 0x9c, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_3[18] =
+ {
+ 0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x15, 0x80, 0x00, 0x00, 0x00,
+ 0x92, 0xc9, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_4[24] =
+ {
+ 0xec, 0x76, 0x00, 0x00, 0x00, 0x00,
+ 0x67, 0x33, 0x80, 0x00, 0x00, 0x00,
+ 0xb1, 0xd8, 0xc0, 0x00, 0x00, 0x00,
+ 0x5a, 0xad, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_5[30] =
+ {
+ 0x4c, 0xa6, 0x40, 0x00, 0x00, 0x00,
+ 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
+ 0x19, 0xd0, 0xc0, 0x00, 0x00, 0x00,
+ 0x9c, 0x89, 0x40, 0x00, 0x00, 0x00,
+ 0xe3, 0x4c, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_6[36] =
+ {
+ 0xcc, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x62, 0x80, 0x00, 0x00, 0x00,
+ 0xb0, 0x98, 0x40, 0x00, 0x00, 0x00,
+ 0x8a, 0x85, 0x40, 0x00, 0x00, 0x00,
+ 0x29, 0x53, 0x00, 0x00, 0x00, 0x00,
+ 0xa6, 0x0a, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_7[42] =
+ {
+ 0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
+ 0x23, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
+ 0x0a, 0x85, 0x40, 0x00, 0x00, 0x00,
+ 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x0b, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0xe0, 0x64, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_8[48] =
+ {
+ 0x16, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0xc2, 0x61, 0x00, 0x00, 0x00, 0x00,
+ 0x60, 0xb0, 0x40, 0x00, 0x00, 0x00,
+ 0x85, 0x42, 0x80, 0x00, 0x00, 0x00,
+ 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x29, 0x14, 0x80, 0x00, 0x00, 0x00,
+ 0x11, 0x88, 0xc0, 0x00, 0x00, 0x00,
+ 0xb0, 0x58, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask18_9[54] =
+ {
+ 0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
+ 0x66, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x90, 0x49, 0x40, 0x00, 0x00, 0x00,
+ 0x01, 0xa5, 0x80, 0x00, 0x00, 0x00,
+ 0x0e, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x13, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0x20, 0xd0, 0x40, 0x00, 0x00, 0x00,
+ 0xc2, 0x51, 0x00, 0x00, 0x00, 0x00,
+ 0x29, 0x0c, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_1[6] =
+ {
+ 0xff, 0xff, 0xe0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_10[60] =
+ {
+ 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
+ 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
+ 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
+ 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
+ 0xcc, 0x75, 0x00, 0x00, 0x00, 0x00,
+ 0x2b, 0x19, 0xc0, 0x00, 0x00, 0x00,
+ 0x32, 0xd2, 0x60, 0x00, 0x00, 0x00,
+ 0x25, 0x8e, 0xa0, 0x00, 0x00, 0x00,
+ 0x50, 0x88, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_11[66] =
+ {
+ 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
+ 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
+ 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
+ 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
+ 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
+ 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
+ 0x89, 0x70, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_12[72] =
+ {
+ 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
+ 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
+ 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
+ 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
+ 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
+ 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
+ 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
+ 0x90, 0xc8, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_13[78] =
+ {
+ 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
+ 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
+ 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
+ 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
+ 0x34, 0x60, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_14[84] =
+ {
+ 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
+ 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
+ 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
+ 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
+ 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
+ 0x6e, 0x27, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_15[90] =
+ {
+ 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
+ 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
+ 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
+ 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
+ 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_16[96] =
+ {
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
+ 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
+ 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
+ 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
+ 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
+ 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
+ 0x7e, 0x75, 0xe0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_17[102] =
+ {
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
+ 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
+ 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
+ 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
+ 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
+ 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
+ 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
+ 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
+ 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_18[108] =
+ {
+ 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
+ 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
+ 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
+ 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
+ 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
+ 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
+ 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
+ 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
+ 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x51, 0x97, 0x20, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_19[114] =
+ {
+ 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
+ 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
+ 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
+ 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
+ 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
+ 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x28, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x50, 0x20, 0x00, 0x00, 0x00,
+ 0x42, 0x82, 0x60, 0x00, 0x00, 0x00,
+ 0xa4, 0x01, 0xc0, 0x00, 0x00, 0x00,
+ 0x13, 0x43, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x94, 0x80, 0x00, 0x00, 0x00,
+ 0x88, 0xa1, 0x20, 0x00, 0x00, 0x00,
+ 0x09, 0x4c, 0x00, 0x00, 0x00, 0x00,
+ 0xcd, 0x98, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_2[12] =
+ {
+ 0xce, 0x77, 0x00, 0x00, 0x00, 0x00,
+ 0x39, 0xcc, 0xe0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_3[18] =
+ {
+ 0xcc, 0x67, 0x00, 0x00, 0x00, 0x00,
+ 0x27, 0x2c, 0xc0, 0x00, 0x00, 0x00,
+ 0x92, 0xd2, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_4[24] =
+ {
+ 0xec, 0x73, 0x00, 0x00, 0x00, 0x00,
+ 0x67, 0x19, 0xc0, 0x00, 0x00, 0x00,
+ 0xb1, 0xcc, 0x60, 0x00, 0x00, 0x00,
+ 0x5a, 0x96, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_5[30] =
+ {
+ 0x4c, 0xe7, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x31, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0xcc, 0x60, 0x00, 0x00, 0x00,
+ 0x92, 0xa6, 0xa0, 0x00, 0x00, 0x00,
+ 0xb8, 0x99, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_6[36] =
+ {
+ 0x4c, 0x36, 0x00, 0x00, 0x00, 0x00,
+ 0x45, 0x68, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0xd0, 0x60, 0x00, 0x00, 0x00,
+ 0x8a, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0x26, 0x0b, 0x40, 0x00, 0x00, 0x00,
+ 0x95, 0x45, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_7[42] =
+ {
+ 0xc4, 0xa3, 0x00, 0x00, 0x00, 0x00,
+ 0x23, 0x19, 0x80, 0x00, 0x00, 0x00,
+ 0x91, 0x1c, 0x20, 0x00, 0x00, 0x00,
+ 0x4a, 0x82, 0xa0, 0x00, 0x00, 0x00,
+ 0x34, 0x49, 0x40, 0x00, 0x00, 0x00,
+ 0x8b, 0x4a, 0x00, 0x00, 0x00, 0x00,
+ 0xc8, 0x24, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_8[48] =
+ {
+ 0x16, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0xc2, 0x44, 0xc0, 0x00, 0x00, 0x00,
+ 0x60, 0xe8, 0x20, 0x00, 0x00, 0x00,
+ 0x85, 0x12, 0x60, 0x00, 0x00, 0x00,
+ 0xcc, 0x21, 0x40, 0x00, 0x00, 0x00,
+ 0x29, 0x63, 0x00, 0x00, 0x00, 0x00,
+ 0x11, 0x98, 0xc0, 0x00, 0x00, 0x00,
+ 0xb0, 0x0c, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask19_9[54] =
+ {
+ 0x44, 0xa7, 0x00, 0x00, 0x00, 0x00,
+ 0x66, 0x70, 0x80, 0x00, 0x00, 0x00,
+ 0x12, 0xc0, 0xe0, 0x00, 0x00, 0x00,
+ 0xc3, 0x10, 0xa0, 0x00, 0x00, 0x00,
+ 0x8c, 0x29, 0x40, 0x00, 0x00, 0x00,
+ 0x11, 0x5b, 0x00, 0x00, 0x00, 0x00,
+ 0x21, 0x93, 0x40, 0x00, 0x00, 0x00,
+ 0xa2, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x18, 0x0c, 0xe0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask1_1[2] =
+ {
+ 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_1[6] =
+ {
+ 0xff, 0xff, 0xf0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_10[60] =
+ {
+ 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
+ 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
+ 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
+ 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
+ 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_11[66] =
+ {
+ 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
+ 0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
+ 0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
+ 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
+ 0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
+ 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
+ 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
+ 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_12[72] =
+ {
+ 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
+ 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
+ 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
+ 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
+ 0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
+ 0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
+ 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
+ 0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
+ 0xf5, 0xdc, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_13[78] =
+ {
+ 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
+ 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
+ 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
+ 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
+ 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
+ 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
+ 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_14[84] =
+ {
+ 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
+ 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
+ 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
+ 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
+ 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
+ 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
+ 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
+ 0x56, 0x3e, 0x20, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_15[90] =
+ {
+ 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
+ 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
+ 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
+ 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
+ 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
+ 0x50, 0x54, 0x10, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_16[96] =
+ {
+ 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
+ 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
+ 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
+ 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
+ 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
+ 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
+ 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
+ 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
+ 0x28, 0x1c, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_17[102] =
+ {
+ 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
+ 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
+ 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
+ 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
+ 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
+ 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
+ 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
+ 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
+ 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
+ 0x19, 0x86, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_18[108] =
+ {
+ 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
+ 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
+ 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
+ 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
+ 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
+ 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
+ 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
+ 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
+ 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
+ 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
+ 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
+ 0x21, 0x7b, 0xf0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_19[114] =
+ {
+ 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
+ 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
+ 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
+ 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
+ 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
+ 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
+ 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
+ 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
+ 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
+ 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
+ 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
+ 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
+ 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_2[12] =
+ {
+ 0xee, 0x3b, 0x80, 0x00, 0x00, 0x00,
+ 0x99, 0xe6, 0x70, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_20[120] =
+ {
+ 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
+ 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
+ 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
+ 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
+ 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00,
+ 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
+ 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
+ 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
+ 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
+ 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
+ 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
+ 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
+ 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
+ 0xf7, 0x8d, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_3[18] =
+ {
+ 0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
+ 0x55, 0x95, 0x60, 0x00, 0x00, 0x00,
+ 0xb1, 0x6a, 0x30, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_4[24] =
+ {
+ 0xe6, 0x39, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x8c, 0xe0, 0x00, 0x00, 0x00,
+ 0x98, 0xe6, 0x30, 0x00, 0x00, 0x00,
+ 0x2d, 0x4b, 0x50, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_5[30] =
+ {
+ 0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
+ 0x63, 0x98, 0xe0, 0x00, 0x00, 0x00,
+ 0x98, 0xe5, 0x30, 0x00, 0x00, 0x00,
+ 0x2b, 0x53, 0x50, 0x00, 0x00, 0x00,
+ 0xb4, 0x5c, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_6[36] =
+ {
+ 0x4c, 0x1b, 0x00, 0x00, 0x00, 0x00,
+ 0x51, 0x34, 0x40, 0x00, 0x00, 0x00,
+ 0x20, 0xe8, 0x30, 0x00, 0x00, 0x00,
+ 0x85, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x06, 0x86, 0xa0, 0x00, 0x00, 0x00,
+ 0x9a, 0x21, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_7[42] =
+ {
+ 0x4e, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x33, 0x2c, 0x00, 0x00, 0x00, 0x00,
+ 0x10, 0x0e, 0xb0, 0x00, 0x00, 0x00,
+ 0x81, 0x51, 0x50, 0x00, 0x00, 0x00,
+ 0x24, 0xc4, 0xa0, 0x00, 0x00, 0x00,
+ 0xd4, 0x23, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0xa2, 0x60, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_8[48] =
+ {
+ 0x27, 0x09, 0xc0, 0x00, 0x00, 0x00,
+ 0x89, 0xa2, 0x60, 0x00, 0x00, 0x00,
+ 0xd0, 0x74, 0x10, 0x00, 0x00, 0x00,
+ 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
+ 0xe2, 0x90, 0xa0, 0x00, 0x00, 0x00,
+ 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
+ 0x31, 0x8c, 0x60, 0x00, 0x00, 0x00,
+ 0x18, 0xc6, 0x30, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask20_9[54] =
+ {
+ 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
+ 0x62, 0x38, 0xc0, 0x00, 0x00, 0x00,
+ 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
+ 0xe1, 0x48, 0x50, 0x00, 0x00, 0x00,
+ 0x13, 0x94, 0xa0, 0x00, 0x00, 0x00,
+ 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
+ 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
+ 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
+ 0x49, 0x86, 0x50, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_1[6] =
+ {
+ 0xff, 0xff, 0xf8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_10[60] =
+ {
+ 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
+ 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
+ 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
+ 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
+ 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_11[66] =
+ {
+ 0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
+ 0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
+ 0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
+ 0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
+ 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
+ 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
+ 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_12[72] =
+ {
+ 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
+ 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
+ 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
+ 0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
+ 0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
+ 0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
+ 0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
+ 0xa0, 0x65, 0x18, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_13[78] =
+ {
+ 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
+ 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
+ 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
+ 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
+ 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
+ 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
+ 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
+ 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_14[84] =
+ {
+ 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
+ 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
+ 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
+ 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
+ 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
+ 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
+ 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
+ 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
+ 0x4d, 0xd0, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_15[90] =
+ {
+ 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
+ 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
+ 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
+ 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
+ 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
+ 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
+ 0x50, 0x45, 0x08, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_16[96] =
+ {
+ 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
+ 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
+ 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
+ 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
+ 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
+ 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
+ 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
+ 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
+ 0x3b, 0xf5, 0x38, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_17[102] =
+ {
+ 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
+ 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
+ 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
+ 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
+ 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
+ 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
+ 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
+ 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_18[108] =
+ {
+ 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
+ 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
+ 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
+ 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
+ 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
+ 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
+ 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
+ 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
+ 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
+ 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
+ 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
+ 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
+ 0x5a, 0x56, 0x58, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_19[114] =
+ {
+ 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
+ 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
+ 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
+ 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
+ 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
+ 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
+ 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
+ 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
+ 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
+ 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
+ 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_2[12] =
+ {
+ 0xee, 0x3b, 0x30, 0x00, 0x00, 0x00,
+ 0x99, 0xe6, 0xe8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_20[120] =
+ {
+ 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
+ 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
+ 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
+ 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
+ 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
+ 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
+ 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
+ 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
+ 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
+ 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
+ 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
+ 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
+ 0x2a, 0x03, 0x30, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_21[126] =
+ {
+ 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
+ 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
+ 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
+ 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
+ 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
+ 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
+ 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
+ 0x4c, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x51, 0x0c, 0xc8, 0x00, 0x00, 0x00,
+ 0xa0, 0x66, 0x40, 0x00, 0x00, 0x00,
+ 0x04, 0xc1, 0x60, 0x00, 0x00, 0x00,
+ 0x03, 0xa0, 0x28, 0x00, 0x00, 0x00,
+ 0x86, 0x21, 0x10, 0x00, 0x00, 0x00,
+ 0x29, 0x10, 0x18, 0x00, 0x00, 0x00,
+ 0x42, 0x42, 0xa0, 0x00, 0x00, 0x00,
+ 0x98, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x30, 0x84, 0x08, 0x00, 0x00, 0x00,
+ 0xdf, 0x4c, 0x10, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_3[18] =
+ {
+ 0xce, 0x32, 0xb0, 0x00, 0x00, 0x00,
+ 0x55, 0xdc, 0x50, 0x00, 0x00, 0x00,
+ 0xa8, 0xed, 0x88, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_4[24] =
+ {
+ 0xe6, 0x31, 0x30, 0x00, 0x00, 0x00,
+ 0x33, 0x8c, 0x58, 0x00, 0x00, 0x00,
+ 0x98, 0xd2, 0xc8, 0x00, 0x00, 0x00,
+ 0x2d, 0x4b, 0x28, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_5[30] =
+ {
+ 0xce, 0x31, 0xb0, 0x00, 0x00, 0x00,
+ 0x63, 0x98, 0xd8, 0x00, 0x00, 0x00,
+ 0x98, 0xc7, 0x68, 0x00, 0x00, 0x00,
+ 0x4d, 0x6b, 0x50, 0x00, 0x00, 0x00,
+ 0xb2, 0x6c, 0xa8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_6[36] =
+ {
+ 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
+ 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
+ 0x20, 0xea, 0x08, 0x00, 0x00, 0x00,
+ 0x85, 0x41, 0x28, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0xd8, 0x00, 0x00, 0x00,
+ 0x8a, 0x24, 0x30, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_7[42] =
+ {
+ 0xc6, 0x11, 0x90, 0x00, 0x00, 0x00,
+ 0x33, 0x04, 0xc8, 0x00, 0x00, 0x00,
+ 0x18, 0x67, 0x40, 0x00, 0x00, 0x00,
+ 0x45, 0x42, 0xd0, 0x00, 0x00, 0x00,
+ 0x12, 0xd4, 0x28, 0x00, 0x00, 0x00,
+ 0xb4, 0x28, 0x30, 0x00, 0x00, 0x00,
+ 0x29, 0x92, 0x18, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_8[48] =
+ {
+ 0x07, 0x0a, 0x70, 0x00, 0x00, 0x00,
+ 0x49, 0xa8, 0x28, 0x00, 0x00, 0x00,
+ 0xb0, 0x7a, 0x00, 0x00, 0x00, 0x00,
+ 0x24, 0xc5, 0xc0, 0x00, 0x00, 0x00,
+ 0x52, 0x80, 0xe8, 0x00, 0x00, 0x00,
+ 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
+ 0x31, 0x94, 0x18, 0x00, 0x00, 0x00,
+ 0x18, 0xc7, 0x08, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask21_9[54] =
+ {
+ 0x4e, 0x11, 0x10, 0x00, 0x00, 0x00,
+ 0x62, 0x1a, 0x08, 0x00, 0x00, 0x00,
+ 0x80, 0xe9, 0x40, 0x00, 0x00, 0x00,
+ 0xa1, 0x50, 0x50, 0x00, 0x00, 0x00,
+ 0x53, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0xa4, 0x24, 0x30, 0x00, 0x00, 0x00,
+ 0x16, 0xa0, 0x88, 0x00, 0x00, 0x00,
+ 0x58, 0x45, 0x20, 0x00, 0x00, 0x00,
+ 0x29, 0x86, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_1[6] =
+ {
+ 0xff, 0xff, 0xfc, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_10[60] =
+ {
+ 0xc0, 0x38, 0x88, 0x00, 0x00, 0x00,
+ 0x30, 0x0e, 0x28, 0x00, 0x00, 0x00,
+ 0xe8, 0x07, 0x00, 0x00, 0x00, 0x00,
+ 0x85, 0x08, 0xa8, 0x00, 0x00, 0x00,
+ 0xd0, 0x92, 0x10, 0x00, 0x00, 0x00,
+ 0x86, 0x50, 0x48, 0x00, 0x00, 0x00,
+ 0x4a, 0x68, 0x0c, 0x00, 0x00, 0x00,
+ 0x01, 0xa0, 0x74, 0x00, 0x00, 0x00,
+ 0x4c, 0x81, 0x90, 0x00, 0x00, 0x00,
+ 0x62, 0x24, 0x04, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_11[66] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
+ 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
+ 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
+ 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
+ 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
+ 0x30, 0x46, 0x08, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_12[72] =
+ {
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
+ 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
+ 0x86, 0x90, 0xd0, 0x00, 0x00, 0x00,
+ 0x23, 0x24, 0x64, 0x00, 0x00, 0x00,
+ 0x16, 0x22, 0xc4, 0x00, 0x00, 0x00,
+ 0x4c, 0x29, 0x84, 0x00, 0x00, 0x00,
+ 0x41, 0xc8, 0x38, 0x00, 0x00, 0x00,
+ 0xf4, 0x18, 0x9c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_13[78] =
+ {
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
+ 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
+ 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
+ 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
+ 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_14[84] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
+ 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
+ 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
+ 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
+ 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
+ 0xc6, 0xca, 0xe8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_15[90] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
+ 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
+ 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
+ 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
+ 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
+ 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
+ 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
+ 0x14, 0x22, 0x84, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_16[96] =
+ {
+ 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
+ 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
+ 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
+ 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
+ 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
+ 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
+ 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
+ 0x86, 0xc1, 0x44, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_17[102] =
+ {
+ 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
+ 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
+ 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
+ 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
+ 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
+ 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
+ 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
+ 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_18[108] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
+ 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
+ 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
+ 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
+ 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
+ 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
+ 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
+ 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
+ 0xe4, 0xd4, 0x6c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_19[114] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
+ 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
+ 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
+ 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
+ 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
+ 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
+ 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
+ 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
+ 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_2[12] =
+ {
+ 0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
+ 0x9b, 0xb3, 0x74, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_20[120] =
+ {
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
+ 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
+ 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
+ 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
+ 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
+ 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
+ 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
+ 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
+ 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
+ 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
+ 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
+ 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
+ 0x3b, 0x48, 0xc4, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_21[126] =
+ {
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
+ 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
+ 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
+ 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
+ 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
+ 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
+ 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
+ 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
+ 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
+ 0x30, 0x46, 0x08, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_22[132] =
+ {
+ 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
+ 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
+ 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
+ 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
+ 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
+ 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
+ 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
+ 0x30, 0x46, 0x08, 0x00, 0x00, 0x00,
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
+ 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
+ 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
+ 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
+ 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
+ 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
+ 0x9e, 0xce, 0x88, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_3[18] =
+ {
+ 0xca, 0xd9, 0x58, 0x00, 0x00, 0x00,
+ 0xf1, 0x5e, 0x28, 0x00, 0x00, 0x00,
+ 0xb6, 0x35, 0xc4, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_4[24] =
+ {
+ 0xc4, 0xd8, 0x98, 0x00, 0x00, 0x00,
+ 0x31, 0x66, 0x2c, 0x00, 0x00, 0x00,
+ 0x4b, 0x29, 0x64, 0x00, 0x00, 0x00,
+ 0x2c, 0xa5, 0x94, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_5[30] =
+ {
+ 0xc6, 0xd8, 0xd8, 0x00, 0x00, 0x00,
+ 0x63, 0x6c, 0x6c, 0x00, 0x00, 0x00,
+ 0x1d, 0xa3, 0xb4, 0x00, 0x00, 0x00,
+ 0xad, 0x55, 0xa8, 0x00, 0x00, 0x00,
+ 0xb2, 0xb6, 0x54, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_6[36] =
+ {
+ 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
+ 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0xa8, 0x35, 0x04, 0x00, 0x00, 0x00,
+ 0xc4, 0xa0, 0x94, 0x00, 0x00, 0x00,
+ 0x03, 0x60, 0x6c, 0x00, 0x00, 0x00,
+ 0x90, 0xd2, 0x18, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_7[42] =
+ {
+ 0xc6, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x13, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x8d, 0x13, 0xa0, 0x00, 0x00, 0x00,
+ 0x8b, 0x41, 0x68, 0x00, 0x00, 0x00,
+ 0x52, 0xaa, 0x14, 0x00, 0x00, 0x00,
+ 0xa2, 0xd4, 0x18, 0x00, 0x00, 0x00,
+ 0x61, 0xa8, 0x2c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_8[48] =
+ {
+ 0x28, 0x85, 0x38, 0x00, 0x00, 0x00,
+ 0x21, 0xf4, 0x04, 0x00, 0x00, 0x00,
+ 0xe9, 0x1d, 0x00, 0x00, 0x00, 0x00,
+ 0x17, 0x02, 0xe0, 0x00, 0x00, 0x00,
+ 0x83, 0xa0, 0x54, 0x00, 0x00, 0x00,
+ 0x46, 0x18, 0xe8, 0x00, 0x00, 0x00,
+ 0x50, 0x6a, 0x0c, 0x00, 0x00, 0x00,
+ 0x1c, 0x23, 0x84, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask22_9[54] =
+ {
+ 0x44, 0x48, 0xc8, 0x00, 0x00, 0x00,
+ 0x28, 0x2d, 0x0c, 0x00, 0x00, 0x00,
+ 0x25, 0x14, 0xa0, 0x00, 0x00, 0x00,
+ 0x59, 0x0a, 0x20, 0x00, 0x00, 0x00,
+ 0x03, 0xa0, 0x34, 0x00, 0x00, 0x00,
+ 0xc0, 0xd0, 0x18, 0x00, 0x00, 0x00,
+ 0xa2, 0x30, 0x44, 0x00, 0x00, 0x00,
+ 0x14, 0x82, 0xd0, 0x00, 0x00, 0x00,
+ 0x9a, 0x03, 0x80, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_1[6] =
+ {
+ 0xff, 0xff, 0xfe, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_10[60] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
+ 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
+ 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
+ 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
+ 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
+ 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
+ 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_11[66] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
+ 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
+ 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
+ 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
+ 0x30, 0x58, 0x04, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_12[72] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
+ 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
+ 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
+ 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
+ 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
+ 0x86, 0x8a, 0x6c, 0x00, 0x00, 0x00,
+ 0x23, 0x2c, 0x84, 0x00, 0x00, 0x00,
+ 0x16, 0x21, 0x98, 0x00, 0x00, 0x00,
+ 0x4c, 0x30, 0x54, 0x00, 0x00, 0x00,
+ 0x41, 0xc1, 0x26, 0x00, 0x00, 0x00,
+ 0x19, 0x56, 0xe4, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_13[78] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
+ 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
+ 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
+ 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
+ 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
+ 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
+ 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
+ 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_14[84] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
+ 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
+ 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
+ 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
+ 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
+ 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
+ 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
+ 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
+ 0x9c, 0x3f, 0xb2, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_15[90] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
+ 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
+ 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
+ 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
+ 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
+ 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
+ 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_16[96] =
+ {
+ 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
+ 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
+ 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
+ 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
+ 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
+ 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
+ 0xfa, 0xd9, 0xf4, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_17[102] =
+ {
+ 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
+ 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
+ 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
+ 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
+ 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
+ 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
+ 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
+ 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
+ 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_18[108] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
+ 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
+ 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
+ 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
+ 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
+ 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
+ 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
+ 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
+ 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
+ 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
+ 0x82, 0x32, 0x56, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_19[114] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
+ 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
+ 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
+ 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
+ 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
+ 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
+ 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
+ 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
+ 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
+ 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
+ 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
+ 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
+ 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_2[12] =
+ {
+ 0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
+ 0x9b, 0xb2, 0x76, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_20[120] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
+ 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
+ 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
+ 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
+ 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
+ 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
+ 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
+ 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
+ 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
+ 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
+ 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
+ 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
+ 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
+ 0xdb, 0x4a, 0x7a, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_21[126] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
+ 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
+ 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
+ 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
+ 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
+ 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
+ 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
+ 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
+ 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
+ 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
+ 0x30, 0x58, 0x04, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_22[132] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
+ 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
+ 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
+ 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
+ 0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
+ 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
+ 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
+ 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
+ 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
+ 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
+ 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
+ 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
+ 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
+ 0xea, 0x8d, 0x1a, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_23[138] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
+ 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
+ 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
+ 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
+ 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
+ 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
+ 0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
+ 0x46, 0x42, 0x0c, 0x00, 0x00, 0x00,
+ 0x33, 0x20, 0x46, 0x00, 0x00, 0x00,
+ 0x99, 0x08, 0x0a, 0x00, 0x00, 0x00,
+ 0x05, 0x84, 0x30, 0x00, 0x00, 0x00,
+ 0x80, 0xb0, 0x22, 0x00, 0x00, 0x00,
+ 0x84, 0x42, 0x90, 0x00, 0x00, 0x00,
+ 0x40, 0x73, 0x00, 0x00, 0x00, 0x00,
+ 0x0a, 0x81, 0x12, 0x00, 0x00, 0x00,
+ 0x68, 0x0c, 0x40, 0x00, 0x00, 0x00,
+ 0x10, 0x24, 0x84, 0x00, 0x00, 0x00,
+ 0x30, 0x51, 0x40, 0x00, 0x00, 0x00,
+ 0x5f, 0x50, 0x88, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_3[18] =
+ {
+ 0xca, 0xd3, 0x64, 0x00, 0x00, 0x00,
+ 0xf1, 0x49, 0x3a, 0x00, 0x00, 0x00,
+ 0x76, 0x27, 0xd0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_4[24] =
+ {
+ 0xc4, 0xd1, 0x64, 0x00, 0x00, 0x00,
+ 0x31, 0x62, 0x96, 0x00, 0x00, 0x00,
+ 0x4b, 0x24, 0x5a, 0x00, 0x00, 0x00,
+ 0x2c, 0xa8, 0xaa, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_5[30] =
+ {
+ 0xc6, 0xca, 0x6c, 0x00, 0x00, 0x00,
+ 0x63, 0x6c, 0x96, 0x00, 0x00, 0x00,
+ 0x1d, 0xa1, 0xdc, 0x00, 0x00, 0x00,
+ 0xad, 0x55, 0x38, 0x00, 0x00, 0x00,
+ 0xb2, 0xb7, 0x06, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_6[36] =
+ {
+ 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
+ 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
+ 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
+ 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
+ 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
+ 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_7[42] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
+ 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
+ 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
+ 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
+ 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_8[48] =
+ {
+ 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
+ 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
+ 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
+ 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
+ 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
+ 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask23_9[54] =
+ {
+ 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
+ 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
+ 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
+ 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
+ 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
+ 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
+ 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
+ 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
+ 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_10[60] =
+ {
+ 0x11, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
+ 0x00, 0x48, 0x05, 0x00, 0x00, 0x00,
+ 0x10, 0x83, 0x09, 0x00, 0x00, 0x00,
+ 0x4a, 0x14, 0xa1, 0x00, 0x00, 0x00,
+ 0x40, 0xa4, 0x0a, 0x00, 0x00, 0x00,
+ 0xa0, 0x6a, 0x02, 0x00, 0x00, 0x00,
+ 0x88, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x54, 0x0d, 0x40, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_11[66] =
+ {
+ 0x53, 0x65, 0x34, 0x00, 0x00, 0x00,
+ 0xa0, 0x32, 0x11, 0x00, 0x00, 0x00,
+ 0x15, 0x11, 0x41, 0x00, 0x00, 0x00,
+ 0x03, 0x50, 0x15, 0x00, 0x00, 0x00,
+ 0x8c, 0x88, 0xc8, 0x00, 0x00, 0x00,
+ 0x28, 0x82, 0x88, 0x00, 0x00, 0x00,
+ 0x08, 0x48, 0x84, 0x00, 0x00, 0x00,
+ 0x99, 0x01, 0x90, 0x00, 0x00, 0x00,
+ 0x22, 0x92, 0x29, 0x00, 0x00, 0x00,
+ 0x46, 0x04, 0x60, 0x00, 0x00, 0x00,
+ 0x8c, 0x2c, 0x02, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_12[72] =
+ {
+ 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
+ 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
+ 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
+ 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
+ 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
+ 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
+ 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
+ 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
+ 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x44, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_13[78] =
+ {
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
+ 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
+ 0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
+ 0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
+ 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
+ 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
+ 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
+ 0x88, 0x68, 0x86, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_14[84] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
+ 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
+ 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
+ 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
+ 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
+ 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
+ 0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
+ 0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
+ 0xf2, 0xd6, 0x8e, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_15[90] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
+ 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
+ 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
+ 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
+ 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
+ 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
+ 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
+ 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
+ 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
+ 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
+ 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_16[96] =
+ {
+ 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
+ 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
+ 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
+ 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
+ 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
+ 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
+ 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
+ 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
+ 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
+ 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
+ 0xff, 0x6e, 0x0a, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_17[102] =
+ {
+ 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
+ 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
+ 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
+ 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
+ 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
+ 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
+ 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
+ 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
+ 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
+ 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_18[108] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
+ 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
+ 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
+ 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
+ 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
+ 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
+ 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
+ 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
+ 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
+ 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
+ 0x34, 0x50, 0xae, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_19[114] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
+ 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
+ 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
+ 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
+ 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
+ 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
+ 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
+ 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_2[12] =
+ {
+ 0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
+ 0x93, 0xb9, 0x3b, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_20[120] =
+ {
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
+ 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
+ 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
+ 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
+ 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
+ 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
+ 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
+ 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0x98, 0xa2, 0x95, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_21[126] =
+ {
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
+ 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
+ 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
+ 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
+ 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
+ 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
+ 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
+ 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
+ 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
+ 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_22[132] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
+ 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
+ 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
+ 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
+ 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
+ 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
+ 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
+ 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
+ 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
+ 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
+ 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
+ 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
+ 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
+ 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
+ 0x1a, 0xaa, 0xee, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_23[138] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
+ 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
+ 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
+ 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
+ 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
+ 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
+ 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
+ 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
+ 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
+ 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
+ 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
+ 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
+ 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
+ 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
+ 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
+ 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x44, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_24[144] =
+ {
+ 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
+ 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
+ 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
+ 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
+ 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
+ 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
+ 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
+ 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
+ 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
+ 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
+ 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
+ 0x84, 0x48, 0x44, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
+ 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
+ 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
+ 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
+ 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
+ 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
+ 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
+ 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
+ 0x88, 0x32, 0x59, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_3[18] =
+ {
+ 0x9b, 0x29, 0xb2, 0x00, 0x00, 0x00,
+ 0x49, 0xd4, 0x9d, 0x00, 0x00, 0x00,
+ 0x3e, 0x83, 0xe8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_4[24] =
+ {
+ 0x8b, 0x28, 0xb2, 0x00, 0x00, 0x00,
+ 0x14, 0xb1, 0x4b, 0x00, 0x00, 0x00,
+ 0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
+ 0x45, 0x54, 0x55, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_5[30] =
+ {
+ 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x64, 0xb6, 0x4b, 0x00, 0x00, 0x00,
+ 0x0e, 0xe0, 0xee, 0x00, 0x00, 0x00,
+ 0xa9, 0xca, 0x9c, 0x00, 0x00, 0x00,
+ 0xb8, 0x3b, 0x83, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_6[36] =
+ {
+ 0xd1, 0x4d, 0x14, 0x00, 0x00, 0x00,
+ 0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
+ 0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
+ 0x16, 0xc1, 0x6c, 0x00, 0x00, 0x00,
+ 0x0b, 0xa0, 0xba, 0x00, 0x00, 0x00,
+ 0xe8, 0x8e, 0x88, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_7[42] =
+ {
+ 0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x25, 0x32, 0x53, 0x00, 0x00, 0x00,
+ 0x30, 0xd3, 0x05, 0x00, 0x00, 0x00,
+ 0x06, 0x48, 0x6c, 0x00, 0x00, 0x00,
+ 0xc0, 0xb8, 0x1b, 0x00, 0x00, 0x00,
+ 0x2a, 0xa2, 0xaa, 0x00, 0x00, 0x00,
+ 0xa8, 0x4e, 0x84, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_8[48] =
+ {
+ 0x81, 0x60, 0x16, 0x00, 0x00, 0x00,
+ 0x40, 0x3c, 0x03, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
+ 0x06, 0x50, 0x65, 0x00, 0x00, 0x00,
+ 0x20, 0x4a, 0x84, 0x00, 0x00, 0x00,
+ 0x8a, 0xa0, 0xaa, 0x00, 0x00, 0x00,
+ 0x33, 0x03, 0x30, 0x00, 0x00, 0x00,
+ 0x4c, 0x84, 0xc8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask24_9[54] =
+ {
+ 0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
+ 0x64, 0x26, 0x42, 0x00, 0x00, 0x00,
+ 0x18, 0x41, 0xc4, 0x00, 0x00, 0x00,
+ 0xa0, 0x4a, 0x04, 0x00, 0x00, 0x00,
+ 0x81, 0x38, 0x13, 0x00, 0x00, 0x00,
+ 0x22, 0xa2, 0x2a, 0x00, 0x00, 0x00,
+ 0x08, 0x70, 0x87, 0x00, 0x00, 0x00,
+ 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
+ 0x01, 0xc0, 0x1c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_10[60] =
+ {
+ 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
+ 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
+ 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
+ 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
+ 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
+ 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
+ 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
+ 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
+ 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_11[66] =
+ {
+ 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
+ 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
+ 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
+ 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
+ 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
+ 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
+ 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_12[72] =
+ {
+ 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
+ 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
+ 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
+ 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
+ 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
+ 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
+ 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
+ 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
+ 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
+ 0x84, 0x45, 0x22, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_13[78] =
+ {
+ 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
+ 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
+ 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
+ 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
+ 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
+ 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
+ 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
+ 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
+ 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
+ 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
+ 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_14[84] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
+ 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
+ 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
+ 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
+ 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
+ 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
+ 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
+ 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
+ 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
+ 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
+ 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
+ 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
+ 0x73, 0x76, 0x61, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_15[90] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
+ 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
+ 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
+ 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
+ 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
+ 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
+ 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
+ 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
+ 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
+ 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
+ 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
+ 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_16[96] =
+ {
+ 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
+ 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
+ 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
+ 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
+ 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
+ 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
+ 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
+ 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
+ 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
+ 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
+ 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
+ 0x16, 0xe8, 0xdc, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_17[102] =
+ {
+ 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
+ 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
+ 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
+ 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
+ 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
+ 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
+ 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
+ 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
+ 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
+ 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_18[108] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
+ 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
+ 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
+ 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
+ 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
+ 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
+ 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
+ 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
+ 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
+ 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
+ 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
+ 0xce, 0x9b, 0xe1, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_19[114] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
+ 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
+ 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
+ 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
+ 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
+ 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
+ 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
+ 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
+ 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
+ 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
+ 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
+ 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
+ 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
+ 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_2[12] =
+ {
+ 0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
+ 0x93, 0xb1, 0xb3, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_20[120] =
+ {
+ 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
+ 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
+ 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
+ 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
+ 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
+ 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
+ 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
+ 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
+ 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
+ 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
+ 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
+ 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
+ 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
+ 0x1b, 0x8a, 0xa0, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_21[126] =
+ {
+ 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
+ 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
+ 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
+ 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
+ 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
+ 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
+ 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
+ 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
+ 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
+ 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
+ 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
+ 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
+ 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
+ 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
+ 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_22[132] =
+ {
+ 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
+ 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
+ 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
+ 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
+ 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
+ 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
+ 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
+ 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
+ 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
+ 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
+ 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
+ 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
+ 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
+ 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
+ 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
+ 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
+ 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
+ 0x15, 0xa2, 0x99, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_23[138] =
+ {
+ 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
+ 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
+ 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
+ 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
+ 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
+ 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
+ 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
+ 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
+ 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
+ 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
+ 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
+ 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
+ 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
+ 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
+ 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
+ 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
+ 0x84, 0x45, 0x22, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_24[144] =
+ {
+ 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
+ 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
+ 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
+ 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
+ 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
+ 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
+ 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
+ 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
+ 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
+ 0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
+ 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
+ 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
+ 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
+ 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
+ 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
+ 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
+ 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
+ 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
+ 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
+ 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
+ 0xf9, 0x0c, 0x14, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_25[150] =
+ {
+ 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
+ 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
+ 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
+ 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
+ 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
+ 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
+ 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
+ 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
+ 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
+ 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
+ 0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
+ 0x10, 0x65, 0x12, 0x00, 0x00, 0x00,
+ 0x02, 0x36, 0x64, 0x00, 0x00, 0x00,
+ 0x40, 0x50, 0x54, 0x80, 0x00, 0x00,
+ 0x21, 0x88, 0x12, 0x00, 0x00, 0x00,
+ 0x81, 0x19, 0x40, 0x00, 0x00, 0x00,
+ 0x14, 0x83, 0x08, 0x00, 0x00, 0x00,
+ 0x98, 0x02, 0x11, 0x00, 0x00, 0x00,
+ 0x08, 0x90, 0x3c, 0x00, 0x00, 0x00,
+ 0x62, 0x0e, 0x80, 0x00, 0x00, 0x00,
+ 0x24, 0x20, 0xa1, 0x00, 0x00, 0x00,
+ 0x8a, 0x08, 0x01, 0x80, 0x00, 0x00,
+ 0x84, 0x40, 0x49, 0x00, 0x00, 0x00,
+ 0x1c, 0x20, 0x8a, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_3[18] =
+ {
+ 0x9b, 0x89, 0x9b, 0x00, 0x00, 0x00,
+ 0x4f, 0x14, 0x6d, 0x80, 0x00, 0x00,
+ 0x3c, 0x63, 0x72, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_4[24] =
+ {
+ 0x8b, 0x24, 0x9b, 0x00, 0x00, 0x00,
+ 0x14, 0xb2, 0x6d, 0x00, 0x00, 0x00,
+ 0x22, 0xd8, 0x56, 0x80, 0x00, 0x00,
+ 0x45, 0x55, 0x25, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_5[30] =
+ {
+ 0x53, 0x65, 0x13, 0x00, 0x00, 0x00,
+ 0x64, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x0c, 0xc0, 0xc6, 0x80, 0x00, 0x00,
+ 0x82, 0xaa, 0x1c, 0x00, 0x00, 0x00,
+ 0x09, 0x32, 0x29, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_6[36] =
+ {
+ 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
+ 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
+ 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
+ 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
+ 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
+ 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_7[42] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
+ 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
+ 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
+ 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
+ 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
+ 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_8[48] =
+ {
+ 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
+ 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
+ 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
+ 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
+ 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
+ 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
+ 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
+ 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask25_9[54] =
+ {
+ 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
+ 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
+ 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
+ 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
+ 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
+ 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
+ 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
+ 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
+ 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_10[60] =
+ {
+ 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
+ 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
+ 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
+ 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
+ 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
+ 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
+ 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
+ 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
+ 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
+ 0x62, 0x23, 0x11, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_11[66] =
+ {
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
+ 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
+ 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
+ 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
+ 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
+ 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
+ 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
+ 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
+ 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
+ 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_12[72] =
+ {
+ 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
+ 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
+ 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
+ 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
+ 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
+ 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
+ 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x52, 0x22, 0x91, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_13[78] =
+ {
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
+ 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
+ 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
+ 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
+ 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
+ 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_14[84] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
+ 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
+ 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
+ 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
+ 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
+ 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
+ 0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
+ 0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
+ 0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
+ 0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
+ 0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
+ 0xaa, 0x25, 0x51, 0x00, 0x00, 0x00,
+ 0x5d, 0xa7, 0x78, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_15[90] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
+ 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
+ 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
+ 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
+ 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
+ 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
+ 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
+ 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
+ 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
+ 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_16[96] =
+ {
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
+ 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
+ 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
+ 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
+ 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
+ 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
+ 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
+ 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
+ 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
+ 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
+ 0x3c, 0xaf, 0x88, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_17[102] =
+ {
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
+ 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
+ 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
+ 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
+ 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
+ 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
+ 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
+ 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_18[108] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
+ 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
+ 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
+ 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
+ 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
+ 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
+ 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
+ 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
+ 0xaa, 0x0c, 0x83, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_19[114] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
+ 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
+ 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
+ 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
+ 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
+ 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
+ 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
+ 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
+ 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
+ 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
+ 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
+ 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
+ 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
+ 0x62, 0x23, 0x11, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_2[12] =
+ {
+ 0xec, 0xc7, 0x66, 0x00, 0x00, 0x00,
+ 0x1b, 0x38, 0xd9, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_20[120] =
+ {
+ 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
+ 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
+ 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
+ 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
+ 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
+ 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
+ 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
+ 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
+ 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
+ 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
+ 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
+ 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
+ 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
+ 0xf4, 0x08, 0xec, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_21[126] =
+ {
+ 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
+ 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
+ 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
+ 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
+ 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
+ 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
+ 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
+ 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
+ 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
+ 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
+ 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
+ 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
+ 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
+ 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
+ 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
+ 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
+ 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
+ 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
+ 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_22[132] =
+ {
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
+ 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
+ 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
+ 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
+ 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
+ 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
+ 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
+ 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
+ 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
+ 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
+ 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
+ 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
+ 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
+ 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
+ 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
+ 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
+ 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
+ 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
+ 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
+ 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
+ 0x13, 0xc6, 0x6b, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_23[138] =
+ {
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
+ 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
+ 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
+ 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
+ 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
+ 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
+ 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
+ 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
+ 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
+ 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
+ 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
+ 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
+ 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
+ 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
+ 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
+ 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
+ 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x52, 0x22, 0x91, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_24[144] =
+ {
+ 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
+ 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
+ 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
+ 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
+ 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
+ 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
+ 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
+ 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
+ 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
+ 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
+ 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
+ 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
+ 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
+ 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
+ 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
+ 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
+ 0xdb, 0x4d, 0xd8, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_25[150] =
+ {
+ 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
+ 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
+ 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
+ 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
+ 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
+ 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
+ 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
+ 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
+ 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
+ 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
+ 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
+ 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_26[156] =
+ {
+ 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
+ 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
+ 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
+ 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
+ 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
+ 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
+ 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00,
+ 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
+ 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
+ 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
+ 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
+ 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
+ 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
+ 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
+ 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
+ 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
+ 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
+ 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
+ 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
+ 0xf9, 0x13, 0x51, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_3[18] =
+ {
+ 0x99, 0xb4, 0xcd, 0x80, 0x00, 0x00,
+ 0x46, 0xda, 0x36, 0xc0, 0x00, 0x00,
+ 0x37, 0x29, 0xb9, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_4[24] =
+ {
+ 0x49, 0xb2, 0x4d, 0x80, 0x00, 0x00,
+ 0x26, 0xd1, 0x36, 0x80, 0x00, 0x00,
+ 0x85, 0x6c, 0x2b, 0x40, 0x00, 0x00,
+ 0x52, 0x5a, 0x92, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_5[30] =
+ {
+ 0x51, 0x32, 0x89, 0x80, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x0c, 0x68, 0x63, 0x40, 0x00, 0x00,
+ 0xa1, 0xc5, 0x0e, 0x00, 0x00, 0x00,
+ 0x22, 0x99, 0x14, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_6[36] =
+ {
+ 0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
+ 0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
+ 0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
+ 0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
+ 0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
+ 0xaa, 0x25, 0x51, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_7[42] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
+ 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
+ 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
+ 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
+ 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
+ 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_8[48] =
+ {
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
+ 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
+ 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
+ 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
+ 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
+ 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
+ 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask26_9[54] =
+ {
+ 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
+ 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
+ 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
+ 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
+ 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
+ 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
+ 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xe0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_10[60] =
+ {
+ 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
+ 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
+ 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
+ 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
+ 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
+ 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
+ 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
+ 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
+ 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
+ 0x62, 0x21, 0x18, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_11[66] =
+ {
+ 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
+ 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
+ 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
+ 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
+ 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
+ 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
+ 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
+ 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
+ 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
+ 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
+ 0x44, 0x19, 0x16, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_12[72] =
+ {
+ 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
+ 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
+ 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
+ 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
+ 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
+ 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
+ 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
+ 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
+ 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
+ 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
+ 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
+ 0x52, 0x20, 0x90, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_13[78] =
+ {
+ 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
+ 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
+ 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
+ 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
+ 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
+ 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
+ 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
+ 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
+ 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
+ 0x08, 0xac, 0x06, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_14[84] =
+ {
+ 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
+ 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
+ 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
+ 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
+ 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
+ 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
+ 0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
+ 0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
+ 0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
+ 0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
+ 0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
+ 0xaa, 0x24, 0x44, 0x60, 0x00, 0x00,
+ 0x0a, 0xe7, 0x3b, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_15[90] =
+ {
+ 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
+ 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
+ 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
+ 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
+ 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
+ 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
+ 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
+ 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
+ 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
+ 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
+ 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
+ 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
+ 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
+ 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_16[96] =
+ {
+ 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
+ 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
+ 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
+ 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
+ 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
+ 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
+ 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
+ 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
+ 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
+ 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
+ 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
+ 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
+ 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
+ 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
+ 0x01, 0x50, 0xfb, 0xe0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_17[102] =
+ {
+ 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
+ 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
+ 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
+ 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
+ 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
+ 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
+ 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
+ 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
+ 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
+ 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
+ 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
+ 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
+ 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
+ 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_18[108] =
+ {
+ 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
+ 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
+ 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
+ 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
+ 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
+ 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
+ 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
+ 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
+ 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
+ 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
+ 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
+ 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
+ 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
+ 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
+ 0x53, 0xc3, 0x33, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_19[114] =
+ {
+ 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
+ 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
+ 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
+ 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
+ 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
+ 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
+ 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
+ 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
+ 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
+ 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
+ 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
+ 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
+ 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
+ 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
+ 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
+ 0x62, 0x21, 0x18, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_2[12] =
+ {
+ 0xec, 0xc7, 0x67, 0x40, 0x00, 0x00,
+ 0x1b, 0x39, 0xdc, 0xe0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_20[120] =
+ {
+ 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
+ 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
+ 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
+ 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
+ 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
+ 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
+ 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
+ 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
+ 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
+ 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
+ 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
+ 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
+ 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
+ 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
+ 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
+ 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
+ 0xcb, 0xff, 0x6f, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_21[126] =
+ {
+ 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
+ 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
+ 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
+ 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
+ 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
+ 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
+ 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
+ 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
+ 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
+ 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
+ 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
+ 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
+ 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
+ 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
+ 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
+ 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
+ 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
+ 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
+ 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
+ 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
+ 0x44, 0x19, 0x16, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_22[132] =
+ {
+ 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
+ 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
+ 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
+ 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
+ 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
+ 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
+ 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
+ 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
+ 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
+ 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
+ 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
+ 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
+ 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
+ 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
+ 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
+ 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
+ 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
+ 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
+ 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
+ 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
+ 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
+ 0xf5, 0x2d, 0x52, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_23[138] =
+ {
+ 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
+ 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
+ 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
+ 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
+ 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
+ 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
+ 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
+ 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
+ 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
+ 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
+ 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
+ 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
+ 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
+ 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
+ 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
+ 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
+ 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
+ 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
+ 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
+ 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
+ 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
+ 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
+ 0x52, 0x20, 0x90, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_24[144] =
+ {
+ 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
+ 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
+ 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
+ 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
+ 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
+ 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
+ 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
+ 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
+ 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
+ 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
+ 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
+ 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
+ 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
+ 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
+ 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
+ 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
+ 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
+ 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
+ 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
+ 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
+ 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
+ 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
+ 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
+ 0xa2, 0x85, 0xdb, 0xa0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_25[150] =
+ {
+ 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
+ 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
+ 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
+ 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
+ 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
+ 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
+ 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
+ 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
+ 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
+ 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
+ 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
+ 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
+ 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
+ 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
+ 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
+ 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
+ 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
+ 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
+ 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
+ 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
+ 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
+ 0x08, 0xac, 0x06, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_26[156] =
+ {
+ 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
+ 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
+ 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
+ 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
+ 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
+ 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
+ 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
+ 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
+ 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
+ 0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
+ 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
+ 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
+ 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
+ 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
+ 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
+ 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
+ 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
+ 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
+ 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
+ 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
+ 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
+ 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
+ 0xcd, 0x41, 0xa2, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_27[162] =
+ {
+ 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
+ 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
+ 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
+ 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
+ 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
+ 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
+ 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
+ 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
+ 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
+ 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
+ 0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
+ 0x51, 0x22, 0x02, 0xa0, 0x00, 0x00,
+ 0x66, 0x40, 0xaa, 0x00, 0x00, 0x00,
+ 0x05, 0x4e, 0x00, 0x20, 0x00, 0x00,
+ 0x81, 0x21, 0x40, 0x80, 0x00, 0x00,
+ 0x94, 0x00, 0x28, 0x60, 0x00, 0x00,
+ 0x30, 0x83, 0x24, 0x00, 0x00, 0x00,
+ 0x21, 0x14, 0x0c, 0x00, 0x00, 0x00,
+ 0x03, 0xc0, 0x84, 0xc0, 0x00, 0x00,
+ 0xe8, 0x04, 0x21, 0x00, 0x00, 0x00,
+ 0x0a, 0x10, 0x91, 0x80, 0x00, 0x00,
+ 0x80, 0x1b, 0x10, 0x00, 0x00, 0x00,
+ 0x04, 0x91, 0x43, 0x00, 0x00, 0x00,
+ 0x08, 0xa8, 0x70, 0x40, 0x00, 0x00,
+ 0x9c, 0xc0, 0x84, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_3[18] =
+ {
+ 0x99, 0xb5, 0x66, 0xc0, 0x00, 0x00,
+ 0x46, 0xda, 0xab, 0x60, 0x00, 0x00,
+ 0x37, 0x29, 0x3d, 0xa0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_4[24] =
+ {
+ 0x49, 0xb1, 0x66, 0xc0, 0x00, 0x00,
+ 0x26, 0xd4, 0x9b, 0x40, 0x00, 0x00,
+ 0x85, 0x68, 0xd5, 0xa0, 0x00, 0x00,
+ 0x52, 0x5a, 0x39, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_5[30] =
+ {
+ 0x51, 0x33, 0x26, 0xc0, 0x00, 0x00,
+ 0x66, 0x45, 0x2b, 0x40, 0x00, 0x00,
+ 0x0c, 0x6a, 0x95, 0xa0, 0x00, 0x00,
+ 0xa1, 0xc0, 0xed, 0x40, 0x00, 0x00,
+ 0x22, 0x9c, 0xe2, 0xa0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_6[36] =
+ {
+ 0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
+ 0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
+ 0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
+ 0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
+ 0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
+ 0xaa, 0x24, 0x44, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_7[42] =
+ {
+ 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
+ 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
+ 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
+ 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
+ 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
+ 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
+ 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_8[48] =
+ {
+ 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
+ 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
+ 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
+ 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
+ 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
+ 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
+ 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
+ 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask27_9[54] =
+ {
+ 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
+ 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
+ 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
+ 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
+ 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
+ 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
+ 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
+ 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
+ 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xf0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_10[60] =
+ {
+ 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
+ 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
+ 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
+ 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
+ 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
+ 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
+ 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
+ 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_11[66] =
+ {
+ 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
+ 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
+ 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
+ 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
+ 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
+ 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
+ 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
+ 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
+ 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
+ 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
+ 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_12[72] =
+ {
+ 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
+ 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
+ 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
+ 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
+ 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
+ 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
+ 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
+ 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
+ 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
+ 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
+ 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_13[78] =
+ {
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
+ 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
+ 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
+ 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
+ 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
+ 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
+ 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
+ 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
+ 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
+ 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_14[84] =
+ {
+ 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
+ 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
+ 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
+ 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
+ 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
+ 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
+ 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
+ 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
+ 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
+ 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
+ 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0x42, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_15[90] =
+ {
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
+ 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
+ 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
+ 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
+ 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
+ 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
+ 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
+ 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
+ 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
+ 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
+ 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
+ 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
+ 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
+ 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_16[96] =
+ {
+ 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
+ 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
+ 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
+ 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
+ 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
+ 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
+ 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
+ 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
+ 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
+ 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
+ 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
+ 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
+ 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
+ 0xed, 0x76, 0x36, 0x50, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_17[102] =
+ {
+ 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
+ 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
+ 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
+ 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
+ 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
+ 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
+ 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
+ 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
+ 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
+ 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
+ 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
+ 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
+ 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
+ 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
+ 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_18[108] =
+ {
+ 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
+ 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
+ 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
+ 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
+ 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
+ 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
+ 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
+ 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
+ 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
+ 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
+ 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
+ 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
+ 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
+ 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
+ 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
+ 0x6e, 0x9f, 0x98, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_19[114] =
+ {
+ 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
+ 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
+ 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
+ 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
+ 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
+ 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
+ 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
+ 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
+ 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
+ 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
+ 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
+ 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
+ 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
+ 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
+ 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_2[12] =
+ {
+ 0xec, 0xeb, 0xb3, 0xa0, 0x00, 0x00,
+ 0x3b, 0x9c, 0xee, 0x70, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_20[120] =
+ {
+ 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
+ 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
+ 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
+ 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
+ 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
+ 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
+ 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
+ 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
+ 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
+ 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
+ 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
+ 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
+ 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
+ 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
+ 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
+ 0xea, 0x1b, 0x3a, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_21[126] =
+ {
+ 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
+ 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
+ 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
+ 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
+ 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
+ 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
+ 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
+ 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
+ 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
+ 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
+ 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
+ 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
+ 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
+ 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
+ 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
+ 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
+ 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
+ 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
+ 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_22[132] =
+ {
+ 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
+ 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
+ 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
+ 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
+ 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
+ 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
+ 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
+ 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
+ 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
+ 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
+ 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
+ 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
+ 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
+ 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
+ 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
+ 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
+ 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
+ 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
+ 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
+ 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
+ 0x45, 0x05, 0x10, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_23[138] =
+ {
+ 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
+ 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
+ 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
+ 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
+ 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
+ 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
+ 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
+ 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
+ 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
+ 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
+ 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
+ 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
+ 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
+ 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
+ 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
+ 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
+ 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
+ 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
+ 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
+ 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
+ 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
+ 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_24[144] =
+ {
+ 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
+ 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
+ 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
+ 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
+ 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
+ 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
+ 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
+ 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
+ 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
+ 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
+ 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
+ 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
+ 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
+ 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
+ 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
+ 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
+ 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
+ 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
+ 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
+ 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
+ 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
+ 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
+ 0x6f, 0xd8, 0xee, 0xa0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_25[150] =
+ {
+ 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
+ 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
+ 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
+ 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
+ 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
+ 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
+ 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
+ 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
+ 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
+ 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
+ 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
+ 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
+ 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
+ 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
+ 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
+ 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
+ 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
+ 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
+ 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
+ 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_26[156] =
+ {
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
+ 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
+ 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
+ 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
+ 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
+ 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
+ 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
+ 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
+ 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
+ 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
+ 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
+ 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
+ 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
+ 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
+ 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
+ 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
+ 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
+ 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
+ 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
+ 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
+ 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
+ 0xf1, 0x64, 0xbe, 0x40, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_27[162] =
+ {
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
+ 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
+ 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
+ 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
+ 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
+ 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
+ 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
+ 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
+ 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
+ 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
+ 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
+ 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
+ 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
+ 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
+ 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
+ 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
+ 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
+ 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
+ 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
+ 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
+ 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0x42, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_28[168] =
+ {
+ 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
+ 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
+ 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
+ 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
+ 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
+ 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
+ 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
+ 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
+ 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
+ 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
+ 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0x42, 0x10, 0x00, 0x00,
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
+ 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
+ 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
+ 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
+ 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
+ 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
+ 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
+ 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
+ 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
+ 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
+ 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
+ 0x36, 0x4f, 0x1f, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_3[18] =
+ {
+ 0xac, 0xda, 0xb3, 0x60, 0x00, 0x00,
+ 0x55, 0x6d, 0x55, 0xb0, 0x00, 0x00,
+ 0x27, 0xb4, 0x9e, 0xd0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_4[24] =
+ {
+ 0x2c, 0xd8, 0xb3, 0x60, 0x00, 0x00,
+ 0x93, 0x6a, 0x4d, 0xa0, 0x00, 0x00,
+ 0x1a, 0xb4, 0x6a, 0xd0, 0x00, 0x00,
+ 0x47, 0x2d, 0x1c, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_5[30] =
+ {
+ 0x64, 0xd9, 0x93, 0x60, 0x00, 0x00,
+ 0xa5, 0x6a, 0x95, 0xa0, 0x00, 0x00,
+ 0x52, 0xb5, 0x4a, 0xd0, 0x00, 0x00,
+ 0x1d, 0xa8, 0x76, 0xa0, 0x00, 0x00,
+ 0x9c, 0x56, 0x71, 0x50, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_6[36] =
+ {
+ 0x4a, 0x55, 0x29, 0x50, 0x00, 0x00,
+ 0x95, 0x4a, 0x55, 0x20, 0x00, 0x00,
+ 0x14, 0xb4, 0x52, 0xd0, 0x00, 0x00,
+ 0x51, 0xa9, 0x46, 0xa0, 0x00, 0x00,
+ 0x22, 0x6c, 0x89, 0xb0, 0x00, 0x00,
+ 0x88, 0x8e, 0x22, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_7[42] =
+ {
+ 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
+ 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
+ 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
+ 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
+ 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
+ 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
+ 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_8[48] =
+ {
+ 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
+ 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
+ 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
+ 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
+ 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
+ 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
+ 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
+ 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask28_9[54] =
+ {
+ 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
+ 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
+ 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
+ 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
+ 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
+ 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
+ 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
+ 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xf8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_10[60] =
+ {
+ 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
+ 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
+ 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
+ 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
+ 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
+ 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
+ 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
+ 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
+ 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
+ 0x23, 0x10, 0x09, 0x88, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_11[66] =
+ {
+ 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
+ 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
+ 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
+ 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
+ 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
+ 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
+ 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
+ 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
+ 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
+ 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
+ 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_12[72] =
+ {
+ 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
+ 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
+ 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
+ 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
+ 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
+ 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
+ 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
+ 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
+ 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
+ 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
+ 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_13[78] =
+ {
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
+ 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
+ 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
+ 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
+ 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
+ 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
+ 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
+ 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
+ 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
+ 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_14[84] =
+ {
+ 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
+ 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
+ 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
+ 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
+ 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
+ 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
+ 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
+ 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
+ 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
+ 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
+ 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_15[90] =
+ {
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
+ 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
+ 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
+ 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
+ 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
+ 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
+ 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
+ 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
+ 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
+ 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
+ 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
+ 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
+ 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
+ 0x72, 0x01, 0x96, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_16[96] =
+ {
+ 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
+ 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
+ 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
+ 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
+ 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
+ 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
+ 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
+ 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
+ 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
+ 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
+ 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
+ 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
+ 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
+ 0x0d, 0x2c, 0xf2, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_17[102] =
+ {
+ 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
+ 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
+ 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
+ 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
+ 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
+ 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
+ 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
+ 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
+ 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
+ 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
+ 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
+ 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
+ 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
+ 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
+ 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
+ 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_18[108] =
+ {
+ 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
+ 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
+ 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
+ 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
+ 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
+ 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
+ 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
+ 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
+ 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
+ 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
+ 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
+ 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
+ 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
+ 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
+ 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
+ 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
+ 0x71, 0x36, 0xf2, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_19[114] =
+ {
+ 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
+ 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
+ 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
+ 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
+ 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
+ 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
+ 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
+ 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
+ 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
+ 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
+ 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
+ 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
+ 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
+ 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
+ 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
+ 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
+ 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
+ 0x23, 0x10, 0x09, 0x88, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_2[12] =
+ {
+ 0xec, 0xeb, 0xb3, 0xa8, 0x00, 0x00,
+ 0x3b, 0x9e, 0xee, 0x70, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_20[120] =
+ {
+ 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
+ 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
+ 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
+ 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
+ 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
+ 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
+ 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
+ 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
+ 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
+ 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
+ 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
+ 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
+ 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
+ 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
+ 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
+ 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
+ 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
+ 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
+ 0xe7, 0xec, 0xdc, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_21[126] =
+ {
+ 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
+ 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
+ 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
+ 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
+ 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
+ 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
+ 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
+ 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
+ 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
+ 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
+ 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
+ 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
+ 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
+ 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
+ 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
+ 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
+ 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
+ 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
+ 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
+ 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
+ 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_22[132] =
+ {
+ 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
+ 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
+ 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
+ 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
+ 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
+ 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
+ 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
+ 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
+ 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
+ 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
+ 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
+ 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
+ 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
+ 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
+ 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
+ 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
+ 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
+ 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
+ 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
+ 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
+ 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
+ 0x1c, 0x90, 0xa9, 0xa0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_23[138] =
+ {
+ 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
+ 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
+ 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
+ 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
+ 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
+ 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
+ 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
+ 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
+ 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
+ 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
+ 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
+ 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
+ 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
+ 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
+ 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
+ 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
+ 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
+ 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
+ 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
+ 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
+ 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
+ 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_24[144] =
+ {
+ 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
+ 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
+ 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
+ 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
+ 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
+ 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
+ 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
+ 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
+ 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
+ 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
+ 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
+ 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
+ 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
+ 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
+ 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
+ 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
+ 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
+ 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
+ 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
+ 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
+ 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
+ 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
+ 0xbd, 0x86, 0x97, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_25[150] =
+ {
+ 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
+ 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
+ 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
+ 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
+ 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
+ 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
+ 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
+ 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
+ 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
+ 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
+ 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
+ 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
+ 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
+ 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
+ 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
+ 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
+ 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
+ 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
+ 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
+ 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_26[156] =
+ {
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
+ 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
+ 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
+ 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
+ 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
+ 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
+ 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
+ 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
+ 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
+ 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
+ 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
+ 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
+ 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
+ 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
+ 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
+ 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
+ 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
+ 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
+ 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
+ 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
+ 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
+ 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
+ 0xb5, 0x4c, 0xa9, 0x70, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_27[162] =
+ {
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
+ 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
+ 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
+ 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
+ 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
+ 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
+ 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
+ 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
+ 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
+ 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
+ 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
+ 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
+ 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
+ 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
+ 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
+ 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
+ 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
+ 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
+ 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
+ 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
+ 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_28[168] =
+ {
+ 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
+ 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
+ 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
+ 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
+ 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
+ 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
+ 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
+ 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
+ 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
+ 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
+ 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
+ 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
+ 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
+ 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
+ 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
+ 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
+ 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
+ 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
+ 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
+ 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
+ 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
+ 0xbe, 0x1f, 0x99, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_29[174] =
+ {
+ 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
+ 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
+ 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
+ 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
+ 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
+ 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
+ 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
+ 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
+ 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
+ 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
+ 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
+ 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
+ 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
+ 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
+ 0x40, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x15, 0x40, 0xc4, 0x40, 0x00, 0x00,
+ 0xc0, 0x05, 0x60, 0x00, 0x00, 0x00,
+ 0x28, 0x10, 0x04, 0x48, 0x00, 0x00,
+ 0x05, 0x0e, 0x20, 0x80, 0x00, 0x00,
+ 0x64, 0x81, 0x10, 0x08, 0x00, 0x00,
+ 0x81, 0x80, 0xa4, 0x10, 0x00, 0x00,
+ 0x10, 0x9a, 0x0a, 0x80, 0x00, 0x00,
+ 0x84, 0x20, 0x28, 0x68, 0x00, 0x00,
+ 0x12, 0x30, 0x47, 0x80, 0x00, 0x00,
+ 0x62, 0x02, 0x10, 0x10, 0x00, 0x00,
+ 0x28, 0x62, 0x19, 0x00, 0x00, 0x00,
+ 0x0e, 0x08, 0x02, 0x18, 0x00, 0x00,
+ 0x10, 0x85, 0x11, 0x20, 0x00, 0x00,
+ 0x29, 0x50, 0x42, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_3[18] =
+ {
+ 0xac, 0xda, 0xb2, 0x48, 0x00, 0x00,
+ 0x55, 0x6d, 0x55, 0x28, 0x00, 0x00,
+ 0x27, 0xb5, 0x0c, 0xd8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_4[24] =
+ {
+ 0x2c, 0xd8, 0x96, 0xa8, 0x00, 0x00,
+ 0x93, 0x6a, 0x55, 0x50, 0x00, 0x00,
+ 0x1a, 0xb4, 0x69, 0xa8, 0x00, 0x00,
+ 0x47, 0x2d, 0x0f, 0x50, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_5[30] =
+ {
+ 0x64, 0xd9, 0x92, 0x88, 0x00, 0x00,
+ 0xa5, 0x68, 0x95, 0x50, 0x00, 0x00,
+ 0x52, 0xb5, 0x25, 0xa0, 0x00, 0x00,
+ 0x1d, 0xa9, 0x4e, 0x40, 0x00, 0x00,
+ 0x9c, 0x56, 0x38, 0xc0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_6[36] =
+ {
+ 0x4a, 0x55, 0x8a, 0x28, 0x00, 0x00,
+ 0x95, 0x48, 0x55, 0x50, 0x00, 0x00,
+ 0x14, 0xb5, 0x31, 0x18, 0x00, 0x00,
+ 0x51, 0xa9, 0x4a, 0x50, 0x00, 0x00,
+ 0x22, 0x6c, 0x8d, 0x90, 0x00, 0x00,
+ 0x88, 0x8e, 0x29, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_7[42] =
+ {
+ 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
+ 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
+ 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
+ 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
+ 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
+ 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
+ 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_8[48] =
+ {
+ 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
+ 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
+ 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
+ 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
+ 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
+ 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
+ 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
+ 0x72, 0x01, 0x96, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask29_9[54] =
+ {
+ 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
+ 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
+ 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
+ 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
+ 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
+ 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
+ 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
+ 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
+ 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask2_1[2] =
+ {
+ 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask2_2[4] =
+ {
+ 0xc0, 0x00,
+ 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xfc, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_10[60] =
+ {
+ 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
+ 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
+ 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
+ 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
+ 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
+ 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
+ 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
+ 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
+ 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_11[66] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
+ 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
+ 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
+ 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
+ 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
+ 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
+ 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
+ 0x98, 0x41, 0x30, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_12[72] =
+ {
+ 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
+ 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
+ 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
+ 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
+ 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
+ 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
+ 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
+ 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
+ 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
+ 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
+ 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
+ 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_13[78] =
+ {
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
+ 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
+ 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
+ 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
+ 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
+ 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
+ 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
+ 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
+ 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
+ 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
+ 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
+ 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_14[84] =
+ {
+ 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
+ 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
+ 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
+ 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
+ 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
+ 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
+ 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
+ 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
+ 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
+ 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
+ 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
+ 0x28, 0x88, 0x51, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_15[90] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
+ 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
+ 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
+ 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
+ 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
+ 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
+ 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
+ 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
+ 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
+ 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
+ 0x10, 0x98, 0x21, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_16[96] =
+ {
+ 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
+ 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
+ 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
+ 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
+ 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
+ 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
+ 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
+ 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
+ 0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
+ 0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
+ 0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
+ 0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
+ 0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00,
+ 0xce, 0x58, 0xa0, 0x14, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_17[102] =
+ {
+ 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
+ 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
+ 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
+ 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
+ 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
+ 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
+ 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
+ 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
+ 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
+ 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
+ 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
+ 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
+ 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
+ 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
+ 0x90, 0x49, 0x20, 0x90, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_18[108] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
+ 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
+ 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
+ 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
+ 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
+ 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
+ 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
+ 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
+ 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
+ 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
+ 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
+ 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
+ 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
+ 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
+ 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
+ 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
+ 0x00, 0xb2, 0x47, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_19[114] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
+ 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
+ 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
+ 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
+ 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
+ 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
+ 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
+ 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
+ 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
+ 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
+ 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
+ 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
+ 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
+ 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
+ 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
+ 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
+ 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_2[12] =
+ {
+ 0xec, 0xeb, 0xd9, 0xd4, 0x00, 0x00,
+ 0xbb, 0x9d, 0x77, 0x38, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_20[120] =
+ {
+ 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
+ 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
+ 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
+ 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
+ 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
+ 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
+ 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
+ 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
+ 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
+ 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
+ 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
+ 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
+ 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
+ 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
+ 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
+ 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
+ 0x51, 0x88, 0xd1, 0x78, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_21[126] =
+ {
+ 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
+ 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
+ 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
+ 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
+ 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
+ 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
+ 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
+ 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
+ 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
+ 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
+ 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
+ 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
+ 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
+ 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
+ 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
+ 0x98, 0x41, 0x30, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_22[132] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
+ 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
+ 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
+ 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
+ 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
+ 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
+ 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
+ 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
+ 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
+ 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
+ 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
+ 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
+ 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
+ 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
+ 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
+ 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
+ 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
+ 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
+ 0x03, 0x10, 0x18, 0x74, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_23[138] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
+ 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
+ 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
+ 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
+ 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
+ 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
+ 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
+ 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
+ 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
+ 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
+ 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
+ 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
+ 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
+ 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
+ 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
+ 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
+ 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
+ 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
+ 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
+ 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_24[144] =
+ {
+ 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
+ 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
+ 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
+ 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
+ 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
+ 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
+ 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
+ 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
+ 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
+ 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
+ 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
+ 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
+ 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
+ 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
+ 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
+ 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
+ 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
+ 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
+ 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
+ 0xf3, 0x4d, 0x1c, 0x70, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_25[150] =
+ {
+ 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
+ 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
+ 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
+ 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
+ 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
+ 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
+ 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
+ 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
+ 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
+ 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
+ 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
+ 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
+ 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
+ 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
+ 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
+ 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
+ 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
+ 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
+ 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
+ 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
+ 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
+ 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
+ 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_26[156] =
+ {
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
+ 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
+ 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
+ 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
+ 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
+ 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
+ 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
+ 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
+ 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
+ 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
+ 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
+ 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
+ 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
+ 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
+ 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
+ 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
+ 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
+ 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
+ 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
+ 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
+ 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
+ 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
+ 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
+ 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
+ 0x83, 0x11, 0xad, 0xe8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_27[162] =
+ {
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
+ 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
+ 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
+ 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
+ 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
+ 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
+ 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
+ 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
+ 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
+ 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
+ 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
+ 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
+ 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
+ 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
+ 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
+ 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
+ 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
+ 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
+ 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
+ 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
+ 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
+ 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
+ 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
+ 0x28, 0x88, 0x51, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_28[168] =
+ {
+ 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
+ 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
+ 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
+ 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
+ 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
+ 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
+ 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
+ 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
+ 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
+ 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
+ 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
+ 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
+ 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
+ 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
+ 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
+ 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
+ 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
+ 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
+ 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
+ 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
+ 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
+ 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
+ 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
+ 0x94, 0x59, 0x03, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_29[174] =
+ {
+ 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
+ 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
+ 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
+ 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
+ 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
+ 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
+ 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
+ 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
+ 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
+ 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
+ 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
+ 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
+ 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
+ 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
+ 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
+ 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
+ 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
+ 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
+ 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
+ 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
+ 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
+ 0x10, 0x98, 0x21, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_3[18] =
+ {
+ 0xac, 0x93, 0x59, 0x24, 0x00, 0x00,
+ 0x55, 0x4a, 0xaa, 0x94, 0x00, 0x00,
+ 0x43, 0x36, 0x86, 0x6c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_30[180] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
+ 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
+ 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
+ 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
+ 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
+ 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
+ 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
+ 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
+ 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
+ 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
+ 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
+ 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
+ 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
+ 0x10, 0x98, 0x21, 0x30, 0x00, 0x00,
+ 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
+ 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
+ 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
+ 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
+ 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
+ 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
+ 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
+ 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
+ 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
+ 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
+ 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
+ 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
+ 0x46, 0xf1, 0xef, 0xec, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_4[24] =
+ {
+ 0x25, 0xaa, 0x4b, 0x54, 0x00, 0x00,
+ 0x95, 0x55, 0x2a, 0xa8, 0x00, 0x00,
+ 0x1a, 0x6a, 0x34, 0xd4, 0x00, 0x00,
+ 0x43, 0xd4, 0x87, 0xa8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_5[30] =
+ {
+ 0x64, 0xa2, 0xc9, 0x44, 0x00, 0x00,
+ 0x25, 0x54, 0x4a, 0xa8, 0x00, 0x00,
+ 0x49, 0x68, 0x92, 0xd0, 0x00, 0x00,
+ 0x53, 0x90, 0xa7, 0x20, 0x00, 0x00,
+ 0x8e, 0x31, 0x1c, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_6[36] =
+ {
+ 0x62, 0x8a, 0xc5, 0x14, 0x00, 0x00,
+ 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
+ 0x4c, 0x46, 0x98, 0x8c, 0x00, 0x00,
+ 0x52, 0x94, 0xa5, 0x28, 0x00, 0x00,
+ 0x23, 0x64, 0x46, 0xc8, 0x00, 0x00,
+ 0x8a, 0x59, 0x14, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_7[42] =
+ {
+ 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
+ 0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
+ 0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
+ 0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
+ 0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
+ 0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
+ 0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_8[48] =
+ {
+ 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
+ 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
+ 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
+ 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
+ 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
+ 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
+ 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
+ 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask30_9[54] =
+ {
+ 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
+ 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
+ 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
+ 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
+ 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
+ 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
+ 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
+ 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
+ 0x90, 0x49, 0x20, 0x90, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xfe, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_10[60] =
+ {
+ 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
+ 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
+ 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
+ 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
+ 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
+ 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
+ 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
+ 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
+ 0x02, 0x63, 0x09, 0x24, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_11[66] =
+ {
+ 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
+ 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
+ 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
+ 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
+ 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
+ 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
+ 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
+ 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
+ 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
+ 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
+ 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_12[72] =
+ {
+ 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
+ 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
+ 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
+ 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
+ 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
+ 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
+ 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
+ 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
+ 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
+ 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
+ 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
+ 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_13[78] =
+ {
+ 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
+ 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
+ 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
+ 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
+ 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
+ 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
+ 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
+ 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
+ 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
+ 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
+ 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
+ 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
+ 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_14[84] =
+ {
+ 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
+ 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
+ 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
+ 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
+ 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
+ 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
+ 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
+ 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
+ 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
+ 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
+ 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
+ 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
+ 0x28, 0x89, 0x05, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_15[90] =
+ {
+ 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
+ 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
+ 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
+ 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
+ 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
+ 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
+ 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
+ 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
+ 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
+ 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
+ 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
+ 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
+ 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
+ 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
+ 0x10, 0x98, 0x30, 0x44, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_16[96] =
+ {
+ 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
+ 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
+ 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
+ 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
+ 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
+ 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
+ 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
+ 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
+ 0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
+ 0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
+ 0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
+ 0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
+ 0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
+ 0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
+ 0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00,
+ 0x8d, 0x94, 0xa9, 0xe0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_17[102] =
+ {
+ 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
+ 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
+ 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
+ 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
+ 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
+ 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
+ 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
+ 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
+ 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
+ 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
+ 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
+ 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
+ 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
+ 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
+ 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
+ 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
+ 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_18[108] =
+ {
+ 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
+ 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
+ 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
+ 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
+ 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
+ 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
+ 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
+ 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
+ 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
+ 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
+ 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
+ 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
+ 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
+ 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
+ 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
+ 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
+ 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
+ 0xe3, 0xd1, 0x2e, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_19[114] =
+ {
+ 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
+ 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
+ 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
+ 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
+ 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
+ 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
+ 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
+ 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
+ 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
+ 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
+ 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
+ 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
+ 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
+ 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
+ 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
+ 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
+ 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
+ 0x02, 0x63, 0x09, 0x24, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_2[12] =
+ {
+ 0xec, 0xeb, 0x5d, 0x5c, 0x00, 0x00,
+ 0xbb, 0x9c, 0xf2, 0xf2, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_20[120] =
+ {
+ 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
+ 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
+ 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
+ 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
+ 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
+ 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
+ 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
+ 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
+ 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
+ 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
+ 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
+ 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
+ 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
+ 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
+ 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
+ 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
+ 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
+ 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
+ 0x9a, 0xd4, 0x6a, 0x36, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_21[126] =
+ {
+ 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
+ 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
+ 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
+ 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
+ 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
+ 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
+ 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
+ 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
+ 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
+ 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
+ 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
+ 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
+ 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
+ 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
+ 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
+ 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
+ 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
+ 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
+ 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
+ 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_22[132] =
+ {
+ 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
+ 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
+ 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
+ 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
+ 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
+ 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
+ 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
+ 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
+ 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
+ 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
+ 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
+ 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
+ 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
+ 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
+ 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
+ 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
+ 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
+ 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
+ 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
+ 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
+ 0x32, 0x23, 0x73, 0x8e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_23[138] =
+ {
+ 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
+ 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
+ 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
+ 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
+ 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
+ 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
+ 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
+ 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
+ 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
+ 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
+ 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
+ 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
+ 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
+ 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
+ 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
+ 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
+ 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
+ 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
+ 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
+ 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
+ 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
+ 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
+ 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_24[144] =
+ {
+ 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
+ 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
+ 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
+ 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
+ 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
+ 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
+ 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
+ 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
+ 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
+ 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
+ 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
+ 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
+ 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
+ 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
+ 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
+ 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
+ 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
+ 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
+ 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
+ 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
+ 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
+ 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
+ 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
+ 0xf0, 0xdf, 0x91, 0xb6, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_25[150] =
+ {
+ 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
+ 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
+ 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
+ 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
+ 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
+ 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
+ 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
+ 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
+ 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
+ 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
+ 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
+ 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
+ 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
+ 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
+ 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
+ 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
+ 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
+ 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
+ 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
+ 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
+ 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
+ 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
+ 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
+ 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
+ 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_26[156] =
+ {
+ 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
+ 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
+ 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
+ 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
+ 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
+ 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
+ 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
+ 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
+ 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
+ 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
+ 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
+ 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
+ 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
+ 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
+ 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
+ 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
+ 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
+ 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
+ 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
+ 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
+ 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
+ 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
+ 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
+ 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
+ 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
+ 0xc5, 0x75, 0x48, 0xba, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_27[162] =
+ {
+ 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
+ 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
+ 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
+ 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
+ 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
+ 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
+ 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
+ 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
+ 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
+ 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
+ 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
+ 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
+ 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
+ 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
+ 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
+ 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
+ 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
+ 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
+ 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
+ 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
+ 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
+ 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
+ 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
+ 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
+ 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
+ 0x28, 0x89, 0x05, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_28[168] =
+ {
+ 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
+ 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
+ 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
+ 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
+ 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
+ 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
+ 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
+ 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
+ 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
+ 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
+ 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
+ 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
+ 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
+ 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
+ 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
+ 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
+ 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
+ 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
+ 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
+ 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
+ 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
+ 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
+ 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
+ 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
+ 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
+ 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
+ 0xbc, 0x0d, 0xca, 0x28, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_29[174] =
+ {
+ 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
+ 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
+ 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
+ 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
+ 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
+ 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
+ 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
+ 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
+ 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
+ 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
+ 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
+ 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
+ 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
+ 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
+ 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
+ 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
+ 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
+ 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
+ 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
+ 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
+ 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
+ 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
+ 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
+ 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
+ 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
+ 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
+ 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
+ 0x10, 0x98, 0x30, 0x44, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_3[18] =
+ {
+ 0xac, 0x93, 0x5a, 0x5a, 0x00, 0x00,
+ 0x55, 0x4a, 0xec, 0x6c, 0x00, 0x00,
+ 0x43, 0x36, 0x4d, 0xb6, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_30[180] =
+ {
+ 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
+ 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
+ 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
+ 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
+ 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
+ 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
+ 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
+ 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
+ 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
+ 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
+ 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
+ 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
+ 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
+ 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
+ 0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
+ 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
+ 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
+ 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
+ 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
+ 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
+ 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
+ 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
+ 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
+ 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
+ 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
+ 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
+ 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
+ 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
+ 0xe1, 0x4f, 0xe0, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_31[186] =
+ {
+ 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
+ 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
+ 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
+ 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
+ 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
+ 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
+ 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
+ 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
+ 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
+ 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
+ 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
+ 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
+ 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
+ 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
+ 0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
+ 0x62, 0x23, 0x48, 0x20, 0x00, 0x00,
+ 0x31, 0x10, 0x02, 0x54, 0x00, 0x00,
+ 0x58, 0x00, 0x0c, 0x84, 0x00, 0x00,
+ 0x01, 0x12, 0x10, 0xd0, 0x00, 0x00,
+ 0x88, 0x21, 0x03, 0x20, 0x00, 0x00,
+ 0x44, 0x02, 0x01, 0xe0, 0x00, 0x00,
+ 0x29, 0x04, 0xa0, 0x0a, 0x00, 0x00,
+ 0x82, 0xa0, 0x40, 0xa2, 0x00, 0x00,
+ 0x0a, 0x1a, 0x86, 0x10, 0x00, 0x00,
+ 0x11, 0xe0, 0xd1, 0x00, 0x00, 0x00,
+ 0x84, 0x05, 0x00, 0x16, 0x00, 0x00,
+ 0x86, 0x40, 0x20, 0x98, 0x00, 0x00,
+ 0x00, 0x86, 0x24, 0x60, 0x00, 0x00,
+ 0x44, 0x48, 0x81, 0x0a, 0x00, 0x00,
+ 0x10, 0x98, 0x1c, 0x08, 0x00, 0x00,
+ 0x87, 0x74, 0x30, 0x24, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_4[24] =
+ {
+ 0x25, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
+ 0x95, 0x55, 0x55, 0x54, 0x00, 0x00,
+ 0x1a, 0x6a, 0x6a, 0x6a, 0x00, 0x00,
+ 0x43, 0xd5, 0x95, 0x94, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_5[30] =
+ {
+ 0x64, 0xa2, 0xaa, 0xaa, 0x00, 0x00,
+ 0x25, 0x54, 0x54, 0x54, 0x00, 0x00,
+ 0x49, 0x68, 0x48, 0x4a, 0x00, 0x00,
+ 0x53, 0x91, 0x09, 0x90, 0x00, 0x00,
+ 0x8e, 0x30, 0x21, 0x6c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_6[36] =
+ {
+ 0x62, 0x8a, 0xa2, 0xa2, 0x00, 0x00,
+ 0x15, 0x54, 0x14, 0x54, 0x00, 0x00,
+ 0x4c, 0x47, 0x44, 0x2a, 0x00, 0x00,
+ 0x52, 0x95, 0x08, 0x94, 0x00, 0x00,
+ 0x23, 0x64, 0x61, 0x24, 0x00, 0x00,
+ 0x8a, 0x58, 0x09, 0x58, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_7[42] =
+ {
+ 0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
+ 0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
+ 0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
+ 0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
+ 0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
+ 0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
+ 0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_8[48] =
+ {
+ 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
+ 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
+ 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
+ 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
+ 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
+ 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
+ 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
+ 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask31_9[54] =
+ {
+ 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
+ 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
+ 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
+ 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
+ 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
+ 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
+ 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
+ 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
+ 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_10[60] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
+ 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
+ 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
+ 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
+ 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
+ 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
+ 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
+ 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
+ 0x84, 0x92, 0x84, 0x92, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_11[66] =
+ {
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
+ 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
+ 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
+ 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
+ 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
+ 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
+ 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
+ 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_12[72] =
+ {
+ 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
+ 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
+ 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
+ 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
+ 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
+ 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
+ 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
+ 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
+ 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
+ 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
+ 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
+ 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_13[78] =
+ {
+ 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
+ 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
+ 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
+ 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
+ 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
+ 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
+ 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
+ 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
+ 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
+ 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
+ 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_14[84] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
+ 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
+ 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
+ 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
+ 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
+ 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
+ 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
+ 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
+ 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
+ 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
+ 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
+ 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
+ 0x82, 0x88, 0x82, 0x88, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_15[90] =
+ {
+ 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
+ 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
+ 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
+ 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
+ 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
+ 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
+ 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
+ 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
+ 0x18, 0x22, 0x18, 0x22, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_16[96] =
+ {
+ 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
+ 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
+ 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
+ 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
+ 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
+ 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
+ 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
+ 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
+ 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
+ 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
+ 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
+ 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
+ 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
+ 0x18, 0x12, 0x18, 0x12, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_17[102] =
+ {
+ 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
+ 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
+ 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
+ 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
+ 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
+ 0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
+ 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
+ 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
+ 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
+ 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
+ 0x44, 0x64, 0x44, 0x64, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_18[108] =
+ {
+ 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
+ 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
+ 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
+ 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
+ 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
+ 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
+ 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
+ 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
+ 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
+ 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
+ 0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
+ 0x1e, 0xb2, 0xd8, 0x53, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_19[114] =
+ {
+ 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
+ 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
+ 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
+ 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
+ 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
+ 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
+ 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
+ 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
+ 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
+ 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
+ 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
+ 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
+ 0x84, 0x92, 0x84, 0x92, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_2[12] =
+ {
+ 0xae, 0xae, 0xae, 0xae, 0x00, 0x00,
+ 0x79, 0x79, 0x79, 0x79, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_20[120] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
+ 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
+ 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
+ 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
+ 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
+ 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
+ 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
+ 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
+ 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
+ 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
+ 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
+ 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
+ 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
+ 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
+ 0x96, 0xd3, 0xf6, 0xac, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_21[126] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
+ 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
+ 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
+ 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
+ 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
+ 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
+ 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
+ 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
+ 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
+ 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
+ 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
+ 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
+ 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
+ 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
+ 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
+ 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_22[132] =
+ {
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
+ 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
+ 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
+ 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
+ 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
+ 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
+ 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
+ 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
+ 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
+ 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
+ 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
+ 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
+ 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
+ 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
+ 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
+ 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
+ 0xeb, 0xb2, 0x22, 0x89, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_23[138] =
+ {
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
+ 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
+ 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
+ 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
+ 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
+ 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
+ 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
+ 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
+ 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
+ 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
+ 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
+ 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
+ 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
+ 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
+ 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
+ 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
+ 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
+ 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
+ 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
+ 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_24[144] =
+ {
+ 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
+ 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
+ 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
+ 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
+ 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
+ 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
+ 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
+ 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
+ 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
+ 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
+ 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
+ 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
+ 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
+ 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
+ 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
+ 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
+ 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
+ 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
+ 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
+ 0xf3, 0x5a, 0x2f, 0x5d, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_25[150] =
+ {
+ 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
+ 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
+ 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
+ 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
+ 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
+ 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
+ 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
+ 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
+ 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
+ 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
+ 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
+ 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
+ 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
+ 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
+ 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
+ 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
+ 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
+ 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
+ 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
+ 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
+ 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
+ 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
+ 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_26[156] =
+ {
+ 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
+ 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
+ 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
+ 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
+ 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
+ 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
+ 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
+ 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
+ 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
+ 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
+ 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
+ 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
+ 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
+ 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
+ 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
+ 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
+ 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
+ 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
+ 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
+ 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
+ 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
+ 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
+ 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
+ 0x52, 0x15, 0x62, 0x0a, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_27[162] =
+ {
+ 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
+ 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
+ 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
+ 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
+ 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
+ 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
+ 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
+ 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
+ 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
+ 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
+ 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
+ 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
+ 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
+ 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
+ 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
+ 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
+ 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
+ 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
+ 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
+ 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
+ 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
+ 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
+ 0x82, 0x88, 0x82, 0x88, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_28[168] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
+ 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
+ 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
+ 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
+ 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
+ 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
+ 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
+ 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
+ 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
+ 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
+ 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
+ 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
+ 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
+ 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
+ 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
+ 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
+ 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
+ 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
+ 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
+ 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
+ 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
+ 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
+ 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
+ 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
+ 0x7f, 0xe2, 0xbc, 0x01, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_29[174] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
+ 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
+ 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
+ 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
+ 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
+ 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
+ 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
+ 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
+ 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
+ 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
+ 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
+ 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
+ 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
+ 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
+ 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
+ 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
+ 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
+ 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
+ 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
+ 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
+ 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
+ 0x18, 0x22, 0x18, 0x22, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_3[18] =
+ {
+ 0xad, 0x2d, 0xad, 0x2d, 0x00, 0x00,
+ 0x76, 0x36, 0x76, 0x36, 0x00, 0x00,
+ 0x26, 0xdb, 0x26, 0xdb, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_30[180] =
+ {
+ 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
+ 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
+ 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
+ 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
+ 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
+ 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
+ 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
+ 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
+ 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
+ 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
+ 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
+ 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
+ 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
+ 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
+ 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
+ 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
+ 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
+ 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
+ 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
+ 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
+ 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
+ 0x1e, 0x27, 0xe2, 0xd8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_31[186] =
+ {
+ 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
+ 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
+ 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
+ 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
+ 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
+ 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
+ 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
+ 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
+ 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
+ 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
+ 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
+ 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
+ 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
+ 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
+ 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
+ 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
+ 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
+ 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
+ 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
+ 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
+ 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
+ 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
+ 0x18, 0x12, 0x18, 0x12, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_32[192] =
+ {
+ 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
+ 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
+ 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
+ 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
+ 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
+ 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
+ 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
+ 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
+ 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
+ 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
+ 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
+ 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
+ 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
+ 0x18, 0x12, 0x18, 0x12, 0x00, 0x00,
+ 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
+ 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
+ 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
+ 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
+ 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
+ 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
+ 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
+ 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
+ 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
+ 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
+ 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
+ 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
+ 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
+ 0x60, 0xc4, 0x02, 0x02, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_4[24] =
+ {
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0xaa, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
+ 0x35, 0x35, 0x35, 0x35, 0x00, 0x00,
+ 0xca, 0xca, 0xca, 0xca, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_5[30] =
+ {
+ 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
+ 0x2a, 0x2a, 0x2a, 0x2a, 0x00, 0x00,
+ 0x24, 0x25, 0x24, 0x25, 0x00, 0x00,
+ 0x84, 0xc8, 0x84, 0xc8, 0x00, 0x00,
+ 0x10, 0xb6, 0x10, 0xb6, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_6[36] =
+ {
+ 0x51, 0x51, 0x51, 0x51, 0x00, 0x00,
+ 0x0a, 0x2a, 0x0a, 0x2a, 0x00, 0x00,
+ 0xa2, 0x15, 0xa2, 0x15, 0x00, 0x00,
+ 0x84, 0x4a, 0x84, 0x4a, 0x00, 0x00,
+ 0x30, 0x92, 0x30, 0x92, 0x00, 0x00,
+ 0x04, 0xac, 0x04, 0xac, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_7[42] =
+ {
+ 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x91, 0x11, 0x91, 0x11, 0x00, 0x00,
+ 0x2e, 0x08, 0x2e, 0x08, 0x00, 0x00,
+ 0x48, 0x34, 0x48, 0x34, 0x00, 0x00,
+ 0x90, 0x29, 0x90, 0x29, 0x00, 0x00,
+ 0x09, 0x86, 0x09, 0x86, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_8[48] =
+ {
+ 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
+ 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
+ 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
+ 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
+ 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
+ 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
+ 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
+ 0x16, 0x30, 0x16, 0x30, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask32_9[54] =
+ {
+ 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
+ 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
+ 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
+ 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
+ 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
+ 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
+ 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
+ 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
+ 0x44, 0x64, 0x44, 0x64, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_10[60] =
+ {
+ 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
+ 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
+ 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
+ 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
+ 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
+ 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
+ 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_11[66] =
+ {
+ 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
+ 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
+ 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
+ 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
+ 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
+ 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
+ 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
+ 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
+ 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_12[72] =
+ {
+ 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
+ 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
+ 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
+ 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
+ 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
+ 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
+ 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
+ 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
+ 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
+ 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
+ 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_13[78] =
+ {
+ 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
+ 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
+ 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
+ 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
+ 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
+ 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
+ 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
+ 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
+ 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
+ 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
+ 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
+ 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_14[84] =
+ {
+ 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
+ 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
+ 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
+ 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
+ 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
+ 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
+ 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
+ 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
+ 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
+ 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
+ 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
+ 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_15[90] =
+ {
+ 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
+ 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
+ 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
+ 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
+ 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
+ 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
+ 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
+ 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
+ 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
+ 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
+ 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
+ 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
+ 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_16[96] =
+ {
+ 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
+ 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
+ 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
+ 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
+ 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
+ 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
+ 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
+ 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
+ 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
+ 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
+ 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
+ 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
+ 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
+ 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
+ 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
+ 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_17[102] =
+ {
+ 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
+ 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
+ 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
+ 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
+ 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
+ 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
+ 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
+ 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
+ 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
+ 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
+ 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
+ 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
+ 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
+ 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
+ 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_18[108] =
+ {
+ 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
+ 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
+ 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
+ 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
+ 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
+ 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
+ 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
+ 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
+ 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
+ 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
+ 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
+ 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
+ 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
+ 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
+ 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
+ 0x89, 0x53, 0x03, 0xad, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_19[114] =
+ {
+ 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
+ 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
+ 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
+ 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
+ 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
+ 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
+ 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
+ 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
+ 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
+ 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
+ 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
+ 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
+ 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
+ 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_2[12] =
+ {
+ 0xae, 0xae, 0xce, 0xce, 0x00, 0x00,
+ 0x79, 0x79, 0xb9, 0x39, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_20[120] =
+ {
+ 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
+ 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
+ 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
+ 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
+ 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
+ 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
+ 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
+ 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
+ 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
+ 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
+ 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
+ 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
+ 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
+ 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
+ 0x73, 0x5f, 0x5b, 0x0e, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_21[126] =
+ {
+ 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
+ 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
+ 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
+ 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
+ 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
+ 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
+ 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
+ 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
+ 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
+ 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
+ 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
+ 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
+ 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
+ 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
+ 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
+ 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_22[132] =
+ {
+ 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
+ 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
+ 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
+ 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
+ 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
+ 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
+ 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
+ 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
+ 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
+ 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
+ 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
+ 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
+ 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
+ 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
+ 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
+ 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
+ 0xcc, 0xe3, 0x42, 0x6b, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_23[138] =
+ {
+ 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
+ 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
+ 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
+ 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
+ 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
+ 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
+ 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
+ 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
+ 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
+ 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
+ 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
+ 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
+ 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
+ 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
+ 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
+ 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
+ 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
+ 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
+ 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
+ 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_24[144] =
+ {
+ 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
+ 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
+ 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
+ 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
+ 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
+ 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
+ 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
+ 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
+ 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
+ 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
+ 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
+ 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
+ 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
+ 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
+ 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
+ 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
+ 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
+ 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
+ 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
+ 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
+ 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
+ 0xdc, 0x4e, 0xfc, 0x70, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_25[150] =
+ {
+ 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
+ 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
+ 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
+ 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
+ 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
+ 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
+ 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
+ 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
+ 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
+ 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
+ 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
+ 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
+ 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
+ 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
+ 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
+ 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
+ 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
+ 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
+ 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
+ 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
+ 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
+ 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
+ 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_26[156] =
+ {
+ 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
+ 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
+ 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
+ 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
+ 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
+ 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
+ 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
+ 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
+ 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
+ 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
+ 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
+ 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
+ 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
+ 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
+ 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
+ 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
+ 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
+ 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
+ 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
+ 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
+ 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
+ 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
+ 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
+ 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
+ 0xa4, 0xa4, 0xfc, 0x91, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_27[162] =
+ {
+ 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
+ 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
+ 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
+ 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
+ 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
+ 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
+ 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
+ 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
+ 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
+ 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
+ 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
+ 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
+ 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
+ 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
+ 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
+ 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
+ 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
+ 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
+ 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
+ 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
+ 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
+ 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
+ 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
+ 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_28[168] =
+ {
+ 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
+ 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
+ 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
+ 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
+ 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
+ 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
+ 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
+ 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
+ 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
+ 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
+ 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
+ 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
+ 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
+ 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
+ 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
+ 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
+ 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
+ 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
+ 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
+ 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
+ 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
+ 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
+ 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
+ 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
+ 0x1b, 0xf4, 0xaa, 0xec, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_29[174] =
+ {
+ 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
+ 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
+ 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
+ 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
+ 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
+ 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
+ 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
+ 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
+ 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
+ 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
+ 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
+ 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
+ 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
+ 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
+ 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
+ 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
+ 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
+ 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
+ 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
+ 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
+ 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
+ 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
+ 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
+ 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
+ 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_3[18] =
+ {
+ 0xad, 0x2d, 0xcd, 0xcc, 0x00, 0x00,
+ 0x76, 0x36, 0x97, 0x27, 0x00, 0x00,
+ 0x26, 0xdb, 0xb8, 0xd1, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_30[180] =
+ {
+ 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
+ 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
+ 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
+ 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
+ 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
+ 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
+ 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
+ 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
+ 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
+ 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
+ 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
+ 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
+ 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
+ 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
+ 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
+ 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
+ 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
+ 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
+ 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
+ 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
+ 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
+ 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
+ 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
+ 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
+ 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
+ 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
+ 0x6d, 0xd2, 0x8c, 0x00, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_31[186] =
+ {
+ 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
+ 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
+ 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
+ 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
+ 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
+ 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
+ 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
+ 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
+ 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
+ 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
+ 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
+ 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
+ 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
+ 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
+ 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
+ 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
+ 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
+ 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
+ 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
+ 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
+ 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
+ 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
+ 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
+ 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
+ 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
+ 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
+ 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
+ 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
+ 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_32[192] =
+ {
+ 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
+ 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
+ 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
+ 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
+ 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
+ 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
+ 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
+ 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
+ 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
+ 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
+ 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
+ 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
+ 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
+ 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
+ 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
+ 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
+ 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
+ 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
+ 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
+ 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
+ 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
+ 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
+ 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
+ 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
+ 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
+ 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
+ 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
+ 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
+ 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
+ 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
+ 0x73, 0x8e, 0x12, 0xca, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_33[198] =
+ {
+ 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
+ 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
+ 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
+ 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
+ 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
+ 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
+ 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
+ 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
+ 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
+ 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
+ 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
+ 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
+ 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
+ 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
+ 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
+ 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
+ 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
+ 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
+ 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
+ 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
+ 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
+ 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
+ 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
+ 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
+ 0x43, 0x08, 0x25, 0x4c, 0x00, 0x00,
+ 0x68, 0x80, 0x8a, 0x66, 0x00, 0x00,
+ 0x80, 0x0b, 0x91, 0x91, 0x00, 0x00,
+ 0x10, 0x4c, 0x68, 0x42, 0x80, 0x00,
+ 0x12, 0x30, 0x32, 0xa4, 0x00, 0x00,
+ 0x40, 0x85, 0x43, 0x13, 0x00, 0x00,
+ 0x0e, 0x04, 0xc4, 0x30, 0x80, 0x00,
+ 0x18, 0x12, 0x1c, 0x88, 0x80, 0x00,
+ 0xdb, 0x10, 0x3c, 0x09, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_4[24] =
+ {
+ 0x55, 0x55, 0xca, 0xec, 0x00, 0x00,
+ 0xaa, 0xaa, 0xa9, 0x67, 0x00, 0x00,
+ 0x35, 0x35, 0x3a, 0xb1, 0x80, 0x00,
+ 0xca, 0xca, 0x55, 0x5a, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_5[30] =
+ {
+ 0x55, 0x55, 0x55, 0x44, 0x80, 0x00,
+ 0x2a, 0x2a, 0x2a, 0x66, 0x00, 0x00,
+ 0x24, 0x25, 0x25, 0xa1, 0x80, 0x00,
+ 0x84, 0xc8, 0xe2, 0x12, 0x80, 0x00,
+ 0x10, 0xb6, 0x99, 0x98, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_6[36] =
+ {
+ 0x51, 0x51, 0xd1, 0x4c, 0x00, 0x00,
+ 0x0a, 0x2a, 0xa2, 0xc5, 0x00, 0x00,
+ 0xa2, 0x15, 0x95, 0x30, 0x80, 0x00,
+ 0x84, 0x4a, 0xca, 0x0a, 0x80, 0x00,
+ 0x30, 0x92, 0xa4, 0xaa, 0x00, 0x00,
+ 0x04, 0xac, 0x78, 0x15, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_7[42] =
+ {
+ 0x45, 0x51, 0x15, 0x44, 0x80, 0x00,
+ 0x22, 0x2a, 0x8a, 0x23, 0x00, 0x00,
+ 0x91, 0x11, 0x85, 0x91, 0x00, 0x00,
+ 0x2e, 0x08, 0x32, 0x0a, 0x80, 0x00,
+ 0x48, 0x34, 0x58, 0x34, 0x00, 0x00,
+ 0x90, 0x29, 0x2c, 0x0d, 0x00, 0x00,
+ 0x09, 0x86, 0x43, 0xc8, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_8[48] =
+ {
+ 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
+ 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
+ 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
+ 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
+ 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
+ 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
+ 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
+ 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask33_9[54] =
+ {
+ 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
+ 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
+ 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
+ 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
+ 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
+ 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
+ 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
+ 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
+ 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_10[60] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_11[66] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_12[72] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_13[78] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_14[84] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_15[90] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_16[96] =
+ {
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_17[102] =
+ {
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
+ 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
+ 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
+ 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
+ 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
+ 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
+ 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_18[108] =
+ {
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
+ 0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
+ 0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
+ 0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
+ 0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
+ 0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
+ 0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
+ 0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00,
+ 0x89, 0x70, 0xf3, 0xf7, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_19[114] =
+ {
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_2[12] =
+ {
+ 0xce, 0xce, 0x67, 0x67, 0x00, 0x00,
+ 0xb9, 0x39, 0xdc, 0x9c, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_20[120] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x5d, 0xc5, 0xfe, 0xd8, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_21[126] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_22[132] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
+ 0x2a, 0xf7, 0x4f, 0xf5, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_23[138] =
+ {
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_24[144] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x4c, 0xb8, 0x04, 0x74, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_25[150] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_26[156] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
+ 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
+ 0x95, 0x20, 0xe9, 0xef, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_27[162] =
+ {
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_28[168] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x10, 0x6c, 0xff, 0x60, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_29[174] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_3[18] =
+ {
+ 0xcd, 0xcc, 0x66, 0xe6, 0x00, 0x00,
+ 0x97, 0x27, 0x4b, 0x93, 0x80, 0x00,
+ 0xb8, 0xd1, 0xdc, 0x68, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_30[180] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
+ 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
+ 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
+ 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
+ 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
+ 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
+ 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
+ 0x87, 0x93, 0x96, 0xc7, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_31[186] =
+ {
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_32[192] =
+ {
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0xa6, 0x27, 0xa9, 0x4a, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_33[198] =
+ {
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
+ 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
+ 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
+ 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
+ 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
+ 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
+ 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_34[204] =
+ {
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
+ 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
+ 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
+ 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
+ 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
+ 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
+ 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00,
+ 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
+ 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
+ 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
+ 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
+ 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
+ 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
+ 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
+ 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
+ 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
+ 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
+ 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
+ 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
+ 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
+ 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
+ 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
+ 0x30, 0x3c, 0xb3, 0xe6, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_4[24] =
+ {
+ 0xca, 0xec, 0x65, 0x76, 0x00, 0x00,
+ 0xa9, 0x67, 0x54, 0xb3, 0x80, 0x00,
+ 0x3a, 0xb1, 0x9d, 0x58, 0xc0, 0x00,
+ 0x55, 0x5a, 0xaa, 0xad, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_5[30] =
+ {
+ 0x55, 0x44, 0xaa, 0xa2, 0x40, 0x00,
+ 0x2a, 0x66, 0x15, 0x33, 0x00, 0x00,
+ 0x25, 0xa1, 0x92, 0xd0, 0xc0, 0x00,
+ 0xe2, 0x12, 0xf1, 0x09, 0x40, 0x00,
+ 0x99, 0x98, 0x4c, 0xcc, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_6[36] =
+ {
+ 0xd1, 0x4c, 0x68, 0xa6, 0x00, 0x00,
+ 0xa2, 0xc5, 0x51, 0x62, 0x80, 0x00,
+ 0x95, 0x30, 0xca, 0x98, 0x40, 0x00,
+ 0xca, 0x0a, 0xe5, 0x05, 0x40, 0x00,
+ 0xa4, 0xaa, 0x52, 0x55, 0x00, 0x00,
+ 0x78, 0x15, 0x3c, 0x0a, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_7[42] =
+ {
+ 0x15, 0x44, 0x8a, 0xa2, 0x40, 0x00,
+ 0x8a, 0x23, 0x45, 0x11, 0x80, 0x00,
+ 0x85, 0x91, 0x42, 0xc8, 0x80, 0x00,
+ 0x32, 0x0a, 0x99, 0x05, 0x40, 0x00,
+ 0x58, 0x34, 0x2c, 0x1a, 0x00, 0x00,
+ 0x2c, 0x0d, 0x16, 0x06, 0x80, 0x00,
+ 0x43, 0xc8, 0x21, 0xe4, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_8[48] =
+ {
+ 0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
+ 0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
+ 0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
+ 0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
+ 0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
+ 0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
+ 0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
+ 0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask34_9[54] =
+ {
+ 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
+ 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
+ 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
+ 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
+ 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
+ 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
+ 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
+ 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
+ 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_10[60] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
+ 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
+ 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_11[66] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_12[72] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_13[78] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_14[84] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_15[90] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_16[96] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_17[102] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
+ 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
+ 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
+ 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_18[108] =
+ {
+ 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
+ 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
+ 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
+ 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
+ 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
+ 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
+ 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
+ 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
+ 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
+ 0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
+ 0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
+ 0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
+ 0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
+ 0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
+ 0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00,
+ 0x40, 0xc9, 0x65, 0xbe, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_19[114] =
+ {
+ 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
+ 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
+ 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
+ 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
+ 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
+ 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
+ 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
+ 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
+ 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
+ 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
+ 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_2[12] =
+ {
+ 0xce, 0xce, 0x67, 0x33, 0x80, 0x00,
+ 0xb9, 0x39, 0x9c, 0xce, 0x60, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_20[120] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
+ 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
+ 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
+ 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
+ 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
+ 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
+ 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
+ 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
+ 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
+ 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
+ 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
+ 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
+ 0x63, 0x36, 0x5c, 0xd3, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_21[126] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
+ 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
+ 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_22[132] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
+ 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
+ 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
+ 0x84, 0xc7, 0xbc, 0xcc, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_23[138] =
+ {
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_24[144] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x0d, 0xfb, 0x06, 0x89, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_25[150] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_26[156] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
+ 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
+ 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
+ 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
+ 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
+ 0xc4, 0x57, 0x70, 0x47, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_27[162] =
+ {
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_28[168] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x34, 0x4a, 0x80, 0x94, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_29[174] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_3[18] =
+ {
+ 0xcd, 0xcc, 0x66, 0x33, 0x00, 0x00,
+ 0x97, 0x27, 0x13, 0x8a, 0xc0, 0x00,
+ 0xb8, 0xd1, 0xc9, 0x64, 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_30[180] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
+ 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
+ 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
+ 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
+ 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
+ 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
+ 0x32, 0x1b, 0x9f, 0x09, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_31[186] =
+ {
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_32[192] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0xeb, 0x31, 0x7b, 0x80, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_33[198] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
+ 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
+ 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
+ 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_34[204] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
+ 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
+ 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
+ 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
+ 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
+ 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
+ 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
+ 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
+ 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
+ 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
+ 0x70, 0x07, 0xcd, 0x8c, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_35[210] =
+ {
+ 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
+ 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
+ 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
+ 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
+ 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
+ 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
+ 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
+ 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
+ 0x25, 0x2c, 0x26, 0x13, 0x00, 0x00,
+ 0x8a, 0x91, 0x33, 0x19, 0x80, 0x00,
+ 0x91, 0xc0, 0xc8, 0xa4, 0x40, 0x00,
+ 0x68, 0x06, 0xa1, 0x50, 0xa0, 0x00,
+ 0x32, 0xc8, 0x52, 0x29, 0x00, 0x00,
+ 0x43, 0x45, 0x09, 0x84, 0xc0, 0x00,
+ 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
+ 0x1c, 0xa2, 0x44, 0x62, 0x20, 0x00,
+ 0x25, 0x4c, 0x04, 0x82, 0x40, 0x00,
+ 0x8a, 0x66, 0x16, 0x0b, 0x00, 0x00,
+ 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x68, 0x42, 0xe0, 0x70, 0x20, 0x00,
+ 0x32, 0xa4, 0x03, 0x41, 0xa0, 0x00,
+ 0x43, 0x13, 0x64, 0x32, 0x00, 0x00,
+ 0xc4, 0x30, 0xa2, 0x91, 0x40, 0x00,
+ 0x1c, 0x88, 0x98, 0x4c, 0x20, 0x00,
+ 0x3c, 0x09, 0x51, 0x28, 0x80, 0x00,
+ 0xc2, 0x1c, 0x68, 0x01, 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_4[24] =
+ {
+ 0xca, 0xec, 0x76, 0x3b, 0x00, 0x00,
+ 0xa9, 0x67, 0x33, 0x99, 0xc0, 0x00,
+ 0x3a, 0xb1, 0xd8, 0xec, 0x60, 0x00,
+ 0x55, 0x5a, 0xad, 0x56, 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_5[30] =
+ {
+ 0x55, 0x44, 0xa6, 0x53, 0x20, 0x00,
+ 0x2a, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x25, 0xa1, 0x8c, 0xe8, 0x60, 0x00,
+ 0xe2, 0x12, 0xce, 0x44, 0xa0, 0x00,
+ 0x99, 0x98, 0x71, 0xa6, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_6[36] =
+ {
+ 0xd1, 0x4c, 0x66, 0x13, 0x00, 0x00,
+ 0xa2, 0xc5, 0x22, 0xb1, 0x40, 0x00,
+ 0x95, 0x30, 0xd8, 0x4c, 0x20, 0x00,
+ 0xca, 0x0a, 0xc5, 0x42, 0xa0, 0x00,
+ 0xa4, 0xaa, 0x14, 0xa9, 0x80, 0x00,
+ 0x78, 0x15, 0x53, 0x05, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_7[42] =
+ {
+ 0x15, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8a, 0x23, 0x11, 0x88, 0xc0, 0x00,
+ 0x85, 0x91, 0x48, 0xa4, 0x40, 0x00,
+ 0x32, 0x0a, 0x85, 0x42, 0xa0, 0x00,
+ 0x58, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x2c, 0x0d, 0x05, 0x83, 0x40, 0x00,
+ 0x43, 0xc8, 0x70, 0x32, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_8[48] =
+ {
+ 0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
+ 0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
+ 0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
+ 0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
+ 0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
+ 0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
+ 0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask35_9[54] =
+ {
+ 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
+ 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
+ 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
+ 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
+ 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
+ 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
+ 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
+ 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
+ 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_10[60] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
+ 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_11[66] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_12[72] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_13[78] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_14[84] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_15[90] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_16[96] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_17[102] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_18[108] =
+ {
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_19[114] =
+ {
+ 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
+ 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
+ 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
+ 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
+ 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
+ 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
+ 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
+ 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
+ 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_2[12] =
+ {
+ 0xce, 0x67, 0x33, 0x99, 0xc0, 0x00,
+ 0x39, 0x9c, 0xce, 0x67, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_20[120] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
+ 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
+ 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
+ 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
+ 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
+ 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
+ 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
+ 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
+ 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
+ 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
+ 0x45, 0xb9, 0x08, 0x16, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_21[126] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
+ 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_22[132] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
+ 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
+ 0x71, 0x04, 0xba, 0x7b, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_23[138] =
+ {
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_24[144] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x76, 0x3a, 0xeb, 0x17, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_25[150] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_26[156] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
+ 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
+ 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
+ 0xec, 0x58, 0x0e, 0x6c, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_27[162] =
+ {
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_28[168] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x29, 0xfd, 0x91, 0x6f, 0xd0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_29[174] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_3[18] =
+ {
+ 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
+ 0x27, 0x15, 0x89, 0xc5, 0x60, 0x00,
+ 0x92, 0xc9, 0x64, 0xb2, 0x50, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_30[180] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
+ 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
+ 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
+ 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
+ 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
+ 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
+ 0xc5, 0x38, 0xbb, 0x98, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_31[186] =
+ {
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_32[192] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x3a, 0x28, 0x9c, 0x2f, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_33[198] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_34[204] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
+ 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
+ 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
+ 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
+ 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
+ 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
+ 0xf7, 0x5e, 0x66, 0x5b, 0x60, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_35[210] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_36[216] =
+ {
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
+ 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
+ 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
+ 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
+ 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
+ 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
+ 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
+ 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
+ 0xa4, 0x9c, 0x31, 0x13, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_4[24] =
+ {
+ 0xec, 0x76, 0x3b, 0x1d, 0x80, 0x00,
+ 0x67, 0x33, 0x99, 0xcc, 0xe0, 0x00,
+ 0xb1, 0xd8, 0xec, 0x76, 0x30, 0x00,
+ 0x5a, 0xad, 0x56, 0xab, 0x50, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_5[30] =
+ {
+ 0x4c, 0xa6, 0x53, 0x29, 0x90, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
+ 0x19, 0xd0, 0xc6, 0x74, 0x30, 0x00,
+ 0x9c, 0x89, 0x67, 0x22, 0x50, 0x00,
+ 0xe3, 0x4c, 0x38, 0xd3, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_6[36] =
+ {
+ 0xcc, 0x26, 0x33, 0x09, 0x80, 0x00,
+ 0x45, 0x62, 0x91, 0x58, 0xa0, 0x00,
+ 0xb0, 0x98, 0x6c, 0x26, 0x10, 0x00,
+ 0x8a, 0x85, 0x62, 0xa1, 0x50, 0x00,
+ 0x29, 0x53, 0x0a, 0x54, 0xc0, 0x00,
+ 0xa6, 0x0a, 0xa9, 0x82, 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_7[42] =
+ {
+ 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
+ 0x23, 0x11, 0x88, 0xc4, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
+ 0x0a, 0x85, 0x42, 0xa1, 0x50, 0x00,
+ 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
+ 0x0b, 0x06, 0x82, 0xc1, 0xa0, 0x00,
+ 0xe0, 0x64, 0x38, 0x19, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_8[48] =
+ {
+ 0x16, 0x0b, 0x05, 0x82, 0xc0, 0x00,
+ 0xc2, 0x61, 0x30, 0x98, 0x40, 0x00,
+ 0x60, 0xb0, 0x58, 0x2c, 0x10, 0x00,
+ 0x85, 0x42, 0xa1, 0x50, 0xa0, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
+ 0x29, 0x14, 0x8a, 0x45, 0x20, 0x00,
+ 0x11, 0x88, 0xc4, 0x62, 0x30, 0x00,
+ 0xb0, 0x58, 0x2c, 0x16, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask36_9[54] =
+ {
+ 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
+ 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
+ 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
+ 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
+ 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
+ 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
+ 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
+ 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xf8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_10[60] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
+ 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
+ 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_11[66] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_12[72] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_13[78] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_14[84] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_15[90] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_16[96] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_17[102] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_18[108] =
+ {
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_19[114] =
+ {
+ 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
+ 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
+ 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
+ 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
+ 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
+ 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
+ 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
+ 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
+ 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
+ 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_2[12] =
+ {
+ 0xce, 0x67, 0x33, 0x9d, 0xc0, 0x00,
+ 0x39, 0x9c, 0xce, 0x73, 0x38, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_20[120] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
+ 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
+ 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
+ 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
+ 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
+ 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
+ 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
+ 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
+ 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
+ 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
+ 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
+ 0xe5, 0x44, 0xda, 0x3a, 0xc8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_21[126] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
+ 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
+ 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_22[132] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
+ 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
+ 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
+ 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
+ 0xe4, 0xd3, 0xff, 0x5a, 0x28, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_23[138] =
+ {
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_24[144] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0xad, 0x58, 0xb2, 0x36, 0x68, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_25[150] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_26[156] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
+ 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
+ 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
+ 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
+ 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
+ 0x7f, 0xb2, 0x5a, 0xaa, 0x20, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_27[162] =
+ {
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_28[168] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x7b, 0xc4, 0x24, 0xbf, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_29[174] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_3[18] =
+ {
+ 0xcc, 0x66, 0x33, 0x19, 0xc0, 0x00,
+ 0x27, 0x15, 0x89, 0xcb, 0x30, 0x00,
+ 0x92, 0xc9, 0x64, 0xb4, 0x98, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_30[180] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
+ 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
+ 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
+ 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
+ 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
+ 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
+ 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
+ 0x1d, 0x8e, 0x11, 0xb0, 0xe8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_31[186] =
+ {
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_32[192] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0xf7, 0x95, 0x57, 0x8c, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_33[198] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_34[204] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
+ 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
+ 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
+ 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
+ 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
+ 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
+ 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
+ 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
+ 0x31, 0x9c, 0xfb, 0x37, 0xc0, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_35[210] =
+ {
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_36[216] =
+ {
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0xc3, 0xc7, 0xce, 0xd8, 0x50, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_37[222] =
+ {
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
+ 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
+ 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
+ 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
+ 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
+ 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
+ 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
+ 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
+ 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
+ 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
+ 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
+ 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
+ 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
+ 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
+ 0x2c, 0x16, 0x13, 0x09, 0x80, 0x00,
+ 0x91, 0x48, 0x99, 0x8a, 0x20, 0x00,
+ 0xc0, 0xe0, 0x64, 0x54, 0x08, 0x00,
+ 0x06, 0x83, 0x50, 0xa0, 0x98, 0x00,
+ 0xc8, 0x64, 0x29, 0x00, 0x70, 0x00,
+ 0x45, 0x22, 0x84, 0xd0, 0xc0, 0x00,
+ 0x30, 0x98, 0x4c, 0x25, 0x20, 0x00,
+ 0xa2, 0x51, 0x22, 0x28, 0x48, 0x00,
+ 0xd0, 0x03, 0x42, 0x53, 0x00, 0x00,
+ 0xee, 0xf5, 0xb3, 0x66, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_4[24] =
+ {
+ 0xec, 0x76, 0x3b, 0x1c, 0xc0, 0x00,
+ 0x67, 0x33, 0x99, 0xc6, 0x70, 0x00,
+ 0xb1, 0xd8, 0xec, 0x73, 0x18, 0x00,
+ 0x5a, 0xad, 0x56, 0xa5, 0xa8, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_5[30] =
+ {
+ 0x4c, 0xa6, 0x53, 0x39, 0xc0, 0x00,
+ 0x66, 0x33, 0x19, 0x8c, 0x70, 0x00,
+ 0x19, 0xd0, 0xe8, 0x73, 0x18, 0x00,
+ 0x9c, 0x89, 0x64, 0xa9, 0xa8, 0x00,
+ 0xe3, 0x4c, 0x2e, 0x26, 0x60, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_6[36] =
+ {
+ 0xcc, 0x26, 0x13, 0x0d, 0x80, 0x00,
+ 0x45, 0x62, 0x91, 0x5a, 0x20, 0x00,
+ 0xb0, 0x98, 0x4c, 0x34, 0x18, 0x00,
+ 0x8a, 0x85, 0x62, 0xa0, 0xa8, 0x00,
+ 0x29, 0x53, 0x09, 0x82, 0xd0, 0x00,
+ 0xa6, 0x0a, 0xa5, 0x51, 0x40, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_7[42] =
+ {
+ 0x44, 0xa2, 0x71, 0x28, 0xc0, 0x00,
+ 0x23, 0x11, 0x88, 0xc6, 0x60, 0x00,
+ 0x91, 0x48, 0xa4, 0x47, 0x08, 0x00,
+ 0x0a, 0x85, 0x52, 0xa0, 0xa8, 0x00,
+ 0x34, 0x1a, 0x0d, 0x12, 0x50, 0x00,
+ 0x0b, 0x06, 0xa2, 0xd2, 0x80, 0x00,
+ 0xe0, 0x64, 0x32, 0x09, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_8[48] =
+ {
+ 0x16, 0x0b, 0x05, 0x84, 0xe0, 0x00,
+ 0xc2, 0x61, 0x30, 0x91, 0x30, 0x00,
+ 0x60, 0xb0, 0x58, 0x3a, 0x08, 0x00,
+ 0x85, 0x42, 0xa1, 0x44, 0x98, 0x00,
+ 0x4c, 0x26, 0x33, 0x08, 0x50, 0x00,
+ 0x29, 0x14, 0x8a, 0x58, 0xc0, 0x00,
+ 0x11, 0x88, 0xc4, 0x66, 0x30, 0x00,
+ 0xb0, 0x58, 0x2c, 0x03, 0x18, 0x00
+ };
+
+ const WebRtc_UWord8 mask37_9[54] =
+ {
+ 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
+ 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
+ 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
+ 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
+ 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
+ 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
+ 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
+ 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
+ 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xfc, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_10[60] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
+ 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
+ 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
+ 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
+ 0x50, 0x88, 0xca, 0x11, 0x18, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_11[66] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_12[72] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_13[78] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_14[84] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_15[90] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_16[96] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_17[102] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_18[108] =
+ {
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_19[114] =
+ {
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
+ 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
+ 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
+ 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
+ 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
+ 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
+ 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
+ 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
+ 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
+ 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_2[12] =
+ {
+ 0xce, 0x77, 0x19, 0xce, 0xe0, 0x00,
+ 0x39, 0xcc, 0xe7, 0x39, 0x9c, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_20[120] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
+ 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
+ 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
+ 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
+ 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
+ 0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
+ 0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
+ 0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
+ 0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
+ 0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
+ 0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
+ 0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
+ 0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
+ 0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00,
+ 0xe6, 0xbc, 0x88, 0xe3, 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_21[126] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
+ 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
+ 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
+ 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
+ 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_22[132] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
+ 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
+ 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
+ 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
+ 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
+ 0x0c, 0x3c, 0x48, 0x3d, 0x58, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_23[138] =
+ {
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_24[144] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x93, 0xc8, 0xb3, 0xbe, 0x5c, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_25[150] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_26[156] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
+ 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
+ 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
+ 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
+ 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
+ 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
+ 0x4b, 0xab, 0xfc, 0xe6, 0xe8, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_27[162] =
+ {
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_28[168] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x70, 0x1b, 0x5b, 0x2c, 0x0c, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_29[174] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_3[18] =
+ {
+ 0xcc, 0x67, 0x19, 0x8c, 0xe0, 0x00,
+ 0x27, 0x2c, 0xc4, 0xe5, 0x98, 0x00,
+ 0x92, 0xd2, 0x72, 0x5a, 0x4c, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_30[180] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
+ 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
+ 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
+ 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
+ 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
+ 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
+ 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
+ 0x5b, 0x16, 0xdf, 0xb8, 0xd0, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_31[186] =
+ {
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_32[192] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x33, 0x10, 0x02, 0x4e, 0x54, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_33[198] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_34[204] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
+ 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
+ 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
+ 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
+ 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
+ 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
+ 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
+ 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
+ 0x91, 0x48, 0xfa, 0xf0, 0xd8, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_35[210] =
+ {
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_36[216] =
+ {
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x80, 0x95, 0xc2, 0x68, 0x28, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_37[222] =
+ {
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
+ 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
+ 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
+ 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
+ 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
+ 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
+ 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
+ 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
+ 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
+ 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_38[228] =
+ {
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
+ 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
+ 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
+ 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
+ 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
+ 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
+ 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
+ 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
+ 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
+ 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00,
+ 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
+ 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
+ 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
+ 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
+ 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
+ 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
+ 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
+ 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
+ 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
+ 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
+ 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
+ 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
+ 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
+ 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
+ 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
+ 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
+ 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
+ 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
+ 0x8c, 0xed, 0x11, 0x5f, 0x24, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_4[24] =
+ {
+ 0xec, 0x73, 0x1d, 0x8e, 0x60, 0x00,
+ 0x67, 0x19, 0xcc, 0xe3, 0x38, 0x00,
+ 0xb1, 0xcc, 0x76, 0x39, 0x8c, 0x00,
+ 0x5a, 0x96, 0xab, 0x52, 0xd4, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_5[30] =
+ {
+ 0x4c, 0xe7, 0x09, 0x9c, 0xe0, 0x00,
+ 0x66, 0x31, 0xcc, 0xc6, 0x38, 0x00,
+ 0xa1, 0xcc, 0x74, 0x39, 0x8c, 0x00,
+ 0x92, 0xa6, 0xb2, 0x54, 0xd4, 0x00,
+ 0xb8, 0x99, 0x97, 0x13, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_6[36] =
+ {
+ 0x4c, 0x36, 0x09, 0x86, 0xc0, 0x00,
+ 0x45, 0x68, 0x88, 0xad, 0x10, 0x00,
+ 0x30, 0xd0, 0x66, 0x1a, 0x0c, 0x00,
+ 0x8a, 0x82, 0xb1, 0x50, 0x54, 0x00,
+ 0x26, 0x0b, 0x44, 0xc1, 0x68, 0x00,
+ 0x95, 0x45, 0x12, 0xa8, 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_7[42] =
+ {
+ 0xc4, 0xa3, 0x18, 0x94, 0x60, 0x00,
+ 0x23, 0x19, 0x84, 0x63, 0x30, 0x00,
+ 0x91, 0x1c, 0x32, 0x23, 0x84, 0x00,
+ 0x4a, 0x82, 0xa9, 0x50, 0x54, 0x00,
+ 0x34, 0x49, 0x46, 0x89, 0x28, 0x00,
+ 0x8b, 0x4a, 0x11, 0x69, 0x40, 0x00,
+ 0xc8, 0x24, 0xd9, 0x04, 0x98, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_8[48] =
+ {
+ 0x16, 0x13, 0x82, 0xc2, 0x70, 0x00,
+ 0xc2, 0x44, 0xd8, 0x48, 0x98, 0x00,
+ 0x60, 0xe8, 0x2c, 0x1d, 0x04, 0x00,
+ 0x85, 0x12, 0x70, 0xa2, 0x4c, 0x00,
+ 0xcc, 0x21, 0x59, 0x84, 0x28, 0x00,
+ 0x29, 0x63, 0x05, 0x2c, 0x60, 0x00,
+ 0x11, 0x98, 0xc2, 0x33, 0x18, 0x00,
+ 0xb0, 0x0c, 0x76, 0x01, 0x8c, 0x00
+ };
+
+ const WebRtc_UWord8 mask38_9[54] =
+ {
+ 0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
+ 0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
+ 0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
+ 0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
+ 0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
+ 0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
+ 0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
+ 0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
+ 0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xfe, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_10[60] =
+ {
+ 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
+ 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
+ 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
+ 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
+ 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
+ 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
+ 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
+ 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
+ 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
+ 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_11[66] =
+ {
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_12[72] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_13[78] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_14[84] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_15[90] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_16[96] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_17[102] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_18[108] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_19[114] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
+ 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
+ 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
+ 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
+ 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
+ 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
+ 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
+ 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
+ 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_2[12] =
+ {
+ 0xce, 0x77, 0x1d, 0xc7, 0x70, 0x00,
+ 0x39, 0xcc, 0xf3, 0x3c, 0xce, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_20[120] =
+ {
+ 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
+ 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
+ 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
+ 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
+ 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
+ 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
+ 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
+ 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
+ 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
+ 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
+ 0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
+ 0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
+ 0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
+ 0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
+ 0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
+ 0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
+ 0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00,
+ 0x0d, 0xba, 0x52, 0x38, 0xbc, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_21[126] =
+ {
+ 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
+ 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
+ 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
+ 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
+ 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
+ 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
+ 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
+ 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
+ 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
+ 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_22[132] =
+ {
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
+ 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
+ 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
+ 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
+ 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
+ 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
+ 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
+ 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
+ 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
+ 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
+ 0xfc, 0x5a, 0xb2, 0x13, 0x12, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_23[138] =
+ {
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_24[144] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0xac, 0xbc, 0xf0, 0xff, 0x62, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_25[150] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_26[156] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
+ 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
+ 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
+ 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
+ 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
+ 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
+ 0x10, 0x17, 0x44, 0x72, 0xec, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_27[162] =
+ {
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_28[168] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x86, 0xb6, 0x04, 0xbc, 0x1e, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_29[174] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_3[18] =
+ {
+ 0xcc, 0x67, 0x19, 0xc6, 0x70, 0x00,
+ 0x27, 0x2c, 0xca, 0xb2, 0xac, 0x00,
+ 0x92, 0xd2, 0x76, 0x2d, 0x46, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_30[180] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
+ 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
+ 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
+ 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
+ 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
+ 0xb3, 0x1d, 0x13, 0x03, 0x5a, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_31[186] =
+ {
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_32[192] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x83, 0x1a, 0x3c, 0x2a, 0x7a, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_33[198] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_34[204] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
+ 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
+ 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
+ 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
+ 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
+ 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
+ 0xc6, 0xbb, 0x7e, 0xd9, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_35[210] =
+ {
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_36[216] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x11, 0x78, 0xfe, 0x43, 0xd6, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_37[222] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
+ 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
+ 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
+ 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
+ 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
+ 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
+ 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
+ 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
+ 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_38[228] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
+ 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
+ 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
+ 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
+ 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
+ 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
+ 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
+ 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
+ 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
+ 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
+ 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
+ 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
+ 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
+ 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
+ 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
+ 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
+ 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
+ 0x9e, 0xd8, 0x3c, 0x7e, 0x2e, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_39[234] =
+ {
+ 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
+ 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
+ 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
+ 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
+ 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
+ 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
+ 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
+ 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
+ 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
+ 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
+ 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
+ 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
+ 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
+ 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
+ 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
+ 0x4c, 0x27, 0x09, 0x82, 0x60, 0x00,
+ 0x66, 0x71, 0x8a, 0x22, 0x88, 0x00,
+ 0x91, 0x40, 0xf4, 0x0d, 0x02, 0x00,
+ 0x42, 0x90, 0xa0, 0x98, 0x26, 0x00,
+ 0xa4, 0x29, 0x40, 0x70, 0x1c, 0x00,
+ 0x13, 0x5a, 0x10, 0xc4, 0x30, 0x00,
+ 0x30, 0x93, 0x45, 0x21, 0x48, 0x00,
+ 0x88, 0xac, 0x28, 0x4a, 0x12, 0x00,
+ 0x09, 0x0c, 0xd3, 0x04, 0xc0, 0x00,
+ 0x4c, 0x26, 0x06, 0x11, 0x84, 0x00,
+ 0x66, 0x28, 0x89, 0xc2, 0x70, 0x00,
+ 0x91, 0x50, 0x3c, 0x67, 0x18, 0x00,
+ 0x42, 0x82, 0x70, 0x3c, 0x0e, 0x00,
+ 0xa4, 0x01, 0xc4, 0x29, 0x0a, 0x00,
+ 0x13, 0x43, 0x0a, 0x52, 0x94, 0x00,
+ 0x30, 0x94, 0x96, 0x85, 0xa0, 0x00,
+ 0x88, 0xa1, 0x24, 0xd1, 0x34, 0x00,
+ 0x09, 0x4c, 0x0b, 0x0a, 0xc2, 0x00,
+ 0xcd, 0x98, 0x43, 0x30, 0xcc, 0x00,
+ 0x1d, 0x04, 0x3e, 0xf1, 0xb4, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_4[24] =
+ {
+ 0xec, 0x73, 0x1c, 0xc7, 0x30, 0x00,
+ 0x67, 0x19, 0xc6, 0x71, 0x9c, 0x00,
+ 0xb1, 0xcc, 0x73, 0x1c, 0xc6, 0x00,
+ 0x5a, 0x96, 0xa5, 0xa9, 0x6a, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_5[30] =
+ {
+ 0x4c, 0xe7, 0x19, 0xc6, 0x70, 0x00,
+ 0x66, 0x31, 0xcc, 0x73, 0x1c, 0x00,
+ 0xa1, 0xcc, 0x73, 0x1c, 0xa6, 0x00,
+ 0x92, 0xa6, 0xa5, 0x6a, 0x6a, 0x00,
+ 0xb8, 0x99, 0x96, 0x8b, 0x94, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_6[36] =
+ {
+ 0x4c, 0x36, 0x09, 0x83, 0x60, 0x00,
+ 0x45, 0x68, 0x8a, 0x26, 0x88, 0x00,
+ 0x30, 0xd0, 0x64, 0x1d, 0x06, 0x00,
+ 0x8a, 0x82, 0xb0, 0xa8, 0x2a, 0x00,
+ 0x26, 0x0b, 0x40, 0xd0, 0xd4, 0x00,
+ 0x95, 0x45, 0x13, 0x44, 0x30, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_7[42] =
+ {
+ 0xc4, 0xa3, 0x09, 0xc2, 0x30, 0x00,
+ 0x23, 0x19, 0x86, 0x65, 0x80, 0x00,
+ 0x91, 0x1c, 0x22, 0x01, 0xd6, 0x00,
+ 0x4a, 0x82, 0xb0, 0x2a, 0x2a, 0x00,
+ 0x34, 0x49, 0x44, 0x98, 0x94, 0x00,
+ 0x8b, 0x4a, 0x1a, 0x84, 0x60, 0x00,
+ 0xc8, 0x24, 0xc1, 0x94, 0x4c, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_8[48] =
+ {
+ 0x16, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xc2, 0x44, 0xd1, 0x34, 0x4c, 0x00,
+ 0x60, 0xe8, 0x3a, 0x0e, 0x82, 0x00,
+ 0x85, 0x12, 0x64, 0x99, 0x26, 0x00,
+ 0xcc, 0x21, 0x5c, 0x52, 0x14, 0x00,
+ 0x29, 0x63, 0x18, 0xc6, 0x30, 0x00,
+ 0x11, 0x98, 0xc6, 0x31, 0x8c, 0x00,
+ 0xb0, 0x0c, 0x63, 0x18, 0xc6, 0x00
+ };
+
+ const WebRtc_UWord8 mask39_9[54] =
+ {
+ 0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
+ 0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
+ 0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
+ 0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
+ 0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
+ 0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
+ 0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
+ 0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
+ 0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00
+ };
+
+ const WebRtc_UWord8 mask3_1[2] =
+ {
+ 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask3_2[4] =
+ {
+ 0xc0, 0x00,
+ 0xa0, 0x00
+ };
+
+ const WebRtc_UWord8 mask3_3[6] =
+ {
+ 0xc0, 0x00,
+ 0xa0, 0x00,
+ 0x60, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_10[60] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_11[66] =
+ {
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_12[72] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_13[78] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_14[84] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_15[90] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_16[96] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_17[102] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_18[108] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_19[114] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_2[12] =
+ {
+ 0xee, 0x3b, 0x8e, 0xe3, 0xb8, 0x00,
+ 0x99, 0xe6, 0x79, 0x9e, 0x67, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_20[120] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_21[126] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_22[132] =
+ {
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x89, 0xee, 0x1f, 0x38, 0xca, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_23[138] =
+ {
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_24[144] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x68, 0xde, 0x83, 0xa9, 0xcf, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_25[150] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_26[156] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
+ 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
+ 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
+ 0x06, 0x8e, 0x8c, 0x1a, 0xd2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_27[162] =
+ {
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_28[168] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x68, 0x0e, 0x9b, 0x52, 0xb6, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_29[174] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_3[18] =
+ {
+ 0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
+ 0x55, 0x95, 0x65, 0x59, 0x56, 0x00,
+ 0xb1, 0x6a, 0x3b, 0x16, 0xa3, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_30[180] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
+ 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
+ 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
+ 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
+ 0xe1, 0x47, 0x04, 0x05, 0x47, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_31[186] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_32[192] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x03, 0x0c, 0x46, 0x10, 0xc5, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_33[198] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_34[204] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
+ 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
+ 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
+ 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
+ 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
+ 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
+ 0x87, 0x3c, 0x08, 0x19, 0x31, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_35[210] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_36[216] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x1e, 0xb9, 0x3d, 0x25, 0xcc, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_37[222] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_38[228] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
+ 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
+ 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
+ 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
+ 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
+ 0xea, 0xaa, 0x20, 0xa2, 0x1b, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_39[234] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_4[24] =
+ {
+ 0xe6, 0x39, 0x8e, 0x63, 0x98, 0x00,
+ 0x33, 0x8c, 0xe3, 0x38, 0xce, 0x00,
+ 0x98, 0xe6, 0x39, 0x8e, 0x63, 0x00,
+ 0x2d, 0x4b, 0x52, 0xd4, 0xb5, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_40[240] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
+ 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
+ 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
+ 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
+ 0xa6, 0xf3, 0xab, 0x1b, 0x87, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_5[30] =
+ {
+ 0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
+ 0x63, 0x98, 0xe6, 0x39, 0x8e, 0x00,
+ 0x98, 0xe5, 0x39, 0x8e, 0x53, 0x00,
+ 0x2b, 0x53, 0x52, 0xb5, 0x35, 0x00,
+ 0xb4, 0x5c, 0xab, 0x45, 0xca, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_6[36] =
+ {
+ 0x4c, 0x1b, 0x04, 0xc1, 0xb0, 0x00,
+ 0x51, 0x34, 0x45, 0x13, 0x44, 0x00,
+ 0x20, 0xe8, 0x32, 0x0e, 0x83, 0x00,
+ 0x85, 0x41, 0x58, 0x54, 0x15, 0x00,
+ 0x06, 0x86, 0xa0, 0x68, 0x6a, 0x00,
+ 0x9a, 0x21, 0x89, 0xa2, 0x18, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_7[42] =
+ {
+ 0x4e, 0x11, 0x84, 0xe1, 0x18, 0x00,
+ 0x33, 0x2c, 0x03, 0x32, 0xc0, 0x00,
+ 0x10, 0x0e, 0xb1, 0x00, 0xeb, 0x00,
+ 0x81, 0x51, 0x58, 0x15, 0x15, 0x00,
+ 0x24, 0xc4, 0xa2, 0x4c, 0x4a, 0x00,
+ 0xd4, 0x23, 0x0d, 0x42, 0x30, 0x00,
+ 0x0c, 0xa2, 0x60, 0xca, 0x26, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_8[48] =
+ {
+ 0x27, 0x09, 0xc2, 0x70, 0x9c, 0x00,
+ 0x89, 0xa2, 0x68, 0x9a, 0x26, 0x00,
+ 0xd0, 0x74, 0x1d, 0x07, 0x41, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
+ 0xe2, 0x90, 0xae, 0x29, 0x0a, 0x00,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x31, 0x8c, 0x63, 0x18, 0xc6, 0x00,
+ 0x18, 0xc6, 0x31, 0x8c, 0x63, 0x00
+ };
+
+ const WebRtc_UWord8 mask40_9[54] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
+ 0x62, 0x38, 0xc6, 0x23, 0x8c, 0x00,
+ 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
+ 0xe1, 0x48, 0x5e, 0x14, 0x85, 0x00,
+ 0x13, 0x94, 0xa1, 0x39, 0x4a, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
+ 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
+ 0x49, 0x86, 0x54, 0x98, 0x65, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_10[60] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_11[66] =
+ {
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_12[72] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_13[78] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_14[84] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_15[90] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_16[96] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_17[102] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_18[108] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_19[114] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_2[12] =
+ {
+ 0xee, 0x3b, 0x8e, 0xe3, 0xb3, 0x00,
+ 0x99, 0xe6, 0x79, 0x9e, 0x6e, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_20[120] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_21[126] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_22[132] =
+ {
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x33, 0x09, 0x6e, 0x49, 0x6b, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_23[138] =
+ {
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_24[144] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x45, 0xa6, 0xef, 0xc9, 0xc3, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_25[150] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_26[156] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
+ 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
+ 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
+ 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
+ 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
+ 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
+ 0x6f, 0x72, 0xf1, 0xe7, 0x1a, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_27[162] =
+ {
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_28[168] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x61, 0x2c, 0xfa, 0x25, 0x38, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_29[174] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_3[18] =
+ {
+ 0xce, 0x33, 0x8c, 0xe3, 0x2b, 0x00,
+ 0x55, 0x95, 0x65, 0x5d, 0xc5, 0x00,
+ 0xb1, 0x6a, 0x3a, 0x8e, 0xd8, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_30[180] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
+ 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
+ 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
+ 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
+ 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
+ 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
+ 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
+ 0x59, 0x53, 0x31, 0x62, 0x15, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_31[186] =
+ {
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_32[192] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0xca, 0xbb, 0xcb, 0x6d, 0xaa, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_33[198] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_34[204] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
+ 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
+ 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
+ 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
+ 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
+ 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
+ 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
+ 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
+ 0xbd, 0x37, 0x3f, 0x75, 0x36, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_35[210] =
+ {
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_36[216] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0xc1, 0xb1, 0x80, 0xbe, 0x3e, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_37[222] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_38[228] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
+ 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
+ 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
+ 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
+ 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
+ 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
+ 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
+ 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
+ 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
+ 0xea, 0xc8, 0xbb, 0xd4, 0x5d, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_39[234] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_4[24] =
+ {
+ 0xe6, 0x39, 0x8e, 0x63, 0x13, 0x00,
+ 0x33, 0x8c, 0xe3, 0x38, 0xc5, 0x80,
+ 0x98, 0xe6, 0x39, 0x8d, 0x2c, 0x80,
+ 0x2d, 0x4b, 0x52, 0xd4, 0xb2, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_40[240] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0xe8, 0x07, 0x18, 0x9a, 0x02, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_41[246] =
+ {
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
+ 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
+ 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
+ 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
+ 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
+ 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
+ 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
+ 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
+ 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
+ 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
+ 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
+ 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
+ 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
+ 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
+ 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
+ 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
+ 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
+ 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
+ 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
+ 0x4e, 0x13, 0x84, 0xc1, 0x19, 0x00,
+ 0xe3, 0x38, 0xc5, 0x10, 0xcc, 0x80,
+ 0x81, 0xe0, 0x7a, 0x06, 0x64, 0x00,
+ 0x21, 0x48, 0x50, 0x4c, 0x16, 0x00,
+ 0x52, 0x94, 0xa0, 0x3a, 0x02, 0x80,
+ 0xb4, 0x2d, 0x08, 0x62, 0x11, 0x00,
+ 0x26, 0x89, 0xa2, 0x91, 0x01, 0x80,
+ 0x58, 0x56, 0x14, 0x24, 0x2a, 0x00,
+ 0x19, 0x86, 0x69, 0x81, 0xa0, 0x00,
+ 0xf7, 0x8d, 0xa3, 0x08, 0x40, 0x80,
+ 0x2b, 0xea, 0x4d, 0xf4, 0xc1, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_5[30] =
+ {
+ 0xce, 0x33, 0x8c, 0xe3, 0x1b, 0x00,
+ 0x63, 0x98, 0xe6, 0x39, 0x8d, 0x80,
+ 0x98, 0xe5, 0x39, 0x8c, 0x76, 0x80,
+ 0x2b, 0x53, 0x54, 0xd6, 0xb5, 0x00,
+ 0xb4, 0x5c, 0xab, 0x26, 0xca, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_6[36] =
+ {
+ 0x4c, 0x1b, 0x04, 0xc1, 0x91, 0x00,
+ 0x51, 0x34, 0x45, 0x11, 0x45, 0x00,
+ 0x20, 0xe8, 0x32, 0x0e, 0xa0, 0x80,
+ 0x85, 0x41, 0x58, 0x54, 0x12, 0x80,
+ 0x06, 0x86, 0xa0, 0x68, 0x0d, 0x80,
+ 0x9a, 0x21, 0x88, 0xa2, 0x43, 0x00
+ };
+
+ const WebRtc_UWord8 mask41_7[42] =
+ {
+ 0x4e, 0x11, 0x8c, 0x61, 0x19, 0x00,
+ 0x33, 0x2c, 0x03, 0x30, 0x4c, 0x80,
+ 0x10, 0x0e, 0xb1, 0x86, 0x74, 0x00,
+ 0x81, 0x51, 0x54, 0x54, 0x2d, 0x00,
+ 0x24, 0xc4, 0xa1, 0x2d, 0x42, 0x80,
+ 0xd4, 0x23, 0x0b, 0x42, 0x83, 0x00,
+ 0x0c, 0xa2, 0x62, 0x99, 0x21, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_8[48] =
+ {
+ 0x27, 0x09, 0xc0, 0x70, 0xa7, 0x00,
+ 0x89, 0xa2, 0x64, 0x9a, 0x82, 0x80,
+ 0xd0, 0x74, 0x1b, 0x07, 0xa0, 0x00,
+ 0x24, 0xc9, 0x32, 0x4c, 0x5c, 0x00,
+ 0xe2, 0x90, 0xa5, 0x28, 0x0e, 0x80,
+ 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
+ 0x31, 0x8c, 0x63, 0x19, 0x41, 0x80,
+ 0x18, 0xc6, 0x31, 0x8c, 0x70, 0x80
+ };
+
+ const WebRtc_UWord8 mask41_9[54] =
+ {
+ 0x4e, 0x13, 0x84, 0xe1, 0x11, 0x00,
+ 0x62, 0x38, 0xc6, 0x21, 0xa0, 0x80,
+ 0x81, 0xe0, 0x78, 0x0e, 0x94, 0x00,
+ 0xe1, 0x48, 0x5a, 0x15, 0x05, 0x00,
+ 0x13, 0x94, 0xa5, 0x30, 0x06, 0x80,
+ 0xb4, 0x2d, 0x0a, 0x42, 0x43, 0x00,
+ 0x26, 0x89, 0xa1, 0x6a, 0x08, 0x80,
+ 0x58, 0x56, 0x15, 0x84, 0x52, 0x00,
+ 0x49, 0x86, 0x52, 0x98, 0x68, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_10[60] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_11[66] =
+ {
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_12[72] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_13[78] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_14[84] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_15[90] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_16[96] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_17[102] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_18[108] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_19[114] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_2[12] =
+ {
+ 0xee, 0x3b, 0x37, 0x71, 0xd9, 0x80,
+ 0x99, 0xe6, 0xec, 0xcf, 0x37, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_20[120] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_21[126] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
+ 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
+ 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
+ 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
+ 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
+ 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
+ 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
+ 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
+ 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
+ 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
+ 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_22[132] =
+ {
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0xdb, 0x36, 0xb0, 0x33, 0x14, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_23[138] =
+ {
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_24[144] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x2e, 0x1c, 0x92, 0xbb, 0x07, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_25[150] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_26[156] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
+ 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
+ 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
+ 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
+ 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
+ 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
+ 0xb8, 0x41, 0xed, 0xa3, 0x77, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_27[162] =
+ {
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_28[168] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0xc3, 0x3c, 0x56, 0xc2, 0x30, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_29[174] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_3[18] =
+ {
+ 0xce, 0x32, 0xb6, 0x71, 0x95, 0x80,
+ 0x55, 0xdc, 0x52, 0xae, 0xe2, 0x80,
+ 0xa8, 0xed, 0x8d, 0x47, 0x6c, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_30[180] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
+ 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
+ 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
+ 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
+ 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
+ 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
+ 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
+ 0xf5, 0xdd, 0x0d, 0x58, 0xeb, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_31[186] =
+ {
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_32[192] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0xf9, 0x1f, 0xb6, 0xe1, 0x09, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_33[198] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_34[204] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
+ 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
+ 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
+ 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
+ 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
+ 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
+ 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
+ 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
+ 0xf8, 0xbf, 0xf6, 0x76, 0x1b, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_35[210] =
+ {
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_36[216] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x57, 0xc7, 0x03, 0xf9, 0xc6, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_37[222] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_38[228] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
+ 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
+ 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
+ 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
+ 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
+ 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
+ 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
+ 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
+ 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
+ 0x05, 0x19, 0x55, 0xee, 0xe2, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_39[234] =
+ {
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_4[24] =
+ {
+ 0xe6, 0x31, 0x37, 0x31, 0x89, 0x80,
+ 0x33, 0x8c, 0x59, 0x9c, 0x62, 0xc0,
+ 0x98, 0xd2, 0xcc, 0xc6, 0x96, 0x40,
+ 0x2d, 0x4b, 0x29, 0x6a, 0x59, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_40[240] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0xf9, 0xdb, 0x5d, 0x7a, 0xd4, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_41[246] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
+ 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
+ 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
+ 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
+ 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
+ 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
+ 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
+ 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
+ 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
+ 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
+ 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_42[252] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
+ 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
+ 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
+ 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
+ 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
+ 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
+ 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
+ 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
+ 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
+ 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
+ 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80,
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
+ 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
+ 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
+ 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
+ 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
+ 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
+ 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
+ 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
+ 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
+ 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
+ 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
+ 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
+ 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
+ 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
+ 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
+ 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
+ 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
+ 0xea, 0x9e, 0x23, 0xb3, 0x65, 0x00
+ };
+
+ const WebRtc_UWord8 mask42_5[30] =
+ {
+ 0xce, 0x31, 0xb6, 0x71, 0x8d, 0x80,
+ 0x63, 0x98, 0xdb, 0x1c, 0xc6, 0xc0,
+ 0x98, 0xc7, 0x6c, 0xc6, 0x3b, 0x40,
+ 0x4d, 0x6b, 0x52, 0x6b, 0x5a, 0x80,
+ 0xb2, 0x6c, 0xad, 0x93, 0x65, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_6[36] =
+ {
+ 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
+ 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
+ 0x20, 0xea, 0x09, 0x07, 0x50, 0x40,
+ 0x85, 0x41, 0x2c, 0x2a, 0x09, 0x40,
+ 0x06, 0x80, 0xd8, 0x34, 0x06, 0xc0,
+ 0x8a, 0x24, 0x34, 0x51, 0x21, 0x80
+ };
+
+ const WebRtc_UWord8 mask42_7[42] =
+ {
+ 0xc6, 0x11, 0x96, 0x30, 0x8c, 0x80,
+ 0x33, 0x04, 0xc9, 0x98, 0x26, 0x40,
+ 0x18, 0x67, 0x40, 0xc3, 0x3a, 0x00,
+ 0x45, 0x42, 0xd2, 0x2a, 0x16, 0x80,
+ 0x12, 0xd4, 0x28, 0x96, 0xa1, 0x40,
+ 0xb4, 0x28, 0x35, 0xa1, 0x41, 0x80,
+ 0x29, 0x92, 0x19, 0x4c, 0x90, 0xc0
+ };
+
+ const WebRtc_UWord8 mask42_8[48] =
+ {
+ 0x07, 0x0a, 0x70, 0x38, 0x53, 0x80,
+ 0x49, 0xa8, 0x2a, 0x4d, 0x41, 0x40,
+ 0xb0, 0x7a, 0x05, 0x83, 0xd0, 0x00,
+ 0x24, 0xc5, 0xc1, 0x26, 0x2e, 0x00,
+ 0x52, 0x80, 0xea, 0x94, 0x07, 0x40,
+ 0xc6, 0x31, 0x86, 0x31, 0x8c, 0x00,
+ 0x31, 0x94, 0x19, 0x8c, 0xa0, 0xc0,
+ 0x18, 0xc7, 0x08, 0xc6, 0x38, 0x40
+ };
+
+ const WebRtc_UWord8 mask42_9[54] =
+ {
+ 0x4e, 0x11, 0x12, 0x70, 0x88, 0x80,
+ 0x62, 0x1a, 0x0b, 0x10, 0xd0, 0x40,
+ 0x80, 0xe9, 0x44, 0x07, 0x4a, 0x00,
+ 0xa1, 0x50, 0x55, 0x0a, 0x82, 0x80,
+ 0x53, 0x00, 0x6a, 0x98, 0x03, 0x40,
+ 0xa4, 0x24, 0x35, 0x21, 0x21, 0x80,
+ 0x16, 0xa0, 0x88, 0xb5, 0x04, 0x40,
+ 0x58, 0x45, 0x22, 0xc2, 0x29, 0x00,
+ 0x29, 0x86, 0x81, 0x4c, 0x34, 0x00
+ };
+
+ const WebRtc_UWord8 mask43_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xe0
+ };
+
+ const WebRtc_UWord8 mask43_10[60] =
+ {
+ 0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
+ 0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
+ 0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
+ 0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
+ 0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
+ 0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
+ 0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
+ 0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
+ 0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
+ 0x30, 0x84, 0xab, 0x11, 0x20, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_11[66] =
+ {
+ 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
+ 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
+ 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
+ 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
+ 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
+ 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
+ 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
+ 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
+ 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
+ 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
+ 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_12[72] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
+ 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
+ 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
+ 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
+ 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
+ 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
+ };
+
+ const WebRtc_UWord8 mask43_13[78] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_14[84] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_15[90] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_16[96] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_17[102] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask43_18[108] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_19[114] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
+ };
+
+ const WebRtc_UWord8 mask43_2[12] =
+ {
+ 0xee, 0x3b, 0x37, 0x66, 0xec, 0xc0,
+ 0x99, 0xe6, 0xec, 0xdd, 0x9b, 0xa0
+ };
+
+ const WebRtc_UWord8 mask43_20[120] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_21[126] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
+ 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
+ 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
+ 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
+ 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
+ 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
+ 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
+ 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_22[132] =
+ {
+ 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
+ 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
+ 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
+ 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
+ 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
+ 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
+ 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
+ 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
+ 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
+ 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
+ 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
+ 0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
+ 0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
+ 0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
+ 0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
+ 0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
+ 0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
+ 0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
+ 0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
+ 0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
+ 0x30, 0x84, 0xab, 0x11, 0x20, 0x20,
+ 0xfe, 0x2c, 0x85, 0xcc, 0x24, 0x80
+ };
+
+ const WebRtc_UWord8 mask43_23[138] =
+ {
+ 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
+ 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
+ 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
+ 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
+ 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
+ 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
+ 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
+ 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
+ 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
+ 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
+ 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
+ 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
+ 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
+ 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
+ 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
+ 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
+ };
+
+ const WebRtc_UWord8 mask43_24[144] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
+ 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
+ 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
+ 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
+ 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
+ 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
+ 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
+ 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
+ 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
+ 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
+ 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
+ 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
+ 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
+ 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
+ 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
+ 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
+ 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
+ 0xf9, 0xb1, 0x26, 0x6c, 0x51, 0xe0
+ };
+
+ const WebRtc_UWord8 mask43_25[150] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
+ 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
+ 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
+ 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
+ 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
+ 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_26[156] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
+ 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
+ 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
+ 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
+ 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
+ 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
+ 0xef, 0x84, 0x77, 0xca, 0x0d, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_27[162] =
+ {
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_28[168] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x16, 0xc9, 0x53, 0x1e, 0xc4, 0x00
+ };
+
+ const WebRtc_UWord8 mask43_29[174] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_3[18] =
+ {
+ 0xce, 0x32, 0xb6, 0x56, 0xca, 0xc0,
+ 0x55, 0xdc, 0x57, 0x8a, 0xf1, 0x40,
+ 0xa8, 0xed, 0x8d, 0xb1, 0xae, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_30[180] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
+ 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
+ 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
+ 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
+ 0x79, 0x4a, 0x8f, 0x42, 0x79, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_31[186] =
+ {
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_32[192] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0xd1, 0xd1, 0x11, 0xa4, 0xed, 0xc0
+ };
+
+ const WebRtc_UWord8 mask43_33[198] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
+ };
+
+ const WebRtc_UWord8 mask43_34[204] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
+ 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
+ 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
+ 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
+ 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
+ 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
+ 0x76, 0x81, 0x4d, 0x33, 0x66, 0x00
+ };
+
+ const WebRtc_UWord8 mask43_35[210] =
+ {
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_36[216] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0xa3, 0x85, 0x0a, 0xb5, 0x11, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_37[222] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
+ };
+
+ const WebRtc_UWord8 mask43_38[228] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
+ 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
+ 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
+ 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
+ 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
+ 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
+ 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
+ 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
+ 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
+ 0x9a, 0x16, 0x97, 0x21, 0xb9, 0x80
+ };
+
+ const WebRtc_UWord8 mask43_39[234] =
+ {
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_4[24] =
+ {
+ 0xe6, 0x31, 0x36, 0x26, 0xc4, 0xc0,
+ 0x33, 0x8c, 0x59, 0x8b, 0x31, 0x60,
+ 0x98, 0xd2, 0xca, 0x59, 0x4b, 0x20,
+ 0x2d, 0x4b, 0x29, 0x65, 0x2c, 0xa0
+ };
+
+ const WebRtc_UWord8 mask43_40[240] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x3a, 0xab, 0x77, 0x63, 0xef, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_41[246] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
+ 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
+ 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
+ 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
+ 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
+ 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
+ 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
+ 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_42[252] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
+ 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
+ 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
+ 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
+ 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
+ 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
+ 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
+ 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
+ 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
+ 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
+ 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
+ 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
+ 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
+ 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
+ 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
+ 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
+ 0x26, 0x84, 0x10, 0xcd, 0xf7, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_43[258] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
+ 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
+ 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
+ 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
+ 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
+ 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
+ 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
+ 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
+ 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
+ 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
+ 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
+ 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
+ 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
+ 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
+ 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
+ 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
+ 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
+ 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
+ 0x4c, 0x19, 0x12, 0x32, 0x46, 0x40,
+ 0x51, 0x14, 0x51, 0x99, 0x33, 0x20,
+ 0xa0, 0x6a, 0x44, 0xc8, 0x99, 0x00,
+ 0x04, 0xc1, 0x30, 0x2c, 0x05, 0x80,
+ 0x03, 0xb4, 0x04, 0x05, 0x80, 0xa0,
+ 0x86, 0x20, 0x94, 0x22, 0x84, 0x40,
+ 0x29, 0x08, 0x4a, 0x03, 0x40, 0x60,
+ 0x42, 0x43, 0x08, 0x54, 0x0a, 0x80,
+ 0x98, 0x12, 0x83, 0x40, 0x68, 0x00,
+ 0x30, 0x84, 0xa8, 0x81, 0x10, 0x20,
+ 0x4c, 0x11, 0x91, 0x82, 0x30, 0x40,
+ 0x51, 0x0c, 0xcb, 0x22, 0x64, 0x40,
+ 0xa0, 0x66, 0x42, 0x8a, 0x51, 0x40,
+ 0x04, 0xc1, 0x65, 0x48, 0xa9, 0x00,
+ 0x03, 0xa0, 0x28, 0x26, 0x04, 0xc0,
+ 0x86, 0x21, 0x16, 0x80, 0xd0, 0x00,
+ 0x29, 0x10, 0x1c, 0x12, 0x82, 0x40,
+ 0x42, 0x42, 0xa1, 0x09, 0x21, 0x20,
+ 0x98, 0x1a, 0x00, 0x61, 0x0c, 0x20,
+ 0x30, 0x84, 0x0a, 0x50, 0x4a, 0x00,
+ 0xdf, 0x4c, 0x10, 0x95, 0x12, 0xa0,
+ 0x72, 0x06, 0x94, 0xf6, 0x74, 0x40
+ };
+
+ const WebRtc_UWord8 mask43_5[30] =
+ {
+ 0xce, 0x31, 0xb6, 0x36, 0xc6, 0xc0,
+ 0x63, 0x98, 0xdb, 0x1b, 0x63, 0x60,
+ 0x98, 0xc7, 0x68, 0xed, 0x1d, 0xa0,
+ 0x4d, 0x6b, 0x55, 0x6a, 0xad, 0x40,
+ 0xb2, 0x6c, 0xad, 0x95, 0xb2, 0xa0
+ };
+
+ const WebRtc_UWord8 mask43_6[36] =
+ {
+ 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
+ 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
+ 0x20, 0xea, 0x0d, 0x41, 0xa8, 0x20,
+ 0x85, 0x41, 0x2e, 0x25, 0x04, 0xa0,
+ 0x06, 0x80, 0xd8, 0x1b, 0x03, 0x60,
+ 0x8a, 0x24, 0x34, 0x86, 0x90, 0xc0
+ };
+
+ const WebRtc_UWord8 mask43_7[42] =
+ {
+ 0xc6, 0x11, 0x96, 0x32, 0x46, 0x40,
+ 0x33, 0x04, 0xc8, 0x99, 0x33, 0x20,
+ 0x18, 0x67, 0x44, 0x68, 0x9d, 0x00,
+ 0x45, 0x42, 0xd4, 0x5a, 0x0b, 0x40,
+ 0x12, 0xd4, 0x2a, 0x95, 0x50, 0xa0,
+ 0xb4, 0x28, 0x35, 0x16, 0xa0, 0xc0,
+ 0x29, 0x92, 0x1b, 0x0d, 0x41, 0x60
+ };
+
+ const WebRtc_UWord8 mask43_8[48] =
+ {
+ 0x07, 0x0a, 0x71, 0x44, 0x29, 0xc0,
+ 0x49, 0xa8, 0x29, 0x0f, 0xa0, 0x20,
+ 0xb0, 0x7a, 0x07, 0x48, 0xe8, 0x00,
+ 0x24, 0xc5, 0xc0, 0xb8, 0x17, 0x00,
+ 0x52, 0x80, 0xec, 0x1d, 0x02, 0xa0,
+ 0xc6, 0x31, 0x82, 0x30, 0xc7, 0x40,
+ 0x31, 0x94, 0x1a, 0x83, 0x50, 0x60,
+ 0x18, 0xc7, 0x08, 0xe1, 0x1c, 0x20
+ };
+
+ const WebRtc_UWord8 mask43_9[54] =
+ {
+ 0x4e, 0x11, 0x12, 0x22, 0x46, 0x40,
+ 0x62, 0x1a, 0x09, 0x41, 0x68, 0x60,
+ 0x80, 0xe9, 0x41, 0x28, 0xa5, 0x00,
+ 0xa1, 0x50, 0x52, 0xc8, 0x51, 0x00,
+ 0x53, 0x00, 0x68, 0x1d, 0x01, 0xa0,
+ 0xa4, 0x24, 0x36, 0x06, 0x80, 0xc0,
+ 0x16, 0xa0, 0x8d, 0x11, 0x82, 0x20,
+ 0x58, 0x45, 0x20, 0xa4, 0x16, 0x80,
+ 0x29, 0x86, 0x84, 0xd0, 0x1c, 0x00
+ };
+
+ const WebRtc_UWord8 mask44_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xf0
+ };
+
+ const WebRtc_UWord8 mask44_10[60] =
+ {
+ 0xc0, 0x38, 0x8b, 0x00, 0xe2, 0x20,
+ 0x30, 0x0e, 0x28, 0xc0, 0x38, 0xa0,
+ 0xe8, 0x07, 0x03, 0xa0, 0x1c, 0x00,
+ 0x85, 0x08, 0xaa, 0x14, 0x22, 0xa0,
+ 0xd0, 0x92, 0x13, 0x42, 0x48, 0x40,
+ 0x86, 0x50, 0x4a, 0x19, 0x41, 0x20,
+ 0x4a, 0x68, 0x0d, 0x29, 0xa0, 0x30,
+ 0x01, 0xa0, 0x74, 0x06, 0x81, 0xd0,
+ 0x4c, 0x81, 0x91, 0x32, 0x06, 0x40,
+ 0x62, 0x24, 0x05, 0x88, 0x90, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_11[66] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_12[72] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
+ 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
+ 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
+ 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
+ 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
+ 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
+ };
+
+ const WebRtc_UWord8 mask44_13[78] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
+ };
+
+ const WebRtc_UWord8 mask44_14[84] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
+ };
+
+ const WebRtc_UWord8 mask44_15[90] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_16[96] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_17[102] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
+ };
+
+ const WebRtc_UWord8 mask44_18[108] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
+ };
+
+ const WebRtc_UWord8 mask44_19[114] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
+ };
+
+ const WebRtc_UWord8 mask44_2[12] =
+ {
+ 0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
+ 0x9b, 0xb3, 0x76, 0x6e, 0xcd, 0xd0
+ };
+
+ const WebRtc_UWord8 mask44_20[120] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_21[126] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_22[132] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_23[138] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
+ 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
+ 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
+ 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
+ 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
+ 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
+ };
+
+ const WebRtc_UWord8 mask44_24[144] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
+ 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
+ 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
+ 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
+ 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
+ 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x15, 0x0f, 0x44, 0x6d, 0x9d, 0xa0
+ };
+
+ const WebRtc_UWord8 mask44_25[150] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
+ 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
+ 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
+ 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
+ 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
+ 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
+ };
+
+ const WebRtc_UWord8 mask44_26[156] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
+ 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
+ 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
+ 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
+ 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
+ 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
+ 0x02, 0xcb, 0x64, 0xb8, 0x55, 0x80
+ };
+
+ const WebRtc_UWord8 mask44_27[162] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
+ };
+
+ const WebRtc_UWord8 mask44_28[168] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x66, 0x26, 0x6c, 0x91, 0xc7, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_29[174] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_3[18] =
+ {
+ 0xca, 0xd9, 0x5b, 0x2b, 0x65, 0x60,
+ 0xf1, 0x5e, 0x2b, 0xc5, 0x78, 0xa0,
+ 0xb6, 0x35, 0xc6, 0xd8, 0xd7, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_30[180] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
+ 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
+ 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
+ 0x60, 0xf4, 0x75, 0x84, 0x90, 0xc0
+ };
+
+ const WebRtc_UWord8 mask44_31[186] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_32[192] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x3e, 0x39, 0x86, 0x5c, 0xd9, 0xd0
+ };
+
+ const WebRtc_UWord8 mask44_33[198] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
+ };
+
+ const WebRtc_UWord8 mask44_34[204] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
+ 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
+ 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
+ 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
+ 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
+ 0xb5, 0xc7, 0xe8, 0x0c, 0xb9, 0x90
+ };
+
+ const WebRtc_UWord8 mask44_35[210] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
+ };
+
+ const WebRtc_UWord8 mask44_36[216] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0xa6, 0x92, 0x01, 0x65, 0x91, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_37[222] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
+ };
+
+ const WebRtc_UWord8 mask44_38[228] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
+ 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
+ 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
+ 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
+ 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
+ 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
+ 0x43, 0x64, 0xf2, 0xe5, 0x5d, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_39[234] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_4[24] =
+ {
+ 0xc4, 0xd8, 0x9b, 0x13, 0x62, 0x60,
+ 0x31, 0x66, 0x2c, 0xc5, 0x98, 0xb0,
+ 0x4b, 0x29, 0x65, 0x2c, 0xa5, 0x90,
+ 0x2c, 0xa5, 0x94, 0xb2, 0x96, 0x50
+ };
+
+ const WebRtc_UWord8 mask44_40[240] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0xd8, 0x2a, 0x16, 0x26, 0x51, 0x40
+ };
+
+ const WebRtc_UWord8 mask44_41[246] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_42[252] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
+ 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
+ 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
+ 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
+ 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
+ 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
+ 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
+ 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
+ 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
+ 0xd9, 0xc1, 0x6f, 0xa8, 0x1c, 0x90
+ };
+
+ const WebRtc_UWord8 mask44_43[258] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
+ };
+
+ const WebRtc_UWord8 mask44_44[264] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20,
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
+ 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
+ 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
+ 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
+ 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
+ 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
+ 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
+ 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
+ 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
+ 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
+ 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
+ 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
+ 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
+ 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
+ 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
+ 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
+ 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
+ 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
+ 0xb5, 0x1c, 0x1c, 0x21, 0xac, 0xa0
+ };
+
+ const WebRtc_UWord8 mask44_5[30] =
+ {
+ 0xc6, 0xd8, 0xdb, 0x1b, 0x63, 0x60,
+ 0x63, 0x6c, 0x6d, 0x8d, 0xb1, 0xb0,
+ 0x1d, 0xa3, 0xb4, 0x76, 0x8e, 0xd0,
+ 0xad, 0x55, 0xaa, 0xb5, 0x56, 0xa0,
+ 0xb2, 0xb6, 0x56, 0xca, 0xd9, 0x50
+ };
+
+ const WebRtc_UWord8 mask44_6[36] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
+ 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
+ 0xa8, 0x35, 0x06, 0xa0, 0xd4, 0x10,
+ 0xc4, 0xa0, 0x97, 0x12, 0x82, 0x50,
+ 0x03, 0x60, 0x6c, 0x0d, 0x81, 0xb0,
+ 0x90, 0xd2, 0x1a, 0x43, 0x48, 0x60
+ };
+
+ const WebRtc_UWord8 mask44_7[42] =
+ {
+ 0xc6, 0x48, 0xcb, 0x19, 0x23, 0x20,
+ 0x13, 0x26, 0x64, 0x4c, 0x99, 0x90,
+ 0x8d, 0x13, 0xa2, 0x34, 0x4e, 0x80,
+ 0x8b, 0x41, 0x6a, 0x2d, 0x05, 0xa0,
+ 0x52, 0xaa, 0x15, 0x4a, 0xa8, 0x50,
+ 0xa2, 0xd4, 0x1a, 0x8b, 0x50, 0x60,
+ 0x61, 0xa8, 0x2d, 0x86, 0xa0, 0xb0
+ };
+
+ const WebRtc_UWord8 mask44_8[48] =
+ {
+ 0x28, 0x85, 0x38, 0xa2, 0x14, 0xe0,
+ 0x21, 0xf4, 0x04, 0x87, 0xd0, 0x10,
+ 0xe9, 0x1d, 0x03, 0xa4, 0x74, 0x00,
+ 0x17, 0x02, 0xe0, 0x5c, 0x0b, 0x80,
+ 0x83, 0xa0, 0x56, 0x0e, 0x81, 0x50,
+ 0x46, 0x18, 0xe9, 0x18, 0x63, 0xa0,
+ 0x50, 0x6a, 0x0d, 0x41, 0xa8, 0x30,
+ 0x1c, 0x23, 0x84, 0x70, 0x8e, 0x10
+ };
+
+ const WebRtc_UWord8 mask44_9[54] =
+ {
+ 0x44, 0x48, 0xc9, 0x11, 0x23, 0x20,
+ 0x28, 0x2d, 0x0c, 0xa0, 0xb4, 0x30,
+ 0x25, 0x14, 0xa0, 0x94, 0x52, 0x80,
+ 0x59, 0x0a, 0x21, 0x64, 0x28, 0x80,
+ 0x03, 0xa0, 0x34, 0x0e, 0x80, 0xd0,
+ 0xc0, 0xd0, 0x1b, 0x03, 0x40, 0x60,
+ 0xa2, 0x30, 0x46, 0x88, 0xc1, 0x10,
+ 0x14, 0x82, 0xd0, 0x52, 0x0b, 0x40,
+ 0x9a, 0x03, 0x82, 0x68, 0x0e, 0x00
+ };
+
+ const WebRtc_UWord8 mask45_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xf8
+ };
+
+ const WebRtc_UWord8 mask45_10[60] =
+ {
+ 0xc0, 0x38, 0x89, 0x91, 0x28, 0xa0,
+ 0x30, 0x0e, 0x29, 0x45, 0x22, 0x88,
+ 0xe8, 0x07, 0x02, 0xa4, 0x40, 0x68,
+ 0x85, 0x08, 0xa8, 0x13, 0x12, 0x10,
+ 0xd0, 0x92, 0x13, 0x40, 0x05, 0x10,
+ 0x86, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x4a, 0x68, 0x0c, 0x84, 0xdc, 0x00,
+ 0x01, 0xa0, 0x74, 0x30, 0x84, 0x88,
+ 0x4c, 0x81, 0x91, 0x28, 0x2b, 0x00,
+ 0x62, 0x24, 0x04, 0x4a, 0xd1, 0x40
+ };
+
+ const WebRtc_UWord8 mask45_11[66] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
+ };
+
+ const WebRtc_UWord8 mask45_12[72] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
+ 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
+ 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
+ 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
+ 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
+ 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
+ };
+
+ const WebRtc_UWord8 mask45_13[78] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
+ };
+
+ const WebRtc_UWord8 mask45_14[84] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
+ };
+
+ const WebRtc_UWord8 mask45_15[90] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
+ };
+
+ const WebRtc_UWord8 mask45_16[96] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
+ };
+
+ const WebRtc_UWord8 mask45_17[102] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
+ };
+
+ const WebRtc_UWord8 mask45_18[108] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
+ };
+
+ const WebRtc_UWord8 mask45_19[114] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
+ };
+
+ const WebRtc_UWord8 mask45_2[12] =
+ {
+ 0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
+ 0x9b, 0xb3, 0x76, 0x6e, 0xc9, 0xd8
+ };
+
+ const WebRtc_UWord8 mask45_20[120] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
+ };
+
+ const WebRtc_UWord8 mask45_21[126] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
+ };
+
+ const WebRtc_UWord8 mask45_22[132] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
+ };
+
+ const WebRtc_UWord8 mask45_23[138] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
+ 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
+ 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
+ 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
+ 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
+ 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
+ };
+
+ const WebRtc_UWord8 mask45_24[144] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
+ 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
+ 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
+ 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
+ 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
+ 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x95, 0x91, 0xad, 0xd9, 0x86, 0x98
+ };
+
+ const WebRtc_UWord8 mask45_25[150] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
+ 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
+ 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
+ 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
+ 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
+ 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
+ };
+
+ const WebRtc_UWord8 mask45_26[156] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
+ 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
+ 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
+ 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
+ 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
+ 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
+ 0xb0, 0xfd, 0xb2, 0xf3, 0x8a, 0xc0
+ };
+
+ const WebRtc_UWord8 mask45_27[162] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
+ };
+
+ const WebRtc_UWord8 mask45_28[168] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x44, 0x46, 0x28, 0xfb, 0x66, 0x80
+ };
+
+ const WebRtc_UWord8 mask45_29[174] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
+ };
+
+ const WebRtc_UWord8 mask45_3[18] =
+ {
+ 0xca, 0xd9, 0x5b, 0x2b, 0x4d, 0x90,
+ 0xf1, 0x5e, 0x2b, 0xc5, 0x24, 0xe8,
+ 0xb6, 0x35, 0xc5, 0xd8, 0x9f, 0x40
+ };
+
+ const WebRtc_UWord8 mask45_30[180] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
+ 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
+ 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
+ 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
+ 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
+ 0x1c, 0xc9, 0x43, 0x25, 0xa7, 0x00
+ };
+
+ const WebRtc_UWord8 mask45_31[186] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
+ };
+
+ const WebRtc_UWord8 mask45_32[192] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x40, 0x7e, 0xc1, 0x30, 0x29, 0x50
+ };
+
+ const WebRtc_UWord8 mask45_33[198] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
+ };
+
+ const WebRtc_UWord8 mask45_34[204] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
+ 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
+ 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
+ 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
+ 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
+ 0x1f, 0x78, 0x45, 0x5e, 0x46, 0x50
+ };
+
+ const WebRtc_UWord8 mask45_35[210] =
+ {
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
+ };
+
+ const WebRtc_UWord8 mask45_36[216] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0xd0, 0x1a, 0xf0, 0x14, 0xf0, 0xe8
+ };
+
+ const WebRtc_UWord8 mask45_37[222] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
+ };
+
+ const WebRtc_UWord8 mask45_38[228] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
+ 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
+ 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
+ 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
+ 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
+ 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
+ 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
+ 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
+ 0x04, 0x67, 0x1b, 0xba, 0x1d, 0xa0
+ };
+
+ const WebRtc_UWord8 mask45_39[234] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
+ };
+
+ const WebRtc_UWord8 mask45_4[24] =
+ {
+ 0xc4, 0xd8, 0x9b, 0x13, 0x45, 0x90,
+ 0x31, 0x66, 0x2c, 0xc5, 0x8a, 0x58,
+ 0x4b, 0x29, 0x65, 0x2c, 0x91, 0x68,
+ 0x2c, 0xa5, 0x94, 0xb2, 0xa2, 0xa8
+ };
+
+ const WebRtc_UWord8 mask45_40[240] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0xd9, 0x40, 0x46, 0xe6, 0x4f, 0xd8
+ };
+
+ const WebRtc_UWord8 mask45_41[246] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
+ };
+
+ const WebRtc_UWord8 mask45_42[252] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
+ 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
+ 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
+ 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
+ 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
+ 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
+ 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
+ 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
+ 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
+ 0xac, 0xcc, 0x04, 0x41, 0x97, 0x30
+ };
+
+ const WebRtc_UWord8 mask45_43[258] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
+ };
+
+ const WebRtc_UWord8 mask45_44[264] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0xf8, 0x40, 0xe3, 0x2e, 0x16, 0x00
+ };
+
+ const WebRtc_UWord8 mask45_45[270] =
+ {
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
+ 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
+ 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
+ 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
+ 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
+ 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
+ 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
+ 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
+ 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
+ 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
+ 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
+ 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
+ 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
+ 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
+ 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
+ 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
+ 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
+ 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
+ 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
+ 0x64, 0x4c, 0x89, 0x19, 0x08, 0x30,
+ 0x51, 0x4a, 0x28, 0xcc, 0x81, 0x18,
+ 0xa9, 0x15, 0x22, 0x64, 0x20, 0x28,
+ 0x04, 0xc0, 0x98, 0x16, 0x10, 0xc0,
+ 0xd0, 0x1a, 0x02, 0x02, 0xc0, 0x88,
+ 0x82, 0x50, 0x4a, 0x11, 0x0a, 0x40,
+ 0x21, 0x24, 0x25, 0x01, 0xcc, 0x00,
+ 0x0c, 0x21, 0x84, 0x2a, 0x04, 0x48,
+ 0x4a, 0x09, 0x41, 0xa0, 0x31, 0x00,
+ 0x12, 0xa2, 0x54, 0x40, 0x92, 0x10,
+ 0x9e, 0xce, 0x88, 0xc1, 0x45, 0x00,
+ 0xfb, 0x97, 0x5d, 0x7d, 0x42, 0x20
+ };
+
+ const WebRtc_UWord8 mask45_5[30] =
+ {
+ 0xc6, 0xd8, 0xdb, 0x1b, 0x29, 0xb0,
+ 0x63, 0x6c, 0x6d, 0x8d, 0xb2, 0x58,
+ 0x1d, 0xa3, 0xb4, 0x76, 0x87, 0x70,
+ 0xad, 0x55, 0xaa, 0xb5, 0x54, 0xe0,
+ 0xb2, 0xb6, 0x56, 0xca, 0xdc, 0x18
+ };
+
+ const WebRtc_UWord8 mask45_6[36] =
+ {
+ 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
+ 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
+ 0xa8, 0x35, 0x04, 0x32, 0x90, 0xc0,
+ 0xc4, 0xa0, 0x96, 0x84, 0x89, 0x18,
+ 0x03, 0x60, 0x6c, 0x4a, 0x84, 0x70,
+ 0x90, 0xd2, 0x1a, 0x29, 0x17, 0x00
+ };
+
+ const WebRtc_UWord8 mask45_7[42] =
+ {
+ 0xc6, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x13, 0x26, 0x64, 0xcc, 0x90, 0x98,
+ 0x8d, 0x13, 0xa2, 0x46, 0x48, 0x48,
+ 0x8b, 0x41, 0x6a, 0x90, 0x81, 0x28,
+ 0x52, 0xaa, 0x15, 0x42, 0x83, 0x50,
+ 0xa2, 0xd4, 0x1a, 0x13, 0x16, 0x00,
+ 0x61, 0xa8, 0x2c, 0x25, 0xc4, 0x30
+ };
+
+ const WebRtc_UWord8 mask45_8[48] =
+ {
+ 0x28, 0x85, 0x38, 0x32, 0x10, 0x30,
+ 0x21, 0xf4, 0x06, 0x01, 0xc0, 0x18,
+ 0xe9, 0x1d, 0x02, 0x82, 0x21, 0x20,
+ 0x17, 0x02, 0xe0, 0x15, 0x00, 0xc8,
+ 0x83, 0xa0, 0x55, 0x0c, 0x0a, 0x08,
+ 0x46, 0x18, 0xe8, 0x68, 0x05, 0x40,
+ 0x50, 0x6a, 0x0d, 0x80, 0x9c, 0x00,
+ 0x1c, 0x23, 0x84, 0x50, 0xe2, 0x80
+ };
+
+ const WebRtc_UWord8 mask45_9[54] =
+ {
+ 0x44, 0x48, 0xc9, 0x19, 0x29, 0xb0,
+ 0x28, 0x2d, 0x0d, 0x89, 0xf2, 0x10,
+ 0x25, 0x14, 0xa2, 0x30, 0x12, 0x20,
+ 0x59, 0x0a, 0x20, 0x05, 0xd0, 0x88,
+ 0x03, 0xa0, 0x34, 0x1e, 0x0c, 0x18,
+ 0xc0, 0xd0, 0x1a, 0x82, 0x01, 0xc8,
+ 0xa2, 0x30, 0x44, 0x62, 0xc5, 0x08,
+ 0x14, 0x82, 0xd2, 0x44, 0x02, 0x48,
+ 0x9a, 0x03, 0x81, 0xe0, 0x00, 0x70
+ };
+
+ const WebRtc_UWord8 mask46_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xfc
+ };
+
+ const WebRtc_UWord8 mask46_10[60] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
+ };
+
+ const WebRtc_UWord8 mask46_11[66] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
+ };
+
+ const WebRtc_UWord8 mask46_12[72] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
+ 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
+ 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
+ 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
+ 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
+ 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8
+ };
+
+ const WebRtc_UWord8 mask46_13[78] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
+ };
+
+ const WebRtc_UWord8 mask46_14[84] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
+ };
+
+ const WebRtc_UWord8 mask46_15[90] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
+ };
+
+ const WebRtc_UWord8 mask46_16[96] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
+ };
+
+ const WebRtc_UWord8 mask46_17[102] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
+ };
+
+ const WebRtc_UWord8 mask46_18[108] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x82, 0x32, 0x57, 0x04, 0x64, 0xac
+ };
+
+ const WebRtc_UWord8 mask46_19[114] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
+ };
+
+ const WebRtc_UWord8 mask46_2[12] =
+ {
+ 0xec, 0xdd, 0x99, 0xd9, 0xbb, 0x30,
+ 0x9b, 0xb2, 0x77, 0x37, 0x64, 0xec
+ };
+
+ const WebRtc_UWord8 mask46_20[120] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
+ };
+
+ const WebRtc_UWord8 mask46_21[126] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
+ };
+
+ const WebRtc_UWord8 mask46_22[132] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
+ };
+
+ const WebRtc_UWord8 mask46_23[138] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
+ 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
+ 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
+ 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
+ 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
+ 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
+ 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
+ 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
+ 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
+ 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
+ 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
+ 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
+ };
+
+ const WebRtc_UWord8 mask46_24[144] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
+ 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
+ 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
+ 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
+ 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
+ 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x18, 0x8b, 0x03, 0xb4, 0x3b, 0x10
+ };
+
+ const WebRtc_UWord8 mask46_25[150] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
+ 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
+ 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
+ 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
+ 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
+ 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
+ };
+
+ const WebRtc_UWord8 mask46_26[156] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
+ 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
+ 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
+ 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
+ 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
+ 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
+ 0x2d, 0x6d, 0xd2, 0x57, 0xd6, 0x2c
+ };
+
+ const WebRtc_UWord8 mask46_27[162] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
+ };
+
+ const WebRtc_UWord8 mask46_28[168] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0x52, 0xf9, 0x72, 0xd9, 0x68
+ };
+
+ const WebRtc_UWord8 mask46_29[174] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
+ };
+
+ const WebRtc_UWord8 mask46_3[18] =
+ {
+ 0xca, 0xd3, 0x65, 0x95, 0xa6, 0xc8,
+ 0xf1, 0x49, 0x3b, 0xe2, 0x92, 0x74,
+ 0x76, 0x27, 0xd0, 0xec, 0x4f, 0xa0
+ };
+
+ const WebRtc_UWord8 mask46_30[180] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
+ 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
+ 0x99, 0xf6, 0x0a, 0xdd, 0x16, 0xb0
+ };
+
+ const WebRtc_UWord8 mask46_31[186] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
+ };
+
+ const WebRtc_UWord8 mask46_32[192] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x69, 0xcd, 0xeb, 0x51, 0xc9, 0xa8
+ };
+
+ const WebRtc_UWord8 mask46_33[198] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
+ };
+
+ const WebRtc_UWord8 mask46_34[204] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
+ 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
+ 0x60, 0xf0, 0x13, 0xf0, 0x4d, 0xe0
+ };
+
+ const WebRtc_UWord8 mask46_35[210] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x82, 0x32, 0x57, 0x04, 0x64, 0xac
+ };
+
+ const WebRtc_UWord8 mask46_36[216] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x6c, 0x3a, 0x45, 0x70, 0xd7, 0x00
+ };
+
+ const WebRtc_UWord8 mask46_37[222] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
+ };
+
+ const WebRtc_UWord8 mask46_38[228] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
+ 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
+ 0x72, 0x2b, 0xa5, 0xd4, 0xb9, 0x30
+ };
+
+ const WebRtc_UWord8 mask46_39[234] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
+ };
+
+ const WebRtc_UWord8 mask46_4[24] =
+ {
+ 0xc4, 0xd1, 0x65, 0x89, 0xa2, 0xc8,
+ 0x31, 0x62, 0x96, 0x62, 0xc5, 0x2c,
+ 0x4b, 0x24, 0x5a, 0x96, 0x48, 0xb4,
+ 0x2c, 0xa8, 0xaa, 0x59, 0x51, 0x54
+ };
+
+ const WebRtc_UWord8 mask46_40[240] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x7c, 0xc8, 0x93, 0x63, 0x3c, 0x80
+ };
+
+ const WebRtc_UWord8 mask46_41[246] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
+ };
+
+ const WebRtc_UWord8 mask46_42[252] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
+ 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
+ 0xfc, 0x6e, 0x89, 0x54, 0x4f, 0x00
+ };
+
+ const WebRtc_UWord8 mask46_43[258] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
+ };
+
+ const WebRtc_UWord8 mask46_44[264] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x1a, 0x8a, 0x00, 0x1c, 0x89, 0x54
+ };
+
+ const WebRtc_UWord8 mask46_45[270] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
+ 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
+ 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
+ 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
+ 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
+ 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
+ 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
+ 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
+ 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
+ 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
+ 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
+ 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
+ };
+
+ const WebRtc_UWord8 mask46_46[276] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
+ 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
+ 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
+ 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
+ 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
+ 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
+ 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
+ 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
+ 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
+ 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
+ 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
+ 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10,
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
+ 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
+ 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
+ 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
+ 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
+ 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
+ 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
+ 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
+ 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
+ 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
+ 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
+ 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
+ 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
+ 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
+ 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
+ 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
+ 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
+ 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
+ 0xd5, 0xdf, 0x59, 0xb9, 0xba, 0x10
+ };
+
+ const WebRtc_UWord8 mask46_5[30] =
+ {
+ 0xc6, 0xca, 0x6d, 0x8d, 0x94, 0xd8,
+ 0x63, 0x6c, 0x96, 0xc6, 0xd9, 0x2c,
+ 0x1d, 0xa1, 0xdc, 0x3b, 0x43, 0xb8,
+ 0xad, 0x55, 0x39, 0x5a, 0xaa, 0x70,
+ 0xb2, 0xb7, 0x07, 0x65, 0x6e, 0x0c
+ };
+
+ const WebRtc_UWord8 mask46_6[36] =
+ {
+ 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
+ 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
+ 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
+ 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
+ 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
+ 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80
+ };
+
+ const WebRtc_UWord8 mask46_7[42] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
+ 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
+ 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
+ 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
+ 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
+ 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
+ };
+
+ const WebRtc_UWord8 mask46_8[48] =
+ {
+ 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
+ 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
+ 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
+ 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
+ 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
+ 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
+ 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
+ 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
+ };
+
+ const WebRtc_UWord8 mask46_9[54] =
+ {
+ 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
+ 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
+ 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
+ 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
+ 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
+ 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
+ 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
+ 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
+ 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
+ };
+
+ const WebRtc_UWord8 mask47_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xfe
+ };
+
+ const WebRtc_UWord8 mask47_10[60] =
+ {
+ 0x64, 0x4a, 0x28, 0x22, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x68, 0xa6,
+ 0xa9, 0x10, 0x1a, 0x00, 0x90, 0x0a,
+ 0x04, 0xc4, 0x84, 0x21, 0x06, 0x12,
+ 0xd0, 0x01, 0x44, 0x94, 0x29, 0x42,
+ 0x82, 0x40, 0x1c, 0x81, 0x48, 0x14,
+ 0x21, 0x37, 0x01, 0x40, 0xd4, 0x04,
+ 0x0c, 0x21, 0x23, 0x11, 0x01, 0x18,
+ 0x4a, 0x0a, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x12, 0xb4, 0x50, 0xa8, 0x1a, 0x80
+ };
+
+ const WebRtc_UWord8 mask47_11[66] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
+ 0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
+ 0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
+ 0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
+ 0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
+ 0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
+ 0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
+ 0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
+ 0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
+ 0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
+ 0x30, 0x58, 0x05, 0x18, 0x58, 0x04
+ };
+
+ const WebRtc_UWord8 mask47_12[72] =
+ {
+ 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
+ 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
+ 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
+ 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
+ 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
+ 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
+ 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
+ 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
+ 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
+ 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
+ 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
+ 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88
+ };
+
+ const WebRtc_UWord8 mask47_13[78] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
+ };
+
+ const WebRtc_UWord8 mask47_14[84] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
+ };
+
+ const WebRtc_UWord8 mask47_15[90] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
+ };
+
+ const WebRtc_UWord8 mask47_16[96] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
+ };
+
+ const WebRtc_UWord8 mask47_17[102] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
+ };
+
+ const WebRtc_UWord8 mask47_18[108] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
+ };
+
+ const WebRtc_UWord8 mask47_19[114] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
+ };
+
+ const WebRtc_UWord8 mask47_2[12] =
+ {
+ 0xec, 0xdd, 0x99, 0xd9, 0x9d, 0x98,
+ 0x9b, 0xb2, 0x77, 0x27, 0x72, 0x76
+ };
+
+ const WebRtc_UWord8 mask47_20[120] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
+ };
+
+ const WebRtc_UWord8 mask47_21[126] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04
+ };
+
+ const WebRtc_UWord8 mask47_22[132] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
+ };
+
+ const WebRtc_UWord8 mask47_23[138] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
+ 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
+ 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
+ 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
+ 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
+ 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
+ 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
+ 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
+ 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
+ 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
+ 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
+ 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
+ };
+
+ const WebRtc_UWord8 mask47_24[144] =
+ {
+ 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
+ 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
+ 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
+ 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
+ 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
+ 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
+ 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
+ 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
+ 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
+ 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
+ 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
+ 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
+ 0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
+ 0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
+ 0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
+ 0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
+ 0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
+ 0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
+ 0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
+ 0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
+ 0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
+ 0x30, 0x58, 0x05, 0x18, 0x58, 0x04,
+ 0x27, 0x41, 0x35, 0x57, 0x7e, 0x6a
+ };
+
+ const WebRtc_UWord8 mask47_25[150] =
+ {
+ 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
+ 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
+ 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
+ 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
+ 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
+ 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
+ 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
+ 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
+ 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
+ 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
+ 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
+ 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
+ };
+
+ const WebRtc_UWord8 mask47_26[156] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
+ 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
+ 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
+ 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
+ 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
+ 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
+ 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
+ 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
+ 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
+ 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
+ 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
+ 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
+ 0x6c, 0xea, 0xc4, 0x42, 0x20, 0x9e
+ };
+
+ const WebRtc_UWord8 mask47_27[162] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
+ };
+
+ const WebRtc_UWord8 mask47_28[168] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x86, 0x1e, 0xa6, 0xaf, 0x3d, 0x04
+ };
+
+ const WebRtc_UWord8 mask47_29[174] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
+ };
+
+ const WebRtc_UWord8 mask47_3[18] =
+ {
+ 0xca, 0xd3, 0x65, 0x36, 0x53, 0x64,
+ 0xf1, 0x49, 0x3a, 0x93, 0xa9, 0x3a,
+ 0x76, 0x27, 0xd0, 0x7d, 0x07, 0xd0
+ };
+
+ const WebRtc_UWord8 mask47_30[180] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
+ 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
+ 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
+ 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
+ 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
+ 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
+ 0x97, 0x43, 0x63, 0xc6, 0x09, 0x9c
+ };
+
+ const WebRtc_UWord8 mask47_31[186] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
+ };
+
+ const WebRtc_UWord8 mask47_32[192] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0xe5, 0x50, 0x45, 0x63, 0xc2, 0xf4
+ };
+
+ const WebRtc_UWord8 mask47_33[198] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
+ };
+
+ const WebRtc_UWord8 mask47_34[204] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
+ 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
+ 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
+ 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
+ 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
+ 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
+ 0xef, 0xbb, 0xa6, 0x23, 0x5c, 0xbe
+ };
+
+ const WebRtc_UWord8 mask47_35[210] =
+ {
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
+ };
+
+ const WebRtc_UWord8 mask47_36[216] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0e, 0xd7, 0x38, 0x20, 0x87, 0x66
+ };
+
+ const WebRtc_UWord8 mask47_37[222] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
+ };
+
+ const WebRtc_UWord8 mask47_38[228] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
+ 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
+ 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
+ 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
+ 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
+ 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
+ 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
+ 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
+ 0x7b, 0x47, 0xa5, 0xde, 0x9a, 0xd4
+ };
+
+ const WebRtc_UWord8 mask47_39[234] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
+ };
+
+ const WebRtc_UWord8 mask47_4[24] =
+ {
+ 0xc4, 0xd1, 0x65, 0x16, 0x51, 0x64,
+ 0x31, 0x62, 0x96, 0x29, 0x62, 0x96,
+ 0x4b, 0x24, 0x5a, 0x45, 0xa4, 0x5a,
+ 0x2c, 0xa8, 0xaa, 0x8a, 0xa8, 0xaa
+ };
+
+ const WebRtc_UWord8 mask47_40[240] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xc4, 0xae, 0x5e, 0x33, 0xf5, 0x1a
+ };
+
+ const WebRtc_UWord8 mask47_41[246] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04
+ };
+
+ const WebRtc_UWord8 mask47_42[252] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
+ 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
+ 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
+ 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
+ 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
+ 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
+ 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
+ 0x3c, 0xb0, 0x36, 0x3b, 0x14, 0xa2
+ };
+
+ const WebRtc_UWord8 mask47_43[258] =
+ {
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
+ };
+
+ const WebRtc_UWord8 mask47_44[264] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0xd4, 0x8a, 0xd4, 0xd3, 0x3f, 0xe6
+ };
+
+ const WebRtc_UWord8 mask47_45[270] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
+ 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
+ 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
+ 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
+ 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
+ 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
+ 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
+ 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
+ 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
+ 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
+ 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
+ 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
+ };
+
+ const WebRtc_UWord8 mask47_46[276] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
+ 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
+ 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
+ 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
+ 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
+ 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
+ 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
+ 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
+ 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
+ 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
+ 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
+ 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
+ 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
+ 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
+ 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
+ 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
+ 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
+ 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
+ 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
+ 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
+ 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
+ 0x37, 0x9d, 0xcf, 0xe0, 0xe4, 0x20
+ };
+
+ const WebRtc_UWord8 mask47_47[282] =
+ {
+ 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
+ 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
+ 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
+ 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
+ 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
+ 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
+ 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
+ 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
+ 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
+ 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
+ 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
+ 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
+ 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
+ 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
+ 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
+ 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
+ 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
+ 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
+ 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
+ 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
+ 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
+ 0x46, 0x4a, 0x6c, 0x20, 0xc2, 0x0c,
+ 0x33, 0x24, 0x26, 0x04, 0x60, 0x46,
+ 0x99, 0x02, 0x12, 0x80, 0xa8, 0x0a,
+ 0x05, 0x80, 0x0e, 0x43, 0x04, 0x30,
+ 0x80, 0xa1, 0x83, 0x02, 0x30, 0x22,
+ 0x84, 0x48, 0x18, 0x29, 0x02, 0x90,
+ 0x40, 0x6d, 0x41, 0x30, 0x13, 0x00,
+ 0x0a, 0x90, 0xc0, 0x11, 0x21, 0x12,
+ 0x68, 0x04, 0x90, 0xc4, 0x0c, 0x40,
+ 0x10, 0x31, 0x20, 0x48, 0x44, 0x84,
+ 0x30, 0x58, 0x05, 0x14, 0x11, 0x40,
+ 0x46, 0x42, 0x0d, 0x08, 0x90, 0x88,
+ 0x33, 0x20, 0x46, 0xa6, 0xca, 0x6c,
+ 0x99, 0x08, 0x0a, 0x42, 0x64, 0x26,
+ 0x05, 0x84, 0x30, 0x21, 0x22, 0x12,
+ 0x80, 0xb0, 0x22, 0x00, 0xe0, 0x0e,
+ 0x84, 0x42, 0x90, 0x18, 0x21, 0x82,
+ 0x40, 0x73, 0x00, 0x81, 0x88, 0x18,
+ 0x0a, 0x81, 0x12, 0xd4, 0x0d, 0x40,
+ 0x68, 0x0c, 0x41, 0x0c, 0x10, 0xc0,
+ 0x10, 0x24, 0x84, 0x49, 0x04, 0x90,
+ 0x30, 0x51, 0x41, 0x12, 0x11, 0x20,
+ 0x5f, 0x50, 0x89, 0x80, 0x58, 0x04,
+ 0x1f, 0x2f, 0x63, 0x10, 0x64, 0xb2
+ };
+
+ const WebRtc_UWord8 mask47_5[30] =
+ {
+ 0xc6, 0xca, 0x6c, 0xa6, 0xca, 0x6c,
+ 0x63, 0x6c, 0x96, 0xc9, 0x6c, 0x96,
+ 0x1d, 0xa1, 0xdc, 0x1d, 0xc1, 0xdc,
+ 0xad, 0x55, 0x39, 0x53, 0x95, 0x38,
+ 0xb2, 0xb7, 0x07, 0x70, 0x77, 0x06
+ };
+
+ const WebRtc_UWord8 mask47_6[36] =
+ {
+ 0x64, 0x4a, 0x29, 0xa2, 0x9a, 0x28,
+ 0x51, 0x58, 0xa2, 0x8a, 0x68, 0xa6,
+ 0x0c, 0xa4, 0x30, 0x45, 0xa4, 0x5a,
+ 0xa1, 0x22, 0x46, 0x2d, 0x82, 0xd8,
+ 0x12, 0xa1, 0x1c, 0x17, 0x41, 0x74,
+ 0x8a, 0x45, 0xc1, 0xd1, 0x1d, 0x10
+ };
+
+ const WebRtc_UWord8 mask47_7[42] =
+ {
+ 0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
+ 0x33, 0x24, 0x26, 0x4a, 0x64, 0xa6,
+ 0x91, 0x92, 0x12, 0x61, 0xa6, 0x0a,
+ 0xa4, 0x20, 0x4a, 0x0c, 0x90, 0xd8,
+ 0x50, 0xa0, 0xd5, 0x81, 0x70, 0x36,
+ 0x84, 0xc5, 0x80, 0x55, 0x45, 0x54,
+ 0x09, 0x71, 0x0d, 0x50, 0x9d, 0x08
+ };
+
+ const WebRtc_UWord8 mask47_8[48] =
+ {
+ 0x0c, 0x84, 0x0d, 0x02, 0xc0, 0x2c,
+ 0x80, 0x70, 0x06, 0x80, 0x78, 0x06,
+ 0xa0, 0x88, 0x48, 0x21, 0x22, 0x12,
+ 0x05, 0x40, 0x32, 0x0c, 0xa0, 0xca,
+ 0x43, 0x02, 0x82, 0x40, 0x95, 0x08,
+ 0x1a, 0x01, 0x51, 0x15, 0x41, 0x54,
+ 0x60, 0x27, 0x00, 0x66, 0x06, 0x60,
+ 0x14, 0x38, 0xa0, 0x99, 0x09, 0x90
+ };
+
+ const WebRtc_UWord8 mask47_9[54] =
+ {
+ 0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
+ 0x62, 0x7c, 0x84, 0xc8, 0x4c, 0x84,
+ 0x8c, 0x04, 0x88, 0x30, 0x83, 0x88,
+ 0x01, 0x74, 0x23, 0x40, 0x94, 0x08,
+ 0x07, 0x83, 0x07, 0x02, 0x70, 0x26,
+ 0xa0, 0x80, 0x72, 0x45, 0x44, 0x54,
+ 0x18, 0xb1, 0x42, 0x10, 0xe1, 0x0e,
+ 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
+ 0x78, 0x00, 0x1c, 0x03, 0x80, 0x38
+ };
+
+ const WebRtc_UWord8 mask48_1[6] =
+ {
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
+ };
+
+ const WebRtc_UWord8 mask48_10[60] =
+ {
+ 0x11, 0x45, 0x14, 0x11, 0x45, 0x14,
+ 0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
+ 0x00, 0x48, 0x05, 0x00, 0x48, 0x05,
+ 0x10, 0x83, 0x09, 0x10, 0x83, 0x09,
+ 0x4a, 0x14, 0xa1, 0x4a, 0x14, 0xa1,
+ 0x40, 0xa4, 0x0a, 0x40, 0xa4, 0x0a,
+ 0xa0, 0x6a, 0x02, 0xa0, 0x6a, 0x02,
+ 0x88, 0x80, 0x8c, 0x88, 0x80, 0x8c,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x54, 0x0d, 0x40, 0x54, 0x0d, 0x40
+ };
+
+ const WebRtc_UWord8 mask48_11[66] =
+ {
+ 0x53, 0x65, 0x34, 0x53, 0x65, 0x34,
+ 0xa0, 0x32, 0x11, 0xa0, 0x32, 0x11,
+ 0x15, 0x11, 0x41, 0x15, 0x11, 0x41,
+ 0x03, 0x50, 0x15, 0x03, 0x50, 0x15,
+ 0x8c, 0x88, 0xc8, 0x8c, 0x88, 0xc8,
+ 0x28, 0x82, 0x88, 0x28, 0x82, 0x88,
+ 0x08, 0x48, 0x84, 0x08, 0x48, 0x84,
+ 0x99, 0x01, 0x90, 0x99, 0x01, 0x90,
+ 0x22, 0x92, 0x29, 0x22, 0x92, 0x29,
+ 0x46, 0x04, 0x60, 0x46, 0x04, 0x60,
+ 0x8c, 0x2c, 0x02, 0x8c, 0x2c, 0x02
+ };
+
+ const WebRtc_UWord8 mask48_12[72] =
+ {
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
+ };
+
+ const WebRtc_UWord8 mask48_13[78] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86
+ };
+
+ const WebRtc_UWord8 mask48_14[84] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
+ };
+
+ const WebRtc_UWord8 mask48_15[90] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
+ };
+
+ const WebRtc_UWord8 mask48_16[96] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
+ };
+
+ const WebRtc_UWord8 mask48_17[102] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
+ };
+
+ const WebRtc_UWord8 mask48_18[108] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x34, 0x50, 0xae, 0x34, 0x50, 0xae
+ };
+
+ const WebRtc_UWord8 mask48_19[114] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
+ };
+
+ const WebRtc_UWord8 mask48_2[12] =
+ {
+ 0xec, 0xce, 0xcc, 0xec, 0xce, 0xcc,
+ 0x93, 0xb9, 0x3b, 0x93, 0xb9, 0x3b
+ };
+
+ const WebRtc_UWord8 mask48_20[120] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
+ };
+
+ const WebRtc_UWord8 mask48_21[126] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
+ };
+
+ const WebRtc_UWord8 mask48_22[132] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
+ };
+
+ const WebRtc_UWord8 mask48_23[138] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
+ };
+
+ const WebRtc_UWord8 mask48_24[144] =
+ {
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x88, 0x32, 0x59, 0x88, 0x32, 0x59
+ };
+
+ const WebRtc_UWord8 mask48_25[150] =
+ {
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86
+ };
+
+ const WebRtc_UWord8 mask48_26[156] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x3e, 0x20, 0x79, 0xe5, 0x55, 0x70
+ };
+
+ const WebRtc_UWord8 mask48_27[162] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
+ };
+
+ const WebRtc_UWord8 mask48_28[168] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x32, 0xe3, 0xc0, 0x4a, 0xf2, 0x2a
+ };
+
+ const WebRtc_UWord8 mask48_29[174] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
+ };
+
+ const WebRtc_UWord8 mask48_3[18] =
+ {
+ 0x9b, 0x29, 0xb2, 0x9b, 0x29, 0xb2,
+ 0x49, 0xd4, 0x9d, 0x49, 0xd4, 0x9d,
+ 0x3e, 0x83, 0xe8, 0x3e, 0x83, 0xe8
+ };
+
+ const WebRtc_UWord8 mask48_30[180] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
+ 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
+ 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
+ 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
+ 0x66, 0xf3, 0x9a, 0xdd, 0x68, 0x93
+ };
+
+ const WebRtc_UWord8 mask48_31[186] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
+ };
+
+ const WebRtc_UWord8 mask48_32[192] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0xd5, 0x4a, 0x4f, 0x48, 0xb5, 0x31
+ };
+
+ const WebRtc_UWord8 mask48_33[198] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
+ };
+
+ const WebRtc_UWord8 mask48_34[204] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
+ 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
+ 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
+ 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
+ 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
+ 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
+ 0x40, 0x72, 0x4c, 0xe8, 0xf2, 0x42
+ };
+
+ const WebRtc_UWord8 mask48_35[210] =
+ {
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x34, 0x50, 0xae, 0x34, 0x50, 0xae
+ };
+
+ const WebRtc_UWord8 mask48_36[216] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x71, 0xba, 0x8b, 0xf3, 0xfa, 0x9d
+ };
+
+ const WebRtc_UWord8 mask48_37[222] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
+ };
+
+ const WebRtc_UWord8 mask48_38[228] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
+ 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
+ 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
+ 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
+ 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
+ 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
+ 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
+ 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
+ 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
+ 0x2a, 0x7a, 0xf6, 0x8c, 0xde, 0x51
+ };
+
+ const WebRtc_UWord8 mask48_39[234] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
+ };
+
+ const WebRtc_UWord8 mask48_4[24] =
+ {
+ 0x8b, 0x28, 0xb2, 0x8b, 0x28, 0xb2,
+ 0x14, 0xb1, 0x4b, 0x14, 0xb1, 0x4b,
+ 0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
+ 0x45, 0x54, 0x55, 0x45, 0x54, 0x55
+ };
+
+ const WebRtc_UWord8 mask48_40[240] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x20, 0x5f, 0x68, 0xd5, 0xa2, 0x1b
+ };
+
+ const WebRtc_UWord8 mask48_41[246] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
+ };
+
+ const WebRtc_UWord8 mask48_42[252] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
+ 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
+ 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
+ 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
+ 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
+ 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
+ 0x66, 0xcf, 0xa3, 0x47, 0x69, 0x00
+ };
+
+ const WebRtc_UWord8 mask48_43[258] =
+ {
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
+ };
+
+ const WebRtc_UWord8 mask48_44[264] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0xc6, 0x40, 0x1f, 0x57, 0xc6, 0xe6
+ };
+
+ const WebRtc_UWord8 mask48_45[270] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
+ };
+
+ const WebRtc_UWord8 mask48_46[276] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
+ 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
+ 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
+ 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
+ 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
+ 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
+ 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
+ 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
+ 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
+ 0x10, 0xf9, 0xab, 0x12, 0x14, 0xef
+ };
+
+ const WebRtc_UWord8 mask48_47[282] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x88, 0x32, 0x59, 0x88, 0x32, 0x59
+ };
+
+ const WebRtc_UWord8 mask48_48[288] =
+ {
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x88, 0x32, 0x59, 0x88, 0x32, 0x59,
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
+ 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
+ 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
+ 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
+ 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
+ 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
+ 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
+ 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
+ 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
+ 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
+ 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
+ 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
+ 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
+ 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
+ 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
+ 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
+ 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
+ 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
+ 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
+ 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
+ 0xff, 0x9b, 0xdf, 0xec, 0xae, 0x0e
+ };
+
+ const WebRtc_UWord8 mask48_5[30] =
+ {
+ 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
+ 0x64, 0xb6, 0x4b, 0x64, 0xb6, 0x4b,
+ 0x0e, 0xe0, 0xee, 0x0e, 0xe0, 0xee,
+ 0xa9, 0xca, 0x9c, 0xa9, 0xca, 0x9c,
+ 0xb8, 0x3b, 0x83, 0xb8, 0x3b, 0x83
+ };
+
+ const WebRtc_UWord8 mask48_6[36] =
+ {
+ 0xd1, 0x4d, 0x14, 0xd1, 0x4d, 0x14,
+ 0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
+ 0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
+ 0x16, 0xc1, 0x6c, 0x16, 0xc1, 0x6c,
+ 0x0b, 0xa0, 0xba, 0x0b, 0xa0, 0xba,
+ 0xe8, 0x8e, 0x88, 0xe8, 0x8e, 0x88
+ };
+
+ const WebRtc_UWord8 mask48_7[42] =
+ {
+ 0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
+ 0x25, 0x32, 0x53, 0x25, 0x32, 0x53,
+ 0x30, 0xd3, 0x05, 0x30, 0xd3, 0x05,
+ 0x06, 0x48, 0x6c, 0x06, 0x48, 0x6c,
+ 0xc0, 0xb8, 0x1b, 0xc0, 0xb8, 0x1b,
+ 0x2a, 0xa2, 0xaa, 0x2a, 0xa2, 0xaa,
+ 0xa8, 0x4e, 0x84, 0xa8, 0x4e, 0x84
+ };
+
+ const WebRtc_UWord8 mask48_8[48] =
+ {
+ 0x81, 0x60, 0x16, 0x81, 0x60, 0x16,
+ 0x40, 0x3c, 0x03, 0x40, 0x3c, 0x03,
+ 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
+ 0x06, 0x50, 0x65, 0x06, 0x50, 0x65,
+ 0x20, 0x4a, 0x84, 0x20, 0x4a, 0x84,
+ 0x8a, 0xa0, 0xaa, 0x8a, 0xa0, 0xaa,
+ 0x33, 0x03, 0x30, 0x33, 0x03, 0x30,
+ 0x4c, 0x84, 0xc8, 0x4c, 0x84, 0xc8
+ };
+
+ const WebRtc_UWord8 mask48_9[54] =
+ {
+ 0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
+ 0x64, 0x26, 0x42, 0x64, 0x26, 0x42,
+ 0x18, 0x41, 0xc4, 0x18, 0x41, 0xc4,
+ 0xa0, 0x4a, 0x04, 0xa0, 0x4a, 0x04,
+ 0x81, 0x38, 0x13, 0x81, 0x38, 0x13,
+ 0x22, 0xa2, 0x2a, 0x22, 0xa2, 0x2a,
+ 0x08, 0x70, 0x87, 0x08, 0x70, 0x87,
+ 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
+ 0x01, 0xc0, 0x1c, 0x01, 0xc0, 0x1c
+ };
+
+ const WebRtc_UWord8 mask4_1[2] =
+ {
+ 0xf0, 0x00
+ };
+
+ const WebRtc_UWord8 mask4_2[4] =
+ {
+ 0xc0, 0x00,
+ 0xb0, 0x00
+ };
+
+ const WebRtc_UWord8 mask4_3[6] =
+ {
+ 0xc0, 0x00,
+ 0xb0, 0x00,
+ 0x60, 0x00
+ };
+
+ const WebRtc_UWord8 mask4_4[8] =
+ {
+ 0xc0, 0x00,
+ 0xa0, 0x00,
+ 0x30, 0x00,
+ 0x50, 0x00
+ };
+
+ const WebRtc_UWord8 mask5_1[2] =
+ {
+ 0xf8, 0x00
+ };
+
+ const WebRtc_UWord8 mask5_2[4] =
+ {
+ 0xa8, 0x00,
+ 0xd0, 0x00
+ };
+
+ const WebRtc_UWord8 mask5_3[6] =
+ {
+ 0xb0, 0x00,
+ 0xc8, 0x00,
+ 0x50, 0x00
+ };
+
+ const WebRtc_UWord8 mask5_4[8] =
+ {
+ 0xc8, 0x00,
+ 0xb0, 0x00,
+ 0x50, 0x00,
+ 0x28, 0x00
+ };
+
+ const WebRtc_UWord8 mask5_5[10] =
+ {
+ 0xc0, 0x00,
+ 0x30, 0x00,
+ 0x18, 0x00,
+ 0xa0, 0x00,
+ 0x48, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_1[2] =
+ {
+ 0xfc, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_2[4] =
+ {
+ 0xa8, 0x00,
+ 0xd4, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_3[6] =
+ {
+ 0xd0, 0x00,
+ 0x68, 0x00,
+ 0xa4, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_4[8] =
+ {
+ 0xa8, 0x00,
+ 0x58, 0x00,
+ 0x64, 0x00,
+ 0x94, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_5[10] =
+ {
+ 0xa8, 0x00,
+ 0x84, 0x00,
+ 0x64, 0x00,
+ 0x90, 0x00,
+ 0x58, 0x00
+ };
+
+ const WebRtc_UWord8 mask6_6[12] =
+ {
+ 0x98, 0x00,
+ 0x64, 0x00,
+ 0x50, 0x00,
+ 0x14, 0x00,
+ 0xa8, 0x00,
+ 0xe0, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_1[2] =
+ {
+ 0xfe, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_2[4] =
+ {
+ 0xd4, 0x00,
+ 0xaa, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_3[6] =
+ {
+ 0xd0, 0x00,
+ 0xaa, 0x00,
+ 0x64, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_4[8] =
+ {
+ 0xd0, 0x00,
+ 0xaa, 0x00,
+ 0x64, 0x00,
+ 0x1c, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_5[10] =
+ {
+ 0x0c, 0x00,
+ 0xb0, 0x00,
+ 0x1a, 0x00,
+ 0xc4, 0x00,
+ 0x62, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_6[12] =
+ {
+ 0x8c, 0x00,
+ 0x4a, 0x00,
+ 0x64, 0x00,
+ 0xd0, 0x00,
+ 0xa0, 0x00,
+ 0x32, 0x00
+ };
+
+ const WebRtc_UWord8 mask7_7[14] =
+ {
+ 0x4a, 0x00,
+ 0x94, 0x00,
+ 0x1a, 0x00,
+ 0xc4, 0x00,
+ 0x28, 0x00,
+ 0xc2, 0x00,
+ 0x34, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_1[2] =
+ {
+ 0xff, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_2[4] =
+ {
+ 0xaa, 0x00,
+ 0xd5, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_3[6] =
+ {
+ 0xc5, 0x00,
+ 0x92, 0x00,
+ 0x6a, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_4[8] =
+ {
+ 0x45, 0x00,
+ 0xb4, 0x00,
+ 0x6a, 0x00,
+ 0x89, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_5[10] =
+ {
+ 0x8c, 0x00,
+ 0x92, 0x00,
+ 0x2b, 0x00,
+ 0x51, 0x00,
+ 0x64, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_6[12] =
+ {
+ 0xa1, 0x00,
+ 0x52, 0x00,
+ 0x91, 0x00,
+ 0x2a, 0x00,
+ 0xc4, 0x00,
+ 0x4c, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_7[14] =
+ {
+ 0x15, 0x00,
+ 0xc2, 0x00,
+ 0x25, 0x00,
+ 0x62, 0x00,
+ 0x58, 0x00,
+ 0x8c, 0x00,
+ 0xa3, 0x00
+ };
+
+ const WebRtc_UWord8 mask8_8[16] =
+ {
+ 0x25, 0x00,
+ 0x8a, 0x00,
+ 0x91, 0x00,
+ 0x68, 0x00,
+ 0x32, 0x00,
+ 0x43, 0x00,
+ 0xc4, 0x00,
+ 0x1c, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_1[2] =
+ {
+ 0xff, 0x80
+ };
+
+ const WebRtc_UWord8 mask9_2[4] =
+ {
+ 0xaa, 0x80,
+ 0xd5, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_3[6] =
+ {
+ 0xa5, 0x00,
+ 0xc8, 0x00,
+ 0x52, 0x80
+ };
+
+ const WebRtc_UWord8 mask9_4[8] =
+ {
+ 0xa2, 0x00,
+ 0xc9, 0x00,
+ 0x52, 0x80,
+ 0x24, 0x80
+ };
+
+ const WebRtc_UWord8 mask9_5[10] =
+ {
+ 0x8c, 0x00,
+ 0x25, 0x00,
+ 0x92, 0x80,
+ 0x41, 0x80,
+ 0x58, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_6[12] =
+ {
+ 0x84, 0x80,
+ 0x27, 0x00,
+ 0x51, 0x80,
+ 0x1a, 0x00,
+ 0x68, 0x00,
+ 0x89, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_7[14] =
+ {
+ 0x8c, 0x00,
+ 0x47, 0x00,
+ 0x81, 0x80,
+ 0x12, 0x80,
+ 0x58, 0x00,
+ 0x28, 0x80,
+ 0xb4, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_8[16] =
+ {
+ 0x2c, 0x00,
+ 0x91, 0x00,
+ 0x40, 0x80,
+ 0x06, 0x80,
+ 0xc8, 0x00,
+ 0x45, 0x00,
+ 0x30, 0x80,
+ 0xa2, 0x00
+ };
+
+ const WebRtc_UWord8 mask9_9[18] =
+ {
+ 0x4c, 0x00,
+ 0x62, 0x00,
+ 0x91, 0x00,
+ 0x42, 0x80,
+ 0xa4, 0x00,
+ 0x13, 0x00,
+ 0x30, 0x80,
+ 0x88, 0x80,
+ 0x09, 0x00
+ };
+
+ const WebRtc_UWord8* packetMask1[1] =
+ {
+ mask1_1
+ };
+
+ const WebRtc_UWord8* packetMask2[2] =
+ {
+ mask2_1,
+ mask2_2
+ };
+
+ const WebRtc_UWord8* packetMask3[3] =
+ {
+ mask3_1,
+ mask3_2,
+ mask3_3
+ };
+
+ const WebRtc_UWord8* packetMask4[4] =
+ {
+ mask4_1,
+ mask4_2,
+ mask4_3,
+ mask4_4
+ };
+
+ const WebRtc_UWord8* packetMask5[5] =
+ {
+ mask5_1,
+ mask5_2,
+ mask5_3,
+ mask5_4,
+ mask5_5
+ };
+
+ const WebRtc_UWord8* packetMask6[6] =
+ {
+ mask6_1,
+ mask6_2,
+ mask6_3,
+ mask6_4,
+ mask6_5,
+ mask6_6
+ };
+
+ const WebRtc_UWord8* packetMask7[7] =
+ {
+ mask7_1,
+ mask7_2,
+ mask7_3,
+ mask7_4,
+ mask7_5,
+ mask7_6,
+ mask7_7
+ };
+
+ const WebRtc_UWord8* packetMask8[8] =
+ {
+ mask8_1,
+ mask8_2,
+ mask8_3,
+ mask8_4,
+ mask8_5,
+ mask8_6,
+ mask8_7,
+ mask8_8
+ };
+
+ const WebRtc_UWord8* packetMask9[9] =
+ {
+ mask9_1,
+ mask9_2,
+ mask9_3,
+ mask9_4,
+ mask9_5,
+ mask9_6,
+ mask9_7,
+ mask9_8,
+ mask9_9
+ };
+
+ const WebRtc_UWord8* packetMask10[10] =
+ {
+ mask10_1,
+ mask10_2,
+ mask10_3,
+ mask10_4,
+ mask10_5,
+ mask10_6,
+ mask10_7,
+ mask10_8,
+ mask10_9,
+ mask10_10
+ };
+
+ const WebRtc_UWord8* packetMask11[11] =
+ {
+ mask11_1,
+ mask11_2,
+ mask11_3,
+ mask11_4,
+ mask11_5,
+ mask11_6,
+ mask11_7,
+ mask11_8,
+ mask11_9,
+ mask11_10,
+ mask11_11
+ };
+
+ const WebRtc_UWord8* packetMask12[12] =
+ {
+ mask12_1,
+ mask12_2,
+ mask12_3,
+ mask12_4,
+ mask12_5,
+ mask12_6,
+ mask12_7,
+ mask12_8,
+ mask12_9,
+ mask12_10,
+ mask12_11,
+ mask12_12
+ };
+
+ const WebRtc_UWord8* packetMask13[13] =
+ {
+ mask13_1,
+ mask13_2,
+ mask13_3,
+ mask13_4,
+ mask13_5,
+ mask13_6,
+ mask13_7,
+ mask13_8,
+ mask13_9,
+ mask13_10,
+ mask13_11,
+ mask13_12,
+ mask13_13
+ };
+
+ const WebRtc_UWord8* packetMask14[14] =
+ {
+ mask14_1,
+ mask14_2,
+ mask14_3,
+ mask14_4,
+ mask14_5,
+ mask14_6,
+ mask14_7,
+ mask14_8,
+ mask14_9,
+ mask14_10,
+ mask14_11,
+ mask14_12,
+ mask14_13,
+ mask14_14
+ };
+
+ const WebRtc_UWord8* packetMask15[15] =
+ {
+ mask15_1,
+ mask15_2,
+ mask15_3,
+ mask15_4,
+ mask15_5,
+ mask15_6,
+ mask15_7,
+ mask15_8,
+ mask15_9,
+ mask15_10,
+ mask15_11,
+ mask15_12,
+ mask15_13,
+ mask15_14,
+ mask15_15
+ };
+
+ const WebRtc_UWord8* packetMask16[16] =
+ {
+ mask16_1,
+ mask16_2,
+ mask16_3,
+ mask16_4,
+ mask16_5,
+ mask16_6,
+ mask16_7,
+ mask16_8,
+ mask16_9,
+ mask16_10,
+ mask16_11,
+ mask16_12,
+ mask16_13,
+ mask16_14,
+ mask16_15,
+ mask16_16
+ };
+
+ const WebRtc_UWord8* packetMask17[17] =
+ {
+ mask17_1,
+ mask17_2,
+ mask17_3,
+ mask17_4,
+ mask17_5,
+ mask17_6,
+ mask17_7,
+ mask17_8,
+ mask17_9,
+ mask17_10,
+ mask17_11,
+ mask17_12,
+ mask17_13,
+ mask17_14,
+ mask17_15,
+ mask17_16,
+ mask17_17
+ };
+
+ const WebRtc_UWord8* packetMask18[18] =
+ {
+ mask18_1,
+ mask18_2,
+ mask18_3,
+ mask18_4,
+ mask18_5,
+ mask18_6,
+ mask18_7,
+ mask18_8,
+ mask18_9,
+ mask18_10,
+ mask18_11,
+ mask18_12,
+ mask18_13,
+ mask18_14,
+ mask18_15,
+ mask18_16,
+ mask18_17,
+ mask18_18
+ };
+
+ const WebRtc_UWord8* packetMask19[19] =
+ {
+ mask19_1,
+ mask19_2,
+ mask19_3,
+ mask19_4,
+ mask19_5,
+ mask19_6,
+ mask19_7,
+ mask19_8,
+ mask19_9,
+ mask19_10,
+ mask19_11,
+ mask19_12,
+ mask19_13,
+ mask19_14,
+ mask19_15,
+ mask19_16,
+ mask19_17,
+ mask19_18,
+ mask19_19
+ };
+
+ const WebRtc_UWord8* packetMask20[20] =
+ {
+ mask20_1,
+ mask20_2,
+ mask20_3,
+ mask20_4,
+ mask20_5,
+ mask20_6,
+ mask20_7,
+ mask20_8,
+ mask20_9,
+ mask20_10,
+ mask20_11,
+ mask20_12,
+ mask20_13,
+ mask20_14,
+ mask20_15,
+ mask20_16,
+ mask20_17,
+ mask20_18,
+ mask20_19,
+ mask20_20
+ };
+
+ const WebRtc_UWord8* packetMask21[21] =
+ {
+ mask21_1,
+ mask21_2,
+ mask21_3,
+ mask21_4,
+ mask21_5,
+ mask21_6,
+ mask21_7,
+ mask21_8,
+ mask21_9,
+ mask21_10,
+ mask21_11,
+ mask21_12,
+ mask21_13,
+ mask21_14,
+ mask21_15,
+ mask21_16,
+ mask21_17,
+ mask21_18,
+ mask21_19,
+ mask21_20,
+ mask21_21
+ };
+
+ const WebRtc_UWord8* packetMask22[22] =
+ {
+ mask22_1,
+ mask22_2,
+ mask22_3,
+ mask22_4,
+ mask22_5,
+ mask22_6,
+ mask22_7,
+ mask22_8,
+ mask22_9,
+ mask22_10,
+ mask22_11,
+ mask22_12,
+ mask22_13,
+ mask22_14,
+ mask22_15,
+ mask22_16,
+ mask22_17,
+ mask22_18,
+ mask22_19,
+ mask22_20,
+ mask22_21,
+ mask22_22
+ };
+
+ const WebRtc_UWord8* packetMask23[23] =
+ {
+ mask23_1,
+ mask23_2,
+ mask23_3,
+ mask23_4,
+ mask23_5,
+ mask23_6,
+ mask23_7,
+ mask23_8,
+ mask23_9,
+ mask23_10,
+ mask23_11,
+ mask23_12,
+ mask23_13,
+ mask23_14,
+ mask23_15,
+ mask23_16,
+ mask23_17,
+ mask23_18,
+ mask23_19,
+ mask23_20,
+ mask23_21,
+ mask23_22,
+ mask23_23
+ };
+
+ const WebRtc_UWord8* packetMask24[24] =
+ {
+ mask24_1,
+ mask24_2,
+ mask24_3,
+ mask24_4,
+ mask24_5,
+ mask24_6,
+ mask24_7,
+ mask24_8,
+ mask24_9,
+ mask24_10,
+ mask24_11,
+ mask24_12,
+ mask24_13,
+ mask24_14,
+ mask24_15,
+ mask24_16,
+ mask24_17,
+ mask24_18,
+ mask24_19,
+ mask24_20,
+ mask24_21,
+ mask24_22,
+ mask24_23,
+ mask24_24
+ };
+
+ const WebRtc_UWord8* packetMask25[25] =
+ {
+ mask25_1,
+ mask25_2,
+ mask25_3,
+ mask25_4,
+ mask25_5,
+ mask25_6,
+ mask25_7,
+ mask25_8,
+ mask25_9,
+ mask25_10,
+ mask25_11,
+ mask25_12,
+ mask25_13,
+ mask25_14,
+ mask25_15,
+ mask25_16,
+ mask25_17,
+ mask25_18,
+ mask25_19,
+ mask25_20,
+ mask25_21,
+ mask25_22,
+ mask25_23,
+ mask25_24,
+ mask25_25
+ };
+
+ const WebRtc_UWord8* packetMask26[26] =
+ {
+ mask26_1,
+ mask26_2,
+ mask26_3,
+ mask26_4,
+ mask26_5,
+ mask26_6,
+ mask26_7,
+ mask26_8,
+ mask26_9,
+ mask26_10,
+ mask26_11,
+ mask26_12,
+ mask26_13,
+ mask26_14,
+ mask26_15,
+ mask26_16,
+ mask26_17,
+ mask26_18,
+ mask26_19,
+ mask26_20,
+ mask26_21,
+ mask26_22,
+ mask26_23,
+ mask26_24,
+ mask26_25,
+ mask26_26
+ };
+
+ const WebRtc_UWord8* packetMask27[27] =
+ {
+ mask27_1,
+ mask27_2,
+ mask27_3,
+ mask27_4,
+ mask27_5,
+ mask27_6,
+ mask27_7,
+ mask27_8,
+ mask27_9,
+ mask27_10,
+ mask27_11,
+ mask27_12,
+ mask27_13,
+ mask27_14,
+ mask27_15,
+ mask27_16,
+ mask27_17,
+ mask27_18,
+ mask27_19,
+ mask27_20,
+ mask27_21,
+ mask27_22,
+ mask27_23,
+ mask27_24,
+ mask27_25,
+ mask27_26,
+ mask27_27
+ };
+
+ const WebRtc_UWord8* packetMask28[28] =
+ {
+ mask28_1,
+ mask28_2,
+ mask28_3,
+ mask28_4,
+ mask28_5,
+ mask28_6,
+ mask28_7,
+ mask28_8,
+ mask28_9,
+ mask28_10,
+ mask28_11,
+ mask28_12,
+ mask28_13,
+ mask28_14,
+ mask28_15,
+ mask28_16,
+ mask28_17,
+ mask28_18,
+ mask28_19,
+ mask28_20,
+ mask28_21,
+ mask28_22,
+ mask28_23,
+ mask28_24,
+ mask28_25,
+ mask28_26,
+ mask28_27,
+ mask28_28
+ };
+
+ const WebRtc_UWord8* packetMask29[29] =
+ {
+ mask29_1,
+ mask29_2,
+ mask29_3,
+ mask29_4,
+ mask29_5,
+ mask29_6,
+ mask29_7,
+ mask29_8,
+ mask29_9,
+ mask29_10,
+ mask29_11,
+ mask29_12,
+ mask29_13,
+ mask29_14,
+ mask29_15,
+ mask29_16,
+ mask29_17,
+ mask29_18,
+ mask29_19,
+ mask29_20,
+ mask29_21,
+ mask29_22,
+ mask29_23,
+ mask29_24,
+ mask29_25,
+ mask29_26,
+ mask29_27,
+ mask29_28,
+ mask29_29
+ };
+
+ const WebRtc_UWord8* packetMask30[30] =
+ {
+ mask30_1,
+ mask30_2,
+ mask30_3,
+ mask30_4,
+ mask30_5,
+ mask30_6,
+ mask30_7,
+ mask30_8,
+ mask30_9,
+ mask30_10,
+ mask30_11,
+ mask30_12,
+ mask30_13,
+ mask30_14,
+ mask30_15,
+ mask30_16,
+ mask30_17,
+ mask30_18,
+ mask30_19,
+ mask30_20,
+ mask30_21,
+ mask30_22,
+ mask30_23,
+ mask30_24,
+ mask30_25,
+ mask30_26,
+ mask30_27,
+ mask30_28,
+ mask30_29,
+ mask30_30
+ };
+
+ const WebRtc_UWord8* packetMask31[31] =
+ {
+ mask31_1,
+ mask31_2,
+ mask31_3,
+ mask31_4,
+ mask31_5,
+ mask31_6,
+ mask31_7,
+ mask31_8,
+ mask31_9,
+ mask31_10,
+ mask31_11,
+ mask31_12,
+ mask31_13,
+ mask31_14,
+ mask31_15,
+ mask31_16,
+ mask31_17,
+ mask31_18,
+ mask31_19,
+ mask31_20,
+ mask31_21,
+ mask31_22,
+ mask31_23,
+ mask31_24,
+ mask31_25,
+ mask31_26,
+ mask31_27,
+ mask31_28,
+ mask31_29,
+ mask31_30,
+ mask31_31
+ };
+
+ const WebRtc_UWord8* packetMask32[32] =
+ {
+ mask32_1,
+ mask32_2,
+ mask32_3,
+ mask32_4,
+ mask32_5,
+ mask32_6,
+ mask32_7,
+ mask32_8,
+ mask32_9,
+ mask32_10,
+ mask32_11,
+ mask32_12,
+ mask32_13,
+ mask32_14,
+ mask32_15,
+ mask32_16,
+ mask32_17,
+ mask32_18,
+ mask32_19,
+ mask32_20,
+ mask32_21,
+ mask32_22,
+ mask32_23,
+ mask32_24,
+ mask32_25,
+ mask32_26,
+ mask32_27,
+ mask32_28,
+ mask32_29,
+ mask32_30,
+ mask32_31,
+ mask32_32
+ };
+
+ const WebRtc_UWord8* packetMask33[33] =
+ {
+ mask33_1,
+ mask33_2,
+ mask33_3,
+ mask33_4,
+ mask33_5,
+ mask33_6,
+ mask33_7,
+ mask33_8,
+ mask33_9,
+ mask33_10,
+ mask33_11,
+ mask33_12,
+ mask33_13,
+ mask33_14,
+ mask33_15,
+ mask33_16,
+ mask33_17,
+ mask33_18,
+ mask33_19,
+ mask33_20,
+ mask33_21,
+ mask33_22,
+ mask33_23,
+ mask33_24,
+ mask33_25,
+ mask33_26,
+ mask33_27,
+ mask33_28,
+ mask33_29,
+ mask33_30,
+ mask33_31,
+ mask33_32,
+ mask33_33
+ };
+
+ const WebRtc_UWord8* packetMask34[34] =
+ {
+ mask34_1,
+ mask34_2,
+ mask34_3,
+ mask34_4,
+ mask34_5,
+ mask34_6,
+ mask34_7,
+ mask34_8,
+ mask34_9,
+ mask34_10,
+ mask34_11,
+ mask34_12,
+ mask34_13,
+ mask34_14,
+ mask34_15,
+ mask34_16,
+ mask34_17,
+ mask34_18,
+ mask34_19,
+ mask34_20,
+ mask34_21,
+ mask34_22,
+ mask34_23,
+ mask34_24,
+ mask34_25,
+ mask34_26,
+ mask34_27,
+ mask34_28,
+ mask34_29,
+ mask34_30,
+ mask34_31,
+ mask34_32,
+ mask34_33,
+ mask34_34
+ };
+
+ const WebRtc_UWord8* packetMask35[35] =
+ {
+ mask35_1,
+ mask35_2,
+ mask35_3,
+ mask35_4,
+ mask35_5,
+ mask35_6,
+ mask35_7,
+ mask35_8,
+ mask35_9,
+ mask35_10,
+ mask35_11,
+ mask35_12,
+ mask35_13,
+ mask35_14,
+ mask35_15,
+ mask35_16,
+ mask35_17,
+ mask35_18,
+ mask35_19,
+ mask35_20,
+ mask35_21,
+ mask35_22,
+ mask35_23,
+ mask35_24,
+ mask35_25,
+ mask35_26,
+ mask35_27,
+ mask35_28,
+ mask35_29,
+ mask35_30,
+ mask35_31,
+ mask35_32,
+ mask35_33,
+ mask35_34,
+ mask35_35
+ };
+
+ const WebRtc_UWord8* packetMask36[36] =
+ {
+ mask36_1,
+ mask36_2,
+ mask36_3,
+ mask36_4,
+ mask36_5,
+ mask36_6,
+ mask36_7,
+ mask36_8,
+ mask36_9,
+ mask36_10,
+ mask36_11,
+ mask36_12,
+ mask36_13,
+ mask36_14,
+ mask36_15,
+ mask36_16,
+ mask36_17,
+ mask36_18,
+ mask36_19,
+ mask36_20,
+ mask36_21,
+ mask36_22,
+ mask36_23,
+ mask36_24,
+ mask36_25,
+ mask36_26,
+ mask36_27,
+ mask36_28,
+ mask36_29,
+ mask36_30,
+ mask36_31,
+ mask36_32,
+ mask36_33,
+ mask36_34,
+ mask36_35,
+ mask36_36
+ };
+
+ const WebRtc_UWord8* packetMask37[37] =
+ {
+ mask37_1,
+ mask37_2,
+ mask37_3,
+ mask37_4,
+ mask37_5,
+ mask37_6,
+ mask37_7,
+ mask37_8,
+ mask37_9,
+ mask37_10,
+ mask37_11,
+ mask37_12,
+ mask37_13,
+ mask37_14,
+ mask37_15,
+ mask37_16,
+ mask37_17,
+ mask37_18,
+ mask37_19,
+ mask37_20,
+ mask37_21,
+ mask37_22,
+ mask37_23,
+ mask37_24,
+ mask37_25,
+ mask37_26,
+ mask37_27,
+ mask37_28,
+ mask37_29,
+ mask37_30,
+ mask37_31,
+ mask37_32,
+ mask37_33,
+ mask37_34,
+ mask37_35,
+ mask37_36,
+ mask37_37
+ };
+
+ const WebRtc_UWord8* packetMask38[38] =
+ {
+ mask38_1,
+ mask38_2,
+ mask38_3,
+ mask38_4,
+ mask38_5,
+ mask38_6,
+ mask38_7,
+ mask38_8,
+ mask38_9,
+ mask38_10,
+ mask38_11,
+ mask38_12,
+ mask38_13,
+ mask38_14,
+ mask38_15,
+ mask38_16,
+ mask38_17,
+ mask38_18,
+ mask38_19,
+ mask38_20,
+ mask38_21,
+ mask38_22,
+ mask38_23,
+ mask38_24,
+ mask38_25,
+ mask38_26,
+ mask38_27,
+ mask38_28,
+ mask38_29,
+ mask38_30,
+ mask38_31,
+ mask38_32,
+ mask38_33,
+ mask38_34,
+ mask38_35,
+ mask38_36,
+ mask38_37,
+ mask38_38
+ };
+
+ const WebRtc_UWord8* packetMask39[39] =
+ {
+ mask39_1,
+ mask39_2,
+ mask39_3,
+ mask39_4,
+ mask39_5,
+ mask39_6,
+ mask39_7,
+ mask39_8,
+ mask39_9,
+ mask39_10,
+ mask39_11,
+ mask39_12,
+ mask39_13,
+ mask39_14,
+ mask39_15,
+ mask39_16,
+ mask39_17,
+ mask39_18,
+ mask39_19,
+ mask39_20,
+ mask39_21,
+ mask39_22,
+ mask39_23,
+ mask39_24,
+ mask39_25,
+ mask39_26,
+ mask39_27,
+ mask39_28,
+ mask39_29,
+ mask39_30,
+ mask39_31,
+ mask39_32,
+ mask39_33,
+ mask39_34,
+ mask39_35,
+ mask39_36,
+ mask39_37,
+ mask39_38,
+ mask39_39
+ };
+
+ const WebRtc_UWord8* packetMask40[40] =
+ {
+ mask40_1,
+ mask40_2,
+ mask40_3,
+ mask40_4,
+ mask40_5,
+ mask40_6,
+ mask40_7,
+ mask40_8,
+ mask40_9,
+ mask40_10,
+ mask40_11,
+ mask40_12,
+ mask40_13,
+ mask40_14,
+ mask40_15,
+ mask40_16,
+ mask40_17,
+ mask40_18,
+ mask40_19,
+ mask40_20,
+ mask40_21,
+ mask40_22,
+ mask40_23,
+ mask40_24,
+ mask40_25,
+ mask40_26,
+ mask40_27,
+ mask40_28,
+ mask40_29,
+ mask40_30,
+ mask40_31,
+ mask40_32,
+ mask40_33,
+ mask40_34,
+ mask40_35,
+ mask40_36,
+ mask40_37,
+ mask40_38,
+ mask40_39,
+ mask40_40
+ };
+
+ const WebRtc_UWord8* packetMask41[41] =
+ {
+ mask41_1,
+ mask41_2,
+ mask41_3,
+ mask41_4,
+ mask41_5,
+ mask41_6,
+ mask41_7,
+ mask41_8,
+ mask41_9,
+ mask41_10,
+ mask41_11,
+ mask41_12,
+ mask41_13,
+ mask41_14,
+ mask41_15,
+ mask41_16,
+ mask41_17,
+ mask41_18,
+ mask41_19,
+ mask41_20,
+ mask41_21,
+ mask41_22,
+ mask41_23,
+ mask41_24,
+ mask41_25,
+ mask41_26,
+ mask41_27,
+ mask41_28,
+ mask41_29,
+ mask41_30,
+ mask41_31,
+ mask41_32,
+ mask41_33,
+ mask41_34,
+ mask41_35,
+ mask41_36,
+ mask41_37,
+ mask41_38,
+ mask41_39,
+ mask41_40,
+ mask41_41
+ };
+
+ const WebRtc_UWord8* packetMask42[42] =
+ {
+ mask42_1,
+ mask42_2,
+ mask42_3,
+ mask42_4,
+ mask42_5,
+ mask42_6,
+ mask42_7,
+ mask42_8,
+ mask42_9,
+ mask42_10,
+ mask42_11,
+ mask42_12,
+ mask42_13,
+ mask42_14,
+ mask42_15,
+ mask42_16,
+ mask42_17,
+ mask42_18,
+ mask42_19,
+ mask42_20,
+ mask42_21,
+ mask42_22,
+ mask42_23,
+ mask42_24,
+ mask42_25,
+ mask42_26,
+ mask42_27,
+ mask42_28,
+ mask42_29,
+ mask42_30,
+ mask42_31,
+ mask42_32,
+ mask42_33,
+ mask42_34,
+ mask42_35,
+ mask42_36,
+ mask42_37,
+ mask42_38,
+ mask42_39,
+ mask42_40,
+ mask42_41,
+ mask42_42
+ };
+
+ const WebRtc_UWord8* packetMask43[43] =
+ {
+ mask43_1,
+ mask43_2,
+ mask43_3,
+ mask43_4,
+ mask43_5,
+ mask43_6,
+ mask43_7,
+ mask43_8,
+ mask43_9,
+ mask43_10,
+ mask43_11,
+ mask43_12,
+ mask43_13,
+ mask43_14,
+ mask43_15,
+ mask43_16,
+ mask43_17,
+ mask43_18,
+ mask43_19,
+ mask43_20,
+ mask43_21,
+ mask43_22,
+ mask43_23,
+ mask43_24,
+ mask43_25,
+ mask43_26,
+ mask43_27,
+ mask43_28,
+ mask43_29,
+ mask43_30,
+ mask43_31,
+ mask43_32,
+ mask43_33,
+ mask43_34,
+ mask43_35,
+ mask43_36,
+ mask43_37,
+ mask43_38,
+ mask43_39,
+ mask43_40,
+ mask43_41,
+ mask43_42,
+ mask43_43
+ };
+
+ const WebRtc_UWord8* packetMask44[44] =
+ {
+ mask44_1,
+ mask44_2,
+ mask44_3,
+ mask44_4,
+ mask44_5,
+ mask44_6,
+ mask44_7,
+ mask44_8,
+ mask44_9,
+ mask44_10,
+ mask44_11,
+ mask44_12,
+ mask44_13,
+ mask44_14,
+ mask44_15,
+ mask44_16,
+ mask44_17,
+ mask44_18,
+ mask44_19,
+ mask44_20,
+ mask44_21,
+ mask44_22,
+ mask44_23,
+ mask44_24,
+ mask44_25,
+ mask44_26,
+ mask44_27,
+ mask44_28,
+ mask44_29,
+ mask44_30,
+ mask44_31,
+ mask44_32,
+ mask44_33,
+ mask44_34,
+ mask44_35,
+ mask44_36,
+ mask44_37,
+ mask44_38,
+ mask44_39,
+ mask44_40,
+ mask44_41,
+ mask44_42,
+ mask44_43,
+ mask44_44
+ };
+
+ const WebRtc_UWord8* packetMask45[45] =
+ {
+ mask45_1,
+ mask45_2,
+ mask45_3,
+ mask45_4,
+ mask45_5,
+ mask45_6,
+ mask45_7,
+ mask45_8,
+ mask45_9,
+ mask45_10,
+ mask45_11,
+ mask45_12,
+ mask45_13,
+ mask45_14,
+ mask45_15,
+ mask45_16,
+ mask45_17,
+ mask45_18,
+ mask45_19,
+ mask45_20,
+ mask45_21,
+ mask45_22,
+ mask45_23,
+ mask45_24,
+ mask45_25,
+ mask45_26,
+ mask45_27,
+ mask45_28,
+ mask45_29,
+ mask45_30,
+ mask45_31,
+ mask45_32,
+ mask45_33,
+ mask45_34,
+ mask45_35,
+ mask45_36,
+ mask45_37,
+ mask45_38,
+ mask45_39,
+ mask45_40,
+ mask45_41,
+ mask45_42,
+ mask45_43,
+ mask45_44,
+ mask45_45
+ };
+
+ const WebRtc_UWord8* packetMask46[46] =
+ {
+ mask46_1,
+ mask46_2,
+ mask46_3,
+ mask46_4,
+ mask46_5,
+ mask46_6,
+ mask46_7,
+ mask46_8,
+ mask46_9,
+ mask46_10,
+ mask46_11,
+ mask46_12,
+ mask46_13,
+ mask46_14,
+ mask46_15,
+ mask46_16,
+ mask46_17,
+ mask46_18,
+ mask46_19,
+ mask46_20,
+ mask46_21,
+ mask46_22,
+ mask46_23,
+ mask46_24,
+ mask46_25,
+ mask46_26,
+ mask46_27,
+ mask46_28,
+ mask46_29,
+ mask46_30,
+ mask46_31,
+ mask46_32,
+ mask46_33,
+ mask46_34,
+ mask46_35,
+ mask46_36,
+ mask46_37,
+ mask46_38,
+ mask46_39,
+ mask46_40,
+ mask46_41,
+ mask46_42,
+ mask46_43,
+ mask46_44,
+ mask46_45,
+ mask46_46
+ };
+
+ const WebRtc_UWord8* packetMask47[47] =
+ {
+ mask47_1,
+ mask47_2,
+ mask47_3,
+ mask47_4,
+ mask47_5,
+ mask47_6,
+ mask47_7,
+ mask47_8,
+ mask47_9,
+ mask47_10,
+ mask47_11,
+ mask47_12,
+ mask47_13,
+ mask47_14,
+ mask47_15,
+ mask47_16,
+ mask47_17,
+ mask47_18,
+ mask47_19,
+ mask47_20,
+ mask47_21,
+ mask47_22,
+ mask47_23,
+ mask47_24,
+ mask47_25,
+ mask47_26,
+ mask47_27,
+ mask47_28,
+ mask47_29,
+ mask47_30,
+ mask47_31,
+ mask47_32,
+ mask47_33,
+ mask47_34,
+ mask47_35,
+ mask47_36,
+ mask47_37,
+ mask47_38,
+ mask47_39,
+ mask47_40,
+ mask47_41,
+ mask47_42,
+ mask47_43,
+ mask47_44,
+ mask47_45,
+ mask47_46,
+ mask47_47
+ };
+
+ const WebRtc_UWord8* packetMask48[48] =
+ {
+ mask48_1,
+ mask48_2,
+ mask48_3,
+ mask48_4,
+ mask48_5,
+ mask48_6,
+ mask48_7,
+ mask48_8,
+ mask48_9,
+ mask48_10,
+ mask48_11,
+ mask48_12,
+ mask48_13,
+ mask48_14,
+ mask48_15,
+ mask48_16,
+ mask48_17,
+ mask48_18,
+ mask48_19,
+ mask48_20,
+ mask48_21,
+ mask48_22,
+ mask48_23,
+ mask48_24,
+ mask48_25,
+ mask48_26,
+ mask48_27,
+ mask48_28,
+ mask48_29,
+ mask48_30,
+ mask48_31,
+ mask48_32,
+ mask48_33,
+ mask48_34,
+ mask48_35,
+ mask48_36,
+ mask48_37,
+ mask48_38,
+ mask48_39,
+ mask48_40,
+ mask48_41,
+ mask48_42,
+ mask48_43,
+ mask48_44,
+ mask48_45,
+ mask48_46,
+ mask48_47,
+ mask48_48
+ };
+
+ const WebRtc_UWord8** packetMaskTbl[48] =
+ {
+ packetMask1,
+ packetMask2,
+ packetMask3,
+ packetMask4,
+ packetMask5,
+ packetMask6,
+ packetMask7,
+ packetMask8,
+ packetMask9,
+ packetMask10,
+ packetMask11,
+ packetMask12,
+ packetMask13,
+ packetMask14,
+ packetMask15,
+ packetMask16,
+ packetMask17,
+ packetMask18,
+ packetMask19,
+ packetMask20,
+ packetMask21,
+ packetMask22,
+ packetMask23,
+ packetMask24,
+ packetMask25,
+ packetMask26,
+ packetMask27,
+ packetMask28,
+ packetMask29,
+ packetMask30,
+ packetMask31,
+ packetMask32,
+ packetMask33,
+ packetMask34,
+ packetMask35,
+ packetMask36,
+ packetMask37,
+ packetMask38,
+ packetMask39,
+ packetMask40,
+ packetMask41,
+ packetMask42,
+ packetMask43,
+ packetMask44,
+ packetMask45,
+ packetMask46,
+ packetMask47,
+ packetMask48
+ };
+} // Anonymous namespace
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc
new file mode 100644
index 0000000..22e1a38
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -0,0 +1,699 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <cassert>
+#include <cstring>
+#include <iterator>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+// FEC header size in bytes.
+const uint8_t kFecHeaderSize = 10;
+
+// ULP header size in bytes (L bit is set).
+const uint8_t kUlpHeaderSizeLBitSet = (2 + kMaskSizeLBitSet);
+
+// ULP header size in bytes (L bit is cleared).
+const uint8_t kUlpHeaderSizeLBitClear = (2 + kMaskSizeLBitClear);
+
+// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
+const uint8_t kTransportOverhead = 28;
+
+enum { kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets };
+
+// Used to link media packets to their protecting FEC packets.
+//
+// TODO(holmer): Refactor into a proper class.
+class ProtectedPacket : public ForwardErrorCorrection::SortablePacket {
+ public:
+ scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+};
+
+typedef std::list<ProtectedPacket*> ProtectedPacketList;
+
+//
+// Used for internal storage of FEC packets in a list.
+//
+// TODO(holmer): Refactor into a proper class.
+class FecPacket : public ForwardErrorCorrection::SortablePacket {
+ public:
+ ProtectedPacketList protectedPktList;
+ uint32_t ssrc; // SSRC of the current frame.
+ scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+};
+
+bool ForwardErrorCorrection::SortablePacket::LessThan(
+ const SortablePacket* first,
+ const SortablePacket* second) {
+ return (first->seqNum != second->seqNum &&
+ LatestSequenceNumber(first->seqNum, second->seqNum) == second->seqNum);
+}
+
+ForwardErrorCorrection::ForwardErrorCorrection(int32_t id)
+ : _id(id),
+ _generatedFecPackets(kMaxMediaPackets),
+ _fecPacketReceived(false) {
+}
+
+ForwardErrorCorrection::~ForwardErrorCorrection() {
+}
+
+// Input packet
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | RTP Header (12 octets) |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | RTP Payload |
+// | |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+// Output packet
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | FEC Header (10 octets) |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | FEC Level 0 Header |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | FEC Level 0 Payload |
+// | |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+int32_t ForwardErrorCorrection::GenerateFEC(
+ const PacketList& mediaPacketList,
+ uint8_t protectionFactor,
+ int numImportantPackets,
+ bool useUnequalProtection,
+ PacketList* fecPacketList) {
+ if (mediaPacketList.empty()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s media packet list is empty", __FUNCTION__);
+ return -1;
+ }
+ if (!fecPacketList->empty()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s FEC packet list is not empty", __FUNCTION__);
+ return -1;
+ }
+ const uint16_t numMediaPackets = mediaPacketList.size();
+ const uint8_t lBit = numMediaPackets > 16 ? 1 : 0;
+ const uint16_t numMaskBytes = (lBit == 1)?
+ kMaskSizeLBitSet : kMaskSizeLBitClear;
+
+ if (numMediaPackets > kMaxMediaPackets) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s can only protect %d media packets per frame; %d requested",
+ __FUNCTION__, kMaxMediaPackets, numMediaPackets);
+ return -1;
+ }
+
+ // Error checking on the number of important packets.
+ // Can't have more important packets than media packets.
+ if (numImportantPackets > numMediaPackets) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Number of important packets (%d) greater than number of media "
+ "packets (%d)", numImportantPackets, numMediaPackets);
+ return -1;
+ }
+ if (numImportantPackets < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Number of important packets (%d) less than zero",
+ numImportantPackets);
+ return -1;
+ }
+ // Do some error checking on the media packets.
+ PacketList::const_iterator mediaListIt = mediaPacketList.begin();
+ while (mediaListIt != mediaPacketList.end()) {
+ Packet* mediaPacket = *mediaListIt;
+ assert(mediaPacket);
+
+ if (mediaPacket->length < kRtpHeaderSize) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s media packet (%d bytes) is smaller than RTP header",
+ __FUNCTION__, mediaPacket->length);
+ return -1;
+ }
+
+ // Ensure our FEC packets will fit in a typical MTU.
+ if (mediaPacket->length + PacketOverhead() + kTransportOverhead >
+ IP_PACKET_SIZE) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s media packet (%d bytes) with overhead is larger than MTU(%d)",
+ __FUNCTION__, mediaPacket->length, IP_PACKET_SIZE);
+ return -1;
+ }
+ mediaListIt++;
+ }
+ // Result in Q0 with an unsigned round.
+ uint32_t numFecPackets = (numMediaPackets * protectionFactor + (1 << 7)) >> 8;
+ // Generate at least one FEC packet if we need protection.
+ if (protectionFactor > 0 && numFecPackets == 0) {
+ numFecPackets = 1;
+ }
+ if (numFecPackets == 0) {
+ return 0;
+ }
+ assert(numFecPackets <= numMediaPackets);
+
+ // Prepare FEC packets by setting them to 0.
+ for (uint32_t i = 0; i < numFecPackets; i++) {
+ memset(_generatedFecPackets[i].data, 0, IP_PACKET_SIZE);
+ _generatedFecPackets[i].length = 0; // Use this as a marker for untouched
+ // packets.
+ fecPacketList->push_back(&_generatedFecPackets[i]);
+ }
+
+ // -- Generate packet masks --
+ uint8_t* packetMask = new uint8_t[numFecPackets * numMaskBytes];
+ memset(packetMask, 0, numFecPackets * numMaskBytes);
+ internal::GeneratePacketMasks(numMediaPackets, numFecPackets,
+ numImportantPackets, useUnequalProtection,
+ packetMask);
+
+ GenerateFecBitStrings(mediaPacketList, packetMask, numFecPackets);
+
+ GenerateFecUlpHeaders(mediaPacketList, packetMask, numFecPackets);
+
+ delete [] packetMask;
+ return 0;
+}
+
+void ForwardErrorCorrection::GenerateFecBitStrings(
+ const PacketList& mediaPacketList,
+ uint8_t* packetMask,
+ uint32_t numFecPackets) {
+ uint8_t mediaPayloadLength[2];
+ const uint8_t lBit = mediaPacketList.size() > 16 ? 1 : 0;
+ const uint16_t numMaskBytes = (lBit == 1) ?
+ kMaskSizeLBitSet : kMaskSizeLBitClear;
+ const uint16_t ulpHeaderSize = (lBit == 1) ?
+ kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
+ const uint16_t fecRtpOffset = kFecHeaderSize + ulpHeaderSize - kRtpHeaderSize;
+
+ for (uint32_t i = 0; i < numFecPackets; i++) {
+ PacketList::const_iterator mediaListIt = mediaPacketList.begin();
+ uint32_t pktMaskIdx = i * numMaskBytes;
+ uint32_t mediaPktIdx = 0;
+ uint16_t fecPacketLength = 0;
+ while (mediaListIt != mediaPacketList.end()) {
+ // Each FEC packet has a multiple byte mask.
+ if (packetMask[pktMaskIdx] & (1 << (7 - mediaPktIdx))) {
+ Packet* mediaPacket = *mediaListIt;
+
+ // Assign network-ordered media payload length.
+ ModuleRTPUtility::AssignUWord16ToBuffer(
+ mediaPayloadLength,
+ mediaPacket->length - kRtpHeaderSize);
+
+ fecPacketLength = mediaPacket->length + fecRtpOffset;
+ // On the first protected packet, we don't need to XOR.
+ if (_generatedFecPackets[i].length == 0) {
+ // Copy the first 2 bytes of the RTP header.
+ memcpy(_generatedFecPackets[i].data, mediaPacket->data, 2);
+ // Copy the 5th to 8th bytes of the RTP header.
+ memcpy(&_generatedFecPackets[i].data[4], &mediaPacket->data[4], 4);
+ // Copy network-ordered payload size.
+ memcpy(&_generatedFecPackets[i].data[8], mediaPayloadLength, 2);
+
+ // Copy RTP payload, leaving room for the ULP header.
+ memcpy(&_generatedFecPackets[i].data[kFecHeaderSize + ulpHeaderSize],
+ &mediaPacket->data[kRtpHeaderSize],
+ mediaPacket->length - kRtpHeaderSize);
+ } else {
+ // XOR with the first 2 bytes of the RTP header.
+ _generatedFecPackets[i].data[0] ^= mediaPacket->data[0];
+ _generatedFecPackets[i].data[1] ^= mediaPacket->data[1];
+
+ // XOR with the 5th to 8th bytes of the RTP header.
+ for (uint32_t j = 4; j < 8; j++) {
+ _generatedFecPackets[i].data[j] ^= mediaPacket->data[j];
+ }
+
+ // XOR with the network-ordered payload size.
+ _generatedFecPackets[i].data[8] ^= mediaPayloadLength[0];
+ _generatedFecPackets[i].data[9] ^= mediaPayloadLength[1];
+
+ // XOR with RTP payload, leaving room for the ULP header.
+ for (int32_t j = kFecHeaderSize + ulpHeaderSize;
+ j < fecPacketLength; j++) {
+ _generatedFecPackets[i].data[j] ^=
+ mediaPacket->data[j - fecRtpOffset];
+ }
+ }
+ if (fecPacketLength > _generatedFecPackets[i].length) {
+ _generatedFecPackets[i].length = fecPacketLength;
+ }
+ }
+ mediaListIt++;
+ mediaPktIdx++;
+ if (mediaPktIdx == 8) {
+ // Switch to the next mask byte.
+ mediaPktIdx = 0;
+ pktMaskIdx++;
+ }
+ }
+ assert(_generatedFecPackets[i].length);
+ //Note: This shouldn't happen: means packet mask is wrong or poorly designed
+ }
+}
+
+void ForwardErrorCorrection::GenerateFecUlpHeaders(
+ const PacketList& mediaPacketList,
+ uint8_t* packetMask,
+ uint32_t numFecPackets) {
+ // -- Generate FEC and ULP headers --
+ //
+ // FEC Header, 10 bytes
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |E|L|P|X| CC |M| PT recovery | SN base |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | TS recovery |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | length recovery |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ //
+ // ULP Header, 4 bytes (for L = 0)
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | Protection Length | mask |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | mask cont. (present only when L = 1) |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ PacketList::const_iterator mediaListIt = mediaPacketList.begin();
+ Packet* mediaPacket = *mediaListIt;
+ assert(mediaPacket != NULL);
+ const uint8_t lBit = mediaPacketList.size() > 16 ? 1 : 0;
+ const uint16_t numMaskBytes = (lBit == 1)?
+ kMaskSizeLBitSet : kMaskSizeLBitClear;
+ const uint16_t ulpHeaderSize = (lBit == 1)?
+ kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
+
+ for (uint32_t i = 0; i < numFecPackets; i++) {
+ // -- FEC header --
+ _generatedFecPackets[i].data[0] &= 0x7f; // Set E to zero.
+ if (lBit == 0) {
+ _generatedFecPackets[i].data[0] &= 0xbf; // Clear the L bit.
+ } else {
+ _generatedFecPackets[i].data[0] |= 0x40; // Set the L bit.
+ }
+ // Two byte sequence number from first RTP packet to SN base.
+ // We use the same sequence number base for every FEC packet,
+ // but that's not required in general.
+ memcpy(&_generatedFecPackets[i].data[2], &mediaPacket->data[2], 2);
+
+ // -- ULP header --
+ // Copy the payload size to the protection length field.
+ // (We protect the entire packet.)
+ ModuleRTPUtility::AssignUWord16ToBuffer(&_generatedFecPackets[i].data[10],
+ _generatedFecPackets[i].length - kFecHeaderSize - ulpHeaderSize);
+
+ // Copy the packet mask.
+ memcpy(&_generatedFecPackets[i].data[12], &packetMask[i * numMaskBytes],
+ numMaskBytes);
+ }
+}
+
+void ForwardErrorCorrection::ResetState(
+ RecoveredPacketList* recoveredPacketList) {
+ _fecPacketReceived = false;
+
+ // Free the memory for any existing recovered packets, if the user hasn't.
+ while (!recoveredPacketList->empty()) {
+ delete recoveredPacketList->front();
+ recoveredPacketList->pop_front();
+ }
+ assert(recoveredPacketList->empty());
+
+ // Free the FEC packet list.
+ while (!_fecPacketList.empty()) {
+ FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
+ FecPacket* fecPacket = *fecPacketListIt;
+ ProtectedPacketList::iterator protectedPacketListIt;
+ protectedPacketListIt = fecPacket->protectedPktList.begin();
+ while (protectedPacketListIt != fecPacket->protectedPktList.end()) {
+ delete *protectedPacketListIt;
+ protectedPacketListIt =
+ fecPacket->protectedPktList.erase(protectedPacketListIt);
+ }
+ assert(fecPacket->protectedPktList.empty());
+ delete fecPacket;
+ _fecPacketList.pop_front();
+ }
+ assert(_fecPacketList.empty());
+}
+
+void ForwardErrorCorrection::InsertMediaPacket(
+ ReceivedPacket* rxPacket,
+ RecoveredPacketList* recoveredPacketList) {
+ RecoveredPacketList::iterator recoveredPacketListIt =
+ recoveredPacketList->begin();
+
+ // Search for duplicate packets.
+ while (recoveredPacketListIt != recoveredPacketList->end()) {
+ if (rxPacket->seqNum == (*recoveredPacketListIt)->seqNum) {
+ // Duplicate packet, no need to add to list.
+ // Delete duplicate media packet data.
+ rxPacket->pkt = NULL;
+ return;
+ }
+ recoveredPacketListIt++;
+ }
+ RecoveredPacket* recoverdPacketToInsert = new RecoveredPacket;
+ recoverdPacketToInsert->wasRecovered = false;
+ recoverdPacketToInsert->returned = false;
+ recoverdPacketToInsert->seqNum = rxPacket->seqNum;
+ recoverdPacketToInsert->pkt = rxPacket->pkt;
+ recoverdPacketToInsert->pkt->length = rxPacket->pkt->length;
+
+ // TODO(holmer): Consider replacing this with a binary search for the right
+ // position, and then just insert the new packet. Would get rid of the sort.
+ recoveredPacketList->push_back(recoverdPacketToInsert);
+ recoveredPacketList->sort(SortablePacket::LessThan);
+ UpdateCoveringFECPackets(recoverdPacketToInsert);
+}
+
+void ForwardErrorCorrection::UpdateCoveringFECPackets(RecoveredPacket* packet) {
+ for (FecPacketList::iterator it = _fecPacketList.begin();
+ it != _fecPacketList.end(); ++it) {
+ // Is this FEC packet protecting the media packet |packet|?
+ ProtectedPacketList::iterator protected_it = std::lower_bound(
+ (*it)->protectedPktList.begin(),
+ (*it)->protectedPktList.end(),
+ packet,
+ SortablePacket::LessThan);
+ if (protected_it != (*it)->protectedPktList.end() &&
+ (*protected_it)->seqNum == packet->seqNum) {
+ // Found an FEC packet which is protecting |packet|.
+ (*protected_it)->pkt = packet->pkt;
+ }
+ }
+}
+
+void ForwardErrorCorrection::InsertFECPacket(
+ ReceivedPacket* rxPacket,
+ const RecoveredPacketList* recoveredPacketList) {
+ _fecPacketReceived = true;
+
+ // Check for duplicate.
+ FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
+ while (fecPacketListIt != _fecPacketList.end()) {
+ if (rxPacket->seqNum == (*fecPacketListIt)->seqNum) {
+ // Delete duplicate FEC packet data.
+ rxPacket->pkt = NULL;
+ return;
+ }
+ fecPacketListIt++;
+ }
+ FecPacket* fecPacket = new FecPacket;
+ fecPacket->pkt = rxPacket->pkt;
+ fecPacket->seqNum = rxPacket->seqNum;
+ fecPacket->ssrc = rxPacket->ssrc;
+
+ const uint16_t seqNumBase = ModuleRTPUtility::BufferToUWord16(
+ &fecPacket->pkt->data[2]);
+ const uint16_t maskSizeBytes = (fecPacket->pkt->data[0] & 0x40) ?
+ kMaskSizeLBitSet : kMaskSizeLBitClear; // L bit set?
+
+ for (uint16_t byteIdx = 0; byteIdx < maskSizeBytes; byteIdx++) {
+ uint8_t packetMask = fecPacket->pkt->data[12 + byteIdx];
+ for (uint16_t bitIdx = 0; bitIdx < 8; bitIdx++) {
+ if (packetMask & (1 << (7 - bitIdx))) {
+ ProtectedPacket* protectedPacket = new ProtectedPacket;
+ fecPacket->protectedPktList.push_back(protectedPacket);
+ // This wraps naturally with the sequence number.
+ protectedPacket->seqNum = static_cast<uint16_t>(seqNumBase +
+ (byteIdx << 3) + bitIdx);
+ protectedPacket->pkt = NULL;
+ }
+ }
+ }
+ if (fecPacket->protectedPktList.empty()) {
+ // All-zero packet mask; we can discard this FEC packet.
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "FEC packet %u has an all-zero packet mask.",
+ fecPacket->seqNum, __FUNCTION__);
+ delete fecPacket;
+ } else {
+ AssignRecoveredPackets(fecPacket,
+ recoveredPacketList);
+ // TODO(holmer): Consider replacing this with a binary search for the right
+ // position, and then just insert the new packet. Would get rid of the sort.
+ _fecPacketList.push_back(fecPacket);
+ _fecPacketList.sort(SortablePacket::LessThan);
+ if (_fecPacketList.size() > kMaxFecPackets) {
+ DiscardFECPacket(_fecPacketList.front());
+ _fecPacketList.pop_front();
+ }
+ assert(_fecPacketList.size() <= kMaxFecPackets);
+ }
+}
+
+void ForwardErrorCorrection::AssignRecoveredPackets(
+ FecPacket* fec_packet,
+ const RecoveredPacketList* recovered_packets) {
+ // Search for missing packets which have arrived or have been recovered by
+ // another FEC packet.
+ ProtectedPacketList* not_recovered = &fec_packet->protectedPktList;
+ RecoveredPacketList already_recovered;
+ std::set_intersection(
+ recovered_packets->begin(), recovered_packets->end(),
+ not_recovered->begin(), not_recovered->end(),
+ std::inserter(already_recovered, already_recovered.end()),
+ SortablePacket::LessThan);
+ // Set the FEC pointers to all recovered packets so that we don't have to
+ // search for them when we are doing recovery.
+ ProtectedPacketList::iterator not_recovered_it = not_recovered->begin();
+ for (RecoveredPacketList::iterator it = already_recovered.begin();
+ it != already_recovered.end(); ++it) {
+ // Search for the next recovered packet in |not_recovered|.
+ while ((*not_recovered_it)->seqNum != (*it)->seqNum)
+ ++not_recovered_it;
+ (*not_recovered_it)->pkt = (*it)->pkt;
+ }
+}
+
+void ForwardErrorCorrection::InsertPackets(
+ ReceivedPacketList* receivedPacketList,
+ RecoveredPacketList* recoveredPacketList) {
+
+ while (!receivedPacketList->empty()) {
+ ReceivedPacket* rxPacket = receivedPacketList->front();
+
+ if (rxPacket->isFec) {
+ InsertFECPacket(rxPacket, recoveredPacketList);
+ } else {
+ // Insert packet at the end of |recoveredPacketList|.
+ InsertMediaPacket(rxPacket, recoveredPacketList);
+ }
+ // Delete the received packet "wrapper", but not the packet data.
+ delete rxPacket;
+ receivedPacketList->pop_front();
+ }
+ assert(receivedPacketList->empty());
+ DiscardOldPackets(recoveredPacketList);
+}
+
+void ForwardErrorCorrection::InitRecovery(
+ const FecPacket* fec_packet,
+ RecoveredPacket* recovered) {
+ // This is the first packet which we try to recover with.
+ const uint16_t ulpHeaderSize = fec_packet->pkt->data[0] & 0x40 ?
+ kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear; // L bit set?
+ recovered->pkt = new Packet;
+ memset(recovered->pkt->data, 0, IP_PACKET_SIZE);
+ recovered->returned = false;
+ recovered->wasRecovered = true;
+ uint8_t protectionLength[2];
+ // Copy the protection length from the ULP header.
+ memcpy(protectionLength, &fec_packet->pkt->data[10], 2);
+ // Copy FEC payload, skipping the ULP header.
+ memcpy(&recovered->pkt->data[kRtpHeaderSize],
+ &fec_packet->pkt->data[kFecHeaderSize + ulpHeaderSize],
+ ModuleRTPUtility::BufferToUWord16(protectionLength));
+ // Copy the length recovery field.
+ memcpy(recovered->length_recovery, &fec_packet->pkt->data[8], 2);
+ // Copy the first 2 bytes of the FEC header.
+ memcpy(recovered->pkt->data, fec_packet->pkt->data, 2);
+ // Copy the 5th to 8th bytes of the FEC header.
+ memcpy(&recovered->pkt->data[4], &fec_packet->pkt->data[4], 4);
+ // Set the SSRC field.
+ ModuleRTPUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8],
+ fec_packet->ssrc);
+}
+
+void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) {
+ // Set the RTP version to 2.
+ recovered->pkt->data[0] |= 0x80; // Set the 1st bit.
+ recovered->pkt->data[0] &= 0xbf; // Clear the 2nd bit.
+
+ // Set the SN field.
+ ModuleRTPUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2],
+ recovered->seqNum);
+ // Recover the packet length.
+ recovered->pkt->length = ModuleRTPUtility::BufferToUWord16(
+ recovered->length_recovery) + kRtpHeaderSize;
+}
+
+void ForwardErrorCorrection::XorPackets(const Packet* src_packet,
+ RecoveredPacket* dst_packet) {
+ // XOR with the first 2 bytes of the RTP header.
+ for (uint32_t i = 0; i < 2; i++) {
+ dst_packet->pkt->data[i] ^= src_packet->data[i];
+ }
+ // XOR with the 5th to 8th bytes of the RTP header.
+ for (uint32_t i = 4; i < 8; i++) {
+ dst_packet->pkt->data[i] ^= src_packet->data[i];
+ }
+ // XOR with the network-ordered payload size.
+ uint8_t mediaPayloadLength[2];
+ ModuleRTPUtility::AssignUWord16ToBuffer(
+ mediaPayloadLength,
+ src_packet->length - kRtpHeaderSize);
+ dst_packet->length_recovery[0] ^= mediaPayloadLength[0];
+ dst_packet->length_recovery[1] ^= mediaPayloadLength[1];
+
+ // XOR with RTP payload.
+ // TODO(marpan/ajm): Are we doing more XORs than required here?
+ for (int32_t i = kRtpHeaderSize; i < src_packet->length; i++) {
+ dst_packet->pkt->data[i] ^= src_packet->data[i];
+ }
+}
+
+void ForwardErrorCorrection::RecoverPacket(
+ const FecPacket* fecPacket,
+ RecoveredPacket* recPacketToInsert) {
+ InitRecovery(fecPacket, recPacketToInsert);
+ ProtectedPacketList::const_iterator protected_it =
+ fecPacket->protectedPktList.begin();
+ while (protected_it != fecPacket->protectedPktList.end()) {
+ if ((*protected_it)->pkt == NULL) {
+ // This is the packet we're recovering.
+ recPacketToInsert->seqNum = (*protected_it)->seqNum;
+ } else {
+ XorPackets((*protected_it)->pkt, recPacketToInsert);
+ }
+ ++protected_it;
+ }
+ FinishRecovery(recPacketToInsert);
+}
+
+void ForwardErrorCorrection::AttemptRecover(
+ RecoveredPacketList* recoveredPacketList) {
+ FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
+ while (fecPacketListIt != _fecPacketList.end()) {
+ // Search for each FEC packet's protected media packets.
+ int packets_missing = NumCoveredPacketsMissing(*fecPacketListIt);
+
+ // We can only recover one packet with an FEC packet.
+ if (packets_missing == 1) {
+ // Recovery possible.
+ RecoveredPacket* packetToInsert = new RecoveredPacket;
+ packetToInsert->pkt = NULL;
+ RecoverPacket(*fecPacketListIt, packetToInsert);
+
+ // Add recovered packet to the list of recovered packets and update any
+ // FEC packets covering this packet with a pointer to the data.
+ // TODO(holmer): Consider replacing this with a binary search for the
+ // right position, and then just insert the new packet. Would get rid of
+ // the sort.
+ recoveredPacketList->push_back(packetToInsert);
+ recoveredPacketList->sort(SortablePacket::LessThan);
+ UpdateCoveringFECPackets(packetToInsert);
+ DiscardOldPackets(recoveredPacketList);
+ DiscardFECPacket(*fecPacketListIt);
+ fecPacketListIt = _fecPacketList.erase(fecPacketListIt);
+
+ // A packet has been recovered. We need to check the FEC list again, as
+ // this may allow additional packets to be recovered.
+ // Restart for first FEC packet.
+ fecPacketListIt = _fecPacketList.begin();
+ } else if (packets_missing == 0) {
+ // Either all protected packets arrived or have been recovered. We can
+ // discard this FEC packet.
+ DiscardFECPacket(*fecPacketListIt);
+ fecPacketListIt = _fecPacketList.erase(fecPacketListIt);
+ } else {
+ fecPacketListIt++;
+ }
+ }
+}
+
+int ForwardErrorCorrection::NumCoveredPacketsMissing(
+ const FecPacket* fec_packet) {
+ int packets_missing = 0;
+ ProtectedPacketList::const_iterator it = fec_packet->protectedPktList.begin();
+ for (; it != fec_packet->protectedPktList.end(); ++it) {
+ if ((*it)->pkt == NULL) {
+ ++packets_missing;
+ if (packets_missing > 1) {
+ break; // We can't recover more than one packet.
+ }
+ }
+ }
+ return packets_missing;
+}
+
+void ForwardErrorCorrection::DiscardFECPacket(FecPacket* fec_packet) {
+ while (!fec_packet->protectedPktList.empty()) {
+ delete fec_packet->protectedPktList.front();
+ fec_packet->protectedPktList.pop_front();
+ }
+ assert(fec_packet->protectedPktList.empty());
+ delete fec_packet;
+}
+
+void ForwardErrorCorrection::DiscardOldPackets(
+ RecoveredPacketList* recoveredPacketList) {
+ while (recoveredPacketList->size() > kMaxMediaPackets) {
+ ForwardErrorCorrection::RecoveredPacket* packet =
+ recoveredPacketList->front();
+ delete packet;
+ recoveredPacketList->pop_front();
+ }
+ assert(recoveredPacketList->size() <= kMaxMediaPackets);
+}
+
+int32_t ForwardErrorCorrection::DecodeFEC(
+ ReceivedPacketList* receivedPacketList,
+ RecoveredPacketList* recoveredPacketList) {
+ // TODO(marpan/ajm): can we check for multiple ULP headers, and return an
+ // error?
+ InsertPackets(receivedPacketList, recoveredPacketList);
+ AttemptRecover(recoveredPacketList);
+ return 0;
+}
+
+uint16_t ForwardErrorCorrection::PacketOverhead() {
+ return kFecHeaderSize + kUlpHeaderSizeLBitSet;
+}
+
+uint16_t ForwardErrorCorrection::LatestSequenceNumber(uint16_t first,
+ uint16_t second) {
+ bool wrap = (first < 0x00ff && second > 0xff00) ||
+ (first > 0xff00 && second < 0x00ff);
+ if (second > first && !wrap)
+ return second;
+ else if (second <= first && !wrap)
+ return first;
+ else if (second < first && wrap)
+ return second;
+ else
+ return first;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h
new file mode 100644
index 0000000..a7588a8
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h
@@ -0,0 +1,289 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+
+#include <list>
+#include <vector>
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "system_wrappers/interface/ref_count.h"
+#include "system_wrappers/interface/scoped_refptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Forward declaration.
+class FecPacket;
+
+/**
+ * Performs codec-independent forward error correction (FEC), based on RFC 5109.
+ * Option exists to enable unequal protection (UEP) across packets.
+ * This is not to be confused with protection within packets
+ * (referred to as uneven level protection (ULP) in RFC 5109).
+ */
+class ForwardErrorCorrection {
+ public:
+ // Maximum number of media packets we can protect
+ static const unsigned int kMaxMediaPackets = 48u;
+
+ // TODO(holmer): As a next step all these struct-like packet classes should be
+ // refactored into proper classes, and their members should be made private.
+ // This will require parts of the functionality in forward_error_correction.cc
+ // and receiver_fec.cc to be refactored into the packet classes.
+ class Packet {
+ public:
+ Packet() : length(0), data(), ref_count_(0) {}
+ virtual ~Packet() {}
+
+ // Add a reference.
+ virtual int32_t AddRef() {
+ return ++ref_count_;
+ }
+
+ // Release a reference. Will delete the object if the reference count
+ // reaches zero.
+ virtual int32_t Release() {
+ int32_t ref_count;
+ ref_count = --ref_count_;
+ if (ref_count == 0)
+ delete this;
+ return ref_count;
+ }
+
+ uint16_t length; // Length of packet in bytes.
+ uint8_t data[IP_PACKET_SIZE]; // Packet data.
+
+ private:
+ int32_t ref_count_; // Counts the number of references to a packet.
+ };
+
+ // TODO(holmer): Refactor into a proper class.
+ class SortablePacket {
+ public:
+ // True if first is <= than second.
+ static bool LessThan(const SortablePacket* first,
+ const SortablePacket* second);
+
+ uint16_t seqNum;
+ };
+
+ /**
+ * The received list parameter of #DecodeFEC() must reference structs of this
+ * type. The lastMediaPktInFrame is not required to be used for correct
+ * recovery, but will reduce delay by allowing #DecodeFEC() to pre-emptively
+ * determine frame completion. If set, we assume a FEC stream, and the
+ * following assumptions must hold:\n
+ *
+ * 1. The media packets in a frame have contiguous sequence numbers, i.e. the
+ * frame's FEC packets have sequence numbers either lower than the first
+ * media packet or higher than the last media packet.\n
+ * 2. All FEC packets have a sequence number base equal to the first media
+ * packet in the corresponding frame.\n
+ *
+ * The ssrc member is needed to ensure we can restore the SSRC field of
+ * recovered packets. In most situations this could be retrieved from other
+ * media packets, but in the case of an FEC packet protecting a single
+ * missing media packet, we have no other means of obtaining it.
+ */
+ // TODO(holmer): Refactor into a proper class.
+ class ReceivedPacket : public SortablePacket {
+ public:
+ uint32_t ssrc; // SSRC of the current frame. Must be set for FEC
+ // packets, but not required for media packets.
+ bool isFec; // Set to true if this is an FEC packet and false
+ // otherwise.
+ scoped_refptr<Packet> pkt; // Pointer to the packet storage.
+ };
+
+ /**
+ * The recovered list parameter of #DecodeFEC() will reference structs of
+ * this type.
+ */
+ // TODO(holmer): Refactor into a proper class.
+ class RecoveredPacket : public SortablePacket {
+ public:
+ bool wasRecovered; // Will be true if this packet was recovered by
+ // the FEC. Otherwise it was a media packet passed in
+ // through the received packet list.
+ bool returned; // True when the packet already has been returned to the
+ // caller through the callback.
+ uint8_t length_recovery[2]; // Two bytes used for recovering the packet
+ // length with XOR operations.
+ scoped_refptr<Packet> pkt; // Pointer to the packet storage.
+ };
+
+ typedef std::list<Packet*> PacketList;
+ typedef std::list<ReceivedPacket*> ReceivedPacketList;
+ typedef std::list<RecoveredPacket*> RecoveredPacketList;
+
+ /**
+ * \param[in] id Module ID
+ */
+ ForwardErrorCorrection(int32_t id);
+
+ virtual ~ForwardErrorCorrection();
+
+ /**
+ * Generates a list of FEC packets from supplied media packets.
+ *
+ * \param[in] mediaPacketList List of media packets to protect, of type
+ * #Packet. All packets must belong to the
+ * same frame and the list must not be empty.
+ * \param[in] protectionFactor FEC protection overhead in the [0, 255]
+ * domain. To obtain 100% overhead, or an
+ * equal number of FEC packets as media
+ * packets, use 255.
+ * \param[in] numImportantPackets The number of "important" packets in the
+ * frame. These packets may receive greater
+ * protection than the remaining packets. The
+ * important packets must be located at the
+ * start of the media packet list. For codecs
+ * with data partitioning, the important
+ * packets may correspond to first partition
+ * packets.
+ * \param[in] useUnequalProtection Parameter to enable/disable unequal
+ * protection (UEP) across packets. Enabling
+ * UEP will allocate more protection to the
+ * numImportantPackets from the start of the
+ * mediaPacketList.
+ * \param[out] fecPacketList List of FEC packets, of type #Packet. Must
+ * be empty on entry. The memory available
+ * through the list will be valid until the
+ * next call to GenerateFEC().
+ *
+ * \return 0 on success, -1 on failure.
+ */
+ int32_t GenerateFEC(const PacketList& mediaPacketList,
+ uint8_t protectionFactor,
+ int numImportantPackets,
+ bool useUnequalProtection,
+ PacketList* fecPacketList);
+
+ /**
+ * Decodes a list of media and FEC packets. It will parse the input received
+ * packet list, storing FEC packets internally and inserting media packets to
+ * the output recovered packet list. The recovered list will be sorted by
+ * ascending sequence number and have duplicates removed. The function
+ * should be called as new packets arrive, with the recovered list being
+ * progressively assembled with each call. The received packet list will be
+ * empty at output.\n
+ *
+ * The user will allocate packets submitted through the received list. The
+ * function will handle allocation of recovered packets and optionally
+ * deleting of all packet memory. The user may delete the recovered list
+ * packets, in which case they must remove deleted packets from the
+ * recovered list.\n
+ *
+ * \param[in] receivedPacketList List of new received packets, of type
+ * #ReceivedPacket, belonging to a single
+ * frame. At output the list will be empty,
+ * with packets either stored internally,
+ * or accessible through the recovered list.
+ * \param[out] recoveredPacketList List of recovered media packets, of type
+ * #RecoveredPacket, belonging to a single
+ * frame. The memory available through the
+ * list will be valid until the next call to
+ * DecodeFEC().
+ *
+ * \return 0 on success, -1 on failure.
+ */
+ int32_t DecodeFEC(ReceivedPacketList* receivedPacketList,
+ RecoveredPacketList* recoveredPacketList);
+ /**
+ * Gets the size in bytes of the FEC/ULP headers, which must be accounted for
+ * as packet overhead.
+ * \return Packet overhead in bytes.
+ */
+ static uint16_t PacketOverhead();
+
+ // Reset internal states from last frame and clear the recoveredPacketList.
+ // Frees all memory allocated by this class.
+ void ResetState(RecoveredPacketList* recoveredPacketList);
+
+ private:
+ typedef std::list<FecPacket*> FecPacketList;
+
+ void GenerateFecUlpHeaders(const PacketList& mediaPacketList,
+ uint8_t* packetMask,
+ uint32_t numFecPackets);
+
+ void GenerateFecBitStrings(const PacketList& mediaPacketList,
+ uint8_t* packetMask,
+ uint32_t numFecPackets);
+
+ // Insert received packets into FEC or recovered list.
+ void InsertPackets(ReceivedPacketList* receivedPacketList,
+ RecoveredPacketList* recoveredPacketList);
+
+ // Insert media packet into recovered packet list. We delete duplicates.
+ void InsertMediaPacket(ReceivedPacket* rxPacket,
+ RecoveredPacketList* recoveredPacketList);
+
+ // Assigns pointers to the recovered packet from all FEC packets which cover
+ // it.
+ // Note: This reduces the complexity when we want to try to recover a packet
+ // since we don't have to find the intersection between recovered packets and
+ // packets covered by the FEC packet.
+ void UpdateCoveringFECPackets(RecoveredPacket* packet);
+
+ // Insert packet into FEC list. We delete duplicates.
+ void InsertFECPacket(ReceivedPacket* rxPacket,
+ const RecoveredPacketList* recoveredPacketList);
+
+ // Assigns pointers to already recovered packets covered by this FEC packet.
+ static void AssignRecoveredPackets(
+ FecPacket* fec_packet,
+ const RecoveredPacketList* recovered_packets);
+
+ // Insert into recovered list in correct position.
+ void InsertRecoveredPacket(
+ RecoveredPacket* recPacketToInsert,
+ RecoveredPacketList* recoveredPacketList);
+
+ // Attempt to recover missing packets.
+ void AttemptRecover(RecoveredPacketList* recoveredPacketList);
+
+ // Initializes the packet recovery using the FEC packet.
+ static void InitRecovery(const FecPacket* fec_packet,
+ RecoveredPacket* recovered);
+
+ // Performs XOR between |src_packet| and |dst_packet| and stores the result
+ // in |dst_packet|.
+ static void XorPackets(const Packet* src_packet,
+ RecoveredPacket* dst_packet);
+
+ // Finish up the recovery of a packet.
+ static void FinishRecovery(RecoveredPacket* recovered);
+
+ // Recover a missing packet.
+ void RecoverPacket(const FecPacket* fecPacket,
+ RecoveredPacket* recPacketToInsert);
+
+ // Get the number of missing media packets which are covered by this
+ // FEC packet. An FEC packet can recover at most one packet, and if zero
+ // packets are missing the FEC packet can be discarded.
+ // This function returns 2 when two or more packets are missing.
+ static int NumCoveredPacketsMissing(const FecPacket* fec_packet);
+
+ static uint16_t LatestSequenceNumber(uint16_t first,
+ uint16_t second);
+
+ static void DiscardFECPacket(FecPacket* fec_packet);
+ static void DiscardOldPackets(RecoveredPacketList* recoveredPacketList);
+
+ int32_t _id;
+ std::vector<Packet> _generatedFecPackets;
+ FecPacketList _fecPacketList;
+ bool _fecPacketReceived;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc
new file mode 100644
index 0000000..5338174
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -0,0 +1,381 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "forward_error_correction_internal.h"
+#include "fec_private_tables.h"
+
+#include <cassert>
+#include <cstring>
+
+namespace {
+
+// Allow for different modes of protection for packets in UEP case.
+enum ProtectionMode
+{
+ kModeNoOverlap,
+ kModeOverlap,
+ kModeBiasFirstPacket,
+};
+
+/**
+ * Fits an input mask (subMask) to an output mask.
+ * The mask is a matrix where the rows are the FEC packets,
+ * and the columns are the source packets the FEC is applied to.
+ * Each row of the mask is represented by a number of mask bytes.
+ *
+ * \param[in] numMaskBytes The number of mask bytes of output mask.
+ * \param[in] numSubMaskBytes The number of mask bytes of input mask.
+ * \param[in] numRows The number of rows of the input mask.
+ * \param[in] subMask A pointer to hold the input mask, of size
+ * [0, numRows * numSubMaskBytes]
+ * \param[out] packetMask A pointer to hold the output mask, of size
+ * [0, x * numMaskBytes], where x >= numRows.
+ */
+void FitSubMask(int numMaskBytes,
+ int numSubMaskBytes,
+ int numRows,
+ const uint8_t* subMask,
+ uint8_t* packetMask)
+{
+ if (numMaskBytes == numSubMaskBytes)
+ {
+ memcpy(packetMask, subMask, numRows * numSubMaskBytes);
+ }
+ else
+ {
+ for (int i = 0; i < numRows; i++)
+ {
+ int pktMaskIdx = i * numMaskBytes;
+ int pktMaskIdx2 = i * numSubMaskBytes;
+ for (int j = 0; j < numSubMaskBytes; j++)
+ {
+ packetMask[pktMaskIdx] = subMask[pktMaskIdx2];
+ pktMaskIdx++;
+ pktMaskIdx2++;
+ }
+ }
+ }
+}
+
+/**
+ * Shifts a mask by number of columns (bits), and fits it to an output mask.
+ * The mask is a matrix where the rows are the FEC packets,
+ * and the columns are the source packets the FEC is applied to.
+ * Each row of the mask is represented by a number of mask bytes.
+ *
+ * \param[in] numMaskBytes The number of mask bytes of output mask.
+ * \param[in] numSubMaskBytes The number of mask bytes of input mask.
+ * \param[in] numColumnShift The number columns to be shifted, and
+ * the starting row for the output mask.
+ * \param[in] endRow The ending row for the output mask.
+ * \param[in] subMask A pointer to hold the input mask, of size
+ * [0, (endRowFEC - startRowFec) * numSubMaskBytes]
+ * \param[out] packetMask A pointer to hold the output mask, of size
+ * [0, x * numMaskBytes], where x >= endRowFEC.
+ */
+// TODO (marpan): This function is doing three things at the same time:
+// shift within a byte, byte shift and resizing.
+// Split up into subroutines.
+void ShiftFitSubMask(int numMaskBytes,
+ int resMaskBytes,
+ int numColumnShift,
+ int endRow,
+ const uint8_t* subMask,
+ uint8_t* packetMask)
+{
+
+ // Number of bit shifts within a byte
+ const int numBitShifts = (numColumnShift % 8);
+ const int numByteShifts = numColumnShift >> 3;
+
+ // Modify new mask with sub-mask21.
+
+ // Loop over the remaining FEC packets.
+ for (int i = numColumnShift; i < endRow; i++)
+ {
+ // Byte index of new mask, for row i and column resMaskBytes,
+ // offset by the number of bytes shifts
+ int pktMaskIdx = i * numMaskBytes + resMaskBytes - 1 + numByteShifts;
+ // Byte index of subMask, for row i and column resMaskBytes
+ int pktMaskIdx2 =
+ (i - numColumnShift) * resMaskBytes + resMaskBytes - 1;
+
+ uint8_t shiftRightCurrByte = 0;
+ uint8_t shiftLeftPrevByte = 0;
+ uint8_t combNewByte = 0;
+
+ // Handle case of numMaskBytes > resMaskBytes:
+ // For a given row, copy the rightmost "numBitShifts" bits
+ // of the last byte of subMask into output mask.
+ if (numMaskBytes > resMaskBytes)
+ {
+ shiftLeftPrevByte =
+ (subMask[pktMaskIdx2] << (8 - numBitShifts));
+ packetMask[pktMaskIdx + 1] = shiftLeftPrevByte;
+ }
+
+ // For each row i (FEC packet), shift the bit-mask of the subMask.
+ // Each row of the mask contains "resMaskBytes" of bytes.
+ // We start from the last byte of the subMask and move to first one.
+ for (int j = resMaskBytes - 1; j > 0; j--)
+ {
+ // Shift current byte of sub21 to the right by "numBitShifts".
+ shiftRightCurrByte =
+ subMask[pktMaskIdx2] >> numBitShifts;
+
+ // Fill in shifted bits with bits from the previous (left) byte:
+ // First shift the previous byte to the left by "8-numBitShifts".
+ shiftLeftPrevByte =
+ (subMask[pktMaskIdx2 - 1] << (8 - numBitShifts));
+
+ // Then combine both shifted bytes into new mask byte.
+ combNewByte = shiftRightCurrByte | shiftLeftPrevByte;
+
+ // Assign to new mask.
+ packetMask[pktMaskIdx] = combNewByte;
+ pktMaskIdx--;
+ pktMaskIdx2--;
+ }
+ // For the first byte in the row (j=0 case).
+ shiftRightCurrByte = subMask[pktMaskIdx2] >> numBitShifts;
+ packetMask[pktMaskIdx] = shiftRightCurrByte;
+
+ }
+}
+
+} //namespace
+
+namespace webrtc {
+namespace internal {
+
+// Remaining protection after important (first partition) packet protection
+void RemainingPacketProtection(int numMediaPackets,
+ int numFecRemaining,
+ int numFecForImpPackets,
+ int numMaskBytes,
+ ProtectionMode mode,
+ uint8_t* packetMask)
+{
+ if (mode == kModeNoOverlap)
+ {
+ // subMask21
+
+ const int lBit =
+ (numMediaPackets - numFecForImpPackets) > 16 ? 1 : 0;
+
+ const int resMaskBytes =
+ (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+
+ const uint8_t* packetMaskSub21 =
+ packetMaskTbl[numMediaPackets - numFecForImpPackets - 1]
+ [numFecRemaining - 1];
+
+ ShiftFitSubMask(numMaskBytes, resMaskBytes, numFecForImpPackets,
+ (numFecForImpPackets + numFecRemaining),
+ packetMaskSub21, packetMask);
+
+ }
+ else if (mode == kModeOverlap || mode == kModeBiasFirstPacket)
+ {
+ // subMask22
+
+ const uint8_t* packetMaskSub22 =
+ packetMaskTbl[numMediaPackets - 1][numFecRemaining - 1];
+
+ FitSubMask(numMaskBytes, numMaskBytes, numFecRemaining, packetMaskSub22,
+ &packetMask[numFecForImpPackets * numMaskBytes]);
+
+ if (mode == kModeBiasFirstPacket)
+ {
+ for (int i = 0; i < numFecRemaining; i++)
+ {
+ int pktMaskIdx = i * numMaskBytes;
+ packetMask[pktMaskIdx] = packetMask[pktMaskIdx] | (1 << 7);
+ }
+ }
+ }
+ else
+ {
+ assert(false);
+ }
+
+}
+
+// Protection for important (first partition) packets
+void ImportantPacketProtection(int numFecForImpPackets,
+ int numImpPackets,
+ int numMaskBytes,
+ uint8_t* packetMask)
+{
+ const int lBit = numImpPackets > 16 ? 1 : 0;
+ const int numImpMaskBytes =
+ (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+
+ // Get subMask1 from table
+ const uint8_t* packetMaskSub1 =
+ packetMaskTbl[numImpPackets - 1][numFecForImpPackets - 1];
+
+ FitSubMask(numMaskBytes, numImpMaskBytes,
+ numFecForImpPackets, packetMaskSub1, packetMask);
+
+}
+
+// This function sets the protection allocation: i.e., how many FEC packets
+// to use for numImp (1st partition) packets, given the: number of media
+// packets, number of FEC packets, and number of 1st partition packets.
+int SetProtectionAllocation(int numMediaPackets,
+ int numFecPackets,
+ int numImpPackets)
+{
+
+ // TODO (marpan): test different cases for protection allocation:
+
+ // Use at most (allocPar * numFecPackets) for important packets.
+ float allocPar = 0.5;
+ int maxNumFecForImp = allocPar * numFecPackets;
+
+ int numFecForImpPackets = (numImpPackets < maxNumFecForImp) ?
+ numImpPackets : maxNumFecForImp;
+
+ // Fall back to equal protection in this case
+ if (numFecPackets == 1 && (numMediaPackets > 2 * numImpPackets))
+ {
+ numFecForImpPackets = 0;
+ }
+
+ return numFecForImpPackets;
+}
+
+// Modification for UEP: reuse the off-line tables for the packet masks.
+// Note: these masks were designed for equal packet protection case,
+// assuming random packet loss.
+
+// Current version has 3 modes (options) to build UEP mask from existing ones.
+// Various other combinations may be added in future versions.
+// Longer-term, we may add another set of tables specifically for UEP cases.
+// TODO (marpan): also consider modification of masks for bursty loss cases.
+
+// Mask is characterized as (#packets_to_protect, #fec_for_protection).
+// Protection factor defined as: (#fec_for_protection / #packets_to_protect).
+
+// Let k=numMediaPackets, n=total#packets, (n-k)=numFecPackets, m=numImpPackets.
+
+// For ProtectionMode 0 and 1:
+// one mask (subMask1) is used for 1st partition packets,
+// the other mask (subMask21/22, for 0/1) is for the remaining FEC packets.
+
+// In both mode 0 and 1, the packets of 1st partition (numImpPackets) are
+// treated equally important, and are afforded more protection than the
+// residual partition packets.
+
+// For numImpPackets:
+// subMask1 = (m, t): protection = t/(m), where t=F(k,n-k,m).
+// t=F(k,n-k,m) is the number of packets used to protect first partition in
+// subMask1. This is determined from the function SetProtectionAllocation().
+
+// For the left-over protection:
+// Mode 0: subMask21 = (k-m,n-k-t): protection = (n-k-t)/(k-m)
+// mode 0 has no protection overlap between the two partitions.
+// For mode 0, we would typically set t = min(m, n-k).
+
+
+// Mode 1: subMask22 = (k, n-k-t), with protection (n-k-t)/(k)
+// mode 1 has protection overlap between the two partitions (preferred).
+
+// For ProtectionMode 2:
+// This gives 1st packet of list (which is 1st packet of 1st partition) more
+// protection. In mode 2, the equal protection mask (which is obtained from
+// mode 1 for t=0) is modified (more "1s" added in 1st column of packet mask)
+// to bias higher protection for the 1st source packet.
+
+// Protection Mode 2 may be extended for a sort of sliding protection
+// (i.e., vary the number/density of "1s" across columns) across packets.
+
+void UnequalProtectionMask(int numMediaPackets,
+ int numFecPackets,
+ int numImpPackets,
+ int numMaskBytes,
+ uint8_t* packetMask)
+{
+
+ // Set Protection type and allocation
+ // TODO (marpan): test/update for best mode and some combinations thereof.
+
+ ProtectionMode mode = kModeOverlap;
+ int numFecForImpPackets = 0;
+
+ if (mode != kModeBiasFirstPacket)
+ {
+ numFecForImpPackets = SetProtectionAllocation(numMediaPackets,
+ numFecPackets,
+ numImpPackets);
+ }
+
+ int numFecRemaining = numFecPackets - numFecForImpPackets;
+ // Done with setting protection type and allocation
+
+ //
+ // Generate subMask1
+ //
+ if (numFecForImpPackets > 0)
+ {
+ ImportantPacketProtection(numFecForImpPackets, numImpPackets,
+ numMaskBytes, packetMask);
+ }
+
+ //
+ // Generate subMask2
+ //
+ if (numFecRemaining > 0)
+ {
+ RemainingPacketProtection(numMediaPackets, numFecRemaining,
+ numFecForImpPackets, numMaskBytes,
+ mode, packetMask);
+ }
+
+}
+
+void GeneratePacketMasks(int numMediaPackets,
+ int numFecPackets,
+ int numImpPackets,
+ bool useUnequalProtection,
+ uint8_t* packetMask)
+{
+ assert(numMediaPackets <= static_cast<int>(sizeof(packetMaskTbl) /
+ sizeof(*packetMaskTbl)));
+ assert(numMediaPackets > 0);
+ assert(numFecPackets <= numMediaPackets && numFecPackets > 0);
+ assert(numImpPackets <= numMediaPackets && numImpPackets >= 0);
+
+ int lBit = numMediaPackets > 16 ? 1 : 0;
+ const int numMaskBytes =
+ (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+
+ // Equal-protection for these cases
+ if (!useUnequalProtection || numImpPackets == 0)
+ {
+ // Retrieve corresponding mask table directly:for equal-protection case.
+ // Mask = (k,n-k), with protection factor = (n-k)/k,
+ // where k = numMediaPackets, n=total#packets, (n-k)=numFecPackets.
+ memcpy(packetMask,
+ packetMaskTbl[numMediaPackets - 1][numFecPackets - 1],
+ numFecPackets * numMaskBytes);
+ }
+ else //UEP case
+ {
+ UnequalProtectionMask(numMediaPackets, numFecPackets, numImpPackets,
+ numMaskBytes, packetMask);
+
+ } // End of UEP modification
+
+} //End of GetPacketMasks
+
+} // namespace internal
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h
new file mode 100644
index 0000000..e137ef2
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Packet mask size in bytes (L bit is set).
+static const int kMaskSizeLBitSet = 6;
+// Packet mask size in bytes (L bit is cleared).
+static const int kMaskSizeLBitClear = 2;
+
+namespace internal {
+
+ /**
+ * Returns an array of packet masks. The mask of a single FEC packet
+ * corresponds to a number of mask bytes. The mask indicates which
+ * media packets should be protected by the FEC packet.
+
+ * \param[in] numMediaPackets The number of media packets to protect.
+ * [1, maxMediaPackets].
+ * \param[in] numFecPackets The number of FEC packets which will
+ * be generated. [1, numMediaPackets].
+ * \param[in] numImpPackets The number of important packets.
+ * [0, numMediaPackets].
+ * numImpPackets = 0 is the equal
+ * protection scenario.
+ * \param[in] useUnequalProtection Enables unequal protection: allocates
+ * more protection to the numImpPackets.
+ * \param[out] packetMask A pointer to hold the packet mask array,
+ * of size:
+ * numFecPackets * "number of mask bytes".
+ */
+void GeneratePacketMasks(int numMediaPackets,
+ int numFecPackets,
+ int numImpPackets,
+ bool useUnequalProtection,
+ uint8_t* packetMask);
+
+} // namespace internal
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h b/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
new file mode 100644
index 0000000..cb7ebba
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
+
+namespace webrtc {
+
+class MockRTPReceiverVideo : public RTPReceiverVideo {
+ public:
+ MOCK_METHOD1(ChangeUniqueId,
+ void(const WebRtc_Word32 id));
+ MOCK_METHOD3(ReceiveRecoveredPacketCallback,
+ WebRtc_Word32(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength));
+ MOCK_METHOD3(CallbackOfReceivedPayloadData,
+ WebRtc_Word32(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader));
+ MOCK_CONST_METHOD0(TimeStamp,
+ WebRtc_UWord32());
+ MOCK_CONST_METHOD0(SequenceNumber,
+ WebRtc_UWord16());
+ MOCK_CONST_METHOD2(PayloadTypeToPayload,
+ WebRtc_UWord32(const WebRtc_UWord8 payloadType,
+ ModuleRTPUtility::Payload*& payload));
+ MOCK_CONST_METHOD2(RetransmitOfOldPacket,
+ bool(const WebRtc_UWord16 sequenceNumber,
+ const WebRtc_UWord32 rtpTimeStamp));
+ MOCK_CONST_METHOD0(REDPayloadType,
+ WebRtc_Word8());
+};
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc b/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc
new file mode 100644
index 0000000..4cffb0f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc
@@ -0,0 +1,450 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if _WIN32
+#include <windows.h>
+#endif
+
+#include "trace.h"
+#include "overuse_detector.h"
+#include "remote_rate_control.h"
+#include "rtp_utility.h"
+#include <math.h>
+#include <stdlib.h> //abs
+
+#ifdef WEBRTC_BWE_MATLAB
+extern MatlabEngine eng; // global variable defined elsewhere
+#endif
+
+#define INIT_CAPACITY_SLOPE 8.0/512.0
+#define DETECTOR_THRESHOLD 25.0
+#define OVER_USING_TIME_THRESHOLD 100
+#define MIN_FRAME_PERIOD_HISTORY_LEN 60
+
+namespace webrtc {
+OverUseDetector::OverUseDetector()
+:
+_firstPacket(true),
+_currentFrame(),
+_prevFrame(),
+_numOfDeltas(0),
+_slope(INIT_CAPACITY_SLOPE),
+_offset(0),
+_E(),
+_processNoise(),
+_avgNoise(0.0),
+_varNoise(500),
+_threshold(DETECTOR_THRESHOLD),
+_tsDeltaHist(),
+_prevOffset(0.0),
+_timeOverUsing(-1),
+_overUseCounter(0),
+_hypothesis(kBwNormal)
+#ifdef WEBRTC_BWE_MATLAB
+,_plot1(NULL),
+_plot2(NULL),
+_plot3(NULL),
+_plot4(NULL)
+#endif
+{
+ _E[0][0] = 100;
+ _E[1][1] = 1e-1;
+ _E[0][1] = _E[1][0] = 0;
+ _processNoise[0] = 1e-10;
+ _processNoise[1] = 1e-2;
+}
+
+OverUseDetector::~OverUseDetector()
+{
+#ifdef WEBRTC_BWE_MATLAB
+ if (_plot1)
+ {
+ eng.DeletePlot(_plot1);
+ _plot1 = NULL;
+ }
+ if (_plot2)
+ {
+ eng.DeletePlot(_plot2);
+ _plot2 = NULL;
+ }
+ if (_plot3)
+ {
+ eng.DeletePlot(_plot3);
+ _plot3 = NULL;
+ }
+ if (_plot4)
+ {
+ eng.DeletePlot(_plot4);
+ _plot4 = NULL;
+ }
+#endif
+
+ _tsDeltaHist.clear();
+}
+
+void OverUseDetector::Reset()
+{
+ _firstPacket = true;
+ _currentFrame._size = 0;
+ _currentFrame._completeTimeMs = -1;
+ _currentFrame._timestamp = -1;
+ _prevFrame._size = 0;
+ _prevFrame._completeTimeMs = -1;
+ _prevFrame._timestamp = -1;
+ _numOfDeltas = 0;
+ _slope = INIT_CAPACITY_SLOPE;
+ _offset = 0;
+ _E[0][0] = 100;
+ _E[1][1] = 1e-1;
+ _E[0][1] = _E[1][0] = 0;
+ _processNoise[0] = 1e-10;
+ _processNoise[1] = 1e-2;
+ _avgNoise = 0.0;
+ _varNoise = 500;
+ _threshold = DETECTOR_THRESHOLD;
+ _prevOffset = 0.0;
+ _timeOverUsing = -1;
+ _overUseCounter = 0;
+ _hypothesis = kBwNormal;
+ _tsDeltaHist.clear();
+}
+
+bool OverUseDetector::Update(const WebRtcRTPHeader& rtpHeader,
+ const WebRtc_UWord16 packetSize,
+ const WebRtc_Word64 nowMS)
+{
+#ifdef WEBRTC_BWE_MATLAB
+ // Create plots
+ const WebRtc_Word64 startTimeMs = nowMS;
+ if (_plot1 == NULL)
+ {
+ _plot1 = eng.NewPlot(new MatlabPlot());
+ _plot1->AddLine(1000, "b.", "scatter");
+ }
+ if (_plot2 == NULL)
+ {
+ _plot2 = eng.NewPlot(new MatlabPlot());
+ _plot2->AddTimeLine(30, "b", "offset", startTimeMs);
+ _plot2->AddTimeLine(30, "r--", "limitPos", startTimeMs);
+ _plot2->AddTimeLine(30, "k.", "trigger", startTimeMs);
+ _plot2->AddTimeLine(30, "ko", "detection", startTimeMs);
+ //_plot2->AddTimeLine(30, "g", "slowMean", startTimeMs);
+ }
+ if (_plot3 == NULL)
+ {
+ _plot3 = eng.NewPlot(new MatlabPlot());
+ _plot3->AddTimeLine(30, "b", "noiseVar", startTimeMs);
+ }
+ if (_plot4 == NULL)
+ {
+ _plot4 = eng.NewPlot(new MatlabPlot());
+ //_plot4->AddTimeLine(60, "b", "p11", startTimeMs);
+ //_plot4->AddTimeLine(60, "r", "p12", startTimeMs);
+ _plot4->AddTimeLine(60, "g", "p22", startTimeMs);
+ //_plot4->AddTimeLine(60, "g--", "p22_hat", startTimeMs);
+ //_plot4->AddTimeLine(30, "b.-", "deltaFs", startTimeMs);
+ }
+
+#endif
+
+ bool wrapped = false;
+ bool completeFrame = false;
+ if (_currentFrame._timestamp == -1)
+ {
+ _currentFrame._timestamp = rtpHeader.header.timestamp;
+ }
+ else if (ModuleRTPUtility::OldTimestamp(
+ rtpHeader.header.timestamp,
+ static_cast<WebRtc_UWord32>(_currentFrame._timestamp),
+ &wrapped))
+ {
+ // Don't update with old data
+ return completeFrame;
+ }
+ else if (rtpHeader.header.timestamp != _currentFrame._timestamp)
+ {
+ // First packet of a later frame, the previous frame sample is ready
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "Frame complete at %I64i", _currentFrame._completeTimeMs);
+ if (_prevFrame._completeTimeMs >= 0) // This is our second frame
+ {
+ WebRtc_Word64 tDelta = 0;
+ double tsDelta = 0;
+ // Check for wrap
+ ModuleRTPUtility::OldTimestamp(
+ static_cast<WebRtc_UWord32>(_prevFrame._timestamp),
+ static_cast<WebRtc_UWord32>(_currentFrame._timestamp),
+ &wrapped);
+ CompensatedTimeDelta(_currentFrame, _prevFrame, tDelta, tsDelta, wrapped);
+ UpdateKalman(tDelta, tsDelta, _currentFrame._size, _prevFrame._size);
+ }
+ // The new timestamp is now the current frame,
+ // and the old timestamp becomes the previous frame.
+ _prevFrame = _currentFrame;
+ _currentFrame._timestamp = rtpHeader.header.timestamp;
+ _currentFrame._size = 0;
+ _currentFrame._completeTimeMs = -1;
+ completeFrame = true;
+ }
+ // Accumulate the frame size
+ _currentFrame._size += packetSize;
+ _currentFrame._completeTimeMs = nowMS;
+ return completeFrame;
+}
+
+BandwidthUsage OverUseDetector::State() const
+{
+ return _hypothesis;
+}
+
+double OverUseDetector::NoiseVar() const
+{
+ return _varNoise;
+}
+
+void OverUseDetector::SetRateControlRegion(RateControlRegion region)
+{
+ switch (region)
+ {
+ case kRcMaxUnknown:
+ {
+ _threshold = DETECTOR_THRESHOLD;
+ break;
+ }
+ case kRcAboveMax:
+ case kRcNearMax:
+ {
+ _threshold = DETECTOR_THRESHOLD / 2;
+ break;
+ }
+ }
+}
+
+void OverUseDetector::CompensatedTimeDelta(const FrameSample& currentFrame, const FrameSample& prevFrame, WebRtc_Word64& tDelta,
+ double& tsDelta, bool wrapped)
+{
+ _numOfDeltas++;
+ if (_numOfDeltas > 1000)
+ {
+ _numOfDeltas = 1000;
+ }
+ // Add wrap-around compensation
+ WebRtc_Word64 wrapCompensation = 0;
+ if (wrapped)
+ {
+ wrapCompensation = static_cast<WebRtc_Word64>(1)<<32;
+ }
+ tsDelta = (currentFrame._timestamp + wrapCompensation - prevFrame._timestamp) / 90.0;
+ tDelta = currentFrame._completeTimeMs - prevFrame._completeTimeMs;
+ assert(tsDelta > 0);
+}
+
+double OverUseDetector::CurrentDrift()
+{
+ return 1.0;
+}
+
+void OverUseDetector::UpdateKalman(WebRtc_Word64 tDelta, double tsDelta, WebRtc_UWord32 frameSize, WebRtc_UWord32 prevFrameSize)
+{
+ const double minFramePeriod = UpdateMinFramePeriod(tsDelta);
+ const double drift = CurrentDrift();
+ // Compensate for drift
+ const double tTsDelta = tDelta - tsDelta / drift;
+ double fsDelta = static_cast<double>(frameSize) - prevFrameSize;
+
+ // Update the Kalman filter
+ const double scaleFactor = minFramePeriod / (1000.0 / 30.0);
+ _E[0][0] += _processNoise[0] * scaleFactor;
+ _E[1][1] += _processNoise[1] * scaleFactor;
+
+ if ((_hypothesis == kBwOverusing && _offset < _prevOffset) ||
+ (_hypothesis == kBwUnderUsing && _offset > _prevOffset))
+ {
+ _E[1][1] += 10 * _processNoise[1] * scaleFactor;
+ }
+
+ const double h[2] = {fsDelta, 1.0};
+ const double Eh[2] = {_E[0][0]*h[0] + _E[0][1]*h[1],
+ _E[1][0]*h[0] + _E[1][1]*h[1]};
+
+ const double residual = tTsDelta - _slope*h[0] - _offset;
+
+ const bool stableState = (BWE_MIN(_numOfDeltas, 60) * abs(_offset) < _threshold);
+ // We try to filter out very late frames. For instance periodic key
+ // frames doesn't fit the Gaussian model well.
+ if (abs(residual) < 3 * sqrt(_varNoise))
+ {
+ UpdateNoiseEstimate(residual, minFramePeriod, stableState);
+ }
+ else
+ {
+ UpdateNoiseEstimate(3 * sqrt(_varNoise), minFramePeriod, stableState);
+ }
+
+ const double denom = _varNoise + h[0]*Eh[0] + h[1]*Eh[1];
+
+ const double K[2] = {Eh[0] / denom,
+ Eh[1] / denom};
+
+ const double IKh[2][2] = {{1.0 - K[0]*h[0], -K[0]*h[1]},
+ {-K[1]*h[0], 1.0 - K[1]*h[1]}};
+ const double e00 = _E[0][0];
+ const double e01 = _E[0][1];
+
+ // Update state
+ _E[0][0] = e00 * IKh[0][0] + _E[1][0] * IKh[0][1];
+ _E[0][1] = e01 * IKh[0][0] + _E[1][1] * IKh[0][1];
+ _E[1][0] = e00 * IKh[1][0] + _E[1][0] * IKh[1][1];
+ _E[1][1] = e01 * IKh[1][0] + _E[1][1] * IKh[1][1];
+
+ // Covariance matrix, must be positive semi-definite
+ assert(_E[0][0] + _E[1][1] >= 0 &&
+ _E[0][0] * _E[1][1] - _E[0][1] * _E[1][0] >= 0 &&
+ _E[0][0] >= 0);
+
+#ifdef WEBRTC_BWE_MATLAB
+ //_plot4->Append("p11",_E[0][0]);
+ //_plot4->Append("p12",_E[0][1]);
+ _plot4->Append("p22",_E[1][1]);
+ //_plot4->Append("p22_hat", 0.5*(_processNoise[1] +
+ // sqrt(_processNoise[1]*(_processNoise[1] + 4*_varNoise))));
+ //_plot4->Append("deltaFs", fsDelta);
+ _plot4->Plot();
+#endif
+ _slope = _slope + K[0] * residual;
+ _prevOffset = _offset;
+ _offset = _offset + K[1] * residual;
+
+ Detect(tsDelta);
+
+#ifdef WEBRTC_BWE_MATLAB
+ _plot1->Append("scatter", static_cast<double>(_currentFrame._size) - _prevFrame._size,
+ static_cast<double>(tDelta-tsDelta));
+ _plot1->MakeTrend("scatter", "slope", _slope, _offset, "k-");
+ _plot1->MakeTrend("scatter", "thresholdPos", _slope, _offset + 2 * sqrt(_varNoise), "r-");
+ _plot1->MakeTrend("scatter", "thresholdNeg", _slope, _offset - 2 * sqrt(_varNoise), "r-");
+ _plot1->Plot();
+
+ _plot2->Append("offset", _offset);
+ _plot2->Append("limitPos", _threshold/BWE_MIN(_numOfDeltas, 60));
+ _plot2->Plot();
+
+ _plot3->Append("noiseVar", _varNoise);
+ _plot3->Plot();
+#endif
+}
+
+double OverUseDetector::UpdateMinFramePeriod(double tsDelta) {
+ double minFramePeriod = tsDelta;
+ if (_tsDeltaHist.size() >= MIN_FRAME_PERIOD_HISTORY_LEN) {
+ std::list<double>::iterator firstItem = _tsDeltaHist.begin();
+ _tsDeltaHist.erase(firstItem);
+ }
+ std::list<double>::iterator it = _tsDeltaHist.begin();
+ for (; it != _tsDeltaHist.end(); it++) {
+ minFramePeriod = BWE_MIN(*it, minFramePeriod);
+ }
+ _tsDeltaHist.push_back(tsDelta);
+ return minFramePeriod;
+}
+
+void OverUseDetector::UpdateNoiseEstimate(double residual, double tsDelta, bool stableState)
+{
+ if (!stableState)
+ {
+ return;
+ }
+ // Faster filter during startup to faster adapt to the jitter level of the network
+ // alpha is tuned for 30 frames per second, but
+ double alpha = 0.01;
+ if (_numOfDeltas > 10*30)
+ {
+ alpha = 0.002;
+ }
+ // Only update the noise estimate if we're not over-using
+ // beta is a function of alpha and the time delta since
+ // the previous update.
+ const double beta = pow(1 - alpha, tsDelta * 30.0 / 1000.0);
+ _avgNoise = beta * _avgNoise + (1 - beta) * residual;
+ _varNoise = beta * _varNoise + (1 - beta) * (_avgNoise - residual) * (_avgNoise - residual);
+ if (_varNoise < 1e-7)
+ {
+ _varNoise = 1e-7;
+ }
+}
+
+BandwidthUsage OverUseDetector::Detect(double tsDelta)
+{
+ if (_numOfDeltas < 2)
+ {
+ return kBwNormal;
+ }
+ const double T = BWE_MIN(_numOfDeltas, 60) * _offset;
+ if (abs(T) > _threshold)
+ {
+ if (_offset > 0)
+ {
+ if (_timeOverUsing == -1)
+ {
+ // Initialize the timer. Assume that we've been
+ // over-using half of the time since the previous
+ // sample.
+ _timeOverUsing = tsDelta / 2;
+ }
+ else
+ {
+ // Increment timer
+ _timeOverUsing += tsDelta;
+ }
+ _overUseCounter++;
+ if (_timeOverUsing > OVER_USING_TIME_THRESHOLD && _overUseCounter > 1)
+ {
+ if (_offset >= _prevOffset)
+ {
+#ifdef _DEBUG
+ if (_hypothesis != kBwOverusing)
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwOverusing");
+#endif
+ _timeOverUsing = 0;
+ _overUseCounter = 0;
+ _hypothesis = kBwOverusing;
+#ifdef WEBRTC_BWE_MATLAB
+ _plot2->Append("detection",_offset); // plot it later
+#endif
+ }
+ }
+#ifdef WEBRTC_BWE_MATLAB
+ _plot2->Append("trigger",_offset); // plot it later
+#endif
+ }
+ else
+ {
+#ifdef _DEBUG
+ if (_hypothesis != kBwUnderUsing)
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwUnderUsing");
+#endif
+ _timeOverUsing = -1;
+ _overUseCounter = 0;
+ _hypothesis = kBwUnderUsing;
+ }
+ }
+ else
+ {
+#ifdef _DEBUG
+ if (_hypothesis != kBwNormal)
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwNormal");
+#endif
+ _timeOverUsing = -1;
+ _overUseCounter = 0;
+ _hypothesis = kBwNormal;
+ }
+ return _hypothesis;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/overuse_detector.h b/trunk/src/modules/rtp_rtcp/source/overuse_detector.h
new file mode 100644
index 0000000..b250e01
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/overuse_detector.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
+
+#include <list>
+
+#include "bwe_defines.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+
+#ifdef WEBRTC_BWE_MATLAB
+#include "../test/BWEStandAlone/MatlabPlot.h"
+#endif
+
+namespace webrtc {
+enum RateControlRegion;
+
+class OverUseDetector
+{
+public:
+ OverUseDetector();
+ ~OverUseDetector();
+ bool Update(const WebRtcRTPHeader& rtpHeader,
+ const WebRtc_UWord16 packetSize,
+ const WebRtc_Word64 nowMS);
+ BandwidthUsage State() const;
+ void Reset();
+ double NoiseVar() const;
+ void SetRateControlRegion(RateControlRegion region);
+
+private:
+ struct FrameSample
+ {
+ FrameSample() : _size(0), _completeTimeMs(-1), _timestamp(-1) {}
+
+ WebRtc_UWord32 _size;
+ WebRtc_Word64 _completeTimeMs;
+ WebRtc_Word64 _timestamp;
+ };
+
+ void CompensatedTimeDelta(const FrameSample& currentFrame,
+ const FrameSample& prevFrame,
+ WebRtc_Word64& tDelta,
+ double& tsDelta,
+ bool wrapped);
+ void UpdateKalman(WebRtc_Word64 tDelta,
+ double tsDelta,
+ WebRtc_UWord32 frameSize,
+ WebRtc_UWord32 prevFrameSize);
+ double UpdateMinFramePeriod(double tsDelta);
+ void UpdateNoiseEstimate(double residual, double tsDelta, bool stableState);
+ BandwidthUsage Detect(double tsDelta);
+ double CurrentDrift();
+
+ bool _firstPacket;
+ FrameSample _currentFrame;
+ FrameSample _prevFrame;
+ WebRtc_UWord16 _numOfDeltas;
+ double _slope;
+ double _offset;
+ double _E[2][2];
+ double _processNoise[2];
+ double _avgNoise;
+ double _varNoise;
+ double _threshold;
+ std::list<double> _tsDeltaHist;
+ double _prevOffset;
+ double _timeOverUsing;
+ WebRtc_UWord16 _overUseCounter;
+ BandwidthUsage _hypothesis;
+
+#ifdef WEBRTC_BWE_MATLAB
+ MatlabPlot* _plot1;
+ MatlabPlot* _plot2;
+ MatlabPlot* _plot3;
+ MatlabPlot* _plot4;
+#endif
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc b/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc
new file mode 100644
index 0000000..77a3465
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc
@@ -0,0 +1,262 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/receiver_fec.h"
+
+#include <cassert>
+
+#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "system_wrappers/interface/trace.h"
+
+// RFC 5109
+namespace webrtc {
+ReceiverFEC::ReceiverFEC(const WebRtc_Word32 id, RTPReceiverVideo* owner)
+ : _id(id),
+ _owner(owner),
+ _fec(new ForwardErrorCorrection(id)),
+ _payloadTypeFEC(-1) {
+}
+
+ReceiverFEC::~ReceiverFEC() {
+ // Clean up DecodeFEC()
+ while (!_receivedPacketList.empty()){
+ ForwardErrorCorrection::ReceivedPacket* receivedPacket =
+ _receivedPacketList.front();
+ delete receivedPacket;
+ _receivedPacketList.pop_front();
+ }
+ assert(_receivedPacketList.empty());
+
+ if (_fec != NULL) {
+ _fec->ResetState(&_recoveredPacketList);
+ delete _fec;
+ }
+}
+
+void ReceiverFEC::SetPayloadTypeFEC(const WebRtc_Word8 payloadType) {
+ _payloadTypeFEC = payloadType;
+}
+
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |F| block PT | timestamp offset | block length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+
+RFC 2198 RTP Payload for Redundant Audio Data September 1997
+
+ The bits in the header are specified as follows:
+
+ F: 1 bit First bit in header indicates whether another header block
+ follows. If 1 further header blocks follow, if 0 this is the
+ last header block.
+ If 0 there is only 1 byte RED header
+
+ block PT: 7 bits RTP payload type for this block.
+
+ timestamp offset: 14 bits Unsigned offset of timestamp of this block
+ relative to timestamp given in RTP header. The use of an unsigned
+ offset implies that redundant data must be sent after the primary
+ data, and is hence a time to be subtracted from the current
+ timestamp to determine the timestamp of the data for which this
+ block is the redundancy.
+
+ block length: 10 bits Length in bytes of the corresponding data
+ block excluding header.
+*/
+
+WebRtc_Word32 ReceiverFEC::AddReceivedFECPacket(
+ const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 payloadDataLength,
+ bool& FECpacket) {
+ if (_payloadTypeFEC == -1) {
+ return -1;
+ }
+
+ WebRtc_UWord8 REDHeaderLength = 1;
+
+ // Add to list without RED header, aka a virtual RTP packet
+ // we remove the RED header
+
+ ForwardErrorCorrection::ReceivedPacket* receivedPacket =
+ new ForwardErrorCorrection::ReceivedPacket;
+ receivedPacket->pkt = new ForwardErrorCorrection::Packet;
+
+ // get payload type from RED header
+ WebRtc_UWord8 payloadType =
+ incomingRtpPacket[rtpHeader->header.headerLength] & 0x7f;
+
+ // use the payloadType to decide if it's FEC or coded data
+ if (_payloadTypeFEC == payloadType) {
+ receivedPacket->isFec = true;
+ FECpacket = true;
+ } else {
+ receivedPacket->isFec = false;
+ FECpacket = false;
+ }
+ receivedPacket->seqNum = rtpHeader->header.sequenceNumber;
+
+ WebRtc_UWord16 blockLength = 0;
+ if(incomingRtpPacket[rtpHeader->header.headerLength] & 0x80) {
+ // f bit set in RED header
+ REDHeaderLength = 4;
+ WebRtc_UWord16 timestampOffset =
+ (incomingRtpPacket[rtpHeader->header.headerLength + 1]) << 8;
+ timestampOffset += incomingRtpPacket[rtpHeader->header.headerLength+2];
+ timestampOffset = timestampOffset >> 2;
+ if(timestampOffset != 0) {
+ // |timestampOffset| should be 0. However, it's possible this is the first
+ // location a corrupt payload can be caught, so don't assert.
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "Corrupt payload found in %s", __FUNCTION__);
+ delete receivedPacket;
+ return -1;
+ }
+
+ blockLength =
+ (0x03 & incomingRtpPacket[rtpHeader->header.headerLength + 2]) << 8;
+ blockLength += (incomingRtpPacket[rtpHeader->header.headerLength + 3]);
+
+ // check next RED header
+ if(incomingRtpPacket[rtpHeader->header.headerLength+4] & 0x80) {
+ // more than 2 blocks in packet not supported
+ delete receivedPacket;
+ assert(false);
+ return -1;
+ }
+ if(blockLength > payloadDataLength - REDHeaderLength) {
+ // block length longer than packet
+ delete receivedPacket;
+ assert(false);
+ return -1;
+ }
+ }
+
+ ForwardErrorCorrection::ReceivedPacket* secondReceivedPacket = NULL;
+ if (blockLength > 0) {
+ // handle block length, split into 2 packets
+ REDHeaderLength = 5;
+
+ // copy the RTP header
+ memcpy(receivedPacket->pkt->data,
+ incomingRtpPacket,
+ rtpHeader->header.headerLength);
+
+ // replace the RED payload type
+ receivedPacket->pkt->data[1] &= 0x80; // reset the payload
+ receivedPacket->pkt->data[1] += payloadType; // set the media payload type
+
+ // copy the payload data
+ memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
+ incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
+ blockLength);
+
+ receivedPacket->pkt->length = blockLength;
+
+ secondReceivedPacket = new ForwardErrorCorrection::ReceivedPacket;
+ secondReceivedPacket->pkt = new ForwardErrorCorrection::Packet;
+
+ secondReceivedPacket->isFec = true;
+ secondReceivedPacket->seqNum = rtpHeader->header.sequenceNumber;
+
+ // copy the FEC payload data
+ memcpy(secondReceivedPacket->pkt->data,
+ incomingRtpPacket + rtpHeader->header.headerLength +
+ REDHeaderLength + blockLength,
+ payloadDataLength - REDHeaderLength - blockLength);
+
+ secondReceivedPacket->pkt->length = payloadDataLength - REDHeaderLength -
+ blockLength;
+
+ } else if(receivedPacket->isFec) {
+ // everything behind the RED header
+ memcpy(receivedPacket->pkt->data,
+ incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
+ payloadDataLength - REDHeaderLength);
+ receivedPacket->pkt->length = payloadDataLength - REDHeaderLength;
+ receivedPacket->ssrc =
+ ModuleRTPUtility::BufferToUWord32(&incomingRtpPacket[8]);
+
+ } else {
+ // copy the RTP header
+ memcpy(receivedPacket->pkt->data,
+ incomingRtpPacket,
+ rtpHeader->header.headerLength);
+
+ // replace the RED payload type
+ receivedPacket->pkt->data[1] &= 0x80; // reset the payload
+ receivedPacket->pkt->data[1] += payloadType; // set the media payload type
+
+ // copy the media payload data
+ memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
+ incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
+ payloadDataLength - REDHeaderLength);
+
+ receivedPacket->pkt->length = rtpHeader->header.headerLength +
+ payloadDataLength - REDHeaderLength;
+ }
+
+ if(receivedPacket->pkt->length == 0) {
+ delete secondReceivedPacket;
+ delete receivedPacket;
+ return 0;
+ }
+
+ _receivedPacketList.push_back(receivedPacket);
+ if (secondReceivedPacket) {
+ _receivedPacketList.push_back(secondReceivedPacket);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ReceiverFEC::ProcessReceivedFEC() {
+ if (!_receivedPacketList.empty()) {
+ if (_fec->DecodeFEC(&_receivedPacketList, &_recoveredPacketList) != 0) {
+ return -1;
+ }
+ assert(_receivedPacketList.empty());
+ }
+ ForwardErrorCorrection::RecoveredPacketList::iterator it =
+ _recoveredPacketList.begin();
+ for (; it != _recoveredPacketList.end(); ++it) {
+ if ((*it)->returned) // Already sent to the VCM and the jitter buffer.
+ continue;
+ if (ParseAndReceivePacket((*it)->pkt) != 0) {
+ return -1;
+ }
+ (*it)->returned = true;
+ }
+ return 0;
+}
+
+int ReceiverFEC::ParseAndReceivePacket(
+ const ForwardErrorCorrection::Packet* packet) {
+ WebRtcRTPHeader header;
+ memset(&header, 0, sizeof(header));
+ ModuleRTPUtility::RTPHeaderParser parser(packet->data,
+ packet->length);
+ if (!parser.Parse(header)) {
+ return -1;
+ }
+ if (_owner->ReceiveRecoveredPacketCallback(
+ &header,
+ &packet->data[header.header.headerLength],
+ packet->length - header.header.headerLength) != 0) {
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec.h b/trunk/src/modules/rtp_rtcp/source/receiver_fec.h
new file mode 100644
index 0000000..63aaa72
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/receiver_fec.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
+
+#include "rtp_rtcp_defines.h"
+// This header is included to get the nested declaration of Packet structure.
+#include "forward_error_correction.h"
+
+#include "typedefs.h"
+
+namespace webrtc {
+class RTPReceiverVideo;
+
+class ReceiverFEC
+{
+public:
+ ReceiverFEC(const WebRtc_Word32 id, RTPReceiverVideo* owner);
+ virtual ~ReceiverFEC();
+
+ WebRtc_Word32 AddReceivedFECPacket(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 payloadDataLength,
+ bool& FECpacket);
+
+ WebRtc_Word32 ProcessReceivedFEC();
+
+ void SetPayloadTypeFEC(const WebRtc_Word8 payloadType);
+
+private:
+ int ParseAndReceivePacket(const ForwardErrorCorrection::Packet* packet);
+
+ int _id;
+ RTPReceiverVideo* _owner;
+ ForwardErrorCorrection* _fec;
+ // TODO(holmer): In the current version _receivedPacketList is never more
+ // than one packet, since we process FEC every time a new packet
+ // arrives. We should remove the list.
+ ForwardErrorCorrection::ReceivedPacketList _receivedPacketList;
+ ForwardErrorCorrection::RecoveredPacketList _recoveredPacketList;
+ WebRtc_Word8 _payloadTypeFEC;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc b/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
new file mode 100644
index 0000000..669018e
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
@@ -0,0 +1,523 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+#include <list>
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h"
+#include "modules/rtp_rtcp/source/receiver_fec.h"
+
+using ::testing::_;
+using ::testing::Args;
+using ::testing::ElementsAreArray;
+using ::testing::InSequence;
+
+namespace webrtc {
+
+typedef ForwardErrorCorrection::Packet Packet;
+
+enum { kRtpHeaderSize = 12 };
+enum { kFecPayloadType = 96 };
+enum { kRedPayloadType = 97 };
+enum { kVp8PayloadType = 120 };
+
+struct RtpPacket : public Packet {
+ WebRtcRTPHeader header;
+};
+
+class FrameGenerator {
+ public:
+ FrameGenerator() : num_packets_(0), seq_num_(0), timestamp_(0) {}
+
+ void NewFrame(int num_packets) {
+ num_packets_ = num_packets;
+ timestamp_ += 3000;
+ }
+
+ RtpPacket* NextPacket(int offset, size_t length) {
+ RtpPacket* rtp_packet = new RtpPacket;
+ for (size_t i = 0; i < length; ++i)
+ rtp_packet->data[i] = offset + i;
+ rtp_packet->length = length;
+ memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader));
+ rtp_packet->header.frameType = kVideoFrameDelta;
+ rtp_packet->header.header.headerLength = kRtpHeaderSize;
+ rtp_packet->header.header.markerBit = (num_packets_ == 1);
+ rtp_packet->header.header.sequenceNumber = seq_num_;
+ rtp_packet->header.header.timestamp = timestamp_;
+ rtp_packet->header.header.payloadType = kVp8PayloadType;
+ BuildRtpHeader(rtp_packet->data, rtp_packet->header.header);
+ ++seq_num_;
+ --num_packets_;
+ return rtp_packet;
+ }
+
+ // Creates a new RtpPacket with the RED header added to the packet.
+ RtpPacket* BuildMediaRedPacket(const RtpPacket* packet) {
+ const int kHeaderLength = packet->header.header.headerLength;
+ RtpPacket* red_packet = new RtpPacket;
+ red_packet->header = packet->header;
+ red_packet->length = packet->length + 1; // 1 byte RED header.
+ memset(red_packet->data, 0, red_packet->length);
+ // Copy RTP header.
+ memcpy(red_packet->data, packet->data, kHeaderLength);
+ SetRedHeader(red_packet, red_packet->data[1] & 0x7f, kHeaderLength);
+ memcpy(red_packet->data + kHeaderLength + 1, packet->data + kHeaderLength,
+ packet->length - kHeaderLength);
+ return red_packet;
+ }
+
+ // Creates a new RtpPacket with FEC payload and red header. Does this by
+ // creating a new fake media RtpPacket, clears the marker bit and adds a RED
+ // header. Finally replaces the payload with the content of |packet->data|.
+ RtpPacket* BuildFecRedPacket(const Packet* packet) {
+ // Create a fake media packet to get a correct header. 1 byte RED header.
+ ++num_packets_;
+ RtpPacket* red_packet = NextPacket(0, packet->length + 1);
+ red_packet->data[1] &= ~0x80; // Clear marker bit.
+ const int kHeaderLength = red_packet->header.header.headerLength;
+ SetRedHeader(red_packet, kFecPayloadType, kHeaderLength);
+ memcpy(red_packet->data + kHeaderLength + 1, packet->data,
+ packet->length);
+ red_packet->length = kHeaderLength + 1 + packet->length;
+ return red_packet;
+ }
+
+ void SetRedHeader(Packet* red_packet, uint8_t payload_type,
+ int header_length) const {
+ // Replace pltype.
+ red_packet->data[1] &= 0x80; // Reset.
+ red_packet->data[1] += kRedPayloadType; // Replace.
+
+ // Add RED header, f-bit always 0.
+ red_packet->data[header_length] = payload_type;
+ }
+
+ private:
+ void BuildRtpHeader(uint8_t* data, RTPHeader header) {
+ data[0] = 0x80; // Version 2.
+ data[1] = header.payloadType;
+ data[1] |= (header.markerBit ? kRtpMarkerBitMask : 0);
+ ModuleRTPUtility::AssignUWord16ToBuffer(data+2, header.sequenceNumber);
+ ModuleRTPUtility::AssignUWord32ToBuffer(data+4, header.timestamp);
+ ModuleRTPUtility::AssignUWord32ToBuffer(data+8, header.ssrc);
+ }
+
+ int num_packets_;
+ uint16_t seq_num_;
+ uint32_t timestamp_;
+};
+
+class ReceiverFecTest : public ::testing::Test {
+ protected:
+ virtual void SetUp() {
+ fec_ = new ForwardErrorCorrection(0);
+ receiver_fec_ = new ReceiverFEC(0, &rtp_receiver_video_);
+ generator_ = new FrameGenerator();
+ receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
+ }
+
+ virtual void TearDown() {
+ delete fec_;
+ delete receiver_fec_;
+ delete generator_;
+ }
+
+ void GenerateAndAddFrames(int num_frames,
+ int num_packets_per_frame,
+ std::list<RtpPacket*>* media_rtp_packets,
+ std::list<Packet*>* media_packets) {
+ for (int i = 0; i < num_frames; ++i) {
+ GenerateFrame(num_packets_per_frame, i, media_rtp_packets,
+ media_packets);
+ }
+ for (std::list<RtpPacket*>::iterator it = media_rtp_packets->begin();
+ it != media_rtp_packets->end(); ++it) {
+ BuildAndAddRedMediaPacket(*it);
+ }
+ }
+
+ void GenerateFEC(std::list<Packet*>* media_packets,
+ std::list<Packet*>* fec_packets,
+ unsigned int num_fec_packets) {
+ EXPECT_EQ(0, fec_->GenerateFEC(
+ *media_packets,
+ num_fec_packets * 255 / media_packets->size(),
+ 0,
+ false,
+ fec_packets));
+ ASSERT_EQ(num_fec_packets, fec_packets->size());
+ }
+
+ void GenerateFrame(int num_media_packets,
+ int frame_offset,
+ std::list<RtpPacket*>* media_rtp_packets,
+ std::list<Packet*>* media_packets) {
+ generator_->NewFrame(num_media_packets);
+ for (int i = 0; i < num_media_packets; ++i) {
+ media_rtp_packets->push_back(generator_->NextPacket(frame_offset + i,
+ kRtpHeaderSize + 10));
+ media_packets->push_back(media_rtp_packets->back());
+ }
+ }
+
+ void VerifyReconstructedMediaPacket(const RtpPacket* packet, int times) {
+ // Verify that the content of the reconstructed packet is equal to the
+ // content of |packet|, and that the same content is received |times| number
+ // of times in a row.
+ EXPECT_CALL(rtp_receiver_video_,
+ ReceiveRecoveredPacketCallback(_, _,
+ packet->length - kRtpHeaderSize))
+ .With(Args<1, 2>(ElementsAreArray(packet->data + kRtpHeaderSize,
+ packet->length - kRtpHeaderSize)))
+ .Times(times);
+ }
+
+ void BuildAndAddRedMediaPacket(RtpPacket* packet) {
+ RtpPacket* red_packet = generator_->BuildMediaRedPacket(packet);
+ bool is_fec = false;
+ EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
+ red_packet->data,
+ red_packet->length -
+ kRtpHeaderSize,
+ is_fec));
+ delete red_packet;
+ EXPECT_FALSE(is_fec);
+ }
+
+ void BuildAndAddRedFecPacket(Packet* packet) {
+ RtpPacket* red_packet = generator_->BuildFecRedPacket(packet);
+ bool is_fec = false;
+ EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
+ red_packet->data,
+ red_packet->length -
+ kRtpHeaderSize,
+ is_fec));
+ delete red_packet;
+ EXPECT_TRUE(is_fec);
+ }
+
+ ForwardErrorCorrection* fec_;
+ MockRTPReceiverVideo rtp_receiver_video_;
+ ReceiverFEC* receiver_fec_;
+ FrameGenerator* generator_;
+};
+
+void DeletePackets(std::list<Packet*>* packets) {
+ while (!packets->empty()) {
+ delete packets->front();
+ packets->pop_front();
+ }
+}
+
+TEST_F(ReceiverFecTest, TwoMediaOneFec) {
+ const unsigned int kNumFecPackets = 1u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
+ BuildAndAddRedMediaPacket(*media_it);
+ // Drop one media packet.
+ std::list<Packet*>::iterator fec_it = fec_packets.begin();
+ BuildAndAddRedFecPacket(*fec_it);
+ {
+ InSequence s;
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ VerifyReconstructedMediaPacket(*it, 1);
+ ++it;
+ VerifyReconstructedMediaPacket(*it, 1);
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TwoMediaTwoFec) {
+ const unsigned int kNumFecPackets = 2u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ // Drop both media packets.
+ std::list<Packet*>::iterator fec_it = fec_packets.begin();
+ BuildAndAddRedFecPacket(*fec_it);
+ ++fec_it;
+ BuildAndAddRedFecPacket(*fec_it);
+ {
+ InSequence s;
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ VerifyReconstructedMediaPacket(*it, 1);
+ ++it;
+ VerifyReconstructedMediaPacket(*it, 1);
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TwoFramesOneFec) {
+ const unsigned int kNumFecPackets = 1u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
+ GenerateFrame(1, 1, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ BuildAndAddRedMediaPacket(media_rtp_packets.front());
+ // Drop one media packet.
+ BuildAndAddRedFecPacket(fec_packets.front());
+ {
+ InSequence s;
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ VerifyReconstructedMediaPacket(*it, 1);
+ ++it;
+ VerifyReconstructedMediaPacket(*it, 1);
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, OneCompleteOneUnrecoverableFrame) {
+ const unsigned int kNumFecPackets = 1u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
+ GenerateFrame(2, 1, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ BuildAndAddRedMediaPacket(*it); // First frame
+ BuildAndAddRedMediaPacket(*it); // First packet of second frame.
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_, _, _))
+ .Times(1);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, MaxFramesOneFec) {
+ const unsigned int kNumFecPackets = 1u;
+ const unsigned int kNumMediaPackets = 48u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ for (unsigned int i = 0; i < kNumMediaPackets; ++i)
+ GenerateFrame(1, i, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ ++it; // Drop first packet.
+ for (; it != media_rtp_packets.end(); ++it)
+ BuildAndAddRedMediaPacket(*it);
+ BuildAndAddRedFecPacket(fec_packets.front());
+ {
+ InSequence s;
+ std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+ for (; it != media_rtp_packets.end(); ++it)
+ VerifyReconstructedMediaPacket(*it, 1);
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TooManyFrames) {
+ const unsigned int kNumFecPackets = 1u;
+ const unsigned int kNumMediaPackets = 49u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ for (unsigned int i = 0; i < kNumMediaPackets; ++i)
+ GenerateFrame(1, i, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ EXPECT_EQ(-1, fec_->GenerateFEC(media_packets,
+ kNumFecPackets * 255 / kNumMediaPackets,
+ 0,
+ false,
+ &fec_packets));
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, PacketNotDroppedTooEarly) {
+ // 1 frame with 2 media packets and one FEC packet. One media packet missing.
+ // Delay the FEC packet.
+ Packet* delayed_fec = NULL;
+ const unsigned int kNumFecPacketsBatch1 = 1u;
+ const unsigned int kNumMediaPacketsBatch1 = 2u;
+ std::list<RtpPacket*> media_rtp_packets_batch1;
+ std::list<Packet*> media_packets_batch1;
+ GenerateFrame(kNumMediaPacketsBatch1, 0, &media_rtp_packets_batch1,
+ &media_packets_batch1);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets_batch1, &fec_packets, kNumFecPacketsBatch1);
+
+ BuildAndAddRedMediaPacket(media_rtp_packets_batch1.front());
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(1);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+ delayed_fec = fec_packets.front();
+
+ // Fill the FEC decoder. No packets should be dropped.
+ const unsigned int kNumMediaPacketsBatch2 = 47u;
+ std::list<RtpPacket*> media_rtp_packets_batch2;
+ std::list<Packet*> media_packets_batch2;
+ GenerateAndAddFrames(kNumMediaPacketsBatch2, 1, &media_rtp_packets_batch2,
+ &media_packets_batch2);
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(media_packets_batch2.size());
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ // Add the delayed FEC packet. One packet should be reconstructed.
+ BuildAndAddRedFecPacket(delayed_fec);
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(1);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets_batch1);
+ DeletePackets(&media_packets_batch2);
+}
+
+TEST_F(ReceiverFecTest, PacketDroppedWhenTooOld) {
+ // 1 frame with 2 media packets and one FEC packet. One media packet missing.
+ // Delay the FEC packet.
+ Packet* delayed_fec = NULL;
+ const unsigned int kNumFecPacketsBatch1 = 1u;
+ const unsigned int kNumMediaPacketsBatch1 = 2u;
+ std::list<RtpPacket*> media_rtp_packets_batch1;
+ std::list<Packet*> media_packets_batch1;
+ GenerateFrame(kNumMediaPacketsBatch1, 0, &media_rtp_packets_batch1,
+ &media_packets_batch1);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets_batch1, &fec_packets, kNumFecPacketsBatch1);
+
+ BuildAndAddRedMediaPacket(media_rtp_packets_batch1.front());
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(1);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+ delayed_fec = fec_packets.front();
+
+ // Fill the FEC decoder and force the last packet to be dropped.
+ const unsigned int kNumMediaPacketsBatch2 = 48u;
+ std::list<RtpPacket*> media_rtp_packets_batch2;
+ std::list<Packet*> media_packets_batch2;
+ GenerateAndAddFrames(kNumMediaPacketsBatch2, 1, &media_rtp_packets_batch2,
+ &media_packets_batch2);
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(media_packets_batch2.size());
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ // Add the delayed FEC packet. No packet should be reconstructed since the
+ // first media packet of that frame has been dropped due to being too old.
+ BuildAndAddRedFecPacket(delayed_fec);
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(0);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets_batch1);
+ DeletePackets(&media_packets_batch2);
+}
+
+TEST_F(ReceiverFecTest, OldFecPacketDropped) {
+ // 49 frames with 2 media packets and one FEC packet. All media packets
+ // missing.
+ const unsigned int kNumMediaPackets = 49 * 2;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ for (unsigned int i = 0; i < kNumMediaPackets / 2; ++i) {
+ std::list<RtpPacket*> frame_media_rtp_packets;
+ std::list<Packet*> frame_media_packets;
+ std::list<Packet*> fec_packets;
+ GenerateFrame(2, 0, &frame_media_rtp_packets, &frame_media_packets);
+ GenerateFEC(&frame_media_packets, &fec_packets, 1);
+ for (std::list<Packet*>::iterator it = fec_packets.begin();
+ it != fec_packets.end(); ++it) {
+ BuildAndAddRedFecPacket(*it);
+ }
+ media_packets.insert(media_packets.end(),
+ frame_media_packets.begin(),
+ frame_media_packets.end());
+ media_rtp_packets.insert(media_rtp_packets.end(),
+ frame_media_rtp_packets.begin(),
+ frame_media_rtp_packets.end());
+ }
+ // Don't insert any media packets.
+ // Only FEC packets inserted. No packets should be recoverable at this time.
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(0);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ // Insert the oldest media packet. The corresponding FEC packet is too old
+ // and should've been dropped. Only the media packet we inserted will be
+ // returned.
+ BuildAndAddRedMediaPacket(media_rtp_packets.front());
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(1);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, PacketsOnlyReturnedOnce) {
+ const unsigned int kNumFecPackets = 1u;
+ std::list<RtpPacket*> media_rtp_packets;
+ std::list<Packet*> media_packets;
+ GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
+ GenerateFrame(2, 1, &media_rtp_packets, &media_packets);
+ std::list<Packet*> fec_packets;
+ GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+
+ // Recovery
+ std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
+ BuildAndAddRedMediaPacket(*media_it); // First frame.
+ {
+ std::list<RtpPacket*>::iterator verify_it = media_rtp_packets.begin();
+ VerifyReconstructedMediaPacket(*verify_it, 1); // First frame
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ ++media_it;
+ BuildAndAddRedMediaPacket(*media_it); // 1st packet of 2nd frame.
+ BuildAndAddRedFecPacket(fec_packets.front()); // Insert FEC packet.
+ {
+ InSequence s;
+ std::list<RtpPacket*>::iterator verify_it = media_rtp_packets.begin();
+ ++verify_it; // First frame has already been returned.
+ VerifyReconstructedMediaPacket(*verify_it, 1); // 1st packet of 2nd frame.
+ ++verify_it;
+ VerifyReconstructedMediaPacket(*verify_it, 1); // 2nd packet of 2nd frame.
+ }
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ ++media_it;
+ BuildAndAddRedMediaPacket(*media_it); // 2nd packet of 2nd frame.
+ EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
+ .Times(0);
+ EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
+
+ DeletePackets(&media_packets);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc
new file mode 100644
index 0000000..fe8477c
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc
@@ -0,0 +1,483 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if _WIN32
+#include <windows.h>
+#endif
+
+#include "remote_rate_control.h"
+#include "trace.h"
+#include <math.h>
+#include <string.h>
+
+#ifdef MATLAB
+extern MatlabEngine eng; // global variable defined elsewhere
+#endif
+
+namespace webrtc {
+RemoteRateControl::RemoteRateControl()
+:
+_minConfiguredBitRate(30000),
+_maxConfiguredBitRate(30000000),
+_currentBitRate(_maxConfiguredBitRate),
+_maxHoldRate(0),
+_avgMaxBitRate(-1.0f),
+_varMaxBitRate(0.4f),
+_rcState(kRcHold),
+_cameFromState(kRcDecrease),
+_rcRegion(kRcMaxUnknown),
+_lastBitRateChange(-1),
+_currentInput(kBwNormal, 0, 1.0),
+_updated(false),
+_timeFirstIncomingEstimate(-1),
+_initializedBitRate(false),
+_avgChangePeriod(1000.0f),
+_lastChangeMs(-1),
+_beta(0.9f)
+#ifdef MATLAB
+,_plot1(NULL),
+_plot2(NULL)
+#endif
+{
+}
+
+RemoteRateControl::~RemoteRateControl()
+{
+#ifdef MATLAB
+ eng.DeletePlot(_plot1);
+ eng.DeletePlot(_plot2);
+#endif
+}
+
+void RemoteRateControl::Reset()
+{
+ _minConfiguredBitRate = 30000;
+ _maxConfiguredBitRate = 30000000;
+ _currentBitRate = _maxConfiguredBitRate;
+ _maxHoldRate = 0;
+ _avgMaxBitRate = -1.0f;
+ _varMaxBitRate = 0.4f;
+ _rcState = kRcHold;
+ _cameFromState = kRcHold;
+ _rcRegion = kRcMaxUnknown;
+ _lastBitRateChange = -1;
+ _avgChangePeriod = 1000.0f;
+ _lastChangeMs = -1;
+ _beta = 0.9f;
+ _currentInput._bwState = kBwNormal;
+ _currentInput._incomingBitRate = 0;
+ _currentInput._noiseVar = 1.0;
+ _updated = false;
+ _timeFirstIncomingEstimate = -1;
+ _initializedBitRate = false;
+}
+
+bool RemoteRateControl::ValidEstimate() const {
+ return _initializedBitRate;
+}
+
+WebRtc_Word32 RemoteRateControl::SetConfiguredBitRates(WebRtc_UWord32 minBitRateBps, WebRtc_UWord32 maxBitRateBps)
+{
+ if (minBitRateBps > maxBitRateBps)
+ {
+ return -1;
+ }
+ _minConfiguredBitRate = minBitRateBps;
+ _maxConfiguredBitRate = maxBitRateBps;
+ _currentBitRate = BWE_MIN(BWE_MAX(minBitRateBps, _currentBitRate), maxBitRateBps);
+ return 0;
+}
+
+WebRtc_UWord32 RemoteRateControl::LatestEstimate() const {
+ return _currentBitRate;
+}
+
+WebRtc_UWord32 RemoteRateControl::UpdateBandwidthEstimate(WebRtc_UWord32 RTT,
+ WebRtc_Word64 nowMS)
+{
+ _currentBitRate = ChangeBitRate(_currentBitRate, _currentInput._incomingBitRate,
+ _currentInput._noiseVar, RTT, nowMS);
+ return _currentBitRate;
+}
+
+RateControlRegion RemoteRateControl::Update(const RateControlInput& input,
+ bool& firstOverUse,
+ WebRtc_Word64 nowMS)
+{
+#ifdef MATLAB
+ // Create plots
+ if (_plot1 == NULL)
+ {
+ _plot1 = eng.NewPlot(new MatlabPlot());
+
+ _plot1->AddTimeLine(30, "b", "current");
+ _plot1->AddTimeLine(30, "r-", "avgMax");
+ _plot1->AddTimeLine(30, "r--", "pStdMax");
+ _plot1->AddTimeLine(30, "r--", "nStdMax");
+ _plot1->AddTimeLine(30, "r+", "max");
+ _plot1->AddTimeLine(30, "g", "incoming");
+ _plot1->AddTimeLine(30, "b+", "recovery");
+ }
+ if (_plot2 == NULL)
+ {
+ _plot2 = eng.NewPlot(new MatlabPlot());
+
+ _plot2->AddTimeLine(30, "b", "alpha");
+ }
+#endif
+
+ firstOverUse = (_currentInput._bwState != kBwOverusing &&
+ input._bwState == kBwOverusing);
+
+ // Set the initial bit rate value to what we're receiving the first second
+ if (!_initializedBitRate)
+ {
+ if (_timeFirstIncomingEstimate < 0)
+ {
+ if (input._incomingBitRate > 0)
+ {
+ _timeFirstIncomingEstimate = nowMS;
+ }
+ }
+ else if (nowMS - _timeFirstIncomingEstimate > 1000 &&
+ input._incomingBitRate > 0)
+ {
+ _currentBitRate = input._incomingBitRate;
+ _initializedBitRate = true;
+ }
+ }
+
+ if (_updated && _currentInput._bwState == kBwOverusing)
+ {
+ // Only update delay factor and incoming bit rate. We always want to react on an over-use.
+ _currentInput._noiseVar = input._noiseVar;
+ _currentInput._incomingBitRate = input._incomingBitRate;
+ return _rcRegion;
+ }
+ _updated = true;
+ _currentInput = input;
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Incoming rate = %u kbps", input._incomingBitRate/1000);
+ return _rcRegion;
+}
+
+WebRtc_UWord32 RemoteRateControl::ChangeBitRate(WebRtc_UWord32 currentBitRate,
+ WebRtc_UWord32 incomingBitRate,
+ double noiseVar,
+ WebRtc_UWord32 RTT,
+ WebRtc_Word64 nowMS)
+{
+ if (!_updated)
+ {
+ return _currentBitRate;
+ }
+ _updated = false;
+ UpdateChangePeriod(nowMS);
+ ChangeState(_currentInput, nowMS);
+ // calculated here because it's used in multiple places
+ const float incomingBitRateKbps = incomingBitRate / 1000.0f;
+ // Calculate the max bit rate std dev given the normalized
+ // variance and the current incoming bit rate.
+ const float stdMaxBitRate = sqrt(_varMaxBitRate * _avgMaxBitRate);
+ bool recovery = false;
+ switch (_rcState)
+ {
+ case kRcHold:
+ {
+ _maxHoldRate = BWE_MAX(_maxHoldRate, incomingBitRate);
+ break;
+ }
+ case kRcIncrease:
+ {
+ if (_avgMaxBitRate >= 0)
+ {
+ if (incomingBitRateKbps > _avgMaxBitRate + 3 * stdMaxBitRate)
+ {
+ ChangeRegion(kRcMaxUnknown);
+ _avgMaxBitRate = -1.0;
+ }
+ else if (incomingBitRateKbps > _avgMaxBitRate + 2.5 * stdMaxBitRate)
+ {
+ ChangeRegion(kRcAboveMax);
+ }
+ }
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "BWE: Response time: %f + %i + 10*33\n",
+ _avgChangePeriod, RTT);
+ const WebRtc_UWord32 responseTime = static_cast<WebRtc_UWord32>(_avgChangePeriod + 0.5f) + RTT + 300;
+ double alpha = RateIncreaseFactor(nowMS, _lastBitRateChange,
+ responseTime, noiseVar);
+
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "BWE: _avgChangePeriod = %f ms; RTT = %u ms", _avgChangePeriod, RTT);
+
+ currentBitRate = static_cast<WebRtc_UWord32>(currentBitRate * alpha) + 1000;
+ if (_maxHoldRate > 0 && _beta * _maxHoldRate > currentBitRate)
+ {
+ currentBitRate = static_cast<WebRtc_UWord32>(_beta * _maxHoldRate);
+ _avgMaxBitRate = _beta * _maxHoldRate / 1000.0f;
+ ChangeRegion(kRcNearMax);
+ recovery = true;
+#ifdef MATLAB
+ _plot1->Append("recovery", _maxHoldRate/1000);
+#endif
+ }
+ _maxHoldRate = 0;
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "BWE: Increase rate to currentBitRate = %u kbps", currentBitRate/1000);
+ _lastBitRateChange = nowMS;
+ break;
+ }
+ case kRcDecrease:
+ {
+ if (incomingBitRate < _minConfiguredBitRate)
+ {
+ currentBitRate = _minConfiguredBitRate;
+ }
+ else
+ {
+ // Set bit rate to something slightly lower than max
+ // to get rid of any self-induced delay.
+ currentBitRate = static_cast<WebRtc_UWord32>(_beta * incomingBitRate + 0.5);
+ if (currentBitRate > _currentBitRate)
+ {
+ // Avoid increasing the rate when over-using.
+ if (_rcRegion != kRcMaxUnknown)
+ {
+ currentBitRate = static_cast<WebRtc_UWord32>(_beta * _avgMaxBitRate * 1000 + 0.5f);
+ }
+ currentBitRate = BWE_MIN(currentBitRate, _currentBitRate);
+ }
+ ChangeRegion(kRcNearMax);
+
+ if (incomingBitRateKbps < _avgMaxBitRate - 3 * stdMaxBitRate)
+ {
+ _avgMaxBitRate = -1.0f;
+ }
+
+ UpdateMaxBitRateEstimate(incomingBitRateKbps);
+
+#ifdef MATLAB
+ _plot1->Append("max", incomingBitRateKbps);
+#endif
+
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Decrease rate to currentBitRate = %u kbps", currentBitRate/1000);
+ }
+ // Stay on hold until the pipes are cleared.
+ ChangeState(kRcHold);
+ _lastBitRateChange = nowMS;
+ break;
+ }
+ }
+ if (!recovery && (incomingBitRate > 100000 || currentBitRate > 150000) &&
+ currentBitRate > 1.5 * incomingBitRate)
+ {
+ // Allow changing the bit rate if we are operating at very low rates
+ // Don't change the bit rate if the send side is too far off
+ currentBitRate = _currentBitRate;
+ _lastBitRateChange = nowMS;
+ }
+#ifdef MATLAB
+ if (_avgMaxBitRate >= 0.0f)
+ {
+ _plot1->Append("avgMax", _avgMaxBitRate);
+ _plot1->Append("pStdMax", _avgMaxBitRate + 3*stdMaxBitRate);
+ _plot1->Append("nStdMax", _avgMaxBitRate - 3*stdMaxBitRate);
+ }
+ _plot1->Append("incoming", incomingBitRate/1000);
+ _plot1->Append("current", currentBitRate/1000);
+ _plot1->Plot();
+#endif
+ return currentBitRate;
+}
+
+double RemoteRateControl::RateIncreaseFactor(WebRtc_Word64 nowMs, WebRtc_Word64 lastMs, WebRtc_UWord32 reactionTimeMs, double noiseVar) const
+{
+ // alpha = 1.02 + B ./ (1 + exp(b*(tr - (c1*s2 + c2))))
+ // Parameters
+ const double B = 0.0407;
+ const double b = 0.0025;
+ const double c1 = -6700.0 / (33 * 33);
+ const double c2 = 800.0;
+ const double d = 0.85;
+
+ double alpha = 1.005 + B / (1 + exp( b * (d * reactionTimeMs - (c1 * noiseVar + c2))));
+
+ if (alpha < 1.005)
+ {
+ alpha = 1.005;
+ }
+ else if (alpha > 1.3)
+ {
+ alpha = 1.3;
+ }
+
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "BWE: alpha = %f", alpha);
+#ifdef MATLAB
+ _plot2->Append("alpha", alpha);
+ _plot2->Plot();
+#endif
+
+ if (lastMs > -1)
+ {
+ alpha = pow(alpha, (nowMs - lastMs) / 1000.0);
+ }
+
+ if (_rcRegion == kRcNearMax)
+ {
+ // We're close to our previous maximum. Try to stabilize the
+ // bit rate in this region, by increasing in smaller steps.
+ alpha = alpha - (alpha - 1.0) / 2.0;
+ }
+ else if (_rcRegion == kRcMaxUnknown)
+ {
+ alpha = alpha + (alpha - 1.0) * 2.0;
+ }
+
+ return alpha;
+}
+
+void RemoteRateControl::UpdateChangePeriod(WebRtc_Word64 nowMs)
+{
+ WebRtc_Word64 changePeriod = 0;
+ if (_lastChangeMs > -1)
+ {
+ changePeriod = nowMs - _lastChangeMs;
+ }
+ _lastChangeMs = nowMs;
+ _avgChangePeriod = 0.9f * _avgChangePeriod + 0.1f * changePeriod;
+}
+
+void RemoteRateControl::UpdateMaxBitRateEstimate(float incomingBitRateKbps)
+{
+ const float alpha = 0.05f;
+ if (_avgMaxBitRate == -1.0f)
+ {
+ _avgMaxBitRate = incomingBitRateKbps;
+ }
+ else
+ {
+ _avgMaxBitRate = (1 - alpha) * _avgMaxBitRate +
+ alpha * incomingBitRateKbps;
+ }
+ // Estimate the max bit rate variance and normalize the variance
+ // with the average max bit rate.
+ const float norm = BWE_MAX(_avgMaxBitRate, 1.0f);
+ _varMaxBitRate = (1 - alpha) * _varMaxBitRate +
+ alpha * (_avgMaxBitRate - incomingBitRateKbps) *
+ (_avgMaxBitRate - incomingBitRateKbps) /
+ norm;
+ // 0.4 ~= 14 kbit/s at 500 kbit/s
+ if (_varMaxBitRate < 0.4f)
+ {
+ _varMaxBitRate = 0.4f;
+ }
+ // 2.5f ~= 35 kbit/s at 500 kbit/s
+ if (_varMaxBitRate > 2.5f)
+ {
+ _varMaxBitRate = 2.5f;
+ }
+}
+
+void RemoteRateControl::ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs)
+{
+ switch (_currentInput._bwState)
+ {
+ case kBwNormal:
+ {
+ if (_rcState == kRcHold)
+ {
+ _lastBitRateChange = nowMs;
+ ChangeState(kRcIncrease);
+ }
+ break;
+ }
+ case kBwOverusing:
+ {
+ if (_rcState != kRcDecrease)
+ {
+ ChangeState(kRcDecrease);
+ }
+ break;
+ }
+ case kBwUnderUsing:
+ {
+ ChangeState(kRcHold);
+ break;
+ }
+ }
+}
+
+void RemoteRateControl::ChangeRegion(RateControlRegion region)
+{
+ _rcRegion = region;
+ switch (_rcRegion)
+ {
+ case kRcAboveMax:
+ case kRcMaxUnknown:
+ {
+ _beta = 0.9f;
+ break;
+ }
+ case kRcNearMax:
+ {
+ _beta = 0.95f;
+ break;
+ }
+ }
+}
+
+void RemoteRateControl::ChangeState(RateControlState newState)
+{
+ _cameFromState = _rcState;
+ _rcState = newState;
+ char state1[15];
+ char state2[15];
+ char state3[15];
+ StateStr(_cameFromState, state1);
+ StateStr(_rcState, state2);
+ StateStr(_currentInput._bwState, state3);
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "\t%s => %s due to %s\n", state1, state2, state3);
+}
+
+void RemoteRateControl::StateStr(RateControlState state, char* str)
+{
+ switch (state)
+ {
+ case kRcDecrease:
+ strncpy(str, "DECREASE", 9);
+ break;
+ case kRcHold:
+ strncpy(str, "HOLD", 5);
+ break;
+ case kRcIncrease:
+ strncpy(str, "INCREASE", 9);
+ break;
+ }
+}
+
+void RemoteRateControl::StateStr(BandwidthUsage state, char* str)
+{
+ switch (state)
+ {
+ case kBwNormal:
+ strncpy(str, "NORMAL", 7);
+ break;
+ case kBwOverusing:
+ strncpy(str, "OVER USING", 11);
+ break;
+ case kBwUnderUsing:
+ strncpy(str, "UNDER USING", 12);
+ break;
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h
new file mode 100644
index 0000000..197bf22
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
+
+#include "bwe_defines.h"
+#include "typedefs.h"
+
+#ifdef MATLAB
+#include "../test/BWEStandAlone/MatlabPlot.h"
+#endif
+
+namespace webrtc {
+class RemoteRateControl
+{
+public:
+ RemoteRateControl();
+ ~RemoteRateControl();
+ WebRtc_Word32 SetConfiguredBitRates(WebRtc_UWord32 minBitRate,
+ WebRtc_UWord32 maxBitRate);
+ WebRtc_UWord32 LatestEstimate() const;
+ WebRtc_UWord32 UpdateBandwidthEstimate(WebRtc_UWord32 RTT,
+ WebRtc_Word64 nowMS);
+ RateControlRegion Update(const RateControlInput& input, bool& firstOverUse,
+ WebRtc_Word64 nowMS);
+ void Reset();
+
+ // Returns true if there is a valid estimate of the incoming bitrate, false
+ // otherwise.
+ bool ValidEstimate() const;
+
+private:
+ WebRtc_UWord32 ChangeBitRate(WebRtc_UWord32 currentBitRate,
+ WebRtc_UWord32 incomingBitRate,
+ double delayFactor, WebRtc_UWord32 RTT,
+ WebRtc_Word64 nowMS);
+ double RateIncreaseFactor(WebRtc_Word64 nowMs,
+ WebRtc_Word64 lastMs,
+ WebRtc_UWord32 reactionTimeMs,
+ double noiseVar) const;
+ void UpdateChangePeriod(WebRtc_Word64 nowMs);
+ void UpdateMaxBitRateEstimate(float incomingBitRateKbps);
+ void ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs);
+ void ChangeState(RateControlState newState);
+ void ChangeRegion(RateControlRegion region);
+ static void StateStr(RateControlState state, char* str);
+ static void StateStr(BandwidthUsage state, char* str);
+
+ WebRtc_UWord32 _minConfiguredBitRate;
+ WebRtc_UWord32 _maxConfiguredBitRate;
+ WebRtc_UWord32 _currentBitRate;
+ WebRtc_UWord32 _maxHoldRate;
+ float _avgMaxBitRate;
+ float _varMaxBitRate;
+ RateControlState _rcState;
+ RateControlState _cameFromState;
+ RateControlRegion _rcRegion;
+ WebRtc_Word64 _lastBitRateChange;
+ RateControlInput _currentInput;
+ bool _updated;
+ WebRtc_Word64 _timeFirstIncomingEstimate;
+ bool _initializedBitRate;
+
+ float _avgChangePeriod;
+ WebRtc_Word64 _lastChangeMs;
+ float _beta;
+#ifdef MATLAB
+ MatlabPlot *_plot1;
+ MatlabPlot *_plot2;
+#endif
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
new file mode 100644
index 0000000..9fdd6ca
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <gtest/gtest.h>
+
+#include "typedefs.h"
+#include "common_types.h"
+#include "rtp_utility.h"
+#include "rtcp_sender.h"
+#include "rtcp_receiver.h"
+#include "rtp_rtcp_impl.h"
+#include "bwe_defines.h"
+
+namespace {
+
+using namespace webrtc;
+
+
+class TestTransport : public Transport {
+ public:
+ TestTransport(RTCPReceiver* rtcp_receiver) :
+ rtcp_receiver_(rtcp_receiver) {
+ }
+
+ virtual int SendPacket(int /*channel*/, const void* /*data*/, int /*len*/) {
+ return -1;
+ }
+ virtual int SendRTCPPacket(int /*channel*/,
+ const void *packet,
+ int packetLength) {
+ RTCPUtility::RTCPParserV2 rtcpParser((WebRtc_UWord8*)packet,
+ (WebRtc_Word32)packetLength,
+ true); // Allow non-compound RTCP
+
+ EXPECT_TRUE(rtcpParser.IsValid());
+ RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
+ EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
+ &rtcpParser));
+
+ EXPECT_EQ((WebRtc_UWord32)kRtcpRemb,
+ rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb);
+ EXPECT_EQ((WebRtc_UWord32)1234,
+ rtcpPacketInformation.receiverEstimatedMaxBitrate);
+ return packetLength;
+ }
+ private:
+ RTCPReceiver* rtcp_receiver_;
+};
+
+
+class RtcpFormatRembTest : public ::testing::Test {
+ protected:
+ RtcpFormatRembTest() {};
+ virtual void SetUp();
+ virtual void TearDown();
+
+ RtpRtcpClock* system_clock_;
+ ModuleRtpRtcpImpl* dummy_rtp_rtcp_impl_;
+ RTCPSender* rtcp_sender_;
+ RTCPReceiver* rtcp_receiver_;
+ TestTransport* test_transport_;
+};
+
+void RtcpFormatRembTest::SetUp() {
+ system_clock_ = ModuleRTPUtility::GetSystemClock();
+ dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
+ rtcp_sender_ = new RTCPSender(0, false, system_clock_, dummy_rtp_rtcp_impl_);
+ rtcp_receiver_ = new RTCPReceiver(0, system_clock_, dummy_rtp_rtcp_impl_);
+ test_transport_ = new TestTransport(rtcp_receiver_);
+
+ EXPECT_EQ(0, rtcp_sender_->Init());
+ EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
+}
+
+void RtcpFormatRembTest::TearDown() {
+ delete rtcp_sender_;
+ delete rtcp_receiver_;
+ delete dummy_rtp_rtcp_impl_;
+ delete test_transport_;
+ delete system_clock_;
+}
+
+TEST_F(RtcpFormatRembTest, TestBasicAPI) {
+ EXPECT_FALSE(rtcp_sender_->REMB());
+ EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(true));
+ EXPECT_TRUE(rtcp_sender_->REMB());
+ EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(false));
+ EXPECT_FALSE(rtcp_sender_->REMB());
+
+ EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 0, NULL));
+}
+
+TEST_F(RtcpFormatRembTest, TestNonCompund) {
+ WebRtc_UWord32 SSRC = 456789;
+ EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpNonCompound));
+ EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC));
+ EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRemb));
+}
+
+TEST_F(RtcpFormatRembTest, TestCompund) {
+ WebRtc_UWord32 SSRCs[2] = {456789, 98765};
+ EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 2, SSRCs));
+ EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRemb));
+}
+
+int main(int argc, char** argv) {
+ ::testing::InitGoogleTest(&argc, argv);
+
+ return RUN_ALL_TESTS();
+}
+
+} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc
new file mode 100644
index 0000000..f4eca71
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -0,0 +1,1500 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtcp_receiver.h"
+
+#include <string.h> //memset
+#include <cassert> //assert
+
+#include "trace.h"
+#include "critical_section_wrapper.h"
+#include "rtcp_utility.h"
+#include "rtp_rtcp_impl.h"
+
+namespace
+{
+ const float FRAC = 4.294967296E9;
+}
+
+namespace webrtc {
+using namespace RTCPUtility;
+using namespace RTCPHelp;
+
+RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id,
+ RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner) :
+ _id(id),
+ _clock(*clock),
+ _method(kRtcpOff),
+ _lastReceived(0),
+ _rtpRtcp(*owner),
+ _criticalSectionFeedbacks(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbRtcpFeedback(NULL),
+ _cbVideoFeedback(NULL),
+ _criticalSectionRTCPReceiver(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _SSRC(0),
+ _remoteSSRC(0),
+ _remoteSenderInfo(),
+ _lastReceivedSRNTPsecs(0),
+ _lastReceivedSRNTPfrac(0),
+ _receivedInfoMap(),
+ _packetTimeOutMS(0),
+ _rtt(0)
+{
+ memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+}
+
+RTCPReceiver::~RTCPReceiver() {
+ delete _criticalSectionRTCPReceiver;
+ delete _criticalSectionFeedbacks;
+
+ while (!_receivedReportBlockMap.empty()) {
+ std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator first =
+ _receivedReportBlockMap.begin();
+ delete first->second;
+ _receivedReportBlockMap.erase(first);
+ }
+ while (!_receivedInfoMap.empty()) {
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator first =
+ _receivedInfoMap.begin();
+ delete first->second;
+ _receivedInfoMap.erase(first);
+ }
+ while (!_receivedCnameMap.empty()) {
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator first =
+ _receivedCnameMap.begin();
+ delete first->second;
+ _receivedCnameMap.erase(first);
+ }
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id,
+ "%s deleted", __FUNCTION__);
+}
+
+void
+RTCPReceiver::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+RTCPMethod
+RTCPReceiver::Status() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ return _method;
+}
+
+WebRtc_Word32
+RTCPReceiver::SetRTCPStatus(const RTCPMethod method)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ _method = method;
+ return 0;
+}
+
+WebRtc_UWord32
+RTCPReceiver::LastReceived()
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ return _lastReceived;
+}
+
+WebRtc_Word32
+RTCPReceiver::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ // new SSRC reset old reports
+ memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
+ _lastReceivedSRNTPsecs = 0;
+ _lastReceivedSRNTPfrac = 0;
+
+ _remoteSSRC = ssrc;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPReceiver::RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+ _cbRtcpFeedback = incomingMessagesCallback;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPReceiver::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+ _cbVideoFeedback = incomingMessagesCallback;
+ return 0;
+}
+
+void
+RTCPReceiver::SetSSRC( const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ _SSRC = ssrc;
+}
+
+WebRtc_Word32 RTCPReceiver::ResetRTT(const WebRtc_UWord32 remoteSSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ RTCPReportBlockInformation* reportBlock =
+ GetReportBlockInformation(remoteSSRC);
+ if (reportBlock == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "\tfailed to GetReportBlockInformation(%u)", remoteSSRC);
+ return -1;
+ }
+ reportBlock->RTT = 0;
+ reportBlock->avgRTT = 0;
+ reportBlock->minRTT = 0;
+ reportBlock->maxRTT = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 RTCPReceiver::RTT(const WebRtc_UWord32 remoteSSRC,
+ WebRtc_UWord16* RTT,
+ WebRtc_UWord16* avgRTT,
+ WebRtc_UWord16* minRTT,
+ WebRtc_UWord16* maxRTT) const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ RTCPReportBlockInformation* reportBlock =
+ GetReportBlockInformation(remoteSSRC);
+
+ if (reportBlock == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "\tfailed to GetReportBlockInformation(%u)",
+ remoteSSRC);
+ return -1;
+ }
+ if (RTT) {
+ *RTT = reportBlock->RTT;
+ }
+ if (avgRTT) {
+ *avgRTT = reportBlock->avgRTT;
+ }
+ if (minRTT) {
+ *minRTT = reportBlock->minRTT;
+ }
+ if (maxRTT) {
+ *maxRTT = reportBlock->maxRTT;
+ }
+ return 0;
+}
+
+WebRtc_UWord16 RTCPReceiver::RTT() const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if (!_receivedReportBlockMap.empty()) {
+ return 0;
+ }
+ return _rtt;
+}
+
+int RTCPReceiver::SetRTT(WebRtc_UWord16 rtt) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if (!_receivedReportBlockMap.empty()) {
+ return -1;
+ }
+ _rtt = rtt;
+ return 0;
+}
+
+void
+RTCPReceiver::UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+ if(_cbRtcpFeedback)
+ {
+ _cbRtcpFeedback->OnLipSyncUpdate(_id,audioVideoOffset);
+ }
+};
+
+WebRtc_Word32
+RTCPReceiver::NTP(WebRtc_UWord32 *ReceivedNTPsecs,
+ WebRtc_UWord32 *ReceivedNTPfrac,
+ WebRtc_UWord32 *RTCPArrivalTimeSecs,
+ WebRtc_UWord32 *RTCPArrivalTimeFrac) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if(ReceivedNTPsecs)
+ {
+ *ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport
+ }
+ if(ReceivedNTPfrac)
+ {
+ *ReceivedNTPfrac = _remoteSenderInfo.NTPfraction;
+ }
+ if(RTCPArrivalTimeFrac)
+ {
+ *RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block
+ }
+ if(RTCPArrivalTimeSecs)
+ {
+ *RTCPArrivalTimeSecs = _lastReceivedSRNTPsecs;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const
+{
+ if(senderInfo == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if(_lastReceivedSRNTPsecs == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__);
+ return -1;
+ }
+ memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
+ return 0;
+}
+
+// statistics
+// we can get multiple receive reports when we receive the report from a CE
+WebRtc_Word32 RTCPReceiver::StatisticsReceived(
+ std::vector<RTCPReportBlock>* receiveBlocks) const {
+ assert(receiveBlocks);
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
+ _receivedReportBlockMap.begin();
+
+ while (it != _receivedReportBlockMap.end()) {
+ receiveBlocks->push_back(it->second->remoteReceiveBlock);
+ it++;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
+ RTCPUtility::RTCPParserV2* rtcpParser)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ _lastReceived = _clock.GetTimeInMS();
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
+ while (pktType != RTCPUtility::kRtcpNotValidCode)
+ {
+ // Each "case" is responsible for iterate the parser to the
+ // next top level packet.
+ switch (pktType)
+ {
+ case RTCPUtility::kRtcpSrCode:
+ case RTCPUtility::kRtcpRrCode:
+ HandleSenderReceiverReport(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpSdesCode:
+ HandleSDES(*rtcpParser);
+ break;
+ case RTCPUtility::kRtcpXrVoipMetricCode:
+ HandleXRVOIPMetric(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpByeCode:
+ HandleBYE(*rtcpParser);
+ break;
+ case RTCPUtility::kRtcpRtpfbNackCode:
+ HandleNACK(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpRtpfbTmmbrCode:
+ HandleTMMBR(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpRtpfbTmmbnCode:
+ HandleTMMBN(*rtcpParser);
+ break;
+ case RTCPUtility::kRtcpRtpfbSrReqCode:
+ HandleSR_REQ(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpPsfbPliCode:
+ HandlePLI(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpPsfbSliCode:
+ HandleSLI(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpPsfbRpsiCode:
+ HandleRPSI(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpExtendedIjCode:
+ HandleIJ(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpPsfbFirCode:
+ HandleFIR(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpPsfbAppCode:
+ HandlePsfbApp(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpAppCode:
+ // generic application messages
+ HandleAPP(*rtcpParser, rtcpPacketInformation);
+ break;
+ case RTCPUtility::kRtcpAppItemCode:
+ // generic application messages
+ HandleAPPItem(*rtcpParser, rtcpPacketInformation);
+ break;
+ default:
+ rtcpParser->Iterate();
+ break;
+ }
+ pktType = rtcpParser->PacketType();
+ }
+ return 0;
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ RTCPUtility::RTCPPacketTypes rtcpPacketType = rtcpParser.PacketType();
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ assert((rtcpPacketType == RTCPUtility::kRtcpRrCode) || (rtcpPacketType == RTCPUtility::kRtcpSrCode));
+
+ // SR.SenderSSRC
+ // The synchronization source identifier for the originator of this SR packet
+
+ // rtcpPacket.RR.SenderSSRC
+ // The source of the packet sender, same as of SR? or is this a CE?
+
+ const WebRtc_UWord32 remoteSSRC = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.SenderSSRC:rtcpPacket.SR.SenderSSRC;
+ const WebRtc_UWord8 numberOfReportBlocks = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.NumberOfReportBlocks:rtcpPacket.SR.NumberOfReportBlocks;
+
+ rtcpPacketInformation.remoteSSRC = remoteSSRC;
+
+ RTCPReceiveInformation* ptrReceiveInfo = CreateReceiveInformation(remoteSSRC);
+ if (!ptrReceiveInfo)
+ {
+ rtcpParser.Iterate();
+ return;
+ }
+
+ if (rtcpPacketType == RTCPUtility::kRtcpSrCode)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
+ "Received SR(%d). SSRC:0x%x, from SSRC:0x%x, to us %d.", _id, _SSRC, remoteSSRC, (_remoteSSRC == remoteSSRC)?1:0);
+
+ if (_remoteSSRC == remoteSSRC) // have I received RTP packets from this party
+ {
+ // only signal that we have received a SR when we accept one
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSr;
+
+ // We will only store the send report from one source, but
+ // we will store all the receive block
+
+ // Save the NTP time of this report
+ _remoteSenderInfo.NTPseconds = rtcpPacket.SR.NTPMostSignificant;
+ _remoteSenderInfo.NTPfraction = rtcpPacket.SR.NTPLeastSignificant;
+ _remoteSenderInfo.RTPtimeStamp = rtcpPacket.SR.RTPTimestamp;
+ _remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount;
+ _remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount;
+
+ _clock.CurrentNTP(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
+ }
+ else
+ {
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
+ }
+ } else
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
+ "Received RR(%d). SSRC:0x%x, from SSRC:0x%x", _id, _SSRC, remoteSSRC);
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
+ }
+ UpdateReceiveInformation(*ptrReceiveInfo);
+
+ rtcpPacketType = rtcpParser.Iterate();
+
+ while (rtcpPacketType == RTCPUtility::kRtcpReportBlockItemCode)
+ {
+ HandleReportBlock(rtcpPacket, rtcpPacketInformation, remoteSSRC, numberOfReportBlocks);
+ rtcpPacketType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation,
+ const WebRtc_UWord32 remoteSSRC,
+ const WebRtc_UWord8 numberOfReportBlocks) {
+ // This will be called once per report block in the RTCP packet.
+ // We filter out all report blocks that are not for us.
+ // Each packet has max 31 RR blocks.
+ //
+ // We can calc RTT if we send a send report and get a report block back.
+
+ // |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to
+ // which the information in this reception report block pertains.
+
+ // Filter out all report blocks that are not for us.
+ if (rtcpPacket.ReportBlockItem.SSRC != _SSRC) {
+ // This block is not for us ignore it.
+ return;
+ }
+
+ // To avoid problem with acquiring _criticalSectionRTCPSender while holding
+ // _criticalSectionRTCPReceiver.
+ _criticalSectionRTCPReceiver->Leave();
+ WebRtc_UWord32 sendTimeMS =
+ _rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR);
+ _criticalSectionRTCPReceiver->Enter();
+
+ RTCPReportBlockInformation* reportBlock =
+ CreateReportBlockInformation(remoteSSRC);
+ if (reportBlock == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC);
+ return;
+ }
+ const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem;
+ reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC;
+ reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
+ reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost;
+ reportBlock->remoteReceiveBlock.cumulativeLost =
+ rb.CumulativeNumOfPacketsLost;
+ reportBlock->remoteReceiveBlock.extendedHighSeqNum =
+ rb.ExtendedHighestSequenceNumber;
+ reportBlock->remoteReceiveBlock.jitter = rb.Jitter;
+ reportBlock->remoteReceiveBlock.delaySinceLastSR = rb.DelayLastSR;
+ reportBlock->remoteReceiveBlock.lastSR = rb.LastSR;
+
+ if (rtcpPacket.ReportBlockItem.Jitter > reportBlock->remoteMaxJitter) {
+ reportBlock->remoteMaxJitter = rtcpPacket.ReportBlockItem.Jitter;
+ }
+
+ WebRtc_UWord32 delaySinceLastSendReport =
+ rtcpPacket.ReportBlockItem.DelayLastSR;
+
+ // local NTP time when we received this
+ WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
+ WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
+
+ _clock.CurrentNTP(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
+
+ // time when we received this in MS
+ WebRtc_UWord32 receiveTimeMS = ModuleRTPUtility::ConvertNTPTimeToMS(
+ lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
+
+ // Estimate RTT
+ WebRtc_UWord32 d = (delaySinceLastSendReport & 0x0000ffff) * 1000;
+ d /= 65536;
+ d += ((delaySinceLastSendReport & 0xffff0000) >> 16) * 1000;
+
+ WebRtc_Word32 RTT = 0;
+
+ if (sendTimeMS > 0) {
+ RTT = receiveTimeMS - d - sendTimeMS;
+ if (RTT <= 0) {
+ RTT = 1;
+ }
+ if (RTT > reportBlock->maxRTT) {
+ // store max RTT
+ reportBlock->maxRTT = (WebRtc_UWord16) RTT;
+ }
+ if (reportBlock->minRTT == 0) {
+ // first RTT
+ reportBlock->minRTT = (WebRtc_UWord16) RTT;
+ } else if (RTT < reportBlock->minRTT) {
+ // Store min RTT
+ reportBlock->minRTT = (WebRtc_UWord16) RTT;
+ }
+ // store last RTT
+ reportBlock->RTT = (WebRtc_UWord16) RTT;
+
+ // store average RTT
+ if (reportBlock->numAverageCalcs != 0) {
+ float ac = static_cast<float> (reportBlock->numAverageCalcs);
+ float newAverage = ((ac / (ac + 1)) * reportBlock->avgRTT)
+ + ((1 / (ac + 1)) * RTT);
+ reportBlock->avgRTT = static_cast<int> (newAverage + 0.5f);
+ } else {
+ // first RTT
+ reportBlock->avgRTT = (WebRtc_UWord16) RTT;
+ }
+ reportBlock->numAverageCalcs++;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
+ " -> Received report block(%d), from SSRC:0x%x, RTT:%d, loss:%d",
+ _id, remoteSSRC, RTT, rtcpPacket.ReportBlockItem.FractionLost);
+
+ // rtcpPacketInformation
+ rtcpPacketInformation.AddReportInfo(
+ reportBlock->remoteReceiveBlock.fractionLost, (WebRtc_UWord16) RTT,
+ reportBlock->remoteReceiveBlock.extendedHighSeqNum,
+ reportBlock->remoteReceiveBlock.jitter);
+}
+
+RTCPReportBlockInformation*
+RTCPReceiver::CreateReportBlockInformation(WebRtc_UWord32 remoteSSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator it =
+ _receivedReportBlockMap.find(remoteSSRC);
+
+ RTCPReportBlockInformation* ptrReportBlockInfo = NULL;
+ if (it != _receivedReportBlockMap.end()) {
+ ptrReportBlockInfo = it->second;
+ } else {
+ ptrReportBlockInfo = new RTCPReportBlockInformation;
+ _receivedReportBlockMap[remoteSSRC] = ptrReportBlockInfo;
+ }
+ return ptrReportBlockInfo;
+}
+
+RTCPReportBlockInformation*
+RTCPReceiver::GetReportBlockInformation(WebRtc_UWord32 remoteSSRC) const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
+ _receivedReportBlockMap.find(remoteSSRC);
+
+ if (it == _receivedReportBlockMap.end()) {
+ return NULL;
+ }
+ return it->second;
+}
+
+RTCPCnameInformation*
+RTCPReceiver::CreateCnameInformation(WebRtc_UWord32 remoteSSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
+ _receivedCnameMap.find(remoteSSRC);
+
+ if (it != _receivedCnameMap.end()) {
+ return it->second;
+ }
+ RTCPCnameInformation* cnameInfo = new RTCPCnameInformation;
+ memset(cnameInfo->name, 0, RTCP_CNAME_SIZE);
+ _receivedCnameMap[remoteSSRC] = cnameInfo;
+ return cnameInfo;
+}
+
+RTCPCnameInformation*
+RTCPReceiver::GetCnameInformation(WebRtc_UWord32 remoteSSRC) const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::const_iterator it =
+ _receivedCnameMap.find(remoteSSRC);
+
+ if (it == _receivedCnameMap.end()) {
+ return NULL;
+ }
+ return it->second;
+}
+
+RTCPReceiveInformation*
+RTCPReceiver::CreateReceiveInformation(WebRtc_UWord32 remoteSSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
+ _receivedInfoMap.find(remoteSSRC);
+
+ if (it != _receivedInfoMap.end()) {
+ return it->second;
+ }
+ RTCPReceiveInformation* receiveInfo = new RTCPReceiveInformation;
+ _receivedInfoMap[remoteSSRC] = receiveInfo;
+ return receiveInfo;
+}
+
+RTCPReceiveInformation*
+RTCPReceiver::GetReceiveInformation(WebRtc_UWord32 remoteSSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
+ _receivedInfoMap.find(remoteSSRC);
+ if (it == _receivedInfoMap.end()) {
+ return NULL;
+ }
+ return it->second;
+}
+
+void RTCPReceiver::UpdateReceiveInformation(
+ RTCPReceiveInformation& receiveInformation) {
+ // Update that this remote is alive
+ receiveInformation.lastTimeReceived = _clock.GetTimeInMS();
+}
+
+bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ bool updateBoundingSet = false;
+ WebRtc_UWord32 timeNow = _clock.GetTimeInMS();
+
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
+ _receivedInfoMap.begin();
+
+ while (receiveInfoIt != _receivedInfoMap.end()) {
+ RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
+ if (receiveInfo == NULL) {
+ return updateBoundingSet;
+ }
+ // time since last received rtcp packet
+ // when we dont have a lastTimeReceived and the object is marked
+ // readyForDelete it's removed from the map
+ if (receiveInfo->lastTimeReceived) {
+ /// use audio define since we don't know what interval the remote peer is
+ // using
+ if ((timeNow - receiveInfo->lastTimeReceived) >
+ 5 * RTCP_INTERVAL_AUDIO_MS) {
+ // no rtcp packet for the last five regular intervals, reset limitations
+ receiveInfo->TmmbrSet.lengthOfSet = 0;
+ // prevent that we call this over and over again
+ receiveInfo->lastTimeReceived = 0;
+ // send new TMMBN to all channels using the default codec
+ updateBoundingSet = true;
+ }
+ receiveInfoIt++;
+ } else if (receiveInfo->readyForDelete) {
+ // store our current receiveInfoItem
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator
+ receiveInfoItemToBeErased = receiveInfoIt;
+ receiveInfoIt++;
+ delete receiveInfoItemToBeErased->second;
+ _receivedInfoMap.erase(receiveInfoItemToBeErased);
+ } else {
+ receiveInfoIt++;
+ }
+ }
+ return updateBoundingSet;
+}
+
+WebRtc_Word32 RTCPReceiver::BoundingSet(bool &tmmbrOwner,
+ TMMBRSet*& boundingSetRec) {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
+ _receivedInfoMap.find(_remoteSSRC);
+
+ if (receiveInfoIt == _receivedInfoMap.end()) {
+ return -1;
+ }
+ RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
+ if (receiveInfo == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s failed to get RTCPReceiveInformation",
+ __FUNCTION__);
+ return -1;
+ }
+ if (receiveInfo->TmmbnBoundingSet.lengthOfSet > 0) {
+ boundingSetRec->VerifyAndAllocateSet(
+ receiveInfo->TmmbnBoundingSet.lengthOfSet + 1);
+ for(WebRtc_UWord32 i=0; i< receiveInfo->TmmbnBoundingSet.lengthOfSet; i++) {
+ if(receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i] == _SSRC) {
+ // owner of bounding set
+ tmmbrOwner = true;
+ }
+ boundingSetRec->ptrTmmbrSet[i] =
+ receiveInfo->TmmbnBoundingSet.ptrTmmbrSet[i];
+ boundingSetRec->ptrPacketOHSet[i] =
+ receiveInfo->TmmbnBoundingSet.ptrPacketOHSet[i];
+ boundingSetRec->ptrSsrcSet[i] =
+ receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i];
+ }
+ }
+ return receiveInfo->TmmbnBoundingSet.lengthOfSet;
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser)
+{
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpSdesChunkCode)
+ {
+ HandleSDESChunk(rtcpParser);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+ RTCPCnameInformation* cnameInfo =
+ CreateCnameInformation(rtcpPacket.CName.SenderSSRC);
+ assert(cnameInfo);
+
+ cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.NACK.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+ if (_SSRC != rtcpPacket.NACK.MediaSSRC)
+ {
+ // Not to us.
+ rtcpParser.Iterate();
+ return;
+ }
+
+ rtcpPacketInformation.ResetNACKPacketIdArray();
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpRtpfbNackItemCode)
+ {
+ HandleNACKItem(rtcpPacket, rtcpPacketInformation);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID);
+
+ WebRtc_UWord16 bitMask = rtcpPacket.NACKItem.BitMask;
+ if(bitMask)
+ {
+ for(int i=1; i <= 16; ++i)
+ {
+ if(bitMask & 0x01)
+ {
+ rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID + i);
+ }
+ bitMask = bitMask >>1;
+ }
+ }
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpNack;
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void RTCPReceiver::HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser) {
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ // clear our lists
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator
+ reportBlockInfoIt = _receivedReportBlockMap.find(
+ rtcpPacket.BYE.SenderSSRC);
+
+ if (reportBlockInfoIt != _receivedReportBlockMap.end()) {
+ delete reportBlockInfoIt->second;
+ _receivedReportBlockMap.erase(reportBlockInfoIt);
+ }
+ // we can't delete it due to TMMBR
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
+ _receivedInfoMap.find(rtcpPacket.BYE.SenderSSRC);
+
+ if (receiveInfoIt != _receivedInfoMap.end()) {
+ receiveInfoIt->second->readyForDelete = true;
+ }
+
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator cnameInfoIt =
+ _receivedCnameMap.find(rtcpPacket.BYE.SenderSSRC);
+
+ if (cnameInfoIt != _receivedCnameMap.end()) {
+ delete cnameInfoIt->second;
+ _receivedCnameMap.erase(cnameInfoIt);
+ }
+ rtcpParser.Iterate();
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ if(rtcpPacket.XRVOIPMetricItem.SSRC == _SSRC)
+ {
+ // Store VoIP metrics block if it's about me
+ // from OriginatorSSRC do we filter it?
+ // rtcpPacket.XR.OriginatorSSRC;
+
+ RTCPVoIPMetric receivedVoIPMetrics;
+ receivedVoIPMetrics.burstDensity = rtcpPacket.XRVOIPMetricItem.burstDensity;
+ receivedVoIPMetrics.burstDuration = rtcpPacket.XRVOIPMetricItem.burstDuration;
+ receivedVoIPMetrics.discardRate = rtcpPacket.XRVOIPMetricItem.discardRate;
+ receivedVoIPMetrics.endSystemDelay = rtcpPacket.XRVOIPMetricItem.endSystemDelay;
+ receivedVoIPMetrics.extRfactor = rtcpPacket.XRVOIPMetricItem.extRfactor;
+ receivedVoIPMetrics.gapDensity = rtcpPacket.XRVOIPMetricItem.gapDensity;
+ receivedVoIPMetrics.gapDuration = rtcpPacket.XRVOIPMetricItem.gapDuration;
+ receivedVoIPMetrics.Gmin = rtcpPacket.XRVOIPMetricItem.Gmin;
+ receivedVoIPMetrics.JBabsMax = rtcpPacket.XRVOIPMetricItem.JBabsMax;
+ receivedVoIPMetrics.JBmax = rtcpPacket.XRVOIPMetricItem.JBmax;
+ receivedVoIPMetrics.JBnominal = rtcpPacket.XRVOIPMetricItem.JBnominal;
+ receivedVoIPMetrics.lossRate = rtcpPacket.XRVOIPMetricItem.lossRate;
+ receivedVoIPMetrics.MOSCQ = rtcpPacket.XRVOIPMetricItem.MOSCQ;
+ receivedVoIPMetrics.MOSLQ = rtcpPacket.XRVOIPMetricItem.MOSLQ;
+ receivedVoIPMetrics.noiseLevel = rtcpPacket.XRVOIPMetricItem.noiseLevel;
+ receivedVoIPMetrics.RERL = rtcpPacket.XRVOIPMetricItem.RERL;
+ receivedVoIPMetrics.Rfactor = rtcpPacket.XRVOIPMetricItem.Rfactor;
+ receivedVoIPMetrics.roundTripDelay = rtcpPacket.XRVOIPMetricItem.roundTripDelay;
+ receivedVoIPMetrics.RXconfig = rtcpPacket.XRVOIPMetricItem.RXconfig;
+ receivedVoIPMetrics.signalLevel = rtcpPacket.XRVOIPMetricItem.signalLevel;
+
+ rtcpPacketInformation.AddVoIPMetric(&receivedVoIPMetrics);
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrVoipMetric; // received signal
+ }
+ rtcpParser.Iterate();
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.PLI.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+ if (_SSRC != rtcpPacket.PLI.MediaSSRC)
+ {
+ // Not to us.
+ rtcpParser.Iterate();
+ return;
+ }
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli; // received signal that we need to send a new key frame
+ rtcpParser.Iterate();
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ WebRtc_UWord32 senderSSRC = rtcpPacket.TMMBR.SenderSSRC;
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(senderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+ if(rtcpPacket.TMMBR.MediaSSRC)
+ {
+ // rtcpPacket.TMMBR.MediaSSRC SHOULD be 0 if same as SenderSSRC
+ // in relay mode this is a valid number
+ senderSSRC = rtcpPacket.TMMBR.MediaSSRC;
+ }
+
+ // Use packet length to calc max number of TMMBR blocks
+ // each TMMBR block is 8 bytes
+ ptrdiff_t maxNumOfTMMBRBlocks = rtcpParser.LengthLeft() / 8;
+
+ // sanity
+ if(maxNumOfTMMBRBlocks > 200) // we can't have more than what's in one packet
+ {
+ assert(false);
+ rtcpParser.Iterate();
+ return;
+ }
+ ptrReceiveInfo->VerifyAndAllocateTMMBRSet((WebRtc_UWord32)maxNumOfTMMBRBlocks);
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpRtpfbTmmbrItemCode)
+ {
+ HandleTMMBRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation, senderSSRC);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation,
+ const WebRtc_UWord32 senderSSRC)
+{
+ if (_SSRC == rtcpPacket.TMMBRItem.SSRC &&
+ rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0)
+ {
+ receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem,
+ _clock.GetTimeInMS());
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr;
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.TMMBN.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+ // Use packet length to calc max number of TMMBN blocks
+ // each TMMBN block is 8 bytes
+ ptrdiff_t maxNumOfTMMBNBlocks = rtcpParser.LengthLeft() / 8;
+
+ // sanity
+ if(maxNumOfTMMBNBlocks > 200) // we cant have more than what's in one packet
+ {
+ assert(false);
+ rtcpParser.Iterate();
+ return;
+ }
+
+ ptrReceiveInfo->VerifyAndAllocateBoundingSet((WebRtc_UWord32)maxNumOfTMMBNBlocks);
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode)
+ {
+ HandleTMMBNItem(*ptrReceiveInfo, rtcpPacket);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSrReq;
+ rtcpParser.Iterate();
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket)
+{
+ const unsigned int idx = receiveInfo.TmmbnBoundingSet.lengthOfSet;
+
+ receiveInfo.TmmbnBoundingSet.ptrTmmbrSet[idx] = rtcpPacket.TMMBNItem.MaxTotalMediaBitRate;
+ receiveInfo.TmmbnBoundingSet.ptrPacketOHSet[idx] = rtcpPacket.TMMBNItem.MeasuredOverhead;
+ receiveInfo.TmmbnBoundingSet.ptrSsrcSet[idx] = rtcpPacket.TMMBNItem.SSRC;
+
+ ++receiveInfo.TmmbnBoundingSet.lengthOfSet;
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.SLI.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpPsfbSliItemCode)
+ {
+ HandleSLIItem(rtcpPacket, rtcpPacketInformation);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ // in theory there could be multiple slices lost
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSli; // received signal that we need to refresh a slice
+ rtcpPacketInformation.sliPictureId = rtcpPacket.SLIItem.PictureId;
+}
+
+void
+RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.RPSI.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ if(pktType == RTCPUtility::kRtcpPsfbRpsiCode)
+ {
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRpsi; // received signal that we have a confirmed reference picture
+ if(rtcpPacket.RPSI.NumberOfValidBits%8 != 0)
+ {
+ // to us unknown
+ // continue
+ rtcpParser.Iterate();
+ return;
+ }
+ rtcpPacketInformation.rpsiPictureId = 0;
+
+ // convert NativeBitString to rpsiPictureId
+ WebRtc_UWord8 numberOfBytes = rtcpPacket.RPSI.NumberOfValidBits /8;
+ for(WebRtc_UWord8 n = 0; n < (numberOfBytes-1); n++)
+ {
+ rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[n] & 0x7f);
+ rtcpPacketInformation.rpsiPictureId <<= 7; // prepare next
+ }
+ rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[numberOfBytes-1] & 0x7f);
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ if (pktType == RTCPUtility::kRtcpPsfbRembItemCode)
+ {
+ HandleREMBItem(rtcpParser, rtcpPacketInformation);
+ rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpExtendedIjItemCode)
+ {
+ HandleIJItem(rtcpPacket, rtcpPacketInformation);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+void
+RTCPReceiver::HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
+ rtcpPacketInformation.interArrivalJitter =
+ rtcpPacket.ExtendedJitterReportItem.Jitter;
+}
+
+void
+RTCPReceiver::HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ rtcpParser.Iterate();
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb;
+ rtcpPacketInformation.receiverEstimatedMaxBitrate = rtcpPacket.REMB.BitRate;
+ // TODO(pwestin) send up SSRCs and do a sanity check
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.FIR.SenderSSRC);
+ if (ptrReceiveInfo == NULL)
+ {
+ // This remote SSRC must be saved before.
+ rtcpParser.Iterate();
+ return;
+ }
+
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
+ while (pktType == RTCPUtility::kRtcpPsfbFirItemCode)
+ {
+ HandleFIRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation);
+ pktType = rtcpParser.Iterate();
+ }
+}
+
+// no need for critsect we have _criticalSectionRTCPReceiver
+void
+RTCPReceiver::HandleFIRItem(RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ if (_SSRC == rtcpPacket.FIRItem.SSRC) // is it our sender that is requested to generate a new keyframe
+ {
+ // rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it
+ // we don't know who this originate from
+
+ // check if we have reported this FIRSequenceNumber before
+ if (rtcpPacket.FIRItem.CommandSequenceNumber != receiveInfo.lastFIRSequenceNumber)
+ {
+ //
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+
+ // extra sanity don't go crazy with the callbacks
+ if( (now - receiveInfo.lastFIRRequest) > RTCP_MIN_FRAME_LENGTH_MS)
+ {
+ receiveInfo.lastFIRRequest = now;
+ receiveInfo.lastFIRSequenceNumber = rtcpPacket.FIRItem.CommandSequenceNumber;
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir; // received signal that we need to send a new key frame
+ }
+ }
+ }
+}
+
+void
+RTCPReceiver::HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpApp;
+ rtcpPacketInformation.applicationSubType = rtcpPacket.APP.SubType;
+ rtcpPacketInformation.applicationName = rtcpPacket.APP.Name;
+
+ rtcpParser.Iterate();
+}
+
+void
+RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+
+ rtcpPacketInformation.AddApplicationData(rtcpPacket.APP.Data, rtcpPacket.APP.Size);
+
+ rtcpParser.Iterate();
+}
+
+void
+RTCPReceiver::OnReceivedIntraFrameRequest(const FrameType frameType,
+ const WebRtc_UWord8 streamIdx) const
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+
+ if(_cbVideoFeedback)
+ {
+ _cbVideoFeedback->OnReceivedIntraFrameRequest(_id, frameType, streamIdx);
+ }
+}
+
+void
+RTCPReceiver::OnReceivedSliceLossIndication(const WebRtc_UWord8 pitureID) const
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+
+ if(_cbRtcpFeedback)
+ {
+ _cbRtcpFeedback->OnSLIReceived(_id, pitureID);
+ }
+}
+
+void
+RTCPReceiver::OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64 pitureID) const
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+
+ if(_cbRtcpFeedback)
+ {
+ _cbRtcpFeedback->OnRPSIReceived(_id, pitureID);
+ }
+}
+
+// Holding no Critical section
+void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
+ RTCPPacketInformation& rtcpPacketInformation)
+{
+ // Process TMMBR and REMB first to avoid multiple callbacks
+ // to OnNetworkChanged.
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+ "SIG [RTCP] Incoming TMMBR to id:%d", _id);
+
+ // Might trigger a OnReceivedBandwidthEstimateUpdate.
+ _rtpRtcp.OnReceivedTMMBR();
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+ "SIG [RTCP] Incoming REMB to id:%d", _id);
+
+ // We need to bounce this to the default channel.
+ _rtpRtcp.OnReceivedEstimatedMaxBitrate(
+ rtcpPacketInformation.receiverEstimatedMaxBitrate);
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
+ rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr)
+ {
+ if (rtcpPacketInformation.reportBlock)
+ {
+ _rtpRtcp.OnPacketLossStatisticsUpdate(
+ rtcpPacketInformation.fractionLost,
+ rtcpPacketInformation.roundTripTime,
+ rtcpPacketInformation.lastReceivedExtendedHighSeqNum);
+ }
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
+ {
+ _rtpRtcp.OnReceivedNTP();
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq)
+ {
+ _rtpRtcp.OnRequestSendReport();
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack)
+ {
+ if (rtcpPacketInformation.nackSequenceNumbersLength > 0)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+ "SIG [RTCP] Incoming NACK to id:%d", _id);
+ _rtpRtcp.OnReceivedNACK(
+ rtcpPacketInformation.nackSequenceNumbersLength,
+ rtcpPacketInformation.nackSequenceNumbers);
+ }
+ }
+ if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
+ (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir))
+ {
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+ "SIG [RTCP] Incoming PLI to id:%d", _id);
+ } else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+ "SIG [RTCP] Incoming FIR to id:%d", _id);
+ }
+ _rtpRtcp.OnReceivedIntraFrameRequest(&_rtpRtcp);
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSli)
+ {
+ // we need use a bounce it up to handle default channel
+ _rtpRtcp.OnReceivedSliceLossIndication(
+ rtcpPacketInformation.sliPictureId);
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi)
+ {
+ // we need use a bounce it up to handle default channel
+ _rtpRtcp.OnReceivedReferencePictureSelectionIndication(
+ rtcpPacketInformation.rpsiPictureId);
+ }
+ {
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+
+ // we need a feedback that we have received a report block(s) so that we can generate a new packet
+ // in a conference relay scenario, one received report can generate several RTCP packets, based
+ // on number relayed/mixed
+ // a send report block should go out to all receivers
+ if(_cbRtcpFeedback)
+ {
+ if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
+ {
+ _cbRtcpFeedback->OnSendReportReceived(_id, rtcpPacketInformation.remoteSSRC);
+ } else
+ {
+ _cbRtcpFeedback->OnReceiveReportReceived(_id, rtcpPacketInformation.remoteSSRC);
+ }
+ if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
+ {
+ _cbRtcpFeedback->OnReceiverEstimatedMaxBitrateReceived(_id,
+ rtcpPacketInformation.receiverEstimatedMaxBitrate);
+ }
+ if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric)
+ {
+ WebRtc_Word8 VoIPmetricBuffer[7*4];
+ VoIPmetricBuffer[0] = rtcpPacketInformation.VoIPMetric->lossRate;
+ VoIPmetricBuffer[1] = rtcpPacketInformation.VoIPMetric->discardRate;
+ VoIPmetricBuffer[2] = rtcpPacketInformation.VoIPMetric->burstDensity;
+ VoIPmetricBuffer[3] = rtcpPacketInformation.VoIPMetric->gapDensity;
+
+ VoIPmetricBuffer[4] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration >> 8);
+ VoIPmetricBuffer[5] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration);
+ VoIPmetricBuffer[6] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration >> 8);
+ VoIPmetricBuffer[7] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration);
+
+ VoIPmetricBuffer[8] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay >> 8);
+ VoIPmetricBuffer[9] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay);
+ VoIPmetricBuffer[10] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay >> 8);
+ VoIPmetricBuffer[11] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay);
+
+ VoIPmetricBuffer[12] = rtcpPacketInformation.VoIPMetric->signalLevel;
+ VoIPmetricBuffer[13] = rtcpPacketInformation.VoIPMetric->noiseLevel;
+ VoIPmetricBuffer[14] = rtcpPacketInformation.VoIPMetric->RERL;
+ VoIPmetricBuffer[15] = rtcpPacketInformation.VoIPMetric->Gmin;
+
+ VoIPmetricBuffer[16] = rtcpPacketInformation.VoIPMetric->Rfactor;
+ VoIPmetricBuffer[17] = rtcpPacketInformation.VoIPMetric->extRfactor;
+ VoIPmetricBuffer[18] = rtcpPacketInformation.VoIPMetric->MOSLQ;
+ VoIPmetricBuffer[19] = rtcpPacketInformation.VoIPMetric->MOSCQ;
+
+ VoIPmetricBuffer[20] = rtcpPacketInformation.VoIPMetric->RXconfig;
+ VoIPmetricBuffer[21] = 0; // reserved
+ VoIPmetricBuffer[22] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal >> 8);
+ VoIPmetricBuffer[23] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal);
+
+ VoIPmetricBuffer[24] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax >> 8);
+ VoIPmetricBuffer[25] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax);
+ VoIPmetricBuffer[26] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax >> 8);
+ VoIPmetricBuffer[27] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax);
+
+ _cbRtcpFeedback->OnXRVoIPMetricReceived(_id, rtcpPacketInformation.VoIPMetric, VoIPmetricBuffer);
+ }
+ if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp)
+ {
+ _cbRtcpFeedback->OnApplicationDataReceived(_id,
+ rtcpPacketInformation.applicationSubType,
+ rtcpPacketInformation.applicationName,
+ rtcpPacketInformation.applicationLength,
+ rtcpPacketInformation.applicationData);
+ }
+ }
+ }
+}
+
+void
+RTCPReceiver::UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+
+ if(_cbRtcpFeedback)
+ {
+ _cbRtcpFeedback->OnTMMBRReceived(_id, bwEstimateKbit);
+ }
+
+}
+
+WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const {
+ if (cName == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
+ assert(cnameInfo);
+
+ cName[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cName, cnameInfo->name, RTCP_CNAME_SIZE - 1);
+ return 0;
+}
+
+// no callbacks allowed inside this function
+WebRtc_Word32 RTCPReceiver::TMMBRReceived(const WebRtc_UWord32 size,
+ const WebRtc_UWord32 accNumCandidates,
+ TMMBRSet* candidateSet) const {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+
+ std::map<WebRtc_UWord32, RTCPReceiveInformation*>::const_iterator
+ receiveInfoIt = _receivedInfoMap.begin();
+ if (receiveInfoIt == _receivedInfoMap.end()) {
+ return -1;
+ }
+ WebRtc_UWord32 num = accNumCandidates;
+ if (candidateSet) {
+ while( num < size && receiveInfoIt != _receivedInfoMap.end()) {
+ RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
+ if (receiveInfo == NULL) {
+ return 0;
+ }
+ for (WebRtc_UWord32 i = 0;
+ (num < size) && (i < receiveInfo->TmmbrSet.lengthOfSet); i++) {
+ if (receiveInfo->GetTMMBRSet(i, num, candidateSet,
+ _clock.GetTimeInMS()) == 0) {
+ num++;
+ }
+ }
+ receiveInfoIt++;
+ }
+ } else {
+ while (receiveInfoIt != _receivedInfoMap.end()) {
+ RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
+ if(receiveInfo == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s failed to get RTCPReceiveInformation",
+ __FUNCTION__);
+ return -1;
+ }
+ num += receiveInfo->TmmbrSet.lengthOfSet;
+ receiveInfoIt++;
+ }
+ }
+ return num;
+}
+
+WebRtc_Word32
+RTCPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ _packetTimeOutMS = timeoutMS;
+ return 0;
+}
+
+void RTCPReceiver::PacketTimeout()
+{
+ if(_packetTimeOutMS == 0)
+ {
+ // not configured
+ return;
+ }
+
+ bool packetTimeOut = false;
+ {
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if(_lastReceived == 0)
+ {
+ // not active
+ return;
+ }
+
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+
+ if(now - _lastReceived > _packetTimeOutMS)
+ {
+ packetTimeOut = true;
+ _lastReceived = 0; // only one callback
+ }
+ }
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
+ if(packetTimeOut && _cbRtcpFeedback)
+ {
+ _cbRtcpFeedback->OnRTCPPacketTimeout(_id);
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h
new file mode 100644
index 0000000..eb1b658
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
+
+#include <map>
+#include <vector>
+
+#include "typedefs.h"
+#include "rtp_utility.h"
+#include "rtcp_utility.h"
+#include "rtp_rtcp_defines.h"
+#include "rtcp_receiver_help.h"
+
+namespace webrtc {
+class ModuleRtpRtcpImpl;
+
+class RTCPReceiver
+{
+public:
+ RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner);
+ virtual ~RTCPReceiver();
+
+ void ChangeUniqueId(const WebRtc_Word32 id);
+
+ RTCPMethod Status() const;
+ WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
+
+ WebRtc_UWord32 LastReceived();
+
+ void SetSSRC( const WebRtc_UWord32 ssrc);
+ void SetRelaySSRC( const WebRtc_UWord32 ssrc);
+ WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
+
+ WebRtc_UWord32 RelaySSRC() const;
+
+ WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback);
+
+ WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback);
+
+ WebRtc_Word32 IncomingRTCPPacket(RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
+ RTCPUtility::RTCPParserV2 *rtcpParser);
+
+ void TriggerCallbacksFromRTCPPacket(RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ // get received cname
+ WebRtc_Word32 CNAME(const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const;
+
+ // get received NTP
+ WebRtc_Word32 NTP(WebRtc_UWord32 *ReceivedNTPsecs,
+ WebRtc_UWord32 *ReceivedNTPfrac,
+ WebRtc_UWord32 *RTCPArrivalTimeSecs,
+ WebRtc_UWord32 *RTCPArrivalTimeFrac) const;
+
+ // get rtt
+ WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
+ WebRtc_UWord16* RTT,
+ WebRtc_UWord16* avgRTT,
+ WebRtc_UWord16* minRTT,
+ WebRtc_UWord16* maxRTT) const;
+
+ WebRtc_UWord16 RTT() const;
+
+ int SetRTT(WebRtc_UWord16 rtt);
+
+ WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
+
+ void UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const;
+
+ WebRtc_Word32 SenderInfoReceived(RTCPSenderInfo* senderInfo) const;
+
+ void OnReceivedIntraFrameRequest(const FrameType frameType,
+ const WebRtc_UWord8 streamIdx) const;
+
+ void OnReceivedSliceLossIndication(const WebRtc_UWord8 pitureID) const;
+ void OnReceivedReferencePictureSelectionIndication(
+ const WebRtc_UWord64 pitureID) const;
+
+ // get statistics
+ WebRtc_Word32 StatisticsReceived(
+ std::vector<RTCPReportBlock>* receiveBlocks) const;
+
+ // Get TMMBR
+ WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
+ const WebRtc_UWord32 accNumCandidates,
+ TMMBRSet* candidateSet) const;
+
+ bool UpdateRTCPReceiveInformationTimers();
+
+ void UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit);
+
+ WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
+ TMMBRSet*& boundingSetRec);
+
+ WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
+ void PacketTimeout();
+
+protected:
+ RTCPHelp::RTCPReportBlockInformation* CreateReportBlockInformation(const WebRtc_UWord32 remoteSSRC);
+ RTCPHelp::RTCPReportBlockInformation* GetReportBlockInformation(const WebRtc_UWord32 remoteSSRC) const;
+
+ RTCPUtility::RTCPCnameInformation* CreateCnameInformation(const WebRtc_UWord32 remoteSSRC);
+ RTCPUtility::RTCPCnameInformation* GetCnameInformation(const WebRtc_UWord32 remoteSSRC) const;
+
+ RTCPHelp::RTCPReceiveInformation* CreateReceiveInformation(const WebRtc_UWord32 remoteSSRC);
+ RTCPHelp::RTCPReceiveInformation* GetReceiveInformation(const WebRtc_UWord32 remoteSSRC);
+
+ void UpdateReceiveInformation( RTCPHelp::RTCPReceiveInformation& receiveInformation);
+
+ void HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
+ const WebRtc_UWord32 remoteSSRC,
+ const WebRtc_UWord8 numberOfReportBlocks);
+
+ void HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser);
+
+ void HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser);
+
+ void HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser);
+
+ void HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleTMMBRItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
+ const WebRtc_UWord32 senderSSRC);
+
+ void HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser);
+
+ void HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleTMMBNItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket);
+
+ void HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleFIRItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
+ const RTCPUtility::RTCPPacket& rtcpPacket,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ void HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
+ RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+
+ private:
+ WebRtc_Word32 _id;
+ RtpRtcpClock& _clock;
+ RTCPMethod _method;
+ WebRtc_UWord32 _lastReceived;
+ ModuleRtpRtcpImpl& _rtpRtcp;
+
+ CriticalSectionWrapper* _criticalSectionFeedbacks;
+ RtcpFeedback* _cbRtcpFeedback;
+ RtpVideoFeedback* _cbVideoFeedback;
+
+ CriticalSectionWrapper* _criticalSectionRTCPReceiver;
+ WebRtc_UWord32 _SSRC;
+ WebRtc_UWord32 _remoteSSRC;
+
+ // Received send report
+ RTCPSenderInfo _remoteSenderInfo;
+ // when did we receive the last send report
+ WebRtc_UWord32 _lastReceivedSRNTPsecs;
+ WebRtc_UWord32 _lastReceivedSRNTPfrac;
+
+ // Received report blocks.
+ std::map<WebRtc_UWord32, RTCPHelp::RTCPReportBlockInformation*>
+ _receivedReportBlockMap;
+ std::map<WebRtc_UWord32, RTCPHelp::RTCPReceiveInformation*>
+ _receivedInfoMap;
+ std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>
+ _receivedCnameMap;
+
+ WebRtc_UWord32 _packetTimeOutMS;
+
+ // Externally set RTT. This value can only be used if there are no valid
+ // RTT estimates.
+ WebRtc_UWord16 _rtt;
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc
new file mode 100644
index 0000000..0058aca
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -0,0 +1,258 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtcp_receiver_help.h"
+#include "rtp_utility.h"
+
+#include <string.h> //memset
+#include <cassert> //assert
+
+namespace webrtc {
+using namespace RTCPHelp;
+
+RTCPPacketInformation::RTCPPacketInformation()
+ : rtcpPacketTypeFlags(0),
+ remoteSSRC(0),
+ nackSequenceNumbers(0),
+ nackSequenceNumbersLength(0),
+ applicationSubType(0),
+ applicationName(0),
+ applicationData(),
+ applicationLength(0),
+ reportBlock(false),
+ fractionLost(0),
+ roundTripTime(0),
+ lastReceivedExtendedHighSeqNum(0),
+ jitter(0),
+ interArrivalJitter(0),
+ sliPictureId(0),
+ rpsiPictureId(0),
+ receiverEstimatedMaxBitrate(0),
+ VoIPMetric(NULL) {
+}
+
+RTCPPacketInformation::~RTCPPacketInformation()
+{
+ delete [] nackSequenceNumbers;
+ delete [] applicationData;
+ delete VoIPMetric;
+}
+
+void
+RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
+{
+ VoIPMetric = new RTCPVoIPMetric();
+ memcpy(VoIPMetric, metric, sizeof(RTCPVoIPMetric));
+}
+
+void
+RTCPPacketInformation::AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size)
+{
+ WebRtc_UWord8* oldData = applicationData;
+ WebRtc_UWord16 oldLength = applicationLength;
+
+ // Don't copy more than kRtcpAppCode_DATA_SIZE bytes.
+ WebRtc_UWord16 copySize = size;
+ if (size > kRtcpAppCode_DATA_SIZE) {
+ copySize = kRtcpAppCode_DATA_SIZE;
+ }
+
+ applicationLength += copySize;
+ applicationData = new WebRtc_UWord8[applicationLength];
+
+ if(oldData)
+ {
+ memcpy(applicationData, oldData, oldLength);
+ memcpy(applicationData+oldLength, data, copySize);
+ delete [] oldData;
+ } else
+ {
+ memcpy(applicationData, data, copySize);
+ }
+}
+
+void
+RTCPPacketInformation::ResetNACKPacketIdArray()
+{
+ if(NULL == nackSequenceNumbers)
+ {
+ nackSequenceNumbers = new WebRtc_UWord16[NACK_PACKETS_MAX_SIZE];
+ }
+ nackSequenceNumbersLength = 0;
+}
+
+void
+RTCPPacketInformation::AddNACKPacket(const WebRtc_UWord16 packetID)
+{
+ assert(nackSequenceNumbers);
+
+ WebRtc_UWord16& idx = nackSequenceNumbersLength;
+ if (idx < NACK_PACKETS_MAX_SIZE)
+ {
+ nackSequenceNumbers[idx++] = packetID;
+ }
+}
+
+void
+RTCPPacketInformation::AddReportInfo(const WebRtc_UWord8 fraction,
+ const WebRtc_UWord16 rtt,
+ const WebRtc_UWord32 extendedHighSeqNum,
+ const WebRtc_UWord32 j)
+{
+ reportBlock = true;
+ fractionLost = fraction;
+ roundTripTime = rtt;
+ jitter = j;
+ lastReceivedExtendedHighSeqNum = extendedHighSeqNum;
+}
+
+RTCPReportBlockInformation::RTCPReportBlockInformation():
+ remoteReceiveBlock(),
+ remoteMaxJitter(0),
+ RTT(0),
+ minRTT(0),
+ maxRTT(0),
+ avgRTT(0),
+ numAverageCalcs(0)
+{
+ memset(&remoteReceiveBlock,0,sizeof(remoteReceiveBlock));
+}
+
+RTCPReportBlockInformation::~RTCPReportBlockInformation()
+{
+}
+
+RTCPReceiveInformation::RTCPReceiveInformation() :
+
+ lastTimeReceived(0),
+ lastFIRSequenceNumber(-1),
+ lastFIRRequest(0),
+ readyForDelete(false),
+ _tmmbrSetTimeouts(NULL)
+{
+}
+
+RTCPReceiveInformation::~RTCPReceiveInformation()
+{
+ if(_tmmbrSetTimeouts)
+ {
+ delete [] _tmmbrSetTimeouts;
+ }
+}
+
+// don't use TmmbrSet.VerifyAndAllocate this version keeps the data
+void
+RTCPReceiveInformation::VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize)
+{
+ if(minimumSize > TmmbrSet.sizeOfSet)
+ {
+ // make sure that our buffers are big enough
+ WebRtc_UWord32* ptrTmmbrSet = new WebRtc_UWord32[minimumSize];
+ WebRtc_UWord32* ptrTmmbrPacketOHSet = new WebRtc_UWord32[minimumSize];
+ WebRtc_UWord32* ptrTmmbrSsrcSet = new WebRtc_UWord32[minimumSize];
+ WebRtc_UWord32* tmmbrSetTimeouts = new WebRtc_UWord32[minimumSize];
+
+ if(TmmbrSet.lengthOfSet > 0)
+ {
+ // copy old values
+ memcpy(ptrTmmbrSet, TmmbrSet.ptrTmmbrSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
+ memcpy(ptrTmmbrPacketOHSet, TmmbrSet.ptrPacketOHSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
+ memcpy(ptrTmmbrSsrcSet, TmmbrSet.ptrSsrcSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
+ memcpy(tmmbrSetTimeouts, _tmmbrSetTimeouts, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
+ }
+ if(TmmbrSet.ptrTmmbrSet)
+ {
+ delete [] TmmbrSet.ptrTmmbrSet;
+ delete [] TmmbrSet.ptrPacketOHSet;
+ delete [] TmmbrSet.ptrSsrcSet;
+ }
+ if(_tmmbrSetTimeouts)
+ {
+ delete [] _tmmbrSetTimeouts;
+ }
+ TmmbrSet.ptrTmmbrSet = ptrTmmbrSet;
+ TmmbrSet.ptrPacketOHSet = ptrTmmbrPacketOHSet;
+ TmmbrSet.ptrSsrcSet = ptrTmmbrSsrcSet;
+ TmmbrSet.sizeOfSet = minimumSize;
+ _tmmbrSetTimeouts = tmmbrSetTimeouts;
+ }
+}
+
+void
+RTCPReceiveInformation::InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
+ const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
+ const WebRtc_UWord32 currentTimeMS)
+{
+ // serach to see if we have it in our list
+ for(WebRtc_UWord32 i = 0; i < TmmbrSet.lengthOfSet; i++)
+ {
+ if(TmmbrSet.ptrSsrcSet[i] == senderSSRC)
+ {
+ // we already have this SSRC in our list
+ // update it
+ TmmbrSet.ptrPacketOHSet[i] = TMMBRItem.MeasuredOverhead;
+ TmmbrSet.ptrTmmbrSet[i] = TMMBRItem.MaxTotalMediaBitRate;
+ _tmmbrSetTimeouts[i] = currentTimeMS;
+ return;
+ }
+ }
+ VerifyAndAllocateTMMBRSet(TmmbrSet.lengthOfSet+1);
+
+ const WebRtc_UWord32 idx = TmmbrSet.lengthOfSet;
+ TmmbrSet.ptrPacketOHSet[idx] = TMMBRItem.MeasuredOverhead;
+ TmmbrSet.ptrTmmbrSet[idx] = TMMBRItem.MaxTotalMediaBitRate;
+ TmmbrSet.ptrSsrcSet[idx] = senderSSRC;
+ _tmmbrSetTimeouts[idx] = currentTimeMS;
+ TmmbrSet.lengthOfSet++;
+}
+
+WebRtc_Word32
+RTCPReceiveInformation::GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
+ const WebRtc_UWord32 targetIdx,
+ TMMBRSet* candidateSet,
+ const WebRtc_UWord32 currentTimeMS)
+{
+ if(sourceIdx >= TmmbrSet.lengthOfSet)
+ {
+ return -1;
+ }
+ if(targetIdx >= candidateSet->sizeOfSet)
+ {
+ return -1;
+ }
+ WebRtc_UWord32 timeNow = currentTimeMS;
+
+ // use audio define since we don't know what interval the remote peer is using
+ if(timeNow - _tmmbrSetTimeouts[sourceIdx] > 5*RTCP_INTERVAL_AUDIO_MS)
+ {
+ // value timed out
+ const WebRtc_UWord32 move = TmmbrSet.lengthOfSet - (sourceIdx + 1);
+ if(move > 0)
+ {
+ memmove(&(TmmbrSet.ptrTmmbrSet[sourceIdx]), &(TmmbrSet.ptrTmmbrSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
+ memmove(&(TmmbrSet.ptrPacketOHSet[sourceIdx]),&(TmmbrSet.ptrPacketOHSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
+ memmove(&(TmmbrSet.ptrSsrcSet[sourceIdx]),&(TmmbrSet.ptrSsrcSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
+ memmove(&(_tmmbrSetTimeouts[sourceIdx]),&(_tmmbrSetTimeouts[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
+ }
+ TmmbrSet.lengthOfSet--;
+ return -1;
+ }
+
+ candidateSet->ptrTmmbrSet[targetIdx] = TmmbrSet.ptrTmmbrSet[sourceIdx];
+ candidateSet->ptrPacketOHSet[targetIdx] = TmmbrSet.ptrPacketOHSet[sourceIdx];
+ candidateSet->ptrSsrcSet[targetIdx] = TmmbrSet.ptrSsrcSet[sourceIdx];
+ return 0;
+}
+
+void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize)
+{
+ TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h
new file mode 100644
index 0000000..6dcaca8
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
+
+#include "typedefs.h"
+
+#include "rtp_rtcp_defines.h" // RTCPReportBlock
+#include "rtp_rtcp_config.h" // RTCP_MAX_REPORT_BLOCKS
+#include "rtcp_utility.h"
+#include "tmmbr_help.h"
+
+namespace webrtc {
+namespace RTCPHelp
+{
+
+class RTCPPacketInformation
+{
+public:
+ RTCPPacketInformation();
+ ~RTCPPacketInformation();
+
+ void AddVoIPMetric(const RTCPVoIPMetric* metric);
+
+ void AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size);
+
+ void AddNACKPacket(const WebRtc_UWord16 packetID);
+ void ResetNACKPacketIdArray();
+
+ void AddReportInfo(const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 rtt,
+ const WebRtc_UWord32 extendedHighSeqNum,
+ const WebRtc_UWord32 jitter);
+
+ WebRtc_UWord32 rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
+ WebRtc_UWord32 remoteSSRC;
+
+ WebRtc_UWord16* nackSequenceNumbers;
+ WebRtc_UWord16 nackSequenceNumbersLength;
+
+ WebRtc_UWord8 applicationSubType;
+ WebRtc_UWord32 applicationName;
+ WebRtc_UWord8* applicationData;
+ WebRtc_UWord16 applicationLength;
+
+ bool reportBlock;
+ WebRtc_UWord8 fractionLost;
+ WebRtc_UWord16 roundTripTime;
+ WebRtc_UWord32 lastReceivedExtendedHighSeqNum;
+ WebRtc_UWord32 jitter;
+
+ WebRtc_UWord32 interArrivalJitter;
+
+ WebRtc_UWord8 sliPictureId;
+ WebRtc_UWord64 rpsiPictureId;
+ WebRtc_UWord32 receiverEstimatedMaxBitrate;
+
+ RTCPVoIPMetric* VoIPMetric;
+};
+
+
+class RTCPReportBlockInformation
+{
+public:
+ RTCPReportBlockInformation();
+ ~RTCPReportBlockInformation();
+
+ // Statistics
+ RTCPReportBlock remoteReceiveBlock;
+ WebRtc_UWord32 remoteMaxJitter;
+
+ // RTT
+ WebRtc_UWord16 RTT;
+ WebRtc_UWord16 minRTT;
+ WebRtc_UWord16 maxRTT;
+ WebRtc_UWord16 avgRTT;
+ WebRtc_UWord32 numAverageCalcs;
+};
+
+class RTCPReceiveInformation
+{
+public:
+ RTCPReceiveInformation();
+ ~RTCPReceiveInformation();
+
+ void VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize);
+ void VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize);
+
+ void InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
+ const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
+ const WebRtc_UWord32 currentTimeMS);
+
+ // get
+ WebRtc_Word32 GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
+ const WebRtc_UWord32 targetIdx,
+ TMMBRSet* candidateSet,
+ const WebRtc_UWord32 currentTimeMS);
+
+ WebRtc_UWord32 lastTimeReceived;
+
+ // FIR
+ WebRtc_Word32 lastFIRSequenceNumber;
+ WebRtc_UWord32 lastFIRRequest;
+
+ // TMMBN
+ TMMBRSet TmmbnBoundingSet;
+
+ // TMMBR
+ TMMBRSet TmmbrSet;
+
+ bool readyForDelete;
+private:
+ WebRtc_UWord32* _tmmbrSetTimeouts;
+};
+
+} // end namespace RTCPHelp
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc
new file mode 100644
index 0000000..58e849f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -0,0 +1,2189 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtcp_sender.h"
+
+#include <string.h> // memcpy
+#include <cassert> // assert
+#include <cstdlib> // rand
+
+#include "trace.h"
+#include "common_types.h"
+#include "critical_section_wrapper.h"
+
+#include "rtp_rtcp_impl.h"
+
+namespace webrtc {
+
+using RTCPUtility::RTCPCnameInformation;
+
+RTCPSender::RTCPSender(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner) :
+ _id(id),
+ _audio(audio),
+ _clock(*clock),
+ _method(kRtcpOff),
+ _rtpRtcp(*owner),
+ _criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbTransport(NULL),
+
+ _criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()),
+ _usingNack(false),
+ _sending(false),
+ _sendTMMBN(false),
+ _REMB(false),
+ _sendREMB(false),
+ _TMMBR(false),
+ _IJ(false),
+ _nextTimeToSendRTCP(0),
+ _SSRC(0),
+ _remoteSSRC(0),
+ _CNAME(),
+ _reportBlocks(),
+ _csrcCNAMEs(),
+
+ _cameraDelayMS(0),
+
+ _lastSendReport(),
+ _lastRTCPTime(),
+
+ _CSRCs(0),
+ _CSRC(),
+ _includeCSRCs(true),
+
+ _sequenceNumberFIR(0),
+
+ _lengthRembSSRC(0),
+ _sizeRembSSRC(0),
+ _rembSSRC(NULL),
+ _rembBitrate(0),
+ _bitrate_observer(NULL),
+
+ _tmmbrHelp(audio),
+ _tmmbr_Send(0),
+ _packetOH_Send(0),
+ _remoteRateControl(),
+
+ _appSend(false),
+ _appSubType(0),
+ _appName(),
+ _appData(NULL),
+ _appLength(0),
+ _xrSendVoIPMetric(false),
+ _xrVoIPMetric()
+{
+ memset(_CNAME, 0, sizeof(_CNAME));
+ memset(_lastSendReport, 0, sizeof(_lastSendReport));
+ memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+}
+
+RTCPSender::~RTCPSender() {
+ delete [] _rembSSRC;
+ delete [] _appData;
+
+ while (!_reportBlocks.empty()) {
+ std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
+ _reportBlocks.begin();
+ delete it->second;
+ _reportBlocks.erase(it);
+ }
+ while (!_csrcCNAMEs.empty()) {
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
+ _csrcCNAMEs.begin();
+ delete it->second;
+ _csrcCNAMEs.erase(it);
+ }
+ delete _criticalSectionTransport;
+ delete _criticalSectionRTCPSender;
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_Word32
+RTCPSender::Init()
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ _method = kRtcpOff;
+ _cbTransport = NULL;
+ _usingNack = false;
+ _sending = false;
+ _sendTMMBN = false;
+ _TMMBR = false;
+ _IJ = false;
+ _REMB = false;
+ _sendREMB = false;
+ _SSRC = 0;
+ _remoteSSRC = 0;
+ _cameraDelayMS = 0;
+ _sequenceNumberFIR = 0;
+ _tmmbr_Send = 0;
+ _packetOH_Send = 0;
+ _remoteRateControl.Reset();
+ _nextTimeToSendRTCP = 0;
+ _CSRCs = 0;
+ _appSend = false;
+ _appSubType = 0;
+
+ if(_appData)
+ {
+ delete [] _appData;
+ _appData = NULL;
+ }
+ _appLength = 0;
+
+ _xrSendVoIPMetric = false;
+
+ memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric));
+ memset(_CNAME, 0, sizeof(_CNAME));
+ memset(_lastSendReport, 0, sizeof(_lastSendReport));
+ memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
+ return 0;
+}
+
+void
+RTCPSender::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+WebRtc_Word32
+RTCPSender::RegisterSendTransport(Transport* outgoingTransport)
+{
+ CriticalSectionScoped lock(_criticalSectionTransport);
+ _cbTransport = outgoingTransport;
+ return 0;
+}
+
+RTCPMethod
+RTCPSender::Status() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _method;
+}
+
+WebRtc_Word32
+RTCPSender::SetRTCPStatus(const RTCPMethod method)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if(method != kRtcpOff)
+ {
+ if(_audio)
+ {
+ _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_AUDIO_MS/2);
+ } else
+ {
+ _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_VIDEO_MS/2);
+ }
+ }
+ _method = method;
+ return 0;
+}
+
+bool
+RTCPSender::Sending() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _sending;
+}
+
+WebRtc_Word32
+RTCPSender::SetSendingStatus(const bool sending)
+{
+ bool sendRTCPBye = false;
+ {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if(_method != kRtcpOff)
+ {
+ if(sending == false && _sending == true)
+ {
+ // Trigger RTCP bye
+ sendRTCPBye = true;
+ }
+ }
+ _sending = sending;
+ }
+ if(sendRTCPBye)
+ {
+ return SendRTCP(kRtcpBye);
+ }
+ return 0;
+}
+
+bool
+RTCPSender::REMB() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _REMB;
+}
+
+WebRtc_Word32
+RTCPSender::SetREMBStatus(const bool enable)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _REMB = enable;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SetREMBData(const WebRtc_UWord32 bitrate,
+ const WebRtc_UWord8 numberOfSSRC,
+ const WebRtc_UWord32* SSRC)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _rembBitrate = bitrate;
+
+ if(_sizeRembSSRC < numberOfSSRC)
+ {
+ delete [] _rembSSRC;
+ _rembSSRC = new WebRtc_UWord32[numberOfSSRC];
+ _sizeRembSSRC = numberOfSSRC;
+ }
+
+ _lengthRembSSRC = numberOfSSRC;
+ for (int i = 0; i < numberOfSSRC; i++)
+ {
+ _rembSSRC[i] = SSRC[i];
+ }
+ _sendREMB = true;
+ return 0;
+}
+
+bool RTCPSender::SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if (observer && _bitrate_observer) {
+ return false;
+ }
+ _bitrate_observer = observer;
+ return true;
+}
+
+void RTCPSender::UpdateRemoteBitrateEstimate(unsigned int target_bitrate) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if (_bitrate_observer) {
+ _bitrate_observer->OnReceiveBitrateChanged(_remoteSSRC, target_bitrate);
+ }
+}
+
+void RTCPSender::ReceivedRemb(unsigned int estimated_bitrate) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if (_bitrate_observer) {
+ _bitrate_observer->OnReceivedRemb(estimated_bitrate);
+ }
+}
+
+bool
+RTCPSender::TMMBR() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _TMMBR;
+}
+
+WebRtc_Word32
+RTCPSender::SetTMMBRStatus(const bool enable)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _TMMBR = enable;
+ return 0;
+}
+
+bool
+RTCPSender::IJ() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _IJ;
+}
+
+WebRtc_Word32
+RTCPSender::SetIJStatus(const bool enable)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _IJ = enable;
+ return 0;
+}
+
+void
+RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if(_SSRC != 0)
+ {
+ // not first SetSSRC, probably due to a collision
+ // schedule a new RTCP report
+ // make sure that we send a RTP packet
+ _nextTimeToSendRTCP = _clock.GetTimeInMS() + 100;
+ }
+ _SSRC = ssrc;
+}
+
+WebRtc_Word32
+RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _remoteSSRC = ssrc;
+ _remoteRateControl.Reset();
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SetCameraDelay(const WebRtc_Word32 delayMS)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if(delayMS > 1000 || delayMS < -1000)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__);
+ return -1;
+ }
+ _cameraDelayMS = delayMS;
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::CNAME(char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ cName[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cName, _CNAME, RTCP_CNAME_SIZE - 1);
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _CNAME[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1);
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if (_csrcCNAMEs.size() >= kRtpCsrcSize) {
+ return -1;
+ }
+ RTCPCnameInformation* ptr = new RTCPCnameInformation();
+ ptr->name[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1);
+ _csrcCNAMEs[SSRC] = ptr;
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
+ _csrcCNAMEs.find(SSRC);
+
+ if (it == _csrcCNAMEs.end()) {
+ return -1;
+ }
+ delete it->second;
+ _csrcCNAMEs.erase(it);
+ return 0;
+}
+
+bool
+RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const
+{
+/*
+ For audio we use a fix 5 sec interval
+
+ For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
+ technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare
+
+
+From RFC 3550
+
+ MAX RTCP BW is 5% if the session BW
+ A send report is approximately 65 bytes inc CNAME
+ A report report is approximately 28 bytes
+
+ The RECOMMENDED value for the reduced minimum in seconds is 360
+ divided by the session bandwidth in kilobits/second. This minimum
+ is smaller than 5 seconds for bandwidths greater than 72 kb/s.
+
+ If the participant has not yet sent an RTCP packet (the variable
+ initial is true), the constant Tmin is set to 2.5 seconds, else it
+ is set to 5 seconds.
+
+ The interval between RTCP packets is varied randomly over the
+ range [0.5,1.5] times the calculated interval to avoid unintended
+ synchronization of all participants
+
+ if we send
+ If the participant is a sender (we_sent true), the constant C is
+ set to the average RTCP packet size (avg_rtcp_size) divided by 25%
+ of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
+ number of senders.
+
+ if we receive only
+ If we_sent is not true, the constant C is set
+ to the average RTCP packet size divided by 75% of the RTCP
+ bandwidth. The constant n is set to the number of receivers
+ (members - senders). If the number of senders is greater than
+ 25%, senders and receivers are treated together.
+
+ reconsideration NOT required for peer-to-peer
+ "timer reconsideration" is
+ employed. This algorithm implements a simple back-off mechanism
+ which causes users to hold back RTCP packet transmission if the
+ group sizes are increasing.
+
+ n = number of members
+ C = avg_size/(rtcpBW/4)
+
+ 3. The deterministic calculated interval Td is set to max(Tmin, n*C).
+
+ 4. The calculated interval T is set to a number uniformly distributed
+ between 0.5 and 1.5 times the deterministic calculated interval.
+
+ 5. The resulting value of T is divided by e-3/2=1.21828 to compensate
+ for the fact that the timer reconsideration algorithm converges to
+ a value of the RTCP bandwidth below the intended average
+*/
+
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if(_method == kRtcpOff)
+ {
+ return false;
+ }
+
+ if(!_audio && sendKeyframeBeforeRTP)
+ {
+ // for video key-frames we want to send the RTCP before the large key-frame
+ // if we have a 100 ms margin
+ now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
+ }
+
+ if(now > _nextTimeToSendRTCP)
+ {
+ return true;
+
+ } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin
+ {
+ // wrap
+ return true;
+ }
+ return false;
+}
+
+WebRtc_UWord32
+RTCPSender::LastSendReport( WebRtc_UWord32& lastRTCPTime)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ lastRTCPTime = _lastRTCPTime[0];
+ return _lastSendReport[0];
+}
+
+WebRtc_UWord32
+RTCPSender::SendTimeOfSendReport(const WebRtc_UWord32 sendReport)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ // This is only saved when we are the sender
+ if((_lastSendReport[0] == 0) || (sendReport == 0))
+ {
+ return 0; // will be ignored
+ } else
+ {
+ for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i)
+ {
+ if( _lastSendReport[i] == sendReport)
+ {
+ return _lastRTCPTime[i];
+ }
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::AddReportBlock(const WebRtc_UWord32 SSRC,
+ const RTCPReportBlock* reportBlock) {
+ if (reportBlock == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if (_reportBlocks.size() >= RTCP_MAX_REPORT_BLOCKS) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ RTCPReportBlock* copyReportBlock = new RTCPReportBlock();
+ memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock));
+ _reportBlocks[SSRC] = copyReportBlock;
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::RemoveReportBlock(const WebRtc_UWord32 SSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
+ _reportBlocks.find(SSRC);
+
+ if (it == _reportBlocks.end()) {
+ return -1;
+ }
+ delete it->second;
+ _reportBlocks.erase(it);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac,
+ const RTCPReportBlock* received)
+{
+ // sanity
+ if(pos + 52 >= IP_PACKET_SIZE)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -2;
+ }
+ WebRtc_UWord32 RTPtime;
+ WebRtc_UWord32 BackTimedNTPsec;
+ WebRtc_UWord32 BackTimedNTPfrac;
+
+ WebRtc_UWord32 posNumberOfReportBlocks = pos;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
+
+ // Sender report
+ rtcpbuffer[pos++]=(WebRtc_UWord8)200;
+
+ for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--)
+ {
+ // shift old
+ _lastSendReport[i+1] = _lastSendReport[i];
+ _lastRTCPTime[i+1] =_lastRTCPTime[i];
+ }
+
+ _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); // before video cam compensation
+
+ if(_cameraDelayMS >= 0)
+ {
+ // fraction of a second as an unsigned word32 4.294 967 296E9
+ WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)_cameraDelayMS* 4294967; // note camera delay can't be larger than +/-1000ms
+ if(NTPfrac > cameraDelayFixFrac)
+ {
+ // no problem just reduce the fraction part
+ BackTimedNTPfrac = NTPfrac - cameraDelayFixFrac;
+ BackTimedNTPsec = NTPsec;
+ } else
+ {
+ // we need to reduce the sec and add that sec to the frac
+ BackTimedNTPsec = NTPsec - 1;
+ BackTimedNTPfrac = 0xffffffff - (cameraDelayFixFrac - NTPfrac);
+ }
+ } else
+ {
+ // fraction of a second as an unsigned word32 4.294 967 296E9
+ WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)(-_cameraDelayMS)* 4294967; // note camera delay can't be larger than +/-1000ms
+ if(NTPfrac > 0xffffffff - cameraDelayFixFrac)
+ {
+ // we need to add the sec and add that sec to the frac
+ BackTimedNTPsec = NTPsec + 1;
+ BackTimedNTPfrac = cameraDelayFixFrac + NTPfrac; // this will wrap but that is ok
+ } else
+ {
+ // no problem just add the fraction part
+ BackTimedNTPsec = NTPsec;
+ BackTimedNTPfrac = NTPfrac + cameraDelayFixFrac;
+ }
+ }
+ _lastSendReport[0] = (BackTimedNTPsec <<16) + (BackTimedNTPfrac >> 16);
+
+ // RTP timestamp
+ // This should have a ramdom start value added
+ // RTP is counted from NTP not the acctual RTP
+ // This reflects the perfect RTP time
+ // we solve this by initiating RTP to our NTP :)
+
+ WebRtc_UWord32 freqHz = 90000; // For video
+ if(_audio)
+ {
+ freqHz = _rtpRtcp.CurrentSendFrequencyHz();
+ RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freqHz);
+ }
+ else // video
+ {
+ // used to be (WebRtc_UWord32)(((float)BackTimedNTPfrac/(float)FRAC)* 90000)
+ WebRtc_UWord32 tmp = 9*(BackTimedNTPfrac/429496);
+ RTPtime = BackTimedNTPsec*freqHz + tmp;
+ }
+
+
+
+
+ // Add sender data
+ // Save for our length field
+ pos++;
+ pos++;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+ // NTP
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPsec);
+ pos += 4;
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPfrac);
+ pos += 4;
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime);
+ pos += 4;
+
+ //sender's packet count
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.PacketCountSent());
+ pos += 4;
+
+ //sender's octet count
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.ByteCountSent());
+ pos += 4;
+
+ WebRtc_UWord8 numberOfReportBlocks = 0;
+ WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac);
+ if(retVal < 0)
+ {
+ //
+ return retVal ;
+ }
+ rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
+
+ WebRtc_UWord16 len = WebRtc_UWord16((pos/4) -1);
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len);
+ return 0;
+}
+
+
+WebRtc_Word32 RTCPSender::BuildSDEC(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos) {
+ size_t lengthCname = strlen(_CNAME);
+ assert(lengthCname < RTCP_CNAME_SIZE);
+
+ // sanity
+ if(pos + 12 + lengthCname >= IP_PACKET_SIZE) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -2;
+ }
+ // SDEC Source Description
+
+ // We always need to add SDES CNAME
+ rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(0x80 + 1 + _csrcCNAMEs.size());
+ rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(202);
+
+ // handle SDES length later on
+ WebRtc_UWord32 SDESLengthPos = pos;
+ pos++;
+ pos++;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // CNAME = 1
+ rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1);
+
+ //
+ rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(lengthCname);
+
+ WebRtc_UWord16 SDESLength = 10;
+
+ memcpy(&rtcpbuffer[pos], _CNAME, lengthCname);
+ pos += lengthCname;
+ SDESLength += (WebRtc_UWord16)lengthCname;
+
+ WebRtc_UWord16 padding = 0;
+ // We must have a zero field even if we have an even multiple of 4 bytes
+ if ((pos % 4) == 0) {
+ padding++;
+ rtcpbuffer[pos++]=0;
+ }
+ while ((pos % 4) != 0) {
+ padding++;
+ rtcpbuffer[pos++]=0;
+ }
+ SDESLength += padding;
+
+ std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>::iterator it =
+ _csrcCNAMEs.begin();
+
+ for(; it != _csrcCNAMEs.end(); it++) {
+ RTCPCnameInformation* cname = it->second;
+ WebRtc_UWord32 SSRC = it->first;
+
+ // Add SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC);
+ pos += 4;
+
+ // CNAME = 1
+ rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1);
+
+ size_t length = strlen(cname->name);
+ assert(length < RTCP_CNAME_SIZE);
+
+ rtcpbuffer[pos++]= static_cast<WebRtc_UWord8>(length);
+ SDESLength += 6;
+
+ memcpy(&rtcpbuffer[pos],cname->name, length);
+
+ pos += length;
+ SDESLength += length;
+ WebRtc_UWord16 padding = 0;
+
+ // We must have a zero field even if we have an even multiple of 4 bytes
+ if((pos % 4) == 0){
+ padding++;
+ rtcpbuffer[pos++]=0;
+ }
+ while((pos % 4) != 0){
+ padding++;
+ rtcpbuffer[pos++] = 0;
+ }
+ SDESLength += padding;
+ }
+ // in 32-bit words minus one and we don't count the header
+ WebRtc_UWord16 buffer_length = (SDESLength / 4) - 1;
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos,
+ buffer_length);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildRR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac,
+ const RTCPReportBlock* received)
+{
+ // sanity one block
+ if(pos + 32 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ WebRtc_UWord32 posNumberOfReportBlocks = pos;
+
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)201;
+
+ // Save for our length field
+ pos++;
+ pos++;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ WebRtc_UWord8 numberOfReportBlocks = 0;
+ WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac);
+ if(retVal < 0)
+ {
+ return retVal;
+ }
+ rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
+
+ WebRtc_UWord16 len = WebRtc_UWord16((pos)/4 -1);
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len);
+ return 0;
+}
+
+// From RFC 5450: Transmission Time Offsets in RTP Streams.
+// 0 1 2 3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// hdr |V=2|P| RC | PT=IJ=195 | length |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | inter-arrival jitter |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// . .
+// . .
+// . .
+// | inter-arrival jitter |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// If present, this RTCP packet must be placed after a receiver report
+// (inside a compound RTCP packet), and MUST have the same value for RC
+// (reception report count) as the receiver report.
+
+WebRtc_Word32
+RTCPSender::BuildExtendedJitterReport(
+ WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 jitterTransmissionTimeOffset)
+{
+ if (_reportBlocks.size() > 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented.");
+ return 0;
+ }
+
+ // sanity
+ if(pos + 8 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add picture loss indicator
+ WebRtc_UWord8 RC = 1;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + RC;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)195;
+
+ // Used fixed length of 2
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(1);
+
+ // Add inter-arrival jitter
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos,
+ jitterTransmissionTimeOffset);
+ pos += 4;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // sanity
+ if(pos + 12 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add picture loss indicator
+ WebRtc_UWord8 FMT = 1;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)206;
+
+ //Used fixed length of 2
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(2);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add the remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::BuildFIR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ bool repeat) {
+ // sanity
+ if(pos + 20 >= IP_PACKET_SIZE) {
+ return -2;
+ }
+ if (!repeat) {
+ _sequenceNumberFIR++; // do not increase if repetition
+ }
+
+ // add full intra request indicator
+ WebRtc_UWord8 FMT = 4;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)206;
+
+ //Length of 4
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(4);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
+ pos += 4;
+
+ // RFC 5104 4.3.1.2. Semantics
+ // SSRC of media source
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+
+ // Additional Feedback Control Information (FCI)
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
+ pos += 4;
+
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_sequenceNumberFIR);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ rtcpbuffer[pos++] = (WebRtc_UWord8)0;
+ return 0;
+}
+
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | First | Number | PictureID |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+WebRtc_Word32
+RTCPSender::BuildSLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord8 pictureID)
+{
+ // sanity
+ if(pos + 16 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add slice loss indicator
+ WebRtc_UWord8 FMT = 2;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)206;
+
+ //Used fixed length of 3
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(3);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add the remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ // Add first, number & picture ID 6 bits
+ // first = 0, 13 - bits
+ // number = 0x1fff, 13 - bits only ones for now
+ WebRtc_UWord32 sliField = (0x1fff << 6)+ (0x3f & pictureID);
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField);
+ pos += 4;
+ return 0;
+}
+
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | PB |0| Payload Type| Native RPSI bit string |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | defined per codec ... | Padding (0) |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+/*
+* Note: not generic made for VP8
+*/
+WebRtc_Word32
+RTCPSender::BuildRPSI(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord64 pictureID,
+ const WebRtc_UWord8 payloadType)
+{
+ // sanity
+ if(pos + 24 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add Reference Picture Selection Indication
+ WebRtc_UWord8 FMT = 3;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)206;
+
+ // calc length
+ WebRtc_UWord32 bitsRequired = 7;
+ WebRtc_UWord8 bytesRequired = 1;
+ while((pictureID>>bitsRequired) > 0)
+ {
+ bitsRequired += 7;
+ bytesRequired++;
+ }
+
+ WebRtc_UWord8 size = 3;
+ if(bytesRequired > 6)
+ {
+ size = 5;
+ } else if(bytesRequired > 2)
+ {
+ size = 4;
+ }
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=size;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add the remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ // calc padding length
+ WebRtc_UWord8 paddingBytes = 4-((2+bytesRequired)%4);
+ if(paddingBytes == 4)
+ {
+ paddingBytes = 0;
+ }
+ // add padding length in bits
+ rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24
+ pos++;
+
+ // add payload type
+ rtcpbuffer[pos] = payloadType;
+ pos++;
+
+ // add picture ID
+ for(int i = bytesRequired-1; i > 0; i--)
+ {
+ rtcpbuffer[pos] = 0x80 | WebRtc_UWord8(pictureID >> (i*7));
+ pos++;
+ }
+ // add last byte of picture ID
+ rtcpbuffer[pos] = WebRtc_UWord8(pictureID & 0x7f);
+ pos++;
+
+ // add padding
+ for(int j = 0; j <paddingBytes; j++)
+ {
+ rtcpbuffer[pos] = 0;
+ pos++;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // sanity
+ if(pos + 20 + 4 * _lengthRembSSRC >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add application layer feedback
+ WebRtc_UWord8 FMT = 15;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)206;
+
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=_lengthRembSSRC + 4;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Remote SSRC must be 0
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0);
+ pos += 4;
+
+ rtcpbuffer[pos++]='R';
+ rtcpbuffer[pos++]='E';
+ rtcpbuffer[pos++]='M';
+ rtcpbuffer[pos++]='B';
+
+ rtcpbuffer[pos++] = _lengthRembSSRC;
+ // 6 bit Exp
+ // 18 bit mantissa
+ WebRtc_UWord8 brExp = 0;
+ for(WebRtc_UWord32 i=0; i<64; i++)
+ {
+ if(_rembBitrate <= ((WebRtc_UWord32)262143 << i))
+ {
+ brExp = i;
+ break;
+ }
+ }
+ const WebRtc_UWord32 brMantissa = (_rembBitrate >> brExp);
+ rtcpbuffer[pos++]=(WebRtc_UWord8)((brExp << 2) + ((brMantissa >> 16) & 0x03));
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa >> 8);
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa);
+
+ for (int i = 0; i < _lengthRembSSRC; i++)
+ {
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]);
+ pos += 4;
+ }
+ return 0;
+}
+
+WebRtc_UWord32
+RTCPSender::CalculateNewTargetBitrate(WebRtc_UWord32 RTT)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ WebRtc_UWord32 target_bitrate =
+ _remoteRateControl.UpdateBandwidthEstimate(RTT, _clock.GetTimeInMS());
+ _tmmbr_Send = target_bitrate / 1000;
+ return target_bitrate;
+}
+
+WebRtc_UWord32 RTCPSender::LatestBandwidthEstimate() const {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _remoteRateControl.LatestEstimate();
+}
+
+bool
+RTCPSender::ValidBitrateEstimate() const {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _remoteRateControl.ValidEstimate();
+}
+
+WebRtc_Word32
+RTCPSender::BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate
+ // If the sender is an owner of the TMMBN -> send TMMBR
+ // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
+
+ // get current bounding set from RTCP receiver
+ bool tmmbrOwner = false;
+ TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet(); // store in candidateSet, allocates one extra slot
+
+ // holding _criticalSectionRTCPSender while calling RTCPreceiver which will accuire _criticalSectionRTCPReceiver
+ // is a potental deadlock but since RTCPreceiver is not doing the revese we should be fine
+ WebRtc_Word32 lengthOfBoundingSet = _rtpRtcp.BoundingSet(tmmbrOwner, candidateSet);
+
+ if(lengthOfBoundingSet > 0)
+ {
+ for (WebRtc_Word32 i = 0; i < lengthOfBoundingSet; i++)
+ {
+ if( candidateSet->ptrTmmbrSet[i] == _tmmbr_Send &&
+ candidateSet->ptrPacketOHSet[i] == _packetOH_Send)
+ {
+ // do not send the same tuple
+ return 0;
+ }
+ }
+ if(!tmmbrOwner)
+ {
+ // use received bounding set as candidate set
+ // add current tuple
+ candidateSet->ptrTmmbrSet[lengthOfBoundingSet] = _tmmbr_Send;
+ candidateSet->ptrPacketOHSet[lengthOfBoundingSet] = _packetOH_Send;
+ candidateSet->ptrSsrcSet[lengthOfBoundingSet] = _SSRC;
+ int numCandidates = lengthOfBoundingSet+ 1;
+
+ // find bounding set
+ TMMBRSet* boundingSet = NULL;
+ int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet);
+ if(numBoundingSet > 0 || numBoundingSet <= numCandidates)
+ {
+ tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet);
+ }
+ if(!tmmbrOwner)
+ {
+ // did not enter bounding set, no meaning to send this request
+ return 0;
+ }
+ }
+ }
+
+ if(_tmmbr_Send)
+ {
+ // sanity
+ if(pos + 20 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ // add TMMBR indicator
+ WebRtc_UWord8 FMT = 3;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)205;
+
+ //Length of 4
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(4);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // RFC 5104 4.2.1.2. Semantics
+
+ // SSRC of media source
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+
+ // Additional Feedback Control Information (FCI)
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ WebRtc_UWord32 bitRate = _tmmbr_Send*1000;
+ WebRtc_UWord32 mmbrExp = 0;
+ for(WebRtc_UWord32 i=0;i<64;i++)
+ {
+ if(bitRate <= ((WebRtc_UWord32)131071 << i))
+ {
+ mmbrExp = i;
+ break;
+ }
+ }
+ WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp);
+
+ rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7);
+ rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01));
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(_packetOH_Send);
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend();
+ if(boundingSet == NULL)
+ {
+ return -1;
+ }
+ // sanity
+ if(pos + 12 + boundingSet->lengthOfSet*8 >= IP_PACKET_SIZE)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -2;
+ }
+ WebRtc_UWord8 FMT = 4;
+ // add TMMBN indicator
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)205;
+
+ //Add length later
+ int posLength = pos;
+ pos++;
+ pos++;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // RFC 5104 4.2.2.2. Semantics
+
+ // SSRC of media source
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+
+ // Additional Feedback Control Information (FCI)
+ int numBoundingSet = 0;
+ for(WebRtc_UWord32 n=0; n< boundingSet->lengthOfSet; n++)
+ {
+ if (boundingSet->ptrTmmbrSet[n] > 0)
+ {
+ WebRtc_UWord32 tmmbrSSRC = boundingSet->ptrSsrcSet[n];
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC);
+ pos += 4;
+
+ WebRtc_UWord32 bitRate = boundingSet->ptrTmmbrSet[n] * 1000;
+ WebRtc_UWord32 mmbrExp = 0;
+ for(int i=0; i<64; i++)
+ {
+ if(bitRate <= ((WebRtc_UWord32)131071 << i))
+ {
+ mmbrExp = i;
+ break;
+ }
+ }
+ WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp);
+ WebRtc_UWord32 measuredOH = boundingSet->ptrPacketOHSet[n];
+
+ rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7);
+ rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01));
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(measuredOH);
+ numBoundingSet++;
+ }
+ }
+ WebRtc_UWord16 length= (WebRtc_UWord16)(2+2*numBoundingSet);
+ rtcpbuffer[posLength++]=(WebRtc_UWord8)(length>>8);
+ rtcpbuffer[posLength]=(WebRtc_UWord8)(length);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // sanity
+ if(_appData == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
+ return -1;
+ }
+ if(pos + 12 + _appLength >= IP_PACKET_SIZE)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -2;
+ }
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + _appSubType;
+
+ // Add APP ID
+ rtcpbuffer[pos++]=(WebRtc_UWord8)204;
+
+ WebRtc_UWord16 length = (_appLength>>2) + 2; // include SSRC and name
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(length>>8);
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(length);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add our application name
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName);
+ pos += 4;
+
+ // Add the data
+ memcpy(rtcpbuffer +pos, _appData,_appLength);
+ pos += _appLength;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_Word32 nackSize,
+ const WebRtc_UWord16* nackList)
+{
+ // sanity
+ if(pos + 16 >= IP_PACKET_SIZE)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -2;
+ }
+
+ // int size, WebRtc_UWord16* nackList
+ // add nack list
+ WebRtc_UWord8 FMT = 1;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)205;
+
+ rtcpbuffer[pos++]=(WebRtc_UWord8) 0;
+ int nackSizePos = pos;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(3); //setting it to one kNACK signal as default
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add the remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ // add the list
+ int i = 0;
+ int numOfNackFields = 0;
+ while(nackSize > i && numOfNackFields < 253)
+ {
+ WebRtc_UWord16 nack = nackList[i];
+ // put dow our sequence number
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack);
+ pos += 2;
+
+ i++;
+ numOfNackFields++;
+ if(nackSize > i)
+ {
+ bool moreThan16Away = (WebRtc_UWord16(nack+16) < nackList[i])?true: false;
+ if(!moreThan16Away)
+ {
+ // check for a wrap
+ if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
+ {
+ // wrap
+ moreThan16Away = true;
+ }
+ }
+ if(moreThan16Away)
+ {
+ // next is more than 16 away
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ } else
+ {
+ // build our bitmask
+ WebRtc_UWord16 bitmask = 0;
+
+ bool within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false;
+ if(within16Away)
+ {
+ // check for a wrap
+ if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
+ {
+ // wrap
+ within16Away = false;
+ }
+ }
+
+ while( nackSize > i && within16Away)
+ {
+ WebRtc_Word16 shift = (nackList[i]-nack)-1;
+ assert(!(shift > 15) && !(shift < 0));
+
+ bitmask += (1<< shift);
+ i++;
+ if(nackSize > i)
+ {
+ within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false;
+ if(within16Away)
+ {
+ // check for a wrap
+ if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
+ {
+ // wrap
+ within16Away = false;
+ }
+ }
+ }
+ }
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, bitmask);
+ pos += 2;
+ }
+ // sanity do we have room from one more 4 byte block?
+ if(pos + 4 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ } else
+ {
+ // no more in the list
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ }
+ }
+ rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // sanity
+ if(pos + 8 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+ if(_includeCSRCs)
+ {
+ // Add a bye packet
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs
+ rtcpbuffer[pos++]=(WebRtc_UWord8)203;
+
+ // length
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)(1 + _CSRCs);
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // add CSRCs
+ for(int i = 0; i < _CSRCs; i++)
+ {
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]);
+ pos += 4;
+ }
+ } else
+ {
+ // Add a bye packet
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1; // number of SSRC+CSRCs
+ rtcpbuffer[pos++]=(WebRtc_UWord8)203;
+
+ // length
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)1;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
+{
+ // sanity
+ if(pos + 44 >= IP_PACKET_SIZE)
+ {
+ return -2;
+ }
+
+ // Add XR header
+ rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
+ rtcpbuffer[pos++]=(WebRtc_UWord8)207;
+
+ WebRtc_UWord32 XRLengthPos = pos;
+
+ // handle length later on
+ pos++;
+ pos++;
+
+ // Add our own SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
+ pos += 4;
+
+ // Add a VoIP metrics block
+ rtcpbuffer[pos++]=7;
+ rtcpbuffer[pos++]=0;
+ rtcpbuffer[pos++]=0;
+ rtcpbuffer[pos++]=8;
+
+ // Add the remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ rtcpbuffer[pos++] = _xrVoIPMetric.lossRate;
+ rtcpbuffer[pos++] = _xrVoIPMetric.discardRate;
+ rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity;
+ rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity;
+
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration);
+
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay);
+
+ rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel;
+ rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel;
+ rtcpbuffer[pos++] = _xrVoIPMetric.RERL;
+ rtcpbuffer[pos++] = _xrVoIPMetric.Gmin;
+
+ rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor;
+ rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor;
+ rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ;
+ rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ;
+
+ rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig;
+ rtcpbuffer[pos++] = 0; // reserved
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal);
+
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax >> 8);
+ rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax);
+
+ rtcpbuffer[XRLengthPos]=(WebRtc_UWord8)(0);
+ rtcpbuffer[XRLengthPos+1]=(WebRtc_UWord8)(10);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
+ const WebRtc_Word32 nackSize, // NACK
+ const WebRtc_UWord16* nackList, // NACK
+ const bool repeat, // FIR
+ const WebRtc_UWord64 pictureID) // SLI & RPSI
+{
+ WebRtc_UWord32 rtcpPacketTypeFlags = packetTypeFlags;
+ WebRtc_UWord32 pos = 0;
+ WebRtc_UWord8 rtcpbuffer[IP_PACKET_SIZE];
+
+ do // only to be able to use break :) (and the critsect must be inside its own scope)
+ {
+ // collect the received information
+ RTCPReportBlock received;
+ bool hasReceived = false;
+ WebRtc_UWord32 NTPsec = 0;
+ WebRtc_UWord32 NTPfrac = 0;
+ bool rtcpCompound = false;
+ WebRtc_UWord32 jitterTransmissionOffset = 0;
+
+ {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if(_method == kRtcpOff)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "%s invalid state", __FUNCTION__);
+ return -1;
+ }
+ rtcpCompound = (_method == kRtcpCompound) ? true : false;
+ }
+
+ if (rtcpCompound ||
+ rtcpPacketTypeFlags & kRtcpReport ||
+ rtcpPacketTypeFlags & kRtcpSr ||
+ rtcpPacketTypeFlags & kRtcpRr)
+ {
+ // get statistics from our RTPreceiver outside critsect
+ if(_rtpRtcp.ReportBlockStatistics(&received.fractionLost,
+ &received.cumulativeLost,
+ &received.extendedHighSeqNum,
+ &received.jitter,
+ &jitterTransmissionOffset) == 0)
+ {
+ hasReceived = true;
+
+ WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
+ WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
+ WebRtc_UWord32 remoteSR = 0;
+
+ // ok even if we have not received a SR, we will send 0 in that case
+ _rtpRtcp.LastReceivedNTP(lastReceivedRRNTPsecs,
+ lastReceivedRRNTPfrac,
+ remoteSR);
+
+ // get our NTP as late as possible to avoid a race
+ _clock.CurrentNTP(NTPsec, NTPfrac);
+
+ // Delay since last received report
+ WebRtc_UWord32 delaySinceLastReceivedSR = 0;
+ if((lastReceivedRRNTPsecs !=0) || (lastReceivedRRNTPfrac !=0))
+ {
+ // get the 16 lowest bits of seconds and the 16 higest bits of fractions
+ WebRtc_UWord32 now=NTPsec&0x0000FFFF;
+ now <<=16;
+ now += (NTPfrac&0xffff0000)>>16;
+
+ WebRtc_UWord32 receiveTime = lastReceivedRRNTPsecs&0x0000FFFF;
+ receiveTime <<=16;
+ receiveTime += (lastReceivedRRNTPfrac&0xffff0000)>>16;
+
+ delaySinceLastReceivedSR = now-receiveTime;
+ }
+ received.delaySinceLastSR = delaySinceLastReceivedSR;
+ received.lastSR = remoteSR;
+ } else
+ {
+ // we need to send our NTP even if we dont have received any reports
+ _clock.CurrentNTP(NTPsec, NTPfrac);
+ }
+ }
+
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if(_TMMBR ) // attach TMMBR to send and receive reports
+ {
+ rtcpPacketTypeFlags |= kRtcpTmmbr;
+ }
+ if(_appSend)
+ {
+ rtcpPacketTypeFlags |= kRtcpApp;
+ _appSend = false;
+ }
+ if(_REMB && _sendREMB)
+ {
+ // Always attach REMB to SR if that is configured. Note that REMB is
+ // only sent on one of the RTP modules in the REMB group.
+ rtcpPacketTypeFlags |= kRtcpRemb;
+ }
+ if(_xrSendVoIPMetric)
+ {
+ rtcpPacketTypeFlags |= kRtcpXrVoipMetric;
+ _xrSendVoIPMetric = false;
+ }
+ if(_sendTMMBN) // set when having received a TMMBR
+ {
+ rtcpPacketTypeFlags |= kRtcpTmmbn;
+ _sendTMMBN = false;
+ }
+
+ if(_method == kRtcpCompound)
+ {
+ if(_sending)
+ {
+ rtcpPacketTypeFlags |= kRtcpSr;
+ } else
+ {
+ rtcpPacketTypeFlags |= kRtcpRr;
+ }
+ if (_IJ && hasReceived)
+ {
+ rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
+ }
+ } else if(_method == kRtcpNonCompound)
+ {
+ if(rtcpPacketTypeFlags & kRtcpReport)
+ {
+ if(_sending)
+ {
+ rtcpPacketTypeFlags |= kRtcpSr;
+ } else
+ {
+ rtcpPacketTypeFlags |= kRtcpRr;
+ }
+ }
+ }
+ if( rtcpPacketTypeFlags & kRtcpRr ||
+ rtcpPacketTypeFlags & kRtcpSr)
+ {
+ // generate next time to send a RTCP report
+ // seeded from RTP constructor
+ WebRtc_Word32 random = rand() % 1000;
+ WebRtc_Word32 timeToNext = RTCP_INTERVAL_AUDIO_MS;
+
+ if(_audio)
+ {
+ timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) + (RTCP_INTERVAL_AUDIO_MS*random/1000);
+ }else
+ {
+ WebRtc_UWord32 minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+ if(_sending)
+ {
+ // calc bw for video 360/sendBW in kbit/s
+ WebRtc_UWord32 sendBitrateKbit = 0;
+ WebRtc_UWord32 videoRate = 0;
+ WebRtc_UWord32 fecRate = 0;
+ WebRtc_UWord32 nackRate = 0;
+ _rtpRtcp.BitrateSent(&sendBitrateKbit,
+ &videoRate,
+ &fecRate,
+ &nackRate);
+ sendBitrateKbit /= 1000;
+ if(sendBitrateKbit != 0)
+ {
+ minIntervalMs = 360000/sendBitrateKbit;
+ }
+ }
+ if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
+ {
+ minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
+ }
+ timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000);
+ }
+ _nextTimeToSendRTCP = _clock.GetTimeInMS() + timeToNext;
+ }
+
+ // if the data does not fitt in the packet we fill it as much as possible
+ WebRtc_Word32 buildVal = 0;
+
+ if(rtcpPacketTypeFlags & kRtcpSr)
+ {
+ if(hasReceived)
+ {
+ buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac, &received);
+ } else
+ {
+ buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac);
+ }
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ buildVal = BuildSDEC(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+
+ }else if(rtcpPacketTypeFlags & kRtcpRr)
+ {
+ if(hasReceived)
+ {
+ buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac,&received);
+ }else
+ {
+ buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac);
+ }
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ // only of set
+ if(_CNAME[0] != 0)
+ {
+ buildVal = BuildSDEC(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+ }
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset)
+ {
+ // If present, this RTCP packet must be placed after a
+ // receiver report.
+ buildVal = BuildExtendedJitterReport(rtcpbuffer,
+ pos,
+ jitterTransmissionOffset);
+ if(buildVal == -1)
+ {
+ return -1; // error
+ }
+ else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpPli)
+ {
+ buildVal = BuildPLI(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpFir)
+ {
+ buildVal = BuildFIR(rtcpbuffer, pos, repeat);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpSli)
+ {
+ buildVal = BuildSLI(rtcpbuffer, pos, (WebRtc_UWord8)pictureID);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpRpsi)
+ {
+ const WebRtc_Word8 payloadType = _rtpRtcp.SendPayloadType();
+ if(payloadType == -1)
+ {
+ return -1;
+ }
+ buildVal = BuildRPSI(rtcpbuffer, pos, pictureID, (WebRtc_UWord8)payloadType);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpRemb)
+ {
+ buildVal = BuildREMB(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpBye)
+ {
+ buildVal = BuildBYE(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpApp)
+ {
+ buildVal = BuildAPP(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpTmmbr)
+ {
+ buildVal = BuildTMMBR(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpTmmbn)
+ {
+ buildVal = BuildTMMBN(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpNack)
+ {
+ buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ if(rtcpPacketTypeFlags & kRtcpXrVoipMetric)
+ {
+ buildVal = BuildVoIPMetric(rtcpbuffer, pos);
+ if(buildVal == -1)
+ {
+ return -1; // error
+
+ }else if(buildVal == -2)
+ {
+ break; // out of buffer
+ }
+ }
+ }while (false);
+ // Sanity don't send empty packets.
+ if (pos == 0)
+ {
+ return -1;
+ }
+ return SendToNetwork(rtcpbuffer, (WebRtc_UWord16)pos);
+}
+
+WebRtc_Word32
+RTCPSender::SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 length)
+{
+ CriticalSectionScoped lock(_criticalSectionTransport);
+ if(_cbTransport)
+ {
+ if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0)
+ {
+ return 0;
+ }
+ }
+ return -1;
+}
+
+WebRtc_Word32
+RTCPSender::SetCSRCStatus(const bool include)
+{
+ _includeCSRCs = include;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength)
+{
+ if(arrLength > kRtpCsrcSize)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ assert(false);
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ for(int i = 0; i < arrLength;i++)
+ {
+ _CSRC[i] = arrOfCSRC[i];
+ }
+ _CSRCs = arrLength;
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SetApplicationSpecificData(const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 length)
+{
+ if(length %4 != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if(_appData)
+ {
+ delete [] _appData;
+ }
+
+ _appSend = true;
+ _appSubType = subType;
+ _appName = name;
+ _appData = new WebRtc_UWord8[length];
+ _appLength = length;
+ memcpy(_appData, data, length);
+ return 0;
+}
+
+WebRtc_Word32
+RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric));
+
+ _xrSendVoIPMetric = true;
+ return 0;
+}
+
+// called under critsect _criticalSectionRTCPSender
+WebRtc_Word32 RTCPSender::AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ WebRtc_UWord8& numberOfReportBlocks,
+ const RTCPReportBlock* received,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac) {
+ // sanity one block
+ if(pos + 24 >= IP_PACKET_SIZE) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ numberOfReportBlocks = _reportBlocks.size();
+ if (received) {
+ // add our multiple RR to numberOfReportBlocks
+ numberOfReportBlocks++;
+ }
+ if (received) {
+ // answer to the one that sends to me
+ _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac);
+
+ // Remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
+ pos += 4;
+
+ // fraction lost
+ rtcpbuffer[pos++]=received->fractionLost;
+
+ // cumulative loss
+ ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos,
+ received->cumulativeLost);
+ pos += 3;
+ // extended highest seq_no, contain the highest sequence number received
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ received->extendedHighSeqNum);
+ pos += 4;
+
+ //Jitter
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->jitter);
+ pos += 4;
+
+ // Last SR timestamp, our NTP time when we received the last report
+ // This is the value that we read from the send report packet not when we
+ // received it...
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->lastSR);
+ pos += 4;
+
+ // Delay since last received report,time since we received the report
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ received->delaySinceLastSR);
+ pos += 4;
+ }
+ if ((pos + _reportBlocks.size() * 24) >= IP_PACKET_SIZE) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
+ _reportBlocks.begin();
+
+ for (; it != _reportBlocks.end(); it++) {
+ // we can have multiple report block in a conference
+ WebRtc_UWord32 remoteSSRC = it->first;
+ RTCPReportBlock* reportBlock = it->second;
+ if (reportBlock) {
+ // Remote SSRC
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSSRC);
+ pos += 4;
+
+ // fraction lost
+ rtcpbuffer[pos++] = reportBlock->fractionLost;
+
+ // cumulative loss
+ ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos,
+ reportBlock->cumulativeLost);
+ pos += 3;
+
+ // extended highest seq_no, contain the highest sequence number received
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ reportBlock->extendedHighSeqNum);
+ pos += 4;
+
+ //Jitter
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ reportBlock->jitter);
+ pos += 4;
+
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ reportBlock->lastSR);
+ pos += 4;
+
+ ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
+ reportBlock->delaySinceLastSR);
+ pos += 4;
+ }
+ }
+ return pos;
+}
+
+// no callbacks allowed inside this function
+WebRtc_Word32
+RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
+ const WebRtc_UWord32 maxBitrateKbit)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+
+ if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit))
+ {
+ _sendTMMBN = true;
+ return 0;
+ }
+ return -1;
+}
+
+WebRtc_Word32
+RTCPSender::RequestTMMBR(WebRtc_UWord32 estimatedBW, WebRtc_UWord32 packetOH)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if(_TMMBR)
+ {
+ _tmmbr_Send = estimatedBW;
+ _packetOH_Send = packetOH;
+
+ return 0;
+ }
+ return -1;
+}
+
+RateControlRegion
+RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse)
+{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ return _remoteRateControl.Update(rateControlInput, firstOverUse,
+ _clock.GetTimeInMS());
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h
new file mode 100644
index 0000000..966b64f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h
@@ -0,0 +1,268 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+
+#include <map>
+
+#include "typedefs.h"
+#include "rtcp_utility.h"
+#include "rtp_utility.h"
+#include "rtp_rtcp_defines.h"
+#include "remote_rate_control.h"
+#include "tmmbr_help.h"
+
+namespace webrtc {
+
+class ModuleRtpRtcpImpl;
+
+class RTCPSender
+{
+public:
+ RTCPSender(const WebRtc_Word32 id, const bool audio,
+ RtpRtcpClock* clock, ModuleRtpRtcpImpl* owner);
+ virtual ~RTCPSender();
+
+ void ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 Init();
+
+ WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
+
+ RTCPMethod Status() const;
+ WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
+
+ bool Sending() const;
+ WebRtc_Word32 SetSendingStatus(const bool enabled); // combine the functions
+
+ WebRtc_Word32 SetNackStatus(const bool enable);
+
+ void SetSSRC( const WebRtc_UWord32 ssrc);
+
+ WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
+
+ WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
+
+ WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
+ WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
+
+ WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]);
+
+ WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
+
+ WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
+
+ bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const;
+
+ WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
+
+ WebRtc_Word32 SendRTCP(const WebRtc_UWord32 rtcpPacketTypeFlags,
+ const WebRtc_Word32 nackSize = 0,
+ const WebRtc_UWord16* nackList = 0,
+ const bool repeat = false,
+ const WebRtc_UWord64 pictureID = 0);
+
+ WebRtc_Word32 AddReportBlock(const WebRtc_UWord32 SSRC,
+ const RTCPReportBlock* receiveBlock);
+
+ WebRtc_Word32 RemoveReportBlock(const WebRtc_UWord32 SSRC);
+
+ /*
+ * REMB
+ */
+ bool REMB() const;
+
+ WebRtc_Word32 SetREMBStatus(const bool enable);
+
+ WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
+ const WebRtc_UWord8 numberOfSSRC,
+ const WebRtc_UWord32* SSRC);
+
+ bool SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer);
+
+ void UpdateRemoteBitrateEstimate(unsigned int target_bitrate);
+
+ void ReceivedRemb(unsigned int estimated_bitrate);
+
+ /*
+ * TMMBR
+ */
+ bool TMMBR() const;
+
+ WebRtc_Word32 SetTMMBRStatus(const bool enable);
+
+ WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
+ const WebRtc_UWord32 maxBitrateKbit);
+
+ WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
+ const WebRtc_UWord32 packetOH);
+
+ /*
+ * Extended jitter report
+ */
+ bool IJ() const;
+
+ WebRtc_Word32 SetIJStatus(const bool enable);
+
+ /*
+ *
+ */
+
+ WebRtc_Word32 SetApplicationSpecificData(const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 length);
+
+ WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
+
+ WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength);
+
+ WebRtc_Word32 SetCSRCStatus(const bool include);
+
+ /*
+ * New bandwidth estimation
+ */
+
+ RateControlRegion UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse);
+
+ WebRtc_UWord32 CalculateNewTargetBitrate(WebRtc_UWord32 RTT);
+
+ WebRtc_UWord32 LatestBandwidthEstimate() const;
+
+ // Returns true if there is a valid estimate of the incoming bitrate, false
+ // otherwise.
+ bool ValidBitrateEstimate() const;
+
+private:
+ WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 length);
+
+ void UpdatePacketRate();
+
+ WebRtc_Word32 AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ WebRtc_UWord8& numberOfReportBlocks,
+ const RTCPReportBlock* received,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac);
+
+ WebRtc_Word32 BuildSR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac,
+ const RTCPReportBlock* received = NULL);
+
+ WebRtc_Word32 BuildRR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 NTPsec,
+ const WebRtc_UWord32 NTPfrac,
+ const RTCPReportBlock* received = NULL);
+
+ WebRtc_Word32 BuildExtendedJitterReport(
+ WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord32 jitterTransmissionTimeOffset);
+
+ WebRtc_Word32 BuildSDEC(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
+ WebRtc_Word32 BuildFIR(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ bool repeat);
+ WebRtc_Word32 BuildSLI(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord8 pictureID);
+ WebRtc_Word32 BuildRPSI(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_UWord64 pictureID,
+ const WebRtc_UWord8 payloadType);
+
+ WebRtc_Word32 BuildNACK(WebRtc_UWord8* rtcpbuffer,
+ WebRtc_UWord32& pos,
+ const WebRtc_Word32 nackSize,
+ const WebRtc_UWord16* nackList);
+
+private:
+ WebRtc_Word32 _id;
+ const bool _audio;
+ RtpRtcpClock& _clock;
+ RTCPMethod _method;
+
+ ModuleRtpRtcpImpl& _rtpRtcp;
+
+ CriticalSectionWrapper* _criticalSectionTransport;
+ Transport* _cbTransport;
+
+ CriticalSectionWrapper* _criticalSectionRTCPSender;
+ bool _usingNack;
+ bool _sending;
+ bool _sendTMMBN;
+ bool _REMB;
+ bool _sendREMB;
+ bool _TMMBR;
+ bool _IJ;
+
+ WebRtc_UWord32 _nextTimeToSendRTCP;
+
+ WebRtc_UWord32 _SSRC;
+ WebRtc_UWord32 _remoteSSRC; // SSRC that we receive on our RTP channel
+ char _CNAME[RTCP_CNAME_SIZE];
+
+ std::map<WebRtc_UWord32, RTCPReportBlock*> _reportBlocks;
+ std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
+
+ WebRtc_Word32 _cameraDelayMS;
+
+ // Sent
+ WebRtc_UWord32 _lastSendReport[RTCP_NUMBER_OF_SR]; // allow packet loss and RTT above 1 sec
+ WebRtc_UWord32 _lastRTCPTime[RTCP_NUMBER_OF_SR];
+
+ // send CSRCs
+ WebRtc_UWord8 _CSRCs;
+ WebRtc_UWord32 _CSRC[kRtpCsrcSize];
+ bool _includeCSRCs;
+
+ // Full intra request
+ WebRtc_UWord8 _sequenceNumberFIR;
+
+ // REMB
+ WebRtc_UWord8 _lengthRembSSRC;
+ WebRtc_UWord8 _sizeRembSSRC;
+ WebRtc_UWord32* _rembSSRC;
+ WebRtc_UWord32 _rembBitrate;
+ RtpRemoteBitrateObserver* _bitrate_observer;
+
+ TMMBRHelp _tmmbrHelp;
+ WebRtc_UWord32 _tmmbr_Send;
+ WebRtc_UWord32 _packetOH_Send;
+ RemoteRateControl _remoteRateControl;
+
+ // APP
+ bool _appSend;
+ WebRtc_UWord8 _appSubType;
+ WebRtc_UWord32 _appName;
+ WebRtc_UWord8* _appData;
+ WebRtc_UWord16 _appLength;
+
+ // XR VoIP metric
+ bool _xrSendVoIPMetric;
+ RTCPVoIPMetric _xrVoIPMetric;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc
new file mode 100644
index 0000000..7b5b057
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc
@@ -0,0 +1,178 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes unit tests for the RTCPSender.
+ */
+
+#include <gtest/gtest.h>
+
+#include "common_types.h"
+#include "rtp_utility.h"
+#include "rtcp_sender.h"
+#include "rtcp_receiver.h"
+#include "rtp_rtcp_impl.h"
+
+namespace webrtc {
+
+void CreateRtpPacket(const bool marker_bit, const WebRtc_UWord8 payload,
+ const WebRtc_UWord16 seq_num, const WebRtc_UWord32 timestamp,
+ const WebRtc_UWord32 ssrc, WebRtc_UWord8* array,
+ WebRtc_UWord16* cur_pos) {
+ ASSERT_TRUE(payload <= 127);
+ array[(*cur_pos)++] = 0x80;
+ array[(*cur_pos)++] = payload | (marker_bit ? 0x80 : 0);
+ array[(*cur_pos)++] = seq_num >> 8;
+ array[(*cur_pos)++] = seq_num;
+ array[(*cur_pos)++] = timestamp >> 24;
+ array[(*cur_pos)++] = timestamp >> 16;
+ array[(*cur_pos)++] = timestamp >> 8;
+ array[(*cur_pos)++] = timestamp;
+ array[(*cur_pos)++] = ssrc >> 24;
+ array[(*cur_pos)++] = ssrc >> 16;
+ array[(*cur_pos)++] = ssrc >> 8;
+ array[(*cur_pos)++] = ssrc;
+ // VP8 payload header
+ array[(*cur_pos)++] = 0x90; // X bit = 1
+ array[(*cur_pos)++] = 0x20; // T bit = 1
+ array[(*cur_pos)++] = 0x00; // TID = 0
+ array[(*cur_pos)++] = 0x00; // Key frame
+ array[(*cur_pos)++] = 0x00;
+ array[(*cur_pos)++] = 0x00;
+ array[(*cur_pos)++] = 0x9d;
+ array[(*cur_pos)++] = 0x01;
+ array[(*cur_pos)++] = 0x2a;
+ array[(*cur_pos)++] = 128;
+ array[(*cur_pos)++] = 0;
+ array[(*cur_pos)++] = 96;
+ array[(*cur_pos)++] = 0;
+}
+
+class TestTransport : public Transport,
+ public RtpData {
+ public:
+ TestTransport(RTCPReceiver* rtcp_receiver) :
+ rtcp_receiver_(rtcp_receiver) {
+ }
+
+ virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
+ return -1;
+ }
+
+ virtual int SendRTCPPacket(int /*ch*/, const void *packet, int packet_len) {
+ RTCPUtility::RTCPParserV2 rtcpParser((WebRtc_UWord8*)packet,
+ (WebRtc_Word32)packet_len,
+ true); // Allow non-compound RTCP
+
+ EXPECT_TRUE(rtcpParser.IsValid());
+ RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
+ EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
+ &rtcpParser));
+ rtcp_packet_info_ = rtcpPacketInformation;
+
+ return packet_len;
+ }
+
+ virtual int OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader)
+ {return 0;}
+ RTCPReceiver* rtcp_receiver_;
+ RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
+};
+
+class RtcpSenderTest : public ::testing::Test {
+ protected:
+ RtcpSenderTest() {
+ system_clock_ = ModuleRTPUtility::GetSystemClock();
+ rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
+ rtcp_sender_ = new RTCPSender(0, false, system_clock_, rtp_rtcp_impl_);
+ rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
+ test_transport_ = new TestTransport(rtcp_receiver_);
+ // Initialize
+ EXPECT_EQ(0, rtcp_sender_->Init());
+ EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
+ EXPECT_EQ(0, rtp_rtcp_impl_->RegisterIncomingDataCallback(test_transport_));
+ }
+ ~RtcpSenderTest() {
+ delete rtcp_sender_;
+ delete rtcp_receiver_;
+ delete rtp_rtcp_impl_;
+ delete test_transport_;
+ delete system_clock_;
+ }
+
+ RtpRtcpClock* system_clock_;
+ ModuleRtpRtcpImpl* rtp_rtcp_impl_;
+ RTCPSender* rtcp_sender_;
+ RTCPReceiver* rtcp_receiver_;
+ TestTransport* test_transport_;
+
+ enum {kMaxPacketLength = 1500};
+ uint8_t packet_[kMaxPacketLength];
+};
+
+TEST_F(RtcpSenderTest, RtcpOff) {
+ EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpOff));
+ EXPECT_EQ(-1, rtcp_sender_->SendRTCP(kRtcpSr));
+}
+
+TEST_F(RtcpSenderTest, IJStatus) {
+ ASSERT_FALSE(rtcp_sender_->IJ());
+ EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
+ ASSERT_TRUE(rtcp_sender_->IJ());
+}
+
+TEST_F(RtcpSenderTest, TestCompound) {
+ const bool marker_bit = false;
+ const WebRtc_UWord8 payload = 100;
+ const WebRtc_UWord16 seq_num = 11111;
+ const WebRtc_UWord32 timestamp = 1234567;
+ const WebRtc_UWord32 ssrc = 0x11111111;
+ WebRtc_UWord16 packet_length = 0;
+ CreateRtpPacket(marker_bit, payload, seq_num, timestamp, ssrc, packet_,
+ &packet_length);
+ EXPECT_EQ(25, packet_length);
+
+ VideoCodec codec_inst;
+ strncpy(codec_inst.plName, "VP8", webrtc::kPayloadNameSize - 1);
+ codec_inst.codecType = webrtc::kVideoCodecVP8;
+ codec_inst.plType = payload;
+ EXPECT_EQ(0, rtp_rtcp_impl_->RegisterReceivePayload(codec_inst));
+
+ // Make sure RTP packet has been received.
+ EXPECT_EQ(0, rtp_rtcp_impl_->IncomingPacket(packet_, packet_length));
+
+ EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
+ EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRr));
+
+ // Transmission time offset packet should be received.
+ ASSERT_TRUE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
+ kRtcpTransmissionTimeOffset);
+}
+
+TEST_F(RtcpSenderTest, TestCompound_NoRtpReceived) {
+ EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
+ EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRr));
+
+ // Transmission time offset packet should not be received.
+ ASSERT_FALSE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
+ kRtcpTransmissionTimeOffset);
+}
+
+int main(int argc, char** argv) {
+ ::testing::InitGoogleTest(&argc, argv);
+
+ return RUN_ALL_TESTS();
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc
new file mode 100644
index 0000000..f421672
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc
@@ -0,0 +1,1509 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtcp_utility.h"
+
+#include <cstring> // memcpy
+#include <cmath> // ceil
+#include <cassert>
+
+
+namespace webrtc {
+// RTCPParserV2 : currently read only
+
+RTCPUtility::RTCPParserV2::RTCPParserV2(const WebRtc_UWord8* rtcpData,
+ size_t rtcpDataLength,
+ bool rtcpReducedSizeEnable)
+ : _ptrRTCPDataBegin(rtcpData),
+ _RTCPReducedSizeEnable(rtcpReducedSizeEnable),
+ _ptrRTCPDataEnd(rtcpData + rtcpDataLength),
+ _validPacket(false),
+ _ptrRTCPData(rtcpData),
+ _ptrRTCPBlockEnd(NULL),
+ _state(State_TopLevel),
+ _numberOfBlocks(0),
+ _packetType(kRtcpNotValidCode) {
+ Validate();
+}
+
+RTCPUtility::RTCPParserV2::~RTCPParserV2() {
+}
+
+ptrdiff_t
+RTCPUtility::RTCPParserV2::LengthLeft() const
+{
+ return (_ptrRTCPDataEnd- _ptrRTCPData);
+}
+
+RTCPUtility::RTCPPacketTypes
+RTCPUtility::RTCPParserV2::PacketType() const
+{
+ return _packetType;
+}
+
+const RTCPUtility::RTCPPacket&
+RTCPUtility::RTCPParserV2::Packet() const
+{
+ return _packet;
+}
+
+RTCPUtility::RTCPPacketTypes
+RTCPUtility::RTCPParserV2::Begin()
+{
+ _ptrRTCPData = _ptrRTCPDataBegin;
+
+ return Iterate();
+}
+
+RTCPUtility::RTCPPacketTypes
+RTCPUtility::RTCPParserV2::Iterate()
+{
+ // Reset packet type
+ _packetType = kRtcpNotValidCode;
+
+ if (IsValid())
+ {
+ switch (_state)
+ {
+ case State_TopLevel:
+ IterateTopLevel();
+ break;
+ case State_ReportBlockItem:
+ IterateReportBlockItem();
+ break;
+ case State_SDESChunk:
+ IterateSDESChunk();
+ break;
+ case State_BYEItem:
+ IterateBYEItem();
+ break;
+ case State_ExtendedJitterItem:
+ IterateExtendedJitterItem();
+ break;
+ case State_RTPFB_NACKItem:
+ IterateNACKItem();
+ break;
+ case State_RTPFB_TMMBRItem:
+ IterateTMMBRItem();
+ break;
+ case State_RTPFB_TMMBNItem:
+ IterateTMMBNItem();
+ break;
+ case State_PSFB_SLIItem:
+ IterateSLIItem();
+ break;
+ case State_PSFB_RPSIItem:
+ IterateRPSIItem();
+ break;
+ case State_PSFB_FIRItem:
+ IterateFIRItem();
+ break;
+ case State_PSFB_AppItem:
+ IteratePsfbAppItem();
+ break;
+ case State_PSFB_REMBItem:
+ IteratePsfbREMBItem();
+ break;
+ case State_AppItem:
+ IterateAppItem();
+ break;
+ default:
+ assert(false); // Invalid state!
+ break;
+ }
+ }
+ return _packetType;
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateTopLevel()
+{
+ for (;;)
+ {
+ RTCPCommonHeader header;
+
+ const bool success = RTCPParseCommonHeader(_ptrRTCPData,
+ _ptrRTCPDataEnd,
+ header);
+
+ if (!success)
+ {
+ return;
+ }
+ _ptrRTCPBlockEnd = _ptrRTCPData + header.LengthInOctets;
+ if (_ptrRTCPBlockEnd > _ptrRTCPDataEnd)
+ {
+ // Bad block!
+ return;
+ }
+
+ switch (header.PT)
+ {
+ case PT_SR:
+ {
+ // number of Report blocks
+ _numberOfBlocks = header.IC;
+ ParseSR();
+ return;
+ }
+ case PT_RR:
+ {
+ // number of Report blocks
+ _numberOfBlocks = header.IC;
+ ParseRR();
+ return;
+ }
+ case PT_SDES:
+ {
+ // number of SDES blocks
+ _numberOfBlocks = header.IC;
+ const bool ok = ParseSDES();
+ if (!ok)
+ {
+ // Nothing supported found, continue to next block!
+ break;
+ }
+ return;
+ }
+ case PT_BYE:
+ {
+ _numberOfBlocks = header.IC;
+ const bool ok = ParseBYE();
+ if (!ok)
+ {
+ // Nothing supported found, continue to next block!
+ break;
+ }
+ return;
+ }
+ case PT_IJ:
+ {
+ // number of Report blocks
+ _numberOfBlocks = header.IC;
+ ParseIJ();
+ return;
+ }
+ case PT_RTPFB: // Fall through!
+ case PT_PSFB:
+ {
+ const bool ok = ParseFBCommon(header);
+ if (!ok)
+ {
+ // Nothing supported found, continue to next block!
+ break;
+ }
+
+ return;
+ }
+ case PT_APP:
+ {
+ const bool ok = ParseAPP(header);
+ if (!ok)
+ {
+ // Nothing supported found, continue to next block!
+ break;
+ }
+ return;
+ }
+ case PT_XR:
+ {
+ const bool ok = ParseXR();
+ if (!ok)
+ {
+ // Nothing supported found, continue to next block!
+ break;
+ }
+ return;
+ }
+ default:
+ // Not supported! Skip!
+ EndCurrentBlock();
+ break;
+ }
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateReportBlockItem()
+{
+ const bool success = ParseReportBlockItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateSDESChunk()
+{
+ const bool success = ParseSDESChunk();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateBYEItem()
+{
+ const bool success = ParseBYEItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateExtendedJitterItem()
+{
+ const bool success = ParseIJItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateNACKItem()
+{
+ const bool success = ParseNACKItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateTMMBRItem()
+{
+ const bool success = ParseTMMBRItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateTMMBNItem()
+{
+ const bool success = ParseTMMBNItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateSLIItem()
+{
+ const bool success = ParseSLIItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateRPSIItem()
+{
+ const bool success = ParseRPSIItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateFIRItem()
+{
+ const bool success = ParseFIRItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IteratePsfbAppItem()
+{
+ const bool success = ParsePsfbAppItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IteratePsfbREMBItem()
+{
+ const bool success = ParsePsfbREMBItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::IterateAppItem()
+{
+ const bool success = ParseAPPItem();
+ if (!success)
+ {
+ Iterate();
+ }
+}
+
+void
+RTCPUtility::RTCPParserV2::Validate()
+{
+ if (_ptrRTCPData == NULL)
+ {
+ return; // NOT VALID
+ }
+
+ RTCPCommonHeader header;
+ const bool success = RTCPParseCommonHeader(_ptrRTCPDataBegin,
+ _ptrRTCPDataEnd,
+ header);
+
+ if (!success)
+ {
+ return; // NOT VALID!
+ }
+
+ // * if (!reducedSize) : first packet must be RR or SR.
+ //
+ // * The padding bit (P) should be zero for the first packet of a
+ // compound RTCP packet because padding should only be applied,
+ // if it is needed, to the last packet. (NOT CHECKED!)
+ //
+ // * The length fields of the individual RTCP packets must add up
+ // to the overall length of the compound RTCP packet as
+ // received. This is a fairly strong check. (NOT CHECKED!)
+
+ if (!_RTCPReducedSizeEnable)
+ {
+ if ((header.PT != PT_SR) && (header.PT != PT_RR))
+ {
+ return; // NOT VALID
+ }
+ }
+
+ _validPacket = true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::IsValid() const
+{
+ return _validPacket;
+}
+
+void
+RTCPUtility::RTCPParserV2::EndCurrentBlock()
+{
+ _ptrRTCPData = _ptrRTCPBlockEnd;
+}
+
+bool
+RTCPUtility::RTCPParseCommonHeader( const WebRtc_UWord8* ptrDataBegin,
+ const WebRtc_UWord8* ptrDataEnd,
+ RTCPCommonHeader& parsedHeader)
+{
+ if (!ptrDataBegin || !ptrDataEnd)
+ {
+ return false;
+ }
+
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // |V=2|P| IC | PT | length |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ //
+ // Common header for all RTCP packets, 4 octets.
+
+ if ((ptrDataEnd - ptrDataBegin) < 4)
+ {
+ return false;
+ }
+
+ parsedHeader.V = ptrDataBegin[0] >> 6;
+ parsedHeader.P = ((ptrDataBegin[0] & 0x20) == 0) ? false : true;
+ parsedHeader.IC = ptrDataBegin[0] & 0x1f;
+ parsedHeader.PT = ptrDataBegin[1];
+
+ parsedHeader.LengthInOctets = (ptrDataBegin[2] << 8) + ptrDataBegin[3] + 1;
+ parsedHeader.LengthInOctets *= 4;
+
+ if(parsedHeader.LengthInOctets == 0)
+ {
+ return false;
+ }
+ // Check if RTP version field == 2
+ if (parsedHeader.V != 2)
+ {
+ return false;
+ }
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseRR()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ return false;
+ }
+
+
+ _ptrRTCPData += 4; // Skip header
+
+ _packetType = kRtcpRrCode;
+
+ _packet.RR.SenderSSRC = *_ptrRTCPData++ << 24;
+ _packet.RR.SenderSSRC += *_ptrRTCPData++ << 16;
+ _packet.RR.SenderSSRC += *_ptrRTCPData++ << 8;
+ _packet.RR.SenderSSRC += *_ptrRTCPData++;
+
+ _packet.RR.NumberOfReportBlocks = _numberOfBlocks;
+
+ // State transition
+ _state = State_ReportBlockItem;
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseSR()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 28)
+ {
+ EndCurrentBlock();
+ return false;
+ }
+
+ _ptrRTCPData += 4; // Skip header
+
+ _packetType = kRtcpSrCode;
+
+ _packet.SR.SenderSSRC = *_ptrRTCPData++ << 24;
+ _packet.SR.SenderSSRC += *_ptrRTCPData++ << 16;
+ _packet.SR.SenderSSRC += *_ptrRTCPData++ << 8;
+ _packet.SR.SenderSSRC += *_ptrRTCPData++;
+
+ _packet.SR.NTPMostSignificant = *_ptrRTCPData++ << 24;
+ _packet.SR.NTPMostSignificant += *_ptrRTCPData++ << 16;
+ _packet.SR.NTPMostSignificant += *_ptrRTCPData++ << 8;
+ _packet.SR.NTPMostSignificant += *_ptrRTCPData++;
+
+ _packet.SR.NTPLeastSignificant = *_ptrRTCPData++ << 24;
+ _packet.SR.NTPLeastSignificant += *_ptrRTCPData++ << 16;
+ _packet.SR.NTPLeastSignificant += *_ptrRTCPData++ << 8;
+ _packet.SR.NTPLeastSignificant += *_ptrRTCPData++;
+
+ _packet.SR.RTPTimestamp = *_ptrRTCPData++ << 24;
+ _packet.SR.RTPTimestamp += *_ptrRTCPData++ << 16;
+ _packet.SR.RTPTimestamp += *_ptrRTCPData++ << 8;
+ _packet.SR.RTPTimestamp += *_ptrRTCPData++;
+
+ _packet.SR.SenderPacketCount = *_ptrRTCPData++ << 24;
+ _packet.SR.SenderPacketCount += *_ptrRTCPData++ << 16;
+ _packet.SR.SenderPacketCount += *_ptrRTCPData++ << 8;
+ _packet.SR.SenderPacketCount += *_ptrRTCPData++;
+
+ _packet.SR.SenderOctetCount = *_ptrRTCPData++ << 24;
+ _packet.SR.SenderOctetCount += *_ptrRTCPData++ << 16;
+ _packet.SR.SenderOctetCount += *_ptrRTCPData++ << 8;
+ _packet.SR.SenderOctetCount += *_ptrRTCPData++;
+
+ _packet.SR.NumberOfReportBlocks = _numberOfBlocks;
+
+ // State transition
+ if(_numberOfBlocks != 0)
+ {
+ _state = State_ReportBlockItem;
+ }else
+ {
+ // don't go to state report block item if 0 report blocks
+ _state = State_TopLevel;
+ EndCurrentBlock();
+ }
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseReportBlockItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 24 || _numberOfBlocks <= 0)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24;
+ _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.SSRC += *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.FractionLost = *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.CumulativeNumOfPacketsLost = *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.CumulativeNumOfPacketsLost += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.CumulativeNumOfPacketsLost += *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.ExtendedHighestSequenceNumber = *_ptrRTCPData++ << 24;
+ _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.Jitter = *_ptrRTCPData++ << 24;
+ _packet.ReportBlockItem.Jitter += *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.Jitter += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.Jitter += *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.LastSR = *_ptrRTCPData++ << 24;
+ _packet.ReportBlockItem.LastSR += *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.LastSR += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.LastSR += *_ptrRTCPData++;
+
+ _packet.ReportBlockItem.DelayLastSR = *_ptrRTCPData++ << 24;
+ _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++ << 16;
+ _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++ << 8;
+ _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++;
+
+ _numberOfBlocks--;
+ _packetType = kRtcpReportBlockItemCode;
+ return true;
+}
+
+/* From RFC 5450: Transmission Time Offsets in RTP Streams.
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ hdr |V=2|P| RC | PT=IJ=195 | length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | inter-arrival jitter |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ . .
+ . .
+ . .
+ | inter-arrival jitter |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+
+bool
+RTCPUtility::RTCPParserV2::ParseIJ()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ return false;
+ }
+
+ _ptrRTCPData += 4; // Skip header
+
+ _packetType = kRtcpExtendedIjCode;
+
+ // State transition
+ _state = State_ExtendedJitterItem;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseIJItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4 || _numberOfBlocks <= 0)
+ {
+ _state = State_TopLevel;
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packet.ExtendedJitterReportItem.Jitter = *_ptrRTCPData++ << 24;
+ _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++ << 16;
+ _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++ << 8;
+ _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++;
+
+ _numberOfBlocks--;
+ _packetType = kRtcpExtendedIjItemCode;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseSDES()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _ptrRTCPData += 4; // Skip header
+
+ _state = State_SDESChunk;
+ _packetType = kRtcpSdesCode;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseSDESChunk()
+{
+ if(_numberOfBlocks <= 0)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _numberOfBlocks--;
+
+ // Find CName item in a SDES chunk.
+ while (_ptrRTCPData < _ptrRTCPBlockEnd)
+ {
+ const ptrdiff_t dataLen = _ptrRTCPBlockEnd - _ptrRTCPData;
+ if (dataLen < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ WebRtc_UWord32 SSRC = *_ptrRTCPData++ << 24;
+ SSRC += *_ptrRTCPData++ << 16;
+ SSRC += *_ptrRTCPData++ << 8;
+ SSRC += *_ptrRTCPData++;
+
+ const bool foundCname = ParseSDESItem();
+ if (foundCname)
+ {
+ _packet.CName.SenderSSRC = SSRC; // Add SSRC
+ return true;
+ }
+ }
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseSDESItem()
+{
+ // Find CName
+ // Only the CNAME item is mandatory. RFC 3550 page 46
+ bool foundCName = false;
+
+ size_t itemOctetsRead = 0;
+ while (_ptrRTCPData < _ptrRTCPBlockEnd)
+ {
+ const WebRtc_UWord8 tag = *_ptrRTCPData++;
+ ++itemOctetsRead;
+
+ if (tag == 0)
+ {
+ // End tag! 4 oct aligned
+ while ((itemOctetsRead++ % 4) != 0)
+ {
+ ++_ptrRTCPData;
+ }
+ return foundCName;
+ }
+
+ if (_ptrRTCPData < _ptrRTCPBlockEnd)
+ {
+ const WebRtc_UWord8 len = *_ptrRTCPData++;
+ ++itemOctetsRead;
+
+ if (tag == 1)
+ {
+ // CNAME
+
+ // Sanity
+ if ((_ptrRTCPData + len) >= _ptrRTCPBlockEnd)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ WebRtc_UWord8 i = 0;
+ for (; i < len; ++i)
+ {
+ const WebRtc_UWord8 c = _ptrRTCPData[i];
+ if ((c < ' ') || (c > '{') || (c == '%') || (c == '\\'))
+ {
+ // Illegal char
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _packet.CName.CName[i] = c;
+ }
+ // Make sure we are null terminated.
+ _packet.CName.CName[i] = 0;
+ _packetType = kRtcpSdesChunkCode;
+
+ foundCName = true;
+ }
+ _ptrRTCPData += len;
+ itemOctetsRead += len;
+ }
+ }
+
+ // No end tag found!
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseBYE()
+{
+ _ptrRTCPData += 4; // Skip header
+
+ _state = State_BYEItem;
+
+ return ParseBYEItem();
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseBYEItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+ if (length < 4 || _numberOfBlocks == 0)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpByeCode;
+
+ _packet.BYE.SenderSSRC = *_ptrRTCPData++ << 24;
+ _packet.BYE.SenderSSRC += *_ptrRTCPData++ << 16;
+ _packet.BYE.SenderSSRC += *_ptrRTCPData++ << 8;
+ _packet.BYE.SenderSSRC += *_ptrRTCPData++;
+
+ // we can have several CSRCs attached
+
+ // sanity
+ if(length >= 4*_numberOfBlocks)
+ {
+ _ptrRTCPData += (_numberOfBlocks -1)*4;
+ }
+ _numberOfBlocks = 0;
+
+ return true;
+}
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |V=2|P|reserved | PT=XR=207 | length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | SSRC |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ : report blocks :
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+bool RTCPUtility::RTCPParserV2::ParseXR()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ EndCurrentBlock();
+ return false;
+ }
+
+ _ptrRTCPData += 4; // Skip header
+
+ _packet.XR.OriginatorSSRC = *_ptrRTCPData++ << 24;
+ _packet.XR.OriginatorSSRC += *_ptrRTCPData++ << 16;
+ _packet.XR.OriginatorSSRC += *_ptrRTCPData++ << 8;
+ _packet.XR.OriginatorSSRC += *_ptrRTCPData++;
+
+ return ParseXRItem();
+}
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | BT | type-specific | block length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ : type-specific block contents :
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+
+bool
+RTCPUtility::RTCPParserV2::ParseXRItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4) //
+ {
+ EndCurrentBlock();
+ return false;
+ }
+
+ WebRtc_UWord8 blockType = *_ptrRTCPData++;
+ WebRtc_UWord8 typeSpecific = *_ptrRTCPData++;
+
+ WebRtc_UWord16 blockLength = *_ptrRTCPData++ << 8;
+ blockLength = *_ptrRTCPData++;
+
+ if(blockType == 7 && typeSpecific == 0)
+ {
+ if(blockLength != 8)
+ {
+ EndCurrentBlock();
+ return false;
+ }
+ return ParseXRVOIPMetricItem();
+ }else
+ {
+ EndCurrentBlock();
+ return false;
+ }
+}
+/*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | BT=7 | reserved | block length = 8 |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | SSRC of source |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | loss rate | discard rate | burst density | gap density |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | burst duration | gap duration |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | round trip delay | end system delay |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | signal level | noise level | RERL | Gmin |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | R factor | ext. R factor | MOS-LQ | MOS-CQ |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | RX config | reserved | JB nominal |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | JB maximum | JB abs max |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+bool
+RTCPUtility::RTCPParserV2::ParseXRVOIPMetricItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 28)
+ {
+ EndCurrentBlock();
+ return false;
+ }
+ _packetType = kRtcpXrVoipMetricCode;
+
+ _packet.XRVOIPMetricItem.SSRC = *_ptrRTCPData++ << 24;
+ _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++ << 16;
+ _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.lossRate = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.discardRate = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.burstDensity = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.gapDensity = *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.burstDuration = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.burstDuration += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.gapDuration = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.gapDuration += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.roundTripDelay = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.roundTripDelay += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.endSystemDelay = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.endSystemDelay += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.signalLevel = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.noiseLevel = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.RERL = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.Gmin = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.Rfactor = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.extRfactor = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.MOSLQ = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.MOSCQ = *_ptrRTCPData++;
+ _packet.XRVOIPMetricItem.RXconfig = *_ptrRTCPData++;
+ _ptrRTCPData++; // skip reserved
+
+ _packet.XRVOIPMetricItem.JBnominal = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.JBnominal += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.JBmax = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.JBmax += *_ptrRTCPData++;
+
+ _packet.XRVOIPMetricItem.JBabsMax = *_ptrRTCPData++ << 8;
+ _packet.XRVOIPMetricItem.JBabsMax += *_ptrRTCPData++;
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
+{
+ assert((header.PT == PT_RTPFB) || (header.PT == PT_PSFB)); // Parser logic check
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 12) // 4 * 3, RFC4585 section 6.1
+ {
+ EndCurrentBlock();
+ return false;
+ }
+
+ _ptrRTCPData += 4; // Skip RTCP header
+
+ WebRtc_UWord32 senderSSRC = *_ptrRTCPData++ << 24;
+ senderSSRC += *_ptrRTCPData++ << 16;
+ senderSSRC += *_ptrRTCPData++ << 8;
+ senderSSRC += *_ptrRTCPData++;
+
+ WebRtc_UWord32 mediaSSRC = *_ptrRTCPData++ << 24;
+ mediaSSRC += *_ptrRTCPData++ << 16;
+ mediaSSRC += *_ptrRTCPData++ << 8;
+ mediaSSRC += *_ptrRTCPData++;
+
+ if (header.PT == PT_RTPFB)
+ {
+ // Transport layer feedback
+
+ switch (header.IC)
+ {
+ case 1:
+ {
+ // NACK
+ _packetType = kRtcpRtpfbNackCode;
+ _packet.NACK.SenderSSRC = senderSSRC;
+ _packet.NACK.MediaSSRC = mediaSSRC;
+
+ _state = State_RTPFB_NACKItem;
+
+ return true;
+ }
+ case 2:
+ {
+ // used to be ACK is this code point, which is removed
+ // conficts with http://tools.ietf.org/html/draft-levin-avt-rtcp-burst-00
+ break;
+ }
+ case 3:
+ {
+ // TMMBR
+ _packetType = kRtcpRtpfbTmmbrCode;
+ _packet.TMMBR.SenderSSRC = senderSSRC;
+ _packet.TMMBR.MediaSSRC = mediaSSRC;
+
+ _state = State_RTPFB_TMMBRItem;
+
+ return true;
+ }
+ case 4:
+ {
+ // TMMBN
+ _packetType = kRtcpRtpfbTmmbnCode;
+ _packet.TMMBN.SenderSSRC = senderSSRC;
+ _packet.TMMBN.MediaSSRC = mediaSSRC;
+
+ _state = State_RTPFB_TMMBNItem;
+
+ return true;
+ }
+ case 5:
+ {
+ // RTCP-SR-REQ Rapid Synchronisation of RTP Flows
+ // draft-perkins-avt-rapid-rtp-sync-03.txt
+ // trigger a new RTCP SR
+ _packetType = kRtcpRtpfbSrReqCode;
+
+ // Note: No state transition, SR REQ is empty!
+ return true;
+ }
+ default:
+ break;
+ }
+ EndCurrentBlock();
+ return false;
+ }
+ else if (header.PT == PT_PSFB)
+ {
+ // Payload specific feedback
+ switch (header.IC)
+ {
+ case 1:
+ // PLI
+ _packetType = kRtcpPsfbPliCode;
+ _packet.PLI.SenderSSRC = senderSSRC;
+ _packet.PLI.MediaSSRC = mediaSSRC;
+
+ // Note: No state transition, PLI FCI is empty!
+ return true;
+ case 2:
+ // SLI
+ _packetType = kRtcpPsfbSliCode;
+ _packet.SLI.SenderSSRC = senderSSRC;
+ _packet.SLI.MediaSSRC = mediaSSRC;
+
+ _state = State_PSFB_SLIItem;
+
+ return true;
+ case 3:
+ _packetType = kRtcpPsfbRpsiCode;
+ _packet.RPSI.SenderSSRC = senderSSRC;
+ _packet.RPSI.MediaSSRC = mediaSSRC;
+
+ _state = State_PSFB_RPSIItem;
+ return true;
+ case 4:
+ // FIR
+ _packetType = kRtcpPsfbFirCode;
+ _packet.FIR.SenderSSRC = senderSSRC;
+ _packet.FIR.MediaSSRC = mediaSSRC;
+
+ _state = State_PSFB_FIRItem;
+ return true;
+ case 15:
+ _packetType = kRtcpPsfbAppCode;
+
+ _state = State_PSFB_AppItem;
+ return true;
+ default:
+ break;
+ }
+
+ EndCurrentBlock();
+ return false;
+ }
+ else
+ {
+ assert(false);
+
+ EndCurrentBlock();
+ return false;
+ }
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseRPSIItem()
+{
+ // RFC 4585 6.3.3. Reference Picture Selection Indication (RPSI)
+ /*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | PB |0| Payload Type| Native RPSI bit string |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | defined per codec ... | Padding (0) |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ if(length > 2+RTCP_RPSI_DATA_SIZE)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpPsfbRpsiCode;
+
+ WebRtc_UWord8 paddingBits = *_ptrRTCPData++;
+ _packet.RPSI.PayloadType = *_ptrRTCPData++;
+
+ memcpy(_packet.RPSI.NativeBitString, _ptrRTCPData, length-2);
+
+ _packet.RPSI.NumberOfValidBits = WebRtc_UWord16(length-2)*8 - paddingBits;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseNACKItem()
+{
+ // RFC 4585 6.2.1. Generic NACK
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpRtpfbNackItemCode;
+
+ _packet.NACKItem.PacketID = *_ptrRTCPData++ << 8;
+ _packet.NACKItem.PacketID += *_ptrRTCPData++;
+
+ _packet.NACKItem.BitMask = *_ptrRTCPData++ << 8;
+ _packet.NACKItem.BitMask += *_ptrRTCPData++;
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParsePsfbAppItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ if(*_ptrRTCPData++ != 'R')
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ if(*_ptrRTCPData++ != 'E')
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ if(*_ptrRTCPData++ != 'M')
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ if(*_ptrRTCPData++ != 'B')
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _packetType = kRtcpPsfbRembItemCode;
+ _state = State_PSFB_REMBItem;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ const WebRtc_UWord8 numSSRC = *_ptrRTCPData++;
+ const WebRtc_UWord8 brExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+
+ WebRtc_UWord32 brMantissa = (_ptrRTCPData[0] & 0x03) << 16;
+ brMantissa += (_ptrRTCPData[1] << 8);
+ brMantissa += (_ptrRTCPData[2]);
+
+ _ptrRTCPData += 3; // Fwd read data
+ _packet.REMB.BitRate = (brMantissa << brExp);
+
+ _ptrRTCPData += 4 * numSSRC; // Ignore the SSRCs for now
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseTMMBRItem()
+{
+ // RFC 5104 4.2.1. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpRtpfbTmmbrItemCode;
+
+ _packet.TMMBRItem.SSRC = *_ptrRTCPData++ << 24;
+ _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 16;
+ _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 8;
+ _packet.TMMBRItem.SSRC += *_ptrRTCPData++;
+
+ WebRtc_UWord8 mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+
+ WebRtc_UWord32 mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
+ mxtbrMantissa += (_ptrRTCPData[1] << 7);
+ mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
+
+ WebRtc_UWord32 measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
+ measuredOH += _ptrRTCPData[3];
+
+ _ptrRTCPData += 4; // Fwd read data
+
+ _packet.TMMBRItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
+ _packet.TMMBRItem.MeasuredOverhead = measuredOH;
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseTMMBNItem()
+{
+ // RFC 5104 4.2.2. Temporary Maximum Media Stream Bit Rate Notification (TMMBN)
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpRtpfbTmmbnItemCode;
+
+ _packet.TMMBNItem.SSRC = *_ptrRTCPData++ << 24;
+ _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 16;
+ _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 8;
+ _packet.TMMBNItem.SSRC += *_ptrRTCPData++;
+
+ WebRtc_UWord8 mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+
+ WebRtc_UWord32 mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
+ mxtbrMantissa += (_ptrRTCPData[1] << 7);
+ mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
+
+ WebRtc_UWord32 measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
+ measuredOH += _ptrRTCPData[3];
+
+ _ptrRTCPData += 4; // Fwd read data
+
+ _packet.TMMBNItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
+ _packet.TMMBNItem.MeasuredOverhead = measuredOH;
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseSLIItem()
+{
+ // RFC 5104 6.3.2. Slice Loss Indication (SLI)
+ /*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | First | Number | PictureID |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _packetType = kRtcpPsfbSliItemCode;
+
+ WebRtc_UWord32 buffer;
+ buffer = *_ptrRTCPData++ << 24;
+ buffer += *_ptrRTCPData++ << 16;
+ buffer += *_ptrRTCPData++ << 8;
+ buffer += *_ptrRTCPData++;
+
+ _packet.SLIItem.FirstMB = WebRtc_UWord16((buffer>>19) & 0x1fff);
+ _packet.SLIItem.NumberOfMB = WebRtc_UWord16((buffer>>6) & 0x1fff);
+ _packet.SLIItem.PictureId = WebRtc_UWord8(buffer & 0x3f);
+
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseFIRItem()
+{
+ // RFC 5104 4.3.1. Full Intra Request (FIR)
+
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 8)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+
+ _packetType = kRtcpPsfbFirItemCode;
+
+ _packet.FIRItem.SSRC = *_ptrRTCPData++ << 24;
+ _packet.FIRItem.SSRC += *_ptrRTCPData++ << 16;
+ _packet.FIRItem.SSRC += *_ptrRTCPData++ << 8;
+ _packet.FIRItem.SSRC += *_ptrRTCPData++;
+
+ _packet.FIRItem.CommandSequenceNumber = *_ptrRTCPData++;
+ _ptrRTCPData += 3; // Skip "Reserved" bytes.
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseAPP( const RTCPCommonHeader& header)
+{
+ ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ if (length < 12) // 4 * 3, RFC 3550 6.7 APP: Application-Defined RTCP Packet
+ {
+ EndCurrentBlock();
+ return false;
+ }
+
+ _ptrRTCPData += 4; // Skip RTCP header
+
+ WebRtc_UWord32 senderSSRC = *_ptrRTCPData++ << 24;
+ senderSSRC += *_ptrRTCPData++ << 16;
+ senderSSRC += *_ptrRTCPData++ << 8;
+ senderSSRC += *_ptrRTCPData++;
+
+ WebRtc_UWord32 name = *_ptrRTCPData++ << 24;
+ name += *_ptrRTCPData++ << 16;
+ name += *_ptrRTCPData++ << 8;
+ name += *_ptrRTCPData++;
+
+ length = _ptrRTCPBlockEnd - _ptrRTCPData;
+
+ _packetType = kRtcpAppCode;
+
+ _packet.APP.SubType = header.IC;
+ _packet.APP.Name = name;
+
+ _state = State_AppItem;
+ return true;
+}
+
+bool
+RTCPUtility::RTCPParserV2::ParseAPPItem()
+{
+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
+ if (length < 4)
+ {
+ _state = State_TopLevel;
+
+ EndCurrentBlock();
+ return false;
+ }
+ _packetType = kRtcpAppItemCode;
+
+ if(length > kRtcpAppCode_DATA_SIZE)
+ {
+ memcpy(_packet.APP.Data, _ptrRTCPData, kRtcpAppCode_DATA_SIZE);
+ _packet.APP.Size = kRtcpAppCode_DATA_SIZE;
+ _ptrRTCPData += kRtcpAppCode_DATA_SIZE;
+ }else
+ {
+ memcpy(_packet.APP.Data, _ptrRTCPData, length);
+ _packet.APP.Size = (WebRtc_UWord16)length;
+ _ptrRTCPData += length;
+ }
+ return true;
+}
+
+RTCPUtility::RTCPPacketIterator::RTCPPacketIterator(WebRtc_UWord8* rtcpData,
+ size_t rtcpDataLength)
+ : _ptrBegin(rtcpData),
+ _ptrEnd(rtcpData + rtcpDataLength),
+ _ptrBlock(NULL) {
+ memset(&_header, 0, sizeof(_header));
+}
+
+RTCPUtility::RTCPPacketIterator::~RTCPPacketIterator() {
+}
+
+const RTCPUtility::RTCPCommonHeader*
+RTCPUtility::RTCPPacketIterator::Begin()
+{
+ _ptrBlock = _ptrBegin;
+
+ return Iterate();
+}
+
+const RTCPUtility::RTCPCommonHeader*
+RTCPUtility::RTCPPacketIterator::Iterate()
+{
+ const bool success = RTCPParseCommonHeader(_ptrBlock, _ptrEnd, _header);
+ if (!success)
+ {
+ _ptrBlock = NULL;
+ return NULL;
+ }
+ _ptrBlock += _header.LengthInOctets;
+
+ if (_ptrBlock > _ptrEnd)
+ {
+ _ptrBlock = NULL;
+ return NULL;
+ }
+
+ return &_header;
+}
+
+const RTCPUtility::RTCPCommonHeader*
+RTCPUtility::RTCPPacketIterator::Current()
+{
+ if (!_ptrBlock)
+ {
+ return NULL;
+ }
+
+ return &_header;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h
new file mode 100644
index 0000000..faba2ef
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h
@@ -0,0 +1,437 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
+
+#include <cstddef> // size_t, ptrdiff_t
+
+#include "typedefs.h"
+#include "rtp_rtcp_config.h"
+#include "rtp_rtcp_defines.h"
+
+namespace webrtc {
+namespace RTCPUtility {
+ // CNAME
+ struct RTCPCnameInformation
+ {
+ char name[RTCP_CNAME_SIZE];
+ };
+ struct RTCPPacketRR
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord8 NumberOfReportBlocks;
+ };
+ struct RTCPPacketSR
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord8 NumberOfReportBlocks;
+
+ // sender info
+ WebRtc_UWord32 NTPMostSignificant;
+ WebRtc_UWord32 NTPLeastSignificant;
+ WebRtc_UWord32 RTPTimestamp;
+ WebRtc_UWord32 SenderPacketCount;
+ WebRtc_UWord32 SenderOctetCount;
+ };
+ struct RTCPPacketReportBlockItem
+ {
+ // report block
+ WebRtc_UWord32 SSRC;
+ WebRtc_UWord8 FractionLost;
+ WebRtc_UWord32 CumulativeNumOfPacketsLost;
+ WebRtc_UWord32 ExtendedHighestSequenceNumber;
+ WebRtc_UWord32 Jitter;
+ WebRtc_UWord32 LastSR;
+ WebRtc_UWord32 DelayLastSR;
+ };
+ struct RTCPPacketSDESCName
+ {
+ // RFC3550
+ WebRtc_UWord32 SenderSSRC;
+ char CName[RTCP_CNAME_SIZE];
+ };
+
+ struct RTCPPacketExtendedJitterReportItem
+ {
+ // RFC 5450
+ WebRtc_UWord32 Jitter;
+ };
+
+ struct RTCPPacketBYE
+ {
+ WebRtc_UWord32 SenderSSRC;
+ };
+ struct RTCPPacketXR
+ {
+ // RFC 3611
+ WebRtc_UWord32 OriginatorSSRC;
+ };
+ struct RTCPPacketXRVOIPMetricItem
+ {
+ // RFC 3611 4.7
+ WebRtc_UWord32 SSRC;
+ WebRtc_UWord8 lossRate;
+ WebRtc_UWord8 discardRate;
+ WebRtc_UWord8 burstDensity;
+ WebRtc_UWord8 gapDensity;
+ WebRtc_UWord16 burstDuration;
+ WebRtc_UWord16 gapDuration;
+ WebRtc_UWord16 roundTripDelay;
+ WebRtc_UWord16 endSystemDelay;
+ WebRtc_UWord8 signalLevel;
+ WebRtc_UWord8 noiseLevel;
+ WebRtc_UWord8 RERL;
+ WebRtc_UWord8 Gmin;
+ WebRtc_UWord8 Rfactor;
+ WebRtc_UWord8 extRfactor;
+ WebRtc_UWord8 MOSLQ;
+ WebRtc_UWord8 MOSCQ;
+ WebRtc_UWord8 RXconfig;
+ WebRtc_UWord16 JBnominal;
+ WebRtc_UWord16 JBmax;
+ WebRtc_UWord16 JBabsMax;
+ };
+
+ struct RTCPPacketRTPFBNACK
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC;
+ };
+ struct RTCPPacketRTPFBNACKItem
+ {
+ // RFC4585
+ WebRtc_UWord16 PacketID;
+ WebRtc_UWord16 BitMask;
+ };
+
+ struct RTCPPacketRTPFBTMMBR
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC; // zero!
+ };
+ struct RTCPPacketRTPFBTMMBRItem
+ {
+ // RFC5104
+ WebRtc_UWord32 SSRC;
+ WebRtc_UWord32 MaxTotalMediaBitRate; // In Kbit/s
+ WebRtc_UWord32 MeasuredOverhead;
+ };
+
+ struct RTCPPacketRTPFBTMMBN
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC; // zero!
+ };
+ struct RTCPPacketRTPFBTMMBNItem
+ {
+ // RFC5104
+ WebRtc_UWord32 SSRC; // "Owner"
+ WebRtc_UWord32 MaxTotalMediaBitRate;
+ WebRtc_UWord32 MeasuredOverhead;
+ };
+
+ struct RTCPPacketPSFBFIR
+ {
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC; // zero!
+ };
+ struct RTCPPacketPSFBFIRItem
+ {
+ // RFC5104
+ WebRtc_UWord32 SSRC;
+ WebRtc_UWord8 CommandSequenceNumber;
+ };
+
+ struct RTCPPacketPSFBPLI
+ {
+ // RFC4585
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC;
+ };
+
+ struct RTCPPacketPSFBSLI
+ {
+ // RFC4585
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC;
+ };
+ struct RTCPPacketPSFBSLIItem
+ {
+ // RFC4585
+ WebRtc_UWord16 FirstMB;
+ WebRtc_UWord16 NumberOfMB;
+ WebRtc_UWord8 PictureId;
+ };
+ struct RTCPPacketPSFBRPSI
+ {
+ // RFC4585
+ WebRtc_UWord32 SenderSSRC;
+ WebRtc_UWord32 MediaSSRC;
+ WebRtc_UWord8 PayloadType;
+ WebRtc_UWord16 NumberOfValidBits;
+ WebRtc_UWord8 NativeBitString[RTCP_RPSI_DATA_SIZE];
+ };
+ struct RTCPPacketPSFBREMB
+ {
+ WebRtc_UWord32 BitRate;
+ };
+ // generic name APP
+ struct RTCPPacketAPP
+ {
+ WebRtc_UWord8 SubType;
+ WebRtc_UWord32 Name;
+ WebRtc_UWord8 Data[kRtcpAppCode_DATA_SIZE];
+ WebRtc_UWord16 Size;
+ };
+
+ union RTCPPacket
+ {
+ RTCPPacketRR RR;
+ RTCPPacketSR SR;
+ RTCPPacketReportBlockItem ReportBlockItem;
+
+ RTCPPacketSDESCName CName;
+ RTCPPacketBYE BYE;
+
+ RTCPPacketExtendedJitterReportItem ExtendedJitterReportItem;
+
+ RTCPPacketRTPFBNACK NACK;
+ RTCPPacketRTPFBNACKItem NACKItem;
+
+ RTCPPacketPSFBPLI PLI;
+ RTCPPacketPSFBSLI SLI;
+ RTCPPacketPSFBSLIItem SLIItem;
+ RTCPPacketPSFBRPSI RPSI;
+ RTCPPacketPSFBREMB REMB;
+
+ RTCPPacketRTPFBTMMBR TMMBR;
+ RTCPPacketRTPFBTMMBRItem TMMBRItem;
+ RTCPPacketRTPFBTMMBN TMMBN;
+ RTCPPacketRTPFBTMMBNItem TMMBNItem;
+ RTCPPacketPSFBFIR FIR;
+ RTCPPacketPSFBFIRItem FIRItem;
+
+ RTCPPacketXR XR;
+ RTCPPacketXRVOIPMetricItem XRVOIPMetricItem;
+
+ RTCPPacketAPP APP;
+ };
+
+ enum RTCPPacketTypes
+ {
+ kRtcpNotValidCode,
+
+ // RFC3550
+ kRtcpRrCode,
+ kRtcpSrCode,
+ kRtcpReportBlockItemCode,
+
+ kRtcpSdesCode,
+ kRtcpSdesChunkCode,
+ kRtcpByeCode,
+
+ // RFC5450
+ kRtcpExtendedIjCode,
+ kRtcpExtendedIjItemCode,
+
+ // RFC4585
+ kRtcpRtpfbNackCode,
+ kRtcpRtpfbNackItemCode,
+
+ kRtcpPsfbPliCode,
+ kRtcpPsfbRpsiCode,
+ kRtcpPsfbSliCode,
+ kRtcpPsfbSliItemCode,
+ kRtcpPsfbAppCode,
+ kRtcpPsfbRembItemCode,
+
+ // RFC5104
+ kRtcpRtpfbTmmbrCode,
+ kRtcpRtpfbTmmbrItemCode,
+ kRtcpRtpfbTmmbnCode,
+ kRtcpRtpfbTmmbnItemCode,
+ kRtcpPsfbFirCode,
+ kRtcpPsfbFirItemCode,
+
+ // draft-perkins-avt-rapid-rtp-sync
+ kRtcpRtpfbSrReqCode,
+
+ // RFC 3611
+ kRtcpXrVoipMetricCode,
+
+ kRtcpAppCode,
+ kRtcpAppItemCode,
+ };
+
+ struct RTCPRawPacket
+ {
+ const WebRtc_UWord8* _ptrPacketBegin;
+ const WebRtc_UWord8* _ptrPacketEnd;
+ };
+
+ struct RTCPModRawPacket
+ {
+ WebRtc_UWord8* _ptrPacketBegin;
+ WebRtc_UWord8* _ptrPacketEnd;
+ };
+
+ struct RTCPCommonHeader
+ {
+ WebRtc_UWord8 V; // Version
+ bool P; // Padding
+ WebRtc_UWord8 IC; // Item count/subtype
+ WebRtc_UWord8 PT; // Packet Type
+ WebRtc_UWord16 LengthInOctets;
+ };
+
+ enum RTCPPT
+ {
+ PT_IJ = 195,
+ PT_SR = 200,
+ PT_RR = 201,
+ PT_SDES = 202,
+ PT_BYE = 203,
+ PT_APP = 204,
+ PT_RTPFB = 205,
+ PT_PSFB = 206,
+ PT_XR = 207
+ };
+
+ bool RTCPParseCommonHeader( const WebRtc_UWord8* ptrDataBegin,
+ const WebRtc_UWord8* ptrDataEnd,
+ RTCPCommonHeader& parsedHeader);
+
+ class RTCPParserV2
+ {
+ public:
+ RTCPParserV2(const WebRtc_UWord8* rtcpData,
+ size_t rtcpDataLength,
+ bool rtcpReducedSizeEnable); // Set to true, to allow non-compound RTCP!
+ ~RTCPParserV2();
+
+ RTCPPacketTypes PacketType() const;
+ const RTCPPacket& Packet() const;
+ const RTCPRawPacket& RawPacket() const;
+ ptrdiff_t LengthLeft() const;
+
+ bool IsValid() const;
+
+ RTCPPacketTypes Begin();
+ RTCPPacketTypes Iterate();
+
+ private:
+ enum ParseState
+ {
+ State_TopLevel, // Top level packet
+ State_ReportBlockItem, // SR/RR report block
+ State_SDESChunk, // SDES chunk
+ State_BYEItem, // BYE item
+ State_ExtendedJitterItem, // Extended jitter report item
+ State_RTPFB_NACKItem, // NACK FCI item
+ State_RTPFB_TMMBRItem, // TMMBR FCI item
+ State_RTPFB_TMMBNItem, // TMMBN FCI item
+ State_PSFB_SLIItem, // SLI FCI item
+ State_PSFB_RPSIItem, // RPSI FCI item
+ State_PSFB_FIRItem, // FIR FCI item
+ State_PSFB_AppItem, // Application specific FCI item
+ State_PSFB_REMBItem, // Application specific REMB item
+ State_XRItem,
+ State_AppItem
+ };
+
+ private:
+ void IterateTopLevel();
+ void IterateReportBlockItem();
+ void IterateSDESChunk();
+ void IterateBYEItem();
+ void IterateExtendedJitterItem();
+ void IterateNACKItem();
+ void IterateTMMBRItem();
+ void IterateTMMBNItem();
+ void IterateSLIItem();
+ void IterateRPSIItem();
+ void IterateFIRItem();
+ void IteratePsfbAppItem();
+ void IteratePsfbREMBItem();
+ void IterateAppItem();
+
+ void Validate();
+ void EndCurrentBlock();
+
+ bool ParseRR();
+ bool ParseSR();
+ bool ParseReportBlockItem();
+
+ bool ParseSDES();
+ bool ParseSDESChunk();
+ bool ParseSDESItem();
+
+ bool ParseBYE();
+ bool ParseBYEItem();
+
+ bool ParseIJ();
+ bool ParseIJItem();
+
+ bool ParseXR();
+ bool ParseXRItem();
+ bool ParseXRVOIPMetricItem();
+
+ bool ParseFBCommon(const RTCPCommonHeader& header);
+ bool ParseNACKItem();
+ bool ParseTMMBRItem();
+ bool ParseTMMBNItem();
+ bool ParseSLIItem();
+ bool ParseRPSIItem();
+ bool ParseFIRItem();
+ bool ParsePsfbAppItem();
+ bool ParsePsfbREMBItem();
+
+ bool ParseAPP(const RTCPCommonHeader& header);
+ bool ParseAPPItem();
+
+ private:
+ const WebRtc_UWord8* const _ptrRTCPDataBegin;
+ const bool _RTCPReducedSizeEnable;
+ const WebRtc_UWord8* const _ptrRTCPDataEnd;
+
+ bool _validPacket;
+ const WebRtc_UWord8* _ptrRTCPData;
+ const WebRtc_UWord8* _ptrRTCPBlockEnd;
+
+ ParseState _state;
+ WebRtc_UWord8 _numberOfBlocks;
+
+ RTCPPacketTypes _packetType;
+ RTCPPacket _packet;
+ };
+
+ class RTCPPacketIterator
+ {
+ public:
+ RTCPPacketIterator(WebRtc_UWord8* rtcpData,
+ size_t rtcpDataLength);
+ ~RTCPPacketIterator();
+
+ const RTCPCommonHeader* Begin();
+ const RTCPCommonHeader* Iterate();
+ const RTCPCommonHeader* Current();
+
+ private:
+ WebRtc_UWord8* const _ptrBegin;
+ WebRtc_UWord8* const _ptrEnd;
+
+ WebRtc_UWord8* _ptrBlock;
+
+ RTCPCommonHeader _header;
+ };
+} // RTCPUtility
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc
new file mode 100644
index 0000000..d794701
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -0,0 +1,550 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+#include <gtest/gtest.h>
+#include <list>
+
+#include "rtp_utility.h"
+
+using webrtc::ForwardErrorCorrection;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
+const uint8_t kTransportOverhead = 28;
+
+// Maximum number of media packets used in the FEC (RFC 5109).
+const uint8_t kMaxNumberMediaPackets = ForwardErrorCorrection::kMaxMediaPackets;
+
+template<typename T> void ClearList(std::list<T*>* my_list) {
+ T* packet = NULL;
+ while (!my_list->empty()) {
+ packet = my_list->front();
+ delete packet;
+ my_list->pop_front();
+ }
+}
+
+class RtpFecTest : public ::testing::Test {
+ protected:
+ RtpFecTest()
+ : fec_(new ForwardErrorCorrection(0)),
+ ssrc_(rand()),
+ fec_seq_num_(0) {
+ }
+
+ ForwardErrorCorrection* fec_;
+ int ssrc_;
+ uint16_t fec_seq_num_;
+
+ std::list<ForwardErrorCorrection::Packet*> media_packet_list_;
+ std::list<ForwardErrorCorrection::Packet*> fec_packet_list_;
+ std::list<ForwardErrorCorrection::ReceivedPacket*> received_packet_list_;
+ std::list<ForwardErrorCorrection::RecoveredPacket*> recovered_packet_list_;
+
+ // Media packet "i" is lost if media_loss_mask_[i] = 1,
+ // received if media_loss_mask_[i] = 0.
+ int media_loss_mask_[kMaxNumberMediaPackets];
+
+ // FEC packet "i" is lost if fec_loss_mask_[i] = 1,
+ // received if fec_loss_mask_[i] = 0.
+ int fec_loss_mask_[kMaxNumberMediaPackets];
+
+ // Construct the media packet list, up to |num_media_packets| packets.
+ // Returns the next sequence number after the last media packet.
+ // (this will be the sequence of the first FEC packet)
+ int ConstructMediaPackets(int num_media_packets);
+
+ // Construct the received packet list: a subset of the media and FEC packets.
+ void NetworkReceivedPackets();
+
+ // Add packet from |packet_list| to list of received packets, using the
+ // |loss_mask|.
+ // The |packet_list| may be a media packet list (is_fec = false), or a
+ // FEC packet list (is_fec = true).
+ void ReceivedPackets(
+ const std::list<ForwardErrorCorrection::Packet*>& packet_list,
+ int* loss_mask,
+ bool is_fec);
+
+ // Check for complete recovery after FEC decoding.
+ bool IsRecoveryComplete();
+
+ // Delete the received packets.
+ void FreeRecoveredPacketList();
+
+ // Delete the media and FEC packets.
+ void TearDown();
+};
+
+TEST_F(RtpFecTest, HandleIncorrectInputs) {
+ int num_important_packets = 0;
+ bool use_unequal_protection = false;
+ uint8_t protection_factor = 60;
+
+ // Media packet list is empty.
+ EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ int num_media_packets = 10;
+ ConstructMediaPackets(num_media_packets);
+
+ num_important_packets = -1;
+ // Number of important packets below 0.
+ EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ num_important_packets = 12;
+ // Number of important packets greater than number of media packets.
+ EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ num_media_packets = kMaxNumberMediaPackets + 1;
+ ConstructMediaPackets(num_media_packets);
+
+ num_important_packets = 0;
+ // Number of media packet is above maximum allowed (kMaxNumberMediaPackets).
+ EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+}
+
+TEST_F(RtpFecTest, FecRecoveryNoLoss) {
+ const int num_important_packets = 0;
+ const bool use_unequal_protection = false;
+ const int num_media_packets = 4;
+ uint8_t protection_factor = 60;
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 1 FEC packet.
+ EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
+
+ // No packets lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // No packets lost, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+}
+
+TEST_F(RtpFecTest, FecRecoveryWithLoss) {
+ const int num_important_packets = 0;
+ const bool use_unequal_protection = false;
+ const int num_media_packets = 4;
+ uint8_t protection_factor = 60;
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 1 FEC packet.
+ EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
+
+ // 1 media packet lost
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // One packet lost, one FEC packet, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+ FreeRecoveredPacketList();
+
+ // 2 media packets lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ media_loss_mask_[1] = 1;
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // 2 packets lost, one FEC packet, cannot get complete recovery.
+ EXPECT_FALSE(IsRecoveryComplete());
+}
+
+TEST_F(RtpFecTest, FecRecoveryWithLoss50perc) {
+ const int num_important_packets = 0;
+ const bool use_unequal_protection = false;
+ const int num_media_packets = 4;
+ const uint8_t protection_factor = 255;
+
+ // Packet Mask for (4,4,0) code:
+ // (num_media_packets = 4; num_fec_packets = 4, num_important_packets = 0)
+
+ // media#0 media#1 media#2 media#3
+ // fec#0: 1 1 0 0
+ // fec#1: 1 0 1 0
+ // fec#2: 0 1 0 1
+ // fec#3: 0 0 1 1
+ //
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 4 FEC packets.
+ EXPECT_EQ(4, static_cast<int>(fec_packet_list_.size()));
+
+ // 4 packets lost: 3 media packets and one FEC packet#2 lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ fec_loss_mask_[2] = 1;
+ media_loss_mask_[0] = 1;
+ media_loss_mask_[2] = 1;
+ media_loss_mask_[3] = 1;
+
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // With media packet#1 and FEC packets #0, #1, #3, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+ FreeRecoveredPacketList();
+
+ // 4 packets lost: all media packets
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 1, sizeof(fec_loss_mask_));
+ media_loss_mask_[0] = 1;
+ media_loss_mask_[1] = 1;
+ media_loss_mask_[2] = 1;
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // Cannot get complete recovery for this loss configuration.
+ EXPECT_FALSE(IsRecoveryComplete());
+}
+
+TEST_F(RtpFecTest, FecRecoveryNoLossUep) {
+ const int num_important_packets = 2;
+ const bool use_unequal_protection = true;
+ const int num_media_packets = 4;
+ const uint8_t protection_factor = 60;
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 1 FEC packet.
+ EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
+
+ // No packets lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // No packets lost, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+}
+
+TEST_F(RtpFecTest, FecRecoveryWithLossUep) {
+ const int num_important_packets = 2;
+ const bool use_unequal_protection = true;
+ const int num_media_packets = 4;
+ const uint8_t protection_factor = 60;
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 1 FEC packet.
+ EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
+
+ // 1 media packet lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // One packet lost, one FEC packet, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+ FreeRecoveredPacketList();
+
+ // 2 media packets lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ media_loss_mask_[1] = 1;
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // 2 packets lost, one FEC packet, cannot get complete recovery.
+ EXPECT_FALSE(IsRecoveryComplete());
+}
+
+TEST_F(RtpFecTest, FecRecoveryWithLoss50percUep) {
+ const int num_important_packets = 1;
+ const bool use_unequal_protection = true;
+ const int num_media_packets = 4;
+ const uint8_t protection_factor = 255;
+
+ // Packet Mask for (4,4,1) code:
+ // (num_media_packets = 4; num_fec_packets = 4, num_important_packets = 2)
+
+ // media#0 media#1 media#2 media#3
+ // fec#0: 1 0 0 0
+ // fec#1: 1 1 0 0
+ // fec#2: 1 0 1 1
+ // fec#3: 0 1 1 0
+ //
+
+ fec_seq_num_ = ConstructMediaPackets(num_media_packets);
+
+ EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
+ protection_factor,
+ num_important_packets,
+ use_unequal_protection,
+ &fec_packet_list_));
+
+ // Expect 4 FEC packets.
+ EXPECT_EQ(4, static_cast<int>(fec_packet_list_.size()));
+
+ // 4 packets lost: 3 media packets and FEC packet#1 lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ fec_loss_mask_[1] = 1;
+ media_loss_mask_[0] = 1;
+ media_loss_mask_[2] = 1;
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // With media packet#1 and FEC packets #0, #2, #3, expect complete recovery.
+ EXPECT_TRUE(IsRecoveryComplete());
+ FreeRecoveredPacketList();
+
+ // 4 packets lost: 3 media packets and one FEC packet#2 lost.
+ memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
+ memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
+ fec_loss_mask_[2] = 1;
+ media_loss_mask_[0] = 1;
+ media_loss_mask_[2] = 1;
+ media_loss_mask_[3] = 1;
+ NetworkReceivedPackets();
+
+ EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
+ &recovered_packet_list_));
+
+ // Cannot get complete recovery for this loss configuration.
+ EXPECT_FALSE(IsRecoveryComplete());
+}
+
+// TODO(marpan): Add more test cases.
+
+void RtpFecTest::TearDown() {
+ fec_->ResetState(&recovered_packet_list_);
+ delete fec_;
+ FreeRecoveredPacketList();
+ ClearList(&media_packet_list_);
+ EXPECT_TRUE(media_packet_list_.empty());
+}
+
+void RtpFecTest::FreeRecoveredPacketList() {
+ ClearList(&recovered_packet_list_);
+}
+
+bool RtpFecTest::IsRecoveryComplete() {
+ // Check that the number of media and recovered packets are equal.
+ if (media_packet_list_.size() != recovered_packet_list_.size()) {
+ return false;
+ }
+
+ ForwardErrorCorrection::Packet* media_packet;
+ ForwardErrorCorrection::RecoveredPacket* recovered_packet;
+
+ bool recovery = true;
+
+ std::list<ForwardErrorCorrection::Packet*>::iterator
+ media_packet_list_item = media_packet_list_.begin();
+ std::list<ForwardErrorCorrection::RecoveredPacket*>::iterator
+ recovered_packet_list_item = recovered_packet_list_.begin();
+ while (media_packet_list_item != media_packet_list_.end()) {
+ if (recovered_packet_list_item == recovered_packet_list_.end()) {
+ return false;
+ }
+ media_packet = *media_packet_list_item;
+ recovered_packet = *recovered_packet_list_item;
+ if (recovered_packet->pkt->length != media_packet->length) {
+ return false;
+ }
+ if (memcmp(recovered_packet->pkt->data, media_packet->data,
+ media_packet->length) != 0) {
+ return false;
+ }
+ media_packet_list_item++;
+ recovered_packet_list_item++;
+ }
+ return recovery;
+}
+
+void RtpFecTest::NetworkReceivedPackets() {
+ const bool kFecPacket = true;
+ ReceivedPackets(media_packet_list_, media_loss_mask_, !kFecPacket);
+ ReceivedPackets(fec_packet_list_, fec_loss_mask_, kFecPacket);
+}
+
+void RtpFecTest:: ReceivedPackets(
+ const std::list<ForwardErrorCorrection::Packet*>& packet_list,
+ int* loss_mask,
+ bool is_fec) {
+ ForwardErrorCorrection::Packet* packet;
+ ForwardErrorCorrection::ReceivedPacket* received_packet;
+ int seq_num = fec_seq_num_;
+ int packet_idx = 0;
+
+ std::list<ForwardErrorCorrection::Packet*>::const_iterator
+ packet_list_item = packet_list.begin();
+
+ while (packet_list_item != packet_list.end()) {
+ packet = *packet_list_item;
+ if (loss_mask[packet_idx] == 0) {
+ received_packet = new ForwardErrorCorrection::ReceivedPacket;
+ received_packet->pkt = new ForwardErrorCorrection::Packet;
+ received_packet_list_.push_back(received_packet);
+ received_packet->pkt->length = packet->length;
+ memcpy(received_packet->pkt->data, packet->data,
+ packet->length);
+ received_packet->isFec = is_fec;
+ if (!is_fec) {
+ // For media packets, the sequence number and marker bit is
+ // obtained from RTP header. These were set in ConstructMediaPackets().
+ received_packet->seqNum =
+ webrtc::ModuleRTPUtility::BufferToUWord16(&packet->data[2]);
+ }
+ else {
+ // The sequence number, marker bit, and ssrc number are defined in the
+ // RTP header of the FEC packet, which is not constructed in this test.
+ // So we set these values below based on the values generated in
+ // ConstructMediaPackets().
+ received_packet->seqNum = seq_num;
+ // The ssrc value for FEC packets is set to the one used for the
+ // media packets in ConstructMediaPackets().
+ received_packet->ssrc = ssrc_;
+ }
+ }
+ packet_idx++;
+ packet_list_item ++;
+ // Sequence number of FEC packets are defined as increment by 1 from
+ // last media packet in frame.
+ if (is_fec) seq_num++;
+ }
+}
+
+int RtpFecTest::ConstructMediaPackets(int num_media_packets) {
+ assert(num_media_packets > 0);
+ ForwardErrorCorrection::Packet* media_packet = NULL;
+ int sequence_number = rand();
+ int time_stamp = rand();
+
+ for (int i = 0; i < num_media_packets; i++) {
+ media_packet = new ForwardErrorCorrection::Packet;
+ media_packet_list_.push_back(media_packet);
+ media_packet->length =
+ static_cast<uint16_t>((static_cast<float>(rand()) / RAND_MAX) *
+ (IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
+ ForwardErrorCorrection::PacketOverhead()));
+
+ if (media_packet->length < kRtpHeaderSize) {
+ media_packet->length = kRtpHeaderSize;
+ }
+ // Generate random values for the first 2 bytes
+ media_packet->data[0] = static_cast<uint8_t>(rand() % 256);
+ media_packet->data[1] = static_cast<uint8_t>(rand() % 256);
+
+ // The first two bits are assumed to be 10 by the FEC encoder.
+ // In fact the FEC decoder will set the two first bits to 10 regardless of
+ // what they actually were. Set the first two bits to 10 so that a memcmp
+ // can be performed for the whole restored packet.
+ media_packet->data[0] |= 0x80;
+ media_packet->data[0] &= 0xbf;
+
+ // FEC is applied to a whole frame.
+ // A frame is signaled by multiple packets without the marker bit set
+ // followed by the last packet of the frame for which the marker bit is set.
+ // Only push one (fake) frame to the FEC.
+ media_packet->data[1] &= 0x7f;
+
+ webrtc::ModuleRTPUtility::AssignUWord16ToBuffer(&media_packet->data[2],
+ sequence_number);
+ webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[4],
+ time_stamp);
+ webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[8],
+ ssrc_);
+
+ // Generate random values for payload.
+ for (int j = 12; j < media_packet->length; j++) {
+ media_packet->data[j] = static_cast<uint8_t> (rand() % 256);
+ }
+ sequence_number++;
+ }
+ // Last packet, set marker bit.
+ assert(media_packet != NULL);
+ media_packet->data[1] |= 0x80;
+ return sequence_number;
+}
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc
new file mode 100644
index 0000000..f066ae2
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -0,0 +1,469 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+
+#include <string.h> // memcpy
+
+#include <cassert> // assert
+#include <vector>
+
+#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+
+namespace webrtc {
+
+// Define how the VP8PacketizerModes are implemented.
+// Modes are: kStrict, kAggregate, kEqualSize.
+const RtpFormatVp8::AggregationMode RtpFormatVp8::aggr_modes_[kNumModes] =
+ { kAggrNone, kAggrPartitions, kAggrFragments };
+const bool RtpFormatVp8::balance_modes_[kNumModes] =
+ { true, true, true };
+const bool RtpFormatVp8::separate_first_modes_[kNumModes] =
+ { true, false, false };
+
+RtpFormatVp8::RtpFormatVp8(const WebRtc_UWord8* payload_data,
+ WebRtc_UWord32 payload_size,
+ const RTPVideoHeaderVP8& hdr_info,
+ int max_payload_len,
+ const RTPFragmentationHeader& fragmentation,
+ VP8PacketizerMode mode)
+ : payload_data_(payload_data),
+ payload_size_(static_cast<int>(payload_size)),
+ vp8_fixed_payload_descriptor_bytes_(1),
+ aggr_mode_(aggr_modes_[mode]),
+ balance_(balance_modes_[mode]),
+ separate_first_(separate_first_modes_[mode]),
+ hdr_info_(hdr_info),
+ num_partitions_(fragmentation.fragmentationVectorSize),
+ max_payload_len_(max_payload_len),
+ packets_calculated_(false) {
+ part_info_ = fragmentation;
+}
+
+RtpFormatVp8::RtpFormatVp8(const WebRtc_UWord8* payload_data,
+ WebRtc_UWord32 payload_size,
+ const RTPVideoHeaderVP8& hdr_info,
+ int max_payload_len)
+ : payload_data_(payload_data),
+ payload_size_(static_cast<int>(payload_size)),
+ part_info_(),
+ vp8_fixed_payload_descriptor_bytes_(1),
+ aggr_mode_(aggr_modes_[kEqualSize]),
+ balance_(balance_modes_[kEqualSize]),
+ separate_first_(separate_first_modes_[kEqualSize]),
+ hdr_info_(hdr_info),
+ num_partitions_(1),
+ max_payload_len_(max_payload_len),
+ packets_calculated_(false) {
+ part_info_.VerifyAndAllocateFragmentationHeader(1);
+ part_info_.fragmentationLength[0] = payload_size;
+ part_info_.fragmentationOffset[0] = 0;
+}
+
+int RtpFormatVp8::NextPacket(WebRtc_UWord8* buffer,
+ int* bytes_to_send,
+ bool* last_packet) {
+ if (!packets_calculated_) {
+ int ret = 0;
+ if (aggr_mode_ == kAggrPartitions && balance_) {
+ ret = GeneratePacketsBalancedAggregates();
+ } else {
+ ret = GeneratePackets();
+ }
+ if (ret < 0) {
+ return ret;
+ }
+ }
+ if (packets_.empty()) {
+ return -1;
+ }
+ InfoStruct packet_info = packets_.front();
+ packets_.pop();
+
+ *bytes_to_send = WriteHeaderAndPayload(packet_info, buffer, max_payload_len_);
+ if (*bytes_to_send < 0) {
+ return -1;
+ }
+
+ *last_packet = packets_.empty();
+ return packet_info.first_partition_ix;
+}
+
+int RtpFormatVp8::CalcNextSize(int max_payload_len, int remaining_bytes,
+ bool split_payload) const {
+ if (max_payload_len == 0 || remaining_bytes == 0) {
+ return 0;
+ }
+ if (!split_payload) {
+ return max_payload_len >= remaining_bytes ? remaining_bytes : 0;
+ }
+
+ if (balance_) {
+ // Balance payload sizes to produce (almost) equal size
+ // fragments.
+ // Number of fragments for remaining_bytes:
+ int num_frags = remaining_bytes / max_payload_len + 1;
+ // Number of bytes in this fragment:
+ return static_cast<int>(static_cast<double>(remaining_bytes)
+ / num_frags + 0.5);
+ } else {
+ return max_payload_len >= remaining_bytes ? remaining_bytes
+ : max_payload_len;
+ }
+}
+
+int RtpFormatVp8::GeneratePackets() {
+ if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
+ + PayloadDescriptorExtraLength() + 1) {
+ // The provided payload length is not long enough for the payload
+ // descriptor and one payload byte. Return an error.
+ return -1;
+ }
+ int total_bytes_processed = 0;
+ bool start_on_new_fragment = true;
+ bool beginning = true;
+ int part_ix = 0;
+ while (total_bytes_processed < payload_size_) {
+ int packet_bytes = 0; // How much data to send in this packet.
+ bool split_payload = true; // Splitting of partitions is initially allowed.
+ int remaining_in_partition = part_info_.fragmentationOffset[part_ix] -
+ total_bytes_processed + part_info_.fragmentationLength[part_ix];
+ int rem_payload_len = max_payload_len_ -
+ (vp8_fixed_payload_descriptor_bytes_ + PayloadDescriptorExtraLength());
+ int first_partition_in_packet = part_ix;
+
+ while (int next_size = CalcNextSize(rem_payload_len, remaining_in_partition,
+ split_payload)) {
+ packet_bytes += next_size;
+ rem_payload_len -= next_size;
+ remaining_in_partition -= next_size;
+
+ if (remaining_in_partition == 0 && !(beginning && separate_first_)) {
+ // Advance to next partition?
+ // Check that there are more partitions; verify that we are either
+ // allowed to aggregate fragments, or that we are allowed to
+ // aggregate intact partitions and that we started this packet
+ // with an intact partition (indicated by first_fragment_ == true).
+ if (part_ix + 1 < num_partitions_ &&
+ ((aggr_mode_ == kAggrFragments) ||
+ (aggr_mode_ == kAggrPartitions && start_on_new_fragment))) {
+ assert(part_ix < num_partitions_);
+ remaining_in_partition = part_info_.fragmentationLength[++part_ix];
+ // Disallow splitting unless kAggrFragments. In kAggrPartitions,
+ // we can only aggregate intact partitions.
+ split_payload = (aggr_mode_ == kAggrFragments);
+ }
+ } else if (balance_ && remaining_in_partition > 0) {
+ break;
+ }
+ }
+ if (remaining_in_partition == 0) {
+ ++part_ix; // Advance to next partition.
+ }
+ assert(packet_bytes > 0);
+
+ QueuePacket(total_bytes_processed, packet_bytes, first_partition_in_packet,
+ start_on_new_fragment);
+ total_bytes_processed += packet_bytes;
+ start_on_new_fragment = (remaining_in_partition == 0);
+ beginning = false; // Next packet cannot be first packet in frame.
+ }
+ packets_calculated_ = true;
+ assert(total_bytes_processed == payload_size_);
+ return 0;
+}
+
+int RtpFormatVp8::GeneratePacketsBalancedAggregates() {
+ if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
+ + PayloadDescriptorExtraLength() + 1) {
+ // The provided payload length is not long enough for the payload
+ // descriptor and one payload byte. Return an error.
+ return -1;
+ }
+ std::vector<int> partition_decision;
+ const int overhead = vp8_fixed_payload_descriptor_bytes_ +
+ PayloadDescriptorExtraLength();
+ const uint32_t max_payload_len = max_payload_len_ - overhead;
+ int min_size, max_size;
+ AggregateSmallPartitions(&partition_decision, &min_size, &max_size);
+
+ int total_bytes_processed = 0;
+ int part_ix = 0;
+ while (part_ix < num_partitions_) {
+ if (partition_decision[part_ix] == -1) {
+ // Split large partitions.
+ int remaining_partition = part_info_.fragmentationLength[part_ix];
+ int num_fragments = Vp8PartitionAggregator::CalcNumberOfFragments(
+ remaining_partition, max_payload_len, overhead, min_size, max_size);
+ const int packet_bytes =
+ (remaining_partition + num_fragments - 1) / num_fragments;
+ for (int n = 0; n < num_fragments; ++n) {
+ const int this_packet_bytes = packet_bytes < remaining_partition ?
+ packet_bytes : remaining_partition;
+ QueuePacket(total_bytes_processed, this_packet_bytes, part_ix,
+ (n == 0));
+ remaining_partition -= this_packet_bytes;
+ total_bytes_processed += this_packet_bytes;
+ if (this_packet_bytes < min_size) {
+ min_size = this_packet_bytes;
+ }
+ if (this_packet_bytes > max_size) {
+ max_size = this_packet_bytes;
+ }
+ }
+ assert(remaining_partition == 0);
+ ++part_ix;
+ } else {
+ int this_packet_bytes = 0;
+ const int first_partition_in_packet = part_ix;
+ const int aggregation_index = partition_decision[part_ix];
+ while (static_cast<size_t>(part_ix) < partition_decision.size() &&
+ partition_decision[part_ix] == aggregation_index) {
+ // Collect all partitions that were aggregated into the same packet.
+ this_packet_bytes += part_info_.fragmentationLength[part_ix];
+ ++part_ix;
+ }
+ QueuePacket(total_bytes_processed, this_packet_bytes,
+ first_partition_in_packet, true);
+ total_bytes_processed += this_packet_bytes;
+ }
+ }
+ packets_calculated_ = true;
+ return 0;
+}
+
+void RtpFormatVp8::AggregateSmallPartitions(std::vector<int>* partition_vec,
+ int* min_size,
+ int* max_size) {
+ assert(min_size && max_size);
+ *min_size = -1;
+ *max_size = -1;
+ assert(partition_vec);
+ partition_vec->assign(num_partitions_, -1);
+ const int overhead = vp8_fixed_payload_descriptor_bytes_ +
+ PayloadDescriptorExtraLength();
+ const uint32_t max_payload_len = max_payload_len_ - overhead;
+ int first_in_set = 0;
+ int last_in_set = 0;
+ int num_aggregate_packets = 0;
+ // Find sets of partitions smaller than max_payload_len_.
+ while (first_in_set < num_partitions_) {
+ if (part_info_.fragmentationLength[first_in_set] < max_payload_len) {
+ // Found start of a set.
+ last_in_set = first_in_set;
+ while (last_in_set + 1 < num_partitions_ &&
+ part_info_.fragmentationLength[last_in_set + 1] < max_payload_len) {
+ ++last_in_set;
+ }
+ // Found end of a set. Run optimized aggregator. It is ok if start == end.
+ Vp8PartitionAggregator aggregator(part_info_, first_in_set,
+ last_in_set);
+ if (*min_size >= 0 && *max_size >= 0) {
+ aggregator.SetPriorMinMax(*min_size, *max_size);
+ }
+ Vp8PartitionAggregator::ConfigVec optimal_config =
+ aggregator.FindOptimalConfiguration(max_payload_len, overhead);
+ aggregator.CalcMinMax(optimal_config, min_size, max_size);
+ for (int i = first_in_set, j = 0; i <= last_in_set; ++i, ++j) {
+ // Transfer configuration for this set of partitions to the joint
+ // partition vector representing all partitions in the frame.
+ (*partition_vec)[i] = num_aggregate_packets + optimal_config[j];
+ }
+ num_aggregate_packets += optimal_config.back() + 1;
+ first_in_set = last_in_set;
+ }
+ ++first_in_set;
+ }
+}
+
+void RtpFormatVp8::QueuePacket(int start_pos,
+ int packet_size,
+ int first_partition_in_packet,
+ bool start_on_new_fragment) {
+ // Write info to packet info struct and store in packet info queue.
+ InfoStruct packet_info;
+ packet_info.payload_start_pos = start_pos;
+ packet_info.size = packet_size;
+ packet_info.first_partition_ix = first_partition_in_packet;
+ packet_info.first_fragment = start_on_new_fragment;
+ packets_.push(packet_info);
+}
+
+int RtpFormatVp8::WriteHeaderAndPayload(const InfoStruct& packet_info,
+ WebRtc_UWord8* buffer,
+ int buffer_length) const {
+ // Write the VP8 payload descriptor.
+ // 0
+ // 0 1 2 3 4 5 6 7 8
+ // +-+-+-+-+-+-+-+-+-+
+ // |X| |N|S| PART_ID |
+ // +-+-+-+-+-+-+-+-+-+
+ // X: |I|L|T|K| | (mandatory if any of the below are used)
+ // +-+-+-+-+-+-+-+-+-+
+ // I: |PictureID (8/16b)| (optional)
+ // +-+-+-+-+-+-+-+-+-+
+ // L: | TL0PIC_IDX | (optional)
+ // +-+-+-+-+-+-+-+-+-+
+ // T/K: |TID:Y| KEYIDX | (optional)
+ // +-+-+-+-+-+-+-+-+-+
+
+ assert(packet_info.size > 0);
+ buffer[0] = 0;
+ if (XFieldPresent()) buffer[0] |= kXBit;
+ if (hdr_info_.nonReference) buffer[0] |= kNBit;
+ if (packet_info.first_fragment) buffer[0] |= kSBit;
+ buffer[0] |= (packet_info.first_partition_ix & kPartIdField);
+
+ const int extension_length = WriteExtensionFields(buffer, buffer_length);
+
+ memcpy(&buffer[vp8_fixed_payload_descriptor_bytes_ + extension_length],
+ &payload_data_[packet_info.payload_start_pos], packet_info.size);
+
+ // Return total length of written data.
+ return packet_info.size + vp8_fixed_payload_descriptor_bytes_
+ + extension_length;
+}
+
+int RtpFormatVp8::WriteExtensionFields(WebRtc_UWord8* buffer,
+ int buffer_length) const {
+ int extension_length = 0;
+ if (XFieldPresent()) {
+ WebRtc_UWord8* x_field = buffer + vp8_fixed_payload_descriptor_bytes_;
+ *x_field = 0;
+ extension_length = 1; // One octet for the X field.
+ if (PictureIdPresent()) {
+ if (WritePictureIDFields(x_field, buffer, buffer_length,
+ &extension_length) < 0) {
+ return -1;
+ }
+ }
+ if (TL0PicIdxFieldPresent()) {
+ if (WriteTl0PicIdxFields(x_field, buffer, buffer_length,
+ &extension_length) < 0) {
+ return -1;
+ }
+ }
+ if (TIDFieldPresent() || KeyIdxFieldPresent()) {
+ if (WriteTIDAndKeyIdxFields(x_field, buffer, buffer_length,
+ &extension_length) < 0) {
+ return -1;
+ }
+ }
+ assert(extension_length == PayloadDescriptorExtraLength());
+ }
+ return extension_length;
+}
+
+int RtpFormatVp8::WritePictureIDFields(WebRtc_UWord8* x_field,
+ WebRtc_UWord8* buffer,
+ int buffer_length,
+ int* extension_length) const {
+ *x_field |= kIBit;
+ const int pic_id_length = WritePictureID(
+ buffer + vp8_fixed_payload_descriptor_bytes_ + *extension_length,
+ buffer_length - vp8_fixed_payload_descriptor_bytes_
+ - *extension_length);
+ if (pic_id_length < 0) return -1;
+ *extension_length += pic_id_length;
+ return 0;
+}
+
+int RtpFormatVp8::WritePictureID(WebRtc_UWord8* buffer,
+ int buffer_length) const {
+ const WebRtc_UWord16 pic_id =
+ static_cast<WebRtc_UWord16> (hdr_info_.pictureId);
+ int picture_id_len = PictureIdLength();
+ if (picture_id_len > buffer_length) return -1;
+ if (picture_id_len == 2) {
+ buffer[0] = 0x80 | ((pic_id >> 8) & 0x7F);
+ buffer[1] = pic_id & 0xFF;
+ } else if (picture_id_len == 1) {
+ buffer[0] = pic_id & 0x7F;
+ }
+ return picture_id_len;
+}
+
+int RtpFormatVp8::WriteTl0PicIdxFields(WebRtc_UWord8* x_field,
+ WebRtc_UWord8* buffer,
+ int buffer_length,
+ int* extension_length) const {
+ if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
+ + 1) {
+ return -1;
+ }
+ *x_field |= kLBit;
+ buffer[vp8_fixed_payload_descriptor_bytes_
+ + *extension_length] = hdr_info_.tl0PicIdx;
+ ++*extension_length;
+ return 0;
+}
+
+int RtpFormatVp8::WriteTIDAndKeyIdxFields(WebRtc_UWord8* x_field,
+ WebRtc_UWord8* buffer,
+ int buffer_length,
+ int* extension_length) const {
+ if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
+ + 1) {
+ return -1;
+ }
+ WebRtc_UWord8* data_field =
+ &buffer[vp8_fixed_payload_descriptor_bytes_ + *extension_length];
+ *data_field = 0;
+ if (TIDFieldPresent()) {
+ *x_field |= kTBit;
+ assert(hdr_info_.temporalIdx >= 0 && hdr_info_.temporalIdx <= 3);
+ *data_field |= hdr_info_.temporalIdx << 6;
+ *data_field |= hdr_info_.layerSync ? kYBit : 0;
+ }
+ if (KeyIdxFieldPresent()) {
+ *x_field |= kKBit;
+ *data_field |= (hdr_info_.keyIdx & kKeyIdxField);
+ }
+ ++*extension_length;
+ return 0;
+}
+
+int RtpFormatVp8::PayloadDescriptorExtraLength() const {
+ int length_bytes = PictureIdLength();
+ if (TL0PicIdxFieldPresent()) ++length_bytes;
+ if (TIDFieldPresent() || KeyIdxFieldPresent()) ++length_bytes;
+ if (length_bytes > 0) ++length_bytes; // Include the extension field.
+ return length_bytes;
+}
+
+int RtpFormatVp8::PictureIdLength() const {
+ if (hdr_info_.pictureId == kNoPictureId) {
+ return 0;
+ }
+ if (hdr_info_.pictureId <= 0x7F) {
+ return 1;
+ }
+ return 2;
+}
+
+bool RtpFormatVp8::XFieldPresent() const {
+ return (TIDFieldPresent() || TL0PicIdxFieldPresent() || PictureIdPresent()
+ || KeyIdxFieldPresent());
+}
+
+bool RtpFormatVp8::TIDFieldPresent() const {
+ assert((hdr_info_.layerSync == false) ||
+ (hdr_info_.temporalIdx != kNoTemporalIdx));
+ return (hdr_info_.temporalIdx != kNoTemporalIdx);
+}
+
+bool RtpFormatVp8::KeyIdxFieldPresent() const {
+ return (hdr_info_.keyIdx != kNoKeyIdx);
+}
+
+bool RtpFormatVp8::TL0PicIdxFieldPresent() const {
+ return (hdr_info_.tl0PicIdx != kNoTl0PicIdx);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h
new file mode 100644
index 0000000..f568f4d
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -0,0 +1,206 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the declaration of the VP8 packetizer class.
+ * A packetizer object is created for each encoded video frame. The
+ * constructor is called with the payload data and size,
+ * together with the fragmentation information and a packetizer mode
+ * of choice. Alternatively, if no fragmentation info is available, the
+ * second constructor can be used with only payload data and size; in that
+ * case the mode kEqualSize is used.
+ *
+ * After creating the packetizer, the method NextPacket is called
+ * repeatedly to get all packets for the frame. The method returns
+ * false as long as there are more packets left to fetch.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
+
+#include <queue>
+#include <vector>
+
+#include "modules/interface/module_common_types.h"
+#include "system_wrappers/interface/constructor_magic.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+enum VP8PacketizerMode {
+ kStrict = 0, // Split partitions if too large;
+ // never aggregate, balance size.
+ kAggregate, // Split partitions if too large; aggregate whole partitions.
+ kEqualSize, // Split entire payload without considering partition limits.
+ // This will produce equal size packets for the whole frame.
+ kNumModes,
+};
+
+// Packetizer for VP8.
+class RtpFormatVp8 {
+ public:
+ // Initialize with payload from encoder and fragmentation info.
+ // The payload_data must be exactly one encoded VP8 frame.
+ RtpFormatVp8(const WebRtc_UWord8* payload_data,
+ WebRtc_UWord32 payload_size,
+ const RTPVideoHeaderVP8& hdr_info,
+ int max_payload_len,
+ const RTPFragmentationHeader& fragmentation,
+ VP8PacketizerMode mode);
+
+ // Initialize without fragmentation info. Mode kEqualSize will be used.
+ // The payload_data must be exactly one encoded VP8 frame.
+ RtpFormatVp8(const WebRtc_UWord8* payload_data,
+ WebRtc_UWord32 payload_size,
+ const RTPVideoHeaderVP8& hdr_info,
+ int max_payload_len);
+
+ // Get the next payload with VP8 payload header.
+ // max_payload_len limits the sum length of payload and VP8 payload header.
+ // buffer is a pointer to where the output will be written.
+ // bytes_to_send is an output variable that will contain number of bytes
+ // written to buffer. Parameter last_packet is true for the last packet of
+ // the frame, false otherwise (i.e., call the function again to get the
+ // next packet).
+ // For the kStrict and kAggregate mode: returns the partition index from which
+ // the first payload byte in the packet is taken, with the first partition
+ // having index 0; returns negative on error.
+ // For the kEqualSize mode: returns 0 on success, return negative on error.
+ int NextPacket(WebRtc_UWord8* buffer,
+ int* bytes_to_send,
+ bool* last_packet);
+
+ private:
+ typedef struct {
+ int payload_start_pos;
+ int size;
+ bool first_fragment;
+ int first_partition_ix;
+ } InfoStruct;
+ typedef std::queue<InfoStruct> InfoQueue;
+ enum AggregationMode {
+ kAggrNone = 0, // No aggregation.
+ kAggrPartitions, // Aggregate intact partitions.
+ kAggrFragments // Aggregate intact and fragmented partitions.
+ };
+
+ static const AggregationMode aggr_modes_[kNumModes];
+ static const bool balance_modes_[kNumModes];
+ static const bool separate_first_modes_[kNumModes];
+ static const int kXBit = 0x80;
+ static const int kNBit = 0x20;
+ static const int kSBit = 0x10;
+ static const int kPartIdField = 0x0F;
+ static const int kKeyIdxField = 0x1F;
+ static const int kIBit = 0x80;
+ static const int kLBit = 0x40;
+ static const int kTBit = 0x20;
+ static const int kKBit = 0x10;
+ static const int kYBit = 0x20;
+
+ // Calculate size of next chunk to send. Returns 0 if none can be sent.
+ int CalcNextSize(int max_payload_len, int remaining_bytes,
+ bool split_payload) const;
+
+ // Calculate all packet sizes and load to packet info queue.
+ int GeneratePackets();
+
+ // Calculate all packet sizes using Vp8PartitionAggregator and load to packet
+ // info queue.
+ int GeneratePacketsBalancedAggregates();
+
+ // Helper function to GeneratePacketsBalancedAggregates(). Find all
+ // continuous sets of partitions smaller than the max payload size (not
+ // max_size), and aggregate them into balanced packets. The result is written
+ // to partition_vec, which is of the same length as the number of partitions.
+ // A value of -1 indicates that the partition is too large and must be split.
+ // Aggregates are numbered 0, 1, 2, etc. For each set of small partitions,
+ // the aggregate numbers restart at 0. Output values min_size and max_size
+ // will hold the smallest and largest resulting aggregates (i.e., not counting
+ // those that must be split).
+ void AggregateSmallPartitions(std::vector<int>* partition_vec,
+ int* min_size,
+ int* max_size);
+
+ // Insert packet into packet queue.
+ void QueuePacket(int start_pos,
+ int packet_size,
+ int first_partition_in_packet,
+ bool start_on_new_fragment);
+
+ // Write the payload header and copy the payload to the buffer.
+ // The info in packet_info determines which part of the payload is written
+ // and what to write in the header fields.
+ int WriteHeaderAndPayload(const InfoStruct& packet_info,
+ WebRtc_UWord8* buffer,
+ int buffer_length) const;
+
+
+ // Write the X field and the appropriate extension fields to buffer.
+ // The function returns the extension length (including X field), or -1
+ // on error.
+ int WriteExtensionFields(WebRtc_UWord8* buffer, int buffer_length) const;
+
+ // Set the I bit in the x_field, and write PictureID to the appropriate
+ // position in buffer. The function returns 0 on success, -1 otherwise.
+ int WritePictureIDFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
+ int buffer_length, int* extension_length) const;
+
+ // Set the L bit in the x_field, and write Tl0PicIdx to the appropriate
+ // position in buffer. The function returns 0 on success, -1 otherwise.
+ int WriteTl0PicIdxFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
+ int buffer_length, int* extension_length) const;
+
+ // Set the T and K bits in the x_field, and write TID, Y and KeyIdx to the
+ // appropriate position in buffer. The function returns 0 on success,
+ // -1 otherwise.
+ int WriteTIDAndKeyIdxFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
+ int buffer_length, int* extension_length) const;
+
+ // Write the PictureID from codec_specific_info_ to buffer. One or two
+ // bytes are written, depending on magnitude of PictureID. The function
+ // returns the number of bytes written.
+ int WritePictureID(WebRtc_UWord8* buffer, int buffer_length) const;
+
+ // Calculate and return length (octets) of the variable header fields in
+ // the next header (i.e., header length in addition to vp8_header_bytes_).
+ int PayloadDescriptorExtraLength() const;
+
+ // Calculate and return length (octets) of PictureID field in the next
+ // header. Can be 0, 1, or 2.
+ int PictureIdLength() const;
+
+ // Check whether each of the optional fields will be included in the header.
+ bool XFieldPresent() const;
+ bool TIDFieldPresent() const;
+ bool KeyIdxFieldPresent() const;
+ bool TL0PicIdxFieldPresent() const;
+ bool PictureIdPresent() const { return (PictureIdLength() > 0); }
+
+ const WebRtc_UWord8* payload_data_;
+ const int payload_size_;
+ RTPFragmentationHeader part_info_;
+ const int vp8_fixed_payload_descriptor_bytes_; // Length of VP8 payload
+ // descriptors's fixed part.
+ const AggregationMode aggr_mode_;
+ const bool balance_;
+ const bool separate_first_;
+ const RTPVideoHeaderVP8 hdr_info_;
+ const int num_partitions_;
+ const int max_payload_len_;
+ InfoQueue packets_;
+ bool packets_calculated_;
+
+ DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8);
+};
+
+} // namespace
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
new file mode 100644
index 0000000..59e76ee
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
@@ -0,0 +1,248 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+
+#include "gtest/gtest.h"
+
+namespace webrtc {
+
+namespace test {
+
+RtpFormatVp8TestHelper::RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr)
+ : payload_data_(NULL),
+ buffer_(NULL),
+ fragmentation_(NULL),
+ hdr_info_(hdr),
+ sloppy_partitioning_(false),
+ inited_(false) {}
+
+RtpFormatVp8TestHelper::~RtpFormatVp8TestHelper() {
+ delete fragmentation_;
+ delete [] payload_data_;
+ delete [] buffer_;
+}
+
+bool RtpFormatVp8TestHelper::Init(const int* partition_sizes,
+ int num_partitions) {
+ if (inited_) return false;
+ fragmentation_ = new RTPFragmentationHeader;
+ fragmentation_->VerifyAndAllocateFragmentationHeader(num_partitions);
+ payload_size_ = 0;
+ // Calculate sum payload size.
+ for (int p = 0; p < num_partitions; ++p) {
+ payload_size_ += partition_sizes[p];
+ }
+ buffer_size_ = payload_size_ + 6; // Add space for payload descriptor.
+ payload_data_ = new WebRtc_UWord8[payload_size_];
+ buffer_ = new WebRtc_UWord8[buffer_size_];
+ int j = 0;
+ // Loop through the partitions again.
+ for (int p = 0; p < num_partitions; ++p) {
+ fragmentation_->fragmentationLength[p] = partition_sizes[p];
+ fragmentation_->fragmentationOffset[p] = j;
+ for (int i = 0; i < partition_sizes[p]; ++i) {
+ assert(j < payload_size_);
+ payload_data_[j++] = p; // Set the payload value to the partition index.
+ }
+ }
+ data_ptr_ = payload_data_;
+ inited_ = true;
+ return true;
+}
+
+void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
+ RtpFormatVp8* packetizer,
+ const int* expected_sizes,
+ const int* expected_part,
+ const bool* expected_frag_start,
+ int expected_num_packets) {
+ ASSERT_TRUE(inited_);
+ int send_bytes = 0;
+ bool last;
+ for (int i = 0; i < expected_num_packets; ++i) {
+ std::ostringstream ss;
+ ss << "Checking packet " << i;
+ SCOPED_TRACE(ss.str());
+ EXPECT_EQ(expected_part[i],
+ packetizer->NextPacket(buffer_, &send_bytes, &last));
+ CheckPacket(send_bytes, expected_sizes[i], last,
+ expected_frag_start[i]);
+ }
+ EXPECT_TRUE(last);
+}
+
+// Payload descriptor
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |X|R|N|S|PartID | (REQUIRED)
+// +-+-+-+-+-+-+-+-+
+// X: |I|L|T|K| RSV | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// I: | PictureID | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// L: | TL0PICIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// T/K: | TID | KEYIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+
+// First octet tests.
+#define EXPECT_BIT_EQ(x, n, a) EXPECT_EQ((((x) >> (n)) & 0x1), a)
+
+#define EXPECT_RSV_ZERO(x) EXPECT_EQ(((x) & 0xE0), 0)
+
+#define EXPECT_BIT_X_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
+
+#define EXPECT_BIT_N_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
+
+#define EXPECT_BIT_S_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
+
+#define EXPECT_PART_ID_EQ(x, a) EXPECT_EQ(((x) & 0x0F), a)
+
+// Extension fields tests
+#define EXPECT_BIT_I_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
+
+#define EXPECT_BIT_L_EQ(x, a) EXPECT_BIT_EQ(x, 6, a)
+
+#define EXPECT_BIT_T_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
+
+#define EXPECT_BIT_K_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
+
+#define EXPECT_TID_EQ(x, a) EXPECT_EQ((((x) & 0xC0) >> 6), a)
+
+#define EXPECT_BIT_Y_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
+
+#define EXPECT_KEYIDX_EQ(x, a) EXPECT_EQ(((x) & 0x1F), a)
+
+void RtpFormatVp8TestHelper::CheckHeader(bool frag_start) {
+ payload_start_ = 1;
+ EXPECT_BIT_EQ(buffer_[0], 6, 0); // Check reserved bit.
+
+ if (hdr_info_->pictureId != kNoPictureId ||
+ hdr_info_->temporalIdx != kNoTemporalIdx ||
+ hdr_info_->tl0PicIdx != kNoTl0PicIdx ||
+ hdr_info_->keyIdx != kNoKeyIdx) {
+ EXPECT_BIT_X_EQ(buffer_[0], 1);
+ ++payload_start_;
+ CheckPictureID();
+ CheckTl0PicIdx();
+ CheckTIDAndKeyIdx();
+ } else {
+ EXPECT_BIT_X_EQ(buffer_[0], 0);
+ }
+
+ EXPECT_BIT_N_EQ(buffer_[0], hdr_info_->nonReference ? 1 : 0);
+ EXPECT_BIT_S_EQ(buffer_[0], frag_start ? 1 : 0);
+
+ // Check partition index.
+ if (!sloppy_partitioning_) {
+ // The test payload data is constructed such that the payload value is the
+ // same as the partition index.
+ EXPECT_EQ(buffer_[0] & 0x0F, buffer_[payload_start_]);
+ } else {
+ // Partition should be set to 0.
+ EXPECT_EQ(buffer_[0] & 0x0F, 0);
+ }
+}
+
+// Verify that the I bit and the PictureID field are both set in accordance
+// with the information in hdr_info_->pictureId.
+void RtpFormatVp8TestHelper::CheckPictureID() {
+ if (hdr_info_->pictureId != kNoPictureId) {
+ EXPECT_BIT_I_EQ(buffer_[1], 1);
+ if (hdr_info_->pictureId > 0x7F) {
+ EXPECT_BIT_EQ(buffer_[payload_start_], 7, 1);
+ EXPECT_EQ(buffer_[payload_start_] & 0x7F,
+ (hdr_info_->pictureId >> 8) & 0x7F);
+ EXPECT_EQ(buffer_[payload_start_ + 1],
+ hdr_info_->pictureId & 0xFF);
+ payload_start_ += 2;
+ } else {
+ EXPECT_BIT_EQ(buffer_[payload_start_], 7, 0);
+ EXPECT_EQ(buffer_[payload_start_] & 0x7F,
+ (hdr_info_->pictureId) & 0x7F);
+ payload_start_ += 1;
+ }
+ } else {
+ EXPECT_BIT_I_EQ(buffer_[1], 0);
+ }
+}
+
+// Verify that the L bit and the TL0PICIDX field are both set in accordance
+// with the information in hdr_info_->tl0PicIdx.
+void RtpFormatVp8TestHelper::CheckTl0PicIdx() {
+ if (hdr_info_->tl0PicIdx != kNoTl0PicIdx) {
+ EXPECT_BIT_L_EQ(buffer_[1], 1);
+ EXPECT_EQ(buffer_[payload_start_], hdr_info_->tl0PicIdx);
+ ++payload_start_;
+ } else {
+ EXPECT_BIT_L_EQ(buffer_[1], 0);
+ }
+}
+
+// Verify that the T bit and the TL0PICIDX field, and the K bit and KEYIDX
+// field are all set in accordance with the information in
+// hdr_info_->temporalIdx and hdr_info_->keyIdx, respectively.
+void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx() {
+ if (hdr_info_->temporalIdx == kNoTemporalIdx &&
+ hdr_info_->keyIdx == kNoKeyIdx) {
+ EXPECT_BIT_T_EQ(buffer_[1], 0);
+ EXPECT_BIT_K_EQ(buffer_[1], 0);
+ return;
+ }
+ if (hdr_info_->temporalIdx != kNoTemporalIdx) {
+ EXPECT_BIT_T_EQ(buffer_[1], 1);
+ EXPECT_TID_EQ(buffer_[payload_start_], hdr_info_->temporalIdx);
+ EXPECT_BIT_Y_EQ(buffer_[payload_start_], hdr_info_->layerSync ? 1 : 0);
+ } else {
+ EXPECT_BIT_T_EQ(buffer_[1], 0);
+ EXPECT_TID_EQ(buffer_[payload_start_], 0);
+ EXPECT_BIT_Y_EQ(buffer_[payload_start_], 0);
+ }
+ if (hdr_info_->keyIdx != kNoKeyIdx) {
+ EXPECT_BIT_K_EQ(buffer_[1], 1);
+ EXPECT_KEYIDX_EQ(buffer_[payload_start_], hdr_info_->keyIdx);
+ } else {
+ EXPECT_BIT_K_EQ(buffer_[1], 0);
+ EXPECT_KEYIDX_EQ(buffer_[payload_start_], 0);
+ }
+ ++payload_start_;
+}
+
+// Verify that the payload (i.e., after the headers) of the packet stored in
+// buffer_ is identical to the expected (as found in data_ptr_).
+void RtpFormatVp8TestHelper::CheckPayload(int payload_end) {
+ for (int i = payload_start_; i < payload_end; ++i, ++data_ptr_)
+ EXPECT_EQ(buffer_[i], *data_ptr_);
+}
+
+// Verify that the input variable "last" agrees with the position of data_ptr_.
+// If data_ptr_ has advanced payload_size_ bytes from the start (payload_data_)
+// we are at the end and last should be true. Otherwise, it should be false.
+void RtpFormatVp8TestHelper::CheckLast(bool last) const {
+ EXPECT_EQ(last, data_ptr_ == payload_data_ + payload_size_);
+}
+
+// Verify the contents of a packet. Check the length versus expected_bytes,
+// the header, payload, and "last" flag.
+void RtpFormatVp8TestHelper::CheckPacket(int send_bytes,
+ int expect_bytes,
+ bool last,
+ bool frag_start) {
+ EXPECT_EQ(expect_bytes, send_bytes);
+ CheckHeader(frag_start);
+ CheckPayload(send_bytes);
+ CheckLast(last);
+}
+
+} // namespace test
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
new file mode 100644
index 0000000..f9bf686
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains the class RtpFormatVp8TestHelper. The class is
+// responsible for setting up a fake VP8 bitstream according to the
+// RTPVideoHeaderVP8 header, and partition information. After initialization,
+// an RTPFragmentationHeader is provided so that the tester can create a
+// packetizer. The packetizer can then be provided to this helper class, which
+// will then extract all packets and compare to the expected outcome.
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
+
+#include "modules/interface/module_common_types.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "system_wrappers/interface/constructor_magic.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+namespace test {
+
+class RtpFormatVp8TestHelper {
+ public:
+ explicit RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr);
+ ~RtpFormatVp8TestHelper();
+ bool Init(const int* partition_sizes, int num_partitions);
+ void GetAllPacketsAndCheck(RtpFormatVp8* packetizer,
+ const int* expected_sizes,
+ const int* expected_part,
+ const bool* expected_frag_start,
+ int expected_num_packets);
+
+ uint8_t* payload_data() const { return payload_data_; }
+ int payload_size() const { return payload_size_; }
+ RTPFragmentationHeader* fragmentation() const { return fragmentation_; }
+ int buffer_size() const { return buffer_size_; }
+ void set_sloppy_partitioning(bool value) { sloppy_partitioning_ = value; }
+
+ private:
+ void CheckHeader(bool frag_start);
+ void CheckPictureID();
+ void CheckTl0PicIdx();
+ void CheckTIDAndKeyIdx();
+ void CheckPayload(int payload_end);
+ void CheckLast(bool last) const;
+ void CheckPacket(int send_bytes, int expect_bytes, bool last,
+ bool frag_start);
+
+ uint8_t* payload_data_;
+ uint8_t* buffer_;
+ uint8_t* data_ptr_;
+ RTPFragmentationHeader* fragmentation_;
+ const RTPVideoHeaderVP8* hdr_info_;
+ int payload_start_;
+ int payload_size_;
+ int buffer_size_;
+ bool sloppy_partitioning_;
+ bool inited_;
+
+ DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper);
+};
+
+} // namespace test
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
new file mode 100644
index 0000000..a6f9632
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -0,0 +1,364 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes unit tests for the VP8 packetizer.
+ */
+
+#include <gtest/gtest.h>
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+template <bool>
+struct CompileAssert {
+};
+
+#undef COMPILE_ASSERT
+#define COMPILE_ASSERT(expr, msg) \
+ typedef CompileAssert<(bool(expr))> msg[bool(expr) ? 1 : -1]
+
+class RtpFormatVp8Test : public ::testing::Test {
+ protected:
+ RtpFormatVp8Test() : helper_(NULL) {}
+ virtual void TearDown() { delete helper_; }
+ bool Init(const int* partition_sizes, int num_partitions) {
+ hdr_info_.pictureId = kNoPictureId;
+ hdr_info_.nonReference = false;
+ hdr_info_.temporalIdx = kNoTemporalIdx;
+ hdr_info_.layerSync = false;
+ hdr_info_.tl0PicIdx = kNoTl0PicIdx;
+ hdr_info_.keyIdx = kNoKeyIdx;
+ if (helper_ != NULL) return false;
+ helper_ = new test::RtpFormatVp8TestHelper(&hdr_info_);
+ return helper_->Init(partition_sizes, num_partitions);
+ }
+
+ RTPVideoHeaderVP8 hdr_info_;
+ test::RtpFormatVp8TestHelper* helper_;
+};
+
+TEST_F(RtpFormatVp8Test, TestStrictMode) {
+ const int kSizeVector[] = {10, 8, 27};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 200; // > 0x7F should produce 2-byte PictureID.
+ const int kMaxSize = 13;
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kStrict);
+
+ // The expected sizes are obtained by running a verified good implementation.
+ const int kExpectedSizes[] = {9, 9, 12, 11, 11, 11, 10};
+ const int kExpectedPart[] = {0, 0, 1, 2, 2, 2, 2};
+ const bool kExpectedFragStart[] =
+ {true, false, true, true, false, false, false};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+TEST_F(RtpFormatVp8Test, TestAggregateMode) {
+ const int kSizeVector[] = {60, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
+ const int kMaxSize = 25;
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // The expected sizes are obtained by running a verified good implementation.
+ const int kExpectedSizes[] = {23, 23, 23, 23};
+ const int kExpectedPart[] = {0, 0, 0, 1};
+ const bool kExpectedFragStart[] = {true, false, false, true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions1) {
+ const int kSizeVector[] = {1600, 200, 200, 200, 200, 200, 200, 200, 200};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
+ const int kMaxSize = 1500;
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // The expected sizes are obtained by running a verified good implementation.
+ const int kExpectedSizes[] = {803, 803, 803, 803};
+ const int kExpectedPart[] = {0, 0, 1, 5};
+ const bool kExpectedFragStart[] = {true, false, true, true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions2) {
+ const int kSizeVector[] = {1599, 200, 200, 200, 1600, 200, 200, 200, 200};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
+ const int kMaxSize = 1500;
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // The expected sizes are obtained by running a verified good implementation.
+ const int kExpectedSizes[] = {803, 802, 603, 803, 803, 803};
+ const int kExpectedPart[] = {0, 0, 1, 4, 4, 5};
+ const bool kExpectedFragStart[] = {true, false, true, true, false, true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+TEST_F(RtpFormatVp8Test, TestAggregateModeTwoLargePartitions) {
+ const int kSizeVector[] = {1654, 2268};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
+ const int kMaxSize = 1460;
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // The expected sizes are obtained by running a verified good implementation.
+ const int kExpectedSizes[] = {830, 830, 1137, 1137};
+ const int kExpectedPart[] = {0, 0, 1, 1};
+ const bool kExpectedFragStart[] = {true, false, true, false};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+// Verify that EqualSize mode is forced if fragmentation info is missing.
+TEST_F(RtpFormatVp8Test, TestEqualSizeModeFallback) {
+ const int kSizeVector[] = {10, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.pictureId = 200; // > 0x7F should produce 2-byte PictureID
+ const int kMaxSize = 12; // Small enough to produce 4 packets.
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize);
+
+ // Expecting three full packets, and one with the remainder.
+ const int kExpectedSizes[] = {12, 11, 12, 11};
+ const int kExpectedPart[] = {0, 0, 0, 0}; // Always 0 for equal size mode.
+ // Frag start only true for first packet in equal size mode.
+ const bool kExpectedFragStart[] = {true, false, false, false};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->set_sloppy_partitioning(true);
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+// Verify that non-reference bit is set. EqualSize mode fallback is expected.
+TEST_F(RtpFormatVp8Test, TestNonReferenceBit) {
+ const int kSizeVector[] = {10, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.nonReference = true;
+ const int kMaxSize = 25; // Small enough to produce two packets.
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize);
+
+ // EqualSize mode => First packet full; other not.
+ const int kExpectedSizes[] = {16, 16};
+ const int kExpectedPart[] = {0, 0}; // Always 0 for equal size mode.
+ // Frag start only true for first packet in equal size mode.
+ const bool kExpectedFragStart[] = {true, false};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->set_sloppy_partitioning(true);
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+// Verify Tl0PicIdx and TID fields, and layerSync bit.
+TEST_F(RtpFormatVp8Test, TestTl0PicIdxAndTID) {
+ const int kSizeVector[] = {10, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.tl0PicIdx = 117;
+ hdr_info_.temporalIdx = 2;
+ hdr_info_.layerSync = true;
+ // kMaxSize is only limited by allocated buffer size.
+ const int kMaxSize = helper_->buffer_size();
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // Expect one single packet of payload_size() + 4 bytes header.
+ const int kExpectedSizes[1] = {helper_->payload_size() + 4};
+ const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
+ const bool kExpectedFragStart[1] = {true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+// Verify KeyIdx field.
+TEST_F(RtpFormatVp8Test, TestKeyIdx) {
+ const int kSizeVector[] = {10, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.keyIdx = 17;
+ // kMaxSize is only limited by allocated buffer size.
+ const int kMaxSize = helper_->buffer_size();
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // Expect one single packet of payload_size() + 3 bytes header.
+ const int kExpectedSizes[1] = {helper_->payload_size() + 3};
+ const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
+ const bool kExpectedFragStart[1] = {true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+// Verify TID field and KeyIdx field in combination.
+TEST_F(RtpFormatVp8Test, TestTIDAndKeyIdx) {
+ const int kSizeVector[] = {10, 10, 10};
+ const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
+ ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
+
+ hdr_info_.temporalIdx = 1;
+ hdr_info_.keyIdx = 5;
+ // kMaxSize is only limited by allocated buffer size.
+ const int kMaxSize = helper_->buffer_size();
+ RtpFormatVp8 packetizer(helper_->payload_data(),
+ helper_->payload_size(),
+ hdr_info_,
+ kMaxSize,
+ *(helper_->fragmentation()),
+ kAggregate);
+
+ // Expect one single packet of payload_size() + 3 bytes header.
+ const int kExpectedSizes[1] = {helper_->payload_size() + 3};
+ const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
+ const bool kExpectedFragStart[1] = {true};
+ const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
+ kExpectedPart_wrong_size);
+ COMPILE_ASSERT(kExpectedNum ==
+ sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
+ kExpectedFragStart_wrong_size);
+
+ helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
+ kExpectedFragStart, kExpectedNum);
+}
+
+} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc
new file mode 100644
index 0000000..37a87d7
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc
@@ -0,0 +1,176 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "common_types.h"
+#include "rtp_header_extension.h"
+
+namespace webrtc {
+
+RtpHeaderExtensionMap::RtpHeaderExtensionMap() {
+}
+
+RtpHeaderExtensionMap::~RtpHeaderExtensionMap() {
+ Erase();
+}
+
+void RtpHeaderExtensionMap::Erase() {
+ while (!extensionMap_.empty()) {
+ std::map<uint8_t, HeaderExtension*>::iterator it =
+ extensionMap_.begin();
+ delete it->second;
+ extensionMap_.erase(it);
+ }
+}
+
+int32_t RtpHeaderExtensionMap::Register(const RTPExtensionType type,
+ const uint8_t id) {
+ if (id < 1 || id > 14) {
+ return -1;
+ }
+ std::map<uint8_t, HeaderExtension*>::iterator it =
+ extensionMap_.find(id);
+ if (it != extensionMap_.end()) {
+ return -1;
+ }
+ extensionMap_[id] = new HeaderExtension(type);
+ return 0;
+}
+
+int32_t RtpHeaderExtensionMap::Deregister(const RTPExtensionType type) {
+ uint8_t id;
+ if (GetId(type, &id) != 0) {
+ return -1;
+ }
+ std::map<uint8_t, HeaderExtension*>::iterator it =
+ extensionMap_.find(id);
+ if (it == extensionMap_.end()) {
+ return -1;
+ }
+ delete it->second;
+ extensionMap_.erase(it);
+ return 0;
+}
+
+int32_t RtpHeaderExtensionMap::GetType(const uint8_t id,
+ RTPExtensionType* type) const {
+ assert(type);
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.find(id);
+ if (it == extensionMap_.end()) {
+ return -1;
+ }
+ HeaderExtension* extension = it->second;
+ *type = extension->type;
+ return 0;
+}
+
+int32_t RtpHeaderExtensionMap::GetId(const RTPExtensionType type,
+ uint8_t* id) const {
+ assert(id);
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.begin();
+
+ while (it != extensionMap_.end()) {
+ HeaderExtension* extension = it->second;
+ if (extension->type == type) {
+ *id = it->first;
+ return 0;
+ }
+ it++;
+ }
+ return -1;
+}
+
+uint16_t RtpHeaderExtensionMap::GetTotalLengthInBytes() const {
+ // Get length for each extension block.
+ uint16_t length = 0;
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.begin();
+ while (it != extensionMap_.end()) {
+ HeaderExtension* extension = it->second;
+ length += extension->length;
+ it++;
+ }
+ // Add RTP extension header length.
+ if (length > 0) {
+ length += RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
+ }
+ return length;
+}
+
+int32_t RtpHeaderExtensionMap::GetLengthUntilBlockStartInBytes(
+ const RTPExtensionType type) const {
+ uint8_t id;
+ if (GetId(type, &id) != 0) {
+ // Not registered.
+ return -1;
+ }
+ // Get length until start of extension block type.
+ uint16_t length = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
+
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.begin();
+ while (it != extensionMap_.end()) {
+ HeaderExtension* extension = it->second;
+ if (extension->type == type) {
+ break;
+ } else {
+ length += extension->length;
+ }
+ it++;
+ }
+ return length;
+}
+
+int32_t RtpHeaderExtensionMap::Size() const {
+ return extensionMap_.size();
+}
+
+RTPExtensionType RtpHeaderExtensionMap::First() const {
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.begin();
+ if (it == extensionMap_.end()) {
+ return kRtpExtensionNone;
+ }
+ HeaderExtension* extension = it->second;
+ return extension->type;
+}
+
+RTPExtensionType RtpHeaderExtensionMap::Next(RTPExtensionType type) const {
+ uint8_t id;
+ if (GetId(type, &id) != 0) {
+ return kRtpExtensionNone;
+ }
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.find(id);
+ if (it == extensionMap_.end()) {
+ return kRtpExtensionNone;
+ }
+ it++;
+ if (it == extensionMap_.end()) {
+ return kRtpExtensionNone;
+ }
+ HeaderExtension* extension = it->second;
+ return extension->type;
+}
+
+void RtpHeaderExtensionMap::GetCopy(RtpHeaderExtensionMap* map) const {
+ assert(map);
+ std::map<uint8_t, HeaderExtension*>::const_iterator it =
+ extensionMap_.begin();
+ while (it != extensionMap_.end()) {
+ HeaderExtension* extension = it->second;
+ map->Register(extension->type, it->first);
+ it++;
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h
new file mode 100644
index 0000000..b1612cc
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
+#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
+
+#include <map>
+
+#include "rtp_rtcp_defines.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+enum {RTP_ONE_BYTE_HEADER_EXTENSION = 0xbede};
+
+enum ExtensionLength {
+ RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES = 4,
+ TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES = 4
+};
+
+struct HeaderExtension {
+ HeaderExtension(RTPExtensionType extension_type)
+ : type(extension_type),
+ length(0) {
+ if (type == kRtpExtensionTransmissionTimeOffset) {
+ length = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
+ }
+ }
+
+ const RTPExtensionType type;
+ uint8_t length;
+};
+
+class RtpHeaderExtensionMap {
+ public:
+ RtpHeaderExtensionMap();
+ ~RtpHeaderExtensionMap();
+
+ void Erase();
+
+ int32_t Register(const RTPExtensionType type, const uint8_t id);
+
+ int32_t Deregister(const RTPExtensionType type);
+
+ int32_t GetType(const uint8_t id, RTPExtensionType* type) const;
+
+ int32_t GetId(const RTPExtensionType type, uint8_t* id) const;
+
+ uint16_t GetTotalLengthInBytes() const;
+
+ int32_t GetLengthUntilBlockStartInBytes(const RTPExtensionType type) const;
+
+ void GetCopy(RtpHeaderExtensionMap* map) const;
+
+ int32_t Size() const;
+
+ RTPExtensionType First() const;
+
+ RTPExtensionType Next(RTPExtensionType type) const;
+
+ private:
+ std::map<uint8_t, HeaderExtension*> extensionMap_;
+};
+}
+#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc
new file mode 100644
index 0000000..003e92c
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes unit tests for the RtpHeaderExtensionMap.
+ */
+
+#include <gtest/gtest.h>
+
+#include "rtp_header_extension.h"
+#include "rtp_rtcp_defines.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class RtpHeaderExtensionTest : public ::testing::Test {
+ protected:
+ RtpHeaderExtensionTest() {}
+ ~RtpHeaderExtensionTest() {}
+
+ RtpHeaderExtensionMap map_;
+ enum {kId = 3};
+};
+
+TEST_F(RtpHeaderExtensionTest, Register) {
+ EXPECT_EQ(0, map_.Size());
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(1, map_.Size());
+ EXPECT_EQ(0, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
+ EXPECT_EQ(0, map_.Size());
+}
+
+TEST_F(RtpHeaderExtensionTest, RegisterIllegalArg) {
+ // Valid range for id: [1-14].
+ EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 0));
+ EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 15));
+}
+
+TEST_F(RtpHeaderExtensionTest, DeregisterIllegalArg) {
+ // Not registered.
+ EXPECT_EQ(-1, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
+}
+
+TEST_F(RtpHeaderExtensionTest, NonUniqueId) {
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+}
+
+TEST_F(RtpHeaderExtensionTest, GetTotalLength) {
+ EXPECT_EQ(0, map_.GetTotalLengthInBytes());
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES +
+ TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES,
+ map_.GetTotalLengthInBytes());
+}
+
+TEST_F(RtpHeaderExtensionTest, GetLengthUntilBlockStart) {
+ EXPECT_EQ(-1, map_.GetLengthUntilBlockStartInBytes(
+ kRtpExtensionTransmissionTimeOffset));
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES,
+ map_.GetLengthUntilBlockStartInBytes(
+ kRtpExtensionTransmissionTimeOffset));
+}
+
+TEST_F(RtpHeaderExtensionTest, GetType) {
+ RTPExtensionType typeOut;
+ EXPECT_EQ(-1, map_.GetType(kId, &typeOut));
+
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(0, map_.GetType(kId, &typeOut));
+ EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, typeOut);
+}
+
+TEST_F(RtpHeaderExtensionTest, GetId) {
+ uint8_t idOut;
+ EXPECT_EQ(-1, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
+
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(0, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
+ EXPECT_EQ(kId, idOut);
+}
+
+TEST_F(RtpHeaderExtensionTest, IterateTypes) {
+ EXPECT_EQ(kRtpExtensionNone, map_.First());
+ EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
+
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+
+ EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, map_.First());
+ EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
+}
+
+TEST_F(RtpHeaderExtensionTest, GetCopy) {
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+
+ RtpHeaderExtensionMap mapOut;
+ map_.GetCopy(&mapOut);
+ EXPECT_EQ(1, mapOut.Size());
+ EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, mapOut.First());
+}
+
+TEST_F(RtpHeaderExtensionTest, Erase) {
+ EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
+ EXPECT_EQ(1, map_.Size());
+ map_.Erase();
+ EXPECT_EQ(0, map_.Size());
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc
new file mode 100644
index 0000000..0de7553
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -0,0 +1,278 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Class for storing RTP packets.
+ */
+
+#include "rtp_packet_history.h"
+
+#include <assert.h>
+#include <cstring> // memset
+
+#include "critical_section_wrapper.h"
+#include "rtp_utility.h"
+#include "trace.h"
+
+namespace webrtc {
+
+RTPPacketHistory::RTPPacketHistory(RtpRtcpClock* clock)
+ : clock_(*clock),
+ critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ store_(false),
+ prev_index_(0),
+ max_packet_length_(0) {
+}
+
+RTPPacketHistory::~RTPPacketHistory() {
+ Free();
+ delete critsect_;
+}
+
+void RTPPacketHistory::SetStorePacketsStatus(bool enable,
+ uint16_t number_to_store) {
+ if (enable) {
+ Allocate(number_to_store);
+ } else {
+ Free();
+ }
+}
+
+void RTPPacketHistory::Allocate(uint16_t number_to_store) {
+ assert(number_to_store > 0);
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (store_) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "SetStorePacketsStatus already set, number: %d", number_to_store);
+ return;
+ }
+
+ store_ = true;
+ stored_packets_.resize(number_to_store);
+ stored_seq_nums_.resize(number_to_store);
+ stored_lengths_.resize(number_to_store);
+ stored_times_.resize(number_to_store);
+ stored_resend_times_.resize(number_to_store);
+ stored_types_.resize(number_to_store);
+}
+
+void RTPPacketHistory::Free() {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (!store_) {
+ return;
+ }
+
+ std::vector<std::vector<uint8_t> >::iterator it;
+ for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) {
+ it->clear();
+ }
+
+ stored_packets_.clear();
+ stored_seq_nums_.clear();
+ stored_lengths_.clear();
+ stored_times_.clear();
+ stored_resend_times_.clear();
+ stored_types_.clear();
+
+ store_ = false;
+ prev_index_ = 0;
+ max_packet_length_ = 0;
+}
+
+bool RTPPacketHistory::StorePackets() const {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ return store_;
+}
+
+// private, lock should already be taken
+void RTPPacketHistory::VerifyAndAllocatePacketLength(uint16_t packet_length) {
+ assert(packet_length > 0);
+ if (!store_) {
+ return;
+ }
+
+ if (packet_length <= max_packet_length_) {
+ return;
+ }
+
+ std::vector<std::vector<uint8_t> >::iterator it;
+ for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) {
+ it->resize(packet_length);
+ }
+ max_packet_length_ = packet_length;
+}
+
+int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
+ uint16_t packet_length,
+ uint16_t max_packet_length,
+ StorageType type) {
+ if (type == kDontStore) {
+ return 0;
+ }
+
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (!store_) {
+ return 0;
+ }
+
+ assert(packet);
+ assert(packet_length > 3);
+
+ VerifyAndAllocatePacketLength(max_packet_length);
+
+ if (packet_length > max_packet_length_) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, -1,
+ "Failed to store RTP packet, length: %d", packet_length);
+ return -1;
+ }
+
+ const uint16_t seq_num = (packet[2] << 8) + packet[3];
+
+ // Store packet
+ std::vector<std::vector<uint8_t> >::iterator it =
+ stored_packets_.begin() + prev_index_;
+ std::copy(packet, packet + packet_length, it->begin());
+
+ stored_seq_nums_[prev_index_] = seq_num;
+ stored_lengths_[prev_index_] = packet_length;
+ stored_times_[prev_index_] = clock_.GetTimeInMS();
+ stored_resend_times_[prev_index_] = 0; // packet not resent
+ stored_types_[prev_index_] = type;
+
+ ++prev_index_;
+ if (prev_index_ >= stored_seq_nums_.size()) {
+ prev_index_ = 0;
+ }
+ return 0;
+}
+
+bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (!store_) {
+ return false;
+ }
+
+ int32_t index = 0;
+ bool found = FindSeqNum(sequence_number, &index);
+ if (!found) {
+ return false;
+ }
+
+ uint16_t length = stored_lengths_.at(index);
+ if (length == 0 || length > max_packet_length_) {
+ // Invalid length.
+ return false;
+ }
+ return true;
+}
+
+bool RTPPacketHistory::GetRTPPacket(uint16_t sequence_number,
+ uint32_t min_elapsed_time_ms,
+ uint8_t* packet,
+ uint16_t* packet_length,
+ uint32_t* stored_time_ms,
+ StorageType* type) const {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (!store_) {
+ return false;
+ }
+
+ int32_t index = 0;
+ bool found = FindSeqNum(sequence_number, &index);
+ if (!found) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "No match for getting seqNum %u", sequence_number);
+ return false;
+ }
+
+ uint16_t length = stored_lengths_.at(index);
+ if (length == 0 || length > max_packet_length_) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "No match for getting seqNum %u, len %d", sequence_number, length);
+ return false;
+ }
+
+ if (length > *packet_length) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "Input buffer too short for packet %u", sequence_number);
+ return false;
+ }
+
+ // Verify elapsed time since last retrieve.
+ uint32_t now = clock_.GetTimeInMS();
+ if (min_elapsed_time_ms > 0 &&
+ ((now - stored_resend_times_.at(index)) < min_elapsed_time_ms)) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "Skip getting packet %u, packet recently resent.", sequence_number);
+ *packet_length = 0;
+ return true;
+ }
+
+ // Get packet.
+ std::vector<std::vector<uint8_t> >::const_iterator it_found_packet =
+ stored_packets_.begin() + index;
+ std::copy(it_found_packet->begin(), it_found_packet->begin() + length, packet);
+ *packet_length = stored_lengths_.at(index);
+ *stored_time_ms = stored_times_.at(index);
+ *type = stored_types_.at(index);
+ return true;
+}
+
+void RTPPacketHistory::UpdateResendTime(uint16_t sequence_number) {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ if (!store_) {
+ return;
+ }
+
+ int32_t index = 0;
+ bool found = FindSeqNum(sequence_number, &index);
+ if (!found) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "Failed to update resend time, seq num: %u.", sequence_number);
+ return;
+ }
+ stored_resend_times_[index] = clock_.GetTimeInMS();
+}
+
+// private, lock should already be taken
+bool RTPPacketHistory::FindSeqNum(uint16_t sequence_number,
+ int32_t* index) const {
+ uint16_t temp_sequence_number = 0;
+ if (prev_index_ > 0) {
+ *index = prev_index_ - 1;
+ temp_sequence_number = stored_seq_nums_[*index];
+ } else {
+ *index = stored_seq_nums_.size() - 1;
+ temp_sequence_number = stored_seq_nums_[*index]; // wrap
+ }
+
+ int32_t idx = (prev_index_ - 1) - (temp_sequence_number - sequence_number);
+ if (idx >= 0 && idx < static_cast<int>(stored_seq_nums_.size())) {
+ *index = idx;
+ temp_sequence_number = stored_seq_nums_[*index];
+ }
+
+ if (temp_sequence_number != sequence_number) {
+ // We did not found a match, search all.
+ for (uint16_t m = 0; m < stored_seq_nums_.size(); m++) {
+ if (stored_seq_nums_[m] == sequence_number) {
+ *index = m;
+ temp_sequence_number = stored_seq_nums_[*index];
+ break;
+ }
+ }
+ }
+ if (temp_sequence_number == sequence_number) {
+ // We found a match.
+ return true;
+ }
+ return false;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h
new file mode 100644
index 0000000..c61fd0b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Class for storing RTP packets.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+
+#include <vector>
+
+#include "module_common_types.h"
+#include "rtp_rtcp_defines.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class RtpRtcpClock;
+class CriticalSectionWrapper;
+
+class RTPPacketHistory {
+ public:
+ RTPPacketHistory(RtpRtcpClock* clock);
+ ~RTPPacketHistory();
+
+ void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
+
+ bool StorePackets() const;
+
+ // Stores RTP packet.
+ int32_t PutRTPPacket(const uint8_t* packet,
+ uint16_t packet_length,
+ uint16_t max_packet_length,
+ StorageType type);
+
+ // Gets stored RTP packet corresponding to the input sequence number.
+ // The packet is copied to the buffer pointed to by ptr_rtp_packet.
+ // The rtp_packet_length should show the available buffer size.
+ // Returns true if packet is found.
+ // rtp_packet_length: returns the copied packet length on success.
+ // min_elapsed_time_ms: the minimum time that must have elapsed since the last
+ // time the packet was resent (parameter is ignored if set to zero).
+ // If the packet is found but the minimum time has not elaped, no bytes are
+ // copied.
+ // stored_time_ms: returns the time when the packet was stored.
+ // type: returns the storage type set in PutRTPPacket.
+ bool GetRTPPacket(uint16_t sequence_number,
+ uint32_t min_elapsed_time_ms,
+ uint8_t* packet,
+ uint16_t* packet_length,
+ uint32_t* stored_time_ms,
+ StorageType* type) const;
+
+ bool HasRTPPacket(uint16_t sequence_number) const;
+
+ void UpdateResendTime(uint16_t sequence_number);
+
+ private:
+ void Allocate(uint16_t number_to_store);
+ void Free();
+ void VerifyAndAllocatePacketLength(uint16_t packet_length);
+ bool FindSeqNum(uint16_t sequence_number, int32_t* index) const;
+
+ private:
+ RtpRtcpClock& clock_;
+ CriticalSectionWrapper* critsect_;
+ bool store_;
+ uint32_t prev_index_;
+ uint16_t max_packet_length_;
+
+ std::vector<std::vector<uint8_t> > stored_packets_;
+ std::vector<uint16_t> stored_seq_nums_;
+ std::vector<uint16_t> stored_lengths_;
+ std::vector<uint32_t> stored_times_;
+ std::vector<uint32_t> stored_resend_times_;
+ std::vector<StorageType> stored_types_;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc
new file mode 100644
index 0000000..c849bc2
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc
@@ -0,0 +1,214 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes unit tests for the RTPPacketHistory.
+ */
+
+#include <gtest/gtest.h>
+
+#include "rtp_packet_history.h"
+#include "rtp_rtcp_defines.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class FakeClock : public RtpRtcpClock {
+ public:
+ FakeClock() {
+ time_in_ms_ = 123456;
+ }
+ // Return a timestamp in milliseconds relative to some arbitrary
+ // source; the source is fixed for this clock.
+ virtual WebRtc_UWord32 GetTimeInMS() {
+ return time_in_ms_;
+ }
+ // Retrieve an NTP absolute timestamp.
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
+ secs = time_in_ms_ / 1000;
+ frac = (time_in_ms_ % 1000) * 4294967;
+ }
+ void IncrementTime(WebRtc_UWord32 time_increment_ms) {
+ time_in_ms_ += time_increment_ms;
+ }
+ private:
+ WebRtc_UWord32 time_in_ms_;
+};
+
+class RtpPacketHistoryTest : public ::testing::Test {
+ protected:
+ RtpPacketHistoryTest()
+ : hist_(new RTPPacketHistory(&fake_clock_)) {
+ }
+ ~RtpPacketHistoryTest() {
+ delete hist_;
+ }
+
+ FakeClock fake_clock_;
+ RTPPacketHistory* hist_;
+ enum {kPayload = 127};
+ enum {kSsrc = 12345678};
+ enum {kSeqNum = 88};
+ enum {kTimestamp = 127};
+ enum {kMaxPacketLength = 1500};
+ uint8_t packet_[kMaxPacketLength];
+ uint8_t packet_out_[kMaxPacketLength];
+
+ void CreateRtpPacket(uint16_t seq_num, uint32_t ssrc, uint8_t payload,
+ uint32_t timestamp, uint8_t* array, uint16_t* cur_pos) {
+ array[(*cur_pos)++] = 0x80;
+ array[(*cur_pos)++] = payload;
+ array[(*cur_pos)++] = seq_num >> 8;
+ array[(*cur_pos)++] = seq_num;
+ array[(*cur_pos)++] = timestamp >> 24;
+ array[(*cur_pos)++] = timestamp >> 16;
+ array[(*cur_pos)++] = timestamp >> 8;
+ array[(*cur_pos)++] = timestamp;
+ array[(*cur_pos)++] = ssrc >> 24;
+ array[(*cur_pos)++] = ssrc >> 16;
+ array[(*cur_pos)++] = ssrc >> 8;
+ array[(*cur_pos)++] = ssrc;
+ }
+};
+
+TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
+ EXPECT_FALSE(hist_->StorePackets());
+ hist_->SetStorePacketsStatus(true, 10);
+ EXPECT_TRUE(hist_->StorePackets());
+ hist_->SetStorePacketsStatus(false, 0);
+ EXPECT_FALSE(hist_->StorePackets());
+}
+
+TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
+ EXPECT_FALSE(hist_->StorePackets());
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kAllowRetransmission));
+ // Packet should not be stored.
+ len = kMaxPacketLength;
+ uint32_t time;
+ StorageType type;
+ EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
+}
+
+TEST_F(RtpPacketHistoryTest, DontStore) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength, kDontStore));
+
+ // Packet should not be stored.
+ len = kMaxPacketLength;
+ uint32_t time;
+ StorageType type;
+ EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
+}
+
+TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) {
+ hist_->SetStorePacketsStatus(true, 10);
+ EXPECT_EQ(-1, hist_->PutRTPPacket(packet_,
+ kMaxPacketLength + 1,
+ kMaxPacketLength,
+ kAllowRetransmission));
+}
+
+TEST_F(RtpPacketHistoryTest, GetRtpPacket_TooSmallBuffer) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kAllowRetransmission));
+ uint16_t len_out = len - 1;
+ uint32_t time;
+ StorageType type;
+ EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len_out, &time,
+ &type));
+}
+
+TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = kMaxPacketLength;
+ uint32_t time;
+ StorageType type;
+ EXPECT_FALSE(hist_->GetRTPPacket(0, 0, packet_, &len, &time, &type));
+}
+
+TEST_F(RtpPacketHistoryTest, PutRtpPacket) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+
+ EXPECT_FALSE(hist_->HasRTPPacket(kSeqNum));
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kAllowRetransmission));
+ EXPECT_TRUE(hist_->HasRTPPacket(kSeqNum));
+}
+
+TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kAllowRetransmission));
+
+ uint16_t len_out = kMaxPacketLength;
+ uint32_t time;
+ StorageType type;
+ EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
+ &type));
+ EXPECT_EQ(len, len_out);
+ EXPECT_EQ(kAllowRetransmission, type);
+ for (int i = 0; i < len; i++) {
+ EXPECT_EQ(packet_[i], packet_out_[i]);
+ }
+}
+
+TEST_F(RtpPacketHistoryTest, DontRetransmit) {
+ hist_->SetStorePacketsStatus(true, 10);
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kDontRetransmit));
+
+ uint16_t len_out = kMaxPacketLength;
+ uint32_t time;
+ StorageType type;
+ EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
+ &type));
+ EXPECT_EQ(len, len_out);
+ EXPECT_EQ(kDontRetransmit, type);
+}
+
+TEST_F(RtpPacketHistoryTest, MinResendTime) {
+ hist_->SetStorePacketsStatus(true, 10);
+ WebRtc_UWord32 store_time = fake_clock_.GetTimeInMS();
+ uint16_t len = 0;
+ CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+ EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
+ kAllowRetransmission));
+
+ hist_->UpdateResendTime(kSeqNum);
+ fake_clock_.IncrementTime(100);
+
+ // Time has elapsed.
+ len = kMaxPacketLength;
+ StorageType type;
+ uint32_t time;
+ EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 100, packet_, &len, &time, &type));
+ EXPECT_GT(len, 0);
+ EXPECT_EQ(store_time, time);
+
+ // Time has not elapsed. Packet should be found, but no bytes copied.
+ len = kMaxPacketLength;
+ EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 101, packet_, &len, &time, &type));
+ EXPECT_EQ(0, len);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc
new file mode 100644
index 0000000..f421966
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -0,0 +1,1685 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "rtp_receiver.h"
+
+#include "rtp_rtcp_defines.h"
+#include "rtp_rtcp_impl.h"
+#include "critical_section_wrapper.h"
+
+#include <cassert>
+#include <string.h> //memcpy
+#include <math.h> // floor
+#include <stdlib.h> // abs
+
+namespace webrtc {
+
+using ModuleRTPUtility::AudioPayload;
+using ModuleRTPUtility::GetCurrentRTP;
+using ModuleRTPUtility::Payload;
+using ModuleRTPUtility::RTPPayloadParser;
+using ModuleRTPUtility::StringCompare;
+using ModuleRTPUtility::VideoPayload;
+
+RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner) :
+ RTPReceiverAudio(id),
+ RTPReceiverVideo(id, owner),
+ Bitrate(clock),
+ _id(id),
+ _audio(audio),
+ _rtpRtcp(*owner),
+ _criticalSectionCbs(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbRtpFeedback(NULL),
+ _cbRtpData(NULL),
+
+ _criticalSectionRTPReceiver(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _lastReceiveTime(0),
+ _lastReceivedPayloadLength(0),
+ _lastReceivedPayloadType(-1),
+ _lastReceivedMediaPayloadType(-1),
+ _lastReceivedAudioSpecific(),
+ _lastReceivedVideoSpecific(),
+
+ _packetTimeOutMS(0),
+
+ _redPayloadType(-1),
+ _payloadTypeMap(),
+ _rtpHeaderExtensionMap(),
+ _SSRC(0),
+ _numCSRCs(0),
+ _currentRemoteCSRC(),
+ _numEnergy(0),
+ _currentRemoteEnergy(),
+ _useSSRCFilter(false),
+ _SSRCFilter(0),
+
+ _jitterQ4(0),
+ _jitterMaxQ4(0),
+ _cumulativeLoss(0),
+ _jitterQ4TransmissionTimeOffset(0),
+ _localTimeLastReceivedTimestamp(0),
+ _lastReceivedTimestamp(0),
+ _lastReceivedSequenceNumber(0),
+ _lastReceivedTransmissionTimeOffset(0),
+
+ _receivedSeqFirst(0),
+ _receivedSeqMax(0),
+ _receivedSeqWraps(0),
+
+ _receivedPacketOH(12), // RTP header
+ _receivedByteCount(0),
+ _receivedOldPacketCount(0),
+ _receivedInorderPacketCount(0),
+
+ _lastReportInorderPackets(0),
+ _lastReportOldPackets(0),
+ _lastReportSeqMax(0),
+ _lastReportFractionLost(0),
+ _lastReportCumulativeLost(0),
+ _lastReportExtendedHighSeqNum(0),
+ _lastReportJitter(0),
+ _lastReportJitterTransmissionTimeOffset(0),
+
+ _nackMethod(kNackOff),
+ _RTX(false),
+ _ssrcRTX(0) {
+ memset(_currentRemoteCSRC, 0, sizeof(_currentRemoteCSRC));
+ memset(_currentRemoteEnergy, 0, sizeof(_currentRemoteEnergy));
+ memset(&_lastReceivedAudioSpecific, 0, sizeof(_lastReceivedAudioSpecific));
+
+ _lastReceivedAudioSpecific.channels = 1;
+ _lastReceivedVideoSpecific.maxRate = 0;
+ _lastReceivedVideoSpecific.videoCodecType = kRtpNoVideo;
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+}
+
+RTPReceiver::~RTPReceiver() {
+ if (_cbRtpFeedback) {
+ for (int i = 0; i < _numCSRCs; i++) {
+ _cbRtpFeedback->OnIncomingCSRCChanged(_id,_currentRemoteCSRC[i], false);
+ }
+ }
+ delete _criticalSectionCbs;
+ delete _criticalSectionRTPReceiver;
+
+ while (!_payloadTypeMap.empty()) {
+ std::map<WebRtc_Word8, Payload*>::iterator it = _payloadTypeMap.begin();
+ delete it->second;
+ _payloadTypeMap.erase(it);
+ }
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_Word32 RTPReceiver::Init() {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ _lastReceiveTime = 0;
+ _lastReceivedPayloadLength = 0;
+ _packetTimeOutMS = 0;
+ _lastReceivedPayloadType = -1;
+ _lastReceivedMediaPayloadType = -1;
+ _redPayloadType = -1;
+
+ memset(&_lastReceivedAudioSpecific, 0, sizeof(_lastReceivedAudioSpecific));
+ _lastReceivedAudioSpecific.channels = 1;
+
+ _lastReceivedVideoSpecific.videoCodecType = kRtpNoVideo;
+ _lastReceivedVideoSpecific.maxRate = 0;
+ _SSRC = 0;
+ _numCSRCs = 0;
+ _numEnergy = 0;
+ _jitterQ4 = 0;
+ _jitterMaxQ4 = 0;
+ _cumulativeLoss = 0;
+ _jitterQ4TransmissionTimeOffset = 0;
+ _useSSRCFilter = false;
+ _SSRCFilter = 0;
+
+ _localTimeLastReceivedTimestamp = 0;
+ _lastReceivedTimestamp = 0;
+ _lastReceivedSequenceNumber = 0;
+ _lastReceivedTransmissionTimeOffset = 0;
+
+ _receivedSeqFirst = 0;
+ _receivedSeqMax = 0;
+ _receivedSeqWraps = 0;
+
+ _receivedPacketOH = 12; // RTP header
+ _receivedByteCount = 0;
+ _receivedOldPacketCount = 0;
+ _receivedInorderPacketCount = 0;
+
+ _lastReportInorderPackets = 0;
+ _lastReportOldPackets = 0;
+ _lastReportSeqMax = 0;
+ _lastReportFractionLost = 0;
+ _lastReportCumulativeLost = 0;
+ _lastReportExtendedHighSeqNum = 0;
+ _lastReportJitter = 0;
+ _lastReportJitterTransmissionTimeOffset = 0;
+
+ _rtpHeaderExtensionMap.Erase();
+
+ while (!_payloadTypeMap.empty()) {
+ std::map<WebRtc_Word8, Payload*>::iterator it = _payloadTypeMap.begin();
+ delete it->second;
+ _payloadTypeMap.erase(it);
+ }
+
+ Bitrate::Init();
+ RTPReceiverAudio::Init();
+ return RTPReceiverVideo::Init();
+}
+
+void
+RTPReceiver::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ RTPReceiverAudio::ChangeUniqueId(id);
+ RTPReceiverVideo::ChangeUniqueId(id);
+}
+
+RtpVideoCodecTypes
+RTPReceiver::VideoCodecType() const
+{
+ return _lastReceivedVideoSpecific.videoCodecType;
+}
+
+WebRtc_UWord32
+RTPReceiver::MaxConfiguredBitrate() const
+{
+ return _lastReceivedVideoSpecific.maxRate;
+}
+
+bool
+RTPReceiver::REDPayloadType(const WebRtc_Word8 payloadType) const
+{
+ return (_redPayloadType == payloadType)?true:false;
+}
+
+WebRtc_Word8
+RTPReceiver::REDPayloadType() const
+{
+ return _redPayloadType;
+}
+
+ // configure a timeout value
+WebRtc_Word32
+RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ _packetTimeOutMS = timeoutMS;
+ return 0;
+}
+
+void RTPReceiver::PacketTimeout()
+{
+ bool packetTimeOut = false;
+ {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ if(_packetTimeOutMS == 0)
+ {
+ // not configured
+ return;
+ }
+
+ if(_lastReceiveTime == 0)
+ {
+ // not active
+ return;
+ }
+
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+
+ if(now - _lastReceiveTime > _packetTimeOutMS)
+ {
+ packetTimeOut = true;
+ _lastReceiveTime = 0; // only one callback
+ _lastReceivedPayloadType = -1; // makes RemotePayload return -1, which we want
+ _lastReceivedMediaPayloadType = -1;
+ }
+ }
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if(packetTimeOut && _cbRtpFeedback)
+ {
+ _cbRtpFeedback->OnPacketTimeout(_id);
+ }
+}
+
+void
+RTPReceiver::ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_UWord32 now)
+{
+ if(_cbRtpFeedback == NULL)
+ {
+ // no callback
+ return;
+ }
+ RTPAliveType alive = kRtpDead;
+
+ if(_lastReceiveTime + 1000 > now)
+ {
+ // always alive if we have received a RTP packet the last sec
+ alive = kRtpAlive;
+
+ } else
+ {
+ if(RTCPalive)
+ {
+ if(_audio)
+ {
+ // alive depends on CNG
+ // if last received size < 10 likely CNG
+ if(_lastReceivedPayloadLength < 10) // our CNG is 9 bytes
+ {
+ // potential CNG
+ // receiver need to check kRtpNoRtp against NetEq speechType kOutputPLCtoCNG
+ alive = kRtpNoRtp;
+ } else
+ {
+ // dead
+ }
+ } else
+ {
+ // dead for video
+ }
+ }else
+ {
+ // no RTP packet for 1 sec and no RTCP
+ // dead
+ }
+ }
+
+
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if(_cbRtpFeedback)
+ {
+ _cbRtpFeedback->OnPeriodicDeadOrAlive(_id, alive);
+ }
+}
+
+WebRtc_UWord16
+RTPReceiver::PacketOHReceived() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _receivedPacketOH;
+}
+
+WebRtc_UWord32
+RTPReceiver::PacketCountReceived() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _receivedInorderPacketCount;
+}
+
+WebRtc_UWord32
+RTPReceiver::ByteCountReceived() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _receivedByteCount;
+}
+
+WebRtc_Word32
+RTPReceiver::RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ _cbRtpFeedback = incomingMessagesCallback;
+ return 0;
+}
+
+WebRtc_Word32
+RTPReceiver::RegisterIncomingDataCallback(RtpData* incomingDataCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ _cbRtpData = incomingDataCallback;
+ return 0;
+}
+
+WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ assert(payloadName);
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ // sanity
+ switch (payloadType) {
+ // reserved payload types to avoid RTCP conflicts when marker bit is set
+ case 64: // 192 Full INTRA-frame request
+ case 72: // 200 Sender report
+ case 73: // 201 Receiver report
+ case 74: // 202 Source description
+ case 75: // 203 Goodbye
+ case 76: // 204 Application-defined
+ case 77: // 205 Transport layer FB message
+ case 78: // 206 Payload-specific FB message
+ case 79: // 207 Extended report
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid payloadtype:%d",
+ __FUNCTION__, payloadType);
+ return -1;
+ default:
+ break;
+ }
+ size_t payloadNameLength = strlen(payloadName);
+
+ std::map<WebRtc_Word8, Payload*>::iterator it =
+ _payloadTypeMap.find(payloadType);
+ if (it != _payloadTypeMap.end()) {
+ // we already use this payload type
+ Payload* payload = it->second;
+ assert(payload);
+
+ size_t nameLength = strlen(payload->name);
+
+ // check if it's the same as we already have
+ // if same ignore sending an error
+ if (payloadNameLength == nameLength &&
+ StringCompare(payload->name, payloadName, payloadNameLength)) {
+ if (_audio &&
+ payload->audio &&
+ payload->typeSpecific.Audio.frequency == frequency &&
+ payload->typeSpecific.Audio.channels == channels &&
+ (payload->typeSpecific.Audio.rate == rate ||
+ payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
+ payload->typeSpecific.Audio.rate = rate;
+ // Ensure that we update the rate if new or old is zero
+ return 0;
+ }
+ if (!_audio && !payload->audio) {
+ // update maxBitrate for video
+ payload->typeSpecific.Video.maxRate = rate;
+ return 0;
+ }
+ }
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument payloadType:%d already registered",
+ __FUNCTION__, payloadType);
+ return -1;
+ }
+ if (_audio) {
+ // remove existing item, hence search for the name
+ // only for audio, for video we allow a codecs to use multiple pltypes
+ std::map<WebRtc_Word8, Payload*>::iterator audio_it =
+ _payloadTypeMap.begin();
+ while (audio_it != _payloadTypeMap.end()) {
+ Payload* payload = audio_it->second;
+ size_t nameLength = strlen(payload->name);
+
+ if (payloadNameLength == nameLength &&
+ StringCompare(payload->name, payloadName, payloadNameLength)) {
+ // we found the payload name in the list
+ // if audio check frequency and rate
+ if (payload->audio) {
+ if (payload->typeSpecific.Audio.frequency == frequency &&
+ (payload->typeSpecific.Audio.rate == rate ||
+ payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
+ // remove old setting
+ delete payload;
+ _payloadTypeMap.erase(audio_it);
+ break;
+ }
+ } else if(StringCompare(payloadName,"red",3)) {
+ delete payload;
+ _payloadTypeMap.erase(audio_it);
+ break;
+ }
+ }
+ audio_it++;
+ }
+ }
+ Payload* payload = NULL;
+
+ // save the RED payload type
+ // used in both audio and video
+ if (StringCompare(payloadName,"red",3)) {
+ _redPayloadType = payloadType;
+ payload = new Payload;
+ payload->audio = false;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ } else {
+ if (_audio) {
+ payload = RegisterReceiveAudioPayload(payloadName, payloadType,
+ frequency, channels, rate);
+ } else {
+ payload = RegisterReceiveVideoPayload(payloadName, payloadType, rate);
+ }
+ }
+ if (payload == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s filed to register payload",
+ __FUNCTION__);
+ return -1;
+ }
+ _payloadTypeMap[payloadType] = payload;
+
+ // Successful set of payload type, clear the value of last receivedPT,
+ // since it might mean something else
+ _lastReceivedPayloadType = -1;
+ _lastReceivedMediaPayloadType = -1;
+ return 0;
+}
+
+WebRtc_Word32 RTPReceiver::DeRegisterReceivePayload(
+ const WebRtc_Word8 payloadType) {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ std::map<WebRtc_Word8, Payload*>::iterator it =
+ _payloadTypeMap.find(payloadType);
+
+ if (it == _payloadTypeMap.end()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s failed to find payloadType:%d",
+ __FUNCTION__, payloadType);
+ return -1;
+ }
+ delete it->second;
+ _payloadTypeMap.erase(it);
+ return 0;
+}
+
+WebRtc_Word32 RTPReceiver::ReceivePayloadType(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ WebRtc_Word8* payloadType) const {
+ if (payloadType == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ size_t payloadNameLength = strlen(payloadName);
+
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ std::map<WebRtc_Word8, Payload*>::const_iterator it =
+ _payloadTypeMap.begin();
+
+ while (it != _payloadTypeMap.end()) {
+ Payload* payload = it->second;
+ assert(payload);
+
+ size_t nameLength = strlen(payload->name);
+ if (payloadNameLength == nameLength &&
+ StringCompare(payload->name, payloadName, payloadNameLength)) {
+ // name match
+ if( payload->audio) {
+ if (rate == 0) {
+ // [default] audio, check freq and channels
+ if (payload->typeSpecific.Audio.frequency == frequency &&
+ payload->typeSpecific.Audio.channels == channels) {
+ *payloadType = it->first;
+ return 0;
+ }
+ } else {
+ // audio, check freq, channels and rate
+ if( payload->typeSpecific.Audio.frequency == frequency &&
+ payload->typeSpecific.Audio.channels == channels &&
+ payload->typeSpecific.Audio.rate == rate) {
+ // extra rate condition added
+ *payloadType = it->first;
+ return 0;
+ }
+ }
+ } else {
+ // video
+ *payloadType = it->first;
+ return 0;
+ }
+ }
+ it++;
+ }
+ return -1;
+}
+
+WebRtc_Word32 RTPReceiver::ReceivePayload(
+ const WebRtc_Word8 payloadType,
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ WebRtc_UWord32* frequency,
+ WebRtc_UWord8* channels,
+ WebRtc_UWord32* rate) const {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ std::map<WebRtc_Word8, Payload*>::const_iterator it =
+ _payloadTypeMap.find(payloadType);
+
+ if (it == _payloadTypeMap.end()) {
+ return -1;
+ }
+ Payload* payload = it->second;
+ assert(payload);
+
+ if(frequency) {
+ if(payload->audio) {
+ *frequency = payload->typeSpecific.Audio.frequency;
+ } else {
+ *frequency = 90000;
+ }
+ }
+ if (channels) {
+ if(payload->audio) {
+ *channels = payload->typeSpecific.Audio.channels;
+ } else {
+ *channels = 1;
+ }
+ }
+ if (rate) {
+ if(payload->audio) {
+ *rate = payload->typeSpecific.Audio.rate;
+ } else {
+ assert(false);
+ *rate = 0;
+ }
+ }
+ if (payloadName) {
+ payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTPReceiver::RemotePayload(
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ WebRtc_Word8* payloadType,
+ WebRtc_UWord32* frequency,
+ WebRtc_UWord8* channels) const {
+ if(_lastReceivedPayloadType == -1) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "%s invalid state", __FUNCTION__);
+ return -1;
+ }
+ std::map<WebRtc_Word8, Payload*>::const_iterator it =
+ _payloadTypeMap.find(_lastReceivedPayloadType);
+
+ if (it == _payloadTypeMap.end()) {
+ return -1;
+ }
+ Payload* payload = it->second;
+ assert(payload);
+ payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+
+ if (payloadType) {
+ *payloadType = _lastReceivedPayloadType;
+ }
+ if (frequency) {
+ if (payload->audio) {
+ *frequency = payload->typeSpecific.Audio.frequency;
+ } else {
+ *frequency = 90000;
+ }
+ }
+ if (channels) {
+ if (payload->audio) {
+ *channels = payload->typeSpecific.Audio.channels;
+ } else {
+ *channels = 1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPReceiver::RegisterRtpHeaderExtension(const RTPExtensionType type,
+ const WebRtc_UWord8 id)
+{
+ CriticalSectionScoped cs(_criticalSectionRTPReceiver);
+ return _rtpHeaderExtensionMap.Register(type, id);
+}
+
+WebRtc_Word32
+RTPReceiver::DeregisterRtpHeaderExtension(const RTPExtensionType type)
+{
+ CriticalSectionScoped cs(_criticalSectionRTPReceiver);
+ return _rtpHeaderExtensionMap.Deregister(type);
+}
+
+void RTPReceiver::GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const
+{
+ CriticalSectionScoped cs(_criticalSectionRTPReceiver);
+ _rtpHeaderExtensionMap.GetCopy(map);
+}
+
+NACKMethod
+RTPReceiver::NACK() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _nackMethod;
+}
+
+ // Turn negative acknowledgement requests on/off
+WebRtc_Word32
+RTPReceiver::SetNACKStatus(const NACKMethod method)
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ _nackMethod = method;
+ return 0;
+}
+
+void RTPReceiver::SetRTXStatus(const bool enable,
+ const WebRtc_UWord32 SSRC) {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ _RTX = enable;
+ _ssrcRTX = SSRC;
+}
+
+void RTPReceiver::RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ *enable = _RTX;
+ *SSRC = _ssrcRTX;
+}
+
+WebRtc_UWord32
+RTPReceiver::SSRC() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _SSRC;
+}
+
+ // Get remote CSRC
+WebRtc_Word32
+RTPReceiver::CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ assert(_numCSRCs <= kRtpCsrcSize);
+
+ if(_numCSRCs >0)
+ {
+ memcpy(arrOfCSRC, _currentRemoteCSRC, sizeof(WebRtc_UWord32)*_numCSRCs);
+ }
+ return _numCSRCs;
+}
+
+WebRtc_Word32
+RTPReceiver::Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ assert(_numEnergy <= kRtpCsrcSize);
+
+ if(_numEnergy >0)
+ {
+ memcpy(arrOfEnergy, _currentRemoteEnergy, sizeof(WebRtc_UWord8)*_numCSRCs);
+ }
+ return _numEnergy;
+}
+
+WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
+ WebRtcRTPHeader* rtp_header,
+ const WebRtc_UWord8* packet,
+ const WebRtc_UWord16 packet_length) {
+ // rtp_header contains the parsed RTP header.
+ // Adjust packet length w r t RTP padding.
+ int length = packet_length - rtp_header->header.paddingLength;
+
+ // length sanity
+ if ((length - rtp_header->header.headerLength) < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument",
+ __FUNCTION__);
+ return -1;
+ }
+ if (_RTX) {
+ if (_ssrcRTX == rtp_header->header.ssrc) {
+ // Sanity check.
+ if (rtp_header->header.headerLength + 2 > packet_length) {
+ return -1;
+ }
+ rtp_header->header.ssrc = _SSRC;
+ rtp_header->header.sequenceNumber =
+ (packet[rtp_header->header.headerLength] << 8) +
+ packet[1 + rtp_header->header.headerLength];
+ // Count the RTX header as part of the RTP header.
+ rtp_header->header.headerLength += 2;
+ }
+ }
+ if (_useSSRCFilter) {
+ if (rtp_header->header.ssrc != _SSRCFilter) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "%s drop packet due to SSRC filter",
+ __FUNCTION__);
+ return -1;
+ }
+ }
+ if (_lastReceiveTime == 0) {
+ // trigger only once
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if (_cbRtpFeedback) {
+ if (length - rtp_header->header.headerLength == 0) {
+ // keepalive packet
+ _cbRtpFeedback->OnReceivedPacket(_id, kPacketKeepAlive);
+ } else {
+ _cbRtpFeedback->OnReceivedPacket(_id, kPacketRtp);
+ }
+ }
+ }
+ WebRtc_Word8 first_payload_byte = 0;
+ if (length > 0) {
+ first_payload_byte = packet[rtp_header->header.headerLength];
+ }
+ // trigger our callbacks
+ CheckSSRCChanged(rtp_header);
+
+ bool is_red = false;
+ VideoPayload video_specific;
+ video_specific.maxRate = 0;
+ video_specific.videoCodecType = kRtpNoVideo;
+
+ AudioPayload audio_specific;
+ audio_specific.bitsPerSample = 0;
+ audio_specific.channels = 0;
+ audio_specific.frequency = 0;
+
+ if (CheckPayloadChanged(rtp_header,
+ first_payload_byte,
+ is_red,
+ audio_specific,
+ video_specific) == -1) {
+ if (length - rtp_header->header.headerLength == 0)
+ {
+ // ok keepalive packet
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "%s received keepalive",
+ __FUNCTION__);
+ return 0;
+ }
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "%s received invalid payloadtype",
+ __FUNCTION__);
+ return -1;
+ }
+ CheckCSRC(rtp_header);
+
+ const WebRtc_UWord8* payload_data =
+ packet + rtp_header->header.headerLength;
+
+ WebRtc_UWord16 payload_data_length =
+ static_cast<WebRtc_UWord16>(length - rtp_header->header.headerLength);
+
+ WebRtc_Word32 retVal = 0;
+ if(_audio) {
+ retVal = ParseAudioCodecSpecific(rtp_header,
+ payload_data,
+ payload_data_length,
+ audio_specific,
+ is_red);
+ } else {
+ retVal = ParseVideoCodecSpecific(rtp_header,
+ payload_data,
+ payload_data_length,
+ video_specific.videoCodecType,
+ is_red,
+ packet,
+ packet_length,
+ _clock.GetTimeInMS());
+ }
+ if(retVal < 0) {
+ return retVal;
+ }
+
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ // this compare to _receivedSeqMax
+ // we store the last received after we have done the callback
+ bool old_packet = RetransmitOfOldPacket(rtp_header->header.sequenceNumber,
+ rtp_header->header.timestamp);
+
+ // this updates _receivedSeqMax and other members
+ UpdateStatistics(rtp_header, payload_data_length, old_packet);
+
+ // Need to be updated after RetransmitOfOldPacket &
+ // RetransmitOfOldPacketUpdateStatistics
+ _lastReceiveTime = _clock.GetTimeInMS();
+ _lastReceivedPayloadLength = payload_data_length;
+
+ if (!old_packet) {
+ if (_lastReceivedTimestamp != rtp_header->header.timestamp) {
+ _lastReceivedTimestamp = rtp_header->header.timestamp;
+ }
+ _lastReceivedSequenceNumber = rtp_header->header.sequenceNumber;
+ _lastReceivedTransmissionTimeOffset =
+ rtp_header->extension.transmissionTimeOffset;
+ }
+ return retVal;
+}
+
+// must not have critsect when called
+WebRtc_Word32
+RTPReceiver::CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader)
+{
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if(_cbRtpData)
+ {
+ return _cbRtpData->OnReceivedPayloadData(payloadData, payloadSize, rtpHeader);
+ }
+ return -1;
+}
+
+// we already have the _criticalSectionRTPReceiver critsect when we call this
+void
+RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord16 bytes,
+ const bool oldPacket)
+{
+ WebRtc_UWord32 freq = 90000;
+ if(_audio)
+ {
+ freq = AudioFrequency();
+ }
+
+ Bitrate::Update(bytes);
+
+ _receivedByteCount += bytes;
+
+ if (_receivedSeqMax == 0 && _receivedSeqWraps == 0)
+ {
+ // First received report
+ _receivedSeqFirst = rtpHeader->header.sequenceNumber;
+ _receivedSeqMax = rtpHeader->header.sequenceNumber;
+ _receivedInorderPacketCount = 1;
+ _localTimeLastReceivedTimestamp =
+ GetCurrentRTP(&_clock, freq); //time in samples
+ return;
+ }
+
+ // count only the new packets received
+ if(InOrderPacket(rtpHeader->header.sequenceNumber))
+ {
+ const WebRtc_UWord32 RTPtime =
+ GetCurrentRTP(&_clock, freq); //time in samples
+ _receivedInorderPacketCount++;
+
+ // wrong if we use RetransmitOfOldPacket
+ WebRtc_Word32 seqDiff = rtpHeader->header.sequenceNumber - _receivedSeqMax;
+ if (seqDiff < 0)
+ {
+ // Wrap around detected
+ _receivedSeqWraps++;
+ }
+ // new max
+ _receivedSeqMax = rtpHeader->header.sequenceNumber;
+
+ if (rtpHeader->header.timestamp != _lastReceivedTimestamp &&
+ _receivedInorderPacketCount > 1)
+ {
+ WebRtc_Word32 timeDiffSamples = (RTPtime - _localTimeLastReceivedTimestamp) -
+ (rtpHeader->header.timestamp - _lastReceivedTimestamp);
+
+ timeDiffSamples = abs(timeDiffSamples);
+
+ // libJingle sometimes deliver crazy jumps in TS for the same stream
+ // If this happen don't update jitter value
+ if(timeDiffSamples < 450000) // Use 5 secs video frequency as border
+ {
+ // note we calculate in Q4 to avoid using float
+ WebRtc_Word32 jitterDiffQ4 = (timeDiffSamples << 4) - _jitterQ4;
+ _jitterQ4 += ((jitterDiffQ4 + 8) >> 4);
+ }
+
+ // Extended jitter report, RFC 5450.
+ // Actual network jitter, excluding the source-introduced jitter.
+ WebRtc_Word32 timeDiffSamplesExt =
+ (RTPtime - _localTimeLastReceivedTimestamp) -
+ ((rtpHeader->header.timestamp +
+ rtpHeader->extension.transmissionTimeOffset) -
+ (_lastReceivedTimestamp +
+ _lastReceivedTransmissionTimeOffset));
+
+ timeDiffSamplesExt = abs(timeDiffSamplesExt);
+
+ if(timeDiffSamplesExt < 450000) // Use 5 secs video freq as border
+ {
+ // note we calculate in Q4 to avoid using float
+ WebRtc_Word32 jitterDiffQ4TransmissionTimeOffset =
+ (timeDiffSamplesExt << 4) - _jitterQ4TransmissionTimeOffset;
+ _jitterQ4TransmissionTimeOffset +=
+ ((jitterDiffQ4TransmissionTimeOffset + 8) >> 4);
+ }
+ }
+ _localTimeLastReceivedTimestamp = RTPtime;
+ } else
+ {
+ if(oldPacket)
+ {
+ _receivedOldPacketCount++;
+ }else
+ {
+ _receivedInorderPacketCount++;
+ }
+ }
+
+ WebRtc_UWord16 packetOH = rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
+
+ // our measured overhead
+ // filter from RFC 5104 4.2.1.2
+ // avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
+ _receivedPacketOH = (15*_receivedPacketOH + packetOH) >> 4;
+}
+
+// we already have the _criticalSectionRTPReceiver critsect when we call this
+bool RTPReceiver::RetransmitOfOldPacket(
+ const WebRtc_UWord16 sequenceNumber,
+ const WebRtc_UWord32 rtpTimeStamp) const {
+ if (InOrderPacket(sequenceNumber)) {
+ return false;
+ }
+ WebRtc_UWord32 frequencyKHz = 90; // Video frequency.
+ if (_audio) {
+ frequencyKHz = AudioFrequency() / 1000;
+ }
+ WebRtc_UWord32 timeDiffMS = _clock.GetTimeInMS() - _lastReceiveTime;
+ // Diff in time stamp since last received in order.
+ WebRtc_Word32 rtpTimeStampDiffMS = static_cast<WebRtc_Word32>(
+ rtpTimeStamp - _lastReceivedTimestamp) / frequencyKHz;
+
+ WebRtc_UWord16 minRTT = 0;
+ WebRtc_Word32 maxDelayMs = 0;
+ _rtpRtcp.RTT(_SSRC, NULL, NULL, &minRTT, NULL);
+ if (minRTT == 0) {
+ float jitter = _jitterQ4 >> 4; // Jitter variance in samples.
+ // Jitter standard deviation in samples.
+ float jitterStd = sqrt(jitter);
+ // 2 times the std deviation => 95% confidence.
+ // And transform to ms by dividing by the frequency in kHz.
+ maxDelayMs = static_cast<WebRtc_Word32>((2 * jitterStd) / frequencyKHz);
+
+ // Min maxDelayMs is 1.
+ if (maxDelayMs == 0) {
+ maxDelayMs = 1;
+ }
+ } else {
+ maxDelayMs = (minRTT / 3) + 1;
+ }
+ if (static_cast<WebRtc_Word32>(timeDiffMS) >
+ rtpTimeStampDiffMS + maxDelayMs) {
+ return true;
+ }
+ return false;
+}
+
+bool
+RTPReceiver::InOrderPacket(const WebRtc_UWord16 sequenceNumber) const
+{
+ if(_receivedSeqMax >= sequenceNumber)
+ {
+ if(!(_receivedSeqMax > 0xff00 && sequenceNumber < 0x0ff ))//detect wrap around
+ {
+ if(_receivedSeqMax - NACK_PACKETS_MAX_SIZE > sequenceNumber)
+ {
+ // we have a restart of the remote side
+ }else
+ {
+ // we received a retransmit of a packet we already have
+ return false;
+ }
+ }
+ }else
+ {
+ // check for a wrap
+ if(sequenceNumber > 0xff00 && _receivedSeqMax < 0x0ff )//detect wrap around
+ {
+ if(_receivedSeqMax - NACK_PACKETS_MAX_SIZE > sequenceNumber)
+ {
+ // we have a restart of the remote side
+ }else
+ {
+ // we received a retransmit of a packet we already have
+ return false;
+ }
+ }
+ }
+ return true;
+}
+
+WebRtc_UWord16
+RTPReceiver::SequenceNumber() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _lastReceivedSequenceNumber;
+}
+
+WebRtc_UWord32
+RTPReceiver::TimeStamp() const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ return _lastReceivedTimestamp;
+}
+
+WebRtc_UWord32 RTPReceiver::PayloadTypeToPayload(
+ const WebRtc_UWord8 payloadType,
+ Payload*& payload) const {
+
+ std::map<WebRtc_Word8, Payload*>::const_iterator it =
+ _payloadTypeMap.find(payloadType);
+
+ // check that this is a registered payload type
+ if (it == _payloadTypeMap.end()) {
+ return -1;
+ }
+ payload = it->second;
+ return 0;
+}
+
+// timeStamp of the last incoming packet that is the first packet of its frame
+WebRtc_Word32
+RTPReceiver::EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ WebRtc_UWord32 freq = 90000;
+ if(_audio)
+ {
+ freq = AudioFrequency();
+ }
+ if(_localTimeLastReceivedTimestamp == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
+ return -1;
+ }
+ //time in samples
+ WebRtc_UWord32 diff = GetCurrentRTP(&_clock, freq)
+ - _localTimeLastReceivedTimestamp;
+
+ timestamp = _lastReceivedTimestamp + diff;
+ return 0;
+}
+
+ // get the currently configured SSRC filter
+WebRtc_Word32
+RTPReceiver::SSRCFilter(WebRtc_UWord32& allowedSSRC) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+ if(_useSSRCFilter)
+ {
+ allowedSSRC = _SSRCFilter;
+ return 0;
+ }
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
+ return -1;
+}
+
+ // set a SSRC to be used as a filter for incoming RTP streams
+WebRtc_Word32
+RTPReceiver::SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC)
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ _useSSRCFilter = enable;
+ if(enable)
+ {
+ _SSRCFilter = allowedSSRC;
+ } else
+ {
+ _SSRCFilter = 0;
+ }
+ return 0;
+}
+
+// no criticalsection when called
+void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader) {
+ bool newSSRC = false;
+ bool reInitializeDecoder = false;
+ char payloadName[RTP_PAYLOAD_NAME_SIZE];
+ WebRtc_UWord32 frequency = 90000; // default video freq
+ WebRtc_UWord8 channels = 1;
+ WebRtc_UWord32 rate = 0;
+
+ {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ if (_SSRC != rtpHeader->header.ssrc ||
+ (_lastReceivedPayloadType == -1 && _SSRC == 0)) {
+ // we need the _payloadType to make the call if the remote SSRC is 0
+ newSSRC = true;
+
+ // reset last report
+ ResetStatistics();
+ RTPReceiverVideo::ResetOverUseDetector();
+
+ _lastReceivedTimestamp = 0;
+ _lastReceivedSequenceNumber = 0;
+ _lastReceivedTransmissionTimeOffset = 0;
+
+ if (_SSRC) { // do we have a SSRC? then the stream is restarted
+ // if we have the same codec? reinit decoder
+ if (rtpHeader->header.payloadType == _lastReceivedPayloadType) {
+ reInitializeDecoder = true;
+
+ std::map<WebRtc_Word8, Payload*>::iterator it =
+ _payloadTypeMap.find(rtpHeader->header.payloadType);
+
+ if (it == _payloadTypeMap.end()) {
+ return;
+ }
+ Payload* payload = it->second;
+ assert(payload);
+ payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ if(payload->audio) {
+ frequency = payload->typeSpecific.Audio.frequency;
+ channels = payload->typeSpecific.Audio.channels;
+ rate = payload->typeSpecific.Audio.rate;
+ } else {
+ frequency = 90000;
+ }
+ }
+ }
+ _SSRC = rtpHeader->header.ssrc;
+ }
+ }
+ if(newSSRC) {
+ // we need to get this to our RTCP sender and receiver
+ // need to do this outside critical section
+ _rtpRtcp.SetRemoteSSRC(rtpHeader->header.ssrc);
+ }
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if(_cbRtpFeedback) {
+ if(newSSRC) {
+ _cbRtpFeedback->OnIncomingSSRCChanged(_id, rtpHeader->header.ssrc);
+ }
+ if(reInitializeDecoder) {
+ if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id,
+ rtpHeader->header.payloadType, payloadName, frequency, channels,
+ rate)) { // new stream same codec
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Failed to create decoder for payload type:%d",
+ rtpHeader->header.payloadType);
+ }
+ }
+ }
+}
+
+// no criticalsection when called
+WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
+ const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_Word8 firstPayloadByte,
+ bool& isRED,
+ AudioPayload& audioSpecificPayload,
+ VideoPayload& videoSpecificPayload) {
+ bool reInitializeDecoder = false;
+
+ char payloadName[RTP_PAYLOAD_NAME_SIZE];
+ WebRtc_Word8 payloadType = rtpHeader->header.payloadType;
+
+ {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ if (payloadType != _lastReceivedPayloadType) {
+ if (REDPayloadType(payloadType)) {
+ // get the real codec payload type
+ payloadType = firstPayloadByte & 0x7f;
+ isRED = true;
+
+ if (REDPayloadType(payloadType)) {
+ // Invalid payload type, traced by caller. If we proceeded here,
+ // this would be set as |_lastReceivedPayloadType|, and we would no
+ // longer catch corrupt packets at this level.
+ return -1;
+ }
+
+ //when we receive RED we need to check the real payload type
+ if (payloadType == _lastReceivedPayloadType) {
+ if(_audio)
+ {
+ memcpy(&audioSpecificPayload, &_lastReceivedAudioSpecific,
+ sizeof(_lastReceivedAudioSpecific));
+ } else {
+ memcpy(&videoSpecificPayload, &_lastReceivedVideoSpecific,
+ sizeof(_lastReceivedVideoSpecific));
+ }
+ return 0;
+ }
+ }
+ if (_audio) {
+ if (TelephoneEventPayloadType(payloadType)) {
+ // don't do callbacks for DTMF packets
+ isRED = false;
+ return 0;
+ }
+ // frequency is updated for CNG
+ if (CNGPayloadType(payloadType, audioSpecificPayload.frequency)) {
+ // don't do callbacks for DTMF packets
+ isRED = false;
+ return 0;
+ }
+ }
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.find(payloadType);
+
+ // check that this is a registered payload type
+ if (it == _payloadTypeMap.end()) {
+ return -1;
+ }
+ Payload* payload = it->second;
+ assert(payload);
+ payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ _lastReceivedPayloadType = payloadType;
+
+ reInitializeDecoder = true;
+
+ if(payload->audio) {
+ memcpy(&_lastReceivedAudioSpecific, &(payload->typeSpecific.Audio),
+ sizeof(_lastReceivedAudioSpecific));
+ memcpy(&audioSpecificPayload, &(payload->typeSpecific.Audio),
+ sizeof(_lastReceivedAudioSpecific));
+ } else {
+ memcpy(&_lastReceivedVideoSpecific, &(payload->typeSpecific.Video),
+ sizeof(_lastReceivedVideoSpecific));
+ memcpy(&videoSpecificPayload, &(payload->typeSpecific.Video),
+ sizeof(_lastReceivedVideoSpecific));
+
+ if (_lastReceivedVideoSpecific.videoCodecType == kRtpFecVideo)
+ {
+ // Only reset the decoder on media packets.
+ reInitializeDecoder = false;
+ } else {
+ if (_lastReceivedMediaPayloadType == _lastReceivedPayloadType) {
+ // Only reset the decoder if the media codec type has changed.
+ reInitializeDecoder = false;
+ }
+ _lastReceivedMediaPayloadType = _lastReceivedPayloadType;
+ }
+ }
+ if (reInitializeDecoder) {
+ // reset statistics
+ ResetStatistics();
+ }
+ } else {
+ if(_audio)
+ {
+ memcpy(&audioSpecificPayload, &_lastReceivedAudioSpecific,
+ sizeof(_lastReceivedAudioSpecific));
+ } else
+ {
+ memcpy(&videoSpecificPayload, &_lastReceivedVideoSpecific,
+ sizeof(_lastReceivedVideoSpecific));
+ }
+ isRED = false;
+ }
+ } // end critsect
+ if (reInitializeDecoder) {
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if (_cbRtpFeedback) {
+ // create new decoder instance
+ if(_audio) {
+ if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id, payloadType,
+ payloadName, audioSpecificPayload.frequency,
+ audioSpecificPayload.channels, audioSpecificPayload.rate)) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Failed to create audio decoder for payload type:%d",
+ payloadType);
+ return -1; // Wrong payload type
+ }
+ } else {
+ if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id, payloadType,
+ payloadName, 90000, 1, 0)) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Failed to create video decoder for payload type:%d",
+ payloadType);
+ return -1; // Wrong payload type
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+// no criticalsection when called
+void RTPReceiver::CheckCSRC(const WebRtcRTPHeader* rtpHeader) {
+ WebRtc_Word32 numCSRCsDiff = 0;
+ WebRtc_UWord32 oldRemoteCSRC[kRtpCsrcSize];
+ WebRtc_UWord8 oldNumCSRCs = 0;
+
+ {
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ if (TelephoneEventPayloadType(rtpHeader->header.payloadType)) {
+ // Don't do this for DTMF packets
+ return;
+ }
+ _numEnergy = rtpHeader->type.Audio.numEnergy;
+ if (rtpHeader->type.Audio.numEnergy > 0 &&
+ rtpHeader->type.Audio.numEnergy <= kRtpCsrcSize) {
+ memcpy(_currentRemoteEnergy,
+ rtpHeader->type.Audio.arrOfEnergy,
+ rtpHeader->type.Audio.numEnergy);
+ }
+ oldNumCSRCs = _numCSRCs;
+ if (oldNumCSRCs > 0) {
+ // Make a copy of old.
+ memcpy(oldRemoteCSRC, _currentRemoteCSRC,
+ _numCSRCs * sizeof(WebRtc_UWord32));
+ }
+ const WebRtc_UWord8 numCSRCs = rtpHeader->header.numCSRCs;
+ if ((numCSRCs > 0) && (numCSRCs <= kRtpCsrcSize)) {
+ // Copy new
+ memcpy(_currentRemoteCSRC,
+ rtpHeader->header.arrOfCSRCs,
+ numCSRCs * sizeof(WebRtc_UWord32));
+ }
+ if (numCSRCs > 0 || oldNumCSRCs > 0) {
+ numCSRCsDiff = numCSRCs - oldNumCSRCs;
+ _numCSRCs = numCSRCs; // Update stored CSRCs.
+ } else {
+ // No change.
+ return;
+ }
+ } // End scoped CriticalSection.
+
+ CriticalSectionScoped lock(_criticalSectionCbs);
+ if (_cbRtpFeedback == NULL) {
+ return;
+ }
+ bool haveCalledCallback = false;
+ // Search for new CSRC in old array.
+ for (WebRtc_UWord8 i = 0; i < rtpHeader->header.numCSRCs; ++i) {
+ const WebRtc_UWord32 csrc = rtpHeader->header.arrOfCSRCs[i];
+
+ bool foundMatch = false;
+ for (WebRtc_UWord8 j = 0; j < oldNumCSRCs; ++j) {
+ if (csrc == oldRemoteCSRC[j]) { // old list
+ foundMatch = true;
+ break;
+ }
+ }
+ if (!foundMatch && csrc) {
+ // Didn't find it, report it as new.
+ haveCalledCallback = true;
+ _cbRtpFeedback->OnIncomingCSRCChanged(_id, csrc, true);
+ }
+ }
+ // Search for old CSRC in new array.
+ for (WebRtc_UWord8 i = 0; i < oldNumCSRCs; ++i) {
+ const WebRtc_UWord32 csrc = oldRemoteCSRC[i];
+
+ bool foundMatch = false;
+ for (WebRtc_UWord8 j = 0; j < rtpHeader->header.numCSRCs; ++j) {
+ if (csrc == rtpHeader->header.arrOfCSRCs[j]) {
+ foundMatch = true;
+ break;
+ }
+ }
+ if (!foundMatch && csrc) {
+ // Did not find it, report as removed.
+ haveCalledCallback = true;
+ _cbRtpFeedback->OnIncomingCSRCChanged(_id, csrc, false);
+ }
+ }
+ if (!haveCalledCallback) {
+ // If the CSRC list contain non-unique entries we will endup here.
+ // Using CSRC 0 to signal this event, not interop safe, other
+ // implementations might have CSRC 0 as avalid value.
+ if (numCSRCsDiff > 0) {
+ _cbRtpFeedback->OnIncomingCSRCChanged(_id, 0, true);
+ } else if (numCSRCsDiff < 0) {
+ _cbRtpFeedback->OnIncomingCSRCChanged(_id, 0, false);
+ }
+ }
+}
+
+WebRtc_Word32
+RTPReceiver::ResetStatistics()
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ _lastReportInorderPackets = 0;
+ _lastReportOldPackets = 0;
+ _lastReportSeqMax = 0;
+ _lastReportFractionLost = 0;
+ _lastReportCumulativeLost = 0;
+ _lastReportExtendedHighSeqNum = 0;
+ _lastReportJitter = 0;
+ _lastReportJitterTransmissionTimeOffset = 0;
+ _jitterQ4 = 0;
+ _jitterMaxQ4 = 0;
+ _cumulativeLoss = 0;
+ _jitterQ4TransmissionTimeOffset = 0;
+ _receivedSeqWraps = 0;
+ _receivedSeqMax = 0;
+ _receivedSeqFirst = 0;
+ _receivedByteCount = 0;
+ _receivedOldPacketCount = 0;
+ _receivedInorderPacketCount = 0;
+ return 0;
+}
+
+WebRtc_Word32
+RTPReceiver::ResetDataCounters()
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ _receivedByteCount = 0;
+ _receivedOldPacketCount = 0;
+ _receivedInorderPacketCount = 0;
+ _lastReportInorderPackets = 0;
+
+ return 0;
+}
+
+WebRtc_Word32
+RTPReceiver::Statistics(WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter,
+ WebRtc_UWord32 *max_jitter,
+ WebRtc_UWord32 *jitter_transmission_time_offset,
+ bool reset) const
+{
+ WebRtc_Word32 missing;
+ return Statistics(fraction_lost,
+ cum_lost,
+ ext_max,
+ jitter,
+ max_jitter,
+ jitter_transmission_time_offset,
+ &missing,
+ reset);
+}
+
+WebRtc_Word32
+RTPReceiver::Statistics(WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter,
+ WebRtc_UWord32 *max_jitter,
+ WebRtc_UWord32 *jitter_transmission_time_offset,
+ WebRtc_Word32 *missing,
+ bool reset) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ if (missing == NULL)
+ {
+ return -1;
+ }
+ if(_receivedSeqFirst == 0 && _receivedByteCount == 0)
+ {
+ // we have not received anything
+ // -1 required by RTCP sender
+ return -1;
+ }
+ if(!reset)
+ {
+ if(_lastReportInorderPackets == 0)
+ {
+ // no report
+ return -1;
+ }
+ // just get last report
+ if(fraction_lost)
+ {
+ *fraction_lost = _lastReportFractionLost;
+ }
+ if(cum_lost)
+ {
+ *cum_lost = _lastReportCumulativeLost; // 24 bits valid
+ }
+ if(ext_max)
+ {
+ *ext_max = _lastReportExtendedHighSeqNum;
+ }
+ if(jitter)
+ {
+ *jitter =_lastReportJitter;
+ }
+ if(max_jitter)
+ {
+ // note that the internal jitter value is in Q4
+ // and needs to be scaled by 1/16
+ *max_jitter = (_jitterMaxQ4 >> 4);
+ }
+ if(jitter_transmission_time_offset)
+ {
+ *jitter_transmission_time_offset =
+ _lastReportJitterTransmissionTimeOffset;
+ }
+ return 0;
+ }
+
+ if (_lastReportInorderPackets == 0)
+ {
+ // First time we send a report
+ _lastReportSeqMax = _receivedSeqFirst-1;
+ }
+ /*
+ * calc fraction lost
+ */
+ WebRtc_UWord16 expSinceLast = (_receivedSeqMax - _lastReportSeqMax);
+
+ if(_lastReportSeqMax > _receivedSeqMax)
+ {
+ // can we assume that the seqNum can't go decrease over a full RTCP period ?
+ expSinceLast = 0;
+ }
+
+ // number of received RTP packets since last report, counts all packets but not re-transmissions
+ WebRtc_UWord32 recSinceLast = _receivedInorderPacketCount - _lastReportInorderPackets;
+
+ if(_nackMethod == kNackOff)
+ {
+ // this is needed for re-ordered packets
+ WebRtc_UWord32 oldPackets = _receivedOldPacketCount - _lastReportOldPackets;
+ recSinceLast += oldPackets;
+ }else
+ {
+ // with NACK we don't know the expected retransmitions during the last second
+ // we know how many "old" packets we have received we just count the numer of
+ // old received to estimate the loss but it still does not guarantee an exact number
+ // since we run this based on time triggered by sending of a RTP packet this
+ // should have a minimum effect
+
+ // with NACK we don't count old packets as received since they are re-transmitted
+ // we use RTT to decide if a packet is re-ordered or re-transmitted
+ }
+
+ *missing = 0;
+ if(expSinceLast > recSinceLast)
+ {
+ *missing = (expSinceLast - recSinceLast);
+ }
+ WebRtc_UWord8 fractionLost = 0;
+ if(expSinceLast)
+ {
+ // scale 0 to 255, where 255 is 100% loss
+ fractionLost = (WebRtc_UWord8) ((255 * (*missing)) / expSinceLast);
+ }
+ if(fraction_lost)
+ {
+ *fraction_lost = fractionLost;
+ }
+ // we need a counter for cumulative loss too
+ _cumulativeLoss += *missing;
+
+ if(_jitterQ4 > _jitterMaxQ4)
+ {
+ _jitterMaxQ4 = _jitterQ4;
+ }
+ if(cum_lost)
+ {
+ *cum_lost = _cumulativeLoss;
+ }
+ if(ext_max)
+ {
+ *ext_max = (_receivedSeqWraps<<16) + _receivedSeqMax;
+ }
+ if(jitter)
+ {
+ // note that the internal jitter value is in Q4
+ // and needs to be scaled by 1/16
+ *jitter = (_jitterQ4 >> 4);
+ }
+ if(max_jitter)
+ {
+ // note that the internal jitter value is in Q4
+ // and needs to be scaled by 1/16
+ *max_jitter = (_jitterMaxQ4 >> 4);
+ }
+ if(jitter_transmission_time_offset)
+ {
+ // note that the internal jitter value is in Q4
+ // and needs to be scaled by 1/16
+ *jitter_transmission_time_offset =
+ (_jitterQ4TransmissionTimeOffset >> 4);
+ }
+ if(reset)
+ {
+ // store this report
+ _lastReportFractionLost = fractionLost;
+ _lastReportCumulativeLost = _cumulativeLoss; // 24 bits valid
+ _lastReportExtendedHighSeqNum = (_receivedSeqWraps<<16) + _receivedSeqMax;
+ _lastReportJitter = (_jitterQ4 >> 4);
+ _lastReportJitterTransmissionTimeOffset =
+ (_jitterQ4TransmissionTimeOffset >> 4);
+
+ // only for report blocks in RTCP SR and RR
+ _lastReportInorderPackets = _receivedInorderPacketCount;
+ _lastReportOldPackets = _receivedOldPacketCount;
+ _lastReportSeqMax = _receivedSeqMax;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPReceiver::DataCounters(WebRtc_UWord32 *bytesReceived,
+ WebRtc_UWord32 *packetsReceived) const
+{
+ CriticalSectionScoped lock(_criticalSectionRTPReceiver);
+
+ if(bytesReceived)
+ {
+ *bytesReceived = _receivedByteCount;
+ }
+ if(packetsReceived)
+ {
+ *packetsReceived = _receivedOldPacketCount + _receivedInorderPacketCount;
+ }
+ return 0;
+}
+
+void
+RTPReceiver::ProcessBitrate()
+{
+ CriticalSectionScoped cs(_criticalSectionRTPReceiver);
+
+ Bitrate::Process();
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h
new file mode 100644
index 0000000..4a14c25
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h
@@ -0,0 +1,264 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
+
+#include <map>
+
+#include "typedefs.h"
+#include "rtp_utility.h"
+
+#include "rtp_header_extension.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+#include "rtp_receiver_audio.h"
+#include "rtp_receiver_video.h"
+#include "rtcp_receiver_help.h"
+#include "Bitrate.h"
+
+namespace webrtc {
+class RtpRtcpFeedback;
+class ModuleRtpRtcpImpl;
+class Trace;
+
+class RTPReceiver : public RTPReceiverAudio, public RTPReceiverVideo, public Bitrate
+{
+public:
+ RTPReceiver(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner);
+
+ virtual ~RTPReceiver();
+
+ virtual void ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 Init();
+
+ RtpVideoCodecTypes VideoCodecType() const;
+ WebRtc_UWord32 MaxConfiguredBitrate() const;
+
+ WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
+ void PacketTimeout();
+
+ void ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_UWord32 now);
+
+ void ProcessBitrate();
+
+ WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
+ WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
+
+ WebRtc_Word32 RegisterReceivePayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
+
+ WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType);
+
+ WebRtc_Word32 ReceivePayloadType(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ WebRtc_Word8* payloadType) const;
+
+ WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ WebRtc_UWord32* frequency,
+ WebRtc_UWord8* channels,
+ WebRtc_UWord32* rate) const;
+
+ WebRtc_Word32 RemotePayload(char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ WebRtc_Word8* payloadType,
+ WebRtc_UWord32* frequency,
+ WebRtc_UWord8* channels) const;
+
+ WebRtc_Word32 IncomingRTPPacket(WebRtcRTPHeader* rtpheader,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 incomingRtpPacketLengt);
+
+ NACKMethod NACK() const ;
+
+ // Turn negative acknowledgement requests on/off
+ WebRtc_Word32 SetNACKStatus(const NACKMethod method);
+
+
+ // last received
+ virtual WebRtc_UWord32 TimeStamp() const;
+ virtual WebRtc_UWord16 SequenceNumber() const;
+
+ WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
+
+ WebRtc_UWord32 SSRC() const;
+
+ WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
+
+ WebRtc_Word32 Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const;
+
+ // get the currently configured SSRC filter
+ WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
+
+ // set a SSRC to be used as a filter for incoming RTP streams
+ WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
+
+ WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter, // will be moved from JB
+ WebRtc_UWord32 *max_jitter,
+ WebRtc_UWord32 *jitter_transmission_time_offset,
+ bool reset) const;
+
+ WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter, // will be moved from JB
+ WebRtc_UWord32 *max_jitter,
+ WebRtc_UWord32 *jitter_transmission_time_offset,
+ WebRtc_Word32 *missing,
+ bool reset) const;
+
+ WebRtc_Word32 DataCounters(WebRtc_UWord32 *bytesReceived,
+ WebRtc_UWord32 *packetsReceived) const;
+
+ WebRtc_Word32 ResetStatistics();
+
+ WebRtc_Word32 ResetDataCounters();
+
+ WebRtc_UWord16 PacketOHReceived() const;
+
+ WebRtc_UWord32 PacketCountReceived() const;
+
+ WebRtc_UWord32 ByteCountReceived() const;
+
+ WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
+ const WebRtc_UWord8 id);
+
+ WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
+
+ void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
+
+ virtual WebRtc_UWord32 PayloadTypeToPayload(const WebRtc_UWord8 payloadType,
+ ModuleRTPUtility::Payload*& payload) const;
+ /*
+ * RTX
+ */
+ void SetRTXStatus(const bool enable, const WebRtc_UWord32 SSRC);
+
+ void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
+
+protected:
+ virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader);
+
+ virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequenceNumber,
+ const WebRtc_UWord32 rtpTimeStamp) const;
+
+
+ void UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord16 bytes,
+ const bool oldPacket);
+
+ virtual WebRtc_Word8 REDPayloadType() const;
+
+private:
+ // Is RED configured with payload type payloadType
+ bool REDPayloadType(const WebRtc_Word8 payloadType) const;
+
+ bool InOrderPacket(const WebRtc_UWord16 sequenceNumber) const;
+
+ void CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader);
+ void CheckCSRC(const WebRtcRTPHeader* rtpHeader);
+ WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
+ const WebRtc_Word8 firstPayloadByte,
+ bool& isRED,
+ ModuleRTPUtility::AudioPayload& audioSpecific,
+ ModuleRTPUtility::VideoPayload& videoSpecific);
+
+ void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
+ bool ProcessNACKBitRate(WebRtc_UWord32 now);
+
+private:
+ WebRtc_Word32 _id;
+ const bool _audio;
+ ModuleRtpRtcpImpl& _rtpRtcp;
+
+ CriticalSectionWrapper* _criticalSectionCbs;
+ RtpFeedback* _cbRtpFeedback;
+ RtpData* _cbRtpData;
+
+ CriticalSectionWrapper* _criticalSectionRTPReceiver;
+ mutable WebRtc_UWord32 _lastReceiveTime;
+ WebRtc_UWord16 _lastReceivedPayloadLength;
+ WebRtc_Word8 _lastReceivedPayloadType;
+ WebRtc_Word8 _lastReceivedMediaPayloadType;
+
+ ModuleRTPUtility::AudioPayload _lastReceivedAudioSpecific;
+ ModuleRTPUtility::VideoPayload _lastReceivedVideoSpecific;
+
+ WebRtc_UWord32 _packetTimeOutMS;
+
+ WebRtc_Word8 _redPayloadType;
+
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
+ RtpHeaderExtensionMap _rtpHeaderExtensionMap;
+
+ // SSRCs
+ WebRtc_UWord32 _SSRC;
+ WebRtc_UWord8 _numCSRCs;
+ WebRtc_UWord32 _currentRemoteCSRC[kRtpCsrcSize];
+ WebRtc_UWord8 _numEnergy;
+ WebRtc_UWord8 _currentRemoteEnergy[kRtpCsrcSize];
+
+ bool _useSSRCFilter;
+ WebRtc_UWord32 _SSRCFilter;
+
+ // stats on received RTP packets
+ WebRtc_UWord32 _jitterQ4;
+ mutable WebRtc_UWord32 _jitterMaxQ4;
+ mutable WebRtc_UWord32 _cumulativeLoss;
+ WebRtc_UWord32 _jitterQ4TransmissionTimeOffset;
+
+ WebRtc_UWord32 _localTimeLastReceivedTimestamp;
+ WebRtc_UWord32 _lastReceivedTimestamp;
+ WebRtc_UWord16 _lastReceivedSequenceNumber;
+ WebRtc_Word32 _lastReceivedTransmissionTimeOffset;
+ WebRtc_UWord16 _receivedSeqFirst;
+ WebRtc_UWord16 _receivedSeqMax;
+ WebRtc_UWord16 _receivedSeqWraps;
+
+ // current counter values
+ WebRtc_UWord16 _receivedPacketOH;
+ WebRtc_UWord32 _receivedByteCount;
+ WebRtc_UWord32 _receivedOldPacketCount;
+ WebRtc_UWord32 _receivedInorderPacketCount;
+
+ // counter values when we sent the last report
+ mutable WebRtc_UWord32 _lastReportInorderPackets;
+ mutable WebRtc_UWord32 _lastReportOldPackets;
+ mutable WebRtc_UWord16 _lastReportSeqMax;
+ mutable WebRtc_UWord8 _lastReportFractionLost;
+ mutable WebRtc_UWord32 _lastReportCumulativeLost; // 24 bits valid
+ mutable WebRtc_UWord32 _lastReportExtendedHighSeqNum;
+ mutable WebRtc_UWord32 _lastReportJitter;
+ mutable WebRtc_UWord32 _lastReportJitterTransmissionTimeOffset;
+
+ NACKMethod _nackMethod;
+
+ bool _RTX;
+ WebRtc_UWord32 _ssrcRTX;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
new file mode 100644
index 0000000..b61157d
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -0,0 +1,590 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_receiver_audio.h"
+
+#include <cassert> //assert
+#include <cstring> // memcpy()
+#include <math.h> // pow()
+
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id):
+ _id(id),
+ _lastReceivedFrequency(8000),
+ _telephoneEvent(false),
+ _telephoneEventForwardToDecoder(false),
+ _telephoneEventDetectEndOfTone(false),
+ _telephoneEventPayloadType(-1),
+ _cngNBPayloadType(-1),
+ _cngWBPayloadType(-1),
+ _cngSWBPayloadType(-1),
+ _cngPayloadType(-1),
+ _G722PayloadType(-1),
+ _lastReceivedG722(false),
+ _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbAudioFeedback(NULL)
+{
+}
+
+RTPReceiverAudio::~RTPReceiverAudio()
+{
+ delete _criticalSectionFeedback;
+}
+
+WebRtc_Word32 RTPReceiverAudio::Init() {
+ _lastReceivedFrequency = 8000;
+ _telephoneEvent = false;
+ _telephoneEventForwardToDecoder = false;
+ _telephoneEventDetectEndOfTone = false;
+ _telephoneEventPayloadType = -1;
+
+ _telephoneEventReported.clear();
+
+ _cngNBPayloadType = -1;
+ _cngWBPayloadType = -1;
+ _cngSWBPayloadType = -1;
+ _cngPayloadType = -1;
+ _G722PayloadType = -1;
+ _lastReceivedG722 = false;
+ return 0;
+}
+
+void
+RTPReceiverAudio::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+WebRtc_Word32
+RTPReceiverAudio::RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedback);
+ _cbAudioFeedback = incomingMessagesCallback;
+ return 0;
+}
+
+WebRtc_UWord32
+RTPReceiverAudio::AudioFrequency() const
+{
+ if(_lastReceivedG722)
+ {
+ return 8000;
+ }
+ return _lastReceivedFrequency;
+}
+
+// Outband TelephoneEvent(DTMF) detection
+WebRtc_Word32
+RTPReceiverAudio::SetTelephoneEventStatus(const bool enable,
+ const bool forwardToDecoder,
+ const bool detectEndOfTone)
+{
+ _telephoneEvent= enable;
+ _telephoneEventDetectEndOfTone = detectEndOfTone;
+ _telephoneEventForwardToDecoder = forwardToDecoder;
+ return 0;
+}
+
+ // Is outband TelephoneEvent(DTMF) turned on/off?
+bool
+RTPReceiverAudio::TelephoneEvent() const
+{
+ return _telephoneEvent;
+}
+
+// Is forwarding of outband telephone events turned on/off?
+bool
+RTPReceiverAudio::TelephoneEventForwardToDecoder() const
+{
+ return _telephoneEventForwardToDecoder;
+}
+
+bool
+RTPReceiverAudio::TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const
+{
+ return (_telephoneEventPayloadType == payloadType)?true:false;
+}
+
+bool
+RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
+ WebRtc_UWord32& frequency)
+{
+ // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
+ if(_cngNBPayloadType == payloadType)
+ {
+ frequency = 8000;
+ if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngNBPayloadType))
+ {
+ ResetStatistics();
+ }
+ _cngPayloadType = _cngNBPayloadType;
+ return true;
+ } else if(_cngWBPayloadType == payloadType)
+ {
+ // if last received codec is G.722 we must use frequency 8000
+ if(_lastReceivedG722)
+ {
+ frequency = 8000;
+ } else
+ {
+ frequency = 16000;
+ }
+ if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngWBPayloadType))
+ {
+ ResetStatistics();
+ }
+ _cngPayloadType = _cngWBPayloadType;
+ return true;
+ }else if(_cngSWBPayloadType == payloadType)
+ {
+ frequency = 32000;
+ if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngSWBPayloadType))
+ {
+ ResetStatistics();
+ }
+ _cngPayloadType = _cngSWBPayloadType;
+ return true;
+ }else
+ {
+ // not CNG
+ if(_G722PayloadType == payloadType)
+ {
+ _lastReceivedG722 = true;
+ }else
+ {
+ _lastReceivedG722 = false;
+ }
+ }
+ return false;
+}
+
+/*
+ Sample based or frame based codecs based on RFC 3551
+
+ NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
+ The correct rate is 4 bits/sample.
+
+ name of sampling default
+ encoding sample/frame bits/sample rate ms/frame ms/packet
+
+ Sample based audio codecs
+ DVI4 sample 4 var. 20
+ G722 sample 4 16,000 20
+ G726-40 sample 5 8,000 20
+ G726-32 sample 4 8,000 20
+ G726-24 sample 3 8,000 20
+ G726-16 sample 2 8,000 20
+ L8 sample 8 var. 20
+ L16 sample 16 var. 20
+ PCMA sample 8 var. 20
+ PCMU sample 8 var. 20
+
+ Frame based audio codecs
+ G723 frame N/A 8,000 30 30
+ G728 frame N/A 8,000 2.5 20
+ G729 frame N/A 8,000 10 20
+ G729D frame N/A 8,000 10 20
+ G729E frame N/A 8,000 10 20
+ GSM frame N/A 8,000 20 20
+ GSM-EFR frame N/A 8,000 20 20
+ LPC frame N/A 8,000 20 20
+ MPA frame N/A var. var.
+
+ G7221 frame N/A
+*/
+
+ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
+ _telephoneEventPayloadType = payloadType;
+ }
+ if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
+ // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
+ if(frequency == 8000){
+ _cngNBPayloadType = payloadType;
+ } else if(frequency == 16000) {
+ _cngWBPayloadType = payloadType;
+ } else if(frequency == 32000) {
+ _cngSWBPayloadType = payloadType;
+ } else {
+ assert(false);
+ return NULL;
+ }
+ }
+ WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
+ bool isTrueStereo = false; // Default value
+ if (ModuleRTPUtility::StringCompare(payloadName, "DVI4", 4)) {
+ bitsPerSample = 4;
+ } else if(ModuleRTPUtility::StringCompare(payloadName, "G722", 4)) {
+ if(ModuleRTPUtility::StringCompare(payloadName, "G7221", 5)) {
+ // frame based
+ } else {
+ _G722PayloadType = payloadType;
+ bitsPerSample = 4;
+ }
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7)) {
+ bitsPerSample = 5;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7)) {
+ bitsPerSample = 4;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7)) {
+ bitsPerSample = 3;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7)) {
+ bitsPerSample = 2;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2)) {
+ bitsPerSample = 8;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3)) {
+ bitsPerSample = 16;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4)) {
+ bitsPerSample = 8;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4)) {
+ bitsPerSample = 8;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"CELT",4))
+ {
+ isTrueStereo = true;
+ }
+ ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ payload->typeSpecific.Audio.frequency = frequency;
+ payload->typeSpecific.Audio.channels = channels;
+ payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
+ payload->typeSpecific.Audio.rate = rate;
+ payload->typeSpecific.Audio.trueStereoCodec = isTrueStereo;
+ payload->audio = true;
+ return payload;
+}
+
+// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
+WebRtc_Word32
+RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadLength,
+ const ModuleRTPUtility::AudioPayload& audioSpecific,
+ const bool isRED)
+{
+ WebRtc_UWord8 newEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
+ WebRtc_UWord8 removedEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
+ WebRtc_UWord8 numberOfNewEvents = 0;
+ WebRtc_UWord8 numberOfRemovedEvents = 0;
+ bool telephoneEventPacket = TelephoneEventPayloadType(rtpHeader->header.payloadType);
+
+ if(payloadLength == 0)
+ {
+ return 0;
+ }
+
+ {
+ CriticalSectionScoped lock(_criticalSectionFeedback);
+
+ if(telephoneEventPacket)
+ {
+ // RFC 4733 2.3
+ /*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | event |E|R| volume | duration |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+ if(payloadLength % 4 != 0)
+ {
+ return -1;
+ }
+ WebRtc_UWord8 numberOfEvents = payloadLength / 4;
+
+ // sanity
+ if(numberOfEvents >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS)
+ {
+ numberOfEvents = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
+ }
+ for (int n = 0; n < numberOfEvents; n++)
+ {
+ bool end = (payloadData[(4*n)+1] & 0x80)? true:false;
+
+ std::set<WebRtc_UWord8>::iterator event =
+ _telephoneEventReported.find(payloadData[4*n]);
+
+ if(event != _telephoneEventReported.end())
+ {
+ // we have already seen this event
+ if(end)
+ {
+ removedEvents[numberOfRemovedEvents]= payloadData[4*n];
+ numberOfRemovedEvents++;
+ _telephoneEventReported.erase(payloadData[4*n]);
+ }
+ }else
+ {
+ if(end)
+ {
+ // don't add if it's a end of a tone
+ }else
+ {
+ newEvents[numberOfNewEvents] = payloadData[4*n];
+ numberOfNewEvents++;
+ _telephoneEventReported.insert(payloadData[4*n]);
+ }
+ }
+ }
+
+ // RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
+ // should not be a problem since we don't care about the duration
+
+ // RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet
+ }
+
+ if(_telephoneEvent && _cbAudioFeedback)
+ {
+ for (int n = 0; n < numberOfNewEvents; n++)
+ {
+ _cbAudioFeedback->OnReceivedTelephoneEvent(_id, newEvents[n], false);
+ }
+ if(_telephoneEventDetectEndOfTone)
+ {
+ for (int n = 0; n < numberOfRemovedEvents; n++)
+ {
+ _cbAudioFeedback->OnReceivedTelephoneEvent(_id, removedEvents[n], true);
+ }
+ }
+ }
+ }
+ if(! telephoneEventPacket )
+ {
+ _lastReceivedFrequency = audioSpecific.frequency;
+ }
+
+ // Check if this is a CNG packet, receiver might want to know
+ WebRtc_UWord32 dummy;
+ if(CNGPayloadType(rtpHeader->header.payloadType, dummy))
+ {
+ rtpHeader->type.Audio.isCNG=true;
+ rtpHeader->frameType = kAudioFrameCN;
+ }else
+ {
+ rtpHeader->frameType = kAudioFrameSpeech;
+ rtpHeader->type.Audio.isCNG=false;
+ }
+
+ // check if it's a DTMF event, hence something we can playout
+ if(telephoneEventPacket)
+ {
+ if(!_telephoneEventForwardToDecoder)
+ {
+ // don't forward event to decoder
+ return 0;
+ }
+ std::set<WebRtc_UWord8>::iterator first =
+ _telephoneEventReported.begin();
+ if(first != _telephoneEventReported.end() && *first > 15)
+ {
+ // don't forward non DTMF events
+ return 0;
+ }
+ }
+ if(isRED && !(payloadData[0] & 0x80))
+ {
+ // we recive only one frame packed in a RED packet remove the RED wrapper
+ rtpHeader->header.payloadType = payloadData[0];
+
+ // only one frame in the RED strip the one byte to help NetEq
+ return CallbackOfReceivedPayloadData(payloadData+1,
+ payloadLength-1,
+ rtpHeader);
+ }
+ if(audioSpecific.channels > 1)
+ {
+ WebRtc_Word32 retVal = 0;
+ WebRtc_UWord16 channelLength = payloadLength/audioSpecific.channels;
+
+ if(audioSpecific.bitsPerSample > 0)
+ {
+ // sanity
+ assert((payloadLength*8)%audioSpecific.bitsPerSample == 0);
+
+ // sample based codec
+
+ // build matrix
+ WebRtc_UWord8 matrix[IP_PACKET_SIZE];
+ WebRtc_UWord32 offsetBytes = 0;
+ WebRtc_UWord32 offsetBytesInsert = 0;
+ // initialize matrix to 0
+ memset(matrix, 0, audioSpecific.channels*channelLength);
+
+ switch(audioSpecific.bitsPerSample)
+ {
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ case 5:
+ case 6:
+ case 7:
+ {
+ WebRtc_UWord32 offsetSamples = 0;
+ WebRtc_UWord32 offsetSamplesInsert = 0;
+ WebRtc_UWord16 bitMask = (WebRtc_UWord16)ModuleRTPUtility::pow2(audioSpecific.bitsPerSample)-1;
+ WebRtc_UWord16 samplesPerChannel =payloadLength*8/audioSpecific.bitsPerSample/audioSpecific.channels;
+
+ for(WebRtc_UWord32 i = 0; i < samplesPerChannel; i++)
+ {
+ WebRtc_UWord8 insertShift = (WebRtc_UWord8)((offsetSamplesInsert+audioSpecific.bitsPerSample)%16);
+ insertShift = 16 - insertShift; // inverse the calculation
+
+ for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
+ {
+ // get sample
+ WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
+
+ // check that we don't read outside the memory
+ if(offsetBytes < (WebRtc_UWord32)payloadLength -2)
+ {
+ s += payloadData[offsetBytes+1];
+ }
+
+ WebRtc_UWord8 readShift = (WebRtc_UWord8)((offsetSamples+audioSpecific.bitsPerSample)%16);
+ readShift = 16 - readShift; // inverse the calculation
+ s >>= readShift;
+ s &= bitMask;
+
+ // prepare for reading next sample
+ offsetSamples += audioSpecific.bitsPerSample;
+ if(readShift <= audioSpecific.bitsPerSample)
+ {
+ // next does not fitt
+ // or fitt exactly
+ offsetSamples -= 8;
+ offsetBytes++;
+ }
+
+ // insert sample into matrix
+ WebRtc_UWord32 columOffset = j*channelLength;
+
+ WebRtc_UWord16 insert = s << insertShift;
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ matrix[columOffset+offsetBytesInsert] |= static_cast<WebRtc_UWord8>(insert>>8);
+ matrix[columOffset+offsetBytesInsert+1] |= static_cast<WebRtc_UWord8>(insert);
+#else
+ WebRtc_UWord16* matrixU16 = (WebRtc_UWord16*)&(matrix[columOffset+offsetBytesInsert]);
+ matrixU16[0] |= (s << insertShift);
+#endif
+ }
+ // prepare for writing next sample
+ offsetSamplesInsert += audioSpecific.bitsPerSample;
+ if(insertShift <= audioSpecific.bitsPerSample)
+ {
+ // next does not fitt
+ // or fitt exactly
+ offsetSamplesInsert -= 8;
+ offsetBytesInsert++;
+ }
+ }
+ }
+ break;
+ case 8:
+ {
+ WebRtc_UWord32 sample = 0;
+ for(WebRtc_UWord32 i = 0; i < channelLength; i++)
+ {
+ for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
+ {
+ WebRtc_UWord32 columOffset = j*channelLength;
+ matrix[columOffset + i] = payloadData[sample++];
+ }
+ }
+ }
+ break;
+ case 16:
+ {
+ WebRtc_UWord32 sample = 0;
+ for(WebRtc_UWord32 i = 0; i < channelLength; i +=2)
+ {
+ for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
+ {
+ WebRtc_UWord32 columOffset = j*channelLength;
+ matrix[columOffset + i] = payloadData[sample++];
+ matrix[columOffset + i + 1] = payloadData[sample++];
+ }
+ }
+ }
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+ // we support 16 bits sample
+ // callback for all channels
+ for(int channel = 0; channel < audioSpecific.channels && retVal == 0; channel++)
+ {
+ // one callback per channel
+ rtpHeader->type.Audio.channel = channel+1;
+
+ if(channel == 0)
+ {
+ // include the original packet only in the first callback
+ retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
+ channelLength,
+ rtpHeader);
+ } else
+ {
+ retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
+ channelLength,
+ rtpHeader);
+ }
+ }
+ } else if (audioSpecific.trueStereoCodec)
+ {
+ // One callback with the whole payload for each channel.
+ for(int channel = 1; (channel <= audioSpecific.channels) &&
+ (retVal == 0); channel++)
+ {
+ // One callback per channel.
+ rtpHeader->type.Audio.channel = channel;
+ retVal = CallbackOfReceivedPayloadData(payloadData,
+ payloadLength,
+ rtpHeader);
+ }
+ } else
+ {
+ for(int channel = 1; channel <= audioSpecific.channels && retVal == 0; channel++)
+ {
+ // one callback per channel
+ rtpHeader->type.Audio.channel = channel;
+
+ if(channel == 1)
+ {
+ // include the original packet only in the first callback
+ retVal = CallbackOfReceivedPayloadData(payloadData,
+ channelLength,
+ rtpHeader);
+ } else
+ {
+ retVal = CallbackOfReceivedPayloadData(payloadData,
+ channelLength,
+ rtpHeader);
+ }
+ payloadData += channelLength;
+ }
+ }
+ return retVal;
+ }else
+ {
+ rtpHeader->type.Audio.channel = 1;
+ return CallbackOfReceivedPayloadData(payloadData,
+ payloadLength,
+ rtpHeader);
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
new file mode 100644
index 0000000..89cd062
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -0,0 +1,98 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
+
+#include <set>
+
+#include "rtp_rtcp_defines.h"
+#include "rtp_utility.h"
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class RTPReceiverAudio
+{
+public:
+ RTPReceiverAudio(const WebRtc_Word32 id);
+ virtual ~RTPReceiverAudio();
+
+ virtual void ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 Init();
+
+ WebRtc_Word32 RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback);
+
+ ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
+
+ WebRtc_UWord32 AudioFrequency() const;
+
+ // Outband TelephoneEvent (DTMF) detection
+ WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
+ const bool forwardToDecoder,
+ const bool detectEndOfTone);
+
+ // Is outband DTMF(AVT) turned on/off?
+ bool TelephoneEvent() const ;
+
+ // Is forwarding of outband telephone events turned on/off?
+ bool TelephoneEventForwardToDecoder() const ;
+
+ // Is TelephoneEvent configured with payload type payloadType
+ bool TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const;
+
+ // Is CNG configured with payload type payloadType
+ bool CNGPayloadType(const WebRtc_Word8 payloadType, WebRtc_UWord32& frequency);
+
+ WebRtc_Word32 ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadLength,
+ const ModuleRTPUtility::AudioPayload& audioSpecific,
+ const bool isRED);
+
+ virtual WebRtc_Word32 ResetStatistics() = 0;
+
+protected:
+ virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader) = 0;
+private:
+ WebRtc_Word32 _id;
+
+ WebRtc_UWord32 _lastReceivedFrequency;
+
+ bool _telephoneEvent;
+ bool _telephoneEventForwardToDecoder;
+ bool _telephoneEventDetectEndOfTone;
+ WebRtc_Word8 _telephoneEventPayloadType;
+ std::set<WebRtc_UWord8> _telephoneEventReported;
+
+ WebRtc_Word8 _cngNBPayloadType;
+ WebRtc_Word8 _cngWBPayloadType;
+ WebRtc_Word8 _cngSWBPayloadType;
+ WebRtc_Word8 _cngPayloadType;
+
+ // G722 is special since it use the wrong number of RTP samples in timestamp VS. number of samples in the frame
+ WebRtc_Word8 _G722PayloadType;
+ bool _lastReceivedG722;
+
+ CriticalSectionWrapper* _criticalSectionFeedback;
+ RtpAudioFeedback* _cbAudioFeedback;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
new file mode 100644
index 0000000..d6a3c17
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -0,0 +1,517 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_receiver_video.h"
+
+#include <cassert> //assert
+#include <cstring> // memcpy()
+#include <math.h>
+
+#include "critical_section_wrapper.h"
+#include "receiver_fec.h"
+#include "rtp_rtcp_impl.h"
+#include "rtp_utility.h"
+
+namespace webrtc {
+WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
+{
+ return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
+}
+
+RTPReceiverVideo::RTPReceiverVideo():
+ _id(0),
+ _rtpRtcp(NULL),
+ _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbVideoFeedback(NULL),
+ _criticalSectionReceiverVideo(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _completeFrame(false),
+ _packetStartTimeMs(0),
+ _receivedBW(),
+ _estimatedBW(0),
+ _currentFecFrameDecoded(false),
+ _receiveFEC(NULL),
+ _overUseDetector(),
+ _videoBitRate(),
+ _lastBitRateChange(0),
+ _packetOverHead(28)
+{
+ memset(_receivedBW, 0,sizeof(_receivedBW));
+}
+
+RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
+ ModuleRtpRtcpImpl* owner):
+ _id(id),
+ _rtpRtcp(owner),
+ _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
+ _cbVideoFeedback(NULL),
+ _criticalSectionReceiverVideo(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _completeFrame(false),
+ _packetStartTimeMs(0),
+ _receivedBW(),
+ _estimatedBW(0),
+ _currentFecFrameDecoded(false),
+ _receiveFEC(NULL),
+ _overUseDetector(),
+ _videoBitRate(),
+ _lastBitRateChange(0),
+ _packetOverHead(28)
+{
+ memset(_receivedBW, 0,sizeof(_receivedBW));
+}
+
+RTPReceiverVideo::~RTPReceiverVideo()
+{
+ delete _criticalSectionFeedback;
+ delete _criticalSectionReceiverVideo;
+ delete _receiveFEC;
+}
+
+WebRtc_Word32
+RTPReceiverVideo::Init()
+{
+ _completeFrame = false;
+ _packetStartTimeMs = 0;
+ _estimatedBW = 0;
+ _currentFecFrameDecoded = false;
+ _packetOverHead = 28;
+ for (int i = 0; i < BW_HISTORY_SIZE; i++)
+ {
+ _receivedBW[i] = 0;
+ }
+ ResetOverUseDetector();
+ return 0;
+}
+
+void
+RTPReceiverVideo::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+WebRtc_Word32
+RTPReceiverVideo::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedback);
+ _cbVideoFeedback = incomingMessagesCallback;
+ return 0;
+}
+
+void
+RTPReceiverVideo::UpdateBandwidthManagement(const WebRtc_UWord32 bitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs)
+{
+ CriticalSectionScoped lock(_criticalSectionFeedback);
+ if(_cbVideoFeedback)
+ {
+ _cbVideoFeedback->OnNetworkChanged(_id,
+ bitrateBps,
+ fractionLost,
+ roundTripTimeMs);
+ }
+}
+
+ModuleRTPUtility::Payload* RTPReceiverVideo::RegisterReceiveVideoPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 maxRate) {
+ RtpVideoCodecTypes videoType = kRtpNoVideo;
+ if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) {
+ videoType = kRtpVp8Video;
+ } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
+ videoType = kRtpNoVideo;
+ } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) {
+ // store this
+ if (_receiveFEC == NULL) {
+ _receiveFEC = new ReceiverFEC(_id, this);
+ }
+ _receiveFEC->SetPayloadTypeFEC(payloadType);
+ videoType = kRtpFecVideo;
+ } else {
+ return NULL;
+ }
+ ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
+
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ payload->typeSpecific.Video.videoCodecType = videoType;
+ payload->typeSpecific.Video.maxRate = maxRate;
+ payload->audio = false;
+ return payload;
+}
+
+void RTPReceiverVideo::ResetOverUseDetector()
+{
+ _overUseDetector.Reset();
+ _videoBitRate.Init();
+ _lastBitRateChange = 0;
+}
+
+// called under _criticalSectionReceiverVideo
+WebRtc_UWord16
+RTPReceiverVideo::EstimateBandwidth(const WebRtc_UWord16 bandwidth)
+{
+ // received fragments
+ // estimate BW
+
+ WebRtc_UWord16 bwSort[BW_HISTORY_SIZE];
+ for(int i = 0; i < BW_HISTORY_SIZE-1; i++)
+ {
+ _receivedBW[i] = _receivedBW[i+1];
+ bwSort[i] = _receivedBW[i+1];
+ }
+ _receivedBW[BW_HISTORY_SIZE-1] = bandwidth;
+ bwSort[BW_HISTORY_SIZE-1] = bandwidth;
+
+ WebRtc_UWord16 temp;
+ for (int i = BW_HISTORY_SIZE-1; i >= 0; i--)
+ {
+ for (int j = 1; j <= i; j++)
+ {
+ if (bwSort[j-1] > bwSort[j])
+ {
+ temp = bwSort[j-1];
+ bwSort[j-1] = bwSort[j];
+ bwSort[j] = temp;
+ }
+ }
+ }
+ int zeroCount = 0;
+ for (; zeroCount < BW_HISTORY_SIZE; zeroCount++)
+ {
+ if (bwSort[zeroCount]!= 0)
+ {
+ break;
+ }
+ }
+ WebRtc_UWord32 indexMedian = (BW_HISTORY_SIZE -1) - (BW_HISTORY_SIZE-zeroCount)/2;
+ WebRtc_UWord16 bandwidthMedian = bwSort[indexMedian];
+
+ if (bandwidthMedian > 0)
+ {
+ if (_estimatedBW == bandwidth)
+ {
+ // don't trigger a callback
+ bandwidthMedian = 0;
+ } else
+ {
+ _estimatedBW = bandwidthMedian;
+ }
+ } else
+ {
+ // can't be negative
+ bandwidthMedian = 0;
+ }
+
+ return bandwidthMedian;
+}
+
+// we have no critext when calling this
+// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
+WebRtc_Word32
+RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength,
+ const RtpVideoCodecTypes videoType,
+ const bool isRED,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 incomingRtpPacketSize,
+ const WebRtc_Word64 nowMS)
+{
+ WebRtc_Word32 retVal = 0;
+
+ _criticalSectionReceiverVideo->Enter();
+
+ _videoBitRate.Update(payloadDataLength + rtpHeader->header.paddingLength,
+ nowMS);
+
+ // Add headers, ideally we would like to include for instance
+ // Ethernet header here as well.
+ const WebRtc_UWord16 packetSize = payloadDataLength + _packetOverHead +
+ rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
+ _overUseDetector.Update(*rtpHeader, packetSize, nowMS);
+
+ if (isRED)
+ {
+ if(_receiveFEC == NULL)
+ {
+ _criticalSectionReceiverVideo->Leave();
+ return -1;
+ }
+ bool FECpacket = false;
+ retVal = _receiveFEC->AddReceivedFECPacket(
+ rtpHeader,
+ incomingRtpPacket,
+ payloadDataLength,
+ FECpacket);
+ if (retVal != -1)
+ retVal = _receiveFEC->ProcessReceivedFEC();
+ _criticalSectionReceiverVideo->Leave();
+
+ if(retVal == 0 && FECpacket)
+ {
+ // Callback with the received FEC packet.
+ // The normal packets are delivered after parsing.
+ // This contains the original RTP packet header but with
+ // empty payload and data length.
+ rtpHeader->frameType = kFrameEmpty;
+ // We need this for the routing.
+ WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
+ if(retVal != 0)
+ {
+ return retVal;
+ }
+ retVal = CallbackOfReceivedPayloadData(NULL, 0, rtpHeader);
+ }
+ }else
+ {
+ // will leave the _criticalSectionReceiverVideo critsect
+ retVal = ParseVideoCodecSpecificSwitch(rtpHeader,
+ payloadData,
+ payloadDataLength,
+ videoType);
+ }
+
+ // Update the remote rate control object and update the overuse
+ // detector with the current rate control region.
+ _criticalSectionReceiverVideo->Enter();
+ const RateControlInput input(_overUseDetector.State(),
+ _videoBitRate.BitRate(nowMS),
+ _overUseDetector.NoiseVar());
+ _criticalSectionReceiverVideo->Leave();
+
+ // Call the callback outside critical section
+ if (_rtpRtcp) {
+ const RateControlRegion region = _rtpRtcp->OnOverUseStateUpdate(input);
+
+ _criticalSectionReceiverVideo->Enter();
+ _overUseDetector.SetRateControlRegion(region);
+ _criticalSectionReceiverVideo->Leave();
+ }
+
+ return retVal;
+}
+
+WebRtc_Word32
+RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
+ WebRtc_UWord8* dataBuffer) const
+{
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(rtpHeader->header.payloadType);
+ if (rtpHeader->header.markerBit)
+ {
+ dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1
+ }
+
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, rtpHeader->header.sequenceNumber);
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp);
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
+
+ WebRtc_Word32 rtpHeaderLength = 12;
+
+ // Add the CSRCs if any
+ if (rtpHeader->header.numCSRCs > 0)
+ {
+ if(rtpHeader->header.numCSRCs > 16)
+ {
+ // error
+ assert(false);
+ }
+ WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
+ for (WebRtc_UWord32 i = 0; i < rtpHeader->header.numCSRCs; ++i)
+ {
+ ModuleRTPUtility::AssignUWord32ToBuffer(ptr, rtpHeader->header.arrOfCSRCs[i]);
+ ptr +=4;
+ }
+ dataBuffer[0] = (dataBuffer[0]&0xf0) | rtpHeader->header.numCSRCs;
+
+ // Update length of header
+ rtpHeaderLength += sizeof(WebRtc_UWord32)*rtpHeader->header.numCSRCs;
+ }
+ return rtpHeaderLength;
+}
+
+WebRtc_Word32
+RTPReceiverVideo::ReceiveRecoveredPacketCallback(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength)
+{
+ _criticalSectionReceiverVideo->Enter();
+
+ _currentFecFrameDecoded = true;
+
+ ModuleRTPUtility::Payload* payload = NULL;
+ if (PayloadTypeToPayload(rtpHeader->header.payloadType, payload) != 0)
+ {
+ return -1;
+ }
+ // here we can re-create the original lost packet so that we can use it for the relay
+ // we need to re-create the RED header too
+ WebRtc_UWord8 recoveredPacket[IP_PACKET_SIZE];
+ WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)BuildRTPheader(rtpHeader, recoveredPacket);
+
+ const WebRtc_UWord8 REDForFECHeaderLength = 1;
+
+ // replace pltype
+ recoveredPacket[1] &= 0x80; // reset
+ recoveredPacket[1] += REDPayloadType(); // replace with RED payload type
+
+ // add RED header
+ recoveredPacket[rtpHeaderLength] = rtpHeader->header.payloadType; // f-bit always 0
+
+ memcpy(recoveredPacket + rtpHeaderLength + REDForFECHeaderLength, payloadData, payloadDataLength);
+
+ return ParseVideoCodecSpecificSwitch(rtpHeader,
+ payloadData,
+ payloadDataLength,
+ payload->typeSpecific.Video.videoCodecType);
+}
+
+WebRtc_Word32 RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
+ WebRtcRTPHeader* rtpHeader) const {
+ switch (videoType) {
+ case kRtpNoVideo:
+ rtpHeader->type.Video.codec = kRTPVideoGeneric;
+ break;
+ case kRtpVp8Video:
+ rtpHeader->type.Video.codec = kRTPVideoVP8;
+ break;
+ case kRtpFecVideo:
+ rtpHeader->type.Video.codec = kRTPVideoFEC;
+ break;
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecificSwitch(
+ WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength,
+ const RtpVideoCodecTypes videoType) {
+ WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
+ if (retVal != 0) {
+ return retVal;
+ }
+ // All receive functions release _criticalSectionReceiverVideo before
+ // returning.
+ switch (videoType) {
+ case kRtpNoVideo:
+ return ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
+ case kRtpVp8Video:
+ return ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
+ case kRtpFecVideo:
+ break;
+ }
+ _criticalSectionReceiverVideo->Leave();
+ return -1;
+}
+
+WebRtc_Word32
+RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength)
+{
+ bool success;
+ ModuleRTPUtility::RTPPayload parsedPacket;
+ if (payloadDataLength == 0)
+ {
+ success = true;
+ parsedPacket.info.VP8.dataLength = 0;
+ } else
+ {
+ ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpVp8Video,
+ payloadData,
+ payloadDataLength,
+ _id);
+
+ success = rtpPayloadParser.Parse(parsedPacket);
+ }
+ // from here down we only work on local data
+ _criticalSectionReceiverVideo->Leave();
+
+ if (!success)
+ {
+ return -1;
+ }
+ if (parsedPacket.info.VP8.dataLength == 0)
+ {
+ // we have an "empty" VP8 packet, it's ok, could be one way video
+ // Inform the jitter buffer about this packet.
+ rtpHeader->frameType = kFrameEmpty;
+ if (CallbackOfReceivedPayloadData(NULL, 0, rtpHeader) != 0)
+ {
+ return -1;
+ }
+ return 0;
+ }
+ rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
+
+ RTPVideoHeaderVP8 *toHeader = &rtpHeader->type.Video.codecHeader.VP8;
+ ModuleRTPUtility::RTPPayloadVP8 *fromHeader = &parsedPacket.info.VP8;
+
+ rtpHeader->type.Video.isFirstPacket = fromHeader->beginningOfPartition
+ && (fromHeader->partitionID == 0);
+ toHeader->pictureId = fromHeader->hasPictureID ? fromHeader->pictureID :
+ kNoPictureId;
+ toHeader->tl0PicIdx = fromHeader->hasTl0PicIdx ? fromHeader->tl0PicIdx :
+ kNoTl0PicIdx;
+ if (fromHeader->hasTID) {
+ toHeader->temporalIdx = fromHeader->tID;
+ toHeader->layerSync = fromHeader->layerSync;
+ } else {
+ toHeader->temporalIdx = kNoTemporalIdx;
+ toHeader->layerSync = false;
+ }
+ toHeader->keyIdx = fromHeader->hasKeyIdx ? fromHeader->keyIdx : kNoKeyIdx;
+
+ toHeader->frameWidth = fromHeader->frameWidth;
+ toHeader->frameHeight = fromHeader->frameHeight;
+
+ toHeader->partitionId = fromHeader->partitionID;
+ toHeader->beginningOfPartition = fromHeader->beginningOfPartition;
+
+ if(CallbackOfReceivedPayloadData(parsedPacket.info.VP8.data,
+ parsedPacket.info.VP8.dataLength,
+ rtpHeader) != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32
+RTPReceiverVideo::ReceiveGenericCodec(WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength)
+{
+ rtpHeader->frameType = kVideoFrameKey;
+
+ if(((SequenceNumber() + 1) == rtpHeader->header.sequenceNumber) &&
+ (TimeStamp() != rtpHeader->header.timestamp))
+ {
+ rtpHeader->type.Video.isFirstPacket = true;
+ }
+ _criticalSectionReceiverVideo->Leave();
+
+ if(CallbackOfReceivedPayloadData(payloadData, payloadDataLength, rtpHeader) != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+void RTPReceiverVideo::SetPacketOverHead(WebRtc_UWord16 packetOverHead)
+{
+ _packetOverHead = packetOverHead;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h
new file mode 100644
index 0000000..5e0138a
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
+
+#include "rtp_rtcp_defines.h"
+#include "rtp_utility.h"
+
+#include "typedefs.h"
+
+#include "overuse_detector.h"
+#include "remote_rate_control.h"
+#include "Bitrate.h"
+
+namespace webrtc {
+class ReceiverFEC;
+class ModuleRtpRtcpImpl;
+class CriticalSectionWrapper;
+
+class RTPReceiverVideo
+{
+public:
+ RTPReceiverVideo();
+ RTPReceiverVideo(const WebRtc_Word32 id, ModuleRtpRtcpImpl* owner);
+
+ virtual ~RTPReceiverVideo();
+
+ virtual void ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 Init();
+
+ WebRtc_Word32 RegisterIncomingVideoCallback(
+ RtpVideoFeedback* incomingMessagesCallback);
+
+ void UpdateBandwidthManagement(const WebRtc_UWord32 bitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs);
+
+ ModuleRTPUtility::Payload* RegisterReceiveVideoPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 maxRate);
+
+ WebRtc_Word32 ParseVideoCodecSpecific(
+ WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength,
+ const RtpVideoCodecTypes videoType,
+ const bool isRED,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 incomingRtpPacketSize,
+ const WebRtc_Word64 nowMS);
+
+ virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
+ WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength);
+
+ void SetPacketOverHead(WebRtc_UWord16 packetOverHead);
+
+protected:
+ void ResetOverUseDetector();
+
+ WebRtc_UWord16 EstimateBandwidth( const WebRtc_UWord16 bufferLength);
+
+ virtual WebRtc_Word32 CallbackOfReceivedPayloadData(
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader) = 0;
+
+ virtual WebRtc_UWord32 TimeStamp() const = 0;
+ virtual WebRtc_UWord16 SequenceNumber() const = 0;
+
+ virtual WebRtc_UWord32 PayloadTypeToPayload(
+ const WebRtc_UWord8 payloadType,
+ ModuleRTPUtility::Payload*& payload) const = 0;
+
+ virtual bool RetransmitOfOldPacket(
+ const WebRtc_UWord16 sequenceNumber,
+ const WebRtc_UWord32 rtpTimeStamp) const = 0;
+
+ virtual WebRtc_Word8 REDPayloadType() const = 0;
+
+ WebRtc_Word32 SetCodecType(const RtpVideoCodecTypes videoType,
+ WebRtcRTPHeader* rtpHeader) const;
+
+ WebRtc_Word32 ParseVideoCodecSpecificSwitch(
+ WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength,
+ const RtpVideoCodecTypes videoType);
+
+ WebRtc_Word32 ReceiveGenericCodec(WebRtcRTPHeader *rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength);
+
+ WebRtc_Word32 ReceiveVp8Codec(WebRtcRTPHeader *rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength);
+
+ WebRtc_Word32 BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
+ WebRtc_UWord8* dataBuffer) const;
+
+private:
+ WebRtc_Word32 _id;
+ ModuleRtpRtcpImpl* _rtpRtcp;
+
+ CriticalSectionWrapper* _criticalSectionFeedback;
+ RtpVideoFeedback* _cbVideoFeedback;
+
+ CriticalSectionWrapper* _criticalSectionReceiverVideo;
+
+ // bandwidth
+ bool _completeFrame;
+ WebRtc_UWord32 _packetStartTimeMs;
+ WebRtc_UWord16 _receivedBW[BW_HISTORY_SIZE];
+ WebRtc_UWord16 _estimatedBW;
+
+ // FEC
+ bool _currentFecFrameDecoded;
+ ReceiverFEC* _receiveFEC;
+
+ // BWE
+ OverUseDetector _overUseDetector;
+ BitRateStats _videoBitRate;
+ WebRtc_Word64 _lastBitRateChange;
+ WebRtc_UWord16 _packetOverHead;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi
new file mode 100644
index 0000000..48fc05f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi
@@ -0,0 +1,102 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'rtp_rtcp',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ },
+ 'sources': [
+ # Common
+ '../interface/rtp_rtcp.h',
+ '../interface/rtp_rtcp_defines.h',
+ 'bitrate.cc',
+ 'Bitrate.h',
+ 'rtp_rtcp_config.h',
+ 'rtp_rtcp_impl.cc',
+ 'rtp_rtcp_impl.h',
+ 'rtcp_receiver.cc',
+ 'rtcp_receiver.h',
+ 'rtcp_receiver_help.cc',
+ 'rtcp_receiver_help.h',
+ 'rtcp_sender.cc',
+ 'rtcp_sender.h',
+ 'rtcp_utility.cc',
+ 'rtcp_utility.h',
+ 'rtp_header_extension.cc',
+ 'rtp_header_extension.h',
+ 'rtp_receiver.cc',
+ 'rtp_receiver.h',
+ 'rtp_sender.cc',
+ 'rtp_sender.h',
+ 'rtp_utility.cc',
+ 'rtp_utility.h',
+ 'ssrc_database.cc',
+ 'ssrc_database.h',
+ 'tmmbr_help.cc',
+ 'tmmbr_help.h',
+ # Audio Files
+ 'dtmf_queue.cc',
+ 'dtmf_queue.h',
+ 'rtp_receiver_audio.cc',
+ 'rtp_receiver_audio.h',
+ 'rtp_sender_audio.cc',
+ 'rtp_sender_audio.h',
+ # Video Files
+ 'bandwidth_management.cc',
+ 'bandwidth_management.h',
+ 'bwe_defines.h',
+ 'fec_private_tables.h',
+ 'forward_error_correction.cc',
+ 'forward_error_correction.h',
+ 'forward_error_correction_internal.cc',
+ 'forward_error_correction_internal.h',
+ 'overuse_detector.cc',
+ 'overuse_detector.h',
+ 'remote_rate_control.cc',
+ 'remote_rate_control.h',
+ 'rtp_packet_history.cc',
+ 'rtp_packet_history.h',
+ 'rtp_receiver_video.cc',
+ 'rtp_receiver_video.h',
+ 'rtp_sender_video.cc',
+ 'rtp_sender_video.h',
+ 'receiver_fec.cc',
+ 'receiver_fec.h',
+ 'video_codec_information.h',
+ 'rtp_format_vp8.cc',
+ 'rtp_format_vp8.h',
+ 'transmission_bucket.cc',
+ 'transmission_bucket.h',
+ 'vp8_partition_aggregator.cc',
+ 'vp8_partition_aggregator.h',
+ # Mocks
+ '../mocks/mock_rtp_rtcp.h',
+ ], # source
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h
new file mode 100644
index 0000000..a0eed88
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+
+// Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
+namespace webrtc {
+enum { kRtpRtcpMaxIdleTimeProcess = 5,
+ kRtpRtcpBitrateProcessTimeMs = 10,
+ kRtpRtcpPacketTimeoutProcessTimeMs = 100 };
+
+enum { NACK_PACKETS_MAX_SIZE = 256 }; // in packets
+enum { NACK_BYTECOUNT_SIZE = 60}; // size of our NACK history
+
+enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
+enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
+enum { RTCP_SEND_BEFORE_KEY_FRAME_MS= 100 };
+enum { RTCP_MAX_REPORT_BLOCKS = 31}; // RFC 3550 page 37
+enum { RTCP_MIN_FRAME_LENGTH_MS = 17};
+enum { kRtcpAppCode_DATA_SIZE = 32*4}; // multiple of 4, this is not a limitation of the size
+enum { RTCP_RPSI_DATA_SIZE = 30};
+enum { RTCP_NUMBER_OF_SR = 60 };
+
+enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
+enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };// RFC
+
+enum { BW_HISTORY_SIZE = 35};
+
+#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
+#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
+
+enum { DTMF_OUTBAND_MAX = 20};
+
+enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
+enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
+enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
+} // namespace webrtc
+
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
new file mode 100644
index 0000000..08da8f8
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -0,0 +1,2971 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_types.h"
+#include "rtp_rtcp_impl.h"
+#include "trace.h"
+
+#ifdef MATLAB
+#include "../test/BWEStandAlone/MatlabPlot.h"
+extern MatlabEngine eng; // global variable defined elsewhere
+#endif
+
+#include <string.h> //memcpy
+#include <cassert> //assert
+
+// local for this file
+namespace {
+
+const float FracMS = 4.294967296E6f;
+
+} // namepace
+
+#ifdef _WIN32
+// disable warning C4355: 'this' : used in base member initializer list
+#pragma warning(disable : 4355)
+#endif
+
+namespace webrtc {
+
+const WebRtc_UWord16 kDefaultRtt = 200;
+
+RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
+ bool audio) {
+ if(audio) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(audio)");
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(video)");
+ }
+ // ModuleRTPUtility::GetSystemClock() creates a new instance of a system
+ // clock implementation. The OwnsClock() function informs the module that
+ // it is responsible for deleting the instance.
+ ModuleRtpRtcpImpl* rtp_rtcp_instance = new ModuleRtpRtcpImpl(id,
+ audio, ModuleRTPUtility::GetSystemClock());
+ rtp_rtcp_instance->OwnsClock();
+ return rtp_rtcp_instance;
+}
+
+RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock) {
+ if (audio) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ id,
+ "CreateRtpRtcp(audio)");
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ id,
+ "CreateRtpRtcp(video)");
+ }
+ return new ModuleRtpRtcpImpl(id, audio, clock);
+}
+
+void RtpRtcp::DestroyRtpRtcp(RtpRtcp* module) {
+ if (module) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ static_cast<ModuleRtpRtcpImpl*>(module)->Id(),
+ "DestroyRtpRtcp()");
+ delete static_cast<ModuleRtpRtcpImpl*>(module);
+ }
+}
+
+ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock):
+ TMMBRHelp(audio),
+ _rtpSender(id, audio, clock),
+ _rtpReceiver(id, audio, clock, this),
+ _rtcpSender(id, audio, clock, this),
+ _rtcpReceiver(id, clock, this),
+ _owns_clock(false),
+ _clock(*clock),
+ _id(id),
+ _audio(audio),
+ _collisionDetected(false),
+ _lastProcessTime(clock->GetTimeInMS()),
+ _lastBitrateProcessTime(clock->GetTimeInMS()),
+ _lastPacketTimeoutProcessTime(clock->GetTimeInMS()),
+ _packetOverHead(28), // IPV4 UDP
+ _criticalSectionModulePtrs(CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionModulePtrsFeedback(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _defaultModule(NULL),
+ _audioModule(NULL),
+ _videoModule(NULL),
+ _deadOrAliveActive(false),
+ _deadOrAliveTimeoutMS(0),
+ _deadOrAliveLastTimer(0),
+ _bandwidthManagement(id),
+ _receivedNTPsecsAudio(0),
+ _receivedNTPfracAudio(0),
+ _RTCPArrivalTimeSecsAudio(0),
+ _RTCPArrivalTimeFracAudio(0),
+ _nackMethod(kNackOff),
+ _nackLastTimeSent(0),
+ _nackLastSeqNumberSent(0),
+ _simulcast(false),
+ _keyFrameReqMethod(kKeyFrameReqFirRtp)
+#ifdef MATLAB
+ , _plot1(NULL)
+#endif
+{
+ _sendVideoCodec.codecType = kVideoCodecUnknown;
+ // make sure that RTCP objects are aware of our SSRC
+ WebRtc_UWord32 SSRC = _rtpSender.SSRC();
+ _rtcpSender.SetSSRC(SSRC);
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+}
+
+ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
+
+ // make sure to unregister this module from other modules
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+
+ if (defaultInstance) {
+ // deregister for the default module
+ // will go in to the child modules and remove it self
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ _childModules.erase(it);
+ if (module) {
+ module->DeRegisterDefaultModule();
+ }
+ it = _childModules.begin();
+ }
+ } else {
+ // deregister for the child modules
+ // will go in to the default and remove it self
+ DeRegisterDefaultModule();
+ }
+
+ if (_audio) {
+ DeRegisterVideoModule();
+ } else {
+ DeRegisterSyncModule();
+ }
+
+#ifdef MATLAB
+ if (_plot1) {
+ eng.DeletePlot(_plot1);
+ _plot1 = NULL;
+ }
+#endif
+
+ delete _criticalSectionModulePtrs;
+ delete _criticalSectionModulePtrsFeedback;
+ if (_owns_clock) {
+ delete &_clock;
+ }
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::ChangeUniqueId(const WebRtc_Word32 id) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "ChangeUniqueId(new id:%d)", id);
+
+ _id = id;
+
+ _rtpReceiver.ChangeUniqueId(id);
+ _rtcpReceiver.ChangeUniqueId(id);
+ _rtpSender.ChangeUniqueId(id);
+ _rtcpSender.ChangeUniqueId(id);
+ return 0;
+}
+
+// default encoder that we need to multiplex out
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterDefaultModule(RtpRtcp* module) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterDefaultModule(module:0x%x)", module);
+
+ if (module == NULL) {
+ return -1;
+ }
+ if (module == this) {
+ WEBRTC_TRACE(kTraceError,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterDefaultModule can't register self as default");
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ if (_defaultModule) {
+ _defaultModule->DeRegisterChildModule(this);
+ }
+ _defaultModule = (ModuleRtpRtcpImpl*)module;
+ _defaultModule->RegisterChildModule(this);
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterDefaultModule() {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterDefaultModule()");
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ _defaultModule->DeRegisterChildModule(this);
+ _defaultModule = NULL;
+ }
+ return 0;
+}
+
+bool ModuleRtpRtcpImpl::DefaultModuleRegistered() {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DefaultModuleRegistered()");
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ return true;
+ }
+ return false;
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::NumberChildModules() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "NumberChildModules");
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
+ // we use two locks for protecting _childModules one
+ // (_criticalSectionModulePtrsFeedback) for incoming messages
+ // (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs for
+ // all outgoing messages sending packets etc
+
+ return _childModules.size();
+}
+
+void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterChildModule(module:0x%x)",
+ module);
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
+ // we use two locks for protecting _childModules one
+ // (_criticalSectionModulePtrsFeedback) for incoming
+ // messages (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs
+ // for all outgoing messages sending packets etc
+ _childModules.push_back((ModuleRtpRtcpImpl*)module);
+}
+
+void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* removeModule) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterChildModule(module:0x%x)", removeModule);
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module == removeModule) {
+ _childModules.erase(it);
+ return;
+ }
+ it++;
+ }
+}
+
+// Lip-sync between voice-video engine,
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSyncModule(RtpRtcp* audioModule) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterSyncModule(module:0x%x)",
+ audioModule);
+
+ if (audioModule == NULL) {
+ return -1;
+ }
+ if (_audio) {
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ _audioModule = (ModuleRtpRtcpImpl*)audioModule;
+ return _audioModule->RegisterVideoModule(this);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSyncModule() {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterSyncModule()");
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_audioModule) {
+ ModuleRtpRtcpImpl* audioModule = _audioModule;
+ _audioModule = NULL;
+ _receivedNTPsecsAudio = 0;
+ _receivedNTPfracAudio = 0;
+ _RTCPArrivalTimeSecsAudio = 0;
+ _RTCPArrivalTimeFracAudio = 0;
+ audioModule->DeRegisterVideoModule();
+ }
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterVideoModule(RtpRtcp* videoModule) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterVideoModule(module:0x%x)",
+ videoModule);
+
+ if (videoModule == NULL) {
+ return -1;
+ }
+ if (!_audio) {
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ _videoModule = (ModuleRtpRtcpImpl*)videoModule;
+ return 0;
+}
+
+void ModuleRtpRtcpImpl::DeRegisterVideoModule() {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterVideoModule()");
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_videoModule) {
+ ModuleRtpRtcpImpl* videoModule = _videoModule;
+ _videoModule = NULL;
+ videoModule->DeRegisterSyncModule();
+ }
+}
+
+// returns the number of milliseconds until the module want a worker thread
+// to call Process
+WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() {
+ const WebRtc_UWord32 now = _clock.GetTimeInMS();
+ return kRtpRtcpMaxIdleTimeProcess - (now - _lastProcessTime);
+}
+
+// Process any pending tasks such as timeouts
+// non time critical events
+WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
+ const WebRtc_UWord32 now = _clock.GetTimeInMS();
+ _lastProcessTime = now;
+
+ _rtpSender.ProcessSendToNetwork();
+
+ if (now >= _lastPacketTimeoutProcessTime +
+ kRtpRtcpPacketTimeoutProcessTimeMs) {
+ _rtpReceiver.PacketTimeout();
+ _rtcpReceiver.PacketTimeout();
+ _lastPacketTimeoutProcessTime = now;
+ }
+
+ if (now >= _lastBitrateProcessTime + kRtpRtcpBitrateProcessTimeMs) {
+ _rtpSender.ProcessBitrate();
+ _rtpReceiver.ProcessBitrate();
+ _lastBitrateProcessTime = now;
+ }
+
+ ProcessDeadOrAliveTimer();
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (!defaultInstance && _rtcpSender.TimeToSendRTCPReport()) {
+ WebRtc_UWord16 max_rtt = 0;
+ if (_rtcpSender.Sending()) {
+ std::vector<RTCPReportBlock> receive_blocks;
+ _rtcpReceiver.StatisticsReceived(&receive_blocks);
+ for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
+ it != receive_blocks.end(); ++it) {
+ WebRtc_UWord16 rtt = 0;
+ _rtcpReceiver.RTT(it->remoteSSRC, &max_rtt, NULL, NULL, NULL);
+ max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
+ }
+ } else {
+ // We're only receiving, i.e. this module doesn't have its own RTT
+ // estimate. Use the RTT set by a sending channel using the same default
+ // module.
+ max_rtt = _rtcpReceiver.RTT();
+ }
+ if (max_rtt == 0) {
+ // No valid estimate available, i.e. no sending channel using the same
+ // default module or no RTCP received yet.
+ max_rtt = kDefaultRtt;
+ }
+ if (REMB() && _rtcpSender.ValidBitrateEstimate()) {
+ unsigned int target_bitrate =
+ _rtcpSender.CalculateNewTargetBitrate(max_rtt);
+ _rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
+ } else if (TMMBR()) {
+ _rtcpSender.CalculateNewTargetBitrate(max_rtt);
+ }
+ _rtcpSender.SendRTCP(kRtcpReport);
+ }
+
+ if (_rtpSender.RTPKeepalive()) {
+ // check time to send RTP keep alive
+ if (_rtpSender.TimeToSendRTPKeepalive()) {
+ _rtpSender.SendRTPKeepalivePacket();
+ }
+ }
+
+ if (UpdateRTCPReceiveInformationTimers()) {
+ // a receiver has timed out
+ UpdateTMMBR();
+ }
+ return 0;
+}
+
+/**
+* Receiver
+*/
+
+WebRtc_Word32 ModuleRtpRtcpImpl::InitReceiver() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitReceiver()");
+
+ _packetOverHead = 28; // default is IPV4 UDP
+ _receivedNTPsecsAudio = 0;
+ _receivedNTPfracAudio = 0;
+ _RTCPArrivalTimeSecsAudio = 0;
+ _RTCPArrivalTimeFracAudio = 0;
+
+ WebRtc_Word32 ret = _rtpReceiver.Init();
+ if (ret < 0) {
+ return ret;
+ }
+ _rtpReceiver.SetPacketOverHead(_packetOverHead);
+ return ret;
+}
+
+void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
+ if (_deadOrAliveActive) {
+ const WebRtc_UWord32 now = _clock.GetTimeInMS();
+ if (now > _deadOrAliveTimeoutMS + _deadOrAliveLastTimer) {
+ // RTCP is alive if we have received a report the last 12 seconds
+ _deadOrAliveLastTimer += _deadOrAliveTimeoutMS;
+
+ bool RTCPalive = false;
+ if (_rtcpReceiver.LastReceived() + 12000 > now) {
+ RTCPalive = true;
+ }
+ _rtpReceiver.ProcessDeadOrAlive(RTCPalive, now);
+ }
+ }
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
+ const bool enable,
+ const WebRtc_UWord8 sampleTimeSeconds) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetPeriodicDeadOrAliveStatus(enable, %d)",
+ sampleTimeSeconds);
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetPeriodicDeadOrAliveStatus(disable)");
+ }
+ if (sampleTimeSeconds == 0) {
+ return -1;
+ }
+ _deadOrAliveActive = enable;
+ _deadOrAliveTimeoutMS = sampleTimeSeconds * 1000;
+ // trigger the first after one period
+ _deadOrAliveLastTimer = _clock.GetTimeInMS();
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::PeriodicDeadOrAliveStatus(
+ bool& enable,
+ WebRtc_UWord8& sampleTimeSeconds) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "PeriodicDeadOrAliveStatus()");
+
+ enable = _deadOrAliveActive;
+ sampleTimeSeconds = (WebRtc_UWord8)(_deadOrAliveTimeoutMS / 1000);
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetPacketTimeout(
+ const WebRtc_UWord32 RTPtimeoutMS,
+ const WebRtc_UWord32 RTCPtimeoutMS) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetPacketTimeout(%u,%u)",
+ RTPtimeoutMS,
+ RTCPtimeoutMS);
+
+ if (_rtpReceiver.SetPacketTimeout(RTPtimeoutMS) == 0) {
+ return _rtcpReceiver.SetPacketTimeout(RTCPtimeoutMS);
+ }
+ return -1;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
+ const CodecInst& voiceCodec) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterReceivePayload(voiceCodec)");
+
+ return _rtpReceiver.RegisterReceivePayload(
+ voiceCodec.plname,
+ voiceCodec.pltype,
+ voiceCodec.plfreq,
+ voiceCodec.channels,
+ (voiceCodec.rate < 0) ? 0 : voiceCodec.rate);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
+ const VideoCodec& videoCodec) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterReceivePayload(videoCodec)");
+
+ return _rtpReceiver.RegisterReceivePayload(videoCodec.plName,
+ videoCodec.plType,
+ 90000,
+ 0,
+ videoCodec.maxBitrate);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
+ const CodecInst& voiceCodec,
+ WebRtc_Word8* plType) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "ReceivePayloadType(voiceCodec)");
+
+ return _rtpReceiver.ReceivePayloadType(
+ voiceCodec.plname,
+ voiceCodec.plfreq,
+ voiceCodec.channels,
+ (voiceCodec.rate < 0) ? 0 : voiceCodec.rate,
+ plType);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
+ const VideoCodec& videoCodec,
+ WebRtc_Word8* plType) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "ReceivePayloadType(videoCodec)");
+
+ return _rtpReceiver.ReceivePayloadType(videoCodec.plName,
+ 90000,
+ 0,
+ videoCodec.maxBitrate,
+ plType);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload(
+ const WebRtc_Word8 payloadType) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterReceivePayload(%d)",
+ payloadType);
+
+ return _rtpReceiver.DeRegisterReceivePayload(payloadType);
+}
+
+// get the currently configured SSRC filter
+WebRtc_Word32 ModuleRtpRtcpImpl::SSRCFilter(WebRtc_UWord32& allowedSSRC) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SSRCFilter()");
+
+ return _rtpReceiver.SSRCFilter(allowedSSRC);
+}
+
+// set a SSRC to be used as a filter for incoming RTP streams
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRCFilter(
+ const bool enable,
+ const WebRtc_UWord32 allowedSSRC) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSSRCFilter(enable, 0x%x)",
+ allowedSSRC);
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSSRCFilter(disable)");
+ }
+
+ return _rtpReceiver.SetSSRCFilter(enable, allowedSSRC);
+}
+
+// Get last received remote timestamp
+WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteTimestamp() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteTimestamp()");
+
+ return _rtpReceiver.TimeStamp();
+}
+
+// Get the current estimated remote timestamp
+WebRtc_Word32 ModuleRtpRtcpImpl::EstimatedRemoteTimeStamp(
+ WebRtc_UWord32& timestamp) const {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "EstimatedRemoteTimeStamp()");
+
+ return _rtpReceiver.EstimatedRemoteTimeStamp(timestamp);
+}
+
+// Get incoming SSRC
+WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteSSRC() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteSSRC()");
+
+ return _rtpReceiver.SSRC();
+}
+
+// Get remote CSRC
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCSRCs(
+ WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteCSRCs()");
+
+ return _rtpReceiver.CSRCs(arrOfCSRC);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXSendStatus(
+ const bool enable,
+ const bool setSSRC,
+ const WebRtc_UWord32 SSRC) {
+ _rtpSender.SetRTXStatus(enable, setSSRC, SSRC);
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RTXSendStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const {
+ _rtpSender.RTXStatus(enable, SSRC);
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXReceiveStatus(
+ const bool enable,
+ const WebRtc_UWord32 SSRC) {
+ _rtpReceiver.SetRTXStatus(enable, SSRC);
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const {
+ _rtpReceiver.RTXStatus(enable, SSRC);
+ return 0;
+}
+
+// called by the network module when we receive a packet
+WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket(
+ const WebRtc_UWord8* incomingPacket,
+ const WebRtc_UWord16 incomingPacketLength) {
+ WEBRTC_TRACE(kTraceStream,
+ kTraceRtpRtcp,
+ _id,
+ "IncomingPacket(packetLength:%u)",
+ incomingPacketLength);
+
+ // minimum RTP is 12 bytes
+ // minimum RTCP is 8 bytes (RTCP BYE)
+ if (incomingPacketLength < 8 || incomingPacket == NULL) {
+ WEBRTC_TRACE(kTraceDebug,
+ kTraceRtpRtcp,
+ _id,
+ "IncomingPacket invalid buffer or length");
+ return -1;
+ }
+ // check RTP version
+ const WebRtc_UWord8 version = incomingPacket[0] >> 6 ;
+ if (version != 2) {
+ WEBRTC_TRACE(kTraceDebug,
+ kTraceRtpRtcp,
+ _id,
+ "IncomingPacket invalid RTP version");
+ return -1;
+ }
+
+ ModuleRTPUtility::RTPHeaderParser rtpParser(incomingPacket,
+ incomingPacketLength);
+
+ if (rtpParser.RTCP()) {
+ // Allow receive of non-compound RTCP packets.
+ RTCPUtility::RTCPParserV2 rtcpParser(incomingPacket,
+ incomingPacketLength,
+ true);
+
+ const bool validRTCPHeader = rtcpParser.IsValid();
+ if (!validRTCPHeader) {
+ WEBRTC_TRACE(kTraceDebug,
+ kTraceRtpRtcp,
+ _id,
+ "IncomingPacket invalid RTCP packet");
+ return -1;
+ }
+ RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
+ WebRtc_Word32 retVal = _rtcpReceiver.IncomingRTCPPacket(
+ rtcpPacketInformation,
+ &rtcpParser);
+ if (retVal == 0) {
+ _rtcpReceiver.TriggerCallbacksFromRTCPPacket(rtcpPacketInformation);
+ }
+ return retVal;
+
+ } else {
+ WebRtcRTPHeader rtpHeader;
+ memset(&rtpHeader, 0, sizeof(rtpHeader));
+
+ RtpHeaderExtensionMap map;
+ _rtpReceiver.GetHeaderExtensionMapCopy(&map);
+
+ const bool validRTPHeader = rtpParser.Parse(rtpHeader, &map);
+ if (!validRTPHeader) {
+ WEBRTC_TRACE(kTraceDebug,
+ kTraceRtpRtcp,
+ _id,
+ "IncomingPacket invalid RTP header");
+ return -1;
+ }
+ return _rtpReceiver.IncomingRTPPacket(&rtpHeader,
+ incomingPacket,
+ incomingPacketLength);
+ }
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::IncomingAudioNTP(
+ const WebRtc_UWord32 audioReceivedNTPsecs,
+ const WebRtc_UWord32 audioReceivedNTPfrac,
+ const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
+ const WebRtc_UWord32 audioRTCPArrivalTimeFrac) {
+ _receivedNTPsecsAudio = audioReceivedNTPsecs;
+ _receivedNTPfracAudio = audioReceivedNTPfrac;
+ _RTCPArrivalTimeSecsAudio = audioRTCPArrivalTimeSecs;
+ _RTCPArrivalTimeFracAudio = audioRTCPArrivalTimeFrac;
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingDataCallback(
+ RtpData* incomingDataCallback) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterIncomingDataCallback(incomingDataCallback:0x%x)",
+ incomingDataCallback);
+
+ return _rtpReceiver.RegisterIncomingDataCallback(incomingDataCallback);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingRTPCallback(
+ RtpFeedback* incomingMessagesCallback) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterIncomingRTPCallback(incomingMessagesCallback:0x%x)",
+ incomingMessagesCallback);
+
+ return _rtpReceiver.RegisterIncomingRTPCallback(incomingMessagesCallback);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingRTCPCallback(
+ RtcpFeedback* incomingMessagesCallback) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterIncomingRTCPCallback(incomingMessagesCallback:0x%x)",
+ incomingMessagesCallback);
+
+ return _rtcpReceiver.RegisterIncomingRTCPCallback(incomingMessagesCallback);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingVideoCallback(
+ RtpVideoFeedback* incomingMessagesCallback) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterIncomingVideoCallback(incomingMessagesCallback:0x%x)",
+ incomingMessagesCallback);
+
+ if (_rtcpReceiver.RegisterIncomingVideoCallback(incomingMessagesCallback)
+ == 0) {
+ return _rtpReceiver.RegisterIncomingVideoCallback(
+ incomingMessagesCallback);
+ }
+ return -1;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterAudioCallback(
+ RtpAudioFeedback* messagesCallback) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterAudioCallback(messagesCallback:0x%x)",
+ messagesCallback);
+
+ if (_rtpSender.RegisterAudioCallback(messagesCallback) == 0) {
+ return _rtpReceiver.RegisterIncomingAudioCallback(messagesCallback);
+ }
+ return -1;
+}
+
+/**
+* Sender
+*/
+
+WebRtc_Word32 ModuleRtpRtcpImpl::InitSender() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitSender()");
+
+ _collisionDetected = false;
+
+ // if we are already receiving inform our sender to avoid collision
+ if (_rtpSender.Init(_rtpReceiver.SSRC()) != 0) {
+ return -1;
+ }
+ WebRtc_Word32 retVal = _rtcpSender.Init();
+
+ // make sure that RTCP objects are aware of our SSRC
+ // (it could have changed due to collision)
+ WebRtc_UWord32 SSRC = _rtpSender.SSRC();
+ _rtcpReceiver.SetSSRC(SSRC);
+ _rtcpSender.SetSSRC(SSRC);
+ return retVal;
+}
+
+bool ModuleRtpRtcpImpl::RTPKeepalive() const {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "RTPKeepalive()");
+
+ return _rtpSender.RTPKeepalive();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RTPKeepaliveStatus(
+ bool* enable,
+ WebRtc_Word8* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTPKeepaliveStatus()");
+
+ return _rtpSender.RTPKeepaliveStatus(enable,
+ unknownPayloadType,
+ deltaTransmitTimeMS);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPKeepaliveStatus(
+ bool enable,
+ WebRtc_Word8 unknownPayloadType,
+ WebRtc_UWord16 deltaTransmitTimeMS) {
+ if (enable) {
+ WEBRTC_TRACE(
+ kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetRTPKeepaliveStatus(true, plType:%d deltaTransmitTimeMS:%u)",
+ unknownPayloadType,
+ deltaTransmitTimeMS);
+
+ // check the transmit keepalive delta time [1,60]
+ if (deltaTransmitTimeMS < 1000 || deltaTransmitTimeMS > 60000) {
+ WEBRTC_TRACE(kTraceError,
+ kTraceRtpRtcp,
+ _id,
+ "\tinvalid deltaTransmitTimeSeconds (%d)",
+ deltaTransmitTimeMS);
+ return -1;
+ }
+
+ // check the payload time [0,127]
+ if (unknownPayloadType < 0) {
+ WEBRTC_TRACE(kTraceError,
+ kTraceRtpRtcp,
+ _id,
+ "\tinvalid unknownPayloadType (%d)",
+ unknownPayloadType);
+ return -1;
+ }
+ // enable RTP keepalive mechanism
+ return _rtpSender.EnableRTPKeepalive(unknownPayloadType,
+ deltaTransmitTimeMS);
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetRTPKeepaliveStatus(disable)");
+ return _rtpSender.DisableRTPKeepalive();
+ }
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload(
+ const CodecInst& voiceCodec) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterSendPayload(plName:%s plType:%d frequency:%u)",
+ voiceCodec.plname,
+ voiceCodec.pltype,
+ voiceCodec.plfreq);
+
+ return _rtpSender.RegisterPayload(
+ voiceCodec.plname,
+ voiceCodec.pltype,
+ voiceCodec.plfreq,
+ voiceCodec.channels,
+ (voiceCodec.rate < 0) ? 0 : voiceCodec.rate);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload(
+ const VideoCodec& videoCodec) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterSendPayload(plName:%s plType:%d)",
+ videoCodec.plName,
+ videoCodec.plType);
+
+ _sendVideoCodec = videoCodec;
+ _simulcast = (videoCodec.numberOfSimulcastStreams > 1) ? true : false;
+ return _rtpSender.RegisterPayload(videoCodec.plName,
+ videoCodec.plType,
+ 90000,
+ 0,
+ videoCodec.maxBitrate);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSendPayload(
+ const WebRtc_Word8 payloadType) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "DeRegisterSendPayload(%d)", payloadType);
+
+ return _rtpSender.DeRegisterSendPayload(payloadType);
+}
+
+WebRtc_Word8 ModuleRtpRtcpImpl::SendPayloadType() const {
+ return _rtpSender.SendPayloadType();
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::StartTimestamp() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "StartTimestamp()");
+
+ return _rtpSender.StartTimestamp();
+}
+
+// configure start timestamp, default is a random number
+WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp(
+ const WebRtc_UWord32 timestamp) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetStartTimestamp(%d)",
+ timestamp);
+
+ return _rtpSender.SetStartTimestamp(timestamp, true);
+}
+
+WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SequenceNumber()");
+
+ return _rtpSender.SequenceNumber();
+}
+
+// Set SequenceNumber, default is a random number
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber(
+ const WebRtc_UWord16 seqNum) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSequenceNumber(%d)",
+ seqNum);
+
+ return _rtpSender.SetSequenceNumber(seqNum);
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SSRC()");
+
+ return _rtpSender.SSRC();
+}
+
+// configure SSRC, default is a random number
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetSSRC(%d)", ssrc);
+
+ if (_rtpSender.SetSSRC(ssrc) == 0) {
+ _rtcpReceiver.SetSSRC(ssrc);
+ _rtcpSender.SetSSRC(ssrc);
+ return 0;
+ }
+ return -1;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
+ _rtcpSender.SetCSRCStatus(include);
+ return _rtpSender.SetCSRCStatus(include);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs(
+ WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "CSRCs()");
+
+ return _rtpSender.CSRCs(arrOfCSRC);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs(
+ const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetCSRCs(arrLength:%d)",
+ arrLength);
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+
+ if (defaultInstance) {
+ // for default we need to update all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ module->SetCSRCs(arrOfCSRC, arrLength);
+ }
+ it++;
+ }
+ return 0;
+
+ } else {
+ for (int i = 0; i < arrLength; i++) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "\tidx:%d CSRC:%u", i,
+ arrOfCSRC[i]);
+ }
+ _rtcpSender.SetCSRCs(arrOfCSRC, arrLength);
+ return _rtpSender.SetCSRCs(arrOfCSRC, arrLength);
+ }
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "PacketCountSent()");
+
+ return _rtpSender.Packets();
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::ByteCountSent() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ByteCountSent()");
+
+ return _rtpSender.Bytes();
+}
+
+int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "CurrentSendFrequencyHz()");
+
+ return _rtpSender.SendPayloadFrequency();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
+ if (sending) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetSendingStatus(sending)");
+ } else {
+ if (_rtpSender.RTPKeepalive()) {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceRtpRtcp,
+ _id,
+ "Can't SetSendingStatus(stopped) when RTP Keepalive is active");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetSendingStatus(stopped)");
+ }
+ if (_rtcpSender.Sending() != sending) {
+ // sends RTCP BYE when going from true to false
+ if (_rtcpSender.SetSendingStatus(sending) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "Failed to send RTCP BYE");
+ }
+
+ _collisionDetected = false;
+
+ // generate a new timeStamp if true and not configured via API
+ // generate a new SSRC for the next "call" if false
+ _rtpSender.SetSendingStatus(sending);
+
+ // make sure that RTCP objects are aware of our SSRC (it could have changed
+ // due to collision)
+ WebRtc_UWord32 SSRC = _rtpSender.SSRC();
+ _rtcpReceiver.SetSSRC(SSRC);
+ _rtcpSender.SetSSRC(SSRC);
+ return 0;
+ }
+ return 0;
+}
+
+bool ModuleRtpRtcpImpl::Sending() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "Sending()");
+
+ return _rtcpSender.Sending();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
+ if (sending) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetSendingMediaStatus(sending)");
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetSendingMediaStatus(stopped)");
+ }
+ _rtpSender.SetSendingMediaStatus(sending);
+ return 0;
+}
+
+bool ModuleRtpRtcpImpl::SendingMedia() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "Sending()");
+
+ const bool haveChildModules(_childModules.empty() ? false : true);
+ if (!haveChildModules) {
+ return _rtpSender.SendingMedia();
+ }
+
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ std::list<ModuleRtpRtcpImpl*>::const_iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RTPSender& rtpSender = (*it)->_rtpSender;
+ if (rtpSender.SendingMedia()) {
+ return true;
+ }
+ it++;
+ }
+ return false;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendTransport(
+ Transport* outgoingTransport) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RegisterSendTransport(0x%x)", outgoingTransport);
+
+ if (_rtpSender.RegisterSendTransport(outgoingTransport) == 0) {
+ return _rtcpSender.RegisterSendTransport(outgoingTransport);
+ }
+ return -1;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SendOutgoingData(
+ FrameType frameType,
+ WebRtc_Word8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ const RTPVideoHeader* rtpVideoHdr) {
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceRtpRtcp,
+ _id,
+ "SendOutgoingData(frameType:%d payloadType:%d timeStamp:%u size:%u)",
+ frameType, payloadType, timeStamp, payloadSize);
+
+ const bool haveChildModules(_childModules.empty() ? false : true);
+ if (!haveChildModules) {
+ // Don't sent RTCP from default module
+ if (_rtcpSender.TimeToSendRTCPReport(kVideoFrameKey == frameType)) {
+ _rtcpSender.SendRTCP(kRtcpReport);
+ }
+ return _rtpSender.SendOutgoingData(frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ NULL,
+ &(rtpVideoHdr->codecHeader));
+ }
+ WebRtc_Word32 retVal = -1;
+ if (_simulcast) {
+ if (rtpVideoHdr == NULL) {
+ return -1;
+ }
+ int idx = 0;
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ for (; idx < rtpVideoHdr->simulcastIdx; idx++) {
+ it++;
+ if (it == _childModules.end()) {
+ return -1;
+ }
+ }
+ RTPSender& rtpSender = (*it)->_rtpSender;
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SendOutgoingData(SimulcastIdx:%u size:%u, ssrc:0x%x)",
+ idx, payloadSize, rtpSender.SSRC());
+ return rtpSender.SendOutgoingData(frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ NULL,
+ &(rtpVideoHdr->codecHeader));
+ } else {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ // TODO(pwestin) remove codecInfo from SendOutgoingData
+ VideoCodecInformation* codecInfo = NULL;
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ if (it != _childModules.end()) {
+ RTPSender& rtpSender = (*it)->_rtpSender;
+ retVal = rtpSender.SendOutgoingData(frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ NULL,
+ &(rtpVideoHdr->codecHeader));
+
+ it++;
+ }
+
+ // send to all remaining "child" modules
+ while (it != _childModules.end()) {
+ RTPSender& rtpSender = (*it)->_rtpSender;
+ retVal = rtpSender.SendOutgoingData(frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ codecInfo,
+ &(rtpVideoHdr->codecHeader));
+
+ it++;
+ }
+ }
+ return retVal;
+}
+
+WebRtc_UWord16 ModuleRtpRtcpImpl::MaxPayloadLength() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "MaxPayloadLength()");
+
+ return _rtpSender.MaxPayloadLength();
+}
+
+WebRtc_UWord16 ModuleRtpRtcpImpl::MaxDataPayloadLength() const {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "MaxDataPayloadLength()");
+
+ WebRtc_UWord16 minDataPayloadLength = IP_PACKET_SIZE - 28; // Assuming IP/UDP
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ // for default we need to update all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ std::list<ModuleRtpRtcpImpl*>::const_iterator it =
+ _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ WebRtc_UWord16 dataPayloadLength =
+ module->MaxDataPayloadLength();
+ if (dataPayloadLength < minDataPayloadLength) {
+ minDataPayloadLength = dataPayloadLength;
+ }
+ }
+ it++;
+ }
+ }
+
+ WebRtc_UWord16 dataPayloadLength = _rtpSender.MaxDataPayloadLength();
+ if (dataPayloadLength < minDataPayloadLength) {
+ minDataPayloadLength = dataPayloadLength;
+ }
+ return minDataPayloadLength;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetTransportOverhead(
+ const bool TCP,
+ const bool IPV6,
+ const WebRtc_UWord8 authenticationOverhead) {
+ WEBRTC_TRACE(
+ kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetTransportOverhead(TCP:%d, IPV6:%d authenticationOverhead:%u)",
+ TCP, IPV6, authenticationOverhead);
+
+ WebRtc_UWord16 packetOverHead = 0;
+ if (IPV6) {
+ packetOverHead = 40;
+ } else {
+ packetOverHead = 20;
+ }
+ if (TCP) {
+ // TCP
+ packetOverHead += 20;
+ } else {
+ // UDP
+ packetOverHead += 8;
+ }
+ packetOverHead += authenticationOverhead;
+
+ if (packetOverHead == _packetOverHead) {
+ // ok same as before
+ return 0;
+ }
+ // calc diff
+ WebRtc_Word16 packetOverHeadDiff = packetOverHead - _packetOverHead;
+
+ // store new
+ _packetOverHead = packetOverHead;
+
+ _rtpReceiver.SetPacketOverHead(_packetOverHead);
+ WebRtc_UWord16 length = _rtpSender.MaxPayloadLength() - packetOverHeadDiff;
+ return _rtpSender.SetMaxPayloadLength(length, _packetOverHead);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetMaxTransferUnit(const WebRtc_UWord16 MTU) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetMaxTransferUnit(%u)",
+ MTU);
+
+ if (MTU > IP_PACKET_SIZE) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "Invalid in argument to SetMaxTransferUnit(%u)", MTU);
+ return -1;
+ }
+ return _rtpSender.SetMaxPayloadLength(MTU - _packetOverHead,
+ _packetOverHead);
+}
+
+/*
+* RTCP
+*/
+RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTCP()");
+
+ if (_rtcpSender.Status() != kRtcpOff) {
+ return _rtcpReceiver.Status();
+ }
+ return kRtcpOff;
+}
+
+// configure RTCP status i.e on/off
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetRTCPStatus(%d)",
+ method);
+
+ if (_rtcpSender.SetRTCPStatus(method) == 0) {
+ return _rtcpReceiver.SetRTCPStatus(method);
+ }
+ return -1;
+}
+
+// only for internal test
+WebRtc_UWord32 ModuleRtpRtcpImpl::LastSendReport(WebRtc_UWord32& lastRTCPTime) {
+ return _rtcpSender.LastSendReport(lastRTCPTime);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetCNAME(%s)", cName);
+ return _rtcpSender.SetCNAME(cName);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::CNAME(char cName[RTCP_CNAME_SIZE]) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "CNAME()");
+ return _rtcpSender.CNAME(cName);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::AddMixedCNAME(
+ const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "AddMixedCNAME(SSRC:%u)", SSRC);
+
+ return _rtcpSender.AddMixedCNAME(SSRC, cName);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "RemoveMixedCNAME(SSRC:%u)", SSRC);
+ return _rtcpSender.RemoveMixedCNAME(SSRC);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCNAME(
+ const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "RemoteCNAME(SSRC:%u)", remoteSSRC);
+
+ return _rtcpReceiver.CNAME(remoteSSRC, cName);
+}
+
+WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteSequenceNumber()");
+
+ return _rtpReceiver.SequenceNumber();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteNTP(
+ WebRtc_UWord32* receivedNTPsecs,
+ WebRtc_UWord32* receivedNTPfrac,
+ WebRtc_UWord32* RTCPArrivalTimeSecs,
+ WebRtc_UWord32* RTCPArrivalTimeFrac) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteNTP()");
+
+ return _rtcpReceiver.NTP(receivedNTPsecs,
+ receivedNTPfrac,
+ RTCPArrivalTimeSecs,
+ RTCPArrivalTimeFrac);
+}
+
+// Get RoundTripTime
+WebRtc_Word32 ModuleRtpRtcpImpl::RTT(const WebRtc_UWord32 remoteSSRC,
+ WebRtc_UWord16* RTT,
+ WebRtc_UWord16* avgRTT,
+ WebRtc_UWord16* minRTT,
+ WebRtc_UWord16* maxRTT) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTT()");
+
+ return _rtcpReceiver.RTT(remoteSSRC, RTT, avgRTT, minRTT, maxRTT);
+}
+
+// Reset RoundTripTime statistics
+WebRtc_Word32
+ModuleRtpRtcpImpl::ResetRTT(const WebRtc_UWord32 remoteSSRC) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ResetRTT(SSRC:%u)",
+ remoteSSRC);
+
+ return _rtcpReceiver.ResetRTT(remoteSSRC);
+}
+
+// Reset RTP statistics
+WebRtc_Word32
+ModuleRtpRtcpImpl::ResetStatisticsRTP() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ResetStatisticsRTP()");
+
+ return _rtpReceiver.ResetStatistics();
+}
+
+// Reset RTP data counters for the receiving side
+WebRtc_Word32 ModuleRtpRtcpImpl::ResetReceiveDataCountersRTP() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "ResetReceiveDataCountersRTP()");
+
+ return _rtpReceiver.ResetDataCounters();
+}
+
+// Reset RTP data counters for the sending side
+WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "ResetSendDataCountersRTP()");
+
+ return _rtpSender.ResetDataCounters();
+}
+
+// Force a send of an RTCP packet
+// normal SR and RR are triggered via the process function
+WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCP(WebRtc_UWord32 rtcpPacketType) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SendRTCP(0x%x)",
+ rtcpPacketType);
+
+ return _rtcpSender.SendRTCP(rtcpPacketType);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
+ const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 length) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetRTCPApplicationSpecificData(subType:%d name:0x%x)", subType,
+ name);
+
+ return _rtcpSender.SetApplicationSpecificData(subType, name, data, length);
+}
+
+/*
+* (XR) VOIP metric
+*/
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
+ const RTCPVoIPMetric* VoIPMetric) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetRTCPVoIPMetrics()");
+
+ return _rtcpSender.SetRTCPVoIPMetrics(VoIPMetric);
+}
+
+// our localy created statistics of the received RTP stream
+WebRtc_Word32 ModuleRtpRtcpImpl::StatisticsRTP(
+ WebRtc_UWord8* fraction_lost,
+ WebRtc_UWord32* cum_lost,
+ WebRtc_UWord32* ext_max,
+ WebRtc_UWord32* jitter,
+ WebRtc_UWord32* max_jitter) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "StatisticsRTP()");
+
+ WebRtc_UWord32 jitter_transmission_time_offset = 0;
+
+ WebRtc_Word32 retVal = _rtpReceiver.Statistics(
+ fraction_lost,
+ cum_lost,
+ ext_max,
+ jitter,
+ max_jitter,
+ &jitter_transmission_time_offset,
+ (_rtcpSender.Status() == kRtcpOff));
+ if (retVal == -1) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "StatisticsRTP() no statisitics availble");
+ }
+ return retVal;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DataCountersRTP(
+ WebRtc_UWord32* bytesSent,
+ WebRtc_UWord32* packetsSent,
+ WebRtc_UWord32* bytesReceived,
+ WebRtc_UWord32* packetsReceived) const {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "DataCountersRTP()");
+
+ if (bytesSent) {
+ *bytesSent = _rtpSender.Bytes();
+ }
+ if (packetsSent) {
+ *packetsSent = _rtpSender.Packets();
+ }
+ return _rtpReceiver.DataCounters(bytesReceived, packetsReceived);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
+ WebRtc_UWord8* fraction_lost,
+ WebRtc_UWord32* cum_lost,
+ WebRtc_UWord32* ext_max,
+ WebRtc_UWord32* jitter,
+ WebRtc_UWord32* jitter_transmission_time_offset) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ReportBlockStatistics()");
+ WebRtc_Word32 missing = 0;
+ WebRtc_Word32 ret = _rtpReceiver.Statistics(fraction_lost,
+ cum_lost,
+ ext_max,
+ jitter,
+ NULL,
+ jitter_transmission_time_offset,
+ &missing,
+ true);
+
+#ifdef MATLAB
+ if (_plot1 == NULL) {
+ _plot1 = eng.NewPlot(new MatlabPlot());
+ _plot1->AddTimeLine(30, "b", "lost", _clock.GetTimeInMS());
+ }
+ _plot1->Append("lost", missing);
+ _plot1->Plot();
+#endif
+
+ return ret;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* senderInfo) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteRTCPStat()");
+
+ return _rtcpReceiver.SenderInfoReceived(senderInfo);
+}
+
+// received RTCP report
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat(
+ std::vector<RTCPReportBlock>* receiveBlocks) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteRTCPStat()");
+
+ return _rtcpReceiver.StatisticsReceived(receiveBlocks);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::AddRTCPReportBlock(
+ const WebRtc_UWord32 SSRC,
+ const RTCPReportBlock* reportBlock) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "AddRTCPReportBlock()");
+
+ return _rtcpSender.AddReportBlock(SSRC, reportBlock);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoveRTCPReportBlock(
+ const WebRtc_UWord32 SSRC) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoveRTCPReportBlock()");
+
+ return _rtcpSender.RemoveReportBlock(SSRC);
+}
+
+/*
+ * (REMB) Receiver Estimated Max Bitrate
+ */
+bool ModuleRtpRtcpImpl::REMB() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "REMB()");
+
+ return _rtcpSender.REMB();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetREMBStatus(enable)");
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetREMBStatus(disable)");
+ }
+ return _rtcpSender.SetREMBStatus(enable);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBData(const WebRtc_UWord32 bitrate,
+ const WebRtc_UWord8 numberOfSSRC,
+ const WebRtc_UWord32* SSRC) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetREMBData(bitrate:%d,?,?)", bitrate);
+ return _rtcpSender.SetREMBData(bitrate, numberOfSSRC, SSRC);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetMaximumBitrateEstimate(
+ const WebRtc_UWord32 bitrate) {
+ if(!_rtcpSender.REMB()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "SetMaximumBitrateEstimate - REMB not enabled.");
+ return -1;
+ }
+ OnReceivedEstimatedMaxBitrate(bitrate);
+ return 0;
+}
+
+bool ModuleRtpRtcpImpl::SetRemoteBitrateObserver(
+ RtpRemoteBitrateObserver* observer) {
+ return _rtcpSender.SetRemoteBitrateObserver(observer);
+}
+
+/*
+ * (IJ) Extended jitter report.
+ */
+bool ModuleRtpRtcpImpl::IJ() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "IJ()");
+
+ return _rtcpSender.IJ();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetIJStatus(const bool enable) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetIJStatus(%s)", enable ? "true" : "false");
+
+ return _rtcpSender.SetIJStatus(enable);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id) {
+ return _rtpSender.RegisterRtpHeaderExtension(type, id);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
+ const RTPExtensionType type) {
+ return _rtpSender.DeregisterRtpHeaderExtension(type);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id) {
+ return _rtpReceiver.RegisterRtpHeaderExtension(type, id);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type) {
+ return _rtpReceiver.DeregisterRtpHeaderExtension(type);
+}
+
+void ModuleRtpRtcpImpl::SetTransmissionSmoothingStatus(const bool enable) {
+ _rtpSender.SetTransmissionSmoothingStatus(enable);
+}
+
+bool ModuleRtpRtcpImpl::TransmissionSmoothingStatus() const {
+ return _rtpSender.TransmissionSmoothingStatus();
+}
+
+/*
+* (TMMBR) Temporary Max Media Bit Rate
+*/
+bool ModuleRtpRtcpImpl::TMMBR() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TMMBR()");
+
+ return _rtcpSender.TMMBR();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetTMMBRStatus(enable)");
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetTMMBRStatus(disable)");
+ }
+ return _rtcpSender.SetTMMBRStatus(enable);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::TMMBRReceived(
+ const WebRtc_UWord32 size,
+ const WebRtc_UWord32 accNumCandidates,
+ TMMBRSet* candidateSet) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TMMBRReceived()");
+
+ return _rtcpReceiver.TMMBRReceived(size, accNumCandidates, candidateSet);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* boundingSet,
+ const WebRtc_UWord32 maxBitrateKbit) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetTMMBN()");
+
+ return _rtcpSender.SetTMMBN(boundingSet, maxBitrateKbit);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RequestTMMBR(const WebRtc_UWord32 estimatedBW,
+ const WebRtc_UWord32 packetOH) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RequestTMMBR()");
+
+ return _rtcpSender.RequestTMMBR(estimatedBW, packetOH);
+}
+
+/*
+* (NACK) Negative acknowledgement
+*/
+
+// Is Negative acknowledgement requests on/off?
+NACKMethod ModuleRtpRtcpImpl::NACK() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "NACK()");
+
+ NACKMethod childMethod = kNackOff;
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ // for default we need to check all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ std::list<ModuleRtpRtcpImpl*>::const_iterator it =
+ _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ NACKMethod nackMethod = module->NACK();
+ if (nackMethod != kNackOff) {
+ childMethod = nackMethod;
+ break;
+ }
+ }
+ it++;
+ }
+ }
+
+ NACKMethod method = _nackMethod;
+ if (childMethod != kNackOff) {
+ method = childMethod;
+ }
+ return method;
+}
+
+// Turn negative acknowledgement requests on/off
+WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus(NACKMethod method) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetNACKStatus(%u)", method);
+
+ _nackMethod = method;
+ _rtpReceiver.SetNACKStatus(method);
+ return 0;
+}
+
+// Returns the currently configured retransmission mode.
+int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SelectiveRetransmissions()");
+ return _rtpSender.SelectiveRetransmissions();
+}
+
+// Enable or disable a retransmission mode, which decides which packets will
+// be retransmitted if NACKed.
+int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSelectiveRetransmissions(%u)",
+ settings);
+ return _rtpSender.SetSelectiveRetransmissions(settings);
+}
+
+// Send a Negative acknowledgement packet
+WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nackList,
+ const WebRtc_UWord16 size) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SendNACK(size:%u)", size);
+
+ if (size > NACK_PACKETS_MAX_SIZE) {
+ RequestKeyFrame();
+ return -1;
+ }
+ WebRtc_UWord16 avgRTT = 0;
+ _rtcpReceiver.RTT(_rtpReceiver.SSRC(), NULL, &avgRTT, NULL, NULL);
+
+ WebRtc_UWord32 waitTime = 5 + ((avgRTT * 3) >> 1); // 5 + RTT*1.5
+ if (waitTime == 5) {
+ waitTime = 100; //During startup we don't have an RTT
+ }
+ const WebRtc_UWord32 now = _clock.GetTimeInMS();
+ const WebRtc_UWord32 timeLimit = now - waitTime;
+
+ if (_nackLastTimeSent < timeLimit) {
+ // send list
+ } else {
+ // only send if extended list
+ if (_nackLastSeqNumberSent == nackList[size - 1]) {
+ // last seq num is the same don't send list
+ return 0;
+ } else {
+ // send list
+ }
+ }
+ _nackLastTimeSent = now;
+ _nackLastSeqNumberSent = nackList[size - 1];
+
+ switch (_nackMethod) {
+ case kNackRtcp:
+ return _rtcpSender.SendRTCP(kRtcpNack, size, nackList);
+ case kNackOff:
+ return -1;
+ };
+ return -1;
+}
+
+// Store the sent packets, needed to answer to a Negative acknowledgement
+// requests
+WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
+ const bool enable,
+ const WebRtc_UWord16 numberToStore) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetStorePacketsStatus(enable, numberToStore:%d)",
+ numberToStore);
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetStorePacketsStatus(disable)");
+ }
+ return _rtpSender.SetStorePacketsStatus(enable, numberToStore);
+}
+
+/*
+* Audio
+*/
+
+// Outband TelephoneEvent detection
+WebRtc_Word32 ModuleRtpRtcpImpl::SetTelephoneEventStatus(
+ const bool enable,
+ const bool forwardToDecoder,
+ const bool detectEndOfTone) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SetTelephoneEventStatus(enable:%d forwardToDecoder:%d"
+ " detectEndOfTone:%d)", enable, forwardToDecoder,
+ detectEndOfTone);
+
+ return _rtpReceiver.SetTelephoneEventStatus(enable, forwardToDecoder,
+ detectEndOfTone);
+}
+
+// Is outband TelephoneEvent turned on/off?
+bool ModuleRtpRtcpImpl::TelephoneEvent() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TelephoneEvent()");
+
+ return _rtpReceiver.TelephoneEvent();
+}
+
+// Is forwarding of outband telephone events turned on/off?
+bool ModuleRtpRtcpImpl::TelephoneEventForwardToDecoder() const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "TelephoneEventForwardToDecoder()");
+
+ return _rtpReceiver.TelephoneEventForwardToDecoder();
+}
+
+// Send a TelephoneEvent tone using RFC 2833 (4733)
+WebRtc_Word32 ModuleRtpRtcpImpl::SendTelephoneEventOutband(
+ const WebRtc_UWord8 key,
+ const WebRtc_UWord16 timeMs,
+ const WebRtc_UWord8 level) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "SendTelephoneEventOutband(key:%u, timeMs:%u, level:%u)", key,
+ timeMs, level);
+
+ return _rtpSender.SendTelephoneEvent(key, timeMs, level);
+}
+
+bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
+ WebRtc_Word8& telephoneEvent) const {
+
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SendTelephoneEventActive()");
+
+ return _rtpSender.SendTelephoneEventActive(telephoneEvent);
+}
+
+// set audio packet size, used to determine when it's time to send a DTMF
+// packet in silence (CNG)
+WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioPacketSize(
+ const WebRtc_UWord16 packetSizeSamples) {
+
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetAudioPacketSize(%u)",
+ packetSizeSamples);
+
+ return _rtpSender.SetAudioPacketSize(packetSizeSamples);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus(
+ const bool enable,
+ const WebRtc_UWord8 ID) {
+
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetRTPAudioLevelIndicationStatus(enable=%d, ID=%u)",
+ enable,
+ ID);
+
+ if (enable) {
+ _rtpReceiver.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, ID);
+ } else {
+ _rtpReceiver.DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
+ }
+ return _rtpSender.SetAudioLevelIndicationStatus(enable, ID);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus(
+ bool& enable,
+ WebRtc_UWord8& ID) const {
+
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "GetRTPAudioLevelIndicationStatus()");
+ return _rtpSender.AudioLevelIndicationStatus(enable, ID);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel(const WebRtc_UWord8 level_dBov) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetAudioLevel(level_dBov:%u)",
+ level_dBov);
+ return _rtpSender.SetAudioLevel(level_dBov);
+}
+
+// Set payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32 ModuleRtpRtcpImpl::SetSendREDPayloadType(
+ const WebRtc_Word8 payloadType) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSendREDPayloadType(%d)",
+ payloadType);
+
+ return _rtpSender.SetRED(payloadType);
+}
+
+// Get payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType(
+ WebRtc_Word8& payloadType) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SendREDPayloadType()");
+
+ return _rtpSender.RED(payloadType);
+}
+
+
+/*
+* Video
+*/
+RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const {
+ return _rtpReceiver.VideoCodecType();
+}
+
+RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const {
+ return _rtpSender.VideoCodecType();
+}
+
+void ModuleRtpRtcpImpl::SetSendBitrate(const WebRtc_UWord32 startBitrate,
+ const WebRtc_UWord16 minBitrateKbit,
+ const WebRtc_UWord16 maxBitrateKbit) {
+
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetSendBitrate start:%ubit/s min:%uKbit/s max:%uKbit/s",
+ startBitrate, minBitrateKbit, maxBitrateKbit);
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+
+ if (defaultInstance) {
+ // for default we need to update all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ module->SetSendBitrate(startBitrate,
+ minBitrateKbit,
+ maxBitrateKbit);
+ }
+ it++;
+ }
+ }
+ // TODO(henrike): this function also returns a value. It never fails so
+ // make it return void.
+ _rtpSender.SetTargetSendBitrate(startBitrate);
+
+ _bandwidthManagement.SetSendBitrate(startBitrate, minBitrateKbit,
+ maxBitrateKbit);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
+ const KeyFrameRequestMethod method) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetKeyFrameRequestMethod(method:%u)",
+ method);
+
+ _keyFrameReqMethod = method;
+ return 0;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::RequestKeyFrame() {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "RequestKeyFrame");
+
+ switch (_keyFrameReqMethod) {
+ case kKeyFrameReqFirRtp:
+ return _rtpSender.SendRTPIntraRequest();
+ case kKeyFrameReqPliRtcp:
+ return _rtcpSender.SendRTCP(kRtcpPli);
+ case kKeyFrameReqFirRtcp:
+ return _rtcpSender.SendRTCP(kRtcpFir);
+ }
+ return -1;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
+ const WebRtc_UWord8 pictureID) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SendRTCPSliceLossIndication (pictureID:%d)",
+ pictureID);
+ return _rtcpSender.SendRTCP(kRtcpSli, 0, 0, false, pictureID);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetCameraDelay(const WebRtc_Word32 delayMS) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetCameraDelay(%d)",
+ delayMS);
+ const bool defaultInstance(_childModules.empty() ? false : true);
+
+ if (defaultInstance) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ module->SetCameraDelay(delayMS);
+ }
+ it++;
+ }
+ return 0;
+ }
+ return _rtcpSender.SetCameraDelay(delayMS);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetGenericFECStatus(
+ const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC) {
+ if (enable) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetGenericFECStatus(enable, %u)",
+ payloadTypeRED);
+ } else {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetGenericFECStatus(disable)");
+ }
+ return _rtpSender.SetGenericFECStatus(enable,
+ payloadTypeRED,
+ payloadTypeFEC);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::GenericFECStatus(
+ bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) {
+
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "GenericFECStatus()");
+
+ bool childEnabled = false;
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ // for default we need to check all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ bool enabled = false;
+ WebRtc_UWord8 dummyPTypeRED = 0;
+ WebRtc_UWord8 dummyPTypeFEC = 0;
+ if (module->GenericFECStatus(enabled,
+ dummyPTypeRED,
+ dummyPTypeFEC) == 0 && enabled) {
+ childEnabled = true;
+ break;
+ }
+ }
+ it++;
+ }
+ }
+ WebRtc_Word32 retVal = _rtpSender.GenericFECStatus(enable,
+ payloadTypeRED,
+ payloadTypeFEC);
+ if (childEnabled) {
+ // returns true if enabled for any child module
+ enable = childEnabled;
+ }
+ return retVal;
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetFECCodeRate(
+ const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp,
+ _id,
+ "SetFECCodeRate(%u, %u)",
+ keyFrameCodeRate,
+ deltaFrameCodeRate);
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ // for default we need to update all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ module->SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
+ }
+ it++;
+ }
+ return 0;
+ }
+ return _rtpSender.SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SetFECUepProtection(
+ const bool keyUseUepProtection,
+ const bool deltaUseUepProtection) {
+ WEBRTC_TRACE(kTraceModuleCall,
+ kTraceRtpRtcp, _id,
+ "SetFECUepProtection(%d, %d)",
+ keyUseUepProtection,
+ deltaUseUepProtection);
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ // for default we need to update all child modules too
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ module->SetFECUepProtection(keyUseUepProtection,
+ deltaUseUepProtection);
+ }
+ it++;
+ }
+ return 0;
+ }
+ return _rtpSender.SetFECUepProtection(keyUseUepProtection,
+ deltaUseUepProtection);
+}
+
+void ModuleRtpRtcpImpl::SetRemoteSSRC(const WebRtc_UWord32 SSRC) {
+ // inform about the incoming SSRC
+ _rtcpSender.SetRemoteSSRC(SSRC);
+ _rtcpReceiver.SetRemoteSSRC(SSRC);
+
+ // check for a SSRC collision
+ if (_rtpSender.SSRC() == SSRC && !_collisionDetected) {
+ // if we detect a collision change the SSRC but only once
+ _collisionDetected = true;
+ WebRtc_UWord32 newSSRC = _rtpSender.GenerateNewSSRC();
+ if (newSSRC == 0) {
+ // configured via API ignore
+ return;
+ }
+ if (kRtcpOff != _rtcpSender.Status()) {
+ // send RTCP bye on the current SSRC
+ _rtcpSender.SendRTCP(kRtcpBye);
+ }
+ // change local SSRC
+
+ // inform all objects about the new SSRC
+ _rtcpSender.SetSSRC(newSSRC);
+ _rtcpReceiver.SetSSRC(newSSRC);
+ }
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::BitrateReceivedNow() const {
+ return _rtpReceiver.BitrateNow();
+}
+
+void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* totalRate,
+ WebRtc_UWord32* videoRate,
+ WebRtc_UWord32* fecRate,
+ WebRtc_UWord32* nackRate) const {
+ const bool defaultInstance(_childModules.empty() ? false : true);
+
+ if (defaultInstance) {
+ // for default we need to update the send bitrate
+ CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
+
+ if (totalRate != NULL)
+ *totalRate = 0;
+ if (videoRate != NULL)
+ *videoRate = 0;
+ if (fecRate != NULL)
+ *fecRate = 0;
+ if (nackRate != NULL)
+ *nackRate = 0;
+
+ std::list<ModuleRtpRtcpImpl*>::const_iterator it =
+ _childModules.begin();
+ while (it != _childModules.end()) {
+ RtpRtcp* module = *it;
+ if (module) {
+ WebRtc_UWord32 childTotalRate = 0;
+ WebRtc_UWord32 childVideoRate = 0;
+ WebRtc_UWord32 childFecRate = 0;
+ WebRtc_UWord32 childNackRate = 0;
+ module->BitrateSent(&childTotalRate,
+ &childVideoRate,
+ &childFecRate,
+ &childNackRate);
+ if (totalRate != NULL && childTotalRate > *totalRate)
+ *totalRate = childTotalRate;
+ if (videoRate != NULL && childVideoRate > *videoRate)
+ *videoRate = childVideoRate;
+ if (fecRate != NULL && childFecRate > *fecRate)
+ *fecRate = childFecRate;
+ if (nackRate != NULL && childNackRate > *nackRate)
+ *nackRate = childNackRate;
+ }
+ it++;
+ }
+ return;
+ }
+ if (totalRate != NULL)
+ *totalRate = _rtpSender.BitrateLast();
+ if (videoRate != NULL)
+ *videoRate = _rtpSender.VideoBitrateSent();
+ if (fecRate != NULL)
+ *fecRate = _rtpSender.FecOverheadRate();
+ if (nackRate != NULL)
+ *nackRate = _rtpSender.NackOverheadRate();
+}
+
+int ModuleRtpRtcpImpl::EstimatedSendBandwidth(
+ WebRtc_UWord32* available_bandwidth) const {
+ return _bandwidthManagement.AvailableBandwidth(available_bandwidth);
+}
+
+int ModuleRtpRtcpImpl::EstimatedReceiveBandwidth(
+ WebRtc_UWord32* available_bandwidth) const {
+ if (!_rtcpSender.ValidBitrateEstimate())
+ return -1;
+ *available_bandwidth = _rtcpSender.LatestBandwidthEstimate();
+ return 0;
+}
+
+// for lip sync
+void ModuleRtpRtcpImpl::OnReceivedNTP() {
+ // don't do anything if we are the audio module
+ // video module is responsible for sync
+ if (!_audio) {
+ WebRtc_Word32 diff = 0;
+ WebRtc_UWord32 receivedNTPsecs = 0;
+ WebRtc_UWord32 receivedNTPfrac = 0;
+ WebRtc_UWord32 RTCPArrivalTimeSecs = 0;
+ WebRtc_UWord32 RTCPArrivalTimeFrac = 0;
+
+ if (0 == _rtcpReceiver.NTP(&receivedNTPsecs,
+ &receivedNTPfrac,
+ &RTCPArrivalTimeSecs,
+ &RTCPArrivalTimeFrac)) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+ if (_audioModule) {
+ if (0 != _audioModule->RemoteNTP(&_receivedNTPsecsAudio,
+ &_receivedNTPfracAudio,
+ &_RTCPArrivalTimeSecsAudio,
+ &_RTCPArrivalTimeFracAudio)) {
+ // failed ot get audio NTP
+ return;
+ }
+ }
+ if (_receivedNTPfracAudio != 0) {
+ // ReceivedNTPxxx is NTP at sender side when sent.
+ // RTCPArrivalTimexxx is NTP at receiver side when received.
+ // can't use ConvertNTPTimeToMS since calculation can be
+ // negative
+
+ WebRtc_Word32 NTPdiff = (WebRtc_Word32)
+ ((_receivedNTPsecsAudio - receivedNTPsecs) *
+ 1000); // ms
+ NTPdiff += (WebRtc_Word32)
+ (_receivedNTPfracAudio / FracMS - receivedNTPfrac / FracMS);
+
+ WebRtc_Word32 RTCPdiff =
+ static_cast<WebRtc_Word32> ((_RTCPArrivalTimeSecsAudio -
+ RTCPArrivalTimeSecs) * 1000);
+ RTCPdiff += (WebRtc_Word32)
+ (_RTCPArrivalTimeFracAudio / FracMS -
+ RTCPArrivalTimeFrac / FracMS);
+
+ diff = NTPdiff - RTCPdiff;
+ // if diff is + video is behind
+ if (diff < -1000 || diff > 1000) {
+ // unresonable ignore value.
+ diff = 0;
+ return;
+ }
+ }
+ }
+ // export via callback
+ // after release of critsect
+ _rtcpReceiver.UpdateLipSync(diff);
+ }
+}
+
+// our local BW estimate is updated
+void ModuleRtpRtcpImpl::OnBandwidthEstimateUpdate(
+ WebRtc_UWord16 bandWidthKbit) {
+
+ WebRtc_UWord32 maxBitrateKbit = _rtpReceiver.MaxConfiguredBitrate() / 1000;
+ if (maxBitrateKbit) {
+ // the app has set a max bitrate
+ if (maxBitrateKbit < bandWidthKbit) {
+ // cap TMMBR at max configured bitrate
+ bandWidthKbit = (WebRtc_UWord16)maxBitrateKbit;
+ }
+ }
+ if (_rtcpSender.TMMBR()) {
+ /* Maximum total media bit rate:
+ The upper limit on total media bit rate for a given media
+ stream at a particular receiver and for its selected protocol
+ layer. Note that this value cannot be measured on the
+ received media stream. Instead, it needs to be calculated or
+ determined through other means, such as quality of service
+ (QoS) negotiations or local resource limitations. Also note
+ that this value is an average (on a timescale that is
+ reasonable for the application) and that it may be different
+ from the instantaneous bit rate seen by packets in the media
+ stream.
+ */
+ /* Overhead:
+ All protocol header information required to convey a packet
+ with media data from sender to receiver, from the application
+ layer down to a pre-defined protocol level (for example, down
+ to, and including, the IP header). Overhead may include, for
+ example, IP, UDP, and RTP headers, any layer 2 headers, any
+ Contributing Sources (CSRCs), RTP padding, and RTP header
+ extensions. Overhead excludes any RTP payload headers and the
+ payload itself.
+ */
+ _rtpReceiver.PacketOHReceived();
+
+ // call RequestTMMBR when our localy created estimate changes
+ _rtcpSender.RequestTMMBR(bandWidthKbit, 0);
+ }
+}
+
+RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
+ const RateControlInput& rateControlInput) {
+
+ bool firstOverUse = false;
+ RateControlRegion region = _rtcpSender.UpdateOverUseState(rateControlInput,
+ firstOverUse);
+ if (firstOverUse) {
+ // Send TMMBR or REMB immediately.
+ WebRtc_UWord16 RTT = 0;
+ _rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
+ // About to send TMMBR, first run remote rate control
+ // to get a target bit rate.
+ unsigned int target_bitrate =
+ _rtcpSender.CalculateNewTargetBitrate(RTT);
+ if (REMB()) {
+ _rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
+ } else if (TMMBR()) {
+ _rtcpSender.SendRTCP(kRtcpTmmbr);
+ }
+ }
+ return region;
+}
+
+// bad state of RTP receiver request a keyframe
+void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
+ RequestKeyFrame();
+}
+
+void ModuleRtpRtcpImpl::OnReceivedIntraFrameRequest(const RtpRtcp* caller) {
+ if (_defaultModule) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ // if we use a default module pass this info to the default module
+ _defaultModule->OnReceivedIntraFrameRequest(caller);
+ return;
+ }
+ }
+
+ WebRtc_UWord8 streamIdx = 0;
+ FrameType frameType = kVideoFrameKey;
+ if (_simulcast) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ // loop though child modules and count idx
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ while (it != _childModules.end()) {
+ ModuleRtpRtcpImpl* childModule = *it;
+ if (childModule == caller) {
+ break;
+ }
+ streamIdx++;
+ it++;
+ }
+ }
+ _rtcpReceiver.OnReceivedIntraFrameRequest(frameType, streamIdx);
+}
+
+void ModuleRtpRtcpImpl::OnReceivedEstimatedMaxBitrate(
+ const WebRtc_UWord32 maxBitrate) {
+ // TODO(mflodman) Split this function in two parts. One for the child module
+ // and one for the default module.
+
+ // We received a REMB.
+ if (_defaultModule) {
+ // Send this update to the REMB instance to take actions.
+ _rtcpSender.ReceivedRemb(maxBitrate);
+ return;
+ }
+
+ WebRtc_UWord32 newBitrate = 0;
+ WebRtc_UWord8 fractionLost = 0;
+ WebRtc_UWord16 roundTripTime = 0;
+ WebRtc_UWord16 bwEstimateKbit = WebRtc_UWord16(maxBitrate / 1000);
+ if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
+ &newBitrate,
+ &fractionLost,
+ &roundTripTime) == 0) {
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ fractionLost,
+ roundTripTime);
+
+ // We've received a new bandwidth estimate lower than the current send
+ // bitrate. For simulcast we need to update the sending bitrate for all
+ // streams.
+ if (_simulcast) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
+ WebRtc_UWord8 idx = 0;
+ for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ it != _childModules.end(); ++it) {
+ // sanity
+ if (idx >= (_sendVideoCodec.numberOfSimulcastStreams - 1)) {
+ return;
+ }
+ ModuleRtpRtcpImpl* module = *it;
+ if (newBitrate >= _sendVideoCodec.simulcastStream[idx].maxBitrate) {
+ module->_bandwidthManagement.SetSendBitrate(
+ _sendVideoCodec.simulcastStream[idx].maxBitrate, 0, 0);
+ module->_rtpSender.SetTargetSendBitrate(
+ _sendVideoCodec.simulcastStream[idx].maxBitrate);
+
+ newBitrate -= _sendVideoCodec.simulcastStream[idx].maxBitrate;
+ } else {
+ module->_bandwidthManagement.SetSendBitrate(newBitrate, 0, 0);
+ module->_rtpSender.SetTargetSendBitrate(newBitrate);
+ newBitrate -= newBitrate;
+ }
+ idx++;
+ }
+ }
+ }
+ // For non-simulcast, update all child modules with the new bandwidth estimate
+ // regardless of the new estimate.
+ if (!_simulcast) {
+ // Update all child modules with the new max bitrate before exiting.
+ CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
+ for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ it != _childModules.end(); ++it) {
+ // Update all child modules with the maximum bitrate estimate.
+ ModuleRtpRtcpImpl* module = *it;
+ WebRtc_UWord32 ignoreBitrate = 0;
+ WebRtc_UWord8 ignoreFractionLost = 0;
+ WebRtc_UWord16 ignoreRoundTripTime = 0;
+ module->_bandwidthManagement.UpdateBandwidthEstimate(
+ bwEstimateKbit,
+ &ignoreBitrate,
+ &ignoreFractionLost,
+ &ignoreRoundTripTime);
+ // We don't need to take care of a possible lowered bitrate, that is
+ // handled earlier in this function for the default module.
+ }
+ }
+}
+
+// received a request for a new SLI
+void ModuleRtpRtcpImpl::OnReceivedSliceLossIndication(
+ const WebRtc_UWord8 pictureID) {
+
+ if (_defaultModule) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ // if we use a default module pass this info to the default module
+ _defaultModule->OnReceivedSliceLossIndication(pictureID);
+ return;
+ }
+ }
+ _rtcpReceiver.OnReceivedSliceLossIndication(pictureID);
+}
+
+// received a new refereence frame
+void ModuleRtpRtcpImpl::OnReceivedReferencePictureSelectionIndication(
+ const WebRtc_UWord64 pictureID) {
+
+ if (_defaultModule) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ // if we use a default module pass this info to the default module
+ _defaultModule->OnReceivedReferencePictureSelectionIndication(
+ pictureID);
+ return;
+ }
+ }
+ _rtcpReceiver.OnReceivedReferencePictureSelectionIndication(pictureID);
+}
+
+void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
+ const WebRtc_UWord16 bwEstimateKbit) {
+
+ // We received a TMMBR
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (defaultInstance) {
+ ProcessDefaultModuleBandwidth();
+ return;
+ }
+ if (_audio) {
+ _rtcpReceiver.UpdateBandwidthEstimate(bwEstimateKbit);
+ } else {
+ WebRtc_UWord32 newBitrate = 0;
+ WebRtc_UWord8 fractionLost = 0;
+ WebRtc_UWord16 roundTripTime = 0;
+ if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
+ &newBitrate,
+ &fractionLost,
+ &roundTripTime) == 0) {
+ if (!_defaultModule) {
+ // No default module check if we should trigger OnNetworkChanged
+ // via video callback
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ fractionLost,
+ roundTripTime);
+ }
+ if (newBitrate > 0) {
+ // update bitrate
+ _rtpSender.SetTargetSendBitrate(newBitrate);
+ }
+ }
+ }
+ if (_defaultModule) {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) {
+ // if we use a default module pass this info to the default module
+ _defaultModule->OnReceivedBandwidthEstimateUpdate(bwEstimateKbit);
+ return;
+ }
+ }
+}
+
+// bw estimation
+// We received a RTCP report block
+void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTime,
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum) {
+
+ const bool defaultInstance(_childModules.empty() ? false : true);
+ if (!defaultInstance) {
+ WebRtc_UWord32 newBitrate = 0;
+ WebRtc_UWord8 loss = fractionLost; // local copy since it can change
+ WebRtc_UWord32 videoRate = 0;
+ WebRtc_UWord32 fecRate = 0;
+ WebRtc_UWord32 nackRate = 0;
+ BitrateSent(NULL, &videoRate, &fecRate, &nackRate);
+ if (_bandwidthManagement.UpdatePacketLoss(
+ lastReceivedExtendedHighSeqNum,
+ videoRate + fecRate + nackRate,
+ roundTripTime,
+ &loss,
+ &newBitrate,
+ _clock.GetTimeInMS()) != 0) {
+ // ignore this update
+ return;
+ }
+ // We need to do update RTP sender before calling default module in
+ // case we'll strip any layers.
+ if (!_simulcast) {
+ // the default module will inform all child modules about
+ // their bitrate
+ _rtpSender.SetTargetSendBitrate(newBitrate);
+ }
+ if (_defaultModule) {
+ // if we have a default module update it
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ if (_defaultModule) { // we need to check again inside the critsect
+ // if we use a default module pass this info to the
+ // default module
+ _defaultModule->OnPacketLossStatisticsUpdate(
+ loss, // send in the filtered loss
+ roundTripTime,
+ lastReceivedExtendedHighSeqNum);
+ }
+ return;
+ }
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ fractionLost,
+ roundTripTime);
+ } else {
+ if (!_simulcast) {
+ ProcessDefaultModuleBandwidth();
+ } else {
+ // default and simulcast
+ WebRtc_UWord32 newBitrate = 0;
+ WebRtc_UWord8 loss = fractionLost; // local copy
+ WebRtc_UWord32 videoRate = 0;
+ WebRtc_UWord32 fecRate = 0;
+ WebRtc_UWord32 nackRate = 0;
+ BitrateSent(NULL, &videoRate, &fecRate, &nackRate);
+ if (_bandwidthManagement.UpdatePacketLoss(0, // we can't use this
+ videoRate + fecRate + nackRate,
+ roundTripTime,
+ &loss,
+ &newBitrate,
+ _clock.GetTimeInMS()) != 0) {
+ // ignore this update
+ return;
+ }
+ _rtpSender.SetTargetSendBitrate(newBitrate);
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ loss,
+ roundTripTime);
+ // sanity
+ if (_sendVideoCodec.codecType == kVideoCodecUnknown) {
+ return;
+ }
+ CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
+ std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ WebRtc_UWord8 idx = 0;
+ while (it != _childModules.end()) {
+ // sanity
+ if (idx >= (_sendVideoCodec.numberOfSimulcastStreams - 1)) {
+ return;
+ }
+ ModuleRtpRtcpImpl* module = *it;
+ // update all child modules
+ if (newBitrate >=
+ _sendVideoCodec.simulcastStream[idx].maxBitrate) {
+ module->_bandwidthManagement.SetSendBitrate(
+ _sendVideoCodec.simulcastStream[idx].maxBitrate, 0, 0);
+ module->_rtpSender.SetTargetSendBitrate(
+ _sendVideoCodec.simulcastStream[idx].maxBitrate);
+
+ newBitrate -=
+ _sendVideoCodec.simulcastStream[idx].maxBitrate;
+ } else {
+ module->_bandwidthManagement.SetSendBitrate(newBitrate,
+ 0,
+ 0);
+ module->_rtpSender.SetTargetSendBitrate(newBitrate);
+ newBitrate -= newBitrate;
+ }
+ idx++;
+ }
+ }
+ }
+}
+
+void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
+
+ WebRtc_UWord32 minBitrateBps = 0xffffffff;
+ WebRtc_UWord32 maxBitrateBps = 0;
+ WebRtc_UWord32 count = 0;
+ WebRtc_UWord32 fractionLostAcc = 0;
+ WebRtc_UWord16 maxRoundTripTime = 0;
+ {
+ // get min and max for the sending channels
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ it != _childModules.end(); ++ it) {
+ // Get child RTP sender and ask for bitrate estimate.
+ ModuleRtpRtcpImpl* childModule = *it;
+ if (childModule->Sending()) {
+ RTPSender& childRtpSender = (*it)->_rtpSender;
+ const WebRtc_UWord32 childEstimateBps =
+ 1000 * childRtpSender.TargetSendBitrateKbit();
+ if (childEstimateBps < minBitrateBps) {
+ minBitrateBps = childEstimateBps;
+ }
+ if (childEstimateBps > maxBitrateBps) {
+ maxBitrateBps = childEstimateBps;
+ }
+ RTCPReceiver& childRtcpReceiver = (*it)->_rtcpReceiver;
+
+ std::vector<RTCPReportBlock> rtcp_blocks;
+ childRtcpReceiver.StatisticsReceived(&rtcp_blocks);
+ for (std::vector<RTCPReportBlock>::iterator rit = rtcp_blocks.begin();
+ rit != rtcp_blocks.end(); ++rit) {
+ count++;
+ fractionLostAcc += rit->fractionLost;
+ WebRtc_UWord16 RTT = 0;
+ childRtcpReceiver.RTT(rit->remoteSSRC, &RTT, NULL, NULL, NULL);
+ maxRoundTripTime = (RTT > maxRoundTripTime) ? RTT : maxRoundTripTime;
+ }
+ }
+ }
+ } // end critsect
+
+ if (count == 0) {
+ // No sending modules and no bitrate estimate.
+ return;
+ }
+
+ // Update RTT to all receive only child modules, they won't have their own RTT
+ // estimate. Assume the receive only channels are on similar links as the
+ // sending channel and have approximately the same RTT.
+ {
+ CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
+ it != _childModules.end(); ++it) {
+ if (!(*it)->Sending()) {
+ (*it)->_rtcpReceiver.SetRTT(maxRoundTripTime);
+ }
+ }
+ }
+
+ _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
+
+ // Update default module bitrate. Don't care about min max.
+ WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
+ _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
+ fractionLostAvg ,
+ maxRoundTripTime);
+}
+
+void ModuleRtpRtcpImpl::OnRequestSendReport() {
+ _rtcpSender.SendRTCP(kRtcpSr);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
+ const WebRtc_UWord64 pictureID) {
+ return _rtcpSender.SendRTCP(kRtcpRpsi, 0, 0, false, pictureID);
+}
+
+WebRtc_UWord32 ModuleRtpRtcpImpl::SendTimeOfSendReport(
+ const WebRtc_UWord32 sendReport) {
+ return _rtcpSender.SendTimeOfSendReport(sendReport);
+}
+
+void ModuleRtpRtcpImpl::OnReceivedNACK(
+ const WebRtc_UWord16 nackSequenceNumbersLength,
+ const WebRtc_UWord16* nackSequenceNumbers) {
+ if (!_rtpSender.StorePackets() ||
+ nackSequenceNumbers == NULL ||
+ nackSequenceNumbersLength == 0) {
+ return;
+ }
+ WebRtc_UWord16 avgRTT = 0;
+ _rtcpReceiver.RTT(_rtpReceiver.SSRC(), NULL, &avgRTT, NULL, NULL);
+ _rtpSender.OnReceivedNACK(nackSequenceNumbersLength,
+ nackSequenceNumbers,
+ avgRTT);
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::LastReceivedNTP(
+ WebRtc_UWord32& RTCPArrivalTimeSecs, // when we received the last report
+ WebRtc_UWord32& RTCPArrivalTimeFrac,
+ WebRtc_UWord32& remoteSR) {
+ // remote SR: NTP inside the last received (mid 16 bits from sec and frac)
+ WebRtc_UWord32 NTPsecs = 0;
+ WebRtc_UWord32 NTPfrac = 0;
+
+ if (-1 == _rtcpReceiver.NTP(&NTPsecs,
+ &NTPfrac,
+ &RTCPArrivalTimeSecs,
+ &RTCPArrivalTimeFrac)) {
+ return -1;
+ }
+ remoteSR = ((NTPsecs & 0x0000ffff) << 16) + ((NTPfrac & 0xffff0000) >> 16);
+ return 0;
+}
+
+void ModuleRtpRtcpImpl::OnReceivedTMMBR() {
+ // we received a TMMBR in a RTCP packet
+ // answer with a TMMBN
+ UpdateTMMBR();
+}
+
+bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
+ // if this returns true this channel has timed out
+ // periodically check if this is true and if so call UpdateTMMBR
+ return _rtcpReceiver.UpdateRTCPReceiveInformationTimers();
+}
+
+WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
+ WebRtc_Word32 numBoundingSet = 0;
+ WebRtc_Word32 newBitrates = 0;
+ WebRtc_UWord32 minBitrateKbit = 0;
+ WebRtc_UWord32 maxBitrateKbit = 0;
+ WebRtc_UWord32 accNumCandidates = 0;
+
+ if (!_childModules.empty()) {
+ // Default module should not handle this
+ return -1;
+ }
+
+ WebRtc_Word32 size = _rtcpReceiver.TMMBRReceived(0, 0, NULL);
+ if (size > 0) {
+ TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
+ // get candidate set from receiver
+ accNumCandidates = _rtcpReceiver.TMMBRReceived(size,
+ accNumCandidates,
+ candidateSet);
+ } else {
+ // candidate set empty
+ VerifyAndAllocateCandidateSet(0); // resets candidate set
+ }
+ // Find bounding set
+ TMMBRSet* boundingSet = NULL;
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ if (numBoundingSet == -1) {
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceRtpRtcp,
+ _id,
+ "Failed to find TMMBR bounding set.");
+ return -1;
+ }
+ // Set bounding set
+ // Inform remote clients about the new bandwidth
+ // inform the remote client
+ _rtcpSender.SetTMMBN(boundingSet,
+ _rtpSender.MaxConfiguredBitrateVideo() / 1000);
+ // might trigger a TMMBN
+ if (numBoundingSet == 0) {
+ // owner of max bitrate request has timed out
+ // empty bounding set has been sent
+ return 0;
+ }
+ // Get net bitrate from bounding set depending on sent packet rate
+ newBitrates = CalcMinMaxBitRate(_rtpSender.PacketRate(),
+ (WebRtc_UWord32)numBoundingSet,
+ minBitrateKbit,
+ maxBitrateKbit);
+
+ // no critsect when calling out to "unknown" code
+ if (newBitrates == 0) {
+ // we have new bitrate
+ // Set new max bitrate
+ // we have a new bandwidth estimate on this channel
+ OnReceivedBandwidthEstimateUpdate((WebRtc_UWord16)minBitrateKbit);
+ WEBRTC_TRACE(kTraceStream,
+ kTraceRtpRtcp,
+ _id,
+ "Set TMMBR request min:%d kbps max:%d kbps, channel: %d",
+ minBitrateKbit, maxBitrateKbit, _id);
+ }
+ return 0;
+}
+
+// called from RTCPsender
+WebRtc_Word32 ModuleRtpRtcpImpl::BoundingSet(bool& tmmbrOwner,
+ TMMBRSet*& boundingSet) {
+ return _rtcpReceiver.BoundingSet(tmmbrOwner,
+ boundingSet);
+}
+
+void ModuleRtpRtcpImpl::SendKeyFrame() {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "SendKeyFrame()");
+ OnReceivedIntraFrameRequest(0);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
new file mode 100644
index 0000000..e16bf71
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -0,0 +1,622 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
+
+#include <list>
+
+#include "bandwidth_management.h"
+#include "rtcp_receiver.h"
+#include "rtcp_sender.h"
+#include "rtp_receiver.h"
+#include "rtp_rtcp.h"
+#include "rtp_sender.h"
+
+#ifdef MATLAB
+class MatlabPlot;
+#endif
+
+namespace webrtc {
+
+class ModuleRtpRtcpImpl : public RtpRtcp, private TMMBRHelp
+{
+public:
+ ModuleRtpRtcpImpl(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock);
+
+ virtual ~ModuleRtpRtcpImpl();
+
+ // get Module ID
+ WebRtc_Word32 Id() {return _id;}
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ // De-muxing functionality for
+ virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module);
+ virtual WebRtc_Word32 DeRegisterDefaultModule();
+ virtual bool DefaultModuleRegistered();
+
+ virtual WebRtc_UWord32 NumberChildModules();
+
+ // Lip-sync between voice-video
+ virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module);
+ virtual WebRtc_Word32 DeRegisterSyncModule();
+
+ virtual WebRtc_Word32 RegisterVideoModule(RtpRtcp* videoModule);
+ virtual void DeRegisterVideoModule();
+
+ // returns the number of milliseconds until the module want a worker thread to call Process
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+
+ // Process any pending tasks such as timeouts
+ virtual WebRtc_Word32 Process();
+
+ /**
+ * Receiver
+ */
+ virtual WebRtc_Word32 InitReceiver();
+
+ // configure a timeout value
+ virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
+ const WebRtc_UWord32 RTCPtimeoutMS);
+
+ // Set periodic dead or alive notification
+ virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
+ const bool enable,
+ const WebRtc_UWord8 sampleTimeSeconds);
+
+ // Get periodic dead or alive notification status
+ virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
+ bool &enable,
+ WebRtc_UWord8 &sampleTimeSeconds);
+
+ virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec);
+
+ virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec);
+
+ virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec,
+ WebRtc_Word8* plType);
+
+ virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec,
+ WebRtc_Word8* plType);
+
+ virtual WebRtc_Word32 DeRegisterReceivePayload(
+ const WebRtc_Word8 payloadType);
+
+ // register RTP header extension
+ virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id);
+
+ virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
+ const RTPExtensionType type);
+
+ // get the currently configured SSRC filter
+ virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
+
+ // set a SSRC to be used as a filter for incoming RTP streams
+ virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
+
+ // Get last received remote timestamp
+ virtual WebRtc_UWord32 RemoteTimestamp() const;
+
+ // Get the current estimated remote timestamp
+ virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
+
+ virtual WebRtc_UWord32 RemoteSSRC() const;
+
+ virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
+
+ virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
+ const WebRtc_UWord32 SSRC);
+
+ virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const;
+
+ // called by the network module when we receive a packet
+ virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket,
+ const WebRtc_UWord16 packetLength);
+
+ virtual WebRtc_Word32 IncomingAudioNTP(const WebRtc_UWord32 audioReceivedNTPsecs,
+ const WebRtc_UWord32 audioReceivedNTPfrac,
+ const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
+ const WebRtc_UWord32 audioRTCPArrivalTimeFrac);
+
+ // Used by the module to deliver the incoming data to the codec module
+ virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
+
+ // Used by the module to deliver messages to the codec module/appliation
+ virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
+
+ virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback);
+
+ virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback);
+
+ virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
+
+ /**
+ * Sender
+ */
+ virtual WebRtc_Word32 InitSender();
+
+ virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable,
+ const WebRtc_Word8 unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS);
+
+ virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
+ WebRtc_Word8* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const;
+
+ virtual bool RTPKeepalive() const;
+
+ virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec);
+
+ virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec);
+
+ virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
+
+ virtual WebRtc_Word8 SendPayloadType() const;
+
+ // register RTP header extension
+ virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
+ const RTPExtensionType type,
+ const WebRtc_UWord8 id);
+
+ virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
+ const RTPExtensionType type);
+
+ virtual void SetTransmissionSmoothingStatus(const bool enable);
+
+ virtual bool TransmissionSmoothingStatus() const;
+
+ // get start timestamp
+ virtual WebRtc_UWord32 StartTimestamp() const;
+
+ // configure start timestamp, default is a random number
+ virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
+
+ virtual WebRtc_UWord16 SequenceNumber() const;
+
+ // Set SequenceNumber, default is a random number
+ virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
+
+ virtual WebRtc_UWord32 SSRC() const;
+
+ // configure SSRC, default is a random number
+ virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
+
+ virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
+
+ virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength);
+
+ virtual WebRtc_Word32 SetCSRCStatus(const bool include);
+
+ virtual WebRtc_UWord32 PacketCountSent() const;
+
+ virtual int CurrentSendFrequencyHz() const;
+
+ virtual WebRtc_UWord32 ByteCountSent() const;
+
+ virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
+ const bool setSSRC,
+ const WebRtc_UWord32 SSRC);
+
+ virtual WebRtc_Word32 RTXSendStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const;
+
+ // sends kRtcpByeCode when going from true to false
+ virtual WebRtc_Word32 SetSendingStatus(const bool sending);
+
+ virtual bool Sending() const;
+
+ // Drops or relays media packets
+ virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
+
+ virtual bool SendingMedia() const;
+
+ // Used by the module to send RTP and RTCP packet to the network module
+ virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
+
+ // Used by the codec module to deliver a video or audio frame for packetization
+ virtual WebRtc_Word32 SendOutgoingData(
+ const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation = NULL,
+ const RTPVideoHeader* rtpVideoHdr = NULL);
+
+ /*
+ * RTCP
+ */
+
+ // Get RTCP status
+ virtual RTCPMethod RTCP() const;
+
+ // configure RTCP status i.e on/off
+ virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
+
+ // Set RTCP CName
+ virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
+
+ // Get RTCP CName
+ virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
+
+ // Get remote CName
+ virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const;
+
+ // Get remote NTP
+ virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
+ WebRtc_UWord32 *ReceivedNTPfrac,
+ WebRtc_UWord32 *RTCPArrivalTimeSecs,
+ WebRtc_UWord32 *RTCPArrivalTimeFrac) const ;
+
+ virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]);
+
+ virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
+
+ // Get RoundTripTime
+ virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
+ WebRtc_UWord16* RTT,
+ WebRtc_UWord16* avgRTT,
+ WebRtc_UWord16* minRTT,
+ WebRtc_UWord16* maxRTT) const;
+
+ // Reset RoundTripTime statistics
+ virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
+
+ // Force a send of an RTCP packet
+ // normal SR and RR are triggered via the process function
+ virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport);
+
+ // statistics of our localy created statistics of the received RTP stream
+ virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter,
+ WebRtc_UWord32 *max_jitter = NULL) const;
+
+ // Reset RTP statistics
+ virtual WebRtc_Word32 ResetStatisticsRTP();
+
+ virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
+
+ virtual WebRtc_Word32 ResetSendDataCountersRTP();
+
+ // statistics of the amount of data sent and received
+ virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
+ WebRtc_UWord32 *packetsSent,
+ WebRtc_UWord32 *bytesReceived,
+ WebRtc_UWord32 *packetsReceived) const;
+
+ virtual WebRtc_Word32 ReportBlockStatistics(
+ WebRtc_UWord8 *fraction_lost,
+ WebRtc_UWord32 *cum_lost,
+ WebRtc_UWord32 *ext_max,
+ WebRtc_UWord32 *jitter,
+ WebRtc_UWord32 *jitter_transmission_time_offset);
+
+ // Get received RTCP report, sender info
+ virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo);
+
+ // Get received RTCP report, report block
+ virtual WebRtc_Word32 RemoteRTCPStat(
+ std::vector<RTCPReportBlock>* receiveBlocks) const;
+
+ // Set received RTCP report block
+ virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC,
+ const RTCPReportBlock* receiveBlock);
+
+ virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC);
+
+ /*
+ * (REMB) Receiver Estimated Max Bitrate
+ */
+ virtual bool REMB() const;
+
+ virtual WebRtc_Word32 SetREMBStatus(const bool enable);
+
+ virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
+ const WebRtc_UWord8 numberOfSSRC,
+ const WebRtc_UWord32* SSRC);
+
+ virtual WebRtc_Word32 SetMaximumBitrateEstimate(
+ const WebRtc_UWord32 bitrate);
+
+ virtual bool SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer);
+ /*
+ * (IJ) Extended jitter report.
+ */
+ virtual bool IJ() const;
+
+ virtual WebRtc_Word32 SetIJStatus(const bool enable);
+
+ /*
+ * (TMMBR) Temporary Max Media Bit Rate
+ */
+ virtual bool TMMBR() const ;
+
+ virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
+
+ virtual WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
+ const WebRtc_UWord32 accNumCandidates,
+ TMMBRSet* candidateSet) const;
+
+ virtual WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
+ const WebRtc_UWord32 maxBitrateKbit);
+
+ virtual WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
+ const WebRtc_UWord32 packetOH);
+
+ virtual WebRtc_UWord16 MaxPayloadLength() const;
+
+ virtual WebRtc_UWord16 MaxDataPayloadLength() const;
+
+ virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
+
+ virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
+ const bool IPV6,
+ const WebRtc_UWord8 authenticationOverhead = 0);
+
+ /*
+ * (NACK) Negative acknowledgement
+ */
+
+ // Is Negative acknowledgement requests on/off?
+ virtual NACKMethod NACK() const ;
+
+ // Turn negative acknowledgement requests on/off
+ virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
+
+ virtual int SelectiveRetransmissions() const;
+
+ virtual int SetSelectiveRetransmissions(uint8_t settings);
+
+ // Send a Negative acknowledgement packet
+ virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
+ const WebRtc_UWord16 size);
+
+ // Store the sent packets, needed to answer to a Negative acknowledgement requests
+ virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200);
+
+ /*
+ * (APP) Application specific data
+ */
+ virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 length);
+ /*
+ * (XR) VOIP metric
+ */
+ virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
+
+ /*
+ * Audio
+ */
+
+ // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+ virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
+
+ // Outband DTMF detection
+ virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
+ const bool forwardToDecoder,
+ const bool detectEndOfTone = false);
+
+ // Is outband DTMF turned on/off?
+ virtual bool TelephoneEvent() const;
+
+ // Is forwarding of outband telephone events turned on/off?
+ virtual bool TelephoneEventForwardToDecoder() const;
+
+ virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
+
+ // Send a TelephoneEvent tone using RFC 2833 (4733)
+ virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level);
+
+ // Set payload type for Redundant Audio Data RFC 2198
+ virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType);
+
+ // Get payload type for Redundant Audio Data RFC 2198
+ virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const;
+
+ // Set status and ID for header-extension-for-audio-level-indication.
+ virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID);
+
+ // Get status and ID for header-extension-for-audio-level-indication.
+ virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const;
+
+ // Store the audio level in dBov for header-extension-for-audio-level-indication.
+ virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
+
+ /*
+ * Video
+ */
+ virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
+
+ virtual RtpVideoCodecTypes SendVideoCodec() const;
+
+ virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID);
+
+ // Set method for requestion a new key frame
+ virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
+
+ // send a request for a keyframe
+ virtual WebRtc_Word32 RequestKeyFrame();
+
+ virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
+
+ virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
+ const WebRtc_UWord16 minBitrateKbit,
+ const WebRtc_UWord16 maxBitrateKbit);
+
+ virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC);
+
+ virtual WebRtc_Word32 GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC);
+
+
+ virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate);
+
+ virtual WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection);
+
+ virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
+ WebRtc_UWord32& NTPfrac,
+ WebRtc_UWord32& remoteSR);
+
+ virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
+ TMMBRSet*& boundingSetRec);
+
+ virtual void BitrateSent(WebRtc_UWord32* totalRate,
+ WebRtc_UWord32* videoRate,
+ WebRtc_UWord32* fecRate,
+ WebRtc_UWord32* nackRate) const;
+
+ virtual int EstimatedSendBandwidth(
+ WebRtc_UWord32* available_bandwidth) const;
+
+ virtual int EstimatedReceiveBandwidth(
+ WebRtc_UWord32* available_bandwidth) const;
+
+ virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC);
+
+ virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
+
+ virtual RateControlRegion OnOverUseStateUpdate(const RateControlInput& rateControlInput);
+
+ // good state of RTP receiver inform sender
+ virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
+
+ virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit);
+
+ void OnReceivedNTP() ;
+
+ // bw estimation
+ void OnPacketLossStatisticsUpdate(
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTime,
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum);
+
+ void OnReceivedTMMBR();
+
+ void OnReceivedEstimatedMaxBitrate(const WebRtc_UWord32 maxBitrate);
+
+ void OnReceivedBandwidthEstimateUpdate(const WebRtc_UWord16 bwEstimateKbit);
+
+ // bad state of RTP receiver request a keyframe
+ void OnRequestIntraFrame();
+
+ void OnReceivedIntraFrameRequest(const RtpRtcp* caller);
+
+ // received a request for a new SLI
+ void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID);
+
+ // received a new refereence frame
+ void OnReceivedReferencePictureSelectionIndication(
+ const WebRtc_UWord64 pitureID);
+
+ void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
+ const WebRtc_UWord16* nackSequenceNumbers);
+
+ void OnRequestSendReport();
+
+ // Following function is only called when constructing the object so no
+ // need to worry about data race.
+ void OwnsClock() { _owns_clock = true; }
+
+protected:
+ void RegisterChildModule(RtpRtcp* module);
+
+ void DeRegisterChildModule(RtpRtcp* module);
+
+ bool UpdateRTCPReceiveInformationTimers();
+
+ void ProcessDeadOrAliveTimer();
+
+ WebRtc_UWord32 BitrateReceivedNow() const;
+
+ // Get remote SequenceNumber
+ WebRtc_UWord16 RemoteSequenceNumber() const;
+
+ WebRtc_Word32 UpdateTMMBR();
+
+ // only for internal testing
+ WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
+
+ RTPSender _rtpSender;
+ RTPReceiver _rtpReceiver;
+
+ RTCPSender _rtcpSender;
+ RTCPReceiver _rtcpReceiver;
+
+ bool _owns_clock;
+ RtpRtcpClock& _clock;
+private:
+ void SendKeyFrame();
+ void ProcessDefaultModuleBandwidth();
+
+ WebRtc_Word32 _id;
+ const bool _audio;
+ bool _collisionDetected;
+ WebRtc_UWord32 _lastProcessTime;
+ WebRtc_UWord32 _lastBitrateProcessTime;
+ WebRtc_UWord32 _lastPacketTimeoutProcessTime;
+ WebRtc_UWord16 _packetOverHead;
+
+ CriticalSectionWrapper* _criticalSectionModulePtrs;
+ CriticalSectionWrapper* _criticalSectionModulePtrsFeedback;
+ ModuleRtpRtcpImpl* _defaultModule;
+ ModuleRtpRtcpImpl* _audioModule;
+ ModuleRtpRtcpImpl* _videoModule;
+ std::list<ModuleRtpRtcpImpl*> _childModules;
+
+ // Dead or alive
+ bool _deadOrAliveActive;
+ WebRtc_UWord32 _deadOrAliveTimeoutMS;
+ WebRtc_UWord32 _deadOrAliveLastTimer;
+
+ // receive side
+ BandwidthManagement _bandwidthManagement;
+
+ WebRtc_UWord32 _receivedNTPsecsAudio;
+ WebRtc_UWord32 _receivedNTPfracAudio;
+ WebRtc_UWord32 _RTCPArrivalTimeSecsAudio;
+ WebRtc_UWord32 _RTCPArrivalTimeFracAudio;
+
+ // send side
+ NACKMethod _nackMethod;
+ WebRtc_UWord32 _nackLastTimeSent;
+ WebRtc_UWord16 _nackLastSeqNumberSent;
+
+ bool _simulcast;
+ VideoCodec _sendVideoCodec;
+ KeyFrameRequestMethod _keyFrameReqMethod;
+
+#ifdef MATLAB
+ MatlabPlot* _plot1;
+#endif
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
new file mode 100644
index 0000000..ea705a8
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
@@ -0,0 +1,47 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'rtp_rtcp_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'rtp_rtcp',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../../../',
+ ],
+ 'sources': [
+ 'receiver_fec_unittest.cc',
+ 'rtp_fec_unittest.cc',
+ 'rtp_format_vp8_unittest.cc',
+ 'rtp_format_vp8_test_helper.cc',
+ 'rtp_format_vp8_test_helper.h',
+ 'rtcp_format_remb_unittest.cc',
+ 'rtp_packet_history_test.cc',
+ 'rtp_utility_test.cc',
+ 'rtp_header_extension_test.cc',
+ 'rtp_sender_test.cc',
+ 'rtcp_sender_test.cc',
+ 'transmission_bucket_test.cc',
+ 'vp8_partition_aggregator_unittest.cc',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc
new file mode 100644
index 0000000..374ce53
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -0,0 +1,1768 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdlib> // srand
+
+#include "rtp_sender.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#include "rtp_packet_history.h"
+#include "rtp_sender_audio.h"
+#include "rtp_sender_video.h"
+
+namespace webrtc {
+RTPSender::RTPSender(const WebRtc_Word32 id,
+ const bool audio,
+ RtpRtcpClock* clock) :
+ Bitrate(clock),
+ _id(id),
+ _audioConfigured(audio),
+ _audio(NULL),
+ _video(NULL),
+ _sendCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _transportCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+
+ _transport(NULL),
+
+ _sendingMedia(true), // Default to sending media
+
+ _maxPayloadLength(IP_PACKET_SIZE-28), // default is IP/UDP
+ _targetSendBitrate(0),
+ _packetOverHead(28),
+
+ _payloadType(-1),
+ _payloadTypeMap(),
+
+ _rtpHeaderExtensionMap(),
+ _transmissionTimeOffset(0),
+
+ _keepAliveIsActive(false),
+ _keepAlivePayloadType(-1),
+ _keepAliveLastSent(0),
+ _keepAliveDeltaTimeSend(0),
+
+ // NACK
+ _nackByteCountTimes(),
+ _nackByteCount(),
+ _nackBitrate(clock),
+
+ _packetHistory(new RTPPacketHistory(clock)),
+ _sendBucket(),
+ _timeLastSendToNetworkUpdate(clock->GetTimeInMS()),
+ _transmissionSmoothing(false),
+
+ // statistics
+ _packetsSent(0),
+ _payloadBytesSent(0),
+
+ // RTP variables
+ _startTimeStampForced(false),
+ _startTimeStamp(0),
+ _ssrcDB(*SSRCDatabase::GetSSRCDatabase()),
+ _remoteSSRC(0),
+ _sequenceNumberForced(false),
+ _sequenceNumber(0),
+ _sequenceNumberRTX(0),
+ _ssrcForced(false),
+ _ssrc(0),
+ _timeStamp(0),
+ _CSRCs(0),
+ _CSRC(),
+ _includeCSRCs(true),
+ _RTX(false),
+ _ssrcRTX(0)
+{
+ memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes));
+ memset(_nackByteCount, 0, sizeof(_nackByteCount));
+
+ memset(_CSRC, 0, sizeof(_CSRC));
+
+ // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
+ srand( (WebRtc_UWord32)_clock.GetTimeInMS() );
+
+ _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
+
+ if(audio)
+ {
+ _audio = new RTPSenderAudio(id, &_clock, this);
+ } else
+ {
+ _video = new RTPSenderVideo(id, &_clock, this);
+ }
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+}
+
+RTPSender::~RTPSender() {
+ if(_remoteSSRC != 0) {
+ _ssrcDB.ReturnSSRC(_remoteSSRC);
+ }
+ _ssrcDB.ReturnSSRC(_ssrc);
+
+ SSRCDatabase::ReturnSSRCDatabase();
+ delete _sendCritsect;
+ delete _transportCritsect;
+ while (!_payloadTypeMap.empty()) {
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.begin();
+ delete it->second;
+ _payloadTypeMap.erase(it);
+ }
+ delete _packetHistory;
+ delete _audio;
+ delete _video;
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_Word32
+RTPSender::Init(const WebRtc_UWord32 remoteSSRC)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+
+ // reset to default generation
+ _ssrcForced = false;
+ _startTimeStampForced = false;
+
+ // register a remote SSRC if we have it to avoid collisions
+ if(remoteSSRC != 0)
+ {
+ if(_ssrc == remoteSSRC)
+ {
+ // collision detected
+ _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
+ }
+ _remoteSSRC = remoteSSRC;
+ _ssrcDB.RegisterSSRC(remoteSSRC);
+ }
+ _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
+ _sequenceNumberRTX = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
+ _packetsSent = 0;
+ _payloadBytesSent = 0;
+ _packetOverHead = 28;
+
+ _keepAlivePayloadType = -1;
+
+ _rtpHeaderExtensionMap.Erase();
+
+ while (!_payloadTypeMap.empty()) {
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.begin();
+ delete it->second;
+ _payloadTypeMap.erase(it);
+ }
+
+ memset(_CSRC, 0, sizeof(_CSRC));
+
+ memset(_nackByteCount, 0, sizeof(_nackByteCount));
+ memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes));
+ _nackBitrate.Init();
+
+ SetStorePacketsStatus(false, 0);
+ _sendBucket.Reset();
+
+ Bitrate::Init();
+
+ if(_audioConfigured)
+ {
+ _audio->Init();
+ } else
+ {
+ _video->Init();
+ }
+ return(0);
+}
+
+void
+RTPSender::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ if(_audioConfigured)
+ {
+ _audio->ChangeUniqueId(id);
+ } else
+ {
+ _video->ChangeUniqueId(id);
+ }
+}
+
+WebRtc_Word32
+RTPSender::SetTargetSendBitrate(const WebRtc_UWord32 bits)
+{
+ _targetSendBitrate = (WebRtc_UWord16)(bits/1000);
+ return 0;
+}
+
+WebRtc_UWord16
+RTPSender::TargetSendBitrateKbit() const
+{
+ return _targetSendBitrate;
+}
+
+WebRtc_UWord16
+RTPSender::ActualSendBitrateKbit() const
+{
+ return (WebRtc_UWord16) (Bitrate::BitrateNow()/1000);
+}
+
+WebRtc_UWord32
+RTPSender::VideoBitrateSent() const {
+ if (_video)
+ return _video->VideoBitrateSent();
+ else
+ return 0;
+}
+
+WebRtc_UWord32
+RTPSender::FecOverheadRate() const {
+ if (_video)
+ return _video->FecOverheadRate();
+ else
+ return 0;
+}
+
+WebRtc_UWord32
+RTPSender::NackOverheadRate() const {
+ return _nackBitrate.BitrateLast();
+}
+
+WebRtc_Word32
+RTPSender::SetTransmissionTimeOffset(
+ const WebRtc_Word32 transmissionTimeOffset)
+{
+ if (transmissionTimeOffset > (0x800000 - 1) ||
+ transmissionTimeOffset < -(0x800000 - 1)) // Word24
+ {
+ return -1;
+ }
+ CriticalSectionScoped cs(_sendCritsect);
+ _transmissionTimeOffset = transmissionTimeOffset;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
+ const WebRtc_UWord8 id)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _rtpHeaderExtensionMap.Register(type, id);
+}
+
+WebRtc_Word32
+RTPSender::DeregisterRtpHeaderExtension(const RTPExtensionType type)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _rtpHeaderExtensionMap.Deregister(type);
+}
+
+WebRtc_UWord16
+RTPSender::RtpHeaderExtensionTotalLength() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _rtpHeaderExtensionMap.GetTotalLengthInBytes();
+}
+
+//can be called multiple times
+WebRtc_Word32 RTPSender::RegisterPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadNumber,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ assert(payloadName);
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if (payloadNumber == _keepAlivePayloadType) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "invalid state",
+ __FUNCTION__);
+ return -1;
+ }
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.find(payloadNumber);
+
+ if (_payloadTypeMap.end() != it) {
+ // we already use this payload type
+ ModuleRTPUtility::Payload* payload = it->second;
+ assert(payload);
+
+ // check if it's the same as we already have
+ if (ModuleRTPUtility::StringCompare(payload->name, payloadName,
+ RTP_PAYLOAD_NAME_SIZE - 1)) {
+ if (_audioConfigured && payload->audio &&
+ payload->typeSpecific.Audio.frequency == frequency &&
+ (payload->typeSpecific.Audio.rate == rate ||
+ payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
+ payload->typeSpecific.Audio.rate = rate;
+ // Ensure that we update the rate if new or old is zero
+ return 0;
+ }
+ if(!_audioConfigured && !payload->audio) {
+ return 0;
+ }
+ }
+ return -1;
+ }
+ WebRtc_Word32 retVal = -1;
+ ModuleRTPUtility::Payload* payload = NULL;
+ if (_audioConfigured) {
+ retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency,
+ channels, rate, payload);
+ } else {
+ retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate,
+ payload);
+ }
+ if(payload) {
+ _payloadTypeMap[payloadNumber] = payload;
+ }
+ return retVal;
+}
+
+WebRtc_Word32 RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType) {
+ CriticalSectionScoped lock(_sendCritsect);
+
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.find(payloadType);
+
+ if (_payloadTypeMap.end() == it) return -1;
+
+ ModuleRTPUtility::Payload* payload = it->second;
+ delete payload;
+ _payloadTypeMap.erase(it);
+ return 0;
+}
+
+WebRtc_Word8 RTPSender::SendPayloadType() const
+{
+ return _payloadType;
+}
+
+
+int RTPSender::SendPayloadFrequency() const
+{
+ return _audio->AudioFrequency();
+}
+
+
+// See http://www.ietf.org/internet-drafts/draft-ietf-avt-app-rtp-keepalive-04.txt
+// for details about this method. Only Section 4.6 is implemented so far.
+bool
+RTPSender::RTPKeepalive() const
+{
+ return _keepAliveIsActive;
+}
+
+WebRtc_Word32
+RTPSender::RTPKeepaliveStatus(bool* enable,
+ WebRtc_Word8* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if(enable)
+ {
+ *enable = _keepAliveIsActive;
+ }
+ if(unknownPayloadType)
+ {
+ *unknownPayloadType = _keepAlivePayloadType;
+ }
+ if(deltaTransmitTimeMS)
+ {
+ *deltaTransmitTimeMS =_keepAliveDeltaTimeSend;
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTPSender::EnableRTPKeepalive(
+ const WebRtc_Word8 unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS) {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.find(unknownPayloadType);
+
+ if (it != _payloadTypeMap.end()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument",
+ __FUNCTION__);
+ return -1;
+ }
+ _keepAliveIsActive = true;
+ _keepAlivePayloadType = unknownPayloadType;
+ _keepAliveLastSent = _clock.GetTimeInMS();
+ _keepAliveDeltaTimeSend = deltaTransmitTimeMS;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSender::DisableRTPKeepalive()
+{
+ _keepAliveIsActive = false;
+ return 0;
+}
+
+bool
+RTPSender::TimeToSendRTPKeepalive() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+
+ bool timeToSend(false);
+
+ WebRtc_UWord32 dT = _clock.GetTimeInMS() - _keepAliveLastSent;
+ if (dT > _keepAliveDeltaTimeSend)
+ {
+ timeToSend = true;
+ }
+ return timeToSend;
+}
+
+// ----------------------------------------------------------------------------
+// From the RFC draft:
+//
+// 4.6. RTP Packet with Unknown Payload Type
+//
+// The application sends an RTP packet of 0 length with a dynamic
+// payload type that has not been negotiated by the peers (e.g. not
+// negotiated within the SDP offer/answer, and thus not mapped to any
+// media format).
+//
+// The sequence number is incremented by one for each packet, as it is
+// sent within the same RTP session as the actual media. The timestamp
+// contains the same value a media packet would have at this time. The
+// marker bit is not significant for the keepalive packets and is thus
+// set to zero.
+//
+// Normally the peer will ignore this packet, as RTP [RFC3550] states
+// that "a receiver MUST ignore packets with payload types that it does
+// not understand".
+//
+// Cons:
+// o [RFC4566] and [RFC3264] mandate not to send media with inactive
+// and recvonly attributes, however this is mitigated as no real
+// media is sent with this mechanism.
+//
+// Recommendation:
+// o This method should be used for RTP keepalive.
+//
+// 7. Timing and Transport Considerations
+//
+// An application supporting this specification must transmit keepalive
+// packets every Tr seconds during the whole duration of the media
+// session. Tr SHOULD be configurable, and otherwise MUST default to 15
+// seconds.
+//
+// Keepalives packets within a particular RTP session MUST use the tuple
+// (source IP address, source TCP/UDP ports, target IP address, target
+// TCP/UDP Port) of the regular RTP packets.
+//
+// The agent SHOULD only send RTP keepalive when it does not send
+// regular RTP packets.
+//
+// http://www.ietf.org/internet-drafts/draft-ietf-avt-app-rtp-keepalive-04.txt
+// ----------------------------------------------------------------------------
+
+WebRtc_Word32
+RTPSender::SendRTPKeepalivePacket()
+{
+ // RFC summary:
+ //
+ // - Send an RTP packet of 0 length;
+ // - dynamic payload type has not been negotiated (not mapped to any media);
+ // - sequence number is incremented by one for each packet;
+ // - timestamp contains the same value a media packet would have at this time;
+ // - marker bit is set to zero.
+
+ WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
+ WebRtc_UWord16 rtpHeaderLength = 12;
+ {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+ WebRtc_UWord32 dT = now -_keepAliveLastSent; // delta time in MS
+
+ WebRtc_UWord32 freqKHz = 90; // video
+ if(_audioConfigured)
+ {
+ freqKHz = _audio->AudioFrequency()/1000;
+ }
+ WebRtc_UWord32 dSamples = dT*freqKHz;
+
+ // set timestamp
+ _timeStamp += dSamples;
+ _keepAliveLastSent = now;
+
+ rtpHeaderLength = RTPHeaderLength();
+
+ // correct seq num, time stamp and payloadtype
+ BuildRTPheader(dataBuffer, _keepAlivePayloadType, false, 0, false);
+ }
+
+ return SendToNetwork(dataBuffer, 0, rtpHeaderLength, kAllowRetransmission);
+}
+
+WebRtc_Word32
+RTPSender::SetMaxPayloadLength(const WebRtc_UWord16 maxPayloadLength, const WebRtc_UWord16 packetOverHead)
+{
+ // sanity check
+ if(maxPayloadLength < 100 || maxPayloadLength > IP_PACKET_SIZE)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_sendCritsect);
+ _maxPayloadLength = maxPayloadLength;
+ _packetOverHead = packetOverHead;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, _id, "SetMaxPayloadLength to %d.", maxPayloadLength);
+ return 0;
+}
+
+WebRtc_UWord16 RTPSender::MaxDataPayloadLength() const {
+ if(_audioConfigured) {
+ return _maxPayloadLength - RTPHeaderLength();
+ } else {
+ return _maxPayloadLength - RTPHeaderLength() -
+ _video->FECPacketOverhead() - ((_RTX) ? 2 : 0);
+ // Include the FEC/ULP/RED overhead.
+ }
+}
+
+WebRtc_UWord16
+RTPSender::MaxPayloadLength() const
+{
+ return _maxPayloadLength;
+}
+
+WebRtc_UWord16
+RTPSender::PacketOverHead() const
+{
+ return _packetOverHead;
+}
+
+void RTPSender::SetTransmissionSmoothingStatus(const bool enable) {
+ CriticalSectionScoped cs(_sendCritsect);
+ _transmissionSmoothing = enable;
+}
+
+bool RTPSender::TransmissionSmoothingStatus() const {
+ CriticalSectionScoped cs(_sendCritsect);
+ return _transmissionSmoothing;
+}
+
+void RTPSender::SetRTXStatus(const bool enable,
+ const bool setSSRC,
+ const WebRtc_UWord32 SSRC) {
+ CriticalSectionScoped cs(_sendCritsect);
+ _RTX = enable;
+ if (enable) {
+ if (setSSRC) {
+ _ssrcRTX = SSRC;
+ } else {
+ _ssrcRTX = _ssrcDB.CreateSSRC(); // can't be 0
+ }
+ }
+}
+
+void RTPSender::RTXStatus(bool* enable,
+ WebRtc_UWord32* SSRC) const {
+ CriticalSectionScoped cs(_sendCritsect);
+ *enable = _RTX;
+ *SSRC = _ssrcRTX;
+}
+
+WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
+ RtpVideoCodecTypes& videoType) {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if (payloadType < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "\tinvalid payloadType (%d)", payloadType);
+ return -1;
+ }
+ if (_audioConfigured) {
+ WebRtc_Word8 redPlType = -1;
+ if (_audio->RED(redPlType) == 0) {
+ // We have configured RED.
+ if(redPlType == payloadType) {
+ // And it's a match...
+ return 0;
+ }
+ }
+ }
+ if (_payloadType == payloadType) {
+ if (!_audioConfigured) {
+ videoType = _video->VideoCodecType();
+ }
+ return 0;
+ }
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
+ _payloadTypeMap.find(payloadType);
+ if (it == _payloadTypeMap.end()) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "\tpayloadType:%d not registered", payloadType);
+ return -1;
+ }
+ _payloadType = payloadType;
+ ModuleRTPUtility::Payload* payload = it->second;
+ assert(payload);
+ if (payload->audio) {
+ if (_audioConfigured) {
+ // Extract payload frequency
+ int payloadFreqHz;
+ if (ModuleRTPUtility::StringCompare(payload->name,"g722",4)&&
+ (payload->name[4] == 0)) {
+ //Check that strings end there, g722.1...
+ // Special case for G.722, bug in spec
+ payloadFreqHz=8000;
+ } else {
+ payloadFreqHz=payload->typeSpecific.Audio.frequency;
+ }
+
+ //we don't do anything if it's CN
+ if ((_audio->AudioFrequency() != payloadFreqHz)&&
+ (!ModuleRTPUtility::StringCompare(payload->name,"cn",2))) {
+ _audio->SetAudioFrequency(payloadFreqHz);
+ // We need to correct the timestamp again,
+ // since this might happen after we've set it
+ WebRtc_UWord32 RTPtime =
+ ModuleRTPUtility::GetCurrentRTP(&_clock, payloadFreqHz);
+ SetStartTimestamp(RTPtime);
+ // will be ignored if it's already configured via API
+ }
+ }
+ } else {
+ if(!_audioConfigured) {
+ _video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
+ videoType = payload->typeSpecific.Video.videoCodecType;
+ _video->SetMaxConfiguredBitrateVideo(
+ payload->typeSpecific.Video.maxRate);
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSender::SendOutgoingData(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ VideoCodecInformation* codecInfo,
+ const RTPVideoTypeHeader* rtpTypeHdr)
+{
+ {
+ // Drop this packet if we're not sending media packets
+ CriticalSectionScoped cs(_sendCritsect);
+ if (!_sendingMedia)
+ {
+ return 0;
+ }
+ }
+ RtpVideoCodecTypes videoType = kRtpNoVideo;
+ if(CheckPayloadType(payloadType, videoType) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument failed to find payloadType:%d", __FUNCTION__, payloadType);
+ return -1;
+ }
+ // update keepalive so that we don't trigger keepalive messages while sending data
+ _keepAliveLastSent = _clock.GetTimeInMS();
+
+ if(_audioConfigured)
+ {
+ // assert video frameTypes
+ assert(frameType == kAudioFrameSpeech ||
+ frameType == kAudioFrameCN ||
+ frameType == kFrameEmpty);
+
+ return _audio->SendAudio(frameType, payloadType, captureTimeStamp, payloadData, payloadSize,fragmentation);
+ } else
+ {
+ // assert audio frameTypes
+ assert(frameType == kVideoFrameKey ||
+ frameType == kVideoFrameDelta ||
+ frameType == kVideoFrameGolden ||
+ frameType == kVideoFrameAltRef);
+
+ return _video->SendVideo(videoType,
+ frameType,
+ payloadType,
+ captureTimeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation,
+ codecInfo,
+ rtpTypeHdr);
+ }
+}
+
+WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
+ WebRtc_UWord32 capture_timestamp,
+ WebRtc_Word32 bytes) {
+ // Drop this packet if we're not sending media packets
+ if (!_sendingMedia) {
+ return 0;
+ }
+ // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
+ int max_length = 224;
+ WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
+
+ for (; bytes > 0; bytes -= max_length) {
+ WebRtc_Word32 header_length;
+ {
+ // Correct seq num, timestamp and payload type.
+ header_length = BuildRTPheader(data_buffer,
+ payload_type,
+ false, // No markerbit.
+ capture_timestamp,
+ true, // Timestamp provided.
+ true); // Increment sequence number.
+ }
+ data_buffer[0] |= 0x20; // Set padding bit.
+ WebRtc_Word32* data =
+ reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
+
+ int padding_bytes_in_packet = max_length;
+ if (bytes < max_length) {
+ padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
+ }
+ if (padding_bytes_in_packet < 32) {
+ // Sanity don't send empty packets.
+ break;
+ }
+ // Fill data buffer with random data.
+ for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
+ data[j] = rand();
+ }
+ // Set number of padding bytes in the last byte of the packet.
+ data_buffer[header_length + padding_bytes_in_packet - 1] =
+ padding_bytes_in_packet;
+ // Send the packet
+ if (0 > SendToNetwork(data_buffer,
+ padding_bytes_in_packet,
+ header_length,
+ kDontRetransmit)) {
+ // Error sending the packet.
+ break;
+ }
+ }
+ if (bytes > 31) { // 31 due to our modulus 32.
+ // We did not manage to send all bytes.
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTPSender::SetStorePacketsStatus(
+ const bool enable,
+ const WebRtc_UWord16 numberToStore) {
+ _packetHistory->SetStorePacketsStatus(enable, numberToStore);
+ return 0;
+}
+
+bool RTPSender::StorePackets() const {
+ return _packetHistory->StorePackets();
+}
+
+WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packet_id,
+ WebRtc_UWord32 min_resend_time) {
+
+ WebRtc_UWord16 length = IP_PACKET_SIZE;
+ WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
+ WebRtc_UWord8* buffer_to_send_ptr = data_buffer;
+
+ WebRtc_UWord32 stored_time_in_ms;
+ StorageType type;
+ bool found = _packetHistory->GetRTPPacket(packet_id,
+ min_resend_time, data_buffer, &length, &stored_time_in_ms, &type);
+ if (!found) {
+ // Packet not found.
+ return -1;
+ }
+
+ if (length == 0 || type == kDontRetransmit) {
+ // No bytes copied (packet recently resent, skip resending) or
+ // packet should not be retransmitted.
+ return 0;
+ }
+
+ WebRtc_UWord8 data_buffer_rtx[IP_PACKET_SIZE];
+ if (_RTX) {
+ buffer_to_send_ptr = data_buffer_rtx;
+
+ CriticalSectionScoped cs(_sendCritsect);
+ // Add RTX header.
+ ModuleRTPUtility::RTPHeaderParser rtpParser(
+ reinterpret_cast<const WebRtc_UWord8*>(data_buffer),
+ length);
+
+ WebRtcRTPHeader rtp_header;
+ rtpParser.Parse(rtp_header);
+
+ // Add original RTP header.
+ memcpy(data_buffer_rtx, data_buffer, rtp_header.header.headerLength);
+
+ // Replace sequence number.
+ WebRtc_UWord8* ptr = data_buffer_rtx + 2;
+ ModuleRTPUtility::AssignUWord16ToBuffer(ptr, _sequenceNumberRTX++);
+
+ // Replace SSRC.
+ ptr += 6;
+ ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _ssrcRTX);
+
+ // Add OSN (original sequence number).
+ ptr = data_buffer_rtx + rtp_header.header.headerLength;
+ ModuleRTPUtility::AssignUWord16ToBuffer(
+ ptr, rtp_header.header.sequenceNumber);
+ ptr += 2;
+
+ // Add original payload data.
+ memcpy(ptr,
+ data_buffer + rtp_header.header.headerLength,
+ length - rtp_header.header.headerLength);
+ length += 2;
+ }
+
+ WebRtc_Word32 bytes_sent = ReSendToNetwork(buffer_to_send_ptr, length);
+ if (bytes_sent <= 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "Transport failed to resend packet_id %u", packet_id);
+ return -1;
+ }
+
+ // Store the time when the packet was last resent.
+ _packetHistory->UpdateResendTime(packet_id);
+
+ return bytes_sent;
+}
+
+WebRtc_Word32 RTPSender::ReSendToNetwork(const WebRtc_UWord8* packet,
+ const WebRtc_UWord32 size) {
+ WebRtc_Word32 bytes_sent = -1;
+ {
+ CriticalSectionScoped lock(_transportCritsect);
+ if (_transport) {
+ bytes_sent = _transport->SendPacket(_id, packet, size);
+ }
+ }
+
+ if (bytes_sent <= 0) {
+ return -1;
+ }
+
+ // Update send statistics
+ CriticalSectionScoped cs(_sendCritsect);
+ Bitrate::Update(bytes_sent);
+ _packetsSent++;
+ // We on purpose don't add to _payloadBytesSent since this is a
+ // re-transmit and not new payload data.
+ return bytes_sent;
+}
+
+int RTPSender::SelectiveRetransmissions() const {
+ if (!_video) return -1;
+ return _video->SelectiveRetransmissions();
+}
+
+int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
+ if (!_video) return -1;
+ return _video->SetSelectiveRetransmissions(settings);
+}
+
+void
+RTPSender::OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
+ const WebRtc_UWord16* nackSequenceNumbers,
+ const WebRtc_UWord16 avgRTT) {
+ const WebRtc_UWord32 now = _clock.GetTimeInMS();
+ WebRtc_UWord32 bytesReSent = 0;
+
+ // Enough bandwidth to send NACK?
+ if (!ProcessNACKBitRate(now)) {
+ WEBRTC_TRACE(kTraceStream,
+ kTraceRtpRtcp,
+ _id,
+ "NACK bitrate reached. Skip sending NACK response. Target %d",
+ TargetSendBitrateKbit());
+ return;
+ }
+
+ for (WebRtc_UWord16 i = 0; i < nackSequenceNumbersLength; ++i) {
+ const WebRtc_Word32 bytesSent = ReSendPacket(nackSequenceNumbers[i],
+ 5+avgRTT);
+ if (bytesSent > 0) {
+ bytesReSent += bytesSent;
+ } else if (bytesSent == 0) {
+ // The packet has previously been resent.
+ // Try resending next packet in the list.
+ continue;
+ } else if (bytesSent < 0) {
+ // Failed to send one Sequence number. Give up the rest in this nack.
+ WEBRTC_TRACE(kTraceWarning,
+ kTraceRtpRtcp,
+ _id,
+ "Failed resending RTP packet %d, Discard rest of packets",
+ nackSequenceNumbers[i]);
+ break;
+ }
+ // delay bandwidth estimate (RTT * BW)
+ if (TargetSendBitrateKbit() != 0 && avgRTT) {
+ // kbits/s * ms = bits => bits/8 = bytes
+ WebRtc_UWord32 targetBytes =
+ (static_cast<WebRtc_UWord32>(TargetSendBitrateKbit()) * avgRTT) >> 3;
+ if (bytesReSent > targetBytes) {
+ break; // ignore the rest of the packets in the list
+ }
+ }
+ }
+ if (bytesReSent > 0) {
+ // TODO(pwestin) consolidate these two methods.
+ UpdateNACKBitRate(bytesReSent, now);
+ _nackBitrate.Update(bytesReSent);
+ }
+}
+
+/**
+* @return true if the nack bitrate is lower than the requested max bitrate
+*/
+bool RTPSender::ProcessNACKBitRate(const WebRtc_UWord32 now) {
+ WebRtc_UWord32 num = 0;
+ WebRtc_Word32 byteCount = 0;
+ const WebRtc_UWord32 avgInterval=1000;
+
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if (_targetSendBitrate == 0) {
+ return true;
+ }
+ for (num = 0; num < NACK_BYTECOUNT_SIZE; num++) {
+ if ((now - _nackByteCountTimes[num]) > avgInterval) {
+ // don't use data older than 1sec
+ break;
+ } else {
+ byteCount += _nackByteCount[num];
+ }
+ }
+ WebRtc_Word32 timeInterval = avgInterval;
+ if (num == NACK_BYTECOUNT_SIZE) {
+ // More than NACK_BYTECOUNT_SIZE nack messages has been received
+ // during the last msgInterval
+ timeInterval = now - _nackByteCountTimes[num-1];
+ if(timeInterval < 0) {
+ timeInterval = avgInterval;
+ }
+ }
+ return (byteCount*8) < (_targetSendBitrate * timeInterval);
+}
+
+void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes,
+ const WebRtc_UWord32 now) {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ // save bitrate statistics
+ if(bytes > 0) {
+ if(now == 0) {
+ // add padding length
+ _nackByteCount[0] += bytes;
+ } else {
+ if(_nackByteCountTimes[0] == 0) {
+ // first no shift
+ } else {
+ // shift
+ for(int i = (NACK_BYTECOUNT_SIZE-2); i >= 0 ; i--) {
+ _nackByteCount[i+1] = _nackByteCount[i];
+ _nackByteCountTimes[i+1] = _nackByteCountTimes[i];
+ }
+ }
+ _nackByteCount[0] = bytes;
+ _nackByteCountTimes[0] = now;
+ }
+ }
+}
+
+void RTPSender::ProcessSendToNetwork() {
+
+ // triggered by timer
+ WebRtc_UWord32 delta_time_ms;
+ {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if (!_transmissionSmoothing) {
+ return;
+ }
+
+ WebRtc_UWord32 now = _clock.GetTimeInMS();
+ delta_time_ms = now - _timeLastSendToNetworkUpdate;
+ _timeLastSendToNetworkUpdate = now;
+ }
+
+ _sendBucket.UpdateBytesPerInterval(delta_time_ms, _targetSendBitrate);
+
+ while (!_sendBucket.Empty()) {
+
+ WebRtc_Word32 seq_num = _sendBucket.GetNextPacket();
+ if (seq_num < 0) {
+ break;
+ }
+
+ WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
+ WebRtc_UWord16 length = IP_PACKET_SIZE;
+ WebRtc_UWord32 stored_time_ms;
+ StorageType type;
+ bool found = _packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length,
+ &stored_time_ms, &type);
+ if (!found) {
+ assert(false);
+ return;
+ }
+ assert(length > 0);
+
+ WebRtc_UWord32 diff_ms = _clock.GetTimeInMS() - stored_time_ms;
+
+ ModuleRTPUtility::RTPHeaderParser rtpParser(data_buffer, length);
+ WebRtcRTPHeader rtp_header;
+ rtpParser.Parse(rtp_header);
+
+ UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms);
+
+ // Send packet
+ WebRtc_Word32 bytes_sent = -1;
+ {
+ CriticalSectionScoped cs(_transportCritsect);
+ if (_transport) {
+ bytes_sent = _transport->SendPacket(_id, data_buffer, length);
+ }
+ }
+
+ // Update send statistics
+ if (bytes_sent > 0) {
+ CriticalSectionScoped cs(_sendCritsect);
+ Bitrate::Update(bytes_sent);
+ _packetsSent++;
+ if (bytes_sent > rtp_header.header.headerLength) {
+ _payloadBytesSent += bytes_sent - rtp_header.header.headerLength;
+ }
+ }
+ }
+}
+
+WebRtc_Word32
+RTPSender::SendToNetwork(const WebRtc_UWord8* buffer,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord16 rtpLength,
+ const StorageType storage)
+{
+ // Used for NACK or to spead out the transmission of packets.
+ if (_packetHistory->PutRTPPacket(
+ buffer, rtpLength + length, _maxPayloadLength, storage) != 0) {
+ return -1;
+ }
+
+ if (_transmissionSmoothing) {
+ const WebRtc_UWord16 sequenceNumber = (buffer[2] << 8) + buffer[3];
+ _sendBucket.Fill(sequenceNumber, rtpLength + length);
+ // Packet will be sent at a later time.
+ return 0;
+ }
+
+ // Send packet
+ WebRtc_Word32 bytes_sent = -1;
+ {
+ CriticalSectionScoped cs(_transportCritsect);
+ if (_transport) {
+ bytes_sent = _transport->SendPacket(_id, buffer, length + rtpLength);
+ }
+ }
+
+ if (bytes_sent <= 0) {
+ return -1;
+ }
+
+ // Update send statistics
+ CriticalSectionScoped cs(_sendCritsect);
+ Bitrate::Update(bytes_sent);
+ _packetsSent++;
+ if (bytes_sent > rtpLength) {
+ _payloadBytesSent += bytes_sent - rtpLength;
+ }
+ return 0;
+}
+
+void
+RTPSender::ProcessBitrate()
+{
+ CriticalSectionScoped cs(_sendCritsect);
+
+ Bitrate::Process();
+ _nackBitrate.Process();
+
+ if (_audioConfigured)
+ return;
+ _video->ProcessBitrate();
+}
+
+WebRtc_UWord16
+RTPSender::RTPHeaderLength() const
+{
+ WebRtc_UWord16 rtpHeaderLength = 12;
+
+ if(_includeCSRCs)
+ {
+ rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
+ }
+ rtpHeaderLength += RtpHeaderExtensionTotalLength();
+
+ return rtpHeaderLength;
+}
+
+WebRtc_UWord16
+RTPSender::IncrementSequenceNumber()
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _sequenceNumber++;
+}
+
+WebRtc_Word32
+RTPSender::ResetDataCounters()
+{
+ _packetsSent = 0;
+ _payloadBytesSent = 0;
+
+ return 0;
+}
+
+// number of sent RTP packets
+// dont use critsect to avoid potental deadlock
+WebRtc_UWord32
+RTPSender::Packets() const
+{
+ return _packetsSent;
+}
+
+// number of sent RTP bytes
+// dont use critsect to avoid potental deadlock
+WebRtc_UWord32
+RTPSender::Bytes() const
+{
+ return _payloadBytesSent;
+}
+
+WebRtc_Word32
+RTPSender::BuildRTPheader(WebRtc_UWord8* dataBuffer,
+ const WebRtc_Word8 payloadType,
+ const bool markerBit,
+ const WebRtc_UWord32 captureTimeStamp,
+ const bool timeStampProvided,
+ const bool incSequenceNumber)
+{
+ assert(payloadType>=0);
+
+ CriticalSectionScoped cs(_sendCritsect);
+
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(payloadType);
+ if (markerBit)
+ {
+ dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is set
+ }
+
+ if(timeStampProvided)
+ {
+ _timeStamp = _startTimeStamp + captureTimeStamp;
+ } else
+ {
+ // make a unique time stamp
+ // used for inband signaling
+ // we can't inc by the actual time, since then we increase the risk of back timing
+ _timeStamp++;
+ }
+
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _sequenceNumber);
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, _timeStamp);
+ ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, _ssrc);
+
+ WebRtc_Word32 rtpHeaderLength = 12;
+
+ // Add the CSRCs if any
+ if (_includeCSRCs && _CSRCs > 0)
+ {
+ if(_CSRCs > kRtpCsrcSize)
+ {
+ // error
+ assert(false);
+ return -1;
+ }
+ WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
+ for (WebRtc_UWord32 i = 0; i < _CSRCs; ++i)
+ {
+ ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _CSRC[i]);
+ ptr +=4;
+ }
+ dataBuffer[0] = (dataBuffer[0]&0xf0) | _CSRCs;
+
+ // Update length of header
+ rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
+ }
+ {
+ _sequenceNumber++; // prepare for next packet
+ }
+
+ WebRtc_UWord16 len = BuildRTPHeaderExtension(dataBuffer + rtpHeaderLength);
+ if (len)
+ {
+ dataBuffer[0] |= 0x10; // set eXtension bit
+ rtpHeaderLength += len;
+ }
+
+ return rtpHeaderLength;
+}
+
+WebRtc_UWord16
+RTPSender::BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const
+{
+ if (_rtpHeaderExtensionMap.Size() <= 0) {
+ return 0;
+ }
+
+ /* RTP header extension, RFC 3550.
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | defined by profile | length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | header extension |
+ | .... |
+ */
+
+ const WebRtc_UWord32 kPosLength = 2;
+ const WebRtc_UWord32 kHeaderLength = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
+
+ // Add extension ID (0xBEDE).
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer,
+ RTP_ONE_BYTE_HEADER_EXTENSION);
+
+ // Add extensions.
+ WebRtc_UWord16 total_block_length = 0;
+
+ RTPExtensionType type = _rtpHeaderExtensionMap.First();
+ while (type != kRtpExtensionNone)
+ {
+ WebRtc_UWord8 block_length = 0;
+ if (type == kRtpExtensionTransmissionTimeOffset)
+ {
+ block_length = BuildTransmissionTimeOffsetExtension(
+ dataBuffer + kHeaderLength + total_block_length);
+ }
+ total_block_length += block_length;
+ type = _rtpHeaderExtensionMap.Next(type);
+ }
+
+ if (total_block_length == 0)
+ {
+ // No extension added.
+ return 0;
+ }
+
+ // Set header length (in number of Word32, header excluded).
+ assert(total_block_length % 4 == 0);
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + kPosLength,
+ total_block_length / 4);
+
+ // Total added length.
+ return kHeaderLength + total_block_length;
+}
+
+WebRtc_UWord8
+RTPSender::BuildTransmissionTimeOffsetExtension(WebRtc_UWord8* dataBuffer) const
+{
+ // From RFC 5450: Transmission Time Offsets in RTP Streams.
+ //
+ // The transmission time is signaled to the receiver in-band using the
+ // general mechanism for RTP header extensions [RFC5285]. The payload
+ // of this extension (the transmitted value) is a 24-bit signed integer.
+ // When added to the RTP timestamp of the packet, it represents the
+ // "effective" RTP transmission time of the packet, on the RTP
+ // timescale.
+ //
+ // The form of the transmission offset extension block:
+ //
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=2 | transmission offset |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ // Get id defined by user.
+ WebRtc_UWord8 id;
+ if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id)
+ != 0) {
+ // Not registered.
+ return 0;
+ }
+
+ int pos = 0;
+ const WebRtc_UWord8 len = 2;
+ dataBuffer[pos++] = (id << 4) + len;
+ ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + pos,
+ _transmissionTimeOffset);
+ pos += 3;
+ assert(pos == TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES);
+ return TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
+}
+
+void RTPSender::UpdateTransmissionTimeOffset(
+ WebRtc_UWord8* rtp_packet,
+ const WebRtc_UWord16 rtp_packet_length,
+ const WebRtcRTPHeader& rtp_header,
+ const WebRtc_UWord32 time_ms) const {
+ CriticalSectionScoped cs(_sendCritsect);
+
+ // Get length until start of transmission block.
+ int transmission_block_pos =
+ _rtpHeaderExtensionMap.GetLengthUntilBlockStartInBytes(
+ kRtpExtensionTransmissionTimeOffset);
+ if (transmission_block_pos < 0) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Failed to update transmission time offset, not registered.");
+ return;
+ }
+
+ int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos;
+ if ((rtp_packet_length < block_pos + 4)) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Failed to update transmission time offset, invalid length.");
+ return;
+ }
+
+ // Verify that header contains extension.
+ if (!((rtp_packet[12 + rtp_header.header.numCSRCs] == 0xBE) &&
+ (rtp_packet[12 + rtp_header.header.numCSRCs + 1] == 0xDE))) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Failed to update transmission time offset, hdr extension not found.");
+ return;
+ }
+
+ // Get id.
+ WebRtc_UWord8 id = 0;
+ if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset,
+ &id) != 0) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Failed to update transmission time offset, no id.");
+ return;
+ }
+
+ // Verify first byte in block.
+ const WebRtc_UWord8 first_block_byte = (id << 4) + 2;
+ if (rtp_packet[block_pos] != first_block_byte) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Failed to update transmission time offset.");
+ return;
+ }
+
+ // Update transmission offset field.
+ ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
+ time_ms * 90); // RTP timestamp
+}
+
+WebRtc_Word32
+RTPSender::RegisterSendTransport(Transport* transport)
+{
+ CriticalSectionScoped cs(_transportCritsect);
+ _transport = transport;
+ return 0;
+}
+
+void
+RTPSender::SetSendingStatus(const bool enabled)
+{
+ if(enabled)
+ {
+ WebRtc_UWord32 freq;
+ if(_audioConfigured)
+ {
+ WebRtc_UWord32 frequency = _audio->AudioFrequency();
+
+ // sanity
+ switch(frequency)
+ {
+ case 8000:
+ case 12000:
+ case 16000:
+ case 24000:
+ case 32000:
+ break;
+ default:
+ assert(false);
+ return;
+ }
+ freq = frequency;
+ } else
+ {
+ freq = 90000; // 90 KHz for all video
+ }
+ WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freq);
+
+ SetStartTimestamp(RTPtime); // will be ignored if it's already configured via API
+
+ } else
+ {
+ if(!_ssrcForced)
+ {
+ // generate a new SSRC
+ _ssrcDB.ReturnSSRC(_ssrc);
+ _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
+
+ }
+ if(!_sequenceNumberForced && !_ssrcForced) // don't initialize seq number if SSRC passed externally
+ {
+ // generate a new sequence number
+ _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
+ }
+ }
+}
+
+void
+RTPSender::SetSendingMediaStatus(const bool enabled)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ _sendingMedia = enabled;
+}
+
+bool
+RTPSender::SendingMedia() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _sendingMedia;
+}
+
+WebRtc_UWord32
+RTPSender::Timestamp() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _timeStamp;
+}
+
+
+WebRtc_Word32
+RTPSender::SetStartTimestamp( const WebRtc_UWord32 timestamp, const bool force)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ if(force)
+ {
+ _startTimeStampForced = force;
+ _startTimeStamp = timestamp;
+ } else
+ {
+ if(!_startTimeStampForced)
+ {
+ _startTimeStamp = timestamp;
+ }
+ }
+ return 0;
+}
+
+WebRtc_UWord32
+RTPSender::StartTimestamp() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _startTimeStamp;
+}
+
+WebRtc_UWord32
+RTPSender::GenerateNewSSRC()
+{
+ // if configured via API, return 0
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if(_ssrcForced)
+ {
+ return 0;
+ }
+ _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
+ return _ssrc;
+}
+
+WebRtc_Word32
+RTPSender::SetSSRC(WebRtc_UWord32 ssrc)
+{
+ // this is configured via the API
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if (_ssrc == ssrc && _ssrcForced)
+ {
+ return 0; // since it's same ssrc, don't reset anything
+ }
+
+ _ssrcForced = true;
+
+ _ssrcDB.ReturnSSRC(_ssrc);
+ _ssrcDB.RegisterSSRC(ssrc);
+ _ssrc = ssrc;
+
+ if(!_sequenceNumberForced)
+ {
+ _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
+ }
+ return 0;
+}
+
+WebRtc_UWord32
+RTPSender::SSRC() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _ssrc;
+}
+
+WebRtc_Word32
+RTPSender::SetCSRCStatus(const bool include)
+{
+ _includeCSRCs = include;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength)
+{
+ if(arrLength > kRtpCsrcSize)
+ {
+ assert(false);
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_sendCritsect);
+
+ for(int i = 0; i < arrLength;i++)
+ {
+ _CSRC[i] = arrOfCSRC[i];
+ }
+ _CSRCs = arrLength;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSender::CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+
+ if(arrOfCSRC == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ for(int i = 0; i < _CSRCs && i < kRtpCsrcSize;i++)
+ {
+ arrOfCSRC[i] = _CSRC[i];
+ }
+ return _CSRCs;
+}
+
+WebRtc_Word32
+RTPSender::SetSequenceNumber(WebRtc_UWord16 seq)
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ _sequenceNumberForced = true;
+ _sequenceNumber = seq;
+ return 0;
+}
+
+WebRtc_UWord16
+RTPSender::SequenceNumber() const
+{
+ CriticalSectionScoped cs(_sendCritsect);
+ return _sequenceNumber;
+}
+
+
+ /*
+ * Audio
+ */
+WebRtc_Word32
+RTPSender::RegisterAudioCallback(RtpAudioFeedback* messagesCallback)
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->RegisterAudioCallback(messagesCallback);
+}
+
+ // Send a DTMF tone, RFC 2833 (4733)
+WebRtc_Word32
+RTPSender::SendTelephoneEvent(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level)
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->SendTelephoneEvent(key, time_ms, level);
+}
+
+bool
+RTPSender::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
+{
+ if(!_audioConfigured)
+ {
+ return false;
+ }
+ return _audio->SendTelephoneEventActive(telephoneEvent);
+}
+
+ // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+WebRtc_Word32
+RTPSender::SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples)
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->SetAudioPacketSize(packetSizeSamples);
+}
+
+WebRtc_Word32
+RTPSender::SetAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID)
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->SetAudioLevelIndicationStatus(enable, ID);
+}
+
+WebRtc_Word32
+RTPSender::AudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const
+{
+ return _audio->AudioLevelIndicationStatus(enable, ID);
+}
+
+WebRtc_Word32
+RTPSender::SetAudioLevel(const WebRtc_UWord8 level_dBov)
+{
+ return _audio->SetAudioLevel(level_dBov);
+}
+
+ // Set payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32
+RTPSender::SetRED(const WebRtc_Word8 payloadType)
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->SetRED(payloadType);
+}
+
+ // Get payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32
+RTPSender::RED(WebRtc_Word8& payloadType) const
+{
+ if(!_audioConfigured)
+ {
+ return -1;
+ }
+ return _audio->RED(payloadType);
+}
+
+ /*
+ * Video
+ */
+VideoCodecInformation*
+RTPSender::CodecInformationVideo()
+{
+ if(_audioConfigured)
+ {
+ return NULL;
+ }
+ return _video->CodecInformationVideo();
+}
+
+RtpVideoCodecTypes
+RTPSender::VideoCodecType() const
+{
+ if(_audioConfigured)
+ {
+ return kRtpNoVideo;
+ }
+ return _video->VideoCodecType();
+}
+
+WebRtc_UWord32
+RTPSender::MaxConfiguredBitrateVideo() const
+{
+ if(_audioConfigured)
+ {
+ return 0;
+ }
+ return _video->MaxConfiguredBitrateVideo();
+}
+
+WebRtc_Word32
+RTPSender::SendRTPIntraRequest()
+{
+ if(_audioConfigured)
+ {
+ return -1;
+ }
+ return _video->SendRTPIntraRequest();
+}
+
+// FEC
+WebRtc_Word32
+RTPSender::SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC)
+{
+ if(_audioConfigured)
+ {
+ return -1;
+ }
+ return _video->SetGenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
+}
+
+WebRtc_Word32
+RTPSender::GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) const
+{
+ if(_audioConfigured)
+ {
+ return -1;
+ }
+ return _video->GenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
+}
+
+WebRtc_Word32
+RTPSender::SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate)
+{
+ if(_audioConfigured)
+ {
+ return -1;
+ }
+ return _video->SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
+}
+
+WebRtc_Word32
+RTPSender::SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection)
+
+{
+ if(_audioConfigured)
+ {
+ return -1;
+ }
+ return _video->SetFECUepProtection(keyUseUepProtection,
+ deltaUseUepProtection);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender.h
new file mode 100644
index 0000000..00012e1
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -0,0 +1,376 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
+
+#include <cassert>
+#include <cmath>
+#include <map>
+
+#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
+#include "rtp_rtcp_defines.h"
+#include "common_types.h" // Encryption
+#include "ssrc_database.h"
+#include "Bitrate.h"
+#include "rtp_header_extension.h"
+#include "video_codec_information.h"
+#include "transmission_bucket.h"
+
+#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class RTPPacketHistory;
+class RTPSenderAudio;
+class RTPSenderVideo;
+
+class RTPSenderInterface
+{
+public:
+ RTPSenderInterface() {}
+ virtual ~RTPSenderInterface() {}
+
+ virtual WebRtc_UWord32 SSRC() const = 0;
+ virtual WebRtc_UWord32 Timestamp() const = 0;
+
+ virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
+ const WebRtc_Word8 payloadType,
+ const bool markerBit,
+ const WebRtc_UWord32 captureTimeStamp,
+ const bool timeStampProvided = true,
+ const bool incSequenceNumber = true) = 0;
+
+ virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
+ virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
+ virtual WebRtc_UWord16 SequenceNumber() const = 0;
+ virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
+ virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
+ virtual WebRtc_UWord16 PacketOverHead() const = 0;
+ virtual WebRtc_UWord16 TargetSendBitrateKbit() const = 0;
+ virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
+
+ virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 payloadLength,
+ const WebRtc_UWord16 rtpHeaderLength,
+ const StorageType storage) = 0;
+};
+
+class RTPSender : public Bitrate, public RTPSenderInterface
+{
+public:
+ RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock);
+ virtual ~RTPSender();
+
+ WebRtc_Word32 Init(const WebRtc_UWord32 remoteSSRC);
+ void ChangeUniqueId(const WebRtc_Word32 id);
+
+ void ProcessBitrate();
+ void ProcessSendToNetwork();
+
+ WebRtc_UWord16 TargetSendBitrateKbit() const;
+ WebRtc_UWord16 ActualSendBitrateKbit() const;
+
+ WebRtc_UWord32 VideoBitrateSent() const;
+ WebRtc_UWord32 FecOverheadRate() const;
+ WebRtc_UWord32 NackOverheadRate() const;
+
+ WebRtc_Word32 SetTargetSendBitrate(const WebRtc_UWord32 bits);
+
+ WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
+
+ // callback
+ WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
+
+ WebRtc_Word32 RegisterPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
+
+ WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
+
+ WebRtc_Word8 SendPayloadType() const;
+
+ int SendPayloadFrequency() const;
+
+ void SetSendingStatus(const bool enabled);
+
+ void SetSendingMediaStatus(const bool enabled);
+ bool SendingMedia() const;
+
+ // number of sent RTP packets
+ WebRtc_UWord32 Packets() const;
+
+ // number of sent RTP bytes
+ WebRtc_UWord32 Bytes() const;
+
+ WebRtc_Word32 ResetDataCounters();
+
+ WebRtc_UWord32 StartTimestamp() const;
+ WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp,
+ const bool force = false);
+
+ WebRtc_UWord32 GenerateNewSSRC();
+ WebRtc_Word32 SetSSRC( const WebRtc_UWord32 ssrc);
+
+ WebRtc_UWord16 SequenceNumber() const;
+ WebRtc_Word32 SetSequenceNumber( WebRtc_UWord16 seq);
+
+ WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
+
+ WebRtc_Word32 SetCSRCStatus(const bool include);
+
+ WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
+ const WebRtc_UWord8 arrLength);
+
+ WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
+ const WebRtc_UWord16 packetOverHead);
+
+ WebRtc_Word32 SendOutgoingData(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ VideoCodecInformation* codecInfo = NULL,
+ const RTPVideoTypeHeader* rtpTypeHdr = NULL);
+
+ WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
+ WebRtc_UWord32 capture_timestamp,
+ WebRtc_Word32 bytes);
+ /*
+ * RTP header extension
+ */
+ WebRtc_Word32 SetTransmissionTimeOffset(
+ const WebRtc_Word32 transmissionTimeOffset);
+
+ WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
+ const WebRtc_UWord8 id);
+
+ WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
+
+ WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
+
+ WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
+
+ WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
+ WebRtc_UWord8* dataBuffer) const;
+
+ void UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
+ const WebRtc_UWord16 rtp_packet_length,
+ const WebRtcRTPHeader& rtp_header,
+ const WebRtc_UWord32 time_ms) const;
+
+ void SetTransmissionSmoothingStatus(const bool enable);
+
+ bool TransmissionSmoothingStatus() const;
+
+ /*
+ * NACK
+ */
+ int SelectiveRetransmissions() const;
+ int SetSelectiveRetransmissions(uint8_t settings);
+ void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
+ const WebRtc_UWord16* nackSequenceNumbers,
+ const WebRtc_UWord16 avgRTT);
+
+ WebRtc_Word32 SetStorePacketsStatus(const bool enable,
+ const WebRtc_UWord16 numberToStore);
+
+ bool StorePackets() const;
+
+ WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
+ WebRtc_UWord32 min_resend_time = 0);
+
+ WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
+ const WebRtc_UWord32 size);
+
+ bool ProcessNACKBitRate(const WebRtc_UWord32 now);
+
+ /*
+ * Keep alive
+ */
+ WebRtc_Word32 EnableRTPKeepalive( const WebRtc_Word8 unknownPayloadType,
+ const WebRtc_UWord16 deltaTransmitTimeMS);
+
+ WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
+ WebRtc_Word8* unknownPayloadType,
+ WebRtc_UWord16* deltaTransmitTimeMS) const;
+
+ WebRtc_Word32 DisableRTPKeepalive();
+
+ bool RTPKeepalive() const;
+
+ bool TimeToSendRTPKeepalive() const;
+
+ WebRtc_Word32 SendRTPKeepalivePacket();
+
+ /*
+ * RTX
+ */
+ void SetRTXStatus(const bool enable,
+ const bool setSSRC,
+ const WebRtc_UWord32 SSRC);
+
+ void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
+
+ /*
+ * Functions wrapping RTPSenderInterface
+ */
+ virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
+ const WebRtc_Word8 payloadType,
+ const bool markerBit,
+ const WebRtc_UWord32 captureTimeStamp,
+ const bool timeStampProvided = true,
+ const bool incSequenceNumber = true);
+
+ virtual WebRtc_UWord16 RTPHeaderLength() const ;
+ virtual WebRtc_UWord16 IncrementSequenceNumber();
+ virtual WebRtc_UWord16 MaxPayloadLength() const;
+ virtual WebRtc_UWord16 PacketOverHead() const;
+
+ // current timestamp
+ virtual WebRtc_UWord32 Timestamp() const;
+ virtual WebRtc_UWord32 SSRC() const;
+
+ virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 payloadLength,
+ const WebRtc_UWord16 rtpHeaderLength,
+ const StorageType storage);
+
+ /*
+ * Audio
+ */
+ WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
+
+ // Send a DTMF tone using RFC 2833 (4733)
+ WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level);
+
+ bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
+
+ // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+ WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
+
+ // Set status and ID for header-extension-for-audio-level-indication.
+ WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID);
+
+ // Get status and ID for header-extension-for-audio-level-indication.
+ WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const;
+
+ // Store the audio level in dBov for header-extension-for-audio-level-indication.
+ WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
+
+ // Set payload type for Redundant Audio Data RFC 2198
+ WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
+
+ // Get payload type for Redundant Audio Data RFC 2198
+ WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
+
+ /*
+ * Video
+ */
+ VideoCodecInformation* CodecInformationVideo();
+
+ RtpVideoCodecTypes VideoCodecType() const;
+
+ WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
+
+ WebRtc_Word32 SendRTPIntraRequest();
+
+ // FEC
+ WebRtc_Word32 SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC);
+
+ WebRtc_Word32 GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) const;
+
+ WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate);
+
+ WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection);
+
+protected:
+ WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
+ RtpVideoCodecTypes& videoType);
+
+private:
+ void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
+ const WebRtc_UWord32 now);
+
+ WebRtc_Word32 _id;
+ const bool _audioConfigured;
+ RTPSenderAudio* _audio;
+ RTPSenderVideo* _video;
+
+ CriticalSectionWrapper* _sendCritsect;
+
+ CriticalSectionWrapper* _transportCritsect;
+ Transport* _transport;
+
+ bool _sendingMedia;
+
+ WebRtc_UWord16 _maxPayloadLength;
+ WebRtc_UWord16 _targetSendBitrate;
+ WebRtc_UWord16 _packetOverHead;
+
+ WebRtc_Word8 _payloadType;
+ std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
+
+ RtpHeaderExtensionMap _rtpHeaderExtensionMap;
+ WebRtc_Word32 _transmissionTimeOffset;
+
+ bool _keepAliveIsActive;
+ WebRtc_Word8 _keepAlivePayloadType;
+ WebRtc_UWord32 _keepAliveLastSent;
+ WebRtc_UWord16 _keepAliveDeltaTimeSend;
+
+ // NACK
+ WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
+ WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
+ Bitrate _nackBitrate;
+
+ RTPPacketHistory* _packetHistory;
+ TransmissionBucket _sendBucket;
+ WebRtc_UWord32 _timeLastSendToNetworkUpdate;
+ bool _transmissionSmoothing;
+
+ // statistics
+ WebRtc_UWord32 _packetsSent;
+ WebRtc_UWord32 _payloadBytesSent;
+
+ // RTP variables
+ bool _startTimeStampForced;
+ WebRtc_UWord32 _startTimeStamp;
+ SSRCDatabase& _ssrcDB;
+ WebRtc_UWord32 _remoteSSRC;
+ bool _sequenceNumberForced;
+ WebRtc_UWord16 _sequenceNumber;
+ WebRtc_UWord16 _sequenceNumberRTX;
+ bool _ssrcForced;
+ WebRtc_UWord32 _ssrc;
+ WebRtc_UWord32 _timeStamp;
+ WebRtc_UWord8 _CSRCs;
+ WebRtc_UWord32 _CSRC[kRtpCsrcSize];
+ bool _includeCSRCs;
+ bool _RTX;
+ WebRtc_UWord32 _ssrcRTX;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
new file mode 100644
index 0000000..ee3c414
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -0,0 +1,627 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_sender_audio.h"
+
+#include <string.h> //memcpy
+#include <cassert> //assert
+
+namespace webrtc {
+RTPSenderAudio::RTPSenderAudio(const WebRtc_Word32 id, RtpRtcpClock* clock,
+ RTPSenderInterface* rtpSender) :
+ _id(id),
+ _clock(*clock),
+ _rtpSender(rtpSender),
+ _audioFeedbackCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _audioFeedback(NULL),
+ _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _frequency(8000),
+ _packetSizeSamples(160),
+ _dtmfEventIsOn(false),
+ _dtmfEventFirstPacketSent(false),
+ _dtmfPayloadType(-1),
+ _dtmfTimestamp(0),
+ _dtmfKey(0),
+ _dtmfLengthSamples(0),
+ _dtmfLevel(0),
+ _dtmfTimeLastSent(0),
+ _dtmfTimestampLastSent(0),
+ _REDPayloadType(-1),
+ _inbandVADactive(false),
+ _cngNBPayloadType(-1),
+ _cngWBPayloadType(-1),
+ _cngSWBPayloadType(-1),
+ _lastPayloadType(-1),
+ _includeAudioLevelIndication(false), // @TODO - reset at Init()?
+ _audioLevelIndicationID(0),
+ _audioLevel_dBov(0) {
+};
+
+RTPSenderAudio::~RTPSenderAudio()
+{
+ delete _sendAudioCritsect;
+ delete _audioFeedbackCritsect;
+}
+
+WebRtc_Word32
+RTPSenderAudio::Init()
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ _dtmfPayloadType = -1;
+ _inbandVADactive = false;
+ _cngNBPayloadType = -1;
+ _cngWBPayloadType = -1;
+ _cngSWBPayloadType = -1;
+ _lastPayloadType = -1;
+ _REDPayloadType = -1;
+ _dtmfTimeLastSent = 0;
+ _dtmfTimestampLastSent = 0;
+ ResetDTMF();
+ return 0;
+}
+
+void
+RTPSenderAudio::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+WebRtc_Word32
+RTPSenderAudio::RegisterAudioCallback(RtpAudioFeedback* messagesCallback)
+{
+ CriticalSectionScoped cs(_audioFeedbackCritsect);
+ _audioFeedback = messagesCallback;
+ return 0;
+}
+
+void
+RTPSenderAudio::SetAudioFrequency(const WebRtc_UWord32 f)
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+ _frequency = f;
+}
+
+int
+RTPSenderAudio::AudioFrequency() const
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+ return _frequency;
+}
+
+ // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+WebRtc_Word32
+RTPSenderAudio::SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples)
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ _packetSizeSamples = packetSizeSamples;
+ return 0;
+}
+
+WebRtc_Word32 RTPSenderAudio::RegisterAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ ModuleRTPUtility::Payload*& payload) {
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
+ // we can have multiple CNG payload types
+ if (frequency == 8000) {
+ _cngNBPayloadType = payloadType;
+
+ } else if (frequency == 16000) {
+ _cngWBPayloadType = payloadType;
+
+ } else if (frequency == 32000) {
+ _cngSWBPayloadType = payloadType;
+ } else {
+ return -1;
+ }
+ }
+ if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
+ // Don't add it to the list
+ // we dont want to allow send with a DTMF payloadtype
+ _dtmfPayloadType = payloadType;
+ return 0;
+ // The default timestamp rate is 8000 Hz, but other rates may be defined.
+ }
+ payload = new ModuleRTPUtility::Payload;
+ payload->typeSpecific.Audio.frequency = frequency;
+ payload->typeSpecific.Audio.channels = channels;
+ payload->typeSpecific.Audio.rate = rate;
+ payload->audio = true;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ return 0;
+}
+
+bool
+RTPSenderAudio::MarkerBit(const FrameType frameType,
+ const WebRtc_Word8 payloadType)
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ // for audio true for first packet in a speech burst
+ bool markerBit = false;
+ if(_lastPayloadType != payloadType)
+ {
+ if(_cngNBPayloadType != -1)
+ {
+ // we have configured NB CNG
+ if(_cngNBPayloadType == payloadType)
+ {
+ // only set a marker bit when we change payload type to a non CNG
+ return false;
+ }
+ }
+ if(_cngWBPayloadType != -1)
+ {
+ // we have configured WB CNG
+ if(_cngWBPayloadType == payloadType)
+ {
+ // only set a marker bit when we change payload type to a non CNG
+ return false;
+ }
+ }
+ if(_cngSWBPayloadType != -1)
+ {
+ // we have configured SWB CNG
+ if(_cngSWBPayloadType == payloadType)
+ {
+ // only set a marker bit when we change payload type to a non CNG
+ return false;
+ }
+ }
+ // payloadType differ
+ if(_lastPayloadType == -1)
+ {
+ if(frameType != kAudioFrameCN)
+ {
+ // first packet and NOT CNG
+ return true;
+
+ }else
+ {
+ // first packet and CNG
+ _inbandVADactive = true;
+ return false;
+ }
+ }
+ // not first packet AND
+ // not CNG AND
+ // payloadType changed
+
+ // set a marker bit when we change payload type
+ markerBit = true;
+ }
+
+ // For G.723 G.729, AMR etc we can have inband VAD
+ if(frameType == kAudioFrameCN)
+ {
+ _inbandVADactive = true;
+
+ } else if(_inbandVADactive)
+ {
+ _inbandVADactive = false;
+ markerBit = true;
+ }
+ return markerBit;
+}
+
+bool
+RTPSenderAudio::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
+{
+ if(_dtmfEventIsOn)
+ {
+ telephoneEvent = _dtmfKey;
+ return true;
+ }
+ WebRtc_UWord32 delaySinceLastDTMF = (_clock.GetTimeInMS() - _dtmfTimeLastSent);
+ if(delaySinceLastDTMF < 100)
+ {
+ telephoneEvent = _dtmfKey;
+ return true;
+ }
+ telephoneEvent = -1;
+ return false;
+}
+
+WebRtc_Word32 RTPSenderAudio::SendAudio(
+ const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 dataSize,
+ const RTPFragmentationHeader* fragmentation) {
+ // TODO(pwestin) Breakup function in smaller functions.
+ WebRtc_UWord16 payloadSize = static_cast<WebRtc_UWord16>(dataSize);
+ WebRtc_UWord16 maxPayloadLength = _rtpSender->MaxPayloadLength();
+ bool dtmfToneStarted = false;
+ WebRtc_UWord16 dtmfLengthMS = 0;
+ WebRtc_UWord8 key = 0;
+
+ // Check if we have pending DTMFs to send
+ if (!_dtmfEventIsOn && PendingDTMF()) {
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ WebRtc_UWord32 delaySinceLastDTMF = _clock.GetTimeInMS() -
+ _dtmfTimeLastSent;
+
+ if (delaySinceLastDTMF > 100) {
+ // New tone to play
+ _dtmfTimestamp = captureTimeStamp;
+ if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0) {
+ _dtmfEventFirstPacketSent = false;
+ _dtmfKey = key;
+ _dtmfLengthSamples = (_frequency / 1000) * dtmfLengthMS;
+ dtmfToneStarted = true;
+ _dtmfEventIsOn = true;
+ }
+ }
+ }
+ if (dtmfToneStarted) {
+ CriticalSectionScoped cs(_audioFeedbackCritsect);
+ if (_audioFeedback) {
+ _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
+ }
+ }
+
+ // A source MAY send events and coded audio packets for the same time
+ // but we don't support it
+ {
+ _sendAudioCritsect->Enter();
+
+ if (_dtmfEventIsOn) {
+ if (frameType == kFrameEmpty) {
+ // kFrameEmpty is used to drive the DTMF when in CN mode
+ // it can be triggered more frequently than we want to send the
+ // DTMF packets.
+ if (_packetSizeSamples > (captureTimeStamp - _dtmfTimestampLastSent)) {
+ // not time to send yet
+ _sendAudioCritsect->Leave();
+ return 0;
+ }
+ }
+ _dtmfTimestampLastSent = captureTimeStamp;
+ WebRtc_UWord32 dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
+ bool ended = false;
+ bool send = true;
+
+ if (_dtmfLengthSamples > dtmfDurationSamples) {
+ if (dtmfDurationSamples <= 0) {
+ // Skip send packet at start, since we shouldn't use duration 0
+ send = false;
+ }
+ } else {
+ ended = true;
+ _dtmfEventIsOn = false;
+ _dtmfTimeLastSent = _clock.GetTimeInMS();
+ }
+ // don't hold the critsect while calling SendTelephoneEventPacket
+ _sendAudioCritsect->Leave();
+ if (send) {
+ if (dtmfDurationSamples > 0xffff) {
+ // RFC 4733 2.5.2.3 Long-Duration Events
+ SendTelephoneEventPacket(ended, _dtmfTimestamp,
+ static_cast<WebRtc_UWord16>(0xffff), false);
+
+ // set new timestap for this segment
+ _dtmfTimestamp = captureTimeStamp;
+ dtmfDurationSamples -= 0xffff;
+ _dtmfLengthSamples -= 0xffff;
+
+ return SendTelephoneEventPacket(
+ ended,
+ _dtmfTimestamp,
+ static_cast<WebRtc_UWord16>(dtmfDurationSamples),
+ false);
+ } else {
+ // set markerBit on the first packet in the burst
+ _dtmfEventFirstPacketSent = true;
+ return SendTelephoneEventPacket(
+ ended,
+ _dtmfTimestamp,
+ static_cast<WebRtc_UWord16>(dtmfDurationSamples),
+ !_dtmfEventFirstPacketSent);
+ }
+ }
+ return 0;
+ }
+ _sendAudioCritsect->Leave();
+ }
+ if (payloadSize == 0 || payloadData == NULL) {
+ if (frameType == kFrameEmpty) {
+ // we don't send empty audio RTP packets
+ // no error since we use it to drive DTMF when we use VAD
+ return 0;
+ }
+ return -1;
+ }
+ WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
+ bool markerBit = MarkerBit(frameType, payloadType);
+
+ WebRtc_Word32 rtpHeaderLength = 0;
+ WebRtc_UWord16 timestampOffset = 0;
+
+ if (_REDPayloadType >= 0 && fragmentation && !markerBit &&
+ fragmentation->fragmentationVectorSize > 1) {
+ // have we configured RED? use its payload type
+ // we need to get the current timestamp to calc the diff
+ WebRtc_UWord32 oldTimeStamp = _rtpSender->Timestamp();
+ rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType,
+ markerBit, captureTimeStamp);
+
+ timestampOffset = WebRtc_UWord16(_rtpSender->Timestamp() - oldTimeStamp);
+ } else {
+ rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType,
+ markerBit, captureTimeStamp);
+ }
+ if (rtpHeaderLength <= 0) {
+ return -1;
+ }
+ {
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+ if (_includeAudioLevelIndication) {
+ dataBuffer[0] |= 0x10; // set eXtension bit
+ /*
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | 0xBE | 0xDE | length=1 |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | ID | len=0 |V| level | 0x00 | 0x00 |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+ // add our ID (0xBEDE)
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
+ RTP_AUDIO_LEVEL_UNIQUE_ID);
+ rtpHeaderLength += 2;
+
+ // add the length (length=1) in number of word32
+ const WebRtc_UWord8 length = 1;
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
+ length);
+ rtpHeaderLength += 2;
+
+ // add ID (defined by the user) and len(=0) byte
+ const WebRtc_UWord8 id = _audioLevelIndicationID;
+ const WebRtc_UWord8 len = 0;
+ dataBuffer[rtpHeaderLength++] = (id << 4) + len;
+
+ // add voice-activity flag (V) bit and the audio level (in dBov)
+ const WebRtc_UWord8 V = (frameType == kAudioFrameSpeech);
+ WebRtc_UWord8 level = _audioLevel_dBov;
+ dataBuffer[rtpHeaderLength++] = (V << 7) + level;
+
+ // add two bytes zero padding
+ ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, 0);
+ rtpHeaderLength += 2;
+ }
+
+ if(maxPayloadLength < rtpHeaderLength + payloadSize ) {
+ // too large payload buffer
+ return -1;
+ }
+
+ if (_REDPayloadType >= 0 && // Have we configured RED?
+ fragmentation &&
+ fragmentation->fragmentationVectorSize > 1 &&
+ !markerBit) {
+ if (timestampOffset <= 0x3fff) {
+ if(fragmentation->fragmentationVectorSize != 2) {
+ // we only support 2 codecs when using RED
+ return -1;
+ }
+ // only 0x80 if we have multiple blocks
+ dataBuffer[rtpHeaderLength++] = 0x80 +
+ fragmentation->fragmentationPlType[1];
+ WebRtc_UWord32 blockLength = fragmentation->fragmentationLength[1];
+
+ // sanity blockLength
+ if(blockLength > 0x3ff) { // block length 10 bits 1023 bytes
+ return -1;
+ }
+ WebRtc_UWord32 REDheader = (timestampOffset << 10) + blockLength;
+ ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength,
+ REDheader);
+ rtpHeaderLength += 3;
+
+ dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+ // copy the RED data
+ memcpy(dataBuffer+rtpHeaderLength,
+ payloadData + fragmentation->fragmentationOffset[1],
+ fragmentation->fragmentationLength[1]);
+
+ // copy the normal data
+ memcpy(dataBuffer+rtpHeaderLength +
+ fragmentation->fragmentationLength[1],
+ payloadData + fragmentation->fragmentationOffset[0],
+ fragmentation->fragmentationLength[0]);
+
+ payloadSize = static_cast<WebRtc_UWord16>(
+ fragmentation->fragmentationLength[0] +
+ fragmentation->fragmentationLength[1]);
+ } else {
+ // silence for too long send only new data
+ dataBuffer[rtpHeaderLength++] = static_cast<WebRtc_UWord8>(payloadType);
+ memcpy(dataBuffer+rtpHeaderLength,
+ payloadData + fragmentation->fragmentationOffset[0],
+ fragmentation->fragmentationLength[0]);
+
+ payloadSize = static_cast<WebRtc_UWord16>(
+ fragmentation->fragmentationLength[0]);
+ }
+ } else {
+ if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
+ // use the fragment info if we have one
+ memcpy( dataBuffer+rtpHeaderLength,
+ payloadData + fragmentation->fragmentationOffset[0],
+ fragmentation->fragmentationLength[0]);
+
+ payloadSize = static_cast<WebRtc_UWord16>(
+ fragmentation->fragmentationLength[0]);
+ } else {
+ memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
+ }
+ }
+ _lastPayloadType = payloadType;
+ } // end critical section
+ return _rtpSender->SendToNetwork(dataBuffer,
+ payloadSize,
+ static_cast<WebRtc_UWord16>(rtpHeaderLength),
+ kAllowRetransmission);
+}
+
+WebRtc_Word32
+RTPSenderAudio::SetAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID)
+{
+ if(ID < 1 || ID > 14)
+ {
+ return -1;
+ }
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ _includeAudioLevelIndication = enable;
+ _audioLevelIndicationID = ID;
+
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderAudio::AudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const
+{
+ CriticalSectionScoped cs(_sendAudioCritsect);
+ enable = _includeAudioLevelIndication;
+ ID = _audioLevelIndicationID;
+ return 0;
+}
+
+ // Audio level magnitude and voice activity flag are set for each RTP packet
+WebRtc_Word32
+RTPSenderAudio::SetAudioLevel(const WebRtc_UWord8 level_dBov)
+{
+ if (level_dBov > 127)
+ {
+ return -1;
+ }
+ CriticalSectionScoped cs(_sendAudioCritsect);
+ _audioLevel_dBov = level_dBov;
+ return 0;
+}
+
+ // Set payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32
+RTPSenderAudio::SetRED(const WebRtc_Word8 payloadType)
+{
+ if(payloadType < -1 )
+ {
+ return -1;
+ }
+ _REDPayloadType = payloadType;
+ return 0;
+}
+
+ // Get payload type for Redundant Audio Data RFC 2198
+WebRtc_Word32
+RTPSenderAudio::RED(WebRtc_Word8& payloadType) const
+{
+ if(_REDPayloadType == -1)
+ {
+ // not configured
+ return -1;
+ }
+ payloadType = _REDPayloadType;
+ return 0;
+}
+
+// Send a TelephoneEvent tone using RFC 2833 (4733)
+WebRtc_Word32
+RTPSenderAudio::SendTelephoneEvent(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level)
+{
+ // DTMF is protected by its own critsect
+ if(_dtmfPayloadType < 0)
+ {
+ // TelephoneEvent payloadtype not configured
+ return -1;
+ }
+ return AddDTMF(key, time_ms, level);
+}
+
+WebRtc_Word32
+RTPSenderAudio::SendTelephoneEventPacket(const bool ended,
+ const WebRtc_UWord32 dtmfTimeStamp,
+ const WebRtc_UWord16 duration,
+ const bool markerBit)
+{
+ WebRtc_UWord8 dtmfbuffer[IP_PACKET_SIZE];
+ WebRtc_UWord8 sendCount = 1;
+ WebRtc_Word32 retVal = 0;
+
+ if(ended)
+ {
+ // resend last packet in an event 3 times
+ sendCount = 3;
+ }
+ do
+ {
+ _sendAudioCritsect->Enter();
+
+ //Send DTMF data
+ _rtpSender->BuildRTPheader(dtmfbuffer, _dtmfPayloadType, markerBit, dtmfTimeStamp);
+
+ // reset CSRC and X bit
+ dtmfbuffer[0] &= 0xe0;
+
+ //Create DTMF data
+ /* From RFC 2833:
+
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | event |E|R| volume | duration |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+ // R bit always cleared
+ WebRtc_UWord8 R = 0x00;
+ WebRtc_UWord8 volume = _dtmfLevel;
+
+ // First packet un-ended
+ WebRtc_UWord8 E = 0x00;
+
+ if(ended)
+ {
+ E = 0x80;
+ }
+
+ // First byte is Event number, equals key number
+ dtmfbuffer[12] = _dtmfKey;
+ dtmfbuffer[13] = E|R|volume;
+ ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration);
+
+ _sendAudioCritsect->Leave();
+ retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12,
+ kAllowRetransmission);
+ sendCount--;
+
+ }while (sendCount > 0 && retVal == 0);
+
+ return retVal;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h
new file mode 100644
index 0000000..5fda2ef
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+
+#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
+#include "common_types.h" // Transport
+#include "typedefs.h"
+
+#include "dtmf_queue.h"
+#include "rtp_utility.h"
+
+#include "rtp_sender.h"
+
+namespace webrtc {
+class RTPSenderAudio: public DTMFqueue
+{
+public:
+ RTPSenderAudio(const WebRtc_Word32 id, RtpRtcpClock* clock,
+ RTPSenderInterface* rtpSender);
+ virtual ~RTPSenderAudio();
+
+ void ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 Init();
+
+ WebRtc_Word32 RegisterAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ ModuleRTPUtility::Payload*& payload);
+
+ WebRtc_Word32 SendAudio(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+ // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
+ WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
+
+ // Set status and ID for header-extension-for-audio-level-indication.
+ // Valid ID range is [1,14].
+ WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
+ const WebRtc_UWord8 ID);
+
+ // Get status and ID for header-extension-for-audio-level-indication.
+ WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
+ WebRtc_UWord8& ID) const;
+
+ // Store the audio level in dBov for header-extension-for-audio-level-indication.
+ // Valid range is [0,100]. Actual value is negative.
+ WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
+
+ // Send a DTMF tone using RFC 2833 (4733)
+ WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
+ const WebRtc_UWord16 time_ms,
+ const WebRtc_UWord8 level);
+
+ bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
+
+ void SetAudioFrequency(const WebRtc_UWord32 f);
+
+ int AudioFrequency() const;
+
+ // Set payload type for Redundant Audio Data RFC 2198
+ WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
+
+ // Get payload type for Redundant Audio Data RFC 2198
+ WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
+
+ WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
+
+protected:
+ WebRtc_Word32 SendTelephoneEventPacket(const bool ended,
+ const WebRtc_UWord32 dtmfTimeStamp,
+ const WebRtc_UWord16 duration,
+ const bool markerBit); // set on first packet in talk burst
+
+ bool MarkerBit(const FrameType frameType,
+ const WebRtc_Word8 payloadType);
+
+private:
+ WebRtc_Word32 _id;
+ RtpRtcpClock& _clock;
+ RTPSenderInterface* _rtpSender;
+ CriticalSectionWrapper* _audioFeedbackCritsect;
+ RtpAudioFeedback* _audioFeedback;
+
+ CriticalSectionWrapper* _sendAudioCritsect;
+
+ WebRtc_UWord32 _frequency;
+ WebRtc_UWord16 _packetSizeSamples;
+
+ // DTMF
+ bool _dtmfEventIsOn;
+ bool _dtmfEventFirstPacketSent;
+ WebRtc_Word8 _dtmfPayloadType;
+ WebRtc_UWord32 _dtmfTimestamp;
+ WebRtc_UWord8 _dtmfKey;
+ WebRtc_UWord32 _dtmfLengthSamples;
+ WebRtc_UWord8 _dtmfLevel;
+ WebRtc_UWord32 _dtmfTimeLastSent;
+ WebRtc_UWord32 _dtmfTimestampLastSent;
+
+ WebRtc_Word8 _REDPayloadType;
+
+ // VAD detection, used for markerbit
+ bool _inbandVADactive;
+ WebRtc_Word8 _cngNBPayloadType;
+ WebRtc_Word8 _cngWBPayloadType;
+ WebRtc_Word8 _cngSWBPayloadType;
+ WebRtc_Word8 _lastPayloadType;
+
+ // Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+ bool _includeAudioLevelIndication;
+ WebRtc_UWord8 _audioLevelIndicationID;
+ WebRtc_UWord8 _audioLevel_dBov;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc
new file mode 100644
index 0000000..4e8c78c
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc
@@ -0,0 +1,227 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes unit tests for the RTPSender.
+ */
+
+#include <gtest/gtest.h>
+
+#include "rtp_header_extension.h"
+#include "rtp_rtcp_defines.h"
+#include "rtp_sender.h"
+#include "rtp_utility.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+namespace {
+const int kId = 1;
+const int kTypeLength = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
+const int kPayload = 100;
+const uint32_t kTimestamp = 10;
+const uint16_t kSeqNum = 33;
+const int kTimeOffset = 22222;
+const int kMaxPacketLength = 1500;
+} // namespace
+
+class FakeClockTest : public RtpRtcpClock {
+ public:
+ FakeClockTest() {
+ time_in_ms_ = 123456;
+ }
+ // Return a timestamp in milliseconds relative to some arbitrary
+ // source; the source is fixed for this clock.
+ virtual WebRtc_UWord32 GetTimeInMS() {
+ return time_in_ms_;
+ }
+ // Retrieve an NTP absolute timestamp.
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
+ secs = time_in_ms_ / 1000;
+ frac = (time_in_ms_ % 1000) * 4294967;
+ }
+ void IncrementTime(WebRtc_UWord32 time_increment_ms) {
+ time_in_ms_ += time_increment_ms;
+ }
+ private:
+ WebRtc_UWord32 time_in_ms_;
+};
+
+class LoopbackTransportTest : public webrtc::Transport {
+ public:
+ LoopbackTransportTest()
+ : packets_sent_(0),
+ last_sent_packet_len_(0) {
+ }
+ virtual int SendPacket(int channel, const void *data, int len) {
+ packets_sent_++;
+ memcpy(last_sent_packet_, data, len);
+ last_sent_packet_len_ = len;
+ return len;
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len) {
+ return -1;
+ }
+ int packets_sent_;
+ int last_sent_packet_len_;
+ uint8_t last_sent_packet_[kMaxPacketLength];
+};
+
+class RtpSenderTest : public ::testing::Test {
+ protected:
+ RtpSenderTest()
+ : fake_clock_(),
+ rtp_sender_(new RTPSender(0, false, &fake_clock_)),
+ transport_(),
+ kMarkerBit(true),
+ kType(kRtpExtensionTransmissionTimeOffset),
+ packet_() {
+ EXPECT_EQ(0, rtp_sender_->SetSequenceNumber(kSeqNum));
+ }
+ ~RtpSenderTest() {
+ delete rtp_sender_;
+ }
+
+ FakeClockTest fake_clock_;
+ RTPSender* rtp_sender_;
+ LoopbackTransportTest transport_;
+ const bool kMarkerBit;
+ RTPExtensionType kType;
+ uint8_t packet_[kMaxPacketLength];
+
+ void VerifyRTPHeaderCommon(const WebRtcRTPHeader& rtp_header) {
+ EXPECT_EQ(kMarkerBit, rtp_header.header.markerBit);
+ EXPECT_EQ(kPayload, rtp_header.header.payloadType);
+ EXPECT_EQ(kSeqNum, rtp_header.header.sequenceNumber);
+ EXPECT_EQ(kTimestamp, rtp_header.header.timestamp);
+ EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.header.ssrc);
+ EXPECT_EQ(0, rtp_header.header.numCSRCs);
+ EXPECT_EQ(0, rtp_header.header.paddingLength);
+ }
+};
+
+TEST_F(RtpSenderTest, RegisterRtpHeaderExtension) {
+ EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
+ EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES + kTypeLength,
+ rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(kType));
+ EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
+}
+
+TEST_F(RtpSenderTest, BuildRTPPacket) {
+ WebRtc_Word32 length = rtp_sender_->BuildRTPheader(packet_,
+ kPayload,
+ kMarkerBit,
+ kTimestamp);
+ EXPECT_EQ(12, length);
+
+ // Verify
+ webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(packet_, length);
+ webrtc::WebRtcRTPHeader rtp_header;
+
+ RtpHeaderExtensionMap map;
+ map.Register(kType, kId);
+ const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
+
+ ASSERT_TRUE(valid_rtp_header);
+ ASSERT_FALSE(rtpParser.RTCP());
+ VerifyRTPHeaderCommon(rtp_header);
+ EXPECT_EQ(length, rtp_header.header.headerLength);
+ EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
+}
+
+TEST_F(RtpSenderTest, BuildRTPPacketWithExtension) {
+ EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
+
+ WebRtc_Word32 length = rtp_sender_->BuildRTPheader(packet_,
+ kPayload,
+ kMarkerBit,
+ kTimestamp);
+ EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
+
+ // Verify
+ webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(packet_, length);
+ webrtc::WebRtcRTPHeader rtp_header;
+
+ RtpHeaderExtensionMap map;
+ map.Register(kType, kId);
+ const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
+
+ ASSERT_TRUE(valid_rtp_header);
+ ASSERT_FALSE(rtpParser.RTCP());
+ VerifyRTPHeaderCommon(rtp_header);
+ EXPECT_EQ(length, rtp_header.header.headerLength);
+ EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
+
+ // Parse without map extension
+ webrtc::WebRtcRTPHeader rtp_header2;
+ const bool valid_rtp_header2 = rtpParser.Parse(rtp_header2, NULL);
+
+ ASSERT_TRUE(valid_rtp_header2);
+ VerifyRTPHeaderCommon(rtp_header2);
+ EXPECT_EQ(length, rtp_header2.header.headerLength);
+ EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
+}
+
+TEST_F(RtpSenderTest, NoTrafficSmoothing) {
+ EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
+
+ WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
+ kPayload,
+ kMarkerBit,
+ kTimestamp);
+
+ // Packet should be sent immediately.
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ kAllowRetransmission));
+ EXPECT_EQ(1, transport_.packets_sent_);
+ EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
+}
+
+TEST_F(RtpSenderTest, TrafficSmoothing) {
+ rtp_sender_->SetTransmissionSmoothingStatus(true);
+ EXPECT_EQ(0, rtp_sender_->SetStorePacketsStatus(true, 10));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
+ EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
+
+ WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
+ kPayload,
+ kMarkerBit,
+ kTimestamp);
+
+ // Packet should be stored in a send bucket.
+ EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+ kAllowRetransmission));
+ EXPECT_EQ(0, transport_.packets_sent_);
+
+ const int kStoredTimeInMs = 100;
+ fake_clock_.IncrementTime(kStoredTimeInMs);
+
+ // Process send bucket. Packet should now be sent.
+ rtp_sender_->ProcessSendToNetwork();
+ EXPECT_EQ(1, transport_.packets_sent_);
+ EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
+
+ // Parse sent packet.
+ webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(
+ transport_.last_sent_packet_, rtp_length);
+ webrtc::WebRtcRTPHeader rtp_header;
+
+ RtpHeaderExtensionMap map;
+ map.Register(kType, kId);
+ const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
+ ASSERT_TRUE(valid_rtp_header);
+
+ // Verify transmission time offset.
+ EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc
new file mode 100644
index 0000000..7a15208
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -0,0 +1,619 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_sender_video.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#include "rtp_utility.h"
+
+#include <string.h> // memcpy
+#include <cassert> // assert
+#include <cstdlib> // srand
+
+#include "rtp_format_vp8.h"
+
+namespace webrtc {
+enum { REDForFECHeaderLength = 1 };
+
+RTPSenderVideo::RTPSenderVideo(const WebRtc_Word32 id,
+ RtpRtcpClock* clock,
+ RTPSenderInterface* rtpSender) :
+ _id(id),
+ _rtpSender(*rtpSender),
+ _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+
+ _videoType(kRtpNoVideo),
+ _videoCodecInformation(NULL),
+ _maxBitrate(0),
+ _retransmissionSettings(kRetransmitBaseLayer),
+
+ // Generic FEC
+ _fec(id),
+ _fecEnabled(false),
+ _payloadTypeRED(-1),
+ _payloadTypeFEC(-1),
+ _codeRateKey(0),
+ _codeRateDelta(0),
+ _useUepProtectionKey(false),
+ _useUepProtectionDelta(false),
+ _fecProtectionFactor(0),
+ _fecUseUepProtection(false),
+ _numberFirstPartition(0),
+ _fecOverheadRate(clock),
+ _videoBitrate(clock) {
+}
+
+RTPSenderVideo::~RTPSenderVideo()
+{
+ if(_videoCodecInformation)
+ {
+ delete _videoCodecInformation;
+ }
+ delete _sendVideoCritsect;
+}
+
+WebRtc_Word32
+RTPSenderVideo::Init()
+{
+ CriticalSectionScoped cs(_sendVideoCritsect);
+
+ _retransmissionSettings = kRetransmitBaseLayer;
+ _fecEnabled = false;
+ _payloadTypeRED = -1;
+ _payloadTypeFEC = -1;
+ _codeRateKey = 0;
+ _codeRateDelta = 0;
+ _useUepProtectionKey = false;
+ _useUepProtectionDelta = false;
+ _fecProtectionFactor = 0;
+ _fecUseUepProtection = false;
+ _numberFirstPartition = 0;
+ _fecOverheadRate.Init();
+ return 0;
+}
+
+void
+RTPSenderVideo::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+}
+
+void
+RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes videoType)
+{
+ CriticalSectionScoped cs(_sendVideoCritsect);
+ _videoType = videoType;
+}
+
+RtpVideoCodecTypes
+RTPSenderVideo::VideoCodecType() const
+{
+ return _videoType;
+}
+
+WebRtc_Word32 RTPSenderVideo::RegisterVideoPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 maxBitRate,
+ ModuleRTPUtility::Payload*& payload) {
+ CriticalSectionScoped cs(_sendVideoCritsect);
+
+ RtpVideoCodecTypes videoType = kRtpNoVideo;
+ if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3)) {
+ videoType = kRtpVp8Video;
+ } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
+ videoType = kRtpNoVideo;
+ } else {
+ return -1;
+ }
+ payload = new ModuleRTPUtility::Payload;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ payload->typeSpecific.Video.videoCodecType = videoType;
+ payload->typeSpecific.Video.maxRate = maxBitRate;
+ payload->audio = false;
+ return 0;
+}
+
+struct RtpPacket
+{
+ WebRtc_UWord16 rtpHeaderLength;
+ ForwardErrorCorrection::Packet* pkt;
+};
+
+WebRtc_Word32
+RTPSenderVideo::SendVideoPacket(const FrameType frameType,
+ const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 payloadLength,
+ const WebRtc_UWord16 rtpHeaderLength,
+ StorageType storage)
+{
+ if(_fecEnabled)
+ {
+ WebRtc_Word32 retVal = 0;
+
+ const bool markerBit = (dataBuffer[1] & kRtpMarkerBitMask)?true:false;
+ RtpPacket* ptrGenericFEC = new RtpPacket;
+ ptrGenericFEC->pkt = new ForwardErrorCorrection::Packet;
+ ptrGenericFEC->pkt->length = payloadLength + rtpHeaderLength;
+ ptrGenericFEC->rtpHeaderLength = rtpHeaderLength;
+ memcpy(ptrGenericFEC->pkt->data, dataBuffer,
+ ptrGenericFEC->pkt->length);
+
+ // Add packet to FEC list
+ _rtpPacketListFec.push_back(ptrGenericFEC);
+ // FEC can only protect up to kMaxMediaPackets packets
+ if (_mediaPacketListFec.size() <
+ ForwardErrorCorrection::kMaxMediaPackets)
+ {
+ _mediaPacketListFec.push_back(ptrGenericFEC->pkt);
+ }
+
+ // Last packet in frame
+ if (markerBit)
+ {
+
+ // Retain the RTP header of the last media packet to construct FEC
+ // packet RTP headers.
+ ForwardErrorCorrection::Packet lastMediaRtpHeader;
+ memcpy(lastMediaRtpHeader.data,
+ ptrGenericFEC->pkt->data,
+ ptrGenericFEC->rtpHeaderLength);
+
+ lastMediaRtpHeader.length = ptrGenericFEC->rtpHeaderLength;
+ // Replace payload and clear marker bit.
+ lastMediaRtpHeader.data[1] = _payloadTypeRED;
+
+ // Number of first partition packets cannot exceed kMaxMediaPackets
+ if (_numberFirstPartition >
+ ForwardErrorCorrection::kMaxMediaPackets)
+ {
+ _numberFirstPartition =
+ ForwardErrorCorrection::kMaxMediaPackets;
+ }
+
+ std::list<ForwardErrorCorrection::Packet*> fecPacketList;
+ retVal = _fec.GenerateFEC(_mediaPacketListFec,
+ _fecProtectionFactor,
+ _numberFirstPartition,
+ _fecUseUepProtection,
+ &fecPacketList);
+
+ int fecOverheadSent = 0;
+ int videoSent = 0;
+
+ while(!_rtpPacketListFec.empty())
+ {
+ WebRtc_UWord8 newDataBuffer[IP_PACKET_SIZE];
+ memset(newDataBuffer, 0, sizeof(newDataBuffer));
+
+ RtpPacket* packetToSend = _rtpPacketListFec.front();
+
+ // Copy RTP header
+ memcpy(newDataBuffer, packetToSend->pkt->data,
+ packetToSend->rtpHeaderLength);
+
+ // Get codec pltype
+ WebRtc_UWord8 payloadType = newDataBuffer[1] & 0x7f;
+
+ // Replace pltype
+ newDataBuffer[1] &= 0x80; // reset
+ newDataBuffer[1] += _payloadTypeRED; // replace
+
+ // Add RED header
+ // f-bit always 0
+ newDataBuffer[packetToSend->rtpHeaderLength] = payloadType;
+
+ // Copy payload data
+ memcpy(newDataBuffer + packetToSend->rtpHeaderLength +
+ REDForFECHeaderLength,
+ packetToSend->pkt->data + packetToSend->rtpHeaderLength,
+ packetToSend->pkt->length -
+ packetToSend->rtpHeaderLength);
+
+ _rtpPacketListFec.pop_front();
+ // Check if _mediaPacketListFec is non-empty.
+ // This list may be smaller than rtpPacketList, if the frame
+ // has more than kMaxMediaPackets.
+ if (!_mediaPacketListFec.empty()) {
+ _mediaPacketListFec.pop_front();
+ }
+
+ // Send normal packet with RED header
+ int packetSuccess = _rtpSender.SendToNetwork(
+ newDataBuffer,
+ packetToSend->pkt->length - packetToSend->rtpHeaderLength +
+ REDForFECHeaderLength,
+ packetToSend->rtpHeaderLength,
+ storage);
+
+ retVal |= packetSuccess;
+
+ if (packetSuccess == 0)
+ {
+ videoSent += packetToSend->pkt->length +
+ REDForFECHeaderLength;
+ }
+
+ delete packetToSend->pkt;
+ delete packetToSend;
+ packetToSend = NULL;
+ }
+ assert(_mediaPacketListFec.empty());
+ assert(_rtpPacketListFec.empty());
+
+ while(!fecPacketList.empty())
+ {
+ WebRtc_UWord8 newDataBuffer[IP_PACKET_SIZE];
+
+ // Build FEC packets
+ ForwardErrorCorrection::Packet* packetToSend = fecPacketList.front();
+
+ // The returned FEC packets have no RTP headers.
+ // Copy the last media packet's modified RTP header.
+ memcpy(newDataBuffer, lastMediaRtpHeader.data,
+ lastMediaRtpHeader.length);
+
+ // Add sequence number
+ ModuleRTPUtility::AssignUWord16ToBuffer(
+ &newDataBuffer[2], _rtpSender.IncrementSequenceNumber());
+
+ // Add RED header
+ // f-bit always 0
+ newDataBuffer[lastMediaRtpHeader.length] = _payloadTypeFEC;
+
+ // Copy payload data
+ memcpy(newDataBuffer + lastMediaRtpHeader.length +
+ REDForFECHeaderLength,
+ packetToSend->data,
+ packetToSend->length);
+
+ fecPacketList.pop_front();
+
+ // Invalid FEC packet
+ assert(packetToSend->length != 0);
+
+ StorageType storage = kDontRetransmit;
+ if (_retransmissionSettings & kRetransmitFECPackets) {
+ storage = kAllowRetransmission;
+ }
+
+ // No marker bit on FEC packets, last media packet have the
+ // marker send FEC packet with RED header
+ int packetSuccess = _rtpSender.SendToNetwork(
+ newDataBuffer,
+ packetToSend->length + REDForFECHeaderLength,
+ lastMediaRtpHeader.length,
+ storage);
+
+ retVal |= packetSuccess;
+
+ if (packetSuccess == 0)
+ {
+ fecOverheadSent += packetToSend->length +
+ REDForFECHeaderLength + lastMediaRtpHeader.length;
+ }
+ }
+ _videoBitrate.Update(videoSent);
+ _fecOverheadRate.Update(fecOverheadSent);
+ }
+ return retVal;
+ }
+ int retVal = _rtpSender.SendToNetwork(dataBuffer,
+ payloadLength,
+ rtpHeaderLength,
+ storage);
+ if (retVal == 0)
+ {
+ _videoBitrate.Update(payloadLength + rtpHeaderLength);
+ }
+ return retVal;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SendRTPIntraRequest()
+{
+ // RFC 2032
+ // 5.2.1. Full intra-frame Request (FIR) packet
+
+ WebRtc_UWord16 length = 8;
+ WebRtc_UWord8 data[8];
+ data[0] = 0x80;
+ data[1] = 192;
+ data[2] = 0;
+ data[3] = 1; // length
+
+ ModuleRTPUtility::AssignUWord32ToBuffer(data+4, _rtpSender.SSRC());
+
+ return _rtpSender.SendToNetwork(data, 0, length, kAllowRetransmission);
+}
+
+WebRtc_Word32
+RTPSenderVideo::SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC)
+{
+ _fecEnabled = enable;
+ _payloadTypeRED = payloadTypeRED;
+ _payloadTypeFEC = payloadTypeFEC;
+ _codeRateKey = 0;
+ _codeRateDelta = 0;
+ _useUepProtectionKey = false;
+ _useUepProtectionDelta = false;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderVideo::GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) const
+{
+ enable = _fecEnabled;
+ payloadTypeRED = _payloadTypeRED;
+ payloadTypeFEC = _payloadTypeFEC;
+ return 0;
+}
+
+WebRtc_UWord16
+RTPSenderVideo::FECPacketOverhead() const
+{
+ if (_fecEnabled)
+ {
+ return ForwardErrorCorrection::PacketOverhead() +
+ REDForFECHeaderLength;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate)
+{
+ _codeRateKey = keyFrameCodeRate;
+ _codeRateDelta = deltaFrameCodeRate;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection)
+{
+ _useUepProtectionKey = keyUseUepProtection;
+ _useUepProtectionDelta = deltaUseUepProtection;
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
+ const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ VideoCodecInformation* codecInfo,
+ const RTPVideoTypeHeader* rtpTypeHdr)
+{
+ if( payloadSize == 0)
+ {
+ return -1;
+ }
+
+ if (frameType == kVideoFrameKey)
+ {
+ _fecProtectionFactor = _codeRateKey;
+ _fecUseUepProtection = _useUepProtectionKey;
+ } else if (videoType == kRtpVp8Video && rtpTypeHdr->VP8.temporalIdx > 0)
+ {
+ // In current version, we only apply FEC on the base layer.
+ _fecProtectionFactor = 0;
+ _fecUseUepProtection = false;
+ } else
+ {
+ _fecProtectionFactor = _codeRateDelta;
+ _fecUseUepProtection = _useUepProtectionDelta;
+ }
+
+ // Default setting for number of first partition packets:
+ // Will be extracted in SendVP8 for VP8 codec; other codecs use 0
+ _numberFirstPartition = 0;
+
+ WebRtc_Word32 retVal = -1;
+ switch(videoType)
+ {
+ case kRtpNoVideo:
+ retVal = SendGeneric(payloadType,captureTimeStamp, payloadData,
+ payloadSize);
+ break;
+ case kRtpVp8Video:
+ retVal = SendVP8(frameType, payloadType, captureTimeStamp,
+ payloadData, payloadSize, fragmentation, rtpTypeHdr);
+ break;
+ default:
+ assert(false);
+ break;
+ }
+ if(retVal <= 0)
+ {
+ return retVal;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize)
+{
+ WebRtc_UWord16 payloadBytesInPacket = 0;
+ WebRtc_UWord32 bytesSent = 0;
+ WebRtc_Word32 payloadBytesToSend = payloadSize;
+
+ const WebRtc_UWord8* data = payloadData;
+ WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
+ WebRtc_UWord16 maxLength = _rtpSender.MaxPayloadLength() -
+ FECPacketOverhead() - rtpHeaderLength;
+ WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
+
+ // Fragment packet into packets of max MaxPayloadLength bytes payload.
+ while (payloadBytesToSend > 0)
+ {
+ if (payloadBytesToSend > maxLength)
+ {
+ payloadBytesInPacket = maxLength;
+ payloadBytesToSend -= payloadBytesInPacket;
+ // MarkerBit is 0
+ if(_rtpSender.BuildRTPheader(dataBuffer,
+ payloadType,
+ false,
+ captureTimeStamp) != rtpHeaderLength)
+ {
+ return -1;
+ }
+ }
+ else
+ {
+ payloadBytesInPacket = (WebRtc_UWord16)payloadBytesToSend;
+ payloadBytesToSend = 0;
+ // MarkerBit is 1
+ if(_rtpSender.BuildRTPheader(dataBuffer, payloadType, true,
+ captureTimeStamp) != rtpHeaderLength)
+ {
+ return -1;
+ }
+ }
+
+ // Put payload in packet
+ memcpy(&dataBuffer[rtpHeaderLength], &data[bytesSent],
+ payloadBytesInPacket);
+ bytesSent += payloadBytesInPacket;
+
+ if(-1 == SendVideoPacket(kVideoFrameKey,
+ dataBuffer,
+ payloadBytesInPacket,
+ rtpHeaderLength,
+ kAllowRetransmission))
+ {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+VideoCodecInformation*
+RTPSenderVideo::CodecInformationVideo()
+{
+ return _videoCodecInformation;
+}
+
+void
+RTPSenderVideo::SetMaxConfiguredBitrateVideo(const WebRtc_UWord32 maxBitrate)
+{
+ _maxBitrate = maxBitrate;
+}
+
+WebRtc_UWord32
+RTPSenderVideo::MaxConfiguredBitrateVideo() const
+{
+ return _maxBitrate;
+}
+
+WebRtc_Word32
+RTPSenderVideo::SendVP8(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ const RTPVideoTypeHeader* rtpTypeHdr)
+{
+ const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
+
+ WebRtc_Word32 payloadBytesToSend = payloadSize;
+ const WebRtc_UWord8* data = payloadData;
+
+ WebRtc_UWord16 maxPayloadLengthVP8 = _rtpSender.MaxDataPayloadLength();
+
+ assert(rtpTypeHdr);
+ // Initialize disregarding partition boundaries: this will use kEqualSize
+ // packetization mode, which produces ~equal size packets for each frame.
+ RtpFormatVp8 packetizer(data, payloadBytesToSend, rtpTypeHdr->VP8,
+ maxPayloadLengthVP8);
+
+ StorageType storage = kAllowRetransmission;
+ if (rtpTypeHdr->VP8.temporalIdx == 0 &&
+ !(_retransmissionSettings & kRetransmitBaseLayer)) {
+ storage = kDontRetransmit;
+ }
+ if (rtpTypeHdr->VP8.temporalIdx > 0 &&
+ !(_retransmissionSettings & kRetransmitHigherLayers)) {
+ storage = kDontRetransmit;
+ }
+
+ bool last = false;
+ _numberFirstPartition = 0;
+ while (!last)
+ {
+ // Write VP8 Payload Descriptor and VP8 payload.
+ WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE] = {0};
+ int payloadBytesInPacket = 0;
+ int packetStartPartition =
+ packetizer.NextPacket(&dataBuffer[rtpHeaderLength],
+ &payloadBytesInPacket, &last);
+ if (packetStartPartition == 0)
+ {
+ ++_numberFirstPartition;
+ }
+ else if (packetStartPartition < 0)
+ {
+ return -1;
+ }
+
+ // Write RTP header.
+ // Set marker bit true if this is the last packet in frame.
+ _rtpSender.BuildRTPheader(dataBuffer, payloadType, last,
+ captureTimeStamp);
+ if (-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket,
+ rtpHeaderLength, storage))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "RTPSenderVideo::SendVP8 failed to send packet number"
+ " %d", _rtpSender.SequenceNumber());
+ }
+ }
+ return 0;
+}
+
+void RTPSenderVideo::ProcessBitrate() {
+ _videoBitrate.Process();
+ _fecOverheadRate.Process();
+}
+
+WebRtc_UWord32 RTPSenderVideo::VideoBitrateSent() const {
+ return _videoBitrate.BitrateLast();
+}
+
+WebRtc_UWord32 RTPSenderVideo::FecOverheadRate() const {
+ return _fecOverheadRate.BitrateLast();
+}
+
+int RTPSenderVideo::SelectiveRetransmissions() const {
+ return _retransmissionSettings;
+}
+
+int RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
+ _retransmissionSettings = settings;
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h
new file mode 100644
index 0000000..1bf7142
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -0,0 +1,149 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
+
+#include <list>
+
+#include "typedefs.h"
+#include "common_types.h" // Transport
+#include "rtp_rtcp_config.h"
+
+#include "rtp_rtcp_defines.h"
+#include "rtp_utility.h"
+
+#include "video_codec_information.h"
+#include "forward_error_correction.h"
+#include "Bitrate.h"
+#include "rtp_sender.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+struct RtpPacket;
+
+class RTPSenderVideo
+{
+public:
+ RTPSenderVideo(const WebRtc_Word32 id, RtpRtcpClock* clock,
+ RTPSenderInterface* rtpSender);
+ virtual ~RTPSenderVideo();
+
+ WebRtc_Word32 Init();
+
+ virtual void ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual RtpVideoCodecTypes VideoCodecType() const;
+
+ WebRtc_UWord16 FECPacketOverhead() const;
+
+ WebRtc_Word32 RegisterVideoPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 maxBitRate,
+ ModuleRTPUtility::Payload*& payload);
+
+ WebRtc_Word32 SendVideo(const RtpVideoCodecTypes videoType,
+ const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ VideoCodecInformation* codecInfo,
+ const RTPVideoTypeHeader* rtpTypeHdr);
+
+ WebRtc_Word32 SendRTPIntraRequest();
+
+ void SetVideoCodecType(RtpVideoCodecTypes type);
+
+ VideoCodecInformation* CodecInformationVideo();
+
+ void SetMaxConfiguredBitrateVideo(const WebRtc_UWord32 maxBitrate);
+
+ WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
+
+ // FEC
+ WebRtc_Word32 SetGenericFECStatus(const bool enable,
+ const WebRtc_UWord8 payloadTypeRED,
+ const WebRtc_UWord8 payloadTypeFEC);
+
+ WebRtc_Word32 GenericFECStatus(bool& enable,
+ WebRtc_UWord8& payloadTypeRED,
+ WebRtc_UWord8& payloadTypeFEC) const;
+
+ WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
+ const WebRtc_UWord8 deltaFrameCodeRate);
+
+ WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
+ const bool deltaUseUepProtection);
+
+ void ProcessBitrate();
+
+ WebRtc_UWord32 VideoBitrateSent() const;
+ WebRtc_UWord32 FecOverheadRate() const;
+
+ int SelectiveRetransmissions() const;
+ int SetSelectiveRetransmissions(uint8_t settings);
+
+protected:
+ virtual WebRtc_Word32 SendVideoPacket(const FrameType frameType,
+ const WebRtc_UWord8* dataBuffer,
+ const WebRtc_UWord16 payloadLength,
+ const WebRtc_UWord16 rtpHeaderLength,
+ StorageType storage);
+
+private:
+ WebRtc_Word32 SendGeneric(const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize);
+
+ WebRtc_Word32 SendVP8(const FrameType frameType,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation,
+ const RTPVideoTypeHeader* rtpTypeHdr);
+
+private:
+ WebRtc_Word32 _id;
+ RTPSenderInterface& _rtpSender;
+
+ CriticalSectionWrapper* _sendVideoCritsect;
+ RtpVideoCodecTypes _videoType;
+ VideoCodecInformation* _videoCodecInformation;
+ WebRtc_UWord32 _maxBitrate;
+ WebRtc_Word32 _retransmissionSettings;
+
+ // FEC
+ ForwardErrorCorrection _fec;
+ bool _fecEnabled;
+ WebRtc_Word8 _payloadTypeRED;
+ WebRtc_Word8 _payloadTypeFEC;
+ WebRtc_UWord8 _codeRateKey;
+ WebRtc_UWord8 _codeRateDelta;
+ bool _useUepProtectionKey;
+ bool _useUepProtectionDelta;
+ WebRtc_UWord8 _fecProtectionFactor;
+ bool _fecUseUepProtection;
+ unsigned int _numberFirstPartition;
+ std::list<ForwardErrorCorrection::Packet*> _mediaPacketListFec;
+ std::list<RtpPacket*> _rtpPacketListFec;
+ // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
+ // and any padding overhead.
+ Bitrate _fecOverheadRate;
+ // Bitrate used for video payload and RTP headers
+ Bitrate _videoBitrate;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc b/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
new file mode 100644
index 0000000..947796d
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
@@ -0,0 +1,914 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_utility.h"
+
+#include <cassert>
+#include <cmath> // ceil
+#include <cstring> // memcpy
+
+#if defined(_WIN32)
+#include <Windows.h> // FILETIME
+#include <WinSock.h> // timeval
+#include <MMSystem.h> // timeGetTime
+#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
+#include <sys/time.h> // gettimeofday
+#include <time.h>
+#endif
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#include <stdio.h>
+#endif
+
+#include "trace.h"
+
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#define DEBUG_PRINT(...) \
+ { \
+ char msg[256]; \
+ sprintf(msg, __VA_ARGS__); \
+ OutputDebugString(msg); \
+ }
+#else
+// special fix for visual 2003
+#define DEBUG_PRINT(exp) ((void)0)
+#endif // defined(_DEBUG) && defined(_WIN32)
+
+namespace webrtc {
+
+namespace ModuleRTPUtility {
+
+/*
+ * Time routines.
+ */
+
+#if defined(_WIN32)
+
+struct reference_point {
+ FILETIME file_time;
+ LARGE_INTEGER counterMS;
+};
+
+struct WindowsHelpTimer {
+ volatile LONG _timeInMs;
+ volatile LONG _numWrapTimeInMs;
+ reference_point _ref_point;
+
+ volatile LONG _sync_flag;
+};
+
+void Synchronize(WindowsHelpTimer* help_timer) {
+ const LONG start_value = 0;
+ const LONG new_value = 1;
+ const LONG synchronized_value = 2;
+
+ LONG compare_flag = new_value;
+ while (help_timer->_sync_flag == start_value) {
+ const LONG new_value = 1;
+ compare_flag = InterlockedCompareExchange(
+ &help_timer->_sync_flag, new_value, start_value);
+ }
+ if (compare_flag != start_value) {
+ // This thread was not the one that incremented the sync flag.
+ // Block until synchronization finishes.
+ while (compare_flag != synchronized_value) {
+ ::Sleep(0);
+ }
+ return;
+ }
+ // Only the synchronizing thread gets here so this part can be
+ // considered single threaded.
+
+ // set timer accuracy to 1 ms
+ timeBeginPeriod(1);
+ FILETIME ft0 = { 0, 0 },
+ ft1 = { 0, 0 };
+ //
+ // Spin waiting for a change in system time. Get the matching
+ // performance counter value for that time.
+ //
+ ::GetSystemTimeAsFileTime(&ft0);
+ do {
+ ::GetSystemTimeAsFileTime(&ft1);
+
+ help_timer->_ref_point.counterMS.QuadPart = ::timeGetTime();
+ ::Sleep(0);
+ } while ((ft0.dwHighDateTime == ft1.dwHighDateTime) &&
+ (ft0.dwLowDateTime == ft1.dwLowDateTime));
+ help_timer->_ref_point.file_time = ft1;
+}
+
+void get_time(WindowsHelpTimer* help_timer, FILETIME& current_time) {
+ // we can't use query performance counter due to speed stepping
+ DWORD t = timeGetTime();
+ // NOTE: we have a missmatch in sign between _timeInMs(LONG) and
+ // (DWORD) however we only use it here without +- etc
+ volatile LONG* timeInMsPtr = &help_timer->_timeInMs;
+ // Make sure that we only inc wrapper once.
+ DWORD old = InterlockedExchange(timeInMsPtr, t);
+ if(old > t) {
+ // wrap
+ help_timer->_numWrapTimeInMs++;
+ }
+ LARGE_INTEGER elapsedMS;
+ elapsedMS.HighPart = help_timer->_numWrapTimeInMs;
+ elapsedMS.LowPart = t;
+
+ elapsedMS.QuadPart = elapsedMS.QuadPart -
+ help_timer->_ref_point.counterMS.QuadPart;
+
+ // Translate to 100-nanoseconds intervals (FILETIME resolution)
+ // and add to reference FILETIME to get current FILETIME.
+ ULARGE_INTEGER filetime_ref_as_ul;
+
+ filetime_ref_as_ul.HighPart =
+ help_timer->_ref_point.file_time.dwHighDateTime;
+ filetime_ref_as_ul.LowPart =
+ help_timer->_ref_point.file_time.dwLowDateTime;
+ filetime_ref_as_ul.QuadPart +=
+ (ULONGLONG)((elapsedMS.QuadPart)*1000*10);
+
+ // Copy to result
+ current_time.dwHighDateTime = filetime_ref_as_ul.HighPart;
+ current_time.dwLowDateTime = filetime_ref_as_ul.LowPart;
+ }
+
+ // A clock reading times from the Windows API.
+ class WindowsSystemClock : public RtpRtcpClock {
+ public:
+ WindowsSystemClock(WindowsHelpTimer* helpTimer)
+ : _helpTimer(helpTimer) {}
+
+ virtual ~WindowsSystemClock() {}
+
+ virtual WebRtc_UWord32 GetTimeInMS();
+
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac);
+
+ private:
+ WindowsHelpTimer* _helpTimer;
+};
+
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+
+// A clock reading times from the POSIX API.
+class UnixSystemClock : public RtpRtcpClock {
+public:
+ UnixSystemClock() {}
+ virtual ~UnixSystemClock() {}
+
+ virtual WebRtc_UWord32 GetTimeInMS();
+
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac);
+};
+#endif
+
+#if defined(_WIN32)
+WebRtc_UWord32 WindowsSystemClock::GetTimeInMS() {
+ return timeGetTime();
+}
+
+// Use the system time (roughly synchronised to the tick, and
+// extrapolated using the system performance counter.
+void WindowsSystemClock::CurrentNTP(WebRtc_UWord32& secs,
+ WebRtc_UWord32& frac) {
+ const WebRtc_UWord64 FILETIME_1970 = 0x019db1ded53e8000;
+
+ FILETIME StartTime;
+ WebRtc_UWord64 Time;
+ struct timeval tv;
+
+ // We can't use query performance counter since they can change depending on
+ // speed steping
+ get_time(_helpTimer, StartTime);
+
+ Time = (((WebRtc_UWord64) StartTime.dwHighDateTime) << 32) +
+ (WebRtc_UWord64) StartTime.dwLowDateTime;
+
+ // Convert the hecto-nano second time to tv format
+ Time -= FILETIME_1970;
+
+ tv.tv_sec = (WebRtc_UWord32)(Time / (WebRtc_UWord64)10000000);
+ tv.tv_usec = (WebRtc_UWord32)((Time % (WebRtc_UWord64)10000000) / 10);
+
+ double dtemp;
+
+ secs = tv.tv_sec + NTP_JAN_1970;
+ dtemp = tv.tv_usec / 1e6;
+
+ if (dtemp >= 1) {
+ dtemp -= 1;
+ secs++;
+ } else if (dtemp < -1) {
+ dtemp += 1;
+ secs--;
+ }
+ dtemp *= NTP_FRAC;
+ frac = (WebRtc_UWord32)dtemp;
+}
+
+#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
+
+WebRtc_UWord32 UnixSystemClock::GetTimeInMS() {
+ struct timeval tv;
+ struct timezone tz;
+ WebRtc_UWord32 val;
+
+ gettimeofday(&tv, &tz);
+ val = (WebRtc_UWord32)(tv.tv_sec * 1000 + tv.tv_usec / 1000);
+ return val;
+}
+
+// Use the system time.
+void UnixSystemClock::CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
+ double dtemp;
+ struct timeval tv;
+ struct timezone tz;
+ tz.tz_minuteswest = 0;
+ tz.tz_dsttime = 0;
+ gettimeofday(&tv, &tz);
+
+ secs = tv.tv_sec + NTP_JAN_1970;
+ dtemp = tv.tv_usec / 1e6;
+ if (dtemp >= 1) {
+ dtemp -= 1;
+ secs++;
+ } else if (dtemp < -1) {
+ dtemp += 1;
+ secs--;
+ }
+ dtemp *= NTP_FRAC;
+ frac = (WebRtc_UWord32)dtemp;
+}
+#endif
+
+#if defined(_WIN32)
+// Keeps the global state for the Windows implementation of RtpRtcpClock.
+// Note that this is a POD. Only PODs are allowed to have static storage
+// duration according to the Google Style guide.
+static WindowsHelpTimer global_help_timer = {0, 0, {{ 0, 0}, 0}, 0};
+#endif
+
+RtpRtcpClock* GetSystemClock() {
+#if defined(_WIN32)
+ return new WindowsSystemClock(&global_help_timer);
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ return new UnixSystemClock();
+#else
+ return NULL;
+#endif
+}
+
+WebRtc_UWord32 GetCurrentRTP(RtpRtcpClock* clock, WebRtc_UWord32 freq) {
+ const bool use_global_clock = (clock == NULL);
+ RtpRtcpClock* local_clock = clock;
+ if (use_global_clock) {
+ local_clock = GetSystemClock();
+ }
+ WebRtc_UWord32 secs = 0, frac = 0;
+ local_clock->CurrentNTP(secs, frac);
+ if (use_global_clock) {
+ delete local_clock;
+ }
+ return ConvertNTPTimeToRTP(secs, frac, freq);
+}
+
+WebRtc_UWord32 ConvertNTPTimeToRTP(WebRtc_UWord32 NTPsec,
+ WebRtc_UWord32 NTPfrac,
+ WebRtc_UWord32 freq) {
+ float ftemp = (float)NTPfrac / (float)NTP_FRAC;
+ WebRtc_UWord32 tmp = (WebRtc_UWord32)(ftemp * freq);
+ return NTPsec * freq + tmp;
+}
+
+WebRtc_UWord32 ConvertNTPTimeToMS(WebRtc_UWord32 NTPsec,
+ WebRtc_UWord32 NTPfrac) {
+ int freq = 1000;
+ float ftemp = (float)NTPfrac / (float)NTP_FRAC;
+ WebRtc_UWord32 tmp = (WebRtc_UWord32)(ftemp * freq);
+ WebRtc_UWord32 MStime = NTPsec * freq + tmp;
+ return MStime;
+}
+
+bool OldTimestamp(uint32_t newTimestamp,
+ uint32_t existingTimestamp,
+ bool* wrapped) {
+ bool tmpWrapped =
+ (newTimestamp < 0x0000ffff && existingTimestamp > 0xffff0000) ||
+ (newTimestamp > 0xffff0000 && existingTimestamp < 0x0000ffff);
+ *wrapped = tmpWrapped;
+ if (existingTimestamp > newTimestamp && !tmpWrapped) {
+ return true;
+ } else if (existingTimestamp <= newTimestamp && !tmpWrapped) {
+ return false;
+ } else if (existingTimestamp < newTimestamp && tmpWrapped) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+/*
+ * Misc utility routines
+ */
+
+#if defined(_WIN32)
+bool StringCompare(const char* str1, const char* str2,
+ const WebRtc_UWord32 length) {
+ return (_strnicmp(str1, str2, length) == 0) ? true : false;
+}
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+bool StringCompare(const char* str1, const char* str2,
+ const WebRtc_UWord32 length) {
+ return (strncasecmp(str1, str2, length) == 0) ? true : false;
+}
+#endif
+
+#if !defined(WEBRTC_LITTLE_ENDIAN) && !defined(WEBRTC_BIG_ENDIAN)
+#error Either WEBRTC_LITTLE_ENDIAN or WEBRTC_BIG_ENDIAN must be defined
+#endif
+
+/* for RTP/RTCP
+ All integer fields are carried in network byte order, that is, most
+ significant byte (octet) first. AKA big-endian.
+*/
+void AssignUWord32ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value) {
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 24);
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 16);
+ dataBuffer[2] = static_cast<WebRtc_UWord8>(value >> 8);
+ dataBuffer[3] = static_cast<WebRtc_UWord8>(value);
+#else
+ WebRtc_UWord32* ptr = reinterpret_cast<WebRtc_UWord32*>(dataBuffer);
+ ptr[0] = value;
+#endif
+}
+
+void AssignUWord24ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value) {
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 16);
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 8);
+ dataBuffer[2] = static_cast<WebRtc_UWord8>(value);
+#else
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(value);
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 8);
+ dataBuffer[2] = static_cast<WebRtc_UWord8>(value >> 16);
+#endif
+}
+
+void AssignUWord16ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord16 value) {
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 8);
+ dataBuffer[1] = static_cast<WebRtc_UWord8>(value);
+#else
+ WebRtc_UWord16* ptr = reinterpret_cast<WebRtc_UWord16*>(dataBuffer);
+ ptr[0] = value;
+#endif
+}
+
+WebRtc_UWord16 BufferToUWord16(const WebRtc_UWord8* dataBuffer) {
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ return (dataBuffer[0] << 8) + dataBuffer[1];
+#else
+ return *reinterpret_cast<const WebRtc_UWord16*>(dataBuffer);
+#endif
+}
+
+WebRtc_UWord32 BufferToUWord24(const WebRtc_UWord8* dataBuffer) {
+ return (dataBuffer[0] << 16) + (dataBuffer[1] << 8) + dataBuffer[2];
+}
+
+WebRtc_UWord32 BufferToUWord32(const WebRtc_UWord8* dataBuffer) {
+#if defined(WEBRTC_LITTLE_ENDIAN)
+ return (dataBuffer[0] << 24) + (dataBuffer[1] << 16) + (dataBuffer[2] << 8) +
+ dataBuffer[3];
+#else
+ return *reinterpret_cast<const WebRtc_UWord32*>(dataBuffer);
+#endif
+}
+
+WebRtc_UWord32 pow2(WebRtc_UWord8 exp) {
+ return 1 << exp;
+}
+
+void RTPPayload::SetType(RtpVideoCodecTypes videoType) {
+ type = videoType;
+
+ switch (type) {
+ case kRtpNoVideo:
+ break;
+ case kRtpVp8Video: {
+ info.VP8.nonReferenceFrame = false;
+ info.VP8.beginningOfPartition = false;
+ info.VP8.partitionID = 0;
+ info.VP8.hasPictureID = false;
+ info.VP8.hasTl0PicIdx = false;
+ info.VP8.hasTID = false;
+ info.VP8.hasKeyIdx = false;
+ info.VP8.pictureID = -1;
+ info.VP8.tl0PicIdx = -1;
+ info.VP8.tID = -1;
+ info.VP8.layerSync = false;
+ info.VP8.frameWidth = 0;
+ info.VP8.frameHeight = 0;
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+RTPHeaderParser::RTPHeaderParser(const WebRtc_UWord8* rtpData,
+ const WebRtc_UWord32 rtpDataLength)
+ : _ptrRTPDataBegin(rtpData),
+ _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {
+}
+
+RTPHeaderParser::~RTPHeaderParser() {
+}
+
+bool RTPHeaderParser::RTCP() const {
+ // 72 to 76 is reserved for RTP
+ // 77 to 79 is not reserver but they are not assigned we will block them
+ // for RTCP 200 SR == marker bit + 72
+ // for RTCP 204 APP == marker bit + 76
+ /*
+ * RTCP
+ *
+ * FIR full INTRA-frame request 192 [RFC2032] supported
+ * NACK negative acknowledgement 193 [RFC2032]
+ * IJ Extended inter-arrival jitter report 195 [RFC-ietf-avt-rtp-toff
+ * set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
+ * SR sender report 200 [RFC3551] supported
+ * RR receiver report 201 [RFC3551] supported
+ * SDES source description 202 [RFC3551] supported
+ * BYE goodbye 203 [RFC3551] supported
+ * APP application-defined 204 [RFC3551] ignored
+ * RTPFB Transport layer FB message 205 [RFC4585] supported
+ * PSFB Payload-specific FB message 206 [RFC4585] supported
+ * XR extended report 207 [RFC3611] supported
+ */
+
+ /* 205 RFC 5104
+ * FMT 1 NACK supported
+ * FMT 2 reserved
+ * FMT 3 TMMBR supported
+ * FMT 4 TMMBN supported
+ */
+
+ /* 206 RFC 5104
+ * FMT 1: Picture Loss Indication (PLI) supported
+ * FMT 2: Slice Lost Indication (SLI)
+ * FMT 3: Reference Picture Selection Indication (RPSI)
+ * FMT 4: Full Intra Request (FIR) Command supported
+ * FMT 5: Temporal-Spatial Trade-off Request (TSTR)
+ * FMT 6: Temporal-Spatial Trade-off Notification (TSTN)
+ * FMT 7: Video Back Channel Message (VBCM)
+ * FMT 15: Application layer FB message
+ */
+
+ const WebRtc_UWord8 payloadType = _ptrRTPDataBegin[1];
+
+ bool RTCP = false;
+
+ // check if this is a RTCP packet
+ switch (payloadType) {
+ case 192:
+ RTCP = true;
+ break;
+ case 193:
+ // not supported
+ // pass through and check for a potential RTP packet
+ break;
+ case 195:
+ case 200:
+ case 201:
+ case 202:
+ case 203:
+ case 204:
+ case 205:
+ case 206:
+ case 207:
+ RTCP = true;
+ break;
+ }
+ return RTCP;
+}
+
+bool RTPHeaderParser::Parse(WebRtcRTPHeader& parsedPacket,
+ RtpHeaderExtensionMap* ptrExtensionMap) const {
+ const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+
+ if (length < 12) {
+ return false;
+ }
+
+ // Version
+ const WebRtc_UWord8 V = _ptrRTPDataBegin[0] >> 6;
+ // Padding
+ const bool P = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
+ // eXtension
+ const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
+ const WebRtc_UWord8 CC = _ptrRTPDataBegin[0] & 0x0f;
+ const bool M = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
+
+ const WebRtc_UWord8 PT = _ptrRTPDataBegin[1] & 0x7f;
+
+ const WebRtc_UWord16 sequenceNumber = (_ptrRTPDataBegin[2] << 8) +
+ _ptrRTPDataBegin[3];
+
+ const WebRtc_UWord8* ptr = &_ptrRTPDataBegin[4];
+
+ WebRtc_UWord32 RTPTimestamp = *ptr++ << 24;
+ RTPTimestamp += *ptr++ << 16;
+ RTPTimestamp += *ptr++ << 8;
+ RTPTimestamp += *ptr++;
+
+ WebRtc_UWord32 SSRC = *ptr++ << 24;
+ SSRC += *ptr++ << 16;
+ SSRC += *ptr++ << 8;
+ SSRC += *ptr++;
+
+ if (V != 2) {
+ return false;
+ }
+
+ const WebRtc_UWord8 CSRCocts = CC * 4;
+
+ if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
+ return false;
+ }
+
+ parsedPacket.header.markerBit = M;
+ parsedPacket.header.payloadType = PT;
+ parsedPacket.header.sequenceNumber = sequenceNumber;
+ parsedPacket.header.timestamp = RTPTimestamp;
+ parsedPacket.header.ssrc = SSRC;
+ parsedPacket.header.numCSRCs = CC;
+ parsedPacket.header.paddingLength = P ? *(_ptrRTPDataEnd - 1) : 0;
+
+ for (unsigned int i = 0; i < CC; ++i) {
+ WebRtc_UWord32 CSRC = *ptr++ << 24;
+ CSRC += *ptr++ << 16;
+ CSRC += *ptr++ << 8;
+ CSRC += *ptr++;
+ parsedPacket.header.arrOfCSRCs[i] = CSRC;
+ }
+ parsedPacket.type.Audio.numEnergy = parsedPacket.header.numCSRCs;
+
+ parsedPacket.header.headerLength = 12 + CSRCocts;
+
+ // If in effect, MAY be omitted for those packets for which the offset
+ // is zero.
+ parsedPacket.extension.transmissionTimeOffset = 0;
+
+ if (X) {
+ /* RTP header extension, RFC 3550.
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | defined by profile | length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | header extension |
+ | .... |
+ */
+ const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
+ if (remain < 4) {
+ return false;
+ }
+
+ parsedPacket.header.headerLength += 4;
+
+ WebRtc_UWord16 definedByProfile = *ptr++ << 8;
+ definedByProfile += *ptr++;
+
+ WebRtc_UWord16 XLen = *ptr++ << 8;
+ XLen += *ptr++; // in 32 bit words
+ XLen *= 4; // in octs
+
+ if (remain < (4 + XLen)) {
+ return false;
+ }
+ if (definedByProfile == RTP_ONE_BYTE_HEADER_EXTENSION) {
+ const WebRtc_UWord8* ptrRTPDataExtensionEnd = ptr + XLen;
+ ParseOneByteExtensionHeader(parsedPacket,
+ ptrExtensionMap,
+ ptrRTPDataExtensionEnd,
+ ptr);
+ }
+ parsedPacket.header.headerLength += XLen;
+ }
+ return true;
+}
+
+void RTPHeaderParser::ParseOneByteExtensionHeader(
+ WebRtcRTPHeader& parsedPacket,
+ const RtpHeaderExtensionMap* ptrExtensionMap,
+ const WebRtc_UWord8* ptrRTPDataExtensionEnd,
+ const WebRtc_UWord8* ptr) const {
+ if (!ptrExtensionMap) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, "No extension map.");
+ return;
+ }
+
+ while (ptrRTPDataExtensionEnd - ptr > 0) {
+ // 0
+ // 0 1 2 3 4 5 6 7
+ // +-+-+-+-+-+-+-+-+
+ // | ID | len |
+ // +-+-+-+-+-+-+-+-+
+
+ const WebRtc_UWord8 id = (*ptr & 0xf0) >> 4;
+ const WebRtc_UWord8 len = (*ptr & 0x0f);
+ ptr++;
+
+ if (id == 15) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "Ext id: 15 encountered, parsing terminated.");
+ return;
+ }
+
+ RTPExtensionType type;
+ if (ptrExtensionMap->GetType(id, &type) != 0) {
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+ "Failed to find extension id: %d", id);
+ return;
+ }
+
+ switch (type) {
+ case kRtpExtensionTransmissionTimeOffset: {
+ if (len != 2) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "Incorrect transmission time offset len: %d", len);
+ return;
+ }
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=2 | transmission offset |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ WebRtc_Word32 transmissionTimeOffset = *ptr++ << 16;
+ transmissionTimeOffset += *ptr++ << 8;
+ transmissionTimeOffset += *ptr++;
+ parsedPacket.extension.transmissionTimeOffset = transmissionTimeOffset;
+ break;
+ }
+ case kRtpExtensionAudioLevel: {
+ // --- Only used for debugging ---
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=0 |V| level | 0x00 | 0x00 |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ //
+
+ // Parse out the fields but only use it for debugging for now.
+ // const WebRtc_UWord8 V = (*ptr & 0x80) >> 7;
+ // const WebRtc_UWord8 level = (*ptr & 0x7f);
+ // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
+ // level=%u", ID, len, V, level);
+ break;
+ }
+ default: {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+ "Extension type not implemented.");
+ return;
+ }
+ }
+ WebRtc_UWord8 num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr);
+ ptr += num_bytes;
+ }
+}
+
+WebRtc_UWord8 RTPHeaderParser::ParsePaddingBytes(
+ const WebRtc_UWord8* ptrRTPDataExtensionEnd,
+ const WebRtc_UWord8* ptr) const {
+
+ WebRtc_UWord8 num_zero_bytes = 0;
+ while (ptrRTPDataExtensionEnd - ptr > 0) {
+ if (*ptr != 0) {
+ return num_zero_bytes;
+ }
+ ptr++;
+ num_zero_bytes++;
+ }
+ return num_zero_bytes;
+}
+
+// RTP payload parser
+RTPPayloadParser::RTPPayloadParser(const RtpVideoCodecTypes videoType,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadDataLength,
+ WebRtc_Word32 id)
+ :
+ _id(id),
+ _dataPtr(payloadData),
+ _dataLength(payloadDataLength),
+ _videoType(videoType) {
+}
+
+RTPPayloadParser::~RTPPayloadParser() {
+}
+
+bool RTPPayloadParser::Parse(RTPPayload& parsedPacket) const {
+ parsedPacket.SetType(_videoType);
+
+ switch (_videoType) {
+ case kRtpNoVideo:
+ return ParseGeneric(parsedPacket);
+ case kRtpVp8Video:
+ return ParseVP8(parsedPacket);
+ default:
+ return false;
+ }
+}
+
+bool RTPPayloadParser::ParseGeneric(RTPPayload& /*parsedPacket*/) const {
+ return false;
+}
+
+//
+// VP8 format:
+//
+// Payload descriptor
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |X|R|N|S|PartID | (REQUIRED)
+// +-+-+-+-+-+-+-+-+
+// X: |I|L|T|K| RSV | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// I: | PictureID | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// L: | TL0PICIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// T/K: |TID:Y| KEYIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+//
+// Payload header (considered part of the actual payload, sent to decoder)
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |Size0|H| VER |P|
+// +-+-+-+-+-+-+-+-+
+// | ... |
+// + +
+
+bool RTPPayloadParser::ParseVP8(RTPPayload& parsedPacket) const {
+ RTPPayloadVP8* vp8 = &parsedPacket.info.VP8;
+ const WebRtc_UWord8* dataPtr = _dataPtr;
+ int dataLength = _dataLength;
+
+ // Parse mandatory first byte of payload descriptor
+ bool extension = (*dataPtr & 0x80) ? true : false; // X bit
+ vp8->nonReferenceFrame = (*dataPtr & 0x20) ? true : false; // N bit
+ vp8->beginningOfPartition = (*dataPtr & 0x10) ? true : false; // S bit
+ vp8->partitionID = (*dataPtr & 0x0F); // PartID field
+
+ if (vp8->partitionID > 8) {
+ // Weak check for corrupt data: PartID MUST NOT be larger than 8.
+ return false;
+ }
+
+ // Advance dataPtr and decrease remaining payload size
+ dataPtr++;
+ dataLength--;
+
+ if (extension) {
+ const int parsedBytes = ParseVP8Extension(vp8, dataPtr, dataLength);
+ if (parsedBytes < 0) return false;
+ dataPtr += parsedBytes;
+ dataLength -= parsedBytes;
+ }
+
+ if (dataLength <= 0) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "Error parsing VP8 payload descriptor; payload too short");
+ return false;
+ }
+
+ // Read P bit from payload header (only at beginning of first partition)
+ if (dataLength > 0 && vp8->beginningOfPartition && vp8->partitionID == 0) {
+ parsedPacket.frameType = (*dataPtr & 0x01) ? kPFrame : kIFrame;
+ } else {
+ parsedPacket.frameType = kPFrame;
+ }
+ if (0 != ParseVP8FrameSize(parsedPacket, dataPtr, dataLength)) {
+ return false;
+ }
+ parsedPacket.info.VP8.data = dataPtr;
+ parsedPacket.info.VP8.dataLength = dataLength;
+ return true;
+}
+
+int RTPPayloadParser::ParseVP8FrameSize(RTPPayload& parsedPacket,
+ const WebRtc_UWord8* dataPtr,
+ int dataLength) const {
+ if (parsedPacket.frameType != kIFrame) {
+ // Included in payload header for I-frames.
+ return 0;
+ }
+ if (dataLength < 10) {
+ // For an I-frame we should always have the uncompressed VP8 header
+ // in the beginning of the partition.
+ return -1;
+ }
+ RTPPayloadVP8* vp8 = &parsedPacket.info.VP8;
+ vp8->frameWidth = ((dataPtr[7] << 8) + dataPtr[6]) & 0x3FFF;
+ vp8->frameHeight = ((dataPtr[9] << 8) + dataPtr[8]) & 0x3FFF;
+ return 0;
+}
+
+int RTPPayloadParser::ParseVP8Extension(RTPPayloadVP8* vp8,
+ const WebRtc_UWord8* dataPtr,
+ int dataLength) const {
+ int parsedBytes = 0;
+ if (dataLength <= 0) return -1;
+ // Optional X field is present
+ vp8->hasPictureID = (*dataPtr & 0x80) ? true : false; // I bit
+ vp8->hasTl0PicIdx = (*dataPtr & 0x40) ? true : false; // L bit
+ vp8->hasTID = (*dataPtr & 0x20) ? true : false; // T bit
+ vp8->hasKeyIdx = (*dataPtr & 0x10) ? true : false; // K bit
+
+ // Advance dataPtr and decrease remaining payload size
+ dataPtr++;
+ parsedBytes++;
+ dataLength--;
+
+ if (vp8->hasPictureID) {
+ if (ParseVP8PictureID(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
+ return -1;
+ }
+ }
+
+ if (vp8->hasTl0PicIdx) {
+ if (ParseVP8Tl0PicIdx(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
+ return -1;
+ }
+ }
+
+ if (vp8->hasTID || vp8->hasKeyIdx) {
+ if (ParseVP8TIDAndKeyIdx(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
+ return -1;
+ }
+ }
+ return parsedBytes;
+}
+
+int RTPPayloadParser::ParseVP8PictureID(RTPPayloadVP8* vp8,
+ const WebRtc_UWord8** dataPtr,
+ int* dataLength,
+ int* parsedBytes) const {
+ if (*dataLength <= 0) return -1;
+ vp8->pictureID = (**dataPtr & 0x7F);
+ if (**dataPtr & 0x80) {
+ (*dataPtr)++;
+ (*parsedBytes)++;
+ if (--(*dataLength) <= 0) return -1;
+ // PictureID is 15 bits
+ vp8->pictureID = (vp8->pictureID << 8) +** dataPtr;
+ }
+ (*dataPtr)++;
+ (*parsedBytes)++;
+ (*dataLength)--;
+ return 0;
+}
+
+int RTPPayloadParser::ParseVP8Tl0PicIdx(RTPPayloadVP8* vp8,
+ const WebRtc_UWord8** dataPtr,
+ int* dataLength,
+ int* parsedBytes) const {
+ if (*dataLength <= 0) return -1;
+ vp8->tl0PicIdx = **dataPtr;
+ (*dataPtr)++;
+ (*parsedBytes)++;
+ (*dataLength)--;
+ return 0;
+}
+
+int RTPPayloadParser::ParseVP8TIDAndKeyIdx(RTPPayloadVP8* vp8,
+ const WebRtc_UWord8** dataPtr,
+ int* dataLength,
+ int* parsedBytes) const {
+ if (*dataLength <= 0) return -1;
+ if (vp8->hasTID) {
+ vp8->tID = ((**dataPtr >> 6) & 0x03);
+ vp8->layerSync = (**dataPtr & 0x20) ? true : false; // Y bit
+ }
+ if (vp8->hasKeyIdx) {
+ vp8->keyIdx = (**dataPtr & 0x1F);
+ }
+ (*dataPtr)++;
+ (*parsedBytes)++;
+ (*dataLength)--;
+ return 0;
+}
+
+} // namespace ModuleRTPUtility
+
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility.h b/trunk/src/modules/rtp_rtcp/source/rtp_utility.h
new file mode 100644
index 0000000..ad88f11
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_utility.h
@@ -0,0 +1,245 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+
+#include <cstddef> // size_t, ptrdiff_t
+
+#include "typedefs.h"
+#include "rtp_header_extension.h"
+#include "rtp_rtcp_config.h"
+#include "rtp_rtcp_defines.h"
+
+namespace webrtc {
+enum RtpVideoCodecTypes
+{
+ kRtpNoVideo = 0,
+ kRtpFecVideo = 10,
+ kRtpVp8Video = 11
+};
+
+const WebRtc_UWord8 kRtpMarkerBitMask = 0x80;
+
+namespace ModuleRTPUtility
+{
+ // January 1970, in NTP seconds.
+ const uint32_t NTP_JAN_1970 = 2208988800UL;
+
+ // Magic NTP fractional unit.
+ const double NTP_FRAC = 4.294967296E+9;
+
+ struct AudioPayload
+ {
+ WebRtc_UWord32 frequency;
+ WebRtc_UWord8 channels;
+ WebRtc_UWord8 bitsPerSample;
+ WebRtc_UWord32 rate;
+ bool trueStereoCodec;
+ };
+ struct VideoPayload
+ {
+ RtpVideoCodecTypes videoCodecType;
+ WebRtc_UWord32 maxRate;
+ };
+ union PayloadUnion
+ {
+ AudioPayload Audio;
+ VideoPayload Video;
+ };
+ struct Payload
+ {
+ char name[RTP_PAYLOAD_NAME_SIZE];
+ bool audio;
+ PayloadUnion typeSpecific;
+ };
+
+ // Return a clock that reads the time as reported by the operating
+ // system. The returned instances are guaranteed to read the same
+ // times; in particular, they return relative times relative to
+ // the same base.
+ // Note that even though the instances returned by this function
+ // read the same times a new object is created every time this
+ // API is called. The ownership of this object belongs to the
+ // caller.
+ RtpRtcpClock* GetSystemClock();
+
+ // Return the current RTP timestamp from the NTP timestamp
+ // returned by the specified clock.
+ WebRtc_UWord32 GetCurrentRTP(RtpRtcpClock* clock, WebRtc_UWord32 freq);
+
+ // Return the current RTP absolute timestamp.
+ WebRtc_UWord32 ConvertNTPTimeToRTP(WebRtc_UWord32 NTPsec,
+ WebRtc_UWord32 NTPfrac,
+ WebRtc_UWord32 freq);
+
+ // Return the time in milliseconds corresponding to the specified
+ // NTP timestamp.
+ WebRtc_UWord32 ConvertNTPTimeToMS(WebRtc_UWord32 NTPsec,
+ WebRtc_UWord32 NTPfrac);
+
+ WebRtc_UWord32 pow2(WebRtc_UWord8 exp);
+
+ // Returns true if |newTimestamp| is older than |existingTimestamp|.
+ // |wrapped| will be set to true if there has been a wraparound between the
+ // two timestamps.
+ bool OldTimestamp(uint32_t newTimestamp,
+ uint32_t existingTimestamp,
+ bool* wrapped);
+
+ bool StringCompare(const char* str1,
+ const char* str2,
+ const WebRtc_UWord32 length);
+
+ void AssignUWord32ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value);
+ void AssignUWord24ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value);
+ void AssignUWord16ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord16 value);
+
+ /**
+ * Converts a network-ordered two-byte input buffer to a host-ordered value.
+ * \param[in] dataBuffer Network-ordered two-byte buffer to convert.
+ * \return Host-ordered value.
+ */
+ WebRtc_UWord16 BufferToUWord16(const WebRtc_UWord8* dataBuffer);
+
+ /**
+ * Converts a network-ordered three-byte input buffer to a host-ordered value.
+ * \param[in] dataBuffer Network-ordered three-byte buffer to convert.
+ * \return Host-ordered value.
+ */
+ WebRtc_UWord32 BufferToUWord24(const WebRtc_UWord8* dataBuffer);
+
+ /**
+ * Converts a network-ordered four-byte input buffer to a host-ordered value.
+ * \param[in] dataBuffer Network-ordered four-byte buffer to convert.
+ * \return Host-ordered value.
+ */
+ WebRtc_UWord32 BufferToUWord32(const WebRtc_UWord8* dataBuffer);
+
+ class RTPHeaderParser
+ {
+ public:
+ RTPHeaderParser(const WebRtc_UWord8* rtpData,
+ const WebRtc_UWord32 rtpDataLength);
+ ~RTPHeaderParser();
+
+ bool RTCP() const;
+ bool Parse(WebRtcRTPHeader& parsedPacket,
+ RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
+
+ private:
+ void ParseOneByteExtensionHeader(
+ WebRtcRTPHeader& parsedPacket,
+ const RtpHeaderExtensionMap* ptrExtensionMap,
+ const WebRtc_UWord8* ptrRTPDataExtensionEnd,
+ const WebRtc_UWord8* ptr) const;
+
+ WebRtc_UWord8 ParsePaddingBytes(
+ const WebRtc_UWord8* ptrRTPDataExtensionEnd,
+ const WebRtc_UWord8* ptr) const;
+
+ const WebRtc_UWord8* const _ptrRTPDataBegin;
+ const WebRtc_UWord8* const _ptrRTPDataEnd;
+ };
+
+ enum FrameTypes
+ {
+ kIFrame, // key frame
+ kPFrame // Delta frame
+ };
+
+ struct RTPPayloadVP8
+ {
+ bool nonReferenceFrame;
+ bool beginningOfPartition;
+ int partitionID;
+ bool hasPictureID;
+ bool hasTl0PicIdx;
+ bool hasTID;
+ bool hasKeyIdx;
+ int pictureID;
+ int tl0PicIdx;
+ int tID;
+ bool layerSync;
+ int keyIdx;
+ int frameWidth;
+ int frameHeight;
+
+ const WebRtc_UWord8* data;
+ WebRtc_UWord16 dataLength;
+ };
+
+ union RTPPayloadUnion
+ {
+ RTPPayloadVP8 VP8;
+ };
+
+ struct RTPPayload
+ {
+ void SetType(RtpVideoCodecTypes videoType);
+
+ RtpVideoCodecTypes type;
+ FrameTypes frameType;
+ RTPPayloadUnion info;
+ };
+
+ // RTP payload parser
+ class RTPPayloadParser
+ {
+ public:
+ RTPPayloadParser(const RtpVideoCodecTypes payloadType,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength, // Length w/o padding.
+ const WebRtc_Word32 id);
+
+ ~RTPPayloadParser();
+
+ bool Parse(RTPPayload& parsedPacket) const;
+
+ private:
+ bool ParseGeneric(RTPPayload& parsedPacket) const;
+
+ bool ParseVP8(RTPPayload& parsedPacket) const;
+
+ int ParseVP8Extension(RTPPayloadVP8 *vp8,
+ const WebRtc_UWord8 *dataPtr,
+ int dataLength) const;
+
+ int ParseVP8PictureID(RTPPayloadVP8 *vp8,
+ const WebRtc_UWord8 **dataPtr,
+ int *dataLength,
+ int *parsedBytes) const;
+
+ int ParseVP8Tl0PicIdx(RTPPayloadVP8 *vp8,
+ const WebRtc_UWord8 **dataPtr,
+ int *dataLength,
+ int *parsedBytes) const;
+
+ int ParseVP8TIDAndKeyIdx(RTPPayloadVP8 *vp8,
+ const WebRtc_UWord8 **dataPtr,
+ int *dataLength,
+ int *parsedBytes) const;
+
+ int ParseVP8FrameSize(RTPPayload& parsedPacket,
+ const WebRtc_UWord8 *dataPtr,
+ int dataLength) const;
+
+ private:
+ WebRtc_Word32 _id;
+ const WebRtc_UWord8* _dataPtr;
+ const WebRtc_UWord16 _dataLength;
+ const RtpVideoCodecTypes _videoType;
+ };
+
+} // namespace ModuleRTPUtility
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc
new file mode 100644
index 0000000..eabc812
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc
@@ -0,0 +1,288 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file conatins unit tests for the ModuleRTPUtility.
+ */
+
+#include "gtest/gtest.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+using ModuleRTPUtility::RTPPayloadParser;
+using ModuleRTPUtility::RTPPayload;
+using ModuleRTPUtility::RTPPayloadVP8;
+
+// Payload descriptor
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |X|R|N|S|PartID | (REQUIRED)
+// +-+-+-+-+-+-+-+-+
+// X: |I|L|T|K| RSV | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// I: | PictureID | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// L: | TL0PICIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+// T/K: |TID:Y| KEYIDX | (OPTIONAL)
+// +-+-+-+-+-+-+-+-+
+//
+// Payload header
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |Size0|H| VER |P|
+// +-+-+-+-+-+-+-+-+
+// | Size1 |
+// +-+-+-+-+-+-+-+-+
+// | Size2 |
+// +-+-+-+-+-+-+-+-+
+// | Bytes 4..N of |
+// | VP8 payload |
+// : :
+// +-+-+-+-+-+-+-+-+
+// | OPTIONAL RTP |
+// | padding |
+// : :
+// +-+-+-+-+-+-+-+-+
+
+void VerifyBasicHeader(const RTPPayloadVP8 &header,
+ bool N, bool S, int PartID) {
+ EXPECT_EQ(N, header.nonReferenceFrame);
+ EXPECT_EQ(S, header.beginningOfPartition);
+ EXPECT_EQ(PartID, header.partitionID);
+}
+
+void VerifyExtensions(const RTPPayloadVP8 &header,
+ bool I, bool L, bool T, bool K) {
+ EXPECT_EQ(I, header.hasPictureID);
+ EXPECT_EQ(L, header.hasTl0PicIdx);
+ EXPECT_EQ(T, header.hasTID);
+ EXPECT_EQ(K, header.hasKeyIdx);
+}
+
+TEST(ParseVP8Test, BasicHeader) {
+ WebRtc_UWord8 payload[4] = {0};
+ payload[0] = 0x14; // Binary 0001 0100; S = 1, PartID = 4.
+ payload[1] = 0x01; // P frame.
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 4, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 4 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
+
+ EXPECT_EQ(payload + 1, parsedPacket.info.VP8.data);
+ EXPECT_EQ(4 - 1, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, PictureID) {
+ WebRtc_UWord8 payload[10] = {0};
+ payload[0] = 0xA0;
+ payload[1] = 0x80;
+ payload[2] = 17;
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
+
+ EXPECT_EQ(17, parsedPacket.info.VP8.pictureID);
+
+ EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
+ EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
+
+
+ // Re-use payload, but change to long PictureID.
+ payload[2] = 0x80 | 17;
+ payload[3] = 17;
+ RTPPayloadParser rtpPayloadParser2(kRtpVp8Video, payload, 10, 0);
+
+ ASSERT_TRUE(rtpPayloadParser2.Parse(parsedPacket));
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
+
+ EXPECT_EQ((17<<8) + 17, parsedPacket.info.VP8.pictureID);
+
+ EXPECT_EQ(payload + 4, parsedPacket.info.VP8.data);
+ EXPECT_EQ(10 - 4, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, Tl0PicIdx) {
+ WebRtc_UWord8 payload[13] = {0};
+ payload[0] = 0x90;
+ payload[1] = 0x40;
+ payload[2] = 17;
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 13, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 0 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 1 /*L*/, 0 /*T*/, 0 /*K*/);
+
+ EXPECT_EQ(17, parsedPacket.info.VP8.tl0PicIdx);
+
+ EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
+ EXPECT_EQ(13 - 3, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, TIDAndLayerSync) {
+ WebRtc_UWord8 payload[10] = {0};
+ payload[0] = 0x88;
+ payload[1] = 0x20;
+ payload[2] = 0x80; // TID(2) + LayerSync(false)
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 1 /*T*/, 0 /*K*/);
+
+ EXPECT_EQ(2, parsedPacket.info.VP8.tID);
+ EXPECT_FALSE(parsedPacket.info.VP8.layerSync);
+
+ EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
+ EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, KeyIdx) {
+ WebRtc_UWord8 payload[10] = {0};
+ payload[0] = 0x88;
+ payload[1] = 0x10; // K = 1.
+ payload[2] = 0x11; // KEYIDX = 17 decimal.
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 0 /*T*/, 1 /*K*/);
+
+ EXPECT_EQ(17, parsedPacket.info.VP8.keyIdx);
+
+ EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
+ EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, MultipleExtensions) {
+ WebRtc_UWord8 payload[10] = {0};
+ payload[0] = 0x88;
+ payload[1] = 0x80 | 0x40 | 0x20 | 0x10;
+ payload[2] = 0x80 | 17; // PictureID, high 7 bits.
+ payload[3] = 17; // PictureID, low 8 bits.
+ payload[4] = 42; // Tl0PicIdx.
+ payload[5] = 0x40 | 0x20 | 0x11; // TID(1) + LayerSync(true) + KEYIDX(17).
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 1 /*L*/, 1 /*T*/, 1 /*K*/);
+
+ EXPECT_EQ((17<<8) + 17, parsedPacket.info.VP8.pictureID);
+ EXPECT_EQ(42, parsedPacket.info.VP8.tl0PicIdx);
+ EXPECT_EQ(1, parsedPacket.info.VP8.tID);
+ EXPECT_EQ(17, parsedPacket.info.VP8.keyIdx);
+
+ EXPECT_EQ(payload + 6, parsedPacket.info.VP8.data);
+ EXPECT_EQ(10 - 6, parsedPacket.info.VP8.dataLength);
+}
+
+TEST(ParseVP8Test, TooShortHeader) {
+ WebRtc_UWord8 payload[4] = {0};
+ payload[0] = 0x88;
+ payload[1] = 0x80 | 0x40 | 0x20 | 0x10; // All extensions are enabled...
+ payload[2] = 0x80 | 17; // ... but only 2 bytes PictureID is provided.
+ payload[3] = 17; // PictureID, low 8 bits.
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 4, 0);
+
+ RTPPayload parsedPacket;
+ EXPECT_FALSE(rtpPayloadParser.Parse(parsedPacket));
+}
+
+TEST(ParseVP8Test, TestWithPacketizer) {
+ WebRtc_UWord8 payload[10] = {0};
+ WebRtc_UWord8 packet[20] = {0};
+ RTPVideoHeaderVP8 inputHeader;
+ inputHeader.nonReference = true;
+ inputHeader.pictureId = 300;
+ inputHeader.temporalIdx = 1;
+ inputHeader.layerSync = false;
+ inputHeader.tl0PicIdx = kNoTl0PicIdx; // Disable.
+ inputHeader.keyIdx = 31;
+ RtpFormatVp8 packetizer(payload, 10, inputHeader, 20);
+ bool last;
+ int send_bytes;
+ ASSERT_EQ(0, packetizer.NextPacket(packet, &send_bytes, &last));
+ ASSERT_TRUE(last);
+
+ RTPPayloadParser rtpPayloadParser(kRtpVp8Video, packet, send_bytes, 0);
+
+ RTPPayload parsedPacket;
+ ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
+
+ EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType);
+ EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
+
+ VerifyBasicHeader(parsedPacket.info.VP8,
+ inputHeader.nonReference /*N*/,
+ 1 /*S*/,
+ 0 /*PartID*/);
+ VerifyExtensions(parsedPacket.info.VP8,
+ 1 /*I*/,
+ 0 /*L*/,
+ 1 /*T*/,
+ 1 /*K*/);
+
+ EXPECT_EQ(inputHeader.pictureId, parsedPacket.info.VP8.pictureID);
+ EXPECT_EQ(inputHeader.temporalIdx, parsedPacket.info.VP8.tID);
+ EXPECT_EQ(inputHeader.layerSync, parsedPacket.info.VP8.layerSync);
+ EXPECT_EQ(inputHeader.keyIdx, parsedPacket.info.VP8.keyIdx);
+
+ EXPECT_EQ(packet + 5, parsedPacket.info.VP8.data);
+ EXPECT_EQ(send_bytes - 5, parsedPacket.info.VP8.dataLength);
+}
+
+} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc b/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc
new file mode 100644
index 0000000..b3e9ab0
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ssrc_database.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#include <stdlib.h>
+#include <cassert>
+
+#ifdef _WIN32
+ #include <windows.h>
+ #include <MMSystem.h> //timeGetTime
+
+// TODO(hellner): investigate if it is necessary to disable these warnings.
+ #pragma warning(disable:4311)
+ #pragma warning(disable:4312)
+#else
+ #include <stdio.h>
+ #include <string.h>
+ #include <time.h>
+ #include <sys/time.h>
+#endif
+
+namespace webrtc {
+SSRCDatabase*
+SSRCDatabase::StaticInstance(CountOperation count_operation)
+{
+ SSRCDatabase* impl =
+ GetStaticInstance<SSRCDatabase>(count_operation);
+ return impl;
+}
+
+SSRCDatabase*
+SSRCDatabase::GetSSRCDatabase()
+{
+ return StaticInstance(kAddRef);
+}
+
+void
+SSRCDatabase::ReturnSSRCDatabase()
+{
+ StaticInstance(kRelease);
+}
+
+WebRtc_UWord32
+SSRCDatabase::CreateSSRC()
+{
+ CriticalSectionScoped lock(_critSect);
+
+ WebRtc_UWord32 ssrc = GenerateRandom();
+
+#ifndef WEBRTC_NO_STL
+
+ while(_ssrcMap.find(ssrc) != _ssrcMap.end())
+ {
+ ssrc = GenerateRandom();
+ }
+ _ssrcMap[ssrc] = 0;
+
+#else
+ if(_sizeOfSSRC <= _numberOfSSRC)
+ {
+ // allocate more space
+ const int newSize = _sizeOfSSRC + 10;
+ WebRtc_UWord32* tempSSRCVector = new WebRtc_UWord32[newSize];
+ memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(WebRtc_UWord32));
+ delete [] _ssrcVector;
+
+ _ssrcVector = tempSSRCVector;
+ _sizeOfSSRC = newSize;
+ }
+
+ // check if in DB
+ if(_ssrcVector)
+ {
+ for (int i=0; i<_numberOfSSRC; i++)
+ {
+ if (_ssrcVector[i] == ssrc)
+ {
+ // we have a match
+ i = 0; // start over with a new ssrc
+ ssrc = GenerateRandom();
+ }
+
+ }
+ // add to database
+ _ssrcVector[_numberOfSSRC] = ssrc;
+ _numberOfSSRC++;
+ }
+#endif
+ return ssrc;
+}
+
+WebRtc_Word32
+SSRCDatabase::RegisterSSRC(const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_critSect);
+
+#ifndef WEBRTC_NO_STL
+
+ _ssrcMap[ssrc] = 0;
+
+#else
+ if(_sizeOfSSRC <= _numberOfSSRC)
+ {
+ // allocate more space
+ const int newSize = _sizeOfSSRC + 10;
+ WebRtc_UWord32* tempSSRCVector = new WebRtc_UWord32[newSize];
+ memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(WebRtc_UWord32));
+ delete [] _ssrcVector;
+
+ _ssrcVector = tempSSRCVector;
+ _sizeOfSSRC = newSize;
+ }
+ // check if in DB
+ if(_ssrcVector)
+ {
+ for (int i=0; i<_numberOfSSRC; i++)
+ {
+ if (_ssrcVector[i] == ssrc)
+ {
+ // we have a match
+ return -1;
+ }
+ }
+ // add to database
+ _ssrcVector[_numberOfSSRC] = ssrc;
+ _numberOfSSRC++;
+ }
+#endif
+ return 0;
+}
+
+WebRtc_Word32
+SSRCDatabase::ReturnSSRC(const WebRtc_UWord32 ssrc)
+{
+ CriticalSectionScoped lock(_critSect);
+
+#ifndef WEBRTC_NO_STL
+ _ssrcMap.erase(ssrc);
+
+#else
+ if(_ssrcVector)
+ {
+ for (int i=0; i<_numberOfSSRC; i++)
+ {
+ if (_ssrcVector[i] == ssrc)
+ {
+ // we have a match
+ // remove from database
+ _ssrcVector[i] = _ssrcVector[_numberOfSSRC-1];
+ _numberOfSSRC--;
+ break;
+ }
+ }
+ }
+#endif
+ return 0;
+}
+
+SSRCDatabase::SSRCDatabase()
+{
+ // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
+#ifdef _WIN32
+ srand(timeGetTime());
+#else
+ struct timeval tv;
+ struct timezone tz;
+ gettimeofday(&tv, &tz);
+ srand(tv.tv_usec);
+#endif
+
+#ifdef WEBRTC_NO_STL
+ _sizeOfSSRC = 10;
+ _numberOfSSRC = 0;
+ _ssrcVector = new WebRtc_UWord32[10];
+#endif
+ _critSect = CriticalSectionWrapper::CreateCriticalSection();
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s created", __FUNCTION__);
+}
+
+SSRCDatabase::~SSRCDatabase()
+{
+#ifdef WEBRTC_NO_STL
+ delete [] _ssrcVector;
+#else
+ _ssrcMap.clear();
+#endif
+ delete _critSect;
+
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_UWord32 SSRCDatabase::GenerateRandom()
+{
+ WebRtc_UWord32 ssrc = 0;
+ do
+ {
+ ssrc = rand();
+ ssrc = ssrc <<16;
+ ssrc += rand();
+
+ } while (ssrc == 0 || ssrc == 0xffffffff);
+
+ return ssrc;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/ssrc_database.h b/trunk/src/modules/rtp_rtcp/source/ssrc_database.h
new file mode 100644
index 0000000..370e549
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/ssrc_database.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
+
+#ifndef WEBRTC_NO_STL
+ #include <map>
+#endif
+
+#include "system_wrappers/interface/static_instance.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class SSRCDatabase
+{
+public:
+ static SSRCDatabase* GetSSRCDatabase();
+ static void ReturnSSRCDatabase();
+
+ WebRtc_UWord32 CreateSSRC();
+ WebRtc_Word32 RegisterSSRC(const WebRtc_UWord32 ssrc);
+ WebRtc_Word32 ReturnSSRC(const WebRtc_UWord32 ssrc);
+
+protected:
+ SSRCDatabase();
+ virtual ~SSRCDatabase();
+
+ static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
+
+private:
+ // Friend function to allow the SSRC destructor to be accessed from the
+ // template class.
+ friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
+ CountOperation count_operation);
+ static SSRCDatabase* StaticInstance(CountOperation count_operation);
+
+ WebRtc_UWord32 GenerateRandom();
+
+#ifdef WEBRTC_NO_STL
+ int _numberOfSSRC;
+ int _sizeOfSSRC;
+
+ WebRtc_UWord32* _ssrcVector;
+#else
+ std::map<WebRtc_UWord32, WebRtc_UWord32> _ssrcMap;
+#endif
+
+ CriticalSectionWrapper* _critSect;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc
new file mode 100644
index 0000000..cf34e08
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -0,0 +1,507 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tmmbr_help.h"
+
+#include "rtp_rtcp_config.h"
+
+namespace webrtc {
+TMMBRSet::TMMBRSet() :
+ ptrTmmbrSet(0),
+ ptrPacketOHSet(0),
+ ptrSsrcSet(0),
+ sizeOfSet(0),
+ lengthOfSet(0)
+{
+}
+
+TMMBRSet::~TMMBRSet()
+{
+ delete [] ptrTmmbrSet;
+ delete [] ptrPacketOHSet;
+ delete [] ptrSsrcSet;
+ ptrTmmbrSet = 0;
+ ptrPacketOHSet = 0;
+ ptrSsrcSet = 0;
+ sizeOfSet = 0;
+ lengthOfSet = 0;
+}
+
+void
+TMMBRSet::VerifyAndAllocateSet(WebRtc_UWord32 minimumSize)
+{
+ if(minimumSize > sizeOfSet)
+ {
+ // make sure that our buffers are big enough
+ if(ptrTmmbrSet)
+ {
+ delete [] ptrTmmbrSet;
+ delete [] ptrPacketOHSet;
+ delete [] ptrSsrcSet;
+ }
+ ptrTmmbrSet = new WebRtc_UWord32[minimumSize];
+ ptrPacketOHSet = new WebRtc_UWord32[minimumSize];
+ ptrSsrcSet = new WebRtc_UWord32[minimumSize];
+ sizeOfSet = minimumSize;
+ }
+ // reset memory
+ for(WebRtc_UWord32 i = 0; i < sizeOfSet; i++)
+ {
+ ptrTmmbrSet[i] = 0;
+ ptrPacketOHSet[i] = 0;
+ ptrSsrcSet[i] = 0;
+ }
+ lengthOfSet = 0;
+}
+
+TMMBRHelp::TMMBRHelp(const bool audio) :
+ _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
+ _audio(audio),
+ _candidateSet(),
+ _boundingSet(),
+ _boundingSetToSend(),
+ _ptrIntersectionBoundingSet(NULL),
+ _ptrMaxPRBoundingSet(NULL)
+{
+}
+
+TMMBRHelp::~TMMBRHelp()
+{
+ delete [] _ptrIntersectionBoundingSet;
+ delete [] _ptrMaxPRBoundingSet;
+ _ptrIntersectionBoundingSet = 0;
+ _ptrMaxPRBoundingSet = 0;
+ delete _criticalSection;
+}
+
+TMMBRSet*
+TMMBRHelp::VerifyAndAllocateBoundingSet(WebRtc_UWord32 minimumSize)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ if(minimumSize > _boundingSet.sizeOfSet)
+ {
+ // make sure that our buffers are big enough
+ if(_ptrIntersectionBoundingSet)
+ {
+ delete [] _ptrIntersectionBoundingSet;
+ delete [] _ptrMaxPRBoundingSet;
+ }
+ _ptrIntersectionBoundingSet = new float[minimumSize];
+ _ptrMaxPRBoundingSet = new float[minimumSize];
+ }
+ _boundingSet.VerifyAndAllocateSet(minimumSize);
+ return &_boundingSet;
+}
+
+TMMBRSet*
+TMMBRHelp::BoundingSet()
+{
+ return &_boundingSet;
+}
+
+WebRtc_Word32
+TMMBRHelp::SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend,
+ const WebRtc_UWord32 maxBitrateKbit)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ if (boundingSetToSend == NULL)
+ {
+ _boundingSetToSend.lengthOfSet = 0;
+ return 0;
+ }
+
+ VerifyAndAllocateBoundingSetToSend(boundingSetToSend->lengthOfSet);
+
+ for (WebRtc_UWord32 i = 0; i < boundingSetToSend->lengthOfSet; i++)
+ {
+ // cap at our configured max bitrate
+ WebRtc_UWord32 bitrate = boundingSetToSend->ptrTmmbrSet[i];
+ if(maxBitrateKbit)
+ {
+ // do we have a configured max bitrate?
+ if(bitrate > maxBitrateKbit)
+ {
+ bitrate = maxBitrateKbit;
+ }
+ }
+
+ _boundingSetToSend.ptrTmmbrSet[i] = bitrate;
+ _boundingSetToSend.ptrPacketOHSet[i] = boundingSetToSend->ptrPacketOHSet[i];
+ _boundingSetToSend.ptrSsrcSet[i] = boundingSetToSend->ptrSsrcSet[i];
+ }
+ _boundingSetToSend.lengthOfSet = boundingSetToSend->lengthOfSet;
+ return 0;
+}
+
+WebRtc_Word32
+TMMBRHelp::VerifyAndAllocateBoundingSetToSend(WebRtc_UWord32 minimumSize)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ _boundingSetToSend.VerifyAndAllocateSet(minimumSize);
+ return 0;
+}
+
+TMMBRSet*
+TMMBRHelp::VerifyAndAllocateCandidateSet(WebRtc_UWord32 minimumSize)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ _candidateSet.VerifyAndAllocateSet(minimumSize);
+ return &_candidateSet;
+}
+
+TMMBRSet*
+TMMBRHelp::CandidateSet()
+{
+ return &_candidateSet;
+}
+
+TMMBRSet*
+TMMBRHelp::BoundingSetToSend()
+{
+ return &_boundingSetToSend;
+}
+
+WebRtc_Word32
+TMMBRHelp::FindTMMBRBoundingSet(TMMBRSet*& boundingSet)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ // Work on local variable, will be modified
+ TMMBRSet candidateSet;
+ candidateSet.VerifyAndAllocateSet(_candidateSet.sizeOfSet);
+
+ // Number of set candidates
+ WebRtc_Word32 numSetCandidates = 0;
+ for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; i++)
+ {
+ if(_candidateSet.ptrTmmbrSet[i])
+ {
+ numSetCandidates++;
+ candidateSet.ptrTmmbrSet[i] = _candidateSet.ptrTmmbrSet[i];
+ candidateSet.ptrPacketOHSet[i] = _candidateSet.ptrPacketOHSet[i];
+ candidateSet.ptrSsrcSet[i] = _candidateSet.ptrSsrcSet[i];
+ }
+ else
+ {
+ // make sure this is zero if tmmbr = 0
+ _candidateSet.ptrPacketOHSet[i] = 0;
+ }
+ }
+ candidateSet.lengthOfSet = numSetCandidates;
+
+ // Find bounding set
+ WebRtc_UWord32 numBoundingSet = 0;
+ if (numSetCandidates > 0)
+ {
+ numBoundingSet = FindTMMBRBoundingSet(numSetCandidates, candidateSet);
+ if(numBoundingSet < 1 || (numBoundingSet > _candidateSet.sizeOfSet))
+ {
+ return -1;
+ }
+ boundingSet = &_boundingSet;
+ }
+ return numBoundingSet;
+}
+
+
+WebRtc_Word32
+TMMBRHelp::FindTMMBRBoundingSet(WebRtc_Word32 numCandidates, TMMBRSet& candidateSet)
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ WebRtc_UWord32 numBoundingSet = 0;
+ VerifyAndAllocateBoundingSet(candidateSet.sizeOfSet);
+
+ if (numCandidates == 1)
+ {
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if(candidateSet.ptrTmmbrSet[i] > 0)
+ {
+ _boundingSet.ptrTmmbrSet[numBoundingSet] = candidateSet.ptrTmmbrSet[i];
+ _boundingSet.ptrPacketOHSet[numBoundingSet] = candidateSet.ptrPacketOHSet[i];
+ _boundingSet.ptrSsrcSet[numBoundingSet] = candidateSet.ptrSsrcSet[i];
+ numBoundingSet++;
+ }
+ }
+ if (numBoundingSet != 1)
+ {
+ numBoundingSet = -1;
+ }
+ } else
+ {
+ // 1. Sort by increasing packetOH
+ WebRtc_UWord32 temp;
+ for (int i = candidateSet.sizeOfSet - 1; i >= 0; i--)
+ {
+ for (int j = 1; j <= i; j++)
+ {
+ if (candidateSet.ptrPacketOHSet[j-1] > candidateSet.ptrPacketOHSet[j])
+ {
+ temp = candidateSet.ptrPacketOHSet[j-1];
+ candidateSet.ptrPacketOHSet[j-1] = candidateSet.ptrPacketOHSet[j];
+ candidateSet.ptrPacketOHSet[j] = temp;
+ temp = candidateSet.ptrTmmbrSet[j-1];
+ candidateSet.ptrTmmbrSet[j-1] = candidateSet.ptrTmmbrSet[j];
+ candidateSet.ptrTmmbrSet[j] = temp;
+ temp = candidateSet.ptrSsrcSet[j-1];
+ candidateSet.ptrSsrcSet[j-1] = candidateSet.ptrSsrcSet[j];
+ candidateSet.ptrSsrcSet[j] = temp;
+ }
+ }
+ }
+ // 2. For tuples with same OH, keep the one w/ the lowest bitrate
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if (candidateSet.ptrTmmbrSet[i] > 0)
+ {
+ // get min bitrate for packets w/ same OH
+ WebRtc_UWord32 currentPacketOH = candidateSet.ptrPacketOHSet[i];
+ WebRtc_UWord32 currentMinTMMBR = candidateSet.ptrTmmbrSet[i];
+ WebRtc_UWord32 currentMinIndexTMMBR = i;
+ for (WebRtc_UWord32 j = i+1; j < candidateSet.sizeOfSet; j++)
+ {
+ if(candidateSet.ptrPacketOHSet[j] == currentPacketOH)
+ {
+ if(candidateSet.ptrTmmbrSet[j] < currentMinTMMBR)
+ {
+ currentMinTMMBR = candidateSet.ptrTmmbrSet[j];
+ currentMinIndexTMMBR = j;
+ }
+ }
+ }
+ // keep lowest bitrate
+ for (WebRtc_UWord32 j = 0; j < candidateSet.sizeOfSet; j++)
+ {
+ if(candidateSet.ptrPacketOHSet[j] == currentPacketOH && j != currentMinIndexTMMBR)
+ {
+ candidateSet.ptrTmmbrSet[j] = 0;
+ candidateSet.ptrPacketOHSet[j] = 0;
+ candidateSet.ptrSsrcSet[j] = 0;
+ numCandidates--;
+ }
+ }
+ }
+ }
+ // 3. Select and remove tuple w/ lowest tmmbr. (If more than 1, choose the one w/ highest OH).
+ WebRtc_UWord32 minTMMBR = 0;
+ WebRtc_UWord32 minIndexTMMBR = 0;
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if (candidateSet.ptrTmmbrSet[i] > 0)
+ {
+ minTMMBR = candidateSet.ptrTmmbrSet[i];
+ minIndexTMMBR = i;
+ break;
+ }
+ }
+
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if (candidateSet.ptrTmmbrSet[i] > 0 && candidateSet.ptrTmmbrSet[i] <= minTMMBR)
+ {
+ // get min bitrate
+ minTMMBR = candidateSet.ptrTmmbrSet[i];
+ minIndexTMMBR = i;
+ }
+ }
+ // first member of selected list
+ _boundingSet.ptrTmmbrSet[numBoundingSet] = candidateSet.ptrTmmbrSet[minIndexTMMBR];
+ _boundingSet.ptrPacketOHSet[numBoundingSet] = candidateSet.ptrPacketOHSet[minIndexTMMBR];
+ _boundingSet.ptrSsrcSet[numBoundingSet] = candidateSet.ptrSsrcSet[minIndexTMMBR];
+ // set intersection value
+ _ptrIntersectionBoundingSet[numBoundingSet] = 0;
+ // calculate its maximum packet rate (where its line crosses x-axis)
+ _ptrMaxPRBoundingSet[numBoundingSet] = _boundingSet.ptrTmmbrSet[numBoundingSet]*1000 / float(8*_boundingSet.ptrPacketOHSet[numBoundingSet]);
+ numBoundingSet++;
+ // remove from candidate list
+ candidateSet.ptrTmmbrSet[minIndexTMMBR] = 0;
+ candidateSet.ptrPacketOHSet[minIndexTMMBR] = 0;
+ candidateSet.ptrSsrcSet[minIndexTMMBR] = 0;
+ numCandidates--;
+
+ // 4. Discard from candidate list all tuple w/ lower OH (next tuple must be steeper)
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if(candidateSet.ptrTmmbrSet[i] > 0 && candidateSet.ptrPacketOHSet[i] < _boundingSet.ptrPacketOHSet[0])
+ {
+ candidateSet.ptrTmmbrSet[i] = 0;
+ candidateSet.ptrPacketOHSet[i] = 0;
+ candidateSet.ptrSsrcSet[i] = 0;
+ numCandidates--;
+ }
+ }
+
+ if (numCandidates == 0)
+ {
+ _boundingSet.lengthOfSet = numBoundingSet;
+ return numBoundingSet;
+ }
+
+ bool getNewCandidate = true;
+ int curCandidateTMMBR = 0;
+ int curCandidateIndex = 0;
+ int curCandidatePacketOH = 0;
+ int curCandidateSSRC = 0;
+ do
+ {
+ if (getNewCandidate)
+ {
+ // 5. Remove first remaining tuple from candidate list
+ for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
+ {
+ if (candidateSet.ptrTmmbrSet[i] > 0)
+ {
+ curCandidateTMMBR = candidateSet.ptrTmmbrSet[i];
+ curCandidatePacketOH = candidateSet.ptrPacketOHSet[i];
+ curCandidateSSRC = candidateSet.ptrSsrcSet[i];
+ curCandidateIndex = i;
+ candidateSet.ptrTmmbrSet[curCandidateIndex] = 0;
+ candidateSet.ptrPacketOHSet[curCandidateIndex] = 0;
+ candidateSet.ptrSsrcSet[curCandidateIndex] = 0;
+ break;
+ }
+ }
+ }
+
+ // 6. Calculate packet rate and intersection of the current line with line of last tuple in selected list
+ float packetRate = float(curCandidateTMMBR - _boundingSet.ptrTmmbrSet[numBoundingSet-1])*1000 / (8*(curCandidatePacketOH - _boundingSet.ptrPacketOHSet[numBoundingSet-1]));
+
+ // 7. If the packet rate is equal or lower than intersection of last tuple in selected list,
+ // remove last tuple in selected list & go back to step 6
+ if(packetRate <= _ptrIntersectionBoundingSet[numBoundingSet-1])
+ {
+ // remove last tuple and goto step 6
+ numBoundingSet--;
+ _boundingSet.ptrTmmbrSet[numBoundingSet] = 0;
+ _boundingSet.ptrPacketOHSet[numBoundingSet] = 0;
+ _boundingSet.ptrSsrcSet[numBoundingSet] = 0;
+ _ptrIntersectionBoundingSet[numBoundingSet] = 0;
+ _ptrMaxPRBoundingSet[numBoundingSet] = 0;
+ getNewCandidate = false;
+ } else
+ {
+ // 8. If packet rate is lower than maximum packet rate of last tuple in selected list, add current tuple to selected list
+ if (packetRate < _ptrMaxPRBoundingSet[numBoundingSet-1])
+ {
+ _boundingSet.ptrTmmbrSet[numBoundingSet] = curCandidateTMMBR;
+ _boundingSet.ptrPacketOHSet[numBoundingSet] = curCandidatePacketOH;
+ _boundingSet.ptrSsrcSet[numBoundingSet] = curCandidateSSRC;
+ _ptrIntersectionBoundingSet[numBoundingSet] = packetRate;
+ _ptrMaxPRBoundingSet[numBoundingSet] = _boundingSet.ptrTmmbrSet[numBoundingSet]*1000 / float(8*_boundingSet.ptrPacketOHSet[numBoundingSet]);
+ numBoundingSet++;
+ }
+ numCandidates--;
+ getNewCandidate = true;
+ }
+
+ // 9. Go back to step 5 if any tuple remains in candidate list
+ } while (numCandidates > 0);
+ }
+ _boundingSet.lengthOfSet = numBoundingSet;
+ return numBoundingSet;
+}
+
+bool
+TMMBRHelp::IsOwner(const WebRtc_UWord32 ssrc,
+ const WebRtc_UWord32 length) const
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ if (length == 0)
+ {
+ // empty bounding set
+ return false;
+ }
+
+ for(WebRtc_UWord32 i = 0; (i < length) && (i < _boundingSet.sizeOfSet); ++i)
+ {
+ if(_boundingSet.ptrSsrcSet[i] == ssrc)
+ {
+ return true;
+ }
+ }
+ return false;
+}
+
+WebRtc_Word32
+TMMBRHelp::CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
+ const WebRtc_UWord32 lengthOfBoundingSet,
+ WebRtc_UWord32& minBitrateKbit,
+ WebRtc_UWord32& maxBitrateKbit) const
+{
+ CriticalSectionScoped lock(_criticalSection);
+
+ if (lengthOfBoundingSet <= 0 || _candidateSet.sizeOfSet == 0)
+ {
+ // empty bounding set
+ return -1;
+ }
+
+ minBitrateKbit = 0xFFFFFFFF;
+ maxBitrateKbit = 0;
+
+ for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; ++i)
+ {
+ if (_candidateSet.ptrTmmbrSet[i])
+ {
+ WebRtc_Word32 curNetBitRate = static_cast<WebRtc_Word32>((_candidateSet.ptrTmmbrSet[i]*1000.0
+ - (totalPacketRate * (_candidateSet.ptrPacketOHSet[i] << 3)))/1000 + 0.5);
+
+ if (curNetBitRate < 0)
+ {
+ // could be negative for a large packet rate
+ if(_audio)
+ {
+ curNetBitRate = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
+ }else
+ {
+ curNetBitRate = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
+ }
+ }
+ minBitrateKbit = (WebRtc_UWord32(curNetBitRate) < minBitrateKbit) ? curNetBitRate : minBitrateKbit;
+ }
+ }
+ maxBitrateKbit = minBitrateKbit;
+
+ if (maxBitrateKbit == 0 || maxBitrateKbit < minBitrateKbit)
+ {
+ return -1;
+ }
+
+ if(_audio)
+ {
+ if (minBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
+ {
+ minBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
+ }
+ if (maxBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
+ {
+ maxBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
+ }
+ }else
+ {
+ if (minBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
+ {
+ minBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
+ }
+ if (maxBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
+ {
+ maxBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
+ }
+ }
+
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h
new file mode 100644
index 0000000..35704fe
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
+
+#include "typedefs.h"
+
+#include "critical_section_wrapper.h"
+
+#ifndef NULL
+ #define NULL 0
+#endif
+
+namespace webrtc {
+class TMMBRSet
+{
+public:
+ TMMBRSet();
+ ~TMMBRSet();
+
+ void VerifyAndAllocateSet(WebRtc_UWord32 minimumSize);
+
+ WebRtc_UWord32* ptrTmmbrSet;
+ WebRtc_UWord32* ptrPacketOHSet;
+ WebRtc_UWord32* ptrSsrcSet;
+ WebRtc_UWord32 sizeOfSet;
+ WebRtc_UWord32 lengthOfSet;
+};
+
+class TMMBRHelp
+{
+public:
+ TMMBRHelp(const bool audio);
+ virtual ~TMMBRHelp();
+
+ TMMBRSet* BoundingSet(); // used for debuging
+ TMMBRSet* CandidateSet();
+ TMMBRSet* BoundingSetToSend();
+
+ TMMBRSet* VerifyAndAllocateCandidateSet(const WebRtc_UWord32 minimumSize);
+ WebRtc_Word32 FindTMMBRBoundingSet(TMMBRSet*& boundingSet);
+ WebRtc_Word32 SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend,
+ const WebRtc_UWord32 maxBitrateKbit);
+
+ bool IsOwner(const WebRtc_UWord32 ssrc,
+ const WebRtc_UWord32 length) const;
+
+ WebRtc_Word32 CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
+ const WebRtc_UWord32 lengthOfBoundingSet,
+ WebRtc_UWord32& minBitrateKbit,
+ WebRtc_UWord32& maxBitrateKbit) const;
+
+protected:
+ TMMBRSet* VerifyAndAllocateBoundingSet(WebRtc_UWord32 minimumSize);
+ WebRtc_Word32 VerifyAndAllocateBoundingSetToSend(WebRtc_UWord32 minimumSize);
+
+ WebRtc_Word32 FindTMMBRBoundingSet(WebRtc_Word32 numCandidates, TMMBRSet& candidateSet);
+
+private:
+ CriticalSectionWrapper* _criticalSection;
+ const bool _audio;
+ TMMBRSet _candidateSet;
+ TMMBRSet _boundingSet;
+ TMMBRSet _boundingSetToSend;
+
+ float* _ptrIntersectionBoundingSet;
+ float* _ptrMaxPRBoundingSet;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc
new file mode 100644
index 0000000..e79d227
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "transmission_bucket.h"
+
+#include <assert.h>
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+
+TransmissionBucket::TransmissionBucket()
+ : critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ accumulator_(0),
+ bytes_rem_total_(0),
+ bytes_rem_interval_(0),
+ packets_(),
+ first_(true) {
+}
+
+TransmissionBucket::~TransmissionBucket() {
+ packets_.clear();
+ delete critsect_;
+}
+
+void TransmissionBucket::Reset() {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ accumulator_ = 0;
+ bytes_rem_total_ = 0;
+ bytes_rem_interval_ = 0;
+ packets_.clear();
+ first_ = true;
+}
+
+void TransmissionBucket::Fill(const uint16_t seq_num,
+ const uint32_t num_bytes) {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ accumulator_ += num_bytes;
+
+ Packet p(seq_num, num_bytes);
+ packets_.push_back(p);
+}
+
+bool TransmissionBucket::Empty() {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+ return packets_.empty();
+}
+
+void TransmissionBucket::UpdateBytesPerInterval(
+ const uint32_t delta_time_ms,
+ const uint16_t target_bitrate_kbps) {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+
+ const float kMargin = 1.05f;
+ uint32_t bytes_per_interval =
+ kMargin * (target_bitrate_kbps * delta_time_ms / 8);
+
+ if (bytes_rem_interval_ < 0) {
+ bytes_rem_interval_ += bytes_per_interval;
+ } else {
+ bytes_rem_interval_ = bytes_per_interval;
+ }
+
+ if (accumulator_) {
+ bytes_rem_total_ += bytes_per_interval;
+ return;
+ }
+ bytes_rem_total_ = bytes_per_interval;
+}
+
+int32_t TransmissionBucket::GetNextPacket() {
+ webrtc::CriticalSectionScoped cs(*critsect_);
+
+ if (accumulator_ == 0) {
+ // Empty.
+ return -1;
+ }
+
+ std::vector<Packet>::const_iterator it_begin = packets_.begin();
+ const uint16_t num_bytes = (*it_begin).length_;
+ const uint16_t seq_num = (*it_begin).sequence_number_;
+
+ if (first_) {
+ // Ok to transmit first packet.
+ first_ = false;
+ packets_.erase(packets_.begin());
+ return seq_num;
+ }
+
+ const float kFrameComplete = 0.80f;
+ if (num_bytes * kFrameComplete > bytes_rem_total_) {
+ // Packet does not fit.
+ return -1;
+ }
+
+ if (bytes_rem_interval_ <= 0) {
+ // All bytes consumed for this interval.
+ return -1;
+ }
+
+ // Ok to transmit packet.
+ bytes_rem_total_ -= num_bytes;
+ bytes_rem_interval_ -= num_bytes;
+
+ assert(accumulator_ >= num_bytes);
+ accumulator_ -= num_bytes;
+
+ packets_.erase(packets_.begin());
+ return seq_num;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h
new file mode 100644
index 0000000..79e45d8
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
+
+#include <vector>
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+class TransmissionBucket {
+ public:
+ TransmissionBucket();
+ ~TransmissionBucket();
+
+ // Resets members to initial state.
+ void Reset();
+
+ // Adds packet to be sent.
+ void Fill(const uint16_t seq_num, const uint32_t num_bytes);
+
+ // Returns true if there is no packet to be sent.
+ bool Empty();
+
+ // Updates the number of bytes that can be sent for the next time interval.
+ void UpdateBytesPerInterval(const uint32_t delta_time_in_ms,
+ const uint16_t target_bitrate_kbps);
+
+ // Checks if next packet in line can be transmitted. Returns the sequence
+ // number of the packet on success, -1 otherwise. The packet is removed from
+ // the vector on success.
+ int32_t GetNextPacket();
+
+ private:
+ struct Packet {
+ Packet(uint16_t sequence_number, uint16_t length_in_bytes)
+ : sequence_number_(sequence_number),
+ length_(length_in_bytes) {
+ }
+ uint16_t sequence_number_;
+ uint16_t length_;
+ };
+
+ CriticalSectionWrapper* critsect_;
+ uint32_t accumulator_;
+ int32_t bytes_rem_total_;
+ int32_t bytes_rem_interval_;
+ std::vector<Packet> packets_;
+ bool first_;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc b/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc
new file mode 100644
index 0000000..a8c9247
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes unit tests for the TransmissionBucket.
+ */
+
+#include <gtest/gtest.h>
+
+#include "transmission_bucket.h"
+
+namespace webrtc {
+
+class TransmissionBucketTest : public ::testing::Test {
+ protected:
+ TransmissionBucket send_bucket_;
+};
+
+TEST_F(TransmissionBucketTest, Fill) {
+ EXPECT_TRUE(send_bucket_.Empty());
+ send_bucket_.Fill(1, 100);
+ EXPECT_FALSE(send_bucket_.Empty());
+}
+
+TEST_F(TransmissionBucketTest, Reset) {
+ send_bucket_.Fill(1, 100);
+ EXPECT_FALSE(send_bucket_.Empty());
+ send_bucket_.Reset();
+ EXPECT_TRUE(send_bucket_.Empty());
+}
+
+TEST_F(TransmissionBucketTest, GetNextPacket) {
+ EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // empty
+ send_bucket_.Fill(1234, 100);
+ EXPECT_EQ(1234, send_bucket_.GetNextPacket()); // first packet ok
+ send_bucket_.Fill(1235, 100);
+ EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // packet does not fit
+}
+
+TEST_F(TransmissionBucketTest, UpdateBytesPerInterval) {
+ const int delta_time_ms = 1;
+ const int target_bitrate_kbps = 800;
+ send_bucket_.UpdateBytesPerInterval(delta_time_ms, target_bitrate_kbps);
+
+ send_bucket_.Fill(1234, 50);
+ send_bucket_.Fill(1235, 50);
+ send_bucket_.Fill(1236, 50);
+
+ EXPECT_EQ(1234, send_bucket_.GetNextPacket()); // first packet ok
+ EXPECT_EQ(1235, send_bucket_.GetNextPacket()); // ok
+ EXPECT_EQ(1236, send_bucket_.GetNextPacket()); // ok
+ EXPECT_TRUE(send_bucket_.Empty());
+
+ send_bucket_.Fill(1237, 50);
+ EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // packet does not fit
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/video_codec_information.h b/trunk/src/modules/rtp_rtcp/source/video_codec_information.h
new file mode 100644
index 0000000..4364f0b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/video_codec_information.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
+
+#include "rtp_rtcp_config.h"
+#include "rtp_utility.h"
+
+namespace webrtc {
+class VideoCodecInformation
+{
+public:
+ virtual void Reset() = 0;
+
+ virtual RtpVideoCodecTypes Type() = 0;
+ virtual ~VideoCodecInformation(){};
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
new file mode 100644
index 0000000..6d5bb75
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
@@ -0,0 +1,268 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+
+#include <assert.h>
+#include <stdlib.h> // NULL
+
+#include <algorithm>
+#include <limits>
+
+namespace webrtc {
+
+PartitionTreeNode::PartitionTreeNode(PartitionTreeNode* parent,
+ const int* size_vector,
+ int num_partitions,
+ int this_size)
+ : parent_(parent),
+ this_size_(this_size),
+ size_vector_(size_vector),
+ num_partitions_(num_partitions),
+ max_parent_size_(0),
+ min_parent_size_(std::numeric_limits<int>::max()),
+ packet_start_(false) {
+ assert(num_partitions >= 0);
+ children_[kLeftChild] = NULL;
+ children_[kRightChild] = NULL;
+}
+
+PartitionTreeNode* PartitionTreeNode::CreateRootNode(const int* size_vector,
+ int num_partitions) {
+ PartitionTreeNode* root_node =
+ new PartitionTreeNode(NULL, &size_vector[1], num_partitions - 1,
+ size_vector[0]);
+ root_node->set_packet_start(true);
+ return root_node;
+}
+
+PartitionTreeNode::~PartitionTreeNode() {
+ delete children_[kLeftChild];
+ delete children_[kRightChild];
+}
+
+int PartitionTreeNode::Cost(int penalty) {
+ assert(penalty >= 0);
+ int cost = 0;
+ if (num_partitions_ == 0) {
+ // This is a solution node.
+ cost = std::max(max_parent_size_, this_size_) -
+ std::min(min_parent_size_, this_size_);
+ } else {
+ cost = std::max(max_parent_size_, this_size_) - min_parent_size_;
+ }
+ return cost + NumPackets() * penalty;
+}
+
+bool PartitionTreeNode::CreateChildren(int max_size) {
+ assert(max_size > 0);
+ bool children_created = false;
+ if (num_partitions_ > 0) {
+ if (this_size_ + size_vector_[0] <= max_size) {
+ assert(!children_[kLeftChild]);
+ children_[kLeftChild] =
+ new PartitionTreeNode(this,
+ &size_vector_[1],
+ num_partitions_ - 1,
+ this_size_ + size_vector_[0]);
+ children_[kLeftChild]->set_max_parent_size(max_parent_size_);
+ children_[kLeftChild]->set_min_parent_size(min_parent_size_);
+ // "Left" child is continuation of same packet.
+ children_[kLeftChild]->set_packet_start(false);
+ children_created = true;
+ }
+ if (this_size_ > 0) {
+ assert(!children_[kRightChild]);
+ children_[kRightChild] = new PartitionTreeNode(this,
+ &size_vector_[1],
+ num_partitions_ - 1,
+ size_vector_[0]);
+ children_[kRightChild]->set_max_parent_size(
+ std::max(max_parent_size_, this_size_));
+ children_[kRightChild]->set_min_parent_size(
+ std::min(min_parent_size_, this_size_));
+ // "Right" child starts a new packet.
+ children_[kRightChild]->set_packet_start(true);
+ children_created = true;
+ }
+ }
+ return children_created;
+}
+
+int PartitionTreeNode::NumPackets() {
+ if (parent_ == NULL) {
+ // Root node is a "right" child by definition.
+ return 1;
+ }
+ if (parent_->children_[kLeftChild] == this) {
+ // This is a "left" child.
+ return parent_->NumPackets();
+ } else {
+ // This is a "right" child.
+ return 1 + parent_->NumPackets();
+ }
+}
+
+PartitionTreeNode* PartitionTreeNode::GetOptimalNode(int max_size,
+ int penalty) {
+ CreateChildren(max_size);
+ PartitionTreeNode* left = children_[kLeftChild];
+ PartitionTreeNode* right = children_[kRightChild];
+ if ((left == NULL) && (right == NULL)) {
+ // This is a solution node; return it.
+ return this;
+ } else if (left == NULL) {
+ // One child empty, return the other.
+ return right->GetOptimalNode(max_size, penalty);
+ } else if (right == NULL) {
+ // One child empty, return the other.
+ return left->GetOptimalNode(max_size, penalty);
+ } else {
+ PartitionTreeNode* first;
+ PartitionTreeNode* second;
+ if (left->Cost(penalty) <= right->Cost(penalty)) {
+ first = left;
+ second = right;
+ } else {
+ first = right;
+ second = left;
+ }
+ first = first->GetOptimalNode(max_size, penalty);
+ if (second->Cost(penalty) <= first->Cost(penalty)) {
+ second = second->GetOptimalNode(max_size, penalty);
+ // Compare cost estimate for "second" with actual cost for "first".
+ if (second->Cost(penalty) < first->Cost(penalty)) {
+ return second;
+ }
+ }
+ return first;
+ }
+}
+
+Vp8PartitionAggregator::Vp8PartitionAggregator(
+ const RTPFragmentationHeader& fragmentation,
+ int first_partition_idx, int last_partition_idx)
+ : root_(NULL),
+ num_partitions_(last_partition_idx - first_partition_idx + 1),
+ size_vector_(new int[num_partitions_]),
+ largest_partition_size_(0) {
+ assert(first_partition_idx >= 0);
+ assert(last_partition_idx >= first_partition_idx);
+ assert(last_partition_idx < fragmentation.fragmentationVectorSize);
+ for (size_t i = 0; i < num_partitions_; ++i) {
+ size_vector_[i] =
+ fragmentation.fragmentationLength[i + first_partition_idx];
+ largest_partition_size_ = std::max(largest_partition_size_,
+ size_vector_[i]);
+ }
+ root_ = PartitionTreeNode::CreateRootNode(size_vector_, num_partitions_);
+}
+
+Vp8PartitionAggregator::~Vp8PartitionAggregator() {
+ delete [] size_vector_;
+ delete root_;
+}
+
+void Vp8PartitionAggregator::SetPriorMinMax(int min_size, int max_size) {
+ assert(root_);
+ assert(min_size >= 0);
+ assert(max_size >= min_size);
+ root_->set_min_parent_size(min_size);
+ root_->set_max_parent_size(max_size);
+}
+
+Vp8PartitionAggregator::ConfigVec
+Vp8PartitionAggregator::FindOptimalConfiguration(int max_size, int penalty) {
+ assert(root_);
+ assert(max_size >= largest_partition_size_);
+ PartitionTreeNode* opt = root_->GetOptimalNode(max_size, penalty);
+ ConfigVec config_vector(num_partitions_, 0);
+ PartitionTreeNode* temp_node = opt;
+ int packet_index = opt->NumPackets() - 1;
+ for (int i = num_partitions_ - 1; i >= 0; --i) {
+ assert(packet_index >= 0);
+ assert(temp_node != NULL);
+ config_vector[i] = packet_index;
+ if (temp_node->packet_start()) --packet_index;
+ temp_node = temp_node->parent();
+ }
+ return config_vector;
+}
+
+void Vp8PartitionAggregator::CalcMinMax(const ConfigVec& config,
+ int* min_size, int* max_size) const {
+ if (*min_size < 0) {
+ *min_size = std::numeric_limits<int>::max();
+ }
+ if (*max_size < 0) {
+ *max_size = 0;
+ }
+ unsigned int i = 0;
+ while (i < config.size()) {
+ int this_size = 0;
+ unsigned int j = i;
+ while (j < config.size() && config[i] == config[j]) {
+ this_size += size_vector_[j];
+ ++j;
+ }
+ i = j;
+ if (this_size < *min_size) {
+ *min_size = this_size;
+ }
+ if (this_size > *max_size) {
+ *max_size = this_size;
+ }
+ }
+}
+
+int Vp8PartitionAggregator::CalcNumberOfFragments(int large_partition_size,
+ int max_payload_size,
+ int penalty,
+ int min_size,
+ int max_size) {
+ assert(max_size <= max_payload_size);
+ assert(min_size <= max_size);
+ assert(max_payload_size > 0);
+ // Divisions with rounding up.
+ const int min_number_of_fragments =
+ (large_partition_size + max_payload_size - 1) / max_payload_size;
+ if (min_size < 0 || max_size < 0) {
+ // No aggregates produced, so we do not have any size boundaries.
+ // Simply split in as few partitions as possible.
+ return min_number_of_fragments;
+ }
+ const int max_number_of_fragments =
+ (large_partition_size + min_size - 1) / min_size;
+ int num_fragments = -1;
+ int best_cost = std::numeric_limits<int>::max();
+ for (int n = min_number_of_fragments; n <= max_number_of_fragments; ++n) {
+ // Round up so that we use the largest fragment.
+ int fragment_size = (large_partition_size + n - 1) / n;
+ int cost = 0;
+ if (fragment_size < min_size) {
+ cost = min_size - fragment_size + n * penalty;
+ } else if (fragment_size > max_size) {
+ cost = fragment_size - max_size + n * penalty;
+ } else {
+ cost = n * penalty;
+ }
+ if (fragment_size <= max_payload_size && cost < best_cost) {
+ num_fragments = n;
+ best_cost = cost;
+ }
+ }
+ assert(num_fragments > 0);
+ // TODO(mflodman) Assert disabled since it's falsely triggered, see issue 293.
+ //assert(large_partition_size / num_fragments + 1 <= max_payload_size);
+ return num_fragments;
+}
+
+} // namespace
+
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h
new file mode 100644
index 0000000..c5d47de
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h
@@ -0,0 +1,135 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
+
+#include <vector>
+
+#include "modules/interface/module_common_types.h"
+#include "system_wrappers/interface/constructor_magic.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+// Class used to solve the VP8 aggregation problem.
+class PartitionTreeNode {
+ public:
+ // Create a tree node.
+ PartitionTreeNode(PartitionTreeNode* parent,
+ const int* size_vector,
+ int num_partitions,
+ int this_size);
+
+ // Create a root node.
+ static PartitionTreeNode* CreateRootNode(const int* size_vector,
+ int num_partitions);
+
+ ~PartitionTreeNode();
+
+ // Calculate the cost for the node. If the node is a solution node, the cost
+ // will be the actual cost associated with that solution. If not, the cost
+ // will be the cost accumulated so far along the current branch (which is a
+ // lower bound for any solution along the branch).
+ int Cost(int penalty);
+
+ // Create the two children for this node.
+ bool CreateChildren(int max_size);
+
+ // Get the number of packets for the configuration that this node represents.
+ int NumPackets();
+
+ // Find the optimal solution given a maximum packet size and a per-packet
+ // penalty. The method will be recursively called while the solver is
+ // probing down the tree of nodes.
+ PartitionTreeNode* GetOptimalNode(int max_size, int penalty);
+
+ // Setters and getters.
+ void set_max_parent_size(int size) { max_parent_size_ = size; }
+ void set_min_parent_size(int size) { min_parent_size_ = size; }
+ PartitionTreeNode* parent() const { return parent_; }
+ PartitionTreeNode* left_child() const { return children_[kLeftChild]; }
+ PartitionTreeNode* right_child() const { return children_[kRightChild]; }
+ int this_size() const { return this_size_; }
+ bool packet_start() const { return packet_start_; }
+
+ private:
+ enum Children {
+ kLeftChild = 0,
+ kRightChild = 1
+ };
+
+ void set_packet_start(bool value) { packet_start_ = value; }
+
+ PartitionTreeNode* parent_;
+ PartitionTreeNode* children_[2];
+ int this_size_;
+ const int* size_vector_;
+ int num_partitions_;
+ int max_parent_size_;
+ int min_parent_size_;
+ bool packet_start_;
+
+ DISALLOW_COPY_AND_ASSIGN(PartitionTreeNode);
+};
+
+// Class that calculates the optimal aggregation of VP8 partitions smaller than
+// the maximum packet size.
+class Vp8PartitionAggregator {
+ public:
+ typedef std::vector<int> ConfigVec;
+
+ // Constructor. All partitions in the fragmentation header from index
+ // first_partition_idx to last_partition_idx must be smaller than
+ // maximum packet size to be used in FindOptimalConfiguration.
+ Vp8PartitionAggregator(const RTPFragmentationHeader& fragmentation,
+ int first_partition_idx, int last_partition_idx);
+
+ ~Vp8PartitionAggregator();
+
+ // Set the smallest and largest payload sizes produces so far.
+ void SetPriorMinMax(int min_size, int max_size);
+
+ // Find the aggregation of VP8 partitions that produces the smallest cost.
+ // The result is given as a vector of the same length as the number of
+ // partitions given to the constructor (i.e., last_partition_idx -
+ // first_partition_idx + 1), where each element indicates the packet index
+ // for that partition. Thus, the output vector starts at 0 and is increasing
+ // up to the number of packets - 1.
+ ConfigVec FindOptimalConfiguration(int max_size, int penalty);
+
+ // Calculate minimum and maximum packet sizes for a given aggregation config.
+ // The extreme packet sizes of the given aggregation are compared with the
+ // values given in min_size and max_size, and if either of these are exceeded,
+ // the new extreme value will be written to the corresponding variable.
+ void CalcMinMax(const ConfigVec& config, int* min_size, int* max_size) const;
+
+ // Calculate the number of fragments to divide a large partition into.
+ // The large partition is of size large_partition_size. The payload must not
+ // be larger than max_payload_size. Each fragment comes at an overhead cost
+ // of penalty bytes. If the size of the fragments fall outside the range
+ // [min_size, max_size], an extra cost is inflicted.
+ static int CalcNumberOfFragments(int large_partition_size,
+ int max_payload_size,
+ int penalty,
+ int min_size,
+ int max_size);
+
+ private:
+ PartitionTreeNode* root_;
+ size_t num_partitions_;
+ int* size_vector_;
+ int largest_partition_size_;
+
+ DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
+};
+} // namespace
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
new file mode 100644
index 0000000..3c274d1
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h> // NULL
+
+#include "gtest/gtest.h"
+#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+
+namespace webrtc {
+
+TEST(PartitionTreeNode, CreateAndDelete) {
+ const int kVector[] = {1, 2, 3};
+ const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
+ PartitionTreeNode* node1 =
+ PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
+ PartitionTreeNode* node2 =
+ new PartitionTreeNode(node1, kVector, kNumPartitions, 17);
+ delete node1;
+ delete node2;
+}
+
+TEST(PartitionTreeNode, CreateChildrenAndDelete) {
+ const int kVector[] = {1, 2, 3};
+ const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
+ const int kMaxSize = 10;
+ const int kPenalty = 5;
+ PartitionTreeNode* root =
+ PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
+ EXPECT_TRUE(root->CreateChildren(kMaxSize));
+ ASSERT_TRUE(NULL != root->left_child());
+ ASSERT_TRUE(NULL != root->right_child());
+ EXPECT_EQ(3, root->left_child()->this_size());
+ EXPECT_EQ(2, root->right_child()->this_size());
+ EXPECT_EQ(11, root->right_child()->Cost(kPenalty));
+ EXPECT_FALSE(root->left_child()->packet_start());
+ EXPECT_TRUE(root->right_child()->packet_start());
+ delete root;
+}
+
+TEST(PartitionTreeNode, FindOptimalConfig) {
+ const int kVector[] = {197, 194, 213, 215, 184, 199, 197, 207};
+ const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
+ const int kMaxSize = 1500;
+ const int kPenalty = 1;
+ PartitionTreeNode* root =
+ PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
+ root->set_max_parent_size(500);
+ root->set_min_parent_size(300);
+ PartitionTreeNode* opt = root->GetOptimalNode(kMaxSize, kPenalty);
+ ASSERT_TRUE(opt != NULL);
+ EXPECT_EQ(4, opt->NumPackets());
+ // Expect optimal sequence to be {1, 0, 1, 0, 1, 0, 1, 0}, which corresponds
+ // to (right)-left-right-left-right-left-right-left, where the root node is
+ // implicitly a "right" node by definition.
+ EXPECT_TRUE(opt->parent()->parent()->parent()->parent()->parent()->
+ parent()->parent()->packet_start());
+ EXPECT_FALSE(opt->parent()->parent()->parent()->parent()->parent()->
+ parent()->packet_start());
+ EXPECT_TRUE(opt->parent()->parent()->parent()->parent()->parent()->
+ packet_start());
+ EXPECT_FALSE(opt->parent()->parent()->parent()->parent()->packet_start());
+ EXPECT_TRUE(opt->parent()->parent()->parent()->packet_start());
+ EXPECT_FALSE(opt->parent()->parent()->packet_start());
+ EXPECT_TRUE(opt->parent()->packet_start());
+ EXPECT_FALSE(opt->packet_start());
+ EXPECT_TRUE(opt == root->left_child()->right_child()->left_child()->
+ right_child()->left_child()->right_child()->left_child());
+ delete root;
+}
+
+TEST(PartitionTreeNode, FindOptimalConfigSinglePartition) {
+ const int kVector[] = {17};
+ const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
+ const int kMaxSize = 1500;
+ const int kPenalty = 1;
+ PartitionTreeNode* root =
+ PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
+ PartitionTreeNode* opt = root->GetOptimalNode(kMaxSize, kPenalty);
+ ASSERT_TRUE(opt != NULL);
+ EXPECT_EQ(1, opt->NumPackets());
+ EXPECT_TRUE(opt == root);
+ delete root;
+}
+
+static void VerifyConfiguration(const int* expected_config,
+ size_t expected_config_len,
+ const std::vector<int>& opt_config,
+ const RTPFragmentationHeader& fragmentation) {
+ ASSERT_EQ(expected_config_len, fragmentation.fragmentationVectorSize);
+ EXPECT_EQ(expected_config_len, opt_config.size());
+ for (size_t i = 0; i < expected_config_len; ++i) {
+ EXPECT_EQ(expected_config[i], opt_config[i]);
+ }
+}
+
+static void VerifyMinMax(const Vp8PartitionAggregator& aggregator,
+ const std::vector<int>& opt_config,
+ int expected_min,
+ int expected_max) {
+ int min_size = -1;
+ int max_size = -1;
+ aggregator.CalcMinMax(opt_config, &min_size, &max_size);
+ EXPECT_EQ(expected_min, min_size);
+ EXPECT_EQ(expected_max, max_size);
+}
+
+TEST(Vp8PartitionAggregator, CreateAndDelete) {
+ RTPFragmentationHeader fragmentation;
+ fragmentation.VerifyAndAllocateFragmentationHeader(3);
+ Vp8PartitionAggregator* aggregator =
+ new Vp8PartitionAggregator(fragmentation, 0, 2);
+ delete aggregator;
+}
+
+TEST(Vp8PartitionAggregator, FindOptimalConfig) {
+ RTPFragmentationHeader fragmentation;
+ fragmentation.VerifyAndAllocateFragmentationHeader(8);
+ fragmentation.fragmentationLength[0] = 197;
+ fragmentation.fragmentationLength[1] = 194;
+ fragmentation.fragmentationLength[2] = 213;
+ fragmentation.fragmentationLength[3] = 215;
+ fragmentation.fragmentationLength[4] = 184;
+ fragmentation.fragmentationLength[5] = 199;
+ fragmentation.fragmentationLength[6] = 197;
+ fragmentation.fragmentationLength[7] = 207;
+ Vp8PartitionAggregator* aggregator =
+ new Vp8PartitionAggregator(fragmentation, 0, 7);
+ aggregator->SetPriorMinMax(300, 500);
+ int kMaxSize = 1500;
+ int kPenalty = 1;
+ std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
+ kPenalty);
+ const int kExpectedConfig[] = {0, 0, 1, 1, 2, 2, 3, 3};
+ const size_t kExpectedConfigSize =
+ sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
+ VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
+ fragmentation);
+ VerifyMinMax(*aggregator, opt_config, 383, 428);
+ // Change min and max and run method again. This time, we expect it to leave
+ // the values unchanged.
+ int min_size = 382;
+ int max_size = 429;
+ aggregator->CalcMinMax(opt_config, &min_size, &max_size);
+ EXPECT_EQ(382, min_size);
+ EXPECT_EQ(429, max_size);
+ delete aggregator;
+}
+
+TEST(Vp8PartitionAggregator, FindOptimalConfigEqualFragments) {
+ RTPFragmentationHeader fragmentation;
+ fragmentation.VerifyAndAllocateFragmentationHeader(8);
+ fragmentation.fragmentationLength[0] = 200;
+ fragmentation.fragmentationLength[1] = 200;
+ fragmentation.fragmentationLength[2] = 200;
+ fragmentation.fragmentationLength[3] = 200;
+ fragmentation.fragmentationLength[4] = 200;
+ fragmentation.fragmentationLength[5] = 200;
+ fragmentation.fragmentationLength[6] = 200;
+ fragmentation.fragmentationLength[7] = 200;
+ Vp8PartitionAggregator* aggregator =
+ new Vp8PartitionAggregator(fragmentation, 0, 7);
+ int kMaxSize = 1500;
+ int kPenalty = 1;
+ std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
+ kPenalty);
+ const int kExpectedConfig[] = {0, 0, 0, 0, 1, 1, 1, 1};
+ const size_t kExpectedConfigSize =
+ sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
+ VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
+ fragmentation);
+ VerifyMinMax(*aggregator, opt_config, 800, 800);
+ delete aggregator;
+}
+
+TEST(Vp8PartitionAggregator, FindOptimalConfigSinglePartition) {
+ RTPFragmentationHeader fragmentation;
+ fragmentation.VerifyAndAllocateFragmentationHeader(1);
+ fragmentation.fragmentationLength[0] = 17;
+ Vp8PartitionAggregator* aggregator =
+ new Vp8PartitionAggregator(fragmentation, 0, 0);
+ int kMaxSize = 1500;
+ int kPenalty = 1;
+ std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
+ kPenalty);
+ const int kExpectedConfig[] = {0};
+ const size_t kExpectedConfigSize =
+ sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
+ VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
+ fragmentation);
+ VerifyMinMax(*aggregator, opt_config, 17, 17);
+ delete aggregator;
+}
+
+TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
+ const int kMTU = 1500;
+ EXPECT_EQ(2,
+ Vp8PartitionAggregator::CalcNumberOfFragments(
+ 1600, kMTU, 1, 300, 900));
+ EXPECT_EQ(3,
+ Vp8PartitionAggregator::CalcNumberOfFragments(
+ 1600, kMTU, 1, 300, 798));
+ EXPECT_EQ(2,
+ Vp8PartitionAggregator::CalcNumberOfFragments(
+ 1600, kMTU, 1, 900, 1000));
+}
+
+} // namespace
+
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc
new file mode 100644
index 0000000..1a55e4e
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "BWEConvergenceTest.h"
+
+#include <fstream>
+#include <string>
+
+#include "TestSenderReceiver.h"
+#include "TestLoadGenerator.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+
+
+BWEConvergenceTestUp::BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps)
+:
+_availBWkbps(availBWkbps),
+BWEOneWayTest(testName, startRateKbps)
+{
+}
+
+
+BWEConvergenceTestUp::~BWEConvergenceTestUp()
+{
+ if (_gen)
+ {
+ delete _gen;
+ _gen = NULL;
+ }
+}
+
+
+int BWEConvergenceTestUp::Init(std::string ip, WebRtc_UWord16 port)
+{
+ // create the load generator object
+ const int rtpSampleRate = 90000;
+ const int frameRate = 30;
+ const double spreadFactor = 0.2;
+
+ if (_master)
+ {
+ _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
+ if (!_gen)
+ {
+ return (-1);
+ }
+ }
+
+ return BWEOneWayTest::Init(ip, port);
+}
+
+
+bool BWEConvergenceTestUp::StoppingCriterionMaster()
+{
+ return ((_sendrec->BitrateSent() / 1000.0) > (0.9 * _availBWkbps));
+}
+
+
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h
new file mode 100644
index 0000000..b830d14
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
+
+#include <string>
+
+#include "BWETestBase.h"
+
+#include "typedefs.h"
+
+#include "TestSenderReceiver.h"
+
+class BWEConvergenceTestUp : public BWEOneWayTest
+{
+public:
+ BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps);
+ virtual ~BWEConvergenceTestUp();
+
+ virtual int Init(std::string ip, WebRtc_UWord16 port);
+
+protected:
+ virtual bool StoppingCriterionMaster();
+
+private:
+ int _availBWkbps;
+};
+
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc
new file mode 100644
index 0000000..1fd19fe
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <fstream>
+#include <math.h>
+
+#include "BWEStabilityTest.h"
+#include "TestLoadGenerator.h"
+#include "tick_util.h"
+#include "critical_section_wrapper.h"
+
+
+BWEStabilityTest::BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds)
+:
+_testDurationSeconds(testDurationSeconds),
+BWEOneWayTest(testName, rateKbps)
+{
+}
+
+
+BWEStabilityTest::~BWEStabilityTest()
+{
+ if (_gen)
+ {
+ delete _gen;
+ _gen = NULL;
+ }
+}
+
+
+int BWEStabilityTest::Init(std::string ip, WebRtc_UWord16 port)
+{
+ // create the load generator object
+ const int rtpSampleRate = 90000;
+ const int frameRate = 30;
+ const double spreadFactor = 0.2;
+ const double keyToDeltaRatio = 7;
+ const int keyFramePeriod = 300;
+
+ if (_master)
+ {
+ _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
+ //_gen = new PeriodicKeyFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate,
+ // spreadFactor, keyToDeltaRatio, keyFramePeriod);
+ if (!_gen)
+ {
+ return (-1);
+ }
+
+ }
+
+ return BWEOneWayTest::Init(ip, port);
+}
+
+
+void BWEStabilityTest::Report(std::fstream &log)
+{
+ // cannot report on a running test
+ if(_running) return;
+
+ BWETest::Report(log);
+
+ CriticalSectionScoped cs(_statCritSect);
+
+ log << "Bitrate statistics\n";
+ log << "\tAverage = " << _rateVecKbps.Mean() << " kbps\n";
+ log << "\tMin = " << _rateVecKbps.Min() << " kbps\n";
+ log << "\tMax = " << _rateVecKbps.Max() << " kbps\n";
+ log << "\tStd = " << _rateVecKbps.Std() << " kbps\n";
+
+}
+
+
+bool BWEStabilityTest::StoppingCriterionMaster()
+{
+ return (TickTime::MillisecondTimestamp() - _startTimeMs >= _testDurationSeconds * 1000);
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h
new file mode 100644
index 0000000..8f213b1
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
+
+#include <string>
+
+#include "BWETestBase.h"
+
+#include "typedefs.h"
+
+#include "TestSenderReceiver.h"
+
+class BWEStabilityTest : public BWEOneWayTest
+{
+public:
+ BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds);
+ virtual ~BWEStabilityTest();
+
+ virtual int Init(std::string ip, WebRtc_UWord16 port);
+ virtual void Report(std::fstream &log);
+
+protected:
+ virtual bool StoppingCriterionMaster();
+
+private:
+ int _testDurationSeconds;
+};
+
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
new file mode 100644
index 0000000..471ea5f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
@@ -0,0 +1,200 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// BWEStandAlone.cpp : Defines the entry point for the console application.
+//
+
+#include <string>
+#include <stdio.h>
+
+#include "event_wrapper.h"
+#include "udp_transport.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+
+#include "TestSenderReceiver.h"
+#include "TestLoadGenerator.h"
+
+#include "MatlabPlot.h"
+
+//#include "vld.h"
+
+class myTransportCB: public UdpTransportData
+{
+public:
+ myTransportCB (RtpRtcp *rtpMod) : _rtpMod(rtpMod) {};
+protected:
+ // Inherited from UdpTransportData
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+private:
+ RtpRtcp *_rtpMod;
+};
+
+void myTransportCB::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ printf("Receiving RTP from IP %s, port %u\n", fromIP, fromPort);
+ _rtpMod->IncomingPacket((WebRtc_UWord8 *) incomingRtpPacket, static_cast<WebRtc_UWord16>(rtpPacketLength));
+}
+
+void myTransportCB::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ printf("Receiving RTCP from IP %s, port %u\n", fromIP, fromPort);
+ _rtpMod->IncomingPacket((WebRtc_UWord8 *) incomingRtcpPacket, static_cast<WebRtc_UWord16>(rtcpPacketLength));
+}
+
+
+int main(int argc, char* argv[])
+{
+ bool isSender = false;
+ bool isReceiver = false;
+ WebRtc_UWord16 port;
+ std::string ip;
+ TestSenderReceiver *sendrec = new TestSenderReceiver();
+ TestLoadGenerator *gen;
+
+ if (argc == 2)
+ {
+ // receiver only
+ isReceiver = true;
+
+ // read port
+ port = atoi(argv[1]);
+ }
+ else if (argc == 3)
+ {
+ // sender and receiver
+ isSender = true;
+ isReceiver = true;
+
+ // read IP
+ ip = argv[1];
+
+ // read port
+ port = atoi(argv[2]);
+ }
+
+ Trace::CreateTrace();
+ Trace::SetTraceFile("BWEStandAloneTrace.txt");
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+ sendrec->InitReceiver(port);
+
+ sendrec->Start();
+
+ if (isSender)
+ {
+ const WebRtc_UWord32 startRateKbps = 1000;
+ //gen = new CBRGenerator(sendrec, 1000, 500);
+ gen = new CBRFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2);
+ //gen = new PeriodicKeyFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2, 7, 300);
+ //const WebRtc_UWord16 numFrameRates = 5;
+ //const WebRtc_UWord8 frameRates[numFrameRates] = {30, 15, 20, 23, 25};
+ //gen = new CBRVarFRGenerator(sendrec, 1000, frameRates, numFrameRates, 90000, 4.0, 0.1, 0.2);
+ //gen = new CBRFrameDropGenerator(sendrec, startRateKbps, 90000, 0.2);
+ sendrec->SetLoadGenerator(gen);
+ sendrec->InitSender(startRateKbps, ip.c_str(), port);
+ gen->Start();
+ }
+
+ while (1)
+ {
+ }
+
+ if (isSender)
+ {
+ gen->Stop();
+ delete gen;
+ }
+
+ delete sendrec;
+
+ //WebRtc_UWord8 numberOfSocketThreads = 1;
+ //UdpTransport* transport = UdpTransport::Create(0, numberOfSocketThreads);
+
+ //RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
+ //if (rtp->InitSender() != 0)
+ //{
+ // exit(1);
+ //}
+ //if (rtp->RegisterSendTransport(transport) != 0)
+ //{
+ // exit(1);
+ //}
+
+// transport->InitializeSendSockets("192.168.200.39", 8000);
+ //transport->InitializeSendSockets("127.0.0.1", 10000);
+ //transport->InitializeSourcePorts(8000);
+
+
+ return(0);
+ // myTransportCB *tp = new myTransportCB(rtp);
+ // transport->InitializeReceiveSockets(tp, 10000, "0.0.0.0");
+ // transport->StartReceiving(500);
+
+ // WebRtc_Word8 data[100];
+ // for (int i = 0; i < 100; data[i] = i++);
+
+ // for (int i = 0; i < 100; i++)
+ // {
+ // transport->SendRaw(data, 100, false);
+ // }
+
+
+
+ // WebRtc_Word32 totTime = 0;
+ // while (totTime < 10000)
+ // {
+ // transport->Process();
+ // WebRtc_Word32 wTime = transport->TimeUntilNextProcess();
+ // totTime += wTime;
+ // Sleep(wTime);
+ // }
+
+
+ //if (transport)
+ //{
+ // // Destroy the Socket Transport module
+ // transport->StopReceiving();
+ // transport->InitializeReceiveSockets(NULL,0);// deregister callback
+ // UdpTransport::Destroy(transport);
+ // transport = NULL;
+ // }
+
+ // if (tp)
+ // {
+ // delete tp;
+ // tp = NULL;
+ // }
+
+ // if (rtp)
+ // {
+ // RtpRtcp::DestroyRtpRtcp(rtp);
+ // rtp = NULL;
+ // }
+
+
+ //return 0;
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc
new file mode 100644
index 0000000..2940abd
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc
@@ -0,0 +1,453 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "BWETestBase.h"
+
+#include <algorithm> // sort
+#include <fstream>
+#include <string>
+#include <vector>
+#include <math.h>
+
+#include "TestSenderReceiver.h"
+#include "TestLoadGenerator.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "critical_section_wrapper.h"
+
+
+double StatVec::Mean()
+{
+ double sum = 0;
+
+ // sanity
+ if (size() <= 0) return (0);
+
+ std::vector<double>::iterator it;
+ for (it = begin(); it < end(); ++it)
+ {
+ sum += (*it);
+ }
+
+ return (sum / size());
+}
+
+double StatVec::Variance()
+{
+ double sumSqaure = 0;
+ double sum = 0;
+
+ std::vector<double>::iterator it;
+ for (it = begin(); it < end(); ++it)
+ {
+ sum += (*it);
+ sumSqaure += (*it) * (*it);
+ }
+
+ // Normalizes by N-1. This produces the best unbiased estimate of the
+ // variance if X is a sample from a normal distribution.
+ int M = static_cast<int> (size() - 1);
+
+ if (M > 0)
+ {
+ double var = (sumSqaure / M) - (sum / (M+1)) * (sum / M);
+ assert(var >= 0);
+ return (var);
+ }
+ else
+ {
+ return (0);
+ }
+}
+
+double StatVec::Std()
+{
+ return (sqrt(Variance()));
+}
+
+double StatVec::Max()
+{
+ // sanity
+ if (size() <= 0) return (0);
+
+ std::vector<double>::iterator it = begin();
+ double maxVal = (*it);
+ ++it;
+
+ for (; it < end(); ++it)
+ {
+ if ((*it) > maxVal) maxVal = (*it);
+ }
+
+ return (maxVal);
+}
+
+double StatVec::Min()
+{
+ // sanity
+ if (size() <= 0) return (0);
+
+ std::vector<double>::iterator it = begin();
+ double minVal = (*it);
+ ++it;
+
+ for (; it < end(); ++it)
+ {
+ if ((*it) < minVal) minVal = (*it);
+ }
+
+ return (minVal);
+}
+
+double StatVec::Median()
+{
+ double median;
+
+ // sanity
+ if (size() <= 0) return (0);
+
+ // sort the vector
+ sort(begin(), end());
+
+ if ((size() % 2) == 0)
+ {
+ // even size; use average of two center elements
+ median = (at(size()/2 - 1) + at(size()/2)) / 2.0;
+ }
+ else
+ {
+ // odd size; take center element
+ median = at(size()/2);
+ }
+
+ return (median);
+}
+
+double StatVec::Percentile(double p)
+{
+ // sanity
+ if (size() <= 0) return (0);
+
+ // sort the vector
+ sort(begin(), end());
+
+ int rank = static_cast<int> (((size() - 1) * p) / 100 + 0.5); // between 1 and size()
+ rank -= 1; // between 0 and size()-1
+
+ assert(rank >= 0);
+ assert(rank < static_cast<int>(size()));
+
+ return (at(rank));
+}
+
+void StatVec::Export(std::fstream &file, bool colVec /*= false*/)
+{
+ // sanity
+ if (size() <= 0) return;
+
+ std::string separator;
+ if (colVec) separator = "\n";
+ else separator = ", ";
+
+ std::vector<double>::iterator it = begin();
+ file << (*it);
+ ++it;
+
+ for (; it < end(); ++it)
+ {
+ file << separator << (*it);
+ }
+
+ file << std::endl;
+}
+
+
+bool BWETestProcThreadFunction(void *obj)
+{
+ if (obj == NULL)
+ {
+ return false;
+ }
+ BWETest *theObj = static_cast<BWETest *>(obj);
+
+ theObj->ProcLoop();
+
+ theObj->Stop();
+
+ return(true);
+}
+
+
+BWETest::BWETest(std::string testName, int startRateKbps):
+_testName(testName),
+_startRateKbps(startRateKbps),
+_master(false),
+_sendrec(NULL),
+_gen(NULL),
+_initialized(false),
+_started(false),
+_running(false),
+_eventPtr(NULL),
+_procThread(NULL),
+_startTimeMs(-1),
+_stopTimeMs(-1),
+_statCritSect(CriticalSectionWrapper::CreateCriticalSection())
+{
+ _sendrec = new TestSenderReceiver();
+}
+
+
+BWETest::~BWETest()
+{
+ if (_running)
+ {
+ Stop();
+ }
+
+ _statCritSect->Enter();
+ delete &_statCritSect;
+
+ if (_sendrec)
+ {
+ delete _sendrec;
+ _sendrec = NULL;
+ }
+}
+
+
+bool BWETest::SetMaster(bool isMaster /*= true*/)
+{
+ if (!_initialized)
+ {
+ // Can only set status before initializing.
+ _master = isMaster;
+ }
+
+ return (_master);
+}
+
+
+int BWETest::Init(std::string ip, WebRtc_UWord16 port)
+{
+ if (_initialized)
+ {
+ // cannot init twice
+ return (-1);
+ }
+
+ if (!_sendrec)
+ {
+ throw "SenderReceiver must be created";
+ exit(1);
+ }
+
+ if (_started)
+ {
+ // cannot init after start
+ return (-1);
+ }
+
+ // initialize receiver port (for feedback)
+ _sendrec->InitReceiver(port);
+
+ // initialize sender
+ _sendrec->SetLoadGenerator(_gen);
+ _sendrec->InitSender(_startRateKbps, ip.c_str(), port);
+ //_gen->Start();
+
+ _sendrec->SetCallback(this);
+
+ _initialized = true;
+
+ return 0;
+}
+
+
+bool BWETest::Start()
+{
+ if (!_initialized)
+ {
+ // must init first
+ return (false);
+ }
+ if (_started)
+ {
+ // already started, do nothing
+ return (true);
+ }
+
+ if (_sendrec->Start() != 0)
+ {
+ // failed
+ return (false);
+ }
+
+ if (_gen)
+ {
+ if (_gen->Start() != 0)
+ {
+ // failed
+ return (false);
+ }
+ }
+
+ _eventPtr = EventWrapper::Create();
+
+ _startTimeMs = TickTime::MillisecondTimestamp();
+ _started = true;
+ _running = true;
+
+ return (true);
+}
+
+
+bool BWETest::Stop()
+{
+ if (_procThread)
+ {
+ _stopTimeMs = TickTime::MillisecondTimestamp();
+ _procThread->SetNotAlive();
+ _running = false;
+ _eventPtr->Set();
+
+ while (!_procThread->Stop())
+ {
+ ;
+ }
+
+ delete _procThread;
+ _procThread = NULL;
+
+ }
+
+ if (_eventPtr)
+ {
+ delete _eventPtr;
+ _eventPtr = NULL;
+ }
+
+ _procThread = NULL;
+
+ if(_gen)
+ {
+ _gen->Stop();
+ }
+
+ return(true);
+}
+
+
+bool BWETest::ProcLoop(void)
+{
+ bool receiving = false;
+
+ // no critSect
+ while (_running)
+ {
+
+ // check stopping criterions
+ if (_master && StoppingCriterionMaster())
+ {
+ printf("StoppingCriterionMaster()\n");
+ _stopTimeMs = TickTime::MillisecondTimestamp();
+ _running = false;
+ }
+ else if (!_master && StoppingCriterionSlave())
+ {
+ printf("StoppingCriterionSlave()\n");
+ _running = false;
+ }
+
+ // wait
+ _eventPtr->Wait(1000); // 1000 ms
+
+ }
+
+ return true;
+}
+
+
+void BWETest::Report(std::fstream &log)
+{
+ // cannot report on a running test
+ if(_running) return;
+
+ CriticalSectionScoped cs(_statCritSect);
+
+ log << "\n\n*** Test name = " << _testName << "\n";
+ log << "Execution time = " << static_cast<double>(_stopTimeMs - _startTimeMs) / 1000 << " s\n";
+ log << "\n";
+ log << "RTT statistics\n";
+ log << "\tMin = " << _rttVecMs.Min() << " ms\n";
+ log << "\tMax = " << _rttVecMs.Max() << " ms\n";
+ log << "\n";
+ log << "Loss statistics\n";
+ log << "\tAverage = " << _lossVec.Mean() << "%\n";
+ log << "\tMax = " << _lossVec.Max() << "%\n";
+
+ log << "\n" << "Rates" << "\n";
+ _rateVecKbps.Export(log);
+
+ log << "\n" << "RTT" << "\n";
+ _rttVecMs.Export(log);
+
+}
+
+
+// SenderReceiver callback
+void BWETest::OnOnNetworkChanged(const WebRtc_UWord32 bitrateTargetBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs,
+ const WebRtc_UWord32 jitterMS,
+ const WebRtc_UWord16 bwEstimateKbitMin,
+ const WebRtc_UWord16 bwEstimateKbitMax)
+{
+ CriticalSectionScoped cs(_statCritSect);
+
+ // bitrate statistics
+ WebRtc_Word32 newBitrateKbps = bitrateTargetBps/1000;
+
+ _rateVecKbps.push_back(newBitrateKbps);
+ _rttVecMs.push_back(roundTripTimeMs);
+ _lossVec.push_back(static_cast<double>(fractionLost) / 255.0);
+}
+
+
+int BWEOneWayTest::Init(std::string ip, WebRtc_UWord16 port)
+{
+
+ if (!_master)
+ {
+ // Use timeout stopping criterion by default for receiver
+ UseRecvTimeout();
+ }
+
+ return (BWETest::Init(ip, port));
+
+}
+
+
+bool BWEOneWayTest::Start()
+{
+ bool ret = BWETest::Start();
+
+ if (!_master)
+ {
+ // send one dummy RTP packet to enable RTT measurements
+ const WebRtc_UWord8 dummy = 0;
+ //_gen->sendPayload(TickTime::MillisecondTimestamp(), &dummy, 0);
+ _sendrec->SendOutgoingData(
+ static_cast<WebRtc_UWord32>(TickTime::MillisecondTimestamp()*90),
+ &dummy, 1, webrtc::kVideoFrameDelta);
+ }
+
+ return ret;
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
new file mode 100644
index 0000000..bab1b94
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
+
+#include <string>
+#include <vector>
+
+#include "typedefs.h"
+
+#include "TestSenderReceiver.h"
+
+
+class StatVec : public std::vector<double>
+{
+public:
+ double Mean();
+ double Variance();
+ double Std();
+ double Max();
+ double Min();
+ double Median();
+ double Percentile(double p); // 0 <= p <= 100%
+ void Export(std::fstream &file, bool colVec = false);
+};
+
+
+class BWETest : public SendRecCB
+{
+public:
+ BWETest(std::string testName, int startRateKbps);
+ virtual ~BWETest();
+
+ bool SetMaster(bool isMaster = true);
+ void UseRecvTimeout() { _sendrec->SetPacketTimeout(1000); };
+ virtual int Init(std::string ip, WebRtc_UWord16 port);
+ virtual bool Start();
+ virtual bool Stop();
+ bool ProcLoop(void);
+ virtual void Report(std::fstream &log);
+ std::string TestName() { return (_testName); };
+
+ // SenderReceiver callback
+ virtual void OnOnNetworkChanged(const WebRtc_UWord32 bitrateTargetBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs,
+ const WebRtc_UWord32 jitterMS,
+ const WebRtc_UWord16 bwEstimateKbitMin,
+ const WebRtc_UWord16 bwEstimateKbitMax);
+
+
+protected:
+ virtual bool StoppingCriterionMaster() = 0;
+ virtual bool StoppingCriterionSlave() { return (_sendrec->timeOutTriggered()); };
+
+ TestSenderReceiver * _sendrec;
+ TestLoadGenerator * _gen;
+
+ std::string _testName;
+ int _startRateKbps;
+ bool _master;
+ bool _initialized;
+ bool _started;
+ bool _running;
+ EventWrapper *_eventPtr;
+ ThreadWrapper* _procThread;
+ WebRtc_Word64 _startTimeMs;
+ WebRtc_Word64 _stopTimeMs;
+
+ // Statistics, protected by separate CritSect
+ CriticalSectionWrapper* _statCritSect;
+ StatVec _rateVecKbps;
+ StatVec _rttVecMs;
+ StatVec _lossVec;
+};
+
+
+class BWEOneWayTest : public BWETest
+{
+public:
+ BWEOneWayTest(std::string testName, int startRateKbps) :
+ BWETest(testName, startRateKbps) {};
+
+ virtual int Init(std::string ip, WebRtc_UWord16 port);
+ virtual bool Start();
+
+protected:
+ virtual bool StoppingCriterionSlave() {return ( _sendrec->timeOutTriggered()); };
+
+private:
+
+};
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc
new file mode 100644
index 0000000..f1d79fe
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc
@@ -0,0 +1,274 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// BWETester.cpp : Defines the entry point for the console application.
+//
+
+#include <fstream>
+#include <string>
+#include <iostream>
+#include <ctime>
+
+#include "event_wrapper.h"
+#include "trace.h"
+
+#include "BWEStabilityTest.h"
+#include "BWEConvergenceTest.h"
+#include "BWETwoWayLimitFinding.h"
+
+#include "MatlabPlot.h"
+
+//#include "vld.h"
+
+#ifdef MATLAB
+MatlabEngine eng;
+#endif
+
+
+class testContainer
+{
+public:
+ testContainer(BWETest *test, bool waitForKeyStroke, int delayStartSec,
+ std::string instruction) :
+ _test(test),
+ _waitMaster(waitForKeyStroke),
+ _waitSlave(waitForKeyStroke),
+ _delayMaster(delayStartSec),
+ _delaySlave(delayStartSec),
+ _instr(instruction) {};
+
+ testContainer(BWETest *test,
+ bool waitForKeyStrokeMaster,
+ bool waitForKeyStrokeSlave,
+ int delayStartSecMaster,
+ int delayStartSecSlave,
+ std::string instruction) :
+ _test(test),
+ _waitMaster(waitForKeyStrokeMaster),
+ _waitSlave(waitForKeyStrokeSlave),
+ _delayMaster(delayStartSecMaster),
+ _delaySlave(delayStartSecSlave),
+ _instr(instruction) {};
+
+ ~testContainer() { if(_test) delete _test; _test = NULL; };
+
+ BWETest *_test;
+ bool _waitMaster;
+ bool _waitSlave;
+ int _delayMaster;
+ int _delaySlave;
+ std::string _instr;
+};
+
+
+// This is were the test cases are created.
+// Syntax:
+// tests->push_back(new testContainer(
+// new _BWETestConstructor_, // constructor for the test case
+// _wait_, // wait for user key press before start
+// _delay_, // delay test start (after a key press if enabled)
+// "Intruction to user.")); // message to show in console before starting
+//
+// Or:
+// tests->push_back(new testContainer(
+// new _BWETestConstructor_, // constructor for the test case
+// _waitMaster_, // master will wait for user key press before start
+// _waitSlave_, // slave will wait for user key press before start
+// _delayMaster_, // delay master test start (after a key press if enabled)
+// _delaySlave_, // delay slave test start (after a key press if enabled)
+// "Intruction to user.")); // message to show in console before starting
+//
+// Valid test cases are:
+// BWEConvergenceTestUp
+// BWEStabilityTest
+// BWETwoWayLimitFinding
+
+
+void PopulateTests(std::vector<testContainer *>* tests, bool isMaster)
+{
+ tests->push_back(new testContainer(
+ new BWEStabilityTest("Stability", 400, 5*60),
+ true, true,
+ 0, 0,
+ "Set bandwidth limit to 512 kbps"));
+
+ tests->push_back(new testContainer(
+ new BWEStabilityTest("Stability", 4000, 5*60),
+ true, true,
+ 0, 0,
+ "Set bandwidth limit to 5120 kbps"));
+
+ tests->push_back(new testContainer(
+ new BWEStabilityTest("Stability", 400, 5*60),
+ true, true,
+ 0, 0,
+ "Set bandwidth limit to 512 kbps and a normal distributed delay\
+ with mean 100 ms and std dev 15 ms"));
+
+ tests->push_back(new testContainer(
+ new BWEConvergenceTestUp("Convergence 256->512", 256, 512),
+ true,
+ 0,
+ "Set bandwith limit to 512 kbps"));
+
+ tests->push_back(new testContainer(
+ new BWEConvergenceTestUp("Convergence 1024->5120", 1024, 5120),
+ true,
+ 0,
+ "Set bandwith limit to 5120 kbps"));
+
+ tests->push_back(new testContainer(
+ new BWETwoWayLimitFinding("Asymmetric limit finding {1024, 2048} kbps",
+ 500, 1024,
+ 500, 2048,
+ isMaster),
+ true,
+ 0,
+ "Set bandwith limit to {1024, 2048} kbps asymmetric"));
+
+ tests->push_back(new testContainer(
+ new BWETwoWayLimitFinding("Symmetric limit finding {1024, 1024} kbps",
+ 500, 1024,
+ 500, 1024,
+ isMaster),
+ true,
+ 0,
+ "Set bandwith limit to 1024 kbps symmetric"));
+}
+
+
+int main(int argc, char* argv[])
+{
+
+ bool isMaster = false;
+ WebRtc_UWord16 port;
+ std::string ip;
+ std::fstream log;
+ log.open("TestLog.txt", std::fstream::out | std::fstream::app);
+
+ log << "\n\nBWE TESTER\n";
+
+ time_t t = time(0); // get time now
+ struct tm * now = localtime( & t );
+ log << (now->tm_year + 1900) << '-'
+ << (now->tm_mon + 1) << '-'
+ << now->tm_mday << " "
+ << now->tm_hour << ":" << now->tm_min
+ << "\n";
+
+ if (argc == 4)
+ {
+ // read IP
+ ip = argv[1];
+
+ // read port
+ port = atoi(argv[2]);
+
+ // read master/slave
+ isMaster = (atoi(argv[3]) != 0);
+
+ std::cout << "Destination: " << ip << "\n";
+ log << "Destination: " << ip << "\n";
+ std::cout << "Port: " << port << "\n";
+ log << "Port: " << port << "\n";
+ if (isMaster)
+ {
+ std::cout << "Master\n";
+ log << "Master\n";
+ }
+ else
+ {
+ std::cout << "Slave\n";
+ log << "Slave\n";
+ }
+
+ }
+ else
+ {
+ printf("Usage\nBWETester dstIP port master\n");
+ exit(1);
+ }
+
+ std::vector<testContainer*> tests;
+ PopulateTests(&tests, isMaster);
+
+ int testIndex = 0;
+ EventWrapper* event = EventWrapper::Create();
+ std::vector<testContainer*>::iterator it;
+ for (it=tests.begin() ; it < tests.end(); it++)
+ {
+ ++testIndex;
+
+ BWETest *theTest = (*it)->_test;
+
+ if (theTest)
+ {
+ std::cout << "\nTest " << testIndex << ": " << theTest->TestName() << "\n";
+ }
+
+ // Print instructions
+ std::cout << "--> " << (*it)->_instr << std::endl;
+
+ if ((isMaster && (*it)->_waitMaster)
+ || (!isMaster && (*it)->_waitSlave))
+ {
+ // Wait for a key press
+ std::cout << "Press enter to start test\n";
+ getc(stdin);
+ }
+
+ if (isMaster)
+ {
+ if ((*it)->_delayMaster > 0)
+ {
+ // Wait
+ std::cout << "Test starting in "
+ << (*it)->_delayMaster
+ << " seconds" << std::endl;
+ event->Wait((*it)->_delayMaster * 1000);
+ }
+ }
+ else
+ {
+ if ((*it)->_delaySlave > 0)
+ {
+ // Wait
+ std::cout << "Test starting in "
+ << (*it)->_delaySlave
+ << " seconds" << std::endl;
+ event->Wait((*it)->_delaySlave * 1000);
+ }
+ }
+
+ // Start execution
+ if (theTest)
+ {
+ theTest->SetMaster(isMaster);
+ if (theTest->Init(ip, port) != 0)
+ {
+ throw "Error initializing sender";
+ exit (1);
+ }
+
+ theTest->Start();
+ theTest->ProcLoop();
+ theTest->Stop();
+ theTest->Report(log);
+ log << std::flush;
+ }
+
+ delete (*it); // deletes the test too
+ }
+ delete event;
+ event = NULL;
+
+ log.close();
+ return (0);
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc
new file mode 100644
index 0000000..043c7b0
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "BWETwoWayLimitFinding.h"
+#include "TestLoadGenerator.h"
+
+
+BWETwoWayLimitFinding::BWETwoWayLimitFinding(
+ std::string testName,
+ int masterStartRateKbps, int masterAvailBWkbps,
+ int slaveStartRateKbps, int slaveAvailBWkbps,
+ bool isMaster /*= false*/)
+ :
+BWETest(testName, (isMaster ? masterStartRateKbps : slaveStartRateKbps)),
+_availBWkbps(isMaster ? masterAvailBWkbps : slaveAvailBWkbps),
+_incomingAvailBWkbps(isMaster ? slaveAvailBWkbps : masterAvailBWkbps),
+_forwLimitReached(false),
+_revLimitReached(false)
+{
+ _master = isMaster;
+}
+
+
+BWETwoWayLimitFinding::~BWETwoWayLimitFinding()
+{
+ if (_gen)
+ {
+ delete _gen;
+ _gen = NULL;
+ }
+}
+
+
+int BWETwoWayLimitFinding::Init(std::string ip, WebRtc_UWord16 port)
+{
+ // create the load generator object
+ const int rtpSampleRate = 90000;
+ const int frameRate = 30;
+ const double spreadFactor = 0.2;
+
+ _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
+ if (!_gen)
+ {
+ return (-1);
+ }
+
+ if (!_master) UseRecvTimeout(); // slave shuts down when incoming stream dies
+
+ return BWETest::Init(ip, port);
+}
+
+
+bool BWETwoWayLimitFinding::StoppingCriterionMaster()
+{
+ if ((_sendrec->BitrateSent() / 1000.0) > (0.95 * _availBWkbps))
+ {
+ _forwLimitReached = true;
+ }
+
+ WebRtc_Word32 revRateKbps = _sendrec->ReceiveBitrateKbps();
+ if (revRateKbps > (0.95 * _incomingAvailBWkbps))
+ {
+ _revLimitReached = true;
+ }
+
+ return (_forwLimitReached && _revLimitReached);
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h
new file mode 100644
index 0000000..fc790e5
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
+
+#include "BWETestBase.h"
+
+class BWETwoWayLimitFinding : public BWETest
+{
+public:
+ BWETwoWayLimitFinding(std::string testName,
+ int masterStartRateKbps, int masterAvailBWkbps,
+ int slaveStartRateKbps, int slaveAvailBWkbps,
+ bool isMaster = false);
+
+ virtual ~BWETwoWayLimitFinding();
+
+ virtual int Init(std::string ip, WebRtc_UWord16 port);
+
+protected:
+ virtual bool StoppingCriterionMaster();
+ //virtual bool StoppingCriterionSlave();
+
+private:
+ int _availBWkbps;
+ int _incomingAvailBWkbps;
+ bool _forwLimitReached;
+ bool _revLimitReached;
+
+};
+
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
new file mode 100644
index 0000000..9c81fd0
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
@@ -0,0 +1,1071 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "MatlabPlot.h"
+#ifdef MATLAB
+#include "engine.h"
+#endif
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "tick_util.h"
+
+#include <sstream>
+#include <algorithm>
+#include <math.h>
+#include <stdio.h>
+
+using namespace webrtc;
+
+#ifdef MATLAB
+MatlabEngine eng;
+
+MatlabLine::MatlabLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
+:
+_xArray(NULL),
+_yArray(NULL),
+_maxLen(maxLen),
+_plotAttribute(),
+_name()
+{
+ if (_maxLen > 0)
+ {
+ _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
+ _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
+ }
+
+ if (plotAttrib)
+ {
+ _plotAttribute = plotAttrib;
+ }
+
+ if (name)
+ {
+ _name = name;
+ }
+}
+
+MatlabLine::~MatlabLine()
+{
+ if (_xArray != NULL)
+ {
+ mxDestroyArray(_xArray);
+ }
+ if (_yArray != NULL)
+ {
+ mxDestroyArray(_yArray);
+ }
+}
+
+void MatlabLine::Append(double x, double y)
+{
+ if (_maxLen > 0 && _xData.size() > static_cast<WebRtc_UWord32>(_maxLen))
+ {
+ _xData.resize(_maxLen);
+ _yData.resize(_maxLen);
+ }
+
+ _xData.push_front(x);
+ _yData.push_front(y);
+}
+
+
+// append y-data with running integer index as x-data
+void MatlabLine::Append(double y)
+{
+ if (_xData.empty())
+ {
+ // first element is index 0
+ Append(0, y);
+ }
+ else
+ {
+ // take last x-value and increment
+ double temp = _xData.back(); // last x-value
+ Append(temp + 1, y);
+ }
+}
+
+
+void MatlabLine::SetMaxLen(int maxLen)
+{
+ if (maxLen <= 0)
+ {
+ // means no maxLen
+ _maxLen = -1;
+ }
+ else
+ {
+ _maxLen = maxLen;
+
+ if (_xArray != NULL)
+ {
+ mxDestroyArray(_xArray);
+ mxDestroyArray(_yArray);
+ }
+ _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
+ _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
+
+ maxLen = ((unsigned int)maxLen <= _xData.size()) ? maxLen : (int)_xData.size();
+ _xData.resize(maxLen);
+ _yData.resize(maxLen);
+
+ //// reserve the right amount of memory
+ //_xData.reserve(_maxLen);
+ //_yData.reserve(_maxLen);
+ }
+}
+
+void MatlabLine::SetAttribute(char *plotAttrib)
+{
+ _plotAttribute = plotAttrib;
+}
+
+void MatlabLine::SetName(char *name)
+{
+ _name = name;
+}
+
+void MatlabLine::GetPlotData(mxArray** xData, mxArray** yData)
+{
+ // Make sure we have enough Matlab allocated memory.
+ // Assuming both arrays (x and y) are of the same size.
+ if (_xData.empty())
+ {
+ return; // No data
+ }
+ unsigned int size = 0;
+ if (_xArray != NULL)
+ {
+ size = (unsigned int)mxGetNumberOfElements(_xArray);
+ }
+ if (size < _xData.size())
+ {
+ if (_xArray != NULL)
+ {
+ mxDestroyArray(_xArray);
+ mxDestroyArray(_yArray);
+ }
+ _xArray = mxCreateDoubleMatrix(1, _xData.size(), mxREAL);
+ _yArray = mxCreateDoubleMatrix(1, _yData.size(), mxREAL);
+ }
+
+ if (!_xData.empty())
+ {
+ double* x = mxGetPr(_xArray);
+
+ std::list<double>::iterator it = _xData.begin();
+
+ for (int i = 0; it != _xData.end(); it++, i++)
+ {
+ x[i] = *it;
+ }
+ }
+
+ if (!_yData.empty())
+ {
+ double* y = mxGetPr(_yArray);
+
+ std::list<double>::iterator it = _yData.begin();
+
+ for (int i = 0; it != _yData.end(); it++, i++)
+ {
+ y[i] = *it;
+ }
+ }
+ *xData = _xArray;
+ *yData = _yArray;
+}
+
+std::string MatlabLine::GetXName()
+{
+ std::ostringstream xString;
+ xString << "x_" << _name;
+ return xString.str();
+}
+
+std::string MatlabLine::GetYName()
+{
+ std::ostringstream yString;
+ yString << "y_" << _name;
+ return yString.str();
+}
+
+std::string MatlabLine::GetPlotString()
+{
+
+ std::ostringstream s;
+
+ if (_xData.size() == 0)
+ {
+ s << "[0 1], [0 1]"; // To get an empty plot
+ }
+ else
+ {
+ s << GetXName() << "(1:" << _xData.size() << "),";
+ s << GetYName() << "(1:" << _yData.size() << ")";
+ }
+
+ s << ", '";
+ s << _plotAttribute;
+ s << "'";
+
+ return s.str();
+}
+
+std::string MatlabLine::GetRefreshString()
+{
+ std::ostringstream s;
+
+ if (_xData.size() > 0)
+ {
+ s << "set(h,'xdata',"<< GetXName() <<"(1:" << _xData.size() << "),'ydata',"<< GetYName() << "(1:" << _yData.size() << "));";
+ }
+ else
+ {
+ s << "set(h,'xdata',[NaN],'ydata',[NaN]);";
+ }
+ return s.str();
+}
+
+std::string MatlabLine::GetLegendString()
+{
+ return ("'" + _name + "'");
+}
+
+bool MatlabLine::hasLegend()
+{
+ return (!_name.empty());
+}
+
+
+// remove data points, but keep attributes
+void MatlabLine::Reset()
+{
+ _xData.clear();
+ _yData.clear();
+}
+
+
+void MatlabLine::UpdateTrendLine(MatlabLine * sourceData, double slope, double offset)
+{
+ Reset(); // reset data, not attributes and name
+
+ double thexMin = sourceData->xMin();
+ double thexMax = sourceData->xMax();
+ Append(thexMin, thexMin * slope + offset);
+ Append(thexMax, thexMax * slope + offset);
+}
+
+double MatlabLine::xMin()
+{
+ if (!_xData.empty())
+ {
+ std::list<double>::iterator theStart = _xData.begin();
+ std::list<double>::iterator theEnd = _xData.end();
+ return(*min_element(theStart, theEnd));
+ }
+ return (0.0);
+}
+
+double MatlabLine::xMax()
+{
+ if (!_xData.empty())
+ {
+ std::list<double>::iterator theStart = _xData.begin();
+ std::list<double>::iterator theEnd = _xData.end();
+ return(*max_element(theStart, theEnd));
+ }
+ return (0.0);
+}
+
+double MatlabLine::yMin()
+{
+ if (!_yData.empty())
+ {
+ std::list<double>::iterator theStart = _yData.begin();
+ std::list<double>::iterator theEnd = _yData.end();
+ return(*min_element(theStart, theEnd));
+ }
+ return (0.0);
+}
+
+double MatlabLine::yMax()
+{
+ if (!_yData.empty())
+ {
+ std::list<double>::iterator theStart = _yData.begin();
+ std::list<double>::iterator theEnd = _yData.end();
+ return(*max_element(theStart, theEnd));
+ }
+ return (0.0);
+}
+
+
+
+MatlabTimeLine::MatlabTimeLine(int horizonSeconds /*= -1*/, const char *plotAttrib /*= NULL*/,
+ const char *name /*= NULL*/,
+ WebRtc_Word64 refTimeMs /* = -1*/)
+ :
+_timeHorizon(horizonSeconds),
+MatlabLine(-1, plotAttrib, name) // infinite number of elements
+{
+ if (refTimeMs < 0)
+ _refTimeMs = TickTime::MillisecondTimestamp();
+ else
+ _refTimeMs = refTimeMs;
+}
+
+void MatlabTimeLine::Append(double y)
+{
+ MatlabLine::Append(static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0, y);
+
+ PurgeOldData();
+}
+
+
+void MatlabTimeLine::PurgeOldData()
+{
+ if (_timeHorizon > 0)
+ {
+ // remove old data
+ double historyLimit = static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0
+ - _timeHorizon; // remove data points older than this
+
+ std::list<double>::reverse_iterator ritx = _xData.rbegin();
+ WebRtc_UWord32 removeCount = 0;
+ while (ritx != _xData.rend())
+ {
+ if (*ritx >= historyLimit)
+ {
+ break;
+ }
+ ritx++;
+ removeCount++;
+ }
+ if (removeCount == 0)
+ {
+ return;
+ }
+
+ // remove the range [begin, it).
+ //if (removeCount > 10)
+ //{
+ // printf("Removing %lu elements\n", removeCount);
+ //}
+ _xData.resize(_xData.size() - removeCount);
+ _yData.resize(_yData.size() - removeCount);
+ }
+}
+
+
+WebRtc_Word64 MatlabTimeLine::GetRefTime()
+{
+ return(_refTimeMs);
+}
+
+
+
+
+MatlabPlot::MatlabPlot()
+:
+_figHandle(-1),
+_smartAxis(false),
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_timeToPlot(false),
+_plotting(false),
+_enabled(true),
+_firstPlot(true),
+_legendEnabled(true),
+_donePlottingEvent(EventWrapper::Create())
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _xlim[0] = 0;
+ _xlim[1] = 0;
+ _ylim[0] = 0;
+ _ylim[1] = 0;
+
+#ifdef PLOT_TESTING
+ _plotStartTime = -1;
+ _plotDelay = 0;
+#endif
+
+}
+
+
+MatlabPlot::~MatlabPlot()
+{
+ _critSect->Enter();
+
+ // delete all line objects
+ while (!_line.empty())
+ {
+ delete *(_line.end() - 1);
+ _line.pop_back();
+ }
+
+ delete _critSect;
+ delete _donePlottingEvent;
+}
+
+
+int MatlabPlot::AddLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ MatlabLine *newLine = new MatlabLine(maxLen, plotAttrib, name);
+ _line.push_back(newLine);
+
+ return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
+}
+
+
+int MatlabPlot::AddTimeLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/,
+ WebRtc_Word64 refTimeMs /*= -1*/)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ MatlabTimeLine *newLine = new MatlabTimeLine(maxLen, plotAttrib, name, refTimeMs);
+ _line.push_back(newLine);
+
+ return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
+}
+
+
+int MatlabPlot::GetLineIx(const char *name)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ // search the list for a matching line name
+ std::vector<MatlabLine*>::iterator it = _line.begin();
+ bool matchFound = false;
+ int lineIx = 0;
+
+ for (; it != _line.end(); it++, lineIx++)
+ {
+ if ((*it)->_name == name)
+ {
+ matchFound = true;
+ break;
+ }
+ }
+
+ if (matchFound)
+ {
+ return (lineIx);
+ }
+ else
+ {
+ return (-1);
+ }
+}
+
+
+void MatlabPlot::Append(int lineIndex, double x, double y)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return;
+ }
+
+ // sanity for index
+ if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
+ {
+ throw "Line index out of range";
+ exit(1);
+ }
+
+ return (_line[lineIndex]->Append(x, y));
+}
+
+
+void MatlabPlot::Append(int lineIndex, double y)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return;
+ }
+
+ // sanity for index
+ if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
+ {
+ throw "Line index out of range";
+ exit(1);
+ }
+
+ return (_line[lineIndex]->Append(y));
+}
+
+
+int MatlabPlot::Append(const char *name, double x, double y)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ // search the list for a matching line name
+ int lineIx = GetLineIx(name);
+
+ if (lineIx < 0) //(!matchFound)
+ {
+ // no match; append new line
+ lineIx = AddLine(-1, NULL, name);
+ }
+
+ // append data to line
+ Append(lineIx, x, y);
+ return (lineIx);
+}
+
+int MatlabPlot::Append(const char *name, double y)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ // search the list for a matching line name
+ int lineIx = GetLineIx(name);
+
+ if (lineIx < 0) //(!matchFound)
+ {
+ // no match; append new line
+ lineIx = AddLine(-1, NULL, name);
+ }
+
+ // append data to line
+ Append(lineIx, y);
+ return (lineIx);
+}
+
+int MatlabPlot::Length(char *name)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return -1;
+ }
+
+ int ix = GetLineIx(name);
+ if (ix >= 0)
+ {
+ return (static_cast<int>(_line[ix]->_xData.size()));
+ }
+ else
+ {
+ return (-1);
+ }
+}
+
+
+void MatlabPlot::SetPlotAttribute(char *name, char *plotAttrib)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_enabled)
+ {
+ return;
+ }
+
+ int lineIx = GetLineIx(name);
+
+ if (lineIx >= 0)
+ {
+ _line[lineIx]->SetAttribute(plotAttrib);
+ }
+}
+
+// Must be called under critical section _critSect
+void MatlabPlot::UpdateData(Engine* ep)
+{
+ if (!_enabled)
+ {
+ return;
+ }
+
+ for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
+ {
+ mxArray* xData = NULL;
+ mxArray* yData = NULL;
+ (*it)->GetPlotData(&xData, &yData);
+ if (xData != NULL)
+ {
+ std::string xName = (*it)->GetXName();
+ std::string yName = (*it)->GetYName();
+ _critSect->Leave();
+#ifdef MATLAB6
+ mxSetName(xData, xName.c_str());
+ mxSetName(yData, yName.c_str());
+ engPutArray(ep, xData);
+ engPutArray(ep, yData);
+#else
+ int ret = engPutVariable(ep, xName.c_str(), xData);
+ assert(ret == 0);
+ ret = engPutVariable(ep, yName.c_str(), yData);
+ assert(ret == 0);
+#endif
+ _critSect->Enter();
+ }
+ }
+}
+
+bool MatlabPlot::GetPlotCmd(std::ostringstream & cmd, Engine* ep)
+{
+ _critSect->Enter();
+
+ if (!DataAvailable())
+ {
+ return false;
+ }
+
+ if (_firstPlot)
+ {
+ GetPlotCmd(cmd);
+ _firstPlot = false;
+ }
+ else
+ {
+ GetRefreshCmd(cmd);
+ }
+
+ UpdateData(ep);
+
+ _critSect->Leave();
+
+ return true;
+}
+
+// Call inside critsect
+void MatlabPlot::GetPlotCmd(std::ostringstream & cmd)
+{
+ // we have something to plot
+ // empty the stream
+ cmd.str(""); // (this seems to be the only way)
+
+ cmd << "figure; h" << _figHandle << "= plot(";
+
+ // first line
+ std::vector<MatlabLine*>::iterator it = _line.begin();
+ cmd << (*it)->GetPlotString();
+
+ it++;
+
+ // remaining lines
+ for (; it != _line.end(); it++)
+ {
+ cmd << ", ";
+ cmd << (*it)->GetPlotString();
+ }
+
+ cmd << "); ";
+
+ if (_legendEnabled)
+ {
+ GetLegendCmd(cmd);
+ }
+
+ if (_smartAxis)
+ {
+ double xMin = _xlim[0];
+ double xMax = _xlim[1];
+ double yMax = _ylim[1];
+ for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
+ {
+ xMax = std::max(xMax, (*it)->xMax());
+ xMin = std::min(xMin, (*it)->xMin());
+
+ yMax = std::max(yMax, (*it)->yMax());
+ yMax = std::max(yMax, fabs((*it)->yMin()));
+ }
+ _xlim[0] = xMin;
+ _xlim[1] = xMax;
+ _ylim[0] = -yMax;
+ _ylim[1] = yMax;
+
+ cmd << "axis([" << _xlim[0] << ", " << _xlim[1] << ", " << _ylim[0] << ", " << _ylim[1] << "]);";
+ }
+
+ int i=1;
+ for (it = _line.begin(); it != _line.end(); i++, it++)
+ {
+ cmd << "set(h" << _figHandle << "(" << i << "), 'Tag', " << (*it)->GetLegendString() << ");";
+ }
+}
+
+// Call inside critsect
+void MatlabPlot::GetRefreshCmd(std::ostringstream & cmd)
+{
+ cmd.str(""); // (this seems to be the only way)
+ std::vector<MatlabLine*>::iterator it = _line.begin();
+ for (it = _line.begin(); it != _line.end(); it++)
+ {
+ cmd << "h = findobj(0, 'Tag', " << (*it)->GetLegendString() << ");";
+ cmd << (*it)->GetRefreshString();
+ }
+ //if (_legendEnabled)
+ //{
+ // GetLegendCmd(cmd);
+ //}
+}
+
+void MatlabPlot::GetLegendCmd(std::ostringstream & cmd)
+{
+ std::vector<MatlabLine*>::iterator it = _line.begin();
+ bool anyLegend = false;
+ for (; it != _line.end(); it++)
+ {
+ anyLegend = anyLegend || (*it)->hasLegend();
+ }
+ if (anyLegend)
+ {
+ // create the legend
+
+ cmd << "legend(h" << _figHandle << ",{";
+
+
+ // iterate lines
+ int i = 0;
+ for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
+ {
+ if (i > 0)
+ {
+ cmd << ", ";
+ }
+ cmd << (*it)->GetLegendString();
+ i++;
+ }
+
+ cmd << "}, 2); "; // place legend in upper-left corner
+ }
+}
+
+// Call inside critsect
+bool MatlabPlot::DataAvailable()
+{
+ if (!_enabled)
+ {
+ return false;
+ }
+
+ for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
+ {
+ (*it)->PurgeOldData();
+ }
+
+ return true;
+}
+
+void MatlabPlot::Plot()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _timeToPlot = true;
+
+#ifdef PLOT_TESTING
+ _plotStartTime = TickTime::MillisecondTimestamp();
+#endif
+}
+
+
+void MatlabPlot::Reset()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _enabled = true;
+
+ for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
+ {
+ (*it)->Reset();
+ }
+
+}
+
+void MatlabPlot::SetFigHandle(int handle)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (handle > 0)
+ _figHandle = handle;
+}
+
+bool
+MatlabPlot::TimeToPlot()
+{
+ CriticalSectionScoped cs(_critSect);
+ return _enabled && _timeToPlot;
+}
+
+void
+MatlabPlot::Plotting()
+{
+ CriticalSectionScoped cs(_critSect);
+ _plotting = true;
+}
+
+void
+MatlabPlot::DonePlotting()
+{
+ CriticalSectionScoped cs(_critSect);
+ _timeToPlot = false;
+ _plotting = false;
+ _donePlottingEvent->Set();
+}
+
+void
+MatlabPlot::DisablePlot()
+{
+ _critSect->Enter();
+ while (_plotting)
+ {
+ _critSect->Leave();
+ _donePlottingEvent->Wait(WEBRTC_EVENT_INFINITE);
+ _critSect->Enter();
+ }
+ _enabled = false;
+}
+
+int MatlabPlot::MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ int sourceIx;
+ int trendIx;
+
+ sourceIx = GetLineIx(sourceName);
+ if (sourceIx < 0)
+ {
+ // could not find source
+ return (-1);
+ }
+
+ trendIx = GetLineIx(trendName);
+ if (trendIx < 0)
+ {
+ // no trend found; add new line
+ trendIx = AddLine(2 /*maxLen*/, plotAttrib, trendName);
+ }
+
+ _line[trendIx]->UpdateTrendLine(_line[sourceIx], slope, offset);
+
+ return (trendIx);
+
+}
+
+
+MatlabEngine::MatlabEngine()
+:
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_eventPtr(NULL),
+_plotThread(NULL),
+_running(false),
+_numPlots(0)
+{
+ _eventPtr = EventWrapper::Create();
+
+ _plotThread = ThreadWrapper::CreateThread(MatlabEngine::PlotThread, this, kLowPriority, "MatlabPlot");
+
+ if (_plotThread == NULL)
+ {
+ throw "Unable to start MatlabEngine thread";
+ exit(1);
+ }
+
+ _running = true;
+
+ unsigned int tid;
+ _plotThread->Start(tid);
+
+}
+
+MatlabEngine::~MatlabEngine()
+{
+ _critSect->Enter();
+
+ if (_plotThread)
+ {
+ _plotThread->SetNotAlive();
+ _running = false;
+ _eventPtr->Set();
+
+ while (!_plotThread->Stop())
+ {
+ ;
+ }
+
+ delete _plotThread;
+ }
+
+ _plots.clear();
+
+ _plotThread = NULL;
+
+ delete _eventPtr;
+ _eventPtr = NULL;
+
+ _critSect->Leave();
+ delete _critSect;
+
+}
+
+MatlabPlot * MatlabEngine::NewPlot(MatlabPlot *newPlot)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ //MatlabPlot *newPlot = new MatlabPlot();
+
+ if (newPlot)
+ {
+ newPlot->SetFigHandle(++_numPlots); // first plot is number 1
+ _plots.push_back(newPlot);
+ }
+
+ return (newPlot);
+
+}
+
+
+void MatlabEngine::DeletePlot(MatlabPlot *plot)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (plot == NULL)
+ {
+ return;
+ }
+
+ std::vector<MatlabPlot *>::iterator it;
+ for (it = _plots.begin(); it < _plots.end(); it++)
+ {
+ if (plot == *it)
+ {
+ break;
+ }
+ }
+
+ assert (plot == *it);
+
+ (*it)->DisablePlot();
+
+ _plots.erase(it);
+ --_numPlots;
+
+ delete plot;
+}
+
+
+bool MatlabEngine::PlotThread(void *obj)
+{
+ if (!obj)
+ {
+ return (false);
+ }
+
+ MatlabEngine *eng = (MatlabEngine *) obj;
+
+ Engine *ep = engOpen(NULL);
+ if (!ep)
+ {
+ throw "Cannot open Matlab engine";
+ return (false);
+ }
+
+ engSetVisible(ep, true);
+ engEvalString(ep, "close all;");
+
+ while (eng->_running)
+ {
+ eng->_critSect->Enter();
+
+ // iterate through all plots
+ for (unsigned int ix = 0; ix < eng->_plots.size(); ix++)
+ {
+ MatlabPlot *plot = eng->_plots[ix];
+ if (plot->TimeToPlot())
+ {
+ plot->Plotting();
+ eng->_critSect->Leave();
+ std::ostringstream cmd;
+
+ if (engEvalString(ep, cmd.str().c_str()))
+ {
+ // engine dead
+ return (false);
+ }
+
+ // empty the stream
+ cmd.str(""); // (this seems to be the only way)
+ if (plot->GetPlotCmd(cmd, ep))
+ {
+ // things to plot, we have already accessed what we need in the plot
+ plot->DonePlotting();
+
+ WebRtc_Word64 start = TickTime::MillisecondTimestamp();
+ // plot it
+ int ret = engEvalString(ep, cmd.str().c_str());
+ printf("time=%I64i\n", TickTime::MillisecondTimestamp() - start);
+ if (ret)
+ {
+ // engine dead
+ return (false);
+ }
+
+#ifdef PLOT_TESTING
+ if(plot->_plotStartTime >= 0)
+ {
+ plot->_plotDelay = TickTime::MillisecondTimestamp() - plot->_plotStartTime;
+ plot->_plotStartTime = -1;
+ }
+#endif
+ }
+ eng->_critSect->Enter();
+ }
+ }
+
+ eng->_critSect->Leave();
+ // wait a while
+ eng->_eventPtr->Wait(66); // 33 ms
+ }
+
+ if (ep)
+ {
+ engClose(ep);
+ ep = NULL;
+ }
+
+ return (true);
+
+}
+
+#endif // MATLAB
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
new file mode 100644
index 0000000..08c7006
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
@@ -0,0 +1,170 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
+
+#include <list>
+#include <string>
+#include <vector>
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+}
+
+//#define PLOT_TESTING
+
+#ifdef MATLAB
+
+typedef struct engine Engine;
+typedef struct mxArray_tag mxArray;
+
+class MatlabLine
+{
+ friend class MatlabPlot;
+
+public:
+ MatlabLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
+ ~MatlabLine();
+ virtual void Append(double x, double y);
+ virtual void Append(double y);
+ void SetMaxLen(int maxLen);
+ void SetAttribute(char *plotAttrib);
+ void SetName(char *name);
+ void Reset();
+ virtual void PurgeOldData() {};
+
+ void UpdateTrendLine(MatlabLine * sourceData, double slope, double offset);
+
+ double xMin();
+ double xMax();
+ double yMin();
+ double yMax();
+
+protected:
+ void GetPlotData(mxArray** xData, mxArray** yData);
+ std::string GetXName();
+ std::string GetYName();
+ std::string GetPlotString();
+ std::string GetRefreshString();
+ std::string GetLegendString();
+ bool hasLegend();
+ std::list<double> _xData;
+ std::list<double> _yData;
+ mxArray* _xArray;
+ mxArray* _yArray;
+ int _maxLen;
+ std::string _plotAttribute;
+ std::string _name;
+};
+
+
+class MatlabTimeLine : public MatlabLine
+{
+public:
+ MatlabTimeLine(int horizonSeconds = -1, const char *plotAttrib = NULL, const char *name = NULL,
+ WebRtc_Word64 refTimeMs = -1);
+ ~MatlabTimeLine() {};
+ void Append(double y);
+ void PurgeOldData();
+ WebRtc_Word64 GetRefTime();
+
+private:
+ WebRtc_Word64 _refTimeMs;
+ int _timeHorizon;
+};
+
+
+class MatlabPlot
+{
+ friend class MatlabEngine;
+
+public:
+ MatlabPlot();
+ ~MatlabPlot();
+
+ int AddLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
+ int AddTimeLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL,
+ WebRtc_Word64 refTimeMs = -1);
+ int GetLineIx(const char *name);
+ void Append(int lineIndex, double x, double y);
+ void Append(int lineIndex, double y);
+ int Append(const char *name, double x, double y);
+ int Append(const char *name, double y);
+ int Length(char *name);
+ void SetPlotAttribute(char *name, char *plotAttrib);
+ void Plot();
+ void Reset();
+ void SmartAxis(bool status = true) { _smartAxis = status; };
+ void SetFigHandle(int handle);
+ void EnableLegend(bool enable) { _legendEnabled = enable; };
+
+ bool TimeToPlot();
+ void Plotting();
+ void DonePlotting();
+ void DisablePlot();
+
+ int MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib = NULL);
+
+#ifdef PLOT_TESTING
+ WebRtc_Word64 _plotStartTime;
+ WebRtc_Word64 _plotDelay;
+#endif
+
+private:
+ void UpdateData(Engine* ep);
+ bool GetPlotCmd(std::ostringstream & cmd, Engine* ep);
+ void GetPlotCmd(std::ostringstream & cmd); // call inside crit sect
+ void GetRefreshCmd(std::ostringstream & cmd); // call inside crit sect
+ void GetLegendCmd(std::ostringstream & cmd);
+ bool DataAvailable();
+
+ std::vector<MatlabLine *> _line;
+ int _figHandle;
+ bool _smartAxis;
+ double _xlim[2];
+ double _ylim[2];
+ webrtc::CriticalSectionWrapper *_critSect;
+ bool _timeToPlot;
+ bool _plotting;
+ bool _enabled;
+ bool _firstPlot;
+ bool _legendEnabled;
+ webrtc::EventWrapper* _donePlottingEvent;
+};
+
+
+class MatlabEngine
+{
+public:
+ MatlabEngine();
+ ~MatlabEngine();
+
+ MatlabPlot * NewPlot(MatlabPlot *newPlot);
+ void DeletePlot(MatlabPlot *plot);
+
+private:
+ static bool PlotThread(void *obj);
+
+ std::vector<MatlabPlot *> _plots;
+ webrtc::CriticalSectionWrapper *_critSect;
+ webrtc::EventWrapper *_eventPtr;
+ webrtc::ThreadWrapper* _plotThread;
+ bool _running;
+ int _numPlots;
+};
+
+#endif //MATLAB
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
new file mode 100644
index 0000000..d322242
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
@@ -0,0 +1,438 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm> // for max function
+#include <stdio.h>
+
+#include "TestLoadGenerator.h"
+#include "TestSenderReceiver.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "tick_util.h"
+
+
+bool SenderThreadFunction(void *obj)
+{
+ if (obj == NULL)
+ {
+ return false;
+ }
+ TestLoadGenerator *_genObj = static_cast<TestLoadGenerator *>(obj);
+
+ return _genObj->GeneratorLoop();
+}
+
+
+TestLoadGenerator::TestLoadGenerator(TestSenderReceiver *sender, WebRtc_Word32 rtpSampleRate)
+:
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_eventPtr(NULL),
+_genThread(NULL),
+_bitrateKbps(0),
+_sender(sender),
+_running(false),
+_rtpSampleRate(rtpSampleRate)
+{
+}
+
+TestLoadGenerator::~TestLoadGenerator ()
+{
+ if (_running)
+ {
+ Stop();
+ }
+
+ delete _critSect;
+}
+
+WebRtc_Word32 TestLoadGenerator::SetBitrate (WebRtc_Word32 newBitrateKbps)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (newBitrateKbps < 0)
+ {
+ return -1;
+ }
+
+ _bitrateKbps = newBitrateKbps;
+
+ printf("New bitrate = %i kbps\n", _bitrateKbps);
+
+ return _bitrateKbps;
+}
+
+
+WebRtc_Word32 TestLoadGenerator::Start (const char *threadName)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _eventPtr = EventWrapper::Create();
+
+ _genThread = ThreadWrapper::CreateThread(SenderThreadFunction, this, kRealtimePriority, threadName);
+ if (_genThread == NULL)
+ {
+ throw "Unable to start generator thread";
+ exit(1);
+ }
+
+ _running = true;
+
+ unsigned int tid;
+ _genThread->Start(tid);
+
+ return 0;
+}
+
+
+WebRtc_Word32 TestLoadGenerator::Stop ()
+{
+ _critSect.Enter();
+
+ if (_genThread)
+ {
+ _genThread->SetNotAlive();
+ _running = false;
+ _eventPtr->Set();
+
+ while (!_genThread->Stop())
+ {
+ _critSect.Leave();
+ _critSect.Enter();
+ }
+
+ delete _genThread;
+ _genThread = NULL;
+
+ delete _eventPtr;
+ _eventPtr = NULL;
+ }
+
+ _genThread = NULL;
+ _critSect.Leave();
+ return (0);
+}
+
+
+int TestLoadGenerator::generatePayload ()
+{
+ return(generatePayload( static_cast<WebRtc_UWord32>( TickTime::MillisecondTimestamp() * _rtpSampleRate / 1000 )));
+}
+
+
+int TestLoadGenerator::sendPayload (const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
+{
+
+ return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType));
+}
+
+
+CBRGenerator::CBRGenerator (TestSenderReceiver *sender, WebRtc_Word32 payloadSizeBytes, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate)
+:
+//_eventPtr(NULL),
+_payloadSizeBytes(payloadSizeBytes),
+_payload(new WebRtc_UWord8[payloadSizeBytes]),
+TestLoadGenerator(sender, rtpSampleRate)
+{
+ SetBitrate (bitrateKbps);
+}
+
+CBRGenerator::~CBRGenerator ()
+{
+ if (_running)
+ {
+ Stop();
+ }
+
+ if (_payload)
+ {
+ delete [] _payload;
+ }
+
+}
+
+bool CBRGenerator::GeneratorLoop ()
+{
+ double periodMs;
+ WebRtc_Word64 nextSendTime = TickTime::MillisecondTimestamp();
+
+
+ // no critSect
+ while (_running)
+ {
+ // send data (critSect inside)
+ generatePayload( static_cast<WebRtc_UWord32>(nextSendTime * _rtpSampleRate / 1000) );
+
+ // calculate wait time
+ periodMs = 8.0 * _payloadSizeBytes / ( _bitrateKbps );
+
+ nextSendTime = static_cast<WebRtc_Word64>(nextSendTime + periodMs);
+
+ WebRtc_Word32 waitTime = static_cast<WebRtc_Word32>(nextSendTime - TickTime::MillisecondTimestamp());
+ if (waitTime < 0)
+ {
+ waitTime = 0;
+ }
+ // wait
+ _eventPtr->Wait(static_cast<WebRtc_Word32>(waitTime));
+ }
+
+ return true;
+}
+
+int CBRGenerator::generatePayload ( WebRtc_UWord32 timestamp )
+{
+ CriticalSectionScoped cs(_critSect);
+
+ //WebRtc_UWord8 *payload = new WebRtc_UWord8[_payloadSizeBytes];
+
+ int ret = sendPayload(timestamp, _payload, _payloadSizeBytes);
+
+ //delete [] payload;
+ return ret;
+}
+
+
+
+
+/////////////////////
+
+CBRFixFRGenerator::CBRFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
+ WebRtc_Word32 rtpSampleRate, WebRtc_Word32 frameRateFps /*= 30*/,
+ double spread /*= 0.0*/)
+:
+//_eventPtr(NULL),
+_payloadSizeBytes(0),
+_payload(NULL),
+_payloadAllocLen(0),
+_frameRateFps(frameRateFps),
+_spreadFactor(spread),
+TestLoadGenerator(sender, rtpSampleRate)
+{
+ SetBitrate (bitrateKbps);
+}
+
+CBRFixFRGenerator::~CBRFixFRGenerator ()
+{
+ if (_running)
+ {
+ Stop();
+ }
+
+ if (_payload)
+ {
+ delete [] _payload;
+ _payloadAllocLen = 0;
+ }
+
+}
+
+bool CBRFixFRGenerator::GeneratorLoop ()
+{
+ double periodMs;
+ WebRtc_Word64 nextSendTime = TickTime::MillisecondTimestamp();
+
+ _critSect.Enter();
+
+ if (_frameRateFps <= 0)
+ {
+ return false;
+ }
+
+ _critSect.Leave();
+
+ // no critSect
+ while (_running)
+ {
+ _critSect.Enter();
+
+ // calculate payload size
+ _payloadSizeBytes = nextPayloadSize();
+
+ if (_payloadSizeBytes > 0)
+ {
+
+ if (_payloadAllocLen < _payloadSizeBytes * (1 + _spreadFactor))
+ {
+ // re-allocate _payload
+ if (_payload)
+ {
+ delete [] _payload;
+ _payload = NULL;
+ }
+
+ _payloadAllocLen = static_cast<WebRtc_Word32>((_payloadSizeBytes * (1 + _spreadFactor) * 3) / 2 + .5); // 50% extra to avoid frequent re-alloc
+ _payload = new WebRtc_UWord8[_payloadAllocLen];
+ }
+
+
+ // send data (critSect inside)
+ generatePayload( static_cast<WebRtc_UWord32>(nextSendTime * _rtpSampleRate / 1000) );
+ }
+
+ _critSect.Leave();
+
+ // calculate wait time
+ periodMs = 1000.0 / _frameRateFps;
+ nextSendTime = static_cast<WebRtc_Word64>(nextSendTime + periodMs + 0.5);
+
+ WebRtc_Word32 waitTime = static_cast<WebRtc_Word32>(nextSendTime - TickTime::MillisecondTimestamp());
+ if (waitTime < 0)
+ {
+ waitTime = 0;
+ }
+ // wait
+ _eventPtr->Wait(waitTime);
+ }
+
+ return true;
+}
+
+WebRtc_Word32 CBRFixFRGenerator::nextPayloadSize()
+{
+ const double periodMs = 1000.0 / _frameRateFps;
+ return static_cast<WebRtc_Word32>(_bitrateKbps * periodMs / 8 + 0.5);
+}
+
+int CBRFixFRGenerator::generatePayload ( WebRtc_UWord32 timestamp )
+{
+ CriticalSectionScoped cs(_critSect);
+
+ double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
+ factor = 1 + 2 * _spreadFactor * factor; // [1 - _spreadFactor ; 1 + _spreadFactor]
+
+ WebRtc_Word32 thisPayloadBytes = static_cast<WebRtc_Word32>(_payloadSizeBytes * factor);
+ // sanity
+ if (thisPayloadBytes > _payloadAllocLen)
+ {
+ thisPayloadBytes = _payloadAllocLen;
+ }
+
+ int ret = sendPayload(timestamp, _payload, thisPayloadBytes);
+ return ret;
+}
+
+
+/////////////////////
+
+PeriodicKeyFixFRGenerator::PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
+ WebRtc_Word32 rtpSampleRate, WebRtc_Word32 frameRateFps /*= 30*/,
+ double spread /*= 0.0*/, double keyFactor /*= 4.0*/, WebRtc_UWord32 keyPeriod /*= 300*/)
+:
+_keyFactor(keyFactor),
+_keyPeriod(keyPeriod),
+_frameCount(0),
+CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRateFps, spread)
+{
+}
+
+WebRtc_Word32 PeriodicKeyFixFRGenerator::nextPayloadSize()
+{
+ // calculate payload size for a delta frame
+ WebRtc_Word32 payloadSizeBytes = static_cast<WebRtc_Word32>(1000 * _bitrateKbps / (8.0 * _frameRateFps * (1.0 + (_keyFactor - 1.0) / _keyPeriod)) + 0.5);
+
+ if (_frameCount % _keyPeriod == 0)
+ {
+ // this is a key frame, scale the payload size
+ payloadSizeBytes = static_cast<WebRtc_Word32>(_keyFactor * _payloadSizeBytes + 0.5);
+ }
+ _frameCount++;
+
+ return payloadSizeBytes;
+}
+
+////////////////////
+
+CBRVarFRGenerator::CBRVarFRGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, const WebRtc_UWord8* frameRates,
+ WebRtc_UWord16 numFrameRates, WebRtc_Word32 rtpSampleRate, double avgFrPeriodMs,
+ double frSpreadFactor, double spreadFactor)
+:
+_avgFrPeriodMs(avgFrPeriodMs),
+_frSpreadFactor(frSpreadFactor),
+_frameRates(NULL),
+_numFrameRates(numFrameRates),
+_frChangeTimeMs(TickTime::MillisecondTimestamp() + _avgFrPeriodMs),
+CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRates[0], spreadFactor)
+{
+ _frameRates = new WebRtc_UWord8[_numFrameRates];
+ memcpy(_frameRates, frameRates, _numFrameRates);
+}
+
+CBRVarFRGenerator::~CBRVarFRGenerator()
+{
+ delete [] _frameRates;
+}
+
+void CBRVarFRGenerator::ChangeFrameRate()
+{
+ const WebRtc_Word64 nowMs = TickTime::MillisecondTimestamp();
+ if (nowMs < _frChangeTimeMs)
+ {
+ return;
+ }
+ // Time to change frame rate
+ WebRtc_UWord16 frIndex = static_cast<WebRtc_UWord16>(static_cast<double>(rand()) / RAND_MAX
+ * (_numFrameRates - 1) + 0.5) ;
+ assert(frIndex < _numFrameRates);
+ _frameRateFps = _frameRates[frIndex];
+ // Update the next frame rate change time
+ double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
+ factor = 1 + 2 * _frSpreadFactor * factor; // [1 - _frSpreadFactor ; 1 + _frSpreadFactor]
+ _frChangeTimeMs = nowMs + static_cast<WebRtc_Word64>(1000.0 * factor *
+ _avgFrPeriodMs + 0.5);
+
+ printf("New frame rate: %d\n", _frameRateFps);
+}
+
+WebRtc_Word32 CBRVarFRGenerator::nextPayloadSize()
+{
+ ChangeFrameRate();
+ return CBRFixFRGenerator::nextPayloadSize();
+}
+
+////////////////////
+
+CBRFrameDropGenerator::CBRFrameDropGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
+ WebRtc_Word32 rtpSampleRate, double spreadFactor)
+:
+_accBits(0),
+CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, 30, spreadFactor)
+{
+}
+
+CBRFrameDropGenerator::~CBRFrameDropGenerator()
+{
+}
+
+WebRtc_Word32 CBRFrameDropGenerator::nextPayloadSize()
+{
+ _accBits -= 1000 * _bitrateKbps / _frameRateFps;
+ if (_accBits < 0)
+ {
+ _accBits = 0;
+ }
+ if (_accBits > 0.3 * _bitrateKbps * 1000)
+ {
+ //printf("drop\n");
+ return 0;
+ }
+ else
+ {
+ //printf("keep\n");
+ const double periodMs = 1000.0 / _frameRateFps;
+ WebRtc_Word32 frameSize = static_cast<WebRtc_Word32>(_bitrateKbps * periodMs / 8 + 0.5);
+ frameSize = std::max(frameSize, static_cast<WebRtc_Word32>(300 * periodMs / 8 + 0.5));
+ _accBits += frameSize * 8;
+ return frameSize;
+ }
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
new file mode 100644
index 0000000..c22591c
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
@@ -0,0 +1,146 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
+
+#include <stdlib.h>
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+class TestSenderReceiver;
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+}
+
+class TestLoadGenerator
+{
+public:
+ TestLoadGenerator (TestSenderReceiver *sender, WebRtc_Word32 rtpSampleRate = 90000);
+ virtual ~TestLoadGenerator ();
+
+ WebRtc_Word32 SetBitrate (WebRtc_Word32 newBitrateKbps);
+ virtual WebRtc_Word32 Start (const char *threadName = NULL);
+ virtual WebRtc_Word32 Stop ();
+ virtual bool GeneratorLoop () = 0;
+
+protected:
+ virtual int generatePayload ( WebRtc_UWord32 timestamp ) = 0;
+ int generatePayload ();
+ int sendPayload (const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
+
+ webrtc::CriticalSectionWrapper* _critSect;
+ webrtc::EventWrapper *_eventPtr;
+ webrtc::ThreadWrapper* _genThread;
+ WebRtc_Word32 _bitrateKbps;
+ TestSenderReceiver *_sender;
+ bool _running;
+ WebRtc_Word32 _rtpSampleRate;
+};
+
+
+class CBRGenerator : public TestLoadGenerator
+{
+public:
+ CBRGenerator (TestSenderReceiver *sender, WebRtc_Word32 payloadSizeBytes, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000);
+ virtual ~CBRGenerator ();
+
+ virtual WebRtc_Word32 Start () {return (TestLoadGenerator::Start("CBRGenerator"));};
+
+ virtual bool GeneratorLoop ();
+
+protected:
+ virtual int generatePayload ( WebRtc_UWord32 timestamp );
+
+ WebRtc_Word32 _payloadSizeBytes;
+ WebRtc_UWord8 *_payload;
+};
+
+
+class CBRFixFRGenerator : public TestLoadGenerator // constant bitrate and fixed frame rate
+{
+public:
+ CBRFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000,
+ WebRtc_Word32 frameRateFps = 30, double spread = 0.0);
+ virtual ~CBRFixFRGenerator ();
+
+ virtual WebRtc_Word32 Start () {return (TestLoadGenerator::Start("CBRFixFRGenerator"));};
+
+ virtual bool GeneratorLoop ();
+
+protected:
+ virtual WebRtc_Word32 nextPayloadSize ();
+ virtual int generatePayload ( WebRtc_UWord32 timestamp );
+
+ WebRtc_Word32 _payloadSizeBytes;
+ WebRtc_UWord8 *_payload;
+ WebRtc_Word32 _payloadAllocLen;
+ WebRtc_Word32 _frameRateFps;
+ double _spreadFactor;
+};
+
+class PeriodicKeyFixFRGenerator : public CBRFixFRGenerator // constant bitrate and fixed frame rate with periodically large frames
+{
+public:
+ PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000,
+ WebRtc_Word32 frameRateFps = 30, double spread = 0.0, double keyFactor = 4.0, WebRtc_UWord32 keyPeriod = 300);
+ virtual ~PeriodicKeyFixFRGenerator () {}
+
+protected:
+ virtual WebRtc_Word32 nextPayloadSize ();
+
+ double _keyFactor;
+ WebRtc_UWord32 _keyPeriod;
+ WebRtc_UWord32 _frameCount;
+};
+
+// Probably better to inherit CBRFixFRGenerator from CBRVarFRGenerator, but since
+// the fix FR version already existed this was easier.
+class CBRVarFRGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
+{
+public:
+ CBRVarFRGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, const WebRtc_UWord8* frameRates,
+ WebRtc_UWord16 numFrameRates, WebRtc_Word32 rtpSampleRate = 90000, double avgFrPeriodMs = 5.0,
+ double frSpreadFactor = 0.05, double spreadFactor = 0.0);
+
+ ~CBRVarFRGenerator();
+
+protected:
+ virtual void ChangeFrameRate();
+ virtual WebRtc_Word32 nextPayloadSize ();
+
+ double _avgFrPeriodMs;
+ double _frSpreadFactor;
+ WebRtc_UWord8* _frameRates;
+ WebRtc_UWord16 _numFrameRates;
+ WebRtc_Word64 _frChangeTimeMs;
+};
+
+class CBRFrameDropGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
+{
+public:
+ CBRFrameDropGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
+ WebRtc_Word32 rtpSampleRate = 90000, double spreadFactor = 0.0);
+
+ ~CBRFrameDropGenerator();
+
+protected:
+ virtual WebRtc_Word32 nextPayloadSize();
+
+ double _accBits;
+};
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
new file mode 100644
index 0000000..1fc0fd3
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
@@ -0,0 +1,442 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "rtp_rtcp.h"
+#include "udp_transport.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "critical_section_wrapper.h"
+#include "TestSenderReceiver.h"
+#include "TestLoadGenerator.h"
+#include <stdlib.h>
+
+#define NR_OF_SOCKET_BUFFERS 500
+
+
+bool ProcThreadFunction(void *obj)
+{
+ if (obj == NULL)
+ {
+ return false;
+ }
+ TestSenderReceiver *theObj = static_cast<TestSenderReceiver *>(obj);
+
+ return theObj->ProcLoop();
+}
+
+
+TestSenderReceiver::TestSenderReceiver (void)
+:
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_eventPtr(NULL),
+_procThread(NULL),
+_running(false),
+_payloadType(0),
+_loadGenerator(NULL),
+_isSender(false),
+_isReceiver(false),
+_timeOut(false),
+_sendRecCB(NULL),
+_lastBytesReceived(0),
+_lastTime(-1)
+{
+ // RTP/RTCP module
+ _rtp = RtpRtcp::CreateRtpRtcp(0, false);
+ if (!_rtp)
+ {
+ throw "Could not create RTP/RTCP module";
+ exit(1);
+ }
+
+ if (_rtp->InitReceiver() != 0)
+ {
+ throw "_rtp->InitReceiver()";
+ exit(1);
+ }
+
+ if (_rtp->InitSender() != 0)
+ {
+ throw "_rtp->InitSender()";
+ exit(1);
+ }
+
+ // SocketTransport module
+ WebRtc_UWord8 numberOfThreads = 1;
+ _transport = UdpTransport::Create(0, numberOfThreads);
+ if (!_transport)
+ {
+ throw "Could not create transport module";
+ exit(1);
+ }
+}
+
+TestSenderReceiver::~TestSenderReceiver (void)
+{
+
+ Stop(); // N.B. without critSect
+
+ _critSect->Enter();
+
+ if (_rtp)
+ {
+ RtpRtcp::DestroyRtpRtcp(_rtp);
+ _rtp = NULL;
+ }
+
+ if (_transport)
+ {
+ UdpTransport::Destroy(_transport);
+ _transport = NULL;
+ }
+
+ delete _critSect;
+
+}
+
+
+WebRtc_Word32 TestSenderReceiver::InitReceiver (const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort,
+ const WebRtc_Word8 payloadType /*= 127*/)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ // init transport
+ if (_transport->InitializeReceiveSockets(this, rtpPort/*, 0, NULL, 0, true*/) != 0)
+ {
+ throw "_transport->InitializeReceiveSockets";
+ exit(1);
+ }
+
+ if (_rtp->RegisterIncomingRTPCallback(this) != 0)
+ {
+ throw "_rtp->RegisterIncomingRTPCallback";
+ exit(1);
+ }
+
+ if (_rtp->RegisterIncomingDataCallback(this) != 0)
+ {
+ throw "_rtp->RegisterIncomingRTPCallback";
+ exit(1);
+ }
+
+ if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
+ {
+ throw "_rtp->SetRTCPStatus";
+ exit(1);
+ }
+
+ if (_rtp->SetTMMBRStatus(true) != 0)
+ {
+ throw "_rtp->SetTMMBRStatus";
+ exit(1);
+ }
+
+ if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0)
+ {
+ throw "_rtp->RegisterReceivePayload";
+ exit(1);
+ }
+
+ _isReceiver = true;
+
+ return (0);
+}
+
+
+WebRtc_Word32 TestSenderReceiver::Start()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _eventPtr = EventWrapper::Create();
+
+ if (_rtp->SetSendingStatus(true) != 0)
+ {
+ throw "_rtp->SetSendingStatus";
+ exit(1);
+ }
+
+ _procThread = ThreadWrapper::CreateThread(ProcThreadFunction, this, kRealtimePriority, "TestSenderReceiver");
+ if (_procThread == NULL)
+ {
+ throw "Unable to create process thread";
+ exit(1);
+ }
+
+ _running = true;
+
+ if (_isReceiver)
+ {
+ if (_transport->StartReceiving(NR_OF_SOCKET_BUFFERS) != 0)
+ {
+ throw "_transport->StartReceiving";
+ exit(1);
+ }
+ }
+
+ unsigned int tid;
+ _procThread->Start(tid);
+
+ return 0;
+
+}
+
+
+WebRtc_Word32 TestSenderReceiver::Stop ()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _transport->StopReceiving();
+
+ if (_procThread)
+ {
+ _procThread->SetNotAlive();
+ _running = false;
+ _eventPtr->Set();
+
+ while (!_procThread->Stop())
+ {
+ ;
+ }
+
+ delete _eventPtr;
+
+ delete _procThread;
+ }
+
+ _procThread = NULL;
+
+ return (0);
+}
+
+
+bool TestSenderReceiver::ProcLoop(void)
+{
+
+ // process RTP/RTCP module
+ _rtp->Process();
+
+ // process SocketTransport module
+ _transport->Process();
+
+ // no critSect
+ while (_running)
+ {
+ // ask RTP/RTCP module for wait time
+ WebRtc_Word32 rtpWait = _rtp->TimeUntilNextProcess();
+
+ // ask SocketTransport module for wait time
+ WebRtc_Word32 tpWait = _transport->TimeUntilNextProcess();
+
+ WebRtc_Word32 minWait = (rtpWait < tpWait) ? rtpWait: tpWait;
+ minWait = (minWait > 0) ? minWait : 0;
+ // wait
+ _eventPtr->Wait(minWait);
+
+ // process RTP/RTCP module
+ _rtp->Process();
+
+ // process SocketTransport module
+ _transport->Process();
+
+ }
+
+ return true;
+}
+
+
+WebRtc_Word32 TestSenderReceiver::ReceiveBitrateKbps ()
+{
+ WebRtc_UWord32 bytesSent;
+ WebRtc_UWord32 packetsSent;
+ WebRtc_UWord32 bytesReceived;
+ WebRtc_UWord32 packetsReceived;
+
+ if (_rtp->DataCountersRTP(&bytesSent, &packetsSent, &bytesReceived, &packetsReceived) == 0)
+ {
+ WebRtc_Word64 now = TickTime::MillisecondTimestamp();
+ WebRtc_Word32 kbps = 0;
+ if (now > _lastTime)
+ {
+ if (_lastTime > 0)
+ {
+ // 8 * bytes / ms = kbps
+ kbps = static_cast<WebRtc_Word32>(
+ (8 * (bytesReceived - _lastBytesReceived)) / (now - _lastTime));
+ }
+ _lastTime = now;
+ _lastBytesReceived = bytesReceived;
+ }
+ return (kbps);
+ }
+
+ return (-1);
+}
+
+
+WebRtc_Word32 TestSenderReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
+{
+ return (_rtp->SetPacketTimeout(timeoutMS, 0 /* RTCP timeout */));
+}
+
+
+void TestSenderReceiver::OnPacketTimeout(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ _timeOut = true;
+}
+
+
+void TestSenderReceiver::OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType)
+{
+ // do nothing
+ //printf("OnReceivedPacket\n");
+
+}
+
+WebRtc_Word32 TestSenderReceiver::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader)
+{
+ //printf("OnReceivedPayloadData\n");
+ return (0);
+}
+
+
+void TestSenderReceiver::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ _rtp->IncomingPacket((WebRtc_UWord8 *) incomingRtpPacket, static_cast<WebRtc_UWord16>(rtpPacketLength));
+}
+
+
+
+void TestSenderReceiver::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ _rtp->IncomingPacket((WebRtc_UWord8 *) incomingRtcpPacket, static_cast<WebRtc_UWord16>(rtcpPacketLength));
+}
+
+
+
+
+
+///////////////////
+
+
+WebRtc_Word32 TestSenderReceiver::InitSender (const WebRtc_UWord32 startBitrateKbps,
+ const WebRtc_Word8* ipAddr,
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort /*= 0*/,
+ const WebRtc_Word8 payloadType /*= 127*/)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _payloadType = payloadType;
+
+ // check load generator valid
+ if (_loadGenerator)
+ {
+ _loadGenerator->SetBitrate(startBitrateKbps);
+ }
+
+ if (_rtp->RegisterSendTransport(_transport) != 0)
+ {
+ throw "_rtp->RegisterSendTransport";
+ exit(1);
+ }
+ if (_rtp->RegisterSendPayload("I420", _payloadType, 90000) != 0)
+ {
+ throw "_rtp->RegisterSendPayload";
+ exit(1);
+ }
+
+ if (_rtp->RegisterIncomingVideoCallback(this) != 0)
+ {
+ throw "_rtp->RegisterIncomingVideoCallback";
+ exit(1);
+ }
+
+ if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
+ {
+ throw "_rtp->SetRTCPStatus";
+ exit(1);
+ }
+
+ if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
+ {
+ throw "_rtp->SetSendBitrate";
+ exit(1);
+ }
+
+
+ // SocketTransport
+ if (_transport->InitializeSendSockets(ipAddr, rtpPort, rtcpPort))
+ {
+ throw "_transport->InitializeSendSockets";
+ exit(1);
+ }
+
+ _isSender = true;
+
+ return (0);
+}
+
+
+
+WebRtc_Word32
+TestSenderReceiver::SendOutgoingData(const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
+{
+ return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
+}
+
+
+WebRtc_Word32 TestSenderReceiver::SetLoadGenerator(TestLoadGenerator *generator)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _loadGenerator = generator;
+ return(0);
+
+}
+
+void TestSenderReceiver::OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 minBitrateBps,
+ const WebRtc_UWord32 maxBitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs,
+ const WebRtc_UWord16 bwEstimateKbitMin,
+ const WebRtc_UWord16 bwEstimateKbitMax)
+{
+ if (_loadGenerator)
+ {
+ _loadGenerator->SetBitrate(maxBitrateBps/1000);
+ }
+
+ if (_sendRecCB)
+ {
+ _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
+ fractionLost,
+ roundTripTimeMs,
+ bwEstimateKbitMin,
+ bwEstimateKbitMax);
+ }
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
new file mode 100644
index 0000000..7f7f2f0
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
@@ -0,0 +1,166 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+#include "udp_transport.h"
+
+class TestLoadGenerator;
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+}
+
+using namespace webrtc;
+
+#define MAX_BITRATE_KBPS 50000
+
+
+class SendRecCB
+{
+public:
+ virtual void OnOnNetworkChanged(const WebRtc_UWord32 bitrateTarget,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs,
+ const WebRtc_UWord16 bwEstimateKbitMin,
+ const WebRtc_UWord16 bwEstimateKbitMax) = 0;
+
+ virtual ~SendRecCB() {};
+};
+
+
+class TestSenderReceiver : public RtpFeedback, public RtpData, public UdpTransportData, public RtpVideoFeedback
+{
+
+public:
+ TestSenderReceiver (void);
+
+ ~TestSenderReceiver (void);
+
+ void SetCallback (SendRecCB *cb) { _sendRecCB = cb; };
+
+ WebRtc_Word32 Start();
+
+ WebRtc_Word32 Stop();
+
+ bool ProcLoop();
+
+ /////////////////////////////////////////////
+ // Receiver methods
+
+ WebRtc_Word32 InitReceiver (const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0,
+ const WebRtc_Word8 payloadType = 127);
+
+ WebRtc_Word32 ReceiveBitrateKbps ();
+
+ WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
+
+ bool timeOutTriggered () { return (_timeOut); };
+
+ // Inherited from RtpFeedback
+ virtual WebRtc_Word32 OnInitializeDecoder(const WebRtc_Word32 id,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) { return(0);};
+
+ virtual void OnPacketTimeout(const WebRtc_Word32 id);
+
+ virtual void OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType);
+
+ virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive) {};
+
+ virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC) {};
+
+ virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added) {};
+
+
+ // Inherited from RtpData
+
+ virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader);
+
+
+ // Inherited from UdpTransportData
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+
+
+ /////////////////////////////////
+ // Sender methods
+
+ WebRtc_Word32 InitSender (const WebRtc_UWord32 startBitrateKbps,
+ const WebRtc_Word8* ipAddr,
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0,
+ const WebRtc_Word8 payloadType = 127);
+
+ WebRtc_Word32 SendOutgoingData(const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
+
+ WebRtc_Word32 SetLoadGenerator(TestLoadGenerator *generator);
+
+ WebRtc_UWord32 BitrateSent() { return (_rtp->BitrateSent()); };
+
+
+ // Inherited from RtpVideoFeedback
+ virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+ const WebRtc_UWord8 message = 0) {};
+
+ virtual void OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 minBitrateBps,
+ const WebRtc_UWord32 maxBitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs,
+ const WebRtc_UWord16 bwEstimateKbitMin,
+ const WebRtc_UWord16 bwEstimateKbitMax);
+
+private:
+ RtpRtcp* _rtp;
+ UdpTransport* _transport;
+ webrtc::CriticalSectionWrapper* _critSect;
+ webrtc::EventWrapper *_eventPtr;
+ webrtc::ThreadWrapper* _procThread;
+ bool _running;
+ WebRtc_Word8 _payloadType;
+ TestLoadGenerator* _loadGenerator;
+ bool _isSender;
+ bool _isReceiver;
+ bool _timeOut;
+ SendRecCB * _sendRecCB;
+ WebRtc_UWord32 _lastBytesReceived;
+ WebRtc_Word64 _lastTime;
+
+};
+
+#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc b/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc
new file mode 100644
index 0000000..38b6e15
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc
@@ -0,0 +1,538 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../../source/BitstreamBuilder.h"
+#include "../../source/BitstreamParser.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <math.h>
+#include <tchar.h>
+#include <windows.h>
+
+WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
+{
+ return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
+}
+
+WebRtc_UWord16 BitRateBPSInv(WebRtc_UWord32 x )
+{
+ // 16383 0x3fff
+ // 1 638 300 exp 0
+ // 16 383 000 exp 1
+ // 163 830 000 exp 2
+ // 1 638 300 000 exp 3
+ const float exp = log10(float(x>>14)) - 2;
+ if(exp < 0.0)
+ {
+ return WebRtc_UWord16(x /100);
+ }else if(exp < 1.0)
+ {
+ return 0x4000 + WebRtc_UWord16(x /1000);
+ }else if(exp < 2.0)
+ {
+ return 0x8000 + WebRtc_UWord16(x /10000);
+ }else if(exp < 3.0)
+ {
+ return 0xC000 + WebRtc_UWord16(x /100000);
+ } else
+ {
+ assert(false);
+ return 0;
+ }
+}
+
+
+int _tmain(int argc, _TCHAR* argv[])
+{
+ WebRtc_UWord8 dataBuffer[128];
+ BitstreamBuilder builder(dataBuffer, sizeof(dataBuffer));
+
+ // test 1 to 4 bits
+ builder.Add1Bit(1);
+ builder.Add1Bit(0);
+ builder.Add1Bit(1);
+
+ builder.Add2Bits(1);
+ builder.Add2Bits(2);
+ builder.Add2Bits(3);
+
+ builder.Add3Bits(1);
+ builder.Add3Bits(3);
+ builder.Add3Bits(7);
+
+ builder.Add4Bits(1);
+ builder.Add4Bits(5);
+ builder.Add4Bits(15);
+
+ assert(4 == builder.Length());
+
+ BitstreamParser parser(dataBuffer, sizeof(dataBuffer));
+
+ assert(1 == parser.Get1Bit());
+ assert(0 == parser.Get1Bit());
+ assert(1 == parser.Get1Bit());
+
+ assert(1 == parser.Get2Bits());
+ assert(2 == parser.Get2Bits());
+ assert(3 == parser.Get2Bits());
+
+ assert(1 == parser.Get3Bits());
+ assert(3 == parser.Get3Bits());
+ assert(7 == parser.Get3Bits());
+
+ assert(1 == parser.Get4Bits());
+ assert(5 == parser.Get4Bits());
+ assert(15 == parser.Get4Bits());
+
+ printf("Test of 1 to 4 bits done\n");
+
+ // test 5 to 7 bits
+ builder.Add5Bits(1);
+ builder.Add5Bits(15);
+ builder.Add5Bits(30);
+
+ builder.Add6Bits(1);
+ builder.Add6Bits(30);
+ builder.Add6Bits(60);
+
+ builder.Add7Bits(1);
+ builder.Add7Bits(60);
+ builder.Add7Bits(120);
+
+ assert(1 == parser.Get5Bits());
+ assert(15 == parser.Get5Bits());
+ assert(30 == parser.Get5Bits());
+
+ assert(1 == parser.Get6Bits());
+ assert(30 == parser.Get6Bits());
+ assert(60 == parser.Get6Bits());
+
+ assert(1 == parser.Get7Bits());
+ assert(60 == parser.Get7Bits());
+ assert(120 == parser.Get7Bits());
+
+ printf("Test of 5 to 7 bits done\n");
+
+ builder.Add8Bits(1);
+ builder.Add1Bit(1);
+ builder.Add8Bits(255);
+ builder.Add1Bit(0);
+ builder.Add8Bits(127);
+ builder.Add1Bit(1);
+ builder.Add8Bits(60);
+ builder.Add1Bit(0);
+ builder.Add8Bits(30);
+ builder.Add1Bit(1);
+ builder.Add8Bits(120);
+ builder.Add1Bit(0);
+ builder.Add8Bits(160);
+ builder.Add1Bit(1);
+ builder.Add8Bits(180);
+
+ assert(1 == parser.Get8Bits());
+ assert(1 == parser.Get1Bit());
+ assert(255 == parser.Get8Bits());
+ assert(0 == parser.Get1Bit());
+ assert(127 == parser.Get8Bits());
+ assert(1 == parser.Get1Bit());
+ assert(60 == parser.Get8Bits());
+ assert(0 == parser.Get1Bit());
+ assert(30 == parser.Get8Bits());
+ assert(1 == parser.Get1Bit());
+ assert(120 == parser.Get8Bits());
+ assert(0 == parser.Get1Bit());
+ assert(160 == parser.Get8Bits());
+ assert(1 == parser.Get1Bit());
+ assert(180 == parser.Get8Bits());
+
+ printf("Test of 8 bits done\n");
+
+ builder.Add16Bits(1);
+ builder.Add1Bit(1);
+ builder.Add16Bits(255);
+ builder.Add1Bit(0);
+ builder.Add16Bits(12756);
+ builder.Add1Bit(1);
+ builder.Add16Bits(60);
+ builder.Add1Bit(0);
+ builder.Add16Bits(30);
+ builder.Add1Bit(1);
+ builder.Add16Bits(30120);
+ builder.Add1Bit(0);
+ builder.Add16Bits(160);
+ builder.Add1Bit(1);
+ builder.Add16Bits(180);
+
+ assert(1 == parser.Get16Bits());
+ assert(1 == parser.Get1Bit());
+ assert(255 == parser.Get16Bits());
+ assert(0 == parser.Get1Bit());
+ assert(12756 == parser.Get16Bits());
+ assert(1 == parser.Get1Bit());
+ assert(60 == parser.Get16Bits());
+ assert(0 == parser.Get1Bit());
+ assert(30 == parser.Get16Bits());
+ assert(1 == parser.Get1Bit());
+ assert(30120 == parser.Get16Bits());
+ assert(0 == parser.Get1Bit());
+ assert(160 == parser.Get16Bits());
+ assert(1 == parser.Get1Bit());
+ assert(180 == parser.Get16Bits());
+
+ printf("Test of 16 bits done\n");
+
+ builder.Add24Bits(1);
+ builder.Add1Bit(1);
+ builder.Add24Bits(255);
+ builder.Add1Bit(0);
+ builder.Add24Bits(12756);
+ builder.Add1Bit(1);
+ builder.Add24Bits(60);
+ builder.Add1Bit(0);
+ builder.Add24Bits(303333);
+ builder.Add1Bit(1);
+ builder.Add24Bits(30120);
+ builder.Add1Bit(0);
+ builder.Add24Bits(160);
+ builder.Add1Bit(1);
+ builder.Add24Bits(8018018);
+
+ assert(1 == parser.Get24Bits());
+ assert(1 == parser.Get1Bit());
+ assert(255 == parser.Get24Bits());
+ assert(0 == parser.Get1Bit());
+ assert(12756 == parser.Get24Bits());
+ assert(1 == parser.Get1Bit());
+ assert(60 == parser.Get24Bits());
+ assert(0 == parser.Get1Bit());
+ assert(303333 == parser.Get24Bits());
+ assert(1 == parser.Get1Bit());
+ assert(30120 == parser.Get24Bits());
+ assert(0 == parser.Get1Bit());
+ assert(160 == parser.Get24Bits());
+ assert(1 == parser.Get1Bit());
+ assert(8018018 == parser.Get24Bits());
+
+ printf("Test of 24 bits done\n");
+
+ builder.Add32Bits(1);
+ builder.Add1Bit(1);
+ builder.Add32Bits(255);
+ builder.Add1Bit(0);
+ builder.Add32Bits(12756);
+ builder.Add1Bit(1);
+ builder.Add32Bits(60);
+ builder.Add1Bit(0);
+ builder.Add32Bits(303333);
+ builder.Add1Bit(1);
+ builder.Add32Bits(3012000012);
+ builder.Add1Bit(0);
+ builder.Add32Bits(1601601601);
+ builder.Add1Bit(1);
+ builder.Add32Bits(8018018);
+
+ assert(1 == parser.Get32Bits());
+ assert(1 == parser.Get1Bit());
+ assert(255 == parser.Get32Bits());
+ assert(0 == parser.Get1Bit());
+ assert(12756 == parser.Get32Bits());
+ assert(1 == parser.Get1Bit());
+ assert(60 == parser.Get32Bits());
+ assert(0 == parser.Get1Bit());
+ assert(303333 == parser.Get32Bits());
+ assert(1 == parser.Get1Bit());
+ assert(3012000012 == parser.Get32Bits());
+ assert(0 == parser.Get1Bit());
+ assert(1601601601 == parser.Get32Bits());
+ assert(1 == parser.Get1Bit());
+ assert(8018018 == parser.Get32Bits());
+
+ printf("Test of 32 bits done\n");
+
+ builder.AddUE(1);
+ builder.AddUE(4);
+ builder.AddUE(9809706);
+ builder.AddUE(2);
+ builder.AddUE(15);
+ builder.AddUE(16998);
+
+ assert( 106 == builder.Length());
+
+ assert(1 == parser.GetUE());
+ assert(4 == parser.GetUE());
+ assert(9809706 == parser.GetUE());
+ assert(2 == parser.GetUE());
+ assert(15 == parser.GetUE());
+ assert(16998 == parser.GetUE());
+
+ printf("Test UE bits done\n");
+
+ BitstreamBuilder builderScalabilityInfo(dataBuffer, sizeof(dataBuffer));
+ BitstreamParser parserScalabilityInfo(dataBuffer, sizeof(dataBuffer));
+
+ const WebRtc_UWord8 numberOfLayers = 4;
+ const WebRtc_UWord8 layerId[numberOfLayers] = {0,1,2,3};
+ const WebRtc_UWord8 priorityId[numberOfLayers] = {0,1,2,3};
+ const WebRtc_UWord8 discardableId[numberOfLayers] = {0,1,1,1};
+
+ const WebRtc_UWord8 dependencyId[numberOfLayers]= {0,1,1,1};
+ const WebRtc_UWord8 qualityId[numberOfLayers]= {0,0,0,1};
+ const WebRtc_UWord8 temporalId[numberOfLayers]= {0,0,1,1};
+
+ const WebRtc_UWord16 avgBitrate[numberOfLayers]= {BitRateBPSInv(100000),
+ BitRateBPSInv(200000),
+ BitRateBPSInv(400000),
+ BitRateBPSInv(800000)};
+
+ // todo which one is the sum?
+ const WebRtc_UWord16 maxBitrateLayer[numberOfLayers]= {BitRateBPSInv(150000),
+ BitRateBPSInv(300000),
+ BitRateBPSInv(500000),
+ BitRateBPSInv(900000)};
+
+ const WebRtc_UWord16 maxBitrateLayerRepresentation[numberOfLayers] = {BitRateBPSInv(150000),
+ BitRateBPSInv(450000),
+ BitRateBPSInv(950000),
+ BitRateBPSInv(1850000)};
+
+ assert( 16300 == BitRateBPS(BitRateBPSInv(16383)));
+ assert( 163800 == BitRateBPS(BitRateBPSInv(163830)));
+ assert( 1638300 == BitRateBPS(BitRateBPSInv(1638300)));
+ assert( 1638000 == BitRateBPS(BitRateBPSInv(1638400)));
+
+ assert( 18500 == BitRateBPS(BitRateBPSInv(18500)));
+ assert( 185000 == BitRateBPS(BitRateBPSInv(185000)));
+ assert( 1850000 == BitRateBPS(BitRateBPSInv(1850000)));
+ assert( 18500000 == BitRateBPS(BitRateBPSInv(18500000)));
+ assert( 185000000 == BitRateBPS(BitRateBPSInv(185000000)));
+
+ const WebRtc_UWord16 maxBitrareCalcWindow[numberOfLayers] = {200, 200,200,200};// in 1/100 of second
+
+ builderScalabilityInfo.Add1Bit(0); // temporal_id_nesting_flag
+ builderScalabilityInfo.Add1Bit(0); // priority_layer_info_present_flag
+ builderScalabilityInfo.Add1Bit(0); // priority_id_setting_flag
+
+ builderScalabilityInfo.AddUE(numberOfLayers-1);
+
+ for(int i = 0; i<= numberOfLayers-1; i++)
+ {
+ builderScalabilityInfo.AddUE(layerId[i]);
+ builderScalabilityInfo.Add6Bits(priorityId[i]);
+ builderScalabilityInfo.Add1Bit(discardableId[i]);
+ builderScalabilityInfo.Add3Bits(dependencyId[i]);
+ builderScalabilityInfo.Add4Bits(qualityId[i]);
+ builderScalabilityInfo.Add3Bits(temporalId[i]);
+
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+
+ builderScalabilityInfo.Add1Bit(1); // bitrate_info_present_flag
+
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+ builderScalabilityInfo.Add1Bit(0);
+
+ builderScalabilityInfo.Add16Bits(avgBitrate[i]);
+ builderScalabilityInfo.Add16Bits(maxBitrateLayer[i]);
+ builderScalabilityInfo.Add16Bits(maxBitrateLayerRepresentation[i]);
+ builderScalabilityInfo.Add16Bits(maxBitrareCalcWindow[i]);
+
+ builderScalabilityInfo.AddUE(0); // layer_dependency_info_src_layer_id_delta
+ builderScalabilityInfo.AddUE(0); // parameter_sets_info_src_layer_id_delta
+ }
+
+ printf("Test builderScalabilityInfo done\n");
+
+ // Scalability Info parser
+ parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+ const WebRtc_UWord8 priority_layer_info_present = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 priority_id_setting_flag = parserScalabilityInfo.Get1Bit();
+
+ WebRtc_UWord32 numberOfLayersMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 j = 0; j<= numberOfLayersMinusOne; j++)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.Get6Bits();
+ parserScalabilityInfo.Get1Bit();
+ parserScalabilityInfo.Get3Bits();
+ parserScalabilityInfo.Get4Bits();
+ parserScalabilityInfo.Get3Bits();
+
+ const WebRtc_UWord8 sub_pic_layer_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 sub_region_layer_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 iroi_division_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 profile_level_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 bitrate_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 frm_rate_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 frm_size_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 layer_dependency_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 parameter_sets_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 bitstream_restriction_info_present_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 exact_inter_layer_pred_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+
+ if(sub_pic_layer_flag || iroi_division_info_present_flag)
+ {
+ parserScalabilityInfo.Get1Bit();
+ }
+ const WebRtc_UWord8 layer_conversion_flag = parserScalabilityInfo.Get1Bit();
+ const WebRtc_UWord8 layer_output_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
+
+ if(profile_level_info_present_flag)
+ {
+ parserScalabilityInfo.Get24Bits();
+ }
+ if(bitrate_info_present_flag)
+ {
+ // this is what we want
+ assert(avgBitrate[j] == parserScalabilityInfo.Get16Bits());
+ assert(maxBitrateLayer[j] == parserScalabilityInfo.Get16Bits());
+ assert(maxBitrateLayerRepresentation[j] == parserScalabilityInfo.Get16Bits());
+ assert(maxBitrareCalcWindow[j] == parserScalabilityInfo.Get16Bits());
+ }else
+ {
+ assert(false);
+ }
+ if(frm_rate_info_present_flag)
+ {
+ parserScalabilityInfo.Get2Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ if(frm_size_info_present_flag || iroi_division_info_present_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }
+ if(sub_region_layer_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ if(sub_pic_layer_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(iroi_division_info_present_flag)
+ {
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }else
+ {
+ const WebRtc_UWord32 numRoisMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= numRoisMinusOne; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }
+ }
+ }
+ if(layer_dependency_info_present_flag)
+ {
+ const WebRtc_UWord32 numDirectlyDependentLayers = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k < numDirectlyDependentLayers; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ } else
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(parameter_sets_info_present_flag)
+ {
+ const WebRtc_UWord32 numSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= numSeqParameterSetMinusOne; k++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ const WebRtc_UWord32 numSubsetSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 l = 0; l <= numSubsetSeqParameterSetMinusOne; l++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ const WebRtc_UWord32 numPicParameterSetMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 m = 0; m <= numPicParameterSetMinusOne; m++)
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ }else
+ {
+ parserScalabilityInfo.GetUE();
+ }
+ if(bitstream_restriction_info_present_flag)
+ {
+ parserScalabilityInfo.Get1Bit();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.GetUE();
+ }
+ if(layer_conversion_flag)
+ {
+ parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <2;k++)
+ {
+ if(parserScalabilityInfo.Get1Bit())
+ {
+ parserScalabilityInfo.Get24Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ }
+ }
+ if(priority_layer_info_present)
+ {
+ const WebRtc_UWord32 prNumDidMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 k = 0; k <= prNumDidMinusOne;k++)
+ {
+ parserScalabilityInfo.Get3Bits();
+ const WebRtc_UWord32 prNumMinusOne = parserScalabilityInfo.GetUE();
+ for(WebRtc_UWord32 l = 0; l <= prNumMinusOne; l++)
+ {
+ parserScalabilityInfo.GetUE();
+ parserScalabilityInfo.Get24Bits();
+ parserScalabilityInfo.Get16Bits();
+ parserScalabilityInfo.Get16Bits();
+ }
+ }
+ }
+ if(priority_id_setting_flag)
+ {
+ WebRtc_UWord8 priorityIdSettingUri;
+ WebRtc_UWord32 priorityIdSettingUriIdx = 0;
+ do
+ {
+ priorityIdSettingUri = parserScalabilityInfo.Get8Bits();
+ } while (priorityIdSettingUri != 0);
+ }
+ printf("Test parserScalabilityInfo done\n");
+
+ printf("\nAPI test of parser for ScalabilityInfo done\n");
+
+ ::Sleep(5000);
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi b/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi
new file mode 100644
index 0000000..36a50de
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi
@@ -0,0 +1,111 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'bwe_standalone',
+ 'type': 'executable',
+ 'dependencies': [
+ 'matlab_plotting',
+ 'rtp_rtcp',
+ 'udp_transport',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ 'sources': [
+ 'BWEStandAlone/BWEStandAlone.cc',
+ 'BWEStandAlone/TestLoadGenerator.cc',
+ 'BWEStandAlone/TestLoadGenerator.h',
+ 'BWEStandAlone/TestSenderReceiver.cc',
+ 'BWEStandAlone/TestSenderReceiver.h',
+ ], # source
+ 'conditions': [
+ ['OS=="linux"', {
+ 'cflags': [
+ '-fexceptions', # enable exceptions
+ ],
+ },
+ ],
+ ],
+
+ 'include_dirs': [
+ ],
+ 'link_settings': {
+ },
+ },
+
+ {
+ 'target_name': 'matlab_plotting',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'matlab_plotting_include',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '/opt/matlab2010a/extern/include',
+ ],
+ # 'direct_dependent_settings': {
+ # 'defines': [
+ # 'MATLAB',
+ # ],
+ # 'include_dirs': [
+ # 'BWEStandAlone',
+ # ],
+ # },
+ 'export_dependent_settings': [
+ 'matlab_plotting_include',
+ ],
+ 'sources': [
+ 'BWEStandAlone/MatlabPlot.cc',
+ 'BWEStandAlone/MatlabPlot.h',
+ ],
+ 'link_settings': {
+ 'ldflags' : [
+ '-L/opt/matlab2010a/bin/glnxa64',
+ '-leng',
+ '-lmx',
+ '-Wl,-rpath,/opt/matlab2010a/bin/glnxa64',
+ ],
+ },
+ 'defines': [
+ 'MATLAB',
+ ],
+ 'conditions': [
+ ['OS=="linux"', {
+ 'cflags': [
+ '-fexceptions', # enable exceptions
+ ],
+ },
+ ],
+ ],
+ },
+
+ {
+ 'target_name': 'matlab_plotting_include',
+ 'type': 'none',
+ 'direct_dependent_settings': {
+ 'defines': [
+# 'MATLAB',
+ ],
+ 'include_dirs': [
+ 'BWEStandAlone',
+ ],
+ },
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc
new file mode 100644
index 0000000..3b5390e
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+#include <gtest/gtest.h>
+
+#include "test_api.h"
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+using namespace webrtc;
+
+class RtpRtcpAPITest : public ::testing::Test {
+ protected:
+ RtpRtcpAPITest() {
+ test_CSRC[0] = 1234;
+ test_CSRC[2] = 2345;
+ test_id = 123;
+ test_ssrc = 3456;
+ test_timestamp = 4567;
+ test_sequence_number = 2345;
+ }
+ ~RtpRtcpAPITest() {}
+
+ virtual void SetUp() {
+ module = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
+ EXPECT_EQ(0, module->InitReceiver());
+ EXPECT_EQ(0, module->InitSender());
+ }
+
+ virtual void TearDown() {
+ RtpRtcp::DestroyRtpRtcp(module);
+ }
+
+ int test_id;
+ RtpRtcp* module;
+ WebRtc_UWord32 test_ssrc;
+ WebRtc_UWord32 test_timestamp;
+ WebRtc_UWord16 test_sequence_number;
+ WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
+ FakeRtpRtcpClock fake_clock;
+};
+
+TEST_F(RtpRtcpAPITest, Basic) {
+ EXPECT_EQ(0, module->SetSequenceNumber(test_sequence_number));
+ EXPECT_EQ(test_sequence_number, module->SequenceNumber());
+
+ EXPECT_EQ(0, module->SetStartTimestamp(test_timestamp));
+ EXPECT_EQ(test_timestamp, module->StartTimestamp());
+
+ EXPECT_FALSE(module->Sending());
+ EXPECT_EQ(0, module->SetSendingStatus(true));
+ EXPECT_TRUE(module->Sending());
+}
+
+TEST_F(RtpRtcpAPITest, MTU) {
+ EXPECT_EQ(-1, module->SetMaxTransferUnit(10));
+ EXPECT_EQ(-1, module->SetMaxTransferUnit(IP_PACKET_SIZE + 1));
+ EXPECT_EQ(0, module->SetMaxTransferUnit(1234));
+ EXPECT_EQ(1234-20-8, module->MaxPayloadLength());
+
+ EXPECT_EQ(0, module->SetTransportOverhead(true, true, 12));
+ EXPECT_EQ(1234 - 20- 20 -20 - 12, module->MaxPayloadLength());
+
+ EXPECT_EQ(0, module->SetTransportOverhead(false, false, 0));
+ EXPECT_EQ(1234 - 20 - 8, module->MaxPayloadLength());
+}
+
+TEST_F(RtpRtcpAPITest, SSRC) {
+ EXPECT_EQ(0, module->SetSSRC(test_ssrc));
+ EXPECT_EQ(test_ssrc, module->SSRC());
+}
+
+TEST_F(RtpRtcpAPITest, CSRC) {
+ EXPECT_EQ(0, module->SetCSRCs(test_CSRC, 2));
+ WebRtc_UWord32 testOfCSRC[webrtc::kRtpCsrcSize];
+ EXPECT_EQ(2, module->CSRCs(testOfCSRC));
+ EXPECT_EQ(test_CSRC[0], testOfCSRC[0]);
+ EXPECT_EQ(test_CSRC[1], testOfCSRC[1]);
+}
+
+TEST_F(RtpRtcpAPITest, TrafficSmoothing) {
+ EXPECT_FALSE(module->TransmissionSmoothingStatus());
+ module->SetTransmissionSmoothingStatus(true);
+ EXPECT_TRUE(module->TransmissionSmoothingStatus());
+}
+
+TEST_F(RtpRtcpAPITest, RTCP) {
+ EXPECT_EQ(kRtcpOff, module->RTCP());
+ EXPECT_EQ(0, module->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(kRtcpCompound, module->RTCP());
+
+ EXPECT_EQ(0, module->SetCNAME("john.doe@test.test"));
+ EXPECT_EQ(-1, module->SetCNAME(NULL));
+
+ char cName[RTCP_CNAME_SIZE];
+ EXPECT_EQ(0, module->CNAME(cName));
+ EXPECT_STRCASEEQ(cName, "john.doe@test.test");
+ EXPECT_EQ(-1, module->CNAME(NULL));
+
+ EXPECT_FALSE(module->TMMBR());
+ EXPECT_EQ(0, module->SetTMMBRStatus(true));
+ EXPECT_TRUE(module->TMMBR());
+ EXPECT_EQ(0, module->SetTMMBRStatus(false));
+ EXPECT_FALSE(module->TMMBR());
+
+ EXPECT_EQ(kNackOff, module->NACK());
+ EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp));
+ EXPECT_EQ(kNackRtcp, module->NACK());
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi
new file mode 100644
index 0000000..d47e6d4
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi
@@ -0,0 +1,42 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'test_rtp_rtcp_api',
+ 'type': 'executable',
+ 'dependencies': [
+ 'rtp_rtcp',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+
+ 'include_dirs': [
+ '../../interface',
+ '../../source',
+ '../../../../system_wrappers/interface',
+ ],
+
+ 'sources': [
+ 'test_api.cc',
+ 'test_api_audio.cc',
+ 'test_api_nack.cc',
+ 'test_api_rtcp.cc',
+ 'test_api_video.cc',
+ ],
+
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h
new file mode 100644
index 0000000..c4ff916
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class FakeRtpRtcpClock : public RtpRtcpClock {
+ public:
+ FakeRtpRtcpClock() {
+ time_in_ms_ = 123456;
+ }
+ // Return a timestamp in milliseconds relative to some arbitrary
+ // source; the source is fixed for this clock.
+ virtual WebRtc_UWord32 GetTimeInMS() {
+ return time_in_ms_;
+ }
+ // Retrieve an NTP absolute timestamp.
+ virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
+ secs = time_in_ms_ / 1000;
+ frac = (time_in_ms_ % 1000) * 4294967;
+ }
+ void IncrementTime(WebRtc_UWord32 time_increment_ms) {
+ time_in_ms_ += time_increment_ms;
+ }
+ private:
+ WebRtc_UWord32 time_in_ms_;
+};
+
+// This class sends all its packet straight to the provided RtpRtcp module.
+// with optional packet loss.
+class LoopBackTransport : public webrtc::Transport {
+ public:
+ LoopBackTransport(RtpRtcp* rtpRtcpModule)
+ : _count(0),
+ _packetLoss(0),
+ _rtpRtcpModule(rtpRtcpModule) {
+ }
+ void DropEveryNthPacket(int n) {
+ _packetLoss = n;
+ }
+ virtual int SendPacket(int channel, const void *data, int len) {
+ _count++;
+ if (_packetLoss > 0) {
+ if ((_count % _packetLoss) == 0) {
+ return len;
+ }
+ }
+ if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+ return len;
+ }
+ return -1;
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len) {
+ if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+ return len;
+ }
+ return -1;
+ }
+ private:
+ int _count;
+ int _packetLoss;
+ RtpRtcp* _rtpRtcpModule;
+};
+
+class RtpReceiver : public RtpData {
+ public:
+ virtual WebRtc_Word32 OnReceivedPayloadData(
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader) {
+ return 0;
+ }
+};
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
new file mode 100644
index 0000000..d066864
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -0,0 +1,446 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+#include <gtest/gtest.h>
+
+#include "test_api.h"
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+using namespace webrtc;
+
+#define test_rate 64000u
+
+class VerifyingAudioReceiver : public RtpData {
+ public:
+ VerifyingAudioReceiver(RtpRtcp* rtpRtcpModule) {}
+
+ virtual WebRtc_Word32 OnReceivedPayloadData(
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader) {
+ if (rtpHeader->header.payloadType == 98 ||
+ rtpHeader->header.payloadType == 99) {
+ EXPECT_EQ(4, payloadSize);
+ char str[5];
+ memcpy(str, payloadData, payloadSize);
+ str[4] = 0;
+ // All our test vectors for payload type 96 and 97 even the stereo is on
+ // a per channel base equal to the 4 chars "test".
+ // Note there is no null termination so we add that to use the
+ // test EXPECT_STRCASEEQ.
+ EXPECT_STRCASEEQ("test", str);
+ return 0;
+ }
+ if (rtpHeader->header.payloadType == 100 ||
+ rtpHeader->header.payloadType == 101 ||
+ rtpHeader->header.payloadType == 102) {
+ if (rtpHeader->type.Audio.channel == 1) {
+ if (payloadData[0] == 0xff) {
+ // All our test vectors for payload type 100, 101 and 102 have the
+ // first channel data being equal to 0xff.
+ return 0;
+ }
+ } else if (rtpHeader->type.Audio.channel == 2) {
+ if (payloadData[0] == 0x0) {
+ // All our test vectors for payload type 100, 101 and 102 have the
+ // second channel data being equal to 0x00.
+ return 0;
+ }
+ } else if (rtpHeader->type.Audio.channel == 3) {
+ // All our test vectors for payload type 100, 101 and 102 have the
+ // third channel data being equal to 0xaa.
+ if (payloadData[0] == 0xaa) {
+ return 0;
+ }
+ }
+ ADD_FAILURE() << "This code path should never happen.";
+ return -1;
+ }
+ return 0;
+ }
+};
+
+class RTPCallback : public RtpFeedback {
+ public:
+ virtual WebRtc_Word32 OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payloadType,
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ if (payloadType == 96) {
+ EXPECT_EQ(test_rate, rate) <<
+ "The rate should be 64K for this payloadType";
+ }
+ return 0;
+ }
+ virtual void OnPacketTimeout(const WebRtc_Word32 id) {
+ }
+ virtual void OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType) {
+ }
+ virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive) {
+ }
+ virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC) {
+ }
+ virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added) {
+ }
+};
+
+class AudioFeedback : public RtpAudioFeedback {
+ virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const bool end) {
+ static WebRtc_UWord8 expectedEvent = 0;
+
+ if (end) {
+ WebRtc_UWord8 oldEvent = expectedEvent-1;
+ if (expectedEvent == 32) {
+ oldEvent = 15;
+ }
+ EXPECT_EQ(oldEvent, event);
+ } else {
+ EXPECT_EQ(expectedEvent, event);
+ expectedEvent++;
+ }
+ if (expectedEvent == 16) {
+ expectedEvent = 32;
+ }
+ }
+ virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const WebRtc_UWord16 lengthMs,
+ const WebRtc_UWord8 volume) {
+ };
+};
+
+class RtpRtcpAudioTest : public ::testing::Test {
+ protected:
+ RtpRtcpAudioTest() {
+ test_CSRC[0] = 1234;
+ test_CSRC[2] = 2345;
+ test_id = 123;
+ test_ssrc = 3456;
+ test_timestamp = 4567;
+ test_sequence_number = 2345;
+ }
+ ~RtpRtcpAudioTest() {}
+
+ virtual void SetUp() {
+ module1 = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
+ module2 = RtpRtcp::CreateRtpRtcp(test_id+1, true, &fake_clock);
+
+ EXPECT_EQ(0, module1->InitReceiver());
+ EXPECT_EQ(0, module1->InitSender());
+ EXPECT_EQ(0, module2->InitReceiver());
+ EXPECT_EQ(0, module2->InitSender());
+ data_receiver1 = new VerifyingAudioReceiver(module1);
+ EXPECT_EQ(0, module1->RegisterIncomingDataCallback(data_receiver1));
+ data_receiver2 = new VerifyingAudioReceiver(module2);
+ EXPECT_EQ(0, module2->RegisterIncomingDataCallback(data_receiver2));
+ transport1 = new LoopBackTransport(module2);
+ EXPECT_EQ(0, module1->RegisterSendTransport(transport1));
+ transport2 = new LoopBackTransport(module1);
+ EXPECT_EQ(0, module2->RegisterSendTransport(transport2));
+ rtp_callback = new RTPCallback();
+ EXPECT_EQ(0, module2->RegisterIncomingRTPCallback(rtp_callback));
+ }
+
+ virtual void TearDown() {
+ RtpRtcp::DestroyRtpRtcp(module1);
+ RtpRtcp::DestroyRtpRtcp(module2);
+ delete transport1;
+ delete transport2;
+ delete data_receiver1;
+ delete data_receiver2;
+ delete rtp_callback;
+ }
+
+ int test_id;
+ RtpRtcp* module1;
+ RtpRtcp* module2;
+ VerifyingAudioReceiver* data_receiver1;
+ VerifyingAudioReceiver* data_receiver2;
+ LoopBackTransport* transport1;
+ LoopBackTransport* transport2;
+ RTPCallback* rtp_callback;
+ WebRtc_UWord32 test_ssrc;
+ WebRtc_UWord32 test_timestamp;
+ WebRtc_UWord16 test_sequence_number;
+ WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
+ FakeRtpRtcpClock fake_clock;
+};
+
+TEST_F(RtpRtcpAudioTest, Basic) {
+ EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
+
+ EXPECT_FALSE(module1->TelephoneEvent());
+
+ // Test detection at the end of a DTMF tone.
+ EXPECT_EQ(0, module2->SetTelephoneEventStatus(true, true, true));
+ EXPECT_EQ(true, module2->TelephoneEvent());
+
+ EXPECT_EQ(0, module1->SetSendingStatus(true));
+
+ // Start basic RTP test.
+
+ // Send an empty RTP packet.
+ // Should fail since we have not registerd the payload type.
+ EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
+ 96, 0, NULL, 0));
+
+ CodecInst voiceCodec;
+ voiceCodec.pltype = 96;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "PCMU", 5);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
+ voiceCodec.rate = test_rate;
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+ printf("4\n");
+
+ const WebRtc_UWord8 test[5] = "test";
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+ 0, test, 4));
+
+ EXPECT_EQ(test_ssrc, module2->RemoteSSRC());
+ EXPECT_EQ(test_timestamp, module2->RemoteTimestamp());
+}
+
+TEST_F(RtpRtcpAudioTest, RED) {
+ CodecInst voiceCodec;
+ voiceCodec.pltype = 96;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "PCMU", 5);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
+ voiceCodec.rate = test_rate;
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
+ EXPECT_EQ(0, module1->SetSendingStatus(true));
+
+ voiceCodec.pltype = 127;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "RED", 4);
+
+ EXPECT_EQ(0, module1->SetSendREDPayloadType(voiceCodec.pltype));
+ WebRtc_Word8 red = 0;
+ EXPECT_EQ(0, module1->SendREDPayloadType(red));
+ EXPECT_EQ(voiceCodec.pltype, red);
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ RTPFragmentationHeader fragmentation;
+ fragmentation.fragmentationVectorSize = 2;
+ fragmentation.fragmentationLength = new WebRtc_UWord32[2];
+ fragmentation.fragmentationLength[0] = 4;
+ fragmentation.fragmentationLength[1] = 4;
+ fragmentation.fragmentationOffset = new WebRtc_UWord32[2];
+ fragmentation.fragmentationOffset[0] = 0;
+ fragmentation.fragmentationOffset[1] = 4;
+ fragmentation.fragmentationTimeDiff = new WebRtc_UWord16[2];
+ fragmentation.fragmentationTimeDiff[0] = 0;
+ fragmentation.fragmentationTimeDiff[1] = 0;
+ fragmentation.fragmentationPlType = new WebRtc_UWord8[2];
+ fragmentation.fragmentationPlType[0] = 96;
+ fragmentation.fragmentationPlType[1] = 96;
+
+ const WebRtc_UWord8 test[5] = "test";
+ // Send a RTP packet.
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
+ 96, 160, test, 4,
+ &fragmentation));
+
+ EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
+ EXPECT_EQ(-1, module1->SendREDPayloadType(red));
+}
+
+TEST_F(RtpRtcpAudioTest, DTMF) {
+ CodecInst voiceCodec;
+ voiceCodec.pltype = 96;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "PCMU", 5);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
+ voiceCodec.rate = test_rate;
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
+ EXPECT_EQ(0, module1->SetSendingStatus(true));
+
+ AudioFeedback* audioFeedback = new AudioFeedback();
+ EXPECT_EQ(0, module2->RegisterAudioCallback(audioFeedback));
+
+ // Prepare for DTMF.
+ voiceCodec.pltype = 97;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "telephone-event", 16);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Start DTMF test.
+ WebRtc_UWord32 timeStamp = 160;
+
+ // Send a DTMF tone using RFC 2833 (4733).
+ for (int i = 0; i < 16; i++) {
+ EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
+ }
+ timeStamp += 160; // Prepare for next packet.
+
+ const WebRtc_UWord8 test[9] = "test";
+
+ // Send RTP packets for 16 tones a 160 ms 100ms
+ // pause between = 2560ms + 1600ms = 4160ms
+ for (;timeStamp <= 250 * 160; timeStamp += 160) {
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+ timeStamp, test, 4));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ }
+ EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
+
+ for (;timeStamp <= 740 * 160; timeStamp += 160) {
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+ timeStamp, test, 4));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ }
+ delete audioFeedback;
+}
+
+TEST_F(RtpRtcpAudioTest, Stereo) {
+ CodecInst voiceCodec;
+ voiceCodec.pltype = 96;
+ voiceCodec.plfreq = 8000;
+ memcpy(voiceCodec.plname, "PCMU", 5);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
+ voiceCodec.rate = test_rate;
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
+ EXPECT_EQ(0, module1->SetSendingStatus(true));
+
+ // Prepare for 3 channel audio 8 bits per sample.
+ voiceCodec.pltype = 98;
+ voiceCodec.channels = 3;
+ memcpy(voiceCodec.plname, "PCMA", 5);
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Prepare for 3 channel audio 16 bits per sample.
+ voiceCodec.pltype = 99;
+ memcpy(voiceCodec.plname, "L16", 4);
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Prepare for 3 channel audio 5 bits per sample.
+ voiceCodec.pltype = 100;
+ memcpy(voiceCodec.plname, "G726-40",8);
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Prepare for 3 channel audio 3 bits per sample.
+ voiceCodec.pltype = 101;
+ memcpy(voiceCodec.plname, "G726-24",8);
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Prepare for 3 channel audio 2 bits per sample.
+ voiceCodec.pltype = 102;
+ memcpy(voiceCodec.plname, "G726-16",8);
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // Test sample based multi channel codec, 3 channels 8 bits.
+ WebRtc_UWord8 test3channels[15] = "ttteeesssttt";
+ WebRtc_UWord32 timeStamp = 160;
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 98,
+ timeStamp, test3channels, 12));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ timeStamp += 160; // Prepare for next packet.
+
+ // Test sample based multi channel codec, 3 channels 16 bits.
+ const WebRtc_UWord8 test3channels16[13] = "teteteststst";
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 99,
+ timeStamp, test3channels16, 12));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ timeStamp += 160; // Prepare for next packet.
+
+ // Test sample based multi channel codec, 3 channels 5 bits.
+ test3channels[0] = 0xf8; // 5 ones 3 zeros.
+ test3channels[1] = 0x2b; // 2 zeros 5 10 1 one.
+ test3channels[2] = 0xf0; // 4 ones 4 zeros.
+ test3channels[3] = 0x2b; // 1 zero 5 01 2 ones.
+ test3channels[4] = 0xe0; // 3 ones 5 zeros.
+ test3channels[5] = 0x0;
+ test3channels[6] = 0x0;
+ test3channels[7] = 0x0;
+ test3channels[8] = 0x0;
+ test3channels[9] = 0x0;
+ test3channels[10] = 0x0;
+ test3channels[11] = 0x0;
+ test3channels[12] = 0x0;
+ test3channels[13] = 0x0;
+ test3channels[14] = 0x0;
+
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 100,
+ timeStamp, test3channels, 15));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ timeStamp += 160; // Prepare for next packet.
+
+ // Test sample based multi channel codec, 3 channels 3 bits.
+ test3channels[0] = 0xe2; // 3 ones 3 zeros 2 10
+ test3channels[1] = 0xf0; // 1 1 3 ones 3 zeros 1 0
+ test3channels[2] = 0xb8; // 2 10 3 ones 3 zeros
+ test3channels[3] = 0xa0; // 3 101 5 zeros
+ test3channels[4] = 0x0;
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 101,
+ timeStamp, test3channels, 15));
+ fake_clock.IncrementTime(20);
+ module1->Process();
+ timeStamp += 160; // Prepare for next packet.
+
+ // Test sample based multi channel codec, 3 channels 2 bits.
+ test3channels[0] = 0xcb; // 2 ones 2 zeros 2 10 2 ones
+ test3channels[1] = 0x2c; // 2 zeros 2 10 2 ones 2 zeros
+ test3channels[2] = 0xb2; // 2 10 2 ones 2 zeros 2 10
+ test3channels[3] = 0xcb; // 2 ones 2 zeros 2 10 2 ones
+ test3channels[4] = 0x2c; // 2 zeros 2 10 2 ones 2 zeros
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 102,
+ timeStamp, test3channels, 15));
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
new file mode 100644
index 0000000..fd3c3df
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
@@ -0,0 +1,249 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+#include <gtest/gtest.h>
+
+#include "test_api.h"
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+using namespace webrtc;
+
+const int kVideoNackListSize = 10;
+const int kTestId = 123;
+const WebRtc_UWord32 kTestSsrc = 3456;
+const WebRtc_UWord16 kTestSequenceNumber = 2345;
+const WebRtc_UWord32 kTestNumberOfPackets = 450;
+const int kTestNumberOfRtxPackets = 49;
+
+class VerifyingNackReceiver : public RtpData
+{
+ public:
+ VerifyingNackReceiver() {}
+
+ virtual WebRtc_Word32 OnReceivedPayloadData(
+ const WebRtc_UWord8* data,
+ const WebRtc_UWord16 size,
+ const webrtc::WebRtcRTPHeader* rtp_header) {
+
+ EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
+ EXPECT_EQ(find(sequence_numbers_.begin(),
+ sequence_numbers_.end(),
+ rtp_header->header.sequenceNumber),
+ sequence_numbers_.end());
+ sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
+ return 0;
+ }
+ std::vector<WebRtc_UWord16 > sequence_numbers_;
+};
+
+class NackLoopBackTransport : public webrtc::Transport {
+ public:
+ NackLoopBackTransport(RtpRtcp* rtp_rtcp_module, uint32_t rtx_ssrc)
+ : count_(0),
+ packet_loss_(0),
+ rtx_ssrc_(rtx_ssrc),
+ count_rtx_ssrc_(0),
+ module_(rtp_rtcp_module) {
+ }
+ void DropEveryNthPacket(int n) {
+ packet_loss_ = n;
+ }
+ virtual int SendPacket(int channel, const void *data, int len) {
+ count_++;
+ const unsigned char* ptr = static_cast<const unsigned char*>(data);
+ uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
+ if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
+
+ if (packet_loss_ > 0) {
+ if ((count_ % packet_loss_) == 0) {
+ return len;
+ }
+ }
+ if (module_->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+ return len;
+ }
+ return -1;
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len) {
+ if (module_->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+ return len;
+ }
+ return -1;
+ }
+ int count_;
+ int packet_loss_;
+ uint32_t rtx_ssrc_;
+ int count_rtx_ssrc_;
+ RtpRtcp* module_;
+};
+
+class RtpRtcpNackTest : public ::testing::Test {
+ protected:
+ RtpRtcpNackTest() {}
+ ~RtpRtcpNackTest() {}
+
+ virtual void SetUp() {
+ video_module_ = RtpRtcp::CreateRtpRtcp(kTestId, false, &fake_clock);
+ EXPECT_EQ(0, video_module_->InitReceiver());
+ EXPECT_EQ(0, video_module_->InitSender());
+ EXPECT_EQ(0, video_module_->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, video_module_->SetSSRC(kTestSsrc));
+ EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp));
+ EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true));
+ EXPECT_EQ(0, video_module_->SetSendingStatus(true));
+ EXPECT_EQ(0, video_module_->SetSequenceNumber(kTestSequenceNumber));
+ EXPECT_EQ(0, video_module_->SetStartTimestamp(111111));
+
+ transport_ = new NackLoopBackTransport(video_module_, kTestSsrc + 1);
+ EXPECT_EQ(0, video_module_->RegisterSendTransport(transport_));
+
+ nack_receiver_ = new VerifyingNackReceiver();
+ EXPECT_EQ(0, video_module_->RegisterIncomingDataCallback(nack_receiver_));
+
+ VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof(video_codec));
+ video_codec.plType = 123;
+ memcpy(video_codec.plName, "I420", 5);
+
+ EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
+ EXPECT_EQ(0, video_module_->RegisterReceivePayload(video_codec));
+
+ payload_data_length = sizeof(payload_data);
+
+ for (int n = 0; n < payload_data_length; n++) {
+ payload_data[n] = n % 10;
+ }
+ }
+
+ virtual void TearDown() {
+ RtpRtcp::DestroyRtpRtcp(video_module_);
+ delete transport_;
+ delete nack_receiver_;
+ }
+
+ RtpRtcp* video_module_;
+ NackLoopBackTransport* transport_;
+ VerifyingNackReceiver* nack_receiver_;
+ WebRtc_UWord8 payload_data[65000];
+ int payload_data_length;
+ FakeRtpRtcpClock fake_clock;
+};
+
+TEST_F(RtpRtcpNackTest, RTCP) {
+ WebRtc_UWord32 timestamp = 3000;
+ WebRtc_UWord16 nack_list[kVideoNackListSize];
+ transport_->DropEveryNthPacket(10);
+
+ for (int frame = 0; frame < 10; ++frame) {
+ EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
+ timestamp,
+ payload_data,
+ payload_data_length));
+
+ std::sort(nack_receiver_->sequence_numbers_.begin(),
+ nack_receiver_->sequence_numbers_.end());
+
+ std::vector<WebRtc_UWord16> missing_sequence_numbers;
+ std::vector<WebRtc_UWord16>::iterator it =
+ nack_receiver_->sequence_numbers_.begin();
+
+ while (it != nack_receiver_->sequence_numbers_.end()) {
+ WebRtc_UWord16 sequence_number_1 = *it;
+ ++it;
+ if (it != nack_receiver_->sequence_numbers_.end()) {
+ WebRtc_UWord16 sequence_number_2 = *it;
+ // Add all missing sequence numbers to list
+ for (WebRtc_UWord16 i = sequence_number_1 + 1; i < sequence_number_2;
+ ++i) {
+ missing_sequence_numbers.push_back(i);
+ }
+ }
+ }
+ int n = 0;
+ for (it = missing_sequence_numbers.begin();
+ it != missing_sequence_numbers.end(); ++it) {
+ nack_list[n++] = (*it);
+ }
+ video_module_->SendNACK(nack_list, n);
+ fake_clock.IncrementTime(33);
+ video_module_->Process();
+
+ // Prepare next frame.
+ timestamp += 3000;
+ }
+ std::sort(nack_receiver_->sequence_numbers_.begin(),
+ nack_receiver_->sequence_numbers_.end());
+ EXPECT_EQ(kTestSequenceNumber, *(nack_receiver_->sequence_numbers_.begin()));
+ EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
+ *(nack_receiver_->sequence_numbers_.rbegin()));
+ EXPECT_EQ(kTestNumberOfPackets, nack_receiver_->sequence_numbers_.size());
+ EXPECT_EQ(0, transport_->count_rtx_ssrc_);
+}
+
+TEST_F(RtpRtcpNackTest, RTX) {
+ EXPECT_EQ(0, video_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
+ EXPECT_EQ(0, video_module_->SetRTXSendStatus(true, true, kTestSsrc + 1));
+
+ transport_->DropEveryNthPacket(10);
+
+ WebRtc_UWord32 timestamp = 3000;
+ WebRtc_UWord16 nack_list[kVideoNackListSize];
+
+ for (int frame = 0; frame < 10; ++frame) {
+ EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
+ timestamp,
+ payload_data,
+ payload_data_length));
+
+ std::sort(nack_receiver_->sequence_numbers_.begin(),
+ nack_receiver_->sequence_numbers_.end());
+
+ std::vector<WebRtc_UWord16> missing_sequence_numbers;
+
+
+ std::vector<WebRtc_UWord16>::iterator it =
+ nack_receiver_->sequence_numbers_.begin();
+ while (it != nack_receiver_->sequence_numbers_.end()) {
+ int sequence_number_1 = *it;
+ ++it;
+ if (it != nack_receiver_->sequence_numbers_.end()) {
+ int sequence_number_2 = *it;
+ // Add all missing sequence numbers to list.
+ for (int i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
+ missing_sequence_numbers.push_back(i);
+ }
+ }
+ }
+ int n = 0;
+ for (it = missing_sequence_numbers.begin();
+ it != missing_sequence_numbers.end(); ++it) {
+ nack_list[n++] = (*it);
+ }
+ video_module_->SendNACK(nack_list, n);
+ fake_clock.IncrementTime(33);
+ video_module_->Process();
+
+ // Prepare next frame.
+ timestamp += 3000;
+ }
+ std::sort(nack_receiver_->sequence_numbers_.begin(),
+ nack_receiver_->sequence_numbers_.end());
+ EXPECT_EQ(kTestSequenceNumber, *(nack_receiver_->sequence_numbers_.begin()));
+ EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
+ *(nack_receiver_->sequence_numbers_.rbegin()));
+ EXPECT_EQ(kTestNumberOfPackets, nack_receiver_->sequence_numbers_.size());
+ EXPECT_EQ(kTestNumberOfRtxPackets, transport_->count_rtx_ssrc_);
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
new file mode 100644
index 0000000..306d47f
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -0,0 +1,333 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+#include <gtest/gtest.h>
+
+#include "test_api.h"
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+using namespace webrtc;
+
+const WebRtc_UWord64 kTestPictureId = 12345678;
+
+class RtcpCallback : public RtcpFeedback {
+ public:
+ RtcpCallback(RtpRtcp* module) {
+ _rtpRtcpModule = module;
+ };
+ virtual void OnRTCPPacketTimeout(const WebRtc_Word32 id) {
+ }
+ virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
+ const WebRtc_Word32 audioVideoOffset) {
+ };
+ virtual void OnTMMBRReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord16 bwEstimateKbit) {
+ };
+ virtual void OnXRVoIPMetricReceived(
+ const WebRtc_Word32 id,
+ const RTCPVoIPMetric* metric,
+ const WebRtc_Word8 VoIPmetricBuffer[28]) {
+ };
+ virtual void OnSLIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 pictureId) {
+ EXPECT_EQ(28, pictureId);
+ };
+
+ virtual void OnRPSIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord64 pictureId) {
+ EXPECT_EQ(kTestPictureId, pictureId);
+ };
+ virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8* data) {
+ char print_name[5];
+ print_name[0] = static_cast<char>(name >> 24);
+ print_name[1] = static_cast<char>(name >> 16);
+ print_name[2] = static_cast<char>(name >> 8);
+ print_name[3] = static_cast<char>(name);
+ print_name[4] = 0;
+
+ EXPECT_STRCASEEQ("test", print_name);
+ };
+
+ virtual void OnSendReportReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord32 senderSSRC) {
+ RTCPSenderInfo senderInfo;
+ EXPECT_EQ(0, _rtpRtcpModule->RemoteRTCPStat(&senderInfo));
+ };
+
+ virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord32 senderSSRC) {
+ };
+ private:
+ RtpRtcp* _rtpRtcpModule;
+};
+
+class RtpRtcpRtcpTest : public ::testing::Test {
+ protected:
+ RtpRtcpRtcpTest() {
+ test_CSRC[0] = 1234;
+ test_CSRC[2] = 2345;
+ test_id = 123;
+ test_ssrc = 3456;
+ test_timestamp = 4567;
+ test_sequence_number = 2345;
+ }
+ ~RtpRtcpRtcpTest() {}
+
+ virtual void SetUp() {
+ module1 = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
+ module2 = RtpRtcp::CreateRtpRtcp(test_id+1, true, &fake_clock);
+
+ EXPECT_EQ(0, module1->InitReceiver());
+ EXPECT_EQ(0, module1->InitSender());
+ EXPECT_EQ(0, module2->InitReceiver());
+ EXPECT_EQ(0, module2->InitSender());
+ receiver = new RtpReceiver();
+ EXPECT_EQ(0, module2->RegisterIncomingDataCallback(receiver));
+ transport1 = new LoopBackTransport(module2);
+ EXPECT_EQ(0, module1->RegisterSendTransport(transport1));
+ transport2 = new LoopBackTransport(module1);
+ EXPECT_EQ(0, module2->RegisterSendTransport(transport2));
+ }
+
+ virtual void TearDown() {
+ RtpRtcp::DestroyRtpRtcp(module1);
+ RtpRtcp::DestroyRtpRtcp(module2);
+ delete transport1;
+ delete transport2;
+ delete receiver;
+ }
+
+ void SetUpCallFromModule1(RtcpCallback* feedback1, RtcpCallback* feedback2 ) {
+ EXPECT_EQ(0, module1->RegisterIncomingRTCPCallback(feedback1));
+ EXPECT_EQ(0, module2->RegisterIncomingRTCPCallback(feedback2));
+
+ EXPECT_EQ(0, module1->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, module2->SetRTCPStatus(kRtcpCompound));
+
+ EXPECT_EQ(0, module2->SetSSRC(test_ssrc + 1));
+ EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, module1->SetSequenceNumber(test_sequence_number));
+ EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
+ EXPECT_EQ(0, module1->SetCSRCs(test_CSRC, 2));
+ EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
+
+ EXPECT_EQ(0, module1->SetSendingStatus(true));
+
+ CodecInst voiceCodec;
+ voiceCodec.pltype = 96;
+ voiceCodec.plfreq = 8000;
+ voiceCodec.rate = 64000;
+ memcpy(voiceCodec.plname, "PCMU", 5);
+
+ EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
+ EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
+
+ // We need to send one RTP packet to get the RTCP packet to be accepted by
+ // the receiving module.
+ // send RTP packet with the data "testtest"
+ const WebRtc_UWord8 test[9] = "testtest";
+ EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+ 0, test, 8));
+ }
+
+ int test_id;
+ RtpRtcp* module1;
+ RtpRtcp* module2;
+ RtpReceiver* receiver;
+ LoopBackTransport* transport1;
+ LoopBackTransport* transport2;
+ WebRtc_UWord32 test_ssrc;
+ WebRtc_UWord32 test_timestamp;
+ WebRtc_UWord16 test_sequence_number;
+ WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
+ FakeRtpRtcpClock fake_clock;
+};
+
+TEST_F(RtpRtcpRtcpTest, RTCP) {
+ RtcpCallback* myRTCPFeedback1 = new RtcpCallback(module1);
+ RtcpCallback* myRTCPFeedback2 = new RtcpCallback(module2);
+
+ SetUpCallFromModule1(myRTCPFeedback1, myRTCPFeedback2);
+
+ EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
+ EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(156));
+
+ WebRtc_UWord32 testOfCSRC[webrtc::kRtpCsrcSize];
+ EXPECT_EQ(2, module2->RemoteCSRCs(testOfCSRC));
+ EXPECT_EQ(test_CSRC[0], testOfCSRC[0]);
+ EXPECT_EQ(test_CSRC[1], testOfCSRC[1]);
+
+ // Set cname of mixed.
+ EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[0], "john@192.168.0.1"));
+ EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[1], "jane@192.168.0.2"));
+ EXPECT_EQ(-1, module1->AddMixedCNAME(test_CSRC[0], NULL));
+
+ EXPECT_EQ(-1, module1->RemoveMixedCNAME(test_CSRC[0] + 1));
+ EXPECT_EQ(0, module1->RemoveMixedCNAME(test_CSRC[1]));
+ EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[1], "jane@192.168.0.2"));
+
+ RTCPReportBlock reportBlock;
+ reportBlock.cumulativeLost = 1;
+ reportBlock.delaySinceLastSR = 2;
+ reportBlock.extendedHighSeqNum = 3;
+ reportBlock.fractionLost= 4;
+ reportBlock.jitter = 5;
+ reportBlock.lastSR = 6;
+
+ // Set report blocks.
+ EXPECT_EQ(-1, module1->AddRTCPReportBlock(test_CSRC[0], NULL));
+ EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[0], &reportBlock));
+
+ reportBlock.lastSR= 7;
+ EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[1], &reportBlock));
+
+ WebRtc_UWord32 name = 't' << 24;
+ name += 'e' << 16;
+ name += 's' << 8;
+ name += 't';
+ EXPECT_EQ(0, module1->SetRTCPApplicationSpecificData(
+ 3,
+ name,
+ (const WebRtc_UWord8 *)"test test test test test test test test test"\
+ " test test test test test test test test test test test test test"\
+ " test test test test test test test test test test test test test"\
+ " test test test test test test test test test test test test test"\
+ " test test test test test test test test test test test test ",
+ 300));
+
+ // send RTCP packet, triggered by timer
+ fake_clock.IncrementTime(7500);
+ module1->Process();
+ fake_clock.IncrementTime(100);
+ module2->Process();
+
+ WebRtc_UWord32 receivedNTPsecs = 0;
+ WebRtc_UWord32 receivedNTPfrac = 0;
+ WebRtc_UWord32 RTCPArrivalTimeSecs = 0;
+ WebRtc_UWord32 RTCPArrivalTimeFrac = 0;
+ char cName[RTCP_CNAME_SIZE];
+
+ EXPECT_EQ(0, module2->RemoteNTP(&receivedNTPsecs, &receivedNTPfrac,
+ &RTCPArrivalTimeSecs, &RTCPArrivalTimeFrac));
+
+ EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC() + 1, cName));
+ EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC(), NULL));
+
+ // Check multiple CNAME.
+ EXPECT_EQ(0, module2->RemoteCNAME(module2->RemoteSSRC(), cName));
+ EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
+
+ EXPECT_EQ(0, module2->RemoteCNAME(test_CSRC[0], cName));
+ EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
+
+ EXPECT_EQ(0, module2->RemoteCNAME(test_CSRC[1], cName));
+ EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
+
+ // get all report blocks
+ std::vector<RTCPReportBlock> report_blocks;
+ EXPECT_EQ(-1, module1->RemoteRTCPStat(NULL));
+ EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
+ EXPECT_EQ(1u, report_blocks.size());
+ const RTCPReportBlock& reportBlockReceived = report_blocks[0];
+
+ float secSinceLastReport =
+ static_cast<float>(reportBlockReceived.delaySinceLastSR) / 65536.0f;
+ EXPECT_GE(0.101f, secSinceLastReport);
+ EXPECT_LE(0.100f, secSinceLastReport);
+ EXPECT_EQ(test_sequence_number, reportBlockReceived.extendedHighSeqNum);
+ EXPECT_EQ(0, reportBlockReceived.fractionLost);
+
+ EXPECT_EQ(static_cast<WebRtc_UWord32>(0),
+ reportBlockReceived.cumulativeLost);
+
+ WebRtc_UWord8 fraction_lost = 0; // scale 0 to 255
+ WebRtc_UWord32 cum_lost = 0; // number of lost packets
+ WebRtc_UWord32 ext_max = 0; // highest sequence number received
+ WebRtc_UWord32 jitter = 0;
+ WebRtc_UWord32 max_jitter = 0;
+ EXPECT_EQ(0, module2->StatisticsRTP(&fraction_lost,
+ &cum_lost,
+ &ext_max,
+ &jitter,
+ &max_jitter));
+ EXPECT_EQ(0, fraction_lost);
+ EXPECT_EQ((WebRtc_UWord32)0, cum_lost);
+ EXPECT_EQ(test_sequence_number, ext_max);
+ EXPECT_EQ(reportBlockReceived.jitter, jitter);
+
+ WebRtc_UWord16 RTT;
+ WebRtc_UWord16 avgRTT;
+ WebRtc_UWord16 minRTT;
+ WebRtc_UWord16 maxRTT;
+
+ // Get RoundTripTime.
+ EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT));
+ EXPECT_GE(10, RTT);
+ EXPECT_GE(10, avgRTT);
+ EXPECT_GE(10, minRTT);
+ EXPECT_GE(10, maxRTT);
+
+ // Set report blocks.
+ EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[0], &reportBlock));
+
+ // Test receive report.
+ EXPECT_EQ(0, module1->SetSendingStatus(false));
+
+ // Test that BYE clears the CNAME
+ EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC(), cName));
+
+ // Send RTCP packet, triggered by timer.
+ fake_clock.IncrementTime(5000);
+ module1->Process();
+ module2->Process();
+
+ delete myRTCPFeedback1;
+ delete myRTCPFeedback2;
+}
+
+TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
+ std::vector<RTCPReportBlock> report_blocks;
+
+ RtcpCallback feedback1(module1);
+ RtcpCallback feedback2(module2);
+
+ SetUpCallFromModule1(&feedback1, &feedback2);
+ EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
+ EXPECT_EQ(0u, report_blocks.size());
+
+ // send RTCP packet, triggered by timer
+ fake_clock.IncrementTime(7500);
+ module1->Process();
+ fake_clock.IncrementTime(100);
+ module2->Process();
+
+ EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
+ ASSERT_EQ(1u, report_blocks.size());
+
+ // |test_ssrc+1| is the SSRC of module2 that send the report.
+ EXPECT_EQ(test_ssrc+1, report_blocks[0].remoteSSRC);
+ EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
+
+ EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
+ EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
+ EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
+ EXPECT_EQ(0u, report_blocks[0].fractionLost);
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
new file mode 100644
index 0000000..7a06ef7
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+#include <gtest/gtest.h>
+
+#include "test_api.h"
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+using namespace webrtc;
+
+class RtpRtcpVideoTest : public ::testing::Test {
+ protected:
+ RtpRtcpVideoTest() {
+ test_id = 123;
+ test_ssrc = 3456;
+ test_timestamp = 4567;
+ test_sequence_number = 2345;
+ }
+ ~RtpRtcpVideoTest() {}
+
+ virtual void SetUp() {
+ video_module = RtpRtcp::CreateRtpRtcp(test_id, false, &fake_clock);
+ EXPECT_EQ(0, video_module->InitReceiver());
+ EXPECT_EQ(0, video_module->InitSender());
+ EXPECT_EQ(0, video_module->SetRTCPStatus(kRtcpCompound));
+ EXPECT_EQ(0, video_module->SetSSRC(test_ssrc));
+ EXPECT_EQ(0, video_module->SetNACKStatus(kNackRtcp));
+ EXPECT_EQ(0, video_module->SetStorePacketsStatus(true));
+ EXPECT_EQ(0, video_module->SetSendingStatus(true));
+
+ transport = new LoopBackTransport(video_module);
+ EXPECT_EQ(0, video_module->RegisterSendTransport(transport));
+
+ receiver = new RtpReceiver();
+ EXPECT_EQ(0, video_module->RegisterIncomingDataCallback(receiver));
+
+ VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof(video_codec));
+ video_codec.plType = 123;
+ memcpy(video_codec.plName, "I420", 5);
+
+ EXPECT_EQ(0, video_module->RegisterSendPayload(video_codec));
+ EXPECT_EQ(0, video_module->RegisterReceivePayload(video_codec));
+
+ payload_data_length = sizeof(payload_data);
+
+ for (int n = 0; n < payload_data_length; n++) {
+ payload_data[n] = n%10;
+ }
+ }
+
+ virtual void TearDown() {
+ RtpRtcp::DestroyRtpRtcp(video_module);
+ delete transport;
+ delete receiver;
+ }
+
+ int test_id;
+ RtpRtcp* video_module;
+ LoopBackTransport* transport;
+ RtpReceiver* receiver;
+ WebRtc_UWord32 test_ssrc;
+ WebRtc_UWord32 test_timestamp;
+ WebRtc_UWord16 test_sequence_number;
+ WebRtc_UWord8 payload_data[65000];
+ int payload_data_length;
+ FakeRtpRtcpClock fake_clock;
+};
+
+TEST_F(RtpRtcpVideoTest, BasicVideo) {
+ WebRtc_UWord32 timestamp = 3000;
+ EXPECT_EQ(0, video_module->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
+ timestamp,
+ payload_data,
+ payload_data_length));
+
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc
new file mode 100644
index 0000000..867c502
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -0,0 +1,492 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Test application for core FEC algorithm. Calls encoding and decoding
+ * functions in ForwardErrorCorrection directly.
+ */
+
+#include <cassert>
+#include <cstdio>
+#include <cstdlib>
+#include <cstring>
+#include <ctime>
+#include <list>
+
+#include "forward_error_correction.h"
+#include "forward_error_correction_internal.h"
+
+#include "rtp_utility.h"
+#include "testsupport/fileutils.h"
+
+//#define VERBOSE_OUTPUT
+
+using namespace webrtc;
+
+void ReceivePackets(
+ ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
+ ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
+ WebRtc_UWord32 numPacketsToDecode, float reorderRate, float duplicateRate);
+
+int main() {
+ enum { kMaxNumberMediaPackets = 48 };
+ enum { kMaxNumberFecPackets = 48 };
+
+ const WebRtc_UWord32 kNumMaskBytesL0 = 2;
+ const WebRtc_UWord32 kNumMaskBytesL1 = 6;
+
+ // FOR UEP
+ const bool kUseUnequalProtection = true;
+
+ WebRtc_UWord32 id = 0;
+ ForwardErrorCorrection fec(id);
+
+ ForwardErrorCorrection::PacketList mediaPacketList;
+ ForwardErrorCorrection::PacketList fecPacketList;
+ ForwardErrorCorrection::ReceivedPacketList toDecodeList;
+ ForwardErrorCorrection::ReceivedPacketList receivedPacketList;
+ ForwardErrorCorrection::RecoveredPacketList recoveredPacketList;
+ std::list<WebRtc_UWord8*> fecMaskList;
+
+ ForwardErrorCorrection::Packet* mediaPacket;
+ // Running over only one loss rate to limit execution time.
+ const float lossRate[] = {0.5f};
+ const WebRtc_UWord32 lossRateSize = sizeof(lossRate)/sizeof(*lossRate);
+ const float reorderRate = 0.1f;
+ const float duplicateRate = 0.1f;
+
+ WebRtc_UWord8 mediaLossMask[kMaxNumberMediaPackets];
+ WebRtc_UWord8 fecLossMask[kMaxNumberFecPackets];
+ WebRtc_UWord8 fecPacketMasks[kMaxNumberFecPackets][kMaxNumberMediaPackets];
+
+ // Seed the random number generator, storing the seed to file in order to
+ // reproduce past results.
+ const unsigned int randomSeed = static_cast<unsigned int>(time(NULL));
+ srand(randomSeed);
+ std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
+ FILE* randomSeedFile = fopen(filename.c_str(), "a");
+ fprintf(randomSeedFile, "%u\n", randomSeed);
+ fclose(randomSeedFile);
+ randomSeedFile = NULL;
+
+ WebRtc_UWord16 seqNum = static_cast<WebRtc_UWord16>(rand());
+ WebRtc_UWord32 timeStamp = static_cast<WebRtc_UWord32>(rand());
+ const WebRtc_UWord32 ssrc = static_cast<WebRtc_UWord32>(rand());
+
+ for (WebRtc_UWord32 lossRateIdx = 0; lossRateIdx < lossRateSize;
+ lossRateIdx++) {
+ WebRtc_UWord8* packetMask =
+ new WebRtc_UWord8[kMaxNumberMediaPackets * kNumMaskBytesL1];
+
+ printf("Loss rate: %.2f\n", lossRate[lossRateIdx]);
+ for (WebRtc_UWord32 numMediaPackets = 1;
+ numMediaPackets <= kMaxNumberMediaPackets;
+ numMediaPackets++) {
+
+ for (WebRtc_UWord32 numFecPackets = 1;
+ numFecPackets <= numMediaPackets &&
+ numFecPackets <= kMaxNumberFecPackets;
+ numFecPackets++) {
+
+ // Loop over numImpPackets: these are usually <= (0.3*numMediaPackets).
+ // For this test we check up to ~ (0.5*numMediaPackets).
+ WebRtc_UWord32 maxNumImpPackets = numMediaPackets / 2 + 1;
+ for (WebRtc_UWord32 numImpPackets = 0;
+ numImpPackets <= maxNumImpPackets &&
+ numImpPackets <= kMaxNumberMediaPackets;
+ numImpPackets++) {
+
+ WebRtc_UWord8 protectionFactor = static_cast<WebRtc_UWord8>
+ (numFecPackets * 255 / numMediaPackets);
+
+ const WebRtc_UWord32 maskBytesPerFecPacket =
+ (numMediaPackets > 16) ? kNumMaskBytesL1 : kNumMaskBytesL0;
+
+ memset(packetMask, 0, numMediaPackets * maskBytesPerFecPacket);
+
+ // Transfer packet masks from bit-mask to byte-mask.
+ internal::GeneratePacketMasks(numMediaPackets,
+ numFecPackets,
+ numImpPackets,
+ kUseUnequalProtection,
+ packetMask);
+
+#ifdef VERBOSE_OUTPUT
+ printf("%u media packets, %u FEC packets, %u numImpPackets, "
+ "loss rate = %.2f \n",
+ numMediaPackets, numFecPackets, numImpPackets, lossRate[lossRateIdx]);
+ printf("Packet mask matrix \n");
+#endif
+
+ for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
+ for (WebRtc_UWord32 j = 0; j < numMediaPackets; j++) {
+ const WebRtc_UWord8 byteMask =
+ packetMask[i * maskBytesPerFecPacket + j / 8];
+ const WebRtc_UWord32 bitPosition = (7 - j % 8);
+ fecPacketMasks[i][j] =
+ (byteMask & (1 << bitPosition)) >> bitPosition;
+#ifdef VERBOSE_OUTPUT
+ printf("%u ", fecPacketMasks[i][j]);
+#endif
+ }
+#ifdef VERBOSE_OUTPUT
+ printf("\n");
+#endif
+ }
+#ifdef VERBOSE_OUTPUT
+ printf("\n");
+#endif
+ // Check for all zero rows or columns: indicates incorrect mask
+ WebRtc_UWord32 rowLimit = numMediaPackets;
+ for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
+ WebRtc_UWord32 rowSum = 0;
+ for (WebRtc_UWord32 j = 0; j < rowLimit; j++) {
+ rowSum += fecPacketMasks[i][j];
+ }
+ if (rowSum == 0) {
+ printf("ERROR: row is all zero %d \n",i);
+ return -1;
+ }
+ }
+ for (WebRtc_UWord32 j = 0; j < rowLimit; j++) {
+ WebRtc_UWord32 columnSum = 0;
+ for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
+ columnSum += fecPacketMasks[i][j];
+ }
+ if (columnSum == 0) {
+ printf("ERROR: column is all zero %d \n",j);
+ return -1;
+ }
+ }
+ // Construct media packets.
+ for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
+ mediaPacket = new ForwardErrorCorrection::Packet;
+ mediaPacketList.push_back(mediaPacket);
+ mediaPacket->length =
+ static_cast<WebRtc_UWord16>((static_cast<float>(rand()) /
+ RAND_MAX) * (IP_PACKET_SIZE - 12 - 28 -
+ ForwardErrorCorrection::PacketOverhead()));
+ if (mediaPacket->length < 12) {
+ mediaPacket->length = 12;
+ }
+ // Generate random values for the first 2 bytes
+ mediaPacket->data[0] = static_cast<WebRtc_UWord8>(rand() % 256);
+ mediaPacket->data[1] = static_cast<WebRtc_UWord8>(rand() % 256);
+
+ // The first two bits are assumed to be 10 by the
+ // FEC encoder. In fact the FEC decoder will set the
+ // two first bits to 10 regardless of what they
+ // actually were. Set the first two bits to 10
+ // so that a memcmp can be performed for the
+ // whole restored packet.
+ mediaPacket->data[0] |= 0x80;
+ mediaPacket->data[0] &= 0xbf;
+
+ // FEC is applied to a whole frame.
+ // A frame is signaled by multiple packets without
+ // the marker bit set followed by the last packet of
+ // the frame for which the marker bit is set.
+ // Only push one (fake) frame to the FEC.
+ mediaPacket->data[1] &= 0x7f;
+
+ ModuleRTPUtility::AssignUWord16ToBuffer(&mediaPacket->data[2],
+ seqNum);
+ ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[4],
+ timeStamp);
+ ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[8],
+ ssrc);
+ // Generate random values for payload
+ for (WebRtc_Word32 j = 12; j < mediaPacket->length; j++) {
+ mediaPacket->data[j] =
+ static_cast<WebRtc_UWord8> (rand() % 256);
+ }
+ seqNum++;
+ }
+ mediaPacket->data[1] |= 0x80;
+
+ if (fec.GenerateFEC(mediaPacketList, protectionFactor,
+ numImpPackets, kUseUnequalProtection,
+ &fecPacketList) != 0) {
+ printf("Error: GenerateFEC() failed\n");
+ return -1;
+ }
+
+ if (fecPacketList.size() != numFecPackets) {
+ printf("Error: we requested %u FEC packets, "
+ "but GenerateFEC() produced %u\n",
+ numFecPackets,
+ static_cast<WebRtc_UWord32>(fecPacketList.size()));
+ return -1;
+ }
+ memset(mediaLossMask, 0, sizeof(mediaLossMask));
+ ForwardErrorCorrection::PacketList::iterator
+ mediaPacketListItem = mediaPacketList.begin();
+ ForwardErrorCorrection::ReceivedPacket* receivedPacket;
+ WebRtc_UWord32 mediaPacketIdx = 0;
+
+ while (mediaPacketListItem != mediaPacketList.end()) {
+ mediaPacket = *mediaPacketListItem;
+ const float lossRandomVariable = (static_cast<float>(rand()) /
+ (RAND_MAX));
+
+ if (lossRandomVariable >= lossRate[lossRateIdx])
+ {
+ mediaLossMask[mediaPacketIdx] = 1;
+ receivedPacket =
+ new ForwardErrorCorrection::ReceivedPacket;
+ receivedPacket->pkt = new ForwardErrorCorrection::Packet;
+ receivedPacketList.push_back(receivedPacket);
+
+ receivedPacket->pkt->length = mediaPacket->length;
+ memcpy(receivedPacket->pkt->data, mediaPacket->data,
+ mediaPacket->length);
+ receivedPacket->seqNum =
+ ModuleRTPUtility::BufferToUWord16(&mediaPacket->data[2]);
+ receivedPacket->isFec = false;
+ }
+ mediaPacketIdx++;
+ ++mediaPacketListItem;
+ }
+ memset(fecLossMask, 0, sizeof(fecLossMask));
+ ForwardErrorCorrection::PacketList::iterator
+ fecPacketListItem = fecPacketList.begin();
+ ForwardErrorCorrection::Packet* fecPacket;
+ WebRtc_UWord32 fecPacketIdx = 0;
+ while (fecPacketListItem != fecPacketList.end()) {
+ fecPacket = *fecPacketListItem;
+ const float lossRandomVariable =
+ (static_cast<float>(rand()) / (RAND_MAX));
+ if (lossRandomVariable >= lossRate[lossRateIdx]) {
+ fecLossMask[fecPacketIdx] = 1;
+ receivedPacket =
+ new ForwardErrorCorrection::ReceivedPacket;
+ receivedPacket->pkt = new ForwardErrorCorrection::Packet;
+
+ receivedPacketList.push_back(receivedPacket);
+
+ receivedPacket->pkt->length = fecPacket->length;
+ memcpy(receivedPacket->pkt->data, fecPacket->data,
+ fecPacket->length);
+
+ receivedPacket->seqNum = seqNum;
+ receivedPacket->isFec = true;
+ receivedPacket->ssrc = ssrc;
+
+ fecMaskList.push_back(fecPacketMasks[fecPacketIdx]);
+ }
+ fecPacketIdx++;
+ seqNum++;
+ ++fecPacketListItem;
+ }
+
+#ifdef VERBOSE_OUTPUT
+ printf("Media loss mask:\n");
+ for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
+ printf("%u ", mediaLossMask[i]);
+ }
+ printf("\n\n");
+
+ printf("FEC loss mask:\n");
+ for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
+ printf("%u ", fecLossMask[i]);
+ }
+ printf("\n\n");
+#endif
+
+ std::list<WebRtc_UWord8*>::iterator fecMaskIt = fecMaskList.begin();
+ WebRtc_UWord8* fecMask;
+ while (fecMaskIt != fecMaskList.end()) {
+ fecMask = *fecMaskIt;
+ WebRtc_UWord32 hammingDist = 0;
+ WebRtc_UWord32 recoveryPosition = 0;
+ for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
+ if (mediaLossMask[i] == 0 && fecMask[i] == 1) {
+ recoveryPosition = i;
+ hammingDist++;
+ }
+ }
+ std::list<WebRtc_UWord8*>::iterator itemToDelete = fecMaskIt;
+ ++fecMaskIt;
+
+ if (hammingDist == 1) {
+ // Recovery possible. Restart search.
+ mediaLossMask[recoveryPosition] = 1;
+ fecMaskIt = fecMaskList.begin();
+ } else if (hammingDist == 0) {
+ // FEC packet cannot provide further recovery.
+ fecMaskList.erase(itemToDelete);
+ }
+ }
+#ifdef VERBOSE_OUTPUT
+ printf("Recovery mask:\n");
+ for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
+ printf("%u ", mediaLossMask[i]);
+ }
+ printf("\n\n");
+#endif
+ bool fecPacketReceived = false; // For error-checking frame completion.
+ while (!receivedPacketList.empty()) {
+ WebRtc_UWord32 numPacketsToDecode = static_cast<WebRtc_UWord32>
+ ((static_cast<float>(rand()) / RAND_MAX) *
+ receivedPacketList.size() + 0.5);
+ if (numPacketsToDecode < 1) {
+ numPacketsToDecode = 1;
+ }
+ ReceivePackets(&toDecodeList, &receivedPacketList,
+ numPacketsToDecode, reorderRate, duplicateRate);
+
+ if (fecPacketReceived == false) {
+ ForwardErrorCorrection::ReceivedPacketList::iterator
+ toDecodeIt = toDecodeList.begin();
+ while (toDecodeIt != toDecodeList.end()) {
+ receivedPacket = *toDecodeIt;
+ if (receivedPacket->isFec) {
+ fecPacketReceived = true;
+ }
+ ++toDecodeIt;
+ }
+ }
+ if (fec.DecodeFEC(&toDecodeList, &recoveredPacketList)
+ != 0) {
+ printf("Error: DecodeFEC() failed\n");
+ return -1;
+ }
+ if (!toDecodeList.empty()) {
+ printf("Error: received packet list is not empty\n");
+ return -1;
+ }
+ }
+ mediaPacketListItem = mediaPacketList.begin();
+ mediaPacketIdx = 0;
+ while (mediaPacketListItem != mediaPacketList.end()) {
+ if (mediaLossMask[mediaPacketIdx] == 1) {
+ // Should have recovered this packet.
+ ForwardErrorCorrection::RecoveredPacketList::iterator
+ recoveredPacketListItem = recoveredPacketList.begin();
+
+ if (recoveredPacketListItem == recoveredPacketList.end()) {
+ printf("Error: insufficient number of recovered packets.\n");
+ return -1;
+ }
+ mediaPacket = *mediaPacketListItem;
+ ForwardErrorCorrection::RecoveredPacket* recoveredPacket =
+ *recoveredPacketListItem;
+
+ if (recoveredPacket->pkt->length != mediaPacket->length) {
+ printf("Error: recovered packet length not identical to "
+ "original media packet\n");
+ return -1;
+ }
+ if (memcmp(recoveredPacket->pkt->data, mediaPacket->data,
+ mediaPacket->length) != 0) {
+ printf("Error: recovered packet payload not identical to "
+ "original media packet\n");
+ return -1;
+ }
+ delete recoveredPacket;
+ recoveredPacketList.pop_front();
+ }
+ mediaPacketIdx++;
+ ++mediaPacketListItem;
+ }
+ fec.ResetState(&recoveredPacketList);
+ if (!recoveredPacketList.empty()) {
+ printf("Error: excessive number of recovered packets.\n");
+ printf("\t size is:%u\n",
+ static_cast<WebRtc_UWord32>(recoveredPacketList.size()));
+ return -1;
+ }
+ // -- Teardown --
+ mediaPacketListItem = mediaPacketList.begin();
+ while (mediaPacketListItem != mediaPacketList.end()) {
+ delete *mediaPacketListItem;
+ ++mediaPacketListItem;
+ mediaPacketList.pop_front();
+ }
+ assert(mediaPacketList.empty());
+
+ fecPacketListItem = fecPacketList.begin();
+ while (fecPacketListItem != fecPacketList.end()) {
+ ++fecPacketListItem;
+ fecPacketList.pop_front();
+ }
+
+ // Delete received packets we didn't pass to DecodeFEC(), due to early
+ // frame completion.
+ ForwardErrorCorrection::ReceivedPacketList::iterator
+ receivedPacketIt = receivedPacketList.begin();
+ while (receivedPacketIt != receivedPacketList.end()) {
+ receivedPacket = *receivedPacketIt;
+ delete receivedPacket;
+ ++receivedPacketIt;
+ receivedPacketList.pop_front();
+ }
+ assert(receivedPacketList.empty());
+
+ while (!fecMaskList.empty()) {
+ fecMaskList.pop_front();
+ }
+ timeStamp += 90000 / 30;
+ } //loop over numImpPackets
+ } //loop over FecPackets
+ } //loop over numMediaPackets
+ delete [] packetMask;
+ } // loop over loss rates
+
+ // Have DecodeFEC free allocated memory.
+ fec.ResetState(&recoveredPacketList);
+ if (!recoveredPacketList.empty()) {
+ printf("Error: recovered packet list is not empty\n");
+ return -1;
+ }
+ printf("\nAll tests passed successfully\n");
+ return 0;
+}
+
+void ReceivePackets(
+ ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
+ ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
+ WebRtc_UWord32 numPacketsToDecode, float reorderRate, float duplicateRate) {
+ assert(toDecodeList->empty());
+ assert(numPacketsToDecode <= receivedPacketList->size());
+
+ ForwardErrorCorrection::ReceivedPacketList::iterator it;
+ for (WebRtc_UWord32 i = 0; i < numPacketsToDecode; i++) {
+ it = receivedPacketList->begin();
+ // Reorder packets.
+ float randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ while (randomVariable < reorderRate) {
+ ++it;
+ if (it == receivedPacketList->end()) {
+ --it;
+ break;
+ }
+ randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ }
+ ForwardErrorCorrection::ReceivedPacket* receivedPacket = *it;
+ toDecodeList->push_back(receivedPacket);
+
+ // Duplicate packets.
+ randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ while (randomVariable < duplicateRate) {
+ ForwardErrorCorrection::ReceivedPacket* duplicatePacket =
+ new ForwardErrorCorrection::ReceivedPacket;
+ *duplicatePacket = *receivedPacket;
+ duplicatePacket->pkt = new ForwardErrorCorrection::Packet;
+ memcpy(duplicatePacket->pkt->data, receivedPacket->pkt->data,
+ receivedPacket->pkt->length);
+ duplicatePacket->pkt->length = receivedPacket->pkt->length;
+
+ toDecodeList->push_back(duplicatePacket);
+ randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ }
+ receivedPacketList->erase(it);
+ }
+}
diff --git a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi
new file mode 100644
index 0000000..b7f933b
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi
@@ -0,0 +1,36 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'test_fec',
+ 'type': 'executable',
+ 'dependencies': [
+ 'rtp_rtcp',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+
+ 'include_dirs': [
+ '../../source',
+ '../../../../system_wrappers/interface',
+ ],
+
+ 'sources': [
+ 'test_fec.cc',
+ ],
+
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc b/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc
new file mode 100644
index 0000000..e4c36d1
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc
@@ -0,0 +1,580 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <windows.h>
+#include <iostream>
+#include <tchar.h>
+
+#include "h263_information.h"
+
+#define TEST_STR "Test H263 parser."
+#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
+#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
+
+
+void PrintInfo(const H263Info* ptr)
+{
+ std::cout << "info = GOBs: {";
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ std::cout << ptr->ptrGOBbuffer[i] << " ";
+ }
+ std::cout << "}" << std::endl;
+ std::cout << " sBit: {";
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ std::cout << (int)ptr->ptrGOBbufferSBit[i] << " ";
+ }
+ std::cout << "}" << std::endl;
+
+ std::cout << "uiH263PTypeFmt: " << (int)ptr->uiH263PTypeFmt << std::endl;
+ std::cout << " codecBits: " << (int)ptr->codecBits << std::endl;
+ std::cout << " pQuant: " << (int)ptr->pQuant << std::endl;
+ std::cout << " fType: " << (int)ptr->fType << std::endl;
+ std::cout << " cpmBit: " << (int)ptr->cpmBit << std::endl;
+ std::cout << " numOfGOBs: " << (int)ptr->numOfGOBs << std::endl;
+ std::cout << " numOfMBs: " << (int)ptr->totalNumOfMBs << std::endl;
+ std::cout << " " << std::endl;
+};
+
+void PrintMBInfo(
+ const H263Info* ptr,
+ const H263MBInfo* ptrMB)
+{
+ std::cout << " gQuant: {";
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ std::cout << (int)ptr->ptrGQuant[i] << " ";
+ }
+ std::cout << "}" << std::endl;
+ std::cout << " " << std::endl;
+
+ std::cout << "MBs:{";
+ int k = 0;
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ std::cout << " {";
+ for (int j = 0; j < ptr->ptrNumOfMBs[i]; j++)
+ {
+ std::cout << ptrMB->ptrBuffer[k++] << " ";
+ }
+ std::cout << "}" << std::endl;
+ }
+ std::cout << "}" << std::endl;
+ PRINT_LINE;
+};
+
+void ValidateResults(
+ const H263Info* ptr,
+ const H263Info* ptrRef)
+{
+ assert(ptr->uiH263PTypeFmt == ptrRef->uiH263PTypeFmt);
+ assert(ptr->codecBits == ptrRef->codecBits);
+ assert(ptr->pQuant == ptrRef->pQuant);
+ assert(ptr->fType == ptrRef->fType);
+ assert(ptr->numOfGOBs == ptrRef->numOfGOBs);
+ assert(ptr->totalNumOfMBs == ptrRef->totalNumOfMBs);
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ assert(ptr->ptrGOBbuffer[i] == ptrRef->ptrGOBbuffer[i]);
+ assert(ptr->ptrGOBbufferSBit[i] == ptrRef->ptrGOBbufferSBit[i]);
+ }
+ PrintInfo(ptr);
+}
+
+void ValidateMBResults(
+ const H263Info* ptr,
+ const H263MBInfo* ptrMB,
+ const H263Info* ptrRef,
+ bool printRes = true)
+{
+ int offset = 0;
+ int numBytes = 0;
+ for (int i = 0; i < ptr->numOfGOBs; i++)
+ {
+ offset = ptr->CalculateMBOffset(i+1);
+ numBytes += ptrMB->ptrBuffer[offset - 1] / 8;
+ int numBytesRem = ptrMB->ptrBuffer[offset - 1] % 8;
+ if (numBytesRem)
+ {
+ numBytes++;
+ }
+ assert(ptr->ptrGQuant[i] == ptrRef->ptrGQuant[i]);
+ }
+ assert(ptr->ptrGOBbuffer[ptr->numOfGOBs] == numBytes);
+ assert(unsigned int( ptr->totalNumOfMBs) <= ptrMB->bufferSize);
+ if (printRes)
+ {
+ PrintMBInfo(ptr, ptrMB);
+ }
+}
+
+
+int _tmain(int argc, _TCHAR* argv[])
+{
+
+ std::string str;
+ std::cout << "--------------------------------" << std::endl;
+ std::cout << "------- Test H.263 Parser ------" << std::endl;
+ std::cout << "--------------------------------" << std::endl;
+ std::cout << " " << std::endl;
+
+ // -----------------------------------------------------
+ // Input data - H.263 encoded stream SQCIF (P-frame)
+ // -----------------------------------------------------
+ const int lengthRefSQCIF = 77;
+ const unsigned char encodedStreamSQCIF[lengthRefSQCIF] = {
+ 0x00, 0x00, 0x81, 0xf6, 0x06, 0x04, 0x3f, 0xb7, 0xbc, 0x00, 0x00, 0x86, 0x23,
+ 0x5b, 0xdb, 0xdf, 0xb1, 0x93, 0xdb, 0xde, 0xd6, 0xf0, 0x00, 0x00, 0x8a, 0x27,
+ 0xdb, 0xcf, 0xad, 0xbe, 0x00, 0x00, 0x8e, 0x27, 0xed, 0xef, 0x80, 0x00, 0x00,
+ 0x92, 0x27, 0x6f, 0x7f, 0x80, 0x00, 0x00, 0x96, 0x20, 0xfc, 0xe2, 0xdb, 0xfe,
+ 0xb7, 0x7d, 0xea, 0x5f, 0xf8, 0xab, 0xd2, 0xff, 0xf6, 0xc9, 0xe5, 0x5e, 0x97,
+ 0xf7, 0xff, 0xad, 0x4f, 0x49, 0x3b, 0xff, 0xd6, 0xa6, 0x75, 0x82, 0x60};
+
+ // Expected results
+ H263Info infoRefSQCIF;
+ infoRefSQCIF.uiH263PTypeFmt = 1;
+ infoRefSQCIF.codecBits = 8;
+ infoRefSQCIF.pQuant = 4;
+ infoRefSQCIF.fType = 1;
+ infoRefSQCIF.cpmBit = 0;
+ infoRefSQCIF.numOfGOBs = 6;
+ infoRefSQCIF.totalNumOfMBs = 8*6;
+
+ infoRefSQCIF.ptrGOBbuffer[0] = 0; infoRefSQCIF.ptrGOBbufferSBit[0] = 0; infoRefSQCIF.ptrGQuant[0] = 0;
+ infoRefSQCIF.ptrGOBbuffer[1] = 9; infoRefSQCIF.ptrGOBbufferSBit[1] = 0; infoRefSQCIF.ptrGQuant[1] = 4;
+ infoRefSQCIF.ptrGOBbuffer[2] = 22; infoRefSQCIF.ptrGOBbufferSBit[2] = 0; infoRefSQCIF.ptrGQuant[2] = 4;
+ infoRefSQCIF.ptrGOBbuffer[3] = 30; infoRefSQCIF.ptrGOBbufferSBit[3] = 0; infoRefSQCIF.ptrGQuant[3] = 4;
+ infoRefSQCIF.ptrGOBbuffer[4] = 37; infoRefSQCIF.ptrGOBbufferSBit[4] = 0; infoRefSQCIF.ptrGQuant[4] = 4;
+ infoRefSQCIF.ptrGOBbuffer[5] = 44; infoRefSQCIF.ptrGOBbufferSBit[5] = 0; infoRefSQCIF.ptrGQuant[5] = 4;
+
+ // ----------------------------------------------------
+ // Input data - H.263 encoded stream QCIF (P-frame)
+ // ----------------------------------------------------
+ const int lengthRefQCIF = 123;
+ const unsigned char encodedStreamQCIF[lengthRefQCIF] = {
+ 0x00, 0x00, 0x81, 0x02, 0x0a, 0x04, 0x3f, 0xf8, 0x00, 0x00, 0x86, 0x27, 0x8b,
+ 0xc6, 0x9f, 0x17, 0x9c, 0x00, 0x00, 0x8a, 0x20, 0xbc, 0x22, 0xf8, 0x5f, 0x46,
+ 0x03, 0xc1, 0x77, 0x15, 0xe0, 0xb8, 0x38, 0x3f, 0x05, 0xa0, 0xbf, 0x8f, 0x00,
+ 0x00, 0x8e, 0x27, 0xfc, 0x5e, 0x5a, 0x33, 0x80, 0x00, 0x00, 0x92, 0x25, 0x8c,
+ 0x1e, 0xbf, 0xfc, 0x7e, 0x35, 0xfc, 0x00, 0x00, 0x96, 0x27, 0xff, 0x00, 0x00,
+ 0x9a, 0x20, 0xdb, 0x34, 0xef, 0xfc, 0x00, 0x00, 0x9e, 0x20, 0xaf, 0x17, 0x0d,
+ 0x3e, 0xde, 0x0f, 0x8f, 0xff, 0x80, 0x00, 0x00, 0xa2, 0x22, 0xbb, 0x27, 0x81,
+ 0xeb, 0xff, 0x5b, 0x07, 0xab, 0xff, 0xad, 0x9e, 0xd8, 0xc9, 0x6b, 0x75, 0x54,
+ 0xbf, 0xbe, 0x8a, 0xbd, 0xf2, 0xfb, 0xfb, 0x3d, 0x3d, 0x25, 0xb7, 0xf7, 0xfc,
+ 0x92, 0x4c, 0xdb, 0x6d, 0x69, 0xc0};
+
+ // Expected results
+ H263Info infoRefQCIF;
+ infoRefQCIF.uiH263PTypeFmt = 2;
+ infoRefQCIF.codecBits = 8;
+ infoRefQCIF.pQuant = 4;
+ infoRefQCIF.fType = 1;
+ infoRefQCIF.cpmBit = 0;
+ infoRefQCIF.numOfGOBs = 9;
+ infoRefQCIF.totalNumOfMBs = 11*9;
+
+ infoRefQCIF.ptrGOBbuffer[0] = 0; infoRefQCIF.ptrGOBbufferSBit[0] = 0; infoRefQCIF.ptrGQuant[0] = 0;
+ infoRefQCIF.ptrGOBbuffer[1] = 8; infoRefQCIF.ptrGOBbufferSBit[1] = 0; infoRefQCIF.ptrGQuant[1] = 4;
+ infoRefQCIF.ptrGOBbuffer[2] = 17; infoRefQCIF.ptrGOBbufferSBit[2] = 0; infoRefQCIF.ptrGQuant[2] = 4;
+ infoRefQCIF.ptrGOBbuffer[3] = 38; infoRefQCIF.ptrGOBbufferSBit[3] = 0; infoRefQCIF.ptrGQuant[3] = 4;
+ infoRefQCIF.ptrGOBbuffer[4] = 47; infoRefQCIF.ptrGOBbufferSBit[4] = 0; infoRefQCIF.ptrGQuant[4] = 4;
+ infoRefQCIF.ptrGOBbuffer[5] = 58; infoRefQCIF.ptrGOBbufferSBit[5] = 0; infoRefQCIF.ptrGQuant[5] = 4;
+ infoRefQCIF.ptrGOBbuffer[6] = 63; infoRefQCIF.ptrGOBbufferSBit[6] = 0; infoRefQCIF.ptrGQuant[6] = 4;
+ infoRefQCIF.ptrGOBbuffer[7] = 71; infoRefQCIF.ptrGOBbufferSBit[7] = 0; infoRefQCIF.ptrGQuant[7] = 4;
+ infoRefQCIF.ptrGOBbuffer[8] = 84; infoRefQCIF.ptrGOBbufferSBit[8] = 0; infoRefQCIF.ptrGQuant[8] = 4;
+
+ // ---------------------------------------------------
+ // Input data - H.263 encoded stream CIF (P-frame)
+ // ---------------------------------------------------
+ const int lengthRefCIF = 212;
+ const unsigned char encodedStreamCIF[lengthRefCIF] = {
+ 0x00, 0x00, 0x82, 0x9a, 0x0e, 0x04, 0x3f, 0xff, 0xff, 0x00, 0x00, 0x86, 0x27,
+ 0xff, 0xff, 0xe0, 0x00, 0x00, 0x8a, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0x8e,
+ 0x27, 0xff, 0xff, 0x6b, 0x09, 0x70, 0x00, 0x00, 0x92, 0x27, 0xff, 0xff, 0xe0,
+ 0x00, 0x00, 0x96, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0x9a, 0x27, 0x6f, 0x7f,
+ 0xff, 0xfe, 0x00, 0x00, 0x9e, 0x27, 0xff, 0xfe, 0xc6, 0x31, 0xe0, 0x00, 0x00,
+ 0xa2, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0xa6, 0x27, 0xf6, 0xb7, 0xfe, 0xa6,
+ 0x14, 0x95, 0xb4, 0xc6, 0x41, 0x6b, 0x3a, 0x2e, 0x8d, 0x42, 0xef, 0xc0, 0x00,
+ 0x00, 0xaa, 0x27, 0xff, 0xb1, 0x95, 0x05, 0x0c, 0xe3, 0x4a, 0x17, 0xff, 0x80,
+ 0x00, 0x00, 0xae, 0x27, 0xff, 0xf6, 0xf7, 0xfe, 0x00, 0x00, 0xb2, 0x27, 0xff,
+ 0x8b, 0xdf, 0xff, 0x00, 0x00, 0xb6, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0xba,
+ 0x26, 0x2f, 0x7f, 0xff, 0xfb, 0x58, 0x5b, 0x80, 0x00, 0x00, 0xbe, 0x20, 0xbc,
+ 0xe4, 0x5e, 0x6f, 0xff, 0xfe, 0xce, 0xf1, 0x94, 0x00, 0x00, 0xc2, 0x23, 0x18,
+ 0x4b, 0x17, 0x87, 0x0f, 0xff, 0xb3, 0xb6, 0x09, 0x63, 0x46, 0x73, 0x40, 0xb2,
+ 0x5f, 0x0a, 0xc6, 0xbe, 0xd7, 0x7a, 0x61, 0xbc, 0x68, 0xde, 0xf0, 0x00, 0x00,
+ 0xc6, 0x26, 0x2f, 0x0a, 0xff, 0xff, 0x17, 0x9a, 0x17, 0x8c, 0xb1, 0x37, 0x67,
+ 0xaf, 0xec, 0xf7, 0xa5, 0xbf, 0xb6, 0x49, 0x27, 0x6d, 0xb7, 0x92, 0x49, 0x1b,
+ 0xb4, 0x9a, 0xe0, 0x62};
+
+ // Expected results
+ H263Info infoRefCIF;
+ infoRefCIF.uiH263PTypeFmt = 3;
+ infoRefCIF.codecBits = 8;
+ infoRefCIF.pQuant = 4;
+ infoRefCIF.fType = 1;
+ infoRefCIF.cpmBit = 0;
+ infoRefCIF.numOfGOBs = 18;
+ infoRefCIF.totalNumOfMBs = 22*18;
+
+ infoRefCIF.ptrGOBbuffer[0] = 0; infoRefCIF.ptrGOBbufferSBit[0] = 0; infoRefCIF.ptrGQuant[0] = 0;
+ infoRefCIF.ptrGOBbuffer[1] = 9; infoRefCIF.ptrGOBbufferSBit[1] = 0; infoRefCIF.ptrGQuant[1] = 4;
+ infoRefCIF.ptrGOBbuffer[2] = 16; infoRefCIF.ptrGOBbufferSBit[2] = 0; infoRefCIF.ptrGQuant[2] = 4;
+ infoRefCIF.ptrGOBbuffer[3] = 23; infoRefCIF.ptrGOBbufferSBit[3] = 0; infoRefCIF.ptrGQuant[3] = 4;
+ infoRefCIF.ptrGOBbuffer[4] = 32; infoRefCIF.ptrGOBbufferSBit[4] = 0; infoRefCIF.ptrGQuant[4] = 4;
+ infoRefCIF.ptrGOBbuffer[5] = 39; infoRefCIF.ptrGOBbufferSBit[5] = 0; infoRefCIF.ptrGQuant[5] = 4;
+ infoRefCIF.ptrGOBbuffer[6] = 46; infoRefCIF.ptrGOBbufferSBit[6] = 0; infoRefCIF.ptrGQuant[6] = 4;
+ infoRefCIF.ptrGOBbuffer[7] = 54; infoRefCIF.ptrGOBbufferSBit[7] = 0; infoRefCIF.ptrGQuant[7] = 4;
+ infoRefCIF.ptrGOBbuffer[8] = 63; infoRefCIF.ptrGOBbufferSBit[8] = 0; infoRefCIF.ptrGQuant[8] = 4;
+ infoRefCIF.ptrGOBbuffer[9] = 70; infoRefCIF.ptrGOBbufferSBit[9] = 0; infoRefCIF.ptrGQuant[9] = 4;
+ infoRefCIF.ptrGOBbuffer[10] = 90; infoRefCIF.ptrGOBbufferSBit[10] = 0; infoRefCIF.ptrGQuant[10] = 4;
+ infoRefCIF.ptrGOBbuffer[11] = 104; infoRefCIF.ptrGOBbufferSBit[11] = 0; infoRefCIF.ptrGQuant[11] = 4;
+ infoRefCIF.ptrGOBbuffer[12] = 112; infoRefCIF.ptrGOBbufferSBit[12] = 0; infoRefCIF.ptrGQuant[12] = 4;
+ infoRefCIF.ptrGOBbuffer[13] = 120; infoRefCIF.ptrGOBbufferSBit[13] = 0; infoRefCIF.ptrGQuant[13] = 4;
+ infoRefCIF.ptrGOBbuffer[14] = 127; infoRefCIF.ptrGOBbufferSBit[14] = 0; infoRefCIF.ptrGQuant[14] = 4;
+ infoRefCIF.ptrGOBbuffer[15] = 138; infoRefCIF.ptrGOBbufferSBit[15] = 0; infoRefCIF.ptrGQuant[15] = 4;
+ infoRefCIF.ptrGOBbuffer[16] = 151; infoRefCIF.ptrGOBbufferSBit[16] = 0; infoRefCIF.ptrGQuant[16] = 4;
+ infoRefCIF.ptrGOBbuffer[17] = 180; infoRefCIF.ptrGOBbufferSBit[17] = 0; infoRefCIF.ptrGQuant[17] = 4;
+
+ // -----------------------------------------------------------------------
+ // Input data - H.263 encoded stream QCIF (I-frame). Non byte aligned GOBs
+ // -----------------------------------------------------------------------
+ const int lengthRefQCIF_N = 2020;
+ const unsigned char encodedStreamQCIF_N[lengthRefQCIF_N] = {
+ 0x00,0x00,0x80,0x06,0x08,0x07,0x34,0xe4,0xf7,0x66,0x75,0x12,0x9b,0x64,0x83,0xe9,0x4c,0xc7,0x3c,0x77,0x83,0xcf,0x67,0x96,
+ 0xe0,0x06,0x69,0x95,0x70,0x60,0x43,0x53,0x96,0x8a,0xa2,0x9e,0x96,0xf8,0x79,0xf0,0xf2,0xb8,0x30,0x21,0xb0,0x0c,0xc0,0x22,
+ 0x0e,0x70,0x12,0xb0,0x30,0x21,0x9e,0x03,0x3e,0x02,0x22,0xa8,0x83,0xcf,0x7d,0xe8,0xf3,0x85,0x8d,0x01,0x47,0x29,0x03,0x02,
+ 0x16,0x5c,0x06,0x7c,0x84,0x56,0x9c,0x0c,0x08,0x49,0x70,0x29,0xe1,0x68,0x39,0xbf,0xd0,0xef,0x6f,0x6e,0x8e,0x64,0x72,0x70,
+ 0x30,0x21,0x03,0xe0,0x53,0x82,0xab,0xe2,0xd4,0xa8,0x3e,0x12,0x80,0xc8,0x2a,0x9f,0x07,0x23,0xdf,0x4f,0xaa,0x5b,0x72,0xaa,
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+ 0xff,0x32,0x44,0x9a,0x53,0x5e,0xfb,0x0b,0xf2,0x6e,0x03,0x8b,0xfd,0xa3,0x6c,0x07,0x17,0xfb,0x1a,0xaa,0x76,0x60,0xf1,0xcf,
+ 0x10,0xff,0x5b,0x60,0x38,0xbf,0xdb,0x3b,0x17,0x18,0xe7,0x8f,0x1d,0xa4,0x3c,0x0c,0x9d,0x22,0x3e,0xd7,0x00,0xc9,0x63,0xf7,
+ 0x38,0xb8,0xc7,0x4c,0x78,0xed,0x31,0xe3,0x9d,0x22,0x3e,0xe7,0x17,0x19,0x6d,0xec,0x27,0xd0,0x8a,0xbc,0x3b,0xe3,0x05,0x2e,
+ 0xd2,0x51,0xe2,0x67,0x58,0x8f,0xb5,0xb0,0x97,0x63,0x61,0x2e,0x82,0x88,0x4d,0x9a,0xa3,0xc4,0xce,0x91,0x3f,0x57,0x61,0x2e,
+ 0x96,0xc2,0x5c,0x95,0x10,0x9b,0x13,0x21,0x33,0xa4,0x4f,0xcd,0x9f,0x84,0xd8,0x92,0xc5,0x86,0x77,0xd2,0x3a,0xc7,0xc2,0x32,
+ 0x38,0x2f,0x73,0x89};
+
+ // Expected results
+ H263Info infoRefQCIF_N;
+ infoRefQCIF_N.uiH263PTypeFmt = 2;
+ infoRefQCIF_N.codecBits = 0;
+ infoRefQCIF_N.pQuant = 7;
+ infoRefQCIF_N.fType = 0;
+ infoRefQCIF_N.cpmBit = 0;
+ infoRefQCIF_N.numOfGOBs = 9;
+ infoRefQCIF_N.totalNumOfMBs = 11*9;
+
+ infoRefQCIF_N.ptrGOBbuffer[0] = 0; infoRefQCIF_N.ptrGOBbufferSBit[0] = 0; infoRefQCIF_N.ptrGQuant[0] = 0;
+ infoRefQCIF_N.ptrGOBbuffer[1] = 215; infoRefQCIF_N.ptrGOBbufferSBit[1] = 2; infoRefQCIF_N.ptrGQuant[1] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[2] = 456; infoRefQCIF_N.ptrGOBbufferSBit[2] = 1; infoRefQCIF_N.ptrGQuant[2] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[3] = 535; infoRefQCIF_N.ptrGOBbufferSBit[3] = 5; infoRefQCIF_N.ptrGQuant[3] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[4] = 615; infoRefQCIF_N.ptrGOBbufferSBit[4] = 7; infoRefQCIF_N.ptrGQuant[4] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[5] = 925; infoRefQCIF_N.ptrGOBbufferSBit[5] = 4; infoRefQCIF_N.ptrGQuant[5] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[6] = 1133; infoRefQCIF_N.ptrGOBbufferSBit[6] = 1; infoRefQCIF_N.ptrGQuant[6] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[7] = 1512; infoRefQCIF_N.ptrGOBbufferSBit[7] = 6; infoRefQCIF_N.ptrGQuant[7] = 7;
+ infoRefQCIF_N.ptrGOBbuffer[8] = 1832; infoRefQCIF_N.ptrGOBbufferSBit[8] = 7; infoRefQCIF_N.ptrGQuant[8] = 7;
+
+ // --------------------------------------------------
+ // Input data - H.263 encoded stream CIF (I-frame)
+ // --------------------------------------------------
+
+ FILE* openFile = fopen("H263Foreman_CIF_Iframe.bin", "rb");
+
+ fseek(openFile, 0, SEEK_END);
+ int lengthRefCIF_I = ftell(openFile);
+ fseek(openFile, 0, SEEK_SET);
+
+ unsigned char* encodedStreamCIF_I = new unsigned char[lengthRefCIF_I];
+ fread(encodedStreamCIF_I, 1, lengthRefCIF_I, openFile);
+ fclose(openFile);
+
+ // Expected results
+ H263Info infoRefCIF_I;
+ infoRefCIF_I.uiH263PTypeFmt = 3;
+ infoRefCIF_I.codecBits = 0;
+ infoRefCIF_I.pQuant = 5;
+ infoRefCIF_I.fType = 0;
+ infoRefCIF_I.cpmBit = 0;
+ infoRefCIF_I.numOfGOBs = 18;
+ infoRefCIF_I.totalNumOfMBs = 22*18;
+
+ infoRefCIF_I.ptrGOBbuffer[0] = 0; infoRefCIF_I.ptrGOBbufferSBit[0] = 0; infoRefCIF_I.ptrGQuant[0] = 0;
+ infoRefCIF_I.ptrGOBbuffer[1] = 1607; infoRefCIF_I.ptrGOBbufferSBit[1] = 0; infoRefCIF_I.ptrGQuant[1] = 5;
+ infoRefCIF_I.ptrGOBbuffer[2] = 2759; infoRefCIF_I.ptrGOBbufferSBit[2] = 0; infoRefCIF_I.ptrGQuant[2] = 5;
+ infoRefCIF_I.ptrGOBbuffer[3] = 3699; infoRefCIF_I.ptrGOBbufferSBit[3] = 0; infoRefCIF_I.ptrGQuant[3] = 5;
+ infoRefCIF_I.ptrGOBbuffer[4] = 4506; infoRefCIF_I.ptrGOBbufferSBit[4] = 0; infoRefCIF_I.ptrGQuant[4] = 5;
+ infoRefCIF_I.ptrGOBbuffer[5] = 5260; infoRefCIF_I.ptrGOBbufferSBit[5] = 0; infoRefCIF_I.ptrGQuant[5] = 5;
+ infoRefCIF_I.ptrGOBbuffer[6] = 6254; infoRefCIF_I.ptrGOBbufferSBit[6] = 0; infoRefCIF_I.ptrGQuant[6] = 5;
+ infoRefCIF_I.ptrGOBbuffer[7] = 7117; infoRefCIF_I.ptrGOBbufferSBit[7] = 0; infoRefCIF_I.ptrGQuant[7] = 5;
+ infoRefCIF_I.ptrGOBbuffer[8] = 7804; infoRefCIF_I.ptrGOBbufferSBit[8] = 0; infoRefCIF_I.ptrGQuant[8] = 5;
+ infoRefCIF_I.ptrGOBbuffer[9] = 8600; infoRefCIF_I.ptrGOBbufferSBit[9] = 0; infoRefCIF_I.ptrGQuant[9] = 5;
+ infoRefCIF_I.ptrGOBbuffer[10] = 9419; infoRefCIF_I.ptrGOBbufferSBit[10] = 0; infoRefCIF_I.ptrGQuant[10] = 5;
+ infoRefCIF_I.ptrGOBbuffer[11] = 10126; infoRefCIF_I.ptrGOBbufferSBit[11] = 0; infoRefCIF_I.ptrGQuant[11] = 5;
+ infoRefCIF_I.ptrGOBbuffer[12] = 10723; infoRefCIF_I.ptrGOBbufferSBit[12] = 0; infoRefCIF_I.ptrGQuant[12] = 5;
+ infoRefCIF_I.ptrGOBbuffer[13] = 11280; infoRefCIF_I.ptrGOBbufferSBit[13] = 0; infoRefCIF_I.ptrGQuant[13] = 5;
+ infoRefCIF_I.ptrGOBbuffer[14] = 11910; infoRefCIF_I.ptrGOBbufferSBit[14] = 0; infoRefCIF_I.ptrGQuant[14] = 5;
+ infoRefCIF_I.ptrGOBbuffer[15] = 12430; infoRefCIF_I.ptrGOBbufferSBit[15] = 0; infoRefCIF_I.ptrGQuant[15] = 5;
+ infoRefCIF_I.ptrGOBbuffer[16] = 12925; infoRefCIF_I.ptrGOBbufferSBit[16] = 0; infoRefCIF_I.ptrGQuant[16] = 5;
+ infoRefCIF_I.ptrGOBbuffer[17] = 13506; infoRefCIF_I.ptrGOBbufferSBit[17] = 0; infoRefCIF_I.ptrGQuant[17] = 5;
+
+ // --------------------------------------------------
+ // Input data - H.263 encoded stream CIF (P-frame)
+ // --------------------------------------------------
+
+ openFile = fopen("H263Foreman_CIF_Pframe.bin", "rb");
+
+ fseek(openFile, 0, SEEK_END);
+ int lengthRefCIF_P = ftell(openFile);
+ fseek(openFile, 0, SEEK_SET);
+
+ unsigned char* encodedStreamCIF_P = new unsigned char[lengthRefCIF_P];
+ fread(encodedStreamCIF_P, 1, lengthRefCIF_P, openFile);
+ fclose(openFile);
+
+ // Expected results
+ H263Info infoRefCIF_P;
+ infoRefCIF_P.uiH263PTypeFmt = 3;
+ infoRefCIF_P.codecBits = 8;
+ infoRefCIF_P.pQuant = 4;
+ infoRefCIF_P.fType = 1;
+ infoRefCIF_P.cpmBit = 0;
+ infoRefCIF_P.numOfGOBs = 18;
+ infoRefCIF_P.totalNumOfMBs = 22*18;
+
+ infoRefCIF_P.ptrGOBbuffer[0] = 0; infoRefCIF_P.ptrGOBbufferSBit[0] = 0; infoRefCIF_P.ptrGQuant[0] = 0;
+ infoRefCIF_P.ptrGOBbuffer[1] = 252; infoRefCIF_P.ptrGOBbufferSBit[1] = 0; infoRefCIF_P.ptrGQuant[1] = 5;
+ infoRefCIF_P.ptrGOBbuffer[2] = 482; infoRefCIF_P.ptrGOBbufferSBit[2] = 0; infoRefCIF_P.ptrGQuant[2] = 6;
+ infoRefCIF_P.ptrGOBbuffer[3] = 581; infoRefCIF_P.ptrGOBbufferSBit[3] = 0; infoRefCIF_P.ptrGQuant[3] = 6;
+ infoRefCIF_P.ptrGOBbuffer[4] = 676; infoRefCIF_P.ptrGOBbufferSBit[4] = 0; infoRefCIF_P.ptrGQuant[4] = 7;
+ infoRefCIF_P.ptrGOBbuffer[5] = 756; infoRefCIF_P.ptrGOBbufferSBit[5] = 0; infoRefCIF_P.ptrGQuant[5] = 7;
+ infoRefCIF_P.ptrGOBbuffer[6] = 855; infoRefCIF_P.ptrGOBbufferSBit[6] = 0; infoRefCIF_P.ptrGQuant[6] = 8;
+ infoRefCIF_P.ptrGOBbuffer[7] = 949; infoRefCIF_P.ptrGOBbufferSBit[7] = 0; infoRefCIF_P.ptrGQuant[7] = 9;
+ infoRefCIF_P.ptrGOBbuffer[8] = 1004; infoRefCIF_P.ptrGOBbufferSBit[8] = 0; infoRefCIF_P.ptrGQuant[8] = 10;
+ infoRefCIF_P.ptrGOBbuffer[9] = 1062; infoRefCIF_P.ptrGOBbufferSBit[9] = 0; infoRefCIF_P.ptrGQuant[9] = 11;
+ infoRefCIF_P.ptrGOBbuffer[10] = 1115; infoRefCIF_P.ptrGOBbufferSBit[10] = 0; infoRefCIF_P.ptrGQuant[10] = 11;
+ infoRefCIF_P.ptrGOBbuffer[11] = 1152; infoRefCIF_P.ptrGOBbufferSBit[11] = 0; infoRefCIF_P.ptrGQuant[11] = 13;
+ infoRefCIF_P.ptrGOBbuffer[12] = 1183; infoRefCIF_P.ptrGOBbufferSBit[12] = 0; infoRefCIF_P.ptrGQuant[12] = 14;
+ infoRefCIF_P.ptrGOBbuffer[13] = 1214; infoRefCIF_P.ptrGOBbufferSBit[13] = 0; infoRefCIF_P.ptrGQuant[13] = 15;
+ infoRefCIF_P.ptrGOBbuffer[14] = 1257; infoRefCIF_P.ptrGOBbufferSBit[14] = 0; infoRefCIF_P.ptrGQuant[14] = 16;
+ infoRefCIF_P.ptrGOBbuffer[15] = 1286; infoRefCIF_P.ptrGOBbufferSBit[15] = 0; infoRefCIF_P.ptrGQuant[15] = 16;
+ infoRefCIF_P.ptrGOBbuffer[16] = 1321; infoRefCIF_P.ptrGOBbufferSBit[16] = 0; infoRefCIF_P.ptrGQuant[16] = 16;
+ infoRefCIF_P.ptrGOBbuffer[17] = 1352; infoRefCIF_P.ptrGOBbufferSBit[17] = 0; infoRefCIF_P.ptrGQuant[17] = 14;
+
+ //---------------------------------------------------------------
+ //---------------------------------------------------------------
+ //---------------------------------------------------------------
+ // Start test
+ const H263Info* ptrInfoSQCIF = NULL;
+ const H263MBInfo* ptrMBInfoSQCIF = NULL;
+ const H263Info* ptrInfoQCIF = NULL;
+ const H263MBInfo* ptrMBInfoQCIF = NULL;
+ const H263Info* ptrInfoCIF = NULL;
+ const H263MBInfo* ptrMBInfoCIF = NULL;
+ const H263Info* ptrInfoQCIF_N = NULL;
+ const H263MBInfo* ptrMBInfoQCIF_N = NULL;
+ const H263Info* ptrInfoCIF_I = NULL;
+ const H263MBInfo* ptrMBInfoCIF_I = NULL;
+ const H263Info* ptrInfoCIF_P = NULL;
+ const H263MBInfo* ptrMBInfoCIF_P = NULL;
+ H263Information h263Information;
+
+ // Input buffer
+ const int length = 3000;
+ unsigned char* encodedBuffer = new unsigned char[lengthRefCIF_P];
+
+ // Test invalid inputs
+ assert(-1 == h263Information.GetInfo(NULL, length, ptrInfoSQCIF));
+ assert(-1 == h263Information.GetInfo(encodedBuffer, 0, ptrInfoSQCIF));
+ assert(-1 == h263Information.GetInfo(encodedBuffer, length, ptrInfoSQCIF)); // invalid H.263 stream
+// assert(-1 == h263Information.GetInfo(encodedStreamSQCIF, lengthRefSQCIF/2, ptrInfoSQCIF)); // invalid H.263 stream
+
+ assert(-1 == h263Information.GetMBInfo(NULL, length, 0, ptrMBInfoSQCIF));
+ assert(-1 == h263Information.GetMBInfo(encodedBuffer, 0, 0, ptrMBInfoSQCIF));
+ assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, 0, ptrMBInfoSQCIF));
+ assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, -1, ptrMBInfoSQCIF)); // incorrect group number
+ assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, 8, ptrMBInfoSQCIF)); // incorrect group number
+
+ // ----------------------------------------------
+ // Get info from encoded H.263 stream - SQCIF
+ // ----------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamSQCIF, lengthRefSQCIF, ptrInfoSQCIF));
+ ValidateResults(ptrInfoSQCIF, &infoRefSQCIF);
+
+ // Get MB info
+ for (int i = 0; i < ptrInfoSQCIF->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamSQCIF, lengthRefSQCIF, i, ptrMBInfoSQCIF));
+ }
+ ValidateMBResults(ptrInfoSQCIF, ptrMBInfoSQCIF, &infoRefSQCIF);
+
+ // ---------------------------------------------
+ // Get info from encoded H.263 stream - QCIF
+ // ---------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamQCIF, lengthRefQCIF, ptrInfoQCIF));
+ ValidateResults(ptrInfoQCIF, &infoRefQCIF);
+
+ // Get MB info
+ for (int i = 0; i < ptrInfoQCIF->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamQCIF, lengthRefQCIF, i, ptrMBInfoQCIF));
+ }
+ ValidateMBResults(ptrInfoQCIF, ptrMBInfoQCIF, &infoRefQCIF);
+
+ // --------------------------------------------
+ // Get info from encoded H.263 stream - CIF
+ // --------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamCIF, lengthRefCIF, ptrInfoCIF));
+ ValidateResults(ptrInfoCIF, &infoRefCIF);
+
+ // Get MB info
+ for (int i = 0; i < ptrInfoCIF->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamCIF, lengthRefCIF, i, ptrMBInfoCIF));
+ }
+ ValidateMBResults(ptrInfoCIF, ptrMBInfoCIF, &infoRefCIF);
+
+ // ----------------------------------------------------------------------
+ // Get info from encoded H.263 stream - QCIF - non byte aligned GOBs
+ // ----------------------------------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamQCIF_N, lengthRefQCIF_N, ptrInfoQCIF_N));
+ ValidateResults(ptrInfoQCIF_N, &infoRefQCIF_N);
+
+ // Get MB info
+ for (int i = 0; i < ptrInfoQCIF_N->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamQCIF_N, lengthRefQCIF_N, i, ptrMBInfoQCIF_N));
+ }
+ ValidateMBResults(ptrInfoQCIF_N, ptrMBInfoQCIF_N, &infoRefQCIF_N);
+
+ // -------------------------------------------
+ // Get info from encoded H.263 stream - CIF
+ // -------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamCIF_I, lengthRefCIF_I, ptrInfoCIF_I));
+ ValidateResults(ptrInfoCIF_I, &infoRefCIF_I);
+
+ // Get MB info
+ unsigned int start = timeGetTime();
+ for (int i = 0; i < ptrInfoCIF_I->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamCIF_I, lengthRefCIF_I, i, ptrMBInfoCIF_I));
+ }
+ unsigned int endMB = timeGetTime();
+ ValidateMBResults(ptrInfoCIF_I, ptrMBInfoCIF_I, &infoRefCIF_I, false);
+
+ std::cout << "I-frame, length: " << lengthRefCIF_I << " bytes. Time: " << endMB - start << " ms." << std::endl;
+ PRINT_LINE;
+
+ // -------------------------------------------
+ // Get info from encoded H.263 stream - CIF
+ // -------------------------------------------
+ h263Information.Reset();
+ assert(0 == h263Information.GetInfo(encodedStreamCIF_P, lengthRefCIF_P, ptrInfoCIF_P));
+ ValidateResults(ptrInfoCIF_P, &infoRefCIF_P);
+
+ // Get MB info
+ start = timeGetTime();
+ for (int i = 0; i < ptrInfoCIF_P->numOfGOBs; i++)
+ {
+ assert(0 == h263Information.GetMBInfo(encodedStreamCIF_P, lengthRefCIF_P, i, ptrMBInfoCIF_P));
+ }
+ endMB = timeGetTime();
+ ValidateMBResults(ptrInfoCIF_P, ptrMBInfoCIF_P, &infoRefCIF_P, false);
+
+ std::cout << "P-frame, length: " << lengthRefCIF_P << " bytes. Time: " << endMB - start << " ms." << std::endl;
+ PRINT_LINE;
+
+ delete [] encodedStreamCIF_I;
+ delete [] encodedStreamCIF_P;
+ delete [] encodedBuffer;
+
+ TEST_PASSED();
+ ::Sleep(5000);
+ return 0;
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc b/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc
new file mode 100644
index 0000000..c282557
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <windows.h>
+#include <iostream>
+#include <tchar.h>
+
+#include "rtp_rtcp.h"
+#include "common_types.h"
+#include "RateControlDetector.h"
+/*#include "rtcp_utility.h"
+#include "tmmbr_help.h"*/
+
+#define TEST_STR "Test RateControl."
+#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
+#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
+
+
+const int maxFileLen = 200;
+WebRtc_UWord8* dataFile[maxFileLen];
+
+
+struct InputSet
+{
+ WebRtc_UWord32 TMMBR;
+ WebRtc_UWord32 packetOH;
+ WebRtc_UWord32 SSRC;
+};
+
+const InputSet set0 = {220, 80, 11111}; // bitRate, packetOH, ssrc
+const InputSet set1 = {180, 90, 22222};
+const InputSet set2 = {100, 210, 33333};
+const InputSet set3 = { 35, 40, 44444};
+const InputSet set4 = { 40, 60, 55555};
+const InputSet set4_1 = {100, 60, 55555};
+const InputSet set4_2 = { 10, 60, 55555};
+const InputSet set5 = {200, 40, 66666};
+const InputSet set00 = { 0, 40, 66666};
+
+
+
+
+WebRtc_Word32 GetFile(char* fileName)
+{
+ if (!fileName[0])
+ {
+ return 0;
+ }
+
+ FILE* openFile = fopen(fileName, "rb");
+ assert(openFile != NULL);
+ fseek(openFile, 0, SEEK_END);
+ int len = (WebRtc_Word16)(ftell(openFile));
+ rewind(openFile);
+ assert(len > 0 && len < maxFileLen);
+ fread(dataFile, 1, len, openFile);
+ fclose(openFile);
+ return len;
+};
+
+
+class LoopBackTransport2 : public webrtc::Transport
+{
+public:
+ LoopBackTransport2(RtpRtcp* rtpRtcpModule) :
+ _rtpRtcpModule(rtpRtcpModule),
+ _cnt(0)
+ {
+ }
+ virtual int SendPacket(int channel, const void *data, int len)
+ {
+ return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len)
+ {
+ char fileName[256] = {0};
+
+
+ // Get stored rtcp packet w/ TMMBR
+ len = GetFile(fileName);
+ if (len == 0)
+ {
+ return 0;
+ }
+
+ // Send in bitrate request
+ return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
+ }
+ RtpRtcp* _rtpRtcpModule;
+ WebRtc_UWord32 _cnt;
+};
+
+
+class LoopBackTransportVideo : public webrtc::Transport
+{
+public:
+ LoopBackTransportVideo(RtpRtcp* rtpRtcpModule) :
+ _rtpRtcpModule(rtpRtcpModule),
+ _cnt(0)
+ {
+ }
+ virtual int SendPacket(int channel, const void *data, int len)
+ {
+ return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len)
+ {
+ char fileName[256] = {0};
+
+ strcpy(fileName, "RTCPPacketTMMBR0.bin");
+
+ ++_cnt;
+
+ // Get stored rtcp packet w/ TMMBR
+ len = GetFile(fileName);
+ if (len == 0)
+ {
+ return 0;
+ }
+
+ // Send in bitrate request*/
+ return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
+ }
+
+ RtpRtcp* _rtpRtcpModule;
+ WebRtc_UWord32 _cnt;
+};
+
+class TestRateControl : private RateControlDetector
+{
+public:
+ TestRateControl():RateControlDetector(0)
+ {
+ }
+ ~TestRateControl()
+ {
+ }
+ void Start()
+ {
+ //Test perfect conditions
+ // But only one packet per frame
+ SetLastUsedBitRate(500);
+ WebRtc_UWord32 rtpTs=1234*90;
+ WebRtc_UWord32 framePeriod=33; // In Ms
+ WebRtc_UWord32 rtpDelta=framePeriod*90;
+ WebRtc_UWord32 netWorkDelay=10;
+ WebRtc_UWord32 arrivalTime=rtpTs/90+netWorkDelay;
+ WebRtc_UWord32 newBitRate=0;
+ for(WebRtc_UWord32 k=0;k<10;k++)
+ {
+ // Receive 10 packets
+ for(WebRtc_UWord32 i=0;i<10;i++)
+ {
+ NotifyNewArrivedPacket(rtpTs,arrivalTime);
+ rtpTs+=rtpDelta;
+ arrivalTime=rtpTs/90+netWorkDelay;
+ }
+ newBitRate=RateControl(2*netWorkDelay);
+ SetLastUsedBitRate(newBitRate);
+ Sleep(10*framePeriod);
+ std::cout << "RTCP Packet " << k << " new bitrate " << newBitRate << std::endl;
+ }
+ Reset();
+
+
+ //Test increasing RTT
+ std::cout << "Test increasing RTT - No Receive timing changes" << std::endl;
+ SetLastUsedBitRate(500);
+
+ for(WebRtc_UWord32 k=0;k<10;k++)
+ {
+ // Receive 10 packets
+ for(WebRtc_UWord32 i=0;i<10;i++)
+ {
+ NotifyNewArrivedPacket(rtpTs,arrivalTime);
+ rtpTs+=rtpDelta;
+ arrivalTime=rtpTs/90+netWorkDelay;
+ }
+ WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
+ newBitRate=RateControl(rtt);
+ Sleep(10*framePeriod);
+ SetLastUsedBitRate(newBitRate);
+ std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
+
+ }
+
+ Reset();
+
+
+ //Test increasing RTT
+ std::cout << "Test increasing RTT - Changed receive timing" << std::endl;
+ SetLastUsedBitRate(500);
+
+ for(WebRtc_UWord32 k=0;k<10;k++)
+ {
+ // Receive 10 packets
+ for(WebRtc_UWord32 i=0;i<10;i++)
+ {
+ NotifyNewArrivedPacket(rtpTs,arrivalTime);
+ rtpTs+=rtpDelta;
+ arrivalTime=rtpTs/90+netWorkDelay+i+(k*20);
+ }
+ WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
+ newBitRate=RateControl(rtt);
+ Sleep(10*framePeriod);
+ SetLastUsedBitRate(newBitRate);
+ std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
+
+ }
+
+
+
+ };
+};
+
+class NULLDataZink: public RtpData
+{
+ virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader)
+ {
+ return 0;
+ };
+};
+
+
+int _tmain(int argc, _TCHAR* argv[])
+{
+
+ std::string str;
+ std::cout << "------------------------" << std::endl;
+ std::cout << "---Test RateControl ----" << std::endl;
+ std::cout << "------------------------" << std::endl;
+ std::cout << " " << std::endl;
+
+ // --------------------
+ // Test TMMBRHelp class
+
+ // --------------------
+ TestRateControl test;
+ test.Start();
+
+ printf("RateControl-class test done.\n");
+
+ // ------------------------
+ // Test RateControl single module
+ // ------------------------
+ RtpRtcp* rtpRtcpModuleVideo = RtpRtcp::CreateRtpRtcp(0, false);
+
+ LoopBackTransportVideo* myLoopBackTransportVideo = new LoopBackTransportVideo(rtpRtcpModuleVideo);
+ assert(0 == rtpRtcpModuleVideo->RegisterSendTransport(myLoopBackTransportVideo));
+ printf("Multi module test done.\n");
+
+
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo);
+ delete myLoopBackTransportVideo;
+
+ TEST_PASSED();
+ ::Sleep(5000);
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc b/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc
new file mode 100644
index 0000000..d1e1572
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc
@@ -0,0 +1,1034 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <windows.h>
+#include <iostream>
+#include <tchar.h>
+
+#include "rtp_rtcp.h"
+#include "common_types.h"
+#include "rtcp_utility.h"
+#include "tmmbr_help.h"
+
+#define TEST_STR "Test TMMBR."
+#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
+#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
+
+
+const int maxFileLen = 200;
+WebRtc_UWord8* dataFile[maxFileLen];
+
+
+struct InputSet
+{
+ WebRtc_UWord32 TMMBR;
+ WebRtc_UWord32 packetOH;
+ WebRtc_UWord32 SSRC;
+};
+
+const InputSet set0 = {220, 80, 11111}; // bitRate, packetOH, ssrc
+const InputSet set1 = {180, 90, 22222};
+const InputSet set2 = {100, 210, 33333};
+const InputSet set3 = { 35, 40, 44444};
+const InputSet set4 = { 40, 60, 55555};
+const InputSet set4_1 = {100, 60, 55555};
+const InputSet set4_2 = { 10, 60, 55555};
+const InputSet set5 = {200, 40, 66666};
+const InputSet set00 = { 0, 40, 66666};
+
+const int maxBitrate = 230; // if this is lower than max in the list above test should fail
+
+void Verify(TMMBRSet* boundingSet, int index, InputSet set)
+{
+ assert(boundingSet->ptrTmmbrSet[index] == set.TMMBR);
+ assert(boundingSet->ptrPacketOHSet[index] == set.packetOH);
+ assert(boundingSet->ptrSsrcSet[index] == set.SSRC);
+};
+
+int ParseRTCPPacket(const void *data, int len, TMMBRSet*& boundingSet)
+{
+ int numItems = -1;
+ RTCPUtility::RTCPParserV2 rtcpParser((const WebRtc_UWord8*)data, len, true);
+ RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Begin();
+ while (pktType != RTCPUtility::kRtcpNotValidCode)
+ {
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+ if (pktType == RTCPUtility::kRtcpRtpfbTmmbnCode)
+ {
+ assert(0 == rtcpPacket.TMMBN.SenderSSRC);
+ assert(0 == rtcpPacket.TMMBN.MediaSSRC);
+ numItems = 0;
+ }
+ if (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode)
+ {
+ boundingSet->ptrTmmbrSet[numItems] = rtcpPacket.TMMBNItem.MaxTotalMediaBitRate;
+ boundingSet->ptrPacketOHSet[numItems] = rtcpPacket.TMMBNItem.MeasuredOverhead;
+ boundingSet->ptrSsrcSet[numItems] = rtcpPacket.TMMBNItem.SSRC;
+ ++numItems;
+ }
+ pktType = rtcpParser.Iterate();
+ }
+ return numItems;
+};
+
+WebRtc_Word32 GetFile(char* fileName)
+{
+ if (!fileName[0])
+ {
+ return 0;
+ }
+
+ FILE* openFile = fopen(fileName, "rb");
+ assert(openFile != NULL);
+ fseek(openFile, 0, SEEK_END);
+ int len = (WebRtc_Word16)(ftell(openFile));
+ rewind(openFile);
+ assert(len > 0 && len < maxFileLen);
+ fread(dataFile, 1, len, openFile);
+ fclose(openFile);
+ return len;
+};
+
+
+class LoopBackTransport2 : public webrtc::Transport, private TMMBRHelp
+{
+public:
+ LoopBackTransport2(RtpRtcp* rtpRtcpModule) :
+ TMMBRHelp(false),
+ _rtpRtcpModule(rtpRtcpModule),
+ _cnt(0)
+ {
+ }
+ virtual int SendPacket(int channel, const void *data, int len)
+ {
+ if( 0 == _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len))
+ {
+ return len;
+ }
+ return -1;
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len)
+ {
+ char fileName[256] = {0};
+ TMMBRSet* boundingSet = BoundingSet();
+ boundingSet->VerifyAndAllocateSet(3);
+
+ if (_cnt == 0)
+ {
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {2,4,0} -> {4,2}
+ assert(2 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set4);
+ Verify(boundingSet, 1, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR3.bin");
+ }
+
+ ++_cnt;
+
+ // Get stored rtcp packet w/ TMMBR
+ len = GetFile(fileName);
+ if (len == 0)
+ {
+ return 1;
+ }
+
+ // Send in bitrate request
+ if(_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len) == 0)
+ {
+ return len;
+ }
+ return -1;
+ }
+ RtpRtcp* _rtpRtcpModule;
+ WebRtc_UWord32 _cnt;
+};
+
+
+class LoopBackTransportVideo : public webrtc::Transport, private TMMBRHelp
+{
+public:
+ LoopBackTransportVideo(RtpRtcp* rtpRtcpModule) :
+ TMMBRHelp(false),
+ _rtpRtcpModule(rtpRtcpModule),
+ _cnt(0)
+ {
+ }
+ virtual int SendPacket(int channel, const void *data, int len)
+ {
+ if(_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len)== 0)
+ {
+ return len;
+ }
+ return -1;
+ }
+ virtual int SendRTCPPacket(int channel, const void *data, int len)
+ {
+ char fileName[256] = {0};
+ TMMBRSet* boundingSet = BoundingSet();
+ boundingSet->VerifyAndAllocateSet(3);
+
+ if (_cnt == 0)
+ {
+ strcpy(fileName, "RTCPPacketTMMBR0.bin");
+ }
+ else if (_cnt == 1)
+ {
+ // TMMBN {0} -> {0}
+ assert(1 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set0);
+
+ strcpy(fileName, "RTCPPacketTMMBR1.bin");
+ }
+ else if (_cnt == 2)
+ {
+ // TMMBN {0,1} -> {1}
+ assert(1 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set1);
+
+ strcpy(fileName, "RTCPPacketTMMBR2.bin");
+ }
+ else if (_cnt == 3)
+ {
+ // TMMBN {0,1,2} -> {2}
+ assert(1 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR3.bin");
+ }
+ else if (_cnt == 4)
+ {
+ // TMMBN {0,1,2,3} -> {3,2}
+ assert(2 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR4.bin");
+ }
+ else if (_cnt == 5)
+ {
+ // TMMBN {0,1,2,3,4} -> {3,4,2}
+ assert(3 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR5.bin");
+ }
+ else if (_cnt == 6)
+ {
+ // TMMBN {0,1,2,3,4,5} -> {3,4,2}
+ assert(3 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR4_2.bin");
+ }
+ else if (_cnt == 7)
+ {
+ // TMMBN {0,1,2,3,4_2,5} -> {4_2}
+ assert(1 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set4_2);
+
+ ++_cnt;
+ ::Sleep(5*RTCP_INTERVAL_AUDIO_MS + 1000); // time out receiver
+ _rtpRtcpModule->Process(); // SendRTCP() (_cnt == 8)
+ // a receiver has timed out -> UpdateTMMBR()
+ }
+ else if (_cnt == 8)
+ {
+ // No TMMBN in this packet
+ assert(-1 == ParseRTCPPacket(data, len, boundingSet));
+ }
+ else if (_cnt == 10)
+ {
+ // TMMBN {} -> {}, empty set
+ assert(0 == ParseRTCPPacket(data, len, boundingSet));
+
+ strcpy(fileName, "RTCPPacketTMMBR2.bin");
+ }
+ else if (_cnt == 11)
+ {
+ // TMMBN {2} -> {2}
+ assert(1 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set2);
+ }
+ else if (_cnt == 12) // ----- multi module -------------
+ {
+ // No TMMBN in this packet
+ assert(-1 == ParseRTCPPacket(data, len, boundingSet));
+
+ strcpy(fileName, "RTCPPacketTMMBR4.bin");
+ }
+ else if (_cnt == 13)
+ {
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {2,4} -> {4,2}
+ assert(2 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set4);
+ Verify(boundingSet, 1, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR0.bin");
+ }
+ else if (_cnt == 14)
+ {
+ // TMMBN {}
+ // TMMBN {3}
+ // TMMBN {}
+ // TMMBN {2,4,0} -> {3,4,2}
+ assert(3 == ParseRTCPPacket(data, len, boundingSet));
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ strcpy(fileName, "RTCPPacketTMMBR1.bin");
+ }
+ //else if (_cnt == 15)
+ //{
+ // // TMMBN {}
+ // // TMMBN {}
+ // // TMMBN {}
+ // // TMMBN {2,4,0,1} -> {4,2}
+ // //assert(2 == ParseRTCPPacket(data, len, boundingSet));
+ // //Verify(boundingSet, 0, set4);
+ // //Verify(boundingSet, 1, set2);
+ //}
+ //else if (_cnt == 15)
+ //{
+ // // No TMMBN in this packet
+ // assert(-1 == ParseRTCPPacket(data, len, boundingSet));
+ //}
+ else if (_cnt == 15)
+ {
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {}
+ // TMMBN {} -> {}, empty set
+ assert(0 == ParseRTCPPacket(data, len, boundingSet));
+ }
+
+ ++_cnt;
+
+ // Get stored rtcp packet w/ TMMBR
+ len = GetFile(fileName);
+ if (len == 0)
+ {
+ return 1;
+ }
+
+ // Send in bitrate request
+ if( 0 == _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len))
+ {
+ return len;
+ }
+ return -1;
+ }
+
+ RtpRtcp* _rtpRtcpModule;
+ WebRtc_UWord32 _cnt;
+};
+
+class TestTMMBR : private TMMBRHelp
+{
+public:
+ TestTMMBR() : TMMBRHelp(false) {};
+
+ void Add(TMMBRSet* candidateSet, int index, InputSet set)
+ {
+ candidateSet->ptrTmmbrSet[index] = set.TMMBR;
+ candidateSet->ptrPacketOHSet[index] = set.packetOH;
+ candidateSet->ptrSsrcSet[index] = set.SSRC;
+ };
+
+ void Start()
+ {
+ // Get sets
+ TMMBRSet* candidateSet = CandidateSet();
+ assert(0 == candidateSet->sizeOfSet);
+ TMMBRSet* boundingSet = BoundingSet();
+ assert(0 == boundingSet->sizeOfSet);
+ TMMBRSet* boundingSetToSend = BoundingSetToSend();
+ assert(0 == boundingSetToSend->sizeOfSet);
+
+ WebRtc_Word32 numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(0 == numBoundingSet); // should be empty
+
+ assert( 0 == SetTMMBRBoundingSetToSend(NULL,0)); // ok to send empty set
+ assert( 0 == SetTMMBRBoundingSetToSend(boundingSet,0)); // ok to send empty set
+
+ WebRtc_UWord32 minBitrateKbit = 0;
+ WebRtc_UWord32 maxBitrateKbit = 0;
+ assert(-1 == CalcMinMaxBitRate(0, 0, 1, false, minBitrateKbit, maxBitrateKbit)); // no bounding set
+
+ // ---------------------------------
+ // Test candidate set {0} -> {0}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(1);
+ assert(1 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(1 == numBoundingSet);
+ Verify(boundingSet, 0, set0);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, 0)); // incorrect length
+ assert(!IsOwner(set1.SSRC, 100)); // incorrect length
+
+ assert( IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set0);
+
+ // Get net bitrate depending on packet rate
+ assert( 0 == CalcMinMaxBitRate(0, numBoundingSet, false,0, minBitrateKbit, maxBitrateKbit));
+ assert(set0.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+ assert(0 == CalcMinMaxBitRate(0, 100, false,0, minBitrateKbit, maxBitrateKbit)); // incorrect length
+ assert(set0.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {0,1} -> {1}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(2);
+ assert(2 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+ Add(candidateSet, 1, set1);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(1 == numBoundingSet);
+ Verify(boundingSet, 0, set1);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert( IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set1);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set1.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {0,1,2} -> {2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(3);
+ assert(3 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+ Add(candidateSet, 1, set1);
+ Add(candidateSet, 2, set2);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(1 == numBoundingSet);
+ Verify(boundingSet, 0, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set2.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {0,1,2,3} -> {3,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(4);
+ assert(4 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+ Add(candidateSet, 1, set1);
+ Add(candidateSet, 2, set2);
+ Add(candidateSet, 3, set3);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {0,1,2,3,4} -> {3,4,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(5);
+ assert(5 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+ Add(candidateSet, 1, set1);
+ Add(candidateSet, 2, set2);
+ Add(candidateSet, 3, set3);
+ Add(candidateSet, 4, set4);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(3 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set4);
+ Verify(boundingSetToSend, 2, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {0,1,2,3,4,5} -> {3,4,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(6);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set0);
+ Add(candidateSet, 1, set1);
+ Add(candidateSet, 2, set2);
+ Add(candidateSet, 3, set3);
+ Add(candidateSet, 4, set4);
+ Add(candidateSet, 5, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(3 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set4);
+ Verify(boundingSetToSend, 2, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set0.TMMBR == maxBitrateKbit);
+
+
+ // ---------------------------------
+ // Test candidate set {1,2,3,4,5} -> {3,4,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(5);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set2);
+ Add(candidateSet, 2, set3);
+ Add(candidateSet, 3, set4);
+ Add(candidateSet, 4, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(3 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+ Verify(boundingSet, 2, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ assert(boundingSetToSend->sizeOfSet == numBoundingSet);
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set4);
+ Verify(boundingSetToSend, 2, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+
+ // ---------------------------------
+ // Test candidate set {1,3,4,5} -> {3,4}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(4);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set3);
+ Add(candidateSet, 2, set4);
+ Add(candidateSet, 3, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set4);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set4);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet,true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {1,2,4,5} -> {4,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(4);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set2);
+ Add(candidateSet, 2, set4);
+ Add(candidateSet, 3, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set4);
+ Verify(boundingSet, 1, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert(!IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set4);
+ Verify(boundingSetToSend, 1, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set4.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {1,2,3,5} -> {3,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(4);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set2);
+ Add(candidateSet, 2, set3);
+ Add(candidateSet, 3, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert(!IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {1,2,3,4_1,5} -> {3,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(5);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set2);
+ Add(candidateSet, 2, set3);
+ Add(candidateSet, 3, set4_1);
+ Add(candidateSet, 4, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set3);
+ Verify(boundingSet, 1, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert( IsOwner(set3.SSRC, numBoundingSet));
+ assert(!IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set3);
+ Verify(boundingSetToSend, 1, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set3.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {1,2,3,4_2,5} -> {4_2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(5);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set1);
+ Add(candidateSet, 1, set2);
+ Add(candidateSet, 2, set3);
+ Add(candidateSet, 3, set4_2);
+ Add(candidateSet, 4, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(1 == numBoundingSet);
+ Verify(boundingSet, 0, set4_2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+ assert(!IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set4_2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(MIN_VIDEO_BW_MANAGEMENT_BITRATE == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {} -> {}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(0);
+ assert(6 == candidateSet->sizeOfSet);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(0 == numBoundingSet);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+ assert(!IsOwner(set3.SSRC, numBoundingSet));
+ assert(!IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+
+ // Get net bitrate depending on packet rate
+ assert(-1 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+
+ // ---------------------------------
+ // Test candidate set {x0,5} -> {5}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(2);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set00);
+ Add(candidateSet, 1, set5);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(1 == numBoundingSet);
+ Verify(boundingSet, 0, set5);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert(!IsOwner(set2.SSRC, numBoundingSet));
+ assert(!IsOwner(set3.SSRC, numBoundingSet));
+ assert(!IsOwner(set4.SSRC, numBoundingSet));
+ assert( IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set5);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set5.TMMBR == minBitrateKbit);
+ assert(set5.TMMBR == maxBitrateKbit);
+
+ // ---------------------------------
+ // Test candidate set {x0,4,2} -> {4,2}
+ // ---------------------------------
+ candidateSet = VerifyAndAllocateCandidateSet(3);
+ assert(6 == candidateSet->sizeOfSet);
+ Add(candidateSet, 0, set00);
+ Add(candidateSet, 1, set4);
+ Add(candidateSet, 2, set2);
+
+ // Find bounding set
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ assert(2 == numBoundingSet);
+ Verify(boundingSet, 0, set4);
+ Verify(boundingSet, 1, set2);
+
+ // Is owner of set
+ assert(!IsOwner(set0.SSRC, numBoundingSet));
+ assert(!IsOwner(set1.SSRC, numBoundingSet));
+ assert( IsOwner(set2.SSRC, numBoundingSet));
+ assert(!IsOwner(set3.SSRC, numBoundingSet));
+ assert( IsOwner(set4.SSRC, numBoundingSet));
+ assert(!IsOwner(set5.SSRC, numBoundingSet));
+
+ // Set boundingSet to send
+ assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
+
+ // Get boundingSet to send
+ boundingSetToSend = BoundingSetToSend();
+ Verify(boundingSetToSend, 0, set4);
+ Verify(boundingSetToSend, 1, set2);
+
+ // Get net bitrate depending on packet rate
+ assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
+ assert(set4.TMMBR == minBitrateKbit);
+ assert(set2.TMMBR == maxBitrateKbit);
+ };
+};
+
+class NULLDataZink: public RtpData
+{
+ virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* incomingRtpPacket,
+ const WebRtc_UWord16 incomingRtpPacketLengt)
+ {
+ return 0;
+ };
+};
+
+
+int _tmain(int argc, _TCHAR* argv[])
+{
+
+ std::string str;
+ std::cout << "------------------------" << std::endl;
+ std::cout << "------ Test TMMBR ------" << std::endl;
+ std::cout << "------------------------" << std::endl;
+ std::cout << " " << std::endl;
+
+ // --------------------
+ // Test TMMBRHelp class
+ // --------------------
+ TestTMMBR test;
+ test.Start();
+
+ printf("TMMBRHelp-class test done.\n");
+
+ // ------------------------
+ // Test TMMBR single module
+ // ------------------------
+ RtpRtcp* rtpRtcpModuleVideo = RtpRtcp::CreateRtpRtcp(0, false);
+
+ LoopBackTransportVideo* myLoopBackTransportVideo = new LoopBackTransportVideo(rtpRtcpModuleVideo);
+ assert(0 == rtpRtcpModuleVideo->RegisterSendTransport(myLoopBackTransportVideo));
+
+ assert(false == rtpRtcpModuleVideo->TMMBR());
+ rtpRtcpModuleVideo->SetTMMBRStatus(true);
+ assert(true == rtpRtcpModuleVideo->TMMBR());
+
+ assert(0 == rtpRtcpModuleVideo->RegisterSendPayload( "I420", 96));
+ assert(0 == rtpRtcpModuleVideo->RegisterReceivePayload( "I420", 96));
+
+ // send a RTP packet with SSRC 11111 to get 11111 as the received SSRC
+ assert(0 == rtpRtcpModuleVideo->SetSSRC(11111));
+ const WebRtc_UWord8 testStream[9] = "testtest";
+ assert(0 == rtpRtcpModuleVideo->RegisterIncomingDataCallback(new NULLDataZink())); // needed to avoid error from parsing the incoming stream
+ assert(0 == rtpRtcpModuleVideo->SendOutgoingData(webrtc::kVideoFrameKey,96, 0, testStream, 8));
+
+ // set the SSRC to 0
+ assert(0 == rtpRtcpModuleVideo->SetSSRC(0));
+
+ //
+ assert(0 == rtpRtcpModuleVideo->SetRTCPStatus(kRtcpCompound));
+
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {0} // should this make us remember a TMMBR?
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {1}, verify TMMBN {0}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {2}, verify TMMBN {1}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {3}, verify TMMBN {2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4}, verify TMMBN {3,2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {5}, verify TMMBN {3,4,2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4_2}, verify TMMBN {3,4,2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> time out receivers, verify TMMBN {4_2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> verify TMMBN {2}
+
+ printf("Single module test done.\n");
+
+ // ------------------------
+ // Test TMMBR multi module
+ // ------------------------
+ RtpRtcp* rtpRtcpModuleVideoDef = RtpRtcp::CreateRtpRtcp(10, false);
+ assert(0 == rtpRtcpModuleVideo->RegisterDefaultModule(rtpRtcpModuleVideoDef));
+
+ RtpRtcp* rtpRtcpModuleVideo1 = RtpRtcp::CreateRtpRtcp(1, false);
+ assert(0 == rtpRtcpModuleVideo1->RegisterDefaultModule(rtpRtcpModuleVideoDef));
+
+ RtpRtcp* rtpRtcpModuleVideo2 = RtpRtcp::CreateRtpRtcp(2, false);
+ assert(0 == rtpRtcpModuleVideo2->RegisterDefaultModule(rtpRtcpModuleVideoDef));
+
+ RtpRtcp* rtpRtcpModuleVideo3 = RtpRtcp::CreateRtpRtcp(3, false);
+ assert(0 == rtpRtcpModuleVideo3->RegisterDefaultModule(rtpRtcpModuleVideoDef));
+
+ LoopBackTransport2* myLoopBackTransport2 = new LoopBackTransport2(rtpRtcpModuleVideo2);
+ assert(0 == rtpRtcpModuleVideo2->RegisterSendTransport(myLoopBackTransport2));
+
+ assert(0 == rtpRtcpModuleVideo2->SetRTCPStatus(kRtcpCompound));
+
+ // set the SSRC to 0
+ assert(0 == rtpRtcpModuleVideo2->SetSSRC(0));
+
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4}, verify no TMMBN in this packet
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {0}, verify TMMBN {4,2}
+ assert(0 == rtpRtcpModuleVideo2->SendRTCP()); // -> incoming TMMBR {3}, verify TMMBN {4,2}
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {1}, verify TMMBN {3,4,2}
+ ::Sleep(5*RTCP_INTERVAL_AUDIO_MS + 1000);
+ rtpRtcpModuleVideo2->Process(); // time out receiver2 -> UpdateTMMBR()
+ assert(0 == rtpRtcpModuleVideo->SendRTCP()); // verify TMMBN {}
+
+ printf("Multi module test done.\n");
+
+
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo);
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo1);
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo2);
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo3);
+ RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideoDef);
+
+ TEST_PASSED();
+ ::Sleep(5000);
+
+ return 0;
+}
+
diff --git a/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi b/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
new file mode 100644
index 0000000..d165e7d
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
@@ -0,0 +1,35 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'test_bwe',
+ 'type': 'executable',
+ 'dependencies': [
+ 'rtp_rtcp',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../../source',
+ ],
+ 'sources': [
+ 'unit_test.cc',
+ '../../source/bitrate.cc',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc b/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc
new file mode 100644
index 0000000..3adb099
--- /dev/null
+++ b/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes unit tests for the bandwidth estimation and management
+ */
+
+#include <gtest/gtest.h>
+
+#include "typedefs.h"
+#include "Bitrate.h"
+
+namespace {
+
+using webrtc::BitRateStats;
+
+class BitRateStatsTest : public ::testing::Test
+{
+protected:
+ BitRateStatsTest() {};
+ BitRateStats bitRate;
+};
+
+TEST_F(BitRateStatsTest, TestStrictMode)
+{
+ WebRtc_Word64 nowMs = 0;
+ // Should be initialized to 0.
+ EXPECT_EQ(0u, bitRate.BitRate(nowMs));
+ bitRate.Update(1500, nowMs);
+ // Expecting 12 kbps given a 1000 window with one 1500 bytes packet.
+ EXPECT_EQ(12000u, bitRate.BitRate(nowMs));
+ bitRate.Init();
+ // Expecting 0 after init.
+ EXPECT_EQ(0u, bitRate.BitRate(nowMs));
+ for (int i = 0; i < 100000; ++i)
+ {
+ if (nowMs % 10 == 0)
+ bitRate.Update(1500, nowMs);
+ // Approximately 1200 kbps expected. Not exact since when packets
+ // are removed we will jump 10 ms to the next packet.
+ if (nowMs > 0 && nowMs % 2000 == 0)
+ EXPECT_NEAR(1200000u, bitRate.BitRate(nowMs), 6000u);
+ nowMs += 1;
+ }
+ nowMs += 2000;
+ // The window is 2 seconds. If nothing has been received for that time
+ // the estimate should be 0.
+ EXPECT_EQ(0u, bitRate.BitRate(nowMs));
+}
+
+}
diff --git a/trunk/src/modules/udp_transport/OWNERS b/trunk/src/modules/udp_transport/OWNERS
new file mode 100644
index 0000000..3b2a444
--- /dev/null
+++ b/trunk/src/modules/udp_transport/OWNERS
@@ -0,0 +1,4 @@
+pwestin@webrtc.org
+henrikg@webrtc.org
+mallinath@webrtc.org
+tomasl@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/udp_transport/interface/udp_transport.h b/trunk/src/modules/udp_transport/interface/udp_transport.h
new file mode 100644
index 0000000..6596fde
--- /dev/null
+++ b/trunk/src/modules/udp_transport/interface/udp_transport.h
@@ -0,0 +1,386 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
+
+#include "common_types.h"
+#include "module.h"
+#include "typedefs.h"
+
+#define SS_MAXSIZE 128
+#define SS_ALIGNSIZE (sizeof (WebRtc_UWord64))
+#define SS_PAD1SIZE (SS_ALIGNSIZE - sizeof(WebRtc_Word16))
+#define SS_PAD2SIZE (SS_MAXSIZE - (sizeof(WebRtc_Word16) + SS_PAD1SIZE +\
+ SS_ALIGNSIZE))
+
+// BSD requires use of HAVE_STRUCT_SOCKADDR_SA_LEN
+namespace webrtc {
+struct SocketAddressIn
+{
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
+#else
+ WebRtc_Word16 sin_family;
+#endif
+ WebRtc_UWord16 sin_port;
+ WebRtc_UWord32 sin_addr;
+ WebRtc_Word8 sin_zero[8];
+};
+
+struct Version6InAddress
+{
+ union
+ {
+ WebRtc_UWord8 _s6_u8[16];
+ WebRtc_UWord32 _s6_u32[4];
+ WebRtc_UWord64 _s6_u64[2];
+ } Version6AddressUnion;
+};
+
+struct SocketAddressInVersion6
+{
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
+#else
+ WebRtc_Word16 sin_family;
+#endif
+ // Transport layer port number.
+ WebRtc_UWord16 sin6_port;
+ // IPv6 traffic class and flow info or ip4 address.
+ WebRtc_UWord32 sin6_flowinfo;
+ // IPv6 address
+ struct Version6InAddress sin6_addr;
+ // Set of interfaces for a scope.
+ WebRtc_UWord32 sin6_scope_id;
+};
+
+struct SocketAddressStorage
+{
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
+#else
+ WebRtc_Word16 sin_family;
+#endif
+ WebRtc_Word8 __ss_pad1[SS_PAD1SIZE];
+ WebRtc_UWord64 __ss_align;
+ WebRtc_Word8 __ss_pad2[SS_PAD2SIZE];
+};
+
+struct SocketAddress
+{
+ union
+ {
+ struct SocketAddressIn _sockaddr_in;
+ struct SocketAddressInVersion6 _sockaddr_in6;
+ struct SocketAddressStorage _sockaddr_storage;
+ };
+};
+
+// Callback class that receives packets from UdpTransport.
+class UdpTransportData
+{
+public:
+ virtual ~UdpTransportData() {};
+
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort) = 0;
+
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort) = 0;
+};
+
+
+class UdpTransport : public Module, public Transport
+{
+public:
+ enum
+ {
+ kIpAddressVersion6Length = 64,
+ kIpAddressVersion4Length = 16
+ };
+ enum ErrorCode
+ {
+ kNoSocketError = 0,
+ kFailedToBindPort = 1,
+ kIpAddressInvalid = 2,
+ kAddressInvalid = 3,
+ kSocketInvalid = 4,
+ kPortInvalid = 5,
+ kTosInvalid = 6,
+ kMulticastAddressInvalid = 7,
+ kQosError = 8,
+ kSocketAlreadyInitialized = 9,
+ kIpVersion6Error = 10,
+ FILTER_ERROR = 11,
+ kStartReceiveError = 12,
+ kStopReceiveError = 13,
+ kCannotFindLocalIp = 14,
+ kTosError = 16,
+ kNotInitialized = 17,
+ kPcpError = 18
+ };
+
+ // Factory method. Constructor disabled.
+ static UdpTransport* Create(const WebRtc_Word32 id,
+ WebRtc_UWord8& numSocketThreads);
+ static void Destroy(UdpTransport* module);
+
+ // Prepares the class for sending RTP packets to ipAddr:rtpPort and RTCP
+ // packets to ipAddr:rtpPort+1 if rtcpPort is zero. Otherwise to
+ // ipAddr:rtcpPort.
+ virtual WebRtc_Word32 InitializeSendSockets(
+ const char* ipAddr,
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0) = 0;
+
+ // Register packetCallback for receiving incoming packets. Set the local
+ // RTP port to rtpPort. Bind local IP address to ipAddr. If ipAddr is NULL
+ // bind to local IP ANY. Set the local rtcp port to rtcpPort or rtpPort + 1
+ // if rtcpPort is 0.
+ virtual WebRtc_Word32 InitializeReceiveSockets(
+ UdpTransportData* const packetCallback,
+ const WebRtc_UWord16 rtpPort,
+ const char* ipAddr = NULL,
+ const char* multicastIpAddr = NULL,
+ const WebRtc_UWord16 rtcpPort = 0) = 0;
+
+ // Set local RTP port to rtpPort and RTCP port to rtcpPort or rtpPort + 1 if
+ // rtcpPort is 0. These ports will be used for sending instead of the local
+ // ports set by InitializeReceiveSockets(..).
+ virtual WebRtc_Word32 InitializeSourcePorts(
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0) = 0;
+
+ // Retrieve local ports used for sending if other than the ports specified
+ // by InitializeReceiveSockets(..). rtpPort is set to the RTP port.
+ // rtcpPort is set to the RTCP port.
+ virtual WebRtc_Word32 SourcePorts(WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const = 0;
+
+ // Set ipAddr to the IP address that is currently being listened on. rtpPort
+ // to the RTP port listened to. rtcpPort to the RTCP port listened on.
+ // multicastIpAddr to the multicast IP address group joined (the address
+ // is NULL terminated).
+ virtual WebRtc_Word32 ReceiveSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort,
+ char multicastIpAddr[kIpAddressVersion6Length]) const = 0;
+
+ // Set ipAddr to the IP address being sent from. rtpPort to the local RTP
+ // port used for sending and rtcpPort to the local RTCP port used for
+ // sending.
+ virtual WebRtc_Word32 SendSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const = 0;
+
+ // Put the IP address, RTP port and RTCP port from the last received packet
+ // into ipAddr, rtpPort and rtcpPort respectively.
+ virtual WebRtc_Word32 RemoteSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const = 0;
+
+ // Enable/disable quality of service if QoS is true or false respectively.
+ // Set the type of service to serviceType, max bitrate in kbit/s to
+ // maxBitrate and override DSCP if overrideDSCP is not 0.
+ // Note: Must be called both InitializeSendSockets() and
+ // InitializeReceiveSockets() has been called.
+ virtual WebRtc_Word32 SetQoS(const bool QoS,
+ const WebRtc_Word32 serviceType,
+ const WebRtc_UWord32 maxBitrate = 0,
+ const WebRtc_Word32 overrideDSCP = 0,
+ const bool audio = false) = 0;
+
+ // Set QoS to true if quality of service has been turned on. If QoS is true,
+ // also set serviceType to type of service and overrideDSCP to override
+ // DSCP.
+ virtual WebRtc_Word32 QoS(bool& QoS,
+ WebRtc_Word32& serviceType,
+ WebRtc_Word32& overrideDSCP) const = 0;
+
+ // Set type of service.
+ virtual WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP,
+ const bool useSetSockOpt = false) = 0;
+
+ // Get type of service configuration.
+ virtual WebRtc_Word32 ToS(WebRtc_Word32& DSCP,
+ bool& useSetSockOpt) const = 0;
+
+ // Set Priority Code Point (IEEE 802.1Q)
+ // Note: for Linux this function will set the priority for the socket,
+ // which then can be mapped to a PCP value with vconfig.
+ virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 PCP) = 0;
+
+ // Get Priority Code Point
+ virtual WebRtc_Word32 PCP(WebRtc_Word32& PCP) const = 0;
+
+ // Enable IPv6.
+ // Note: this API must be called before any call to
+ // InitializeReceiveSockets() or InitializeSendSockets(). It is not
+ // possible to go back to IPv4 (default) after this call.
+ virtual WebRtc_Word32 EnableIpV6() = 0;
+
+ // Return true if IPv6 has been enabled.
+ virtual bool IpV6Enabled() const = 0;
+
+ // Only allow packets received from filterIPAddress to be processed.
+ // Note: must be called after EnableIPv6(), if IPv6 is used.
+ virtual WebRtc_Word32 SetFilterIP(
+ const char filterIPAddress[kIpAddressVersion6Length]) = 0;
+
+ // Write the filter IP address (if any) to filterIPAddress.
+ virtual WebRtc_Word32 FilterIP(
+ char filterIPAddress[kIpAddressVersion6Length]) const = 0;
+
+ // Only allow RTP packets from rtpFilterPort and RTCP packets from
+ // rtcpFilterPort be processed.
+ // Note: must be called after EnableIPv6(), if IPv6 is used.
+ virtual WebRtc_Word32 SetFilterPorts(
+ const WebRtc_UWord16 rtpFilterPort,
+ const WebRtc_UWord16 rtcpFilterPort) = 0;
+
+ // Set rtpFilterPort to the filter RTP port and rtcpFilterPort to the
+ // filter RTCP port (if filtering based on port is enabled).
+ virtual WebRtc_Word32 FilterPorts(WebRtc_UWord16& rtpFilterPort,
+ WebRtc_UWord16& rtcpFilterPort) const = 0;
+
+ // Set the number of buffers that the socket implementation may use for
+ // receiving packets to numberOfSocketBuffers. I.e. the number of packets
+ // that can be received in parallell.
+ // Note: this API only has effect on Windows.
+ virtual WebRtc_Word32 StartReceiving(
+ const WebRtc_UWord32 numberOfSocketBuffers) = 0;
+
+ // Stop receive incoming packets.
+ virtual WebRtc_Word32 StopReceiving() = 0;
+
+ // Return true incoming packets are received.
+ virtual bool Receiving() const = 0;
+
+ // Return true if send sockets have been initialized.
+ virtual bool SendSocketsInitialized() const = 0;
+
+ // Return true if local ports for sending has been set.
+ virtual bool SourcePortsInitialized() const = 0;
+
+ // Return true if receive sockets have been initialized.
+ virtual bool ReceiveSocketsInitialized() const = 0;
+
+ // Send data with size length to ip:portnr. The same port as the set
+ // with InitializeSendSockets(..) is used if portnr is 0. The same IP
+ // address as set with InitializeSendSockets(..) is used if ip is NULL.
+ // If isRTCP is true the port used will be the RTCP port.
+ virtual WebRtc_Word32 SendRaw(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ WebRtc_Word32 isRTCP,
+ WebRtc_UWord16 portnr = 0,
+ const char* ip = NULL) = 0;
+
+ // Send RTP data with size length to the address specified by to.
+ virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to) = 0;
+
+
+ // Send RTCP data with size length to the address specified by to.
+ virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to) = 0;
+
+ // Send RTP data with size length to ip:rtpPort where ip is the ip set by
+ // the InitializeSendSockets(..) call.
+ virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ WebRtc_UWord16 rtpPort) = 0;
+
+
+ // Send RTCP data with size length to ip:rtcpPort where ip is the ip set by
+ // the InitializeSendSockets(..) call.
+ virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ WebRtc_UWord16 rtcpPort) = 0;
+
+ // Set the IP address to which packets are sent to ipaddr.
+ virtual WebRtc_Word32 SetSendIP(
+ const char ipaddr[kIpAddressVersion6Length]) = 0;
+
+ // Set the send RTP and RTCP port to rtpPort and rtcpPort respectively.
+ virtual WebRtc_Word32 SetSendPorts(const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0) = 0;
+
+ // Retreive the last registered error code.
+ virtual ErrorCode LastError() const = 0;
+
+ // Put the local IPv4 address in localIP.
+ // Note: this API is for IPv4 only.
+ static WebRtc_Word32 LocalHostAddress(WebRtc_UWord32& localIP);
+
+ // Put the local IP6 address in localIP.
+ // Note: this API is for IPv6 only.
+ static WebRtc_Word32 LocalHostAddressIPV6(char localIP[16]);
+
+ // Return a copy of hostOrder (host order) in network order.
+ static WebRtc_UWord16 Htons(WebRtc_UWord16 hostOrder);
+
+ // Return a copy of hostOrder (host order) in network order.
+ static WebRtc_UWord32 Htonl(WebRtc_UWord32 hostOrder);
+
+ // Return IPv4 address in ip as 32 bit integer.
+ static WebRtc_UWord32 InetAddrIPV4(const char* ip);
+
+ // Convert the character string src into a network address structure in
+ // the af address family and put it in dst.
+ // Note: same functionality as inet_pton(..)
+ static WebRtc_Word32 InetPresentationToNumeric(WebRtc_Word32 af,
+ const char* src,
+ void* dst);
+
+ // Set ip and sourcePort according to address. As input parameter ipSize
+ // is the length of ip. As output parameter it's the number of characters
+ // written to ip (not counting the '\0' character).
+ // Note: this API is only implemented on Windows and Linux.
+ static WebRtc_Word32 IPAddress(const SocketAddress& address,
+ char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort);
+
+ // Set ip and sourcePort according to address. As input parameter ipSize
+ // is the length of ip. As output parameter it's the number of characters
+ // written to ip (not counting the '\0' character).
+ // Note: this API is only implemented on Windows and Linux.
+ // Additional note: this API caches the address of the last call to it. If
+ // address is likley to be the same for multiple calls it may be beneficial
+ // to call this API instead of IPAddress().
+ virtual WebRtc_Word32 IPAddressCached(const SocketAddress& address,
+ char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort) = 0;
+
+ // Return true if ipaddr is a valid IP address.
+ // If ipV6 is false ipaddr is interpreted as an IPv4 address otherwise it
+ // is interptreted as IPv6.
+ static bool IsIpAddressValid(const char* ipaddr, const bool ipV6);
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
diff --git a/trunk/src/modules/udp_transport/source/Android.mk b/trunk/src/modules/udp_transport/source/Android.mk
new file mode 100644
index 0000000..f3cb0e8
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/Android.mk
@@ -0,0 +1,45 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_udp_transport
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ udp_transport_impl.cc \
+ udp_socket_wrapper.cc \
+ udp_socket_manager_wrapper.cc \
+ udp_socket_manager_posix.cc \
+ udp_socket_posix.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/udp_transport/source/traffic_control_windows.cc b/trunk/src/modules/udp_transport/source/traffic_control_windows.cc
new file mode 100644
index 0000000..09038c0
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/traffic_control_windows.cc
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "traffic_control_windows.h"
+
+#include <assert.h>
+
+#include "trace.h"
+
+namespace webrtc {
+TrafficControlWindows* TrafficControlWindows::instance = NULL;
+WebRtc_UWord32 TrafficControlWindows::refCounter = 0;
+
+TrafficControlWindows::TrafficControlWindows(const WebRtc_Word32 id) : _id(id)
+{
+}
+
+TrafficControlWindows* TrafficControlWindows::GetInstance(
+ const WebRtc_Word32 id)
+{
+ if(instance != NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ id,
+ "TrafficControlWindows - Returning already created object");
+ refCounter++;
+ return instance;
+ }
+
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
+ "TrafficControlWindows - Creating new object");
+ instance = new TrafficControlWindows(id);
+ if(instance == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
+ "TrafficControlWindows - Error allocating memory");
+ return NULL;
+ }
+
+ instance->tcRegister = NULL;
+ instance->tcDeregister = NULL;
+
+ instance->tcEnumerate = NULL;
+ instance->tcOpenInterface = NULL;
+ instance->tcCloseInterface = NULL;
+
+ instance->tcAddFlow = NULL;
+ instance->tcDeleteFlow = NULL;
+
+ instance->tcAddFilter = NULL;
+ instance->tcDeleteFilter = NULL;
+
+ HMODULE trafficLib = LoadLibrary(TEXT("traffic.dll"));
+ if(trafficLib == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceTransport,
+ id,
+ "TrafficControlWindows - No QOS support, LoadLibrary returned NULL,\
+ last error: %d\n",
+ GetLastError());
+ delete instance;
+ instance = NULL;
+ return NULL;
+ }
+
+ instance->tcRegister = (registerFn)GetProcAddress(trafficLib,
+ "TcRegisterClient");
+ instance->tcDeregister = (deregisterFn)GetProcAddress(trafficLib,
+ "TcDeregisterClient");
+ instance->tcEnumerate = (enumerateFn)GetProcAddress(
+ trafficLib,
+ "TcEnumerateInterfaces");
+ instance->tcOpenInterface = (openInterfaceFn)GetProcAddress(
+ trafficLib,
+ "TcOpenInterfaceW");
+ instance->tcCloseInterface = (closeInterfaceFn)GetProcAddress(
+ trafficLib,
+ "TcCloseInterface");
+ instance->tcAddFlow = (flowAddFn)GetProcAddress(trafficLib,
+ "TcAddFlow");
+ instance->tcDeleteFlow = (flowDeleteFn)GetProcAddress(trafficLib,
+ "TcDeleteFlow");
+
+ instance->tcAddFilter = (filterAddFn)GetProcAddress(trafficLib,
+ "TcAddFilter");
+ instance->tcDeleteFilter = (filterDeleteFn)GetProcAddress(trafficLib,
+ "TcDeleteFilter");
+
+ if(instance->tcRegister == NULL ||
+ instance->tcDeregister == NULL ||
+ instance->tcEnumerate == NULL ||
+ instance->tcOpenInterface == NULL ||
+ instance->tcCloseInterface == NULL ||
+ instance->tcAddFlow == NULL ||
+ instance->tcAddFilter == NULL ||
+ instance->tcDeleteFlow == NULL ||
+ instance->tcDeleteFilter == NULL)
+ {
+ delete instance;
+ instance = NULL;
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ id,
+ "TrafficControlWindows - Could not find function pointer for\
+ traffic control functions");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ id,
+ "Tcregister : %x, tcDeregister: %x, tcEnumerate: %x,\
+ tcOpenInterface: %x, tcCloseInterface: %x, tcAddFlow: %x, tcAddFilter: %x,\
+ tcDeleteFlow: %x, tcDeleteFilter: %x",
+ instance->tcRegister,
+ instance->tcDeregister,
+ instance->tcEnumerate,
+ instance->tcOpenInterface,
+ instance->tcCloseInterface,
+ instance->tcAddFlow,
+ instance->tcAddFilter,
+ instance->tcDeleteFlow,
+ instance->tcDeleteFilter );
+ return NULL;
+ }
+ refCounter++;
+ return instance;
+}
+
+void TrafficControlWindows::Release(TrafficControlWindows* gtc)
+{
+ if (0 == refCounter)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, -1,
+ "TrafficControlWindows - Cannot release, refCounter is 0");
+ return;
+ }
+ if (NULL == gtc)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, -1,
+ "TrafficControlWindows - Not releasing, gtc is NULL");
+ return;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, gtc->_id,
+ "TrafficControlWindows - Releasing object");
+ refCounter--;
+ if ((0 == refCounter) && instance)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, gtc->_id,
+ "TrafficControlWindows - Deleting object");
+ delete instance;
+ instance = NULL;
+ }
+}
+WebRtc_Word32 TrafficControlWindows::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+ULONG TrafficControlWindows::TcRegisterClient(
+ ULONG TciVersion,
+ HANDLE ClRegCtx,
+ PTCI_CLIENT_FUNC_LIST ClientHandlerList,
+ PHANDLE pClientHandle)
+{
+ assert(tcRegister != NULL);
+
+ return tcRegister(TciVersion, ClRegCtx, ClientHandlerList, pClientHandle);
+}
+
+ULONG TrafficControlWindows::TcDeregisterClient(HANDLE clientHandle)
+{
+ assert(tcDeregister != NULL);
+
+ return tcDeregister(clientHandle);
+}
+
+
+ULONG TrafficControlWindows::TcEnumerateInterfaces(
+ HANDLE ClientHandle,
+ PULONG pBufferSize,
+ PTC_IFC_DESCRIPTOR interfaceBuffer)
+{
+ assert(tcEnumerate != NULL);
+
+ return tcEnumerate(ClientHandle, pBufferSize, interfaceBuffer);
+}
+
+
+ULONG TrafficControlWindows::TcOpenInterfaceW(LPWSTR pInterfaceName,
+ HANDLE ClientHandle,
+ HANDLE ClIfcCtx,
+ PHANDLE pIfcHandle)
+{
+ assert(tcOpenInterface != NULL);
+
+ return tcOpenInterface(pInterfaceName, ClientHandle, ClIfcCtx, pIfcHandle);
+
+}
+
+ULONG TrafficControlWindows::TcCloseInterface(HANDLE IfcHandle)
+{
+ assert(tcCloseInterface != NULL);
+
+ return tcCloseInterface(IfcHandle);
+}
+
+ULONG TrafficControlWindows::TcAddFlow(HANDLE IfcHandle, HANDLE ClFlowCtx,
+ ULONG Flags, PTC_GEN_FLOW pGenericFlow,
+ PHANDLE pFlowHandle)
+{
+ assert(tcAddFlow != NULL);
+ return tcAddFlow(IfcHandle, ClFlowCtx, Flags, pGenericFlow, pFlowHandle);
+}
+
+ULONG TrafficControlWindows::TcAddFilter(HANDLE FlowHandle,
+ PTC_GEN_FILTER pGenericFilter,
+ PHANDLE pFilterHandle)
+{
+ assert(tcAddFilter != NULL);
+ return tcAddFilter(FlowHandle, pGenericFilter, pFilterHandle);
+}
+
+ULONG TrafficControlWindows::TcDeleteFlow(HANDLE FlowHandle)
+{
+ assert(tcDeleteFlow != NULL);
+ return tcDeleteFlow(FlowHandle);
+
+}
+
+ULONG TrafficControlWindows::TcDeleteFilter(HANDLE FilterHandle)
+{
+ assert(tcDeleteFilter != NULL);
+ return tcDeleteFilter(FilterHandle);
+}
+
+void MyClNotifyHandler(HANDLE ClRegCtx, HANDLE ClIfcCtx, ULONG Event,
+ HANDLE SubCode, ULONG BufSize, PVOID Buffer)
+{
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/traffic_control_windows.h b/trunk/src/modules/udp_transport/source/traffic_control_windows.h
new file mode 100644
index 0000000..cfa52ce
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/traffic_control_windows.h
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
+
+#ifndef WIN32_LEAN_AND_MEAN
+#define WIN32_LEAN_AND_MEAN
+#endif
+
+// Disable deprication warning from traffic.h
+#pragma warning(disable : 4995)
+
+#include <windows.h>
+#include <qos.h>
+#include <ntddndis.h>
+#include <traffic.h>
+
+#include "trace.h"
+
+namespace webrtc {
+void MyClNotifyHandler(HANDLE ClRegCtx, HANDLE ClIfcCtx, ULONG Event,
+ HANDLE SubCode, ULONG BufSize, PVOID Buffer);
+
+
+typedef ULONG (WINAPI *registerFn)(ULONG, HANDLE, PTCI_CLIENT_FUNC_LIST,
+ PHANDLE);
+typedef ULONG (WINAPI *deregisterFn)(HANDLE);
+typedef ULONG (WINAPI *enumerateFn)(HANDLE, PULONG, PTC_IFC_DESCRIPTOR);
+typedef ULONG (WINAPI *openInterfaceFn)(LPWSTR, HANDLE, HANDLE, PHANDLE);
+typedef ULONG (WINAPI *closeInterfaceFn)(HANDLE);
+typedef ULONG (WINAPI *flowAddFn)(HANDLE, HANDLE, ULONG, PTC_GEN_FLOW, PHANDLE);
+typedef ULONG (WINAPI *filterAddFn)(HANDLE, PTC_GEN_FILTER, PHANDLE);
+typedef ULONG (WINAPI *flowDeleteFn)(HANDLE);
+typedef ULONG (WINAPI *filterDeleteFn)(HANDLE);
+
+class TrafficControlWindows
+{
+ public:
+ // Factory method. Constructor disabled.
+ static TrafficControlWindows* GetInstance(const WebRtc_Word32 id);
+ static void Release(TrafficControlWindows* gtc);
+
+ WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ ULONG TcRegisterClient(ULONG TciVersion, HANDLE ClRegCtx,
+ PTCI_CLIENT_FUNC_LIST ClientHandlerList,
+ PHANDLE pClientHandle);
+
+ ULONG TcDeregisterClient(HANDLE clientHandle);
+
+ ULONG TcEnumerateInterfaces(HANDLE ClientHandle, PULONG pBufferSize,
+ PTC_IFC_DESCRIPTOR interfaceBuffer);
+
+ ULONG TcOpenInterfaceW(LPWSTR pInterfaceName, HANDLE ClientHandle,
+ HANDLE ClIfcCtx, PHANDLE pIfcHandle);
+
+ ULONG TcCloseInterface(HANDLE IfcHandle);
+
+ ULONG TcAddFlow(HANDLE IfcHandle, HANDLE ClFlowCtx, ULONG Flags,
+ PTC_GEN_FLOW pGenericFlow, PHANDLE pFlowHandle);
+
+ ULONG TcAddFilter(HANDLE FlowHandle, PTC_GEN_FILTER pGenericFilter,
+ PHANDLE pFilterHandle);
+
+ ULONG TcDeleteFlow(HANDLE FlowHandle);
+ ULONG TcDeleteFilter(HANDLE FilterHandle);
+private:
+ TrafficControlWindows(const WebRtc_Word32 id);
+ WebRtc_Word32 _id;
+ TCI_CLIENT_FUNC_LIST QoSFunctions;
+
+ static TrafficControlWindows* instance;
+
+ registerFn tcRegister;
+ deregisterFn tcDeregister;
+
+ enumerateFn tcEnumerate;
+ openInterfaceFn tcOpenInterface;
+ closeInterfaceFn tcCloseInterface;
+
+ flowAddFn tcAddFlow;
+ flowDeleteFn tcDeleteFlow;
+
+ filterAddFn tcAddFilter;
+ filterDeleteFn tcDeleteFilter;
+
+ static WebRtc_UWord32 refCounter;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc
new file mode 100644
index 0000000..0356331
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc
@@ -0,0 +1,658 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket2_manager_windows.h"
+
+#include <assert.h>
+#include <stdio.h>
+
+#include "aligned_malloc.h"
+#include "udp_socket2_windows.h"
+
+namespace webrtc {
+WebRtc_UWord32 UdpSocket2ManagerWindows::_numOfActiveManagers = 0;
+bool UdpSocket2ManagerWindows::_wsaInit = false;
+
+UdpSocket2ManagerWindows::UdpSocket2ManagerWindows()
+ : UdpSocketManager(),
+ _id(-1),
+ _stopped(false),
+ _init(false),
+ _pCrit(CriticalSectionWrapper::CreateCriticalSection()),
+ _ioCompletionHandle(NULL),
+ _numActiveSockets(0),
+ _event(EventWrapper::Create())
+{
+ _managerNumber = _numOfActiveManagers++;
+
+ if(_numOfActiveManagers == 1)
+ {
+ WORD wVersionRequested = MAKEWORD(2, 2);
+ WSADATA wsaData;
+ _wsaInit = WSAStartup(wVersionRequested, &wsaData) == 0;
+ // TODO (hellner): seems safer to use RAII for this. E.g. what happens
+ // if a UdpSocket2ManagerWindows() created and destroyed
+ // without being initialized.
+ }
+}
+
+UdpSocket2ManagerWindows::~UdpSocket2ManagerWindows()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2ManagerWindows(%d)::~UdpSocket2ManagerWindows()",
+ _managerNumber);
+
+ if(_init)
+ {
+ _pCrit->Enter();
+ if(_numActiveSockets)
+ {
+ _pCrit->Leave();
+ _event->Wait(INFINITE);
+ }
+ else
+ {
+ _pCrit->Leave();
+ }
+ StopWorkerThreads();
+
+ // All threads are stopped. Safe to delete them.
+ ListItem* pItem = NULL;
+ while((pItem = _workerThreadsList.First()) != NULL)
+ {
+ delete static_cast<UdpSocket2WorkerWindows*>(pItem->GetItem());
+ _workerThreadsList.PopFront();
+ }
+
+ _ioContextPool.Free();
+
+ _numOfActiveManagers--;
+ if(_ioCompletionHandle)
+ {
+ CloseHandle(_ioCompletionHandle);
+ }
+ if (_numOfActiveManagers == 0)
+ {
+ if(_wsaInit)
+ {
+ WSACleanup();
+ }
+ }
+ }
+ if(_pCrit)
+ {
+ delete _pCrit;
+ }
+ if(_event)
+ {
+ delete _event;
+ }
+}
+
+bool UdpSocket2ManagerWindows::Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads) {
+ CriticalSectionScoped cs(_pCrit);
+ if ((_id != -1) || (_numOfWorkThreads != 0)) {
+ assert(_id != -1);
+ assert(_numOfWorkThreads != 0);
+ return false;
+ }
+ _id = id;
+ _numOfWorkThreads = numOfWorkThreads;
+ return true;
+}
+
+WebRtc_Word32 UdpSocket2ManagerWindows::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+bool UdpSocket2ManagerWindows::Start()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2ManagerWindows(%d)::Start()",_managerNumber);
+ if(!_init)
+ {
+ StartWorkerThreads();
+ }
+
+ if(!_init)
+ {
+ return false;
+ }
+ _pCrit->Enter();
+ // Start worker threads.
+ _stopped = false;
+ WebRtc_Word32 i = 0;
+ WebRtc_Word32 error = 0;
+ ListItem* pItem = _workerThreadsList.First();
+ UdpSocket2WorkerWindows* pWorker;
+ while(pItem != NULL && !error)
+ {
+ pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
+ if(!pWorker->Start())
+ error = 1;
+ pItem = _workerThreadsList.Next(pItem);
+ }
+ if(error)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::Start() error starting worker\
+ threads",
+ _managerNumber);
+ _pCrit->Leave();
+ return false;
+ }
+ _pCrit->Leave();
+ return true;
+}
+
+bool UdpSocket2ManagerWindows::StartWorkerThreads()
+{
+ if(!_init)
+ {
+ _pCrit->Enter();
+
+ _ioCompletionHandle = CreateIoCompletionPort(INVALID_HANDLE_VALUE, NULL,
+ 0, 0);
+ if(_ioCompletionHandle == NULL)
+ {
+ WebRtc_Word32 error = GetLastError();
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::StartWorkerThreads()"
+ "_ioCompletioHandle == NULL: error:%d",
+ _managerNumber,error);
+ _pCrit->Leave();
+ return false;
+ }
+
+ // Create worker threads.
+ WebRtc_UWord32 i = 0;
+ bool error = false;
+ while(i < _numOfWorkThreads && !error)
+ {
+ UdpSocket2WorkerWindows* pWorker =
+ new UdpSocket2WorkerWindows(_ioCompletionHandle);
+ if(pWorker->Init() != 0)
+ {
+ error = true;
+ delete pWorker;
+ break;
+ }
+ _workerThreadsList.PushFront(pWorker);
+ i++;
+ }
+ if(error)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::StartWorkerThreads() error "
+ "creating work threads",
+ _managerNumber);
+ // Delete worker threads.
+ ListItem* pItem = NULL;
+ while((pItem = _workerThreadsList.First()) != NULL)
+ {
+ delete static_cast<UdpSocket2WorkerWindows*>(pItem->GetItem());
+ _workerThreadsList.PopFront();
+ }
+ _pCrit->Leave();
+ return false;
+ }
+ if(_ioContextPool.Init())
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::StartWorkerThreads() error "
+ "initiating _ioContextPool",
+ _managerNumber);
+ _pCrit->Leave();
+ return false;
+ }
+ _init = true;
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows::StartWorkerThreads %d number of work "
+ "threads created and initialized",
+ _numOfWorkThreads);
+ _pCrit->Leave();
+ }
+ return true;
+}
+
+bool UdpSocket2ManagerWindows::Stop()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2ManagerWindows(%d)::Stop()",_managerNumber);
+
+ if(!_init)
+ {
+ return false;
+ }
+ _pCrit->Enter();
+ _stopped = true;
+ if(_numActiveSockets)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::Stop() there is still active\
+ sockets",
+ _managerNumber);
+ _pCrit->Leave();
+ return false;
+ }
+ // No active sockets. Stop all worker threads.
+ bool result = StopWorkerThreads();
+ _pCrit->Leave();
+ return result;
+}
+
+bool UdpSocket2ManagerWindows::StopWorkerThreads()
+{
+ WebRtc_Word32 error = 0;
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::StopWorkerThreads() Worker\
+ threadsStoped, numActicve Sockets=%d",
+ _managerNumber,
+ _numActiveSockets);
+ UdpSocket2WorkerWindows* pWorker;
+ ListItem* pItem = _workerThreadsList.First();
+
+ // Set worker threads to not alive so that they will stop calling
+ // UdpSocket2WorkerWindows::Run().
+ while(pItem != NULL)
+ {
+ pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
+ pWorker->SetNotAlive();
+ pItem = _workerThreadsList.Next(pItem);
+ }
+ // Release all threads waiting for GetQueuedCompletionStatus(..).
+ if(_ioCompletionHandle)
+ {
+ WebRtc_UWord32 i = 0;
+ for(i = 0; i < _workerThreadsList.GetSize(); i++)
+ {
+ PostQueuedCompletionStatus(_ioCompletionHandle, 0 ,0 , NULL);
+ }
+ }
+ pItem = _workerThreadsList.First();
+
+ while(pItem != NULL)
+ {
+ pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
+ if(pWorker->Stop() == false)
+ {
+ error = -1;
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
+ "failed to stop worker thread");
+ }
+ pItem = _workerThreadsList.Next(pItem);
+ }
+
+ if(error)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::StopWorkerThreads() error stopping\
+ worker threads",
+ _managerNumber);
+ return false;
+ }
+ return true;
+}
+
+bool UdpSocket2ManagerWindows::AddSocketPrv(UdpSocket2Windows* s)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2ManagerWindows(%d)::AddSocketPrv()",_managerNumber);
+ if(!_init)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::AddSocketPrv() manager not\
+ initialized",
+ _managerNumber);
+ return false;
+ }
+ _pCrit->Enter();
+ if(s == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::AddSocketPrv() socket == NULL",
+ _managerNumber);
+ _pCrit->Leave();
+ return false;
+ }
+ if(s->GetFd() == NULL || s->GetFd() == INVALID_SOCKET)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::AddSocketPrv() socket->GetFd() ==\
+ %d",
+ _managerNumber,
+ (WebRtc_Word32)s->GetFd());
+ _pCrit->Leave();
+ return false;
+
+ }
+ _ioCompletionHandle = CreateIoCompletionPort((HANDLE)s->GetFd(),
+ _ioCompletionHandle,
+ (ULONG_PTR)(s), 0);
+ if(_ioCompletionHandle == NULL)
+ {
+ WebRtc_Word32 error = GetLastError();
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::AddSocketPrv() Error adding to IO\
+ completion: %d",
+ _managerNumber,
+ error);
+ _pCrit->Leave();
+ return false;
+ }
+ _numActiveSockets++;
+ _pCrit->Leave();
+ return true;
+}
+bool UdpSocket2ManagerWindows::RemoveSocketPrv(UdpSocket2Windows* s)
+{
+ if(!_init)
+ {
+ return false;
+ }
+ _pCrit->Enter();
+ _numActiveSockets--;
+ if(_numActiveSockets == 0)
+ {
+ _event->Set();
+ }
+ _pCrit->Leave();
+ return true;
+}
+
+PerIoContext* UdpSocket2ManagerWindows::PopIoContext()
+{
+ if(!_init)
+ {
+ return NULL;
+ }
+
+ PerIoContext* pIoC = NULL;
+ if(!_stopped)
+ {
+ pIoC = _ioContextPool.PopIoContext();
+ }else
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2ManagerWindows(%d)::PopIoContext() Manager Not started",
+ _managerNumber);
+ }
+ return pIoC;
+}
+
+WebRtc_Word32 UdpSocket2ManagerWindows::PushIoContext(PerIoContext* pIoContext)
+{
+ return _ioContextPool.PushIoContext(pIoContext);
+}
+
+IoContextPool::IoContextPool()
+ : _pListHead(NULL),
+ _init(false),
+ _size(0),
+ _inUse(0)
+{
+}
+
+IoContextPool::~IoContextPool()
+{
+ Free();
+ assert(_size.Value() == 0);
+ AlignedFree(_pListHead);
+}
+
+WebRtc_Word32 IoContextPool::Init(WebRtc_UWord32 /*increaseSize*/)
+{
+ if(_init)
+ {
+ return 0;
+ }
+
+ _pListHead = (PSLIST_HEADER)AlignedMalloc(sizeof(SLIST_HEADER),
+ MEMORY_ALLOCATION_ALIGNMENT);
+ if(_pListHead == NULL)
+ {
+ return -1;
+ }
+ InitializeSListHead(_pListHead);
+ _init = true;
+ return 0;
+}
+
+PerIoContext* IoContextPool::PopIoContext()
+{
+ if(!_init)
+ {
+ return NULL;
+ }
+
+ PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
+ if(pListEntry == NULL)
+ {
+ IoContextPoolItem* item = (IoContextPoolItem*)
+ AlignedMalloc(
+ sizeof(IoContextPoolItem),
+ MEMORY_ALLOCATION_ALIGNMENT);
+ if(item == NULL)
+ {
+ return NULL;
+ }
+ memset(&item->payload.ioContext,0,sizeof(PerIoContext));
+ item->payload.base = item;
+ pListEntry = &(item->itemEntry);
+ ++_size;
+ }
+ ++_inUse;
+ return &((IoContextPoolItem*)pListEntry)->payload.ioContext;
+}
+
+WebRtc_Word32 IoContextPool::PushIoContext(PerIoContext* pIoContext)
+{
+ // TODO (hellner): Overlapped IO should be completed at this point. Perhaps
+ // add an assert?
+ const bool overlappedIOCompleted = HasOverlappedIoCompleted(
+ (LPOVERLAPPED)pIoContext);
+
+ IoContextPoolItem* item = ((IoContextPoolItemPayload*)pIoContext)->base;
+
+ const WebRtc_Word32 usedItems = --_inUse;
+ const WebRtc_Word32 totalItems = _size.Value();
+ const WebRtc_Word32 freeItems = totalItems - usedItems;
+ if(freeItems < 0)
+ {
+ assert(false);
+ AlignedFree(item);
+ return -1;
+ }
+ if((freeItems >= totalItems>>1) &&
+ overlappedIOCompleted)
+ {
+ AlignedFree(item);
+ --_size;
+ return 0;
+ }
+ InterlockedPushEntrySList(_pListHead, &(item->itemEntry));
+ return 0;
+}
+
+WebRtc_Word32 IoContextPool::Free()
+{
+ if(!_init)
+ {
+ return 0;
+ }
+
+ WebRtc_Word32 itemsFreed = 0;
+ PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
+ while(pListEntry != NULL)
+ {
+ IoContextPoolItem* item = ((IoContextPoolItem*)pListEntry);
+ AlignedFree(item);
+ --_size;
+ itemsFreed++;
+ pListEntry = InterlockedPopEntrySList(_pListHead);
+ }
+ return itemsFreed;
+}
+
+WebRtc_Word32 UdpSocket2WorkerWindows::_numOfWorkers = 0;
+
+UdpSocket2WorkerWindows::UdpSocket2WorkerWindows(HANDLE ioCompletionHandle)
+ : _ioCompletionHandle(ioCompletionHandle),
+ _pThread(NULL),
+ _init(false)
+{
+ _workerNumber = _numOfWorkers++;
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
+ "UdpSocket2WorkerWindows created");
+}
+
+UdpSocket2WorkerWindows::~UdpSocket2WorkerWindows()
+{
+ if(_pThread)
+ {
+ delete _pThread;
+ }
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
+ "UdpSocket2WorkerWindows deleted");
+}
+
+bool UdpSocket2WorkerWindows::Start()
+{
+ unsigned int id = 0;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
+ "Start UdpSocket2WorkerWindows");
+ return _pThread->Start(id);
+}
+
+bool UdpSocket2WorkerWindows::Stop()
+{
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
+ "Stop UdpSocket2WorkerWindows");
+ return _pThread->Stop();
+}
+
+void UdpSocket2WorkerWindows::SetNotAlive()
+{
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
+ "SetNotAlive UdpSocket2WorkerWindows");
+ _pThread->SetNotAlive();
+}
+
+WebRtc_Word32 UdpSocket2WorkerWindows::Init()
+{
+ if(!_init)
+ {
+ const WebRtc_Word8* threadName = "UdpSocket2ManagerWindows_thread";
+ _pThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
+ threadName);
+ if(_pThread == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ -1,
+ "UdpSocket2WorkerWindows(%d)::Init(), error creating thread!",
+ _workerNumber);
+ return -1;
+ }
+ _init = true;
+ }
+ return 0;
+}
+
+bool UdpSocket2WorkerWindows::Run(ThreadObj obj)
+{
+ UdpSocket2WorkerWindows* pWorker =
+ static_cast<UdpSocket2WorkerWindows*>(obj);
+ return pWorker->Process();
+}
+
+// Process should always return true. Stopping the worker threads is done in
+// the UdpSocket2ManagerWindows::StopWorkerThreads() function.
+bool UdpSocket2WorkerWindows::Process()
+{
+ WebRtc_Word32 success = 0;
+ DWORD ioSize = 0;
+ UdpSocket2Windows* pSocket = NULL;
+ PerIoContext* pIOContext = 0;
+ OVERLAPPED* pOverlapped = 0;
+ success = GetQueuedCompletionStatus(_ioCompletionHandle,
+ &ioSize,
+ (ULONG_PTR*)&pSocket, &pOverlapped, 200);
+
+ WebRtc_UWord32 error = 0;
+ if(!success)
+ {
+ error = GetLastError();
+ if(error == WAIT_TIMEOUT)
+ {
+ return true;
+ }
+ // This may happen if e.g. PostQueuedCompletionStatus() has been called.
+ // The IO context still needs to be reclaimed or re-used which is done
+ // in UdpSocket2Windows::IOCompleted(..).
+ }
+ if(pSocket == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ -1,
+ "UdpSocket2WorkerWindows(%d)::Process(), pSocket == 0, end thread",
+ _workerNumber);
+ return true;
+ }
+ pIOContext = (PerIoContext*)pOverlapped;
+ pSocket->IOCompleted(pIOContext,ioSize,error);
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h
new file mode 100644
index 0000000..5adb293
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
+
+#if _MSC_VER > 1000
+#pragma once
+#endif
+
+#include <winsock2.h>
+
+#include "atomic32_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "list_wrapper.h"
+#include "thread_wrapper.h"
+#include "udp_socket2_windows.h"
+#include "udp_socket_manager_wrapper.h"
+
+#define MAX_IO_BUFF_SIZE 1600
+
+namespace webrtc {
+enum IO_OPERATION {
+ OP_READ,
+ OP_WRITE
+};
+
+class UdpSocket2Windows;
+
+// Struct used for all socket I/O operations.
+struct PerIoContext {
+ WSAOVERLAPPED overlapped;
+ char buffer[MAX_IO_BUFF_SIZE];
+ WSABUF wsabuf;
+ int nTotalBytes;
+ int nSentBytes;
+ int bytes;
+ IO_OPERATION ioOperation;
+ SocketAddress from;
+ int fromLen;
+ // Should be set to true if the I/O context was passed to the system by
+ // a thread not controlled by the socket implementation.
+ bool ioInitiatedByThreadWrapper;
+ // TODO (hellner): Not used. Delete it.
+ PerIoContext* pNextFree;
+};
+
+struct IoContextPoolItem;
+struct IoContextPoolItemPayload
+{
+ PerIoContext ioContext;
+ IoContextPoolItem* base;
+};
+
+struct IoContextPoolItem
+{
+ // Atomic single linked list entry header.
+ SLIST_ENTRY itemEntry;
+ // Atomic single linked list payload
+ IoContextPoolItemPayload payload;
+};
+
+class IoContextPool
+{
+public:
+ IoContextPool();
+ virtual ~IoContextPool();
+ virtual WebRtc_Word32 Init(WebRtc_UWord32 increaseSize = 128);
+ // Re-use an old unused IO context or create a new one.
+ virtual PerIoContext* PopIoContext();
+ virtual WebRtc_Word32 PushIoContext(PerIoContext* pIoContext);
+ virtual inline WebRtc_Word32 GetSize(WebRtc_UWord32* inUse = 0)
+ {return _size.Value();}
+ virtual WebRtc_Word32 Free();
+private:
+ // Sample code for use of msfts single linked atomic list can be found here:
+ // http://msdn.microsoft.com/en-us/library/ms686962(VS.85).aspx
+
+ // Atomic single linked list head.
+ PSLIST_HEADER _pListHead;
+
+ bool _init;
+ Atomic32Wrapper _size;
+ Atomic32Wrapper _inUse;
+};
+
+
+class UdpSocket2ManagerWindows : public UdpSocketManager
+{
+public:
+ UdpSocket2ManagerWindows();
+ virtual ~UdpSocket2ManagerWindows();
+
+ virtual bool Init(WebRtc_Word32 id, WebRtc_UWord8& numOfWorkThreads);
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool Start();
+ virtual bool Stop();
+
+ virtual inline bool AddSocket(UdpSocketWrapper* s)
+ {if(s) return AddSocketPrv(reinterpret_cast<UdpSocket2Windows*>(s));
+ return false;}
+ virtual bool RemoveSocket(UdpSocketWrapper* s)
+ {if(s) return RemoveSocketPrv(reinterpret_cast<UdpSocket2Windows*>(s));
+ return false;}
+
+ PerIoContext* PopIoContext(void);
+ WebRtc_Word32 PushIoContext(PerIoContext* pIoContext);
+
+private:
+ bool StopWorkerThreads();
+ bool StartWorkerThreads();
+ bool AddSocketPrv(UdpSocket2Windows* s);
+ bool RemoveSocketPrv(UdpSocket2Windows* s);
+
+ static WebRtc_UWord32 _numOfActiveManagers;
+ static bool _wsaInit;
+
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper* _pCrit;
+ WebRtc_Word32 _managerNumber;
+ volatile bool _stopped;
+ bool _init;
+ WebRtc_Word32 _numActiveSockets;
+ ListWrapper _workerThreadsList;
+ EventWrapper* _event;
+
+ HANDLE _ioCompletionHandle;
+ IoContextPool _ioContextPool;
+};
+
+class UdpSocket2WorkerWindows
+{
+public:
+ UdpSocket2WorkerWindows(HANDLE ioCompletionHandle);
+ virtual ~UdpSocket2WorkerWindows();
+
+ virtual bool Start();
+ virtual bool Stop();
+ virtual WebRtc_Word32 Init();
+ virtual void SetNotAlive();
+protected:
+ static bool Run(ThreadObj obj);
+ bool Process();
+private:
+ HANDLE _ioCompletionHandle;
+ ThreadWrapper*_pThread;
+ static WebRtc_Word32 _numOfWorkers;
+ WebRtc_Word32 _workerNumber;
+ volatile bool _stop;
+ bool _init;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc
new file mode 100644
index 0000000..32e3598
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc
@@ -0,0 +1,1411 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket2_windows.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <winsock2.h>
+
+#include "traffic_control_windows.h"
+#include "udp_socket2_manager_windows.h"
+
+#pragma warning(disable : 4311)
+
+namespace webrtc {
+typedef struct _QOS_DESTADDR
+{
+ QOS_OBJECT_HDR ObjectHdr;
+ const struct sockaddr* SocketAddress;
+ ULONG SocketAddressLength;
+} QOS_DESTADDR, *LPQOS_DESTADDR;
+
+typedef const QOS_DESTADDR* LPCQOS_DESTADDR;
+
+// TODO (patrikw): seems to be defined in ws2ipdef.h as 3. How come it's
+// redefined here (as a different value)?
+#define IP_TOS 8
+
+#define QOS_GENERAL_ID_BASE 2000
+#define QOS_OBJECT_DESTADDR (0x00000004 + QOS_GENERAL_ID_BASE)
+
+UdpSocket2Windows::UdpSocket2Windows(const WebRtc_Word32 id,
+ UdpSocketManager* mgr, bool ipV6Enable,
+ bool disableGQOS)
+ : _id(id),
+ _qos(true),
+ _iProtocol(0),
+ _outstandingCalls(0),
+ _outstandingCallComplete(0),
+ _terminate(false),
+ _addedToMgr(false),
+ _safeTodelete(false),
+ _outstandingCallsDisabled(0),
+ _clientHandle(NULL),
+ _flowHandle(NULL),
+ _filterHandle(NULL),
+ _flow(NULL),
+ _gtc(NULL),
+ _pcp(-2),
+ _receiveBuffers(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
+ "UdpSocket2Windows::UdpSocket2Windows()");
+
+ _wantsIncoming = false;
+ _mgr = static_cast<UdpSocket2ManagerWindows *>(mgr);
+
+ _obj = NULL;
+ _incomingCb = NULL;
+ _socket = INVALID_SOCKET;
+ _pCrit = CriticalSectionWrapper::CreateCriticalSection();
+ _ptrCbRWLock = RWLockWrapper::CreateRWLock();
+ _ptrDestRWLock = RWLockWrapper::CreateRWLock();
+ _ptrSocketRWLock = RWLockWrapper::CreateRWLock();
+ _ptrDeleteCrit = CriticalSectionWrapper::CreateCriticalSection();
+ _ptrDeleteCond = ConditionVariableWrapper::CreateConditionVariable();
+
+ // Check if QoS is supported.
+ BOOL bProtocolFound = FALSE;
+ WSAPROTOCOL_INFO *lpProtocolBuf = NULL;
+ WSAPROTOCOL_INFO pProtocolInfo;
+
+ if(!disableGQOS)
+ {
+ DWORD dwBufLen = 0;
+ // Set dwBufLen to the size needed to retreive all the requested
+ // information from WSAEnumProtocols.
+ WebRtc_Word32 nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
+ lpProtocolBuf = (WSAPROTOCOL_INFO*)malloc(dwBufLen);
+ nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
+
+ if (ipV6Enable)
+ {
+ _iProtocol=AF_INET6;
+ } else {
+ _iProtocol=AF_INET;
+ }
+
+ for (WebRtc_Word32 i=0; i<nRet; i++)
+ {
+ if (_iProtocol == lpProtocolBuf[i].iAddressFamily &&
+ IPPROTO_UDP == lpProtocolBuf[i].iProtocol)
+ {
+ if ((XP1_QOS_SUPPORTED ==
+ (XP1_QOS_SUPPORTED & lpProtocolBuf[i].dwServiceFlags1)))
+ {
+ pProtocolInfo = lpProtocolBuf[i];
+ bProtocolFound = TRUE;
+ break;
+ }
+ }
+ }
+ }
+
+ if(!bProtocolFound)
+ {
+ free(lpProtocolBuf);
+ _qos=false;
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR_NO_QOS,\
+ !bProtocolFound");
+ } else {
+
+ _socket = WSASocket(FROM_PROTOCOL_INFO, FROM_PROTOCOL_INFO,
+ FROM_PROTOCOL_INFO,&pProtocolInfo, 0,
+ WSA_FLAG_OVERLAPPED);
+ free(lpProtocolBuf);
+
+ if (_socket != INVALID_SOCKET)
+ {
+ return;
+ } else {
+ _qos = false;
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR_NO_QOS");
+ }
+ }
+ // QoS not supported.
+ if(ipV6Enable)
+ {
+ _socket = WSASocket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP, 0 , 0,
+ WSA_FLAG_OVERLAPPED);
+ }else
+ {
+ _socket = WSASocket(AF_INET, SOCK_DGRAM, IPPROTO_UDP, 0 , 0,
+ WSA_FLAG_OVERLAPPED);
+ }
+ if (_socket == INVALID_SOCKET)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::UdpSocket2Windows(), INVALID_SOCKET,\
+ WSAerror: %d",
+ WSAGetLastError());
+ }
+
+ // Disable send buffering on the socket to improve CPU usage.
+ // This is done by setting SO_SNDBUF to 0.
+ WebRtc_Word32 nZero = 0;
+ WebRtc_Word32 nRet = setsockopt(_socket, SOL_SOCKET, SO_SNDBUF,
+ (WebRtc_Word8*)&nZero, sizeof(nZero));
+ if( nRet == SOCKET_ERROR )
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR,\
+ WSAerror: %d",
+ WSAGetLastError());
+ }
+}
+
+UdpSocket2Windows::~UdpSocket2Windows()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
+ "UdpSocket2Windows::~UdpSocket2Windows()");
+
+ WaitForOutstandingCalls();
+
+ delete _ptrCbRWLock;
+ delete _ptrDeleteCrit;
+ delete _ptrDeleteCond;
+ delete _ptrDestRWLock;
+ delete _ptrSocketRWLock;
+
+ if(_pCrit)
+ delete _pCrit;
+
+ if (_flow)
+ {
+ free(_flow);
+ _flow = NULL;
+ }
+
+ if (_gtc)
+ {
+ if(_filterHandle)
+ {
+ _gtc->TcDeleteFilter(_filterHandle);
+ }
+ if(_flowHandle)
+ {
+ _gtc->TcDeleteFlow(_flowHandle);
+ }
+ TrafficControlWindows::Release( _gtc);
+ }
+}
+
+WebRtc_Word32 UdpSocket2Windows::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ if (_gtc)
+ {
+ _gtc->ChangeUniqueId(id);
+ }
+ return 0;
+}
+
+bool UdpSocket2Windows::ValidHandle()
+{
+ return GetFd() != INVALID_SOCKET;
+}
+
+bool UdpSocket2Windows::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
+{
+ _ptrCbRWLock->AcquireLockExclusive();
+ _obj = obj;
+ _incomingCb = cb;
+ _ptrCbRWLock->ReleaseLockExclusive();
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SetCallback ",(WebRtc_Word32)this);
+ if(_addedToMgr)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SetCallback alreadey added",
+ (WebRtc_Word32) this);
+ return false;
+
+ }
+ if (_mgr->AddSocket(this))
+ {
+ WEBRTC_TRACE(
+ kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SetCallback socket added to manager",
+ (WebRtc_Word32)this);
+ _addedToMgr = true;
+ return true;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SetCallback error adding me to mgr",
+ (WebRtc_Word32) this);
+ return false;
+}
+
+bool UdpSocket2Windows::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval,
+ WebRtc_Word32 optlen)
+{
+ bool returnValue = true;
+ if(!AquireSocket())
+ {
+ return false;
+ }
+ if(0 != setsockopt(_socket, level, optname, optval, optlen ))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::SetSockopt(), WSAerror:%d",
+ WSAGetLastError());
+ returnValue = false;
+ }
+ ReleaseSocket();
+ return returnValue;
+}
+
+bool UdpSocket2Windows::StartReceiving(WebRtc_UWord32 receiveBuffers)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::StartReceiving(%d)",
+ (WebRtc_Word32)this, receiveBuffers);
+
+ _wantsIncoming = true;
+
+ WebRtc_Word32 numberOfReceiveBuffersToCreate =
+ receiveBuffers - _receiveBuffers.Value();
+ numberOfReceiveBuffersToCreate = (numberOfReceiveBuffersToCreate < 0) ?
+ 0 : numberOfReceiveBuffersToCreate;
+
+ WebRtc_Word32 error = 0;
+ for(WebRtc_Word32 i = 0;
+ i < numberOfReceiveBuffersToCreate;
+ i++)
+ {
+ if(PostRecv())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::StartReceiving() i=%d", i);
+ error = -1;
+ break;
+ }
+ ++_receiveBuffers;
+ }
+ if(error == -1)
+ {
+ return false;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Socket receiving using:%d number of buffers",
+ _receiveBuffers.Value());
+ return true;
+}
+
+bool UdpSocket2Windows::StopReceiving()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocket2Windows::StopReceiving()");
+ _wantsIncoming = false;
+ return true;
+}
+
+bool UdpSocket2Windows::Bind(const SocketAddress& name)
+{
+ const struct sockaddr* addr =
+ reinterpret_cast<const struct sockaddr*>(&name);
+ bool returnValue = true;
+ if(!AquireSocket())
+ {
+ return false;
+ }
+ if (0 != bind(_socket, addr, sizeof(SocketAddress)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::Bind() WSAerror: %d",
+ WSAGetLastError());
+ returnValue = false;
+ }
+ ReleaseSocket();
+ return returnValue;
+}
+
+WebRtc_Word32 UdpSocket2Windows::SendTo(const WebRtc_Word8* buf,
+ WebRtc_Word32 len,
+ const SocketAddress& to)
+{
+ WebRtc_Word32 retVal = 0;
+ WebRtc_Word32 error = 0;
+ if(len < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SendTo(), len= %d < 0",
+ (WebRtc_Word32)this, len);
+ return -1;
+ }
+
+ PerIoContext* pIoContext = _mgr->PopIoContext();
+ if(pIoContext == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::SendTo(), pIoContext==0",
+ (WebRtc_Word32) this);
+ return -1;
+ }
+ // sizeof(pIoContext->buffer) is smaller than the highest number that
+ // can be represented by a WebRtc_Word32.
+ if(len >= (WebRtc_Word32) sizeof(pIoContext->buffer))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows(%d)::SendTo(), len= %d > buffer_size = %d",
+ (WebRtc_Word32) this,
+ len,sizeof(pIoContext->buffer));
+ len = sizeof(pIoContext->buffer);
+ }
+
+ memcpy(pIoContext->buffer,buf,len);
+ pIoContext->wsabuf.buf = pIoContext->buffer;
+ pIoContext->wsabuf.len = len;
+ pIoContext->fromLen=sizeof(SocketAddress);
+ pIoContext->ioOperation = OP_WRITE;
+ pIoContext->nTotalBytes = len;
+ pIoContext->nSentBytes=0;
+
+ DWORD numOfbytesSent = 0;
+ const struct sockaddr* addr = reinterpret_cast<const struct sockaddr*>(&to);
+
+ if(!AquireSocket())
+ {
+ _mgr->PushIoContext(pIoContext);
+ return -1;
+ }
+ // Assume that the WSASendTo call will be successfull to make sure that
+ // _outstandingCalls is positive. Roll back if WSASendTo failed.
+ if(!NewOutstandingCall())
+ {
+ _mgr->PushIoContext(pIoContext);
+ ReleaseSocket();
+ return -1;
+ }
+ retVal = WSASendTo(_socket, &pIoContext->wsabuf, 1, &numOfbytesSent,
+ 0, addr, sizeof(SocketAddress),
+ &(pIoContext->overlapped), 0);
+ ReleaseSocket();
+
+ if( retVal == SOCKET_ERROR )
+ {
+ error = WSAGetLastError();
+ if(error != ERROR_IO_PENDING)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::SendTo() WSAerror: %d",error);
+ }
+ }
+ if(retVal == 0 || (retVal == SOCKET_ERROR && error == ERROR_IO_PENDING))
+ {
+ return len;
+ }
+ if((error = _mgr->PushIoContext(pIoContext)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows(%d)::SendTo(), error:%d pushing ioContext",
+ (WebRtc_Word32)this, error);
+ }
+
+ // Roll back.
+ OutstandingCallCompleted();
+ return -1;
+}
+
+void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
+ WebRtc_UWord32 ioSize, WebRtc_UWord32 error)
+{
+ if(pIOContext == NULL || error == ERROR_OPERATION_ABORTED)
+ {
+ if ((pIOContext != NULL) &&
+ !pIOContext->ioInitiatedByThreadWrapper &&
+ (error == ERROR_OPERATION_ABORTED) &&
+ (pIOContext->ioOperation == OP_READ) &&
+ _outstandingCallsDisabled.Value() == 1)
+ {
+ // !pIOContext->initiatedIOByThreadWrapper indicate that the I/O
+ // was not initiaded by a ThreadWrapper thread.
+ // This may happen if the thread that initiated receiving (e.g.
+ // by calling StartListen())) is deleted before any packets have
+ // been received.
+ // In this case there is no packet in the PerIoContext. Re-use it
+ // to post a new PostRecv(..).
+ // Note 1: the PerIoContext will henceforth be posted by a thread
+ // that is controlled by the socket implementation.
+ // Note 2: This is more likely to happen to RTCP packets as
+ // they are less frequent than RTP packets.
+ // Note 3: _outstandingCallsDisabled being false (= 1) indicates
+ // that the socket isn't being shut down.
+ // Note 4: This should only happen buffers set to recevie packets
+ // (OP_READ).
+ if (_outstandingCallsDisabled.Value() != 1)
+ {
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::IOCompleted(pIOContext=%p,\
+ ioSize=%.lu, error=%.lu) Received operation aborted but continuing since\
+ pIOContext->ioInitiatedByThreadWrapper == false",
+ pIOContext,
+ ioSize,
+ error);
+ WebRtc_Word32 ioOp = pIOContext ?
+ (WebRtc_Word32)pIOContext->ioOperation : -1;
+ WebRtc_Word32 ioInit = pIOContext ?
+ (WebRtc_Word32)pIOContext->ioInitiatedByThreadWrapper : -1;
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ _id,
+ "pIOContext->ioOperation=%d,\
+ pIOContext->ioInitiatedByThreadWrapper=%d, _outstandingCallsDisabled=%d,\
+ _incomingCb=%p, this=%p",
+ ioOp,
+ ioInit,
+ (WebRtc_Word32)_outstandingCallsDisabled.Value(),
+ _incomingCb,
+ this);
+ }
+ } else {
+ if(pIOContext == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::IOCompleted(%d,%d,%d), %d",
+ (WebRtc_Word32)pIOContext,
+ ioSize,
+ error,
+ pIOContext ? (WebRtc_Word32)pIOContext->ioOperation : -1);
+ } else {
+ WEBRTC_TRACE(
+ kTraceDebug,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::IOCompleted() Operation aborted");
+ }
+ if(pIOContext)
+ {
+ WebRtc_Word32 remainingReceiveBuffers = --_receiveBuffers;
+ if(remainingReceiveBuffers < 0)
+ {
+ assert(false);
+ }
+ WebRtc_Word32 err = 0;
+ if((err = _mgr->PushIoContext(pIOContext)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows::IOCompleted(), err = %d, when\
+ pushing ioContext after error",
+ err);
+ }
+ }
+ OutstandingCallCompleted();
+ return;
+ }
+ } // if (pIOContext == NULL || error == ERROR_OPERATION_ABORTED)
+
+ if(pIOContext->ioOperation == OP_WRITE)
+ {
+ _mgr->PushIoContext(pIOContext);
+ }
+ else if(pIOContext->ioOperation == OP_READ)
+ {
+ if(!error && ioSize != 0)
+ {
+ _ptrCbRWLock->AcquireLockShared();
+ if(_wantsIncoming && _incomingCb)
+ {
+ _incomingCb(_obj,pIOContext->wsabuf.buf, ioSize,
+ &pIOContext->from);
+ }
+ _ptrCbRWLock->ReleaseLockShared();
+ }
+ WebRtc_Word32 err = PostRecv(pIOContext);
+ if(err == 0)
+ {
+ // The PerIoContext was posted by a thread controlled by the socket
+ // implementation.
+ pIOContext->ioInitiatedByThreadWrapper = true;
+ }
+ OutstandingCallCompleted();
+ return;
+ } else {
+ // Unknown operation. Should not happen. Return pIOContext to avoid
+ // memory leak.
+ assert(false);
+ _mgr->PushIoContext(pIOContext);
+ }
+ OutstandingCallCompleted();
+ // Don't touch any members after OutstandingCallCompleted() since the socket
+ // may be deleted at this point.
+}
+
+WebRtc_Word32 UdpSocket2Windows::PostRecv()
+{
+ PerIoContext* pIoContext=_mgr->PopIoContext();
+ if(pIoContext == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::PostRecv(), pIoContext == 0",
+ (WebRtc_Word32)this);
+ return -1;
+ }
+ // This function may have been called by thread not controlled by the socket
+ // implementation.
+ pIoContext->ioInitiatedByThreadWrapper = false;
+ return PostRecv(pIoContext);
+}
+
+WebRtc_Word32 UdpSocket2Windows::PostRecv(PerIoContext* pIoContext)
+{
+ if(pIoContext==0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::PostRecv(?), pIoContext==0",
+ (WebRtc_Word32)this);
+ return -1;
+ }
+
+ DWORD numOfRecivedBytes = 0;
+ DWORD flags = 0;
+ pIoContext->wsabuf.buf = pIoContext->buffer;
+ pIoContext->wsabuf.len = sizeof(pIoContext->buffer);
+ pIoContext->fromLen = sizeof(SocketAddress);
+ pIoContext->ioOperation = OP_READ;
+ WebRtc_Word32 rxError = 0;
+ WebRtc_Word32 nRet = 0;
+ WebRtc_Word32 postingSucessfull = false;
+
+ if(!AquireSocket())
+ {
+ _mgr->PushIoContext(pIoContext);
+ return -1;
+ }
+
+ // Assume that the WSARecvFrom() call will be successfull to make sure that
+ // _outstandingCalls is positive. Roll back if WSARecvFrom() failed.
+ if(!NewOutstandingCall())
+ {
+ _mgr->PushIoContext(pIoContext);
+ ReleaseSocket();
+ return -1;
+ }
+ for(WebRtc_Word32 tries = 0; tries < 10; tries++)
+ {
+ nRet = WSARecvFrom(
+ _socket,
+ &(pIoContext->wsabuf),
+ 1,
+ &numOfRecivedBytes,
+ &flags,
+ reinterpret_cast<struct sockaddr*>(&(pIoContext->from)),
+ &(pIoContext->fromLen),
+ &(pIoContext->overlapped),
+ 0);
+
+ if( nRet == SOCKET_ERROR)
+ {
+ rxError = WSAGetLastError();
+ if(rxError != ERROR_IO_PENDING)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows(%d)::PostRecv(?), WSAerror:%d when\
+ posting new recieve,trie:%d",
+ (WebRtc_Word32)this,
+ rxError,
+ tries);
+ // Tell the OS that this is a good place to context switch if
+ // it wants to.
+ Sleep(0);
+ }
+ }
+ if((rxError == ERROR_IO_PENDING) || (nRet == 0))
+ {
+ postingSucessfull = true;
+ break;
+ }
+ }
+ ReleaseSocket();
+
+ if(postingSucessfull)
+ {
+ return 0;
+ }
+ WebRtc_Word32 remainingReceiveBuffers = --_receiveBuffers;
+ if(remainingReceiveBuffers < 0)
+ {
+ assert(false);
+ }
+ WebRtc_Word32 error = 0;
+ if((error = _mgr->PushIoContext(pIoContext)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocket2Windows(%d)::PostRecv(?), error:%d when PushIoContext",
+ (WebRtc_Word32)this,
+ error);
+ }
+ // Roll back.
+ OutstandingCallCompleted();
+ return -1;
+}
+
+void UdpSocket2Windows::CloseBlocking()
+{
+ LINGER lingerStruct;
+
+ lingerStruct.l_onoff = 1;
+ lingerStruct.l_linger = 0;
+ if(AquireSocket())
+ {
+ setsockopt(_socket, SOL_SOCKET, SO_LINGER,
+ (WebRtc_Word8 *)&lingerStruct, sizeof(lingerStruct));
+ ReleaseSocket();
+ }
+
+ _wantsIncoming = false;
+ // Reclaims the socket and prevents it from being used again.
+ InvalidateSocket();
+ DisableNewOutstandingCalls();
+ WaitForOutstandingCalls();
+ delete this;
+}
+
+bool UdpSocket2Windows::SetQos(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize,
+ const SocketAddress &stRemName,
+ WebRtc_Word32 overrideDSCP)
+{
+ if(_qos == false)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::SetQos(), socket not capable of QOS");
+ return false;
+ }
+ if(overrideDSCP != 0)
+ {
+ FLOWSPEC f;
+ WebRtc_Word32 err = CreateFlowSpec(serviceType, tokenRate, bucketSize,
+ peekBandwith, minPolicedSize,
+ maxSduSize, &f);
+ if(err == -1)
+ {
+ return false;
+ }
+
+ SocketAddress socketName;
+ struct sockaddr_in* name =
+ reinterpret_cast<struct sockaddr_in*>(&socketName);
+ int nameLength = sizeof(SocketAddress);
+ if(AquireSocket())
+ {
+ getsockname(_socket, (struct sockaddr*)name, &nameLength);
+ ReleaseSocket();
+ }
+
+ if(serviceType == 0)
+ {
+ // Disable TOS byte setting.
+ return SetTrafficControl(0, -1, name, &f, &f) == 0;
+ }
+ return SetTrafficControl(overrideDSCP, -1, name, &f, &f) == 0;
+ }
+
+ QOS Qos;
+ WebRtc_Word32 result ;
+ DWORD BytesRet;
+ QOS_DESTADDR QosDestaddr;
+
+ memset (&Qos, QOS_NOT_SPECIFIED, sizeof(QOS));
+
+ Qos.SendingFlowspec.ServiceType = serviceType;
+ Qos.SendingFlowspec.TokenRate = tokenRate;
+ Qos.SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+
+ // Only ServiceType is needed for receiving.
+ Qos.ReceivingFlowspec.ServiceType = serviceType;
+ Qos.ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+
+ Qos.ProviderSpecific.len = 0;
+
+ Qos.ProviderSpecific.buf = NULL;
+
+ ZeroMemory((WebRtc_Word8 *)&QosDestaddr, sizeof(QosDestaddr));
+
+ OSVERSIONINFOEX osvie;
+ osvie.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+ GetVersionEx((LPOSVERSIONINFO)&osvie);
+
+// Operating system Version number dwMajorVersion dwMinorVersion
+// Windows 7 6.1 6 1
+// Windows Server 2008 R2 6.1 6 1
+// Windows Server 2008 6.0 6 0
+// Windows Vista 6.0 6 0
+// Windows Server 2003 R2 5.2 5 2
+// Windows Server 2003 5.2 5 2
+// Windows XP 5.1 5 1
+// Windows 2000 5.0 5 0
+
+ // SERVICE_NO_QOS_SIGNALING and QOS_DESTADDR should not be used if version
+ // is 6.0 or greater.
+ if(osvie.dwMajorVersion >= 6)
+ {
+ Qos.SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.ServiceType = serviceType;
+
+ } else {
+ Qos.SendingFlowspec.MinimumPolicedSize =
+ QOS_NOT_SPECIFIED | SERVICE_NO_QOS_SIGNALING;
+ Qos.ReceivingFlowspec.ServiceType =
+ serviceType | SERVICE_NO_QOS_SIGNALING;
+
+ QosDestaddr.ObjectHdr.ObjectType = QOS_OBJECT_DESTADDR;
+ QosDestaddr.ObjectHdr.ObjectLength = sizeof(QosDestaddr);
+ QosDestaddr.SocketAddress = (SOCKADDR *)&stRemName;
+ if (AF_INET6 == _iProtocol)
+ {
+ QosDestaddr.SocketAddressLength = sizeof(SocketAddressInVersion6);
+ } else {
+ QosDestaddr.SocketAddressLength = sizeof(SocketAddressIn);
+ }
+
+ Qos.ProviderSpecific.len = QosDestaddr.ObjectHdr.ObjectLength;
+ Qos.ProviderSpecific.buf = (WebRtc_Word8*)&QosDestaddr;
+ }
+
+ if(AquireSocket())
+ {
+ // To set QoS with SIO_SET_QOS the socket must be locally bound first
+ // or the call will fail with error code 10022.
+ result = WSAIoctl(GetFd(), SIO_SET_QOS, &Qos, sizeof(QOS), NULL, 0,
+ &BytesRet, NULL,NULL);
+ ReleaseSocket();
+ }
+ if (result == SOCKET_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::SetQos() WSAerror : %d",
+ WSAGetLastError());
+ return false;
+ }
+ return true;
+}
+
+WebRtc_Word32 UdpSocket2Windows::SetTOS(WebRtc_Word32 serviceType)
+{
+ SocketAddress socketName;
+
+ struct sockaddr_in* name =
+ reinterpret_cast<struct sockaddr_in*>(&socketName);
+ int nameLength = sizeof(SocketAddress);
+ if(AquireSocket())
+ {
+ getsockname(_socket, (struct sockaddr*)name, &nameLength);
+ ReleaseSocket();
+ }
+
+ WebRtc_Word32 res = SetTrafficControl(serviceType, -1, name);
+ if (res == -1)
+ {
+ OSVERSIONINFO OsVersion;
+ OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
+ GetVersionEx (&OsVersion);
+
+ if ((OsVersion.dwMajorVersion == 4)) // NT 4.0
+ {
+ if(SetSockopt(IPPROTO_IP,IP_TOS ,
+ (WebRtc_Word8*)&serviceType, 4) != 0)
+ {
+ return -1;
+ }
+ }
+ }
+ return res;
+}
+
+WebRtc_Word32 UdpSocket2Windows::SetPCP(WebRtc_Word32 pcp)
+{
+ SocketAddress socketName;
+ struct sockaddr_in* name =
+ reinterpret_cast<struct sockaddr_in*>(&socketName);
+ int nameLength = sizeof(SocketAddress);
+ if(AquireSocket())
+ {
+ getsockname(_socket, (struct sockaddr*)name, &nameLength);
+ ReleaseSocket();
+ }
+ return SetTrafficControl(-1, pcp, name);
+}
+
+WebRtc_Word32 UdpSocket2Windows::SetTrafficControl(
+ WebRtc_Word32 dscp,
+ WebRtc_Word32 pcp,
+ const struct sockaddr_in* name,
+ FLOWSPEC* send, FLOWSPEC* recv)
+{
+ if (pcp == _pcp)
+ {
+ // No change.
+ pcp = -1;
+ }
+ if ((-1 == pcp) && (-1 == dscp))
+ {
+ return 0;
+ }
+ if (!_gtc)
+ {
+ _gtc = TrafficControlWindows::GetInstance(_id);
+ }
+ if (!_gtc)
+ {
+ return -1;
+ }
+ if(_filterHandle)
+ {
+ _gtc->TcDeleteFilter(_filterHandle);
+ _filterHandle = NULL;
+ }
+ if(_flowHandle)
+ {
+ _gtc->TcDeleteFlow(_flowHandle);
+ _flowHandle = NULL;
+ }
+ if(_clientHandle)
+ {
+ _gtc->TcDeregisterClient(_clientHandle);
+ _clientHandle = NULL;
+ }
+ if ((0 == dscp) && (-2 == _pcp) && (-1 == pcp))
+ {
+ // TODO (pwestin): why is this not done before deleting old filter and
+ // flow? This scenario should probably be documented in
+ // the function declaration.
+ return 0;
+ }
+
+ TCI_CLIENT_FUNC_LIST QoSFunctions;
+ QoSFunctions.ClAddFlowCompleteHandler = NULL;
+ QoSFunctions.ClDeleteFlowCompleteHandler = NULL;
+ QoSFunctions.ClModifyFlowCompleteHandler = NULL;
+ QoSFunctions.ClNotifyHandler = (TCI_NOTIFY_HANDLER)MyClNotifyHandler;
+ // Register the client with Traffic control interface.
+ HANDLE ClientHandle;
+ ULONG result = _gtc->TcRegisterClient(CURRENT_TCI_VERSION, NULL,
+ &QoSFunctions,&ClientHandle);
+ if(result != NO_ERROR)
+ {
+ // This is likely caused by the application not being run as
+ // administrator.
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "TcRegisterClient returned %d", result);
+ return result;
+ }
+
+ // Find traffic control-enabled network interfaces that matches this
+ // socket's IP address.
+ ULONG BufferSize = 0;
+ result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize, NULL);
+
+ if(result != NO_ERROR && result != ERROR_INSUFFICIENT_BUFFER)
+ {
+ _gtc->TcDeregisterClient(ClientHandle);
+ return result;
+ }
+
+ if(result != ERROR_INSUFFICIENT_BUFFER)
+ {
+ // Empty buffer contains all control-enabled network interfaces. I.e.
+ // QoS is not enabled.
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS faild since QOS is not installed on the interface");
+
+ _gtc->TcDeregisterClient(ClientHandle);
+ return -1;
+ }
+
+ PTC_IFC_DESCRIPTOR pInterfaceBuffer =
+ (PTC_IFC_DESCRIPTOR)malloc(BufferSize);
+ if(pInterfaceBuffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Out ot memory failure");
+ _gtc->TcDeregisterClient(ClientHandle);
+ return ERROR_NOT_ENOUGH_MEMORY;
+ }
+
+ result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize,
+ pInterfaceBuffer);
+
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "Critical: error enumerating interfaces when passing in correct\
+ buffer size: %d", result);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+ PTC_IFC_DESCRIPTOR oneinterface;
+ HANDLE ifcHandle, iFilterHandle, iflowHandle;
+ bool addrFound = false;
+ ULONG filterSourceAddress = ULONG_MAX;
+
+ // Find the interface corresponding to the local address.
+ for(oneinterface = pInterfaceBuffer;
+ oneinterface != (PTC_IFC_DESCRIPTOR)
+ (((WebRtc_Word8*)pInterfaceBuffer) + BufferSize);
+ oneinterface = (PTC_IFC_DESCRIPTOR)
+ ((WebRtc_Word8 *)oneinterface + oneinterface->Length))
+ {
+
+ WebRtc_Word8 interfaceName[500];
+ WideCharToMultiByte(CP_ACP, 0, oneinterface->pInterfaceName, -1,
+ interfaceName, sizeof(interfaceName), 0, 0 );
+
+ PNETWORK_ADDRESS_LIST addresses =
+ &(oneinterface->AddressListDesc.AddressList);
+ for(LONG i = 0; i < addresses->AddressCount ; i++)
+ {
+ // Only look at TCP/IP addresses.
+ if(addresses->Address[i].AddressType != NDIS_PROTOCOL_ID_TCP_IP)
+ {
+ continue;
+ }
+
+ NETWORK_ADDRESS_IP* pIpAddr =
+ (NETWORK_ADDRESS_IP*)&(addresses->Address[i].Address);
+ struct in_addr in;
+ in.S_un.S_addr = pIpAddr->in_addr;
+ if(pIpAddr->in_addr == name->sin_addr.S_un.S_addr)
+ {
+ filterSourceAddress = pIpAddr->in_addr;
+ addrFound = true;
+ }
+ }
+ if(!addrFound)
+ {
+ continue;
+ } else
+ {
+ break;
+ }
+ }
+ if(!addrFound)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS faild since address is not found");
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return -1;
+ }
+ result = _gtc->TcOpenInterfaceW(oneinterface->pInterfaceName, ClientHandle,
+ NULL, &ifcHandle);
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Error opening interface: %d", result);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+ // Create flow if one doesn't exist.
+ if (!_flow)
+ {
+ bool addPCP = ((pcp >= 0) || ((-1 == pcp) && (_pcp >= 0)));
+ int allocSize = sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS) +
+ (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
+ _flow = (PTC_GEN_FLOW)malloc(allocSize);
+
+ _flow->SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
+ _flow->SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
+
+ _flow->ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
+ _flow->ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
+
+ QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
+ dsClass->DSField = 0;
+ dsClass->ObjectHdr.ObjectType = QOS_OBJECT_DS_CLASS;
+ dsClass->ObjectHdr.ObjectLength = sizeof(QOS_DS_CLASS);
+
+ if (addPCP)
+ {
+ QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
+ trafficClass->TrafficClass = 0;
+ trafficClass->ObjectHdr.ObjectType = QOS_OBJECT_TRAFFIC_CLASS;
+ trafficClass->ObjectHdr.ObjectLength = sizeof(QOS_TRAFFIC_CLASS);
+ }
+
+ _flow->TcObjectsLength = sizeof(QOS_DS_CLASS) +
+ (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
+ } else if (-1 != pcp) {
+ // Reallocate memory since pcp has changed.
+ PTC_GEN_FLOW oldFlow = _flow;
+ bool addPCP = (pcp >= 0);
+ int allocSize = sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS) +
+ (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
+ _flow = (PTC_GEN_FLOW)malloc(allocSize);
+
+ // Copy old flow.
+ _flow->ReceivingFlowspec = oldFlow->ReceivingFlowspec;
+ _flow->SendingFlowspec = oldFlow->SendingFlowspec;
+ // The DS info is always the first object.
+ QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
+ QOS_DS_CLASS* oldDsClass = (QOS_DS_CLASS*)oldFlow->TcObjects;
+ dsClass->DSField = oldDsClass->DSField;
+ dsClass->ObjectHdr.ObjectType = oldDsClass->ObjectHdr.ObjectType;
+ dsClass->ObjectHdr.ObjectLength = oldDsClass->ObjectHdr.ObjectLength;
+
+ if (addPCP)
+ {
+ QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
+ trafficClass->TrafficClass = 0;
+ trafficClass->ObjectHdr.ObjectType = QOS_OBJECT_TRAFFIC_CLASS;
+ trafficClass->ObjectHdr.ObjectLength = sizeof(QOS_TRAFFIC_CLASS);
+ }
+
+ _flow->TcObjectsLength = sizeof(QOS_DS_CLASS) +
+ (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
+ free(oldFlow);
+ }
+
+ // Setup send and receive flow and DS object.
+ if (dscp >= 0)
+ {
+ if (!send || (0 == dscp))
+ {
+ _flow->SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ _flow->SendingFlowspec.PeakBandwidth =
+ (0 == dscp ? QOS_NOT_SPECIFIED : POSITIVE_INFINITY_RATE);
+ _flow->SendingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
+ _flow->SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ // 128000 * 10 is 10mbit/s.
+ _flow->SendingFlowspec.TokenRate =
+ (0 == dscp ? QOS_NOT_SPECIFIED : 128000 * 10);
+ }
+ else
+ {
+ _flow->SendingFlowspec.DelayVariation = send->DelayVariation;
+ _flow->SendingFlowspec.Latency = send->Latency;
+ _flow->SendingFlowspec.MaxSduSize = send->MaxSduSize;
+ _flow->SendingFlowspec.MinimumPolicedSize =
+ send->MinimumPolicedSize;
+ _flow->SendingFlowspec.PeakBandwidth = send->PeakBandwidth;
+ _flow->SendingFlowspec.PeakBandwidth = POSITIVE_INFINITY_RATE;
+ _flow->SendingFlowspec.ServiceType = send->ServiceType;
+ _flow->SendingFlowspec.TokenBucketSize = send->TokenBucketSize;
+ _flow->SendingFlowspec.TokenRate = send->TokenRate;
+ }
+
+ if (!recv || (0 == dscp))
+ {
+ _flow->ReceivingFlowspec.DelayVariation =
+ _flow->SendingFlowspec.DelayVariation;
+ _flow->ReceivingFlowspec.Latency = _flow->SendingFlowspec.Latency;
+ _flow->ReceivingFlowspec.MaxSduSize =
+ _flow->SendingFlowspec.MaxSduSize;
+ _flow->ReceivingFlowspec.MinimumPolicedSize =
+ _flow->SendingFlowspec.MinimumPolicedSize;
+ _flow->ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ _flow->ReceivingFlowspec.ServiceType =
+ 0 == dscp ? SERVICETYPE_BESTEFFORT : SERVICETYPE_CONTROLLEDLOAD;
+ _flow->ReceivingFlowspec.TokenBucketSize =
+ _flow->SendingFlowspec.TokenBucketSize;
+ _flow->ReceivingFlowspec.TokenRate =
+ _flow->SendingFlowspec.TokenRate;
+ } else {
+ _flow->ReceivingFlowspec.DelayVariation = recv->DelayVariation;
+ _flow->ReceivingFlowspec.Latency = recv->Latency;
+ _flow->ReceivingFlowspec.MaxSduSize = recv->MaxSduSize;
+ _flow->ReceivingFlowspec.MinimumPolicedSize =
+ recv->MinimumPolicedSize;
+ _flow->ReceivingFlowspec.PeakBandwidth = recv->PeakBandwidth;
+ _flow->ReceivingFlowspec.ServiceType = recv->ServiceType;
+ _flow->ReceivingFlowspec.TokenBucketSize = recv->TokenBucketSize;
+ _flow->ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
+ }
+
+ // Setup DS (for DSCP value).
+ // DS is always the first object.
+ QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
+ dsClass->DSField = dscp;
+ }
+
+ // Setup PCP (802.1p priority in 802.1Q/VLAN tagging)
+ if (pcp >= 0)
+ {
+ // DS is always first object.
+ QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
+ QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
+ trafficClass->TrafficClass = pcp;
+ }
+
+ result = _gtc->TcAddFlow(ifcHandle, NULL, 0, _flow, &iflowHandle);
+ if(result != NO_ERROR)
+ {
+ _gtc->TcCloseInterface(ifcHandle);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return -1;
+ }
+
+ IP_PATTERN filterPattern, mask;
+
+ ZeroMemory((WebRtc_Word8*)&filterPattern, sizeof(IP_PATTERN));
+ ZeroMemory((WebRtc_Word8*)&mask, sizeof(IP_PATTERN));
+
+ filterPattern.ProtocolId = IPPROTO_UDP;
+ // "name" fields already in network order.
+ filterPattern.S_un.S_un_ports.s_srcport = name->sin_port;
+ filterPattern.SrcAddr = filterSourceAddress;
+
+ // Unsigned max of a type corresponds to a bitmask with all bits set to 1.
+ // I.e. the filter should allow all ProtocolIds, any source port and any
+ // IP address
+ mask.ProtocolId = UCHAR_MAX;
+ mask.S_un.S_un_ports.s_srcport = USHRT_MAX;
+ mask.SrcAddr = ULONG_MAX;
+
+ TC_GEN_FILTER filter;
+
+ filter.AddressType = NDIS_PROTOCOL_ID_TCP_IP;
+ filter.Mask = (LPVOID)&mask;
+ filter.Pattern = (LPVOID)&filterPattern;
+ filter.PatternSize = sizeof(IP_PATTERN);
+
+ result = _gtc->TcAddFilter(iflowHandle, &filter, &iFilterHandle);
+ if(result != NO_ERROR)
+ {
+ _gtc->TcDeleteFlow(iflowHandle);
+ _gtc->TcCloseInterface(ifcHandle);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+ _flowHandle = iflowHandle;
+ _filterHandle = iFilterHandle;
+ _clientHandle = ClientHandle;
+ if (-1 != pcp)
+ {
+ _pcp = pcp;
+ }
+
+ _gtc->TcCloseInterface(ifcHandle);
+ free(pInterfaceBuffer);
+
+ return 0;
+}
+
+WebRtc_Word32 UdpSocket2Windows::CreateFlowSpec(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize,
+ FLOWSPEC* f)
+{
+ if (!f)
+ {
+ return -1;
+ }
+
+ f->ServiceType = serviceType;
+ f->TokenRate = tokenRate;
+ f->TokenBucketSize = QOS_NOT_SPECIFIED;
+ f->PeakBandwidth = QOS_NOT_SPECIFIED;
+ f->DelayVariation = QOS_NOT_SPECIFIED;
+ f->Latency = QOS_NOT_SPECIFIED;
+ f->MaxSduSize = QOS_NOT_SPECIFIED;
+ f->MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ return 0;
+}
+
+bool UdpSocket2Windows::NewOutstandingCall()
+{
+ assert(_outstandingCallsDisabled.Value() == 0);
+
+ const WebRtc_Word32 outstandingCalls = ++_outstandingCalls;
+ return true;
+}
+
+void UdpSocket2Windows::OutstandingCallCompleted()
+{
+ _ptrDestRWLock->AcquireLockShared();
+ ++_outstandingCallComplete;
+ if((--_outstandingCalls == 0) && (_outstandingCallsDisabled.Value() == 1))
+ {
+ // When there are no outstanding calls and new outstandning calls are
+ // disabled it is time to terminate.
+ _terminate = true;
+ }
+ _ptrDestRWLock->ReleaseLockShared();
+
+ if((--_outstandingCallComplete == 0) &&
+ (_terminate))
+ {
+ // Only one thread will enter here. The thread with the last outstanding
+ // call.
+ CriticalSectionScoped cs(_ptrDeleteCrit);
+ _safeTodelete = true;
+ _ptrDeleteCond->Wake();
+ }
+}
+
+void UdpSocket2Windows::DisableNewOutstandingCalls()
+{
+ _ptrDestRWLock->AcquireLockExclusive();
+ if(_outstandingCallsDisabled.Value() == 1)
+ {
+ // Outstandning calls are already disabled.
+ _ptrDestRWLock->ReleaseLockExclusive();
+ return;
+ }
+ _outstandingCallsDisabled = 1;
+ const bool noOutstandingCalls = (_outstandingCalls.Value() == 0);
+ _ptrDestRWLock->ReleaseLockExclusive();
+
+ RemoveSocketFromManager();
+
+ if(noOutstandingCalls)
+ {
+ CriticalSectionScoped cs(_ptrDeleteCrit);
+ _safeTodelete = true;
+ _ptrDeleteCond->Wake();
+ }
+}
+
+void UdpSocket2Windows::WaitForOutstandingCalls()
+{
+ CriticalSectionScoped cs(_ptrDeleteCrit);
+ while(!_safeTodelete)
+ {
+ _ptrDeleteCond->SleepCS(*_ptrDeleteCrit);
+ }
+}
+
+void UdpSocket2Windows::RemoveSocketFromManager()
+{
+ // New outstanding calls should be disabled at this point.
+ assert(_outstandingCallsDisabled.Value() != 0);
+
+ if(_addedToMgr)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "calling UdpSocketManager::RemoveSocket()");
+ if(_mgr->RemoveSocket(this))
+ {
+ _addedToMgr=false;
+ }
+ }
+}
+
+bool UdpSocket2Windows::AquireSocket()
+{
+ _ptrSocketRWLock->AcquireLockShared();
+ const bool returnValue = _socket != INVALID_SOCKET;
+ if(!returnValue)
+ {
+ _ptrSocketRWLock->ReleaseLockShared();
+ }
+ return returnValue;
+}
+
+void UdpSocket2Windows::ReleaseSocket()
+{
+ _ptrSocketRWLock->ReleaseLockShared();
+}
+
+bool UdpSocket2Windows::InvalidateSocket()
+{
+ _ptrSocketRWLock->AcquireLockExclusive();
+ if(_socket == INVALID_SOCKET)
+ {
+ _ptrSocketRWLock->ReleaseLockExclusive();
+ return true;
+ }
+ // Give the socket back to the system. All socket calls will fail from now
+ // on.
+ if(closesocket(_socket) == SOCKET_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows(%d)::InvalidateSocket() WSAerror: %d",
+ (WebRtc_Word32)this, WSAGetLastError());
+ }
+ _socket = INVALID_SOCKET;
+ _ptrSocketRWLock->ReleaseLockExclusive();
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_windows.h b/trunk/src/modules/udp_transport/source/udp_socket2_windows.h
new file mode 100644
index 0000000..c0d68bb
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket2_windows.h
@@ -0,0 +1,171 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
+
+// Disable deprication warning from traffic.h
+#pragma warning(disable : 4995)
+
+// Don't change include order for these header files.
+#include <Winsock2.h>
+#include <Ntddndis.h>
+#include <traffic.h>
+
+#include "atomic32_wrapper.h"
+#include "condition_variable_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "list_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+#include "udp_socket_wrapper.h"
+#include "udp_socket2_manager_windows.h"
+
+namespace webrtc {
+class UdpSocket2ManagerWindows;
+class TrafficControlWindows;
+struct PerIoContext;
+
+class UdpSocket2Windows : public UdpSocketWrapper
+{
+public:
+ UdpSocket2Windows(const WebRtc_Word32 id, UdpSocketManager* mgr,
+ bool ipV6Enable = false, bool disableGQOS = false);
+ virtual ~UdpSocket2Windows();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool ValidHandle();
+
+ virtual bool SetCallback(CallbackObj, IncomingSocketCallback);
+
+ virtual bool Bind(const SocketAddress& name);
+ virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval, WebRtc_Word32 optlen);
+
+ virtual bool StartReceiving(const WebRtc_UWord32 receiveBuffers);
+ virtual inline bool StartReceiving() {return StartReceiving(8);}
+ virtual bool StopReceiving();
+
+ virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
+ const SocketAddress& to);
+
+ virtual void CloseBlocking();
+
+ virtual SOCKET GetFd() { return _socket;}
+ virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
+ const SocketAddress &stRemName,
+ WebRtc_Word32 overrideDSCP = 0);
+
+ virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
+ virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 pcp);
+
+ virtual WebRtc_UWord32 ReceiveBuffers(){return _receiveBuffers.Value();}
+
+protected:
+ void IOCompleted(PerIoContext* pIOContext, WebRtc_UWord32 ioSize,
+ WebRtc_UWord32 error);
+
+ WebRtc_Word32 PostRecv();
+ // Use pIoContext to post a new WSARecvFrom(..).
+ WebRtc_Word32 PostRecv(PerIoContext* pIoContext);
+
+private:
+ friend class UdpSocket2WorkerWindows;
+
+ // Set traffic control (TC) flow adding it the interface that matches this
+ // sockets address.
+ // A filter is created and added to the flow.
+ // The flow consists of:
+ // (1) QoS send and receive information (flow specifications).
+ // (2) A DS object (for specifying exact DSCP value).
+ // (3) Possibly a traffic object (for specifying exact 802.1p priority (PCP)
+ // value).
+ //
+ // dscp values:
+ // -1 don't change the current dscp value.
+ // 0 don't add any flow to TC, unless pcp is specified.
+ // 1-63 Add a flow to TC with the specified dscp value.
+ // pcp values:
+ // -2 Don't add pcp info to the flow, (3) will not be added.
+ // -1 Don't change the current value.
+ // 0-7 Add pcp info to the flow with the specified value,
+ // (3) will be added.
+ //
+ // If both dscp and pcp are -1 no flow will be created or added to TC.
+ // If dscp is 0 and pcp is 0-7 (1), (2) and (3) will be created.
+ // Note: input parameter values are assumed to be in valid range, checks
+ // must be done by caller.
+ WebRtc_Word32 SetTrafficControl(WebRtc_Word32 dscp, WebRtc_Word32 pcp,
+ const struct sockaddr_in* name,
+ FLOWSPEC* send = NULL,
+ FLOWSPEC* recv = NULL);
+ WebRtc_Word32 CreateFlowSpec(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize, FLOWSPEC *f);
+
+ WebRtc_Word32 _id;
+ RWLockWrapper* _ptrCbRWLock;
+ IncomingSocketCallback _incomingCb;
+ CallbackObj _obj;
+ bool _qos;
+
+ SocketAddress _remoteAddr;
+ SOCKET _socket;
+ WebRtc_Word32 _iProtocol;
+ UdpSocket2ManagerWindows* _mgr;
+
+ CriticalSectionWrapper* _pCrit;
+ Atomic32Wrapper _outstandingCalls;
+ Atomic32Wrapper _outstandingCallComplete;
+ volatile bool _terminate;
+ volatile bool _addedToMgr;
+
+ CriticalSectionWrapper* _ptrDeleteCrit;
+ ConditionVariableWrapper* _ptrDeleteCond;
+ bool _safeTodelete;
+
+ RWLockWrapper* _ptrDestRWLock;
+ Atomic32Wrapper _outstandingCallsDisabled; // 0 = false, 1 = true
+ bool NewOutstandingCall();
+ void OutstandingCallCompleted();
+ void DisableNewOutstandingCalls();
+ void WaitForOutstandingCalls();
+
+ void RemoveSocketFromManager();
+
+ // RWLockWrapper is used as a reference counter for the socket. Write lock
+ // is used for creating and deleting socket. Read lock is used for
+ // accessing the socket.
+ RWLockWrapper* _ptrSocketRWLock;
+ bool AquireSocket();
+ void ReleaseSocket();
+ bool InvalidateSocket();
+
+ // Traffic control handles and structure pointers.
+ HANDLE _clientHandle;
+ HANDLE _flowHandle;
+ HANDLE _filterHandle;
+ PTC_GEN_FLOW _flow;
+ // TrafficControlWindows implements TOS and PCP.
+ TrafficControlWindows* _gtc;
+ // Holds the current pcp value. Can be -2 or 0 - 7.
+ int _pcp;
+
+ Atomic32Wrapper _receiveBuffers;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc
new file mode 100644
index 0000000..889fb2f
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc
@@ -0,0 +1,429 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_manager_posix.h"
+
+#include <strings.h>
+#include <sys/time.h>
+#include <sys/types.h>
+#include <time.h>
+#include <unistd.h>
+
+#include "trace.h"
+#include "udp_socket_posix.h"
+
+namespace webrtc {
+UdpSocketManagerPosix::UdpSocketManagerPosix()
+ : UdpSocketManager(),
+ _id(-1),
+ _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _numberOfSocketMgr(-1),
+ _incSocketMgrNextTime(0),
+ _nextSocketMgrToAssign(0),
+ _socketMgr()
+{
+}
+
+bool UdpSocketManagerPosix::Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads) {
+ CriticalSectionScoped cs(_critSect);
+ if ((_id != -1) || (_numOfWorkThreads != 0)) {
+ assert(_id != -1);
+ assert(_numOfWorkThreads != 0);
+ return false;
+ }
+
+ _id = id;
+ _numberOfSocketMgr = numOfWorkThreads;
+ _numOfWorkThreads = numOfWorkThreads;
+
+ if(MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX < _numberOfSocketMgr)
+ {
+ _numberOfSocketMgr = MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX;
+ }
+ for(int i = 0;i < _numberOfSocketMgr; i++)
+ {
+ _socketMgr[i] = new UdpSocketManagerPosixImpl();
+ }
+ return true;
+}
+
+
+UdpSocketManagerPosix::~UdpSocketManagerPosix()
+{
+ Stop();
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketManagerPosix(%d)::UdpSocketManagerPosix()",
+ _numberOfSocketMgr);
+
+ for(int i = 0;i < _numberOfSocketMgr; i++)
+ {
+ delete _socketMgr[i];
+ }
+ delete _critSect;
+}
+
+WebRtc_Word32 UdpSocketManagerPosix::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+bool UdpSocketManagerPosix::Start()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketManagerPosix(%d)::Start()",
+ _numberOfSocketMgr);
+
+ _critSect->Enter();
+ bool retVal = true;
+ for(int i = 0;i < _numberOfSocketMgr && retVal; i++)
+ {
+ retVal = _socketMgr[i]->Start();
+ }
+ if(!retVal)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocketManagerPosix(%d)::Start() error starting socket managers",
+ _numberOfSocketMgr);
+ }
+ _critSect->Leave();
+ return retVal;
+}
+
+bool UdpSocketManagerPosix::Stop()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketManagerPosix(%d)::Stop()",_numberOfSocketMgr);
+
+ _critSect->Enter();
+ bool retVal = true;
+ for(int i = 0; i < _numberOfSocketMgr && retVal; i++)
+ {
+ retVal = _socketMgr[i]->Stop();
+ }
+ if(!retVal)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocketManagerPosix(%d)::Stop() there are still active socket "
+ "managers",
+ _numberOfSocketMgr);
+ }
+ _critSect->Leave();
+ return retVal;
+}
+
+bool UdpSocketManagerPosix::AddSocket(UdpSocketWrapper* s)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketManagerPosix(%d)::AddSocket()",_numberOfSocketMgr);
+
+ _critSect->Enter();
+ bool retVal = _socketMgr[_nextSocketMgrToAssign]->AddSocket(s);
+ if(!retVal)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocketManagerPosix(%d)::AddSocket() failed to add socket to\
+ manager",
+ _numberOfSocketMgr);
+ }
+
+ // Distribute sockets on UdpSocketManagerPosixImpls in a round-robin
+ // fashion.
+ if(_incSocketMgrNextTime == 0)
+ {
+ _incSocketMgrNextTime++;
+ } else {
+ _incSocketMgrNextTime = 0;
+ _nextSocketMgrToAssign++;
+ if(_nextSocketMgrToAssign >= _numberOfSocketMgr)
+ {
+ _nextSocketMgrToAssign = 0;
+ }
+ }
+ _critSect->Leave();
+ return retVal;
+}
+
+bool UdpSocketManagerPosix::RemoveSocket(UdpSocketWrapper* s)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketManagerPosix(%d)::RemoveSocket()",
+ _numberOfSocketMgr);
+
+ _critSect->Enter();
+ bool retVal = false;
+ for(int i = 0;i < _numberOfSocketMgr && (retVal == false); i++)
+ {
+ retVal = _socketMgr[i]->RemoveSocket(s);
+ }
+ if(!retVal)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpSocketManagerPosix(%d)::RemoveSocket() failed to remove socket\
+ from manager",
+ _numberOfSocketMgr);
+ }
+ _critSect->Leave();
+ return retVal;
+}
+
+
+UdpSocketManagerPosixImpl::UdpSocketManagerPosixImpl()
+{
+ _critSectList = CriticalSectionWrapper::CreateCriticalSection();
+ _thread = ThreadWrapper::CreateThread(UdpSocketManagerPosixImpl::Run, this,
+ kRealtimePriority,
+ "UdpSocketManagerPosixImplThread");
+ FD_ZERO(&_readFds);
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
+ "UdpSocketManagerPosix created");
+}
+
+UdpSocketManagerPosixImpl::~UdpSocketManagerPosixImpl()
+{
+ if(_thread != NULL)
+ {
+ delete _thread;
+ }
+
+ if (_critSectList != NULL)
+ {
+ UpdateSocketMap();
+
+ _critSectList->Enter();
+
+ MapItem* item = _socketMap.First();
+ while(item)
+ {
+ UdpSocketPosix* s = static_cast<UdpSocketPosix*>(item->GetItem());
+ _socketMap.Erase(item);
+ item = _socketMap.First();
+ delete s;
+ }
+ _critSectList->Leave();
+
+ delete _critSectList;
+ }
+
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
+ "UdpSocketManagerPosix deleted");
+}
+
+bool UdpSocketManagerPosixImpl::Start()
+{
+ unsigned int id = 0;
+ if (_thread == NULL)
+ {
+ return false;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
+ "Start UdpSocketManagerPosix");
+ return _thread->Start(id);
+}
+
+bool UdpSocketManagerPosixImpl::Stop()
+{
+ if (_thread == NULL)
+ {
+ return true;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
+ "Stop UdpSocketManagerPosix");
+ return _thread->Stop();
+}
+
+bool UdpSocketManagerPosixImpl::Process()
+{
+ bool doSelect = false;
+ // Timeout = 1 second.
+ struct timeval timeout;
+ timeout.tv_sec = 0;
+ timeout.tv_usec = 10000;
+ MapItem* it;
+
+ FD_ZERO(&_readFds);
+
+ UpdateSocketMap();
+
+ unsigned int maxFd = 0;
+ for (it = _socketMap.First(); it != NULL; it=_socketMap.Next(it))
+ {
+ doSelect = true;
+ maxFd = maxFd > it->GetUnsignedId() ? maxFd : it->GetUnsignedId();
+ FD_SET(it->GetUnsignedId(), &_readFds);
+
+ maxFd = maxFd > it->GetUnsignedId() ? maxFd : it->GetUnsignedId();
+ doSelect = true;
+ }
+
+ int num = 0;
+ if (doSelect)
+ {
+ num = select(maxFd+1, &_readFds, NULL, NULL, &timeout);
+
+ if (num == SOCKET_ERROR)
+ {
+ // Timeout = 10 ms.
+ timespec t;
+ t.tv_sec = 0;
+ t.tv_nsec = 10000*1000;
+ nanosleep(&t, NULL);
+ return true;
+ }
+ }else
+ {
+ // Timeout = 10 ms.
+ timespec t;
+ t.tv_sec = 0;
+ t.tv_nsec = 10000*1000;
+ nanosleep(&t, NULL);
+ return true;
+ }
+
+ for (it = _socketMap.First(); it != NULL && num > 0;
+ it = _socketMap.Next(it))
+ {
+ UdpSocketPosix* s = static_cast<UdpSocketPosix*>(it->GetItem());
+ if (FD_ISSET(it->GetUnsignedId(), &_readFds))
+ {
+ s->HasIncoming();
+ num--;
+ }
+ }
+ return true;
+}
+
+bool UdpSocketManagerPosixImpl::Run(ThreadObj obj)
+{
+ UdpSocketManagerPosixImpl* mgr =
+ static_cast<UdpSocketManagerPosixImpl*>(obj);
+ return mgr->Process();
+}
+
+bool UdpSocketManagerPosixImpl::AddSocket(UdpSocketWrapper* s)
+{
+ UdpSocketPosix* sl = static_cast<UdpSocketPosix*>(s);
+ if(sl->GetFd() == INVALID_SOCKET || !(sl->GetFd() < FD_SETSIZE))
+ {
+ return false;
+ }
+ _critSectList->Enter();
+ _addList.PushBack(s);
+ _critSectList->Leave();
+ return true;
+}
+
+bool UdpSocketManagerPosixImpl::RemoveSocket(UdpSocketWrapper* s)
+{
+ // Put in remove list if this is the correct UdpSocketManagerPosixImpl.
+ _critSectList->Enter();
+
+ // If the socket is in the add list it's safe to remove and delete it.
+ ListItem* addListItem = _addList.First();
+ while(addListItem)
+ {
+ UdpSocketPosix* addSocket = (UdpSocketPosix*)addListItem->GetItem();
+ unsigned int addFD = addSocket->GetFd();
+ unsigned int removeFD = static_cast<UdpSocketPosix*>(s)->GetFd();
+ if(removeFD == addFD)
+ {
+ _removeList.PushBack(removeFD);
+ _critSectList->Leave();
+ return true;
+ }
+ addListItem = _addList.Next(addListItem);
+ }
+
+ // Checking the socket map is safe since all Erase and Insert calls to this
+ // map are also protected by _critSectList.
+ if(_socketMap.Find(static_cast<UdpSocketPosix*>(s)->GetFd()) != NULL)
+ {
+ _removeList.PushBack(static_cast<UdpSocketPosix*>(s)->GetFd());
+ _critSectList->Leave();
+ return true;
+ }
+ _critSectList->Leave();
+ return false;
+}
+
+void UdpSocketManagerPosixImpl::UpdateSocketMap()
+{
+ // Remove items in remove list.
+ _critSectList->Enter();
+ while(!_removeList.Empty())
+ {
+ UdpSocketPosix* deleteSocket = NULL;
+ unsigned int removeFD = _removeList.First()->GetUnsignedItem();
+
+ // If the socket is in the add list it hasn't been added to the socket
+ // map yet. Just remove the socket from the add list.
+ ListItem* addListItem = _addList.First();
+ while(addListItem)
+ {
+ UdpSocketPosix* addSocket = (UdpSocketPosix*)addListItem->GetItem();
+ unsigned int addFD = addSocket->GetFd();
+ if(removeFD == addFD)
+ {
+ deleteSocket = addSocket;
+ _addList.Erase(addListItem);
+ break;
+ }
+ addListItem = _addList.Next(addListItem);
+ }
+
+ // Find and remove socket from _socketMap.
+ MapItem* it = _socketMap.Find(removeFD);
+ if(it != NULL)
+ {
+ UdpSocketPosix* socket =
+ static_cast<UdpSocketPosix*>(it->GetItem());
+ if(socket)
+ {
+ deleteSocket = socket;
+ }
+ _socketMap.Erase(it);
+ }
+ if(deleteSocket)
+ {
+ deleteSocket->ReadyForDeletion();
+ delete deleteSocket;
+ }
+ _removeList.PopFront();
+ }
+
+ // Add sockets from add list.
+ while(!_addList.Empty())
+ {
+ UdpSocketPosix* s =
+ static_cast<UdpSocketPosix*>(_addList.First()->GetItem());
+ if(s)
+ {
+ _socketMap.Insert(s->GetFd(), s);
+ }
+ _addList.PopFront();
+ }
+ _critSectList->Leave();
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h
new file mode 100644
index 0000000..c89aa13
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
+
+#include <sys/types.h>
+#include <unistd.h>
+
+#include "critical_section_wrapper.h"
+#include "list_wrapper.h"
+#include "map_wrapper.h"
+#include "thread_wrapper.h"
+#include "udp_socket_manager_wrapper.h"
+#include "udp_socket_wrapper.h"
+
+#define MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX 8
+
+namespace webrtc {
+
+class ConditionVariableWrapper;
+class UdpSocketManagerPosixImpl;
+
+class UdpSocketManagerPosix : public UdpSocketManager
+{
+public:
+ UdpSocketManagerPosix();
+ virtual ~UdpSocketManagerPosix();
+
+ virtual bool Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads);
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool Start();
+ virtual bool Stop();
+
+ virtual bool AddSocket(UdpSocketWrapper* s);
+ virtual bool RemoveSocket(UdpSocketWrapper* s);
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper* _critSect;
+ WebRtc_UWord8 _numberOfSocketMgr;
+ WebRtc_UWord8 _incSocketMgrNextTime;
+ WebRtc_UWord8 _nextSocketMgrToAssign;
+ UdpSocketManagerPosixImpl* _socketMgr[MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX];
+};
+
+class UdpSocketManagerPosixImpl
+{
+public:
+ UdpSocketManagerPosixImpl();
+ virtual ~UdpSocketManagerPosixImpl();
+
+ virtual bool Start();
+ virtual bool Stop();
+
+ virtual bool AddSocket(UdpSocketWrapper* s);
+ virtual bool RemoveSocket(UdpSocketWrapper* s);
+
+protected:
+ static bool Run(ThreadObj obj);
+ bool Process();
+ void UpdateSocketMap();
+
+private:
+ ThreadWrapper* _thread;
+ CriticalSectionWrapper* _critSectList;
+
+ fd_set _readFds;
+
+ MapWrapper _socketMap;
+ ListWrapper _addList;
+ ListWrapper _removeList;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc
new file mode 100644
index 0000000..d817d74
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc
@@ -0,0 +1,283 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_manager_windows.h"
+#include "udp_socket_windows.h"
+
+namespace webrtc {
+WebRtc_UWord32 UdpSocketManagerWindows::_numOfActiveManagers = 0;
+
+UdpSocketManagerWindows::UdpSocketManagerWindows()
+ : UdpSocketManager(),
+ _id(-1)
+{
+ const WebRtc_Word8* threadName = "UdpSocketManagerWindows_Thread";
+ _critSectList = CriticalSectionWrapper::CreateCriticalSection();
+ _thread = ThreadWrapper::CreateThread(UdpSocketManagerWindows::Run,
+ this, kRealtimePriority, threadName);
+ FD_ZERO(&_readFds);
+ FD_ZERO(&_writeFds);
+ FD_ZERO(&_exceptFds);
+ _numOfActiveManagers++;
+}
+
+bool UdpSocketManagerWindows::Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads) {
+ CriticalSectionScoped cs(_critSectList);
+ if ((_id != -1) || (_numOfWorkThreads != 0)) {
+ assert(_id == -1);
+ assert(_numOfWorkThreads == 0);
+ return false;
+ }
+ _id = id;
+ _numOfWorkThreads = numOfWorkThreads;
+ return true;
+}
+
+UdpSocketManagerWindows::~UdpSocketManagerWindows()
+{
+ Stop();
+ if(_thread != NULL)
+ {
+ delete _thread;
+ }
+
+ if (_critSectList != NULL)
+ {
+ _critSectList->Enter();
+
+ while(!_socketMap.empty())
+ {
+ std::map<SOCKET, UdpSocketWindows*>::iterator it =
+ _socketMap.begin();
+ UdpSocketWindows* s = static_cast<UdpSocketWindows*>(it->second);
+ _socketMap.erase(it);
+ delete s;
+ }
+ _removeList.erase(_removeList.begin(), _removeList.end());
+
+ while(!_addList.empty())
+ {
+ std::list<UdpSocketWindows*>::iterator it = _addList.begin();
+ UdpSocketWindows* s = static_cast<UdpSocketWindows*>(*it);
+ _addList.erase(it);
+ delete s;
+ }
+ _critSectList->Leave();
+
+ delete _critSectList;
+ }
+
+ _numOfActiveManagers--;
+
+ if (_numOfActiveManagers == 0)
+ WSACleanup();
+}
+
+WebRtc_Word32 UdpSocketManagerWindows::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+bool UdpSocketManagerWindows::Start()
+{
+ unsigned int id;
+ if (_thread == NULL)
+ return false;
+
+ return _thread->Start(id);
+}
+
+bool UdpSocketManagerWindows::Stop()
+{
+ if (_thread == NULL)
+ return true;
+
+ return _thread->Stop();
+}
+
+bool UdpSocketManagerWindows::Process()
+{
+ bool doSelect = false;
+ // Timeout = 1 second.
+ timeval timeout;
+ timeout.tv_sec = 0;
+ timeout.tv_usec = 10000;
+
+ FD_ZERO(&_readFds);
+ FD_ZERO(&_writeFds);
+ FD_ZERO(&_exceptFds);
+
+ _critSectList->Enter();
+ // Remove sockets that have been registered for removal.
+ while(!_removeList.empty())
+ {
+ SOCKET id = *_removeList.begin();
+ std::map<SOCKET, UdpSocketWindows*>::iterator it = _socketMap.find(id);
+ if(it != _socketMap.end())
+ {
+ UdpSocketWindows* s = static_cast<UdpSocketWindows*>(it->second);
+ _socketMap.erase(it);
+ _removeList.pop_front();
+ delete s;
+ }
+ }
+
+ // Add sockets that have been registered for being added.
+ while (!_addList.empty())
+ {
+ UdpSocketWindows* s = *_addList.begin();
+ if(s)
+ {
+ _socketMap[s->GetFd()] = s;
+ }
+ _addList.pop_front();
+ }
+ _critSectList->Leave();
+
+ std::map<SOCKET, UdpSocketWindows*>::iterator it = _socketMap.begin();
+ while(it != _socketMap.end())
+ {
+ UdpSocketWindows* s = it->second;
+ if (s->WantsIncoming())
+ {
+ doSelect = true;
+ FD_SET(it->first, &_readFds);
+ }
+ if(!s->IsWritable())
+ {
+ FD_SET(it->first, &_writeFds);
+ doSelect = true;
+ }
+ it++;
+ }
+
+ WebRtc_Word32 num = 0;
+ if (doSelect)
+ {
+ num = select(0, &_readFds, &_writeFds, &_exceptFds, &timeout);
+ if (num == SOCKET_ERROR)
+ {
+ Sleep(10);
+ return true;
+ }
+ }else
+ {
+ Sleep(10);
+ return true;
+ }
+
+ it = _socketMap.begin();
+ while (it != _socketMap.end() && num > 0)
+ {
+ if (FD_ISSET(it->first, &_readFds))
+ {
+ static_cast<UdpSocketWindows*>(it->second)->HasIncoming();
+ num--;
+ }
+ if (FD_ISSET(it->first, &_writeFds))
+ {
+ // Socket available for writing.
+ static_cast<UdpSocketWindows*>(it->second)->SetWritable();
+ num--;
+ }
+ }
+ return true;
+}
+
+bool UdpSocketManagerWindows::Run(ThreadObj obj)
+{
+ UdpSocketManagerWindows* mgr = static_cast<UdpSocketManagerWindows*>(obj);
+ return mgr->Process();
+};
+
+bool UdpSocketManagerWindows::AddSocket(UdpSocketWrapper* s)
+{
+ UdpSocketWindows* winSock = static_cast<UdpSocketWindows*>(s);
+
+ _critSectList->Enter();
+ std::map<SOCKET, UdpSocketWindows*>::iterator it =
+ _socketMap.find(winSock->GetFd());
+ if (it != _socketMap.end())
+ {
+ if (!_removeList.empty())
+ {
+ // File descriptors are re-used so it's possible that a socket has
+ // been added with the same file descriptor as a socket that is to
+ // be removed. I.e. the socket that is to be removed is no longer
+ // in use, delete it.
+ // TODO (hellner): removing items from _socketMap may cause race
+ // condition. Fix this.
+ std::list<SOCKET>::iterator removeIt = _removeList.begin();
+ while(removeIt != _removeList.end())
+ {
+ if (*removeIt == winSock->GetFd())
+ {
+ it = _socketMap.find(*removeIt);
+ UdpSocketWindows* delete_socket = it->second;
+ _socketMap.erase(it);
+ _removeList.erase(removeIt);
+ delete delete_socket;
+ _addList.push_back(winSock);
+ _critSectList->Leave();
+ return true;
+ }
+ removeIt++;
+ }
+ }
+ _critSectList->Leave();
+ return false;
+ }
+
+ _addList.push_back(winSock);
+ _critSectList->Leave();
+ return true;
+}
+
+bool UdpSocketManagerWindows::RemoveSocket(UdpSocketWrapper* s)
+{
+ UdpSocketWindows* winSock = static_cast<UdpSocketWindows*>(s);
+
+ _critSectList->Enter();
+ // If socket is in the add list its safe to just remove it from the list.
+ if (!_addList.empty())
+ {
+ std::list<UdpSocketWindows*>::iterator it = _addList.begin();
+ while(it != _addList.end())
+ {
+ UdpSocketWindows* tempSocket = (*it);
+ if (tempSocket->GetFd() == winSock->GetFd())
+ {
+ _addList.erase(it);
+ delete winSock;
+ _critSectList->Leave();
+ return true;
+ }
+ it++;
+ }
+ }
+
+ // If the socket is not even added to the UdpSocketManagerWindows it's
+ // safe to delete the socket.
+ std::map<SOCKET, UdpSocketWindows*>::iterator findIt =
+ _socketMap.find(winSock->GetFd());
+ if (findIt == _socketMap.end())
+ {
+ delete winSock;
+ _critSectList->Leave();
+ return false;
+ }
+
+ _removeList.push_back(winSock->GetFd());
+ _critSectList->Leave();
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h
new file mode 100644
index 0000000..06d905a
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
+
+#define FD_SETSIZE 1024
+
+#include <winsock2.h>
+#include <map>
+#include <list>
+
+// Don't change the include order.
+// TODO (hellner): all header files should be compilable separately. Fix this.
+#include "udp_socket_manager_wrapper.h"
+#include "thread_wrapper.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+class UdpSocketWindows;
+
+class UdpSocketManagerWindows : public UdpSocketManager
+{
+public:
+ UdpSocketManagerWindows();
+ virtual ~UdpSocketManagerWindows();
+
+ virtual bool Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads);
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool Start();
+ virtual bool Stop();
+
+ virtual bool AddSocket(UdpSocketWrapper* s);
+ virtual bool RemoveSocket(UdpSocketWrapper* s);
+
+protected:
+ static bool Run(ThreadObj obj);
+ bool Process();
+
+private:
+ WebRtc_Word32 _id;
+ ThreadWrapper* _thread;
+
+ fd_set _readFds;
+ fd_set _writeFds;
+ fd_set _exceptFds;
+
+ CriticalSectionWrapper* _critSectList;
+
+ std::map<SOCKET, UdpSocketWindows*> _socketMap;
+ std::list<UdpSocketWindows*> _addList;
+ std::list<SOCKET> _removeList;
+
+ static WebRtc_UWord32 _numOfActiveManagers;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc
new file mode 100644
index 0000000..9f1162e
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_manager_wrapper.h"
+
+#include <cassert>
+
+#ifdef _WIN32
+#include "fix_interlocked_exchange_pointer_win.h"
+#include "udp_socket_manager_windows.h"
+#include "udp_socket2_manager_windows.h"
+#else
+#include "udp_socket_manager_posix.h"
+#endif
+
+namespace webrtc {
+UdpSocketManager* UdpSocketManager::CreateInstance()
+{
+#if defined(_WIN32)
+ #if (defined(USE_WINSOCK2))
+ return static_cast<UdpSocketManager*>(
+ new UdpSocket2ManagerWindows());
+ #else
+ numOfWorkThreads = 1;
+ return static_cast<UdpSocketManager*>(
+ new UdpSocketManagerWindows());
+ #endif
+#else
+ return new UdpSocketManagerPosix();
+#endif
+}
+
+UdpSocketManager* UdpSocketManager::StaticInstance(
+ CountOperation count_operation,
+ const WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads)
+{
+ UdpSocketManager* impl =
+ GetStaticInstance<UdpSocketManager>(count_operation);
+ if (count_operation == kAddRef && impl != NULL) {
+ if (impl->Init(id, numOfWorkThreads)) {
+ impl->Start();
+ }
+ }
+ return impl;
+}
+
+UdpSocketManager* UdpSocketManager::Create(const WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads)
+{
+ return UdpSocketManager::StaticInstance(kAddRef, id,
+ numOfWorkThreads);
+}
+
+void UdpSocketManager::Return()
+{
+ WebRtc_UWord8 numOfWorkThreads = 0;
+ UdpSocketManager::StaticInstance(kRelease, -1,
+ numOfWorkThreads);
+}
+
+UdpSocketManager::UdpSocketManager() : _numOfWorkThreads(0)
+{
+}
+
+WebRtc_UWord8 UdpSocketManager::WorkThreads() const
+{
+ return _numOfWorkThreads;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h
new file mode 100644
index 0000000..e7bd09e
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
+
+#include "system_wrappers/interface/static_instance.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class UdpSocketWrapper;
+
+class UdpSocketManager
+{
+public:
+ static UdpSocketManager* Create(const WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads);
+ static void Return();
+
+ // Initializes the socket manager. Returns true if the manager wasn't
+ // already initialized.
+ virtual bool Init(WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads) = 0;
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ // Start listening to sockets that have been registered via the
+ // AddSocket(..) API.
+ virtual bool Start() = 0;
+ // Stop listening to sockets.
+ virtual bool Stop() = 0;
+
+ virtual WebRtc_UWord8 WorkThreads() const;
+
+ // Register a socket with the socket manager.
+ virtual bool AddSocket(UdpSocketWrapper* s) = 0;
+ // Unregister a socket from the manager.
+ virtual bool RemoveSocket(UdpSocketWrapper* s) = 0;
+
+protected:
+ UdpSocketManager();
+ virtual ~UdpSocketManager() {}
+
+ WebRtc_UWord8 _numOfWorkThreads;
+
+ // Factory method.
+ static UdpSocketManager* CreateInstance();
+
+private:
+ // Friend function to allow the UDP destructor to be accessed from the
+ // instance template.
+ friend UdpSocketManager*
+ GetStaticInstance<UdpSocketManager>(CountOperation count_operation);
+
+ static UdpSocketManager* StaticInstance(
+ CountOperation count_operation,
+ const WebRtc_Word32 id,
+ WebRtc_UWord8& numOfWorkThreads);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_posix.cc b/trunk/src/modules/udp_transport/source/udp_socket_posix.cc
new file mode 100644
index 0000000..9d3564d
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_posix.cc
@@ -0,0 +1,275 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_posix.h"
+
+#include <errno.h>
+#include <fcntl.h>
+#include <netdb.h>
+#include <string.h>
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <time.h>
+#include <unistd.h>
+
+#include "trace.h"
+#include "udp_socket_manager_wrapper.h"
+#include "udp_socket_wrapper.h"
+
+namespace webrtc {
+UdpSocketPosix::UdpSocketPosix(const WebRtc_Word32 id, UdpSocketManager* mgr,
+ bool ipV6Enable)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
+ "UdpSocketPosix::UdpSocketPosix()");
+
+ _wantsIncoming = false;
+ _error = 0;
+ _mgr = mgr;
+
+ _id = id;
+ _obj = NULL;
+ _incomingCb = NULL;
+ _readyForDeletionCond = ConditionVariableWrapper::CreateConditionVariable();
+ _closeBlockingCompletedCond =
+ ConditionVariableWrapper::CreateConditionVariable();
+ _cs = CriticalSectionWrapper::CreateCriticalSection();
+ _readyForDeletion = false;
+ _closeBlockingActive = false;
+ _closeBlockingCompleted= false;
+ if(ipV6Enable)
+ {
+ _socket = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
+ }
+ else {
+ _socket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ }
+
+ // Set socket to nonblocking mode.
+ int enable_non_blocking = 1;
+ if(ioctl(_socket, FIONBIO, &enable_non_blocking) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, id,
+ "Failed to make socket nonblocking");
+ }
+ // Enable close on fork for file descriptor so that it will not block until
+ // forked process terminates.
+ if(fcntl(_socket, F_SETFD, FD_CLOEXEC) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, id,
+ "Failed to set FD_CLOEXEC for socket");
+ }
+}
+
+UdpSocketPosix::~UdpSocketPosix()
+{
+ if(_socket != INVALID_SOCKET)
+ {
+ close(_socket);
+ _socket = INVALID_SOCKET;
+ }
+ if(_readyForDeletionCond)
+ {
+ delete _readyForDeletionCond;
+ }
+
+ if(_closeBlockingCompletedCond)
+ {
+ delete _closeBlockingCompletedCond;
+ }
+
+ if(_cs)
+ {
+ delete _cs;
+ }
+}
+
+WebRtc_Word32 UdpSocketPosix::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+bool UdpSocketPosix::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
+{
+ _obj = obj;
+ _incomingCb = cb;
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketPosix(%p)::SetCallback", this);
+
+ if (_mgr->AddSocket(this))
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketPosix(%p)::SetCallback socket added to manager",
+ this);
+ return true; // socket is now ready for action
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketPosix(%p)::SetCallback error adding me to mgr",
+ this);
+ return false;
+}
+
+bool UdpSocketPosix::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval, WebRtc_Word32 optlen)
+{
+ if(0 == setsockopt(_socket, level, optname, optval, optlen ))
+ {
+ return true;
+ }
+
+ _error = errno;
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocketPosix::SetSockopt(), error:%d", _error);
+ return false;
+}
+
+WebRtc_Word32 UdpSocketPosix::SetTOS(WebRtc_Word32 serviceType)
+{
+ if (SetSockopt(IPPROTO_IP, IP_TOS ,(WebRtc_Word8*)&serviceType ,4) != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+bool UdpSocketPosix::Bind(const SocketAddress& name)
+{
+ int size = sizeof(sockaddr);
+ if (0 == bind(_socket, reinterpret_cast<const sockaddr*>(&name),size))
+ {
+ return true;
+ }
+ _error = errno;
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocketPosix::Bind() error: %d",_error);
+ return false;
+}
+
+WebRtc_Word32 UdpSocketPosix::SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
+ const SocketAddress& to)
+{
+ int size = sizeof(sockaddr);
+ int retVal = sendto(_socket,buf, len, 0,
+ reinterpret_cast<const sockaddr*>(&to), size);
+ if(retVal == SOCKET_ERROR)
+ {
+ _error = errno;
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocketPosix::SendTo() error: %d", _error);
+ }
+
+ return retVal;
+}
+
+bool UdpSocketPosix::ValidHandle()
+{
+ return _socket != INVALID_SOCKET;
+}
+
+void UdpSocketPosix::HasIncoming()
+{
+ char buf[2048];
+ int retval;
+ SocketAddress from;
+#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
+ sockaddr sockaddrfrom;
+ memset(&from, 0, sizeof(from));
+ memset(&sockaddrfrom, 0, sizeof(sockaddrfrom));
+ socklen_t fromlen = sizeof(sockaddrfrom);
+#else
+ memset(&from, 0, sizeof(from));
+ socklen_t fromlen = sizeof(from);
+#endif
+
+#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
+ retval = recvfrom(_socket,buf, sizeof(buf), 0,
+ reinterpret_cast<sockaddr*>(&sockaddrfrom), &fromlen);
+ memcpy(&from, &sockaddrfrom, fromlen);
+ from._sockaddr_storage.sin_family = sockaddrfrom.sa_family;
+#else
+ retval = recvfrom(_socket,buf, sizeof(buf), 0,
+ reinterpret_cast<sockaddr*>(&from), &fromlen);
+#endif
+
+ switch(retval)
+ {
+ case 0:
+ // The peer has performed an orderly shutdown.
+ break;
+ case SOCKET_ERROR:
+ break;
+ default:
+ if(_wantsIncoming && _incomingCb)
+ {
+ _incomingCb(_obj,buf, retval, &from);
+ }
+ break;
+ }
+}
+
+void UdpSocketPosix::CloseBlocking()
+{
+ _cs->Enter();
+ _closeBlockingActive = true;
+ if(!CleanUp())
+ {
+ _closeBlockingActive = false;
+ _cs->Leave();
+ return;
+ }
+
+ while(!_readyForDeletion)
+ {
+ _readyForDeletionCond->SleepCS(*_cs);
+ }
+ _closeBlockingCompleted = true;
+ _closeBlockingCompletedCond->Wake();
+ _cs->Leave();
+}
+
+void UdpSocketPosix::ReadyForDeletion()
+{
+ _cs->Enter();
+ if(!_closeBlockingActive)
+ {
+ _cs->Leave();
+ return;
+ }
+ close(_socket);
+ _socket = INVALID_SOCKET;
+ _readyForDeletion = true;
+ _readyForDeletionCond->Wake();
+ while(!_closeBlockingCompleted)
+ {
+ _closeBlockingCompletedCond->SleepCS(*_cs);
+ }
+ _cs->Leave();
+}
+
+bool UdpSocketPosix::CleanUp()
+{
+ _wantsIncoming = false;
+
+ if (_socket == INVALID_SOCKET)
+ {
+ return false;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "calling UdpSocketManager::RemoveSocket()...");
+ _mgr->RemoveSocket(this);
+ // After this, the socket should may be or will be as deleted. Return
+ // immediately.
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_posix.h b/trunk/src/modules/udp_transport/source/udp_socket_posix.h
new file mode 100644
index 0000000..ee76abb
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_posix.h
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
+
+#include <arpa/inet.h>
+#include <netinet/in.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+
+#include "condition_variable_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "udp_socket_wrapper.h"
+
+#define SOCKET_ERROR -1
+
+namespace webrtc {
+class UdpSocketPosix : public UdpSocketWrapper
+{
+public:
+ UdpSocketPosix(const WebRtc_Word32 id, UdpSocketManager* mgr,
+ bool ipV6Enable = false);
+
+ virtual ~UdpSocketPosix();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb);
+
+ virtual bool Bind(const SocketAddress& name);
+
+ virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval, WebRtc_Word32 optlen);
+
+ virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
+
+ virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
+ const SocketAddress& to);
+
+ // Deletes socket in addition to closing it.
+ // TODO (hellner): make destructor protected.
+ virtual void CloseBlocking();
+
+ virtual SOCKET GetFd() {return _socket;}
+ virtual WebRtc_Word32 GetError() {return _error;}
+
+ virtual bool ValidHandle();
+
+ virtual bool SetQos(WebRtc_Word32 /*serviceType*/,
+ WebRtc_Word32 /*tokenRate*/,
+ WebRtc_Word32 /*bucketSize*/,
+ WebRtc_Word32 /*peekBandwith*/,
+ WebRtc_Word32 /*minPolicedSize*/,
+ WebRtc_Word32 /*maxSduSize*/,
+ const SocketAddress& /*stRemName*/,
+ WebRtc_Word32 /*overrideDSCP*/) {return false;}
+
+ bool CleanUp();
+ void HasIncoming();
+ bool WantsIncoming() {return _wantsIncoming;}
+ void ReadyForDeletion();
+private:
+ friend class UdpSocketManagerPosix;
+
+ WebRtc_Word32 _id;
+ IncomingSocketCallback _incomingCb;
+ CallbackObj _obj;
+ WebRtc_Word32 _error;
+
+ SOCKET _socket;
+ UdpSocketManager* _mgr;
+ ConditionVariableWrapper* _closeBlockingCompletedCond;
+ ConditionVariableWrapper* _readyForDeletionCond;
+
+ bool _closeBlockingActive;
+ bool _closeBlockingCompleted;
+ bool _readyForDeletion;
+
+ CriticalSectionWrapper* _cs;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket_windows.cc
new file mode 100644
index 0000000..05381e1
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_windows.cc
@@ -0,0 +1,772 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_windows.h"
+
+// Disable deprication warning from traffic.h
+#pragma warning(disable : 4995)
+
+#ifndef WIN32_LEAN_AND_MEAN
+#define WIN32_LEAN_AND_MEAN
+#endif
+
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <windows.h>
+#include <Qos.h>
+// Don't change include order for these header files.
+#include <Winsock2.h>
+#include <Ntddndis.h>
+#include <traffic.h>
+
+#include "traffic_control_windows.h"
+#include "udp_socket_manager_wrapper.h"
+
+namespace webrtc {
+typedef struct _QOS_DESTADDR
+{
+ QOS_OBJECT_HDR ObjectHdr;
+ const struct sockaddr* SocketAddress;
+ ULONG SocketAddressLength;
+} QOS_DESTADDR, *LPQOS_DESTADDR;
+
+typedef const QOS_DESTADDR* LPCQOS_DESTADDR;
+
+#define QOS_GENERAL_ID_BASE 2000
+#define QOS_OBJECT_DESTADDR (0x00000004 + QOS_GENERAL_ID_BASE)
+
+#define MAX_PACKET_SIZE 2048
+
+class UDPPacket
+{
+public:
+ UDPPacket()
+ {
+ _length = 0;
+ }
+ WebRtc_Word32 Set(const WebRtc_Word8* buf, WebRtc_Word32 length)
+ {
+ if(length > MAX_PACKET_SIZE)
+ return 0;
+
+ _length = length;
+ memcpy(_buffer,buf,length);
+ return length;
+ }
+ WebRtc_Word32 Set(const WebRtc_Word8* buf, WebRtc_Word32 length,
+ const SocketAddress* addr)
+ {
+ if(length > MAX_PACKET_SIZE)
+ {
+ return 0;
+ }
+
+ _length = length;
+ memcpy(&_remoteAddr,addr,sizeof(SocketAddress));
+ memcpy(_buffer,buf,length);
+ return length;
+ }
+
+ SocketAddress _remoteAddr;
+ WebRtc_Word8 _buffer[MAX_PACKET_SIZE];
+ WebRtc_Word32 _length;
+};
+
+UdpSocketWindows::UdpSocketWindows(const WebRtc_Word32 id,
+ UdpSocketManager* mgr, bool ipV6Enable)
+ : _id(id),
+ _qos(true)
+{
+ _wantsIncoming = false;
+ _error = 0;
+ _mgr = mgr;
+ _addedToMgr = false;
+
+ _obj = NULL;
+ _incomingCb = NULL;
+ _socket = INVALID_SOCKET;
+ _terminate=false;
+
+ _clientHandle = INVALID_HANDLE_VALUE;
+ _flowHandle = INVALID_HANDLE_VALUE;
+ _filterHandle = INVALID_HANDLE_VALUE;
+
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
+ "UdpSocketWindows::UdpSocketWindows()");
+
+ _gtc = NULL;
+
+ // Check if QoS is supported.
+ WSAPROTOCOL_INFO pProtocolInfo;
+ DWORD dwBufLen = 0;
+ BOOL bProtocolFound = FALSE;
+ WSAPROTOCOL_INFO* lpProtocolBuf = NULL;
+
+ // Set dwBufLen to the size needed to retreive all the requested information
+ // from WSAEnumProtocols.
+ WebRtc_Word32 nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
+ lpProtocolBuf = (WSAPROTOCOL_INFO*)malloc(dwBufLen);
+ nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
+
+ WebRtc_Word32 iProtocol;
+ if (ipV6Enable)
+ {
+ iProtocol = AF_INET6;
+ } else {
+ iProtocol = AF_INET;
+ }
+
+ for (WebRtc_Word32 i = 0; i < nRet; i++)
+ {
+ if (iProtocol == lpProtocolBuf[i].iAddressFamily && IPPROTO_UDP ==
+ lpProtocolBuf[i].iProtocol)
+ {
+ if ((XP1_QOS_SUPPORTED ==
+ (XP1_QOS_SUPPORTED & lpProtocolBuf[i].dwServiceFlags1)))
+ {
+ pProtocolInfo = lpProtocolBuf[i];
+ bProtocolFound = TRUE;
+ break;
+ }
+ }
+ }
+
+ if(!bProtocolFound)
+ {
+ _socket = INVALID_SOCKET;
+ _qos = false;
+ free(lpProtocolBuf);
+ _error = SOCKET_ERROR_NO_QOS;
+ }else {
+ _socket = WSASocket(FROM_PROTOCOL_INFO, FROM_PROTOCOL_INFO,
+ FROM_PROTOCOL_INFO,&pProtocolInfo, 0,
+ WSA_FLAG_OVERLAPPED);
+ free(lpProtocolBuf);
+ if (_socket != INVALID_SOCKET)
+ {
+ return;
+ }else
+ {
+ _qos = false;
+ _error = SOCKET_ERROR_NO_QOS;
+ }
+ }
+ // QoS not supported.
+ if(ipV6Enable)
+ {
+ _socket = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
+ }else
+ {
+ _socket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ }
+ // Non-blocking mode.
+ WebRtc_Word32 iMode = 1;
+ ioctlsocket(_socket, FIONBIO, (u_long FAR*) &iMode);
+}
+
+UdpSocketWindows::~UdpSocketWindows()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
+ "UdpSocketWindows::~UdpSocketWindows()");
+ if (_gtc)
+ {
+ TrafficControlWindows::Release(_gtc);
+ }
+}
+
+WebRtc_Word32 UdpSocketWindows::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ if (_gtc)
+ {
+ _gtc->ChangeUniqueId(id);
+ }
+ return 0;
+}
+
+bool UdpSocketWindows::ValidHandle()
+{
+ return GetFd() != INVALID_SOCKET;
+}
+
+bool UdpSocketWindows::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
+{
+ _obj = obj;
+ _incomingCb = cb;
+
+ if (_mgr->AddSocket(this))
+ {
+ _addedToMgr = true;
+ return true;
+ }
+ return false;
+}
+
+bool UdpSocketWindows::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval,
+ WebRtc_Word32 optlen)
+{
+ if(0 == setsockopt(_socket, level, optname, optval, optlen))
+ {
+ return true;
+ }
+ _error = WSAGetLastError();
+ return false;
+}
+
+bool UdpSocketWindows::Bind(const SocketAddress& name)
+{
+ const struct sockaddr* socketName =
+ reinterpret_cast<const struct sockaddr*>(&name);
+
+ if (0 == bind(_socket, socketName, sizeof(SocketAddress)))
+ {
+ _localAddr = name;
+ return true;
+ }
+ _error = WSAGetLastError();
+ return false;
+}
+
+WebRtc_Word32 UdpSocketWindows::SendTo(const WebRtc_Word8* buf,
+ WebRtc_Word32 len,
+ const SocketAddress& to)
+{
+ // Don't try to send this packet if there are older packets queued up.
+ if(!_notSentPackets.Empty())
+ {
+ UDPPacket* packet = new UDPPacket();
+ packet->Set(buf, len, &to);
+ if(!_notSentPackets.Empty())
+ {
+ _notSentPackets.PushBack(packet);
+ return len;
+ }else {
+ // No old packets queued up. Free to try to send.
+ delete packet;
+ }
+ }
+
+ WebRtc_Word32 retVal;
+ retVal = sendto(_socket, buf, len, 0,
+ reinterpret_cast<const struct sockaddr*>(&to),
+ sizeof(SocketAddress));
+
+ if(retVal == SOCKET_ERROR)
+ {
+ _error = WSAGetLastError();
+ if (_error == WSAEWOULDBLOCK)
+ {
+ UDPPacket* packet = new UDPPacket();
+ packet->Set(buf,len, &to);
+ _notSentPackets.PushBack(packet);
+ return len;
+ }
+ }
+ return retVal;
+}
+
+void UdpSocketWindows::HasIncoming()
+{
+ WebRtc_Word8 buf[MAX_PACKET_SIZE];
+ SocketAddress from;
+ int fromlen = sizeof(from);
+ WebRtc_Word32 retval = recvfrom(_socket,buf, sizeof(buf), 0,
+ reinterpret_cast<struct sockaddr*>(&from),
+ &fromlen);
+
+ switch(retval)
+ {
+ case 0:
+ // The connection has been gracefully closed.
+ break;
+ case SOCKET_ERROR:
+ _error = WSAGetLastError();
+ break;
+ default:
+ if(_wantsIncoming && _incomingCb)
+ _incomingCb(_obj,buf, retval, &from);
+ break;
+ }
+}
+
+void UdpSocketWindows::CleanUp()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "UdpSocketWindows::CleanUp()");
+ _wantsIncoming = false;
+
+ if(_clientHandle != INVALID_HANDLE_VALUE)
+ {
+ assert(_filterHandle != INVALID_HANDLE_VALUE);
+ assert(_flowHandle != INVALID_HANDLE_VALUE);
+
+ if (_gtc)
+ {
+ _gtc->TcDeleteFilter(_filterHandle);
+ _gtc->TcDeleteFlow(_flowHandle);
+ _gtc->TcDeregisterClient(_clientHandle);
+ }
+
+ _clientHandle = INVALID_HANDLE_VALUE;
+ _filterHandle = INVALID_HANDLE_VALUE;
+ _flowHandle = INVALID_HANDLE_VALUE;
+ }
+
+ while(!_notSentPackets.Empty())
+ {
+ UDPPacket* packet = (UDPPacket*)_notSentPackets.First()->GetItem();
+ if(!packet)
+ {
+ break;
+ }
+ delete packet;
+ _notSentPackets.PopFront();
+ }
+
+ if (_socket != INVALID_SOCKET)
+ {
+ if (closesocket(_socket) == SOCKET_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "closesocket() => error = %d", WSAGetLastError());
+ }
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "WinSock::closesocket() done");
+
+ if(_addedToMgr)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "calling UdpSocketManager::RemoveSocket()");
+ _mgr->RemoveSocket(this);
+ }
+ }
+}
+
+void UdpSocketWindows::SetWritable()
+{
+ // Try to send packets that have been queued up.
+ while(!_notSentPackets.Empty())
+ {
+ UDPPacket* packet = (UDPPacket*)_notSentPackets.First()->GetItem();
+ if(!packet)
+ {
+ break;
+ }
+ if(sendto(
+ _socket,packet->_buffer,
+ packet->_length,
+ 0,
+ reinterpret_cast<const struct sockaddr*>(
+ &(packet->_remoteAddr)),
+ sizeof(SocketAddress)) == SOCKET_ERROR)
+ {
+ _error = WSAGetLastError();
+ if (_error == WSAEWOULDBLOCK)
+ {
+ return;
+ }
+ } else {
+ delete packet;
+ _notSentPackets.PopFront();
+ }
+ }
+}
+
+bool UdpSocketWindows::SetQos(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize,
+ const SocketAddress &stRemName,
+ WebRtc_Word32 overrideDSCP)
+{
+ if(_qos == false)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "UdpSocket2Windows::SetQos(), socket not capable of QOS");
+ return false;
+ }
+ QOS Qos;
+ WebRtc_Word32 result;
+ DWORD BytesRet;
+
+ if(overrideDSCP != 0)
+ {
+ FLOWSPEC f;
+ WebRtc_Word32 err = CreateFlowSpec(serviceType, tokenRate, bucketSize,
+ peekBandwith, minPolicedSize,
+ maxSduSize, &f);
+ if(err == -1)
+ {
+ return false;
+ }
+ return SetTOSByte(overrideDSCP, &f, &f) == 0;
+ }
+ memset(&Qos, QOS_NOT_SPECIFIED, sizeof(QOS));
+
+ Qos.SendingFlowspec.ServiceType = serviceType;
+ Qos.SendingFlowspec.TokenRate = tokenRate;
+ Qos.SendingFlowspec.TokenBucketSize = bucketSize;
+ Qos.SendingFlowspec.PeakBandwidth = peekBandwith;
+ Qos.SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ Qos.SendingFlowspec.MinimumPolicedSize = minPolicedSize;
+ Qos.SendingFlowspec.MaxSduSize = maxSduSize;
+
+ // Only ServiceType is needed for receiving.
+ Qos.ReceivingFlowspec.ServiceType = serviceType;
+ Qos.ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ Qos.ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
+
+ Qos.ProviderSpecific.len = 0;
+ Qos.ProviderSpecific.buf = NULL;
+ WebRtc_Word8* p = (WebRtc_Word8*)malloc(sizeof(QOS_DESTADDR) +
+ sizeof(QOS_DS_CLASS));
+
+ QOS_DESTADDR* QosDestaddr = (QOS_DESTADDR*)p;
+ ZeroMemory((WebRtc_Word8 *)QosDestaddr, sizeof(QOS_DESTADDR));
+ QosDestaddr->ObjectHdr.ObjectType = QOS_OBJECT_DESTADDR;
+ QosDestaddr->ObjectHdr.ObjectLength = sizeof(QOS_DESTADDR);
+ QosDestaddr->SocketAddress = (SOCKADDR*)&stRemName;
+ QosDestaddr->SocketAddressLength = sizeof(SocketAddress);
+ Qos.ProviderSpecific.len = QosDestaddr->ObjectHdr.ObjectLength;
+ Qos.ProviderSpecific.buf = (WebRtc_Word8*)p;
+
+ // Socket must be bound for this call to be successfull. If socket is not
+ // bound WSAGetLastError() will return 10022.
+ result = WSAIoctl(GetFd(),SIO_SET_QOS, &Qos,sizeof(QOS),NULL, 0, &BytesRet,
+ NULL,NULL);
+ if (result == SOCKET_ERROR)
+ {
+ _error = WSAGetLastError();
+ free(p);
+ return false;
+ }
+ free(p);
+ return true;
+}
+
+WebRtc_Word32 UdpSocketWindows::SetTOS(WebRtc_Word32 serviceType)
+{
+ WebRtc_Word32 res = SetTOSByte(serviceType, NULL, NULL);
+
+ if (res == -1)
+ {
+ OSVERSIONINFO OsVersion;
+ OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
+ GetVersionEx (&OsVersion);
+
+ if ((OsVersion.dwMajorVersion == 4))
+ {
+ return -1;
+ }
+ }
+ return res;
+}
+
+WebRtc_Word32 UdpSocketWindows::CreateFlowSpec(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize,
+ FLOWSPEC *f)
+{
+ if(!f)
+ {
+ return -1;
+ }
+
+ f->ServiceType = serviceType;
+ f->TokenRate = tokenRate;
+ f->TokenBucketSize = bucketSize;
+ f->PeakBandwidth = peekBandwith;
+ f->DelayVariation = QOS_NOT_SPECIFIED;
+ f->Latency = QOS_NOT_SPECIFIED;
+ f->MinimumPolicedSize = minPolicedSize;
+ f->MaxSduSize = maxSduSize;
+ return 0;
+}
+
+WebRtc_Word32 UdpSocketWindows::SetTOSByte(WebRtc_Word32 serviceType,
+ FLOWSPEC* send, FLOWSPEC* recv)
+{
+ if(_socket == INVALID_SOCKET)
+ {
+ return -1;
+ }
+ if (!_gtc)
+ {
+ _gtc = TrafficControlWindows::GetInstance(_id);
+ }
+ if (!_gtc)
+ {
+ return -1;
+ }
+
+ TCI_CLIENT_FUNC_LIST QoSFunctions;
+ QoSFunctions.ClAddFlowCompleteHandler = NULL;
+ QoSFunctions.ClDeleteFlowCompleteHandler = NULL;
+ QoSFunctions.ClModifyFlowCompleteHandler = NULL;
+ QoSFunctions.ClNotifyHandler = (TCI_NOTIFY_HANDLER)MyClNotifyHandler;
+ // Register the client with Traffic control interface.
+ HANDLE ClientHandle;
+ ULONG result = _gtc->TcRegisterClient(CURRENT_TCI_VERSION, NULL,
+ &QoSFunctions,&ClientHandle);
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "TcRegisterClient returned %d", result);
+ return result;
+ }
+
+ // Find traffic control-enabled network interfaces.
+ ULONG BufferSize = 0;
+ result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize, NULL);
+
+ if(result != NO_ERROR && result != ERROR_INSUFFICIENT_BUFFER)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Error enumerating interfaces, %d", result);
+ _gtc->TcDeregisterClient(ClientHandle);
+ return result;
+ }
+
+ if(result != ERROR_INSUFFICIENT_BUFFER)
+ {
+ // Empty buffer contains all control-enabled network interfaces. I.e.
+ // ToS is not enabled.
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "Error enumerating interfaces: passed in 0 and received\
+ NO_ERROR when expecting INSUFFICIENT_BUFFER, %d");
+ _gtc->TcDeregisterClient(ClientHandle);
+ return -1;
+ }
+
+ PTC_IFC_DESCRIPTOR pInterfaceBuffer =
+ (PTC_IFC_DESCRIPTOR)malloc(BufferSize);
+ if(pInterfaceBuffer == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Out ot memory failure");
+ _gtc->TcDeregisterClient(ClientHandle);
+ return ERROR_NOT_ENOUGH_MEMORY;
+ }
+
+ result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize,
+ pInterfaceBuffer);
+
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "Critical: error enumerating interfaces when passing in correct\
+ buffer size: %d", result);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+
+ PTC_IFC_DESCRIPTOR oneinterface;
+ HANDLE ifcHandle, iFilterHandle, iflowHandle;
+ bool addrFound = false;
+ ULONG filterSourceAddress = ULONG_MAX;
+
+ const struct sockaddr_in* name;
+ name = reinterpret_cast<const struct sockaddr_in*>(&_localAddr);
+
+ // Find the interface corresponding to the local address.
+ for(oneinterface = pInterfaceBuffer;
+ oneinterface != (PTC_IFC_DESCRIPTOR)
+ (((WebRtc_Word8*)pInterfaceBuffer) + BufferSize);
+ oneinterface = (PTC_IFC_DESCRIPTOR)
+ ((WebRtc_Word8*)oneinterface + oneinterface->Length))
+ {
+
+ WebRtc_Word8 interfaceName[500];
+ WideCharToMultiByte(CP_ACP, 0, oneinterface->pInterfaceName, -1,
+ interfaceName, sizeof(interfaceName), 0, 0);
+
+ PNETWORK_ADDRESS_LIST addresses =
+ &(oneinterface->AddressListDesc.AddressList);
+ for(LONG i = 0; i < addresses->AddressCount; i++)
+ {
+ // Only look at TCP/IP addresses.
+ if(addresses->Address[i].AddressType != NDIS_PROTOCOL_ID_TCP_IP)
+ {
+ continue;
+ }
+
+ NETWORK_ADDRESS_IP* pIpAddr =
+ (NETWORK_ADDRESS_IP*)&(addresses->Address[i].Address);
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Examining Interface %s", interfaceName);
+ if(pIpAddr->in_addr == name->sin_addr.S_un.S_addr)
+ {
+ filterSourceAddress = pIpAddr->in_addr;
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Found ip addr: %s", inet_ntoa(name->sin_addr));
+ addrFound = true;
+ }
+ }
+ if(!addrFound)
+ {
+ continue;
+ }
+ else
+ {
+ break;
+ }
+ }
+
+ if(!addrFound)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "IP Address not found");
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return -1;
+ }
+
+
+ result = _gtc->TcOpenInterfaceW(oneinterface->pInterfaceName, ClientHandle,
+ NULL, &ifcHandle);
+
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Error opening interface: %d", result);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+ FLOWSPEC defaultSend, defaultRecv;
+ if(send == NULL)
+ {
+ defaultSend.DelayVariation = QOS_NOT_SPECIFIED;
+ defaultSend.Latency = QOS_NOT_SPECIFIED;
+ defaultSend.MaxSduSize = QOS_NOT_SPECIFIED;
+ defaultSend.MinimumPolicedSize = QOS_NOT_SPECIFIED;
+ defaultSend.PeakBandwidth = QOS_NOT_SPECIFIED;
+ defaultSend.ServiceType = SERVICETYPE_BESTEFFORT;
+ defaultSend.TokenBucketSize = QOS_NOT_SPECIFIED;
+ defaultSend.TokenRate = 10000;
+ }
+ else
+ {
+ defaultSend = *send;
+ }
+ if(recv == NULL)
+ {
+ defaultRecv = defaultSend;
+ defaultRecv.ServiceType = SERVICETYPE_CONTROLLEDLOAD;
+ }
+ else
+ {
+ defaultRecv = *recv;
+ }
+
+
+ PTC_GEN_FLOW flow =
+ (PTC_GEN_FLOW)malloc(sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS));
+
+ flow->ReceivingFlowspec = defaultRecv;
+ flow->SendingFlowspec = defaultSend;
+
+ QOS_DS_CLASS dsClass;
+
+ ZeroMemory((WebRtc_Word8*)&dsClass, sizeof(QOS_DS_CLASS));
+
+ dsClass.DSField = serviceType;
+
+ dsClass.ObjectHdr.ObjectType = QOS_OBJECT_DS_CLASS;
+ dsClass.ObjectHdr.ObjectLength = sizeof(dsClass);
+
+ memcpy(flow->TcObjects, (void*)&dsClass, sizeof(QOS_DS_CLASS));
+ flow->TcObjectsLength = sizeof(dsClass);
+
+ result = _gtc->TcAddFlow(ifcHandle, NULL, 0, flow, &iflowHandle);
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Error adding flow: %d", result);
+ _gtc->TcCloseInterface(ifcHandle);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return -1;
+ }
+
+ free(flow);
+ IP_PATTERN filterPattern, mask;
+
+ ZeroMemory((WebRtc_Word8*)&filterPattern, sizeof(IP_PATTERN));
+ ZeroMemory((WebRtc_Word8*)&mask, sizeof(IP_PATTERN));
+
+ filterPattern.ProtocolId = IPPROTO_UDP;
+ // "name" fields are in network order.
+ filterPattern.S_un.S_un_ports.s_srcport = name->sin_port;
+ filterPattern.SrcAddr = filterSourceAddress;
+
+ // Unsigned max of a type corresponds to a bitmask with all bits set to 1.
+ // I.e. the filter should allow all ProtocolIds, any source port and any
+ // IP address.
+ mask.ProtocolId = UCHAR_MAX;
+ mask.S_un.S_un_ports.s_srcport = USHRT_MAX;
+ mask.SrcAddr = ULONG_MAX;
+
+ TC_GEN_FILTER filter;
+ filter.AddressType = NDIS_PROTOCOL_ID_TCP_IP;
+ filter.Mask = (LPVOID)&mask;
+ filter.Pattern = (LPVOID)&filterPattern;
+ filter.PatternSize = sizeof(IP_PATTERN);
+ if(_filterHandle != INVALID_HANDLE_VALUE)
+ {
+ _gtc->TcDeleteFilter(_filterHandle);
+ }
+
+ result = _gtc->TcAddFilter(iflowHandle, &filter, &iFilterHandle);
+ if(result != NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Error adding filter: %d", result);
+ _gtc->TcDeleteFlow(iflowHandle);
+ _gtc->TcCloseInterface(ifcHandle);
+ _gtc->TcDeregisterClient(ClientHandle);
+ free(pInterfaceBuffer);
+ return result;
+ }
+
+ _flowHandle = iflowHandle;
+ _filterHandle = iFilterHandle;
+ _clientHandle = ClientHandle;
+
+ _gtc->TcCloseInterface(ifcHandle);
+ free(pInterfaceBuffer);
+
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Successfully created flow and filter.");
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_windows.h b/trunk/src/modules/udp_transport/source/udp_socket_windows.h
new file mode 100644
index 0000000..0bbd20d
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_windows.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
+
+#include <Winsock2.h>
+
+#include "list_wrapper.h"
+#include "udp_socket_manager_wrapper.h"
+#include "udp_socket_wrapper.h"
+
+namespace webrtc {
+class TrafficControlWindows;
+
+class UdpSocketWindows : public UdpSocketWrapper
+{
+public:
+ UdpSocketWindows(const WebRtc_Word32 id, UdpSocketManager* mgr,
+ bool ipV6Enable = false);
+ virtual ~UdpSocketWindows();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual bool ValidHandle();
+
+ virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb);
+
+ virtual bool Bind(const SocketAddress& name);
+
+ virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval, WebRtc_Word32 optlen);
+
+ virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
+ const SocketAddress& to);
+
+ virtual SOCKET GetFd() {return _socket;}
+ virtual WebRtc_Word32 GetError() {return _error;}
+
+ virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
+ const SocketAddress& stRemName,
+ WebRtc_Word32 overrideDSCP = 0);
+
+ virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
+
+protected:
+ void CleanUp();
+ void HasIncoming();
+ // Socket is free to process pending packets.
+ void SetWritable();
+ bool IsWritable() {return _notSentPackets.Empty();}
+ bool WantsIncoming() {return _wantsIncoming;}
+
+private:
+ friend class UdpSocketManagerWindows;
+
+ WebRtc_Word32 CreateFlowSpec(WebRtc_Word32 serviceType,
+ WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize,
+ WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize,
+ WebRtc_Word32 maxSduSize, FLOWSPEC* f);
+
+ WebRtc_Word32 SetTOSByte(WebRtc_Word32 serviceType, FLOWSPEC* send,
+ FLOWSPEC* recv);
+
+ WebRtc_Word32 _id;
+ IncomingSocketCallback _incomingCb;
+
+ CallbackObj _obj;
+ bool _qos;
+ WebRtc_Word32 _error;
+
+ volatile bool _addedToMgr;
+
+ SocketAddress _remoteAddr;
+ SocketAddress _localAddr;
+ SOCKET _socket;
+ UdpSocketManager* _mgr;
+
+ ListWrapper _notSentPackets;
+ bool _terminate;
+
+ // QoS handles.
+ HANDLE _clientHandle;
+ HANDLE _flowHandle;
+ HANDLE _filterHandle;
+
+ // TOS implementation.
+ TrafficControlWindows* _gtc;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc
new file mode 100644
index 0000000..53093a1
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_socket_wrapper.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "event_wrapper.h"
+#include "trace.h"
+#include "udp_socket_manager_wrapper.h"
+
+#if defined(_WIN32)
+ #include "udp_socket_windows.h"
+ #include "udp_socket2_windows.h"
+#else
+ #include "udp_socket_posix.h"
+#endif
+
+
+namespace webrtc {
+bool UdpSocketWrapper::_initiated = false;
+
+// Temporary Android hack. The value 1024 is taken from
+// <ndk>/build/platforms/android-1.5/arch-arm/usr/include/linux/posix_types.h
+// TODO (tomasl): can we remove this now?
+#ifndef FD_SETSIZE
+#define FD_SETSIZE 1024
+#endif
+
+UdpSocketWrapper::UdpSocketWrapper()
+ : _wantsIncoming(false),
+ _deleteEvent(NULL)
+{
+}
+
+UdpSocketWrapper::~UdpSocketWrapper()
+{
+ if(_deleteEvent)
+ {
+ _deleteEvent->Set();
+ _deleteEvent = NULL;
+ }
+}
+
+void UdpSocketWrapper::SetEventToNull()
+{
+ if (_deleteEvent)
+ {
+ _deleteEvent = NULL;
+ }
+}
+
+#ifdef USE_WINSOCK2
+UdpSocketWrapper* UdpSocketWrapper::CreateSocket(const WebRtc_Word32 id,
+ UdpSocketManager* mgr,
+ CallbackObj obj,
+ IncomingSocketCallback cb,
+ bool ipV6Enable,
+ bool disableGQOS)
+#else
+UdpSocketWrapper* UdpSocketWrapper::CreateSocket(const WebRtc_Word32 id,
+ UdpSocketManager* mgr,
+ CallbackObj obj,
+ IncomingSocketCallback cb,
+ bool ipV6Enable,
+ bool /*disableGQOS*/)
+#endif
+
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
+ "UdpSocketWrapper::CreateSocket");
+
+ UdpSocketWrapper* s = 0;
+
+#ifdef _WIN32
+ if (!_initiated)
+ {
+ WSADATA wsaData;
+ WORD wVersionRequested = MAKEWORD( 2, 2 );
+ WebRtc_Word32 err = WSAStartup( wVersionRequested, &wsaData);
+ if (err != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ id,
+ "UdpSocketWrapper::CreateSocket failed to initialize sockets\
+ WSAStartup error:%d",
+ err);
+ return NULL;
+ }
+
+ _initiated = true;
+ }
+
+#ifdef USE_WINSOCK2
+ s = new UdpSocket2Windows(id, mgr, ipV6Enable, disableGQOS);
+#else
+ #pragma message("Error: No non-Winsock2 implementation for WinCE")
+ s = new UdpSocketWindows(id, mgr, ipV6Enable);
+#endif
+
+#else
+ if (!_initiated)
+ {
+ _initiated = true;
+ }
+ s = new UdpSocketPosix(id, mgr, ipV6Enable);
+ if (s)
+ {
+ UdpSocketPosix* sl = static_cast<UdpSocketPosix*>(s);
+ if (sl->GetFd() != INVALID_SOCKET && sl->GetFd() < FD_SETSIZE)
+ {
+ // ok
+ } else
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ id,
+ "UdpSocketWrapper::CreateSocket failed to initialize socket");
+ delete s;
+ s = NULL;
+ }
+ }
+#endif
+ if (s)
+ {
+ s->_deleteEvent = NULL;
+ if (!s->SetCallback(obj, cb))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ id,
+ "UdpSocketWrapper::CreateSocket failed to ser callback");
+ return(NULL);
+ }
+ }
+ return s;
+}
+
+bool UdpSocketWrapper::StartReceiving()
+{
+ _wantsIncoming = true;
+ return true;
+}
+
+bool UdpSocketWrapper::StopReceiving()
+{
+ _wantsIncoming = false;
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h
new file mode 100644
index 0000000..9a3b585
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
+
+#include "udp_transport.h"
+
+namespace webrtc {
+class EventWrapper;
+class UdpSocketManager;
+
+#define SOCKET_ERROR_NO_QOS -1000
+
+#ifndef _WIN32
+typedef int SOCKET;
+#endif
+
+#ifndef INVALID_SOCKET
+#define INVALID_SOCKET (SOCKET)(~0)
+
+#ifndef AF_INET
+#define AF_INET 2
+#endif
+
+#endif
+
+typedef void* CallbackObj;
+typedef void(*IncomingSocketCallback)(CallbackObj obj, const WebRtc_Word8* buf,
+ WebRtc_Word32 len,
+ const SocketAddress* from);
+
+class UdpSocketWrapper
+{
+public:
+ virtual ~UdpSocketWrapper();
+ static UdpSocketWrapper* CreateSocket(const WebRtc_Word32 id,
+ UdpSocketManager* mgr,
+ CallbackObj obj,
+ IncomingSocketCallback cb,
+ bool ipV6Enable = false,
+ bool disableGQOS = false);
+
+ // Set the unique identifier of this class to id.
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ // Register cb for receiving callbacks when there are incoming packets.
+ // Register obj so that it will be passed in calls to cb.
+ virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb) = 0;
+
+ // Socket to local address specified by name.
+ virtual bool Bind(const SocketAddress& name) = 0;
+
+ // Start receiving UDP data.
+ virtual bool StartReceiving();
+ virtual inline bool StartReceiving(const WebRtc_UWord32 /*receiveBuffers*/)
+ {return StartReceiving();}
+ // Stop receiving UDP data.
+ virtual bool StopReceiving();
+
+ virtual bool ValidHandle() = 0;
+
+ // Set socket options.
+ virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
+ const WebRtc_Word8* optval,
+ WebRtc_Word32 optlen) = 0;
+
+ // Set TOS for outgoing packets.
+ virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType) = 0;
+
+ // Set 802.1Q PCP field (802.1p) for outgoing VLAN traffic.
+ virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 /*pcp*/) {return -1;}
+
+ // Send buf of length len to the address specified by to.
+ virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
+ const SocketAddress& to) = 0;
+
+ virtual void SetEventToNull();
+
+ // Close socket and don't return until completed.
+ virtual void CloseBlocking() {}
+
+ // tokenRate is in bit/s. peakBandwidt is in byte/s
+ virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
+ WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
+ WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
+ const SocketAddress &stRemName,
+ WebRtc_Word32 overrideDSCP = 0) = 0;
+
+ virtual WebRtc_UWord32 ReceiveBuffers() {return 0;};
+
+protected:
+ UdpSocketWrapper();
+
+ bool _wantsIncoming;
+ EventWrapper* _deleteEvent;
+
+private:
+ static bool _initiated;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_transport.gypi b/trunk/src/modules/udp_transport/source/udp_transport.gypi
new file mode 100644
index 0000000..c30e167
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_transport.gypi
@@ -0,0 +1,120 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'udp_transport',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ ],
+ },
+ 'sources': [
+ # PLATFORM INDEPENDENT SOURCE FILES
+ '../interface/udp_transport.h',
+ 'udp_transport_impl.cc',
+ 'udp_socket_wrapper.cc',
+ 'udp_socket_manager_wrapper.cc',
+ 'udp_transport_impl.h',
+ 'udp_socket_wrapper.h',
+ 'udp_socket_manager_wrapper.h',
+ # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
+ # Posix (Linux/Mac)
+ 'udp_socket_posix.cc',
+ 'udp_socket_posix.h',
+ 'udp_socket_manager_posix.cc',
+ 'udp_socket_manager_posix.h',
+ # Windows
+ 'udp_socket_manager_windows.cc',
+ 'udp_socket_manager_windows.h',
+ 'udp_socket2_manager_windows.cc',
+ 'udp_socket2_manager_windows.h',
+ 'udp_socket_windows.cc',
+ 'udp_socket_windows.h',
+ 'udp_socket2_windows.cc',
+ 'udp_socket2_windows.h',
+ 'traffic_control_windows.cc',
+ 'traffic_control_windows.h',
+ ], # source
+ 'conditions': [
+ # DEFINE PLATFORM SPECIFIC SOURCE FILES
+ ['os_posix==0', {
+ 'sources!': [
+ 'udp_socket_posix.cc',
+ 'udp_socket_posix.h',
+ 'udp_socket_manager_posix.cc',
+ 'udp_socket_manager_posix.h',
+ ],
+ }],
+ ['OS!="win"', {
+ 'sources!': [
+ 'udp_socket_manager_windows.cc',
+ 'udp_socket_manager_windows.h',
+ 'udp_socket2_manager_windows.cc',
+ 'udp_socket2_manager_windows.h',
+ 'udp_socket_windows.cc',
+ 'udp_socket_windows.h',
+ 'udp_socket2_windows.cc',
+ 'udp_socket2_windows.h',
+ 'traffic_control_windows.cc',
+ 'traffic_control_windows.h',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'cflags': [
+ '-fno-strict-aliasing',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ 'OTHER_CPLUSPLUSFLAGS': '-fno-strict-aliasing',
+ },
+ }],
+ ['OS=="win"', {
+ 'defines': [
+ 'USE_WINSOCK2',
+ ],
+ }],
+ ] # conditions
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'udp_transport_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'udp_transport',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'udp_transport_unittest.cc',
+ ],
+ }, # udp_transport_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_impl.cc b/trunk/src/modules/udp_transport/source/udp_transport_impl.cc
new file mode 100644
index 0000000..9ce2d92
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_transport_impl.cc
@@ -0,0 +1,3009 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "udp_transport_impl.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+
+#if defined(_WIN32)
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ // Disable warning for default initialized arrays on VS2005
+ #pragma warning(disable:4351)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ #include <arpa/inet.h>
+ #include <ctype.h>
+ #include <fcntl.h>
+ #include <netdb.h>
+ #include <net/if.h>
+ #include <netinet/in.h>
+ #include <stdlib.h>
+ #include <sys/ioctl.h>
+ #include <sys/socket.h>
+ #include <sys/time.h>
+ #include <unistd.h>
+#ifndef MAC_IPHONE
+ #include <net/if_arp.h>
+#endif
+#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+
+#if defined(WEBRTC_MAC)
+#include <ifaddrs.h>
+#include <machine/types.h>
+#endif
+#if defined(WEBRTC_LINUX)
+#include <linux/netlink.h>
+#include <linux/rtnetlink.h>
+#endif
+
+#include "common_types.h"
+#include "critical_section_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "trace.h"
+#include "typedefs.h"
+#include "udp_socket_manager_wrapper.h"
+
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#define GetLastError() errno
+
+#define IFRSIZE ((int)(size * sizeof (struct ifreq)))
+
+#define NLMSG_OK_NO_WARNING(nlh,len) \
+ ((len) >= (int)sizeof(struct nlmsghdr) && \
+ (int)(nlh)->nlmsg_len >= (int)sizeof(struct nlmsghdr) && \
+ (int)(nlh)->nlmsg_len <= (len))
+
+#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+
+namespace webrtc {
+UdpTransport* UdpTransport::Create(const WebRtc_Word32 id,
+ WebRtc_UWord8& numSocketThreads)
+{
+ return new UdpTransportImpl(id, numSocketThreads);
+}
+
+void UdpTransport::Destroy(UdpTransport* module)
+{
+ if(module)
+ {
+ delete module;
+ }
+}
+
+UdpTransportImpl::UdpTransportImpl(const WebRtc_Word32 id,
+ WebRtc_UWord8& numSocketThreads)
+ : _id(id),
+ _crit(CriticalSectionWrapper::CreateCriticalSection()),
+ _critFilter(CriticalSectionWrapper::CreateCriticalSection()),
+ _critPacketCallback(CriticalSectionWrapper::CreateCriticalSection()),
+ _mgr(UdpSocketManager::Create(id, numSocketThreads)),
+ _lastError(kNoSocketError),
+ _destPort(0),
+ _destPortRTCP(0),
+ _localPort(0),
+ _localPortRTCP(0),
+ _srcPort(0),
+ _srcPortRTCP(0),
+ _fromPort(0),
+ _fromPortRTCP(0),
+ _fromIP(),
+ _destIP(),
+ _localIP(),
+ _localMulticastIP(),
+ _ptrRtpSocket(NULL),
+ _ptrRtcpSocket(NULL),
+ _ptrSendRtpSocket(NULL),
+ _ptrSendRtcpSocket(NULL),
+ _remoteRTPAddr(),
+ _remoteRTCPAddr(),
+ _localRTPAddr(),
+ _localRTCPAddr(),
+ _tos(0),
+ _receiving(false),
+ _useSetSockOpt(false),
+ _qos(false),
+ _pcp(0),
+ _ipV6Enabled(false),
+ _serviceType(0),
+ _overrideDSCP(0),
+ _maxBitrate(0),
+ _cachLock(RWLockWrapper::CreateRWLock()),
+ _previousAddress(),
+ _previousIP(),
+ _previousIPSize(0),
+ _previousSourcePort(0),
+ _filterIPAddress(),
+ _rtpFilterPort(0),
+ _rtcpFilterPort(0),
+ _packetCallback(0)
+{
+ memset(&_remoteRTPAddr, 0, sizeof(_remoteRTPAddr));
+ memset(&_remoteRTCPAddr, 0, sizeof(_remoteRTCPAddr));
+ memset(&_localRTPAddr, 0, sizeof(_localRTPAddr));
+ memset(&_localRTCPAddr, 0, sizeof(_localRTCPAddr));
+
+ memset(_fromIP, 0, sizeof(_fromIP));
+ memset(_destIP, 0, sizeof(_destIP));
+ memset(_localIP, 0, sizeof(_localIP));
+ memset(_localMulticastIP, 0, sizeof(_localMulticastIP));
+
+ memset(&_filterIPAddress, 0, sizeof(_filterIPAddress));
+ if(_mgr == NULL)
+ {
+ _mgr = UdpSocketManager::Create(id, numSocketThreads);
+ }
+
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, id, "%s created", __FUNCTION__);
+}
+
+UdpTransportImpl::~UdpTransportImpl()
+{
+ CloseSendSockets();
+ CloseReceiveSockets();
+ delete _crit;
+ delete _critFilter;
+ delete _critPacketCallback;
+ delete _cachLock;
+
+ UdpSocketManager::Return();
+ WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id, "%s deleted",
+ __FUNCTION__);
+}
+
+WebRtc_Word32 UdpTransportImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+
+ CriticalSectionScoped cs(_crit);
+ _id = id;
+ if(_mgr)
+ {
+ _mgr->ChangeUniqueId(id);
+ }
+ if(_ptrRtpSocket)
+ {
+ _ptrRtpSocket->ChangeUniqueId(id);
+ }
+ if(_ptrRtcpSocket)
+ {
+ _ptrRtcpSocket->ChangeUniqueId(id);
+ }
+ if(_ptrSendRtpSocket)
+ {
+ _ptrSendRtpSocket->ChangeUniqueId(id);
+ }
+ if(_ptrSendRtcpSocket)
+ {
+ _ptrSendRtcpSocket->ChangeUniqueId(id);
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::TimeUntilNextProcess()
+{
+ return 100;
+}
+
+WebRtc_Word32 UdpTransportImpl::Process()
+{
+ return 0;
+}
+
+UdpTransport::ErrorCode UdpTransportImpl::LastError() const
+{
+ return _lastError;
+}
+
+bool SameAddress(const SocketAddress& address1, const SocketAddress& address2)
+{
+ return (memcmp(&address1,&address2,sizeof(address1)) == 0);
+}
+
+void UdpTransportImpl::GetCachedAddress(char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort)
+{
+ const WebRtc_UWord32 originalIPSize = ipSize;
+ // If the incoming string is too small, fill it as much as there is room
+ // for. Make sure that there is room for the '\0' character.
+ ipSize = (ipSize - 1 < _previousIPSize) ? ipSize - 1 : _previousIPSize;
+ memcpy(ip,_previousIP,sizeof(WebRtc_Word8)*(ipSize + 1));
+ ip[originalIPSize - 1] = '\0';
+ sourcePort = _previousSourcePort;
+}
+
+WebRtc_Word32 UdpTransportImpl::IPAddressCached(const SocketAddress& address,
+ char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort)
+{
+ {
+ ReadLockScoped rl(*_cachLock);
+ // Check if the old address can be re-used (is the same).
+ if(SameAddress(address,_previousAddress))
+ {
+ GetCachedAddress(ip,ipSize,sourcePort);
+ return 0;
+ }
+ }
+ // Get the new address and store it.
+ WriteLockScoped wl(*_cachLock);
+ ipSize = kIpAddressVersion6Length;
+ if(IPAddress(address,_previousIP,ipSize,_previousSourcePort) != 0)
+ {
+ return -1;
+ }
+ _previousIPSize = ipSize;
+ memcpy(&_previousAddress, &address, sizeof(address));
+ // Address has been cached at this point.
+ GetCachedAddress(ip,ipSize,sourcePort);
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::InitializeReceiveSockets(
+ UdpTransportData* const packetCallback,
+ const WebRtc_UWord16 portnr,
+ const char* ip,
+ const char* multicastIpAddr,
+ const WebRtc_UWord16 rtcpPort)
+{
+
+ {
+ CriticalSectionScoped cs(_critPacketCallback);
+ _packetCallback = packetCallback;
+
+ if(packetCallback == NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
+ "Closing down receive sockets");
+ return 0;
+ }
+ }
+
+ CriticalSectionScoped cs(_crit);
+ CloseReceiveSockets();
+
+ if(portnr == 0)
+ {
+ // TODO (hellner): why not just fail here?
+ if(_destPort == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets port 0 not allowed");
+ _lastError = kPortInvalid;
+ return -1;
+ }
+ _localPort = _destPort;
+ } else {
+ _localPort = portnr;
+ }
+ if(rtcpPort)
+ {
+ _localPortRTCP = rtcpPort;
+ }else {
+ _localPortRTCP = _localPort + 1;
+ WEBRTC_TRACE(
+ kTraceStateInfo,
+ kTraceTransport,
+ _id,
+ "InitializeReceiveSockets RTCP port not configured using RTP\
+ port+1=%d",
+ _localPortRTCP);
+ }
+
+ if(ip)
+ {
+ if(IsIpAddressValid(ip,IpV6Enabled()))
+ {
+ strncpy(_localIP, ip,kIpAddressVersion6Length);
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets invalid IP address");
+ _lastError = kIpAddressInvalid;
+ return -1;
+ }
+ }else
+ {
+ // Don't bind to a specific IP address.
+ if(! IpV6Enabled())
+ {
+ strncpy(_localIP, "0.0.0.0",16);
+ } else
+ {
+ strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
+ kIpAddressVersion6Length);
+ }
+ }
+ if(multicastIpAddr && !IpV6Enabled())
+ {
+ if(IsIpAddressValid(multicastIpAddr,IpV6Enabled()))
+ {
+ strncpy(_localMulticastIP, multicastIpAddr,
+ kIpAddressVersion6Length);
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets invalid IP address");
+ _lastError = kIpAddressInvalid;
+ return -1;
+ }
+ }
+ if(_mgr == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets no socket manager");
+ return -1;
+ }
+
+ _useSetSockOpt=false;
+ _tos=0;
+ _pcp=0;
+
+ _ptrRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
+ IncomingRTPCallback,
+ IpV6Enabled());
+
+ _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
+ IncomingRTCPCallback,
+ IpV6Enabled());
+
+ ErrorCode retVal = BindLocalRTPSocket();
+ if(retVal != kNoSocketError)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets faild to bind RTP socket");
+ _lastError = retVal;
+ CloseReceiveSockets();
+ return -1;
+ }
+ retVal = BindLocalRTCPSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeReceiveSockets faild to bind RTCP socket");
+ CloseReceiveSockets();
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::ReceiveSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort,
+ char multicastIpAddr[kIpAddressVersion6Length]) const
+{
+ CriticalSectionScoped cs(_crit);
+ rtpPort = _localPort;
+ rtcpPort = _localPortRTCP;
+ if (ipAddr)
+ {
+ strncpy(ipAddr, _localIP, IpV6Enabled() ?
+ UdpTransport::kIpAddressVersion6Length :
+ UdpTransport::kIpAddressVersion4Length);
+ }
+ if (multicastIpAddr)
+ {
+ strncpy(multicastIpAddr, _localMulticastIP, IpV6Enabled() ?
+ UdpTransport::kIpAddressVersion6Length :
+ UdpTransport::kIpAddressVersion4Length);
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SendSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const
+{
+ CriticalSectionScoped cs(_crit);
+ rtpPort = _destPort;
+ rtcpPort = _destPortRTCP;
+ strncpy(ipAddr, _destIP, IpV6Enabled() ?
+ UdpTransport::kIpAddressVersion6Length :
+ UdpTransport::kIpAddressVersion4Length);
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::RemoteSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const
+{
+ CriticalSectionScoped cs(_crit);
+ rtpPort = _fromPort;
+ rtcpPort = _fromPortRTCP;
+ if(ipAddr)
+ {
+ strncpy(ipAddr, _fromIP, IpV6Enabled() ?
+ kIpAddressVersion6Length :
+ kIpAddressVersion4Length);
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::FilterPorts(
+ WebRtc_UWord16& rtpFilterPort,
+ WebRtc_UWord16& rtcpFilterPort) const
+{
+ CriticalSectionScoped cs(_critFilter);
+ rtpFilterPort = _rtpFilterPort;
+ rtcpFilterPort = _rtcpFilterPort;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetQoS(bool QoS, WebRtc_Word32 serviceType,
+ WebRtc_UWord32 maxBitrate,
+ WebRtc_Word32 overrideDSCP, bool audio)
+{
+ if(QoS)
+ {
+ return EnableQoS(serviceType, audio, maxBitrate, overrideDSCP);
+ }else
+ {
+ return DisableQoS();
+ }
+}
+
+WebRtc_Word32 UdpTransportImpl::EnableQoS(WebRtc_Word32 serviceType,
+ bool audio, WebRtc_UWord32 maxBitrate,
+ WebRtc_Word32 overrideDSCP)
+{
+ if (_ipV6Enabled)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but will be ignored since IPv6 is enabled");
+ _lastError = kQosError;
+ return -1;
+ }
+ if (_tos)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "TOS already enabled, can't use TOS and QoS at the same time");
+ _lastError = kQosError;
+ return -1;
+ }
+ if (_pcp)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "PCP already enabled, can't use PCP and QoS at the same time");
+ _lastError = kQosError;
+ return -1;
+ }
+ if(_destPort == 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but not started since we have not yet configured\
+ the send destination");
+ return -1;
+ }
+ if(_qos)
+ {
+ if(_overrideDSCP == 0 && overrideDSCP != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is already enabled and overrideDSCP differs, not allowed");
+ return -1;
+ }
+ }
+ CriticalSectionScoped cs(_crit);
+
+ UdpSocketWrapper* rtpSock = _ptrSendRtpSocket ?
+ _ptrSendRtpSocket :
+ _ptrRtpSocket;
+ if (!rtpSock || !rtpSock->ValidHandle())
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but not started since we have not yet created the\
+ RTP socket");
+ return -1;
+ }
+ UdpSocketWrapper* rtcpSock = _ptrSendRtcpSocket ?
+ _ptrSendRtcpSocket :
+ _ptrRtcpSocket;
+ if (!rtcpSock || !rtcpSock->ValidHandle())
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but not started since we have not yet created the\
+ RTCP socket");
+ return -1;
+ }
+
+ // Minimum packet size in bytes for which the requested quality of service
+ // will be provided. The smallest RTP header is 12 byte.
+ const WebRtc_Word32 min_policed_size = 12;
+ // Max SDU, maximum packet size permitted or used in the traffic flow, in
+ // bytes.
+ const WebRtc_Word32 max_sdu_size = 1500;
+
+ // Enable QoS for RTP sockets.
+ if(maxBitrate)
+ {
+ // Note: 1 kbit is 125 bytes.
+ // Token Rate is typically set to the average bit rate from peak to
+ // peak.
+ // Bucket size is normally set to the largest average frame size.
+ if(audio)
+ {
+ WEBRTC_TRACE(kTraceStateInfo,
+ kTraceTransport,
+ _id,
+ "Enable QOS for audio with max bitrate:%d",
+ maxBitrate);
+
+ const WebRtc_Word32 token_rate = maxBitrate*125;
+ // The largest audio packets are 60ms frames. This is a fraction
+ // more than 16 packets/second. These 16 frames are sent, at max,
+ // at a bitrate of maxBitrate*125 -> 1 frame is maxBitrate*125/16 ~
+ // maxBitrate * 8.
+ const WebRtc_Word32 bucket_size = maxBitrate * 8;
+ const WebRtc_Word32 peek_bandwith = maxBitrate * 125;
+ if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size,
+ max_sdu_size, _remoteRTPAddr, overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS failed on the RTP socket");
+ _lastError = kQosError;
+ return -1;
+ }
+ }else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
+ "Enable QOS for video with max bitrate:%d",
+ maxBitrate);
+
+ // Allow for a token rate that is twice that of the maximum bitrate
+ // (in bytes).
+ const WebRtc_Word32 token_rate = maxBitrate*250;
+ // largest average frame size (key frame size). Assuming that a
+ // keyframe is 25% of the bitrate during the second its sent
+ // Assume that a key frame is 25% of the bitrate the second that it
+ // is sent. The largest frame size is then maxBitrate* 125 * 0.25 ~
+ // 31.
+ const WebRtc_Word32 bucket_size = maxBitrate*31;
+ const WebRtc_Word32 peek_bandwith = maxBitrate*125;
+ if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size, max_sdu_size,
+ _remoteRTPAddr, overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS failed on the RTP socket");
+ _lastError = kQosError;
+ return -1;
+ }
+ }
+ } else if(audio)
+ {
+ // No max bitrate set. Audio.
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
+ "Enable QOS for audio with default max bitrate");
+
+ // Let max bitrate be 240kbit/s.
+ const WebRtc_Word32 token_rate = 30000;
+ const WebRtc_Word32 bucket_size = 2000;
+ const WebRtc_Word32 peek_bandwith = 30000;
+ if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size, max_sdu_size,
+ _remoteRTPAddr, overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS failed on the RTP socket");
+ _lastError = kQosError;
+ return -1;
+ }
+ }else
+ {
+ // No max bitrate set. Video.
+ WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
+ "Enable QOS for video with default max bitrate");
+
+ // Let max bitrate be 10mbit/s.
+ const WebRtc_Word32 token_rate = 128000*10;
+ const WebRtc_Word32 bucket_size = 32000;
+ const WebRtc_Word32 peek_bandwith = 256000;
+ if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size, max_sdu_size,
+ _remoteRTPAddr, overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "QOS failed on the RTP socket");
+ _lastError = kQosError;
+ return -1;
+ }
+ }
+
+ // Enable QoS for RTCP sockets.
+ // TODO (hellner): shouldn't RTCP be based on 5% of the maximum bandwidth?
+ if(audio)
+ {
+ const WebRtc_Word32 token_rate = 200;
+ const WebRtc_Word32 bucket_size = 200;
+ const WebRtc_Word32 peek_bandwith = 400;
+ if (!rtcpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size, max_sdu_size,
+ _remoteRTCPAddr, overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "QOS failed on the RTCP socket");
+ _lastError = kQosError;
+ }
+ }else
+ {
+ const WebRtc_Word32 token_rate = 5000;
+ const WebRtc_Word32 bucket_size = 100;
+ const WebRtc_Word32 peek_bandwith = 10000;
+ if (!rtcpSock->SetQos(serviceType, token_rate, bucket_size,
+ peek_bandwith, min_policed_size, max_sdu_size,
+ _remoteRTCPAddr, _overrideDSCP))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "QOS failed on the RTCP socket");
+ _lastError = kQosError;
+ }
+ }
+ _qos = true;
+ _serviceType = serviceType;
+ _maxBitrate = maxBitrate;
+ _overrideDSCP = overrideDSCP;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::DisableQoS()
+{
+ if(_qos == false)
+ {
+ return 0;
+ }
+ CriticalSectionScoped cs(_crit);
+
+ UdpSocketWrapper* rtpSock = (_ptrSendRtpSocket ?
+ _ptrSendRtpSocket : _ptrRtpSocket);
+ if (!rtpSock || !rtpSock->ValidHandle())
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but not started since we have not yet created the\
+ RTP socket");
+ return -1;
+ }
+ UdpSocketWrapper* rtcpSock = (_ptrSendRtcpSocket ?
+ _ptrSendRtcpSocket : _ptrRtcpSocket);
+ if (!rtcpSock || !rtcpSock->ValidHandle())
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but not started since we have not yet created the\
+ RTCP socket");
+ return -1;
+ }
+
+ const WebRtc_Word32 service_type = 0; // = SERVICETYPE_NOTRAFFIC
+ const WebRtc_Word32 not_specified = -1;
+ if (!rtpSock->SetQos(service_type, not_specified, not_specified,
+ not_specified, not_specified, not_specified,
+ _remoteRTPAddr, _overrideDSCP))
+ {
+ _lastError = kQosError;
+ return -1;
+ }
+ if (!rtcpSock->SetQos(service_type, not_specified, not_specified,
+ not_specified, not_specified, not_specified,
+ _remoteRTCPAddr,_overrideDSCP))
+ {
+ _lastError = kQosError;
+ }
+ _qos = false;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::QoS(bool& QoS, WebRtc_Word32& serviceType,
+ WebRtc_Word32& overrideDSCP) const
+{
+ CriticalSectionScoped cs(_crit);
+ QoS = _qos;
+ serviceType = _serviceType;
+ overrideDSCP = _overrideDSCP;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetToS(WebRtc_Word32 DSCP, bool useSetSockOpt)
+{
+ if (_qos)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "QoS already enabled");
+ _lastError = kQosError;
+ return -1;
+ }
+ if (DSCP < 0 || DSCP > 63)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Invalid DSCP");
+ _lastError = kTosInvalid;
+ return -1;
+ }
+ if(_tos)
+ {
+ if(useSetSockOpt != _useSetSockOpt)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "Can't switch SetSockOpt method without disabling TOS first");
+ _lastError = kTosInvalid;
+ return -1;
+ }
+ }
+ CriticalSectionScoped cs(_crit);
+ UdpSocketWrapper* rtpSock = NULL;
+ UdpSocketWrapper* rtcpSock = NULL;
+ if(_ptrSendRtpSocket)
+ {
+ rtpSock = _ptrSendRtpSocket;
+ }else
+ {
+ rtpSock = _ptrRtpSocket;
+ }
+ if (rtpSock == NULL)
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(!rtpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(_ptrSendRtcpSocket)
+ {
+ rtcpSock = _ptrSendRtcpSocket;
+ }else
+ {
+ rtcpSock = _ptrRtcpSocket;
+ }
+ if (rtcpSock == NULL)
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(!rtcpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+
+ if (useSetSockOpt)
+ {
+#ifdef _WIN32
+ OSVERSIONINFO OsVersion;
+ OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
+ GetVersionEx(&OsVersion);
+ // Disable QoS before setting ToS on Windows XP. This is done by closing
+ // and re-opening the sockets.
+ // TODO (hellner): why not just fail here and force the user to
+ // re-initialize sockets? Doing this may trick the user
+ // into thinking that the sockets are in a state which
+ // they aren't.
+ if (OsVersion.dwMajorVersion == 5 &&
+ OsVersion.dwMinorVersion == 1)
+ {
+ if(!_useSetSockOpt)
+ {
+ if(_ptrSendRtpSocket)
+ {
+ CloseSendSockets();
+ _ptrSendRtpSocket =
+ UdpSocketWrapper::CreateSocket(_id, _mgr, NULL,
+ NULL, IpV6Enabled(),
+ true);
+ _ptrSendRtcpSocket =
+ UdpSocketWrapper::CreateSocket(_id, _mgr, NULL,
+ NULL, IpV6Enabled(),
+ true);
+ rtpSock=_ptrSendRtpSocket;
+ rtcpSock=_ptrSendRtcpSocket;
+ ErrorCode retVal = BindRTPSendSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ retVal = BindRTCPSendSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ }
+ else
+ {
+ bool receiving=_receiving;
+ WebRtc_UWord32 noOfReceiveBuffers = 0;
+ if(receiving)
+ {
+ noOfReceiveBuffers=_ptrRtpSocket->ReceiveBuffers();
+ if(StopReceiving()!=0)
+ {
+ return -1;
+ }
+ }
+ CloseReceiveSockets();
+ _ptrRtpSocket = UdpSocketWrapper::CreateSocket(
+ _id, _mgr, this, IncomingRTPCallback,
+ IpV6Enabled(), true);
+ _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(
+ _id, _mgr, this, IncomingRTCPCallback,
+ IpV6Enabled(),true);
+ rtpSock=_ptrRtpSocket;
+ rtcpSock=_ptrRtcpSocket;
+ ErrorCode retVal = BindLocalRTPSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ retVal = BindLocalRTCPSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ if(receiving)
+ {
+ if(StartReceiving(noOfReceiveBuffers) !=
+ kNoSocketError)
+ {
+ return -1;
+ }
+ }
+ }
+ }
+ }
+#endif // #ifdef _WIN32
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Setting TOS using SetSockopt");
+ WebRtc_Word32 TOSShifted = DSCP << 2;
+ if (!rtpSock->SetSockopt(IPPROTO_IP, IP_TOS,
+ (WebRtc_Word8*) &TOSShifted, 4))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not SetSockopt tos value on RTP socket");
+ _lastError = kTosInvalid;
+ return -1;
+ }
+ if (!rtcpSock->SetSockopt(IPPROTO_IP, IP_TOS,
+ (WebRtc_Word8*) &TOSShifted, 4))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not sSetSockopt tos value on RTCP socket");
+ _lastError = kTosInvalid;
+ return -1;
+ }
+ } else
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
+ "Setting TOS NOT using SetSockopt");
+ if (rtpSock->SetTOS(DSCP) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not set tos value on RTP socket");
+ _lastError = kTosError;
+ return -1;
+ }
+ if (rtcpSock->SetTOS(DSCP) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not set tos value on RTCP socket");
+ _lastError = kTosError;
+ return -1;
+ }
+ }
+ _useSetSockOpt = useSetSockOpt;
+ _tos = DSCP;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::ToS(WebRtc_Word32& DSCP,
+ bool& useSetSockOpt) const
+{
+ CriticalSectionScoped cs(_crit);
+ DSCP = _tos;
+ useSetSockOpt = _useSetSockOpt;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetPCP(WebRtc_Word32 PCP)
+{
+
+ if (_qos)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "QoS already enabled");
+ _lastError = kQosError;
+ return -1;
+ }
+ if ((PCP < 0) || (PCP > 7))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Invalid PCP");
+ _lastError = kPcpError;
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_crit);
+ UdpSocketWrapper* rtpSock = NULL;
+ UdpSocketWrapper* rtcpSock = NULL;
+ if(_ptrSendRtpSocket)
+ {
+ rtpSock = _ptrSendRtpSocket;
+ }else
+ {
+ rtpSock = _ptrRtpSocket;
+ }
+ if (rtpSock == NULL)
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(!rtpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(_ptrSendRtcpSocket)
+ {
+ rtcpSock = _ptrSendRtcpSocket;
+ }else
+ {
+ rtcpSock = _ptrRtcpSocket;
+ }
+ if (rtcpSock == NULL)
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ if(!rtcpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+
+#if defined(_WIN32)
+ if (rtpSock->SetPCP(PCP) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not set PCP value on RTP socket");
+ _lastError = kPcpError;
+ return -1;
+ }
+ if (rtcpSock->SetPCP(PCP) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not set PCP value on RTCP socket");
+ _lastError = kPcpError;
+ return -1;
+ }
+
+#elif defined(WEBRTC_LINUX)
+ if (!rtpSock->SetSockopt(SOL_SOCKET, SO_PRIORITY, (WebRtc_Word8*) &PCP,
+ sizeof(PCP)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not SetSockopt PCP value on RTP socket");
+ _lastError = kPcpError;
+ return -1;
+ }
+ if (!rtcpSock->SetSockopt(SOL_SOCKET, SO_PRIORITY, (WebRtc_Word8*) &PCP,
+ sizeof(PCP)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Could not SetSockopt PCP value on RTCP socket");
+ _lastError = kPcpError;
+ return -1;
+ }
+#else
+ // Not supported on other platforms (WEBRTC_MAC)
+ _lastError = kPcpError;
+ return -1;
+#endif
+ _pcp = PCP;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::PCP(WebRtc_Word32& PCP) const
+{
+ CriticalSectionScoped cs(_crit);
+ PCP = _pcp;
+ return 0;
+}
+
+bool UdpTransportImpl::SetSockOptUsed()
+{
+ return _useSetSockOpt;
+}
+
+WebRtc_Word32 UdpTransportImpl::EnableIpV6() {
+
+ CriticalSectionScoped cs(_crit);
+ const bool initialized = (_ptrSendRtpSocket || _ptrRtpSocket);
+
+ if (_ipV6Enabled) {
+ return 0;
+ }
+ if (initialized) {
+ _lastError = kIpVersion6Error;
+ return -1;
+ }
+ _ipV6Enabled = true;
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::FilterIP(
+ char filterIPAddress[kIpAddressVersion6Length]) const
+{
+
+ if(filterIPAddress == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "FilterIP: Invalid argument");
+ return -1;
+ }
+ if(_filterIPAddress._sockaddr_storage.sin_family == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "No Filter configured");
+ return -1;
+ }
+ CriticalSectionScoped cs(_critFilter);
+ WebRtc_UWord32 ipSize = kIpAddressVersion6Length;
+ WebRtc_UWord16 sourcePort;
+ return IPAddress(_filterIPAddress, filterIPAddress, ipSize, sourcePort);
+}
+
+WebRtc_Word32 UdpTransportImpl::SetFilterIP(
+ const char filterIPAddress[kIpAddressVersion6Length])
+{
+ if(filterIPAddress == NULL)
+ {
+ memset(&_filterIPAddress, 0, sizeof(_filterIPAddress));
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id, "Filter IP reset");
+ return 0;
+ }
+ CriticalSectionScoped cs(_critFilter);
+ if (_ipV6Enabled)
+ {
+ _filterIPAddress._sockaddr_storage.sin_family = AF_INET6;
+
+ if (InetPresentationToNumeric(
+ AF_INET6,
+ filterIPAddress,
+ &_filterIPAddress._sockaddr_in6.sin6_addr) < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Failed to set\
+ filter IP for IPv6");
+ _lastError = FILTER_ERROR;
+ return -1;
+ }
+ }
+ else
+ {
+ _filterIPAddress._sockaddr_storage.sin_family = AF_INET;
+
+ if(InetPresentationToNumeric(
+ AF_INET,
+ filterIPAddress,
+ &_filterIPAddress._sockaddr_in.sin_addr) < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to set filter IP for IPv4");
+ _lastError = FILTER_ERROR;
+ return -1;
+ }
+ }
+ WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id, "Filter IP set");
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetFilterPorts(WebRtc_UWord16 rtpFilterPort,
+ WebRtc_UWord16 rtcpFilterPort)
+{
+ CriticalSectionScoped cs(_critFilter);
+ _rtpFilterPort = rtpFilterPort;
+ _rtcpFilterPort = rtcpFilterPort;
+ return 0;
+}
+
+bool UdpTransportImpl::SendSocketsInitialized() const
+{
+ CriticalSectionScoped cs(_crit);
+ if(_ptrSendRtpSocket)
+ {
+ return true;
+ }
+ if(_destPort !=0)
+ {
+ return true;
+ }
+ return false;
+}
+
+bool UdpTransportImpl::ReceiveSocketsInitialized() const
+{
+ if(_ptrRtpSocket)
+ {
+ return true;
+ }
+ return false;
+}
+
+bool UdpTransportImpl::SourcePortsInitialized() const
+{
+ if(_ptrSendRtpSocket)
+ {
+ return true;
+ }
+ return false;
+}
+
+bool UdpTransportImpl::IpV6Enabled() const
+{
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
+ return _ipV6Enabled;
+}
+
+void UdpTransportImpl::BuildRemoteRTPAddr()
+{
+ if(_ipV6Enabled)
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _remoteRTPAddr.sin_length = 0;
+ _remoteRTPAddr.sin_family = PF_INET6;
+#else
+ _remoteRTPAddr._sockaddr_storage.sin_family = PF_INET6;
+#endif
+
+ _remoteRTPAddr._sockaddr_in6.sin6_flowinfo=0;
+ _remoteRTPAddr._sockaddr_in6.sin6_scope_id=0;
+ _remoteRTPAddr._sockaddr_in6.sin6_port = Htons(_destPort);
+ InetPresentationToNumeric(AF_INET6,_destIP,
+ &_remoteRTPAddr._sockaddr_in6.sin6_addr);
+ } else
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _remoteRTPAddr.sin_length = 0;
+ _remoteRTPAddr.sin_family = PF_INET;
+#else
+ _remoteRTPAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ _remoteRTPAddr._sockaddr_in.sin_port = Htons(_destPort);
+ _remoteRTPAddr._sockaddr_in.sin_addr = InetAddrIPV4(_destIP);
+ }
+}
+
+void UdpTransportImpl::BuildRemoteRTCPAddr()
+{
+ if(_ipV6Enabled)
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _remoteRTCPAddr.sin_length = 0;
+ _remoteRTCPAddr.sin_family = PF_INET6;
+#else
+ _remoteRTCPAddr._sockaddr_storage.sin_family = PF_INET6;
+#endif
+
+ _remoteRTCPAddr._sockaddr_in6.sin6_flowinfo=0;
+ _remoteRTCPAddr._sockaddr_in6.sin6_scope_id=0;
+ _remoteRTCPAddr._sockaddr_in6.sin6_port = Htons(_destPortRTCP);
+ InetPresentationToNumeric(AF_INET6,_destIP,
+ &_remoteRTCPAddr._sockaddr_in6.sin6_addr);
+
+ } else
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _remoteRTCPAddr.sin_length = 0;
+ _remoteRTCPAddr.sin_family = PF_INET;
+#else
+ _remoteRTCPAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ _remoteRTCPAddr._sockaddr_in.sin_port = Htons(_destPortRTCP);
+ _remoteRTCPAddr._sockaddr_in.sin_addr= InetAddrIPV4(_destIP);
+ }
+}
+
+UdpTransportImpl::ErrorCode UdpTransportImpl::BindRTPSendSocket()
+{
+ if(!_ptrSendRtpSocket)
+ {
+ return kSocketInvalid;
+ }
+ if(!_ptrSendRtpSocket->ValidHandle())
+ {
+ return kIpAddressInvalid;
+ }
+ if(_ipV6Enabled)
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _localRTPAddr.sin_length = 0;
+ _localRTPAddr.sin_family = PF_INET6;
+#else
+ _localRTPAddr._sockaddr_storage.sin_family = PF_INET6;
+#endif
+ _localRTPAddr._sockaddr_in6.sin6_flowinfo=0;
+ _localRTPAddr._sockaddr_in6.sin6_scope_id=0;
+ _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[0] =
+ 0; // = INADDR_ANY
+ _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[1] =
+ 0;
+ _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[2] =
+ 0;
+ _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[3] =
+ 0;
+ _localRTPAddr._sockaddr_in6.sin6_port = Htons(_srcPort);
+ if(_ptrSendRtpSocket->Bind(_localRTPAddr) == false)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _srcPort);
+ return kFailedToBindPort;
+ }
+ } else {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _localRTPAddr.sin_length = 0;
+ _localRTPAddr.sin_family = PF_INET;
+#else
+ _localRTPAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ _localRTPAddr._sockaddr_in.sin_addr = 0;
+ _localRTPAddr._sockaddr_in.sin_port = Htons(_srcPort);
+ if(_ptrSendRtpSocket->Bind(_localRTPAddr) == false)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _srcPort);
+ return kFailedToBindPort;
+ }
+ }
+ return kNoSocketError;
+}
+
+UdpTransportImpl::ErrorCode UdpTransportImpl::BindRTCPSendSocket()
+{
+ if(!_ptrSendRtcpSocket)
+ {
+ return kSocketInvalid;
+ }
+
+ if(_ipV6Enabled)
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _localRTCPAddr.sin_length = 0;
+ _localRTCPAddr.sin_family = PF_INET6;
+#else
+ _localRTCPAddr._sockaddr_storage.sin_family = PF_INET6;
+#endif
+ _localRTCPAddr._sockaddr_in6.sin6_flowinfo=0;
+ _localRTCPAddr._sockaddr_in6.sin6_scope_id=0;
+ _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[0] =
+ 0; // = INADDR_ANY
+ _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[1] =
+ 0;
+ _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[2] =
+ 0;
+ _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[3] =
+ 0;
+ _localRTCPAddr._sockaddr_in6.sin6_port = Htons(_srcPortRTCP);
+ if(_ptrSendRtcpSocket->Bind(_localRTCPAddr) == false)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _srcPortRTCP);
+ return kFailedToBindPort;
+ }
+ } else {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ _localRTCPAddr.sin_length = 0;
+ _localRTCPAddr.sin_family = PF_INET;
+#else
+ _localRTCPAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ _localRTCPAddr._sockaddr_in.sin_addr= 0;
+ _localRTCPAddr._sockaddr_in.sin_port = Htons(_srcPortRTCP);
+ if(_ptrSendRtcpSocket->Bind(_localRTCPAddr) == false)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _srcPortRTCP);
+ return kFailedToBindPort;
+ }
+ }
+ return kNoSocketError;
+}
+
+UdpTransportImpl::ErrorCode UdpTransportImpl::BindLocalRTPSocket()
+{
+ if(!_ptrRtpSocket)
+ {
+ return kSocketInvalid;
+ }
+ if(!IpV6Enabled())
+ {
+ SocketAddress recAddr;
+ memset(&recAddr, 0, sizeof(SocketAddress));
+ recAddr._sockaddr_storage.sin_family = AF_INET;
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ recAddr.sin_length = 0;
+ recAddr.sin_family = PF_INET;
+#else
+ recAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ recAddr._sockaddr_in.sin_addr = InetAddrIPV4(_localIP);
+ recAddr._sockaddr_in.sin_port = Htons(_localPort);
+
+ if (!_ptrRtpSocket->Bind(recAddr))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _localPort);
+ return kFailedToBindPort;
+ }
+ }
+ else
+ {
+ SocketAddress stLclName;
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ stLclName.sin_lenght = 0;
+ stLclName.sin_family = PF_INET6;
+#else
+ stLclName._sockaddr_storage.sin_family = PF_INET6;
+#endif
+ InetPresentationToNumeric(AF_INET6,_localIP,
+ &stLclName._sockaddr_in6.sin6_addr);
+ stLclName._sockaddr_in6.sin6_port = Htons(_localPort);
+ stLclName._sockaddr_in6.sin6_flowinfo = 0;
+ stLclName._sockaddr_in6.sin6_scope_id = 0;
+
+ if (!_ptrRtpSocket->Bind(stLclName))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _localPort);
+ return kFailedToBindPort;
+ }
+ }
+
+ if(_localMulticastIP[0] != 0)
+ {
+ // Join the multicast group from which to receive datagrams.
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr.s_addr = InetAddrIPV4(_localMulticastIP);
+ mreq.imr_interface.s_addr = INADDR_ANY;
+
+ if (!_ptrRtpSocket->SetSockopt(IPPROTO_IP,IP_ADD_MEMBERSHIP,
+ (WebRtc_Word8*)&mreq,sizeof (mreq)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "setsockopt() for multicast failed, not closing socket");
+ }else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceTransport, _id,
+ "multicast group successfully joined");
+ }
+ }
+ return kNoSocketError;
+}
+
+UdpTransportImpl::ErrorCode UdpTransportImpl::BindLocalRTCPSocket()
+{
+ if(!_ptrRtcpSocket)
+ {
+ return kSocketInvalid;
+ }
+ if(! IpV6Enabled())
+ {
+ SocketAddress recAddr;
+ memset(&recAddr, 0, sizeof(SocketAddress));
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ recAddr.sin_length = 0;
+ recAddr.sin_family = AF_INET;
+#else
+ recAddr._sockaddr_storage.sin_family = AF_INET;
+#endif
+ recAddr._sockaddr_in.sin_addr = InetAddrIPV4(_localIP);
+ recAddr._sockaddr_in.sin_port = Htons(_localPortRTCP);
+
+ if (!_ptrRtcpSocket->Bind(recAddr))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _localPortRTCP);
+ return kFailedToBindPort;
+ }
+ }
+ else
+ {
+ SocketAddress stLclName;
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ stLclName.sin_length = 0;
+ stLclName.sin_family = PF_INET6;
+#else
+ stLclName._sockaddr_storage.sin_family = PF_INET6;
+#endif
+ stLclName._sockaddr_in6.sin6_flowinfo = 0;
+ stLclName._sockaddr_in6.sin6_scope_id = 0;
+ stLclName._sockaddr_in6.sin6_port = Htons(_localPortRTCP);
+
+ InetPresentationToNumeric(AF_INET6,_localIP,
+ &stLclName._sockaddr_in6.sin6_addr);
+ if (!_ptrRtcpSocket->Bind(stLclName))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
+ "Failed to bind to port:%d ", _localPortRTCP);
+ return kFailedToBindPort;
+ }
+ }
+ if(_localMulticastIP[0] != 0)
+ {
+ // Join the multicast group from which to receive datagrams.
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr.s_addr = InetAddrIPV4(_localMulticastIP);
+ mreq.imr_interface.s_addr = INADDR_ANY;
+
+ if (!_ptrRtcpSocket->SetSockopt(IPPROTO_IP,IP_ADD_MEMBERSHIP,
+ (WebRtc_Word8*)&mreq,sizeof (mreq)))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "setsockopt() for multicast failed, not closing socket");
+ }else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceTransport, _id,
+ "multicast group successfully joined");
+ }
+ }
+ return kNoSocketError;
+}
+
+WebRtc_Word32 UdpTransportImpl::InitializeSourcePorts(WebRtc_UWord16 rtpPort,
+ WebRtc_UWord16 rtcpPort)
+{
+
+ if(rtpPort == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "InitializeSourcePorts port 0 not allowed");
+ _lastError = kPortInvalid;
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_crit);
+
+ CloseSendSockets();
+
+ if(_mgr == NULL)
+ {
+ return -1;
+ }
+
+ _srcPort = rtpPort;
+ if(rtcpPort == 0)
+ {
+ _srcPortRTCP = rtpPort+1;
+ } else
+ {
+ _srcPortRTCP = rtcpPort;
+ }
+ _useSetSockOpt =false;
+ _tos=0;
+ _pcp=0;
+
+ _ptrSendRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, NULL, NULL,
+ IpV6Enabled());
+ _ptrSendRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, NULL, NULL,
+ IpV6Enabled());
+
+ ErrorCode retVal = BindRTPSendSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ retVal = BindRTCPSendSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SourcePorts(WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const
+{
+ CriticalSectionScoped cs(_crit);
+
+ rtpPort = (_srcPort != 0) ? _srcPort : _localPort;
+ rtcpPort = (_srcPortRTCP != 0) ? _srcPortRTCP : _localPortRTCP;
+ return 0;
+}
+
+
+#ifdef _WIN32
+WebRtc_Word32 UdpTransportImpl::StartReceiving(
+ WebRtc_UWord32 numberOfSocketBuffers)
+#else
+WebRtc_Word32 UdpTransportImpl::StartReceiving(
+ WebRtc_UWord32 /*numberOfSocketBuffers*/)
+#endif
+{
+ CriticalSectionScoped cs(_crit);
+ if(_receiving)
+ {
+ return 0;
+ }
+ if(_ptrRtpSocket)
+ {
+#ifdef _WIN32
+ if(!_ptrRtpSocket->StartReceiving(numberOfSocketBuffers))
+#else
+ if(!_ptrRtpSocket->StartReceiving())
+#endif
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to start receive on RTP socket");
+ _lastError = kStartReceiveError;
+ return -1;
+ }
+ }
+ if(_ptrRtcpSocket)
+ {
+ if(!_ptrRtcpSocket->StartReceiving())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to start receive on RTCP socket");
+ _lastError = kStartReceiveError;
+ return -1;
+ }
+ }
+ if( _ptrRtpSocket == NULL &&
+ _ptrRtcpSocket == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to StartReceiving, no socket initialized");
+ _lastError = kStartReceiveError;
+ return -1;
+ }
+ _receiving = true;
+ return 0;
+}
+
+bool UdpTransportImpl::Receiving() const
+{
+ return _receiving;
+}
+
+WebRtc_Word32 UdpTransportImpl::StopReceiving()
+{
+
+ CriticalSectionScoped cs(_crit);
+
+ _receiving = false;
+
+ if (_ptrRtpSocket)
+ {
+ if (!_ptrRtpSocket->StopReceiving())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to stop receiving on RTP socket");
+ _lastError = kStopReceiveError;
+ return -1;
+ }
+ }
+ if (_ptrRtcpSocket)
+ {
+ if (!_ptrRtcpSocket->StopReceiving())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "Failed to stop receiving on RTCP socket");
+ _lastError = kStopReceiveError;
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::InitializeSendSockets(
+ const char* ipaddr,
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort)
+{
+ {
+ CriticalSectionScoped cs(_crit);
+ _destPort = rtpPort;
+ if(rtcpPort == 0)
+ {
+ _destPortRTCP = _destPort+1;
+ } else
+ {
+ _destPortRTCP = rtcpPort;
+ }
+
+ if(ipaddr == NULL)
+ {
+ if (!IsIpAddressValid(_destIP, IpV6Enabled()))
+ {
+ _destPort = 0;
+ _destPortRTCP = 0;
+ _lastError = kIpAddressInvalid;
+ return -1;
+ }
+ } else
+ {
+ if (IsIpAddressValid(ipaddr, IpV6Enabled()))
+ {
+ strncpy(
+ _destIP,
+ ipaddr,
+ IpV6Enabled() ? kIpAddressVersion6Length :
+ kIpAddressVersion4Length);
+ } else {
+ _destPort = 0;
+ _destPortRTCP = 0;
+ _lastError = kIpAddressInvalid;
+ return -1;
+ }
+ }
+ BuildRemoteRTPAddr();
+ BuildRemoteRTCPAddr();
+ }
+
+ if (_ipV6Enabled)
+ {
+ if (_qos)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceTransport,
+ _id,
+ "QOS is enabled but will be ignored since IPv6 is enabled");
+ }
+ }else
+ {
+ // TODO (grunell): Multicast support is experimantal.
+
+ // Put the first digit of the remote address in val.
+ WebRtc_Word32 val = ntohl(_remoteRTPAddr._sockaddr_in.sin_addr)>> 24;
+
+ if((val > 223) && (val < 240))
+ {
+ // Multicast address.
+ CriticalSectionScoped cs(_crit);
+
+ UdpSocketWrapper* rtpSock = (_ptrSendRtpSocket ?
+ _ptrSendRtpSocket : _ptrRtpSocket);
+ if (!rtpSock || !rtpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+ UdpSocketWrapper* rtcpSock = (_ptrSendRtcpSocket ?
+ _ptrSendRtcpSocket : _ptrRtcpSocket);
+ if (!rtcpSock || !rtcpSock->ValidHandle())
+ {
+ _lastError = kSocketInvalid;
+ return -1;
+ }
+
+ // Set Time To Live to same region
+ WebRtc_Word32 iOptVal = 64;
+ if (!rtpSock->SetSockopt(IPPROTO_IP, IP_MULTICAST_TTL,
+ (WebRtc_Word8*)&iOptVal,
+ sizeof (WebRtc_Word32)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "setsockopt for multicast error on RTP socket");
+ _ptrRtpSocket->CloseBlocking();
+ _ptrRtpSocket = NULL;
+ _lastError = kMulticastAddressInvalid;
+ return -1;
+ }
+ if (!rtcpSock->SetSockopt(IPPROTO_IP, IP_MULTICAST_TTL,
+ (WebRtc_Word8*)&iOptVal,
+ sizeof (WebRtc_Word32)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "setsockopt for multicast error on RTCP socket");
+ _ptrRtpSocket->CloseBlocking();
+ _ptrRtpSocket = NULL;
+ _lastError = kMulticastAddressInvalid;
+ return -1;
+ }
+ }
+ }
+ return 0;
+}
+
+void UdpTransportImpl::BuildSockaddrIn(WebRtc_UWord16 portnr,
+ const char* ip,
+ SocketAddress& remoteAddr) const
+{
+ if(_ipV6Enabled)
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ remoteAddr.sin_length = 0;
+ remoteAddr.sin_family = PF_INET6;
+#else
+ remoteAddr._sockaddr_storage.sin_family = PF_INET6;
+#endif
+ remoteAddr._sockaddr_in6.sin6_port = Htons(portnr);
+ InetPresentationToNumeric(AF_INET6, ip,
+ &remoteAddr._sockaddr_in6.sin6_addr);
+ remoteAddr._sockaddr_in6.sin6_flowinfo=0;
+ remoteAddr._sockaddr_in6.sin6_scope_id=0;
+ } else
+ {
+#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
+ remoteAddr.sin_length = 0;
+ remoteAddr.sin_family = PF_INET;
+#else
+ remoteAddr._sockaddr_storage.sin_family = PF_INET;
+#endif
+ remoteAddr._sockaddr_in.sin_port = Htons(portnr);
+ remoteAddr._sockaddr_in.sin_addr= InetAddrIPV4(
+ const_cast<char*>(ip));
+ }
+}
+
+WebRtc_Word32 UdpTransportImpl::SendRaw(const WebRtc_Word8 *data,
+ WebRtc_UWord32 length,
+ WebRtc_Word32 isRTCP,
+ WebRtc_UWord16 portnr,
+ const char* ip)
+{
+ CriticalSectionScoped cs(_crit);
+ if(isRTCP)
+ {
+ UdpSocketWrapper* rtcpSock = NULL;
+ if(_ptrSendRtcpSocket)
+ {
+ rtcpSock = _ptrSendRtcpSocket;
+ } else if(_ptrRtcpSocket)
+ {
+ rtcpSock = _ptrRtcpSocket;
+ } else
+ {
+ return -1;
+ }
+ if(portnr == 0 && ip == NULL)
+ {
+ return rtcpSock->SendTo(data,length,_remoteRTCPAddr);
+
+ } else if(portnr != 0 && ip != NULL)
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(portnr, ip, remoteAddr);
+ return rtcpSock->SendTo(data,length,remoteAddr);
+ } else if(ip != NULL)
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(_destPortRTCP, ip, remoteAddr);
+ return rtcpSock->SendTo(data,length,remoteAddr);
+ } else
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(portnr, _destIP, remoteAddr);
+ return rtcpSock->SendTo(data,length,remoteAddr);
+ }
+ } else {
+ UdpSocketWrapper* rtpSock = NULL;
+ if(_ptrSendRtpSocket)
+ {
+ rtpSock = _ptrSendRtpSocket;
+
+ } else if(_ptrRtpSocket)
+ {
+ rtpSock = _ptrRtpSocket;
+ } else
+ {
+ return -1;
+ }
+ if(portnr == 0 && ip == NULL)
+ {
+ return rtpSock->SendTo(data,length,_remoteRTPAddr);
+
+ } else if(portnr != 0 && ip != NULL)
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(portnr, ip, remoteAddr);
+ return rtpSock->SendTo(data,length,remoteAddr);
+ } else if(ip != NULL)
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(_destPort, ip, remoteAddr);
+ return rtpSock->SendTo(data,length,remoteAddr);
+ } else
+ {
+ SocketAddress remoteAddr;
+ BuildSockaddrIn(portnr, _destIP, remoteAddr);
+ return rtpSock->SendTo(data,length,remoteAddr);
+ }
+ }
+}
+
+WebRtc_Word32 UdpTransportImpl::SendRTPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to)
+{
+ CriticalSectionScoped cs(_crit);
+ if(_ptrSendRtpSocket)
+ {
+ return _ptrSendRtpSocket->SendTo(data,length,to);
+
+ } else if(_ptrRtpSocket)
+ {
+ return _ptrRtpSocket->SendTo(data,length,to);
+ }
+ return -1;
+}
+
+WebRtc_Word32 UdpTransportImpl::SendRTCPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to)
+{
+
+ CriticalSectionScoped cs(_crit);
+
+ if(_ptrSendRtcpSocket)
+ {
+ return _ptrSendRtcpSocket->SendTo(data,length,to);
+
+ } else if(_ptrRtcpSocket)
+ {
+ return _ptrRtcpSocket->SendTo(data,length,to);
+ }
+ return -1;
+}
+
+WebRtc_Word32 UdpTransportImpl::SendRTPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const WebRtc_UWord16 rtpPort)
+{
+
+ CriticalSectionScoped cs(_crit);
+ // Use the current SocketAdress but update it with rtpPort.
+ SocketAddress to;
+ memcpy(&to, &_remoteRTPAddr, sizeof(SocketAddress));
+
+ if(_ipV6Enabled)
+ {
+ to._sockaddr_in6.sin6_port = Htons(rtpPort);
+ } else
+ {
+ to._sockaddr_in.sin_port = Htons(rtpPort);
+ }
+
+ if(_ptrSendRtpSocket)
+ {
+ return _ptrSendRtpSocket->SendTo(data,length,to);
+
+ } else if(_ptrRtpSocket)
+ {
+ return _ptrRtpSocket->SendTo(data,length,to);
+ }
+ return -1;
+}
+
+WebRtc_Word32 UdpTransportImpl::SendRTCPPacketTo(const WebRtc_Word8* data,
+ WebRtc_UWord32 length,
+ const WebRtc_UWord16 rtcpPort)
+{
+ CriticalSectionScoped cs(_crit);
+
+ // Use the current SocketAdress but update it with rtcpPort.
+ SocketAddress to;
+ memcpy(&to, &_remoteRTCPAddr, sizeof(SocketAddress));
+
+ if(_ipV6Enabled)
+ {
+ to._sockaddr_in6.sin6_port = Htons(rtcpPort);
+ } else
+ {
+ to._sockaddr_in.sin_port = Htons(rtcpPort);
+ }
+
+ if(_ptrSendRtcpSocket)
+ {
+ return _ptrSendRtcpSocket->SendTo(data,length,to);
+
+ } else if(_ptrRtcpSocket)
+ {
+ return _ptrRtcpSocket->SendTo(data,length,to);
+ }
+ return -1;
+}
+
+int UdpTransportImpl::SendPacket(int /*channel*/, const void* data, int length)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
+
+ CriticalSectionScoped cs(_crit);
+
+ if(_destIP[0] == 0)
+ {
+ return -1;
+ }
+ if(_destPort == 0)
+ {
+ return -1;
+ }
+
+ // Create socket if it hasn't been set up already.
+ // TODO (hellner): why not fail here instead. Sockets not being initialized
+ // indicates that there is a problem somewhere.
+ if( _ptrSendRtpSocket == NULL &&
+ _ptrRtpSocket == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceStateInfo,
+ kTraceTransport,
+ _id,
+ "Creating RTP socket since no receive or source socket is\
+ configured");
+
+ _ptrRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
+ IncomingRTPCallback,
+ IpV6Enabled());
+
+ // Don't bind to a specific IP address.
+ if(! IpV6Enabled())
+ {
+ strncpy(_localIP, "0.0.0.0",16);
+ } else
+ {
+ strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
+ kIpAddressVersion6Length);
+ }
+ _localPort = _destPort;
+
+ ErrorCode retVal = BindLocalRTPSocket();
+ if(retVal != kNoSocketError)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "SendPacket() failed to bind RTP socket");
+ _lastError = retVal;
+ CloseReceiveSockets();
+ return -1;
+ }
+ }
+
+ if(_ptrSendRtpSocket)
+ {
+ return _ptrSendRtpSocket->SendTo((const WebRtc_Word8*)data, length,
+ _remoteRTPAddr);
+
+ } else if(_ptrRtpSocket)
+ {
+ return _ptrRtpSocket->SendTo((const WebRtc_Word8*)data, length,
+ _remoteRTPAddr);
+ }
+ return -1;
+}
+
+int UdpTransportImpl::SendRTCPPacket(int /*channel*/, const void* data,
+ int length)
+{
+
+ CriticalSectionScoped cs(_crit);
+ if(_destIP[0] == 0)
+ {
+ return -1;
+ }
+ if(_destPortRTCP == 0)
+ {
+ return -1;
+ }
+
+ // Create socket if it hasn't been set up already.
+ // TODO (hellner): why not fail here instead. Sockets not being initialized
+ // indicates that there is a problem somewhere.
+ if( _ptrSendRtcpSocket == NULL &&
+ _ptrRtcpSocket == NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceStateInfo,
+ kTraceTransport,
+ _id,
+ "Creating RTCP socket since no receive or source socket is\
+ configured");
+
+ _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
+ IncomingRTCPCallback,
+ IpV6Enabled());
+
+ // Don't bind to a specific IP address.
+ if(! IpV6Enabled())
+ {
+ strncpy(_localIP, "0.0.0.0",16);
+ } else
+ {
+ strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
+ kIpAddressVersion6Length);
+ }
+ _localPortRTCP = _destPortRTCP;
+
+ ErrorCode retVal = BindLocalRTCPSocket();
+ if(retVal != kNoSocketError)
+ {
+ _lastError = retVal;
+ WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
+ "SendRTCPPacket() failed to bind RTCP socket");
+ CloseReceiveSockets();
+ return -1;
+ }
+ }
+
+ if(_ptrSendRtcpSocket)
+ {
+ return _ptrSendRtcpSocket->SendTo((const WebRtc_Word8*)data, length,
+ _remoteRTCPAddr);
+ } else if(_ptrRtcpSocket)
+ {
+ return _ptrRtcpSocket->SendTo((const WebRtc_Word8*)data, length,
+ _remoteRTCPAddr);
+ }
+ return -1;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetSendIP(const char* ipaddr)
+{
+ if(!IsIpAddressValid(ipaddr,IpV6Enabled()))
+ {
+ return kIpAddressInvalid;
+ }
+ CriticalSectionScoped cs(_crit);
+ strncpy(_destIP, ipaddr,kIpAddressVersion6Length);
+ BuildRemoteRTPAddr();
+ BuildRemoteRTCPAddr();
+ return 0;
+}
+
+WebRtc_Word32 UdpTransportImpl::SetSendPorts(WebRtc_UWord16 rtpPort,
+ WebRtc_UWord16 rtcpPort)
+{
+ CriticalSectionScoped cs(_crit);
+ _destPort = rtpPort;
+ if(rtcpPort == 0)
+ {
+ _destPortRTCP = _destPort+1;
+ } else
+ {
+ _destPortRTCP = rtcpPort;
+ }
+ BuildRemoteRTPAddr();
+ BuildRemoteRTCPAddr();
+ return 0;
+}
+
+void UdpTransportImpl::IncomingRTPCallback(CallbackObj obj,
+ const WebRtc_Word8* rtpPacket,
+ WebRtc_Word32 rtpPacketLength,
+ const SocketAddress* from)
+{
+ if (rtpPacket && rtpPacketLength > 0)
+ {
+ UdpTransportImpl* socketTransport = (UdpTransportImpl*) obj;
+ socketTransport->IncomingRTPFunction(rtpPacket, rtpPacketLength, from);
+ }
+}
+
+void UdpTransportImpl::IncomingRTCPCallback(CallbackObj obj,
+ const WebRtc_Word8* rtcpPacket,
+ WebRtc_Word32 rtcpPacketLength,
+ const SocketAddress* from)
+{
+ if (rtcpPacket && rtcpPacketLength > 0)
+ {
+ UdpTransportImpl* socketTransport = (UdpTransportImpl*) obj;
+ socketTransport->IncomingRTCPFunction(rtcpPacket, rtcpPacketLength,
+ from);
+ }
+}
+
+void UdpTransportImpl::IncomingRTPFunction(const WebRtc_Word8* rtpPacket,
+ WebRtc_Word32 rtpPacketLength,
+ const SocketAddress* fromSocket)
+{
+ char ipAddress[kIpAddressVersion6Length];
+ WebRtc_UWord32 ipAddressLength = kIpAddressVersion6Length;
+ WebRtc_UWord16 portNr = 0;
+
+ {
+ CriticalSectionScoped cs(_critFilter);
+ if (FilterIPAddress(fromSocket) == false)
+ {
+ // Packet should be filtered out. Drop it.
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
+ "Incoming RTP packet blocked by IP filter");
+ return;
+ }
+
+ if (IPAddressCached(*fromSocket, ipAddress, ipAddressLength, portNr) <
+ 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpTransportImpl::IncomingRTPFunction - Cannot get sender\
+ information");
+ }else
+ {
+ strncpy(_fromIP, ipAddress, kIpAddressVersion6Length);
+ }
+
+ // Filter based on port.
+ if (_rtpFilterPort != 0 &&
+ _rtpFilterPort != portNr)
+ {
+ // Drop packet.
+ memset(_fromIP, 0, sizeof(_fromIP));
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceTransport,
+ _id,
+ "Incoming RTP packet blocked by filter incoming from port:%d\
+ allowed port:%d",
+ portNr,
+ _rtpFilterPort);
+ return;
+ }
+ _fromPort = portNr;
+ }
+
+ CriticalSectionScoped cs(_critPacketCallback);
+ if (_packetCallback)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
+ "Incoming RTP packet from ip:%s port:%d", ipAddress, portNr);
+ _packetCallback->IncomingRTPPacket(rtpPacket, rtpPacketLength,
+ ipAddress, portNr);
+ }
+}
+
+void UdpTransportImpl::IncomingRTCPFunction(const WebRtc_Word8* rtcpPacket,
+ WebRtc_Word32 rtcpPacketLength,
+ const SocketAddress* fromSocket)
+{
+ char ipAddress[kIpAddressVersion6Length];
+ WebRtc_UWord32 ipAddressLength = kIpAddressVersion6Length;
+ WebRtc_UWord16 portNr = 0;
+
+ {
+ CriticalSectionScoped cs(_critFilter);
+ if (FilterIPAddress(fromSocket) == false)
+ {
+ // Packet should be filtered out. Drop it.
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
+ "Incoming RTCP packet blocked by IP filter");
+ return;
+ }
+ if (IPAddress(*fromSocket, ipAddress, ipAddressLength, portNr) < 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpTransportImpl::IncomingRTCPFunction - Cannot get sender\
+ information");
+ }else {
+ strncpy(_fromIP, ipAddress, kIpAddressVersion6Length);
+ }
+
+ // Filter based on port.
+ if (_rtcpFilterPort != 0 &&
+ _rtcpFilterPort != portNr)
+ {
+ // Drop packet.
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceTransport,
+ _id,
+ "Incoming RTCP packet blocked by filter incoming from port:%d\
+ allowed port:%d",
+ portNr,
+ _rtpFilterPort);
+ return;
+ }
+ _fromPortRTCP = portNr;
+ }
+
+ CriticalSectionScoped cs(_critPacketCallback);
+ if (_packetCallback)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
+ "Incoming RTCP packet from ip:%s port:%d", ipAddress,
+ portNr);
+ _packetCallback->IncomingRTCPPacket(rtcpPacket, rtcpPacketLength,
+ ipAddress, portNr);
+ }
+}
+
+bool UdpTransportImpl::FilterIPAddress(const SocketAddress* fromAddress)
+{
+ if(fromAddress->_sockaddr_storage.sin_family == AF_INET)
+ {
+ if (_filterIPAddress._sockaddr_storage.sin_family == AF_INET)
+ {
+ // IP is stored in sin_addr.
+ if (_filterIPAddress._sockaddr_in.sin_addr != 0 &&
+ (_filterIPAddress._sockaddr_in.sin_addr !=
+ fromAddress->_sockaddr_in.sin_addr))
+ {
+ return false;
+ }
+ }
+ }
+ else if(fromAddress->_sockaddr_storage.sin_family == AF_INET6)
+ {
+ if (_filterIPAddress._sockaddr_storage.sin_family == AF_INET6)
+ {
+ // IP is stored in sin_6addr.
+ for (WebRtc_Word32 i = 0; i < 4; i++)
+ {
+ if (_filterIPAddress._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i] != 0 &&
+ _filterIPAddress._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i] != fromAddress->_sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i])
+ {
+ return false;
+ }
+ }
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ _id,
+ "UdpTransportImpl::FilterIPAddress() unknown address family");
+ return false;
+ }
+ return true;
+}
+
+void UdpTransportImpl::CloseReceiveSockets()
+{
+ if(_ptrRtpSocket)
+ {
+ _ptrRtpSocket->CloseBlocking();
+ _ptrRtpSocket = NULL;
+ }
+ if(_ptrRtcpSocket)
+ {
+ _ptrRtcpSocket->CloseBlocking();
+ _ptrRtcpSocket = NULL;
+ }
+ _receiving = false;
+}
+
+void UdpTransportImpl::CloseSendSockets()
+{
+ if(_ptrSendRtpSocket)
+ {
+ _ptrSendRtpSocket->CloseBlocking();
+ _ptrSendRtpSocket = 0;
+ }
+ if(_ptrSendRtcpSocket)
+ {
+ _ptrSendRtcpSocket->CloseBlocking();
+ _ptrSendRtcpSocket = 0;
+ }
+}
+
+WebRtc_UWord16 UdpTransport::Htons(const WebRtc_UWord16 port)
+{
+ return htons(port);
+}
+
+WebRtc_UWord32 UdpTransport::Htonl(const WebRtc_UWord32 a)
+{
+ return htonl(a);
+}
+
+WebRtc_UWord32 UdpTransport::InetAddrIPV4(const char* ip)
+{
+ return ::inet_addr(ip);
+}
+
+WebRtc_Word32 UdpTransport::InetPresentationToNumeric(WebRtc_Word32 af,
+ const char* src,
+ void* dst)
+{
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ const WebRtc_Word32 result = inet_pton(af, src, dst);
+ return result > 0 ? 0 : -1;
+
+#elif defined(_WIN32)
+ SocketAddress temp;
+ int length=sizeof(SocketAddress);
+
+ if(af == AF_INET)
+ {
+ WebRtc_Word32 result = WSAStringToAddressA(
+ (const LPSTR)src,
+ af,
+ 0,
+ reinterpret_cast<struct sockaddr*>(&temp),
+ &length);
+ if(result != 0)
+ {
+ return -1;
+ }
+ memcpy(dst,&(temp._sockaddr_in.sin_addr),
+ sizeof(temp._sockaddr_in.sin_addr));
+ return 0;
+ }
+ else if(af == AF_INET6)
+ {
+ WebRtc_Word32 result = WSAStringToAddressA(
+ (const LPSTR)src,
+ af,
+ 0,
+ reinterpret_cast<struct sockaddr*>(&temp),
+ &length);
+ if(result !=0)
+ {
+ return -1;
+ }
+ memcpy(dst,&(temp._sockaddr_in6.sin6_addr),
+ sizeof(temp._sockaddr_in6.sin6_addr));
+ return 0;
+
+ }else
+ {
+ return -1;
+ }
+#else
+ return -1;
+#endif
+}
+
+WebRtc_Word32 UdpTransport::LocalHostAddressIPV6(char n_localIP[16])
+{
+
+#if defined(_WIN32)
+ struct addrinfo *result = NULL;
+ struct addrinfo *ptr = NULL;
+ struct addrinfo hints;
+
+ ZeroMemory(&hints, sizeof(hints));
+ hints.ai_family = AF_INET6;
+
+ char szHostName[256] = "";
+ if(::gethostname(szHostName, sizeof(szHostName) - 1))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1, "gethostname failed");
+ return -1;
+ }
+
+ DWORD dwRetval = getaddrinfo(szHostName, NULL, &hints, &result);
+ if ( dwRetval != 0 )
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
+ "getaddrinfo failed, error:%d", dwRetval);
+ return -1;
+ }
+ for(ptr=result; ptr != NULL ;ptr=ptr->ai_next)
+ {
+ switch (ptr->ai_family)
+ {
+ case AF_INET6:
+ {
+ for(int i = 0; i< 16; i++)
+ {
+ n_localIP[i] = (*(SocketAddress*)ptr->ai_addr).
+ _sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u8[i];
+ }
+ bool islocalIP = true;
+
+ for(int n = 0; n< 15; n++)
+ {
+ if(n_localIP[n] != 0)
+ {
+ islocalIP = false;
+ break;
+ }
+ }
+
+ if(islocalIP && n_localIP[15] != 1)
+ {
+ islocalIP = false;
+ }
+
+ if(islocalIP && ptr->ai_next)
+ {
+ continue;
+ }
+ if(n_localIP[0] == 0xfe &&
+ n_localIP[1] == 0x80 && ptr->ai_next)
+ {
+ continue;
+ }
+ freeaddrinfo(result);
+ }
+ return 0;
+ default:
+ break;
+ };
+ }
+ freeaddrinfo(result);
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
+ "getaddrinfo failed to find address");
+ return -1;
+
+#elif defined(WEBRTC_MAC)
+ struct ifaddrs* ptrIfAddrs = NULL;
+ struct ifaddrs* ptrIfAddrsStart = NULL;
+
+ getifaddrs(&ptrIfAddrsStart);
+ ptrIfAddrs = ptrIfAddrsStart;
+ while(ptrIfAddrs)
+ {
+ if(ptrIfAddrs->ifa_addr->sa_family == AF_INET6)
+ {
+ const struct sockaddr_in6* sock_in6 =
+ reinterpret_cast<struct sockaddr_in6*>(ptrIfAddrs->ifa_addr);
+ const struct in6_addr* sin6_addr = &sock_in6->sin6_addr;
+
+ if (IN6_IS_ADDR_LOOPBACK(sin6_addr) ||
+ IN6_IS_ADDR_LINKLOCAL(sin6_addr)) {
+ ptrIfAddrs = ptrIfAddrs->ifa_next;
+ continue;
+ }
+ memcpy(n_localIP, sin6_addr->s6_addr, sizeof(sin6_addr->s6_addr));
+ freeifaddrs(ptrIfAddrsStart);
+ return 0;
+ }
+ ptrIfAddrs = ptrIfAddrs->ifa_next;
+ }
+ freeifaddrs(ptrIfAddrsStart);
+ return -1;
+#elif defined(WEBRTC_ANDROID)
+ return -1;
+#else // WEBRTC_LINUX
+ struct
+ {
+ struct nlmsghdr n;
+ struct ifaddrmsg r;
+ } req;
+
+ struct rtattr* rta = NULL;
+ int status;
+ char buf[16384]; // = 16 * 1024 (16 kB)
+ struct nlmsghdr* nlmp;
+ struct ifaddrmsg* rtmp;
+ struct rtattr* rtatp;
+ int rtattrlen;
+ struct in6_addr* in6p;
+
+ int fd = socket(PF_NETLINK, SOCK_DGRAM, NETLINK_ROUTE);
+ if (fd == -1)
+ {
+ return -1;
+ }
+
+ // RTM_GETADDR is used to fetch the ip address from the kernel interface
+ // table. Populate the msg structure (req) the size of the message buffer
+ // is specified to netlinkmessage header, and flags values are set as
+ // NLM_F_ROOT | NLM_F_REQUEST.
+ // The request flag must be set for all messages requesting the data from
+ // kernel. The root flag is used to notify the kernel to return the full
+ // tabel. Another flag (not used) is NLM_F_MATCH. This is used to get only
+ // specified entries in the table. At the time of writing this program this
+ // flag is not implemented in kernel
+
+ memset(&req, 0, sizeof(req));
+ req.n.nlmsg_len = NLMSG_LENGTH(sizeof(struct ifaddrmsg));
+ req.n.nlmsg_flags = NLM_F_REQUEST | NLM_F_ROOT;
+ req.n.nlmsg_type = RTM_GETADDR;
+ req.r.ifa_family = AF_INET6;
+
+ // Fill up all the attributes for the rtnetlink header.
+ // The lenght is very important. 16 signifies the ipv6 address.
+ rta = (struct rtattr*)(((char*)&req) + NLMSG_ALIGN(req.n.nlmsg_len));
+ rta->rta_len = RTA_LENGTH(16);
+
+ status = send(fd, &req, req.n.nlmsg_len, 0);
+ if (status < 0)
+ {
+ close(fd);
+ return -1;
+ }
+ status = recv(fd, buf, sizeof(buf), 0);
+ if (status < 0)
+ {
+ close(fd);
+ return -1;
+ }
+ if(status == 0)
+ {
+ close(fd);
+ return -1;
+ }
+ close(fd);
+
+ // The message is stored in buff. Parse the message to get the requested
+ // data.
+ {
+ nlmp = (struct nlmsghdr*)buf;
+ int len = nlmp->nlmsg_len;
+ int req_len = len - sizeof(*nlmp);
+
+ if (req_len < 0 || len > status)
+ {
+ return -1;
+ }
+ if (!NLMSG_OK_NO_WARNING(nlmp, status))
+ {
+ return -1;
+ }
+ rtmp = (struct ifaddrmsg*)NLMSG_DATA(nlmp);
+ rtatp = (struct rtattr*)IFA_RTA(rtmp);
+
+ rtattrlen = IFA_PAYLOAD(nlmp);
+
+ for (; RTA_OK(rtatp, rtattrlen); rtatp = RTA_NEXT(rtatp, rtattrlen))
+ {
+
+ // Here we hit the fist chunk of the message. Time to validate the
+ // type. For more info on the different types see
+ // "man(7) rtnetlink" The table below is taken from man pages.
+ // Attributes
+ // rta_type value type description
+ // -------------------------------------------------------------
+ // IFA_UNSPEC - unspecified.
+ // IFA_ADDRESS raw protocol address interface address
+ // IFA_LOCAL raw protocol address local address
+ // IFA_LABEL asciiz string name of the interface
+ // IFA_BROADCAST raw protocol address broadcast address.
+ // IFA_ANYCAST raw protocol address anycast address
+ // IFA_CACHEINFO struct ifa_cacheinfo Address information.
+
+ if(rtatp->rta_type == IFA_ADDRESS)
+ {
+ bool islocalIP = true;
+ in6p = (struct in6_addr*)RTA_DATA(rtatp);
+ for(int n = 0; n< 15; n++)
+ {
+ if(in6p->s6_addr[n] != 0)
+ {
+ islocalIP = false;
+ break;
+ }
+ }
+ if(islocalIP && in6p->s6_addr[15] != 1)
+ {
+ islocalIP = false;
+ }
+ if(!islocalIP)
+ {
+ for(int i = 0; i< 16; i++)
+ {
+ n_localIP[i] = in6p->s6_addr[i];
+ }
+ if(n_localIP[0] == static_cast<char> (0xfe)
+ && n_localIP[1] == static_cast<char>(0x80) )
+ {
+ // Auto configured IP.
+ continue;
+ }
+ break;
+ }
+ }
+ }
+ }
+ return 0;
+#endif
+}
+
+WebRtc_Word32 UdpTransport::LocalHostAddress(WebRtc_UWord32& localIP)
+{
+ #if defined(_WIN32)
+ hostent* localHost;
+ localHost = gethostbyname( "" );
+ if(localHost)
+ {
+ if(localHost->h_addrtype != AF_INET)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ -1,
+ "LocalHostAddress can only get local IP for IP Version 4");
+ return -1;
+ }
+ localIP= Htonl(
+ (*(struct in_addr *)localHost->h_addr_list[0]).S_un.S_addr);
+ return 0;
+ }
+ else
+ {
+ WebRtc_Word32 error = WSAGetLastError();
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
+ "gethostbyname failed, error:%d", error);
+ return -1;
+ }
+#elif (defined(WEBRTC_MAC))
+ char localname[255];
+ if (gethostname(localname, 255) != -1)
+ {
+ hostent* localHost;
+ localHost = gethostbyname(localname);
+ if(localHost)
+ {
+ if(localHost->h_addrtype != AF_INET)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceTransport,
+ -1,
+ "LocalHostAddress can only get local IP for IP Version 4");
+ return -1;
+ }
+ localIP = Htonl((*(struct in_addr*)*localHost->h_addr_list).s_addr);
+ return 0;
+ }
+ }
+ WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1, "gethostname failed");
+ return -1;
+#else // WEBRTC_LINUX
+ int sockfd, size = 1;
+ struct ifreq* ifr;
+ struct ifconf ifc;
+
+ if (0 > (sockfd = socket(AF_INET, SOCK_DGRAM, IPPROTO_IP)))
+ {
+ return -1;
+ }
+ ifc.ifc_len = IFRSIZE;
+ ifc.ifc_req = NULL;
+ do
+ {
+ ++size;
+ // Buffer size needed is unknown. Try increasing it until no overflow
+ // occurs.
+ if (NULL == (ifc.ifc_req = (ifreq*)realloc(ifc.ifc_req, IFRSIZE))) {
+ fprintf(stderr, "Out of memory.\n");
+ exit(EXIT_FAILURE);
+ }
+ ifc.ifc_len = IFRSIZE;
+ if (ioctl(sockfd, SIOCGIFCONF, &ifc))
+ {
+ close(sockfd);
+ return -1;
+ }
+ } while (IFRSIZE <= ifc.ifc_len);
+
+ ifr = ifc.ifc_req;
+ for (;(char *) ifr < (char *) ifc.ifc_req + ifc.ifc_len; ++ifr)
+ {
+ if (ifr->ifr_addr.sa_data == (ifr+1)->ifr_addr.sa_data)
+ {
+ continue; // duplicate, skip it
+ }
+ if (ioctl(sockfd, SIOCGIFFLAGS, ifr))
+ {
+ continue; // failed to get flags, skip it
+ }
+ if(strncmp(ifr->ifr_name, "lo",3) == 0)
+ {
+ continue;
+ }else
+ {
+ struct sockaddr* saddr = &(ifr->ifr_addr);
+ SocketAddress* socket_addess = reinterpret_cast<SocketAddress*>(
+ saddr);
+ localIP = Htonl(socket_addess->_sockaddr_in.sin_addr);
+ close(sockfd);
+ return 0;
+ }
+ }
+ close(sockfd);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 UdpTransport::IPAddress(const SocketAddress& address,
+ char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort)
+{
+ #if defined(_WIN32)
+ DWORD dwIPSize = ipSize;
+ WebRtc_Word32 returnvalue = WSAAddressToStringA((LPSOCKADDR)(&address),
+ sizeof(SocketAddress),
+ NULL,
+ ip,
+ &dwIPSize);
+ if(returnvalue == -1)
+ {
+ return -1;
+ }
+
+ WebRtc_UWord16 source_port = 0;
+ if(address._sockaddr_storage.sin_family == AF_INET)
+ {
+ // Parse IP assuming format "a.b.c.d:port".
+ WebRtc_Word8* ipEnd = strchr(ip,':');
+ if(ipEnd != NULL)
+ {
+ *ipEnd = '\0';
+ }
+ ipSize = (WebRtc_Word32)strlen(ip);
+ if(ipSize == 0)
+ {
+ return -1;
+ }
+ source_port = address._sockaddr_in.sin_port;
+ }
+ else
+ {
+ // Parse IP assuming format "[address]:port".
+ WebRtc_Word8* ipEnd = strchr(ip,']');
+ if(ipEnd != NULL)
+ {
+ // Calculate length
+ WebRtc_Word32 adrSize = WebRtc_Word32(ipEnd - ip) - 1;
+ memmove(ip, &ip[1], adrSize); // Remove '['
+ *(ipEnd - 1) = '\0';
+ }
+ ipSize = (WebRtc_Word32)strlen(ip);
+ if(ipSize == 0)
+ {
+ return -1;
+ }
+
+ source_port = address._sockaddr_in6.sin6_port;
+ }
+ // Convert port number to network byte order.
+ sourcePort = htons(source_port);
+ return 0;
+
+ #elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ WebRtc_Word32 ipFamily = address._sockaddr_storage.sin_family;
+ const void* ptrNumericIP = NULL;
+
+ if(ipFamily == AF_INET)
+ {
+ ptrNumericIP = &(address._sockaddr_in.sin_addr);
+ }
+ else if(ipFamily == AF_INET6)
+ {
+ ptrNumericIP = &(address._sockaddr_in6.sin6_addr);
+ }
+ else
+ {
+ return -1;
+ }
+ if(inet_ntop(ipFamily, ptrNumericIP, ip, ipSize) == NULL)
+ {
+ return -1;
+ }
+ WebRtc_UWord16 source_port;
+ if(ipFamily == AF_INET)
+ {
+ source_port = address._sockaddr_in.sin_port;
+ } else
+ {
+ source_port = address._sockaddr_in6.sin6_port;
+ }
+ // Convert port number to network byte order.
+ sourcePort = htons(source_port);
+ return 0;
+ #else
+ return -1;
+ #endif
+}
+
+bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
+{
+ if(ipV6)
+ {
+ WebRtc_Word32 len = (WebRtc_Word32)strlen(ipadr);
+ if( len>39 || len == 0)
+ {
+ return false;
+ }
+
+ WebRtc_Word32 i;
+ WebRtc_Word32 colonPos[7] = {0,0,0,0,0,0,0};
+ WebRtc_Word32 lastColonPos = -2;
+ WebRtc_Word32 nColons = 0;
+ WebRtc_Word32 nDubbleColons = 0;
+ WebRtc_Word32 nDots = 0;
+ WebRtc_Word32 error = 0;
+ char c;
+ for(i = 0; i < len ; i++)
+ {
+ c=ipadr[i];
+ if(isxdigit(c))
+ ;
+ else if(c == ':')
+ {
+ if(nColons < 7)
+ colonPos[nColons] = i;
+ if((i-lastColonPos)==1)
+ nDubbleColons++;
+ lastColonPos=i;
+ if(nDots != 0)
+ {
+ error = 1;
+ }
+ nColons++;
+ }
+ else if(c == '.')
+ {
+ nDots++;
+ }
+ else
+ {
+ error = 1;
+ }
+
+ }
+ if(error)
+ {
+ return false;
+ }
+ if(nDubbleColons > 1)
+ {
+ return false;
+ }
+ if(nColons > 7 || nColons < 2)
+ {
+ return false;
+ }
+ if(!(nDots == 3 || nDots == 0))
+ {
+ return false;
+ }
+ lastColonPos = -1;
+ WebRtc_Word32 charsBeforeColon = 0;
+ for(i = 0; i < nColons; i++)
+ {
+ charsBeforeColon=colonPos[i]-lastColonPos-1;
+ if(charsBeforeColon > 4)
+ {
+ return false;
+ }
+ lastColonPos=colonPos[i];
+ }
+ WebRtc_Word32 lengthAfterLastColon = len - lastColonPos - 1;
+ if(nDots == 0)
+ {
+ if(lengthAfterLastColon > 4)
+ return false;
+ }
+ if(nDots == 3 && lengthAfterLastColon > 0)
+ {
+ return IsIpAddressValid((ipadr+lastColonPos+1),false);
+ }
+
+ }
+ else
+ {
+ WebRtc_Word32 len = (WebRtc_Word32)strlen(ipadr);
+ if((len>15)||(len==0))
+ {
+ return false;
+ }
+
+ // IPv4 should be [0-255].[0-255].[0-255].[0-255]
+ WebRtc_Word32 i;
+ WebRtc_Word32 nDots = 0;
+ WebRtc_Word32 iDotPos[4] = {0,0,0,0};
+
+ for (i = 0; (i < len) && (nDots < 4); i++)
+ {
+ if (ipadr[i] == (char)'.')
+ {
+ // Store index of dots and count number of dots.
+ iDotPos[nDots++] = i;
+ }
+ }
+
+ bool allUnder256 = false;
+ // TODO (hellner): while loop seems to be abused here to get
+ // label like functionality. Fix later to avoid introducing bugs now.
+
+ // Check that all numbers are smaller than 256.
+ do
+ {
+ if (nDots != 3 )
+ {
+ break;
+ }
+
+ if (iDotPos[0] <= 3)
+ {
+ char nr[4];
+ memset(nr,0,4);
+ strncpy(nr,&ipadr[0],iDotPos[0]);
+ WebRtc_Word32 num = atoi(nr);
+ if (num > 255)
+ {
+ break;
+ }
+ } else {
+ break;
+ }
+
+ if (iDotPos[1] - iDotPos[0] <= 4)
+ {
+ char nr[4];
+ memset(nr,0,4);
+ strncpy(nr,&ipadr[iDotPos[0]+1], iDotPos[1] - iDotPos[0] - 1);
+ WebRtc_Word32 num = atoi(nr);
+ if (num > 255)
+ break;
+ } else {
+ break;
+ }
+
+ if (iDotPos[2] - iDotPos[1] <= 4)
+ {
+ char nr[4];
+ memset(nr,0,4);
+ strncpy(nr,&ipadr[iDotPos[1]+1], iDotPos[1] - iDotPos[0] - 1);
+ WebRtc_Word32 num = atoi(nr);
+ if (num > 255)
+ break;
+
+ memset(nr,0,4);
+ strncpy(nr,&ipadr[iDotPos[2]+1], len - iDotPos[2] -1);
+ num = atoi(nr);
+ if (num > 255)
+ break;
+ else
+ allUnder256 = true;
+ } else
+ break;
+ } while(false);
+
+ if (nDots != 3 || !allUnder256)
+ {
+ return false;
+ }
+ }
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_impl.h b/trunk/src/modules/udp_transport/source/udp_transport_impl.h
new file mode 100644
index 0000000..c5a73f9
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_transport_impl.h
@@ -0,0 +1,246 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
+#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
+
+#include "udp_transport.h"
+#include "udp_socket_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class RWLockWrapper;
+class UdpSocketManager;
+
+class UdpTransportImpl : public UdpTransport
+{
+public:
+ // Factory method. Constructor disabled.
+ UdpTransportImpl(const WebRtc_Word32 id, WebRtc_UWord8& numSocketThreads);
+ virtual ~UdpTransportImpl();
+
+ // Module functions
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ // UdpTransport functions
+ virtual WebRtc_Word32 InitializeSendSockets(
+ const char* ipAddr,
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0);
+ virtual WebRtc_Word32 InitializeReceiveSockets(
+ UdpTransportData* const packetCallback,
+ const WebRtc_UWord16 rtpPort,
+ const char* ipAddr = NULL,
+ const char* multicastIpAddr = NULL,
+ const WebRtc_UWord16 rtcpPort = 0);
+ virtual WebRtc_Word32 InitializeSourcePorts(
+ const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0);
+ virtual WebRtc_Word32 SourcePorts(WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const;
+ virtual WebRtc_Word32 ReceiveSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort,
+ char multicastIpAddr[kIpAddressVersion6Length]) const;
+ virtual WebRtc_Word32 SendSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const;
+ virtual WebRtc_Word32 RemoteSocketInformation(
+ char ipAddr[kIpAddressVersion6Length],
+ WebRtc_UWord16& rtpPort,
+ WebRtc_UWord16& rtcpPort) const;
+ virtual WebRtc_Word32 SetQoS(const bool QoS,
+ const WebRtc_Word32 serviceType,
+ const WebRtc_UWord32 maxBitrate = 0,
+ const WebRtc_Word32 overrideDSCP = 0,
+ const bool audio = false);
+ virtual WebRtc_Word32 QoS(bool& QoS, WebRtc_Word32& serviceType,
+ WebRtc_Word32& overrideDSCP) const;
+ virtual WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP,
+ const bool useSetSockOpt = false);
+ virtual WebRtc_Word32 ToS(WebRtc_Word32& DSCP,
+ bool& useSetSockOpt) const;
+ virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 PCP);
+ virtual WebRtc_Word32 PCP(WebRtc_Word32& PCP) const;
+ virtual WebRtc_Word32 EnableIpV6();
+ virtual bool IpV6Enabled() const;
+ virtual WebRtc_Word32 SetFilterIP(
+ const char filterIPAddress[kIpAddressVersion6Length]);
+ virtual WebRtc_Word32 FilterIP(
+ char filterIPAddress[kIpAddressVersion6Length]) const;
+ virtual WebRtc_Word32 SetFilterPorts(const WebRtc_UWord16 rtpFilterPort,
+ const WebRtc_UWord16 rtcpFilterPort);
+ virtual WebRtc_Word32 FilterPorts(WebRtc_UWord16& rtpFilterPort,
+ WebRtc_UWord16& rtcpFilterPort) const;
+ virtual WebRtc_Word32 StartReceiving(
+ const WebRtc_UWord32 numberOfSocketBuffers);
+ virtual WebRtc_Word32 StopReceiving();
+ virtual bool Receiving() const;
+ virtual bool SendSocketsInitialized() const;
+ virtual bool SourcePortsInitialized() const;
+ virtual bool ReceiveSocketsInitialized() const;
+ virtual WebRtc_Word32 SendRaw(const WebRtc_Word8* data,
+ WebRtc_UWord32 length, WebRtc_Word32 isRTCP,
+ WebRtc_UWord16 portnr = 0,
+ const char* ip = NULL);
+ virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8 *data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to);
+ virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8 *data,
+ WebRtc_UWord32 length,
+ const SocketAddress& to);
+ virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8 *data,
+ WebRtc_UWord32 length,
+ WebRtc_UWord16 rtpPort);
+ virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8 *data,
+ WebRtc_UWord32 length,
+ WebRtc_UWord16 rtcpPort);
+ // Transport functions
+ virtual int SendPacket(int channel, const void* data, int length);
+ virtual int SendRTCPPacket(int channel, const void* data, int length);
+
+ // UdpTransport functions continue.
+ virtual WebRtc_Word32 SetSendIP(const char* ipaddr);
+ virtual WebRtc_Word32 SetSendPorts(const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort = 0);
+
+ virtual ErrorCode LastError() const;
+
+ virtual WebRtc_Word32 IPAddressCached(const SocketAddress& address,
+ char* ip,
+ WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort);
+
+ WebRtc_Word32 Id() const {return _id;}
+protected:
+ // IncomingSocketCallback signature functions for receiving callbacks from
+ // UdpSocketWrapper.
+ static void IncomingRTPCallback(CallbackObj obj,
+ const WebRtc_Word8* rtpPacket,
+ WebRtc_Word32 rtpPacketLength,
+ const SocketAddress* from);
+ static void IncomingRTCPCallback(CallbackObj obj,
+ const WebRtc_Word8* rtcpPacket,
+ WebRtc_Word32 rtcpPacketLength,
+ const SocketAddress* from);
+
+ void CloseSendSockets();
+ void CloseReceiveSockets();
+
+ // Update _remoteRTPAddr according to _destPort and _destIP
+ void BuildRemoteRTPAddr();
+ // Update _remoteRTCPAddr according to _destPortRTCP and _destIP
+ void BuildRemoteRTCPAddr();
+
+ void BuildSockaddrIn(WebRtc_UWord16 portnr, const char* ip,
+ SocketAddress& remoteAddr) const;
+
+ ErrorCode BindLocalRTPSocket();
+ ErrorCode BindLocalRTCPSocket();
+
+ ErrorCode BindRTPSendSocket();
+ ErrorCode BindRTCPSendSocket();
+
+ void IncomingRTPFunction(const WebRtc_Word8* rtpPacket,
+ WebRtc_Word32 rtpPacketLength,
+ const SocketAddress* from);
+ void IncomingRTCPFunction(const WebRtc_Word8* rtcpPacket,
+ WebRtc_Word32 rtcpPacketLength,
+ const SocketAddress* from);
+
+ bool FilterIPAddress(const SocketAddress* fromAddress);
+
+ bool SetSockOptUsed();
+
+ WebRtc_Word32 EnableQoS(WebRtc_Word32 serviceType, bool audio,
+ WebRtc_UWord32 maxBitrate,
+ WebRtc_Word32 overrideDSCP);
+
+ WebRtc_Word32 DisableQoS();
+
+private:
+ void GetCachedAddress(char* ip, WebRtc_UWord32& ipSize,
+ WebRtc_UWord16& sourcePort);
+
+ WebRtc_Word32 _id;
+ // Protects the sockets from being re-configured while receiving packets.
+ CriticalSectionWrapper* _crit;
+ CriticalSectionWrapper* _critFilter;
+ // _packetCallback's critical section.
+ CriticalSectionWrapper* _critPacketCallback;
+ UdpSocketManager* _mgr;
+ ErrorCode _lastError;
+
+ // Remote RTP and RTCP ports.
+ WebRtc_UWord16 _destPort;
+ WebRtc_UWord16 _destPortRTCP;
+
+ // Local RTP and RTCP ports.
+ WebRtc_UWord16 _localPort;
+ WebRtc_UWord16 _localPortRTCP;
+
+ // Local port number when the local port for receiving and local port number
+ // for sending are not the same.
+ WebRtc_UWord16 _srcPort;
+ WebRtc_UWord16 _srcPortRTCP;
+
+ // Remote port from which last received packet was sent.
+ WebRtc_UWord16 _fromPort;
+ WebRtc_UWord16 _fromPortRTCP;
+
+ char _fromIP[kIpAddressVersion6Length];
+ char _destIP[kIpAddressVersion6Length];
+ char _localIP[kIpAddressVersion6Length];
+ char _localMulticastIP[kIpAddressVersion6Length];
+
+ UdpSocketWrapper* _ptrRtpSocket;
+ UdpSocketWrapper* _ptrRtcpSocket;
+
+ // Local port when the local port for receiving and local port for sending
+ // are not the same.
+ UdpSocketWrapper* _ptrSendRtpSocket;
+ UdpSocketWrapper* _ptrSendRtcpSocket;
+
+ SocketAddress _remoteRTPAddr;
+ SocketAddress _remoteRTCPAddr;
+
+ SocketAddress _localRTPAddr;
+ SocketAddress _localRTCPAddr;
+
+ WebRtc_Word32 _tos;
+ bool _receiving;
+ bool _useSetSockOpt;
+ bool _qos;
+ WebRtc_Word32 _pcp;
+ bool _ipV6Enabled;
+ WebRtc_Word32 _serviceType;
+ WebRtc_Word32 _overrideDSCP;
+ WebRtc_UWord32 _maxBitrate;
+
+ // Cache used by GetCachedAddress(..).
+ RWLockWrapper* _cachLock;
+ SocketAddress _previousAddress;
+ char _previousIP[kIpAddressVersion6Length];
+ WebRtc_UWord32 _previousIPSize;
+ WebRtc_UWord16 _previousSourcePort;
+
+ SocketAddress _filterIPAddress;
+ WebRtc_UWord16 _rtpFilterPort;
+ WebRtc_UWord16 _rtcpFilterPort;
+
+ UdpTransportData* _packetCallback;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc b/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc
new file mode 100644
index 0000000..b894435
--- /dev/null
+++ b/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "udp_transport.h"
+#include "gtest/gtest.h"
+
+TEST(UDPTransportTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/udp_transport/test/SocketManagerTest.cc b/trunk/src/modules/udp_transport/test/SocketManagerTest.cc
new file mode 100644
index 0000000..03119be
--- /dev/null
+++ b/trunk/src/modules/udp_transport/test/SocketManagerTest.cc
@@ -0,0 +1,449 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <iostream>
+
+#ifdef _WIN32
+#include <windows.h>
+#include <tchar.h>
+#else
+#include <stdio.h>
+#define Sleep(x) usleep(x*1000)
+#endif
+
+#include "udp_transport.h"
+#include "common_types.h"
+#include "trace.h"
+
+//#define QOS_TEST
+//#define QOS_TEST_WITH_OVERRIDE // require admin on Win7
+//#define TOS_TEST // require admin on Win7
+//#define TOS_TEST_USING_SETSOCKOPT
+//#define PCP_TEST
+
+class UdpTransportDataA: public UdpTransportData
+{
+public:
+ UdpTransportDataA() :
+ _counterRTP(0),
+ _counterRTCP(0)
+ {
+ };
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort)
+ {
+ _counterRTP++;
+ };
+
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort)
+ {
+ _counterRTCP++;
+ };
+ WebRtc_UWord32 _counterRTP;
+ WebRtc_UWord32 _counterRTCP;
+};
+
+class UdpTransportDataB: public UdpTransportData
+{
+public:
+ UdpTransportDataB() :
+ _counterRTP(0),
+ _counterRTCP(0)
+ {
+ };
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort)
+ {
+ _counterRTP++;
+ };
+
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort)
+ {
+ _counterRTCP++;
+ };
+ WebRtc_UWord32 _counterRTP;
+ WebRtc_UWord32 _counterRTCP;
+};
+
+#ifdef _WIN32
+int _tmain(int argc, _TCHAR* argv[])
+#else
+int main(int argc, char* argv[])
+#endif
+{
+ Trace::CreateTrace();
+ Trace::SetTraceFile("testTrace.txt");
+ Trace::SetEncryptedTraceFile("testTraceDebug.txt");
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+ printf("Start UdpTransport test\n");
+
+ WebRtc_UWord8 numberOfSocketThreads = 5;
+ UdpTransport* client1 = UdpTransport::Create(1,numberOfSocketThreads,NULL);
+ numberOfSocketThreads = 0;
+ UdpTransport* client2 = UdpTransport::Create(2,numberOfSocketThreads,NULL);
+ assert(5 == numberOfSocketThreads);
+
+ UdpTransportDataA* client1Callback = new UdpTransportDataA();
+ UdpTransportDataB* client2Callback = new UdpTransportDataB();
+
+ WebRtc_UWord32 localIP = 0;
+ char localIPAddr[64];
+ assert( 0 == client1->LocalHostAddress(localIP)); // network host order aka big-endian
+
+ sprintf(localIPAddr,"%lu.%lu.%lu.%lu",(localIP>>24)& 0x0ff,(localIP>>16)& 0x0ff ,(localIP>>8)& 0x0ff, localIP & 0x0ff);
+ printf("\tLocal IP:%s\n", localIPAddr);
+
+ char localIPV6[16];
+ char localIPAddrV6[128];
+ if( 0 == client1->LocalHostAddressIPV6(localIPV6))
+ {
+ sprintf(localIPAddrV6,"%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x", localIPV6[0],localIPV6[1],localIPV6[2],localIPV6[3],localIPV6[4],localIPV6[5],localIPV6[6],localIPV6[7], localIPV6[8],localIPV6[9],localIPV6[10],localIPV6[11],localIPV6[12],localIPV6[13],localIPV6[14],localIPV6[15]);
+ printf("\tLocal IPV6:%s\n", localIPAddrV6);
+ }
+
+ char test[9] = "testtest";
+ assert( 0 == client1->InitializeReceiveSockets(client1Callback,1234, localIPAddr));
+
+#if defined QOS_TEST_WITH_OVERRIDE || defined QOS_TEST || defined TOS_TEST || defined TOS_TEST_USING_SETSOCKOPT
+ assert( -1 == client1->SetQoS(true, 3, 1000)); // should fail
+ assert( 0 == client1->InitializeSendSockets("192.168.200.1", 1236,1237));
+#else
+ assert( 0 == client1->InitializeSendSockets(localIPAddr, 1236,1237));
+#endif
+ assert( 0 == client1->StartReceiving(20));
+
+ assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
+ assert( 0 == client2->InitializeSendSockets(localIPAddr, 1234,1235));
+ assert( 0 == client2->StartReceiving(20));
+
+ Sleep(10);
+
+#ifdef TOS_TEST
+ // note: you need to have QOS installed on your interface for this test
+ // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
+ assert( 0 == client1->SetToS(2));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetToS(3));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetToS(0));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+
+ printf("Tested TOS \n");
+ Sleep(5000);
+ return 0;
+#endif
+
+#ifdef TOS_TEST_USING_SETSOCKOPT
+ // note: you need to have QOS installed on your interface for this test
+ // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
+ assert( 0 == client1->SetToS(2, true));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetToS(3, true));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetToS(0, true));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+
+ printf("Tested TOS using setsockopt \n");
+ Sleep(5000);
+ return 0;
+#endif
+
+#ifdef QOS_TEST
+ // note: you need to have QOS installed on your interface for this test
+ // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
+ assert( 0 == client1->SetQoS(true, 2, 1000)); // SERVICETYPE_CONTROLLEDLOAD 2
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetQoS(true, 3, 1000)); // SERVICETYPE_GUARANTEED 3
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetQoS(false, 0)); //
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+
+ printf("Tested QOS \n");
+ Sleep(5000);
+ return 0;
+#endif
+
+#ifdef QOS_TEST_WITH_OVERRIDE
+ // note: you need to have QOS installed on your interface for this test
+ // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
+ assert( 0 == client1->SetQoS(true, 2, 1000, 1)); // SERVICETYPE_CONTROLLEDLOAD 2
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetQoS(true, 2, 1000, 2)); // SERVICETYPE_GUARANTEED 3
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ Sleep(10);
+ assert( 0 == client1->SetQoS(false, 0)); //
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ printf("Tested QOS with override \n");
+ Sleep(5000);
+ return 0;
+#endif
+
+#ifdef PCP_TEST
+ // Note: We currently don't know how to test that the bits are actually set in the frame,
+ // this test simply tests the API and that we can send a packet after setting PCP.
+ assert( -1 == client1->SetPCP(-1)); // should fail
+ assert( -1 == client1->SetPCP(8)); // should fail
+ printf("Setting PCP to 7 returned %d \n", client1->SetPCP(7));
+ printf("(Failing is normal, requires the CAP_NET_ADMIN capability to succeed.) \n");
+ Sleep(10);
+ for (int pcp = 6; pcp >= 0; --pcp)
+ {
+ assert( 0 == client1->SetPCP(pcp));
+ Sleep(10);
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ }
+ printf("Tested PCP \n");
+ Sleep(5000);
+ return 0;
+#endif
+
+ Sleep(10);
+
+ assert( 9 == client1->SendPacket(-1, test, 9));
+
+ // test start rec after a socket has revceived data
+ // result: packets received before first startReceive is saved by the OS
+/*
+ for(int i = 0; i < 100; i++)
+ {
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ }
+ Sleep(10);
+ assert( 0 == client2->StartReceiving(20));
+
+// assert( 0 == client2->StopReceiving());
+
+ Sleep(10);
+ for(int i = 0; i < 100; i++)
+ {
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ }
+
+ assert( 0 == client2->StartReceiving(20));
+
+ for(int i = 0; i < 100; i++)
+ {
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ }
+*/
+ Sleep(10);
+
+ assert( 0 == client1Callback->_counterRTP);
+ assert( 1 == client2Callback->_counterRTP);
+ assert( 0 == client1Callback->_counterRTCP);
+ assert( 0 == client2Callback->_counterRTCP);
+
+ printf("Sent 1 packet on one socket \n");
+
+ char ipAddr[64];
+ char tempIpAddr[64];
+ char ipMulticastAddr[64];
+ WebRtc_UWord16 rtpPort = 0;
+ WebRtc_UWord16 rtcpPort = 0;
+ bool reusableSocket = true;
+ assert( 0 == client2->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
+ assert( rtpPort == 1234);
+ assert( strncmp(ipAddr, localIPAddr, 16) == 0);
+
+ assert( 0 == client2->ReceiveSocketInformation(ipAddr, rtpPort, rtcpPort, ipMulticastAddr, reusableSocket));
+ assert( rtpPort == 1236);
+ assert( rtcpPort == 1237);
+ assert( strncmp(ipAddr, "0.0.0.0", 16) == 0);
+ assert( ipMulticastAddr[0] == 0);
+ assert( reusableSocket == false);
+
+ assert( 0 == client2->SendSocketInformation(ipAddr, rtpPort, rtcpPort));
+ assert( rtpPort == 1234);
+ assert( rtcpPort == 1235);
+ assert( strncmp(ipAddr,localIPAddr, 16) == 0);
+
+ const int numberOfPackets = 1000;
+ int n = 0;
+ while(n < numberOfPackets)
+ {
+ assert( 9 == client1->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client1->SendRTCPPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+ n++;
+ }
+ int loops = 0;
+ for(; loops < 100 &&
+ !(client1Callback->_counterRTP == numberOfPackets &&
+ client1Callback->_counterRTCP == numberOfPackets &&
+ client2Callback->_counterRTP == numberOfPackets+1 &&
+ client2Callback->_counterRTCP == numberOfPackets);
+ loops++)
+ {
+ Sleep(10);
+ }
+ printf("\tSent %d packets on 4 sockets in:%d ms\n", numberOfPackets, loops*10);
+
+ assert( numberOfPackets == client1Callback->_counterRTP);
+ assert( numberOfPackets+1 == client2Callback->_counterRTP);
+ assert( numberOfPackets == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+
+ assert( 0 == client1->StopReceiving());
+ assert( 0 == client2->StopReceiving());
+
+ printf("Tear down client 2\n");
+
+ // configure that fail
+ assert( -1 == client2->InitializeReceiveSockets(client2Callback,1234, localIPAddr)); // port in use
+ assert( !client2->ReceiveSocketsInitialized());
+ assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
+ assert( 0 == client2->StartReceiving(20));
+
+ printf("Client 2 re-configured\n");
+
+ assert( client1->SendSocketsInitialized());
+ assert( client1->ReceiveSocketsInitialized());
+ assert( client2->SendSocketsInitialized());
+ assert( client2->ReceiveSocketsInitialized());
+
+ assert( 9 == client1->SendPacket(-1, test, 9));
+
+ // this should not be received since we dont receive in client 1
+ assert( 9 == client2->SendPacket(-1, test, 9));
+
+ Sleep(10);
+
+ assert( numberOfPackets == client1Callback->_counterRTP);
+ assert( numberOfPackets+2 == client2Callback->_counterRTP);
+ assert( numberOfPackets == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+ printf("\tSent 1 packet on one socket \n");
+
+ printf("Start filter test\n");
+
+ assert( 0 == client1->StartReceiving(20));
+
+ assert( 0 == client1->SetFilterPorts(1234, 1235)); // should filter out what we send
+ assert( 0 == client1->SetFilterIP(localIPAddr));
+
+ assert( 0 == client1->FilterIP(tempIpAddr));
+ assert( strncmp(tempIpAddr, localIPAddr, 16) == 0);
+
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+
+ Sleep(10);
+
+ assert( numberOfPackets == client1Callback->_counterRTP);
+ assert( numberOfPackets+2 == client2Callback->_counterRTP);
+ assert( numberOfPackets == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+
+ assert( 0 == client1->SetFilterPorts(1236, 1237)); // should pass through
+
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+ printf("\tSent 1 packet on two sockets \n");
+
+ Sleep(10);
+
+ assert( numberOfPackets+1 == client1Callback->_counterRTP);
+ assert( numberOfPackets+2 == client2Callback->_counterRTP);
+ assert( numberOfPackets+1 == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+
+ assert( 0 == client1->SetFilterIP("127.0.0.2"));
+
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+ printf("\tSent 1 packet on two sockets \n");
+
+ Sleep(10);
+
+ assert( numberOfPackets+1 == client1Callback->_counterRTP);
+ assert( numberOfPackets+2 == client2Callback->_counterRTP);
+ assert( numberOfPackets+1 == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+
+ assert( 0 == client1->SetFilterIP(NULL));
+ assert( 0 == client1->SetFilterPorts(0, 0));
+
+ printf("Tested filter \n");
+
+ assert( 0 == client2->InitializeSourcePorts(1238, 1239));
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+ printf("\tSent 1 packet on two sockets \n");
+
+ Sleep(10);
+
+ assert( numberOfPackets+2 == client1Callback->_counterRTP);
+ assert( numberOfPackets+2 == client2Callback->_counterRTP);
+ assert( numberOfPackets+2 == client1Callback->_counterRTCP);
+ assert( numberOfPackets == client2Callback->_counterRTCP);
+
+ assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
+ assert( rtpPort == 1238);
+ assert( rtcpPort == 1239);
+ assert( strncmp(ipAddr, localIPAddr, 16) == 0);
+
+ printf("Tested source port \n");
+
+ assert( 0 == client2->InitializeSourcePorts(1240 ));
+ assert( 9 == client2->SendPacket(-1, test, 9));
+ assert( 9 == client2->SendRTCPPacket(-1, test, 9));
+ printf("\tSent 1 packet on two sockets \n");
+
+ Sleep(10);
+
+ assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
+ assert( rtpPort == 1240);
+ assert( rtcpPort == 1241);
+
+ printf("Tested SetSendPorts source port \n");
+
+ UdpTransport::Destroy(client1);
+ UdpTransport::Destroy(client2);
+
+ printf("\n\nUdpTransport test done\n");
+
+ delete client1Callback;
+ delete client2Callback;
+
+ Sleep(5000);
+ Trace::ReturnTrace();
+};
diff --git a/trunk/src/modules/utility/OWNERS b/trunk/src/modules/utility/OWNERS
new file mode 100644
index 0000000..674c738
--- /dev/null
+++ b/trunk/src/modules/utility/OWNERS
@@ -0,0 +1,4 @@
+henrike@webrtc.org
+pwestin@webrtc.org
+asapersson@webrtc.org
+perkj@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/utility/interface/file_player.h b/trunk/src/modules/utility/interface/file_player.h
new file mode 100644
index 0000000..29da8f1
--- /dev/null
+++ b/trunk/src/modules/utility/interface/file_player.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class FileCallback;
+
+class FilePlayer
+{
+public:
+ // The largest decoded frame size in samples (60ms with 32kHz sample rate).
+ enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
+ enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+
+ // Note: will return NULL for video file formats (e.g. AVI) if the flag
+ // WEBRTC_MODULE_UTILITY_VIDEO is not defined.
+ static FilePlayer* CreateFilePlayer(const WebRtc_UWord32 instanceID,
+ const FileFormats fileFormat);
+
+ static void DestroyFilePlayer(FilePlayer* player);
+
+ virtual WebRtc_Word32 Get10msAudioFromFile(
+ WebRtc_Word16* decodedDataBuffer,
+ WebRtc_UWord32& decodedDataLengthInSamples,
+ const WebRtc_UWord32 frequencyInHz) = 0;
+
+ // Register callback for receiving file playing notifications.
+ virtual WebRtc_Word32 RegisterModuleFileCallback(
+ FileCallback* callback) = 0;
+
+ // API for playing audio from fileName to channel.
+ // Note: codecInst is used for pre-encoded files.
+ virtual WebRtc_Word32 StartPlayingFile(
+ const char* fileName,
+ bool loop,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ // Note: codecInst is used for pre-encoded files.
+ virtual WebRtc_Word32 StartPlayingFile(
+ InStream& sourceStream,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ virtual WebRtc_Word32 StopPlayingFile() = 0;
+
+ virtual bool IsPlayingFile() const = 0;
+
+ virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs) = 0;
+
+ // Set audioCodec to the currently used audio codec.
+ virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const = 0;
+
+ virtual WebRtc_Word32 Frequency() const = 0;
+
+ // Note: scaleFactor is in the range [0.0 - 2.0]
+ virtual WebRtc_Word32 SetAudioScaling(float scaleFactor) = 0;
+
+ // Return the time in ms until next video frame should be pulled (by
+ // calling GetVideoFromFile(..)).
+ // Note: this API reads one video frame from file. This means that it should
+ // be called exactly once per GetVideoFromFile(..) API call.
+ virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;}
+
+ virtual WebRtc_Word32 StartPlayingVideoFile(
+ const char* /*fileName*/,
+ bool /*loop*/,
+ bool /*videoOnly*/) { return -1;}
+
+ virtual WebRtc_Word32 video_codec_info(VideoCodec& /*videoCodec*/) const
+ {return -1;}
+
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/)
+ { return -1;}
+
+ // Same as GetVideoFromFile(). videoFrame will have the resolution specified
+ // by the width outWidth and height outHeight in pixels.
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/,
+ const WebRtc_UWord32 /*outWidth*/,
+ const WebRtc_UWord32 /*outHeight*/)
+ {return -1;}
+protected:
+ virtual ~FilePlayer() {}
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
diff --git a/trunk/src/modules/utility/interface/file_recorder.h b/trunk/src/modules/utility/interface/file_recorder.h
new file mode 100644
index 0000000..eb460ae
--- /dev/null
+++ b/trunk/src/modules/utility/interface/file_recorder.h
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+
+#include "audio_coding_module_typedefs.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "media_file_defines.h"
+#include "module_common_types.h"
+#include "tick_util.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class FileRecorder
+{
+public:
+
+ // Note: will return NULL for video file formats (e.g. AVI) if the flag
+ // WEBRTC_MODULE_UTILITY_VIDEO is not defined.
+ static FileRecorder* CreateFileRecorder(const WebRtc_UWord32 instanceID,
+ const FileFormats fileFormat);
+
+ static void DestroyFileRecorder(FileRecorder* recorder);
+
+ virtual WebRtc_Word32 RegisterModuleFileCallback(
+ FileCallback* callback) = 0;
+
+ virtual FileFormats RecordingFileFormat() const = 0;
+
+ virtual WebRtc_Word32 StartRecordingAudioFile(
+ const char* fileName,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notification,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
+
+ virtual WebRtc_Word32 StartRecordingAudioFile(
+ OutStream& destStream,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notification,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
+
+ // Stop recording.
+ // Note: this API is for both audio and video.
+ virtual WebRtc_Word32 StopRecording() = 0;
+
+ // Return true if recording.
+ // Note: this API is for both audio and video.
+ virtual bool IsRecording() const = 0;
+
+ virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0;
+
+ // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
+ virtual WebRtc_Word32 RecordAudioToFile(
+ const AudioFrame& frame,
+ const TickTime* playoutTS = NULL) = 0;
+
+ // Open/create the file specified by fileName for writing audio/video data
+ // (relative path is allowed). audioCodecInst specifies the encoding of the
+ // audio data. videoCodecInst specifies the encoding of the video data.
+ // Only video data will be recorded if videoOnly is true. amrFormat
+ // specifies the amr/amrwb storage format.
+ // Note: the file format is AVI.
+ virtual WebRtc_Word32 StartRecordingVideoFile(
+ const char* fileName,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage,
+ bool videoOnly = false) = 0;
+
+ // Record the video frame in videoFrame to AVI file.
+ virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame) = 0;
+
+protected:
+ virtual ~FileRecorder() {}
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
diff --git a/trunk/src/modules/utility/interface/process_thread.h b/trunk/src/modules/utility/interface/process_thread.h
new file mode 100644
index 0000000..6c51404
--- /dev/null
+++ b/trunk/src/modules/utility/interface/process_thread.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class Module;
+
+class ProcessThread
+{
+public:
+ static ProcessThread* CreateProcessThread();
+ static void DestroyProcessThread(ProcessThread* module);
+
+ virtual WebRtc_Word32 Start() = 0;
+ virtual WebRtc_Word32 Stop() = 0;
+
+ virtual WebRtc_Word32 RegisterModule(const Module* module) = 0;
+ virtual WebRtc_Word32 DeRegisterModule(const Module* module) = 0;
+protected:
+ virtual ~ProcessThread();
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
diff --git a/trunk/src/modules/utility/interface/rtp_dump.h b/trunk/src/modules/utility/interface/rtp_dump.h
new file mode 100644
index 0000000..9291a1c
--- /dev/null
+++ b/trunk/src/modules/utility/interface/rtp_dump.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file implements a class that writes a stream of RTP and RTCP packets
+// to a file according to the format specified by rtpplay. See
+// http://www.cs.columbia.edu/irt/software/rtptools/.
+// Notes: supported platforms are Windows, Linux and Mac OSX
+
+#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
+
+#include "typedefs.h"
+#include "file_wrapper.h"
+
+namespace webrtc {
+class RtpDump
+{
+public:
+ // Factory method.
+ static RtpDump* CreateRtpDump();
+
+ // Delete function. Destructor disabled.
+ static void DestroyRtpDump(RtpDump* object);
+
+ // Open the file fileNameUTF8 for writing RTP/RTCP packets.
+ // Note: this API also adds the rtpplay header.
+ virtual WebRtc_Word32 Start(const char* fileNameUTF8) = 0;
+
+ // Close the existing file. No more packets will be recorded.
+ virtual WebRtc_Word32 Stop() = 0;
+
+ // Return true if a file is open for recording RTP/RTCP packets.
+ virtual bool IsActive() const = 0;
+
+ // Writes the RTP/RTCP packet in packet with length packetLength in bytes.
+ // Note: packet should contain the RTP/RTCP part of the packet. I.e. the
+ // first bytes of packet should be the RTP/RTCP header.
+ virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
+ WebRtc_UWord16 packetLength) = 0;
+
+protected:
+ virtual ~RtpDump();
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
diff --git a/trunk/src/modules/utility/source/Android.mk b/trunk/src/modules/utility/source/Android.mk
new file mode 100644
index 0000000..452a95b
--- /dev/null
+++ b/trunk/src/modules/utility/source/Android.mk
@@ -0,0 +1,55 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_utility
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := coder.cc \
+ file_player_impl.cc \
+ file_recorder_impl.cc \
+ process_thread_impl.cc \
+ rtp_dump_impl.cc \
+ frame_scaler.cc \
+ video_coder.cc \
+ video_frames_queue.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../audio_coding/main/interface \
+ $(LOCAL_PATH)/../../media_file/interface \
+ $(LOCAL_PATH)/../../video_coding/main/interface \
+ $(LOCAL_PATH)/../../video_coding/codecs/interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../common_audio/resampler/include \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+
diff --git a/trunk/src/modules/utility/source/coder.cc b/trunk/src/modules/utility/source/coder.cc
new file mode 100644
index 0000000..b858da1
--- /dev/null
+++ b/trunk/src/modules/utility/source/coder.cc
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "coder.h"
+#include "common_types.h"
+#include "module_common_types.h"
+
+// OS independent case insensitive string comparison.
+#ifdef WIN32
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+ #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+namespace webrtc {
+AudioCoder::AudioCoder(WebRtc_UWord32 instanceID)
+ : _instanceID(instanceID),
+ _acm(AudioCodingModule::Create(instanceID)),
+ _receiveCodec(),
+ _encodeTimestamp(0),
+ _encodedData(NULL),
+ _encodedLengthInBytes(0),
+ _decodeTimestamp(0)
+{
+ _acm->InitializeSender();
+ _acm->InitializeReceiver();
+ _acm->RegisterTransportCallback(this);
+}
+
+AudioCoder::~AudioCoder()
+{
+ AudioCodingModule::Destroy(_acm);
+}
+
+WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat)
+{
+ if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat)
+{
+ if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
+ {
+ return -1;
+ }
+ memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst));
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
+ WebRtc_UWord32 sampFreqHz,
+ const WebRtc_Word8* incomingPayload,
+ WebRtc_Word32 payloadLength)
+{
+ if (payloadLength > 0)
+ {
+ const WebRtc_UWord8 payloadType = _receiveCodec.pltype;
+ _decodeTimestamp += _receiveCodec.pacsize;
+ if(_acm->IncomingPayload(incomingPayload,
+ payloadLength,
+ payloadType,
+ _decodeTimestamp) == -1)
+ {
+ return -1;
+ }
+ }
+ return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz,
+ (AudioFrame&)decodedAudio);
+}
+
+WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
+ WebRtc_UWord16& sampFreqHz)
+{
+ return _acm->PlayoutData10Ms(sampFreqHz, (AudioFrame&)decodedAudio);
+}
+
+WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
+ WebRtc_Word8* encodedData,
+ WebRtc_UWord32& encodedLengthInBytes)
+{
+ // Fake a timestamp in case audio doesn't contain a correct timestamp.
+ // Make a local copy of the audio frame since audio is const
+ AudioFrame audioFrame = audio;
+ audioFrame._timeStamp = _encodeTimestamp;
+ _encodeTimestamp += audioFrame._payloadDataLengthInSamples;
+
+ // For any codec with a frame size that is longer than 10 ms the encoded
+ // length in bytes should be zero until a a full frame has been encoded.
+ _encodedLengthInBytes = 0;
+ if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
+ {
+ return -1;
+ }
+ _encodedData = encodedData;
+ if(_acm->Process() == -1)
+ {
+ return -1;
+ }
+ encodedLengthInBytes = _encodedLengthInBytes;
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::SendData(
+ FrameType /* frameType */,
+ WebRtc_UWord8 /* payloadType */,
+ WebRtc_UWord32 /* timeStamp */,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* /* fragmentation*/)
+{
+ memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize);
+ _encodedLengthInBytes = payloadSize;
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/coder.h b/trunk/src/modules/utility/source/coder.h
new file mode 100644
index 0000000..e96f455
--- /dev/null
+++ b/trunk/src/modules/utility/source/coder.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+
+#include "audio_coding_module.h"
+#include "common_types.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class AudioFrame;
+
+class AudioCoder : public AudioPacketizationCallback
+{
+public:
+ AudioCoder(WebRtc_UWord32 instanceID);
+ ~AudioCoder();
+
+ WebRtc_Word32 SetEncodeCodec(
+ const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
+
+ WebRtc_Word32 SetDecodeCodec(
+ const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
+
+ WebRtc_Word32 Decode(AudioFrame& decodedAudio, WebRtc_UWord32 sampFreqHz,
+ const WebRtc_Word8* incomingPayload,
+ WebRtc_Word32 payloadLength);
+
+ WebRtc_Word32 PlayoutData(AudioFrame& decodedAudio,
+ WebRtc_UWord16& sampFreqHz);
+
+ WebRtc_Word32 Encode(const AudioFrame& audio,
+ WebRtc_Word8* encodedData,
+ WebRtc_UWord32& encodedLengthInBytes);
+
+protected:
+ virtual WebRtc_Word32 SendData(FrameType frameType,
+ WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+
+private:
+ WebRtc_UWord32 _instanceID;
+
+ AudioCodingModule* _acm;
+
+ CodecInst _receiveCodec;
+
+ WebRtc_UWord32 _encodeTimestamp;
+ WebRtc_Word8* _encodedData;
+ WebRtc_UWord32 _encodedLengthInBytes;
+
+ WebRtc_UWord32 _decodeTimestamp;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
diff --git a/trunk/src/modules/utility/source/file_player_impl.cc b/trunk/src/modules/utility/source/file_player_impl.cc
new file mode 100644
index 0000000..0e22872
--- /dev/null
+++ b/trunk/src/modules/utility/source/file_player_impl.cc
@@ -0,0 +1,723 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "file_player_impl.h"
+#include "trace.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ #include "cpu_wrapper.h"
+ #include "frame_scaler.h"
+ #include "tick_util.h"
+ #include "video_coder.h"
+#endif
+
+// OS independent case insensitive string comparison.
+#ifdef WIN32
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+ #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+namespace webrtc {
+FilePlayer* FilePlayer::CreateFilePlayer(WebRtc_UWord32 instanceID,
+ FileFormats fileFormat)
+{
+ switch(fileFormat)
+ {
+ case kFileFormatWavFile:
+ case kFileFormatCompressedFile:
+ case kFileFormatPreencodedFile:
+ case kFileFormatPcm16kHzFile:
+ case kFileFormatPcm8kHzFile:
+ case kFileFormatPcm32kHzFile:
+ // audio formats
+ return new FilePlayerImpl(instanceID, fileFormat);
+ case kFileFormatAviFile:
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ return new VideoFilePlayerImpl(instanceID, fileFormat);
+#else
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "Invalid file format: %d", kFileFormatAviFile);
+ assert(false);
+ return NULL;
+#endif
+ }
+ assert(false);
+ return NULL;
+}
+
+void FilePlayer::DestroyFilePlayer(FilePlayer* player)
+{
+ delete player;
+}
+
+FilePlayerImpl::FilePlayerImpl(const WebRtc_UWord32 instanceID,
+ const FileFormats fileFormat)
+ : _instanceID(instanceID),
+ _fileFormat(fileFormat),
+ _fileModule(*MediaFile::CreateMediaFile(instanceID)),
+ _decodedLengthInMS(0),
+ _decodedAudioBuffer(),
+ _audioDecoder(instanceID),
+ _codec(),
+ _numberOf10MsPerFrame(0),
+ _numberOf10MsInDecoder(0),
+ _resampler(),
+ _scaling(1.0)
+{
+ _codec.plfreq = 0;
+}
+
+FilePlayerImpl::~FilePlayerImpl()
+{
+ MediaFile::DestroyMediaFile(&_fileModule);
+}
+
+WebRtc_Word32 FilePlayerImpl::Frequency() const
+{
+ if(_codec.plfreq == 0)
+ {
+ return -1;
+ }
+ // Make sure that sample rate is 8,16 or 32 kHz. E.g. WAVE files may have
+ // other sampling rates.
+ if(_codec.plfreq == 11000)
+ {
+ return 16000;
+ }
+ else if(_codec.plfreq == 22000)
+ {
+ return 32000;
+ }
+ else if(_codec.plfreq == 44000)
+ {
+ return 32000;
+ }
+ else if(_codec.plfreq == 48000)
+ {
+ return 32000;
+ }
+ else
+ {
+ return _codec.plfreq;
+ }
+}
+
+WebRtc_Word32 FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const
+{
+ audioCodec = _codec;
+ return 0;
+}
+
+WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile(
+ WebRtc_Word16* outBuffer,
+ WebRtc_UWord32& lengthInSamples,
+ WebRtc_UWord32 frequencyInHz)
+{
+ if(_codec.plfreq == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
+ "FilePlayerImpl::Get10msAudioFromFile() playing not started!\
+ codecFreq = %d, wantedFreq = %d",
+ _codec.plfreq, frequencyInHz);
+ return -1;
+ }
+
+ AudioFrame unresampledAudioFrame;
+ if(STR_CASE_CMP(_codec.plname, "L16") == 0)
+ {
+ unresampledAudioFrame._frequencyInHz = _codec.plfreq;
+
+ // L16 is un-encoded data. Just pull 10 ms.
+ WebRtc_UWord32 lengthInBytes =
+ sizeof(unresampledAudioFrame._payloadData);
+ if (_fileModule.PlayoutAudioData(
+ (WebRtc_Word8*)unresampledAudioFrame._payloadData,
+ lengthInBytes) == -1)
+ {
+ // End of file reached.
+ return -1;
+ }
+ if(lengthInBytes == 0)
+ {
+ lengthInSamples = 0;
+ return 0;
+ }
+ // One sample is two bytes.
+ unresampledAudioFrame._payloadDataLengthInSamples =
+ (WebRtc_UWord16)lengthInBytes >> 1;
+
+ }else {
+ // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
+ // expects a full frame. If the frame size is larger than 10 ms,
+ // PlayoutAudioData(..) data should be called proportionally less often.
+ WebRtc_Word16 encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
+ WebRtc_UWord32 encodedLengthInBytes = 0;
+ if(++_numberOf10MsInDecoder >= _numberOf10MsPerFrame)
+ {
+ _numberOf10MsInDecoder = 0;
+ WebRtc_UWord32 bytesFromFile = sizeof(encodedBuffer);
+ if (_fileModule.PlayoutAudioData((WebRtc_Word8*)encodedBuffer,
+ bytesFromFile) == -1)
+ {
+ // End of file reached.
+ return -1;
+ }
+ encodedLengthInBytes = bytesFromFile;
+ }
+ if(_audioDecoder.Decode(unresampledAudioFrame,frequencyInHz,
+ (WebRtc_Word8*)encodedBuffer,
+ encodedLengthInBytes) == -1)
+ {
+ return -1;
+ }
+ }
+
+ int outLen = 0;
+ if(_resampler.ResetIfNeeded(unresampledAudioFrame._frequencyInHz,
+ frequencyInHz, kResamplerSynchronous))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
+ "FilePlayerImpl::Get10msAudioFromFile() unexpected codec");
+
+ // New sampling frequency. Update state.
+ outLen = frequencyInHz / 100;
+ memset(outBuffer, 0, outLen * sizeof(WebRtc_Word16));
+ return 0;
+ }
+ _resampler.Push(unresampledAudioFrame._payloadData,
+ unresampledAudioFrame._payloadDataLengthInSamples,
+ outBuffer,
+ MAX_AUDIO_BUFFER_IN_SAMPLES,
+ outLen);
+
+ lengthInSamples = outLen;
+
+ if(_scaling != 1.0)
+ {
+ for (int i = 0;i < outLen; i++)
+ {
+ outBuffer[i] = (WebRtc_Word16)(outBuffer[i] * _scaling);
+ }
+ }
+ _decodedLengthInMS += 10;
+ return 0;
+}
+
+WebRtc_Word32 FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback)
+{
+ return _fileModule.SetModuleFileCallback(callback);
+}
+
+WebRtc_Word32 FilePlayerImpl::SetAudioScaling(float scaleFactor)
+{
+ if((scaleFactor >= 0)&&(scaleFactor <= 2.0))
+ {
+ _scaling = scaleFactor;
+ return 0;
+ }
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
+ "FilePlayerImpl::SetAudioScaling() not allowed scale factor");
+ return -1;
+}
+
+WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const char* fileName,
+ bool loop,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition,
+ const CodecInst* codecInst)
+{
+ if (_fileFormat == kFileFormatPcm16kHzFile ||
+ _fileFormat == kFileFormatPcm8kHzFile||
+ _fileFormat == kFileFormatPcm32kHzFile )
+ {
+ CodecInst codecInstL16;
+ strncpy(codecInstL16.plname,"L16",32);
+ codecInstL16.pltype = 93;
+ codecInstL16.channels = 1;
+
+ if (_fileFormat == kFileFormatPcm8kHzFile)
+ {
+ codecInstL16.rate = 128000;
+ codecInstL16.plfreq = 8000;
+ codecInstL16.pacsize = 80;
+
+ } else if(_fileFormat == kFileFormatPcm16kHzFile)
+ {
+ codecInstL16.rate = 256000;
+ codecInstL16.plfreq = 16000;
+ codecInstL16.pacsize = 160;
+
+ }else if(_fileFormat == kFileFormatPcm32kHzFile)
+ {
+ codecInstL16.rate = 512000;
+ codecInstL16.plfreq = 32000;
+ codecInstL16.pacsize = 160;
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID,
+ "FilePlayerImpl::StartPlayingFile() sample frequency\
+ specifed not supported for PCM format.");
+ return -1;
+ }
+
+ if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
+ _fileFormat, &codecInstL16,
+ startPosition,
+ stopPosition) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to initialize file\
+ %s playout.", fileName);
+ return -1;
+ }
+ SetAudioScaling(volumeScaling);
+ }else if(_fileFormat == kFileFormatPreencodedFile)
+ {
+ if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
+ _fileFormat, codecInst) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingPreEncodedFile() failed to\
+ initialize pre-encoded file %s playout.",
+ fileName);
+ return -1;
+ }
+ } else
+ {
+ CodecInst* no_inst = NULL;
+ if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
+ _fileFormat, no_inst,
+ startPosition,
+ stopPosition) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to initialize file\
+ %s playout.", fileName);
+ return -1;
+ }
+ SetAudioScaling(volumeScaling);
+ }
+ if (SetUpAudioDecoder() == -1)
+ {
+ StopPlayingFile();
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition,
+ const CodecInst* codecInst)
+{
+ if (_fileFormat == kFileFormatPcm16kHzFile ||
+ _fileFormat == kFileFormatPcm32kHzFile ||
+ _fileFormat == kFileFormatPcm8kHzFile)
+ {
+ CodecInst codecInstL16;
+ strncpy(codecInstL16.plname,"L16",32);
+ codecInstL16.pltype = 93;
+ codecInstL16.channels = 1;
+
+ if (_fileFormat == kFileFormatPcm8kHzFile)
+ {
+ codecInstL16.rate = 128000;
+ codecInstL16.plfreq = 8000;
+ codecInstL16.pacsize = 80;
+
+ }else if (_fileFormat == kFileFormatPcm16kHzFile)
+ {
+ codecInstL16.rate = 256000;
+ codecInstL16.plfreq = 16000;
+ codecInstL16.pacsize = 160;
+
+ }else if (_fileFormat == kFileFormatPcm32kHzFile)
+ {
+ codecInstL16.rate = 512000;
+ codecInstL16.plfreq = 32000;
+ codecInstL16.pacsize = 160;
+ }else
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() sample frequency specifed\
+ not supported for PCM format.");
+ return -1;
+ }
+ if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
+ _fileFormat, &codecInstL16,
+ startPosition,
+ stopPosition) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to initialize stream\
+ playout.");
+ return -1;
+ }
+
+ }else if(_fileFormat == kFileFormatPreencodedFile)
+ {
+ if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
+ _fileFormat, codecInst) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to initialize stream\
+ playout.");
+ return -1;
+ }
+ } else {
+ CodecInst* no_inst = NULL;
+ if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
+ _fileFormat, no_inst,
+ startPosition,
+ stopPosition) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to initialize\
+ stream playout.");
+ return -1;
+ }
+ }
+ SetAudioScaling(volumeScaling);
+
+ if (SetUpAudioDecoder() == -1)
+ {
+ StopPlayingFile();
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 FilePlayerImpl::StopPlayingFile()
+{
+ memset(&_codec, 0, sizeof(CodecInst));
+ _numberOf10MsPerFrame = 0;
+ _numberOf10MsInDecoder = 0;
+ return _fileModule.StopPlaying();
+}
+
+bool FilePlayerImpl::IsPlayingFile() const
+{
+ return _fileModule.IsPlaying();
+}
+
+WebRtc_Word32 FilePlayerImpl::GetPlayoutPosition(WebRtc_UWord32& durationMs)
+{
+ return _fileModule.PlayoutPositionMs(durationMs);
+}
+
+WebRtc_Word32 FilePlayerImpl::SetUpAudioDecoder()
+{
+ if ((_fileModule.codec_info(_codec) == -1))
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() failed to retrieve Codec info\
+ of file data.");
+ return -1;
+ }
+ if( STR_CASE_CMP(_codec.plname, "L16") != 0 &&
+ _audioDecoder.SetDecodeCodec(_codec,AMRFileStorage) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FilePlayerImpl::StartPlayingFile() codec %s not supported",
+ _codec.plname);
+ return -1;
+ }
+ _numberOf10MsPerFrame = _codec.pacsize / (_codec.plfreq / 100);
+ _numberOf10MsInDecoder = 0;
+ return 0;
+}
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+VideoFilePlayerImpl::VideoFilePlayerImpl(WebRtc_UWord32 instanceID,
+ FileFormats fileFormat)
+ : FilePlayerImpl(instanceID,fileFormat),
+ _videoDecoder(*new VideoCoder(instanceID)),
+ video_codec_info_(),
+ _decodedVideoFrames(0),
+ _encodedData(*new EncodedVideoData()),
+ _frameScaler(*new FrameScaler()),
+ _critSec(CriticalSectionWrapper::CreateCriticalSection()),
+ _startTime(),
+ _accumulatedRenderTimeMs(0),
+ _frameLengthMS(0),
+ _numberOfFramesRead(0),
+ _videoOnly(false)
+{
+ memset(&video_codec_info_, 0, sizeof(video_codec_info_));
+}
+
+VideoFilePlayerImpl::~VideoFilePlayerImpl()
+{
+ delete _critSec;
+ delete &_frameScaler;
+ delete &_videoDecoder;
+ delete &_encodedData;
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::StartPlayingVideoFile(
+ const char* fileName,
+ bool loop,
+ bool videoOnly)
+{
+ CriticalSectionScoped lock( _critSec);
+
+ if(_fileModule.StartPlayingVideoFile(fileName, loop, videoOnly,
+ _fileFormat) != 0)
+ {
+ return -1;
+ }
+
+ _decodedVideoFrames = 0;
+ _accumulatedRenderTimeMs = 0;
+ _frameLengthMS = 0;
+ _numberOfFramesRead = 0;
+ _videoOnly = videoOnly;
+
+ // Set up video_codec_info_ according to file,
+ if(SetUpVideoDecoder() != 0)
+ {
+ StopPlayingFile();
+ return -1;
+ }
+ if(!videoOnly)
+ {
+ // Set up _codec according to file,
+ if(SetUpAudioDecoder() != 0)
+ {
+ StopPlayingFile();
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::StopPlayingFile()
+{
+ CriticalSectionScoped lock( _critSec);
+
+ _decodedVideoFrames = 0;
+ _videoDecoder.ResetDecoder();
+
+ return FilePlayerImpl::StopPlayingFile();
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(VideoFrame& videoFrame,
+ WebRtc_UWord32 outWidth,
+ WebRtc_UWord32 outHeight)
+{
+ CriticalSectionScoped lock( _critSec);
+
+ WebRtc_Word32 retVal = GetVideoFromFile(videoFrame);
+ if(retVal != 0)
+ {
+ return retVal;
+ }
+ if( videoFrame.Length() > 0)
+ {
+ retVal = _frameScaler.ResizeFrameIfNeeded(&videoFrame, outWidth,
+ outHeight);
+ }
+ return retVal;
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(VideoFrame& videoFrame)
+{
+ CriticalSectionScoped lock( _critSec);
+ // No new video data read from file.
+ if(_encodedData.payloadSize == 0)
+ {
+ videoFrame.SetLength(0);
+ return -1;
+ }
+ WebRtc_Word32 retVal = 0;
+ if(strncmp(video_codec_info_.plName, "I420", 5) == 0)
+ {
+ videoFrame.CopyFrame(_encodedData.payloadSize,_encodedData.payloadData);
+ videoFrame.SetLength(_encodedData.payloadSize);
+ videoFrame.SetWidth(video_codec_info_.width);
+ videoFrame.SetHeight(video_codec_info_.height);
+ }else
+ {
+ // Set the timestamp manually since there is no timestamp in the file.
+ // Update timestam according to 90 kHz stream.
+ _encodedData.timeStamp += (90000 / video_codec_info_.maxFramerate);
+ retVal = _videoDecoder.Decode(videoFrame, _encodedData);
+ }
+
+ WebRtc_Word64 renderTimeMs = TickTime::MillisecondTimestamp();
+ videoFrame.SetRenderTime(renderTimeMs);
+
+ // Indicate that the current frame in the encoded buffer is old/has
+ // already been read.
+ _encodedData.payloadSize = 0;
+ if( retVal == 0)
+ {
+ _decodedVideoFrames++;
+ }
+ return retVal;
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::video_codec_info(
+ VideoCodec& videoCodec) const
+{
+ if(video_codec_info_.plName[0] == 0)
+ {
+ return -1;
+ }
+ memcpy(&videoCodec, &video_codec_info_, sizeof(VideoCodec));
+ return 0;
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame()
+{
+ if(_fileFormat != kFileFormatAviFile)
+ {
+ return -1;
+ }
+ if(!_fileModule.IsPlaying())
+ {
+ return -1;
+ }
+ if(_encodedData.payloadSize <= 0)
+ {
+ // Read next frame from file.
+ CriticalSectionScoped lock( _critSec);
+
+ if(_fileFormat == kFileFormatAviFile)
+ {
+ // Get next video frame
+ WebRtc_UWord32 encodedBufferLengthInBytes = _encodedData.bufferSize;
+ if(_fileModule.PlayoutAVIVideoData(
+ reinterpret_cast< WebRtc_Word8*>(_encodedData.payloadData),
+ encodedBufferLengthInBytes) != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ _instanceID,
+ "FilePlayerImpl::TimeUntilNextVideoFrame() error reading\
+ video data");
+ return -1;
+ }
+ _encodedData.payloadSize = encodedBufferLengthInBytes;
+ _encodedData.codec = video_codec_info_.codecType;
+ _numberOfFramesRead++;
+
+ if(_accumulatedRenderTimeMs == 0)
+ {
+ _startTime = TickTime::Now();
+ // This if-statement should only trigger once.
+ _accumulatedRenderTimeMs = 1;
+ } else {
+ // A full seconds worth of frames have been read.
+ if(_numberOfFramesRead % video_codec_info_.maxFramerate == 0)
+ {
+ // Frame rate is in frames per seconds. Frame length is
+ // calculated as an integer division which means it may
+ // be rounded down. Compensate for this every second.
+ WebRtc_UWord32 rest = 1000%_frameLengthMS;
+ _accumulatedRenderTimeMs += rest;
+ }
+ _accumulatedRenderTimeMs += _frameLengthMS;
+ }
+ }
+ }
+
+ WebRtc_Word64 timeToNextFrame;
+ if(_videoOnly)
+ {
+ timeToNextFrame = _accumulatedRenderTimeMs -
+ (TickTime::Now() - _startTime).Milliseconds();
+
+ } else {
+ // Synchronize with the audio stream instead of system clock.
+ timeToNextFrame = _accumulatedRenderTimeMs - _decodedLengthInMS;
+ }
+ if(timeToNextFrame < 0)
+ {
+ return 0;
+
+ } else if(timeToNextFrame > 0x0fffffff)
+ {
+ // Wraparound or audio stream has gone to far ahead of the video stream.
+ return -1;
+ }
+ return static_cast<WebRtc_Word32>(timeToNextFrame);
+}
+
+WebRtc_Word32 VideoFilePlayerImpl::SetUpVideoDecoder()
+{
+ if (_fileModule.VideoCodecInst(video_codec_info_) != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ _instanceID,
+ "FilePlayerImpl::SetVideoDecoder() failed to retrieve Codec info of\
+ file data.");
+ return -1;
+ }
+
+ WebRtc_Word32 useNumberOfCores = 1;
+ if(_videoDecoder.SetDecodeCodec(video_codec_info_, useNumberOfCores) != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ _instanceID,
+ "FilePlayerImpl::SetUpVideoDecoder() codec %s not supported",
+ video_codec_info_.plName);
+ return -1;
+ }
+
+ _frameLengthMS = 1000/video_codec_info_.maxFramerate;
+
+ // Size of unencoded data (I420) should be the largest possible frame size
+ // in a file.
+ const WebRtc_UWord32 KReadBufferSize = 3 * video_codec_info_.width *
+ video_codec_info_.height / 2;
+ _encodedData.VerifyAndAllocate(KReadBufferSize);
+ _encodedData.encodedHeight = video_codec_info_.height;
+ _encodedData.encodedWidth = video_codec_info_.width;
+ _encodedData.payloadType = video_codec_info_.plType;
+ _encodedData.timeStamp = 0;
+ return 0;
+}
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/file_player_impl.h b/trunk/src/modules/utility/source/file_player_impl.h
new file mode 100644
index 0000000..e15cb09
--- /dev/null
+++ b/trunk/src/modules/utility/source/file_player_impl.h
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
+
+#include "coder.h"
+#include "common_types.h"
+#include "critical_section_wrapper.h"
+#include "engine_configurations.h"
+#include "file_player.h"
+#include "media_file_defines.h"
+#include "media_file.h"
+#include "resampler.h"
+#include "tick_util.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class VideoCoder;
+class FrameScaler;
+
+class FilePlayerImpl : public FilePlayer
+{
+public:
+ FilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+ ~FilePlayerImpl();
+
+ // FilePlayer functions.
+ virtual WebRtc_Word32 Get10msAudioFromFile(
+ WebRtc_Word16* decodedDataBuffer,
+ WebRtc_UWord32& decodedDataLengthInSamples,
+ const WebRtc_UWord32 frequencyInHz);
+ virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
+ virtual WebRtc_Word32 StartPlayingFile(
+ const char* fileName,
+ bool loop,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL);
+ virtual WebRtc_Word32 StartPlayingFile(
+ InStream& sourceStream,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL);
+ virtual WebRtc_Word32 StopPlayingFile();
+ virtual bool IsPlayingFile() const;
+ virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs);
+ virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const;
+ virtual WebRtc_Word32 Frequency() const;
+ virtual WebRtc_Word32 SetAudioScaling(float scaleFactor);
+
+protected:
+ WebRtc_Word32 SetUpAudioDecoder();
+
+ WebRtc_UWord32 _instanceID;
+ const FileFormats _fileFormat;
+ MediaFile& _fileModule;
+
+ WebRtc_UWord32 _decodedLengthInMS;
+
+private:
+ WebRtc_Word16 _decodedAudioBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
+ AudioCoder _audioDecoder;
+
+ CodecInst _codec;
+ WebRtc_Word32 _numberOf10MsPerFrame;
+ WebRtc_Word32 _numberOf10MsInDecoder;
+
+ Resampler _resampler;
+ float _scaling;
+};
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+class VideoFilePlayerImpl: public FilePlayerImpl
+{
+public:
+ VideoFilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+ ~VideoFilePlayerImpl();
+
+ // FilePlayer functions.
+ virtual WebRtc_Word32 TimeUntilNextVideoFrame();
+ virtual WebRtc_Word32 StartPlayingVideoFile(const char* fileName,
+ bool loop,
+ bool videoOnly);
+ virtual WebRtc_Word32 StopPlayingFile();
+ virtual WebRtc_Word32 video_codec_info(VideoCodec& videoCodec) const;
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& videoFrame);
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& videoFrame,
+ const WebRtc_UWord32 outWidth,
+ const WebRtc_UWord32 outHeight);
+
+private:
+ WebRtc_Word32 SetUpVideoDecoder();
+
+ VideoCoder& _videoDecoder;
+ VideoCodec video_codec_info_;
+ WebRtc_Word32 _decodedVideoFrames;
+
+ EncodedVideoData& _encodedData;
+
+ FrameScaler& _frameScaler;
+ CriticalSectionWrapper* _critSec;
+ TickTime _startTime;
+ WebRtc_Word64 _accumulatedRenderTimeMs;
+ WebRtc_UWord32 _frameLengthMS;
+
+ WebRtc_Word32 _numberOfFramesRead;
+ bool _videoOnly;
+};
+#endif //WEBRTC_MODULE_UTILITY_VIDEO
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
diff --git a/trunk/src/modules/utility/source/file_player_unittest.cc b/trunk/src/modules/utility/source/file_player_unittest.cc
new file mode 100644
index 0000000..2e76905
--- /dev/null
+++ b/trunk/src/modules/utility/source/file_player_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "file_player.h"
+#include "gtest/gtest.h"
+
+TEST(FilePlayerTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/utility/source/file_recorder_impl.cc b/trunk/src/modules/utility/source/file_recorder_impl.cc
new file mode 100644
index 0000000..07d16aa
--- /dev/null
+++ b/trunk/src/modules/utility/source/file_recorder_impl.cc
@@ -0,0 +1,794 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#include "file_recorder_impl.h"
+#include "media_file.h"
+#include "trace.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ #include "cpu_wrapper.h"
+ #include "critical_section_wrapper.h"
+ #include "frame_scaler.h"
+ #include "video_coder.h"
+ #include "video_frames_queue.h"
+#endif
+
+// OS independent case insensitive string comparison.
+#ifdef WIN32
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+ #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+namespace webrtc {
+FileRecorder* FileRecorder::CreateFileRecorder(WebRtc_UWord32 instanceID,
+ FileFormats fileFormat)
+{
+ switch(fileFormat)
+ {
+ case kFileFormatWavFile:
+ case kFileFormatCompressedFile:
+ case kFileFormatPreencodedFile:
+ case kFileFormatPcm16kHzFile:
+ case kFileFormatPcm8kHzFile:
+ case kFileFormatPcm32kHzFile:
+ return new FileRecorderImpl(instanceID, fileFormat);
+ case kFileFormatAviFile:
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ return new AviRecorder(instanceID, fileFormat);
+#else
+ WEBRTC_TRACE(kTraceError, kTraceFile, -1,
+ "Invalid file format: %d", kFileFormatAviFile);
+ assert(false);
+ return NULL;
+#endif
+ }
+ assert(false);
+ return NULL;
+}
+
+void FileRecorder::DestroyFileRecorder(FileRecorder* recorder)
+{
+ delete recorder;
+}
+
+FileRecorderImpl::FileRecorderImpl(WebRtc_UWord32 instanceID,
+ FileFormats fileFormat)
+ : _instanceID(instanceID),
+ _fileFormat(fileFormat),
+ _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
+ _stream(NULL),
+ codec_info_(),
+ _amrFormat(AMRFileStorage),
+ _audioBuffer(),
+ _audioEncoder(instanceID),
+ _audioResampler()
+{
+}
+
+FileRecorderImpl::~FileRecorderImpl()
+{
+ MediaFile::DestroyMediaFile(_moduleFile);
+}
+
+FileFormats FileRecorderImpl::RecordingFileFormat() const
+{
+ return _fileFormat;
+}
+
+WebRtc_Word32 FileRecorderImpl::RegisterModuleFileCallback(
+ FileCallback* callback)
+{
+ if(_moduleFile == NULL)
+ {
+ return -1;
+ }
+ return _moduleFile->SetModuleFileCallback(callback);
+}
+
+WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile(
+ const char* fileName,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notificationTimeMs,
+ ACMAMRPackingFormat amrFormat)
+{
+ if(_moduleFile == NULL)
+ {
+ return -1;
+ }
+ codec_info_ = codecInst;
+ _amrFormat = amrFormat;
+
+ WebRtc_Word32 retVal = 0;
+ if(_fileFormat != kFileFormatAviFile)
+ {
+ // AVI files should be started using StartRecordingVideoFile(..) all
+ // other formats should use this API.
+ retVal =_moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
+ codecInst,
+ notificationTimeMs);
+ }
+
+ if( retVal == 0)
+ {
+ retVal = SetUpAudioEncoder();
+ }
+ if( retVal != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FileRecorder::StartRecording() failed to initialize file %s for\
+ recording.",
+ fileName);
+
+ if(IsRecording())
+ {
+ StopRecording();
+ }
+ }
+ return retVal;
+}
+
+WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile(
+ OutStream& destStream,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notificationTimeMs,
+ ACMAMRPackingFormat amrFormat)
+{
+ codec_info_ = codecInst;
+ _amrFormat = amrFormat;
+
+ WebRtc_Word32 retVal = _moduleFile->StartRecordingAudioStream(
+ destStream,
+ _fileFormat,
+ codecInst,
+ notificationTimeMs);
+
+ if( retVal == 0)
+ {
+ retVal = SetUpAudioEncoder();
+ }
+ if( retVal != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FileRecorder::StartRecording() failed to initialize outStream for\
+ recording.");
+
+ if(IsRecording())
+ {
+ StopRecording();
+ }
+ }
+ return retVal;
+}
+
+WebRtc_Word32 FileRecorderImpl::StopRecording()
+{
+ memset(&codec_info_, 0, sizeof(CodecInst));
+ return _moduleFile->StopRecording();
+}
+
+bool FileRecorderImpl::IsRecording() const
+{
+ return _moduleFile->IsRecording();
+}
+
+WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
+ const AudioFrame& incomingAudioFrame,
+ const TickTime* playoutTS)
+{
+ if (codec_info_.plfreq == 0)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FileRecorder::RecordAudioToFile() recording audio is not turned\
+ on");
+ return -1;
+ }
+ AudioFrame tempAudioFrame;
+ tempAudioFrame._payloadDataLengthInSamples = 0;
+ if( incomingAudioFrame._audioChannel == 2 &&
+ !_moduleFile->IsStereo())
+ {
+ // Recording mono but incoming audio is (interleaved) stereo.
+ tempAudioFrame._audioChannel = 1;
+ tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
+ tempAudioFrame._payloadDataLengthInSamples =
+ incomingAudioFrame._payloadDataLengthInSamples;
+ for (WebRtc_UWord16 i = 0;
+ i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
+ {
+ // Sample value is the average of left and right buffer rounded to
+ // closest integer value. Note samples can be either 1 or 2 byte.
+ tempAudioFrame._payloadData[i] =
+ ((incomingAudioFrame._payloadData[2 * i] +
+ incomingAudioFrame._payloadData[(2 * i) + 1] + 1) >> 1);
+ }
+ }
+ else if( incomingAudioFrame._audioChannel == 1 &&
+ _moduleFile->IsStereo())
+ {
+ // Recording stereo but incoming audio is mono.
+ tempAudioFrame._audioChannel = 2;
+ tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
+ tempAudioFrame._payloadDataLengthInSamples =
+ incomingAudioFrame._payloadDataLengthInSamples;
+ for (WebRtc_UWord16 i = 0;
+ i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
+ {
+ // Duplicate sample to both channels
+ tempAudioFrame._payloadData[2*i] =
+ incomingAudioFrame._payloadData[i];
+ tempAudioFrame._payloadData[2*i+1] =
+ incomingAudioFrame._payloadData[i];
+ }
+ }
+
+ const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
+ if(tempAudioFrame._payloadDataLengthInSamples != 0)
+ {
+ // If ptrAudioFrame is not empty it contains the audio to be recorded.
+ ptrAudioFrame = &tempAudioFrame;
+ }
+
+ // Encode the audio data before writing to file. Don't encode if the codec
+ // is PCM.
+ // NOTE: stereo recording is only supported for WAV files.
+ // TODO (hellner): WAV expect PCM in little endian byte order. Not
+ // "encoding" with PCM coder should be a problem for big endian systems.
+ WebRtc_UWord32 encodedLenInBytes = 0;
+ if (_fileFormat == kFileFormatPreencodedFile ||
+ STR_CASE_CMP(codec_info_.plname, "L16") != 0)
+ {
+ if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
+ encodedLenInBytes) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ _instanceID,
+ "FileRecorder::RecordAudioToFile() codec %s not supported or\
+ failed to encode stream",
+ codec_info_.plname);
+ return -1;
+ }
+ } else {
+ int outLen = 0;
+ if(ptrAudioFrame->_audioChannel == 2)
+ {
+ // ptrAudioFrame contains interleaved stereo audio.
+ _audioResampler.ResetIfNeeded(ptrAudioFrame->_frequencyInHz,
+ codec_info_.plfreq,
+ kResamplerSynchronousStereo);
+ _audioResampler.Push(ptrAudioFrame->_payloadData,
+ ptrAudioFrame->_payloadDataLengthInSamples *
+ ptrAudioFrame->_audioChannel,
+ (WebRtc_Word16*)_audioBuffer,
+ MAX_AUDIO_BUFFER_IN_BYTES, outLen);
+ } else {
+ _audioResampler.ResetIfNeeded(ptrAudioFrame->_frequencyInHz,
+ codec_info_.plfreq,
+ kResamplerSynchronous);
+ _audioResampler.Push(ptrAudioFrame->_payloadData,
+ ptrAudioFrame->_payloadDataLengthInSamples,
+ (WebRtc_Word16*)_audioBuffer,
+ MAX_AUDIO_BUFFER_IN_BYTES, outLen);
+ }
+ encodedLenInBytes = outLen * sizeof(WebRtc_Word16);
+ }
+
+ // Codec may not be operating at a frame rate of 10 ms. Whenever enough
+ // 10 ms chunks of data has been pushed to the encoder an encoded frame
+ // will be available. Wait until then.
+ if (encodedLenInBytes)
+ {
+ WebRtc_UWord16 msOfData =
+ ptrAudioFrame->_payloadDataLengthInSamples /
+ WebRtc_UWord16(ptrAudioFrame->_frequencyInHz / 1000);
+ if (WriteEncodedAudioData(_audioBuffer,
+ (WebRtc_UWord16)encodedLenInBytes,
+ msOfData, playoutTS) == -1)
+ {
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 FileRecorderImpl::SetUpAudioEncoder()
+{
+ if (_fileFormat == kFileFormatPreencodedFile ||
+ STR_CASE_CMP(codec_info_.plname, "L16") != 0)
+ {
+ if(_audioEncoder.SetEncodeCodec(codec_info_,_amrFormat) == -1)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVoice,
+ _instanceID,
+ "FileRecorder::StartRecording() codec %s not supported",
+ codec_info_.plname);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 FileRecorderImpl::codec_info(CodecInst& codecInst) const
+{
+ if(codec_info_.plfreq == 0)
+ {
+ return -1;
+ }
+ codecInst = codec_info_;
+ return 0;
+}
+
+WebRtc_Word32 FileRecorderImpl::WriteEncodedAudioData(
+ const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord16 bufferLength,
+ WebRtc_UWord16 /*millisecondsOfData*/,
+ const TickTime* /*playoutTS*/)
+{
+ return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
+}
+
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+class AudioFrameFileInfo
+{
+ public:
+ AudioFrameFileInfo(const WebRtc_Word8* audioData,
+ const WebRtc_UWord16 audioSize,
+ const WebRtc_UWord16 audioMS,
+ const TickTime& playoutTS)
+ : _audioData(), _audioSize(audioSize), _audioMS(audioMS),
+ _playoutTS(playoutTS)
+ {
+ if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES)
+ {
+ assert(false);
+ _audioSize = 0;
+ return;
+ }
+ memcpy(_audioData, audioData, audioSize);
+ };
+ // TODO (hellner): either turn into a struct or provide get/set functions.
+ WebRtc_Word8 _audioData[MAX_AUDIO_BUFFER_IN_BYTES];
+ WebRtc_UWord16 _audioSize;
+ WebRtc_UWord16 _audioMS;
+ TickTime _playoutTS;
+};
+
+AviRecorder::AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat)
+ : FileRecorderImpl(instanceID, fileFormat),
+ _videoOnly(false),
+ _thread( 0),
+ _timeEvent(*EventWrapper::Create()),
+ _critSec(CriticalSectionWrapper::CreateCriticalSection()),
+ _writtenVideoFramesCounter(0),
+ _writtenAudioMS(0),
+ _writtenVideoMS(0)
+{
+ _videoEncoder = new VideoCoder(instanceID);
+ _frameScaler = new FrameScaler();
+ _videoFramesQueue = new VideoFramesQueue();
+ _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
+ "AviRecorder()");
+}
+
+AviRecorder::~AviRecorder( )
+{
+ StopRecording( );
+
+ delete _videoEncoder;
+ delete _frameScaler;
+ delete _videoFramesQueue;
+ delete _thread;
+ delete &_timeEvent;
+ delete _critSec;
+}
+
+WebRtc_Word32 AviRecorder::StartRecordingVideoFile(
+ const char* fileName,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ ACMAMRPackingFormat amrFormat,
+ bool videoOnly)
+{
+ _firstAudioFrameReceived = false;
+ _videoCodecInst = videoCodecInst;
+ _videoOnly = videoOnly;
+
+ if(_moduleFile->StartRecordingVideoFile(fileName, _fileFormat,
+ audioCodecInst, videoCodecInst,
+ videoOnly) != 0)
+ {
+ return -1;
+ }
+
+ if(!videoOnly)
+ {
+ if(FileRecorderImpl::StartRecordingAudioFile(fileName,audioCodecInst, 0,
+ amrFormat) !=0)
+ {
+ StopRecording();
+ return -1;
+ }
+ }
+ if( SetUpVideoEncoder() != 0)
+ {
+ StopRecording();
+ return -1;
+ }
+ if(_videoOnly)
+ {
+ // Writing to AVI file is non-blocking.
+ // Start non-blocking timer if video only. If recording both video and
+ // audio let the pushing of audio frames be the timer.
+ _timeEvent.StartTimer(true, 1000 / _videoCodecInst.maxFramerate);
+ }
+ StartThread();
+ return 0;
+}
+
+WebRtc_Word32 AviRecorder::StopRecording()
+{
+ _timeEvent.StopTimer();
+
+ StopThread();
+ return FileRecorderImpl::StopRecording();
+}
+
+WebRtc_Word32 AviRecorder::CalcI420FrameSize( ) const
+{
+ return 3 * _videoCodecInst.width * _videoCodecInst.height / 2;
+}
+
+WebRtc_Word32 AviRecorder::SetUpVideoEncoder()
+{
+ // Size of unencoded data (I420) should be the largest possible frame size
+ // in a file.
+ _videoMaxPayloadSize = CalcI420FrameSize();
+ _videoEncodedData.VerifyAndAllocate(_videoMaxPayloadSize);
+
+ _videoCodecInst.plType = _videoEncoder->DefaultPayloadType(
+ _videoCodecInst.plName);
+
+ WebRtc_Word32 useNumberOfCores = 1;
+ // Set the max payload size to 16000. This means that the codec will try to
+ // create slices that will fit in 16000 kByte packets. However, the
+ // Encode() call will still generate one full frame.
+ if(_videoEncoder->SetEncodeCodec(_videoCodecInst, useNumberOfCores,
+ 16000))
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AviRecorder::RecordVideoToFile(const VideoFrame& videoFrame)
+{
+ CriticalSectionScoped lock(_critSec);
+
+ if(!IsRecording() || ( videoFrame.Length() == 0))
+ {
+ return -1;
+ }
+ // The frame is written to file in AviRecorder::Process().
+ WebRtc_Word32 retVal = _videoFramesQueue->AddFrame(videoFrame);
+ if(retVal != 0)
+ {
+ StopRecording();
+ }
+ return retVal;
+}
+
+bool AviRecorder::StartThread()
+{
+ unsigned int id;
+ if( _thread == 0)
+ {
+ return false;
+ }
+
+ return _thread->Start(id);
+}
+
+bool AviRecorder::StopThread()
+{
+ _critSec->Enter();
+
+ if(_thread)
+ {
+ _thread->SetNotAlive();
+
+ ThreadWrapper* thread = _thread;
+ _thread = NULL;
+
+ _timeEvent.Set();
+
+ _critSec->Leave();
+
+ if(thread->Stop())
+ {
+ delete thread;
+ } else {
+ return false;
+ }
+ } else {
+ _critSec->Leave();
+ }
+ return true;
+}
+
+bool AviRecorder::Run( ThreadObj threadObj)
+{
+ return static_cast<AviRecorder*>( threadObj)->Process();
+}
+
+WebRtc_Word32 AviRecorder::ProcessAudio()
+{
+ if (_writtenVideoFramesCounter == 0)
+ {
+ // Get the most recent frame that is due for writing to file. Since
+ // frames are unencoded it's safe to throw away frames if necessary
+ // for synchronizing audio and video.
+ VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord();
+ if(frameToProcess)
+ {
+ // Syncronize audio to the current frame to process by throwing away
+ // audio samples with older timestamp than the video frame.
+ WebRtc_UWord32 numberOfAudioElements =
+ _audioFramesToWrite.GetSize();
+ for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i)
+ {
+ AudioFrameFileInfo* frameInfo =
+ (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem();
+ if(frameInfo)
+ {
+ if(TickTime::TicksToMilliseconds(
+ frameInfo->_playoutTS.Ticks()) <
+ frameToProcess->RenderTimeMs())
+ {
+ delete frameInfo;
+ _audioFramesToWrite.PopFront();
+ } else
+ {
+ break;
+ }
+ }
+ }
+ }
+ }
+ // Write all audio up to current timestamp.
+ WebRtc_Word32 error = 0;
+ WebRtc_UWord32 numberOfAudioElements = _audioFramesToWrite.GetSize();
+ for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i)
+ {
+ AudioFrameFileInfo* frameInfo =
+ (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem();
+ if(frameInfo)
+ {
+ if((TickTime::Now() - frameInfo->_playoutTS).Milliseconds() > 0)
+ {
+ _moduleFile->IncomingAudioData(frameInfo->_audioData,
+ frameInfo->_audioSize);
+ _writtenAudioMS += frameInfo->_audioMS;
+ delete frameInfo;
+ _audioFramesToWrite.PopFront();
+ } else {
+ break;
+ }
+ } else {
+ _audioFramesToWrite.PopFront();
+ }
+ }
+ return error;
+}
+
+bool AviRecorder::Process()
+{
+ switch(_timeEvent.Wait(500))
+ {
+ case kEventSignaled:
+ if(_thread == NULL)
+ {
+ return false;
+ }
+ break;
+ case kEventError:
+ return false;
+ case kEventTimeout:
+ // No events triggered. No work to do.
+ return true;
+ }
+ CriticalSectionScoped lock( _critSec);
+
+ // Get the most recent frame to write to file (if any). Synchronize it with
+ // the audio stream (if any). Synchronization the video based on its render
+ // timestamp (i.e. VideoFrame::RenderTimeMS())
+ VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord();
+ if( frameToProcess == NULL)
+ {
+ return true;
+ }
+ WebRtc_Word32 error = 0;
+ if(!_videoOnly)
+ {
+ if(!_firstAudioFrameReceived)
+ {
+ // Video and audio can only be synchronized if both have been
+ // received.
+ return true;
+ }
+ error = ProcessAudio();
+
+ while (_writtenAudioMS > _writtenVideoMS)
+ {
+ error = EncodeAndWriteVideoToFile( *frameToProcess);
+ if( error != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
+ "AviRecorder::Process() error writing to file.");
+ break;
+ } else {
+ WebRtc_UWord32 frameLengthMS = 1000 /
+ _videoCodecInst.maxFramerate;
+ _writtenVideoFramesCounter++;
+ _writtenVideoMS += frameLengthMS;
+ // A full seconds worth of frames have been written.
+ if(_writtenVideoFramesCounter%_videoCodecInst.maxFramerate == 0)
+ {
+ // Frame rate is in frames per seconds. Frame length is
+ // calculated as an integer division which means it may
+ // be rounded down. Compensate for this every second.
+ WebRtc_UWord32 rest = 1000 % frameLengthMS;
+ _writtenVideoMS += rest;
+ }
+ }
+ }
+ } else {
+ // Frame rate is in frames per seconds. Frame length is calculated as an
+ // integer division which means it may be rounded down. This introduces
+ // drift. Once a full frame worth of drift has happened, skip writing
+ // one frame. Note that frame rate is in frames per second so the
+ // drift is completely compensated for.
+ WebRtc_UWord32 frameLengthMS = 1000/_videoCodecInst.maxFramerate;
+ WebRtc_UWord32 restMS = 1000 % frameLengthMS;
+ WebRtc_UWord32 frameSkip = (_videoCodecInst.maxFramerate *
+ frameLengthMS) / restMS;
+
+ _writtenVideoFramesCounter++;
+ if(_writtenVideoFramesCounter % frameSkip == 0)
+ {
+ _writtenVideoMS += frameLengthMS;
+ return true;
+ }
+
+ error = EncodeAndWriteVideoToFile( *frameToProcess);
+ if(error != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
+ "AviRecorder::Process() error writing to file.");
+ } else {
+ _writtenVideoMS += frameLengthMS;
+ }
+ }
+ return error == 0;
+}
+
+WebRtc_Word32 AviRecorder::EncodeAndWriteVideoToFile(VideoFrame& videoFrame)
+{
+ if(!IsRecording() || (videoFrame.Length() == 0))
+ {
+ return -1;
+ }
+
+ if(_frameScaler->ResizeFrameIfNeeded(&videoFrame, _videoCodecInst.width,
+ _videoCodecInst.height) != 0)
+ {
+ return -1;
+ }
+
+ _videoEncodedData.payloadSize = 0;
+
+ if( STR_CASE_CMP(_videoCodecInst.plName, "I420") == 0)
+ {
+ _videoEncodedData.VerifyAndAllocate(videoFrame.Length());
+
+ // I420 is raw data. No encoding needed (each sample is represented by
+ // 1 byte so there is no difference depending on endianness).
+ memcpy(_videoEncodedData.payloadData, videoFrame.Buffer(),
+ videoFrame.Length());
+
+ _videoEncodedData.payloadSize = videoFrame.Length();
+ _videoEncodedData.frameType = kVideoFrameKey;
+ }else {
+ if( _videoEncoder->Encode(videoFrame, _videoEncodedData) != 0)
+ {
+ return -1;
+ }
+ }
+
+ if(_videoEncodedData.payloadSize > 0)
+ {
+ if(_moduleFile->IncomingAVIVideoData(
+ (WebRtc_Word8*)(_videoEncodedData.payloadData),
+ _videoEncodedData.payloadSize))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
+ "Error writing AVI file");
+ return -1;
+ }
+ } else {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ _instanceID,
+ "FileRecorder::RecordVideoToFile() frame dropped by encoder bitrate\
+ likely to low.");
+ }
+ return 0;
+}
+
+// Store audio frame in the _audioFramesToWrite buffer. The writing to file
+// happens in AviRecorder::Process().
+WebRtc_Word32 AviRecorder::WriteEncodedAudioData(
+ const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord16 bufferLength,
+ WebRtc_UWord16 millisecondsOfData,
+ const TickTime* playoutTS)
+{
+ if (!IsRecording())
+ {
+ return -1;
+ }
+ if (bufferLength > MAX_AUDIO_BUFFER_IN_BYTES)
+ {
+ return -1;
+ }
+ if (_videoOnly)
+ {
+ return -1;
+ }
+ if (_audioFramesToWrite.GetSize() > kMaxAudioBufferQueueLength)
+ {
+ StopRecording();
+ return -1;
+ }
+ _firstAudioFrameReceived = true;
+
+ if(playoutTS)
+ {
+ _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer,
+ bufferLength,
+ millisecondsOfData,
+ *playoutTS));
+ } else {
+ _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer,
+ bufferLength,
+ millisecondsOfData,
+ TickTime::Now()));
+ }
+ _timeEvent.Set();
+ return 0;
+}
+
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/file_recorder_impl.h b/trunk/src/modules/utility/source/file_recorder_impl.h
new file mode 100644
index 0000000..3684166
--- /dev/null
+++ b/trunk/src/modules/utility/source/file_recorder_impl.h
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains a class that can write audio and/or video to file in
+// multiple file formats. The unencoded input data is written to file in the
+// encoded format specified.
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
+
+#include "coder.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "event_wrapper.h"
+#include "file_recorder.h"
+#include "media_file_defines.h"
+#include "media_file.h"
+#include "module_common_types.h"
+#include "resampler.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "typedefs.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+ #include "frame_scaler.h"
+ #include "video_coder.h"
+ #include "video_frames_queue.h"
+#endif
+
+namespace webrtc {
+// The largest decoded frame size in samples (60ms with 32kHz sample rate).
+enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
+enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+enum { kMaxAudioBufferQueueLength = 100 };
+
+class FileRecorderImpl : public FileRecorder
+{
+public:
+ FileRecorderImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+ virtual ~FileRecorderImpl();
+
+ // FileRecorder functions.
+ virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
+ virtual FileFormats RecordingFileFormat() const;
+ virtual WebRtc_Word32 StartRecordingAudioFile(
+ const char* fileName,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notificationTimeMs,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage);
+ virtual WebRtc_Word32 StartRecordingAudioFile(
+ OutStream& destStream,
+ const CodecInst& codecInst,
+ WebRtc_UWord32 notificationTimeMs,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage);
+ virtual WebRtc_Word32 StopRecording();
+ virtual bool IsRecording() const;
+ virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const;
+ virtual WebRtc_Word32 RecordAudioToFile(
+ const AudioFrame& frame,
+ const TickTime* playoutTS = NULL);
+ virtual WebRtc_Word32 StartRecordingVideoFile(
+ const char* fileName,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage,
+ bool videoOnly = false)
+ {
+ return -1;
+ }
+ virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame)
+ {
+ return -1;
+ }
+
+protected:
+ virtual WebRtc_Word32 WriteEncodedAudioData(
+ const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord16 bufferLength,
+ WebRtc_UWord16 millisecondsOfData,
+ const TickTime* playoutTS);
+
+ WebRtc_Word32 SetUpAudioEncoder();
+
+ WebRtc_UWord32 _instanceID;
+ FileFormats _fileFormat;
+ MediaFile* _moduleFile;
+
+private:
+ OutStream* _stream;
+ CodecInst codec_info_;
+ ACMAMRPackingFormat _amrFormat;
+
+ WebRtc_Word8 _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
+ AudioCoder _audioEncoder;
+ Resampler _audioResampler;
+};
+
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+class AviRecorder : public FileRecorderImpl
+{
+public:
+ AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+ virtual ~AviRecorder();
+
+ // FileRecorder functions.
+ virtual WebRtc_Word32 StartRecordingVideoFile(
+ const char* fileName,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ ACMAMRPackingFormat amrFormat = AMRFileStorage,
+ bool videoOnly = false);
+ virtual WebRtc_Word32 StopRecording();
+ virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame);
+
+protected:
+ virtual WebRtc_Word32 WriteEncodedAudioData(
+ const WebRtc_Word8* audioBuffer,
+ WebRtc_UWord16 bufferLength,
+ WebRtc_UWord16 millisecondsOfData,
+ const TickTime* playoutTS);
+private:
+ static bool Run(ThreadObj threadObj);
+ bool Process();
+
+ bool StartThread();
+ bool StopThread();
+
+ WebRtc_Word32 EncodeAndWriteVideoToFile(VideoFrame& videoFrame);
+ WebRtc_Word32 ProcessAudio();
+
+ WebRtc_Word32 CalcI420FrameSize() const;
+ WebRtc_Word32 SetUpVideoEncoder();
+
+ VideoCodec _videoCodecInst;
+ bool _videoOnly;
+
+ ListWrapper _audioFramesToWrite;
+ bool _firstAudioFrameReceived;
+
+ VideoFramesQueue* _videoFramesQueue;
+
+ FrameScaler* _frameScaler;
+ VideoCoder* _videoEncoder;
+ WebRtc_Word32 _videoMaxPayloadSize;
+ EncodedVideoData _videoEncodedData;
+
+ ThreadWrapper* _thread;
+ EventWrapper& _timeEvent;
+ CriticalSectionWrapper* _critSec;
+ WebRtc_Word64 _writtenVideoFramesCounter;
+ WebRtc_Word64 _writtenAudioMS;
+ WebRtc_Word64 _writtenVideoMS;
+};
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
diff --git a/trunk/src/modules/utility/source/frame_scaler.cc b/trunk/src/modules/utility/source/frame_scaler.cc
new file mode 100644
index 0000000..c012e89
--- /dev/null
+++ b/trunk/src/modules/utility/source/frame_scaler.cc
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/utility/source/frame_scaler.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include "common_video/libyuv/include/scaler.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+FrameScaler::FrameScaler()
+ : scaler_(new Scaler()),
+ scaled_frame_() {}
+
+FrameScaler::~FrameScaler() {}
+
+int FrameScaler::ResizeFrameIfNeeded(VideoFrame* video_frame,
+ WebRtc_UWord32 out_width,
+ WebRtc_UWord32 out_height) {
+ if (video_frame->Length() == 0) {
+ return -1;
+ }
+
+ if ((video_frame->Width() != out_width) ||
+ (video_frame->Height() != out_height)) {
+ // Set correct scale settings and scale |video_frame| into |scaled_frame_|.
+ scaler_->Set(video_frame->Width(), video_frame->Height(), out_width,
+ out_height, kI420, kI420, kScaleBox);
+ int out_length = CalcBufferSize(kI420, out_width, out_height);
+ scaled_frame_.VerifyAndAllocate(out_length);
+ int ret = scaler_->Scale(video_frame->Buffer(), scaled_frame_.Buffer(),
+ out_length);
+ if (ret < 0) {
+ return ret;
+ }
+
+ scaled_frame_.SetWidth(out_width);
+ scaled_frame_.SetHeight(out_height);
+ scaled_frame_.SetLength(out_length);
+ scaled_frame_.SetRenderTime(video_frame->RenderTimeMs());
+ scaled_frame_.SetTimeStamp(video_frame->TimeStamp());
+ video_frame->SwapFrame(scaled_frame_);
+ }
+ return 0;
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/frame_scaler.h b/trunk/src/modules/utility/source/frame_scaler.h
new file mode 100644
index 0000000..f86a933
--- /dev/null
+++ b/trunk/src/modules/utility/source/frame_scaler.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file implements a class that can be used for scaling frames.
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include "engine_configurations.h"
+#include "modules/interface/module_common_types.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class Scaler;
+class VideoFrame;
+
+class FrameScaler {
+ public:
+ FrameScaler();
+ ~FrameScaler();
+
+ // Re-sizes |video_frame| so that it has the width |out_width| and height
+ // |out_height|.
+ int ResizeFrameIfNeeded(VideoFrame* video_frame,
+ WebRtc_UWord32 out_width,
+ WebRtc_UWord32 out_height);
+
+ private:
+ scoped_ptr<Scaler> scaler_;
+ VideoFrame scaled_frame_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
diff --git a/trunk/src/modules/utility/source/process_thread_impl.cc b/trunk/src/modules/utility/source/process_thread_impl.cc
new file mode 100644
index 0000000..9028c44
--- /dev/null
+++ b/trunk/src/modules/utility/source/process_thread_impl.cc
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "process_thread_impl.h"
+#include "module.h"
+#include "trace.h"
+
+namespace webrtc {
+ProcessThread::~ProcessThread()
+{
+}
+
+ProcessThread* ProcessThread::CreateProcessThread()
+{
+ return new ProcessThreadImpl();
+}
+
+void ProcessThread::DestroyProcessThread(ProcessThread* module)
+{
+ delete module;
+}
+
+ProcessThreadImpl::ProcessThreadImpl()
+ : _timeEvent(*EventWrapper::Create()),
+ _critSectModules(CriticalSectionWrapper::CreateCriticalSection()),
+ _thread(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__);
+}
+
+ProcessThreadImpl::~ProcessThreadImpl()
+{
+ delete _critSectModules;
+ delete &_timeEvent;
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_Word32 ProcessThreadImpl::Start()
+{
+ CriticalSectionScoped lock(_critSectModules);
+ if(_thread)
+ {
+ return -1;
+ }
+ _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
+ "ProcessThread");
+ unsigned int id;
+ WebRtc_Word32 retVal = _thread->Start(id);
+ if(retVal >= 0)
+ {
+ return 0;
+ }
+ delete _thread;
+ _thread = NULL;
+ return -1;
+}
+
+WebRtc_Word32 ProcessThreadImpl::Stop()
+{
+ _critSectModules->Enter();
+ if(_thread)
+ {
+ _thread->SetNotAlive();
+
+ ThreadWrapper* thread = _thread;
+ _thread = NULL;
+
+ _timeEvent.Set();
+ _critSectModules->Leave();
+
+ if(thread->Stop())
+ {
+ delete thread;
+ } else {
+ return -1;
+ }
+ } else {
+ _critSectModules->Leave();
+ }
+ return 0;
+}
+
+WebRtc_Word32 ProcessThreadImpl::RegisterModule(const Module* module)
+{
+ CriticalSectionScoped lock(_critSectModules);
+
+ // Only allow module to be registered once.
+ ListItem* item = _modules.First();
+ for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
+ {
+ if(module == item->GetItem())
+ {
+ return -1;
+ }
+ item = _modules.Next(item);
+ }
+
+ _modules.PushFront(module);
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1,
+ "number of registered modules has increased to %d",
+ _modules.GetSize());
+ // Wake the thread calling ProcessThreadImpl::Process() to update the
+ // waiting time. The waiting time for the just registered module may be
+ // shorter than all other registered modules.
+ _timeEvent.Set();
+ return 0;
+}
+
+WebRtc_Word32 ProcessThreadImpl::DeRegisterModule(const Module* module)
+{
+ CriticalSectionScoped lock(_critSectModules);
+
+ ListItem* item = _modules.First();
+ for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
+ {
+ if(module == item->GetItem())
+ {
+ int res = _modules.Erase(item);
+ WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1,
+ "number of registered modules has decreased to %d",
+ _modules.GetSize());
+ return res;
+ }
+ item = _modules.Next(item);
+ }
+ return -1;
+}
+
+bool ProcessThreadImpl::Run(void* obj)
+{
+ return static_cast<ProcessThreadImpl*>(obj)->Process();
+}
+
+bool ProcessThreadImpl::Process()
+{
+ // Wait for the module that should be called next, but don't block thread
+ // longer than 100 ms.
+ WebRtc_Word32 minTimeToNext = 100;
+ {
+ CriticalSectionScoped lock(_critSectModules);
+ ListItem* item = _modules.First();
+ for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
+ {
+ WebRtc_Word32 timeToNext =
+ static_cast<Module*>(item->GetItem())->TimeUntilNextProcess();
+ if(minTimeToNext > timeToNext)
+ {
+ minTimeToNext = timeToNext;
+ }
+ item = _modules.Next(item);
+ }
+ }
+
+ if(minTimeToNext > 0)
+ {
+ if(kEventError == _timeEvent.Wait(minTimeToNext))
+ {
+ return true;
+ }
+ if(!_thread)
+ {
+ return false;
+ }
+ }
+ {
+ CriticalSectionScoped lock(_critSectModules);
+ ListItem* item = _modules.First();
+ for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
+ {
+ WebRtc_Word32 timeToNext =
+ static_cast<Module*>(item->GetItem())->TimeUntilNextProcess();
+ if(timeToNext < 1)
+ {
+ static_cast<Module*>(item->GetItem())->Process();
+ }
+ item = _modules.Next(item);
+ }
+ }
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/process_thread_impl.h b/trunk/src/modules/utility/source/process_thread_impl.h
new file mode 100644
index 0000000..79b1272
--- /dev/null
+++ b/trunk/src/modules/utility/source/process_thread_impl.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "list_wrapper.h"
+#include "process_thread.h"
+#include "thread_wrapper.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class ProcessThreadImpl : public ProcessThread
+{
+public:
+ ProcessThreadImpl();
+ virtual ~ProcessThreadImpl();
+
+ virtual WebRtc_Word32 Start();
+ virtual WebRtc_Word32 Stop();
+
+ virtual WebRtc_Word32 RegisterModule(const Module* module);
+ virtual WebRtc_Word32 DeRegisterModule(const Module* module);
+
+protected:
+ static bool Run(void* obj);
+
+ bool Process();
+
+private:
+ EventWrapper& _timeEvent;
+ CriticalSectionWrapper* _critSectModules;
+ ListWrapper _modules;
+ ThreadWrapper* _thread;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
diff --git a/trunk/src/modules/utility/source/rtp_dump_impl.cc b/trunk/src/modules/utility/source/rtp_dump_impl.cc
new file mode 100644
index 0000000..69a52ec
--- /dev/null
+++ b/trunk/src/modules/utility/source/rtp_dump_impl.cc
@@ -0,0 +1,282 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_dump_impl.h"
+
+#include <cassert>
+#include <stdio.h>
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#if defined(_WIN32)
+#include <Windows.h>
+#include <mmsystem.h>
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#include <string.h>
+#include <sys/time.h>
+#include <time.h>
+#endif
+
+#if (defined(_DEBUG) && defined(_WIN32))
+#define DEBUG_PRINT(expr) OutputDebugString(##expr)
+#define DEBUG_PRINTP(expr, p) \
+{ \
+ char msg[128]; \
+ sprintf(msg, ##expr, p); \
+ OutputDebugString(msg); \
+}
+#else
+#define DEBUG_PRINT(expr) ((void)0)
+#define DEBUG_PRINTP(expr,p) ((void)0)
+#endif // defined(_DEBUG) && defined(_WIN32)
+
+namespace webrtc {
+const char RTPFILE_VERSION[] = "1.0";
+const WebRtc_UWord32 MAX_UWORD32 = 0xffffffff;
+
+// This stucture is specified in the rtpdump documentation.
+// This struct corresponds to RD_packet_t in
+// http://www.cs.columbia.edu/irt/software/rtptools/
+typedef struct
+{
+ // Length of packet, including this header (may be smaller than plen if not
+ // whole packet recorded).
+ WebRtc_UWord16 length;
+ // Actual header+payload length for RTP, 0 for RTCP.
+ WebRtc_UWord16 plen;
+ // Milliseconds since the start of recording.
+ WebRtc_UWord32 offset;
+} rtpDumpPktHdr_t;
+
+RtpDump* RtpDump::CreateRtpDump()
+{
+ return new RtpDumpImpl();
+}
+
+void RtpDump::DestroyRtpDump(RtpDump* object)
+{
+ delete object;
+}
+
+RtpDumpImpl::RtpDumpImpl()
+ : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _file(*FileWrapper::Create()),
+ _startTime(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__);
+}
+
+RtpDump::~RtpDump()
+{
+}
+
+RtpDumpImpl::~RtpDumpImpl()
+{
+ _file.Flush();
+ _file.CloseFile();
+ delete &_file;
+ delete _critSect;
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__);
+}
+
+WebRtc_Word32 RtpDumpImpl::Start(const char* fileNameUTF8)
+{
+
+ if (fileNameUTF8 == NULL)
+ {
+ return -1;
+ }
+
+ CriticalSectionScoped lock(_critSect);
+ _file.Flush();
+ _file.CloseFile();
+ if (_file.OpenFile(fileNameUTF8, false, false, false) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "failed to open the specified file");
+ return -1;
+ }
+
+ // Store start of RTP dump (to be used for offset calculation later).
+ _startTime = GetTimeInMS();
+
+ // All rtp dump files start with #!rtpplay.
+ char magic[16];
+ sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION);
+ if (_file.WriteText(magic) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "error writing to file");
+ return -1;
+ }
+
+ // The header according to the rtpdump documentation is sizeof(RD_hdr_t)
+ // which is 8 + 4 + 2 = 14 bytes for 32-bit architecture (and 22 bytes on
+ // 64-bit architecture). However, Wireshark use 16 bytes for the header
+ // regardless of if the binary is 32-bit or 64-bit. Go by the same approach
+ // as Wireshark since it makes more sense.
+ // http://wiki.wireshark.org/rtpdump explains that an additional 2 bytes
+ // of padding should be added to the header.
+ char dummyHdr[16];
+ memset(dummyHdr, 0, 16);
+ if (!_file.Write(dummyHdr, sizeof(dummyHdr)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "error writing to file");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 RtpDumpImpl::Stop()
+{
+ CriticalSectionScoped lock(_critSect);
+ _file.Flush();
+ _file.CloseFile();
+ return 0;
+}
+
+bool RtpDumpImpl::IsActive() const
+{
+ CriticalSectionScoped lock(_critSect);
+ return _file.Open();
+}
+
+WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet,
+ WebRtc_UWord16 packetLength)
+{
+ CriticalSectionScoped lock(_critSect);
+ if (!IsActive())
+ {
+ return 0;
+ }
+
+ if (packet == NULL)
+ {
+ return -1;
+ }
+
+ if (packetLength < 1)
+ {
+ return -1;
+ }
+
+ // If the packet doesn't contain a valid RTCP header the packet will be
+ // considered RTP (without further verification).
+ bool isRTCP = RTCP(packet);
+
+ rtpDumpPktHdr_t hdr;
+ WebRtc_UWord32 offset;
+
+ // Offset is relative to when recording was started.
+ offset = GetTimeInMS();
+ if (offset < _startTime)
+ {
+ // Compensate for wraparound.
+ offset += MAX_UWORD32 - _startTime + 1;
+ } else {
+ offset -= _startTime;
+ }
+ hdr.offset = RtpDumpHtonl(offset);
+
+ hdr.length = RtpDumpHtons((WebRtc_UWord16)(packetLength + sizeof(hdr)));
+ if (isRTCP)
+ {
+ hdr.plen = 0;
+ }
+ else
+ {
+ hdr.plen = RtpDumpHtons((WebRtc_UWord16)packetLength);
+ }
+
+ if (!_file.Write(&hdr, sizeof(hdr)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "error writing to file");
+ return -1;
+ }
+ if (!_file.Write(packet, packetLength))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "error writing to file");
+ return -1;
+ }
+
+ return 0;
+}
+
+bool RtpDumpImpl::RTCP(const WebRtc_UWord8* packet) const
+{
+ const WebRtc_UWord8 payloadType = packet[1];
+ bool is_rtcp = false;
+
+ switch(payloadType)
+ {
+ case 192:
+ is_rtcp = true;
+ break;
+ case 193: case 195:
+ break;
+ case 200: case 201: case 202: case 203:
+ case 204: case 205: case 206: case 207:
+ is_rtcp = true;
+ break;
+ }
+ return is_rtcp;
+}
+
+// TODO (hellner): why is TickUtil not used here?
+inline WebRtc_UWord32 RtpDumpImpl::GetTimeInMS() const
+{
+#if defined(_WIN32)
+ return timeGetTime();
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ struct timeval tv;
+ struct timezone tz;
+ unsigned long val;
+
+ gettimeofday(&tv, &tz);
+ val = tv.tv_sec * 1000 + tv.tv_usec / 1000;
+ return val;
+#else
+ #error Either _WIN32 or LINUX or WEBRTC_MAC has to be defined!
+ assert(false);
+ return 0;
+#endif
+}
+
+inline WebRtc_UWord32 RtpDumpImpl::RtpDumpHtonl(WebRtc_UWord32 x) const
+{
+#if defined(WEBRTC_BIG_ENDIAN)
+ return x;
+#elif defined(WEBRTC_LITTLE_ENDIAN)
+ return (x >> 24) + ((((x >> 16) & 0xFF) << 8) + ((((x >> 8) & 0xFF) << 16) +
+ ((x & 0xFF) << 24)));
+#else
+#error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
+ assert(false);
+ return 0;
+#endif
+}
+
+inline WebRtc_UWord16 RtpDumpImpl::RtpDumpHtons(WebRtc_UWord16 x) const
+{
+#if defined(WEBRTC_BIG_ENDIAN)
+ return x;
+#elif defined(WEBRTC_LITTLE_ENDIAN)
+ return (x >> 8) + ((x & 0xFF) << 8);
+#else
+ #error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
+ assert(false);
+ return 0;
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/rtp_dump_impl.h b/trunk/src/modules/utility/source/rtp_dump_impl.h
new file mode 100644
index 0000000..9715c35
--- /dev/null
+++ b/trunk/src/modules/utility/source/rtp_dump_impl.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
+
+#include "rtp_dump.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class FileWrapper;
+class RtpDumpImpl : public RtpDump
+{
+public:
+ RtpDumpImpl();
+ virtual ~RtpDumpImpl();
+
+ virtual WebRtc_Word32 Start(const char* fileNameUTF8);
+ virtual WebRtc_Word32 Stop();
+ virtual bool IsActive() const;
+ virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
+ WebRtc_UWord16 packetLength);
+private:
+ // Return the system time in ms.
+ inline WebRtc_UWord32 GetTimeInMS() const;
+ // Return x in network byte order (big endian).
+ inline WebRtc_UWord32 RtpDumpHtonl(WebRtc_UWord32 x) const;
+ // Return x in network byte order (big endian).
+ inline WebRtc_UWord16 RtpDumpHtons(WebRtc_UWord16 x) const;
+
+ // Return true if the packet starts with a valid RTCP header.
+ // Note: See ModuleRTPUtility::RTPHeaderParser::RTCP() for details on how
+ // to determine if the packet is an RTCP packet.
+ bool RTCP(const WebRtc_UWord8* packet) const;
+
+private:
+ CriticalSectionWrapper* _critSect;
+ FileWrapper& _file;
+ WebRtc_UWord32 _startTime;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
diff --git a/trunk/src/modules/utility/source/utility.gypi b/trunk/src/modules/utility/source/utility.gypi
new file mode 100644
index 0000000..7dcce8e
--- /dev/null
+++ b/trunk/src/modules/utility/source/utility.gypi
@@ -0,0 +1,92 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_utility',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'audio_coding_module',
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ '../../media_file/interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../interface',
+ '../../audio_coding/main/interface',
+ ],
+ },
+ 'sources': [
+ '../interface/file_player.h',
+ '../interface/file_recorder.h',
+ '../interface/process_thread.h',
+ '../interface/rtp_dump.h',
+ 'coder.cc',
+ 'coder.h',
+ 'file_player_impl.cc',
+ 'file_player_impl.h',
+ 'file_recorder_impl.cc',
+ 'file_recorder_impl.h',
+ 'process_thread_impl.cc',
+ 'process_thread_impl.h',
+ 'rtp_dump_impl.cc',
+ 'rtp_dump_impl.h',
+ ],
+ 'conditions': [
+ ['enable_video==1', {
+ # Adds support for video recording.
+ 'defines': [
+ 'WEBRTC_MODULE_UTILITY_VIDEO',
+ ],
+ 'dependencies': [
+ 'webrtc_video_coding',
+ ],
+ 'include_dirs': [
+ '../../video_coding/main/interface',
+ ],
+ 'sources': [
+ 'frame_scaler.cc',
+ 'video_coder.cc',
+ 'video_frames_queue.cc',
+ ],
+ }],
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'webrtc_utility_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'webrtc_utility',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'file_player_unittest.cc',
+ ],
+ }, # webrtc_utility_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/utility/source/video_coder.cc b/trunk/src/modules/utility/source/video_coder.cc
new file mode 100644
index 0000000..e17c3e0
--- /dev/null
+++ b/trunk/src/modules/utility/source/video_coder.cc
@@ -0,0 +1,152 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include "video_coder.h"
+
+namespace webrtc {
+VideoCoder::VideoCoder(WebRtc_UWord32 instanceID)
+ : _instanceID( instanceID),
+ _vcm(VideoCodingModule::Create(instanceID)),
+ _decodedVideo(0)
+{
+ _vcm->InitializeSender();
+ _vcm->InitializeReceiver();
+
+ _vcm->RegisterTransportCallback(this);
+ _vcm->RegisterReceiveCallback(this);
+}
+
+VideoCoder::~VideoCoder()
+{
+ VideoCodingModule::Destroy(_vcm);
+}
+
+WebRtc_Word32 VideoCoder::ResetDecoder()
+{
+ _vcm->ResetDecoder();
+
+ _vcm->InitializeSender();
+ _vcm->InitializeReceiver();
+
+ _vcm->RegisterTransportCallback(this);
+ _vcm->RegisterReceiveCallback(this);
+ return 0;
+}
+
+WebRtc_Word32 VideoCoder::SetEncodeCodec(VideoCodec& videoCodecInst,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize)
+{
+ if(_vcm->RegisterSendCodec(&videoCodecInst, numberOfCores,
+ maxPayloadSize) != VCM_OK)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32 VideoCoder::SetDecodeCodec(VideoCodec& videoCodecInst,
+ WebRtc_Word32 numberOfCores)
+{
+ if (videoCodecInst.plType == 0)
+ {
+ WebRtc_Word8 plType = DefaultPayloadType(videoCodecInst.plName);
+ if (plType == -1)
+ {
+ return -1;
+ }
+ videoCodecInst.plType = plType;
+ }
+
+ if(_vcm->RegisterReceiveCodec(&videoCodecInst, numberOfCores) != VCM_OK)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCoder::Decode(VideoFrame& decodedVideo,
+ const EncodedVideoData& encodedData)
+{
+ decodedVideo.SetLength(0);
+ if(encodedData.payloadSize <= 0)
+ {
+ return -1;
+ }
+
+ _decodedVideo = &decodedVideo;
+ if(_vcm->DecodeFromStorage(encodedData) != VCM_OK)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+
+WebRtc_Word32 VideoCoder::Encode(const VideoFrame& videoFrame,
+ EncodedVideoData& videoEncodedData)
+{
+ // The AddVideoFrame(..) call will (indirectly) call SendData(). Store a
+ // pointer to videoFrame so that it can be updated.
+ _videoEncodedData = &videoEncodedData;
+ videoEncodedData.payloadSize = 0;
+ if(_vcm->AddVideoFrame(videoFrame) != VCM_OK)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word8 VideoCoder::DefaultPayloadType(const char* plName)
+{
+ VideoCodec tmpCodec;
+ WebRtc_Word32 numberOfCodecs = _vcm->NumberOfCodecs();
+ for (WebRtc_UWord8 i = 0; i < numberOfCodecs; i++)
+ {
+ _vcm->Codec(i, &tmpCodec);
+ if(strncmp(tmpCodec.plName, plName, kPayloadNameSize) == 0)
+ {
+ return tmpCodec.plType;
+ }
+ }
+ return -1;
+}
+
+WebRtc_Word32 VideoCoder::FrameToRender(VideoFrame& videoFrame)
+{
+ return _decodedVideo->CopyFrame(videoFrame);
+}
+
+WebRtc_Word32 VideoCoder::SendData(
+ FrameType frameType,
+ WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* /*rtpVideoHdr*/)
+{
+ // Store the data in _videoEncodedData which is a pointer to videoFrame in
+ // Encode(..)
+ _videoEncodedData->VerifyAndAllocate(payloadSize);
+ _videoEncodedData->frameType = frameType;
+ _videoEncodedData->payloadType = payloadType;
+ _videoEncodedData->timeStamp = timeStamp;
+ _videoEncodedData->fragmentationHeader = fragmentationHeader;
+ memcpy(_videoEncodedData->payloadData, payloadData,
+ sizeof(WebRtc_UWord8) * payloadSize);
+ _videoEncodedData->payloadSize = payloadSize;
+ return 0;
+}
+} // namespace webrtc
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/video_coder.h b/trunk/src/modules/utility/source/video_coder.h
new file mode 100644
index 0000000..5c4b0ae
--- /dev/null
+++ b/trunk/src/modules/utility/source/video_coder.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include "engine_configurations.h"
+#include "video_coding.h"
+
+namespace webrtc {
+class VideoCoder : public VCMPacketizationCallback, public VCMReceiveCallback
+{
+public:
+ VideoCoder(WebRtc_UWord32 instanceID);
+ ~VideoCoder();
+
+ WebRtc_Word32 ResetDecoder();
+
+ WebRtc_Word32 SetEncodeCodec(VideoCodec& videoCodecInst,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize);
+
+
+ // Select the codec that should be used for decoding. videoCodecInst.plType
+ // will be set to the codec's default payload type.
+ WebRtc_Word32 SetDecodeCodec(VideoCodec& videoCodecInst,
+ WebRtc_Word32 numberOfCores);
+
+ WebRtc_Word32 Decode(VideoFrame& decodedVideo,
+ const EncodedVideoData& encodedData);
+
+ WebRtc_Word32 Encode(const VideoFrame& videoFrame,
+ EncodedVideoData& videoEncodedData);
+
+ WebRtc_Word8 DefaultPayloadType(const char* plName);
+
+private:
+ // VCMReceiveCallback function.
+ // Note: called by VideoCodingModule when decoding finished.
+ WebRtc_Word32 FrameToRender(VideoFrame& videoFrame);
+
+ // VCMPacketizationCallback function.
+ // Note: called by VideoCodingModule when encoding finished.
+ WebRtc_Word32 SendData(
+ const FrameType /*frameType*/,
+ const WebRtc_UWord8 /*payloadType*/,
+ const WebRtc_UWord32 /*timeStamp*/,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& /* fragmentationHeader*/,
+ const RTPVideoHeader* rtpTypeHdr);
+
+ WebRtc_UWord32 _instanceID;
+ VideoCodingModule* _vcm;
+ VideoFrame* _decodedVideo;
+ EncodedVideoData* _videoEncodedData;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
diff --git a/trunk/src/modules/utility/source/video_frames_queue.cc b/trunk/src/modules/utility/source/video_frames_queue.cc
new file mode 100644
index 0000000..ab590c4
--- /dev/null
+++ b/trunk/src/modules/utility/source/video_frames_queue.cc
@@ -0,0 +1,150 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_frames_queue.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include <cassert>
+
+#include "module_common_types.h"
+#include "tick_util.h"
+#include "trace.h"
+
+namespace webrtc {
+VideoFramesQueue::VideoFramesQueue()
+ : _incomingFrames(),
+ _renderDelayMs(10)
+{
+}
+
+VideoFramesQueue::~VideoFramesQueue()
+{
+ while (!_incomingFrames.Empty())
+ {
+ ListItem* item = _incomingFrames.First();
+ if (item)
+ {
+ VideoFrame* ptrFrame = static_cast<VideoFrame*>(item->GetItem());
+ assert(ptrFrame != NULL);
+ ptrFrame->Free();
+ delete ptrFrame;
+ }
+ _incomingFrames.Erase(item);
+ }
+ while (!_emptyFrames.Empty())
+ {
+ ListItem* item = _emptyFrames.First();
+ if (item)
+ {
+ VideoFrame* ptrFrame = static_cast<VideoFrame*>(item->GetItem());
+ assert(ptrFrame != NULL);
+ ptrFrame->Free();
+ delete ptrFrame;
+ }
+ _emptyFrames.Erase(item);
+ }
+}
+
+WebRtc_Word32 VideoFramesQueue::AddFrame(const VideoFrame& newFrame)
+{
+ VideoFrame* ptrFrameToAdd = NULL;
+ // Try to re-use a VideoFrame. Only allocate new memory if it is necessary.
+ if (!_emptyFrames.Empty())
+ {
+ ListItem* item = _emptyFrames.First();
+ if (item)
+ {
+ ptrFrameToAdd = static_cast<VideoFrame*>(item->GetItem());
+ _emptyFrames.Erase(item);
+ }
+ }
+ if (!ptrFrameToAdd)
+ {
+ if (_emptyFrames.GetSize() + _incomingFrames.GetSize() >
+ KMaxNumberOfFrames)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
+ "%s: too many frames, limit: %d", __FUNCTION__,
+ KMaxNumberOfFrames);
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1,
+ "%s: allocating buffer %d", __FUNCTION__,
+ _emptyFrames.GetSize() + _incomingFrames.GetSize());
+
+ ptrFrameToAdd = new VideoFrame();
+ if (!ptrFrameToAdd)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "%s: could not create new frame for", __FUNCTION__);
+ return -1;
+ }
+ }
+ ptrFrameToAdd->CopyFrame(newFrame);
+ _incomingFrames.PushBack(ptrFrameToAdd);
+ return 0;
+}
+
+// Find the most recent frame that has a VideoFrame::RenderTimeMs() that is
+// lower than current time in ms (TickTime::MillisecondTimestamp()).
+// Note _incomingFrames is sorted so that the oldest frame is first.
+// Recycle all frames that are older than the most recent frame.
+VideoFrame* VideoFramesQueue::FrameToRecord()
+{
+ VideoFrame* ptrRenderFrame = NULL;
+ ListItem* item = _incomingFrames.First();
+ while(item)
+ {
+ VideoFrame* ptrOldestFrameInList =
+ static_cast<VideoFrame*>(item->GetItem());
+ if (ptrOldestFrameInList->RenderTimeMs() <=
+ TickTime::MillisecondTimestamp() + _renderDelayMs)
+ {
+ if (ptrRenderFrame)
+ {
+ // List is traversed beginning to end. If ptrRenderFrame is not
+ // NULL it must be the first, and thus oldest, VideoFrame in the
+ // queue. It can be recycled.
+ ReturnFrame(ptrRenderFrame);
+ _incomingFrames.PopFront();
+ }
+ item = _incomingFrames.Next(item);
+ ptrRenderFrame = ptrOldestFrameInList;
+ }else
+ {
+ // All VideoFrames following this one will be even newer. No match
+ // will be found.
+ break;
+ }
+ }
+ return ptrRenderFrame;
+}
+
+WebRtc_Word32 VideoFramesQueue::ReturnFrame(VideoFrame* ptrOldFrame)
+{
+ ptrOldFrame->SetTimeStamp(0);
+ ptrOldFrame->SetWidth(0);
+ ptrOldFrame->SetHeight(0);
+ ptrOldFrame->SetRenderTime(0);
+ ptrOldFrame->SetLength(0);
+ _emptyFrames.PushBack(ptrOldFrame);
+ return 0;
+}
+
+//
+WebRtc_Word32 VideoFramesQueue::SetRenderDelay(WebRtc_UWord32 renderDelay)
+{
+ _renderDelayMs = renderDelay;
+ return 0;
+}
+} // namespace webrtc
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/video_frames_queue.h b/trunk/src/modules/utility/source/video_frames_queue.h
new file mode 100644
index 0000000..6c9be1c
--- /dev/null
+++ b/trunk/src/modules/utility/source/video_frames_queue.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+
+#include "engine_configurations.h"
+#include "list_wrapper.h"
+#include "typedefs.h"
+
+namespace webrtc {
+class VideoFrame;
+
+class VideoFramesQueue
+{
+public:
+ VideoFramesQueue();
+ ~VideoFramesQueue();
+
+ // Put newFrame (last) in the queue.
+ WebRtc_Word32 AddFrame(const VideoFrame& newFrame);
+
+ // Return the most current frame. I.e. the frame with the highest
+ // VideoFrame::RenderTimeMs() that is lower than
+ // TickTime::MillisecondTimestamp().
+ VideoFrame* FrameToRecord();
+
+ // Set the render delay estimate to renderDelay ms.
+ WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 renderDelay);
+
+protected:
+ // Make ptrOldFrame available for re-use. I.e. put it in the empty frames
+ // queue.
+ WebRtc_Word32 ReturnFrame(VideoFrame* ptrOldFrame);
+
+private:
+ // Don't allow the buffer to expand beyond KMaxNumberOfFrames VideoFrames.
+ // 300 frames correspond to 10 seconds worth of frames at 30 fps.
+ enum {KMaxNumberOfFrames = 300};
+
+ // List of VideoFrame pointers. The list is sorted in the order of when the
+ // VideoFrame was inserted into the list. The first VideoFrame in the list
+ // was inserted first.
+ ListWrapper _incomingFrames;
+ // A list of frames that are free to be re-used.
+ ListWrapper _emptyFrames;
+
+ // Estimated render delay.
+ WebRtc_UWord32 _renderDelayMs;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
diff --git a/trunk/src/modules/utility/test/testAPI.cc b/trunk/src/modules/utility/test/testAPI.cc
new file mode 100644
index 0000000..96664a3
--- /dev/null
+++ b/trunk/src/modules/utility/test/testAPI.cc
@@ -0,0 +1,368 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// testAPI.cpp : Defines the entry point for the console application.
+//
+// NOTES:
+// 1. MediaFile library and testAPI.cpp must be built in DEBUG mode for testing.
+//
+
+#include <iostream>
+#include <stdio.h>
+#include <assert.h>
+
+#ifdef WIN32
+ #include <windows.h>
+ #include <tchar.h>
+#endif
+
+#include "common_types.h"
+#include "trace.h"
+
+#include "Engineconfigurations.h"
+#include "media_file.h"
+#include "file_player.h"
+#include "file_recorder.h"
+
+
+bool notify = false, playing = false, recording = false;
+
+// callback class for FileModule
+class MyFileModuleCallback : public FileCallback
+{
+public:
+ virtual void PlayNotification( const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs )
+ {
+ printf("\tReceived PlayNotification from module %ld, durationMs = %ld\n",
+ id, durationMs);
+ notify = true;
+ };
+
+ virtual void RecordNotification( const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs )
+ {
+ printf("\tReceived RecordNotification from module %ld, durationMs = %ld\n",
+ id, durationMs);
+ notify = true;
+ };
+
+ virtual void PlayFileEnded(const WebRtc_Word32 id)
+ {
+ printf("\tReceived PlayFileEnded notification from module %ld.\n", id);
+ playing = false;
+ };
+
+ virtual void RecordFileEnded(const WebRtc_Word32 id)
+ {
+ printf("\tReceived RecordFileEnded notification from module %ld.\n", id);
+ recording = false;
+ }
+};
+
+// main test app
+#ifdef WIN32
+int _tmain(int argc, _TCHAR* argv[])
+#else
+int main(int /*argc*/, char** /*argv*/)
+#endif
+{
+ Trace::CreateTrace();
+ Trace::SetTraceFile("testTrace.txt");
+ Trace::SetEncryptedTraceFile("testTraceDebug.txt");
+
+ int playId = 1;
+ int recordId = 2;
+
+ printf("Welcome to test of FilePlayer and FileRecorder\n");
+
+
+ ///////////////////////////////////////////////
+ //
+ // avi test case 1
+ //
+ ///////////////////////////////////////////////
+
+
+ // todo PW we need more AVI tests Mp4
+
+ {
+ FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(1, webrtc::kFileFormatAviFile));
+ FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(1, webrtc::kFileFormatAviFile));
+
+ const char* KFileName = "./tmpAviFileTestCase1_audioI420CIF30fps.avi";
+
+ printf("\tReading from an avi file and writing the information to another \n");
+ printf("\tin the same format (I420 CIF 30fps) \n");
+ printf("\t\t check file named %s\n", KFileName);
+
+ assert(filePlayer.StartPlayingVideoFile(
+ "../../../MediaFile/main/test/files/aviTestCase1_audioI420CIF30fps.avi",
+ false, false) == 0);
+
+ // init codecs
+ webrtc::VideoCodec videoCodec;
+ webrtc::VideoCodec recVideoCodec;
+ webrtc::CodecInst audioCodec;
+ assert(filePlayer.VideoCodec( videoCodec ) == 0);
+ assert(filePlayer.AudioCodec( audioCodec) == 0);
+
+ recVideoCodec = videoCodec;
+
+ assert( fileRecorder.StartRecordingVideoFile(KFileName,
+ audioCodec,
+ recVideoCodec) == 0);
+
+ assert(fileRecorder.IsRecording());
+
+ WebRtc_UWord32 videoReadSize = static_cast<WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3.0) / 2.0);
+
+ webrtc::VideoFrame videoFrame;
+ videoFrame.VerifyAndAllocate(videoReadSize);
+
+ int frameCount = 0;
+ bool audioNotDone = true;
+ bool videoNotDone = true;
+ AudioFrame audioFrame;
+
+ while( audioNotDone || videoNotDone)
+ {
+ if(filePlayer.TimeUntilNextVideoFrame() <= 0)
+ {
+ if(filePlayer.GetVideoFromFile( videoFrame) != 0)
+ {
+ // no more video frames
+ break;
+ }
+ frameCount++;
+ videoNotDone = ( videoFrame.Length() > 0);
+ videoFrame.SetRenderTime(TickTime::MillisecondTimestamp());
+ if( videoNotDone)
+ {
+ assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
+ ::Sleep(10);
+ }
+ }
+ WebRtc_UWord32 decodedDataLengthInSamples;
+ if( 0 != filePlayer.Get10msAudioFromFile( audioFrame._payloadData, decodedDataLengthInSamples, audioCodec.plfreq))
+ {
+ audioNotDone = false;
+ } else
+ {
+ audioFrame._frequencyInHz = filePlayer.Frequency();
+ audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)decodedDataLengthInSamples;
+ fileRecorder.RecordAudioToFile(audioFrame, &TickTime::Now());
+ }
+ }
+ ::Sleep(100);
+ assert(fileRecorder.StopRecording() == 0);
+ assert( !fileRecorder.IsRecording());
+ assert(frameCount == 135);
+ printf("\tGenerated %s\n\n", KFileName);
+ }
+ ///////////////////////////////////////////////
+ //
+ // avi test case 2
+ //
+ ///////////////////////////////////////////////
+ {
+ FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(2, webrtc::kFileFormatAviFile));
+ FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(2, webrtc::kFileFormatAviFile));
+
+ const char* KFileName = "./tmpAviFileTestCase2_audioI420CIF20fps.avi";
+
+ printf("\tWriting information to a avi file and check the written file by \n");
+ printf("\treopening it and control codec information.\n");
+ printf("\t\t check file named %s all frames should be light green.\n", KFileName);
+ // init codecs
+ webrtc::VideoCodec videoCodec;
+ webrtc::CodecInst audioCodec;
+
+ memset(&videoCodec, 0, sizeof(videoCodec));
+
+ const char* KVideoCodecName = "I420";
+ strcpy(videoCodec.plName, KVideoCodecName);
+ videoCodec.plType = 124;
+ videoCodec.maxFramerate = 20;
+ videoCodec.height = 288;
+ videoCodec.width = 352;
+
+ const char* KAudioCodecName = "PCMU";
+ strcpy(audioCodec.plname, KAudioCodecName);
+ audioCodec.pltype = 0;
+ audioCodec.plfreq = 8000;
+ audioCodec.pacsize = 80;
+ audioCodec.channels = 1;
+ audioCodec.rate = 64000;
+
+ assert( fileRecorder.StartRecordingVideoFile(
+ KFileName,
+ audioCodec,
+ videoCodec) == 0);
+
+ assert(fileRecorder.IsRecording());
+
+ const WebRtc_UWord32 KVideoWriteSize = static_cast< WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3) / 2);
+ webrtc::VideoFrame videoFrame;
+
+ // 10 ms
+ AudioFrame audioFrame;
+ audioFrame._payloadDataLengthInSamples = audioCodec.plfreq/100;
+ memset(audioFrame._payloadData, 0, 2*audioFrame._payloadDataLengthInSamples);
+ audioFrame._frequencyInHz = 8000;
+
+ // prepare the video frame
+ videoFrame.VerifyAndAllocate(KVideoWriteSize);
+ memset(videoFrame.Buffer(), 127, videoCodec.width * videoCodec.height);
+ memset(videoFrame.Buffer() +(videoCodec.width * videoCodec.height), 0, videoCodec.width * videoCodec.height/2);
+ videoFrame.SetLength(KVideoWriteSize);
+ videoFrame.SetHeight(videoCodec.height);
+ videoFrame.SetWidth(videoCodec.width);
+
+ // write avi file, with 20 video frames
+ const int KWriteNumFrames = 20;
+ int writeFrameCount = 0;
+ while(writeFrameCount < KWriteNumFrames)
+ {
+ // add a video frame
+ assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
+
+ // add 50 ms of audio
+ for(int i=0; i<5; i++)
+ {
+ assert( fileRecorder.RecordAudioToFile(audioFrame) == 0);
+ }// for i
+ writeFrameCount++;
+ }
+ ::Sleep(10); // enough tim eto write the queued data to the file
+ assert(writeFrameCount == 20);
+ assert(fileRecorder.StopRecording() == 0);
+ assert( ! fileRecorder.IsRecording());
+
+ assert(filePlayer.StartPlayingVideoFile(KFileName,false, false) == 0);
+ assert(filePlayer.IsPlayingFile( ));
+
+ // compare codecs read from file to the ones used when writing the file
+ webrtc::VideoCodec readVideoCodec;
+ assert(filePlayer.VideoCodec( readVideoCodec ) == 0);
+ assert(strcmp(readVideoCodec.plName, videoCodec.plName) == 0);
+ assert(readVideoCodec.width == videoCodec.width);
+ assert(readVideoCodec.height == videoCodec.height);
+ assert(readVideoCodec.maxFramerate == videoCodec.maxFramerate);
+
+ webrtc::CodecInst readAudioCodec;
+ assert(filePlayer.AudioCodec( readAudioCodec) == 0);
+ assert(strcmp(readAudioCodec.plname, audioCodec.plname) == 0);
+ assert(readAudioCodec.pltype == audioCodec.pltype);
+ assert(readAudioCodec.plfreq == audioCodec.plfreq);
+ assert(readAudioCodec.pacsize == audioCodec.pacsize);
+ assert(readAudioCodec.channels == audioCodec.channels);
+ assert(readAudioCodec.rate == audioCodec.rate);
+
+ assert(filePlayer.StopPlayingFile() == 0);
+ assert( ! filePlayer.IsPlayingFile());
+ printf("\tGenerated %s\n\n", KFileName);
+ }
+ ///////////////////////////////////////////////
+ //
+ // avi test case 3
+ //
+ ///////////////////////////////////////////////
+
+ {
+ FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(2, webrtc::kFileFormatAviFile));
+ FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(3, webrtc::kFileFormatAviFile));
+
+ printf("\tReading from an avi file and writing the information to another \n");
+ printf("\tin a different format (H.263 CIF 30fps) \n");
+ printf("\t\t check file named tmpAviFileTestCase1_audioH263CIF30fps.avi\n");
+
+ assert(filePlayer.StartPlayingVideoFile(
+ "../../../MediaFile/main/test/files/aviTestCase1_audioI420CIF30fps.avi",
+ false,
+ false) == 0);
+
+ // init codecs
+ webrtc::VideoCodec videoCodec;
+ webrtc::VideoCodec recVideoCodec;
+ webrtc::CodecInst audioCodec;
+ assert(filePlayer.VideoCodec( videoCodec ) == 0);
+ assert(filePlayer.AudioCodec( audioCodec) == 0);
+ recVideoCodec = videoCodec;
+
+ memcpy(recVideoCodec.plName, "H263",5);
+ recVideoCodec.startBitrate = 1000;
+ recVideoCodec.codecSpecific.H263.quality = 1;
+ recVideoCodec.plType = 34;
+ recVideoCodec.codecType = webrtc::kVideoCodecH263;
+
+ assert( fileRecorder.StartRecordingVideoFile(
+ "./tmpAviFileTestCase1_audioH263CIF30fps.avi",
+ audioCodec,
+ recVideoCodec) == 0);
+
+ assert(fileRecorder.IsRecording());
+
+ WebRtc_UWord32 videoReadSize = static_cast<WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3.0) / 2.0);
+
+ webrtc::VideoFrame videoFrame;
+ videoFrame.VerifyAndAllocate(videoReadSize);
+
+ int videoFrameCount = 0;
+ int audioFrameCount = 0;
+ bool audioNotDone = true;
+ bool videoNotDone = true;
+ AudioFrame audioFrame;
+
+ while( audioNotDone || videoNotDone)
+ {
+ if(filePlayer.TimeUntilNextVideoFrame() <= 0)
+ {
+ if(filePlayer.GetVideoFromFile( videoFrame) != 0)
+ {
+ break;
+ }
+ videoFrameCount++;
+ videoNotDone = ( videoFrame.Length() > 0);
+ if( videoNotDone)
+ {
+ assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
+ }
+ }
+
+ WebRtc_UWord32 decodedDataLengthInSamples;
+ if( 0 != filePlayer.Get10msAudioFromFile( audioFrame._payloadData, decodedDataLengthInSamples, audioCodec.plfreq))
+ {
+ audioNotDone = false;
+
+ } else
+ {
+ ::Sleep(5);
+ audioFrame._frequencyInHz = filePlayer.Frequency();
+ audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)decodedDataLengthInSamples;
+ assert(0 == fileRecorder.RecordAudioToFile(audioFrame));
+
+ audioFrameCount++;
+ }
+ }
+ assert(videoFrameCount == 135);
+ assert(audioFrameCount == 446); // we will start & stop with a video frame
+
+ assert(fileRecorder.StopRecording() == 0);
+ assert( !fileRecorder.IsRecording());
+ printf("\tGenerated ./tmpAviFileTestCase1_audioH263CIF30fps.avi\n\n");
+ }
+
+
+ printf("\nTEST completed.\n");
+
+ Trace::ReturnTrace();
+ return 0;
+}
diff --git a/trunk/src/modules/video_capture/OWNERS b/trunk/src/modules/video_capture/OWNERS
new file mode 100644
index 0000000..9034747
--- /dev/null
+++ b/trunk/src/modules/video_capture/OWNERS
@@ -0,0 +1,4 @@
+mallinath@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
+wu@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture.h b/trunk/src/modules/video_capture/main/interface/video_capture.h
new file mode 100644
index 0000000..e2f2f23
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/interface/video_capture.h
@@ -0,0 +1,160 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
+
+#include "module.h"
+#include "video_capture_defines.h"
+
+namespace webrtc {
+
+class VideoCaptureModule: public RefCountedModule {
+ public:
+ // Interface for receiving information about available camera devices.
+ class DeviceInfo {
+ public:
+ virtual WebRtc_UWord32 NumberOfDevices() = 0;
+
+ // Returns the available capture devices.
+ // deviceNumber - Index of capture device.
+ // deviceNameUTF8 - Friendly name of the capture device.
+ // deviceUniqueIdUTF8 - Unique name of the capture device if it exist.
+ // Otherwise same as deviceNameUTF8.
+ // productUniqueIdUTF8 - Unique product id if it exist.
+ // Null terminated otherwise.
+ virtual WebRtc_Word32 GetDeviceName(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8 = 0,
+ WebRtc_UWord32 productUniqueIdUTF8Length = 0) = 0;
+
+
+ // Returns the number of capabilities this device.
+ virtual WebRtc_Word32 NumberOfCapabilities(
+ const WebRtc_UWord8* deviceUniqueIdUTF8) = 0;
+
+ // Gets the capabilities of the named device.
+ virtual WebRtc_Word32 GetCapability(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability) = 0;
+
+ // Gets clockwise angle the captured frames should be rotated in order
+ // to be displayed correctly on a normally rotated display.
+ virtual WebRtc_Word32 GetOrientation(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ VideoCaptureRotation& orientation) = 0;
+
+ // Gets the capability that best matches the requested width, height and
+ // frame rate.
+ // Returns the deviceCapabilityNumber on success.
+ virtual WebRtc_Word32 GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested,
+ VideoCaptureCapability& resulting) = 0;
+
+ // Display OS /capture device specific settings dialog
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8,
+ void* parentWindow,
+ WebRtc_UWord32 positionX,
+ WebRtc_UWord32 positionY) = 0;
+
+ virtual ~DeviceInfo() {}
+ };
+
+ class VideoCaptureEncodeInterface {
+ public:
+ virtual WebRtc_Word32 ConfigureEncoder(const VideoCodec& codec,
+ WebRtc_UWord32 maxPayloadSize) = 0;
+ // Inform the encoder about the new target bit rate.
+ // - newBitRate : New target bit rate in Kbit/s.
+ // - frameRate : The target frame rate.
+ virtual WebRtc_Word32 SetRates(WebRtc_Word32 newBitRate,
+ WebRtc_Word32 frameRate) = 0;
+ // Inform the encoder about the packet loss and the round-trip time.
+ // - packetLoss : Fraction lost
+ // (loss rate in percent = 100 * packetLoss / 255).
+ // - rtt : Round-trip time in milliseconds.
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
+ int rtt) = 0;
+
+ // Encode the next frame as key frame.
+ virtual WebRtc_Word32 EncodeFrameType(const FrameType type) = 0;
+ protected:
+ virtual ~VideoCaptureEncodeInterface() {
+ }
+ };
+
+ // Register capture data callback
+ virtual WebRtc_Word32 RegisterCaptureDataCallback(
+ VideoCaptureDataCallback& dataCallback) = 0;
+
+ // Remove capture data callback
+ virtual WebRtc_Word32 DeRegisterCaptureDataCallback() = 0;
+
+ // Register capture callback.
+ virtual WebRtc_Word32 RegisterCaptureCallback(
+ VideoCaptureFeedBack& callBack) = 0;
+
+ // Remove capture callback.
+ virtual WebRtc_Word32 DeRegisterCaptureCallback() = 0;
+
+ // Start capture device
+ virtual WebRtc_Word32 StartCapture(
+ const VideoCaptureCapability& capability) = 0;
+
+ virtual WebRtc_Word32 StopCapture() = 0;
+
+ // Send an image when the capture device is not running.
+ virtual WebRtc_Word32 StartSendImage(const VideoFrame& videoFrame,
+ WebRtc_Word32 frameRate = 1) = 0;
+
+ virtual WebRtc_Word32 StopSendImage() = 0;
+
+ // Returns the name of the device used by this module.
+ virtual const WebRtc_UWord8* CurrentDeviceName() const = 0;
+
+ // Returns true if the capture device is running
+ virtual bool CaptureStarted() = 0;
+
+ // Gets the current configuration.
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings) = 0;
+
+ virtual WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delayMS) = 0;
+
+ // Returns the current CaptureDelay. Only valid when the camera is running.
+ virtual WebRtc_Word32 CaptureDelay() = 0;
+
+ // Set the rotation of the captured frames.
+ // If the rotation is set to the same as returned by
+ // DeviceInfo::GetOrientation the captured frames are
+ // displayed correctly if rendered.
+ virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation) = 0;
+
+ // Gets a pointer to an encode interface if the capture device supports the
+ // requested type and size. NULL otherwise.
+ virtual VideoCaptureEncodeInterface* GetEncodeInterface(
+ const VideoCodec& codec) = 0;
+
+ virtual WebRtc_Word32 EnableFrameRateCallback(const bool enable) = 0;
+ virtual WebRtc_Word32 EnableNoPictureAlarm(const bool enable) = 0;
+
+protected:
+ virtual ~VideoCaptureModule() {};
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture_defines.h b/trunk/src/modules/video_capture/main/interface/video_capture_defines.h
new file mode 100644
index 0000000..2a3408b
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/interface/video_capture_defines.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
+
+// Includes
+#include "typedefs.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+// Defines
+#ifndef NULL
+ #define NULL 0
+#endif
+
+enum {kVideoCaptureUniqueNameLength =1024}; //Max unique capture device name lenght
+enum {kVideoCaptureDeviceNameLength =256}; //Max capture device name lenght
+enum {kVideoCaptureProductIdLength =128}; //Max product id length
+
+// Enums
+enum VideoCaptureRotation
+{
+ kCameraRotate0 = 0,
+ kCameraRotate90 = 5,
+ kCameraRotate180 = 10,
+ kCameraRotate270 = 15
+};
+
+struct VideoCaptureCapability
+{
+ WebRtc_Word32 width;
+ WebRtc_Word32 height;
+ WebRtc_Word32 maxFPS;
+ WebRtc_Word32 expectedCaptureDelay;
+ RawVideoType rawType;
+ VideoCodecType codecType;
+ bool interlaced;
+
+ VideoCaptureCapability()
+ {
+ width = 0;
+ height = 0;
+ maxFPS = 0;
+ expectedCaptureDelay = 0;
+ rawType = kVideoUnknown;
+ codecType = kVideoCodecUnknown;
+ interlaced = false;
+ }
+ ;
+ bool operator!=(const VideoCaptureCapability &other) const
+ {
+ if (width != other.width)
+ return true;
+ if (height != other.height)
+ return true;
+ if (maxFPS != other.maxFPS)
+ return true;
+ if (rawType != other.rawType)
+ return true;
+ if (codecType != other.codecType)
+ return true;
+ if (interlaced != other.interlaced)
+ return true;
+ return false;
+ }
+ bool operator==(const VideoCaptureCapability &other) const
+ {
+ return !operator!=(other);
+ }
+};
+
+enum VideoCaptureAlarm
+{
+ Raised = 0,
+ Cleared = 1
+};
+
+// VideoFrameI420 doesn't take the ownership of the buffer.
+// It's mostly used to group the parameters for external capture.
+struct VideoFrameI420
+{
+ VideoFrameI420() {
+ y_plane = NULL;
+ u_plane = NULL;
+ v_plane = NULL;
+ y_pitch = 0;
+ u_pitch = 0;
+ v_pitch = 0;
+ width = 0;
+ height = 0;
+ }
+
+ unsigned char* y_plane;
+ unsigned char* u_plane;
+ unsigned char* v_plane;
+
+ int y_pitch;
+ int u_pitch;
+ int v_pitch;
+
+ unsigned short width;
+ unsigned short height;
+};
+
+/* External Capture interface. Returned by Create
+ and implemented by the capture module.
+ */
+class VideoCaptureExternal
+{
+public:
+ virtual WebRtc_Word32 IncomingFrame(WebRtc_UWord8* videoFrame,
+ WebRtc_Word32 videoFrameLength,
+ const VideoCaptureCapability& frameInfo,
+ WebRtc_Word64 captureTime = 0) = 0;
+ virtual WebRtc_Word32 IncomingFrameI420(const VideoFrameI420& video_frame,
+ WebRtc_Word64 captureTime = 0) = 0;
+protected:
+ ~VideoCaptureExternal() {}
+};
+
+// Callback class to be implemented by module user
+class VideoCaptureDataCallback
+{
+public:
+ virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
+ VideoFrame& videoFrame,
+ VideoCodecType codecType) = 0;
+ virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
+ const WebRtc_Word32 delay) = 0;
+protected:
+ virtual ~VideoCaptureDataCallback(){}
+};
+
+class VideoCaptureFeedBack
+{
+public:
+ virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
+ const WebRtc_UWord32 frameRate) = 0;
+ virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
+ const VideoCaptureAlarm alarm) = 0;
+protected:
+ virtual ~VideoCaptureFeedBack(){}
+};
+
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture_factory.h b/trunk/src/modules/video_capture/main/interface/video_capture_factory.h
new file mode 100644
index 0000000..6cb7a77
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/interface/video_capture_factory.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains interfaces used for creating the VideoCaptureModule
+// and DeviceInfo.
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
+
+#include "video_capture.h"
+
+namespace webrtc {
+
+class VideoCaptureFactory {
+ public:
+ // Create a video capture module object
+ // id - unique identifier of this video capture module object.
+ // deviceUniqueIdUTF8 - name of the device.
+ // Available names can be found by using GetDeviceName
+ static VideoCaptureModule* Create(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ // Create a video capture module object used for external capture.
+ // id - unique identifier of this video capture module object
+ // externalCapture - [out] interface to call when a new frame is captured.
+ static VideoCaptureModule* Create(const WebRtc_Word32 id,
+ VideoCaptureExternal*& externalCapture);
+
+ static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(
+ const WebRtc_Word32 id);
+
+#ifdef WEBRTC_ANDROID
+ static WebRtc_Word32 SetAndroidObjects(void* javaVM, void* javaContext);
+#endif
+
+ private:
+ ~VideoCaptureFactory();
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
diff --git a/trunk/src/modules/video_capture/main/source/Android.mk b/trunk/src/modules/video_capture/main/source/Android.mk
new file mode 100644
index 0000000..8976f19
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android.mk
@@ -0,0 +1,50 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_video_capture
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ video_capture_impl.cc \
+ device_info_impl.cc \
+ video_capture_factory.cc \
+ Android/video_capture_android.cc \
+ Android/device_info_android.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/Android \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../source \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../utility/interface \
+ $(LOCAL_PATH)/../../../audio_coding/main/interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc b/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc
new file mode 100644
index 0000000..3c5e052
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc
@@ -0,0 +1,363 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "device_info_android.h"
+
+#include <stdio.h>
+
+#include "ref_count.h"
+#include "trace.h"
+#include "video_capture_android.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo (
+ const WebRtc_Word32 id)
+{
+ videocapturemodule::DeviceInfoAndroid *deviceInfo =
+ new videocapturemodule::DeviceInfoAndroid(id);
+ if (deviceInfo && deviceInfo->Init() != 0) // Failed to init
+ {
+ delete deviceInfo;
+ deviceInfo = NULL;
+ }
+ return deviceInfo;
+}
+
+DeviceInfoAndroid::DeviceInfoAndroid(const WebRtc_Word32 id) :
+ DeviceInfoImpl(id)
+{
+}
+
+WebRtc_Word32 DeviceInfoAndroid::Init()
+{
+ return 0;
+}
+
+DeviceInfoAndroid::~DeviceInfoAndroid()
+{
+}
+
+WebRtc_UWord32 DeviceInfoAndroid::NumberOfDevices()
+{
+
+ JNIEnv *env;
+ jclass javaCmDevInfoClass;
+ jobject javaCmDevInfoObject;
+ bool attached = false;
+ if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+ env,
+ javaCmDevInfoClass,
+ javaCmDevInfoObject,
+ attached) != 0)
+ {
+ return 0;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s GetMethodId", __FUNCTION__);
+ // get the method ID for the Android Java GetDeviceUniqueName name.
+ jmethodID cid = env->GetMethodID(javaCmDevInfoClass,
+ "NumberOfDevices",
+ "()I");
+
+ jint numberOfDevices = 0;
+ if (cid != NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s Calling Number of devices", __FUNCTION__);
+ numberOfDevices = env->CallIntMethod(javaCmDevInfoObject, cid);
+ }
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+
+ if (numberOfDevices > 0)
+ return numberOfDevices;
+ return 0;
+}
+
+WebRtc_Word32 DeviceInfoAndroid::GetDeviceName(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* /*productUniqueIdUTF8*/,
+ WebRtc_UWord32 /*productUniqueIdUTF8Length*/)
+{
+
+ JNIEnv *env;
+ jclass javaCmDevInfoClass;
+ jobject javaCmDevInfoObject;
+ WebRtc_Word32 result = 0;
+ bool attached = false;
+ if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+ env,
+ javaCmDevInfoClass,
+ javaCmDevInfoObject,
+ attached)!= 0)
+ {
+ return -1;
+ }
+
+ // get the method ID for the Android Java GetDeviceUniqueName name.
+ jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetDeviceUniqueName",
+ "(I)Ljava/lang/String;");
+ if (cid != NULL)
+ {
+
+ jobject javaDeviceNameObj = env->CallObjectMethod(javaCmDevInfoObject,
+ cid, deviceNumber);
+ if (javaDeviceNameObj == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Failed to get device name for device %d.",
+ __FUNCTION__, (int) deviceNumber);
+ result = -1;
+ }
+ else
+ {
+ jboolean isCopy;
+ const char* javaDeviceNameChar = env->GetStringUTFChars(
+ (jstring) javaDeviceNameObj
+ ,&isCopy);
+ const jsize javaDeviceNameCharLength = env->GetStringUTFLength(
+ (jstring) javaDeviceNameObj);
+ if ((WebRtc_UWord32) javaDeviceNameCharLength < deviceUniqueIdUTF8Length)
+ {
+ memcpy(deviceUniqueIdUTF8,
+ javaDeviceNameChar,
+ javaDeviceNameCharLength + 1);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id, "%s: deviceUniqueIdUTF8 to short.",
+ __FUNCTION__);
+ result = -1;
+ }
+ if ((WebRtc_UWord32) javaDeviceNameCharLength < deviceNameLength)
+ {
+ memcpy(deviceNameUTF8,
+ javaDeviceNameChar,
+ javaDeviceNameCharLength + 1);
+ }
+ env->ReleaseStringUTFChars((jstring) javaDeviceNameObj,
+ javaDeviceNameChar);
+ }//javaDeviceNameObj==NULL
+
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Failed to find GetDeviceUniqueName function id",
+ __FUNCTION__);
+ result = -1;
+ }
+
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: result %d", __FUNCTION__, (int) result);
+ return result;
+
+}
+
+WebRtc_Word32 DeviceInfoAndroid::CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ MapItem* item = NULL;
+ while ((item = _captureCapabilities.Last()))
+ {
+ delete (VideoCaptureCapability*) item->GetItem();
+ _captureCapabilities.Erase(item);
+ }
+
+ JNIEnv *env;
+ jclass javaCmDevInfoClass;
+ jobject javaCmDevInfoObject;
+ bool attached = false;
+ if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+ env,
+ javaCmDevInfoClass,
+ javaCmDevInfoObject,
+ attached) != 0)
+ {
+ return -1;
+ }
+
+ // Find the capability class
+ jclass javaCapClassLocal = env->FindClass(AndroidJavaCaptureCapabilityClass);
+ if (javaCapClassLocal == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Can't find java class VideoCaptureCapabilityAndroid.",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // get the method ID for the Android Java GetCapabilityArray .
+ char signature[256];
+ sprintf(signature,
+ "(Ljava/lang/String;)[L%s;",
+ AndroidJavaCaptureCapabilityClass);
+ jmethodID cid = env->GetMethodID(javaCmDevInfoClass,
+ "GetCapabilityArray",
+ signature);
+ if (cid == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Can't find method GetCapabilityArray.", __FUNCTION__);
+ return -1;
+ }
+ // Create a jstring so we can pass the deviceUniquName to the java method.
+ jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
+
+ if (capureIdString == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Can't create string for method GetCapabilityArray.",
+ __FUNCTION__);
+ return -1;
+ }
+ // Call the java class and get an array with capabilities back.
+ jobject javaCapabilitiesObj = env->CallObjectMethod(javaCmDevInfoObject,
+ cid, capureIdString);
+ if (!javaCapabilitiesObj)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Failed to call java GetCapabilityArray.",
+ __FUNCTION__);
+ return -1;
+ }
+
+ jfieldID widthField = env->GetFieldID(javaCapClassLocal, "width", "I");
+ jfieldID heigtField = env->GetFieldID(javaCapClassLocal, "height", "I");
+ jfieldID maxFpsField = env->GetFieldID(javaCapClassLocal, "maxFPS", "I");
+ if (widthField == NULL || heigtField == NULL || maxFpsField == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Failed to get field Id.", __FUNCTION__);
+ return -1;
+ }
+
+ const jsize numberOfCapabilities =
+ env->GetArrayLength((jarray) javaCapabilitiesObj);
+
+ for (jsize i = 0; i < numberOfCapabilities; ++i)
+ {
+ VideoCaptureCapability *cap = new VideoCaptureCapability();
+ jobject capabilityElement = env->GetObjectArrayElement(
+ (jobjectArray) javaCapabilitiesObj,
+ i);
+
+ cap->width = env->GetIntField(capabilityElement, widthField);
+ cap->height = env->GetIntField(capabilityElement, heigtField);
+ cap->expectedCaptureDelay = _expectedCaptureDelay;
+ cap->rawType = kVideoNV21;
+ cap->maxFPS = env->GetIntField(capabilityElement, maxFpsField);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Cap width %d, height %d, fps %d", __FUNCTION__,
+ cap->width, cap->height, cap->maxFPS);
+ _captureCapabilities.Insert(i, cap);
+ }
+
+ _lastUsedDeviceNameLength = strlen((char*) deviceUniqueIdUTF8);
+ _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
+ _lastUsedDeviceNameLength + 1);
+ memcpy(_lastUsedDeviceName,
+ deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength + 1);
+
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "CreateCapabilityMap %d", _captureCapabilities.Size());
+
+ return _captureCapabilities.Size();
+}
+
+WebRtc_Word32 DeviceInfoAndroid::GetOrientation(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ VideoCaptureRotation& orientation)
+{
+
+ JNIEnv *env;
+ jclass javaCmDevInfoClass;
+ jobject javaCmDevInfoObject;
+ WebRtc_Word32 result = 0;
+ bool attached = false;
+ if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+ env,
+ javaCmDevInfoClass,
+ javaCmDevInfoObject,
+ attached) != 0)
+ {
+ return -1;
+ }
+
+ // get the method ID for the Android Java GetOrientation .
+ jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetOrientation",
+ "(Ljava/lang/String;)I");
+ if (cid == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Can't find method GetOrientation.", __FUNCTION__);
+ return -1;
+ }
+ // Create a jstring so we can pass the deviceUniquName to the java method.
+ jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
+ if (capureIdString == NULL)
+ {
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Can't create string for method GetCapabilityArray.",
+ __FUNCTION__);
+ return -1;
+ }
+ // Call the java class and get the orientation.
+ jint jorientation = env->CallIntMethod(javaCmDevInfoObject, cid,
+ capureIdString);
+ VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+
+ WebRtc_Word32 retValue = 0;
+ switch (jorientation)
+ {
+ case -1: //Error
+ orientation = kCameraRotate0;
+ retValue = -1;
+ break;
+ case 0:
+ orientation = kCameraRotate0;
+ break;
+ case 90:
+ orientation = kCameraRotate90;
+ break;
+ case 180:
+ orientation = kCameraRotate180;
+ break;
+ case 270:
+ orientation = kCameraRotate270;
+ break;
+ case 360:
+ orientation = kCameraRotate0;
+ break;
+ }
+ return retValue;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Android/device_info_android.h b/trunk/src/modules/video_capture/main/source/Android/device_info_android.h
new file mode 100644
index 0000000..33b22c3
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/device_info_android.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
+
+#include <jni.h>
+#include "video_capture_impl.h"
+#include "device_info_impl.h"
+
+#define AndroidJavaCaptureDeviceInfoClass "org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid"
+#define AndroidJavaCaptureCapabilityClass "org/webrtc/videoengine/CaptureCapabilityAndroid"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+// Android logging, uncomment to print trace to logcat instead of trace file/callback
+//#include <android/log.h>
+//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
+
+class DeviceInfoAndroid: public DeviceInfoImpl
+{
+public:
+
+ DeviceInfoAndroid(const WebRtc_Word32 id);
+ WebRtc_Word32 Init();
+ virtual ~DeviceInfoAndroid();
+ virtual WebRtc_UWord32 NumberOfDevices();
+ virtual WebRtc_Word32 GetDeviceName(WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8 = 0,
+ WebRtc_UWord32 productUniqueIdUTF8Length = 0);
+ virtual WebRtc_Word32 CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
+ const WebRtc_UWord8* /*dialogTitleUTF8*/,
+ void* /*parentWindow*/,
+ WebRtc_UWord32 /*positionX*/,
+ WebRtc_UWord32 /*positionY*/){return -1;}
+ virtual WebRtc_Word32 GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ VideoCaptureRotation& orientation);
+private:
+ bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
+ enum {_expectedCaptureDelay = 190};
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java
new file mode 100644
index 0000000..0443953
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+public class CaptureCapabilityAndroid {
+ public int width = 0;
+ public int height = 0;
+ public int maxFPS = 0;
+}
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java
new file mode 100644
index 0000000..a4c39a8
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java
@@ -0,0 +1,262 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+import java.io.IOException;
+import java.util.Locale;
+import java.util.concurrent.locks.ReentrantLock;
+
+import org.webrtc.videoengine.CaptureCapabilityAndroid;
+import org.webrtc.videoengine.VideoCaptureDeviceInfoAndroid.AndroidVideoCaptureDevice;
+
+import android.graphics.ImageFormat;
+import android.graphics.PixelFormat;
+import android.hardware.Camera;
+import android.hardware.Camera.PreviewCallback;
+import android.util.Log;
+import android.view.SurfaceHolder;
+import android.view.SurfaceHolder.Callback;
+
+public class VideoCaptureAndroid implements PreviewCallback, Callback {
+
+ private Camera camera;
+ private AndroidVideoCaptureDevice currentDevice = null;
+ public ReentrantLock previewBufferLock = new ReentrantLock();
+ private int PIXEL_FORMAT = ImageFormat.NV21;
+ PixelFormat pixelFormat = new PixelFormat();
+ // True when the C++ layer has ordered the camera to be started.
+ private boolean isRunning=false;
+
+ private final int numCaptureBuffers = 3;
+ private int expectedFrameSize = 0;
+ private int orientation = 0;
+ private int id = 0;
+ // C++ callback context variable.
+ private long context = 0;
+ private SurfaceHolder localPreview = null;
+ // True if this class owns the preview video buffers.
+ private boolean ownsBuffers = false;
+
+ // Set this to 2 for VERBOSE logging. 1 for DEBUG
+ private static int LOGLEVEL = 0;
+ private static boolean VERBOSE = LOGLEVEL > 2;
+ private static boolean DEBUG = LOGLEVEL > 1;
+
+ CaptureCapabilityAndroid currentCapability = null;
+
+ public static
+ void DeleteVideoCaptureAndroid(VideoCaptureAndroid captureAndroid) {
+ if(DEBUG) Log.d("*WEBRTC*", "DeleteVideoCaptureAndroid");
+
+ captureAndroid.StopCapture();
+ captureAndroid.camera.release();
+ captureAndroid.camera = null;
+ captureAndroid.context = 0;
+
+ if(DEBUG) Log.v("*WEBRTC*", "DeleteVideoCaptureAndroid ended");
+
+ }
+
+ public VideoCaptureAndroid(int in_id,
+ long in_context,
+ Camera in_camera,
+ AndroidVideoCaptureDevice in_device) {
+ id = in_id;
+ context = in_context;
+ camera = in_camera;
+ currentDevice = in_device;
+ }
+
+ public int StartCapture(int width, int height, int frameRate) {
+ if(DEBUG) Log.d("*WEBRTC*", "StartCapture width" + width +
+ " height " + height +" frame rate " + frameRate);
+ try {
+ if (camera == null) {
+ Log.e("*WEBRTC*",
+ String.format(Locale.US,"Camera not initialized %d",id));
+ return -1;
+ }
+ currentCapability = new CaptureCapabilityAndroid();
+ currentCapability.width = width;
+ currentCapability.height = height;
+ currentCapability.maxFPS = frameRate;
+ PixelFormat.getPixelFormatInfo(PIXEL_FORMAT, pixelFormat);
+
+ Camera.Parameters parameters = camera.getParameters();
+ parameters.setPreviewSize(currentCapability.width,
+ currentCapability.height);
+ parameters.setPreviewFormat(PIXEL_FORMAT );
+ parameters.setPreviewFrameRate(currentCapability.maxFPS);
+ camera.setParameters(parameters);
+
+ // Get the local preview SurfaceHolder from the static render class
+ localPreview = ViERenderer.GetLocalRenderer();
+ if(localPreview != null) {
+ localPreview.addCallback(this);
+ }
+
+ int bufSize = width * height * pixelFormat.bitsPerPixel / 8;
+ if(android.os.Build.VERSION.SDK_INT >= 7) {
+ // According to Doc addCallbackBuffer belongs to API level 8.
+ // But it seems like it works on Android 2.1 as well.
+ // At least SE X10 and Milestone
+ byte[] buffer = null;
+ for (int i = 0; i < numCaptureBuffers; i++) {
+ buffer = new byte[bufSize];
+ camera.addCallbackBuffer(buffer);
+ }
+
+ camera.setPreviewCallbackWithBuffer(this);
+ ownsBuffers = true;
+ }
+ else {
+ camera.setPreviewCallback(this);
+ }
+
+ camera.startPreview();
+ previewBufferLock.lock();
+ expectedFrameSize = bufSize;
+ isRunning = true;
+ previewBufferLock.unlock();
+ }
+ catch (Exception ex) {
+ Log.e("*WEBRTC*", "Failed to start camera");
+ return -1;
+ }
+ return 0;
+ }
+
+ public int StopCapture() {
+ if(DEBUG) Log.d("*WEBRTC*", "StopCapture");
+ try {
+ previewBufferLock.lock();
+ isRunning = false;
+ previewBufferLock.unlock();
+
+ camera.stopPreview();
+
+ if(android.os.Build.VERSION.SDK_INT > 7) {
+ camera.setPreviewCallbackWithBuffer(null);
+ }
+ else {
+ camera.setPreviewCallback(null);
+ }
+ }
+ catch (Exception ex) {
+ Log.e("*WEBRTC*", "Failed to stop camera");
+ return -1;
+ }
+
+ if(DEBUG) {
+ Log.d("*WEBRTC*", "StopCapture ended");
+ }
+ return 0;
+ }
+
+ native void ProvideCameraFrame(byte[] data,int length, long captureObject);
+
+ public void onPreviewFrame(byte[] data, Camera camera) {
+ previewBufferLock.lock();
+
+ if(VERBOSE) {
+ Log.v("*WEBRTC*",
+ String.format(Locale.US, "preview frame length %d context %x",
+ data.length, context));
+ }
+ if(isRunning) {
+ // If StartCapture has been called but not StopCapture
+ // Call the C++ layer with the captured frame
+ if (data.length == expectedFrameSize) {
+ ProvideCameraFrame(data, expectedFrameSize, context);
+ if (VERBOSE) {
+ Log.v("*WEBRTC*", String.format(Locale.US, "frame delivered"));
+ }
+ if(ownsBuffers) {
+ // Give the video buffer to the camera service again.
+ camera.addCallbackBuffer(data);
+ }
+ }
+ }
+ previewBufferLock.unlock();
+ }
+
+
+ public void surfaceChanged(SurfaceHolder holder,
+ int format, int width, int height) {
+
+ try {
+ if(camera != null) {
+ camera.setPreviewDisplay(localPreview);
+ }
+ } catch (IOException e) {
+ Log.e("*WEBRTC*",
+ String.format(Locale.US,
+ "Failed to set Local preview. " + e.getMessage()));
+ }
+ }
+
+ // Sets the rotation of the preview render window.
+ // Does not affect the captured video image.
+ public void SetPreviewRotation(int rotation) {
+ if(camera != null) {
+ previewBufferLock.lock();
+ final boolean running = isRunning;
+ int width = 0;
+ int height = 0;
+ int framerate = 0;
+
+ if(running) {
+ width = currentCapability.width;
+ height = currentCapability.height;
+ framerate = currentCapability.maxFPS;
+
+ StopCapture();
+ }
+
+ int resultRotation = 0;
+ if(currentDevice.frontCameraType ==
+ VideoCaptureDeviceInfoAndroid.FrontFacingCameraType.Android23) {
+ // this is a 2.3 or later front facing camera.
+ // SetDisplayOrientation will flip the image horizontally
+ // before doing the rotation.
+ resultRotation=(360-rotation) % 360; // compensate the mirror
+ }
+ else {
+ // Back facing or 2.2 or previous front camera
+ resultRotation=rotation;
+ }
+ if(android.os.Build.VERSION.SDK_INT>7) {
+ camera.setDisplayOrientation(resultRotation);
+ }
+ else {
+ // Android 2.1 and previous
+ // This rotation unfortunately does not seems to work.
+ // http://code.google.com/p/android/issues/detail?id=1193
+ Camera.Parameters parameters = camera.getParameters();
+ parameters.setRotation(resultRotation);
+ camera.setParameters(parameters);
+ }
+
+ if(running) {
+ StartCapture(width, height, framerate);
+ }
+ previewBufferLock.unlock();
+ }
+ }
+
+ public void surfaceCreated(SurfaceHolder holder) {
+ }
+
+
+ public void surfaceDestroyed(SurfaceHolder holder) {
+ }
+
+}
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
new file mode 100644
index 0000000..4ccf060
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
@@ -0,0 +1,432 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+import java.io.File;
+import java.lang.reflect.InvocationTargetException;
+import java.lang.reflect.Method;
+import java.util.ArrayList;
+import java.util.List;
+import java.util.Locale;
+
+import dalvik.system.DexClassLoader;
+
+import android.content.Context;
+import android.hardware.Camera;
+import android.hardware.Camera.Size;
+import android.util.Log;
+
+public class VideoCaptureDeviceInfoAndroid {
+
+ //Context
+ Context context;
+
+ // Set this to 2 for VERBOSE logging. 1 for DEBUG
+ private static int LOGLEVEL = 0;
+ private static boolean VERBOSE = LOGLEVEL > 2;
+ private static boolean DEBUG = LOGLEVEL > 1;
+
+ // Private class with info about all available cameras and the capabilities
+ public class AndroidVideoCaptureDevice {
+ AndroidVideoCaptureDevice() {
+ frontCameraType = FrontFacingCameraType.None;
+ index = 0;
+ }
+
+ public String deviceUniqueName;
+ public CaptureCapabilityAndroid captureCapabilies[];
+ public FrontFacingCameraType frontCameraType;
+
+ // Orientation of camera as described in
+ // android.hardware.Camera.CameraInfo.Orientation
+ public int orientation;
+ // Camera index used in Camera.Open on Android 2.3 and onwards
+ public int index;
+ }
+
+ public enum FrontFacingCameraType {
+ None, // This is not a front facing camera
+ GalaxyS, // Galaxy S front facing camera.
+ HTCEvo, // HTC Evo front facing camera
+ Android23, // Android 2.3 front facing camera.
+ }
+
+ String currentDeviceUniqueId;
+ int id;
+ List<AndroidVideoCaptureDevice> deviceList;
+
+ public static VideoCaptureDeviceInfoAndroid
+ CreateVideoCaptureDeviceInfoAndroid(int in_id, Context in_context) {
+ if(DEBUG) {
+ Log.d("*WEBRTC*",
+ String.format(Locale.US, "VideoCaptureDeviceInfoAndroid"));
+ }
+
+ VideoCaptureDeviceInfoAndroid self =
+ new VideoCaptureDeviceInfoAndroid(in_id, in_context);
+ if(self != null && self.Init() == 0) {
+ return self;
+ }
+ else {
+ if(DEBUG) {
+ Log.d("*WEBRTC*", "Failed to create VideoCaptureDeviceInfoAndroid.");
+ }
+ }
+ return null;
+ }
+
+ private VideoCaptureDeviceInfoAndroid(int in_id,
+ Context in_context) {
+ id = in_id;
+ context = in_context;
+ deviceList = new ArrayList<AndroidVideoCaptureDevice>();
+ }
+
+ private int Init() {
+ // Populate the deviceList with available cameras and their capabilities.
+ Camera camera = null;
+ try{
+ if(android.os.Build.VERSION.SDK_INT > 8) {
+ // From Android 2.3 and onwards
+ for(int i = 0; i < Camera.getNumberOfCameras(); ++i) {
+ AndroidVideoCaptureDevice newDevice = new AndroidVideoCaptureDevice();
+
+ Camera.CameraInfo info = new Camera.CameraInfo();
+ Camera.getCameraInfo(i, info);
+ newDevice.index = i;
+ newDevice.orientation=info.orientation;
+ if(info.facing == Camera.CameraInfo.CAMERA_FACING_BACK) {
+ newDevice.deviceUniqueName =
+ "Camera " + i +", Facing back, Orientation "+ info.orientation;
+ }
+ else {
+ newDevice.deviceUniqueName =
+ "Camera " + i +", Facing front, Orientation "+ info.orientation;
+ newDevice.frontCameraType = FrontFacingCameraType.Android23;
+ }
+
+ camera = Camera.open(i);
+ Camera.Parameters parameters = camera.getParameters();
+ AddDeviceInfo(newDevice, parameters);
+ camera.release();
+ camera = null;
+ deviceList.add(newDevice);
+ }
+ }
+ else {
+ // Prior to Android 2.3
+ AndroidVideoCaptureDevice newDevice;
+ Camera.Parameters parameters;
+
+ newDevice = new AndroidVideoCaptureDevice();
+ camera = Camera.open();
+ parameters = camera.getParameters();
+ newDevice.deviceUniqueName = "Camera 1, Facing back";
+ newDevice.orientation = 90;
+ AddDeviceInfo(newDevice, parameters);
+
+ deviceList.add(newDevice);
+ camera.release();
+ camera=null;
+
+ newDevice = new AndroidVideoCaptureDevice();
+ newDevice.deviceUniqueName = "Camera 2, Facing front";
+ parameters = SearchOldFrontFacingCameras(newDevice);
+ if(parameters != null) {
+ AddDeviceInfo(newDevice, parameters);
+ deviceList.add(newDevice);
+ }
+ }
+ }
+ catch (Exception ex) {
+ Log.e("*WEBRTC*", "Failed to init VideoCaptureDeviceInfo ex" +
+ ex.getLocalizedMessage());
+ return -1;
+ }
+ VerifyCapabilities();
+ return 0;
+ }
+
+ // Adds the capture capabilities of the currently opened device
+ private void AddDeviceInfo(AndroidVideoCaptureDevice newDevice,
+ Camera.Parameters parameters) {
+
+ List<Size> sizes = parameters.getSupportedPreviewSizes();
+ List<Integer> frameRates = parameters.getSupportedPreviewFrameRates();
+ int maxFPS=0;
+ for(Integer frameRate:frameRates) {
+ if(VERBOSE) {
+ Log.v("*WEBRTC*",
+ "VideoCaptureDeviceInfoAndroid:frameRate " + frameRate);
+ }
+ if(frameRate > maxFPS) {
+ maxFPS = frameRate;
+ }
+ }
+
+ newDevice.captureCapabilies = new CaptureCapabilityAndroid[sizes.size()];
+ for(int i = 0; i < sizes.size(); ++i) {
+ Size s = sizes.get(i);
+ newDevice.captureCapabilies[i] = new CaptureCapabilityAndroid();
+ newDevice.captureCapabilies[i].height = s.height;
+ newDevice.captureCapabilies[i].width = s.width;
+ newDevice.captureCapabilies[i].maxFPS = maxFPS;
+ }
+ }
+
+ // Function that make sure device specific capabilities are
+ // in the capability list.
+ // Ie Galaxy S supports CIF but does not list CIF as a supported capability.
+ // Motorola Droid Camera does not work with frame rate above 15fps.
+ // http://code.google.com/p/android/issues/detail?id=5514#c0
+ private void VerifyCapabilities() {
+ // Nexus S or Galaxy S
+ if(android.os.Build.DEVICE.equals("GT-I9000") ||
+ android.os.Build.DEVICE.equals("crespo")) {
+ CaptureCapabilityAndroid specificCapability =
+ new CaptureCapabilityAndroid();
+ specificCapability.width = 352;
+ specificCapability.height = 288;
+ specificCapability.maxFPS = 15;
+ AddDeviceSpecificCapability(specificCapability);
+
+ specificCapability = new CaptureCapabilityAndroid();
+ specificCapability.width = 176;
+ specificCapability.height = 144;
+ specificCapability.maxFPS = 15;
+ AddDeviceSpecificCapability(specificCapability);
+
+ specificCapability = new CaptureCapabilityAndroid();
+ specificCapability.width = 320;
+ specificCapability.height = 240;
+ specificCapability.maxFPS = 15;
+ AddDeviceSpecificCapability(specificCapability);
+ }
+ // Motorola Milestone Camera server does not work at 30fps
+ // even though it reports that it can
+ if(android.os.Build.MANUFACTURER.equals("motorola") &&
+ android.os.Build.DEVICE.equals("umts_sholes")) {
+ for(AndroidVideoCaptureDevice device:deviceList) {
+ for(CaptureCapabilityAndroid capability:device.captureCapabilies) {
+ capability.maxFPS=15;
+ }
+ }
+ }
+ }
+
+ private void AddDeviceSpecificCapability(
+ CaptureCapabilityAndroid specificCapability) {
+ for(AndroidVideoCaptureDevice device:deviceList) {
+ boolean foundCapability = false;
+ for(CaptureCapabilityAndroid capability:device.captureCapabilies) {
+ if(capability.width == specificCapability.width &&
+ capability.height == specificCapability.height) {
+ foundCapability = true;
+ break;
+ }
+ }
+ if(foundCapability==false) {
+ CaptureCapabilityAndroid newCaptureCapabilies[]=
+ new CaptureCapabilityAndroid[device.captureCapabilies.length+1];
+ for(int i = 0; i < device.captureCapabilies.length; ++i) {
+ newCaptureCapabilies[i+1] = device.captureCapabilies[i];
+ }
+ newCaptureCapabilies[0] = specificCapability;
+ device.captureCapabilies = newCaptureCapabilies;
+ }
+ }
+ }
+
+ // Returns the number of Capture devices that is supported
+ public int NumberOfDevices() {
+ return deviceList.size();
+ }
+
+ public String GetDeviceUniqueName(int deviceNumber) {
+ if(deviceNumber < 0 || deviceNumber >= deviceList.size()) {
+ return null;
+ }
+ return deviceList.get(deviceNumber).deviceUniqueName;
+ }
+
+ public CaptureCapabilityAndroid[] GetCapabilityArray (String deviceUniqueId)
+ {
+ for (AndroidVideoCaptureDevice device: deviceList) {
+ if(device.deviceUniqueName.equals(deviceUniqueId)) {
+ return (CaptureCapabilityAndroid[]) device.captureCapabilies;
+ }
+ }
+ return null;
+ }
+
+ // Returns the camera orientation as described by
+ // android.hardware.Camera.CameraInfo.orientation
+ public int GetOrientation(String deviceUniqueId) {
+ for (AndroidVideoCaptureDevice device: deviceList) {
+ if(device.deviceUniqueName.equals(deviceUniqueId)) {
+ return device.orientation;
+ }
+ }
+ return -1;
+ }
+
+ // Returns an instance of VideoCaptureAndroid.
+ public VideoCaptureAndroid AllocateCamera(int id, long context,
+ String deviceUniqueId) {
+ try {
+ if(DEBUG) Log.d("*WEBRTC*", "AllocateCamera " + deviceUniqueId);
+
+ Camera camera = null;
+ AndroidVideoCaptureDevice deviceToUse = null;
+ for (AndroidVideoCaptureDevice device: deviceList) {
+ if(device.deviceUniqueName.equals(deviceUniqueId)) {
+ // Found the wanted camera
+ deviceToUse = device;
+ switch(device.frontCameraType) {
+ case GalaxyS:
+ camera = AllocateGalaxySFrontCamera();
+ break;
+ case HTCEvo:
+ camera = AllocateEVOFrontFacingCamera();
+ break;
+ default:
+ // From Android 2.3 and onwards)
+ if(android.os.Build.VERSION.SDK_INT>8)
+ camera=Camera.open(device.index);
+ else
+ camera=Camera.open(); // Default camera
+ }
+ }
+ }
+
+ if(camera == null) {
+ return null;
+ }
+ if(VERBOSE) {
+ Log.v("*WEBRTC*", "AllocateCamera - creating VideoCaptureAndroid");
+ }
+
+ return new VideoCaptureAndroid(id,context,camera,deviceToUse);
+
+ }catch (Exception ex) {
+ Log.e("*WEBRTC*", "AllocateCamera Failed to open camera- ex " +
+ ex.getLocalizedMessage());
+ }
+ return null;
+ }
+
+ // Searches for a front facing camera device. This is device specific code.
+ private Camera.Parameters
+ SearchOldFrontFacingCameras(AndroidVideoCaptureDevice newDevice)
+ throws SecurityException, IllegalArgumentException,
+ NoSuchMethodException, ClassNotFoundException,
+ IllegalAccessException, InvocationTargetException {
+ // Check the id of the opened camera device
+ // Returns null on X10 and 1 on Samsung Galaxy S.
+ Camera camera = Camera.open();
+ Camera.Parameters parameters = camera.getParameters();
+ String cameraId = parameters.get("camera-id");
+ if(cameraId != null && cameraId.equals("1")) {
+ // This might be a Samsung Galaxy S with a front facing camera.
+ try {
+ parameters.set("camera-id", 2);
+ camera.setParameters(parameters);
+ parameters = camera.getParameters();
+ newDevice.frontCameraType = FrontFacingCameraType.GalaxyS;
+ newDevice.orientation = 0;
+ camera.release();
+ return parameters;
+ }
+ catch (Exception ex) {
+ //Nope - it did not work.
+ Log.e("*WEBRTC*", "Init Failed to open front camera camera - ex " +
+ ex.getLocalizedMessage());
+ }
+ }
+ camera.release();
+
+ //Check for Evo front facing camera
+ File file =
+ new File("/system/framework/com.htc.hardware.twinCamDevice.jar");
+ boolean exists = file.exists();
+ if (!exists){
+ file =
+ new File("/system/framework/com.sprint.hardware.twinCamDevice.jar");
+ exists = file.exists();
+ }
+ if(exists) {
+ newDevice.frontCameraType = FrontFacingCameraType.HTCEvo;
+ newDevice.orientation = 0;
+ Camera evCamera = AllocateEVOFrontFacingCamera();
+ parameters = evCamera.getParameters();
+ evCamera.release();
+ return parameters;
+ }
+ return null;
+ }
+
+ // Returns a handle to HTC front facing camera.
+ // The caller is responsible to release it on completion.
+ private Camera AllocateEVOFrontFacingCamera()
+ throws SecurityException, NoSuchMethodException,
+ ClassNotFoundException, IllegalArgumentException,
+ IllegalAccessException, InvocationTargetException {
+ String classPath = null;
+ File file =
+ new File("/system/framework/com.htc.hardware.twinCamDevice.jar");
+ classPath = "com.htc.hardware.twinCamDevice.FrontFacingCamera";
+ boolean exists = file.exists();
+ if (!exists){
+ file =
+ new File("/system/framework/com.sprint.hardware.twinCamDevice.jar");
+ classPath = "com.sprint.hardware.twinCamDevice.FrontFacingCamera";
+ exists = file.exists();
+ }
+ if(!exists) {
+ return null;
+ }
+
+ String dexOutputDir = "";
+ if(context != null) {
+ dexOutputDir = context.getFilesDir().getAbsolutePath();
+ File mFilesDir = new File(dexOutputDir, "dexfiles");
+ if(!mFilesDir.exists()){
+ //Log.e("*WEBRTCN*", "Directory doesn't exists");
+ if(!mFilesDir.mkdirs()) {
+ //Log.e("*WEBRTCN*", "Unable to create files directory");
+ }
+ }
+ }
+
+ dexOutputDir += "/dexfiles";
+
+ DexClassLoader loader =
+ new DexClassLoader(file.getAbsolutePath(), dexOutputDir,
+ null, ClassLoader.getSystemClassLoader());
+
+ Method method = loader.loadClass(classPath).getDeclaredMethod(
+ "getFrontFacingCamera", (Class[]) null);
+ Camera camera = (Camera) method.invoke((Object[])null,(Object[]) null);
+ return camera;
+ }
+
+ // Returns a handle to Galaxy S front camera.
+ // The caller is responsible to release it on completion.
+ private Camera AllocateGalaxySFrontCamera()
+ {
+ Camera camera = Camera.open();
+ Camera.Parameters parameters = camera.getParameters();
+ parameters.set("camera-id",2);
+ camera.setParameters(parameters);
+ return camera;
+ }
+
+}
diff --git a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc
new file mode 100644
index 0000000..acf7e3b
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc
@@ -0,0 +1,703 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_capture_android.h"
+
+#include <stdio.h>
+
+#include "critical_section_wrapper.h"
+#include "ref_count.h"
+#include "trace.h"
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule* VideoCaptureImpl::Create(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ RefCountImpl<videocapturemodule::VideoCaptureAndroid>* implementation =
+ new RefCountImpl<videocapturemodule::VideoCaptureAndroid>(id);
+
+ if (!implementation || implementation->Init(id, deviceUniqueIdUTF8) != 0)
+ {
+ delete implementation;
+ implementation = NULL;
+ }
+ return implementation;
+}
+
+// Android logging, uncomment to print trace to logcat instead of trace file/callback
+//#include <android/log.h>
+//#undef WEBRTC_TRACE
+//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
+
+JavaVM* VideoCaptureAndroid::g_jvm = NULL;
+jclass VideoCaptureAndroid::g_javaCmClass = NULL; //VideoCaptureAndroid.java
+jclass VideoCaptureAndroid::g_javaCmDevInfoClass = NULL; //VideoCaptureDeviceInfoAndroid.java
+jobject VideoCaptureAndroid::g_javaCmDevInfoObject = NULL; //static instance of VideoCaptureDeviceInfoAndroid.java
+jobject VideoCaptureAndroid::g_javaContext = NULL;
+
+/*
+ * Register references to Java Capture class.
+ */
+WebRtc_Word32 VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
+ void* javaContext)
+{
+
+ g_jvm = static_cast<JavaVM*> (javaVM);
+ g_javaContext = static_cast<jobject> (javaContext);
+
+ if (javaVM)
+ {
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: could not get Java environment", __FUNCTION__);
+ return -1;
+ }
+ // get java capture class type (note path to class packet)
+ jclass javaCmClassLocal = env->FindClass(AndroidJavaCaptureClass);
+ if (!javaCmClassLocal)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: could not find java class", __FUNCTION__);
+ return -1;
+ }
+ // create a global reference to the class (to tell JNI that we are referencing it
+ // after this function has returned)
+ g_javaCmClass = static_cast<jclass>
+ (env->NewGlobalRef(javaCmClassLocal));
+ if (!g_javaCmClass)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: InitVideoEngineJava(): could not create"
+ " Java Camera class reference",
+ __FUNCTION__);
+ return -1;
+ }
+ // Delete local class ref, we only use the global ref
+ env->DeleteLocalRef(javaCmClassLocal);
+ JNINativeMethod nativeFunctions = { "ProvideCameraFrame", "([BIJ)V",
+ (void*) &VideoCaptureAndroid::ProvideCameraFrame };
+ if (env->RegisterNatives(g_javaCmClass, &nativeFunctions, 1) == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: Registered native functions", __FUNCTION__);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Failed to register native functions",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // get java capture class type (note path to class packet)
+ jclass javaCmDevInfoClassLocal = env->FindClass(
+ AndroidJavaCaptureDeviceInfoClass);
+ if (!javaCmDevInfoClassLocal)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: could not find java class", __FUNCTION__);
+ return -1;
+ }
+
+ // create a global reference to the class (to tell JNI that we are referencing it
+ // after this function has returned)
+ g_javaCmDevInfoClass = static_cast<jclass>
+ (env->NewGlobalRef(javaCmDevInfoClassLocal));
+ if (!g_javaCmDevInfoClass)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: InitVideoEngineJava(): could not create Java "
+ "Camera Device info class reference",
+ __FUNCTION__);
+ return -1;
+ }
+ // Delete local class ref, we only use the global ref
+ env->DeleteLocalRef(javaCmDevInfoClassLocal);
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "VideoCaptureDeviceInfoAndroid get method id");
+
+ // get the method ID for the Android Java CaptureClass static
+ //CreateVideoCaptureAndroid factory method.
+ jmethodID cid = env->GetStaticMethodID(g_javaCmDevInfoClass,
+ "CreateVideoCaptureDeviceInfoAndroid",
+ "(ILandroid/content/Context;)"
+ "Lorg/webrtc/videoengine/VideoCaptureDeviceInfoAndroid;");
+ if (cid == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: could not get java VideoCaptureDeviceInfoAndroid constructor ID",
+ __FUNCTION__);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: construct static java device object", __FUNCTION__);
+
+ // construct the object by calling the static constructor object
+ jobject javaCameraDeviceInfoObjLocal = env->CallStaticObjectMethod(
+ g_javaCmDevInfoClass,
+ cid, (int) -1,
+ g_javaContext);
+ if (!javaCameraDeviceInfoObjLocal)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
+ "%s: could not create Java Capture Device info object",
+ __FUNCTION__);
+ return -1;
+ }
+ // create a reference to the object (to tell JNI that we are referencing it
+ // after this function has returned)
+ g_javaCmDevInfoObject = env->NewGlobalRef(javaCameraDeviceInfoObjLocal);
+ if (!g_javaCmDevInfoObject)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioDevice, -1,
+ "%s: could not create Java cameradevinceinfo object reference",
+ __FUNCTION__);
+ return -1;
+ }
+ // Delete local object ref, we only use the global ref
+ env->DeleteLocalRef(javaCameraDeviceInfoObjLocal);
+ return 0;
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: JVM is NULL, assuming deinit", __FUNCTION__);
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: SetAndroidObjects not called with a valid JVM.",
+ __FUNCTION__);
+ return -1;
+ }
+ JNIEnv* env = NULL;
+ bool attached = false;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ -1, "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ attached = true;
+ }
+ env->DeleteGlobalRef(g_javaCmDevInfoObject);
+ env->DeleteGlobalRef(g_javaCmDevInfoClass);
+ env->DeleteGlobalRef(g_javaCmClass);
+ if (attached && g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ return -1;
+ }
+ return 0;
+ env = (JNIEnv *) NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+ JNIEnv*& env,
+ jclass& javaCmDevInfoClass,
+ jobject& javaCmDevInfoObject,
+ bool& attached)
+{
+ // get the JNI env for this thread
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: SetAndroidObjects not called with a valid JVM.",
+ __FUNCTION__);
+ return -1;
+ }
+ attached = false;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ attached = true;
+ }
+ javaCmDevInfoClass = g_javaCmDevInfoClass;
+ javaCmDevInfoObject = g_javaCmDevInfoObject;
+ return 0;
+
+}
+
+WebRtc_Word32 VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(bool attached)
+{
+ if (attached && g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ return -1;
+ }
+ return 0;
+}
+
+/*
+ * JNI callback from Java class. Called when the camera has a new frame to deliver
+ * Class: org_webrtc_capturemodule_VideoCaptureAndroid
+ * Method: ProvideCameraFrame
+ * Signature: ([BIJ)V
+ */
+void JNICALL VideoCaptureAndroid::ProvideCameraFrame(JNIEnv * env,
+ jobject,
+ jbyteArray javaCameraFrame,
+ jint length,
+ jlong context)
+{
+ VideoCaptureAndroid* captureModule=reinterpret_cast<VideoCaptureAndroid*>(context);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture,
+ -1, "%s: IncomingFrame %d", __FUNCTION__,length);
+ jbyte* cameraFrame= env->GetByteArrayElements(javaCameraFrame,NULL);
+ captureModule->IncomingFrame((WebRtc_UWord8*) cameraFrame,
+ length,captureModule->_frameInfo,0);
+ env->ReleaseByteArrayElements(javaCameraFrame,cameraFrame,JNI_ABORT);
+}
+
+
+
+VideoCaptureAndroid::VideoCaptureAndroid(const WebRtc_Word32 id)
+ : VideoCaptureImpl(id), _capInfo(id), _javaCaptureObj(NULL),
+ _captureStarted(false)
+
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: context %x", __FUNCTION__, (int) this);
+}
+// ----------------------------------------------------------------------------
+// Init
+//
+// Initializes needed Java resources like the JNI interface to
+// VideoCaptureAndroid.java
+// ----------------------------------------------------------------------------
+WebRtc_Word32 VideoCaptureAndroid::Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ const int nameLength = strlen((char*) deviceUniqueIdUTF8);
+ if (nameLength >= kVideoCaptureUniqueNameLength)
+ {
+ return -1;
+ }
+
+ // Store the device name
+ _deviceUniqueId = new WebRtc_UWord8[nameLength + 1];
+ memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
+
+ if (_capInfo.Init() != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Failed to initialize CaptureDeviceInfo", __FUNCTION__);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1, "%s:",
+ __FUNCTION__);
+ // use the jvm that has been set
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Not a valid Java VM pointer", __FUNCTION__);
+ return -1;
+ }
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "get method id");
+
+ // get the method ID for the Android Java CaptureDeviceInfoClass AllocateCamera factory method.
+ char signature[256];
+ sprintf(signature, "(IJLjava/lang/String;)L%s;", AndroidJavaCaptureClass);
+
+ jmethodID cid = env->GetMethodID(g_javaCmDevInfoClass, "AllocateCamera",
+ signature);
+ if (cid == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: could not get constructor ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
+ // construct the object by calling the static constructor object
+ jobject javaCameraObjLocal = env->CallObjectMethod(g_javaCmDevInfoObject,
+ cid, (jint) id,
+ (jlong) this,
+ capureIdString);
+ if (!javaCameraObjLocal)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s: could not create Java Capture object", __FUNCTION__);
+ return -1;
+ }
+
+ // create a reference to the object (to tell JNI that we are referencing it
+ // after this function has returned)
+ _javaCaptureObj = env->NewGlobalRef(javaCameraObjLocal);
+ if (!_javaCaptureObj)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioDevice, _id,
+ "%s: could not create Java camera object reference",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local object ref, we only use the global ref
+ env->DeleteLocalRef(javaCameraObjLocal);
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ return 0;
+}
+
+VideoCaptureAndroid::~VideoCaptureAndroid()
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1, "%s:",
+ __FUNCTION__);
+ if (_javaCaptureObj == NULL || g_jvm == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Nothing to clean", __FUNCTION__);
+ }
+ else
+ {
+ bool isAttached = false;
+ // get the JNI env for this thread
+ JNIEnv *env;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+
+ // get the method ID for the Android Java CaptureClass static
+ // DeleteVideoCaptureAndroid method. Call this to release the camera so
+ // another application can use it.
+ jmethodID cid = env->GetStaticMethodID(g_javaCmClass,
+ "DeleteVideoCaptureAndroid",
+ "(Lorg/webrtc/videoengine/VideoCaptureAndroid;)V");
+ if (cid != NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: Call DeleteVideoCaptureAndroid", __FUNCTION__);
+ // Close the camera by calling the static destruct function.
+ env->CallStaticVoidMethod(g_javaCmClass, cid, _javaCaptureObj);
+
+ // Delete global object ref to the camera.
+ env->DeleteGlobalRef(_javaCaptureObj);
+ _javaCaptureObj = NULL;
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Failed to find DeleteVideoCaptureAndroid id",
+ __FUNCTION__);
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
+ _id, "%s: Could not detach thread from JVM",
+ __FUNCTION__);
+ }
+ }
+ }
+}
+
+WebRtc_Word32 VideoCaptureAndroid::StartCapture(
+ const VideoCaptureCapability& capability)
+{
+ CriticalSectionScoped cs(_apiCs);
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: ", __FUNCTION__);
+
+ bool isAttached = false;
+ WebRtc_Word32 result = 0;
+ // get the JNI env for this thread
+ JNIEnv *env;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+
+ if (_capInfo.GetBestMatchedCapability(_deviceUniqueId, capability,
+ _frameInfo) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: GetBestMatchedCapability failed. Req cap w%d h%d",
+ __FUNCTION__, capability.width, capability.height);
+ return -1;
+ }
+
+ // Store the new expected capture delay
+ _captureDelay = _frameInfo.expectedCaptureDelay;
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: _frameInfo w%d h%d", __FUNCTION__, _frameInfo.width,
+ _frameInfo.height);
+
+ // get the method ID for the Android Java CaptureClass static StartCapture method.
+ jmethodID cid = env->GetMethodID(g_javaCmClass, "StartCapture", "(III)I");
+ if (cid != NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: Call StartCapture", __FUNCTION__);
+ // Close the camera by calling the static destruct function.
+ result = env->CallIntMethod(_javaCaptureObj, cid, _frameInfo.width,
+ _frameInfo.height, _frameInfo.maxFPS);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Failed to find StartCapture id", __FUNCTION__);
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+ if (result == 0)
+ {
+ _requestedCapability = capability;
+ _captureStarted = true;
+ }
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: result %d", __FUNCTION__, result);
+ return result;
+}
+WebRtc_Word32 VideoCaptureAndroid::StopCapture()
+{
+ CriticalSectionScoped cs(_apiCs);
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: ", __FUNCTION__);
+
+ bool isAttached = false;
+ WebRtc_Word32 result = 0;
+ // get the JNI env for this thread
+ JNIEnv *env = NULL;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+
+ memset(&_requestedCapability, 0, sizeof(_requestedCapability));
+ memset(&_frameInfo, 0, sizeof(_frameInfo));
+
+ // get the method ID for the Android Java CaptureClass StopCapture method.
+ jmethodID cid = env->GetMethodID(g_javaCmClass, "StopCapture", "()I");
+ if (cid != NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+ "%s: Call StopCapture", __FUNCTION__);
+ // Close the camera by calling the static destruct function.
+ result = env->CallIntMethod(_javaCaptureObj, cid);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: Failed to find StopCapture id", __FUNCTION__);
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+ _captureStarted = false;
+
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: result %d", __FUNCTION__, result);
+ return result;
+}
+
+bool VideoCaptureAndroid::CaptureStarted()
+{
+ CriticalSectionScoped cs(_apiCs);
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: ", __FUNCTION__);
+ return _captureStarted;
+}
+WebRtc_Word32 VideoCaptureAndroid::CaptureSettings(
+ VideoCaptureCapability& settings)
+{
+ CriticalSectionScoped cs(_apiCs);
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+ "%s: ", __FUNCTION__);
+ settings = _requestedCapability;
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureAndroid::SetCaptureRotation(
+ VideoCaptureRotation rotation)
+{
+ CriticalSectionScoped cs(_apiCs);
+ if (VideoCaptureImpl::SetCaptureRotation(rotation) == 0)
+ {
+ if (!g_jvm)
+ return -1;
+
+ // get the JNI env for this thread
+ JNIEnv *env;
+ bool isAttached = false;
+
+ // get the JNI env for this thread
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ jmethodID cid = env->GetMethodID(g_javaCmClass, "SetPreviewRotation",
+ "(I)V");
+ if (cid == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "%s: could not get java SetPreviewRotation ID",
+ __FUNCTION__);
+ return -1;
+ }
+ jint rotateFrame = 0;
+ switch (rotation)
+ {
+ case kCameraRotate0:
+ rotateFrame = 0;
+ break;
+ case kCameraRotate90:
+ rotateFrame = 90;
+ break;
+ case kCameraRotate180:
+ rotateFrame = 180;
+ break;
+ case kCameraRotate270:
+ rotateFrame = 270;
+ break;
+ }
+ env->CallVoidMethod(_javaCaptureObj, cid, rotateFrame);
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
+ _id, "%s: Could not detach thread from JVM",
+ __FUNCTION__);
+ }
+ }
+
+ }
+ return 0;
+}
+} //namespace videocapturemodule
+} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h
new file mode 100644
index 0000000..abb4dc9
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
+
+#include <jni.h>
+#include "device_info_android.h"
+#include "video_capture_impl.h"
+
+#define AndroidJavaCaptureClass "org/webrtc/videoengine/VideoCaptureAndroid"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+class VideoCaptureAndroid: public VideoCaptureImpl
+{
+public:
+ static WebRtc_Word32 SetAndroidObjects(void* javaVM, void* javaContext);
+ static WebRtc_Word32 AttachAndUseAndroidDeviceInfoObjects(JNIEnv*& env,
+ jclass& javaCmDevInfoClass,
+ jobject& javaCmDevInfoObject,
+ bool& attached);
+ static WebRtc_Word32 ReleaseAndroidDeviceInfoObjects(bool attached);
+
+ VideoCaptureAndroid(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+
+ virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability);
+ virtual WebRtc_Word32 StopCapture();
+ virtual bool CaptureStarted();
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
+ virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation);
+
+protected:
+ virtual ~VideoCaptureAndroid();
+ static void JNICALL ProvideCameraFrame (JNIEnv * env,
+ jobject,
+ jbyteArray javaCameraFrame,
+ jint length, jlong context);
+ DeviceInfoAndroid _capInfo;
+ jobject _javaCaptureObj; // Java Camera object.
+ VideoCaptureCapability _frameInfo;
+ bool _captureStarted;
+
+ static JavaVM* g_jvm;
+ static jclass g_javaCmClass;
+ static jclass g_javaCmDevInfoClass;
+ static jobject g_javaCmDevInfoObject; //Static java object implementing the needed device info functions;
+ static jobject g_javaContext; // Java Application context
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
diff --git a/trunk/src/modules/video_capture/main/source/External/device_info_external.cc b/trunk/src/modules/video_capture/main/source/External/device_info_external.cc
new file mode 100644
index 0000000..2aad081
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/External/device_info_external.cc
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../device_info_impl.h"
+#include "../video_capture_impl.h"
+
+namespace webrtc {
+
+namespace videocapturemodule {
+
+class ExternalDeviceInfo : public DeviceInfoImpl {
+ public:
+ ExternalDeviceInfo(const WebRtc_Word32 id)
+ : DeviceInfoImpl(id) {
+ }
+ virtual ~ExternalDeviceInfo() {}
+ virtual WebRtc_UWord32 NumberOfDevices() { return 0; }
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
+ const WebRtc_UWord8* /*dialogTitleUTF8*/,
+ void* /*parentWindow*/,
+ WebRtc_UWord32 /*positionX*/,
+ WebRtc_UWord32 /*positionY*/) { return -1; }
+ virtual WebRtc_Word32 GetDeviceName(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8=0,
+ WebRtc_UWord32 productUniqueIdUTF8Length=0) {
+ return -1;
+ }
+ virtual WebRtc_Word32 CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8) { return 0; }
+ virtual WebRtc_Word32 Init() { return 0; }
+};
+
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo(
+ const WebRtc_Word32 id) {
+ return new ExternalDeviceInfo(id);
+}
+
+} // namespace videocapturemodule
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc b/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc
new file mode 100644
index 0000000..e8cbccb
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../video_capture_impl.h"
+#include "ref_count.h"
+
+namespace webrtc {
+
+namespace videocapturemodule {
+
+VideoCaptureModule* VideoCaptureImpl::Create(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8) {
+ RefCountImpl<VideoCaptureImpl>* implementation =
+ new RefCountImpl<VideoCaptureImpl>(id);
+ return implementation;
+}
+
+} // namespace videocapturemodule
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc
new file mode 100644
index 0000000..c67885f
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc
@@ -0,0 +1,330 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "device_info_linux.h"
+
+#include <errno.h>
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+//v4l includes
+#include <linux/videodev2.h>
+
+#include "ref_count.h"
+#include "trace.h"
+
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule::DeviceInfo*
+VideoCaptureImpl::CreateDeviceInfo(const WebRtc_Word32 id)
+{
+ videocapturemodule::DeviceInfoLinux *deviceInfo =
+ new videocapturemodule::DeviceInfoLinux(id);
+ if (!deviceInfo)
+ {
+ deviceInfo = NULL;
+ }
+
+ return deviceInfo;
+}
+
+DeviceInfoLinux::DeviceInfoLinux(const WebRtc_Word32 id)
+ : DeviceInfoImpl(id)
+{
+}
+
+WebRtc_Word32 DeviceInfoLinux::Init()
+{
+ return 0;
+}
+
+DeviceInfoLinux::~DeviceInfoLinux()
+{
+}
+
+WebRtc_UWord32 DeviceInfoLinux::NumberOfDevices()
+{
+ WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__);
+
+ WebRtc_UWord32 count = 0;
+ char device[20];
+ int fd = -1;
+
+ /* detect /dev/video [0-63]VideoCaptureModule entries */
+ for (int n = 0; n < 64; n++)
+ {
+ sprintf(device, "/dev/video%d", n);
+ if ((fd = open(device, O_RDONLY)) != -1)
+ {
+ close(fd);
+ count++;
+ }
+ }
+
+ return count;
+}
+
+WebRtc_Word32 DeviceInfoLinux::GetDeviceName(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* /*productUniqueIdUTF8*/,
+ WebRtc_UWord32 /*productUniqueIdUTF8Length*/)
+{
+ WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__);
+
+ // Travel through /dev/video [0-63]
+ WebRtc_UWord32 count = 0;
+ char device[20];
+ int fd = -1;
+ bool found = false;
+ for (int n = 0; n < 64; n++)
+ {
+ sprintf(device, "/dev/video%d", n);
+ if ((fd = open(device, O_RDONLY)) != -1)
+ {
+ if (count == deviceNumber) {
+ // Found the device
+ found = true;
+ break;
+ } else {
+ close(fd);
+ count++;
+ }
+ }
+ }
+
+ if (!found)
+ return -1;
+
+ // query device capabilities
+ struct v4l2_capability cap;
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "error in querying the device capability for device %s. errno = %d",
+ device, errno);
+ close(fd);
+ return -1;
+ }
+
+ close(fd);
+
+ char cameraName[64];
+ memset(deviceNameUTF8, 0, deviceNameLength);
+ memcpy(cameraName, cap.card, sizeof(cap.card));
+
+ if (deviceNameLength >= strlen(cameraName))
+ {
+ memcpy(deviceNameUTF8, cameraName, strlen(cameraName));
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "buffer passed is too small");
+ return -1;
+ }
+
+ if (cap.bus_info[0] != 0) // may not available in all drivers
+ {
+ // copy device id
+ if (deviceUniqueIdUTF8Length >= strlen((const char*) cap.bus_info))
+ {
+ memset(deviceUniqueIdUTF8, 0, deviceUniqueIdUTF8Length);
+ memcpy(deviceUniqueIdUTF8, cap.bus_info,
+ strlen((const char*) cap.bus_info));
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "buffer passed is too small");
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 DeviceInfoLinux::CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ int fd;
+ char device[32];
+ bool found = false;
+
+ const WebRtc_Word32 deviceUniqueIdUTF8Length =
+ (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
+ if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "Device name too long");
+ return -1;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "CreateCapabilityMap called for device %s", deviceUniqueIdUTF8);
+
+ /* detect /dev/video [0-63] entries */
+ for (int n = 0; n < 64; ++n)
+ {
+ sprintf(device, "/dev/video%d", n);
+ fd = open(device, O_RDONLY);
+ if (fd == -1)
+ continue;
+
+ // query device capabilities
+ struct v4l2_capability cap;
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0)
+ {
+ if (cap.bus_info[0] != 0)
+ {
+ if (strncmp((const char*) cap.bus_info,
+ (const char*) deviceUniqueIdUTF8,
+ strlen((const char*) deviceUniqueIdUTF8)) == 0) //match with device id
+ {
+ found = true;
+ break; // fd matches with device unique id supplied
+ }
+ }
+ else //match for device name
+ {
+ if (IsDeviceNameMatches((const char*) cap.card,
+ (const char*) deviceUniqueIdUTF8))
+ {
+ found = true;
+ break;
+ }
+ }
+ }
+ close(fd); // close since this is not the matching device
+ }
+
+ if (!found)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "no matching device found");
+ return -1;
+ }
+
+ // now fd will point to the matching device
+ // reset old capability map
+ MapItem* item = NULL;
+ while ((item = _captureCapabilities.Last()))
+ {
+ delete static_cast<VideoCaptureCapability*> (item->GetItem());
+ _captureCapabilities.Erase(item);
+ }
+
+ int size = FillCapabilityMap(fd);
+ close(fd);
+
+ // Store the new used device name
+ _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
+ _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
+ _lastUsedDeviceNameLength + 1);
+ memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8, _lastUsedDeviceNameLength + 1);
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "CreateCapabilityMap %d",
+ _captureCapabilities.Size());
+
+ return size;
+}
+
+bool DeviceInfoLinux::IsDeviceNameMatches(const char* name,
+ const char* deviceUniqueIdUTF8)
+{
+ if (strncmp(deviceUniqueIdUTF8, name, strlen(name)) == 0)
+ return true;
+ return false;
+}
+
+WebRtc_Word32 DeviceInfoLinux::FillCapabilityMap(int fd)
+{
+
+ // set image format
+ struct v4l2_format video_fmt;
+ memset(&video_fmt, 0, sizeof(struct v4l2_format));
+
+ video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ video_fmt.fmt.pix.sizeimage = 0;
+
+ int totalFmts = 3;
+ unsigned int videoFormats[] = {
+ V4L2_PIX_FMT_MJPEG,
+ V4L2_PIX_FMT_YUV420,
+ V4L2_PIX_FMT_YUYV };
+
+ int sizes = 13;
+ unsigned int size[][2] = { { 128, 96 }, { 160, 120 }, { 176, 144 },
+ { 320, 240 }, { 352, 288 }, { 640, 480 },
+ { 704, 576 }, { 800, 600 }, { 960, 720 },
+ { 1280, 720 }, { 1024, 768 }, { 1440, 1080 },
+ { 1920, 1080 } };
+
+ int index = 0;
+ for (int fmts = 0; fmts < totalFmts; fmts++)
+ {
+ for (int i = 0; i < sizes; i++)
+ {
+ video_fmt.fmt.pix.pixelformat = videoFormats[fmts];
+ video_fmt.fmt.pix.width = size[i][0];
+ video_fmt.fmt.pix.height = size[i][1];
+
+ if (ioctl(fd, VIDIOC_TRY_FMT, &video_fmt) >= 0)
+ {
+ if ((video_fmt.fmt.pix.width == size[i][0])
+ && (video_fmt.fmt.pix.height == size[i][1]))
+ {
+ VideoCaptureCapability *cap = new VideoCaptureCapability();
+ cap->width = video_fmt.fmt.pix.width;
+ cap->height = video_fmt.fmt.pix.height;
+ cap->expectedCaptureDelay = 120;
+ if (videoFormats[fmts] == V4L2_PIX_FMT_YUYV)
+ {
+ cap->rawType = kVideoYUY2;
+ }
+ else if (videoFormats[fmts] == V4L2_PIX_FMT_MJPEG)
+ {
+ cap->rawType = kVideoMJPEG;
+ }
+
+ // get fps of current camera mode
+ // V4l2 does not have a stable method of knowing so we just guess.
+ if(cap->width >= 800 && cap->rawType != kVideoMJPEG)
+ {
+ cap->maxFPS = 15;
+ }
+ else
+ {
+ cap->maxFPS = 30;
+ }
+
+ _captureCapabilities.Insert(index, cap);
+ index++;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "Camera capability, width:%d height:%d type:%d fps:%d",
+ cap->width, cap->height, cap->rawType, cap->maxFPS);
+ }
+ }
+ }
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "CreateCapabilityMap %d",
+ _captureCapabilities.Size());
+ return _captureCapabilities.Size();
+}
+
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h
new file mode 100644
index 0000000..3d0e031
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
+
+#include "../video_capture_impl.h"
+#include "../device_info_impl.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+class DeviceInfoLinux: public DeviceInfoImpl
+{
+public:
+ DeviceInfoLinux(const WebRtc_Word32 id);
+ virtual ~DeviceInfoLinux();
+ virtual WebRtc_UWord32 NumberOfDevices();
+ virtual WebRtc_Word32 GetDeviceName(WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8=0,
+ WebRtc_UWord32 productUniqueIdUTF8Length=0);
+ /*
+ * Fills the membervariable _captureCapabilities with capabilites for the given device name.
+ */
+ virtual WebRtc_Word32 CreateCapabilityMap (const WebRtc_UWord8* deviceUniqueIdUTF8);
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
+ const WebRtc_UWord8* /*dialogTitleUTF8*/,
+ void* /*parentWindow*/,
+ WebRtc_UWord32 /*positionX*/,
+ WebRtc_UWord32 /*positionY*/) { return -1;}
+ WebRtc_Word32 FillCapabilityMap(int fd);
+ WebRtc_Word32 Init();
+private:
+
+ bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
diff --git a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc
new file mode 100644
index 0000000..6b47f24
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc
@@ -0,0 +1,464 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <linux/videodev2.h>
+#include <errno.h>
+#include <stdio.h>
+#include <sys/mman.h>
+#include <string.h>
+
+#include <iostream>
+#include <new>
+
+#include "ref_count.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "video_capture_linux.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule* VideoCaptureImpl::Create(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueId)
+{
+ RefCountImpl<videocapturemodule::VideoCaptureModuleV4L2>* implementation =
+ new RefCountImpl<videocapturemodule::VideoCaptureModuleV4L2>(id);
+
+ if (!implementation || implementation->Init(deviceUniqueId) != 0)
+ {
+ delete implementation;
+ implementation = NULL;
+ }
+
+ return implementation;
+}
+
+VideoCaptureModuleV4L2::VideoCaptureModuleV4L2(const WebRtc_Word32 id)
+ : VideoCaptureImpl(id),
+ _captureThread(NULL),
+ _captureCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _deviceId(-1),
+ _deviceFd(-1),
+ _buffersAllocatedByDevice(-1),
+ _currentWidth(-1),
+ _currentHeight(-1),
+ _currentFrameRate(-1),
+ _captureStarted(false),
+ _captureVideoType(kVideoI420),
+ _pool(NULL)
+{
+}
+
+WebRtc_Word32 VideoCaptureModuleV4L2::Init(const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ int len = strlen((const char*) deviceUniqueIdUTF8);
+ _deviceUniqueId = new (std::nothrow) WebRtc_UWord8[len + 1];
+ if (_deviceUniqueId)
+ {
+ memcpy(_deviceUniqueId, deviceUniqueIdUTF8, len + 1);
+ }
+
+ int fd;
+ char device[32];
+ bool found = false;
+
+ /* detect /dev/video [0-63] entries */
+ int n;
+ for (n = 0; n < 64; n++)
+ {
+ sprintf(device, "/dev/video%d", n);
+ if ((fd = open(device, O_RDONLY)) != -1)
+ {
+ // query device capabilities
+ struct v4l2_capability cap;
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0)
+ {
+ if (cap.bus_info[0] != 0)
+ {
+ if (strncmp((const char*) cap.bus_info,
+ (const char*) deviceUniqueIdUTF8,
+ strlen((const char*) deviceUniqueIdUTF8)) == 0) //match with device id
+ {
+ close(fd);
+ found = true;
+ break; // fd matches with device unique id supplied
+ }
+ }
+ }
+ close(fd); // close since this is not the matching device
+ }
+ }
+ if (!found)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "no matching device found");
+ return -1;
+ }
+ _deviceId = n; //store the device id
+ return 0;
+}
+
+VideoCaptureModuleV4L2::~VideoCaptureModuleV4L2()
+{
+ StopCapture();
+ if (_captureCritSect)
+ {
+ delete _captureCritSect;
+ }
+ if (_deviceFd != -1)
+ close(_deviceFd);
+}
+
+WebRtc_Word32 VideoCaptureModuleV4L2::StartCapture(
+ const VideoCaptureCapability& capability)
+{
+ if (_captureStarted)
+ {
+ if (capability.width == _currentWidth &&
+ capability.height == _currentHeight &&
+ _captureVideoType == capability.rawType)
+ {
+ return 0;
+ }
+ else
+ {
+ StopCapture();
+ }
+ }
+
+ CriticalSectionScoped cs(*_captureCritSect);
+ //first open /dev/video device
+ char device[20];
+ sprintf(device, "/dev/video%d", (int) _deviceId);
+
+ if ((_deviceFd = open(device, O_RDWR | O_NONBLOCK, 0)) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "error in opening %s errono = %d", device, errno);
+ return -1;
+ }
+
+ // Supported video formats in preferred order.
+ // If the requested resolution is larger than VGA, we prefer MJPEG. Go for
+ // I420 otherwise.
+ const int nFormats = 3;
+ unsigned int fmts[nFormats];
+ if (capability.width > 640 || capability.height > 480) {
+ fmts[0] = V4L2_PIX_FMT_MJPEG;
+ fmts[1] = V4L2_PIX_FMT_YUV420;
+ fmts[2] = V4L2_PIX_FMT_YUYV;
+ } else {
+ fmts[0] = V4L2_PIX_FMT_YUV420;
+ fmts[1] = V4L2_PIX_FMT_YUYV;
+ fmts[2] = V4L2_PIX_FMT_MJPEG;
+ }
+
+ struct v4l2_format video_fmt;
+ memset(&video_fmt, 0, sizeof(struct v4l2_format));
+ video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ video_fmt.fmt.pix.sizeimage = 0;
+ video_fmt.fmt.pix.width = capability.width;
+ video_fmt.fmt.pix.height = capability.height;
+
+ bool formatMatch = false;
+ for (int i = 0; i < nFormats; i++)
+ {
+ video_fmt.fmt.pix.pixelformat = fmts[i];
+ if (ioctl(_deviceFd, VIDIOC_TRY_FMT, &video_fmt) < 0)
+ {
+ continue;
+ }
+ else
+ {
+ formatMatch = true;
+ break;
+ }
+ }
+ if (!formatMatch)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "no supporting video formats found");
+ return -1;
+ }
+
+ if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUYV)
+ _captureVideoType = kVideoYUY2;
+ else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUV420)
+ _captureVideoType = kVideoI420;
+ else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_MJPEG)
+ _captureVideoType = kVideoMJPEG;
+
+ //set format and frame size now
+ if (ioctl(_deviceFd, VIDIOC_S_FMT, &video_fmt) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "error in VIDIOC_S_FMT, errno = %d", errno);
+ return -1;
+ }
+
+ // initialize current width and height
+ _currentWidth = video_fmt.fmt.pix.width;
+ _currentHeight = video_fmt.fmt.pix.height;
+ _captureDelay = 120;
+ // No way of knowing frame rate, make a guess.
+ if(_currentWidth >= 800 && _captureVideoType != kVideoMJPEG)
+ _currentFrameRate = 15;
+ else
+ _currentFrameRate = 30;
+
+ if (!AllocateVideoBuffers())
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "failed to allocate video capture buffers");
+ return -1;
+ }
+
+ //start capture thread;
+ if (!_captureThread)
+ {
+ _captureThread = ThreadWrapper::CreateThread(
+ VideoCaptureModuleV4L2::CaptureThread, this, kHighPriority);
+ unsigned int id;
+ _captureThread->Start(id);
+ }
+
+ // Needed to start UVC camera - from the uvcview application
+ enum v4l2_buf_type type;
+ type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ if (ioctl(_deviceFd, VIDIOC_STREAMON, &type) == -1)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to turn on stream");
+ return -1;
+ }
+
+ _captureStarted = true;
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureModuleV4L2::StopCapture()
+{
+ if (_captureThread)
+ _captureThread->SetNotAlive();// Make sure the capture thread stop stop using the critsect.
+
+
+ CriticalSectionScoped cs(*_captureCritSect);
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, -1, "StopCapture(), was running: %d",
+ _captureStarted);
+
+ if (!_captureStarted)
+ {
+ // we were not capturing!
+ return 0;
+ }
+
+ _captureStarted = false;
+
+ // stop the capture thread
+ // Delete capture update thread and event
+ if (_captureThread)
+ {
+ ThreadWrapper* temp = _captureThread;
+ _captureThread = NULL;
+ temp->SetNotAlive();
+ if (temp->Stop())
+ {
+ delete temp;
+ }
+ }
+
+ DeAllocateVideoBuffers();
+ close(_deviceFd);
+ _deviceFd = -1;
+
+ return 0;
+}
+
+//critical section protected by the caller
+
+bool VideoCaptureModuleV4L2::AllocateVideoBuffers()
+{
+ struct v4l2_requestbuffers rbuffer;
+ memset(&rbuffer, 0, sizeof(v4l2_requestbuffers));
+
+ rbuffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ rbuffer.memory = V4L2_MEMORY_MMAP;
+ rbuffer.count = kNoOfV4L2Bufffers;
+
+ if (ioctl(_deviceFd, VIDIOC_REQBUFS, &rbuffer) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Could not get buffers from device. errno = %d", errno);
+ return false;
+ }
+
+ if (rbuffer.count > kNoOfV4L2Bufffers)
+ rbuffer.count = kNoOfV4L2Bufffers;
+
+ _buffersAllocatedByDevice = rbuffer.count;
+
+ //Map the buffers
+ _pool = new Buffer[rbuffer.count];
+
+ for (unsigned int i = 0; i < rbuffer.count; i++)
+ {
+ struct v4l2_buffer buffer;
+ memset(&buffer, 0, sizeof(v4l2_buffer));
+ buffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ buffer.memory = V4L2_MEMORY_MMAP;
+ buffer.index = i;
+
+ if (ioctl(_deviceFd, VIDIOC_QUERYBUF, &buffer) < 0)
+ {
+ return false;
+ }
+
+ _pool[i].start = mmap(NULL, buffer.length, PROT_READ | PROT_WRITE, MAP_SHARED,
+ _deviceFd, buffer.m.offset);
+
+ if (MAP_FAILED == _pool[i].start)
+ {
+ for (unsigned int j = 0; j < i; j++)
+ munmap(_pool[j].start, _pool[j].length);
+ return false;
+ }
+
+ _pool[i].length = buffer.length;
+
+ if (ioctl(_deviceFd, VIDIOC_QBUF, &buffer) < 0)
+ {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool VideoCaptureModuleV4L2::DeAllocateVideoBuffers()
+{
+ // unmap buffers
+ for (int i = 0; i < _buffersAllocatedByDevice; i++)
+ munmap(_pool[i].start, _pool[i].length);
+
+ delete[] _pool;
+
+ // turn off stream
+ enum v4l2_buf_type type;
+ type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ if (ioctl(_deviceFd, VIDIOC_STREAMOFF, &type) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "VIDIOC_STREAMOFF error. errno: %d", errno);
+ }
+
+ return true;
+}
+
+bool VideoCaptureModuleV4L2::CaptureStarted()
+{
+ return _captureStarted;
+}
+
+bool VideoCaptureModuleV4L2::CaptureThread(void* obj)
+{
+ return static_cast<VideoCaptureModuleV4L2*> (obj)->CaptureProcess();
+}
+bool VideoCaptureModuleV4L2::CaptureProcess()
+{
+ int retVal = 0;
+ fd_set rSet;
+ struct timeval timeout;
+
+ _captureCritSect->Enter();
+ if (!_captureThread)
+ {
+ // terminating
+ _captureCritSect->Leave();
+ return false;
+ }
+
+ FD_ZERO(&rSet);
+ FD_SET(_deviceFd, &rSet);
+ timeout.tv_sec = 1;
+ timeout.tv_usec = 0;
+
+ retVal = select(_deviceFd + 1, &rSet, NULL, NULL, &timeout);
+ if (retVal < 0 && errno != EINTR) // continue if interrupted
+ {
+ // select failed
+ _captureCritSect->Leave();
+ return false;
+ }
+ else if (retVal == 0)
+ {
+ // select timed out
+ _captureCritSect->Leave();
+ return true;
+ }
+ else if (!FD_ISSET(_deviceFd, &rSet))
+ {
+ // not event on camera handle
+ _captureCritSect->Leave();
+ return true;
+ }
+
+ if (_captureStarted)
+ {
+ struct v4l2_buffer buf;
+ memset(&buf, 0, sizeof(struct v4l2_buffer));
+ buf.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+ buf.memory = V4L2_MEMORY_MMAP;
+ // dequeue a buffer - repeat until dequeued properly!
+ while (ioctl(_deviceFd, VIDIOC_DQBUF, &buf) < 0)
+ {
+ if (errno != EINTR)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "could not sync on a buffer on device %s", strerror(errno));
+ _captureCritSect->Leave();
+ return true;
+ }
+ }
+ VideoCaptureCapability frameInfo;
+ frameInfo.width = _currentWidth;
+ frameInfo.height = _currentHeight;
+ frameInfo.rawType = _captureVideoType;
+
+ // convert to to I420 if needed
+ IncomingFrame((unsigned char*) _pool[buf.index].start,
+ buf.bytesused, frameInfo);
+ // enqueue the buffer again
+ if (ioctl(_deviceFd, VIDIOC_QBUF, &buf) == -1)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "Failed to enqueue capture buffer");
+ }
+ }
+ _captureCritSect->Leave();
+ usleep(0);
+ return true;
+}
+
+WebRtc_Word32 VideoCaptureModuleV4L2::CaptureSettings(VideoCaptureCapability& settings)
+{
+ settings.width = _currentWidth;
+ settings.height = _currentHeight;
+ settings.maxFPS = _currentFrameRate;
+ settings.rawType=_captureVideoType;
+
+ return 0;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h
new file mode 100644
index 0000000..1cb6702
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
+
+#include "common_types.h"
+#include "../video_capture_impl.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class ThreadWrapper;
+namespace videocapturemodule
+{
+class VideoCaptureModuleV4L2: public VideoCaptureImpl
+{
+public:
+ VideoCaptureModuleV4L2(WebRtc_Word32 id);
+ virtual ~VideoCaptureModuleV4L2();
+ virtual WebRtc_Word32 Init(const WebRtc_UWord8* deviceUniqueId);
+ virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability);
+ virtual WebRtc_Word32 StopCapture();
+ virtual bool CaptureStarted();
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
+
+private:
+ enum {kNoOfV4L2Bufffers=4};
+
+ static bool CaptureThread(void*);
+ bool CaptureProcess();
+ bool AllocateVideoBuffers();
+ bool DeAllocateVideoBuffers();
+
+ ThreadWrapper* _captureThread;
+ CriticalSectionWrapper* _captureCritSect;
+
+ WebRtc_Word32 _deviceId;
+ WebRtc_Word32 _deviceFd;
+
+ WebRtc_Word32 _buffersAllocatedByDevice;
+ WebRtc_Word32 _currentWidth;
+ WebRtc_Word32 _currentHeight;
+ WebRtc_Word32 _currentFrameRate;
+ bool _captureStarted;
+ RawVideoType _captureVideoType;
+ struct Buffer
+ {
+ void *start;
+ size_t length;
+ };
+ Buffer *_pool;
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h
new file mode 100644
index 0000000..14ca344
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
+
+#import <QTKit/QTKit.h>
+
+#include <stdio.h>
+
+#include "../../video_capture_impl.h"
+#include "video_capture_qtkit_utility.h"
+#include "../../device_info_impl.h"
+
+
+// Forward declaraion
+@class VideoCaptureMacQTKitObjC;
+@class VideoCaptureMacQTKitInfoObjC;
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+class VideoCaptureMacQTKit : public VideoCaptureImpl
+{
+public:
+ VideoCaptureMacQTKit(const WebRtc_Word32 id);
+ virtual ~VideoCaptureMacQTKit();
+
+ /*
+ * Create a video capture module object
+ *
+ * id - unique identifier of this video capture module object
+ * deviceUniqueIdUTF8 - name of the device. Available names can be found
+ * by using GetDeviceName
+ * deviceUniqueIdUTF8Length - length of deviceUniqueIdUTF8
+ */
+ static void Destroy(VideoCaptureModule* module);
+
+ WebRtc_Word32 Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+
+ // Start/Stop
+ virtual WebRtc_Word32 StartCapture(
+ const VideoCaptureCapability& capability);
+ virtual WebRtc_Word32 StopCapture();
+
+ // Properties of the set device
+
+ virtual bool CaptureStarted();
+
+ WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
+
+protected:
+ // Help functions
+ WebRtc_Word32 SetCameraOutput();
+
+private:
+ VideoCaptureMacQTKitObjC* _captureDevice;
+ VideoCaptureMacQTKitInfoObjC* _captureInfo;
+ bool _isCapturing;
+ WebRtc_Word32 _id;
+ WebRtc_Word32 _captureWidth;
+ WebRtc_Word32 _captureHeight;
+ WebRtc_Word32 _captureFrameRate;
+ WebRtc_UWord8 _currentDeviceNameUTF8[MAX_NAME_LENGTH];
+ WebRtc_UWord8 _currentDeviceUniqueIdUTF8[MAX_NAME_LENGTH];
+ WebRtc_UWord8 _currentDeviceProductUniqueIDUTF8[MAX_NAME_LENGTH];
+ WebRtc_Word32 _frameCount;
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm
new file mode 100644
index 0000000..c4a02c9
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm
@@ -0,0 +1,223 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_capture_qtkit.h"
+#import "video_capture_qtkit_objc.h"
+#import "video_capture_qtkit_info_objc.h"
+#include "trace.h"
+#include "critical_section_wrapper.h"
+#include "../../video_capture_config.h"
+
+namespace webrtc
+{
+
+namespace videocapturemodule
+{
+
+VideoCaptureMacQTKit::VideoCaptureMacQTKit(const WebRtc_Word32 id) :
+ VideoCaptureImpl(id),
+ _captureDevice(NULL),
+ _captureInfo(NULL),
+ _isCapturing(false),
+ _id(id),
+ _captureWidth(QTKIT_DEFAULT_WIDTH),
+ _captureHeight(QTKIT_DEFAULT_HEIGHT),
+ _captureFrameRate(QTKIT_DEFAULT_FRAME_RATE),
+ _frameCount(0)
+{
+
+ memset(_currentDeviceNameUTF8, 0, MAX_NAME_LENGTH);
+ memset(_currentDeviceUniqueIdUTF8, 0, MAX_NAME_LENGTH);
+ memset(_currentDeviceProductUniqueIDUTF8, 0, MAX_NAME_LENGTH);
+}
+
+VideoCaptureMacQTKit::~VideoCaptureMacQTKit()
+{
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "~VideoCaptureMacQTKit() called");
+ if(_captureDevice)
+ {
+ [_captureDevice stopCapture];
+ [_captureDevice release];
+ }
+
+ if(_captureInfo)
+ {
+ [_captureInfo release];
+ }
+}
+
+WebRtc_Word32 VideoCaptureMacQTKit::Init(
+ const WebRtc_Word32 id, const WebRtc_UWord8* iDeviceUniqueIdUTF8)
+{
+ CriticalSectionScoped cs(_apiCs);
+
+
+ const WebRtc_Word32 nameLength =
+ (WebRtc_Word32) strlen((char*)iDeviceUniqueIdUTF8);
+ if(nameLength>kVideoCaptureUniqueNameLength)
+ return -1;
+
+ // Store the device name
+ _deviceUniqueId = new WebRtc_UWord8[nameLength+1];
+ memcpy(_deviceUniqueId, iDeviceUniqueIdUTF8,nameLength+1);
+
+ _captureDevice = [[VideoCaptureMacQTKitObjC alloc] init];
+ if(NULL == _captureDevice)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "Failed to create an instance of "
+ "VideoCaptureMacQTKitObjC");
+ return -1;
+ }
+
+ if(-1 == [[_captureDevice registerOwner:this]intValue])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "Failed to register owner for _captureDevice");
+ return -1;
+ }
+
+ if(0 == strcmp((char*)iDeviceUniqueIdUTF8, ""))
+ {
+ // the user doesn't want to set a capture device at this time
+ return 0;
+ }
+
+ _captureInfo = [[VideoCaptureMacQTKitInfoObjC alloc]init];
+ if(nil == _captureInfo)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id, "Failed to create an instance of VideoCaptureMacQTKitInfoObjC");
+ return -1;
+ }
+
+ int captureDeviceCount = [[_captureInfo getCaptureDeviceCount]intValue];
+ if(captureDeviceCount < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "No Capture Devices Present");
+ return -1;
+ }
+
+ const int NAME_LENGTH = 1024;
+ WebRtc_UWord8 deviceNameUTF8[1024] = "";
+ WebRtc_UWord8 deviceUniqueIdUTF8[1024] = "";
+ WebRtc_UWord8 deviceProductUniqueIDUTF8[1024] = "";
+
+ bool captureDeviceFound = false;
+ for(int index = 0; index < captureDeviceCount; index++){
+
+ memset(deviceNameUTF8, 0, NAME_LENGTH);
+ memset(deviceUniqueIdUTF8, 0, NAME_LENGTH);
+ memset(deviceProductUniqueIDUTF8, 0, NAME_LENGTH);
+ if(-1 == [[_captureInfo getDeviceNamesFromIndex:index
+ DefaultName:deviceNameUTF8 WithLength:NAME_LENGTH
+ AndUniqueID:deviceUniqueIdUTF8 WithLength:NAME_LENGTH
+ AndProductID:deviceProductUniqueIDUTF8
+ WithLength:NAME_LENGTH]intValue])
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "GetDeviceName returned -1 for index %d", index);
+ return -1;
+ }
+ if(0 == strcmp((const char*)iDeviceUniqueIdUTF8,
+ (char*)deviceUniqueIdUTF8))
+ {
+ // we have a match
+ captureDeviceFound = true;
+ break;
+ }
+ }
+
+ if(false == captureDeviceFound)
+ {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "Failed to find capture device unique ID %s",
+ iDeviceUniqueIdUTF8);
+ return -1;
+ }
+
+ // at this point we know that the user has passed in a valid camera. Let's
+ // set it as the current.
+ if(-1 == [[_captureDevice
+ setCaptureDeviceById:(char*)deviceUniqueIdUTF8]intValue])
+ {
+ strcpy((char*)_deviceUniqueId, (char*)deviceUniqueIdUTF8);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to set capture device %s (unique ID %s) even "
+ "though it was a valid return from "
+ "VideoCaptureMacQTKitInfo");
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "successfully Init VideoCaptureMacQTKit" );
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureMacQTKit::StartCapture(
+ const VideoCaptureCapability& capability)
+{
+
+ _captureWidth = capability.width;
+ _captureHeight = capability.height;
+ _captureFrameRate = capability.maxFPS;
+
+ if(-1 == [[_captureDevice setCaptureHeight:_captureHeight
+ AndWidth:_captureWidth AndFrameRate:_captureFrameRate]intValue])
+ {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "Could not set width=%d height=%d frameRate=%d",
+ _captureWidth, _captureHeight, _captureFrameRate);
+ return -1;
+ }
+
+ if(-1 == [[_captureDevice startCapture]intValue])
+ {
+ return -1;
+ }
+ _isCapturing = true;
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureMacQTKit::StopCapture()
+{
+ [_captureDevice stopCapture];
+
+ _isCapturing = false;
+ return 0;
+}
+
+bool VideoCaptureMacQTKit::CaptureStarted()
+{
+ return _isCapturing;
+}
+
+WebRtc_Word32 VideoCaptureMacQTKit::CaptureSettings(VideoCaptureCapability& settings)
+{
+ settings.width = _captureWidth;
+ settings.height = _captureHeight;
+ settings.maxFPS = _captureFrameRate;
+ return 0;
+}
+
+
+// ********** begin functions inherited from DeviceInfoImpl **********
+
+struct VideoCaptureCapabilityMacQTKit:public VideoCaptureCapability
+{
+ VideoCaptureCapabilityMacQTKit()
+ {
+ }
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h
new file mode 100644
index 0000000..b6d05ad
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
+
+#include "../../video_capture_impl.h"
+#include "../../device_info_impl.h"
+#include "video_capture_qtkit_utility.h"
+
+#include "map_wrapper.h"
+
+
+@class VideoCaptureMacQTKitInfoObjC;
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+class VideoCaptureMacQTKitInfo: public DeviceInfoImpl
+{
+public:
+
+ VideoCaptureMacQTKitInfo(const WebRtc_Word32 id);
+ virtual ~VideoCaptureMacQTKitInfo();
+
+ WebRtc_Word32 Init();
+
+ virtual WebRtc_UWord32 NumberOfDevices();
+
+ /*
+ * Returns the available capture devices.
+ * deviceNumber -[in] index of capture device
+ * deviceNameUTF8 - friendly name of the capture device
+ * deviceUniqueIdUTF8 - unique name of the capture device if it exist.
+ * Otherwise same as deviceNameUTF8
+ * productUniqueIdUTF8 - unique product id if it exist. Null terminated
+ * otherwise.
+ */
+ virtual WebRtc_Word32 GetDeviceName(
+ WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8 = 0,
+ WebRtc_UWord32 productUniqueIdUTF8Length = 0);
+
+ /*
+ * Returns the number of capabilities for this device
+ */
+ virtual WebRtc_Word32 NumberOfCapabilities(
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ /*
+ * Gets the capabilities of the named device
+ */
+ virtual WebRtc_Word32 GetCapability(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability);
+
+ /*
+ * Gets the capability that best matches the requested width, height and frame rate.
+ * Returns the deviceCapabilityNumber on success.
+ */
+ virtual WebRtc_Word32 GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested,
+ VideoCaptureCapability& resulting);
+
+ /*
+ * Display OS /capture device specific settings dialog
+ */
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
+ WebRtc_UWord32 positionX, WebRtc_UWord32 positionY);
+
+protected:
+ virtual WebRtc_Word32 CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ VideoCaptureMacQTKitInfoObjC* _captureInfo;
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm
new file mode 100644
index 0000000..b13a8ab
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm
@@ -0,0 +1,126 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "../../video_capture_config.h"
+#import "video_capture_qtkit_info_objc.h"
+
+#include "video_capture.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+VideoCaptureMacQTKitInfo::VideoCaptureMacQTKitInfo(const WebRtc_Word32 id) :
+ DeviceInfoImpl(id)
+{
+ _captureInfo = [[VideoCaptureMacQTKitInfoObjC alloc] init];
+}
+
+VideoCaptureMacQTKitInfo::~VideoCaptureMacQTKitInfo()
+{
+ [_captureInfo release];
+
+}
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::Init()
+{
+
+ return 0;
+}
+
+WebRtc_UWord32 VideoCaptureMacQTKitInfo::NumberOfDevices()
+{
+
+ WebRtc_UWord32 captureDeviceCount =
+ [[_captureInfo getCaptureDeviceCount]intValue];
+ return captureDeviceCount;
+
+}
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::GetDeviceName(
+ WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+{
+ int errNum = [[_captureInfo getDeviceNamesFromIndex:deviceNumber
+ DefaultName:deviceNameUTF8 WithLength:deviceNameLength
+ AndUniqueID:deviceUniqueIdUTF8
+ WithLength:deviceUniqueIdUTF8Length
+ AndProductID:productUniqueIdUTF8
+ WithLength:productUniqueIdUTF8Length]intValue];
+ return errNum;
+}
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::NumberOfCapabilities(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ // Not implemented. Mac doesn't use discrete steps in capabilities, rather
+ // "analog". QTKit will do it's best to convert frames to what ever format
+ // you ask for.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::GetCapability(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability)
+{
+ // Not implemented. Mac doesn't use discrete steps in capabilities, rather
+ // "analog". QTKit will do it's best to convert frames to what ever format
+ // you ask for.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested, VideoCaptureCapability& resulting)
+{
+ // Not implemented. Mac doesn't use discrete steps in capabilities, rather
+ // "analog". QTKit will do it's best to convert frames to what ever format
+ // you ask for.
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
+ WebRtc_UWord32 positionX, WebRtc_UWord32 positionY)
+{
+
+ return [[_captureInfo
+ displayCaptureSettingsDialogBoxWithDevice:deviceUniqueIdUTF8
+ AndTitle:dialogTitleUTF8
+ AndParentWindow:parentWindow AtX:positionX AndY:positionY]
+ intValue];
+}
+
+WebRtc_Word32 VideoCaptureMacQTKitInfo::CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ // Not implemented. Mac doesn't use discrete steps in capabilities, rather
+ // "analog". QTKit will do it's best to convert frames to what ever format
+ // you ask for.
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h
new file mode 100644
index 0000000..1d3ac86
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// video_capture_qtkit_info_objc.h
+//
+//
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
+
+#import <QTKit/QTKit.h>
+#import <Foundation/Foundation.h>
+#include "video_capture_qtkit_utility.h"
+#include "video_capture_qtkit_info.h"
+
+@interface VideoCaptureMacQTKitInfoObjC : NSObject{
+ bool _OSSupportedInfo;
+ NSArray* _captureDevicesInfo;
+ NSAutoreleasePool* _poolInfo;
+ int _captureDeviceCountInfo;
+
+}
+
+/**************************************************************************
+ *
+ * The following functions are considered to be private
+ *
+ ***************************************************************************/
+
+- (NSNumber*)getCaptureDevices;
+- (NSNumber*)initializeVariables;
+- (void)checkOSSupported;
+
+
+/**************************************************************************
+ *
+ * The following functions are considered to be public and called by VideoCaptureMacQTKitInfo class
+ *
+ ***************************************************************************/
+
+- (NSNumber*)getCaptureDeviceCount;
+
+- (NSNumber*)getDeviceNamesFromIndex:(WebRtc_UWord32)index
+ DefaultName:(WebRtc_UWord8*)deviceName
+ WithLength:(WebRtc_UWord32)deviceNameLength
+ AndUniqueID:(WebRtc_UWord8*)deviceUniqueID
+ WithLength:(WebRtc_UWord32)deviceUniqueIDLength
+ AndProductID:(WebRtc_UWord8*)deviceProductID
+ WithLength:(WebRtc_UWord32)deviceProductIDLength;
+
+- (NSNumber*)displayCaptureSettingsDialogBoxWithDevice:
+ (const WebRtc_UWord8*)deviceUniqueIdUTF8
+ AndTitle:(const WebRtc_UWord8*)dialogTitleUTF8
+ AndParentWindow:(void*) parentWindow AtX:(WebRtc_UWord32)positionX
+ AndY:(WebRtc_UWord32) positionY;
+@end
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm
new file mode 100644
index 0000000..57ab3d0
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma mark **** imports/includes
+
+#import "video_capture_qtkit_info_objc.h"
+
+#include "trace.h"
+
+using namespace webrtc;
+
+#pragma mark **** hidden class interface
+
+@implementation VideoCaptureMacQTKitInfoObjC
+
+// ****************** over-written OS methods ***********************
+#pragma mark **** over-written OS methods
+
+/// ***** Objective-C. Similar to C++ constructor, although invoked manually
+/// ***** Potentially returns an instance of self
+-(id)init{
+ self = [super init];
+ if(nil != self){
+ [self checkOSSupported];
+ [self initializeVariables];
+ }
+ else
+ {
+ return nil;
+ }
+ return self;
+}
+
+/// ***** Objective-C. Similar to C++ destructor
+/// ***** Returns nothing
+- (void)dealloc {
+ [super dealloc];
+}
+
+// ****************** public methods ******************
+#pragma mark **** public method implementations
+
+/// ***** Creates a message box with Cocoa framework
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)displayCaptureSettingsDialogBoxWithDevice:(const WebRtc_UWord8*)deviceUniqueIdUTF8
+ AndTitle:(const WebRtc_UWord8*)dialogTitleUTF8
+ AndParentWindow:(void*) parentWindow
+ AtX:(WebRtc_UWord32)positionX
+ AndY:(WebRtc_UWord32) positionY
+{
+ NSString* strTitle = [NSString stringWithFormat:@"%s", dialogTitleUTF8];
+ NSString* strButton = @"Alright";
+ NSString* strMessage = [NSString stringWithFormat:@"Device %s is capturing:\nWidth:%d\n:Height:%d\n@%dfps", deviceUniqueIdUTF8];
+ NSAlert* alert = [NSAlert alertWithMessageText:strTitle
+ defaultButton:strButton
+ alternateButton:nil otherButton:nil
+ informativeTextWithFormat:strMessage];
+ [alert setAlertStyle:NSInformationalAlertStyle];
+ [alert runModal];
+ return [NSNumber numberWithInt:0];
+}
+
+- (NSNumber*)getCaptureDeviceCount{
+ [self getCaptureDevices];
+ return [NSNumber numberWithInt:_captureDeviceCountInfo];
+}
+
+
+- (NSNumber*)getDeviceNamesFromIndex:(WebRtc_UWord32)index
+ DefaultName:(WebRtc_UWord8*)deviceName
+ WithLength:(WebRtc_UWord32)deviceNameLength
+ AndUniqueID:(WebRtc_UWord8*)deviceUniqueID
+ WithLength:(WebRtc_UWord32)deviceUniqueIDLength
+ AndProductID:(WebRtc_UWord8*)deviceProductID
+ WithLength:(WebRtc_UWord32)deviceProductIDLength
+{
+ if(NO == _OSSupportedInfo)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(index > (WebRtc_UWord32)_captureDeviceCountInfo)
+ {
+ return [NSNumber numberWithInt:-1];
+ }
+
+ QTCaptureDevice* tempCaptureDevice =
+ (QTCaptureDevice*)[_captureDevicesInfo objectAtIndex:index];
+ if(!tempCaptureDevice)
+ {
+ return [NSNumber numberWithInt:-1];
+ }
+
+ memset(deviceName, 0, deviceNameLength);
+ memset(deviceUniqueID, 0, deviceUniqueIDLength);
+
+ bool successful = NO;
+
+ NSString* tempString = [tempCaptureDevice localizedDisplayName];
+ successful = [tempString getCString:(char*)deviceName
+ maxLength:deviceNameLength encoding:NSUTF8StringEncoding];
+ if(NO == successful)
+ {
+ memset(deviceName, 0, deviceNameLength);
+ return [NSNumber numberWithInt:-1];
+ }
+
+ tempString = [tempCaptureDevice uniqueID];
+ successful = [tempString getCString:(char*)deviceUniqueID
+ maxLength:deviceUniqueIDLength encoding:NSUTF8StringEncoding];
+ if(NO == successful)
+ {
+ memset(deviceUniqueID, 0, deviceNameLength);
+ return [NSNumber numberWithInt:-1];
+ }
+
+ return [NSNumber numberWithInt:0];
+
+}
+
+// ****************** "private" category functions below here ******************
+#pragma mark **** "private" method implementations
+
+- (NSNumber*)initializeVariables
+{
+ if(NO == _OSSupportedInfo)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ _poolInfo = [[NSAutoreleasePool alloc]init];
+ _captureDeviceCountInfo = 0;
+ [self getCaptureDevices];
+
+ return [NSNumber numberWithInt:0];
+}
+
+// ***** Checks to see if the QTCaptureSession framework is available in the OS
+// ***** If it is not, isOSSupprted = NO
+// ***** Throughout the rest of the class isOSSupprted is checked and functions
+// ***** are/aren't called depending
+// ***** The user can use weak linking to the QTKit framework and run on older
+// ***** versions of the OS
+// ***** I.E. Backwards compaitibility
+// ***** Returns nothing. Sets member variable
+- (void)checkOSSupported
+{
+ Class osSupportedTest = NSClassFromString(@"QTCaptureSession");
+ _OSSupportedInfo = NO;
+ if(nil == osSupportedTest)
+ {
+ }
+ _OSSupportedInfo = YES;
+}
+
+/// ***** Retrieves the number of capture devices currently available
+/// ***** Stores them in an NSArray instance
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)getCaptureDevices
+{
+ if(NO == _OSSupportedInfo)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(_captureDevicesInfo)
+ {
+ [_captureDevicesInfo release];
+ }
+ _captureDevicesInfo = [[NSArray alloc]
+ initWithArray:[QTCaptureDevice
+ inputDevicesWithMediaType:QTMediaTypeVideo]];
+
+ _captureDeviceCountInfo = _captureDevicesInfo.count;
+ if(_captureDeviceCountInfo < 1){
+ return [NSNumber numberWithInt:0];
+ }
+ return [NSNumber numberWithInt:0];
+}
+
+@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h
new file mode 100644
index 0000000..d48dbf1
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// video_capture_qtkit_objc.h
+//
+//
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
+
+#import <Foundation/Foundation.h>
+#import <QTKit/QTKit.h>
+#import <AppKit/AppKit.h>
+#import <CoreData/CoreData.h>
+#import <CoreFoundation/CoreFoundation.h>
+#import <CoreVideo/CoreVideo.h>
+
+#import "video_capture_recursive_lock.h"
+
+#include "video_capture_qtkit.h"
+
+@interface VideoCaptureMacQTKitObjC : NSObject{
+ // class properties
+ bool _capturing;
+ int _counter;
+ int _frameRate;
+ int _frameWidth;
+ int _frameHeight;
+ int _framesDelivered;
+ int _framesRendered;
+ bool _OSSupported;
+ bool _captureInitialized;
+
+ // WebRTC Custom classes
+ webrtc::videocapturemodule::VideoCaptureMacQTKit* _owner;
+ VideoCaptureRecursiveLock* _rLock;
+
+ // QTKit variables
+ QTCaptureSession* _captureSession;
+ QTCaptureDeviceInput* _captureVideoDeviceInput;
+ QTCaptureDecompressedVideoOutput* _captureDecompressedVideoOutput;
+ NSArray* _captureDevices;
+ int _captureDeviceCount;
+ int _captureDeviceIndex;
+ NSString* _captureDeviceName;
+ char _captureDeviceNameUTF8[1024];
+ char _captureDeviceNameUniqueID[1024];
+ char _captureDeviceNameProductID[1024];
+ NSString* _key;
+ NSNumber* _val;
+ NSDictionary* _videoSettings;
+ NSString* _captureQuality;
+
+ // other
+ NSAutoreleasePool* _pool;
+
+}
+/**************************************************************************
+ *
+ * The following functions are considered to be private.
+ *
+ ***************************************************************************/
+
+- (NSNumber*)getCaptureDevices;
+- (NSNumber*)initializeVideoCapture;
+- (NSNumber*)initializeVariables;
+- (void)checkOSSupported;
+
+
+/**************************************************************************
+ *
+ * The following functions are considered public and to be called by the VideoCaptureMacQTKit class.
+ *
+ ***************************************************************************/
+
+
+- (NSNumber*)registerOwner:(webrtc::videocapturemodule::VideoCaptureMacQTKit*)owner;
+- (NSNumber*)setCaptureDeviceById:(char*)uniqueId;
+- (NSNumber*)setCaptureHeight:(int)height AndWidth:(int)width AndFrameRate:(int)frameRate;
+- (NSNumber*)startCapture;
+- (NSNumber*)stopCapture;
+
+@end
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm
new file mode 100644
index 0000000..48f734c
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm
@@ -0,0 +1,462 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#define DEFAULT_CAPTURE_DEVICE_INDEX 1
+#define DEFAULT_FRAME_RATE 30
+#define DEFAULT_FRAME_WIDTH 352
+#define DEFAULT_FRAME_HEIGHT 288
+#define ROTATE_CAPTURED_FRAME 1
+#define LOW_QUALITY 1
+
+#import "video_capture_qtkit_objc.h"
+#include "video_capture_qtkit_utility.h"
+#include "trace.h"
+
+using namespace webrtc;
+using namespace videocapturemodule;
+
+@implementation VideoCaptureMacQTKitObjC
+
+#pragma mark **** over-written OS methods
+
+/// ***** Objective-C. Similar to C++ constructor, although must be invoked
+/// manually.
+/// ***** Potentially returns an instance of self
+-(id)init{
+ self = [super init];
+ if(nil != self)
+ {
+ [self checkOSSupported];
+ [self initializeVariables];
+ }
+ else
+ {
+ return nil;
+ }
+ return self;
+}
+
+/// ***** Objective-C. Similar to C++ destructor
+/// ***** Returns nothing
+- (void)dealloc {
+ if(_captureSession)
+ {
+ [_captureSession stopRunning];
+ [_captureSession release];
+ }
+ [super dealloc];
+}
+
+#pragma mark **** public methods
+
+
+
+/// ***** Registers the class's owner, which is where the delivered frames are
+/// sent
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)registerOwner:(VideoCaptureMacQTKit*)owner{
+ if(!owner){
+ return [NSNumber numberWithInt:-1];
+ }
+ _owner = owner;
+ return [NSNumber numberWithInt:0];
+}
+
+/// ***** Sets the QTCaptureSession's input device from a char*
+/// ***** Sets several member variables. Can signal the error system if one has
+/// occurred
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)setCaptureDeviceById:(char*)uniqueId{
+ if(NO == _OSSupported)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d OS version does not support necessary APIs",
+ __FUNCTION__, __LINE__);
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(!uniqueId || (0 == strcmp("", uniqueId)))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d \"\" was passed in for capture device name",
+ __FUNCTION__, __LINE__);
+ memset(_captureDeviceNameUTF8, 0, 1024);
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(0 == strcmp(uniqueId, _captureDeviceNameUniqueID))
+ {
+ // camera already set
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d Capture device is already set to %s", __FUNCTION__,
+ __LINE__, _captureDeviceNameUTF8);
+ return [NSNumber numberWithInt:0];
+ }
+
+ bool success = NO;
+ QTCaptureDevice* tempCaptureDevice;
+ for(int index = 0; index < _captureDeviceCount; index++)
+ {
+ tempCaptureDevice = (QTCaptureDevice*)[_captureDevices
+ objectAtIndex:index];
+ char tempCaptureDeviceId[1024] = "";
+ [[tempCaptureDevice uniqueID]
+ getCString:tempCaptureDeviceId maxLength:1024
+ encoding:NSUTF8StringEncoding];
+ if(0 == strcmp(uniqueId, tempCaptureDeviceId))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d Found capture device id %s as index %d",
+ __FUNCTION__, __LINE__, tempCaptureDeviceId, index);
+ success = YES;
+ [[tempCaptureDevice localizedDisplayName]
+ getCString:_captureDeviceNameUTF8
+ maxLength:1024
+ encoding:NSUTF8StringEncoding];
+ [[tempCaptureDevice uniqueID]
+ getCString:_captureDeviceNameUniqueID
+ maxLength:1024
+ encoding:NSUTF8StringEncoding];
+ break;
+ }
+
+ }
+
+ if(NO == success)
+ {
+ // camera not found
+ // nothing has been changed yet, so capture device will stay in it's
+ // state
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d Capture device id %s was not found in list of "
+ "available devices.", __FUNCTION__, __LINE__, uniqueId);
+ return [NSNumber numberWithInt:0];
+ }
+
+ NSError* error;
+ success = [tempCaptureDevice open:&error];
+ if(!success)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoCapture, 0,
+ "%s:%d Failed to open capture device: %s",
+ __FUNCTION__, __LINE__, _captureDeviceNameUTF8);
+ return [NSNumber numberWithInt:-1];
+ }
+
+ if(_captureVideoDeviceInput)
+ {
+ [_captureVideoDeviceInput release];
+ }
+ _captureVideoDeviceInput = [[QTCaptureDeviceInput alloc]
+ initWithDevice:tempCaptureDevice];
+
+ success = [_captureSession addInput:_captureVideoDeviceInput error:&error];
+ if(!success)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoCapture, 0,
+ "%s:%d Failed to add input from %s to the capture session",
+ __FUNCTION__, __LINE__, _captureDeviceNameUTF8);
+ return [NSNumber numberWithInt:-1];
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
+ "%s:%d successfully added capture device: %s", __FUNCTION__,
+ __LINE__, _captureDeviceNameUTF8);
+ return [NSNumber numberWithInt:0];
+}
+
+
+/// ***** Updates the capture devices size and frequency
+/// ***** Sets member variables _frame* and _captureDecompressedVideoOutput
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)setCaptureHeight:(int)height AndWidth:(int)width
+ AndFrameRate:(int)frameRate{
+ if(NO == _OSSupported)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ _frameWidth = width;
+ _frameHeight = height;
+ _frameRate = frameRate;
+
+ // TODO(mflodman) Check fps settings.
+ // [_captureDecompressedVideoOutput
+ // setMinimumVideoFrameInterval:(NSTimeInterval)1/(float)_frameRate];
+ NSDictionary* captureDictionary = [NSDictionary dictionaryWithObjectsAndKeys:
+ [NSNumber numberWithDouble:_frameWidth], (id)kCVPixelBufferWidthKey,
+ [NSNumber numberWithDouble:_frameHeight], (id)kCVPixelBufferHeightKey,
+ [NSNumber numberWithUnsignedInt:kCVPixelFormatType_32ARGB],
+ (id)kCVPixelBufferPixelFormatTypeKey, nil];
+ [_captureDecompressedVideoOutput performSelectorOnMainThread:@selector(setPixelBufferAttributes:) withObject:captureDictionary waitUntilDone:NO];
+// [_captureDecompressedVideoOutput setPixelBufferAttributes:captureDictionary];
+
+
+ // these methods return type void so there isn't much we can do about
+ // checking success
+ return [NSNumber numberWithInt:0];
+}
+
+/// ***** Starts the QTCaptureSession, assuming correct state. Also ensures that
+/// an NSRunLoop is running
+/// ***** Without and NSRunLoop to process events, the OS doesn't check for a
+/// new frame.
+/// ***** Sets member variables _capturing
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)startCapture{
+ if(NO == _OSSupported)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(YES == _capturing)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+// NSLog(@"--------------- before ---------------");
+ [[NSRunLoop mainRunLoop] runUntilDate:[NSDate distantFuture]];
+// NSLog(@"--------------- after ---------------");
+
+ if(NO == _captureInitialized)
+ {
+ // this should never be called..... it is initialized on class init
+ [self initializeVideoCapture];
+ }
+ [_captureSession startRunning];
+
+
+ _capturing = YES;
+
+ return [NSNumber numberWithInt:0];
+}
+
+/// ***** Stops the QTCaptureSession, assuming correct state
+/// ***** Sets member variables _capturing
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)stopCapture{
+
+ if(NO == _OSSupported)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(nil == _captureSession)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(NO == _capturing)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(YES == _capturing)
+ {
+ [_captureSession stopRunning];
+ }
+
+ _capturing = NO;
+ return [NSNumber numberWithInt:0];
+}
+
+// ********** "private" functions below here **********
+#pragma mark **** "private" methods
+
+/// ***** Class member variables are initialized here
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)initializeVariables{
+
+ if(NO == _OSSupported)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ _pool = [[NSAutoreleasePool alloc]init];
+
+ memset(_captureDeviceNameUTF8, 0, 1024);
+ _counter = 0;
+ _framesDelivered = 0;
+ _framesRendered = 0;
+ _captureDeviceCount = 0;
+ _capturing = NO;
+ _captureInitialized = NO;
+ _frameRate = DEFAULT_FRAME_RATE;
+ _frameWidth = DEFAULT_FRAME_WIDTH;
+ _frameHeight = DEFAULT_FRAME_HEIGHT;
+ _captureDeviceName = [[NSString alloc] initWithFormat:@""];
+ _rLock = [[VideoCaptureRecursiveLock alloc] init];
+ _captureSession = [[QTCaptureSession alloc] init];
+ _captureDecompressedVideoOutput = [[QTCaptureDecompressedVideoOutput alloc]
+ init];
+ [_captureDecompressedVideoOutput setDelegate:self];
+
+ [self getCaptureDevices];
+ [self initializeVideoCapture];
+
+ return [NSNumber numberWithInt:0];
+
+}
+
+// Checks to see if the QTCaptureSession framework is available in the OS
+// If it is not, isOSSupprted = NO.
+// Throughout the rest of the class isOSSupprted is checked and functions
+// are/aren't called depending
+// The user can use weak linking to the QTKit framework and run on older
+// versions of the OS. I.E. Backwards compaitibility
+// Returns nothing. Sets member variable
+- (void)checkOSSupported{
+
+ Class osSupportedTest = NSClassFromString(@"QTCaptureSession");
+ _OSSupported = NO;
+ if(nil == osSupportedTest)
+ {
+ }
+ _OSSupported = YES;
+}
+
+/// ***** Retrieves the number of capture devices currently available
+/// ***** Stores them in an NSArray instance
+/// ***** Returns 0 on success, -1 otherwise.
+- (NSNumber*)getCaptureDevices{
+
+ if(NO == _OSSupported)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+
+ if(_captureDevices)
+ {
+ [_captureDevices release];
+ }
+ _captureDevices = [[NSArray alloc] initWithArray:
+ [QTCaptureDevice inputDevicesWithMediaType:QTMediaTypeVideo]];
+
+ _captureDeviceCount = _captureDevices.count;
+ if(_captureDeviceCount < 1)
+ {
+ return [NSNumber numberWithInt:0];
+ }
+ return [NSNumber numberWithInt:0];
+}
+
+// Initializes a QTCaptureSession (member variable) to deliver frames via
+// callback
+// QTCapture* member variables affected
+// The image format and frequency are setup here
+// Returns 0 on success, -1 otherwise.
+- (NSNumber*)initializeVideoCapture{
+
+ if(YES == _captureInitialized)
+ {
+ return [NSNumber numberWithInt:-1];
+ }
+
+ QTCaptureDevice* videoDevice =
+ (QTCaptureDevice*)[_captureDevices objectAtIndex:0];
+
+ bool success = NO;
+ NSError* error;
+
+ success = [videoDevice open:&error];
+ if(!success)
+ {
+ return [NSNumber numberWithInt:-1];
+ }
+
+ [_captureDecompressedVideoOutput setPixelBufferAttributes:
+ [NSDictionary dictionaryWithObjectsAndKeys:
+ [NSNumber numberWithDouble:_frameWidth], (id)kCVPixelBufferWidthKey,
+ [NSNumber numberWithDouble:_frameHeight], (id)kCVPixelBufferHeightKey,
+ [NSNumber numberWithUnsignedInt:kCVPixelFormatType_32ARGB],
+ (id)kCVPixelBufferPixelFormatTypeKey, nil]];
+
+ // TODO(mflodman) Check fps settings.
+ //[_captureDecompressedVideoOutput setMinimumVideoFrameInterval:
+ // (NSTimeInterval)1/(float)_frameRate];
+ //[_captureDecompressedVideoOutput setAutomaticallyDropsLateVideoFrames:YES];
+
+ success = [_captureSession addOutput:_captureDecompressedVideoOutput
+ error:&error];
+
+ if(!success)
+ {
+ return [NSNumber numberWithInt:-1];
+ }
+
+ _captureInitialized = YES;
+
+ return [NSNumber numberWithInt:0];
+}
+
+// This is the callback that is called when the OS has a frame to deliver to us.
+// Starts being called when [_captureSession startRunning] is called. Stopped
+// similarly.
+// Parameter videoFrame contains the image. The format, size, and frequency
+// were setup earlier.
+// Returns 0 on success, -1 otherwise.
+- (void)captureOutput:(QTCaptureOutput *)captureOutput
+ didOutputVideoFrame:(CVImageBufferRef)videoFrame
+ withSampleBuffer:(QTSampleBuffer *)sampleBuffer
+ fromConnection:(QTCaptureConnection *)connection{
+
+ if(YES == [_rLock tryLock])
+ {
+ [_rLock lock];
+ }
+ else
+ {
+ return;
+ }
+
+ if(NO == _OSSupported)
+ {
+ return;
+ }
+
+ const int LOCK_FLAGS = 0; // documentation says to pass 0
+
+ // get size of the frame
+ CVPixelBufferLockBaseAddress(videoFrame, LOCK_FLAGS);
+ void* baseAddress = CVPixelBufferGetBaseAddress(videoFrame);
+ size_t bytesPerRow = CVPixelBufferGetBytesPerRow(videoFrame);
+ int frameHeight = CVPixelBufferGetHeight(videoFrame);
+ CVPixelBufferUnlockBaseAddress(videoFrame, LOCK_FLAGS);
+
+ if(_owner)
+ {
+
+ int frameSize = bytesPerRow * frameHeight; // 32 bit ARGB format
+ CVBufferRetain(videoFrame);
+ VideoCaptureCapability tempCaptureCapability;
+ tempCaptureCapability.width = _frameWidth;
+ tempCaptureCapability.height = _frameHeight;
+ tempCaptureCapability.maxFPS = _frameRate;
+ // TODO(wu) : Update actual type and not hard-coded value.
+ tempCaptureCapability.rawType = kVideoBGRA;
+
+ _owner->IncomingFrame((unsigned char*)baseAddress,
+ frameSize,
+ tempCaptureCapability,
+ 0);
+
+ CVBufferRelease(videoFrame);
+ }
+
+ _framesDelivered++;
+ _framesRendered++;
+
+ if(YES == [_rLock locked])
+ {
+ [_rLock unlock];
+ }
+}
+
+@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h
new file mode 100644
index 0000000..5ef0b96
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_qtkit_utility.h
+ *
+ */
+
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
+
+#define MAX_NAME_LENGTH 1024
+
+#define QTKIT_MIN_WIDTH 0
+#define QTKIT_MAX_WIDTH 2560
+#define QTKIT_DEFAULT_WIDTH 352
+
+#define QTKIT_MIN_HEIGHT 0
+#define QTKIT_MAX_HEIGHT 1440
+#define QTKIT_DEFAULT_HEIGHT 288
+
+#define QTKIT_MIN_FRAME_RATE 1
+#define QTKIT_MAX_FRAME_RATE 60
+#define QTKIT_DEFAULT_FRAME_RATE 30
+
+#define RELEASE_AND_CLEAR(p) if (p) { (p) -> Release () ; (p) = NULL ; }
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h
new file mode 100644
index 0000000..f4008a4
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// video_capture_recursive_lock.h
+//
+//
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
+
+#import <Foundation/Foundation.h>
+
+@interface VideoCaptureRecursiveLock : NSRecursiveLock <NSLocking> {
+ BOOL _locked;
+}
+
+@property BOOL locked;
+
+- (void)lock;
+- (void)unlock;
+
+@end
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm
new file mode 100644
index 0000000..d9df5cb
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm
@@ -0,0 +1,33 @@
+//
+// video_capture_recursive_lock.mm
+//
+//
+
+#import "video_capture_recursive_lock.h"
+
+@implementation VideoCaptureRecursiveLock
+
+@synthesize locked = _locked;
+
+- (id)init{
+ self = [super init];
+ if(nil == self){
+ return nil;
+ }
+
+ [self setLocked:NO];
+ return self;
+}
+
+- (void)lock{
+ [self setLocked:YES];
+ [super lock];
+}
+
+- (void)unlock{
+ [self setLocked:NO];
+ [super unlock];
+}
+
+
+@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc
new file mode 100644
index 0000000..f70923e
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc
@@ -0,0 +1,1387 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_quick_time.cc
+ *
+ */
+
+
+#include "video_capture_quick_time.h"
+
+#include "CriticalSectionWrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "trace.h"
+#include <unistd.h>
+
+namespace webrtc
+{
+
+VideoCaptureMacQuickTime::VideoCaptureMacQuickTime(WebRtc_Word32 iID) :
+ VideoCaptureImpl(iID), // super class constructor
+ _id(iID),
+ _isCapturing(false),
+ _captureCapability(),
+ _grabberCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _videoMacCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _terminated(true), _grabberUpdateThread(NULL),
+ _grabberUpdateEvent(NULL), _captureGrabber(NULL), _captureDevice(NULL),
+ _captureVideoType(kVideoUnknown), _captureIsInitialized(false),
+ _gWorld(NULL), _captureChannel(0), _captureSequence(NULL),
+ _sgPrepared(false), _sgStarted(false), _trueCaptureWidth(0),
+ _trueCaptureHeight(0), _captureDeviceList(),
+ _captureDeviceListTime(0), _captureCapabilityList()
+
+{
+ _captureCapability.width = START_CODEC_WIDTH;
+ _captureCapability.height = START_CODEC_HEIGHT;
+ memset(_captureDeviceDisplayName, 0, sizeof(_captureDeviceDisplayName));
+}
+
+VideoCaptureMacQuickTime::~VideoCaptureMacQuickTime()
+{
+
+
+ VideoCaptureTerminate();
+
+ if (_videoMacCritsect)
+ {
+ delete _videoMacCritsect;
+ }
+ if (_grabberCritsect)
+ {
+ delete _grabberCritsect;
+ }
+
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTime::Init(
+ const WebRtc_Word32 id, const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ const WebRtc_Word32 nameLength =
+ (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
+ if (nameLength > kVideoCaptureUniqueNameLength)
+ return -1;
+
+ // Store the device name
+ _deviceUniqueId = new WebRtc_UWord8[nameLength + 1];
+ memset(_deviceUniqueId, 0, nameLength + 1);
+ memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
+
+ // Check OSX version
+ OSErr err = noErr;
+ long version;
+
+ _videoMacCritsect->Enter();
+ if (!_terminated)
+ {
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Already Initialized", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ err = Gestalt(gestaltSystemVersion, &version);
+ if (err != noErr)
+ {
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not retrieve OS version", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d OS X version: %x,", __FUNCTION__, __LINE__, version);
+ if (version < 0x00001040) // Older version than Mac OSX 10.4
+ {
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d OS version not supported", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ err = Gestalt(gestaltQuickTime, &version);
+ if (err != noErr)
+ {
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not retrieve QuickTime version",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d QuickTime version: %x", __FUNCTION__, __LINE__,
+ version);
+ if (version < 0x07000000) // QT v. 7.x or newer (QT 5.0.2 0x05020000)
+ {
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d EnterMovies()", __FUNCTION__, __LINE__);
+ EnterMovies();
+
+ if (VideoCaptureSetCaptureDevice((char*) deviceUniqueIdUTF8,
+ kVideoCaptureProductIdLength) == -1)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d failed to set capture device: %s", __FUNCTION__,
+ __LINE__, deviceUniqueIdUTF8);
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+
+ _terminated = false;
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d successful initialization", __FUNCTION__, __LINE__);
+ _videoMacCritsect->Leave();
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTime::StartCapture(
+ const VideoCaptureCapability& capability)
+{
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "%s:%d "
+ "capability.width=%d, capability.height=%d ,capability.maxFPS=%d "
+ "capability.expectedCaptureDelay=%d, capability.interlaced=%d",
+ __FUNCTION__, __LINE__, capability.width, capability.height,
+ capability.maxFPS, capability.expectedCaptureDelay,
+ capability.interlaced);
+
+ _captureCapability.width = capability.width;
+ _captureCapability.height = capability.height;
+
+ if (VideoCaptureRun() == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTime::StopCapture()
+{
+
+ if (VideoCaptureStop() == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+bool VideoCaptureMacQuickTime::CaptureStarted()
+{
+ return _isCapturing;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTime::CaptureSettings(
+ VideoCaptureCapability& settings)
+{
+ settings.width = _captureCapability.width;
+ settings.height = _captureCapability.height;
+ settings.maxFPS = 0;
+ return 0;
+}
+
+int VideoCaptureMacQuickTime::VideoCaptureTerminate()
+{
+ VideoCaptureStop();
+
+ _videoMacCritsect->Enter();
+ if (_terminated)
+ {
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Already terminated", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ _grabberCritsect->Enter();
+
+ // Stop the camera/sequence grabber
+ // Resets: _captureSequence, _sgStarted
+ StopQuickTimeCapture();
+
+ // Remove local video settings
+ // Resets: _gWorld, _captureCapability.width, _captureCapability.height
+ RemoveLocalGWorld();
+ DisconnectCaptureDevice();
+
+ if (_grabberUpdateThread)
+ _grabberUpdateThread->SetNotAlive();
+
+ _grabberCritsect->Leave();
+
+ if (_grabberUpdateEvent)
+ _grabberUpdateEvent->Set();
+
+ SLEEP(1);
+ _grabberCritsect->Enter();
+
+ if (_grabberUpdateThread)
+ {
+ _grabberUpdateThread->Stop();
+ delete _grabberUpdateThread;
+ _grabberUpdateThread = NULL;
+ }
+ if (_grabberUpdateEvent)
+ {
+ delete _grabberUpdateEvent;
+ _grabberUpdateEvent = NULL;
+ }
+
+ // Close the sequence grabber
+ if (_captureGrabber)
+ {
+ SGRelease(_captureGrabber);
+ _captureGrabber = NULL;
+ CloseComponent(_captureGrabber);
+ _captureDevice = NULL;
+ }
+ _captureVideoType = kVideoUnknown;
+
+ // Delete capture device list
+ ListItem* item = _captureDeviceList.First();
+ while (item)
+ {
+ delete static_cast<unsigned char*> (item->GetItem());
+ _captureDeviceList.Erase(item);
+ item = _captureDeviceList.First();
+ }
+ _captureDeviceListTime = 0;
+
+ _terminated = true;
+
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+
+ return 0;
+}
+
+int VideoCaptureMacQuickTime::UpdateCaptureSettings(int channel,
+ webrtc::VideoCodec& inst,
+ bool def)
+{
+
+ if (channel < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Invalid channel number: %d", __FUNCTION__,
+ __LINE__, channel);
+ return -1;
+ }
+
+ // the size has changed, we need to change our setup
+ _videoMacCritsect->Enter();
+
+ // Stop capturing, if we are...
+ _grabberCritsect->Enter();
+
+ bool wasCapturing = false;
+ StopQuickTimeCapture(&wasCapturing);
+
+ // Create a new offline GWorld to receive captured frames
+ RemoveLocalGWorld();
+
+ if (CreateLocalGWorld(inst.width, inst.height) == -1)
+ {
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ // Error already logged
+ return -1;
+ }
+ _captureCapability.width = inst.width;
+ _captureCapability.height = inst.height;
+
+ // Connect the capture device to our offline GWorld
+ // if we already have a capture device selected.
+ if (_captureDevice)
+ {
+ DisconnectCaptureDevice();
+ if (ConnectCaptureDevice() == -1)
+ {
+ // Error already logged
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+ }
+
+ // Start capture if we did before
+ if (wasCapturing)
+ {
+ if (StartQuickTimeCapture() == -1)
+ {
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Failed to start capturing", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+ }
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+
+ return 0;
+}
+
+// Creates an off screen graphics world used for converting
+// captured video frames if we can't get a format we want.
+// Assumed protected by critsects
+int VideoCaptureMacQuickTime::CreateLocalGWorld(int width, int height)
+{
+ if (_gWorld)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d GWorld already created", __FUNCTION__, __LINE__);
+ return -1;
+ }
+ if (width == 0 || height == 0)
+ {
+ return -1;
+ }
+
+ Rect captureRect;
+ captureRect.left = 0;
+ captureRect.top = 0;
+ captureRect.right = width;
+ captureRect.bottom = height;
+
+ // Create a GWorld in same size as we want to send to the codec
+ if (QTNewGWorld(&(_gWorld), k2vuyPixelFormat, &captureRect, 0, NULL, 0)
+ != noErr)
+ {
+ return -1;
+ }
+ _captureCapability.width = width;
+ _captureCapability.height = height;
+
+ if (!LockPixels(GetGWorldPixMap(_gWorld)))
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not lock pixmap. Continuing anyhow",
+ __FUNCTION__, __LINE__);
+ }
+
+ CGrafPtr theOldPort;
+ GDHandle theOldDevice;
+ GetGWorld(&theOldPort, &theOldDevice); // Gets the result from QTGetNewGWorld
+ SetGWorld(_gWorld, NULL); // Sets the new GWorld
+ BackColor( blackColor); // Changes the color on the graphic port
+ ForeColor( whiteColor);
+ EraseRect(&captureRect);
+ SetGWorld(theOldPort, theOldDevice);
+
+ return 0;
+}
+
+// Assumed critsect protected
+int VideoCaptureMacQuickTime::RemoveLocalGWorld()
+{
+ if (!_gWorld)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d !gWorld", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ DisposeGWorld(_gWorld);
+ _gWorld = NULL;
+ _captureCapability.width = START_CODEC_WIDTH;
+ _captureCapability.height = START_CODEC_HEIGHT;
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d GWorld has been removed", __FUNCTION__, __LINE__);
+ return 0;
+}
+
+// ConnectCaptureDevice
+// This function prepares the capture device
+// with the wanted settings, but the capture
+// device isn't started.
+//
+// Assumed critsect protected
+int VideoCaptureMacQuickTime::ConnectCaptureDevice()
+{
+ // Prepare the capture grabber if a capture device is already set
+ if (!_captureGrabber)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d No capture device is selected", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+ if (_captureIsInitialized)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Capture device is already initialized",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ if (!_gWorld)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d No GWorld is created", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ OSErr err = noErr;
+ long flags = 0;
+
+ // Connect the camera to our offline GWorld
+ // We won't use the GWorld if we get the format we want
+ // from the camera.
+ if (SGSetGWorld(_captureGrabber, _gWorld, NULL ) != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not connect capture device", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+ if (SGSetDataRef(_captureGrabber, 0, 0, seqGrabDontMakeMovie) != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not configure capture device", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Set our capture callback
+ if (SGSetDataProc(_captureGrabber, NewSGDataUPP(SendProcess), (long) this)
+ != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not set capture callback. Unable to receive "
+ "frames", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ // Create a video channel to the sequence grabber
+ if (SGNewChannel(_captureGrabber, VideoMediaType, &_captureChannel)
+ != noErr) // Takes time!!!
+ {
+ return -1;
+ }
+
+ // Get a list with all capture devices to choose the one we want.
+ SGDeviceList deviceList = NULL;
+ if (SGGetChannelDeviceList(_captureChannel, sgDeviceListIncludeInputs,
+ &deviceList) != noErr)
+ {
+
+ }
+
+ int numDevicesTypes = (*deviceList)->count;
+ bool captureDeviceFound = false;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Found %d channel devices", __FUNCTION__, __LINE__,
+ numDevicesTypes);
+
+ // Loop through all devices to get the one we want.
+ for (int i = 0; i < numDevicesTypes; i++)
+ {
+ SGDeviceName deviceTypeName = (*deviceList)->entry[i];
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Inspecting device number: %d", __FUNCTION__,
+ __LINE__, i);
+ // Get the list with input devices
+ if (deviceTypeName.inputs)
+ {
+ SGDeviceInputList inputList = deviceTypeName.inputs;
+ int numInputDev = (*inputList)->count;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Device has %d inputs", __FUNCTION__, __LINE__,
+ numInputDev);
+ for (int inputDevIndex = 0;
+ inputDevIndex < numInputDev;
+ inputDevIndex++)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
+ _id, "%s:%d Inspecting input number: %d",
+ __FUNCTION__, __LINE__, inputDevIndex);
+ SGDeviceInputName deviceInputName =
+ (*inputList)->entry[inputDevIndex];
+ char devInName[64];
+ memset(devInName, 0, 64);
+
+ // SGDeviceInputName::name is a Str63, defined as a Pascal string.
+ // (Refer to MacTypes.h)
+ CFIndex devInNameLength =
+ PascalStringToCString(deviceInputName.name, devInName,
+ sizeof(devInName));
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Converted pascal string with length:%d "
+ "to: %s", __FUNCTION__, __LINE__,
+ sizeof(devInName), devInName);
+ if (devInNameLength < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Failed to convert device name from "
+ "pascal string to c string", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ if (!strcmp(devInName, _captureDeviceDisplayName))
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d We have found our device: %s",
+ __FUNCTION__, __LINE__,
+ _captureDeviceDisplayName);
+
+ if (SGSetChannelDevice(_captureChannel, deviceTypeName.name)
+ != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not set capture device type: "
+ "%s",__FUNCTION__, __LINE__,
+ deviceTypeName.name);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Capture device type is: %s",
+ __FUNCTION__, __LINE__, deviceTypeName.name);
+ if (SGSetChannelDeviceInput(_captureChannel, inputDevIndex)
+ != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not set SG device",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Capture device: %s has successfully "
+ "been set", __FUNCTION__, __LINE__,
+ _captureDeviceDisplayName);
+ captureDeviceFound = true;
+ break;
+ }
+ }
+ if (captureDeviceFound)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Capture device found, breaking from loops",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+ }
+ }
+ err = SGDisposeDeviceList(_captureGrabber, deviceList);
+
+ if (!captureDeviceFound)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Failed to find capture device: %s. Returning -1",
+ __FUNCTION__, __LINE__, _captureDeviceDisplayName);
+ return -1;
+ }
+
+ // Set the size we want from the capture device
+ Rect captureSize;
+ captureSize.left = 0;
+ captureSize.top = 0;
+ captureSize.right = _captureCapability.width;
+ captureSize.bottom = _captureCapability.height;
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Using capture rect: l:%d t:%d r:%d b:%d", __FUNCTION__,
+ __LINE__, captureSize.left, captureSize.top,
+ captureSize.right, captureSize.bottom);
+
+ err = SGSetChannelBounds(_captureChannel, &captureSize);
+ if (err == noErr)
+ {
+ err = SGSetChannelUsage(_captureChannel, flags | seqGrabRecord);
+ }
+ if (err != noErr)
+ {
+ SGDisposeChannel(_captureGrabber, _captureChannel);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error setting SG channel to device", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Find out what video format we'll get from the capture device.
+ OSType compType;
+ err = SGGetVideoCompressorType(_captureChannel, &compType);
+
+ // Convert the Apple video format name to a VideoCapture name.
+ if (compType == k2vuyPixelFormat)
+ {
+ _captureVideoType = kVideoUYVY;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Device delivers UYUV formatted frames",
+ __FUNCTION__, __LINE__);
+ }
+ else if (compType == kYUVSPixelFormat)
+ {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Device delivers YUY2 formatted frames",
+ __FUNCTION__, __LINE__);
+ _captureVideoType = kVideoYUY2;
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Device delivers frames in an unknown format: 0x%x. "
+ "Consult QuickdrawTypes.h",
+ __FUNCTION__, __LINE__, compType);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Device delivers frames in an unknown format.",
+ __FUNCTION__, __LINE__);
+ _captureVideoType = kVideoUnknown;
+ }
+
+ if (SGPrepare(_captureGrabber, false, true) != noErr)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error starting sequence grabber", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Try to set the codec size as capture size.
+ err = SGSetChannelBounds(_captureChannel, &captureSize);
+
+ // Check if we really will get the size we asked for.
+ ImageDescriptionHandle imageDesc = (ImageDescriptionHandle) NewHandle(0);
+ err = SGGetChannelSampleDescription(_captureChannel, (Handle) imageDesc);
+
+ _trueCaptureWidth = (**imageDesc).width;
+ _trueCaptureHeight = (**imageDesc).height;
+
+ DisposeHandle((Handle) imageDesc);
+
+ _captureIsInitialized = true;
+ _sgPrepared = true;
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Success starting sequence grabber", __FUNCTION__,
+ __LINE__);
+
+ return 0;
+}
+
+// Assumed critsect protected
+int VideoCaptureMacQuickTime::DisconnectCaptureDevice()
+{
+ if (_sgStarted)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Capture device is still running. Returning -1",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ if (!_sgPrepared)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d No capture device connected", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Close the capture channel
+ SGStop(_captureGrabber);
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d !!!! releasing sg stuff", __FUNCTION__, __LINE__);
+ SGDisposeChannel(_captureGrabber, _captureChannel);
+ SGRelease(_captureGrabber);
+ CloseComponent(_captureGrabber);
+
+ // Reset all values
+ _captureChannel = NULL;
+ _captureVideoType = kVideoUnknown;
+ _trueCaptureWidth = 0;
+ _trueCaptureHeight = 0;
+ _captureIsInitialized = false;
+ _sgPrepared = false;
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Sequence grabber removed", __FUNCTION__, __LINE__);
+
+ return 0;
+}
+
+// StartQuickTimeCapture
+//
+// Actually starts the camera
+//
+int VideoCaptureMacQuickTime::StartQuickTimeCapture()
+{
+ _grabberCritsect->Enter();
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Attempting to start sequence grabber", __FUNCTION__,
+ __LINE__);
+
+ if (_sgStarted)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Sequence grabber already started", __FUNCTION__,
+ __LINE__);
+ return 0;
+ }
+ if (!_sgPrepared)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Sequence grabber not prepared properly",
+ __FUNCTION__, __LINE__);
+ return 0;
+ }
+
+ if (SGStartRecord(_captureGrabber) != noErr)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error starting sequence grabber", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ Rect captureRect = { 0, 0, 0, 0 };
+ MatrixRecord scaleMatrix;
+ ImageDescriptionHandle imageDesc = (ImageDescriptionHandle) NewHandle(0);
+
+ // Get the sample description for the channel, which is the same as for the
+ // capture device
+ if (SGGetChannelSampleDescription(_captureChannel, (Handle) imageDesc)
+ != noErr)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error accessing device properties", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Create a scale matrix to scale the captured image
+ // Needed if we don't get the size wanted from the camera
+ captureRect.right = (**imageDesc).width;
+ captureRect.bottom = (**imageDesc).height;
+
+ Rect codecRect;
+ codecRect.left = 0;
+ codecRect.top = 0;
+ codecRect.right = _captureCapability.width;
+ codecRect.bottom = _captureCapability.height;
+ RectMatrix(&scaleMatrix, &captureRect, &codecRect);
+
+ // Start grabbing images from the capture device to _gWorld
+ if (DecompressSequenceBegin(&_captureSequence, imageDesc, _gWorld, NULL,
+ NULL, &scaleMatrix, srcCopy, (RgnHandle) NULL,
+ NULL, codecNormalQuality, bestSpeedCodec)
+ != noErr)
+ {
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error starting decompress sequence", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+ DisposeHandle((Handle) imageDesc);
+ _sgStarted = true;
+ _grabberCritsect->Leave();
+ return 0;
+}
+
+int VideoCaptureMacQuickTime::StopQuickTimeCapture(bool* wasCapturing)
+{
+ _grabberCritsect->Enter();
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "%s:%d wasCapturing=%d", __FUNCTION__, __LINE__, wasCapturing);
+
+ if (!_sgStarted)
+ {
+ if (wasCapturing)
+ *wasCapturing = false;
+
+ _grabberCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Sequence grabber was never started", __FUNCTION__,
+ __LINE__);
+ return 0;
+ }
+
+ if (wasCapturing)
+ *wasCapturing = true;
+
+ OSErr error = noErr;
+ error = SGStop(_captureGrabber);
+ CDSequenceEnd(_captureSequence);
+ _captureSequence = NULL;
+ _sgStarted = false;
+
+ _grabberCritsect->Leave();
+ if (error != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not stop sequence grabber", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ return 0;
+}
+
+//-------------------------------------------------
+//
+// Thread/function to keep capture device working
+//
+//-------------------------------------------------
+
+
+//
+// GrabberUpdateThread / GrabberUpdateProcess
+//
+// Called at a certain time interval to tell
+// the capture device / SequenceGrabber to
+// actually work.
+bool VideoCaptureMacQuickTime::GrabberUpdateThread(void* obj)
+{
+ return static_cast<VideoCaptureMacQuickTime*> (obj)->GrabberUpdateProcess();
+}
+
+bool VideoCaptureMacQuickTime::GrabberUpdateProcess()
+{
+ _grabberUpdateEvent->Wait(30);
+
+ if (_isCapturing == false)
+ return false;
+
+ _grabberCritsect->Enter();
+ if (_captureGrabber)
+ {
+ if (SGIdle(_captureGrabber) != noErr)
+ {
+ }
+ }
+ _grabberCritsect->Leave();
+ return true;
+}
+
+//
+// VideoCaptureStop
+//
+// Stops the capture device
+//
+int VideoCaptureMacQuickTime::VideoCaptureStop()
+{
+ if (_grabberUpdateThread)
+ {
+ _grabberUpdateThread->Stop();
+ }
+
+ _videoMacCritsect->Enter();
+ _grabberCritsect->Enter();
+ int retVal = StopQuickTimeCapture();
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ if (retVal == -1)
+ {
+ return -1;
+ }
+
+ _isCapturing = false;
+ return 0;
+}
+
+//
+// VideoCaptureRun
+//
+// Starts the capture device and creates
+// the update thread.
+//
+int VideoCaptureMacQuickTime::VideoCaptureRun()
+{
+ _videoMacCritsect->Enter();
+ _grabberCritsect->Enter();
+
+ int res = StartQuickTimeCapture();
+
+ // Create the thread for updating sequence grabber if not created earlier
+ if (!_grabberUpdateThread)
+ {
+ _grabberUpdateEvent = EventWrapper::Create();
+ _grabberUpdateThread = ThreadWrapper::CreateThread(
+ VideoCaptureMacQuickTime::GrabberUpdateThread, this, kHighPriority);
+ unsigned int id;
+ _grabberUpdateThread->Start(id);
+ }
+ else
+ {
+ unsigned int id;
+ _grabberUpdateThread->Start(id);
+ }
+
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+
+ _isCapturing = true;
+ return res;
+}
+
+// ----------------------------------------------------------------------
+//
+// SendProcess
+// sequence grabber data procedure
+//
+// This function is called by the capture device as soon as a new
+// frame is available.
+//
+//
+// SendFrame
+//
+// The non-static function used by the capture device callback
+//
+// Input:
+// sgChannel: the capture device channel generating the callback
+// data: the video frame
+// length: the data length in bytes
+// grabTime: time stamp generated by the capture device / sequece grabber
+//
+// ----------------------------------------------------------------------
+
+OSErr VideoCaptureMacQuickTime::SendProcess(SGChannel sgChannel, Ptr p,
+ long len, long* /*offset*/,
+ long /*chRefCon*/, TimeValue time,
+ short /*writeType*/, long refCon)
+{
+ VideoCaptureMacQuickTime* videoEngine =
+ reinterpret_cast<VideoCaptureMacQuickTime*> (refCon);
+ return videoEngine->SendFrame(sgChannel, (char*) p, len, time);
+}
+
+int VideoCaptureMacQuickTime::SendFrame(SGChannel /*sgChannel*/, char* data,
+ long length, TimeValue /*grabTime*/)
+{
+ if (!_sgPrepared)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Sequence Grabber is not initialized", __FUNCTION__,
+ __LINE__);
+ return 0;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Frame has been delivered\n", __FUNCTION__, __LINE__);
+
+ CodecFlags ignore;
+ _grabberCritsect->Enter();
+ if (_gWorld)
+ {
+ // Will be set to true if we don't recognize the size and/or video
+ // format.
+ bool convertFrame = false;
+ WebRtc_Word32 width = 352;
+ WebRtc_Word32 height = 288;
+ WebRtc_Word32 frameSize = 0;
+
+ VideoCaptureCapability captureCapability;
+ captureCapability.width = width;
+ captureCapability.height = height;
+ captureCapability.maxFPS = 30;
+
+ switch (_captureVideoType)
+ {
+ case kVideoUYVY:
+ captureCapability.rawType = kVideoUYVY;
+ break;
+ case kVideoYUY2:
+ captureCapability.rawType = kVideoYUY2;
+ break;
+ case kVideoI420:
+ captureCapability.rawType = kVideoI420;
+ break;
+ default:
+ captureCapability.rawType = kVideoI420;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
+ _id, "%s:%d raw = I420 by default\n",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+
+ // Convert the camera video type to something VideoEngine can work with
+ // Check if we need to downsample the incomming frame.
+ switch (_captureVideoType)
+ {
+ case kVideoUYVY:
+ case kVideoYUY2:
+ frameSize = (width * height * 16) >> 3; // 16 is for YUY2 format
+ if (width == _captureCapability.width || height
+ == _captureCapability.height)
+ {
+ // Ok format and size, send the frame to super class
+ IncomingFrame((WebRtc_UWord8*) data,
+ (WebRtc_Word32) frameSize, captureCapability,
+ TickTime::MillisecondTimestamp());
+
+ }
+ else if (width == _trueCaptureWidth && height
+ == _trueCaptureHeight)
+ {
+ // We need to scale the picture to correct size...
+ // This happens for cameras not supporting all sizes.
+ // E.g. older built-in iSight doesn't support QCIF.
+
+ // Convert the incoming frame into our GWorld.
+ int res =
+ DecompressSequenceFrameS(_captureSequence, data,
+ length, 0, &ignore, NULL);
+ if (res != noErr && res != -8976) // 8796 == black frame
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Captured black frame. Not "
+ "processing it", __FUNCTION__, __LINE__);
+ _grabberCritsect->Leave();
+ return 0;
+ }
+
+ // Copy the frame from the PixMap to our video buffer
+ PixMapHandle pixMap = GetGWorldPixMap(_gWorld);
+
+ // Lock the image data in the GWorld.
+ LockPixels(pixMap);
+
+ // Get a pointer to the pixel data.
+ Ptr capturedFrame = GetPixBaseAddr(pixMap);
+
+ // Send the converted frame out to super class
+ IncomingFrame((WebRtc_UWord8*) data,
+ (WebRtc_Word32) frameSize, captureCapability,
+ TickTime::MillisecondTimestamp());
+
+ // Unlock the image data to get ready for the next frame.
+ UnlockPixels(pixMap);
+ }
+ else
+ {
+ // Not a size we recognize, use the Mac internal scaling...
+ convertFrame = true;
+ WEBRTC_TRACE(webrtc::kTraceDebug,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Not correct incoming stream size for "
+ "the format and configured size",
+ __FUNCTION__, __LINE__);
+ }
+ break;
+ default:
+
+ // Not a video format we recognize, use the Mac internal scaling
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture,
+ _id, "%s:%d Unknown video frame format (default)",
+ __FUNCTION__, __LINE__);
+ convertFrame = true;
+ break;
+ }
+
+ if (convertFrame)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Unrecognized frame format. Converting frame",
+ __FUNCTION__, __LINE__);
+
+ // We don't recognise the input format. Convert to UYVY, I420 is not
+ // supported on osx. Decompress the grabbed frame into the GWorld,
+ // i.e from webcam format to ARGB (RGB24), and extract the frame.
+ int res = DecompressSequenceFrameS(_captureSequence, data, length,
+ 0, &ignore, NULL);
+ if (res != noErr && res != -8976) // 8796 means a black frame
+ {
+ _grabberCritsect->Leave();
+ return 0;
+ }
+
+ // Copy the frame from the PixMap to our video buffer
+ PixMapHandle rgbPixMap = GetGWorldPixMap(_gWorld);
+ LockPixels(rgbPixMap);
+ Ptr capturedFrame = GetPixBaseAddr(rgbPixMap);
+
+ // Get the picture size
+ int width = (*rgbPixMap)->bounds.right;
+ int height = (*rgbPixMap)->bounds.bottom;
+
+ // 16 is for YUY2 format.
+ WebRtc_Word32 frameSize = (width * height * 16) >> 3;
+
+ // Ok format and size, send the frame to super class
+ IncomingFrame((WebRtc_UWord8*) data, (WebRtc_Word32) frameSize,
+ captureCapability, TickTime::MillisecondTimestamp());
+
+ UnlockPixels(rgbPixMap);
+ }
+
+ // Tell the capture device it's ok to update.
+ SGUpdate(_captureGrabber, NULL);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "%s:%d No GWorld created, but frames are being delivered",
+ __FUNCTION__, __LINE__);
+ }
+
+ _grabberCritsect->Leave();
+ return 0;
+}
+
+int VideoCaptureMacQuickTime::VideoCaptureInitThreadContext()
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d ", __FUNCTION__, __LINE__);
+ _videoMacCritsect->Enter();
+ EnterMoviesOnThread( kQTEnterMoviesFlagDontSetComponentsThreadMode);
+ _videoMacCritsect->Leave();
+ return 0;
+}
+
+//
+//
+// Functions for handling capture devices
+//
+//
+
+VideoCaptureMacQuickTime::VideoCaptureMacName::VideoCaptureMacName() :
+ _size(0)
+{
+ memset(_name, 0, kVideoCaptureMacNameMaxSize);
+}
+
+int VideoCaptureMacQuickTime::VideoCaptureSetCaptureDevice(
+ const char* deviceName, int size)
+{
+
+
+ _videoMacCritsect->Enter();
+ bool wasCapturing = false;
+
+ _grabberCritsect->Enter();
+ if (_captureGrabber)
+ {
+ // Stop grabbing, disconnect and close the old capture device
+ StopQuickTimeCapture(&wasCapturing);
+ DisconnectCaptureDevice();
+ CloseComponent(_captureGrabber);
+ _captureDevice = NULL;
+ _captureGrabber = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Old capture device removed", __FUNCTION__,
+ __LINE__);
+ }
+
+ if (deviceName == NULL || size == 0)
+ {
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return 0;
+ }
+
+ if (size < 0)
+ {
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d 'size' is not valid", __FUNCTION__, __LINE__);
+ return 0;
+ }
+
+ ComponentDescription compCaptureType;
+
+ // Define the component we want to open
+ compCaptureType.componentType = SeqGrabComponentType;
+ compCaptureType.componentSubType = 0;
+ compCaptureType.componentManufacturer = 0;
+ compCaptureType.componentFlags = 0;
+ compCaptureType.componentFlagsMask = 0;
+
+ long numSequenceGrabbers = CountComponents(&compCaptureType);
+
+ // loop through the available grabbers and open the first possible
+ for (int i = 0; i < numSequenceGrabbers; i++)
+ {
+ _captureDevice = FindNextComponent(0, &compCaptureType);
+ _captureGrabber = OpenComponent(_captureDevice);
+ if (_captureGrabber != NULL)
+ {
+ // We've found a sequencegrabber that we could open
+ if (SGInitialize(_captureGrabber) != noErr)
+ {
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Could not initialize sequence grabber",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ break;
+ }
+ if (i == numSequenceGrabbers - 1)
+ {
+ // Couldn't open a sequence grabber
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not open a sequence grabber",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ }
+
+ if (!_gWorld)
+ {
+ // We don't have a GWorld. Create one to enable early preview
+ // without calling SetSendCodec
+ if (CreateLocalGWorld(_captureCapability.width,
+ _captureCapability.height) == -1)
+ {
+ // Error already logged
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+ }
+ // Connect the camera with our GWorld
+ int cpySize = size;
+ if ((unsigned int) size > sizeof(_captureDeviceDisplayName))
+ {
+ cpySize = sizeof(_captureDeviceDisplayName);
+ }
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Copying %d chars from deviceName to "
+ "_captureDeviceDisplayName (size=%d)\n",
+ __FUNCTION__, __LINE__, cpySize, size);
+ memcpy(_captureDeviceDisplayName, deviceName, cpySize);
+ if (ConnectCaptureDevice() == -1)
+ {
+ // Error already logged
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+
+ if (StartQuickTimeCapture() == -1)
+ {
+ // Error already logged
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return -1;
+ }
+ _grabberCritsect->Leave();
+ _videoMacCritsect->Leave();
+ return 0;
+}
+
+bool VideoCaptureMacQuickTime::IsCaptureDeviceSelected()
+{
+ _grabberCritsect->Leave();
+ return (_captureIsInitialized) ? true : false;
+ _grabberCritsect->Leave();
+}
+
+/**
+ Convert a Pascal string to a C string.
+
+ \param[in] pascalString
+ Pascal string to convert. Pascal strings contain the number of
+ characters in the first byte and are not null-terminated.
+
+ \param[out] cString
+ The C string buffer into which to copy the converted string.
+
+ \param[in] bufferSize
+ The size of the C string buffer in bytes.
+
+ \return The number of characters in the string on success and -1 on failure.
+ */
+CFIndex VideoCaptureMacQuickTime::PascalStringToCString(
+ const unsigned char* pascalString, char* cString, CFIndex bufferSize)
+{
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
+ "%s:%d Converting pascal string to c string", __FUNCTION__,
+ __LINE__);
+ if (pascalString == NULL)
+ {
+ return -1;
+ }
+
+ if (cString == NULL)
+ {
+ return -1;
+ }
+
+ if (bufferSize == 0)
+ {
+ return -1;
+ }
+
+ CFIndex cStringLength = 0;
+ CFIndex maxStringLength = bufferSize - 1;
+
+ CFStringRef cfString = CFStringCreateWithPascalString(
+ NULL, pascalString, kCFStringEncodingMacRoman);
+ if (cfString == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
+ "%s:%d Error in CFStringCreateWithPascalString()",
+ __FUNCTION__, __LINE__);
+ CFRelease(cfString);
+ return -1;
+ }
+
+ CFIndex cfLength = CFStringGetLength(cfString);
+ cStringLength = cfLength;
+ if (cfLength > maxStringLength)
+ {
+ cStringLength = maxStringLength;
+ }
+
+ Boolean success = CFStringGetCString(cfString, cString, bufferSize,
+ kCFStringEncodingMacRoman);
+
+ // Ensure the problem isn't insufficient buffer length.
+ // This is fine; we will return a partial string.
+ if (success == false && cfLength <= maxStringLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
+ "%s:%d Error in CFStringGetCString()", __FUNCTION__,
+ __LINE__);
+ CFRelease(cfString);
+ return -1;
+ }
+
+ CFRelease(cfString);
+ return cStringLength;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h
new file mode 100644
index 0000000..84e0667
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_quick_time.h
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
+
+#include <QuickTime/QuickTime.h>
+
+
+#include "../../device_info_impl.h"
+#include "../../video_capture_impl.h"
+#include "list_wrapper.h"
+
+
+#define START_CODEC_WIDTH 352
+#define START_CODEC_HEIGHT 288
+#define SLEEP(x) usleep(x * 1000);
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+
+class VideoCaptureMacQuickTime : public VideoCaptureImpl
+{
+
+public:
+ VideoCaptureMacQuickTime(const WebRtc_Word32 id);
+ virtual ~VideoCaptureMacQuickTime();
+
+ static void Destroy(VideoCaptureModule* module);
+
+ WebRtc_Word32 Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+ virtual WebRtc_Word32 StartCapture(
+ const VideoCaptureCapability& capability);
+ virtual WebRtc_Word32 StopCapture();
+ virtual bool CaptureStarted();
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
+
+ // TODO: remove?
+ int VideoCaptureInitThreadContext();
+ int VideoCaptureTerminate();
+ int VideoCaptureSetCaptureDevice(const char* deviceName, int size);
+ int UpdateCaptureSettings(int channel, webrtc::VideoCodec& inst, bool def);
+ int VideoCaptureRun();
+ int VideoCaptureStop();
+
+protected:
+
+private: // functions
+
+ struct VideoCaptureMacName
+ {
+ VideoCaptureMacName();
+
+ enum { kVideoCaptureMacNameMaxSize = 64};
+ char _name[kVideoCaptureMacNameMaxSize];
+ CFIndex _size;
+ };
+
+ // Timeout value [ms] if we want to create a new device list or not
+ enum { kVideoCaptureDeviceListTimeout = 5000};
+ // Temporary constant allowing this size from builtin iSight webcams.
+ enum { kYuy2_1280_1024_length = 2621440};
+
+private:
+
+ // Capture device callback
+ static OSErr SendProcess(SGChannel sgChannel, Ptr p, long len, long *offset,
+ long chRefCon, TimeValue time, short writeType,
+ long refCon);
+ int SendFrame(SGChannel sgChannel, char* data, long length, TimeValue time);
+
+ // Capture device functions
+ int CreateLocalGWorld(int width, int height);
+ int RemoveLocalGWorld();
+ int ConnectCaptureDevice();
+ int DisconnectCaptureDevice();
+ virtual bool IsCaptureDeviceSelected();
+
+ // Process to make sure the capture device won't stop
+ static bool GrabberUpdateThread(void*);
+ bool GrabberUpdateProcess();
+
+ // Starts and stops the capture
+ int StartQuickTimeCapture();
+ int StopQuickTimeCapture(bool* wasCapturing = NULL);
+
+ static CFIndex PascalStringToCString(const unsigned char* pascalString,
+ char* cString,
+ CFIndex bufferSize);
+
+private: // variables
+ WebRtc_Word32 _id;
+ bool _isCapturing;
+ VideoCaptureCapability _captureCapability;
+ CriticalSectionWrapper* _grabberCritsect;
+ CriticalSectionWrapper* _videoMacCritsect;
+ bool _terminated;
+ webrtc::ThreadWrapper* _grabberUpdateThread;
+ webrtc::EventWrapper* _grabberUpdateEvent;
+ SeqGrabComponent _captureGrabber;
+ Component _captureDevice;
+ char _captureDeviceDisplayName[64];
+ RawVideoType _captureVideoType;
+ bool _captureIsInitialized;
+ GWorldPtr _gWorld;
+ SGChannel _captureChannel;
+ ImageSequence _captureSequence;
+ bool _sgPrepared;
+ bool _sgStarted;
+ int _trueCaptureWidth;
+ int _trueCaptureHeight;
+ ListWrapper _captureDeviceList;
+ unsigned long _captureDeviceListTime;
+ ListWrapper _captureCapabilityList;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc
new file mode 100644
index 0000000..4f3cc1a
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc
@@ -0,0 +1,391 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_quick_time_info.cc
+ *
+ */
+
+#include "../../video_capture_config.h"
+#include "video_capture_quick_time_info.h"
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "trace.h"
+#include "video_capture.h"
+
+namespace webrtc
+{
+
+VideoCaptureMacQuickTimeInfo::VideoCaptureMacQuickTimeInfo(
+ const WebRtc_Word32 iID) :
+ DeviceInfoImpl(iID), _id(iID),
+ _grabberCritsect(CriticalSectionWrapper::CreateCriticalSection())
+{
+}
+
+VideoCaptureMacQuickTimeInfo::~VideoCaptureMacQuickTimeInfo()
+{
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::Init()
+{
+
+ return 0;
+}
+
+WebRtc_UWord32 VideoCaptureMacQuickTimeInfo::NumberOfDevices()
+{
+ int numOfDevices = 0;
+
+ // don't care about these variables... dummy vars to call GetCaptureDevices
+ const int kNameLength = 1024;
+ WebRtc_UWord8 deviceNameUTF8[kNameLength] = "";
+ WebRtc_UWord8 deviceUniqueIdUTF8[kNameLength] = "";
+ WebRtc_UWord8 productUniqueIdUTF8[kNameLength] = "";
+
+ if (GetCaptureDevices(0, deviceNameUTF8, kNameLength, deviceUniqueIdUTF8,
+ kNameLength, productUniqueIdUTF8, kNameLength,
+ numOfDevices) != 0)
+ {
+ return 0;
+ }
+
+ return numOfDevices;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetDeviceName(
+ WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameUTF8Length, WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+{
+
+ int numOfDevices = 0; // not needed for this function
+ return GetCaptureDevices(deviceNumber, deviceNameUTF8,
+ deviceNameUTF8Length, deviceUniqueIdUTF8,
+ deviceUniqueIdUTF8Length, productUniqueIdUTF8,
+ productUniqueIdUTF8Length, numOfDevices);
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::NumberOfCapabilities(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetCapability(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability)
+{
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested, VideoCaptureCapability& resulting)
+{
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
+ WebRtc_UWord32 positionX, WebRtc_UWord32 positionY)
+{
+ return -1;
+}
+
+WebRtc_Word32 VideoCaptureMacQuickTimeInfo::CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "NumberOfCapabilities is not supported on the Mac platform.");
+ return -1;
+}
+
+int VideoCaptureMacQuickTimeInfo::GetCaptureDevices(
+ WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameUTF8Length, WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length, int& numberOfDevices)
+{
+
+
+ numberOfDevices = 0;
+ memset(deviceNameUTF8, 0, deviceNameUTF8Length);
+ memset(deviceUniqueIdUTF8, 0, deviceUniqueIdUTF8Length);
+ memset(productUniqueIdUTF8, 0, productUniqueIdUTF8Length);
+
+ if (deviceNumber < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Invalid deviceNumber", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ Component captureDevice = NULL;
+ SeqGrabComponent captureGrabber = NULL;
+ SGChannel captureChannel = NULL;
+ bool closeChannel = false;
+
+ ComponentDescription compCaptureType;
+
+ compCaptureType.componentType = SeqGrabComponentType;
+ compCaptureType.componentSubType = 0;
+ compCaptureType.componentManufacturer = 0;
+ compCaptureType.componentFlags = 0;
+ compCaptureType.componentFlagsMask = 0;
+
+ // Get the number of sequence grabbers
+ long numSequenceGrabbers = CountComponents(&compCaptureType);
+
+ if (deviceNumber > numSequenceGrabbers)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Invalid deviceNumber", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ if (numSequenceGrabbers <= 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d No sequence grabbers available", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Open a sequence grabber
+ for (int seqGrabberIndex = 0;
+ seqGrabberIndex < numSequenceGrabbers;
+ seqGrabberIndex++)
+ {
+ captureDevice = FindNextComponent(0, &compCaptureType);
+ captureGrabber = OpenComponent(captureDevice);
+ if (captureGrabber != NULL)
+ {
+ // We've found a sequencegrabber
+ if (SGInitialize(captureGrabber) != noErr)
+ {
+ CloseComponent(captureGrabber);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id, "%s:%d Could not init the sequence grabber",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ break;
+ }
+ if (seqGrabberIndex == numSequenceGrabbers - 1)
+ {
+ // Couldn't open a sequence grabber
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not open a sequence grabber",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ }
+
+ // Create a temporary channel to get the names of the capture devices.
+ // Takes time, make this in a nother way...
+ if (SGNewChannel(captureGrabber, VideoMediaType, &captureChannel) != noErr)
+ {
+ // Could not create a video channel...
+ SGRelease(captureGrabber);
+ CloseComponent(captureGrabber);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not create a sequence grabber video channel",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ closeChannel = true;
+
+ // Find the type of capture devices, e.g. USB-devices, Firewire, DV, ...
+ SGDeviceList deviceList = NULL;
+ if (SGGetChannelDeviceList(captureChannel, sgDeviceListIncludeInputs,
+ &deviceList) != noErr)
+ {
+ if (closeChannel)
+ SGDisposeChannel(captureGrabber, captureChannel);
+ if (captureGrabber)
+ {
+ SGRelease(captureGrabber);
+ CloseComponent(captureGrabber);
+ }
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Could not create a device list", __FUNCTION__,
+ __LINE__);
+ return -1;
+ }
+
+ // Loop through all device types and all devices for each type
+ // and store in a list.
+ int numDevices = (*deviceList)->count;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d Found %d devices", __FUNCTION__, __LINE__, numDevices);
+
+ for (int i = 0; i < numDevices; i++)
+ {
+
+ SGDeviceName sgDeviceName = (*deviceList)->entry[i];
+ // Get the list with input devices for this type of device
+ if (sgDeviceName.inputs)
+ {
+ SGDeviceInputList inputList = sgDeviceName.inputs;
+ int numInputDev = (*inputList)->count;
+
+ for (int inputDevIndex = 0;
+ inputDevIndex < numInputDev;
+ inputDevIndex++)
+ {
+ // Get the name for this capture device
+ SGDeviceInputName deviceInputName =
+ (*inputList)->entry[inputDevIndex];
+
+ VideoCaptureMacName* deviceName = new VideoCaptureMacName();
+
+ deviceName->_size = PascalStringToCString(
+ deviceInputName.name, deviceName->_name,
+ sizeof(deviceName->_name));
+
+ if (deviceName->_size > 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug,webrtc::kTraceVideoCapture,
+ _id,
+ "%s:%d Capture device %d: %s was successfully "
+ "set", __FUNCTION__, __LINE__, numberOfDevices,
+ deviceName->_name);
+
+ if (numberOfDevices == deviceNumber)
+ {
+ strcpy((char*) deviceNameUTF8, deviceName->_name);
+ strcpy((char*) deviceUniqueIdUTF8, deviceName->_name);
+ }
+ numberOfDevices++;
+ }
+ else
+ {
+ delete deviceName;
+
+ if (deviceName->_size < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "%s:%d Error in PascalStringToCString",
+ __FUNCTION__, __LINE__);
+ return -1;
+ }
+ }
+ }
+ }
+ }
+
+ // clean up
+ SGDisposeDeviceList(captureGrabber, deviceList);
+ if (closeChannel)
+ {
+ SGDisposeChannel(captureGrabber, captureChannel);
+ }
+ if (captureGrabber)
+ {
+ SGRelease(captureGrabber);
+ CloseComponent(captureGrabber);
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ "%s:%d End function successfully", __FUNCTION__, __LINE__);
+ return 0;
+}
+
+/**
+ Convert a Pascal string to a C string.
+
+ \param[in] pascalString
+ Pascal string to convert. Pascal strings contain the number of
+ characters in the first byte and are not null-terminated.
+
+ \param[out] cString
+ The C string buffer into which to copy the converted string.
+
+ \param[in] bufferSize
+ The size of the C string buffer in bytes.
+
+ \return The number of characters in the string on success and -1 on failure.
+ */
+CFIndex VideoCaptureMacQuickTimeInfo::PascalStringToCString(
+ const unsigned char* pascalString, char* cString, CFIndex bufferSize)
+{
+ if (pascalString == NULL)
+ {
+ return -1;
+ }
+
+ if (cString == NULL)
+ {
+ return -1;
+ }
+
+ if (bufferSize == 0)
+ {
+ return -1;
+ }
+
+ CFIndex cStringLength = 0;
+ CFIndex maxStringLength = bufferSize - 1;
+
+ CFStringRef cfString = CFStringCreateWithPascalString(
+ NULL, pascalString, kCFStringEncodingMacRoman);
+ if (cfString == NULL)
+ {
+ CFRelease(cfString);
+ return -1;
+ }
+
+ CFIndex cfLength = CFStringGetLength(cfString);
+ cStringLength = cfLength;
+ if (cfLength > maxStringLength)
+ {
+ cStringLength = maxStringLength;
+ }
+
+ Boolean success = CFStringGetCString(cfString, cString, bufferSize,
+ kCFStringEncodingMacRoman);
+
+ // Ensure the problem isn't insufficient buffer length.
+ // This is fine; we will return a partial string.
+ if (success == false && cfLength <= maxStringLength)
+ {
+ CFRelease(cfString);
+ return -1;
+ }
+
+ CFRelease(cfString);
+ return cStringLength;
+}
+
+//
+//
+// Functions for handling capture devices
+//
+//
+
+VideoCaptureMacQuickTimeInfo::VideoCaptureMacName::VideoCaptureMacName() :
+ _size(0)
+{
+ memset(_name, 0, kVideoCaptureMacNameMaxSize);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h
new file mode 100644
index 0000000..5114d11
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h
@@ -0,0 +1,166 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_quick_time_info.h
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
+
+#include <QuickTime/QuickTime.h>
+
+#include "../../video_capture_impl.h"
+#include "../../device_info_impl.h"
+#include "list_wrapper.h"
+#include "map_wrapper.h"
+
+class VideoRenderCallback;
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class Trace;
+
+class VideoCaptureMacQuickTimeInfo: public DeviceInfoImpl
+{
+public:
+
+ static DeviceInfo* Create(const WebRtc_Word32 id);
+ static void Destroy(DeviceInfo* deviceInfo);
+
+ VideoCaptureMacQuickTimeInfo(const WebRtc_Word32 id);
+ virtual ~VideoCaptureMacQuickTimeInfo();
+
+ WebRtc_Word32 Init();
+
+ virtual WebRtc_UWord32 NumberOfDevices();
+
+ /*
+ * Returns the available capture devices.
+ * deviceNumber -[in] index of capture device
+ * deviceNameUTF8 - friendly name of the capture device
+ * deviceUniqueIdUTF8 - unique name of the capture device if it exist.
+ * Otherwise same as deviceNameUTF8
+ * productUniqueIdUTF8 - unique product id if it exist. Null terminated
+ * otherwise.
+ */
+ virtual WebRtc_Word32 GetDeviceName(
+ WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8 = 0,
+ WebRtc_UWord32 productUniqueIdUTF8Length = 0);
+
+
+ // ************** The remaining public functions are not supported on Mac
+
+ /*
+ * Returns the number of capabilities for this device
+ */
+ virtual WebRtc_Word32 NumberOfCapabilities(const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ /*
+ * Gets the capabilities of the named device
+ */
+ virtual WebRtc_Word32 GetCapability(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability);
+
+ /*
+ * Gets the capability that best matches the requested width, height and frame rate.
+ * Returns the deviceCapabilityNumber on success.
+ */
+ virtual WebRtc_Word32 GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested,
+ VideoCaptureCapability& resulting);
+
+ /*
+ * Display OS /capture device specific settings dialog
+ */
+ virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
+ WebRtc_UWord32 positionX, WebRtc_UWord32 positionY);
+
+protected:
+ virtual WebRtc_Word32 CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+private:
+
+ struct VideoCaptureMacName
+ {
+ VideoCaptureMacName();
+
+ enum
+ {
+ kVideoCaptureMacNameMaxSize = 64
+ };
+ char _name[kVideoCaptureMacNameMaxSize];
+ CFIndex _size;
+ };
+
+ enum
+ {
+ kVideoCaptureMacDeviceListTimeout = 5000
+ }; // Timeout value [ms] if we want to create a new device list or not
+ enum
+ {
+ kYuy2_1280_1024_length = 2621440
+ }; // Temporary constant allowing this size from built-in iSight webcams.
+
+private:
+ // private methods
+
+ int GetCaptureDevices(WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameUTF8Length,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length,
+ int& numberOfDevices);
+
+ static CFIndex PascalStringToCString(const unsigned char* pascalString,
+ char* cString, CFIndex bufferSize);
+
+private:
+ // member vars
+ WebRtc_Word32 _id;
+ bool _terminated;
+ CriticalSectionWrapper* _grabberCritsect;
+ webrtc::Trace* _trace;
+ webrtc::ThreadWrapper* _grabberUpdateThread;
+ webrtc::EventWrapper* _grabberUpdateEvent;
+ SeqGrabComponent _captureGrabber;
+ Component _captureDevice;
+ char _captureDeviceDisplayName[64];
+ bool _captureIsInitialized;
+ GWorldPtr _gWorld;
+ SGChannel _captureChannel;
+ ImageSequence _captureSequence;
+ bool _sgPrepared;
+ bool _sgStarted;
+ int _codecWidth;
+ int _codecHeight;
+ int _trueCaptureWidth;
+ int _trueCaptureHeight;
+ ListWrapper _captureDeviceList;
+ WebRtc_Word64 _captureDeviceListTime;
+ ListWrapper _captureCapabilityList;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h
diff --git a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm
new file mode 100644
index 0000000..d329936
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm
@@ -0,0 +1,275 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_capture_mac.cc
+ *
+ */
+
+
+// super class stuff
+#include "../video_capture_impl.h"
+#include "../device_info_impl.h"
+#include "../video_capture_config.h"
+#include "ref_count.h"
+
+#include "trace.h"
+
+#include <QuickTime/QuickTime.h>
+
+// 10.4 support must be decided runtime. We will just decide which framework to
+// use at compile time "work" classes. One for QTKit, one for QuickTime
+#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
+#include "QuickTime/video_capture_quick_time.h"
+#include "QuickTime/video_capture_quick_time_info.h"
+#else
+#include "QTKit/video_capture_qtkit.h"
+#include "QTKit/video_capture_qtkit_info.h"
+#endif
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+// static
+bool CheckOSVersion()
+{
+ // Check OSX version
+ OSErr err = noErr;
+
+ SInt32 version;
+
+ err = Gestalt(gestaltSystemVersion, &version);
+ if (err != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
+ "Could not get OS version");
+ return false;
+ }
+
+ if (version < 0x00001040) // Older version than Mac OSX 10.4
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
+ "OS version too old: 0x%x", version);
+ return false;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
+ "OS version compatible: 0x%x", version);
+
+ return true;
+}
+
+// static
+bool CheckQTVersion()
+{
+ // Check OSX version
+ OSErr err = noErr;
+
+ SInt32 version;
+
+ err = Gestalt(gestaltQuickTime, &version);
+ if (err != noErr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
+ "Could not get QuickTime version");
+ return false;
+ }
+
+ if (version < 0x07000000) // QT v. 7.x or newer (QT 5.0.2 0x05020000)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
+ "QuickTime version too old: 0x%x", version);
+ return false;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
+ "QuickTime version compatible: 0x%x", version);
+ return true;
+}
+
+/**************************************************************************
+ *
+ * Create/Destroy a VideoCaptureModule
+ *
+ ***************************************************************************/
+
+/*
+ * Returns version of the module and its components
+ *
+ * version - buffer to which the version will be written
+ * remainingBufferInBytes - remaining number of WebRtc_Word8 in the version
+ * buffer
+ * position - position of the next empty WebRtc_Word8 in the
+ * version buffer
+ */
+
+VideoCaptureModule* VideoCaptureImpl::Create(
+ const WebRtc_Word32 id, const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ if (webrtc::videocapturemodule::CheckOSVersion() == false)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "OS version is too old. Could not create video capture "
+ "module. Returning NULL");
+ return NULL;
+ }
+
+#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
+ if (webrtc::videocapturemodule::CheckQTVersion() == false)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "QuickTime version is too old. Could not create video "
+ "capture module. Returning NULL");
+ return NULL;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "%s line %d. QTKit is not supported on this machine. Using "
+ "QuickTime framework to capture video",
+ __FILE__, __LINE__);
+
+ RefCountImpl<videocapturemodule::VideoCaptureMacQuickTime>*
+ newCaptureModule =
+ new RefCountImpl<videocapturemodule::VideoCaptureMacQuickTime>(id);
+
+ if (!newCaptureModule)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
+ "could not Create for unique device %s, !newCaptureModule",
+ deviceUniqueIdUTF8);
+ return NULL;
+ }
+
+ if (newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
+ "could not Create for unique device %s, "
+ "newCaptureModule->Init()!=0",
+ deviceUniqueIdUTF8);
+ delete newCaptureModule;
+ return NULL;
+ }
+
+ // Successfully created VideoCaptureMacQuicktime. Return it
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "Module created for unique device %s. Will use QuickTime "
+ "framework to capture",
+ deviceUniqueIdUTF8);
+ return newCaptureModule;
+
+#else // QTKit version
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "Using QTKit framework to capture video", id);
+
+ RefCountImpl<videocapturemodule::VideoCaptureMacQTKit>* newCaptureModule =
+ new RefCountImpl<videocapturemodule::VideoCaptureMacQTKit>(id);
+
+ if(!newCaptureModule)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
+ "could not Create for unique device %s, !newCaptureModule",
+ deviceUniqueIdUTF8);
+ return NULL;
+ }
+ if(newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
+ "could not Create for unique device %s, "
+ "newCaptureModule->Init()!=0", deviceUniqueIdUTF8);
+ delete newCaptureModule;
+ return NULL;
+ }
+
+ // Successfully created VideoCaptureMacQuicktime. Return it
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "Module created for unique device %s, will use QTKit "
+ "framework",deviceUniqueIdUTF8);
+ return newCaptureModule;
+#endif
+}
+
+/**************************************************************************
+ *
+ * Create/Destroy a DeviceInfo
+ *
+ ***************************************************************************/
+
+VideoCaptureModule::DeviceInfo*
+VideoCaptureImpl::CreateDeviceInfo(const WebRtc_Word32 id)
+{
+
+
+ if (webrtc::videocapturemodule::CheckOSVersion() == false)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "OS version is too old. Could not create video capture "
+ "module. Returning NULL");
+ return NULL;
+ }
+
+#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
+ if (webrtc::videocapturemodule::CheckQTVersion() == false)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
+ "QuickTime version is too old. Could not create video "
+ "capture module. Returning NULL");
+ return NULL;
+ }
+
+ webrtc::videocapturemodule::VideoCaptureMacQuickTimeInfo* newCaptureInfoModule =
+ new webrtc::videocapturemodule::VideoCaptureMacQuickTimeInfo(id);
+
+ if (!newCaptureInfoModule || newCaptureInfoModule->Init() != 0)
+ {
+ Destroy(newCaptureInfoModule);
+ newCaptureInfoModule = NULL;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "Failed to Init newCaptureInfoModule created with id %d "
+ "and device \"\" ", id);
+ return NULL;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "VideoCaptureModule created for id", id);
+ return newCaptureInfoModule;
+
+#else // QTKit version
+ webrtc::videocapturemodule::VideoCaptureMacQTKitInfo* newCaptureInfoModule =
+ new webrtc::videocapturemodule::VideoCaptureMacQTKitInfo(id);
+
+ if(!newCaptureInfoModule || newCaptureInfoModule->Init() != 0)
+ {
+ //Destroy(newCaptureInfoModule);
+ delete newCaptureInfoModule;
+ newCaptureInfoModule = NULL;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "Failed to Init newCaptureInfoModule created with id %d "
+ "and device \"\" ", id);
+ return NULL;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
+ "VideoCaptureModule created for id", id);
+ return newCaptureInfoModule;
+
+#endif
+
+}
+
+/**************************************************************************
+ *
+ * End Create/Destroy VideoCaptureModule
+ *
+ ***************************************************************************/
+} // namespace videocapturemodule
+} // namespace webrtc
+
diff --git a/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h b/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h
new file mode 100644
index 0000000..a4d9da6
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
+
+#include "../video_capture_delay.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+const WebRtc_Word32 NoWindowsCaptureDelays=1;
+const DelayValues WindowsCaptureDelays[NoWindowsCaptureDelays]=
+{
+ "Microsoft LifeCam Cinema","usb#vid_045e&pid_075d",{{640,480,125},{640,360,117},{424,240,111},{352,288,111},{320,240,116},{176,144,101},{160,120,109},{1280,720,166},{960,544,126},{800,448,120},{800,600,127}},
+};
+
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc
new file mode 100644
index 0000000..5adcc96
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc
@@ -0,0 +1,792 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "device_info_windows.h"
+
+#include "../video_capture_config.h"
+#include "help_functions_windows.h"
+#include "capture_delay_values_windows.h"
+#include "ref_count.h"
+#include "trace.h"
+
+#include <Streams.h>
+#include <Dvdmedia.h>
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo(
+ const WebRtc_Word32 id)
+{
+ videocapturemodule::DeviceInfoWindows* dsInfo =
+ new videocapturemodule::DeviceInfoWindows(id);
+
+ if (!dsInfo || dsInfo->Init() != 0)
+ {
+ delete dsInfo;
+ dsInfo = NULL;
+ }
+ return dsInfo;
+}
+
+DeviceInfoWindows::DeviceInfoWindows(const WebRtc_Word32 id)
+ : DeviceInfoImpl(id), _dsDevEnum(NULL), _dsMonikerDevEnum(NULL),
+ _CoUninitializeIsRequired(true)
+{
+ // 1) Initialize the COM library (make Windows load the DLLs).
+ //
+ // CoInitializeEx must be called at least once, and is usually called only once,
+ // for each thread that uses the COM library. Multiple calls to CoInitializeEx
+ // by the same thread are allowed as long as they pass the same concurrency flag,
+ // but subsequent valid calls return S_FALSE.
+ // To close the COM library gracefully on a thread, each successful call to
+ // CoInitializeEx, including any call that returns S_FALSE, must be balanced
+ // by a corresponding call to CoUninitialize.
+ //
+
+ /*Apartment-threading, while allowing for multiple threads of execution,
+ serializes all incoming calls by requiring that calls to methods of objects created by this thread always run on the same thread
+ the apartment/thread that created them. In addition, calls can arrive only at message-queue boundaries (i.e., only during a
+ PeekMessage, SendMessage, DispatchMessage, etc.). Because of this serialization, it is not typically necessary to write concurrency control into
+ the code for the object, other than to avoid calls to PeekMessage and SendMessage during processing that must not be interrupted by other method
+ invocations or calls to other objects in the same apartment/thread.*/
+
+ ///CoInitializeEx(NULL, COINIT_APARTMENTTHREADED ); //| COINIT_SPEED_OVER_MEMORY
+ HRESULT hr = CoInitializeEx(NULL, COINIT_MULTITHREADED); // Use COINIT_MULTITHREADED since Voice Engine uses COINIT_MULTITHREADED
+ if (FAILED(hr))
+ {
+ // Avoid calling CoUninitialize() since CoInitializeEx() failed.
+ _CoUninitializeIsRequired = FALSE;
+
+ if (hr == RPC_E_CHANGED_MODE)
+ {
+ // Calling thread has already initialized COM to be used in a single-threaded
+ // apartment (STA). We are then prevented from using STA.
+ // Details: hr = 0x80010106 <=> "Cannot change thread mode after it is set".
+ //
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "VideoCaptureWindowsDSInfo::VideoCaptureWindowsDSInfo "
+ "CoInitializeEx(NULL, COINIT_APARTMENTTHREADED) => "
+ "RPC_E_CHANGED_MODE, error 0x%x",
+ hr);
+ }
+ }
+}
+
+DeviceInfoWindows::~DeviceInfoWindows()
+{
+ RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+ RELEASE_AND_CLEAR(_dsDevEnum);
+ if (_CoUninitializeIsRequired)
+ {
+ CoUninitialize();
+ }
+}
+
+WebRtc_Word32 DeviceInfoWindows::Init()
+{
+ HRESULT hr = CoCreateInstance(CLSID_SystemDeviceEnum, NULL, CLSCTX_INPROC,
+ IID_ICreateDevEnum, (void **) &_dsDevEnum);
+ if (hr != NOERROR)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create CLSID_SystemDeviceEnum, error 0x%x", hr);
+ return -1;
+ }
+ return 0;
+}
+WebRtc_UWord32 DeviceInfoWindows::NumberOfDevices()
+{
+ ReadLockScoped cs(_apiLock);
+ return GetDeviceInfo(0, 0, 0, 0, 0, 0, 0);
+
+}
+WebRtc_Word32 DeviceInfoWindows::GetDeviceName(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+{
+ ReadLockScoped cs(_apiLock);
+ const WebRtc_Word32 result = GetDeviceInfo(deviceNumber, deviceNameUTF8,
+ deviceNameLength,
+ deviceUniqueIdUTF8,
+ deviceUniqueIdUTF8Length,
+ productUniqueIdUTF8,
+ productUniqueIdUTF8Length);
+ return result > (WebRtc_Word32) deviceNumber ? 0 : -1;
+}
+
+WebRtc_Word32 DeviceInfoWindows::GetDeviceInfo(
+ WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+
+{
+
+ // enumerate all video capture devices
+ RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+ HRESULT hr =
+ _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
+ &_dsMonikerDevEnum, 0);
+ if (hr != NOERROR)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to enumerate CLSID_SystemDeviceEnum, error 0x%x."
+ " No webcam exist?", hr);
+ return 0;
+ }
+
+ _dsMonikerDevEnum->Reset();
+ ULONG cFetched;
+ IMoniker *pM;
+ int index = 0;
+ while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched))
+ {
+ IPropertyBag *pBag;
+ hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void **) &pBag);
+ if (S_OK == hr)
+ {
+ // Find the description or friendly name.
+ VARIANT varName;
+ VariantInit(&varName);
+ hr = pBag->Read(L"Description", &varName, 0);
+ if (FAILED(hr))
+ {
+ hr = pBag->Read(L"FriendlyName", &varName, 0);
+ }
+ if (SUCCEEDED(hr))
+ {
+ // ignore all VFW drivers
+ if ((wcsstr(varName.bstrVal, (L"(VFW)")) == NULL) &&
+ (_wcsnicmp(varName.bstrVal, (L"Google Camera Adapter"),21)
+ != 0))
+ {
+ // Found a valid device
+ if (index == deviceNumber) // This is the device we are interested in.
+ {
+ int convResult = 0;
+ if (deviceNameLength > 0)
+ {
+ convResult = WideCharToMultiByte(CP_UTF8, 0,
+ varName.bstrVal, -1,
+ (char*) deviceNameUTF8,
+ deviceNameLength, NULL,
+ NULL);
+ if (convResult == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "Failed to convert device name to UTF8. %d",
+ GetLastError());
+ return -1;
+ }
+ }
+ if (deviceUniqueIdUTF8Length > 0)
+ {
+ hr = pBag->Read(L"DevicePath", &varName, 0);
+ if (FAILED(hr))
+ {
+ strncpy_s((char *) deviceUniqueIdUTF8,
+ deviceUniqueIdUTF8Length,
+ (char *) deviceNameUTF8, convResult);
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "Failed to get deviceUniqueIdUTF8 using deviceNameUTF8");
+ }
+ else
+ {
+ convResult = WideCharToMultiByte(
+ CP_UTF8,
+ 0,
+ varName.bstrVal,
+ -1,
+ (char*) deviceUniqueIdUTF8,
+ deviceUniqueIdUTF8Length,
+ NULL, NULL);
+ if (convResult == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCapture, _id,
+ "Failed to convert device name to UTF8. %d",
+ GetLastError());
+ return -1;
+ }
+ if (productUniqueIdUTF8
+ && productUniqueIdUTF8Length > 0)
+ {
+ GetProductId(deviceUniqueIdUTF8,
+ productUniqueIdUTF8,
+ productUniqueIdUTF8Length);
+ }
+ }
+ }
+
+ }
+ ++index; // increase the number of valid devices
+ }
+ }
+ VariantClear(&varName);
+ pBag->Release();
+ pM->Release();
+ }
+
+ }
+ if (deviceNameLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id, "%s %s",
+ __FUNCTION__, deviceNameUTF8);
+ }
+ return index;
+}
+
+IBaseFilter * DeviceInfoWindows::GetDeviceFilter(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+{
+
+ const WebRtc_Word32 deviceUniqueIdUTF8Length =
+ (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8); // UTF8 is also NULL terminated
+ if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Device name too long");
+ return NULL;
+ }
+
+ // enumerate all video capture devices
+ RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+ HRESULT hr = _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
+ &_dsMonikerDevEnum, 0);
+ if (hr != NOERROR)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to enumerate CLSID_SystemDeviceEnum, error 0x%x."
+ " No webcam exist?", hr);
+ return 0;
+ }
+ _dsMonikerDevEnum->Reset();
+ ULONG cFetched;
+ IMoniker *pM;
+
+ IBaseFilter *captureFilter = NULL;
+ bool deviceFound = false;
+ while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched) && !deviceFound)
+ {
+ IPropertyBag *pBag;
+ hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void **) &pBag);
+ if (S_OK == hr)
+ {
+ // Find the description or friendly name.
+ VARIANT varName;
+ VariantInit(&varName);
+ if (deviceUniqueIdUTF8Length > 0)
+ {
+ hr = pBag->Read(L"DevicePath", &varName, 0);
+ if (FAILED(hr))
+ {
+ hr = pBag->Read(L"Description", &varName, 0);
+ if (FAILED(hr))
+ {
+ hr = pBag->Read(L"FriendlyName", &varName, 0);
+ }
+ }
+ if (SUCCEEDED(hr))
+ {
+ char tempDevicePathUTF8[256];
+ tempDevicePathUTF8[0] = 0;
+ const int compresult =
+ WideCharToMultiByte(CP_UTF8, 0, varName.bstrVal, -1,
+ tempDevicePathUTF8,
+ sizeof(tempDevicePathUTF8), NULL,
+ NULL);
+ if (strncmp(tempDevicePathUTF8,
+ (const char*) deviceUniqueIdUTF8,
+ deviceUniqueIdUTF8Length) == 0)
+ {
+ // We have found the requested device
+ deviceFound = true;
+ hr = pM->BindToObject(0, 0, IID_IBaseFilter,
+ (void**) &captureFilter);
+ if FAILED(hr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id, "Failed to bind to the selected capture device %d",hr);
+ }
+
+ if (productUniqueIdUTF8
+ && productUniqueIdUTF8Length > 0) // Get the device name
+ {
+
+ GetProductId(deviceUniqueIdUTF8,
+ productUniqueIdUTF8,
+ productUniqueIdUTF8Length);
+ }
+
+ }
+ }
+ }
+ VariantClear(&varName);
+ pBag->Release();
+ pM->Release();
+ }
+ }
+ return captureFilter;
+}
+
+WebRtc_Word32 DeviceInfoWindows::GetWindowsCapability(
+ const WebRtc_Word32 capabilityIndex,
+ VideoCaptureCapabilityWindows& windowsCapability)
+
+{
+ ReadLockScoped cs(_apiLock);
+ // Make sure the number is valid
+ if (capabilityIndex >= _captureCapabilities.Size() || capabilityIndex < 0)
+ return -1;
+
+ MapItem* item = _captureCapabilities.Find(capabilityIndex);
+ if (!item)
+ return -1;
+
+ VideoCaptureCapabilityWindows* capPointer =
+ static_cast<VideoCaptureCapabilityWindows*> (item->GetItem());
+ windowsCapability = *capPointer;
+ return 0;
+}
+
+WebRtc_Word32 DeviceInfoWindows::CreateCapabilityMap(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+
+{
+ // Reset old capability list
+ MapItem* item = NULL;
+ while (item = _captureCapabilities.Last())
+ {
+ delete item->GetItem();
+ _captureCapabilities.Erase(item);
+ }
+
+ const WebRtc_Word32 deviceUniqueIdUTF8Length =
+ (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
+ if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Device name too long");
+ return -1;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "CreateCapabilityMap called for device %s", deviceUniqueIdUTF8);
+
+
+ WebRtc_UWord8 productId[kVideoCaptureProductIdLength];
+ IBaseFilter* captureDevice = DeviceInfoWindows::GetDeviceFilter(
+ deviceUniqueIdUTF8,
+ productId,
+ kVideoCaptureProductIdLength);
+ if (!captureDevice)
+ return -1;
+ IPin* outputCapturePin = GetOutputPin(captureDevice);
+ if (!outputCapturePin)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to get capture device output pin");
+ RELEASE_AND_CLEAR(captureDevice);
+ return -1;
+ }
+ IAMExtDevice* extDevice = NULL;
+ HRESULT hr = captureDevice->QueryInterface(IID_IAMExtDevice,
+ (void **) &extDevice);
+ if (SUCCEEDED(hr) && extDevice)
+ {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "This is an external device");
+ extDevice->Release();
+ }
+
+ IAMStreamConfig* streamConfig = NULL;
+ hr = outputCapturePin->QueryInterface(IID_IAMStreamConfig,
+ (void**) &streamConfig);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to get IID_IAMStreamConfig interface from capture device");
+ return -1;
+ }
+
+ // this gets the FPS
+ IAMVideoControl* videoControlConfig = NULL;
+ HRESULT hrVC = captureDevice->QueryInterface(IID_IAMVideoControl,
+ (void**) &videoControlConfig);
+ if (FAILED(hrVC))
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "IID_IAMVideoControl Interface NOT SUPPORTED");
+ }
+
+ AM_MEDIA_TYPE *pmt = NULL;
+ VIDEO_STREAM_CONFIG_CAPS caps;
+ int count, size;
+
+ hr = streamConfig->GetNumberOfCapabilities(&count, &size);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to GetNumberOfCapabilities");
+ RELEASE_AND_CLEAR(videoControlConfig);
+ RELEASE_AND_CLEAR(streamConfig);
+ RELEASE_AND_CLEAR(outputCapturePin);
+ RELEASE_AND_CLEAR(captureDevice);
+ return -1;
+ }
+
+ WebRtc_Word32 index = 0; // Index in created _capabilities map
+ // Check if the device support formattype == FORMAT_VideoInfo2 and FORMAT_VideoInfo.
+ // Prefer FORMAT_VideoInfo since some cameras (ZureCam) has been seen having problem with MJPEG and FORMAT_VideoInfo2
+ // Interlace flag is only supported in FORMAT_VideoInfo2
+ bool supportFORMAT_VideoInfo2 = false;
+ bool supportFORMAT_VideoInfo = false;
+ bool foundInterlacedFormat = false;
+ GUID preferedVideoFormat = FORMAT_VideoInfo;
+ for (WebRtc_Word32 tmp = 0; tmp < count; ++tmp)
+ {
+ hr = streamConfig->GetStreamCaps(tmp, &pmt,
+ reinterpret_cast<BYTE*> (&caps));
+ if (!FAILED(hr))
+ {
+ if (pmt->majortype == MEDIATYPE_Video
+ && pmt->formattype == FORMAT_VideoInfo2)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ " Device support FORMAT_VideoInfo2");
+ supportFORMAT_VideoInfo2 = true;
+ VIDEOINFOHEADER2* h =
+ reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
+ assert(h);
+ foundInterlacedFormat |= h->dwInterlaceFlags
+ & (AMINTERLACE_IsInterlaced
+ | AMINTERLACE_DisplayModeBobOnly);
+ }
+ if (pmt->majortype == MEDIATYPE_Video
+ && pmt->formattype == FORMAT_VideoInfo)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
+ " Device support FORMAT_VideoInfo2");
+ supportFORMAT_VideoInfo = true;
+ }
+ }
+ }
+ if (supportFORMAT_VideoInfo2)
+ {
+ if (supportFORMAT_VideoInfo && !foundInterlacedFormat)
+ {
+ preferedVideoFormat = FORMAT_VideoInfo;
+ }
+ else
+ {
+ preferedVideoFormat = FORMAT_VideoInfo2;
+ }
+ }
+
+ for (WebRtc_Word32 tmp = 0; tmp < count; ++tmp)
+ {
+ hr = streamConfig->GetStreamCaps(tmp, &pmt,
+ reinterpret_cast<BYTE*> (&caps));
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to GetStreamCaps");
+ RELEASE_AND_CLEAR(videoControlConfig);
+ RELEASE_AND_CLEAR(streamConfig);
+ RELEASE_AND_CLEAR(outputCapturePin);
+ RELEASE_AND_CLEAR(captureDevice);
+ return -1;
+ }
+
+ if (pmt->majortype == MEDIATYPE_Video
+ && pmt->formattype == preferedVideoFormat)
+ {
+
+ VideoCaptureCapabilityWindows* capability =
+ new VideoCaptureCapabilityWindows();
+ WebRtc_Word64 avgTimePerFrame = 0;
+ bool interlaced = false;
+
+ if (pmt->formattype == FORMAT_VideoInfo)
+ {
+ VIDEOINFOHEADER* h =
+ reinterpret_cast<VIDEOINFOHEADER*> (pmt->pbFormat);
+ assert(h);
+ capability->directShowCapabilityIndex = tmp;
+ capability->width = h->bmiHeader.biWidth;
+ capability->height = h->bmiHeader.biHeight;
+ avgTimePerFrame = h->AvgTimePerFrame;
+ }
+ if (pmt->formattype == FORMAT_VideoInfo2)
+ {
+ VIDEOINFOHEADER2* h =
+ reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
+ assert(h);
+ capability->directShowCapabilityIndex = tmp;
+ capability->width = h->bmiHeader.biWidth;
+ capability->height = h->bmiHeader.biHeight;
+ capability->interlaced = h->dwInterlaceFlags
+ & (AMINTERLACE_IsInterlaced
+ | AMINTERLACE_DisplayModeBobOnly);
+ avgTimePerFrame = h->AvgTimePerFrame;
+ }
+
+ if (hrVC == S_OK)
+ {
+ LONGLONG *maxFps; // array
+ long listSize;
+ SIZE size;
+ size.cx = capability->width;
+ size.cy = capability->height;
+
+ // GetMaxAvailableFrameRate doesn't return max frame rate always
+ // eg: Logitech Notebook. This may be due to a bug in that API
+ // because GetFrameRateList array is reversed in the above camera. So
+ // a util method written. Can't assume the first value will return
+ // the max fps.
+ hrVC = videoControlConfig->GetFrameRateList(outputCapturePin,
+ tmp, size,
+ &listSize,
+ &maxFps);
+
+ if (hrVC == S_OK && listSize > 0)
+ {
+ LONGLONG maxFPS = GetMaxOfFrameArray(maxFps, listSize);
+ capability->maxFPS = static_cast<int> (10000000
+ / maxFPS);
+ capability->supportFrameRateControl = true;
+ }
+ else // use existing method
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+ _id,
+ "GetMaxAvailableFrameRate NOT SUPPORTED");
+ if (avgTimePerFrame > 0)
+ capability->maxFPS = static_cast<int> (10000000
+ / avgTimePerFrame);
+ else
+ capability->maxFPS = 0;
+ }
+ }
+ else // use existing method in case IAMVideoControl is not supported
+ {
+ if (avgTimePerFrame > 0)
+ capability->maxFPS = static_cast<int> (10000000
+ / avgTimePerFrame);
+ else
+ capability->maxFPS = 0;
+ }
+
+ // can't switch MEDIATYPE :~(
+ if (pmt->subtype == MEDIASUBTYPE_I420)
+ {
+ capability->rawType = kVideoI420;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_IYUV)
+ {
+ capability->rawType = kVideoIYUV;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_RGB24)
+ {
+ capability->rawType = kVideoRGB24;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_YUY2)
+ {
+ capability->rawType = kVideoYUY2;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_RGB565)
+ {
+ capability->rawType = kVideoRGB565;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_MJPG)
+ {
+ capability->rawType = kVideoMJPEG;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_dvsl
+ || pmt->subtype == MEDIASUBTYPE_dvsd
+ || pmt->subtype == MEDIASUBTYPE_dvhd) // If this is an external DV camera
+ {
+ capability->rawType = kVideoYUY2;// MS DV filter seems to create this type
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_UYVY) // Seen used by Declink capture cards
+ {
+ capability->rawType = kVideoUYVY;
+ }
+ else if (pmt->subtype == MEDIASUBTYPE_HDYC) // Seen used by Declink capture cards. Uses BT. 709 color. Not entiry correct to use UYVY. http://en.wikipedia.org/wiki/YCbCr
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "Device support HDYC.");
+ capability->rawType = kVideoUYVY;
+ }
+ else
+ {
+ WCHAR strGuid[39];
+ StringFromGUID2(pmt->subtype, strGuid, 39);
+ WEBRTC_TRACE( webrtc::kTraceWarning,
+ webrtc::kTraceVideoCapture, _id,
+ "Device support unknown media type %ls, width %d, height %d",
+ strGuid);
+ delete capability;
+ continue;
+ }
+
+ // Get the expected capture delay from the static list
+ capability->expectedCaptureDelay
+ = GetExpectedCaptureDelay(WindowsCaptureDelays,
+ NoWindowsCaptureDelays,
+ productId,
+ capability->width,
+ capability->height);
+ _captureCapabilities.Insert(index++, capability);
+ WEBRTC_TRACE( webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "Camera capability, width:%d height:%d type:%d fps:%d",
+ capability->width, capability->height,
+ capability->rawType, capability->maxFPS);
+ }
+ DeleteMediaType(pmt);
+ pmt = NULL;
+ }
+ RELEASE_AND_CLEAR(streamConfig);
+ RELEASE_AND_CLEAR(videoControlConfig);
+ RELEASE_AND_CLEAR(outputCapturePin);
+ RELEASE_AND_CLEAR(captureDevice); // Release the capture device
+
+ // Store the new used device name
+ _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
+ _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
+ _lastUsedDeviceNameLength
+ + 1);
+ memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8, _lastUsedDeviceNameLength+ 1);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "CreateCapabilityMap %d", _captureCapabilities.Size());
+
+ return _captureCapabilities.Size();
+}
+
+/* Constructs a product ID from the Windows DevicePath. on a USB device the devicePath contains product id and vendor id.
+ This seems to work for firewire as well
+ /* Example of device path
+ "\\?\usb#vid_0408&pid_2010&mi_00#7&258e7aaf&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
+ "\\?\avc#sony&dv-vcr&camcorder&dv#65b2d50301460008#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
+ */
+void DeviceInfoWindows::GetProductId(const WebRtc_UWord8* devicePath,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length)
+{
+ *productUniqueIdUTF8 = '\0';
+ char* startPos = strstr((char*) devicePath, "\\\\?\\");
+ if (!startPos)
+ {
+ strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "Failed to get the product Id");
+ return;
+ }
+ startPos += 4;
+
+ char* pos = strchr(startPos, '&');
+ if (!pos || pos >= (char*) devicePath + strlen((char*) devicePath))
+ {
+ strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "Failed to get the product Id");
+ return;
+ }
+ // Find the second occurence
+ pos = strchr(pos + 1, '&');
+ WebRtc_UWord32 bytesToCopy = (WebRtc_UWord32)(pos - startPos);
+ if (pos && (bytesToCopy <= productUniqueIdUTF8Length) && bytesToCopy
+ <= kVideoCaptureProductIdLength)
+ {
+ strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length,
+ (char*) startPos, bytesToCopy);
+ }
+ else
+ {
+ strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+ "Failed to get the product Id");
+ }
+}
+
+WebRtc_Word32 DeviceInfoWindows::DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8,
+ void* parentWindow,
+ WebRtc_UWord32 positionX,
+ WebRtc_UWord32 positionY)
+{
+ ReadLockScoped cs(_apiLock);
+ HWND window = (HWND) parentWindow;
+
+ IBaseFilter* filter = GetDeviceFilter(deviceUniqueIdUTF8, NULL, 0);
+ if (!filter)
+ return -1;
+
+ ISpecifyPropertyPages* pPages = NULL;
+ CAUUID uuid;
+ HRESULT hr = S_OK;
+
+ hr = filter->QueryInterface(IID_ISpecifyPropertyPages, (LPVOID*) &pPages);
+ if (!SUCCEEDED(hr))
+ {
+ filter->Release();
+ return -1;
+ }
+ hr = pPages->GetPages(&uuid);
+ if (!SUCCEEDED(hr))
+ {
+ filter->Release();
+ return -1;
+ }
+
+ WCHAR tempDialogTitleWide[256];
+ tempDialogTitleWide[0] = 0;
+ int size = 255;
+
+ // UTF-8 to wide char
+ MultiByteToWideChar(CP_UTF8, 0, (char*) dialogTitleUTF8, -1,
+ tempDialogTitleWide, size);
+
+ // Invoke a dialog box to display.
+
+ hr = OleCreatePropertyFrame(window, // You must create the parent window.
+ positionX, // Horizontal position for the dialog box.
+ positionY, // Vertical position for the dialog box.
+ tempDialogTitleWide,// String used for the dialog box caption.
+ 1, // Number of pointers passed in pPlugin.
+ (LPUNKNOWN*) &filter, // Pointer to the filter.
+ uuid.cElems, // Number of property pages.
+ uuid.pElems, // Array of property page CLSIDs.
+ LOCALE_USER_DEFAULT, // Locale ID for the dialog box.
+ 0, NULL); // Reserved
+ // Release memory.
+ if (uuid.pElems)
+ {
+ CoTaskMemFree(uuid.pElems);
+ }
+ filter->Release();
+ return 0;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h
new file mode 100644
index 0000000..b05a63c
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h
@@ -0,0 +1,105 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
+
+#include "../video_capture_impl.h"
+#include "../device_info_impl.h"
+
+#include <Dshow.h>
+#include "map_wrapper.h"
+
+// forward declarations
+namespace webrtc
+{
+namespace videocapturemodule
+{
+struct VideoCaptureCapabilityWindows: public VideoCaptureCapability
+{
+ WebRtc_UWord32 directShowCapabilityIndex;
+ bool supportFrameRateControl;
+ VideoCaptureCapabilityWindows()
+ {
+ directShowCapabilityIndex = 0;
+ supportFrameRateControl = false;
+ }
+
+};
+class DeviceInfoWindows: public DeviceInfoImpl
+{
+public:
+ DeviceInfoWindows(const WebRtc_Word32 id);
+ virtual ~DeviceInfoWindows();
+
+ WebRtc_Word32 Init();
+ virtual WebRtc_UWord32 NumberOfDevices();
+
+ /*
+ * Returns the available capture devices.
+ */
+ virtual WebRtc_Word32
+ GetDeviceName(WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length);
+
+ /*
+ * Display OS /capture device specific settings dialog
+ */
+ virtual WebRtc_Word32
+ DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord8* dialogTitleUTF8,
+ void* parentWindow,
+ WebRtc_UWord32 positionX,
+ WebRtc_UWord32 positionY);
+
+ // Windows specific
+
+ /* Gets a capture device filter
+ The user of this API is responsible for releasing the filter when it not needed.
+ */
+ IBaseFilter * GetDeviceFilter(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord8* productUniqueIdUTF8 = NULL,
+ WebRtc_UWord32 productUniqueIdUTF8Length = 0);
+
+ WebRtc_Word32
+ GetWindowsCapability(const WebRtc_Word32 capabilityIndex,
+ VideoCaptureCapabilityWindows& windowsCapability);
+
+ static void GetProductId(const WebRtc_UWord8* devicePath,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length);
+protected:
+
+ WebRtc_Word32 GetDeviceInfo(WebRtc_UWord32 deviceNumber,
+ WebRtc_UWord8* deviceNameUTF8,
+ WebRtc_UWord32 deviceNameLength,
+ WebRtc_UWord8* deviceUniqueIdUTF8,
+ WebRtc_UWord32 deviceUniqueIdUTF8Length,
+ WebRtc_UWord8* productUniqueIdUTF8,
+ WebRtc_UWord32 productUniqueIdUTF8Length);
+
+ virtual WebRtc_Word32
+ CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+private:
+ ICreateDevEnum* _dsDevEnum;
+ IEnumMoniker* _dsMonikerDevEnum;
+ bool _CoUninitializeIsRequired;
+
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc
new file mode 100644
index 0000000..dd533a3
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc
@@ -0,0 +1,116 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "help_functions_windows.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+// This returns minimum :), which will give max frame rate...
+LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size)
+{
+ LONGLONG maxFPS = maxFps[0];
+ for (int i = 0; i < size; i++)
+ {
+ if (maxFPS > maxFps[i])
+ maxFPS = maxFps[i];
+ }
+ return maxFPS;
+}
+
+IPin* GetInputPin(IBaseFilter* filter)
+{
+ HRESULT hr;
+ IPin* pin = NULL;
+ IEnumPins* pPinEnum = NULL;
+ filter->EnumPins(&pPinEnum);
+ if (pPinEnum == NULL)
+ {
+ return NULL;
+ }
+
+ // get first unconnected pin
+ hr = pPinEnum->Reset(); // set to first pin
+
+ int count = 0;
+ while (S_OK == pPinEnum->Next(1, &pin, NULL))
+ {
+ PIN_DIRECTION pPinDir;
+ pin->QueryDirection(&pPinDir);
+ if (PINDIR_INPUT == pPinDir) // This is an input pin
+ {
+ IPin* tempPin = NULL;
+ if (S_OK != pin->ConnectedTo(&tempPin)) // The pint is not connected
+ {
+ pPinEnum->Release();
+ return pin;
+ }
+ }
+ pin->Release();
+ }
+ pPinEnum->Release();
+ return NULL;
+}
+
+IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category)
+{
+ HRESULT hr;
+ IPin* pin = NULL;
+ IEnumPins* pPinEnum = NULL;
+ filter->EnumPins(&pPinEnum);
+ if (pPinEnum == NULL)
+ {
+ return NULL;
+ }
+ // get first unconnected pin
+ hr = pPinEnum->Reset(); // set to first pin
+ int count = 0;
+ while (S_OK == pPinEnum->Next(1, &pin, NULL))
+ {
+ PIN_DIRECTION pPinDir;
+ pin->QueryDirection(&pPinDir);
+ if (PINDIR_OUTPUT == pPinDir) // This is an output pin
+ {
+ if (Category == GUID_NULL || PinMatchesCategory(pin, Category))
+ {
+ pPinEnum->Release();
+ return pin;
+ }
+ }
+ pin->Release();
+ pin = NULL;
+ }
+ pPinEnum->Release();
+ return NULL;
+}
+
+BOOL PinMatchesCategory(IPin *pPin, REFGUID Category)
+{
+ BOOL bFound = FALSE;
+ IKsPropertySet *pKs = NULL;
+ HRESULT hr = pPin->QueryInterface(IID_PPV_ARGS(&pKs));
+ if (SUCCEEDED(hr))
+ {
+ GUID PinCategory;
+ DWORD cbReturned;
+ hr = pKs->Get(AMPROPSETID_Pin, AMPROPERTY_PIN_CATEGORY, NULL, 0, &PinCategory,
+ sizeof(GUID), &cbReturned);
+ if (SUCCEEDED(hr) && (cbReturned == sizeof(GUID)))
+ {
+ bFound = (PinCategory == Category);
+ }
+ pKs->Release();
+ }
+ return bFound;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
+
diff --git a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h
new file mode 100644
index 0000000..0020877
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
+
+#include <dshow.h>
+#include <initguid.h>
+
+DEFINE_GUID(MEDIASUBTYPE_I420, 0x30323449, 0x0000, 0x0010, 0x80, 0x00, 0x00,
+ 0xAA, 0x00, 0x38, 0x9B, 0x71);
+DEFINE_GUID(MEDIASUBTYPE_V210, 0x30313276, 0x0000, 0x0010, 0x80, 0x00, 0x00,
+ 0xAA, 0x00, 0x38, 0x9B, 0x71);
+DEFINE_GUID(MEDIASUBTYPE_HDYC, 0x43594448, 0x0000, 0x0010, 0x80, 0x00, 0x00,
+ 0xAA, 0x00, 0x38, 0x9B, 0x71);
+
+#define RELEASE_AND_CLEAR(p) if (p) { (p) -> Release () ; (p) = NULL ; }
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size);
+
+IPin* GetInputPin(IBaseFilter* filter);
+IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category = GUID_NULL);
+BOOL PinMatchesCategory(IPin *pPin, REFGUID Category);
+
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc
new file mode 100644
index 0000000..88e47a0
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc
@@ -0,0 +1,506 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sink_filter_windows.h"
+
+#include "trace.h"
+#include "help_functions_windows.h"
+
+#include <Dvdmedia.h> // VIDEOINFOHEADER2
+#include <initguid.h>
+
+#define DELETE_RESET(p) { delete (p) ; (p) = NULL ;}
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+typedef struct tagTHREADNAME_INFO
+{
+ DWORD dwType; // must be 0x1000
+ LPCSTR szName; // pointer to name (in user addr space)
+ DWORD dwThreadID; // thread ID (-1=caller thread)
+ DWORD dwFlags; // reserved for future use, must be zero
+} THREADNAME_INFO;
+
+DEFINE_GUID(CLSID_SINKFILTER, 0x88cdbbdc, 0xa73b, 0x4afa, 0xac, 0xbf, 0x15, 0xd5,
+ 0xe2, 0xce, 0x12, 0xc3);
+
+CaptureInputPin::CaptureInputPin (WebRtc_Word32 moduleId,
+ IN TCHAR * szName,
+ IN CaptureSinkFilter* pFilter,
+ IN CCritSec * pLock,
+ OUT HRESULT * pHr,
+ IN LPCWSTR pszName)
+ : CBaseInputPin (szName, pFilter, pLock, pHr, pszName),
+ _requestedCapability(),
+ _resultingCapability()
+{
+ _moduleId=moduleId;
+ _threadHandle = NULL;
+}
+
+CaptureInputPin::~CaptureInputPin()
+{
+}
+
+HRESULT
+CaptureInputPin::GetMediaType (IN int iPosition, OUT CMediaType * pmt)
+{
+ // reset the thread handle
+ _threadHandle = NULL;
+
+ if(iPosition < 0)
+ return E_INVALIDARG;
+
+ VIDEOINFOHEADER* pvi = (VIDEOINFOHEADER*) pmt->AllocFormatBuffer(
+ sizeof(VIDEOINFOHEADER));
+ if(NULL == pvi)
+ {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
+ "CheckMediaType VIDEOINFOHEADER is NULL. Returning...Line:%d\n", __LINE__);
+ return(E_OUTOFMEMORY);
+ }
+
+ ZeroMemory(pvi, sizeof(VIDEOINFOHEADER));
+ pvi->bmiHeader.biSize = sizeof(BITMAPINFOHEADER);
+ pvi->bmiHeader.biPlanes = 1;
+ pvi->bmiHeader.biClrImportant = 0;
+ pvi->bmiHeader.biClrUsed = 0;
+ pvi->AvgTimePerFrame = 10000000/_requestedCapability.maxFPS;
+
+ SetRectEmpty(&(pvi->rcSource)); // we want the whole image area rendered.
+ SetRectEmpty(&(pvi->rcTarget)); // no particular destination rectangle
+
+ pmt->SetType(&MEDIATYPE_Video);
+ pmt->SetFormatType(&FORMAT_VideoInfo);
+ pmt->SetTemporalCompression(FALSE);
+
+ WebRtc_Word32 positionOffset=1;
+ if(_requestedCapability.codecType!=kVideoCodecUnknown)
+ {
+ positionOffset=0;
+ }
+
+ switch (iPosition+positionOffset)
+ {
+ case 0:
+ {
+ pvi->bmiHeader.biCompression = MAKEFOURCC('I','4','2','0');
+ pvi->bmiHeader.biBitCount = 12; //bit per pixel
+ pvi->bmiHeader.biWidth = _requestedCapability.width;
+ pvi->bmiHeader.biHeight = _requestedCapability.height;
+ pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
+ *_requestedCapability.width/2;
+ pmt->SetSubtype(&MEDIASUBTYPE_I420);
+ }
+ break;
+ case 1:
+ {
+ pvi->bmiHeader.biCompression = MAKEFOURCC('Y','U','Y','2');;
+ pvi->bmiHeader.biBitCount = 16; //bit per pixel
+ pvi->bmiHeader.biWidth = _requestedCapability.width;
+ pvi->bmiHeader.biHeight = _requestedCapability.height;
+ pvi->bmiHeader.biSizeImage = 2*_requestedCapability.width
+ *_requestedCapability.height;
+ pmt->SetSubtype(&MEDIASUBTYPE_YUY2);
+ }
+ break;
+ case 2:
+ {
+ pvi->bmiHeader.biCompression = BI_RGB;
+ pvi->bmiHeader.biBitCount = 24; //bit per pixel
+ pvi->bmiHeader.biWidth = _requestedCapability.width;
+ pvi->bmiHeader.biHeight = _requestedCapability.height;
+ pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
+ *_requestedCapability.width;
+ pmt->SetSubtype(&MEDIASUBTYPE_RGB24);
+ }
+ break;
+ case 3:
+ {
+ pvi->bmiHeader.biCompression = MAKEFOURCC('U','Y','V','Y');
+ pvi->bmiHeader.biBitCount = 16; //bit per pixel
+ pvi->bmiHeader.biWidth = _requestedCapability.width;
+ pvi->bmiHeader.biHeight = _requestedCapability.height;
+ pvi->bmiHeader.biSizeImage = 2*_requestedCapability.height
+ *_requestedCapability.width;
+ pmt->SetSubtype(&MEDIASUBTYPE_UYVY);
+ }
+ break;
+ case 4:
+ {
+ pvi->bmiHeader.biCompression = MAKEFOURCC('M','J','P','G');
+ pvi->bmiHeader.biBitCount = 12; //bit per pixel
+ pvi->bmiHeader.biWidth = _requestedCapability.width;
+ pvi->bmiHeader.biHeight = _requestedCapability.height;
+ pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
+ *_requestedCapability.width/2;
+ pmt->SetSubtype(&MEDIASUBTYPE_MJPG);
+ }
+ break;
+ default :
+ return VFW_S_NO_MORE_ITEMS;
+ }
+ pmt->SetSampleSize(pvi->bmiHeader.biSizeImage);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
+ "GetMediaType position %d, width %d, height %d, biCompression 0x%x",
+ iPosition, _requestedCapability.width,
+ _requestedCapability.height,pvi->bmiHeader.biCompression);
+ return NOERROR;
+}
+
+HRESULT
+CaptureInputPin::CheckMediaType ( IN const CMediaType * pMediaType)
+{
+ // reset the thread handle
+ _threadHandle = NULL;
+
+ const GUID *type = pMediaType->Type();
+ if (*type != MEDIATYPE_Video)
+ return E_INVALIDARG;
+
+ const GUID *formatType = pMediaType->FormatType();
+
+ // Check for the subtypes we support
+ const GUID *SubType = pMediaType->Subtype();
+ if (SubType == NULL)
+ {
+ return E_INVALIDARG;
+ }
+
+ if(*formatType == FORMAT_VideoInfo)
+ {
+ VIDEOINFOHEADER *pvi = (VIDEOINFOHEADER *) pMediaType->Format();
+ if(pvi == NULL)
+ {
+ return E_INVALIDARG;
+ }
+
+ // Store the incoming width and height
+ _resultingCapability.width = pvi->bmiHeader.biWidth;
+ _resultingCapability.height = abs(pvi->bmiHeader.biHeight);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
+ "CheckMediaType width:%d height:%d Compression:0x%x\n",
+ pvi->bmiHeader.biWidth,pvi->bmiHeader.biHeight,
+ pvi->bmiHeader.biCompression);
+
+ if(*SubType == MEDIASUBTYPE_MJPG
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('M','J','P','G'))
+ {
+ _resultingCapability.rawType = kVideoMJPEG;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_I420
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('I','4','2','0'))
+ {
+ _resultingCapability.rawType = kVideoI420;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_YUY2
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('Y','U','Y','2'))
+ {
+ _resultingCapability.rawType = kVideoYUY2;
+ ::Sleep(60); // workaround for bad driver
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_UYVY
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('U','Y','V','Y'))
+ {
+ _resultingCapability.rawType = kVideoUYVY;
+ return S_OK; // This format is acceptable.
+ }
+
+ if(*SubType == MEDIASUBTYPE_HDYC)
+ {
+ _resultingCapability.rawType = kVideoUYVY;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_RGB24
+ && pvi->bmiHeader.biCompression == BI_RGB)
+ {
+ _resultingCapability.rawType = kVideoRGB24;
+ return S_OK; // This format is acceptable.
+ }
+ }
+ if(*formatType == FORMAT_VideoInfo2)
+ {
+ // VIDEOINFOHEADER2 that has dwInterlaceFlags
+ VIDEOINFOHEADER2 *pvi = (VIDEOINFOHEADER2 *) pMediaType->Format();
+
+ if(pvi == NULL)
+ {
+ return E_INVALIDARG;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
+ "CheckMediaType width:%d height:%d Compression:0x%x\n",
+ pvi->bmiHeader.biWidth,pvi->bmiHeader.biHeight,
+ pvi->bmiHeader.biCompression);
+
+ _resultingCapability.width = pvi->bmiHeader.biWidth;
+ _resultingCapability.height = abs(pvi->bmiHeader.biHeight);
+
+ if(*SubType == MEDIASUBTYPE_MJPG
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('M','J','P','G'))
+ {
+ _resultingCapability.rawType = kVideoMJPEG;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_I420
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('I','4','2','0'))
+ {
+ _resultingCapability.rawType = kVideoI420;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_YUY2
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('Y','U','Y','2'))
+ {
+ _resultingCapability.rawType = kVideoYUY2;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_UYVY
+ && pvi->bmiHeader.biCompression == MAKEFOURCC('U','Y','V','Y'))
+ {
+ _resultingCapability.rawType = kVideoUYVY;
+ return S_OK; // This format is acceptable.
+ }
+
+ if(*SubType == MEDIASUBTYPE_HDYC)
+ {
+ _resultingCapability.rawType = kVideoUYVY;
+ return S_OK; // This format is acceptable.
+ }
+ if(*SubType == MEDIASUBTYPE_RGB24
+ && pvi->bmiHeader.biCompression == BI_RGB)
+ {
+ _resultingCapability.rawType = kVideoRGB24;
+ return S_OK; // This format is acceptable.
+ }
+ }
+ return E_INVALIDARG;
+}
+
+HRESULT
+CaptureInputPin::Receive ( IN IMediaSample * pIMediaSample )
+{
+ HRESULT hr = S_OK;
+
+ ASSERT (m_pFilter);
+ ASSERT (pIMediaSample);
+
+ // get the thread handle of the delivering thread inc its priority
+ if( _threadHandle == NULL)
+ {
+ HANDLE handle= GetCurrentThread();
+ SetThreadPriority(handle, THREAD_PRIORITY_HIGHEST);
+ _threadHandle = handle;
+ // See http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx for details on the code
+ // in this function. Name od article is "Setting a Thread Name (Unmanaged)".
+
+ THREADNAME_INFO info;
+ info.dwType = 0x1000;
+ info.szName = "capture_thread";
+ info.dwThreadID = (DWORD)-1;
+ info.dwFlags = 0;
+
+ __try
+ {
+ RaiseException( 0x406D1388, 0, sizeof(info)/sizeof(DWORD),
+ (DWORD_PTR*)&info );
+ }
+ __except (EXCEPTION_CONTINUE_EXECUTION)
+ {
+ }
+
+ }
+
+ reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->LockReceive();
+ hr = CBaseInputPin::Receive (pIMediaSample);
+
+ if (SUCCEEDED (hr))
+ {
+ const WebRtc_Word32 length = pIMediaSample->GetActualDataLength();
+
+ unsigned char* pBuffer = NULL;
+ if(S_OK != pIMediaSample->GetPointer(&pBuffer))
+ {
+ reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
+ return S_FALSE;
+ }
+
+ // NOTE: filter unlocked within Send call
+ reinterpret_cast <CaptureSinkFilter *> (m_pFilter)->ProcessCapturedFrame(
+ pBuffer,length,_resultingCapability);
+ }
+ else
+ {
+ reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
+ }
+
+ return hr;
+}
+
+// called under LockReceive
+HRESULT CaptureInputPin::SetMatchingMediaType(
+ const VideoCaptureCapability& capability)
+{
+
+ _requestedCapability = capability;
+ _resultingCapability = VideoCaptureCapability();
+ return S_OK;
+}
+// ----------------------------------------------------------------------------
+CaptureSinkFilter::CaptureSinkFilter (IN TCHAR * tszName,
+ IN LPUNKNOWN punk,
+ OUT HRESULT * phr,
+ VideoCaptureExternal& captureObserver,
+ WebRtc_Word32 moduleId)
+ : CBaseFilter(tszName,punk,& m_crtFilter,CLSID_SINKFILTER),
+ m_pInput(NULL),
+ _captureObserver(captureObserver),
+ _moduleId(moduleId)
+{
+ (* phr) = S_OK;
+ m_pInput = new CaptureInputPin(moduleId,NAME ("VideoCaptureInputPin"),
+ this,
+ & m_crtFilter,
+ phr, L"VideoCapture");
+ if (m_pInput == NULL || FAILED (* phr))
+ {
+ (* phr) = FAILED (* phr) ? (* phr) : E_OUTOFMEMORY;
+ goto cleanup;
+ }
+ cleanup :
+ return;
+}
+
+CaptureSinkFilter::~CaptureSinkFilter()
+{
+ delete m_pInput;
+}
+
+int CaptureSinkFilter::GetPinCount()
+{
+ return 1;
+}
+
+CBasePin *
+CaptureSinkFilter::GetPin(IN int Index)
+{
+ CBasePin * pPin;
+ LockFilter ();
+ if (Index == 0)
+ {
+ pPin = m_pInput;
+ }
+ else
+ {
+ pPin = NULL;
+ }
+ UnlockFilter ();
+ return pPin;
+}
+
+STDMETHODIMP CaptureSinkFilter::Pause()
+{
+ HRESULT hr = S_OK;
+ LockFilter();
+ if (m_State == State_Stopped)
+ {
+ // change the state, THEN activate the input pin
+ m_State = State_Paused;
+ if (m_pInput && m_pInput->IsConnected())
+ {
+ m_pInput->Active();
+ }
+ if (m_pInput && !m_pInput->IsConnected())
+ {
+ m_State = State_Running;
+ }
+ }
+ else if (m_State == State_Running)
+ {
+ m_State = State_Paused;
+ }
+ UnlockFilter();
+ return S_OK;
+}
+
+STDMETHODIMP CaptureSinkFilter::Stop()
+{
+ LockReceive();
+ LockFilter();
+
+ // set the state
+ m_State = State_Stopped;
+
+ // inactivate the pins
+ if (m_pInput)
+ m_pInput->Inactive();
+
+ UnlockFilter();
+ UnlockReceive();
+ return S_OK;
+}
+
+void CaptureSinkFilter::SetFilterGraph(IGraphBuilder* graph)
+{
+ LockFilter();
+ m_pGraph = graph;
+ UnlockFilter();
+}
+
+void CaptureSinkFilter::ProcessCapturedFrame(unsigned char* pBuffer,
+ WebRtc_Word32 length,
+ const VideoCaptureCapability& frameInfo)
+{
+ // we have the receiver lock
+ if (m_State == State_Running)
+ {
+ _captureObserver.IncomingFrame(pBuffer, length, frameInfo);
+
+ // trying to hold it since it's only a memcpy
+ // IMPROVEMENT if this work move critsect
+ UnlockReceive();
+ return;
+ }
+ UnlockReceive();
+ return;
+}
+
+STDMETHODIMP CaptureSinkFilter::SetMatchingMediaType(
+ const VideoCaptureCapability& capability)
+{
+ LockReceive();
+ LockFilter();
+ HRESULT hr;
+ if (m_pInput)
+ {
+ hr = m_pInput->SetMatchingMediaType(capability);
+ }
+ else
+ {
+ hr = E_UNEXPECTED;
+ }
+ UnlockFilter();
+ UnlockReceive();
+ return hr;
+}
+
+STDMETHODIMP CaptureSinkFilter::GetClassID( OUT CLSID * pCLSID )
+{
+ (* pCLSID) = CLSID_SINKFILTER;
+ return S_OK;
+}
+
+} // namespace videocapturemodule
+} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h
new file mode 100644
index 0000000..efeb9c7
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
+
+#include <Streams.h> // Include base DS filter header files
+
+#include "video_capture_defines.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+//forward declaration
+
+class CaptureSinkFilter;
+/**
+ * input pin for camera input
+ *
+ */
+class CaptureInputPin: public CBaseInputPin
+{
+public:
+ WebRtc_Word32 _moduleId;
+
+ VideoCaptureCapability _requestedCapability;
+ VideoCaptureCapability _resultingCapability;
+ HANDLE _threadHandle;
+
+ CaptureInputPin ( WebRtc_Word32 moduleId,
+ IN TCHAR* szName,
+ IN CaptureSinkFilter* pFilter,
+ IN CCritSec * pLock,
+ OUT HRESULT * pHr,
+ IN LPCWSTR pszName);
+ virtual ~CaptureInputPin();
+
+ HRESULT GetMediaType (IN int iPos, OUT CMediaType * pmt);
+ HRESULT CheckMediaType (IN const CMediaType * pmt);
+ STDMETHODIMP Receive (IN IMediaSample *);
+ HRESULT SetMatchingMediaType(const VideoCaptureCapability& capability);
+};
+
+class CaptureSinkFilter: public CBaseFilter
+{
+
+public:
+ CaptureSinkFilter (IN TCHAR * tszName,
+ IN LPUNKNOWN punk,
+ OUT HRESULT * phr,
+ VideoCaptureExternal& captureObserver,
+ WebRtc_Word32 moduleId);
+ virtual ~CaptureSinkFilter();
+
+ // --------------------------------------------------------------------
+ // class methods
+
+ void ProcessCapturedFrame(unsigned char* pBuffer, WebRtc_Word32 length,
+ const VideoCaptureCapability& frameInfo);
+ // explicit receiver lock aquisition and release
+ void LockReceive() { m_crtRecv.Lock();}
+ void UnlockReceive() {m_crtRecv.Unlock();}
+ // explicit filter lock aquisition and release
+ void LockFilter() {m_crtFilter.Lock();}
+ void UnlockFilter() { m_crtFilter.Unlock(); }
+ void SetFilterGraph(IGraphBuilder* graph); // Used if EVR
+
+ // --------------------------------------------------------------------
+ // COM interfaces
+DECLARE_IUNKNOWN ;
+ STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
+
+ // --------------------------------------------------------------------
+ // CBaseFilter methods
+ int GetPinCount ();
+ CBasePin * GetPin ( IN int Index);
+ STDMETHODIMP Pause ();
+ STDMETHODIMP Stop ();
+ STDMETHODIMP GetClassID ( OUT CLSID * pCLSID);
+ // --------------------------------------------------------------------
+ // class factory calls this
+ static CUnknown * CreateInstance (IN LPUNKNOWN punk, OUT HRESULT * phr);
+private:
+ CCritSec m_crtFilter; // filter lock
+ CCritSec m_crtRecv; // receiver lock; always acquire before filter lock
+ CaptureInputPin * m_pInput;
+ VideoCaptureExternal& _captureObserver;
+ WebRtc_Word32 _moduleId;
+};
+} // namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc
new file mode 100644
index 0000000..bfc7a30
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ref_count.h"
+#include "video_capture_windows.h"
+#include "trace.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule* VideoCaptureImpl::Create(
+ const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ if (deviceUniqueIdUTF8 == NULL)
+ {
+ return NULL;
+ }
+
+ WebRtc_UWord8 productId[kVideoCaptureProductIdLength];
+ videocapturemodule::DeviceInfoWindows::GetProductId(deviceUniqueIdUTF8,
+ productId,
+ sizeof(productId));
+
+ RefCountImpl<videocapturemodule::VideoCaptureDS>* newCaptureModule =
+ new RefCountImpl<videocapturemodule::VideoCaptureDS>(id);
+
+ if (newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
+ {
+ delete newCaptureModule;
+ newCaptureModule = NULL;
+ }
+ return newCaptureModule;
+}
+} //namespace videocapturemodule
+} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc
new file mode 100644
index 0000000..889b7ef
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc
@@ -0,0 +1,413 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_capture_windows.h"
+
+#include "../video_capture_config.h"
+#include "critical_section_wrapper.h"
+#include "help_functions_windows.h"
+#include "sink_filter_windows.h"
+#include "trace.h"
+
+#include <Dvdmedia.h> // VIDEOINFOHEADER2
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureDS::VideoCaptureDS(const WebRtc_Word32 id)
+ : VideoCaptureImpl(id), _dsInfo(id), _captureFilter(NULL),
+ _graphBuilder(NULL), _mediaControl(NULL), _sinkFilter(NULL),
+ _inputSendPin(NULL), _outputCapturePin(NULL), _dvFilter(NULL),
+ _inputDvPin(NULL), _outputDvPin(NULL)
+{
+}
+
+VideoCaptureDS::~VideoCaptureDS()
+{
+ if (_mediaControl)
+ {
+ _mediaControl->Stop();
+ }
+ if (_graphBuilder)
+ {
+ if (_sinkFilter)
+ _graphBuilder->RemoveFilter(_sinkFilter);
+ if (_captureFilter)
+ _graphBuilder->RemoveFilter(_captureFilter);
+ if (_dvFilter)
+ _graphBuilder->RemoveFilter(_dvFilter);
+ }
+ RELEASE_AND_CLEAR(_captureFilter); // release the capture device
+ RELEASE_AND_CLEAR(_sinkFilter);
+ RELEASE_AND_CLEAR(_dvFilter);
+
+ RELEASE_AND_CLEAR(_mediaControl);
+ RELEASE_AND_CLEAR(_inputSendPin);
+ RELEASE_AND_CLEAR(_outputCapturePin);
+
+ RELEASE_AND_CLEAR(_inputDvPin);
+ RELEASE_AND_CLEAR(_outputDvPin);
+
+ RELEASE_AND_CLEAR(_graphBuilder);
+}
+
+WebRtc_Word32 VideoCaptureDS::Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+ WebRtc_Word32 result = 0;
+
+ const WebRtc_Word32 nameLength =
+ (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
+ if (nameLength > kVideoCaptureUniqueNameLength)
+ return -1;
+
+ // Store the device name
+ _deviceUniqueId = new (std::nothrow) WebRtc_UWord8[nameLength + 1];
+ memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
+
+ if (_dsInfo.Init() != 0)
+ return -1;
+
+ _captureFilter = _dsInfo.GetDeviceFilter(deviceUniqueIdUTF8);
+ if (!_captureFilter)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create capture filter.");
+ return -1;
+ }
+
+ // Get the interface for DirectShow's GraphBuilder
+ HRESULT hr = CoCreateInstance(CLSID_FilterGraph, NULL,
+ CLSCTX_INPROC_SERVER, IID_IGraphBuilder,
+ (void **) &_graphBuilder);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create graph builder.");
+ return -1;
+ }
+
+ hr = _graphBuilder->QueryInterface(IID_IMediaControl,
+ (void **) &_mediaControl);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create media control builder.");
+ return -1;
+ }
+ hr = _graphBuilder->AddFilter(_captureFilter, CAPTURE_FILTER_NAME);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to add the capture device to the graph.");
+ return -1;
+ }
+
+ _outputCapturePin = GetOutputPin(_captureFilter, PIN_CATEGORY_CAPTURE);
+
+ // Create the sink filte used for receiving Captured frames.
+ _sinkFilter = new CaptureSinkFilter(SINK_FILTER_NAME, NULL, &hr,
+ *this, _id);
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create send filter");
+ return -1;
+ }
+ _sinkFilter->AddRef();
+
+ hr = _graphBuilder->AddFilter(_sinkFilter, SINK_FILTER_NAME);
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to add the send filter to the graph.");
+ return -1;
+ }
+ _inputSendPin = GetInputPin(_sinkFilter);
+
+ // Temporary connect here.
+ // This is done so that no one else can use the capture device.
+ if (SetCameraOutput(_requestedCapability) != 0)
+ {
+ return -1;
+ }
+ hr = _mediaControl->Pause();
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to Pause the Capture device. Is it already occupied? %d.",
+ hr);
+ return -1;
+ }
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, _id,
+ "Capture device '%s' initialized.", deviceUniqueIdUTF8);
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureDS::StartCapture(
+ const VideoCaptureCapability& capability)
+{
+ CriticalSectionScoped cs(_apiCs);
+
+ if (capability != _requestedCapability)
+ {
+ DisconnectGraph();
+
+ if (SetCameraOutput(capability) != 0)
+ {
+ return -1;
+ }
+ }
+ HRESULT hr = _mediaControl->Run();
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to start the Capture device.");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureDS::StopCapture()
+{
+ CriticalSectionScoped cs(_apiCs);
+
+ HRESULT hr = _mediaControl->Pause();
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to stop the capture graph. %d", hr);
+ return -1;
+ }
+ return 0;
+}
+bool VideoCaptureDS::CaptureStarted()
+{
+ OAFilterState state = 0;
+ HRESULT hr = _mediaControl->GetState(1000, &state);
+ if (hr != S_OK && hr != VFW_S_CANT_CUE)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to get the CaptureStarted status");
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "CaptureStarted %d", state);
+ return state == State_Running;
+
+}
+WebRtc_Word32 VideoCaptureDS::CaptureSettings(
+ VideoCaptureCapability& settings)
+{
+ settings = _requestedCapability;
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureDS::SetCameraOutput(
+ const VideoCaptureCapability& requestedCapability)
+{
+
+ // Get the best matching capability
+ VideoCaptureCapability capability;
+ WebRtc_Word32 capabilityIndex;
+
+ // Store the new requested size
+ _requestedCapability = requestedCapability;
+ // Match the requested capability with the supported.
+ if ((capabilityIndex = _dsInfo.GetBestMatchedCapability(_deviceUniqueId,
+ _requestedCapability,
+ capability)) < 0)
+ {
+ return -1;
+ }
+ //Reduce the frame rate if possible.
+ if (capability.maxFPS > requestedCapability.maxFPS)
+ {
+ capability.maxFPS = requestedCapability.maxFPS;
+ }
+ // Store the new expected capture delay
+ _captureDelay = capability.expectedCaptureDelay;
+
+ // Convert it to the windows capability index since they are not nexessary
+ // the same
+ VideoCaptureCapabilityWindows windowsCapability;
+ if (_dsInfo.GetWindowsCapability(capabilityIndex, windowsCapability) != 0)
+ {
+ return -1;
+ }
+
+ IAMStreamConfig* streamConfig = NULL;
+ AM_MEDIA_TYPE *pmt = NULL;
+ VIDEO_STREAM_CONFIG_CAPS caps;
+
+ HRESULT hr = _outputCapturePin->QueryInterface(IID_IAMStreamConfig,
+ (void**) &streamConfig);
+ if (hr)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Can't get the Capture format settings.");
+ return -1;
+ }
+
+ //Get the windows capability from the capture device
+ bool isDVCamera = false;
+ hr = streamConfig->GetStreamCaps(
+ windowsCapability.directShowCapabilityIndex,
+ &pmt, reinterpret_cast<BYTE*> (&caps));
+ if (!FAILED(hr))
+ {
+ if (pmt->formattype == FORMAT_VideoInfo2)
+ {
+ VIDEOINFOHEADER2* h =
+ reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
+ if (capability.maxFPS > 0
+ && windowsCapability.supportFrameRateControl)
+ {
+ h->AvgTimePerFrame = REFERENCE_TIME(10000000.0
+ / capability.maxFPS);
+ }
+ }
+ else
+ {
+ VIDEOINFOHEADER* h = reinterpret_cast<VIDEOINFOHEADER*>
+ (pmt->pbFormat);
+ if (capability.maxFPS > 0
+ && windowsCapability.supportFrameRateControl)
+ {
+ h->AvgTimePerFrame = REFERENCE_TIME(10000000.0
+ / capability.maxFPS);
+ }
+
+ }
+
+ // Set the sink filter to request this capability
+ _sinkFilter->SetMatchingMediaType(capability);
+ //Order the capture device to use this capability
+ hr += streamConfig->SetFormat(pmt);
+
+ //Check if this is a DV camera and we need to add MS DV Filter
+ if (pmt->subtype == MEDIASUBTYPE_dvsl
+ || pmt->subtype == MEDIASUBTYPE_dvsd
+ || pmt->subtype == MEDIASUBTYPE_dvhd)
+ isDVCamera = true; // This is a DV camera. Use MS DV filter
+ }
+ RELEASE_AND_CLEAR(streamConfig);
+
+ if (FAILED(hr))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to set capture device output format");
+ return -1;
+ }
+
+ if (isDVCamera)
+ {
+ hr = ConnectDVCamera();
+ }
+ else
+ {
+ hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputSendPin,
+ NULL);
+ }
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to connect the Capture graph %d", hr);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureDS::DisconnectGraph()
+{
+ HRESULT hr = _mediaControl->Stop();
+ hr += _graphBuilder->Disconnect(_outputCapturePin);
+ hr += _graphBuilder->Disconnect(_inputSendPin);
+
+ //if the DV camera filter exist
+ if (_dvFilter)
+ {
+ _graphBuilder->Disconnect(_inputDvPin);
+ _graphBuilder->Disconnect(_outputDvPin);
+ }
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE( webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to Stop the Capture device for reconfiguration %d",
+ hr);
+ return -1;
+ }
+ return 0;
+}
+HRESULT VideoCaptureDS::ConnectDVCamera()
+{
+ HRESULT hr = S_OK;
+
+ if (!_dvFilter)
+ {
+ hr = CoCreateInstance(CLSID_DVVideoCodec, NULL, CLSCTX_INPROC,
+ IID_IBaseFilter, (void **) &_dvFilter);
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to create the dv decoder: %x", hr);
+ return hr;
+ }
+ hr = _graphBuilder->AddFilter(_dvFilter, L"VideoDecoderDV");
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to add the dv decoder to the graph: %x", hr);
+ return hr;
+ }
+ _inputDvPin = GetInputPin(_dvFilter);
+ if (_inputDvPin == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to get input pin from DV decoder");
+ return -1;
+ }
+ _outputDvPin = GetOutputPin(_dvFilter);
+ if (_outputDvPin == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to get output pin from DV decoder");
+ return -1;
+ }
+ }
+ hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputDvPin, NULL);
+ if (hr != S_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to connect capture device to the dv devoder: %x",
+ hr);
+ return hr;
+ }
+
+ hr = _graphBuilder->ConnectDirect(_outputDvPin, _inputSendPin, NULL);
+ if (hr != S_OK)
+ {
+ if (hr == 0x80070004)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to connect the capture device, busy");
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to connect capture device to the send graph: 0x%x",
+ hr);
+ }
+ return hr;
+ }
+ return hr;
+}
+} // namespace videocapturemodule
+} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h
new file mode 100644
index 0000000..69e2db5
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
+
+#include "../video_capture_impl.h"
+#include <tchar.h>
+
+#include "device_info_windows.h"
+
+#define CAPTURE_FILTER_NAME L"VideoCaptureFilter"
+#define SINK_FILTER_NAME L"SinkFilter"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+// Forward declaraion
+class CaptureSinkFilter;
+
+class VideoCaptureDS: public VideoCaptureImpl
+{
+public:
+
+ static VideoCaptureModule* Create(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ VideoCaptureDS(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 Init(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ /*************************************************************************
+ *
+ * Start/Stop
+ *
+ *************************************************************************/
+ virtual WebRtc_Word32
+ StartCapture(const VideoCaptureCapability& capability);
+ virtual WebRtc_Word32 StopCapture();
+
+ /**************************************************************************
+ *
+ * Properties of the set device
+ *
+ **************************************************************************/
+
+ virtual bool CaptureStarted();
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
+
+protected:
+ virtual ~VideoCaptureDS();
+
+ // Help functions
+
+ WebRtc_Word32
+ SetCameraOutput(const VideoCaptureCapability& requestedCapability);
+ WebRtc_Word32 DisconnectGraph();
+ HRESULT VideoCaptureDS::ConnectDVCamera();
+
+ DeviceInfoWindows _dsInfo;
+
+ IBaseFilter* _captureFilter;
+ IGraphBuilder* _graphBuilder;
+ IMediaControl* _mediaControl;
+ CaptureSinkFilter* _sinkFilter;
+ IPin* _inputSendPin;
+ IPin* _outputCapturePin;
+
+ // Microsoft DV interface (external DV cameras)
+ IBaseFilter* _dvFilter;
+ IPin* _inputDvPin;
+ IPin* _outputDvPin;
+
+};
+} // namespace videocapturemodule
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/device_info_impl.cc b/trunk/src/modules/video_capture/main/source/device_info_impl.cc
new file mode 100644
index 0000000..e41e0bb
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/device_info_impl.cc
@@ -0,0 +1,400 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "device_info_impl.h"
+#include "video_capture_config.h"
+#include "trace.h"
+#include <stdlib.h>
+
+#ifndef abs
+#define abs(a) (a>=0?a:-a)
+#endif
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+DeviceInfoImpl::DeviceInfoImpl(const WebRtc_Word32 id)
+ : _id(id), _apiLock(*RWLockWrapper::CreateRWLock()), _lastUsedDeviceName(NULL),
+ _lastUsedDeviceNameLength(0)
+{
+}
+
+DeviceInfoImpl::~DeviceInfoImpl(void)
+{
+ _apiLock.AcquireLockExclusive();
+ // Reset old capability list
+ MapItem* item = NULL;
+ while ((item = _captureCapabilities.Last()))
+ {
+ delete (VideoCaptureCapability*) item->GetItem();
+ _captureCapabilities.Erase(item);
+ }
+ free(_lastUsedDeviceName);
+ _apiLock.ReleaseLockExclusive();
+
+ delete &_apiLock;
+}
+WebRtc_Word32 DeviceInfoImpl::NumberOfCapabilities(
+ const WebRtc_UWord8* deviceUniqueIdUTF8)
+{
+
+ if (!deviceUniqueIdUTF8)
+ return -1;
+
+ _apiLock.AcquireLockShared();
+
+ if (_lastUsedDeviceNameLength == strlen((char*) deviceUniqueIdUTF8))
+ {
+ // Is it the same device that is asked for again.
+#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
+ if(strncasecmp((char*)_lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength)==0)
+#else
+ if (_strnicmp((char*) _lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength) == 0)
+#endif
+ {
+ //yes
+ _apiLock.ReleaseLockShared();
+ return _captureCapabilities.Size();
+ }
+ }
+ // Need to get exclusive rights to create the new capability map.
+ _apiLock.ReleaseLockShared();
+ WriteLockScoped cs2(_apiLock);
+
+ WebRtc_Word32 ret = CreateCapabilityMap(deviceUniqueIdUTF8);
+ return ret;
+}
+
+WebRtc_Word32 DeviceInfoImpl::GetCapability(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability)
+{
+
+ if (!deviceUniqueIdUTF8)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "deviceUniqueIdUTF8 parameter not set in call to GetCapability");
+ return -1;
+ }
+ ReadLockScoped cs(_apiLock);
+
+ if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
+#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
+ || (strncasecmp((char*)_lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength)!=0))
+#else
+ || (_strnicmp((char*) _lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength) != 0))
+#endif
+
+ {
+ _apiLock.ReleaseLockShared();
+ _apiLock.AcquireLockExclusive();
+ if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8))
+ {
+ _apiLock.ReleaseLockExclusive();
+ _apiLock.AcquireLockShared();
+ return -1;
+ }
+ _apiLock.ReleaseLockExclusive();
+ _apiLock.AcquireLockShared();
+ }
+
+ // Make sure the number is valid
+ if (deviceCapabilityNumber >= (unsigned int) _captureCapabilities.Size())
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "deviceCapabilityNumber %d is invalid in call to GetCapability",
+ deviceCapabilityNumber);
+ return -1;
+ }
+
+ MapItem* item = _captureCapabilities.Find(deviceCapabilityNumber);
+ if (!item)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to find capability number %d of %d possible",
+ deviceCapabilityNumber, _captureCapabilities.Size());
+ return -1;
+ }
+
+ VideoCaptureCapability* capPointer = static_cast<VideoCaptureCapability*>
+ (item->GetItem());
+ if (!capPointer)
+ {
+ return -1;
+ }
+
+ capability = *capPointer;
+ return 0;
+}
+
+WebRtc_Word32 DeviceInfoImpl::GetBestMatchedCapability(
+ const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested,
+ VideoCaptureCapability& resulting)
+{
+
+
+ if (!deviceUniqueIdUTF8)
+ return -1;
+
+ ReadLockScoped cs(_apiLock);
+ if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
+#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
+ || (strncasecmp((char*)_lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength)!=0))
+#else
+ || (_strnicmp((char*) _lastUsedDeviceName,
+ (char*) deviceUniqueIdUTF8,
+ _lastUsedDeviceNameLength) != 0))
+#endif
+ {
+ _apiLock.ReleaseLockShared();
+ _apiLock.AcquireLockExclusive();
+ if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8))
+ {
+ return -1;
+ }
+ _apiLock.ReleaseLockExclusive();
+ _apiLock.AcquireLockShared();
+ }
+
+ WebRtc_Word32 bestformatIndex = -1;
+ WebRtc_Word32 bestWidth = 0;
+ WebRtc_Word32 bestHeight = 0;
+ WebRtc_Word32 bestFrameRate = 0;
+ RawVideoType bestRawType = kVideoUnknown;
+ webrtc::VideoCodecType bestCodecType = webrtc::kVideoCodecUnknown;
+
+ const WebRtc_Word32 numberOfCapabilies = _captureCapabilities.Size();
+
+ for (WebRtc_Word32 tmp = 0; tmp < numberOfCapabilies; ++tmp) // Loop through all capabilities
+ {
+ MapItem* item = _captureCapabilities.Find(tmp);
+ if (!item)
+ return -1;
+
+ VideoCaptureCapability& capability = *static_cast<VideoCaptureCapability*>
+ (item->GetItem());
+
+ const WebRtc_Word32 diffWidth = capability.width - requested.width;
+ const WebRtc_Word32 diffHeight = capability.height - requested.height;
+ const WebRtc_Word32 diffFrameRate = capability.maxFPS - requested.maxFPS;
+
+ const WebRtc_Word32 currentbestDiffWith = bestWidth - requested.width;
+ const WebRtc_Word32 currentbestDiffHeight = bestHeight - requested.height;
+ const WebRtc_Word32 currentbestDiffFrameRate = bestFrameRate - requested.maxFPS;
+
+ if ((diffHeight >= 0 && diffHeight <= abs(currentbestDiffHeight)) // Height better or equalt that previouse.
+ || (currentbestDiffHeight < 0 && diffHeight >= currentbestDiffHeight))
+ {
+
+ if (diffHeight == currentbestDiffHeight) // Found best height. Care about the width)
+ {
+ if ((diffWidth >= 0 && diffWidth <= abs(currentbestDiffWith)) // Width better or equal
+ || (currentbestDiffWith < 0 && diffWidth >= currentbestDiffWith))
+ {
+ if (diffWidth == currentbestDiffWith && diffHeight
+ == currentbestDiffHeight) // Same size as previously
+ {
+ //Also check the best frame rate if the diff is the same as previouse
+ if (((diffFrameRate >= 0 &&
+ diffFrameRate <= currentbestDiffFrameRate) // Frame rate to high but better match than previouse and we have not selected IUV
+ ||
+ (currentbestDiffFrameRate < 0 &&
+ diffFrameRate >= currentbestDiffFrameRate)) // Current frame rate is lower than requested. This is better.
+ )
+ {
+ if ((currentbestDiffFrameRate == diffFrameRate) // Same frame rate as previous or frame rate allready good enough
+ || (currentbestDiffFrameRate >= 0))
+ {
+ if (bestRawType != requested.rawType
+ && requested.rawType != kVideoUnknown
+ && (capability.rawType == requested.rawType
+ || capability.rawType == kVideoI420
+ || capability.rawType == kVideoYUY2
+ || capability.rawType == kVideoYV12))
+ {
+ bestCodecType = capability.codecType;
+ bestRawType = capability.rawType;
+ bestformatIndex = tmp;
+ }
+ // If width height and frame rate is full filled we can use the camera for encoding if it is supported.
+ if (capability.height == requested.height
+ && capability.width == requested.width
+ && capability.maxFPS >= requested.maxFPS)
+ {
+ if (capability.codecType == requested.codecType
+ && bestCodecType != requested.codecType)
+ {
+ bestCodecType = capability.codecType;
+ bestformatIndex = tmp;
+ }
+ }
+ }
+ else // Better frame rate
+ {
+ if (requested.codecType == capability.codecType)
+ {
+
+ bestWidth = capability.width;
+ bestHeight = capability.height;
+ bestFrameRate = capability.maxFPS;
+ bestCodecType = capability.codecType;
+ bestRawType = capability.rawType;
+ bestformatIndex = tmp;
+ }
+ }
+ }
+ }
+ else // Better width than previously
+ {
+ if (requested.codecType == capability.codecType)
+ {
+ bestWidth = capability.width;
+ bestHeight = capability.height;
+ bestFrameRate = capability.maxFPS;
+ bestCodecType = capability.codecType;
+ bestRawType = capability.rawType;
+ bestformatIndex = tmp;
+ }
+ }
+ }// else width no good
+ }
+ else // Better height
+ {
+ if (requested.codecType == capability.codecType)
+ {
+ bestWidth = capability.width;
+ bestHeight = capability.height;
+ bestFrameRate = capability.maxFPS;
+ bestCodecType = capability.codecType;
+ bestRawType = capability.rawType;
+ bestformatIndex = tmp;
+ }
+ }
+ }// else height not good
+ }//end for
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
+ "Best camera format: Width %d, Height %d, Frame rate %d, Color format %d",
+ bestWidth, bestHeight, bestFrameRate, bestRawType);
+
+ // Copy the capability
+ MapItem* item = _captureCapabilities.Find(bestformatIndex);
+ if (!item)
+ return -1;
+ VideoCaptureCapability* capPointer =
+ static_cast<VideoCaptureCapability*> (item->GetItem());
+ if (!capPointer)
+ return -1;
+
+ resulting = *capPointer;
+
+ return bestformatIndex;
+}
+
+/* Returns the expected Capture delay*/
+WebRtc_Word32 DeviceInfoImpl::GetExpectedCaptureDelay(
+ const DelayValues delayValues[],
+ const WebRtc_UWord32 sizeOfDelayValues,
+ const WebRtc_UWord8* productId,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ WebRtc_Word32 bestDelay = kDefaultCaptureDelay;
+
+ for (WebRtc_UWord32 device = 0; device < sizeOfDelayValues; ++device)
+ {
+ if (delayValues[device].productId && strncmp((char*) productId,
+ (char*) delayValues[device].productId,
+ kVideoCaptureProductIdLength) == 0)
+ {
+ // We have found the camera
+
+ WebRtc_Word32 bestWidth = 0;
+ WebRtc_Word32 bestHeight = 0;
+
+ //Loop through all tested sizes and find one that seems fitting
+ for (WebRtc_UWord32 delayIndex = 0; delayIndex < NoOfDelayValues; ++delayIndex)
+ {
+ const DelayValue& currentValue = delayValues[device].delayValues[delayIndex];
+
+ const WebRtc_Word32 diffWidth = currentValue.width - width;
+ const WebRtc_Word32 diffHeight = currentValue.height - height;
+
+ const WebRtc_Word32 currentbestDiffWith = bestWidth - width;
+ const WebRtc_Word32 currentbestDiffHeight = bestHeight - height;
+
+ if ((diffHeight >= 0 && diffHeight <= abs(currentbestDiffHeight)) // Height better or equal than previous.
+ || (currentbestDiffHeight < 0 && diffHeight >= currentbestDiffHeight))
+ {
+
+ if (diffHeight == currentbestDiffHeight) // Found best height. Care about the width)
+ {
+ if ((diffWidth >= 0 && diffWidth <= abs(currentbestDiffWith)) // Width better or equal
+ || (currentbestDiffWith < 0 && diffWidth >= currentbestDiffWith))
+ {
+ if (diffWidth == currentbestDiffWith && diffHeight
+ == currentbestDiffHeight) // Same size as previous
+ {
+ }
+ else // Better width than previously
+ {
+ bestWidth = currentValue.width;
+ bestHeight = currentValue.height;
+ bestDelay = currentValue.delay;
+ }
+ }// else width no good
+ }
+ else // Better height
+ {
+ bestWidth = currentValue.width;
+ bestHeight = currentValue.height;
+ bestDelay = currentValue.delay;
+ }
+ }// else height not good
+ }//end for
+ break;
+ }
+ }
+ if (bestDelay > kMaxCaptureDelay)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "Expected capture delay too high. %dms, will use %d", bestDelay,
+ kMaxCaptureDelay);
+ bestDelay = kMaxCaptureDelay;
+
+ }
+
+ return bestDelay;
+
+}
+
+//Default implementation. This should be overridden by Mobile implementations.
+WebRtc_Word32 DeviceInfoImpl::GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ VideoCaptureRotation& orientation)
+{
+ orientation = kCameraRotate0;
+ return -1;
+}
+} //namespace videocapturemodule
+} // namespace webrtc
+
+
diff --git a/trunk/src/modules/video_capture/main/source/device_info_impl.h b/trunk/src/modules/video_capture/main/source/device_info_impl.h
new file mode 100644
index 0000000..34ef65d
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/device_info_impl.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
+
+#include "video_capture.h"
+
+#include "map_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "video_capture_delay.h"
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+class DeviceInfoImpl: public VideoCaptureModule::DeviceInfo
+{
+public:
+ DeviceInfoImpl(const WebRtc_Word32 id);
+ virtual ~DeviceInfoImpl(void);
+ virtual WebRtc_Word32 NumberOfCapabilities(const WebRtc_UWord8* deviceUniqueIdUTF8);
+ virtual WebRtc_Word32 GetCapability(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ const WebRtc_UWord32 deviceCapabilityNumber,
+ VideoCaptureCapability& capability);
+
+ virtual WebRtc_Word32 GetBestMatchedCapability(const WebRtc_UWord8*deviceUniqueIdUTF8,
+ const VideoCaptureCapability& requested,
+ VideoCaptureCapability& resulting);
+ virtual WebRtc_Word32 GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
+ VideoCaptureRotation& orientation);
+
+protected:
+ /* Initialize this object*/
+
+ virtual WebRtc_Word32 Init()=0;
+ /*
+ * Fills the member variable _captureCapabilities with capabilities for the given device name.
+ */
+ virtual WebRtc_Word32 CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8)=0;
+
+ /* Returns the expected Capture delay*/
+ WebRtc_Word32 GetExpectedCaptureDelay(const DelayValues delayValues[],
+ const WebRtc_UWord32 sizeOfDelayValues,
+ const WebRtc_UWord8* productId,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height);
+protected:
+ // Data members
+ WebRtc_Word32 _id;
+ MapWrapper _captureCapabilities;
+ RWLockWrapper& _apiLock;
+ WebRtc_UWord8* _lastUsedDeviceName;
+ WebRtc_UWord32 _lastUsedDeviceNameLength;
+};
+} //namespace videocapturemodule
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture.gypi b/trunk/src/modules/video_capture/main/source/video_capture.gypi
new file mode 100644
index 0000000..de696ae
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture.gypi
@@ -0,0 +1,259 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'video_capture_module',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_utility',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ '<(webrtc_root)/common_video/libyuv/include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ '<(webrtc_root)/common_video/libyuv/include',
+ ],
+ },
+ 'sources': [
+ # interfaces
+ '../interface/video_capture.h',
+ '../interface/video_capture_defines.h',
+ '../interface/video_capture_factory.h',
+ # headers
+ 'video_capture_config.h',
+ 'video_capture_delay.h',
+ 'video_capture_impl.h',
+ 'device_info_impl.h',
+
+ # DEFINE PLATFORM INDEPENDENT SOURCE FILES
+ 'video_capture_factory.cc',
+ 'video_capture_impl.cc',
+ 'device_info_impl.cc',
+ ],
+ 'conditions': [
+ ['include_internal_video_capture==0', {
+ 'sources': [
+ 'External/device_info_external.cc',
+ 'External/video_capture_external.cc',
+ ],
+ },{ # include_internal_video_capture == 1
+ 'conditions': [
+ # DEFINE PLATFORM SPECIFIC SOURCE FILES
+ ['OS=="linux"', {
+ 'include_dirs': [
+ 'Linux',
+ ],
+ 'sources': [
+ 'Linux/device_info_linux.h',
+ 'Linux/video_capture_linux.h',
+ 'Linux/device_info_linux.cc',
+ 'Linux/video_capture_linux.cc',
+ ],
+ }], # linux
+ ['OS=="mac"', {
+ 'sources': [
+ 'Mac/QTKit/video_capture_recursive_lock.h',
+ 'Mac/QTKit/video_capture_qtkit.h',
+ 'Mac/QTKit/video_capture_qtkit_info.h',
+ 'Mac/QTKit/video_capture_qtkit_info_objc.h',
+ 'Mac/QTKit/video_capture_qtkit_objc.h',
+ 'Mac/QTKit/video_capture_qtkit_utility.h',
+ 'Mac/video_capture_mac.mm',
+ 'Mac/QTKit/video_capture_qtkit.mm',
+ 'Mac/QTKit/video_capture_qtkit_objc.mm',
+ 'Mac/QTKit/video_capture_recursive_lock.mm',
+ 'Mac/QTKit/video_capture_qtkit_info.mm',
+ 'Mac/QTKit/video_capture_qtkit_info_objc.mm',
+ ],
+ 'include_dirs': [
+ 'Mac',
+ ],
+ 'link_settings': {
+ 'xcode_settings': {
+ 'OTHER_LDFLAGS': [
+ '-framework QTKit',
+ ],
+ },
+ },
+ }], # mac
+ ['OS=="win"', {
+ 'include_dirs': [
+ 'Windows',
+ '<(direct_show_base_classes)',
+ ],
+ 'defines!': [
+ 'NOMINMAX',
+ ],
+ 'sources': [
+ 'Windows/help_functions_windows.h',
+ 'Windows/sink_filter_windows.h',
+ 'Windows/video_capture_windows.h',
+ 'Windows/device_info_windows.h',
+ 'Windows/capture_delay_values_windows.h',
+ 'Windows/help_functions_windows.cc',
+ 'Windows/sink_filter_windows.cc',
+ 'Windows/video_capture_windows.cc',
+ 'Windows/device_info_windows.cc',
+ 'Windows/video_capture_factory_windows.cc',
+ '<(direct_show_base_classes)amextra.cpp',
+ '<(direct_show_base_classes)amextra.h',
+ '<(direct_show_base_classes)amfilter.cpp',
+ '<(direct_show_base_classes)amfilter.h',
+ '<(direct_show_base_classes)amvideo.cpp',
+ '<(direct_show_base_classes)arithutil.cpp',
+ '<(direct_show_base_classes)cache.h',
+ '<(direct_show_base_classes)checkbmi.h',
+ '<(direct_show_base_classes)combase.cpp',
+ '<(direct_show_base_classes)combase.h',
+ '<(direct_show_base_classes)cprop.cpp',
+ '<(direct_show_base_classes)cprop.h',
+ '<(direct_show_base_classes)ctlutil.cpp',
+ '<(direct_show_base_classes)ctlutil.h',
+ '<(direct_show_base_classes)ddmm.cpp',
+ '<(direct_show_base_classes)ddmm.h',
+ '<(direct_show_base_classes)dllentry.cpp',
+ '<(direct_show_base_classes)dllsetup.cpp',
+ '<(direct_show_base_classes)dllsetup.h',
+ '<(direct_show_base_classes)dxmperf.h',
+ '<(direct_show_base_classes)fourcc.h',
+ '<(direct_show_base_classes)measure.h',
+ '<(direct_show_base_classes)msgthrd.h',
+ '<(direct_show_base_classes)mtype.cpp',
+ '<(direct_show_base_classes)mtype.h',
+ '<(direct_show_base_classes)outputq.cpp',
+ '<(direct_show_base_classes)outputq.h',
+ '<(direct_show_base_classes)perflog.cpp',
+ '<(direct_show_base_classes)perflog.h',
+ '<(direct_show_base_classes)perfstruct.h',
+ '<(direct_show_base_classes)pstream.cpp',
+ '<(direct_show_base_classes)pstream.h',
+ '<(direct_show_base_classes)pullpin.cpp',
+ '<(direct_show_base_classes)pullpin.h',
+ '<(direct_show_base_classes)refclock.cpp',
+ '<(direct_show_base_classes)refclock.h',
+ '<(direct_show_base_classes)reftime.h',
+ '<(direct_show_base_classes)renbase.cpp',
+ '<(direct_show_base_classes)renbase.h',
+ '<(direct_show_base_classes)schedule.cpp',
+ '<(direct_show_base_classes)schedule.h',
+ '<(direct_show_base_classes)seekpt.cpp',
+ '<(direct_show_base_classes)seekpt.h',
+ '<(direct_show_base_classes)source.cpp',
+ '<(direct_show_base_classes)source.h',
+ '<(direct_show_base_classes)streams.h',
+ '<(direct_show_base_classes)strmctl.cpp',
+ '<(direct_show_base_classes)strmctl.h',
+ '<(direct_show_base_classes)sysclock.cpp',
+ '<(direct_show_base_classes)sysclock.h',
+ '<(direct_show_base_classes)transfrm.cpp',
+ '<(direct_show_base_classes)transfrm.h',
+ '<(direct_show_base_classes)transip.cpp',
+ '<(direct_show_base_classes)transip.h',
+ '<(direct_show_base_classes)videoctl.cpp',
+ '<(direct_show_base_classes)videoctl.h',
+ '<(direct_show_base_classes)vtrans.cpp',
+ '<(direct_show_base_classes)vtrans.h',
+ '<(direct_show_base_classes)winctrl.cpp',
+ '<(direct_show_base_classes)winctrl.h',
+ '<(direct_show_base_classes)winutil.cpp',
+ '<(direct_show_base_classes)winutil.h',
+ '<(direct_show_base_classes)wxdebug.cpp',
+ '<(direct_show_base_classes)wxdebug.h',
+ '<(direct_show_base_classes)wxlist.cpp',
+ '<(direct_show_base_classes)wxlist.h',
+ '<(direct_show_base_classes)wxutil.cpp',
+ '<(direct_show_base_classes)wxutil.h',
+ ],
+ 'msvs_settings': {
+ 'VCLibrarianTool': {
+ 'AdditionalDependencies': 'Strmiids.lib',
+ },
+ },
+ }], # win
+ ['OS=="android"', {
+ 'include_dirs': [
+ 'Android',
+ ],
+ 'sources': [
+ 'Android/device_info_android.cc',
+ 'Android/device_info_android.h',
+ 'Android/video_capture_android.cc',
+ 'Android/video_capture_android.h',
+ ],
+ }], # android
+ ], # conditions
+ }], # include_internal_video_capture
+ ], # conditions
+ },
+ ],
+ # Exclude the test targets when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'video_capture_module_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_capture_module',
+ 'webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'include_dirs': [
+ '../interface',
+ ],
+ 'sources': [
+ '../test/video_capture_unittest.cc',
+ ],
+ 'conditions': [
+ # DEFINE PLATFORM SPECIFIC INCLUDE AND CFLAGS
+ ['OS=="mac" or OS=="linux"', {
+ 'cflags': [
+ '-Wno-write-strings',
+ ],
+ 'ldflags': [
+ '-lpthread -lm',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'libraries': [
+ '-lrt',
+ '-lXext',
+ '-lX11',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ # TODO(andrew): CoreAudio and AudioToolbox shouldn't be needed.
+ 'OTHER_LDFLAGS': [
+ '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL -framework CoreVideo -framework CoreAudio -framework AudioToolbox',
+ ],
+ },
+ }],
+ ] # conditions
+ },
+ ],
+ }],
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_config.h b/trunk/src/modules/video_capture/main/source/video_capture_config.h
new file mode 100644
index 0000000..ab4010c
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture_config.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+enum {kDefaultWidth = 640}; // Start width
+enum {kDefaultHeight = 480}; // Start heigt
+enum {kDefaultFrameRate = 30}; // Start frame rate
+
+enum {kMaxFrameRate =60}; // Max allowed frame rate of the start image
+
+enum {kDefaultCaptureDelay = 120};
+enum {kMaxCaptureDelay = 270}; // Max capture delay allowed in the precompiled capture delay values.
+
+enum {kProcessInterval = 300};
+enum {kFrameRateCallbackInterval = 1000};
+enum {kFrameRateCountHistorySize = 90};
+enum {kFrameRateHistoryWindowMs = 2000};
+} // namespace videocapturemodule
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_delay.h b/trunk/src/modules/video_capture/main/source/video_capture_delay.h
new file mode 100644
index 0000000..9f5b76e
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture_delay.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+
+struct DelayValue
+{
+ WebRtc_Word32 width;
+ WebRtc_Word32 height;
+ WebRtc_Word32 delay;
+};
+
+enum { NoOfDelayValues = 40 };
+struct DelayValues
+{
+ char * deviceName;
+ char* productId;
+ DelayValue delayValues[NoOfDelayValues];
+};
+
+} //namespace videocapturemodule
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_factory.cc b/trunk/src/modules/video_capture/main/source/video_capture_factory.cc
new file mode 100644
index 0000000..ac0e641
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture_factory.cc
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_capture_factory.h"
+#include "video_capture_impl.h"
+#ifdef WEBRTC_ANDROID
+#include "video_capture_android.h"
+#endif
+
+namespace webrtc
+{
+
+VideoCaptureModule* VideoCaptureFactory::Create(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8) {
+ return videocapturemodule::VideoCaptureImpl::Create(id, deviceUniqueIdUTF8);
+}
+
+VideoCaptureModule* VideoCaptureFactory::Create(const WebRtc_Word32 id,
+ VideoCaptureExternal*& externalCapture) {
+ return videocapturemodule::VideoCaptureImpl::Create(id, externalCapture);
+}
+
+VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo(
+ const WebRtc_Word32 id) {
+ return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo(id);
+}
+
+#ifdef WEBRTC_ANDROID
+WebRtc_Word32 VideoCaptureFactory::SetAndroidObjects(void* javaVM,
+ void* javaContext) {
+ return videocapturemodule::VideoCaptureAndroid::SetAndroidObjects(javaVM,
+ javaContext);
+}
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_impl.cc b/trunk/src/modules/video_capture/main/source/video_capture_impl.cc
new file mode 100644
index 0000000..f976d10
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture_impl.cc
@@ -0,0 +1,501 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_capture_impl.h"
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "critical_section_wrapper.h"
+#include "module_common_types.h"
+#include "ref_count.h"
+#include "tick_util.h"
+#include "trace.h"
+#include "video_capture_config.h"
+
+#ifdef WEBRTC_ANDROID
+#include "video_capture_android.h" // Need inclusion here to set Java environment.
+#endif
+
+namespace webrtc
+{
+namespace videocapturemodule
+{
+VideoCaptureModule* VideoCaptureImpl::Create(
+ const WebRtc_Word32 id,
+ VideoCaptureExternal*& externalCapture)
+{
+ RefCountImpl<VideoCaptureImpl>* implementation =
+ new RefCountImpl<VideoCaptureImpl>(id);
+ externalCapture = implementation;
+ return implementation;
+}
+
+const WebRtc_UWord8* VideoCaptureImpl::CurrentDeviceName() const
+{
+ return _deviceUniqueId;
+}
+
+WebRtc_Word32 VideoCaptureImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+// returns the number of milliseconds until the module want a worker thread to call Process
+WebRtc_Word32 VideoCaptureImpl::TimeUntilNextProcess()
+{
+ TickTime timeNow = TickTime::Now();
+
+ WebRtc_Word32 timeToNormalProcess = kProcessInterval
+ - (WebRtc_Word32)((TickTime::Now() - _lastProcessTime).Milliseconds());
+ WebRtc_Word32 timeToStartImage = timeToNormalProcess;
+ if (_startImageFrameIntervall)
+ {
+ timeToStartImage = _startImageFrameIntervall
+ - (WebRtc_Word32)((timeNow - _lastSentStartImageTime).Milliseconds());
+ if (timeToStartImage < 0)
+ {
+ timeToStartImage = 0;
+ }
+ }
+ return (timeToStartImage < timeToNormalProcess)
+ ? timeToStartImage : timeToNormalProcess;
+}
+
+// Process any pending tasks such as timeouts
+WebRtc_Word32 VideoCaptureImpl::Process()
+{
+ CriticalSectionScoped cs(_callBackCs);
+
+ const TickTime now = TickTime::Now();
+ _lastProcessTime = TickTime::Now();
+
+ // Handle No picture alarm
+
+ if (_lastProcessFrameCount.Ticks() == _incomingFrameTimes[0].Ticks() &&
+ _captureAlarm != Raised)
+ {
+ if (_noPictureAlarmCallBack && _captureCallBack)
+ {
+ _captureAlarm = Raised;
+ _captureCallBack->OnNoPictureAlarm(_id, _captureAlarm);
+ }
+ }
+ else if (_lastProcessFrameCount.Ticks() != _incomingFrameTimes[0].Ticks() &&
+ _captureAlarm != Cleared)
+ {
+ if (_noPictureAlarmCallBack && _captureCallBack)
+ {
+ _captureAlarm = Cleared;
+ _captureCallBack->OnNoPictureAlarm(_id, _captureAlarm);
+
+ }
+ }
+
+ // Handle frame rate callback
+ if ((now - _lastFrameRateCallbackTime).Milliseconds()
+ > kFrameRateCallbackInterval)
+ {
+ if (_frameRateCallBack && _captureCallBack)
+ {
+ const WebRtc_UWord32 frameRate = CalculateFrameRate(now);
+ _captureCallBack->OnCaptureFrameRate(_id, frameRate);
+ }
+ _lastFrameRateCallbackTime = now; // Can be set by EnableFrameRateCallback
+
+ }
+
+ _lastProcessFrameCount = _incomingFrameTimes[0];
+
+ // Handle start image frame rates.
+ if (_startImageFrameIntervall
+ && (now - _lastSentStartImageTime).Milliseconds() >= _startImageFrameIntervall)
+ {
+ _lastSentStartImageTime = now;
+ if (_dataCallBack)
+ {
+ _captureFrame.CopyFrame(_startImage);
+ _captureFrame.SetRenderTime(TickTime::MillisecondTimestamp());
+ _dataCallBack->OnIncomingCapturedFrame(_id, _captureFrame,
+ kVideoCodecUnknown);
+ }
+ }
+ return 0;
+}
+
+VideoCaptureImpl::VideoCaptureImpl(const WebRtc_Word32 id)
+ : _id(id), _deviceUniqueId(NULL), _apiCs(*CriticalSectionWrapper::CreateCriticalSection()),
+ _captureDelay(0), _requestedCapability(),
+ _callBackCs(*CriticalSectionWrapper::CreateCriticalSection()),
+ _lastProcessTime(TickTime::Now()),
+ _lastFrameRateCallbackTime(TickTime::Now()), _frameRateCallBack(false),
+ _noPictureAlarmCallBack(false), _captureAlarm(Cleared), _setCaptureDelay(0),
+ _dataCallBack(NULL), _captureCallBack(NULL),
+ _startImage(), _startImageFrameIntervall(0),
+ _lastSentStartImageTime(TickTime::Now()),
+ _lastProcessFrameCount(TickTime::Now()), _rotateFrame(kRotateNone),
+ last_capture_time_(TickTime::MillisecondTimestamp())
+
+{
+ _requestedCapability.width = kDefaultWidth;
+ _requestedCapability.height = kDefaultHeight;
+ _requestedCapability.maxFPS = 30;
+ _requestedCapability.rawType = kVideoI420;
+ _requestedCapability.codecType = kVideoCodecUnknown;
+ memset(_incomingFrameTimes, 0, sizeof(_incomingFrameTimes));
+}
+
+VideoCaptureImpl::~VideoCaptureImpl()
+{
+ DeRegisterCaptureDataCallback();
+ DeRegisterCaptureCallback();
+ delete &_callBackCs;
+ delete &_apiCs;
+
+ if (_deviceUniqueId)
+ delete[] _deviceUniqueId;
+}
+
+WebRtc_Word32 VideoCaptureImpl::RegisterCaptureDataCallback(
+ VideoCaptureDataCallback& dataCallBack)
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _dataCallBack = &dataCallBack;
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::DeRegisterCaptureDataCallback()
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _dataCallBack = NULL;
+ return 0;
+}
+WebRtc_Word32 VideoCaptureImpl::RegisterCaptureCallback(VideoCaptureFeedBack& callBack)
+{
+
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _captureCallBack = &callBack;
+ return 0;
+}
+WebRtc_Word32 VideoCaptureImpl::DeRegisterCaptureCallback()
+{
+
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _captureCallBack = NULL;
+ return 0;
+
+}
+WebRtc_Word32 VideoCaptureImpl::SetCaptureDelay(WebRtc_Word32 delayMS)
+{
+ CriticalSectionScoped cs(_apiCs);
+ _captureDelay = delayMS;
+ return 0;
+}
+WebRtc_Word32 VideoCaptureImpl::CaptureDelay()
+{
+ CriticalSectionScoped cs(_apiCs);
+ return _setCaptureDelay;
+}
+
+WebRtc_Word32 VideoCaptureImpl::DeliverCapturedFrame(VideoFrame& captureFrame,
+ WebRtc_Word32 width, WebRtc_Word32 height, WebRtc_Word64 capture_time,
+ VideoCodecType codec_type) {
+ UpdateFrameCount();// frame count used for local frame rate callback.
+ _startImageFrameIntervall = 0; // prevent the start image to be displayed.
+
+ const bool callOnCaptureDelayChanged = _setCaptureDelay != _captureDelay;
+ // Capture delay changed
+ if (_setCaptureDelay != _captureDelay) {
+ _setCaptureDelay = _captureDelay;
+ }
+
+ // Set the capture time
+ if (capture_time != 0) {
+ captureFrame.SetRenderTime(capture_time);
+ }
+ else {
+ captureFrame.SetRenderTime(TickTime::MillisecondTimestamp());
+ }
+
+ if (captureFrame.RenderTimeMs() == last_capture_time_) {
+ // We don't allow the same capture time for two frames, drop this one.
+ return -1;
+ }
+ last_capture_time_ = captureFrame.RenderTimeMs();
+
+ captureFrame.SetHeight(height);
+ captureFrame.SetWidth(width);
+
+ if (_dataCallBack) {
+ if (callOnCaptureDelayChanged) {
+ _dataCallBack->OnCaptureDelayChanged(_id, _captureDelay);
+ }
+ _dataCallBack->OnIncomingCapturedFrame(_id, captureFrame, codec_type);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::IncomingFrame(
+ WebRtc_UWord8* videoFrame,
+ WebRtc_Word32 videoFrameLength,
+ const VideoCaptureCapability& frameInfo,
+ WebRtc_Word64 captureTime/*=0*/)
+{
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideoCapture, _id,
+ "IncomingFrame width %d, height %d", (int) frameInfo.width,
+ (int) frameInfo.height);
+
+ TickTime startProcessTime = TickTime::Now();
+
+ CriticalSectionScoped cs(_callBackCs);
+
+ const WebRtc_Word32 width = frameInfo.width;
+ const WebRtc_Word32 height = frameInfo.height;
+
+ if (frameInfo.codecType == kVideoCodecUnknown)
+ {
+ // Not encoded, convert to I420.
+ const VideoType commonVideoType =
+ RawVideoTypeToCommonVideoVideoType(frameInfo.rawType);
+
+ if (frameInfo.rawType != kVideoMJPEG &&
+ CalcBufferSize(commonVideoType, width, height) != videoFrameLength)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Wrong incoming frame length.");
+ return -1;
+ }
+
+ // Allocate I420 buffer.
+ int requiredLength = CalcBufferSize(kI420, width, height);
+ _captureFrame.VerifyAndAllocate(requiredLength);
+ if (!_captureFrame.Buffer())
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to allocate frame buffer.");
+ return -1;
+ }
+
+ memset(_captureFrame.Buffer(), 0, _captureFrame.Size());
+ // Keeping stride = width for I420 destination.
+ int dstStride = width;
+ const int conversionResult = ConvertToI420(commonVideoType,
+ videoFrame,
+ 0, 0, // No cropping
+ width, height,
+ videoFrameLength,
+ width, height, dstStride,
+ _rotateFrame,
+ _captureFrame.Buffer());
+ if (conversionResult < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to convert capture frame from type %d to I420",
+ frameInfo.rawType);
+ return -1;
+ }
+ _captureFrame.SetLength(requiredLength);
+ }
+ else // Encoded format
+ {
+ if (_captureFrame.CopyFrame(videoFrameLength, videoFrame) != 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to copy captured frame of length %d", (int) videoFrameLength);
+ }
+ }
+
+ DeliverCapturedFrame(_captureFrame, width, height, captureTime, frameInfo.codecType);
+
+
+ const WebRtc_UWord32 processTime =
+ (WebRtc_UWord32)(TickTime::Now() - startProcessTime).Milliseconds();
+ if (processTime > 10) // If the process time is too long MJPG will not work well.
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
+ "Too long processing time of Incoming frame: %ums",
+ (unsigned int) processTime);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::IncomingFrameI420(
+ const VideoFrameI420& video_frame, WebRtc_Word64 captureTime) {
+
+ CriticalSectionScoped cs(_callBackCs);
+
+ // Allocate I420 buffer
+ int frame_size = CalcBufferSize(kI420,
+ video_frame.width,
+ video_frame.height);
+ _captureFrame.VerifyAndAllocate(frame_size);
+ if (!_captureFrame.Buffer()) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "Failed to allocate frame buffer.");
+ return -1;
+ }
+
+ // Copy planes to the _captureFrame
+ int y_width = video_frame.width;
+ int uv_width = video_frame.width / 2;
+ int y_rows = video_frame.height;
+ int uv_rows = video_frame.height / 2; // I420
+ unsigned char* current_pointer = _captureFrame.Buffer();
+ unsigned char* y_plane = video_frame.y_plane;
+ unsigned char* u_plane = video_frame.u_plane;
+ unsigned char* v_plane = video_frame.v_plane;
+ // Copy Y
+ for (int i = 0; i < y_rows; ++i) {
+ memcpy(current_pointer, y_plane, y_width);
+ current_pointer += video_frame.y_pitch;
+ y_plane += video_frame.y_pitch;
+ }
+ // Copy U
+ for (int i = 0; i < uv_rows; ++i) {
+ memcpy(current_pointer, u_plane, uv_width);
+ current_pointer += video_frame.u_pitch;
+ u_plane += video_frame.u_pitch;
+ }
+ // Copy V
+ for (int i = 0; i < uv_rows; ++i) {
+ memcpy(current_pointer, v_plane, uv_width);
+ current_pointer += video_frame.v_pitch;
+ v_plane += video_frame.v_pitch;
+ }
+ _captureFrame.SetLength(frame_size);
+
+ DeliverCapturedFrame(_captureFrame,
+ video_frame.width,
+ video_frame.height,
+ captureTime,
+ kVideoCodecUnknown);
+
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::SetCaptureRotation(VideoCaptureRotation rotation)
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ switch (rotation)
+ {
+ case kCameraRotate0:
+ _rotateFrame = kRotateNone;
+ break;
+ case kCameraRotate90:
+ _rotateFrame = kRotate90;
+ break;
+ case kCameraRotate180:
+ _rotateFrame = kRotate180;
+ break;
+ case kCameraRotate270:
+ _rotateFrame = kRotate270;
+ break;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::StartSendImage(const VideoFrame& videoFrame,
+ WebRtc_Word32 frameRate)
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ if (frameRate < 1 || frameRate > kMaxFrameRate)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+ "StartSendImage Invalid parameter. frameRate %d", (int) frameRate);
+ return -1;;
+ }
+ _startImage.CopyFrame(videoFrame);
+ _startImageFrameIntervall = 1000 / frameRate;
+ _lastSentStartImageTime = TickTime::Now();
+ return 0;
+
+}
+WebRtc_Word32 VideoCaptureImpl::StopSendImage()
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _startImageFrameIntervall = 0;
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::EnableFrameRateCallback(const bool enable)
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _frameRateCallBack = enable;
+ if (enable)
+ {
+ _lastFrameRateCallbackTime = TickTime::Now();
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoCaptureImpl::EnableNoPictureAlarm(const bool enable)
+{
+ CriticalSectionScoped cs(_apiCs);
+ CriticalSectionScoped cs2(_callBackCs);
+ _noPictureAlarmCallBack = enable;
+ return 0;
+}
+
+void VideoCaptureImpl::UpdateFrameCount()
+{
+ if (_incomingFrameTimes[0].MicrosecondTimestamp() == 0)
+ {
+ // first no shift
+ }
+ else
+ {
+ // shift
+ for (int i = (kFrameRateCountHistorySize - 2); i >= 0; i--)
+ {
+ _incomingFrameTimes[i + 1] = _incomingFrameTimes[i];
+ }
+ }
+ _incomingFrameTimes[0] = TickTime::Now();
+}
+
+WebRtc_UWord32 VideoCaptureImpl::CalculateFrameRate(const TickTime& now)
+{
+ WebRtc_Word32 num = 0;
+ WebRtc_Word32 nrOfFrames = 0;
+ for (num = 1; num < (kFrameRateCountHistorySize - 1); num++)
+ {
+ if (_incomingFrameTimes[num].Ticks() <= 0
+ || (now - _incomingFrameTimes[num]).Milliseconds() > kFrameRateHistoryWindowMs) // don't use data older than 2sec
+ {
+ break;
+ }
+ else
+ {
+ nrOfFrames++;
+ }
+ }
+ if (num > 1)
+ {
+ WebRtc_Word64 diff = (now - _incomingFrameTimes[num - 1]).Milliseconds();
+ if (diff > 0)
+ {
+ return WebRtc_UWord32((nrOfFrames * 1000.0f / diff) + 0.5f);
+ }
+ }
+
+ return nrOfFrames;
+}
+} // namespace videocapturemodule
+} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_impl.h b/trunk/src/modules/video_capture/main/source/video_capture_impl.h
new file mode 100644
index 0000000..eae3f39
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/source/video_capture_impl.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
+
+/*
+ * video_capture_impl.h
+ */
+
+#include "video_capture.h"
+#include "video_capture_config.h"
+#include "tick_util.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+
+namespace videocapturemodule {
+// Class definitions
+class VideoCaptureImpl: public VideoCaptureModule, public VideoCaptureExternal
+{
+public:
+
+ /*
+ * Create a video capture module object
+ *
+ * id - unique identifier of this video capture module object
+ * deviceUniqueIdUTF8 - name of the device. Available names can be found by using GetDeviceName
+ */
+ static VideoCaptureModule* Create(const WebRtc_Word32 id,
+ const WebRtc_UWord8* deviceUniqueIdUTF8);
+
+ /*
+ * Create a video capture module object used for external capture.
+ *
+ * id - unique identifier of this video capture module object
+ * externalCapture - [out] interface to call when a new frame is captured.
+ */
+ static VideoCaptureModule* Create(const WebRtc_Word32 id,
+ VideoCaptureExternal*& externalCapture);
+
+ static DeviceInfo* CreateDeviceInfo(const WebRtc_Word32 id);
+
+ // Implements Module declared functions.
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ //Call backs
+ virtual WebRtc_Word32 RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback);
+ virtual WebRtc_Word32 DeRegisterCaptureDataCallback();
+ virtual WebRtc_Word32 RegisterCaptureCallback(VideoCaptureFeedBack& callBack);
+ virtual WebRtc_Word32 DeRegisterCaptureCallback();
+
+ virtual WebRtc_Word32 StartSendImage(const VideoFrame& videoFrame,
+ WebRtc_Word32 frameRate = 1);
+ virtual WebRtc_Word32 StopSendImage();
+
+ virtual WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delayMS);
+ virtual WebRtc_Word32 CaptureDelay();
+ virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation);
+
+ virtual WebRtc_Word32 EnableFrameRateCallback(const bool enable);
+ virtual WebRtc_Word32 EnableNoPictureAlarm(const bool enable);
+
+ virtual const WebRtc_UWord8* CurrentDeviceName() const;
+
+ // Module handling
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ // Implement VideoCaptureExternal
+ virtual WebRtc_Word32 IncomingFrame(WebRtc_UWord8* videoFrame,
+ WebRtc_Word32 videoFrameLength,
+ const VideoCaptureCapability& frameInfo,
+ WebRtc_Word64 captureTime = 0);
+ virtual WebRtc_Word32 IncomingFrameI420(
+ const VideoFrameI420& video_frame,
+ WebRtc_Word64 captureTime = 0);
+
+ // Platform dependent
+ virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability)
+ {
+ _requestedCapability = capability;
+ return -1;
+ }
+ virtual WebRtc_Word32 StopCapture() { return -1; }
+ virtual bool CaptureStarted() {return false; }
+ virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& /*settings*/)
+ { return -1; }
+ VideoCaptureEncodeInterface* GetEncodeInterface(const VideoCodec& /*codec*/)
+ { return NULL; }
+
+protected:
+ VideoCaptureImpl(const WebRtc_Word32 id);
+ virtual ~VideoCaptureImpl();
+ WebRtc_Word32 DeliverCapturedFrame(
+ VideoFrame& captureFrame, WebRtc_Word32 width, WebRtc_Word32 height,
+ WebRtc_Word64 capture_time, VideoCodecType codec_type);
+
+ WebRtc_Word32 _id; // Module ID
+ WebRtc_UWord8* _deviceUniqueId; // current Device unique name;
+ CriticalSectionWrapper& _apiCs;
+ WebRtc_Word32 _captureDelay; // Current capture delay. May be changed of platform dependent parts.
+ VideoCaptureCapability _requestedCapability; // Should be set by platform dependent code in StartCapture.
+private:
+ void UpdateFrameCount();
+ WebRtc_UWord32 CalculateFrameRate(const TickTime& now);
+
+ CriticalSectionWrapper& _callBackCs;
+
+ TickTime _lastProcessTime; // last time the module process function was called.
+ TickTime _lastFrameRateCallbackTime; // last time the frame rate callback function was called.
+ bool _frameRateCallBack; // true if EnableFrameRateCallback
+ bool _noPictureAlarmCallBack; //true if EnableNoPictureAlarm
+ VideoCaptureAlarm _captureAlarm; // current value of the noPictureAlarm
+
+ WebRtc_Word32 _setCaptureDelay; // The currently used capture delay
+ VideoCaptureDataCallback* _dataCallBack;
+ VideoCaptureFeedBack* _captureCallBack;
+
+ VideoFrame _startImage;
+ WebRtc_Word32 _startImageFrameIntervall;
+ TickTime _lastSentStartImageTime; // last time the start image was sent
+ TickTime _lastProcessFrameCount;
+ TickTime _incomingFrameTimes[kFrameRateCountHistorySize];// timestamp for local captured frames
+ VideoRotationMode _rotateFrame; //Set if the frame should be rotated by the capture module.
+
+ VideoFrame _captureFrame;
+
+ // Used to make sure incoming timestamp is increasing for every frame.
+ WebRtc_Word64 last_capture_time_;
+};
+} // namespace videocapturemodule
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
diff --git a/trunk/src/modules/video_capture/main/test/android/.classpath b/trunk/src/modules/video_capture/main/test/android/.classpath
new file mode 100644
index 0000000..841ac51
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/.classpath
@@ -0,0 +1,9 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+ <classpathentry kind="src" path="src"/>
+ <classpathentry kind="src" path="renderer"/>
+ <classpathentry kind="src" path="java"/>
+ <classpathentry kind="src" path="gen"/>
+ <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+ <classpathentry kind="output" path="bin"/>
+</classpath>
diff --git a/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml b/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml
new file mode 100644
index 0000000..1a4ec9b
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ package="org.webrtc.capturemoduleandroidtest"
+ android:versionCode="1"
+ android:versionName="1.0">
+ <application android:icon="@drawable/icon"
+ android:label="@string/app_name"
+ android:debuggable="true">
+ <activity android:label="@string/app_name"
+ android:name="VideoCaptureModuleTest"
+ android:configChanges="orientation|keyboardHidden"
+ android:launchMode="singleTask"
+ android:multiprocess="false">
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+ </application>
+
+ <uses-feature android:required="true"
+ android:name="android.hardware.camera">
+ </uses-feature>
+
+ <uses-permission android:name="android.permission.CAMERA">
+ </uses-permission>
+ <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+ <uses-sdk android:targetSdkVersion="7"
+ android:minSdkVersion="7">
+ </uses-sdk>
+</manifest>
diff --git a/trunk/src/modules/video_capture/main/test/android/default.properties b/trunk/src/modules/video_capture/main/test/android/default.properties
new file mode 100644
index 0000000..2ad44a4
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target.
+target=android-9
diff --git a/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java b/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java
new file mode 100644
index 0000000..94bf93b
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java
@@ -0,0 +1,33 @@
+/* AUTO-GENERATED FILE. DO NOT MODIFY.
+ *
+ * This class was automatically generated by the
+ * aapt tool from the resource data it found. It
+ * should not be modified by hand.
+ */
+
+package org.webrtc.capturemoduleandroidtest;
+
+public final class R {
+ public static final class attr {
+ }
+ public static final class drawable {
+ public static final int icon=0x7f020000;
+ }
+ public static final class id {
+ public static final int Button01=0x7f050000;
+ public static final int Button02=0x7f050001;
+ public static final int Button03=0x7f050002;
+ public static final int Button04=0x7f050003;
+ public static final int renderView=0x7f050004;
+ }
+ public static final class layout {
+ public static final int main=0x7f030000;
+ }
+ public static final class string {
+ public static final int app_name=0x7f040000;
+ public static final int run_button=0x7f040001;
+ public static final int run_button2=0x7f040002;
+ public static final int run_button3=0x7f040003;
+ public static final int run_button4=0x7f040004;
+ }
+}
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/Android.mk b/trunk/src/modules/video_capture/main/test/android/jni/Android.mk
new file mode 100644
index 0000000..7eba756
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/jni/Android.mk
@@ -0,0 +1,19 @@
+# Android makefile for VideoCapture Module
+
+LOCAL_PATH := $(call my-dir)
+
+WEBRTC_INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../../build/interface
+WEBRTC_LIBS_PATH := $(LOCAL_PATH)/../../../../../../../build/libraries
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := VideoCaptureModuleAndroidTestJniAPI
+LOCAL_SRC_FILES := video_capture_module_android_test_jni.cc
+LOCAL_CFLAGS := -DWEBRTC_TARGET_PC # For typedefs.h
+LOCAL_C_INCLUDES := $(WEBRTC_INTERFACES_PATH)
+LOCAL_LDLIBS := \
+ $(WEBRTC_LIBS_PATH)/testVideoCaptureAPI_android_gcc.a \
+ $(WEBRTC_LIBS_PATH)/VideoCaptureModuleTestApiLib_android_gcc.a \
+ -llog -lgcc -lGLESv2
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h b/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h
new file mode 100644
index 0000000..0320df0
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* DO NOT EDIT THIS FILE - it is machine generated */
+#include <jni.h>
+/* Header for class org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest */
+
+#ifndef _Included_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
+#define _Included_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
+#ifdef __cplusplus
+extern "C" {
+#endif
+/*
+ * Class: org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
+ * Method: RunTest
+ * Signature: (Landroid/content/Context;)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RunTest
+ (JNIEnv *, jobject, jobject);
+
+JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RenderInit
+(JNIEnv * env, jobject context,jobject surface);
+
+JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StartCapture
+(JNIEnv *, jobject);
+
+JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StopCapture
+(JNIEnv *, jobject);
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc b/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc
new file mode 100644
index 0000000..ac2f6e1
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc
@@ -0,0 +1,149 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h> // memset
+#include <android/log.h>
+
+#include "org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h"
+#include "../../../interface/video_capture_factory.h"
+#include "../../../../../video_render/main/interface/video_render.h"
+#include "../../testAPI/testPlatformDependent.h"
+#include "../../testAPI/testPlatformDependent.h"
+#ifdef RENDER_PREVIEW
+#include "../../testAPI/Renderer.h"
+#endif
+
+using namespace webrtc;
+#define WEBRTC_LOG_TAG "*WEBRTCN*" // As in WEBRTC Native...
+// ADM data struct
+typedef struct
+{
+ // Other
+ JavaVM* jvm;
+ Renderer* renderer;
+ VideoCaptureModule* _videoCapture;
+ VideoCaptureModule::DeviceInfo*_captureInfo;
+} JniData;
+
+// Global variables visible in this file
+static JniData jniData;
+
+//////////////////////////////////////////////////////////////////
+// General functions
+//////////////////////////////////////////////////////////////////
+
+/////////////////////////////////////////////
+// JNI_OnLoad
+//
+jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "JNI_OnLoad");
+ if (!vm)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad did not receive a valid VM pointer");
+ return -1;
+ }
+
+ // Get JNI
+ JNIEnv* env;
+ if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env), JNI_VERSION_1_4))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad could not get JNI env");
+ return -1;
+ }
+
+ // Init JniData data
+ memset(&jniData, 0, sizeof(jniData));
+
+ // Store the JVM
+ jniData.jvm = vm;
+
+ return JNI_VERSION_1_4;
+}
+
+/////////////////////////////////////////////
+// Run Test
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RunTest(
+ JNIEnv * env,
+ jobject context,
+ jobject surface)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Run test");
+ // Set instance independent Java objects
+ VideoCaptureModule::SetAndroidObjects(jniData.jvm, context);
+
+ // Start test
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "Create testPlatformDependent");
+ testPlatformDependent testPlatformDependent;
+ testPlatformDependent.SetRenderer(jniData.renderer);
+ testPlatformDependent.DoTest();
+
+ // Clear instance independent Java objects
+ VideoCaptureModule::SetAndroidObjects(NULL, NULL);
+
+ return 0;
+}
+
+JNIEXPORT jint JNICALL
+Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RenderInit(
+ JNIEnv * env,
+ jobject context,
+ jobject surface)
+{
+ VideoRender::SetAndroidObjects(jniData.jvm);
+#ifdef RENDER_PREVIEW
+ Renderer::SetRenderWindow(surface);
+ jniData.renderer=new Renderer(true);
+#endif
+}
+
+JNIEXPORT jint JNICALL
+Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StartCapture(
+ JNIEnv * env,
+ jobject context)
+{
+ if (!jniData._captureInfo) {
+ VideoCaptureModule::SetAndroidObjects(jniData.jvm, context);
+ jniData._captureInfo = VideoCaptureFactory::CreateDeviceInfo(5);
+ WebRtc_UWord8 id[256];
+ WebRtc_UWord8 name[256];
+ jniData._captureInfo->GetDeviceName(0, name, 256, id, 256);
+ jniData._videoCapture = VideoCaptureFactory::Create(0, id);
+ VideoCaptureCapability capability;
+
+ jniData._captureInfo->GetCapability(id, 0, capability);
+ capability.width = 176;
+ capability.height = 144;
+ capability.maxFPS = 15;
+
+ jniData._videoCapture->StartCapture(capability);
+ }
+ return 0;
+}
+
+JNIEXPORT jint JNICALL
+Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StopCapture(
+ JNIEnv * env,
+ jobject context)
+{
+ if (jniData._videoCapture) {
+ jniData._videoCapture->StopCapture();
+ delete jniData._captureInfo;
+ VideoCaptureModule::Destroy(jniData._videoCapture);
+ jniData._videoCapture = NULL;
+ jniData._captureInfo = NULL;
+ }
+ return 0;
+}
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png
new file mode 100644
index 0000000..8074c4c
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png
new file mode 100644
index 0000000..1095584
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png
new file mode 100644
index 0000000..a07c69f
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml b/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml
new file mode 100644
index 0000000..3642733
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <Button android:text="@string/run_button"
+ android:id="@+id/Button01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ <Button android:text="@string/run_button2"
+ android:id="@+id/Button02"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ <Button android:text="@string/run_button3"
+ android:id="@+id/Button03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ <Button android:text="@string/run_button4"
+ android:id="@+id/Button04"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ <LinearLayout
+ android:id="@+id/renderView"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:layout_weight="1">
+ </LinearLayout>
+</LinearLayout>
diff --git a/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml b/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml
new file mode 100644
index 0000000..bd9a752
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml
@@ -0,0 +1,11 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+
+ <string name="app_name">VideoCaptureModuleAndroidTest</string>
+<string name="run_button">Run Test</string>
+
+<string name="run_button2">Run Java test</string>
+<string name="run_button3">Start c++ Capture</string>
+<string name="run_button4">Stop c++ Capture</string>
+
+</resources>
diff --git a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java
new file mode 100644
index 0000000..7a92c48
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java
@@ -0,0 +1,61 @@
+package org.webrtc.capturemoduleandroidtest;
+
+import java.util.List;
+
+import android.content.Context;
+import android.util.Log;
+
+import org.webrtc.videoengine.CaptureCapabilityAndroid;
+import org.webrtc.videoengine.VideoCaptureAndroid;
+import org.webrtc.videoengine.VideoCaptureDeviceInfoAndroid;
+
+public class VideoCaptureJavaTest {
+ void DoTest(Context context)
+ {
+ VideoCaptureDeviceInfoAndroid videoCaptureDeviceInfo =
+ VideoCaptureDeviceInfoAndroid.CreateVideoCaptureDeviceInfoAndroid(
+ 5,context);
+ for(int i = 0; i < videoCaptureDeviceInfo.NumberOfDevices(); i++) {
+ String deviceUniqueId=videoCaptureDeviceInfo.GetDeviceUniqueName(i);
+ VideoCaptureAndroid videoCapture =
+ videoCaptureDeviceInfo.AllocateCamera(i,0,deviceUniqueId);
+
+ CaptureCapabilityAndroid capArray[] =
+ videoCaptureDeviceInfo.GetCapabilityArray(deviceUniqueId);
+ for(CaptureCapabilityAndroid cap: capArray) {
+ Log.d("*WEBRTC*", "Capability widht" + cap.width +
+ " height " +cap.height+ " frameRate " +cap.maxFPS);
+ int result=videoCapture.StartCapture(cap.width,
+ cap.height,
+ cap.maxFPS);
+ try{
+ Thread.sleep(2000);//sleep for 2000 ms
+ }
+ catch(InterruptedException ie){
+ //If this thread was interrupted by another thread
+ }
+ result+=videoCapture.StopCapture();
+ Log.d("*WEBRTC*", "Start stop result " + result);
+ }
+ VideoCaptureAndroid.DeleteVideoCaptureAndroid(videoCapture);
+ videoCapture=null;
+ }
+ Log.d("*WEBRTC*", "Test complete");
+ }
+
+ VideoCaptureDeviceInfoAndroid _videoCaptureDeviceInfo;
+ VideoCaptureAndroid _videoCapture;
+ void StartCapture(Context context) {
+ _videoCaptureDeviceInfo =
+ VideoCaptureDeviceInfoAndroid.CreateVideoCaptureDeviceInfoAndroid(
+ 5,context);
+ String deviceUniqueId=_videoCaptureDeviceInfo.GetDeviceUniqueName(0);
+ _videoCapture=_videoCaptureDeviceInfo.AllocateCamera(5,0,deviceUniqueId);
+ _videoCapture.StartCapture(176,144,15);
+ }
+ void StopCapture() {
+ _videoCapture.StopCapture();
+ VideoCaptureAndroid.DeleteVideoCaptureAndroid(_videoCapture);
+ }
+
+}
diff --git a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java
new file mode 100644
index 0000000..7191b1c
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java
@@ -0,0 +1,142 @@
+package org.webrtc.capturemoduleandroidtest;
+
+import javax.microedition.khronos.egl.EGLConfig;
+import javax.microedition.khronos.opengles.GL10;
+
+import org.webrtc.capturemoduleandroidtest.R;
+import org.webrtc.videoengine.ViERenderer;
+
+import android.app.Activity;
+import android.opengl.GLSurfaceView;
+import android.os.Bundle;
+import android.util.Log;
+import android.view.SurfaceHolder;
+import android.view.SurfaceView;
+import android.view.View;
+import android.view.View.OnClickListener;
+import android.widget.Button;
+import android.widget.LinearLayout;
+
+public class VideoCaptureModuleTest
+ extends Activity implements OnClickListener {
+ // Set to 1 if OpenGL shall be used. 0 Otherwise
+ private final int _useOpenGL=1;
+ private Thread _testThread;
+ private SurfaceView _view=null;
+ private VideoCaptureModuleTest _thisPointer;
+ private VideoCaptureJavaTest _videoCaptureJavaTest;
+ /** Called when the activity is first created. */
+ @Override
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ setContentView(R.layout.main);
+
+ final Button buttonStartCP = (Button) findViewById(R.id.Button01);
+ buttonStartCP.setOnClickListener(this);
+ final Button buttonStartJava = (Button) findViewById(R.id.Button02);
+ buttonStartJava.setOnClickListener(this);
+ final Button buttonStartCPP = (Button) findViewById(R.id.Button03);
+ buttonStartCPP.setOnClickListener(this);
+ final Button buttonStopCPP = (Button) findViewById(R.id.Button04);
+ buttonStopCPP.setOnClickListener(this);
+ }
+
+ private Runnable _testProc = new Runnable() {
+ public void run() {
+ // TODO: choose test from GUI
+ // Select test here, 0 for API test, 1-> for Func tests
+ RunTest(_view);
+ }
+ };
+
+ @Override
+ protected void onStart()
+ {
+ super.onStart();
+ }
+ @Override
+ protected void onRestart()
+ {
+ super.onRestart();
+ }
+ @Override
+ protected void onPause()
+ {
+ super.onPause();
+ }
+ @Override
+ protected void onStop()
+ {
+ super.onStop();
+ }
+
+ // Function used to call test
+ private native int RunTest(Object view);
+ private native int RenderInit(Object view);
+
+ private native int StartCapture();
+ private native int StopCapture();
+
+ static {
+ Log.d("*WEBRTC*",
+ "Loading ModuleVideoCaptureModuleAndroidTest...");
+ System.loadLibrary(
+ "ModuleVideoCaptureModuleAndroidTestJniAPI");
+ }
+
+ public void onClick(View v) {
+ //get the handle to the layout
+ LinearLayout renderLayout=(LinearLayout) findViewById(R.id.renderView);
+ switch(v.getId())
+ {
+ case R.id.Button01:
+ renderLayout.removeAllViews();
+ _view=ViERenderer.CreateLocalRenderer(this);
+ if(_useOpenGL==1)
+ {
+ _view= ViERenderer.CreateRenderer(this, true);
+ }
+ else
+ {
+ _view= new SurfaceView(this);
+ }
+ // add the surfaceview to the layout,
+ // the surfaceview will be the same size as the layout (container)
+ renderLayout.addView(_view);
+ RenderInit(_view);
+ _testThread = new Thread(_testProc);
+ _testThread.start();
+ break;
+ case R.id.Button02:
+ _view=ViERenderer.CreateLocalRenderer(this);
+ renderLayout.removeAllViews();
+ if(_videoCaptureJavaTest==null)
+ {
+ _videoCaptureJavaTest=new VideoCaptureJavaTest();
+ _videoCaptureJavaTest.StartCapture(this);
+ // add the surfaceview to the layout,
+ // the surfaceview will be the same size as the layout (container)
+ renderLayout.addView(_view);
+ }
+ else
+ {
+ _videoCaptureJavaTest.StopCapture();
+ _videoCaptureJavaTest=null;
+ }
+ break;
+
+ case R.id.Button03:
+ _view=ViERenderer.CreateLocalRenderer(this);
+ renderLayout.removeAllViews();
+ StartCapture();
+ // add the surfaceview to the layout,
+ // the surfaceview will be the same size as the layout (container)
+ renderLayout.addView(_view);
+ break;
+ case R.id.Button04:
+ renderLayout.removeAllViews();
+ StopCapture();
+ break;
+ }
+ }
+}
\ No newline at end of file
diff --git a/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc b/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc
new file mode 100644
index 0000000..ac49601
--- /dev/null
+++ b/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc
@@ -0,0 +1,455 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "gtest/gtest.h"
+#include "process_thread.h"
+#include "scoped_ptr.h"
+#include "scoped_refptr.h"
+#include "tick_util.h"
+#include "video_capture.h"
+#include "video_capture_factory.h"
+
+using webrtc::TickTime;
+using webrtc::VideoCaptureAlarm;
+using webrtc::VideoCaptureCapability;
+using webrtc::VideoCaptureDataCallback;
+using webrtc::VideoCaptureFactory;
+using webrtc::VideoCaptureFeedBack;
+using webrtc::VideoCaptureModule;
+
+#if defined(_WIN32)
+#define SLEEP(x) Sleep(x)
+#elif defined(WEBRTC_ANDROID)
+#define SLEEP(x) usleep(x*1000)
+#else
+#include <unistd.h>
+#define SLEEP(x) usleep(x * 1000)
+#endif
+
+#define WAIT_(ex, timeout, res) \
+ do { \
+ res = (ex); \
+ WebRtc_Word64 start = TickTime::MillisecondTimestamp(); \
+ while (!res && TickTime::MillisecondTimestamp() < start + timeout) { \
+ SLEEP(5); \
+ res = (ex); \
+ } \
+ } while (0);\
+
+#define EXPECT_TRUE_WAIT(ex, timeout) \
+ do { \
+ bool res; \
+ WAIT_(ex, timeout, res); \
+ if (!res) EXPECT_TRUE(ex); \
+ } while (0);
+
+
+static const int kTimeOut = 5000;
+static const int kTestHeight = 288;
+static const int kTestWidth = 352;
+static const int kTestFramerate = 30;
+
+// Compares the content of two video frames.
+static bool CompareFrames(const webrtc::VideoFrame& frame1,
+ const webrtc::VideoFrame& frame2) {
+ bool result =
+ (frame1.Length() == frame2.Length()) &&
+ (frame1.Width() == frame2.Width()) &&
+ (frame1.Height() == frame2.Height());
+
+ for (unsigned int i = 0; i < frame1.Length() && result; ++i)
+ result = (*(frame1.Buffer()+i) == *(frame2.Buffer()+i));
+ return result;
+}
+
+// Compares the content of a I420 frame in planar form and video frame.
+static bool CompareFrames(const webrtc::VideoFrameI420& frame1,
+ const webrtc::VideoFrame& frame2) {
+ if (frame1.width != frame2.Width() ||
+ frame1.height != frame2.Height()) {
+ return false;
+ }
+
+ // Compare Y
+ unsigned char* y_plane = frame1.y_plane;
+ for (unsigned int i = 0; i < frame2.Height(); ++i) {
+ for (unsigned int j = 0; j < frame2.Width(); ++j) {
+ if (*y_plane != *(frame2.Buffer()+i*frame2.Width() +j))
+ return false;
+ ++y_plane;
+ }
+ y_plane += frame1.y_pitch - frame1.width;
+ }
+
+ // Compare U
+ unsigned char* u_plane = frame1.u_plane;
+ for (unsigned int i = 0; i < frame2.Height() /2; ++i) {
+ for (unsigned int j = 0; j < frame2.Width() /2; ++j) {
+ if (*u_plane !=*(
+ frame2.Buffer()+frame2.Width() * frame2.Height() +
+ i*frame2.Width() / 2 + j)) {
+ return false;
+ }
+ ++u_plane;
+ }
+ u_plane += frame1.u_pitch - frame1.width / 2;
+ }
+
+ // Compare V
+ unsigned char* v_plane = frame1.v_plane;
+ for (unsigned int i = 0; i < frame2.Height() /2; ++i) {
+ for (unsigned int j = 0; j < frame2.Width() /2; ++j) {
+ if (*v_plane != *(
+ frame2.Buffer()+frame2.Width() * frame2.Height()* 5 / 4 +
+ i*frame2.Width() / 2 + j)) {
+ return false;
+ }
+ ++v_plane;
+ }
+ v_plane += frame1.v_pitch - frame1.width / 2;
+ }
+ return true;
+}
+
+class TestVideoCaptureCallback : public VideoCaptureDataCallback {
+ public:
+ TestVideoCaptureCallback()
+ : capture_delay(0),
+ last_render_time_ms(0),
+ incoming_frames(0),
+ timing_warnings(0) {
+ }
+
+ ~TestVideoCaptureCallback() {
+ if (timing_warnings > 0)
+ printf("No of timing warnings %d\n", timing_warnings);
+ }
+
+ virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
+ webrtc::VideoFrame& videoFrame,
+ webrtc::VideoCodecType codecType) {
+ int height = static_cast<int>(videoFrame.Height());
+ int width = static_cast<int>(videoFrame.Width());
+ EXPECT_EQ(height, capability.height);
+ EXPECT_EQ(width, capability.width);
+ // RenderTimstamp should be the time now.
+ EXPECT_TRUE(
+ videoFrame.RenderTimeMs() >= TickTime::MillisecondTimestamp()-30 &&
+ videoFrame.RenderTimeMs() <= TickTime::MillisecondTimestamp());
+
+ if ((videoFrame.RenderTimeMs() >
+ last_render_time_ms + (1000 * 1.1) / capability.maxFPS &&
+ last_render_time_ms > 0) ||
+ (videoFrame.RenderTimeMs() <
+ last_render_time_ms + (1000 * 0.9) / capability.maxFPS &&
+ last_render_time_ms > 0)) {
+ timing_warnings++;
+ }
+
+ incoming_frames++;
+ last_render_time_ms = videoFrame.RenderTimeMs();
+ last_frame.CopyFrame(videoFrame);
+ }
+
+ virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
+ const WebRtc_Word32 delay) {
+ capture_delay = delay;
+ }
+
+ VideoCaptureCapability capability;
+ int capture_delay;
+ WebRtc_Word64 last_render_time_ms;
+ int incoming_frames;
+ int timing_warnings;
+ webrtc::VideoFrame last_frame;
+};
+
+class TestVideoCaptureFeedBack : public VideoCaptureFeedBack {
+ public:
+ TestVideoCaptureFeedBack() : frame_rate(0), alarm(webrtc::Cleared) {}
+
+ virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
+ const WebRtc_UWord32 frameRate) {
+ frame_rate = frameRate;
+ }
+
+ virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
+ const VideoCaptureAlarm reported_alarm) {
+ alarm = reported_alarm;
+ }
+ unsigned int frame_rate;
+ VideoCaptureAlarm alarm;
+};
+
+class VideoCaptureTest : public testing::Test {
+ public:
+ VideoCaptureTest() : number_of_devices_(0) {}
+
+ void SetUp() {
+ device_info_.reset(VideoCaptureFactory::CreateDeviceInfo(5));
+ number_of_devices_ = device_info_->NumberOfDevices();
+ ASSERT_GT(number_of_devices_, 0u);
+ }
+
+ webrtc::scoped_refptr<VideoCaptureModule> OpenVideoCaptureDevice(
+ unsigned int device,
+ VideoCaptureDataCallback* callback) {
+ WebRtc_UWord8 device_name[256];
+ WebRtc_UWord8 unique_name[256];
+
+ EXPECT_EQ(0, device_info_->GetDeviceName(
+ device, device_name, 256, unique_name, 256));
+
+ webrtc::scoped_refptr<VideoCaptureModule> module(
+ VideoCaptureFactory::Create(device, unique_name));
+ if (module.get() == NULL)
+ return NULL;
+
+ EXPECT_FALSE(module->CaptureStarted());
+
+ EXPECT_EQ(0, module->RegisterCaptureDataCallback(*callback));
+ return module;
+ }
+
+ void StartCapture(VideoCaptureModule* capture_module,
+ VideoCaptureCapability capability) {
+ EXPECT_EQ(0, capture_module->StartCapture(capability));
+ EXPECT_TRUE(capture_module->CaptureStarted());
+
+ VideoCaptureCapability resulting_capability;
+ EXPECT_EQ(0, capture_module->CaptureSettings(resulting_capability));
+ EXPECT_EQ(capability, resulting_capability);
+ }
+
+ webrtc::scoped_ptr<VideoCaptureModule::DeviceInfo> device_info_;
+ unsigned int number_of_devices_;
+};
+
+TEST_F(VideoCaptureTest, CreateDelete) {
+ for (int i = 0; i < 5; ++i) {
+ WebRtc_Word64 start_time = TickTime::MillisecondTimestamp();
+ TestVideoCaptureCallback capture_observer;
+ webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
+ 0, &capture_observer));
+ ASSERT_TRUE(module.get() != NULL);
+
+#ifndef WEBRTC_MAC
+ device_info_->GetCapability(module->CurrentDeviceName(), 0,
+ capture_observer.capability);
+#else
+ capture_observer.capability.width = kTestWidth;
+ capture_observer.capability.height = kTestHeight;
+ capture_observer.capability.maxFPS = kTestFramerate;
+ capture_observer.capability.rawType = webrtc::kVideoUnknown;
+#endif
+
+ StartCapture(module.get(), capture_observer.capability);
+
+ // Less than 4s to start the camera.
+ EXPECT_LE(TickTime::MillisecondTimestamp() - start_time, 4000);
+
+ // Make sure 5 frames are captured.
+ EXPECT_TRUE_WAIT(capture_observer.incoming_frames >= 5, kTimeOut);
+
+ EXPECT_GT(capture_observer.capture_delay, 0);
+
+ WebRtc_Word64 stop_time = TickTime::MillisecondTimestamp();
+ EXPECT_EQ(0, module->StopCapture());
+ EXPECT_FALSE(module->CaptureStarted());
+
+ // Less than 3s to stop the camera.
+ EXPECT_LE(TickTime::MillisecondTimestamp() - stop_time, 3000);
+ }
+}
+
+TEST_F(VideoCaptureTest, Capabilities) {
+#ifdef WEBRTC_MAC
+ printf("Video capture capabilities are not supported on Mac.\n");
+ return;
+#endif
+
+ TestVideoCaptureCallback capture_observer;
+
+ webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
+ 0, &capture_observer));
+ ASSERT_TRUE(module.get() != NULL);
+
+ int number_of_capabilities = device_info_->NumberOfCapabilities(
+ module->CurrentDeviceName());
+ EXPECT_GT(number_of_capabilities, 0);
+ for (int i = 0; i < number_of_capabilities; ++i) {
+ device_info_->GetCapability(module->CurrentDeviceName(), i,
+ capture_observer.capability);
+ StartCapture(module.get(), capture_observer.capability);
+ // Make sure 5 frames are captured.
+ EXPECT_TRUE_WAIT(capture_observer.incoming_frames >= 5, kTimeOut);
+ capture_observer.incoming_frames = 0;
+ EXPECT_EQ(0, module->StopCapture());
+ }
+}
+
+TEST_F(VideoCaptureTest, TestTwoCameras) {
+ if (number_of_devices_ < 2) {
+ printf("There are not two cameras available. Aborting test. \n");
+ return;
+ }
+
+ TestVideoCaptureCallback capture_observer1;
+ webrtc::scoped_refptr<VideoCaptureModule> module1(OpenVideoCaptureDevice(
+ 0, &capture_observer1));
+ ASSERT_TRUE(module1.get() != NULL);
+
+#ifndef WEBRTC_MAC
+ device_info_->GetCapability(module1->CurrentDeviceName(), 0,
+ capture_observer1.capability);
+#else
+ capture_observer1.capability.width = kTestWidth;
+ capture_observer1.capability.height = kTestHeight;
+ capture_observer1.capability.maxFPS = kTestFramerate;
+ capture_observer1.capability.rawType = webrtc::kVideoUnknown;
+#endif
+
+ TestVideoCaptureCallback capture_observer2;
+ webrtc::scoped_refptr<VideoCaptureModule> module2(OpenVideoCaptureDevice(
+ 1, &capture_observer2));
+ ASSERT_TRUE(module1.get() != NULL);
+
+
+#ifndef WEBRTC_MAC
+ device_info_->GetCapability(module2->CurrentDeviceName(), 0,
+ capture_observer2.capability);
+#else
+ capture_observer2.capability.width = kTestWidth;
+ capture_observer2.capability.height = kTestHeight;
+ capture_observer2.capability.maxFPS = kTestFramerate;
+ capture_observer2.capability.rawType = webrtc::kVideoUnknown;
+#endif
+
+ StartCapture(module1.get(), capture_observer1.capability);
+ StartCapture(module2.get(), capture_observer2.capability);
+ EXPECT_TRUE_WAIT(capture_observer1.incoming_frames >= 5, kTimeOut);
+ EXPECT_TRUE_WAIT(capture_observer2.incoming_frames >= 5, kTimeOut);
+}
+
+// Test class for testing external capture and capture feedback information
+// such as frame rate and picture alarm.
+class VideoCaptureExternalTest : public testing::Test {
+ public:
+ void SetUp() {
+ capture_module_ = VideoCaptureFactory::Create(0, capture_input_interface_);
+ process_module_ = webrtc::ProcessThread::CreateProcessThread();
+ process_module_->Start();
+ process_module_->RegisterModule(capture_module_);
+
+ capture_callback_.capability.width = kTestWidth;
+ capture_callback_.capability.height = kTestHeight;
+ capture_callback_.capability.rawType = webrtc::kVideoYV12;
+ capture_callback_.capability.maxFPS = kTestFramerate;
+
+ test_frame_.VerifyAndAllocate(kTestWidth * kTestHeight * 3 / 2);
+ test_frame_.SetLength(kTestWidth * kTestHeight * 3 / 2);
+ test_frame_.SetHeight(kTestHeight);
+ test_frame_.SetWidth(kTestWidth);
+ SLEEP(1); // Wait 1ms so that two tests can't have the same timestamp.
+ memset(test_frame_.Buffer(), 127, test_frame_.Length());
+
+ EXPECT_EQ(0, capture_module_->RegisterCaptureDataCallback(
+ capture_callback_));
+ EXPECT_EQ(0, capture_module_->RegisterCaptureCallback(capture_feedback_));
+ EXPECT_EQ(0, capture_module_->EnableFrameRateCallback(true));
+ EXPECT_EQ(0, capture_module_->EnableNoPictureAlarm(true));
+ }
+
+ void TearDown() {
+ process_module_->Stop();
+ webrtc::ProcessThread::DestroyProcessThread(process_module_);
+ }
+
+ webrtc::VideoCaptureExternal* capture_input_interface_;
+ webrtc::scoped_refptr<VideoCaptureModule> capture_module_;
+ webrtc::ProcessThread* process_module_;
+ webrtc::VideoFrame test_frame_;
+ TestVideoCaptureCallback capture_callback_;
+ TestVideoCaptureFeedBack capture_feedback_;
+};
+
+// Test input of external video frames.
+TEST_F(VideoCaptureExternalTest , TestExternalCapture) {
+ EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
+ test_frame_.Buffer(), test_frame_.Length(), capture_callback_.capability,
+ 0));
+ EXPECT_TRUE(CompareFrames(test_frame_, capture_callback_.last_frame));
+}
+
+// Test input of planar I420 frames.
+TEST_F(VideoCaptureExternalTest , TestExternalCaptureI420) {
+ webrtc::VideoFrameI420 frame_i420;
+ frame_i420.width = kTestWidth;
+ frame_i420.height = kTestHeight;
+ frame_i420.y_plane = test_frame_.Buffer();
+ frame_i420.u_plane = frame_i420.y_plane + (kTestWidth * kTestHeight);
+ frame_i420.v_plane = frame_i420.u_plane + ((kTestWidth * kTestHeight) >> 2);
+ frame_i420.y_pitch = kTestWidth;
+ frame_i420.u_pitch = kTestWidth / 2;
+ frame_i420.v_pitch = kTestWidth / 2;
+ EXPECT_EQ(0, capture_input_interface_->IncomingFrameI420(frame_i420, 0));
+
+ EXPECT_TRUE(CompareFrames(frame_i420, capture_callback_.last_frame));
+}
+
+// Test frame rate and no picture alarm.
+TEST_F(VideoCaptureExternalTest , FrameRate) {
+ WebRtc_Word64 testTime = 3;
+ TickTime startTime = TickTime::Now();
+ capture_callback_.capability.maxFPS = 10;
+ while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
+ EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
+ test_frame_.Buffer(), test_frame_.Length(),
+ capture_callback_.capability, 0));
+ SLEEP(1000 / capture_callback_.capability.maxFPS);
+ }
+ EXPECT_TRUE(capture_feedback_.frame_rate >= 8 &&
+ capture_feedback_.frame_rate <= 10);
+ SLEEP(500);
+ EXPECT_EQ(webrtc::Raised, capture_feedback_.alarm);
+
+ startTime = TickTime::Now();
+ capture_callback_.capability.maxFPS = 30;
+ while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
+ EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
+ test_frame_.Buffer(), test_frame_.Length(),
+ capture_callback_.capability, 0));
+ SLEEP(1000 / capture_callback_.capability.maxFPS);
+ }
+ EXPECT_EQ(webrtc::Cleared, capture_feedback_.alarm);
+ // Frame rate might be less than 33 since we have paused providing
+ // frames for a while.
+ EXPECT_TRUE(capture_feedback_.frame_rate >= 25 &&
+ capture_feedback_.frame_rate <= 33);
+}
+
+// Test start image
+TEST_F(VideoCaptureExternalTest , StartImage) {
+ capture_callback_.capability.maxFPS = 10;
+ EXPECT_EQ(0, capture_module_->StartSendImage(
+ test_frame_, capture_callback_.capability.maxFPS));
+
+ EXPECT_TRUE_WAIT(capture_callback_.incoming_frames == 5, kTimeOut);
+ EXPECT_EQ(0, capture_module_->StopSendImage());
+
+ SLEEP(200);
+ // Test that no more start images have arrived.
+ EXPECT_TRUE(capture_callback_.incoming_frames >= 4 &&
+ capture_callback_.incoming_frames <= 5);
+ EXPECT_TRUE(CompareFrames(test_frame_, capture_callback_.last_frame));
+}
+
diff --git a/trunk/src/modules/video_coding/codecs/OWNERS b/trunk/src/modules/video_coding/codecs/OWNERS
new file mode 100644
index 0000000..7183cf2
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/OWNERS
@@ -0,0 +1,4 @@
+stefan@webrtc.org
+mikhal@webrtc.org
+marpan@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h b/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h
new file mode 100644
index 0000000..fb4ba8c
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h
@@ -0,0 +1,156 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
+
+#include "video_codec_interface.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class I420Encoder : public VideoEncoder
+{
+public:
+
+ I420Encoder();
+
+ virtual ~I420Encoder();
+
+// Initialize the encoder with the information from the VideoCodec
+//
+// Input:
+// - codecSettings : Codec settings
+// - numberOfCores : Number of cores available for the encoder
+// - maxPayloadSize : The maximum size each payload is allowed
+// to have. Usually MTU - overhead.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK
+// <0 - Error
+ virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/, WebRtc_UWord32 /*maxPayloadSize*/);
+
+// "Encode" an I420 image (as a part of a video stream). The encoded image
+// will be returned to the user via the encode complete callback.
+//
+// Input:
+// - inputImage : Image to be encoded
+// - codecSpecificInfo : Pointer to codec specific data
+// - frameType : Frame type to be sent (Key /Delta) .
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK
+// <0 - Error
+ virtual WebRtc_Word32
+ Encode(const RawImage& inputImage,
+ const CodecSpecificInfo* /*codecSpecificInfo*/,
+ const VideoFrameType* /*frameTypes*/);
+
+// Register an encode complete callback object.
+//
+// Input:
+// - callback : Callback object which handles encoded images.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
+
+// Free encoder memory.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 Release();
+
+ virtual WebRtc_Word32 SetRates(WebRtc_UWord32 /*newBitRate*/,
+ WebRtc_UWord32 /*frameRate*/)
+ {return WEBRTC_VIDEO_CODEC_OK;}
+
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 /*packetLoss*/,
+ int /*rtt*/)
+ {return WEBRTC_VIDEO_CODEC_OK;}
+
+ virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
+ WebRtc_Word32 /*size*/)
+ {return WEBRTC_VIDEO_CODEC_OK;}
+
+private:
+ bool _inited;
+ EncodedImage _encodedImage;
+ EncodedImageCallback* _encodedCompleteCallback;
+
+}; // end of WebRtcI420DEncoder class
+
+class I420Decoder : public VideoDecoder
+{
+public:
+
+ I420Decoder();
+
+ virtual ~I420Decoder();
+
+// Initialize the decoder.
+// The user must notify the codec of width and height values.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK.
+// <0 - Errors
+ virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/);
+
+ virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/){return WEBRTC_VIDEO_CODEC_OK;};
+
+// Decode encoded image (as a part of a video stream). The decoded image
+// will be returned to the user through the decode complete callback.
+//
+// Input:
+// - inputImage : Encoded image to be decoded
+// - missingFrames : True if one or more frames have been lost
+// since the previous decode call.
+// - codecSpecificInfo : pointer to specific codec data
+// - renderTimeMs : Render time in Ms
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK
+// <0 - Error
+ virtual WebRtc_Word32 Decode(
+ const EncodedImage& inputImage,
+ bool missingFrames,
+ const RTPFragmentationHeader* /*fragmentation*/,
+ const CodecSpecificInfo* /*codecSpecificInfo*/,
+ WebRtc_Word64 /*renderTimeMs*/);
+
+// Register a decode complete callback object.
+//
+// Input:
+// - callback : Callback object which handles decoded images.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
+
+// Free decoder memory.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK if OK
+// <0 - Error
+ virtual WebRtc_Word32 Release();
+
+// Reset decoder state and prepare for a new call.
+//
+// Return value : WEBRTC_VIDEO_CODEC_OK.
+// <0 - Error
+ virtual WebRtc_Word32 Reset();
+
+private:
+
+ RawImage _decodedImage;
+ WebRtc_Word32 _width;
+ WebRtc_Word32 _height;
+ bool _inited;
+ DecodedImageCallback* _decodeCompleteCallback;
+
+
+}; // end of WebRtcI420Decoder class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk b/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk
new file mode 100644
index 0000000..d73e77d
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk
@@ -0,0 +1,41 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_i420
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := i420.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../common_video/interface \
+ $(LOCAL_PATH)/../../../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc
new file mode 100644
index 0000000..b73a00a
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "i420.h"
+#include <string.h>
+
+namespace webrtc
+{
+
+I420Encoder::I420Encoder():
+_inited(false),
+_encodedImage(),
+_encodedCompleteCallback(NULL)
+{
+ //
+}
+
+I420Encoder::~I420Encoder()
+{
+ _inited = false;
+ if (_encodedImage._buffer != NULL)
+ {
+ delete [] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ }
+}
+
+WebRtc_Word32
+I420Encoder::Release()
+{
+ // should allocate an encoded frame and then release it here, for that we actaully need an init flag
+ if (_encodedImage._buffer != NULL)
+ {
+ delete [] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ }
+ _inited = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Encoder::InitEncode(const VideoCodec* codecSettings,
+ WebRtc_Word32 /*numberOfCores*/,
+ WebRtc_UWord32 /*maxPayloadSize */)
+{
+ if (codecSettings == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (codecSettings->width < 1 || codecSettings->height < 1)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+
+ // allocating encoded memory
+
+ if (_encodedImage._buffer != NULL)
+ {
+ delete [] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ _encodedImage._size = 0;
+ }
+ const WebRtc_UWord32 newSize = (3 * codecSettings->width *
+ codecSettings->height) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _encodedImage._size = newSize;
+ _encodedImage._buffer = newBuffer;
+
+ // if no memory allocation, no point to init
+ _inited = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+
+WebRtc_Word32
+I420Encoder::Encode(const RawImage& inputImage,
+ const CodecSpecificInfo* /*codecSpecificInfo*/,
+ const VideoFrameType* /*frameTypes*/)
+{
+ if (!_inited)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (_encodedCompleteCallback == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+
+ _encodedImage._frameType = kKeyFrame; // no coding
+ _encodedImage._timeStamp = inputImage._timeStamp;
+ _encodedImage._encodedHeight = inputImage._height;
+ _encodedImage._encodedWidth = inputImage._width;
+ if (inputImage._length > _encodedImage._size)
+ {
+
+ // allocating encoded memory
+ if (_encodedImage._buffer != NULL)
+ {
+ delete [] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ _encodedImage._size = 0;
+ }
+ const WebRtc_UWord32 newSize = (3 * _encodedImage._encodedWidth * _encodedImage._encodedHeight) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _encodedImage._size = newSize;
+ _encodedImage._buffer = newBuffer;
+ }
+ memcpy(_encodedImage._buffer, inputImage._buffer, inputImage._length);
+ _encodedImage._length = inputImage._length;
+ _encodedCompleteCallback->Encoded(_encodedImage);
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+WebRtc_Word32
+I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
+{
+ _encodedCompleteCallback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+I420Decoder::I420Decoder():
+_decodedImage(),
+_width(0),
+_height(0),
+_inited(false),
+_decodeCompleteCallback(NULL)
+{
+ //
+}
+
+I420Decoder::~I420Decoder()
+{
+ Release();
+}
+
+WebRtc_Word32
+I420Decoder::Reset()
+{
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+WebRtc_Word32
+I420Decoder::InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores */)
+{
+ if (codecSettings == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ else if (codecSettings->width < 1 || codecSettings->height < 1)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ _width = codecSettings->width;
+ _height = codecSettings->height;
+ _inited = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::Decode(const EncodedImage& inputImage,
+ bool /*missingFrames*/,
+ const RTPFragmentationHeader* /*fragmentation*/,
+ const CodecSpecificInfo* /*codecSpecificInfo*/,
+ WebRtc_Word64 /*renderTimeMs*/)
+{
+ if (inputImage._buffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (_decodeCompleteCallback == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (inputImage._length <= 0)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (!_inited)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+
+ //Allocate memory for decoded image
+
+ if (_decodedImage._buffer != NULL)
+ {
+ delete [] _decodedImage._buffer;
+ _decodedImage._buffer = NULL;
+ _decodedImage._size = 0;
+ }
+ if (_decodedImage._buffer == NULL)
+ {
+ const WebRtc_UWord32 newSize = (3*_width*_height) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _decodedImage._size = newSize;
+ _decodedImage._buffer = newBuffer;
+ }
+
+ // Set decoded image parameters
+ _decodedImage._height = _height;
+ _decodedImage._width = _width;
+ _decodedImage._timeStamp = inputImage._timeStamp;
+ memcpy(_decodedImage._buffer, inputImage._buffer, inputImage._length);
+ _decodedImage._length = inputImage._length;
+ //_decodedImage._buffer = inputImage._buffer;
+
+ _decodeCompleteCallback->Decoded(_decodedImage);
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::RegisterDecodeCompleteCallback(DecodedImageCallback* callback)
+{
+ _decodeCompleteCallback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::Release()
+{
+ if (_decodedImage._buffer != NULL)
+ {
+ delete [] _decodedImage._buffer;
+ _decodedImage._buffer = NULL;
+ }
+ _inited = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi
new file mode 100644
index 0000000..af13f8d
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_i420',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ '../../../../../../common_video/interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../../../../../common_video/interface',
+ ],
+ },
+ 'sources': [
+ '../interface/i420.h',
+ 'i420.cc',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
new file mode 100644
index 0000000..f61fce5
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+
+#include <string>
+
+#include "gmock/gmock.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class MockEncodedImageCallback : public EncodedImageCallback {
+ public:
+ MOCK_METHOD3(Encoded,
+ WebRtc_Word32(EncodedImage& encodedImage,
+ const CodecSpecificInfo* codecSpecificInfo,
+ const RTPFragmentationHeader* fragmentation));
+};
+
+class MockVideoEncoder : public VideoEncoder {
+ public:
+ MOCK_CONST_METHOD2(Version,
+ WebRtc_Word32(WebRtc_Word8 *version,
+ WebRtc_Word32 length));
+ MOCK_METHOD3(InitEncode,
+ WebRtc_Word32(const VideoCodec* codecSettings,
+ WebRtc_Word32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize));
+ MOCK_METHOD3(Encode,
+ WebRtc_Word32(const RawImage& inputImage,
+ const CodecSpecificInfo* codecSpecificInfo,
+ const VideoFrameType* frameType));
+ MOCK_METHOD1(RegisterEncodeCompleteCallback,
+ WebRtc_Word32(EncodedImageCallback* callback));
+ MOCK_METHOD0(Release, WebRtc_Word32());
+ MOCK_METHOD0(Reset, WebRtc_Word32());
+ MOCK_METHOD2(SetChannelParameters, WebRtc_Word32(WebRtc_UWord32 packetLoss,
+ int rtt));
+ MOCK_METHOD2(SetRates,
+ WebRtc_Word32(WebRtc_UWord32 newBitRate,
+ WebRtc_UWord32 frameRate));
+ MOCK_METHOD1(SetPeriodicKeyFrames, WebRtc_Word32(bool enable));
+ MOCK_METHOD2(CodecConfigParameters,
+ WebRtc_Word32(WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
+};
+
+class MockDecodedImageCallback : public DecodedImageCallback {
+ public:
+ MOCK_METHOD1(Decoded,
+ WebRtc_Word32(RawImage& decodedImage));
+ MOCK_METHOD1(ReceivedDecodedReferenceFrame,
+ WebRtc_Word32(const WebRtc_UWord64 pictureId));
+ MOCK_METHOD1(ReceivedDecodedFrame,
+ WebRtc_Word32(const WebRtc_UWord64 pictureId));
+};
+
+class MockVideoDecoder : public VideoDecoder {
+ public:
+ MOCK_METHOD2(InitDecode,
+ WebRtc_Word32(const VideoCodec* codecSettings,
+ WebRtc_Word32 numberOfCores));
+ MOCK_METHOD5(Decode,
+ WebRtc_Word32(const EncodedImage& inputImage,
+ bool missingFrames,
+ const RTPFragmentationHeader* fragmentation,
+ const CodecSpecificInfo* codecSpecificInfo,
+ WebRtc_Word64 renderTimeMs));
+ MOCK_METHOD1(RegisterDecodeCompleteCallback,
+ WebRtc_Word32(DecodedImageCallback* callback));
+ MOCK_METHOD0(Release, WebRtc_Word32());
+ MOCK_METHOD0(Reset, WebRtc_Word32());
+ MOCK_METHOD2(SetCodecConfigParameters,
+ WebRtc_Word32(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
+ MOCK_METHOD0(Copy, VideoDecoder*());
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
diff --git a/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h b/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h
new file mode 100644
index 0000000..9aa5cb7
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -0,0 +1,247 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+
+#include "common_types.h"
+#include "common_video/interface/video_image.h"
+#include "modules/video_coding/codecs/interface/video_error_codes.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class RTPFragmentationHeader; // forward declaration
+
+// Note: if any pointers are added to this struct, it must be fitted
+// with a copy-constructor. See below.
+struct CodecSpecificInfoVP8
+{
+ bool hasReceivedSLI;
+ WebRtc_UWord8 pictureIdSLI;
+ bool hasReceivedRPSI;
+ WebRtc_UWord64 pictureIdRPSI;
+ WebRtc_Word16 pictureId; // negative value to skip pictureId
+ bool nonReference;
+ WebRtc_UWord8 simulcastIdx;
+ WebRtc_UWord8 temporalIdx;
+ bool layerSync;
+ int tl0PicIdx; // Negative value to skip tl0PicIdx
+ WebRtc_Word8 keyIdx; // negative value to skip keyIdx
+};
+
+union CodecSpecificInfoUnion
+{
+ CodecSpecificInfoVP8 VP8;
+};
+
+// Note: if any pointers are added to this struct or its sub-structs, it
+// must be fitted with a copy-constructor. This is because it is copied
+// in the copy-constructor of VCMEncodedFrame.
+struct CodecSpecificInfo
+{
+ VideoCodecType codecType;
+ CodecSpecificInfoUnion codecSpecific;
+};
+
+class EncodedImageCallback
+{
+public:
+ virtual ~EncodedImageCallback() {};
+
+ // Callback function which is called when an image has been encoded.
+ //
+ // Input:
+ // - encodedImage : The encoded image
+ //
+ // Return value : > 0, signals to the caller that one or more future frames
+ // should be dropped to keep bit rate or frame rate.
+ // = 0, if OK.
+ // < 0, on error.
+ virtual WebRtc_Word32
+ Encoded(EncodedImage& encodedImage,
+ const CodecSpecificInfo* codecSpecificInfo = NULL,
+ const RTPFragmentationHeader* fragmentation = NULL) = 0;
+};
+
+class VideoEncoder
+{
+public:
+ virtual ~VideoEncoder() {};
+
+ // Initialize the encoder with the information from the VideoCodec.
+ //
+ // Input:
+ // - codecSettings : Codec settings
+ // - numberOfCores : Number of cores available for the encoder
+ // - maxPayloadSize : The maximum size each payload is allowed
+ // to have. Usually MTU - overhead.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores, WebRtc_UWord32 maxPayloadSize) = 0;
+
+ // Encode an I420 image (as a part of a video stream). The encoded image
+ // will be returned to the user through the encode complete callback.
+ //
+ // Input:
+ // - inputImage : Image to be encoded
+ // - codecSpecificInfo : Pointer to codec specific data
+ // - frameType : The frame type to encode
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32
+ Encode(const RawImage& inputImage,
+ const CodecSpecificInfo* codecSpecificInfo,
+ const VideoFrameType* frameTypes) = 0;
+
+ // Register an encode complete callback object.
+ //
+ // Input:
+ // - callback : Callback object which handles encoded images.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback) = 0;
+
+ // Free encoder memory.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 Release() = 0;
+
+ // Inform the encoder about the packet loss and round trip time on the
+ // network used to decide the best pattern and signaling.
+ //
+ // - packetLoss : Fraction lost (loss rate in percent =
+ // 100 * packetLoss / 255)
+ // - rtt : Round-trip time in milliseconds
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
+ int rtt) = 0;
+
+ // Inform the encoder about the new target bit rate.
+ //
+ // - newBitRate : New target bit rate
+ // - frameRate : The target frame rate
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate) = 0;
+
+ // Use this function to enable or disable periodic key frames. Can be useful for codecs
+ // which have other ways of stopping error propagation.
+ //
+ // - enable : Enable or disable periodic key frames
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable) { return WEBRTC_VIDEO_CODEC_ERROR; }
+
+ // Codec configuration data to send out-of-band, i.e. in SIP call setup
+ //
+ // - buffer : Buffer pointer to where the configuration data
+ // should be stored
+ // - size : The size of the buffer in bytes
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
+};
+
+class DecodedImageCallback
+{
+public:
+ virtual ~DecodedImageCallback() {};
+
+ // Callback function which is called when an image has been decoded.
+ //
+ // Input:
+ // - decodedImage : The decoded image
+ //
+ // Return value : 0 if OK, < 0 otherwise.
+ virtual WebRtc_Word32 Decoded(RawImage& decodedImage) = 0;
+
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
+
+ virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId) {return -1;}
+};
+
+class VideoDecoder
+{
+public:
+ virtual ~VideoDecoder() {};
+
+ // Initialize the decoder with the information from the VideoCodec.
+ //
+ // Input:
+ // - inst : Codec settings
+ // - numberOfCores : Number of cores available for the decoder
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores) = 0;
+
+ // Decode encoded image (as a part of a video stream). The decoded image
+ // will be returned to the user through the decode complete callback.
+ //
+ // Input:
+ // - inputImage : Encoded image to be decoded
+ // - missingFrames : True if one or more frames have been lost
+ // since the previous decode call.
+ // - fragmentation : Specifies where the encoded frame can be
+ // split into separate fragments. The meaning
+ // of fragment is codec specific, but often
+ // means that each fragment is decodable by
+ // itself.
+ // - codecSpecificInfo : Pointer to codec specific data
+ // - renderTimeMs : System time to render in milliseconds. Only
+ // used by decoders with internal rendering.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32
+ Decode(const EncodedImage& inputImage,
+ bool missingFrames,
+ const RTPFragmentationHeader* fragmentation,
+ const CodecSpecificInfo* codecSpecificInfo = NULL,
+ WebRtc_Word64 renderTimeMs = -1) = 0;
+
+ // Register an decode complete callback object.
+ //
+ // Input:
+ // - callback : Callback object which handles decoded images.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback) = 0;
+
+ // Free decoder memory.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 Release() = 0;
+
+ // Reset decoder state and prepare for a new call.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 Reset() = 0;
+
+ // Codec configuration data sent out-of-band, i.e. in SIP call setup
+ //
+ // Input/Output:
+ // - buffer : Buffer pointer to the configuration data
+ // - size : The size of the configuration data in
+ // bytes
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
+
+ // Create a copy of the codec and its internal state.
+ //
+ // Return value : A copy of the instance if OK, NULL otherwise.
+ virtual VideoDecoder* Copy() { return NULL; }
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
diff --git a/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h b/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h
new file mode 100644
index 0000000..dfa3f53
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+
+// NOTE: in sync with video_coding_module_defines.h
+
+// Define return values
+
+#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
+#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
+#define WEBRTC_VIDEO_CODEC_OK 0
+#define WEBRTC_VIDEO_CODEC_ERROR -1
+#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
+#define WEBRTC_VIDEO_CODEC_MEMORY -3
+#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
+#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
+#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
+#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
+#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
diff --git a/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
new file mode 100644
index 0000000..57d21ca
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
+
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <string>
+
+#include "common_video/interface/video_image.h"
+#include "gmock/gmock.h"
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class MockPacketManipulator : public PacketManipulator {
+ public:
+ MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image));
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc
new file mode 100644
index 0000000..acdb2e5
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <cassert>
+#include <cstdio>
+
+namespace webrtc {
+namespace test {
+
+PacketManipulatorImpl::PacketManipulatorImpl(PacketReader* packet_reader,
+ const NetworkingConfig& config,
+ bool verbose)
+ : packet_reader_(packet_reader),
+ config_(config),
+ active_burst_packets_(0),
+ critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ random_seed_(1),
+ verbose_(verbose) {
+ assert(packet_reader);
+}
+
+PacketManipulatorImpl::~PacketManipulatorImpl() {
+ delete critsect_;
+}
+
+int PacketManipulatorImpl::ManipulatePackets(
+ webrtc::EncodedImage* encoded_image) {
+ assert(encoded_image);
+ int nbr_packets_dropped = 0;
+ // There's no need to build a copy of the image data since viewing an
+ // EncodedImage object, setting the length to a new lower value represents
+ // that everything is dropped after that position in the byte array.
+ // EncodedImage._size is the allocated bytes.
+ // EncodedImage._length is how many that are filled with data.
+ int new_length = 0;
+ packet_reader_->InitializeReading(encoded_image->_buffer,
+ encoded_image->_length,
+ config_.packet_size_in_bytes);
+ WebRtc_UWord8* packet = NULL;
+ int nbr_bytes_to_read;
+ // keep track of if we've lost any packets, since then we shall loose
+ // the remains of the current frame:
+ bool packet_loss_has_occurred = false;
+ while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
+ // Check if we're currently in a packet loss burst that is not completed:
+ if (active_burst_packets_ > 0) {
+ active_burst_packets_--;
+ nbr_packets_dropped++;
+ } else if (RandomUniform() < config_.packet_loss_probability ||
+ packet_loss_has_occurred) {
+ packet_loss_has_occurred = true;
+ nbr_packets_dropped++;
+ if (config_.packet_loss_mode == kBurst) {
+ // Initiate a new burst
+ active_burst_packets_ = config_.packet_loss_burst_length - 1;
+ }
+ } else {
+ new_length += nbr_bytes_to_read;
+ }
+ }
+ encoded_image->_length = new_length;
+ if (nbr_packets_dropped > 0) {
+ // Must set completeFrame to false to inform the decoder about this:
+ encoded_image->_completeFrame = false;
+ if (verbose_) {
+ printf("Dropped %d packets for frame %d (frame length: %d)\n",
+ nbr_packets_dropped, encoded_image->_timeStamp,
+ encoded_image->_length);
+ }
+ }
+ return nbr_packets_dropped;
+}
+
+void PacketManipulatorImpl::InitializeRandomSeed(unsigned int seed) {
+ random_seed_ = seed;
+}
+
+inline double PacketManipulatorImpl::RandomUniform() {
+ // Use the previous result as new seed before each rand() call. Doing this
+ // it doesn't matter if other threads are calling rand() since we'll always
+ // get the same behavior as long as we're using a fixed initial seed.
+ critsect_->Enter();
+ srand(random_seed_);
+ random_seed_ = std::rand();
+ critsect_->Leave();
+ return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
+}
+
+const char* PacketLossModeToStr(PacketLossMode e) {
+ switch (e) {
+ case kUniform:
+ return "Uniform";
+ case kBurst:
+ return "Burst";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+} // namespace test
+} // namespace webrtcc
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h
new file mode 100644
index 0000000..e3891cb
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h
@@ -0,0 +1,113 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
+
+#include <cstdlib>
+
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+// Which mode the packet loss shall be performed according to.
+enum PacketLossMode {
+ // Drops packets with a configured probability independently for each packet
+ kUniform,
+ // Drops packets similar to uniform but when a packet is being dropped,
+ // the number of lost packets in a row is equal to the configured burst
+ // length.
+ kBurst
+};
+// Returns a string representation of the enum value.
+const char* PacketLossModeToStr(PacketLossMode e);
+
+// Contains configurations related to networking and simulation of
+// scenarios caused by network interference.
+struct NetworkingConfig {
+ NetworkingConfig()
+ : packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
+ packet_loss_mode(kUniform), packet_loss_probability(0.0),
+ packet_loss_burst_length(1) {
+ }
+
+ // Packet size in bytes. Default: 1500 bytes.
+ int packet_size_in_bytes;
+
+ // Encoder specific setting of maximum size in bytes of each payload.
+ // Default: 1440 bytes.
+ int max_payload_size_in_bytes;
+
+ // Packet loss mode. Two different packet loss models are supported:
+ // uniform or burst. This setting has no effect unless
+ // packet_loss_probability is >0.
+ // Default: uniform.
+ PacketLossMode packet_loss_mode;
+
+ // Packet loss probability. A value between 0.0 and 1.0 that defines the
+ // probability of a packet being lost. 0.1 means 10% and so on.
+ // Default: 0 (no loss).
+ double packet_loss_probability;
+
+ // Packet loss burst length. Defines how many packets will be lost in a burst
+ // when a packet has been decided to be lost. Must be >=1. Default: 1.
+ int packet_loss_burst_length;
+};
+
+// Class for simulating packet loss on the encoded frame data.
+// When a packet loss has occurred in a frame, the remaining data in that
+// frame is lost (even if burst length is only a single packet).
+// TODO(kjellander): Support discarding only individual packets in the frame
+// when CL 172001 has been submitted. This also requires a correct
+// fragmentation header to be passed to the decoder.
+//
+// To get a repeatable packet drop pattern, re-initialize the random seed
+// using InitializeRandomSeed before each test run.
+class PacketManipulator {
+ public:
+ virtual ~PacketManipulator() {}
+
+ // Manipulates the data of the encoded_image to simulate parts being lost
+ // during transport.
+ // If packets are dropped from frame data, the completedFrame field will be
+ // set to false.
+ // Returns the number of packets being dropped.
+ virtual int
+ ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
+};
+
+class PacketManipulatorImpl : public PacketManipulator {
+ public:
+ PacketManipulatorImpl(PacketReader* packet_reader,
+ const NetworkingConfig& config,
+ bool verbose);
+ virtual ~PacketManipulatorImpl();
+ virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image);
+ virtual void InitializeRandomSeed(unsigned int seed);
+ protected:
+ // Returns a uniformly distributed random value between 0.0 and 1.0
+ virtual double RandomUniform();
+ private:
+ PacketReader* packet_reader_;
+ const NetworkingConfig& config_;
+ // Used to simulate a burst over several frames.
+ int active_burst_packets_;
+ CriticalSectionWrapper* critsect_;
+ unsigned int random_seed_;
+ bool verbose_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
new file mode 100644
index 0000000..a5d8bc3
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
@@ -0,0 +1,153 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <queue>
+
+#include "gtest/gtest.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
+#include "testsupport/unittest_utils.h"
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+const double kNeverDropProbability = 0.0;
+const double kAlwaysDropProbability = 1.0;
+const int kBurstLength = 1;
+
+class PacketManipulatorTest: public PacketRelatedTest {
+ protected:
+ PacketReader packet_reader_;
+ EncodedImage image_;
+ NetworkingConfig drop_config_;
+ NetworkingConfig no_drop_config_;
+
+ PacketManipulatorTest() {
+ image_._buffer = packet_data_;
+ image_._length = kPacketDataLength;
+ image_._size = kPacketDataLength;
+
+ drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
+ drop_config_.packet_loss_probability = kAlwaysDropProbability;
+ drop_config_.packet_loss_burst_length = kBurstLength;
+ drop_config_.packet_loss_mode = kUniform;
+
+ no_drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
+ no_drop_config_.packet_loss_probability = kNeverDropProbability;
+ no_drop_config_.packet_loss_burst_length = kBurstLength;
+ no_drop_config_.packet_loss_mode = kUniform;
+ }
+
+ virtual ~PacketManipulatorTest() {}
+
+ void SetUp() {
+ PacketRelatedTest::SetUp();
+ }
+
+ void TearDown() {
+ PacketRelatedTest::TearDown();
+ }
+
+ void VerifyPacketLoss(int expected_nbr_packets_dropped,
+ int actual_nbr_packets_dropped,
+ int expected_packet_data_length,
+ WebRtc_UWord8* expected_packet_data,
+ EncodedImage& actual_image) {
+ EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
+ EXPECT_EQ(expected_packet_data_length, static_cast<int>(image_._length));
+ EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
+ expected_packet_data_length));
+ }
+};
+
+TEST_F(PacketManipulatorTest, Constructor) {
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+}
+
+TEST_F(PacketManipulatorTest, DropNone) {
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+ VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
+ packet_data_, image_);
+}
+
+TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
+ int data_length = 400; // smaller than the packet size
+ image_._length = data_length;
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ VerifyPacketLoss(0, nbr_packets_dropped, data_length,
+ packet_data_, image_);
+}
+
+TEST_F(PacketManipulatorTest, UniformDropAll) {
+ PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+ VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
+ 0, packet_data_, image_);
+}
+
+// Use our customized test class to make the second packet being lost
+TEST_F(PacketManipulatorTest, UniformDropSinglePacket) {
+ drop_config_.packet_loss_probability = 0.5;
+ PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
+ manipulator.AddRandomResult(1.0);
+ manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
+ manipulator.AddRandomResult(1.0);
+
+ // Execute the test target method:
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ // Since we setup the predictive packet manipulator, it will throw away the
+ // second packet. The third packet is also lost because when we have lost one,
+ // the remains shall also be discarded (in the current implementation).
+ VerifyPacketLoss(2, nbr_packets_dropped, kPacketSizeInBytes, packet1_,
+ image_);
+}
+
+// Use our customized test class to make the second packet being lost
+TEST_F(PacketManipulatorTest, BurstDropNinePackets) {
+ // Create a longer packet data structure (10 packets)
+ const int kNbrPackets = 10;
+ const int kDataLength = kPacketSizeInBytes * kNbrPackets;
+ WebRtc_UWord8 data[kDataLength];
+ WebRtc_UWord8* data_pointer = data;
+ // Fill with 0s, 1s and so on to be able to easily verify which were dropped:
+ for (int i = 0; i < kNbrPackets; ++i) {
+ memset(data_pointer + i * kPacketSizeInBytes, i, kPacketSizeInBytes);
+ }
+ // Overwrite the defaults from the test fixture:
+ image_._buffer = data;
+ image_._length = kDataLength;
+ image_._size = kDataLength;
+
+ drop_config_.packet_loss_probability = 0.5;
+ drop_config_.packet_loss_burst_length = 5;
+ drop_config_.packet_loss_mode = kBurst;
+ PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
+ manipulator.AddRandomResult(1.0);
+ manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
+ for (int i = 0; i < kNbrPackets - 2; ++i) {
+ manipulator.AddRandomResult(1.0);
+ }
+
+ // Execute the test target method:
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ // Should discard every packet after the first one.
+ VerifyPacketLoss(9, nbr_packets_dropped, kPacketSizeInBytes, data, image_);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
new file mode 100644
index 0000000..5668378
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
+
+#include <cassert>
+#include <cstdio>
+
+#include "testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+PredictivePacketManipulator::PredictivePacketManipulator(
+ PacketReader* packet_reader, const NetworkingConfig& config)
+ : PacketManipulatorImpl(packet_reader, config, false) {
+}
+
+PredictivePacketManipulator::~PredictivePacketManipulator() {
+}
+
+
+void PredictivePacketManipulator::AddRandomResult(double result) {
+ assert(result >= 0.0 && result <= 1.0);
+ random_results_.push(result);
+}
+
+double PredictivePacketManipulator::RandomUniform() {
+ if(random_results_.size() == 0u) {
+ fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
+ "is called same amount of times you're going to invoke the "
+ "RandomUniform() function, i.e. once per packet.\n");
+ assert(false);
+ }
+ double result = random_results_.front();
+ random_results_.pop();
+ return result;
+}
+
+} // namespace test
+} // namespace webrtcc
diff --git a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h
new file mode 100644
index 0000000..22a0ce9
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
+
+#include <queue>
+
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+#include "testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+// Predictive packet manipulator that allows for setup of the result of
+// the random invocations.
+class PredictivePacketManipulator : public PacketManipulatorImpl {
+ public:
+ PredictivePacketManipulator(PacketReader* packet_reader,
+ const NetworkingConfig& config);
+ virtual ~PredictivePacketManipulator();
+ // Adds a result. You must add at least the same number of results as the
+ // expected calls to the RandomUniform method. The results are added to a
+ // FIFO queue so they will be returned in the same order they were added.
+ // Result parameter must be 0.0 to 1.0.
+ void AddRandomResult(double result);
+ protected:
+ // Returns a uniformly distributed random value between 0.0 and 1.0
+ virtual double RandomUniform();
+
+ private:
+ std::queue<double> random_results_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/stats.cc b/trunk/src/modules/video_coding/codecs/test/stats.cc
new file mode 100644
index 0000000..5cdf364
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/stats.cc
@@ -0,0 +1,172 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/stats.h"
+
+#include <algorithm> // min_element, max_element
+#include <cassert>
+#include <cstdio>
+
+namespace webrtc {
+namespace test {
+
+Stats::Stats() {}
+
+Stats::~Stats() {}
+
+bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.encode_time_in_us < s2.encode_time_in_us;
+}
+
+bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.decode_time_in_us < s2.decode_time_in_us;
+}
+
+bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
+}
+
+bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
+}
+
+FrameStatistic& Stats::NewFrame(int frame_number) {
+ assert(frame_number >= 0);
+ FrameStatistic stat;
+ stat.frame_number = frame_number;
+ stats_.push_back(stat);
+ return stats_[frame_number];
+}
+
+void Stats::PrintSummary() {
+ printf("Processing summary:\n");
+ if (stats_.size() == 0) {
+ printf("No frame statistics have been logged yet.\n");
+ return;
+ }
+
+ // Calculate min, max, average and total encoding time
+ int total_encoding_time_in_us = 0;
+ int total_decoding_time_in_us = 0;
+ int total_encoded_frames_lengths = 0;
+ int total_encoded_key_frames_lengths = 0;
+ int total_encoded_nonkey_frames_lengths = 0;
+ int nbr_keyframes = 0;
+ int nbr_nonkeyframes = 0;
+
+ for (FrameStatisticsIterator it = stats_.begin();
+ it != stats_.end(); ++it) {
+ total_encoding_time_in_us += it->encode_time_in_us;
+ total_decoding_time_in_us += it->decode_time_in_us;
+ total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
+ if (it->frame_type == webrtc::kKeyFrame) {
+ total_encoded_key_frames_lengths += it->encoded_frame_length_in_bytes;
+ nbr_keyframes++;
+ } else {
+ total_encoded_nonkey_frames_lengths += it->encoded_frame_length_in_bytes;
+ nbr_nonkeyframes++;
+ }
+ }
+
+ FrameStatisticsIterator frame;
+
+ // ENCODING
+ printf("Encoding time:\n");
+ frame = min_element(stats_.begin(),
+ stats_.end(), LessForEncodeTime);
+ printf(" Min : %7d us (frame %d)\n",
+ frame->encode_time_in_us, frame->frame_number);
+
+ frame = max_element(stats_.begin(),
+ stats_.end(), LessForEncodeTime);
+ printf(" Max : %7d us (frame %d)\n",
+ frame->encode_time_in_us, frame->frame_number);
+
+ printf(" Average : %7d us\n",
+ static_cast<int>(total_encoding_time_in_us / stats_.size()));
+
+ // DECODING
+ printf("Decoding time:\n");
+ // only consider frames that were successfully decoded (packet loss may cause
+ // failures)
+ std::vector<FrameStatistic> decoded_frames;
+ for (std::vector<FrameStatistic>::iterator it = stats_.begin();
+ it != stats_.end(); ++it) {
+ if (it->decoding_successful) {
+ decoded_frames.push_back(*it);
+ }
+ }
+ if (decoded_frames.size() == 0) {
+ printf("No successfully decoded frames exist in this statistics.\n");
+ } else {
+ frame = min_element(decoded_frames.begin(),
+ decoded_frames.end(), LessForDecodeTime);
+ printf(" Min : %7d us (frame %d)\n",
+ frame->decode_time_in_us, frame->frame_number);
+
+ frame = max_element(decoded_frames.begin(),
+ decoded_frames.end(), LessForDecodeTime);
+ printf(" Max : %7d us (frame %d)\n",
+ frame->decode_time_in_us, frame->frame_number);
+
+ printf(" Average : %7d us\n",
+ static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
+ printf(" Failures: %d frames failed to decode.\n",
+ static_cast<int>(stats_.size() - decoded_frames.size()));
+ }
+
+ // SIZE
+ printf("Frame sizes:\n");
+ frame = min_element(stats_.begin(),
+ stats_.end(), LessForEncodedSize);
+ printf(" Min : %7d bytes (frame %d)\n",
+ frame->encoded_frame_length_in_bytes, frame->frame_number);
+
+ frame = max_element(stats_.begin(),
+ stats_.end(), LessForEncodedSize);
+ printf(" Max : %7d bytes (frame %d)\n",
+ frame->encoded_frame_length_in_bytes, frame->frame_number);
+
+ printf(" Average : %7d bytes\n",
+ static_cast<int>(total_encoded_frames_lengths / stats_.size()));
+ if (nbr_keyframes > 0) {
+ printf(" Average key frame size : %7d bytes (%d keyframes)\n",
+ total_encoded_key_frames_lengths / nbr_keyframes,
+ nbr_keyframes);
+ }
+ if (nbr_nonkeyframes > 0) {
+ printf(" Average non-key frame size: %7d bytes (%d frames)\n",
+ total_encoded_nonkey_frames_lengths / nbr_nonkeyframes,
+ nbr_nonkeyframes);
+ }
+
+ // BIT RATE
+ printf("Bit rates:\n");
+ frame = min_element(stats_.begin(),
+ stats_.end(), LessForBitRate);
+ printf(" Min bit rate: %7d kbps (frame %d)\n",
+ frame->bit_rate_in_kbps, frame->frame_number);
+
+ frame = max_element(stats_.begin(),
+ stats_.end(), LessForBitRate);
+ printf(" Max bit rate: %7d kbps (frame %d)\n",
+ frame->bit_rate_in_kbps, frame->frame_number);
+
+ printf("\n");
+ printf("Total encoding time : %7d ms.\n",
+ total_encoding_time_in_us / 1000);
+ printf("Total decoding time : %7d ms.\n",
+ total_decoding_time_in_us / 1000);
+ printf("Total processing time: %7d ms.\n",
+ (total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/stats.h b/trunk/src/modules/video_coding/codecs/test/stats.h
new file mode 100644
index 0000000..ec2bb9d
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/stats.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
+
+#include <vector>
+
+#include "common_video/interface/video_image.h"
+
+namespace webrtc {
+namespace test {
+
+// Contains statistics of a single frame that has been processed.
+struct FrameStatistic {
+ FrameStatistic() :
+ encoding_successful(false), decoding_successful(false),
+ encode_return_code(0), decode_return_code(0),
+ encode_time_in_us(0), decode_time_in_us(0),
+ frame_number(0), packets_dropped(0), total_packets(0),
+ bit_rate_in_kbps(0), encoded_frame_length_in_bytes(0),
+ frame_type(kDeltaFrame) {
+ };
+ bool encoding_successful;
+ bool decoding_successful;
+ int encode_return_code;
+ int decode_return_code;
+ int encode_time_in_us;
+ int decode_time_in_us;
+ int frame_number;
+ // How many packets were discarded of the encoded frame data (if any)
+ int packets_dropped;
+ int total_packets;
+
+ // Current bit rate. Calculated out of the size divided with the time
+ // interval per frame.
+ int bit_rate_in_kbps;
+
+ // Copied from EncodedImage
+ int encoded_frame_length_in_bytes;
+ webrtc::VideoFrameType frame_type;
+};
+
+// Handles statistics from a single video processing run.
+// Contains calculation methods for interesting metrics from these stats.
+class Stats {
+ public:
+ typedef std::vector<FrameStatistic>::iterator FrameStatisticsIterator;
+
+ Stats();
+ virtual ~Stats();
+
+ // Add a new statistic data object.
+ // The frame number must be incrementing and start at zero in order to use
+ // it as an index for the frame_statistics_ vector.
+ // Returns the newly created statistic object.
+ FrameStatistic& NewFrame(int frame_number);
+
+ // Prints a summary of all the statistics that have been gathered during the
+ // processing
+ void PrintSummary();
+
+ std::vector<FrameStatistic> stats_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc b/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc
new file mode 100644
index 0000000..53a50d7
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/stats.h"
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class StatsTest: public testing::Test {
+ protected:
+ StatsTest() {
+ }
+
+ virtual ~StatsTest() {
+ }
+
+ void SetUp() {
+ stats_ = new Stats();
+ }
+
+ void TearDown() {
+ delete stats_;
+ }
+
+ Stats* stats_;
+};
+
+// Test empty object
+TEST_F(StatsTest, Uninitialized) {
+ EXPECT_EQ(0u, stats_->stats_.size());
+ stats_->PrintSummary(); // should not crash
+}
+
+// Add single frame stats and verify
+TEST_F(StatsTest, AddOne) {
+ stats_->NewFrame(0u);
+ FrameStatistic* frameStat = &stats_->stats_[0];
+ EXPECT_EQ(0, frameStat->frame_number);
+}
+
+// Add multiple frame stats and verify
+TEST_F(StatsTest, AddMany) {
+ int nbr_of_frames = 1000;
+ for (int i = 0; i < nbr_of_frames; ++i) {
+ FrameStatistic& frameStat = stats_->NewFrame(i);
+ EXPECT_EQ(i, frameStat.frame_number);
+ }
+ EXPECT_EQ(nbr_of_frames, static_cast<int>(stats_->stats_.size()));
+
+ stats_->PrintSummary(); // should not crash
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi b/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi
new file mode 100644
index 0000000..662d09d
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi
@@ -0,0 +1,72 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'video_codecs_test_framework',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ ],
+ 'sources': [
+ 'mock/mock_packet_manipulator.h',
+ 'packet_manipulator.h',
+ 'packet_manipulator.cc',
+ 'predictive_packet_manipulator.h',
+ 'predictive_packet_manipulator.cc',
+ 'stats.h',
+ 'stats.cc',
+ 'videoprocessor.h',
+ 'videoprocessor.cc',
+ ],
+ },
+ {
+ 'target_name': 'video_codecs_test_framework_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_codecs_test_framework',
+ 'webrtc_video_coding',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'packet_manipulator_unittest.cc',
+ 'stats_unittest.cc',
+ 'videoprocessor_unittest.cc',
+ ],
+ },
+ {
+ 'target_name': 'video_codecs_test_framework_integrationtests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_codecs_test_framework',
+ 'webrtc_video_coding',
+ 'webrtc_vp8',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/metrics.gyp:metrics',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'videoprocessor_integrationtest.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc
new file mode 100644
index 0000000..8b13513
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc
@@ -0,0 +1,287 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+
+#include <cassert>
+#include <cstring>
+#include <limits>
+
+#include "system_wrappers/interface/cpu_info.h"
+
+namespace webrtc {
+namespace test {
+
+VideoProcessorImpl::VideoProcessorImpl(webrtc::VideoEncoder* encoder,
+ webrtc::VideoDecoder* decoder,
+ FrameReader* frame_reader,
+ FrameWriter* frame_writer,
+ PacketManipulator* packet_manipulator,
+ const TestConfig& config,
+ Stats* stats)
+ : encoder_(encoder),
+ decoder_(decoder),
+ frame_reader_(frame_reader),
+ frame_writer_(frame_writer),
+ packet_manipulator_(packet_manipulator),
+ config_(config),
+ stats_(stats),
+ encode_callback_(NULL),
+ decode_callback_(NULL),
+ source_buffer_(NULL),
+ first_key_frame_has_been_excluded_(false),
+ last_frame_missing_(false),
+ initialized_(false) {
+ assert(encoder);
+ assert(decoder);
+ assert(frame_reader);
+ assert(frame_writer);
+ assert(packet_manipulator);
+ assert(stats);
+}
+
+bool VideoProcessorImpl::Init() {
+ // Calculate a factor used for bit rate calculations:
+ bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits
+
+ int frame_length_in_bytes = frame_reader_->FrameLength();
+
+ // Initialize data structures used by the encoder/decoder APIs
+ source_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
+ last_successful_frame_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
+
+ // Set fixed properties common for all frames:
+ source_frame_._width = config_.codec_settings->width;
+ source_frame_._height = config_.codec_settings->height;
+ source_frame_._length = frame_length_in_bytes;
+ source_frame_._size = frame_length_in_bytes;
+
+ // Setup required callbacks for the encoder/decoder:
+ encode_callback_ = new VideoProcessorEncodeCompleteCallback(this);
+ decode_callback_ = new VideoProcessorDecodeCompleteCallback(this);
+ WebRtc_Word32 register_result =
+ encoder_->RegisterEncodeCompleteCallback(encode_callback_);
+ if (register_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to register encode complete callback, return code: "
+ "%d\n", register_result);
+ return false;
+ }
+ register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
+ if (register_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to register decode complete callback, return code: "
+ "%d\n", register_result);
+ return false;
+ }
+ // Init the encoder and decoder
+ WebRtc_UWord32 nbr_of_cores = 1;
+ if (!config_.use_single_core) {
+ nbr_of_cores = CpuInfo::DetectNumberOfCores();
+ }
+ WebRtc_Word32 init_result =
+ encoder_->InitEncode(config_.codec_settings, nbr_of_cores,
+ config_.networking_config.max_payload_size_in_bytes);
+ if (init_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to initialize VideoEncoder, return code: %d\n",
+ init_result);
+ return false;
+ }
+ init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores);
+ if (init_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to initialize VideoDecoder, return code: %d\n",
+ init_result);
+ return false;
+ }
+
+ if (config_.verbose) {
+ printf("Video Processor:\n");
+ printf(" #CPU cores used : %d\n", nbr_of_cores);
+ printf(" Total # of frames: %d\n", frame_reader_->NumberOfFrames());
+ printf(" Codec settings:\n");
+ printf(" Start bitrate : %d kbps\n",
+ config_.codec_settings->startBitrate);
+ printf(" Width : %d\n", config_.codec_settings->width);
+ printf(" Height : %d\n", config_.codec_settings->height);
+ }
+ initialized_ = true;
+ return true;
+}
+
+VideoProcessorImpl::~VideoProcessorImpl() {
+ delete[] source_buffer_;
+ delete[] last_successful_frame_buffer_;
+ encoder_->RegisterEncodeCompleteCallback(NULL);
+ delete encode_callback_;
+ decoder_->RegisterDecodeCompleteCallback(NULL);
+ delete decode_callback_;
+}
+
+bool VideoProcessorImpl::ProcessFrame(int frame_number) {
+ assert(frame_number >=0);
+ if (!initialized_) {
+ fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
+ return false;
+ }
+ if (frame_reader_->ReadFrame(source_buffer_)) {
+ // point the source frame buffer to the newly read frame data:
+ source_frame_._buffer = source_buffer_;
+
+ // Ensure we have a new statistics data object we can fill:
+ FrameStatistic& stat = stats_->NewFrame(frame_number);
+
+ encode_start_ = TickTime::Now();
+ // Use the frame number as "timestamp" to identify frames
+ source_frame_._timeStamp = frame_number;
+
+ // Decide if we're going to force a keyframe:
+ VideoFrameType frame_type = kDeltaFrame;
+ if (config_.keyframe_interval > 0 &&
+ frame_number % config_.keyframe_interval == 0) {
+ frame_type = kKeyFrame;
+ }
+ WebRtc_Word32 encode_result = encoder_->Encode(source_frame_, NULL,
+ &frame_type);
+ if (encode_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to encode frame %d, return code: %d\n",
+ frame_number, encode_result);
+ }
+ stat.encode_return_code = encode_result;
+ return true;
+ } else {
+ return false; // we've reached the last frame
+ }
+}
+
+void VideoProcessorImpl::FrameEncoded(EncodedImage* encoded_image) {
+ TickTime encode_stop = TickTime::Now();
+ int frame_number = encoded_image->_timeStamp;
+ FrameStatistic& stat = stats_->stats_[frame_number];
+ stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
+ encode_stop);
+ stat.encoding_successful = true;
+ stat.encoded_frame_length_in_bytes = encoded_image->_length;
+ stat.frame_number = encoded_image->_timeStamp;
+ stat.frame_type = encoded_image->_frameType;
+ stat.bit_rate_in_kbps = encoded_image->_length * bit_rate_factor_;
+ stat.total_packets = encoded_image->_length /
+ config_.networking_config.packet_size_in_bytes + 1;
+
+ // Perform packet loss if criteria is fullfilled:
+ bool exclude_this_frame = false;
+ // Only keyframes can be excluded
+ if (encoded_image->_frameType == kKeyFrame) {
+ switch (config_.exclude_frame_types) {
+ case kExcludeOnlyFirstKeyFrame:
+ if (!first_key_frame_has_been_excluded_) {
+ first_key_frame_has_been_excluded_ = true;
+ exclude_this_frame = true;
+ }
+ break;
+ case kExcludeAllKeyFrames:
+ exclude_this_frame = true;
+ break;
+ default:
+ assert(false);
+ }
+ }
+ if (!exclude_this_frame) {
+ stat.packets_dropped =
+ packet_manipulator_->ManipulatePackets(encoded_image);
+ }
+
+ // Keep track of if frames are lost due to packet loss so we can tell
+ // this to the encoder (this is handled by the RTP logic in the full stack)
+ decode_start_ = TickTime::Now();
+ // TODO(kjellander): Pass fragmentation header to the decoder when
+ // CL 172001 has been submitted and PacketManipulator supports this.
+ WebRtc_Word32 decode_result = decoder_->Decode(*encoded_image,
+ last_frame_missing_, NULL);
+ stat.decode_return_code = decode_result;
+ if (decode_result != WEBRTC_VIDEO_CODEC_OK) {
+ // Write the last successful frame the output file to avoid getting it out
+ // of sync with the source file for SSIM and PSNR comparisons:
+ frame_writer_->WriteFrame(last_successful_frame_buffer_);
+ }
+ // save status for losses so we can inform the decoder for the next frame:
+ last_frame_missing_ = encoded_image->_length == 0;
+}
+
+void VideoProcessorImpl::FrameDecoded(const RawImage& image) {
+ TickTime decode_stop = TickTime::Now();
+ int frame_number = image._timeStamp;
+ // Report stats
+ FrameStatistic& stat = stats_->stats_[frame_number];
+ stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
+ decode_stop);
+ stat.decoding_successful = true;
+ // Update our copy of the last successful frame:
+ memcpy(last_successful_frame_buffer_, image._buffer, image._length);
+
+ bool write_success = frame_writer_->WriteFrame(image._buffer);
+ if (!write_success) {
+ fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
+ }
+}
+
+int VideoProcessorImpl::GetElapsedTimeMicroseconds(
+ const webrtc::TickTime& start, const webrtc::TickTime& stop) {
+ WebRtc_UWord64 encode_time = (stop - start).Microseconds();
+ assert(encode_time <
+ static_cast<unsigned int>(std::numeric_limits<int>::max()));
+ return static_cast<int>(encode_time);
+}
+
+const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e) {
+ switch (e) {
+ case kExcludeOnlyFirstKeyFrame:
+ return "ExcludeOnlyFirstKeyFrame";
+ case kExcludeAllKeyFrames:
+ return "ExcludeAllKeyFrames";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
+ switch (e) {
+ case kVideoCodecVP8:
+ return "VP8";
+ case kVideoCodecI420:
+ return "I420";
+ case kVideoCodecRED:
+ return "RED";
+ case kVideoCodecULPFEC:
+ return "ULPFEC";
+ case kVideoCodecUnknown:
+ return "Unknown";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+// Callbacks
+WebRtc_Word32
+VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
+ EncodedImage& encoded_image,
+ const webrtc::CodecSpecificInfo* codec_specific_info,
+ const webrtc::RTPFragmentationHeader* fragmentation) {
+ video_processor_->FrameEncoded(&encoded_image); // forward to parent class
+ return 0;
+}
+WebRtc_Word32
+VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
+ RawImage& image) {
+ video_processor_->FrameDecoded(image); // forward to parent class
+ return 0;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor.h b/trunk/src/modules/video_coding/codecs/test/videoprocessor.h
new file mode 100644
index 0000000..7c2c14f
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/videoprocessor.h
@@ -0,0 +1,229 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+
+#include <string>
+
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+#include "modules/video_coding/codecs/test/stats.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/frame_reader.h"
+#include "testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Defines which frame types shall be excluded from packet loss and when.
+enum ExcludeFrameTypes {
+ // Will exclude the first keyframe in the video sequence from packet loss.
+ // Following keyframes will be targeted for packet loss.
+ kExcludeOnlyFirstKeyFrame,
+ // Exclude all keyframes from packet loss, no matter where in the video
+ // sequence they occur.
+ kExcludeAllKeyFrames
+};
+// Returns a string representation of the enum value.
+const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
+
+// Test configuration for a test run
+struct TestConfig {
+ TestConfig()
+ : name(""), description(""), test_number(0),
+ input_filename(""), output_filename(""), output_dir("out"),
+ networking_config(), exclude_frame_types(kExcludeOnlyFirstKeyFrame),
+ frame_length_in_bytes(-1), use_single_core(false), keyframe_interval(0),
+ codec_settings(NULL), verbose(true) {
+ };
+
+ // Name of the test. This is purely metadata and does not affect
+ // the test in any way.
+ std::string name;
+
+ // More detailed description of the test. This is purely metadata and does
+ // not affect the test in any way.
+ std::string description;
+
+ // Number of this test. Useful if multiple runs of the same test with
+ // different configurations shall be managed.
+ int test_number;
+
+ // File to process for the test. This must be a video file in the YUV format.
+ std::string input_filename;
+
+ // File to write to during processing for the test. Will be a video file
+ // in the YUV format.
+ std::string output_filename;
+
+ // Path to the directory where encoded files will be put
+ // (absolute or relative to the executable). Default: "out".
+ std::string output_dir;
+
+ // Configurations related to networking.
+ NetworkingConfig networking_config;
+
+ // Decides how the packet loss simulations shall exclude certain frames
+ // from packet loss. Default: kExcludeOnlyFirstKeyFrame.
+ ExcludeFrameTypes exclude_frame_types;
+
+ // The length of a single frame of the input video file. This value is
+ // calculated out of the width and height according to the video format
+ // specification. Must be set before processing.
+ int frame_length_in_bytes;
+
+ // Force the encoder and decoder to use a single core for processing.
+ // Using a single core is necessary to get a deterministic behavior for the
+ // encoded frames - using multiple cores will produce different encoded frames
+ // since multiple cores are competing to consume the byte budget for each
+ // frame in parallel.
+ // If set to false, the maximum number of available cores will be used.
+ // Default: false.
+ bool use_single_core;
+
+ // If set to a value >0 this setting forces the encoder to create a keyframe
+ // every Nth frame. Note that the encoder may create a keyframe in other
+ // locations in addition to the interval that is set using this parameter.
+ // Forcing key frames may also affect encoder planning optimizations in
+ // a negative way, since it will suddenly be forced to produce an expensive
+ // key frame.
+ // Default: 0.
+ int keyframe_interval;
+
+ // The codec settings to use for the test (target bitrate, video size,
+ // framerate and so on). This struct must be created and filled in using
+ // the VideoCodingModule::Codec() method.
+ webrtc::VideoCodec* codec_settings;
+
+ // If printing of information to stdout shall be performed during processing.
+ bool verbose;
+};
+
+// Returns a string representation of the enum value.
+const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
+
+// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
+// interfaces. This is done in a sequential manner in order to be able to
+// measure times properly.
+// The class processes a frame at the time for the configured input file.
+// It maintains state of where in the source input file the processing is at.
+//
+// Regarding packet loss: Note that keyframes are excluded (first or all
+// depending on the ExcludeFrameTypes setting). This is because if key frames
+// would be altered, all the following delta frames would be pretty much
+// worthless. VP8 has an error-resilience feature that makes it able to handle
+// packet loss in key non-first keyframes, which is why only the first is
+// excluded by default.
+// Packet loss in such important frames is handled on a higher level in the
+// Video Engine, where signaling would request a retransmit of the lost packets,
+// since they're so important.
+//
+// Note this class is not thread safe in any way and is meant for simple testing
+// purposes.
+class VideoProcessor {
+ public:
+ virtual ~VideoProcessor() {}
+
+ // Performs initial calculations about frame size, sets up callbacks etc.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Processes a single frame. Returns true as long as there's more frames
+ // available in the source clip.
+ // Frame number must be an integer >=0.
+ virtual bool ProcessFrame(int frame_number) = 0;
+};
+
+class VideoProcessorImpl : public VideoProcessor {
+ public:
+ VideoProcessorImpl(webrtc::VideoEncoder* encoder,
+ webrtc::VideoDecoder* decoder,
+ FrameReader* frame_reader,
+ FrameWriter* frame_writer,
+ PacketManipulator* packet_manipulator,
+ const TestConfig& config,
+ Stats* stats);
+ virtual ~VideoProcessorImpl();
+ virtual bool Init();
+ virtual bool ProcessFrame(int frame_number);
+
+ private:
+ // Invoked by the callback when a frame has completed encoding.
+ void FrameEncoded(EncodedImage* encodedImage);
+ // Invoked by the callback when a frame has completed decoding.
+ void FrameDecoded(const RawImage& image);
+ // Used for getting a 32-bit integer representing time
+ // (checks the size is within signed 32-bit bounds before casting it)
+ int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
+ const webrtc::TickTime& stop);
+
+ webrtc::VideoEncoder* encoder_;
+ webrtc::VideoDecoder* decoder_;
+ FrameReader* frame_reader_;
+ FrameWriter* frame_writer_;
+ PacketManipulator* packet_manipulator_;
+ const TestConfig& config_;
+ Stats* stats_;
+
+ EncodedImageCallback* encode_callback_;
+ DecodedImageCallback* decode_callback_;
+ // Buffer used for reading the source video file:
+ WebRtc_UWord8* source_buffer_;
+ // Keep track of the last successful frame, since we need to write that
+ // when decoding fails:
+ WebRtc_UWord8* last_successful_frame_buffer_;
+ webrtc::RawImage source_frame_;
+ // To keep track of if we have excluded the first key frame from packet loss:
+ bool first_key_frame_has_been_excluded_;
+ // To tell the decoder previous frame have been dropped due to packet loss:
+ bool last_frame_missing_;
+ // If Init() has executed successfully.
+ bool initialized_;
+
+ // Statistics
+ double bit_rate_factor_; // multiply frame length with this to get bit rate
+ webrtc::TickTime encode_start_;
+ webrtc::TickTime decode_start_;
+
+ // Callback class required to implement according to the VideoEncoder API.
+ class VideoProcessorEncodeCompleteCallback
+ : public webrtc::EncodedImageCallback {
+ public:
+ explicit VideoProcessorEncodeCompleteCallback(VideoProcessorImpl* vp)
+ : video_processor_(vp) {
+ }
+ WebRtc_Word32 Encoded(
+ webrtc::EncodedImage& encoded_image,
+ const webrtc::CodecSpecificInfo* codec_specific_info = NULL,
+ const webrtc::RTPFragmentationHeader* fragmentation = NULL);
+
+ private:
+ VideoProcessorImpl* video_processor_;
+ };
+
+ // Callback class required to implement according to the VideoDecoder API.
+ class VideoProcessorDecodeCompleteCallback
+ : public webrtc::DecodedImageCallback {
+ public:
+ explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
+ : video_processor_(vp) {
+ }
+ WebRtc_Word32 Decoded(webrtc::RawImage& image);
+
+ private:
+ VideoProcessorImpl* video_processor_;
+ };
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
new file mode 100644
index 0000000..19c40ad
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -0,0 +1,173 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+#include "modules/video_coding/codecs/vp8/main/interface/vp8.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/frame_reader.h"
+#include "testsupport/frame_writer.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "testsupport/packet_reader.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+const int kNbrFrames = 61; // foreman_cif_short.yuv
+const int kCIFWidth = 352;
+const int kCIFHeight = 288;
+const int kBitRateKbps = 500;
+
+// Integration test for video processor. Encodes+decodes a small clip and
+// writes it to the output directory. After completion, PSNR and SSIM
+// measurements are performed on the original and the processed clip to verify
+// the quality is acceptable.
+// The limits for the PSNR and SSIM values must be set quite low, since we have
+// no control over the random function used for packet loss in this test.
+class VideoProcessorIntegrationTest: public testing::Test {
+ protected:
+ VideoEncoder* encoder_;
+ VideoDecoder* decoder_;
+ webrtc::test::FrameReader* frame_reader_;
+ webrtc::test::FrameWriter* frame_writer_;
+ webrtc::test::PacketReader packet_reader_;
+ webrtc::test::PacketManipulator* packet_manipulator_;
+ webrtc::test::Stats stats_;
+ webrtc::test::TestConfig config_;
+ VideoCodec codec_settings_;
+ webrtc::test::VideoProcessor* processor_;
+
+ VideoProcessorIntegrationTest() {}
+ virtual ~VideoProcessorIntegrationTest() {}
+
+ void SetUp() {
+ encoder_ = VP8Encoder::Create();
+ decoder_ = VP8Decoder::Create();
+
+ // Setup the TestConfig struct for processing of a clip in CIF resolution.
+ config_.input_filename =
+ webrtc::test::ResourcePath("foreman_cif_short", "yuv");
+ config_.output_filename = webrtc::test::OutputPath() +
+ "foreman_cif_short_video_codecs_test_framework_integrationtests.yuv";
+ config_.frame_length_in_bytes = 3 * kCIFWidth * kCIFHeight / 2;
+ config_.verbose = false;
+ // Only allow encoder/decoder to use single core, for predictability.
+ config_.use_single_core = true;
+
+ // Get a codec configuration struct and configure it.
+ VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
+ config_.codec_settings = &codec_settings_;
+ config_.codec_settings->startBitrate = kBitRateKbps;
+ config_.codec_settings->width = kCIFWidth;
+ config_.codec_settings->height = kCIFHeight;
+
+ frame_reader_ =
+ new webrtc::test::FrameReaderImpl(config_.input_filename,
+ config_.frame_length_in_bytes);
+ frame_writer_ =
+ new webrtc::test::FrameWriterImpl(config_.output_filename,
+ config_.frame_length_in_bytes);
+ ASSERT_TRUE(frame_reader_->Init());
+ ASSERT_TRUE(frame_writer_->Init());
+
+ packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
+ &packet_reader_, config_.networking_config, config_.verbose);
+ processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
+ frame_reader_,
+ frame_writer_,
+ packet_manipulator_,
+ config_, &stats_);
+ ASSERT_TRUE(processor_->Init());
+ }
+
+ void TearDown() {
+ delete processor_;
+ delete packet_manipulator_;
+ delete frame_writer_;
+ delete frame_reader_;
+ delete decoder_;
+ delete encoder_;
+ }
+
+ // Processes all frames in the clip and verifies the result.
+ void ProcessFramesAndVerify(double minimum_avg_psnr,
+ double minimum_min_psnr,
+ double minimum_avg_ssim,
+ double minimum_min_ssim) {
+ int frame_number = 0;
+ while (processor_->ProcessFrame(frame_number)) {
+ frame_number++;
+ }
+ EXPECT_EQ(kNbrFrames, frame_number);
+ EXPECT_EQ(kNbrFrames, static_cast<int>(stats_.stats_.size()));
+
+ // Release encoder and decoder to make sure they have finished processing:
+ EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+ EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
+ // Close the files before we start using them for SSIM/PSNR calculations.
+ frame_reader_->Close();
+ frame_writer_->Close();
+
+ webrtc::test::QualityMetricsResult psnr_result, ssim_result;
+ EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
+ config_.input_filename.c_str(),
+ config_.output_filename.c_str(),
+ config_.codec_settings->width,
+ config_.codec_settings->height,
+ &psnr_result,
+ &ssim_result));
+ printf("PSNR avg: %f, min: %f SSIM avg: %f, min: %f\n",
+ psnr_result.average, psnr_result.min,
+ ssim_result.average, ssim_result.min);
+ EXPECT_GT(psnr_result.average, minimum_avg_psnr);
+ EXPECT_GT(psnr_result.min, minimum_min_psnr);
+ EXPECT_GT(ssim_result.average, minimum_avg_ssim);
+ EXPECT_GT(ssim_result.min, minimum_min_ssim);
+ }
+};
+
+// Run with no packet loss. Quality should be very high.
+TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
+ config_.networking_config.packet_loss_probability = 0;
+ double minimum_avg_psnr = 36;
+ double minimum_min_psnr = 34;
+ double minimum_avg_ssim = 0.9;
+ double minimum_min_ssim = 0.9;
+ ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
+ minimum_avg_ssim, minimum_min_ssim);
+}
+
+// Run with 5% packet loss. Quality should be a bit lower.
+// TODO(mflodman): Reenable this once it's not flaky.
+TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
+ config_.networking_config.packet_loss_probability = 0.05;
+ double minimum_avg_psnr = 21;
+ double minimum_min_psnr = 17;
+ double minimum_avg_ssim = 0.6;
+ double minimum_min_ssim = 0.4;
+ ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
+ minimum_avg_ssim, minimum_min_ssim);
+}
+
+// Run with 10% packet loss. Quality should be even lower.
+TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
+ config_.networking_config.packet_loss_probability = 0.10;
+ double minimum_avg_psnr = 19;
+ double minimum_min_psnr = 16;
+ double minimum_avg_ssim = 0.6;
+ double minimum_min_ssim = 0.4;
+ ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
+ minimum_avg_ssim, minimum_min_ssim);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc
new file mode 100644
index 0000000..d51ef6b
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+#include "gmock/gmock.h"
+#include "modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "testsupport/mock/mock_frame_reader.h"
+#include "testsupport/mock/mock_frame_writer.h"
+#include "testsupport/packet_reader.h"
+#include "testsupport/unittest_utils.h"
+#include "typedefs.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Return;
+
+namespace webrtc {
+namespace test {
+
+// Very basic testing for VideoProcessor. It's mostly tested by running the
+// video_quality_measurement program.
+class VideoProcessorTest: public testing::Test {
+ protected:
+ MockVideoEncoder encoder_mock_;
+ MockVideoDecoder decoder_mock_;
+ MockFrameReader frame_reader_mock_;
+ MockFrameWriter frame_writer_mock_;
+ MockPacketManipulator packet_manipulator_mock_;
+ Stats stats_;
+ TestConfig config_;
+ VideoCodec codec_settings_;
+
+ VideoProcessorTest() {}
+ virtual ~VideoProcessorTest() {}
+ void SetUp() {
+ // Get a codec configuration struct and configure it.
+ VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
+ config_.codec_settings = &codec_settings_;
+ config_.codec_settings->startBitrate = 100;
+ config_.codec_settings->width = 352;
+ config_.codec_settings->height = 288;
+ }
+ void TearDown() {}
+
+ void ExpectInit() {
+ EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
+ .Times(1);
+ EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
+ .Times(AtLeast(1));
+ EXPECT_CALL(decoder_mock_, InitDecode(_, _))
+ .Times(1);
+ EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
+ .Times(AtLeast(1));
+ EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
+ .WillOnce(Return(1));
+ EXPECT_CALL(frame_reader_mock_, FrameLength())
+ .WillOnce(Return(150000));
+ }
+};
+
+TEST_F(VideoProcessorTest, Init) {
+ ExpectInit();
+ VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
+ &frame_reader_mock_,
+ &frame_writer_mock_,
+ &packet_manipulator_mock_, config_,
+ &stats_);
+ ASSERT_TRUE(video_processor.Init());
+}
+
+TEST_F(VideoProcessorTest, ProcessFrame) {
+ ExpectInit();
+ EXPECT_CALL(encoder_mock_, Encode(_, _, _))
+ .Times(1);
+ EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
+ .WillOnce(Return(true));
+ // Since we don't return any callback from the mock, the decoder will not
+ // be more than initialized...
+ VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
+ &frame_reader_mock_,
+ &frame_writer_mock_,
+ &packet_manipulator_mock_, config_,
+ &stats_);
+ ASSERT_TRUE(video_processor.Init());
+ video_processor.ProcessFrame(0);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc
new file mode 100644
index 0000000..883330c
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc
@@ -0,0 +1,310 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "benchmark.h"
+
+#include <cassert>
+#include <iostream>
+#include <sstream>
+#include <vector>
+#if defined(_WIN32)
+ #include <windows.h>
+#endif
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "modules/video_coding/codecs/test_framework/video_source.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/metrics/video_metrics.h"
+
+#define SSIM_CALC 0 // by default, don't compute SSIM
+
+using namespace webrtc;
+
+Benchmark::Benchmark()
+:
+NormalAsyncTest("Benchmark", "Codec benchmark over a range of test cases", 6),
+_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
+_codecName("Default")
+{
+}
+
+Benchmark::Benchmark(std::string name, std::string description)
+:
+NormalAsyncTest(name, description, 6),
+_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
+_codecName("Default")
+{
+}
+
+Benchmark::Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName)
+:
+NormalAsyncTest(name, description, 6),
+_resultsFileName(resultsFileName),
+_codecName(codecName)
+{
+}
+
+void
+Benchmark::Perform()
+{
+ std::vector<const VideoSource*> sources;
+ std::vector<const VideoSource*>::iterator it;
+
+ // Configuration --------------------------
+ sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
+ "resources/foreman_cif.yuv", kCIF));
+// sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
+// "resources/akiyo_cif.yuv", kCIF));
+
+ const VideoSize size[] = {kQCIF, kCIF};
+ const int frameRate[] = {10, 15, 30};
+ // Specifies the framerates for which to perform a speed test.
+ const bool speedTestMask[] = {false, false, false};
+ const int bitRate[] = {50, 100, 200, 300, 400, 500, 600, 1000};
+ // Determines the number of iterations to perform to arrive at the speed result.
+ enum { kSpeedTestIterations = 10 };
+ // ----------------------------------------
+
+ const int nFrameRates = sizeof(frameRate)/sizeof(*frameRate);
+ assert(sizeof(speedTestMask)/sizeof(*speedTestMask) == nFrameRates);
+ const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
+ int testIterations = 10;
+
+ webrtc::test::QualityMetricsResult psnr[nBitrates];
+ webrtc::test::QualityMetricsResult ssim[nBitrates];
+ double fps[nBitrates];
+ double totalEncodeTime[nBitrates];
+ double totalDecodeTime[nBitrates];
+
+ _results.open(_resultsFileName.c_str(), std::fstream::out);
+ _results << GetMagicStr() << std::endl;
+ _results << _codecName << std::endl;
+
+ for (it = sources.begin() ; it < sources.end(); it++)
+ {
+ for (int i = 0; i < static_cast<int>(sizeof(size)/sizeof(*size)); i++)
+ {
+ for (int j = 0; j < nFrameRates; j++)
+ {
+ std::stringstream ss;
+ std::string strFrameRate;
+ std::string outFileName;
+ ss << frameRate[j];
+ ss >> strFrameRate;
+ outFileName = (*it)->GetFilePath() + "/" + (*it)->GetName() + "_" +
+ VideoSource::GetSizeString(size[i]) + "_" + strFrameRate + ".yuv";
+
+ _target = new const VideoSource(outFileName, size[i], frameRate[j]);
+ (*it)->Convert(*_target);
+ if (VideoSource::FileExists(outFileName.c_str()))
+ {
+ _inname = outFileName;
+ }
+ else
+ {
+ _inname = (*it)->GetFileName();
+ }
+
+ std::cout << (*it)->GetName() << ", " << VideoSource::GetSizeString(size[i])
+ << ", " << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]:";
+ _results << (*it)->GetName() << "," << VideoSource::GetSizeString(size[i])
+ << "," << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]";
+
+ if (speedTestMask[j])
+ {
+ testIterations = kSpeedTestIterations;
+ }
+ else
+ {
+ testIterations = 1;
+ }
+
+ for (int k = 0; k < nBitrates; k++)
+ {
+ _bitRate = (bitRate[k]);
+ double avgFps = 0.0;
+ totalEncodeTime[k] = 0;
+ totalDecodeTime[k] = 0;
+
+ for (int l = 0; l < testIterations; l++)
+ {
+ PerformNormalTest();
+ _appendNext = false;
+
+ avgFps += _framecnt / (_totalEncodeTime + _totalDecodeTime);
+ totalEncodeTime[k] += _totalEncodeTime;
+ totalDecodeTime[k] += _totalDecodeTime;
+
+ }
+ avgFps /= testIterations;
+ totalEncodeTime[k] /= testIterations;
+ totalDecodeTime[k] /= testIterations;
+
+ double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
+ std::cout << " " << actualBitRate;
+ _results << "," << actualBitRate;
+ webrtc::test::QualityMetricsResult psnr_result;
+ I420PSNRFromFiles(_inname.c_str(), _outname.c_str(),
+ _inst.width, _inst.height, &psnr[k]);
+ if (SSIM_CALC)
+ {
+ webrtc::test::QualityMetricsResult ssim_result;
+ I420SSIMFromFiles(_inname.c_str(), _outname.c_str(),
+ _inst.width, _inst.height, &ssim[k]);
+
+ }
+ fps[k] = avgFps;
+ }
+ std::cout << std::endl << "Y-PSNR [dB]:";
+ _results << std::endl << "Y-PSNR [dB]";
+ for (int k = 0; k < nBitrates; k++)
+ {
+ std::cout << " " << psnr[k].average;
+ _results << "," << psnr[k].average;
+
+ }
+ if (SSIM_CALC)
+ {
+ std::cout << std::endl << "SSIM: ";
+ _results << std::endl << "SSIM ";
+ for (int k = 0; k < nBitrates; k++)
+ {
+ std::cout << " " << ssim[k].average;
+ _results << "," << ssim[k].average;
+ }
+
+ }
+
+ std::cout << std::endl << "Encode Time[ms]:";
+ _results << std::endl << "Encode Time[ms]";
+ for (int k = 0; k < nBitrates; k++)
+ {
+ std::cout << " " << totalEncodeTime[k];
+ _results << "," << totalEncodeTime[k];
+
+ }
+
+ std::cout << std::endl << "Decode Time[ms]:";
+ _results << std::endl << "Decode Time[ms]";
+ for (int k = 0; k < nBitrates; k++)
+ {
+ std::cout << " " << totalDecodeTime[k];
+ _results << "," << totalDecodeTime[k];
+
+ }
+
+ if (speedTestMask[j])
+ {
+ std::cout << std::endl << "Speed [fps]:";
+ _results << std::endl << "Speed [fps]";
+ for (int k = 0; k < nBitrates; k++)
+ {
+ std::cout << " " << static_cast<int>(fps[k] + 0.5);
+ _results << "," << static_cast<int>(fps[k] + 0.5);
+ }
+ }
+ std::cout << std::endl << std::endl;
+ _results << std::endl << std::endl;
+
+ delete _target;
+ }
+ }
+ delete *it;
+ }
+ _results.close();
+}
+
+void
+Benchmark::PerformNormalTest()
+{
+ _encoder = GetNewEncoder();
+ _decoder = GetNewDecoder();
+ CodecSettings(_target->GetWidth(), _target->GetHeight(), _target->GetFrameRate(), _bitRate);
+ Setup();
+ EventWrapper* waitEvent = EventWrapper::Create();
+
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _encoder->InitEncode(&_inst, 4, 1440);
+ CodecSpecific_InitBitrate();
+ _decoder->InitDecode(&_inst,1);
+
+ FrameQueue frameQueue;
+ VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+ VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
+ _encoder->RegisterEncodeCompleteCallback(&encCallback);
+ _decoder->RegisterDecodeCompleteCallback(&decCallback);
+
+ SetCodecSpecificParameters();
+
+ _totalEncodeTime = _totalDecodeTime = 0;
+ _totalEncodePipeTime = _totalDecodePipeTime = 0;
+ bool complete = false;
+ _framecnt = 0;
+ _encFrameCnt = 0;
+ _sumEncBytes = 0;
+ _lengthEncFrame = 0;
+ while (!complete)
+ {
+ complete = Encode();
+ if (!frameQueue.Empty() || complete)
+ {
+ while (!frameQueue.Empty())
+ {
+ _frameToDecode = static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+ DoPacketLoss();
+ int ret = Decode();
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ if (ret < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+ exit(EXIT_FAILURE);
+ }
+ else if (ret == 0)
+ {
+ _framecnt++;
+ }
+ else
+ {
+ fprintf(stderr, "\n\nPositive return value from decode!\n\n");
+ }
+ }
+ }
+ waitEvent->Wait(5);
+ }
+
+ _inputVideoBuffer.Free();
+ //_encodedVideoBuffer.Reset(); ?
+ _encodedVideoBuffer.Free();
+ _decodedVideoBuffer.Free();
+
+ _encoder->Release();
+ _decoder->Release();
+ delete waitEvent;
+ delete _encoder;
+ delete _decoder;
+ Teardown();
+}
+
+void
+Benchmark::CodecSpecific_InitBitrate()
+{
+ if (_bitRate == 0)
+ {
+ _encoder->SetRates(600, _inst.maxFramerate);
+ }
+ else
+ {
+ _encoder->SetRates(_bitRate, _inst.maxFramerate);
+ }
+}
+
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h
new file mode 100644
index 0000000..57806e5
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+
+#include "normal_async_test.h"
+
+class VideoSource;
+
+class Benchmark : public NormalAsyncTest
+{
+public:
+ Benchmark();
+ virtual void Perform();
+
+protected:
+ Benchmark(std::string name, std::string description);
+ Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName);
+ virtual webrtc::VideoEncoder* GetNewEncoder() = 0;
+ virtual webrtc::VideoDecoder* GetNewDecoder() = 0;
+ virtual void PerformNormalTest();
+ virtual void CodecSpecific_InitBitrate();
+ static const char* GetMagicStr() { return "#!benchmark1.0"; }
+
+ const VideoSource* _target;
+ std::string _resultsFileName;
+ std::ofstream _results;
+ std::string _codecName;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m b/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m
new file mode 100644
index 0000000..d0d5ed9
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m
@@ -0,0 +1,500 @@
+function exportfig(varargin)
+%EXPORTFIG Export a figure to Encapsulated Postscript.
+% EXPORTFIG(H, FILENAME) writes the figure H to FILENAME. H is
+% a figure handle and FILENAME is a string that specifies the
+% name of the output file.
+%
+% EXPORTFIG(...,PARAM1,VAL1,PARAM2,VAL2,...) specifies
+% parameters that control various characteristics of the output
+% file.
+%
+% Format Paramter:
+% 'Format' one of the strings 'eps','eps2','jpeg','png','preview'
+% specifies the output format. Defaults to 'eps'.
+% The output format 'preview' does not generate an output
+% file but instead creates a new figure window with a
+% preview of the exported figure. In this case the
+% FILENAME parameter is ignored.
+%
+% 'Preview' one of the strings 'none', 'tiff'
+% specifies a preview for EPS files. Defaults to 'none'.
+%
+% Size Parameters:
+% 'Width' a positive scalar
+% specifies the width in the figure's PaperUnits
+% 'Height' a positive scalar
+% specifies the height in the figure's PaperUnits
+%
+% Specifying only one dimension sets the other dimension
+% so that the exported aspect ratio is the same as the
+% figure's current aspect ratio.
+% If neither dimension is specified the size defaults to
+% the width and height from the figure's PaperPosition.
+%
+% Rendering Parameters:
+% 'Color' one of the strings 'bw', 'gray', 'cmyk'
+% 'bw' specifies that lines and text are exported in
+% black and all other objects in grayscale
+% 'gray' specifies that all objects are exported in grayscale
+% 'cmyk' specifies that all objects are exported in color
+% using the CMYK color space
+% 'Renderer' one of the strings 'painters', 'zbuffer', 'opengl'
+% specifies the renderer to use
+% 'Resolution' a positive scalar
+% specifies the resolution in dots-per-inch.
+%
+% The default color setting is 'bw'.
+%
+% Font Parameters:
+% 'FontMode' one of the strings 'scaled', 'fixed'
+% 'FontSize' a positive scalar
+% in 'scaled' mode multiplies with the font size of each
+% text object to obtain the exported font size
+% in 'fixed' mode specifies the font size of all text
+% objects in points
+% 'FontEncoding' one of the strings 'latin1', 'adobe'
+% specifies the character encoding of the font
+%
+% If FontMode is 'scaled' but FontSize is not specified then a
+% scaling factor is computed from the ratio of the size of the
+% exported figure to the size of the actual figure. The minimum
+% font size allowed after scaling is 5 points.
+% If FontMode is 'fixed' but FontSize is not specified then the
+% exported font sizes of all text objects is 7 points.
+%
+% The default 'FontMode' setting is 'scaled'.
+%
+% Line Width Parameters:
+% 'LineMode' one of the strings 'scaled', 'fixed'
+% 'LineWidth' a positive scalar
+% the semantics of LineMode and LineWidth are exactly the
+% same as FontMode and FontSize, except that they apply
+% to line widths instead of font sizes. The minumum line
+% width allowed after scaling is 0.5 points.
+% If LineMode is 'fixed' but LineWidth is not specified
+% then the exported line width of all line objects is 1
+% point.
+%
+% Examples:
+% exportfig(gcf,'fig1.eps','height',3);
+% Exports the current figure to the file named 'fig1.eps' with
+% a height of 3 inches (assuming the figure's PaperUnits is
+% inches) and an aspect ratio the same as the figure's aspect
+% ratio on screen.
+%
+% exportfig(gcf, 'fig2.eps', 'FontMode', 'fixed',...
+% 'FontSize', 10, 'color', 'cmyk' );
+% Exports the current figure to 'fig2.eps' in color with all
+% text in 10 point fonts. The size of the exported figure is
+% the figure's PaperPostion width and height.
+
+
+if (nargin < 2)
+ error('Too few input arguments');
+end
+
+% exportfig(H, filename, ...)
+H = varargin{1};
+if ~ishandle(H) | ~strcmp(get(H,'type'), 'figure')
+ error('First argument must be a handle to a figure.');
+end
+filename = varargin{2};
+if ~ischar(filename)
+ error('Second argument must be a string.');
+end
+paramPairs = varargin(3:end);
+
+% Do some validity checking on param-value pairs
+if (rem(length(paramPairs),2) ~= 0)
+ error(['Invalid input syntax. Optional parameters and values' ...
+ ' must be in pairs.']);
+end
+
+format = 'eps';
+preview = 'none';
+width = -1;
+height = -1;
+color = 'bw';
+fontsize = -1;
+fontmode='scaled';
+linewidth = -1;
+linemode=[];
+fontencoding = 'latin1';
+renderer = [];
+resolution = [];
+
+% Process param-value pairs
+args = {};
+for k = 1:2:length(paramPairs)
+ param = lower(paramPairs{k});
+ if (~ischar(param))
+ error('Optional parameter names must be strings');
+ end
+ value = paramPairs{k+1};
+
+ switch (param)
+ case 'format'
+ format = value;
+ if (~strcmp(format,{'eps','eps2','jpeg','png','preview'}))
+ error(['Format must be ''eps'', ''eps2'', ''jpeg'', ''png'' or' ...
+ ' ''preview''.']);
+ end
+ case 'preview'
+ preview = value;
+ if (~strcmp(preview,{'none','tiff'}))
+ error('Preview must be ''none'' or ''tiff''.');
+ end
+ case 'width'
+ width = LocalToNum(value);
+ if(~LocalIsPositiveScalar(width))
+ error('Width must be a numeric scalar > 0');
+ end
+ case 'height'
+ height = LocalToNum(value);
+ if(~LocalIsPositiveScalar(height))
+ error('Height must be a numeric scalar > 0');
+ end
+ case 'color'
+ color = lower(value);
+ if (~strcmp(color,{'bw','gray','cmyk'}))
+ error('Color must be ''bw'', ''gray'' or ''cmyk''.');
+ end
+ case 'fontmode'
+ fontmode = lower(value);
+ if (~strcmp(fontmode,{'scaled','fixed'}))
+ error('FontMode must be ''scaled'' or ''fixed''.');
+ end
+ case 'fontsize'
+ fontsize = LocalToNum(value);
+ if(~LocalIsPositiveScalar(fontsize))
+ error('FontSize must be a numeric scalar > 0');
+ end
+ case 'fontencoding'
+ fontencoding = lower(value);
+ if (~strcmp(fontencoding,{'latin1','adobe'}))
+ error('FontEncoding must be ''latin1'' or ''adobe''.');
+ end
+ case 'linemode'
+ linemode = lower(value);
+ if (~strcmp(linemode,{'scaled','fixed'}))
+ error('LineMode must be ''scaled'' or ''fixed''.');
+ end
+ case 'linewidth'
+ linewidth = LocalToNum(value);
+ if(~LocalIsPositiveScalar(linewidth))
+ error('LineWidth must be a numeric scalar > 0');
+ end
+ case 'renderer'
+ renderer = lower(value);
+ if (~strcmp(renderer,{'painters','zbuffer','opengl'}))
+ error('Renderer must be ''painters'', ''zbuffer'' or ''opengl''.');
+ end
+ case 'resolution'
+ resolution = LocalToNum(value);
+ if ~(isnumeric(value) & (prod(size(value)) == 1) & (value >= 0));
+ error('Resolution must be a numeric scalar >= 0');
+ end
+ otherwise
+ error(['Unrecognized option ' param '.']);
+ end
+end
+
+allLines = findall(H, 'type', 'line');
+allText = findall(H, 'type', 'text');
+allAxes = findall(H, 'type', 'axes');
+allImages = findall(H, 'type', 'image');
+allLights = findall(H, 'type', 'light');
+allPatch = findall(H, 'type', 'patch');
+allSurf = findall(H, 'type', 'surface');
+allRect = findall(H, 'type', 'rectangle');
+allFont = [allText; allAxes];
+allColor = [allLines; allText; allAxes; allLights];
+allMarker = [allLines; allPatch; allSurf];
+allEdge = [allPatch; allSurf];
+allCData = [allImages; allPatch; allSurf];
+
+old.objs = {};
+old.prop = {};
+old.values = {};
+
+% Process format and preview parameter
+showPreview = strcmp(format,'preview');
+if showPreview
+ format = 'png';
+ filename = [tempName '.png'];
+end
+if strncmp(format,'eps',3) & ~strcmp(preview,'none')
+ args = {args{:}, ['-' preview]};
+end
+
+hadError = 0;
+try
+ % Process size parameters
+ paperPos = get(H, 'PaperPosition');
+ old = LocalPushOldData(old, H, 'PaperPosition', paperPos);
+ figureUnits = get(H, 'Units');
+ set(H, 'Units', get(H,'PaperUnits'));
+ figurePos = get(H, 'Position');
+ aspectRatio = figurePos(3)/figurePos(4);
+ set(H, 'Units', figureUnits);
+ if (width == -1) & (height == -1)
+ width = paperPos(3);
+ height = paperPos(4);
+ elseif (width == -1)
+ width = height * aspectRatio;
+ elseif (height == -1)
+ height = width / aspectRatio;
+ end
+ set(H, 'PaperPosition', [0 0 width height]);
+ paperPosMode = get(H, 'PaperPositionMode');
+ old = LocalPushOldData(old, H, 'PaperPositionMode', paperPosMode);
+ set(H, 'PaperPositionMode', 'manual');
+
+ % Process rendering parameters
+ switch (color)
+ case {'bw', 'gray'}
+ if ~strcmp(color,'bw') & strncmp(format,'eps',3)
+ format = [format 'c'];
+ end
+ args = {args{:}, ['-d' format]};
+
+ %compute and set gray colormap
+ oldcmap = get(H,'Colormap');
+ newgrays = 0.30*oldcmap(:,1) + 0.59*oldcmap(:,2) + 0.11*oldcmap(:,3);
+ newcmap = [newgrays newgrays newgrays];
+ old = LocalPushOldData(old, H, 'Colormap', oldcmap);
+ set(H, 'Colormap', newcmap);
+
+ %compute and set ColorSpec and CData properties
+ old = LocalUpdateColors(allColor, 'color', old);
+ old = LocalUpdateColors(allAxes, 'xcolor', old);
+ old = LocalUpdateColors(allAxes, 'ycolor', old);
+ old = LocalUpdateColors(allAxes, 'zcolor', old);
+ old = LocalUpdateColors(allMarker, 'MarkerEdgeColor', old);
+ old = LocalUpdateColors(allMarker, 'MarkerFaceColor', old);
+ old = LocalUpdateColors(allEdge, 'EdgeColor', old);
+ old = LocalUpdateColors(allEdge, 'FaceColor', old);
+ old = LocalUpdateColors(allCData, 'CData', old);
+
+ case 'cmyk'
+ if strncmp(format,'eps',3)
+ format = [format 'c'];
+ args = {args{:}, ['-d' format], '-cmyk'};
+ else
+ args = {args{:}, ['-d' format]};
+ end
+ otherwise
+ error('Invalid Color parameter');
+ end
+ if (~isempty(renderer))
+ args = {args{:}, ['-' renderer]};
+ end
+ if (~isempty(resolution)) | ~strncmp(format,'eps',3)
+ if isempty(resolution)
+ resolution = 0;
+ end
+ args = {args{:}, ['-r' int2str(resolution)]};
+ end
+
+ % Process font parameters
+ if (~isempty(fontmode))
+ oldfonts = LocalGetAsCell(allFont,'FontSize');
+ switch (fontmode)
+ case 'fixed'
+ oldfontunits = LocalGetAsCell(allFont,'FontUnits');
+ old = LocalPushOldData(old, allFont, {'FontUnits'}, oldfontunits);
+ set(allFont,'FontUnits','points');
+ if (fontsize == -1)
+ set(allFont,'FontSize',7);
+ else
+ set(allFont,'FontSize',fontsize);
+ end
+ case 'scaled'
+ if (fontsize == -1)
+ wscale = width/figurePos(3);
+ hscale = height/figurePos(4);
+ scale = min(wscale, hscale);
+ else
+ scale = fontsize;
+ end
+ newfonts = LocalScale(oldfonts,scale,5);
+ set(allFont,{'FontSize'},newfonts);
+ otherwise
+ error('Invalid FontMode parameter');
+ end
+ % make sure we push the size after the units
+ old = LocalPushOldData(old, allFont, {'FontSize'}, oldfonts);
+ end
+ if strcmp(fontencoding,'adobe') & strncmp(format,'eps',3)
+ args = {args{:}, '-adobecset'};
+ end
+
+ % Process linewidth parameters
+ if (~isempty(linemode))
+ oldlines = LocalGetAsCell(allMarker,'LineWidth');
+ old = LocalPushOldData(old, allMarker, {'LineWidth'}, oldlines);
+ switch (linemode)
+ case 'fixed'
+ if (linewidth == -1)
+ set(allMarker,'LineWidth',1);
+ else
+ set(allMarker,'LineWidth',linewidth);
+ end
+ case 'scaled'
+ if (linewidth == -1)
+ wscale = width/figurePos(3);
+ hscale = height/figurePos(4);
+ scale = min(wscale, hscale);
+ else
+ scale = linewidth;
+ end
+ newlines = LocalScale(oldlines, scale, 0.5);
+ set(allMarker,{'LineWidth'},newlines);
+ otherwise
+ error('Invalid LineMode parameter');
+ end
+ end
+
+ % Export
+ print(H, filename, args{:});
+
+catch
+ hadError = 1;
+end
+
+% Restore figure settings
+for n=1:length(old.objs)
+ set(old.objs{n}, old.prop{n}, old.values{n});
+end
+
+if hadError
+ error(deblank(lasterr));
+end
+
+% Show preview if requested
+if showPreview
+ X = imread(filename,'png');
+ delete(filename);
+ f = figure( 'Name', 'Preview', ...
+ 'Menubar', 'none', ...
+ 'NumberTitle', 'off', ...
+ 'Visible', 'off');
+ image(X);
+ axis image;
+ ax = findobj(f, 'type', 'axes');
+ set(ax, 'Units', get(H,'PaperUnits'), ...
+ 'Position', [0 0 width height], ...
+ 'Visible', 'off');
+ set(ax, 'Units', 'pixels');
+ axesPos = get(ax,'Position');
+ figPos = get(f,'Position');
+ rootSize = get(0,'ScreenSize');
+ figPos(3:4) = axesPos(3:4);
+ if figPos(1) + figPos(3) > rootSize(3)
+ figPos(1) = rootSize(3) - figPos(3) - 50;
+ end
+ if figPos(2) + figPos(4) > rootSize(4)
+ figPos(2) = rootSize(4) - figPos(4) - 50;
+ end
+ set(f, 'Position',figPos, ...
+ 'Visible', 'on');
+end
+
+%
+% Local Functions
+%
+
+function outData = LocalPushOldData(inData, objs, prop, values)
+outData.objs = {inData.objs{:}, objs};
+outData.prop = {inData.prop{:}, prop};
+outData.values = {inData.values{:}, values};
+
+function cellArray = LocalGetAsCell(fig,prop);
+cellArray = get(fig,prop);
+if (~isempty(cellArray)) & (~iscell(cellArray))
+ cellArray = {cellArray};
+end
+
+function newArray = LocalScale(inArray, scale, minValue)
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+ newArray{k} = max(minValue,scale*inArray{k}(1));
+end
+
+function newArray = LocalMapToGray(inArray);
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+ color = inArray{k};
+ if (~isempty(color))
+ if ischar(color)
+ switch color(1)
+ case 'y'
+ color = [1 1 0];
+ case 'm'
+ color = [1 0 1];
+ case 'c'
+ color = [0 1 1];
+ case 'r'
+ color = [1 0 0];
+ case 'g'
+ color = [0 1 0];
+ case 'b'
+ color = [0 0 1];
+ case 'w'
+ color = [1 1 1];
+ case 'k'
+ color = [0 0 0];
+ otherwise
+ newArray{k} = color;
+ end
+ end
+ if ~ischar(color)
+ color = 0.30*color(1) + 0.59*color(2) + 0.11*color(3);
+ end
+ end
+ if isempty(color) | ischar(color)
+ newArray{k} = color;
+ else
+ newArray{k} = [color color color];
+ end
+end
+
+function newArray = LocalMapCData(inArray);
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+ color = inArray{k};
+ if (ndims(color) == 3) & isa(color,'double')
+ gray = 0.30*color(:,:,1) + 0.59*color(:,:,2) + 0.11*color(:,:,3);
+ color(:,:,1) = gray;
+ color(:,:,2) = gray;
+ color(:,:,3) = gray;
+ end
+ newArray{k} = color;
+end
+
+function outData = LocalUpdateColors(inArray, prop, inData)
+value = LocalGetAsCell(inArray,prop);
+outData.objs = {inData.objs{:}, inArray};
+outData.prop = {inData.prop{:}, {prop}};
+outData.values = {inData.values{:}, value};
+if (~isempty(value))
+ if strcmp(prop,'CData')
+ value = LocalMapCData(value);
+ else
+ value = LocalMapToGray(value);
+ end
+ set(inArray,{prop},value);
+end
+
+function bool = LocalIsPositiveScalar(value)
+bool = isnumeric(value) & ...
+ prod(size(value)) == 1 & ...
+ value > 0;
+
+function value = LocalToNum(value)
+if ischar(value)
+ value = str2num(value);
+end
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc
new file mode 100644
index 0000000..e379758
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc
@@ -0,0 +1,593 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_async_test.h"
+
+#include <assert.h>
+#include <string.h>
+#include <sstream>
+#include <queue>
+
+#include "gtest/gtest.h"
+#include "tick_util.h"
+#include "testsupport/fileutils.h"
+#include "typedefs.h"
+
+using namespace webrtc;
+
+NormalAsyncTest::NormalAsyncTest()
+:
+NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
+ _testNo),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_encFrameCnt(0),
+_decFrameCnt(0),
+_requestKeyFrame(false),
+_testNo(1),
+_appendNext(false),
+_missingFrames(false),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedRPSI(false),
+_hasReceivedPLI(false),
+_waitForKey(false)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(WebRtc_UWord32 bitRate)
+:
+NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
+ bitRate, _testNo),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_encFrameCnt(0),
+_decFrameCnt(0),
+_requestKeyFrame(false),
+_testNo(1),
+_appendNext(false),
+_missingFrames(false),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedRPSI(false),
+_hasReceivedPLI(false),
+_waitForKey(false)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+ unsigned int testNo)
+:
+NormalTest(name, description, _testNo),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_encFrameCnt(0),
+_decFrameCnt(0),
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_missingFrames(false),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedRPSI(false),
+_hasReceivedPLI(false),
+_waitForKey(false)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+ WebRtc_UWord32 bitRate, unsigned int testNo)
+:
+NormalTest(name, description, bitRate, _testNo),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_encFrameCnt(0),
+_decFrameCnt(0),
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_missingFrames(false),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedRPSI(false),
+_hasReceivedPLI(false),
+_waitForKey(false)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+ WebRtc_UWord32 bitRate, unsigned int testNo,
+ unsigned int rttFrames)
+:
+NormalTest(name, description, bitRate, _testNo),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_encFrameCnt(0),
+_decFrameCnt(0),
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_missingFrames(false),
+_rttFrames(rttFrames),
+_hasReceivedSLI(false),
+_hasReceivedRPSI(false),
+_hasReceivedPLI(false),
+_waitForKey(false)
+{
+}
+
+void
+NormalAsyncTest::Setup()
+{
+ Test::Setup();
+ std::stringstream ss;
+ std::string strTestNo;
+ ss << _testNo;
+ ss >> strTestNo;
+
+ // Check if settings exist. Otherwise use defaults.
+ if (_outname == "")
+ {
+ _outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
+ ".yuv";
+ }
+
+ if (_encodedName == "")
+ {
+ _encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
+ strTestNo + ".yuv";
+ }
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+
+ char mode[3] = "wb";
+ if (_appendNext)
+ {
+ strncpy(mode, "ab", 3);
+ }
+
+ if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
+ {
+ printf("Cannot write file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ _appendNext = true;
+}
+
+void
+NormalAsyncTest::Teardown()
+{
+ Test::Teardown();
+ fclose(_sourceFile);
+ fclose(_encodedFile);
+ fclose(_decodedFile);
+}
+
+FrameQueueTuple::~FrameQueueTuple()
+{
+ if (_codecSpecificInfo != NULL)
+ {
+ delete _codecSpecificInfo;
+ }
+ if (_frame != NULL)
+ {
+ delete _frame;
+ }
+}
+
+void FrameQueue::PushFrame(TestVideoEncodedBuffer *frame,
+ webrtc::CodecSpecificInfo* codecSpecificInfo)
+{
+ WriteLockScoped cs(_queueRWLock);
+ _frameBufferQueue.push(new FrameQueueTuple(frame, codecSpecificInfo));
+}
+
+FrameQueueTuple* FrameQueue::PopFrame()
+{
+ WriteLockScoped cs(_queueRWLock);
+ if (_frameBufferQueue.empty())
+ {
+ return NULL;
+ }
+ FrameQueueTuple* tuple = _frameBufferQueue.front();
+ _frameBufferQueue.pop();
+ return tuple;
+}
+
+bool FrameQueue::Empty()
+{
+ ReadLockScoped cs(_queueRWLock);
+ return _frameBufferQueue.empty();
+}
+
+WebRtc_UWord32 VideoEncodeCompleteCallback::EncodedBytes()
+{
+ return _encodedBytes;
+}
+
+WebRtc_Word32
+VideoEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo,
+ const webrtc::RTPFragmentationHeader*
+ fragmentation)
+{
+ _test.Encoded(encodedImage);
+ TestVideoEncodedBuffer *newBuffer = new TestVideoEncodedBuffer();
+ //newBuffer->VerifyAndAllocate(encodedImage._length);
+ newBuffer->VerifyAndAllocate(encodedImage._size);
+ _encodedBytes += encodedImage._length;
+ // If _frameQueue would have been a fixed sized buffer we could have asked
+ // it for an empty frame and then just do:
+ // emptyFrame->SwapBuffers(encodedBuffer);
+ // This is how it should be done in Video Engine to save in on memcpys
+ webrtc::CodecSpecificInfo* codecSpecificInfoCopy =
+ _test.CopyCodecSpecificInfo(codecSpecificInfo);
+ _test.CopyEncodedImage(*newBuffer, encodedImage, codecSpecificInfoCopy);
+ if (_encodedFile != NULL)
+ {
+ fwrite(newBuffer->GetBuffer(), 1, newBuffer->GetLength(), _encodedFile);
+ }
+ _frameQueue->PushFrame(newBuffer, codecSpecificInfoCopy);
+ return 0;
+}
+
+WebRtc_UWord32 VideoDecodeCompleteCallback::DecodedBytes()
+{
+ return _decodedBytes;
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::Decoded(RawImage& image)
+{
+ _test.Decoded(image);
+ _decodedBytes += image._length;
+ if (_decodedFile != NULL)
+ {
+ fwrite(image._buffer, 1, image._length, _decodedFile);
+ }
+ return 0;
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 pictureId)
+{
+ return _test.ReceivedDecodedReferenceFrame(pictureId);
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::ReceivedDecodedFrame(
+ const WebRtc_UWord64 pictureId)
+{
+ return _test.ReceivedDecodedFrame(pictureId);
+}
+
+void
+NormalAsyncTest::Encoded(const EncodedImage& encodedImage)
+{
+ _encodeCompleteTime = tGetTime();
+ _encFrameCnt++;
+ _totalEncodePipeTime += _encodeCompleteTime -
+ _encodeTimes[encodedImage._timeStamp];
+}
+
+void
+NormalAsyncTest::Decoded(const RawImage& decodedImage)
+{
+ _decodeCompleteTime = tGetTime();
+ _decFrameCnt++;
+ _totalDecodePipeTime += _decodeCompleteTime -
+ _decodeTimes[decodedImage._timeStamp];
+ _decodedWidth = decodedImage._width;
+ _decodedHeight = decodedImage._height;
+}
+
+void
+NormalAsyncTest::Perform()
+{
+ _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
+ CodecSettings(352, 288, 30, _bitRate);
+ Setup();
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ if(_encoder->InitEncode(&_inst, 1, 1440) < 0)
+ {
+ exit(EXIT_FAILURE);
+ }
+ _decoder->InitDecode(&_inst, 1);
+ FrameQueue frameQueue;
+ VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+ VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
+ _encoder->RegisterEncodeCompleteCallback(&encCallback);
+ _decoder->RegisterDecodeCompleteCallback(&decCallback);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+ _totalEncodeTime = _totalDecodeTime = 0;
+ _totalEncodePipeTime = _totalDecodePipeTime = 0;
+ bool complete = false;
+ _framecnt = 0;
+ _encFrameCnt = 0;
+ _decFrameCnt = 0;
+ _sumEncBytes = 0;
+ _lengthEncFrame = 0;
+ double starttime = tGetTime();
+ while (!complete)
+ {
+ CodecSpecific_InitBitrate();
+ complete = Encode();
+ if (!frameQueue.Empty() || complete)
+ {
+ while (!frameQueue.Empty())
+ {
+ _frameToDecode =
+ static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+ int lost = DoPacketLoss();
+ if (lost == 2)
+ {
+ // Lost the whole frame, continue
+ _missingFrames = true;
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ continue;
+ }
+ int ret = Decode(lost);
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ if (ret < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+ exit(EXIT_FAILURE);
+ }
+ else if (ret == 0)
+ {
+ _framecnt++;
+ }
+ else
+ {
+ fprintf(stderr,
+ "\n\nPositive return value from decode!\n\n");
+ }
+ }
+ }
+ }
+ double endtime = tGetTime();
+ double totalExecutionTime = endtime - starttime;
+ printf("Total execution time: %.1f s\n", totalExecutionTime);
+ _sumEncBytes = encCallback.EncodedBytes();
+ double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
+ double avgEncTime = _totalEncodeTime / _encFrameCnt;
+ double avgDecTime = _totalDecodeTime / _decFrameCnt;
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
+ printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
+ printf("Average encode pipeline time: %.1f ms\n",
+ 1000 * _totalEncodePipeTime / _encFrameCnt);
+ printf("Average decode pipeline time: %.1f ms\n",
+ 1000 * _totalDecodePipeTime / _decFrameCnt);
+ printf("Number of encoded frames: %u\n", _encFrameCnt);
+ printf("Number of decoded frames: %u\n", _decFrameCnt);
+ (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
+ _bitRate << " kbps" << std::endl;
+ (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+ _encoder->Release();
+ _decoder->Release();
+ Teardown();
+}
+
+bool
+NormalAsyncTest::Encode()
+{
+ _lengthEncFrame = 0;
+ EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp((unsigned int)
+ (_encFrameCnt * 9e4 / _inst.maxFramerate));
+ _inputVideoBuffer.SetWidth(_inst.width);
+ _inputVideoBuffer.SetHeight(_inst.height);
+ RawImage rawImage;
+ VideoBufferToRawImage(_inputVideoBuffer, rawImage);
+ if (feof(_sourceFile) != 0)
+ {
+ return true;
+ }
+ _encodeCompleteTime = 0;
+ _encodeTimes[rawImage._timeStamp] = tGetTime();
+ VideoFrameType frameType = kDeltaFrame;
+
+ // check SLI queue
+ _hasReceivedSLI = false;
+ while (!_signalSLI.empty() && _signalSLI.front().delay == 0)
+ {
+ // SLI message has arrived at sender side
+ _hasReceivedSLI = true;
+ _pictureIdSLI = _signalSLI.front().id;
+ _signalSLI.pop_front();
+ }
+ // decrement SLI queue times
+ for (std::list<fbSignal>::iterator it = _signalSLI.begin();
+ it !=_signalSLI.end(); it++)
+ {
+ (*it).delay--;
+ }
+
+ // check PLI queue
+ _hasReceivedPLI = false;
+ while (!_signalPLI.empty() && _signalPLI.front().delay == 0)
+ {
+ // PLI message has arrived at sender side
+ _hasReceivedPLI = true;
+ _signalPLI.pop_front();
+ }
+ // decrement PLI queue times
+ for (std::list<fbSignal>::iterator it = _signalPLI.begin();
+ it != _signalPLI.end(); it++)
+ {
+ (*it).delay--;
+ }
+
+ if (_hasReceivedPLI)
+ {
+ // respond to PLI by encoding a key frame
+ frameType = kKeyFrame;
+ _hasReceivedPLI = false;
+ _hasReceivedSLI = false; // don't trigger both at once
+ }
+
+ webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
+ int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
+ EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
+ if (codecSpecificInfo != NULL)
+ {
+ delete codecSpecificInfo;
+ codecSpecificInfo = NULL;
+ }
+ if (_encodeCompleteTime > 0)
+ {
+ _totalEncodeTime += _encodeCompleteTime -
+ _encodeTimes[rawImage._timeStamp];
+ }
+ else
+ {
+ _totalEncodeTime += tGetTime() - _encodeTimes[rawImage._timeStamp];
+ }
+ assert(ret >= 0);
+ return false;
+}
+
+int
+NormalAsyncTest::Decode(int lossValue)
+{
+ _sumEncBytes += _frameToDecode->_frame->GetLength();
+ EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+ encodedImage._completeFrame = !lossValue;
+ _decodeCompleteTime = 0;
+ _decodeTimes[encodedImage._timeStamp] = tGetTime();
+ int ret = WEBRTC_VIDEO_CODEC_OK;
+ if (!_waitForKey || encodedImage._frameType == kKeyFrame)
+ {
+ _waitForKey = false;
+ ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo);
+
+ if (ret >= 0)
+ {
+ _missingFrames = false;
+ }
+ }
+
+ // check for SLI
+ if (ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
+ {
+ // add an SLI feedback to the feedback "queue"
+ // to be delivered to encoder with _rttFrames delay
+ _signalSLI.push_back(fbSignal(_rttFrames,
+ static_cast<WebRtc_UWord8>((_lastDecPictureId) & 0x3f))); // 6 lsb
+
+ ret = WEBRTC_VIDEO_CODEC_OK;
+ }
+ else if (ret == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI)
+ {
+ // add an SLI feedback to the feedback "queue"
+ // to be delivered to encoder with _rttFrames delay
+ _signalSLI.push_back(fbSignal(_rttFrames,
+ static_cast<WebRtc_UWord8>((_lastDecPictureId + 1) & 0x3f)));//6 lsb
+
+ ret = WEBRTC_VIDEO_CODEC_OK;
+ }
+ else if (ret == WEBRTC_VIDEO_CODEC_ERROR)
+ {
+ // wait for new key frame
+ // add an PLI feedback to the feedback "queue"
+ // to be delivered to encoder with _rttFrames delay
+ _signalPLI.push_back(fbSignal(_rttFrames, 0 /* picId not used*/));
+ _waitForKey = true;
+
+ ret = WEBRTC_VIDEO_CODEC_OK;
+ }
+
+ if (_decodeCompleteTime > 0)
+ {
+ _totalDecodeTime += _decodeCompleteTime -
+ _decodeTimes[encodedImage._timeStamp];
+ }
+ else
+ {
+ _totalDecodeTime += tGetTime() - _decodeTimes[encodedImage._timeStamp];
+ }
+ return ret;
+}
+
+webrtc::CodecSpecificInfo*
+NormalAsyncTest::CopyCodecSpecificInfo(
+ const webrtc::CodecSpecificInfo* codecSpecificInfo) const
+{
+ webrtc::CodecSpecificInfo* info = new webrtc::CodecSpecificInfo;
+ *info = *codecSpecificInfo;
+ return info;
+}
+
+void NormalAsyncTest::CodecSpecific_InitBitrate()
+{
+ if (_bitRate == 0)
+ {
+ _encoder->SetRates(600, _inst.maxFramerate);
+ }
+ else
+ {
+ _encoder->SetRates(_bitRate, _inst.maxFramerate);
+ }
+}
+
+void NormalAsyncTest::CopyEncodedImage(TestVideoEncodedBuffer& dest,
+ EncodedImage& src,
+ void* /*codecSpecificInfo*/) const
+{
+ dest.CopyBuffer(src._length, src._buffer);
+ dest.SetFrameType(src._frameType);
+ dest.SetCaptureWidth((WebRtc_UWord16)src._encodedWidth);
+ dest.SetCaptureHeight((WebRtc_UWord16)src._encodedHeight);
+ dest.SetTimeStamp(src._timeStamp);
+}
+
+WebRtc_Word32 NormalAsyncTest::ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 pictureId) {
+ _lastDecRefPictureId = pictureId;
+ return 0;
+}
+
+WebRtc_Word32 NormalAsyncTest::ReceivedDecodedFrame(
+ const WebRtc_UWord64 pictureId) {
+ _lastDecPictureId = pictureId;
+ return 0;
+}
+
+double
+NormalAsyncTest::tGetTime()
+{// return time in sec
+ return ((double) (TickTime::MillisecondTimestamp())/1000);
+ }
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h
new file mode 100644
index 0000000..e520601
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h
@@ -0,0 +1,189 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
+
+#include "common_types.h"
+
+#include "normal_test.h"
+#include "rw_lock_wrapper.h"
+#include <list>
+#include <map>
+#include <queue>
+
+class FrameQueueTuple
+{
+public:
+ FrameQueueTuple(TestVideoEncodedBuffer *frame,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL)
+ :
+ _frame(frame),
+ _codecSpecificInfo(codecSpecificInfo)
+ {};
+ ~FrameQueueTuple();
+ TestVideoEncodedBuffer* _frame;
+ const webrtc::CodecSpecificInfo* _codecSpecificInfo;
+};
+
+class FrameQueue
+{
+public:
+ FrameQueue()
+ :
+ _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()),
+ _prevTS(-1)
+ {
+ }
+
+ ~FrameQueue()
+ {
+ delete &_queueRWLock;
+ }
+
+ void PushFrame(TestVideoEncodedBuffer *frame,
+ webrtc::CodecSpecificInfo* codecSpecificInfo = NULL);
+ FrameQueueTuple* PopFrame();
+ bool Empty();
+
+private:
+ webrtc::RWLockWrapper& _queueRWLock;
+ std::queue<FrameQueueTuple *> _frameBufferQueue;
+ WebRtc_Word64 _prevTS;
+};
+
+// feedback signal to encoder
+struct fbSignal
+{
+ fbSignal(int d, WebRtc_UWord8 pid) : delay(d), id(pid) {};
+ int delay;
+ WebRtc_UWord8 id;
+};
+
+class NormalAsyncTest : public NormalTest
+{
+public:
+ NormalAsyncTest();
+ NormalAsyncTest(WebRtc_UWord32 bitRate);
+ NormalAsyncTest(std::string name, std::string description,
+ unsigned int testNo);
+ NormalAsyncTest(std::string name, std::string description,
+ WebRtc_UWord32 bitRate, unsigned int testNo);
+ NormalAsyncTest(std::string name, std::string description,
+ WebRtc_UWord32 bitRate, unsigned int testNo,
+ unsigned int rttFrames);
+ virtual ~NormalAsyncTest() {};
+ virtual void Perform();
+ virtual void Encoded(const webrtc::EncodedImage& encodedImage);
+ virtual void Decoded(const webrtc::RawImage& decodedImage);
+ virtual webrtc::CodecSpecificInfo*
+ CopyCodecSpecificInfo(
+ const webrtc::CodecSpecificInfo* codecSpecificInfo) const;
+ virtual void CopyEncodedImage(TestVideoEncodedBuffer& dest,
+ webrtc::EncodedImage& src,
+ void* /*codecSpecificInfo*/) const;
+ virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const
+ {
+ return NULL;
+ };
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 pictureId);
+ virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
+
+protected:
+ virtual void Setup();
+ virtual void Teardown();
+ virtual bool Encode();
+ virtual int Decode(int lossValue = 0);
+ virtual void CodecSpecific_InitBitrate();
+ virtual int SetCodecSpecificParameters() {return 0;};
+ double tGetTime();// return time in sec
+
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ WebRtc_UWord32 _decodedWidth;
+ WebRtc_UWord32 _decodedHeight;
+ double _totalEncodeTime;
+ double _totalDecodeTime;
+ double _decodeCompleteTime;
+ double _encodeCompleteTime;
+ double _totalEncodePipeTime;
+ double _totalDecodePipeTime;
+ int _framecnt;
+ int _encFrameCnt;
+ int _decFrameCnt;
+ bool _requestKeyFrame;
+ unsigned int _testNo;
+ unsigned int _lengthEncFrame;
+ FrameQueueTuple* _frameToDecode;
+ bool _appendNext;
+ std::map<WebRtc_UWord32, double> _encodeTimes;
+ std::map<WebRtc_UWord32, double> _decodeTimes;
+ bool _missingFrames;
+ std::list<fbSignal> _signalSLI;
+ int _rttFrames;
+ mutable bool _hasReceivedSLI;
+ mutable bool _hasReceivedRPSI;
+ WebRtc_UWord8 _pictureIdSLI;
+ WebRtc_UWord16 _pictureIdRPSI;
+ WebRtc_UWord64 _lastDecRefPictureId;
+ WebRtc_UWord64 _lastDecPictureId;
+ std::list<fbSignal> _signalPLI;
+ bool _hasReceivedPLI;
+ bool _waitForKey;
+};
+
+class VideoEncodeCompleteCallback : public webrtc::EncodedImageCallback
+{
+public:
+ VideoEncodeCompleteCallback(FILE* encodedFile, FrameQueue *frameQueue,
+ NormalAsyncTest& test)
+ :
+ _encodedFile(encodedFile),
+ _frameQueue(frameQueue),
+ _test(test),
+ _encodedBytes(0)
+ {}
+
+ WebRtc_Word32
+ Encoded(webrtc::EncodedImage& encodedImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
+ const webrtc::RTPFragmentationHeader* fragmentation = NULL);
+ WebRtc_UWord32 EncodedBytes();
+private:
+ FILE* _encodedFile;
+ FrameQueue* _frameQueue;
+ NormalAsyncTest& _test;
+ WebRtc_UWord32 _encodedBytes;
+};
+
+class VideoDecodeCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+ VideoDecodeCompleteCallback(FILE* decodedFile, NormalAsyncTest& test)
+ :
+ _decodedFile(decodedFile),
+ _test(test),
+ _decodedBytes(0)
+ {}
+
+ virtual WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
+ virtual WebRtc_Word32
+ ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+ virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
+
+ WebRtc_UWord32 DecodedBytes();
+private:
+ FILE* _decodedFile;
+ NormalAsyncTest& _test;
+ WebRtc_UWord32 _decodedBytes;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc
new file mode 100644
index 0000000..2dd7bce
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_test.h"
+
+#include <time.h>
+#include <sstream>
+#include <string.h>
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+NormalTest::NormalTest()
+:
+Test("Normal Test 1", "A test of normal execution of the codec"),
+_testNo(1),
+_lengthEncFrame(0),
+_appendNext(false)
+{
+}
+
+NormalTest::NormalTest(std::string name, std::string description,
+ unsigned int testNo)
+:
+Test(name, description),
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false)
+{
+}
+
+NormalTest::NormalTest(std::string name, std::string description,
+ WebRtc_UWord32 bitRate, unsigned int testNo)
+:
+Test(name, description, bitRate),
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false)
+{
+}
+
+void
+NormalTest::Setup()
+{
+ Test::Setup();
+ std::stringstream ss;
+ std::string strTestNo;
+ ss << _testNo;
+ ss >> strTestNo;
+
+ // Check if settings exist. Otherwise use defaults.
+ if (_outname == "")
+ {
+ _outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
+ ".yuv";
+ }
+
+ if (_encodedName == "")
+ {
+ _encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
+ strTestNo + ".yuv";
+ }
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+
+ char mode[3] = "wb";
+ if (_appendNext)
+ {
+ strncpy(mode, "ab", 3);
+ }
+
+ if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
+ {
+ printf("Cannot write file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ _appendNext = true;
+}
+
+void
+NormalTest::Teardown()
+{
+ Test::Teardown();
+ fclose(_sourceFile);
+ fclose(_decodedFile);
+}
+
+void
+NormalTest::Perform()
+{
+ _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
+ CodecSettings(352, 288, 30, _bitRate);
+ Setup();
+
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+
+ _encoder->InitEncode(&_inst, 1, 1460);
+ CodecSpecific_InitBitrate();
+ _decoder->InitDecode(&_inst,1);
+
+ _totalEncodeTime = _totalDecodeTime = 0;
+ _framecnt = 0;
+ _sumEncBytes = 0;
+ _lengthEncFrame = 0;
+ int decodeLength = 0;
+ while (!Encode())
+ {
+ DoPacketLoss();
+ _encodedVideoBuffer.UpdateLength(_encodedVideoBuffer.GetLength());
+ fwrite(_encodedVideoBuffer.GetBuffer(), 1, _encodedVideoBuffer.GetLength(), _encodedFile);
+ decodeLength = Decode();
+ if (decodeLength < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
+ exit(EXIT_FAILURE);
+ }
+ fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
+ CodecSpecific_InitBitrate();
+ _framecnt++;
+ }
+
+ // Ensure we empty the decoding queue.
+ while (decodeLength > 0)
+ {
+ decodeLength = Decode();
+ if (decodeLength < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
+ exit(EXIT_FAILURE);
+ }
+ fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
+ }
+
+ double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
+ double avgEncTime = _totalEncodeTime / _framecnt;
+ double avgDecTime = _totalDecodeTime / _framecnt;
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Average encode time: %f s\n", avgEncTime);
+ printf("Average decode time: %f s\n", avgDecTime);
+ (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
+ (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+
+ _inputVideoBuffer.Free();
+ _encodedVideoBuffer.Reset();
+ _decodedVideoBuffer.Free();
+
+ _encoder->Release();
+ _decoder->Release();
+
+ Teardown();
+}
+
+bool
+NormalTest::Encode()
+{
+ _lengthEncFrame = 0;
+ EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
+ if (feof(_sourceFile) != 0)
+ {
+ return true;
+ }
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp(_framecnt);
+
+ // This multiple attempt ridiculousness is to accomodate VP7:
+ // 1. The wrapper can unilaterally reduce the framerate for low bitrates.
+ // 2. The codec inexplicably likes to reject some frames. Perhaps there
+ // is a good reason for this...
+ int encodingAttempts = 0;
+ double starttime = 0;
+ double endtime = 0;
+ while (_lengthEncFrame == 0)
+ {
+ starttime = clock()/(double)CLOCKS_PER_SEC;
+
+ _inputVideoBuffer.SetWidth(_inst.width);
+ _inputVideoBuffer.SetHeight(_inst.height);
+ //_lengthEncFrame = _encoder->Encode(_inputVideoBuffer, _encodedVideoBuffer, _frameInfo,
+ // _inst.frameRate, _requestKeyFrame && !(_framecnt%50));
+
+ endtime = clock()/(double)CLOCKS_PER_SEC;
+
+ _encodedVideoBuffer.SetCaptureHeight(_inst.height);
+ _encodedVideoBuffer.SetCaptureWidth(_inst.width);
+ if (_lengthEncFrame < 0)
+ {
+ (*_log) << "Error in encoder: " << _lengthEncFrame << std::endl;
+ fprintf(stderr,"\n\nError in encoder: %d\n\n", _lengthEncFrame);
+ exit(EXIT_FAILURE);
+ }
+ _sumEncBytes += _lengthEncFrame;
+
+ encodingAttempts++;
+ if (encodingAttempts > 50)
+ {
+ (*_log) << "Unable to encode frame: " << _framecnt << std::endl;
+ fprintf(stderr,"\n\nUnable to encode frame: %d\n\n", _framecnt);
+ exit(EXIT_FAILURE);
+ }
+ }
+ _totalEncodeTime += endtime - starttime;
+
+ if (encodingAttempts > 1)
+ {
+ (*_log) << encodingAttempts << " attempts required to encode frame: " <<
+ _framecnt + 1 << std::endl;
+ fprintf(stderr,"\n%d attempts required to encode frame: %d\n", encodingAttempts,
+ _framecnt + 1);
+ }
+
+ return false;
+}
+
+int
+NormalTest::Decode(int lossValue)
+{
+ _encodedVideoBuffer.SetWidth(_inst.width);
+ _encodedVideoBuffer.SetHeight(_inst.height);
+ int lengthDecFrame = 0;
+ //int lengthDecFrame = _decoder->Decode(_encodedVideoBuffer, _decodedVideoBuffer);
+ //_totalDecodeTime += (double)((clock()/(double)CLOCKS_PER_SEC) - starttime);
+ if (lengthDecFrame < 0)
+ {
+ return lengthDecFrame;
+ }
+ _encodedVideoBuffer.Reset();
+ _encodedVideoBuffer.UpdateLength(0);
+ return lengthDecFrame;
+}
+
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h
new file mode 100644
index 0000000..0dcd9c2
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+
+#include "test.h"
+
+class NormalTest : public Test
+{
+public:
+ NormalTest();
+ NormalTest(std::string name, std::string description, unsigned int testNo);
+ NormalTest(std::string name, std::string description, WebRtc_UWord32 bitRate, unsigned int testNo);
+ virtual ~NormalTest() {};
+ virtual void Perform();
+
+protected:
+ virtual void Setup();
+ virtual void Teardown();
+ virtual bool Encode();
+ virtual int Decode(int lossValue = 0);
+ virtual void CodecSpecific_InitBitrate()=0;
+ virtual int DoPacketLoss() {return 0;};
+
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ FILE* _encodedFile;
+ double _totalEncodeTime;
+ double _totalDecodeTime;
+ unsigned int _framecnt;
+ bool _requestKeyFrame;
+ unsigned int _testNo;
+ int _lengthEncFrame;
+ bool _appendNext;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc
new file mode 100644
index 0000000..c94fc9a
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc
@@ -0,0 +1,248 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet_loss_test.h"
+#include "video_source.h"
+#include <sstream>
+#include <cassert>
+#include <string.h>
+
+using namespace webrtc;
+
+PacketLossTest::PacketLossTest()
+:
+NormalAsyncTest("PacketLossTest", "Encode, remove lost packets, decode", 300,
+ 5),
+_lossRate(0.1),
+_lossProbability(0.1),
+_lastFrame(NULL),
+_lastFrameLength(0)
+{
+}
+
+PacketLossTest::PacketLossTest(std::string name, std::string description)
+:
+NormalAsyncTest(name, description, 300, 5),
+_lossRate(0.1),
+_lossProbability(0.1),
+_lastFrame(NULL),
+_lastFrameLength(0)
+{
+}
+
+PacketLossTest::PacketLossTest(std::string name, std::string description, double lossRate, bool useNack, unsigned int rttFrames /* = 0*/)
+:
+NormalAsyncTest(name, description, 300, 5, rttFrames),
+_lossRate(lossRate),
+_lastFrame(NULL),
+_lastFrameLength(0)
+{
+ assert(lossRate >= 0 && lossRate <= 1);
+ if (useNack)
+ {
+ _lossProbability = 0;
+ }
+ else
+ {
+ _lossProbability = lossRate;
+ }
+}
+
+void
+PacketLossTest::Encoded(const EncodedImage& encodedImage)
+{
+ // push timestamp to queue
+ _frameQueue.push_back(encodedImage._timeStamp);
+ NormalAsyncTest::Encoded(encodedImage);
+}
+
+void
+PacketLossTest::Decoded(const RawImage& decodedImage)
+{
+ // check the frame queue if any frames have gone missing
+ assert(!_frameQueue.empty()); // decoded frame is not in the queue
+ while(_frameQueue.front() < decodedImage._timeStamp)
+ {
+ // this frame is missing
+ // write previous decoded frame again (frame freeze)
+ if (_decodedFile && _lastFrame)
+ {
+ fwrite(_lastFrame, 1, _lastFrameLength, _decodedFile);
+ }
+
+ // remove frame from queue
+ _frameQueue.pop_front();
+ }
+ assert(_frameQueue.front() == decodedImage._timeStamp); // decoded frame is not in the queue
+
+ // pop the current frame
+ _frameQueue.pop_front();
+
+ // save image for future freeze-frame
+ if (_lastFrameLength < decodedImage._length)
+ {
+ if (_lastFrame) delete [] _lastFrame;
+
+ _lastFrame = new WebRtc_UWord8[decodedImage._length];
+ }
+ memcpy(_lastFrame, decodedImage._buffer, decodedImage._length);
+ _lastFrameLength = decodedImage._length;
+
+ NormalAsyncTest::Decoded(decodedImage);
+}
+
+void
+PacketLossTest::Teardown()
+{
+ if (_totalKept + _totalThrown > 0)
+ {
+ printf("Target packet loss rate: %.4f\n", _lossProbability);
+ printf("Actual packet loss rate: %.4f\n", (_totalThrown * 1.0f) / (_totalKept + _totalThrown));
+ printf("Channel rate: %.2f kbps\n",
+ 0.001 * 8.0 * _sumChannelBytes / ((_framecnt * 1.0f) / _inst.maxFramerate));
+ }
+ else
+ {
+ printf("No packet losses inflicted\n");
+ }
+
+ NormalAsyncTest::Teardown();
+}
+
+void
+PacketLossTest::Setup()
+{
+ const VideoSource source(_inname, _inst.width, _inst.height, _inst.maxFramerate);
+
+ std::stringstream ss;
+ std::string lossRateStr;
+ ss << _lossRate;
+ ss >> lossRateStr;
+ _encodedName = source.GetName() + "-" + lossRateStr;
+ _outname = "out-" + source.GetName() + "-" + lossRateStr;
+
+ if (_lossProbability != _lossRate)
+ {
+ _encodedName += "-nack";
+ _outname += "-nack";
+ }
+ _encodedName += ".vp8";
+ _outname += ".yuv";
+
+ _totalKept = 0;
+ _totalThrown = 0;
+ _sumChannelBytes = 0;
+
+ NormalAsyncTest::Setup();
+}
+
+void
+PacketLossTest::CodecSpecific_InitBitrate()
+{
+ assert(_bitRate > 0);
+ WebRtc_UWord32 simulatedBitRate;
+ if (_lossProbability != _lossRate)
+ {
+ // Simulating NACK
+ simulatedBitRate = WebRtc_UWord32(_bitRate / (1 + _lossRate));
+ }
+ else
+ {
+ simulatedBitRate = _bitRate;
+ }
+ int rtt = 0;
+ if (_inst.maxFramerate > 0)
+ rtt = _rttFrames * (1000 / _inst.maxFramerate);
+ _encoder->SetChannelParameters((WebRtc_UWord32)(_lossProbability * 255.0),
+ rtt);
+ _encoder->SetRates(simulatedBitRate, _inst.maxFramerate);
+}
+
+int PacketLossTest::DoPacketLoss()
+{
+ // Only packet loss for delta frames
+ if (_frameToDecode->_frame->GetLength() == 0 || _frameToDecode->_frame->GetFrameType() != kDeltaFrame)
+ {
+ _sumChannelBytes += _frameToDecode->_frame->GetLength();
+ return 0;
+ }
+ unsigned char *packet = NULL;
+ TestVideoEncodedBuffer newEncBuf;
+ newEncBuf.VerifyAndAllocate(_lengthSourceFrame);
+ _inBufIdx = 0;
+ _outBufIdx = 0;
+ int size = 1;
+ int kept = 0;
+ int thrown = 0;
+ while ((size = NextPacket(1500, &packet)) > 0)
+ {
+ if (!PacketLoss(_lossProbability, thrown))
+ {
+ InsertPacket(&newEncBuf, packet, size);
+ kept++;
+ }
+ else
+ {
+ // Use the ByteLoss function if you want to lose only
+ // parts of a packet, and not the whole packet.
+
+ //int size2 = ByteLoss(size, packet, 15);
+ thrown++;
+ //if (size2 != size)
+ //{
+ // InsertPacket(&newEncBuf, packet, size2);
+ //}
+ }
+ }
+ int lossResult = (thrown!=0); // 0 = no loss 1 = loss(es)
+ if (lossResult)
+ {
+ lossResult += (kept==0); // 2 = all lost = full frame
+ }
+ _frameToDecode->_frame->CopyBuffer(newEncBuf.GetLength(), newEncBuf.GetBuffer());
+ _sumChannelBytes += newEncBuf.GetLength();
+ _totalKept += kept;
+ _totalThrown += thrown;
+
+ return lossResult;
+ //printf("Threw away: %d out of %d packets\n", thrown, thrown + kept);
+ //printf("Encoded left: %d bytes\n", _encodedVideoBuffer.GetLength());
+}
+
+int PacketLossTest::NextPacket(int mtu, unsigned char **pkg)
+{
+ unsigned char *buf = _frameToDecode->_frame->GetBuffer();
+ *pkg = buf + _inBufIdx;
+ if (static_cast<long>(_frameToDecode->_frame->GetLength()) - _inBufIdx <= mtu)
+ {
+ int size = _frameToDecode->_frame->GetLength() - _inBufIdx;
+ _inBufIdx = _frameToDecode->_frame->GetLength();
+ return size;
+ }
+ _inBufIdx += mtu;
+ return mtu;
+}
+
+int PacketLossTest::ByteLoss(int size, unsigned char *pkg, int bytesToLose)
+{
+ return size;
+}
+
+void PacketLossTest::InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size)
+{
+ if (static_cast<long>(buf->GetSize()) - _outBufIdx < size)
+ {
+ printf("InsertPacket error!\n");
+ return;
+ }
+ memcpy(buf->GetBuffer() + _outBufIdx, pkg, size);
+ buf->UpdateLength(buf->GetLength() + size);
+ _outBufIdx += size;
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h
new file mode 100644
index 0000000..ea37681
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
+
+#include <list>
+
+#include "normal_async_test.h"
+
+class PacketLossTest : public NormalAsyncTest
+{
+public:
+ PacketLossTest();
+ virtual ~PacketLossTest() {if(_lastFrame) {delete [] _lastFrame; _lastFrame = NULL;}}
+ virtual void Encoded(const webrtc::EncodedImage& encodedImage);
+ virtual void Decoded(const webrtc::RawImage& decodedImage);
+protected:
+ PacketLossTest(std::string name, std::string description);
+ PacketLossTest(std::string name,
+ std::string description,
+ double lossRate,
+ bool useNack,
+ unsigned int rttFrames = 0);
+
+ virtual void Setup();
+ virtual void Teardown();
+ virtual void CodecSpecific_InitBitrate();
+ virtual int DoPacketLoss();
+ virtual int NextPacket(int size, unsigned char **pkg);
+ virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
+ virtual void InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size);
+ int _inBufIdx;
+ int _outBufIdx;
+
+ // When NACK is being simulated _lossProbabilty is zero,
+ // otherwise it is set equal to _lossRate.
+ // Desired channel loss rate.
+ double _lossRate;
+ // Probability used to simulate packet drops.
+ double _lossProbability;
+
+ int _totalKept;
+ int _totalThrown;
+ int _sumChannelBytes;
+ std::list<WebRtc_UWord32> _frameQueue;
+ WebRtc_UWord8* _lastFrame;
+ WebRtc_UWord32 _lastFrameLength;
+};
+
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc
new file mode 100644
index 0000000..250400c
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc
@@ -0,0 +1,296 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "performance_test.h"
+
+#include <assert.h>
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+#include "tick_util.h"
+
+using namespace webrtc;
+
+#define NUM_FRAMES 300
+
+PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate)
+:
+NormalAsyncTest(bitRate),
+_numCodecs(0),
+_tests(NULL),
+_encoders(NULL),
+_decoders(NULL),
+_threads(NULL),
+_rawImageLock(NULL),
+_encodeEvents(new EventWrapper*[1]),
+_stopped(true),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL)
+{
+}
+
+PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs)
+:
+NormalAsyncTest(bitRate),
+_numCodecs(numCodecs),
+_tests(new PerformanceTest*[_numCodecs]),
+_encoders(new VideoEncoder*[_numCodecs]),
+_decoders(new VideoDecoder*[_numCodecs]),
+_threads(new ThreadWrapper*[_numCodecs]),
+_rawImageLock(RWLockWrapper::CreateRWLock()),
+_encodeEvents(new EventWrapper*[_numCodecs]),
+_stopped(true),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL)
+{
+ for (int i=0; i < _numCodecs; i++)
+ {
+ _tests[i] = new PerformanceTest(bitRate);
+ _encodeEvents[i] = EventWrapper::Create();
+ }
+}
+
+PerformanceTest::~PerformanceTest()
+{
+ if (_encoders != NULL)
+ {
+ delete [] _encoders;
+ }
+ if (_decoders != NULL)
+ {
+ delete [] _decoders;
+ }
+ if (_tests != NULL)
+ {
+ delete [] _tests;
+ }
+ if (_threads != NULL)
+ {
+ delete [] _threads;
+ }
+ if (_rawImageLock != NULL)
+ {
+ delete _rawImageLock;
+ }
+ if (_encodeEvents != NULL)
+ {
+ delete [] _encodeEvents;
+ }
+}
+
+void
+PerformanceTest::Setup()
+{
+ _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
+ NormalAsyncTest::Setup(); // Setup input and output files
+ CodecSettings(352, 288, 30, _bitRate); // common to all codecs
+ for (int i=0; i < _numCodecs; i++)
+ {
+ _encoders[i] = CreateEncoder();
+ _decoders[i] = CreateDecoder();
+ if (_encoders[i] == NULL)
+ {
+ printf("Must create a codec specific test!\n");
+ exit(EXIT_FAILURE);
+ }
+ if(_encoders[i]->InitEncode(&_inst, 4, 1440) < 0)
+ {
+ exit(EXIT_FAILURE);
+ }
+ if (_decoders[i]->InitDecode(&_inst, 1))
+ {
+ exit(EXIT_FAILURE);
+ }
+ _tests[i]->SetEncoder(_encoders[i]);
+ _tests[i]->SetDecoder(_decoders[i]);
+ _tests[i]->_rawImageLock = _rawImageLock;
+ _encodeEvents[i]->Reset();
+ _tests[i]->_encodeEvents[0] = _encodeEvents[i];
+ _tests[i]->_inst = _inst;
+ _threads[i] = ThreadWrapper::CreateThread(PerformanceTest::RunThread, _tests[i]);
+ unsigned int id = 0;
+ _tests[i]->_stopped = false;
+ _threads[i]->Start(id);
+ }
+}
+
+void
+PerformanceTest::Perform()
+{
+ Setup();
+ EventWrapper& sleepEvent = *EventWrapper::Create();
+ const WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+ for (int i=0; i < NUM_FRAMES; i++)
+ {
+ {
+ // Read a new frame from file
+ WriteLockScoped imageLock(*_rawImageLock);
+ _lengthEncFrame = 0;
+ EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile),
+ 0u);
+ if (feof(_sourceFile) != 0)
+ {
+ rewind(_sourceFile);
+ }
+ _inputVideoBuffer.VerifyAndAllocate(_inst.width*_inst.height*3/2);
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp((unsigned int) (_encFrameCnt * 9e4 / static_cast<float>(_inst.maxFramerate)));
+ _inputVideoBuffer.SetWidth(_inst.width);
+ _inputVideoBuffer.SetHeight(_inst.height);
+ for (int i=0; i < _numCodecs; i++)
+ {
+ _tests[i]->_inputVideoBuffer.CopyPointer(_inputVideoBuffer);
+ _encodeEvents[i]->Set();
+ }
+ }
+ if (i < NUM_FRAMES - 1)
+ {
+ sleepEvent.Wait(33);
+ }
+ }
+ for (int i=0; i < _numCodecs; i++)
+ {
+ _tests[i]->_stopped = true;
+ _encodeEvents[i]->Set();
+ _threads[i]->Stop();
+ }
+ const WebRtc_UWord32 totalTime =
+ static_cast<WebRtc_UWord32>(TickTime::MillisecondTimestamp() - startTime);
+ printf("Total time: %u\n", totalTime);
+ delete &sleepEvent;
+ Teardown();
+}
+
+void PerformanceTest::Teardown()
+{
+ if (_encodeCompleteCallback != NULL)
+ {
+ delete _encodeCompleteCallback;
+ }
+ if (_decodeCompleteCallback != NULL)
+ {
+ delete _decodeCompleteCallback;
+ }
+ // main test only, all others have numCodecs = 0:
+ if (_numCodecs > 0)
+ {
+ WriteLockScoped imageLock(*_rawImageLock);
+ _inputVideoBuffer.Free();
+ NormalAsyncTest::Teardown();
+ }
+ for (int i=0; i < _numCodecs; i++)
+ {
+ _encoders[i]->Release();
+ delete _encoders[i];
+ _decoders[i]->Release();
+ delete _decoders[i];
+ _tests[i]->_inputVideoBuffer.ClearPointer();
+ _tests[i]->_rawImageLock = NULL;
+ _tests[i]->Teardown();
+ delete _tests[i];
+ delete _encodeEvents[i];
+ delete _threads[i];
+ }
+}
+
+bool
+PerformanceTest::RunThread(void* obj)
+{
+ PerformanceTest& test = *static_cast<PerformanceTest*>(obj);
+ return test.PerformSingleTest();
+}
+
+bool
+PerformanceTest::PerformSingleTest()
+{
+ if (_encodeCompleteCallback == NULL)
+ {
+ _encodeCompleteCallback = new VideoEncodeCompleteCallback(NULL, &_frameQueue, *this);
+ _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
+ }
+ if (_decodeCompleteCallback == NULL)
+ {
+ _decodeCompleteCallback = new VideoDecodeCompleteCallback(NULL, *this);
+ _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
+ }
+ (*_encodeEvents)->Wait(WEBRTC_EVENT_INFINITE); // The first event is used for every single test
+ CodecSpecific_InitBitrate();
+ bool complete = false;
+ {
+ ReadLockScoped imageLock(*_rawImageLock);
+ complete = Encode();
+ }
+ if (!_frameQueue.Empty() || complete)
+ {
+ while (!_frameQueue.Empty())
+ {
+ _frameToDecode = static_cast<FrameQueueTuple *>(_frameQueue.PopFrame());
+ int lost = DoPacketLoss();
+ if (lost == 2)
+ {
+ // Lost the whole frame, continue
+ _missingFrames = true;
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ continue;
+ }
+ int ret = Decode(lost);
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ if (ret < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+ return false;
+ }
+ else if (ret < 0)
+ {
+ fprintf(stderr, "\n\nPositive return value from decode!\n\n");
+ return false;
+ }
+ }
+ }
+ if (_stopped)
+ {
+ return false;
+ }
+ return true;
+}
+
+bool PerformanceTest::Encode()
+{
+ RawImage rawImage;
+ VideoBufferToRawImage(_inputVideoBuffer, rawImage);
+ VideoFrameType frameType = kDeltaFrame;
+ if (_requestKeyFrame && !(_encFrameCnt%50))
+ {
+ frameType = kKeyFrame;
+ }
+ webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
+ int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
+ EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
+ if (codecSpecificInfo != NULL)
+ {
+ delete codecSpecificInfo;
+ codecSpecificInfo = NULL;
+ }
+ assert(ret >= 0);
+ return false;
+}
+
+int PerformanceTest::Decode(int lossValue)
+{
+ EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+ encodedImage._completeFrame = !lossValue;
+ int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo);
+ _missingFrames = false;
+ return ret;
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h
new file mode 100644
index 0000000..d060832
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
+
+#include "normal_async_test.h"
+#include "thread_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "event_wrapper.h"
+
+class PerformanceTest : public NormalAsyncTest
+{
+public:
+ PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs);
+ virtual ~PerformanceTest();
+
+ virtual void Perform();
+ virtual void Print() {};
+
+protected:
+ PerformanceTest(WebRtc_UWord32 bitRate);
+ virtual void Setup();
+ virtual bool Encode();
+ virtual int Decode(int lossValue = 0);
+ virtual void Teardown();
+ static bool RunThread(void* obj);
+ bool PerformSingleTest();
+
+ virtual webrtc::VideoEncoder* CreateEncoder() const { return NULL; };
+ virtual webrtc::VideoDecoder* CreateDecoder() const { return NULL; };
+
+ WebRtc_UWord8 _numCodecs;
+ PerformanceTest** _tests;
+ webrtc::VideoEncoder** _encoders;
+ webrtc::VideoDecoder** _decoders;
+ webrtc::ThreadWrapper** _threads;
+ webrtc::RWLockWrapper* _rawImageLock;
+ webrtc::EventWrapper** _encodeEvents;
+ FrameQueue _frameQueue;
+ bool _stopped;
+ webrtc::EncodedImageCallback* _encodeCompleteCallback;
+ webrtc::DecodedImageCallback* _decodeCompleteCallback;
+ FILE* _outFile;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m b/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m
new file mode 100644
index 0000000..33c8eb6
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m
@@ -0,0 +1,427 @@
+function plotBenchmark(fileNames, export)
+%PLOTBENCHMARK Plots and exports video codec benchmarking results.
+% PLOTBENCHMARK(FILENAMES, EXPORT) parses the video codec benchmarking result
+% files given by the cell array of strings FILENAME. It plots the results and
+% optionally exports each plot to an appropriately named file.
+%
+% EXPORT parameter:
+% 'none' No file exports.
+% 'eps' Exports to eps files (default).
+% 'pdf' Exports to eps files and uses the command-line utility
+% epstopdf to obtain pdf files.
+%
+% Example:
+% plotBenchmark({'H264Benchmark.txt' 'LSVXBenchmark.txt'}, 'pdf')
+
+if (nargin < 1)
+ error('Too few input arguments');
+elseif (nargin < 2)
+ export = 'eps';
+end
+
+if ~iscell(fileNames)
+ if ischar(fileNames)
+ % one single file name as a string is ok
+ if size(fileNames,1) > 1
+ % this is a char matrix, not ok
+ error('First argument must not be a char matrix');
+ end
+ % wrap in a cell array
+ fileNames = {fileNames};
+ else
+ error('First argument must be a cell array of strings');
+ end
+end
+
+if ~ischar(export)
+ error('Second argument must be a string');
+end
+
+outpath = 'BenchmarkPlots';
+[status, errMsg] = mkdir(outpath);
+if status == 0
+ error(errMsg);
+end
+
+nCases = 0;
+testCases = [];
+% Read each test result file
+for fileIdx = 1:length(fileNames)
+ if ~isstr(fileNames{fileIdx})
+ error('First argument must be a cell array of strings');
+ end
+
+ fid = fopen(fileNames{fileIdx}, 'rt');
+ if fid == -1
+ error(['Unable to open ' fileNames{fileIdx}]);
+ end
+
+ version = '1.0';
+ if ~strcmp(fgetl(fid), ['#!benchmark' version])
+ fclose(fid);
+ error(['Requires benchmark file format version ' version]);
+ end
+
+ % Parse results file into testCases struct
+ codec = fgetl(fid);
+ tline = fgetl(fid);
+ while(tline ~= -1)
+ nCases = nCases + 1;
+
+ delim = strfind(tline, ',');
+ name = tline(1:delim(1)-1);
+ % Drop underscored suffix from name
+ underscore = strfind(name, '_');
+ if ~isempty(underscore)
+ name = name(1:underscore(1)-1);
+ end
+
+ resolution = tline(delim(1)+1:delim(2)-1);
+ frameRate = tline(delim(2)+1:end);
+
+ tline = fgetl(fid);
+ delim = strfind(tline, ',');
+ bitrateLabel = tline(1:delim(1)-1);
+ bitrate = sscanf(tline(delim(1):end),',%f');
+
+ tline = fgetl(fid);
+ delim = strfind(tline, ',');
+ psnrLabel = tline(1:delim(1)-1);
+ psnr = sscanf(tline(delim(1):end),',%f');
+
+
+ % Default data for the optional lines
+ speedLabel = 'Default';
+ speed = 0;
+ ssimLabel = 'Default';
+ ssim = 0;
+
+ tline = fgetl(fid);
+ delim = strfind(tline, ',');
+
+ while ~isempty(delim)
+ % More data
+ % Check type of data
+ if strncmp(lower(tline), 'speed', 5)
+ % Speed data included
+ speedLabel = tline(1:delim(1)-1);
+ speed = sscanf(tline(delim(1):end), ',%f');
+
+ tline = fgetl(fid);
+
+ elseif strncmp(lower(tline), 'encode time', 11)
+ % Encode and decode times included
+ % TODO: take care of the data
+
+ % pop two lines from file
+ tline = fgetl(fid);
+ tline = fgetl(fid);
+
+ elseif strncmp(tline, 'SSIM', 4)
+ % SSIM data included
+ ssimLabel = tline(1:delim(1)-1);
+ ssim = sscanf(tline(delim(1):end), ',%f');
+
+ tline = fgetl(fid);
+ end
+ delim = strfind(tline, ',');
+ end
+
+ testCases = [testCases struct('codec', codec, 'name', name, 'resolution', ...
+ resolution, 'frameRate', frameRate, 'bitrate', bitrate, 'psnr', psnr, ...
+ 'speed', speed, 'bitrateLabel', bitrateLabel, 'psnrLabel', psnrLabel, ...
+ 'speedLabel', speedLabel, ...
+ 'ssim', ssim, 'ssimLabel', ssimLabel)];
+
+ tline = fgetl(fid);
+ end
+
+ fclose(fid);
+end
+
+i = 0;
+casesPsnr = testCases;
+while ~isempty(casesPsnr)
+ i = i + 1;
+ casesPsnr = plotOnePsnr(casesPsnr, i, export, outpath);
+end
+
+casesSSIM = testCases;
+while ~isempty(casesSSIM)
+ i = i + 1;
+ casesSSIM = plotOneSSIM(casesSSIM, i, export, outpath);
+end
+
+casesSpeed = testCases;
+while ~isempty(casesSpeed)
+ if casesSpeed(1).speed == 0
+ casesSpeed = casesSpeed(2:end);
+ else
+ i = i + 1;
+ casesSpeed = plotOneSpeed(casesSpeed, i, export, outpath);
+ end
+end
+
+
+
+%%%%%%%%%%%%%%%%%%
+%% SUBFUNCTIONS %%
+%%%%%%%%%%%%%%%%%%
+
+function casesOut = plotOnePsnr(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+ if strcmp(cases(1).resolution, cases(i).resolution) & ...
+ strcmp(cases(1).frameRate, cases(i).frameRate)
+ plotIdx = [plotIdx i];
+ end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+ simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+ while ~isempty(simIndx)
+ diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
+ cases(i).psnr = cases(i).psnr(diffIndx);
+ cases(i).bitrate = cases(i).bitrate(diffIndx);
+ simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+ end
+end
+
+% Prepare figure with axis labels and so on
+hFig = figure(num);
+clf;
+hold on;
+grid on;
+axis([0 1100 20 50]);
+set(gca, 'XTick', 0:200:1000);
+set(gca, 'YTick', 20:10:60);
+xlabel(cases(1).bitrateLabel);
+ylabel(cases(1).psnrLabel);
+res = cases(1).resolution;
+frRate = cases(1).frameRate;
+title([res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+ i = i + 1;
+ [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'psnr', i, sequences, 1);
+
+ % Stored to generate the legend
+ hLines = [hLines ; hLine];
+ codecs = {codecs{:} codec};
+end
+legend(hLines, codecs, 4);
+hold off;
+
+if ~strcmp(export, 'none')
+ % Export figure to an eps file
+ res = stripws(res);
+ frRate = stripws(frRate);
+ exportName = [outpath '/psnr-' res '-' frRate];
+ exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+ % Use the epstopdf utility to convert to pdf
+ system(['epstopdf ' exportName '.eps']);
+end
+
+
+function casesOut = plotOneSSIM(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+ if strcmp(cases(1).resolution, cases(i).resolution) & ...
+ strcmp(cases(1).frameRate, cases(i).frameRate)
+ plotIdx = [plotIdx i];
+ end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+ simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+ while ~isempty(simIndx)
+ diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
+ cases(i).ssim = cases(i).ssim(diffIndx);
+ cases(i).bitrate = cases(i).bitrate(diffIndx);
+ simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+ end
+end
+
+% Prepare figure with axis labels and so on
+hFig = figure(num);
+clf;
+hold on;
+grid on;
+axis([0 1100 0.5 1]); % y-limit are set to 'auto' below
+set(gca, 'XTick', 0:200:1000);
+%set(gca, 'YTick', 20:10:60);
+xlabel(cases(1).bitrateLabel);
+ylabel(cases(1).ssimLabel);
+res = cases(1).resolution;
+frRate = cases(1).frameRate;
+title([res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+ i = i + 1;
+ [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'ssim', i, sequences, 1);
+
+ % Stored to generate the legend
+ hLines = [hLines ; hLine];
+ codecs = {codecs{:} codec};
+end
+%set(gca,'YLimMode','auto')
+set(gca,'YLim',[0.5 1])
+set(gca,'YScale','log')
+legend(hLines, codecs, 4);
+hold off;
+
+if ~strcmp(export, 'none')
+ % Export figure to an eps file
+ res = stripws(res);
+ frRate = stripws(frRate);
+ exportName = [outpath '/psnr-' res '-' frRate];
+ exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+ % Use the epstopdf utility to convert to pdf
+ system(['epstopdf ' exportName '.eps']);
+end
+
+
+function casesOut = plotOneSpeed(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+ if strcmp(cases(1).resolution, cases(i).resolution) & ...
+ strcmp(cases(1).frameRate, cases(i).frameRate) & ...
+ strcmp(cases(1).name, cases(i).name)
+ plotIdx = [plotIdx i];
+ end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+ simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
+ while ~isempty(simIndx)
+ diffIndx = setdiff(1:length(cases(i).psnr), simIndx);
+ cases(i).psnr = cases(i).psnr(diffIndx);
+ cases(i).speed = cases(i).speed(diffIndx);
+ simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
+ end
+end
+
+hFig = figure(num);
+clf;
+hold on;
+%grid on;
+xlabel(cases(1).psnrLabel);
+ylabel(cases(1).speedLabel);
+res = cases(1).resolution;
+name = cases(1).name;
+frRate = cases(1).frameRate;
+title([name ', ' res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+ i = i + 1;
+ [cases, hLine, codec, sequences] = plotOneCodec(cases, 'psnr', 'speed', i, sequences, 0);
+
+ % Stored to generate the legend
+ hLines = [hLines ; hLine];
+ codecs = {codecs{:} codec};
+end
+legend(hLines, codecs, 1);
+hold off;
+
+if ~strcmp(export, 'none')
+ % Export figure to an eps file
+ res = stripws(res);
+ frRate = stripws(frRate);
+ exportName = [outpath '/speed-' name '-' res '-' frRate];
+ exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+ % Use the epstopdf utility to convert to pdf
+ system(['epstopdf ' exportName '.eps']);
+end
+
+
+function [casesOut, hLine, codec, sequences] = plotOneCodec(cases, xfield, yfield, num, sequences, annotatePlot)
+plotStr = {'gx-', 'bo-', 'r^-', 'kd-', 'cx-', 'go--', 'b^--'};
+% Find matching codecs
+plotIdx = 1;
+for i = 2:length(cases)
+ if strcmp(cases(1).codec, cases(i).codec)
+ plotIdx = [plotIdx i];
+ end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+for i = 1:length(cases)
+ % Plot a single case
+ hLine = plot(getfield(cases(i), xfield), getfield(cases(i), yfield), plotStr{num}, ...
+ 'LineWidth', 1.1, 'MarkerSize', 6);
+end
+
+% hLine handle and codec are returned to construct the legend afterwards
+codec = cases(1).codec;
+
+if annotatePlot == 0
+ return;
+end
+
+for i = 1:length(cases)
+ % Print the codec name as a text label
+ % Ensure each codec is only printed once
+ sequencePlotted = 0;
+ for j = 1:length(sequences)
+ if strcmp(cases(i).name, sequences{j})
+ sequencePlotted = 1;
+ break;
+ end
+ end
+
+ if sequencePlotted == 0
+ text(getfield(cases(i), xfield, {1}), getfield(cases(i), yfield, {1}), ...
+ [' ' cases(i).name]);
+ sequences = {sequences{:} cases(i).name};
+ end
+end
+
+
+% Strip whitespace from string
+function str = stripws(str)
+if ~isstr(str)
+ error('String required');
+end
+str = str(setdiff(1:length(str), find(isspace(str) == 1)));
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test.cc b/trunk/src/modules/video_coding/codecs/test_framework/test.cc
new file mode 100644
index 0000000..3eb6935
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/test.cc
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test.h"
+
+#include <cstring>
+#include <iostream>
+
+#include "testsupport/metrics/video_metrics.h"
+
+using namespace webrtc;
+
+long filesize(const char *filename); // local function defined at end of file
+
+Test::Test(std::string name, std::string description)
+:
+_bitRate(0),
+_inname(""),
+_outname(""),
+_encodedName(""),
+_name(name),
+_description(description)
+{
+ memset(&_inst, 0, sizeof(_inst));
+ unsigned int seed = static_cast<unsigned int>(0);
+ std::srand(seed);
+}
+
+Test::Test(std::string name, std::string description, WebRtc_UWord32 bitRate)
+:
+_bitRate(bitRate),
+_inname(""),
+_outname(""),
+_encodedName(""),
+_name(name),
+_description(description)
+{
+ memset(&_inst, 0, sizeof(_inst));
+ unsigned int seed = static_cast<unsigned int>(0);
+ std::srand(seed);
+}
+
+void
+Test::Print()
+{
+ std::cout << _name << " completed!" << std::endl;
+ (*_log) << _name << std::endl;
+ (*_log) << _description << std::endl;
+ (*_log) << "Input file: " << _inname << std::endl;
+ (*_log) << "Output file: " << _outname << std::endl;
+ webrtc::test::QualityMetricsResult psnr;
+ webrtc::test::QualityMetricsResult ssim;
+ I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
+ _inst.height, &psnr);
+ I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
+ _inst.height, &ssim);
+
+ (*_log) << "PSNR: " << psnr.average << std::endl;
+ std::cout << "PSNR: " << psnr.average << std::endl << std::endl;
+ (*_log) << "SSIM: " << ssim.average << std::endl;
+ std::cout << "SSIM: " << ssim.average << std::endl << std::endl;
+ (*_log) << std::endl;
+}
+
+void
+Test::Setup()
+{
+ int widhei = _inst.width*_inst.height;
+ _lengthSourceFrame = 3*widhei/2;
+ _sourceBuffer = new unsigned char[_lengthSourceFrame];
+}
+
+void
+Test::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
+{
+ if (bitRate > 0)
+ {
+ _bitRate = bitRate;
+ }
+ else if (_bitRate == 0)
+ {
+ _bitRate = 600;
+ }
+ _inst.codecType = kVideoCodecVP8;
+ _inst.codecSpecific.VP8.feedbackModeOn = true;
+ _inst.maxFramerate = (unsigned char)frameRate;
+ _inst.startBitrate = (int)_bitRate;
+ _inst.maxBitrate = 8000;
+ _inst.width = width;
+ _inst.height = height;
+}
+
+void
+Test::Teardown()
+{
+ delete [] _sourceBuffer;
+}
+
+void
+Test::SetEncoder(webrtc::VideoEncoder*encoder)
+{
+ _encoder = encoder;
+}
+
+void
+Test::SetDecoder(VideoDecoder*decoder)
+{
+ _decoder = decoder;
+}
+
+void
+Test::SetLog(std::fstream* log)
+{
+ _log = log;
+}
+
+double Test::ActualBitRate(int nFrames)
+{
+ return 8.0 * _sumEncBytes / (nFrames / _inst.maxFramerate);
+}
+
+bool Test::PacketLoss(double lossRate, int /*thrown*/)
+{
+ return RandUniform() < lossRate;
+}
+
+void
+Test::VideoBufferToRawImage(TestVideoBuffer& videoBuffer, RawImage &image)
+{
+ image._buffer = videoBuffer.GetBuffer();
+ image._size = videoBuffer.GetSize();
+ image._length = videoBuffer.GetLength();
+ image._width = videoBuffer.GetWidth();
+ image._height = videoBuffer.GetHeight();
+ image._timeStamp = videoBuffer.GetTimeStamp();
+}
+void
+Test::VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer, EncodedImage &image)
+{
+ image._buffer = videoBuffer.GetBuffer();
+ image._length = videoBuffer.GetLength();
+ image._size = videoBuffer.GetSize();
+ image._frameType = static_cast<VideoFrameType>(videoBuffer.GetFrameType());
+ image._timeStamp = videoBuffer.GetTimeStamp();
+ image._encodedWidth = videoBuffer.GetCaptureWidth();
+ image._encodedHeight = videoBuffer.GetCaptureHeight();
+ image._completeFrame = true;
+}
+
+long filesize(const char *filename)
+{
+ FILE *f = fopen(filename,"rb"); /* open the file in read only */
+ long size = 0;
+ if (fseek(f,0,SEEK_END)==0) /* seek was successful */
+ size = ftell(f);
+ fclose(f);
+ return size;
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test.h b/trunk/src/modules/video_coding/codecs/test_framework/test.h
new file mode 100644
index 0000000..a31868b
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/test.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
+
+#include "video_codec_interface.h"
+#include "video_buffer.h"
+#include <string>
+#include <fstream>
+#include <cstdlib>
+
+class Test
+{
+public:
+ Test(std::string name, std::string description);
+ Test(std::string name, std::string description, WebRtc_UWord32 bitRate);
+ virtual ~Test() {};
+ virtual void Perform()=0;
+ virtual void Print();
+ void SetEncoder(webrtc::VideoEncoder *encoder);
+ void SetDecoder(webrtc::VideoDecoder *decoder);
+ void SetLog(std::fstream* log);
+
+protected:
+ virtual void Setup();
+ virtual void CodecSettings(int width,
+ int height,
+ WebRtc_UWord32 frameRate=30,
+ WebRtc_UWord32 bitRate=0);
+ virtual void Teardown();
+ double ActualBitRate(int nFrames);
+ virtual bool PacketLoss(double lossRate, int /*thrown*/);
+ static double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
+ static void VideoBufferToRawImage(TestVideoBuffer& videoBuffer,
+ webrtc::RawImage &image);
+ static void VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer,
+ webrtc::EncodedImage &image);
+
+ webrtc::VideoEncoder* _encoder;
+ webrtc::VideoDecoder* _decoder;
+ WebRtc_UWord32 _bitRate;
+ unsigned int _lengthSourceFrame;
+ unsigned char* _sourceBuffer;
+ TestVideoBuffer _inputVideoBuffer;
+ TestVideoEncodedBuffer _encodedVideoBuffer;
+ TestVideoBuffer _decodedVideoBuffer;
+ webrtc::VideoCodec _inst;
+ std::fstream* _log;
+ std::string _inname;
+ std::string _outname;
+ std::string _encodedName;
+ int _sumEncBytes;
+
+private:
+ std::string _name;
+ std::string _description;
+
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi b/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi
new file mode 100644
index 0000000..3639e9a
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi
@@ -0,0 +1,71 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'test_framework',
+ 'type': '<(library)',
+
+ 'dependencies': [
+ '<(webrtc_root)/../test/metrics.gyp:metrics',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+
+ 'include_dirs': [
+ '../interface',
+ '<(webrtc_root)/../testing/gtest/include',
+ '../../../../common_video/interface',
+ ],
+
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ ],
+ },
+
+ 'sources': [
+ # header files
+ 'benchmark.h',
+ 'normal_async_test.h',
+ 'normal_test.h',
+ 'packet_loss_test.h',
+ 'performance_test.h',
+ 'test.h',
+ 'unit_test.h',
+ 'video_buffer.h',
+ 'video_source.h',
+
+ # source files
+ 'benchmark.cc',
+ 'normal_async_test.cc',
+ 'normal_test.cc',
+ 'packet_loss_test.cc',
+ 'performance_test.cc',
+ 'test.cc',
+ 'unit_test.cc',
+ 'video_buffer.cc',
+ 'video_source.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc
new file mode 100644
index 0000000..99bd7fb
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc
@@ -0,0 +1,817 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <cassert>
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+#include "tick_util.h"
+#include "unit_test.h"
+#include "video_source.h"
+
+using namespace webrtc;
+
+UnitTest::UnitTest()
+:
+Test("UnitTest", "Unit test"),
+_tests(0),
+_errors(0),
+_source(NULL),
+_refFrame(NULL),
+_refEncFrame(NULL),
+_refDecFrame(NULL),
+_refEncFrameLength(0),
+_sourceFile(NULL),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL)
+{
+}
+
+UnitTest::UnitTest(std::string name, std::string description)
+:
+Test(name, description),
+_tests(0),
+_errors(0),
+_source(NULL),
+_refFrame(NULL),
+_refEncFrame(NULL),
+_refDecFrame(NULL),
+_refEncFrameLength(0),
+_sourceFile(NULL),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL)
+{
+}
+
+UnitTest::~UnitTest()
+{
+ if (_encodeCompleteCallback) {
+ delete _encodeCompleteCallback;
+ }
+
+ if (_decodeCompleteCallback) {
+ delete _decodeCompleteCallback;
+ }
+
+ if (_source) {
+ delete _source;
+ }
+
+ if (_refFrame) {
+ delete [] _refFrame;
+ }
+
+ if (_refDecFrame) {
+ delete [] _refDecFrame;
+ }
+
+ if (_sourceBuffer) {
+ delete [] _sourceBuffer;
+ }
+
+ if (_sourceFile) {
+ fclose(_sourceFile);
+ }
+
+ if (_refEncFrame) {
+ delete [] _refEncFrame;
+ }
+}
+
+WebRtc_Word32
+UnitTestEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo,
+ const webrtc::RTPFragmentationHeader*
+ fragmentation)
+{
+ _encodedVideoBuffer->VerifyAndAllocate(encodedImage._size);
+ _encodedVideoBuffer->CopyBuffer(encodedImage._size, encodedImage._buffer);
+ _encodedVideoBuffer->UpdateLength(encodedImage._length);
+ _encodedVideoBuffer->SetFrameType(encodedImage._frameType);
+ _encodedVideoBuffer->SetCaptureWidth(
+ (WebRtc_UWord16)encodedImage._encodedWidth);
+ _encodedVideoBuffer->SetCaptureHeight(
+ (WebRtc_UWord16)encodedImage._encodedHeight);
+ _encodedVideoBuffer->SetTimeStamp(encodedImage._timeStamp);
+ _encodeComplete = true;
+ _encodedFrameType = encodedImage._frameType;
+ return 0;
+}
+
+WebRtc_Word32 UnitTestDecodeCompleteCallback::Decoded(RawImage& image)
+{
+ _decodedVideoBuffer->VerifyAndAllocate(image._length);
+ _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
+ _decodedVideoBuffer->SetWidth(image._width);
+ _decodedVideoBuffer->SetHeight(image._height);
+ _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
+ _decodeComplete = true;
+ return 0;
+}
+
+bool
+UnitTestEncodeCompleteCallback::EncodeComplete()
+{
+ if (_encodeComplete)
+ {
+ _encodeComplete = false;
+ return true;
+ }
+ return false;
+}
+
+VideoFrameType
+UnitTestEncodeCompleteCallback::EncodedFrameType() const
+{
+ return _encodedFrameType;
+}
+
+bool
+UnitTestDecodeCompleteCallback::DecodeComplete()
+{
+ if (_decodeComplete)
+ {
+ _decodeComplete = false;
+ return true;
+ }
+ return false;
+}
+
+WebRtc_UWord32
+UnitTest::WaitForEncodedFrame() const
+{
+ WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+ while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs)
+ {
+ if (_encodeCompleteCallback->EncodeComplete())
+ {
+ return _encodedVideoBuffer.GetLength();
+ }
+ }
+ return 0;
+}
+
+WebRtc_UWord32
+UnitTest::WaitForDecodedFrame() const
+{
+ WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+ while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitDecTimeMs)
+ {
+ if (_decodeCompleteCallback->DecodeComplete())
+ {
+ return _decodedVideoBuffer.GetLength();
+ }
+ }
+ return 0;
+}
+
+WebRtc_UWord32
+UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
+ WebRtc_UWord32 /* frameRate */)
+{
+ return _encoder->SetRates(bitRate, _inst.maxFramerate);
+}
+
+void
+UnitTest::Setup()
+{
+ // Use _sourceFile as a check to prevent multiple Setup() calls.
+ if (_sourceFile != NULL)
+ {
+ return;
+ }
+
+ if (_encodeCompleteCallback == NULL)
+ {
+ _encodeCompleteCallback =
+ new UnitTestEncodeCompleteCallback(&_encodedVideoBuffer);
+ }
+ if (_decodeCompleteCallback == NULL)
+ {
+ _decodeCompleteCallback =
+ new UnitTestDecodeCompleteCallback(&_decodedVideoBuffer);
+ }
+
+ _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
+ _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
+
+ _source = new VideoSource(webrtc::test::ProjectRootPath() +
+ "resources/foreman_cif.yuv", kCIF);
+
+ _lengthSourceFrame = _source->GetFrameLength();
+ _refFrame = new unsigned char[_lengthSourceFrame];
+ _refDecFrame = new unsigned char[_lengthSourceFrame];
+ _sourceBuffer = new unsigned char [_lengthSourceFrame];
+ _sourceFile = fopen(_source->GetFileName().c_str(), "rb");
+ ASSERT_TRUE(_sourceFile != NULL);
+
+ _inst.maxFramerate = _source->GetFrameRate();
+ _bitRate = 300;
+ _inst.startBitrate = 300;
+ _inst.maxBitrate = 4000;
+ _inst.width = _source->GetWidth();
+ _inst.height = _source->GetHeight();
+
+ // Get input frame.
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame, _sourceFile)
+ == _lengthSourceFrame);
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+ rewind(_sourceFile);
+
+ // Get a reference encoded frame.
+ _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+
+ RawImage image;
+ VideoBufferToRawImage(_inputVideoBuffer, image);
+
+ // Ensures our initial parameters are valid.
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ VideoFrameType videoFrameType = kDeltaFrame;
+ _encoder->Encode(image, NULL, &videoFrameType);
+ _refEncFrameLength = WaitForEncodedFrame();
+ ASSERT_TRUE(_refEncFrameLength > 0);
+ _refEncFrame = new unsigned char[_refEncFrameLength];
+ memcpy(_refEncFrame, _encodedVideoBuffer.GetBuffer(), _refEncFrameLength);
+
+ // Get a reference decoded frame.
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ unsigned int frameLength = 0;
+ int i=0;
+ while (frameLength == 0)
+ {
+ if (i > 0)
+ {
+ // Insert yet another frame
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame,
+ _sourceFile) == _lengthSourceFrame);
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+ _inputVideoBuffer.SetWidth(_source->GetWidth());
+ _inputVideoBuffer.SetHeight(_source->GetHeight());
+ VideoBufferToRawImage(_inputVideoBuffer, image);
+ _encoder->Encode(image, NULL, &videoFrameType);
+ ASSERT_TRUE(WaitForEncodedFrame() > 0);
+ }
+ EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ ASSERT_TRUE(_decoder->Decode(encodedImage, 0, NULL)
+ == WEBRTC_VIDEO_CODEC_OK);
+ frameLength = WaitForDecodedFrame();
+ _encodedVideoBuffer.Reset();
+ _encodedVideoBuffer.UpdateLength(0);
+ i++;
+ }
+ rewind(_sourceFile);
+ EXPECT_TRUE(frameLength == _lengthSourceFrame);
+ memcpy(_refDecFrame, _decodedVideoBuffer.GetBuffer(), _lengthSourceFrame);
+}
+
+void
+UnitTest::Teardown()
+{
+ // Use _sourceFile as a check to prevent multiple Teardown() calls.
+ if (_sourceFile == NULL)
+ {
+ return;
+ }
+
+ _encoder->Release();
+ _decoder->Release();
+
+ fclose(_sourceFile);
+ _sourceFile = NULL;
+ delete [] _refFrame;
+ _refFrame = NULL;
+ delete [] _refEncFrame;
+ _refEncFrame = NULL;
+ delete [] _refDecFrame;
+ _refDecFrame = NULL;
+ delete [] _sourceBuffer;
+ _sourceBuffer = NULL;
+}
+
+void
+UnitTest::Print()
+{
+ printf("Unit Test\n\n%i tests completed\n", _tests);
+ if (_errors > 0)
+ {
+ printf("%i FAILED\n\n", _errors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+
+int
+UnitTest::DecodeWithoutAssert()
+{
+ EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ int ret = _decoder->Decode(encodedImage, 0, NULL);
+ int frameLength = WaitForDecodedFrame();
+ _encodedVideoBuffer.Reset();
+ _encodedVideoBuffer.UpdateLength(0);
+ return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
+}
+
+int
+UnitTest::Decode()
+{
+ EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ if (encodedImage._length == 0)
+ {
+ return WEBRTC_VIDEO_CODEC_OK;
+ }
+ int ret = _decoder->Decode(encodedImage, 0, NULL);
+ unsigned int frameLength = WaitForDecodedFrame();
+ assert(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
+ == _lengthSourceFrame));
+ EXPECT_TRUE(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
+ == _lengthSourceFrame));
+ _encodedVideoBuffer.Reset();
+ _encodedVideoBuffer.UpdateLength(0);
+ return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
+}
+
+// Test pure virtual VideoEncoder and VideoDecoder APIs.
+void
+UnitTest::Perform()
+{
+ UnitTest::Setup();
+ int frameLength;
+ RawImage inputImage;
+ EncodedImage encodedImage;
+ VideoFrameType videoFrameType = kDeltaFrame;
+
+ //----- Encoder parameter tests -----
+
+ //-- Calls before InitEncode() --
+ // We want to revert the initialization done in Setup().
+ EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType )
+ == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+ //-- InitEncode() errors --
+ // Null pointer.
+ EXPECT_TRUE(_encoder->InitEncode(NULL, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ // bit rate exceeds max bit rate
+ WebRtc_Word32 tmpBitRate = _inst.startBitrate;
+ WebRtc_Word32 tmpMaxBitRate = _inst.maxBitrate;
+ _inst.startBitrate = 4000;
+ _inst.maxBitrate = 3000;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _inst.startBitrate = tmpBitRate;
+ _inst.maxBitrate = tmpMaxBitRate; //unspecified value
+
+ // Bad framerate.
+ _inst.maxFramerate = 0;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ // Seems like we should allow any framerate in range [0, 255].
+ //_inst.frameRate = 100;
+ //EXPECT_TRUE(_encoder->InitEncode(&_inst, 1) == -1); // FAILS
+ _inst.maxFramerate = 30;
+
+ // Bad bitrate.
+ _inst.startBitrate = -1;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _inst.maxBitrate = _inst.startBitrate - 1;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _inst.maxBitrate = 0;
+ _inst.startBitrate = 300;
+
+ // Bad maxBitRate.
+ _inst.maxBitrate = 200;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _inst.maxBitrate = 4000;
+
+ // Bad width.
+ _inst.width = 0;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
+ // Should there be a width and height cap?
+ //_inst.width = 10000;
+ //EXPECT_TRUE(_encoder->InitEncode(&_inst, 1) == -1);
+ _inst.width = _source->GetWidth();
+
+ // Bad height.
+ _inst.height = 0;
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
+ _inst.height = _source->GetHeight();
+
+ // Bad number of cores.
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, -1, 1440) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+
+ //-- Encode() errors --
+
+ // inputVideoBuffer unallocated.
+ _inputVideoBuffer.Free();
+ VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+
+ //----- Encoder stress tests -----
+
+ // Vary frame rate and I-frame request.
+ VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+ for (int i = 1; i <= 60; i++)
+ {
+ VideoFrameType frameType = !(i % 2) ? kKeyFrame : kDeltaFrame;
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &frameType) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(WaitForEncodedFrame() > 0);
+ }
+
+ // Init then encode.
+ _encodedVideoBuffer.UpdateLength(0);
+ _encodedVideoBuffer.Reset();
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(WaitForEncodedFrame() > 0);
+
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ _encoder->Encode(inputImage, NULL, &videoFrameType);
+ frameLength = WaitForEncodedFrame();
+ EXPECT_TRUE(frameLength > 0);
+ EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+ _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+ // Reset then encode.
+ _encodedVideoBuffer.UpdateLength(0);
+ _encodedVideoBuffer.Reset();
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ WaitForEncodedFrame();
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ _encoder->Encode(inputImage, NULL, &videoFrameType);
+ frameLength = WaitForEncodedFrame();
+ EXPECT_TRUE(frameLength > 0);
+ EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+ _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+ // Release then encode.
+ _encodedVideoBuffer.UpdateLength(0);
+ _encodedVideoBuffer.Reset();
+ EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ WaitForEncodedFrame();
+ EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ _encoder->Encode(inputImage, NULL, &videoFrameType);
+ frameLength = WaitForEncodedFrame();
+ EXPECT_TRUE(frameLength > 0);
+ EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+ _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+ //----- Decoder parameter tests -----
+
+ //-- Calls before InitDecode() --
+ // We want to revert the initialization done in Setup().
+ EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ EXPECT_TRUE(_decoder->Decode(encodedImage, false, NULL) ==
+ WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+ WaitForDecodedFrame();
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ //-- Decode() errors --
+ // Unallocated encodedVideoBuffer.
+ _encodedVideoBuffer.Free();
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ encodedImage._length = 10; // Buffer NULL but length > 0
+ EXPECT_EQ(_decoder->Decode(encodedImage, false, NULL),
+ WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+
+ //----- Decoder stress tests -----
+ unsigned char* tmpBuf = new unsigned char[_lengthSourceFrame];
+
+ // "Random" and zero data.
+ // We either expect an error, or at the least, no output.
+ // This relies on the codec's ability to detect an erroneous bitstream.
+ /*
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+ for (int i = 0; i < 100; i++)
+ {
+ ASSERT_TRUE(fread(tmpBuf, 1, _refEncFrameLength, _sourceFile)
+ == _refEncFrameLength);
+ _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ FillDecoderSpecificInfo(encodedImage);
+ int ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
+ EXPECT_TRUE(ret <= 0);
+ if (ret == 0)
+ {
+ EXPECT_TRUE(WaitForDecodedFrame() == 0);
+ }
+
+ memset(tmpBuf, 0, _refEncFrameLength);
+ _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ FillDecoderSpecificInfo(encodedImage);
+ ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
+ EXPECT_TRUE(ret <= 0);
+ if (ret == 0)
+ {
+ EXPECT_TRUE(WaitForDecodedFrame() == 0);
+ }
+ }
+ */
+ rewind(_sourceFile);
+
+ _encodedVideoBuffer.UpdateLength(_refEncFrameLength);
+ _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, _refEncFrame);
+
+ // Init then decode.
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+ frameLength = 0;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ while (frameLength == 0)
+ {
+ _decoder->Decode(encodedImage, false, NULL);
+ frameLength = WaitForDecodedFrame();
+ }
+ EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+ _refDecFrame, _lengthSourceFrame) == true);
+
+ // Reset then decode.
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ frameLength = 0;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ while (frameLength == 0)
+ {
+ _decoder->Decode(encodedImage, false, NULL);
+ frameLength = WaitForDecodedFrame();
+ }
+ EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+ _refDecFrame, _lengthSourceFrame) == true);
+
+ // Decode with other size, reset, then decode with original size again
+ // to verify that decoder is reset to a "fresh" state upon Reset().
+ {
+ // assert that input frame size is a factor of two, so that we can use
+ // quarter size below
+ EXPECT_TRUE((_inst.width % 2 == 0) && (_inst.height % 2 == 0));
+
+ VideoCodec tempInst;
+ memcpy(&tempInst, &_inst, sizeof(VideoCodec));
+ tempInst.width /= 2;
+ tempInst.height /= 2;
+
+ // Encode reduced (quarter) frame size
+ EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_encoder->InitEncode(&tempInst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ RawImage tempInput(inputImage._buffer, inputImage._length/4,
+ inputImage._size/4);
+ VideoFrameType videoFrameType = kDeltaFrame;
+ _encoder->Encode(tempInput, NULL, &videoFrameType);
+ frameLength = WaitForEncodedFrame();
+ EXPECT_TRUE(frameLength > 0);
+
+ // Reset then decode.
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ frameLength = 0;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ while (frameLength == 0)
+ {
+ _decoder->Decode(encodedImage, false, NULL);
+ frameLength = WaitForDecodedFrame();
+ }
+
+ // Encode original frame again
+ EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ _encoder->Encode(inputImage, NULL, &videoFrameType);
+ frameLength = WaitForEncodedFrame();
+ EXPECT_TRUE(frameLength > 0);
+
+ // Reset then decode original frame again.
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ frameLength = 0;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ while (frameLength == 0)
+ {
+ _decoder->Decode(encodedImage, false, NULL);
+ frameLength = WaitForDecodedFrame();
+ }
+
+ // check that decoded frame matches with reference
+ EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+ _refDecFrame, _lengthSourceFrame) == true);
+
+ }
+
+ // Release then decode.
+ EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+ frameLength = 0;
+ VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+ while (frameLength == 0)
+ {
+ _decoder->Decode(encodedImage, false, NULL);
+ frameLength = WaitForDecodedFrame();
+ }
+ EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+ _refDecFrame, _lengthSourceFrame) == true);
+ _encodedVideoBuffer.UpdateLength(0);
+ _encodedVideoBuffer.Reset();
+
+ delete [] tmpBuf;
+
+ //----- Function tests -----
+ int frames = 0;
+ // Do not specify maxBitRate (as in ViE).
+ _inst.maxBitrate = 0;
+
+ //-- Timestamp propagation --
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ printf("\nTimestamp propagation test...\n");
+ frames = 0;
+ int frameDelay = 0;
+ int encTimeStamp;
+ _decodedVideoBuffer.SetTimeStamp(0);
+ while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+ _lengthSourceFrame)
+ {
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp(frames);
+ VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+ VideoFrameType videoFrameType = kDeltaFrame;
+ ASSERT_TRUE(_encoder->Encode(inputImage,
+ NULL,
+ &videoFrameType) ==
+ WEBRTC_VIDEO_CODEC_OK);
+ frameLength = WaitForEncodedFrame();
+ //ASSERT_TRUE(frameLength);
+ EXPECT_TRUE(frameLength > 0);
+ encTimeStamp = _encodedVideoBuffer.GetTimeStamp();
+ EXPECT_TRUE(_inputVideoBuffer.GetTimeStamp() ==
+ static_cast<unsigned>(encTimeStamp));
+
+ frameLength = Decode();
+ if (frameLength == 0)
+ {
+ frameDelay++;
+ }
+
+ encTimeStamp -= frameDelay;
+ if (encTimeStamp < 0)
+ {
+ encTimeStamp = 0;
+ }
+ EXPECT_TRUE(_decodedVideoBuffer.GetTimeStamp() ==
+ static_cast<unsigned>(encTimeStamp));
+ frames++;
+ }
+ ASSERT_TRUE(feof(_sourceFile) != 0);
+ rewind(_sourceFile);
+
+ RateControlTests();
+
+ Teardown();
+}
+
+void
+UnitTest::RateControlTests()
+{
+ std::string outFileName;
+ int frames = 0;
+ RawImage inputImage;
+ WebRtc_UWord32 frameLength;
+
+ // Do not specify maxBitRate (as in ViE).
+ _inst.maxBitrate = 0;
+ //-- Verify rate control --
+ EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+ // add: should also be 0, and 1
+ const int bitRate[] =
+ {100, 200, 300, 400, 500, 600, 800, 1000, 2000, 3000, 4000, 10000};
+ const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
+
+ printf("\nRate control test\n");
+ for (int i = 0; i < nBitrates; i++)
+ {
+ _bitRate = bitRate[i];
+ int totalBytes = 0;
+ _inst.startBitrate = _bitRate;
+ _encoder->InitEncode(&_inst, 4, 1440);
+ _decoder->Reset();
+ _decoder->InitDecode(&_inst, 1);
+ frames = 0;
+
+ if (_bitRate > _inst.maxBitrate)
+ {
+ CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
+ }
+ else
+ {
+ CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
+ }
+
+ while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+ _lengthSourceFrame)
+ {
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp(_inputVideoBuffer.GetTimeStamp() +
+ static_cast<WebRtc_UWord32>(9e4 /
+ static_cast<float>(_inst.maxFramerate)));
+ VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+ VideoFrameType videoFrameType = kDeltaFrame;
+ ASSERT_EQ(_encoder->Encode(inputImage, NULL, &videoFrameType),
+ WEBRTC_VIDEO_CODEC_OK);
+ frameLength = WaitForEncodedFrame();
+ ASSERT_GE(frameLength, 0u);
+ totalBytes += frameLength;
+ frames++;
+
+ _encodedVideoBuffer.UpdateLength(0);
+ _encodedVideoBuffer.Reset();
+ }
+ WebRtc_UWord32 actualBitrate =
+ (totalBytes / frames * _inst.maxFramerate * 8)/1000;
+ printf("Target bitrate: %d kbps, actual bitrate: %d kbps\n", _bitRate,
+ actualBitrate);
+ // Test for close match over reasonable range.
+ if (_bitRate >= 100 && _bitRate <= 2500)
+ {
+ EXPECT_TRUE(abs(WebRtc_Word32(actualBitrate - _bitRate)) <
+ 0.1 * _bitRate); // for VP8
+ }
+ ASSERT_TRUE(feof(_sourceFile) != 0);
+ rewind(_sourceFile);
+ }
+}
+
+bool
+UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+ const void* ptrB, unsigned int bLengthBytes)
+{
+ if (aLengthBytes != bLengthBytes)
+ {
+ return false;
+ }
+
+ return memcmp(ptrA, ptrB, aLengthBytes) == 0;
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h
new file mode 100644
index 0000000..01b348d
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+
+#include "test.h"
+#include "event_wrapper.h"
+
+// Disable "conditional expression is constant" warnings on the perfectly
+// acceptable
+// do { ... } while (0) constructions below.
+// Refer to http://stackoverflow.com/questions/1946445/
+// is-there-better-way-to-write-do-while0-construct-to-avoid-compiler-warnings
+// for some discussion of the issue.
+#ifdef _WIN32
+#pragma warning(disable : 4127)
+#endif
+
+class VideoSource;
+class UnitTestEncodeCompleteCallback;
+class UnitTestDecodeCompleteCallback;
+
+class UnitTest : public Test
+{
+public:
+ UnitTest();
+ virtual ~UnitTest();
+ virtual void Perform();
+ virtual void Print();
+
+protected:
+ UnitTest(std::string name, std::string description);
+ virtual WebRtc_UWord32 CodecSpecific_SetBitrate(
+ WebRtc_UWord32 bitRate,
+ WebRtc_UWord32 /* frameRate */);
+ virtual void Setup();
+ virtual void Teardown();
+ virtual void RateControlTests();
+ virtual int Decode();
+ virtual int DecodeWithoutAssert();
+ virtual int SetCodecSpecificParameters() {return 0;};
+
+ virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+ const void *ptrB, unsigned int bLengthBytes);
+
+ WebRtc_UWord32 WaitForEncodedFrame() const;
+ WebRtc_UWord32 WaitForDecodedFrame() const;
+
+ int _tests;
+ int _errors;
+
+ VideoSource* _source;
+ unsigned char* _refFrame;
+ unsigned char* _refEncFrame;
+ unsigned char* _refDecFrame;
+ int _refEncFrameLength;
+ FILE* _sourceFile;
+
+ UnitTestEncodeCompleteCallback* _encodeCompleteCallback;
+ UnitTestDecodeCompleteCallback* _decodeCompleteCallback;
+ enum { kMaxWaitEncTimeMs = 100 };
+ enum { kMaxWaitDecTimeMs = 25 };
+};
+
+class UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback
+{
+public:
+ UnitTestEncodeCompleteCallback(TestVideoEncodedBuffer* buffer,
+ WebRtc_UWord32 decoderSpecificSize = 0,
+ void* decoderSpecificInfo = NULL) :
+ _encodedVideoBuffer(buffer),
+ _decoderSpecificInfo(decoderSpecificInfo),
+ _decoderSpecificSize(decoderSpecificSize),
+ _encodeComplete(false) {}
+ WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo,
+ const webrtc::RTPFragmentationHeader*
+ fragmentation = NULL);
+ bool EncodeComplete();
+ // Note that this only makes sense if an encode has been completed
+ webrtc::VideoFrameType EncodedFrameType() const;
+private:
+ TestVideoEncodedBuffer* _encodedVideoBuffer;
+ void* _decoderSpecificInfo;
+ WebRtc_UWord32 _decoderSpecificSize;
+ bool _encodeComplete;
+ webrtc::VideoFrameType _encodedFrameType;
+};
+
+class UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+ UnitTestDecodeCompleteCallback(TestVideoBuffer* buffer) :
+ _decodedVideoBuffer(buffer), _decodeComplete(false) {}
+ WebRtc_Word32 Decoded(webrtc::RawImage& image);
+ bool DecodeComplete();
+private:
+ TestVideoBuffer* _decodedVideoBuffer;
+ bool _decodeComplete;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc
new file mode 100644
index 0000000..3958e90
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc
@@ -0,0 +1,319 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <string.h>
+#include "video_buffer.h"
+
+using namespace webrtc;
+
+TestVideoBuffer::TestVideoBuffer():
+_buffer(0),
+_bufferSize(0),
+_bufferLength(0),
+_startOffset(0),
+_timeStamp(0),
+_width(0),
+_height(0)
+{
+ //
+}
+
+
+TestVideoBuffer::~TestVideoBuffer()
+{
+ _timeStamp = 0;
+ _startOffset = 0;
+ _bufferLength = 0;
+ _bufferSize = 0;
+
+ if(_buffer)
+ {
+ delete [] _buffer;
+ _buffer = 0;
+ }
+}
+
+TestVideoBuffer::TestVideoBuffer(const TestVideoBuffer& rhs)
+:
+_buffer(0),
+_bufferSize(rhs._bufferSize),
+_bufferLength(rhs._bufferLength),
+_startOffset(rhs._startOffset),
+_timeStamp(rhs._timeStamp),
+_width(rhs._width),
+_height(rhs._height)
+{
+ // make sure that our buffer is big enough
+ _buffer = new unsigned char[_bufferSize];
+
+ // only copy required length
+ memcpy(_buffer + _startOffset, rhs._buffer, _bufferLength); // GetBuffer() includes _startOffset
+}
+
+void TestVideoBuffer::SetTimeStamp(unsigned int timeStamp)
+{
+ _timeStamp = timeStamp;
+}
+
+unsigned int
+TestVideoBuffer::GetWidth() const
+{
+ return _width;
+}
+
+unsigned int
+TestVideoBuffer::GetHeight() const
+{
+ return _height;
+}
+
+void
+TestVideoBuffer::SetWidth(unsigned int width)
+{
+ _width = width;
+}
+
+void
+TestVideoBuffer::SetHeight(unsigned int height)
+{
+ _height = height;
+}
+
+
+void TestVideoBuffer::Free()
+{
+ _timeStamp = 0;
+ _startOffset = 0;
+ _bufferLength = 0;
+ _bufferSize = 0;
+ _height = 0;
+ _width = 0;
+
+ if(_buffer)
+ {
+ delete [] _buffer;
+ _buffer = 0;
+ }
+}
+
+void TestVideoBuffer::VerifyAndAllocate(unsigned int minimumSize)
+{
+ if(minimumSize > _bufferSize)
+ {
+ // make sure that our buffer is big enough
+ unsigned char * newBufferBuffer = new unsigned char[minimumSize];
+ if(_buffer)
+ {
+ // copy the old data
+ memcpy(newBufferBuffer, _buffer, _bufferSize);
+ delete [] _buffer;
+ }
+ _buffer = newBufferBuffer;
+ _bufferSize = minimumSize;
+ }
+}
+
+int TestVideoBuffer::SetOffset(unsigned int length)
+{
+ if (length > _bufferSize ||
+ length > _bufferLength)
+ {
+ return -1;
+ }
+
+ unsigned int oldOffset = _startOffset;
+
+ if(oldOffset > length)
+ {
+ unsigned int newLength = _bufferLength + (oldOffset-length);// increase by the diff
+ assert(newLength <= _bufferSize);
+ _bufferLength = newLength;
+ }
+ if(oldOffset < length)
+ {
+ if(_bufferLength > (length-oldOffset))
+ {
+ _bufferLength -= (length-oldOffset); // decrease by the diff
+ }
+ }
+ _startOffset = length; // update
+
+ return 0;
+}
+
+void TestVideoBuffer::UpdateLength(unsigned int newLength)
+{
+ assert(newLength +_startOffset <= _bufferSize);
+ _bufferLength = newLength;
+}
+
+void TestVideoBuffer::CopyBuffer(unsigned int length, const unsigned char* buffer)
+{
+ assert(length+_startOffset <= _bufferSize);
+ memcpy(_buffer+_startOffset, buffer, length);
+ _bufferLength = length;
+}
+
+void TestVideoBuffer::CopyBuffer(TestVideoBuffer& fromVideoBuffer)
+{
+ assert(fromVideoBuffer.GetLength() + fromVideoBuffer.GetStartOffset() <= _bufferSize);
+ assert(fromVideoBuffer.GetSize() <= _bufferSize);
+
+ _bufferLength = fromVideoBuffer.GetLength();
+ _startOffset = fromVideoBuffer.GetStartOffset();
+ _timeStamp = fromVideoBuffer.GetTimeStamp();
+ _height = fromVideoBuffer.GetHeight();
+ _width = fromVideoBuffer.GetWidth();
+
+ // only copy required length
+ memcpy(_buffer+_startOffset, fromVideoBuffer.GetBuffer(), fromVideoBuffer.GetLength()); // GetBuffer() includes _startOffset
+
+}
+
+void TestVideoBuffer::CopyPointer(const TestVideoBuffer& fromVideoBuffer)
+{
+ _bufferSize = fromVideoBuffer.GetSize();
+ _bufferLength = fromVideoBuffer.GetLength();
+ _startOffset = fromVideoBuffer.GetStartOffset();
+ _timeStamp = fromVideoBuffer.GetTimeStamp();
+ _height = fromVideoBuffer.GetHeight();
+ _width = fromVideoBuffer.GetWidth();
+
+ _buffer = fromVideoBuffer.GetBuffer();
+}
+
+void TestVideoBuffer::ClearPointer()
+{
+ _buffer = NULL;
+}
+
+void TestVideoBuffer::SwapBuffers(TestVideoBuffer& videoBuffer)
+{
+ unsigned char* tempBuffer = _buffer;
+ unsigned int tempSize = _bufferSize;
+ unsigned int tempLength =_bufferLength;
+ unsigned int tempOffset = _startOffset;
+ unsigned int tempTime = _timeStamp;
+ unsigned int tempHeight = _height;
+ unsigned int tempWidth = _width;
+
+ _buffer = videoBuffer.GetBuffer();
+ _bufferSize = videoBuffer.GetSize();
+ _bufferLength = videoBuffer.GetLength();
+ _startOffset = videoBuffer.GetStartOffset();
+ _timeStamp = videoBuffer.GetTimeStamp();
+ _height = videoBuffer.GetHeight();
+ _width = videoBuffer.GetWidth();
+
+
+ videoBuffer.Set(tempBuffer, tempSize, tempLength, tempOffset, tempTime);
+ videoBuffer.SetHeight(tempHeight);
+ videoBuffer.SetWidth(tempWidth);
+}
+
+void TestVideoBuffer::Set(unsigned char* tempBuffer,unsigned int tempSize,unsigned int tempLength, unsigned int tempOffset,unsigned int timeStamp)
+{
+ _buffer = tempBuffer;
+ _bufferSize = tempSize;
+ _bufferLength = tempLength;
+ _startOffset = tempOffset;
+ _timeStamp = timeStamp;
+}
+
+unsigned char* TestVideoBuffer::GetBuffer() const
+{
+ return _buffer+_startOffset;
+}
+
+unsigned int TestVideoBuffer::GetStartOffset() const
+{
+ return _startOffset;
+}
+
+unsigned int TestVideoBuffer::GetSize() const
+{
+ return _bufferSize;
+}
+
+unsigned int TestVideoBuffer::GetLength() const
+{
+ return _bufferLength;
+}
+
+unsigned int TestVideoBuffer::GetTimeStamp() const
+{
+ return _timeStamp;
+}
+
+/**
+* TestVideoEncodedBuffer
+*
+*/
+
+TestVideoEncodedBuffer::TestVideoEncodedBuffer() :
+ _captureWidth(0),
+ _captureHeight(0),
+ _frameRate(-1)
+{
+ _frameType = kDeltaFrame;
+}
+
+TestVideoEncodedBuffer::~TestVideoEncodedBuffer()
+{
+}
+
+void TestVideoEncodedBuffer::SetCaptureWidth(unsigned short width)
+{
+ _captureWidth = width;
+}
+
+void TestVideoEncodedBuffer::SetCaptureHeight(unsigned short height)
+{
+ _captureHeight = height;
+}
+
+unsigned short TestVideoEncodedBuffer::GetCaptureWidth()
+{
+ return _captureWidth;
+}
+
+unsigned short TestVideoEncodedBuffer::GetCaptureHeight()
+{
+ return _captureHeight;
+}
+
+VideoFrameType TestVideoEncodedBuffer::GetFrameType()
+{
+ return _frameType;
+}
+
+void TestVideoEncodedBuffer::SetFrameType(VideoFrameType frametype)
+{
+ _frameType = frametype;
+}
+
+void TestVideoEncodedBuffer::Reset()
+{
+ _captureWidth = 0;
+ _captureHeight = 0;
+ _frameRate = -1;
+ _frameType = kDeltaFrame;
+}
+
+void TestVideoEncodedBuffer::SetFrameRate(float frameRate)
+{
+ _frameRate = frameRate;
+}
+
+float TestVideoEncodedBuffer::GetFrameRate()
+{
+ return _frameRate;
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h
new file mode 100644
index 0000000..824440e
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
+
+#include "typedefs.h"
+#include "video_image.h"
+
+class TestVideoBuffer
+{
+public:
+ TestVideoBuffer();
+
+ virtual ~TestVideoBuffer();
+
+ TestVideoBuffer(const TestVideoBuffer& rhs);
+
+ /**
+ * Verifies that current allocated buffer size is larger than or equal to the input size.
+ * If the current buffer size is smaller, a new allocation is made and the old buffer data is copied to the new buffer.
+ */
+ void VerifyAndAllocate(unsigned int minimumSize);
+
+ void UpdateLength(unsigned int newLength);
+
+ void SwapBuffers(TestVideoBuffer& videoBuffer);
+
+ void CopyBuffer(unsigned int length, const unsigned char* fromBuffer);
+
+ void CopyBuffer(TestVideoBuffer& fromVideoBuffer);
+
+ // Use with care, and remember to call ClearPointer() when done.
+ void CopyPointer(const TestVideoBuffer& fromVideoBuffer);
+
+ void ClearPointer();
+
+ int SetOffset(unsigned int length); // Sets offset to beginning of frame in buffer
+
+ void Free(); // Deletes frame buffer and resets members to zero
+
+ void SetTimeStamp(unsigned int timeStamp); // Sets timestamp of frame (90kHz)
+
+ /**
+ * Gets pointer to frame buffer
+ */
+ unsigned char* GetBuffer() const;
+
+ /**
+ * Gets allocated buffer size
+ */
+ unsigned int GetSize() const;
+
+ /**
+ * Gets length of frame
+ */
+ unsigned int GetLength() const;
+
+ /**
+ * Gets timestamp of frame (90kHz)
+ */
+ unsigned int GetTimeStamp() const;
+
+ unsigned int GetWidth() const;
+ unsigned int GetHeight() const;
+
+ void SetWidth(unsigned int width);
+ void SetHeight(unsigned int height);
+
+private:
+ TestVideoBuffer& operator=(const TestVideoBuffer& inBuffer);
+
+private:
+ void Set(unsigned char* buffer,unsigned int size,unsigned int length,unsigned int offset, unsigned int timeStamp);
+ unsigned int GetStartOffset() const;
+
+ unsigned char* _buffer; // Pointer to frame buffer
+ unsigned int _bufferSize; // Allocated buffer size
+ unsigned int _bufferLength; // Length (in bytes) of frame
+ unsigned int _startOffset; // Offset (in bytes) to beginning of frame in buffer
+ unsigned int _timeStamp; // Timestamp of frame (90kHz)
+ unsigned int _width;
+ unsigned int _height;
+};
+
+class TestVideoEncodedBuffer: public TestVideoBuffer
+{
+public:
+ TestVideoEncodedBuffer();
+ ~TestVideoEncodedBuffer();
+
+ void SetCaptureWidth(unsigned short width);
+ void SetCaptureHeight(unsigned short height);
+ unsigned short GetCaptureWidth();
+ unsigned short GetCaptureHeight();
+
+ webrtc::VideoFrameType GetFrameType();
+ void SetFrameType(webrtc::VideoFrameType frametype);
+
+ void Reset();
+
+ void SetFrameRate(float frameRate);
+ float GetFrameRate();
+
+private:
+ TestVideoEncodedBuffer& operator=(const TestVideoEncodedBuffer& inBuffer);
+
+private:
+ unsigned short _captureWidth;
+ unsigned short _captureHeight;
+ webrtc::VideoFrameType _frameType;
+ float _frameRate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc b/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc
new file mode 100644
index 0000000..8abad1b
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc
@@ -0,0 +1,419 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_source.h"
+
+#include <stdio.h>
+#include <cassert>
+
+#include "testsupport/fileutils.h"
+
+VideoSource::VideoSource()
+:
+_fileName(webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv"),
+_width(352),
+_height(288),
+_type(webrtc::kI420),
+_frameRate(30)
+{
+}
+
+VideoSource::VideoSource(std::string fileName, VideoSize size,
+ int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
+:
+_fileName(fileName),
+_type(type),
+_frameRate(frameRate)
+{
+ assert(size != kUndefined && size != kNumberOfVideoSizes);
+ assert(type != webrtc::kUnknown);
+ assert(frameRate > 0);
+ assert(GetWidthHeight(size, _width, _height) == 0);
+}
+
+VideoSource::VideoSource(std::string fileName, int width, int height,
+ int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
+:
+_fileName(fileName),
+_width(width),
+_height(height),
+_type(type),
+_frameRate(frameRate)
+{
+ assert(width > 0);
+ assert(height > 0);
+ assert(type != webrtc::kUnknown);
+ assert(frameRate > 0);
+}
+
+VideoSize
+VideoSource::GetSize() const
+{
+ return GetSize(_width, _height);
+}
+
+VideoSize
+VideoSource::GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height)
+{
+ if(width == 128 && height == 96)
+ {
+ return kSQCIF;
+ }else if(width == 160 && height == 120)
+ {
+ return kQQVGA;
+ }else if(width == 176 && height == 144)
+ {
+ return kQCIF;
+ }else if(width == 320 && height == 240)
+ {
+ return kQVGA;
+ }else if(width == 352 && height == 288)
+ {
+ return kCIF;
+ }else if(width == 640 && height == 480)
+ {
+ return kVGA;
+ }else if(width == 720 && height == 480)
+ {
+ return kNTSC;
+ }else if(width == 704 && height == 576)
+ {
+ return k4CIF;
+ }else if(width == 800 && height == 600)
+ {
+ return kSVGA;
+ }else if(width == 960 && height == 720)
+ {
+ return kHD;
+ }else if(width == 1024 && height == 768)
+ {
+ return kXGA;
+ }else if(width == 1440 && height == 1080)
+ {
+ return kFullHD;
+ }else if(width == 400 && height == 240)
+ {
+ return kWQVGA;
+ }else if(width == 800 && height == 480)
+ {
+ return kWVGA;
+ }else if(width == 1280 && height == 720)
+ {
+ return kWHD;
+ }else if(width == 1920 && height == 1080)
+ {
+ return kWFullHD;
+ }
+ return kUndefined;
+}
+
+unsigned int
+VideoSource::GetFrameLength() const
+{
+ return webrtc::CalcBufferSize(_type, _width, _height);
+}
+
+const char*
+VideoSource::GetMySizeString() const
+{
+ return VideoSource::GetSizeString(GetSize());
+}
+
+const char*
+VideoSource::GetSizeString(VideoSize size)
+{
+ switch (size)
+ {
+ case kSQCIF:
+ return "SQCIF";
+ case kQQVGA:
+ return "QQVGA";
+ case kQCIF:
+ return "QCIF";
+ case kQVGA:
+ return "QVGA";
+ case kCIF:
+ return "CIF";
+ case kVGA:
+ return "VGA";
+ case kNTSC:
+ return "NTSC";
+ case k4CIF:
+ return "4CIF";
+ case kSVGA:
+ return "SVGA";
+ case kHD:
+ return "HD";
+ case kXGA:
+ return "XGA";
+ case kFullHD:
+ return "Full_HD";
+ case kWQVGA:
+ return "WQVGA";
+ case kWHD:
+ return "WHD";
+ case kWFullHD:
+ return "WFull_HD";
+ default:
+ return "Undefined";
+ }
+}
+
+std::string
+VideoSource::GetFilePath() const
+{
+ size_t slashPos = _fileName.find_last_of("/\\");
+ if (slashPos == std::string::npos)
+ {
+ return ".";
+ }
+
+ return _fileName.substr(0, slashPos);
+}
+
+std::string
+VideoSource::GetName() const
+{
+ // Remove path.
+ size_t slashPos = _fileName.find_last_of("/\\");
+ if (slashPos == std::string::npos)
+ {
+ slashPos = 0;
+ }
+ else
+ {
+ slashPos++;
+ }
+
+ // Remove extension and underscored suffix if it exists.
+ return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
+ _fileName.find_last_of(".")) - slashPos);
+}
+
+void
+VideoSource::Convert(const VideoSource &target, bool force /* = false */) const
+{
+ // Ensure target rate is less than or equal to source
+ // (i.e. we are only temporally downsampling).
+ assert(target.GetFrameRate() <= _frameRate);
+ // Only supports YUV420 currently.
+ assert(_type == webrtc::kI420 && target.GetType() == webrtc::kI420);
+ if (!force && (FileExists(target.GetFileName().c_str()) ||
+ (target.GetWidth() == _width && target.GetHeight() == _height && target.GetFrameRate() == _frameRate)))
+ {
+ // Assume that the filename uniquely defines the content.
+ // If the file already exists, it is the correct file.
+ return;
+ }
+ FILE *inFile = NULL;
+ FILE *outFile = NULL;
+
+ inFile = fopen(_fileName.c_str(), "rb");
+ assert(inFile != NULL);
+
+ outFile = fopen(target.GetFileName().c_str(), "wb");
+ assert(outFile != NULL);
+
+ FrameDropper fd;
+ fd.SetFrameRate(target.GetFrameRate(), _frameRate);
+
+ const size_t lengthOutFrame = webrtc::CalcBufferSize(target.GetType(),
+ target.GetWidth(), target.GetHeight());
+ assert(lengthOutFrame > 0);
+ unsigned char *outFrame = new unsigned char[lengthOutFrame];
+
+ const size_t lengthInFrame = webrtc::CalcBufferSize(_type, _width, _height);
+ assert(lengthInFrame > 0);
+ unsigned char *inFrame = new unsigned char[lengthInFrame];
+
+ while (fread(inFrame, 1, lengthInFrame, inFile) == lengthInFrame)
+ {
+ if (!fd.DropFrame())
+ {
+ assert(target.GetWidth() == _width &&
+ target.GetHeight() == _height); // Add video interpolator here!
+ fwrite(outFrame, 1, lengthOutFrame, outFile);
+ }
+ }
+
+ delete inFrame;
+ delete outFrame;
+ fclose(inFile);
+ fclose(outFile);
+}
+
+bool VideoSource::FileExists(const char* fileName)
+{
+ FILE* fp = NULL;
+ fp = fopen(fileName, "rb");
+ if(fp != NULL)
+ {
+ fclose(fp);
+ return true;
+ }
+ return false;
+}
+
+
+int
+VideoSource::GetWidthHeight( VideoSize size, int & width, int& height)
+{
+ switch(size)
+ {
+ case kSQCIF:
+ width = 128;
+ height = 96;
+ return 0;
+ case kQQVGA:
+ width = 160;
+ height = 120;
+ return 0;
+ case kQCIF:
+ width = 176;
+ height = 144;
+ return 0;
+ case kCGA:
+ width = 320;
+ height = 200;
+ return 0;
+ case kQVGA:
+ width = 320;
+ height = 240;
+ return 0;
+ case kSIF:
+ width = 352;
+ height = 240;
+ return 0;
+ case kWQVGA:
+ width = 400;
+ height = 240;
+ return 0;
+ case kCIF:
+ width = 352;
+ height = 288;
+ return 0;
+ case kW288p:
+ width = 512;
+ height = 288;
+ return 0;
+ case k448p:
+ width = 576;
+ height = 448;
+ return 0;
+ case kVGA:
+ width = 640;
+ height = 480;
+ return 0;
+ case k432p:
+ width = 720;
+ height = 432;
+ return 0;
+ case kW432p:
+ width = 768;
+ height = 432;
+ return 0;
+ case k4SIF:
+ width = 704;
+ height = 480;
+ return 0;
+ case kW448p:
+ width = 768;
+ height = 448;
+ return 0;
+ case kNTSC:
+ width = 720;
+ height = 480;
+ return 0;
+ case kFW448p:
+ width = 800;
+ height = 448;
+ return 0;
+ case kWVGA:
+ width = 800;
+ height = 480;
+ return 0;
+ case k4CIF:
+ width = 704;
+ height = 576;
+ return 0;
+ case kSVGA:
+ width = 800;
+ height = 600;
+ return 0;
+ case kW544p:
+ width = 960;
+ height = 544;
+ return 0;
+ case kW576p:
+ width = 1024;
+ height = 576;
+ return 0;
+ case kHD:
+ width = 960;
+ height = 720;
+ return 0;
+ case kXGA:
+ width = 1024;
+ height = 768;
+ return 0;
+ case kFullHD:
+ width = 1440;
+ height = 1080;
+ return 0;
+ case kWHD:
+ width = 1280;
+ height = 720;
+ return 0;
+ case kWFullHD:
+ width = 1920;
+ height = 1080;
+ return 0;
+ default:
+ return -1;
+ }
+}
+
+FrameDropper::FrameDropper()
+:
+_dropsBetweenRenders(0),
+_frameCounter(0)
+{
+}
+
+bool
+FrameDropper::DropFrame()
+{
+ _frameCounter++;
+ if (_frameCounter > _dropsBetweenRenders)
+ {
+ _frameCounter = 0;
+ return false;
+ }
+ return true;
+}
+
+unsigned int
+FrameDropper::DropsBetweenRenders()
+{
+ return _dropsBetweenRenders;
+}
+
+void
+FrameDropper::SetFrameRate(double frameRate, double maxFrameRate)
+{
+ if (frameRate >= 1.0)
+ {
+ _dropsBetweenRenders = static_cast<unsigned int>(maxFrameRate / frameRate + 0.5) - 1;
+ }
+ else
+ {
+ _dropsBetweenRenders = 0;
+ }
+}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_source.h b/trunk/src/modules/video_coding/codecs/test_framework/video_source.h
new file mode 100644
index 0000000..4f785ba
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/test_framework/video_source.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+
+#include <string>
+#include "common_video/libyuv/include/libyuv.h"
+
+enum VideoSize
+ {
+ kUndefined,
+ kSQCIF, // 128*96 = 12 288
+ kQQVGA, // 160*120 = 19 200
+ kQCIF, // 176*144 = 25 344
+ kCGA, // 320*200 = 64 000
+ kQVGA, // 320*240 = 76 800
+ kSIF, // 352*240 = 84 480
+ kWQVGA, // 400*240 = 96 000
+ kCIF, // 352*288 = 101 376
+ kW288p, // 512*288 = 147 456 (WCIF)
+ k448p, // 576*448 = 281 088
+ kVGA, // 640*480 = 307 200
+ k432p, // 720*432 = 311 040
+ kW432p, // 768*432 = 331 776
+ k4SIF, // 704*480 = 337 920
+ kW448p, // 768*448 = 344 064
+ kNTSC, // 720*480 = 345 600
+ kFW448p, // 800*448 = 358 400
+ kWVGA, // 800*480 = 384 000
+ k4CIF, // 704�576 = 405 504
+ kSVGA, // 800*600 = 480 000
+ kW544p, // 960*544 = 522 240
+ kW576p, // 1024*576 = 589 824 (W4CIF)
+ kHD, // 960*720 = 691 200
+ kXGA, // 1024*768 = 786 432
+ kWHD, // 1280*720 = 921 600
+ kFullHD, // 1440*1080 = 1 555 200
+ kWFullHD, // 1920*1080 = 2 073 600
+
+ kNumberOfVideoSizes
+ };
+
+class VideoSource
+{
+public:
+ VideoSource();
+ VideoSource(std::string fileName, VideoSize size, int frameRate = 30,
+ webrtc::VideoType type = webrtc::kI420);
+ VideoSource(std::string fileName, int width, int height, int frameRate = 30,
+ webrtc::VideoType type = webrtc::kI420);
+
+ std::string GetFileName() const { return _fileName; }
+ int GetWidth() const { return _width; }
+ int GetHeight() const { return _height; }
+ webrtc::VideoType GetType() const { return _type; }
+ int GetFrameRate() const { return _frameRate; }
+
+ // Returns the file path without a trailing slash.
+ std::string GetFilePath() const;
+
+ // Returns the filename with the path (including the leading slash) removed.
+ std::string GetName() const;
+
+ VideoSize GetSize() const;
+ static VideoSize GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
+ unsigned int GetFrameLength() const;
+
+ // Returns a human-readable size string.
+ static const char* GetSizeString(VideoSize size);
+ const char* GetMySizeString() const;
+
+ // Opens the video source, converting and writing to the specified target.
+ // If force is true, the conversion will be done even if the target file
+ // already exists.
+ void Convert(const VideoSource& target, bool force = false) const;
+ static bool FileExists(const char* fileName);
+private:
+ static int GetWidthHeight( VideoSize size, int& width, int& height);
+ std::string _fileName;
+ int _width;
+ int _height;
+ webrtc::VideoType _type;
+ int _frameRate;
+};
+
+class FrameDropper
+{
+public:
+ FrameDropper();
+ bool DropFrame();
+ unsigned int DropsBetweenRenders();
+ void SetFrameRate(double frameRate, double maxFrameRate);
+
+private:
+ unsigned int _dropsBetweenRenders;
+ unsigned int _frameCounter;
+};
+
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+
diff --git a/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi b/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi
new file mode 100644
index 0000000..f43f285
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi
@@ -0,0 +1,37 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'video_quality_measurement',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_codecs_test_framework',
+ 'webrtc_video_coding',
+ 'webrtc_vp8',
+ '<(webrtc_root)/../test/metrics.gyp:metrics',
+ '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
+ ],
+ 'sources': [
+ 'video_quality_measurement.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc b/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc
new file mode 100644
index 0000000..953e3b4
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc
@@ -0,0 +1,526 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdarg.h>
+#include <sys/stat.h> // To check for directory existence.
+
+#include <cassert>
+#include <cstdio>
+#include <ctime>
+
+#ifndef S_ISDIR // Not defined in stat.h on Windows.
+#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
+#endif
+
+#include "common_types.h"
+#include "google/gflags.h"
+#include "modules/video_coding/codecs/test/packet_manipulator.h"
+#include "modules/video_coding/codecs/test/stats.h"
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+#include "modules/video_coding/codecs/vp8/main/interface/vp8.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "system_wrappers/interface/trace.h"
+#include "testsupport/frame_reader.h"
+#include "testsupport/frame_writer.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "testsupport/packet_reader.h"
+
+DEFINE_string(test_name, "Quality test", "The name of the test to run. ");
+DEFINE_string(test_description, "", "A more detailed description about what "
+ "the current test is about.");
+DEFINE_string(input_filename, "", "Input file. "
+ "The source video file to be encoded and decoded. Must be in "
+ ".yuv format");
+DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
+DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
+DEFINE_int32(framerate, 30, "Frame rate of the input file, in FPS "
+ "(frames-per-second). ");
+DEFINE_string(output_dir, ".", "Output directory. "
+ "The directory where the output file will be put. Must already "
+ "exist.");
+DEFINE_bool(use_single_core, false, "Force using a single core. If set to "
+ "true, only one core will be used for processing. Using a single "
+ "core is necessary to get a deterministic behavior for the"
+ "encoded frames - using multiple cores will produce different "
+ "encoded frames since multiple cores are competing to consume the "
+ "byte budget for each frame in parallel. If set to false, "
+ "the maximum detected number of cores will be used. ");
+DEFINE_bool(disable_fixed_random_seed , false, "Set this flag to disable the"
+ "usage of a fixed random seed for the random generator used "
+ "for packet loss. Disabling this will cause consecutive runs "
+ "loose packets at different locations, which is bad for "
+ "reproducibility.");
+DEFINE_string(output_filename, "", "Output file. "
+ "The name of the output video file resulting of the processing "
+ "of the source file. By default this is the same name as the "
+ "input file with '_out' appended before the extension.");
+DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
+DEFINE_int32(keyframe_interval, 0, "Forces a keyframe every Nth frame. "
+ "0 means the encoder decides when to insert keyframes. Note that "
+ "the encoder may create a keyframe in other locations in addition "
+ "to the interval that is set using this parameter.");
+DEFINE_int32(temporal_layers, 0, "The number of temporal layers to use "
+ "(VP8 specific codec setting). Must be 0-4.");
+DEFINE_int32(packet_size, 1500, "Simulated network packet size in bytes (MTU). "
+ "Used for packet loss simulation.");
+DEFINE_int32(max_payload_size, 1440, "Max payload size in bytes for the "
+ "encoder.");
+DEFINE_string(packet_loss_mode, "uniform", "Packet loss mode. Two different "
+ "packet loss models are supported: uniform or burst. This "
+ "setting has no effect unless packet_loss_rate is >0. ");
+DEFINE_double(packet_loss_probability, 0.0, "Packet loss probability. A value "
+ "between 0.0 and 1.0 that defines the probability of a packet "
+ "being lost. 0.1 means 10% and so on.");
+DEFINE_int32(packet_loss_burst_length, 1, "Packet loss burst length. Defines "
+ "how many packets will be lost in a burst when a packet has been "
+ "decided to be lost. Must be >=1.");
+DEFINE_bool(csv, false, "CSV output. Enabling this will output all frame "
+ "statistics at the end of execution. Recommended to run combined "
+ "with --noverbose to avoid mixing output.");
+DEFINE_bool(python, false, "Python output. Enabling this will output all frame "
+ "statistics as a Python script at the end of execution. "
+ "Recommended to run combine with --noverbose to avoid mixing "
+ "output.");
+DEFINE_bool(verbose, true, "Verbose mode. Prints a lot of debugging info. "
+ "Suitable for tracking progress but not for capturing output. "
+ "Disable with --noverbose flag.");
+
+// Custom log method that only prints if the verbose flag is given.
+// Supports all the standard printf parameters and formatting (just forwarded).
+int Log(const char *format, ...) {
+ int result = 0;
+ if (FLAGS_verbose) {
+ va_list args;
+ va_start(args, format);
+ result = vprintf(format, args);
+ va_end(args);
+ }
+ return result;
+}
+
+// Validates the arguments given as command line flags and fills in the
+// TestConfig struct with all configurations needed for video processing.
+// Returns 0 if everything is OK, otherwise an exit code.
+int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
+ // Validate the mandatory flags:
+ if (FLAGS_input_filename == "" || FLAGS_width == -1 || FLAGS_height == -1) {
+ printf("%s\n", google::ProgramUsage());
+ return 1;
+ }
+ config->name = FLAGS_test_name;
+ config->description = FLAGS_test_description;
+
+ // Verify the input file exists and is readable.
+ FILE* test_file;
+ test_file = fopen(FLAGS_input_filename.c_str(), "rb");
+ if (test_file == NULL) {
+ fprintf(stderr, "Cannot read the specified input file: %s\n",
+ FLAGS_input_filename.c_str());
+ return 2;
+ }
+ fclose(test_file);
+ config->input_filename = FLAGS_input_filename;
+
+ // Verify the output dir exists.
+ struct stat dir_info;
+ if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
+ S_ISDIR(dir_info.st_mode))) {
+ fprintf(stderr, "Cannot find output directory: %s\n",
+ FLAGS_output_dir.c_str());
+ return 3;
+ }
+ config->output_dir = FLAGS_output_dir;
+
+ // Manufacture an output filename if none was given.
+ if (FLAGS_output_filename == "") {
+ // Cut out the filename without extension from the given input file
+ // (which may include a path)
+ int startIndex = FLAGS_input_filename.find_last_of("/") + 1;
+ if (startIndex == 0) {
+ startIndex = 0;
+ }
+ FLAGS_output_filename =
+ FLAGS_input_filename.substr(startIndex,
+ FLAGS_input_filename.find_last_of(".")
+ - startIndex) + "_out.yuv";
+ }
+
+ // Verify output file can be written.
+ if (FLAGS_output_dir == ".") {
+ config->output_filename = FLAGS_output_filename;
+ } else {
+ config->output_filename = FLAGS_output_dir + "/"+ FLAGS_output_filename;
+ }
+ test_file = fopen(config->output_filename.c_str(), "wb");
+ if (test_file == NULL) {
+ fprintf(stderr, "Cannot write output file: %s\n",
+ config->output_filename.c_str());
+ return 4;
+ }
+ fclose(test_file);
+
+ // Check single core flag.
+ config->use_single_core = FLAGS_use_single_core;
+
+ // Get codec specific configuration.
+ webrtc::VideoCodingModule::Codec(webrtc::kVideoCodecVP8,
+ config->codec_settings);
+
+ // Check the temporal layers.
+ if (FLAGS_temporal_layers < 0 ||
+ FLAGS_temporal_layers > webrtc::kMaxTemporalStreams) {
+ fprintf(stderr, "Temporal layers number must be 0-4, was: %d\n",
+ FLAGS_temporal_layers);
+ return 13;
+ }
+ config->codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
+ FLAGS_temporal_layers;
+
+ // Check the bit rate.
+ if (FLAGS_bitrate <= 0) {
+ fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAGS_bitrate);
+ return 5;
+ }
+ config->codec_settings->startBitrate = FLAGS_bitrate;
+
+ // Check the keyframe interval.
+ if (FLAGS_keyframe_interval < 0) {
+ fprintf(stderr, "Keyframe interval must be >=0, was: %d\n",
+ FLAGS_keyframe_interval);
+ return 6;
+ }
+ config->keyframe_interval = FLAGS_keyframe_interval;
+
+ // Check packet size and max payload size.
+ if (FLAGS_packet_size <= 0) {
+ fprintf(stderr, "Packet size must be >0 bytes, was: %d\n",
+ FLAGS_packet_size);
+ return 7;
+ }
+ config->networking_config.packet_size_in_bytes = FLAGS_packet_size;
+
+ if (FLAGS_max_payload_size <= 0) {
+ fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
+ FLAGS_max_payload_size);
+ return 8;
+ }
+ config->networking_config.max_payload_size_in_bytes =
+ FLAGS_max_payload_size;
+
+ // Check the width and height
+ if (FLAGS_width <= 0 || FLAGS_height <= 0) {
+ fprintf(stderr, "Width and height must be >0.");
+ return 9;
+ }
+ config->codec_settings->width = FLAGS_width;
+ config->codec_settings->height = FLAGS_height;
+ config->codec_settings->maxFramerate = FLAGS_framerate;
+
+ // Calculate the size of each frame to read (according to YUV spec).
+ config->frame_length_in_bytes =
+ 3 * config->codec_settings->width * config->codec_settings->height / 2;
+
+ // Check packet loss settings
+ if (FLAGS_packet_loss_mode != "uniform" &&
+ FLAGS_packet_loss_mode != "burst") {
+ fprintf(stderr, "Unsupported packet loss mode, must be 'uniform' or "
+ "'burst'\n.");
+ return 10;
+ }
+ config->networking_config.packet_loss_mode = webrtc::test::kUniform;
+ if (FLAGS_packet_loss_mode == "burst") {
+ config->networking_config.packet_loss_mode = webrtc::test::kBurst;
+ }
+
+ if (FLAGS_packet_loss_probability < 0.0 ||
+ FLAGS_packet_loss_probability > 1.0) {
+ fprintf(stderr, "Invalid packet loss probability. Must be 0.0 - 1.0, "
+ "was: %f\n", FLAGS_packet_loss_probability);
+ return 11;
+ }
+ config->networking_config.packet_loss_probability =
+ FLAGS_packet_loss_probability;
+
+ if (FLAGS_packet_loss_burst_length < 1) {
+ fprintf(stderr, "Invalid packet loss burst length, must be >=1, "
+ "was: %d\n", FLAGS_packet_loss_burst_length);
+ return 12;
+ }
+ config->networking_config.packet_loss_burst_length =
+ FLAGS_packet_loss_burst_length;
+ config->verbose = FLAGS_verbose;
+ return 0;
+}
+
+void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
+ webrtc::test::QualityMetricsResult* result) {
+ Log("Calculating SSIM...\n");
+ I420SSIMFromFiles(config->input_filename.c_str(),
+ config->output_filename.c_str(),
+ config->codec_settings->width,
+ config->codec_settings->height, result);
+ Log(" Average: %3.2f\n", result->average);
+ Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
+ Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
+}
+
+void CalculatePsnrVideoMetrics(webrtc::test::TestConfig* config,
+ webrtc::test::QualityMetricsResult* result) {
+ Log("Calculating PSNR...\n");
+ I420PSNRFromFiles(config->input_filename.c_str(),
+ config->output_filename.c_str(),
+ config->codec_settings->width,
+ config->codec_settings->height, result);
+ Log(" Average: %3.2f\n", result->average);
+ Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
+ Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
+}
+
+void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
+ Log("Quality test with parameters:\n");
+ Log(" Test name : %s\n", config.name.c_str());
+ Log(" Description : %s\n", config.description.c_str());
+ Log(" Input filename : %s\n", config.input_filename.c_str());
+ Log(" Output directory : %s\n", config.output_dir.c_str());
+ Log(" Output filename : %s\n", config.output_filename.c_str());
+ Log(" Frame length : %d bytes\n", config.frame_length_in_bytes);
+ Log(" Packet size : %d bytes\n",
+ config.networking_config.packet_size_in_bytes);
+ Log(" Max payload size : %d bytes\n",
+ config.networking_config.max_payload_size_in_bytes);
+ Log(" Packet loss:\n");
+ Log(" Mode : %s\n",
+ PacketLossModeToStr(config.networking_config.packet_loss_mode));
+ Log(" Probability : %2.1f\n",
+ config.networking_config.packet_loss_probability);
+ Log(" Burst length : %d packets\n",
+ config.networking_config.packet_loss_burst_length);
+}
+
+void PrintCsvOutput(const webrtc::test::Stats& stats,
+ const webrtc::test::QualityMetricsResult& ssim_result,
+ const webrtc::test::QualityMetricsResult& psnr_result) {
+ Log("\nCSV output (recommended to run with --noverbose to skip the "
+ "above output)\n");
+ printf("frame_number encoding_successful decoding_successful "
+ "encode_return_code decode_return_code "
+ "encode_time_in_us decode_time_in_us "
+ "bit_rate_in_kbps encoded_frame_length_in_bytes frame_type "
+ "packets_dropped total_packets "
+ "ssim psnr\n");
+
+ for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
+ const webrtc::test::FrameStatistic& f = stats.stats_[i];
+ const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
+ const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
+ printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7d, %d, %2d, %2d, "
+ "%5.3f, %5.2f\n",
+ f.frame_number,
+ f.encoding_successful,
+ f.decoding_successful,
+ f.encode_return_code,
+ f.decode_return_code,
+ f.encode_time_in_us,
+ f.decode_time_in_us,
+ f.bit_rate_in_kbps,
+ f.encoded_frame_length_in_bytes,
+ f.frame_type,
+ f.packets_dropped,
+ f.total_packets,
+ ssim.value,
+ psnr.value);
+ }
+}
+
+void PrintPythonOutput(const webrtc::test::TestConfig& config,
+ const webrtc::test::Stats& stats,
+ const webrtc::test::QualityMetricsResult& ssim_result,
+ const webrtc::test::QualityMetricsResult& psnr_result) {
+ Log("\nPython output (recommended to run with --noverbose to skip the "
+ "above output)\n");
+ printf("test_configuration = ["
+ "{'name': 'name', 'value': '%s'},\n"
+ "{'name': 'description', 'value': '%s'},\n"
+ "{'name': 'test_number', 'value': '%d'},\n"
+ "{'name': 'input_filename', 'value': '%s'},\n"
+ "{'name': 'output_filename', 'value': '%s'},\n"
+ "{'name': 'output_dir', 'value': '%s'},\n"
+ "{'name': 'packet_size_in_bytes', 'value': '%d'},\n"
+ "{'name': 'max_payload_size_in_bytes', 'value': '%d'},\n"
+ "{'name': 'packet_loss_mode', 'value': '%s'},\n"
+ "{'name': 'packet_loss_probability', 'value': '%f'},\n"
+ "{'name': 'packet_loss_burst_length', 'value': '%d'},\n"
+ "{'name': 'exclude_frame_types', 'value': '%s'},\n"
+ "{'name': 'frame_length_in_bytes', 'value': '%d'},\n"
+ "{'name': 'use_single_core', 'value': '%s'},\n"
+ "{'name': 'keyframe_interval;', 'value': '%d'},\n"
+ "{'name': 'video_codec_type', 'value': '%s'},\n"
+ "{'name': 'width', 'value': '%d'},\n"
+ "{'name': 'height', 'value': '%d'},\n"
+ "{'name': 'bit_rate_in_kbps', 'value': '%d'},\n"
+ "]\n",
+ config.name.c_str(),
+ config.description.c_str(),
+ config.test_number,
+ config.input_filename.c_str(),
+ config.output_filename.c_str(),
+ config.output_dir.c_str(),
+ config.networking_config.packet_size_in_bytes,
+ config.networking_config.max_payload_size_in_bytes,
+ PacketLossModeToStr(config.networking_config.packet_loss_mode),
+ config.networking_config.packet_loss_probability,
+ config.networking_config.packet_loss_burst_length,
+ ExcludeFrameTypesToStr(config.exclude_frame_types),
+ config.frame_length_in_bytes,
+ config.use_single_core ? "True " : "False",
+ config.keyframe_interval,
+ webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
+ config.codec_settings->width,
+ config.codec_settings->height,
+ config.codec_settings->startBitrate);
+ printf("frame_data_types = {"
+ "'frame_number': ('number', 'Frame number'),\n"
+ "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
+ "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
+ "'encode_time': ('number', 'Encode time (us)'),\n"
+ "'decode_time': ('number', 'Decode time (us)'),\n"
+ "'encode_return_code': ('number', 'Encode return code'),\n"
+ "'decode_return_code': ('number', 'Decode return code'),\n"
+ "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
+ "'encoded_frame_length': "
+ "('number', 'Encoded frame length (bytes)'),\n"
+ "'frame_type': ('string', 'Frame type'),\n"
+ "'packets_dropped': ('number', 'Packets dropped'),\n"
+ "'total_packets': ('number', 'Total packets'),\n"
+ "'ssim': ('number', 'SSIM'),\n"
+ "'psnr': ('number', 'PSNR (dB)'),\n"
+ "}\n");
+ printf("frame_data = [");
+ for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
+ const webrtc::test::FrameStatistic& f = stats.stats_[i];
+ const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
+ const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
+ printf("{'frame_number': %d, "
+ "'encoding_successful': %s, 'decoding_successful': %s, "
+ "'encode_time': %d, 'decode_time': %d, "
+ "'encode_return_code': %d, 'decode_return_code': %d, "
+ "'bit_rate': %d, 'encoded_frame_length': %d, 'frame_type': %s, "
+ "'packets_dropped': %d, 'total_packets': %d, "
+ "'ssim': %f, 'psnr': %f},\n",
+ f.frame_number,
+ f.encoding_successful ? "True " : "False",
+ f.decoding_successful ? "True " : "False",
+ f.encode_time_in_us,
+ f.decode_time_in_us,
+ f.encode_return_code,
+ f.decode_return_code,
+ f.bit_rate_in_kbps,
+ f.encoded_frame_length_in_bytes,
+ f.frame_type == webrtc::kDeltaFrame ? "'Delta'" : "'Other'",
+ f.packets_dropped,
+ f.total_packets,
+ ssim.value,
+ psnr.value);
+ }
+ printf("]\n");
+}
+
+// Runs a quality measurement on the input file supplied to the program.
+// The input file must be in YUV format.
+int main(int argc, char* argv[]) {
+ std::string program_name = argv[0];
+ std::string usage = "Quality test application for video comparisons.\n"
+ "Run " + program_name + " --helpshort for usage.\n"
+ "Example usage:\n" + program_name +
+ " --input_filename=filename.yuv --width=352 --height=288\n";
+ google::SetUsageMessage(usage);
+
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ // Create TestConfig and codec settings struct.
+ webrtc::test::TestConfig config;
+ webrtc::VideoCodec codec_settings;
+ config.codec_settings = &codec_settings;
+
+ int return_code = HandleCommandLineFlags(&config);
+ // Exit if an invalid argument is supplied.
+ if (return_code != 0) {
+ return return_code;
+ }
+
+ PrintConfigurationSummary(config);
+
+ webrtc::VP8Encoder* encoder = webrtc::VP8Encoder::Create();
+ webrtc::VP8Decoder* decoder = webrtc::VP8Decoder::Create();
+ webrtc::test::Stats stats;
+ webrtc::test::FrameReaderImpl frame_reader(config.input_filename,
+ config.frame_length_in_bytes);
+ webrtc::test::FrameWriterImpl frame_writer(config.output_filename,
+ config.frame_length_in_bytes);
+ frame_reader.Init();
+ frame_writer.Init();
+ webrtc::test::PacketReader packet_reader;
+
+ webrtc::test::PacketManipulatorImpl packet_manipulator(
+ &packet_reader, config.networking_config, config.verbose);
+ // By default the packet manipulator is seeded with a fixed random.
+ // If disabled we must generate a new seed.
+ if (FLAGS_disable_fixed_random_seed) {
+ packet_manipulator.InitializeRandomSeed(time(NULL));
+ }
+ webrtc::test::VideoProcessor* processor =
+ new webrtc::test::VideoProcessorImpl(encoder, decoder,
+ &frame_reader,
+ &frame_writer,
+ &packet_manipulator,
+ config, &stats);
+ processor->Init();
+
+ int frame_number = 0;
+ while (processor->ProcessFrame(frame_number)) {
+ if (frame_number % 80 == 0) {
+ Log("\n"); // make the output a bit nicer.
+ }
+ Log(".");
+ frame_number++;
+ }
+ Log("\n");
+ Log("Processed %d frames\n", frame_number);
+
+ // Release encoder and decoder to make sure they have finished processing.
+ encoder->Release();
+ decoder->Release();
+
+ // Verify statistics are correct:
+ assert(frame_number == static_cast<int>(stats.stats_.size()));
+
+ // Close the files before we start using them for SSIM/PSNR calculations.
+ frame_reader.Close();
+ frame_writer.Close();
+
+ stats.PrintSummary();
+
+ webrtc::test::QualityMetricsResult ssim_result;
+ CalculateSsimVideoMetrics(&config, &ssim_result);
+ webrtc::test::QualityMetricsResult psnr_result;
+ CalculatePsnrVideoMetrics(&config, &psnr_result);
+
+ if (FLAGS_csv) {
+ PrintCsvOutput(stats, ssim_result, psnr_result);
+ }
+ if (FLAGS_python) {
+ PrintPythonOutput(config, stats, ssim_result, psnr_result);
+ }
+ delete processor;
+ delete encoder;
+ delete decoder;
+ Log("Quality test finished!");
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h
new file mode 100644
index 0000000..89cf067
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h
@@ -0,0 +1,236 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ *
+ * WEBRTC VP8 wrapper interface
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
+
+#include "video_codec_interface.h"
+
+// VPX forward declaration
+typedef struct vpx_codec_ctx vpx_codec_ctx_t;
+typedef struct vpx_codec_ctx vpx_dec_ctx_t;
+typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
+typedef struct vpx_image vpx_image_t;
+typedef struct vpx_ref_frame vpx_ref_frame_t;
+struct vpx_codec_cx_pkt;
+
+namespace webrtc
+{
+class TemporalLayers;
+class ReferencePictureSelection;
+
+class VP8Encoder : public VideoEncoder {
+ public:
+ static VP8Encoder* Create();
+
+ virtual ~VP8Encoder();
+
+ // Free encoder memory.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual int Release();
+
+ // Initialize the encoder with the information from the codecSettings
+ //
+ // Input:
+ // - codec_settings : Codec settings
+ // - number_of_cores : Number of cores available for the encoder
+ // - max_payload_size : The maximum size each payload is allowed
+ // to have. Usually MTU - overhead.
+ //
+ // Return value : Set bit rate if OK
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+ // WEBRTC_VIDEO_CODEC_ERR_SIZE
+ // WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED
+ // WEBRTC_VIDEO_CODEC_MEMORY
+ // WEBRTC_VIDEO_CODEC_ERROR
+ virtual int InitEncode(const VideoCodec* codec_settings,
+ int number_of_cores,
+ uint32_t max_payload_size);
+
+ // Encode an I420 image (as a part of a video stream). The encoded image
+ // will be returned to the user through the encode complete callback.
+ //
+ // Input:
+ // - input_image : Image to be encoded
+ // - frame_types : Frame type to be generated by the encoder.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+ // WEBRTC_VIDEO_CODEC_MEMORY
+ // WEBRTC_VIDEO_CODEC_ERROR
+ // WEBRTC_VIDEO_CODEC_TIMEOUT
+
+ virtual int Encode(const RawImage& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const VideoFrameType* frame_types);
+
+ // Register an encode complete callback object.
+ //
+ // Input:
+ // - callback : Callback object which handles encoded images.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
+
+ // Inform the encoder of the new packet loss rate and the round-trip time of
+ // the network.
+ //
+ // - packet_loss : Fraction lost
+ // (loss rate in percent = 100 * packetLoss / 255)
+ // - rtt : Round-trip time in milliseconds
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK
+ // <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
+ //
+ virtual int SetChannelParameters(uint32_t packet_loss, int rtt);
+
+ // Inform the encoder about the new target bit rate.
+ //
+ // - new_bitrate_kbit : New target bit rate
+ // - frame_rate : The target frame rate
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate);
+
+ private:
+ VP8Encoder();
+
+ // Call encoder initialize function and set control settings.
+ int InitAndSetControlSettings();
+
+ void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+ const vpx_codec_cx_pkt& pkt);
+
+ int GetEncodedFrame(const RawImage& input_image);
+
+ int GetEncodedPartitions(const RawImage& input_image);
+
+ // Determine maximum target for Intra frames
+ //
+ // Input:
+ // - optimal_buffer_size : Optimal buffer size
+ // Return Value : Max target size for Intra frames represented as
+ // percentage of the per frame bandwidth
+ uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
+
+ EncodedImage encoded_image_;
+ EncodedImageCallback* encoded_complete_callback_;
+ VideoCodec codec_;
+ bool inited_;
+ uint32_t timestamp_;
+ uint16_t picture_id_;
+ bool feedback_mode_;
+ int cpu_speed_;
+ uint32_t rc_max_intra_target_;
+ int token_partitions_;
+ ReferencePictureSelection* rps_;
+ TemporalLayers* temporal_layers_;
+ vpx_codec_ctx_t* encoder_;
+ vpx_codec_enc_cfg_t* config_;
+ vpx_image_t* raw_;
+}; // end of VP8Encoder class
+
+
+class VP8Decoder : public VideoDecoder {
+ public:
+ static VP8Decoder* Create();
+
+ virtual ~VP8Decoder();
+
+ // Initialize the decoder.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK.
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_ERROR
+ virtual int InitDecode(const VideoCodec* inst, int number_of_cores);
+
+ // Decode encoded image (as a part of a video stream). The decoded image
+ // will be returned to the user through the decode complete callback.
+ //
+ // Input:
+ // - input_image : Encoded image to be decoded
+ // - missing_frames : True if one or more frames have been lost
+ // since the previous decode call.
+ // - fragmentation : Specifies the start and length of each VP8
+ // partition.
+ // - codec_specific_info : pointer to specific codec data
+ // - render_time_ms : Render time in Ms
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_ERROR
+ // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+ virtual int Decode(const EncodedImage& input_image,
+ bool missing_frames,
+ const RTPFragmentationHeader* fragmentation,
+ const CodecSpecificInfo* codec_specific_info,
+ int64_t /*render_time_ms*/);
+
+ // Register a decode complete callback object.
+ //
+ // Input:
+ // - callback : Callback object which handles decoded images.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+ virtual int RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
+
+ // Free decoder memory.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK if OK
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_ERROR
+ virtual int Release();
+
+ // Reset decoder state and prepare for a new call.
+ //
+ // Return value : WEBRTC_VIDEO_CODEC_OK.
+ // <0 - Errors:
+ // WEBRTC_VIDEO_CODEC_UNINITIALIZED
+ // WEBRTC_VIDEO_CODEC_ERROR
+ virtual int Reset();
+
+ // Create a copy of the codec and its internal state.
+ //
+ // Return value : A copy of the instance if OK, NULL otherwise.
+ virtual VideoDecoder* Copy();
+
+ private:
+ VP8Decoder();
+
+ // Copy reference image from this _decoder to the _decoder in copyTo. Set
+ // which frame type to copy in _refFrame->frame_type before the call to
+ // this function.
+ int CopyReference(VP8Decoder* copy);
+
+ int DecodePartitions(const EncodedImage& input_image,
+ const RTPFragmentationHeader* fragmentation);
+
+ int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp);
+
+ RawImage decoded_image_;
+ DecodedImageCallback* decode_complete_callback_;
+ bool inited_;
+ bool feedback_mode_;
+ vpx_dec_ctx_t* decoder_;
+ VideoCodec codec_;
+ EncodedImage last_keyframe_;
+ int image_format_;
+ vpx_ref_frame_t* ref_frame_;
+ int propagation_cnt_;
+ bool latest_keyframe_complete_;
+ bool mfqe_enabled_;
+}; // end of VP8Decoder class
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h
new file mode 100644
index 0000000..6f347cd
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+// Ratio allocation between temporal streams:
+// Values as required for the VP8 codec (accumulating).
+static const float
+ kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
+ {1.0f, 0, 0, 0}, // 1 layer
+ {0.6f, 1.0f , 0 , 0}, // 2 layers {60%, 40%}
+ {0.4f, 0.6f , 1.0f, 0}, // 3 layers {40%, 20%, 40%}
+ {0.25f, 0.4f, 0.6f, 1.0f} // 4 layers {25%, 15%, 20%, 40%}
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk b/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk
new file mode 100644
index 0000000..eb6c1af
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk
@@ -0,0 +1,47 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_vp8
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ reference_picture_selection.cc \
+ vp8.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+# TODO(leozwang) Enable WEBRTC_LIBVPX_VERSION after libvpx is updateed
+# to a new version and also add temporal_layers.cc
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../common_video/interface \
+ $(LOCAL_PATH)/../../../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../../../modules/interface \
+ $(LOCAL_PATH)/../../../../../../system_wrappers/interface \
+ external/libvpx
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc
new file mode 100644
index 0000000..3ae6f19
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc
@@ -0,0 +1,131 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "reference_picture_selection.h"
+
+#include "typedefs.h"
+#include "vpx/vpx_encoder.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+
+ReferencePictureSelection::ReferencePictureSelection()
+ : kRttConfidence(1.33),
+ update_golden_next_(true),
+ established_golden_(false),
+ received_ack_(false),
+ last_sent_ref_picture_id_(0),
+ last_sent_ref_update_time_(0),
+ established_ref_picture_id_(0),
+ last_refresh_time_(0),
+ rtt_(0) {
+}
+
+void ReferencePictureSelection::Init() {
+ update_golden_next_ = true;
+ established_golden_ = false;
+ received_ack_ = false;
+ last_sent_ref_picture_id_ = 0;
+ last_sent_ref_update_time_ = 0;
+ established_ref_picture_id_ = 0;
+ last_refresh_time_ = 0;
+ rtt_ = 0;
+}
+
+void ReferencePictureSelection::ReceivedRPSI(int rpsi_picture_id) {
+ // Assume RPSI is signaled with 14 bits.
+ if ((rpsi_picture_id & 0x3fff) == (last_sent_ref_picture_id_ & 0x3fff)) {
+ // Remote peer has received our last reference frame, switch frame type.
+ received_ack_ = true;
+ established_golden_ = update_golden_next_;
+ update_golden_next_ = !update_golden_next_;
+ established_ref_picture_id_ = last_sent_ref_picture_id_;
+ }
+}
+
+bool ReferencePictureSelection::ReceivedSLI(uint32_t now_ts) {
+ bool send_refresh = false;
+ // Don't send a refresh more than once per round-trip time.
+ // This is to avoid too frequent refreshes, since the receiver
+ // will signal an SLI for every corrupt frame.
+ if (TimestampDiff(now_ts, last_refresh_time_) > rtt_) {
+ send_refresh = true;
+ last_refresh_time_ = now_ts;
+ }
+ return send_refresh;
+}
+
+int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
+ uint32_t now_ts) {
+ int flags = 0;
+ // We can't refresh the decoder until we have established the key frame.
+ if (send_refresh && received_ack_) {
+ flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame
+ if (established_golden_)
+ flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
+ else
+ flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame
+ }
+
+ // Make sure we don't update the reference frames too often. We must wait long
+ // enough for an RPSI to arrive after the decoder decoded the reference frame.
+ // Ideally that should happen after one round-trip time.
+ // Add a margin defined by |kRttConfidence|.
+ uint32_t update_interval = kRttConfidence * rtt_;
+ if (update_interval < kMinUpdateInterval)
+ update_interval = kMinUpdateInterval;
+ // Don't send reference frame updates until we have an established reference.
+ if (TimestampDiff(now_ts, last_sent_ref_update_time_) > update_interval &&
+ received_ack_) {
+ flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame.
+ if (update_golden_next_) {
+ flags |= VP8_EFLAG_FORCE_GF; // Update the golden reference.
+ flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update alt-ref.
+ flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
+ } else {
+ flags |= VP8_EFLAG_FORCE_ARF; // Update the alt-ref reference.
+ flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
+ flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
+ }
+ last_sent_ref_picture_id_ = picture_id;
+ last_sent_ref_update_time_ = now_ts;
+ } else {
+ // No update of golden or alt-ref. We can therefore freely reference the
+ // established reference frame and the last frame.
+ if (established_golden_)
+ flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
+ else
+ flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
+ flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
+ flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update the alt-ref frame.
+ }
+ return flags;
+}
+
+void ReferencePictureSelection::EncodedKeyFrame(int picture_id) {
+ last_sent_ref_picture_id_ = picture_id;
+ received_ack_ = false;
+}
+
+void ReferencePictureSelection::SetRtt(int rtt) {
+ // Convert from milliseconds to timestamp frequency.
+ rtt_ = 90 * rtt;
+}
+
+uint32_t ReferencePictureSelection::TimestampDiff(uint32_t new_ts,
+ uint32_t old_ts) {
+ if (old_ts > new_ts) {
+ // Assuming this is a wrap, doing a compensated subtraction.
+ return (new_ts + (static_cast<int64_t>(1) << 32)) - old_ts;
+ }
+ return new_ts - old_ts;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h
new file mode 100644
index 0000000..59e5940
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file defines classes for doing reference picture selection, primarily
+ * with VP8.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+class ReferencePictureSelection {
+ public:
+ ReferencePictureSelection();
+ void Init();
+
+ // Report a received reference picture selection indication. This will
+ // introduce a new established reference if the received RPSI isn't too late.
+ void ReceivedRPSI(int rpsi_picture_id);
+
+ // Report a received slice loss indication. Returns true if a refresh frame
+ // must be sent to the receiver, which is accomplished by only predicting
+ // from the established reference.
+ // |now_ts| is the RTP timestamp corresponding to the current time. Typically
+ // the capture timestamp of the frame currently being processed.
+ // Returns true if it's time to encode a decoder refresh, otherwise false.
+ bool ReceivedSLI(uint32_t now_ts);
+
+ // Returns the recommended VP8 encode flags needed. May refresh the decoder
+ // and/or update the reference buffers.
+ // |picture_id| picture id of the frame to be encoded.
+ // |send_refresh| should be set to true if a decoder refresh should be
+ // encoded, otherwise false.
+ // |now_ts| is the RTP timestamp corresponding to the current time. Typically
+ // the capture timestamp of the frame currently being processed.
+ // Returns the flags to be given to the libvpx encoder when encoding the next
+ // frame.
+ int EncodeFlags(int picture_id, bool send_refresh, uint32_t now_ts);
+
+ // Notify the RPS that the frame with picture id |picture_id| was encoded as
+ // a key frame, effectively updating all reference buffers.
+ void EncodedKeyFrame(int picture_id);
+
+ // Set the round-trip time between the sender and the receiver to |rtt|
+ // milliseconds.
+ void SetRtt(int rtt);
+
+ private:
+ static uint32_t TimestampDiff(uint32_t new_ts, uint32_t old_ts);
+
+ // The minimum time between reference frame updates.
+ enum { kMinUpdateInterval = 90 * 10 }; // Timestamp frequency
+ const double kRttConfidence;
+
+ bool update_golden_next_;
+ bool established_golden_;
+ bool received_ack_;
+ int last_sent_ref_picture_id_;
+ uint32_t last_sent_ref_update_time_;
+ int established_ref_picture_id_;
+ uint32_t last_refresh_time_;
+ uint32_t rtt_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc
new file mode 100644
index 0000000..cdac76c
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+#include "reference_picture_selection.h"
+#include "vpx/vpx_encoder.h"
+#include "vpx/vp8cx.h"
+
+using webrtc::ReferencePictureSelection;
+
+// The minimum time between reference frame updates. Should match the values
+// set in reference_picture_selection.h
+enum { kMinUpdateInterval = 10 };
+// The minimum time between decoder refreshes through restricted prediction.
+// Should match the values set in reference_picture_selection.h
+enum { kRtt = 10 };
+
+enum {
+ kNoPropagationGolden = VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_ARF,
+ kNoPropagationAltRef = VP8_EFLAG_NO_REF_GF |
+ VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_ARF,
+ kPropagateGolden = VP8_EFLAG_FORCE_GF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_REF_GF |
+ VP8_EFLAG_NO_REF_LAST,
+ kPropagateAltRef = VP8_EFLAG_FORCE_ARF |
+ VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_REF_LAST,
+ kRefreshFromGolden = VP8_EFLAG_NO_REF_LAST |
+ VP8_EFLAG_NO_REF_ARF,
+ kRefreshFromAltRef = VP8_EFLAG_NO_REF_LAST |
+ VP8_EFLAG_NO_REF_GF
+};
+
+class TestRPS : public ::testing::Test {
+ protected:
+ virtual void SetUp() {
+ rps_.Init();
+ // Initialize with sending a key frame and acknowledging it.
+ rps_.EncodedKeyFrame(0);
+ rps_.ReceivedRPSI(0);
+ rps_.SetRtt(kRtt);
+ }
+
+ ReferencePictureSelection rps_;
+};
+
+TEST_F(TestRPS, TestPropagateReferenceFrames) {
+ // Should propagate the alt-ref reference.
+ uint32_t time = (4 * kMinUpdateInterval) / 3 + 1;
+ EXPECT_EQ(rps_.EncodeFlags(1, false, 90 * time), kPropagateAltRef);
+ rps_.ReceivedRPSI(1);
+ time += (4 * (time + kMinUpdateInterval)) / 3 + 1;
+ // Should propagate the golden reference.
+ EXPECT_EQ(rps_.EncodeFlags(2, false, 90 * time), kPropagateGolden);
+ rps_.ReceivedRPSI(2);
+ // Should propagate the alt-ref reference.
+ time = (4 * (time + kMinUpdateInterval)) / 3 + 1;
+ EXPECT_EQ(rps_.EncodeFlags(3, false, 90 * time), kPropagateAltRef);
+ rps_.ReceivedRPSI(3);
+ // Shouldn't propagate any reference frames (except last), and the established
+ // reference is alt-ref.
+ time = time + kMinUpdateInterval;
+ EXPECT_EQ(rps_.EncodeFlags(4, false, 90 * time), kNoPropagationAltRef);
+}
+
+TEST_F(TestRPS, TestDecoderRefresh) {
+ uint32_t time = kRtt + 1;
+ // No more than one refresh per RTT.
+ EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
+ time += 5;
+ EXPECT_EQ(rps_.ReceivedSLI(90 * time), false);
+ time += kRtt - 4;
+ EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
+ // Enough time have elapsed since the previous reference propagation, we will
+ // therefore get both a refresh from golden and a propagation of alt-ref.
+ EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time), kRefreshFromGolden |
+ kPropagateAltRef);
+ rps_.ReceivedRPSI(5);
+ time += kRtt + 1;
+ // Enough time for a new refresh, but not enough time for a reference
+ // propagation.
+ EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
+ EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time), kRefreshFromAltRef |
+ kNoPropagationAltRef);
+}
+
+TEST_F(TestRPS, TestWrap) {
+ EXPECT_EQ(rps_.ReceivedSLI(0xffffffff), true);
+ EXPECT_EQ(rps_.ReceivedSLI(1), false);
+ EXPECT_EQ(rps_.ReceivedSLI(90 * 100), true);
+
+ EXPECT_EQ(rps_.EncodeFlags(7, false, 0xffffffff), kPropagateAltRef);
+ EXPECT_EQ(rps_.EncodeFlags(8, false, 1), kNoPropagationGolden);
+ EXPECT_EQ(rps_.EncodeFlags(10, false, 90 * 100), kPropagateAltRef);
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc
new file mode 100644
index 0000000..59ad9c2
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc
@@ -0,0 +1,219 @@
+/* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+*
+* Use of this source code is governed by a BSD-style license
+* that can be found in the LICENSE file in the root of the source
+* tree. An additional intellectual property rights grant can be found
+* in the file PATENTS. All contributing project authors may
+* be found in the AUTHORS file in the root of the source tree.
+*/
+
+#include "temporal_layers.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include <cassert>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h"
+
+#include "vpx/vpx_encoder.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+
+TemporalLayers::TemporalLayers(int numberOfTemporalLayers)
+ : number_of_temporal_layers_(numberOfTemporalLayers),
+ temporal_ids_length_(0),
+ temporal_pattern_length_(0),
+ tl0_pic_idx_(rand()),
+ pattern_idx_(255) {
+ assert(kMaxTemporalStreams >= numberOfTemporalLayers);
+ memset(temporal_ids_, 0, sizeof(temporal_ids_));
+ memset(temporal_pattern_, 0, sizeof(temporal_pattern_));
+}
+
+bool TemporalLayers::ConfigureBitrates(int bitrateKbit,
+ vpx_codec_enc_cfg_t* cfg) {
+ switch (number_of_temporal_layers_) {
+ case 0:
+ case 1:
+ // Do nothing.
+ break;
+ case 2:
+ temporal_ids_length_ = 2;
+ temporal_ids_[0] = 0;
+ temporal_ids_[1] = 1;
+ cfg->ts_number_layers = number_of_temporal_layers_;
+ cfg->ts_periodicity = temporal_ids_length_;
+ // Split stream 60% 40%.
+ // Bitrate API for VP8 is the agregated bitrate for all lower layers.
+ cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[1][0];
+ cfg->ts_target_bitrate[1] = bitrateKbit;
+ cfg->ts_rate_decimator[0] = 2;
+ cfg->ts_rate_decimator[1] = 1;
+ memcpy(cfg->ts_layer_id,
+ temporal_ids_,
+ sizeof(unsigned int) * temporal_ids_length_);
+ temporal_pattern_length_ = 8;
+ temporal_pattern_[0] = kTemporalUpdateLast;
+ temporal_pattern_[1] = kTemporalUpdateGoldenWithoutDependency;
+ temporal_pattern_[2] = kTemporalUpdateLast;
+ temporal_pattern_[3] = kTemporalUpdateGolden;
+ temporal_pattern_[4] = kTemporalUpdateLast;
+ temporal_pattern_[5] = kTemporalUpdateGolden;
+ temporal_pattern_[6] = kTemporalUpdateLast;
+ temporal_pattern_[7] = kTemporalUpdateNoneNoRefAltref;
+ break;
+ case 3:
+ temporal_ids_length_ = 4;
+ temporal_ids_[0] = 0;
+ temporal_ids_[1] = 2;
+ temporal_ids_[2] = 1;
+ temporal_ids_[3] = 2;
+ cfg->ts_number_layers = number_of_temporal_layers_;
+ cfg->ts_periodicity = temporal_ids_length_;
+ // Split stream 40% 20% 40%.
+ // Bitrate API for VP8 is the agregated bitrate for all lower layers.
+ cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[2][0];
+ cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[2][1];
+ cfg->ts_target_bitrate[2] = bitrateKbit;
+ cfg->ts_rate_decimator[0] = 4;
+ cfg->ts_rate_decimator[1] = 2;
+ cfg->ts_rate_decimator[2] = 1;
+ memcpy(cfg->ts_layer_id,
+ temporal_ids_,
+ sizeof(unsigned int) * temporal_ids_length_);
+ temporal_pattern_length_ = 8;
+ temporal_pattern_[0] = kTemporalUpdateLast;
+ temporal_pattern_[1] = kTemporalUpdateAltrefWithoutDependency;
+ temporal_pattern_[2] = kTemporalUpdateGoldenWithoutDependency;
+ temporal_pattern_[3] = kTemporalUpdateAltref;
+ temporal_pattern_[4] = kTemporalUpdateLast;
+ temporal_pattern_[5] = kTemporalUpdateAltref;
+ temporal_pattern_[6] = kTemporalUpdateGolden;
+ temporal_pattern_[7] = kTemporalUpdateNone;
+ break;
+ case 4:
+ temporal_ids_length_ = 8;
+ temporal_ids_[0] = 0;
+ temporal_ids_[1] = 3;
+ temporal_ids_[2] = 2;
+ temporal_ids_[3] = 3;
+ temporal_ids_[4] = 1;
+ temporal_ids_[5] = 3;
+ temporal_ids_[6] = 2;
+ temporal_ids_[7] = 3;
+ // Split stream 25% 15% 20% 40%.
+ // Bitrate API for VP8 is the agregated bitrate for all lower layers.
+ cfg->ts_number_layers = 4;
+ cfg->ts_periodicity = temporal_ids_length_;
+ cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[3][0];
+ cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[3][1];
+ cfg->ts_target_bitrate[2] = bitrateKbit * kVp8LayerRateAlloction[3][2];
+ cfg->ts_target_bitrate[3] = bitrateKbit;
+ cfg->ts_rate_decimator[0] = 8;
+ cfg->ts_rate_decimator[1] = 4;
+ cfg->ts_rate_decimator[2] = 2;
+ cfg->ts_rate_decimator[3] = 1;
+ memcpy(cfg->ts_layer_id,
+ temporal_ids_,
+ sizeof(unsigned int) * temporal_ids_length_);
+ temporal_pattern_length_ = 16;
+ temporal_pattern_[0] = kTemporalUpdateLast;
+ temporal_pattern_[1] = kTemporalUpdateNone;
+ temporal_pattern_[2] = kTemporalUpdateAltrefWithoutDependency;
+ temporal_pattern_[3] = kTemporalUpdateNone;
+ temporal_pattern_[4] = kTemporalUpdateGoldenWithoutDependency;
+ temporal_pattern_[5] = kTemporalUpdateNone;
+ temporal_pattern_[6] = kTemporalUpdateAltref;
+ temporal_pattern_[7] = kTemporalUpdateNone;
+ temporal_pattern_[8] = kTemporalUpdateLast;
+ temporal_pattern_[9] = kTemporalUpdateNone;
+ temporal_pattern_[10] = kTemporalUpdateAltref;
+ temporal_pattern_[11] = kTemporalUpdateNone;
+ temporal_pattern_[12] = kTemporalUpdateGolden;
+ temporal_pattern_[13] = kTemporalUpdateNone;
+ temporal_pattern_[14] = kTemporalUpdateAltref;
+ temporal_pattern_[15] = kTemporalUpdateNone;
+ break;
+ default:
+ assert(false);
+ return false;
+ }
+ return true;
+}
+
+int TemporalLayers::EncodeFlags() {
+ assert(number_of_temporal_layers_ > 1);
+ assert(kMaxTemporalPattern >= temporal_pattern_length_);
+ assert(0 < temporal_pattern_length_);
+
+ int flags = 0;
+ int patternIdx = ++pattern_idx_ % temporal_pattern_length_;
+ assert(kMaxTemporalPattern >= patternIdx);
+ switch (temporal_pattern_[patternIdx]) {
+ case kTemporalUpdateLast:
+ flags |= VP8_EFLAG_NO_UPD_GF;
+ flags |= VP8_EFLAG_NO_UPD_ARF;
+ flags |= VP8_EFLAG_NO_REF_GF;
+ flags |= VP8_EFLAG_NO_REF_ARF;
+ break;
+ case kTemporalUpdateGoldenWithoutDependency:
+ flags |= VP8_EFLAG_NO_REF_GF;
+ // Deliberately no break here.
+ case kTemporalUpdateGolden:
+ flags |= VP8_EFLAG_NO_REF_ARF;
+ flags |= VP8_EFLAG_NO_UPD_ARF;
+ flags |= VP8_EFLAG_NO_UPD_LAST;
+ break;
+ case kTemporalUpdateAltrefWithoutDependency:
+ flags |= VP8_EFLAG_NO_REF_ARF;
+ flags |= VP8_EFLAG_NO_REF_GF;
+ // Deliberately no break here.
+ case kTemporalUpdateAltref:
+ flags |= VP8_EFLAG_NO_UPD_GF;
+ flags |= VP8_EFLAG_NO_UPD_LAST;
+ break;
+ case kTemporalUpdateNoneNoRefAltref:
+ flags |= VP8_EFLAG_NO_REF_ARF;
+ // Deliberately no break here.
+ case kTemporalUpdateNone:
+ flags |= VP8_EFLAG_NO_UPD_GF;
+ flags |= VP8_EFLAG_NO_UPD_ARF;
+ flags |= VP8_EFLAG_NO_UPD_LAST;
+ flags |= VP8_EFLAG_NO_UPD_ENTROPY;
+ break;
+ }
+ return flags;
+}
+
+void TemporalLayers::PopulateCodecSpecific(bool key_frame,
+ CodecSpecificInfoVP8 *vp8_info) {
+ assert(number_of_temporal_layers_ > 1);
+ assert(0 < temporal_ids_length_);
+
+ if (key_frame) {
+ // Keyframe is always temporal layer 0
+ vp8_info->temporalIdx = 0;
+ } else {
+ vp8_info->temporalIdx = temporal_ids_[pattern_idx_ % temporal_ids_length_];
+ }
+ TemporalReferences temporal_reference =
+ temporal_pattern_[pattern_idx_ % temporal_pattern_length_];
+
+ if (temporal_reference == kTemporalUpdateAltrefWithoutDependency ||
+ temporal_reference == kTemporalUpdateGoldenWithoutDependency ||
+ (temporal_reference == kTemporalUpdateNone &&
+ number_of_temporal_layers_ == 4)) {
+ vp8_info->layerSync = true;
+ } else {
+ vp8_info->layerSync = false;
+ }
+
+ if (vp8_info->temporalIdx == 0) {
+ tl0_pic_idx_++;
+ }
+ vp8_info->tl0PicIdx = tl0_pic_idx_;
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h
new file mode 100644
index 0000000..2ca7229
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h
@@ -0,0 +1,67 @@
+/* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+*
+* Use of this source code is governed by a BSD-style license
+* that can be found in the LICENSE file in the root of the source
+* tree. An additional intellectual property rights grant can be found
+* in the file PATENTS. All contributing project authors may
+* be found in the AUTHORS file in the root of the source tree.
+*/
+/*
+* This file defines classes for doing temporal layers with VP8.
+*/
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
+
+#include <typedefs.h>
+
+ // VPX forward declaration
+typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
+
+namespace webrtc {
+
+struct CodecSpecificInfoVP8;
+
+class TemporalLayers {
+ public:
+ TemporalLayers(int number_of_temporal_layers);
+
+ // Returns the recommended VP8 encode flags needed. May refresh the decoder
+ // and/or update the reference buffers.
+ int EncodeFlags();
+
+ bool ConfigureBitrates(int bitrate_kbit, vpx_codec_enc_cfg_t* cfg);
+
+ void PopulateCodecSpecific(bool key_frame, CodecSpecificInfoVP8 *vp8_info);
+
+ private:
+ enum TemporalReferences {
+ // Second layer and last frame in cycle, for 2 layers.
+ kTemporalUpdateNoneNoRefAltref = 6,
+ // Highest enhancement layer.
+ kTemporalUpdateNone = 5,
+ // Second enhancement layer.
+ kTemporalUpdateAltref = 4,
+ // Second enhancement layer without dependency on previous frames in
+ // the second enhancement layer.
+ kTemporalUpdateAltrefWithoutDependency = 3,
+ // First enhancement layer.
+ kTemporalUpdateGolden = 2,
+ // First enhancement layer without dependency on previous frames in
+ // the first enhancement layer.
+ kTemporalUpdateGoldenWithoutDependency = 1,
+ // Base layer.
+ kTemporalUpdateLast = 0,
+ };
+ enum { kMaxTemporalPattern = 16 };
+
+ int number_of_temporal_layers_;
+ int temporal_ids_length_;
+ int temporal_ids_[kMaxTemporalPattern];
+ int temporal_pattern_length_;
+ TemporalReferences temporal_pattern_[kMaxTemporalPattern];
+ uint8_t tl0_pic_idx_;
+ uint8_t pattern_idx_;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
+
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc
new file mode 100644
index 0000000..6525dda
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc
@@ -0,0 +1,202 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include "gtest/gtest.h"
+#include "temporal_layers.h"
+#include "video_codec_interface.h"
+
+#include "vpx/vpx_encoder.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+
+enum {
+ kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_REF_GF |
+ VP8_EFLAG_NO_REF_ARF,
+ kTemporalUpdateGoldenWithoutDependency = VP8_EFLAG_NO_REF_GF |
+ VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_UPD_LAST,
+ kTemporalUpdateGolden = VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_UPD_LAST,
+ kTemporalUpdateAltrefWithoutDependency = VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_REF_GF |
+ VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_LAST,
+ kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_LAST,
+ kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_UPD_LAST |
+ VP8_EFLAG_NO_UPD_ENTROPY,
+ kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF |
+ VP8_EFLAG_NO_UPD_GF |
+ VP8_EFLAG_NO_UPD_ARF |
+ VP8_EFLAG_NO_UPD_LAST |
+ VP8_EFLAG_NO_UPD_ENTROPY,
+};
+
+TEST(TemporalLayersTest, 2Layers) {
+ TemporalLayers tl(2);
+ vpx_codec_enc_cfg_t cfg;
+ CodecSpecificInfoVP8 vp8_info;
+ tl.ConfigureBitrates(500, &cfg);
+
+ int expected_flags[16] = { kTemporalUpdateLast,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateLast,
+ kTemporalUpdateGolden,
+ kTemporalUpdateLast,
+ kTemporalUpdateGolden,
+ kTemporalUpdateLast,
+ kTemporalUpdateNoneNoRefAltRef,
+ kTemporalUpdateLast,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateLast,
+ kTemporalUpdateGolden,
+ kTemporalUpdateLast,
+ kTemporalUpdateGolden,
+ kTemporalUpdateLast,
+ kTemporalUpdateNoneNoRefAltRef
+ };
+ int expected_temporal_idx[16] =
+ { 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
+
+ bool expected_layer_sync[16] =
+ { false, true, false, false, false, false, false, false,
+ false, true, false, false, false, false, false, false };
+
+ for (int i = 0; i < 16; ++i) {
+ EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
+ tl.PopulateCodecSpecific(false, &vp8_info);
+ EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
+ bool expected_sync = expected_layer_sync[i];
+ EXPECT_EQ(expected_sync, vp8_info.layerSync);
+ }
+}
+
+TEST(TemporalLayersTest, 3Layers) {
+ TemporalLayers tl(3);
+ vpx_codec_enc_cfg_t cfg;
+ CodecSpecificInfoVP8 vp8_info;
+ tl.ConfigureBitrates(500, &cfg);
+
+ int expected_flags[16] = { kTemporalUpdateLast,
+ kTemporalUpdateAltrefWithoutDependency,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateAltref,
+ kTemporalUpdateLast,
+ kTemporalUpdateAltref,
+ kTemporalUpdateGolden,
+ kTemporalUpdateNone,
+ kTemporalUpdateLast,
+ kTemporalUpdateAltrefWithoutDependency,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateAltref,
+ kTemporalUpdateLast,
+ kTemporalUpdateAltref,
+ kTemporalUpdateGolden,
+ kTemporalUpdateNone,
+ };
+ int expected_temporal_idx[16] =
+ { 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2 };
+
+ bool expected_layer_sync[16] =
+ { false, true, true, false, false, false, false, false,
+ false, true, true, false, false, false, false, false };
+
+ for (int i = 0; i < 16; ++i) {
+ EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
+ tl.PopulateCodecSpecific(false, &vp8_info);
+ EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
+ bool expected_sync = expected_layer_sync[i];
+ EXPECT_EQ(expected_sync, vp8_info.layerSync);
+ }
+}
+
+TEST(TemporalLayersTest, 4Layers) {
+ TemporalLayers tl(4);
+ vpx_codec_enc_cfg_t cfg;
+ CodecSpecificInfoVP8 vp8_info;
+ tl.ConfigureBitrates(500, &cfg);
+ int expected_flags[16] = {
+ kTemporalUpdateLast,
+ kTemporalUpdateNone,
+ kTemporalUpdateAltrefWithoutDependency,
+ kTemporalUpdateNone,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateNone,
+ kTemporalUpdateAltref,
+ kTemporalUpdateNone,
+ kTemporalUpdateLast,
+ kTemporalUpdateNone,
+ kTemporalUpdateAltref,
+ kTemporalUpdateNone,
+ kTemporalUpdateGolden,
+ kTemporalUpdateNone,
+ kTemporalUpdateAltref,
+ kTemporalUpdateNone,
+ };
+ int expected_temporal_idx[16] =
+ { 0, 3, 2, 3, 1, 3, 2, 3, 0, 3, 2, 3, 1, 3, 2, 3 };
+
+ bool expected_layer_sync[16] =
+ { false, true, true, true, true, true, false, true,
+ false, true, false, true, false, true, false, true };
+
+ for (int i = 0; i < 16; ++i) {
+ EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
+ tl.PopulateCodecSpecific(false, &vp8_info);
+ EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
+ bool expected_sync = expected_layer_sync[i];
+ EXPECT_EQ(expected_sync, vp8_info.layerSync);
+ }
+}
+
+TEST(TemporalLayersTest, KeyFrame) {
+ TemporalLayers tl(3);
+ vpx_codec_enc_cfg_t cfg;
+ CodecSpecificInfoVP8 vp8_info;
+ tl.ConfigureBitrates(500, &cfg);
+
+ int expected_flags[8] = {
+ kTemporalUpdateLast,
+ kTemporalUpdateAltrefWithoutDependency,
+ kTemporalUpdateGoldenWithoutDependency,
+ kTemporalUpdateAltref,
+ kTemporalUpdateLast,
+ kTemporalUpdateAltref,
+ kTemporalUpdateGolden,
+ kTemporalUpdateNone,
+ };
+ int expected_temporal_idx[8] =
+ { 0, 0, 0, 0, 0, 0, 0, 2};
+
+ bool expected_layer_sync[8] =
+ { false, true, true, false, false, false, false, false };
+
+ for (int i = 0; i < 7; ++i) {
+ EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
+ tl.PopulateCodecSpecific(true, &vp8_info);
+ EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
+ bool expected_sync = expected_layer_sync[i];
+ EXPECT_EQ(expected_sync, vp8_info.layerSync);
+ }
+ EXPECT_EQ(expected_flags[7], tl.EncodeFlags());
+ tl.PopulateCodecSpecific(false, &vp8_info);
+ EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx);
+ bool expected_sync = expected_layer_sync[7];
+ EXPECT_EQ(expected_sync, vp8_info.layerSync);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
new file mode 100644
index 0000000..a477237
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
@@ -0,0 +1,977 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ *
+ * This file contains the WEBRTC VP8 wrapper implementation
+ *
+ */
+#include "vp8.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+
+#include "module_common_types.h"
+#include "reference_picture_selection.h"
+#include "temporal_layers.h"
+#include "tick_util.h"
+#include "vpx/vpx_encoder.h"
+#include "vpx/vpx_decoder.h"
+#include "vpx/vp8cx.h"
+#include "vpx/vp8dx.h"
+
+enum { kVp8ErrorPropagationTh = 30 };
+
+namespace webrtc
+{
+
+VP8Encoder* VP8Encoder::Create() {
+ return new VP8Encoder();
+}
+
+VP8Encoder::VP8Encoder()
+ : encoded_image_(),
+ encoded_complete_callback_(NULL),
+ inited_(false),
+ timestamp_(0),
+ picture_id_(0),
+ feedback_mode_(false),
+ cpu_speed_(-6), // default value
+ rc_max_intra_target_(0),
+ token_partitions_(VP8_ONE_TOKENPARTITION),
+ rps_(new ReferencePictureSelection),
+#if WEBRTC_LIBVPX_VERSION >= 971
+ temporal_layers_(NULL),
+#endif
+ encoder_(NULL),
+ config_(NULL),
+ raw_(NULL) {
+ uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
+ srand(seed);
+}
+
+VP8Encoder::~VP8Encoder() {
+ Release();
+ delete rps_;
+}
+
+int VP8Encoder::Release() {
+ if (encoded_image_._buffer != NULL) {
+ delete [] encoded_image_._buffer;
+ encoded_image_._buffer = NULL;
+ }
+ if (encoder_ != NULL) {
+ if (vpx_codec_destroy(encoder_)) {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ delete encoder_;
+ encoder_ = NULL;
+ }
+ if (config_ != NULL) {
+ delete config_;
+ config_ = NULL;
+ }
+ if (raw_ != NULL) {
+ vpx_img_free(raw_);
+ delete raw_;
+ raw_ = NULL;
+ }
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_ != NULL) {
+ delete temporal_layers_;
+ temporal_layers_ = NULL;
+ }
+#endif
+ inited_ = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Encoder::SetRates(uint32_t new_bitrate_kbit, uint32_t new_framerate) {
+ if (!inited_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (encoder_->err) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ if (new_framerate < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ // update bit rate
+ if (codec_.maxBitrate > 0 && new_bitrate_kbit > codec_.maxBitrate) {
+ new_bitrate_kbit = codec_.maxBitrate;
+ }
+ config_->rc_target_bitrate = new_bitrate_kbit; // in kbit/s
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_) {
+ temporal_layers_->ConfigureBitrates(new_bitrate_kbit, config_);
+ }
+#endif
+ codec_.maxFramerate = new_framerate;
+
+ // update encoder context
+ if (vpx_codec_enc_config_set(encoder_, config_)) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Encoder::InitEncode(const VideoCodec* inst,
+ int number_of_cores,
+ uint32_t /*max_payload_size*/) {
+ if (inst == NULL) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (inst->maxFramerate < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ // allow zero to represent an unspecified maxBitRate
+ if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (inst->width < 1 || inst->height < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (number_of_cores < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
+
+ int retVal = Release();
+ if (retVal < 0) {
+ return retVal;
+ }
+ if (encoder_ == NULL) {
+ encoder_ = new vpx_codec_ctx_t;
+ }
+ if (config_ == NULL) {
+ config_ = new vpx_codec_enc_cfg_t;
+ }
+ if (raw_ == NULL) {
+ raw_ = new vpx_image_t;
+ }
+ timestamp_ = 0;
+
+ codec_ = *inst;
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (inst->codecSpecific.VP8.numberOfTemporalLayers > 1) {
+ assert(temporal_layers_ == NULL);
+ temporal_layers_ =
+ new TemporalLayers(inst->codecSpecific.VP8.numberOfTemporalLayers);
+ }
+#endif
+ // random start 16 bits is enough.
+ picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
+
+ // allocate memory for encoded image
+ if (encoded_image_._buffer != NULL) {
+ delete [] encoded_image_._buffer;
+ }
+ encoded_image_._size = (3 * codec_.width * codec_.height) >> 1;
+ encoded_image_._buffer = new uint8_t[encoded_image_._size];
+ encoded_image_._completeFrame = true;
+
+ vpx_img_alloc(raw_, IMG_FMT_I420, codec_.width, codec_.height, 1);
+ // populate encoder configuration with default values
+ if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), config_, 0)) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ config_->g_w = codec_.width;
+ config_->g_h = codec_.height;
+ config_->rc_target_bitrate = inst->startBitrate; // in kbit/s
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_) {
+ temporal_layers_->ConfigureBitrates(inst->startBitrate, config_);
+ }
+#endif
+ // setting the time base of the codec
+ config_->g_timebase.num = 1;
+ config_->g_timebase.den = 90000;
+
+ // Set the error resilience mode according to user settings.
+ switch (inst->codecSpecific.VP8.resilience) {
+ case kResilienceOff:
+ config_->g_error_resilient = 0;
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_) {
+ // Must be on for temporal layers.
+ config_->g_error_resilient = 1;
+ }
+#endif
+ break;
+ case kResilientStream:
+ config_->g_error_resilient = 1; // TODO(holmer): Replace with
+ // VPX_ERROR_RESILIENT_DEFAULT when we
+ // drop support for libvpx 9.6.0.
+ break;
+ case kResilientFrames:
+#ifdef INDEPENDENT_PARTITIONS
+ config_->g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT |
+ VPX_ERROR_RESILIENT_PARTITIONS;
+ break;
+#else
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; // Not supported
+#endif
+ }
+ config_->g_lag_in_frames = 0; // 0- no frame lagging
+
+ // Determining number of threads based on the image size
+ if (codec_.width * codec_.height > 704 * 576 && number_of_cores > 1) {
+ // 2 threads when larger than 4CIF
+ config_->g_threads = 2;
+ } else {
+ config_->g_threads = 1;
+ }
+
+ // rate control settings
+ config_->rc_dropframe_thresh = 30;
+ config_->rc_end_usage = VPX_CBR;
+ config_->g_pass = VPX_RC_ONE_PASS;
+ config_->rc_resize_allowed = 0;
+ config_->rc_min_quantizer = 8;
+ config_->rc_max_quantizer = 56;
+ config_->rc_undershoot_pct = 100;
+ config_->rc_overshoot_pct = 15;
+ config_->rc_buf_initial_sz = 500;
+ config_->rc_buf_optimal_sz = 600;
+ config_->rc_buf_sz = 1000;
+ // set the maximum target size of any key-frame.
+ rc_max_intra_target_ = MaxIntraTarget(config_->rc_buf_optimal_sz);
+
+ if (feedback_mode_) {
+ // Disable periodic key frames if we get feedback from the decoder
+ // through SLI and RPSI.
+ config_->kf_mode = VPX_KF_DISABLED;
+ } else {
+ config_->kf_mode = VPX_KF_AUTO;
+ config_->kf_max_dist = 3000;
+ }
+ switch (inst->codecSpecific.VP8.complexity) {
+ case kComplexityHigh:
+ cpu_speed_ = -5;
+ break;
+ case kComplexityHigher:
+ cpu_speed_ = -4;
+ break;
+ case kComplexityMax:
+ cpu_speed_ = -3;
+ break;
+ default:
+ cpu_speed_ = -6;
+ break;
+ }
+ rps_->Init();
+ return InitAndSetControlSettings();
+}
+
+int VP8Encoder::InitAndSetControlSettings() {
+ vpx_codec_flags_t flags = 0;
+ // TODO(holmer): We should make a smarter decision on the number of
+ // partitions. Eight is probably not the optimal number for low resolution
+ // video.
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
+#endif
+ if (vpx_codec_enc_init(encoder_, vpx_codec_vp8_cx(), config_, flags)) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ vpx_codec_control(encoder_, VP8E_SET_STATIC_THRESHOLD, 800);
+ vpx_codec_control(encoder_, VP8E_SET_CPUUSED, cpu_speed_);
+ vpx_codec_control(encoder_, VP8E_SET_TOKEN_PARTITIONS,
+ static_cast<vp8e_token_partitions>(token_partitions_));
+ vpx_codec_control(encoder_, VP8E_SET_NOISE_SENSITIVITY, 2);
+#if WEBRTC_LIBVPX_VERSION >= 971
+ vpx_codec_control(encoder_, VP8E_SET_MAX_INTRA_BITRATE_PCT,
+ rc_max_intra_target_);
+#endif
+ inited_ = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+uint32_t VP8Encoder::MaxIntraTarget(uint32_t optimalBuffersize) {
+ // Set max to the optimal buffer level (normalized by target BR),
+ // and scaled by a scalePar.
+ // Max target size = scalePar * optimalBufferSize * targetBR[Kbps].
+ // This values is presented in percentage of perFrameBw:
+ // perFrameBw = targetBR[Kbps] * 1000 / frameRate.
+ // The target in % is as follows:
+
+ float scalePar = 0.5;
+ uint32_t targetPct = optimalBuffersize * scalePar * codec_.maxFramerate / 10;
+
+ // Don't go below 3 times the per frame bandwidth.
+ const uint32_t minIntraTh = 300;
+ return (targetPct < minIntraTh) ? minIntraTh: targetPct;
+}
+
+int VP8Encoder::Encode(const RawImage& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const VideoFrameType* frame_types) {
+ if (!inited_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (input_image._buffer == NULL) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (encoded_complete_callback_ == NULL) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ // image in vpx_image_t format
+ raw_->planes[PLANE_Y] = input_image._buffer;
+ raw_->planes[PLANE_U] = &input_image._buffer[codec_.height * codec_.width];
+ raw_->planes[PLANE_V] =
+ &input_image._buffer[codec_.height * codec_.width * 5 >> 2];
+
+ int flags = 0;
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_) {
+ flags |= temporal_layers_->EncodeFlags();
+ }
+#endif
+ bool send_keyframe = frame_types && (*frame_types == kKeyFrame);
+ if (send_keyframe) {
+ // Key frame request from caller.
+ // Will update both golden and alt-ref.
+ flags = VPX_EFLAG_FORCE_KF;
+ } else if (feedback_mode_ && codec_specific_info) {
+ // Handle RPSI and SLI messages and set up the appropriate encode flags.
+ bool sendRefresh = false;
+ if (codec_specific_info->codecType == kVideoCodecVP8) {
+ if (codec_specific_info->codecSpecific.VP8.hasReceivedRPSI) {
+ rps_->ReceivedRPSI(
+ codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
+ }
+ if (codec_specific_info->codecSpecific.VP8.hasReceivedSLI) {
+ sendRefresh = rps_->ReceivedSLI(input_image._timeStamp);
+ }
+ }
+ flags = rps_->EncodeFlags(picture_id_, sendRefresh, input_image._timeStamp);
+ }
+
+ // TODO(holmer): Ideally the duration should be the timestamp diff of this
+ // frame and the next frame to be encoded, which we don't have. Instead we
+ // would like to use the duration of the previous frame. Unfortunately the
+ // rate control seems to be off with that setup. Using the average input
+ // frame rate to calculate an average duration for now.
+ assert(codec_.maxFramerate > 0);
+ uint32_t duration = 90000 / codec_.maxFramerate;
+ if (vpx_codec_encode(encoder_, raw_, timestamp_, duration, flags,
+ VPX_DL_REALTIME)) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ timestamp_ += duration;
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ return GetEncodedPartitions(input_image);
+#else
+ return GetEncodedFrame(input_image);
+#endif
+}
+
+void VP8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+ const vpx_codec_cx_pkt& pkt) {
+ assert(codec_specific != NULL);
+ codec_specific->codecType = kVideoCodecVP8;
+ CodecSpecificInfoVP8 *vp8Info = &(codec_specific->codecSpecific.VP8);
+ vp8Info->pictureId = picture_id_;
+ vp8Info->simulcastIdx = 0;
+ vp8Info->keyIdx = kNoKeyIdx; // TODO(hlundin) populate this
+ vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (temporal_layers_) {
+ temporal_layers_->PopulateCodecSpecific(
+ (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? true : false, vp8Info);
+ } else {
+#endif
+ vp8Info->temporalIdx = kNoTemporalIdx;
+ vp8Info->layerSync = false;
+ vp8Info->tl0PicIdx = kNoTl0PicIdx;
+#if WEBRTC_LIBVPX_VERSION >= 971
+ }
+#endif
+ picture_id_ = (picture_id_ + 1) & 0x7FFF; // prepare next
+}
+
+int VP8Encoder::GetEncodedFrame(const RawImage& input_image) {
+ vpx_codec_iter_t iter = NULL;
+ encoded_image_._frameType = kDeltaFrame;
+ const vpx_codec_cx_pkt_t *pkt= vpx_codec_get_cx_data(encoder_, &iter);
+ if (pkt == NULL) {
+ if (!encoder_->err) {
+ // dropped frame
+ return WEBRTC_VIDEO_CODEC_OK;
+ } else {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ } else if (pkt->kind == VPX_CODEC_CX_FRAME_PKT) {
+ CodecSpecificInfo codecSpecific;
+ PopulateCodecSpecific(&codecSpecific, *pkt);
+
+ assert(pkt->data.frame.sz <= encoded_image_._size);
+ memcpy(encoded_image_._buffer, pkt->data.frame.buf, pkt->data.frame.sz);
+ encoded_image_._length = uint32_t(pkt->data.frame.sz);
+ encoded_image_._encodedHeight = raw_->h;
+ encoded_image_._encodedWidth = raw_->w;
+
+ // check if encoded frame is a key frame
+ if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
+ encoded_image_._frameType = kKeyFrame;
+ rps_->EncodedKeyFrame(picture_id_);
+ }
+
+ if (encoded_image_._length > 0) {
+ encoded_image_._timeStamp = input_image._timeStamp;
+
+ // Figure out where partition boundaries are located.
+ RTPFragmentationHeader fragInfo;
+ fragInfo.VerifyAndAllocateFragmentationHeader(2);
+ // two partitions: 1st and 2nd
+
+ // First partition
+ fragInfo.fragmentationOffset[0] = 0;
+ uint8_t *firstByte = encoded_image_._buffer;
+ uint32_t tmpSize = (firstByte[2] << 16) | (firstByte[1] << 8)
+ | firstByte[0];
+ fragInfo.fragmentationLength[0] = (tmpSize >> 5) & 0x7FFFF;
+ fragInfo.fragmentationPlType[0] = 0; // not known here
+ fragInfo.fragmentationTimeDiff[0] = 0;
+
+ // Second partition
+ fragInfo.fragmentationOffset[1] = fragInfo.fragmentationLength[0];
+ fragInfo.fragmentationLength[1] = encoded_image_._length -
+ fragInfo.fragmentationLength[0];
+ fragInfo.fragmentationPlType[1] = 0; // not known here
+ fragInfo.fragmentationTimeDiff[1] = 0;
+
+ encoded_complete_callback_->Encoded(encoded_image_, &codecSpecific,
+ &fragInfo);
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+ }
+ return WEBRTC_VIDEO_CODEC_ERROR;
+}
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+int VP8Encoder::GetEncodedPartitions(const RawImage& input_image) {
+ vpx_codec_iter_t iter = NULL;
+ int part_idx = 0;
+ encoded_image_._length = 0;
+ encoded_image_._frameType = kDeltaFrame;
+ RTPFragmentationHeader frag_info;
+ frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_) + 1);
+ CodecSpecificInfo codec_specific;
+
+ const vpx_codec_cx_pkt_t *pkt = NULL;
+ while ((pkt = vpx_codec_get_cx_data(encoder_, &iter)) != NULL) {
+ switch(pkt->kind) {
+ case VPX_CODEC_CX_FRAME_PKT: {
+ memcpy(&encoded_image_._buffer[encoded_image_._length],
+ pkt->data.frame.buf,
+ pkt->data.frame.sz);
+ frag_info.fragmentationOffset[part_idx] = encoded_image_._length;
+ frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
+ frag_info.fragmentationPlType[part_idx] = 0; // not known here
+ frag_info.fragmentationTimeDiff[part_idx] = 0;
+ encoded_image_._length += pkt->data.frame.sz;
+ assert(encoded_image_._length <= encoded_image_._size);
+ ++part_idx;
+ break;
+ }
+ default: {
+ break;
+ }
+ }
+ // End of frame
+ if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
+ // check if encoded frame is a key frame
+ if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
+ encoded_image_._frameType = kKeyFrame;
+ rps_->EncodedKeyFrame(picture_id_);
+ }
+ PopulateCodecSpecific(&codec_specific, *pkt);
+ break;
+ }
+ }
+ if (encoded_image_._length > 0) {
+ encoded_image_._timeStamp = input_image._timeStamp;
+ encoded_image_._encodedHeight = raw_->h;
+ encoded_image_._encodedWidth = raw_->w;
+ encoded_complete_callback_->Encoded(encoded_image_, &codec_specific,
+ &frag_info);
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+#endif
+
+int VP8Encoder::SetChannelParameters(uint32_t /*packet_loss*/, int rtt) {
+ rps_->SetRtt(rtt);
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Encoder::RegisterEncodeCompleteCallback(
+ EncodedImageCallback* callback) {
+ encoded_complete_callback_ = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VP8Decoder* VP8Decoder::Create() {
+ return new VP8Decoder();
+}
+
+VP8Decoder::VP8Decoder()
+ : decode_complete_callback_(NULL),
+ inited_(false),
+ feedback_mode_(false),
+ decoder_(NULL),
+ last_keyframe_(),
+ image_format_(VPX_IMG_FMT_NONE),
+ ref_frame_(NULL),
+ propagation_cnt_(-1),
+ latest_keyframe_complete_(false),
+ mfqe_enabled_(false) {
+}
+
+VP8Decoder::~VP8Decoder() {
+ inited_ = true; // in order to do the actual release
+ Release();
+}
+
+int VP8Decoder::Reset() {
+ if (!inited_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ InitDecode(&codec_, 1);
+ propagation_cnt_ = -1;
+ latest_keyframe_complete_ = false;
+ mfqe_enabled_ = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::InitDecode(const VideoCodec* inst, int number_of_cores) {
+ int ret_val = Release();
+ if (ret_val < 0 ) {
+ return ret_val;
+ }
+ if (decoder_ == NULL) {
+ decoder_ = new vpx_dec_ctx_t;
+ }
+ if (inst && inst->codecType == kVideoCodecVP8) {
+ feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
+ }
+ vpx_codec_dec_cfg_t cfg;
+ // Setting number of threads to a constant value (1)
+ cfg.threads = 1;
+ cfg.h = cfg.w = 0; // set after decode
+
+ vpx_codec_flags_t flags = 0;
+#if WEBRTC_LIBVPX_VERSION >= 971
+ flags = VPX_CODEC_USE_ERROR_CONCEALMENT | VPX_CODEC_USE_POSTPROC;
+#ifdef INDEPENDENT_PARTITIONS
+ flags |= VPX_CODEC_USE_INPUT_PARTITION;
+#endif
+#endif
+
+ if (vpx_codec_dec_init(decoder_, vpx_codec_vp8_dx(), &cfg, flags)) {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ vp8_postproc_cfg_t ppcfg;
+ // Disable deblocking for now due to uninitialized memory being returned.
+ ppcfg.post_proc_flag = 0;
+ // Strength of deblocking filter. Valid range:[0,16]
+ //ppcfg.deblocking_level = 3;
+ vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
+#endif
+
+ // Save VideoCodec instance for later; mainly for duplicating the decoder.
+ codec_ = *inst;
+ propagation_cnt_ = -1;
+ latest_keyframe_complete_ = false;
+
+ inited_ = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::Decode(const EncodedImage& input_image,
+ bool missing_frames,
+ const RTPFragmentationHeader* fragmentation,
+ const CodecSpecificInfo* codec_specific_info,
+ int64_t /*render_time_ms*/) {
+ if (!inited_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (decode_complete_callback_ == NULL) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (input_image._buffer == NULL && input_image._length > 0) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+
+#ifdef INDEPENDENT_PARTITIONS
+ if (fragmentation == NULL) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+#endif
+
+#if WEBRTC_LIBVPX_VERSION >= 971
+ if (!mfqe_enabled_ && codec_specific_info &&
+ codec_specific_info->codecSpecific.VP8.temporalIdx > 0) {
+ // Enable MFQE if we are receiving layers.
+ // temporalIdx is set in the jitter buffer according to what the RTP
+ // header says.
+ mfqe_enabled_ = true;
+ vp8_postproc_cfg_t ppcfg;
+ ppcfg.post_proc_flag = VP8_MFQE;
+ vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
+ }
+#endif
+
+ // Restrict error propagation using key frame requests. Disabled when
+ // the feedback mode is enabled (RPS).
+ // Reset on a key frame refresh.
+ if (!feedback_mode_) {
+ if (input_image._frameType == kKeyFrame && input_image._completeFrame)
+ propagation_cnt_ = -1;
+ // Start count on first loss.
+ else if ((!input_image._completeFrame || missing_frames) &&
+ propagation_cnt_ == -1)
+ propagation_cnt_ = 0;
+ if (propagation_cnt_ >= 0)
+ propagation_cnt_++;
+ }
+
+ vpx_dec_iter_t iter = NULL;
+ vpx_image_t* img;
+ int ret;
+
+ // Check for missing frames.
+ if (missing_frames) {
+ // Call decoder with zero data length to signal missing frames.
+ if (vpx_codec_decode(decoder_, NULL, 0, 0, VPX_DL_REALTIME)) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ img = vpx_codec_get_frame(decoder_, &iter);
+ iter = NULL;
+ }
+
+#ifdef INDEPENDENT_PARTITIONS
+ if (DecodePartitions(inputImage, fragmentation)) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0) {
+ propagation_cnt_ = 0;
+ }
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+#else
+ uint8_t* buffer = input_image._buffer;
+ if (input_image._length == 0) {
+ buffer = NULL; // Triggers full frame concealment.
+ }
+ if (vpx_codec_decode(decoder_,
+ buffer,
+ input_image._length,
+ 0,
+ VPX_DL_REALTIME)) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+#endif
+
+ // Store encoded frame if key frame. (Used in Copy method.)
+ if (input_image._frameType == kKeyFrame && input_image._buffer != NULL) {
+ const uint32_t bytes_to_copy = input_image._length;
+ if (last_keyframe_._size < bytes_to_copy) {
+ delete [] last_keyframe_._buffer;
+ last_keyframe_._buffer = NULL;
+ last_keyframe_._size = 0;
+ }
+
+ uint8_t* temp_buffer = last_keyframe_._buffer; // Save buffer ptr.
+ uint32_t temp_size = last_keyframe_._size; // Save size.
+ last_keyframe_ = input_image; // Shallow copy.
+ last_keyframe_._buffer = temp_buffer; // Restore buffer ptr.
+ last_keyframe_._size = temp_size; // Restore buffer size.
+ if (!last_keyframe_._buffer) {
+ // Allocate memory.
+ last_keyframe_._size = bytes_to_copy;
+ last_keyframe_._buffer = new uint8_t[last_keyframe_._size];
+ }
+ // Copy encoded frame.
+ memcpy(last_keyframe_._buffer, input_image._buffer, bytes_to_copy);
+ last_keyframe_._length = bytes_to_copy;
+ }
+
+ img = vpx_codec_get_frame(decoder_, &iter);
+ ret = ReturnFrame(img, input_image._timeStamp);
+ if (ret != 0) {
+ // Reset to avoid requesting key frames too often.
+ if (ret < 0 && propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return ret;
+ }
+ if (feedback_mode_) {
+ // Whenever we receive an incomplete key frame all reference buffers will
+ // be corrupt. If that happens we must request new key frames until we
+ // decode a complete.
+ if (input_image._frameType == kKeyFrame)
+ latest_keyframe_complete_ = input_image._completeFrame;
+ if (!latest_keyframe_complete_)
+ return WEBRTC_VIDEO_CODEC_ERROR;
+
+ // Check for reference updates and last reference buffer corruption and
+ // signal successful reference propagation or frame corruption to the
+ // encoder.
+ int reference_updates = 0;
+ if (vpx_codec_control(decoder_, VP8D_GET_LAST_REF_UPDATES,
+ &reference_updates)) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ int corrupted = 0;
+ if (vpx_codec_control(decoder_, VP8D_GET_FRAME_CORRUPTED, &corrupted)) {
+ // Reset to avoid requesting key frames too often.
+ if (propagation_cnt_ > 0)
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ int16_t picture_id = -1;
+ if (codec_specific_info) {
+ picture_id = codec_specific_info->codecSpecific.VP8.pictureId;
+ }
+ if (picture_id > -1) {
+ if (((reference_updates & VP8_GOLD_FRAME) ||
+ (reference_updates & VP8_ALTR_FRAME)) && !corrupted) {
+ decode_complete_callback_->ReceivedDecodedReferenceFrame(picture_id);
+ }
+ decode_complete_callback_->ReceivedDecodedFrame(picture_id);
+ }
+ if (corrupted) {
+ // we can decode but with artifacts
+ return WEBRTC_VIDEO_CODEC_REQUEST_SLI;
+ }
+ }
+ // Check Vs. threshold
+ if (propagation_cnt_ > kVp8ErrorPropagationTh) {
+ // Reset to avoid requesting key frames too often.
+ propagation_cnt_ = 0;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::DecodePartitions(
+ const EncodedImage& input_image,
+ const RTPFragmentationHeader* fragmentation) {
+ for (int i = 0; i < fragmentation->fragmentationVectorSize; ++i) {
+ const uint8_t* partition = input_image._buffer +
+ fragmentation->fragmentationOffset[i];
+ const uint32_t partition_length =
+ fragmentation->fragmentationLength[i];
+ if (vpx_codec_decode(decoder_,
+ partition,
+ partition_length,
+ 0,
+ VPX_DL_REALTIME)) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ }
+ // Signal end of frame data. If there was no frame data this will trigger
+ // a full frame concealment.
+ if (vpx_codec_decode(decoder_, NULL, 0, 0, VPX_DL_REALTIME))
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) {
+ if (img == NULL) {
+ // Decoder OK and NULL image => No show frame
+ return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+ }
+
+ // Allocate memory for decoded image
+ uint32_t required_size = (3 * img->d_h * img->d_w) >> 1;
+ if (required_size > decoded_image_._size) {
+ delete [] decoded_image_._buffer;
+ decoded_image_._buffer = NULL;
+ }
+ if (decoded_image_._buffer == NULL) {
+ decoded_image_._size = required_size;
+ decoded_image_._buffer = new uint8_t[decoded_image_._size];
+ }
+ uint8_t* buf;
+ uint32_t pos = 0;
+ uint32_t plane, y;
+
+ for (plane = 0; plane < 3; plane++) {
+ unsigned int width = (plane ? (img->d_w + 1) >> 1 : img->d_w);
+ unsigned int height = (plane ? (img->d_h + 1) >> 1 : img->d_h);
+ buf = img->planes[plane];
+ for(y = 0; y < height; y++) {
+ memcpy(&decoded_image_._buffer[pos], buf, width);
+ pos += width;
+ buf += img->stride[plane];
+ }
+ }
+
+ // Set image parameters
+ decoded_image_._height = img->d_h;
+ decoded_image_._width = img->d_w;
+ decoded_image_._length = (3 * img->d_h * img->d_w) >> 1;
+ decoded_image_._timeStamp = timestamp;
+ int ret = decode_complete_callback_->Decoded(decoded_image_);
+ if (ret != 0)
+ return ret;
+
+ // Remember image format for later
+ image_format_ = img->fmt;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::RegisterDecodeCompleteCallback(
+ DecodedImageCallback* callback) {
+ decode_complete_callback_ = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int VP8Decoder::Release() {
+ if (decoded_image_._buffer != NULL) {
+ delete [] decoded_image_._buffer;
+ decoded_image_._buffer = NULL;
+ }
+ if (last_keyframe_._buffer != NULL) {
+ delete [] last_keyframe_._buffer;
+ last_keyframe_._buffer = NULL;
+ }
+ if (decoder_ != NULL) {
+ if(vpx_codec_destroy(decoder_)) {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ delete decoder_;
+ decoder_ = NULL;
+ }
+ if (ref_frame_ != NULL) {
+ vpx_img_free(&ref_frame_->img);
+ delete ref_frame_;
+ ref_frame_ = NULL;
+ }
+ inited_ = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VideoDecoder* VP8Decoder::Copy() {
+ // Sanity checks.
+ if (!inited_) {
+ // Not initialized.
+ assert(false);
+ return NULL;
+ }
+ if (decoded_image_._buffer == NULL) {
+ // Nothing has been decoded before; cannot clone.
+ return NULL;
+ }
+ if (last_keyframe_._buffer == NULL) {
+ // Cannot clone if we have no key frame to start with.
+ return NULL;
+ }
+ // Create a new VideoDecoder object
+ VP8Decoder *copy = new VP8Decoder;
+
+ // Initialize the new decoder
+ if (copy->InitDecode(&codec_, 1) != WEBRTC_VIDEO_CODEC_OK) {
+ delete copy;
+ return NULL;
+ }
+ // Inject last key frame into new decoder.
+ if (vpx_codec_decode(copy->decoder_, last_keyframe_._buffer,
+ last_keyframe_._length, NULL, VPX_DL_REALTIME)) {
+ delete copy;
+ return NULL;
+ }
+ // Allocate memory for reference image copy
+ assert(decoded_image_._width > 0);
+ assert(decoded_image_._height > 0);
+ assert(image_format_ > VPX_IMG_FMT_NONE);
+ // Check if frame format has changed.
+ if (ref_frame_ &&
+ (decoded_image_._width != ref_frame_->img.d_w ||
+ decoded_image_._height != ref_frame_->img.d_h ||
+ image_format_ != ref_frame_->img.fmt)) {
+ vpx_img_free(&ref_frame_->img);
+ delete ref_frame_;
+ ref_frame_ = NULL;
+ }
+
+
+ if (!ref_frame_) {
+ ref_frame_ = new vpx_ref_frame_t;
+
+ if (!vpx_img_alloc(&ref_frame_->img,
+ static_cast<vpx_img_fmt_t>(image_format_),
+ decoded_image_._width, decoded_image_._height, 1)) {
+ assert(false);
+ delete copy;
+ return NULL;
+ }
+ }
+ const vpx_ref_frame_type_t type_vec[] = { VP8_LAST_FRAME, VP8_GOLD_FRAME,
+ VP8_ALTR_FRAME };
+ for (uint32_t ix = 0;
+ ix < sizeof(type_vec) / sizeof(vpx_ref_frame_type_t); ++ix) {
+ ref_frame_->frame_type = type_vec[ix];
+ if (CopyReference(copy) < 0) {
+ delete copy;
+ return NULL;
+ }
+ }
+ // Copy all member variables (that are not set in initialization).
+ copy->feedback_mode_ = feedback_mode_;
+ copy->image_format_ = image_format_;
+ copy->last_keyframe_ = last_keyframe_; // Shallow copy.
+ // Allocate memory. (Discard copied _buffer pointer.)
+ copy->last_keyframe_._buffer = new uint8_t[last_keyframe_._size];
+ memcpy(copy->last_keyframe_._buffer, last_keyframe_._buffer,
+ last_keyframe_._length);
+
+ // Initialize decoded_image_.
+ copy->decoded_image_ = decoded_image_; // Shallow copy
+ copy->decoded_image_._buffer = NULL;
+ if (decoded_image_._size) {
+ copy->decoded_image_._buffer = new uint8_t[decoded_image_._size];
+ }
+ return static_cast<VideoDecoder*>(copy);
+}
+
+int VP8Decoder::CopyReference(VP8Decoder* copyTo) {
+ // The type of frame to copy should be set in ref_frame_->frame_type
+ // before the call to this function.
+ if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_)
+ != VPX_CODEC_OK) {
+ return -1;
+ }
+ if (vpx_codec_control(copyTo->decoder_, VP8_SET_REFERENCE, ref_frame_)
+ != VPX_CODEC_OK) {
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi
new file mode 100644
index 0000000..89d8c78
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi
@@ -0,0 +1,121 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_vp8',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '<(webrtc_root)/common_video/interface',
+ '<(webrtc_root)/modules/video_coding/codecs/interface',
+ '<(webrtc_root)/modules/interface',
+ ],
+ 'conditions': [
+ ['build_with_chromium==1', {
+ 'dependencies': [
+ '<(webrtc_root)/../libvpx/libvpx.gyp:libvpx',
+ ],
+ 'defines': [
+ 'WEBRTC_LIBVPX_VERSION=960' # Bali
+ ],
+ },{
+ 'dependencies': [
+ '<(webrtc_root)/../third_party/libvpx/libvpx.gyp:libvpx',
+ ],
+ 'defines': [
+ 'WEBRTC_LIBVPX_VERSION=971' # Cayuga
+ ],
+ 'sources': [
+ 'temporal_layers.h',
+ 'temporal_layers.cc',
+ ],
+ }],
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '<(webrtc_root)/common_video/interface',
+ '<(webrtc_root)/modules/video_coding/codecs/interface',
+ ],
+ },
+ 'sources': [
+ 'reference_picture_selection.h',
+ 'reference_picture_selection.cc',
+ '../interface/vp8.h',
+ '../interface/vp8_common_types.h',
+ 'vp8.cc',
+ ],
+ },
+ ], # targets
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'vp8_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'test_framework',
+ 'webrtc_vp8',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ # header files
+ '../test/benchmark.h',
+ '../test/dual_decoder_test.h',
+ '../test/normal_async_test.h',
+ '../test/packet_loss_test.h',
+ '../test/rps_test.h',
+ '../test/unit_test.h',
+
+ # source files
+ '../test/benchmark.cc',
+ '../test/dual_decoder_test.cc',
+ '../test/normal_async_test.cc',
+ '../test/packet_loss_test.cc',
+ '../test/rps_test.cc',
+ '../test/tester.cc',
+ '../test/unit_test.cc',
+ ],
+ },
+ {
+ 'target_name': 'vp8_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../third_party/libvpx/libvpx.gyp:libvpx',
+ 'webrtc_vp8',
+ ],
+ 'include_dirs': [
+ '<(webrtc_root)/../third_party/libvpx/source/libvpx',
+ ],
+ 'sources': [
+ 'reference_picture_selection_unittest.cc',
+ 'temporal_layers_unittest.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc
new file mode 100644
index 0000000..4fc0e25
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "benchmark.h"
+#include "testsupport/fileutils.h"
+#include "vp8.h"
+
+using namespace webrtc;
+
+VP8Benchmark::VP8Benchmark()
+ : Benchmark("VP8Benchmark", "VP8 benchmark over a range of test cases",
+ webrtc::test::OutputPath() + "VP8Benchmark.txt", "VP8") {
+}
+
+VP8Benchmark::VP8Benchmark(std::string name, std::string description)
+ : Benchmark(name, description,
+ webrtc::test::OutputPath() + "VP8Benchmark.txt",
+ "VP8") {
+}
+
+VP8Benchmark::VP8Benchmark(std::string name, std::string description,
+ std::string resultsFileName)
+ : Benchmark(name, description, resultsFileName, "VP8") {
+}
+
+VideoEncoder* VP8Benchmark::GetNewEncoder() {
+ return VP8Encoder::Create();
+}
+
+VideoDecoder* VP8Benchmark::GetNewDecoder() {
+ return VP8Decoder::Create();
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h
new file mode 100644
index 0000000..5143de3
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
+
+#include "../../../test_framework/benchmark.h"
+
+class VP8Benchmark : public Benchmark
+{
+public:
+ VP8Benchmark();
+ VP8Benchmark(std::string name, std::string description);
+ VP8Benchmark(std::string name, std::string description, std::string resultsFileName);
+
+protected:
+ virtual webrtc::VideoEncoder* GetNewEncoder();
+ virtual webrtc::VideoDecoder* GetNewDecoder();
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc
new file mode 100644
index 0000000..478b034
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc
@@ -0,0 +1,221 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dual_decoder_test.h"
+
+#include <assert.h>
+#include <string.h> // memcmp
+#include <time.h>
+
+#include "testsupport/fileutils.h"
+
+VP8DualDecoderTest::VP8DualDecoderTest(float bitRate)
+:
+VP8NormalAsyncTest(bitRate)
+{
+ _decoder2 = NULL;
+}
+
+VP8DualDecoderTest::VP8DualDecoderTest()
+:
+VP8NormalAsyncTest("VP8 Dual Decoder Test", "Tests VP8 dual decoder", 1),
+_decoder2(NULL)
+{}
+
+VP8DualDecoderTest::~VP8DualDecoderTest()
+{
+ if(_decoder2)
+ {
+ _decoder2->Release();
+ delete _decoder2;
+ }
+
+ _decodedVideoBuffer2.Free();
+}
+
+void
+VP8DualDecoderTest::Perform()
+{
+ _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
+ CodecSettings(352, 288, 30, _bitRate);
+ Setup();
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer2.VerifyAndAllocate(_lengthSourceFrame);
+ if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
+ {
+ exit(EXIT_FAILURE);
+ }
+ _decoder->InitDecode(&_inst,1);
+
+ FrameQueue frameQueue;
+ VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+ DualDecoderCompleteCallback decCallback(&_decodedVideoBuffer);
+ DualDecoderCompleteCallback decCallback2(&_decodedVideoBuffer2);
+ _encoder->RegisterEncodeCompleteCallback(&encCallback);
+ _decoder->RegisterDecodeCompleteCallback(&decCallback);
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ {
+ exit(EXIT_FAILURE);
+ }
+ _totalEncodeTime = _totalDecodeTime = 0;
+ _totalEncodePipeTime = _totalDecodePipeTime = 0;
+ bool complete = false;
+ _framecnt = 0;
+ _encFrameCnt = 0;
+ _decFrameCnt = 0;
+ _sumEncBytes = 0;
+ _lengthEncFrame = 0;
+ double starttime = clock()/(double)CLOCKS_PER_SEC;
+ while (!complete)
+ {
+ if (_encFrameCnt == 10)
+ {
+ // initialize second decoder and copy state
+ _decoder2 = static_cast<webrtc::VP8Decoder *>(_decoder->Copy());
+ assert(_decoder2 != NULL);
+ _decoder2->RegisterDecodeCompleteCallback(&decCallback2);
+ }
+ CodecSpecific_InitBitrate();
+ complete = Encode();
+ if (!frameQueue.Empty() || complete)
+ {
+ while (!frameQueue.Empty())
+ {
+ _frameToDecode =
+ static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+ int lost = DoPacketLoss();
+ if (lost == 2)
+ {
+ // Lost the whole frame, continue
+ _missingFrames = true;
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ continue;
+ }
+ int ret = Decode(lost);
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ if (ret < 0)
+ {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+ exit(EXIT_FAILURE);
+ }
+ else if (ret == 0)
+ {
+ _framecnt++;
+ }
+ else
+ {
+ fprintf(stderr,
+ "\n\nPositive return value from decode!\n\n");
+ }
+ }
+ }
+ }
+ double endtime = clock()/(double)CLOCKS_PER_SEC;
+ double totalExecutionTime = endtime - starttime;
+ printf("Total execution time: %.1f s\n", totalExecutionTime);
+ _sumEncBytes = encCallback.EncodedBytes();
+ double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
+ double avgEncTime = _totalEncodeTime / _encFrameCnt;
+ double avgDecTime = _totalDecodeTime / _decFrameCnt;
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
+ printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
+ printf("Average encode pipeline time: %.1f ms\n",
+ 1000 * _totalEncodePipeTime / _encFrameCnt);
+ printf("Average decode pipeline time: %.1f ms\n",
+ 1000 * _totalDecodePipeTime / _decFrameCnt);
+ printf("Number of encoded frames: %u\n", _encFrameCnt);
+ printf("Number of decoded frames: %u\n", _decFrameCnt);
+ (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
+ _bitRate << " kbps" << std::endl;
+ (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+ _encoder->Release();
+ _decoder->Release();
+ Teardown();
+}
+
+
+int
+VP8DualDecoderTest::Decode(int lossValue)
+{
+ _sumEncBytes += _frameToDecode->_frame->GetLength();
+ webrtc::EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+ encodedImage._completeFrame = !lossValue;
+ _decodeCompleteTime = 0;
+ _decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
+ int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo);
+ // second decoder
+ if (_decoder2)
+ {
+ int ret2 = _decoder2->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo,
+ 0 /* dummy */);
+
+ // check return values
+ if (ret < 0 || ret2 < 0 || ret2 != ret)
+ {
+ exit(EXIT_FAILURE);
+ }
+
+ // compare decoded images
+ if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
+ _decodedVideoBuffer.GetLength(),
+ _decodedVideoBuffer2.GetBuffer(), _decodedVideoBuffer.GetLength()))
+ {
+ fprintf(stderr,"\n\nClone output different from master.\n\n");
+ exit(EXIT_FAILURE);
+ }
+
+ }
+
+ _missingFrames = false;
+ return ret;
+}
+
+
+bool
+VP8DualDecoderTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+ const void* ptrB, unsigned int bLengthBytes)
+{
+ if (aLengthBytes != bLengthBytes)
+ {
+ return false;
+ }
+
+ return memcmp(ptrA, ptrB, aLengthBytes) == 0;
+}
+
+WebRtc_Word32 DualDecoderCompleteCallback::Decoded(webrtc::RawImage& image)
+{
+ _decodedVideoBuffer->VerifyAndAllocate(image._length);
+ _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
+ _decodedVideoBuffer->SetWidth(image._width);
+ _decodedVideoBuffer->SetHeight(image._height);
+ _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
+ _decodeComplete = true;
+ return 0;
+}
+
+bool DualDecoderCompleteCallback::DecodeComplete()
+{
+ if (_decodeComplete)
+ {
+ _decodeComplete = false;
+ return true;
+ }
+ return false;
+}
+
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h
new file mode 100644
index 0000000..4af4e3e
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
+
+#include "vp8.h"
+#include "normal_async_test.h"
+
+class DualDecoderCompleteCallback;
+
+class VP8DualDecoderTest : public VP8NormalAsyncTest
+{
+public:
+ VP8DualDecoderTest(float bitRate);
+ VP8DualDecoderTest();
+ virtual ~VP8DualDecoderTest();
+ virtual void Perform();
+protected:
+ VP8DualDecoderTest(std::string name, std::string description,
+ unsigned int testNo)
+ : VP8NormalAsyncTest(name, description, testNo) {}
+ virtual int Decode(int lossValue = 0);
+
+ webrtc::VP8Decoder* _decoder2;
+ TestVideoBuffer _decodedVideoBuffer2;
+ static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+ const void *ptrB, unsigned int bLengthBytes);
+private:
+};
+
+class DualDecoderCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+ DualDecoderCompleteCallback(TestVideoBuffer* buffer)
+ : _decodedVideoBuffer(buffer), _decodeComplete(false) {}
+ WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
+ bool DecodeComplete();
+private:
+ TestVideoBuffer* _decodedVideoBuffer;
+ bool _decodeComplete;
+};
+
+
+#endif
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc
new file mode 100644
index 0000000..9ed508b
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_async_test.h"
+
+using namespace webrtc;
+
+VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate) :
+ NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, 1),
+ _hasReceivedRPSI(false)
+{
+}
+
+VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo):
+ NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, testNo),
+ _hasReceivedRPSI(false)
+{
+}
+
+void
+VP8NormalAsyncTest::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
+{
+ if (bitRate > 0)
+ {
+ _bitRate = bitRate;
+
+ }else if (_bitRate == 0)
+ {
+ _bitRate = 600;
+ }
+ _inst.codecType = kVideoCodecVP8;
+ _inst.codecSpecific.VP8.feedbackModeOn = true;
+ _inst.codecSpecific.VP8.pictureLossIndicationOn = true;
+ _inst.codecSpecific.VP8.complexity = kComplexityNormal;
+ _inst.maxFramerate = (unsigned char)frameRate;
+ _inst.startBitrate = _bitRate;
+ _inst.maxBitrate = 8000;
+ _inst.width = width;
+ _inst.height = height;
+}
+
+void
+VP8NormalAsyncTest::CodecSpecific_InitBitrate()
+{
+ if (_bitRate == 0)
+ {
+ _encoder->SetRates(600, _inst.maxFramerate);
+ }else
+ {
+ _encoder->SetRates(_bitRate, _inst.maxFramerate);
+ }
+}
+
+WebRtc_Word32
+VP8NormalAsyncTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
+{
+ _pictureIdRPSI = pictureId;
+ _hasReceivedRPSI = true;
+ return 0;
+}
+
+CodecSpecificInfo*
+VP8NormalAsyncTest::CreateEncoderSpecificInfo() const
+{
+ CodecSpecificInfo* vp8CodecSpecificInfo = new CodecSpecificInfo();
+ vp8CodecSpecificInfo->codecType = kVideoCodecVP8;
+ vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
+
+ _hasReceivedSLI = false;
+ _hasReceivedRPSI = false;
+
+ return vp8CodecSpecificInfo;
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h
new file mode 100644
index 0000000..f49c812
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
+
+#include "../../../test_framework/normal_async_test.h"
+
+class VP8NormalAsyncTest : public NormalAsyncTest
+{
+public:
+ VP8NormalAsyncTest(WebRtc_UWord32 bitRate);
+ VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo);
+ VP8NormalAsyncTest() : NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", 1) {}
+protected:
+ VP8NormalAsyncTest(std::string name, std::string description, unsigned int testNo) : NormalAsyncTest(name, description, testNo) {}
+ virtual void CodecSpecific_InitBitrate();
+ virtual void CodecSettings(int width, int height, WebRtc_UWord32 frameRate=30, WebRtc_UWord32 bitRate=0);
+ virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+private:
+ mutable bool _hasReceivedRPSI;
+ WebRtc_UWord64 _pictureIdRPSI;
+};
+
+#endif
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc
new file mode 100644
index 0000000..69e028a
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet_loss_test.h"
+#include <cassert>
+
+VP8PacketLossTest::VP8PacketLossTest()
+:
+PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode")
+{
+}
+
+VP8PacketLossTest::VP8PacketLossTest(std::string name, std::string description)
+:
+PacketLossTest(name, description)
+{
+}
+
+VP8PacketLossTest::VP8PacketLossTest(double lossRate,
+ bool useNack,
+ int rttFrames)
+:
+PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode",
+ lossRate, useNack, rttFrames)
+{
+}
+
+int VP8PacketLossTest::ByteLoss(int size, unsigned char* /* pkg */, int bytesToLose)
+{
+ int retLength = size - bytesToLose;
+ if (retLength < 4)
+ {
+ retLength = 4;
+ }
+ return retLength;
+}
+
+WebRtc_Word32
+VP8PacketLossTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
+{
+ _pictureIdRPSI = pictureId;
+ _hasReceivedRPSI = true;
+ return 0;
+}
+
+webrtc::CodecSpecificInfo*
+VP8PacketLossTest::CreateEncoderSpecificInfo() const
+{
+ webrtc::CodecSpecificInfo* vp8CodecSpecificInfo =
+ new webrtc::CodecSpecificInfo();
+ vp8CodecSpecificInfo->codecType = webrtc::kVideoCodecVP8;
+ vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
+ vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
+
+ _hasReceivedSLI = false;
+ _hasReceivedRPSI = false;
+
+ return vp8CodecSpecificInfo;
+}
+
+bool VP8PacketLossTest::PacketLoss(double lossRate, int numLosses) {
+ if (numLosses)
+ return true;
+ return RandUniform() < lossRate;
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h
new file mode 100644
index 0000000..262ee2f
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
+
+#include "../../../test_framework/packet_loss_test.h"
+
+class VP8PacketLossTest : public PacketLossTest
+{
+public:
+ VP8PacketLossTest();
+ VP8PacketLossTest(double lossRate, bool useNack, int rttFrames);
+
+protected:
+ VP8PacketLossTest(std::string name, std::string description);
+ virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
+ WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+ // |lossRate| is the probability of packet loss between 0 and 1.
+ // |numLosses| is the number of packets already lost in the current frame.
+ virtual bool PacketLoss(double lossRate, int numLosses);
+
+ webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
+
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc
new file mode 100644
index 0000000..cd1c42e
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc
@@ -0,0 +1,311 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rps_test.h"
+
+#include <assert.h>
+#include <string.h> // memcmp
+#include <time.h>
+
+#include "vp8.h"
+
+VP8RpsTest::VP8RpsTest(float bitRate)
+ : VP8NormalAsyncTest(bitRate),
+ decoder2_(webrtc::VP8Decoder::Create()),
+ sli_(false) {
+}
+
+VP8RpsTest::VP8RpsTest()
+ : VP8NormalAsyncTest("VP8 Reference Picture Selection Test",
+ "VP8 Reference Picture Selection Test", 1),
+ decoder2_(webrtc::VP8Decoder::Create()),
+ sli_(false) {
+}
+
+VP8RpsTest::~VP8RpsTest() {
+ if (decoder2_) {
+ decoder2_->Release();
+ delete decoder2_;
+ }
+ decoded_frame2_.Free();
+}
+
+void VP8RpsTest::Perform() {
+ _inname = "test/testFiles/foreman_cif.yuv";
+ CodecSettings(352, 288, 30, _bitRate);
+ Setup();
+ _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+ decoded_frame2_.VerifyAndAllocate(_lengthSourceFrame);
+
+ // Enable RPS functionality
+ _inst.codecSpecific.VP8.pictureLossIndicationOn = true;
+ _inst.codecSpecific.VP8.feedbackModeOn = true;
+
+ if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
+ exit(EXIT_FAILURE);
+
+ _decoder->InitDecode(&_inst,1);
+ decoder2_->InitDecode(&_inst,1);
+
+ FrameQueue frameQueue;
+ VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+ RpsDecodeCompleteCallback decCallback(&_decodedVideoBuffer);
+ RpsDecodeCompleteCallback decCallback2(&decoded_frame2_);
+ _encoder->RegisterEncodeCompleteCallback(&encCallback);
+ _decoder->RegisterDecodeCompleteCallback(&decCallback);
+ decoder2_->RegisterDecodeCompleteCallback(&decCallback2);
+
+ if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+ exit(EXIT_FAILURE);
+
+ _totalEncodeTime = _totalDecodeTime = 0;
+ _totalEncodePipeTime = _totalDecodePipeTime = 0;
+ bool complete = false;
+ _framecnt = 0;
+ _encFrameCnt = 0;
+ _decFrameCnt = 0;
+ _sumEncBytes = 0;
+ _lengthEncFrame = 0;
+ double starttime = clock()/(double)CLOCKS_PER_SEC;
+ while (!complete) {
+ CodecSpecific_InitBitrate();
+ complete = EncodeRps(&decCallback2);
+ if (!frameQueue.Empty() || complete) {
+ while (!frameQueue.Empty()) {
+ _frameToDecode =
+ static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+ int lost = DoPacketLoss();
+ if (lost == 2) {
+ // Lost the whole frame, continue
+ _missingFrames = true;
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ continue;
+ }
+ int ret = Decode(lost);
+ delete _frameToDecode;
+ _frameToDecode = NULL;
+ if (ret < 0) {
+ fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+ exit(EXIT_FAILURE);
+ }
+ else if (ret == 0) {
+ _framecnt++;
+ }
+ else {
+ fprintf(stderr,
+ "\n\nPositive return value from decode!\n\n");
+ }
+ }
+ }
+ }
+ double endtime = clock()/(double)CLOCKS_PER_SEC;
+ double totalExecutionTime = endtime - starttime;
+ printf("Total execution time: %.1f s\n", totalExecutionTime);
+ _sumEncBytes = encCallback.EncodedBytes();
+ double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
+ double avgEncTime = _totalEncodeTime / _encFrameCnt;
+ double avgDecTime = _totalDecodeTime / _decFrameCnt;
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
+ printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
+ printf("Average encode pipeline time: %.1f ms\n",
+ 1000 * _totalEncodePipeTime / _encFrameCnt);
+ printf("Average decode pipeline time: %.1f ms\n",
+ 1000 * _totalDecodePipeTime / _decFrameCnt);
+ printf("Number of encoded frames: %u\n", _encFrameCnt);
+ printf("Number of decoded frames: %u\n", _decFrameCnt);
+ (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
+ _bitRate << " kbps" << std::endl;
+ (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+ _encoder->Release();
+ _decoder->Release();
+ Teardown();
+}
+
+bool VP8RpsTest::EncodeRps(RpsDecodeCompleteCallback* decodeCallback) {
+ _lengthEncFrame = 0;
+ size_t bytes_read = fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+ if (bytes_read < _lengthSourceFrame)
+ return true;
+ _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+ _inputVideoBuffer.SetTimeStamp((unsigned int)
+ (_encFrameCnt * 9e4 / _inst.maxFramerate));
+ _inputVideoBuffer.SetWidth(_inst.width);
+ _inputVideoBuffer.SetHeight(_inst.height);
+ webrtc::RawImage rawImage;
+ VideoBufferToRawImage(_inputVideoBuffer, rawImage);
+ if (feof(_sourceFile) != 0) {
+ return true;
+ }
+ _encodeCompleteTime = 0;
+ _encodeTimes[rawImage._timeStamp] = tGetTime();
+ webrtc::VideoFrameType frameType = webrtc::kDeltaFrame;
+
+ webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
+ codecSpecificInfo->codecSpecific.VP8.pictureIdRPSI =
+ decodeCallback->LastDecodedRefPictureId(
+ &codecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI);
+ if (sli_) {
+ codecSpecificInfo->codecSpecific.VP8.pictureIdSLI =
+ decodeCallback->LastDecodedPictureId();
+ codecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = true;
+ sli_ = false;
+ }
+ printf("Encoding: %u\n", _framecnt);
+ int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
+ if (ret < 0)
+ printf("Failed to encode: %u\n", _framecnt);
+
+ if (codecSpecificInfo != NULL) {
+ delete codecSpecificInfo;
+ codecSpecificInfo = NULL;
+ }
+ if (_encodeCompleteTime > 0) {
+ _totalEncodeTime += _encodeCompleteTime -
+ _encodeTimes[rawImage._timeStamp];
+ }
+ else {
+ _totalEncodeTime += tGetTime() - _encodeTimes[rawImage._timeStamp];
+ }
+ return false;
+}
+
+//#define FRAME_LOSS 1
+
+int VP8RpsTest::Decode(int lossValue) {
+ _sumEncBytes += _frameToDecode->_frame->GetLength();
+ webrtc::EncodedImage encodedImage;
+ VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+ encodedImage._completeFrame = !lossValue;
+ _decodeCompleteTime = 0;
+ _decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
+ int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo);
+ // Drop every 10th frame for the second decoder
+#if FRAME_LOSS
+ if (_framecnt == 0 || _framecnt % 10 != 0) {
+ printf("Decoding: %u\n", _framecnt);
+ if (_framecnt > 1 && (_framecnt - 1) % 10 == 0)
+ _missingFrames = true;
+#else
+ if (true) {
+ if (_framecnt > 0 && _framecnt % 10 == 0) {
+ encodedImage._length = std::rand() % encodedImage._length;
+ printf("Decoding with loss: %u\n", _framecnt);
+ }
+ else
+ printf("Decoding: %u\n", _framecnt);
+#endif
+ int ret2 = decoder2_->Decode(encodedImage, _missingFrames, NULL,
+ _frameToDecode->_codecSpecificInfo,
+ 0 /* dummy */);
+
+ // check return values
+ if (ret < 0 || ret2 < 0) {
+ return -1;
+ } else if (ret2 == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI ||
+ ret2 == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
+ sli_ = true;
+ }
+
+ // compare decoded images
+#if FRAME_LOSS
+ if (!_missingFrames) {
+ if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
+ _decodedVideoBuffer.GetLength(),
+ decoded_frame2_.GetBuffer(), _decodedVideoBuffer.GetLength())) {
+ fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
+ _framecnt);
+ return -1;
+ }
+ }
+#else
+ if (_framecnt > 0 && _framecnt % 10 != 0) {
+ if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
+ _decodedVideoBuffer.GetLength(),
+ decoded_frame2_.GetBuffer(), _decodedVideoBuffer.GetLength())) {
+ fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
+ _framecnt);
+ return -1;
+ }
+ }
+#endif
+ }
+#if FRAME_LOSS
+ else
+ printf("Dropping %u\n", _framecnt);
+#endif
+ _missingFrames = false;
+ return 0;
+}
+
+
+bool
+VP8RpsTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+ const void* ptrB, unsigned int bLengthBytes) {
+ if (aLengthBytes != bLengthBytes)
+ return false;
+ return memcmp(ptrA, ptrB, aLengthBytes) == 0;
+}
+
+RpsDecodeCompleteCallback::RpsDecodeCompleteCallback(TestVideoBuffer* buffer)
+ : decoded_frame_(buffer),
+ decode_complete_(false),
+ last_decoded_picture_id_(0),
+ last_decoded_ref_picture_id_(0),
+ updated_ref_picture_id_(false) {
+}
+
+WebRtc_Word32 RpsDecodeCompleteCallback::Decoded(webrtc::RawImage& image) {
+ decoded_frame_->VerifyAndAllocate(image._length);
+ decoded_frame_->CopyBuffer(image._length, image._buffer);
+ decoded_frame_->SetWidth(image._width);
+ decoded_frame_->SetHeight(image._height);
+ decoded_frame_->SetTimeStamp(image._timeStamp);
+ decode_complete_ = true;
+ return 0;
+}
+
+bool RpsDecodeCompleteCallback::DecodeComplete() {
+ if (decode_complete_)
+ {
+ decode_complete_ = false;
+ return true;
+ }
+ return false;
+}
+
+WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 picture_id) {
+ last_decoded_ref_picture_id_ = picture_id & 0x7FFF;
+ updated_ref_picture_id_ = true;
+ return 0;
+}
+
+WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedFrame(
+ const WebRtc_UWord64 picture_id) {
+ last_decoded_picture_id_ = picture_id & 0x3F;
+ return 0;
+}
+
+WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedPictureId() const {
+ return last_decoded_picture_id_;
+}
+
+WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedRefPictureId(
+ bool *updated) {
+ if (updated)
+ *updated = updated_ref_picture_id_;
+ updated_ref_picture_id_ = false;
+ return last_decoded_ref_picture_id_;
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h
new file mode 100644
index 0000000..cbdbfa3
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
+
+#include "vp8.h"
+#include "normal_async_test.h"
+
+class RpsDecodeCompleteCallback;
+
+class VP8RpsTest : public VP8NormalAsyncTest {
+ public:
+ VP8RpsTest(float bitRate);
+ VP8RpsTest();
+ virtual ~VP8RpsTest();
+ virtual void Perform();
+ private:
+ VP8RpsTest(std::string name, std::string description, unsigned int testNo)
+ : VP8NormalAsyncTest(name, description, testNo) {}
+ virtual bool EncodeRps(RpsDecodeCompleteCallback* decodeCallback);
+ virtual int Decode(int lossValue = 0);
+
+ static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+ const void *ptrB, unsigned int bLengthBytes);
+
+ webrtc::VP8Decoder* decoder2_;
+ TestVideoBuffer decoded_frame2_;
+ bool sli_;
+};
+
+class RpsDecodeCompleteCallback : public webrtc::DecodedImageCallback {
+ public:
+ RpsDecodeCompleteCallback(TestVideoBuffer* buffer);
+ WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
+ bool DecodeComplete();
+ WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 picture_id);
+ WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 picture_id);
+ WebRtc_UWord64 LastDecodedPictureId() const;
+ WebRtc_UWord64 LastDecodedRefPictureId(bool *updated);
+
+ private:
+ TestVideoBuffer* decoded_frame_;
+ bool decode_complete_;
+ WebRtc_UWord64 last_decoded_picture_id_;
+ WebRtc_UWord64 last_decoded_ref_picture_id_;
+ bool updated_ref_picture_id_;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc
new file mode 100644
index 0000000..350d9a3
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+#include "benchmark.h"
+#include "dual_decoder_test.h"
+#include "normal_async_test.h"
+#include "packet_loss_test.h"
+#include "unit_test.h"
+#include "rps_test.h"
+#include "testsupport/fileutils.h"
+#include "vp8.h"
+
+using namespace webrtc;
+
+void PopulateTests(std::vector<Test*>* tests)
+{
+// tests->push_back(new VP8RpsTest());
+// tests->push_back(new VP8UnitTest());
+// tests->push_back(new VP8DualDecoderTest());
+// tests->push_back(new VP8Benchmark());
+// tests->push_back(new VP8PacketLossTest(0.05, false, 5));
+ tests->push_back(new VP8NormalAsyncTest());
+}
+
+int main()
+{
+ VP8Encoder* enc;
+ VP8Decoder* dec;
+ std::vector<Test*> tests;
+ PopulateTests(&tests);
+ std::fstream log;
+ std::string log_file = webrtc::test::OutputPath() + "VP8_test_log.txt";
+ log.open(log_file.c_str(), std::fstream::out | std::fstream::app);
+ std::vector<Test*>::iterator it;
+ for (it = tests.begin() ; it < tests.end(); it++)
+ {
+ enc = VP8Encoder::Create();
+ dec = VP8Decoder::Create();
+ (*it)->SetEncoder(enc);
+ (*it)->SetDecoder(dec);
+ (*it)->SetLog(&log);
+ (*it)->Perform();
+ (*it)->Print();
+ delete enc;
+ delete dec;
+ delete *it;
+ }
+ log.close();
+ tests.pop_back();
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc
new file mode 100644
index 0000000..5ba35ec
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc
@@ -0,0 +1,161 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "unit_test.h"
+
+#include <string.h>
+
+#include "../../../test_framework/video_source.h"
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+#include "vp8.h"
+
+using namespace webrtc;
+
+VP8UnitTest::VP8UnitTest()
+:
+UnitTest("VP8UnitTest", "Unit test")
+{
+}
+
+VP8UnitTest::VP8UnitTest(std::string name, std::string description)
+:
+UnitTest(name, description)
+{
+}
+
+WebRtc_UWord32
+VP8UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate, WebRtc_UWord32 /*frameRate*/)
+{
+ int rate = _encoder->SetRates(bitRate, _inst.maxFramerate);
+ EXPECT_TRUE(rate >= 0);
+ return rate;
+}
+
+bool
+VP8UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+ const void* ptrB, unsigned int bLengthBytes)
+{
+ const unsigned char* cPtrA = (const unsigned char*)ptrA;
+ const unsigned char* cPtrB = (const unsigned char*)ptrB;
+ // Skip picture ID before comparing
+ int aSkip = PicIdLength(cPtrA);
+ int bSkip = PicIdLength(cPtrB);
+ return UnitTest::CheckIfBitExact(cPtrA + aSkip, aLengthBytes,
+ cPtrB + bSkip, bLengthBytes);
+}
+
+int
+VP8UnitTest::PicIdLength(const unsigned char* ptr)
+{
+ WebRtc_UWord8 numberOfBytes;
+ WebRtc_UWord64 pictureID = 0;
+ for (numberOfBytes = 0; (ptr[numberOfBytes] & 0x80) && numberOfBytes < 8; numberOfBytes++)
+ {
+ pictureID += ptr[numberOfBytes] & 0x7f;
+ pictureID <<= 7;
+ }
+ pictureID += ptr[numberOfBytes] & 0x7f;
+ numberOfBytes++;
+ return numberOfBytes;
+}
+
+void
+VP8UnitTest::Perform()
+{
+ Setup();
+ FILE *outFile = NULL;
+ std::string outFileName;
+ VP8Encoder* enc = (VP8Encoder*)_encoder;
+ VP8Decoder* dec = (VP8Decoder*)_decoder;
+
+ //----- Encoder parameter tests -----
+ //-- Calls before InitEncode() --
+ EXPECT_EQ(enc->Release(), WEBRTC_VIDEO_CODEC_OK);
+ EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
+ WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+ EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
+ WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+ // EXPECT_TRUE(enc->GetCodecConfigParameters(configParameters, sizeof(configParameters)) ==
+ // WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+
+ VideoCodec codecInst;
+ strncpy(codecInst.plName, "VP8", 31);
+ codecInst.plType = 126;
+ codecInst.maxBitrate = 0;
+ codecInst.minBitrate = 0;
+ codecInst.width = 1440;
+ codecInst.height = 1080;
+ codecInst.maxFramerate = 30;
+ codecInst.startBitrate = 300;
+ codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+ EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
+
+
+ //-- Test two problematic level settings --
+ strncpy(codecInst.plName, "VP8", 31);
+ codecInst.plType = 126;
+ codecInst.maxBitrate = 0;
+ codecInst.minBitrate = 0;
+ codecInst.width = 352;
+ codecInst.height = 288;
+ codecInst.maxFramerate = 30;
+ codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+ codecInst.startBitrate = 300;
+ EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
+
+ // Settings not correct for this profile
+ strncpy(codecInst.plName, "VP8", 31);
+ codecInst.plType = 126;
+ codecInst.maxBitrate = 0;
+ codecInst.minBitrate = 0;
+ codecInst.width = 176;
+ codecInst.height = 144;
+ codecInst.maxFramerate = 15;
+ codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+ codecInst.startBitrate = 300;
+ //EXPECT_TRUE(enc->InitEncode(&codecInst, 1, 1440) == WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED);
+
+ ASSERT_EQ(enc->InitEncode(&_inst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
+
+
+ //-- ProcessNewBitrate() errors --
+ // Bad bitrate.
+ EXPECT_EQ(enc->SetRates(_inst.maxBitrate + 1, _inst.maxFramerate),
+ WEBRTC_VIDEO_CODEC_OK);
+
+ // Signaling not used.
+
+ // Bad packetloss.
+// EXPECT_TRUE(enc->SetPacketLoss(300) < 0);
+
+ //----- Decoder parameter tests -----
+ //-- Calls before InitDecode() --
+ EXPECT_TRUE(dec->Release() == 0);
+ ASSERT_TRUE(dec->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+ //-- SetCodecConfigParameters() errors --
+ unsigned char tmpBuf[128];
+ EXPECT_TRUE(dec->SetCodecConfigParameters(NULL, sizeof(tmpBuf)) == -1);
+ EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, 1) == -1);
+ // Garbage data.
+ EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, sizeof(tmpBuf)) == -1);
+
+ //----- Function tests -----
+ outFileName = webrtc::test::OutputPath() + _source->GetName() + "-errResTest.yuv";
+ outFile = fopen(outFileName.c_str(), "wb");
+ ASSERT_TRUE(outFile != NULL);
+
+ UnitTest::Perform();
+ Teardown();
+
+}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h
new file mode 100644
index 0000000..e9287e6
--- /dev/null
+++ b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
+
+#include "../../../test_framework/unit_test.h"
+
+class VP8UnitTest : public UnitTest
+{
+public:
+ VP8UnitTest();
+ VP8UnitTest(std::string name, std::string description);
+ virtual void Perform();
+
+protected:
+ virtual WebRtc_UWord32 CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
+ WebRtc_UWord32 /*frameRate*/);
+ virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+ const void *ptrB, unsigned int bLengthBytes);
+ static int PicIdLength(const unsigned char* ptr);
+};
+
+////////////////////////////////////////////////////////////////
+// RESERVATIONS TO PASSING UNIT TEST ON VP8 CODEC //
+// Test that will not pass: //
+// 1. Check bit exact for decoded images. //
+// 2. Target bitrate - Allow a margin of 10% instead of 5% //
+// 3. Detecting errors in bit stream - NA. //
+////////////////////////////////////////////////////////////////
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/OWNERS b/trunk/src/modules/video_coding/main/OWNERS
new file mode 100644
index 0000000..7183cf2
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/OWNERS
@@ -0,0 +1,4 @@
+stefan@webrtc.org
+mikhal@webrtc.org
+marpan@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h b/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
new file mode 100644
index 0000000..c84d5e7
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+
+#include "gmock/gmock.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class MockVCMFrameTypeCallback : public VCMFrameTypeCallback {
+ public:
+ MOCK_METHOD0(RequestKeyFrame, int32_t());
+ MOCK_METHOD1(SliceLossIndicationRequest,
+ WebRtc_Word32(const WebRtc_UWord64 pictureId));
+};
+
+class MockPacketRequestCallback : public VCMPacketRequestCallback {
+ public:
+ MOCK_METHOD2(ResendPackets, int32_t(const uint16_t* sequenceNumbers,
+ uint16_t length));
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
diff --git a/trunk/src/modules/video_coding/main/interface/video_coding.h b/trunk/src/modules/video_coding/main/interface/video_coding.h
new file mode 100644
index 0000000..54e6742
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/interface/video_coding.h
@@ -0,0 +1,556 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
+#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
+
+#include "modules/interface/module.h"
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+
+namespace webrtc
+{
+
+class TickTimeBase;
+class VideoEncoder;
+class VideoDecoder;
+struct CodecSpecificInfo;
+
+class VideoCodingModule : public Module
+{
+public:
+ enum SenderNackMode {
+ kNackNone,
+ kNackAll,
+ kNackSelective
+ };
+
+ enum ReceiverRobustness {
+ kNone,
+ kHardNack,
+ kSoftNack,
+ kDualDecoder,
+ kReferenceSelection
+ };
+
+ enum DecodeErrors {
+ kNoDecodeErrors,
+ kAllowDecodeErrors
+ };
+
+ static VideoCodingModule* Create(const WebRtc_Word32 id);
+
+ static VideoCodingModule* Create(const WebRtc_Word32 id,
+ TickTimeBase* clock);
+
+ static void Destroy(VideoCodingModule* module);
+
+ // Get number of supported codecs
+ //
+ // Return value : Number of supported codecs
+ static WebRtc_UWord8 NumberOfCodecs();
+
+ // Get supported codec settings with using id
+ //
+ // Input:
+ // - listId : Id or index of the codec to look up
+ // - codec : Memory where the codec settings will be stored
+ //
+ // Return value : VCM_OK, on success
+ // VCM_PARAMETER_ERROR if codec not supported or id too high
+ static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, VideoCodec* codec);
+
+ // Get supported codec settings using codec type
+ //
+ // Input:
+ // - codecType : The codec type to get settings for
+ // - codec : Memory where the codec settings will be stored
+ //
+ // Return value : VCM_OK, on success
+ // VCM_PARAMETER_ERROR if codec not supported
+ static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* codec);
+
+ /*
+ * Sender
+ */
+
+ // Any encoder-related state of VCM will be initialized to the
+ // same state as when the VCM was created. This will not interrupt
+ // or effect decoding functionality of VCM. VCM will lose all the
+ // encoding-related settings by calling this function.
+ // For instance, a send codec has to be registered again.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 InitializeSender() = 0;
+
+ // Registers a codec to be used for encoding. Calling this
+ // API multiple times overwrites any previously registered codecs.
+ //
+ // Input:
+ // - sendCodec : Settings for the codec to be registered.
+ // - numberOfCores : The number of cores the codec is allowed
+ // to use.
+ // - maxPayloadSize : The maximum size each payload is allowed
+ // to have. Usually MTU - overhead.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize) = 0;
+
+ // API to get the current send codec in use.
+ //
+ // Input:
+ // - currentSendCodec : Address where the sendCodec will be written.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const = 0;
+
+ // API to get the current send codec type
+ //
+ // Return value : Codec type, on success.
+ // kVideoCodecUnknown, on error or if no send codec is set
+ virtual VideoCodecType SendCodec() const = 0;
+
+ // Register an external encoder object. This can not be used together with
+ // external decoder callbacks.
+ //
+ // Input:
+ // - externalEncoder : Encoder object to be used for encoding frames inserted
+ // with the AddVideoFrame API.
+ // - payloadType : The payload type bound which this encoder is bound to.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
+ WebRtc_UWord8 payloadType,
+ bool internalSource = false) = 0;
+
+ // API to get codec config parameters to be sent out-of-band to a receiver.
+ //
+ // Input:
+ // - buffer : Memory where the codec config parameters should be written.
+ // - size : Size of the memory available.
+ //
+ // Return value : Number of bytes written, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size) = 0;
+
+ // API to get currently configured encoder target bitrate in kbit/s.
+ //
+ // Return value : 0, on success.
+ // < 0, on error.
+ virtual int Bitrate(unsigned int* bitrate) const = 0;
+
+ // API to get currently configured encoder target frame rate.
+ //
+ // Return value : 0, on success.
+ // < 0, on error.
+ virtual int FrameRate(unsigned int* framerate) const = 0;
+
+ // Sets the parameters describing the send channel. These parameters are inputs to the
+ // Media Optimization inside the VCM and also specifies the target bit rate for the
+ // encoder. Bit rate used by NACK should already be compensated for by the user.
+ //
+ // Input:
+ // - availableBandWidth : Band width available for the VCM in kbit/s.
+ // - lossRate : Fractions of lost packets the past second.
+ // (loss rate in percent = 100 * packetLoss / 255)
+ // - rtt : Current round-trip time in ms.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 availableBandWidth,
+ WebRtc_UWord8 lossRate,
+ WebRtc_UWord32 rtt) = 0;
+
+ // Sets the parameters describing the receive channel. These parameters are inputs to the
+ // Media Optimization inside the VCM.
+ //
+ // Input:
+ // - rtt : Current round-trip time in ms.
+ // with the most amount available bandwidth in a conference
+ // scenario
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 rtt) = 0;
+
+ // Register a transport callback which will be called to deliver the encoded data and
+ // side information.
+ //
+ // Input:
+ // - transport : The callback object to register.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
+
+ // Register video output information callback which will be called to deliver information
+ // about the video stream produced by the encoder, for instance the average frame rate and
+ // bit rate.
+ //
+ // Input:
+ // - outputInformation : The callback object to register.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterSendStatisticsCallback(
+ VCMSendStatisticsCallback* sendStats) = 0;
+
+ // Register a video quality settings callback which will be called when
+ // frame rate/dimensions need to be updated for video quality optimization
+ //
+ // Input:
+ // - videoQMSettings : The callback object to register.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error
+ virtual WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings) = 0;
+
+ // Register a video protection callback which will be called to deliver
+ // the requested FEC rate and NACK status (on/off).
+ //
+ // Input:
+ // - protection : The callback object to register.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
+
+ // Enable or disable a video protection method.
+ //
+ // Input:
+ // - videoProtection : The method to enable or disable.
+ // - enable : True if the method should be enabled, false if
+ // it should be disabled.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
+ bool enable) = 0;
+
+ // Add one raw video frame to the encoder. This function does all the necessary
+ // processing, then decides what frame type to encode, or if the frame should be
+ // dropped. If the frame should be encoded it passes the frame to the encoder
+ // before it returns.
+ //
+ // Input:
+ // - videoFrame : Video frame to encode.
+ // - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 AddVideoFrame(
+ const VideoFrame& videoFrame,
+ const VideoContentMetrics* contentMetrics = NULL,
+ const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
+
+ // Next frame encoded should be of the type frameType.
+ //
+ // Input:
+ // - frameType : The frame type to encode next time a VideoFrame
+ // is added to the module.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 FrameTypeRequest(FrameType frameType,
+ WebRtc_UWord8 simulcastIdx) = 0;
+
+ // Frame Dropper enable. Can be used to disable the frame dropping when the encoder
+ // over-uses its bit rate. This API is designed to be used when the encoded frames
+ // are supposed to be stored to an AVI file, or when the I420 codec is used and the
+ // target bit rate shouldn't affect the frame rate.
+ //
+ // Input:
+ // - enable : True to enable the setting, false to disable it.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 EnableFrameDropper(bool enable) = 0;
+
+ // Sent frame counters
+ virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const = 0;
+
+ /*
+ * Receiver
+ */
+
+ // The receiver state of the VCM will be initialized to the
+ // same state as when the VCM was created. This will not interrupt
+ // or effect the send side functionality of VCM. VCM will lose all the
+ // decoding-related settings by calling this function. All frames
+ // inside the jitter buffer are flushed and the delay is reset.
+ // For instance, a receive codec has to be registered again.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 InitializeReceiver() = 0;
+
+ // Register possible receive codecs, can be called multiple times for different codecs.
+ // The module will automatically switch between registered codecs depending on the
+ // payload type of incoming frames. The actual decoder will be created when needed.
+ //
+ // Input:
+ // - receiveCodec : Settings for the codec to be registered.
+ // - numberOfCores : Number of CPU cores that the decoder is allowed to use.
+ // - requireKeyFrame : Set this to true if you don't want any delta frames
+ // to be decoded until the first key frame has been decoded.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
+ WebRtc_Word32 numberOfCores,
+ bool requireKeyFrame = false) = 0;
+
+ // Register an externally defined decoder/renderer object. Can be a decoder only or a
+ // decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
+ // be used for decoding incoming streams.
+ //
+ // Input:
+ // - externalDecoder : The external decoder/renderer object.
+ // - payloadType : The payload type which this decoder should be
+ // registered to.
+ // - internalRenderTiming : True if the internal renderer (if any) of the decoder
+ // object can make sure to render at a given time in ms.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming) = 0;
+
+ // Register a receive callback. Will be called whenever there is a new frame ready
+ // for rendering.
+ //
+ // Input:
+ // - receiveCallback : The callback object to be used by the module when a
+ // frame is ready for rendering.
+ // De-register with a NULL pointer.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
+
+ // Register a receive statistics callback which will be called to deliver information
+ // about the video stream received by the receiving side of the VCM, for instance the
+ // average frame rate and bit rate.
+ //
+ // Input:
+ // - receiveStats : The callback object to register.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
+ VCMReceiveStatisticsCallback* receiveStats) = 0;
+
+ // Register a frame type request callback. This callback will be called when the
+ // module needs to request specific frame types from the send side.
+ //
+ // Input:
+ // - frameTypeCallback : The callback object to be used by the module when
+ // requesting a specific type of frame from the send side.
+ // De-register with a NULL pointer.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterFrameTypeCallback(
+ VCMFrameTypeCallback* frameTypeCallback) = 0;
+
+ // Register a frame storage callback. This callback will be called right before an
+ // encoded frame is given to the decoder. Useful for recording the incoming video sequence.
+ //
+ // Input:
+ // - frameStorageCallback : The callback object used by the module
+ // to store a received encoded frame.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 RegisterFrameStorageCallback(
+ VCMFrameStorageCallback* frameStorageCallback) = 0;
+
+ // Registers a callback which is called whenever the receive side of the VCM
+ // encounters holes in the packet sequence and needs packets to be retransmitted.
+ //
+ // Input:
+ // - callback : The callback to be registered in the VCM.
+ //
+ // Return value : VCM_OK, on success.
+ // <0, on error.
+ virtual WebRtc_Word32 RegisterPacketRequestCallback(
+ VCMPacketRequestCallback* callback) = 0;
+
+ // Waits for the next frame in the jitter buffer to become complete
+ // (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
+ // Should be called as often as possible to get the most out of the decoder.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
+
+ // Waits for the next frame in the dual jitter buffer to become complete
+ // (waits no longer than maxWaitTimeMs), then passes it to the dual decoder
+ // for decoding. This will never trigger a render callback. Should be
+ // called frequently, and as long as it returns 1 it should be called again
+ // as soon as possible.
+ //
+ // Return value : 1, if a frame was decoded
+ // 0, if no frame was decoded
+ // < 0, on error.
+ virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
+
+ // Decodes a frame and sets an appropriate render time in ms relative to the system time.
+ // Should be used in conjunction with VCMFrameStorageCallback.
+ //
+ // Input:
+ // - frameFromStorage : Encoded frame read from file or received through
+ // the VCMFrameStorageCallback callback.
+ //
+ // Return value: : VCM_OK, on success
+ // < 0, on error
+ virtual WebRtc_Word32 DecodeFromStorage(const EncodedVideoData& frameFromStorage) = 0;
+
+ // Reset the decoder state to the initial state.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 ResetDecoder() = 0;
+
+ // API to get the codec which is currently used for decoding by the module.
+ //
+ // Input:
+ // - currentReceiveCodec : Settings for the codec to be registered.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
+
+ // API to get the codec type currently used for decoding by the module.
+ //
+ // Return value : codecy type, on success.
+ // kVideoCodecUnknown, on error or if no receive codec is registered
+ virtual VideoCodecType ReceiveCodec() const = 0;
+
+ // Insert a parsed packet into the receiver side of the module. Will be placed in the
+ // jitter buffer waiting for the frame to become complete. Returns as soon as the packet
+ // has been placed in the jitter buffer.
+ //
+ // Input:
+ // - incomingPayload : Payload of the packet.
+ // - payloadLength : Length of the payload.
+ // - rtpInfo : The parsed header.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
+ WebRtc_UWord32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo) = 0;
+
+ // Minimum playout delay (Used for lip-sync). This is the minimum delay required
+ // to sync with audio. Not included in VideoCodingModule::Delay()
+ // Defaults to 0 ms.
+ //
+ // Input:
+ // - minPlayoutDelayMs : Additional delay in ms.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs) = 0;
+
+ // Set the time required by the renderer to render a frame.
+ //
+ // Input:
+ // - timeMS : The time in ms required by the renderer to render a frame.
+ //
+ // Return value : VCM_OK, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS) = 0;
+
+ // The total delay desired by the VCM. Can be less than the minimum
+ // delay set with SetMinimumPlayoutDelay.
+ //
+ // Return value : Total delay in ms, on success.
+ // < 0, on error.
+ virtual WebRtc_Word32 Delay() const = 0;
+
+ // Get the received frame counters. Keeps track of the number of each frame type
+ // received since the start of the call.
+ //
+ // Output:
+ // - frameCount : Struct to be filled with the number of frames received.
+ //
+ // Return value : VCM_OK, on success.
+ // <0, on error.
+ virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const = 0;
+
+ // Returns the number of packets discarded by the jitter buffer due to being
+ // too late. This can include duplicated packets which arrived after the
+ // frame was sent to the decoder. Therefore packets which were prematurely
+ // NACKed will be counted.
+ virtual WebRtc_UWord32 DiscardedPackets() const = 0;
+
+
+ // Robustness APIs
+
+ // Set the sender RTX/NACK mode.
+ // Input:
+ // - mode : the selected NACK mode.
+ //
+ // Return value : VCM_OK, on success;
+ // < 0, on error.
+ virtual int SetSenderNackMode(SenderNackMode mode) = 0;
+
+ // Set the sender reference picture selection (RPS) mode.
+ // Input:
+ // - enable : true or false, for enable and disable, respectively.
+ //
+ // Return value : VCM_OK, on success;
+ // < 0, on error.
+ virtual int SetSenderReferenceSelection(bool enable) = 0;
+
+ // Set the sender forward error correction (FEC) mode.
+ // Input:
+ // - enable : true or false, for enable and disable, respectively.
+ //
+ // Return value : VCM_OK, on success;
+ // < 0, on error.
+ virtual int SetSenderFEC(bool enable) = 0;
+
+ // Set the key frame period, or disable periodic key frames (I-frames).
+ // Input:
+ // - periodMs : period in ms; <= 0 to disable periodic key frames.
+ //
+ // Return value : VCM_OK, on success;
+ // < 0, on error.
+ virtual int SetSenderKeyFramePeriod(int periodMs) = 0;
+
+ // Set the receiver robustness mode. The mode decides how the receiver
+ // responds to losses in the stream. The type of counter-measure (soft or
+ // hard NACK, dual decoder, RPS, etc.) is selected through the
+ // robustnessMode parameter. The errorMode parameter decides if it is
+ // allowed to display frames corrupted by losses. Note that not all
+ // combinations of the two parameters are feasible. An error will be
+ // returned for invalid combinations.
+ // Input:
+ // - robustnessMode : selected robustness mode.
+ // - errorMode : selected error mode.
+ //
+ // Return value : VCM_OK, on success;
+ // < 0, on error.
+ virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
+ DecodeErrors errorMode) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
diff --git a/trunk/src/modules/video_coding/main/interface/video_coding_defines.h b/trunk/src/modules/video_coding/main/interface/video_coding_defines.h
new file mode 100644
index 0000000..3755f6d
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/interface/video_coding_defines.h
@@ -0,0 +1,195 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+
+#include "typedefs.h"
+#include "modules/interface/module_common_types.h"
+
+namespace webrtc
+{
+
+// Error codes
+#define VCM_FRAME_NOT_READY 3
+#define VCM_REQUEST_SLI 2
+#define VCM_MISSING_CALLBACK 1
+#define VCM_OK 0
+#define VCM_GENERAL_ERROR -1
+#define VCM_LEVEL_EXCEEDED -2
+#define VCM_MEMORY -3
+#define VCM_PARAMETER_ERROR -4
+#define VCM_UNKNOWN_PAYLOAD -5
+#define VCM_CODEC_ERROR -6
+#define VCM_UNINITIALIZED -7
+#define VCM_NO_CODEC_REGISTERED -8
+#define VCM_JITTER_BUFFER_ERROR -9
+#define VCM_OLD_PACKET_ERROR -10
+#define VCM_NO_FRAME_DECODED -11
+#define VCM_ERROR_REQUEST_SLI -12
+#define VCM_NOT_IMPLEMENTED -20
+
+#define VCM_RED_PAYLOAD_TYPE 96
+#define VCM_ULPFEC_PAYLOAD_TYPE 97
+#define VCM_VP8_PAYLOAD_TYPE 120
+#define VCM_I420_PAYLOAD_TYPE 124
+
+enum VCMNackProperties
+{
+ kNackHistoryLength = 450
+};
+
+enum VCMVideoProtection
+{
+ kProtectionNack, // Both send-side and receive-side
+ kProtectionNackSender, // Send-side only
+ kProtectionNackReceiver, // Receive-side only
+ kProtectionDualDecoder,
+ kProtectionFEC,
+ kProtectionNackFEC,
+ kProtectionKeyOnLoss,
+ kProtectionKeyOnKeyLoss,
+ kProtectionPeriodicKeyFrames
+};
+
+enum VCMTemporalDecimation
+{
+ kBitrateOverUseDecimation,
+};
+
+struct VCMFrameCount
+{
+ WebRtc_UWord32 numKeyFrames;
+ WebRtc_UWord32 numDeltaFrames;
+};
+
+
+// Callback class used for sending data ready to be packetized
+class VCMPacketizationCallback
+{
+public:
+ virtual WebRtc_Word32 SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* rtpVideoHdr) = 0;
+protected:
+ virtual ~VCMPacketizationCallback() {}
+};
+
+// Callback class used for passing decoded frames which are ready to be rendered.
+class VCMFrameStorageCallback
+{
+public:
+ virtual WebRtc_Word32 StoreReceivedFrame(const EncodedVideoData& frameToStore) = 0;
+
+protected:
+ virtual ~VCMFrameStorageCallback() {}
+};
+
+// Callback class used for passing decoded frames which are ready to be rendered.
+class VCMReceiveCallback
+{
+public:
+ virtual WebRtc_Word32 FrameToRender(VideoFrame& videoFrame) = 0;
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
+
+protected:
+ virtual ~VCMReceiveCallback() {}
+};
+
+// Callback class used for informing the user of the bit rate and frame rate produced by the
+// encoder.
+class VCMSendStatisticsCallback
+{
+public:
+ virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
+ const WebRtc_UWord32 frameRate) = 0;
+
+protected:
+ virtual ~VCMSendStatisticsCallback() {}
+};
+
+// Callback class used for informing the user of the incoming bit rate and frame rate.
+class VCMReceiveStatisticsCallback
+{
+public:
+ virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bitRate,
+ const WebRtc_UWord32 frameRate) = 0;
+
+protected:
+ virtual ~VCMReceiveStatisticsCallback() {}
+};
+
+// Callback class used for telling the user about the requested amount of
+// bit stream protection: FEC rate for key and delta frame;
+// whether the FEC uses unequal protection (UEP) across packets,
+// for key and delta frame;
+// and whether NACK should be on or off.
+class VCMProtectionCallback
+{
+public:
+ virtual int ProtectionRequest(
+ uint8_t delta_fec_rate,
+ uint8_t key_fec_rate,
+ bool delta_use_uep_protection,
+ bool key_use_uep_protection,
+ bool nack_enabled,
+ uint32_t* sent_video_rate_bps,
+ uint32_t* sent_nack_rate_bps,
+ uint32_t* sent_fec_rate_bps) = 0;
+
+protected:
+ virtual ~VCMProtectionCallback() {}
+};
+
+// Callback class used for telling the user about what frame type needed to continue decoding.
+// Typically a key frame when the stream has been corrupted in some way.
+class VCMFrameTypeCallback
+{
+public:
+ virtual WebRtc_Word32 RequestKeyFrame() = 0;
+ virtual WebRtc_Word32 SliceLossIndicationRequest(const WebRtc_UWord64 pictureId) {return -1;}
+
+protected:
+ virtual ~VCMFrameTypeCallback() {}
+};
+
+// Callback class used for telling the user about which packet sequence numbers are currently
+// missing and need to be resent.
+class VCMPacketRequestCallback
+{
+public:
+ virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
+ WebRtc_UWord16 length) = 0;
+
+protected:
+ virtual ~VCMPacketRequestCallback() {}
+};
+
+// Callback used to inform the user of the the desired resolution
+// as subscribed by Media Optimization (Quality Modes)
+class VCMQMSettingsCallback
+{
+public:
+ virtual WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height) = 0;
+
+protected:
+ virtual ~VCMQMSettingsCallback() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
diff --git a/trunk/src/modules/video_coding/main/source/Android.mk b/trunk/src/modules/video_coding/main/source/Android.mk
new file mode 100644
index 0000000..2ecf45d
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/Android.mk
@@ -0,0 +1,70 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_video_coding
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ codec_database.cc \
+ codec_timer.cc \
+ content_metrics_processing.cc \
+ decoding_state.cc \
+ encoded_frame.cc \
+ exp_filter.cc \
+ frame_buffer.cc \
+ frame_dropper.cc \
+ generic_decoder.cc \
+ generic_encoder.cc \
+ inter_frame_delay.cc \
+ jitter_buffer.cc \
+ jitter_buffer_common.cc \
+ jitter_estimator.cc \
+ media_opt_util.cc \
+ media_optimization.cc \
+ packet.cc \
+ qm_select.cc \
+ receiver.cc \
+ rtt_filter.cc \
+ session_info.cc \
+ timestamp_extrapolator.cc \
+ timestamp_map.cc \
+ timing.cc \
+ video_coding_impl.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../codecs/interface \
+ $(LOCAL_PATH)/../../codecs/i420/main/interface \
+ $(LOCAL_PATH)/../../codecs/vp8/main/interface \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../common_video/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/main/source/codec_database.cc b/trunk/src/modules/video_coding/main/source/codec_database.cc
new file mode 100644
index 0000000..6f81895
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/codec_database.cc
@@ -0,0 +1,719 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "codec_database.h"
+
+#include <assert.h>
+
+#include "../../../../engine_configurations.h"
+#include "internal_defines.h"
+#include "trace.h"
+
+#if defined(_WIN32)
+ // VS 2005: Don't warn for default initialized arrays. See help for more info.
+ // Don't warn for strncpy being unsecure.
+ // switch statement contains 'default' but no 'case' labels
+#pragma warning(disable:4351; disable:4996; disable:4065)
+#endif
+
+// Supported codecs
+#ifdef VIDEOCODEC_VP8
+ #include "vp8.h"
+#endif
+#ifdef VIDEOCODEC_I420
+ #include "i420.h"
+#endif
+
+namespace webrtc
+{
+
+VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
+ WebRtc_UWord32 numberOfCores,
+ bool requireKeyFrame)
+:
+_settings(settings),
+_numberOfCores(numberOfCores),
+_requireKeyFrame(requireKeyFrame)
+{
+}
+
+VCMExtDecoderMapItem::VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming)
+:
+_payloadType(payloadType),
+_externalDecoderInstance(externalDecoderInstance),
+_internalRenderTiming(internalRenderTiming)
+{
+}
+
+VCMCodecDataBase::VCMCodecDataBase(WebRtc_Word32 id):
+_id(id),
+_numberOfCores(0),
+_maxPayloadSize(kDefaultPayloadSize),
+_periodicKeyFrames(false),
+_currentEncIsExternal(false),
+_sendCodec(),
+_receiveCodec(),
+_externalPayloadType(0),
+_externalEncoder(NULL),
+_internalSource(false),
+_ptrEncoder(NULL),
+_ptrDecoder(NULL),
+_currentDecIsExternal(false),
+_decMap(),
+_decExternalMap()
+{
+ //
+}
+
+VCMCodecDataBase::~VCMCodecDataBase()
+{
+ Reset();
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Reset()
+{
+ WebRtc_Word32 ret = ResetReceiver();
+ if (ret < 0)
+ {
+ return ret;
+ }
+ ret = ResetSender();
+ if (ret < 0)
+ {
+ return ret;
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ResetSender()
+{
+ DeleteEncoder();
+ _periodicKeyFrames = false;
+ return VCM_OK;
+}
+
+VCMGenericEncoder* VCMCodecDataBase::CreateEncoder(
+ const VideoCodecType type) const {
+
+ switch(type)
+ {
+#ifdef VIDEOCODEC_VP8
+ case kVideoCodecVP8:
+ return new VCMGenericEncoder(*(VP8Encoder::Create()));
+#endif
+#ifdef VIDEOCODEC_I420
+ case kVideoCodecI420:
+ return new VCMGenericEncoder(*(new I420Encoder));
+#endif
+ default:
+ return NULL;
+ }
+}
+
+void
+VCMCodecDataBase::DeleteEncoder()
+{
+ if (_ptrEncoder)
+ {
+ _ptrEncoder->Release();
+ if (!_currentEncIsExternal)
+ {
+ delete &_ptrEncoder->_encoder;
+ }
+ delete _ptrEncoder;
+ _ptrEncoder = NULL;
+ }
+}
+
+WebRtc_UWord8
+VCMCodecDataBase::NumberOfCodecs()
+{
+ return VCM_NUM_VIDEO_CODECS_AVAILABLE;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Codec(WebRtc_UWord8 listId, VideoCodec *settings)
+{
+ if (settings == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+
+ if (listId >= VCM_NUM_VIDEO_CODECS_AVAILABLE)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ memset(settings, 0, sizeof(VideoCodec));
+ switch (listId)
+ {
+#ifdef VIDEOCODEC_VP8
+ case VCM_VP8_IDX:
+ {
+ strncpy(settings->plName, "VP8", 4);
+ settings->codecType = kVideoCodecVP8;
+ // 96 to 127 dynamic payload types for video codecs
+ settings->plType = VCM_VP8_PAYLOAD_TYPE;
+ settings->startBitrate = 100;
+ settings->minBitrate = VCM_MIN_BITRATE;
+ settings->maxBitrate = 0;
+ settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
+ settings->width = VCM_DEFAULT_CODEC_WIDTH;
+ settings->height = VCM_DEFAULT_CODEC_HEIGHT;
+ settings->numberOfSimulcastStreams = 0;
+ settings->codecSpecific.VP8.resilience = kResilientStream;
+ settings->codecSpecific.VP8.numberOfTemporalLayers = 1;
+ break;
+ }
+#endif
+#ifdef VIDEOCODEC_I420
+ case VCM_I420_IDX:
+ {
+ strncpy(settings->plName, "I420", 5);
+ settings->codecType = kVideoCodecI420;
+ // 96 to 127 dynamic payload types for video codecs
+ settings->plType = VCM_I420_PAYLOAD_TYPE;
+ // Bitrate needed for this size and framerate
+ settings->startBitrate = 3*VCM_DEFAULT_CODEC_WIDTH*
+ VCM_DEFAULT_CODEC_HEIGHT*8*
+ VCM_DEFAULT_FRAME_RATE/1000/2;
+ settings->maxBitrate = settings->startBitrate;
+ settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
+ settings->width = VCM_DEFAULT_CODEC_WIDTH;
+ settings->height = VCM_DEFAULT_CODEC_HEIGHT;
+ settings->minBitrate = VCM_MIN_BITRATE;
+ settings->numberOfSimulcastStreams = 0;
+ break;
+ }
+#endif
+ default:
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ }
+
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Codec(VideoCodecType codecType, VideoCodec* settings)
+{
+ for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++)
+ {
+ const WebRtc_Word32 ret = VCMCodecDataBase::Codec(i, settings);
+ if (ret != VCM_OK)
+ {
+ return ret;
+ }
+ if (codecType == settings->codecType)
+ {
+ return VCM_OK;
+ }
+ }
+ return VCM_PARAMETER_ERROR;
+}
+
+// assuming only one registered encoder - since only one used, no need for more
+WebRtc_Word32
+VCMCodecDataBase::RegisterSendCodec(const VideoCodec* sendCodec,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize)
+ {
+ if (sendCodec == NULL)
+ {
+ return VCM_UNINITIALIZED;
+ }
+ if (maxPayloadSize == 0)
+ {
+ maxPayloadSize = kDefaultPayloadSize;
+ }
+ if (numberOfCores > 32)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ if (sendCodec->plType <= 0)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ // Make sure the start bit rate is sane...
+ if (sendCodec->startBitrate > 1000000)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ if (sendCodec->codecType == kVideoCodecUnknown)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ _numberOfCores = numberOfCores;
+ _maxPayloadSize = maxPayloadSize;
+
+ memcpy(&_sendCodec, sendCodec, sizeof(VideoCodec));
+
+ if (_sendCodec.maxBitrate == 0)
+ {
+ // max is one bit per pixel
+ _sendCodec.maxBitrate = ((WebRtc_Word32)_sendCodec.height *
+ (WebRtc_Word32)_sendCodec.width *
+ (WebRtc_Word32)_sendCodec.maxFramerate) / 1000;
+ if (_sendCodec.startBitrate > _sendCodec.maxBitrate)
+ {
+ // but if the customer tries to set a higher start bit rate we will increase
+ // the max accordingly
+ _sendCodec.maxBitrate = _sendCodec.startBitrate;
+ }
+ }
+
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::SendCodec(VideoCodec* currentSendCodec) const
+{
+ WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id), "SendCodec");
+
+ if(_ptrEncoder == NULL)
+ {
+ return VCM_UNINITIALIZED;
+ }
+ memcpy(currentSendCodec, &_sendCodec, sizeof(VideoCodec));
+ return VCM_OK;
+}
+
+VideoCodecType
+VCMCodecDataBase::SendCodec() const
+{
+ WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id),
+ "SendCodec type");
+ if (_ptrEncoder == NULL)
+ {
+ return kVideoCodecUnknown;
+ }
+ return _sendCodec.codecType;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec)
+{
+ wasSendCodec = false;
+ if (_externalPayloadType != payloadType)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ if (_sendCodec.plType == payloadType)
+ {
+ //De-register as send codec if needed
+ DeleteEncoder();
+ memset(&_sendCodec, 0, sizeof(VideoCodec));
+ _currentEncIsExternal = false;
+ wasSendCodec = true;
+ }
+ _externalPayloadType = 0;
+ _externalEncoder = NULL;
+ _internalSource = false;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* externalEncoder,
+ WebRtc_UWord8 payloadType,
+ bool internalSource)
+{
+ // since only one encoder can be used at a given time,
+ // only one external encoder can be registered/used
+ _externalEncoder = externalEncoder;
+ _externalPayloadType = payloadType;
+ _internalSource = internalSource;
+
+ return VCM_OK;
+}
+
+VCMGenericEncoder*
+VCMCodecDataBase::SetEncoder(const VideoCodec* settings,
+ VCMEncodedFrameCallback* VCMencodedFrameCallback)
+
+{
+ // if encoder exists, will destroy it and create new one
+ DeleteEncoder();
+
+ if (settings->plType == _externalPayloadType)
+ {
+ // External encoder
+ _ptrEncoder = new VCMGenericEncoder(*_externalEncoder, _internalSource);
+ _currentEncIsExternal = true;
+ }
+ else
+ {
+ _ptrEncoder = CreateEncoder(settings->codecType);
+ _currentEncIsExternal = false;
+ }
+ VCMencodedFrameCallback->SetPayloadType(settings->plType);
+
+ if (_ptrEncoder == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to create encoder: %s.",
+ settings->plName);
+ return NULL;
+ }
+ if (_ptrEncoder->InitEncode(settings, _numberOfCores, _maxPayloadSize) < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to initialize encoder: %s.",
+ settings->plName);
+ DeleteEncoder();
+ return NULL;
+ }
+ else if (_ptrEncoder->RegisterEncodeCallback(VCMencodedFrameCallback) < 0)
+ {
+ DeleteEncoder();
+ return NULL;
+ }
+ // Intentionally don't check return value since the encoder registration
+ // shouldn't fail because the codec doesn't support changing the
+ // periodic key frame setting.
+ _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
+ return _ptrEncoder;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::SetPeriodicKeyFrames(bool enable)
+{
+ _periodicKeyFrames = enable;
+ if (_ptrEncoder != NULL)
+ {
+ return _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receiveCodec,
+ WebRtc_UWord32 numberOfCores,
+ bool requireKeyFrame)
+{
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCoding, VCMId(_id),
+ "Codec: %s, Payload type %d, Height %d, Width %d, Bitrate %d, Framerate %d.",
+ receiveCodec->plName, receiveCodec->plType,
+ receiveCodec->height, receiveCodec->width,
+ receiveCodec->startBitrate, receiveCodec->maxFramerate);
+
+ // check if payload value already exists, if so - erase old and insert new
+ DeRegisterReceiveCodec(receiveCodec->plType);
+ if (receiveCodec->codecType == kVideoCodecUnknown)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ VideoCodec* newReceiveCodec = new VideoCodec(*receiveCodec);
+ _decMap[receiveCodec->plType] =
+ new VCMDecoderMapItem(newReceiveCodec, numberOfCores, requireKeyFrame);
+
+ return VCM_OK;
+}
+
+WebRtc_Word32 VCMCodecDataBase::DeRegisterReceiveCodec(
+ WebRtc_UWord8 payloadType)
+{
+ DecoderMap::iterator it = _decMap.find(payloadType);
+ if (it == _decMap.end())
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ VCMDecoderMapItem* decItem = (*it).second;
+ delete decItem->_settings;
+ delete decItem;
+ _decMap.erase(it);
+ if (_receiveCodec.plType == payloadType)
+ {
+ // This codec is currently in use.
+ memset(&_receiveCodec, 0, sizeof(VideoCodec));
+ _currentDecIsExternal = false;
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ResetReceiver()
+{
+ ReleaseDecoder(_ptrDecoder);
+ _ptrDecoder = NULL;
+ memset(&_receiveCodec, 0, sizeof(VideoCodec));
+ DecoderMap::iterator it = _decMap.begin();
+ while (it != _decMap.end()) {
+ if ((*it).second->_settings != NULL)
+ {
+ delete (*it).second->_settings;
+ }
+ delete (*it).second;
+ _decMap.erase(it);
+ it = _decMap.begin();
+ }
+ ExternalDecoderMap::iterator exterit = _decExternalMap.begin();
+ while (exterit != _decExternalMap.end()) {
+ delete (*exterit).second;
+ _decExternalMap.erase(exterit);
+ exterit = _decExternalMap.begin();
+ }
+
+ _currentDecIsExternal = false;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::DeRegisterExternalDecoder(WebRtc_UWord8 payloadType)
+{
+ ExternalDecoderMap::iterator it = _decExternalMap.find(payloadType);
+ if (it == _decExternalMap.end())
+ {
+ // Not found
+ return VCM_PARAMETER_ERROR;
+ }
+ if (_receiveCodec.plType == payloadType)
+ {
+ // Release it if it was registered and in use
+ ReleaseDecoder(_ptrDecoder);
+ _ptrDecoder = NULL;
+ }
+ DeRegisterReceiveCodec(payloadType);
+ delete (*it).second;
+ _decExternalMap.erase(it);
+ return VCM_OK;
+}
+
+// Add the external encoder object to the list of external decoders.
+// Won't be registered as a receive codec until RegisterReceiveCodec is called.
+WebRtc_Word32
+VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming)
+{
+ // check if payload value already exists, if so - erase old and insert new
+ VCMExtDecoderMapItem* extDecoder = new VCMExtDecoderMapItem(externalDecoder,
+ payloadType,
+ internalRenderTiming);
+ if (extDecoder == NULL)
+ {
+ return VCM_MEMORY;
+ }
+ DeRegisterExternalDecoder(payloadType);
+ _decExternalMap[payloadType] = extDecoder;
+
+ return VCM_OK;
+}
+
+bool
+VCMCodecDataBase::DecoderRegistered() const
+{
+ return !_decMap.empty();
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ReceiveCodec(VideoCodec* currentReceiveCodec) const
+{
+ if (_ptrDecoder == NULL)
+ {
+ return VCM_NO_FRAME_DECODED;
+ }
+ memcpy(currentReceiveCodec, &_receiveCodec, sizeof(VideoCodec));
+ return VCM_OK;
+}
+
+VideoCodecType
+VCMCodecDataBase::ReceiveCodec() const
+{
+ if (_ptrDecoder == NULL)
+ {
+ return kVideoCodecUnknown;
+ }
+ return _receiveCodec.codecType;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::SetDecoder(WebRtc_UWord8 payloadType,
+ VCMDecodedFrameCallback& callback)
+{
+ if (payloadType == _receiveCodec.plType || payloadType == 0)
+ {
+ return _ptrDecoder;
+ }
+ // check for exisitng decoder, if exists - delete
+ if (_ptrDecoder)
+ {
+ ReleaseDecoder(_ptrDecoder);
+ _ptrDecoder = NULL;
+ memset(&_receiveCodec, 0, sizeof(VideoCodec));
+ }
+ _ptrDecoder = CreateAndInitDecoder(payloadType, _receiveCodec,
+ _currentDecIsExternal);
+ if (_ptrDecoder == NULL)
+ {
+ return NULL;
+ }
+ if (_ptrDecoder->RegisterDecodeCompleteCallback(&callback) < 0)
+ {
+ ReleaseDecoder(_ptrDecoder);
+ _ptrDecoder = NULL;
+ memset(&_receiveCodec, 0, sizeof(VideoCodec));
+ return NULL;
+ }
+ return _ptrDecoder;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateAndInitDecoder(WebRtc_UWord8 payloadType,
+ VideoCodec& newCodec,
+ bool &external) const
+{
+ VCMDecoderMapItem* decoderItem = FindDecoderItem(payloadType);
+ if (decoderItem == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id),
+ "Unknown payload type: %u", payloadType);
+ return NULL;
+ }
+ VCMGenericDecoder* ptrDecoder = NULL;
+ VCMExtDecoderMapItem* externalDecItem = FindExternalDecoderItem(
+ payloadType);
+ if (externalDecItem != NULL)
+ {
+ // External codec
+ ptrDecoder = new VCMGenericDecoder(
+ *externalDecItem->_externalDecoderInstance,
+ _id,
+ true);
+ external = true;
+ }
+ else
+ {
+ // create decoder
+ ptrDecoder = CreateDecoder(decoderItem->_settings->codecType);
+ external = false;
+ }
+ if (ptrDecoder == NULL)
+ {
+ return NULL;
+ }
+
+ if (ptrDecoder->InitDecode(decoderItem->_settings,
+ decoderItem->_numberOfCores,
+ decoderItem->_requireKeyFrame) < 0)
+ {
+ ReleaseDecoder(ptrDecoder);
+ return NULL;
+ }
+ memcpy(&newCodec, decoderItem->_settings, sizeof(VideoCodec));
+ return ptrDecoder;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateDecoderCopy() const
+{
+ if (_ptrDecoder == NULL)
+ {
+ return NULL;
+ }
+ VideoDecoder* decoderCopy = _ptrDecoder->_decoder.Copy();
+ if (decoderCopy == NULL)
+ {
+ return NULL;
+ }
+ return new VCMGenericDecoder(*decoderCopy, _id, _ptrDecoder->External());
+}
+
+void
+VCMCodecDataBase::CopyDecoder(const VCMGenericDecoder& decoder)
+{
+ VideoDecoder* decoderCopy = decoder._decoder.Copy();
+ if (decoderCopy != NULL)
+ {
+ VCMDecodedFrameCallback* cb = _ptrDecoder->_callback;
+ ReleaseDecoder(_ptrDecoder);
+ _ptrDecoder = new VCMGenericDecoder(*decoderCopy, _id,
+ decoder.External());
+ if (cb && _ptrDecoder->RegisterDecodeCompleteCallback(cb))
+ {
+ assert(false);
+ }
+ }
+}
+
+bool
+VCMCodecDataBase::RenderTiming() const
+{
+ bool renderTiming = true;
+ if (_currentDecIsExternal)
+ {
+ VCMExtDecoderMapItem* extItem = FindExternalDecoderItem(_receiveCodec.plType);
+ renderTiming = extItem->_internalRenderTiming;
+ }
+ return renderTiming;
+}
+
+void
+VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const
+{
+ if (decoder != NULL)
+ {
+ assert(&decoder->_decoder != NULL);
+ decoder->Release();
+ if (!decoder->External())
+ {
+ delete &decoder->_decoder;
+ }
+ delete decoder;
+ }
+}
+
+VCMDecoderMapItem*
+VCMCodecDataBase::FindDecoderItem(WebRtc_UWord8 payloadType) const
+{
+ DecoderMap::const_iterator it = _decMap.find(payloadType);
+ if (it != _decMap.end())
+ {
+ return (*it).second;
+ }
+ return NULL;
+}
+
+VCMExtDecoderMapItem*
+VCMCodecDataBase::FindExternalDecoderItem(WebRtc_UWord8 payloadType) const
+{
+ ExternalDecoderMap::const_iterator it = _decExternalMap.find(payloadType);
+ if (it != _decExternalMap.end())
+ {
+ return (*it).second;
+ }
+ return NULL;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateDecoder(VideoCodecType type) const
+{
+ switch(type)
+ {
+#ifdef VIDEOCODEC_VP8
+ case kVideoCodecVP8:
+ return new VCMGenericDecoder(*(VP8Decoder::Create()), _id);
+#endif
+#ifdef VIDEOCODEC_I420
+ case kVideoCodecI420:
+ return new VCMGenericDecoder(*(new I420Decoder), _id);
+#endif
+ default:
+ return NULL;
+ }
+}
+}
diff --git a/trunk/src/modules/video_coding/main/source/codec_database.h b/trunk/src/modules/video_coding/main/source/codec_database.h
new file mode 100644
index 0000000..aab8229
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/codec_database.h
@@ -0,0 +1,210 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+
+#include <map>
+
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_coding/main/source/generic_decoder.h"
+#include "modules/video_coding/main/source/generic_encoder.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+enum VCMCodecDBProperties
+{
+ kDefaultPayloadSize = 1440
+};
+
+class VCMDecoderMapItem {
+public:
+ VCMDecoderMapItem(VideoCodec* settings,
+ WebRtc_UWord32 numberOfCores,
+ bool requireKeyFrame);
+
+ VideoCodec* _settings;
+ WebRtc_UWord32 _numberOfCores;
+ bool _requireKeyFrame;
+};
+
+class VCMExtDecoderMapItem {
+public:
+ VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming);
+
+ WebRtc_UWord8 _payloadType;
+ VideoDecoder* _externalDecoderInstance;
+ bool _internalRenderTiming;
+};
+
+/*******************************/
+/* VCMCodecDataBase class */
+/*******************************/
+class VCMCodecDataBase
+{
+public:
+ VCMCodecDataBase(WebRtc_Word32 id);
+ ~VCMCodecDataBase();
+ /**
+ * Release codecdatabase - release all memory for both send and receive side
+ */
+ WebRtc_Word32 Reset();
+ /**
+ * Sender Side
+ */
+ /**
+ * Returns the number of supported codecs (or -1 in case of error).
+ */
+ static WebRtc_UWord8 NumberOfCodecs();
+ /**
+ * Get supported codecs with ID
+ * Input Values:
+ * listnr : Requested codec id number
+ * codec_inst: Pointer to the struct in which the returned codec information is copied
+ * Return Values: 0 if successful, otherwise
+ */
+ static WebRtc_Word32 Codec(WebRtc_UWord8 listId, VideoCodec* settings);
+ static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* settings);
+ /**
+ * Reset Sender side
+ */
+ WebRtc_Word32 ResetSender();
+ /**
+ * Setting the sender side codec and initiaiting the desired codec given the VideoCodec
+ * struct.
+ * Return Value: 0 if the codec and the settings are supported, otherwise
+ */
+ WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize);
+ /**
+ * Get current send side codec. Relevant for internal codecs only.
+ */
+ WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const;
+ /**
+ * Get current send side codec type. Relevant for internal codecs only.
+ */
+ VideoCodecType SendCodec() const;
+ /**
+ * Register external encoder - current assumption - if one is registered then it will also
+ * be used, and therefore it is also initialized
+ * Return value: A pointer to the encoder on success, or null, in case of an error.
+ */
+ WebRtc_Word32 DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec);
+ WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
+ WebRtc_UWord8 payloadType,
+ bool internalSource);
+ /**
+ * Returns a encoder given a payloadname - to be used with internal encoders only.
+ * Special cases:
+ * Encoder exists - If payload matches, returns existing one, otherwise,
+ * deletes existing one and creates new one.
+ * No match found / Error - returns NULL.
+ */
+ VCMGenericEncoder* SetEncoder(const VideoCodec* settings,
+ VCMEncodedFrameCallback* VCMencodedFrameCallback);
+
+ WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
+
+ bool InternalSource() const;
+
+ /*
+ * Receiver Side
+ */
+ WebRtc_Word32 ResetReceiver();
+ /**
+ * Register external decoder/render object
+ */
+ WebRtc_Word32 DeRegisterExternalDecoder(WebRtc_UWord8 payloadType);
+ WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming);
+
+ bool DecoderRegistered() const;
+ /**
+ * Register recieve codec
+ */
+ WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
+ WebRtc_UWord32 numberOfCores,
+ bool requireKeyFrame);
+ WebRtc_Word32 DeRegisterReceiveCodec(WebRtc_UWord8 payloadType);
+ /**
+ * Get current receive side codec. Relevant for internal codecs only.
+ */
+ WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const;
+ /**
+ * Get current receive side codec type. Relevant for internal codecs only.
+ */
+ VideoCodecType ReceiveCodec() const;
+ /**
+ * Returns a decoder given which matches a payload type.
+ * Special cases:
+ * Decoder exists - If payload matches, returns existing one, otherwise, deletes
+ * existing one, and creates new one.
+ * No match found / Error - returns NULL.
+ */
+ VCMGenericDecoder* SetDecoder(WebRtc_UWord8 payloadType, VCMDecodedFrameCallback& callback);
+
+ VCMGenericDecoder* CreateAndInitDecoder(WebRtc_UWord8 payloadType,
+ VideoCodec& newCodec,
+ bool &external) const;
+
+ VCMGenericDecoder* CreateDecoderCopy() const;
+
+ void ReleaseDecoder(VCMGenericDecoder* decoder) const;
+
+ void CopyDecoder(const VCMGenericDecoder& decoder);
+
+ bool RenderTiming() const;
+
+protected:
+ /**
+ * Create an internal Encoder given a codec type
+ */
+ VCMGenericEncoder* CreateEncoder(const VideoCodecType type) const;
+
+ void DeleteEncoder();
+ /*
+ * Create an internal Decoder given a codec type
+ */
+ VCMGenericDecoder* CreateDecoder(VideoCodecType type) const;
+
+ VCMDecoderMapItem* FindDecoderItem(WebRtc_UWord8 payloadType) const;
+
+ VCMExtDecoderMapItem* FindExternalDecoderItem(WebRtc_UWord8 payloadType) const;
+
+private:
+ typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
+ typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
+ WebRtc_Word32 _id;
+ WebRtc_UWord32 _numberOfCores;
+ WebRtc_UWord32 _maxPayloadSize;
+ bool _periodicKeyFrames;
+ bool _currentEncIsExternal;
+ VideoCodec _sendCodec;
+ VideoCodec _receiveCodec;
+ WebRtc_UWord8 _externalPayloadType;
+ VideoEncoder* _externalEncoder;
+ bool _internalSource;
+ VCMGenericEncoder* _ptrEncoder;
+ VCMGenericDecoder* _ptrDecoder;
+ bool _currentDecIsExternal;
+ DecoderMap _decMap;
+ ExternalDecoderMap _decExternalMap;
+}; // end of VCMCodecDataBase class definition
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
diff --git a/trunk/src/modules/video_coding/main/source/codec_timer.cc b/trunk/src/modules/video_coding/main/source/codec_timer.cc
new file mode 100644
index 0000000..1d112fa
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/codec_timer.cc
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "codec_timer.h"
+
+#include <assert.h>
+
+namespace webrtc
+{
+
+VCMCodecTimer::VCMCodecTimer()
+:
+_filteredMax(0),
+_firstDecodeTime(true),
+_shortMax(0),
+_history()
+{
+ Reset();
+}
+
+WebRtc_Word32 VCMCodecTimer::StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs)
+{
+ const WebRtc_Word32 timeDiff = static_cast<WebRtc_Word32>(nowMs - startTimeMs);
+ MaxFilter(timeDiff, nowMs);
+ return timeDiff;
+}
+
+void VCMCodecTimer::Reset()
+{
+ _filteredMax = 0;
+ _firstDecodeTime = true;
+ _shortMax = 0;
+ for (int i=0; i < MAX_HISTORY_SIZE; i++)
+ {
+ _history[i].shortMax = 0;
+ _history[i].timeMs = -1;
+ }
+}
+
+// Update the max-value filter
+void VCMCodecTimer::MaxFilter(WebRtc_Word32 decodeTime, WebRtc_Word64 nowMs)
+{
+ if (!_firstDecodeTime)
+ {
+ UpdateMaxHistory(decodeTime, nowMs);
+ ProcessHistory(nowMs);
+ }
+ else
+ {
+ _firstDecodeTime = false;
+ }
+}
+
+void
+VCMCodecTimer::UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now)
+{
+ if (_history[0].timeMs >= 0 &&
+ now - _history[0].timeMs < SHORT_FILTER_MS)
+ {
+ if (decodeTime > _shortMax)
+ {
+ _shortMax = decodeTime;
+ }
+ }
+ else
+ {
+ // Only add a new value to the history once a second
+ if(_history[0].timeMs == -1)
+ {
+ // First, no shift
+ _shortMax = decodeTime;
+ }
+ else
+ {
+ // Shift
+ for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
+ {
+ _history[i+1].shortMax = _history[i].shortMax;
+ _history[i+1].timeMs = _history[i].timeMs;
+ }
+ }
+ if (_shortMax == 0)
+ {
+ _shortMax = decodeTime;
+ }
+
+ _history[0].shortMax = _shortMax;
+ _history[0].timeMs = now;
+ _shortMax = 0;
+ }
+}
+
+void
+VCMCodecTimer::ProcessHistory(WebRtc_Word64 nowMs)
+{
+ _filteredMax = _shortMax;
+ if (_history[0].timeMs == -1)
+ {
+ return;
+ }
+ for (int i=0; i < MAX_HISTORY_SIZE; i++)
+ {
+ if (_history[i].timeMs == -1)
+ {
+ break;
+ }
+ if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
+ {
+ // This sample (and all samples after this) is too old
+ break;
+ }
+ if (_history[i].shortMax > _filteredMax)
+ {
+ // This sample is the largest one this far into the history
+ _filteredMax = _history[i].shortMax;
+ }
+ }
+}
+
+// Get the maximum observed time within a time window
+WebRtc_Word32 VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
+{
+ return _filteredMax;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/codec_timer.h b/trunk/src/modules/video_coding/main/source/codec_timer.h
new file mode 100644
index 0000000..e03c5bf
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/codec_timer.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+
+// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
+#define MAX_HISTORY_SIZE 20
+#define SHORT_FILTER_MS 1000
+
+class VCMShortMaxSample
+{
+public:
+ VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
+
+ WebRtc_Word32 shortMax;
+ WebRtc_Word64 timeMs;
+};
+
+class VCMCodecTimer
+{
+public:
+ VCMCodecTimer();
+
+ // Updates and returns the max filtered decode time.
+ WebRtc_Word32 StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs);
+
+ // Empty the list of timers.
+ void Reset();
+
+ // Get the required decode time in ms.
+ WebRtc_Word32 RequiredDecodeTimeMs(FrameType frameType) const;
+
+private:
+ void UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now);
+ void MaxFilter(WebRtc_Word32 newTime, WebRtc_Word64 nowMs);
+ void ProcessHistory(WebRtc_Word64 nowMs);
+
+ WebRtc_Word32 _filteredMax;
+ bool _firstDecodeTime;
+ WebRtc_Word32 _shortMax;
+ VCMShortMaxSample _history[MAX_HISTORY_SIZE];
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc b/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc
new file mode 100644
index 0000000..99160c9
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc
@@ -0,0 +1,125 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/content_metrics_processing.h"
+
+#include <math.h>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+
+namespace webrtc {
+//////////////////////////////////
+/// VCMContentMetricsProcessing //
+//////////////////////////////////
+
+VCMContentMetricsProcessing::VCMContentMetricsProcessing()
+ : recursive_avg_factor_(1 / 150.0f), // matched to 30fps.
+ frame_cnt_uniform_avg_(0),
+ avg_motion_level_(0.0f),
+ avg_spatial_level_(0.0f) {
+ recursive_avg_ = new VideoContentMetrics();
+ uniform_avg_ = new VideoContentMetrics();
+}
+
+VCMContentMetricsProcessing::~VCMContentMetricsProcessing() {
+ delete recursive_avg_;
+ delete uniform_avg_;
+}
+
+int VCMContentMetricsProcessing::Reset() {
+ recursive_avg_->Reset();
+ uniform_avg_->Reset();
+ frame_cnt_uniform_avg_ = 0;
+ avg_motion_level_ = 0.0f;
+ avg_spatial_level_ = 0.0f;
+ return VCM_OK;
+}
+
+void VCMContentMetricsProcessing::UpdateFrameRate(uint32_t frameRate) {
+ // Update factor for recursive averaging.
+ recursive_avg_factor_ = static_cast<float> (1000.0f) /
+ static_cast<float>(frameRate * kQmMinIntervalMs);
+}
+
+VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
+ return recursive_avg_;
+}
+
+VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
+ if (frame_cnt_uniform_avg_ == 0) {
+ return NULL;
+ }
+ // Two metrics are used: motion and spatial level.
+ uniform_avg_->motion_magnitude = avg_motion_level_ /
+ static_cast<float>(frame_cnt_uniform_avg_);
+ uniform_avg_->spatial_pred_err = avg_spatial_level_ /
+ static_cast<float>(frame_cnt_uniform_avg_);
+ return uniform_avg_;
+}
+
+void VCMContentMetricsProcessing::ResetShortTermAvgData() {
+ // Reset.
+ avg_motion_level_ = 0.0f;
+ avg_spatial_level_ = 0.0f;
+ frame_cnt_uniform_avg_ = 0;
+}
+
+int VCMContentMetricsProcessing::UpdateContentData(
+ const VideoContentMetrics *contentMetrics) {
+ if (contentMetrics == NULL) {
+ return VCM_OK;
+ }
+ return ProcessContent(contentMetrics);
+}
+
+int VCMContentMetricsProcessing::ProcessContent(
+ const VideoContentMetrics *contentMetrics) {
+ // Update the recursive averaged metrics: average is over longer window
+ // of time: over QmMinIntervalMs ms.
+ UpdateRecursiveAvg(contentMetrics);
+ // Update the uniform averaged metrics: average is over shorter window
+ // of time: based on ~RTCP reports.
+ UpdateUniformAvg(contentMetrics);
+ return VCM_OK;
+}
+
+void VCMContentMetricsProcessing::UpdateUniformAvg(
+ const VideoContentMetrics *contentMetrics) {
+ // Update frame counter.
+ frame_cnt_uniform_avg_ += 1;
+ // Update averaged metrics: motion and spatial level are used.
+ avg_motion_level_ += contentMetrics->motion_magnitude;
+ avg_spatial_level_ += contentMetrics->spatial_pred_err;
+ return;
+}
+
+void VCMContentMetricsProcessing::UpdateRecursiveAvg(
+ const VideoContentMetrics *contentMetrics) {
+
+ // Spatial metrics: 2x2, 1x2(H), 2x1(V).
+ recursive_avg_->spatial_pred_err = (1 - recursive_avg_factor_) *
+ recursive_avg_->spatial_pred_err +
+ recursive_avg_factor_ * contentMetrics->spatial_pred_err;
+
+ recursive_avg_->spatial_pred_err_h = (1 - recursive_avg_factor_) *
+ recursive_avg_->spatial_pred_err_h +
+ recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
+
+ recursive_avg_->spatial_pred_err_v = (1 - recursive_avg_factor_) *
+ recursive_avg_->spatial_pred_err_v +
+ recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
+
+ // Motion metric: Derived from NFD (normalized frame difference).
+ recursive_avg_->motion_magnitude = (1 - recursive_avg_factor_) *
+ recursive_avg_->motion_magnitude +
+ recursive_avg_factor_ * contentMetrics->motion_magnitude;
+}
+} // end of namespace
diff --git a/trunk/src/modules/video_coding/main/source/content_metrics_processing.h b/trunk/src/modules/video_coding/main/source/content_metrics_processing.h
new file mode 100644
index 0000000..0317add
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/content_metrics_processing.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+struct VideoContentMetrics;
+
+// QM interval time (in ms)
+enum {
+ kQmMinIntervalMs = 10000
+};
+
+// Flag for NFD metric vs motion metric
+enum {
+ kNfdMetric = 1
+};
+
+/**********************************/
+/* Content Metrics Processing */
+/**********************************/
+class VCMContentMetricsProcessing {
+ public:
+ VCMContentMetricsProcessing();
+ ~VCMContentMetricsProcessing();
+
+ // Update class with latest metrics.
+ int UpdateContentData(const VideoContentMetrics *contentMetrics);
+
+ // Reset the short-term averaged content data.
+ void ResetShortTermAvgData();
+
+ // Initialize.
+ int Reset();
+
+ // Inform class of current frame rate.
+ void UpdateFrameRate(uint32_t frameRate);
+
+ // Returns the long-term averaged content data: recursive average over longer
+ // time scale.
+ VideoContentMetrics* LongTermAvgData();
+
+ // Returns the short-term averaged content data: uniform average over
+ // shorter time scalE.
+ VideoContentMetrics* ShortTermAvgData();
+
+ private:
+ // Compute working average.
+ int ProcessContent(const VideoContentMetrics *contentMetrics);
+
+ // Update the recursive averaged metrics: longer time average (~5/10 secs).
+ void UpdateRecursiveAvg(const VideoContentMetrics *contentMetrics);
+
+ // Update the uniform averaged metrics: shorter time average (~RTCP report).
+ void UpdateUniformAvg(const VideoContentMetrics *contentMetrics);
+
+ VideoContentMetrics* recursive_avg_;
+ VideoContentMetrics* uniform_avg_;
+ float recursive_avg_factor_;
+ uint32_t frame_cnt_uniform_avg_;
+ float avg_motion_level_;
+ float avg_spatial_level_;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state.cc b/trunk/src/modules/video_coding/main/source/decoding_state.cc
new file mode 100644
index 0000000..eea0a2c
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/decoding_state.cc
@@ -0,0 +1,202 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/decoding_state.h"
+
+#include "modules/video_coding/main/source/frame_buffer.h"
+#include "modules/video_coding/main/source/jitter_buffer_common.h"
+#include "modules/video_coding/main/source/packet.h"
+#include "modules/interface/module_common_types.h"
+
+namespace webrtc {
+
+VCMDecodingState::VCMDecodingState()
+ : sequence_num_(0),
+ time_stamp_(0),
+ picture_id_(kNoPictureId),
+ temporal_id_(kNoTemporalIdx),
+ tl0_pic_id_(kNoTl0PicIdx),
+ full_sync_(true),
+ init_(true) {}
+
+VCMDecodingState::~VCMDecodingState() {}
+
+void VCMDecodingState::Reset() {
+ // TODO(mikhal): Verify - not always would want to reset the sync
+ sequence_num_ = 0;
+ time_stamp_ = 0;
+ picture_id_ = kNoPictureId;
+ temporal_id_ = kNoTemporalIdx;
+ tl0_pic_id_ = kNoTl0PicIdx;
+ full_sync_ = true;
+ init_ = true;
+}
+
+uint32_t VCMDecodingState::time_stamp() const {
+ return time_stamp_;
+}
+
+uint16_t VCMDecodingState::sequence_num() const {
+ return sequence_num_;
+}
+
+bool VCMDecodingState::IsOldFrame(const VCMFrameBuffer* frame) const {
+ assert(frame != NULL);
+ if (init_)
+ return false;
+ return (LatestTimestamp(time_stamp_, frame->TimeStamp(), NULL)
+ == time_stamp_);
+}
+
+bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
+ assert(packet != NULL);
+ if (init_)
+ return false;
+ return (LatestTimestamp(time_stamp_, packet->timestamp, NULL)
+ == time_stamp_);
+}
+
+void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
+ assert(frame != NULL && frame->GetHighSeqNum() >= 0);
+ UpdateSyncState(frame);
+ sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
+ time_stamp_ = frame->TimeStamp();
+ picture_id_ = frame->PictureId();
+ temporal_id_ = frame->TemporalId();
+ tl0_pic_id_ = frame->Tl0PicId();
+ init_ = false;
+}
+
+void VCMDecodingState::SetStateOneBack(const VCMFrameBuffer* frame) {
+ assert(frame != NULL && frame->GetHighSeqNum() >= 0);
+ sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum()) - 1u;
+ time_stamp_ = frame->TimeStamp() - 1u;
+ temporal_id_ = frame->TemporalId();
+ if (frame->PictureId() != kNoPictureId) {
+ if (frame->PictureId() == 0)
+ picture_id_ = 0x7FFF;
+ else
+ picture_id_ = frame->PictureId() - 1;
+ }
+ if (frame->Tl0PicId() != kNoTl0PicIdx) {
+ if (frame->Tl0PicId() == 0)
+ tl0_pic_id_ = 0x00FF;
+ else
+ tl0_pic_id_ = frame->Tl0PicId() - 1;
+ }
+ init_ = false;
+}
+
+void VCMDecodingState::UpdateOldPacket(const VCMPacket* packet) {
+ assert(packet != NULL);
+ if (packet->timestamp == time_stamp_) {
+ // Late packet belonging to the last decoded frame - make sure we update the
+ // last decoded sequence number.
+ sequence_num_ = LatestSequenceNumber(packet->seqNum, sequence_num_, NULL);
+ }
+}
+
+void VCMDecodingState::SetSeqNum(uint16_t new_seq_num) {
+ sequence_num_ = new_seq_num;
+}
+
+bool VCMDecodingState::init() const {
+ return init_;
+}
+
+bool VCMDecodingState::full_sync() const {
+ return full_sync_;
+}
+
+void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
+ if (init_)
+ return;
+ if (frame->TemporalId() == kNoTemporalIdx ||
+ frame->Tl0PicId() == kNoTl0PicIdx) {
+ full_sync_ = true;
+ } else if (frame->FrameType() == kVideoFrameKey || frame->LayerSync()) {
+ full_sync_ = true;
+ } else if (full_sync_) {
+ // Verify that we are still in sync.
+ // Sync will be broken if continuity is true for layers but not for the
+ // other methods (PictureId and SeqNum).
+ if (!ContinuousPictureId(frame->PictureId()) &&
+ !ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()))) {
+ // Non-layered methods have failed.
+ full_sync_ = false;
+ }
+ }
+}
+
+bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
+ // Check continuity based on the following hierarchy:
+ // - Temporal layers (stop here if out of sync).
+ // - Picture Id when available.
+ // - Sequence numbers.
+ // Return true when in initial state.
+ // Note that when a method is not applicable it will return false.
+ assert(frame != NULL);
+ if (init_)
+ return true;
+
+ if (!ContinuousLayer(frame->TemporalId(), frame->Tl0PicId())) {
+ // Base layers are not continuous or temporal layers are inactive.
+ // In the presence of temporal layers, check for Picture ID/sequence number
+ // continuity if sync can be restored by this frame.
+ if (!full_sync_ && !frame->LayerSync())
+ return false;
+ else if (!ContinuousPictureId(frame->PictureId()))
+ return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
+ }
+ return true;
+}
+
+bool VCMDecodingState::ContinuousPictureId(int picture_id) const {
+ // First, check if applicable.
+ if (picture_id == kNoPictureId || picture_id_ == kNoPictureId)
+ return false;
+
+ int next_picture_id = picture_id_ + 1;
+ if (picture_id < picture_id_) {
+ // Wrap
+ if (picture_id_ >= 0x80) {
+ // 15 bits used for picture id
+ return ((next_picture_id & 0x7FFF) == picture_id);
+ } else {
+ // 7 bits used for picture id
+ return ((next_picture_id & 0x7F) == picture_id);
+ }
+ }
+ // No wrap
+ return (next_picture_id == picture_id);
+}
+
+bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
+ return (seq_num == static_cast<uint16_t>(sequence_num_ + 1));
+}
+
+bool VCMDecodingState::ContinuousLayer(int temporal_id,
+ int tl0_pic_id) const {
+ // First, check if applicable.
+ if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
+ return false;
+ // If this is the first frame to use temporal layers, make sure we start
+ // from base.
+ else if (tl0_pic_id_ == kNoTl0PicIdx && temporal_id_ == kNoTemporalIdx &&
+ temporal_id == 0)
+ return true;
+
+ // Current implementation: Look for base layer continuity.
+ if (temporal_id != 0)
+ return false;
+ return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state.h b/trunk/src/modules/video_coding/main/source/decoding_state.h
new file mode 100644
index 0000000..afec3ba
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/decoding_state.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations
+class VCMFrameBuffer;
+class VCMPacket;
+
+class VCMDecodingState {
+ public:
+ VCMDecodingState();
+ ~VCMDecodingState();
+ // Check for old frame
+ bool IsOldFrame(const VCMFrameBuffer* frame) const;
+ // Check for old packet
+ bool IsOldPacket(const VCMPacket* packet) const;
+ // Check for frame continuity based on current decoded state. Use best method
+ // possible, i.e. temporal info, picture ID or sequence number.
+ bool ContinuousFrame(const VCMFrameBuffer* frame) const;
+ void SetState(const VCMFrameBuffer* frame);
+ // Set the decoding state one frame back.
+ void SetStateOneBack(const VCMFrameBuffer* frame);
+ // Update the sequence number if the timestamp matches current state and the
+ // sequence number is higher than the current one. This accounts for packets
+ // arriving late.
+ void UpdateOldPacket(const VCMPacket* packet);
+ void SetSeqNum(uint16_t new_seq_num);
+ void Reset();
+ uint32_t time_stamp() const;
+ uint16_t sequence_num() const;
+ // Return true if at initial state.
+ bool init() const;
+ // Return true when sync is on - decode all layers.
+ bool full_sync() const;
+
+ private:
+ void UpdateSyncState(const VCMFrameBuffer* frame);
+ // Designated continuity functions
+ bool ContinuousPictureId(int picture_id) const;
+ bool ContinuousSeqNum(uint16_t seq_num) const;
+ bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
+
+ // Keep state of last decoded frame.
+ // TODO(mikhal/stefan): create designated classes to handle these types.
+ uint16_t sequence_num_;
+ uint32_t time_stamp_;
+ int picture_id_;
+ int temporal_id_;
+ int tl0_pic_id_;
+ bool full_sync_; // Sync flag when temporal layers are used.
+ bool init_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc b/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc
new file mode 100644
index 0000000..853f42a
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc
@@ -0,0 +1,430 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "modules/video_coding/main/source/decoding_state.h"
+#include "modules/video_coding/main/source/frame_buffer.h"
+#include "gtest/gtest.h"
+#include "modules/video_coding/main/source/jitter_buffer_common.h"
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/source/packet.h"
+
+namespace webrtc {
+
+
+TEST(TestDecodingState, Sanity) {
+ VCMDecodingState dec_state;
+ dec_state.Reset();
+ EXPECT_TRUE(dec_state.init());
+ EXPECT_TRUE(dec_state.full_sync());
+}
+
+TEST(TestDecodingState, FrameContinuity) {
+ VCMDecodingState dec_state;
+ // Check that makes decision based on correct method.
+ VCMFrameBuffer frame;
+ frame.SetState(kStateEmpty);
+ VCMPacket* packet = new VCMPacket();
+ packet->isFirstPacket = 1;
+ packet->timestamp = 1;
+ packet->seqNum = 0xffff;
+ packet->frameType = kVideoFrameDelta;
+ packet->codecSpecificHeader.codec = kRTPVideoVP8;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x007F;
+ frame.InsertPacket(*packet, 0, false, 0);
+ // Should return true on init.
+ dec_state.Reset();
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ frame.Reset();
+ // Use pictureId
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x0002;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
+ packet->seqNum = 10;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+ // Use sequence numbers.
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->seqNum = dec_state.sequence_num() - 1u;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->seqNum = dec_state.sequence_num() + 1u;
+ frame.InsertPacket(*packet, 0, false, 0);
+ // Insert another packet to this frame
+ packet->seqNum++;
+ frame.InsertPacket(*packet, 0, false, 0);
+ // Verify wrap.
+ EXPECT_EQ(dec_state.sequence_num(), 0xffff);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+
+ // Insert packet with temporal info.
+ dec_state.Reset();
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
+ packet->seqNum = 1;
+ packet->timestamp = 1;
+ EXPECT_TRUE(dec_state.full_sync());
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ // 1 layer up - still good.
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
+ packet->seqNum = 2;
+ packet->timestamp = 2;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ // Lost non-base layer packet => should update sync parameter.
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
+ packet->seqNum = 4;
+ packet->timestamp = 4;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ // Now insert the next non-base layer (belonging to a next tl0PicId).
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
+ packet->seqNum = 5;
+ packet->timestamp = 5;
+ frame.InsertPacket(*packet, 0, false, 0);
+ // Checking continuity and not updating the state - this should not trigger
+ // an update of sync state.
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ EXPECT_TRUE(dec_state.full_sync());
+ // Next base layer (dropped interim non-base layers) - should update sync.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
+ packet->seqNum = 6;
+ packet->timestamp = 6;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.full_sync());
+
+ // Check wrap for temporal layers.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x00FF;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
+ packet->seqNum = 7;
+ packet->timestamp = 7;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.full_sync());
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x0000;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 7;
+ packet->seqNum = 8;
+ packet->timestamp = 8;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ // The current frame is not continuous
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ delete packet;
+}
+
+TEST(TestDecodingState, SetStateOneBack) {
+ VCMDecodingState dec_state;
+ VCMFrameBuffer frame;
+ frame.SetState(kStateEmpty);
+ VCMPacket* packet = new VCMPacket();
+ // Based on PictureId.
+ packet->frameType = kVideoFrameDelta;
+ packet->codecSpecificHeader.codec = kRTPVideoVP8;
+ packet->timestamp = 0;
+ packet->seqNum = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
+ packet->frameType = kVideoFrameDelta;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetStateOneBack(&frame);
+ EXPECT_EQ(dec_state.sequence_num(), 0xFFFF);
+ // Check continuity.
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+ // Based on Temporal layers.
+ packet->timestamp = 0;
+ packet->seqNum = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
+ packet->frameType = kVideoFrameDelta;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetStateOneBack(&frame);
+ // Check continuity
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ delete packet;
+}
+
+TEST(TestDecodingState, UpdateOldPacket) {
+ VCMDecodingState dec_state;
+ // Update only if zero size and newer than previous.
+ // Should only update if the timeStamp match.
+ VCMFrameBuffer frame;
+ frame.SetState(kStateEmpty);
+ VCMPacket* packet = new VCMPacket();
+ packet->timestamp = 1;
+ packet->seqNum = 1;
+ packet->frameType = kVideoFrameDelta;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ EXPECT_EQ(dec_state.sequence_num(), 1);
+ // Insert an empty packet that does not belong to the same frame.
+ // => Sequence num should be the same.
+ packet->timestamp = 2;
+ dec_state.UpdateOldPacket(packet);
+ EXPECT_EQ(dec_state.sequence_num(), 1);
+ // Now insert empty packet belonging to the same frame.
+ packet->timestamp = 1;
+ packet->seqNum = 2;
+ packet->frameType = kFrameEmpty;
+ packet->sizeBytes = 0;
+ dec_state.UpdateOldPacket(packet);
+ EXPECT_EQ(dec_state.sequence_num(), 2);
+ // Now insert delta packet belonging to the same frame.
+ packet->timestamp = 1;
+ packet->seqNum = 3;
+ packet->frameType = kVideoFrameDelta;
+ packet->sizeBytes = 1400;
+ dec_state.UpdateOldPacket(packet);
+ EXPECT_EQ(dec_state.sequence_num(), 3);
+ // Insert a packet belonging to an older timestamp - should not update the
+ // sequence number.
+ packet->timestamp = 0;
+ packet->seqNum = 4;
+ packet->frameType = kFrameEmpty;
+ packet->sizeBytes = 0;
+ dec_state.UpdateOldPacket(packet);
+ EXPECT_EQ(dec_state.sequence_num(), 3);
+
+ delete packet;
+}
+
+TEST(TestDecodingState, MultiLayerBehavior) {
+ // Identify sync/non-sync when more than one layer.
+ VCMDecodingState dec_state;
+ // Identify packets belonging to old frames/packets.
+ // Set state for current frames.
+ // tl0PicIdx 0, temporal id 0.
+ VCMFrameBuffer frame;
+ VCMPacket* packet = new VCMPacket();
+ packet->frameType = kVideoFrameDelta;
+ packet->codecSpecificHeader.codec = kRTPVideoVP8;
+ frame.SetState(kStateEmpty);
+ packet->timestamp = 0;
+ packet->seqNum = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ // tl0PicIdx 0, temporal id 1.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->timestamp = 1;
+ packet->seqNum = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+ // Lost tl0PicIdx 0, temporal id 2.
+ // Insert tl0PicIdx 0, temporal id 3.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->timestamp = 3;
+ packet->seqNum = 3;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.full_sync());
+ // Insert next base layer
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->timestamp = 4;
+ packet->seqNum = 4;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.full_sync());
+ // Insert key frame - should update sync value.
+ // A key frame is always a base layer.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameKey;
+ packet->isFirstPacket = 1;
+ packet->timestamp = 5;
+ packet->seqNum = 5;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 2;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+ // After sync, a continuous PictureId is required
+ // (continuous base layer is not enough )
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->timestamp = 6;
+ packet->seqNum = 6;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 3;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ EXPECT_TRUE(dec_state.full_sync());
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->isFirstPacket = 1;
+ packet->timestamp = 8;
+ packet->seqNum = 8;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 8;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ EXPECT_TRUE(dec_state.full_sync());
+ dec_state.SetState(&frame);
+ EXPECT_FALSE(dec_state.full_sync());
+
+ // Insert a non-ref frame - should update sync value.
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->isFirstPacket = 1;
+ packet->timestamp = 9;
+ packet->seqNum = 9;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 9;
+ packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+
+ // The following test will verify the sync flag behavior after a loss.
+ // Create the following pattern:
+ // Update base layer, lose packet 1 (sync flag on, layer 2), insert packet 3
+ // (sync flag on, layer 2) check continuity and sync flag after inserting
+ // packet 2 (sync flag on, layer 1).
+ // Base layer.
+ frame.Reset();
+ dec_state.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->isFirstPacket = 1;
+ packet->markerBit = 1;
+ packet->timestamp = 0;
+ packet->seqNum = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.layerSync = false;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ EXPECT_TRUE(dec_state.full_sync());
+ // Layer 2 - 2 packets (insert one, lose one).
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->isFirstPacket = 1;
+ packet->markerBit = 0;
+ packet->timestamp = 1;
+ packet->seqNum = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+ // Layer 1
+ frame.Reset();
+ frame.SetState(kStateEmpty);
+ packet->frameType = kVideoFrameDelta;
+ packet->isFirstPacket = 1;
+ packet->markerBit = 1;
+ packet->timestamp = 2;
+ packet->seqNum = 3;
+ packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
+ packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
+ packet->codecSpecificHeader.codecHeader.VP8.pictureId = 2;
+ packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+ EXPECT_TRUE(dec_state.full_sync());
+
+ delete packet;
+}
+
+TEST(TestDecodingState, OldInput) {
+ VCMDecodingState dec_state;
+ // Identify packets belonging to old frames/packets.
+ // Set state for current frames.
+ VCMFrameBuffer frame;
+ frame.SetState(kStateEmpty);
+ VCMPacket* packet = new VCMPacket();
+ packet->timestamp = 10;
+ packet->seqNum = 1;
+ frame.InsertPacket(*packet, 0, false, 0);
+ dec_state.SetState(&frame);
+ packet->timestamp = 9;
+ EXPECT_TRUE(dec_state.IsOldPacket(packet));
+ // Check for old frame
+ frame.Reset();
+ frame.InsertPacket(*packet, 0, false, 0);
+ EXPECT_TRUE(dec_state.IsOldFrame(&frame));
+
+
+ delete packet;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/encoded_frame.cc b/trunk/src/modules/video_coding/main/source/encoded_frame.cc
new file mode 100644
index 0000000..8e5d745
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/encoded_frame.cc
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "encoded_frame.h"
+#include "generic_encoder.h"
+#include "jitter_buffer_common.h"
+#include "video_coding_defines.h"
+
+namespace webrtc {
+
+VCMEncodedFrame::VCMEncodedFrame()
+:
+webrtc::EncodedImage(),
+_renderTimeMs(-1),
+_payloadType(0),
+_missingFrame(false),
+_codec(kVideoCodecUnknown),
+_fragmentation()
+{
+ _codecSpecificInfo.codecType = kVideoCodecUnknown;
+}
+
+VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
+:
+webrtc::EncodedImage(rhs),
+_renderTimeMs(-1),
+_payloadType(0),
+_missingFrame(false),
+_codec(kVideoCodecUnknown),
+_fragmentation()
+{
+ _codecSpecificInfo.codecType = kVideoCodecUnknown;
+ _buffer = NULL;
+ _size = 0;
+ _length = 0;
+ if (rhs._buffer != NULL)
+ {
+ VerifyAndAllocate(rhs._length);
+ memcpy(_buffer, rhs._buffer, rhs._length);
+ }
+}
+
+VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
+ :
+ webrtc::EncodedImage(rhs),
+ _renderTimeMs(rhs._renderTimeMs),
+ _payloadType(rhs._payloadType),
+ _missingFrame(rhs._missingFrame),
+ _codecSpecificInfo(rhs._codecSpecificInfo),
+ _codec(rhs._codec),
+ _fragmentation() {
+ _buffer = NULL;
+ _size = 0;
+ _length = 0;
+ if (rhs._buffer != NULL)
+ {
+ VerifyAndAllocate(rhs._length);
+ memcpy(_buffer, rhs._buffer, rhs._length);
+ _length = rhs._length;
+ }
+ // Deep operator=
+ _fragmentation = rhs._fragmentation;
+}
+
+VCMEncodedFrame::~VCMEncodedFrame()
+{
+ Free();
+}
+
+void VCMEncodedFrame::Free()
+{
+ Reset();
+ if (_buffer != NULL)
+ {
+ delete [] _buffer;
+ _buffer = NULL;
+ }
+}
+
+void VCMEncodedFrame::Reset()
+{
+ _renderTimeMs = -1;
+ _timeStamp = 0;
+ _payloadType = 0;
+ _frameType = kDeltaFrame;
+ _encodedWidth = 0;
+ _encodedHeight = 0;
+ _completeFrame = false;
+ _missingFrame = false;
+ _length = 0;
+ _codecSpecificInfo.codecType = kVideoCodecUnknown;
+ _codec = kVideoCodecUnknown;
+}
+
+void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
+{
+ if (header)
+ {
+ switch (header->codec)
+ {
+ case kRTPVideoVP8:
+ {
+ if (_codecSpecificInfo.codecType != kVideoCodecVP8)
+ {
+ // This is the first packet for this frame.
+ _codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
+ _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
+ _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
+ _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
+ _codecSpecificInfo.codecType = kVideoCodecVP8;
+ }
+ _codecSpecificInfo.codecSpecific.VP8.nonReference =
+ header->codecHeader.VP8.nonReference;
+ if (header->codecHeader.VP8.pictureId != kNoPictureId)
+ {
+ _codecSpecificInfo.codecSpecific.VP8.pictureId =
+ header->codecHeader.VP8.pictureId;
+ }
+ if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx)
+ {
+ _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
+ header->codecHeader.VP8.temporalIdx;
+ _codecSpecificInfo.codecSpecific.VP8.layerSync =
+ header->codecHeader.VP8.layerSync;
+ }
+ if (header->codecHeader.VP8.keyIdx != kNoKeyIdx)
+ {
+ _codecSpecificInfo.codecSpecific.VP8.keyIdx =
+ header->codecHeader.VP8.keyIdx;
+ }
+ break;
+ }
+ default:
+ {
+ _codecSpecificInfo.codecType = kVideoCodecUnknown;
+ break;
+ }
+ }
+ }
+}
+
+const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
+ return &_fragmentation;
+}
+
+WebRtc_Word32
+VCMEncodedFrame::Store(VCMFrameStorageCallback& storeCallback) const
+{
+ EncodedVideoData frameToStore;
+ frameToStore.codec = _codec;
+ if (_buffer != NULL)
+ {
+ frameToStore.VerifyAndAllocate(_length);
+ memcpy(frameToStore.payloadData, _buffer, _length);
+ frameToStore.payloadSize = _length;
+ }
+ frameToStore.completeFrame = _completeFrame;
+ frameToStore.encodedWidth = _encodedWidth;
+ frameToStore.encodedHeight = _encodedHeight;
+ frameToStore.frameType = ConvertFrameType(_frameType);
+ frameToStore.missingFrame = _missingFrame;
+ frameToStore.payloadType = _payloadType;
+ frameToStore.renderTimeMs = _renderTimeMs;
+ frameToStore.timeStamp = _timeStamp;
+ storeCallback.StoreReceivedFrame(frameToStore);
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMEncodedFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
+{
+ if(minimumSize > _size)
+ {
+ // create buffer of sufficient size
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[minimumSize];
+ if (newBuffer == NULL)
+ {
+ return -1;
+ }
+ if(_buffer)
+ {
+ // copy old data
+ memcpy(newBuffer, _buffer, _size);
+ delete [] _buffer;
+ }
+ _buffer = newBuffer;
+ _size = minimumSize;
+ }
+ return 0;
+}
+
+webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
+{
+ switch(frameType)
+ {
+ case kKeyFrame:
+ {
+ return kVideoFrameKey;
+ }
+ case kDeltaFrame:
+ {
+ return kVideoFrameDelta;
+ }
+ case kGoldenFrame:
+ {
+ return kVideoFrameGolden;
+ }
+ case kAltRefFrame:
+ {
+ return kVideoFrameAltRef;
+ }
+ default:
+ {
+ return kVideoFrameDelta;
+ }
+ }
+}
+
+VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frameType)
+{
+ switch (frameType)
+ {
+ case kVideoFrameKey:
+ {
+ return kKeyFrame;
+ }
+ case kVideoFrameDelta:
+ {
+ return kDeltaFrame;
+ }
+ case kVideoFrameGolden:
+ {
+ return kGoldenFrame;
+ }
+ case kVideoFrameAltRef:
+ {
+ return kAltRefFrame;
+ }
+ default:
+ {
+ return kDeltaFrame;
+ }
+ }
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/encoded_frame.h b/trunk/src/modules/video_coding/main/source/encoded_frame.h
new file mode 100644
index 0000000..6289e9e
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/encoded_frame.h
@@ -0,0 +1,120 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+
+#include "common_types.h"
+#include "common_video/interface/video_image.h"
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+
+namespace webrtc
+{
+
+class VCMEncodedFrame : protected EncodedImage
+{
+public:
+ VCMEncodedFrame();
+ VCMEncodedFrame(const webrtc::EncodedImage& rhs);
+ VCMEncodedFrame(const VCMEncodedFrame& rhs);
+
+ ~VCMEncodedFrame();
+ /**
+ * Delete VideoFrame and resets members to zero
+ */
+ void Free();
+ /**
+ * Set render time in milliseconds
+ */
+ void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
+
+ /**
+ * Set the encoded frame size
+ */
+ void SetEncodedSize(WebRtc_UWord32 width, WebRtc_UWord32 height)
+ { _encodedWidth = width; _encodedHeight = height; }
+ /**
+ * Get the encoded image
+ */
+ const webrtc::EncodedImage& EncodedImage() const
+ { return static_cast<const webrtc::EncodedImage&>(*this); }
+ /**
+ * Get pointer to frame buffer
+ */
+ const WebRtc_UWord8* Buffer() const {return _buffer;}
+ /**
+ * Get frame length
+ */
+ WebRtc_UWord32 Length() const {return _length;}
+ /**
+ * Get frame timestamp (90kHz)
+ */
+ WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
+ /**
+ * Get render time in milliseconds
+ */
+ WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
+ /**
+ * Get frame type
+ */
+ webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
+ /**
+ * True if this frame is complete, false otherwise
+ */
+ bool Complete() const { return _completeFrame; }
+ /**
+ * True if there's a frame missing before this frame
+ */
+ bool MissingFrame() const { return _missingFrame; }
+ /**
+ * Payload type of the encoded payload
+ */
+ WebRtc_UWord8 PayloadType() const { return _payloadType; }
+ /**
+ * Get codec specific info.
+ * The returned pointer is only valid as long as the VCMEncodedFrame
+ * is valid. Also, VCMEncodedFrame owns the pointer and will delete
+ * the object.
+ */
+ const CodecSpecificInfo* CodecSpecific() const {return &_codecSpecificInfo;}
+
+ const RTPFragmentationHeader* FragmentationHeader() const;
+
+ WebRtc_Word32 Store(VCMFrameStorageCallback& storeCallback) const;
+
+ static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
+ static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
+
+protected:
+ /**
+ * Verifies that current allocated buffer size is larger than or equal to the input size.
+ * If the current buffer size is smaller, a new allocation is made and the old buffer data
+ * is copied to the new buffer.
+ * Buffer size is updated to minimumSize.
+ */
+ WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
+
+ void Reset();
+
+ void CopyCodecSpecific(const RTPVideoHeader* header);
+
+ WebRtc_Word64 _renderTimeMs;
+ WebRtc_UWord8 _payloadType;
+ bool _missingFrame;
+ CodecSpecificInfo _codecSpecificInfo;
+ webrtc::VideoCodecType _codec;
+ RTPFragmentationHeader _fragmentation;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/trunk/src/modules/video_coding/main/source/er_tables_xor.h b/trunk/src/modules/video_coding/main/source/er_tables_xor.h
new file mode 100644
index 0000000..99163ba
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/er_tables_xor.h
@@ -0,0 +1,38742 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
+
+// This is a private header for media_opt_util.cc.
+// It should not be included by other files.
+
+namespace webrtc {
+
+// Table for average FEC recovery from packet loss, for XOR code.
+// From RPL model of random loss.
+// Input is the received packet loss (up to 50%), and FEC code parameters
+// (up to 24x24):
+// i.e., kAvgFECRecoveryXOR[k] where k = code_i*129 + loss_j;
+// code_i=1x1,2x1,2x2,..24x24, loss_j = 0,1,..128.
+
+// Maximum number of source packets in off-line model
+static const int kMaxNumPackets = 24;
+// Max value of loss rates in off-line model
+static const int kPacketLossMax = 129;
+
+// Table size for model is: kPacketLossMax * numberOfFecCodes = 38700
+// numberOfFecCodes is determined as:
+// {(1,1), (2,1), (2,2),...(n,1),..(n,n-1), (n,n)} = n*(n+1)/2
+// for n = kMaxNumPackets.
+static const int kSizeAvgFECRecoveryXOR = 38700;
+static const unsigned char kAvgFECRecoveryXOR[kSizeAvgFECRecoveryXOR] = {
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+18,
+18,
+17,
+16,
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/event.h b/trunk/src/modules/video_coding/main/source/event.h
new file mode 100644
index 0000000..39fd494
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/event.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
+#define WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
+
+#include "event_wrapper.h"
+
+namespace webrtc
+{
+
+//#define EVENT_DEBUG
+
+class VCMEvent : public EventWrapper
+{
+public:
+ VCMEvent() : _event(*EventWrapper::Create()) {};
+
+ virtual ~VCMEvent() { delete &_event; };
+
+ /**
+ * Release waiting threads
+ */
+ bool Set() { return _event.Set(); };
+
+ bool Reset() { return _event.Reset(); };
+
+ /**
+ * Wait for this event
+ */
+ EventTypeWrapper Wait(unsigned long maxTime)
+ {
+#ifdef EVENT_DEBUG
+ return kEventTimeout;
+#else
+ return _event.Wait(maxTime);
+#endif
+ };
+
+ /**
+ * Start a timer
+ */
+ bool StartTimer(bool periodic, unsigned long time)
+ { return _event.StartTimer(periodic, time); };
+ /**
+ * Stop the timer
+ */
+ bool StopTimer() { return _event.StopTimer(); };
+
+private:
+ EventWrapper& _event;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
diff --git a/trunk/src/modules/video_coding/main/source/exp_filter.cc b/trunk/src/modules/video_coding/main/source/exp_filter.cc
new file mode 100644
index 0000000..1d6f9a7
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/exp_filter.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "exp_filter.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+void
+VCMExpFilter::Reset(float alpha)
+{
+ _alpha = alpha;
+ _filtered = -1.0;
+}
+
+float
+VCMExpFilter::Apply(float exp, float sample)
+{
+ if (_filtered == -1.0)
+ {
+ // Initialize filtered bit rates
+ _filtered = sample;
+ }
+ else if (exp == 1.0)
+ {
+ _filtered = _alpha * _filtered + (1 - _alpha) * sample;
+ }
+ else
+ {
+ float alpha = pow(_alpha, exp);
+ _filtered = alpha * _filtered + (1 - alpha) * sample;
+ }
+ if (_max != -1 && _filtered > _max)
+ {
+ _filtered = _max;
+ }
+ return _filtered;
+}
+
+void
+VCMExpFilter::UpdateBase(float alpha)
+{
+ _alpha = alpha;
+}
+
+float
+VCMExpFilter::Value() const
+{
+ return _filtered;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/exp_filter.h b/trunk/src/modules/video_coding/main/source/exp_filter.h
new file mode 100644
index 0000000..46d206a
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/exp_filter.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
+
+namespace webrtc
+{
+
+/**********************/
+/* ExpFilter class */
+/**********************/
+
+class VCMExpFilter
+{
+public:
+ VCMExpFilter(float alpha, float max = -1.0) : _alpha(alpha), _filtered(-1.0), _max(max) {}
+
+ // Resets the filter to its initial state, and resets alpha to the given value
+ //
+ // Input:
+ // - alpha : the new value of the filter factor base.
+ void Reset(float alpha);
+
+ // Applies the filter with the given exponent on the provided sample
+ //
+ // Input:
+ // - exp : Exponent T in y(k) = alpha^T * y(k-1) + (1 - alpha^T) * x(k)
+ // - sample : x(k) in the above filter equation
+ float Apply(float exp, float sample);
+
+ // Return current filtered value: y(k)
+ //
+ // Return value : The current filter output
+ float Value() const;
+
+ // Change the filter factor base
+ //
+ // Input:
+ // - alpha : The new filter factor base.
+ void UpdateBase(float alpha);
+
+private:
+ float _alpha; // Filter factor base
+ float _filtered; // Current filter output
+ const float _max;
+}; // end of ExpFilter class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
diff --git a/trunk/src/modules/video_coding/main/source/fec_tables_xor.h b/trunk/src/modules/video_coding/main/source/fec_tables_xor.h
new file mode 100644
index 0000000..27db9a4
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/fec_tables_xor.h
@@ -0,0 +1,6481 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
+
+// This is a private header for media_opt_util.cc.
+// It should not be included by other files.
+
+namespace webrtc {
+
+// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
+// Input is the packet loss and an effective rate (bits/frame).
+// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
+// loss_j = 0,1,..128, and rate_i varies over some range.
+static const int kSizeCodeRateXORTable = 6450;
+static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
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+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+127,
+
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/frame_buffer.cc b/trunk/src/modules/video_coding/main/source/frame_buffer.cc
new file mode 100644
index 0000000..6339b05
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/frame_buffer.cc
@@ -0,0 +1,415 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "frame_buffer.h"
+#include "packet.h"
+
+#include <cassert>
+#include <string.h>
+
+#if defined(_WIN32)
+ // VS 2005: Don't warn for default initialized arrays. See help for more info.
+ #pragma warning(disable:4351)
+#endif
+
+namespace webrtc {
+
+VCMFrameBuffer::VCMFrameBuffer()
+ :
+ _state(kStateFree),
+ _frameCounted(false),
+ _nackCount(0),
+ _latestPacketTimeMs(-1) {
+}
+
+VCMFrameBuffer::~VCMFrameBuffer() {
+}
+
+VCMFrameBuffer::VCMFrameBuffer(VCMFrameBuffer& rhs)
+:
+VCMEncodedFrame(rhs),
+_state(rhs._state),
+_frameCounted(rhs._frameCounted),
+_sessionInfo(),
+_nackCount(rhs._nackCount),
+_latestPacketTimeMs(rhs._latestPacketTimeMs)
+{
+ _sessionInfo = rhs._sessionInfo;
+ _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
+}
+
+webrtc::FrameType
+VCMFrameBuffer::FrameType() const
+{
+ return _sessionInfo.FrameType();
+}
+
+void
+VCMFrameBuffer::SetPreviousFrameLoss()
+{
+ _sessionInfo.SetPreviousFrameLoss();
+}
+
+WebRtc_Word32
+VCMFrameBuffer::GetLowSeqNum() const
+{
+ return _sessionInfo.LowSequenceNumber();
+}
+
+WebRtc_Word32
+VCMFrameBuffer::GetHighSeqNum() const
+{
+ return _sessionInfo.HighSequenceNumber();
+}
+
+int VCMFrameBuffer::PictureId() const {
+ return _sessionInfo.PictureId();
+}
+
+int VCMFrameBuffer::TemporalId() const {
+ return _sessionInfo.TemporalId();
+}
+
+bool VCMFrameBuffer::LayerSync() const {
+ return _sessionInfo.LayerSync();
+}
+
+int VCMFrameBuffer::Tl0PicId() const {
+ return _sessionInfo.Tl0PicId();
+}
+
+bool VCMFrameBuffer::NonReference() const {
+ return _sessionInfo.NonReference();
+}
+
+bool
+VCMFrameBuffer::IsSessionComplete() const
+{
+ return _sessionInfo.complete();
+}
+
+// Insert packet
+VCMFrameBufferEnum
+VCMFrameBuffer::InsertPacket(const VCMPacket& packet, WebRtc_Word64 timeInMs,
+ bool enableDecodableState, WebRtc_UWord32 rttMS)
+{
+ if (_state == kStateDecoding)
+ {
+ // Do not insert packet
+ return kNoError;
+ }
+
+ // Sanity to check if the frame has been freed. (Too old for example)
+ if (_state == kStateFree)
+ {
+ return kStateError;
+ }
+
+ // is this packet part of this frame
+ if (TimeStamp() && (TimeStamp() != packet.timestamp))
+ {
+ return kTimeStampError;
+ }
+
+ // sanity checks
+ if (_size + packet.sizeBytes +
+ (packet.insertStartCode ? kH264StartCodeLengthBytes : 0 )
+ > kMaxJBFrameSizeBytes)
+ {
+ return kSizeError;
+ }
+ if (NULL == packet.dataPtr && packet.sizeBytes > 0)
+ {
+ return kSizeError;
+ }
+ if (packet.dataPtr != NULL)
+ {
+ _payloadType = packet.payloadType;
+ }
+
+ if (kStateEmpty == _state)
+ {
+ // First packet (empty and/or media) inserted into this frame.
+ // store some info and set some initial values.
+ _timeStamp = packet.timestamp;
+ _codec = packet.codec;
+ if (packet.frameType != kFrameEmpty)
+ {
+ // first media packet
+ SetState(kStateIncomplete);
+ }
+ }
+
+ WebRtc_UWord32 requiredSizeBytes = Length() + packet.sizeBytes +
+ (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+ if (requiredSizeBytes >= _size)
+ {
+ const WebRtc_UWord8* prevBuffer = _buffer;
+ const WebRtc_UWord32 increments = requiredSizeBytes /
+ kBufferIncStepSizeBytes +
+ (requiredSizeBytes %
+ kBufferIncStepSizeBytes > 0);
+ const WebRtc_UWord32 newSize = _size +
+ increments * kBufferIncStepSizeBytes;
+ if (newSize > kMaxJBFrameSizeBytes)
+ {
+ return kSizeError;
+ }
+ if (VerifyAndAllocate(newSize) == -1)
+ {
+ return kSizeError;
+ }
+ _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
+ }
+
+ CopyCodecSpecific(&packet.codecSpecificHeader);
+
+ int retVal = _sessionInfo.InsertPacket(packet, _buffer,
+ enableDecodableState,
+ rttMS);
+ if (retVal == -1)
+ {
+ return kSizeError;
+ }
+ else if (retVal == -2)
+ {
+ return kDuplicatePacket;
+ }
+ // update length
+ _length = Length() + static_cast<WebRtc_UWord32>(retVal);
+
+ _latestPacketTimeMs = timeInMs;
+
+ if (_sessionInfo.complete()) {
+ return kCompleteSession;
+ } else if (_sessionInfo.decodable()) {
+ SetState(kStateDecodable);
+ return kDecodableSession;
+ } else {
+ // this layer is not complete
+ if (_state == kStateComplete) {
+ // we already have a complete layer
+ // wait for all independent layers belonging to the same frame
+ _state = kStateIncomplete;
+ }
+ }
+ return kIncomplete;
+}
+
+WebRtc_Word64
+VCMFrameBuffer::LatestPacketTimeMs()
+{
+ return _latestPacketTimeMs;
+}
+
+// Build hard NACK list:Zero out all entries in list up to and including the
+// (first) entry equal to _lowSeqNum.
+int VCMFrameBuffer::BuildHardNackList(int* list, int num) {
+ if (_sessionInfo.BuildHardNackList(list, num) != 0) {
+ return -1;
+ }
+ return 0;
+}
+
+// Build selective NACK list: Create a soft (selective) list of entries to zero
+// out up to and including the (first) entry equal to _lowSeqNum.
+int VCMFrameBuffer::BuildSoftNackList(int* list, int num, int rttMs) {
+ return _sessionInfo.BuildSoftNackList(list, num, rttMs);
+}
+
+void
+VCMFrameBuffer::IncrementNackCount()
+{
+ _nackCount++;
+}
+
+WebRtc_Word16
+VCMFrameBuffer::GetNackCount() const
+{
+ return _nackCount;
+}
+
+bool
+VCMFrameBuffer::HaveLastPacket() const
+{
+ return _sessionInfo.HaveLastPacket();
+}
+
+void
+VCMFrameBuffer::Reset()
+{
+ _length = 0;
+ _timeStamp = 0;
+ _sessionInfo.Reset();
+ _frameCounted = false;
+ _payloadType = 0;
+ _nackCount = 0;
+ _latestPacketTimeMs = -1;
+ _state = kStateFree;
+ VCMEncodedFrame::Reset();
+}
+
+// Makes sure the session contains a decodable stream.
+void
+VCMFrameBuffer::MakeSessionDecodable()
+{
+ WebRtc_UWord32 retVal;
+#ifdef INDEPENDENT_PARTITIONS
+ if (_codec != kVideoCodecVP8) {
+ retVal = _sessionInfo.MakeDecodable();
+ _length -= retVal;
+ }
+#else
+ retVal = _sessionInfo.MakeDecodable();
+ _length -= retVal;
+#endif
+}
+
+// Set state of frame
+void
+VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state)
+{
+ if (_state == state)
+ {
+ return;
+ }
+ switch (state)
+ {
+ case kStateFree:
+ // Reset everything
+ // We can go to this state from all other states.
+ // The one setting the state to free must ensure
+ // that the frame is removed from the timestamp
+ // ordered frame list in the jb.
+ Reset();
+ break;
+
+ case kStateIncomplete:
+ // we can go to this state from state kStateEmpty
+ assert(_state == kStateEmpty ||
+ _state == kStateDecoding);
+
+ // Do nothing, we received a packet
+ break;
+
+ case kStateComplete:
+ assert(_state == kStateEmpty ||
+ _state == kStateIncomplete ||
+ _state == kStateDecodable);
+
+ break;
+
+ case kStateEmpty:
+ assert(_state == kStateFree);
+ // Do nothing
+ break;
+
+ case kStateDecoding:
+ // A frame might have received empty packets, or media packets might
+ // have been removed when making the frame decodable. The frame can
+ // still be set to decodable since it can be used to inform the
+ // decoder of a frame loss.
+ assert(_state == kStateComplete || _state == kStateIncomplete ||
+ _state == kStateDecodable || _state == kStateEmpty);
+ // Transfer frame information to EncodedFrame and create any codec
+ // specific information
+ RestructureFrameInformation();
+ break;
+
+ case kStateDecodable:
+ assert(_state == kStateEmpty ||
+ _state == kStateIncomplete);
+ break;
+ }
+ _state = state;
+}
+
+void
+VCMFrameBuffer::RestructureFrameInformation()
+{
+ PrepareForDecode();
+ _frameType = ConvertFrameType(_sessionInfo.FrameType());
+ _completeFrame = _sessionInfo.complete();
+ _missingFrame = _sessionInfo.PreviousFrameLoss();
+}
+
+WebRtc_Word32
+VCMFrameBuffer::ExtractFromStorage(const EncodedVideoData& frameFromStorage)
+{
+ _frameType = ConvertFrameType(frameFromStorage.frameType);
+ _timeStamp = frameFromStorage.timeStamp;
+ _payloadType = frameFromStorage.payloadType;
+ _encodedWidth = frameFromStorage.encodedWidth;
+ _encodedHeight = frameFromStorage.encodedHeight;
+ _missingFrame = frameFromStorage.missingFrame;
+ _completeFrame = frameFromStorage.completeFrame;
+ _renderTimeMs = frameFromStorage.renderTimeMs;
+ _codec = frameFromStorage.codec;
+ const WebRtc_UWord8 *prevBuffer = _buffer;
+ if (VerifyAndAllocate(frameFromStorage.payloadSize) < 0)
+ {
+ return VCM_MEMORY;
+ }
+ _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
+ memcpy(_buffer, frameFromStorage.payloadData, frameFromStorage.payloadSize);
+ _length = frameFromStorage.payloadSize;
+ return VCM_OK;
+}
+
+int VCMFrameBuffer::NotDecodablePackets() const {
+ return _sessionInfo.packets_not_decodable();
+}
+
+// Set counted status (as counted by JB or not)
+void VCMFrameBuffer::SetCountedFrame(bool frameCounted)
+{
+ _frameCounted = frameCounted;
+}
+
+bool VCMFrameBuffer::GetCountedFrame() const
+{
+ return _frameCounted;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum
+VCMFrameBuffer::GetState() const
+{
+ return _state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum
+VCMFrameBuffer::GetState(WebRtc_UWord32& timeStamp) const
+{
+ timeStamp = TimeStamp();
+ return GetState();
+}
+
+bool
+VCMFrameBuffer::IsRetransmitted() const
+{
+ return _sessionInfo.session_nack();
+}
+
+void
+VCMFrameBuffer::PrepareForDecode()
+{
+#ifdef INDEPENDENT_PARTITIONS
+ if (_codec == kVideoCodecVP8)
+ {
+ _length =
+ _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
+ &_fragmentation);
+ }
+#endif
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/frame_buffer.h b/trunk/src/modules/video_coding/main/source/frame_buffer.h
new file mode 100644
index 0000000..ea05754
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/frame_buffer.h
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/source/encoded_frame.h"
+#include "modules/video_coding/main/source/jitter_buffer_common.h"
+#include "modules/video_coding/main/source/session_info.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class VCMFrameBuffer : public VCMEncodedFrame
+{
+public:
+ VCMFrameBuffer();
+ virtual ~VCMFrameBuffer();
+
+ VCMFrameBuffer(VCMFrameBuffer& rhs);
+
+ virtual void Reset();
+
+ VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
+ WebRtc_Word64 timeInMs,
+ bool enableDecodableState,
+ WebRtc_UWord32 rttMs);
+
+ // State
+ // Get current state of frame
+ VCMFrameBufferStateEnum GetState() const;
+ // Get current state and timestamp of frame
+ VCMFrameBufferStateEnum GetState(WebRtc_UWord32& timeStamp) const;
+ void SetState(VCMFrameBufferStateEnum state); // Set state of frame
+
+ bool IsRetransmitted() const;
+ bool IsSessionComplete() const;
+ bool HaveLastPacket() const;
+ // Makes sure the session contain a decodable stream.
+ void MakeSessionDecodable();
+
+ // Sequence numbers
+ // Get lowest packet sequence number in frame
+ WebRtc_Word32 GetLowSeqNum() const;
+ // Get highest packet sequence number in frame
+ WebRtc_Word32 GetHighSeqNum() const;
+
+ int PictureId() const;
+ int TemporalId() const;
+ bool LayerSync() const;
+ int Tl0PicId() const;
+ bool NonReference() const;
+
+ // Set counted status (as counted by JB or not)
+ void SetCountedFrame(bool frameCounted);
+ bool GetCountedFrame() const;
+
+ // NACK - Building the NACK lists.
+ // Build hard NACK list: Zero out all entries in list up to and including
+ // _lowSeqNum.
+ int BuildHardNackList(int* list, int num);
+ // Build soft NACK list: Zero out only a subset of the packets, discard
+ // empty packets.
+ int BuildSoftNackList(int* list, int num, int rttMs);
+ void IncrementNackCount();
+ WebRtc_Word16 GetNackCount() const;
+
+ WebRtc_Word64 LatestPacketTimeMs();
+
+ webrtc::FrameType FrameType() const;
+ void SetPreviousFrameLoss();
+
+ WebRtc_Word32 ExtractFromStorage(const EncodedVideoData& frameFromStorage);
+
+ // The number of packets discarded because the decoder can't make use of
+ // them.
+ int NotDecodablePackets() const;
+
+protected:
+ void RestructureFrameInformation();
+ void PrepareForDecode();
+
+private:
+ VCMFrameBufferStateEnum _state; // Current state of the frame
+ bool _frameCounted; // Was this frame counted by JB?
+ VCMSessionInfo _sessionInfo;
+ WebRtc_UWord16 _nackCount;
+ WebRtc_Word64 _latestPacketTimeMs;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/main/source/frame_dropper.cc b/trunk/src/modules/video_coding/main/source/frame_dropper.cc
new file mode 100644
index 0000000..065e452
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/frame_dropper.cc
@@ -0,0 +1,331 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "frame_dropper.h"
+#include "internal_defines.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+VCMFrameDropper::VCMFrameDropper(WebRtc_Word32 vcmId)
+:
+_vcmId(vcmId),
+_keyFrameSizeAvgKbits(0.9f),
+_keyFrameRatio(0.99f),
+_dropRatio(0.9f, 0.96f)
+{
+ Reset();
+}
+
+void
+VCMFrameDropper::Reset()
+{
+ _keyFrameRatio.Reset(0.99f);
+ _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
+ _keyFrameSizeAvgKbits.Reset(0.9f);
+ _keyFrameCount = 0;
+ _accumulator = 0.0f;
+ _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
+ _targetBitRate = 300.0f;
+ _userFrameRate = 30;
+ _keyFrameSpreadFrames = 0.5f * _userFrameRate;
+ _dropNext = false;
+ _dropRatio.Reset(0.9f);
+ _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
+ _dropCount = 0;
+ _windowSize = 0.5f;
+ _wasBelowMax = true;
+ _enabled = true;
+ _fastMode = false; // start with normal (non-aggressive) mode
+}
+
+void
+VCMFrameDropper::Enable(bool enable)
+{
+ _enabled = enable;
+}
+
+void
+VCMFrameDropper::Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame)
+{
+ if (!_enabled)
+ {
+ return;
+ }
+ float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
+ if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
+ {
+ _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
+ _keyFrameRatio.Apply(1.0, 1.0);
+ if (frameSizeKbits > _keyFrameSizeAvgKbits.Value())
+ {
+ // Remove the average key frame size since we
+ // compensate for key frames when adding delta
+ // frames.
+ frameSizeKbits -= _keyFrameSizeAvgKbits.Value();
+ }
+ else
+ {
+ // Shouldn't be negative, so zero is the lower bound.
+ frameSizeKbits = 0;
+ }
+ if (_keyFrameRatio.Value() > 1e-5 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
+ {
+ // We are sending key frames more often than our upper bound for
+ // how much we allow the key frame compensation to be spread
+ // out in time. Therefor we must use the key frame ratio rather
+ // than keyFrameSpreadFrames.
+ _keyFrameCount = static_cast<WebRtc_Word32>(1 / _keyFrameRatio.Value() + 0.5);
+ }
+ else
+ {
+ // Compensate for the key frame the following frames
+ _keyFrameCount = static_cast<WebRtc_Word32>(_keyFrameSpreadFrames + 0.5);
+ }
+ }
+ else
+ {
+ // Decrease the keyFrameRatio
+ _keyFrameRatio.Apply(1.0, 0.0);
+ }
+ // Change the level of the accumulator (bucket)
+ _accumulator += frameSizeKbits;
+}
+
+void
+VCMFrameDropper::Leak(WebRtc_UWord32 inputFrameRate)
+{
+ if (!_enabled)
+ {
+ return;
+ }
+ if (inputFrameRate < 1)
+ {
+ return;
+ }
+ if (_targetBitRate < 0.0f)
+ {
+ return;
+ }
+ _keyFrameSpreadFrames = 0.5f * inputFrameRate;
+ // T is the expected bits per frame (target). If all frames were the same size,
+ // we would get T bits per frame. Notice that T is also weighted to be able to
+ // force a lower frame rate if wanted.
+ float T = _targetBitRate / inputFrameRate;
+ if (_keyFrameCount > 0)
+ {
+ // Perform the key frame compensation
+ if (_keyFrameRatio.Value() > 0 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
+ {
+ T -= _keyFrameSizeAvgKbits.Value() * _keyFrameRatio.Value();
+ }
+ else
+ {
+ T -= _keyFrameSizeAvgKbits.Value() / _keyFrameSpreadFrames;
+ }
+ _keyFrameCount--;
+ }
+ _accumulator -= T;
+ UpdateRatio();
+
+}
+
+void
+VCMFrameDropper::UpdateNack(WebRtc_UWord32 nackBytes)
+{
+ if (!_enabled)
+ {
+ return;
+ }
+ _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
+}
+
+void
+VCMFrameDropper::FillBucket(float inKbits, float outKbits)
+{
+ _accumulator += (inKbits - outKbits);
+}
+
+void
+VCMFrameDropper::UpdateRatio()
+{
+ if (_accumulator > 1.3f * _accumulatorMax)
+ {
+ // Too far above accumulator max, react faster
+ _dropRatio.UpdateBase(0.8f);
+ }
+ else
+ {
+ // Go back to normal reaction
+ _dropRatio.UpdateBase(0.9f);
+ }
+ if (_accumulator > _accumulatorMax)
+ {
+ // We are above accumulator max, and should ideally
+ // drop a frame. Increase the dropRatio and drop
+ // the frame later.
+ if (_wasBelowMax)
+ {
+ _dropNext = true;
+ }
+ if (_fastMode)
+ {
+ // always drop in aggressive mode
+ _dropNext = true;
+ }
+
+ _dropRatio.Apply(1.0f, 1.0f);
+ _dropRatio.UpdateBase(0.9f);
+ }
+ else
+ {
+ _dropRatio.Apply(1.0f, 0.0f);
+ }
+ if (_accumulator < 0.0f)
+ {
+ _accumulator = 0.0f;
+ }
+ _wasBelowMax = _accumulator < _accumulatorMax;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId), "FrameDropper: dropRatio = %f accumulator = %f, accumulatorMax = %f", _dropRatio.Value(), _accumulator, _accumulatorMax);
+}
+
+// This function signals when to drop frames to the caller. It makes use of the dropRatio
+// to smooth out the drops over time.
+bool
+VCMFrameDropper::DropFrame()
+{
+ if (!_enabled)
+ {
+ return false;
+ }
+ if (_dropNext)
+ {
+ _dropNext = false;
+ _dropCount = 0;
+ }
+
+ if (_dropRatio.Value() >= 0.5f) // Drops per keep
+ {
+ // limit is the number of frames we should drop between each kept frame
+ // to keep our drop ratio. limit is positive in this case.
+ float denom = 1.0f - _dropRatio.Value();
+ if (denom < 1e-5)
+ {
+ denom = (float)1e-5;
+ }
+ WebRtc_Word32 limit = static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
+ if (_dropCount < 0)
+ {
+ // Reset the _dropCount since it was negative and should be positive.
+ if (_dropRatio.Value() > 0.4f)
+ {
+ _dropCount = -_dropCount;
+ }
+ else
+ {
+ _dropCount = 0;
+ }
+ }
+ if (_dropCount < limit)
+ {
+ // As long we are below the limit we should drop frames.
+ _dropCount++;
+ return true;
+ }
+ else
+ {
+ // Only when we reset _dropCount a frame should be kept.
+ _dropCount = 0;
+ return false;
+ }
+ }
+ else if (_dropRatio.Value() > 0.0f && _dropRatio.Value() < 0.5f) // Keeps per drop
+ {
+ // limit is the number of frames we should keep between each drop
+ // in order to keep the drop ratio. limit is negative in this case,
+ // and the _dropCount is also negative.
+ float denom = _dropRatio.Value();
+ if (denom < 1e-5)
+ {
+ denom = (float)1e-5;
+ }
+ WebRtc_Word32 limit = -static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
+ if (_dropCount > 0)
+ {
+ // Reset the _dropCount since we have a positive
+ // _dropCount, and it should be negative.
+ if (_dropRatio.Value() < 0.6f)
+ {
+ _dropCount = -_dropCount;
+ }
+ else
+ {
+ _dropCount = 0;
+ }
+ }
+ if (_dropCount > limit)
+ {
+ if (_dropCount == 0)
+ {
+ // Drop frames when we reset _dropCount.
+ _dropCount--;
+ return true;
+ }
+ else
+ {
+ // Keep frames as long as we haven't reached limit.
+ _dropCount--;
+ return false;
+ }
+ }
+ else
+ {
+ _dropCount = 0;
+ return false;
+ }
+ }
+ _dropCount = 0;
+ return false;
+
+ // A simpler version, unfiltered and quicker
+ //bool dropNext = _dropNext;
+ //_dropNext = false;
+ //return dropNext;
+}
+
+void
+VCMFrameDropper::SetRates(float bitRate, float userFrameRate)
+{
+ // Bit rate of -1 means infinite bandwidth.
+ _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
+ if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
+ {
+ // Rescale the accumulator level if the accumulator max decreases
+ _accumulator = bitRate / _targetBitRate * _accumulator;
+ }
+ _targetBitRate = bitRate;
+ if (userFrameRate > 0.0f)
+ {
+ _userFrameRate = userFrameRate;
+ }
+}
+
+float
+VCMFrameDropper::ActualFrameRate(WebRtc_UWord32 inputFrameRate) const
+{
+ if (!_enabled)
+ {
+ return static_cast<float>(inputFrameRate);
+ }
+ return inputFrameRate * (1.0f - _dropRatio.Value());
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/frame_dropper.h b/trunk/src/modules/video_coding/main/source/frame_dropper.h
new file mode 100644
index 0000000..5e7e8a1
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/frame_dropper.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
+
+#include "exp_filter.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+/******************************/
+/* VCMFrameDropper class */
+/****************************/
+// The Frame Dropper implements a variant of the leaky bucket algorithm
+// for keeping track of when to drop frames to avoid bit rate
+// over use when the encoder can't keep its bit rate.
+class VCMFrameDropper
+{
+public:
+ VCMFrameDropper(WebRtc_Word32 vcmId = 0);
+ // Resets the FrameDropper to its initial state.
+ // This means that the frameRateWeight is set to its
+ // default value as well.
+ void Reset();
+
+ void Enable(bool enable);
+ // Answers the question if it's time to drop a frame
+ // if we want to reach a given frame rate. Must be
+ // called for every frame.
+ //
+ // Return value : True if we should drop the current frame
+ bool DropFrame();
+ // Updates the FrameDropper with the size of the latest encoded
+ // frame. The FrameDropper calculates a new drop ratio (can be
+ // seen as the probability to drop a frame) and updates its
+ // internal statistics.
+ //
+ // Input:
+ // - frameSizeBytes : The size of the latest frame
+ // returned from the encoder.
+ // - deltaFrame : True if the encoder returned
+ // a key frame.
+ void Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame);
+
+ void Leak(WebRtc_UWord32 inputFrameRate);
+
+ void UpdateNack(WebRtc_UWord32 nackBytes);
+
+ // Sets the target bit rate and the frame rate produced by
+ // the camera.
+ //
+ // Input:
+ // - bitRate : The target bit rate
+ void SetRates(float bitRate, float userFrameRate);
+
+ // Return value : The current average frame rate produced
+ // if the DropFrame() function is used as
+ // instruction of when to drop frames.
+ float ActualFrameRate(WebRtc_UWord32 inputFrameRate) const;
+
+private:
+ void FillBucket(float inKbits, float outKbits);
+ void UpdateRatio();
+
+ WebRtc_Word32 _vcmId;
+ VCMExpFilter _keyFrameSizeAvgKbits;
+ VCMExpFilter _keyFrameRatio;
+ float _keyFrameSpreadFrames;
+ WebRtc_Word32 _keyFrameCount;
+ float _accumulator;
+ float _accumulatorMax;
+ float _targetBitRate;
+ bool _dropNext;
+ VCMExpFilter _dropRatio;
+ WebRtc_Word32 _dropCount;
+ float _windowSize;
+ float _userFrameRate;
+ bool _wasBelowMax;
+ bool _enabled;
+ bool _fastMode;
+}; // end of VCMFrameDropper class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
diff --git a/trunk/src/modules/video_coding/main/source/generic_decoder.cc b/trunk/src/modules/video_coding/main/source/generic_decoder.cc
new file mode 100644
index 0000000..217c21d
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/generic_decoder.cc
@@ -0,0 +1,227 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_coding.h"
+#include "trace.h"
+#include "generic_decoder.h"
+#include "internal_defines.h"
+#include "tick_time_base.h"
+
+namespace webrtc {
+
+VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
+ TickTimeBase* clock)
+:
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_clock(clock),
+_receiveCallback(NULL),
+_timing(timing),
+_timestampMap(kDecoderFrameMemoryLength),
+_lastReceivedPictureID(0)
+{
+}
+
+VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
+{
+ delete _critSect;
+}
+
+void VCMDecodedFrameCallback::SetUserReceiveCallback(
+ VCMReceiveCallback* receiveCallback)
+{
+ CriticalSectionScoped cs(_critSect);
+ _receiveCallback = receiveCallback;
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Decoded(RawImage& decodedImage)
+{
+ // TODO(holmer): We should improve this so that we can handle multiple
+ // callbacks from one call to Decode().
+ CriticalSectionScoped cs(_critSect);
+ VCMFrameInformation* frameInfo = static_cast<VCMFrameInformation*>(
+ _timestampMap.Pop(decodedImage._timeStamp));
+ if (frameInfo == NULL)
+ {
+ // The map should never be empty or full if this callback is called.
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+
+ _timing.StopDecodeTimer(
+ decodedImage._timeStamp,
+ frameInfo->decodeStartTimeMs,
+ _clock->MillisecondTimestamp());
+
+ if (_receiveCallback != NULL)
+ {
+ _frame.Swap(decodedImage._buffer,
+ decodedImage._length,
+ decodedImage._size);
+ _frame.SetWidth(decodedImage._width);
+ _frame.SetHeight(decodedImage._height);
+ _frame.SetTimeStamp(decodedImage._timeStamp);
+ _frame.SetRenderTime(frameInfo->renderTimeMs);
+ // Convert raw image to video frame
+ WebRtc_Word32 callbackReturn = _receiveCallback->FrameToRender(_frame);
+ if (callbackReturn < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug,
+ webrtc::kTraceVideoCoding,
+ -1,
+ "Render callback returned error: %d", callbackReturn);
+ }
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 pictureId)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (_receiveCallback != NULL)
+ {
+ return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
+ }
+ return -1;
+}
+
+WebRtc_Word32
+VCMDecodedFrameCallback::ReceivedDecodedFrame(const WebRtc_UWord64 pictureId)
+{
+ _lastReceivedPictureID = pictureId;
+ return 0;
+}
+
+WebRtc_UWord64 VCMDecodedFrameCallback::LastReceivedPictureID() const
+{
+ return _lastReceivedPictureID;
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo)
+{
+ CriticalSectionScoped cs(_critSect);
+ return _timestampMap.Add(timestamp, frameInfo);
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Pop(WebRtc_UWord32 timestamp)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (_timestampMap.Pop(timestamp) == NULL)
+ {
+ return VCM_GENERAL_ERROR;
+ }
+ return VCM_OK;
+}
+
+VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id, bool isExternal)
+:
+_id(id),
+_callback(NULL),
+_frameInfos(),
+_nextFrameInfoIdx(0),
+_decoder(decoder),
+_codecType(kVideoCodecUnknown),
+_isExternal(isExternal),
+_requireKeyFrame(false),
+_keyFrameDecoded(false)
+{
+}
+
+VCMGenericDecoder::~VCMGenericDecoder()
+{
+}
+
+WebRtc_Word32 VCMGenericDecoder::InitDecode(const VideoCodec* settings,
+ WebRtc_Word32 numberOfCores,
+ bool requireKeyFrame)
+{
+ _requireKeyFrame = requireKeyFrame;
+ _keyFrameDecoded = false;
+ _codecType = settings->codecType;
+
+ return _decoder.InitDecode(settings, numberOfCores);
+}
+
+WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
+ int64_t nowMs)
+{
+ if (_requireKeyFrame &&
+ !_keyFrameDecoded &&
+ frame.FrameType() != kVideoFrameKey &&
+ frame.FrameType() != kVideoFrameGolden)
+ {
+ // Require key frame is enabled, meaning that one key frame must be decoded
+ // before we can decode delta frames.
+ return VCM_CODEC_ERROR;
+ }
+ _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
+ _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
+ _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
+
+ WEBRTC_TRACE(webrtc::kTraceDebug,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Decoding timestamp %u", frame.TimeStamp());
+
+ _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
+
+ WebRtc_Word32 ret = _decoder.Decode(frame.EncodedImage(),
+ frame.MissingFrame(),
+ frame.FragmentationHeader(),
+ frame.CodecSpecific(),
+ frame.RenderTimeMs());
+
+ if (ret < WEBRTC_VIDEO_CODEC_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id), "Decoder error: %d\n", ret);
+ _callback->Pop(frame.TimeStamp());
+ return ret;
+ }
+ else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
+ ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
+ {
+ // No output
+ _callback->Pop(frame.TimeStamp());
+ }
+ // Update the key frame decoded variable so that we know whether or not we've decoded a key frame since reset.
+ _keyFrameDecoded = (frame.FrameType() == kVideoFrameKey || frame.FrameType() == kVideoFrameGolden);
+ return ret;
+}
+
+WebRtc_Word32
+VCMGenericDecoder::Release()
+{
+ _keyFrameDecoded = false;
+ return _decoder.Release();
+}
+
+WebRtc_Word32 VCMGenericDecoder::Reset()
+{
+ _keyFrameDecoded = false;
+ return _decoder.Reset();
+}
+
+WebRtc_Word32 VCMGenericDecoder::SetCodecConfigParameters(const WebRtc_UWord8* buffer, WebRtc_Word32 size)
+{
+ return _decoder.SetCodecConfigParameters(buffer, size);
+}
+
+WebRtc_Word32 VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
+{
+ _callback = callback;
+ return _decoder.RegisterDecodeCompleteCallback(callback);
+}
+
+bool VCMGenericDecoder::External() const
+{
+ return _isExternal;
+}
+
+} // namespace
diff --git a/trunk/src/modules/video_coding/main/source/generic_decoder.h b/trunk/src/modules/video_coding/main/source/generic_decoder.h
new file mode 100644
index 0000000..58dabc7
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/generic_decoder.h
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+
+#include "timing.h"
+#include "timestamp_map.h"
+#include "video_codec_interface.h"
+#include "encoded_frame.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+
+class VCMReceiveCallback;
+
+enum { kDecoderFrameMemoryLength = 10 };
+
+struct VCMFrameInformation
+{
+ WebRtc_Word64 renderTimeMs;
+ WebRtc_Word64 decodeStartTimeMs;
+ void* userData;
+};
+
+class VCMDecodedFrameCallback : public DecodedImageCallback
+{
+public:
+ VCMDecodedFrameCallback(VCMTiming& timing, TickTimeBase* clock);
+ virtual ~VCMDecodedFrameCallback();
+ void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
+
+ virtual WebRtc_Word32 Decoded(RawImage& decodedImage);
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+ virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
+
+ WebRtc_UWord64 LastReceivedPictureID() const;
+
+ WebRtc_Word32 Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo);
+ WebRtc_Word32 Pop(WebRtc_UWord32 timestamp);
+
+private:
+ CriticalSectionWrapper* _critSect;
+ TickTimeBase* _clock;
+ VideoFrame _frame;
+ VCMReceiveCallback* _receiveCallback;
+ VCMTiming& _timing;
+ VCMTimestampMap _timestampMap;
+ WebRtc_UWord64 _lastReceivedPictureID;
+};
+
+
+class VCMGenericDecoder
+{
+ friend class VCMCodecDataBase;
+public:
+ VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id = 0, bool isExternal = false);
+ ~VCMGenericDecoder();
+
+ /**
+ * Initialize the decoder with the information from the VideoCodec
+ */
+ WebRtc_Word32 InitDecode(const VideoCodec* settings,
+ WebRtc_Word32 numberOfCores,
+ bool requireKeyFrame);
+
+ /**
+ * Decode to a raw I420 frame,
+ *
+ * inputVideoBuffer reference to encoded video frame
+ */
+ WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
+
+ /**
+ * Free the decoder memory
+ */
+ WebRtc_Word32 Release();
+
+ /**
+ * Reset the decoder state, prepare for a new call
+ */
+ WebRtc_Word32 Reset();
+
+ /**
+ * Codec configuration data sent out-of-band, i.e. in SIP call setup
+ *
+ * buffer pointer to the configuration data
+ * size the size of the configuration data in bytes
+ */
+ WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/,
+ WebRtc_Word32 /*size*/);
+
+ WebRtc_Word32 RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
+
+ bool External() const;
+
+protected:
+
+ WebRtc_Word32 _id;
+ VCMDecodedFrameCallback* _callback;
+ VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength];
+ WebRtc_UWord32 _nextFrameInfoIdx;
+ VideoDecoder& _decoder;
+ VideoCodecType _codecType;
+ bool _isExternal;
+ bool _requireKeyFrame;
+ bool _keyFrameDecoded;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/trunk/src/modules/video_coding/main/source/generic_encoder.cc b/trunk/src/modules/video_coding/main/source/generic_encoder.cc
new file mode 100644
index 0000000..58cdd99
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/generic_encoder.cc
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "encoded_frame.h"
+#include "generic_encoder.h"
+#include "media_optimization.h"
+#include "../../../../engine_configurations.h"
+
+namespace webrtc {
+
+//#define DEBUG_ENCODER_BIT_STREAM
+
+VCMGenericEncoder::VCMGenericEncoder(VideoEncoder& encoder, bool internalSource /*= false*/)
+:
+_encoder(encoder),
+_codecType(kVideoCodecUnknown),
+_VCMencodedFrameCallback(NULL),
+_bitRate(0),
+_frameRate(0),
+_internalSource(false)
+{
+}
+
+
+VCMGenericEncoder::~VCMGenericEncoder()
+{
+}
+
+WebRtc_Word32 VCMGenericEncoder::Release()
+{
+ _bitRate = 0;
+ _frameRate = 0;
+ _VCMencodedFrameCallback = NULL;
+ return _encoder.Release();
+}
+
+WebRtc_Word32
+VCMGenericEncoder::InitEncode(const VideoCodec* settings,
+ WebRtc_Word32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize)
+{
+ _bitRate = settings->startBitrate;
+ _frameRate = settings->maxFramerate;
+ _codecType = settings->codecType;
+ if (_VCMencodedFrameCallback != NULL)
+ {
+ _VCMencodedFrameCallback->SetCodecType(_codecType);
+ }
+ return _encoder.InitEncode(settings, numberOfCores, maxPayloadSize);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
+ const CodecSpecificInfo* codecSpecificInfo,
+ FrameType* frameType)
+{
+ RawImage rawImage(inputFrame.Buffer(),
+ inputFrame.Length(),
+ inputFrame.Size());
+ rawImage._width = inputFrame.Width();
+ rawImage._height = inputFrame.Height();
+ rawImage._timeStamp = inputFrame.TimeStamp();
+
+ VideoFrameType videoFrameTypes[kMaxSimulcastStreams];
+ for (int i = 0; i < kMaxSimulcastStreams; i++)
+ {
+ videoFrameTypes[i] = VCMEncodedFrame::ConvertFrameType(frameType[i]);
+ }
+ return _encoder.Encode(rawImage, codecSpecificInfo, videoFrameTypes);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetChannelParameters(WebRtc_Word32 packetLoss, int rtt)
+{
+ return _encoder.SetChannelParameters(packetLoss, rtt);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate)
+{
+ WebRtc_Word32 ret = _encoder.SetRates(newBitRate, frameRate);
+ if (ret < 0)
+ {
+ return ret;
+ }
+ _bitRate = newBitRate;
+ _frameRate = frameRate;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMGenericEncoder::CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size)
+{
+ WebRtc_Word32 ret = _encoder.CodecConfigParameters(buffer, size);
+ if (ret < 0)
+ {
+ return ret;
+ }
+ return ret;
+}
+
+WebRtc_UWord32 VCMGenericEncoder::BitRate() const
+{
+ return _bitRate;
+}
+
+WebRtc_UWord32 VCMGenericEncoder::FrameRate() const
+{
+ return _frameRate;
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
+{
+ return _encoder.SetPeriodicKeyFrames(enable);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::RequestFrame(FrameType* frameTypes)
+{
+ RawImage image;
+ VideoFrameType videoFrameTypes[kMaxSimulcastStreams];
+ for (int i = 0; i < kMaxSimulcastStreams; i++)
+ {
+ videoFrameTypes[i] = VCMEncodedFrame::ConvertFrameType(frameTypes[i]);
+ }
+ return _encoder.Encode(image, NULL, videoFrameTypes);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback)
+{
+ _VCMencodedFrameCallback = VCMencodedFrameCallback;
+
+ _VCMencodedFrameCallback->SetCodecType(_codecType);
+ _VCMencodedFrameCallback->SetInternalSource(_internalSource);
+ return _encoder.RegisterEncodeCompleteCallback(_VCMencodedFrameCallback);
+}
+
+bool
+VCMGenericEncoder::InternalSource() const
+{
+ return _internalSource;
+}
+
+ /***************************
+ * Callback Implementation
+ ***************************/
+VCMEncodedFrameCallback::VCMEncodedFrameCallback():
+_sendCallback(),
+_mediaOpt(NULL),
+_encodedBytes(0),
+_payloadType(0),
+_bitStreamAfterEncoder(NULL)
+{
+#ifdef DEBUG_ENCODER_BIT_STREAM
+ _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
+#endif
+}
+
+VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
+{
+#ifdef DEBUG_ENCODER_BIT_STREAM
+ fclose(_bitStreamAfterEncoder);
+#endif
+}
+
+WebRtc_Word32
+VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
+{
+ _sendCallback = transport;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMEncodedFrameCallback::Encoded(
+ EncodedImage &encodedImage,
+ const CodecSpecificInfo* codecSpecificInfo,
+ const RTPFragmentationHeader* fragmentationHeader)
+{
+ FrameType frameType = VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
+
+ WebRtc_UWord32 encodedBytes = 0;
+ if (_sendCallback != NULL)
+ {
+ encodedBytes = encodedImage._length;
+
+#ifdef DEBUG_ENCODER_BIT_STREAM
+ if (_bitStreamAfterEncoder != NULL)
+ {
+ fwrite(encodedImage._buffer, 1, encodedImage._length, _bitStreamAfterEncoder);
+ }
+#endif
+
+ RTPVideoHeader rtpVideoHeader;
+ RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
+ if (codecSpecificInfo)
+ {
+ CopyCodecSpecific(*codecSpecificInfo, &rtpVideoHeaderPtr);
+ }
+ else
+ {
+ rtpVideoHeaderPtr = NULL;
+ }
+
+ WebRtc_Word32 callbackReturn = _sendCallback->SendData(
+ frameType,
+ _payloadType,
+ encodedImage._timeStamp,
+ encodedImage._buffer,
+ encodedBytes,
+ *fragmentationHeader,
+ rtpVideoHeaderPtr);
+ if (callbackReturn < 0)
+ {
+ return callbackReturn;
+ }
+ }
+ else
+ {
+ return VCM_UNINITIALIZED;
+ }
+ _encodedBytes = encodedBytes;
+ if (_mediaOpt != NULL) {
+ _mediaOpt->UpdateWithEncodedData(_encodedBytes, frameType);
+ if (_internalSource)
+ {
+ return _mediaOpt->DropFrame(); // Signal to encoder to drop next frame
+ }
+ }
+ return VCM_OK;
+}
+
+WebRtc_UWord32
+VCMEncodedFrameCallback::EncodedBytes()
+{
+ return _encodedBytes;
+}
+
+void
+VCMEncodedFrameCallback::SetMediaOpt(VCMMediaOptimization *mediaOpt)
+{
+ _mediaOpt = mediaOpt;
+}
+
+void VCMEncodedFrameCallback::CopyCodecSpecific(const CodecSpecificInfo& info,
+ RTPVideoHeader** rtp) {
+ switch (info.codecType) {
+ case kVideoCodecVP8: {
+ (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8();
+ (*rtp)->codecHeader.VP8.pictureId =
+ info.codecSpecific.VP8.pictureId;
+ (*rtp)->codecHeader.VP8.nonReference =
+ info.codecSpecific.VP8.nonReference;
+ (*rtp)->codecHeader.VP8.temporalIdx =
+ info.codecSpecific.VP8.temporalIdx;
+ (*rtp)->codecHeader.VP8.layerSync =
+ info.codecSpecific.VP8.layerSync;
+ (*rtp)->codecHeader.VP8.tl0PicIdx =
+ info.codecSpecific.VP8.tl0PicIdx;
+ (*rtp)->codecHeader.VP8.keyIdx =
+ info.codecSpecific.VP8.keyIdx;
+ (*rtp)->simulcastIdx = info.codecSpecific.VP8.simulcastIdx;
+ return;
+ }
+ default: {
+ // No codec specific info. Change RTP header pointer to NULL.
+ *rtp = NULL;
+ return;
+ }
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/generic_encoder.h b/trunk/src/modules/video_coding/main/source/generic_encoder.h
new file mode 100644
index 0000000..99c2ce9
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/generic_encoder.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+
+#include "video_codec_interface.h"
+
+#include <stdio.h>
+
+namespace webrtc
+{
+
+class VCMMediaOptimization;
+
+/*************************************/
+/* VCMEncodeFrameCallback class */
+/***********************************/
+class VCMEncodedFrameCallback : public EncodedImageCallback
+{
+public:
+ VCMEncodedFrameCallback();
+ virtual ~VCMEncodedFrameCallback();
+
+ /*
+ * Callback implementation - codec encode complete
+ */
+ WebRtc_Word32 Encoded(
+ EncodedImage& encodedImage,
+ const CodecSpecificInfo* codecSpecificInfo = NULL,
+ const RTPFragmentationHeader* fragmentationHeader = NULL);
+ /*
+ * Get number of encoded bytes
+ */
+ WebRtc_UWord32 EncodedBytes();
+ /*
+ * Callback implementation - generic encoder encode complete
+ */
+ WebRtc_Word32 SetTransportCallback(VCMPacketizationCallback* transport);
+ /**
+ * Set media Optimization
+ */
+ void SetMediaOpt (VCMMediaOptimization* mediaOpt);
+
+ void SetPayloadType(WebRtc_UWord8 payloadType) { _payloadType = payloadType; };
+ void SetCodecType(VideoCodecType codecType) {_codecType = codecType;};
+ void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
+
+private:
+ /*
+ * Map information from info into rtp. If no relevant information is found
+ * in info, rtp is set to NULL.
+ */
+ static void CopyCodecSpecific(const CodecSpecificInfo& info,
+ RTPVideoHeader** rtp);
+
+ VCMPacketizationCallback* _sendCallback;
+ VCMMediaOptimization* _mediaOpt;
+ WebRtc_UWord32 _encodedBytes;
+ WebRtc_UWord8 _payloadType;
+ VideoCodecType _codecType;
+ bool _internalSource;
+ FILE* _bitStreamAfterEncoder;
+};// end of VCMEncodeFrameCallback class
+
+
+/******************************/
+/* VCMGenericEncoder class */
+/******************************/
+class VCMGenericEncoder
+{
+ friend class VCMCodecDataBase;
+public:
+ VCMGenericEncoder(VideoEncoder& encoder, bool internalSource = false);
+ ~VCMGenericEncoder();
+ /**
+ * Free encoder memory
+ */
+ WebRtc_Word32 Release();
+ /**
+ * Initialize the encoder with the information from the VideoCodec
+ */
+ WebRtc_Word32 InitEncode(const VideoCodec* settings,
+ WebRtc_Word32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize);
+ /**
+ * Encode raw image
+ * inputFrame : Frame containing raw image
+ * codecSpecificInfo : Specific codec data
+ * cameraFrameRate : request or information from the remote side
+ * frameType : The requested frame type to encode
+ */
+ WebRtc_Word32 Encode(const VideoFrame& inputFrame,
+ const CodecSpecificInfo* codecSpecificInfo,
+ FrameType* frameType);
+ /**
+ * Set new target bit rate and frame rate
+ * Return Value: new bit rate if OK, otherwise <0s
+ */
+ WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate);
+ /**
+ * Set a new packet loss rate and a new round-trip time in milliseconds.
+ */
+ WebRtc_Word32 SetChannelParameters(WebRtc_Word32 packetLoss, int rtt);
+ WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size);
+ /**
+ * Register a transport callback which will be called to deliver the encoded buffers
+ */
+ WebRtc_Word32 RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback);
+ /**
+ * Get encoder bit rate
+ */
+ WebRtc_UWord32 BitRate() const;
+ /**
+ * Get encoder frame rate
+ */
+ WebRtc_UWord32 FrameRate() const;
+
+ WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
+
+ WebRtc_Word32 RequestFrame(FrameType* frameTypes);
+
+ bool InternalSource() const;
+
+private:
+ VideoEncoder& _encoder;
+ VideoCodecType _codecType;
+ VCMEncodedFrameCallback* _VCMencodedFrameCallback;
+ WebRtc_UWord32 _bitRate;
+ WebRtc_UWord32 _frameRate;
+ bool _internalSource;
+}; // end of VCMGenericEncoder class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc b/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc
new file mode 100644
index 0000000..3b520b3
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "inter_frame_delay.h"
+
+namespace webrtc {
+
+VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
+{
+ Reset(currentWallClock);
+}
+
+// Resets the delay estimate
+void
+VCMInterFrameDelay::Reset(int64_t currentWallClock)
+{
+ _zeroWallClock = currentWallClock;
+ _wrapArounds = 0;
+ _prevWallClock = 0;
+ _prevTimestamp = 0;
+ _dTS = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+bool
+VCMInterFrameDelay::CalculateDelay(WebRtc_UWord32 timestamp,
+ WebRtc_Word64 *delay,
+ int64_t currentWallClock)
+{
+ if (_prevWallClock == 0)
+ {
+ // First set of data, initialization, wait for next frame
+ _prevWallClock = currentWallClock;
+ _prevTimestamp = timestamp;
+ *delay = 0;
+ return true;
+ }
+
+ WebRtc_Word32 prevWrapArounds = _wrapArounds;
+ CheckForWrapArounds(timestamp);
+
+ // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
+ WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+ // Account for reordering in jitter variance estimate in the future?
+ // Note that this also captures incomplete frames which are grabbed
+ // for decoding after a later frame has been complete, i.e. real
+ // packet losses.
+ if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
+ {
+ *delay = 0;
+ return false;
+ }
+
+ // Compute the compensated timestamp difference and convert it to ms and
+ // round it to closest integer.
+ _dTS = static_cast<WebRtc_Word64>((timestamp + wrapAroundsSincePrev *
+ (static_cast<WebRtc_Word64>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
+
+ // frameDelay is the difference of dT and dTS -- i.e. the difference of
+ // the wall clock time difference and the timestamp difference between
+ // two following frames.
+ *delay = static_cast<WebRtc_Word64>(currentWallClock - _prevWallClock - _dTS);
+
+ _prevTimestamp = timestamp;
+ _prevWallClock = currentWallClock;
+
+ return true;
+}
+
+// Returns the current difference between incoming timestamps
+WebRtc_UWord32 VCMInterFrameDelay::CurrentTimeStampDiffMs() const
+{
+ if (_dTS < 0)
+ {
+ return 0;
+ }
+ return static_cast<WebRtc_UWord32>(_dTS);
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp and
+// keeps track of the number of wrap arounds since reset.
+void
+VCMInterFrameDelay::CheckForWrapArounds(WebRtc_UWord32 timestamp)
+{
+ if (timestamp < _prevTimestamp)
+ {
+ // This difference will probably be less than -2^31 if we have had a wrap around
+ // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
+ // it should be positive.
+ if (static_cast<WebRtc_Word32>(timestamp - _prevTimestamp) > 0)
+ {
+ // Forward wrap around
+ _wrapArounds++;
+ }
+ }
+ // This difference will probably be less than -2^31 if we have had a backward wrap around.
+ // Since it is cast to a Word32, it should be positive.
+ else if (static_cast<WebRtc_Word32>(_prevTimestamp - timestamp) > 0)
+ {
+ // Backward wrap around
+ _wrapArounds--;
+ }
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/inter_frame_delay.h b/trunk/src/modules/video_coding/main/source/inter_frame_delay.h
new file mode 100644
index 0000000..807c64b
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/inter_frame_delay.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class VCMInterFrameDelay
+{
+public:
+ VCMInterFrameDelay(int64_t currentWallClock);
+
+ // Resets the estimate. Zeros are given as parameters.
+ void Reset(int64_t currentWallClock);
+
+ // Calculates the delay of a frame with the given timestamp.
+ // This method is called when the frame is complete.
+ //
+ // Input:
+ // - timestamp : RTP timestamp of a received frame
+ // - *delay : Pointer to memory where the result should be stored
+ // - currentWallClock : The current time in milliseconds.
+ // Should be -1 for normal operation, only used for testing.
+ // Return value : true if OK, false when reordered timestamps
+ bool CalculateDelay(WebRtc_UWord32 timestamp,
+ WebRtc_Word64 *delay,
+ int64_t currentWallClock);
+
+ // Returns the current difference between incoming timestamps
+ //
+ // Return value : Wrap-around compensated difference between incoming
+ // timestamps.
+ WebRtc_UWord32 CurrentTimeStampDiffMs() const;
+
+private:
+ // Controls if the RTP timestamp counter has had a wrap around
+ // between the current and the previously received frame.
+ //
+ // Input:
+ // - timestmap : RTP timestamp of the current frame.
+ void CheckForWrapArounds(WebRtc_UWord32 timestamp);
+
+ WebRtc_Word64 _zeroWallClock; // Local timestamp of the first video packet received
+ WebRtc_Word32 _wrapArounds; // Number of wrapArounds detected
+ // The previous timestamp passed to the delay estimate
+ WebRtc_UWord32 _prevTimestamp;
+ // The previous wall clock timestamp used by the delay estimate
+ WebRtc_Word64 _prevWallClock;
+ // Wrap-around compensated difference between incoming timestamps
+ WebRtc_Word64 _dTS;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/trunk/src/modules/video_coding/main/source/internal_defines.h b/trunk/src/modules/video_coding/main/source/internal_defines.h
new file mode 100644
index 0000000..0ba6385
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/internal_defines.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
+
+inline WebRtc_UWord32 MaskWord64ToUWord32(WebRtc_Word64 w64)
+{
+ return static_cast<WebRtc_UWord32>(MASK_32_BITS(w64));
+}
+
+#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
+#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
+
+#define VCM_DEFAULT_CODEC_WIDTH 352
+#define VCM_DEFAULT_CODEC_HEIGHT 288
+#define VCM_DEFAULT_FRAME_RATE 30
+#define VCM_MIN_BITRATE 30
+#define VCM_FLUSH_INDICATOR 4
+
+// Helper macros for creating the static codec list
+#define VCM_NO_CODEC_IDX -1
+#ifdef VIDEOCODEC_VP8
+ #define VCM_VP8_IDX VCM_NO_CODEC_IDX + 1
+#else
+ #define VCM_VP8_IDX VCM_NO_CODEC_IDX
+#endif
+#ifdef VIDEOCODEC_I420
+ #define VCM_I420_IDX VCM_VP8_IDX + 1
+#else
+ #define VCM_I420_IDX VCM_VP8_IDX
+#endif
+#define VCM_NUM_VIDEO_CODECS_AVAILABLE VCM_I420_IDX + 1
+
+#define VCM_NO_RECEIVER_ID 0
+
+inline WebRtc_Word32 VCMId(const WebRtc_Word32 vcmId, const WebRtc_Word32 receiverId = 0)
+{
+ return static_cast<WebRtc_Word32>((vcmId << 16) + receiverId);
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer.cc
new file mode 100644
index 0000000..4b146b6
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_buffer.cc
@@ -0,0 +1,1775 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/main/source/jitter_buffer.h"
+
+#include <algorithm>
+#include <cassert>
+
+#include "modules/video_coding/main/source/event.h"
+#include "modules/video_coding/main/source/frame_buffer.h"
+#include "modules/video_coding/main/source/inter_frame_delay.h"
+#include "modules/video_coding/main/source/internal_defines.h"
+#include "modules/video_coding/main/source/jitter_buffer_common.h"
+#include "modules/video_coding/main/source/jitter_estimator.h"
+#include "modules/video_coding/main/source/packet.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+
+#if defined(_WIN32)
+ // VS 2005: Don't warn for default initialized arrays. See help for more info.
+ #pragma warning(disable:4351)
+#endif
+
+namespace webrtc {
+
+// Predicates used when searching for frames in the frame buffer list
+class FrameSmallerTimestamp {
+ public:
+ FrameSmallerTimestamp(uint32_t timestamp) : timestamp_(timestamp) {}
+ bool operator()(VCMFrameBuffer* frame) {
+ return (LatestTimestamp(timestamp_, frame->TimeStamp(), NULL) ==
+ timestamp_);
+ }
+
+ private:
+ uint32_t timestamp_;
+};
+
+class FrameEqualTimestamp {
+ public:
+ FrameEqualTimestamp(uint32_t timestamp) : timestamp_(timestamp) {}
+ bool operator()(VCMFrameBuffer* frame) {
+ return (timestamp_ == frame->TimeStamp());
+ }
+
+ private:
+ uint32_t timestamp_;
+};
+
+class CompleteDecodableKeyFrameCriteria {
+ public:
+ bool operator()(VCMFrameBuffer* frame) {
+ return (frame->FrameType() == kVideoFrameKey) &&
+ (frame->GetState() == kStateComplete ||
+ frame->GetState() == kStateDecodable);
+ }
+};
+
+// Constructor
+VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
+ WebRtc_Word32 vcmId,
+ WebRtc_Word32 receiverId,
+ bool master) :
+ _vcmId(vcmId),
+ _receiverId(receiverId),
+ _clock(clock),
+ _running(false),
+ _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _master(master),
+ _frameEvent(),
+ _packetEvent(),
+ _maxNumberOfFrames(kStartNumberOfFrames),
+ _frameBuffers(),
+ _frameList(),
+ _lastDecodedState(),
+ _packetsNotDecodable(0),
+ _receiveStatistics(),
+ _incomingFrameRate(0),
+ _incomingFrameCount(0),
+ _timeLastIncomingFrameCount(0),
+ _incomingBitCount(0),
+ _incomingBitRate(0),
+ _dropCount(0),
+ _numConsecutiveOldFrames(0),
+ _numConsecutiveOldPackets(0),
+ _discardedPackets(0),
+ _jitterEstimate(vcmId, receiverId),
+ _delayEstimate(_clock->MillisecondTimestamp()),
+ _rttMs(0),
+ _nackMode(kNoNack),
+ _lowRttNackThresholdMs(-1),
+ _highRttNackThresholdMs(-1),
+ _NACKSeqNum(),
+ _NACKSeqNumLength(0),
+ _waitingForKeyFrame(false),
+ _firstPacket(true)
+{
+ memset(_frameBuffers, 0, sizeof(_frameBuffers));
+ memset(_receiveStatistics, 0, sizeof(_receiveStatistics));
+ memset(_NACKSeqNumInternal, -1, sizeof(_NACKSeqNumInternal));
+
+ for (int i = 0; i< kStartNumberOfFrames; i++)
+ {
+ _frameBuffers[i] = new VCMFrameBuffer();
+ }
+}
+
+// Destructor
+VCMJitterBuffer::~VCMJitterBuffer()
+{
+ Stop();
+ for (int i = 0; i< kMaxNumberOfFrames; i++)
+ {
+ if (_frameBuffers[i])
+ {
+ delete _frameBuffers[i];
+ }
+ }
+ delete _critSect;
+}
+
+void
+VCMJitterBuffer::CopyFrom(const VCMJitterBuffer& rhs)
+{
+ if (this != &rhs)
+ {
+ _critSect->Enter();
+ rhs._critSect->Enter();
+ _vcmId = rhs._vcmId;
+ _receiverId = rhs._receiverId;
+ _running = rhs._running;
+ _master = !rhs._master;
+ _maxNumberOfFrames = rhs._maxNumberOfFrames;
+ _incomingFrameRate = rhs._incomingFrameRate;
+ _incomingFrameCount = rhs._incomingFrameCount;
+ _timeLastIncomingFrameCount = rhs._timeLastIncomingFrameCount;
+ _incomingBitCount = rhs._incomingBitCount;
+ _incomingBitRate = rhs._incomingBitRate;
+ _dropCount = rhs._dropCount;
+ _numConsecutiveOldFrames = rhs._numConsecutiveOldFrames;
+ _numConsecutiveOldPackets = rhs._numConsecutiveOldPackets;
+ _discardedPackets = rhs._discardedPackets;
+ _jitterEstimate = rhs._jitterEstimate;
+ _delayEstimate = rhs._delayEstimate;
+ _waitingForCompletion = rhs._waitingForCompletion;
+ _rttMs = rhs._rttMs;
+ _NACKSeqNumLength = rhs._NACKSeqNumLength;
+ _waitingForKeyFrame = rhs._waitingForKeyFrame;
+ _firstPacket = rhs._firstPacket;
+ _lastDecodedState = rhs._lastDecodedState;
+ _packetsNotDecodable = rhs._packetsNotDecodable;
+ memcpy(_receiveStatistics, rhs._receiveStatistics,
+ sizeof(_receiveStatistics));
+ memcpy(_NACKSeqNumInternal, rhs._NACKSeqNumInternal,
+ sizeof(_NACKSeqNumInternal));
+ memcpy(_NACKSeqNum, rhs._NACKSeqNum, sizeof(_NACKSeqNum));
+ for (int i = 0; i < kMaxNumberOfFrames; i++)
+ {
+ if (_frameBuffers[i] != NULL)
+ {
+ delete _frameBuffers[i];
+ _frameBuffers[i] = NULL;
+ }
+ }
+ _frameList.clear();
+ for (int i = 0; i < _maxNumberOfFrames; i++)
+ {
+ _frameBuffers[i] = new VCMFrameBuffer(*(rhs._frameBuffers[i]));
+ if (_frameBuffers[i]->Length() > 0)
+ {
+ FrameList::reverse_iterator rit = std::find_if(
+ _frameList.rbegin(), _frameList.rend(),
+ FrameSmallerTimestamp(_frameBuffers[i]->TimeStamp()));
+ _frameList.insert(rit.base(), _frameBuffers[i]);
+ }
+ }
+ rhs._critSect->Leave();
+ _critSect->Leave();
+ }
+}
+
+// Start jitter buffer
+void
+VCMJitterBuffer::Start()
+{
+ CriticalSectionScoped cs(_critSect);
+ _running = true;
+ _incomingFrameCount = 0;
+ _incomingFrameRate = 0;
+ _incomingBitCount = 0;
+ _incomingBitRate = 0;
+ _timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
+ memset(_receiveStatistics, 0, sizeof(_receiveStatistics));
+
+ _numConsecutiveOldFrames = 0;
+ _numConsecutiveOldPackets = 0;
+ _discardedPackets = 0;
+
+ _frameEvent.Reset(); // start in a non-signaled state
+ _packetEvent.Reset(); // start in a non-signaled state
+ _waitingForCompletion.frameSize = 0;
+ _waitingForCompletion.timestamp = 0;
+ _waitingForCompletion.latestPacketTime = -1;
+ _firstPacket = true;
+ _NACKSeqNumLength = 0;
+ _waitingForKeyFrame = false;
+ _rttMs = 0;
+ _packetsNotDecodable = 0;
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
+ _receiverId), "JB(0x%x): Jitter buffer: start", this);
+}
+
+
+// Stop jitter buffer
+void
+VCMJitterBuffer::Stop()
+{
+ _critSect->Enter();
+ _running = false;
+ _lastDecodedState.Reset();
+ _frameList.clear();
+ for (int i = 0; i < kMaxNumberOfFrames; i++)
+ {
+ if (_frameBuffers[i] != NULL)
+ {
+ static_cast<VCMFrameBuffer*>(_frameBuffers[i])->SetState(kStateFree);
+ }
+ }
+
+ _critSect->Leave();
+ _frameEvent.Set(); // Make sure we exit from trying to get a frame to decoder
+ _packetEvent.Set(); // Make sure we exit from trying to get a sequence number
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
+ _receiverId), "JB(0x%x): Jitter buffer: stop", this);
+}
+
+bool
+VCMJitterBuffer::Running() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return _running;
+}
+
+// Flush jitter buffer
+void
+VCMJitterBuffer::Flush()
+{
+ CriticalSectionScoped cs(_critSect);
+ FlushInternal();
+}
+
+// Must be called under the critical section _critSect
+void
+VCMJitterBuffer::FlushInternal()
+{
+ // Erase all frames from the sorted list and set their state to free.
+ _frameList.clear();
+ for (WebRtc_Word32 i = 0; i < _maxNumberOfFrames; i++)
+ {
+ ReleaseFrameInternal(_frameBuffers[i]);
+ }
+ _lastDecodedState.Reset(); // TODO (mikhal): sync reset
+ _packetsNotDecodable = 0;
+
+ _frameEvent.Reset();
+ _packetEvent.Reset();
+
+ _numConsecutiveOldFrames = 0;
+ _numConsecutiveOldPackets = 0;
+
+ // Also reset the jitter and delay estimates
+ _jitterEstimate.Reset();
+ _delayEstimate.Reset(_clock->MillisecondTimestamp());
+
+ _waitingForCompletion.frameSize = 0;
+ _waitingForCompletion.timestamp = 0;
+ _waitingForCompletion.latestPacketTime = -1;
+
+ _firstPacket = true;
+
+ _NACKSeqNumLength = 0;
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
+ _receiverId), "JB(0x%x): Jitter buffer: flush", this);
+}
+
+// Set the frame state to free and remove it from the sorted
+// frame list. Must be called from inside the critical section _critSect.
+void
+VCMJitterBuffer::ReleaseFrameInternal(VCMFrameBuffer* frame)
+{
+ if (frame != NULL && frame->GetState() != kStateDecoding)
+ {
+ frame->SetState(kStateFree);
+ }
+}
+
+// Update frame state (set as complete if conditions are met)
+// Doing it here increases the degree of freedom for e.g. future
+// reconstructability of separate layers. Must be called under the
+// critical section _critSect.
+VCMFrameBufferEnum
+VCMJitterBuffer::UpdateFrameState(VCMFrameBuffer* frame)
+{
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId), "JB(0x%x) FB(0x%x): "
+ "UpdateFrameState NULL frame pointer", this, frame);
+ return kNoError;
+ }
+
+ int length = frame->Length();
+ if (_master)
+ {
+ // Only trace the primary jitter buffer to make it possible to parse
+ // and plot the trace file.
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "JB(0x%x) FB(0x%x): Complete frame added to jitter buffer,"
+ " size:%d type %d",
+ this, frame,length,frame->FrameType());
+ }
+
+ if (length != 0 && !frame->GetCountedFrame())
+ {
+ // ignore Ack frames
+ _incomingFrameCount++;
+ frame->SetCountedFrame(true);
+ }
+
+ // Check if we should drop frame
+ // an old complete frame can arrive too late
+ if (_lastDecodedState.IsOldFrame(frame))
+ {
+ // Frame is older than the latest decoded frame, drop it. Will be
+ // released by CleanUpOldFrames later.
+ frame->Reset();
+ frame->SetState(kStateEmpty);
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "JB(0x%x) FB(0x%x): Dropping old frame in Jitter buffer",
+ this, frame);
+ _dropCount++;
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Jitter buffer drop count: %d, consecutive drops: %u",
+ _dropCount, _numConsecutiveOldFrames);
+ // Flush() if this happens consistently.
+ _numConsecutiveOldFrames++;
+ if (_numConsecutiveOldFrames > kMaxConsecutiveOldFrames) {
+ FlushInternal();
+ return kFlushIndicator;
+ }
+ return kNoError;
+ }
+ _numConsecutiveOldFrames = 0;
+ frame->SetState(kStateComplete);
+
+
+ // Update receive statistics. We count all layers, thus when you use layers
+ // adding all key and delta frames might differ from frame count
+ if (frame->IsSessionComplete())
+ {
+ switch (frame->FrameType())
+ {
+ case kVideoFrameKey:
+ {
+ _receiveStatistics[0]++;
+ break;
+ }
+ case kVideoFrameDelta:
+ {
+ _receiveStatistics[1]++;
+ break;
+ }
+ case kVideoFrameGolden:
+ {
+ _receiveStatistics[2]++;
+ break;
+ }
+ case kVideoFrameAltRef:
+ {
+ _receiveStatistics[3]++;
+ break;
+ }
+ default:
+ assert(false);
+
+ }
+ }
+ const FrameList::iterator it = FindOldestCompleteContinuousFrame(false);
+ VCMFrameBuffer* oldFrame = NULL;
+ if (it != _frameList.end())
+ {
+ oldFrame = *it;
+ }
+
+ // Only signal if this is the oldest frame.
+ // Not necessary the case due to packet reordering or NACK.
+ if (!WaitForNack() || (oldFrame != NULL && oldFrame == frame))
+ {
+ _frameEvent.Set();
+ }
+ return kNoError;
+}
+
+// Get received key and delta frames
+WebRtc_Word32
+VCMJitterBuffer::GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
+ WebRtc_UWord32& receivedKeyFrames) const
+{
+ {
+ CriticalSectionScoped cs(_critSect);
+ receivedDeltaFrames = _receiveStatistics[1] + _receiveStatistics[3];
+ receivedKeyFrames = _receiveStatistics[0] + _receiveStatistics[2];
+ }
+ return 0;
+}
+
+WebRtc_UWord32 VCMJitterBuffer::NumNotDecodablePackets() const {
+ CriticalSectionScoped cs(_critSect);
+ return _packetsNotDecodable;
+}
+
+WebRtc_UWord32 VCMJitterBuffer::DiscardedPackets() const {
+ CriticalSectionScoped cs(_critSect);
+ return _discardedPackets;
+}
+
+// Gets frame to use for this timestamp. If no match, get empty frame.
+WebRtc_Word32
+VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMEncodedFrame*& frame)
+{
+ if (!_running) // don't accept incoming packets until we are started
+ {
+ return VCM_UNINITIALIZED;
+ }
+
+ _critSect->Enter();
+ // Does this packet belong to an old frame?
+ if (_lastDecodedState.IsOldPacket(&packet))
+ {
+ // Account only for media packets
+ if (packet.sizeBytes > 0)
+ {
+ _discardedPackets++;
+ _numConsecutiveOldPackets++;
+ }
+ // Update last decoded sequence number if the packet arrived late and
+ // belongs to a frame with a timestamp equal to the last decoded
+ // timestamp.
+ _lastDecodedState.UpdateOldPacket(&packet);
+
+ if (_numConsecutiveOldPackets > kMaxConsecutiveOldPackets)
+ {
+ FlushInternal();
+ _critSect->Leave();
+ return VCM_FLUSH_INDICATOR;
+ }
+ _critSect->Leave();
+ return VCM_OLD_PACKET_ERROR;
+ }
+ _numConsecutiveOldPackets = 0;
+
+ FrameList::iterator it = std::find_if(
+ _frameList.begin(),
+ _frameList.end(),
+ FrameEqualTimestamp(packet.timestamp));
+
+ if (it != _frameList.end()) {
+ frame = *it;
+ _critSect->Leave();
+ return VCM_OK;
+ }
+
+ _critSect->Leave();
+
+ // No match, return empty frame
+ frame = GetEmptyFrame();
+ if (frame != NULL)
+ {
+ return VCM_OK;
+ }
+ // No free frame! Try to reclaim some...
+ _critSect->Enter();
+ RecycleFramesUntilKeyFrame();
+ _critSect->Leave();
+
+ frame = GetEmptyFrame();
+ if (frame != NULL)
+ {
+ return VCM_OK;
+ }
+ return VCM_JITTER_BUFFER_ERROR;
+}
+
+// Deprecated! Kept for testing purposes.
+VCMEncodedFrame*
+VCMJitterBuffer::GetFrame(const VCMPacket& packet)
+{
+ VCMEncodedFrame* frame = NULL;
+ if (GetFrame(packet, frame) < 0)
+ {
+ return NULL;
+ }
+ return frame;
+}
+
+// Get empty frame, creates new (i.e. increases JB size) if necessary
+VCMFrameBuffer*
+VCMJitterBuffer::GetEmptyFrame()
+{
+ if (!_running) // don't accept incoming packets until we are started
+ {
+ return NULL;
+ }
+
+ _critSect->Enter();
+
+ for (int i = 0; i <_maxNumberOfFrames; ++i)
+ {
+ if (kStateFree == _frameBuffers[i]->GetState())
+ {
+ // found a free buffer
+ _frameBuffers[i]->SetState(kStateEmpty);
+ _critSect->Leave();
+ return _frameBuffers[i];
+ }
+ }
+
+ // Check if we can increase JB size
+ if (_maxNumberOfFrames < kMaxNumberOfFrames)
+ {
+ VCMFrameBuffer* ptrNewBuffer = new VCMFrameBuffer();
+ ptrNewBuffer->SetState(kStateEmpty);
+ _frameBuffers[_maxNumberOfFrames] = ptrNewBuffer;
+ _maxNumberOfFrames++;
+
+ _critSect->Leave();
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId), "JB(0x%x) FB(0x%x): Jitter buffer "
+ "increased to:%d frames", this, ptrNewBuffer, _maxNumberOfFrames);
+ return ptrNewBuffer;
+ }
+ _critSect->Leave();
+
+ // We have reached max size, cannot increase JB size
+ return NULL;
+}
+
+
+// Find oldest complete frame used for getting next frame to decode
+// Must be called under critical section
+FrameList::iterator
+VCMJitterBuffer::FindOldestCompleteContinuousFrame(bool enable_decodable) {
+ // If we have more than one frame done since last time, pick oldest.
+ VCMFrameBuffer* oldest_frame = NULL;
+ FrameList::iterator it = _frameList.begin();
+
+ // When temporal layers are available, we search for a complete or decodable
+ // frame until we hit one of the following:
+ // 1. Continuous base or sync layer.
+ // 2. The end of the list was reached.
+ for (; it != _frameList.end(); ++it) {
+ oldest_frame = *it;
+ VCMFrameBufferStateEnum state = oldest_frame->GetState();
+ // Is this frame complete or decodable and continuous?
+ if ((state == kStateComplete ||
+ (enable_decodable && state == kStateDecodable)) &&
+ _lastDecodedState.ContinuousFrame(oldest_frame)) {
+ break;
+ } else {
+ int temporal_id = oldest_frame->TemporalId();
+ oldest_frame = NULL;
+ if (temporal_id <= 0) {
+ // When temporal layers are disabled or we have hit a base layer
+ // we break (regardless of continuity and completeness).
+ break;
+ }
+ }
+ }
+
+ if (oldest_frame == NULL) {
+ // No complete frame no point to continue.
+ return _frameList.end();
+ } else if (_waitingForKeyFrame &&
+ oldest_frame->FrameType() != kVideoFrameKey) {
+ // We are waiting for a key frame.
+ return _frameList.end();
+ }
+
+ // We have a complete continuous frame.
+ return it;
+}
+
+// Call from inside the critical section _critSect
+void
+VCMJitterBuffer::RecycleFrame(VCMFrameBuffer* frame)
+{
+ if (frame == NULL)
+ {
+ return;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "JB(0x%x) FB(0x%x): RecycleFrame, size:%d",
+ this, frame, frame->Length());
+
+ ReleaseFrameInternal(frame);
+}
+
+// Calculate frame and bit rates
+WebRtc_Word32
+VCMJitterBuffer::GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate)
+{
+ CriticalSectionScoped cs(_critSect);
+ const WebRtc_Word64 now = _clock->MillisecondTimestamp();
+ WebRtc_Word64 diff = now - _timeLastIncomingFrameCount;
+ if (diff < 1000 && _incomingFrameRate > 0 && _incomingBitRate > 0)
+ {
+ // Make sure we report something even though less than
+ // 1 second has passed since last update.
+ frameRate = _incomingFrameRate;
+ bitRate = _incomingBitRate;
+ }
+ else if (_incomingFrameCount != 0)
+ {
+ // We have received frame(s) since last call to this function
+
+ // Prepare calculations
+ if (diff <= 0)
+ {
+ diff = 1;
+ }
+ // we add 0.5f for rounding
+ float rate = 0.5f + ((_incomingFrameCount * 1000.0f) / diff);
+ if (rate < 1.0f) // don't go below 1, can crash
+ {
+ rate = 1.0f;
+ }
+
+ // Calculate frame rate
+ // Let r be rate.
+ // r(0) = 1000*framecount/delta_time.
+ // (I.e. frames per second since last calculation.)
+ // frameRate = r(0)/2 + r(-1)/2
+ // (I.e. fr/s average this and the previous calculation.)
+ frameRate = (_incomingFrameRate + (WebRtc_Word32)rate) >> 1;
+ _incomingFrameRate = (WebRtc_UWord8)rate;
+
+ // Calculate bit rate
+ if (_incomingBitCount == 0)
+ {
+ bitRate = 0;
+ }
+ else
+ {
+ bitRate = 10 * ((100 * _incomingBitCount) /
+ static_cast<WebRtc_UWord32>(diff));
+ }
+ _incomingBitRate = bitRate;
+
+ // Reset count
+ _incomingFrameCount = 0;
+ _incomingBitCount = 0;
+ _timeLastIncomingFrameCount = now;
+
+ }
+ else
+ {
+ // No frames since last call
+ _timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
+ frameRate = 0;
+ bitRate = 0;
+ _incomingBitRate = 0;
+ }
+
+ return 0;
+}
+
+// Returns immediately or a X ms event hang waiting for a complete frame,
+// X decided by caller
+VCMEncodedFrame*
+VCMJitterBuffer::GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS)
+{
+ if (!_running)
+ {
+ return NULL;
+ }
+
+ _critSect->Enter();
+
+ CleanUpOldFrames();
+
+ if (_lastDecodedState.init() && WaitForNack()) {
+ _waitingForKeyFrame = true;
+ }
+
+ FrameList::iterator it = FindOldestCompleteContinuousFrame(false);
+ if (it == _frameList.end())
+ {
+ if (maxWaitTimeMS == 0)
+ {
+ _critSect->Leave();
+ return NULL;
+ }
+ const WebRtc_Word64 endWaitTimeMs = _clock->MillisecondTimestamp()
+ + maxWaitTimeMS;
+ WebRtc_Word64 waitTimeMs = maxWaitTimeMS;
+ while (waitTimeMs > 0)
+ {
+ _critSect->Leave();
+ const EventTypeWrapper ret =
+ _frameEvent.Wait(static_cast<WebRtc_UWord32>(waitTimeMs));
+ _critSect->Enter();
+ if (ret == kEventSignaled)
+ {
+ // are we closing down the Jitter buffer
+ if (!_running)
+ {
+ _critSect->Leave();
+ return NULL;
+ }
+
+ // Finding oldest frame ready for decoder, but check
+ // sequence number and size
+ CleanUpOldFrames();
+ it = FindOldestCompleteContinuousFrame(false);
+ if (it == _frameList.end())
+ {
+ waitTimeMs = endWaitTimeMs -
+ _clock->MillisecondTimestamp();
+ }
+ else
+ {
+ break;
+ }
+ }
+ else
+ {
+ _critSect->Leave();
+ return NULL;
+ }
+ }
+ // Inside critSect
+ }
+ else
+ {
+ // we already have a frame reset the event
+ _frameEvent.Reset();
+ }
+
+ if (it == _frameList.end())
+ {
+ // Even after signaling we're still missing a complete continuous frame
+ _critSect->Leave();
+ return NULL;
+ }
+
+ VCMFrameBuffer* oldestFrame = *it;
+ it = _frameList.erase(it);
+
+ // Update jitter estimate
+ const bool retransmitted = (oldestFrame->GetNackCount() > 0);
+ if (retransmitted)
+ {
+ _jitterEstimate.FrameNacked();
+ }
+ else if (oldestFrame->Length() > 0)
+ {
+ // Ignore retransmitted and empty frames.
+ UpdateJitterAndDelayEstimates(*oldestFrame, false);
+ }
+
+ oldestFrame->SetState(kStateDecoding);
+
+ CleanUpOldFrames();
+
+ if (oldestFrame->FrameType() == kVideoFrameKey)
+ {
+ _waitingForKeyFrame = false;
+ }
+
+ _critSect->Leave();
+
+ // We have a frame - update decoded state with frame info.
+ _lastDecodedState.SetState(oldestFrame);
+
+ return oldestFrame;
+}
+
+WebRtc_UWord32
+VCMJitterBuffer::GetEstimatedJitterMS()
+{
+ CriticalSectionScoped cs(_critSect);
+ return GetEstimatedJitterMsInternal();
+}
+
+WebRtc_UWord32
+VCMJitterBuffer::GetEstimatedJitterMsInternal()
+{
+ WebRtc_UWord32 estimate = VCMJitterEstimator::OPERATING_SYSTEM_JITTER;
+
+ // Compute RTT multiplier for estimation
+ // _lowRttNackThresholdMs == -1 means no FEC.
+ double rttMult = 1.0f;
+ if (_nackMode == kNackHybrid && (_lowRttNackThresholdMs >= 0 &&
+ static_cast<int>(_rttMs) > _lowRttNackThresholdMs))
+ {
+ // from here we count on FEC
+ rttMult = 0.0f;
+ }
+ estimate += static_cast<WebRtc_UWord32>
+ (_jitterEstimate.GetJitterEstimate(rttMult) + 0.5);
+ return estimate;
+}
+
+void
+VCMJitterBuffer::UpdateRtt(WebRtc_UWord32 rttMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ _rttMs = rttMs;
+ _jitterEstimate.UpdateRtt(rttMs);
+}
+
+// wait for the first packet in the next frame to arrive
+WebRtc_Word64
+VCMJitterBuffer::GetNextTimeStamp(WebRtc_UWord32 maxWaitTimeMS,
+ FrameType& incomingFrameType,
+ WebRtc_Word64& renderTimeMs)
+{
+ if (!_running)
+ {
+ return -1;
+ }
+
+ _critSect->Enter();
+
+ // Finding oldest frame ready for decoder, check sequence number and size
+ CleanUpOldFrames();
+
+ FrameList::iterator it = _frameList.begin();
+
+ if (it == _frameList.end())
+ {
+ _packetEvent.Reset();
+ _critSect->Leave();
+
+ if (_packetEvent.Wait(maxWaitTimeMS) == kEventSignaled)
+ {
+ // are we closing down the Jitter buffer
+ if (!_running)
+ {
+ return -1;
+ }
+ _critSect->Enter();
+
+ CleanUpOldFrames();
+ it = _frameList.begin();
+ }
+ else
+ {
+ _critSect->Enter();
+ }
+ }
+
+ if (it == _frameList.end())
+ {
+ _critSect->Leave();
+ return -1;
+ }
+ // we have a frame
+
+ // return frame type
+ // All layers are assumed to have the same type
+ incomingFrameType = (*it)->FrameType();
+
+ renderTimeMs = (*it)->RenderTimeMs();
+
+ const WebRtc_UWord32 timestamp = (*it)->TimeStamp();
+
+ _critSect->Leave();
+
+ // return current time
+ return timestamp;
+}
+
+// Answers the question:
+// Will the packet sequence be complete if the next frame is grabbed for
+// decoding right now? That is, have we lost a frame between the last decoded
+// frame and the next, or is the next
+// frame missing one or more packets?
+bool
+VCMJitterBuffer::CompleteSequenceWithNextFrame()
+{
+ CriticalSectionScoped cs(_critSect);
+ // Finding oldest frame ready for decoder, check sequence number and size
+ CleanUpOldFrames();
+
+ if (_frameList.empty())
+ return true;
+
+ VCMFrameBuffer* oldestFrame = _frameList.front();
+ if (_frameList.size() <= 1 &&
+ oldestFrame->GetState() != kStateComplete)
+ {
+ // Frame not ready to be decoded.
+ return true;
+ }
+ if (!oldestFrame->Complete())
+ {
+ return false;
+ }
+
+ // See if we have lost a frame before this one.
+ if (_lastDecodedState.init())
+ {
+ // Following start, reset or flush -> check for key frame.
+ if (oldestFrame->FrameType() != kVideoFrameKey)
+ {
+ return false;
+ }
+ }
+ else if (oldestFrame->GetLowSeqNum() == -1)
+ {
+ return false;
+ }
+ else if (!_lastDecodedState.ContinuousFrame(oldestFrame))
+ {
+ return false;
+ }
+ return true;
+}
+
+// Returns immediately
+VCMEncodedFrame*
+VCMJitterBuffer::GetFrameForDecoding()
+{
+ CriticalSectionScoped cs(_critSect);
+ if (!_running)
+ {
+ return NULL;
+ }
+
+ if (WaitForNack())
+ {
+ return GetFrameForDecodingNACK();
+ }
+
+ CleanUpOldFrames();
+
+ if (_frameList.empty()) {
+ return NULL;
+ }
+
+ VCMFrameBuffer* oldestFrame = _frameList.front();
+ if (_frameList.size() <= 1 &&
+ oldestFrame->GetState() != kStateComplete) {
+ return NULL;
+ }
+
+ // Incomplete frame pulled out from jitter buffer,
+ // update the jitter estimate with what we currently know.
+ // This frame shouldn't have been retransmitted, but if we recently
+ // turned off NACK this might still happen.
+ const bool retransmitted = (oldestFrame->GetNackCount() > 0);
+ if (retransmitted)
+ {
+ _jitterEstimate.FrameNacked();
+ }
+ else if (oldestFrame->Length() > 0)
+ {
+ // Ignore retransmitted and empty frames.
+ // Update with the previous incomplete frame first
+ if (_waitingForCompletion.latestPacketTime >= 0)
+ {
+ UpdateJitterAndDelayEstimates(_waitingForCompletion, true);
+ }
+ // Then wait for this one to get complete
+ _waitingForCompletion.frameSize = oldestFrame->Length();
+ _waitingForCompletion.latestPacketTime =
+ oldestFrame->LatestPacketTimeMs();
+ _waitingForCompletion.timestamp = oldestFrame->TimeStamp();
+ }
+ _frameList.erase(_frameList.begin());
+
+ // Look for previous frame loss
+ VerifyAndSetPreviousFrameLost(*oldestFrame);
+
+ // The state must be changed to decoding before cleaning up zero sized
+ // frames to avoid empty frames being cleaned up and then given to the
+ // decoder.
+ // Set as decoding. Propagates the missingFrame bit.
+ oldestFrame->SetState(kStateDecoding);
+
+ CleanUpOldFrames();
+
+ if (oldestFrame->FrameType() == kVideoFrameKey)
+ {
+ _waitingForKeyFrame = false;
+ }
+
+ _packetsNotDecodable += oldestFrame->NotDecodablePackets();
+
+ // We have a frame - update decoded state with frame info.
+ _lastDecodedState.SetState(oldestFrame);
+
+ return oldestFrame;
+}
+
+VCMEncodedFrame*
+VCMJitterBuffer::GetFrameForDecodingNACK()
+{
+ // when we use NACK we don't release non complete frames
+ // unless we have a complete key frame.
+ // In hybrid mode, we may release decodable frames (non-complete)
+
+ // Clean up old frames and empty frames
+ CleanUpOldFrames();
+
+ // First look for a complete _continuous_ frame.
+ // When waiting for nack, wait for a key frame, if a continuous frame cannot
+ // be determined (i.e. initial decoding state).
+ if (_lastDecodedState.init()) {
+ _waitingForKeyFrame = true;
+ }
+
+ // Allow for a decodable frame when in Hybrid mode.
+ bool enableDecodable = _nackMode == kNackHybrid ? true : false;
+ FrameList::iterator it = FindOldestCompleteContinuousFrame(enableDecodable);
+ if (it == _frameList.end())
+ {
+ // If we didn't find one we're good with a complete key/decodable frame.
+ it = find_if(_frameList.begin(), _frameList.end(),
+ CompleteDecodableKeyFrameCriteria());
+ if (it == _frameList.end())
+ {
+ return NULL;
+ }
+ }
+ VCMFrameBuffer* oldestFrame = *it;
+ // Update jitter estimate
+ const bool retransmitted = (oldestFrame->GetNackCount() > 0);
+ if (retransmitted)
+ {
+ _jitterEstimate.FrameNacked();
+ }
+ else if (oldestFrame->Length() > 0)
+ {
+ // Ignore retransmitted and empty frames.
+ UpdateJitterAndDelayEstimates(*oldestFrame, false);
+ }
+ it = _frameList.erase(it);
+
+ // Look for previous frame loss
+ VerifyAndSetPreviousFrameLost(*oldestFrame);
+
+ // The state must be changed to decoding before cleaning up zero sized
+ // frames to avoid empty frames being cleaned up and then given to the
+ // decoder.
+ oldestFrame->SetState(kStateDecoding);
+
+ // Clean up old frames and empty frames
+ CleanUpOldFrames();
+
+ if (oldestFrame->FrameType() == kVideoFrameKey)
+ {
+ _waitingForKeyFrame = false;
+ }
+
+ // We have a frame - update decoded state with frame info.
+ _lastDecodedState.SetState(oldestFrame);
+
+ return oldestFrame;
+}
+
+// Must be called under the critical section _critSect. Should never be called
+// with retransmitted frames, they must be filtered out before this function is
+// called.
+void
+VCMJitterBuffer::UpdateJitterAndDelayEstimates(VCMJitterSample& sample,
+ bool incompleteFrame)
+{
+ if (sample.latestPacketTime == -1)
+ {
+ return;
+ }
+ if (incompleteFrame)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId), "Received incomplete frame "
+ "timestamp %u frame size %u at time %u",
+ sample.timestamp, sample.frameSize,
+ MaskWord64ToUWord32(sample.latestPacketTime));
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId), "Received complete frame "
+ "timestamp %u frame size %u at time %u",
+ sample.timestamp, sample.frameSize,
+ MaskWord64ToUWord32(sample.latestPacketTime));
+ }
+ UpdateJitterAndDelayEstimates(sample.latestPacketTime,
+ sample.timestamp,
+ sample.frameSize,
+ incompleteFrame);
+}
+
+// Must be called under the critical section _critSect. Should never be
+// called with retransmitted frames, they must be filtered out before this
+// function is called.
+void
+VCMJitterBuffer::UpdateJitterAndDelayEstimates(VCMFrameBuffer& frame,
+ bool incompleteFrame)
+{
+ if (frame.LatestPacketTimeMs() == -1)
+ {
+ return;
+ }
+ // No retransmitted frames should be a part of the jitter
+ // estimate.
+ if (incompleteFrame)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Received incomplete frame timestamp %u frame type %d "
+ "frame size %u at time %u, jitter estimate was %u",
+ frame.TimeStamp(), frame.FrameType(), frame.Length(),
+ MaskWord64ToUWord32(frame.LatestPacketTimeMs()),
+ GetEstimatedJitterMsInternal());
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),"Received complete frame "
+ "timestamp %u frame type %d frame size %u at time %u, "
+ "jitter estimate was %u",
+ frame.TimeStamp(), frame.FrameType(), frame.Length(),
+ MaskWord64ToUWord32(frame.LatestPacketTimeMs()),
+ GetEstimatedJitterMsInternal());
+ }
+ UpdateJitterAndDelayEstimates(frame.LatestPacketTimeMs(), frame.TimeStamp(),
+ frame.Length(), incompleteFrame);
+}
+
+// Must be called under the critical section _critSect. Should never be called
+// with retransmitted frames, they must be filtered out before this function
+// is called.
+void
+VCMJitterBuffer::UpdateJitterAndDelayEstimates(WebRtc_Word64 latestPacketTimeMs,
+ WebRtc_UWord32 timestamp,
+ WebRtc_UWord32 frameSize,
+ bool incompleteFrame)
+{
+ if (latestPacketTimeMs == -1)
+ {
+ return;
+ }
+ WebRtc_Word64 frameDelay;
+ // Calculate the delay estimate
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Packet received and sent to jitter estimate with: "
+ "timestamp=%u wallClock=%u", timestamp,
+ MaskWord64ToUWord32(latestPacketTimeMs));
+ bool notReordered = _delayEstimate.CalculateDelay(timestamp,
+ &frameDelay,
+ latestPacketTimeMs);
+ // Filter out frames which have been reordered in time by the network
+ if (notReordered)
+ {
+ // Update the jitter estimate with the new samples
+ _jitterEstimate.UpdateEstimate(frameDelay, frameSize, incompleteFrame);
+ }
+}
+
+WebRtc_UWord16*
+VCMJitterBuffer::GetNackList(WebRtc_UWord16& nackSize,bool& listExtended)
+{
+ return CreateNackList(nackSize,listExtended);
+}
+
+// Assume called internally with critsect
+WebRtc_Word32
+VCMJitterBuffer::GetLowHighSequenceNumbers(WebRtc_Word32& lowSeqNum,
+ WebRtc_Word32& highSeqNum) const
+{
+ // TODO (mikhal/stefan): refactor to use lastDecodedState
+ WebRtc_Word32 i = 0;
+ WebRtc_Word32 seqNum = -1;
+
+ highSeqNum = -1;
+ lowSeqNum = -1;
+ if (!_lastDecodedState.init())
+ lowSeqNum = _lastDecodedState.sequence_num();
+
+ // find highest seq numbers
+ for (i = 0; i < _maxNumberOfFrames; ++i)
+ {
+ seqNum = _frameBuffers[i]->GetHighSeqNum();
+
+ // Ignore free / empty frames
+ VCMFrameBufferStateEnum state = _frameBuffers[i]->GetState();
+
+ if ((kStateFree != state) &&
+ (kStateEmpty != state) &&
+ (kStateDecoding != state) &&
+ seqNum != -1)
+ {
+ bool wrap;
+ highSeqNum = LatestSequenceNumber(seqNum, highSeqNum, &wrap);
+ }
+ } // for
+ return 0;
+}
+
+
+WebRtc_UWord16*
+VCMJitterBuffer::CreateNackList(WebRtc_UWord16& nackSize, bool& listExtended)
+{
+ // TODO (mikhal/stefan): Refactor to use lastDecodedState.
+ CriticalSectionScoped cs(_critSect);
+ int i = 0;
+ WebRtc_Word32 lowSeqNum = -1;
+ WebRtc_Word32 highSeqNum = -1;
+ listExtended = false;
+
+ // Don't create list, if we won't wait for it
+ if (!WaitForNack())
+ {
+ nackSize = 0;
+ return NULL;
+ }
+
+ // Find the lowest (last decoded) sequence number and
+ // the highest (highest sequence number of the newest frame)
+ // sequence number. The nack list is a subset of the range
+ // between those two numbers.
+ GetLowHighSequenceNumbers(lowSeqNum, highSeqNum);
+
+ // write a list of all seq num we have
+ if (lowSeqNum == -1 || highSeqNum == -1)
+ {
+ // This happens if we lose the first packet, nothing is popped
+ if (highSeqNum == -1)
+ {
+ // we have not received any packets yet
+ nackSize = 0;
+ }
+ else
+ {
+ // signal that we want a key frame request to be sent
+ nackSize = 0xffff;
+ }
+ return NULL;
+ }
+
+ int numberOfSeqNum = 0;
+ if (lowSeqNum > highSeqNum)
+ {
+ if (lowSeqNum - highSeqNum > 0x00ff)
+ {
+ // wrap
+ numberOfSeqNum = (0xffff-lowSeqNum) + highSeqNum + 1;
+ }
+ }
+ else
+ {
+ numberOfSeqNum = highSeqNum - lowSeqNum;
+ }
+
+ if (numberOfSeqNum > kNackHistoryLength)
+ {
+ // Nack list is too big, flush and try to restart.
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Nack list too large, try to find a key frame and restart "
+ "from seq: %d. Lowest seq in jb %d", highSeqNum,lowSeqNum);
+
+ // This nack size will trigger a key request...
+ bool foundKeyFrame = false;
+
+ while (numberOfSeqNum > kNackHistoryLength)
+ {
+ foundKeyFrame = RecycleFramesUntilKeyFrame();
+
+ if (!foundKeyFrame)
+ {
+ break;
+ }
+
+ // Check if we still have too many packets in JB
+ lowSeqNum = -1;
+ highSeqNum = -1;
+ GetLowHighSequenceNumbers(lowSeqNum, highSeqNum);
+
+ if (highSeqNum == -1)
+ {
+ assert(lowSeqNum != -1); // This should never happen
+ // We can't calculate the nack list length...
+ return NULL;
+ }
+
+ numberOfSeqNum = 0;
+ if (lowSeqNum > highSeqNum)
+ {
+ if (lowSeqNum - highSeqNum > 0x00ff)
+ {
+ // wrap
+ numberOfSeqNum = (0xffff-lowSeqNum) + highSeqNum + 1;
+ highSeqNum=lowSeqNum;
+ }
+ }
+ else
+ {
+ numberOfSeqNum = highSeqNum - lowSeqNum;
+ }
+
+ } // end while
+
+ if (!foundKeyFrame)
+ {
+ // No key frame in JB.
+
+ // Set the last decoded sequence number to current high.
+ // This is to not get a large nack list again right away
+ _lastDecodedState.SetSeqNum(static_cast<uint16_t>(highSeqNum));
+ // Set to trigger key frame signal
+ nackSize = 0xffff;
+ listExtended = true;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
+ "\tNo key frame found, request one. _lastDecodedSeqNum[0] "
+ "%d", _lastDecodedState.sequence_num());
+ }
+ else
+ {
+ // We have cleaned up the jb and found a key frame
+ // The function itself has set last decoded seq.
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
+ "\tKey frame found. _lastDecodedSeqNum[0] %d",
+ _lastDecodedState.sequence_num());
+ nackSize = 0;
+ }
+
+ return NULL;
+ }
+
+ WebRtc_UWord16 seqNumberIterator = (WebRtc_UWord16)(lowSeqNum + 1);
+ for (i = 0; i < numberOfSeqNum; i++)
+ {
+ _NACKSeqNumInternal[i] = seqNumberIterator;
+ seqNumberIterator++;
+ }
+
+ // now we have a list of all sequence numbers that could have been sent
+
+ // zero out the ones we have received
+ for (i = 0; i < _maxNumberOfFrames; i++)
+ {
+ // loop all created frames
+ // We don't need to check if frame is decoding since lowSeqNum is based
+ // on _lastDecodedSeqNum
+ // Ignore free frames
+ VCMFrameBufferStateEnum state = _frameBuffers[i]->GetState();
+
+ if ((kStateFree != state) &&
+ (kStateEmpty != state) &&
+ (kStateDecoding != state))
+ {
+ // Reaching thus far means we are going to update the nack list
+ // When in hybrid mode, we use the soft NACKing feature.
+ if (_nackMode == kNackHybrid)
+ {
+ _frameBuffers[i]->BuildSoftNackList(_NACKSeqNumInternal,
+ numberOfSeqNum,
+ _rttMs);
+ }
+ else
+ {
+ // Used when the frame is being processed by the decoding thread
+ // don't need to use that info in this loop.
+ _frameBuffers[i]->BuildHardNackList(_NACKSeqNumInternal,
+ numberOfSeqNum);
+ }
+ }
+ }
+
+ // compress list
+ int emptyIndex = -1;
+ for (i = 0; i < numberOfSeqNum; i++)
+ {
+ if (_NACKSeqNumInternal[i] == -1 || _NACKSeqNumInternal[i] == -2 )
+ {
+ // this is empty
+ if (emptyIndex == -1)
+ {
+ // no empty index before, remember this position
+ emptyIndex = i;
+ }
+ }
+ else
+ {
+ // this is not empty
+ if (emptyIndex == -1)
+ {
+ // no empty index, continue
+ }
+ else
+ {
+ _NACKSeqNumInternal[emptyIndex] = _NACKSeqNumInternal[i];
+ _NACKSeqNumInternal[i] = -1;
+ emptyIndex++;
+ }
+ }
+ } // for
+
+ if (emptyIndex == -1)
+ {
+ // no empty
+ nackSize = numberOfSeqNum;
+ }
+ else
+ {
+ nackSize = emptyIndex;
+ }
+
+ if (nackSize > _NACKSeqNumLength)
+ {
+ // Larger list: nack list was extended since the last call.
+ listExtended = true;
+ }
+
+ for (WebRtc_UWord32 j = 0; j < nackSize; j++)
+ {
+ // Check if the list has been extended since it was last created. I.e,
+ // new items have been added
+ if (_NACKSeqNumLength > j && !listExtended)
+ {
+ WebRtc_UWord32 k = 0;
+ for (k = j; k < _NACKSeqNumLength; k++)
+ {
+ // Found the item in the last list, i.e, no new items found yet.
+ if (_NACKSeqNum[k] == (WebRtc_UWord16)_NACKSeqNumInternal[j])
+ {
+ break;
+ }
+ }
+ if (k == _NACKSeqNumLength) // New item not found in last list.
+ {
+ listExtended = true;
+ }
+ }
+ else
+ {
+ listExtended = true;
+ }
+ _NACKSeqNum[j] = (WebRtc_UWord16)_NACKSeqNumInternal[j];
+ }
+
+ _NACKSeqNumLength = nackSize;
+
+ return _NACKSeqNum;
+}
+
+// Release frame when done with decoding. Should never be used to release
+// frames from within the jitter buffer.
+void
+VCMJitterBuffer::ReleaseFrame(VCMEncodedFrame* frame)
+{
+ CriticalSectionScoped cs(_critSect);
+ VCMFrameBuffer* frameBuffer = static_cast<VCMFrameBuffer*>(frame);
+ if (frameBuffer != NULL)
+ frameBuffer->SetState(kStateFree);
+}
+
+WebRtc_Word64
+VCMJitterBuffer::LastPacketTime(VCMEncodedFrame* frame,
+ bool& retransmitted) const
+{
+ CriticalSectionScoped cs(_critSect);
+ retransmitted = (static_cast<VCMFrameBuffer*>(frame)->GetNackCount() > 0);
+ return static_cast<VCMFrameBuffer*>(frame)->LatestPacketTimeMs();
+}
+
+WebRtc_Word64
+VCMJitterBuffer::LastDecodedTimestamp() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return _lastDecodedState.time_stamp();
+}
+
+// Insert packet
+// Takes crit sect, and inserts packet in frame buffer, possibly does logging
+VCMFrameBufferEnum
+VCMJitterBuffer::InsertPacket(VCMEncodedFrame* buffer, const VCMPacket& packet)
+{
+ CriticalSectionScoped cs(_critSect);
+ WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
+ VCMFrameBufferEnum bufferReturn = kSizeError;
+ VCMFrameBufferEnum ret = kSizeError;
+ VCMFrameBuffer* frame = static_cast<VCMFrameBuffer*>(buffer);
+
+ // We are keeping track of the first seq num, the latest seq num and
+ // the number of wraps to be able to calculate how many packets we expect.
+ if (_firstPacket)
+ {
+ // Now it's time to start estimating jitter
+ // reset the delay estimate.
+ _delayEstimate.Reset(_clock->MillisecondTimestamp());
+ _firstPacket = false;
+ }
+
+ // Empty packets may bias the jitter estimate (lacking size component),
+ // therefore don't let empty packet trigger the following updates:
+ if (packet.frameType != kFrameEmpty)
+ {
+ if (_waitingForCompletion.timestamp == packet.timestamp)
+ {
+ // This can get bad if we have a lot of duplicate packets,
+ // we will then count some packet multiple times.
+ _waitingForCompletion.frameSize += packet.sizeBytes;
+ _waitingForCompletion.latestPacketTime = nowMs;
+ }
+ else if (_waitingForCompletion.latestPacketTime >= 0 &&
+ _waitingForCompletion.latestPacketTime + 2000 <= nowMs)
+ {
+ // A packet should never be more than two seconds late
+ UpdateJitterAndDelayEstimates(_waitingForCompletion, true);
+ _waitingForCompletion.latestPacketTime = -1;
+ _waitingForCompletion.frameSize = 0;
+ _waitingForCompletion.timestamp = 0;
+ }
+ }
+
+ if (frame != NULL)
+ {
+ VCMFrameBufferStateEnum state = frame->GetState();
+ _lastDecodedState.UpdateOldPacket(&packet);
+ // Insert packet
+ // Check for first packet
+ // High sequence number will be -1 if neither an empty packet nor
+ // a media packet has been inserted.
+ bool first = (frame->GetHighSeqNum() == -1);
+ // When in Hybrid mode, we allow for a decodable state
+ // Note: Under current version, a decodable frame will never be
+ // triggered, as the body of the function is empty.
+ // TODO (mikhal): Update when decodable is enabled.
+ bufferReturn = frame->InsertPacket(packet, nowMs,
+ _nackMode == kNackHybrid,
+ _rttMs);
+ ret = bufferReturn;
+
+ if (bufferReturn > 0)
+ {
+ _incomingBitCount += packet.sizeBytes << 3;
+
+ // Has this packet been nacked or is it about to be nacked?
+ if (IsPacketRetransmitted(packet))
+ {
+ frame->IncrementNackCount();
+ }
+
+ // Insert each frame once on the arrival of the first packet
+ // belonging to that frame (media or empty)
+ if (state == kStateEmpty && first)
+ {
+ ret = kFirstPacket;
+ FrameList::reverse_iterator rit = std::find_if(
+ _frameList.rbegin(), _frameList.rend(),
+ FrameSmallerTimestamp(frame->TimeStamp()));
+ _frameList.insert(rit.base(), frame);
+ }
+ }
+ }
+ switch(bufferReturn)
+ {
+ case kStateError:
+ case kTimeStampError:
+ case kSizeError:
+ {
+ if (frame != NULL)
+ {
+ // Will be released when it gets old.
+ frame->Reset();
+ frame->SetState(kStateEmpty);
+ }
+ break;
+ }
+ case kCompleteSession:
+ {
+ // Only update return value for a JB flush indicator.
+ if (UpdateFrameState(frame) == kFlushIndicator)
+ ret = kFlushIndicator;
+ // Signal that we have a received packet
+ _packetEvent.Set();
+ break;
+ }
+ case kDecodableSession:
+ case kIncomplete:
+ {
+ // Signal that we have a received packet
+ _packetEvent.Set();
+ break;
+ }
+ case kNoError:
+ case kDuplicatePacket:
+ {
+ break;
+ }
+ default:
+ {
+ assert(!"JitterBuffer::InsertPacket: Undefined value");
+ }
+ }
+ return ret;
+}
+
+// Must be called from within _critSect
+void
+VCMJitterBuffer::UpdateOldJitterSample(const VCMPacket& packet)
+{
+ if (_waitingForCompletion.timestamp != packet.timestamp &&
+ LatestTimestamp(_waitingForCompletion.timestamp, packet.timestamp,
+ NULL) == packet.timestamp)
+ {
+ // This is a newer frame than the one waiting for completion.
+ _waitingForCompletion.frameSize = packet.sizeBytes;
+ _waitingForCompletion.timestamp = packet.timestamp;
+ }
+ else
+ {
+ // This can get bad if we have a lot of duplicate packets,
+ // we will then count some packet multiple times.
+ _waitingForCompletion.frameSize += packet.sizeBytes;
+ _jitterEstimate.UpdateMaxFrameSize(_waitingForCompletion.frameSize);
+ }
+}
+
+// Must be called from within _critSect
+bool
+VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const
+{
+ if (_NACKSeqNum && _NACKSeqNumLength > 0)
+ {
+ for (WebRtc_UWord16 i = 0; i < _NACKSeqNumLength; i++)
+ {
+ if (packet.seqNum == _NACKSeqNum[i])
+ {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+// Get nack status (enabled/disabled)
+VCMNackMode
+VCMJitterBuffer::GetNackMode() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return _nackMode;
+}
+
+// Set NACK mode
+void
+VCMJitterBuffer::SetNackMode(VCMNackMode mode,
+ int lowRttNackThresholdMs,
+ int highRttNackThresholdMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ _nackMode = mode;
+ assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
+ assert(highRttNackThresholdMs == -1 ||
+ lowRttNackThresholdMs <= highRttNackThresholdMs);
+ assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
+ _lowRttNackThresholdMs = lowRttNackThresholdMs;
+ _highRttNackThresholdMs = highRttNackThresholdMs;
+ if (_nackMode == kNoNack)
+ {
+ _jitterEstimate.ResetNackCount();
+ }
+}
+
+
+// Recycle oldest frames up to a key frame, used if JB is completely full
+bool
+VCMJitterBuffer::RecycleFramesUntilKeyFrame()
+{
+ // Remove up to oldest key frame
+ while (_frameList.size() > 0)
+ {
+ // Throw at least one frame.
+ _dropCount++;
+ FrameList::iterator it = _frameList.begin();
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Jitter buffer drop count:%d, lowSeq %d", _dropCount,
+ (*it)->GetLowSeqNum());
+ RecycleFrame(*it);
+ it = _frameList.erase(it);
+ if (it != _frameList.end() && (*it)->FrameType() == kVideoFrameKey)
+ {
+ // Fake the lastDecodedState to match this key frame.
+ _lastDecodedState.SetStateOneBack(*it);
+ return true;
+ }
+ }
+ _waitingForKeyFrame = true;
+ _lastDecodedState.Reset(); // TODO (mikhal): no sync
+ return false;
+}
+
+// Must be called under the critical section _critSect.
+void VCMJitterBuffer::CleanUpOldFrames() {
+ while (_frameList.size() > 0) {
+ VCMFrameBuffer* oldestFrame = _frameList.front();
+ bool nextFrameEmpty = (_lastDecodedState.ContinuousFrame(oldestFrame) &&
+ oldestFrame->GetState() == kStateEmpty);
+ if (_lastDecodedState.IsOldFrame(oldestFrame) ||
+ (nextFrameEmpty && _frameList.size() > 1)) {
+ ReleaseFrameInternal(_frameList.front());
+ _frameList.erase(_frameList.begin());
+ } else {
+ break;
+ }
+ }
+}
+
+// Used in GetFrameForDecoding
+void VCMJitterBuffer::VerifyAndSetPreviousFrameLost(VCMFrameBuffer& frame) {
+ frame.MakeSessionDecodable(); // Make sure the session can be decoded.
+ if (frame.FrameType() == kVideoFrameKey)
+ return;
+
+ if (!_lastDecodedState.ContinuousFrame(&frame))
+ frame.SetPreviousFrameLoss();
+}
+
+bool
+VCMJitterBuffer::WaitForNack()
+{
+ // NACK disabled -> can't wait
+ if (_nackMode == kNoNack)
+ {
+ return false;
+ }
+ // NACK only -> always wait
+ else if (_nackMode == kNackInfinite)
+ {
+ return true;
+ }
+ // else: hybrid mode, evaluate
+ // RTT high, don't wait
+ if (_highRttNackThresholdMs >= 0 &&
+ _rttMs >= static_cast<unsigned int>(_highRttNackThresholdMs))
+ {
+ return false;
+ }
+ // Either NACK only or hybrid
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer.h b/trunk/src/modules/video_coding/main/source/jitter_buffer.h
new file mode 100644
index 0000000..d951187
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_buffer.h
@@ -0,0 +1,259 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+
+#include <list>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+#include "modules/video_coding/main/source/decoding_state.h"
+#include "modules/video_coding/main/source/event.h"
+#include "modules/video_coding/main/source/inter_frame_delay.h"
+#include "modules/video_coding/main/source/jitter_buffer_common.h"
+#include "modules/video_coding/main/source/jitter_estimator.h"
+#include "system_wrappers/interface/constructor_magic.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+enum VCMNackMode
+{
+ kNackInfinite,
+ kNackHybrid,
+ kNoNack
+};
+
+typedef std::list<VCMFrameBuffer*> FrameList;
+
+// forward declarations
+class TickTimeBase;
+class VCMFrameBuffer;
+class VCMPacket;
+class VCMEncodedFrame;
+
+class VCMJitterSample
+{
+public:
+ VCMJitterSample() : timestamp(0), frameSize(0), latestPacketTime(-1) {}
+ WebRtc_UWord32 timestamp;
+ WebRtc_UWord32 frameSize;
+ WebRtc_Word64 latestPacketTime;
+};
+
+class VCMJitterBuffer
+{
+public:
+ VCMJitterBuffer(TickTimeBase* clock,
+ WebRtc_Word32 vcmId = -1,
+ WebRtc_Word32 receiverId = -1,
+ bool master = true);
+ virtual ~VCMJitterBuffer();
+
+ void CopyFrom(const VCMJitterBuffer& rhs);
+
+ // We need a start and stop to break out of the wait event
+ // used in GetCompleteFrameForDecoding
+ void Start();
+ void Stop();
+ bool Running() const;
+
+ // Empty the Jitter buffer of all its data
+ void Flush();
+
+ // Statistics, Get received key and delta frames
+ WebRtc_Word32 GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
+ WebRtc_UWord32& receivedKeyFrames) const;
+
+ // The number of packets discarded by the jitter buffer because the decoder
+ // won't be able to decode them.
+ WebRtc_UWord32 NumNotDecodablePackets() const;
+ // Get number of packets discarded by the jitter buffer
+ WebRtc_UWord32 DiscardedPackets() const;
+
+ // Statistics, Calculate frame and bit rates
+ WebRtc_Word32 GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate);
+
+ // Wait for the first packet in the next frame to arrive, blocks
+ // for <= maxWaitTimeMS ms
+ WebRtc_Word64 GetNextTimeStamp(WebRtc_UWord32 maxWaitTimeMS,
+ FrameType& incomingFrameType,
+ WebRtc_Word64& renderTimeMs);
+
+ // Will the packet sequence be complete if the next frame is grabbed
+ // for decoding right now? That is, have we lost a frame between the
+ // last decoded frame and the next, or is the next frame missing one
+ // or more packets?
+ bool CompleteSequenceWithNextFrame();
+
+ // TODO (mikhal/stefan): Merge all GetFrameForDecoding into one.
+ // Wait maxWaitTimeMS for a complete frame to arrive. After timeout NULL
+ // is returned.
+ VCMEncodedFrame* GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS);
+
+ // Get a frame for decoding (even an incomplete) without delay.
+ VCMEncodedFrame* GetFrameForDecoding();
+
+ VCMEncodedFrame* GetFrameForDecodingNACK();
+
+ // Release frame (when done with decoding)
+ void ReleaseFrame(VCMEncodedFrame* frame);
+
+ // Get frame to use for this timestamp
+ WebRtc_Word32 GetFrame(const VCMPacket& packet, VCMEncodedFrame*&);
+ VCMEncodedFrame* GetFrame(const VCMPacket& packet); // deprecated
+
+ // Returns the time in ms when the latest packet was inserted into the frame.
+ // Retransmitted is set to true if any of the packets belonging to the frame
+ // has been retransmitted.
+ WebRtc_Word64 LastPacketTime(VCMEncodedFrame* frame,
+ bool& retransmitted) const;
+
+ // Insert a packet into a frame
+ VCMFrameBufferEnum InsertPacket(VCMEncodedFrame* frame,
+ const VCMPacket& packet);
+
+ // Sync
+ WebRtc_UWord32 GetEstimatedJitterMS();
+ void UpdateRtt(WebRtc_UWord32 rttMs);
+
+ // NACK
+ // Set the NACK mode. "highRttNackThreshold" is an RTT threshold in ms above
+ // which NACK will be disabled if the NACK mode is "kNackHybrid",
+ // -1 meaning that NACK is always enabled in the Hybrid mode.
+ // "lowRttNackThreshold" is an RTT threshold in ms below which we expect to
+ // rely on NACK only, and therefore are using larger buffers to have time to
+ // wait for retransmissions.
+ void SetNackMode(VCMNackMode mode,
+ int lowRttNackThresholdMs,
+ int highRttNackThresholdMs);
+ VCMNackMode GetNackMode() const; // Get nack mode
+ // Get list of missing sequence numbers (size in number of elements)
+ WebRtc_UWord16* GetNackList(WebRtc_UWord16& nackSize,
+ bool& listExtended);
+
+ WebRtc_Word64 LastDecodedTimestamp() const;
+
+private:
+ // Misc help functions
+ // Recycle (release) frame, used if we didn't receive whole frame
+ void RecycleFrame(VCMFrameBuffer* frame);
+ void ReleaseFrameInternal(VCMFrameBuffer* frame);
+ // Flush and reset the jitter buffer. Call under critical section.
+ void FlushInternal();
+
+ // Help functions for insert packet
+ // Get empty frame, creates new (i.e. increases JB size) if necessary
+ VCMFrameBuffer* GetEmptyFrame();
+ // Recycle oldest frames up to a key frame, used if JB is completely full
+ bool RecycleFramesUntilKeyFrame();
+ // Update frame state
+ // (set as complete or reconstructable if conditions are met)
+ VCMFrameBufferEnum UpdateFrameState(VCMFrameBuffer* frameListItem);
+
+ // Help functions for getting a frame
+ // Find oldest complete frame, used for getting next frame to decode
+ // When enabled, will return a decodable frame
+ FrameList::iterator FindOldestCompleteContinuousFrame(bool enableDecodable);
+
+ void CleanUpOldFrames();
+
+ void VerifyAndSetPreviousFrameLost(VCMFrameBuffer& frame);
+ bool IsPacketRetransmitted(const VCMPacket& packet) const;
+
+ void UpdateJitterAndDelayEstimates(VCMJitterSample& sample,
+ bool incompleteFrame);
+ void UpdateJitterAndDelayEstimates(VCMFrameBuffer& frame,
+ bool incompleteFrame);
+ void UpdateJitterAndDelayEstimates(WebRtc_Word64 latestPacketTimeMs,
+ WebRtc_UWord32 timestamp,
+ WebRtc_UWord32 frameSize,
+ bool incompleteFrame);
+ void UpdateOldJitterSample(const VCMPacket& packet);
+ WebRtc_UWord32 GetEstimatedJitterMsInternal();
+
+ // NACK help
+ WebRtc_UWord16* CreateNackList(WebRtc_UWord16& nackSize,
+ bool& listExtended);
+ WebRtc_Word32 GetLowHighSequenceNumbers(WebRtc_Word32& lowSeqNum,
+ WebRtc_Word32& highSeqNum) const;
+
+ // Decide whether should wait for NACK (mainly relevant for hybrid mode)
+ bool WaitForNack();
+
+ WebRtc_Word32 _vcmId;
+ WebRtc_Word32 _receiverId;
+ TickTimeBase* _clock;
+ // If we are running (have started) or not
+ bool _running;
+ CriticalSectionWrapper* _critSect;
+ bool _master;
+ // Event to signal when we have a frame ready for decoder
+ VCMEvent _frameEvent;
+ // Event to signal when we have received a packet
+ VCMEvent _packetEvent;
+ // Number of allocated frames
+ WebRtc_Word32 _maxNumberOfFrames;
+ // Array of pointers to the frames in JB
+ VCMFrameBuffer* _frameBuffers[kMaxNumberOfFrames];
+ FrameList _frameList;
+
+ // timing
+ VCMDecodingState _lastDecodedState;
+ WebRtc_UWord32 _packetsNotDecodable;
+
+ // Statistics
+ // Frame counter for each type (key, delta, golden, key-delta)
+ WebRtc_UWord8 _receiveStatistics[4];
+ // Latest calculated frame rates of incoming stream
+ WebRtc_UWord8 _incomingFrameRate;
+ // Frame counter, reset in GetUpdate
+ WebRtc_UWord32 _incomingFrameCount;
+ // Real time for last _frameCount reset
+ WebRtc_Word64 _timeLastIncomingFrameCount;
+ // Received bits counter, reset in GetUpdate
+ WebRtc_UWord32 _incomingBitCount;
+ WebRtc_UWord32 _incomingBitRate;
+ WebRtc_UWord32 _dropCount; // Frame drop counter
+ // Number of frames in a row that have been too old
+ WebRtc_UWord32 _numConsecutiveOldFrames;
+ // Number of packets in a row that have been too old
+ WebRtc_UWord32 _numConsecutiveOldPackets;
+ // Number of packets discarded by the jitter buffer
+ WebRtc_UWord32 _discardedPackets;
+
+ // Filters for estimating jitter
+ VCMJitterEstimator _jitterEstimate;
+ // Calculates network delays used for jitter calculations
+ VCMInterFrameDelay _delayEstimate;
+ VCMJitterSample _waitingForCompletion;
+ WebRtc_UWord32 _rttMs;
+
+ // NACK
+ VCMNackMode _nackMode;
+ int _lowRttNackThresholdMs;
+ int _highRttNackThresholdMs;
+ // Holds the internal nack list (the missing sequence numbers)
+ WebRtc_Word32 _NACKSeqNumInternal[kNackHistoryLength];
+ WebRtc_UWord16 _NACKSeqNum[kNackHistoryLength];
+ WebRtc_UWord32 _NACKSeqNumLength;
+ bool _waitingForKeyFrame;
+
+ bool _firstPacket;
+
+ DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc
new file mode 100644
index 0000000..79a21b4
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "jitter_buffer_common.h"
+
+#include <cstdlib>
+
+namespace webrtc {
+
+WebRtc_UWord32 LatestTimestamp(WebRtc_UWord32 timestamp1,
+ WebRtc_UWord32 timestamp2,
+ bool* has_wrapped) {
+ bool wrap = (timestamp2 < 0x0000ffff && timestamp1 > 0xffff0000) ||
+ (timestamp2 > 0xffff0000 && timestamp1 < 0x0000ffff);
+ if (has_wrapped != NULL)
+ *has_wrapped = wrap;
+ if (timestamp1 > timestamp2 && !wrap)
+ return timestamp1;
+ else if (timestamp1 <= timestamp2 && !wrap)
+ return timestamp2;
+ else if (timestamp1 < timestamp2 && wrap)
+ return timestamp1;
+ else
+ return timestamp2;
+}
+
+WebRtc_Word32 LatestSequenceNumber(WebRtc_Word32 seq_num1,
+ WebRtc_Word32 seq_num2,
+ bool* has_wrapped) {
+ if (seq_num1 < 0 && seq_num2 < 0)
+ return -1;
+ else if (seq_num1 < 0)
+ return seq_num2;
+ else if (seq_num2 < 0)
+ return seq_num1;
+
+ bool wrap = (seq_num1 < 0x00ff && seq_num2 > 0xff00) ||
+ (seq_num1 > 0xff00 && seq_num2 < 0x00ff);
+
+ if (has_wrapped != NULL)
+ *has_wrapped = wrap;
+
+ if (seq_num2 > seq_num1 && !wrap)
+ return seq_num2;
+ else if (seq_num2 <= seq_num1 && !wrap)
+ return seq_num1;
+ else if (seq_num2 < seq_num1 && wrap)
+ return seq_num2;
+ else
+ return seq_num1;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h
new file mode 100644
index 0000000..38cea42
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+enum { kMaxNumberOfFrames = 100 };
+enum { kStartNumberOfFrames = 6 }; // in packets, 6 packets are approximately 198 ms,
+ // we need at least one more for process
+enum { kMaxVideoDelayMs = 2000 }; // in ms
+
+enum VCMJitterBufferEnum
+{
+ kMaxConsecutiveOldFrames = 60,
+ kMaxConsecutiveOldPackets = 300,
+ kMaxPacketsInSession = 800,
+ kBufferIncStepSizeBytes = 30000, // >20 packets
+ kMaxJBFrameSizeBytes = 4000000 // sanity don't go above 4Mbyte
+};
+
+enum VCMFrameBufferEnum
+{
+ kStateError = -4,
+ kFlushIndicator = -3, // Indicator that a flush has occurred.
+ kTimeStampError = -2,
+ kSizeError = -1,
+ kNoError = 0,
+ kIncomplete = 1, // Frame incomplete
+ kFirstPacket = 2,
+ kCompleteSession = 3, // at least one layer in the frame complete.
+ kDecodableSession = 4, // Frame incomplete, but ready to be decoded
+ kDuplicatePacket = 5 // We're receiving a duplicate packet.
+};
+
+enum VCMFrameBufferStateEnum
+{
+ kStateFree, // Unused frame in the JB
+ kStateEmpty, // frame popped by the RTP receiver
+ kStateIncomplete, // frame that have one or more packet(s) stored
+ kStateComplete, // frame that have all packets
+ kStateDecoding, // frame popped by the decoding thread
+ kStateDecodable // Hybrid mode - frame can be decoded
+};
+
+enum { kH264StartCodeLengthBytes = 4};
+
+// Used to indicate if a received packet contain a complete NALU (or equivalent)
+enum VCMNaluCompleteness
+{
+ kNaluUnset = 0, //Packet has not been filled.
+ kNaluComplete = 1, //Packet can be decoded as is.
+ kNaluStart, // Packet contain beginning of NALU
+ kNaluIncomplete, //Packet is not beginning or end of NALU
+ kNaluEnd, // Packet is the end of a NALU
+};
+
+// Returns the latest of the two timestamps, compensating for wrap arounds.
+// This function assumes that the two timestamps are close in time.
+WebRtc_UWord32 LatestTimestamp(WebRtc_UWord32 timestamp1,
+ WebRtc_UWord32 timestamp2,
+ bool* has_wrapped);
+
+// Returns the latest of the two sequence numbers, compensating for wrap
+// arounds. This function assumes that the two sequence numbers are close in
+// time.
+WebRtc_Word32 LatestSequenceNumber(WebRtc_Word32 seq_num1,
+ WebRtc_Word32 seq_num2,
+ bool* has_wrapped);
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc
new file mode 100644
index 0000000..6e48ea7
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -0,0 +1,376 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include <list>
+
+#include "gtest/gtest.h"
+#include "modules/video_coding/main/source/jitter_buffer.h"
+#include "modules/video_coding/main/source/media_opt_util.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "modules/video_coding/main/source/packet.h"
+
+namespace webrtc {
+
+class StreamGenerator {
+ public:
+ StreamGenerator(uint16_t start_seq_num, uint32_t start_timestamp,
+ int64_t current_time)
+ : packets_(),
+ sequence_number_(start_seq_num),
+ timestamp_(start_timestamp),
+ start_time_(current_time) {}
+
+ void Init(uint16_t start_seq_num, uint32_t start_timestamp,
+ int64_t current_time) {
+ packets_.clear();
+ sequence_number_ = start_seq_num;
+ timestamp_ = start_timestamp;
+ start_time_ = current_time;
+ }
+
+ void GenerateFrame(FrameType type, int num_media_packets,
+ int num_empty_packets, int64_t current_time) {
+ timestamp_ += 90 * (current_time - start_time_);
+ // Move the sequence number counter if all packets from the previous frame
+ // wasn't collected.
+ sequence_number_ += packets_.size();
+ packets_.clear();
+ for (int i = 0; i < num_media_packets; ++i) {
+ packets_.push_back(GeneratePacket(sequence_number_,
+ timestamp_,
+ (i == 0),
+ (i == num_media_packets - 1),
+ type));
+ ++sequence_number_;
+ }
+ for (int i = 0; i < num_empty_packets; ++i) {
+ packets_.push_back(GeneratePacket(sequence_number_,
+ timestamp_,
+ false,
+ false,
+ kFrameEmpty));
+ ++sequence_number_;
+ }
+ }
+
+ static VCMPacket GeneratePacket(uint16_t sequence_number,
+ uint32_t timestamp,
+ bool first_packet,
+ bool marker_bit,
+ FrameType type) {
+ VCMPacket packet;
+ packet.seqNum = sequence_number;
+ packet.timestamp = timestamp;
+ packet.frameType = type;
+ packet.isFirstPacket = first_packet;
+ packet.markerBit = marker_bit;
+ if (packet.isFirstPacket)
+ packet.completeNALU = kNaluStart;
+ else if (packet.markerBit)
+ packet.completeNALU = kNaluEnd;
+ else
+ packet.completeNALU = kNaluIncomplete;
+ return packet;
+ }
+
+ bool PopPacket(VCMPacket* packet, int index) {
+ std::list<VCMPacket>::iterator it = GetPacketIterator(index);
+ if (it == packets_.end())
+ return false;
+ if (packet)
+ *packet = (*it);
+ packets_.erase(it);
+ return true;
+ }
+
+ bool GetPacket(VCMPacket* packet, int index) {
+ std::list<VCMPacket>::iterator it = GetPacketIterator(index);
+ if (it == packets_.end())
+ return false;
+ if (packet)
+ *packet = (*it);
+ return true;
+ }
+
+ bool NextPacket(VCMPacket* packet) {
+ if (packets_.empty())
+ return false;
+ if (packet != NULL)
+ *packet = packets_.front();
+ packets_.pop_front();
+ return true;
+ }
+
+ uint16_t NextSequenceNumber() const {
+ if (packets_.empty())
+ return sequence_number_;
+ return packets_.front().seqNum;
+ }
+
+ int PacketsRemaining() const {
+ return packets_.size();
+ }
+
+ private:
+ std::list<VCMPacket>::iterator GetPacketIterator(int index) {
+ std::list<VCMPacket>::iterator it = packets_.begin();
+ for (int i = 0; i < index; ++i) {
+ ++it;
+ if (it == packets_.end()) break;
+ }
+ return it;
+ }
+
+ std::list<VCMPacket> packets_;
+ uint16_t sequence_number_;
+ uint32_t timestamp_;
+ int64_t start_time_;
+
+ DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
+};
+
+class TestRunningJitterBuffer : public ::testing::Test {
+ protected:
+ enum { kDataBufferSize = 10 };
+ enum { kDefaultFrameRate = 25 };
+ enum { kDefaultFramePeriodMs = 1000 / kDefaultFrameRate };
+
+ virtual void SetUp() {
+ clock_ = new FakeTickTime(0);
+ jitter_buffer_ = new VCMJitterBuffer(clock_);
+ stream_generator = new StreamGenerator(0, 0,
+ clock_->MillisecondTimestamp());
+ jitter_buffer_->Start();
+ memset(data_buffer_, 0, kDataBufferSize);
+ }
+
+ virtual void TearDown() {
+ jitter_buffer_->Stop();
+ delete stream_generator;
+ delete jitter_buffer_;
+ delete clock_;
+ }
+
+ VCMFrameBufferEnum InsertPacketAndPop(int index) {
+ VCMPacket packet;
+ VCMEncodedFrame* frame;
+
+ packet.dataPtr = data_buffer_;
+ bool packet_available = stream_generator->PopPacket(&packet, index);
+ EXPECT_TRUE(packet_available);
+ if (!packet_available)
+ return kStateError; // Return here to avoid crashes below.
+ EXPECT_EQ(VCM_OK, jitter_buffer_->GetFrame(packet, frame));
+ return jitter_buffer_->InsertPacket(frame, packet);
+ }
+
+ VCMFrameBufferEnum InsertPacket(int index) {
+ VCMPacket packet;
+ VCMEncodedFrame* frame;
+
+ packet.dataPtr = data_buffer_;
+ bool packet_available = stream_generator->GetPacket(&packet, index);
+ EXPECT_TRUE(packet_available);
+ if (!packet_available)
+ return kStateError; // Return here to avoid crashes below.
+ EXPECT_EQ(VCM_OK, jitter_buffer_->GetFrame(packet, frame));
+ return jitter_buffer_->InsertPacket(frame, packet);
+ }
+
+ void InsertFrame(FrameType frame_type) {
+ stream_generator->GenerateFrame(frame_type,
+ (frame_type != kFrameEmpty) ? 1 : 0,
+ (frame_type == kFrameEmpty) ? 1 : 0,
+ clock_->MillisecondTimestamp());
+ EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
+ clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ }
+
+ void InsertFrames(int num_frames, FrameType frame_type) {
+ for (int i = 0; i < num_frames; ++i) {
+ InsertFrame(frame_type);
+ }
+ }
+
+ void DropFrame(int num_packets) {
+ stream_generator->GenerateFrame(kVideoFrameDelta, num_packets, 0,
+ clock_->MillisecondTimestamp());
+ clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ }
+
+ bool DecodeCompleteFrame() {
+ VCMEncodedFrame* frame = jitter_buffer_->GetCompleteFrameForDecoding(0);
+ bool ret = (frame != NULL);
+ jitter_buffer_->ReleaseFrame(frame);
+ return ret;
+ }
+
+ bool DecodeFrame() {
+ VCMEncodedFrame* frame = jitter_buffer_->GetFrameForDecoding();
+ bool ret = (frame != NULL);
+ jitter_buffer_->ReleaseFrame(frame);
+ return ret;
+ }
+
+ VCMJitterBuffer* jitter_buffer_;
+ StreamGenerator* stream_generator;
+ FakeTickTime* clock_;
+ uint8_t data_buffer_[kDataBufferSize];
+};
+
+class TestJitterBufferNack : public TestRunningJitterBuffer {
+ protected:
+ virtual void SetUp() {
+ TestRunningJitterBuffer::SetUp();
+ jitter_buffer_->SetNackMode(kNackInfinite, -1, -1);
+ }
+
+ virtual void TearDown() {
+ TestRunningJitterBuffer::TearDown();
+ }
+};
+
+TEST_F(TestRunningJitterBuffer, TestFull) {
+ // Insert a key frame and decode it.
+ InsertFrame(kVideoFrameKey);
+ EXPECT_TRUE(DecodeCompleteFrame());
+ DropFrame(1);
+ // Fill the jitter buffer.
+ InsertFrames(kMaxNumberOfFrames, kVideoFrameDelta);
+ // Make sure we can't decode these frames.
+ EXPECT_FALSE(DecodeCompleteFrame());
+ // This frame will make the jitter buffer recycle frames until a key frame.
+ // Since none is found it will have to wait until the next key frame before
+ // decoding.
+ InsertFrame(kVideoFrameDelta);
+ EXPECT_FALSE(DecodeCompleteFrame());
+}
+
+TEST_F(TestRunningJitterBuffer, TestEmptyPackets) {
+ // Make sure a frame can get complete even though empty packets are missing.
+ stream_generator->GenerateFrame(kVideoFrameKey, 3, 3,
+ clock_->MillisecondTimestamp());
+ EXPECT_EQ(kFirstPacket, InsertPacketAndPop(4));
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(4));
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+}
+
+TEST_F(TestJitterBufferNack, TestEmptyPackets) {
+ // Make sure empty packets doesn't clog the jitter buffer.
+ jitter_buffer_->SetNackMode(kNackHybrid, kLowRttNackMs, -1);
+ InsertFrames(kMaxNumberOfFrames, kFrameEmpty);
+ InsertFrame(kVideoFrameKey);
+ EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, TestNackListFull) {
+ // Insert a key frame and decode it.
+ InsertFrame(kVideoFrameKey);
+ EXPECT_TRUE(DecodeCompleteFrame());
+
+ // Generate and drop |kNackHistoryLength| packets to fill the NACK list.
+ DropFrame(kNackHistoryLength);
+ // Insert a frame which should trigger a recycle until the next key frame.
+ InsertFrame(kVideoFrameDelta);
+ EXPECT_FALSE(DecodeCompleteFrame());
+
+ uint16_t nack_list_length = kNackHistoryLength;
+ bool extended;
+ uint16_t* nack_list = jitter_buffer_->GetNackList(nack_list_length, extended);
+ // Verify that the jitter buffer requests a key frame.
+ EXPECT_TRUE(nack_list_length == 0xffff && nack_list == NULL);
+
+ InsertFrame(kVideoFrameDelta);
+ EXPECT_FALSE(DecodeCompleteFrame());
+ EXPECT_FALSE(DecodeFrame());
+}
+
+TEST_F(TestJitterBufferNack, TestNackBeforeDecode) {
+ DropFrame(10);
+ // Insert a frame and try to generate a NACK list. Shouldn't get one.
+ InsertFrame(kVideoFrameDelta);
+ uint16_t nack_list_size = 0;
+ bool extended = false;
+ uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
+ // No list generated, and a key frame request is signaled.
+ EXPECT_TRUE(list == NULL);
+ EXPECT_EQ(0xFFFF, nack_list_size);
+}
+
+TEST_F(TestJitterBufferNack, TestNormalOperation) {
+ EXPECT_EQ(kNackInfinite, jitter_buffer_->GetNackMode());
+
+ InsertFrame(kVideoFrameKey);
+ EXPECT_TRUE(DecodeFrame());
+
+ // ----------------------------------------------------------------
+ // | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
+ // ----------------------------------------------------------------
+ stream_generator->GenerateFrame(kVideoFrameKey, 100, 0,
+ clock_->MillisecondTimestamp());
+ clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
+ // Verify that the frame is incomplete.
+ EXPECT_FALSE(DecodeCompleteFrame());
+ while (stream_generator->PacketsRemaining() > 1) {
+ if (stream_generator->NextSequenceNumber() % 10 != 0)
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ else
+ stream_generator->NextPacket(NULL); // Drop packet
+ }
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ EXPECT_EQ(0, stream_generator->PacketsRemaining());
+ EXPECT_FALSE(DecodeCompleteFrame());
+ EXPECT_FALSE(DecodeFrame());
+ uint16_t nack_list_size = 0;
+ bool extended = false;
+ uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
+ // Verify the NACK list.
+ const int kExpectedNackSize = 9;
+ ASSERT_EQ(kExpectedNackSize, nack_list_size);
+ for (int i = 0; i < nack_list_size; ++i)
+ EXPECT_EQ((1 + i) * 10, list[i]);
+}
+
+TEST_F(TestJitterBufferNack, TestNormalOperationWrap) {
+ // ------- ------------------------------------------------------------
+ // | 65532 | | 65533 | 65534 | 65535 | x | 1 | .. | 9 | x | 11 |.....| 96 |
+ // ------- ------------------------------------------------------------
+ stream_generator->Init(65532, 0, clock_->MillisecondTimestamp());
+ InsertFrame(kVideoFrameKey);
+ EXPECT_TRUE(DecodeCompleteFrame());
+ stream_generator->GenerateFrame(kVideoFrameDelta, 100, 0,
+ clock_->MillisecondTimestamp());
+ EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
+ while (stream_generator->PacketsRemaining() > 1) {
+ if (stream_generator->NextSequenceNumber() % 10 != 0)
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ else
+ stream_generator->NextPacket(NULL); // Drop packet
+ }
+ EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+ EXPECT_EQ(0, stream_generator->PacketsRemaining());
+ EXPECT_FALSE(DecodeCompleteFrame());
+ EXPECT_FALSE(DecodeCompleteFrame());
+ uint16_t nack_list_size = 0;
+ bool extended = false;
+ uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
+ // Verify the NACK list.
+ const int kExpectedNackSize = 10;
+ ASSERT_EQ(kExpectedNackSize, nack_list_size);
+ for (int i = 0; i < nack_list_size; ++i)
+ EXPECT_EQ(i * 10, list[i]);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_estimator.cc b/trunk/src/modules/video_coding/main/source/jitter_estimator.cc
new file mode 100644
index 0000000..93c6ccb
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_estimator.cc
@@ -0,0 +1,422 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "jitter_estimator.h"
+#include "rtt_filter.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+namespace webrtc {
+
+VCMJitterEstimator::VCMJitterEstimator(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId) :
+_vcmId(vcmId),
+_receiverId(receiverId),
+_phi(0.97),
+_psi(0.9999),
+_alphaCountMax(400),
+_beta(0.9994),
+_thetaLow(0.000001),
+_nackLimit(3),
+_numStdDevDelayOutlier(15),
+_numStdDevFrameSizeOutlier(3),
+_noiseStdDevs(2.33), // ~Less than 1% chance
+ // (look up in normal distribution table)...
+_noiseStdDevOffset(30.0), // ...of getting 30 ms freezes
+_rttFilter(vcmId, receiverId)
+{
+ Reset();
+}
+
+VCMJitterEstimator&
+VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs)
+{
+ if (this != &rhs)
+ {
+ memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
+ memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
+
+ _vcmId = rhs._vcmId;
+ _receiverId = rhs._receiverId;
+ _avgFrameSize = rhs._avgFrameSize;
+ _varFrameSize = rhs._varFrameSize;
+ _maxFrameSize = rhs._maxFrameSize;
+ _fsSum = rhs._fsSum;
+ _fsCount = rhs._fsCount;
+ _lastUpdateT = rhs._lastUpdateT;
+ _prevEstimate = rhs._prevEstimate;
+ _prevFrameSize = rhs._prevFrameSize;
+ _avgNoise = rhs._avgNoise;
+ _alphaCount = rhs._alphaCount;
+ _filterJitterEstimate = rhs._filterJitterEstimate;
+ _startupCount = rhs._startupCount;
+ _latestNackTimestamp = rhs._latestNackTimestamp;
+ _nackCount = rhs._nackCount;
+ _rttFilter = rhs._rttFilter;
+ }
+ return *this;
+}
+
+// Resets the JitterEstimate
+void
+VCMJitterEstimator::Reset()
+{
+ _theta[0] = 1/(512e3/8);
+ _theta[1] = 0;
+ _varNoise = 4.0;
+
+ _thetaCov[0][0] = 1e-4;
+ _thetaCov[1][1] = 1e2;
+ _thetaCov[0][1] = _thetaCov[1][0] = 0;
+ _Qcov[0][0] = 2.5e-10;
+ _Qcov[1][1] = 1e-10;
+ _Qcov[0][1] = _Qcov[1][0] = 0;
+ _avgFrameSize = 500;
+ _maxFrameSize = 500;
+ _varFrameSize = 100;
+ _lastUpdateT = -1;
+ _prevEstimate = -1.0;
+ _prevFrameSize = 0;
+ _avgNoise = 0.0;
+ _alphaCount = 1;
+ _filterJitterEstimate = 0.0;
+ _latestNackTimestamp = 0;
+ _nackCount = 0;
+ _fsSum = 0;
+ _fsCount = 0;
+ _startupCount = 0;
+ _rttFilter.Reset();
+}
+
+void
+VCMJitterEstimator::ResetNackCount()
+{
+ _nackCount = 0;
+}
+
+// Updates the estimates with the new measurements
+void
+VCMJitterEstimator::UpdateEstimate(WebRtc_Word64 frameDelayMS, WebRtc_UWord32 frameSizeBytes,
+ bool incompleteFrame /* = false */)
+{
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Jitter estimate updated with: frameSize=%d frameDelayMS=%d",
+ frameSizeBytes, frameDelayMS);
+ if (frameSizeBytes == 0)
+ {
+ return;
+ }
+ int deltaFS = frameSizeBytes - _prevFrameSize;
+ if (_fsCount < kFsAccuStartupSamples)
+ {
+ _fsSum += frameSizeBytes;
+ _fsCount++;
+ }
+ else if (_fsCount == kFsAccuStartupSamples)
+ {
+ // Give the frame size filter
+ _avgFrameSize = static_cast<double>(_fsSum) /
+ static_cast<double>(_fsCount);
+ _fsCount++;
+ }
+ if (!incompleteFrame || frameSizeBytes > _avgFrameSize)
+ {
+ double avgFrameSize = _phi * _avgFrameSize +
+ (1 - _phi) * frameSizeBytes;
+ if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize))
+ {
+ // Only update the average frame size if this sample wasn't a
+ // key frame
+ _avgFrameSize = avgFrameSize;
+ }
+ // Update the variance anyway since we want to capture cases where we only get
+ // key frames.
+ _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) *
+ (frameSizeBytes - avgFrameSize) *
+ (frameSizeBytes - avgFrameSize), 1.0);
+ }
+
+ // Update max frameSize estimate
+ _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
+
+ if (_prevFrameSize == 0)
+ {
+ _prevFrameSize = frameSizeBytes;
+ return;
+ }
+ _prevFrameSize = frameSizeBytes;
+
+ // Only update the Kalman filter if the sample is not considered
+ // an extreme outlier. Even if it is an extreme outlier from a
+ // delay point of view, if the frame size also is large the
+ // deviation is probably due to an incorrect line slope.
+ double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
+
+ if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
+ frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
+ {
+ // Update the variance of the deviation from the
+ // line given by the Kalman filter
+ EstimateRandomJitter(deviation, incompleteFrame);
+ // Prevent updating with frames which have been congested by a large
+ // frame, and therefore arrives almost at the same time as that frame.
+ // This can occur when we receive a large frame (key frame) which
+ // has been delayed. The next frame is of normal size (delta frame),
+ // and thus deltaFS will be << 0. This removes all frame samples
+ // which arrives after a key frame.
+ if ((!incompleteFrame || deviation >= 0.0) &&
+ static_cast<double>(deltaFS) > - 0.25 * _maxFrameSize)
+ {
+ // Update the Kalman filter with the new data
+ KalmanEstimateChannel(frameDelayMS, deltaFS);
+ }
+ }
+ else
+ {
+ int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
+ EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
+ }
+ // Post process the total estimated jitter
+ if (_startupCount >= kStartupDelaySamples)
+ {
+ PostProcessEstimate();
+ }
+ else
+ {
+ _startupCount++;
+ }
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Framesize statistics: max=%f average=%f", _maxFrameSize, _avgFrameSize);
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "The estimated slope is: theta=(%f, %f)", _theta[0], _theta[1]);
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Random jitter: mean=%f variance=%f", _avgNoise, _varNoise);
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Current jitter estimate: %f", _filterJitterEstimate);
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Current max RTT: %u", _rttFilter.RttMs());
+}
+
+// Updates the nack/packet ratio
+void
+VCMJitterEstimator::FrameNacked()
+{
+ // Wait until _nackLimit retransmissions has been received,
+ // then always add ~1 RTT delay.
+ // TODO(holmer): Should we ever remove the additional delay if the
+ // the packet losses seem to have stopped? We could for instance scale
+ // the number of RTTs to add with the amount of retransmissions in a given
+ // time interval, or similar.
+ if (_nackCount < _nackLimit)
+ {
+ _nackCount++;
+ }
+}
+
+// Updates Kalman estimate of the channel
+// The caller is expected to sanity check the inputs.
+void
+VCMJitterEstimator::KalmanEstimateChannel(WebRtc_Word64 frameDelayMS,
+ WebRtc_Word32 deltaFSBytes)
+{
+ double Mh[2];
+ double hMh_sigma;
+ double kalmanGain[2];
+ double measureRes;
+ double t00, t01;
+
+ // Kalman filtering
+
+ // Prediction
+ // M = M + Q
+ _thetaCov[0][0] += _Qcov[0][0];
+ _thetaCov[0][1] += _Qcov[0][1];
+ _thetaCov[1][0] += _Qcov[1][0];
+ _thetaCov[1][1] += _Qcov[1][1];
+
+ // Kalman gain
+ // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
+ // h = [dFS 1]
+ // Mh = M*h'
+ // hMh_sigma = h*M*h' + R
+ Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
+ Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
+ // sigma weights measurements with a small deltaFS as noisy and
+ // measurements with large deltaFS as good
+ if (_maxFrameSize < 1.0)
+ {
+ return;
+ }
+ double sigma = (300.0 * exp(-abs(static_cast<double>(deltaFSBytes)) /
+ (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
+ if (sigma < 1.0)
+ {
+ sigma = 1.0;
+ }
+ hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
+ if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0))
+ {
+ assert(false);
+ return;
+ }
+ kalmanGain[0] = Mh[0] / hMh_sigma;
+ kalmanGain[1] = Mh[1] / hMh_sigma;
+
+ // Correction
+ // theta = theta + K*(dT - h*theta)
+ measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
+ _theta[0] += kalmanGain[0] * measureRes;
+ _theta[1] += kalmanGain[1] * measureRes;
+
+ if (_theta[0] < _thetaLow)
+ {
+ _theta[0] = _thetaLow;
+ }
+
+ // M = (I - K*h)*M
+ t00 = _thetaCov[0][0];
+ t01 = _thetaCov[0][1];
+ _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
+ kalmanGain[0] * _thetaCov[1][0];
+ _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
+ kalmanGain[0] * _thetaCov[1][1];
+ _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
+ kalmanGain[1] * deltaFSBytes * t00;
+ _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
+ kalmanGain[1] * deltaFSBytes * t01;
+
+ // Covariance matrix, must be positive semi-definite
+ assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
+ _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 &&
+ _thetaCov[0][0] >= 0);
+}
+
+// Calculate difference in delay between a sample and the
+// expected delay estimated by the Kalman filter
+double
+VCMJitterEstimator::DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
+ WebRtc_Word32 deltaFSBytes) const
+{
+ return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
+}
+
+// Estimates the random jitter by calculating the variance of the
+// sample distance from the line given by theta.
+void
+VCMJitterEstimator::EstimateRandomJitter(double d_dT, bool incompleteFrame)
+{
+ double alpha;
+ if (_alphaCount == 0)
+ {
+ assert(_alphaCount > 0);
+ return;
+ }
+ alpha = static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
+ _alphaCount++;
+ if (_alphaCount > _alphaCountMax)
+ {
+ _alphaCount = _alphaCountMax;
+ }
+ double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
+ double varNoise = alpha * _varNoise +
+ (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
+ if (!incompleteFrame || varNoise > _varNoise)
+ {
+ _avgNoise = avgNoise;
+ _varNoise = varNoise;
+ }
+ if (_varNoise < 1.0)
+ {
+ // The variance should never be zero, since we might get
+ // stuck and consider all samples as outliers.
+ _varNoise = 1.0;
+ }
+}
+
+double
+VCMJitterEstimator::NoiseThreshold() const
+{
+ double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
+ if (noiseThreshold < 1.0)
+ {
+ noiseThreshold = 1.0;
+ }
+ return noiseThreshold;
+}
+
+// Calculates the current jitter estimate from the filtered estimates
+double
+VCMJitterEstimator::CalculateEstimate()
+{
+ double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
+
+ // A very low estimate (or negative) is neglected
+ if (ret < 1.0) {
+ if (_prevEstimate <= 0.01)
+ {
+ ret = 1.0;
+ }
+ else
+ {
+ ret = _prevEstimate;
+ }
+ }
+ if (ret > 10000.0) // Sanity
+ {
+ ret = 10000.0;
+ }
+ _prevEstimate = ret;
+ return ret;
+}
+
+void
+VCMJitterEstimator::PostProcessEstimate()
+{
+ _filterJitterEstimate = CalculateEstimate();
+}
+
+void
+VCMJitterEstimator::UpdateRtt(WebRtc_UWord32 rttMs)
+{
+ _rttFilter.Update(rttMs);
+}
+
+void
+VCMJitterEstimator::UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes)
+{
+ if (_maxFrameSize < frameSizeBytes)
+ {
+ _maxFrameSize = frameSizeBytes;
+ }
+}
+
+// Returns the current filtered estimate if available,
+// otherwise tries to calculate an estimate.
+double
+VCMJitterEstimator::GetJitterEstimate(double rttMultiplier)
+{
+ double jitterMS = CalculateEstimate();
+ if (_filterJitterEstimate > jitterMS)
+ {
+ jitterMS = _filterJitterEstimate;
+ }
+ if (_nackCount >= _nackLimit)
+ {
+ return jitterMS + _rttFilter.RttMs() * rttMultiplier;
+ }
+ return jitterMS;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/jitter_estimator.h b/trunk/src/modules/video_coding/main/source/jitter_estimator.h
new file mode 100644
index 0000000..6fc4703
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/jitter_estimator.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+
+#include "typedefs.h"
+#include "rtt_filter.h"
+
+namespace webrtc
+{
+
+class VCMJitterEstimator
+{
+public:
+ VCMJitterEstimator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
+
+ VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
+
+ // Resets the estimate to the initial state
+ void Reset();
+ void ResetNackCount();
+
+ // Updates the jitter estimate with the new data.
+ //
+ // Input:
+ // - frameDelay : Delay-delta calculated by UTILDelayEstimate in milliseconds
+ // - frameSize : Frame size of the current frame.
+ // - incompleteFrame : Flags if the frame is used to update the estimate before it
+ // was complete. Default is false.
+ void UpdateEstimate(WebRtc_Word64 frameDelayMS,
+ WebRtc_UWord32 frameSizeBytes,
+ bool incompleteFrame = false);
+
+ // Returns the current jitter estimate in milliseconds and adds
+ // also adds an RTT dependent term in cases of retransmission.
+ // Input:
+ // - rttMultiplier : RTT param multiplier (when applicable).
+ //
+ // Return value : Jitter estimate in milliseconds
+ double GetJitterEstimate(double rttMultiplier);
+
+ // Updates the nack counter.
+ void FrameNacked();
+
+ // Updates the RTT filter.
+ //
+ // Input:
+ // - rttMs : RTT in ms
+ void UpdateRtt(WebRtc_UWord32 rttMs);
+
+ void UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes);
+
+ // A constant describing the delay from the jitter buffer
+ // to the delay on the receiving side which is not accounted
+ // for by the jitter buffer nor the decoding delay estimate.
+ static const WebRtc_UWord32 OPERATING_SYSTEM_JITTER = 10;
+
+protected:
+ // These are protected for better testing possibilities
+ double _theta[2]; // Estimated line parameters (slope, offset)
+ double _varNoise; // Variance of the time-deviation from the line
+
+private:
+ // Updates the Kalman filter for the line describing
+ // the frame size dependent jitter.
+ //
+ // Input:
+ // - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
+ // - deltaFSBytes : Frame size delta, i.e.
+ // : frame size at time T minus frame size at time T-1
+ void KalmanEstimateChannel(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes);
+
+ // Updates the random jitter estimate, i.e. the variance
+ // of the time deviations from the line given by the Kalman filter.
+ //
+ // Input:
+ // - d_dT : The deviation from the kalman estimate
+ // - incompleteFrame : True if the frame used to update the estimate
+ // with was incomplete
+ void EstimateRandomJitter(double d_dT, bool incompleteFrame);
+
+ double NoiseThreshold() const;
+
+ // Calculates the current jitter estimate.
+ //
+ // Return value : The current jitter estimate in milliseconds
+ double CalculateEstimate();
+
+ // Post process the calculated estimate
+ void PostProcessEstimate();
+
+ // Calculates the difference in delay between a sample and the
+ // expected delay estimated by the Kalman filter.
+ //
+ // Input:
+ // - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
+ // - deltaFS : Frame size delta, i.e. frame size at time
+ // T minus frame size at time T-1
+ //
+ // Return value : The difference in milliseconds
+ double DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
+ WebRtc_Word32 deltaFSBytes) const;
+
+ // Constants, filter parameters
+ WebRtc_Word32 _vcmId;
+ WebRtc_Word32 _receiverId;
+ const double _phi;
+ const double _psi;
+ const WebRtc_UWord32 _alphaCountMax;
+ const double _beta;
+ const double _thetaLow;
+ const WebRtc_UWord32 _nackLimit;
+ const WebRtc_Word32 _numStdDevDelayOutlier;
+ const WebRtc_Word32 _numStdDevFrameSizeOutlier;
+ const double _noiseStdDevs;
+ const double _noiseStdDevOffset;
+
+ double _thetaCov[2][2]; // Estimate covariance
+ double _Qcov[2][2]; // Process noise covariance
+ double _avgFrameSize; // Average frame size
+ double _varFrameSize; // Frame size variance
+ double _maxFrameSize; // Largest frame size received (descending
+ // with a factor _psi)
+ WebRtc_UWord32 _fsSum;
+ WebRtc_UWord32 _fsCount;
+
+ WebRtc_Word64 _lastUpdateT;
+ double _prevEstimate; // The previously returned jitter estimate
+ WebRtc_UWord32 _prevFrameSize; // Frame size of the previous frame
+ double _avgNoise; // Average of the random jitter
+ WebRtc_UWord32 _alphaCount;
+ double _filterJitterEstimate; // The filtered sum of jitter estimates
+
+ WebRtc_UWord32 _startupCount;
+
+ WebRtc_Word64 _latestNackTimestamp; // Timestamp in ms when the latest nack was seen
+ WebRtc_UWord32 _nackCount; // Keeps track of the number of nacks received,
+ // but never goes above _nackLimit
+ VCMRttFilter _rttFilter;
+
+ enum { kStartupDelaySamples = 30 };
+ enum { kFsAccuStartupSamples = 5 };
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/media_opt_util.cc b/trunk/src/modules/video_coding/main/source/media_opt_util.cc
new file mode 100644
index 0000000..b520278
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/media_opt_util.cc
@@ -0,0 +1,893 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/media_opt_util.h"
+
+#include <math.h>
+#include <float.h>
+#include <limits.h>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+#include "modules/video_coding/main/source/er_tables_xor.h"
+#include "modules/video_coding/main/source/fec_tables_xor.h"
+#include "modules/video_coding/main/source/nack_fec_tables.h"
+
+namespace webrtc {
+
+VCMProtectionMethod::VCMProtectionMethod():
+_effectivePacketLoss(0),
+_protectionFactorK(0),
+_protectionFactorD(0),
+_residualPacketLossFec(0.0f),
+_scaleProtKey(2.0f),
+_maxPayloadSize(1460),
+_qmRobustness(new VCMQmRobustness()),
+_useUepProtectionK(false),
+_useUepProtectionD(true),
+_corrFecCost(1.0),
+_type(kNone),
+_efficiency(0)
+{
+ //
+}
+
+VCMProtectionMethod::~VCMProtectionMethod()
+{
+ delete _qmRobustness;
+}
+void
+VCMProtectionMethod::UpdateContentMetrics(const
+ VideoContentMetrics* contentMetrics)
+{
+ _qmRobustness->UpdateContent(contentMetrics);
+}
+
+VCMNackFecMethod::VCMNackFecMethod(int lowRttNackThresholdMs,
+ int highRttNackThresholdMs)
+ : VCMFecMethod(),
+ _lowRttNackMs(lowRttNackThresholdMs),
+ _highRttNackMs(highRttNackThresholdMs) {
+ assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
+ assert(highRttNackThresholdMs == -1 ||
+ lowRttNackThresholdMs <= highRttNackThresholdMs);
+ assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
+ _type = kNackFec;
+}
+
+VCMNackFecMethod::~VCMNackFecMethod()
+{
+ //
+}
+bool
+VCMNackFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
+{
+ // Hybrid Nack FEC has three operational modes:
+ // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
+ // (_protectionFactorD) to zero. -1 means no FEC.
+ // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
+ // -1 means always allow NACK.
+ // 3. Medium RTT values - Hybrid mode: We will only nack the
+ // residual following the decoding of the FEC (refer to JB logic). FEC
+ // delta protection factor will be adjusted based on the RTT.
+
+ // Otherwise: we count on FEC; if the RTT is below a threshold, then we
+ // nack the residual, based on a decision made in the JB.
+
+ // Compute the protection factors
+ VCMFecMethod::ProtectionFactor(parameters);
+ if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs)
+ {
+ _protectionFactorD = 0;
+ VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+ }
+
+ // When in Hybrid mode (RTT range), adjust FEC rates based on the
+ // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
+ else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
+ {
+ // TODO(mikhal): Disabling adjustment temporarily.
+ // WebRtc_UWord16 rttIndex = (WebRtc_UWord16) parameters->rtt;
+ float adjustRtt = 1.0f;// (float)VCMNackFecTable[rttIndex] / 100.0f;
+
+ // Adjust FEC with NACK on (for delta frame only)
+ // table depends on RTT relative to rttMax (NACK Threshold)
+ _protectionFactorD = static_cast<WebRtc_UWord8>
+ (adjustRtt *
+ static_cast<float>(_protectionFactorD));
+ // update FEC rates after applying adjustment
+ VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+ }
+
+ return true;
+}
+
+bool
+VCMNackFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
+{
+ // Set the effective packet loss for encoder (based on FEC code).
+ // Compute the effective packet loss and residual packet loss due to FEC.
+ VCMFecMethod::EffectivePacketLoss(parameters);
+ return true;
+}
+
+bool
+VCMNackFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+ ProtectionFactor(parameters);
+ EffectivePacketLoss(parameters);
+
+ // Efficiency computation is based on FEC and NACK
+
+ // Add FEC cost: ignore I frames for now
+ float fecRate = static_cast<float> (_protectionFactorD) / 255.0f;
+ _efficiency = parameters->bitRate * fecRate * _corrFecCost;
+
+ // Add NACK cost, when applicable
+ if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
+ {
+ // nackCost = (bitRate - nackCost) * (lossPr)
+ _efficiency += parameters->bitRate * _residualPacketLossFec /
+ (1.0f + _residualPacketLossFec);
+ }
+
+ // Protection/fec rates obtained above are defined relative to total number
+ // of packets (total rate: source + fec) FEC in RTP module assumes
+ // protection factor is defined relative to source number of packets so we
+ // should convert the factor to reduce mismatch between mediaOpt's rate and
+ // the actual one
+ _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
+ _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
+
+ return true;
+}
+
+VCMNackMethod::VCMNackMethod():
+VCMProtectionMethod()
+{
+ _type = kNack;
+}
+
+VCMNackMethod::~VCMNackMethod()
+{
+ //
+}
+
+bool
+VCMNackMethod::EffectivePacketLoss(const VCMProtectionParameters* parameter)
+{
+ // Effective Packet Loss, NA in current version.
+ _effectivePacketLoss = 0;
+ return true;
+}
+
+bool
+VCMNackMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+ // Compute the effective packet loss
+ EffectivePacketLoss(parameters);
+
+ // nackCost = (bitRate - nackCost) * (lossPr)
+ _efficiency = parameters->bitRate * parameters->lossPr /
+ (1.0f + parameters->lossPr);
+ return true;
+}
+
+VCMFecMethod::VCMFecMethod():
+VCMProtectionMethod()
+{
+ _type = kFec;
+}
+VCMFecMethod::~VCMFecMethod()
+{
+ //
+}
+
+WebRtc_UWord8
+VCMFecMethod::BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta,
+ WebRtc_UWord8 packetFrameKey) const
+{
+ WebRtc_UWord8 boostRateKey = 2;
+ // Default: ratio scales the FEC protection up for I frames
+ WebRtc_UWord8 ratio = 1;
+
+ if (packetFrameDelta > 0)
+ {
+ ratio = (WebRtc_Word8) (packetFrameKey / packetFrameDelta);
+ }
+ ratio = VCM_MAX(boostRateKey, ratio);
+
+ return ratio;
+}
+
+WebRtc_UWord8
+VCMFecMethod::ConvertFECRate(WebRtc_UWord8 codeRateRTP) const
+{
+ return static_cast<WebRtc_UWord8> (VCM_MIN(255,(0.5 + 255.0 * codeRateRTP /
+ (float)(255 - codeRateRTP))));
+}
+
+// Update FEC with protectionFactorD
+void
+VCMFecMethod::UpdateProtectionFactorD(WebRtc_UWord8 protectionFactorD)
+{
+ _protectionFactorD = protectionFactorD;
+}
+
+// Update FEC with protectionFactorK
+void
+VCMFecMethod::UpdateProtectionFactorK(WebRtc_UWord8 protectionFactorK)
+{
+ _protectionFactorK = protectionFactorK;
+}
+
+// AvgRecoveryFEC: computes the residual packet loss (RPL) function.
+// This is the average recovery from the FEC, assuming random packet loss model.
+// Computed off-line for a range of FEC code parameters and loss rates.
+float
+VCMFecMethod::AvgRecoveryFEC(const VCMProtectionParameters* parameters) const
+{
+ // Total (avg) bits available per frame: total rate over actual/sent frame
+ // rate units are kbits/frame
+ const WebRtc_UWord16 bitRatePerFrame = static_cast<WebRtc_UWord16>
+ (parameters->bitRate / (parameters->frameRate));
+
+ // Total (average) number of packets per frame (source and fec):
+ const WebRtc_UWord8 avgTotPackets = 1 + static_cast<WebRtc_UWord8>
+ (static_cast<float> (bitRatePerFrame * 1000.0) /
+ static_cast<float> (8.0 * _maxPayloadSize) + 0.5);
+
+ const float protectionFactor = static_cast<float>(_protectionFactorD) /
+ 255.0;
+
+ WebRtc_UWord8 fecPacketsPerFrame = static_cast<WebRtc_UWord8>
+ (0.5 + protectionFactor * avgTotPackets);
+
+ WebRtc_UWord8 sourcePacketsPerFrame = avgTotPackets - fecPacketsPerFrame;
+
+ if ( (fecPacketsPerFrame == 0) || (sourcePacketsPerFrame == 0) )
+ {
+ // No protection, or rate too low: so average recovery from FEC == 0.
+ return 0.0;
+ }
+
+ // Table defined up to kMaxNumPackets
+ if (sourcePacketsPerFrame > kMaxNumPackets)
+ {
+ sourcePacketsPerFrame = kMaxNumPackets;
+ }
+
+ // Table defined up to kMaxNumPackets
+ if (fecPacketsPerFrame > kMaxNumPackets)
+ {
+ fecPacketsPerFrame = kMaxNumPackets;
+ }
+
+ // Code index for tables: up to (kMaxNumPackets * kMaxNumPackets)
+ WebRtc_UWord16 codeIndexTable[kMaxNumPackets * kMaxNumPackets];
+ WebRtc_UWord16 k = 0;
+ for (WebRtc_UWord8 i = 1; i <= kMaxNumPackets; i++)
+ {
+ for (WebRtc_UWord8 j = 1; j <= i; j++)
+ {
+ codeIndexTable[(j - 1) * kMaxNumPackets + i - 1] = k;
+ k += 1;
+ }
+ }
+
+ WebRtc_UWord8 lossRate = static_cast<WebRtc_UWord8> (255.0 *
+ parameters->lossPr + 0.5f);
+
+ // Constrain lossRate to 50%: tables defined up to 50%
+ if (lossRate >= kPacketLossMax)
+ {
+ lossRate = kPacketLossMax - 1;
+ }
+
+ const WebRtc_UWord16 codeIndex = (fecPacketsPerFrame - 1) * kMaxNumPackets +
+ (sourcePacketsPerFrame - 1);
+
+ const WebRtc_UWord16 indexTable = codeIndexTable[codeIndex] * kPacketLossMax +
+ lossRate;
+
+ // Check on table index
+ assert(indexTable < kSizeAvgFECRecoveryXOR);
+ float avgFecRecov = static_cast<float>(kAvgFECRecoveryXOR[indexTable]);
+
+ return avgFecRecov;
+}
+
+bool
+VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
+{
+ // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
+
+ // No protection if (filtered) packetLoss is 0
+ WebRtc_UWord8 packetLoss = (WebRtc_UWord8) (255 * parameters->lossPr);
+ if (packetLoss == 0)
+ {
+ _protectionFactorK = 0;
+ _protectionFactorD = 0;
+ return true;
+ }
+
+ // Parameters for FEC setting:
+ // first partition size, thresholds, table pars, spatial resoln fac.
+
+ // First partition protection: ~ 20%
+ WebRtc_UWord8 firstPartitionProt = (WebRtc_UWord8) (255 * 0.20);
+
+ // Minimum protection level needed to generate one FEC packet for one
+ // source packet/frame (in RTP sender)
+ WebRtc_UWord8 minProtLevelFec = 85;
+
+ // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
+ // above which we allocate protection to cover at least first partition.
+ WebRtc_UWord8 lossThr = 0;
+ WebRtc_UWord8 packetNumThr = 1;
+
+ // Parameters for range of rate index of table.
+ const WebRtc_UWord8 ratePar1 = 5;
+ const WebRtc_UWord8 ratePar2 = 49;
+
+ // Spatial resolution size, relative to a reference size.
+ float spatialSizeToRef = static_cast<float>
+ (parameters->codecWidth * parameters->codecHeight) /
+ (static_cast<float>(704 * 576));
+ // resolnFac: This parameter will generally increase/decrease the FEC rate
+ // (for fixed bitRate and packetLoss) based on system size.
+ // Use a smaller exponent (< 1) to control/soften system size effect.
+ const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
+
+ const int bitRatePerFrame = BitsPerFrame(parameters);
+
+
+ // Average number of packets per frame (source and fec):
+ const WebRtc_UWord8 avgTotPackets = 1 + (WebRtc_UWord8)
+ ((float) bitRatePerFrame * 1000.0
+ / (float) (8.0 * _maxPayloadSize) + 0.5);
+
+ // FEC rate parameters: for P and I frame
+ WebRtc_UWord8 codeRateDelta = 0;
+ WebRtc_UWord8 codeRateKey = 0;
+
+ // Get index for table: the FEC protection depends on an effective rate.
+ // The range on the rate index corresponds to rates (bps)
+ // from ~200k to ~8000k, for 30fps
+ const WebRtc_UWord16 effRateFecTable = static_cast<WebRtc_UWord16>
+ (resolnFac * bitRatePerFrame);
+ WebRtc_UWord8 rateIndexTable =
+ (WebRtc_UWord8) VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) /
+ ratePar1, ratePar2), 0);
+
+ // Restrict packet loss range to 50:
+ // current tables defined only up to 50%
+ if (packetLoss >= kPacketLossMax)
+ {
+ packetLoss = kPacketLossMax - 1;
+ }
+ WebRtc_UWord16 indexTable = rateIndexTable * kPacketLossMax + packetLoss;
+
+ // Check on table index
+ assert(indexTable < kSizeCodeRateXORTable);
+
+ // Protection factor for P frame
+ codeRateDelta = kCodeRateXORTable[indexTable];
+
+ if (packetLoss > lossThr && avgTotPackets > packetNumThr)
+ {
+ // Set a minimum based on first partition size.
+ if (codeRateDelta < firstPartitionProt)
+ {
+ codeRateDelta = firstPartitionProt;
+ }
+ }
+
+ // Check limit on amount of protection for P frame; 50% is max.
+ if (codeRateDelta >= kPacketLossMax)
+ {
+ codeRateDelta = kPacketLossMax - 1;
+ }
+
+ float adjustFec = 1.0f;
+ // Avoid additional adjustments when layers are active.
+ // TODO(mikhal/marco): Update adjusmtent based on layer info.
+ if (parameters->numLayers == 1)
+ {
+ adjustFec = _qmRobustness->AdjustFecFactor(codeRateDelta,
+ parameters->bitRate,
+ parameters->frameRate,
+ parameters->rtt,
+ packetLoss);
+ }
+
+ codeRateDelta = static_cast<WebRtc_UWord8>(codeRateDelta * adjustFec);
+
+ // For Key frame:
+ // Effectively at a higher rate, so we scale/boost the rate
+ // The boost factor may depend on several factors: ratio of packet
+ // number of I to P frames, how much protection placed on P frames, etc.
+ const WebRtc_UWord8 packetFrameDelta = (WebRtc_UWord8)
+ (0.5 + parameters->packetsPerFrame);
+ const WebRtc_UWord8 packetFrameKey = (WebRtc_UWord8)
+ (0.5 + parameters->packetsPerFrameKey);
+ const WebRtc_UWord8 boostKey = BoostCodeRateKey(packetFrameDelta,
+ packetFrameKey);
+
+ rateIndexTable = (WebRtc_UWord8) VCM_MAX(VCM_MIN(
+ 1 + (boostKey * effRateFecTable - ratePar1) /
+ ratePar1,ratePar2),0);
+ WebRtc_UWord16 indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
+
+ indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
+
+ // Check on table index
+ assert(indexTableKey < kSizeCodeRateXORTable);
+
+ // Protection factor for I frame
+ codeRateKey = kCodeRateXORTable[indexTableKey];
+
+ // Boosting for Key frame.
+ int boostKeyProt = _scaleProtKey * codeRateDelta;
+ if (boostKeyProt >= kPacketLossMax)
+ {
+ boostKeyProt = kPacketLossMax - 1;
+ }
+
+ // Make sure I frame protection is at least larger than P frame protection,
+ // and at least as high as filtered packet loss.
+ codeRateKey = static_cast<WebRtc_UWord8> (VCM_MAX(packetLoss,
+ VCM_MAX(boostKeyProt, codeRateKey)));
+
+ // Check limit on amount of protection for I frame: 50% is max.
+ if (codeRateKey >= kPacketLossMax)
+ {
+ codeRateKey = kPacketLossMax - 1;
+ }
+
+ _protectionFactorK = codeRateKey;
+ _protectionFactorD = codeRateDelta;
+
+ // Generally there is a rate mis-match between the FEC cost estimated
+ // in mediaOpt and the actual FEC cost sent out in RTP module.
+ // This is more significant at low rates (small # of source packets), where
+ // the granularity of the FEC decreases. In this case, non-zero protection
+ // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
+ // is based on rounding off protectionFactor on actual source packet number).
+ // The correction factor (_corrFecCost) attempts to corrects this, at least
+ // for cases of low rates (small #packets) and low protection levels.
+
+ float numPacketsFl = 1.0f + ((float) bitRatePerFrame * 1000.0
+ / (float) (8.0 * _maxPayloadSize) + 0.5);
+
+ const float estNumFecGen = 0.5f + static_cast<float> (_protectionFactorD *
+ numPacketsFl / 255.0f);
+
+
+ // We reduce cost factor (which will reduce overhead for FEC and
+ // hybrid method) and not the protectionFactor.
+ _corrFecCost = 1.0f;
+ if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec)
+ {
+ _corrFecCost = 0.5f;
+ }
+ if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec)
+ {
+ _corrFecCost = 0.0f;
+ }
+
+ // TODO (marpan): Set the UEP protection on/off for Key and Delta frames
+ _useUepProtectionK = _qmRobustness->SetUepProtection(codeRateKey,
+ parameters->bitRate,
+ packetLoss,
+ 0);
+
+ _useUepProtectionD = _qmRobustness->SetUepProtection(codeRateDelta,
+ parameters->bitRate,
+ packetLoss,
+ 1);
+
+ // DONE WITH FEC PROTECTION SETTINGS
+ return true;
+}
+
+int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
+ // When temporal layers are available FEC will only be applied on the base
+ // layer.
+ const float bitRateRatio =
+ kVp8LayerRateAlloction[parameters->numLayers - 1][0];
+ float frameRateRatio = powf(1 / 2, parameters->numLayers - 1);
+ float bitRate = parameters->bitRate * bitRateRatio;
+ float frameRate = parameters->frameRate * frameRateRatio;
+
+ // TODO(mikhal): Update factor following testing.
+ float adjustmentFactor = 1;
+
+ // Average bits per frame (units of kbits)
+ return static_cast<int>(adjustmentFactor * bitRate / frameRate);
+}
+
+bool
+VCMFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
+{
+ // Effective packet loss to encoder is based on RPL (residual packet loss)
+ // this is a soft setting based on degree of FEC protection
+ // RPL = received/input packet loss - average_FEC_recovery
+ // note: received/input packet loss may be filtered based on FilteredLoss
+
+ // The packet loss:
+ WebRtc_UWord8 packetLoss = (WebRtc_UWord8) (255 * parameters->lossPr);
+
+ float avgFecRecov = AvgRecoveryFEC(parameters);
+
+ // Residual Packet Loss:
+ _residualPacketLossFec = (float) (packetLoss - avgFecRecov) / 255.0f;
+
+ // Effective Packet Loss, NA in current version.
+ _effectivePacketLoss = 0;
+
+ return true;
+}
+
+bool
+VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+ // Compute the protection factor
+ ProtectionFactor(parameters);
+
+ // Compute the effective packet loss
+ EffectivePacketLoss(parameters);
+
+ // Compute the bit cost
+ // Ignore key frames for now.
+ float fecRate = static_cast<float> (_protectionFactorD) / 255.0f;
+ if (fecRate >= 0.0f)
+ {
+ // use this formula if the fecRate (protection factor) is defined
+ // relative to number of source packets
+ // this is the case for the previous tables:
+ // _efficiency = parameters->bitRate * ( 1.0 - 1.0 / (1.0 + fecRate));
+
+ // in the new tables, the fecRate is defined relative to total number of
+ // packets (total rate), so overhead cost is:
+ _efficiency = parameters->bitRate * fecRate * _corrFecCost;
+ }
+ else
+ {
+ _efficiency = 0.0f;
+ }
+
+ // Protection/fec rates obtained above is defined relative to total number
+ // of packets (total rate: source+fec) FEC in RTP module assumes protection
+ // factor is defined relative to source number of packets so we should
+ // convert the factor to reduce mismatch between mediaOpt suggested rate and
+ // the actual rate
+ _protectionFactorK = ConvertFECRate(_protectionFactorK);
+ _protectionFactorD = ConvertFECRate(_protectionFactorD);
+
+ return true;
+}
+VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs):
+_selectedMethod(NULL),
+_currentParameters(),
+_rtt(0),
+_lossPr(0.0f),
+_bitRate(0.0f),
+_frameRate(0.0f),
+_keyFrameSize(0.0f),
+_fecRateKey(0),
+_fecRateDelta(0),
+_lastPrUpdateT(0),
+_lossPr255(0.9999f),
+_lossPrHistory(),
+_shortMaxLossPr255(0),
+_packetsPerFrame(0.9999f),
+_packetsPerFrameKey(0.9999f),
+_residualPacketLossFec(0),
+_boostRateKey(2),
+_codecWidth(0),
+_codecHeight(0),
+_numLayers(1)
+{
+ Reset(nowMs);
+}
+
+VCMLossProtectionLogic::~VCMLossProtectionLogic()
+{
+ Release();
+}
+
+bool
+VCMLossProtectionLogic::SetMethod(enum VCMProtectionMethodEnum newMethodType)
+{
+ if (_selectedMethod != NULL)
+ {
+ if (_selectedMethod->Type() == newMethodType)
+ {
+ // Nothing to update
+ return false;
+ }
+ // New method - delete existing one
+ delete _selectedMethod;
+ }
+ VCMProtectionMethod *newMethod = NULL;
+ switch (newMethodType)
+ {
+ case kNack:
+ {
+ newMethod = new VCMNackMethod();
+ break;
+ }
+ case kFec:
+ {
+ newMethod = new VCMFecMethod();
+ break;
+ }
+ case kNackFec:
+ {
+ // Default to always having NACK enabled for the hybrid mode.
+ newMethod = new VCMNackFecMethod(kLowRttNackMs, -1);
+ break;
+ }
+ default:
+ {
+ return false;
+ break;
+ }
+
+ }
+ _selectedMethod = newMethod;
+ return true;
+}
+bool
+VCMLossProtectionLogic::RemoveMethod(enum VCMProtectionMethodEnum method)
+{
+ if (_selectedMethod == NULL)
+ {
+ return false;
+ }
+ else if (_selectedMethod->Type() == method)
+ {
+ delete _selectedMethod;
+ _selectedMethod = NULL;
+ }
+ return true;
+}
+
+float
+VCMLossProtectionLogic::RequiredBitRate() const
+{
+ float RequiredBitRate = 0.0f;
+ if (_selectedMethod != NULL)
+ {
+ RequiredBitRate = _selectedMethod->RequiredBitRate();
+ }
+ return RequiredBitRate;
+}
+
+void
+VCMLossProtectionLogic::UpdateRtt(WebRtc_UWord32 rtt)
+{
+ _rtt = rtt;
+}
+
+void
+VCMLossProtectionLogic::UpdateResidualPacketLoss(float residualPacketLoss)
+{
+ _residualPacketLossFec = residualPacketLoss;
+}
+
+void
+VCMLossProtectionLogic::UpdateMaxLossHistory(WebRtc_UWord8 lossPr255,
+ WebRtc_Word64 now)
+{
+ if (_lossPrHistory[0].timeMs >= 0 &&
+ now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs)
+ {
+ if (lossPr255 > _shortMaxLossPr255)
+ {
+ _shortMaxLossPr255 = lossPr255;
+ }
+ }
+ else
+ {
+ // Only add a new value to the history once a second
+ if (_lossPrHistory[0].timeMs == -1)
+ {
+ // First, no shift
+ _shortMaxLossPr255 = lossPr255;
+ }
+ else
+ {
+ // Shift
+ for (WebRtc_Word32 i = (kLossPrHistorySize - 2); i >= 0; i--)
+ {
+ _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
+ _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
+ }
+ }
+ if (_shortMaxLossPr255 == 0)
+ {
+ _shortMaxLossPr255 = lossPr255;
+ }
+
+ _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
+ _lossPrHistory[0].timeMs = now;
+ _shortMaxLossPr255 = 0;
+ }
+}
+
+WebRtc_UWord8
+VCMLossProtectionLogic::MaxFilteredLossPr(WebRtc_Word64 nowMs) const
+{
+ WebRtc_UWord8 maxFound = _shortMaxLossPr255;
+ if (_lossPrHistory[0].timeMs == -1)
+ {
+ return maxFound;
+ }
+ for (WebRtc_Word32 i = 0; i < kLossPrHistorySize; i++)
+ {
+ if (_lossPrHistory[i].timeMs == -1)
+ {
+ break;
+ }
+ if (nowMs - _lossPrHistory[i].timeMs >
+ kLossPrHistorySize * kLossPrShortFilterWinMs)
+ {
+ // This sample (and all samples after this) is too old
+ break;
+ }
+ if (_lossPrHistory[i].lossPr255 > maxFound)
+ {
+ // This sample is the largest one this far into the history
+ maxFound = _lossPrHistory[i].lossPr255;
+ }
+ }
+ return maxFound;
+}
+
+WebRtc_UWord8 VCMLossProtectionLogic::FilteredLoss(
+ int64_t nowMs,
+ FilterPacketLossMode filter_mode,
+ WebRtc_UWord8 lossPr255) {
+
+ // Update the max window filter.
+ UpdateMaxLossHistory(lossPr255, nowMs);
+
+ // Update the recursive average filter.
+ _lossPr255.Apply(static_cast<float> (nowMs - _lastPrUpdateT),
+ static_cast<float> (lossPr255));
+ _lastPrUpdateT = nowMs;
+
+ // Filtered loss: default is received loss (no filtering).
+ WebRtc_UWord8 filtered_loss = lossPr255;
+
+ switch (filter_mode) {
+ case kNoFilter:
+ break;
+ case kAvgFilter:
+ filtered_loss = static_cast<WebRtc_UWord8> (_lossPr255.Value() + 0.5);
+ break;
+ case kMaxFilter:
+ filtered_loss = MaxFilteredLossPr(nowMs);
+ break;
+ }
+
+ return filtered_loss;
+}
+
+void
+VCMLossProtectionLogic::UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc)
+{
+ _lossPr = (float) packetLossEnc / (float) 255.0;
+}
+
+void
+VCMLossProtectionLogic::UpdateBitRate(float bitRate)
+{
+ _bitRate = bitRate;
+}
+
+void
+VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets, int64_t nowMs)
+{
+ _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
+ nPackets);
+ _lastPacketPerFrameUpdateT = nowMs;
+}
+
+void
+VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs)
+{
+ _packetsPerFrameKey.Apply(static_cast<float>(nowMs -
+ _lastPacketPerFrameUpdateTKey), nPackets);
+ _lastPacketPerFrameUpdateTKey = nowMs;
+}
+
+void
+VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize)
+{
+ _keyFrameSize = keyFrameSize;
+}
+
+void
+VCMLossProtectionLogic::UpdateFrameSize(WebRtc_UWord16 width,
+ WebRtc_UWord16 height)
+{
+ _codecWidth = width;
+ _codecHeight = height;
+}
+
+void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
+ _numLayers = (numLayers == 0) ? 1 : numLayers;
+}
+
+bool
+VCMLossProtectionLogic::UpdateMethod()
+{
+ if (_selectedMethod == NULL)
+ {
+ return false;
+ }
+ _currentParameters.rtt = _rtt;
+ _currentParameters.lossPr = _lossPr;
+ _currentParameters.bitRate = _bitRate;
+ _currentParameters.frameRate = _frameRate; // rename actual frame rate?
+ _currentParameters.keyFrameSize = _keyFrameSize;
+ _currentParameters.fecRateDelta = _fecRateDelta;
+ _currentParameters.fecRateKey = _fecRateKey;
+ _currentParameters.packetsPerFrame = _packetsPerFrame.Value();
+ _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.Value();
+ _currentParameters.residualPacketLossFec = _residualPacketLossFec;
+ _currentParameters.codecWidth = _codecWidth;
+ _currentParameters.codecHeight = _codecHeight;
+ _currentParameters.numLayers = _numLayers;
+ return _selectedMethod->UpdateParameters(&_currentParameters);
+}
+
+VCMProtectionMethod*
+VCMLossProtectionLogic::SelectedMethod() const
+{
+ return _selectedMethod;
+}
+
+VCMProtectionMethodEnum
+VCMLossProtectionLogic::SelectedType() const
+{
+ return _selectedMethod->Type();
+}
+
+void
+VCMLossProtectionLogic::Reset(int64_t nowMs)
+{
+ _lastPrUpdateT = nowMs;
+ _lastPacketPerFrameUpdateT = nowMs;
+ _lastPacketPerFrameUpdateTKey = nowMs;
+ _lossPr255.Reset(0.9999f);
+ _packetsPerFrame.Reset(0.9999f);
+ _fecRateDelta = _fecRateKey = 0;
+ for (WebRtc_Word32 i = 0; i < kLossPrHistorySize; i++)
+ {
+ _lossPrHistory[i].lossPr255 = 0;
+ _lossPrHistory[i].timeMs = -1;
+ }
+ _shortMaxLossPr255 = 0;
+ Release();
+}
+
+void
+VCMLossProtectionLogic::Release()
+{
+ delete _selectedMethod;
+ _selectedMethod = NULL;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/media_opt_util.h b/trunk/src/modules/video_coding/main/source/media_opt_util.h
new file mode 100644
index 0000000..1f808de
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/media_opt_util.h
@@ -0,0 +1,375 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+
+#include "typedefs.h"
+#include "trace.h"
+#include "exp_filter.h"
+#include "internal_defines.h"
+#include "qm_select.h"
+
+#include <cmath>
+#include <cstdlib>
+
+
+namespace webrtc
+{
+// Number of time periods used for (max) window filter for packet loss
+// TODO (marpan): set reasonable window size for filtered packet loss,
+// adjustment should be based on logged/real data of loss stats/correlation.
+enum { kLossPrHistorySize = 10 };
+
+// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
+enum { kLossPrShortFilterWinMs = 1000 };
+
+// The type of filter used on the received packet loss reports.
+enum FilterPacketLossMode {
+ kNoFilter, // No filtering on received loss.
+ kAvgFilter, // Recursive average filter.
+ kMaxFilter // Max-window filter, over the time interval of:
+ // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
+};
+
+// Thresholds for hybrid NACK/FEC
+// common to media optimization and the jitter buffer.
+enum HybridNackTH {
+ kHighRttNackMs = 100,
+ kLowRttNackMs = 20
+};
+
+struct VCMProtectionParameters
+{
+ VCMProtectionParameters() : rtt(0), lossPr(0.0f), bitRate(0.0f),
+ packetsPerFrame(0.0f), packetsPerFrameKey(0.0f), frameRate(0.0f),
+ keyFrameSize(0.0f), fecRateDelta(0), fecRateKey(0),
+ residualPacketLossFec(0.0f), codecWidth(0), codecHeight(0),
+ numLayers(1)
+ {}
+
+ int rtt;
+ float lossPr;
+ float bitRate;
+ float packetsPerFrame;
+ float packetsPerFrameKey;
+ float frameRate;
+ float keyFrameSize;
+ WebRtc_UWord8 fecRateDelta;
+ WebRtc_UWord8 fecRateKey;
+ float residualPacketLossFec;
+ WebRtc_UWord16 codecWidth;
+ WebRtc_UWord16 codecHeight;
+ int numLayers;
+};
+
+
+/******************************/
+/* VCMProtectionMethod class */
+/****************************/
+
+enum VCMProtectionMethodEnum
+{
+ kNack,
+ kFec,
+ kNackFec,
+ kNone
+};
+
+class VCMLossProbabilitySample
+{
+public:
+ VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {};
+
+ WebRtc_UWord8 lossPr255;
+ WebRtc_Word64 timeMs;
+};
+
+
+class VCMProtectionMethod
+{
+public:
+ VCMProtectionMethod();
+ virtual ~VCMProtectionMethod();
+
+ // Updates the efficiency of the method using the parameters provided
+ //
+ // Input:
+ // - parameters : Parameters used to calculate efficiency
+ //
+ // Return value : True if this method is recommended in
+ // the given conditions.
+ virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
+
+ // Returns the protection type
+ //
+ // Return value : The protection type
+ enum VCMProtectionMethodEnum Type() const { return _type; }
+
+ // Returns the bit rate required by this protection method
+ // during these conditions.
+ //
+ // Return value : Required bit rate
+ virtual float RequiredBitRate() { return _efficiency; }
+
+ // Returns the effective packet loss for ER, required by this protection method
+ //
+ // Return value : Required effective packet loss
+ virtual WebRtc_UWord8 RequiredPacketLossER() { return _effectivePacketLoss; }
+
+ // Extracts the FEC protection factor for Key frame, required by this protection method
+ //
+ // Return value : Required protectionFactor for Key frame
+ virtual WebRtc_UWord8 RequiredProtectionFactorK() { return _protectionFactorK; }
+
+ // Extracts the FEC protection factor for Delta frame, required by this protection method
+ //
+ // Return value : Required protectionFactor for delta frame
+ virtual WebRtc_UWord8 RequiredProtectionFactorD() { return _protectionFactorD; }
+
+ // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
+ //
+ // Return value : Required Unequal protection on/off state.
+ virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
+
+ // Extracts whether the the FEC Unequal protection (UEP) is used for Delta frame.
+ //
+ // Return value : Required Unequal protection on/off state.
+ virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
+
+ // Updates content metrics
+ void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
+
+protected:
+
+ WebRtc_UWord8 _effectivePacketLoss;
+ WebRtc_UWord8 _protectionFactorK;
+ WebRtc_UWord8 _protectionFactorD;
+ // Estimation of residual loss after the FEC
+ float _residualPacketLossFec;
+ float _scaleProtKey;
+ WebRtc_Word32 _maxPayloadSize;
+
+ VCMQmRobustness* _qmRobustness;
+ bool _useUepProtectionK;
+ bool _useUepProtectionD;
+ float _corrFecCost;
+ enum VCMProtectionMethodEnum _type;
+ float _efficiency;
+};
+
+class VCMNackMethod : public VCMProtectionMethod
+{
+public:
+ VCMNackMethod();
+ virtual ~VCMNackMethod();
+ virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+ // Get the effective packet loss
+ bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
+};
+
+class VCMFecMethod : public VCMProtectionMethod
+{
+public:
+ VCMFecMethod();
+ virtual ~VCMFecMethod();
+ virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+ // Get the effective packet loss for ER
+ bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+ // Get the FEC protection factors
+ bool ProtectionFactor(const VCMProtectionParameters* parameters);
+ // Get the boost for key frame protection
+ WebRtc_UWord8 BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta,
+ WebRtc_UWord8 packetFrameKey) const;
+ // Convert the rates: defined relative to total# packets or source# packets
+ WebRtc_UWord8 ConvertFECRate(WebRtc_UWord8 codeRate) const;
+ // Get the average effective recovery from FEC: for random loss model
+ float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
+ // Update FEC with protectionFactorD
+ void UpdateProtectionFactorD(WebRtc_UWord8 protectionFactorD);
+ // Update FEC with protectionFactorK
+ void UpdateProtectionFactorK(WebRtc_UWord8 protectionFactorK);
+ // Compute the bits per frame. Account for temporal layers when applicable.
+ int BitsPerFrame(const VCMProtectionParameters* parameters);
+};
+
+
+class VCMNackFecMethod : public VCMFecMethod
+{
+public:
+ VCMNackFecMethod(int lowRttNackThresholdMs,
+ int highRttNackThresholdMs);
+ virtual ~VCMNackFecMethod();
+ virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+ // Get the effective packet loss for ER
+ bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+ // Get the protection factors
+ bool ProtectionFactor(const VCMProtectionParameters* parameters);
+
+private:
+ int _lowRttNackMs;
+ int _highRttNackMs;
+};
+
+class VCMLossProtectionLogic
+{
+public:
+ VCMLossProtectionLogic(int64_t nowMs);
+ ~VCMLossProtectionLogic();
+
+ // Set the protection method to be used
+ //
+ // Input:
+ // - newMethodType : New requested protection method type. If one
+ // is already set, it will be deleted and replaced
+ // Return value: Returns true on update
+ bool SetMethod(enum VCMProtectionMethodEnum newMethodType);
+
+ // Remove requested protection method
+ // Input:
+ // - method : method to be removed (if currently selected)
+ //
+ // Return value: Returns true on update
+ bool RemoveMethod(enum VCMProtectionMethodEnum method);
+
+ // Return required bit rate per selected protectin method
+ float RequiredBitRate() const;
+
+ // Update the round-trip time
+ //
+ // Input:
+ // - rtt : Round-trip time in seconds.
+ void UpdateRtt(WebRtc_UWord32 rtt);
+
+ // Update residual packet loss
+ //
+ // Input:
+ // - residualPacketLoss : residual packet loss:
+ // effective loss after FEC recovery
+ void UpdateResidualPacketLoss(float _residualPacketLoss);
+
+ // Update the filtered packet loss.
+ //
+ // Input:
+ // - packetLossEnc : The reported packet loss filtered
+ // (max window or average)
+ void UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc);
+
+ // Update the current target bit rate.
+ //
+ // Input:
+ // - bitRate : The current target bit rate in kbits/s
+ void UpdateBitRate(float bitRate);
+
+ // Update the number of packets per frame estimate, for delta frames
+ //
+ // Input:
+ // - nPackets : Number of packets in the latest sent frame.
+ void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
+
+ // Update the number of packets per frame estimate, for key frames
+ //
+ // Input:
+ // - nPackets : umber of packets in the latest sent frame.
+ void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
+
+ // Update the keyFrameSize estimate
+ //
+ // Input:
+ // - keyFrameSize : The size of the latest sent key frame.
+ void UpdateKeyFrameSize(float keyFrameSize);
+
+ // Update the frame rate
+ //
+ // Input:
+ // - frameRate : The current target frame rate.
+ void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
+
+ // Update the frame size
+ //
+ // Input:
+ // - width : The codec frame width.
+ // - height : The codec frame height.
+ void UpdateFrameSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
+
+ // Update the number of active layers
+ //
+ // Input:
+ // - numLayers : Number of layers used.
+ void UpdateNumLayers(int numLayers);
+
+ // The amount of packet loss to cover for with FEC.
+ //
+ // Input:
+ // - fecRateKey : Packet loss to cover for with FEC when
+ // sending key frames.
+ // - fecRateDelta : Packet loss to cover for with FEC when
+ // sending delta frames.
+ void UpdateFECRates(WebRtc_UWord8 fecRateKey, WebRtc_UWord8 fecRateDelta)
+ { _fecRateKey = fecRateKey;
+ _fecRateDelta = fecRateDelta; }
+
+ // Update the protection methods with the current VCMProtectionParameters
+ // and set the requested protection settings.
+ // Return value : Returns true on update
+ bool UpdateMethod();
+
+ // Returns the method currently selected.
+ //
+ // Return value : The protection method currently selected.
+ VCMProtectionMethod* SelectedMethod() const;
+
+ // Return the protection type of the currently selected method
+ VCMProtectionMethodEnum SelectedType() const;
+
+ // Updates the filtered loss for the average and max window packet loss,
+ // and returns the filtered loss probability in the interval [0, 255].
+ // The returned filtered loss value depends on the parameter |filter_mode|.
+ // The input parameter |lossPr255| is the received packet loss.
+
+ // Return value : The filtered loss probability
+ WebRtc_UWord8 FilteredLoss(int64_t nowMs, FilterPacketLossMode filter_mode,
+ WebRtc_UWord8 lossPr255);
+
+ void Reset(int64_t nowMs);
+
+ void Release();
+
+private:
+ // Sets the available loss protection methods.
+ void UpdateMaxLossHistory(WebRtc_UWord8 lossPr255, WebRtc_Word64 now);
+ WebRtc_UWord8 MaxFilteredLossPr(WebRtc_Word64 nowMs) const;
+ VCMProtectionMethod* _selectedMethod;
+ VCMProtectionParameters _currentParameters;
+ WebRtc_UWord32 _rtt;
+ float _lossPr;
+ float _bitRate;
+ float _frameRate;
+ float _keyFrameSize;
+ WebRtc_UWord8 _fecRateKey;
+ WebRtc_UWord8 _fecRateDelta;
+ WebRtc_Word64 _lastPrUpdateT;
+ WebRtc_Word64 _lastPacketPerFrameUpdateT;
+ WebRtc_Word64 _lastPacketPerFrameUpdateTKey;
+ VCMExpFilter _lossPr255;
+ VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
+ WebRtc_UWord8 _shortMaxLossPr255;
+ VCMExpFilter _packetsPerFrame;
+ VCMExpFilter _packetsPerFrameKey;
+ float _residualPacketLossFec;
+ WebRtc_UWord8 _boostRateKey;
+ WebRtc_UWord16 _codecWidth;
+ WebRtc_UWord16 _codecHeight;
+ int _numLayers;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/trunk/src/modules/video_coding/main/source/media_optimization.cc b/trunk/src/modules/video_coding/main/source/media_optimization.cc
new file mode 100644
index 0000000..cae4912
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/media_optimization.cc
@@ -0,0 +1,697 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "media_optimization.h"
+#include "content_metrics_processing.h"
+#include "frame_dropper.h"
+#include "qm_select.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+
+namespace webrtc {
+
+VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id,
+ TickTimeBase* clock):
+_id(id),
+_clock(clock),
+_maxBitRate(0),
+_sendCodecType(kVideoCodecUnknown),
+_codecWidth(0),
+_codecHeight(0),
+_initCodecWidth(0),
+_initCodecHeight(0),
+_userFrameRate(0),
+_packetLossEnc(0),
+_fractionLost(0),
+_sendStatisticsZeroEncode(0),
+_maxPayloadSize(1460),
+_targetBitRate(0),
+_incomingFrameRate(0),
+_enableQm(false),
+_videoProtectionCallback(NULL),
+_videoQMSettingsCallback(NULL),
+_encodedFrameSamples(),
+_avgSentBitRateBps(0.0f),
+_keyFrameCnt(0),
+_deltaFrameCnt(0),
+_lastQMUpdateTime(0),
+_lastChangeTime(0),
+_numLayers(0)
+{
+ memset(_sendStatistics, 0, sizeof(_sendStatistics));
+ memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
+
+ _frameDropper = new VCMFrameDropper(_id);
+ _lossProtLogic = new VCMLossProtectionLogic(_clock->MillisecondTimestamp());
+ _content = new VCMContentMetricsProcessing();
+ _qmResolution = new VCMQmResolution();
+}
+
+VCMMediaOptimization::~VCMMediaOptimization(void)
+{
+ _lossProtLogic->Release();
+ delete _lossProtLogic;
+ delete _frameDropper;
+ delete _content;
+ delete _qmResolution;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::Reset()
+{
+ memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
+ _incomingFrameRate = 0.0;
+ _frameDropper->Reset();
+ _lossProtLogic->Reset(_clock->MillisecondTimestamp());
+ _frameDropper->SetRates(0, 0);
+ _content->Reset();
+ _qmResolution->Reset();
+ _lossProtLogic->UpdateFrameRate(_incomingFrameRate);
+ _lossProtLogic->Reset(_clock->MillisecondTimestamp());
+ _sendStatisticsZeroEncode = 0;
+ _targetBitRate = 0;
+ _codecWidth = 0;
+ _codecHeight = 0;
+ _userFrameRate = 0;
+ _keyFrameCnt = 0;
+ _deltaFrameCnt = 0;
+ _lastQMUpdateTime = 0;
+ _lastChangeTime = 0;
+ for (WebRtc_Word32 i = 0; i < kBitrateMaxFrameSamples; i++)
+ {
+ _encodedFrameSamples[i]._sizeBytes = -1;
+ _encodedFrameSamples[i]._timeCompleteMs = -1;
+ }
+ _avgSentBitRateBps = 0.0f;
+ _numLayers = 1;
+ return VCM_OK;
+}
+
+WebRtc_UWord32
+VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
+ WebRtc_UWord8 &fractionLost,
+ WebRtc_UWord32 roundTripTimeMs)
+{
+ VCMProtectionMethod *selectedMethod = _lossProtLogic->SelectedMethod();
+ _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
+ _lossProtLogic->UpdateRtt(roundTripTimeMs);
+ _lossProtLogic->UpdateResidualPacketLoss(static_cast<float>(fractionLost));
+
+ // Get frame rate for encoder: this is the actual/sent frame rate
+ float actualFrameRate = SentFrameRate();
+
+ // sanity
+ if (actualFrameRate < 1.0)
+ {
+ actualFrameRate = 1.0;
+ }
+
+ // Update frame rate for the loss protection logic class: frame rate should
+ // be the actual/sent rate
+ _lossProtLogic->UpdateFrameRate(actualFrameRate);
+
+ _fractionLost = fractionLost;
+
+ // Returns the filtered packet loss, used for the protection setting.
+ // The filtered loss may be the received loss (no filter), or some
+ // filtered value (average or max window filter).
+ // Use max window filter for now.
+ FilterPacketLossMode filter_mode = kMaxFilter;
+ WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss(
+ _clock->MillisecondTimestamp(), filter_mode, fractionLost);
+
+ // For now use the filtered loss for computing the robustness settings
+ _lossProtLogic->UpdateFilteredLossPr(packetLossEnc);
+
+ // Rate cost of the protection methods
+ uint32_t protection_overhead_kbps = 0;
+
+ // Update protection settings, when applicable
+ float sent_video_rate = 0.0f;
+ if (selectedMethod)
+ {
+ // Update protection method with content metrics
+ selectedMethod->UpdateContentMetrics(_content->ShortTermAvgData());
+
+ // Update method will compute the robustness settings for the given
+ // protection method and the overhead cost
+ // the protection method is set by the user via SetVideoProtection.
+ _lossProtLogic->UpdateMethod();
+
+ // Update protection callback with protection settings.
+ uint32_t sent_video_rate_bps = 0;
+ uint32_t sent_nack_rate_bps = 0;
+ uint32_t sent_fec_rate_bps = 0;
+ // Get the bit cost of protection method, based on the amount of
+ // overhead data actually transmitted (including headers) the last
+ // second.
+ UpdateProtectionCallback(selectedMethod,
+ &sent_video_rate_bps,
+ &sent_nack_rate_bps,
+ &sent_fec_rate_bps);
+ uint32_t sent_total_rate_bps = sent_video_rate_bps +
+ sent_nack_rate_bps + sent_fec_rate_bps;
+ // Estimate the overhead costs of the next second as staying the same
+ // wrt the source bitrate.
+ if (sent_total_rate_bps > 0) {
+ protection_overhead_kbps = static_cast<uint32_t>(bitRate *
+ static_cast<double>(sent_nack_rate_bps + sent_fec_rate_bps) /
+ sent_total_rate_bps + 0.5);
+ }
+ // Cap the overhead estimate to 50%.
+ if (protection_overhead_kbps > bitRate / 2)
+ protection_overhead_kbps = bitRate / 2;
+
+ // Get the effective packet loss for encoder ER
+ // when applicable, should be passed to encoder via fractionLost
+ packetLossEnc = selectedMethod->RequiredPacketLossER();
+ sent_video_rate = static_cast<float>(sent_video_rate_bps / 1000.0);
+ }
+
+ // Source coding rate: total rate - protection overhead
+ _targetBitRate = bitRate - protection_overhead_kbps;
+
+ // Update encoding rates following protection settings
+ _frameDropper->SetRates(static_cast<float>(_targetBitRate), 0);
+
+ if (_enableQm && _numLayers == 1)
+ {
+ // Update QM with rates
+ _qmResolution->UpdateRates((float)_targetBitRate, sent_video_rate,
+ _incomingFrameRate, _fractionLost);
+ // Check for QM selection
+ bool selectQM = checkStatusForQMchange();
+ if (selectQM)
+ {
+ SelectQuality();
+ }
+ // Reset the short-term averaged content data.
+ _content->ResetShortTermAvgData();
+ }
+
+ return _targetBitRate;
+}
+
+int VCMMediaOptimization::UpdateProtectionCallback(
+ VCMProtectionMethod *selected_method,
+ uint32_t* video_rate_bps,
+ uint32_t* nack_overhead_rate_bps,
+ uint32_t* fec_overhead_rate_bps)
+{
+ if (!_videoProtectionCallback)
+ {
+ return VCM_OK;
+ }
+ // Get the FEC code rate for Key frames (set to 0 when NA)
+ const WebRtc_UWord8
+ codeRateKeyRTP = selected_method->RequiredProtectionFactorK();
+
+ // Get the FEC code rate for Delta frames (set to 0 when NA)
+ const WebRtc_UWord8
+ codeRateDeltaRTP = selected_method->RequiredProtectionFactorD();
+
+ // Get the FEC-UEP protection status for Key frames: UEP on/off
+ const bool
+ useUepProtectionKeyRTP = selected_method->RequiredUepProtectionK();
+
+ // Get the FEC-UEP protection status for Delta frames: UEP on/off
+ const bool
+ useUepProtectionDeltaRTP = selected_method->RequiredUepProtectionD();
+
+ // NACK is on for NACK and NackFec protection method: off for FEC method
+ bool nackStatus = (selected_method->Type() == kNackFec ||
+ selected_method->Type() == kNack);
+
+ // TODO(Marco): Pass FEC protection values per layer.
+
+ return _videoProtectionCallback->ProtectionRequest(codeRateDeltaRTP,
+ codeRateKeyRTP,
+ useUepProtectionDeltaRTP,
+ useUepProtectionKeyRTP,
+ nackStatus,
+ video_rate_bps,
+ nack_overhead_rate_bps,
+ fec_overhead_rate_bps);
+}
+
+bool
+VCMMediaOptimization::DropFrame()
+{
+ // leak appropriate number of bytes
+ _frameDropper->Leak((WebRtc_UWord32)(InputFrameRate() + 0.5f));
+ return _frameDropper->DropFrame();
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SentFrameCount(VCMFrameCount &frameCount) const
+{
+ frameCount.numDeltaFrames = _deltaFrameCnt;
+ frameCount.numKeyFrames = _keyFrameCnt;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SetEncodingData(VideoCodecType sendCodecType,
+ WebRtc_Word32 maxBitRate,
+ WebRtc_UWord32 frameRate,
+ WebRtc_UWord32 bitRate,
+ WebRtc_UWord16 width,
+ WebRtc_UWord16 height,
+ int numLayers)
+{
+ // Everything codec specific should be reset here since this means the codec
+ // has changed. If native dimension values have changed, then either user
+ // initiated change, or QM initiated change. Will be able to determine only
+ // after the processing of the first frame.
+ _lastChangeTime = _clock->MillisecondTimestamp();
+ _content->Reset();
+ _content->UpdateFrameRate(frameRate);
+
+ _maxBitRate = maxBitRate;
+ _sendCodecType = sendCodecType;
+ _targetBitRate = bitRate;
+ _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
+ _lossProtLogic->UpdateFrameRate(static_cast<float>(frameRate));
+ _lossProtLogic->UpdateFrameSize(width, height);
+ _lossProtLogic->UpdateNumLayers(numLayers);
+ _frameDropper->Reset();
+ _frameDropper->SetRates(static_cast<float>(bitRate),
+ static_cast<float>(frameRate));
+ _userFrameRate = static_cast<float>(frameRate);
+ _codecWidth = width;
+ _codecHeight = height;
+ _initCodecWidth = width;
+ _initCodecHeight = height;
+ _numLayers = (numLayers <= 1) ? 1 : numLayers; // Can also be zero.
+ WebRtc_Word32 ret = VCM_OK;
+ ret = _qmResolution->Initialize((float)_targetBitRate, _userFrameRate,
+ _codecWidth, _codecHeight);
+ return ret;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::RegisterProtectionCallback(VCMProtectionCallback*
+ protectionCallback)
+{
+ _videoProtectionCallback = protectionCallback;
+ return VCM_OK;
+
+}
+
+void
+VCMMediaOptimization::EnableFrameDropper(bool enable)
+{
+ _frameDropper->Enable(enable);
+}
+
+void
+VCMMediaOptimization::EnableProtectionMethod(bool enable,
+ VCMProtectionMethodEnum method)
+{
+ bool updated = false;
+ if (enable)
+ {
+ updated = _lossProtLogic->SetMethod(method);
+ }
+ else
+ {
+ _lossProtLogic->RemoveMethod(method);
+ }
+ if (updated)
+ {
+ _lossProtLogic->UpdateMethod();
+ }
+}
+
+bool
+VCMMediaOptimization::IsProtectionMethodEnabled(VCMProtectionMethodEnum method)
+{
+ return (_lossProtLogic->SelectedType() == method);
+}
+
+void
+VCMMediaOptimization::SetMtu(WebRtc_Word32 mtu)
+{
+ _maxPayloadSize = mtu;
+}
+
+float
+VCMMediaOptimization::SentFrameRate()
+{
+ if (_frameDropper)
+ {
+ return _frameDropper->ActualFrameRate((WebRtc_UWord32)(InputFrameRate()
+ + 0.5f));
+ }
+
+ return VCM_CODEC_ERROR;
+}
+
+float
+VCMMediaOptimization::SentBitRate()
+{
+ UpdateBitRateEstimate(-1, _clock->MillisecondTimestamp());
+ return _avgSentBitRateBps / 1000.0f;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::MaxBitRate()
+{
+ return _maxBitRate;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
+ FrameType encodedFrameType)
+{
+ // look into the ViE version - debug mode - needs also number of layers.
+ UpdateBitRateEstimate(encodedLength, _clock->MillisecondTimestamp());
+ if(encodedLength > 0)
+ {
+ const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
+ encodedFrameType != kVideoFrameGolden);
+
+ _frameDropper->Fill(encodedLength, deltaFrame);
+ if (_maxPayloadSize > 0 && encodedLength > 0)
+ {
+ const float minPacketsPerFrame = encodedLength /
+ static_cast<float>(_maxPayloadSize);
+ if (deltaFrame)
+ {
+ _lossProtLogic->UpdatePacketsPerFrame(
+ minPacketsPerFrame, _clock->MillisecondTimestamp());
+ }
+ else
+ {
+ _lossProtLogic->UpdatePacketsPerFrameKey(
+ minPacketsPerFrame, _clock->MillisecondTimestamp());
+ }
+
+ if (_enableQm)
+ {
+ // update quality select with encoded length
+ _qmResolution->UpdateEncodedSize(encodedLength,
+ encodedFrameType);
+ }
+ }
+ if (!deltaFrame && encodedLength > 0)
+ {
+ _lossProtLogic->UpdateKeyFrameSize(static_cast<float>(encodedLength));
+ }
+
+ // updating counters
+ if (deltaFrame)
+ {
+ _deltaFrameCnt++;
+ }
+ else
+ {
+ _keyFrameCnt++;
+ }
+
+ }
+
+ return VCM_OK;
+
+}
+
+void VCMMediaOptimization::UpdateBitRateEstimate(WebRtc_Word64 encodedLength,
+ WebRtc_Word64 nowMs)
+{
+ int i = kBitrateMaxFrameSamples - 1;
+ WebRtc_UWord32 frameSizeSum = 0;
+ WebRtc_Word64 timeOldest = -1;
+ // Find an empty slot for storing the new sample and at the same time
+ // accumulate the history.
+ for (; i >= 0; i--)
+ {
+ if (_encodedFrameSamples[i]._sizeBytes == -1)
+ {
+ // Found empty slot
+ break;
+ }
+ if (nowMs - _encodedFrameSamples[i]._timeCompleteMs <
+ kBitrateAverageWinMs)
+ {
+ frameSizeSum += static_cast<WebRtc_UWord32>
+ (_encodedFrameSamples[i]._sizeBytes);
+ if (timeOldest == -1)
+ {
+ timeOldest = _encodedFrameSamples[i]._timeCompleteMs;
+ }
+ }
+ }
+ if (encodedLength > 0)
+ {
+ if (i < 0)
+ {
+ // No empty slot, shift
+ for (i = kBitrateMaxFrameSamples - 2; i >= 0; i--)
+ {
+ _encodedFrameSamples[i + 1] = _encodedFrameSamples[i];
+ }
+ i++;
+ }
+ // Insert new sample
+ _encodedFrameSamples[i]._sizeBytes = encodedLength;
+ _encodedFrameSamples[i]._timeCompleteMs = nowMs;
+ }
+ if (timeOldest > -1)
+ {
+ // Update average bit rate
+ float denom = static_cast<float>(nowMs - timeOldest);
+ if (denom < 1.0)
+ {
+ denom = 1.0;
+ }
+ _avgSentBitRateBps = (frameSizeSum + encodedLength) * 8 * 1000 / denom;
+ }
+ else if (encodedLength > 0)
+ {
+ _avgSentBitRateBps = static_cast<float>(encodedLength * 8);
+ }
+ else
+ {
+ _avgSentBitRateBps = 0;
+ }
+}
+
+
+WebRtc_Word32
+VCMMediaOptimization::RegisterVideoQMCallback(VCMQMSettingsCallback*
+ videoQMSettings)
+{
+ _videoQMSettingsCallback = videoQMSettings;
+ // Callback setting controls QM
+ if (_videoQMSettingsCallback != NULL)
+ {
+ _enableQm = true;
+ }
+ else
+ {
+ _enableQm = false;
+ }
+ return VCM_OK;
+}
+
+void
+VCMMediaOptimization::updateContentData(const VideoContentMetrics*
+ contentMetrics)
+{
+ // Updating content metrics
+ if (contentMetrics == NULL)
+ {
+ // Disable QM if metrics are NULL
+ _enableQm = false;
+ _qmResolution->Reset();
+ }
+ else
+ {
+ _content->UpdateContentData(contentMetrics);
+ }
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SelectQuality()
+{
+ // Reset quantities for QM select
+ _qmResolution->ResetQM();
+
+ // Update QM will long-term averaged content metrics.
+ _qmResolution->UpdateContent(_content->LongTermAvgData());
+
+ // Select quality mode
+ VCMResolutionScale* qm = NULL;
+ WebRtc_Word32 ret = _qmResolution->SelectResolution(&qm);
+ if (ret < 0)
+ {
+ return ret;
+ }
+
+ // Check for updates to spatial/temporal modes
+ QMUpdate(qm);
+
+ // Reset all the rate and related frame counters quantities
+ _qmResolution->ResetRates();
+
+ // Reset counters
+ _lastQMUpdateTime = _clock->MillisecondTimestamp();
+
+ // Reset content metrics
+ _content->Reset();
+
+ return VCM_OK;
+}
+
+
+// Check timing constraints and look for significant change in:
+// (1) scene content
+// (2) target bit rate
+
+bool
+VCMMediaOptimization::checkStatusForQMchange()
+{
+
+ bool status = true;
+
+ // Check that we do not call QMSelect too often, and that we waited some time
+ // (to sample the metrics) from the event lastChangeTime
+ // lastChangeTime is the time where user changed the size/rate/frame rate
+ // (via SetEncodingData)
+ WebRtc_Word64 now = _clock->MillisecondTimestamp();
+ if ((now - _lastQMUpdateTime) < kQmMinIntervalMs ||
+ (now - _lastChangeTime) < kQmMinIntervalMs)
+ {
+ status = false;
+ }
+
+ return status;
+
+}
+
+bool
+VCMMediaOptimization::QMUpdate(VCMResolutionScale* qm)
+{
+ // Check for no change
+ if (qm->spatialHeightFact == 1 &&
+ qm->spatialWidthFact == 1 &&
+ qm->temporalFact == 1) {
+ return false;
+ }
+
+ // Temporal
+ WebRtc_UWord32 frameRate = static_cast<WebRtc_UWord32>
+ (_incomingFrameRate + 0.5f);
+
+ // Check if go back up in temporal resolution
+ if (qm->temporalFact == 0) {
+ // Currently only allow for 1/2 frame rate reduction per action.
+ // TODO (marpan): allow for 2/3 reduction.
+ frameRate = (WebRtc_UWord32) 2 * _incomingFrameRate;
+ }
+ // go down in temporal resolution
+ else {
+ frameRate = (WebRtc_UWord32)(_incomingFrameRate / qm->temporalFact + 1);
+ }
+ // Reset _incomingFrameRate if temporal action was selected.
+ if (qm->temporalFact != 1) {
+ memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
+ _incomingFrameRate = frameRate;
+ }
+
+ // Spatial
+ WebRtc_UWord32 height = _codecHeight;
+ WebRtc_UWord32 width = _codecWidth;
+ // Check if go back up in spatial resolution, and update frame sizes.
+ // Currently only allow for 2x2 spatial down-sampling.
+ // TODO (marpan): allow for 1x2, 2x1, and 4/3x4/3 (or 3/2x3/2).
+ if (qm->spatialHeightFact == 0 && qm->spatialWidthFact == 0) {
+ width = _codecWidth * 2;
+ height = _codecHeight * 2;
+ } else {
+ width = _codecWidth / qm->spatialWidthFact;
+ height = _codecHeight / qm->spatialHeightFact;
+ }
+ _codecWidth = width;
+ _codecHeight = height;
+
+ // New frame sizes should never exceed the original sizes
+ // from SetEncodingData().
+ assert(_codecWidth <= _initCodecWidth);
+ assert(_codecHeight <= _initCodecHeight);
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, _id,
+ "Quality Mode Update: W = %d, H = %d, FR = %f",
+ width, height, frameRate);
+
+ // Update VPM with new target frame rate and size
+ _videoQMSettingsCallback->SetVideoQMSettings(frameRate, width, height);
+
+ _content->UpdateFrameRate(frameRate);
+ _qmResolution->UpdateCodecFrameSize(width, height);
+
+ return true;
+}
+
+void
+VCMMediaOptimization::UpdateIncomingFrameRate()
+{
+ WebRtc_Word64 now = _clock->MillisecondTimestamp();
+ if (_incomingFrameTimes[0] == 0)
+ {
+ // first no shift
+ } else
+ {
+ // shift
+ for(WebRtc_Word32 i = (kFrameCountHistorySize - 2); i >= 0 ; i--)
+ {
+ _incomingFrameTimes[i+1] = _incomingFrameTimes[i];
+ }
+ }
+ _incomingFrameTimes[0] = now;
+ ProcessIncomingFrameRate(now);
+}
+
+// allowing VCM to keep track of incoming frame rate
+void
+VCMMediaOptimization::ProcessIncomingFrameRate(WebRtc_Word64 now)
+{
+ WebRtc_Word32 num = 0;
+ WebRtc_Word32 nrOfFrames = 0;
+ for (num = 1; num < (kFrameCountHistorySize - 1); num++)
+ {
+ if (_incomingFrameTimes[num] <= 0 ||
+ // don't use data older than 2 s
+ now - _incomingFrameTimes[num] > kFrameHistoryWinMs)
+ {
+ break;
+ } else
+ {
+ nrOfFrames++;
+ }
+ }
+ if (num > 1)
+ {
+ const WebRtc_Word64 diff = now - _incomingFrameTimes[num-1];
+ _incomingFrameRate = 1.0;
+ if(diff >0)
+ {
+ _incomingFrameRate = nrOfFrames * 1000.0f / static_cast<float>(diff);
+ }
+ }
+}
+
+WebRtc_UWord32
+VCMMediaOptimization::InputFrameRate()
+{
+ ProcessIncomingFrameRate(_clock->MillisecondTimestamp());
+ return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/media_optimization.h b/trunk/src/modules/video_coding/main/source/media_optimization.h
new file mode 100644
index 0000000..14e5d1a
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/media_optimization.h
@@ -0,0 +1,212 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+
+#include "module_common_types.h"
+#include "video_coding.h"
+#include "trace.h"
+#include "media_opt_util.h"
+#include "qm_select.h"
+
+namespace webrtc
+{
+
+enum { kBitrateMaxFrameSamples = 60 };
+enum { kBitrateAverageWinMs = 1000 };
+
+class TickTimeBase;
+class VCMContentMetricsProcessing;
+class VCMFrameDropper;
+
+struct VCMEncodedFrameSample
+{
+ VCMEncodedFrameSample() : _sizeBytes(-1), _timeCompleteMs(-1) {}
+
+ WebRtc_Word64 _sizeBytes;
+ WebRtc_Word64 _timeCompleteMs;
+};
+
+class VCMMediaOptimization
+{
+public:
+ VCMMediaOptimization(WebRtc_Word32 id, TickTimeBase* clock);
+ ~VCMMediaOptimization(void);
+ /*
+ * Reset the Media Optimization module
+ */
+ WebRtc_Word32 Reset();
+ /**
+ * Set target Rates for the encoder given the channel parameters
+ * Inputs: bitRate - target bitRate, in the conference case this is the rate
+ * between the sending client and the server
+ * fractionLost - packet loss in % in the network
+ * roundTripTimeMs - round trip time in miliseconds
+ * minBitRate - the bit rate of the end-point with lowest rate
+ * maxBitRate - the bit rate of the end-point with highest rate
+ */
+ WebRtc_UWord32 SetTargetRates(WebRtc_UWord32 bitRate,
+ WebRtc_UWord8 &fractionLost,
+ WebRtc_UWord32 roundTripTimeMs);
+
+ /**
+ * Inform media optimization of initial encoding state
+ */
+ WebRtc_Word32 SetEncodingData(VideoCodecType sendCodecType,
+ WebRtc_Word32 maxBitRate,
+ WebRtc_UWord32 frameRate,
+ WebRtc_UWord32 bitRate,
+ WebRtc_UWord16 width,
+ WebRtc_UWord16 height,
+ int numTemporalLayers);
+ /**
+ * Enable protection method
+ */
+ void EnableProtectionMethod(bool enable, VCMProtectionMethodEnum method);
+ /**
+ * Returns weather or not protection method is enabled
+ */
+ bool IsProtectionMethodEnabled(VCMProtectionMethodEnum method);
+ /**
+ * Updates the max pay load size
+ */
+ void SetMtu(WebRtc_Word32 mtu);
+ /*
+ * Get actual input frame rate
+ */
+ WebRtc_UWord32 InputFrameRate();
+
+ /*
+ * Get actual sent frame rate
+ */
+ float SentFrameRate();
+ /*
+ * Get actual sent bit rate
+ */
+ float SentBitRate();
+ /*
+ * Get maximum allowed bit rate
+ */
+ WebRtc_Word32 MaxBitRate();
+ /*
+ * Inform Media Optimization of encoding output: Length and frame type
+ */
+ WebRtc_Word32 UpdateWithEncodedData(WebRtc_Word32 encodedLength,
+ FrameType encodedFrameType);
+ /*
+ * Register a protection callback to be used to inform the user about the
+ * protection methods used
+ */
+ WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback*
+ protectionCallback);
+ /*
+ * Register a quality settings callback to be used to inform VPM/user about
+ */
+ WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings);
+ void EnableFrameDropper(bool enable);
+
+ bool DropFrame();
+
+ /*
+ * Get number of key/delta frames encoded
+ */
+ WebRtc_Word32 SentFrameCount(VCMFrameCount &frameCount) const;
+
+ /*
+ * update incoming frame rate value
+ */
+ void UpdateIncomingFrameRate();
+
+ /**
+ * Update content metric Data
+ */
+ void updateContentData(const VideoContentMetrics* contentMetrics);
+
+ /**
+ * Compute new Quality Mode
+ */
+ WebRtc_Word32 SelectQuality();
+
+private:
+
+ /*
+ * Update protection callback with protection settings
+ */
+ int UpdateProtectionCallback(VCMProtectionMethod *selected_method,
+ uint32_t* total_video_rate_bps,
+ uint32_t* nack_overhead_rate_bps,
+ uint32_t* fec_overhead_rate_bps);
+
+ void UpdateBitRateEstimate(WebRtc_Word64 encodedLength, WebRtc_Word64 nowMs);
+ /*
+ * verify if QM settings differ from default, i.e. if an update is required
+ * Compute actual values, as will be sent to the encoder
+ */
+ bool QMUpdate(VCMResolutionScale* qm);
+ /**
+ * check if we should make a QM change
+ * will return 1 if yes, 0 otherwise
+ */
+ bool checkStatusForQMchange();
+
+ void ProcessIncomingFrameRate(WebRtc_Word64 now);
+
+ enum { kFrameCountHistorySize = 90};
+ enum { kFrameHistoryWinMs = 2000};
+
+ WebRtc_Word32 _id;
+ TickTimeBase* _clock;
+ WebRtc_Word32 _maxBitRate;
+ VideoCodecType _sendCodecType;
+ WebRtc_UWord16 _codecWidth;
+ WebRtc_UWord16 _codecHeight;
+ WebRtc_UWord16 _initCodecWidth;
+ WebRtc_UWord16 _initCodecHeight;
+ float _userFrameRate;
+
+ VCMFrameDropper* _frameDropper;
+ VCMLossProtectionLogic* _lossProtLogic;
+ WebRtc_UWord8 _packetLossEnc;
+ WebRtc_UWord8 _fractionLost;
+
+
+ WebRtc_UWord32 _sendStatistics[4];
+ WebRtc_UWord32 _sendStatisticsZeroEncode;
+ WebRtc_Word32 _maxPayloadSize;
+ WebRtc_UWord32 _targetBitRate;
+
+ float _incomingFrameRate;
+ WebRtc_Word64 _incomingFrameTimes[kFrameCountHistorySize];
+
+ bool _enableQm;
+
+ VCMProtectionCallback* _videoProtectionCallback;
+ VCMQMSettingsCallback* _videoQMSettingsCallback;
+
+ VCMEncodedFrameSample _encodedFrameSamples[kBitrateMaxFrameSamples];
+ float _avgSentBitRateBps;
+
+ WebRtc_UWord32 _keyFrameCnt;
+ WebRtc_UWord32 _deltaFrameCnt;
+
+ VCMContentMetricsProcessing* _content;
+ VCMQmResolution* _qmResolution;
+
+ WebRtc_Word64 _lastQMUpdateTime;
+ WebRtc_Word64 _lastChangeTime; // content/user triggered
+ int _numLayers;
+
+
+}; // end of VCMMediaOptimization class definition
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
diff --git a/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h b/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h
new file mode 100644
index 0000000..c6da348
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
+
+#include <assert.h>
+
+#include <limits>
+
+#include "modules/video_coding/main/source/tick_time_base.h"
+
+namespace webrtc {
+
+// Provides a fake implementation of TickTimeBase, intended for offline
+// testing. This implementation does not query the system clock, but returns a
+// time value set by the user when creating the object, and incremented with
+// the method IncrementDebugClock.
+class FakeTickTime : public TickTimeBase {
+ public:
+ explicit FakeTickTime(int64_t start_time_ms) : fake_now_ms_(start_time_ms) {}
+ virtual ~FakeTickTime() {}
+ virtual int64_t MillisecondTimestamp() const {
+ return fake_now_ms_;
+ }
+ virtual int64_t MicrosecondTimestamp() const {
+ return 1000 * fake_now_ms_;
+ }
+ virtual void IncrementDebugClock(int64_t increase_ms) {
+ assert(increase_ms <= std::numeric_limits<int64_t>::max() - fake_now_ms_);
+ fake_now_ms_ += increase_ms;
+ }
+
+ private:
+ int64_t fake_now_ms_;
+};
+
+} // namespace
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
diff --git a/trunk/src/modules/video_coding/main/source/nack_fec_tables.h b/trunk/src/modules/video_coding/main/source/nack_fec_tables.h
new file mode 100644
index 0000000..88e225d
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/nack_fec_tables.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
+
+namespace webrtc
+{
+
+// Table for adjusting FEC rate for NACK/FEC protection method
+// Table values are built as a sigmoid function, ranging from 0 to
+// kHighRttNackMs (100), based on the HybridNackTH values defined in
+// media_opt_util.h.
+const WebRtc_UWord16 VCMNackFecTable[100] = {
+0,
+0,
+0,
+0,
+0,
+0,
+0,
+0,
+0,
+0,
+0,
+1,
+1,
+1,
+1,
+1,
+2,
+2,
+2,
+3,
+3,
+4,
+5,
+6,
+7,
+9,
+10,
+12,
+15,
+18,
+21,
+24,
+28,
+32,
+37,
+41,
+46,
+51,
+56,
+61,
+66,
+70,
+74,
+78,
+81,
+84,
+86,
+89,
+90,
+92,
+93,
+95,
+95,
+96,
+97,
+97,
+98,
+98,
+99,
+99,
+99,
+99,
+99,
+99,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+100,
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
diff --git a/trunk/src/modules/video_coding/main/source/packet.cc b/trunk/src/modules/video_coding/main/source/packet.cc
new file mode 100644
index 0000000..e52cbdd
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/packet.cc
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet.h"
+#include "module_common_types.h"
+
+#include <assert.h>
+
+namespace webrtc {
+
+VCMPacket::VCMPacket()
+ :
+ payloadType(0),
+ timestamp(0),
+ seqNum(0),
+ dataPtr(NULL),
+ sizeBytes(0),
+ markerBit(false),
+ frameType(kFrameEmpty),
+ codec(kVideoCodecUnknown),
+ isFirstPacket(false),
+ completeNALU(kNaluUnset),
+ insertStartCode(false),
+ codecSpecificHeader() {
+}
+
+VCMPacket::VCMPacket(const WebRtc_UWord8* ptr,
+ const WebRtc_UWord32 size,
+ const WebRtcRTPHeader& rtpHeader) :
+ payloadType(rtpHeader.header.payloadType),
+ timestamp(rtpHeader.header.timestamp),
+ seqNum(rtpHeader.header.sequenceNumber),
+ dataPtr(ptr),
+ sizeBytes(size),
+ markerBit(rtpHeader.header.markerBit),
+
+ frameType(rtpHeader.frameType),
+ codec(kVideoCodecUnknown),
+ isFirstPacket(rtpHeader.type.Video.isFirstPacket),
+ completeNALU(kNaluComplete),
+ insertStartCode(false),
+ codecSpecificHeader(rtpHeader.type.Video)
+{
+ CopyCodecSpecifics(rtpHeader.type.Video);
+}
+
+VCMPacket::VCMPacket(const WebRtc_UWord8* ptr, WebRtc_UWord32 size, WebRtc_UWord16 seq, WebRtc_UWord32 ts, bool mBit) :
+ payloadType(0),
+ timestamp(ts),
+ seqNum(seq),
+ dataPtr(ptr),
+ sizeBytes(size),
+ markerBit(mBit),
+
+ frameType(kVideoFrameDelta),
+ codec(kVideoCodecUnknown),
+ isFirstPacket(false),
+ completeNALU(kNaluComplete),
+ insertStartCode(false),
+ codecSpecificHeader()
+{}
+
+void VCMPacket::Reset() {
+ payloadType = 0;
+ timestamp = 0;
+ seqNum = 0;
+ dataPtr = NULL;
+ sizeBytes = 0;
+ markerBit = false;
+ frameType = kFrameEmpty;
+ codec = kVideoCodecUnknown;
+ isFirstPacket = false;
+ completeNALU = kNaluUnset;
+ insertStartCode = false;
+ memset(&codecSpecificHeader, 0, sizeof(RTPVideoHeader));
+}
+
+void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader)
+{
+ switch(videoHeader.codec)
+ {
+ case kRTPVideoVP8:
+ {
+ // Handle all packets within a frame as depending on the previous packet
+ // TODO(holmer): This should be changed to make fragments independent
+ // when the VP8 RTP receiver supports fragments.
+ if (isFirstPacket && markerBit)
+ completeNALU = kNaluComplete;
+ else if (isFirstPacket)
+ completeNALU = kNaluStart;
+ else if (markerBit)
+ completeNALU = kNaluEnd;
+ else
+ completeNALU = kNaluIncomplete;
+
+ codec = kVideoCodecVP8;
+ break;
+ }
+ case kRTPVideoI420:
+ {
+ codec = kVideoCodecI420;
+ break;
+ }
+ default:
+ {
+ codec = kVideoCodecUnknown;
+ break;
+ }
+ }
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/packet.h b/trunk/src/modules/video_coding/main/source/packet.h
new file mode 100644
index 0000000..2035653
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/packet.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+#include "jitter_buffer_common.h"
+
+namespace webrtc
+{
+
+class VCMPacket
+{
+public:
+ VCMPacket();
+ VCMPacket(const WebRtc_UWord8* ptr,
+ const WebRtc_UWord32 size,
+ const WebRtcRTPHeader& rtpHeader);
+ VCMPacket(const WebRtc_UWord8* ptr,
+ WebRtc_UWord32 size,
+ WebRtc_UWord16 seqNum,
+ WebRtc_UWord32 timestamp,
+ bool markerBit);
+
+ void Reset();
+
+ WebRtc_UWord8 payloadType;
+ WebRtc_UWord32 timestamp;
+ WebRtc_UWord16 seqNum;
+ const WebRtc_UWord8* dataPtr;
+ WebRtc_UWord32 sizeBytes;
+ bool markerBit;
+
+ FrameType frameType;
+ webrtc::VideoCodecType codec;
+
+ bool isFirstPacket; // Is this first packet in a frame.
+ VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
+ bool insertStartCode; // True if a start code should be inserted before this
+ // packet.
+ RTPVideoHeader codecSpecificHeader;
+
+protected:
+ void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/trunk/src/modules/video_coding/main/source/qm_select.cc b/trunk/src/modules/video_coding/main/source/qm_select.cc
new file mode 100644
index 0000000..c4bd707
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/qm_select.cc
@@ -0,0 +1,664 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/qm_select.h"
+
+#include <math.h>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/source/internal_defines.h"
+#include "modules/video_coding/main/source/qm_select_data.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+// QM-METHOD class
+
+VCMQmMethod::VCMQmMethod()
+ : _contentMetrics(NULL),
+ _width(0),
+ _height(0),
+ _nativeWidth(0),
+ _nativeHeight(0),
+ _frameRateLevel(kDefault),
+ _init(false) {
+ ResetQM();
+}
+
+VCMQmMethod::~VCMQmMethod() {
+}
+
+void VCMQmMethod::ResetQM() {
+ _aspectRatio = 1.0f;
+ _imageType = 2;
+ _motion.Reset();
+ _spatial.Reset();
+ _contentClass = 0;
+}
+
+uint8_t VCMQmMethod::ComputeContentClass() {
+ ComputeMotionNFD();
+ ComputeSpatial();
+ return _contentClass = 3 * _motion.level + _spatial.level;
+}
+
+void VCMQmMethod::UpdateContent(const VideoContentMetrics* contentMetrics) {
+ _contentMetrics = contentMetrics;
+}
+
+void VCMQmMethod::ComputeMotionNFD() {
+ if (_contentMetrics) {
+ _motion.value = _contentMetrics->motion_magnitude;
+ }
+ // Determine motion level.
+ if (_motion.value < kLowMotionNfd) {
+ _motion.level = kLow;
+ } else if (_motion.value > kHighMotionNfd) {
+ _motion.level = kHigh;
+ } else {
+ _motion.level = kDefault;
+ }
+}
+
+void VCMQmMethod::ComputeSpatial() {
+ float spatialErr = 0.0;
+ float spatialErrH = 0.0;
+ float spatialErrV = 0.0;
+ if (_contentMetrics) {
+ spatialErr = _contentMetrics->spatial_pred_err;
+ spatialErrH = _contentMetrics->spatial_pred_err_h;
+ spatialErrV = _contentMetrics->spatial_pred_err_v;
+ }
+ // Spatial measure: take average of 3 prediction errors.
+ _spatial.value = (spatialErr + spatialErrH + spatialErrV) / 3.0f;
+
+ // Reduce thresholds for large scenes/higher pixel correlation (~>=WHD).
+ float scale2 = _imageType > 3 ? kScaleTexture : 1.0;
+
+ if (_spatial.value > scale2 * kHighTexture) {
+ _spatial.level = kHigh;
+ } else if (_spatial.value < scale2 * kLowTexture) {
+ _spatial.level = kLow;
+ } else {
+ _spatial.level = kDefault;
+ }
+}
+
+uint8_t VCMQmMethod::GetImageType(uint16_t width,
+ uint16_t height) {
+ // Get the closest image type for encoder frame size.
+ uint32_t imageSize = width * height;
+ if (imageSize < kFrameSizeTh[0]) {
+ return 0; // QCIF
+ } else if (imageSize < kFrameSizeTh[1]) {
+ return 1; // CIF
+ } else if (imageSize < kFrameSizeTh[2]) {
+ return 2; // VGA
+ } else if (imageSize < kFrameSizeTh[3]) {
+ return 3; // 4CIF
+ } else if (imageSize < kFrameSizeTh[4]) {
+ return 4; // 720,4:3
+ } else if (imageSize < kFrameSizeTh[5]) {
+ return 5; // WHD
+ } else {
+ return 6; // HD
+ }
+}
+
+LevelClass VCMQmMethod::FrameRateLevel(float avgFrameRate) {
+ if (avgFrameRate < kLowFrameRate) {
+ return kLow;
+ } else if (avgFrameRate > kHighFrameRate) {
+ return kHigh;
+ } else {
+ return kDefault;
+ }
+}
+
+// RESOLUTION CLASS
+
+VCMQmResolution::VCMQmResolution()
+ : _qm(new VCMResolutionScale()) {
+ Reset();
+}
+
+VCMQmResolution::~VCMQmResolution() {
+ delete _qm;
+}
+
+void VCMQmResolution::ResetRates() {
+ _sumTargetRate = 0.0f;
+ _sumIncomingFrameRate = 0.0f;
+ _sumRateMM = 0.0f;
+ _sumRateMMSgn = 0;
+ _sumPacketLoss = 0.0f;
+ _frameCnt = 0;
+ _frameCntDelta = 0;
+ _lowBufferCnt = 0;
+ _updateRateCnt = 0;
+}
+
+void VCMQmResolution::ResetDownSamplingState() {
+ _stateDecFactorSpatial = 1;
+ _stateDecFactorTemp = 1;
+}
+
+void VCMQmResolution::Reset() {
+ _targetBitRate = 0.0f;
+ _userFrameRate = 0.0f;
+ _incomingFrameRate = 0.0f;
+ _perFrameBandwidth =0.0f;
+ _bufferLevel = 0.0f;
+ _avgTargetRate = 0.0f;
+ _avgIncomingFrameRate = 0.0f;
+ _avgRatioBufferLow = 0.0f;
+ _avgRateMisMatch = 0.0f;
+ _avgRateMisMatchSgn = 0.0f;
+ _avgPacketLoss = 0.0f;
+ _encoderState = kStableEncoding;
+ ResetRates();
+ ResetDownSamplingState();
+ ResetQM();
+}
+
+EncoderState VCMQmResolution::GetEncoderState() {
+ return _encoderState;
+}
+
+// Initialize state after re-initializing the encoder,
+// i.e., after SetEncodingData() in mediaOpt.
+int VCMQmResolution::Initialize(float bitRate,
+ float userFrameRate,
+ uint16_t width,
+ uint16_t height) {
+ if (userFrameRate == 0.0f || width == 0 || height == 0) {
+ return VCM_PARAMETER_ERROR;
+ }
+ Reset();
+ _targetBitRate = bitRate;
+ _userFrameRate = userFrameRate;
+ _incomingFrameRate = userFrameRate;
+ UpdateCodecFrameSize(width, height);
+ _nativeWidth = width;
+ _nativeHeight = height;
+ // Initial buffer level.
+ _bufferLevel = kInitBufferLevel * _targetBitRate;
+ // Per-frame bandwidth.
+ _perFrameBandwidth = _targetBitRate / _userFrameRate;
+ _init = true;
+ return VCM_OK;
+}
+
+void VCMQmResolution::UpdateCodecFrameSize(uint16_t width, uint16_t height) {
+ _width = width;
+ _height = height;
+ // Set the imageType for the encoder width/height.
+ _imageType = GetImageType(width, height);
+}
+
+// Update rate data after every encoded frame.
+void VCMQmResolution::UpdateEncodedSize(int encodedSize,
+ FrameType encodedFrameType) {
+ _frameCnt++;
+ // Convert to Kbps.
+ float encodedSizeKbits = static_cast<float>((encodedSize * 8.0) / 1000.0);
+
+ // Update the buffer level:
+ // Note this is not the actual encoder buffer level.
+ // |_bufferLevel| is reset to 0 every time SelectResolution is called, and
+ // does not account for frame dropping by encoder or VCM.
+ _bufferLevel += _perFrameBandwidth - encodedSizeKbits;
+ // Counter for occurrences of low buffer level:
+ // low/negative values means encoder is likely dropping frames.
+ if (_bufferLevel <= kPercBufferThr * kOptBufferLevel * _targetBitRate) {
+ _lowBufferCnt++;
+ }
+}
+
+// Update various quantities after SetTargetRates in MediaOpt.
+void VCMQmResolution::UpdateRates(float targetBitRate,
+ float encoderSentRate,
+ float incomingFrameRate,
+ uint8_t packetLoss) {
+ // Sum the target bitrate and incoming frame rate:
+ // these values are the encoder rates (from previous update ~1sec),
+ // i.e, before the update for next ~1sec.
+ _sumTargetRate += _targetBitRate;
+ _sumIncomingFrameRate += _incomingFrameRate;
+ _updateRateCnt++;
+
+ // Sum the received (from RTCP reports) packet loss rates.
+ _sumPacketLoss += static_cast<float>(packetLoss / 255.0);
+
+ // Sum the sequence rate mismatch:
+ // Mismatch here is based on the difference between the target rate
+ // used (in previous ~1sec) and the average actual encoding rate measured
+ // at previous ~1sec.
+ float diff = _targetBitRate - encoderSentRate;
+ if (_targetBitRate > 0.0)
+ _sumRateMM += fabs(diff) / _targetBitRate;
+ int sgnDiff = diff > 0 ? 1 : (diff < 0 ? -1 : 0);
+ // To check for consistent under(+)/over_shooting(-) of target rate.
+ _sumRateMMSgn += sgnDiff;
+
+ // Update with the current new target and frame rate:
+ // these values are ones the encoder will use for the current/next ~1sec
+ _targetBitRate = targetBitRate;
+ _incomingFrameRate = incomingFrameRate;
+
+ // Update the per_frame_bandwidth:
+ // this is the per_frame_bw for the current/next ~1sec
+ _perFrameBandwidth = 0.0f;
+ if (_incomingFrameRate > 0.0f) {
+ _perFrameBandwidth = _targetBitRate / _incomingFrameRate;
+ }
+}
+
+// Select the resolution factors: frame size and frame rate change (qm scales).
+// Selection is for going down in resolution, or for going back up
+// (if a previous down-sampling action was taken).
+
+// In the current version the following constraints are imposed:
+// 1) we only allow for one action (either down or back up) at a given time.
+// 2) the possible down-sampling actions are: 2x2 spatial and 1/2 temporal.
+// 3) the total amount of down-sampling (spatial and/or temporal) from the
+// initial (native) resolution is limited by various factors.
+
+// TODO(marpan): extend to allow options for: 4/3x4/3, 1x2, 2x1 spatial,
+// and 2/3 temporal (i.e., skip every third frame).
+int VCMQmResolution::SelectResolution(VCMResolutionScale** qm) {
+ if (!_init) {
+ return VCM_UNINITIALIZED;
+ }
+ if (_contentMetrics == NULL) {
+ Reset();
+ *qm = _qm;
+ return VCM_OK;
+ }
+
+ // Default settings: no action.
+ _qm->spatialWidthFact = 1;
+ _qm->spatialHeightFact = 1;
+ _qm->temporalFact = 1;
+ *qm = _qm;
+
+ // Compute content class for selection.
+ _contentClass = ComputeContentClass();
+
+ // Compute various rate quantities for selection.
+ ComputeRatesForSelection();
+
+ // Get the encoder state.
+ ComputeEncoderState();
+
+ // Check for going back up in resolution, if we have had some down-sampling
+ // relative to native state in Initialize (i.e., after SetEncodingData()
+ // in mediaOpt.).
+ if (_stateDecFactorSpatial > 1 || _stateDecFactorTemp > 1) {
+ if (GoingUpResolution()) {
+ *qm = _qm;
+ return VCM_OK;
+ }
+ }
+
+ // Check for going down in resolution, only if current total amount of
+ // down-sampling state is below threshold.
+ if (_stateDecFactorTemp * _stateDecFactorSpatial < kMaxDownSample) {
+ if (GoingDownResolution()) {
+ *qm = _qm;
+ return VCM_OK;
+ }
+ }
+ return VCM_OK;
+}
+
+void VCMQmResolution::ComputeRatesForSelection() {
+ _avgTargetRate = 0.0f;
+ _avgIncomingFrameRate = 0.0f;
+ _avgRatioBufferLow = 0.0f;
+ _avgRateMisMatch = 0.0f;
+ _avgRateMisMatchSgn = 0.0f;
+ _avgPacketLoss = 0.0f;
+ if (_frameCnt > 0) {
+ _avgRatioBufferLow = static_cast<float>(_lowBufferCnt) /
+ static_cast<float>(_frameCnt);
+ }
+ if (_updateRateCnt > 0) {
+ _avgRateMisMatch = static_cast<float>(_sumRateMM) /
+ static_cast<float>(_updateRateCnt);
+ _avgRateMisMatchSgn = static_cast<float>(_sumRateMMSgn) /
+ static_cast<float>(_updateRateCnt);
+ _avgTargetRate = static_cast<float>(_sumTargetRate) /
+ static_cast<float>(_updateRateCnt);
+ _avgIncomingFrameRate = static_cast<float>(_sumIncomingFrameRate) /
+ static_cast<float>(_updateRateCnt);
+ _avgPacketLoss = static_cast<float>(_sumPacketLoss) /
+ static_cast<float>(_updateRateCnt);
+ }
+ // For selection we may want to weight some quantities more heavily
+ // with the current (i.e., next ~1sec) rate values.
+ float weight = 0.7f;
+ _avgTargetRate = weight * _avgTargetRate + (1.0 - weight) * _targetBitRate;
+ _avgIncomingFrameRate = weight * _avgIncomingFrameRate +
+ (1.0 - weight) * _incomingFrameRate;
+ _frameRateLevel = FrameRateLevel(_avgIncomingFrameRate);
+}
+
+void VCMQmResolution::ComputeEncoderState() {
+ // Default.
+ _encoderState = kStableEncoding;
+
+ // Assign stressed state if:
+ // 1) occurrences of low buffer levels is high, or
+ // 2) rate mis-match is high, and consistent over-shooting by encoder.
+ if ((_avgRatioBufferLow > kMaxBufferLow) ||
+ ((_avgRateMisMatch > kMaxRateMisMatch) &&
+ (_avgRateMisMatchSgn < -kRateOverShoot))) {
+ _encoderState = kStressedEncoding;
+ }
+ // Assign easy state if:
+ // 1) rate mis-match is high, and
+ // 2) consistent under-shooting by encoder.
+ if ((_avgRateMisMatch > kMaxRateMisMatch) &&
+ (_avgRateMisMatchSgn > kRateUnderShoot)) {
+ _encoderState = kEasyEncoding;
+ }
+}
+
+bool VCMQmResolution::GoingUpResolution() {
+ // Check if we should go up both spatially and temporally.
+ if (_stateDecFactorSpatial > 1 && _stateDecFactorTemp > 1) {
+ if (ConditionForGoingUp(2, 2, 2, kTransRateScaleUpSpatialTemp)) {
+ _qm->spatialHeightFact = 0;
+ _qm->spatialWidthFact = 0;
+ _qm->temporalFact = 0;
+ UpdateDownsamplingState(kUpResolution);
+ return true;
+ }
+ } else {
+ // Check if we should go up either spatially or temporally.
+ bool selectedUpS = false;
+ bool selectedUpT = false;
+ if (_stateDecFactorSpatial > 1) {
+ selectedUpS = ConditionForGoingUp(2, 2, 1, kTransRateScaleUpSpatial);
+ }
+ if (_stateDecFactorTemp > 1) {
+ selectedUpT = ConditionForGoingUp(1, 1, 2, kTransRateScaleUpTemp);
+ }
+ if (selectedUpS && !selectedUpT) {
+ _qm->spatialHeightFact = 0;
+ _qm->spatialWidthFact = 0;
+ UpdateDownsamplingState(kUpResolution);
+ return true;
+ } else if (!selectedUpS && selectedUpT) {
+ _qm->temporalFact = 0;
+ UpdateDownsamplingState(kUpResolution);
+ return true;
+ } else if (selectedUpS && selectedUpT) {
+ // TODO(marpan): which one to pick?
+ // pickSpatialOrTemporal()
+ // For now take spatial over temporal.
+ _qm->spatialHeightFact = 0;
+ _qm->spatialWidthFact = 0;
+ UpdateDownsamplingState(kUpResolution);
+ return true;
+ }
+ }
+ return false;
+}
+
+bool VCMQmResolution::ConditionForGoingUp(uint8_t facWidth,
+ uint8_t facHeight,
+ uint8_t facTemp,
+ float scaleFac) {
+ float estimatedTransitionRateUp = GetTransitionRate(facWidth, facHeight,
+ facTemp, scaleFac);
+ // Go back up if:
+ // 1) target rate is above threshold and current encoder state is stable, or
+ // 2) encoder state is easy (encoder is significantly under-shooting target).
+ if (((_avgTargetRate > estimatedTransitionRateUp) &&
+ (_encoderState == kStableEncoding)) ||
+ (_encoderState == kEasyEncoding)) {
+ return true;
+ } else {
+ return false;
+ }
+}
+
+bool VCMQmResolution::GoingDownResolution() {
+ float estimatedTransitionRateDown = GetTransitionRate(1, 1, 1, 1.0);
+ float maxRate = kFrameRateFac[_frameRateLevel] * kMaxRateQm[_imageType];
+
+ // TODO(marpan): Bias down-sampling based on packet loss conditions.
+
+ // Resolution reduction if:
+ // (1) target rate is below transition rate, or
+ // (2) encoder is in stressed state and target rate below a max threshold.
+ if ((_avgTargetRate < estimatedTransitionRateDown ) ||
+ (_encoderState == kStressedEncoding && _avgTargetRate < maxRate)) {
+ // Get the down-sampling action.
+ uint8_t spatialFact = kSpatialAction[_contentClass];
+ uint8_t tempFact = kTemporalAction[_contentClass];
+
+ switch (spatialFact) {
+ case 4: {
+ _qm->spatialWidthFact = 2;
+ _qm->spatialHeightFact = 2;
+ break;
+ }
+ case 2: {
+ assert(false); // Currently not used.
+ // Select 1x2,2x1, or 4/3x4/3.
+ // SelectSpatialDirectionMode((float) estimatedTransitionRateDown);
+ break;
+ }
+ case 1: {
+ _qm->spatialWidthFact = 1;
+ _qm->spatialHeightFact = 1;
+ break;
+ }
+ default: {
+ assert(false);
+ }
+ }
+ switch (tempFact) {
+ case 2: {
+ _qm->temporalFact = 2;
+ break;
+ }
+ case 1: {
+ _qm->temporalFact = 1;
+ break;
+ }
+ default: {
+ assert(false);
+ }
+ }
+ // Adjust some cases based on frame rate.
+ // TODO(marpan): will be modified when we add 1/2 spatial and 2/3 temporal.
+ AdjustAction();
+
+ // Sanity checks on down-sampling selection:
+ // override the settings for too small image size and/or frame rate.
+ // Also check the limit on current down-sampling states.
+
+ // No spatial sampling if current frame size is too small (QCIF),
+ // or if amount of spatial down-sampling is already too much.
+ if ((_width * _height) <= kMinImageSize ||
+ _stateDecFactorSpatial >= kMaxSpatialDown) {
+ _qm->spatialWidthFact = 1;
+ _qm->spatialHeightFact = 1;
+ }
+ // No frame rate reduction if average frame rate is below some point,
+ // or if the amount of temporal down-sampling is already too much.
+ if (_avgIncomingFrameRate <= kMinFrameRate ||
+ _stateDecFactorTemp >= kMaxTempDown) {
+ _qm->temporalFact = 1;
+ }
+
+ // Update down-sampling state.
+ if (_qm->spatialWidthFact != 1 || _qm->spatialHeightFact != 1 ||
+ _qm->temporalFact != 1) {
+ UpdateDownsamplingState(kDownResolution);
+ return true;
+ }
+ }
+ return false;
+}
+
+float VCMQmResolution::GetTransitionRate(uint8_t facWidth,
+ uint8_t facHeight,
+ uint8_t facTemp,
+ float scaleFac) {
+ uint8_t imageType = GetImageType(facWidth * _width,
+ facHeight * _height);
+ LevelClass frameRateLevel = FrameRateLevel(facTemp * _avgIncomingFrameRate);
+
+ // The maximum allowed rate below which down-sampling is allowed:
+ // Nominal values based on image format (frame size and frame rate).
+ float maxRate = kFrameRateFac[frameRateLevel] * kMaxRateQm[imageType];
+
+ uint8_t imageClass = imageType > 3 ? 1: 0;
+ uint8_t tableIndex = imageClass * 9 + _contentClass;
+ // Scale factor for down-sampling transition threshold:
+ // factor based on the content class and the image size.
+ float scaleTransRate = kScaleTransRateQm[tableIndex];
+
+ // Threshold bitrate for resolution action.
+ return static_cast<float> (scaleFac * facTemp * _incomingFrameRate *
+ scaleTransRate * maxRate / 30);
+}
+
+void VCMQmResolution::UpdateDownsamplingState(ResolutionAction action) {
+ // Assumes for now only actions are 1/2 frame rate of 2x2 spatial.
+ if (action == kUpResolution) {
+ if (_qm->spatialHeightFact == 0 && _qm->spatialWidthFact == 0) {
+ _stateDecFactorSpatial = _stateDecFactorSpatial / 4;
+ assert(_stateDecFactorSpatial >= 1);
+ }
+ if (_qm->temporalFact == 0) {
+ _stateDecFactorTemp = _stateDecFactorTemp / 2;
+ assert(_stateDecFactorTemp >= 1);
+ }
+ } else if (action == kDownResolution) {
+ _stateDecFactorSpatial = _stateDecFactorSpatial * _qm->spatialWidthFact
+ * _qm->spatialHeightFact;
+ _stateDecFactorTemp = _stateDecFactorTemp * _qm->temporalFact;
+ assert(_stateDecFactorSpatial >= 1);
+ assert(_stateDecFactorTemp >= 1);
+ } else {
+ assert(false);
+ }
+}
+
+void VCMQmResolution::AdjustAction() {
+ if (_spatial.level == kDefault && _motion.level != kHigh &&
+ _frameRateLevel == kHigh) {
+ _qm->temporalFact = 2;
+ _qm->spatialWidthFact = 1;
+ _qm->spatialHeightFact = 1;
+ }
+}
+
+// TODO(marpan): Update this when we allow for 1/2 spatial down-sampling.
+void VCMQmResolution::SelectSpatialDirectionMode(float transRate) {
+ // Default is 1x2 (H)
+ // For bit rates well below transitional rate, we select 2x2.
+ if (_targetBitRate < transRate * kRateRedSpatial2X2) {
+ _qm->spatialWidthFact = 2;
+ _qm->spatialHeightFact = 2;
+ }
+ // Otherwise check prediction errors and aspect ratio.
+ float spatialErr = 0.0;
+ float spatialErrH = 0.0;
+ float spatialErrV = 0.0;
+ if (_contentMetrics) {
+ spatialErr = _contentMetrics->spatial_pred_err;
+ spatialErrH = _contentMetrics->spatial_pred_err_h;
+ spatialErrV = _contentMetrics->spatial_pred_err_v;
+ }
+
+ // Favor 1x2 if aspect_ratio is 16:9.
+ if (_aspectRatio >= 16.0f / 9.0f) {
+ // Check if 1x2 has lowest prediction error.
+ if (spatialErrH < spatialErr && spatialErrH < spatialErrV) {
+ _qm->spatialWidthFact = 2;
+ _qm->spatialHeightFact = 1;
+ }
+ }
+ // Check for 2x2 selection: favor 2x2 over 1x2 and 2x1.
+ if (spatialErr < spatialErrH * (1.0f + kSpatialErr2x2VsHoriz) &&
+ spatialErr < spatialErrV * (1.0f + kSpatialErr2X2VsVert)) {
+ _qm->spatialWidthFact = 2;
+ _qm->spatialHeightFact = 2;
+ }
+ // Check for 2x1 selection.
+ if (spatialErrV < spatialErrH * (1.0f - kSpatialErrVertVsHoriz) &&
+ spatialErrV < spatialErr * (1.0f - kSpatialErr2X2VsVert)) {
+ _qm->spatialWidthFact = 1;
+ _qm->spatialHeightFact = 2;
+ }
+}
+
+// ROBUSTNESS CLASS
+
+VCMQmRobustness::VCMQmRobustness() {
+ Reset();
+}
+
+VCMQmRobustness::~VCMQmRobustness() {
+}
+
+void VCMQmRobustness::Reset() {
+ _prevTotalRate = 0.0f;
+ _prevRttTime = 0;
+ _prevPacketLoss = 0;
+ _prevCodeRateDelta = 0;
+ ResetQM();
+}
+
+// Adjust the FEC rate based on the content and the network state
+// (packet loss rate, total rate/bandwidth, round trip time).
+// Note that packetLoss here is the filtered loss value.
+float VCMQmRobustness::AdjustFecFactor(uint8_t codeRateDelta,
+ float totalRate,
+ float frameRate,
+ uint32_t rttTime,
+ uint8_t packetLoss) {
+ // Default: no adjustment
+ float adjustFec = 1.0f;
+ if (_contentMetrics == NULL) {
+ return adjustFec;
+ }
+ // Compute class state of the content.
+ ComputeMotionNFD();
+ ComputeSpatial();
+
+ // TODO(marpan): Set FEC adjustment factor.
+
+ // Keep track of previous values of network state:
+ // adjustment may be also based on pattern of changes in network state.
+ _prevTotalRate = totalRate;
+ _prevRttTime = rttTime;
+ _prevPacketLoss = packetLoss;
+ _prevCodeRateDelta = codeRateDelta;
+ return adjustFec;
+}
+
+// Set the UEP (unequal-protection across packets) on/off for the FEC.
+bool VCMQmRobustness::SetUepProtection(uint8_t codeRateDelta,
+ float totalRate,
+ uint8_t packetLoss,
+ bool frameType) {
+ // Default.
+ return false;
+}
+} // end of namespace
diff --git a/trunk/src/modules/video_coding/main/source/qm_select.h b/trunk/src/modules/video_coding/main/source/qm_select.h
new file mode 100644
index 0000000..1859530
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/qm_select.h
@@ -0,0 +1,259 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
+#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
+
+#include "common_types.h"
+#include "typedefs.h"
+
+/******************************************************/
+/* Quality Modes: Resolution and Robustness settings */
+/******************************************************/
+
+namespace webrtc {
+struct VideoContentMetrics;
+
+struct VCMResolutionScale {
+ VCMResolutionScale()
+ : spatialWidthFact(1),
+ spatialHeightFact(1),
+ temporalFact(1) {
+ }
+ uint8_t spatialWidthFact;
+ uint8_t spatialHeightFact;
+ uint8_t temporalFact;
+};
+
+enum LevelClass {
+ kLow,
+ kHigh,
+ kDefault
+};
+
+struct VCMContFeature {
+ VCMContFeature()
+ : value(0.0f),
+ level(kDefault) {
+ }
+ void Reset() {
+ value = 0.0f;
+ level = kDefault;
+ }
+ float value;
+ LevelClass level;
+};
+
+enum ResolutionAction {
+ kDownResolution,
+ kUpResolution,
+ kNoChangeResolution
+};
+
+enum EncoderState {
+ kStableEncoding, // Low rate mis-match, stable buffer levels.
+ kStressedEncoding, // Significant over-shooting of target rate,
+ // Buffer under-flow, etc.
+ kEasyEncoding // Significant under-shooting of target rate.
+};
+
+// QmMethod class: main class for resolution and robustness settings
+
+class VCMQmMethod {
+ public:
+ VCMQmMethod();
+ virtual ~VCMQmMethod();
+
+ // Reset values
+ void ResetQM();
+ virtual void Reset() = 0;
+
+ // Compute content class.
+ uint8_t ComputeContentClass();
+
+ // Update with the content metrics.
+ void UpdateContent(const VideoContentMetrics* contentMetrics);
+
+ // Compute spatial texture magnitude and level.
+ // Spatial texture is a spatial prediction error measure.
+ void ComputeSpatial();
+
+ // Compute motion magnitude and level for NFD metric.
+ // NFD is normalized frame difference (normalized by spatial variance).
+ void ComputeMotionNFD();
+
+ // Get the imageType (CIF, VGA, HD, etc) for the system width/height.
+ uint8_t GetImageType(uint16_t width, uint16_t height);
+
+ // Get the frame rate level.
+ LevelClass FrameRateLevel(float frame_rate);
+
+ protected:
+ // Content Data.
+ const VideoContentMetrics* _contentMetrics;
+
+ // Encoder frame sizes and native frame sizes.
+ uint16_t _width;
+ uint16_t _height;
+ uint16_t _nativeWidth;
+ uint16_t _nativeHeight;
+ float _aspectRatio;
+ // Image type and frame rate leve, for the current encoder resolution.
+ uint8_t _imageType;
+ LevelClass _frameRateLevel;
+ // Content class data.
+ VCMContFeature _motion;
+ VCMContFeature _spatial;
+ uint8_t _contentClass;
+ bool _init;
+};
+
+// Resolution settings class
+
+class VCMQmResolution : public VCMQmMethod {
+ public:
+ VCMQmResolution();
+ virtual ~VCMQmResolution();
+
+ // Reset all quantities.
+ virtual void Reset();
+
+ // Reset rate quantities and counters after every SelectResolution() call.
+ void ResetRates();
+
+ // Reset down-sampling state.
+ void ResetDownSamplingState();
+
+ // Get the encoder state.
+ EncoderState GetEncoderState();
+
+ // Initialize after SetEncodingData in media_opt.
+ int Initialize(float bitRate, float userFrameRate,
+ uint16_t width, uint16_t height);
+
+ // Update the encoder frame size.
+ void UpdateCodecFrameSize(uint16_t width, uint16_t height);
+
+ // Update with actual bit rate (size of the latest encoded frame)
+ // and frame type, after every encoded frame.
+ void UpdateEncodedSize(int encodedSize,
+ FrameType encodedFrameType);
+
+ // Update with new target bitrate, actual encoder sent rate, frame_rate,
+ // loss rate: every ~1 sec from SetTargetRates in media_opt.
+ void UpdateRates(float targetBitRate, float encoderSentRate,
+ float incomingFrameRate, uint8_t packetLoss);
+
+ // Extract ST (spatio-temporal) resolution action.
+ // Inputs: qm: Reference to the quality modes pointer.
+ // Output: the spatial and/or temporal scale change.
+ int SelectResolution(VCMResolutionScale** qm);
+
+ // Compute rates for the selection of down-sampling action.
+ void ComputeRatesForSelection();
+
+ // Compute the encoder state.
+ void ComputeEncoderState();
+
+ // Return true if the action is to go back up in resolution.
+ bool GoingUpResolution();
+
+ // Return true if the action is to go down in resolution.
+ bool GoingDownResolution();
+
+ // Check the condition for going up in resolution by the scale factors:
+ // |facWidth|, |facHeight|, |facTemp|.
+ // |scaleFac| is a scale factor for the transition rate.
+ bool ConditionForGoingUp(uint8_t facWidth, uint8_t facHeight,
+ uint8_t facTemp,
+ float scaleFac);
+
+ // Get the bitrate threshold for the resolution action.
+ // The case |facWidth|=|facHeight|=|facTemp|==1 is for down-sampling action.
+ // |scaleFac| is a scale factor for the transition rate.
+ float GetTransitionRate(uint8_t facWidth, uint8_t facHeight,
+ uint8_t facTemp, float scaleFac);
+
+ // Update the downsampling state.
+ void UpdateDownsamplingState(ResolutionAction action);
+
+ void AdjustAction();
+
+ // Select the directional (1x2 or 2x1) spatial down-sampling action.
+ void SelectSpatialDirectionMode(float transRate);
+
+ private:
+ VCMResolutionScale* _qm;
+ // Encoder rate control parameters.
+ float _targetBitRate;
+ float _userFrameRate;
+ float _incomingFrameRate;
+ float _perFrameBandwidth;
+ float _bufferLevel;
+
+ // Data accumulated every ~1sec from MediaOpt.
+ float _sumTargetRate;
+ float _sumIncomingFrameRate;
+ float _sumRateMM;
+ float _sumRateMMSgn;
+ float _sumPacketLoss;
+ // Counters.
+ uint32_t _frameCnt;
+ uint32_t _frameCntDelta;
+ uint32_t _updateRateCnt;
+ uint32_t _lowBufferCnt;
+
+ // Resolution state parameters.
+ uint8_t _stateDecFactorSpatial;
+ uint8_t _stateDecFactorTemp;
+
+ // Quantities used for selection.
+ float _avgTargetRate;
+ float _avgIncomingFrameRate;
+ float _avgRatioBufferLow;
+ float _avgRateMisMatch;
+ float _avgRateMisMatchSgn;
+ float _avgPacketLoss;
+ EncoderState _encoderState;
+};
+
+// Robustness settings class.
+
+class VCMQmRobustness : public VCMQmMethod {
+ public:
+ VCMQmRobustness();
+ ~VCMQmRobustness();
+
+ virtual void Reset();
+
+ // Adjust FEC rate based on content: every ~1 sec from SetTargetRates.
+ // Returns an adjustment factor.
+ float AdjustFecFactor(uint8_t codeRateDelta,
+ float totalRate,
+ float frameRate,
+ uint32_t rttTime,
+ uint8_t packetLoss);
+
+ // Set the UEP protection on/off.
+ bool SetUepProtection(uint8_t codeRateDelta,
+ float totalRate,
+ uint8_t packetLoss,
+ bool frameType);
+
+ private:
+ // Previous state of network parameters.
+ float _prevTotalRate;
+ uint32_t _prevRttTime;
+ uint8_t _prevPacketLoss;
+ uint8_t _prevCodeRateDelta;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
+
diff --git a/trunk/src/modules/video_coding/main/source/qm_select_data.h b/trunk/src/modules/video_coding/main/source/qm_select_data.h
new file mode 100644
index 0000000..d4af642
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/qm_select_data.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+
+/***************************************************************
+*QMSelectData.h
+* This file includes parameters for content-aware media optimization
+****************************************************************/
+
+#include "typedefs.h"
+
+namespace webrtc {
+//
+// PARAMETERS FOR RESOLUTION ADAPTATION
+//
+
+// Initial level of buffer in secs: should corresponds to wrapper settings.
+const float kInitBufferLevel = 0.5f;
+
+// Optimal level of buffer in secs: should corresponds to wrapper settings.
+const float kOptBufferLevel = 0.6f;
+
+// Threshold of (max) buffer size below which we consider too low (underflow).
+const float kPercBufferThr = 0.10f;
+
+// Threshold on the occurrences of low buffer levels.
+const float kMaxBufferLow = 0.5f;
+
+// Threshold on rate mismatch
+const float kMaxRateMisMatch = 0.5f;
+
+// Threshold on amount of under/over encoder shooting.
+const float kRateOverShoot = 0.75f;
+const float kRateUnderShoot = 0.75f;
+
+// Factor for transitional rate for going back up in resolution.
+const float kTransRateScaleUpSpatial = 1.25f;
+const float kTransRateScaleUpTemp = 1.25f;
+const float kTransRateScaleUpSpatialTemp = 1.25f;
+
+// Threshold on packet loss rate, above which favor resolution reduction.
+const float kPacketLossThr = 0.1f;
+
+// Factor for reducing transitonal bitrate under packet loss.
+const float kPacketLossRateFac = 1.0f;
+
+// Maximum possible transitional rate for down-sampling:
+// (units in kbps), for 30fps.
+const uint16_t kMaxRateQm[7] = {
+ 100, // QCIF
+ 250, // CIF
+ 500, // VGA
+ 800, // 4CIF
+ 1000, // 720 HD 4:3,
+ 1500, // 720 HD 16:9
+ 2000 // 1080HD
+};
+
+// Frame rate scale for maximum transition rate.
+const float kFrameRateFac[3] = {
+ 0.7f, // L
+ 1.0f, // H
+ 0.8f // D
+};
+
+// Scale for transitional rate: based on content class
+// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
+const float kScaleTransRateQm[18] = {
+ // 4CIF and lower
+ 0.50f, // L, L
+ 0.50f, // L, H
+ 0.50f, // L, D
+ 0.50f, // H ,L
+ 0.25f, // H, H
+ 0.25f, // H, D
+ 0.50f, // D, L
+ 0.50f, // D, D
+ 0.25f, // D, H
+
+ // over 4CIF: WHD, HD
+ 0.50f, // L, L
+ 0.50f, // L, H
+ 0.50f, // L, D
+ 0.50f, // H ,L
+ 0.25f, // H, H
+ 0.25f, // H, D
+ 0.50f, // D, L
+ 0.50f, // D, D
+ 0.25f, // D, H
+};
+
+// Action for down-sampling:
+// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
+const uint8_t kSpatialAction[9] = {
+ 1, // L, L
+ 1, // L, H
+ 1, // L, D
+ 4, // H ,L
+ 1, // H, H
+ 4, // H, D
+ 4, // D, L
+ 1, // D, H
+ 1, // D, D
+};
+
+const uint8_t kTemporalAction[9] = {
+ 1, // L, L
+ 2, // L, H
+ 2, // L, D
+ 1, // H ,L
+ 2, // H, H
+ 1, // H, D
+ 1, // D, L
+ 2, // D, H
+ 1, // D, D
+};
+
+// Control the total amount of down-sampling allowed.
+const int kMaxSpatialDown = 16;
+const int kMaxTempDown = 4;
+const int kMaxDownSample = 16;
+
+// Minimum image size for a spatial down-sampling.
+const int kMinImageSize= 176 * 144;
+
+// Minimum frame rate for temporal down-sampling:
+// no frame rate reduction if incomingFrameRate <= MIN_FRAME_RATE
+const int kMinFrameRate = 8;
+
+// Boundaries for the closest standard frame size
+const uint32_t kFrameSizeTh[6] = {
+ 63360, // between 176*144 and 352*288
+ 204288, // between 352*288 and 640*480
+ 356352, // between 640*480 and 704*576
+ 548352, // between 704*576 and 960*720
+ 806400, // between 960*720 and 1280*720
+ 1497600, // between 1280*720 and 1920*1080
+};
+
+//
+// PARAMETERS FOR FEC ADJUSTMENT: TODO (marpan)
+//
+
+//
+// PARAMETETS FOR SETTING LOW/HIGH STATES OF CONTENT METRICS:
+//
+
+// Thresholds for frame rate:
+const int kLowFrameRate = 10;
+const int kHighFrameRate = 25;
+
+// Thresholds for motion: motion level is from NFD
+const float kHighMotionNfd = 0.075f;
+const float kLowMotionNfd = 0.04f;
+
+// Thresholds for spatial prediction error:
+// this is applied on the min(2x2,1x2,2x1)
+const float kHighTexture = 0.035f;
+const float kLowTexture = 0.025f;
+
+// Used to reduce thresholds for larger/HD scenes: correction factor since
+// higher correlation in HD scenes means lower spatial prediction error.
+const float kScaleTexture = 0.9f;
+
+// percentage reduction in transitional bitrate for 2x2 selected over 1x2/2x1
+const float kRateRedSpatial2X2 = 0.6f;
+
+const float kSpatialErr2x2VsHoriz = 0.1f; // percentage to favor 2x2 over H
+const float kSpatialErr2X2VsVert = 0.1f; // percentage to favor 2x2 over V
+const float kSpatialErrVertVsHoriz = 0.1f; // percentage to favor H over V
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+
diff --git a/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc b/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc
new file mode 100644
index 0000000..e0ab7bf
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc
@@ -0,0 +1,834 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes unit tests the QmResolution class
+ * In particular, for the selection of spatial and/or temporal down-sampling.
+ */
+
+#include <gtest/gtest.h>
+
+#include "modules/video_coding/main/source/qm_select.h"
+#include "modules/interface/module_common_types.h"
+
+namespace webrtc {
+
+class QmSelectTest : public ::testing::Test {
+ protected:
+ QmSelectTest()
+ : qm_resolution_(new VCMQmResolution()),
+ content_metrics_(new VideoContentMetrics()),
+ qm_scale_(NULL) {
+ }
+ VCMQmResolution* qm_resolution_;
+ VideoContentMetrics* content_metrics_;
+ VCMResolutionScale* qm_scale_;
+
+ void InitQmNativeData(float initial_bit_rate, int user_frame_rate,
+ int native_width, int native_height);
+
+ void UpdateQmEncodedFrame(int* encoded_size, int num_updates);
+
+ void UpdateQmRateData(int* target_rate,
+ int* encoder_sent_rate,
+ int* incoming_frame_rate,
+ uint8_t* fraction_lost,
+ int num_updates);
+
+ void UpdateQmContentData(float motion_metric,
+ float spatial_metric,
+ float spatial_metric_horiz,
+ float spatial_metric_vert);
+
+ bool IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
+ uint8_t fac_width,
+ uint8_t fac_height,
+ uint8_t fac_temp);
+
+ void TearDown() {
+ delete qm_resolution_;
+ delete content_metrics_;
+ }
+};
+
+TEST_F(QmSelectTest, HandleInputs) {
+ // Expect parameter error. Initialize with invalid inputs.
+ EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 0, 640, 480));
+ EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 640, 0));
+ EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 0, 480));
+
+ // Expect uninitialized error.: No valid initialization before selection.
+ EXPECT_EQ(-7, qm_resolution_->SelectResolution(&qm_scale_));
+
+ VideoContentMetrics* content_metrics = NULL;
+ EXPECT_EQ(0, qm_resolution_->Initialize(1000, 30, 640, 480));
+ qm_resolution_->UpdateContent(content_metrics);
+ // Content metrics are NULL: Expect success and no down-sampling action.
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// No down-sampling action at high rates.
+TEST_F(QmSelectTest, NoActionHighRate) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(800, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {800, 800, 800};
+ int encoder_sent_rate[] = {800, 800, 800};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ UpdateQmContentData(0.01f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// Rate is well below transition, down-sampling action is taken,
+// depending on the content state.
+TEST_F(QmSelectTest, DownActionLowRate) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial: 2x2 spatial expected.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, low spatial: no action expected: content is too low.
+ UpdateQmContentData(0.01f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Medium motion, low spatial: 2x2 spatial expected.
+ UpdateQmContentData(0.06f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // High motion, high spatial: 1/2 temporal expected.
+ UpdateQmContentData(0.1f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(4, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, high spatial: 1/2 temporal expected.
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Medium motion, high spatial: 1/2 temporal expected.
+ UpdateQmContentData(0.06f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ qm_resolution_->ResetDownSamplingState();
+ // High motion, medium spatial: 2x2 spatial expected.
+ UpdateQmContentData(0.1f, 0.03f, 0.03f, 0.03f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, medium spatial: high frame rate, so 1/2 temporal expected.
+ UpdateQmContentData(0.01f, 0.03f, 0.03f, 0.03f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Medium motion, medium spatial: high frame rate, so 1/2 temporal expected.
+ UpdateQmContentData(0.06f, 0.03f, 0.03f, 0.03f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(8, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+}
+
+// Rate mis-match is high, and we have over-shooting.
+// since target rate is below max for down-sampling, down-sampling is selected.
+TEST_F(QmSelectTest, DownActionHighRateMMOvershoot) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(450, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {450, 450, 450};
+ int encoder_sent_rate[] = {900, 900, 900};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, high spatial
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+}
+
+// Rate mis-match is high, target rate is below max for down-sampling,
+// but since we have consistent under-shooting, no down-sampling action.
+TEST_F(QmSelectTest, NoActionHighRateMMUndershoot) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(450, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {450, 450, 450};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, high spatial
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// Buffer is underflowing, and target rate is below max for down-sampling,
+// so action is taken.
+TEST_F(QmSelectTest, DownActionBufferUnderflow) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(450, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update with encoded size over a number of frames.
+ // per-frame bandwidth = 15 = 450/30: simulate (decoder) buffer underflow:
+ int encoded_size[] = {200, 100, 50, 30, 60, 40, 20, 30, 20, 40};
+ UpdateQmEncodedFrame(encoded_size, 10);
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {450, 450, 450};
+ int encoder_sent_rate[] = {450, 450, 450};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, high spatial
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+}
+
+// Target rate is below max for down-sampling, but buffer level is stable,
+// so no action is taken.
+TEST_F(QmSelectTest, NoActionBufferStable) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(450, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update with encoded size over a number of frames.
+ // per-frame bandwidth = 15 = 450/30: simulate stable (decoder) buffer levels.
+ int32_t encoded_size[] = {40, 10, 10, 16, 18, 20, 17, 20, 16, 15};
+ UpdateQmEncodedFrame(encoded_size, 10);
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {450, 450, 450};
+ int encoder_sent_rate[] = {450, 450, 450};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+
+ qm_resolution_->ResetDownSamplingState();
+ // Low motion, high spatial
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// Very low rate, but no spatial down-sampling below some size (QCIF).
+TEST_F(QmSelectTest, LimitDownSpatialAction) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(10, 30, 176, 144);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 176;
+ uint16_t codec_height = 144;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(0, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {10, 10, 10};
+ int encoder_sent_rate[] = {10, 10, 10};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// Very low rate, but no frame reduction below some frame_rate (8fps).
+TEST_F(QmSelectTest, LimitDownTemporalAction) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(10, 8, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {10, 10, 10};
+ int encoder_sent_rate[] = {10, 10, 10};
+ int incoming_frame_rate[] = {8, 8, 8};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // Low motion, medium spatial.
+ UpdateQmContentData(0.01f, 0.03f, 0.03f, 0.03f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+// Two stages: spatial down-sample and then back up spatially,
+// as rate as increased.
+TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset and go up in rate: expected to go back up.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(320, 240);
+ EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {400, 400, 400, 400, 400};
+ int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
+ int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 1));
+}
+
+// Two stages: spatial down-sample and then back up spatially, since encoder
+// is under-shooting target even though rate has not increased much.
+TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset rates and simulate under-shooting scenario.: expect to go back up.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(320, 240);
+ EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {200, 200, 200, 200, 200};
+ int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
+ int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 1));
+}
+
+// Two stages: spatial down-sample and then no action to go up,
+// as encoding rate mis-match is too high.
+TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset and simulate large rate mis-match: expect no action to go back up.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(320, 240);
+ EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {400, 400, 400, 400, 400};
+ int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
+ int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+// Two stages: temporally down-sample and then back up temporally,
+// as rate as increased.
+TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // Low motion, high spatial.
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ // Reset rates and go up in rate: expect to go back up.
+ qm_resolution_->ResetRates();
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {400, 400, 400, 400, 400};
+ int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
+ int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 0));
+}
+
+// Two stages: temporal down-sample and then back up temporally, since encoder
+// is under-shooting target even though rate has not increased much.
+TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // Low motion, high spatial.
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ // Reset rates and simulate under-shooting scenario.: expect to go back up.
+ qm_resolution_->ResetRates();
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {200, 200, 200, 200, 200};
+ int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
+ int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 0));
+}
+
+// Two stages: temporal down-sample and then no action to go up,
+// as encoding rate mis-match is too high.
+TEST_F(QmSelectTest, 2StageDownTemporalNoActionUp) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // Low motion, high spatial.
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ // Reset and simulate large rate mis-match: expect no action to go back up.
+ qm_resolution_->ResetRates();
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {600, 600, 600, 600, 600};
+ int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
+ int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+// 3 stages: spatial down-sample, followed by temporal down-sample,
+// and then go up to full state, as encoding rate has increased.
+TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(100, 30, 640, 480);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 640;
+ uint16_t codec_height = 480;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {100, 100, 100};
+ int encoder_sent_rate[] = {100, 100, 100};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset rate and change content data: expect temporal down-sample.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(320, 240);
+ EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // Low motion, high spatial.
+ UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
+
+ // Reset rates and go high up in rate: expect to go back up both spatial
+ // and temporally.
+ qm_resolution_->ResetRates();
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {1000, 1000, 1000, 1000, 1000};
+ int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
+ int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 0));
+}
+
+// No down-sampling below some totol amount (factor of 16)
+TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
+ // Initialize with bitrate, frame rate, and native system width/height.
+ InitQmNativeData(400, 30, 1280, 720);
+
+ // Update with encoder frame size.
+ uint16_t codec_width = 1280;
+ uint16_t codec_height = 720;
+ qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
+ EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
+
+ // Update rates for a sequence of intervals.
+ int target_rate[] = {400, 400, 400};
+ int encoder_sent_rate[] = {400, 400, 400};
+ int incoming_frame_rate[] = {30, 30, 30};
+ uint8_t fraction_lost[] = {10, 10, 10};
+ UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
+ fraction_lost, 3);
+
+ // Update content: motion level, and 3 spatial prediction errors.
+ // High motion, low spatial: 2x2 spatial expected.
+ UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset and lower rates to get another spatial action.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(640, 360);
+ EXPECT_EQ(2, qm_resolution_->GetImageType(640, 360));
+ // Update rates for a sequence of intervals.
+ int target_rate2[] = {100, 100, 100, 100, 100};
+ int encoder_sent_rate2[] = {100, 100, 100, 100, 100};
+ int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
+ uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
+ fraction_lost2, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
+
+ // Reset and go to low rate: no action should be taken,
+ // we went down too much already.
+ qm_resolution_->ResetRates();
+ qm_resolution_->UpdateCodecFrameSize(320, 180);
+ EXPECT_EQ(0, qm_resolution_->GetImageType(320, 180));
+ // Update rates for a sequence of intervals.
+ int target_rate3[] = {10, 10, 10, 10, 10};
+ int encoder_sent_rate3[] = {10, 10, 10, 10, 10};
+ int incoming_frame_rate3[] = {30, 30, 30, 30, 30};
+ uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
+ UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
+ fraction_lost3, 5);
+ EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
+ EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
+ EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
+ EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
+}
+
+void QmSelectTest::InitQmNativeData(float initial_bit_rate,
+ int user_frame_rate,
+ int native_width,
+ int native_height) {
+ EXPECT_EQ(0, qm_resolution_->Initialize(initial_bit_rate, user_frame_rate,
+ native_width, native_height));
+}
+
+void QmSelectTest::UpdateQmContentData(float motion_metric,
+ float spatial_metric,
+ float spatial_metric_horiz,
+ float spatial_metric_vert) {
+ content_metrics_->motion_magnitude = motion_metric;
+ content_metrics_->spatial_pred_err = spatial_metric;
+ content_metrics_->spatial_pred_err_h = spatial_metric_horiz;
+ content_metrics_->spatial_pred_err_v = spatial_metric_vert;
+ qm_resolution_->UpdateContent(content_metrics_);
+}
+
+void QmSelectTest::UpdateQmEncodedFrame(int* encoded_size, int num_updates) {
+ FrameType frame_type = kVideoFrameDelta;
+ for (int i = 0; i < num_updates; i++) {
+ // Convert to bytes.
+ int32_t encoded_size_update = 1000 * encoded_size[i] / 8;
+ qm_resolution_->UpdateEncodedSize(encoded_size_update, frame_type);
+ }
+}
+
+void QmSelectTest::UpdateQmRateData(int* target_rate,
+ int* encoder_sent_rate,
+ int* incoming_frame_rate,
+ uint8_t* fraction_lost,
+ int num_updates) {
+ for (int i = 0; i < num_updates; i++) {
+ float target_rate_update = target_rate[i];
+ float encoder_sent_rate_update = encoder_sent_rate[i];
+ float incoming_frame_rate_update = incoming_frame_rate[i];
+ uint8_t fraction_lost_update = fraction_lost[i];
+ qm_resolution_->UpdateRates(target_rate_update,
+ encoder_sent_rate_update,
+ incoming_frame_rate_update,
+ fraction_lost_update);
+ }
+}
+
+// Check is the selected action from the QmResolution class is the same
+// as the expected scales from |fac_width|, |fac_height|, |fac_temp|.
+bool QmSelectTest::IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
+ uint8_t fac_width,
+ uint8_t fac_height,
+ uint8_t fac_temp) {
+ if (qm_scale->spatialWidthFact == fac_width &&
+ qm_scale->spatialHeightFact == fac_height &&
+ qm_scale->temporalFact == fac_temp) {
+ return true;
+ } else {
+ return false;
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/receiver.cc b/trunk/src/modules/video_coding/main/source/receiver.cc
new file mode 100644
index 0000000..6be5336
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/receiver.cc
@@ -0,0 +1,493 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/receiver.h"
+
+#include <assert.h>
+
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_coding/main/source/encoded_frame.h"
+#include "modules/video_coding/main/source/internal_defines.h"
+#include "modules/video_coding/main/source/media_opt_util.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+VCMReceiver::VCMReceiver(VCMTiming& timing,
+ TickTimeBase* clock,
+ WebRtc_Word32 vcmId,
+ WebRtc_Word32 receiverId,
+ bool master)
+ : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _vcmId(vcmId),
+ _clock(clock),
+ _receiverId(receiverId),
+ _master(master),
+ _jitterBuffer(_clock, vcmId, receiverId, master),
+ _timing(timing),
+ _renderWaitEvent(*new VCMEvent()),
+ _state(kPassive) {}
+
+VCMReceiver::~VCMReceiver()
+{
+ _renderWaitEvent.Set();
+ delete &_renderWaitEvent;
+ delete _critSect;
+}
+
+void
+VCMReceiver::Reset()
+{
+ CriticalSectionScoped cs(_critSect);
+ if (!_jitterBuffer.Running())
+ {
+ _jitterBuffer.Start();
+ }
+ else
+ {
+ _jitterBuffer.Flush();
+ }
+ _renderWaitEvent.Reset();
+ if (_master)
+ {
+ _state = kReceiving;
+ }
+ else
+ {
+ _state = kPassive;
+ }
+}
+
+WebRtc_Word32
+VCMReceiver::Initialize()
+{
+ CriticalSectionScoped cs(_critSect);
+ Reset();
+ if (!_master)
+ {
+ SetNackMode(kNoNack);
+ }
+ return VCM_OK;
+}
+
+void VCMReceiver::UpdateRtt(WebRtc_UWord32 rtt)
+{
+ _jitterBuffer.UpdateRtt(rtt);
+}
+
+WebRtc_Word32
+VCMReceiver::InsertPacket(const VCMPacket& packet,
+ WebRtc_UWord16 frameWidth,
+ WebRtc_UWord16 frameHeight)
+{
+ // Find an empty frame
+ VCMEncodedFrame *buffer = NULL;
+ const WebRtc_Word32 error = _jitterBuffer.GetFrame(packet, buffer);
+ if (error == VCM_OLD_PACKET_ERROR)
+ {
+ return VCM_OK;
+ }
+ else if (error != VCM_OK)
+ {
+ return error;
+ }
+ assert(buffer);
+ {
+ CriticalSectionScoped cs(_critSect);
+
+ if (frameWidth && frameHeight)
+ {
+ buffer->SetEncodedSize(static_cast<WebRtc_UWord32>(frameWidth),
+ static_cast<WebRtc_UWord32>(frameHeight));
+ }
+
+ if (_master)
+ {
+ // Only trace the primary receiver to make it possible
+ // to parse and plot the trace file.
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Packet seqNo %u of frame %u at %u",
+ packet.seqNum, packet.timestamp,
+ MaskWord64ToUWord32(_clock->MillisecondTimestamp()));
+ }
+
+ const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
+
+ WebRtc_Word64 renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
+
+ if (renderTimeMs < 0)
+ {
+ // Render time error. Assume that this is due to some change in
+ // the incoming video stream and reset the JB and the timing.
+ _jitterBuffer.Flush();
+ _timing.Reset(_clock->MillisecondTimestamp());
+ return VCM_FLUSH_INDICATOR;
+ }
+ else if (renderTimeMs < nowMs - kMaxVideoDelayMs)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "This frame should have been rendered more than %u ms ago."
+ "Flushing jitter buffer and resetting timing.", kMaxVideoDelayMs);
+ _jitterBuffer.Flush();
+ _timing.Reset(_clock->MillisecondTimestamp());
+ return VCM_FLUSH_INDICATOR;
+ }
+ else if (_timing.TargetVideoDelay() > kMaxVideoDelayMs)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "More than %u ms target delay. Flushing jitter buffer and resetting timing.",
+ kMaxVideoDelayMs);
+ _jitterBuffer.Flush();
+ _timing.Reset(_clock->MillisecondTimestamp());
+ return VCM_FLUSH_INDICATOR;
+ }
+
+ // First packet received belonging to this frame.
+ if (buffer->Length() == 0)
+ {
+ const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
+ if (_master)
+ {
+ // Only trace the primary receiver to make it possible to parse and plot the trace file.
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "First packet of frame %u at %u", packet.timestamp,
+ MaskWord64ToUWord32(nowMs));
+ }
+ renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
+ if (renderTimeMs >= 0)
+ {
+ buffer->SetRenderTime(renderTimeMs);
+ }
+ else
+ {
+ buffer->SetRenderTime(nowMs);
+ }
+ }
+
+ // Insert packet into the jitter buffer
+ // both media and empty packets
+ const VCMFrameBufferEnum
+ ret = _jitterBuffer.InsertPacket(buffer, packet);
+ if (ret == kFlushIndicator) {
+ return VCM_FLUSH_INDICATOR;
+ } else if (ret < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
+ VCMId(_vcmId, _receiverId),
+ "Error inserting packet seqNo=%u, timeStamp=%u",
+ packet.seqNum, packet.timestamp);
+ return VCM_JITTER_BUFFER_ERROR;
+ }
+ }
+ return VCM_OK;
+}
+
+VCMEncodedFrame* VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64& nextRenderTimeMs,
+ bool renderTiming,
+ VCMReceiver* dualReceiver)
+{
+ // No need to enter the critical section here since the jitter buffer
+ // is thread-safe.
+ FrameType incomingFrameType = kVideoFrameDelta;
+ nextRenderTimeMs = -1;
+ const WebRtc_Word64 startTimeMs = _clock->MillisecondTimestamp();
+ WebRtc_Word64 ret = _jitterBuffer.GetNextTimeStamp(maxWaitTimeMs,
+ incomingFrameType,
+ nextRenderTimeMs);
+ if (ret < 0)
+ {
+ // No timestamp in jitter buffer at the moment
+ return NULL;
+ }
+ const WebRtc_UWord32 timeStamp = static_cast<WebRtc_UWord32>(ret);
+
+ // Update the timing
+ _timing.SetRequiredDelay(_jitterBuffer.GetEstimatedJitterMS());
+ _timing.UpdateCurrentDelay(timeStamp);
+
+ const WebRtc_Word32 tempWaitTime = maxWaitTimeMs -
+ static_cast<WebRtc_Word32>(_clock->MillisecondTimestamp() - startTimeMs);
+ WebRtc_UWord16 newMaxWaitTime = static_cast<WebRtc_UWord16>(VCM_MAX(tempWaitTime, 0));
+
+ VCMEncodedFrame* frame = NULL;
+
+ if (renderTiming)
+ {
+ frame = FrameForDecoding(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
+ }
+ else
+ {
+ frame = FrameForRendering(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
+ }
+
+ if (frame != NULL)
+ {
+ bool retransmitted = false;
+ const WebRtc_Word64 lastPacketTimeMs =
+ _jitterBuffer.LastPacketTime(frame, retransmitted);
+ if (lastPacketTimeMs >= 0 && !retransmitted)
+ {
+ // We don't want to include timestamps which have suffered from retransmission
+ // here, since we compensate with extra retransmission delay within
+ // the jitter estimate.
+ _timing.IncomingTimestamp(timeStamp, lastPacketTimeMs);
+ }
+ if (dualReceiver != NULL)
+ {
+ dualReceiver->UpdateState(*frame);
+ }
+ }
+ return frame;
+}
+
+VCMEncodedFrame*
+VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64 nextRenderTimeMs,
+ VCMReceiver* dualReceiver)
+{
+ // How long can we wait until we must decode the next frame
+ WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
+ _clock->MillisecondTimestamp());
+
+ // Try to get a complete frame from the jitter buffer
+ VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
+
+ if (frame == NULL && maxWaitTimeMs == 0 && waitTimeMs > 0)
+ {
+ // If we're not allowed to wait for frames to get complete we must
+ // calculate if it's time to decode, and if it's not we will just return
+ // for now.
+ return NULL;
+ }
+
+ if (frame == NULL && VCM_MIN(waitTimeMs, maxWaitTimeMs) == 0)
+ {
+ // No time to wait for a complete frame,
+ // check if we have an incomplete
+ const bool dualReceiverEnabledAndPassive = (dualReceiver != NULL &&
+ dualReceiver->State() == kPassive &&
+ dualReceiver->NackMode() == kNackInfinite);
+ if (dualReceiverEnabledAndPassive &&
+ !_jitterBuffer.CompleteSequenceWithNextFrame())
+ {
+ // Jitter buffer state might get corrupt with this frame.
+ dualReceiver->CopyJitterBufferStateFromReceiver(*this);
+ frame = _jitterBuffer.GetFrameForDecoding();
+ assert(frame);
+ } else {
+ frame = _jitterBuffer.GetFrameForDecoding();
+ }
+ }
+ if (frame == NULL)
+ {
+ // Wait for a complete frame
+ frame = _jitterBuffer.GetCompleteFrameForDecoding(maxWaitTimeMs);
+ }
+ if (frame == NULL)
+ {
+ // Get an incomplete frame
+ if (_timing.MaxWaitingTime(nextRenderTimeMs,
+ _clock->MillisecondTimestamp()) > 0)
+ {
+ // Still time to wait for a complete frame
+ return NULL;
+ }
+
+ // No time left to wait, we must decode this frame now.
+ const bool dualReceiverEnabledAndPassive = (dualReceiver != NULL &&
+ dualReceiver->State() == kPassive &&
+ dualReceiver->NackMode() == kNackInfinite);
+ if (dualReceiverEnabledAndPassive &&
+ !_jitterBuffer.CompleteSequenceWithNextFrame())
+ {
+ // Jitter buffer state might get corrupt with this frame.
+ dualReceiver->CopyJitterBufferStateFromReceiver(*this);
+ }
+
+ frame = _jitterBuffer.GetFrameForDecoding();
+ }
+ return frame;
+}
+
+VCMEncodedFrame*
+VCMReceiver::FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64 nextRenderTimeMs,
+ VCMReceiver* dualReceiver)
+{
+ // How long MUST we wait until we must decode the next frame. This is different for the case
+ // where we have a renderer which can render at a specified time. Here we must wait as long
+ // as possible before giving the frame to the decoder, which will render the frame as soon
+ // as it has been decoded.
+ WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
+ _clock->MillisecondTimestamp());
+ if (maxWaitTimeMs < waitTimeMs)
+ {
+ // If we're not allowed to wait until the frame is supposed to be rendered
+ // we will have to return NULL for now.
+ return NULL;
+ }
+ // Wait until it's time to render
+ _renderWaitEvent.Wait(waitTimeMs);
+
+ // Get a complete frame if possible
+ VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
+
+ if (frame == NULL)
+ {
+ // Get an incomplete frame
+ const bool dualReceiverEnabledAndPassive = dualReceiver != NULL &&
+ dualReceiver->State() == kPassive &&
+ dualReceiver->NackMode() == kNackInfinite;
+ if (dualReceiverEnabledAndPassive && !_jitterBuffer.CompleteSequenceWithNextFrame())
+ {
+ // Jitter buffer state might get corrupt with this frame.
+ dualReceiver->CopyJitterBufferStateFromReceiver(*this);
+ }
+
+ frame = _jitterBuffer.GetFrameForDecoding();
+ }
+ return frame;
+}
+
+void
+VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame)
+{
+ _jitterBuffer.ReleaseFrame(frame);
+}
+
+WebRtc_Word32
+VCMReceiver::ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate)
+{
+ const WebRtc_Word32 ret = _jitterBuffer.GetUpdate(frameRate, bitRate);
+ bitRate /= 1000; // Should be in kbps
+ return ret;
+}
+
+WebRtc_Word32
+VCMReceiver::ReceivedFrameCount(VCMFrameCount& frameCount) const
+{
+ return _jitterBuffer.GetFrameStatistics(frameCount.numDeltaFrames,
+ frameCount.numKeyFrames);
+}
+
+WebRtc_UWord32 VCMReceiver::DiscardedPackets() const {
+ return _jitterBuffer.DiscardedPackets();
+}
+
+void
+VCMReceiver::SetNackMode(VCMNackMode nackMode)
+{
+ CriticalSectionScoped cs(_critSect);
+ // Default to always having NACK enabled in hybrid mode.
+ _jitterBuffer.SetNackMode(nackMode, kLowRttNackMs, -1);
+ if (!_master)
+ {
+ _state = kPassive; // The dual decoder defaults to passive
+ }
+}
+
+VCMNackMode
+VCMReceiver::NackMode() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return _jitterBuffer.GetNackMode();
+}
+
+VCMNackStatus
+VCMReceiver::NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size)
+{
+ bool extended = false;
+ WebRtc_UWord16 nackListSize = 0;
+ WebRtc_UWord16* internalNackList = _jitterBuffer.GetNackList(nackListSize, extended);
+ if (internalNackList == NULL && nackListSize == 0xffff)
+ {
+ // This combination is used to trigger key frame requests.
+ size = 0;
+ return kNackKeyFrameRequest;
+ }
+ if (nackListSize > size)
+ {
+ size = nackListSize;
+ return kNackNeedMoreMemory;
+ }
+ if (internalNackList != NULL && nackListSize > 0) {
+ memcpy(nackList, internalNackList, nackListSize * sizeof(WebRtc_UWord16));
+ }
+ size = nackListSize;
+ return kNackOk;
+}
+
+// Decide whether we should change decoder state. This should be done if the dual decoder
+// has caught up with the decoder decoding with packet losses.
+bool
+VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const
+{
+ if (dualFrame == NULL)
+ {
+ return false;
+ }
+ if (_jitterBuffer.LastDecodedTimestamp() == dualFrame->TimeStamp())
+ {
+ dualReceiver.UpdateState(kWaitForPrimaryDecode);
+ return true;
+ }
+ return false;
+}
+
+void
+VCMReceiver::CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver)
+{
+ _jitterBuffer.CopyFrom(receiver._jitterBuffer);
+}
+
+VCMReceiverState
+VCMReceiver::State() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return _state;
+}
+
+void
+VCMReceiver::UpdateState(VCMReceiverState newState)
+{
+ CriticalSectionScoped cs(_critSect);
+ assert(!(_state == kPassive && newState == kWaitForPrimaryDecode));
+// assert(!(_state == kReceiving && newState == kPassive));
+ _state = newState;
+}
+
+void
+VCMReceiver::UpdateState(VCMEncodedFrame& frame)
+{
+ if (_jitterBuffer.GetNackMode() == kNoNack)
+ {
+ // Dual decoder mode has not been enabled.
+ return;
+ }
+ // Update the dual receiver state
+ if (frame.Complete() && frame.FrameType() == kVideoFrameKey)
+ {
+ UpdateState(kPassive);
+ }
+ if (State() == kWaitForPrimaryDecode &&
+ frame.Complete() && !frame.MissingFrame())
+ {
+ UpdateState(kPassive);
+ }
+ if (frame.MissingFrame() || !frame.Complete())
+ {
+ // State was corrupted, enable dual receiver.
+ UpdateState(kReceiving);
+ }
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/receiver.h b/trunk/src/modules/video_coding/main/source/receiver.h
new file mode 100644
index 0000000..0081ed1
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/receiver.h
@@ -0,0 +1,101 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+
+#include "critical_section_wrapper.h"
+#include "jitter_buffer.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "timing.h"
+#include "packet.h"
+
+namespace webrtc
+{
+
+class VCMEncodedFrame;
+
+enum VCMNackStatus
+{
+ kNackOk,
+ kNackNeedMoreMemory,
+ kNackKeyFrameRequest
+};
+
+
+enum VCMReceiverState
+{
+ kReceiving,
+ kPassive,
+ kWaitForPrimaryDecode
+};
+
+class VCMReceiver
+{
+public:
+ VCMReceiver(VCMTiming& timing,
+ TickTimeBase* clock,
+ WebRtc_Word32 vcmId = -1,
+ WebRtc_Word32 receiverId = -1,
+ bool master = true);
+ ~VCMReceiver();
+
+ void Reset();
+ WebRtc_Word32 Initialize();
+ void UpdateRtt(WebRtc_UWord32 rtt);
+ WebRtc_Word32 InsertPacket(const VCMPacket& packet,
+ WebRtc_UWord16 frameWidth,
+ WebRtc_UWord16 frameHeight);
+ VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64& nextRenderTimeMs,
+ bool renderTiming = true,
+ VCMReceiver* dualReceiver = NULL);
+ void ReleaseFrame(VCMEncodedFrame* frame);
+ WebRtc_Word32 ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate);
+ WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
+ WebRtc_UWord32 DiscardedPackets() const;
+
+ // NACK
+ void SetNackMode(VCMNackMode nackMode);
+ VCMNackMode NackMode() const;
+ VCMNackStatus NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
+
+ // Dual decoder
+ bool DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const;
+ VCMReceiverState State() const;
+
+private:
+ VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64 nextrenderTimeMs,
+ VCMReceiver* dualReceiver);
+ VCMEncodedFrame* FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
+ WebRtc_Word64 nextrenderTimeMs,
+ VCMReceiver* dualReceiver);
+ void CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver);
+ void UpdateState(VCMReceiverState newState);
+ void UpdateState(VCMEncodedFrame& frame);
+ static WebRtc_Word32 GenerateReceiverId();
+
+ CriticalSectionWrapper* _critSect;
+ WebRtc_Word32 _vcmId;
+ TickTimeBase* _clock;
+ WebRtc_Word32 _receiverId;
+ bool _master;
+ VCMJitterBuffer _jitterBuffer;
+ VCMTiming& _timing;
+ VCMEvent& _renderWaitEvent;
+ VCMReceiverState _state;
+
+ static WebRtc_Word32 _receiverIdCounter;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
diff --git a/trunk/src/modules/video_coding/main/source/rtt_filter.cc b/trunk/src/modules/video_coding/main/source/rtt_filter.cc
new file mode 100644
index 0000000..36f7660
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/rtt_filter.cc
@@ -0,0 +1,214 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "rtt_filter.h"
+
+#include <cmath>
+#include <stdlib.h>
+#include <string.h>
+
+namespace webrtc {
+
+VCMRttFilter::VCMRttFilter(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId)
+:
+_vcmId(vcmId),
+_receiverId(receiverId),
+_filtFactMax(35),
+_jumpStdDevs(2.5),
+_driftStdDevs(3.5),
+_detectThreshold(kMaxDriftJumpCount)
+{
+ Reset();
+}
+
+VCMRttFilter&
+VCMRttFilter::operator=(const VCMRttFilter& rhs)
+{
+ if (this != &rhs)
+ {
+ _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
+ _avgRtt = rhs._avgRtt;
+ _varRtt = rhs._varRtt;
+ _maxRtt = rhs._maxRtt;
+ _filtFactCount = rhs._filtFactCount;
+ _jumpCount = rhs._jumpCount;
+ _driftCount = rhs._driftCount;
+ memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
+ memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
+ }
+ return *this;
+}
+
+void
+VCMRttFilter::Reset()
+{
+ _gotNonZeroUpdate = false;
+ _avgRtt = 0;
+ _varRtt = 0;
+ _maxRtt = 0;
+ _filtFactCount = 1;
+ _jumpCount = 0;
+ _driftCount = 0;
+ memset(_jumpBuf, 0, kMaxDriftJumpCount);
+ memset(_driftBuf, 0, kMaxDriftJumpCount);
+}
+
+void
+VCMRttFilter::Update(WebRtc_UWord32 rttMs)
+{
+ if (!_gotNonZeroUpdate)
+ {
+ if (rttMs == 0)
+ {
+ return;
+ }
+ _gotNonZeroUpdate = true;
+ }
+
+ // Sanity check
+ if (rttMs > 3000)
+ {
+ rttMs = 3000;
+ }
+
+ double filtFactor = 0;
+ if (_filtFactCount > 1)
+ {
+ filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
+ }
+ _filtFactCount++;
+ if (_filtFactCount > _filtFactMax)
+ {
+ // This prevents filtFactor from going above
+ // (_filtFactMax - 1) / _filtFactMax,
+ // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
+ _filtFactCount = _filtFactMax;
+ }
+ double oldAvg = _avgRtt;
+ double oldVar = _varRtt;
+ _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
+ _varRtt = filtFactor * _varRtt + (1 - filtFactor) *
+ (rttMs - _avgRtt) * (rttMs - _avgRtt);
+ _maxRtt = VCM_MAX(rttMs, _maxRtt);
+ if (!JumpDetection(rttMs) || !DriftDetection(rttMs))
+ {
+ // In some cases we don't want to update the statistics
+ _avgRtt = oldAvg;
+ _varRtt = oldVar;
+ }
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "RttFilter Update: sample=%u avgRtt=%f varRtt=%f maxRtt=%u",
+ rttMs, _avgRtt, _varRtt, _maxRtt);
+}
+
+bool
+VCMRttFilter::JumpDetection(WebRtc_UWord32 rttMs)
+{
+ double diffFromAvg = _avgRtt - rttMs;
+ if (abs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
+ {
+ int diffSign = (diffFromAvg >= 0) ? 1 : -1;
+ int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
+ if (diffSign != jumpCountSign)
+ {
+ // Since the signs differ the samples currently
+ // in the buffer is useless as they represent a
+ // jump in a different direction.
+ _jumpCount = 0;
+ }
+ if (abs(_jumpCount) < kMaxDriftJumpCount)
+ {
+ // Update the buffer used for the short time
+ // statistics.
+ // The sign of the diff is used for updating the counter since
+ // we want to use the same buffer for keeping track of when
+ // the RTT jumps down and up.
+ _jumpBuf[abs(_jumpCount)] = rttMs;
+ _jumpCount += diffSign;
+ }
+ if (abs(_jumpCount) >= _detectThreshold)
+ {
+ // Detected an RTT jump
+ ShortRttFilter(_jumpBuf, abs(_jumpCount));
+ _filtFactCount = _detectThreshold + 1;
+ _jumpCount = 0;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Detected an RTT jump");
+ }
+ else
+ {
+ return false;
+ }
+ }
+ else
+ {
+ _jumpCount = 0;
+ }
+ return true;
+}
+
+bool
+VCMRttFilter::DriftDetection(WebRtc_UWord32 rttMs)
+{
+ if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt))
+ {
+ if (_driftCount < kMaxDriftJumpCount)
+ {
+ // Update the buffer used for the short time
+ // statistics.
+ _driftBuf[_driftCount] = rttMs;
+ _driftCount++;
+ }
+ if (_driftCount >= _detectThreshold)
+ {
+ // Detected an RTT drift
+ ShortRttFilter(_driftBuf, _driftCount);
+ _filtFactCount = _detectThreshold + 1;
+ _driftCount = 0;
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+ "Detected an RTT drift");
+ }
+ }
+ else
+ {
+ _driftCount = 0;
+ }
+ return true;
+}
+
+void
+VCMRttFilter::ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length)
+{
+ if (length == 0)
+ {
+ return;
+ }
+ _maxRtt = 0;
+ _avgRtt = 0;
+ for (WebRtc_UWord32 i=0; i < length; i++)
+ {
+ if (buf[i] > _maxRtt)
+ {
+ _maxRtt = buf[i];
+ }
+ _avgRtt += buf[i];
+ }
+ _avgRtt = _avgRtt / static_cast<double>(length);
+}
+
+WebRtc_UWord32
+VCMRttFilter::RttMs() const
+{
+ return static_cast<WebRtc_UWord32>(_maxRtt + 0.5);
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/rtt_filter.h b/trunk/src/modules/video_coding/main/source/rtt_filter.h
new file mode 100644
index 0000000..5ec85fd
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/rtt_filter.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class VCMRttFilter
+{
+public:
+ VCMRttFilter(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
+
+ VCMRttFilter& operator=(const VCMRttFilter& rhs);
+
+ // Resets the filter.
+ void Reset();
+ // Updates the filter with a new sample.
+ void Update(WebRtc_UWord32 rttMs);
+ // A getter function for the current RTT level in ms.
+ WebRtc_UWord32 RttMs() const;
+
+private:
+ // The size of the drift and jump memory buffers
+ // and thus also the detection threshold for these
+ // detectors in number of samples.
+ enum { kMaxDriftJumpCount = 5 };
+ // Detects RTT jumps by comparing the difference between
+ // samples and average to the standard deviation.
+ // Returns true if the long time statistics should be updated
+ // and false otherwise
+ bool JumpDetection(WebRtc_UWord32 rttMs);
+ // Detects RTT drifts by comparing the difference between
+ // max and average to the standard deviation.
+ // Returns true if the long time statistics should be updated
+ // and false otherwise
+ bool DriftDetection(WebRtc_UWord32 rttMs);
+ // Computes the short time average and maximum of the vector buf.
+ void ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length);
+
+ WebRtc_Word32 _vcmId;
+ WebRtc_Word32 _receiverId;
+ bool _gotNonZeroUpdate;
+ double _avgRtt;
+ double _varRtt;
+ WebRtc_UWord32 _maxRtt;
+ WebRtc_UWord32 _filtFactCount;
+ const WebRtc_UWord32 _filtFactMax;
+ const double _jumpStdDevs;
+ const double _driftStdDevs;
+ WebRtc_Word32 _jumpCount;
+ WebRtc_Word32 _driftCount;
+ const WebRtc_Word32 _detectThreshold;
+ WebRtc_UWord32 _jumpBuf[kMaxDriftJumpCount];
+ WebRtc_UWord32 _driftBuf[kMaxDriftJumpCount];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/trunk/src/modules/video_coding/main/source/session_info.cc b/trunk/src/modules/video_coding/main/source/session_info.cc
new file mode 100644
index 0000000..7261403
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/session_info.cc
@@ -0,0 +1,595 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/main/source/session_info.h"
+
+#include "modules/video_coding/main/source/packet.h"
+
+namespace webrtc {
+
+VCMSessionInfo::VCMSessionInfo()
+ : session_nack_(false),
+ complete_(false),
+ decodable_(false),
+ frame_type_(kVideoFrameDelta),
+ previous_frame_loss_(false),
+ packets_(),
+ empty_seq_num_low_(-1),
+ empty_seq_num_high_(-1),
+ packets_not_decodable_(0) {
+}
+
+void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
+ const uint8_t* new_base_ptr) {
+ for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it)
+ if ((*it).dataPtr != NULL) {
+ assert(old_base_ptr != NULL && new_base_ptr != NULL);
+ (*it).dataPtr = new_base_ptr + ((*it).dataPtr - old_base_ptr);
+ }
+}
+
+int VCMSessionInfo::LowSequenceNumber() const {
+ if (packets_.empty())
+ return empty_seq_num_low_;
+ return packets_.front().seqNum;
+}
+
+int VCMSessionInfo::HighSequenceNumber() const {
+ if (packets_.empty())
+ return empty_seq_num_high_;
+ return LatestSequenceNumber(packets_.back().seqNum, empty_seq_num_high_,
+ NULL);
+}
+
+int VCMSessionInfo::PictureId() const {
+ if (packets_.empty() ||
+ packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
+ return kNoPictureId;
+ return packets_.front().codecSpecificHeader.codecHeader.VP8.pictureId;
+}
+
+int VCMSessionInfo::TemporalId() const {
+ if (packets_.empty() ||
+ packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
+ return kNoTemporalIdx;
+ return packets_.front().codecSpecificHeader.codecHeader.VP8.temporalIdx;
+}
+
+bool VCMSessionInfo::LayerSync() const {
+ if (packets_.empty() ||
+ packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
+ return false;
+ return packets_.front().codecSpecificHeader.codecHeader.VP8.layerSync;
+}
+
+int VCMSessionInfo::Tl0PicId() const {
+ if (packets_.empty() ||
+ packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
+ return kNoTl0PicIdx;
+ return packets_.front().codecSpecificHeader.codecHeader.VP8.tl0PicIdx;
+}
+
+bool VCMSessionInfo::NonReference() const {
+ if (packets_.empty() ||
+ packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
+ return false;
+ return packets_.front().codecSpecificHeader.codecHeader.VP8.nonReference;
+}
+
+void VCMSessionInfo::Reset() {
+ session_nack_ = false;
+ complete_ = false;
+ decodable_ = false;
+ frame_type_ = kVideoFrameDelta;
+ previous_frame_loss_ = false;
+ packets_.clear();
+ empty_seq_num_low_ = -1;
+ empty_seq_num_high_ = -1;
+ packets_not_decodable_ = 0;
+}
+
+int VCMSessionInfo::SessionLength() const {
+ int length = 0;
+ for (PacketIteratorConst it = packets_.begin(); it != packets_.end(); ++it)
+ length += (*it).sizeBytes;
+ return length;
+}
+
+int VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
+ PacketIterator packet_it) {
+ VCMPacket& packet = *packet_it;
+ PacketIterator it;
+
+ int packet_size = packet.sizeBytes;
+ packet_size += (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+
+ // Calculate the offset into the frame buffer for this packet.
+ int offset = 0;
+ for (it = packets_.begin(); it != packet_it; ++it)
+ offset += (*it).sizeBytes;
+
+ // Set the data pointer to pointing to the start of this packet in the
+ // frame buffer.
+ const uint8_t* data = packet.dataPtr;
+ packet.dataPtr = frame_buffer + offset;
+ packet.sizeBytes = packet_size;
+
+ ShiftSubsequentPackets(packet_it, packet_size);
+
+ const unsigned char startCode[] = {0, 0, 0, 1};
+ if (packet.insertStartCode) {
+ memcpy(const_cast<uint8_t*>(packet.dataPtr), startCode,
+ kH264StartCodeLengthBytes);
+ }
+ memcpy(const_cast<uint8_t*>(packet.dataPtr
+ + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0)),
+ data,
+ packet.sizeBytes);
+
+ return packet_size;
+}
+
+void VCMSessionInfo::ShiftSubsequentPackets(PacketIterator it,
+ int steps_to_shift) {
+ ++it;
+ if (it == packets_.end())
+ return;
+ uint8_t* first_packet_ptr = const_cast<WebRtc_UWord8*>((*it).dataPtr);
+ int shift_length = 0;
+ // Calculate the total move length and move the data pointers in advance.
+ for (; it != packets_.end(); ++it) {
+ shift_length += (*it).sizeBytes;
+ if ((*it).dataPtr != NULL)
+ (*it).dataPtr += steps_to_shift;
+ }
+ memmove(first_packet_ptr + steps_to_shift, first_packet_ptr, shift_length);
+}
+
+void VCMSessionInfo::UpdateCompleteSession() {
+ if (packets_.front().isFirstPacket && packets_.back().markerBit) {
+ // Do we have all the packets in this session?
+ bool complete_session = true;
+ PacketIterator it = packets_.begin();
+ PacketIterator prev_it = it;
+ ++it;
+ for (; it != packets_.end(); ++it) {
+ if (!InSequence(it, prev_it)) {
+ complete_session = false;
+ break;
+ }
+ prev_it = it;
+ }
+ complete_ = complete_session;
+ }
+}
+
+void VCMSessionInfo::UpdateDecodableSession(int rttMs) {
+ // Irrelevant if session is already complete or decodable
+ if (complete_ || decodable_)
+ return;
+ // First iteration - do nothing
+}
+
+bool VCMSessionInfo::complete() const {
+ return complete_;
+}
+
+bool VCMSessionInfo::decodable() const {
+ return decodable_;
+}
+
+// Find the end of the NAL unit which the packet pointed to by |packet_it|
+// belongs to. Returns an iterator to the last packet of the frame if the end
+// of the NAL unit wasn't found.
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindNaluEnd(
+ PacketIterator packet_it) const {
+ if ((*packet_it).completeNALU == kNaluEnd ||
+ (*packet_it).completeNALU == kNaluComplete) {
+ return packet_it;
+ }
+ // Find the end of the NAL unit.
+ for (; packet_it != packets_.end(); ++packet_it) {
+ if (((*packet_it).completeNALU == kNaluComplete &&
+ (*packet_it).sizeBytes > 0) ||
+ // Found next NALU.
+ (*packet_it).completeNALU == kNaluStart)
+ return --packet_it;
+ if ((*packet_it).completeNALU == kNaluEnd)
+ return packet_it;
+ }
+ // The end wasn't found.
+ return --packet_it;
+}
+
+int VCMSessionInfo::DeletePacketData(PacketIterator start,
+ PacketIterator end) {
+ int bytes_to_delete = 0; // The number of bytes to delete.
+ PacketIterator packet_after_end = end;
+ ++packet_after_end;
+
+ // Get the number of bytes to delete.
+ // Clear the size of these packets.
+ for (PacketIterator it = start; it != packet_after_end; ++it) {
+ bytes_to_delete += (*it).sizeBytes;
+ (*it).sizeBytes = 0;
+ (*it).dataPtr = NULL;
+ ++packets_not_decodable_;
+ }
+ if (bytes_to_delete > 0)
+ ShiftSubsequentPackets(end, -bytes_to_delete);
+ return bytes_to_delete;
+}
+
+int VCMSessionInfo::BuildVP8FragmentationHeader(
+ uint8_t* frame_buffer,
+ int frame_buffer_length,
+ RTPFragmentationHeader* fragmentation) {
+ int new_length = 0;
+ // Allocate space for max number of partitions
+ fragmentation->VerifyAndAllocateFragmentationHeader(kMaxVP8Partitions);
+ fragmentation->fragmentationVectorSize = 0;
+ memset(fragmentation->fragmentationLength, 0,
+ kMaxVP8Partitions * sizeof(WebRtc_UWord32));
+ if (packets_.empty())
+ return new_length;
+ PacketIterator it = FindNextPartitionBeginning(packets_.begin(),
+ &packets_not_decodable_);
+ while (it != packets_.end()) {
+ const int partition_id =
+ (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
+ PacketIterator partition_end = FindPartitionEnd(it);
+ fragmentation->fragmentationOffset[partition_id] =
+ (*it).dataPtr - frame_buffer;
+ assert(fragmentation->fragmentationOffset[partition_id] <
+ static_cast<WebRtc_UWord32>(frame_buffer_length));
+ fragmentation->fragmentationLength[partition_id] =
+ (*partition_end).dataPtr + (*partition_end).sizeBytes - (*it).dataPtr;
+ assert(fragmentation->fragmentationLength[partition_id] <=
+ static_cast<WebRtc_UWord32>(frame_buffer_length));
+ new_length += fragmentation->fragmentationLength[partition_id];
+ ++partition_end;
+ it = FindNextPartitionBeginning(partition_end, &packets_not_decodable_);
+ if (partition_id + 1 > fragmentation->fragmentationVectorSize)
+ fragmentation->fragmentationVectorSize = partition_id + 1;
+ }
+ // Set all empty fragments to start where the previous fragment ends,
+ // and have zero length.
+ if (fragmentation->fragmentationLength[0] == 0)
+ fragmentation->fragmentationOffset[0] = 0;
+ for (int i = 1; i < fragmentation->fragmentationVectorSize; ++i) {
+ if (fragmentation->fragmentationLength[i] == 0)
+ fragmentation->fragmentationOffset[i] =
+ fragmentation->fragmentationOffset[i - 1] +
+ fragmentation->fragmentationLength[i - 1];
+ assert(i == 0 ||
+ fragmentation->fragmentationOffset[i] >=
+ fragmentation->fragmentationOffset[i - 1]);
+ }
+ assert(new_length <= frame_buffer_length);
+ return new_length;
+}
+
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindNextPartitionBeginning(
+ PacketIterator it, int* packets_skipped) const {
+ while (it != packets_.end()) {
+ if ((*it).codecSpecificHeader.codecHeader.VP8.beginningOfPartition) {
+ return it;
+ } else if (packets_skipped != NULL) {
+ // This packet belongs to a partition with a previous loss and can't
+ // be decoded.
+ ++(*packets_skipped);
+ }
+ ++it;
+ }
+ return it;
+}
+
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindPartitionEnd(
+ PacketIterator it) const {
+ assert((*it).codec == kVideoCodecVP8);
+ PacketIterator prev_it = it;
+ const int partition_id =
+ (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
+ while (it != packets_.end()) {
+ bool beginning =
+ (*it).codecSpecificHeader.codecHeader.VP8.beginningOfPartition;
+ int current_partition_id =
+ (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
+ bool packet_loss_found = (!beginning && !InSequence(it, prev_it));
+ if (packet_loss_found ||
+ (beginning && current_partition_id != partition_id)) {
+ // Missing packet, the previous packet was the last in sequence.
+ return prev_it;
+ }
+ prev_it = it;
+ ++it;
+ }
+ return prev_it;
+}
+
+bool VCMSessionInfo::InSequence(const PacketIterator& packet_it,
+ const PacketIterator& prev_packet_it) {
+ // If the two iterators are pointing to the same packet they are considered
+ // to be in sequence.
+ return (packet_it == prev_packet_it ||
+ (static_cast<WebRtc_UWord16>((*prev_packet_it).seqNum + 1) ==
+ (*packet_it).seqNum));
+}
+
+int VCMSessionInfo::MakeDecodable() {
+ int return_length = 0;
+ if (packets_.empty()) {
+ return 0;
+ }
+ PacketIterator it = packets_.begin();
+ // Make sure we remove the first NAL unit if it's not decodable.
+ if ((*it).completeNALU == kNaluIncomplete ||
+ (*it).completeNALU == kNaluEnd) {
+ PacketIterator nalu_end = FindNaluEnd(it);
+ return_length += DeletePacketData(it, nalu_end);
+ it = nalu_end;
+ }
+ PacketIterator prev_it = it;
+ // Take care of the rest of the NAL units.
+ for (; it != packets_.end(); ++it) {
+ bool start_of_nalu = ((*it).completeNALU == kNaluStart ||
+ (*it).completeNALU == kNaluComplete);
+ if (!start_of_nalu && !InSequence(it, prev_it)) {
+ // Found a sequence number gap due to packet loss.
+ PacketIterator nalu_end = FindNaluEnd(it);
+ return_length += DeletePacketData(it, nalu_end);
+ it = nalu_end;
+ }
+ prev_it = it;
+ }
+ return return_length;
+}
+
+int VCMSessionInfo::BuildHardNackList(int* seq_num_list,
+ int seq_num_list_length) {
+ if (NULL == seq_num_list || seq_num_list_length < 1) {
+ return -1;
+ }
+ if (packets_.empty()) {
+ return 0;
+ }
+
+ // Find end point (index of entry equals the sequence number of the first
+ // packet).
+ int index = 0;
+ for (; index < seq_num_list_length; ++index) {
+ if (seq_num_list[index] == packets_.front().seqNum) {
+ seq_num_list[index] = -1;
+ ++index;
+ break;
+ }
+ }
+
+ // Zero out between the first entry and the end point.
+ PacketIterator it = packets_.begin();
+ PacketIterator prev_it = it;
+ ++it;
+ while (it != packets_.end() && index < seq_num_list_length) {
+ if (!InSequence(it, prev_it)) {
+ // Found a sequence number gap due to packet loss.
+ index += PacketsMissing(it, prev_it);
+ session_nack_ = true;
+ }
+ seq_num_list[index] = -1;
+ ++index;
+ prev_it = it;
+ ++it;
+ }
+ if (!packets_.front().isFirstPacket)
+ session_nack_ = true;
+ return 0;
+}
+
+int VCMSessionInfo::BuildSoftNackList(int* seq_num_list,
+ int seq_num_list_length,
+ int rtt_ms) {
+ if (NULL == seq_num_list || seq_num_list_length < 1) {
+ return -1;
+ }
+ if (packets_.empty() && empty_seq_num_low_ == -1) {
+ return 0;
+ }
+
+ int index = 0;
+ int low_seq_num = (packets_.empty()) ? empty_seq_num_low_:
+ packets_.front().seqNum;
+ // Find entrance point (index of entry equals the sequence number of the
+ // first packet).
+ for (; index < seq_num_list_length; ++index) {
+ if (seq_num_list[index] == low_seq_num) {
+ seq_num_list[index] = -1;
+ break;
+ }
+ }
+
+ // TODO(mikhal): 1. Update score based on RTT value 2. Add partition data.
+ // Use the previous available.
+ bool base_available = false;
+ if ((index > 0) && (seq_num_list[index] == -1)) {
+ // Found first packet, for now let's go only one back.
+ if ((seq_num_list[index - 1] == -1) || (seq_num_list[index - 1] == -2)) {
+ // This is indeed the first packet, as previous packet was populated.
+ base_available = true;
+ }
+ }
+ bool allow_nack = ((packets_.size() > 0 && !packets_.front().isFirstPacket)
+ || !base_available);
+
+ // Zero out between first entry and end point.
+
+ int media_high_seq_num;
+ if (HaveLastPacket()) {
+ media_high_seq_num = packets_.back().seqNum;
+ } else {
+ // Estimation.
+ if (empty_seq_num_low_ >= 0) {
+ // Assuming empty packets have later sequence numbers than media packets.
+ media_high_seq_num = empty_seq_num_low_ - 1;
+ } else {
+ // Since this frame doesn't have the marker bit we can assume it should
+ // contain at least one more packet.
+ media_high_seq_num = static_cast<uint16_t>(packets_.back().seqNum + 1);
+ }
+ }
+
+ // Compute session/packet scores and thresholds:
+ // based on RTT and layer info (when available).
+ float nack_score_threshold = 0.25f;
+ float layer_score = TemporalId() > 0 ? 0.0f : 1.0f;
+ float rtt_score = 1.0f;
+ float score_multiplier = rtt_score * layer_score;
+ // Zero out between first entry and end point.
+ if (!packets_.empty()) {
+ PacketIterator it = packets_.begin();
+ PacketIterator prev_it = it;
+ ++index;
+ ++it;
+ // TODO(holmer): Rewrite this in a way which better makes use of the list.
+ while (it != packets_.end() && index < seq_num_list_length) {
+ // Only process media packet sequence numbers.
+ if (LatestSequenceNumber((*it).seqNum, media_high_seq_num, NULL) ==
+ (*it).seqNum && (*it).seqNum != media_high_seq_num)
+ break;
+ if (!InSequence(it, prev_it)) {
+ // Found a sequence number gap due to packet loss.
+ int num_lost = PacketsMissing(it, prev_it);
+ for (int i = 0 ; i < num_lost; ++i) {
+ // Compute score of the packet.
+ float score = 1.0f;
+ // Multiply internal score (packet) by score multiplier.
+ score *= score_multiplier;
+ if (score > nack_score_threshold) {
+ allow_nack = true;
+ } else {
+ seq_num_list[index] = -1;
+ }
+ ++index;
+ }
+ }
+ seq_num_list[index] = -1;
+ ++index;
+ prev_it = it;
+ ++it;
+ }
+ }
+
+ // Empty packets follow the data packets, and therefore have a higher
+ // sequence number. We do not want to NACK empty packets.
+ if ((empty_seq_num_low_ != -1) && (empty_seq_num_high_ != -1) &&
+ (index < seq_num_list_length)) {
+ // First make sure that we are at least at the minimum value (if not we are
+ // missing last packet(s)).
+ while (seq_num_list[index] < empty_seq_num_low_ &&
+ index < seq_num_list_length) {
+ ++index;
+ }
+
+ // Mark empty packets.
+ while (seq_num_list[index] <= empty_seq_num_high_ &&
+ index < seq_num_list_length) {
+ seq_num_list[index] = -2;
+ ++index;
+ }
+ }
+
+ session_nack_ = allow_nack;
+ return 0;
+}
+
+int VCMSessionInfo::PacketsMissing(const PacketIterator& packet_it,
+ const PacketIterator& prev_packet_it) {
+ if (packet_it == prev_packet_it)
+ return 0;
+ if ((*prev_packet_it).seqNum > (*packet_it).seqNum) // Wrap.
+ return static_cast<WebRtc_UWord16>(
+ static_cast<WebRtc_UWord32>((*packet_it).seqNum + 0x10000) -
+ (*prev_packet_it).seqNum) - 1;
+ else
+ return (*packet_it).seqNum - (*prev_packet_it).seqNum - 1;
+}
+
+bool
+VCMSessionInfo::HaveLastPacket() const {
+ return (!packets_.empty() && packets_.back().markerBit);
+}
+
+bool
+VCMSessionInfo::session_nack() const {
+ return session_nack_;
+}
+
+int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
+ uint8_t* frame_buffer,
+ bool enable_decodable_state,
+ int rtt_ms) {
+ // Check if this is first packet (only valid for some codecs)
+ if (packet.isFirstPacket) {
+ // The first packet in a frame signals the frame type.
+ frame_type_ = packet.frameType;
+ } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
+ // Update the frame type with the first media packet.
+ frame_type_ = packet.frameType;
+ }
+ if (packet.frameType == kFrameEmpty) {
+ // Update sequence number of an empty packet.
+ // Only media packets are inserted into the packet list.
+ InformOfEmptyPacket(packet.seqNum);
+ return 0;
+ }
+
+ if (packets_.size() == kMaxPacketsInSession)
+ return -1;
+
+ // Find the position of this packet in the packet list in sequence number
+ // order and insert it. Loop over the list in reverse order.
+ ReversePacketIterator rit = packets_.rbegin();
+ for (; rit != packets_.rend(); ++rit)
+ if (LatestSequenceNumber((*rit).seqNum, packet.seqNum, NULL) ==
+ packet.seqNum)
+ break;
+
+ // Check for duplicate packets.
+ if (rit != packets_.rend() &&
+ (*rit).seqNum == packet.seqNum && (*rit).sizeBytes > 0)
+ return -2;
+
+ // The insert operation invalidates the iterator |rit|.
+ PacketIterator packet_list_it = packets_.insert(rit.base(), packet);
+
+ int returnLength = InsertBuffer(frame_buffer, packet_list_it);
+ UpdateCompleteSession();
+ if (enable_decodable_state)
+ UpdateDecodableSession(rtt_ms);
+ return returnLength;
+}
+
+void VCMSessionInfo::InformOfEmptyPacket(uint16_t seq_num) {
+ // Empty packets may be FEC or filler packets. They are sequential and
+ // follow the data packets, therefore, we should only keep track of the high
+ // and low sequence numbers and may assume that the packets in between are
+ // empty packets belonging to the same frame (timestamp).
+ empty_seq_num_high_ = LatestSequenceNumber(seq_num, empty_seq_num_high_,
+ NULL);
+ if (empty_seq_num_low_ == -1 ||
+ LatestSequenceNumber(seq_num, empty_seq_num_low_, NULL) ==
+ empty_seq_num_low_)
+ empty_seq_num_low_ = seq_num;
+}
+
+int VCMSessionInfo::packets_not_decodable() const {
+ return packets_not_decodable_;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/session_info.h b/trunk/src/modules/video_coding/main/source/session_info.h
new file mode 100644
index 0000000..27533ce
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/session_info.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+
+#include <list>
+
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/source/packet.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+class VCMSessionInfo {
+ public:
+ VCMSessionInfo();
+
+ void UpdateDataPointers(const uint8_t* old_base_ptr,
+ const uint8_t* new_base_ptr);
+ // NACK - Building the NACK lists.
+ // Build hard NACK list: Zero out all entries in list up to and including
+ // _lowSeqNum.
+ int BuildHardNackList(int* seq_num_list,
+ int seq_num_list_length);
+
+ // Build soft NACK list: Zero out only a subset of the packets, discard
+ // empty packets.
+ int BuildSoftNackList(int* seq_num_list,
+ int seq_num_list_length,
+ int rtt_ms);
+ void Reset();
+ int InsertPacket(const VCMPacket& packet,
+ uint8_t* frame_buffer,
+ bool enable_decodable_state,
+ int rtt_ms);
+ bool complete() const;
+ bool decodable() const;
+
+ // Builds fragmentation headers for VP8, each fragment being a decodable
+ // VP8 partition. Returns the total number of bytes which are decodable. Is
+ // used instead of MakeDecodable for VP8.
+ int BuildVP8FragmentationHeader(uint8_t* frame_buffer,
+ int frame_buffer_length,
+ RTPFragmentationHeader* fragmentation);
+
+ // Makes the frame decodable. I.e., only contain decodable NALUs. All
+ // non-decodable NALUs will be deleted and packets will be moved to in
+ // memory to remove any empty space.
+ // Returns the number of bytes deleted from the session.
+ int MakeDecodable();
+ int SessionLength() const;
+ bool HaveLastPacket() const;
+ bool session_nack() const;
+ webrtc::FrameType FrameType() const { return frame_type_; }
+ int LowSequenceNumber() const;
+
+ // Returns highest sequence number, media or empty.
+ int HighSequenceNumber() const;
+ int PictureId() const;
+ int TemporalId() const;
+ bool LayerSync() const;
+ int Tl0PicId() const;
+ bool NonReference() const;
+ void SetPreviousFrameLoss() { previous_frame_loss_ = true; }
+ bool PreviousFrameLoss() const { return previous_frame_loss_; }
+
+ // The number of packets discarded because the decoder can't make use of
+ // them.
+ int packets_not_decodable() const;
+
+ private:
+ enum { kMaxVP8Partitions = 9 };
+
+ typedef std::list<VCMPacket> PacketList;
+ typedef PacketList::iterator PacketIterator;
+ typedef PacketList::const_iterator PacketIteratorConst;
+ typedef PacketList::reverse_iterator ReversePacketIterator;
+
+ void InformOfEmptyPacket(uint16_t seq_num);
+
+ // Finds the packet of the beginning of the next VP8 partition. If
+ // none is found the returned iterator points to |packets_.end()|.
+ // |it| is expected to point to the last packet of the previous partition,
+ // or to the first packet of the frame. |packets_skipped| is incremented
+ // for each packet found which doesn't have the beginning bit set.
+ PacketIterator FindNextPartitionBeginning(PacketIterator it,
+ int* packets_skipped) const;
+
+ // Returns an iterator pointing to the last packet of the partition pointed to
+ // by |it|.
+ PacketIterator FindPartitionEnd(PacketIterator it) const;
+ static bool InSequence(const PacketIterator& it,
+ const PacketIterator& prev_it);
+ static int PacketsMissing(const PacketIterator& packet_it,
+ const PacketIterator& prev_packet_it);
+ int InsertBuffer(uint8_t* frame_buffer,
+ PacketIterator packetIterator);
+ void ShiftSubsequentPackets(PacketIterator it, int steps_to_shift);
+ PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
+ // Deletes the data of all packets between |start| and |end|, inclusively.
+ // Note that this function doesn't delete the actual packets.
+ int DeletePacketData(PacketIterator start,
+ PacketIterator end);
+ void UpdateCompleteSession();
+
+ // When enabled, determine if session is decodable, i.e. incomplete but
+ // would be sent to the decoder.
+ void UpdateDecodableSession(int rtt_ms);
+
+ // If this session has been NACKed by the jitter buffer.
+ bool session_nack_;
+ bool complete_;
+ bool decodable_;
+ webrtc::FrameType frame_type_;
+ bool previous_frame_loss_;
+ // Packets in this frame.
+ PacketList packets_;
+ int empty_seq_num_low_;
+ int empty_seq_num_high_;
+ // Number of packets discarded because the decoder can't use them.
+ int packets_not_decodable_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
diff --git a/trunk/src/modules/video_coding/main/source/session_info_unittest.cc b/trunk/src/modules/video_coding/main/source/session_info_unittest.cc
new file mode 100644
index 0000000..e017735
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/session_info_unittest.cc
@@ -0,0 +1,931 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "gtest/gtest.h"
+#include "modules/interface/module_common_types.h"
+#include "modules/video_coding/main/source/packet.h"
+#include "modules/video_coding/main/source/session_info.h"
+
+namespace webrtc {
+
+class TestSessionInfo : public ::testing::Test {
+ protected:
+ enum { kPacketBufferSize = 10 };
+ enum { kFrameBufferSize = 10 * kPacketBufferSize };
+
+ virtual void SetUp() {
+ memset(packet_buffer_, 0, kPacketBufferSize);
+ memset(frame_buffer_, 0, kFrameBufferSize);
+ session_.Reset();
+ packet_.Reset();
+ packet_.frameType = kVideoFrameDelta;
+ packet_.sizeBytes = kPacketBufferSize;
+ packet_.dataPtr = packet_buffer_;
+ packet_.seqNum = 0;
+ packet_.timestamp = 0;
+ }
+
+ void FillPacket(uint8_t start_value) {
+ for (int i = 0; i < kPacketBufferSize; ++i)
+ packet_buffer_[i] = start_value + i;
+ }
+
+ void VerifyPacket(uint8_t* start_ptr, uint8_t start_value) {
+ for (int j = 0; j < kPacketBufferSize; ++j) {
+ ASSERT_EQ(start_value + j, start_ptr[j]);
+ }
+ }
+
+ uint8_t packet_buffer_[kPacketBufferSize];
+ uint8_t frame_buffer_[kFrameBufferSize];
+ VCMSessionInfo session_;
+ VCMPacket packet_;
+};
+
+class TestVP8Partitions : public TestSessionInfo {
+ protected:
+ enum { kMaxVP8Partitions = 9 };
+
+ virtual void SetUp() {
+ TestSessionInfo::SetUp();
+ vp8_header_ = &packet_header_.type.Video.codecHeader.VP8;
+ packet_header_.frameType = kVideoFrameDelta;
+ packet_header_.type.Video.codec = kRTPVideoVP8;
+ vp8_header_->InitRTPVideoHeaderVP8();
+ fragmentation_.VerifyAndAllocateFragmentationHeader(kMaxVP8Partitions);
+ }
+
+ bool VerifyPartition(int partition_id,
+ int packets_expected,
+ int start_value) {
+ EXPECT_EQ(static_cast<uint32_t>(packets_expected * kPacketBufferSize),
+ fragmentation_.fragmentationLength[partition_id]);
+ for (int i = 0; i < packets_expected; ++i) {
+ int packet_index = fragmentation_.fragmentationOffset[partition_id] +
+ i * kPacketBufferSize;
+ if (packet_index + kPacketBufferSize > kFrameBufferSize)
+ return false;
+ VerifyPacket(frame_buffer_ + packet_index, start_value + i);
+ }
+ return true;
+ }
+
+ WebRtcRTPHeader packet_header_;
+ RTPVideoHeaderVP8* vp8_header_;
+ RTPFragmentationHeader fragmentation_;
+};
+
+class TestNalUnits : public TestSessionInfo {
+ protected:
+ virtual void SetUp() {
+ TestSessionInfo::SetUp();
+ packet_.codec = kVideoCodecVP8;
+ }
+
+ bool VerifyNalu(int offset, int packets_expected, int start_value) {
+ EXPECT_GE(session_.SessionLength(),
+ packets_expected * kPacketBufferSize);
+ for (int i = 0; i < packets_expected; ++i) {
+ int packet_index = offset * kPacketBufferSize + i * kPacketBufferSize;
+ VerifyPacket(frame_buffer_ + packet_index, start_value + i);
+ }
+ return true;
+ }
+};
+
+class TestNackList : public TestSessionInfo {
+ protected:
+ enum { kMaxSeqNumListLength = 30 };
+
+ virtual void SetUp() {
+ TestSessionInfo::SetUp();
+ seq_num_list_length_ = 0;
+ memset(seq_num_list_, 0, sizeof(seq_num_list_));
+ }
+
+ void BuildSeqNumList(uint16_t low,
+ uint16_t high) {
+ int i = 0;
+ while (low != high + 1) {
+ EXPECT_LT(i, kMaxSeqNumListLength);
+ if (i >= kMaxSeqNumListLength) {
+ seq_num_list_length_ = kMaxSeqNumListLength;
+ return;
+ }
+ seq_num_list_[i] = low;
+ low++;
+ i++;
+ }
+ seq_num_list_length_ = i;
+ }
+
+ void VerifyAll(int value) {
+ for (int i = 0; i < seq_num_list_length_; ++i)
+ EXPECT_EQ(seq_num_list_[i], value);
+ }
+
+ int seq_num_list_[kMaxSeqNumListLength];
+ int seq_num_list_length_;
+};
+
+TEST_F(TestSessionInfo, TestSimpleAPIs) {
+ packet_.isFirstPacket = true;
+ packet_.seqNum = 0xFFFE;
+ packet_.sizeBytes = kPacketBufferSize;
+ packet_.frameType = kVideoFrameKey;
+ FillPacket(0);
+ ASSERT_EQ(kPacketBufferSize,
+ session_.InsertPacket(packet_, frame_buffer_, false, 0));
+ EXPECT_FALSE(session_.HaveLastPacket());
+ EXPECT_EQ(kVideoFrameKey, session_.FrameType());
+
+ packet_.isFirstPacket = false;
+ packet_.markerBit = true;
+ packet_.seqNum += 1;
+ ASSERT_EQ(kPacketBufferSize,
+ session_.InsertPacket(packet_, frame_buffer_, false, 0));
+ EXPECT_TRUE(session_.HaveLastPacket());
+ EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
+ EXPECT_EQ(0xFFFE, session_.LowSequenceNumber());
+
+ // Insert empty packet which will be the new high sequence number.
+ // To make things more difficult we will make sure to have a wrap here.
+ packet_.isFirstPacket = false;
+ packet_.markerBit = true;
+ packet_.seqNum = 2;
+ packet_.sizeBytes = 0;
+ packet_.frameType = kFrameEmpty;
+ ASSERT_EQ(0,
+ session_.InsertPacket(packet_, frame_buffer_, false, 0));
+ EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
+}
+
+TEST_F(TestSessionInfo, NormalOperation) {
+ packet_.seqNum = 0xFFFF;
+ packet_.isFirstPacket = true;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ for (int i = 1; i < 9; ++i) {
+ packet_.seqNum += 1;
+ FillPacket(i);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ }
+
+ packet_.seqNum += 1;
+ packet_.markerBit = true;
+ FillPacket(9);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(0, session_.packets_not_decodable());
+ EXPECT_EQ(10 * kPacketBufferSize, session_.SessionLength());
+ for (int i = 0; i < 10; ++i) {
+ SCOPED_TRACE("Calling VerifyPacket");
+ VerifyPacket(frame_buffer_ + i * kPacketBufferSize, i);
+ }
+}
+
+TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
+ // Partition 0 | Partition 1
+ // [ 0 ] [ 2 ] | [ 3 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 0;
+ FillPacket(0);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 2;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(3);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // One packet should be removed (end of partition 0).
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 2*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 1, 0));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(1, 1, 3));
+}
+
+TEST_F(TestVP8Partitions, TwoPartitionsOneLoss2) {
+ // Partition 0 | Partition 1
+ // [ 1 ] [ 2 ] | [ 3 ] [ 5 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 1;
+ FillPacket(1);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0)
+ , kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(3);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 2;
+ FillPacket(5);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // One packet should be removed (end of partition 2), 3 left.
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 3*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 1));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(1, 1, 3));
+ EXPECT_EQ(1, session_.packets_not_decodable());
+}
+
+TEST_F(TestVP8Partitions, TwoPartitionsNoLossWrap) {
+ // Partition 0 | Partition 1
+ // [ fffd ] [ fffe ] | [ ffff ] [ 0 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 0xfffd;
+ FillPacket(0);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(1);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(3);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // No packet should be removed.
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 4*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 0));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(1, 2, 2));
+ EXPECT_EQ(0, session_.packets_not_decodable());
+}
+
+TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
+ // Partition 0 | Partition 1
+ // [ fffd ] [ fffe ] | [ ffff ] [ 1 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 0xfffd;
+ FillPacket(0);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(1);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 2;
+ FillPacket(3);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // One packet should be removed from the last partition
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 3*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 0));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(1, 1, 2));
+ EXPECT_EQ(1, session_.packets_not_decodable());
+}
+
+
+TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
+ // Partition 1 |Partition 2 | Partition 3
+ // [ 1 ] [ 2 ] | | [ 5 ] | [ 6 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 1;
+ FillPacket(1);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 2;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 3;
+ FillPacket(5);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 2;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(6);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // No packet should be removed.
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 4*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 1));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(2, 2, 5));
+ EXPECT_EQ(0, session_.packets_not_decodable());
+}
+
+TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
+ // Partition 0 |Partition 1 | Partition 2
+ // [ 1 ] [ 2 ] | [ 4 ] [ 5 ] | [ 6 ] [ 7 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 1;
+ FillPacket(1);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 0;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 2;
+ FillPacket(4);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(5);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 2;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(6);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 2;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(7);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // 2 partitions left. 2 packets removed from second partition
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 4*kPacketBufferSize);
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 1));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(2, 2, 6));
+ EXPECT_EQ(2, session_.packets_not_decodable());
+}
+
+TEST_F(TestVP8Partitions, AggregationOverTwoPackets) {
+ // Partition 0 | Partition 1 | Partition 2
+ // [ 0 | ] [ 1 ] | [ 2 ]
+ packet_header_.type.Video.isFirstPacket = true;
+ vp8_header_->beginningOfPartition = true;
+ vp8_header_->partitionId = 0;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber = 0;
+ FillPacket(0);
+ VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
+ packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 1;
+ vp8_header_->beginningOfPartition = false;
+ packet_header_.header.markerBit = false;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(1);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ packet_header_.type.Video.isFirstPacket = false;
+ vp8_header_->partitionId = 2;
+ vp8_header_->beginningOfPartition = true;
+ packet_header_.header.markerBit = true;
+ packet_header_.header.sequenceNumber += 1;
+ FillPacket(2);
+ packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
+ ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ delete packet;
+
+ // No packets removed.
+ EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
+ kFrameBufferSize,
+ &fragmentation_),
+ 3*kPacketBufferSize);
+ EXPECT_EQ(0, session_.packets_not_decodable());
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(0, 2, 0));
+ // This partition is aggregated in partition 0
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(1, 0, 0));
+ SCOPED_TRACE("Calling VerifyPartition");
+ EXPECT_TRUE(VerifyPartition(2, 1, 2));
+}
+
+TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluComplete;
+ packet_.frameType = kFrameEmpty;
+ packet_.sizeBytes = 0;
+ packet_.seqNum = 0;
+ packet_.markerBit = false;
+ ASSERT_EQ(0, session_.InsertPacket(packet_, frame_buffer_, false, 0));
+
+ EXPECT_EQ(0, session_.MakeDecodable());
+ EXPECT_EQ(0, session_.SessionLength());
+ EXPECT_EQ(0, session_.packets_not_decodable());
+}
+
+TEST_F(TestNalUnits, OneIsolatedNaluLoss) {
+ packet_.isFirstPacket = true;
+ packet_.completeNALU = kNaluComplete;
+ packet_.seqNum = 0;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluComplete;
+ packet_.seqNum += 2;
+ packet_.markerBit = true;
+ FillPacket(2);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(0, session_.MakeDecodable());
+ EXPECT_EQ(2 * kPacketBufferSize, session_.SessionLength());
+ EXPECT_EQ(0, session_.packets_not_decodable());
+ SCOPED_TRACE("Calling VerifyNalu");
+ EXPECT_TRUE(VerifyNalu(0, 1, 0));
+ SCOPED_TRACE("Calling VerifyNalu");
+ EXPECT_TRUE(VerifyNalu(1, 1, 2));
+}
+
+TEST_F(TestNalUnits, LossInMiddleOfNalu) {
+ packet_.isFirstPacket = true;
+ packet_.completeNALU = kNaluComplete;
+ packet_.seqNum = 0;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluEnd;
+ packet_.seqNum += 2;
+ packet_.markerBit = true;
+ FillPacket(2);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(kPacketBufferSize, session_.MakeDecodable());
+ EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
+ EXPECT_EQ(1, session_.packets_not_decodable());
+ SCOPED_TRACE("Calling VerifyNalu");
+ EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, StartAndEndOfLastNalUnitLost) {
+ packet_.isFirstPacket = true;
+ packet_.completeNALU = kNaluComplete;
+ packet_.seqNum = 0;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluIncomplete;
+ packet_.seqNum += 2;
+ packet_.markerBit = false;
+ FillPacket(1);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(kPacketBufferSize, session_.MakeDecodable());
+ EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
+ EXPECT_EQ(1, session_.packets_not_decodable());
+ SCOPED_TRACE("Calling VerifyNalu");
+ EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, ReorderWrapNoLoss) {
+ packet_.seqNum = 0xFFFF;
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluIncomplete;
+ packet_.seqNum += 1;
+ packet_.markerBit = false;
+ FillPacket(1);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = true;
+ packet_.completeNALU = kNaluComplete;
+ packet_.seqNum -= 1;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluEnd;
+ packet_.seqNum += 2;
+ packet_.markerBit = true;
+ FillPacket(2);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(0, session_.MakeDecodable());
+ EXPECT_EQ(0, session_.packets_not_decodable());
+ EXPECT_EQ(3*kPacketBufferSize, session_.SessionLength());
+ SCOPED_TRACE("Calling VerifyNalu");
+ EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, WrapLosses) {
+ packet_.seqNum = 0xFFFF;
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluIncomplete;
+ packet_.markerBit = false;
+ FillPacket(1);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluEnd;
+ packet_.seqNum += 2;
+ packet_.markerBit = true;
+ FillPacket(2);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(2 * kPacketBufferSize, session_.MakeDecodable());
+ EXPECT_EQ(0, session_.SessionLength());
+ EXPECT_EQ(2, session_.packets_not_decodable());
+}
+
+TEST_F(TestNalUnits, ReorderWrapLosses) {
+ packet_.seqNum = 0xFFFF;
+
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluEnd;
+ packet_.seqNum += 2;
+ packet_.markerBit = true;
+ FillPacket(2);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ packet_.seqNum -= 2;
+ packet_.isFirstPacket = false;
+ packet_.completeNALU = kNaluIncomplete;
+ packet_.markerBit = false;
+ FillPacket(1);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(2 * kPacketBufferSize, session_.MakeDecodable());
+ EXPECT_EQ(0, session_.SessionLength());
+ EXPECT_EQ(2, session_.packets_not_decodable());
+}
+
+TEST_F(TestNackList, NoLosses) {
+ uint16_t low = 0xFFFF - 5;
+
+ packet_.seqNum = low;
+ packet_.isFirstPacket = true;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ for (int i = 1; i < 9; ++i) {
+ packet_.seqNum += 1;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = false;
+ FillPacket(i + 1);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ }
+
+ packet_.seqNum += 1;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = true;
+ FillPacket(10);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(10 * kPacketBufferSize, session_.SessionLength());
+ BuildSeqNumList(low, packet_.seqNum);
+ EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
+ EXPECT_FALSE(session_.session_nack());
+ SCOPED_TRACE("Calling VerifyAll");
+ VerifyAll(-1);
+
+ BuildSeqNumList(low, packet_.seqNum);
+ EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
+ 60));
+ SCOPED_TRACE("Calling VerifyAll");
+ VerifyAll(-1);
+}
+
+TEST_F(TestNackList, FiveLossesSpreadOut) {
+ uint16_t low = 0xFFFF - 5;
+
+ packet_.seqNum = low;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = true;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ for (int i = 1; i < 9; ++i) {
+ packet_.seqNum += 1;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = false;
+ FillPacket(i);
+ if ((i + 1) % 2)
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+ }
+
+ packet_.seqNum++; // Simulate loss of last packet.
+
+ EXPECT_EQ(5 * kPacketBufferSize, session_.SessionLength());
+ BuildSeqNumList(low, packet_.seqNum);
+ EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
+ for (int i = 0; i < seq_num_list_length_; ++i) {
+ if (i % 2)
+ EXPECT_EQ(static_cast<uint16_t>(low + i), seq_num_list_[i]);
+ else
+ EXPECT_EQ(-1, seq_num_list_[i]);
+ }
+
+ BuildSeqNumList(low, packet_.seqNum);
+ EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
+ 60));
+ EXPECT_EQ(true, session_.session_nack());
+ for (int i = 0; i < seq_num_list_length_; ++i) {
+ if (i % 2)
+ EXPECT_EQ(static_cast<uint16_t>(low + i), seq_num_list_[i]);
+ else
+ EXPECT_EQ(-1, seq_num_list_[i]);
+ }
+}
+
+TEST_F(TestNackList, FirstAndLastLost) {
+ uint16_t low = 0xFFFF;
+
+ packet_.seqNum = low + 1;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = false;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
+ kPacketBufferSize);
+
+ EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
+ BuildSeqNumList(low, packet_.seqNum + 1);
+ EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
+ EXPECT_EQ(0xFFFF, seq_num_list_[0]);
+ EXPECT_EQ(-1, seq_num_list_[1]);
+ EXPECT_EQ(1, seq_num_list_[2]);
+
+ BuildSeqNumList(low, packet_.seqNum + 1);
+ EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_,seq_num_list_length_,
+ 60));
+ EXPECT_EQ(true, session_.session_nack());
+ EXPECT_EQ(0xFFFF, seq_num_list_[0]);
+ EXPECT_EQ(-1, seq_num_list_[1]);
+ EXPECT_EQ(1, seq_num_list_[2]);
+}
+
+TEST_F(TestNackList, LostAllButEmptyPackets) {
+ uint16_t low = 0;
+ packet_.seqNum = low + 1;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = false;
+ packet_.frameType = kFrameEmpty;
+ packet_.sizeBytes = 0;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0), 0);
+
+ packet_.seqNum = low + 3;
+ packet_.isFirstPacket = false;
+ packet_.markerBit = false;
+ packet_.frameType = kFrameEmpty;
+ packet_.sizeBytes = 0;
+ FillPacket(0);
+ ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0), 0);
+
+ EXPECT_EQ(0, session_.SessionLength());
+ BuildSeqNumList(low, packet_.seqNum + 1);
+ EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
+ 60));
+ EXPECT_EQ(true, session_.session_nack());
+ EXPECT_EQ(0, seq_num_list_[0]);
+ EXPECT_EQ(-1, seq_num_list_[1]);
+ EXPECT_EQ(-2, seq_num_list_[2]);
+ EXPECT_EQ(-2, seq_num_list_[3]);
+ EXPECT_EQ(4, seq_num_list_[4]);
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/tick_time_base.h b/trunk/src/modules/video_coding/main/source/tick_time_base.h
new file mode 100644
index 0000000..a212591
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/tick_time_base.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
+
+#include "system_wrappers/interface/tick_util.h"
+
+namespace webrtc {
+
+// This class provides a mockable wrapper to TickTime.
+class TickTimeBase {
+ public:
+ virtual ~TickTimeBase() {}
+
+ // "Now" in milliseconds.
+ virtual int64_t MillisecondTimestamp() const {
+ return TickTime::MillisecondTimestamp();
+ }
+
+ // "Now" in microseconds.
+ virtual int64_t MicrosecondTimestamp() const {
+ return TickTime::MicrosecondTimestamp();
+ }
+};
+
+} // namespace
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc
new file mode 100644
index 0000000..e272eb9
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc
@@ -0,0 +1,262 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "internal_defines.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "timestamp_extrapolator.h"
+#include "trace.h"
+
+namespace webrtc {
+
+VCMTimestampExtrapolator::VCMTimestampExtrapolator(TickTimeBase* clock,
+ WebRtc_Word32 vcmId,
+ WebRtc_Word32 id)
+:
+_rwLock(RWLockWrapper::CreateRWLock()),
+_vcmId(vcmId),
+_id(id),
+_clock(clock),
+_startMs(0),
+_firstTimestamp(0),
+_wrapArounds(0),
+_prevTs90khz(0),
+_lambda(1),
+_firstAfterReset(true),
+_packetCount(0),
+_startUpFilterDelayInPackets(2),
+_detectorAccumulatorPos(0),
+_detectorAccumulatorNeg(0),
+_alarmThreshold(60e3),
+_accDrift(6600), // in timestamp ticks, i.e. 15 ms
+_accMaxError(7000),
+_P11(1e10)
+{
+ Reset(_clock->MillisecondTimestamp());
+}
+
+VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
+{
+ delete _rwLock;
+}
+
+void
+VCMTimestampExtrapolator::Reset(const WebRtc_Word64 nowMs /* = -1 */)
+{
+ WriteLockScoped wl(*_rwLock);
+ if (nowMs > -1)
+ {
+ _startMs = nowMs;
+ }
+ else
+ {
+ _startMs = _clock->MillisecondTimestamp();
+ }
+ _prevMs = _startMs;
+ _firstTimestamp = 0;
+ _w[0] = 90.0;
+ _w[1] = 0;
+ _P[0][0] = 1;
+ _P[1][1] = _P11;
+ _P[0][1] = _P[1][0] = 0;
+ _firstAfterReset = true;
+ _prevTs90khz = 0;
+ _wrapArounds = 0;
+ _packetCount = 0;
+ _detectorAccumulatorPos = 0;
+ _detectorAccumulatorNeg = 0;
+}
+
+void
+VCMTimestampExtrapolator::Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace)
+{
+
+ _rwLock->AcquireLockExclusive();
+ if (tMs - _prevMs > 10e3)
+ {
+ // Ten seconds without a complete frame.
+ // Reset the extrapolator
+ _rwLock->ReleaseLockExclusive();
+ Reset();
+ _rwLock->AcquireLockExclusive();
+ }
+ else
+ {
+ _prevMs = tMs;
+ }
+
+ // Remove offset to prevent badly scaled matrices
+ tMs -= _startMs;
+
+ WebRtc_Word32 prevWrapArounds = _wrapArounds;
+ CheckForWrapArounds(ts90khz);
+ WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+ if (wrapAroundsSincePrev == 0 && ts90khz < _prevTs90khz)
+ {
+ _rwLock->ReleaseLockExclusive();
+ return;
+ }
+
+ if (_firstAfterReset)
+ {
+ // Make an initial guess of the offset,
+ // should be almost correct since tMs - _startMs
+ // should about zero at this time.
+ _w[1] = -_w[0] * tMs;
+ _firstTimestamp = ts90khz;
+ _firstAfterReset = false;
+ }
+
+ // Compensate for wraparounds by changing the line offset
+ _w[1] = _w[1] - wrapAroundsSincePrev * ((static_cast<WebRtc_Word64>(1)<<32) - 1);
+
+ double residual = (static_cast<double>(ts90khz) - _firstTimestamp) - static_cast<double>(tMs) * _w[0] - _w[1];
+ if (DelayChangeDetection(residual, trace) &&
+ _packetCount >= _startUpFilterDelayInPackets)
+ {
+ // A sudden change of average network delay has been detected.
+ // Force the filter to adjust its offset parameter by changing
+ // the offset uncertainty. Don't do this during startup.
+ _P[1][1] = _P11;
+ }
+ //T = [t(k) 1]';
+ //that = T'*w;
+ //K = P*T/(lambda + T'*P*T);
+ double K[2];
+ K[0] = _P[0][0] * tMs + _P[0][1];
+ K[1] = _P[1][0] * tMs + _P[1][1];
+ double TPT = _lambda + tMs * K[0] + K[1];
+ K[0] /= TPT;
+ K[1] /= TPT;
+ //w = w + K*(ts(k) - that);
+ _w[0] = _w[0] + K[0] * residual;
+ _w[1] = _w[1] + K[1] * residual;
+ //P = 1/lambda*(P - K*T'*P);
+ double p00 = 1 / _lambda * (_P[0][0] - (K[0] * tMs * _P[0][0] + K[0] * _P[1][0]));
+ double p01 = 1 / _lambda * (_P[0][1] - (K[0] * tMs * _P[0][1] + K[0] * _P[1][1]));
+ _P[1][0] = 1 / _lambda * (_P[1][0] - (K[1] * tMs * _P[0][0] + K[1] * _P[1][0]));
+ _P[1][1] = 1 / _lambda * (_P[1][1] - (K[1] * tMs * _P[0][1] + K[1] * _P[1][1]));
+ _P[0][0] = p00;
+ _P[0][1] = p01;
+ if (_packetCount < _startUpFilterDelayInPackets)
+ {
+ _packetCount++;
+ }
+ if (trace)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "w[0]=%f w[1]=%f ts=%u tMs=%u", _w[0], _w[1], ts90khz, tMs);
+ }
+ _rwLock->ReleaseLockExclusive();
+}
+
+WebRtc_UWord32
+VCMTimestampExtrapolator::ExtrapolateTimestamp(WebRtc_Word64 tMs) const
+{
+ ReadLockScoped rl(*_rwLock);
+ WebRtc_UWord32 timestamp = 0;
+ if (_packetCount == 0)
+ {
+ timestamp = 0;
+ }
+ else if (_packetCount < _startUpFilterDelayInPackets)
+ {
+ timestamp = static_cast<WebRtc_UWord32>(90.0 * (tMs - _prevMs) + _prevTs90khz + 0.5);
+ }
+ else
+ {
+ timestamp = static_cast<WebRtc_UWord32>(_w[0] * (tMs - _startMs) + _w[1] + _firstTimestamp + 0.5);
+ }
+ return timestamp;
+}
+
+WebRtc_Word64
+VCMTimestampExtrapolator::ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const
+{
+ ReadLockScoped rl(*_rwLock);
+ WebRtc_Word64 localTimeMs = 0;
+ if (_packetCount == 0)
+ {
+ localTimeMs = -1;
+ }
+ else if (_packetCount < _startUpFilterDelayInPackets)
+ {
+ localTimeMs = _prevMs + static_cast<WebRtc_Word64>(static_cast<double>(timestamp90khz - _prevTs90khz) / 90.0 + 0.5);
+ }
+ else
+ {
+ if (_w[0] < 1e-3)
+ {
+ localTimeMs = _startMs;
+ }
+ else
+ {
+ double timestampDiff = static_cast<double>(timestamp90khz) - static_cast<double>(_firstTimestamp);
+ localTimeMs = static_cast<WebRtc_Word64>(static_cast<double>(_startMs) + (timestampDiff - _w[1]) / _w[0] + 0.5);
+ }
+ }
+ return localTimeMs;
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp and
+// keeps track of the number of wrap arounds since reset.
+void
+VCMTimestampExtrapolator::CheckForWrapArounds(WebRtc_UWord32 ts90khz)
+{
+ if (_prevTs90khz == 0)
+ {
+ _prevTs90khz = ts90khz;
+ return;
+ }
+ if (ts90khz < _prevTs90khz)
+ {
+ // This difference will probably be less than -2^31 if we have had a wrap around
+ // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is casted to a Word32,
+ // it should be positive.
+ if (static_cast<WebRtc_Word32>(ts90khz - _prevTs90khz) > 0)
+ {
+ // Forward wrap around
+ _wrapArounds++;
+ }
+ }
+ // This difference will probably be less than -2^31 if we have had a backward wrap around.
+ // Since it is casted to a Word32, it should be positive.
+ else if (static_cast<WebRtc_Word32>(_prevTs90khz - ts90khz) > 0)
+ {
+ // Backward wrap around
+ _wrapArounds--;
+ }
+ _prevTs90khz = ts90khz;
+}
+
+bool
+VCMTimestampExtrapolator::DelayChangeDetection(double error, bool trace)
+{
+ // CUSUM detection of sudden delay changes
+ error = (error > 0) ? VCM_MIN(error, _accMaxError) : VCM_MAX(error, -_accMaxError);
+ _detectorAccumulatorPos = VCM_MAX(_detectorAccumulatorPos + error - _accDrift, (double)0);
+ _detectorAccumulatorNeg = VCM_MIN(_detectorAccumulatorNeg + error + _accDrift, (double)0);
+ if (_detectorAccumulatorPos > _alarmThreshold || _detectorAccumulatorNeg < -_alarmThreshold)
+ {
+ // Alarm
+ if (trace)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=1", _detectorAccumulatorPos, _detectorAccumulatorNeg);
+ }
+ _detectorAccumulatorPos = _detectorAccumulatorNeg = 0;
+ return true;
+ }
+ if (trace)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=0", _detectorAccumulatorPos, _detectorAccumulatorNeg);
+ }
+ return false;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h
new file mode 100644
index 0000000..901d8d4
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+
+#include "typedefs.h"
+#include "rw_lock_wrapper.h"
+
+namespace webrtc
+{
+
+class TickTimeBase;
+
+class VCMTimestampExtrapolator
+{
+public:
+ VCMTimestampExtrapolator(TickTimeBase* clock,
+ WebRtc_Word32 vcmId = 0,
+ WebRtc_Word32 receiverId = 0);
+ ~VCMTimestampExtrapolator();
+ void Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace = true);
+ WebRtc_UWord32 ExtrapolateTimestamp(WebRtc_Word64 tMs) const;
+ WebRtc_Word64 ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const;
+ void Reset(WebRtc_Word64 nowMs = -1);
+
+private:
+ void CheckForWrapArounds(WebRtc_UWord32 ts90khz);
+ bool DelayChangeDetection(double error, bool trace = true);
+ RWLockWrapper* _rwLock;
+ WebRtc_Word32 _vcmId;
+ WebRtc_Word32 _id;
+ TickTimeBase* _clock;
+ double _w[2];
+ double _P[2][2];
+ WebRtc_Word64 _startMs;
+ WebRtc_Word64 _prevMs;
+ WebRtc_UWord32 _firstTimestamp;
+ WebRtc_Word32 _wrapArounds;
+ WebRtc_UWord32 _prevTs90khz;
+ const double _lambda;
+ bool _firstAfterReset;
+ WebRtc_UWord32 _packetCount;
+ const WebRtc_UWord32 _startUpFilterDelayInPackets;
+
+ double _detectorAccumulatorPos;
+ double _detectorAccumulatorNeg;
+ const double _alarmThreshold;
+ const double _accDrift;
+ const double _accMaxError;
+ const double _P11;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_map.cc b/trunk/src/modules/video_coding/main/source/timestamp_map.cc
new file mode 100644
index 0000000..f19819b
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timestamp_map.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "timestamp_map.h"
+#include <stdlib.h>
+#include <assert.h>
+
+namespace webrtc {
+
+// Constructor. Optional parameter specifies maximum number of
+// coexisting timers.
+VCMTimestampMap::VCMTimestampMap(WebRtc_Word32 length):
+ _nextAddIx(0),
+ _nextPopIx(0)
+{
+ if (length <= 0)
+ {
+ // default
+ length = 10;
+ }
+
+ _map = new VCMTimestampDataTuple[length];
+ _length = length;
+}
+
+// Destructor.
+VCMTimestampMap::~VCMTimestampMap()
+{
+ delete [] _map;
+}
+
+// Empty the list of timers.
+void
+VCMTimestampMap::Reset()
+{
+ _nextAddIx = 0;
+ _nextPopIx = 0;
+}
+
+WebRtc_Word32
+VCMTimestampMap::Add(WebRtc_UWord32 timestamp, void* data)
+{
+ _map[_nextAddIx].timestamp = timestamp;
+ _map[_nextAddIx].data = data;
+ _nextAddIx = (_nextAddIx + 1) % _length;
+
+ if (_nextAddIx == _nextPopIx)
+ {
+ // Circular list full; forget oldest entry
+ _nextPopIx = (_nextPopIx + 1) % _length;
+ return -1;
+ }
+ return 0;
+}
+
+void*
+VCMTimestampMap::Pop(WebRtc_UWord32 timestamp)
+{
+ while (!IsEmpty())
+ {
+ if (_map[_nextPopIx].timestamp == timestamp)
+ {
+ // found start time for this timestamp
+ void* data = _map[_nextPopIx].data;
+ _map[_nextPopIx].data = NULL;
+ _nextPopIx = (_nextPopIx + 1) % _length;
+ return data;
+ }
+ else if (_map[_nextPopIx].timestamp > timestamp)
+ {
+ // the timestamp we are looking for is not in the list
+ assert(_nextPopIx < _length && _nextPopIx >= 0);
+ return NULL;
+ }
+
+ // not in this position, check next (and forget this position)
+ _nextPopIx = (_nextPopIx + 1) % _length;
+ }
+
+ // could not find matching timestamp in list
+ assert(_nextPopIx < _length && _nextPopIx >= 0);
+ return NULL;
+}
+
+// Check if no timers are currently running
+bool
+VCMTimestampMap::IsEmpty() const
+{
+ return (_nextAddIx == _nextPopIx);
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_map.h b/trunk/src/modules/video_coding/main/source/timestamp_map.h
new file mode 100644
index 0000000..fd532bc
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timestamp_map.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+struct VCMTimestampDataTuple
+{
+ WebRtc_UWord32 timestamp;
+ void* data;
+};
+
+class VCMTimestampMap
+{
+public:
+ // Constructor. Optional parameter specifies maximum number of
+ // timestamps in map.
+ VCMTimestampMap(const WebRtc_Word32 length = 10);
+
+ // Destructor.
+ ~VCMTimestampMap();
+
+ // Empty the map
+ void Reset();
+
+ WebRtc_Word32 Add(WebRtc_UWord32 timestamp, void* data);
+ void* Pop(WebRtc_UWord32 timestamp);
+
+private:
+ bool IsEmpty() const;
+
+ VCMTimestampDataTuple* _map;
+ WebRtc_Word32 _nextAddIx;
+ WebRtc_Word32 _nextPopIx;
+ WebRtc_Word32 _length;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/trunk/src/modules/video_coding/main/source/timing.cc b/trunk/src/modules/video_coding/main/source/timing.cc
new file mode 100644
index 0000000..aca05fa
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timing.cc
@@ -0,0 +1,337 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "jitter_buffer_common.h"
+#include "timing.h"
+#include "timestamp_extrapolator.h"
+
+namespace webrtc {
+
+VCMTiming::VCMTiming(TickTimeBase* clock,
+ WebRtc_Word32 vcmId,
+ WebRtc_Word32 timingId,
+ VCMTiming* masterTiming)
+:
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
+_vcmId(vcmId),
+_clock(clock),
+_timingId(timingId),
+_master(false),
+_tsExtrapolator(),
+_codecTimer(),
+_renderDelayMs(kDefaultRenderDelayMs),
+_minTotalDelayMs(0),
+_requiredDelayMs(0),
+_currentDelayMs(0),
+_prevFrameTimestamp(0)
+{
+ if (masterTiming == NULL)
+ {
+ _master = true;
+ _tsExtrapolator = new VCMTimestampExtrapolator(_clock, vcmId, timingId);
+ }
+ else
+ {
+ _tsExtrapolator = masterTiming->_tsExtrapolator;
+ }
+}
+
+VCMTiming::~VCMTiming()
+{
+ if (_master)
+ {
+ delete _tsExtrapolator;
+ }
+ delete _critSect;
+}
+
+void
+VCMTiming::Reset(WebRtc_Word64 nowMs /* = -1 */)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (nowMs > -1)
+ {
+ _tsExtrapolator->Reset(nowMs);
+ }
+ else
+ {
+ _tsExtrapolator->Reset();
+ }
+ _codecTimer.Reset();
+ _renderDelayMs = kDefaultRenderDelayMs;
+ _minTotalDelayMs = 0;
+ _requiredDelayMs = 0;
+ _currentDelayMs = 0;
+ _prevFrameTimestamp = 0;
+}
+
+void VCMTiming::ResetDecodeTime()
+{
+ _codecTimer.Reset();
+}
+
+void
+VCMTiming::SetRenderDelay(WebRtc_UWord32 renderDelayMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ _renderDelayMs = renderDelayMs;
+}
+
+void
+VCMTiming::SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ _minTotalDelayMs = minTotalDelayMs;
+}
+
+void
+VCMTiming::SetRequiredDelay(WebRtc_UWord32 requiredDelayMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (requiredDelayMs != _requiredDelayMs)
+ {
+ if (_master)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Desired jitter buffer level: %u ms", requiredDelayMs);
+ }
+ _requiredDelayMs = requiredDelayMs;
+ }
+}
+
+void VCMTiming::UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp)
+{
+ CriticalSectionScoped cs(_critSect);
+ WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
+
+ // Make sure we try to sync with audio
+ if (targetDelayMs < _minTotalDelayMs)
+ {
+ targetDelayMs = _minTotalDelayMs;
+ }
+
+ if (_currentDelayMs == 0)
+ {
+ // Not initialized, set current delay to target.
+ _currentDelayMs = targetDelayMs;
+ }
+ else if (targetDelayMs != _currentDelayMs)
+ {
+ WebRtc_Word64 delayDiffMs = static_cast<WebRtc_Word64>(targetDelayMs) -
+ _currentDelayMs;
+ // Never change the delay with more than 100 ms every second. If we're changing the
+ // delay in too large steps we will get noticable freezes. By limiting the change we
+ // can increase the delay in smaller steps, which will be experienced as the video is
+ // played in slow motion. When lowering the delay the video will be played at a faster
+ // pace.
+ WebRtc_Word64 maxChangeMs = 0;
+ if (frameTimestamp < 0x0000ffff && _prevFrameTimestamp > 0xffff0000)
+ {
+ // wrap
+ maxChangeMs = kDelayMaxChangeMsPerS * (frameTimestamp +
+ (static_cast<WebRtc_Word64>(1)<<32) - _prevFrameTimestamp) / 90000;
+ }
+ else
+ {
+ maxChangeMs = kDelayMaxChangeMsPerS *
+ (frameTimestamp - _prevFrameTimestamp) / 90000;
+ }
+ if (maxChangeMs <= 0)
+ {
+ // Any changes less than 1 ms are truncated and
+ // will be postponed. Negative change will be due
+ // to reordering and should be ignored.
+ return;
+ }
+ else if (delayDiffMs < -maxChangeMs)
+ {
+ delayDiffMs = -maxChangeMs;
+ }
+ else if (delayDiffMs > maxChangeMs)
+ {
+ delayDiffMs = maxChangeMs;
+ }
+ _currentDelayMs = _currentDelayMs + static_cast<WebRtc_Word32>(delayDiffMs);
+ }
+ _prevFrameTimestamp = frameTimestamp;
+}
+
+void VCMTiming::UpdateCurrentDelay(WebRtc_Word64 renderTimeMs,
+ WebRtc_Word64 actualDecodeTimeMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
+ // Make sure we try to sync with audio
+ if (targetDelayMs < _minTotalDelayMs)
+ {
+ targetDelayMs = _minTotalDelayMs;
+ }
+ WebRtc_Word64 delayedMs = actualDecodeTimeMs -
+ (renderTimeMs - MaxDecodeTimeMs() - _renderDelayMs);
+ if (delayedMs < 0)
+ {
+ return;
+ }
+ else if (_currentDelayMs + delayedMs <= targetDelayMs)
+ {
+ _currentDelayMs += static_cast<WebRtc_UWord32>(delayedMs);
+ }
+ else
+ {
+ _currentDelayMs = targetDelayMs;
+ }
+}
+
+WebRtc_Word32
+VCMTiming::StopDecodeTimer(WebRtc_UWord32 timeStamp,
+ WebRtc_Word64 startTimeMs,
+ WebRtc_Word64 nowMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ const WebRtc_Word32 maxDecTime = MaxDecodeTimeMs();
+ WebRtc_Word32 timeDiffMs = _codecTimer.StopTimer(startTimeMs, nowMs);
+ if (timeDiffMs < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Codec timer error: %d", timeDiffMs);
+ assert(false);
+ }
+
+ if (_master)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Frame decoded: timeStamp=%u decTime=%d maxDecTime=%u, at %u",
+ timeStamp, timeDiffMs, maxDecTime, MaskWord64ToUWord32(nowMs));
+ }
+ return 0;
+}
+
+void
+VCMTiming::IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 nowMs)
+{
+ CriticalSectionScoped cs(_critSect);
+ _tsExtrapolator->Update(nowMs, timeStamp, _master);
+}
+
+WebRtc_Word64
+VCMTiming::RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
+{
+ CriticalSectionScoped cs(_critSect);
+ const WebRtc_Word64 renderTimeMs = RenderTimeMsInternal(frameTimestamp, nowMs);
+ if (renderTimeMs < 0)
+ {
+ return renderTimeMs;
+ }
+ if (_master)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Render frame %u at %u. Render delay %u, required delay %u,"
+ " max decode time %u, min total delay %u",
+ frameTimestamp, MaskWord64ToUWord32(renderTimeMs), _renderDelayMs,
+ _requiredDelayMs, MaxDecodeTimeMs(),_minTotalDelayMs);
+ }
+ return renderTimeMs;
+}
+
+WebRtc_Word64
+VCMTiming::RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
+{
+ WebRtc_Word64 estimatedCompleteTimeMs =
+ _tsExtrapolator->ExtrapolateLocalTime(frameTimestamp);
+ if (estimatedCompleteTimeMs - nowMs > kMaxVideoDelayMs)
+ {
+ if (_master)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Timestamp arrived 2 seconds early, reset statistics",
+ frameTimestamp, estimatedCompleteTimeMs);
+ }
+ return -1;
+ }
+ if (_master)
+ {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "ExtrapolateLocalTime(%u)=%u ms",
+ frameTimestamp, MaskWord64ToUWord32(estimatedCompleteTimeMs));
+ }
+ if (estimatedCompleteTimeMs == -1)
+ {
+ estimatedCompleteTimeMs = nowMs;
+ }
+
+ return estimatedCompleteTimeMs + _currentDelayMs;
+}
+
+// Must be called from inside a critical section
+WebRtc_Word32
+VCMTiming::MaxDecodeTimeMs(FrameType frameType /*= kVideoFrameDelta*/) const
+{
+ const WebRtc_Word32 decodeTimeMs = _codecTimer.RequiredDecodeTimeMs(frameType);
+
+ if (decodeTimeMs < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+ "Negative maximum decode time: %d", decodeTimeMs);
+ return -1;
+ }
+ return decodeTimeMs;
+}
+
+WebRtc_UWord32
+VCMTiming::MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const
+{
+ CriticalSectionScoped cs(_critSect);
+
+ const WebRtc_Word64 maxWaitTimeMs = renderTimeMs - nowMs -
+ MaxDecodeTimeMs() - _renderDelayMs;
+
+ if (maxWaitTimeMs < 0)
+ {
+ return 0;
+ }
+ return static_cast<WebRtc_UWord32>(maxWaitTimeMs);
+}
+
+bool
+VCMTiming::EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const
+{
+ CriticalSectionScoped cs(_critSect);
+ WebRtc_Word32 maxDecodeTimeMs = MaxDecodeTimeMs();
+ if (maxDecodeTimeMs < 0)
+ {
+ // Haven't decoded any frames yet, try decoding one to get an estimate
+ // of the decode time.
+ return true;
+ }
+ else if (maxDecodeTimeMs == 0)
+ {
+ // Decode time is less than 1, set to 1 for now since
+ // we don't have any better precision. Count ticks later?
+ maxDecodeTimeMs = 1;
+ }
+ return static_cast<WebRtc_Word32>(availableProcessingTimeMs) - maxDecodeTimeMs > 0;
+}
+
+WebRtc_UWord32
+VCMTiming::TargetVideoDelay() const
+{
+ CriticalSectionScoped cs(_critSect);
+ return TargetDelayInternal();
+}
+
+WebRtc_UWord32
+VCMTiming::TargetDelayInternal() const
+{
+ return _requiredDelayMs + MaxDecodeTimeMs() + _renderDelayMs;
+}
+
+}
diff --git a/trunk/src/modules/video_coding/main/source/timing.h b/trunk/src/modules/video_coding/main/source/timing.h
new file mode 100644
index 0000000..41a4945
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/timing.h
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+
+#include "typedefs.h"
+#include "critical_section_wrapper.h"
+#include "codec_timer.h"
+
+namespace webrtc
+{
+
+class TickTimeBase;
+class VCMTimestampExtrapolator;
+
+class VCMTiming
+{
+public:
+ // The primary timing component should be passed
+ // if this is the dual timing component.
+ VCMTiming(TickTimeBase* clock,
+ WebRtc_Word32 vcmId = 0,
+ WebRtc_Word32 timingId = 0,
+ VCMTiming* masterTiming = NULL);
+ ~VCMTiming();
+
+ // Resets the timing to the initial state.
+ void Reset(WebRtc_Word64 nowMs = -1);
+ void ResetDecodeTime();
+
+ // The amount of time needed to render an image. Defaults to 10 ms.
+ void SetRenderDelay(WebRtc_UWord32 renderDelayMs);
+
+ // The minimum time the video must be delayed on the receiver to
+ // get the desired jitter buffer level.
+ void SetRequiredDelay(WebRtc_UWord32 requiredDelayMs);
+
+ // Minimum total delay required to sync video with audio.
+ void SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs);
+
+ // Increases or decreases the current delay to get closer to the target delay.
+ // Calculates how long it has been since the previous call to this function,
+ // and increases/decreases the delay in proportion to the time difference.
+ void UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp);
+
+ // Increases or decreases the current delay to get closer to the target delay.
+ // Given the actual decode time in ms and the render time in ms for a frame, this
+ // function calculates how late the frame is and increases the delay accordingly.
+ void UpdateCurrentDelay(WebRtc_Word64 renderTimeMs, WebRtc_Word64 actualDecodeTimeMs);
+
+ // Stops the decoder timer, should be called when the decoder returns a frame
+ // or when the decoded frame callback is called.
+ WebRtc_Word32 StopDecodeTimer(WebRtc_UWord32 timeStamp,
+ WebRtc_Word64 startTimeMs,
+ WebRtc_Word64 nowMs);
+
+ // Used to report that a frame is passed to decoding. Updates the timestamp filter
+ // which is used to map between timestamps and receiver system time.
+ void IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 lastPacketTimeMs);
+
+ // Returns the receiver system time when the frame with timestamp frameTimestamp
+ // should be rendered, assuming that the system time currently is nowMs.
+ WebRtc_Word64 RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const;
+
+ // Returns the maximum time in ms that we can wait for a frame to become complete
+ // before we must pass it to the decoder.
+ WebRtc_UWord32 MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const;
+
+ // Returns the current target delay which is required delay + decode time + render
+ // delay.
+ WebRtc_UWord32 TargetVideoDelay() const;
+
+ // Calculates whether or not there is enough time to decode a frame given a
+ // certain amount of processing time.
+ bool EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const;
+
+ enum { kDefaultRenderDelayMs = 10 };
+ enum { kDelayMaxChangeMsPerS = 100 };
+
+protected:
+ WebRtc_Word32 MaxDecodeTimeMs(FrameType frameType = kVideoFrameDelta) const;
+ WebRtc_Word64 RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp,
+ WebRtc_Word64 nowMs) const;
+ WebRtc_UWord32 TargetDelayInternal() const;
+
+private:
+ CriticalSectionWrapper* _critSect;
+ WebRtc_Word32 _vcmId;
+ TickTimeBase* _clock;
+ WebRtc_Word32 _timingId;
+ bool _master;
+ VCMTimestampExtrapolator* _tsExtrapolator;
+ VCMCodecTimer _codecTimer;
+ WebRtc_UWord32 _renderDelayMs;
+ WebRtc_UWord32 _minTotalDelayMs;
+ WebRtc_UWord32 _requiredDelayMs;
+ WebRtc_UWord32 _currentDelayMs;
+ WebRtc_UWord32 _prevFrameTimestamp;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
diff --git a/trunk/src/modules/video_coding/main/source/video_coding.gypi b/trunk/src/modules/video_coding/main/source/video_coding.gypi
new file mode 100644
index 0000000..3a99283
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/video_coding.gypi
@@ -0,0 +1,106 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_video_coding',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_i420',
+ 'webrtc_vp8',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ '../../codecs/interface',
+ '../../../../common_video/interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../codecs/interface',
+ ],
+ },
+ 'sources': [
+ # interfaces
+ '../interface/video_coding.h',
+ '../interface/video_coding_defines.h',
+
+ # headers
+ 'codec_database.h',
+ 'codec_timer.h',
+ 'content_metrics_processing.h',
+ 'decoding_state.h',
+ 'encoded_frame.h',
+ 'er_tables_xor.h',
+ 'event.h',
+ 'exp_filter.h',
+ 'fec_tables_xor.h',
+ 'frame_buffer.h',
+ 'frame_dropper.h',
+ 'generic_decoder.h',
+ 'generic_encoder.h',
+ 'inter_frame_delay.h',
+ 'internal_defines.h',
+ 'jitter_buffer.h',
+ 'jitter_buffer_common.h',
+ 'jitter_estimator.h',
+ 'media_opt_util.h',
+ 'media_optimization.h',
+ 'nack_fec_tables.h',
+ 'packet.h',
+ 'qm_select_data.h',
+ 'qm_select.h',
+ 'receiver.h',
+ 'rtt_filter.h',
+ 'session_info.h',
+ 'tick_time_base.h',
+ 'timestamp_extrapolator.h',
+ 'timestamp_map.h',
+ 'timing.h',
+ 'video_coding_impl.h',
+
+ # sources
+ 'codec_database.cc',
+ 'codec_timer.cc',
+ 'content_metrics_processing.cc',
+ 'decoding_state.cc',
+ 'encoded_frame.cc',
+ 'exp_filter.cc',
+ 'frame_buffer.cc',
+ 'frame_dropper.cc',
+ 'generic_decoder.cc',
+ 'generic_encoder.cc',
+ 'inter_frame_delay.cc',
+ 'jitter_buffer.cc',
+ 'jitter_buffer_common.cc',
+ 'jitter_estimator.cc',
+ 'media_opt_util.cc',
+ 'media_optimization.cc',
+ 'packet.cc',
+ 'qm_select.cc',
+ 'receiver.cc',
+ 'rtt_filter.cc',
+ 'session_info.cc',
+ 'timestamp_extrapolator.cc',
+ 'timestamp_map.cc',
+ 'timing.cc',
+ 'video_coding_impl.cc',
+ ], # source
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_impl.cc b/trunk/src/modules/video_coding/main/source/video_coding_impl.cc
new file mode 100644
index 0000000..771f702
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/video_coding_impl.cc
@@ -0,0 +1,1385 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_coding_impl.h"
+#include "common_types.h"
+#include "encoded_frame.h"
+#include "jitter_buffer.h"
+#include "packet.h"
+#include "trace.h"
+#include "video_codec_interface.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+
+namespace webrtc
+{
+
+//#define DEBUG_DECODER_BIT_STREAM
+//#define DEBUG_ENCODER_INPUT
+
+WebRtc_UWord32
+VCMProcessTimer::Period() const
+{
+ return _periodMs;
+}
+
+WebRtc_UWord32
+VCMProcessTimer::TimeUntilProcess() const
+{
+ return static_cast<WebRtc_UWord32>(
+ VCM_MAX(static_cast<WebRtc_Word64>(_periodMs) -
+ (_clock->MillisecondTimestamp() - _latestMs), 0));
+}
+
+void
+VCMProcessTimer::Processed()
+{
+ _latestMs = _clock->MillisecondTimestamp();
+}
+
+VideoCodingModuleImpl::VideoCodingModuleImpl(const WebRtc_Word32 id,
+ TickTimeBase* clock,
+ bool delete_clock_on_destroy)
+:
+_id(id),
+clock_(clock),
+delete_clock_on_destroy_(delete_clock_on_destroy),
+_receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+_receiverInited(false),
+_timing(clock_, id, 1),
+_dualTiming(clock_, id, 2, &_timing),
+_receiver(_timing, clock_, id, 1),
+_dualReceiver(_dualTiming, clock_, id, 2, false),
+_decodedFrameCallback(_timing, clock_),
+_dualDecodedFrameCallback(_dualTiming, clock_),
+_frameTypeCallback(NULL),
+_frameStorageCallback(NULL),
+_receiveStatsCallback(NULL),
+_packetRequestCallback(NULL),
+_decoder(NULL),
+_dualDecoder(NULL),
+_bitStreamBeforeDecoder(NULL),
+_frameFromFile(),
+_keyRequestMode(kKeyOnError),
+_scheduleKeyRequest(false),
+
+_sendCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+_encoder(),
+_encodedFrameCallback(),
+_mediaOpt(id, clock_),
+_sendCodecType(kVideoCodecUnknown),
+_sendStatsCallback(NULL),
+_encoderInputFile(NULL),
+
+_codecDataBase(id),
+_receiveStatsTimer(1000, clock_),
+_sendStatsTimer(1000, clock_),
+_retransmissionTimer(10, clock_),
+_keyRequestTimer(500, clock_)
+{
+ assert(clock_);
+ for (int i = 0; i < kMaxSimulcastStreams; i++)
+ {
+ _nextFrameType[i] = kVideoFrameDelta;
+ }
+#ifdef DEBUG_DECODER_BIT_STREAM
+ _bitStreamBeforeDecoder = fopen("decoderBitStream.bit", "wb");
+#endif
+#ifdef DEBUG_ENCODER_INPUT
+ _encoderInputFile = fopen("encoderInput.yuv", "wb");
+#endif
+}
+
+VideoCodingModuleImpl::~VideoCodingModuleImpl()
+{
+ if (_dualDecoder != NULL)
+ {
+ _codecDataBase.ReleaseDecoder(_dualDecoder);
+ }
+ delete _receiveCritSect;
+ delete _sendCritSect;
+ if (delete_clock_on_destroy_) delete clock_;
+#ifdef DEBUG_DECODER_BIT_STREAM
+ fclose(_bitStreamBeforeDecoder);
+#endif
+#ifdef DEBUG_ENCODER_INPUT
+ fclose(_encoderInputFile);
+#endif
+}
+
+VideoCodingModule*
+VideoCodingModule::Create(const WebRtc_Word32 id)
+{
+ return new VideoCodingModuleImpl(id, new TickTimeBase(), true);
+}
+
+VideoCodingModule*
+VideoCodingModule::Create(const WebRtc_Word32 id, TickTimeBase* clock)
+{
+ assert(clock);
+ return new VideoCodingModuleImpl(id, clock, false);
+}
+
+void
+VideoCodingModule::Destroy(VideoCodingModule* module)
+{
+ if (module != NULL)
+ {
+ delete static_cast<VideoCodingModuleImpl*>(module);
+ }
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::Process()
+{
+ WebRtc_Word32 returnValue = VCM_OK;
+
+ // Receive-side statistics
+ if (_receiveStatsTimer.TimeUntilProcess() == 0)
+ {
+ _receiveStatsTimer.Processed();
+ if (_receiveStatsCallback != NULL)
+ {
+ WebRtc_UWord32 bitRate;
+ WebRtc_UWord32 frameRate;
+ const WebRtc_Word32 ret = _receiver.ReceiveStatistics(bitRate,
+ frameRate);
+ if (ret == 0)
+ {
+ _receiveStatsCallback->ReceiveStatistics(bitRate, frameRate);
+ }
+ else if (returnValue == VCM_OK)
+ {
+ returnValue = ret;
+ }
+ }
+ }
+
+ // Send-side statistics
+ if (_sendStatsTimer.TimeUntilProcess() == 0)
+ {
+ _sendStatsTimer.Processed();
+ if (_sendStatsCallback != NULL)
+ {
+ WebRtc_UWord32 bitRate;
+ WebRtc_UWord32 frameRate;
+ {
+ CriticalSectionScoped cs(_sendCritSect);
+ bitRate = static_cast<WebRtc_UWord32>(
+ _mediaOpt.SentBitRate() + 0.5f);
+ frameRate = static_cast<WebRtc_UWord32>(
+ _mediaOpt.SentFrameRate() + 0.5f);
+ }
+ _sendStatsCallback->SendStatistics(bitRate, frameRate);
+ }
+ }
+
+ // Packet retransmission requests
+ if (_retransmissionTimer.TimeUntilProcess() == 0)
+ {
+ _retransmissionTimer.Processed();
+ if (_packetRequestCallback != NULL)
+ {
+ WebRtc_UWord16 nackList[kNackHistoryLength];
+ WebRtc_UWord16 length = kNackHistoryLength;
+ const WebRtc_Word32 ret = NackList(nackList, length);
+ if (ret != VCM_OK && returnValue == VCM_OK)
+ {
+ returnValue = ret;
+ }
+ if (length > 0)
+ {
+ _packetRequestCallback->ResendPackets(nackList, length);
+ }
+ }
+ }
+
+ // Key frame requests
+ if (_keyRequestTimer.TimeUntilProcess() == 0)
+ {
+ _keyRequestTimer.Processed();
+ if (_scheduleKeyRequest && _frameTypeCallback != NULL)
+ {
+ const WebRtc_Word32 ret = RequestKeyFrame();
+ if (ret != VCM_OK && returnValue == VCM_OK)
+ {
+ returnValue = ret;
+ }
+ }
+ }
+
+ return returnValue;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::Id() const
+{
+ CriticalSectionScoped receiveCs(_receiveCritSect);
+ {
+ CriticalSectionScoped sendCs(_sendCritSect);
+ return _id;
+ }
+}
+
+// Change the unique identifier of this object
+WebRtc_Word32
+VideoCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped receiveCs(_receiveCritSect);
+ {
+ CriticalSectionScoped sendCs(_sendCritSect);
+ _id = id;
+ return VCM_OK;
+ }
+}
+
+// Returns the number of milliseconds until the module wants a worker thread to
+// call Process
+WebRtc_Word32
+VideoCodingModuleImpl::TimeUntilNextProcess()
+{
+ WebRtc_UWord32 timeUntilNextProcess = VCM_MIN(
+ _receiveStatsTimer.TimeUntilProcess(),
+ _sendStatsTimer.TimeUntilProcess());
+ if ((_receiver.NackMode() != kNoNack) ||
+ (_dualReceiver.State() != kPassive))
+ {
+ // We need a Process call more often if we are relying on
+ // retransmissions
+ timeUntilNextProcess = VCM_MIN(timeUntilNextProcess,
+ _retransmissionTimer.TimeUntilProcess());
+ }
+ timeUntilNextProcess = VCM_MIN(timeUntilNextProcess,
+ _keyRequestTimer.TimeUntilProcess());
+
+ return timeUntilNextProcess;
+}
+
+// Get number of supported codecs
+WebRtc_UWord8
+VideoCodingModule::NumberOfCodecs()
+{
+ return VCMCodecDataBase::NumberOfCodecs();
+}
+
+// Get supported codec with id
+WebRtc_Word32
+VideoCodingModule::Codec(WebRtc_UWord8 listId, VideoCodec* codec)
+{
+ if (codec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ return VCMCodecDataBase::Codec(listId, codec);
+}
+
+// Get supported codec with type
+WebRtc_Word32
+VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec)
+{
+ if (codec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ return VCMCodecDataBase::Codec(codecType, codec);
+}
+
+/*
+* Sender
+*/
+
+// Reset send side to initial state - all components
+WebRtc_Word32
+VideoCodingModuleImpl::InitializeSender()
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ _codecDataBase.ResetSender();
+ _encoder = NULL;
+ _encodedFrameCallback.SetTransportCallback(NULL);
+ // setting default bitRate and frameRate to 0
+ _mediaOpt.SetEncodingData(kVideoCodecUnknown, 0, 0, 0, 0, 0, 0);
+ _mediaOpt.Reset(); // Resetting frame dropper
+ return VCM_OK;
+}
+
+// Register the send codec to be used.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterSendCodec(const VideoCodec* sendCodec,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ if (sendCodec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ WebRtc_Word32 ret = _codecDataBase.RegisterSendCodec(sendCodec,
+ numberOfCores,
+ maxPayloadSize);
+ if (ret < 0)
+ {
+ return ret;
+ }
+
+ _encoder = _codecDataBase.SetEncoder(sendCodec, &_encodedFrameCallback);
+ if (_encoder == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to initialize encoder");
+ return VCM_CODEC_ERROR;
+ }
+ _sendCodecType = sendCodec->codecType;
+ int numLayers = (_sendCodecType != kVideoCodecVP8) ? 1 :
+ sendCodec->codecSpecific.VP8.numberOfTemporalLayers;
+
+ _mediaOpt.SetEncodingData(_sendCodecType,
+ sendCodec->maxBitrate,
+ sendCodec->maxFramerate,
+ sendCodec->startBitrate,
+ sendCodec->width,
+ sendCodec->height,
+ numLayers);
+ _mediaOpt.SetMtu(maxPayloadSize);
+
+ return VCM_OK;
+}
+
+// Get current send codec
+WebRtc_Word32
+VideoCodingModuleImpl::SendCodec(VideoCodec* currentSendCodec) const
+{
+ CriticalSectionScoped cs(_sendCritSect);
+
+ if (currentSendCodec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ return _codecDataBase.SendCodec(currentSendCodec);
+}
+
+// Get the current send codec type
+VideoCodecType
+VideoCodingModuleImpl::SendCodec() const
+{
+ CriticalSectionScoped cs(_sendCritSect);
+
+ return _codecDataBase.SendCodec();
+}
+
+// Register an external decoder object.
+// This can not be used together with external decoder callbacks.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterExternalEncoder(VideoEncoder* externalEncoder,
+ WebRtc_UWord8 payloadType,
+ bool internalSource /*= false*/)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+
+ if (externalEncoder == NULL)
+ {
+ bool wasSendCodec = false;
+ const WebRtc_Word32 ret = _codecDataBase.DeRegisterExternalEncoder(
+ payloadType,
+ wasSendCodec);
+ if (wasSendCodec)
+ {
+ // Make sure the VCM doesn't use the de-registered codec
+ _encoder = NULL;
+ }
+ return ret;
+ }
+ return _codecDataBase.RegisterExternalEncoder(externalEncoder,
+ payloadType,
+ internalSource);
+}
+
+// Get codec config parameters
+WebRtc_Word32
+VideoCodingModuleImpl::CodecConfigParameters(WebRtc_UWord8* buffer,
+ WebRtc_Word32 size)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ if (_encoder != NULL)
+ {
+ return _encoder->CodecConfigParameters(buffer, size);
+ }
+ return VCM_UNINITIALIZED;
+}
+
+// Get encode bitrate
+int VideoCodingModuleImpl::Bitrate(unsigned int* bitrate) const
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ // return the bit rate which the encoder is set to
+ if (!_encoder) {
+ return VCM_UNINITIALIZED;
+ }
+ *bitrate = _encoder->BitRate();
+ return 0;
+}
+
+// Get encode frame rate
+int VideoCodingModuleImpl::FrameRate(unsigned int* framerate) const
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ // input frame rate, not compensated
+ if (!_encoder) {
+ return VCM_UNINITIALIZED;
+ }
+ *framerate = _encoder->FrameRate();
+ return 0;
+}
+
+// Set channel parameters
+WebRtc_Word32
+VideoCodingModuleImpl::SetChannelParameters(WebRtc_UWord32 availableBandWidth,
+ WebRtc_UWord8 lossRate,
+ WebRtc_UWord32 rtt)
+{
+ WebRtc_Word32 ret = 0;
+ {
+ CriticalSectionScoped sendCs(_sendCritSect);
+ WebRtc_UWord32 targetRate = _mediaOpt.SetTargetRates(availableBandWidth,
+ lossRate,
+ rtt);
+ if (_encoder != NULL)
+ {
+ ret = _encoder->SetChannelParameters(lossRate, rtt);
+ if (ret < 0 )
+ {
+ return ret;
+ }
+ ret = (WebRtc_Word32)_encoder->SetRates(targetRate,
+ _mediaOpt.InputFrameRate());
+ if (ret < 0)
+ {
+ return ret;
+ }
+ }
+ else
+ {
+ return VCM_UNINITIALIZED;
+ } // encoder
+ }// send side
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::SetReceiveChannelParameters(WebRtc_UWord32 rtt)
+{
+ CriticalSectionScoped receiveCs(_receiveCritSect);
+ _receiver.UpdateRtt(rtt);
+ return 0;
+}
+
+// Register a transport callback which will be called to deliver the encoded
+// buffers
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterTransportCallback(
+ VCMPacketizationCallback* transport)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ _encodedFrameCallback.SetMediaOpt(&_mediaOpt);
+ _encodedFrameCallback.SetTransportCallback(transport);
+ return VCM_OK;
+}
+
+// Register video output information callback which will be called to deliver
+// information about the video stream produced by the encoder, for instance the
+// average frame rate and bit rate.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterSendStatisticsCallback(
+ VCMSendStatisticsCallback* sendStats)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ _sendStatsCallback = sendStats;
+ return VCM_OK;
+}
+
+// Register a video quality settings callback which will be called when frame
+// rate/dimensions need to be updated for video quality optimization
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterVideoQMCallback(
+ VCMQMSettingsCallback* videoQMSettings)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ return _mediaOpt.RegisterVideoQMCallback(videoQMSettings);
+}
+
+
+// Register a video protection callback which will be called to deliver the
+// requested FEC rate and NACK status (on/off).
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterProtectionCallback(
+ VCMProtectionCallback* protection)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.RegisterProtectionCallback(protection);
+ return VCM_OK;
+}
+
+// Enable or disable a video protection method.
+WebRtc_Word32
+VideoCodingModuleImpl::SetVideoProtection(VCMVideoProtection videoProtection,
+ bool enable)
+{
+
+ switch (videoProtection)
+ {
+
+ case kProtectionNack:
+ {
+ // Both send-side and receive-side
+ SetVideoProtection(kProtectionNackSender, enable);
+ SetVideoProtection(kProtectionNackReceiver, enable);
+ break;
+ }
+
+ case kProtectionNackSender:
+ {
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.EnableProtectionMethod(enable, kNack);
+ break;
+ }
+
+ case kProtectionNackReceiver:
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (enable)
+ {
+ _receiver.SetNackMode(kNackInfinite);
+ }
+ else
+ {
+ _receiver.SetNackMode(kNoNack);
+ }
+ break;
+ }
+
+ case kProtectionDualDecoder:
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (enable)
+ {
+ _receiver.SetNackMode(kNoNack);
+ _dualReceiver.SetNackMode(kNackInfinite);
+ }
+ else
+ {
+ _dualReceiver.SetNackMode(kNoNack);
+ }
+ break;
+ }
+
+ case kProtectionKeyOnLoss:
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (enable)
+ {
+ _keyRequestMode = kKeyOnLoss;
+ }
+ else if (_keyRequestMode == kKeyOnLoss)
+ {
+ _keyRequestMode = kKeyOnError; // default mode
+ }
+ else
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ break;
+ }
+
+ case kProtectionKeyOnKeyLoss:
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (enable)
+ {
+ _keyRequestMode = kKeyOnKeyLoss;
+ }
+ else if (_keyRequestMode == kKeyOnKeyLoss)
+ {
+ _keyRequestMode = kKeyOnError; // default mode
+ }
+ else
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ break;
+ }
+
+ case kProtectionNackFEC:
+ {
+ {
+ // Receive side
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (enable)
+ {
+ _receiver.SetNackMode(kNackHybrid);
+ }
+ else
+ {
+ _receiver.SetNackMode(kNoNack);
+ }
+ }
+ // Send Side
+ {
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.EnableProtectionMethod(enable, kNackFec);
+ }
+ break;
+ }
+
+ case kProtectionFEC:
+ {
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.EnableProtectionMethod(enable, kFec);
+ break;
+ }
+
+ case kProtectionPeriodicKeyFrames:
+ {
+ CriticalSectionScoped cs(_sendCritSect);
+ return _codecDataBase.SetPeriodicKeyFrames(enable);
+ break;
+ }
+ }
+ return VCM_OK;
+}
+
+// Add one raw video frame to the encoder, blocking.
+WebRtc_Word32
+VideoCodingModuleImpl::AddVideoFrame(const VideoFrame& videoFrame,
+ const VideoContentMetrics* contentMetrics,
+ const CodecSpecificInfo* codecSpecificInfo)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+
+ if (_encoder == NULL)
+ {
+ return VCM_UNINITIALIZED;
+ }
+ if (_nextFrameType[0] == kFrameEmpty)
+ {
+ return VCM_OK;
+ }
+ _mediaOpt.UpdateIncomingFrameRate();
+
+ if (_mediaOpt.DropFrame())
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Drop frame due to bitrate");
+ }
+ else
+ {
+ _mediaOpt.updateContentData(contentMetrics);
+ WebRtc_Word32 ret = _encoder->Encode(videoFrame,
+ codecSpecificInfo,
+ _nextFrameType);
+#ifdef DEBUG_ENCODER_INPUT
+ if (_encoderInputFile != NULL)
+ {
+ fwrite(videoFrame.Buffer(), 1, videoFrame.Length(),
+ _encoderInputFile);
+ }
+#endif
+ if (ret < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Encode error: %d", ret);
+ return ret;
+ }
+ for (int i = 0; i < kMaxSimulcastStreams; i++)
+ {
+ _nextFrameType[i] = kVideoFrameDelta; // default frame type
+ }
+ }
+ return VCM_OK;
+}
+
+// Next frame encoded should be of the type frameType
+// Good for only one frame
+WebRtc_Word32
+VideoCodingModuleImpl::FrameTypeRequest(FrameType frameType,
+ WebRtc_UWord8 simulcastIdx)
+{
+ assert(simulcastIdx < kMaxSimulcastStreams);
+
+
+ CriticalSectionScoped cs(_sendCritSect);
+ _nextFrameType[simulcastIdx] = frameType;
+ if (_encoder != NULL && _encoder->InternalSource())
+ {
+ // Try to request the frame if we have an external encoder with
+ // internal source since AddVideoFrame never will be called.
+ if (_encoder->RequestFrame(_nextFrameType) == WEBRTC_VIDEO_CODEC_OK)
+ {
+ _nextFrameType[simulcastIdx] = kVideoFrameDelta;
+ }
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::EnableFrameDropper(bool enable)
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.EnableFrameDropper(enable);
+ return VCM_OK;
+}
+
+
+WebRtc_Word32
+VideoCodingModuleImpl::SentFrameCount(VCMFrameCount &frameCount) const
+{
+ CriticalSectionScoped cs(_sendCritSect);
+ return _mediaOpt.SentFrameCount(frameCount);
+}
+
+// Initialize receiver, resets codec database etc
+WebRtc_Word32
+VideoCodingModuleImpl::InitializeReceiver()
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ WebRtc_Word32 ret = _receiver.Initialize();
+ if (ret < 0)
+ {
+ return ret;
+ }
+
+ ret = _dualReceiver.Initialize();
+ if (ret < 0)
+ {
+ return ret;
+ }
+ _codecDataBase.ResetReceiver();
+ _timing.Reset();
+
+ _decoder = NULL;
+ _decodedFrameCallback.SetUserReceiveCallback(NULL);
+ _receiverInited = true;
+ _frameTypeCallback = NULL;
+ _frameStorageCallback = NULL;
+ _receiveStatsCallback = NULL;
+ _packetRequestCallback = NULL;
+ _keyRequestMode = kKeyOnError;
+ _scheduleKeyRequest = false;
+
+ return VCM_OK;
+}
+
+// Register a receive callback. Will be called whenever there is a new frame
+// ready for rendering.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterReceiveCallback(
+ VCMReceiveCallback* receiveCallback)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ _decodedFrameCallback.SetUserReceiveCallback(receiveCallback);
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterReceiveStatisticsCallback(
+ VCMReceiveStatisticsCallback* receiveStats)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ _receiveStatsCallback = receiveStats;
+ return VCM_OK;
+}
+
+// Register an externally defined decoder/render object.
+// Can be a decoder only or a decoder coupled with a renderer.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (externalDecoder == NULL)
+ {
+ // Make sure the VCM updates the decoder next time it decodes.
+ _decoder = NULL;
+ return _codecDataBase.DeRegisterExternalDecoder(payloadType);
+ }
+ else
+ {
+ return _codecDataBase.RegisterExternalDecoder(externalDecoder,
+ payloadType,
+ internalRenderTiming);
+ }
+}
+
+// Register a frame type request callback.
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterFrameTypeCallback(
+ VCMFrameTypeCallback* frameTypeCallback)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ _frameTypeCallback = frameTypeCallback;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterFrameStorageCallback(
+ VCMFrameStorageCallback* frameStorageCallback)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ _frameStorageCallback = frameStorageCallback;
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterPacketRequestCallback(
+ VCMPacketRequestCallback* callback)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ _packetRequestCallback = callback;
+ return VCM_OK;
+}
+
+// Decode next frame, blocking.
+// Should be called as often as possible to get the most out of the decoder.
+WebRtc_Word32
+VideoCodingModuleImpl::Decode(WebRtc_UWord16 maxWaitTimeMs)
+{
+ WebRtc_Word64 nextRenderTimeMs;
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (!_receiverInited)
+ {
+ return VCM_UNINITIALIZED;
+ }
+ if (!_codecDataBase.DecoderRegistered())
+ {
+ return VCM_NO_CODEC_REGISTERED;
+ }
+ }
+
+ const bool dualReceiverEnabledNotReceiving =
+ (_dualReceiver.State() != kReceiving &&
+ _dualReceiver.NackMode() == kNackInfinite);
+
+ VCMEncodedFrame* frame = _receiver.FrameForDecoding(
+ maxWaitTimeMs,
+ nextRenderTimeMs,
+ _codecDataBase.RenderTiming(),
+ &_dualReceiver);
+
+ if (dualReceiverEnabledNotReceiving && _dualReceiver.State() == kReceiving)
+ {
+ // Dual receiver is enabled (kNACK enabled), but was not receiving
+ // before the call to FrameForDecoding(). After the call the state
+ // changed to receiving, and therefore we must copy the primary decoder
+ // state to the dual decoder to make it possible for the dual decoder to
+ // start decoding retransmitted frames and recover.
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (_dualDecoder != NULL)
+ {
+ _codecDataBase.ReleaseDecoder(_dualDecoder);
+ }
+ _dualDecoder = _codecDataBase.CreateDecoderCopy();
+ if (_dualDecoder != NULL)
+ {
+ _dualDecoder->RegisterDecodeCompleteCallback(
+ &_dualDecodedFrameCallback);
+ }
+ else
+ {
+ _dualReceiver.Reset();
+ }
+ }
+
+ if (frame == NULL)
+ return VCM_FRAME_NOT_READY;
+ else
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+
+ // If this frame was too late, we should adjust the delay accordingly
+ _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
+ clock_->MillisecondTimestamp());
+
+#ifdef DEBUG_DECODER_BIT_STREAM
+ if (_bitStreamBeforeDecoder != NULL)
+ {
+ // Write bit stream to file for debugging purposes
+ fwrite(frame->Buffer(), 1, frame->Length(),
+ _bitStreamBeforeDecoder);
+ }
+#endif
+ if (_frameStorageCallback != NULL)
+ {
+ WebRtc_Word32 ret = frame->Store(*_frameStorageCallback);
+ if (ret < 0)
+ {
+ return ret;
+ }
+ }
+
+ const WebRtc_Word32 ret = Decode(*frame);
+ _receiver.ReleaseFrame(frame);
+ frame = NULL;
+ if (ret != VCM_OK)
+ {
+ return ret;
+ }
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::RequestSliceLossIndication(
+ const WebRtc_UWord64 pictureID) const
+{
+ if (_frameTypeCallback != NULL)
+ {
+ const WebRtc_Word32 ret =
+ _frameTypeCallback->SliceLossIndicationRequest(pictureID);
+ if (ret < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to request key frame");
+ return ret;
+ }
+ } else
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "No frame type request callback registered");
+ return VCM_MISSING_CALLBACK;
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::RequestKeyFrame()
+{
+ if (_frameTypeCallback != NULL)
+ {
+ const WebRtc_Word32 ret = _frameTypeCallback->RequestKeyFrame();
+ if (ret < 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to request key frame");
+ return ret;
+ }
+ _scheduleKeyRequest = false;
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "No frame type request callback registered");
+ return VCM_MISSING_CALLBACK;
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (_dualReceiver.State() != kReceiving ||
+ _dualReceiver.NackMode() != kNackInfinite)
+ {
+ // The dual receiver is currently not receiving or
+ // dual decoder mode is disabled.
+ return VCM_OK;
+ }
+ WebRtc_Word64 dummyRenderTime;
+ WebRtc_Word32 decodeCount = 0;
+ VCMEncodedFrame* dualFrame = _dualReceiver.FrameForDecoding(
+ maxWaitTimeMs,
+ dummyRenderTime);
+ if (dualFrame != NULL && _dualDecoder != NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Decoding frame %u with dual decoder",
+ dualFrame->TimeStamp());
+ // Decode dualFrame and try to catch up
+ WebRtc_Word32 ret = _dualDecoder->Decode(*dualFrame,
+ clock_->MillisecondTimestamp());
+ if (ret != WEBRTC_VIDEO_CODEC_OK)
+ {
+ WEBRTC_TRACE(webrtc::kTraceWarning,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to decode frame with dual decoder");
+ _dualReceiver.ReleaseFrame(dualFrame);
+ return VCM_CODEC_ERROR;
+ }
+ if (_receiver.DualDecoderCaughtUp(dualFrame, _dualReceiver))
+ {
+ // Copy the complete decoder state of the dual decoder
+ // to the primary decoder.
+ WEBRTC_TRACE(webrtc::kTraceStream,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Dual decoder caught up");
+ _codecDataBase.CopyDecoder(*_dualDecoder);
+ _codecDataBase.ReleaseDecoder(_dualDecoder);
+ _dualDecoder = NULL;
+ }
+ decodeCount++;
+ }
+ _dualReceiver.ReleaseFrame(dualFrame);
+ return decodeCount;
+}
+
+
+// Must be called from inside the receive side critical section.
+WebRtc_Word32
+VideoCodingModuleImpl::Decode(const VCMEncodedFrame& frame)
+{
+ // Change decoder if payload type has changed
+ const bool renderTimingBefore = _codecDataBase.RenderTiming();
+ _decoder = _codecDataBase.SetDecoder(frame.PayloadType(),
+ _decodedFrameCallback);
+ if (renderTimingBefore != _codecDataBase.RenderTiming())
+ {
+ // Make sure we reset the decode time estimate since it will
+ // be zero for codecs without render timing.
+ _timing.ResetDecodeTime();
+ }
+ if (_decoder == NULL)
+ {
+ return VCM_NO_CODEC_REGISTERED;
+ }
+ // Decode a frame
+ WebRtc_Word32 ret = _decoder->Decode(frame, clock_->MillisecondTimestamp());
+
+ // Check for failed decoding, run frame type request callback if needed.
+ if (ret < 0)
+ {
+ if (ret == VCM_ERROR_REQUEST_SLI)
+ {
+ return RequestSliceLossIndication(
+ _decodedFrameCallback.LastReceivedPictureID() + 1);
+ }
+ else
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to decode frame %u, requesting key frame",
+ frame.TimeStamp());
+ ret = RequestKeyFrame();
+ }
+ }
+ else if (ret == VCM_REQUEST_SLI)
+ {
+ ret = RequestSliceLossIndication(
+ _decodedFrameCallback.LastReceivedPictureID() + 1);
+ }
+ if (!frame.Complete() || frame.MissingFrame())
+ {
+ switch (_keyRequestMode)
+ {
+ case kKeyOnKeyLoss:
+ {
+ if (frame.FrameType() == kVideoFrameKey)
+ {
+ _scheduleKeyRequest = true;
+ return VCM_OK;
+ }
+ break;
+ }
+ case kKeyOnLoss:
+ {
+ _scheduleKeyRequest = true;
+ return VCM_OK;
+ }
+ default:
+ break;
+ }
+ }
+ return ret;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::DecodeFromStorage(
+ const EncodedVideoData& frameFromStorage)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ WebRtc_Word32 ret = _frameFromFile.ExtractFromStorage(frameFromStorage);
+ if (ret < 0)
+ {
+ return ret;
+ }
+ return Decode(_frameFromFile);
+}
+
+// Reset the decoder state
+WebRtc_Word32
+VideoCodingModuleImpl::ResetDecoder()
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (_decoder != NULL)
+ {
+ _receiver.Initialize();
+ _timing.Reset();
+ _scheduleKeyRequest = false;
+ _decoder->Reset();
+ }
+ if (_dualReceiver.State() != kPassive)
+ {
+ _dualReceiver.Initialize();
+ }
+ if (_dualDecoder != NULL)
+ {
+ _codecDataBase.ReleaseDecoder(_dualDecoder);
+ _dualDecoder = NULL;
+ }
+ return VCM_OK;
+}
+
+// Register possible receive codecs, can be called multiple times
+WebRtc_Word32
+VideoCodingModuleImpl::RegisterReceiveCodec(const VideoCodec* receiveCodec,
+ WebRtc_Word32 numberOfCores,
+ bool requireKeyFrame)
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (receiveCodec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ return _codecDataBase.RegisterReceiveCodec(receiveCodec, numberOfCores,
+ requireKeyFrame);
+}
+
+// Get current received codec
+WebRtc_Word32
+VideoCodingModuleImpl::ReceiveCodec(VideoCodec* currentReceiveCodec) const
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ if (currentReceiveCodec == NULL)
+ {
+ return VCM_PARAMETER_ERROR;
+ }
+ return _codecDataBase.ReceiveCodec(currentReceiveCodec);
+}
+
+// Get current received codec
+VideoCodecType
+VideoCodingModuleImpl::ReceiveCodec() const
+{
+ CriticalSectionScoped cs(_receiveCritSect);
+ return _codecDataBase.ReceiveCodec();
+}
+
+// Incoming packet from network parsed and ready for decode, non blocking.
+WebRtc_Word32
+VideoCodingModuleImpl::IncomingPacket(const WebRtc_UWord8* incomingPayload,
+ WebRtc_UWord32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo)
+{
+ const VCMPacket packet(incomingPayload, payloadLength, rtpInfo);
+ WebRtc_Word32 ret;
+ if (_dualReceiver.State() != kPassive)
+ {
+ ret = _dualReceiver.InsertPacket(packet,
+ rtpInfo.type.Video.width,
+ rtpInfo.type.Video.height);
+ if (ret == VCM_FLUSH_INDICATOR) {
+ RequestKeyFrame();
+ ResetDecoder();
+ } else if (ret < 0) {
+ return ret;
+ }
+ }
+ ret = _receiver.InsertPacket(packet, rtpInfo.type.Video.width,
+ rtpInfo.type.Video.height);
+ if (ret == VCM_FLUSH_INDICATOR) {
+ RequestKeyFrame();
+ ResetDecoder();
+ } else if (ret < 0) {
+ return ret;
+ }
+ return VCM_OK;
+}
+
+// Minimum playout delay (used for lip-sync). This is the minimum delay required
+// to sync with audio. Not included in VideoCodingModule::Delay()
+// Defaults to 0 ms.
+WebRtc_Word32
+VideoCodingModuleImpl::SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs)
+{
+ _timing.SetMinimumTotalDelay(minPlayoutDelayMs);
+ return VCM_OK;
+}
+
+// The estimated delay caused by rendering, defaults to
+// kDefaultRenderDelayMs = 10 ms
+WebRtc_Word32
+VideoCodingModuleImpl::SetRenderDelay(WebRtc_UWord32 timeMS)
+{
+ _timing.SetRenderDelay(timeMS);
+ return VCM_OK;
+}
+
+// Current video delay
+WebRtc_Word32
+VideoCodingModuleImpl::Delay() const
+{
+ return _timing.TargetVideoDelay();
+}
+
+// Nack list
+WebRtc_Word32
+VideoCodingModuleImpl::NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size)
+{
+ VCMNackStatus nackStatus = kNackOk;
+ // Collect sequence numbers from the default receiver
+ // if in normal nack mode. Otherwise collect them from
+ // the dual receiver if the dual receiver is receiving.
+ if (_receiver.NackMode() != kNoNack)
+ {
+ nackStatus = _receiver.NackList(nackList, size);
+ }
+ else if (_dualReceiver.State() != kPassive)
+ {
+ nackStatus = _dualReceiver.NackList(nackList, size);
+ }
+ else
+ {
+ size = 0;
+ }
+
+ switch (nackStatus)
+ {
+ case kNackNeedMoreMemory:
+ {
+ WEBRTC_TRACE(webrtc::kTraceError,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Out of memory");
+ return VCM_MEMORY;
+ }
+ case kNackKeyFrameRequest:
+ {
+ CriticalSectionScoped cs(_receiveCritSect);
+ WEBRTC_TRACE(webrtc::kTraceWarning,
+ webrtc::kTraceVideoCoding,
+ VCMId(_id),
+ "Failed to get NACK list, requesting key frame");
+ return RequestKeyFrame();
+ }
+ default:
+ break;
+ }
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VideoCodingModuleImpl::ReceivedFrameCount(VCMFrameCount& frameCount) const
+{
+ return _receiver.ReceivedFrameCount(frameCount);
+}
+
+WebRtc_UWord32 VideoCodingModuleImpl::DiscardedPackets() const {
+ return _receiver.DiscardedPackets();
+}
+
+int VideoCodingModuleImpl::SetSenderNackMode(SenderNackMode mode) {
+ CriticalSectionScoped cs(_sendCritSect);
+
+ switch (mode) {
+ case kNackNone:
+ _mediaOpt.EnableProtectionMethod(false, kNack);
+ break;
+ case kNackAll:
+ _mediaOpt.EnableProtectionMethod(true, kNack);
+ break;
+ case kNackSelective:
+ return VCM_NOT_IMPLEMENTED;
+ break;
+ }
+ return VCM_OK;
+}
+
+int VideoCodingModuleImpl::SetSenderReferenceSelection(bool enable) {
+ return VCM_NOT_IMPLEMENTED;
+}
+
+int VideoCodingModuleImpl::SetSenderFEC(bool enable) {
+ CriticalSectionScoped cs(_sendCritSect);
+ _mediaOpt.EnableProtectionMethod(enable, kFec);
+ return VCM_OK;
+}
+
+int VideoCodingModuleImpl::SetSenderKeyFramePeriod(int periodMs) {
+ return VCM_NOT_IMPLEMENTED;
+}
+
+int VideoCodingModuleImpl::SetReceiverRobustnessMode(
+ ReceiverRobustness robustnessMode,
+ DecodeErrors errorMode) {
+ CriticalSectionScoped cs(_receiveCritSect);
+ switch (robustnessMode) {
+ case kNone:
+ _receiver.SetNackMode(kNoNack);
+ _dualReceiver.SetNackMode(kNoNack);
+ if (errorMode == kNoDecodeErrors) {
+ _keyRequestMode = kKeyOnLoss;
+ } else {
+ _keyRequestMode = kKeyOnError;
+ }
+ break;
+ case kHardNack:
+ if (errorMode == kAllowDecodeErrors) {
+ return VCM_PARAMETER_ERROR;
+ }
+ _receiver.SetNackMode(kNackInfinite);
+ _dualReceiver.SetNackMode(kNoNack);
+ _keyRequestMode = kKeyOnError; // TODO(hlundin): On long NACK list?
+ break;
+ case kSoftNack:
+ assert(false); // TODO(hlundin): Not completed.
+ return VCM_NOT_IMPLEMENTED;
+ _receiver.SetNackMode(kNackHybrid);
+ _dualReceiver.SetNackMode(kNoNack);
+ _keyRequestMode = kKeyOnError;
+ break;
+ case kDualDecoder:
+ if (errorMode == kNoDecodeErrors) {
+ return VCM_PARAMETER_ERROR;
+ }
+ _receiver.SetNackMode(kNoNack);
+ _dualReceiver.SetNackMode(kNackInfinite);
+ _keyRequestMode = kKeyOnError;
+ break;
+ case kReferenceSelection:
+ assert(false); // TODO(hlundin): Not completed.
+ return VCM_NOT_IMPLEMENTED;
+ if (errorMode == kNoDecodeErrors) {
+ return VCM_PARAMETER_ERROR;
+ }
+ _receiver.SetNackMode(kNoNack);
+ _dualReceiver.SetNackMode(kNoNack);
+ break;
+ }
+ return VCM_OK;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_impl.h b/trunk/src/modules/video_coding/main/source/video_coding_impl.h
new file mode 100644
index 0000000..490c847
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/video_coding_impl.h
@@ -0,0 +1,311 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
+
+#include "video_coding.h"
+#include "critical_section_wrapper.h"
+#include "frame_buffer.h"
+#include "receiver.h"
+#include "timing.h"
+#include "jitter_buffer.h"
+#include "codec_database.h"
+#include "generic_decoder.h"
+#include "generic_encoder.h"
+#include "media_optimization.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+
+#include <stdio.h>
+
+namespace webrtc
+{
+
+class VCMProcessTimer
+{
+public:
+ VCMProcessTimer(WebRtc_UWord32 periodMs, TickTimeBase* clock)
+ : _clock(clock),
+ _periodMs(periodMs),
+ _latestMs(_clock->MillisecondTimestamp()) {}
+ WebRtc_UWord32 Period() const;
+ WebRtc_UWord32 TimeUntilProcess() const;
+ void Processed();
+
+private:
+ TickTimeBase* _clock;
+ WebRtc_UWord32 _periodMs;
+ WebRtc_Word64 _latestMs;
+};
+
+enum VCMKeyRequestMode
+{
+ kKeyOnError, // Normal mode, request key frames on decoder error
+ kKeyOnKeyLoss, // Request key frames on decoder error and on packet loss
+ // in key frames.
+ kKeyOnLoss, // Request key frames on decoder error and on packet loss
+ // in any frame
+};
+
+class VideoCodingModuleImpl : public VideoCodingModule
+{
+public:
+ VideoCodingModuleImpl(const WebRtc_Word32 id,
+ TickTimeBase* clock,
+ bool delete_clock_on_destroy);
+
+ virtual ~VideoCodingModuleImpl();
+
+ WebRtc_Word32 Id() const;
+
+ // Change the unique identifier of this object
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ // Returns the number of milliseconds until the module want a worker thread
+ // to call Process
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+
+ virtual WebRtc_Word32 Process();
+
+ /*
+ * Sender
+ */
+
+ // Initialize send codec
+ virtual WebRtc_Word32 InitializeSender();
+
+ // Register the send codec to be used.
+ virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
+ WebRtc_UWord32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize);
+
+ // Get current send codec
+ virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const;
+
+ // Get current send codec type
+ virtual VideoCodecType SendCodec() const;
+
+ // Register an external encoder object.
+ virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
+ WebRtc_UWord8 payloadType,
+ bool internalSource = false);
+
+ // Get codec config parameters
+ virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer,
+ WebRtc_Word32 size);
+
+ // Get encode bitrate
+ virtual int Bitrate(unsigned int* bitrate) const;
+
+ // Get encode frame rate
+ virtual int FrameRate(unsigned int* framerate) const;
+
+ // Set channel parameters
+ virtual WebRtc_Word32 SetChannelParameters(
+ WebRtc_UWord32 availableBandWidth,
+ WebRtc_UWord8 lossRate,
+ WebRtc_UWord32 rtt);
+
+ // Set recieve channel parameters
+ virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 rtt);
+
+ // Register a transport callback which will be called to deliver the
+ // encoded buffers
+ virtual WebRtc_Word32 RegisterTransportCallback(
+ VCMPacketizationCallback* transport);
+
+ // Register a send statistics callback which will be called to deliver
+ // information about the video stream produced by the encoder,
+ // for instance the average frame rate and bit rate.
+ virtual WebRtc_Word32 RegisterSendStatisticsCallback(
+ VCMSendStatisticsCallback* sendStats);
+
+ // Register a video quality settings callback which will be called when
+ // frame rate/dimensions need to be updated for video quality optimization
+ virtual WebRtc_Word32 RegisterVideoQMCallback(
+ VCMQMSettingsCallback* videoQMSettings);
+
+ // Register a video protection callback which will be called to deliver
+ // the requested FEC rate and NACK status (on/off).
+ virtual WebRtc_Word32 RegisterProtectionCallback(
+ VCMProtectionCallback* protection);
+
+ // Enable or disable a video protection method.
+ virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
+ bool enable);
+
+ // Add one raw video frame to the encoder, blocking.
+ virtual WebRtc_Word32 AddVideoFrame(
+ const VideoFrame& videoFrame,
+ const VideoContentMetrics* _contentMetrics = NULL,
+ const CodecSpecificInfo* codecSpecificInfo = NULL);
+
+ // Next frame encoded should be of the type frameType.
+ virtual WebRtc_Word32 FrameTypeRequest(FrameType frameType,
+ WebRtc_UWord8 simulcastIdx);
+
+ //Enable frame dropper
+ virtual WebRtc_Word32 EnableFrameDropper(bool enable);
+
+ // Sent frame counters
+ virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const;
+
+ /*
+ * Receiver
+ */
+
+ // Initialize receiver, resets codec database etc
+ virtual WebRtc_Word32 InitializeReceiver();
+
+ // Register possible reveive codecs, can be called multiple times
+ virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
+ WebRtc_Word32 numberOfCores,
+ bool requireKeyFrame = false);
+
+ // Register an externally defined decoder/render object.
+ // Can be a decoder only or a decoder coupled with a renderer.
+ virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
+ WebRtc_UWord8 payloadType,
+ bool internalRenderTiming);
+
+ // Register a receive callback. Will be called whenever there are a new
+ // frame ready for rendering.
+ virtual WebRtc_Word32 RegisterReceiveCallback(
+ VCMReceiveCallback* receiveCallback);
+
+ // Register a receive statistics callback which will be called to deliver
+ // information about the video stream received by the receiving side of the
+ // VCM, for instance the average frame rate and bit rate.
+ virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
+ VCMReceiveStatisticsCallback* receiveStats);
+
+ // Register a frame type request callback.
+ virtual WebRtc_Word32 RegisterFrameTypeCallback(
+ VCMFrameTypeCallback* frameTypeCallback);
+
+ // Register a frame storage callback.
+ virtual WebRtc_Word32 RegisterFrameStorageCallback(
+ VCMFrameStorageCallback* frameStorageCallback);
+
+ // Nack callback
+ virtual WebRtc_Word32 RegisterPacketRequestCallback(
+ VCMPacketRequestCallback* callback);
+
+ // Decode next frame, blocks for a maximum of maxWaitTimeMs milliseconds.
+ // Should be called as often as possible to get the most out of the decoder.
+ virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200);
+
+ // Decode next dual frame, blocks for a maximum of maxWaitTimeMs
+ // milliseconds.
+ virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200);
+
+ // Reset the decoder state
+ virtual WebRtc_Word32 ResetDecoder();
+
+ // Get current received codec
+ virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const;
+
+ // Get current received codec type
+ virtual VideoCodecType ReceiveCodec() const;
+
+ // Incoming packet from network parsed and ready for decode, non blocking.
+ virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
+ WebRtc_UWord32 payloadLength,
+ const WebRtcRTPHeader& rtpInfo);
+
+ // A part of an encoded frame to be decoded.
+ // Used in conjunction with VCMFrameStorageCallback.
+ virtual WebRtc_Word32 DecodeFromStorage(
+ const EncodedVideoData& frameFromStorage);
+
+ // Minimum playout delay (Used for lip-sync). This is the minimum delay
+ // required to sync with audio. Not included in VideoCodingModule::Delay()
+ // Defaults to 0 ms.
+ virtual WebRtc_Word32 SetMinimumPlayoutDelay(
+ WebRtc_UWord32 minPlayoutDelayMs);
+
+ // The estimated delay caused by rendering
+ virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS);
+
+ // Current delay
+ virtual WebRtc_Word32 Delay() const;
+
+ // Received frame counters
+ virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
+
+ // Returns the number of packets discarded by the jitter buffer.
+ virtual WebRtc_UWord32 DiscardedPackets() const;
+
+
+ // Robustness APIs
+
+ // Set the sender RTX/NACK mode.
+ virtual int SetSenderNackMode(SenderNackMode mode);
+
+ // Set the sender reference picture selection (RPS) mode.
+ virtual int SetSenderReferenceSelection(bool enable);
+
+ // Set the sender forward error correction (FEC) mode.
+ virtual int SetSenderFEC(bool enable);
+
+ // Set the key frame period, or disable periodic key frames (I-frames).
+ virtual int SetSenderKeyFramePeriod(int periodMs);
+
+ // Set the receiver robustness mode.
+ virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
+ DecodeErrors errorMode);
+
+protected:
+ WebRtc_Word32 Decode(const webrtc::VCMEncodedFrame& frame);
+ WebRtc_Word32 RequestKeyFrame();
+ WebRtc_Word32 RequestSliceLossIndication(
+ const WebRtc_UWord64 pictureID) const;
+ WebRtc_Word32 NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
+
+private:
+ WebRtc_Word32 _id;
+ TickTimeBase* clock_;
+ bool delete_clock_on_destroy_;
+ CriticalSectionWrapper* _receiveCritSect;
+ bool _receiverInited;
+ VCMTiming _timing;
+ VCMTiming _dualTiming;
+ VCMReceiver _receiver;
+ VCMReceiver _dualReceiver;
+ VCMDecodedFrameCallback _decodedFrameCallback;
+ VCMDecodedFrameCallback _dualDecodedFrameCallback;
+ VCMFrameTypeCallback* _frameTypeCallback;
+ VCMFrameStorageCallback* _frameStorageCallback;
+ VCMReceiveStatisticsCallback* _receiveStatsCallback;
+ VCMPacketRequestCallback* _packetRequestCallback;
+ VCMGenericDecoder* _decoder;
+ VCMGenericDecoder* _dualDecoder;
+ FILE* _bitStreamBeforeDecoder;
+ VCMFrameBuffer _frameFromFile;
+ VCMKeyRequestMode _keyRequestMode;
+ bool _scheduleKeyRequest;
+
+ CriticalSectionWrapper* _sendCritSect; // Critical section for send side
+ VCMGenericEncoder* _encoder;
+ VCMEncodedFrameCallback _encodedFrameCallback;
+ FrameType _nextFrameType[kMaxSimulcastStreams];
+ VCMMediaOptimization _mediaOpt;
+ VideoCodecType _sendCodecType;
+ VCMSendStatisticsCallback* _sendStatsCallback;
+ FILE* _encoderInputFile;
+
+ VCMCodecDataBase _codecDataBase;
+ VCMProcessTimer _receiveStatsTimer;
+ VCMProcessTimer _sendStatsTimer;
+ VCMProcessTimer _retransmissionTimer;
+ VCMProcessTimer _keyRequestTimer;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc
new file mode 100644
index 0000000..0ee9657
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc
@@ -0,0 +1,396 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+
+namespace webrtc {
+
+using ::testing::Return;
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::AllOf;
+using ::testing::Args;
+using ::testing::Field;
+using ::testing::Pointee;
+using ::testing::NiceMock;
+using ::testing::Sequence;
+
+class VCMRobustnessTest : public ::testing::Test {
+ protected:
+ static const size_t kPayloadLen = 10;
+
+ virtual void SetUp() {
+ clock_ = new FakeTickTime(0);
+ ASSERT_TRUE(clock_ != NULL);
+ vcm_ = VideoCodingModule::Create(0, clock_);
+ ASSERT_TRUE(vcm_ != NULL);
+ ASSERT_EQ(0, vcm_->InitializeReceiver());
+ ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
+ ASSERT_EQ(0, vcm_->RegisterPacketRequestCallback(&request_callback_));
+ ASSERT_EQ(VCM_OK, vcm_->Codec(kVideoCodecVP8, &video_codec_));
+ ASSERT_EQ(VCM_OK, vcm_->RegisterReceiveCodec(&video_codec_, 1));
+ ASSERT_EQ(VCM_OK, vcm_->RegisterExternalDecoder(&decoder_,
+ video_codec_.plType,
+ true));
+ }
+
+ virtual void TearDown() {
+ VideoCodingModule::Destroy(vcm_);
+ delete clock_;
+ }
+
+ void InsertPacket(uint32_t timestamp,
+ uint16_t seq_no,
+ bool first,
+ bool marker_bit,
+ FrameType frame_type) {
+ const uint8_t payload[kPayloadLen] = {0};
+ WebRtcRTPHeader rtp_info;
+ memset(&rtp_info, 0, sizeof(rtp_info));
+ rtp_info.frameType = frame_type;
+ rtp_info.header.timestamp = timestamp;
+ rtp_info.header.sequenceNumber = seq_no;
+ rtp_info.header.markerBit = marker_bit;
+ rtp_info.header.payloadType = video_codec_.plType;
+ rtp_info.type.Video.codec = kRTPVideoVP8;
+ rtp_info.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
+ rtp_info.type.Video.isFirstPacket = first;
+
+ ASSERT_EQ(VCM_OK, vcm_->IncomingPacket(payload, kPayloadLen, rtp_info));
+ }
+
+ VideoCodingModule* vcm_;
+ VideoCodec video_codec_;
+ MockVCMFrameTypeCallback frame_type_callback_;
+ MockPacketRequestCallback request_callback_;
+ NiceMock<MockVideoDecoder> decoder_;
+ NiceMock<MockVideoDecoder> decoderCopy_;
+ FakeTickTime* clock_;
+};
+
+TEST_F(VCMRobustnessTest, TestHardNack) {
+ Sequence s;
+ EXPECT_CALL(request_callback_, ResendPackets(_, 2))
+ .With(Args<0, 1>(ElementsAre(6, 7)))
+ .Times(1);
+ for (int ts = 0; ts <= 6000; ts += 3000) {
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
+ Field(&EncodedImage::_length,
+ kPayloadLen * 3),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s);
+ }
+
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
+ VideoCodingModule::kHardNack,
+ VideoCodingModule::kNoDecodeErrors));
+
+ InsertPacket(0, 0, true, false, kVideoFrameKey);
+ InsertPacket(0, 1, false, false, kVideoFrameKey);
+ InsertPacket(0, 2, false, true, kVideoFrameKey);
+
+ InsertPacket(3000, 3, true, false, kVideoFrameDelta);
+ InsertPacket(3000, 4, false, false, kVideoFrameDelta);
+ InsertPacket(3000, 5, false, true, kVideoFrameDelta);
+
+ ASSERT_EQ(VCM_OK, vcm_->Decode(0));
+ ASSERT_EQ(VCM_OK, vcm_->Decode(0));
+ ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+
+ clock_->IncrementDebugClock(10);
+
+ ASSERT_EQ(VCM_OK, vcm_->Process());
+
+ ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+
+ InsertPacket(6000, 8, false, true, kVideoFrameDelta);
+ clock_->IncrementDebugClock(10);
+ ASSERT_EQ(VCM_OK, vcm_->Process());
+
+ ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+
+ InsertPacket(6000, 6, true, false, kVideoFrameDelta);
+ InsertPacket(6000, 7, false, false, kVideoFrameDelta);
+ clock_->IncrementDebugClock(10);
+ ASSERT_EQ(VCM_OK, vcm_->Process());
+
+ ASSERT_EQ(VCM_OK, vcm_->Decode(0));
+}
+
+TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
+ EXPECT_CALL(request_callback_, ResendPackets(_, _))
+ .Times(0);
+ EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
+ .Times(1);
+
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
+ VideoCodingModule::kHardNack,
+ VideoCodingModule::kNoDecodeErrors));
+
+ InsertPacket(3000, 3, true, false, kVideoFrameDelta);
+ InsertPacket(3000, 4, false, false, kVideoFrameDelta);
+ InsertPacket(3000, 5, false, true, kVideoFrameDelta);
+
+ EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+ ASSERT_EQ(VCM_OK, vcm_->Process());
+
+ clock_->IncrementDebugClock(10);
+
+ EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+ ASSERT_EQ(VCM_OK, vcm_->Process());
+}
+
+TEST_F(VCMRobustnessTest, TestDualDecoder) {
+ Sequence s1, s2;
+ EXPECT_CALL(request_callback_, ResendPackets(_, 1))
+ .With(Args<0, 1>(ElementsAre(4)))
+ .Times(1);
+
+ EXPECT_CALL(decoder_, Copy())
+ .Times(1)
+ .WillOnce(Return(&decoderCopy_));
+ EXPECT_CALL(decoderCopy_, Copy())
+ .Times(1)
+ .WillOnce(Return(&decoder_));
+
+ // Decode operations
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+ Field(&EncodedImage::_completeFrame,
+ false)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+
+ EXPECT_CALL(decoderCopy_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s2);
+ EXPECT_CALL(decoderCopy_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s2);
+
+
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
+ VideoCodingModule::kDualDecoder,
+ VideoCodingModule::kAllowDecodeErrors));
+
+ InsertPacket(0, 0, true, false, kVideoFrameKey);
+ InsertPacket(0, 1, false, false, kVideoFrameKey);
+ InsertPacket(0, 2, false, true, kVideoFrameKey);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(3000, 3, true, false, kVideoFrameDelta);
+ // Packet 4 missing
+ InsertPacket(3000, 5, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(6000, 6, true, false, kVideoFrameDelta);
+ InsertPacket(6000, 7, false, false, kVideoFrameDelta);
+ InsertPacket(6000, 8, false, true, kVideoFrameDelta);
+
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
+ // Spawn a decoder copy.
+ EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
+
+ clock_->IncrementDebugClock(10);
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Generate NACK list.
+
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
+ EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
+
+ InsertPacket(3000, 4, false, false, kVideoFrameDelta);
+ EXPECT_EQ(1, vcm_->DecodeDualFrame(0)); // Dual decode of timestamp 3000.
+ EXPECT_EQ(1, vcm_->DecodeDualFrame(0)); // Dual decode of timestamp 6000.
+ EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // No more frames.
+
+ InsertPacket(9000, 9, true, false, kVideoFrameDelta);
+ InsertPacket(9000, 10, false, false, kVideoFrameDelta);
+ InsertPacket(9000, 11, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 9000 complete.
+ EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
+}
+
+TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
+ EXPECT_CALL(decoder_, InitDecode(_, _)).Times(1);
+ EXPECT_CALL(decoder_, Release()).Times(1);
+ Sequence s1;
+ EXPECT_CALL(request_callback_, ResendPackets(_, 1))
+ .With(Args<0, 1>(ElementsAre(4)))
+ .Times(0);
+
+ EXPECT_CALL(decoder_, Copy())
+ .Times(0);
+ EXPECT_CALL(decoderCopy_, Copy())
+ .Times(0);
+
+ // Decode operations
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+ Field(&EncodedImage::_completeFrame,
+ false)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
+ VideoCodingModule::kNone,
+ VideoCodingModule::kAllowDecodeErrors));
+
+ InsertPacket(0, 0, true, false, kVideoFrameKey);
+ InsertPacket(0, 1, false, false, kVideoFrameKey);
+ InsertPacket(0, 2, false, true, kVideoFrameKey);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(3000, 3, true, false, kVideoFrameDelta);
+ // Packet 4 missing
+ InsertPacket(3000, 5, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(6000, 6, true, false, kVideoFrameDelta);
+ InsertPacket(6000, 7, false, false, kVideoFrameDelta);
+ InsertPacket(6000, 8, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(10);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(23);
+ InsertPacket(3000, 4, false, false, kVideoFrameDelta);
+
+ InsertPacket(9000, 9, true, false, kVideoFrameDelta);
+ InsertPacket(9000, 10, false, false, kVideoFrameDelta);
+ InsertPacket(9000, 11, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 9000 complete.
+}
+
+TEST_F(VCMRobustnessTest, TestModeNoneWithoutErrors) {
+ Sequence s1;
+ EXPECT_CALL(decoder_, InitDecode(_, _)).Times(1);
+ EXPECT_CALL(decoder_, Release()).Times(1);
+ EXPECT_CALL(request_callback_, ResendPackets(_, 1))
+ .With(Args<0, 1>(ElementsAre(4)))
+ .Times(0);
+
+ EXPECT_CALL(decoder_, Copy())
+ .Times(0);
+ EXPECT_CALL(decoderCopy_, Copy())
+ .Times(0);
+
+ // Decode operations
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+ Field(&EncodedImage::_completeFrame,
+ false)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+ Field(&EncodedImage::_completeFrame,
+ true)),
+ false, _, _, _))
+ .Times(1)
+ .InSequence(s1);
+ EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
+ .Times(1);
+
+ ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
+ VideoCodingModule::kNone,
+ VideoCodingModule::kNoDecodeErrors));
+
+ InsertPacket(0, 0, true, false, kVideoFrameKey);
+ InsertPacket(0, 1, false, false, kVideoFrameKey);
+ InsertPacket(0, 2, false, true, kVideoFrameKey);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(3000, 3, true, false, kVideoFrameDelta);
+ // Packet 4 missing
+ InsertPacket(3000, 5, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(33);
+ InsertPacket(6000, 6, true, false, kVideoFrameDelta);
+ InsertPacket(6000, 7, false, false, kVideoFrameDelta);
+ InsertPacket(6000, 8, false, true, kVideoFrameDelta);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
+ // Schedule key frame request.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(10);
+ EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
+
+ clock_->IncrementDebugClock(500); // Wait for the key request timer to set.
+ EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect key frame request.
+}
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_test.gypi b/trunk/src/modules/video_coding/main/source/video_coding_test.gypi
new file mode 100644
index 0000000..7ab265c
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/source/video_coding_test.gypi
@@ -0,0 +1,96 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [{
+ 'target_name': 'video_coding_test',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../test/metrics.gyp:metrics',
+ 'webrtc_video_coding',
+ 'rtp_rtcp',
+ 'webrtc_utility',
+ 'video_processing',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+ 'include_dirs': [
+ '../../../interface',
+ '../../codecs/vp8/main/interface',
+ '../../../../system_wrappers/interface',
+ '../../../../common_video/interface',
+ '../source',
+ ],
+ 'sources': [
+ # headers
+ '../test/codec_database_test.h',
+ '../test/generic_codec_test.h',
+ '../test/jitter_estimate_test.h',
+ '../test/media_opt_test.h',
+ '../test/mt_test_common.h',
+ '../test/normal_test.h',
+ '../test/quality_modes_test.h',
+ '../test/receiver_tests.h',
+ '../test/release_test.h',
+ '../test/rtp_player.h',
+ '../test/test_callbacks.h',
+ '../test/test_util.h',
+ '../test/video_source.h',
+
+ # sources
+ '../test/codec_database_test.cc',
+ '../test/decode_from_storage_test.cc',
+ '../test/generic_codec_test.cc',
+ '../test/jitter_buffer_test.cc',
+ '../test/media_opt_test.cc',
+ '../test/mt_test_common.cc',
+ '../test/mt_rx_tx_test.cc',
+ '../test/normal_test.cc',
+ '../test/quality_modes_test.cc',
+ '../test/receiver_timing_tests.cc',
+ '../test/rtp_player.cc',
+ '../test/test_callbacks.cc',
+ '../test/test_util.cc',
+ '../test/tester_main.cc',
+ '../test/video_rtp_play_mt.cc',
+ '../test/video_rtp_play.cc',
+ '../test/video_source.cc',
+ ], # source
+ },
+ {
+ 'target_name': 'video_coding_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'webrtc_video_coding',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../../../interface',
+ '../../codecs/interface',
+ ],
+ 'sources': [
+ '../interface/mock/mock_vcm_callbacks.h',
+ 'decoding_state_unittest.cc',
+ 'jitter_buffer_unittest.cc',
+ 'session_info_unittest.cc',
+ 'video_coding_robustness_unittest.cc',
+ 'qm_select_unittest.cc',
+ ],
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/main/test/codec_database_test.cc b/trunk/src/modules/video_coding/main/test/codec_database_test.cc
new file mode 100644
index 0000000..f9ea3e5
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/codec_database_test.cc
@@ -0,0 +1,408 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Implementation of codec data base test
+// testing is done via the VCM module, no specific CodecDataBase module functionality.
+
+#include "codec_database_test.h"
+
+#include <assert.h>
+#include <stdio.h>
+
+#include "../../../../engine_configurations.h"
+#include "../source/event.h"
+#include "test_callbacks.h"
+#include "test_macros.h"
+#include "test_util.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "vp8.h" // for external codecs test
+
+
+using namespace webrtc;
+
+int CodecDataBaseTest::RunTest(CmdArgs& args)
+{
+ VideoCodingModule* vcm = VideoCodingModule::Create(1);
+ CodecDataBaseTest* cdbt = new CodecDataBaseTest(vcm);
+ cdbt->Perform(args);
+ VideoCodingModule::Destroy(vcm);
+ delete cdbt;
+ return 0;
+
+}
+
+CodecDataBaseTest::CodecDataBaseTest(VideoCodingModule* vcm):
+_vcm(vcm),
+_width(0),
+_height(0),
+_lengthSourceFrame(0),
+_timeStamp(0)
+{
+ //
+}
+CodecDataBaseTest::~CodecDataBaseTest()
+{
+ //
+}
+void
+CodecDataBaseTest::Setup(CmdArgs& args)
+{
+ _inname= args.inputFile;
+ _width = args.width;
+ _height = args.height;
+ _frameRate = args.frameRate;
+ _lengthSourceFrame = 3*_width*_height/2;
+ if (args.outputFile.compare(""))
+ _outname = test::OutputPath() + "CDBtest_decoded.yuv";
+ else
+ _outname = args.outputFile;
+ _outname = args.outputFile;
+ _encodedName = test::OutputPath() + "CDBtest_encoded.vp8";
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+
+ if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ return;
+}
+
+
+
+WebRtc_Word32
+CodecDataBaseTest::Perform(CmdArgs& args)
+{
+#ifndef VIDEOCODEC_VP8
+ assert(false);
+#endif
+ Setup(args);
+ EventWrapper* waitEvent = EventWrapper::Create();
+
+ /**************************/
+ /* General Sanity Checks */
+ /************************/
+ VideoCodec sendCodec, receiveCodec;
+ TEST(VideoCodingModule::NumberOfCodecs() > 0);
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ VCMDecodeCompleteCallback *_decodeCallback = new VCMDecodeCompleteCallback(_decodedFile);
+ VCMEncodeCompleteCallback *_encodeCompleteCallback = new VCMEncodeCompleteCallback(_encodedFile);
+ _vcm->RegisterReceiveCallback(_decodeCallback);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ _encodeCompleteCallback->SetFrameDimensions(_width, _height);
+ // registering the callback - encode and decode with the same vcm (could be later changed)
+ _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
+ // preparing a frame to be encoded
+ VideoFrame sourceFrame;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ // Encoder registration
+ TEST (VideoCodingModule::NumberOfCodecs() > 0);
+ TEST(VideoCodingModule::Codec(-1, &sendCodec) == VCM_PARAMETER_ERROR);
+ TEST(VideoCodingModule::Codec(VideoCodingModule::NumberOfCodecs() + 1, &sendCodec) == VCM_PARAMETER_ERROR);
+ VideoCodingModule::Codec(1, &sendCodec);
+ sendCodec.plType = 0; // random value
+ TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0);
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ _vcm->RegisterReceiveCallback(_decodeCallback);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ printf(" \nNumber of Registered Codecs: %d \n\n", VideoCodingModule::NumberOfCodecs());
+ printf("Registered codec names: ");
+ for (int i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
+ {
+ VideoCodingModule::Codec(i, &sendCodec);
+ printf("%s ", sendCodec.plName);
+ }
+ printf("\n\nVerify that all requested codecs are used\n \n \n");
+
+ // Testing with VP8.
+ VideoCodingModule::Codec(kVideoCodecVP8, &sendCodec);
+ _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
+ _encodeCompleteCallback->SetCodecType(kRTPVideoVP8);
+ _vcm->InitializeReceiver();
+ TEST (_vcm->AddVideoFrame(sourceFrame) == VCM_OK );
+ _vcm->InitializeSender();
+ TEST (_vcm->AddVideoFrame(sourceFrame) < 0 );
+
+ // Test changing frame size while keeping the same payload type
+ VideoCodingModule::Codec(0, &sendCodec);
+ sendCodec.width = 352;
+ sendCodec.height = 288;
+ VideoCodec currentSendCodec;
+ _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
+ _vcm->SendCodec(¤tSendCodec);
+ TEST(currentSendCodec.width == sendCodec.width &&
+ currentSendCodec.height == sendCodec.height);
+ sendCodec.width = 352/2;
+ sendCodec.height = 288/2;
+ _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
+ _vcm->SendCodec(¤tSendCodec);
+ TEST(currentSendCodec.width == sendCodec.width &&
+ currentSendCodec.height == sendCodec.height);
+
+ delete _decodeCallback;
+ _decodeCallback = NULL;
+ delete _encodeCompleteCallback;
+ _encodeCompleteCallback = NULL;
+
+ VCMEncodeCompleteCallback *_encodeCallback = new VCMEncodeCompleteCallback(_encodedFile);
+
+ /*************************/
+ /* External codecs */
+ /*************************/
+
+
+ _vcm->InitializeReceiver();
+ VP8Decoder* decoder = VP8Decoder::Create();
+ VideoCodec vp8DecSettings;
+ VideoCodingModule::Codec(kVideoCodecVP8, &vp8DecSettings);
+ TEST(_vcm->RegisterExternalDecoder(decoder, vp8DecSettings.plType, false) == VCM_OK);
+ TEST(_vcm->RegisterReceiveCodec(&vp8DecSettings, 1, false) == VCM_OK);
+ VP8Encoder* encoder = VP8Encoder::Create();
+ VideoCodec vp8EncSettings;
+ VideoCodingModule::Codec(kVideoCodecVP8, &vp8EncSettings);
+ _vcm->RegisterTransportCallback(_encodeCallback); // encode returns error if callback uninitialized
+ _encodeCallback->RegisterReceiverVCM(_vcm);
+ _encodeCallback->SetCodecType(kRTPVideoVP8);
+ TEST(_vcm->RegisterExternalEncoder(encoder, vp8EncSettings.plType) == VCM_OK);
+ TEST(_vcm->RegisterSendCodec(&vp8EncSettings, 4, 1440) == VCM_OK);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+ waitEvent->Wait(33);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+
+ // De-register and try again.
+ TEST(_vcm->RegisterExternalDecoder(NULL, vp8DecSettings.plType, false) == VCM_OK);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() < 0); // Expect an error since we have de-registered the decoder
+ TEST(_vcm->RegisterExternalEncoder(NULL, vp8DecSettings.plType) == VCM_OK);
+ TEST(_vcm->AddVideoFrame(sourceFrame) < 0); // No send codec registered
+
+ delete decoder;
+ decoder = NULL;
+ delete encoder;
+ encoder = NULL;
+
+ /***************************************
+ * Test the "require key frame" setting*
+ ***************************************/
+
+ TEST(_vcm->InitializeSender() == VCM_OK);
+ TEST(_vcm->InitializeReceiver() == VCM_OK);
+ VideoCodingModule::Codec(kVideoCodecVP8, &receiveCodec);
+ receiveCodec.height = _height;
+ receiveCodec.width = _width;
+ TEST(_vcm->RegisterSendCodec(&receiveCodec, 4, 1440) == VCM_OK);
+ TEST(_vcm->RegisterReceiveCodec(&receiveCodec, 1, true) == VCM_OK); // Require key frame
+ _vcm->RegisterTransportCallback(_encodeCallback); // encode returns error if callback uninitialized
+ _encodeCallback->RegisterReceiverVCM(_vcm);
+ _encodeCallback->SetCodecType(kRTPVideoVP8);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+ TEST(_vcm->ResetDecoder() == VCM_OK);
+ waitEvent->Wait(33);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ // Try to decode a delta frame. Should get a warning since we have enabled the "require key frame" setting
+ // and because no frame type request callback has been registered.
+ TEST(_vcm->Decode() == VCM_MISSING_CALLBACK);
+ TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+
+ // Make sure we can register another codec with the same
+ // payload type without crash.
+ _vcm->InitializeReceiver();
+ sendCodec.width = _width;
+ sendCodec.height = _height;
+ TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
+ TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
+ waitEvent->Wait(33);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+ TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
+ waitEvent->Wait(33);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+ TEST(_vcm->ResetDecoder() == VCM_OK);
+
+ delete _encodeCallback;
+
+ /*************************/
+ /* Send/Receive Control */
+ /***********************/
+ /*
+ 1. check available codecs (N)
+ 2. register all corresponding decoders
+ 3. encode 300/N frames with each encoder, and hope to properly decode
+ 4. encode without a matching decoder - expect an error
+ */
+ rewind(_sourceFile);
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ sourceFrame.Free();
+ VCMDecodeCompleteCallback* decodeCallCDT = new VCMDecodeCompleteCallback(_decodedFile);
+ VCMEncodeCompleteCallback* encodeCallCDT = new VCMEncodeCompleteCallback(_encodedFile);
+ _vcm->RegisterReceiveCallback(decodeCallCDT);
+ _vcm->RegisterTransportCallback(encodeCallCDT);
+ encodeCallCDT->RegisterReceiverVCM(_vcm);
+ if (VideoCodingModule::NumberOfCodecs() > 0)
+ {
+ // Register all available decoders.
+ int i, j;
+ //double psnr;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ _vcm->RegisterReceiveCallback(decodeCallCDT);
+ for (i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
+ {
+ VideoCodingModule::Codec(i, &receiveCodec);
+ if (strcmp(receiveCodec.plName, "I420") == 0)
+ {
+ receiveCodec.height = _height;
+ receiveCodec.width = _width;
+ }
+ _vcm->RegisterReceiveCodec(&receiveCodec, 1);
+ }
+ // start encoding - iterating over available encoders
+ _vcm->RegisterTransportCallback(encodeCallCDT);
+ encodeCallCDT->RegisterReceiverVCM(_vcm);
+ encodeCallCDT->Initialize();
+ int frameCnt = 0;
+ for (i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
+ {
+ encodeCallCDT->ResetByteCount();
+ VideoCodingModule::Codec(i, &sendCodec);
+ sendCodec.height = _height;
+ sendCodec.width = _width;
+ sendCodec.startBitrate = 1000;
+ sendCodec.maxBitrate = 8000;
+ encodeCallCDT->SetFrameDimensions(_width, _height);
+ encodeCallCDT->SetCodecType(ConvertCodecType(sendCodec.plName));
+ TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) == VCM_OK);
+
+ // We disable the frame dropper to avoid dropping frames due to
+ // bad rate control. This isn't a codec performance test, and the
+ // I420 codec is expected to produce too many bits.
+ _vcm->EnableFrameDropper(false);
+
+ printf("Encoding with %s \n\n", sendCodec.plName);
+ for (j=0; j < int(300/VideoCodingModule::NumberOfCodecs()); j++)// assuming 300 frames, NumberOfCodecs <= 10
+ {
+ frameCnt++;
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ // building source frame
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ sourceFrame.SetLength(_lengthSourceFrame);
+ _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
+ sourceFrame.SetTimeStamp(_timeStamp);
+ // send frame to the encoder
+ TEST (_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ waitEvent->Wait(33); // was 100
+
+ int ret =_vcm->Decode();
+ TEST(ret == 0);
+ if (ret < 0)
+ {
+ printf("Error #%d in frame number %d \n",ret, frameCnt);
+ }
+ // verifying matching payload types:
+ _vcm->SendCodec(&sendCodec);
+ _vcm->ReceiveCodec(&receiveCodec);
+ TEST(sendCodec.plType == receiveCodec.plType);
+ if (sendCodec.plType != receiveCodec.plType)
+ {
+ printf("frame number:%d\n",frameCnt);
+ }
+ } // end for:encode-decode
+ // byte count for codec specific
+
+ printf("Total bytes encoded: %f \n\n",(8.0/1000)*(encodeCallCDT->EncodedBytes()/((int)10/VideoCodingModule::NumberOfCodecs())));
+ // decode what's left in the buffer....
+ _vcm->Decode();
+ _vcm->Decode();
+ // Don't measure PSNR for I420 since it will be perfect.
+ if (sendCodec.codecType != kVideoCodecI420) {
+ webrtc::test::QualityMetricsResult psnr;
+ I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _width,
+ _height, &psnr);
+ printf("\n @ %d KBPS: ", sendCodec.startBitrate);
+ printf("PSNR from encoder-decoder send-receive control test"
+ "is %f\n\n", psnr.average);
+ }
+ } // end: iterate codecs
+ rewind(_sourceFile);
+ sourceFrame.Free();
+ delete [] tmpBuffer;
+ delete decodeCallCDT;
+ delete encodeCallCDT;
+ // closing and calculating PSNR for prior encoder-decoder test
+ TearDown(); // closing open files
+ } // end of #codecs >1
+
+ delete waitEvent;
+ Print();
+ return 0;
+}
+void
+CodecDataBaseTest::Print()
+{
+ printf("\nVCM Codec DataBase Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+
+void
+CodecDataBaseTest::TearDown()
+{
+ fclose(_sourceFile);
+ fclose(_decodedFile);
+ fclose(_encodedFile);
+ return;
+}
diff --git a/trunk/src/modules/video_coding/main/test/codec_database_test.h b/trunk/src/modules/video_coding/main/test/codec_database_test.h
new file mode 100644
index 0000000..cc33e05
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/codec_database_test.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
+
+#include "video_coding.h"
+#include "test_util.h"
+
+#include <string.h>
+
+/*
+Test consists of:
+1. Sanity chacks: Send and Receive side (bad input, etc. )
+2. Send-side control (encoder registration etc.)
+3. Decoder-side control - encode with various encoders, and verify correct decoding
+*/
+
+class CodecDataBaseTest
+{
+public:
+ CodecDataBaseTest(webrtc::VideoCodingModule* vcm);
+ ~CodecDataBaseTest();
+ static int RunTest(CmdArgs& args);
+ WebRtc_Word32 Perform(CmdArgs& args);
+private:
+ void TearDown();
+ void Setup(CmdArgs& args);
+ void Print();
+ webrtc::VideoCodingModule* _vcm;
+ std::string _inname;
+ std::string _outname;
+ std::string _encodedName;
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ FILE* _encodedFile;
+ WebRtc_UWord16 _width;
+ WebRtc_UWord16 _height;
+ WebRtc_UWord32 _lengthSourceFrame;
+ WebRtc_UWord32 _timeStamp;
+ float _frameRate;
+}; // end of codecDBTest class definition
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc b/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc
new file mode 100644
index 0000000..628d509
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "receiver_tests.h"
+#include "video_coding.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+#include "../source/event.h"
+#include "rtp_player.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+
+using namespace webrtc;
+
+class FrameStorageCallback : public VCMFrameStorageCallback
+{
+public:
+ FrameStorageCallback(VideoCodingModule* vcm) : _vcm(vcm) {}
+
+ WebRtc_Word32 StoreReceivedFrame(const EncodedVideoData& frameToStore)
+ {
+ _vcm->DecodeFromStorage(frameToStore);
+ return VCM_OK;
+ }
+
+private:
+ VideoCodingModule* _vcm;
+};
+
+int DecodeFromStorageTest(CmdArgs& args)
+{
+ // Make sure this test isn't executed without simulated events.
+#if !defined(EVENT_DEBUG)
+ return -1;
+#endif
+ // BEGIN Settings
+
+ bool protectionEnabled = false;
+ VCMVideoProtection protectionMethod = kProtectionNack;
+ WebRtc_UWord32 rttMS = 100;
+ float lossRate = 0.00f;
+ bool reordering = false;
+ WebRtc_UWord32 renderDelayMs = 0;
+ WebRtc_UWord32 minPlayoutDelayMs = 0;
+ const WebRtc_Word64 MAX_RUNTIME_MS = -1;
+ std::string rtpFilename = args.inputFile;
+ std::string outFilename = args.outputFile;
+ if (outFilename == "")
+ outFilename = test::OutputPath() + "DecodeFromStorage.yuv";
+
+ FrameReceiveCallback receiveCallback(outFilename.c_str());
+
+ // END Settings
+
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (test::OutputPath() + "decodeFromStorageTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+
+ FakeTickTime clock(0);
+ // TODO(hlundin): This test was not verified after changing to FakeTickTime.
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ VideoCodingModule* vcmPlayback = VideoCodingModule::Create(2, &clock);
+ FrameStorageCallback storageCallback(vcmPlayback);
+ RtpDataCallback dataCallback(vcm);
+ WebRtc_Word32 ret = vcm->InitializeReceiver();
+ if (ret < 0)
+ {
+ return -1;
+ }
+ ret = vcmPlayback->InitializeReceiver();
+ if (ret < 0)
+ {
+ return -1;
+ }
+ vcm->RegisterFrameStorageCallback(&storageCallback);
+ vcmPlayback->RegisterReceiveCallback(&receiveCallback);
+ RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
+ PayloadTypeList payloadTypes;
+ payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
+ kVideoCodecVP8));
+
+ // Register receive codecs in VCM
+ for (PayloadTypeList::iterator it = payloadTypes.begin();
+ it != payloadTypes.end(); ++it) {
+ PayloadCodecTuple* payloadType = *it;
+ if (payloadType != NULL)
+ {
+ VideoCodec codec;
+ memset(&codec, 0, sizeof(codec));
+ strncpy(codec.plName, payloadType->name.c_str(), payloadType->name.length());
+ codec.plName[payloadType->name.length()] = '\0';
+ codec.plType = payloadType->payloadType;
+ codec.codecType = payloadType->codecType;
+ if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
+ {
+ return -1;
+ }
+ if (vcmPlayback->RegisterReceiveCodec(&codec, 1) < 0)
+ {
+ return -1;
+ }
+ }
+ }
+ if (rtpStream.Initialize(&payloadTypes) < 0)
+ {
+ return -1;
+ }
+ bool nackEnabled = protectionEnabled && (protectionMethod == kProtectionNack ||
+ protectionMethod == kProtectionDualDecoder);
+ rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
+ rtpStream.SetReordering(reordering);
+ vcm->SetChannelParameters(0, 0, rttMS);
+ vcm->SetVideoProtection(protectionMethod, protectionEnabled);
+ vcm->SetRenderDelay(renderDelayMs);
+ vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+
+ ret = 0;
+
+ // RTP stream main loop
+ while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
+ {
+ if (clock.MillisecondTimestamp() % 5 == 0)
+ {
+ ret = vcm->Decode();
+ if (ret < 0)
+ {
+ return -1;
+ }
+ }
+ if (vcm->TimeUntilNextProcess() <= 0)
+ {
+ vcm->Process();
+ }
+ if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
+ {
+ break;
+ }
+ clock.IncrementDebugClock(1);
+ }
+
+ switch (ret)
+ {
+ case 1:
+ printf("Success\n");
+ break;
+ case -1:
+ printf("Failed\n");
+ break;
+ case 0:
+ printf("Timeout\n");
+ break;
+ }
+
+ rtpStream.Print();
+
+ // Tear down
+ while (!payloadTypes.empty())
+ {
+ delete payloadTypes.front();
+ payloadTypes.pop_front();
+ }
+ VideoCodingModule::Destroy(vcm);
+ vcm = NULL;
+ VideoCodingModule::Destroy(vcmPlayback);
+ vcmPlayback = NULL;
+ Trace::ReturnTrace();
+
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/main/test/generic_codec_test.cc b/trunk/src/modules/video_coding/main/test/generic_codec_test.cc
new file mode 100644
index 0000000..56b3c86
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/generic_codec_test.cc
@@ -0,0 +1,587 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "generic_codec_test.h"
+#include <cmath>
+#include <stdio.h>
+#include "../source/event.h"
+#include "rtp_rtcp.h"
+#include "module_common_types.h"
+#include "test_macros.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+
+using namespace webrtc;
+
+enum { kMaxWaitEncTimeMs = 100 };
+
+int GenericCodecTest::RunTest(CmdArgs& args)
+{
+#if !defined(EVENT_DEBUG)
+ printf("\n\nEnable debug events to run this test!\n\n");
+ return -1;
+#endif
+ FakeTickTime clock(0);
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ GenericCodecTest* get = new GenericCodecTest(vcm, &clock);
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (test::OutputPath() + "genericCodecTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ get->Perform(args);
+ Trace::ReturnTrace();
+ delete get;
+ VideoCodingModule::Destroy(vcm);
+ return 0;
+}
+
+GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm, FakeTickTime* clock):
+_clock(clock),
+_vcm(vcm),
+_width(0),
+_height(0),
+_frameRate(0),
+_lengthSourceFrame(0),
+_timeStamp(0)
+{
+}
+
+GenericCodecTest::~GenericCodecTest()
+{
+}
+
+void
+GenericCodecTest::Setup(CmdArgs& args)
+{
+ _timeStamp = 0;
+
+ /* Test Sequence parameters */
+
+ _inname= args.inputFile;
+ if (args.outputFile.compare(""))
+ _outname = test::OutputPath() + "GCTest_decoded.yuv";
+ else
+ _outname = args.outputFile;
+ _encodedName = test::OutputPath() + "GCTest_encoded.vp8";
+ _width = args.width;
+ _height = args.height;
+ _frameRate = args.frameRate;
+ _lengthSourceFrame = 3*_width*_height/2;
+
+ /* File settings */
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+ if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ return;
+}
+WebRtc_Word32
+GenericCodecTest::Perform(CmdArgs& args)
+{
+ WebRtc_Word32 ret;
+ Setup(args);
+ /*
+ 1. sanity checks
+ 2. encode/decoder individuality
+ 3. API testing
+ 4. Target bitrate (within a specific timespan)
+ 5. Pipeline Delay
+ */
+
+ /*******************************/
+ /* sanity checks on inputs */
+ /*****************************/
+ VideoCodec sendCodec, receiveCodec;
+ sendCodec.maxBitrate = 8000;
+ TEST(_vcm->NumberOfCodecs() > 0); // This works since we now initialize the list in the constructor
+ TEST(_vcm->Codec(0, &sendCodec) == VCM_OK);
+ _vcm->InitializeSender();
+ _vcm->InitializeReceiver();
+ WebRtc_Word32 NumberOfCodecs = _vcm->NumberOfCodecs();
+ // registration of first codec in the list
+ int i = 0;
+ _vcm->Codec(0, &_sendCodec);
+ TEST(_vcm->RegisterSendCodec(&_sendCodec, 4, 1440) == VCM_OK);
+ // sanity on encoder registration
+ VideoFrame sourceFrame;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ _vcm->InitializeSender();
+ TEST(_vcm->Codec(kVideoCodecVP8, &sendCodec) == 0);
+ TEST(_vcm->RegisterSendCodec(&sendCodec, -1, 1440) < 0); // bad number of cores
+ sendCodec.maxBitrate = 8000;
+ _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
+ _vcm->InitializeSender();
+ _vcm->Codec(kVideoCodecVP8, &sendCodec);
+ sendCodec.height = 0;
+ TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0); // bad height
+ _vcm->Codec(kVideoCodecVP8, &sendCodec);
+ sendCodec.startBitrate = -2;
+ TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0); // bad bit rate
+ _vcm->Codec(kVideoCodecVP8, &sendCodec);
+ _vcm->InitializeSender();
+ TEST(_vcm->SetChannelParameters(100, 0, 0) < 0);// setting rate when encoder uninitialized
+ // register all availbale decoders -- need to have more for this test
+ for (i=0; i< NumberOfCodecs; i++)
+ {
+ _vcm->Codec(i, &receiveCodec);
+ _vcm->RegisterReceiveCodec(&receiveCodec, 1);
+ }
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ // building source frame
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ sourceFrame.SetTimeStamp(_timeStamp++);
+ // encode/decode
+ TEST(_vcm->AddVideoFrame(sourceFrame) < 0 ); // encoder uninitialized
+ _vcm->InitializeReceiver();
+ TEST(_vcm->SetChannelParameters(100, 0, 0) < 0);// setting rtt when receiver uninitialized
+
+ /**************************************/
+ /* encoder/decoder individuality test */
+ /**************************************/
+ //Register both encoder and decoder, reset decoder - encode, set up decoder, reset encoder - decode.
+ rewind(_sourceFile);
+ sourceFrame.Free();
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ NumberOfCodecs = _vcm->NumberOfCodecs();
+ // Register VP8
+ _vcm->Codec(kVideoCodecVP8, &_sendCodec);
+ _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
+ _vcm->SendCodec(&sendCodec);
+ sendCodec.startBitrate = 2000;
+
+ // Set target frame rate to half of the incoming frame rate
+ // to test the frame rate control in the VCM
+ sendCodec.maxFramerate = (WebRtc_UWord8)(_frameRate / 2);
+ sendCodec.width = _width;
+ sendCodec.height = _height;
+ TEST(strncmp(_sendCodec.plName, "VP8", 3) == 0); // was VP8
+
+ _decodeCallback = new VCMDecodeCompleteCallback(_decodedFile);
+ _encodeCompleteCallback = new VCMEncodeCompleteCallback(_encodedFile);
+ _vcm->RegisterReceiveCallback(_decodeCallback);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
+
+ _vcm->RegisterSendCodec(&sendCodec, 4, 1440);
+ _encodeCompleteCallback->SetCodecType(ConvertCodecType(sendCodec.plName));
+
+ _vcm->InitializeReceiver();
+ _vcm->Process();
+
+ //encoding 1 second of video
+ for (i = 0; i < _frameRate; i++)
+ {
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ IncrementDebugClock(_frameRate);
+ _vcm->Process();
+ }
+ sendCodec.maxFramerate = (WebRtc_UWord8)_frameRate;
+ _vcm->InitializeSender();
+ TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK); // same codec for encode and decode
+ ret = 0;
+ i = 0;
+ while ((i < 25) && (ret == 0) )
+ {
+ ret = _vcm->Decode();
+ TEST(ret == VCM_OK);
+ if (ret < 0)
+ {
+ printf("error in frame # %d \n", i);
+ }
+ IncrementDebugClock(_frameRate);
+ i++;
+ }
+ //TEST((ret == 0) && (i = 50));
+ if (ret == 0)
+ {
+ printf("Encoder/Decoder individuality test complete - View output files \n");
+ }
+ // last frame - not decoded
+ _vcm->InitializeReceiver();
+ TEST(_vcm->Decode() < 0); // frame to be encoded exists, decoder uninitialized
+
+
+ // Test key frame request on packet loss mode.
+ // This a frame as a key frame and fooling the receiver
+ // that the last packet was lost. The decoding will succeed,
+ // but the VCM will see a packet loss and request a new key frame.
+ VCMEncComplete_KeyReqTest keyReqTest_EncCompleteCallback(*_vcm);
+ KeyFrameReqTest frameTypeCallback;
+ _vcm->RegisterTransportCallback(&keyReqTest_EncCompleteCallback);
+ _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
+ _vcm->RegisterSendCodec(&sendCodec, 4, 1440);
+ _encodeCompleteCallback->SetCodecType(ConvertCodecType(sendCodec.plName));
+ TEST(_vcm->SetVideoProtection(kProtectionKeyOnKeyLoss, true) == VCM_OK);
+ TEST(_vcm->RegisterFrameTypeCallback(&frameTypeCallback) == VCM_OK);
+ TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ // First packet of a subsequent frame required before the jitter buffer
+ // will allow decoding an incomplete frame.
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ TEST(_vcm->Decode() == VCM_OK);
+
+ printf("API tests complete \n");
+
+ /*******************/
+ /* Bit Rate Tests */
+ /*****************/
+ /* Requirements:
+ * 1. OneSecReq = 15 % above/below target over a time period of 1s (_frameRate number of frames)
+ * 3. FullReq = 10% for total seq. (for 300 frames/seq. coincides with #1)
+ * 4. Test will go over all registered codecs
+ //NOTE: time requirements are not part of the release tests
+ */
+ double FullReq = 0.1;
+ //double OneSecReq = 0.15;
+ printf("\n RATE CONTROL TEST\n");
+ // initializing....
+ _vcm->InitializeSender();
+ _vcm->InitializeReceiver();
+ rewind(_sourceFile);
+ sourceFrame.Free();
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ const float bitRate[] = {100, 400, 600, 1000, 2000};
+ const float nBitrates = sizeof(bitRate)/sizeof(*bitRate);
+ float _bitRate = 0;
+ int _frameCnt = 0;
+ float totalBytesOneSec;//, totalBytesTenSec;
+ float totalBytes, actualBitrate;
+ VCMFrameCount frameCount; // testing frame type counters
+ // start test
+ NumberOfCodecs = _vcm->NumberOfCodecs();
+ // going over all available codecs
+ _encodeCompleteCallback->SetFrameDimensions(_width, _height);
+ SendStatsTest sendStats;
+ for (int k = 0; k < NumberOfCodecs; k++)
+ //for (int k = NumberOfCodecs - 1; k >=0; k--)
+ {// static list starts from 0
+ //just checking
+ _vcm->InitializeSender();
+ _sendCodec.maxBitrate = 8000;
+ TEST(_vcm->Codec(k, &_sendCodec)== VCM_OK);
+ _vcm->RegisterSendCodec(&_sendCodec, 1, 1440);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ _encodeCompleteCallback->SetCodecType(ConvertCodecType(_sendCodec.plName));
+ printf (" \n\n Codec type = %s \n\n",_sendCodec.plName);
+ for (i = 0; i < nBitrates; i++)
+ {
+ _bitRate = static_cast<float>(bitRate[i]);
+ // just testing
+ _vcm->InitializeSender();
+ _sendCodec.startBitrate = (int)_bitRate;
+ _sendCodec.maxBitrate = 8000;
+ _sendCodec.maxFramerate = _frameRate;
+ _vcm->RegisterSendCodec(&_sendCodec, 1, 1440);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ // up to here
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
+ _frameCnt = 0;
+ totalBytes = 0;
+ _encodeCompleteCallback->Initialize();
+ sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
+ _vcm->RegisterSendStatisticsCallback(&sendStats);
+ while (fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+ _lengthSourceFrame)
+ {
+ _frameCnt++;
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+
+ ret = _vcm->AddVideoFrame(sourceFrame);
+ IncrementDebugClock(_frameRate);
+ // The following should be uncommneted for timing tests. Release tests only include
+ // compliance with full sequence bit rate.
+
+
+ //totalBytes = WaitForEncodedFrame();
+ //currentTime = VCMTickTime::MillisecondTimestamp();//clock()/(double)CLOCKS_PER_SEC;
+ if (_frameCnt == _frameRate)// @ 1sec
+ {
+ totalBytesOneSec = _encodeCompleteCallback->EncodedBytes();//totalBytes;
+ }
+ TEST(_vcm->TimeUntilNextProcess() >= 0);
+ } // video seq. encode done
+ TEST(_vcm->TimeUntilNextProcess() == 0);
+ _vcm->Process(); // Let the module calculate its send bit rate estimate
+ // estimating rates
+ // complete sequence
+ // bit rate assumes input frame rate is as specified
+ totalBytes = _encodeCompleteCallback->EncodedBytes();
+ actualBitrate = (float)(8.0/1000)*(totalBytes / (_frameCnt / _frameRate));
+
+ printf("Complete Seq.: target bitrate: %.0f kbps, actual bitrate: %.1f kbps\n", _bitRate, actualBitrate);
+ TEST((fabs(actualBitrate - _bitRate) < FullReq * _bitRate) ||
+ (strncmp(_sendCodec.plName, "I420", 4) == 0));
+
+ // 1 Sec.
+ actualBitrate = (float)(8.0/1000)*(totalBytesOneSec);
+ //actualBitrate = (float)(8.0*totalBytesOneSec)/(oneSecTime - startTime);
+ //printf("First 1Sec: target bitrate: %.0f kbps, actual bitrate: %.1f kbps\n", _bitRate, actualBitrate);
+ //TEST(fabs(actualBitrate - _bitRate) < OneSecReq * _bitRate);
+ rewind(_sourceFile);
+
+ //checking key/delta frame count
+ _vcm->SentFrameCount(frameCount);
+ printf("frame count: %d delta, %d key\n", frameCount.numDeltaFrames, frameCount.numKeyFrames);
+ }// end per codec
+
+ } // end rate control test
+ /********************************/
+ /* Encoder Pipeline Delay Test */
+ /******************************/
+ _vcm->InitializeSender();
+ sourceFrame.Free();
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ NumberOfCodecs = _vcm->NumberOfCodecs();
+ bool encodeComplete = false;
+ // going over all available codecs
+ for (int k = 0; k < NumberOfCodecs; k++)
+ {
+ _vcm->Codec(k, &_sendCodec);
+ _vcm->InitializeSender();
+ _sendCodec.maxBitrate = 8000;
+ _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+
+ _frameCnt = 0;
+ encodeComplete = false;
+ while (encodeComplete == false)
+ {
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ _frameCnt++;
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ _vcm->AddVideoFrame(sourceFrame);
+ encodeComplete = _encodeCompleteCallback->EncodeComplete();
+ } // first frame encoded
+ printf ("\n Codec type = %s \n", _sendCodec.plName);
+ printf(" Encoder pipeline delay = %d frames\n", _frameCnt - 1);
+ } // end for all codecs
+
+ /********************************/
+ /* Encoder Packet Size Test */
+ /********************************/
+ RtpRtcp& rtpModule = *RtpRtcp::CreateRtpRtcp(1, false);
+ TEST(rtpModule.InitSender() == 0);
+ RTPSendCallback_SizeTest sendCallback;
+ rtpModule.RegisterSendTransport(&sendCallback);
+
+ VCMRTPEncodeCompleteCallback encCompleteCallback(&rtpModule);
+ _vcm->InitializeSender();
+
+ // TEST DISABLED FOR NOW SINCE VP8 DOESN'T HAVE THIS FEATURE
+
+// sourceFrame.Free();
+// sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+// NumberOfCodecs = _vcm->NumberOfCodecs();
+// WebRtc_UWord32 targetPayloadSize = 500;
+// rtpModule.SetMaxTransferUnit(targetPayloadSize);
+// // going over all available codecs
+// for (int k = 0; k < NumberOfCodecs; k++)
+// {
+// _vcm->Codec(k, &_sendCodec);
+// if (strncmp(_sendCodec.plName, "VP8", 3) == 0)
+// {
+// // Only test with VP8
+// continue;
+// }
+// rtpModule.RegisterSendPayload(_sendCodec.plName, _sendCodec.plType);
+// // Make sure we only get one NAL unit per packet
+// _vcm->InitializeSender();
+// _vcm->RegisterSendCodec(&_sendCodec, 4, targetPayloadSize);
+// sendCallback.SetMaxPayloadSize(targetPayloadSize);
+// _vcm->RegisterTransportCallback(&encCompleteCallback);
+// sendCallback.Reset();
+// _frameCnt = 0;
+// rewind(_sourceFile);
+// while (!feof(_sourceFile))
+// {
+// fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+// _frameCnt++;
+// sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+// sourceFrame.SetHeight(_height);
+// sourceFrame.SetWidth(_width);
+// _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+// sourceFrame.SetTimeStamp(_timeStamp);
+// ret = _vcm->AddVideoFrame(sourceFrame);
+// } // first frame encoded
+// printf ("\n Codec type = %s \n",_sendCodec.plName);
+// printf(" Average payload size = %f bytes, target = %u bytes\n", sendCallback.AveragePayloadSize(), targetPayloadSize);
+// } // end for all codecs
+
+
+ // Test temporal decimation settings
+ for (int k = 0; k < NumberOfCodecs; k++)
+ {
+ _vcm->Codec(k, &_sendCodec);
+ if (strncmp(_sendCodec.plName, "I420", 4) == 0)
+ {
+ // Only test with I420
+ break;
+ }
+ }
+ TEST(strncmp(_sendCodec.plName, "I420", 4) == 0);
+ _vcm->InitializeSender();
+ _sendCodec.maxFramerate = static_cast<WebRtc_UWord8>(_frameRate / 2.0 + 0.5f);
+ _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
+ _vcm->SetChannelParameters(2000, 0, 0);
+ _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+ // up to here
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
+ _encodeCompleteCallback->Initialize();
+ sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
+ _vcm->RegisterSendStatisticsCallback(&sendStats);
+ rewind(_sourceFile);
+ while (fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+ _lengthSourceFrame)
+ {
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ ret = _vcm->AddVideoFrame(sourceFrame);
+ if (_vcm->TimeUntilNextProcess() <= 0)
+ {
+ _vcm->Process();
+ }
+ IncrementDebugClock(_frameRate);
+ } // first frame encoded
+
+ RtpRtcp::DestroyRtpRtcp(&rtpModule);
+ Print();
+ delete tmpBuffer;
+ delete _decodeCallback;
+ delete _encodeCompleteCallback;
+ return 0;
+}
+
+
+void
+GenericCodecTest::Print()
+{
+ printf(" \n\n VCM Generic Encoder Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+
+float
+GenericCodecTest::WaitForEncodedFrame() const
+{
+ WebRtc_Word64 startTime = _clock->MillisecondTimestamp();
+ while (_clock->MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs*10)
+ {
+ if (_encodeCompleteCallback->EncodeComplete())
+ {
+ return _encodeCompleteCallback->EncodedBytes();
+ }
+ }
+ return 0;
+}
+
+void
+GenericCodecTest::IncrementDebugClock(float frameRate)
+{
+ _clock->IncrementDebugClock(1000/frameRate);
+}
+
+int
+RTPSendCallback_SizeTest::SendPacket(int channel, const void *data, int len)
+{
+ _nPackets++;
+ _payloadSizeSum += len;
+ // Make sure no payloads (len - header size) are larger than maxPayloadSize
+ TEST(len > 0 && static_cast<WebRtc_UWord32>(len - 12) <= _maxPayloadSize);
+ return 0;
+}
+
+void
+RTPSendCallback_SizeTest::SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize)
+{
+ _maxPayloadSize = maxPayloadSize;
+}
+
+void
+RTPSendCallback_SizeTest::Reset()
+{
+ _nPackets = 0;
+ _payloadSizeSum = 0;
+}
+
+float
+RTPSendCallback_SizeTest::AveragePayloadSize() const
+{
+ if (_nPackets > 0)
+ {
+ return _payloadSizeSum / static_cast<float>(_nPackets);
+ }
+ return 0;
+}
+
+WebRtc_Word32
+VCMEncComplete_KeyReqTest::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& /*fragmentationHeader*/,
+ const webrtc::RTPVideoHeader* /*videoHdr*/)
+{
+ WebRtcRTPHeader rtpInfo;
+ rtpInfo.header.markerBit = true; // end of frame
+ rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
+ rtpInfo.type.Video.codec = kRTPVideoVP8;
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.sequenceNumber = _seqNo;
+ _seqNo += 2;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.timestamp = _timeStamp;
+ _timeStamp += 3000;
+ rtpInfo.type.Video.isFirstPacket = false;
+ rtpInfo.frameType = kVideoFrameKey;
+ return _vcm.IncomingPacket(payloadData, payloadSize, rtpInfo);
+}
diff --git a/trunk/src/modules/video_coding/main/test/generic_codec_test.h b/trunk/src/modules/video_coding/main/test/generic_codec_test.h
new file mode 100644
index 0000000..c88280f
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/generic_codec_test.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
+
+#include "video_coding.h"
+
+#include <string.h>
+#include <fstream>
+
+#include "test_callbacks.h"
+#include "test_util.h"
+/*
+Test consists of:
+1. Sanity checks
+2. Bit rate validation
+3. Encoder control test / General API functionality
+4. Decoder control test / General API functionality
+
+*/
+
+namespace webrtc {
+
+int VCMGenericCodecTest(CmdArgs& args);
+
+class FakeTickTime;
+
+class GenericCodecTest
+{
+public:
+ GenericCodecTest(webrtc::VideoCodingModule* vcm,
+ webrtc::FakeTickTime* clock);
+ ~GenericCodecTest();
+ static int RunTest(CmdArgs& args);
+ WebRtc_Word32 Perform(CmdArgs& args);
+ float WaitForEncodedFrame() const;
+
+private:
+ void Setup(CmdArgs& args);
+ void Print();
+ WebRtc_Word32 TearDown();
+ void IncrementDebugClock(float frameRate);
+
+ webrtc::FakeTickTime* _clock;
+ webrtc::VideoCodingModule* _vcm;
+ webrtc::VideoCodec _sendCodec;
+ webrtc::VideoCodec _receiveCodec;
+ std::string _inname;
+ std::string _outname;
+ std::string _encodedName;
+ WebRtc_Word32 _sumEncBytes;
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ FILE* _encodedFile;
+ WebRtc_UWord16 _width;
+ WebRtc_UWord16 _height;
+ float _frameRate;
+ WebRtc_UWord32 _lengthSourceFrame;
+ WebRtc_UWord32 _timeStamp;
+ VCMDecodeCompleteCallback* _decodeCallback;
+ VCMEncodeCompleteCallback* _encodeCompleteCallback;
+
+}; // end of GenericCodecTest class definition
+
+class RTPSendCallback_SizeTest : public webrtc::Transport
+{
+public:
+ // constructor input: (receive side) rtp module to send encoded data to
+ RTPSendCallback_SizeTest() : _maxPayloadSize(0), _payloadSizeSum(0), _nPackets(0) {}
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len) {return 0;}
+ void SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize);
+ void Reset();
+ float AveragePayloadSize() const;
+private:
+ WebRtc_UWord32 _maxPayloadSize;
+ WebRtc_UWord32 _payloadSizeSum;
+ WebRtc_UWord32 _nPackets;
+};
+
+class VCMEncComplete_KeyReqTest : public webrtc::VCMPacketizationCallback
+{
+public:
+ VCMEncComplete_KeyReqTest(webrtc::VideoCodingModule &vcm) : _vcm(vcm), _seqNo(0), _timeStamp(0) {}
+ WebRtc_Word32 SendData(
+ const webrtc::FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::RTPFragmentationHeader& fragmentationHeader,
+ const webrtc::RTPVideoHeader* videoHdr);
+private:
+ webrtc::VideoCodingModule& _vcm;
+ WebRtc_UWord16 _seqNo;
+ WebRtc_UWord32 _timeStamp;
+}; // end of VCMEncodeCompleteCallback
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc b/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc
new file mode 100644
index 0000000..2066983
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc
@@ -0,0 +1,1936 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <stdio.h>
+
+#include "common_types.h"
+#include "../source/event.h"
+#include "frame_buffer.h"
+#include "inter_frame_delay.h"
+#include "jitter_buffer.h"
+#include "jitter_estimate_test.h"
+#include "jitter_estimator.h"
+#include "media_opt_util.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "packet.h"
+#include "test_util.h"
+#include "test_macros.h"
+
+// TODO(holmer): Get rid of this to conform with style guide.
+using namespace webrtc;
+
+// TODO (Mikhal/Stefan): Update as gtest and separate to specific tests.
+
+int CheckOutFrame(VCMEncodedFrame* frameOut, unsigned int size, bool startCode)
+{
+ if (frameOut == 0)
+ {
+ return -1;
+ }
+
+ const WebRtc_UWord8* outData = frameOut->Buffer();
+
+ unsigned int i = 0;
+
+ if(startCode)
+ {
+ if (outData[0] != 0 || outData[1] != 0 || outData[2] != 0 ||
+ outData[3] != 1)
+ {
+ return -2;
+ }
+ i+= 4;
+ }
+
+ // check the frame data
+ int count = 3;
+
+ // check the frame length
+ if (frameOut->Length() != size)
+ {
+ return -3;
+ }
+
+ for(; i < size; i++)
+ {
+ if (outData[i] == 0 && outData[i + 1] == 0 && outData[i + 2] == 0x80)
+ {
+ i += 2;
+ }
+ else if(startCode && outData[i] == 0 && outData[i + 1] == 0)
+ {
+ if (outData[i] != 0 || outData[i + 1] != 0 ||
+ outData[i + 2] != 0 || outData[i + 3] != 1)
+ {
+ return -3;
+ }
+ i += 3;
+ }
+ else
+ {
+ if (outData[i] != count)
+ {
+ return -4;
+ }
+ count++;
+ if(count == 10)
+ {
+ count = 3;
+ }
+ }
+ }
+ return 0;
+}
+
+
+int JitterBufferTest(CmdArgs& args)
+{
+ // Don't run these tests with debug event.
+#if defined(EVENT_DEBUG)
+ return -1;
+#endif
+ TickTimeBase clock;
+
+ // Start test
+ WebRtc_UWord16 seqNum = 1234;
+ WebRtc_UWord32 timeStamp = 0;
+ int size = 1400;
+ WebRtc_UWord8 data[1500];
+ VCMPacket packet(data, size, seqNum, timeStamp, true);
+
+ VCMJitterBuffer jb(&clock);
+
+ seqNum = 1234;
+ timeStamp = 123*90;
+ FrameType incomingFrameType(kVideoFrameKey);
+ VCMEncodedFrame* frameOut=NULL;
+ WebRtc_Word64 renderTimeMs = 0;
+ packet.timestamp = timeStamp;
+ packet.seqNum = seqNum;
+
+ // build a data vector with 0, 0, 0x80, 3, 4, 5, 6, 7, 8, 9, 0, 0, 0x80, 3....
+ data[0] = 0;
+ data[1] = 0;
+ data[2] = 0x80;
+ int count = 3;
+ for (unsigned int i = 3; i < sizeof(data) - 3; ++i)
+ {
+ data[i] = count;
+ count++;
+ if(count == 10)
+ {
+ data[i+1] = 0;
+ data[i+2] = 0;
+ data[i+3] = 0x80;
+ count = 3;
+ i += 3;
+ }
+ }
+
+ // Test out of range inputs
+ TEST(kSizeError == jb.InsertPacket(0, packet));
+ jb.ReleaseFrame(0);
+
+ // Not started
+ TEST(0 == jb.GetFrame(packet));
+ TEST(-1 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(0 == jb.GetCompleteFrameForDecoding(10));
+ TEST(0 == jb.GetFrameForDecoding());
+
+ // Start
+ jb.Start();
+
+ // Get frame to use for this timestamp
+ VCMEncodedFrame* frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // No packets inserted
+ TEST(0 == jb.GetCompleteFrameForDecoding(10));
+
+
+ //
+ // TEST single packet frame
+ //
+ // --------
+ // | 1234 |
+ // --------
+
+ // packet.frameType;
+ // packet.dataPtr;
+ // packet.sizeBytes;
+ // packet.timestamp;
+ // packet.seqNum;
+ // packet.isFirstPacket;
+ // packet.markerBit;
+ //
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 1 packet\n");
+
+ //
+ // TEST dual packet frame
+ //
+ // -----------------
+ // | 1235 | 1236 |
+ // -----------------
+ //
+
+ seqNum++;
+ timeStamp += 33*90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 2 packets\n");
+
+
+ //
+ // TEST 100 packets frame Key frame
+ //
+ // ----------------------------------
+ // | 1237 | 1238 | .... | 1336 |
+ // ----------------------------------
+
+ // insert first packet
+ timeStamp += 33*90;
+ seqNum++;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameKey);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert 98 frames
+ int loop = 0;
+ do
+ {
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ loop++;
+ } while (loop < 98);
+
+ // insert last packet
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*100, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameKey);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE key frame 100 packets\n");
+
+ //
+ // TEST 100 packets frame Delta frame
+ //
+ // ----------------------------------
+ // | 1337 | 1238 | .... | 1436 |
+ // ----------------------------------
+
+ // insert first packet
+ timeStamp += 33*90;
+ seqNum++;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert 98 frames
+ loop = 0;
+ do
+ {
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ loop++;
+ } while (loop < 98);
+
+ // insert last packet
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*100, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 100 packets\n");
+
+ //
+ // TEST packet re-ordering reverse order
+ //
+ // ----------------------------------
+ // | 1437 | 1238 | .... | 1536 |
+ // ----------------------------------
+ // <----------
+
+ // insert "first" packet last seqnum
+ timeStamp += 33*90;
+ seqNum += 100;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert 98 packets
+ loop = 0;
+ do
+ {
+ seqNum--;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ loop++;
+ } while (loop < 98);
+
+ // insert last packet
+ seqNum--;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*100, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 100 packets reverse order\n");
+
+ seqNum+= 100;
+
+ //
+ // TEST frame re-ordering 2 frames 2 packets each
+ //
+ // ----------------- -----------------
+ // | 1539 | 1540 | | 1537 | 1538 |
+ // ----------------- -----------------
+
+ seqNum += 2;
+ timeStamp += 2* 33 * 90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // check that we fail to get frame since seqnum is not continuous
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(frameOut == 0);
+
+ seqNum -= 3;
+ timeStamp -= 33*90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ seqNum += 2;
+ //printf("DONE frame re-ordering 2 frames 2 packets\n");
+
+ // restore
+ packet.dataPtr = data;
+ packet.codec = kVideoCodecUnknown;
+
+ //
+ // TEST duplicate packets
+ //
+ // -----------------
+ // | 1543 | 1543 |
+ // -----------------
+ //
+
+ seqNum++;
+ timeStamp += 2*33*90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kDuplicatePacket == jb.InsertPacket(frameIn, packet));
+
+ seqNum++;
+ packet.seqNum = seqNum;
+
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE test duplicate packets\n");
+
+ //
+ // TEST H.264 insert start code
+ //
+ // -----------------
+ // | 1544 | 1545 |
+ // -----------------
+ // insert start code, both packets
+
+ seqNum++;
+ timeStamp += 33 * 90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.insertStartCode = true;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size * 2 + 4 * 2, true) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ // reset
+ packet.insertStartCode = false;
+ //printf("DONE H.264 insert start code test 2 packets\n");
+
+ //
+ // TEST statistics
+ //
+ WebRtc_UWord32 numDeltaFrames = 0;
+ WebRtc_UWord32 numKeyFrames = 0;
+ TEST(jb.GetFrameStatistics(numDeltaFrames, numKeyFrames) == 0);
+
+ TEST(numDeltaFrames == 8);
+ TEST(numKeyFrames == 1);
+
+ WebRtc_UWord32 frameRate;
+ WebRtc_UWord32 bitRate;
+ TEST(jb.GetUpdate(frameRate, bitRate) == 0);
+
+ // these depend on CPU speed works on a T61
+ TEST(frameRate > 30);
+ TEST(bitRate > 10000000);
+
+
+ jb.Flush();
+
+ //
+ // TEST packet loss. Verify missing packets statistics and not decodable
+ // packets statistics.
+ // Insert 10 frames consisting of 4 packets and remove one from all of them.
+ // The last packet is an empty (non-media) packet
+ //
+
+ // Select a start seqNum which triggers a difficult wrap situation
+ // The JB will only output (incomplete)frames if the next one has started
+ // to arrive. Start by inserting one frame (key).
+ seqNum = 0xffff - 4;
+ seqNum++;
+ timeStamp += 33*90;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.completeNALU = kNaluStart;
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ for (int i = 0; i < 11; ++i) {
+ webrtc::FrameType frametype = kVideoFrameDelta;
+ seqNum++;
+ timeStamp += 33*90;
+ packet.frameType = frametype;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.completeNALU = kNaluStart;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(timeStamp - 33 * 90 == jb.GetNextTimeStamp(10, incomingFrameType,
+ renderTimeMs));
+
+ // Check incoming frame type
+ if (i == 0)
+ {
+ TEST(incomingFrameType == kVideoFrameKey);
+ }
+ else
+ {
+ TEST(incomingFrameType == frametype);
+ }
+
+ // Get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // Should not be complete
+ TEST(frameOut == 0);
+
+ seqNum += 2;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.completeNALU = kNaluEnd;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+
+ // Insert an empty (non-media) packet
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.completeNALU = kNaluEnd;
+ packet.frameType = kFrameEmpty;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ // Get the frame
+ frameOut = jb.GetFrameForDecoding();
+
+ // One of the packets has been discarded by the jitter buffer.
+ // Last frame can't be extracted yet.
+ if (i < 10)
+ {
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ if (i == 0)
+ {
+ TEST(frameOut->FrameType() == kVideoFrameKey);
+ }
+ else
+ {
+ TEST(frameOut->FrameType() == frametype);
+ }
+ TEST(frameOut->Complete() == false);
+ TEST(frameOut->MissingFrame() == false);
+ }
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+ }
+
+ TEST(jb.NumNotDecodablePackets() == 10);
+
+ // Insert 3 old packets and verify that we have 3 discarded packets
+ // Match value to actual latest timestamp decoded
+ timeStamp -= 33 * 90;
+ packet.timestamp = timeStamp - 1000;
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn == NULL);
+
+ packet.timestamp = timeStamp - 500;
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn == NULL);
+
+ packet.timestamp = timeStamp - 100;
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn == NULL);
+
+ TEST(jb.DiscardedPackets() == 3);
+
+ jb.Flush();
+
+ // This statistic shouldn't be reset by a flush.
+ TEST(jb.DiscardedPackets() == 3);
+
+ //printf("DONE Statistics\n");
+
+
+ // Temporarily do this to make the rest of the test work:
+ timeStamp += 33*90;
+ seqNum += 4;
+
+
+ //
+ // TEST delta frame 100 packets with seqNum wrap
+ //
+ // ---------------------------------------
+ // | 65520 | 65521 | ... | 82 | 83 |
+ // ---------------------------------------
+ //
+
+ jb.Flush();
+
+ // insert first packet
+ timeStamp += 33*90;
+ seqNum = 0xfff0;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert 98 packets
+ loop = 0;
+ do
+ {
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(2, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(2);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ loop++;
+ } while (loop < 98);
+
+ // insert last packet
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*100, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 100 packets with wrap in seqNum\n");
+
+ //
+ // TEST packet re-ordering reverse order with neg seqNum warp
+ //
+ // ----------------------------------------
+ // | 65447 | 65448 | ... | 9 | 10 |
+ // ----------------------------------------
+ // <-------------
+
+ // test flush
+ jb.Flush();
+
+ // insert "first" packet last seqnum
+ timeStamp += 33*90;
+ seqNum = 10;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert 98 frames
+ loop = 0;
+ do
+ {
+ seqNum--;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(2, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(2);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ loop++;
+ } while (loop < 98);
+
+ // insert last packet
+ seqNum--;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*100, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 100 packets reverse order with wrap in seqNum \n");
+
+ // test flush
+ jb.Flush();
+
+ //
+ // TEST packet re-ordering with seqNum wrap
+ //
+ // -----------------------
+ // | 1 | 65535 | 0 |
+ // -----------------------
+
+ // insert "first" packet last seqnum
+ timeStamp += 33*90;
+ seqNum = 1;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ // insert last packet
+ seqNum -= 2;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*3, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE delta frame 3 packets re-ordering with wrap in seqNum \n");
+
+ // test flush
+ jb.Flush();
+
+ //
+ // TEST insert old frame
+ //
+ // ------- -------
+ // | 2 | | 1 |
+ // ------- -------
+ // t = 3000 t = 2000
+
+ seqNum = 2;
+ timeStamp = 3000;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(3000 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(3000 == frameOut->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ jb.ReleaseFrame(frameOut);
+
+ seqNum--;
+ timeStamp = 2000;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ // Changed behavior, never insert packets into frames older than the
+ // last decoded frame.
+ TEST(frameIn == 0);
+
+ //printf("DONE insert old frame\n");
+
+ jb.Flush();
+
+ //
+ // TEST insert old frame with wrap in timestamp
+ //
+ // ------- -------
+ // | 2 | | 1 |
+ // ------- -------
+ // t = 3000 t = 0xffffff00
+
+ seqNum = 2;
+ timeStamp = 3000;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(timeStamp == frameOut->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ jb.ReleaseFrame(frameOut);
+
+ seqNum--;
+ timeStamp = 0xffffff00;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ // This timestamp is old
+ TEST(frameIn == 0);
+
+ jb.Flush();
+
+ //
+ // TEST wrap in timeStamp
+ //
+ // --------------- ---------------
+ // | 1 | 2 | | 3 | 4 |
+ // --------------- ---------------
+ // t = 0xffffff00 t = 33*90
+
+ seqNum = 1;
+ timeStamp = 0xffffff00;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ jb.ReleaseFrame(frameOut);
+
+ seqNum++;
+ timeStamp += 33*90;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ // it should not be complete
+ TEST(frameOut == 0);
+
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+
+ TEST(CheckOutFrame(frameOut, size*2, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ //printf("DONE time stamp wrap 2 frames 2 packets\n");
+
+ jb.Flush();
+
+ //
+ // TEST insert 2 frames with wrap in timeStamp
+ //
+ // ------- -------
+ // | 1 | | 2 |
+ // ------- -------
+ // t = 0xffffff00 t = 2700
+
+ seqNum = 1;
+ timeStamp = 0xffffff00;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert first frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Insert next frame
+ seqNum++;
+ timeStamp = 2700;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(0xffffff00 == frameOut->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // Get packet notification
+ TEST(2700 == jb.GetNextTimeStamp(0, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get frame
+ VCMEncodedFrame* frameOut2 = jb.GetCompleteFrameForDecoding(10);
+ TEST(2700 == frameOut2->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut2, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut2->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+ jb.ReleaseFrame(frameOut2);
+
+ //printf("DONE insert 2 frames (1 packet) with wrap in timestamp\n");
+
+ jb.Flush();
+
+ //
+ // TEST insert 2 frames re-ordered with wrap in timeStamp
+ //
+ // ------- -------
+ // | 2 | | 1 |
+ // ------- -------
+ // t = 2700 t = 0xffffff00
+
+ seqNum = 2;
+ timeStamp = 2700;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert first frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(2700 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Insert second frame
+ seqNum--;
+ timeStamp = 0xffffff00;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(0xffffff00 == frameOut->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameDelta);
+
+ // get packet notification
+ TEST(2700 == jb.GetNextTimeStamp(0, incomingFrameType, renderTimeMs));
+ TEST(kVideoFrameDelta == incomingFrameType);
+
+ // Get frame
+ frameOut2 = jb.GetCompleteFrameForDecoding(10);
+ TEST(2700 == frameOut2->TimeStamp());
+
+ TEST(CheckOutFrame(frameOut2, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut2->FrameType() == kVideoFrameDelta);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+ jb.ReleaseFrame(frameOut2);
+
+ //printf("DONE insert 2 frames (1 packet) re-ordered with wrap in timestamp\n");
+
+ //
+ // TEST delta frame with more than max number of packets
+ //
+
+ jb.Start();
+
+ loop = 0;
+ packet.timestamp += 33*90;
+ bool firstPacket = true;
+ // insert kMaxPacketsInJitterBuffer into frame
+ do
+ {
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert frame
+ if (firstPacket)
+ {
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+ firstPacket = false;
+ }
+ else
+ {
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ }
+
+ // get packet notification
+ TEST(packet.timestamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ loop++;
+ } while (loop < kMaxPacketsInSession);
+
+ // Max number of packets inserted
+
+ // Insert one more packet
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ // Insert the packet -> frame recycled
+ TEST(kSizeError == jb.InsertPacket(frameIn, packet));
+
+ TEST(0 == jb.GetCompleteFrameForDecoding(10));
+
+ //printf("DONE fill frame - packets > max number of packets\n");
+
+ //
+ // TEST fill JB with more than max number of frame (50 delta frames +
+ // 51 key frames) with wrap in seqNum
+ //
+ // --------------------------------------------------------------
+ // | 65485 | 65486 | 65487 | .... | 65535 | 0 | 1 | 2 | .....| 50 |
+ // --------------------------------------------------------------
+ // |<-----------delta frames------------->|<------key frames----->|
+
+ jb.Flush();
+
+ loop = 0;
+ seqNum = 65485;
+ WebRtc_UWord32 timeStampStart = timeStamp + 33*90;
+ WebRtc_UWord32 timeStampFirstKey = 0;
+ VCMEncodedFrame* ptrLastDeltaFrame = NULL;
+ VCMEncodedFrame* ptrFirstKeyFrame = NULL;
+ // insert MAX_NUMBER_OF_FRAMES frames
+ do
+ {
+ timeStamp += 33*90;
+ seqNum++;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ frameIn = jb.GetFrame(packet);
+ TEST(frameIn != 0);
+
+ if (loop == 49) // last delta
+ {
+ ptrLastDeltaFrame = frameIn;
+ }
+ if (loop == 50) // first key
+ {
+ ptrFirstKeyFrame = frameIn;
+ packet.frameType = kVideoFrameKey;
+ timeStampFirstKey = packet.timestamp;
+ }
+
+ // Insert frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification, should be first inserted frame
+ TEST(timeStampStart == jb.GetNextTimeStamp(10, incomingFrameType,
+ renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameDelta);
+
+ loop++;
+ } while (loop < kMaxNumberOfFrames);
+
+ // Max number of frames inserted
+
+ // Insert one more frame
+ timeStamp += 33*90;
+ seqNum++;
+ packet.isFirstPacket = true;
+ packet.markerBit = true;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+
+ // Now, no free frame - frames will be recycled until first key frame
+ frameIn = jb.GetFrame(packet);
+ // ptr to last inserted delta frame should be returned
+ TEST(frameIn != 0 && frameIn && ptrLastDeltaFrame);
+
+ // Insert frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // First inserted key frame should be oldest in buffer
+ TEST(timeStampFirstKey == jb.GetNextTimeStamp(10, incomingFrameType,
+ renderTimeMs));
+
+ // check incoming frame type
+ TEST(incomingFrameType == kVideoFrameKey);
+
+ // get the first key frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ TEST(ptrFirstKeyFrame == frameOut);
+
+ TEST(CheckOutFrame(frameOut, size, false) == 0);
+
+ // check the frame type
+ TEST(frameOut->FrameType() == kVideoFrameKey);
+
+ // Release frame (when done with decoding)
+ jb.ReleaseFrame(frameOut);
+
+ jb.Flush();
+
+ // printf("DONE fill JB - nr of delta + key frames (w/ wrap in seqNum) >
+ // max nr of frames\n");
+
+ // Testing that 1 empty packet inserted last will not be set for decoding
+ seqNum = 3;
+ // Insert one empty packet per frame, should never return the last timestamp
+ // inserted. Only return empty frames in the presence of subsequent frames.
+ int maxSize = 1000;
+ for (int i = 0; i < maxSize + 10; i++)
+ {
+ timeStamp += 33 * 90;
+ seqNum++;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kFrameEmpty;
+ VCMEncodedFrame* testFrame = jb.GetFrameForDecoding();
+ // timestamp should bever be the last TS inserted
+ if (testFrame != NULL)
+ {
+ TEST(testFrame->TimeStamp() < timeStamp);
+ printf("Not null TS = %d\n",testFrame->TimeStamp());
+ }
+ }
+
+ jb.Flush();
+
+
+ // printf(DONE testing inserting empty packets to the JB)
+
+
+ // H.264 tests
+ // Test incomplete NALU frames
+
+ jb.Flush();
+ jb.SetNackMode(kNoNack, -1, -1);
+ seqNum ++;
+ timeStamp += 33 * 90;
+ int insertedLength = 0;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = true;
+ packet.completeNALU = kNaluStart;
+ packet.markerBit = false;
+
+ frameIn = jb.GetFrame(packet);
+
+ // Insert a packet into a frame
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ seqNum += 2; // Skip one packet
+ packet.seqNum = seqNum;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluIncomplete;
+ packet.markerBit = false;
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ seqNum++;
+ packet.seqNum = seqNum;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluEnd;
+ packet.markerBit = false;
+
+ // Insert a packet into a frame
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+ seqNum++;
+ packet.seqNum = seqNum;
+ packet.completeNALU = kNaluComplete;
+ packet.markerBit = true; // Last packet
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+
+
+ // The JB will only output (incomplete) frames if a packet belonging to a
+ // subsequent frame was already inserted. Insert one packet of a subsequent
+ // frame. place high timestamp so the JB would always have a next frame
+ // (otherwise, for every inserted frame we need to take care of the next
+ // frame as well).
+ packet.seqNum = 1;
+ packet.timestamp = timeStamp + 33 * 90 * 10;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluStart;
+ packet.markerBit = false;
+ frameIn = jb.GetFrame(packet);
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ // Get packet notification
+ TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
+ frameOut = jb.GetFrameForDecoding();
+
+ // We can decode everything from a NALU until a packet has been lost.
+ // Thus we can decode the first packet of the first NALU and the second NALU
+ // which consists of one packet.
+ TEST(CheckOutFrame(frameOut, packet.sizeBytes * 2, false) == 0);
+ jb.ReleaseFrame(frameOut);
+
+ // Test reordered start frame + 1 lost
+ seqNum += 2; // Reoreder 1 frame
+ timeStamp += 33*90;
+ insertedLength = 0;
+
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluEnd;
+ packet.markerBit = false;
+
+ TEST(frameIn = jb.GetFrame(packet));
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+ insertedLength += packet.sizeBytes; // This packet should be decoded
+
+ seqNum--;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = true;
+ packet.completeNALU = kNaluStart;
+ packet.markerBit = false;
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ insertedLength += packet.sizeBytes; // This packet should be decoded
+
+ seqNum += 3; // One packet drop
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluComplete;
+ packet.markerBit = false;
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ insertedLength += packet.sizeBytes; // This packet should be decoded
+
+ seqNum += 1;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluStart;
+ packet.markerBit = false;
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ // This packet should be decoded since it's the beginning of a NAL
+ insertedLength += packet.sizeBytes;
+
+ seqNum += 2;
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = false;
+ packet.completeNALU = kNaluEnd;
+ packet.markerBit = true;
+ TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
+ // This packet should not be decoded because it is an incomplete NAL if it
+ // is the last
+
+ frameOut = jb.GetFrameForDecoding();
+ // Only last NALU is complete
+ TEST(CheckOutFrame(frameOut, insertedLength, false) == 0);
+ jb.ReleaseFrame(frameOut);
+
+
+ // Test to insert empty packet
+ seqNum += 1;
+ timeStamp += 33 * 90;
+ VCMPacket emptypacket(data, 0, seqNum, timeStamp, true);
+ emptypacket.seqNum = seqNum;
+ emptypacket.timestamp = timeStamp;
+ emptypacket.frameType = kVideoFrameKey;
+ emptypacket.isFirstPacket = true;
+ emptypacket.completeNALU = kNaluComplete;
+ emptypacket.markerBit = true;
+ TEST(frameIn = jb.GetFrame(emptypacket));
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, emptypacket));
+ // This packet should not be decoded because it is an incomplete NAL if it
+ // is the last
+ insertedLength += 0;
+
+ // Will be sent to the decoder, as a packet belonging to a subsequent frame
+ // has arrived.
+ frameOut = jb.GetFrameForDecoding();
+
+
+ // Test that a frame can include an empty packet.
+ seqNum += 1;
+ timeStamp += 33 * 90;
+
+ packet.seqNum = seqNum;
+ packet.timestamp = timeStamp;
+ packet.frameType = kVideoFrameKey;
+ packet.isFirstPacket = true;
+ packet.completeNALU = kNaluComplete;
+ packet.markerBit = false;
+ TEST(frameIn = jb.GetFrame(packet));
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ seqNum += 1;
+ emptypacket.seqNum = seqNum;
+ emptypacket.timestamp = timeStamp;
+ emptypacket.frameType = kVideoFrameKey;
+ emptypacket.isFirstPacket = true;
+ emptypacket.completeNALU = kNaluComplete;
+ emptypacket.markerBit = true;
+ TEST(kCompleteSession == jb.InsertPacket(frameIn, emptypacket));
+
+ // get the frame
+ frameOut = jb.GetCompleteFrameForDecoding(10);
+ // Only last NALU is complete
+ TEST(CheckOutFrame(frameOut, packet.sizeBytes, false) == 0);
+
+ jb.ReleaseFrame(frameOut);
+
+ jb.Flush();
+
+ // Test that a we cannot get incomplete frames from the JB if we haven't
+ // received the marker bit, unless we have received a packet from a later
+ // timestamp.
+
+ packet.seqNum += 2;
+ packet.frameType = kVideoFrameDelta;
+ packet.isFirstPacket = false;
+ packet.markerBit = false;
+
+ TEST(frameIn = jb.GetFrame(packet));
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ frameOut = jb.GetFrameForDecoding();
+ TEST(frameOut == NULL);
+
+ packet.seqNum += 2;
+ packet.timestamp += 33 * 90;
+
+ TEST(frameIn = jb.GetFrame(packet));
+ TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
+
+ frameOut = jb.GetFrameForDecoding();
+
+ TEST(frameOut != NULL);
+ TEST(CheckOutFrame(frameOut, packet.sizeBytes, false) == 0);
+ jb.ReleaseFrame(frameOut);
+
+ jb.Stop();
+
+ printf("DONE !!!\n");
+
+ printf("\nVCM Jitter Buffer Test: \n\n%i tests completed\n",
+ vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+
+ return 0;
+
+}
diff --git a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc
new file mode 100644
index 0000000..ac74a8a
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc
@@ -0,0 +1,109 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <ctime>
+#include "JitterEstimateTest.h"
+
+using namespace webrtc;
+
+JitterEstimateTest::JitterEstimateTest(unsigned int frameRate) :
+_frameRate(frameRate),
+_capacity(2000),
+_rate(500),
+_jitter(5, 0),
+_keyFrameRate(1.0),
+_deltaFrameSize(10000, 1e6),
+_counter(0),
+_lossrate(0.0)
+{
+ // Assign to random value between 0 and max of unsigned int
+ _seed = static_cast<unsigned>(std::time(0));
+ std::srand(_seed);
+ _prevTimestamp = static_cast<unsigned int>((std::rand() + 1.0)/(RAND_MAX + 1.0)*(pow((float) 2, (long) sizeof(unsigned int)*8)-1));
+ _prevWallClock = VCMTickTime::MillisecondTimestamp();
+}
+
+FrameSample
+JitterEstimateTest::GenerateFrameSample()
+{
+ double increment = 1.0/_frameRate;
+ unsigned int frameSize = static_cast<unsigned int>(_deltaFrameSize.RandValue());
+ bool keyFrame = false;
+ bool resent = false;
+ _prevTimestamp += static_cast<unsigned int>(90000*increment + 0.5);
+ double deltaFrameRate = _frameRate - _keyFrameRate;
+ double ratio = deltaFrameRate/static_cast<double>(_keyFrameRate);
+ if (ratio < 1.0)
+ {
+ ratio = 1.0/ratio;
+ if (_counter >= ratio)
+ _counter = 0;
+ else
+ {
+ _counter++;
+ frameSize += static_cast<unsigned int>(3*_deltaFrameSize.GetAverage());
+ keyFrame = true;
+ }
+ }
+ else
+ {
+ if (_counter >= ratio)
+ {
+ frameSize += static_cast<unsigned int>(3*_deltaFrameSize.GetAverage());
+ _counter = 0;
+ keyFrame = true;
+ }
+ else
+ _counter++;
+ }
+ WebRtc_Word64 jitter = static_cast<WebRtc_Word64>(_jitter.RandValue() + 1.0/_capacity * frameSize + 0.5);
+ _prevWallClock += static_cast<WebRtc_Word64>(1000*increment + 0.5);
+ double rndValue = RandUniform();
+ resent = (rndValue < _lossrate);
+ //printf("rndValue = %f\n", rndValue);
+ return FrameSample(_prevTimestamp, _prevWallClock + jitter, frameSize, keyFrame, resent);
+}
+
+void
+JitterEstimateTest::SetCapacity(unsigned int c)
+{
+ _capacity = c;
+}
+
+void
+JitterEstimateTest::SetRate(unsigned int r)
+{
+ _rate = r;
+}
+
+void
+JitterEstimateTest::SetJitter(double m, double v)
+{
+ _jitter.SetParams(m, v);
+}
+
+void
+JitterEstimateTest::SetFrameSizeStats(double m, double v)
+{
+ _deltaFrameSize.SetParams(m, v);
+}
+
+void
+JitterEstimateTest::SetKeyFrameRate(int rate)
+{
+ _keyFrameRate = rate;
+}
+
+void
+JitterEstimateTest::SetLossRate(double rate)
+{
+ _lossrate = rate;
+}
diff --git a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h
new file mode 100644
index 0000000..cd7338a
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h
@@ -0,0 +1,105 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
+
+#include "typedefs.h"
+#include "jitter_buffer.h"
+#include "jitter_estimator.h"
+#include <cstdlib>
+#include <cmath>
+
+double const pi = 4*std::atan(1.0);
+
+class GaussDist
+{
+public:
+ GaussDist(double m, double v): _mu(m), _sigma(sqrt(v)) {}
+
+ double RandValue() // returns a single normally distributed number
+ {
+ double r1 = (std::rand() + 1.0)/(RAND_MAX + 1.0); // gives equal distribution in (0, 1]
+ double r2 = (std::rand() + 1.0)/(RAND_MAX + 1.0);
+ return _mu + _sigma * std::sqrt(-2*std::log(r1))*std::cos(2*pi*r2);
+ }
+
+ double GetAverage()
+ {
+ return _mu;
+ }
+
+ double GetVariance()
+ {
+ return _sigma*_sigma;
+ }
+
+ void SetParams(double m, double v)
+ {
+ _mu = m;
+ _sigma = sqrt(v);
+ }
+
+private:
+ double _mu, _sigma;
+};
+
+class JitterEstimateTestWrapper : public webrtc::VCMJitterEstimator
+{
+public:
+ JitterEstimateTestWrapper() : VCMJitterEstimator() {}
+ double GetTheta() { return _theta[0]; }
+ double GetVarNoise() { return _varNoise; }
+};
+
+class FrameSample
+{
+public:
+ FrameSample() {FrameSample(0, 0, 0, false, false);}
+ FrameSample(unsigned int ts, WebRtc_Word64 wallClk, unsigned int fs, bool _keyFrame, bool _resent):
+ timestamp90Khz(ts), wallClockMs(wallClk), frameSize(fs), keyFrame(_keyFrame), resent(_resent) {}
+
+ unsigned int timestamp90Khz;
+ WebRtc_Word64 wallClockMs;
+ unsigned int frameSize;
+ bool keyFrame;
+ bool resent;
+};
+
+class JitterEstimateTest
+{
+public:
+ JitterEstimateTest(unsigned int frameRate);
+ FrameSample GenerateFrameSample();
+ void SetCapacity(unsigned int c);
+ void SetRate(unsigned int r);
+ void SetJitter(double m, double v);
+ void SetFrameSizeStats(double m, double v);
+ void SetKeyFrameRate(int rate);
+ void SetLossRate(double rate);
+
+private:
+ double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
+ unsigned int _frameRate;
+ unsigned int _capacity;
+ unsigned int _rate;
+ GaussDist _jitter;
+ //GaussDist _noResend;
+ GaussDist _deltaFrameSize;
+ unsigned int _prevTimestamp;
+ WebRtc_Word64 _prevWallClock;
+ unsigned int _nextDelay;
+ double _keyFrameRate;
+ unsigned int _counter;
+ unsigned int _seed;
+ double _lossrate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/media_opt_test.cc b/trunk/src/modules/video_coding/main/test/media_opt_test.cc
new file mode 100644
index 0000000..f767919
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/media_opt_test.cc
@@ -0,0 +1,560 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Implementation of Media Optimization Test
+// testing is done via the VCM module, no specific Media opt functionality.
+
+#include "media_opt_test.h"
+
+#include <string.h>
+#include <stdio.h>
+#include <time.h>
+#include <vector>
+
+#include "../source/event.h"
+#include "receiver_tests.h" // receive side callbacks
+#include "test_callbacks.h"
+#include "test_macros.h"
+#include "test_util.h" // send side callback
+#include "testsupport/metrics/video_metrics.h"
+#include "video_coding.h"
+
+
+using namespace webrtc;
+
+int MediaOptTest::RunTest(int testNum, CmdArgs& args)
+{
+ Trace::CreateTrace();
+ Trace::SetTraceFile((test::OutputPath() + "mediaOptTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ TickTimeBase clock;
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ MediaOptTest* mot = new MediaOptTest(vcm, &clock);
+ if (testNum == 0)
+ { // regular
+ mot->Setup(0, args);
+ mot->GeneralSetup();
+ mot->Perform();
+ mot->Print(1);// print to screen
+ mot->TearDown();
+ }
+ if (testNum == 1)
+ { // release test
+ mot->Setup(0, args);
+ mot->RTTest();
+ }
+ if (testNum == 2)
+ { // release test, running from script
+ mot->Setup(1, args);
+ mot->GeneralSetup();
+ mot->Perform();
+ mot->Print(1);// print to screen
+ mot->TearDown();
+ }
+
+ VideoCodingModule::Destroy(vcm);
+ delete mot;
+ Trace::ReturnTrace();
+ return 0;
+
+}
+
+
+MediaOptTest::MediaOptTest(VideoCodingModule* vcm, TickTimeBase* clock):
+_vcm(vcm),
+_clock(clock),
+_width(0),
+_height(0),
+_lengthSourceFrame(0),
+_timeStamp(0),
+_frameRate(30.0f),
+_nackEnabled(false),
+_fecEnabled(false),
+_rttMS(0),
+_bitRate(300.0f),
+_lossRate(0.0f),
+_renderDelayMs(0),
+_frameCnt(0),
+_sumEncBytes(0),
+_numFramesDropped(0),
+_numberOfCores(4)
+{
+ _rtp = RtpRtcp::CreateRtpRtcp(1, false);
+}
+
+MediaOptTest::~MediaOptTest()
+{
+ RtpRtcp::DestroyRtpRtcp(_rtp);
+}
+void
+MediaOptTest::Setup(int testType, CmdArgs& args)
+{
+ /*TEST USER SETTINGS*/
+ // test parameters
+ _inname = args.inputFile;
+ if (args.outputFile == "")
+ _outname = test::OutputPath() + "MOTest_out.vp8";
+ else
+ _outname = args.outputFile;
+ // actual source after frame dropping
+ _actualSourcename = test::OutputPath() + "MOTestSource.yuv";
+ _codecName = args.codecName;
+ _sendCodecType = args.codecType;
+ _width = args.width;
+ _height = args.height;
+ _frameRate = args.frameRate;
+ _bitRate = args.bitRate;
+ _numberOfCores = 4;
+
+ // error resilience
+ _nackEnabled = false;
+ _fecEnabled = true;
+ _nackFecEnabled = false;
+
+ _rttMS = 100;
+ _lossRate = 0.00*255; // no packet loss
+
+ _testType = testType;
+
+ //For multiple runs with script
+ if (_testType == 1)
+ {
+ float rateTest,lossTest;
+ int numRuns;
+ _fpinp = fopen("dat_inp","rb");
+ _fpout = fopen("test_runs/dat_out","ab");
+ _fpout2 = fopen("test_runs/dat_out2","ab");
+ TEST(fscanf(_fpinp,"%f %f %d \n",&rateTest,&lossTest,&numRuns) > 0);
+ _bitRate = rateTest;
+ _lossRate = lossTest;
+ _testNum = 0;
+
+ // for bit rates: 500, 1000, 2000, 3000,4000
+ // for loss rates: 0, 1, 3, 5, 10%
+ _numParRuns = 25;
+
+ _testNum = numRuns + 1;
+ if (rateTest == 0.0) _lossRate = 0.0;
+ else
+ {
+ if (rateTest == 4000) //final bit rate
+ {
+ if (lossTest == 0.1*255) _lossRate = 0.0; //start at 1%
+ else
+ if (lossTest == 0.05*255) _lossRate = 0.1*255; //final loss rate
+ else
+ if (lossTest == 0.0) _lossRate = 0.01*255;
+ else _lossRate = lossTest + 0.02*255;
+ }
+ }
+
+ if (rateTest == 0.0 || rateTest == 4000) _bitRate = 500; //starting bit rate
+ else
+ if (rateTest == 500) _bitRate = 1000;
+ else _bitRate = rateTest + 1000;
+ }
+ //
+
+ _renderDelayMs = 0;
+ /* test settings end*/
+
+ _lengthSourceFrame = 3*_width*_height/2;
+ _log.open((test::OutputPath() + "VCM_MediaOptLog.txt").c_str(),
+ std::fstream::out | std::fstream::app);
+ return;
+}
+
+void
+MediaOptTest::GeneralSetup()
+{
+ WebRtc_UWord32 minPlayoutDelayMs = 0;
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+
+ if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ if ((_actualSourceFile = fopen(_actualSourcename.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _actualSourcename.c_str());
+ exit(1);
+ }
+
+ if (_rtp->InitReceiver() < 0)
+ {
+ exit(1);
+ }
+ if (_rtp->InitSender() < 0)
+ {
+ exit(1);
+ }
+ if (_vcm->InitializeReceiver() < 0)
+ {
+ exit(1);
+ }
+ if (_vcm->InitializeSender())
+ {
+ exit(1);
+ }
+
+ // Registering codecs for the RTP module
+
+ // Register receive and send payload
+ VideoCodec videoCodec;
+ strncpy(videoCodec.plName, "VP8", 32);
+ videoCodec.plType = VCM_VP8_PAYLOAD_TYPE;
+ _rtp->RegisterReceivePayload(videoCodec);
+ _rtp->RegisterSendPayload(videoCodec);
+
+ strncpy(videoCodec.plName, "ULPFEC", 32);
+ videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
+ _rtp->RegisterReceivePayload(videoCodec);
+ _rtp->RegisterSendPayload(videoCodec);
+
+ strncpy(videoCodec.plName, "RED", 32);
+ videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
+ _rtp->RegisterReceivePayload(videoCodec);
+ _rtp->RegisterSendPayload(videoCodec);
+
+ if (_nackFecEnabled == 1)
+ _rtp->SetGenericFECStatus(_nackFecEnabled, VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ else
+ _rtp->SetGenericFECStatus(_fecEnabled, VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+
+ // VCM: Registering codecs
+ VideoCodec sendCodec;
+ _vcm->InitializeSender();
+ _vcm->InitializeReceiver();
+ WebRtc_Word32 numberOfCodecs = _vcm->NumberOfCodecs();
+ if (numberOfCodecs < 1)
+ {
+ exit(1);
+ }
+
+ if (_vcm->Codec(_sendCodecType, &sendCodec) != 0)
+ {
+ printf("Unknown codec\n");
+ exit(1);
+ }
+ // register codec
+ sendCodec.startBitrate = (int) _bitRate;
+ sendCodec.height = _height;
+ sendCodec.width = _width;
+ sendCodec.maxFramerate = (WebRtc_UWord8)_frameRate;
+ _vcm->RegisterSendCodec(&sendCodec, _numberOfCores, 1440);
+ _vcm->RegisterReceiveCodec(&sendCodec, _numberOfCores); // same settings for encode and decode
+
+ _vcm->SetRenderDelay(_renderDelayMs);
+ _vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+
+ return;
+}
+// The following test shall be conducted under release tests
+
+
+
+WebRtc_Word32
+MediaOptTest::Perform()
+{
+ //Setup();
+ EventWrapper* waitEvent = EventWrapper::Create();
+
+ // callback settings
+ VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(_rtp);
+ _vcm->RegisterTransportCallback(encodeCompleteCallback);
+ encodeCompleteCallback->SetCodecType(ConvertCodecType(_codecName.c_str()));
+ encodeCompleteCallback->SetFrameDimensions(_width, _height);
+ // frame ready to be sent to network
+ RTPSendCompleteCallback* outgoingTransport =
+ new RTPSendCompleteCallback(_rtp, _clock);
+ _rtp->RegisterSendTransport(outgoingTransport);
+ //FrameReceiveCallback
+ VCMDecodeCompleteCallback receiveCallback(_decodedFile);
+ RtpDataCallback dataCallback(_vcm);
+ _rtp->RegisterIncomingDataCallback(&dataCallback);
+
+ VideoProtectionCallback protectionCallback;
+ protectionCallback.RegisterRtpModule(_rtp);
+ _vcm->RegisterProtectionCallback(&protectionCallback);
+
+ // set error resilience / test parameters:
+ outgoingTransport->SetLossPct(_lossRate);
+ if (_nackFecEnabled == 1)
+ _vcm->SetVideoProtection(kProtectionNackFEC, _nackFecEnabled);
+ else
+ {
+ _vcm->SetVideoProtection(kProtectionNack, _nackEnabled);
+ _vcm->SetVideoProtection(kProtectionFEC, _fecEnabled);
+ }
+
+ // START TEST
+ VideoFrame sourceFrame;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, (WebRtc_UWord8)_lossRate, _rttMS);
+ _vcm->RegisterReceiveCallback(&receiveCallback);
+
+ _frameCnt = 0;
+ _sumEncBytes = 0.0;
+ _numFramesDropped = 0;
+
+ while (feof(_sourceFile)== 0)
+ {
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ _frameCnt++;
+
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
+ // inform RTP Module of error resilience features
+ //_rtp->SetFECCodeRate(protectionCallback.FECKeyRate(),protectionCallback.FECDeltaRate());
+ //_rtp->SetNACKStatus(protectionCallback.NACKMethod());
+
+ WebRtc_Word32 ret = _vcm->Decode();
+ if (ret < 0 )
+ {
+ TEST(ret == 0);
+ printf ("Decode error in frame # %d",_frameCnt);
+ }
+
+ float encBytes = encodeCompleteCallback->EncodedBytes();
+ if (encBytes == 0)
+ {
+ _numFramesDropped += 1;
+ //printf("frame #%d dropped \n", _frameCnt );
+ }
+ else
+ {
+ // write frame to file
+ fwrite(sourceFrame.Buffer(), 1, sourceFrame.Length(), _actualSourceFile);
+ }
+
+ _sumEncBytes += encBytes;
+ //waitEvent->Wait(33);
+ }
+
+ //END TEST
+ delete waitEvent;
+ delete encodeCompleteCallback;
+ delete outgoingTransport;
+ delete tmpBuffer;
+
+return 0;
+
+}
+
+void
+MediaOptTest::RTTest()
+{
+ // will only calculate PSNR - not create output files for all
+ // SET UP
+ // Set bit rates
+ const float bitRateVec[] = {500, 1000, 2000,3000, 4000};
+ //const float bitRateVec[] = {1000};
+ // Set Packet loss values ([0,255])
+ const double lossPctVec[] = {0.0*255, 0.0*255, 0.01*255, 0.01*255, 0.03*255, 0.03*255, 0.05*255, 0.05*255, 0.1*255, 0.1*255};
+ const bool nackEnabledVec[] = {false , false, false, false, false, false, false, false , false, false};
+ const bool fecEnabledVec[] = {false , true, false, true , false, true , false, true , false, true};
+ // fec and nack are set according to the packet loss values
+
+ const float nBitrates = sizeof(bitRateVec)/sizeof(*bitRateVec);
+ const float nlossPct = sizeof(lossPctVec)/sizeof(*lossPctVec);
+
+ std::vector<const VideoSource*> sources;
+ std::vector<const VideoSource*>::iterator it;
+
+ sources.push_back(new const VideoSource(_inname, _width, _height));
+ int numOfSrc = 1;
+
+ // constant settings (valid for entire run time)
+ _rttMS = 20;
+ _renderDelayMs = 0;
+
+ // same out name for all
+ _outname = test::OutputPath() + "RTMOTest_out.yuv";
+ // actual source after frame dropping
+ _actualSourcename = test::OutputPath() + "RTMOTestSource.yuv";
+
+ _codecName = "VP8"; // for now just this one - later iterate over all codec types
+ _log.open((test::OutputPath() + "/VCM_RTMediaOptLog.txt").c_str(),
+ std::fstream::out | std::fstream::app);
+ _outputRes=fopen((test::OutputPath() + "VCM_MediaOptResults.txt").c_str(),
+ "ab");
+
+ //char filename[128];
+ /* test settings end*/
+
+ // START TEST
+ // iterate over test sequences
+ printf("\n****START TEST OVER ALL RUNS ****\n");
+ int runCnt = 0;
+ for (it = sources.begin() ; it < sources.end(); it++)
+ {
+
+ // test set up
+ _inname = (*it)->GetFileName();
+ _width = (*it)->GetWidth();
+ _height = (*it)->GetHeight();
+ _lengthSourceFrame = 3*_width*_height/2;
+ _frameRate = (*it)->GetFrameRate();
+ //GeneralSetup();
+
+
+ // iterate over all bit rates
+ for (int i = 0; i < nBitrates; i++)
+ {
+ _bitRate = static_cast<float>(bitRateVec[i]);
+ // iterate over all packet loss values
+ for (int j = 0; j < nlossPct; j++)
+ {
+ _lossRate = static_cast<float>(lossPctVec[j]);
+ _nackEnabled = static_cast<bool>(nackEnabledVec[j]);
+ _fecEnabled = static_cast<bool>(fecEnabledVec[j]);
+
+ runCnt++;
+ printf("run #%d out of %d \n", runCnt,(int)(nlossPct*nBitrates*numOfSrc));
+
+ //printf("**FOR RUN: **%d %d %d %d \n",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
+
+ /*
+ int ch = sprintf(filename,"../test_mediaOpt/RTMOTest_%d_%d_%d_%d.yuv",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
+ _outname = filename;
+
+ printf("**FOR RUN: **%d %d %d %d \n",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
+ */
+ if (_rtp != NULL)
+ {
+ RtpRtcp::DestroyRtpRtcp(_rtp);
+ }
+ _rtp = RtpRtcp::CreateRtpRtcp(1, false);
+ GeneralSetup();
+ Perform();
+ Print(1);
+ TearDown();
+ RtpRtcp::DestroyRtpRtcp(_rtp);
+ _rtp = NULL;
+
+ printf("\n");
+ //printf("**DONE WITH RUN: **%d %d %f %d \n",_nackEnabled,_fecEnabled,lossPctVec[j],int(_bitRate));
+ //
+
+ }// end of packet loss loop
+ }// end of bit rate loop
+ delete *it;
+ }// end of video sequence loop
+ // at end of sequence
+ fclose(_outputRes);
+ printf("\nVCM Media Optimization Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+
+
+void
+MediaOptTest::Print(int mode)
+{
+ double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _frameRate));
+ double actualBitRate = ActualBitRate / 1000.0;
+ webrtc::test::QualityMetricsResult psnr;
+ I420PSNRFromFiles(_actualSourcename.c_str(), _outname.c_str(), _width,
+ _height, &psnr);
+
+ (_log) << "VCM: Media Optimization Test Cycle Completed!" << std::endl;
+ (_log) << "Input file: " << _inname << std::endl;
+ (_log) << "Output file:" << _outname << std::endl;
+ ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
+ (_log) << "Error Reslience: NACK:" << _nackEnabled << "; FEC: " << _fecEnabled << std::endl;
+ (_log) << "Packet Loss applied= %f " << _lossRate << std::endl;
+ (_log) << _numFramesDropped << " FRames were dropped" << std::endl;
+ ( _log) << "PSNR: " << psnr.average << std::endl;
+ (_log) << std::endl;
+
+ if (_testType == 2)
+ {
+ fprintf(_outputRes,"************\n");
+ fprintf(_outputRes,"\n\n\n");
+ fprintf(_outputRes,"Actual bitrate: %f kbps\n", actualBitRate);
+ fprintf(_outputRes,"Target bitrate: %f kbps\n", _bitRate);
+ fprintf(_outputRes,"NACK: %s ",(_nackEnabled)?"true":"false");
+ fprintf(_outputRes,"FEC: %s \n ",(_fecEnabled)?"true":"false");
+ fprintf(_outputRes,"Packet loss applied = %f\n", _lossRate);
+ fprintf(_outputRes,"%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
+ fprintf(_outputRes,"PSNR: %f \n", psnr.average);
+ fprintf(_outputRes,"************\n");
+ }
+
+
+ //
+ if (_testType == 1)
+ {
+ fprintf(_fpout,"************\n");
+ fprintf(_fpout,"\n\n\n");
+ fprintf(_fpout,"Actual bitrate: %f kbps\n", actualBitRate);
+ fprintf(_fpout,"Target bitrate: %f kbps\n", _bitRate);
+ fprintf(_fpout,"NACK: %s ",(_nackEnabled)?"true":"false");
+ fprintf(_fpout,"FEC: %s \n ",(_fecEnabled)?"true":"false");
+ fprintf(_fpout,"Packet loss applied = %f\n", _lossRate);
+ fprintf(_fpout,"%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
+ fprintf(_fpout,"PSNR: %f \n", psnr.average);
+ fprintf(_fpout,"************\n");
+
+ int testNum1 = _testNum/(_numParRuns +1) + 1;
+ int testNum2 = _testNum%_numParRuns;
+ if (testNum2 == 0) testNum2 = _numParRuns;
+ fprintf(_fpout2,"%d %d %f %f %f %f \n",testNum1,testNum2,_bitRate,actualBitRate,_lossRate,psnr.average);
+ fclose(_fpinp);
+ _fpinp = fopen("dat_inp","wb");
+ fprintf(_fpinp,"%f %f %d \n",_bitRate,_lossRate,_testNum);
+ }
+ //
+
+
+ if (mode == 1)
+ {
+ // print to screen
+ printf("\n\n\n");
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Target bitrate: %f kbps\n", _bitRate);
+ printf("NACK: %s ",(_nackEnabled)?"true":"false");
+ printf("FEC: %s \n",(_fecEnabled)?"true":"false");
+ printf("Packet loss applied = %f\n", _lossRate);
+ printf("%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
+ printf("PSNR: %f \n", psnr.average);
+ }
+ TEST(psnr.average > 10); // low becuase of possible frame dropping (need to verify that OK for all packet loss values/ rates)
+}
+
+void
+MediaOptTest::TearDown()
+{
+ _log.close();
+ fclose(_sourceFile);
+ fclose(_decodedFile);
+ fclose(_actualSourceFile);
+ return;
+}
diff --git a/trunk/src/modules/video_coding/main/test/media_opt_test.h b/trunk/src/modules/video_coding/main/test/media_opt_test.h
new file mode 100644
index 0000000..7d4e226
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/media_opt_test.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// VCM Media Optimization Test
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
+
+
+#include <string>
+
+#include "rtp_rtcp.h"
+#include "test_util.h"
+#include "video_coding.h"
+#include "video_source.h"
+
+// media optimization test
+// This test simulates a complete encode-decode cycle via the RTP module.
+// allows error resilience tests, packet loss tests, etc.
+// Does not test the media optimization deirectly, but via the VCM API only.
+// The test allows two modes:
+// 1 - Standard, basic settings, one run
+// 2 - Release test - iterates over a number of video sequences, bit rates, packet loss values ,etc.
+
+class MediaOptTest
+{
+public:
+ MediaOptTest(webrtc::VideoCodingModule* vcm,
+ webrtc::TickTimeBase* clock);
+ ~MediaOptTest();
+
+ static int RunTest(int testNum, CmdArgs& args);
+ // perform encode-decode of an entire sequence
+ WebRtc_Word32 Perform();
+ // Set up for a single mode test
+ void Setup(int testType, CmdArgs& args);
+ // General set up - applicable for both modes
+ void GeneralSetup();
+ // Run release testing
+ void RTTest();
+ void TearDown();
+ // mode = 1; will print to screen, otherwise only to log file
+ void Print(int mode);
+
+private:
+
+ webrtc::VideoCodingModule* _vcm;
+ webrtc::RtpRtcp* _rtp;
+ webrtc::TickTimeBase* _clock;
+ std::string _inname;
+ std::string _outname;
+ std::string _actualSourcename;
+ std::fstream _log;
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ FILE* _actualSourceFile;
+ FILE* _outputRes;
+ WebRtc_UWord16 _width;
+ WebRtc_UWord16 _height;
+ WebRtc_UWord32 _lengthSourceFrame;
+ WebRtc_UWord32 _timeStamp;
+ float _frameRate;
+ bool _nackEnabled;
+ bool _fecEnabled;
+ bool _nackFecEnabled;
+ WebRtc_UWord8 _rttMS;
+ float _bitRate;
+ double _lossRate;
+ WebRtc_UWord32 _renderDelayMs;
+ WebRtc_Word32 _frameCnt;
+ float _sumEncBytes;
+ WebRtc_Word32 _numFramesDropped;
+ std::string _codecName;
+ webrtc::VideoCodecType _sendCodecType;
+ WebRtc_Word32 _numberOfCores;
+
+ //for release test#2
+ FILE* _fpinp;
+ FILE* _fpout;
+ FILE* _fpout2;
+ int _testType;
+ int _testNum;
+ int _numParRuns;
+
+}; // end of MediaOptTest class definition
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc b/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc
new file mode 100644
index 0000000..d0a39ad
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc
@@ -0,0 +1,362 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*************************************************
+ *
+ * Testing multi thread - receive and send sides
+ *
+ **************************************************/
+
+#include <string.h>
+
+#include "../source/event.h"
+#include "media_opt_test.h"
+#include "mt_test_common.h"
+#include "receiver_tests.h" // shared RTP state and receive side threads
+#include "rtp_rtcp.h"
+#include "test_macros.h"
+#include "test_util.h" // send side callback
+#include "thread_wrapper.h"
+#include "video_coding.h"
+
+using namespace webrtc;
+
+bool
+MainSenderThread(void* obj)
+{
+ SendSharedState* state = static_cast<SendSharedState*>(obj);
+ EventWrapper& waitEvent = *EventWrapper::Create();
+ // preparing a frame for encoding
+ VideoFrame sourceFrame;
+ WebRtc_Word32 width = state->_args.width;
+ WebRtc_Word32 height = state->_args.height;
+ float frameRate = state->_args.frameRate;
+ WebRtc_Word32 lengthSourceFrame = 3*width*height/2;
+ sourceFrame.VerifyAndAllocate(lengthSourceFrame);
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[lengthSourceFrame];
+
+ if (state->_sourceFile == NULL)
+ {
+ state->_sourceFile = fopen(state->_args.inputFile.c_str(), "rb");
+ if (state->_sourceFile == NULL)
+ {
+ printf ("Error when opening file \n");
+ delete &waitEvent;
+ delete [] tmpBuffer;
+ return false;
+ }
+ }
+ if (feof(state->_sourceFile) == 0)
+ {
+ TEST(fread(tmpBuffer, 1, lengthSourceFrame,state->_sourceFile) > 0 ||
+ feof(state->_sourceFile));
+ state->_frameCnt++;
+ sourceFrame.CopyFrame(lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(height);
+ sourceFrame.SetWidth(width);
+ state->_timestamp += (WebRtc_UWord32)(9e4 / frameRate);
+ sourceFrame.SetTimeStamp(state->_timestamp);
+
+ WebRtc_Word32 ret = state->_vcm.AddVideoFrame(sourceFrame);
+ if (ret < 0)
+ {
+ printf("Add Frame error: %d\n", ret);
+ delete &waitEvent;
+ delete [] tmpBuffer;
+ return false;
+ }
+ waitEvent.Wait(33);
+ }
+
+ delete &waitEvent;
+ delete [] tmpBuffer;
+
+ return true;
+}
+
+bool
+IntSenderThread(void* obj)
+{
+ SendSharedState* state = static_cast<SendSharedState*>(obj);
+ state->_vcm.SetChannelParameters(1000,30,0);
+
+ return true;
+}
+
+
+int MTRxTxTest(CmdArgs& args)
+{
+ /* TEST SETTINGS */
+ std::string inname = args.inputFile;
+ std::string outname;
+ if (args.outputFile == "")
+ outname = test::OutputPath() + "MTRxTxTest_decoded.yuv";
+ else
+ outname = args.outputFile;
+
+ WebRtc_UWord16 width = args.width;
+ WebRtc_UWord16 height = args.height;
+
+ float frameRate = args.frameRate;
+ float bitRate = args.bitRate;
+ WebRtc_Word32 numberOfCores = 1;
+
+ // error resilience/network
+ // Nack support is currently not implemented in this test.
+ bool nackEnabled = false;
+ bool fecEnabled = false;
+ WebRtc_UWord8 rttMS = 20;
+ float lossRate = 0.0*255; // no packet loss
+ WebRtc_UWord32 renderDelayMs = 0;
+ WebRtc_UWord32 minPlayoutDelayMs = 0;
+
+ /* TEST SET-UP */
+
+ // Set up trace
+ Trace::CreateTrace();
+ Trace::SetTraceFile((test::OutputPath() + "MTRxTxTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+ FILE* sourceFile;
+ FILE* decodedFile;
+
+ if ((sourceFile = fopen(inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", inname.c_str());
+ return -1;
+ }
+
+ if ((decodedFile = fopen(outname.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", outname.c_str());
+ return -1;
+ }
+
+ //RTP
+ RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
+ if (rtp->InitReceiver() < 0)
+ {
+ return -1;
+ }
+ if (rtp->InitSender() < 0)
+ {
+ return -1;
+ }
+ // registering codecs for the RTP module
+ VideoCodec videoCodec;
+ strncpy(videoCodec.plName, "ULPFEC", 32);
+ videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
+ TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
+
+ strncpy(videoCodec.plName, "RED", 32);
+ videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
+ TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
+
+ strncpy(videoCodec.plName, args.codecName.c_str(), 32);
+ videoCodec.plType = VCM_VP8_PAYLOAD_TYPE;
+ videoCodec.maxBitrate = 10000;
+ videoCodec.codecType = args.codecType;
+ TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
+ TEST(rtp->RegisterSendPayload(videoCodec) == 0);
+
+ // inform RTP Module of error resilience features
+ TEST(rtp->SetGenericFECStatus(fecEnabled, VCM_RED_PAYLOAD_TYPE, VCM_ULPFEC_PAYLOAD_TYPE) == 0);
+
+ //VCM
+ TickTimeBase clock;
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ if (vcm->InitializeReceiver() < 0)
+ {
+ return -1;
+ }
+ if (vcm->InitializeSender())
+ {
+ return -1;
+ }
+ // registering codecs for the VCM module
+ VideoCodec sendCodec;
+ vcm->InitializeSender();
+ WebRtc_Word32 numberOfCodecs = vcm->NumberOfCodecs();
+ if (numberOfCodecs < 1)
+ {
+ return -1;
+ }
+
+ if (vcm->Codec(args.codecType, &sendCodec) != 0)
+ {
+ // desired codec unavailable
+ printf("Codec not registered\n");
+ return -1;
+ }
+ // register codec
+ sendCodec.startBitrate = (int) bitRate;
+ sendCodec.height = height;
+ sendCodec.width = width;
+ sendCodec.maxFramerate = (WebRtc_UWord8)frameRate;
+ vcm->RegisterSendCodec(&sendCodec, numberOfCores, 1440);
+ vcm->RegisterReceiveCodec(&sendCodec, numberOfCores); // same settings for encode and decode
+
+ vcm->SetRenderDelay(renderDelayMs);
+ vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+
+ // Callback Settings
+
+ PacketRequester packetRequester(*rtp);
+ vcm->RegisterPacketRequestCallback(&packetRequester);
+
+ VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(rtp);
+ vcm->RegisterTransportCallback(encodeCompleteCallback);
+ encodeCompleteCallback->SetCodecType(ConvertCodecType(args.codecName.c_str()));
+ encodeCompleteCallback->SetFrameDimensions(width, height);
+ // frame ready to be sent to network
+ RTPSendCompleteCallback* outgoingTransport =
+ new RTPSendCompleteCallback(rtp, &clock, "dump.rtp");
+ rtp->RegisterSendTransport(outgoingTransport);
+ // FrameReceiveCallback
+ VCMDecodeCompleteCallback receiveCallback(decodedFile);
+ RtpDataCallback dataCallback(vcm);
+ rtp->RegisterIncomingDataCallback(&dataCallback);
+ vcm->RegisterReceiveCallback(&receiveCallback);
+
+ VideoProtectionCallback protectionCallback;
+ vcm->RegisterProtectionCallback(&protectionCallback);
+
+ outgoingTransport->SetLossPct(lossRate);
+ // Nack support is currently not implemented in this test
+ assert(nackEnabled == false);
+ vcm->SetVideoProtection(kProtectionNack, nackEnabled);
+ vcm->SetVideoProtection(kProtectionFEC, fecEnabled);
+
+ // inform RTP Module of error resilience features
+ rtp->SetFECCodeRate(protectionCallback.FECKeyRate(),
+ protectionCallback.FECDeltaRate());
+ rtp->SetNACKStatus(protectionCallback.NACKMethod());
+
+ vcm->SetChannelParameters((WebRtc_UWord32) bitRate,
+ (WebRtc_UWord8) lossRate, rttMS);
+
+ SharedRTPState mtState(*vcm, *rtp); // receive side
+ SendSharedState mtSendState(*vcm, *rtp, args); // send side
+
+ /*START TEST*/
+
+ // Create and start all threads
+ // send side threads
+ ThreadWrapper* mainSenderThread = ThreadWrapper::CreateThread(MainSenderThread,
+ &mtSendState, kNormalPriority, "MainSenderThread");
+ ThreadWrapper* intSenderThread = ThreadWrapper::CreateThread(IntSenderThread,
+ &mtSendState, kNormalPriority, "IntThread");
+
+ if (mainSenderThread != NULL)
+ {
+ unsigned int tid;
+ mainSenderThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start main sender thread\n");
+ return -1;
+ }
+
+ if (intSenderThread != NULL)
+ {
+ unsigned int tid;
+ intSenderThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start sender interference thread\n");
+ return -1;
+ }
+
+ // Receive side threads
+ ThreadWrapper* processingThread = ThreadWrapper::CreateThread(ProcessingThread,
+ &mtState, kNormalPriority, "ProcessingThread");
+ ThreadWrapper* decodeThread = ThreadWrapper::CreateThread(DecodeThread,
+ &mtState, kNormalPriority, "DecodeThread");
+
+ if (processingThread != NULL)
+ {
+ unsigned int tid;
+ processingThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start processing thread\n");
+ return -1;
+ }
+
+ if (decodeThread != NULL)
+ {
+ unsigned int tid;
+ decodeThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start decode thread\n");
+ return -1;
+ }
+
+ EventWrapper& waitEvent = *EventWrapper::Create();
+
+ // Decode for 10 seconds and then tear down and exit.
+ waitEvent.Wait(30000);
+
+ // Tear down
+
+ while (!mainSenderThread->Stop())
+ {
+ ;
+ }
+
+ while (!intSenderThread->Stop())
+ {
+ ;
+ }
+
+
+ while (!processingThread->Stop())
+ {
+ ;
+ }
+
+ while (!decodeThread->Stop())
+ {
+ ;
+ }
+
+ printf("\nVCM MT RX/TX Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+
+ delete &waitEvent;
+ delete mainSenderThread;
+ delete intSenderThread;
+ delete processingThread;
+ delete decodeThread;
+ delete encodeCompleteCallback;
+ delete outgoingTransport;
+ VideoCodingModule::Destroy(vcm);
+ RtpRtcp::DestroyRtpRtcp(rtp);
+ rtp = NULL;
+ vcm = NULL;
+ Trace::ReturnTrace();
+ fclose(decodedFile);
+ printf("Multi-Thread test Done: View output file \n");
+ return 0;
+
+}
+
diff --git a/trunk/src/modules/video_coding/main/test/mt_test_common.cc b/trunk/src/modules/video_coding/main/test/mt_test_common.cc
new file mode 100644
index 0000000..e3ebd97
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/mt_test_common.cc
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "mt_test_common.h"
+
+#include <cmath>
+
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "rtp_dump.h"
+
+namespace webrtc {
+
+TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
+ TickTimeBase* clock,
+ const char* filename):
+RTPSendCompleteCallback(rtp, clock, filename)
+{
+ //
+}
+
+TransportCallback::~TransportCallback()
+{
+ //
+}
+
+int
+TransportCallback::SendPacket(int channel, const void *data, int len)
+{
+ _sendCount++;
+ _totalSentLength += len;
+
+ if (_rtpDump != NULL)
+ {
+ if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
+ {
+ return -1;
+ }
+ }
+
+ bool transmitPacket = true;
+ // Off-line tests, don't drop first Key frame (approx.)
+ if (_sendCount > 20)
+ {
+ transmitPacket = PacketLoss();
+ }
+
+ TickTimeBase clock;
+ int64_t now = clock.MillisecondTimestamp();
+ // Insert outgoing packet into list
+ if (transmitPacket)
+ {
+ RtpPacket* newPacket = new RtpPacket();
+ memcpy(newPacket->data, data, len);
+ newPacket->length = len;
+ // Simulate receive time = network delay + packet jitter
+ // simulated as a Normal distribution random variable with
+ // mean = networkDelay and variance = jitterVar
+ WebRtc_Word32
+ simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
+ sqrt(_jitterVar));
+ newPacket->receiveTime = now + simulatedDelay;
+ _rtpPackets.push_back(newPacket);
+ }
+ return 0;
+}
+
+int
+TransportCallback::TransportPackets()
+{
+ // Are we ready to send packets to the receiver?
+ RtpPacket* packet = NULL;
+ TickTimeBase clock;
+ int64_t now = clock.MillisecondTimestamp();
+
+ while (!_rtpPackets.empty())
+ {
+ // Take first packet in list
+ packet = _rtpPackets.front();
+ WebRtc_Word64 timeToReceive = packet->receiveTime - now;
+ if (timeToReceive > 0)
+ {
+ // No available packets to send
+ break;
+ }
+
+ _rtpPackets.pop_front();
+ // Send to receive side
+ if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
+ packet->length) < 0)
+ {
+ delete packet;
+ packet = NULL;
+ // Will return an error after the first packet that goes wrong
+ return -1;
+ }
+ delete packet;
+ packet = NULL;
+ }
+ return 0; // OK
+}
+
+
+
+bool VCMProcessingThread(void* obj)
+{
+ SharedRTPState* state = static_cast<SharedRTPState*>(obj);
+ if (state->_vcm.TimeUntilNextProcess() <= 0)
+ {
+ if (state->_vcm.Process() < 0)
+ {
+ return false;
+ }
+ }
+ return true;
+}
+
+
+bool VCMDecodeThread(void* obj)
+{
+ SharedRTPState* state = static_cast<SharedRTPState*>(obj);
+ state->_vcm.Decode();
+ return true;
+}
+
+bool TransportThread(void *obj)
+{
+ SharedTransportState* state = static_cast<SharedTransportState*>(obj);
+ state->_transport.TransportPackets();
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/test/mt_test_common.h b/trunk/src/modules/video_coding/main/test/mt_test_common.h
new file mode 100644
index 0000000..438f0be
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/mt_test_common.h
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Common multi-thread functionality across video coding module tests
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
+
+#include "rtp_rtcp.h"
+#include "test_callbacks.h"
+#include "test_util.h"
+#include "video_coding.h"
+
+namespace webrtc {
+
+class SendSharedState
+{
+public:
+ SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
+ CmdArgs args) :
+ _vcm(vcm),
+ _rtp(rtp),
+ _args(args),
+ _sourceFile(NULL),
+ _frameCnt(0),
+ _timestamp(0) {}
+
+ webrtc::VideoCodingModule& _vcm;
+ webrtc::RtpRtcp& _rtp;
+ CmdArgs _args;
+ FILE* _sourceFile;
+ WebRtc_Word32 _frameCnt;
+ WebRtc_Word32 _timestamp;
+};
+
+// MT implementation of the RTPSendCompleteCallback (Transport)
+class TransportCallback:public RTPSendCompleteCallback
+{
+ public:
+ // constructor input: (receive side) rtp module to send encoded data to
+ TransportCallback(webrtc::RtpRtcp* rtp, TickTimeBase* clock,
+ const char* filename = NULL);
+ virtual ~TransportCallback();
+ // Add packets to list
+ // Incorporate network conditions - delay and packet loss
+ // Actual transmission will occur on a separate thread
+ int SendPacket(int channel, const void *data, int len);
+ // Send to the receiver packets which are ready to be submitted
+ int TransportPackets();
+};
+
+class SharedRTPState
+{
+public:
+ SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) :
+ _vcm(vcm),
+ _rtp(rtp) {}
+ webrtc::VideoCodingModule& _vcm;
+ webrtc::RtpRtcp& _rtp;
+};
+
+
+class SharedTransportState
+{
+public:
+ SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport):
+ _rtp(rtp),
+ _transport(transport) {}
+ webrtc::RtpRtcp& _rtp;
+ TransportCallback& _transport;
+};
+
+bool VCMProcessingThread(void* obj);
+bool VCMDecodeThread(void* obj);
+bool TransportThread(void *obj);
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
diff --git a/trunk/src/modules/video_coding/main/test/normal_test.cc b/trunk/src/modules/video_coding/main/test/normal_test.cc
new file mode 100644
index 0000000..923d649
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/normal_test.cc
@@ -0,0 +1,402 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_test.h"
+
+#include <assert.h>
+#include <iostream>
+#include <sstream>
+#include <time.h>
+
+#include "../source/event.h"
+#include "common_types.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "test_callbacks.h"
+#include "test_macros.h"
+#include "test_util.h"
+#include "trace.h"
+#include "testsupport/metrics/video_metrics.h"
+
+using namespace webrtc;
+
+int NormalTest::RunTest(CmdArgs& args)
+{
+#if defined(EVENT_DEBUG)
+ printf("SIMULATION TIME\n");
+ FakeTickTime clock(0);
+#else
+ printf("REAL-TIME\n");
+ TickTimeBase clock;
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (test::OutputPath() + "VCMNormalTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ NormalTest VCMNTest(vcm, &clock);
+ VCMNTest.Perform(args);
+ VideoCodingModule::Destroy(vcm);
+ Trace::ReturnTrace();
+ return 0;
+}
+
+////////////////
+// Callback Implementation
+//////////////
+
+VCMNTEncodeCompleteCallback::VCMNTEncodeCompleteCallback(FILE* encodedFile,
+ NormalTest& test):
+ _encodedFile(encodedFile),
+ _encodedBytes(0),
+ _skipCnt(0),
+ _VCMReceiver(NULL),
+ _seqNo(0),
+ _test(test)
+{
+ //
+}
+VCMNTEncodeCompleteCallback::~VCMNTEncodeCompleteCallback()
+{
+}
+
+void
+VCMNTEncodeCompleteCallback::RegisterTransportCallback(VCMPacketizationCallback* transport)
+{
+}
+
+WebRtc_Word32
+VCMNTEncodeCompleteCallback::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& /*fragmentationHeader*/,
+ const webrtc::RTPVideoHeader* videoHdr)
+
+{
+ // will call the VCMReceiver input packet
+ _frameType = frameType;
+ // writing encodedData into file
+ fwrite(payloadData, 1, payloadSize, _encodedFile);
+ WebRtcRTPHeader rtpInfo;
+ rtpInfo.header.markerBit = true;
+ rtpInfo.type.Video.width = 0;
+ rtpInfo.type.Video.height = 0;
+ switch (_test.VideoType())
+ {
+ case kVideoCodecVP8:
+ rtpInfo.type.Video.codec = kRTPVideoVP8;
+ rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
+ rtpInfo.type.Video.codecHeader.VP8.nonReference =
+ videoHdr->codecHeader.VP8.nonReference;
+ rtpInfo.type.Video.codecHeader.VP8.pictureId =
+ videoHdr->codecHeader.VP8.pictureId;
+ break;
+ case kVideoCodecI420:
+ rtpInfo.type.Video.codec = kRTPVideoI420;
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.sequenceNumber = _seqNo++;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.timestamp = timeStamp;
+ rtpInfo.frameType = frameType;
+ rtpInfo.type.Video.isFirstPacket = true;
+ // Size should also be received from that table, since the payload type
+ // defines the size.
+
+ _encodedBytes += payloadSize;
+ if (payloadSize < 20)
+ {
+ _skipCnt++;
+ }
+ _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
+ return 0;
+}
+void
+VCMNTEncodeCompleteCallback::RegisterReceiverVCM(VideoCodingModule *vcm)
+{
+ _VCMReceiver = vcm;
+ return;
+}
+ WebRtc_Word32
+VCMNTEncodeCompleteCallback::EncodedBytes()
+{
+ return _encodedBytes;
+}
+
+WebRtc_UWord32
+VCMNTEncodeCompleteCallback::SkipCnt()
+{
+ return _skipCnt;
+}
+
+// Decoded Frame Callback Implementation
+VCMNTDecodeCompleCallback::~VCMNTDecodeCompleCallback()
+{
+ if (_decodedFile)
+ fclose(_decodedFile);
+}
+ WebRtc_Word32
+VCMNTDecodeCompleCallback::FrameToRender(webrtc::VideoFrame& videoFrame)
+{
+ if (videoFrame.Width() != _currentWidth ||
+ videoFrame.Height() != _currentHeight)
+ {
+ _currentWidth = videoFrame.Width();
+ _currentHeight = videoFrame.Height();
+ if (_decodedFile != NULL)
+ {
+ fclose(_decodedFile);
+ _decodedFile = NULL;
+ }
+ _decodedFile = fopen(_outname.c_str(), "wb");
+ }
+ fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
+ _decodedBytes+= videoFrame.Length();
+ return VCM_OK;
+}
+
+ WebRtc_Word32
+VCMNTDecodeCompleCallback::DecodedBytes()
+{
+ return _decodedBytes;
+}
+
+ //VCM Normal Test Class implementation
+
+NormalTest::NormalTest(VideoCodingModule* vcm, TickTimeBase* clock)
+:
+_clock(clock),
+_vcm(vcm),
+_sumEncBytes(0),
+_timeStamp(0),
+_totalEncodeTime(0),
+_totalDecodeTime(0),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_totalEncodePipeTime(0),
+_totalDecodePipeTime(0),
+_frameCnt(0),
+_encFrameCnt(0),
+_decFrameCnt(0)
+{
+ //
+}
+
+NormalTest::~NormalTest()
+{
+ //
+}
+void
+NormalTest::Setup(CmdArgs& args)
+{
+ _inname = args.inputFile;
+ _encodedName = test::OutputPath() + "encoded_normaltest.yuv";
+ _width = args.width;
+ _height = args.height;
+ _frameRate = args.frameRate;
+ _bitRate = args.bitRate;
+ if (args.outputFile == "")
+ {
+ std::ostringstream filename;
+ filename << test::OutputPath() << "NormalTest_" <<
+ _width << "x" << _height << "_" << _frameRate << "Hz_P420.yuv";
+ _outname = filename.str();
+ }
+ else
+ {
+ _outname = args.outputFile;
+ }
+ _lengthSourceFrame = 3*_width*_height/2;
+ _videoType = args.codecType;
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+
+ _log.open((test::OutputPath() + "TestLog.txt").c_str(),
+ std::fstream::out | std::fstream::app);
+ return;
+}
+
+WebRtc_Word32
+NormalTest::Perform(CmdArgs& args)
+{
+ Setup(args);
+ EventWrapper* waitEvent = EventWrapper::Create();
+ VideoCodec _sendCodec;//, _receiveCodec; // tmp - sendCodecd used as receive codec
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ TEST(VideoCodingModule::Codec(_videoType, &_sendCodec) == VCM_OK);
+ _sendCodec.startBitrate = (int)_bitRate; // should be later on changed via the API
+ _sendCodec.width = static_cast<WebRtc_UWord16>(_width);
+ _sendCodec.height = static_cast<WebRtc_UWord16>(_height);
+ _sendCodec.maxFramerate = _frameRate;
+ TEST(_vcm->RegisterSendCodec(&_sendCodec, 4, 1400) == VCM_OK);// will also set and init the desired codec
+ // register a decoder (same codec for decoder and encoder )
+ TEST(_vcm->RegisterReceiveCodec(&_sendCodec, 1) == VCM_OK);
+ /* Callback Settings */
+ VCMNTDecodeCompleCallback _decodeCallback(_outname);
+ _vcm->RegisterReceiveCallback(&_decodeCallback);
+ VCMNTEncodeCompleteCallback _encodeCompleteCallback(_encodedFile, *this);
+ _vcm->RegisterTransportCallback(&_encodeCompleteCallback);
+ // encode and decode with the same vcm
+ _encodeCompleteCallback.RegisterReceiverVCM(_vcm);
+ ///////////////////////
+ /// Start Test
+ ///////////////////////
+ VideoFrame sourceFrame;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
+ double startTime = clock()/(double)CLOCKS_PER_SEC;
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 0);
+
+ SendStatsTest sendStats;
+ sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
+ _vcm->RegisterSendStatisticsCallback(&sendStats);
+
+ while (feof(_sourceFile) == 0)
+ {
+#if !defined(EVENT_DEBUG)
+ WebRtc_Word64 processStartTime = _clock->MillisecondTimestamp();
+#endif
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0 ||
+ feof(_sourceFile));
+ _frameCnt++;
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_height);
+ sourceFrame.SetWidth(_width);
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_sendCodec.maxFramerate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+ _encodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC;
+ WebRtc_Word32 ret = _vcm->AddVideoFrame(sourceFrame);
+ double encodeTime = clock()/(double)CLOCKS_PER_SEC - _encodeTimes[int(sourceFrame.TimeStamp())];
+ _totalEncodeTime += encodeTime;
+ if (ret < 0)
+ {
+ printf("Error in AddFrame: %d\n", ret);
+ //exit(1);
+ }
+ _decodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC; // same timestamp value for encode and decode
+ ret = _vcm->Decode();
+ _totalDecodeTime += clock()/(double)CLOCKS_PER_SEC - _decodeTimes[int(sourceFrame.TimeStamp())];
+ if (ret < 0)
+ {
+ printf("Error in Decode: %d\n", ret);
+ //exit(1);
+ }
+ if (_vcm->TimeUntilNextProcess() <= 0)
+ {
+ _vcm->Process();
+ }
+ WebRtc_UWord32 framePeriod = static_cast<WebRtc_UWord32>(1000.0f/static_cast<float>(_sendCodec.maxFramerate) + 0.5f);
+#if defined(EVENT_DEBUG)
+ static_cast<FakeTickTime*>(_clock)->IncrementDebugClock(framePeriod);
+#else
+ WebRtc_Word64 timeSpent = _clock->MillisecondTimestamp() - processStartTime;
+ if (timeSpent < framePeriod)
+ {
+ waitEvent->Wait(framePeriod - timeSpent);
+ }
+#endif
+ }
+ double endTime = clock()/(double)CLOCKS_PER_SEC;
+ _testTotalTime = endTime - startTime;
+ _sumEncBytes = _encodeCompleteCallback.EncodedBytes();
+
+ delete [] tmpBuffer;
+ delete waitEvent;
+ Teardown();
+ Print();
+ return 0;
+}
+
+void
+NormalTest::FrameEncoded(WebRtc_UWord32 timeStamp)
+{
+ _encodeCompleteTime = clock()/(double)CLOCKS_PER_SEC;
+ _encFrameCnt++;
+ _totalEncodePipeTime += _encodeCompleteTime - _encodeTimes[int(timeStamp)];
+
+}
+
+void
+NormalTest::FrameDecoded(WebRtc_UWord32 timeStamp)
+{
+ _decodeCompleteTime = clock()/(double)CLOCKS_PER_SEC;
+ _decFrameCnt++;
+ _totalDecodePipeTime += _decodeCompleteTime - _decodeTimes[timeStamp];
+}
+
+void
+NormalTest::Print()
+{
+ std::cout << "Normal Test Completed!" << std::endl;
+ (_log) << "Normal Test Completed!" << std::endl;
+ (_log) << "Input file: " << _inname << std::endl;
+ (_log) << "Output file: " << _outname << std::endl;
+ (_log) << "Total run time: " << _testTotalTime << std::endl;
+ printf("Total run time: %f s \n", _testTotalTime);
+ double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _frameRate));
+ double actualBitRate = ActualBitRate / 1000.0;
+ double avgEncTime = _totalEncodeTime / _frameCnt;
+ double avgDecTime = _totalDecodeTime / _frameCnt;
+ webrtc::test::QualityMetricsResult psnr, ssim;
+ I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _width, _height,
+ &psnr);
+ I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _width, _height,
+ &ssim);
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Target bitrate: %f kbps\n", _bitRate);
+ ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
+ printf("Average encode time: %f s\n", avgEncTime);
+ ( _log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ printf("Average decode time: %f s\n", avgDecTime);
+ ( _log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+ printf("PSNR: %f \n", psnr.average);
+ ( _log) << "PSNR: " << psnr.average << std::endl;
+ printf("SSIM: %f \n", ssim.average);
+ ( _log) << "SSIM: " << ssim.average << std::endl;
+ (_log) << std::endl;
+
+ printf("\nVCM Normal Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+void
+NormalTest::Teardown()
+{
+ //_log.close();
+ fclose(_sourceFile);
+ fclose(_encodedFile);
+ return;
+}
+
+
+
+
diff --git a/trunk/src/modules/video_coding/main/test/normal_test.h b/trunk/src/modules/video_coding/main/test/normal_test.h
new file mode 100644
index 0000000..6f75dfb
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/normal_test.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
+
+#include "video_coding.h"
+#include "test_util.h"
+
+#include <map>
+
+class NormalTest;
+
+//Send Side - Packetization callback - will create and send a packet to the VCMReceiver
+class VCMNTEncodeCompleteCallback : public webrtc::VCMPacketizationCallback
+{
+public:
+ // constructor input: file in which encoded data will be written
+ VCMNTEncodeCompleteCallback(FILE* encodedFile, NormalTest& test);
+ virtual ~VCMNTEncodeCompleteCallback();
+ // Register transport callback
+ void RegisterTransportCallback(webrtc::VCMPacketizationCallback* transport);
+ // process encoded data received from the encoder, pass stream to the VCMReceiver module
+ WebRtc_Word32 SendData(const webrtc::FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const webrtc::RTPFragmentationHeader& fragmentationHeader,
+ const webrtc::RTPVideoHeader* videoHdr);
+
+ // Register exisitng VCM. Currently - encode and decode with the same vcm module.
+ void RegisterReceiverVCM(webrtc::VideoCodingModule *vcm);
+ // Return sum of encoded data (all frames in the sequence)
+ WebRtc_Word32 EncodedBytes();
+ // return number of encoder-skipped frames
+ WebRtc_UWord32 SkipCnt();;
+ // conversion function for payload type (needed for the callback function)
+// RTPVideoVideoCodecTypes ConvertPayloadType(WebRtc_UWord8 payloadType);
+
+private:
+ FILE* _encodedFile;
+ WebRtc_UWord32 _encodedBytes;
+ WebRtc_UWord32 _skipCnt;
+ webrtc::VideoCodingModule* _VCMReceiver;
+ webrtc::FrameType _frameType;
+ WebRtc_UWord8* _payloadData; // max payload size??
+ WebRtc_UWord16 _seqNo;
+ NormalTest& _test;
+}; // end of VCMEncodeCompleteCallback
+
+class VCMNTDecodeCompleCallback: public webrtc::VCMReceiveCallback
+{
+public:
+ VCMNTDecodeCompleCallback(std::string outname): // or should it get a name?
+ _decodedFile(NULL),
+ _outname(outname),
+ _decodedBytes(0),
+ _currentWidth(0),
+ _currentHeight(0) {}
+ virtual ~VCMNTDecodeCompleCallback();
+ void SetUserReceiveCallback(webrtc::VCMReceiveCallback* receiveCallback);
+ // will write decoded frame into file
+ WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
+ WebRtc_Word32 DecodedBytes();
+private:
+ FILE* _decodedFile;
+ std::string _outname;
+ WebRtc_UWord32 _decodedBytes;
+ WebRtc_UWord32 _currentWidth;
+ WebRtc_UWord32 _currentHeight;
+
+}; // end of VCMDecodeCompleCallback class
+
+
+class NormalTest
+{
+public:
+ NormalTest(webrtc::VideoCodingModule* vcm,
+ webrtc::TickTimeBase* clock);
+ ~NormalTest();
+ static int RunTest(CmdArgs& args);
+ WebRtc_Word32 Perform(CmdArgs& args);
+ // option:: turn into private and call from perform
+ WebRtc_UWord32 Width() const { return _width; };
+ WebRtc_UWord32 Height() const { return _height; };
+ webrtc::VideoCodecType VideoType() const { return _videoType; };
+
+
+protected:
+ // test setup - open files, general initializations
+ void Setup(CmdArgs& args);
+ // close open files, delete used memory
+ void Teardown();
+ // print results to std output and to log file
+ void Print();
+ // calculating pipeline delay, and encoding time
+ void FrameEncoded(WebRtc_UWord32 timeStamp);
+ // calculating pipeline delay, and decoding time
+ void FrameDecoded(WebRtc_UWord32 timeStamp);
+
+ webrtc::TickTimeBase* _clock;
+ webrtc::VideoCodingModule* _vcm;
+ webrtc::VideoCodec _sendCodec;
+ webrtc::VideoCodec _receiveCodec;
+ std::string _inname;
+ std::string _outname;
+ std::string _encodedName;
+ WebRtc_Word32 _sumEncBytes;
+ FILE* _sourceFile;
+ FILE* _decodedFile;
+ FILE* _encodedFile;
+ std::fstream _log;
+ WebRtc_UWord32 _width;
+ WebRtc_UWord32 _height;
+ float _frameRate;
+ float _bitRate;
+ WebRtc_UWord32 _lengthSourceFrame;
+ WebRtc_UWord32 _timeStamp;
+ webrtc::VideoCodecType _videoType;
+ double _totalEncodeTime;
+ double _totalDecodeTime;
+ double _decodeCompleteTime;
+ double _encodeCompleteTime;
+ double _totalEncodePipeTime;
+ double _totalDecodePipeTime;
+ double _testTotalTime;
+ std::map<int, double> _encodeTimes;
+ std::map<int, double> _decodeTimes;
+ WebRtc_Word32 _frameCnt;
+ WebRtc_Word32 _encFrameCnt;
+ WebRtc_Word32 _decFrameCnt;
+
+}; // end of VCMNormalTestClass
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m b/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m
new file mode 100644
index 0000000..d6185f5
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m
@@ -0,0 +1,52 @@
+function plotJitterEstimate(filename)
+
+[timestamps, framedata, slopes, randJitters, framestats, timetable, filtjitter, rtt, rttStatsVec] = jitterBufferTraceParser(filename);
+
+x = 1:size(framestats, 1);
+%figure(2);
+subfigure(3, 2, 1);
+hold on;
+plot(x, slopes(x, 1).*(framestats(x, 1) - framestats(x, 2)) + 3*sqrt(randJitters(x,2)), 'b'); title('Estimate ms');
+plot(x, filtjitter, 'r');
+plot(x, slopes(x, 1).*(framestats(x, 1) - framestats(x, 2)), 'g');
+subfigure(3, 2, 2);
+%subplot(211);
+plot(x, slopes(x, 1)); title('Line slope');
+%subplot(212);
+%plot(x, slopes(x, 2)); title('Line offset');
+subfigure(3, 2, 3); hold on;
+plot(x, framestats); plot(x, framedata(x, 1)); title('frame size and average frame size');
+subfigure(3, 2, 4);
+plot(x, framedata(x, 2)); title('Delay');
+subfigure(3, 2, 5);
+hold on;
+plot(x, randJitters(x,1),'r');
+plot(x, randJitters(x,2)); title('Random jitter');
+
+subfigure(3, 2, 6);
+delays = framedata(:,2);
+dL = [0; framedata(2:end, 1) - framedata(1:end-1, 1)];
+hold on;
+plot(dL, delays, '.');
+s = [min(dL) max(dL)];
+plot(s, slopes(end, 1)*s + slopes(end, 2), 'g');
+plot(s, slopes(end, 1)*s + slopes(end, 2) + 3*sqrt(randJitters(end,2)), 'r');
+plot(s, slopes(end, 1)*s + slopes(end, 2) - 3*sqrt(randJitters(end,2)), 'r');
+title('theta(1)*x+theta(2), (dT-dTS)/dL');
+if sum(size(rttStatsVec)) > 0
+ figure; hold on;
+ rttNstdDevsDrift = 3.5;
+ rttNstdDevsJump = 2.5;
+ rttSamples = rttStatsVec(:, 1);
+ rttAvgs = rttStatsVec(:, 2);
+ rttStdDevs = sqrt(rttStatsVec(:, 3));
+ rttMax = rttStatsVec(:, 4);
+ plot(rttSamples, 'ko-');
+ plot(rttAvgs, 'g');
+ plot(rttAvgs + rttNstdDevsDrift*rttStdDevs, 'b--');
+ plot(rttAvgs + rttNstdDevsJump*rttStdDevs, 'b');
+ plot(rttAvgs - rttNstdDevsJump*rttStdDevs, 'b');
+ plot(rttMax, 'r');
+ %plot(driftRestarts*max(maxRtts), '.');
+ %plot(jumpRestarts*max(maxRtts), '.');
+end
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m b/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m
new file mode 100644
index 0000000..4d262aa
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m
@@ -0,0 +1,213 @@
+function [t, TS] = plotReceiveTrace(filename, flat)
+fid=fopen(filename);
+%DEBUG ; ( 8:32:33:375 | 0) VIDEO:1 ; 5260; First packet of frame 1869537938
+%DEBUG ; ( 8:32:33:375 | 0) VIDEO CODING:1 ; 5260; Decoding timestamp 1869534934
+%DEBUG ; ( 8:32:33:375 | 0) VIDEO:1 ; 5260; Render frame 1869534934 at 20772610
+%DEBUG ; ( 8:32:33:375 | 0) VIDEO CODING:-1 ; 5260; Frame decoded: timeStamp=1870511259 decTime=0 maxDecTime=0, at 19965
+%DEBUG ; ( 7:59:42:500 | 0) VIDEO:-1 ; 2500; Received complete frame timestamp 1870514263 frame type 1 frame size 7862 at time 19965, jitter estimate was 130
+%DEBUG ; ( 8: 5:51:774 | 0) VIDEO:-1 ; 3968; ExtrapolateLocalTime(1870967878)=24971 ms
+
+if nargin == 1
+ flat = 0;
+end
+line = fgetl(fid);
+estimatedArrivalTime = [];
+packetTime = [];
+firstPacketTime = [];
+decodeTime = [];
+decodeCompleteTime = [];
+renderTime = [];
+completeTime = [];
+while ischar(line)%line ~= -1
+ if length(line) == 0
+ line = fgetl(fid);
+ continue;
+ end
+ % Parse the trace line header
+ [tempres, count] = sscanf(line, 'DEBUG ; (%u:%u:%u:%u |%*lu)%13c:');
+ if count < 5
+ line = fgetl(fid);
+ continue;
+ end
+ hr=tempres(1);
+ mn=tempres(2);
+ sec=tempres(3);
+ ms=tempres(4);
+ timeInMs=hr*60*60*1000 + mn*60*1000 + sec*1000 + ms;
+ label = tempres(5:end);
+ I = find(label ~= 32);
+ label = label(I(1):end); % remove white spaces
+ if ~strncmp(char(label), 'VIDEO', 5) & ~strncmp(char(label), 'VIDEO CODING', 12)
+ line = fgetl(fid);
+ continue;
+ end
+ message = line(72:end);
+
+ % Parse message
+ [p, count] = sscanf(message, 'ExtrapolateLocalTime(%lu)=%lu ms');
+ if count == 2
+ estimatedArrivalTime = [estimatedArrivalTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'Packet seqNo %u of frame %lu at %lu');
+ if count == 3
+ packetTime = [packetTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'First packet of frame %lu at %lu');
+ if count == 2
+ firstPacketTime = [firstPacketTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'Decoding timestamp %lu at %lu');
+ if count == 2
+ decodeTime = [decodeTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'Render frame %lu at %lu. Render delay %lu, required delay %lu, max decode time %lu, min total delay %lu');
+ if count == 6
+ renderTime = [renderTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'Frame decoded: timeStamp=%lu decTime=%d maxDecTime=%lu, at %lu');
+ if count == 4
+ decodeCompleteTime = [decodeCompleteTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ [p, count] = sscanf(message, 'Received complete frame timestamp %lu frame type %u frame size %*u at time %lu, jitter estimate was %lu');
+ if count == 4
+ completeTime = [completeTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+
+ line = fgetl(fid);
+end
+fclose(fid);
+
+t = completeTime(:,3);
+TS = completeTime(:,1);
+
+figure;
+subplot(211);
+hold on;
+slope = 0;
+
+if sum(size(packetTime)) > 0
+ % Plot the time when each packet arrives
+ firstTimeStamp = packetTime(1,2);
+ x = (packetTime(:,2) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ firstTime = packetTime(1,3);
+ plot(x, packetTime(:,3) - firstTime - slope, 'b.');
+else
+ % Plot the time when the first packet of a frame arrives
+ firstTimeStamp = firstPacketTime(1,1);
+ x = (firstPacketTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ firstTime = firstPacketTime(1,2);
+ plot(x, firstPacketTime(:,2) - firstTime - slope, 'b.');
+end
+
+% Plot the frame complete time
+if prod(size(completeTime)) > 0
+ x = (completeTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, completeTime(:,3) - firstTime - slope, 'ks');
+end
+
+% Plot the time the decode starts
+if prod(size(decodeTime)) > 0
+ x = (decodeTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, decodeTime(:,2) - firstTime - slope, 'r.');
+end
+
+% Plot the decode complete time
+if prod(size(decodeCompleteTime)) > 0
+ x = (decodeCompleteTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, decodeCompleteTime(:,4) - firstTime - slope, 'g.');
+end
+
+if prod(size(renderTime)) > 0
+ % Plot the wanted render time in ms
+ x = (renderTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, renderTime(:,2) - firstTime - slope, 'c-');
+
+ % Plot the render time if there were no render delay or decoding delay.
+ x = (renderTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, renderTime(:,2) - firstTime - slope - renderTime(:, 3) - renderTime(:, 5), 'c--');
+
+ % Plot the render time if there were no render delay.
+ x = (renderTime(:,1) - firstTimeStamp)/90;
+ if flat
+ slope = x;
+ end
+ plot(x, renderTime(:,2) - firstTime - slope - renderTime(:, 3) - renderTime(:, 5), 'b-');
+end
+
+%plot(x, 90*x, 'r-');
+
+xlabel('RTP timestamp (in ms)');
+ylabel('Time (ms)');
+legend('Packet arrives', 'Frame complete', 'Decode', 'Decode complete', 'Time to render', 'Only jitter', 'Must decode');
+
+% subplot(312);
+% hold on;
+% completeTs = completeTime(:, 1);
+% arrivalTs = estimatedArrivalTime(:, 1);
+% [c, completeIdx, arrivalIdx] = intersect(completeTs, arrivalTs);
+% %plot(completeTs(completeIdx), completeTime(completeIdx, 3) - estimatedArrivalTime(arrivalIdx, 2));
+% timeUntilComplete = completeTime(completeIdx, 3) - estimatedArrivalTime(arrivalIdx, 2);
+% devFromAvgCompleteTime = timeUntilComplete - mean(timeUntilComplete);
+% plot(completeTs(completeIdx) - completeTs(completeIdx(1)), devFromAvgCompleteTime);
+% plot(completeTime(:, 1) - completeTime(1, 1), completeTime(:, 4), 'r');
+% plot(decodeCompleteTime(:, 1) - decodeCompleteTime(1, 1), decodeCompleteTime(:, 2), 'g');
+% plot(decodeCompleteTime(:, 1) - decodeCompleteTime(1, 1), decodeCompleteTime(:, 3), 'k');
+% xlabel('RTP timestamp');
+% ylabel('Time (ms)');
+% legend('Complete time - Estimated arrival time', 'Desired jitter buffer level', 'Actual decode time', 'Max decode time', 0);
+
+if prod(size(renderTime)) > 0
+ subplot(212);
+ hold on;
+ firstTime = renderTime(1, 1);
+ targetDelay = max(renderTime(:, 3) + renderTime(:, 4) + renderTime(:, 5), renderTime(:, 6));
+ plot(renderTime(:, 1) - firstTime, renderTime(:, 3), 'r-');
+ plot(renderTime(:, 1) - firstTime, renderTime(:, 4), 'b-');
+ plot(renderTime(:, 1) - firstTime, renderTime(:, 5), 'g-');
+ plot(renderTime(:, 1) - firstTime, renderTime(:, 6), 'k-');
+ plot(renderTime(:, 1) - firstTime, targetDelay, 'c-');
+ xlabel('RTP timestamp');
+ ylabel('Time (ms)');
+ legend('Render delay', 'Jitter delay', 'Decode delay', 'Extra delay', 'Min total delay');
+end
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/plotTimingTest.m b/trunk/src/modules/video_coding/main/test/plotTimingTest.m
new file mode 100644
index 0000000..52a6f30
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/plotTimingTest.m
@@ -0,0 +1,62 @@
+function plotTimingTest(filename)
+fid=fopen(filename);
+
+%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; Stochastic test 1
+%DEBUG ; ( 9:53:33:859 | 0) VIDEO CODING:-1 ; 7132; Frame decoded: timeStamp=3000 decTime=10 at 10012
+%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; timeStamp=3000 clock=10037 maxWaitTime=0
+%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; timeStampMs=33 renderTime=54
+line = fgetl(fid);
+decTime = [];
+waitTime = [];
+renderTime = [];
+foundStart = 0;
+testName = 'Stochastic test 1';
+while ischar(line)
+ if length(line) == 0
+ line = fgetl(fid);
+ continue;
+ end
+ lineOrig = line;
+ line = line(72:end);
+ if ~foundStart
+ if strncmp(line, testName, length(testName))
+ foundStart = 1;
+ end
+ line = fgetl(fid);
+ continue;
+ end
+ [p, count] = sscanf(line, 'Frame decoded: timeStamp=%lu decTime=%d maxDecTime=%d, at %lu');
+ if count == 4
+ decTime = [decTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+ [p, count] = sscanf(line, 'timeStamp=%u clock=%u maxWaitTime=%u');
+ if count == 3
+ waitTime = [waitTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+ [p, count] = sscanf(line, 'timeStamp=%u renderTime=%u');
+ if count == 2
+ renderTime = [renderTime; p'];
+ line = fgetl(fid);
+ continue;
+ end
+ line = fgetl(fid);
+end
+fclose(fid);
+
+% Compensate for wrap arounds and start counting from zero.
+timeStamps = waitTime(:, 1);
+tsDiff = diff(timeStamps);
+wrapIdx = find(tsDiff < 0);
+timeStamps(wrapIdx+1:end) = hex2dec('ffffffff') + timeStamps(wrapIdx+1:end);
+timeStamps = timeStamps - timeStamps(1);
+
+figure;
+hold on;
+plot(timeStamps, decTime(:, 2), 'r');
+plot(timeStamps, waitTime(:, 3), 'g');
+plot(timeStamps(2:end), diff(renderTime(:, 2)), 'b');
+legend('Decode time', 'Max wait time', 'Render time diff');
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/quality_modes_test.cc b/trunk/src/modules/video_coding/main/test/quality_modes_test.cc
new file mode 100644
index 0000000..e7da691
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/quality_modes_test.cc
@@ -0,0 +1,485 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "quality_modes_test.h"
+
+#include <iostream>
+#include <string>
+#include <time.h>
+
+#include "../source/event.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "test_callbacks.h"
+#include "test_macros.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+using namespace webrtc;
+
+int qualityModeTest()
+{
+ // Don't run this test with debug events.
+#if defined(EVENT_DEBUG)
+ return -1;
+#endif
+ TickTimeBase clock;
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ QualityModesTest QMTest(vcm, &clock);
+ QMTest.Perform();
+ VideoCodingModule::Destroy(vcm);
+ return 0;
+}
+
+
+QualityModesTest::QualityModesTest(VideoCodingModule* vcm,
+ TickTimeBase* clock):
+NormalTest(vcm, clock),
+_vpm()
+{
+ //
+}
+
+
+QualityModesTest::~QualityModesTest()
+{
+ //
+}
+
+void
+QualityModesTest::Setup()
+{
+
+
+ _inname= test::ProjectRootPath() + "resources/crew_30f_4CIF.yuv";
+ _outname = test::OutputPath() + "out_qmtest.yuv";
+ _encodedName = test::OutputPath() + "encoded_qmtest.yuv";
+
+ //NATIVE/SOURCE VALUES
+ _nativeWidth = 2*352;
+ _nativeHeight = 2*288;
+ _nativeFrameRate = 30;
+
+
+ //TARGET/ENCODER VALUES
+ _width = 2*352;
+ _height = 2*288;
+ _frameRate = 30;
+ //
+ _bitRate = 400;
+
+ _flagSSIM = false;
+
+ _lengthSourceFrame = 3*_nativeWidth*_nativeHeight/2;
+
+ if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+ {
+ printf("Cannot read file %s.\n", _inname.c_str());
+ exit(1);
+ }
+ if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write encoded file.\n");
+ exit(1);
+ }
+ if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
+ {
+ printf("Cannot write file %s.\n", _outname.c_str());
+ exit(1);
+ }
+
+ _log.open((test::OutputPath() + "TestLog.txt").c_str(),
+ std::fstream::out | std::fstream::app);
+ return;
+}
+
+
+void
+QualityModesTest::Print()
+{
+ std::cout << "Quality Modes Test Completed!" << std::endl;
+ (_log) << "Quality Modes Test Completed!" << std::endl;
+ (_log) << "Input file: " << _inname << std::endl;
+ (_log) << "Output file: " << _outname << std::endl;
+ (_log) << "Total run time: " << _testTotalTime << std::endl;
+ printf("Total run time: %f s \n", _testTotalTime);
+ double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _nativeFrameRate));
+ double actualBitRate = ActualBitRate / 1000.0;
+ double avgEncTime = _totalEncodeTime / _frameCnt;
+ double avgDecTime = _totalDecodeTime / _frameCnt;
+ webrtc::test::QualityMetricsResult psnr,ssim;
+ I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _nativeWidth,
+ _nativeHeight, &psnr);
+ printf("Actual bitrate: %f kbps\n", actualBitRate);
+ printf("Target bitrate: %f kbps\n", _bitRate);
+ ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
+ printf("Average encode time: %f s\n", avgEncTime);
+ ( _log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+ printf("Average decode time: %f s\n", avgDecTime);
+ ( _log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+ printf("PSNR: %f \n", psnr.average);
+ printf("**Number of frames dropped in VPM***%d \n",_numFramesDroppedVPM);
+ ( _log) << "PSNR: " << psnr.average << std::endl;
+ if (_flagSSIM == 1)
+ {
+ printf("***computing SSIM***\n");
+ I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _nativeWidth,
+ _nativeHeight, &ssim);
+ printf("SSIM: %f \n", ssim.average);
+ }
+ (_log) << std::endl;
+
+ printf("\nVCM Qualit Modes Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+}
+void
+QualityModesTest::Teardown()
+{
+ _log.close();
+ fclose(_sourceFile);
+ fclose(_decodedFile);
+ fclose(_encodedFile);
+ return;
+}
+
+
+WebRtc_Word32
+QualityModesTest::Perform()
+{
+ Setup();
+ // changing bit/frame rate during the test
+ const float bitRateUpdate[] = {1000};
+ const float frameRateUpdate[] = {30};
+ const int updateFrameNum[] = {10000}; // frame numbers at which an update will occur
+
+ WebRtc_UWord32 numChanges = sizeof(updateFrameNum)/sizeof(*updateFrameNum);
+ WebRtc_UWord8 change = 0;// change counter
+
+ _vpm = VideoProcessingModule::Create(1);
+
+ EventWrapper* waitEvent = EventWrapper::Create();
+ VideoCodec codec;//both send and receive
+ _vcm->InitializeReceiver();
+ _vcm->InitializeSender();
+ WebRtc_Word32 NumberOfCodecs = _vcm->NumberOfCodecs();
+ for (int i = 0; i < NumberOfCodecs; i++)
+ {
+ _vcm->Codec(i, &codec);
+ if(strncmp(codec.plName,"VP8" , 5) == 0)
+ {
+ codec.startBitrate = (int)_bitRate;
+ codec.maxFramerate = (WebRtc_UWord8) _frameRate;
+ codec.width = (WebRtc_UWord16)_width;
+ codec.height = (WebRtc_UWord16)_height;
+ TEST(_vcm->RegisterSendCodec(&codec, 2, 1440) == VCM_OK);// will also set and init the desired codec
+ i = NumberOfCodecs;
+ }
+ }
+
+ // register a decoder (same codec for decoder and encoder )
+ TEST(_vcm->RegisterReceiveCodec(&codec, 2) == VCM_OK);
+ /* Callback Settings */
+ VCMQMDecodeCompleCallback _decodeCallback(_decodedFile);
+ _vcm->RegisterReceiveCallback(&_decodeCallback);
+ VCMNTEncodeCompleteCallback _encodeCompleteCallback(_encodedFile, *this);
+ _vcm->RegisterTransportCallback(&_encodeCompleteCallback);
+ // encode and decode with the same vcm
+ _encodeCompleteCallback.RegisterReceiverVCM(_vcm);
+
+ //quality modes callback
+ QMTestVideoSettingsCallback QMCallback;
+ QMCallback.RegisterVCM(_vcm);
+ QMCallback.RegisterVPM(_vpm);
+ _vcm->RegisterVideoQMCallback(&QMCallback);
+
+ ///////////////////////
+ /// Start Test
+ ///////////////////////
+ _vpm->EnableTemporalDecimation(true);
+ _vpm->EnableContentAnalysis(true);
+ _vpm->SetInputFrameResampleMode(kFastRescaling);
+
+ // disabling internal VCM frame dropper
+ _vcm->EnableFrameDropper(false);
+
+ VideoFrame sourceFrame;
+ VideoFrame *decimatedFrame = NULL;
+ sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
+ WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
+ double startTime = clock()/(double)CLOCKS_PER_SEC;
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 0);
+
+ SendStatsTest sendStats;
+ sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
+ _vcm->RegisterSendStatisticsCallback(&sendStats);
+
+ VideoContentMetrics* contentMetrics = NULL;
+ // setting user frame rate
+ _vpm->SetMaxFrameRate((WebRtc_UWord32)(_nativeFrameRate+ 0.5f));
+ // for starters: keeping native values:
+ _vpm->SetTargetResolution(_width, _height, (WebRtc_UWord32)(_frameRate+ 0.5f));
+ _decodeCallback.SetOriginalFrameDimensions(_nativeWidth, _nativeHeight);
+
+ //tmp - disabling VPM frame dropping
+ _vpm->EnableTemporalDecimation(false);
+
+
+ WebRtc_Word32 ret = 0;
+ _numFramesDroppedVPM = 0;
+
+ while (feof(_sourceFile)== 0)
+ {
+ TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
+ _frameCnt++;
+ sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
+ sourceFrame.SetHeight(_nativeHeight);
+ sourceFrame.SetWidth(_nativeWidth);
+
+ _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(codec.maxFramerate));
+ sourceFrame.SetTimeStamp(_timeStamp);
+
+ ret = _vpm->PreprocessFrame(&sourceFrame, &decimatedFrame);
+ if (ret == 1)
+ {
+ printf("VD: frame drop %d \n",_frameCnt);
+ _numFramesDroppedVPM += 1;
+ continue; // frame drop
+ }
+ else if (ret < 0)
+ {
+ printf("Error in PreprocessFrame: %d\n", ret);
+ //exit(1);
+ }
+ contentMetrics = _vpm->ContentMetrics();
+ if (contentMetrics == NULL)
+ {
+ printf("error: contentMetrics = NULL\n");
+ }
+
+ // counting only encoding time
+ _encodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC;
+
+ WebRtc_Word32 ret = _vcm->AddVideoFrame(*decimatedFrame, contentMetrics);
+
+ _totalEncodeTime += clock()/(double)CLOCKS_PER_SEC - _encodeTimes[int(sourceFrame.TimeStamp())];
+
+ if (ret < 0)
+ {
+ printf("Error in AddFrame: %d\n", ret);
+ //exit(1);
+ }
+ _decodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC; // same timestamp value for encode and decode
+ ret = _vcm->Decode();
+ _totalDecodeTime += clock()/(double)CLOCKS_PER_SEC - _decodeTimes[int(sourceFrame.TimeStamp())];
+ if (ret < 0)
+ {
+ printf("Error in Decode: %d\n", ret);
+ //exit(1);
+ }
+ if (_vcm->TimeUntilNextProcess() <= 0)
+ {
+ _vcm->Process();
+ }
+ // mimicking setTargetRates - update every 1 sec
+ // this will trigger QMSelect
+ if (_frameCnt%((int)_frameRate) == 0)
+ {
+ _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 1);
+ waitEvent->Wait(33);
+ }
+ waitEvent->Wait(33);
+ // check for bit rate update
+ if (change < numChanges && _frameCnt == updateFrameNum[change])
+ {
+ _bitRate = bitRateUpdate[change];
+ _frameRate = frameRateUpdate[change];
+ codec.startBitrate = (int)_bitRate;
+ codec.maxFramerate = (WebRtc_UWord8) _frameRate;
+ TEST(_vcm->RegisterSendCodec(&codec, 2, 1440) == VCM_OK);// will also set and init the desired codec
+ change++;
+ }
+ }
+
+ double endTime = clock()/(double)CLOCKS_PER_SEC;
+ _testTotalTime = endTime - startTime;
+ _sumEncBytes = _encodeCompleteCallback.EncodedBytes();
+
+ delete tmpBuffer;
+ delete waitEvent;
+ _vpm->Reset();
+ Teardown();
+ Print();
+ VideoProcessingModule::Destroy(_vpm);
+ return 0;
+}
+
+
+// implementing callback to be called from VCM to update VPM of frame rate and size
+QMTestVideoSettingsCallback::QMTestVideoSettingsCallback():
+_vpm(NULL),
+_vcm(NULL)
+{
+ //
+}
+
+void
+QMTestVideoSettingsCallback::RegisterVPM(VideoProcessingModule *vpm)
+{
+ _vpm = vpm;
+}
+void
+QMTestVideoSettingsCallback::RegisterVCM(VideoCodingModule *vcm)
+{
+ _vcm = vcm;
+}
+
+bool
+QMTestVideoSettingsCallback::Updated()
+{
+ if (_updated)
+ {
+ _updated = false;
+ return true;
+ }
+ return false;
+}
+
+WebRtc_Word32
+QMTestVideoSettingsCallback::SetVideoQMSettings(const WebRtc_UWord32 frameRate,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ WebRtc_Word32 retVal = 0;
+ printf("QM updates: W = %d, H = %d, FR = %d, \n", width, height, frameRate);
+ retVal = _vpm->SetTargetResolution(width, height, frameRate);
+ //Initialize codec with new values - is this the best place to do it?
+ if (!retVal)
+ {
+ // first get current settings
+ VideoCodec currentCodec;
+ _vcm->SendCodec(¤tCodec);
+ // now set new values:
+ currentCodec.height = (WebRtc_UWord16)height;
+ currentCodec.width = (WebRtc_UWord16)width;
+ currentCodec.maxFramerate = (WebRtc_UWord8)frameRate;
+
+ // re-register encoder
+ retVal = _vcm->RegisterSendCodec(¤tCodec, 2, 1440);
+ _updated = true;
+ }
+
+ return retVal;
+}
+
+
+// Decoded Frame Callback Implmentation
+VCMQMDecodeCompleCallback::VCMQMDecodeCompleCallback(FILE* decodedFile):
+_decodedFile(decodedFile),
+_decodedBytes(0),
+//_test(test),
+_origWidth(0),
+_origHeight(0),
+_decWidth(0),
+_decHeight(0),
+//_interpolator(NULL),
+_decBuffer(NULL),
+_frameCnt(0)
+{
+ //
+}
+
+VCMQMDecodeCompleCallback::~VCMQMDecodeCompleCallback()
+ {
+// if (_interpolator != NULL)
+// {
+// deleteInterpolator(_interpolator);
+// _interpolator = NULL;
+// }
+ if (_decBuffer != NULL)
+ {
+ delete [] _decBuffer;
+ _decBuffer = NULL;
+ }
+ }
+WebRtc_Word32
+VCMQMDecodeCompleCallback::FrameToRender(VideoFrame& videoFrame)
+{
+ if ((_origWidth == videoFrame.Width()) && (_origHeight == videoFrame.Height()))
+ {
+ fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
+ _frameCnt++;
+ //printf("frame dec # %d", _frameCnt);
+ // no need for interpolator and decBuffer
+ if (_decBuffer != NULL)
+ {
+ delete [] _decBuffer;
+ _decBuffer = NULL;
+ }
+// if (_interpolator != NULL)
+// {
+// deleteInterpolator(_interpolator);
+// _interpolator = NULL;
+// }
+ _decWidth = 0;
+ _decHeight = 0;
+ }
+ else
+ {
+ if ((_decWidth != videoFrame.Width()) || (_decHeight != videoFrame.Height()))
+ {
+ _decWidth = videoFrame.Width();
+ _decHeight = videoFrame.Height();
+ buildInterpolator();
+ }
+
+// interpolateFrame(_interpolator, videoFrame.Buffer(),_decBuffer);
+ fwrite(_decBuffer, 1, _origWidth*_origHeight*3/2, _decodedFile);
+ _frameCnt++;
+ }
+
+ _decodedBytes += videoFrame.Length();
+ return VCM_OK;
+}
+
+WebRtc_Word32
+VCMQMDecodeCompleCallback::DecodedBytes()
+{
+ return _decodedBytes;
+}
+
+void
+VCMQMDecodeCompleCallback::SetOriginalFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
+{
+ _origWidth = width;
+ _origHeight = height;
+}
+
+WebRtc_Word32
+VCMQMDecodeCompleCallback::buildInterpolator()
+{
+ WebRtc_UWord32 decFrameLength = _origWidth*_origHeight*3 >> 1;
+ if (_decBuffer != NULL)
+ {
+ delete [] _decBuffer;
+ }
+ _decBuffer = new WebRtc_UWord8[decFrameLength];
+ if (_decBuffer == NULL)
+ {
+ return -1;
+ }
+
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/main/test/quality_modes_test.h b/trunk/src/modules/video_coding/main/test/quality_modes_test.h
new file mode 100644
index 0000000..87fa01f
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/quality_modes_test.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
+
+#include "video_processing.h"
+#include "normal_test.h"
+#include "video_coding_defines.h"
+
+int qualityModeTest();
+
+class QualityModesTest : public NormalTest
+{
+public:
+ QualityModesTest(webrtc::VideoCodingModule* vcm,
+ webrtc::TickTimeBase* clock);
+ virtual ~QualityModesTest();
+ WebRtc_Word32 Perform();
+
+private:
+
+ void Setup();
+ void Print();
+ void Teardown();
+ void SsimComp();
+
+ webrtc::VideoProcessingModule* _vpm;
+
+ WebRtc_UWord32 _width;
+ WebRtc_UWord32 _height;
+ float _frameRate;
+ WebRtc_UWord32 _nativeWidth;
+ WebRtc_UWord32 _nativeHeight;
+ float _nativeFrameRate;
+
+ WebRtc_UWord32 _numFramesDroppedVPM;
+ bool _flagSSIM;
+
+}; // end of QualityModesTest class
+
+
+class VCMQMDecodeCompleCallback: public webrtc::VCMReceiveCallback
+{
+public:
+ VCMQMDecodeCompleCallback(FILE* decodedFile);
+ virtual ~VCMQMDecodeCompleCallback();
+ void SetUserReceiveCallback(webrtc::VCMReceiveCallback* receiveCallback);
+ // will write decoded frame into file
+ WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
+ WebRtc_Word32 DecodedBytes();
+ void SetOriginalFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height);
+ WebRtc_Word32 buildInterpolator();
+private:
+ FILE* _decodedFile;
+ WebRtc_UWord32 _decodedBytes;
+ // QualityModesTest& _test;
+ WebRtc_UWord32 _origWidth;
+ WebRtc_UWord32 _origHeight;
+ WebRtc_UWord32 _decWidth;
+ WebRtc_UWord32 _decHeight;
+// VideoInterpolator* _interpolator;
+ WebRtc_UWord8* _decBuffer;
+ WebRtc_UWord32 _frameCnt; // debug
+
+}; // end of VCMQMDecodeCompleCallback class
+
+class QMTestVideoSettingsCallback : public webrtc::VCMQMSettingsCallback
+{
+public:
+ QMTestVideoSettingsCallback();
+ // update VPM with QM settings
+ WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height);
+ // register VPM used by test
+ void RegisterVPM(webrtc::VideoProcessingModule* vpm);
+ void RegisterVCM(webrtc::VideoCodingModule* vcm);
+ bool Updated();
+
+private:
+ webrtc::VideoProcessingModule* _vpm;
+ webrtc::VideoCodingModule* _vcm;
+ bool _updated;
+};
+
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/receiver_tests.h b/trunk/src/modules/video_coding/main/test/receiver_tests.h
new file mode 100644
index 0000000..33d9f5f
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/receiver_tests.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+
+#include "video_coding.h"
+#include "module_common_types.h"
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "typedefs.h"
+#include "rtp_player.h"
+#include "test_util.h"
+
+#include <string>
+#include <stdio.h>
+
+class RtpDataCallback : public webrtc::RtpData
+{
+public:
+ RtpDataCallback(webrtc::VideoCodingModule* vcm)
+ : _vcm(vcm) {};
+
+ virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const webrtc::WebRtcRTPHeader* rtpHeader);
+private:
+ webrtc::VideoCodingModule* _vcm;
+};
+
+class FrameReceiveCallback : public webrtc::VCMReceiveCallback
+{
+public:
+ FrameReceiveCallback(std::string outFilename) :
+ _outFilename(outFilename),
+ _outFile(NULL),
+ _timingFile(NULL) {}
+
+ virtual ~FrameReceiveCallback();
+
+ WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
+
+private:
+ std::string _outFilename;
+ FILE* _outFile;
+ FILE* _timingFile;
+};
+
+class SharedState
+{
+public:
+ SharedState(webrtc::VideoCodingModule& vcm, RTPPlayer& rtpPlayer) :
+ _vcm(vcm),
+ _rtpPlayer(rtpPlayer) {}
+ webrtc::VideoCodingModule& _vcm;
+ RTPPlayer& _rtpPlayer;
+};
+
+
+int RtpPlay(CmdArgs& args);
+int RtpPlayMT(CmdArgs& args,
+ int releaseTest = 0,
+ webrtc::VideoCodecType releaseTestVideoType = webrtc::kVideoCodecVP8);
+int ReceiverTimingTests(CmdArgs& args);
+int JitterBufferTest(CmdArgs& args);
+int DecodeFromStorageTest(CmdArgs& args);
+
+// Thread functions:
+bool ProcessingThread(void* obj);
+bool RtpReaderThread(void* obj);
+bool DecodeThread(void* obj);
+bool NackThread(void* obj);
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
diff --git a/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc b/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc
new file mode 100644
index 0000000..0b09256
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc
@@ -0,0 +1,235 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "receiver_tests.h"
+#include "video_coding.h"
+#include "trace.h"
+#include "../source/event.h"
+#include "../source/internal_defines.h"
+#include "timing.h"
+#include "test_macros.h"
+#include "test_util.h"
+
+#include <cstdio>
+#include <cstdlib>
+#include <cmath>
+
+using namespace webrtc;
+
+float vcmFloatMax(float a, float b)
+{
+ return a > b ? a : b;
+}
+
+float vcmFloatMin(float a, float b)
+{
+ return a < b ? a : b;
+}
+
+double const pi = 4*std::atan(1.0);
+
+class GaussDist
+{
+public:
+ static float RandValue(float m, float stdDev) // returns a single normally distributed number
+ {
+ float r1 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0)); // gives equal distribution in (0, 1]
+ float r2 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0));
+ return m + stdDev * static_cast<float>(std::sqrt(-2*std::log(r1))*std::cos(2*pi*r2));
+ }
+};
+
+int ReceiverTimingTests(CmdArgs& args)
+{
+ // Make sure this test is never executed with simulated events.
+#if defined(EVENT_DEBUG)
+ return -1;
+#endif
+
+ // Set up trace
+ Trace::CreateTrace();
+ Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+ // A static random seed
+ srand(0);
+
+ TickTimeBase clock;
+ VCMTiming timing(&clock);
+ float clockInMs = 0.0;
+ WebRtc_UWord32 waitTime = 0;
+ WebRtc_UWord32 jitterDelayMs = 0;
+ WebRtc_UWord32 maxDecodeTimeMs = 0;
+ WebRtc_UWord32 timeStamp = 0;
+
+ timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
+
+ timing.UpdateCurrentDelay(timeStamp);
+
+ timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
+
+ timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ jitterDelayMs = 20;
+ timing.SetRequiredDelay(jitterDelayMs);
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ // First update initializes the render time. Since we have no decode delay
+ // we get waitTime = renderTime - now - renderDelay = jitter
+ TEST(waitTime == jitterDelayMs);
+
+ jitterDelayMs += VCMTiming::kDelayMaxChangeMsPerS + 10;
+ timeStamp += 90000;
+ clockInMs += 1000.0f;
+ timing.SetRequiredDelay(jitterDelayMs);
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ // Since we gradually increase the delay we only get
+ // 100 ms every second.
+ TEST(waitTime == jitterDelayMs - 10);
+
+ timeStamp += 90000;
+ clockInMs += 1000.0;
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ TEST(waitTime == jitterDelayMs);
+
+ // 300 incoming frames without jitter, verify that this gives the exact wait time
+ for (int i=0; i < 300; i++)
+ {
+ clockInMs += 1000.0f/30.0f;
+ timeStamp += 3000;
+ timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ }
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ TEST(waitTime == jitterDelayMs);
+
+ // Add decode time estimates
+ for (int i=0; i < 10; i++)
+ {
+ WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
+ clockInMs += 10.0f;
+ timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ timeStamp += 3000;
+ clockInMs += 1000.0f/30.0f - 10.0f;
+ timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ }
+ maxDecodeTimeMs = 10;
+ timing.SetRequiredDelay(jitterDelayMs);
+ clockInMs += 1000.0f;
+ timeStamp += 90000;
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ TEST(waitTime == jitterDelayMs);
+
+ WebRtc_UWord32 totalDelay1 = timing.TargetVideoDelay();
+ WebRtc_UWord32 minTotalDelayMs = 200;
+ timing.SetMinimumTotalDelay(minTotalDelayMs);
+ clockInMs += 5000.0f;
+ timeStamp += 5*90000;
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ WebRtc_UWord32 totalDelay2 = timing.TargetVideoDelay();
+ // We should at least have minTotalDelayMs - decodeTime (10) - renderTime (10) to wait
+ TEST(waitTime == minTotalDelayMs - maxDecodeTimeMs - 10);
+ // The total video delay should not increase with the extra delay,
+ // the extra delay should be independent.
+ TEST(totalDelay1 == totalDelay2);
+
+ // Reset min total delay
+ timing.SetMinimumTotalDelay(0);
+ clockInMs += 5000.0f;
+ timeStamp += 5*90000;
+ timing.UpdateCurrentDelay(timeStamp);
+
+ // A sudden increase in timestamp of 2.1 seconds
+ clockInMs += 1000.0f/30.0f;
+ timeStamp += static_cast<WebRtc_UWord32>(2.1*90000 + 0.5);
+ WebRtc_Word64 ret = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+ TEST(ret == -1);
+ timing.Reset();
+
+ // This test produces a trace which can be parsed with plotTimingTest.m. The plot
+ // can be used to see that the timing is reasonable under noise, and that the
+ // gradual transition between delays works as expected.
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "Stochastic test 1");
+
+ jitterDelayMs = 60;
+ maxDecodeTimeMs = 10;
+
+ timeStamp = static_cast<WebRtc_UWord32>(-10000); // To produce a wrap
+ clockInMs = 10000.0f;
+ timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
+
+ float noise = 0.0f;
+ for (int i=0; i < 1400; i++)
+ {
+ if (i == 400)
+ {
+ jitterDelayMs = 30;
+ }
+ else if (i == 700)
+ {
+ jitterDelayMs = 100;
+ }
+ else if (i == 1000)
+ {
+ minTotalDelayMs = 200;
+ timing.SetMinimumTotalDelay(minTotalDelayMs);
+ }
+ else if (i == 1200)
+ {
+ minTotalDelayMs = 0;
+ timing.SetMinimumTotalDelay(minTotalDelayMs);
+ }
+ WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
+ noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 2), -10.0f), 30.0f);
+ clockInMs += 10.0f;
+ timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
+ timeStamp += 3000;
+ clockInMs += 1000.0f/30.0f - 10.0f;
+ noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 8), -15.0f), 15.0f);
+ timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
+ timing.SetRequiredDelay(jitterDelayMs);
+ timing.UpdateCurrentDelay(timeStamp);
+ waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
+ static_cast<WebRtc_Word64>(clockInMs + 0.5));
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "timeStamp=%u clock=%u maxWaitTime=%u", timeStamp,
+ static_cast<WebRtc_UWord32>(clockInMs + 0.5), waitTime);
+
+ WebRtc_Word64 renderTimeMs = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
+
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
+ "timeStamp=%u renderTime=%u",
+ timeStamp,
+ MaskWord64ToUWord32(renderTimeMs));
+ }
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "End Stochastic test 1");
+
+ printf("\nVCM Timing Test: \n\n%i tests completed\n", vcmMacrosTests);
+ if (vcmMacrosErrors > 0)
+ {
+ printf("%i FAILED\n\n", vcmMacrosErrors);
+ }
+ else
+ {
+ printf("ALL PASSED\n\n");
+ }
+
+ Trace::ReturnTrace();
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/main/test/release_test.cc b/trunk/src/modules/video_coding/main/test/release_test.cc
new file mode 100644
index 0000000..8e3a073
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/release_test.cc
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ReleaseTest.h"
+#include "ReceiverTests.h"
+#include "TestMacros.h"
+#include "MediaOptTest.h"
+#include "CodecDataBaseTest.h"
+#include "GenericCodecTest.h"
+
+
+
+
+int ReleaseTest()
+{
+ printf("VCM RELEASE TESTS \n\n");
+
+ // Automatic tests
+
+ printf("Testing receive side timing...\n");
+ TEST(ReceiverTimingTests() == 0);
+
+ printf("Testing jitter buffer...\n");
+ TEST(JitterBufferTest() == 0);
+
+ printf("Testing Codec Data Base...\n");
+ TEST(CodecDBTest() == 0);
+
+ printf("Testing Media Optimization....\n");
+ TEST(VCMMediaOptTest(1) == 0);
+
+ // Tests requiring verification
+
+ printf("Testing Multi thread send-receive....\n");
+ TEST(MTRxTxTest() == 0);
+ printf("Verify by viewing output file MTRxTx_out.yuv \n");
+
+ return 0;
+}
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/release_test.h b/trunk/src/modules/video_coding/main/test/release_test.h
new file mode 100644
index 0000000..2578160
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/release_test.h
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RELEASE_TEST_H
+#define RELEASE_TEST_H
+
+int ReleaseTest();
+int ReleaseTestPart2();
+
+#endif
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/release_test_pt2.cc b/trunk/src/modules/video_coding/main/test/release_test_pt2.cc
new file mode 100644
index 0000000..5ff48e5
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/release_test_pt2.cc
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ReleaseTest.h"
+#include "ReceiverTests.h"
+#include "TestMacros.h"
+#include "MediaOptTest.h"
+#include "CodecDataBaseTest.h"
+#include "GenericCodecTest.h"
+
+
+
+
+int ReleaseTestPart2()
+{
+ printf("Verify that TICK_TIME_DEBUG and EVENT_DEBUG are uncommented");
+ // Tests requiring verification
+
+ printf("Testing Generic Codecs...\n");
+ TEST(VCMGenericCodecTest() == 0);
+ printf("Verify by viewing output file GCTest_out.yuv \n");
+
+ return 0;
+}
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/rtp_player.cc b/trunk/src/modules/video_coding/main/test/rtp_player.cc
new file mode 100644
index 0000000..5361e31
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/rtp_player.cc
@@ -0,0 +1,443 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtp_player.h"
+
+#include <cstdlib>
+#ifdef WIN32
+#include <windows.h>
+#include <Winsock2.h>
+#else
+#include <arpa/inet.h>
+#endif
+
+#include "../source/internal_defines.h"
+#include "gtest/gtest.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "rtp_rtcp.h"
+
+using namespace webrtc;
+
+RawRtpPacket::RawRtpPacket(uint8_t* rtp_data, uint16_t rtp_length)
+ : data(rtp_data),
+ length(rtp_length),
+ resend_time_ms(-1) {
+ data = new uint8_t[length];
+ memcpy(data, rtp_data, length);
+}
+
+RawRtpPacket::~RawRtpPacket() {
+ delete [] data;
+}
+
+LostPackets::LostPackets()
+ : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+ loss_count_(0),
+ debug_file_(NULL),
+ packets_() {
+ debug_file_ = fopen("PacketLossDebug.txt", "w");
+}
+
+LostPackets::~LostPackets() {
+ if (debug_file_) {
+ fclose(debug_file_);
+ }
+ while (!packets_.empty()) {
+ delete packets_.front();
+ packets_.pop_front();
+ }
+ delete crit_sect_;
+}
+
+void LostPackets::AddPacket(RawRtpPacket* packet) {
+ CriticalSectionScoped cs(crit_sect_);
+ packets_.push_back(packet);
+ uint16_t seq_num = (packet->data[2] << 8) + packet->data[3];
+ if (debug_file_ != NULL) {
+ fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_, seq_num);
+ }
+ ++loss_count_;
+}
+
+void LostPackets::SetResendTime(uint16_t resend_seq_num,
+ int64_t resend_time_ms,
+ int64_t now_ms) {
+ CriticalSectionScoped cs(crit_sect_);
+ for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
+ const uint16_t seq_num = ((*it)->data[2] << 8) +
+ (*it)->data[3];
+ if (resend_seq_num == seq_num) {
+ if ((*it)->resend_time_ms + 10 < now_ms) {
+ if (debug_file_ != NULL) {
+ fprintf(debug_file_, "Resend %u at %u\n", seq_num,
+ MaskWord64ToUWord32(resend_time_ms));
+ }
+ (*it)->resend_time_ms = resend_time_ms;
+ }
+ return;
+ }
+ }
+ assert(false);
+}
+
+RawRtpPacket* LostPackets::NextPacketToResend(int64_t timeNow) {
+ CriticalSectionScoped cs(crit_sect_);
+ for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
+ if (timeNow >= (*it)->resend_time_ms && (*it)->resend_time_ms != -1) {
+ RawRtpPacket* packet = *it;
+ it = packets_.erase(it);
+ return packet;
+ }
+ }
+ return NULL;
+}
+
+int LostPackets::NumberOfPacketsToResend() const {
+ CriticalSectionScoped cs(crit_sect_);
+ int count = 0;
+ for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
+ ++it) {
+ if ((*it)->resend_time_ms >= 0) {
+ count++;
+ }
+ }
+ return count;
+}
+
+void LostPackets::SetPacketResent(uint16_t seq_num, int64_t now_ms) {
+ CriticalSectionScoped cs(crit_sect_);
+ if (debug_file_ != NULL) {
+ fprintf(debug_file_, "Resent %u at %u\n", seq_num,
+ MaskWord64ToUWord32(now_ms));
+ }
+}
+
+void LostPackets::Print() const {
+ CriticalSectionScoped cs(crit_sect_);
+ printf("Lost packets: %u\n", loss_count_);
+ printf("Packets waiting to be resent: %u\n",
+ NumberOfPacketsToResend());
+ printf("Packets still lost: %u\n",
+ static_cast<unsigned int>(packets_.size()));
+ printf("Sequence numbers:\n");
+ for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
+ ++it) {
+ uint16_t seq_num = ((*it)->data[2] << 8) + (*it)->data[3];
+ printf("%u, ", seq_num);
+ }
+ printf("\n");
+}
+
+RTPPlayer::RTPPlayer(const char* filename,
+ RtpData* callback,
+ TickTimeBase* clock)
+:
+_clock(clock),
+_rtpModule(*RtpRtcp::CreateRtpRtcp(1, false)),
+_nextRtpTime(0),
+_dataCallback(callback),
+_firstPacket(true),
+_lossRate(0.0f),
+_nackEnabled(false),
+_resendPacketCount(0),
+_noLossStartup(100),
+_endOfFile(false),
+_rttMs(0),
+_firstPacketRtpTime(0),
+_firstPacketTimeMs(0),
+_reorderBuffer(NULL),
+_reordering(false),
+_nextPacket(),
+_nextPacketLength(0),
+_randVec(),
+_randVecPos(0)
+{
+ _rtpFile = fopen(filename, "rb");
+ memset(_nextPacket, 0, sizeof(_nextPacket));
+}
+
+RTPPlayer::~RTPPlayer()
+{
+ RtpRtcp::DestroyRtpRtcp(&_rtpModule);
+ if (_rtpFile != NULL)
+ {
+ fclose(_rtpFile);
+ }
+ if (_reorderBuffer != NULL)
+ {
+ delete _reorderBuffer;
+ _reorderBuffer = NULL;
+ }
+}
+
+WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
+{
+ std::srand(321);
+ for (int i=0; i < RAND_VEC_LENGTH; i++)
+ {
+ _randVec[i] = rand();
+ }
+ _randVecPos = 0;
+ WebRtc_Word32 ret = _rtpModule.SetNACKStatus(kNackOff);
+ if (ret < 0)
+ {
+ return -1;
+ }
+ ret = _rtpModule.InitReceiver();
+ if (ret < 0)
+ {
+ return -1;
+ }
+
+ _rtpModule.InitSender();
+ _rtpModule.SetRTCPStatus(kRtcpNonCompound);
+ _rtpModule.SetTMMBRStatus(true);
+
+ ret = _rtpModule.RegisterIncomingDataCallback(_dataCallback);
+ if (ret < 0)
+ {
+ return -1;
+ }
+ // Register payload types
+ for (PayloadTypeList::const_iterator it = payloadList->begin();
+ it != payloadList->end(); ++it) {
+ PayloadCodecTuple* payloadType = *it;
+ if (payloadType != NULL)
+ {
+ VideoCodec videoCodec;
+ strncpy(videoCodec.plName, payloadType->name.c_str(), 32);
+ videoCodec.plType = payloadType->payloadType;
+ if (_rtpModule.RegisterReceivePayload(videoCodec) < 0)
+ {
+ return -1;
+ }
+ }
+ }
+ if (ReadHeader() < 0)
+ {
+ return -1;
+ }
+ memset(_nextPacket, 0, sizeof(_nextPacket));
+ _nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
+ return 0;
+}
+
+WebRtc_Word32 RTPPlayer::ReadHeader()
+{
+ char firstline[FIRSTLINELEN];
+ if (_rtpFile == NULL)
+ {
+ return -1;
+ }
+ EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, _rtpFile) != NULL);
+ if(strncmp(firstline,"#!rtpplay",9) == 0) {
+ if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
+ printf("ERROR: wrong rtpplay version, must be 1.0\n");
+ return -1;
+ }
+ }
+ else if (strncmp(firstline,"#!RTPencode",11) == 0) {
+ if(strncmp(firstline,"#!RTPencode1.0",14) != 0){
+ printf("ERROR: wrong RTPencode version, must be 1.0\n");
+ return -1;
+ }
+ }
+ else {
+ printf("ERROR: wrong file format of input file\n");
+ return -1;
+ }
+
+ WebRtc_UWord32 start_sec;
+ WebRtc_UWord32 start_usec;
+ WebRtc_UWord32 source;
+ WebRtc_UWord16 port;
+ WebRtc_UWord16 padding;
+
+ EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
+ start_sec=ntohl(start_sec);
+ EXPECT_GT(fread(&start_usec, 4, 1, _rtpFile), 0u);
+ start_usec=ntohl(start_usec);
+ EXPECT_GT(fread(&source, 4, 1, _rtpFile), 0u);
+ source=ntohl(source);
+ EXPECT_GT(fread(&port, 2, 1, _rtpFile), 0u);
+ port=ntohs(port);
+ EXPECT_GT(fread(&padding, 2, 1, _rtpFile), 0u);
+ padding=ntohs(padding);
+ return 0;
+}
+
+WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
+{
+ WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (_clock->MillisecondTimestamp() - _firstPacketTimeMs);
+ if (timeLeft < 0)
+ {
+ return 0;
+ }
+ return static_cast<WebRtc_UWord32>(timeLeft);
+}
+
+WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
+{
+ // Send any packets ready to be resent,
+ RawRtpPacket* resend_packet = _lostPackets.NextPacketToResend(timeNow);
+ while (resend_packet != NULL) {
+ const uint16_t seqNo = (resend_packet->data[2] << 8) +
+ resend_packet->data[3];
+ printf("Resend: %u\n", seqNo);
+ int ret = SendPacket(resend_packet->data, resend_packet->length);
+ delete resend_packet;
+ _resendPacketCount++;
+ if (ret > 0) {
+ _lostPackets.SetPacketResent(seqNo, _clock->MillisecondTimestamp());
+ } else if (ret < 0) {
+ return ret;
+ }
+ resend_packet = _lostPackets.NextPacketToResend(timeNow);
+ }
+
+ // Send any packets from rtp file
+ if (!_endOfFile && (TimeUntilNextPacket() == 0 || _firstPacket))
+ {
+ _rtpModule.Process();
+ if (_firstPacket)
+ {
+ _firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
+ _firstPacketTimeMs = _clock->MillisecondTimestamp();
+ }
+ if (_reordering && _reorderBuffer == NULL)
+ {
+ _reorderBuffer = new RawRtpPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
+ return 0;
+ }
+ WebRtc_Word32 ret = SendPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
+ if (_reordering && _reorderBuffer != NULL)
+ {
+ RawRtpPacket* rtpPacket = _reorderBuffer;
+ _reorderBuffer = NULL;
+ SendPacket(rtpPacket->data, rtpPacket->length);
+ delete rtpPacket;
+ }
+ _firstPacket = false;
+ if (ret < 0)
+ {
+ return ret;
+ }
+ _nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
+ if (_nextPacketLength < 0)
+ {
+ _endOfFile = true;
+ return 0;
+ }
+ else if (_nextPacketLength == 0)
+ {
+ return 0;
+ }
+ }
+ if (_endOfFile && _lostPackets.NumberOfPacketsToResend() == 0)
+ {
+ return 1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
+{
+ if ((_randVec[(_randVecPos++) % RAND_VEC_LENGTH] + 1.0)/(RAND_MAX + 1.0) < _lossRate &&
+ _noLossStartup < 0)
+ {
+ if (_nackEnabled)
+ {
+ const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
+ printf("Throw: %u\n", seqNo);
+ _lostPackets.AddPacket(new RawRtpPacket(rtpData, rtpLen));
+ return 0;
+ }
+ }
+ else if (rtpLen > 0)
+ {
+ WebRtc_Word32 ret = _rtpModule.IncomingPacket(rtpData, rtpLen);
+ if (ret < 0)
+ {
+ return -1;
+ }
+ }
+ if (_noLossStartup >= 0)
+ {
+ _noLossStartup--;
+ }
+ return 1;
+}
+
+WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset)
+{
+ WebRtc_UWord16 length, plen;
+
+ if (fread(&length,2,1,_rtpFile)==0)
+ return(-1);
+ length=ntohs(length);
+
+ if (fread(&plen,2,1,_rtpFile)==0)
+ return(-1);
+ plen=ntohs(plen);
+
+ if (fread(offset,4,1,_rtpFile)==0)
+ return(-1);
+ *offset=ntohl(*offset);
+
+ // Use length here because a plen of 0 specifies rtcp
+ length = (WebRtc_UWord16) (length - HDR_SIZE);
+ if (fread((unsigned short *) rtpdata,1,length,_rtpFile) != length)
+ return(-1);
+
+#ifdef JUNK_DATA
+ // destroy the RTP payload with random data
+ if (plen > 12) { // ensure that we have more than just a header
+ for ( int ix = 12; ix < plen; ix=ix+2 ) {
+ rtpdata[ix>>1] = (short) (rtpdata[ix>>1] + (short) rand());
+ }
+ }
+#endif
+ return plen;
+}
+
+WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, WebRtc_UWord32 rttMs)
+{
+ _nackEnabled = enableNack;
+ _lossRate = lossRate;
+ _rttMs = rttMs;
+ return 0;
+}
+
+WebRtc_Word32 RTPPlayer::SetReordering(bool enabled)
+{
+ _reordering = enabled;
+ return 0;
+}
+
+WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
+{
+ if (sequenceNumbers == NULL)
+ {
+ return 0;
+ }
+ for (int i=0; i < length; i++)
+ {
+ _lostPackets.SetResendTime(sequenceNumbers[i],
+ _clock->MillisecondTimestamp() + _rttMs,
+ _clock->MillisecondTimestamp());
+ }
+ return 0;
+}
+
+void RTPPlayer::Print() const
+{
+ printf("Resent packets: %u\n", _resendPacketCount);
+ _lostPackets.Print();
+}
diff --git a/trunk/src/modules/video_coding/main/test/rtp_player.h b/trunk/src/modules/video_coding/main/test/rtp_player.h
new file mode 100644
index 0000000..eac6ba8
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/rtp_player.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp.h"
+#include "critical_section_wrapper.h"
+#include "video_coding_defines.h"
+#include "modules/video_coding/main/source/tick_time_base.h"
+
+#include <stdio.h>
+#include <list>
+#include <string>
+
+#define HDR_SIZE 8 // rtpplay packet header size in bytes
+#define FIRSTLINELEN 40
+#define RAND_VEC_LENGTH 4096
+
+struct PayloadCodecTuple;
+
+struct RawRtpPacket
+{
+public:
+ RawRtpPacket(WebRtc_UWord8* rtp_data, WebRtc_UWord16 rtp_length);
+ ~RawRtpPacket();
+
+ uint8_t* data;
+ uint16_t length;
+ int64_t resend_time_ms;
+};
+
+typedef std::list<PayloadCodecTuple*> PayloadTypeList;
+typedef std::list<RawRtpPacket*> RtpPacketList;
+typedef RtpPacketList::iterator RtpPacketIterator;
+typedef RtpPacketList::const_iterator ConstRtpPacketIterator;
+
+class LostPackets {
+ public:
+ LostPackets();
+ ~LostPackets();
+
+ void AddPacket(RawRtpPacket* packet);
+ void SetResendTime(uint16_t sequenceNumber,
+ int64_t resendTime,
+ int64_t nowMs);
+ RawRtpPacket* NextPacketToResend(int64_t timeNow);
+ int NumberOfPacketsToResend() const;
+ void SetPacketResent(uint16_t seqNo, int64_t nowMs);
+ void Print() const;
+
+ private:
+ webrtc::CriticalSectionWrapper* crit_sect_;
+ int loss_count_;
+ FILE* debug_file_;
+ RtpPacketList packets_;
+};
+
+struct PayloadCodecTuple
+{
+ PayloadCodecTuple(WebRtc_UWord8 plType, std::string codecName, webrtc::VideoCodecType type) :
+ name(codecName), payloadType(plType), codecType(type) {};
+ const std::string name;
+ const WebRtc_UWord8 payloadType;
+ const webrtc::VideoCodecType codecType;
+};
+
+class RTPPlayer : public webrtc::VCMPacketRequestCallback
+{
+public:
+ RTPPlayer(const char* filename,
+ webrtc::RtpData* callback,
+ webrtc::TickTimeBase* clock);
+ virtual ~RTPPlayer();
+
+ WebRtc_Word32 Initialize(const PayloadTypeList* payloadList);
+ WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
+ WebRtc_UWord32 TimeUntilNextPacket() const;
+ WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);
+ WebRtc_Word32 SetReordering(bool enabled);
+ WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length);
+ void Print() const;
+
+private:
+ WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
+ WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
+ WebRtc_Word32 ReadHeader();
+ webrtc::TickTimeBase* _clock;
+ FILE* _rtpFile;
+ webrtc::RtpRtcp& _rtpModule;
+ WebRtc_UWord32 _nextRtpTime;
+ webrtc::RtpData* _dataCallback;
+ bool _firstPacket;
+ float _lossRate;
+ bool _nackEnabled;
+ LostPackets _lostPackets;
+ WebRtc_UWord32 _resendPacketCount;
+ WebRtc_Word32 _noLossStartup;
+ bool _endOfFile;
+ WebRtc_UWord32 _rttMs;
+ WebRtc_Word64 _firstPacketRtpTime;
+ WebRtc_Word64 _firstPacketTimeMs;
+ RawRtpPacket* _reorderBuffer;
+ bool _reordering;
+ WebRtc_Word16 _nextPacket[8000];
+ WebRtc_Word32 _nextPacketLength;
+ int _randVec[RAND_VEC_LENGTH];
+ int _randVecPos;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
diff --git a/trunk/src/modules/video_coding/main/test/subfigure.m b/trunk/src/modules/video_coding/main/test/subfigure.m
new file mode 100644
index 0000000..eadfcb6
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/subfigure.m
@@ -0,0 +1,30 @@
+function H = subfigure(m, n, p)
+%
+% H = SUBFIGURE(m, n, p)
+%
+% Create a new figure window and adjust position and size such that it will
+% become the p-th tile in an m-by-n matrix of windows. (The interpretation of
+% m, n, and p is the same as for SUBPLOT.
+%
+% Henrik Lundin, 2009-01-19
+%
+
+
+h = figure;
+
+[j, i] = ind2sub([n m], p);
+scrsz = get(0,'ScreenSize'); % get screen size
+%scrsz = [1, 1, 1600, 1200];
+
+taskbarSize = 58;
+windowbarSize = 68;
+windowBorder = 4;
+
+scrsz(2) = scrsz(2) + taskbarSize;
+scrsz(4) = scrsz(4) - taskbarSize;
+
+set(h, 'position', [(j-1)/n * scrsz(3) + scrsz(1) + windowBorder,...
+ (m-i)/m * scrsz(4) + scrsz(2) + windowBorder, ...
+ scrsz(3)/n - (windowBorder + windowBorder),...
+ scrsz(4)/m - (windowbarSize + windowBorder + windowBorder)]);
+
diff --git a/trunk/src/modules/video_coding/main/test/test_callbacks.cc b/trunk/src/modules/video_coding/main/test/test_callbacks.cc
new file mode 100644
index 0000000..1ec0e59
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/test_callbacks.cc
@@ -0,0 +1,509 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test_callbacks.h"
+
+#include <cmath>
+
+#include "modules/video_coding/main/source/tick_time_base.h"
+#include "rtp_dump.h"
+#include "test_macros.h"
+
+namespace webrtc {
+
+/******************************
+ * VCMEncodeCompleteCallback
+ *****************************/
+// Basic callback implementation
+// passes the encoded frame directly to the encoder
+// Packetization callback implementation
+VCMEncodeCompleteCallback::VCMEncodeCompleteCallback(FILE* encodedFile):
+ _encodedFile(encodedFile),
+ _encodedBytes(0),
+ _VCMReceiver(NULL),
+ _seqNo(0),
+ _encodeComplete(false),
+ _width(0),
+ _height(0),
+ _codecType(kRTPVideoNoVideo)
+{
+ //
+}
+VCMEncodeCompleteCallback::~VCMEncodeCompleteCallback()
+{
+}
+
+void
+VCMEncodeCompleteCallback::RegisterTransportCallback(
+ VCMPacketizationCallback* transport)
+{
+}
+
+WebRtc_Word32
+VCMEncodeCompleteCallback::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr)
+{
+ // will call the VCMReceiver input packet
+ _frameType = frameType;
+ // writing encodedData into file
+ fwrite(payloadData, 1, payloadSize, _encodedFile);
+ WebRtcRTPHeader rtpInfo;
+ rtpInfo.header.markerBit = true; // end of frame
+ rtpInfo.type.Video.isFirstPacket = true;
+ rtpInfo.type.Video.codec = _codecType;
+ rtpInfo.type.Video.height = (WebRtc_UWord16)_height;
+ rtpInfo.type.Video.width = (WebRtc_UWord16)_width;
+ switch (_codecType)
+ {
+ case webrtc::kRTPVideoVP8:
+ rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
+ rtpInfo.type.Video.codecHeader.VP8.nonReference =
+ videoHdr->codecHeader.VP8.nonReference;
+ rtpInfo.type.Video.codecHeader.VP8.pictureId =
+ videoHdr->codecHeader.VP8.pictureId;
+ break;
+ case webrtc::kRTPVideoI420:
+ break;
+ default:
+ assert(false);
+ return -1;
+ }
+
+ rtpInfo.header.payloadType = payloadType;
+ rtpInfo.header.sequenceNumber = _seqNo++;
+ rtpInfo.header.ssrc = 0;
+ rtpInfo.header.timestamp = timeStamp;
+ rtpInfo.frameType = frameType;
+ // Size should also be received from that table, since the payload type
+ // defines the size.
+
+ _encodedBytes += payloadSize;
+ // directly to receiver
+ int ret = _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
+ _encodeComplete = true;
+
+ return ret;
+}
+
+float
+VCMEncodeCompleteCallback::EncodedBytes()
+{
+ return _encodedBytes;
+}
+
+bool
+VCMEncodeCompleteCallback::EncodeComplete()
+{
+ if (_encodeComplete)
+ {
+ _encodeComplete = false;
+ return true;
+ }
+ return false;
+}
+
+void
+VCMEncodeCompleteCallback::Initialize()
+{
+ _encodeComplete = false;
+ _encodedBytes = 0;
+ _seqNo = 0;
+ return;
+}
+
+void
+VCMEncodeCompleteCallback::ResetByteCount()
+{
+ _encodedBytes = 0;
+}
+
+/***********************************/
+/* VCMRTPEncodeCompleteCallback */
+/***********************************/
+// Encode Complete callback implementation
+// passes the encoded frame via the RTP module to the decoder
+// Packetization callback implementation
+
+WebRtc_Word32
+VCMRTPEncodeCompleteCallback::SendData(
+ const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr)
+{
+ _frameType = frameType;
+ _encodedBytes+= payloadSize;
+ _encodeComplete = true;
+ return _RTPModule->SendOutgoingData(frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ &fragmentationHeader,
+ videoHdr);
+}
+
+float
+VCMRTPEncodeCompleteCallback::EncodedBytes()
+{
+ // only good for one call - after which will reset value;
+ float tmp = _encodedBytes;
+ _encodedBytes = 0;
+ return tmp;
+ }
+
+bool
+VCMRTPEncodeCompleteCallback::EncodeComplete()
+{
+ if (_encodeComplete)
+ {
+ _encodeComplete = false;
+ return true;
+ }
+ return false;
+}
+
+// Decoded Frame Callback Implementation
+
+WebRtc_Word32
+VCMDecodeCompleteCallback::FrameToRender(VideoFrame& videoFrame)
+{
+ fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
+ _decodedBytes+= videoFrame.Length();
+ return VCM_OK;
+ }
+
+WebRtc_Word32
+VCMDecodeCompleteCallback::DecodedBytes()
+{
+ return _decodedBytes;
+}
+
+RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp,
+ TickTimeBase* clock,
+ const char* filename):
+ _clock(clock),
+ _sendCount(0),
+ _rtp(rtp),
+ _lossPct(0),
+ _burstLength(0),
+ _networkDelayMs(0),
+ _jitterVar(0),
+ _prevLossState(0),
+ _totalSentLength(0),
+ _rtpPackets(),
+ _rtpDump(NULL)
+{
+ if (filename != NULL)
+ {
+ _rtpDump = RtpDump::CreateRtpDump();
+ _rtpDump->Start(filename);
+ }
+}
+
+RTPSendCompleteCallback::~RTPSendCompleteCallback()
+{
+ if (_rtpDump != NULL)
+ {
+ _rtpDump->Stop();
+ RtpDump::DestroyRtpDump(_rtpDump);
+ }
+ // Delete remaining packets
+ while (!_rtpPackets.empty())
+ {
+ // Take first packet in list
+ delete _rtpPackets.front();
+ _rtpPackets.pop_front();
+ }
+}
+
+int
+RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
+{
+ _sendCount++;
+ _totalSentLength += len;
+
+ if (_rtpDump != NULL)
+ {
+ if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
+ {
+ return -1;
+ }
+ }
+
+ bool transmitPacket = true;
+ transmitPacket = PacketLoss();
+
+ WebRtc_UWord64 now = _clock->MillisecondTimestamp();
+ // Insert outgoing packet into list
+ if (transmitPacket)
+ {
+ RtpPacket* newPacket = new RtpPacket();
+ memcpy(newPacket->data, data, len);
+ newPacket->length = len;
+ // Simulate receive time = network delay + packet jitter
+ // simulated as a Normal distribution random variable with
+ // mean = networkDelay and variance = jitterVar
+ WebRtc_Word32
+ simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
+ sqrt(_jitterVar));
+ newPacket->receiveTime = now + simulatedDelay;
+ _rtpPackets.push_back(newPacket);
+ }
+
+ // Are we ready to send packets to the receiver?
+ RtpPacket* packet = NULL;
+
+ while (!_rtpPackets.empty())
+ {
+ // Take first packet in list
+ packet = _rtpPackets.front();
+ WebRtc_Word64 timeToReceive = packet->receiveTime - now;
+ if (timeToReceive > 0)
+ {
+ // No available packets to send
+ break;
+ }
+
+ _rtpPackets.pop_front();
+ // Send to receive side
+ if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
+ packet->length) < 0)
+ {
+ delete packet;
+ packet = NULL;
+ // Will return an error after the first packet that goes wrong
+ return -1;
+ }
+ delete packet;
+ packet = NULL;
+ }
+ return len; // OK
+}
+
+int
+RTPSendCompleteCallback::SendRTCPPacket(int channel, const void *data, int len)
+{
+ // Incorporate network conditions
+ return SendPacket(channel, data, len);
+}
+
+void
+RTPSendCompleteCallback::SetLossPct(double lossPct)
+{
+ _lossPct = lossPct;
+ return;
+}
+
+void
+RTPSendCompleteCallback::SetBurstLength(double burstLength)
+{
+ _burstLength = burstLength;
+ return;
+}
+
+bool
+RTPSendCompleteCallback::PacketLoss()
+{
+ bool transmitPacket = true;
+ if (_burstLength <= 1.0)
+ {
+ // Random loss: if _burstLength parameter is not set, or <=1
+ if (UnifomLoss(_lossPct))
+ {
+ // drop
+ transmitPacket = false;
+ }
+ }
+ else
+ {
+ // Simulate bursty channel (Gilbert model)
+ // (1st order) Markov chain model with memory of the previous/last
+ // packet state (loss or received)
+
+ // 0 = received state
+ // 1 = loss state
+
+ // probTrans10: if previous packet is lost, prob. to -> received state
+ // probTrans11: if previous packet is lost, prob. to -> loss state
+
+ // probTrans01: if previous packet is received, prob. to -> loss state
+ // probTrans00: if previous packet is received, prob. to -> received
+
+ // Map the two channel parameters (average loss rate and burst length)
+ // to the transition probabilities:
+ double probTrans10 = 100 * (1.0 / _burstLength);
+ double probTrans11 = (100.0 - probTrans10);
+ double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct)));
+
+ // Note: Random loss (Bernoulli) model is a special case where:
+ // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
+
+ if (_prevLossState == 0 )
+ {
+ // previous packet was received
+ if (UnifomLoss(probTrans01))
+ {
+ // drop, update previous state to loss
+ _prevLossState = 1;
+ transmitPacket = false;
+ }
+ }
+ else if (_prevLossState == 1)
+ {
+ _prevLossState = 0;
+ // previous packet was lost
+ if (UnifomLoss(probTrans11))
+ {
+ // drop, update previous state to loss
+ _prevLossState = 1;
+ transmitPacket = false;
+ }
+ }
+ }
+ return transmitPacket;
+}
+
+
+bool
+RTPSendCompleteCallback::UnifomLoss(double lossPct)
+{
+ double randVal = (std::rand() + 1.0)/(RAND_MAX + 1.0);
+ return randVal < lossPct/100;
+}
+
+WebRtc_Word32
+PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers,
+ WebRtc_UWord16 length)
+{
+ return _rtp.SendNACK(sequenceNumbers, length);
+}
+
+WebRtc_Word32
+SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate,
+ const WebRtc_UWord32 frameRate)
+{
+ TEST(frameRate <= _frameRate);
+ TEST(bitRate > 0 && bitRate < 100000);
+ printf("VCM 1 sec: Bit rate: %u\tFrame rate: %u\n", bitRate, frameRate);
+ return 0;
+}
+
+WebRtc_Word32 KeyFrameReqTest::RequestKeyFrame() {
+ printf("Key frame requested\n");
+ return 0;
+}
+
+
+VideoProtectionCallback::VideoProtectionCallback():
+_deltaFECRate(0),
+_keyFECRate(0),
+_deltaUseUepProtection(0),
+_keyUseUepProtection(0),
+_nack(kNackOff)
+{
+ //
+}
+
+VideoProtectionCallback::~VideoProtectionCallback()
+{
+ //
+}
+
+WebRtc_Word32
+VideoProtectionCallback::ProtectionRequest(WebRtc_UWord8 deltaFECRate,
+ WebRtc_UWord8 keyFECRate,
+ bool deltaUseUepProtection,
+ bool keyUseUepProtection,
+ bool nack_enabled,
+ WebRtc_UWord32* sent_video_rate_bps,
+ WebRtc_UWord32* sent_nack_rate_bps,
+ WebRtc_UWord32* sent_fec_rate_bps)
+{
+ _deltaFECRate = deltaFECRate;
+ _keyFECRate = keyFECRate;
+ _deltaUseUepProtection = deltaUseUepProtection;
+ _keyUseUepProtection = keyUseUepProtection;
+ if (nack_enabled)
+ {
+ _nack = kNackRtcp;
+ }
+ else
+ {
+ _nack = kNackOff;
+ }
+
+ // Update RTP
+ if (_rtp->SetFECCodeRate(keyFECRate, deltaFECRate) != 0)
+ {
+ printf("Error in Setting FEC rate\n");
+ return -1;
+
+ }
+ if (_rtp->SetFECUepProtection(keyUseUepProtection,
+ deltaUseUepProtection) != 0)
+ {
+ printf("Error in Setting FEC UEP protection\n");
+ return -1;
+ }
+ return 0;
+
+}
+NACKMethod
+VideoProtectionCallback::NACKMethod()
+{
+ return _nack;
+}
+
+WebRtc_UWord8
+VideoProtectionCallback::FECDeltaRate()
+{
+ return _deltaFECRate;
+}
+
+WebRtc_UWord8
+VideoProtectionCallback::FECKeyRate()
+{
+ return _keyFECRate;
+}
+
+bool
+VideoProtectionCallback::FECDeltaUepProtection()
+{
+ return _deltaUseUepProtection;
+}
+
+bool
+VideoProtectionCallback::FECKeyUepProtection()
+{
+ return _keyUseUepProtection;
+}
+
+void
+RTPFeedbackCallback::OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 bitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs)
+{
+
+ _vcm->SetChannelParameters(bitrateBps / 1000, fractionLost,
+ (WebRtc_UWord8)roundTripTimeMs);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/test/test_callbacks.h b/trunk/src/modules/video_coding/main/test/test_callbacks.h
new file mode 100644
index 0000000..07820bb
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/test_callbacks.h
@@ -0,0 +1,279 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_CALLBACKS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_CALLBACKS_H_
+
+/*
+ * Declaration of general callbacks that are used throughout VCM's offline tests
+ */
+
+
+#include <string.h>
+#include <cstdlib>
+#include <fstream>
+#include <list>
+
+#include "module_common_types.h"
+#include "rtp_rtcp.h"
+#include "test_util.h"
+#include "trace.h"
+#include "video_coding.h"
+
+namespace webrtc
+{
+class RtpDump;
+
+// Send Side - Packetization callback - send an encoded frame to the VCMReceiver
+class VCMEncodeCompleteCallback: public VCMPacketizationCallback
+{
+public:
+ // Constructor input: file in which encoded data will be written
+ VCMEncodeCompleteCallback(FILE* encodedFile);
+ virtual ~VCMEncodeCompleteCallback();
+ // Register transport callback
+ void RegisterTransportCallback(VCMPacketizationCallback* transport);
+ // Process encoded data received from the encoder, pass stream to the
+ // VCMReceiver module
+ WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr);
+ // Register exisitng VCM. Currently - encode and decode under same module.
+ void RegisterReceiverVCM(VideoCodingModule *vcm) {_VCMReceiver = vcm;}
+ // Return size of last encoded frame data (all frames in the sequence)
+ // Good for only one call - after which will reset value
+ // (to allow detection of frame drop)
+ float EncodedBytes();
+ // Return encode complete (true/false)
+ bool EncodeComplete();
+ // Inform callback of codec used
+ void SetCodecType(RTPVideoCodecTypes codecType)
+ {_codecType = codecType;}
+ // Inform callback of frame dimensions
+ void SetFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
+ {
+ _width = width;
+ _height = height;
+ }
+ // Initialize callback data
+ void Initialize();
+ void ResetByteCount();
+
+ // Conversion function for payload type (needed for the callback function)
+
+private:
+ FILE* _encodedFile;
+ float _encodedBytes;
+ VideoCodingModule* _VCMReceiver;
+ FrameType _frameType;
+ WebRtc_UWord8* _payloadData;
+ WebRtc_UWord16 _seqNo;
+ bool _encodeComplete;
+ WebRtc_Word32 _width;
+ WebRtc_Word32 _height;
+ RTPVideoCodecTypes _codecType;
+
+}; // end of VCMEncodeCompleteCallback
+
+// Send Side - Packetization callback - packetize an encoded frame via the
+// RTP module
+class VCMRTPEncodeCompleteCallback: public VCMPacketizationCallback
+{
+public:
+ VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) :
+ _encodedBytes(0),
+ _seqNo(0),
+ _encodeComplete(false),
+ _RTPModule(rtp) {}
+
+ virtual ~VCMRTPEncodeCompleteCallback() {}
+ // Process encoded data received from the encoder, pass stream to the
+ // RTP module
+ WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader& fragmentationHeader,
+ const RTPVideoHeader* videoHdr);
+ // Return size of last encoded frame. Value good for one call
+ // (resets to zero after call to inform test of frame drop)
+ float EncodedBytes();
+ // Return encode complete (true/false)
+ bool EncodeComplete();
+ // Inform callback of codec used
+ void SetCodecType(RTPVideoCodecTypes codecType)
+ {_codecType = codecType;}
+
+ // Inform callback of frame dimensions
+ void SetFrameDimensions(WebRtc_Word16 width, WebRtc_Word16 height)
+ {
+ _width = width;
+ _height = height;
+ }
+
+private:
+ float _encodedBytes;
+ FrameType _frameType;
+ WebRtc_UWord8* _payloadData;
+ WebRtc_UWord16 _seqNo;
+ bool _encodeComplete;
+ RtpRtcp* _RTPModule;
+ WebRtc_Word16 _width;
+ WebRtc_Word16 _height;
+ RTPVideoCodecTypes _codecType;
+}; // end of VCMEncodeCompleteCallback
+
+// Decode Complete callback
+// Writes the decoded frames to a given file.
+class VCMDecodeCompleteCallback: public VCMReceiveCallback
+{
+public:
+ VCMDecodeCompleteCallback(FILE* decodedFile) :
+ _decodedFile(decodedFile), _decodedBytes(0) {}
+ virtual ~VCMDecodeCompleteCallback() {}
+ // Write decoded frame into file
+ WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
+ WebRtc_Word32 DecodedBytes();
+private:
+ FILE* _decodedFile;
+ WebRtc_UWord32 _decodedBytes;
+}; // end of VCMDecodeCompleCallback class
+
+// Transport callback
+// Called by the RTP Sender - simulates sending packets through a network to the
+// RTP receiver. User can set network conditions as: RTT, packet loss,
+// burst length and jitter.
+class RTPSendCompleteCallback: public Transport
+{
+public:
+ // Constructor input: (receive side) rtp module to send encoded data to
+ RTPSendCompleteCallback(RtpRtcp* rtp, TickTimeBase* clock,
+ const char* filename = NULL);
+ virtual ~RTPSendCompleteCallback();
+ // Send Packet to receive side RTP module
+ virtual int SendPacket(int channel, const void *data, int len);
+ // Send RTCP Packet to receive side RTP module
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+ // Set percentage of channel loss in the network
+ void SetLossPct(double lossPct);
+ // Set average size of burst loss
+ void SetBurstLength(double burstLength);
+ // Set network delay in the network
+ void SetNetworkDelay(WebRtc_UWord32 networkDelayMs)
+ {_networkDelayMs = networkDelayMs;};
+ // Set Packet jitter delay
+ void SetJitterVar(WebRtc_UWord32 jitterVar)
+ {_jitterVar = jitterVar;};
+ // Return send count
+ int SendCount() {return _sendCount; }
+ // Return accumulated length in bytes of transmitted packets
+ WebRtc_UWord32 TotalSentLength() {return _totalSentLength;}
+protected:
+ // Randomly decide whether to drop packets, based on the channel model
+ bool PacketLoss();
+ // Random uniform loss model
+ bool UnifomLoss(double lossPct);
+
+ TickTimeBase* _clock;
+ WebRtc_UWord32 _sendCount;
+ RtpRtcp* _rtp;
+ double _lossPct;
+ double _burstLength;
+ WebRtc_UWord32 _networkDelayMs;
+ double _jitterVar;
+ bool _prevLossState;
+ WebRtc_UWord32 _totalSentLength;
+ std::list<RtpPacket*> _rtpPackets;
+ RtpDump* _rtpDump;
+};
+
+// Request re-transmission of packets (NACK)
+class PacketRequester: public VCMPacketRequestCallback
+{
+public:
+ PacketRequester(RtpRtcp& rtp) :
+ _rtp(rtp) {}
+ WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
+ WebRtc_UWord16 length);
+private:
+ webrtc::RtpRtcp& _rtp;
+};
+
+// Key frame request
+class KeyFrameReqTest: public VCMFrameTypeCallback
+{
+public:
+ WebRtc_Word32 RequestKeyFrame();
+};
+
+
+// VCM statistics
+class SendStatsTest: public webrtc::VCMSendStatisticsCallback
+{
+public:
+ SendStatsTest() : _frameRate(15) {}
+ WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
+ const WebRtc_UWord32 frameRate);
+ void SetTargetFrameRate(WebRtc_UWord32 frameRate) {_frameRate = frameRate;}
+private:
+ WebRtc_UWord32 _frameRate;
+};
+
+// Protection callback - allows the VCM (media optimization) to inform the RTP
+// module of the required protection(FEC rates/settings and NACK mode).
+class VideoProtectionCallback: public VCMProtectionCallback
+{
+public:
+ VideoProtectionCallback();
+ virtual ~VideoProtectionCallback();
+ void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;}
+ WebRtc_Word32 ProtectionRequest(WebRtc_UWord8 deltaFECRate,
+ WebRtc_UWord8 keyFECRate,
+ bool deltaUseUepProtection,
+ bool keyUseUepProtection,
+ bool nack_enabled,
+ WebRtc_UWord32* sent_video_rate_bps,
+ WebRtc_UWord32* sent_nack_rate_bps,
+ WebRtc_UWord32* sent_fec_rate_bps);
+ enum NACKMethod NACKMethod();
+ WebRtc_UWord8 FECDeltaRate();
+ WebRtc_UWord8 FECKeyRate();
+ bool FECDeltaUepProtection();
+ bool FECKeyUepProtection();
+private:
+ RtpRtcp* _rtp;
+ WebRtc_UWord8 _deltaFECRate;
+ WebRtc_UWord8 _keyFECRate;
+ bool _deltaUseUepProtection;
+ bool _keyUseUepProtection;
+ enum NACKMethod _nack;
+};
+
+// Feed back from the RTP Module callback
+class RTPFeedbackCallback : public RtpVideoFeedback {
+ public:
+ RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
+ void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+ const FrameType type,
+ const WebRtc_UWord8 streamIdx) {};
+
+ void OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 bitrateBps,
+ const WebRtc_UWord8 fractionLost,
+ const WebRtc_UWord16 roundTripTimeMs);
+
+ private:
+ VideoCodingModule* _vcm;
+};
+
+} // namespace webrtc
+
+#endif
diff --git a/trunk/src/modules/video_coding/main/test/test_macros.h b/trunk/src/modules/video_coding/main/test/test_macros.h
new file mode 100644
index 0000000..31693b5
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/test_macros.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VCM_TEST_MACROS_H
+#define VCM_TEST_MACROS_H
+
+#include <cstdio>
+#include <cstdlib>
+
+extern int vcmMacrosTests;
+extern int vcmMacrosErrors;
+
+#define PRINT_ERR_MSG(msg) \
+ do { \
+ fprintf(stderr, "Error at line %i of %s\n%s", \
+ __LINE__, __FILE__, msg); \
+ } while(0)
+
+#define TEST(expr) \
+ do { \
+ vcmMacrosTests++; \
+ if (!(expr)) { \
+ PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
+ vcmMacrosErrors++; \
+ } \
+ } while(0)
+
+#define TEST_EXIT_ON_FAIL(expr) \
+ do { \
+ vcmMacrosTests++; \
+ if (!(expr)) { \
+ PRINT_ERR_MSG("Assertion failed: " #expr "\nExiting...\n\n"); \
+ vcmMacrosErrors++; \
+ exit(EXIT_FAILURE); \
+ } \
+ } while(0)
+
+#endif
diff --git a/trunk/src/modules/video_coding/main/test/test_util.cc b/trunk/src/modules/video_coding/main/test/test_util.cc
new file mode 100644
index 0000000..1f89168
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/test_util.cc
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test_util.h"
+#include "test_macros.h"
+#include "rtp_dump.h"
+#include <cmath>
+
+using namespace webrtc;
+
+// Normal Distribution
+#define PI 3.14159265
+double
+NormalDist(double mean, double stdDev)
+{
+ // Creating a Normal distribution variable from two independent uniform
+ // variables based on the Box-Muller transform
+ double uniform1 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
+ double uniform2 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
+ return (mean + stdDev * sqrt(-2 * log(uniform1)) * cos(2 * PI * uniform2));
+}
+
+RTPVideoCodecTypes
+ConvertCodecType(const char* plname)
+{
+ if (strncmp(plname,"VP8" , 3) == 0)
+ {
+ return kRTPVideoVP8;
+ }
+ else if (strncmp(plname,"I420" , 5) == 0)
+ {
+ return kRTPVideoI420;
+ }
+ else
+ {
+ return kRTPVideoNoVideo; // Default value
+ }
+}
+
diff --git a/trunk/src/modules/video_coding/main/test/test_util.h b/trunk/src/modules/video_coding/main/test/test_util.h
new file mode 100644
index 0000000..d705434
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/test_util.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
+
+/*
+ * General declarations used through out VCM offline tests.
+ */
+
+#include <string.h>
+#include <fstream>
+#include <cstdlib>
+
+#include "module_common_types.h"
+#include "testsupport/fileutils.h"
+
+// Class used for passing command line arguments to tests
+class CmdArgs
+{
+ public:
+ CmdArgs()
+ : codecName("VP8"),
+ codecType(webrtc::kVideoCodecVP8),
+ width(352),
+ height(288),
+ bitRate(500),
+ frameRate(30),
+ packetLoss(0),
+ rtt(0),
+ protectionMode(0),
+ camaEnable(0),
+ inputFile(webrtc::test::ProjectRootPath() +
+ "/resources/foreman_cif.yuv"),
+ outputFile(webrtc::test::OutputPath() +
+ "video_coding_test_output_352x288.yuv"),
+ testNum(11) {}
+ std::string codecName;
+ webrtc::VideoCodecType codecType;
+ int width;
+ int height;
+ int bitRate;
+ int frameRate;
+ int packetLoss;
+ int rtt;
+ int protectionMode;
+ int camaEnable;
+ std::string inputFile;
+ std::string outputFile;
+ int testNum;
+};
+
+// forward declaration
+int MTRxTxTest(CmdArgs& args);
+double NormalDist(double mean, double stdDev);
+
+struct RtpPacket {
+ WebRtc_Word8 data[1650]; // max packet size
+ WebRtc_Word32 length;
+ WebRtc_Word64 receiveTime;
+};
+
+
+// Codec type conversion
+webrtc::RTPVideoCodecTypes
+ConvertCodecType(const char* plname);
+
+#endif
diff --git a/trunk/src/modules/video_coding/main/test/tester_main.cc b/trunk/src/modules/video_coding/main/test/tester_main.cc
new file mode 100644
index 0000000..e5d7cd3
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/tester_main.cc
@@ -0,0 +1,208 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "receiver_tests.h"
+#include "normal_test.h"
+#include "codec_database_test.h"
+#include "generic_codec_test.h"
+#include "../source/event.h"
+#include "media_opt_test.h"
+#include "quality_modes_test.h"
+#include "test_util.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#ifdef _WIN32
+//#include "vld.h"
+#endif
+
+using namespace webrtc;
+
+/*
+ * Build with EVENT_DEBUG defined
+ * to build the tests with simulated events.
+ */
+
+int vcmMacrosTests = 0;
+int vcmMacrosErrors = 0;
+
+int ParseArguments(int argc, char **argv, CmdArgs& args)
+{
+ int i = 1;
+ while (i < argc)
+ {
+ if (argv[i][0] != '-')
+ {
+ return -1;
+ }
+ switch (argv[i][1])
+ {
+ case 'w':
+ {
+ int w = atoi(argv[i+1]);
+ if (w < 1)
+ return -1;
+ args.width = w;
+ break;
+ }
+ case 'h':
+ {
+ int h = atoi(argv[i+1]);
+ if (h < 1)
+ return -1;
+ args.height = h;
+ break;
+ }
+ case 'b':
+ {
+ int b = atoi(argv[i+1]);
+ if (b < 1)
+ return -1;
+ args.bitRate = b;
+ break;
+ }
+ case 'f':
+ {
+ int f = atoi(argv[i+1]);
+ if (f < 1)
+ return -1;
+ args.frameRate = f;
+ break;
+ }
+ case 'c':
+ {
+ // TODO(holmer): This should be replaced with a map if more codecs
+ // are added
+ args.codecName = argv[i+1];
+ if (strncmp(argv[i+1], "VP8", 3) == 0)
+ {
+ args.codecType = kVideoCodecVP8;
+ }
+ else if (strncmp(argv[i+1], "I420", 4) == 0)
+ {
+ args.codecType = kVideoCodecI420;
+ }
+ else
+ return -1;
+
+ break;
+ }
+ case 'i':
+ {
+ args.inputFile = argv[i+1];
+ break;
+ }
+ case 'o':
+ args.outputFile = argv[i+1];
+ break;
+ case 'n':
+ {
+ int n = atoi(argv[i+1]);
+ if (n < 1)
+ return -1;
+ args.testNum = n;
+ break;
+ }
+ case 'p':
+ {
+ args.packetLoss = atoi(argv[i+1]);
+ break;
+ }
+ case 'r':
+ {
+ args.rtt = atoi(argv[i+1]);
+ break;
+ }
+ case 'm':
+ {
+ args.protectionMode = atoi(argv[i+1]);
+ break;
+ }
+ case 'e':
+ {
+ args.camaEnable = atoi(argv[i+1]);
+ break;
+ }
+ default:
+ return -1;
+ }
+ i += 2;
+ }
+ return 0;
+}
+
+int main(int argc, char **argv)
+{
+ CmdArgs args;
+
+ if (ParseArguments(argc, argv, args) != 0)
+ {
+ printf("Unable to parse input arguments\n");
+ printf("args: -n <test #> -w <width> -h <height> -f <fps> -b <bps> "
+ "-c <codec> -i <input file> -o <output file> -p <packet loss> "
+ "-r <round-trip-time> -e <cama enable> -m <protection mode> \n");
+ return -1;
+ }
+
+ int ret = 0;
+ switch (args.testNum)
+ {
+ case 1:
+ ret = NormalTest::RunTest(args);
+ break;
+ case 2:
+ ret = MTRxTxTest(args);
+ break;
+ case 3:
+ ret = GenericCodecTest::RunTest(args);
+ break;
+ case 4:
+ ret = CodecDataBaseTest::RunTest(args);
+ break;
+ case 5:
+ // 0- normal, 1-Release test(50 runs) 2- from file
+ ret = MediaOptTest::RunTest(0, args);
+ break;
+ case 6:
+ ret = ReceiverTimingTests(args);
+ break;
+ case 7:
+ ret = RtpPlay(args);
+ break;
+ case 8:
+ ret = RtpPlayMT(args);
+ break;
+ case 9:
+ ret = JitterBufferTest(args);
+ break;
+ case 10:
+ ret = DecodeFromStorageTest(args);
+ break;
+ case 11:
+ ret = NormalTest::RunTest(args);
+ ret |= CodecDataBaseTest::RunTest(args);
+ ret |= ReceiverTimingTests(args);
+ ret |= JitterBufferTest(args);
+ break;
+ default:
+ ret = -1;
+ break;
+ }
+ if (ret != 0)
+ {
+ printf("Test failed!\n");
+ return -1;
+ }
+ return 0;
+}
+
+
+
diff --git a/trunk/src/modules/video_coding/main/test/video_rtp_play.cc b/trunk/src/modules/video_coding/main/test/video_rtp_play.cc
new file mode 100644
index 0000000..577408f
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/video_rtp_play.cc
@@ -0,0 +1,202 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "receiver_tests.h"
+#include "video_coding.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+#include "../source/event.h"
+#include "../source/internal_defines.h"
+#include "test_macros.h"
+#include "rtp_player.h"
+#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+
+#include <stdio.h>
+#include <string.h>
+
+using namespace webrtc;
+
+WebRtc_Word32
+RtpDataCallback::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader)
+{
+ return _vcm->IncomingPacket(payloadData, payloadSize, *rtpHeader);
+}
+
+FrameReceiveCallback::~FrameReceiveCallback()
+{
+ if (_timingFile != NULL)
+ {
+ fclose(_timingFile);
+ }
+ if (_outFile != NULL)
+ {
+ fclose(_outFile);
+ }
+}
+
+WebRtc_Word32
+FrameReceiveCallback::FrameToRender(VideoFrame& videoFrame)
+{
+ if (_timingFile == NULL)
+ {
+ _timingFile = fopen((test::OutputPath() + "renderTiming.txt").c_str(),
+ "w");
+ if (_timingFile == NULL)
+ {
+ return -1;
+ }
+ }
+ if (_outFile == NULL)
+ {
+ _outFile = fopen(_outFilename.c_str(), "wb");
+ if (_outFile == NULL)
+ {
+ return -1;
+ }
+ }
+ fprintf(_timingFile, "%u, %u\n",
+ videoFrame.TimeStamp(),
+ MaskWord64ToUWord32(videoFrame.RenderTimeMs()));
+ fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _outFile);
+ return 0;
+}
+
+int RtpPlay(CmdArgs& args)
+{
+ // Make sure this test isn't executed without simulated events.
+#if !defined(EVENT_DEBUG)
+ return -1;
+#endif
+ // BEGIN Settings
+
+ bool protectionEnabled = false;
+ VCMVideoProtection protectionMethod = kProtectionNack;
+ WebRtc_UWord32 rttMS = 0;
+ float lossRate = 0.0f;
+ bool reordering = false;
+ WebRtc_UWord32 renderDelayMs = 0;
+ WebRtc_UWord32 minPlayoutDelayMs = 0;
+ const WebRtc_Word64 MAX_RUNTIME_MS = -1;
+ std::string outFile = args.outputFile;
+ if (outFile == "")
+ outFile = test::OutputPath() + "RtpPlay_decoded.yuv";
+ FrameReceiveCallback receiveCallback(outFile);
+ FakeTickTime clock(0);
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ RtpDataCallback dataCallback(vcm);
+ RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback, &clock);
+
+
+ PayloadTypeList payloadTypes;
+ payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
+ kVideoCodecVP8));
+
+ Trace::CreateTrace();
+ Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ // END Settings
+
+ // Set up
+
+ WebRtc_Word32 ret = vcm->InitializeReceiver();
+ if (ret < 0)
+ {
+ return -1;
+ }
+ vcm->RegisterReceiveCallback(&receiveCallback);
+ vcm->RegisterPacketRequestCallback(&rtpStream);
+
+ // Register receive codecs in VCM
+ for (PayloadTypeList::iterator it = payloadTypes.begin();
+ it != payloadTypes.end(); ++it) {
+ PayloadCodecTuple* payloadType = *it;
+ if (payloadType != NULL)
+ {
+ VideoCodec codec;
+ if (VideoCodingModule::Codec(payloadType->codecType, &codec) < 0)
+ {
+ return -1;
+ }
+ codec.plType = payloadType->payloadType;
+ if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
+ {
+ return -1;
+ }
+ }
+ }
+
+ if (rtpStream.Initialize(&payloadTypes) < 0)
+ {
+ return -1;
+ }
+ bool nackEnabled = protectionEnabled &&
+ (protectionMethod == kProtectionNack ||
+ protectionMethod == kProtectionDualDecoder);
+ rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
+ rtpStream.SetReordering(reordering);
+ vcm->SetChannelParameters(0, 0, rttMS);
+ vcm->SetVideoProtection(protectionMethod, protectionEnabled);
+ vcm->SetRenderDelay(renderDelayMs);
+ vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+
+ ret = 0;
+
+ // RTP stream main loop
+ while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
+ {
+ if (clock.MillisecondTimestamp() % 5 == 0)
+ {
+ ret = vcm->Decode();
+ if (ret < 0)
+ {
+ return -1;
+ }
+ }
+ while (vcm->DecodeDualFrame(0) == 1);
+ if (vcm->TimeUntilNextProcess() <= 0)
+ {
+ vcm->Process();
+ }
+ if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >=
+ MAX_RUNTIME_MS)
+ {
+ break;
+ }
+ clock.IncrementDebugClock(1);
+ }
+
+ switch (ret)
+ {
+ case 1:
+ printf("Success\n");
+ break;
+ case -1:
+ printf("Failed\n");
+ break;
+ case 0:
+ printf("Timeout\n");
+ break;
+ }
+
+ rtpStream.Print();
+
+ // Tear down
+ while (!payloadTypes.empty())
+ {
+ delete payloadTypes.front();
+ payloadTypes.pop_front();
+ }
+ delete vcm;
+ vcm = NULL;
+ Trace::ReturnTrace();
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc b/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc
new file mode 100644
index 0000000..21b9f73
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc
@@ -0,0 +1,261 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "receiver_tests.h"
+#include "video_coding.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "../source/event.h"
+#include "test_macros.h"
+#include "rtp_player.h"
+
+#include <string.h>
+
+using namespace webrtc;
+
+bool ProcessingThread(void* obj)
+{
+ SharedState* state = static_cast<SharedState*>(obj);
+ if (state->_vcm.TimeUntilNextProcess() <= 0)
+ {
+ if (state->_vcm.Process() < 0)
+ {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool RtpReaderThread(void* obj)
+{
+ SharedState* state = static_cast<SharedState*>(obj);
+ EventWrapper& waitEvent = *EventWrapper::Create();
+ // RTP stream main loop
+ TickTimeBase clock;
+ if (state->_rtpPlayer.NextPacket(clock.MillisecondTimestamp()) < 0)
+ {
+ return false;
+ }
+ waitEvent.Wait(state->_rtpPlayer.TimeUntilNextPacket());
+ delete &waitEvent;
+ return true;
+}
+
+bool DecodeThread(void* obj)
+{
+ SharedState* state = static_cast<SharedState*>(obj);
+ state->_vcm.Decode(10000);
+ while (state->_vcm.DecodeDualFrame(0) == 1);
+ return true;
+}
+
+int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTestVideoType)
+{
+ // Don't run these tests with debug events.
+#if defined(EVENT_DEBUG)
+ return -1;
+#endif
+
+ // BEGIN Settings
+
+ bool protectionEnabled = true;
+ VCMVideoProtection protection = kProtectionDualDecoder;
+ WebRtc_UWord8 rttMS = 50;
+ float lossRate = 0.05f;
+ WebRtc_UWord32 renderDelayMs = 0;
+ WebRtc_UWord32 minPlayoutDelayMs = 0;
+ const WebRtc_Word64 MAX_RUNTIME_MS = 10000;
+ std::string outFilename = args.outputFile;
+ if (outFilename == "")
+ outFilename = test::OutputPath() + "RtpPlayMT_decoded.yuv";
+
+ bool nackEnabled = (protectionEnabled &&
+ (protection == kProtectionDualDecoder ||
+ protection == kProtectionNack ||
+ kProtectionNackFEC));
+ TickTimeBase clock;
+ VideoCodingModule* vcm =
+ VideoCodingModule::Create(1, &clock);
+ RtpDataCallback dataCallback(vcm);
+ std::string rtpFilename;
+ rtpFilename = args.inputFile;
+ if (releaseTestNo > 0)
+ {
+ // Setup a release test
+ switch (releaseTestVideoType)
+ {
+ case webrtc::kVideoCodecVP8:
+ rtpFilename = args.inputFile;
+ outFilename = test::OutputPath() + "MTReceiveTest_VP8";
+ break;
+ default:
+ return -1;
+ }
+ switch (releaseTestNo)
+ {
+ case 1:
+ // Normal execution
+ protectionEnabled = false;
+ nackEnabled = false;
+ rttMS = 0;
+ lossRate = 0.0f;
+ outFilename += "_Normal.yuv";
+ break;
+ case 2:
+ // Packet loss
+ protectionEnabled = false;
+ nackEnabled = false;
+ rttMS = 0;
+ lossRate = 0.05f;
+ outFilename += "_0.05.yuv";
+ break;
+ case 3:
+ // Packet loss and NACK
+ protection = kProtectionNack;
+ nackEnabled = true;
+ protectionEnabled = true;
+ rttMS = 100;
+ lossRate = 0.05f;
+ outFilename += "_0.05_NACK_100ms.yuv";
+ break;
+ case 4:
+ // Packet loss and dual decoder
+ // Not implemented
+ return 0;
+ break;
+ default:
+ return -1;
+ }
+ printf("Watch %s to verify that the output is reasonable\n", outFilename.c_str());
+ }
+ RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
+ PayloadTypeList payloadTypes;
+ payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
+ kVideoCodecVP8));
+ Trace::CreateTrace();
+ Trace::SetTraceFile("receiverTestTrace.txt");
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+
+ // END Settings
+
+ // Set up
+
+ SharedState mtState(*vcm, rtpStream);
+
+ if (rtpStream.Initialize(&payloadTypes) < 0)
+ {
+ return -1;
+ }
+ rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
+
+ WebRtc_Word32 ret = vcm->InitializeReceiver();
+ if (ret < 0)
+ {
+ return -1;
+ }
+
+ // Create and start all threads
+ ThreadWrapper* processingThread = ThreadWrapper::CreateThread(ProcessingThread,
+ &mtState, kNormalPriority, "ProcessingThread");
+ ThreadWrapper* rtpReaderThread = ThreadWrapper::CreateThread(RtpReaderThread,
+ &mtState, kNormalPriority, "RtpReaderThread");
+ ThreadWrapper* decodeThread = ThreadWrapper::CreateThread(DecodeThread,
+ &mtState, kNormalPriority, "DecodeThread");
+
+ // Register receive codecs in VCM
+ for (PayloadTypeList::iterator it = payloadTypes.begin();
+ it != payloadTypes.end(); ++it) {
+ PayloadCodecTuple* payloadType = *it;
+ if (payloadType != NULL)
+ {
+ VideoCodec codec;
+ VideoCodingModule::Codec(payloadType->codecType, &codec);
+ codec.plType = payloadType->payloadType;
+ if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
+ {
+ return -1;
+ }
+ }
+ }
+
+ if (processingThread != NULL)
+ {
+ unsigned int tid;
+ processingThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start processing thread\n");
+ return -1;
+ }
+ if (rtpReaderThread != NULL)
+ {
+ unsigned int tid;
+ rtpReaderThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start RTP reader thread\n");
+ return -1;
+ }
+ if (decodeThread != NULL)
+ {
+ unsigned int tid;
+ decodeThread->Start(tid);
+ }
+ else
+ {
+ printf("Unable to start decode thread\n");
+ return -1;
+ }
+
+ FrameReceiveCallback receiveCallback(outFilename);
+ vcm->RegisterReceiveCallback(&receiveCallback);
+ vcm->RegisterPacketRequestCallback(&rtpStream);
+
+ vcm->SetChannelParameters(0, 0, rttMS);
+ vcm->SetVideoProtection(protection, protectionEnabled);
+ vcm->SetRenderDelay(renderDelayMs);
+ vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+
+ EventWrapper& waitEvent = *EventWrapper::Create();
+
+ // Decode for 10 seconds and then tear down and exit.
+ waitEvent.Wait(MAX_RUNTIME_MS);
+
+ // Tear down
+ while (!payloadTypes.empty())
+ {
+ delete payloadTypes.front();
+ payloadTypes.pop_front();
+ }
+ while (!processingThread->Stop())
+ {
+ ;
+ }
+ while (!rtpReaderThread->Stop())
+ {
+ ;
+ }
+ while (!decodeThread->Stop())
+ {
+ ;
+ }
+ VideoCodingModule::Destroy(vcm);
+ vcm = NULL;
+ delete &waitEvent;
+ delete processingThread;
+ delete decodeThread;
+ delete rtpReaderThread;
+ rtpStream.Print();
+ Trace::ReturnTrace();
+ return 0;
+}
diff --git a/trunk/src/modules/video_coding/main/test/video_source.cc b/trunk/src/modules/video_coding/main/test/video_source.cc
new file mode 100644
index 0000000..d7ba0b9
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/video_source.cc
@@ -0,0 +1,202 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_source.h"
+
+#include <cassert>
+
+#include "testsupport/fileutils.h"
+
+VideoSource::VideoSource()
+:
+_fileName(webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv"),
+_width(352),
+_height(288),
+_type(webrtc::kI420),
+_frameRate(30)
+{
+ //
+}
+
+VideoSource::VideoSource(std::string fileName, VideoSize size,
+ float frameRate, webrtc::VideoType type /*= webrtc::kI420*/)
+:
+_fileName(fileName),
+_width(0),
+_height(0),
+_type(type),
+_frameRate(frameRate)
+{
+ assert(size != kUndefined && size != kNumberOfVideoSizes);
+ assert(type != webrtc::kUnknown);
+ assert(frameRate > 0);
+ GetWidthHeight(size);
+}
+
+VideoSource::VideoSource(std::string fileName, WebRtc_UWord16 width, WebRtc_UWord16 height,
+ float frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
+:
+_fileName(fileName),
+_width(width),
+_height(height),
+_type(type),
+_frameRate(frameRate)
+{
+ assert(width > 0);
+ assert(height > 0);
+ assert(type != webrtc::kUnknown);
+ assert(frameRate > 0);
+}
+
+WebRtc_Word32
+VideoSource::GetFrameLength() const
+{
+ return webrtc::CalcBufferSize(_type, _width, _height);
+}
+
+std::string
+VideoSource::GetName() const
+{
+ // Remove path.
+ size_t slashPos = _fileName.find_last_of("/\\");
+ if (slashPos == std::string::npos)
+ {
+ slashPos = 0;
+ }
+ else
+ {
+ slashPos++;
+ }
+
+ // Remove extension and underscored suffix if it exists.
+ //return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
+ // _fileName.find_last_of(".")) - slashPos);
+ // MS: Removing suffix, not underscore....keeping full file name
+ return _fileName.substr(slashPos, _fileName.find_last_of(".") - slashPos);
+
+}
+
+int
+VideoSource::GetWidthHeight( VideoSize size)
+{
+ switch(size)
+ {
+ case kSQCIF:
+ _width = 128;
+ _height = 96;
+ return 0;
+ case kQQVGA:
+ _width = 160;
+ _height = 120;
+ return 0;
+ case kQCIF:
+ _width = 176;
+ _height = 144;
+ return 0;
+ case kCGA:
+ _width = 320;
+ _height = 200;
+ return 0;
+ case kQVGA:
+ _width = 320;
+ _height = 240;
+ return 0;
+ case kSIF:
+ _width = 352;
+ _height = 240;
+ return 0;
+ case kWQVGA:
+ _width = 400;
+ _height = 240;
+ return 0;
+ case kCIF:
+ _width = 352;
+ _height = 288;
+ return 0;
+ case kW288p:
+ _width = 512;
+ _height = 288;
+ return 0;
+ case k448p:
+ _width = 576;
+ _height = 448;
+ return 0;
+ case kVGA:
+ _width = 640;
+ _height = 480;
+ return 0;
+ case k432p:
+ _width = 720;
+ _height = 432;
+ return 0;
+ case kW432p:
+ _width = 768;
+ _height = 432;
+ return 0;
+ case k4SIF:
+ _width = 704;
+ _height = 480;
+ return 0;
+ case kW448p:
+ _width = 768;
+ _height = 448;
+ return 0;
+ case kNTSC:
+ _width = 720;
+ _height = 480;
+ return 0;
+ case kFW448p:
+ _width = 800;
+ _height = 448;
+ return 0;
+ case kWVGA:
+ _width = 800;
+ _height = 480;
+ return 0;
+ case k4CIF:
+ _width = 704;
+ _height = 576;
+ return 0;
+ case kSVGA:
+ _width = 800;
+ _height = 600;
+ return 0;
+ case kW544p:
+ _width = 960;
+ _height = 544;
+ return 0;
+ case kW576p:
+ _width = 1024;
+ _height = 576;
+ return 0;
+ case kHD:
+ _width = 960;
+ _height = 720;
+ return 0;
+ case kXGA:
+ _width = 1024;
+ _height = 768;
+ return 0;
+ case kFullHD:
+ _width = 1440;
+ _height = 1080;
+ return 0;
+ case kWHD:
+ _width = 1280;
+ _height = 720;
+ return 0;
+ case kWFullHD:
+ _width = 1920;
+ _height = 1080;
+ return 0;
+ default:
+ return -1;
+ }
+}
diff --git a/trunk/src/modules/video_coding/main/test/video_source.h b/trunk/src/modules/video_coding/main/test/video_source.h
new file mode 100644
index 0000000..d9ad3d8
--- /dev/null
+++ b/trunk/src/modules/video_coding/main/test/video_source.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "typedefs.h"
+
+#include <string>
+
+enum VideoSize
+ {
+ kUndefined,
+ kSQCIF, // 128*96 = 12 288
+ kQQVGA, // 160*120 = 19 200
+ kQCIF, // 176*144 = 25 344
+ kCGA, // 320*200 = 64 000
+ kQVGA, // 320*240 = 76 800
+ kSIF, // 352*240 = 84 480
+ kWQVGA, // 400*240 = 96 000
+ kCIF, // 352*288 = 101 376
+ kW288p, // 512*288 = 147 456 (WCIF)
+ k448p, // 576*448 = 281 088
+ kVGA, // 640*480 = 307 200
+ k432p, // 720*432 = 311 040
+ kW432p, // 768*432 = 331 776
+ k4SIF, // 704*480 = 337 920
+ kW448p, // 768*448 = 344 064
+ kNTSC, // 720*480 = 345 600
+ kFW448p, // 800*448 = 358 400
+ kWVGA, // 800*480 = 384 000
+ k4CIF, // 704*576 = 405 504
+ kSVGA, // 800*600 = 480 000
+ kW544p, // 960*544 = 522 240
+ kW576p, // 1024*576 = 589 824 (W4CIF)
+ kHD, // 960*720 = 691 200
+ kXGA, // 1024*768 = 786 432
+ kWHD, // 1280*720 = 921 600
+ kFullHD, // 1440*1080 = 1 555 200
+ kWFullHD, // 1920*1080 = 2 073 600
+
+ kNumberOfVideoSizes
+ };
+
+
+class VideoSource
+{
+public:
+ VideoSource();
+ VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
+ VideoSource(std::string fileName, WebRtc_UWord16 width, WebRtc_UWord16 height,
+ float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
+
+ std::string GetFileName() const { return _fileName; }
+ WebRtc_UWord16 GetWidth() const { return _width; }
+ WebRtc_UWord16 GetHeight() const { return _height; }
+ webrtc::VideoType GetType() const { return _type; }
+ float GetFrameRate() const { return _frameRate; }
+ int GetWidthHeight( VideoSize size);
+
+ // Returns the filename with the path (including the leading slash) removed.
+ std::string GetName() const;
+
+ WebRtc_Word32 GetFrameLength() const;
+
+private:
+ std::string _fileName;
+ WebRtc_UWord16 _width;
+ WebRtc_UWord16 _height;
+ webrtc::VideoType _type;
+ float _frameRate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+
diff --git a/trunk/src/modules/video_processing/main/OWNERS b/trunk/src/modules/video_processing/main/OWNERS
new file mode 100644
index 0000000..7183cf2
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/OWNERS
@@ -0,0 +1,4 @@
+stefan@webrtc.org
+mikhal@webrtc.org
+marpan@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_processing/main/interface/video_processing.h b/trunk/src/modules/video_processing/main/interface/video_processing.h
new file mode 100644
index 0000000..512cace
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/interface/video_processing.h
@@ -0,0 +1,382 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_processing.h
+ * This header file contains the API required for the video
+ * processing module class.
+ */
+
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
+#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
+
+#include "module.h"
+#include "module_common_types.h"
+#include "video_processing_defines.h"
+
+/**
+ The module is largely intended to process video streams, except functionality
+ provided by static functions which operate independent of previous frames. It
+ is recommended, but not required that a unique instance be used for each
+ concurrently processed stream. Similarly, it is recommended to call Reset()
+ before switching to a new stream, but this is not absolutely required.
+
+ The module provides basic thread safety by permitting only a single function to
+ execute concurrently.
+*/
+
+namespace webrtc {
+
+class VideoProcessingModule : public Module
+{
+public:
+ /**
+ Structure to hold frame statistics. Populate it with GetFrameStats().
+ */
+ struct FrameStats
+ {
+ FrameStats() :
+ mean(0),
+ sum(0),
+ numPixels(0),
+ subSamplWidth(0),
+ subSamplHeight(0)
+ {
+ memset(hist, 0, sizeof(hist));
+ }
+
+ WebRtc_UWord32 hist[256]; /**< Histogram of frame */
+ WebRtc_UWord32 mean; /**< Mean value of frame */
+ WebRtc_UWord32 sum; /**< Sum of frame */
+ WebRtc_UWord32 numPixels; /**< Number of pixels */
+ WebRtc_UWord8 subSamplWidth; /**< Subsampling rate of width in powers of 2 */
+ WebRtc_UWord8 subSamplHeight; /**< Subsampling rate of height in powers of 2 */
+ };
+
+ /**
+ Specifies the warning types returned by BrightnessDetection().
+ */
+ enum BrightnessWarning
+ {
+ kNoWarning, /**< Frame has acceptable brightness */
+ kDarkWarning, /**< Frame is too dark */
+ kBrightWarning /**< Frame is too bright */
+ };
+
+ /*
+ Creates a VPM object.
+
+ \param[in] id
+ Unique identifier of this object.
+
+ \return Pointer to a VPM object.
+ */
+ static VideoProcessingModule* Create(WebRtc_Word32 id);
+
+ /**
+ Destroys a VPM object.
+
+ \param[in] module
+ Pointer to the VPM object to destroy.
+ */
+ static void Destroy(VideoProcessingModule* module);
+
+ /**
+ Not supported.
+ */
+ virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; }
+
+ /**
+ Not supported.
+ */
+ virtual WebRtc_Word32 Process() { return -1; }
+
+ /**
+ Resets all processing components to their initial states. This should be
+ called whenever a new video stream is started.
+ */
+ virtual void Reset() = 0;
+
+ /**
+ Retrieves statistics for the input frame. This function must be used to
+ prepare a FrameStats struct for use in certain VPM functions.
+
+ \param[out] stats
+ The frame statistics will be stored here on return.
+
+ \param[in] frame
+ Pointer to the video frame.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \return 0 on success, -1 on failure.
+ */
+ static WebRtc_Word32 GetFrameStats(FrameStats& stats,
+ const WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height);
+
+ /**
+ \overload
+ */
+ static WebRtc_Word32 GetFrameStats(FrameStats& stats,
+ const VideoFrame& frame);
+
+ /**
+ Checks the validity of a FrameStats struct. Currently, valid implies only
+ that is had changed from its initialized state.
+
+ \param[in] stats
+ Frame statistics.
+
+ \return True on valid stats, false on invalid stats.
+ */
+ static bool ValidFrameStats(const FrameStats& stats);
+
+ /**
+ Returns a FrameStats struct to its intialized state.
+
+ \param[in,out] stats
+ Frame statistics.
+ */
+ static void ClearFrameStats(FrameStats& stats);
+
+ /**
+ Enhances the color of an image through a constant mapping. Only the
+ chrominance is altered. Has a fixed-point implementation.
+
+ \param[in,out] frame
+ Pointer to the video frame.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \return 0 on success, -1 on failure.
+ */
+ static WebRtc_Word32 ColorEnhancement(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height);
+
+ /**
+ \overload
+ */
+ static WebRtc_Word32 ColorEnhancement(VideoFrame& frame);
+
+ /**
+ Increases/decreases the luminance value.
+
+ \param[in,out] frame
+ Pointer to the video frame buffer.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \param[in] delta
+ The amount to change the chrominance value of every single pixel.
+ Can be < 0 also.
+
+ \return 0 on success, -1 on failure.
+ */
+ static WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
+ int width, int height, int delta);
+ /**
+ \overload
+ */
+ static WebRtc_Word32 Brighten(VideoFrame& frame, int delta);
+
+ /**
+ Detects and removes camera flicker from a video stream. Every frame from the
+ stream must be passed in. A frame will only be altered if flicker has been
+ detected. Has a fixed-point implementation.
+
+ \param[in,out] frame
+ Pointer to the video frame.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \param[in] timestamp
+ Frame timestamp in 90 kHz format.
+
+ \param[in,out] stats
+ Frame statistics provided by GetFrameStats(). On return the stats will
+ be reset to zero if the frame was altered. Call GetFrameStats() again
+ if the statistics for the altered frame are required.
+
+ \return 0 on success, -1 on failure.
+ */
+ virtual WebRtc_Word32 Deflickering(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ WebRtc_UWord32 timestamp,
+ FrameStats& stats) = 0;
+
+ /**
+ \overload
+ */
+ virtual WebRtc_Word32 Deflickering(VideoFrame& frame,
+ FrameStats& stats) = 0;
+
+ /**
+ Denoises a video frame. Every frame from the stream should be passed in.
+ Has a fixed-point implementation.
+
+ \param[in,out] frame
+ Pointer to the video frame.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \return The number of modified pixels on success, -1 on failure.
+ */
+ virtual WebRtc_Word32 Denoising(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height) = 0;
+
+ /**
+ \overload
+ */
+ virtual WebRtc_Word32 Denoising(VideoFrame& frame) = 0;
+
+ /**
+ Detects if a video frame is excessively bright or dark. Returns a warning if
+ this is the case. Multiple frames should be passed in before expecting a
+ warning. Has a floating-point implementation.
+
+ \param[in] frame
+ Pointer to the video frame.
+
+ \param[in] width
+ Frame width in pixels.
+
+ \param[in] height
+ Frame height in pixels.
+
+ \param[in] stats
+ Frame statistics provided by GetFrameStats().
+
+ \return A member of BrightnessWarning on success, -1 on error
+ */
+ virtual WebRtc_Word32 BrightnessDetection(const WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ const FrameStats& stats) = 0;
+
+ /**
+ \overload
+ */
+ virtual WebRtc_Word32 BrightnessDetection(const VideoFrame& frame,
+ const FrameStats& stats) = 0;
+
+
+ /**
+ The following functions refer to the pre-processor unit within VPM. The pre-processor
+ perfoms spatial/temporal decimation and content analysis on the frames prior to encoding.
+ */
+
+ /**
+ Enable/disable temporal decimation
+
+ \param[in] enable when true, temporal decimation is enabled
+ */
+ virtual void EnableTemporalDecimation(bool enable) = 0;
+
+ /**
+ Set target resolution
+
+ \param[in] width
+ Target width
+
+ \param[in] height
+ Target height
+
+ \param[in] frameRate
+ Target frameRate
+
+ \return VPM_OK on success, a negative value on error (see error codes)
+
+ */
+ virtual WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate) = 0;
+
+ /**
+ Set max frame rate
+ \param[in] maxFrameRate: maximum frame rate (limited to native frame rate)
+
+ \return VPM_OK on success, a negative value on error (see error codes)
+ */
+ virtual WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate) = 0;
+
+ /**
+ Get decimated(target) frame rate
+ */
+ virtual WebRtc_UWord32 DecimatedFrameRate() = 0;
+
+ /**
+ Get decimated(target) frame width
+ */
+ virtual WebRtc_UWord32 DecimatedWidth() const = 0;
+
+ /**
+ Get decimated(target) frame height
+ */
+ virtual WebRtc_UWord32 DecimatedHeight() const = 0 ;
+
+ /**
+ Set the spatial resampling settings of the VPM: The resampler may either be disabled or one of the following:
+ scaling to a close to target dimension followed by crop/pad
+
+ \param[in] resamplingMode
+ Set resampling mode (a member of VideoFrameResampling)
+ */
+ virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode) = 0;
+
+ /**
+ Get Processed (decimated) frame
+
+ \param[in] frame pointer to the video frame.
+
+ \param[in] processedFrame pointer (double) to the processed frame
+
+ \return VPM_OK on success, a negative value on error (see error codes)
+ */
+ virtual WebRtc_Word32 PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame) = 0;
+
+ /**
+ Return content metrics for the last processed frame
+ */
+ virtual VideoContentMetrics* ContentMetrics() const = 0 ;
+
+ /**
+ Enable content analysis
+ */
+ virtual void EnableContentAnalysis(bool enable) = 0;
+
+};
+
+} //namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/interface/video_processing_defines.h b/trunk/src/modules/video_processing/main/interface/video_processing_defines.h
new file mode 100644
index 0000000..d9bebd4
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/interface/video_processing_defines.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_processing_defines.h
+ * This header file includes the definitions used in the video processor module
+ */
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Error codes
+#define VPM_OK 0
+#define VPM_GENERAL_ERROR -1
+#define VPM_MEMORY -2
+#define VPM_PARAMETER_ERROR -3
+#define VPM_SCALE_ERROR -4
+#define VPM_UNINITIALIZED -5
+#define VPM_UNIMPLEMENTED -6
+
+enum VideoFrameResampling
+{
+ // TODO: Do we still need crop/pad?
+ kNoRescaling, // disables rescaling
+ kFastRescaling, // point
+ kBiLinear, // bi-linear interpolation
+ kBox, // Box inteprolation
+};
+
+} //namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/source/Android.mk b/trunk/src/modules/video_processing/main/source/Android.mk
new file mode 100644
index 0000000..829fa96
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/Android.mk
@@ -0,0 +1,59 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_video_processing
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ brighten.cc \
+ brightness_detection.cc \
+ color_enhancement.cc \
+ content_analysis.cc \
+ deflickering.cc \
+ denoising.cc \
+ frame_preprocessor.cc \
+ spatial_resampler.cc \
+ video_decimator.cc \
+ video_processing_impl.cc
+
+ifeq ($(TARGET_ARCH),x86)
+LOCAL_SRC_FILES += \
+ content_analysis_sse2.cc
+endif
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../audio_coding/main/interface \
+ $(LOCAL_PATH)/../../../utility/interface \
+ $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_processing/main/source/brighten.cc b/trunk/src/modules/video_processing/main/source/brighten.cc
new file mode 100644
index 0000000..51e4b6b
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/brighten.cc
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_processing/main/source/brighten.h"
+
+#include <cstdlib>
+
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+namespace VideoProcessing {
+
+WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
+ int width, int height, int delta) {
+ if (frame == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1,
+ "Null frame pointer");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ if (width <= 0 || height <= 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1,
+ "Invalid frame size");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ int numPixels = width * height;
+
+ int lookUp[256];
+ for (int i = 0; i < 256; i++) {
+ int val = i + delta;
+ lookUp[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
+ }
+
+ WebRtc_UWord8* tempPtr = frame;
+
+ for (int i = 0; i < numPixels; i++) {
+ *tempPtr = static_cast<WebRtc_UWord8>(lookUp[*tempPtr]);
+ tempPtr++;
+ }
+ return VPM_OK;
+}
+
+} // namespace VideoProcessing
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/brighten.h b/trunk/src/modules/video_processing/main/source/brighten.h
new file mode 100644
index 0000000..b7e6fb7
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/brighten.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
+#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
+
+#include "typedefs.h"
+#include "modules/video_processing/main/interface/video_processing.h"
+
+namespace webrtc {
+namespace VideoProcessing {
+
+WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
+ int width, int height, int delta);
+
+} // namespace VideoProcessing
+} // namespace webrtc
+
+#endif // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
diff --git a/trunk/src/modules/video_processing/main/source/brightness_detection.cc b/trunk/src/modules/video_processing/main/source/brightness_detection.cc
new file mode 100644
index 0000000..6840df2
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/brightness_detection.cc
@@ -0,0 +1,195 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_processing.h"
+#include "brightness_detection.h"
+#include "trace.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+VPMBrightnessDetection::VPMBrightnessDetection() :
+ _id(0)
+{
+ Reset();
+}
+
+VPMBrightnessDetection::~VPMBrightnessDetection()
+{
+}
+
+WebRtc_Word32
+VPMBrightnessDetection::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return VPM_OK;
+}
+
+void
+VPMBrightnessDetection::Reset()
+{
+ _frameCntBright = 0;
+ _frameCntDark = 0;
+}
+
+WebRtc_Word32
+VPMBrightnessDetection::ProcessFrame(const WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height,
+ const VideoProcessingModule::FrameStats& stats)
+{
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ if (width == 0 || height == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ if (!VideoProcessingModule::ValidFrameStats(stats))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame stats");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ const WebRtc_UWord8 frameCntAlarm = 2;
+
+ // Get proportion in lowest bins
+ WebRtc_UWord8 lowTh = 20;
+ float propLow = 0;
+ for (WebRtc_UWord32 i = 0; i < lowTh; i++)
+ {
+ propLow += stats.hist[i];
+ }
+ propLow /= stats.numPixels;
+
+ // Get proportion in highest bins
+ unsigned char highTh = 230;
+ float propHigh = 0;
+ for (WebRtc_UWord32 i = highTh; i < 256; i++)
+ {
+ propHigh += stats.hist[i];
+ }
+ propHigh /= stats.numPixels;
+
+ if(propHigh < 0.4)
+ {
+ if (stats.mean < 90 || stats.mean > 170)
+ {
+ // Standard deviation of Y
+ float stdY = 0;
+ for (WebRtc_UWord32 h = 0; h < height; h += (1 << stats.subSamplHeight))
+ {
+ WebRtc_UWord32 row = h*width;
+ for (WebRtc_UWord32 w = 0; w < width; w += (1 << stats.subSamplWidth))
+ {
+ stdY += (frame[w + row] - stats.mean) * (frame[w + row] - stats.mean);
+ }
+ }
+ stdY = sqrt(stdY / stats.numPixels);
+
+ // Get percentiles
+ WebRtc_UWord32 sum = 0;
+ WebRtc_UWord32 medianY = 140;
+ WebRtc_UWord32 perc05 = 0;
+ WebRtc_UWord32 perc95 = 255;
+ float posPerc05 = stats.numPixels * 0.05f;
+ float posMedian = stats.numPixels * 0.5f;
+ float posPerc95 = stats.numPixels * 0.95f;
+ for (WebRtc_UWord32 i = 0; i < 256; i++)
+ {
+ sum += stats.hist[i];
+
+ if (sum < posPerc05)
+ {
+ perc05 = i; // 5th perc
+ }
+ if (sum < posMedian)
+ {
+ medianY = i; // 50th perc
+ }
+ if (sum < posPerc95)
+ {
+ perc95 = i; // 95th perc
+ }
+ else
+ {
+ break;
+ }
+ }
+
+ // Check if image is too dark
+ if ((stdY < 55) && (perc05 < 50))
+ {
+ if (medianY < 60 || stats.mean < 80 || perc95 < 130 || propLow > 0.20)
+ {
+ _frameCntDark++;
+ }
+ else
+ {
+ _frameCntDark = 0;
+ }
+ }
+ else
+ {
+ _frameCntDark = 0;
+ }
+
+ // Check if image is too bright
+ if ((stdY < 52) && (perc95 > 200) && (medianY > 160))
+ {
+ if (medianY > 185 || stats.mean > 185 || perc05 > 140 || propHigh > 0.25)
+ {
+ _frameCntBright++;
+ }
+ else
+ {
+ _frameCntBright = 0;
+ }
+ }
+ else
+ {
+ _frameCntBright = 0;
+ }
+
+ }
+ else
+ {
+ _frameCntDark = 0;
+ _frameCntBright = 0;
+ }
+
+ }
+ else
+ {
+ _frameCntBright++;
+ _frameCntDark = 0;
+ }
+
+ if (_frameCntDark > frameCntAlarm)
+ {
+ return VideoProcessingModule::kDarkWarning;
+ }
+ else if (_frameCntBright > frameCntAlarm)
+ {
+ return VideoProcessingModule::kBrightWarning;
+ }
+ else
+ {
+ return VideoProcessingModule::kNoWarning;
+ }
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/brightness_detection.h b/trunk/src/modules/video_processing/main/source/brightness_detection.h
new file mode 100644
index 0000000..7bed556
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/brightness_detection.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * brightness_detection.h
+ */
+#ifndef VPM_BRIGHTNESS_DETECTION_H
+#define VPM_BRIGHTNESS_DETECTION_H
+
+#include "typedefs.h"
+#include "video_processing.h"
+
+namespace webrtc {
+
+class VPMBrightnessDetection
+{
+public:
+ VPMBrightnessDetection();
+ ~VPMBrightnessDetection();
+
+ WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
+
+ void Reset();
+
+ WebRtc_Word32 ProcessFrame(const WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ const VideoProcessingModule::FrameStats& stats);
+
+private:
+ WebRtc_Word32 _id;
+
+ WebRtc_UWord32 _frameCntBright;
+ WebRtc_UWord32 _frameCntDark;
+};
+
+} //namespace
+
+#endif // VPM_BRIGHTNESS_DETECTION_H
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement.cc b/trunk/src/modules/video_processing/main/source/color_enhancement.cc
new file mode 100644
index 0000000..426596f
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/color_enhancement.cc
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "color_enhancement.h"
+#include "color_enhancement_private.h"
+#include "trace.h"
+#include <cstdlib> // NULL
+
+namespace webrtc {
+
+namespace VideoProcessing
+{
+ WebRtc_Word32
+ ColorEnhancement(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+ {
+ // pointers to U and V color pixels
+ WebRtc_UWord8* ptrU;
+ WebRtc_UWord8* ptrV;
+ WebRtc_UWord8 tempChroma;
+ const WebRtc_UWord32 numPixels = width * height;
+
+
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Null frame pointer");
+ return VPM_GENERAL_ERROR;
+ }
+
+ if (width == 0 || height == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Invalid frame size");
+ return VPM_GENERAL_ERROR;
+ }
+
+ // set pointers to first U and V pixels
+
+ // stream format:
+ // | numPixels bytes luminance | numPixels/4 bytes chroma U | numPixels/4 chroma V |
+
+ ptrU = frame + numPixels; // skip luminance
+ ptrV = ptrU + (numPixels>>2);
+
+ // loop through all chrominance pixels and modify color
+ for (WebRtc_UWord32 ix = 0; ix < (numPixels>>2); ix++)
+ {
+ tempChroma = colorTable[*ptrU][*ptrV];
+ *ptrV = colorTable[*ptrV][*ptrU];
+ *ptrU = tempChroma;
+
+ // increment pointers
+ ptrU++;
+ ptrV++;
+ }
+ return VPM_OK;
+ }
+
+} //namespace
+
+} //namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement.h b/trunk/src/modules/video_processing/main/source/color_enhancement.h
new file mode 100644
index 0000000..87fabc3
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/color_enhancement.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * color_enhancement.h
+ */
+#ifndef VPM_COLOR_ENHANCEMENT_H
+#define VPM_COLOR_ENHANCEMENT_H
+
+#include "typedefs.h"
+#include "video_processing.h"
+
+namespace webrtc {
+
+namespace VideoProcessing
+{
+ WebRtc_Word32 ColorEnhancement(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height);
+}
+
+} //namespace
+
+#endif // VPM_COLOR_ENHANCEMENT_H
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement_private.h b/trunk/src/modules/video_processing/main/source/color_enhancement_private.h
new file mode 100644
index 0000000..b88fc1a
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/color_enhancement_private.h
@@ -0,0 +1,273 @@
+#ifndef VPM_COLOR_ENHANCEMENT_PRIVATE_H
+#define VPM_COLOR_ENHANCEMENT_PRIVATE_H
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+//Table created with Matlab script createTable.m
+//Usage:
+// Umod=colorTable[U][V]
+// Vmod=colorTable[V][U]
+static const WebRtc_UWord8 colorTable[256][256] = {
+ {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+ {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+ {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3},
+ {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4},
+ {5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5},
+ {6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6},
+ {7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7},
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+ {253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253},
+ {254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254},
+ {255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255}
+};
+
+} //namespace
+
+#endif // VPM_COLOR_ENHANCEMENT_PRIVATE_H
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis.cc b/trunk/src/modules/video_processing/main/source/content_analysis.cc
new file mode 100644
index 0000000..8ea319c
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/content_analysis.cc
@@ -0,0 +1,338 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "content_analysis.h"
+#include "tick_util.h"
+#include "system_wrappers/interface/cpu_features_wrapper.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+namespace webrtc {
+
+VPMContentAnalysis::VPMContentAnalysis(bool runtime_cpu_detection):
+_origFrame(NULL),
+_prevFrame(NULL),
+_width(0),
+_height(0),
+_skipNum(1),
+_border(8),
+_motionMagnitude(0.0f),
+_spatialPredErr(0.0f),
+_spatialPredErrH(0.0f),
+_spatialPredErrV(0.0f),
+_firstFrame(true),
+_CAInit(false),
+_cMetrics(NULL)
+{
+ ComputeSpatialMetrics = &VPMContentAnalysis::ComputeSpatialMetrics_C;
+ TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_C;
+
+ if (runtime_cpu_detection)
+ {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+ if (WebRtc_GetCPUInfo(kSSE2))
+ {
+ ComputeSpatialMetrics =
+ &VPMContentAnalysis::ComputeSpatialMetrics_SSE2;
+ TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_SSE2;
+ }
+#endif
+ }
+
+ Release();
+}
+
+VPMContentAnalysis::~VPMContentAnalysis()
+{
+ Release();
+}
+
+
+VideoContentMetrics*
+VPMContentAnalysis::ComputeContentMetrics(const VideoFrame* inputFrame)
+{
+ if (inputFrame == NULL)
+ {
+ return NULL;
+ }
+
+ // Init if needed (native dimension change)
+ if (_width != inputFrame->Width() || _height != inputFrame->Height())
+ {
+ if (VPM_OK != Initialize((WebRtc_UWord16)inputFrame->Width(),
+ (WebRtc_UWord16)inputFrame->Height()))
+ {
+ return NULL;
+ }
+ }
+
+ _origFrame = inputFrame->Buffer();
+
+ // compute spatial metrics: 3 spatial prediction errors
+ (this->*ComputeSpatialMetrics)();
+
+ // compute motion metrics
+ if (_firstFrame == false)
+ ComputeMotionMetrics();
+
+ // saving current frame as previous one: Y only
+ memcpy(_prevFrame, _origFrame, _width * _height);
+
+ _firstFrame = false;
+ _CAInit = true;
+
+ return ContentMetrics();
+}
+
+WebRtc_Word32
+VPMContentAnalysis::Release()
+{
+ if (_cMetrics != NULL)
+ {
+ delete _cMetrics;
+ _cMetrics = NULL;
+ }
+
+ if (_prevFrame != NULL)
+ {
+ delete [] _prevFrame;
+ _prevFrame = NULL;
+ }
+
+ _width = 0;
+ _height = 0;
+ _firstFrame = true;
+
+ return VPM_OK;
+}
+
+WebRtc_Word32
+VPMContentAnalysis::Initialize(WebRtc_UWord16 width, WebRtc_UWord16 height)
+{
+ _width = width;
+ _height = height;
+ _firstFrame = true;
+
+ // skip parameter: # of skipped rows: for complexity reduction
+ // temporal also currently uses it for column reduction.
+ _skipNum = 1;
+
+ // use skipNum = 2 for 4CIF, WHD
+ if ( (_height >= 576) && (_width >= 704) )
+ {
+ _skipNum = 2;
+ }
+ // use skipNum = 4 for FULLL_HD images
+ if ( (_height >= 1080) && (_width >= 1920) )
+ {
+ _skipNum = 4;
+ }
+
+ if (_cMetrics != NULL)
+ {
+ delete _cMetrics;
+ }
+
+ if (_prevFrame != NULL)
+ {
+ delete [] _prevFrame;
+ }
+
+ // Spatial Metrics don't work on a border of 8. Minimum processing
+ // block size is 16 pixels. So make sure the width and height support this.
+ if (_width <= 32 || _height <= 32)
+ {
+ _CAInit = false;
+ return VPM_PARAMETER_ERROR;
+ }
+
+ _cMetrics = new VideoContentMetrics();
+ if (_cMetrics == NULL)
+ {
+ return VPM_MEMORY;
+ }
+
+ _prevFrame = new WebRtc_UWord8[_width * _height] ; // Y only
+ if (_prevFrame == NULL)
+ {
+ return VPM_MEMORY;
+ }
+
+ return VPM_OK;
+}
+
+
+// Compute motion metrics: magnitude over non-zero motion vectors,
+// and size of zero cluster
+WebRtc_Word32
+VPMContentAnalysis::ComputeMotionMetrics()
+{
+
+ // Motion metrics: only one is derived from normalized
+ // (MAD) temporal difference
+ (this->*TemporalDiffMetric)();
+
+ return VPM_OK;
+}
+
+// Normalized temporal difference (MAD): used as a motion level metric
+// Normalize MAD by spatial contrast: images with more contrast
+// (pixel variance) likely have larger temporal difference
+// To reduce complexity, we compute the metric for a reduced set of points.
+WebRtc_Word32
+VPMContentAnalysis::TemporalDiffMetric_C()
+{
+ // size of original frame
+ WebRtc_UWord16 sizei = _height;
+ WebRtc_UWord16 sizej = _width;
+
+ WebRtc_UWord32 tempDiffSum = 0;
+ WebRtc_UWord32 pixelSum = 0;
+ WebRtc_UWord64 pixelSqSum = 0;
+
+ WebRtc_UWord32 numPixels = 0; // counter for # of pixels
+ WebRtc_UWord32 ssn;
+
+ const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
+
+ for(WebRtc_UWord16 i = _border; i < sizei - _border; i += _skipNum)
+ {
+ for(WebRtc_UWord16 j = _border; j < width_end; j++)
+ {
+ numPixels += 1;
+ ssn = i * sizej + j;
+
+ WebRtc_UWord8 currPixel = _origFrame[ssn];
+ WebRtc_UWord8 prevPixel = _prevFrame[ssn];
+
+ tempDiffSum += (WebRtc_UWord32)
+ abs((WebRtc_Word16)(currPixel - prevPixel));
+ pixelSum += (WebRtc_UWord32) currPixel;
+ pixelSqSum += (WebRtc_UWord64) (currPixel * currPixel);
+ }
+ }
+
+ // default
+ _motionMagnitude = 0.0f;
+
+ if (tempDiffSum == 0)
+ {
+ return VPM_OK;
+ }
+
+ // normalize over all pixels
+ float const tempDiffAvg = (float)tempDiffSum / (float)(numPixels);
+ float const pixelSumAvg = (float)pixelSum / (float)(numPixels);
+ float const pixelSqSumAvg = (float)pixelSqSum / (float)(numPixels);
+ float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
+
+ if (contrast > 0.0)
+ {
+ contrast = sqrt(contrast);
+ _motionMagnitude = tempDiffAvg/contrast;
+ }
+
+ return VPM_OK;
+
+}
+
+// Compute spatial metrics:
+// To reduce complexity, we compute the metric for a reduced set of points.
+// The spatial metrics are rough estimates of the prediction error cost for
+// each QM spatial mode: 2x2,1x2,2x1
+// The metrics are a simple estimate of the up-sampling prediction error,
+// estimated assuming sub-sampling for decimation (no filtering),
+// and up-sampling back up with simple bilinear interpolation.
+WebRtc_Word32
+VPMContentAnalysis::ComputeSpatialMetrics_C()
+{
+ //size of original frame
+ const WebRtc_UWord16 sizei = _height;
+ const WebRtc_UWord16 sizej = _width;
+
+ // pixel mean square average: used to normalize the spatial metrics
+ WebRtc_UWord32 pixelMSA = 0;
+
+ WebRtc_UWord32 spatialErrSum = 0;
+ WebRtc_UWord32 spatialErrVSum = 0;
+ WebRtc_UWord32 spatialErrHSum = 0;
+
+ // make sure work section is a multiple of 16
+ const WebRtc_UWord32 width_end = ((sizej - 2*_border) & -16) + _border;
+
+ for(WebRtc_UWord16 i = _border; i < sizei - _border; i += _skipNum)
+ {
+ for(WebRtc_UWord16 j = _border; j < width_end; j++)
+ {
+ WebRtc_UWord32 ssn1,ssn2,ssn3,ssn4,ssn5;
+
+ ssn1= i * sizej + j;
+ ssn2 = (i + 1) * sizej + j; // bottom
+ ssn3 = (i - 1) * sizej + j; // top
+ ssn4 = i * sizej + j + 1; // right
+ ssn5 = i * sizej + j - 1; // left
+
+ WebRtc_UWord16 refPixel1 = _origFrame[ssn1] << 1;
+ WebRtc_UWord16 refPixel2 = _origFrame[ssn1] << 2;
+
+ WebRtc_UWord8 bottPixel = _origFrame[ssn2];
+ WebRtc_UWord8 topPixel = _origFrame[ssn3];
+ WebRtc_UWord8 rightPixel = _origFrame[ssn4];
+ WebRtc_UWord8 leftPixel = _origFrame[ssn5];
+
+ spatialErrSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel2
+ - (WebRtc_UWord16)(bottPixel + topPixel
+ + leftPixel + rightPixel)));
+ spatialErrVSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel1
+ - (WebRtc_UWord16)(bottPixel + topPixel)));
+ spatialErrHSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel1
+ - (WebRtc_UWord16)(leftPixel + rightPixel)));
+
+ pixelMSA += _origFrame[ssn1];
+ }
+ }
+
+ // normalize over all pixels
+ const float spatialErr = (float)(spatialErrSum >> 2);
+ const float spatialErrH = (float)(spatialErrHSum >> 1);
+ const float spatialErrV = (float)(spatialErrVSum >> 1);
+ const float norm = (float)pixelMSA;
+
+ // 2X2:
+ _spatialPredErr = spatialErr / norm;
+
+ // 1X2:
+ _spatialPredErrH = spatialErrH / norm;
+
+ // 2X1:
+ _spatialPredErrV = spatialErrV / norm;
+
+ return VPM_OK;
+}
+
+VideoContentMetrics*
+VPMContentAnalysis::ContentMetrics()
+{
+ if (_CAInit == false)
+ {
+ return NULL;
+ }
+
+ _cMetrics->spatial_pred_err = _spatialPredErr;
+ _cMetrics->spatial_pred_err_h = _spatialPredErrH;
+ _cMetrics->spatial_pred_err_v = _spatialPredErrV;
+ // Motion metric: normalized temporal difference (MAD)
+ _cMetrics->motion_magnitude = _motionMagnitude;
+
+ return _cMetrics;
+
+}
+
+} // namespace
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis.h b/trunk/src/modules/video_processing/main/source/content_analysis.h
new file mode 100644
index 0000000..588712a
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/content_analysis.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VPM_CONTENT_ANALYSIS_H
+#define VPM_CONTENT_ANALYSIS_H
+
+#include "typedefs.h"
+#include "module_common_types.h"
+#include "video_processing_defines.h"
+
+namespace webrtc {
+
+class VPMContentAnalysis
+{
+public:
+ // When |runtime_cpu_detection| is true, runtime selection of an optimized
+ // code path is allowed.
+ VPMContentAnalysis(bool runtime_cpu_detection);
+ ~VPMContentAnalysis();
+
+ // Initialize ContentAnalysis - should be called prior to
+ // extractContentFeature
+ // Inputs: width, height
+ // Return value: 0 if OK, negative value upon error
+ WebRtc_Word32 Initialize(WebRtc_UWord16 width, WebRtc_UWord16 height);
+
+ // Extract content Feature - main function of ContentAnalysis
+ // Input: new frame
+ // Return value: pointer to structure containing content Analysis
+ // metrics or NULL value upon error
+ VideoContentMetrics* ComputeContentMetrics(const VideoFrame* inputFrame);
+
+ // Release all allocated memory
+ // Output: 0 if OK, negative value upon error
+ WebRtc_Word32 Release();
+
+private:
+
+ // return motion metrics
+ VideoContentMetrics* ContentMetrics();
+
+ // Normalized temporal difference metric: for motion magnitude
+ typedef WebRtc_Word32 (VPMContentAnalysis::*TemporalDiffMetricFunc)();
+ TemporalDiffMetricFunc TemporalDiffMetric;
+ WebRtc_Word32 TemporalDiffMetric_C();
+
+ // Motion metric method: call 2 metrics (magnitude and size)
+ WebRtc_Word32 ComputeMotionMetrics();
+
+ // Spatial metric method: computes the 3 frame-average spatial
+ // prediction errors (1x2,2x1,2x2)
+ typedef WebRtc_Word32 (VPMContentAnalysis::*ComputeSpatialMetricsFunc)();
+ ComputeSpatialMetricsFunc ComputeSpatialMetrics;
+ WebRtc_Word32 ComputeSpatialMetrics_C();
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+ WebRtc_Word32 ComputeSpatialMetrics_SSE2();
+ WebRtc_Word32 TemporalDiffMetric_SSE2();
+#endif
+
+ const WebRtc_UWord8* _origFrame;
+ WebRtc_UWord8* _prevFrame;
+ WebRtc_UWord16 _width;
+ WebRtc_UWord16 _height;
+ WebRtc_UWord32 _skipNum;
+ WebRtc_Word32 _border;
+
+ // Content Metrics:
+ // stores the local average of the metrics
+ float _motionMagnitude; // motion class
+ float _spatialPredErr; // spatial class
+ float _spatialPredErrH; // spatial class
+ float _spatialPredErrV; // spatial class
+ bool _firstFrame;
+ bool _CAInit;
+
+ VideoContentMetrics* _cMetrics;
+
+}; // end of VPMContentAnalysis class definition
+
+} // namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc b/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc
new file mode 100644
index 0000000..f505850
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc
@@ -0,0 +1,300 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "content_analysis.h"
+
+#include <emmintrin.h>
+#include <math.h>
+
+namespace webrtc {
+
+WebRtc_Word32
+VPMContentAnalysis::TemporalDiffMetric_SSE2()
+{
+ WebRtc_UWord32 numPixels = 0; // counter for # of pixels
+
+ const WebRtc_UWord8* imgBufO = _origFrame + _border*_width + _border;
+ const WebRtc_UWord8* imgBufP = _prevFrame + _border*_width + _border;
+
+ const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
+
+ __m128i sad_64 = _mm_setzero_si128();
+ __m128i sum_64 = _mm_setzero_si128();
+ __m128i sqsum_64 = _mm_setzero_si128();
+ const __m128i z = _mm_setzero_si128();
+
+ for(WebRtc_UWord16 i = 0; i < (_height - 2*_border); i += _skipNum)
+ {
+ __m128i sqsum_32 = _mm_setzero_si128();
+
+ const WebRtc_UWord8 *lineO = imgBufO;
+ const WebRtc_UWord8 *lineP = imgBufP;
+
+ // Work on 16 pixels at a time. For HD content with a width of 1920
+ // this loop will run ~67 times (depending on border). Maximum for
+ // abs(o-p) and sum(o) will be 255. _mm_sad_epu8 produces 2 64 bit
+ // results which are then accumulated. There is no chance of
+ // rollover for these two accumulators.
+ // o*o will have a maximum of 255*255 = 65025. This will roll over
+ // a 16 bit accumulator as 67*65025 > 65535, but will fit in a
+ // 32 bit accumulator.
+ for(WebRtc_UWord16 j = 0; j < width_end - _border; j += 16)
+ {
+ const __m128i o = _mm_loadu_si128((__m128i*)(lineO));
+ const __m128i p = _mm_loadu_si128((__m128i*)(lineP));
+
+ lineO += 16;
+ lineP += 16;
+
+ // abs pixel difference between frames
+ sad_64 = _mm_add_epi64 (sad_64, _mm_sad_epu8(o, p));
+
+ // sum of all pixels in frame
+ sum_64 = _mm_add_epi64 (sum_64, _mm_sad_epu8(o, z));
+
+ // squared sum of all pixels in frame
+ const __m128i olo = _mm_unpacklo_epi8(o,z);
+ const __m128i ohi = _mm_unpackhi_epi8(o,z);
+
+ const __m128i sqsum_32_lo = _mm_madd_epi16(olo, olo);
+ const __m128i sqsum_32_hi = _mm_madd_epi16(ohi, ohi);
+
+ sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_lo);
+ sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_hi);
+ }
+
+ // Add to 64 bit running sum as to not roll over.
+ sqsum_64 = _mm_add_epi64(sqsum_64,
+ _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32,z),
+ _mm_unpacklo_epi32(sqsum_32,z)));
+
+ imgBufO += _width * _skipNum;
+ imgBufP += _width * _skipNum;
+ numPixels += (width_end - _border);
+ }
+
+ __m128i sad_final_128;
+ __m128i sum_final_128;
+ __m128i sqsum_final_128;
+
+ // bring sums out of vector registers and into integer register
+ // domain, summing them along the way
+ _mm_store_si128 (&sad_final_128, sad_64);
+ _mm_store_si128 (&sum_final_128, sum_64);
+ _mm_store_si128 (&sqsum_final_128, sqsum_64);
+
+ WebRtc_UWord64 *sad_final_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&sad_final_128);
+ WebRtc_UWord64 *sum_final_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&sum_final_128);
+ WebRtc_UWord64 *sqsum_final_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&sqsum_final_128);
+
+ const WebRtc_UWord32 pixelSum = sum_final_64[0] + sum_final_64[1];
+ const WebRtc_UWord64 pixelSqSum = sqsum_final_64[0] + sqsum_final_64[1];
+ const WebRtc_UWord32 tempDiffSum = sad_final_64[0] + sad_final_64[1];
+
+ // default
+ _motionMagnitude = 0.0f;
+
+ if (tempDiffSum == 0)
+ {
+ return VPM_OK;
+ }
+
+ // normalize over all pixels
+ const float tempDiffAvg = (float)tempDiffSum / (float)(numPixels);
+ const float pixelSumAvg = (float)pixelSum / (float)(numPixels);
+ const float pixelSqSumAvg = (float)pixelSqSum / (float)(numPixels);
+ float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
+
+ if (contrast > 0.0)
+ {
+ contrast = sqrt(contrast);
+ _motionMagnitude = tempDiffAvg/contrast;
+ }
+
+ return VPM_OK;
+}
+
+WebRtc_Word32
+VPMContentAnalysis::ComputeSpatialMetrics_SSE2()
+{
+ const WebRtc_UWord8* imgBuf = _origFrame + _border*_width;
+ const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
+
+ __m128i se_32 = _mm_setzero_si128();
+ __m128i sev_32 = _mm_setzero_si128();
+ __m128i seh_32 = _mm_setzero_si128();
+ __m128i msa_32 = _mm_setzero_si128();
+ const __m128i z = _mm_setzero_si128();
+
+ // Error is accumulated as a 32 bit value. Looking at HD content with a
+ // height of 1080 lines, or about 67 macro blocks. If the 16 bit row
+ // value is maxed out at 65529 for every row, 65529*1080 = 70777800, which
+ // will not roll over a 32 bit accumulator.
+ // _skipNum is also used to reduce the number of rows
+ for(WebRtc_Word32 i = 0; i < (_height - 2*_border); i += _skipNum)
+ {
+ __m128i se_16 = _mm_setzero_si128();
+ __m128i sev_16 = _mm_setzero_si128();
+ __m128i seh_16 = _mm_setzero_si128();
+ __m128i msa_16 = _mm_setzero_si128();
+
+ // Row error is accumulated as a 16 bit value. There are 8
+ // accumulators. Max value of a 16 bit number is 65529. Looking
+ // at HD content, 1080p, has a width of 1920, 120 macro blocks.
+ // A mb at a time is processed at a time. Absolute max error at
+ // a point would be abs(0-255+255+255+255) which equals 1020.
+ // 120*1020 = 122400. The probability of hitting this is quite low
+ // on well behaved content. A specially crafted image could roll over.
+ // _border could also be adjusted to concentrate on just the center of
+ // the images for an HD capture in order to reduce the possiblity of
+ // rollover.
+ const WebRtc_UWord8 *lineTop = imgBuf - _width + _border;
+ const WebRtc_UWord8 *lineCen = imgBuf + _border;
+ const WebRtc_UWord8 *lineBot = imgBuf + _width + _border;
+
+ for(WebRtc_Word32 j = 0; j < width_end - _border; j += 16)
+ {
+ const __m128i t = _mm_loadu_si128((__m128i*)(lineTop));
+ const __m128i l = _mm_loadu_si128((__m128i*)(lineCen - 1));
+ const __m128i c = _mm_loadu_si128((__m128i*)(lineCen));
+ const __m128i r = _mm_loadu_si128((__m128i*)(lineCen + 1));
+ const __m128i b = _mm_loadu_si128((__m128i*)(lineBot));
+
+ lineTop += 16;
+ lineCen += 16;
+ lineBot += 16;
+
+ // center pixel unpacked
+ __m128i clo = _mm_unpacklo_epi8(c,z);
+ __m128i chi = _mm_unpackhi_epi8(c,z);
+
+ // left right pixels unpacked and added together
+ const __m128i lrlo = _mm_add_epi16(_mm_unpacklo_epi8(l,z),
+ _mm_unpacklo_epi8(r,z));
+ const __m128i lrhi = _mm_add_epi16(_mm_unpackhi_epi8(l,z),
+ _mm_unpackhi_epi8(r,z));
+
+ // top & bottom pixels unpacked and added together
+ const __m128i tblo = _mm_add_epi16(_mm_unpacklo_epi8(t,z),
+ _mm_unpacklo_epi8(b,z));
+ const __m128i tbhi = _mm_add_epi16(_mm_unpackhi_epi8(t,z),
+ _mm_unpackhi_epi8(b,z));
+
+ // running sum of all pixels
+ msa_16 = _mm_add_epi16(msa_16, _mm_add_epi16(chi, clo));
+
+ clo = _mm_slli_epi16(clo, 1);
+ chi = _mm_slli_epi16(chi, 1);
+ const __m128i sevtlo = _mm_subs_epi16(clo, tblo);
+ const __m128i sevthi = _mm_subs_epi16(chi, tbhi);
+ const __m128i sehtlo = _mm_subs_epi16(clo, lrlo);
+ const __m128i sehthi = _mm_subs_epi16(chi, lrhi);
+
+ clo = _mm_slli_epi16(clo, 1);
+ chi = _mm_slli_epi16(chi, 1);
+ const __m128i setlo = _mm_subs_epi16(clo,
+ _mm_add_epi16(lrlo, tblo));
+ const __m128i sethi = _mm_subs_epi16(chi,
+ _mm_add_epi16(lrhi, tbhi));
+
+ // Add to 16 bit running sum
+ se_16 = _mm_add_epi16(se_16,
+ _mm_max_epi16(setlo,
+ _mm_subs_epi16(z, setlo)));
+ se_16 = _mm_add_epi16(se_16,
+ _mm_max_epi16(sethi,
+ _mm_subs_epi16(z, sethi)));
+ sev_16 = _mm_add_epi16(sev_16,
+ _mm_max_epi16(sevtlo,
+ _mm_subs_epi16(z, sevtlo)));
+ sev_16 = _mm_add_epi16(sev_16,
+ _mm_max_epi16(sevthi,
+ _mm_subs_epi16(z, sevthi)));
+ seh_16 = _mm_add_epi16(seh_16,
+ _mm_max_epi16(sehtlo,
+ _mm_subs_epi16(z, sehtlo)));
+ seh_16 = _mm_add_epi16(seh_16,
+ _mm_max_epi16(sehthi,
+ _mm_subs_epi16(z, sehthi)));
+ }
+
+ // Add to 32 bit running sum as to not roll over.
+ se_32 = _mm_add_epi32(se_32,
+ _mm_add_epi32(_mm_unpackhi_epi16(se_16,z),
+ _mm_unpacklo_epi16(se_16,z)));
+ sev_32 = _mm_add_epi32(sev_32,
+ _mm_add_epi32(_mm_unpackhi_epi16(sev_16,z),
+ _mm_unpacklo_epi16(sev_16,z)));
+ seh_32 = _mm_add_epi32(seh_32,
+ _mm_add_epi32(_mm_unpackhi_epi16(seh_16,z),
+ _mm_unpacklo_epi16(seh_16,z)));
+ msa_32 = _mm_add_epi32(msa_32,
+ _mm_add_epi32(_mm_unpackhi_epi16(msa_16,z),
+ _mm_unpacklo_epi16(msa_16,z)));
+
+ imgBuf += _width * _skipNum;
+ }
+
+ __m128i se_128;
+ __m128i sev_128;
+ __m128i seh_128;
+ __m128i msa_128;
+
+ // bring sums out of vector registers and into integer register
+ // domain, summing them along the way
+ _mm_store_si128 (&se_128,
+ _mm_add_epi64(_mm_unpackhi_epi32(se_32,z),
+ _mm_unpacklo_epi32(se_32,z)));
+ _mm_store_si128 (&sev_128,
+ _mm_add_epi64(_mm_unpackhi_epi32(sev_32,z),
+ _mm_unpacklo_epi32(sev_32,z)));
+ _mm_store_si128 (&seh_128,
+ _mm_add_epi64(_mm_unpackhi_epi32(seh_32,z),
+ _mm_unpacklo_epi32(seh_32,z)));
+ _mm_store_si128 (&msa_128,
+ _mm_add_epi64(_mm_unpackhi_epi32(msa_32,z),
+ _mm_unpacklo_epi32(msa_32,z)));
+
+ WebRtc_UWord64 *se_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&se_128);
+ WebRtc_UWord64 *sev_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&sev_128);
+ WebRtc_UWord64 *seh_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&seh_128);
+ WebRtc_UWord64 *msa_64 =
+ reinterpret_cast<WebRtc_UWord64*>(&msa_128);
+
+ const WebRtc_UWord32 spatialErrSum = se_64[0] + se_64[1];
+ const WebRtc_UWord32 spatialErrVSum = sev_64[0] + sev_64[1];
+ const WebRtc_UWord32 spatialErrHSum = seh_64[0] + seh_64[1];
+ const WebRtc_UWord32 pixelMSA = msa_64[0] + msa_64[1];
+
+ // normalize over all pixels
+ const float spatialErr = (float)(spatialErrSum >> 2);
+ const float spatialErrH = (float)(spatialErrHSum >> 1);
+ const float spatialErrV = (float)(spatialErrVSum >> 1);
+ const float norm = (float)pixelMSA;
+
+ // 2X2:
+ _spatialPredErr = spatialErr / norm;
+
+ // 1X2:
+ _spatialPredErrH = spatialErrH / norm;
+
+ // 2X1:
+ _spatialPredErrV = spatialErrV / norm;
+
+ return VPM_OK;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/deflickering.cc b/trunk/src/modules/video_processing/main/source/deflickering.cc
new file mode 100644
index 0000000..d0b8d3b
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/deflickering.cc
@@ -0,0 +1,445 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "deflickering.h"
+#include "trace.h"
+#include "signal_processing_library.h"
+#include "sort.h"
+
+namespace webrtc {
+
+// Detection constants
+enum { kFrequencyDeviation = 39 }; // (Q4) Maximum allowed deviation for detection
+enum { kMinFrequencyToDetect = 32 }; // (Q4) Minimum frequency that can be detected
+enum { kNumFlickerBeforeDetect = 2 }; // Number of flickers before we accept detection
+enum { kMeanValueScaling = 4 }; // (Q4) In power of 2
+enum { kZeroCrossingDeadzone = 10 }; // Deadzone region in terms of pixel values
+
+// Deflickering constants
+// Compute the quantiles over 1 / DownsamplingFactor of the image.
+enum { kDownsamplingFactor = 8 };
+enum { kLog2OfDownsamplingFactor = 3 };
+
+// To generate in Matlab:
+// >> probUW16 = round(2^11 * [0.05,0.1,0.2,0.3,0.4,0.5,0.6,0.7,0.8,0.9,0.95,0.97]);
+// >> fprintf('%d, ', probUW16)
+// Resolution reduced to avoid overflow when multiplying with the (potentially) large
+// number of pixels.
+const WebRtc_UWord16 VPMDeflickering::_probUW16[kNumProbs] =
+ {102, 205, 410, 614, 819, 1024, 1229, 1434, 1638, 1843, 1946, 1987}; // <Q11>
+
+// To generate in Matlab:
+// >> numQuants = 14; maxOnlyLength = 5;
+// >> weightUW16 = round(2^15 * [linspace(0.5, 1.0, numQuants - maxOnlyLength)]);
+// >> fprintf('%d, %d,\n ', weightUW16);
+const WebRtc_UWord16 VPMDeflickering::_weightUW16[kNumQuants - kMaxOnlyLength] =
+ {16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768}; // <Q15>
+
+VPMDeflickering::VPMDeflickering() :
+ _id(0)
+{
+ Reset();
+}
+
+VPMDeflickering::~VPMDeflickering()
+{
+}
+
+WebRtc_Word32
+VPMDeflickering::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return 0;
+}
+
+void
+VPMDeflickering::Reset()
+{
+ _meanBufferLength = 0;
+ _detectionState = 0;
+ _frameRate = 0;
+
+ memset(_meanBuffer, 0, sizeof(WebRtc_Word32) * kMeanBufferLength);
+ memset(_timestampBuffer, 0, sizeof(WebRtc_Word32) * kMeanBufferLength);
+
+ // Initialize the history with a uniformly distributed histogram
+ _quantHistUW8[0][0] = 0;
+ _quantHistUW8[0][kNumQuants - 1] = 255;
+ for (WebRtc_Word32 i = 0; i < kNumProbs; i++)
+ {
+ _quantHistUW8[0][i + 1] = static_cast<WebRtc_UWord8>((WEBRTC_SPL_UMUL_16_16(
+ _probUW16[i], 255) + (1 << 10)) >> 11); // Unsigned round. <Q0>
+ }
+
+ for (WebRtc_Word32 i = 1; i < kFrameHistorySize; i++)
+ {
+ memcpy(_quantHistUW8[i], _quantHistUW8[0], sizeof(WebRtc_UWord8) * kNumQuants);
+ }
+}
+
+WebRtc_Word32
+VPMDeflickering::ProcessFrame(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height,
+ const WebRtc_UWord32 timestamp,
+ VideoProcessingModule::FrameStats& stats)
+{
+ WebRtc_UWord32 frameMemory;
+ WebRtc_UWord8 quantUW8[kNumQuants];
+ WebRtc_UWord8 maxQuantUW8[kNumQuants];
+ WebRtc_UWord8 minQuantUW8[kNumQuants];
+ WebRtc_UWord16 targetQuantUW16[kNumQuants];
+ WebRtc_UWord16 incrementUW16;
+ WebRtc_UWord8 mapUW8[256];
+
+ WebRtc_UWord16 tmpUW16;
+ WebRtc_UWord32 tmpUW32;
+
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
+ return VPM_GENERAL_ERROR;
+ }
+
+ // Stricter height check due to subsampling size calculation below.
+ if (width == 0 || height < 2)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
+ return VPM_GENERAL_ERROR;
+ }
+
+ if (!VideoProcessingModule::ValidFrameStats(stats))
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame stats");
+ return VPM_GENERAL_ERROR;
+ }
+
+ if (PreDetection(timestamp, stats) == -1)
+ {
+ return VPM_GENERAL_ERROR;
+ }
+
+ // Flicker detection
+ WebRtc_Word32 detFlicker = DetectFlicker();
+ if (detFlicker < 0)
+ { // Error
+ return VPM_GENERAL_ERROR;
+ }
+ else if (detFlicker != 1)
+ {
+ return 0;
+ }
+
+ // Size of luminance component
+ const WebRtc_UWord32 ySize = height * width;
+
+ const WebRtc_UWord32 ySubSize = width * (((height - 1) >>
+ kLog2OfDownsamplingFactor) + 1);
+ WebRtc_UWord8* ySorted = new WebRtc_UWord8[ySubSize];
+ WebRtc_UWord32 sortRowIdx = 0;
+ for (WebRtc_UWord32 i = 0; i < height; i += kDownsamplingFactor)
+ {
+ memcpy(ySorted + sortRowIdx * width, frame + i * width, width);
+ sortRowIdx++;
+ }
+
+ webrtc::Sort(ySorted, ySubSize, webrtc::TYPE_UWord8);
+
+ WebRtc_UWord32 probIdxUW32 = 0;
+ quantUW8[0] = 0;
+ quantUW8[kNumQuants - 1] = 255;
+
+ // Ensure we won't get an overflow below.
+ // In practice, the number of subsampled pixels will not become this large.
+ if (ySubSize > (1 << 21) - 1)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id,
+ "Subsampled number of pixels too large");
+ return -1;
+ }
+
+ for (WebRtc_Word32 i = 0; i < kNumProbs; i++)
+ {
+ probIdxUW32 = WEBRTC_SPL_UMUL_32_16(ySubSize, _probUW16[i]) >> 11; // <Q0>
+ quantUW8[i + 1] = ySorted[probIdxUW32];
+ }
+
+ delete [] ySorted;
+ ySorted = NULL;
+
+ // Shift history for new frame.
+ memmove(_quantHistUW8[1], _quantHistUW8[0], (kFrameHistorySize - 1) * kNumQuants *
+ sizeof(WebRtc_UWord8));
+ // Store current frame in history.
+ memcpy(_quantHistUW8[0], quantUW8, kNumQuants * sizeof(WebRtc_UWord8));
+
+ // We use a frame memory equal to the ceiling of half the frame rate to ensure we
+ // capture an entire period of flicker.
+ frameMemory = (_frameRate + (1 << 5)) >> 5; // Unsigned ceiling. <Q0>
+ // _frameRate in Q4.
+ if (frameMemory > kFrameHistorySize)
+ {
+ frameMemory = kFrameHistorySize;
+ }
+
+ // Get maximum and minimum.
+ for (WebRtc_Word32 i = 0; i < kNumQuants; i++)
+ {
+ maxQuantUW8[i] = 0;
+ minQuantUW8[i] = 255;
+ for (WebRtc_UWord32 j = 0; j < frameMemory; j++)
+ {
+ if (_quantHistUW8[j][i] > maxQuantUW8[i])
+ {
+ maxQuantUW8[i] = _quantHistUW8[j][i];
+ }
+
+ if (_quantHistUW8[j][i] < minQuantUW8[i])
+ {
+ minQuantUW8[i] = _quantHistUW8[j][i];
+ }
+ }
+ }
+
+ // Get target quantiles.
+ for (WebRtc_Word32 i = 0; i < kNumQuants - kMaxOnlyLength; i++)
+ {
+ targetQuantUW16[i] = static_cast<WebRtc_UWord16>((WEBRTC_SPL_UMUL_16_16(
+ _weightUW16[i], maxQuantUW8[i]) + WEBRTC_SPL_UMUL_16_16((1 << 15) -
+ _weightUW16[i], minQuantUW8[i])) >> 8); // <Q7>
+ }
+
+ for (WebRtc_Word32 i = kNumQuants - kMaxOnlyLength; i < kNumQuants; i++)
+ {
+ targetQuantUW16[i] = ((WebRtc_UWord16)maxQuantUW8[i]) << 7;
+ }
+
+ // Compute the map from input to output pixels.
+ WebRtc_UWord16 mapUW16; // <Q7>
+ for (WebRtc_Word32 i = 1; i < kNumQuants; i++)
+ {
+ // As quant and targetQuant are limited to UWord8, we're safe to use Q7 here.
+ tmpUW32 = static_cast<WebRtc_UWord32>(targetQuantUW16[i] -
+ targetQuantUW16[i - 1]); // <Q7>
+ tmpUW16 = static_cast<WebRtc_UWord16>(quantUW8[i] - quantUW8[i - 1]); // <Q0>
+
+ if (tmpUW16 > 0)
+ {
+ incrementUW16 = static_cast<WebRtc_UWord16>(WebRtcSpl_DivU32U16(tmpUW32,
+ tmpUW16)); // <Q7>
+ }
+ else
+ {
+ // The value is irrelevant; the loop below will only iterate once.
+ incrementUW16 = 0;
+ }
+
+ mapUW16 = targetQuantUW16[i - 1];
+ for (WebRtc_UWord32 j = quantUW8[i - 1]; j < (WebRtc_UWord32)(quantUW8[i] + 1); j++)
+ {
+ mapUW8[j] = (WebRtc_UWord8)((mapUW16 + (1 << 6)) >> 7); // Unsigned round. <Q0>
+ mapUW16 += incrementUW16;
+ }
+ }
+
+ // Map to the output frame.
+ for (WebRtc_UWord32 i = 0; i < ySize; i++)
+ {
+ frame[i] = mapUW8[frame[i]];
+ }
+
+ // Frame was altered, so reset stats.
+ VideoProcessingModule::ClearFrameStats(stats);
+
+ return 0;
+}
+
+/**
+ Performs some pre-detection operations. Must be called before
+ DetectFlicker().
+
+ \param[in] timestamp Timestamp of the current frame.
+ \param[in] stats Statistics of the current frame.
+
+ \return 0: Success\n
+ 2: Detection not possible due to flickering frequency too close to
+ zero.\n
+ -1: Error
+*/
+WebRtc_Word32
+VPMDeflickering::PreDetection(const WebRtc_UWord32 timestamp,
+ const VideoProcessingModule::FrameStats& stats)
+{
+ WebRtc_Word32 meanVal; // Mean value of frame (Q4)
+ WebRtc_UWord32 frameRate = 0;
+ WebRtc_Word32 meanBufferLength; // Temp variable
+
+ meanVal = ((stats.sum << kMeanValueScaling) / stats.numPixels);
+ /* Update mean value buffer.
+ * This should be done even though we might end up in an unreliable detection.
+ */
+ memmove(_meanBuffer + 1, _meanBuffer, (kMeanBufferLength - 1) * sizeof(WebRtc_Word32));
+ _meanBuffer[0] = meanVal;
+
+ /* Update timestamp buffer.
+ * This should be done even though we might end up in an unreliable detection.
+ */
+ memmove(_timestampBuffer + 1, _timestampBuffer, (kMeanBufferLength - 1) *
+ sizeof(WebRtc_UWord32));
+ _timestampBuffer[0] = timestamp;
+
+ /* Compute current frame rate (Q4) */
+ if (_timestampBuffer[kMeanBufferLength - 1] != 0)
+ {
+ frameRate = ((90000 << 4) * (kMeanBufferLength - 1));
+ frameRate /= (_timestampBuffer[0] - _timestampBuffer[kMeanBufferLength - 1]);
+ }else if (_timestampBuffer[1] != 0)
+ {
+ frameRate = (90000 << 4) / (_timestampBuffer[0] - _timestampBuffer[1]);
+ }
+
+ /* Determine required size of mean value buffer (_meanBufferLength) */
+ if (frameRate == 0) {
+ meanBufferLength = 1;
+ }
+ else {
+ meanBufferLength = (kNumFlickerBeforeDetect * frameRate) / kMinFrequencyToDetect;
+ }
+ /* Sanity check of buffer length */
+ if (meanBufferLength >= kMeanBufferLength)
+ {
+ /* Too long buffer. The flickering frequency is too close to zero, which
+ * makes the estimation unreliable.
+ */
+ _meanBufferLength = 0;
+ return 2;
+ }
+ _meanBufferLength = meanBufferLength;
+
+ if ((_timestampBuffer[_meanBufferLength - 1] != 0) && (_meanBufferLength != 1))
+ {
+ frameRate = ((90000 << 4) * (_meanBufferLength - 1));
+ frameRate /= (_timestampBuffer[0] - _timestampBuffer[_meanBufferLength - 1]);
+ }else if (_timestampBuffer[1] != 0)
+ {
+ frameRate = (90000 << 4) / (_timestampBuffer[0] - _timestampBuffer[1]);
+ }
+ _frameRate = frameRate;
+
+ return 0;
+}
+
+/**
+ This function detects flicker in the video stream. As a side effect the mean value
+ buffer is updated with the new mean value.
+
+ \return 0: No flickering detected\n
+ 1: Flickering detected\n
+ 2: Detection not possible due to unreliable frequency interval
+ -1: Error
+*/
+WebRtc_Word32 VPMDeflickering::DetectFlicker()
+{
+ /* Local variables */
+ WebRtc_UWord32 i;
+ WebRtc_Word32 freqEst; // (Q4) Frequency estimate to base detection upon
+ WebRtc_Word32 retVal = -1;
+
+ /* Sanity check for _meanBufferLength */
+ if (_meanBufferLength < 2)
+ {
+ /* Not possible to estimate frequency */
+ return(2);
+ }
+ /* Count zero crossings with a dead zone to be robust against noise.
+ * If the noise std is 2 pixel this corresponds to about 95% confidence interval.
+ */
+ WebRtc_Word32 deadzone = (kZeroCrossingDeadzone << kMeanValueScaling); // Q4
+ WebRtc_Word32 meanOfBuffer = 0; // Mean value of mean value buffer
+ WebRtc_Word32 numZeros = 0; // Number of zeros that cross the deadzone
+ WebRtc_Word32 cntState = 0; // State variable for zero crossing regions
+ WebRtc_Word32 cntStateOld = 0; // Previous state variable for zero crossing regions
+
+ for (i = 0; i < _meanBufferLength; i++)
+ {
+ meanOfBuffer += _meanBuffer[i];
+ }
+ meanOfBuffer += (_meanBufferLength >> 1); // Rounding, not truncation
+ meanOfBuffer /= _meanBufferLength;
+
+ /* Count zero crossings */
+ cntStateOld = (_meanBuffer[0] >= (meanOfBuffer + deadzone));
+ cntStateOld -= (_meanBuffer[0] <= (meanOfBuffer - deadzone));
+ for (i = 1; i < _meanBufferLength; i++)
+ {
+ cntState = (_meanBuffer[i] >= (meanOfBuffer + deadzone));
+ cntState -= (_meanBuffer[i] <= (meanOfBuffer - deadzone));
+ if (cntStateOld == 0)
+ {
+ cntStateOld = -cntState;
+ }
+ if (((cntState + cntStateOld) == 0) && (cntState != 0))
+ {
+ numZeros++;
+ cntStateOld = cntState;
+ }
+ }
+ /* END count zero crossings */
+
+ /* Frequency estimation according to:
+ * freqEst = numZeros * frameRate / 2 / _meanBufferLength;
+ *
+ * Resolution is set to Q4
+ */
+ freqEst = ((numZeros * 90000) << 3);
+ freqEst /= (_timestampBuffer[0] - _timestampBuffer[_meanBufferLength - 1]);
+
+ /* Translate frequency estimate to regions close to 100 and 120 Hz */
+ WebRtc_UWord8 freqState = 0; // Current translation state;
+ // (0) Not in interval,
+ // (1) Within valid interval,
+ // (2) Out of range
+ WebRtc_Word32 freqAlias = freqEst;
+ if (freqEst > kMinFrequencyToDetect)
+ {
+ WebRtc_UWord8 aliasState = 1;
+ while(freqState == 0)
+ {
+ /* Increase frequency */
+ freqAlias += (aliasState * _frameRate);
+ freqAlias += ((freqEst << 1) * (1 - (aliasState << 1)));
+ /* Compute state */
+ freqState = (abs(freqAlias - (100 << 4)) <= kFrequencyDeviation);
+ freqState += (abs(freqAlias - (120 << 4)) <= kFrequencyDeviation);
+ freqState += 2 * (freqAlias > ((120 << 4) + kFrequencyDeviation));
+ /* Switch alias state */
+ aliasState++;
+ aliasState &= 0x01;
+ }
+ }
+ /* Is frequency estimate within detection region? */
+ if (freqState == 1)
+ {
+ retVal = 1;
+ }else if (freqState == 0)
+ {
+ retVal = 2;
+ }else
+ {
+ retVal = 0;
+ }
+ return retVal;
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/deflickering.h b/trunk/src/modules/video_processing/main/source/deflickering.h
new file mode 100644
index 0000000..ee5f90d
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/deflickering.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * deflickering.h
+ */
+
+#ifndef VPM_DEFLICKERING_H
+#define VPM_DEFLICKERING_H
+
+#include "typedefs.h"
+#include "video_processing.h"
+
+#include <cstring> // NULL
+
+namespace webrtc {
+
+class VPMDeflickering
+{
+public:
+ VPMDeflickering();
+ ~VPMDeflickering();
+
+ WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
+
+ void Reset();
+
+ WebRtc_Word32 ProcessFrame(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ WebRtc_UWord32 timestamp,
+ VideoProcessingModule::FrameStats& stats);
+private:
+ WebRtc_Word32 PreDetection(WebRtc_UWord32 timestamp,
+ const VideoProcessingModule::FrameStats& stats);
+
+ WebRtc_Word32 DetectFlicker();
+
+ enum { kMeanBufferLength = 32 };
+ enum { kFrameHistorySize = 15 };
+ enum { kNumProbs = 12 };
+ enum { kNumQuants = kNumProbs + 2 };
+ enum { kMaxOnlyLength = 5 };
+
+ WebRtc_Word32 _id;
+
+ WebRtc_UWord32 _meanBufferLength;
+ WebRtc_UWord8 _detectionState; // 0: No flickering
+ // 1: Flickering detected
+ // 2: In flickering
+ WebRtc_Word32 _meanBuffer[kMeanBufferLength];
+ WebRtc_UWord32 _timestampBuffer[kMeanBufferLength];
+ WebRtc_UWord32 _frameRate;
+ static const WebRtc_UWord16 _probUW16[kNumProbs];
+ static const WebRtc_UWord16 _weightUW16[kNumQuants - kMaxOnlyLength];
+ WebRtc_UWord8 _quantHistUW8[kFrameHistorySize][kNumQuants];
+};
+
+} //namespace
+
+#endif // VPM_DEFLICKERING_H
+
diff --git a/trunk/src/modules/video_processing/main/source/denoising.cc b/trunk/src/modules/video_processing/main/source/denoising.cc
new file mode 100644
index 0000000..d8931c9
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/denoising.cc
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "denoising.h"
+#include "trace.h"
+
+#include <cstring>
+
+namespace webrtc {
+
+enum { kSubsamplingTime = 0 }; // Down-sampling in time (unit: number of frames)
+enum { kSubsamplingWidth = 0 }; // Sub-sampling in width (unit: power of 2)
+enum { kSubsamplingHeight = 0 }; // Sub-sampling in height (unit: power of 2)
+enum { kDenoiseFiltParam = 179 }; // (Q8) De-noising filter parameter
+enum { kDenoiseFiltParamRec = 77 }; // (Q8) 1 - filter parameter
+enum { kDenoiseThreshold = 19200 }; // (Q8) De-noising threshold level
+
+VPMDenoising::VPMDenoising() :
+ _id(0),
+ _moment1(NULL),
+ _moment2(NULL)
+{
+ Reset();
+}
+
+VPMDenoising::~VPMDenoising()
+{
+ if (_moment1)
+ {
+ delete [] _moment1;
+ _moment1 = NULL;
+ }
+
+ if (_moment2)
+ {
+ delete [] _moment2;
+ _moment2 = NULL;
+ }
+}
+
+WebRtc_Word32
+VPMDenoising::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return VPM_OK;
+}
+
+void
+VPMDenoising::Reset()
+{
+ _frameSize = 0;
+ _denoiseFrameCnt = 0;
+
+ if (_moment1)
+ {
+ delete [] _moment1;
+ _moment1 = NULL;
+ }
+
+ if (_moment2)
+ {
+ delete [] _moment2;
+ _moment2 = NULL;
+ }
+}
+
+WebRtc_Word32
+VPMDenoising::ProcessFrame(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ WebRtc_Word32 thevar;
+ WebRtc_UWord32 k;
+ WebRtc_UWord32 jsub, ksub;
+ WebRtc_Word32 diff0;
+ WebRtc_UWord32 tmpMoment1;
+ WebRtc_UWord32 tmpMoment2;
+ WebRtc_UWord32 tmp;
+ WebRtc_Word32 numPixelsChanged = 0;
+
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
+ return VPM_GENERAL_ERROR;
+ }
+
+ if (width == 0 || height == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
+ return VPM_GENERAL_ERROR;
+ }
+
+ /* Size of luminance component */
+ const WebRtc_UWord32 ysize = height * width;
+
+ /* Initialization */
+ if (ysize != _frameSize)
+ {
+ delete [] _moment1;
+ _moment1 = NULL;
+
+ delete [] _moment2;
+ _moment2 = NULL;
+ }
+ _frameSize = ysize;
+
+ if (!_moment1)
+ {
+ _moment1 = new WebRtc_UWord32[ysize];
+ memset(_moment1, 0, sizeof(WebRtc_UWord32)*ysize);
+ }
+
+ if (!_moment2)
+ {
+ _moment2 = new WebRtc_UWord32[ysize];
+ memset(_moment2, 0, sizeof(WebRtc_UWord32)*ysize);
+ }
+
+ /* Apply de-noising on each pixel, but update variance sub-sampled */
+ for (WebRtc_UWord32 i = 0; i < height; i++)
+ { // Collect over height
+ k = i * width;
+ ksub = ((i >> kSubsamplingHeight) << kSubsamplingHeight) * width;
+ for (WebRtc_UWord32 j = 0; j < width; j++)
+ { // Collect over width
+ jsub = ((j >> kSubsamplingWidth) << kSubsamplingWidth);
+ /* Update mean value for every pixel and every frame */
+ tmpMoment1 = _moment1[k + j];
+ tmpMoment1 *= kDenoiseFiltParam; // Q16
+ tmpMoment1 += ((kDenoiseFiltParamRec * ((WebRtc_UWord32)frame[k + j])) << 8);
+ tmpMoment1 >>= 8; // Q8
+ _moment1[k + j] = tmpMoment1;
+
+ tmpMoment2 = _moment2[ksub + jsub];
+ if ((ksub == k) && (jsub == j) && (_denoiseFrameCnt == 0))
+ {
+ tmp = ((WebRtc_UWord32)frame[k + j] * (WebRtc_UWord32)frame[k + j]);
+ tmpMoment2 *= kDenoiseFiltParam; // Q16
+ tmpMoment2 += ((kDenoiseFiltParamRec * tmp)<<8);
+ tmpMoment2 >>= 8; // Q8
+ }
+ _moment2[k + j] = tmpMoment2;
+ /* Current event = deviation from mean value */
+ diff0 = ((WebRtc_Word32)frame[k + j] << 8) - _moment1[k + j];
+ /* Recent events = variance (variations over time) */
+ thevar = _moment2[k + j];
+ thevar -= ((_moment1[k + j] * _moment1[k + j]) >> 8);
+ /***************************************************************************
+ * De-noising criteria, i.e., when should we replace a pixel by its mean
+ *
+ * 1) recent events are minor
+ * 2) current events are minor
+ ***************************************************************************/
+ if ((thevar < kDenoiseThreshold)
+ && ((diff0 * diff0 >> 8) < kDenoiseThreshold))
+ { // Replace with mean
+ frame[k + j] = (WebRtc_UWord8)(_moment1[k + j] >> 8);
+ numPixelsChanged++;
+ }
+ }
+ }
+
+ /* Update frame counter */
+ _denoiseFrameCnt++;
+ if (_denoiseFrameCnt > kSubsamplingTime)
+ {
+ _denoiseFrameCnt = 0;
+ }
+
+ return numPixelsChanged;
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/denoising.h b/trunk/src/modules/video_processing/main/source/denoising.h
new file mode 100644
index 0000000..f53157c
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/denoising.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * denoising.h
+ */
+#ifndef VPM_DENOISING_H
+#define VPM_DENOISING_H
+
+#include "typedefs.h"
+#include "video_processing.h"
+
+namespace webrtc {
+
+class VPMDenoising
+{
+public:
+ VPMDenoising();
+ ~VPMDenoising();
+
+ WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
+
+ void Reset();
+
+ WebRtc_Word32 ProcessFrame(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height);
+
+private:
+ WebRtc_Word32 _id;
+
+ WebRtc_UWord32* _moment1; // (Q8) First order moment (mean)
+ WebRtc_UWord32* _moment2; // (Q8) Second order moment
+ WebRtc_UWord32 _frameSize; // Size (# of pixels) of frame
+ WebRtc_Word32 _denoiseFrameCnt; // Counter for subsampling in time
+};
+
+} //namespace
+
+#endif // VPM_DENOISING_H
+
diff --git a/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc b/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc
new file mode 100644
index 0000000..76fdac8
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc
@@ -0,0 +1,182 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "frame_preprocessor.h"
+#include "trace.h"
+
+namespace webrtc {
+
+VPMFramePreprocessor::VPMFramePreprocessor():
+_id(0),
+_contentMetrics(NULL),
+_maxFrameRate(0),
+_resampledFrame(),
+_enableCA(false)
+{
+ _spatialResampler = new VPMSimpleSpatialResampler();
+ _ca = new VPMContentAnalysis(true);
+ _vd = new VPMVideoDecimator();
+}
+
+VPMFramePreprocessor::~VPMFramePreprocessor()
+{
+ Reset();
+ delete _spatialResampler;
+ delete _ca;
+ delete _vd;
+ _resampledFrame.Free(); // is this needed?
+}
+
+WebRtc_Word32
+VPMFramePreprocessor::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ _id = id;
+ return VPM_OK;
+}
+
+void
+VPMFramePreprocessor::Reset()
+{
+ _ca->Release();
+ _vd->Reset();
+ _contentMetrics = NULL;
+ _spatialResampler->Reset();
+ _enableCA = false;
+}
+
+
+void
+VPMFramePreprocessor::EnableTemporalDecimation(bool enable)
+{
+ _vd->EnableTemporalDecimation(enable);
+}
+void
+VPMFramePreprocessor::EnableContentAnalysis(bool enable)
+{
+ _enableCA = enable;
+}
+
+void
+VPMFramePreprocessor::SetInputFrameResampleMode(VideoFrameResampling resamplingMode)
+{
+ _spatialResampler->SetInputFrameResampleMode(resamplingMode);
+}
+
+
+WebRtc_Word32
+VPMFramePreprocessor::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
+{
+ if (maxFrameRate == 0)
+ {
+ return VPM_PARAMETER_ERROR;
+ }
+ //Max allowed frame rate
+ _maxFrameRate = maxFrameRate;
+
+ return _vd->SetMaxFrameRate(maxFrameRate);
+}
+
+
+WebRtc_Word32
+VPMFramePreprocessor::SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate)
+{
+ if ( (width == 0) || (height == 0) || (frameRate == 0))
+ {
+ return VPM_PARAMETER_ERROR;
+ }
+ WebRtc_Word32 retVal = 0;
+ retVal = _spatialResampler->SetTargetFrameSize(width, height);
+ if (retVal < 0)
+ {
+ return retVal;
+ }
+ retVal = _vd->SetTargetFrameRate(frameRate);
+ if (retVal < 0)
+ {
+ return retVal;
+ }
+
+ return VPM_OK;
+}
+
+void
+VPMFramePreprocessor::UpdateIncomingFrameRate()
+{
+ _vd->UpdateIncomingFrameRate();
+}
+
+WebRtc_UWord32
+VPMFramePreprocessor::DecimatedFrameRate()
+{
+ return _vd->DecimatedFrameRate();
+}
+
+
+WebRtc_UWord32
+VPMFramePreprocessor::DecimatedWidth() const
+{
+ return _spatialResampler->TargetWidth();
+}
+
+
+WebRtc_UWord32
+VPMFramePreprocessor::DecimatedHeight() const
+{
+ return _spatialResampler->TargetHeight();
+}
+
+
+WebRtc_Word32
+VPMFramePreprocessor::PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame)
+{
+ if (frame == NULL || frame->Height() == 0 || frame->Width() == 0)
+ {
+ return VPM_PARAMETER_ERROR;
+ }
+
+ _vd->UpdateIncomingFrameRate();
+
+ if (_vd->DropFrame())
+ {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, _id, "Drop frame due to frame rate");
+ return 1; // drop 1 frame
+ }
+
+ // Resizing incoming frame if needed.
+ // Note that we must make a copy of it.
+ // We are not allowed to resample the input frame.
+ *processedFrame = NULL;
+ if (_spatialResampler->ApplyResample(frame->Width(), frame->Height())) {
+ WebRtc_Word32 ret = _spatialResampler->ResampleFrame(*frame, _resampledFrame);
+ if (ret != VPM_OK)
+ return ret;
+ *processedFrame = &_resampledFrame;
+ }
+
+ // Perform content analysis on the frame to be encoded
+ if (_enableCA)
+ {
+ if (*processedFrame == NULL) {
+ _contentMetrics = _ca->ComputeContentMetrics(frame);
+ } else {
+ _contentMetrics = _ca->ComputeContentMetrics(&_resampledFrame);
+ }
+ }
+ return VPM_OK;
+}
+
+
+VideoContentMetrics*
+VPMFramePreprocessor::ContentMetrics() const
+{
+ return _contentMetrics;
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/frame_preprocessor.h b/trunk/src/modules/video_processing/main/source/frame_preprocessor.h
new file mode 100644
index 0000000..3c07a47
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/frame_preprocessor.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * frame_preprocessor.h
+ */
+#ifndef VPM_FRAME_PREPROCESSOR_H
+#define VPM_FRAME_PREPROCESSOR_H
+
+#include "typedefs.h"
+#include "video_processing.h"
+#include "content_analysis.h"
+#include "spatial_resampler.h"
+#include "video_decimator.h"
+
+namespace webrtc {
+
+
+class VPMFramePreprocessor
+{
+public:
+
+ VPMFramePreprocessor();
+ ~VPMFramePreprocessor();
+
+ WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ void Reset();
+
+ // Enable temporal decimation
+ void EnableTemporalDecimation(bool enable);
+
+ void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
+
+ //Enable content analysis
+ void EnableContentAnalysis(bool enable);
+
+ //Set max frame rate
+ WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
+
+ //Set target resolution: frame rate and dimension
+ WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate);
+
+ //Update incoming frame rate/dimension
+ void UpdateIncomingFrameRate();
+
+ WebRtc_Word32 updateIncomingFrameSize(WebRtc_UWord32 width, WebRtc_UWord32 height);
+
+ //Set decimated values: frame rate/dimension
+ WebRtc_UWord32 DecimatedFrameRate();
+ WebRtc_UWord32 DecimatedWidth() const;
+ WebRtc_UWord32 DecimatedHeight() const;
+
+ //Preprocess output:
+ WebRtc_Word32 PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame);
+ VideoContentMetrics* ContentMetrics() const;
+
+private:
+
+ WebRtc_Word32 _id;
+ VideoContentMetrics* _contentMetrics;
+ WebRtc_UWord32 _maxFrameRate;
+ VideoFrame _resampledFrame;
+ VPMSpatialResampler* _spatialResampler;
+ VPMContentAnalysis* _ca;
+ VPMVideoDecimator* _vd;
+ bool _enableCA;
+
+}; // end of VPMFramePreprocessor class definition
+
+} //namespace
+
+#endif // VPM_FRAME_PREPROCESS_H
diff --git a/trunk/src/modules/video_processing/main/source/spatial_resampler.cc b/trunk/src/modules/video_processing/main/source/spatial_resampler.cc
new file mode 100644
index 0000000..b48c322
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/spatial_resampler.cc
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "spatial_resampler.h"
+
+
+namespace webrtc {
+
+VPMSimpleSpatialResampler::VPMSimpleSpatialResampler()
+:
+_resamplingMode(kFastRescaling),
+_targetWidth(0),
+_targetHeight(0),
+_scaler()
+{
+}
+
+VPMSimpleSpatialResampler::~VPMSimpleSpatialResampler()
+{
+ //
+}
+
+
+WebRtc_Word32
+VPMSimpleSpatialResampler::SetTargetFrameSize(WebRtc_Word32 width,
+ WebRtc_Word32 height)
+{
+ if (_resamplingMode == kNoRescaling) {
+ return VPM_OK;
+ }
+
+ if (width < 1 || height < 1) {
+ return VPM_PARAMETER_ERROR;
+ }
+
+ _targetWidth = width;
+ _targetHeight = height;
+
+ return VPM_OK;
+}
+
+void
+VPMSimpleSpatialResampler::SetInputFrameResampleMode(VideoFrameResampling
+ resamplingMode)
+{
+ _resamplingMode = resamplingMode;
+}
+
+void
+VPMSimpleSpatialResampler::Reset()
+{
+ _resamplingMode = kFastRescaling;
+ _targetWidth = 0;
+ _targetHeight = 0;
+}
+
+WebRtc_Word32
+VPMSimpleSpatialResampler::ResampleFrame(const VideoFrame& inFrame,
+ VideoFrame& outFrame)
+{
+ if (_resamplingMode == kNoRescaling)
+ return outFrame.CopyFrame(inFrame);
+ // Check if re-sampling is needed
+ if ((inFrame.Width() == (WebRtc_UWord32)_targetWidth) &&
+ (inFrame.Height() == (WebRtc_UWord32)_targetHeight)) {
+ return outFrame.CopyFrame(inFrame);
+ }
+
+ // Setting scaler
+ //TODO: Modify scaler types
+ int retVal = 0;
+ retVal = _scaler.Set(inFrame.Width(), inFrame.Height(),
+ _targetWidth, _targetHeight, kI420, kI420, kScaleBox);
+ if (retVal < 0)
+ return retVal;
+
+
+ // Disabling cut/pad for now - only scaling.
+ int requiredSize = (WebRtc_UWord32)(_targetWidth * _targetHeight * 3 >> 1);
+ outFrame.VerifyAndAllocate(requiredSize);
+ outFrame.SetTimeStamp(inFrame.TimeStamp());
+ outFrame.SetWidth(_targetWidth);
+ outFrame.SetHeight(_targetHeight);
+
+ retVal = _scaler.Scale(inFrame.Buffer(), outFrame.Buffer(), requiredSize);
+ outFrame.SetLength(requiredSize);
+ if (retVal == 0)
+ return VPM_OK;
+ else
+ return VPM_SCALE_ERROR;
+}
+
+WebRtc_Word32
+VPMSimpleSpatialResampler::TargetHeight()
+{
+ return _targetHeight;
+}
+
+WebRtc_Word32
+VPMSimpleSpatialResampler::TargetWidth()
+{
+ return _targetWidth;
+}
+
+bool
+VPMSimpleSpatialResampler::ApplyResample(WebRtc_Word32 width,
+ WebRtc_Word32 height)
+{
+ if ((width == _targetWidth && height == _targetHeight) ||
+ _resamplingMode == kNoRescaling)
+ return false;
+ else
+ return true;
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/spatial_resampler.h b/trunk/src/modules/video_processing/main/source/spatial_resampler.h
new file mode 100644
index 0000000..76a63fb
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/spatial_resampler.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * spatial_resampler.h
+ */
+
+#ifndef VPM_SPATIAL_RESAMPLER_H
+#define VPM_SPATIAL_RESAMPLER_H
+
+#include "typedefs.h"
+
+#include "module_common_types.h"
+#include "video_processing_defines.h"
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "common_video/libyuv/include/scaler.h"
+
+namespace webrtc {
+
+class VPMSpatialResampler
+{
+public:
+ virtual ~VPMSpatialResampler() {};
+ virtual WebRtc_Word32 SetTargetFrameSize(WebRtc_Word32 width,
+ WebRtc_Word32 height) = 0;
+ virtual void SetInputFrameResampleMode(VideoFrameResampling
+ resamplingMode) = 0;
+ virtual void Reset() = 0;
+ virtual WebRtc_Word32 ResampleFrame(const VideoFrame& inFrame,
+ VideoFrame& outFrame) = 0;
+ virtual WebRtc_Word32 TargetWidth() = 0;
+ virtual WebRtc_Word32 TargetHeight() = 0;
+ virtual bool ApplyResample(WebRtc_Word32 width, WebRtc_Word32 height) = 0;
+};
+
+class VPMSimpleSpatialResampler : public VPMSpatialResampler
+{
+public:
+ VPMSimpleSpatialResampler();
+ ~VPMSimpleSpatialResampler();
+ virtual WebRtc_Word32 SetTargetFrameSize(WebRtc_Word32 width,
+ WebRtc_Word32 height);
+ virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
+ virtual void Reset();
+ virtual WebRtc_Word32 ResampleFrame(const VideoFrame& inFrame,
+ VideoFrame& outFrame);
+ virtual WebRtc_Word32 TargetWidth();
+ virtual WebRtc_Word32 TargetHeight();
+ virtual bool ApplyResample(WebRtc_Word32 width, WebRtc_Word32 height);
+
+private:
+
+ VideoFrameResampling _resamplingMode;
+ WebRtc_Word32 _targetWidth;
+ WebRtc_Word32 _targetHeight;
+ Scaler _scaler;
+};
+
+} //namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/source/video_decimator.cc b/trunk/src/modules/video_processing/main/source/video_decimator.cc
new file mode 100644
index 0000000..43bda08
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/video_decimator.cc
@@ -0,0 +1,235 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_decimator.h"
+#include "tick_util.h"
+#include "video_processing.h"
+
+#define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
+
+namespace webrtc {
+
+VPMVideoDecimator::VPMVideoDecimator()
+:
+_overShootModifier(0),
+_dropCount(0),
+_keepCount(0),
+_targetFrameRate(30),
+_incomingFrameRate(0.0f),
+_maxFrameRate(30),
+_incomingFrameTimes(),
+_enableTemporalDecimation(true)
+{
+ Reset();
+}
+
+VPMVideoDecimator::~VPMVideoDecimator()
+{
+ //
+}
+
+void
+VPMVideoDecimator::Reset()
+{
+ _overShootModifier = 0;
+ _dropCount = 0;
+ _keepCount = 0;
+ _targetFrameRate = 30;
+ _incomingFrameRate = 0.0f;
+ _maxFrameRate = 30;
+ memset(_incomingFrameTimes, 0, sizeof(_incomingFrameTimes));
+ _enableTemporalDecimation = true;
+}
+
+void
+VPMVideoDecimator::EnableTemporalDecimation(bool enable)
+{
+ _enableTemporalDecimation = enable;
+}
+WebRtc_Word32
+VPMVideoDecimator::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
+{
+ if (maxFrameRate == 0)
+ {
+ return VPM_PARAMETER_ERROR;
+ }
+
+ _maxFrameRate = maxFrameRate;
+
+ if (_targetFrameRate > _maxFrameRate)
+ {
+ _targetFrameRate = _maxFrameRate;
+
+ }
+ return VPM_OK;
+}
+
+WebRtc_Word32
+VPMVideoDecimator::SetTargetFrameRate(WebRtc_UWord32 frameRate)
+{
+ if (frameRate == 0)
+ {
+ return VPM_PARAMETER_ERROR;
+ }
+ if (frameRate > _maxFrameRate)
+ {
+ //override
+ _targetFrameRate = _maxFrameRate;
+ }
+ else
+ {
+ _targetFrameRate = frameRate;
+ }
+ return VPM_OK;
+}
+
+bool
+VPMVideoDecimator::DropFrame()
+{
+ if (!_enableTemporalDecimation)
+ {
+ return false;
+ }
+
+ if (_incomingFrameRate <= 0)
+ {
+ return false;
+ }
+
+ const WebRtc_UWord32 incomingFrameRate = static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
+
+ if (_targetFrameRate == 0)
+ {
+ return true;
+ }
+
+ bool drop = false;
+ if (incomingFrameRate > _targetFrameRate)
+ {
+ WebRtc_Word32 overshoot = _overShootModifier + (incomingFrameRate - _targetFrameRate);
+ if(overshoot < 0)
+ {
+ overshoot = 0;
+ _overShootModifier = 0;
+ }
+
+ if (overshoot && 2 * overshoot < (WebRtc_Word32) incomingFrameRate)
+ {
+
+ if (_dropCount) // Just got here so drop to be sure.
+ {
+ _dropCount = 0;
+ return true;
+ }
+ const WebRtc_UWord32 dropVar = incomingFrameRate / overshoot;
+
+ if (_keepCount >= dropVar)
+ {
+ drop = true;
+ _overShootModifier = -((WebRtc_Word32) incomingFrameRate % overshoot) / 3;
+ _keepCount = 1;
+ }
+ else
+ {
+
+ _keepCount++;
+ }
+ }
+ else
+ {
+ _keepCount = 0;
+ const WebRtc_UWord32 dropVar = overshoot / _targetFrameRate;
+ if (_dropCount < dropVar)
+ {
+ drop = true;
+ _dropCount++;
+ }
+ else
+ {
+ _overShootModifier = overshoot % _targetFrameRate;
+ drop = false;
+ _dropCount = 0;
+ }
+ }
+ }
+
+ return drop;
+}
+
+
+WebRtc_UWord32
+VPMVideoDecimator::DecimatedFrameRate()
+{
+ ProcessIncomingFrameRate(TickTime::MillisecondTimestamp());
+ if (!_enableTemporalDecimation)
+ {
+ return static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
+ }
+ return VD_MIN(_targetFrameRate, static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f));
+}
+
+WebRtc_UWord32
+VPMVideoDecimator::InputFrameRate()
+{
+ ProcessIncomingFrameRate(TickTime::MillisecondTimestamp());
+ return static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
+}
+
+void
+VPMVideoDecimator::UpdateIncomingFrameRate()
+{
+ WebRtc_Word64 now = TickTime::MillisecondTimestamp();
+ if(_incomingFrameTimes[0] == 0)
+ {
+ // first no shift
+ } else
+ {
+ // shift
+ for(int i = (kFrameCountHistorySize - 2); i >= 0 ; i--)
+ {
+ _incomingFrameTimes[i+1] = _incomingFrameTimes[i];
+ }
+ }
+ _incomingFrameTimes[0] = now;
+ ProcessIncomingFrameRate(now);
+}
+
+void
+VPMVideoDecimator::ProcessIncomingFrameRate(WebRtc_Word64 now)
+{
+ WebRtc_Word32 num = 0;
+ WebRtc_Word32 nrOfFrames = 0;
+ for(num = 1; num < (kFrameCountHistorySize - 1); num++)
+ {
+ if (_incomingFrameTimes[num] <= 0 ||
+ now - _incomingFrameTimes[num] > kFrameHistoryWindowMs) // don't use data older than 2sec
+ {
+ break;
+ } else
+ {
+ nrOfFrames++;
+ }
+ }
+ if (num > 1)
+ {
+ WebRtc_Word64 diff = now - _incomingFrameTimes[num-1];
+ _incomingFrameRate = 1.0;
+ if(diff >0)
+ {
+ _incomingFrameRate = nrOfFrames * 1000.0f / static_cast<float>(diff);
+ }
+ }
+ else
+ {
+ _incomingFrameRate = static_cast<float>(nrOfFrames);
+ }
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/video_decimator.h b/trunk/src/modules/video_processing/main/source/video_decimator.h
new file mode 100644
index 0000000..e152bb9
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/video_decimator.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * video_decimator.h
+ */
+#ifndef VPM_VIDEO_DECIMATOR_H
+#define VPM_VIDEO_DECIMATOR_H
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+
+class VPMVideoDecimator
+{
+public:
+ VPMVideoDecimator();
+ ~VPMVideoDecimator();
+
+ void Reset();
+
+ void EnableTemporalDecimation(bool enable);
+
+ WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
+ WebRtc_Word32 SetTargetFrameRate(WebRtc_UWord32 frameRate);
+
+ bool DropFrame();
+
+ void UpdateIncomingFrameRate();
+
+ // Get Decimated Frame Rate/Dimensions
+ WebRtc_UWord32 DecimatedFrameRate();
+
+ //Get input frame rate
+ WebRtc_UWord32 InputFrameRate();
+
+private:
+ void ProcessIncomingFrameRate(WebRtc_Word64 now);
+
+ enum { kFrameCountHistorySize = 90};
+ enum { kFrameHistoryWindowMs = 2000};
+
+ // Temporal decimation
+ WebRtc_Word32 _overShootModifier;
+ WebRtc_UWord32 _dropCount;
+ WebRtc_UWord32 _keepCount;
+ WebRtc_UWord32 _targetFrameRate;
+ float _incomingFrameRate;
+ WebRtc_UWord32 _maxFrameRate;
+ WebRtc_Word64 _incomingFrameTimes[kFrameCountHistorySize];
+ bool _enableTemporalDecimation;
+
+};
+
+} //namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/source/video_processing.gypi b/trunk/src/modules/video_processing/main/source/video_processing.gypi
new file mode 100644
index 0000000..3bc03bc
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/video_processing.gypi
@@ -0,0 +1,88 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'video_processing',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_utility',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/video_processing.h',
+ '../interface/video_processing_defines.h',
+ 'brighten.cc',
+ 'brighten.h',
+ 'brightness_detection.cc',
+ 'brightness_detection.h',
+ 'color_enhancement.cc',
+ 'color_enhancement.h',
+ 'color_enhancement_private.h',
+ 'content_analysis.cc',
+ 'content_analysis.h',
+ 'deflickering.cc',
+ 'deflickering.h',
+ 'denoising.cc',
+ 'denoising.h',
+ 'frame_preprocessor.cc',
+ 'frame_preprocessor.h',
+ 'spatial_resampler.cc',
+ 'spatial_resampler.h',
+ 'video_decimator.cc',
+ 'video_decimator.h',
+ 'video_processing_impl.cc',
+ 'video_processing_impl.h',
+ ],
+ 'conditions': [
+ ['target_arch=="ia32" or target_arch=="x64"', {
+ 'dependencies': [ 'video_processing_sse2', ],
+ }],
+ ],
+ },
+ ],
+ 'conditions': [
+ ['target_arch=="ia32" or target_arch=="x64"', {
+ 'targets': [
+ {
+ 'target_name': 'video_processing_sse2',
+ 'type': '<(library)',
+ 'sources': [
+ 'content_analysis_sse2.cc',
+ ],
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ ],
+ 'conditions': [
+ ['os_posix==1 and OS!="mac"', {
+ 'cflags': [ '-msse2', ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ 'OTHER_CFLAGS': [ '-msse2', ],
+ },
+ }],
+ ],
+ },
+ ],
+ }],
+ ],
+}
+
diff --git a/trunk/src/modules/video_processing/main/source/video_processing_impl.cc b/trunk/src/modules/video_processing/main/source/video_processing_impl.cc
new file mode 100644
index 0000000..3619996
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/video_processing_impl.cc
@@ -0,0 +1,340 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_processing_impl.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+#include <cassert>
+
+namespace webrtc {
+
+namespace
+{
+ void
+ SetSubSampling(VideoProcessingModule::FrameStats& stats,
+ const WebRtc_Word32 width,
+ const WebRtc_Word32 height)
+ {
+ if (width * height >= 640 * 480)
+ {
+ stats.subSamplWidth = 3;
+ stats.subSamplHeight = 3;
+ }
+ else if (width * height >= 352 * 288)
+ {
+ stats.subSamplWidth = 2;
+ stats.subSamplHeight = 2;
+ }
+ else if (width * height >= 176 * 144)
+ {
+ stats.subSamplWidth = 1;
+ stats.subSamplHeight = 1;
+ }
+ else
+ {
+ stats.subSamplWidth = 0;
+ stats.subSamplHeight = 0;
+ }
+ }
+}
+
+VideoProcessingModule*
+VideoProcessingModule::Create(const WebRtc_Word32 id)
+{
+
+ return new VideoProcessingModuleImpl(id);
+}
+
+void
+VideoProcessingModule::Destroy(VideoProcessingModule* module)
+{
+ if (module)
+ {
+ delete static_cast<VideoProcessingModuleImpl*>(module);
+ }
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped mutex(_mutex);
+ _id = id;
+ _brightnessDetection.ChangeUniqueId(id);
+ _deflickering.ChangeUniqueId(id);
+ _denoising.ChangeUniqueId(id);
+ _framePreProcessor.ChangeUniqueId(id);
+ return VPM_OK;
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::Id() const
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _id;
+}
+
+VideoProcessingModuleImpl::VideoProcessingModuleImpl(const WebRtc_Word32 id) :
+ _id(id),
+ _mutex(*CriticalSectionWrapper::CreateCriticalSection())
+{
+ _brightnessDetection.ChangeUniqueId(id);
+ _deflickering.ChangeUniqueId(id);
+ _denoising.ChangeUniqueId(id);
+ _framePreProcessor.ChangeUniqueId(id);
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, _id, "Created");
+}
+
+
+VideoProcessingModuleImpl::~VideoProcessingModuleImpl()
+{
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, _id, "Destroyed");
+
+ delete &_mutex;
+}
+
+void
+VideoProcessingModuleImpl::Reset()
+{
+ CriticalSectionScoped mutex(_mutex);
+ _deflickering.Reset();
+ _denoising.Reset();
+ _brightnessDetection.Reset();
+ _framePreProcessor.Reset();
+
+}
+
+WebRtc_Word32
+VideoProcessingModule::GetFrameStats(FrameStats& stats,
+ const VideoFrame& frame)
+{
+ return GetFrameStats(stats, frame.Buffer(), frame.Width(), frame.Height());
+}
+
+WebRtc_Word32
+VideoProcessingModule::GetFrameStats(FrameStats& stats,
+ const WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ if (frame == NULL)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Null frame pointer");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ if (width == 0 || height == 0)
+ {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Invalid frame size");
+ return VPM_PARAMETER_ERROR;
+ }
+
+ ClearFrameStats(stats); // The histogram needs to be zeroed out.
+ SetSubSampling(stats, width, height);
+
+ // Compute histogram and sum of frame
+ for (WebRtc_UWord32 i = 0; i < height; i += (1 << stats.subSamplHeight))
+ {
+ WebRtc_Word32 k = i * width;
+ for (WebRtc_UWord32 j = 0; j < width; j += (1 << stats.subSamplWidth))
+ {
+ stats.hist[frame[k + j]]++;
+ stats.sum += frame[k + j];
+ }
+ }
+
+ stats.numPixels = (width * height) / ((1 << stats.subSamplWidth) * (1 << stats.subSamplHeight));
+ assert(stats.numPixels > 0);
+
+ // Compute mean value of frame
+ stats.mean = stats.sum / stats.numPixels;
+
+ return VPM_OK;
+}
+
+bool
+VideoProcessingModule::ValidFrameStats(const FrameStats& stats)
+{
+ if (stats.numPixels == 0)
+ {
+ return false;
+ }
+
+ return true;
+}
+
+void
+VideoProcessingModule::ClearFrameStats(FrameStats& stats)
+{
+ stats.mean = 0;
+ stats.sum = 0;
+ stats.numPixels = 0;
+ stats.subSamplWidth = 0;
+ stats.subSamplHeight = 0;
+ memset(stats.hist, 0, sizeof(stats.hist));
+}
+
+WebRtc_Word32
+VideoProcessingModule::ColorEnhancement(VideoFrame& frame)
+{
+ return ColorEnhancement(frame.Buffer(), frame.Width(), frame.Height());
+}
+
+WebRtc_Word32
+VideoProcessingModule::ColorEnhancement(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ return VideoProcessing::ColorEnhancement(frame, width, height);
+}
+
+WebRtc_Word32
+VideoProcessingModule::Brighten(VideoFrame& frame, int delta)
+{
+ return Brighten(frame.Buffer(), frame.Width(), frame.Height(), delta);
+}
+
+WebRtc_Word32
+VideoProcessingModule::Brighten(WebRtc_UWord8* frame,
+ int width,
+ int height,
+ int delta)
+{
+ return VideoProcessing::Brighten(frame, width, height, delta);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::Deflickering(VideoFrame& frame,
+ FrameStats& stats)
+{
+ return Deflickering(frame.Buffer(), frame.Width(), frame.Height(),
+ frame.TimeStamp(), stats);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::Deflickering(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height,
+ const WebRtc_UWord32 timestamp,
+ FrameStats& stats)
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _deflickering.ProcessFrame(frame, width, height, timestamp, stats);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::Denoising(VideoFrame& frame)
+{
+ return Denoising(frame.Buffer(), frame.Width(), frame.Height());
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::Denoising(WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height)
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _denoising.ProcessFrame(frame, width, height);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::BrightnessDetection(const VideoFrame& frame,
+ const FrameStats& stats)
+{
+ return BrightnessDetection(frame.Buffer(), frame.Width(), frame.Height(), stats);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::BrightnessDetection(const WebRtc_UWord8* frame,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height,
+ const FrameStats& stats)
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _brightnessDetection.ProcessFrame(frame, width, height, stats);
+}
+
+
+void
+VideoProcessingModuleImpl::EnableTemporalDecimation(bool enable)
+{
+ CriticalSectionScoped mutex(_mutex);
+ _framePreProcessor.EnableTemporalDecimation(enable);
+}
+
+
+void
+VideoProcessingModuleImpl::SetInputFrameResampleMode(VideoFrameResampling resamplingMode)
+{
+ CriticalSectionScoped cs(_mutex);
+ _framePreProcessor.SetInputFrameResampleMode(resamplingMode);
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
+{
+ CriticalSectionScoped cs(_mutex);
+ return _framePreProcessor.SetMaxFrameRate(maxFrameRate);
+
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate)
+{
+ CriticalSectionScoped cs(_mutex);
+ return _framePreProcessor.SetTargetResolution(width, height, frameRate);
+}
+
+
+WebRtc_UWord32
+VideoProcessingModuleImpl::DecimatedFrameRate()
+{
+ CriticalSectionScoped cs(_mutex);
+ return _framePreProcessor.DecimatedFrameRate();
+}
+
+
+WebRtc_UWord32
+VideoProcessingModuleImpl::DecimatedWidth() const
+{
+ CriticalSectionScoped cs(_mutex);
+ return _framePreProcessor.DecimatedWidth();
+}
+
+WebRtc_UWord32
+VideoProcessingModuleImpl::DecimatedHeight() const
+{
+ CriticalSectionScoped cs(_mutex);
+ return _framePreProcessor.DecimatedHeight();
+}
+
+WebRtc_Word32
+VideoProcessingModuleImpl::PreprocessFrame(const VideoFrame *frame, VideoFrame **processedFrame)
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _framePreProcessor.PreprocessFrame(frame, processedFrame);
+}
+
+VideoContentMetrics*
+VideoProcessingModuleImpl::ContentMetrics() const
+{
+ CriticalSectionScoped mutex(_mutex);
+ return _framePreProcessor.ContentMetrics();
+}
+
+
+void
+VideoProcessingModuleImpl::EnableContentAnalysis(bool enable)
+{
+ CriticalSectionScoped mutex(_mutex);
+ _framePreProcessor.EnableContentAnalysis(enable);
+}
+
+} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/video_processing_impl.h b/trunk/src/modules/video_processing/main/source/video_processing_impl.h
new file mode 100644
index 0000000..3170ab1
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/source/video_processing_impl.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
+#define WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
+
+#include "video_processing.h"
+#include "brighten.h"
+#include "brightness_detection.h"
+#include "color_enhancement.h"
+#include "deflickering.h"
+#include "denoising.h"
+#include "frame_preprocessor.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class VideoProcessingModuleImpl : public VideoProcessingModule
+{
+public:
+
+ VideoProcessingModuleImpl(WebRtc_Word32 id);
+
+ virtual ~VideoProcessingModuleImpl();
+
+ WebRtc_Word32 Id() const;
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual void Reset();
+
+ virtual WebRtc_Word32 Deflickering(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ WebRtc_UWord32 timestamp,
+ FrameStats& stats);
+
+ virtual WebRtc_Word32 Deflickering(VideoFrame& frame,
+ FrameStats& stats);
+
+ virtual WebRtc_Word32 Denoising(WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height);
+
+ virtual WebRtc_Word32 Denoising(VideoFrame& frame);
+
+ virtual WebRtc_Word32 BrightnessDetection(const WebRtc_UWord8* frame,
+ WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ const FrameStats& stats);
+
+ virtual WebRtc_Word32 BrightnessDetection(const VideoFrame& frame,
+ const FrameStats& stats);
+
+
+ //Frame pre-processor functions
+
+ //Enable temporal decimation
+ virtual void EnableTemporalDecimation(bool enable);
+
+ virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
+
+ //Enable content analysis
+ virtual void EnableContentAnalysis(bool enable);
+
+ //Set max frame rate
+ virtual WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
+
+ // Set Target Resolution: frame rate and dimension
+ virtual WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width,
+ WebRtc_UWord32 height,
+ WebRtc_UWord32 frameRate);
+
+
+ // Get decimated values: frame rate/dimension
+ virtual WebRtc_UWord32 DecimatedFrameRate();
+ virtual WebRtc_UWord32 DecimatedWidth() const;
+ virtual WebRtc_UWord32 DecimatedHeight() const;
+
+ // Preprocess:
+ // Pre-process incoming frame: Sample when needed and compute content
+ // metrics when enabled.
+ // If no resampling takes place - processedFrame is set to NULL.
+ virtual WebRtc_Word32 PreprocessFrame(const VideoFrame* frame,
+ VideoFrame** processedFrame);
+ virtual VideoContentMetrics* ContentMetrics() const;
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _mutex;
+
+ VPMDeflickering _deflickering;
+ VPMDenoising _denoising;
+ VPMBrightnessDetection _brightnessDetection;
+ VPMFramePreprocessor _framePreProcessor;
+};
+
+} // namespace
+
+#endif
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
new file mode 100644
index 0000000..6510a5c
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "unit_test.h"
+#include "video_processing.h"
+
+using namespace webrtc;
+
+TEST_F(VideoProcessingModuleTest, BrightnessDetection)
+{
+ WebRtc_UWord32 frameNum = 0;
+ WebRtc_Word32 brightnessWarning = 0;
+ WebRtc_UWord32 warningCount = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
+ {
+ frameNum++;
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
+ if (brightnessWarning != VideoProcessingModule::kNoWarning)
+ {
+ warningCount++;
+ }
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ // Expect few warnings
+ float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+ printf("\nWarning proportions:\n");
+ printf("Stock foreman: %.1f %%\n", warningProportion);
+ EXPECT_LT(warningProportion, 10);
+
+ rewind(_sourceFile);
+ frameNum = 0;
+ warningCount = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength &&
+ frameNum < 300)
+ {
+ frameNum++;
+
+ WebRtc_UWord8* frame = _videoFrame.Buffer();
+ WebRtc_UWord32 yTmp = 0;
+ for (WebRtc_UWord32 yIdx = 0; yIdx < _width * _height; yIdx++)
+ {
+ yTmp = frame[yIdx] << 1;
+ if (yTmp > 255)
+ {
+ yTmp = 255;
+ }
+ frame[yIdx] = static_cast<WebRtc_UWord8>(yTmp);
+ }
+
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
+ EXPECT_NE(VideoProcessingModule::kDarkWarning, brightnessWarning);
+ if (brightnessWarning == VideoProcessingModule::kBrightWarning)
+ {
+ warningCount++;
+ }
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ // Expect many brightness warnings
+ warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+ printf("Bright foreman: %.1f %%\n", warningProportion);
+ EXPECT_GT(warningProportion, 95);
+
+ rewind(_sourceFile);
+ frameNum = 0;
+ warningCount = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength &&
+ frameNum < 300)
+ {
+ frameNum++;
+
+ WebRtc_UWord8* frame = _videoFrame.Buffer();
+ WebRtc_Word32 yTmp = 0;
+ for (WebRtc_UWord32 yIdx = 0; yIdx < _width * _height; yIdx++)
+ {
+ yTmp = frame[yIdx] >> 1;
+ frame[yIdx] = static_cast<WebRtc_UWord8>(yTmp);
+ }
+
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
+ EXPECT_NE(VideoProcessingModule::kBrightWarning, brightnessWarning);
+ if (brightnessWarning == VideoProcessingModule::kDarkWarning)
+ {
+ warningCount++;
+ }
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ // Expect many darkness warnings
+ warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+ printf("Dark foreman: %.1f %%\n\n", warningProportion);
+ EXPECT_GT(warningProportion, 90);
+}
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
new file mode 100644
index 0000000..c494c85
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
@@ -0,0 +1,125 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+#include <cstdlib>
+
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_processing/main/test/unit_test/unit_test.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingModuleTest, ColorEnhancement)
+{
+ TickTime t0;
+ TickTime t1;
+ TickInterval accTicks;
+
+ // Use a shorter version of the Foreman clip for this test.
+ fclose(_sourceFile);
+ const std::string video_file =
+ webrtc::test::ResourcePath("foreman_cif_short", "yuv");
+ _sourceFile = fopen(video_file.c_str(), "rb");
+ ASSERT_TRUE(_sourceFile != NULL) <<
+ "Cannot read source file: " + video_file + "\n";
+
+ std::string output_file = webrtc::test::OutputPath() +
+ "foremanColorEnhancedVPM_cif_short.yuv";
+ FILE* modFile = fopen(output_file.c_str(), "w+b");
+ ASSERT_TRUE(modFile != NULL) << "Could not open output file.\n";
+
+ WebRtc_UWord32 frameNum = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
+ {
+ frameNum++;
+ t0 = TickTime::Now();
+ ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(_videoFrame));
+ t1 = TickTime::Now();
+ accTicks += t1 - t0;
+ fwrite(_videoFrame.Buffer(), 1, _frameLength, modFile);
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ printf("\nTime per frame: %d us \n",
+ static_cast<int>(accTicks.Microseconds() / frameNum));
+ rewind(modFile);
+
+ printf("Comparing files...\n\n");
+ std::string reference_filename =
+ webrtc::test::ResourcePath("foremanColorEnhanced_cif_short", "yuv");
+ FILE* refFile = fopen(reference_filename.c_str(), "rb");
+ ASSERT_TRUE(refFile != NULL) << "Cannot open reference file: " <<
+ reference_filename << "\n"
+ "Create the reference by running Matlab script createTable.m.";
+
+ // get file lenghts
+ ASSERT_EQ(0, fseek(refFile, 0L, SEEK_END));
+ long refLen = ftell(refFile);
+ ASSERT_NE(-1L, refLen);
+ rewind(refFile);
+ ASSERT_EQ(0, fseek(modFile, 0L, SEEK_END));
+ long testLen = ftell(modFile);
+ ASSERT_NE(-1L, testLen);
+ rewind(modFile);
+ ASSERT_EQ(refLen, testLen) << "File lengths differ.";
+
+ VideoFrame refVideoFrame;
+ refVideoFrame.VerifyAndAllocate(_frameLength);
+ refVideoFrame.SetWidth(_width);
+ refVideoFrame.SetHeight(_height);
+
+ // Compare frame-by-frame.
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, modFile) == _frameLength)
+ {
+ ASSERT_EQ(_frameLength, fread(refVideoFrame.Buffer(), 1, _frameLength, refFile));
+ EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), refVideoFrame.Buffer(), _frameLength));
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ // Verify that all color pixels are enhanced, that no luminance values are altered,
+ // and that the function does not write outside the vector.
+ WebRtc_UWord32 safeGuard = 1000;
+ WebRtc_UWord32 numPixels = 352*288; // CIF size
+ WebRtc_UWord8 *testFrame = new WebRtc_UWord8[numPixels + (numPixels / 2) + (2 * safeGuard)];
+ WebRtc_UWord8 *refFrame = new WebRtc_UWord8[numPixels + (numPixels / 2) + (2 * safeGuard)];
+
+ // use value 128 as probe value, since we know that this will be changed in the enhancement
+ memset(testFrame, 128, safeGuard);
+ memset(&testFrame[safeGuard], 128, numPixels);
+ memset(&testFrame[safeGuard + numPixels], 128, numPixels / 2);
+ memset(&testFrame[safeGuard + numPixels + (numPixels / 2)], 128, safeGuard);
+
+ memcpy(refFrame, testFrame, numPixels + (numPixels / 2) + (2 * safeGuard));
+
+ ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(&testFrame[safeGuard], 352, 288));
+
+ EXPECT_EQ(0, memcmp(testFrame, refFrame, safeGuard)) <<
+ "Function is writing outside the frame memory.";
+
+ EXPECT_EQ(0, memcmp(&testFrame[safeGuard + numPixels + (numPixels / 2)],
+ &refFrame[safeGuard + numPixels + (numPixels / 2)], safeGuard)) <<
+ "Function is writing outside the frame memory.";
+
+ EXPECT_EQ(0, memcmp(&testFrame[safeGuard], &refFrame[safeGuard], numPixels)) <<
+ "Function is modifying the luminance.";
+
+ EXPECT_NE(0, memcmp(&testFrame[safeGuard + numPixels],
+ &refFrame[safeGuard + numPixels], numPixels / 2)) <<
+ "Function is not modifying all chrominance pixels";
+
+ ASSERT_EQ(0, fclose(refFile));
+ ASSERT_EQ(0, fclose(modFile));
+ delete [] testFrame;
+ delete [] refFrame;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc
new file mode 100644
index 0000000..54a1390
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_processing/main/source/content_analysis.h"
+#include "modules/video_processing/main/test/unit_test/unit_test.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingModuleTest, ContentAnalysis)
+{
+ VPMContentAnalysis _ca_c(false);
+ VPMContentAnalysis _ca_sse(true);
+ VideoContentMetrics *_cM_c, *_cM_SSE;
+
+ _ca_c.Initialize(_width,_height);
+ _ca_sse.Initialize(_width,_height);
+
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile)
+ == _frameLength)
+ {
+ _cM_c = _ca_c.ComputeContentMetrics(&_videoFrame);
+ _cM_SSE = _ca_sse.ComputeContentMetrics(&_videoFrame);
+
+ ASSERT_EQ(_cM_c->spatial_pred_err, _cM_SSE->spatial_pred_err);
+ ASSERT_EQ(_cM_c->spatial_pred_err_v, _cM_SSE->spatial_pred_err_v);
+ ASSERT_EQ(_cM_c->spatial_pred_err_h, _cM_SSE->spatial_pred_err_h);
+ ASSERT_EQ(_cM_c->motion_magnitude, _cM_SSE->motion_magnitude);
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/createTable.m b/trunk/src/modules/video_processing/main/test/unit_test/createTable.m
new file mode 100644
index 0000000..2c7fb52
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/createTable.m
@@ -0,0 +1,179 @@
+% Create the color enhancement look-up table and write it to
+% file colorEnhancementTable.cpp. Copy contents of that file into
+% the source file for the color enhancement function.
+
+clear
+close all
+
+
+% First, define the color enhancement in a normalized domain
+
+% Compander function is defined in three radial zones.
+% 1. From 0 to radius r0, the compander function
+% is a second-order polynomial intersecting the points (0,0)
+% and (r0, r0), and with a slope B in (0,0).
+% 2. From r0 to r1, the compander is a third-order polynomial
+% intersecting the points (r0, r0) and (r1, r1), and with the
+% same slope as the first part in the point (r0, r0) and slope
+% equal to 1 in (r1, r1).
+% 3. For radii larger than r1, the compander function is the
+% unity scale function (no scaling at all).
+
+r0=0.07; % Dead zone radius (must be > 0)
+r1=0.6; % Enhancement zone radius (must be > r0 and < 1)
+B=0.2; % initial slope of compander function (between 0 and 1)
+
+x0=linspace(0,r0).'; % zone 1
+x1=linspace(r0,r1).'; % zone 2
+x2=linspace(r1,1).'; % zone 3
+
+A=(1-B)/r0;
+f0=A*x0.^2+B*x0; % compander function in zone 1
+
+% equation system for finding second zone parameters
+M=[r0^3 r0^2 r0 1;
+ 3*r0^2 2*r0 1 0;
+ 3*r1^2 2*r1 1 0;
+ r1^3 r1^2 r1 1];
+m=[A*r0^2+B*r0; 2*A*r0+B; 1; r1];
+% solve equations
+theta=M\m;
+
+% compander function in zone 1
+f1=[x1.^3 x1.^2 x1 ones(size(x1))]*theta;
+
+x=[x0; x1; x2];
+f=[f0; f1; x2];
+
+% plot it
+figure(1)
+plot(x,f,x,x,':')
+xlabel('Normalized radius')
+ylabel('Modified radius')
+
+
+% Now, create the look-up table in the integer color space
+[U,V]=meshgrid(0:255, 0:255); % U-V space
+U0=U;
+V0=V;
+
+% Conversion matrix from normalized YUV to RGB
+T=[1 0 1.13983; 1 -0.39465 -0.58060; 1 2.03211 0];
+Ylum=0.5;
+
+figure(2)
+Z(:,:,1)=Ylum + (U-127)/256*T(1,2) + (V-127)/256*T(1,3);
+Z(:,:,2)=Ylum + (U-127)/256*T(2,2) + (V-127)/256*T(2,3);
+Z(:,:,3)=Ylum + (U-127)/256*T(3,2) + (V-127)/256*T(3,3);
+Z=max(Z,0);
+Z=min(Z,1);
+subplot(121)
+image(Z);
+axis square
+axis off
+set(gcf,'color','k')
+
+R = sqrt((U-127).^2 + (V-127).^2);
+Rnorm = R/127;
+RnormMod = Rnorm;
+RnormMod(RnormMod==0)=1; % avoid division with zero
+
+% find indices to pixels in dead-zone (zone 1)
+ix=find(Rnorm<=r0);
+scaleMatrix = (A*Rnorm(ix).^2 + B*Rnorm(ix))./RnormMod(ix);
+U(ix)=(U(ix)-127).*scaleMatrix+127;
+V(ix)=(V(ix)-127).*scaleMatrix+127;
+
+% find indices to pixels in zone 2
+ix=find(Rnorm>r0 & Rnorm<=r1);
+scaleMatrix = (theta(1)*Rnorm(ix).^3 + theta(2)*Rnorm(ix).^2 + ...
+ theta(3)*Rnorm(ix) + theta(4)) ./ RnormMod(ix);
+U(ix)=(U(ix)-127).*scaleMatrix + 127;
+V(ix)=(V(ix)-127).*scaleMatrix + 127;
+
+% round to integer values and saturate
+U=round(U);
+V=round(V);
+U=max(min(U,255),0);
+V=max(min(V,255),0);
+
+Z(:,:,1)=Ylum + (U-127)/256*T(1,2) + (V-127)/256*T(1,3);
+Z(:,:,2)=Ylum + (U-127)/256*T(2,2) + (V-127)/256*T(2,3);
+Z(:,:,3)=Ylum + (U-127)/256*T(3,2) + (V-127)/256*T(3,3);
+Z=max(Z,0);
+Z=min(Z,1);
+subplot(122)
+image(Z);
+axis square
+axis off
+
+figure(3)
+subplot(121)
+mesh(U-U0)
+subplot(122)
+mesh(V-V0)
+
+
+
+% Last, write to file
+% Write only one matrix, since U=V'
+
+fid = fopen('../out/Debug/colorEnhancementTable.h','wt');
+if fid==-1
+ error('Cannot open file colorEnhancementTable.cpp');
+end
+
+fprintf(fid,'//colorEnhancementTable.h\n\n');
+fprintf(fid,'//Copy the constant table to the appropriate header file.\n\n');
+
+fprintf(fid,'//Table created with Matlab script createTable.m\n\n');
+fprintf(fid,'//Usage:\n');
+fprintf(fid,'// Umod=colorTable[U][V]\n');
+fprintf(fid,'// Vmod=colorTable[V][U]\n');
+
+fprintf(fid,'static unsigned char colorTable[%i][%i] = {\n', size(U,1), size(U,2));
+
+for u=1:size(U,2)
+ fprintf(fid,' {%i', U(1,u));
+ for v=2:size(U,1)
+ fprintf(fid,', %i', U(v,u));
+ end
+ fprintf(fid,'}');
+ if u<size(U,2)
+ fprintf(fid,',');
+ end
+ fprintf(fid,'\n');
+end
+fprintf(fid,'};\n\n');
+fclose(fid);
+fprintf('done');
+
+
+answ=input('Create test vector (takes some time...)? y/n : ','s');
+if answ ~= 'y'
+ return
+end
+
+% Also, create test vectors
+
+% Read test file foreman.yuv
+fprintf('Reading test file...')
+[y,u,v]=readYUV420file('../out/Debug/testFiles/foreman_cif.yuv',352,288);
+fprintf(' done\n');
+unew=uint8(zeros(size(u)));
+vnew=uint8(zeros(size(v)));
+
+% traverse all frames
+for k=1:size(y,3)
+ fprintf('Frame %i\n', k);
+ for r=1:size(u,1)
+ for c=1:size(u,2)
+ unew(r,c,k) = uint8(U(double(v(r,c,k))+1, double(u(r,c,k))+1));
+ vnew(r,c,k) = uint8(V(double(v(r,c,k))+1, double(u(r,c,k))+1));
+ end
+ end
+end
+
+fprintf('\nWriting modified test file...')
+writeYUV420file('../out/Debug/foremanColorEnhanced.yuv',y,unew,vnew);
+fprintf(' done\n');
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc
new file mode 100644
index 0000000..f91dc13
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+#include <cstdlib>
+
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_processing/main/test/unit_test/unit_test.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingModuleTest, Deflickering)
+{
+ enum { NumRuns = 30 };
+ WebRtc_UWord32 frameNum = 0;
+ const WebRtc_UWord32 frameRate = 15;
+
+ WebRtc_Word64 minRuntime = 0;
+ WebRtc_Word64 avgRuntime = 0;
+
+ // Close automatically opened Foreman.
+ fclose(_sourceFile);
+ const std::string input_file =
+ webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
+ _sourceFile = fopen(input_file.c_str(), "rb");
+ ASSERT_TRUE(_sourceFile != NULL) <<
+ "Cannot read input file: " << input_file << "\n";
+
+ const std::string output_file =
+ webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
+ FILE* deflickerFile = fopen(output_file.c_str(), "wb");
+ ASSERT_TRUE(deflickerFile != NULL) <<
+ "Could not open output file: " << output_file << "\n";
+
+ printf("\nRun time [us / frame]:\n");
+ for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
+ {
+ TickTime t0;
+ TickTime t1;
+ TickInterval accTicks;
+ WebRtc_UWord32 timeStamp = 1;
+
+ frameNum = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
+ {
+ frameNum++;
+ _videoFrame.SetTimeStamp(timeStamp);
+
+ t0 = TickTime::Now();
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ ASSERT_EQ(0, _vpm->Deflickering(_videoFrame, stats));
+ t1 = TickTime::Now();
+ accTicks += t1 - t0;
+
+ if (runIdx == 0)
+ {
+ fwrite(_videoFrame.Buffer(), 1, _frameLength, deflickerFile);
+ }
+ timeStamp += (90000 / frameRate);
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ printf("%u\n", static_cast<int>(accTicks.Microseconds() / frameNum));
+ if (accTicks.Microseconds() < minRuntime || runIdx == 0)
+ {
+ minRuntime = accTicks.Microseconds();
+ }
+ avgRuntime += accTicks.Microseconds();
+
+ rewind(_sourceFile);
+ }
+ ASSERT_EQ(0, fclose(deflickerFile));
+ // TODO(kjellander): Add verification of deflicker output file.
+
+ printf("\nAverage run time = %d us / frame\n",
+ static_cast<int>(avgRuntime / frameNum / NumRuns));
+ printf("Min run time = %d us / frame\n\n",
+ static_cast<int>(minRuntime / frameNum));
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc
new file mode 100644
index 0000000..173ceb8
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+#include <cstdlib>
+
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_processing/main/test/unit_test/unit_test.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingModuleTest, Denoising)
+{
+ enum { NumRuns = 10 };
+ WebRtc_UWord32 frameNum = 0;
+
+ WebRtc_Word64 minRuntime = 0;
+ WebRtc_Word64 avgRuntime = 0;
+
+ const std::string denoise_filename =
+ webrtc::test::OutputPath() + "denoise_testfile.yuv";
+ FILE* denoiseFile = fopen(denoise_filename.c_str(), "wb");
+ ASSERT_TRUE(denoiseFile != NULL) <<
+ "Could not open output file: " << denoise_filename << "\n";
+
+ const std::string noise_filename =
+ webrtc::test::OutputPath() + "noise_testfile.yuv";
+ FILE* noiseFile = fopen(noise_filename.c_str(), "wb");
+ ASSERT_TRUE(noiseFile != NULL) <<
+ "Could not open noisy file: " << noise_filename << "\n";
+
+ printf("\nRun time [us / frame]:\n");
+ for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
+ {
+ TickTime t0;
+ TickTime t1;
+ TickInterval accTicks;
+ WebRtc_Word32 modifiedPixels = 0;
+
+ frameNum = 0;
+ while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
+ {
+ frameNum++;
+ WebRtc_UWord8* sourceBuffer = _videoFrame.Buffer();
+
+ // Add noise to a part in video stream
+ // Random noise
+ // TODO: investigate the effectiveness of this test.
+
+ //for(WebRtc_UWord32 ir = 0; ir < _frameLength; ir++)
+ // sourceBuffer[ir] = 128
+ for (WebRtc_UWord32 ir = 0; ir < _height; ir++)
+ {
+ WebRtc_UWord32 ik = ir * _width;
+ for (WebRtc_UWord32 ic = 0; ic < _width; ic++)
+ {
+ WebRtc_UWord8 r = rand() % 16;
+ r -= 8;
+ if (ir < _height / 4)
+ r = 0;
+ if (ir >= 3 * _height / 4)
+ r = 0;
+ if (ic < _width / 4)
+ r = 0;
+ if (ic >= 3 * _width / 4)
+ r = 0;
+
+ /*WebRtc_UWord8 pixelValue = 0;
+ if (ir >= _height / 2)
+ { // Region 3 or 4
+ pixelValue = 170;
+ }
+ if (ic >= _width / 2)
+ { // Region 2 or 4
+ pixelValue += 85;
+ }
+ pixelValue += r;
+ sourceBuffer[ik + ic] = pixelValue;
+ */
+ sourceBuffer[ik + ic] += r;
+ }
+ }
+
+ if (runIdx == 0)
+ {
+ fwrite(_videoFrame.Buffer(), 1, _frameLength, noiseFile);
+ }
+
+ t0 = TickTime::Now();
+ ASSERT_GE(modifiedPixels = _vpm->Denoising(_videoFrame), 0);
+ t1 = TickTime::Now();
+ accTicks += t1 - t0;
+
+ if (runIdx == 0)
+ {
+ fwrite(_videoFrame.Buffer(), 1, _frameLength, denoiseFile);
+ }
+ }
+ ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
+
+ printf("%u\n", static_cast<int>(accTicks.Microseconds() / frameNum));
+ if (accTicks.Microseconds() < minRuntime || runIdx == 0)
+ {
+ minRuntime = accTicks.Microseconds();
+ }
+ avgRuntime += accTicks.Microseconds();
+
+ rewind(_sourceFile);
+ }
+ ASSERT_EQ(0, fclose(denoiseFile));
+ ASSERT_EQ(0, fclose(noiseFile));
+ printf("\nAverage run time = %d us / frame\n",
+ static_cast<int>(avgRuntime / frameNum / NumRuns));
+ printf("Min run time = %d us / frame\n\n",
+ static_cast<int>(minRuntime / frameNum));
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m b/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m
new file mode 100644
index 0000000..03013ef
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m
@@ -0,0 +1,45 @@
+function [Y,U,V] = readYUV420file(filename, width, height)
+% [Y,U,V] = readYUVfile(filename, width, height)
+
+fid = fopen(filename,'rb');
+if fid==-1
+ error(['Cannot open file ' filename]);
+end
+
+% Number of pixels per image
+nPx=width*height;
+
+% nPx bytes luminance, nPx/4 bytes U, nPx/4 bytes V
+frameSizeBytes = nPx*1.5;
+
+% calculate number of frames
+fseek(fid,0,'eof'); % move to end of file
+fileLen=ftell(fid); % number of bytes
+fseek(fid,0,'bof'); % rewind to start
+
+% calculate number of frames
+numFrames = floor(fileLen/frameSizeBytes);
+
+Y=uint8(zeros(height,width,numFrames));
+U=uint8(zeros(height/2,width/2,numFrames));
+V=uint8(zeros(height/2,width/2,numFrames));
+
+[X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
+
+for k=1:numFrames
+
+ % Store luminance
+ Y(:,:,k)=uint8(reshape(X(1:nPx), width, height).');
+
+ % Store U channel
+ U(:,:,k)=uint8(reshape(X(nPx + (1:nPx/4)), width/2, height/2).');
+
+ % Store V channel
+ V(:,:,k)=uint8(reshape(X(nPx + nPx/4 + (1:nPx/4)), width/2, height/2).');
+
+ % Read next frame
+ [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
+end
+
+
+fclose(fid);
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc
new file mode 100644
index 0000000..1dbe070
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc
@@ -0,0 +1,343 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_processing/main/test/unit_test/unit_test.h"
+
+#include <string>
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+void TestSize(VideoFrame& sourceFrame,
+ WebRtc_UWord32 targetWidth, WebRtc_UWord32 targetHeight,
+ WebRtc_UWord32 mode, VideoProcessingModule *vpm);
+
+VideoProcessingModuleTest::VideoProcessingModuleTest() :
+ _vpm(NULL),
+ _sourceFile(NULL),
+ _width(352),
+ _height(288),
+ _frameLength(CalcBufferSize(kI420, 352, 288))
+{
+}
+
+void VideoProcessingModuleTest::SetUp()
+{
+ _vpm = VideoProcessingModule::Create(0);
+ ASSERT_TRUE(_vpm != NULL);
+
+ ASSERT_EQ(0, _videoFrame.VerifyAndAllocate(_frameLength));
+ _videoFrame.SetWidth(_width);
+ _videoFrame.SetHeight(_height);
+
+ const std::string video_file =
+ webrtc::test::ResourcePath("foreman_cif", "yuv");
+ _sourceFile = fopen(video_file.c_str(),"rb");
+ ASSERT_TRUE(_sourceFile != NULL) <<
+ "Cannot read source file: " + video_file + "\n";
+}
+
+void VideoProcessingModuleTest::TearDown()
+{
+ if (_sourceFile != NULL) {
+ ASSERT_EQ(0, fclose(_sourceFile));
+ }
+ _sourceFile = NULL;
+
+ if (_vpm != NULL) {
+ VideoProcessingModule::Destroy(_vpm);
+ }
+ _vpm = NULL;
+}
+
+TEST_F(VideoProcessingModuleTest, HandleNullBuffer)
+{
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ // Video frame with unallocated buffer.
+ VideoFrame videoFrame;
+ videoFrame.SetWidth(_width);
+ videoFrame.SetHeight(_height);
+
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, NULL, _width, _height));
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, videoFrame));
+
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(NULL, _width, _height));
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(videoFrame));
+
+ EXPECT_EQ(-1, _vpm->Deflickering(NULL, _width, _height, 0, stats));
+ EXPECT_EQ(-1, _vpm->Deflickering(videoFrame, stats));
+
+ EXPECT_EQ(-1, _vpm->Denoising(NULL, _width, _height));
+ EXPECT_EQ(-1, _vpm->Denoising(videoFrame));
+
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(NULL, _width, _height, stats));
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(videoFrame, stats));
+
+ EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->PreprocessFrame(NULL, NULL));
+}
+
+TEST_F(VideoProcessingModuleTest, HandleBadStats)
+{
+ VideoProcessingModule::FrameStats stats;
+
+ ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
+ _sourceFile));
+
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), _width, _height, 0,
+ stats));
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
+
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), _width,
+ _height, stats));
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
+}
+
+TEST_F(VideoProcessingModuleTest, HandleBadSize)
+{
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+
+ // Bad width
+ _videoFrame.SetWidth(0);
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame.Buffer(), 0, _height));
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame));
+
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame.Buffer(), 0, _height));
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame));
+
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), 0, _height, 0,
+ stats));
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
+
+ EXPECT_EQ(-1, _vpm->Denoising(_videoFrame.Buffer(), 0, _height));
+ EXPECT_EQ(-1, _vpm->Denoising(_videoFrame));
+
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), 0, _height,
+ stats));
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
+
+
+ // Bad height
+ _videoFrame.SetWidth(_width);
+ _videoFrame.SetHeight(0);
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame.Buffer(), _width, 0));
+ EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame));
+
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame.Buffer(), _width, 0));
+ EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame));
+
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), _width, 0, 0,
+ stats));
+ EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
+
+ EXPECT_EQ(-1, _vpm->Denoising(_videoFrame.Buffer(), _width, 0));
+ EXPECT_EQ(-1, _vpm->Denoising(_videoFrame));
+
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), _width, 0,
+ stats));
+ EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
+
+ EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->SetTargetResolution(0,0,0));
+ EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->SetMaxFrameRate(0));
+
+ VideoFrame *outFrame = NULL;
+ EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->PreprocessFrame(&_videoFrame,
+ &outFrame));
+}
+
+TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset)
+{
+ VideoFrame videoFrame2;
+ VideoProcessingModule::FrameStats stats;
+
+ ASSERT_EQ(0, videoFrame2.VerifyAndAllocate(_frameLength));
+ videoFrame2.SetWidth(_width);
+ videoFrame2.SetHeight(_height);
+
+ // Only testing non-static functions here.
+ ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
+ _sourceFile));
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
+ ASSERT_EQ(0, _vpm->Deflickering(_videoFrame, stats));
+ _vpm->Reset();
+ // Retrieve frame stats again in case Deflickering() has zeroed them.
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, videoFrame2));
+ ASSERT_EQ(0, _vpm->Deflickering(videoFrame2, stats));
+ EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
+ _frameLength));
+
+ ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
+ _sourceFile));
+ memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
+ ASSERT_GE(_vpm->Denoising(_videoFrame), 0);
+ _vpm->Reset();
+ ASSERT_GE(_vpm->Denoising(videoFrame2), 0);
+ EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
+ _frameLength));
+
+ ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
+ _sourceFile));
+ ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
+ ASSERT_EQ(0, _vpm->BrightnessDetection(_videoFrame, stats));
+ _vpm->Reset();
+ ASSERT_EQ(0, _vpm->BrightnessDetection(videoFrame2, stats));
+ EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
+ _frameLength));
+}
+
+TEST_F(VideoProcessingModuleTest, FrameStats)
+{
+ VideoProcessingModule::FrameStats stats;
+ ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
+ _sourceFile));
+
+ EXPECT_FALSE(_vpm->ValidFrameStats(stats));
+ EXPECT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
+ EXPECT_TRUE(_vpm->ValidFrameStats(stats));
+
+ printf("\nFrameStats\n");
+ printf("mean: %u\nnumPixels: %u\nsubSamplWidth: "
+ "%u\nsumSamplHeight: %u\nsum: %u\n\n",
+ static_cast<unsigned int>(stats.mean),
+ static_cast<unsigned int>(stats.numPixels),
+ static_cast<unsigned int>(stats.subSamplHeight),
+ static_cast<unsigned int>(stats.subSamplWidth),
+ static_cast<unsigned int>(stats.sum));
+
+ _vpm->ClearFrameStats(stats);
+ EXPECT_FALSE(_vpm->ValidFrameStats(stats));
+}
+
+TEST_F(VideoProcessingModuleTest, PreprocessorLogic)
+{
+ // Disable temporal sampling
+ _vpm->EnableTemporalDecimation(false);
+ ASSERT_EQ(VPM_OK, _vpm->SetMaxFrameRate(30));
+ ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 15));
+ // Revert
+ _vpm->EnableTemporalDecimation(true);
+ ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 30));
+ // Disable spatial sampling
+ _vpm->SetInputFrameResampleMode(kNoRescaling);
+ ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 30));
+ VideoFrame *outFrame = NULL;
+ ASSERT_EQ(VPM_OK, _vpm->PreprocessFrame(&_videoFrame, &outFrame));
+ // No rescaling=> output frame = NULL
+ ASSERT_TRUE(outFrame == NULL);
+}
+
+TEST_F(VideoProcessingModuleTest, Resampler)
+{
+ enum { NumRuns = 1 };
+
+ WebRtc_Word64 minRuntime = 0;
+ WebRtc_Word64 avgRuntime = 0;
+
+ TickTime t0;
+ TickTime t1;
+ TickInterval accTicks;
+ WebRtc_Word32 height = 288;
+ WebRtc_Word32 width = 352;
+ WebRtc_Word32 lengthSourceFrame = width*height*3/2;
+
+ rewind(_sourceFile);
+ ASSERT_TRUE(_sourceFile != NULL) <<
+ "Cannot read input file \n";
+
+ // CA not needed here
+ _vpm->EnableContentAnalysis(false);
+ // no temporal decimation
+ _vpm->EnableTemporalDecimation(false);
+
+ // Reading test frame
+ VideoFrame sourceFrame;
+ ASSERT_EQ(0, sourceFrame.VerifyAndAllocate(lengthSourceFrame));
+ EXPECT_GT(fread(sourceFrame.Buffer(), 1, lengthSourceFrame, _sourceFile), 0u);
+ ASSERT_EQ(0, sourceFrame.SetLength(lengthSourceFrame));
+ sourceFrame.SetHeight(height);
+ sourceFrame.SetWidth(width);
+
+ for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
+ {
+ // initiate test timer
+ t0 = TickTime::Now();
+
+ // kFastRescaling
+ _vpm->SetInputFrameResampleMode(kFastRescaling);
+ // TESTING DIFFERENT SIZES
+ TestSize(sourceFrame, 100, 50, 1, _vpm); // Cut, decimation 1x, interpolate
+ TestSize(sourceFrame, 352/2, 288/2, 1, _vpm); // Even decimation
+ TestSize(sourceFrame, 352, 288, 1, _vpm); // No resampling
+ TestSize(sourceFrame, 2*352, 2*288,1, _vpm); // Even upsampling
+ TestSize(sourceFrame, 400, 256, 1, _vpm); // Upsampling 1.5x and cut
+ TestSize(sourceFrame, 960, 720, 1, _vpm); // Upsampling 3.5x and cut
+ TestSize(sourceFrame, 1280, 720, 1, _vpm); // Upsampling 4x and cut
+
+ // kBiLinear
+ _vpm->SetInputFrameResampleMode(kBiLinear);
+ // TESTING DIFFERENT SIZES
+ TestSize(sourceFrame, 352/4, 288/4, 2, _vpm);
+ TestSize(sourceFrame, 352/2, 288/2, 2, _vpm);
+ TestSize(sourceFrame, 2*352, 2*288,2, _vpm);
+ TestSize(sourceFrame, 480, 640, 2, _vpm);
+ TestSize(sourceFrame, 960, 720, 2, _vpm);
+ TestSize(sourceFrame, 1280, 720, 2, _vpm);
+ // stop timer
+ t1 = TickTime::Now();
+ accTicks += t1 - t0;
+
+ if (accTicks.Microseconds() < minRuntime || runIdx == 0) {
+ minRuntime = accTicks.Microseconds();
+ }
+ avgRuntime += accTicks.Microseconds();
+ }
+
+ sourceFrame.Free();
+
+ printf("\nAverage run time = %d us / frame\n",
+ //static_cast<int>(avgRuntime / frameNum / NumRuns));
+ static_cast<int>(avgRuntime));
+ printf("Min run time = %d us / frame\n\n",
+ //static_cast<int>(minRuntime / frameNum));
+ static_cast<int>(minRuntime));
+}
+
+void TestSize(VideoFrame& sourceFrame, WebRtc_UWord32 targetWidth,
+ WebRtc_UWord32 targetHeight,
+ WebRtc_UWord32 mode, VideoProcessingModule *vpm)
+{
+ VideoFrame *outFrame = NULL;
+ std::ostringstream filename;
+ filename << webrtc::test::OutputPath() << "Resampler_"<< mode << "_" <<
+ targetWidth << "x" << targetHeight << "_30Hz_P420.yuv";
+ // TODO(kjellander): Add automatic verification of these output files:
+ std::cout << "Watch " << filename.str() << " and verify that it is okay."
+ << std::endl;
+ FILE* standAloneFile = fopen(filename.str().c_str(), "wb");
+ ASSERT_EQ(VPM_OK, vpm->SetTargetResolution(targetWidth, targetHeight, 30));
+ ASSERT_EQ(VPM_OK, vpm->PreprocessFrame(&sourceFrame, &outFrame));
+ // Length should be updated only if frame was resampled
+ if (targetWidth != sourceFrame.Width() ||
+ targetHeight != sourceFrame.Height()) {
+ ASSERT_EQ((targetWidth * targetHeight * 3 / 2), outFrame->Length());
+ // Write to file for visual inspection
+ fwrite(outFrame->Buffer(), 1, outFrame->Length(), standAloneFile);
+ outFrame->Free();
+ }
+ fclose(standAloneFile);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h
new file mode 100644
index 0000000..2363e1a
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
+#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
+
+#include "gtest/gtest.h"
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "system_wrappers/interface/trace.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+class VideoProcessingModuleTest : public ::testing::Test
+{
+protected:
+ VideoProcessingModuleTest();
+ virtual void SetUp();
+ virtual void TearDown();
+ static void SetUpTestCase()
+ {
+ Trace::CreateTrace();
+ std::string trace_file = webrtc::test::OutputPath() + "VPMTrace.txt";
+ ASSERT_EQ(0, Trace::SetTraceFile(trace_file.c_str()));
+ }
+ static void TearDownTestCase()
+ {
+ Trace::ReturnTrace();
+ }
+ VideoProcessingModule* _vpm;
+ FILE* _sourceFile;
+ VideoFrame _videoFrame;
+ const WebRtc_UWord32 _width;
+ const WebRtc_UWord32 _height;
+ const WebRtc_UWord32 _frameLength;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m b/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m
new file mode 100644
index 0000000..69a8808
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m
@@ -0,0 +1,22 @@
+function writeYUV420file(filename, Y, U, V)
+% writeYUV420file(filename, Y, U, V)
+
+fid = fopen(filename,'wb');
+if fid==-1
+ error(['Cannot open file ' filename]);
+end
+
+numFrames=size(Y,3);
+
+for k=1:numFrames
+ % Write luminance
+ fwrite(fid,uint8(Y(:,:,k).'), 'uchar');
+
+ % Write U channel
+ fwrite(fid,uint8(U(:,:,k).'), 'uchar');
+
+ % Write V channel
+ fwrite(fid,uint8(V(:,:,k).'), 'uchar');
+end
+
+fclose(fid);
diff --git a/trunk/src/modules/video_processing/main/test/vpm_tests.gypi b/trunk/src/modules/video_processing/main/test/vpm_tests.gypi
new file mode 100644
index 0000000..e53ac07
--- /dev/null
+++ b/trunk/src/modules/video_processing/main/test/vpm_tests.gypi
@@ -0,0 +1,39 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'video_processing_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_processing',
+ 'webrtc_utility',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ # headers
+ 'unit_test/unit_test.h',
+ # sources
+ 'unit_test/brightness_detection_test.cc',
+ 'unit_test/color_enhancement_test.cc',
+ 'unit_test/content_metrics_test.cc',
+ 'unit_test/deflickering_test.cc',
+ 'unit_test/denoising_test.cc',
+ 'unit_test/unit_test.cc',
+ ], # sources
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_render/OWNERS b/trunk/src/modules/video_render/OWNERS
new file mode 100644
index 0000000..ac607bd
--- /dev/null
+++ b/trunk/src/modules/video_render/OWNERS
@@ -0,0 +1,4 @@
+mflodman@webrtc.org
+perkj@webrtc.org
+wu@webrtc.org
+mallinath@webrtc.org
diff --git a/trunk/src/modules/video_render/main/interface/video_render.h b/trunk/src/modules/video_render/main/interface/video_render.h
new file mode 100644
index 0000000..42fce70
--- /dev/null
+++ b/trunk/src/modules/video_render/main/interface/video_render.h
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+
+/*
+ * video_render.h
+ *
+ * This header file together with module.h and module_common_types.h
+ * contains all of the APIs that are needed for using the video render
+ * module class.
+ *
+ */
+
+#include "module.h"
+#include "video_render_defines.h"
+
+namespace webrtc {
+// Class definitions
+class VideoRender: public Module
+{
+public:
+ /*
+ * Create a video render module object
+ *
+ * id - unique identifier of this video render module object
+ * window - pointer to the window to render to
+ * fullscreen - true if this is a fullscreen renderer
+ * videoRenderType - type of renderer to create
+ */
+ static VideoRender
+ * CreateVideoRender(
+ const WebRtc_Word32 id,
+ void* window,
+ const bool fullscreen,
+ const VideoRenderType videoRenderType =
+ kRenderDefault);
+
+ /*
+ * Destroy a video render module object
+ *
+ * module - object to destroy
+ */
+ static void DestroyVideoRender(VideoRender* module);
+
+ /*
+ * Change the unique identifier of this object
+ *
+ * id - new unique identifier of this video render module object
+ */
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ virtual WebRtc_Word32 TimeUntilNextProcess() = 0;
+ virtual WebRtc_Word32 Process() = 0;
+
+ /**************************************************************************
+ *
+ * Window functions
+ *
+ ***************************************************************************/
+
+ /*
+ * Get window for this renderer
+ */
+ virtual void* Window() = 0;
+
+ /*
+ * Change render window
+ *
+ * window - the new render window, assuming same type as originally created.
+ */
+ virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ /*
+ * Add incoming render stream
+ *
+ * streamID - id of the stream to add
+ * zOrder - relative render order for the streams, 0 = on top
+ * left - position of the stream in the window, [0.0f, 1.0f]
+ * top - position of the stream in the window, [0.0f, 1.0f]
+ * right - position of the stream in the window, [0.0f, 1.0f]
+ * bottom - position of the stream in the window, [0.0f, 1.0f]
+ *
+ * Return - callback class to use for delivering new frames to render.
+ */
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom) = 0;
+ /*
+ * Delete incoming render stream
+ *
+ * streamID - id of the stream to add
+ */
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId) = 0;
+
+ /*
+ * Add incoming render callback, used for external rendering
+ *
+ * streamID - id of the stream the callback is used for
+ * renderObject - the VideoRenderCallback to use for this stream, NULL to remove
+ *
+ * Return - callback class to use for delivering new frames to render.
+ */
+ virtual WebRtc_Word32
+ AddExternalRenderCallback(const WebRtc_UWord32 streamId,
+ VideoRenderCallback* renderObject) = 0;
+
+ /*
+ * Get the porperties for an incoming render stream
+ *
+ * streamID - [in] id of the stream to get properties for
+ * zOrder - [out] relative render order for the streams, 0 = on top
+ * left - [out] position of the stream in the window, [0.0f, 1.0f]
+ * top - [out] position of the stream in the window, [0.0f, 1.0f]
+ * right - [out] position of the stream in the window, [0.0f, 1.0f]
+ * bottom - [out] position of the stream in the window, [0.0f, 1.0f]
+ */
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const = 0;
+ /*
+ * The incoming frame rate to the module, not the rate rendered in the window.
+ */
+ virtual WebRtc_UWord32
+ GetIncomingFrameRate(const WebRtc_UWord32 streamId) = 0;
+
+ /*
+ * Returns the number of incoming streams added to this render module
+ */
+ virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const = 0;
+
+ /*
+ * Returns true if this render module has the streamId added, false otherwise.
+ */
+ virtual bool
+ HasIncomingRenderStream(const WebRtc_UWord32 streamId) const = 0;
+
+ /*
+ * Registers a callback to get raw images in the same time as sent
+ * to the renderer. To be used for external rendering.
+ */
+ virtual WebRtc_Word32
+ RegisterRawFrameCallback(const WebRtc_UWord32 streamId,
+ VideoRenderCallback* callbackObj) = 0;
+
+ /*
+ * This method is usefull to get last rendered frame for the stream specified
+ */
+ virtual WebRtc_Word32
+ GetLastRenderedFrame(const WebRtc_UWord32 streamId,
+ VideoFrame &frame) const = 0;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ /*
+ * Starts rendering the specified stream
+ */
+ virtual WebRtc_Word32 StartRender(const WebRtc_UWord32 streamId) = 0;
+
+ /*
+ * Stops the renderer
+ */
+ virtual WebRtc_Word32 StopRender(const WebRtc_UWord32 streamId) = 0;
+
+ /*
+ * Resets the renderer
+ * No streams are removed. The state should be as after AddStream was called.
+ */
+ virtual WebRtc_Word32 ResetRender() = 0;
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ /*
+ * Returns the preferred render video type
+ */
+ virtual RawVideoType PreferredVideoType() const = 0;
+
+ /*
+ * Returns true if the renderer is in fullscreen mode, otherwise false.
+ */
+ virtual bool IsFullScreen() = 0;
+
+ /*
+ * Gets screen resolution in pixels
+ */
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const = 0;
+
+ /*
+ * Get the actual render rate for this stream. I.e rendered frame rate,
+ * not frames delivered to the renderer.
+ */
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId) = 0;
+
+ /*
+ * Set cropping of incoming stream
+ */
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom) = 0;
+
+ /*
+ * re-configure renderer
+ */
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
+
+ virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable) = 0;
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float right, const float bottom) = 0;
+
+ /*
+ * Set a start image. The image is rendered before the first image has been delivered
+ */
+ virtual WebRtc_Word32
+ SetStartImage(const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame) = 0;
+
+ /*
+ * Set a timout image. The image is rendered if no videoframe has been delivered
+ */
+ virtual WebRtc_Word32 SetTimeoutImage(const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame,
+ const WebRtc_UWord32 timeout)= 0;
+
+ virtual WebRtc_Word32 MirrorRenderStream(const int renderId,
+ const bool enable,
+ const bool mirrorXAxis,
+ const bool mirrorYAxis) = 0;
+
+ static WebRtc_Word32 SetAndroidObjects(void* javaVM);
+};
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/interface/video_render_defines.h b/trunk/src/modules/video_render/main/interface/video_render_defines.h
new file mode 100644
index 0000000..2a87f3f
--- /dev/null
+++ b/trunk/src/modules/video_render/main/interface/video_render_defines.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+
+// Includes
+#include "common_types.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+// Defines
+#ifndef NULL
+#define NULL 0
+#endif
+
+// Enums
+enum VideoRenderType
+{
+ kRenderExternal = 0, // External
+ kRenderWindows = 1, // Windows
+ kRenderCocoa = 2, // Mac
+ kRenderCarbon = 3,
+ kRenderiPhone = 4, // iPhone
+ kRenderAndroid = 5, // Android
+ kRenderX11 = 6, // Linux
+ kRenderDefault
+};
+
+// Runtime errors
+enum VideoRenderError
+{
+ kRenderShutDown = 0,
+ kRenderPerformanceAlarm = 1
+};
+
+// The object a module user uses to send new frames to the renderer
+// One object is used for each incoming stream
+class VideoRenderCallback
+{
+public:
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame) = 0;
+
+protected:
+ virtual ~VideoRenderCallback()
+ {
+ }
+};
+
+// Feedback class to be implemented by module user
+class VideoRenderFeedback
+{
+public:
+ virtual void OnRenderError(const WebRtc_Word32 streamId,
+ const VideoRenderError error) = 0;
+
+protected:
+ virtual ~VideoRenderFeedback()
+ {
+ }
+};
+
+// Mobile enums
+enum StretchMode
+{
+ kStretchToInsideEdge = 1,
+ kStretchToOutsideEdge = 2,
+ kStretchMatchWidth = 3,
+ kStretchMatchHeight = 4,
+ kStretchNone = 5
+};
+
+enum Rotation
+{
+ kRotation0 = 0,
+ kRotation90 = 1,
+ kRotation180 = 2,
+ kRotation270 = 3
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
diff --git a/trunk/src/modules/video_render/main/source/Android.mk b/trunk/src/modules/video_render/main/source/Android.mk
new file mode 100644
index 0000000..73eec93
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android.mk
@@ -0,0 +1,53 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_video_render
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ incoming_video_stream.cc \
+ video_render_frames.cc \
+ video_render_impl.cc \
+ external/video_render_external_impl.cc \
+ Android/video_render_android_impl.cc \
+ Android/video_render_android_native_opengl2.cc \
+ Android/video_render_android_surface_view.cc \
+ Android/video_render_opengles20.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/Android \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../audio_coding/main/interface \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../../utility/interface \
+ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java
new file mode 100644
index 0000000..fc30607
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java
@@ -0,0 +1,263 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+import java.util.concurrent.locks.ReentrantLock;
+
+import javax.microedition.khronos.egl.EGL10;
+import javax.microedition.khronos.egl.EGLConfig;
+import javax.microedition.khronos.egl.EGLContext;
+import javax.microedition.khronos.egl.EGLDisplay;
+import javax.microedition.khronos.opengles.GL10;
+
+import android.app.ActivityManager;
+import android.content.Context;
+import android.content.pm.ConfigurationInfo;
+import android.opengl.GLSurfaceView;
+import android.util.Log;
+
+public class ViEAndroidGLES20 extends GLSurfaceView
+ implements GLSurfaceView.Renderer {
+ // True if onSurfaceCreated has been called.
+ private boolean surfaceCreated = false;
+ private boolean openGLCreated = false;
+ // True if NativeFunctionsRegistered has been called.
+ private boolean nativeFunctionsRegisted = false;
+ private ReentrantLock nativeFunctionLock = new ReentrantLock();
+ // Address of Native object that will do the drawing.
+ private long nativeObject = 0;
+ private int viewWidth = 0;
+ private int viewHeight = 0;
+
+ public static boolean UseOpenGL2(Object renderWindow) {
+ return ViEAndroidGLES20.class.isInstance(renderWindow);
+ }
+
+ public ViEAndroidGLES20(Context context) {
+ super(context);
+
+ // Setup the context factory for 2.0 rendering.
+ // See ContextFactory class definition below
+ setEGLContextFactory(new ContextFactory());
+
+ // We need to choose an EGLConfig that matches the format of
+ // our surface exactly. This is going to be done in our
+ // custom config chooser. See ConfigChooser class definition below
+ // Use RGB 565 without an alpha channel.
+ setEGLConfigChooser( new ConfigChooser(5, 6, 5, 0, 0, 0) );
+
+ this.setRenderer(this);
+ this.setRenderMode(GLSurfaceView.RENDERMODE_WHEN_DIRTY);
+ }
+
+ // IsSupported
+ // Return true if this device support Open GL ES 2.0 rendering.
+ public static boolean IsSupported(Context context) {
+ ActivityManager am =
+ (ActivityManager) context.getSystemService(Context.ACTIVITY_SERVICE);
+ ConfigurationInfo info = am.getDeviceConfigurationInfo();
+ if(info.reqGlEsVersion >= 0x20000) {
+ // Open GL ES 2.0 is supported.
+ return true;
+ }
+ return false;
+ }
+
+ public void onDrawFrame(GL10 gl) {
+ nativeFunctionLock.lock();
+ if(!nativeFunctionsRegisted || !surfaceCreated) {
+ nativeFunctionLock.unlock();
+ return;
+ }
+
+ if(!openGLCreated) {
+ if(0 != CreateOpenGLNative(nativeObject, viewWidth, viewHeight)) {
+ return; // Failed to create OpenGL
+ }
+ openGLCreated = true; // Created OpenGL successfully
+ }
+ DrawNative(nativeObject); // Draw the new frame
+ nativeFunctionLock.unlock();
+ }
+
+ public void onSurfaceChanged(GL10 gl, int width, int height) {
+ surfaceCreated = true;
+ viewWidth = width;
+ viewHeight = height;
+
+ nativeFunctionLock.lock();
+ if(nativeFunctionsRegisted) {
+ if(CreateOpenGLNative(nativeObject,width,height) == 0)
+ openGLCreated = true;
+ }
+ nativeFunctionLock.unlock();
+ }
+
+ public void onSurfaceCreated(GL10 gl, EGLConfig config) {
+ }
+
+ public void RegisterNativeObject(long nativeObject) {
+ nativeFunctionLock.lock();
+ nativeObject = nativeObject;
+ nativeFunctionsRegisted = true;
+ nativeFunctionLock.unlock();
+ }
+
+ public void DeRegisterNativeObject() {
+ nativeFunctionLock.lock();
+ nativeFunctionsRegisted = false;
+ openGLCreated = false;
+ nativeObject = 0;
+ nativeFunctionLock.unlock();
+ }
+
+ public void ReDraw() {
+ if(surfaceCreated) {
+ // Request the renderer to redraw using the render thread context.
+ this.requestRender();
+ }
+ }
+
+ // EGL Context factory used for creating EGL 2.0 context
+ // on Android 2.1(and later,
+ // though there are simpler ways in 2.2)
+ // Code is from the NDK samples\hello-gl2\src\com\android\gl2jni.
+ private static class ContextFactory
+ implements GLSurfaceView.EGLContextFactory {
+ private static int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
+ public EGLContext createContext(EGL10 egl,
+ EGLDisplay display,
+ EGLConfig eglConfig) {
+ //checkEglError("Before eglCreateContext", egl);
+ int[] attrib_list = {EGL_CONTEXT_CLIENT_VERSION, 2, EGL10.EGL_NONE };
+ // Create an Open GL ES 2.0 context
+ EGLContext context = egl.eglCreateContext(display,
+ eglConfig,
+ EGL10.EGL_NO_CONTEXT,
+ attrib_list);
+ checkEglError("ContextFactory eglCreateContext", egl);
+ return context;
+ }
+
+ public void destroyContext(EGL10 egl, EGLDisplay display,
+ EGLContext context) {
+ egl.eglDestroyContext(display, context);
+ }
+ }
+
+ private static void checkEglError(String prompt, EGL10 egl) {
+ int error;
+ while ((error = egl.eglGetError()) != EGL10.EGL_SUCCESS) {
+ Log.e("*WEBRTC*", String.format("%s: EGL error: 0x%x", prompt, error));
+ }
+ }
+
+ // Code is from the NDK samples\hello-gl2\src\com\android\gl2jni
+ private static class ConfigChooser
+ implements GLSurfaceView.EGLConfigChooser {
+
+ public ConfigChooser(int r, int g, int b, int a, int depth, int stencil) {
+ mRedSize = r;
+ mGreenSize = g;
+ mBlueSize = b;
+ mAlphaSize = a;
+ mDepthSize = depth;
+ mStencilSize = stencil;
+ }
+
+ // This EGL config specification is used to specify 2.0 rendering.
+ // We use a minimum size of 4 bits for red/green/blue, but will
+ // perform actual matching in chooseConfig() below.
+ private static int EGL_OPENGL_ES2_BIT = 4;
+ private static int[] s_configAttribs2 =
+ {
+ EGL10.EGL_RED_SIZE, 4,
+ EGL10.EGL_GREEN_SIZE, 4,
+ EGL10.EGL_BLUE_SIZE, 4,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL10.EGL_NONE
+ };
+
+ public EGLConfig chooseConfig(EGL10 egl, EGLDisplay display) {
+
+ // Get the number of minimally matching EGL configurations
+ int[] num_config = new int[1];
+ egl.eglChooseConfig(display, s_configAttribs2, null, 0, num_config);
+
+ int numConfigs = num_config[0];
+
+ if (numConfigs <= 0) {
+ throw new IllegalArgumentException("No configs match configSpec");
+ }
+
+ // Allocate then read the array of minimally matching EGL configs
+ EGLConfig[] configs = new EGLConfig[numConfigs];
+ egl.eglChooseConfig(display, s_configAttribs2, configs,
+ numConfigs, num_config);
+
+ // Now return the "best" one
+ return chooseConfig(egl, display, configs);
+ }
+
+ public EGLConfig chooseConfig(EGL10 egl, EGLDisplay display,
+ EGLConfig[] configs) {
+ for(EGLConfig config : configs) {
+ int d = findConfigAttrib(egl, display, config,
+ EGL10.EGL_DEPTH_SIZE, 0);
+ int s = findConfigAttrib(egl, display, config,
+ EGL10.EGL_STENCIL_SIZE, 0);
+
+ // We need at least mDepthSize and mStencilSize bits
+ if (d < mDepthSize || s < mStencilSize)
+ continue;
+
+ // We want an *exact* match for red/green/blue/alpha
+ int r = findConfigAttrib(egl, display, config,
+ EGL10.EGL_RED_SIZE, 0);
+ int g = findConfigAttrib(egl, display, config,
+ EGL10.EGL_GREEN_SIZE, 0);
+ int b = findConfigAttrib(egl, display, config,
+ EGL10.EGL_BLUE_SIZE, 0);
+ int a = findConfigAttrib(egl, display, config,
+ EGL10.EGL_ALPHA_SIZE, 0);
+
+ if (r == mRedSize && g == mGreenSize &&
+ b == mBlueSize && a == mAlphaSize)
+ return config;
+ }
+ return null;
+ }
+
+ private int findConfigAttrib(EGL10 egl, EGLDisplay display,
+ EGLConfig config, int attribute,
+ int defaultValue) {
+
+ if (egl.eglGetConfigAttrib(display, config, attribute, mValue)) {
+ return mValue[0];
+ }
+ return defaultValue;
+ }
+
+ // Subclasses can adjust these values:
+ protected int mRedSize;
+ protected int mGreenSize;
+ protected int mBlueSize;
+ protected int mAlphaSize;
+ protected int mDepthSize;
+ protected int mStencilSize;
+ private int[] mValue = new int[1];
+ }
+
+ private native int CreateOpenGLNative(long nativeObject,
+ int width, int height);
+ private native void DrawNative(long nativeObject);
+
+}
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java
new file mode 100644
index 0000000..56d5261
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+import android.content.Context;
+import android.view.SurfaceHolder;
+import android.view.SurfaceView;
+
+public class ViERenderer {
+
+ // View used for local rendering that Cameras can use for Video Overlay.
+ private static SurfaceHolder g_localRenderer;
+
+ public static SurfaceView CreateRenderer(Context context) {
+ return CreateRenderer(context,false);
+ }
+
+ public static SurfaceView CreateRenderer(Context context,
+ boolean useOpenGLES2) {
+ if(useOpenGLES2 == true && ViEAndroidGLES20.IsSupported(context))
+ return new ViEAndroidGLES20(context);
+ else
+ return new SurfaceView(context);
+ }
+
+ // Creates a SurfaceView to be used by Android Camera
+ // service to display a local preview.
+ // This needs to be used on Android prior to version 2.1
+ // in order to run the camera.
+ // Call this function before ViECapture::StartCapture.
+ // The created view needs to be added to a visible layout
+ // after a camera has been allocated
+ // (with the call ViECapture::AllocateCaptureDevice).
+ // IE.
+ // CreateLocalRenderer
+ // ViECapture::AllocateCaptureDevice
+ // LinearLayout.addview
+ // ViECapture::StartCapture
+ public static SurfaceView CreateLocalRenderer(Context context) {
+ SurfaceView localRender = new SurfaceView(context);
+ g_localRenderer = localRender.getHolder();
+ g_localRenderer.setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS);
+ return localRender;
+ }
+
+ public static SurfaceHolder GetLocalRenderer() {
+ return g_localRenderer;
+ }
+
+}
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java
new file mode 100644
index 0000000..3412582
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengine;
+
+import java.nio.ByteBuffer;
+
+import android.graphics.Bitmap;
+import android.graphics.Canvas;
+import android.graphics.Rect;
+import android.util.Log;
+import android.view.SurfaceHolder;
+import android.view.SurfaceView;
+import android.view.SurfaceHolder.Callback;
+
+public class ViESurfaceRenderer implements Callback {
+
+ // the bitmap used for drawing.
+ private Bitmap bitmap = null;
+ private ByteBuffer byteBuffer;
+ private SurfaceHolder surfaceHolder;
+ // Rect of the source bitmap to draw
+ private Rect srcRect = new Rect();
+ // Rect of the destination canvas to draw to
+ private Rect dstRect = new Rect();
+ private int dstHeight = 0;
+ private int dstWidth = 0;
+ private float dstTopScale = 0;
+ private float dstBottomScale = 1;
+ private float dstLeftScale = 0;
+ private float dstRightScale = 1;
+
+ public ViESurfaceRenderer(SurfaceView view) {
+ surfaceHolder = view.getHolder();
+ if(surfaceHolder == null)
+ return;
+
+ Canvas canvas = surfaceHolder.lockCanvas();
+ if(canvas != null) {
+ Rect dst =surfaceHolder.getSurfaceFrame();
+ if(dst != null) {
+ dstRect = dst;
+ dstHeight =dstRect.bottom-dstRect.top;
+ dstWidth = dstRect.right-dstRect.left;
+ }
+ surfaceHolder.unlockCanvasAndPost(canvas);
+ }
+
+ surfaceHolder.addCallback(this);
+ }
+
+ public void surfaceChanged(SurfaceHolder holder, int format,
+ int in_width, int in_height) {
+
+ dstHeight = in_height;
+ dstWidth = in_width;
+ dstRect.left = (int)(dstLeftScale*dstWidth);
+ dstRect.top = (int)(dstTopScale*dstHeight);
+ dstRect.bottom = (int)(dstBottomScale*dstHeight);
+ dstRect.right = (int) (dstRightScale*dstWidth);
+ }
+
+ public void surfaceCreated(SurfaceHolder holder) {
+ // TODO(leozwang) Auto-generated method stub
+ }
+
+ public void surfaceDestroyed(SurfaceHolder holder) {
+ // TODO(leozwang) Auto-generated method stub
+ }
+
+ public Bitmap CreateBitmap(int width, int height) {
+ if (bitmap == null) {
+ try {
+ android.os.Process.setThreadPriority(
+ android.os.Process.THREAD_PRIORITY_DISPLAY);
+ }
+ catch (Exception e) {
+ }
+ }
+ bitmap = Bitmap.createBitmap(width, height, Bitmap.Config.RGB_565);
+ srcRect.left = 0;
+ srcRect.top = 0;
+ srcRect.bottom = height;
+ srcRect.right = width;
+
+ return bitmap;
+ }
+
+ public ByteBuffer CreateByteBuffer(int width, int height) {
+ if (bitmap == null) {
+ try {
+ android.os.Process
+ .setThreadPriority(android.os.Process.THREAD_PRIORITY_DISPLAY);
+ }
+ catch (Exception e) {
+ }
+ }
+
+ try {
+ bitmap = Bitmap.createBitmap(width, height, Bitmap.Config.RGB_565);
+ byteBuffer = ByteBuffer.allocateDirect(width*height*2);
+ srcRect.left = 0;
+ srcRect.top = 0;
+ srcRect.bottom = height;
+ srcRect.right = width;
+ }
+ catch (Exception ex) {
+ Log.e("*WEBRTC*", "Failed to CreateByteBuffer");
+ bitmap = null;
+ byteBuffer = null;
+ }
+
+ return byteBuffer;
+ }
+
+ public void SetCoordinates(float left, float top,
+ float right, float bottom) {
+ dstLeftScale = left;
+ dstTopScale = top;
+ dstRightScale = right;
+ dstBottomScale = bottom;
+
+ dstRect.left = (int)(dstLeftScale*dstWidth);
+ dstRect.top = (int)(dstTopScale*dstHeight);
+ dstRect.bottom = (int)(dstBottomScale*dstHeight);
+ dstRect.right = (int) (dstRightScale*dstWidth);
+ }
+
+ public void DrawByteBuffer() {
+ if(byteBuffer == null)
+ return;
+ byteBuffer.rewind();
+ bitmap.copyPixelsFromBuffer(byteBuffer);
+ DrawBitmap();
+ }
+
+ public void DrawBitmap() {
+ if(bitmap == null)
+ return;
+
+ Canvas canvas = surfaceHolder.lockCanvas();
+ if(canvas != null) {
+ canvas.drawBitmap(bitmap, srcRect, dstRect, null);
+ surfaceHolder.unlockCanvasAndPost(canvas);
+ }
+ }
+
+}
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc
new file mode 100644
index 0000000..22fa659
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc
@@ -0,0 +1,392 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_android_impl.h"
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+
+#ifdef ANDROID_LOG
+#include <stdio.h>
+#include <android/log.h>
+
+#undef WEBRTC_TRACE
+#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
+#else
+#include "trace.h"
+#endif
+
+namespace webrtc {
+JavaVM* VideoRenderAndroid::g_jvm = NULL;
+
+WebRtc_Word32 VideoRenderAndroid::SetAndroidEnvVariables(void* javaVM)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1, "%s", __FUNCTION__);
+
+ g_jvm = (JavaVM*) javaVM;
+
+ return 0;
+
+}
+
+VideoRenderAndroid::VideoRenderAndroid(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool /*fullscreen*/):
+ _id(id),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _renderType(videoRenderType),
+ _ptrWindow((jobject)(window)),
+ _streamsMap(),
+ _javaShutDownFlag(false),
+ _javaShutdownEvent(*EventWrapper::Create()),
+ _javaRenderEvent(*EventWrapper::Create()),
+ _lastJavaRenderEvent(0),
+ _javaRenderJniEnv(NULL),
+ _javaRenderThread(NULL)
+{
+}
+
+VideoRenderAndroid::~VideoRenderAndroid()
+{
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
+ "VideoRenderAndroid dtor");
+
+ if (_javaRenderThread)
+ StopRender();
+
+ for (MapItem* item = _streamsMap.First(); item != NULL; item
+ = _streamsMap.Next(item))
+ { // Delete streams
+ delete static_cast<AndroidStream*> (item->GetItem());
+ }
+ delete &_javaShutdownEvent;
+ delete &_javaRenderEvent;
+ delete &_critSect;
+}
+
+WebRtc_Word32 VideoRenderAndroid::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_critSect);
+ _id = id;
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderAndroid::ChangeWindow(void* /*window*/)
+{
+ return -1;
+}
+
+VideoRenderCallback*
+VideoRenderAndroid::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ AndroidStream* renderStream = NULL;
+ MapItem* item = _streamsMap.Find(streamId);
+ if (item)
+ {
+ renderStream = (AndroidStream*) (item->GetItem());
+ if (NULL != renderStream)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
+ "%s: Render stream already exists", __FUNCTION__);
+ return renderStream;
+ }
+ }
+
+ renderStream = CreateAndroidRenderChannel(streamId, zOrder, left, top,
+ right, bottom, *this);
+ if (renderStream)
+ {
+ _streamsMap.Insert(streamId, renderStream);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
+ return NULL;
+ }
+ return renderStream;
+}
+
+WebRtc_Word32 VideoRenderAndroid::DeleteIncomingRenderStream(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ MapItem* item = _streamsMap.Find(streamId);
+ if (item)
+ {
+ delete (AndroidStream*) item->GetItem();
+ _streamsMap.Erase(streamId);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderAndroid::GetIncomingRenderStreamProperties(
+ const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::StartRender()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ if (_javaRenderThread)
+ {
+ // StartRender is called when this stream should start render.
+ // However StopRender is not called when the streams stop rendering. Thus the the thread is only deleted when the renderer is removed.
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s, Render thread already exist", __FUNCTION__);
+ return 0;
+ }
+
+ _javaRenderThread = ThreadWrapper::CreateThread(JavaRenderThreadFun, this,
+ kRealtimePriority,
+ "AndroidRenderThread");
+ if (!_javaRenderThread)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No thread", __FUNCTION__);
+ return -1;
+ }
+
+ unsigned int tId = 0;
+ if (_javaRenderThread->Start(tId))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
+ "%s: thread started: %u", __FUNCTION__, tId);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not start send thread", __FUNCTION__);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderAndroid::StopRender()
+{
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:", __FUNCTION__);
+ {
+ CriticalSectionScoped cs(_critSect);
+ if (!_javaRenderThread)
+ {
+ return -1;
+ }
+ _javaShutDownFlag = true;
+ _javaRenderEvent.Set();
+ }
+
+ _javaShutdownEvent.Wait(3000);
+ CriticalSectionScoped cs(_critSect);
+ _javaRenderThread->SetNotAlive();
+ if (_javaRenderThread->Stop())
+ {
+ delete _javaRenderThread;
+ _javaRenderThread = NULL;
+ }
+ else
+ {
+ assert(false);
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Not able to stop thread, leaking", __FUNCTION__);
+ _javaRenderThread = NULL;
+ }
+ return 0;
+}
+
+void VideoRenderAndroid::ReDraw()
+{
+ CriticalSectionScoped cs(_critSect);
+ if (_lastJavaRenderEvent < TickTime::MillisecondTimestamp() - 20) // Allow redraw if it was more than 20ms since last.
+ {
+ _lastJavaRenderEvent = TickTime::MillisecondTimestamp();
+ _javaRenderEvent.Set();
+ }
+}
+
+bool VideoRenderAndroid::JavaRenderThreadFun(void* obj)
+{
+ return static_cast<VideoRenderAndroid*> (obj)->JavaRenderThreadProcess();
+}
+
+bool VideoRenderAndroid::JavaRenderThreadProcess()
+{
+ _javaRenderEvent.Wait(1000);
+
+ CriticalSectionScoped cs(_critSect);
+ if (!_javaRenderJniEnv)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&_javaRenderJniEnv, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !_javaRenderJniEnv)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, _javaRenderJniEnv);
+ return false;
+ }
+ }
+
+ for (MapItem* item = _streamsMap.First(); item != NULL; item
+ = _streamsMap.Next(item))
+ {
+ static_cast<AndroidStream*> (item->GetItem())->DeliverFrame(
+ _javaRenderJniEnv);
+ }
+
+ if (_javaShutDownFlag)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
+ "%s: Java thread detached", __FUNCTION__);
+ }
+ _javaRenderJniEnv = false;
+ _javaShutDownFlag = false;
+ _javaShutdownEvent.Set();
+ return false; // Do not run this thread again.
+ }
+ return true;
+}
+
+VideoRenderType VideoRenderAndroid::RenderType()
+{
+ return _renderType;
+}
+
+RawVideoType VideoRenderAndroid::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool VideoRenderAndroid::FullScreen()
+{
+ return false;
+}
+
+WebRtc_Word32 VideoRenderAndroid::GetGraphicsMemory(
+ WebRtc_UWord64& /*totalGraphicsMemory*/,
+ WebRtc_UWord64& /*availableGraphicsMemory*/) const
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::GetScreenResolution(
+ WebRtc_UWord32& /*screenWidth*/,
+ WebRtc_UWord32& /*screenHeight*/) const
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_UWord32 VideoRenderAndroid::RenderFrameRate(const WebRtc_UWord32 /*streamId*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::SetStreamCropping(
+ const WebRtc_UWord32 /*streamId*/,
+ const float /*left*/,
+ const float /*top*/,
+ const float /*right*/,
+ const float /*bottom*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::SetTransparentBackground(const bool enable)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::ConfigureRenderer(
+ const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::SetText(
+ const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float rigth, const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAndroid::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left, const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Android", __FUNCTION__);
+ return -1;
+}
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h
new file mode 100644
index 0000000..7058871
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
+
+#include <jni.h>
+#include "i_video_render.h"
+#include "map_wrapper.h"
+
+
+namespace webrtc {
+
+//#define ANDROID_LOG
+
+
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+
+
+// The object a module user uses to send new frames to the java renderer
+// Base class for android render streams.
+
+class AndroidStream: public VideoRenderCallback
+{
+public:
+ /*
+ * DeliverFrame is called from a thread connected to the Java VM.
+ * Used for Delivering frame for rendering.
+ */
+ virtual void DeliverFrame(JNIEnv* jniEnv)=0;
+
+ virtual ~AndroidStream()
+ {
+ };
+};
+
+class VideoRenderAndroid: IVideoRender
+{
+public:
+ static WebRtc_Word32 SetAndroidEnvVariables(void* javaVM);
+
+ VideoRenderAndroid(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType, void* window,
+ const bool fullscreen);
+
+ virtual ~VideoRenderAndroid();
+
+ virtual WebRtc_Word32 Init()=0;
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const;
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ virtual void ReDraw();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32
+ GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float rigth, const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+protected:
+ virtual AndroidStream
+ * CreateAndroidRenderChannel(WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom,
+ VideoRenderAndroid& renderer) = 0;
+
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _critSect;
+ VideoRenderType _renderType;
+ jobject _ptrWindow;
+
+ static JavaVM* g_jvm;
+
+private:
+ static bool JavaRenderThreadFun(void* obj);
+ bool JavaRenderThreadProcess();
+
+ MapWrapper _streamsMap; // Map with streams to render.
+ bool _javaShutDownFlag; // True if the _javaRenderThread thread shall be detached from the JVM.
+ EventWrapper& _javaShutdownEvent;
+ EventWrapper& _javaRenderEvent;
+ WebRtc_Word64 _lastJavaRenderEvent;
+ JNIEnv* _javaRenderJniEnv; // JNIEnv for the java render thread.
+ ThreadWrapper* _javaRenderThread;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc
new file mode 100644
index 0000000..a6d7133
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc
@@ -0,0 +1,495 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_android_native_opengl2.h"
+#include "critical_section_wrapper.h"
+#include "tick_util.h"
+
+#ifdef ANDROID_LOG
+#include <stdio.h>
+#include <android/log.h>
+
+#undef WEBRTC_TRACE
+#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
+#else
+#include "trace.h"
+#endif
+
+namespace webrtc {
+
+AndroidNativeOpenGl2Renderer::AndroidNativeOpenGl2Renderer(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+ VideoRenderAndroid(id, videoRenderType, window, fullscreen),
+ _javaRenderObj(NULL),
+ _javaRenderClass(NULL)
+{
+}
+
+bool AndroidNativeOpenGl2Renderer::UseOpenGL2(void* window)
+{
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "RendererAndroid():UseOpenGL No JVM set.");
+ return false;
+ }
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideoRenderer,
+ -1,
+ "RendererAndroid(): Could not attach thread to JVM (%d, %p)",
+ res, env);
+ return false;
+ }
+ isAttached = true;
+ }
+
+ // get the renderer class
+ jclass javaRenderClassLocal =
+ env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
+ if (!javaRenderClassLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "%s: could not find ViEAndroidRenderer class",
+ __FUNCTION__);
+ return false;
+ }
+
+ // get the method ID for UseOpenGL
+ jmethodID cidUseOpenGL = env->GetStaticMethodID(javaRenderClassLocal,
+ "UseOpenGL2",
+ "(Ljava/lang/Object;)Z");
+ if (cidUseOpenGL == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "%s: could not get UseOpenGL ID", __FUNCTION__);
+ return false;
+ }
+ jboolean res = env->CallStaticBooleanMethod(javaRenderClassLocal,
+ cidUseOpenGL, (jobject) window);
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+ return res;
+}
+
+AndroidNativeOpenGl2Renderer::~AndroidNativeOpenGl2Renderer()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
+ "AndroidNativeOpenGl2Renderer dtor");
+ if (g_jvm)
+ {
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ env = NULL;
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+
+ env->DeleteGlobalRef(_javaRenderObj);
+ env->DeleteGlobalRef(_javaRenderClass);
+
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Could not detach thread from JVM",
+ __FUNCTION__);
+ }
+ }
+ }
+}
+
+WebRtc_Word32 AndroidNativeOpenGl2Renderer::Init()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "(%s): Not a valid Java VM pointer.", __FUNCTION__);
+ return -1;
+ }
+ if (!_ptrWindow)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "(%s): No window have been provided.", __FUNCTION__);
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the ViEAndroidGLES20 class
+ jclass javaRenderClassLocal =
+ env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
+ if (!javaRenderClassLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not find ViEAndroidGLES20", __FUNCTION__);
+ return -1;
+ }
+
+ // create a global reference to the class (to tell JNI that we are referencing it after this function has returned)
+ _javaRenderClass
+ = reinterpret_cast<jclass> (env->NewGlobalRef(javaRenderClassLocal));
+ if (!_javaRenderClass)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not create Java SurfaceHolder class reference",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local class ref, we only use the global ref
+ env->DeleteLocalRef(javaRenderClassLocal);
+
+ // create a reference to the object (to tell JNI that we are referencing it
+ // after this function has returned)
+ _javaRenderObj = env->NewGlobalRef(_ptrWindow);
+ if (!_javaRenderObj)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideoRenderer,
+ _id,
+ "%s: could not create Java SurfaceRender object reference",
+ __FUNCTION__);
+ return -1;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done",
+ __FUNCTION__);
+ return 0;
+
+}
+AndroidStream*
+AndroidNativeOpenGl2Renderer::CreateAndroidRenderChannel(
+ WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom,
+ VideoRenderAndroid& renderer)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: Id %d",
+ __FUNCTION__, streamId);
+ AndroidNativeOpenGl2Channel* stream =
+ new AndroidNativeOpenGl2Channel(streamId, g_jvm, renderer,
+ _javaRenderObj);
+ if (stream && stream->Init(zOrder, left, top, right, bottom) == 0)
+ return stream;
+ else
+ {
+ delete stream;
+ }
+ return NULL;
+}
+
+AndroidNativeOpenGl2Channel::AndroidNativeOpenGl2Channel(WebRtc_UWord32 streamId,
+ JavaVM* jvm,
+ VideoRenderAndroid& renderer,jobject javaRenderObj):
+ _id(streamId),
+ _renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _renderer(renderer), _jvm(jvm), _javaRenderObj(javaRenderObj),
+ _registerNativeCID(NULL), _deRegisterNativeCID(NULL),
+ _openGLRenderer(streamId)
+{
+
+}
+AndroidNativeOpenGl2Channel::~AndroidNativeOpenGl2Channel()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
+ "AndroidNativeOpenGl2Channel dtor");
+ delete &_renderCritSect;
+ if (_jvm)
+ {
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ env = NULL;
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+ if (env && _deRegisterNativeCID)
+ {
+ env->CallVoidMethod(_javaRenderObj, _deRegisterNativeCID);
+ }
+
+ if (isAttached)
+ {
+ if (_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Could not detach thread from JVM",
+ __FUNCTION__);
+ }
+ }
+ }
+}
+
+WebRtc_Word32 AndroidNativeOpenGl2Channel::Init(WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s: AndroidNativeOpenGl2Channel", __FUNCTION__);
+ if (!_jvm)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Not a valid Java VM pointer", __FUNCTION__);
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not attach thread to JVM (%d, %p)",
+ __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ jclass javaRenderClass =
+ env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
+ if (!javaRenderClass)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not find ViESurfaceRenderer", __FUNCTION__);
+ return -1;
+ }
+
+ // get the method ID for the ReDraw function
+ _redrawCid = env->GetMethodID(javaRenderClass, "ReDraw", "()V");
+ if (_redrawCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not get ReDraw ID", __FUNCTION__);
+ return -1;
+ }
+
+ _registerNativeCID = env->GetMethodID(javaRenderClass,
+ "RegisterNativeObject", "(J)V");
+ if (_registerNativeCID == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not get RegisterNativeObject ID", __FUNCTION__);
+ return -1;
+ }
+
+ _deRegisterNativeCID = env->GetMethodID(javaRenderClass,
+ "DeRegisterNativeObject", "()V");
+ if (_deRegisterNativeCID == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not get DeRegisterNativeObject ID",
+ __FUNCTION__);
+ return -1;
+ }
+
+ JNINativeMethod
+ nativeFunctions[2] = {
+ "DrawNative",
+ "(J)V",
+ (void*) &AndroidNativeOpenGl2Channel::DrawNativeStatic,
+ "CreateOpenGLNative",
+ "(JII)I",
+ (void*) &AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic };
+ if (env->RegisterNatives(javaRenderClass, nativeFunctions, 2) == 0)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1,
+ "%s: Registered native functions", __FUNCTION__);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "%s: Failed to register native functions", __FUNCTION__);
+ return -1;
+ }
+
+ env->CallVoidMethod(_javaRenderObj, _registerNativeCID, (jlong) this);
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
+ "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ if (_openGLRenderer.SetCoordinates(zOrder, left, top, right, bottom) != 0)
+ {
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s: AndroidNativeOpenGl2Channel done", __FUNCTION__);
+ return 0;
+}
+
+WebRtc_Word32 AndroidNativeOpenGl2Channel::RenderFrame(const WebRtc_UWord32 /*streamId*/,
+ VideoFrame& videoFrame)
+{
+ // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
+ _renderCritSect.Enter();
+ _bufferToRender.SwapFrame(videoFrame);
+ _renderCritSect.Leave();
+ _renderer.ReDraw();
+ return 0;
+}
+
+/*Implements AndroidStream
+ * Calls the Java object and render the buffer in _bufferToRender
+ */
+void AndroidNativeOpenGl2Channel::DeliverFrame(JNIEnv* jniEnv)
+{
+ //TickTime timeNow=TickTime::Now();
+
+ //Draw the Surface
+ jniEnv->CallVoidMethod(_javaRenderObj, _redrawCid);
+
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s: time to deliver %lld" ,__FUNCTION__,(TickTime::Now()-timeNow).Milliseconds());
+}
+
+/*
+ * JNI callback from Java class. Called when the render want to render a frame. Called from the GLRenderThread
+ * Method: DrawNative
+ * Signature: (J)V
+ */
+void JNICALL AndroidNativeOpenGl2Channel::DrawNativeStatic
+(JNIEnv * env, jobject, jlong context)
+{
+ AndroidNativeOpenGl2Channel* renderChannel=reinterpret_cast<AndroidNativeOpenGl2Channel*>(context);
+ renderChannel->DrawNative();
+}
+
+void AndroidNativeOpenGl2Channel::DrawNative()
+{
+ _openGLRenderer.Render(_bufferToRender);
+}
+/*
+ * JNI callback from Java class. Called when the GLSurfaceview have created a surface. Called from the GLRenderThread
+ * Method: CreateOpenGLNativeStatic
+ * Signature: (JII)I
+ */
+jint JNICALL AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic(JNIEnv * env,
+ jobject,
+ jlong context,
+ jint width,
+ jint height)
+{
+ AndroidNativeOpenGl2Channel* renderChannel =
+ reinterpret_cast<AndroidNativeOpenGl2Channel*> (context);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1, "%s:", __FUNCTION__);
+ return renderChannel->CreateOpenGLNative(width, height);
+}
+
+jint AndroidNativeOpenGl2Channel::CreateOpenGLNative(int width, int height)
+{
+
+ return _openGLRenderer.Setup(width, height);
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h
new file mode 100644
index 0000000..54532a6
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
+
+#include <jni.h>
+
+#include "video_render_defines.h"
+
+#include "video_render_android_impl.h"
+#include "video_render_opengles20.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class AndroidNativeOpenGl2Channel: public AndroidStream
+{
+
+public:
+ AndroidNativeOpenGl2Channel(WebRtc_UWord32 streamId,JavaVM* jvm,VideoRenderAndroid& renderer,jobject javaRenderObj);
+ ~AndroidNativeOpenGl2Channel();
+
+ WebRtc_Word32 Init(WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ //Implement VideoRenderCallback
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
+
+ //Implements AndroidStream
+ virtual void DeliverFrame(JNIEnv* jniEnv);
+
+private:
+ static jint CreateOpenGLNativeStatic(JNIEnv * env,jobject, jlong context, jint width, jint height);
+ jint CreateOpenGLNative(int width, int height);
+
+ static void DrawNativeStatic(JNIEnv * env,jobject, jlong context);
+ void DrawNative();
+ WebRtc_UWord32 _id;
+ CriticalSectionWrapper& _renderCritSect;
+
+ VideoFrame _bufferToRender;
+ VideoRenderAndroid& _renderer;
+ JavaVM* _jvm;
+ jobject _javaRenderObj;
+
+ jmethodID _redrawCid;
+ jmethodID _registerNativeCID;
+ jmethodID _deRegisterNativeCID;
+ VideoRenderOpenGles20 _openGLRenderer;
+
+};
+
+
+class AndroidNativeOpenGl2Renderer: private VideoRenderAndroid
+{
+public:
+ AndroidNativeOpenGl2Renderer(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen);
+
+ ~AndroidNativeOpenGl2Renderer();
+ static bool UseOpenGL2(void* window);
+
+ WebRtc_Word32 Init();
+ virtual AndroidStream* CreateAndroidRenderChannel(WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom,
+ VideoRenderAndroid& renderer);
+
+private:
+ jobject _javaRenderObj;
+ jclass _javaRenderClass;
+
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc
new file mode 100644
index 0000000..253d831
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc
@@ -0,0 +1,470 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_android_surface_view.h"
+#include "critical_section_wrapper.h"
+#include "common_video/libyuv/include/libyuv.h"
+#include "tick_util.h"
+#ifdef ANDROID_NDK_8_OR_ABOVE
+ #include <android/bitmap.h>
+#endif
+
+
+#ifdef ANDROID_LOG
+#include <stdio.h>
+#include <android/log.h>
+
+#undef WEBRTC_TRACE
+#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
+#else
+#include "trace.h"
+#endif
+
+namespace webrtc {
+
+AndroidSurfaceViewRenderer::AndroidSurfaceViewRenderer(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen)
+:
+ VideoRenderAndroid(id,videoRenderType,window,fullscreen),
+ _javaRenderObj(NULL),
+ _javaRenderClass(NULL)
+{
+}
+
+AndroidSurfaceViewRenderer::~AndroidSurfaceViewRenderer()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "AndroidSurfaceViewRenderer dtor");
+ if(g_jvm)
+ {
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
+ env=NULL;
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+ env->DeleteGlobalRef(_javaRenderObj);
+ env->DeleteGlobalRef(_javaRenderClass);
+
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+ }
+}
+
+
+WebRtc_Word32
+AndroidSurfaceViewRenderer::Init()
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ if (!g_jvm)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "(%s): Not a valid Java VM pointer.", __FUNCTION__);
+ return -1;
+ }
+ if(!_ptrWindow)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "(%s): No window have been provided.", __FUNCTION__);
+ return -1;
+ }
+
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = g_jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ // get the ViESurfaceRender class
+ jclass javaRenderClassLocal = env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
+ if (!javaRenderClassLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not find ViESurfaceRenderer", __FUNCTION__);
+ return -1;
+ }
+
+ // create a global reference to the class (to tell JNI that we are referencing it after this function has returned)
+ _javaRenderClass = reinterpret_cast<jclass>(env->NewGlobalRef(javaRenderClassLocal));
+ if (!_javaRenderClass)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java ViESurfaceRenderer class reference", __FUNCTION__);
+ return -1;
+ }
+
+ // Delete local class ref, we only use the global ref
+ env->DeleteLocalRef(javaRenderClassLocal);
+
+ // get the method ID for the constructor
+ jmethodID cid = env->GetMethodID(_javaRenderClass, "<init>", "(Landroid/view/SurfaceView;)V");
+ if (cid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get constructor ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ // construct the object
+ jobject javaRenderObjLocal = env->NewObject(_javaRenderClass, cid, _ptrWindow);
+ if (!javaRenderObjLocal)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java Render", __FUNCTION__);
+ return -1;
+ }
+
+ // create a reference to the object (to tell JNI that we are referencing it
+ // after this function has returned)
+ _javaRenderObj = env->NewGlobalRef(javaRenderObjLocal);
+ if (!_javaRenderObj)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java SurfaceRender object reference", __FUNCTION__);
+ return -1;
+ }
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (g_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done", __FUNCTION__);
+ return 0;
+
+}
+AndroidStream*
+AndroidSurfaceViewRenderer::CreateAndroidRenderChannel(WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom,
+ VideoRenderAndroid& renderer)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: Id %d", __FUNCTION__,streamId);
+ AndroidSurfaceViewChannel* stream=new AndroidSurfaceViewChannel(streamId,g_jvm,renderer,_javaRenderObj);
+ if(stream && stream->Init(zOrder,left,top,right,bottom)==0)
+ return stream;
+ else
+ delete stream;
+ return NULL;
+}
+
+
+
+
+
+
+AndroidSurfaceViewChannel::AndroidSurfaceViewChannel(WebRtc_UWord32 streamId,JavaVM* jvm,VideoRenderAndroid& renderer,jobject javaRenderObj)
+:
+_id(streamId),
+_renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_renderer(renderer),
+_jvm(jvm),
+_javaRenderObj(javaRenderObj),
+_bitmapWidth(0),
+_bitmapHeight(0)
+{
+
+}
+AndroidSurfaceViewChannel::~AndroidSurfaceViewChannel()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "AndroidSurfaceViewChannel dtor");
+ delete &_renderCritSect;
+ if(_jvm)
+ {
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
+ env=NULL;
+ }
+ else
+ {
+ isAttached = true;
+ }
+ }
+
+#ifdef ANDROID_NDK_8_OR_ABOVE
+ env->DeleteGlobalRef(_javaBitmapObj);
+#else
+ env->DeleteGlobalRef(_javaByteBufferObj);
+#endif
+ if (isAttached)
+ {
+ if (_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+ }
+}
+
+WebRtc_Word32
+AndroidSurfaceViewChannel::Init(WebRtc_Word32 /*zOrder*/,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: AndroidSurfaceViewChannel", __FUNCTION__);
+ if (!_jvm)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer,_id, "%s: Not a valid Java VM pointer", __FUNCTION__);
+ return -1;
+ }
+
+ if((top>1 || top<0) || (right>1 || right<0) || (bottom>1 || bottom<0) || (left>1 || left<0))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Wrong coordinates",
+ __FUNCTION__);
+ return -1;
+ }
+
+
+ // get the JNI env for this thread
+ bool isAttached = false;
+ JNIEnv* env = NULL;
+ if (_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
+ {
+ // try to attach the thread and get the env
+ // Attach this thread to JVM
+ jint res = _jvm->AttachCurrentThread(&env, NULL);
+
+ // Get the JNI env for this thread
+ if ((res < 0) || !env)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
+ return -1;
+ }
+ isAttached = true;
+ }
+
+ jclass javaRenderClass = env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
+ if (!javaRenderClass)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not find ViESurfaceRenderer", __FUNCTION__);
+ return -1;
+ }
+#ifdef ANDROID_NDK_8_OR_ABOVE
+ // get the method ID for the CreateBitmap
+ _createBitmapCid = env->GetMethodID(_javaRenderClass, "CreateBitmap", "(II)Landroid/graphics/Bitmap;");
+ if (_createBitmapCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get CreateBitmap ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+ // get the method ID for the DrawBitmap function
+ _drawBitmapCid = env->GetMethodID(_javaRenderClass, "DrawBitmap", "()V");
+ if (_drawBitmapCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get DrawBitmap ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+#else
+ // get the method ID for the CreateIntArray
+ _createByteBufferCid = env->GetMethodID(javaRenderClass, "CreateByteBuffer", "(II)Ljava/nio/ByteBuffer;");
+ if (_createByteBufferCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get CreateByteBuffer ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ // get the method ID for the DrawByteBuffer function
+ _drawByteBufferCid = env->GetMethodID(javaRenderClass, "DrawByteBuffer", "()V");
+ if (_drawByteBufferCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get DrawByteBuffer ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+#endif
+
+ // get the method ID for the SetCoordinates function
+ _setCoordinatesCid = env->GetMethodID(javaRenderClass, "SetCoordinates", "(FFFF)V");
+ if (_setCoordinatesCid == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get SetCoordinates ID", __FUNCTION__);
+ return -1; /* exception thrown */
+ }
+
+ env->CallVoidMethod(_javaRenderObj,_setCoordinatesCid,left,top,right,bottom);
+
+ // Detach this thread if it was attached
+ if (isAttached)
+ {
+ if (_jvm->DetachCurrentThread() < 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
+ }
+ }
+
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: AndroidSurfaceViewChannel done", __FUNCTION__);
+ return 0;
+}
+
+
+WebRtc_Word32 AndroidSurfaceViewChannel::RenderFrame(const WebRtc_UWord32 /*streamId*/, VideoFrame& videoFrame)
+{
+ // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
+ _renderCritSect.Enter();
+ _bufferToRender.SwapFrame(videoFrame);
+ _renderCritSect.Leave();
+ _renderer.ReDraw();
+ return 0;
+}
+
+
+/*Implements AndroidStream
+ * Calls the Java object and render the buffer in _bufferToRender
+ */
+void AndroidSurfaceViewChannel::DeliverFrame(JNIEnv* jniEnv) {
+ _renderCritSect.Enter();
+
+#ifdef ANDROID_NDK_8_OR_ABOVE
+ if (_bitmapWidth != _bufferToRender.Width() ||
+ _bitmapHeight != _bufferToRender.Height()) {
+ // Create the bitmap to write to
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: Creating bitmap %u "
+ "%u", __FUNCTION__, _bufferToRender.Width(),
+ _bufferToRender.Height());
+ if (_javaBitmapObj) {
+ jniEnv->DeleteGlobalRef(_javaBitmapObj);
+ _javaBitmapObj = NULL;
+ }
+ jobject javaBitmap = jniEnv->CallObjectMethod(_javaRenderObj,
+ _createBitmapCid,
+ videoFrame.Width(),
+ videoFrame.Height());
+ _javaBitmapObj = jniEnv->NewGlobalRef(javaBitmap);
+ if (!_javaBitmapObj) {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not "
+ "create Java Bitmap object reference", __FUNCTION__);
+ _renderCritSect.Leave();
+ return;
+ } else {
+ _bitmapWidth=_bufferToRender.Width();
+ _bitmapHeight=_bufferToRender.Height();
+ }
+ }
+ void* pixels;
+ if (_javaBitmapObj &&
+ AndroidBitmap_lockPixels(jniEnv, _javaBitmapObj, &pixels) >= 0) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: Locked bitmap",
+ __FUNCTION__);
+ // Convert I420 straight into the Java bitmap.
+ int ret = ConvertI420ToRGB565((unsigned char* )_bufferToRender.Buffer(),
+ (unsigned char* ) pixels,
+ _bitmapWidth, _bitmapHeight);
+ if (ret < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Color conversion "
+ "failed.", __FUNCTION__);
+ }
+
+ AndroidBitmap_unlockPixels(jniEnv, _javaBitmapObj);
+ // Draw the Surface.
+ jniEnv->CallVoidMethod(_javaRenderObj,_drawCid);
+
+ } else {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not lock "
+ "bitmap", __FUNCTION__);
+ }
+ _renderCritSect.Leave();
+
+#else
+ if (_bitmapWidth != _bufferToRender.Width() ||
+ _bitmapHeight != _bufferToRender.Height()) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: New render size %d "
+ "%d",__FUNCTION__,
+ _bufferToRender.Width(), _bufferToRender.Height());
+ if (_javaByteBufferObj) {
+ jniEnv->DeleteGlobalRef(_javaByteBufferObj);
+ _javaByteBufferObj = NULL;
+ _directBuffer = NULL;
+ }
+ jobject javaByteBufferObj =
+ jniEnv->CallObjectMethod(_javaRenderObj, _createByteBufferCid,
+ _bufferToRender.Width(),
+ _bufferToRender.Height());
+ _javaByteBufferObj = jniEnv->NewGlobalRef(javaByteBufferObj);
+ if (!_javaByteBufferObj) {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not "
+ "create Java ByteBuffer object reference", __FUNCTION__);
+ _renderCritSect.Leave();
+ return;
+ } else {
+ _directBuffer = static_cast<unsigned char*>
+ (jniEnv->GetDirectBufferAddress(_javaByteBufferObj));
+ _bitmapWidth = _bufferToRender.Width();
+ _bitmapHeight = _bufferToRender.Height();
+ }
+ }
+
+ if(_javaByteBufferObj && _bitmapWidth && _bitmapHeight) {
+ // Android requires a vertically flipped image compared to std convert.
+ // This is done by giving a negative height input.
+ const int conversionResult =
+ ConvertI420ToRGB565((unsigned char* )_bufferToRender.Buffer(),
+ _directBuffer, _bitmapWidth, -_bitmapHeight);
+ if (conversionResult < 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Color conversion"
+ " failed.", __FUNCTION__);
+ _renderCritSect.Leave();
+ return;
+ }
+ }
+ _renderCritSect.Leave();
+ // Draw the Surface
+ jniEnv->CallVoidMethod(_javaRenderObj, _drawByteBufferCid);
+#endif
+}
+
+} // namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h
new file mode 100644
index 0000000..f55e60b
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
+
+#include <jni.h>
+
+#include "video_render_defines.h"
+
+#include "video_render_android_impl.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+
+class AndroidSurfaceViewChannel: public AndroidStream
+{
+
+public:
+ AndroidSurfaceViewChannel(WebRtc_UWord32 streamId,
+ JavaVM* jvm,
+ VideoRenderAndroid& renderer,
+ jobject javaRenderObj);
+ ~AndroidSurfaceViewChannel();
+
+ WebRtc_Word32 Init(WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ //Implement VideoRenderCallback
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+ //Implements AndroidStream
+ virtual void DeliverFrame(JNIEnv* jniEnv);
+
+private:
+ WebRtc_UWord32 _id;
+ CriticalSectionWrapper& _renderCritSect;
+
+ VideoFrame _bufferToRender;
+ VideoRenderAndroid& _renderer;
+ JavaVM* _jvm;
+ jobject _javaRenderObj;
+
+#ifdef ANDROID_NDK_8_OR_ABOVE
+ jclass _javaBitmapClass;
+ jmethodID _createBitmapCid;
+ jobject _javaBitmapObj;
+ jmethodID _drawBitmapCid;
+#else
+ jobject _javaByteBufferObj;
+ unsigned char* _directBuffer;
+ jmethodID _createByteBufferCid;
+ jmethodID _drawByteBufferCid;
+#endif
+ jmethodID _setCoordinatesCid;
+ unsigned int _bitmapWidth;
+ unsigned int _bitmapHeight;
+};
+
+class AndroidSurfaceViewRenderer: private VideoRenderAndroid
+{
+public:
+ AndroidSurfaceViewRenderer(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen);
+ ~AndroidSurfaceViewRenderer();
+ WebRtc_Word32 Init();
+ virtual AndroidStream* CreateAndroidRenderChannel(WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom,
+ VideoRenderAndroid& renderer);
+private:
+ jobject _javaRenderObj;
+ jclass _javaRenderClass;
+
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc
new file mode 100644
index 0000000..8f4e5c5
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc
@@ -0,0 +1,446 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <GLES2/gl2.h>
+#include <GLES2/gl2ext.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "video_render_opengles20.h"
+
+//#define ANDROID_LOG
+
+#ifdef ANDROID_LOG
+#include <stdio.h>
+#include <android/log.h>
+
+#undef WEBRTC_TRACE
+#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
+#else
+#include "trace.h"
+#endif
+
+namespace webrtc {
+
+const char VideoRenderOpenGles20::g_indices[] = { 0, 3, 2, 0, 2, 1 };
+
+const char VideoRenderOpenGles20::g_vertextShader[] = {
+ "attribute vec4 aPosition;\n"
+ "attribute vec2 aTextureCoord;\n"
+ "varying vec2 vTextureCoord;\n"
+ "void main() {\n"
+ " gl_Position = aPosition;\n"
+ " vTextureCoord = aTextureCoord;\n"
+ "}\n" };
+
+// The fragment shader.
+// Do YUV to RGB565 conversion.
+const char VideoRenderOpenGles20::g_fragmentShader[] = {
+ "precision mediump float;\n"
+ "uniform sampler2D Ytex;\n"
+ "uniform sampler2D Utex,Vtex;\n"
+ "varying vec2 vTextureCoord;\n"
+ "void main(void) {\n"
+ " float nx,ny,r,g,b,y,u,v;\n"
+ " mediump vec4 txl,ux,vx;"
+ " nx=vTextureCoord[0];\n"
+ " ny=vTextureCoord[1];\n"
+ " y=texture2D(Ytex,vec2(nx,ny)).r;\n"
+ " u=texture2D(Utex,vec2(nx,ny)).r;\n"
+ " v=texture2D(Vtex,vec2(nx,ny)).r;\n"
+
+ //" y = v;\n"+
+ " y=1.1643*(y-0.0625);\n"
+ " u=u-0.5;\n"
+ " v=v-0.5;\n"
+
+ " r=y+1.5958*v;\n"
+ " g=y-0.39173*u-0.81290*v;\n"
+ " b=y+2.017*u;\n"
+ " gl_FragColor=vec4(r,g,b,1.0);\n"
+ "}\n" };
+
+VideoRenderOpenGles20::VideoRenderOpenGles20(WebRtc_Word32 id) :
+ _id(id),
+ _textureWidth(-1),
+ _textureHeight(-1)
+
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
+ __FUNCTION__, (int) _id);
+
+ const GLfloat vertices[20] = {
+ // X, Y, Z, U, V
+ -1, -1, 0, 0, 1, // Bottom Left
+ 1, -1, 0, 1, 1, //Bottom Right
+ 1, 1, 0, 1, 0, //Top Right
+ -1, 1, 0, 0, 0 }; //Top Left
+
+ memcpy(_vertices, vertices, sizeof(_vertices));
+}
+
+VideoRenderOpenGles20::~VideoRenderOpenGles20()
+{
+
+}
+
+WebRtc_Word32 VideoRenderOpenGles20::Setup(WebRtc_Word32 width,
+ WebRtc_Word32 height)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s: width %d, height %d", __FUNCTION__, (int) width,
+ (int) height);
+
+ printGLString("Version", GL_VERSION);
+ printGLString("Vendor", GL_VENDOR);
+ printGLString("Renderer", GL_RENDERER);
+ printGLString("Extensions", GL_EXTENSIONS);
+
+ int maxTextureImageUnits[2];
+ int maxTextureSize[2];
+ glGetIntegerv(GL_MAX_TEXTURE_IMAGE_UNITS, maxTextureImageUnits);
+ glGetIntegerv(GL_MAX_TEXTURE_SIZE, maxTextureSize);
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s: number of textures %d, size %d", __FUNCTION__,
+ (int) maxTextureImageUnits[0], (int) maxTextureSize[0]);
+
+ _program = createProgram(g_vertextShader, g_fragmentShader);
+ if (!_program)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not create program", __FUNCTION__);
+ return -1;
+ }
+
+ int positionHandle = glGetAttribLocation(_program, "aPosition");
+ checkGlError("glGetAttribLocation aPosition");
+ if (positionHandle == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not get aPosition handle", __FUNCTION__);
+ return -1;
+ }
+ int textureHandle = glGetAttribLocation(_program, "aTextureCoord");
+ checkGlError("glGetAttribLocation aTextureCoord");
+ if (textureHandle == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not get aTextureCoord handle", __FUNCTION__);
+ return -1;
+ }
+
+ // set the vertices array in the shader
+ // _vertices contains 4 vertices with 5 coordinates. 3 for (xyz) for the vertices and 2 for the texture
+ glVertexAttribPointer(positionHandle, 3, GL_FLOAT, false, 5
+ * sizeof(GLfloat), _vertices);
+ checkGlError("glVertexAttribPointer aPosition");
+
+ glEnableVertexAttribArray(positionHandle);
+ checkGlError("glEnableVertexAttribArray positionHandle");
+
+ // set the texture coordinate array in the shader
+ // _vertices contains 4 vertices with 5 coordinates. 3 for (xyz) for the vertices and 2 for the texture
+ glVertexAttribPointer(textureHandle, 2, GL_FLOAT, false, 5
+ * sizeof(GLfloat), &_vertices[3]);
+ checkGlError("glVertexAttribPointer maTextureHandle");
+ glEnableVertexAttribArray(textureHandle);
+ checkGlError("glEnableVertexAttribArray textureHandle");
+
+ glUseProgram(_program);
+ int i = glGetUniformLocation(_program, "Ytex");
+ checkGlError("glGetUniformLocation");
+ glUniform1i(i, 0); /* Bind Ytex to texture unit 0 */
+ checkGlError("glUniform1i Ytex");
+
+ i = glGetUniformLocation(_program, "Utex");
+ checkGlError("glGetUniformLocation Utex");
+ glUniform1i(i, 1); /* Bind Utex to texture unit 1 */
+ checkGlError("glUniform1i Utex");
+
+ i = glGetUniformLocation(_program, "Vtex");
+ checkGlError("glGetUniformLocation");
+ glUniform1i(i, 2); /* Bind Vtex to texture unit 2 */
+ checkGlError("glUniform1i");
+
+ glViewport(0, 0, width, height);
+ checkGlError("glViewport");
+ return 0;
+
+}
+/*
+ * SetCoordinates
+ * Sets the coordinates where the stream shall be rendered. Values must be between 0 and 1.
+ */
+WebRtc_Word32 VideoRenderOpenGles20::SetCoordinates(WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ if ((top > 1 || top < 0) || (right > 1 || right < 0) || (bottom > 1
+ || bottom < 0) || (left > 1 || left < 0))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Wrong coordinates", __FUNCTION__);
+ return -1;
+ }
+ /*
+ // X, Y, Z, U, V
+ -1, -1, 0, 0, 1, // Bottom Left
+ 1, -1, 0, 1, 1, //Bottom Right
+ 1, 1, 0, 1, 0, //Top Right
+ -1, 1, 0, 0, 0 }; //Top Left
+ */
+ // Bottom Left
+ _vertices[0] = (left * 2) - 1;
+ _vertices[1] = -1 * (2 * bottom) + 1;
+ _vertices[2] = zOrder;
+
+ //Bottom Right
+ _vertices[5] = (right * 2) - 1;
+ _vertices[6] = -1 * (2 * bottom) + 1;
+ _vertices[7] = zOrder;
+
+ //Top Right
+ _vertices[10] = (right * 2) - 1;
+ _vertices[11] = -1 * (2 * top) + 1;
+ _vertices[12] = zOrder;
+
+ //Top Left
+ _vertices[15] = (left * 2) - 1;
+ _vertices[16] = -1 * (2 * top) + 1;
+ _vertices[17] = zOrder;
+
+ return 0;
+
+}
+WebRtc_Word32 VideoRenderOpenGles20::Render(const VideoFrame& frameToRender)
+{
+
+ if (frameToRender.Length() == 0)
+ {
+ return -1;
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
+ __FUNCTION__, (int) _id);
+
+ //glClearColor(0.0f, 0.0f, 1.0f, 1.0f);
+ //glClear(GL_DEPTH_BUFFER_BIT | GL_COLOR_BUFFER_BIT);
+
+ glUseProgram(_program);
+ checkGlError("glUseProgram");
+
+ if (_textureWidth != (GLsizei) frameToRender.Width() || _textureHeight
+ != (GLsizei) frameToRender.Height())
+ {
+ SetupTextures(frameToRender);
+ }
+ else
+ {
+ UpdateTextures(frameToRender);
+ }
+
+ glDrawElements(GL_TRIANGLES, 6, GL_UNSIGNED_BYTE, g_indices);
+ checkGlError("glDrawArrays");
+
+ return 0;
+}
+
+GLuint VideoRenderOpenGles20::loadShader(GLenum shaderType,
+ const char* pSource)
+{
+ GLuint shader = glCreateShader(shaderType);
+ if (shader)
+ {
+ glShaderSource(shader, 1, &pSource, NULL);
+ glCompileShader(shader);
+ GLint compiled = 0;
+ glGetShaderiv(shader, GL_COMPILE_STATUS, &compiled);
+ if (!compiled)
+ {
+ GLint infoLen = 0;
+ glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &infoLen);
+ if (infoLen)
+ {
+ char* buf = (char*) malloc(infoLen);
+ if (buf)
+ {
+ glGetShaderInfoLog(shader, infoLen, NULL, buf);
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not compile shader %d: %s",
+ __FUNCTION__, shaderType, buf);
+ free(buf);
+ }
+ glDeleteShader(shader);
+ shader = 0;
+ }
+ }
+ }
+ return shader;
+}
+
+GLuint VideoRenderOpenGles20::createProgram(const char* pVertexSource,
+ const char* pFragmentSource)
+{
+ GLuint vertexShader = loadShader(GL_VERTEX_SHADER, pVertexSource);
+ if (!vertexShader)
+ {
+ return 0;
+ }
+
+ GLuint pixelShader = loadShader(GL_FRAGMENT_SHADER, pFragmentSource);
+ if (!pixelShader)
+ {
+ return 0;
+ }
+
+ GLuint program = glCreateProgram();
+ if (program)
+ {
+ glAttachShader(program, vertexShader);
+ checkGlError("glAttachShader");
+ glAttachShader(program, pixelShader);
+ checkGlError("glAttachShader");
+ glLinkProgram(program);
+ GLint linkStatus = GL_FALSE;
+ glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
+ if (linkStatus != GL_TRUE)
+ {
+ GLint bufLength = 0;
+ glGetProgramiv(program, GL_INFO_LOG_LENGTH, &bufLength);
+ if (bufLength)
+ {
+ char* buf = (char*) malloc(bufLength);
+ if (buf)
+ {
+ glGetProgramInfoLog(program, bufLength, NULL, buf);
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Could not link program: %s",
+ __FUNCTION__, buf);
+ free(buf);
+ }
+ }
+ glDeleteProgram(program);
+ program = 0;
+ }
+ }
+ return program;
+}
+
+void VideoRenderOpenGles20::printGLString(const char *name, GLenum s)
+{
+ const char *v = (const char *) glGetString(s);
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "GL %s = %s\n",
+ name, v);
+}
+
+void VideoRenderOpenGles20::checkGlError(const char* op)
+{
+#ifdef ANDROID_LOG
+ for (GLint error = glGetError(); error; error
+ = glGetError())
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "after %s() glError (0x%x)\n", op, error);
+ }
+#else
+ return;
+#endif
+}
+
+void VideoRenderOpenGles20::SetupTextures(const VideoFrame& frameToRender)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
+ "%s: width %d, height %d length %u", __FUNCTION__,
+ frameToRender.Width(), frameToRender.Height(),
+ frameToRender.Length());
+
+ const GLsizei width = frameToRender.Width();
+ const GLsizei height = frameToRender.Height();
+
+ glGenTextures(3, _textureIds); //Generate the Y, U and V texture
+ GLuint currentTextureId = _textureIds[0]; // Y
+ glActiveTexture( GL_TEXTURE0);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+
+ glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width, height, 0,
+ GL_LUMINANCE, GL_UNSIGNED_BYTE,
+ (const GLvoid*) frameToRender.Buffer());
+
+ currentTextureId = _textureIds[1]; // U
+ glActiveTexture( GL_TEXTURE1);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+
+ const WebRtc_UWord8* uComponent = frameToRender.Buffer() + width * height;
+ glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width / 2, height / 2, 0,
+ GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) uComponent);
+
+ currentTextureId = _textureIds[2]; // V
+ glActiveTexture( GL_TEXTURE2);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+
+ const WebRtc_UWord8* vComponent = uComponent + (width * height) / 4;
+ glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width / 2, height / 2, 0,
+ GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) vComponent);
+ checkGlError("SetupTextures");
+
+ _textureWidth = width;
+ _textureHeight = height;
+}
+
+void VideoRenderOpenGles20::UpdateTextures(const VideoFrame& frameToRender)
+{
+ const GLsizei width = frameToRender.Width();
+ const GLsizei height = frameToRender.Height();
+
+ GLuint currentTextureId = _textureIds[0]; // Y
+ glActiveTexture( GL_TEXTURE0);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_LUMINANCE,
+ GL_UNSIGNED_BYTE, (const GLvoid*) frameToRender.Buffer());
+
+ currentTextureId = _textureIds[1]; // U
+ glActiveTexture( GL_TEXTURE1);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+ const WebRtc_UWord8* uComponent = frameToRender.Buffer() + width * height;
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width / 2, height / 2,
+ GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) uComponent);
+
+ currentTextureId = _textureIds[2]; // V
+ glActiveTexture( GL_TEXTURE2);
+ glBindTexture(GL_TEXTURE_2D, currentTextureId);
+ const WebRtc_UWord8* vComponent = uComponent + (width * height) / 4;
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width / 2, height / 2,
+ GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) vComponent);
+ checkGlError("UpdateTextures");
+
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h
new file mode 100644
index 0000000..379b1e7
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
+
+#include "video_render_defines.h"
+
+#include <GLES2/gl2.h>
+#include <GLES2/gl2ext.h>
+
+namespace webrtc
+{
+
+class VideoRenderOpenGles20
+{
+public:
+ VideoRenderOpenGles20(WebRtc_Word32 id);
+ ~VideoRenderOpenGles20();
+
+ WebRtc_Word32 Setup(WebRtc_Word32 widht, WebRtc_Word32 height);
+ WebRtc_Word32 Render(const VideoFrame& frameToRender);
+ WebRtc_Word32 SetCoordinates(WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+private:
+ void printGLString(const char *name, GLenum s);
+ void checkGlError(const char* op);
+ GLuint loadShader(GLenum shaderType, const char* pSource);
+ GLuint createProgram(const char* pVertexSource, const char* pFragmentSource);
+ void SetupTextures(const VideoFrame& frameToRender);
+ void UpdateTextures(const VideoFrame& frameToRender);
+
+ WebRtc_Word32 _id;
+ GLuint _textureIds[3]; // Texture id of Y,U and V texture.
+ GLuint _program;
+ GLuint _vPositionHandle;
+ GLsizei _textureWidth;
+ GLsizei _textureHeight;
+
+ GLfloat _vertices[20];
+ static const char g_indices[];
+
+ static const char g_vertextShader[];
+ static const char g_fragmentShader[];
+
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
diff --git a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc
new file mode 100644
index 0000000..7abb09a
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_external_impl.h"
+
+namespace webrtc {
+
+VideoRenderExternalImpl::VideoRenderExternalImpl(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+ _id(id), _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _window(window), _fullscreen(fullscreen)
+{
+}
+
+VideoRenderExternalImpl::~VideoRenderExternalImpl()
+{
+ delete &_critSect;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::Init()
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_critSect);
+ _id = id;
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::ChangeWindow(void* window)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+VideoRenderCallback*
+VideoRenderExternalImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ return this;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::DeleteIncomingRenderStream(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::GetIncomingRenderStreamProperties(
+ const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ CriticalSectionScoped cs(_critSect);
+
+ zOrder = 0;
+ left = 0;
+ top = 0;
+ right = 0;
+ bottom = 0;
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::StartRender()
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::StopRender()
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+VideoRenderType VideoRenderExternalImpl::RenderType()
+{
+ return kRenderExternal;
+}
+
+RawVideoType VideoRenderExternalImpl::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool VideoRenderExternalImpl::FullScreen()
+{
+ CriticalSectionScoped cs(_critSect);
+ return _fullscreen;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::GetGraphicsMemory(
+ WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const
+{
+ totalGraphicsMemory = 0;
+ availableGraphicsMemory = 0;
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::GetScreenResolution(
+ WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const
+{
+ CriticalSectionScoped cs(_critSect);
+ screenWidth = 0;
+ screenHeight = 0;
+ return 0;
+}
+
+WebRtc_UWord32 VideoRenderExternalImpl::RenderFrameRate(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::SetStreamCropping(
+ const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::ConfigureRenderer(
+ const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::SetTransparentBackground(
+ const bool enable)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::SetText(
+ const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderExternalImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ return 0;
+}
+
+// VideoRenderCallback
+WebRtc_Word32 VideoRenderExternalImpl::RenderFrame(
+ const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+{
+ return 0;
+}
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h
new file mode 100644
index 0000000..e1374f4
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
+
+#include "i_video_render.h"
+#include "critical_section_wrapper.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+
+// Class definitions
+class VideoRenderExternalImpl: IVideoRender, public VideoRenderCallback
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ VideoRenderExternalImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window, const bool fullscreen);
+
+ virtual ~VideoRenderExternalImpl();
+
+ virtual WebRtc_Word32 Init();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32
+ GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+ // VideoRenderCallback
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _critSect;
+ void* _window;
+ bool _fullscreen;
+};
+
+} //namespace webrtc
+
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/i_video_render.h b/trunk/src/modules/video_render/main/source/i_video_render.h
new file mode 100644
index 0000000..2799a79
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/i_video_render.h
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
+
+#include "video_render.h"
+
+namespace webrtc {
+
+// Class definitions
+class IVideoRender
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ virtual ~IVideoRender()
+ {
+ };
+
+ virtual WebRtc_Word32 Init() = 0;
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
+
+ virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId) = 0;
+
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const = 0;
+ // Implemented in common code?
+ //virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const = 0;
+ //virtual bool HasIncomingRenderStream(const WebRtc_UWord16 stramId) const = 0;
+
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender() = 0;
+
+ virtual WebRtc_Word32 StopRender() = 0;
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+ virtual VideoRenderType RenderType() = 0;
+
+ virtual RawVideoType PerferedVideoType() = 0;
+
+ virtual bool FullScreen() = 0;
+
+ // TODO: This should be treated in platform specific code only
+ virtual WebRtc_Word32
+ GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const = 0;
+
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const = 0;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId) = 0;
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float rigth,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+};
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/source/incoming_video_stream.cc b/trunk/src/modules/video_render/main/source/incoming_video_stream.cc
new file mode 100644
index 0000000..418a325
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/incoming_video_stream.cc
@@ -0,0 +1,413 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "incoming_video_stream.h"
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "video_render_frames.h"
+#include "tick_util.h"
+#include "map_wrapper.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+#include <cassert>
+
+// Platform specifics
+#if defined(_WIN32)
+#include <windows.h>
+#elif defined(WEBRTC_LINUX)
+#include <ctime>
+#include <sys/time.h>
+#else
+#include <sys/time.h>
+#endif
+
+namespace webrtc {
+IncomingVideoStream::IncomingVideoStream(const WebRtc_Word32 moduleId,
+ const WebRtc_UWord32 streamId) :
+ _moduleId(moduleId),
+ _streamId(streamId),
+ _streamCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _threadCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _bufferCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrIncomingRenderThread(),
+ _deliverBufferEvent(*EventWrapper::Create()),
+ _running(false),
+ _ptrExternalCallback(NULL),
+ _ptrRenderCallback(NULL),
+ _renderBuffers(*(new VideoRenderFrames)),
+ _callbackVideoType(kVideoI420),
+ _callbackWidth(0),
+ _callbackHeight(0),
+ _incomingRate(0),
+ _lastRateCalculationTimeMs(0),
+ _numFramesSinceLastCalculation(0),
+ _lastRenderedFrame(),
+ _tempFrame(),
+ _startImage(),
+ _timeoutImage(),
+ _timeoutTime(),
+ _mirrorFramesEnabled(false),
+ _mirroring(),
+ _transformedVideoFrame()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, _moduleId,
+ "%s created for stream %d", __FUNCTION__, streamId);
+}
+
+IncomingVideoStream::~IncomingVideoStream()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, _moduleId,
+ "%s deleted for stream %d", __FUNCTION__, _streamId);
+
+ Stop();
+
+ // _ptrIncomingRenderThread - Delete in stop
+ delete &_renderBuffers;
+ delete &_streamCritsect;
+ delete &_bufferCritsect;
+ delete &_threadCritsect;
+ delete &_deliverBufferEvent;
+
+}
+
+WebRtc_Word32 IncomingVideoStream::ChangeModuleId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_streamCritsect);
+
+ _moduleId = id;
+ return 0;
+}
+
+VideoRenderCallback*
+IncomingVideoStream::ModuleCallback()
+{
+ CriticalSectionScoped cs(_streamCritsect);
+ return this;
+}
+
+WebRtc_Word32 IncomingVideoStream::RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+{
+
+ CriticalSectionScoped csS(_streamCritsect);
+ WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
+ "%s for stream %d, render time: %u", __FUNCTION__, _streamId,
+ videoFrame.RenderTimeMs());
+
+ if (!_running)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
+ "%s: Not running", __FUNCTION__);
+ return -1;
+ }
+
+ if (true == _mirrorFramesEnabled)
+ {
+ _transformedVideoFrame.VerifyAndAllocate(videoFrame.Length());
+ if (_mirroring.mirrorXAxis)
+ {
+ MirrorI420UpDown(videoFrame.Buffer(),
+ _transformedVideoFrame.Buffer(),
+ videoFrame.Width(), videoFrame.Height());
+ _transformedVideoFrame.SetLength(videoFrame.Length());
+ _transformedVideoFrame.SetWidth(videoFrame.Width());
+ _transformedVideoFrame.SetHeight(videoFrame.Height());
+ videoFrame.SwapFrame(_transformedVideoFrame);
+ }
+ if (_mirroring.mirrorYAxis)
+ {
+ MirrorI420LeftRight(videoFrame.Buffer(),
+ _transformedVideoFrame.Buffer(),
+ videoFrame.Width(), videoFrame.Height());
+ _transformedVideoFrame.SetLength(videoFrame.Length());
+ _transformedVideoFrame.SetWidth(videoFrame.Width());
+ _transformedVideoFrame.SetHeight(videoFrame.Height());
+ videoFrame.SwapFrame(_transformedVideoFrame);
+ }
+ }
+
+ // Rate statistics
+ _numFramesSinceLastCalculation++;
+ WebRtc_Word64 nowMs = TickTime::MillisecondTimestamp();
+ if (nowMs >= _lastRateCalculationTimeMs + KFrameRatePeriodMs)
+ {
+ _incomingRate = (WebRtc_UWord32) (1000 * _numFramesSinceLastCalculation
+ / (nowMs - _lastRateCalculationTimeMs));
+ _numFramesSinceLastCalculation = 0;
+ _lastRateCalculationTimeMs = nowMs;
+ }
+
+ // Insert frame
+ CriticalSectionScoped csB(_bufferCritsect);
+ if (_renderBuffers.AddFrame(&videoFrame) == 1)
+ _deliverBufferEvent.Set();
+
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::SetStartImage(const VideoFrame& videoFrame)
+{
+ CriticalSectionScoped csS(_threadCritsect);
+ return _startImage.CopyFrame(videoFrame);
+}
+
+WebRtc_Word32 IncomingVideoStream::SetTimeoutImage(const VideoFrame& videoFrame,
+ const WebRtc_UWord32 timeout)
+{
+ CriticalSectionScoped csS(_threadCritsect);
+ _timeoutTime = timeout;
+ return _timeoutImage.CopyFrame(videoFrame);
+}
+
+WebRtc_Word32 IncomingVideoStream::SetRenderCallback(VideoRenderCallback* renderCallback)
+{
+ CriticalSectionScoped cs(_streamCritsect);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
+ "%s(%x) for stream %d", __FUNCTION__, renderCallback,
+ _streamId);
+ _ptrRenderCallback = renderCallback;
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::EnableMirroring(const bool enable,
+ const bool mirrorXAxis,
+ const bool mirrorYAxis)
+{
+ CriticalSectionScoped cs(_streamCritsect);
+ _mirrorFramesEnabled = enable;
+ _mirroring.mirrorXAxis = mirrorXAxis;
+ _mirroring.mirrorYAxis = mirrorYAxis;
+
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::SetExternalCallback(VideoRenderCallback* externalCallback)
+{
+ CriticalSectionScoped cs(_streamCritsect);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
+ "%s(%x) for stream %d", __FUNCTION__, externalCallback,
+ _streamId);
+ _ptrExternalCallback = externalCallback;
+ _callbackVideoType = kVideoI420;
+ _callbackWidth = 0;
+ _callbackHeight = 0;
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::Start()
+{
+ CriticalSectionScoped csS(_streamCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
+ "%s for stream %d", __FUNCTION__, _streamId);
+ if (_running)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
+ "%s: Already running", __FUNCTION__);
+ return 0;
+ }
+
+ CriticalSectionScoped csT(_threadCritsect);
+ assert(_ptrIncomingRenderThread == NULL);
+
+ _ptrIncomingRenderThread
+ = ThreadWrapper::CreateThread(IncomingVideoStreamThreadFun, this,
+ kRealtimePriority,
+ "IncomingVideoStreamThread");
+ if (!_ptrIncomingRenderThread)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _moduleId,
+ "%s: No thread", __FUNCTION__);
+ return -1;
+ }
+
+ unsigned int tId = 0;
+ if (_ptrIncomingRenderThread->Start(tId))
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
+ "%s: thread started: %u", __FUNCTION__, tId);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _moduleId,
+ "%s: Could not start send thread", __FUNCTION__);
+ return -1;
+ }
+ _deliverBufferEvent.StartTimer(false, KEventStartupTimeMS);
+
+ _running = true;
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::Stop()
+{
+ CriticalSectionScoped csStream(_streamCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
+ "%s for stream %d", __FUNCTION__, _streamId);
+
+ if (!_running)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
+ "%s: Not running", __FUNCTION__);
+ return 0;
+ }
+
+ _threadCritsect.Enter();
+ if (_ptrIncomingRenderThread)
+ {
+ ThreadWrapper* ptrThread = _ptrIncomingRenderThread;
+ _ptrIncomingRenderThread = NULL;
+ ptrThread->SetNotAlive();
+#ifndef _WIN32
+ _deliverBufferEvent.StopTimer();
+#endif
+ _threadCritsect.Leave();
+ if (ptrThread->Stop())
+ {
+ delete ptrThread;
+ }
+ else
+ {
+ assert(false);
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
+ "%s: Not able to stop thread, leaking", __FUNCTION__);
+ }
+ }
+ else
+ {
+ _threadCritsect.Leave();
+ }
+ _running = false;
+ return 0;
+}
+
+WebRtc_Word32 IncomingVideoStream::Reset()
+{
+ CriticalSectionScoped csStream(_streamCritsect);
+ CriticalSectionScoped csBuffer(_bufferCritsect);
+
+ _renderBuffers.ReleaseAllFrames();
+ return 0;
+}
+
+WebRtc_UWord32 IncomingVideoStream::StreamId() const
+{
+ CriticalSectionScoped csStream(_streamCritsect);
+ return _streamId;
+}
+
+WebRtc_UWord32 IncomingVideoStream::IncomingRate() const
+{
+ CriticalSectionScoped cs(_streamCritsect);
+ return _incomingRate;
+}
+
+bool IncomingVideoStream::IncomingVideoStreamThreadFun(void* obj)
+{
+ return static_cast<IncomingVideoStream*> (obj)->IncomingVideoStreamProcess();
+}
+
+bool IncomingVideoStream::IncomingVideoStreamProcess()
+{
+ if (kEventError != _deliverBufferEvent.Wait(KEventMaxWaitTimeMs))
+ {
+ if (_ptrIncomingRenderThread == NULL)
+ {
+ // Terminating
+ return false;
+ }
+
+ _threadCritsect.Enter();
+
+ VideoFrame* ptrFrameToRender = NULL;
+
+ // Get a new frame to render and the time for the frame after this one.
+ _bufferCritsect.Enter();
+ ptrFrameToRender = _renderBuffers.FrameToRender();
+ WebRtc_UWord32 waitTime = _renderBuffers.TimeToNextFrameRelease();
+ _bufferCritsect.Leave();
+
+ // Set timer for next frame to render
+ if (waitTime > KEventMaxWaitTimeMs)
+ {
+ waitTime = KEventMaxWaitTimeMs;
+ }
+ _deliverBufferEvent.StartTimer(false, waitTime);
+
+ if (!ptrFrameToRender)
+ {
+ if (_ptrRenderCallback)
+ {
+ if (_lastRenderedFrame.RenderTimeMs() == 0
+ && _startImage.Size()) // And we have not rendered anything and have a start image
+ {
+ _tempFrame.CopyFrame(_startImage);// Copy the startimage if the renderer modifies the render buffer.
+ _ptrRenderCallback->RenderFrame(_streamId, _tempFrame);
+ }
+ else if (_timeoutImage.Size()
+ && _lastRenderedFrame.RenderTimeMs() + _timeoutTime
+ < TickTime::MillisecondTimestamp()) // We have rendered something a long time ago and have a timeout image
+ {
+ _tempFrame.CopyFrame(_timeoutImage); // Copy the timeoutImage if the renderer modifies the render buffer.
+ _ptrRenderCallback->RenderFrame(_streamId, _tempFrame);
+ }
+ }
+
+ // No frame
+ _threadCritsect.Leave();
+ return true;
+ }
+
+ // Send frame for rendering
+ if (_ptrExternalCallback)
+ {
+ WEBRTC_TRACE(kTraceStream,
+ kTraceVideoRenderer,
+ _moduleId,
+ "%s: executing external renderer callback to deliver frame",
+ __FUNCTION__, ptrFrameToRender->RenderTimeMs());
+ _ptrExternalCallback->RenderFrame(_streamId, *ptrFrameToRender);
+ }
+ else
+ {
+ if (_ptrRenderCallback)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
+ "%s: Render frame, time: ", __FUNCTION__,
+ ptrFrameToRender->RenderTimeMs());
+ _ptrRenderCallback->RenderFrame(_streamId, *ptrFrameToRender);
+ }
+ }
+
+ // Release critsect before calling the module user
+ _threadCritsect.Leave();
+
+ // We're done with this frame, delete it.
+ if (ptrFrameToRender)
+ {
+ CriticalSectionScoped cs(_bufferCritsect);
+ _lastRenderedFrame.SwapFrame(*ptrFrameToRender);
+ _renderBuffers.ReturnFrame(ptrFrameToRender);
+ }
+ }
+ return true;
+}
+WebRtc_Word32 IncomingVideoStream::GetLastRenderedFrame(VideoFrame& videoFrame) const
+{
+ CriticalSectionScoped cs(_bufferCritsect);
+ return videoFrame.CopyFrame(_lastRenderedFrame);
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/incoming_video_stream.h b/trunk/src/modules/video_render/main/source/incoming_video_stream.h
new file mode 100644
index 0000000..cd2785d
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/incoming_video_stream.h
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
+
+#include "video_render.h"
+#include "map_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class VideoRenderCallback;
+class VideoRenderFrames;
+
+struct VideoMirroring
+{
+ bool mirrorXAxis;
+ bool mirrorYAxis;
+ VideoMirroring() :
+ mirrorXAxis(false), mirrorYAxis(false)
+ {
+ }
+};
+
+// Class definitions
+class IncomingVideoStream: public VideoRenderCallback
+{
+public:
+ /*
+ * VideoRenderer constructor/destructor
+ */
+ IncomingVideoStream(const WebRtc_Word32 moduleId,
+ const WebRtc_UWord32 streamId);
+ ~IncomingVideoStream();
+
+ WebRtc_Word32 ChangeModuleId(const WebRtc_Word32 id);
+
+ // Get callbck to deliver frames to the module
+ VideoRenderCallback* ModuleCallback();
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+ // Set callback to the platform dependant code
+ WebRtc_Word32 SetRenderCallback(VideoRenderCallback* renderCallback);
+
+ // Callback for file recording, snapshot, ...
+ WebRtc_Word32 SetExternalCallback(VideoRenderCallback* renderObject);
+
+ /*
+ * Start/Stop
+ */
+ WebRtc_Word32 Start();
+ WebRtc_Word32 Stop();
+
+ // Clear all buffers
+ WebRtc_Word32 Reset();
+
+ /*
+ * Properties
+ */
+ WebRtc_UWord32 StreamId() const;
+ WebRtc_UWord32 IncomingRate() const;
+
+ /*
+ *
+ */
+ WebRtc_Word32 GetLastRenderedFrame(VideoFrame& videoFrame) const;
+
+ WebRtc_Word32 SetStartImage(const VideoFrame& videoFrame);
+
+ WebRtc_Word32 SetTimeoutImage(const VideoFrame& videoFrame,
+ const WebRtc_UWord32 timeout);
+
+ WebRtc_Word32 EnableMirroring(const bool enable,
+ const bool mirrorXAxis,
+ const bool mirrorYAxis);
+
+protected:
+ static bool IncomingVideoStreamThreadFun(void* obj);
+ bool IncomingVideoStreamProcess();
+
+private:
+
+ // Enums
+ enum
+ {
+ KEventStartupTimeMS = 10
+ };
+ enum
+ {
+ KEventMaxWaitTimeMs = 100
+ };
+ enum
+ {
+ KFrameRatePeriodMs = 1000
+ };
+
+ WebRtc_Word32 _moduleId;
+ WebRtc_UWord32 _streamId;
+ CriticalSectionWrapper& _streamCritsect; // Critsects in allowed to enter order
+ CriticalSectionWrapper& _threadCritsect;
+ CriticalSectionWrapper& _bufferCritsect;
+ ThreadWrapper* _ptrIncomingRenderThread;
+ EventWrapper& _deliverBufferEvent;
+ bool _running;
+
+ VideoRenderCallback* _ptrExternalCallback;
+ VideoRenderCallback* _ptrRenderCallback;
+ VideoRenderFrames& _renderBuffers;
+
+ RawVideoType _callbackVideoType;
+ WebRtc_UWord32 _callbackWidth;
+ WebRtc_UWord32 _callbackHeight;
+
+ WebRtc_UWord32 _incomingRate;
+ WebRtc_Word64 _lastRateCalculationTimeMs;
+ WebRtc_UWord16 _numFramesSinceLastCalculation;
+ VideoFrame _lastRenderedFrame;
+ VideoFrame _tempFrame;
+ VideoFrame _startImage;
+ VideoFrame _timeoutImage;
+ WebRtc_UWord32 _timeoutTime;
+
+ bool _mirrorFramesEnabled;
+ VideoMirroring _mirroring;
+ VideoFrame _transformedVideoFrame;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc
new file mode 100644
index 0000000..a5e311a
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_linux_impl.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "video_x11_render.h"
+
+#include <X11/Xlib.h>
+
+namespace webrtc {
+
+VideoRenderLinuxImpl::VideoRenderLinuxImpl(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window, const bool fullscreen) :
+ _id(id),
+ _renderLinuxCritsect(
+ *CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrWindow(window), _fullscreen(fullscreen), _ptrX11Render(NULL),
+ _renderType(videoRenderType)
+{
+}
+
+VideoRenderLinuxImpl::~VideoRenderLinuxImpl()
+{
+ if (_ptrX11Render)
+ delete _ptrX11Render;
+
+ delete &_renderLinuxCritsect;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::Init()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+ _ptrX11Render = new VideoX11Render((Window) _ptrWindow);
+ if (!_ptrX11Render)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s",
+ "Failed to create instance of VideoX11Render object");
+ return -1;
+ }
+ int retVal = _ptrX11Render->Init();
+ if (retVal == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+
+ _id = id;
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::ChangeWindow(void* window)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+ _ptrWindow = window;
+
+ if (_ptrX11Render)
+ {
+ return _ptrX11Render->ChangeWindow((Window) window);
+ }
+
+ return -1;
+}
+
+VideoRenderCallback* VideoRenderLinuxImpl::AddIncomingRenderStream(
+ const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+
+ VideoRenderCallback* renderCallback = NULL;
+ if (_ptrX11Render)
+ {
+ VideoX11Channel* renderChannel =
+ _ptrX11Render->CreateX11RenderChannel(streamId, zOrder, left,
+ top, right, bottom);
+ if (!renderChannel)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "Render channel creation failed for stream id: %d",
+ streamId);
+ return NULL;
+ }
+ renderCallback = (VideoRenderCallback *) renderChannel;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "_ptrX11Render is NULL");
+ return NULL;
+ }
+ return renderCallback;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::DeleteIncomingRenderStream(
+ const WebRtc_UWord32 streamId)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+
+ if (_ptrX11Render)
+ {
+ return _ptrX11Render->DeleteX11RenderChannel(streamId);
+ }
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::GetIncomingRenderStreamProperties(
+ const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_renderLinuxCritsect);
+
+ if (_ptrX11Render)
+ {
+ return _ptrX11Render->GetIncomingStreamProperties(streamId, zOrder,
+ left, top, right,
+ bottom);
+ }
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::StartRender()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::StopRender()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
+ __FUNCTION__);
+ return 0;
+}
+
+VideoRenderType VideoRenderLinuxImpl::RenderType()
+{
+ return kRenderX11;
+}
+
+RawVideoType VideoRenderLinuxImpl::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool VideoRenderLinuxImpl::FullScreen()
+{
+ return false;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::GetGraphicsMemory(
+ WebRtc_UWord64& /*totalGraphicsMemory*/,
+ WebRtc_UWord64& /*availableGraphicsMemory*/) const
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::GetScreenResolution(
+ WebRtc_UWord32& /*screenWidth*/,
+ WebRtc_UWord32& /*screenHeight*/) const
+{
+ return -1;
+}
+
+WebRtc_UWord32 VideoRenderLinuxImpl::RenderFrameRate(const WebRtc_UWord32 /*streamId*/)
+{
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::SetStreamCropping(
+ const WebRtc_UWord32 /*streamId*/,
+ const float /*left*/,
+ const float /*top*/,
+ const float /*right*/,
+ const float /*bottom*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::SetTransparentBackground(const bool /*enable*/)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::ConfigureRenderer(
+ const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::SetText(
+ const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float rigth,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderLinuxImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s - not supported on Linux", __FUNCTION__);
+ return -1;
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h
new file mode 100644
index 0000000..10460ec
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h
@@ -0,0 +1,136 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
+
+#include "i_video_render.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class VideoX11Render;
+
+// Class definitions
+class VideoRenderLinuxImpl: IVideoRender
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ VideoRenderLinuxImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window, const bool fullscreen);
+
+ virtual ~VideoRenderLinuxImpl();
+
+ virtual WebRtc_Word32 Init();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32
+ GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float rigth, const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _renderLinuxCritsect;
+
+ void* _ptrWindow;
+ bool _fullscreen;
+
+ // X11 Render
+ VideoX11Render* _ptrX11Render;
+
+ VideoRenderType _renderType;
+
+};
+
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc
new file mode 100644
index 0000000..5f8bfa1
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc
@@ -0,0 +1,332 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_x11_channel.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+
+#define DISP_MAX 128
+
+static Display *dispArray[DISP_MAX];
+static int dispCount = 0;
+
+
+VideoX11Channel::VideoX11Channel(WebRtc_Word32 id) :
+ _crit(*CriticalSectionWrapper::CreateCriticalSection()), _display(NULL),
+ _shminfo(), _image(NULL), _window(0L), _gc(NULL),
+ _width(DEFAULT_RENDER_FRAME_WIDTH),
+ _height(DEFAULT_RENDER_FRAME_HEIGHT), _outWidth(0), _outHeight(0),
+ _xPos(0), _yPos(0), _prepared(false), _dispCount(0), _buffer(NULL),
+ _top(0.0), _left(0.0), _right(0.0), _bottom(0.0),
+ _Id(id)
+{
+}
+
+VideoX11Channel::~VideoX11Channel()
+{
+ if (_prepared)
+ {
+ _crit.Enter();
+ ReleaseWindow();
+ _crit.Leave();
+ }
+ delete &_crit;
+}
+
+WebRtc_Word32 VideoX11Channel::RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+{
+ CriticalSectionScoped cs(_crit);
+ if (_width != (WebRtc_Word32) videoFrame.Width() || _height
+ != (WebRtc_Word32) videoFrame.Height())
+ {
+ if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
+ {
+ return -1;
+ }
+ }
+ return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
+ videoFrame.TimeStamp());
+}
+
+WebRtc_Word32 VideoX11Channel::FrameSizeChange(WebRtc_Word32 width,
+ WebRtc_Word32 height,
+ WebRtc_Word32 /*numberOfStreams */)
+{
+ CriticalSectionScoped cs(_crit);
+ if (_prepared)
+ {
+ RemoveRenderer();
+ }
+ if (CreateLocalRenderer(width, height) == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::DeliverFrame(unsigned char* buffer,
+ WebRtc_Word32 bufferSize,
+ unsigned WebRtc_Word32 /*timeStamp90kHz*/)
+{
+ CriticalSectionScoped cs(_crit);
+ if (!_prepared)
+ {
+ return 0;
+ }
+
+ if (!dispArray[_dispCount])
+ {
+ return -1;
+ }
+
+ unsigned char *pBuf = buffer;
+ // convert to RGB32, setting stride = width.
+ ConvertFromI420(pBuf, _width, kARGB, 0, _width, _height, _buffer);
+
+ // put image in window
+ XShmPutImage(_display, _window, _gc, _image, 0, 0, _xPos, _yPos, _width,
+ _height, True);
+
+ // very important for the image to update properly!
+ XSync(_display, False);
+ return 0;
+
+}
+
+WebRtc_Word32 VideoX11Channel::GetFrameSize(WebRtc_Word32& width,
+ WebRtc_Word32& height)
+{
+ width = _width;
+ height = _height;
+
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::Init(Window window, float left, float top,
+ float right, float bottom)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_crit);
+
+ _window = window;
+ _left = left;
+ _right = right;
+ _top = top;
+ _bottom = bottom;
+
+ _display = XOpenDisplay(NULL); // Use default display
+ if (!_window || !_display)
+ {
+ return -1;
+ }
+
+ if (dispCount < DISP_MAX)
+ {
+ dispArray[dispCount] = _display;
+ _dispCount = dispCount;
+ dispCount++;
+ }
+ else
+ {
+ return -1;
+ }
+
+ if ((1 < left || left < 0) || (1 < top || top < 0) || (1 < right || right
+ < 0) || (1 < bottom || bottom < 0))
+ {
+ return -1;
+ }
+
+ // calculate position and size of rendered video
+ int x, y;
+ unsigned int winWidth, winHeight, borderwidth, depth;
+ Window rootret;
+ if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
+ &winHeight, &borderwidth, &depth) == 0)
+ {
+ return -1;
+ }
+
+ _xPos = (WebRtc_Word32) (winWidth * left);
+ _yPos = (WebRtc_Word32) (winHeight * top);
+ _outWidth = (WebRtc_Word32) (winWidth * (right - left));
+ _outHeight = (WebRtc_Word32) (winHeight * (bottom - top));
+ if (_outWidth % 2)
+ _outWidth++; // the renderer want's sizes that are multiples of two
+ if (_outHeight % 2)
+ _outHeight++;
+
+ _gc = XCreateGC(_display, _window, 0, 0);
+ if (!_gc) {
+ // Failed to create the graphics context.
+ assert(false);
+ return -1;
+ }
+
+ if (CreateLocalRenderer(winWidth, winHeight) == -1)
+ {
+ return -1;
+ }
+ return 0;
+
+}
+
+WebRtc_Word32 VideoX11Channel::ChangeWindow(Window window)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_crit);
+
+ // Stop the rendering, if we are rendering...
+ RemoveRenderer();
+ _window = window;
+
+ // calculate position and size of rendered video
+ int x, y;
+ unsigned int winWidth, winHeight, borderwidth, depth;
+ Window rootret;
+ if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
+ &winHeight, &borderwidth, &depth) == -1)
+ {
+ return -1;
+ }
+ _xPos = (int) (winWidth * _left);
+ _yPos = (int) (winHeight * _top);
+ _outWidth = (int) (winWidth * (_right - _left));
+ _outHeight = (int) (winHeight * (_bottom - _top));
+ if (_outWidth % 2)
+ _outWidth++; // the renderer want's sizes that are multiples of two
+ if (_outHeight % 2)
+ _outHeight++;
+
+ // Prepare rendering using the
+ if (CreateLocalRenderer(_width, _height) == -1)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::ReleaseWindow()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_crit);
+
+ RemoveRenderer();
+ if (_gc) {
+ XFreeGC(_display, _gc);
+ _gc = NULL;
+ }
+ if (_display)
+ {
+ XCloseDisplay(_display);
+ _display = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::CreateLocalRenderer(WebRtc_Word32 width,
+ WebRtc_Word32 height)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(_crit);
+
+ if (!_window || !_display)
+ {
+ return -1;
+ }
+
+ if (_prepared)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _Id,
+ "Renderer already prepared, exits.");
+ return -1;
+ }
+
+ _width = width;
+ _height = height;
+
+ // create shared memory image
+ _image = XShmCreateImage(_display, CopyFromParent, 24, ZPixmap, NULL,
+ &_shminfo, _width, _height); // this parameter needs to be the same for some reason.
+ _shminfo.shmid = shmget(IPC_PRIVATE, (_image->bytes_per_line
+ * _image->height), IPC_CREAT | 0777);
+ _shminfo.shmaddr = _image->data = (char*) shmat(_shminfo.shmid, 0, 0);
+ if (_image->data == reinterpret_cast<char*>(-1))
+ {
+ return -1;
+ }
+ _buffer = (unsigned char*) _image->data;
+ _shminfo.readOnly = False;
+
+ // attach image to display
+ if (!XShmAttach(_display, &_shminfo))
+ {
+ //printf("XShmAttach failed !\n");
+ return -1;
+ }
+ XSync(_display, False);
+
+ _prepared = true;
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::RemoveRenderer()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+
+ if (!_prepared)
+ {
+ return 0;
+ }
+ _prepared = false;
+
+ // Free the memory.
+ XShmDetach(_display, &_shminfo);
+ XDestroyImage( _image );
+ _image = NULL;
+ shmdt(_shminfo.shmaddr);
+ _shminfo.shmaddr = NULL;
+ _buffer = NULL;
+ shmctl(_shminfo.shmid, IPC_RMID, 0);
+ _shminfo.shmid = 0;
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Channel::GetStreamProperties(WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right,
+ float& bottom) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
+ __FUNCTION__);
+
+ zOrder = 0; // no z-order support yet
+ left = _left;
+ top = _top;
+ right = _right;
+ bottom = _bottom;
+
+ return 0;
+}
+
+
+} //namespace webrtc
+
+
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h
new file mode 100644
index 0000000..1fb2dab
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h
@@ -0,0 +1,97 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
+
+#include "video_render_defines.h"
+#include "common_video/libyuv/include/libyuv.h"
+#include <sys/shm.h>
+
+#include <X11/Xlib.h>
+#include <X11/Xutil.h>
+#include <X11/extensions/XShm.h>
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+#define DEFAULT_RENDER_FRAME_WIDTH 352
+#define DEFAULT_RENDER_FRAME_HEIGHT 288
+
+
+class VideoX11Channel: public VideoRenderCallback
+{
+public:
+ VideoX11Channel(WebRtc_Word32 id);
+
+ virtual ~VideoX11Channel();
+
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+ WebRtc_Word32 FrameSizeChange(WebRtc_Word32 width, WebRtc_Word32 height,
+ WebRtc_Word32 numberOfStreams);
+ WebRtc_Word32 DeliverFrame(unsigned char* buffer, WebRtc_Word32 bufferSize,
+ unsigned WebRtc_Word32 /*timeStamp90kHz*/);
+ WebRtc_Word32 GetFrameSize(WebRtc_Word32& width, WebRtc_Word32& height);
+ WebRtc_Word32 Init(Window window, float left, float top, float right,
+ float bottom);
+ WebRtc_Word32 ChangeWindow(Window window);
+ WebRtc_Word32
+ GetStreamProperties(WebRtc_UWord32& zOrder, float& left,
+ float& top, float& right, float& bottom) const;
+ WebRtc_Word32 ReleaseWindow();
+
+ bool IsPrepared()
+ {
+ return _prepared;
+ }
+
+private:
+
+ WebRtc_Word32
+ CreateLocalRenderer(WebRtc_Word32 width, WebRtc_Word32 height);
+ WebRtc_Word32 RemoveRenderer();
+
+ //FIXME a better place for this method? the GetWidthHeight no longer
+ // supported by common_video.
+ int GetWidthHeight(VideoType type, int bufferSize, int& width,
+ int& height);
+
+ CriticalSectionWrapper& _crit;
+
+ Display* _display;
+ XShmSegmentInfo _shminfo;
+ XImage* _image;
+ Window _window;
+ GC _gc;
+ WebRtc_Word32 _width; // incoming frame width
+ WebRtc_Word32 _height; // incoming frame height
+ WebRtc_Word32 _outWidth; // render frame width
+ WebRtc_Word32 _outHeight; // render frame height
+ WebRtc_Word32 _xPos; // position within window
+ WebRtc_Word32 _yPos;
+ bool _prepared; // true if ready to use
+ WebRtc_Word32 _dispCount;
+
+ unsigned char* _buffer;
+ float _top;
+ float _left;
+ float _right;
+ float _bottom;
+
+ WebRtc_Word32 _Id;
+
+};
+
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc b/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc
new file mode 100644
index 0000000..a3543e3
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc
@@ -0,0 +1,154 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_x11_render.h"
+#include "video_x11_channel.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+
+VideoX11Render::VideoX11Render(Window window) :
+ _window(window),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection())
+{
+}
+
+VideoX11Render::~VideoX11Render()
+{
+ delete &_critSect;
+}
+
+WebRtc_Word32 VideoX11Render::Init()
+{
+ CriticalSectionScoped cs(_critSect);
+
+ _streamIdToX11ChannelMap.clear();
+
+ return 0;
+}
+
+WebRtc_Word32 VideoX11Render::ChangeWindow(Window window)
+{
+ CriticalSectionScoped cs(_critSect);
+ VideoX11Channel* renderChannel = NULL;
+
+ std::map<int, VideoX11Channel*>::iterator iter =
+ _streamIdToX11ChannelMap.begin();
+
+ while (iter != _streamIdToX11ChannelMap.end())
+ {
+ renderChannel = iter->second;
+ if (renderChannel)
+ {
+ renderChannel->ChangeWindow(window);
+ }
+ iter++;
+ }
+
+ _window = window;
+
+ return 0;
+}
+
+VideoX11Channel* VideoX11Render::CreateX11RenderChannel(
+ WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+ VideoX11Channel* renderChannel = NULL;
+
+ std::map<int, VideoX11Channel*>::iterator iter =
+ _streamIdToX11ChannelMap.find(streamId);
+
+ if (iter == _streamIdToX11ChannelMap.end())
+ {
+ renderChannel = new VideoX11Channel(streamId);
+ if (!renderChannel)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideoRenderer,
+ -1,
+ "Failed to create VideoX11Channel for streamId : %d",
+ streamId);
+ return NULL;
+ }
+ renderChannel->Init(_window, left, top, right, bottom);
+ _streamIdToX11ChannelMap[streamId] = renderChannel;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
+ "Render Channel already exists for streamId: %d", streamId);
+ renderChannel = iter->second;
+ }
+
+ return renderChannel;
+}
+
+WebRtc_Word32 VideoX11Render::DeleteX11RenderChannel(WebRtc_Word32 streamId)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ std::map<int, VideoX11Channel*>::iterator iter =
+ _streamIdToX11ChannelMap.find(streamId);
+ if (iter != _streamIdToX11ChannelMap.end())
+ {
+ VideoX11Channel *renderChannel = iter->second;
+ if (renderChannel)
+ {
+ renderChannel->ReleaseWindow();
+ delete renderChannel;
+ renderChannel = NULL;
+ }
+ _streamIdToX11ChannelMap.erase(iter);
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "No VideoX11Channel object exists for stream id: %d",
+ streamId);
+ return -1;
+}
+
+WebRtc_Word32 VideoX11Render::GetIncomingStreamProperties(
+ WebRtc_Word32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ std::map<int, VideoX11Channel*>::iterator iter =
+ _streamIdToX11ChannelMap.find(streamId);
+ if (iter != _streamIdToX11ChannelMap.end())
+ {
+ VideoX11Channel *renderChannel = iter->second;
+ if (renderChannel)
+ {
+ renderChannel->GetStreamProperties(zOrder, left, top, right, bottom);
+ }
+ }
+
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "No VideoX11Channel object exists for stream id: %d",
+ streamId);
+ return -1;
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_render.h b/trunk/src/modules/video_render/main/source/linux/video_x11_render.h
new file mode 100644
index 0000000..9b140ef
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/linux/video_x11_render.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
+
+#include "video_render_defines.h"
+
+#include <X11/Xlib.h>
+#include <map>
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class VideoX11Channel;
+
+class VideoX11Render
+{
+
+public:
+ VideoX11Render(Window window);
+ ~VideoX11Render();
+
+ WebRtc_Word32 Init();
+ WebRtc_Word32 ChangeWindow(Window window);
+
+ VideoX11Channel* CreateX11RenderChannel(WebRtc_Word32 streamId,
+ WebRtc_Word32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ WebRtc_Word32 DeleteX11RenderChannel(WebRtc_Word32 streamId);
+
+ WebRtc_Word32 GetIncomingStreamProperties(WebRtc_Word32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom);
+
+private:
+ Window _window;
+ CriticalSectionWrapper& _critSect;
+ std::map<int, VideoX11Channel*> _streamIdToX11ChannelMap;
+
+};
+
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h
new file mode 100644
index 0000000..c8e98bb
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// cocoa_full_screen_window.h
+//
+//
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
+
+#import <Cocoa/Cocoa.h>
+//#define GRAB_ALL_SCREENS 1
+
+@interface CocoaFullScreenWindow : NSObject {
+ NSWindow* _window;
+}
+
+-(id)init;
+-(void)grabFullScreen;
+-(void)releaseFullScreen;
+-(NSWindow*)window;
+
+@end
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm
new file mode 100644
index 0000000..e86bab1
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "cocoa_full_screen_window.h"
+#include "trace.h"
+
+using namespace webrtc;
+
+@implementation CocoaFullScreenWindow
+
+-(id)init{
+
+ self = [super init];
+ if(!self){
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d COULD NOT CREATE INSTANCE", __FUNCTION__, __LINE__);
+ return nil;
+ }
+
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, 0, "%s:%d Created instance", __FUNCTION__, __LINE__);
+ return self;
+}
+
+-(void)grabFullScreen{
+
+#ifdef GRAB_ALL_SCREENS
+ if(CGCaptureAllDisplays() != kCGErrorSuccess)
+#else
+ if(CGDisplayCapture(kCGDirectMainDisplay) != kCGErrorSuccess)
+#endif
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not capture main level", __FUNCTION__, __LINE__);
+ }
+
+ // get the shielding window level
+ int windowLevel = CGShieldingWindowLevel();
+
+ // get the screen rect of main display
+ NSRect screenRect = [[NSScreen mainScreen]frame];
+
+ _window = [[NSWindow alloc]initWithContentRect:screenRect
+ styleMask:NSBorderlessWindowMask
+ backing:NSBackingStoreBuffered
+ defer:NO
+ screen:[NSScreen mainScreen]];
+
+ [_window setLevel:windowLevel];
+ [_window setBackgroundColor:[NSColor blackColor]];
+ [_window makeKeyAndOrderFront:nil];
+
+}
+
+-(void)releaseFullScreen
+{
+ [_window orderOut:self];
+
+#ifdef GRAB_ALL_SCREENS
+ if(CGReleaseAllDisplays() != kCGErrorSuccess)
+#else
+ if(CGDisplayRelease(kCGDirectMainDisplay) != kCGErrorSuccess)
+#endif
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not release the displays", __FUNCTION__, __LINE__);
+ }
+}
+
+- (NSWindow*)window
+{
+ return _window;
+}
+
+- (void) dealloc
+{
+ [self releaseFullScreen];
+ [super dealloc];
+}
+
+
+
+@end
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h
new file mode 100644
index 0000000..15a8108
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// cocoa_render_view.h
+//
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
+
+#import <Cocoa/Cocoa.h>
+#import <OpenGL/gl.h>
+#import <OpenGL/glu.h>
+#import <OpenGL/OpenGL.h>
+
+@interface CocoaRenderView : NSOpenGLView {
+ NSOpenGLContext* _nsOpenGLContext;
+}
+
+-(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt;
+-(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt;
+-(NSOpenGLContext*)nsOpenGLContext;
+@end
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm
new file mode 100644
index 0000000..567d171
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Cocoa/Cocoa.h>
+#import <AppKit/AppKit.h>
+#import "cocoa_render_view.h"
+#include "trace.h"
+
+using namespace webrtc;
+
+@implementation CocoaRenderView
+
+-(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt{
+
+ self = [super initWithFrame:[self frame] pixelFormat:[fmt autorelease]];
+ if (self == nil){
+
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__);
+ }
+
+
+ _nsOpenGLContext = [self openGLContext];
+
+}
+
+-(NSOpenGLContext*)nsOpenGLContext {
+ return _nsOpenGLContext;
+}
+
+-(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt{
+
+ NSRect screenRect = [[NSScreen mainScreen]frame];
+// [_windowRef setFrame:screenRect];
+// [_windowRef setBounds:screenRect];
+ self = [super initWithFrame:screenRect pixelFormat:[fmt autorelease]];
+ if (self == nil){
+
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__);
+ }
+
+ _nsOpenGLContext = [self openGLContext];
+
+}
+
+@end
+
+
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc b/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc
new file mode 100644
index 0000000..e968865
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc
@@ -0,0 +1,2007 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(CARBON_RENDERING)
+
+#include "video_render_agl.h"
+
+// includes
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc {
+
+/*
+ *
+ * VideoChannelAGL
+ *
+ */
+
+#pragma mark VideoChannelAGL constructor
+
+VideoChannelAGL::VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner) :
+ _aglContext( aglContext),
+ _id( iId),
+ _owner( owner),
+ _width( 0),
+ _height( 0),
+ _stretchedWidth( 0),
+ _stretchedHeight( 0),
+ _startWidth( 0.0f),
+ _startHeight( 0.0f),
+ _stopWidth( 0.0f),
+ _stopHeight( 0.0f),
+ _xOldWidth( 0),
+ _yOldHeight( 0),
+ _oldStretchedHeight(0),
+ _oldStretchedWidth( 0),
+ _buffer( 0),
+ _bufferSize( 0),
+ _incommingBufferSize(0),
+ _bufferIsUpdated( false),
+ _sizeInitialized( false),
+ _numberOfStreams( 0),
+ _bVideoSizeStartedChanging(false),
+ _pixelFormat( GL_RGBA),
+ _pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
+ _texture( 0)
+
+{
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Constructor", __FUNCTION__, __LINE__);
+}
+
+VideoChannelAGL::~VideoChannelAGL()
+{
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Destructor", __FUNCTION__, __LINE__);
+ if (_buffer)
+ {
+ delete [] _buffer;
+ _buffer = NULL;
+ }
+
+ aglSetCurrentContext(_aglContext);
+
+ if (_texture != 0)
+ {
+ glDeleteTextures(1, (const GLuint*) &_texture);
+ _texture = 0;
+ }
+}
+
+WebRtc_Word32 VideoChannelAGL::RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame)
+{
+ _owner->LockAGLCntx();
+ if(_width != videoFrame.Width() ||
+ _height != videoFrame.Height())
+ {
+ if(FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
+ { //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d FrameSizeChange returned an error", __FUNCTION__, __LINE__);
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+ }
+
+ _owner->UnlockAGLCntx();
+ return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(), videoFrame.TimeStamp());
+}
+
+int VideoChannelAGL::UpdateSize(int /*width*/, int /*height*/)
+{
+ _owner->LockAGLCntx();
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelAGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
+{
+
+ _owner->LockAGLCntx();
+ _stretchedHeight = stretchHeight;
+ _stretchedWidth = stretchWidth;
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelAGL::FrameSizeChange(int width, int height, int numberOfStreams)
+{
+ // We'll get a new frame size from VideoAPI, prepare the buffer
+
+ _owner->LockAGLCntx();
+
+ if (width == _width && _height == height)
+ {
+ // We already have a correct buffer size
+ _numberOfStreams = numberOfStreams;
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ _width = width;
+ _height = height;
+
+ // Delete the old buffer, create a new one with correct size.
+ if (_buffer)
+ {
+ delete [] _buffer;
+ _bufferSize = 0;
+ }
+
+ _incommingBufferSize = CalcBufferSize(kI420, _width, _height);
+ _bufferSize = CalcBufferSize(kARGB, _width, _height);//_width * _height * bytesPerPixel;
+ _buffer = new unsigned char [_bufferSize];
+ memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
+
+ if (aglSetCurrentContext(_aglContext) == false)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ // Delete a possible old texture
+ if (_texture != 0)
+ {
+ glDeleteTextures(1, (const GLuint*) &_texture);
+ _texture = 0;
+ }
+
+ // Create a new texture
+ glGenTextures(1, (GLuint *) &_texture);
+
+ GLenum glErr = glGetError();
+
+ if (glErr != GL_NO_ERROR)
+ {
+ }
+
+ // Do the setup for both textures
+ // Note: we setup two textures even if we're not running full screen
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
+
+ // Set texture parameters
+ glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
+
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+ //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
+ //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+
+ glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
+
+ glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
+
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
+
+ // Maximum width/height for a texture
+ GLint texSize;
+ glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
+
+ if (texSize < _width || texSize < _height)
+ {
+ // Image too big for memory
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ // Set up th texture type and size
+ glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
+ 0, // level
+ GL_RGBA, // internal format
+ _width, // width
+ _height, // height
+ 0, // border 0/1 = off/on
+ _pixelFormat, // format, GL_BGRA
+ _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
+ _buffer); // pixel data
+
+ glErr = glGetError();
+ if (glErr != GL_NO_ERROR)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+// Called from video engine when a new frame should be rendered.
+int VideoChannelAGL::DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int /*timeStamp90kHz*/)
+{
+ _owner->LockAGLCntx();
+
+ if (_texture == 0)
+ {
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ if (bufferSize != _incommingBufferSize)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ // Setting stride = width.
+ int rgbret = ConvertFromYV12(buffer, _width, kBGRA, 0, _width, _height,
+ _buffer);
+ if (rgbret < 0)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ aglSetCurrentContext(_aglContext);
+
+ // Put the new frame into the graphic card texture.
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture); // Make sure this texture is the active one
+ GLenum glErr = glGetError();
+ if (glErr != GL_NO_ERROR)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ // Copy buffer to texture
+ glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
+ 0, // Level, not use
+ 0, // start point x, (low left of pic)
+ 0, // start point y,
+ _width, // width
+ _height, // height
+ _pixelFormat, // pictue format for _buffer
+ _pixelDataType, // data type of _buffer
+ (const GLvoid*) _buffer); // the pixel data
+
+ if (glGetError() != GL_NO_ERROR)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ _bufferIsUpdated = true;
+ _owner->UnlockAGLCntx();
+
+ return 0;
+}
+
+int VideoChannelAGL::RenderOffScreenBuffer()
+{
+
+ _owner->LockAGLCntx();
+
+ if (_texture == 0)
+ {
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ GLfloat xStart = 2.0f * _startWidth - 1.0f;
+ GLfloat xStop = 2.0f * _stopWidth - 1.0f;
+ GLfloat yStart = 1.0f - 2.0f * _stopHeight;
+ GLfloat yStop = 1.0f - 2.0f * _startHeight;
+
+ aglSetCurrentContext(_aglContext);
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
+
+ if(_stretchedWidth != _oldStretchedWidth || _stretchedHeight != _oldStretchedHeight)
+ {
+ glViewport(0, 0, _stretchedWidth, _stretchedHeight);
+ }
+ _oldStretchedHeight = _stretchedHeight;
+ _oldStretchedWidth = _stretchedWidth;
+
+ // Now really put the texture into the framebuffer
+ glLoadIdentity();
+
+ glEnable(GL_TEXTURE_RECTANGLE_EXT);
+
+ glBegin(GL_POLYGON);
+ {
+ glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
+ glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
+ glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
+ glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
+ }
+ glEnd();
+
+ glDisable(GL_TEXTURE_RECTANGLE_EXT);
+
+ _bufferIsUpdated = false;
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelAGL::IsUpdated(bool& isUpdated)
+{
+ _owner->LockAGLCntx();
+ isUpdated = _bufferIsUpdated;
+ _owner->UnlockAGLCntx();
+
+ return 0;
+}
+
+int VideoChannelAGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+
+ _owner->LockAGLCntx();
+
+ _startWidth = startWidth;
+ _stopWidth = stopWidth;
+ _startHeight = startHeight;
+ _stopHeight = stopHeight;
+
+ int oldWidth = _width;
+ int oldHeight = _height;
+ int oldNumberOfStreams = _numberOfStreams;
+
+ _width = 0;
+ _height = 0;
+
+ int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
+
+ _owner->UnlockAGLCntx();
+
+ return retVal;
+}
+
+int VideoChannelAGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
+{
+ return -1;
+}
+
+#pragma mark VideoRenderAGL WindowRef constructor
+
+VideoRenderAGL::VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId) :
+_hiviewRef( 0),
+_windowRef( windowRef),
+_fullScreen( fullscreen),
+_id( iId),
+_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
+_screenUpdateThread( 0),
+_screenUpdateEvent( 0),
+_isHIViewRef( false),
+_aglContext( 0),
+_windowWidth( 0),
+_windowHeight( 0),
+_lastWindowWidth( -1),
+_lastWindowHeight( -1),
+_lastHiViewWidth( -1),
+_lastHiViewHeight( -1),
+_currentParentWindowHeight( 0),
+_currentParentWindowWidth( 0),
+_currentParentWindowBounds( ),
+_windowHasResized( false),
+_lastParentWindowBounds( ),
+_currentHIViewBounds( ),
+_lastHIViewBounds( ),
+_windowRect( ),
+_aglChannels( ),
+_zOrderToChannel( ),
+_hiviewEventHandlerRef( NULL),
+_windowEventHandlerRef( NULL),
+_currentViewBounds( ),
+_lastViewBounds( ),
+_renderingIsPaused( false),
+_threadID( )
+
+{
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s");
+
+ _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
+ _screenUpdateEvent = EventWrapper::Create();
+
+ if(!IsValidWindowPtr(_windowRef))
+ {
+ //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef:0x%x", __FUNCTION__, __LINE__, _windowRef);
+ }
+ else
+ {
+ //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d WindowRef 0x%x is valid", __FUNCTION__, __LINE__, _windowRef);
+ }
+
+ GetWindowRect(_windowRect);
+
+ _lastViewBounds.origin.x = 0;
+ _lastViewBounds.origin.y = 0;
+ _lastViewBounds.size.width = 0;
+ _lastViewBounds.size.height = 0;
+
+}
+
+// this is a static function. It has been registered (in class constructor) to be called on various window redrawing or resizing.
+// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
+#pragma mark WindowRef Event Handler
+pascal OSStatus VideoRenderAGL::sHandleWindowResized (EventHandlerCallRef /*nextHandler*/,
+ EventRef theEvent,
+ void* userData)
+{
+ WindowRef windowRef = NULL;
+
+ int eventType = GetEventKind(theEvent);
+
+ // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
+ GetEventParameter (theEvent,
+ kEventParamDirectObject,
+ typeWindowRef,
+ NULL,
+ sizeof (WindowRef),
+ NULL,
+ &windowRef);
+
+ VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
+
+ bool updateUI = true;
+ if(kEventWindowBoundsChanged == eventType)
+ {
+ }
+ else if(kEventWindowBoundsChanging == eventType)
+ {
+ }
+ else if(kEventWindowZoomed == eventType)
+ {
+ }
+ else if(kEventWindowExpanding == eventType)
+ {
+ }
+ else if(kEventWindowExpanded == eventType)
+ {
+ }
+ else if(kEventWindowClickResizeRgn == eventType)
+ {
+ }
+ else if(kEventWindowClickDragRgn == eventType)
+ {
+ }
+ else
+ {
+ updateUI = false;
+ }
+
+ if(true == updateUI)
+ {
+ obj->ParentWindowResized(windowRef);
+ obj->UpdateClipping();
+ obj->RenderOffScreenBuffers();
+ }
+
+ return noErr;
+}
+
+#pragma mark VideoRenderAGL HIViewRef constructor
+
+VideoRenderAGL::VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId) :
+_hiviewRef( windowRef),
+_windowRef( 0),
+_fullScreen( fullscreen),
+_id( iId),
+_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
+_screenUpdateThread( 0),
+_screenUpdateEvent( 0),
+_isHIViewRef( false),
+_aglContext( 0),
+_windowWidth( 0),
+_windowHeight( 0),
+_lastWindowWidth( -1),
+_lastWindowHeight( -1),
+_lastHiViewWidth( -1),
+_lastHiViewHeight( -1),
+_currentParentWindowHeight( 0),
+_currentParentWindowWidth( 0),
+_currentParentWindowBounds( ),
+_windowHasResized( false),
+_lastParentWindowBounds( ),
+_currentHIViewBounds( ),
+_lastHIViewBounds( ),
+_windowRect( ),
+_aglChannels( ),
+_zOrderToChannel( ),
+_hiviewEventHandlerRef( NULL),
+_windowEventHandlerRef( NULL),
+_currentViewBounds( ),
+_lastViewBounds( ),
+_renderingIsPaused( false),
+_threadID( )
+{
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Constructor", __FUNCTION__, __LINE__);
+ // _renderCritSec = CriticalSectionWrapper::CreateCriticalSection();
+
+ _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
+ _screenUpdateEvent = EventWrapper::Create();
+
+ GetWindowRect(_windowRect);
+
+ _lastViewBounds.origin.x = 0;
+ _lastViewBounds.origin.y = 0;
+ _lastViewBounds.size.width = 0;
+ _lastViewBounds.size.height = 0;
+
+#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
+ // This gets the parent window of the HIViewRef that's passed in and installs a WindowRef event handler on it
+ // The event handler looks for window resize events and adjusts the offset of the controls.
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef's parent window", __FUNCTION__, __LINE__);
+
+
+ static const EventTypeSpec windowEventTypes[] =
+ {
+ kEventClassWindow, kEventWindowBoundsChanged,
+ kEventClassWindow, kEventWindowBoundsChanging,
+ kEventClassWindow, kEventWindowZoomed,
+ kEventClassWindow, kEventWindowExpanded,
+ kEventClassWindow, kEventWindowClickResizeRgn,
+ kEventClassWindow, kEventWindowClickDragRgn
+ };
+
+ WindowRef parentWindow = HIViewGetWindow(windowRef);
+
+ InstallWindowEventHandler (parentWindow,
+ NewEventHandlerUPP (sHandleWindowResized),
+ GetEventTypeCount(windowEventTypes),
+ windowEventTypes,
+ (void *) this, // this is an arbitrary parameter that will be passed on to your event handler when it is called later
+ &_windowEventHandlerRef);
+
+#endif
+
+#ifdef NEW_HIVIEW_EVENT_HANDLER
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef", __FUNCTION__, __LINE__);
+
+ static const EventTypeSpec hiviewEventTypes[] =
+ {
+ kEventClassControl, kEventControlBoundsChanged,
+ kEventClassControl, kEventControlDraw
+ // kEventControlDragLeave
+ // kEventControlDragReceive
+ // kEventControlGetFocusPart
+ // kEventControlApplyBackground
+ // kEventControlDraw
+ // kEventControlHit
+
+ };
+
+ HIViewInstallEventHandler(_hiviewRef,
+ NewEventHandlerUPP(sHandleHiViewResized),
+ GetEventTypeCount(hiviewEventTypes),
+ hiviewEventTypes,
+ (void *) this,
+ &_hiviewEventHandlerRef);
+
+#endif
+}
+
+// this is a static function. It has been registered (in constructor) to be called on various window redrawing or resizing.
+// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
+#pragma mark HIViewRef Event Handler
+pascal OSStatus VideoRenderAGL::sHandleHiViewResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
+{
+ //static int callbackCounter = 1;
+ HIViewRef hiviewRef = NULL;
+
+ // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
+ int eventType = GetEventKind(theEvent);
+ OSStatus status = noErr;
+ status = GetEventParameter (theEvent,
+ kEventParamDirectObject,
+ typeControlRef,
+ NULL,
+ sizeof (ControlRef),
+ NULL,
+ &hiviewRef);
+
+ VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
+ WindowRef parentWindow = HIViewGetWindow(hiviewRef);
+ bool updateUI = true;
+
+ if(kEventControlBoundsChanged == eventType)
+ {
+ }
+ else if(kEventControlDraw == eventType)
+ {
+ }
+ else
+ {
+ updateUI = false;
+ }
+
+ if(true == updateUI)
+ {
+ obj->ParentWindowResized(parentWindow);
+ obj->UpdateClipping();
+ obj->RenderOffScreenBuffers();
+ }
+
+ return status;
+}
+
+VideoRenderAGL::~VideoRenderAGL()
+{
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Destructor", __FUNCTION__, __LINE__);
+
+
+#ifdef USE_EVENT_HANDLERS
+ // remove event handlers
+ OSStatus status;
+ if(_isHIViewRef)
+ {
+ status = RemoveEventHandler(_hiviewEventHandlerRef);
+ }
+ else
+ {
+ status = RemoveEventHandler(_windowEventHandlerRef);
+ }
+ if(noErr != status)
+ {
+ if(_isHIViewRef)
+ {
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
+ }
+ else
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
+ }
+ }
+
+#endif
+
+ OSStatus status;
+#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
+ if(_windowEventHandlerRef)
+ {
+ status = RemoveEventHandler(_windowEventHandlerRef);
+ if(status != noErr)
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
+ }
+ }
+#endif
+
+#ifdef NEW_HIVIEW_EVENT_HANDLER
+ if(_hiviewEventHandlerRef)
+ {
+ status = RemoveEventHandler(_hiviewEventHandlerRef);
+ if(status != noErr)
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
+ }
+ }
+#endif
+
+ // Signal event to exit thread, then delete it
+ ThreadWrapper* tmpPtr = _screenUpdateThread;
+ _screenUpdateThread = NULL;
+
+ if (tmpPtr)
+ {
+ tmpPtr->SetNotAlive();
+ _screenUpdateEvent->Set();
+ _screenUpdateEvent->StopTimer();
+
+ if (tmpPtr->Stop())
+ {
+ delete tmpPtr;
+ }
+ delete _screenUpdateEvent;
+ _screenUpdateEvent = NULL;
+ }
+
+ if (_aglContext != 0)
+ {
+ aglSetCurrentContext(_aglContext);
+ aglDestroyContext(_aglContext);
+ _aglContext = 0;
+ }
+
+ // Delete all channels
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
+ while (it!= _aglChannels.end())
+ {
+ delete it->second;
+ _aglChannels.erase(it);
+ it = _aglChannels.begin();
+ }
+ _aglChannels.clear();
+
+ // Clean the zOrder map
+ std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
+ while(zIt != _zOrderToChannel.end())
+ {
+ _zOrderToChannel.erase(zIt);
+ zIt = _zOrderToChannel.begin();
+ }
+ _zOrderToChannel.clear();
+
+ //delete _renderCritSec;
+
+
+}
+
+int VideoRenderAGL::GetOpenGLVersion(int& aglMajor, int& aglMinor)
+{
+ aglGetVersion((GLint *) &aglMajor, (GLint *) &aglMinor);
+ return 0;
+}
+
+int VideoRenderAGL::Init()
+{
+ LockAGLCntx();
+
+ // Start rendering thread...
+ if (!_screenUpdateThread)
+ {
+ UnlockAGLCntx();
+ //WEBRTC_TRACE(kTraceError, "%s:%d Thread not created", __FUNCTION__, __LINE__);
+ return -1;
+ }
+ unsigned int threadId;
+ _screenUpdateThread->Start(threadId);
+
+ // Start the event triggering the render process
+ unsigned int monitorFreq = 60;
+ _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
+
+ // Create mixing textures
+ if (CreateMixingContext() == -1)
+ {
+ //WEBRTC_TRACE(kTraceError, "%s:%d Could not create a mixing context", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+VideoChannelAGL* VideoRenderAGL::CreateAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+
+ LockAGLCntx();
+
+ //WEBRTC_TRACE(kTraceInfo, "%s:%d Creating AGL channel: %d", __FUNCTION__, __LINE__, channel);
+
+ if (HasChannel(channel))
+ {
+ //WEBRTC_TRACE(kTraceError, "%s:%d Channel already exists", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();k
+ return NULL;
+ }
+
+ if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
+ {
+ // There are already one channel using this zOrder
+ // TODO: Allow multiple channels with same zOrder
+ }
+
+ VideoChannelAGL* newAGLChannel = new VideoChannelAGL(_aglContext, _id, this);
+
+ if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
+ {
+ if (newAGLChannel)
+ {
+ delete newAGLChannel;
+ newAGLChannel = NULL;
+ }
+ //WEBRTC_LOG(kTraceError, "Could not create AGL channel");
+ //WEBRTC_TRACE(kTraceError, "%s:%d Could not create AGL channel", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return NULL;
+ }
+k
+ _aglChannels[channel] = newAGLChannel;
+ _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
+
+ UnlockAGLCntx();
+ return newAGLChannel;
+}
+
+int VideoRenderAGL::DeleteAllAGLChannels()
+{
+ CriticalSectionScoped cs(_renderCritSec);
+
+ //WEBRTC_TRACE(kTraceInfo, "%s:%d Deleting all AGL channels", __FUNCTION__, __LINE__);
+ //int i = 0 ;
+ std::map<int, VideoChannelAGL*>::iterator it;
+ it = _aglChannels.begin();
+
+ while (it != _aglChannels.end())
+ {
+ VideoChannelAGL* channel = it->second;
+ if (channel)
+ delete channel;
+
+ _aglChannels.erase(it);
+ it = _aglChannels.begin();
+ }
+ _aglChannels.clear();
+ return 0;
+}
+
+int VideoRenderAGL::DeleteAGLChannel(int channel)
+{
+ CriticalSectionScoped cs(_renderCritSec);
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Deleting AGL channel %d", __FUNCTION__, __LINE__, channel);
+
+ std::map<int, VideoChannelAGL*>::iterator it;
+ it = _aglChannels.find(channel);
+ if (it != _aglChannels.end())
+ {
+ delete it->second;
+ _aglChannels.erase(it);
+ }
+ else
+ {
+ //WEBRTC_TRACE(kTraceWarning, "%s:%d Channel not found", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
+ while( zIt != _zOrderToChannel.end())
+ {
+ if (zIt->second == channel)
+ {
+ _zOrderToChannel.erase(zIt);
+ break;
+ }
+ zIt++;// = _zOrderToChannel.begin();
+ }
+
+ return 0;
+}
+
+int VideoRenderAGL::StopThread()
+{
+ CriticalSectionScoped cs(_renderCritSec);
+ ThreadWrapper* tmpPtr = _screenUpdateThread;
+ //_screenUpdateThread = NULL;
+
+ if (tmpPtr)
+ {
+ tmpPtr->SetNotAlive();
+ _screenUpdateEvent->Set();
+ if (tmpPtr->Stop())
+ {
+ delete tmpPtr;
+ }
+ }
+
+ delete _screenUpdateEvent;
+ _screenUpdateEvent = NULL;
+
+ return 0;
+}
+
+bool VideoRenderAGL::IsFullScreen()
+{
+ CriticalSectionScoped cs(_renderCritSec);
+ return _fullScreen;
+}
+
+bool VideoRenderAGL::HasChannels()
+{
+
+ CriticalSectionScoped cs(_renderCritSec);
+
+ if (_aglChannels.begin() != _aglChannels.end())
+ {
+ return true;
+ }
+
+ return false;
+}
+
+bool VideoRenderAGL::HasChannel(int channel)
+{
+ CriticalSectionScoped cs(_renderCritSec);
+
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
+ if (it != _aglChannels.end())
+ {
+ return true;
+ }
+
+ return false;
+}
+
+int VideoRenderAGL::GetChannels(std::list<int>& channelList)
+{
+
+ CriticalSectionScoped cs(_renderCritSec);
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
+
+ while (it != _aglChannels.end())
+ {
+ channelList.push_back(it->first);
+ it++;
+ }
+
+ return 0;
+}
+
+VideoChannelAGL* VideoRenderAGL::ConfigureAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+
+ CriticalSectionScoped cs(_renderCritSec);
+
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
+
+ if (it != _aglChannels.end())
+ {
+ VideoChannelAGL* aglChannel = it->second;
+ if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
+ {
+ return NULL;
+ }
+
+ std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
+ while(it != _zOrderToChannel.end())
+ {
+ if (it->second == channel)
+ {
+ if (it->first != zOrder)
+ {
+ _zOrderToChannel.erase(it);
+ _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
+ }
+ break;
+ }
+ it++;
+ }
+ return aglChannel;
+ }
+
+ return NULL;
+}
+
+bool VideoRenderAGL::ScreenUpdateThreadProc(void* obj)
+{
+ return static_cast<VideoRenderAGL*>(obj)->ScreenUpdateProcess();
+}
+
+bool VideoRenderAGL::ScreenUpdateProcess()
+{
+ _screenUpdateEvent->Wait(100);
+
+ LockAGLCntx();
+
+ if (!_screenUpdateThread)
+ {
+ UnlockAGLCntx();
+ return false;
+ }
+
+ if (aglSetCurrentContext(_aglContext) == GL_FALSE)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+
+ if (_windowWidth != (_windowRect.right - _windowRect.left)
+ || _windowHeight != (_windowRect.bottom - _windowRect.top))
+ {
+ // We have a new window size, update the context.
+ if (aglUpdateContext(_aglContext) == GL_FALSE)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+ _windowWidth = _windowRect.right - _windowRect.left;
+ _windowHeight = _windowRect.bottom - _windowRect.top;
+ }
+
+ // this section will poll to see if the window size has changed
+ // this is causing problem w/invalid windowRef
+ // this code has been modified and exists now in the window event handler
+#ifndef NEW_HIVIEW_PARENT_EVENT_HANDLER
+ if (_isHIViewRef)
+ {
+
+ if(FALSE == HIViewIsValid(_hiviewRef))
+ {
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalid windowRef", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return true;
+ }
+ WindowRef window = HIViewGetWindow(_hiviewRef);
+
+ if(FALSE == IsValidWindowPtr(window))
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalide hiviewRef", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return true;
+ }
+ if (window == NULL)
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d WindowRef = NULL", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return true;
+ }
+
+ if(FALSE == MacIsWindowVisible(window))
+ {
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d MacIsWindowVisible == FALSE. Returning early", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return true;
+ }
+
+ HIRect viewBounds; // Placement and size for HIView
+ int windowWidth = 0; // Parent window width
+ int windowHeight = 0; // Parent window height
+
+ // NOTE: Calling GetWindowBounds with kWindowStructureRgn will crash intermittentaly if the OS decides it needs to push it into the back for a moment.
+ // To counter this, we get the titlebar height on class construction and then add it to the content region here. Content regions seems not to crash
+ Rect contentBounds =
+ { 0, 0, 0, 0}; // The bounds for the parent window
+
+#if defined(USE_CONTENT_RGN)
+ GetWindowBounds(window, kWindowContentRgn, &contentBounds);
+#elif defined(USE_STRUCT_RGN)
+ GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
+#endif
+
+ Rect globalBounds =
+ { 0, 0, 0, 0}; // The bounds for the parent window
+ globalBounds.top = contentBounds.top;
+ globalBounds.right = contentBounds.right;
+ globalBounds.bottom = contentBounds.bottom;
+ globalBounds.left = contentBounds.left;
+
+ windowHeight = globalBounds.bottom - globalBounds.top;
+ windowWidth = globalBounds.right - globalBounds.left;
+
+ // Get the size of the HIViewRef
+ HIViewGetBounds(_hiviewRef, &viewBounds);
+ HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
+
+ // Check if this is the first call..
+ if (_lastWindowHeight == -1 &&
+ _lastWindowWidth == -1)
+ {
+ _lastWindowWidth = windowWidth;
+ _lastWindowHeight = windowHeight;
+
+ _lastViewBounds.origin.x = viewBounds.origin.x;
+ _lastViewBounds.origin.y = viewBounds.origin.y;
+ _lastViewBounds.size.width = viewBounds.size.width;
+ _lastViewBounds.size.height = viewBounds.size.height;
+ }
+ sfasdfasdf
+
+ bool resized = false;
+
+ // Check if parent window size has changed
+ if (windowHeight != _lastWindowHeight ||
+ windowWidth != _lastWindowWidth)
+ {
+ resized = true;
+ }
+
+ // Check if the HIView has new size or is moved in the parent window
+ if (_lastViewBounds.origin.x != viewBounds.origin.x ||
+ _lastViewBounds.origin.y != viewBounds.origin.y ||
+ _lastViewBounds.size.width != viewBounds.size.width ||
+ _lastViewBounds.size.height != viewBounds.size.height)
+ {
+ // The HiView is resized or has moved.
+ resized = true;
+ }
+
+ if (resized)
+ {
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d Window has resized", __FUNCTION__, __LINE__);
+
+ // Calculate offset between the windows
+ // {x, y, widht, height}, x,y = lower left corner
+ const GLint offs[4] =
+ { (int)(0.5f + viewBounds.origin.x),
+ (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
+ viewBounds.size.width, viewBounds.size.height};
+
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d contentBounds t:%d r:%d b:%d l:%d", __FUNCTION__, __LINE__,
+ contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d windowHeight=%d", __FUNCTION__, __LINE__, windowHeight);
+ //WEBRTC_TRACE(kTraceDebug, "%s:%d offs[4] = %d, %d, %d, %d", __FUNCTION__, __LINE__, offs[0], offs[1], offs[2], offs[3]);
+
+ aglSetDrawable (_aglContext, GetWindowPort(window));
+ aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
+ aglEnable(_aglContext, AGL_BUFFER_RECT);
+
+ // We need to change the viewport too if the HIView size has changed
+ glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
+
+ }
+ _lastWindowWidth = windowWidth;
+ _lastWindowHeight = windowHeight;
+
+ _lastViewBounds.origin.x = viewBounds.origin.x;
+ _lastViewBounds.origin.y = viewBounds.origin.y;
+ _lastViewBounds.size.width = viewBounds.size.width;
+ _lastViewBounds.size.height = viewBounds.size.height;
+
+ }
+#endif
+ if (_fullScreen)
+ {
+ // TODO
+ // We use double buffers, must always update
+ //RenderOffScreenBuffersToBackBuffer();
+ }
+ else
+ {
+ // Check if there are any updated buffers
+ bool updated = false;
+
+ // TODO: check if window size is updated!
+ // TODO Improvement: Walk through the zOrder Map to only render the ones in need of update
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
+ while (it != _aglChannels.end())
+ {
+
+ VideoChannelAGL* aglChannel = it->second;
+ aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
+ aglChannel->IsUpdated(updated);
+ if (updated)
+ {
+ break;
+ }
+ it++;
+ }
+
+ if (updated)
+ {
+ // At least on buffers is updated, we need to repaint the texture
+ if (RenderOffScreenBuffers() != -1)
+ {
+ // MF
+ //SwapAndDisplayBuffers();
+ }
+ else
+ {
+ // Error updating the mixing texture, don't swap.
+ }
+ }
+ }
+
+ UnlockAGLCntx();
+
+ //WEBRTC_LOG(kTraceDebug, "Leaving ScreenUpdateProcess()");
+ return true;
+}
+
+void VideoRenderAGL::ParentWindowResized(WindowRef window)
+{
+ //WEBRTC_LOG(kTraceDebug, "%s HIViewRef:%d owner window has resized", __FUNCTION__, (int)_hiviewRef);
+
+ LockAGLCntx();
+k
+ // set flag
+ _windowHasResized = false;
+
+ if(FALSE == HIViewIsValid(_hiviewRef))
+ {
+ //WEBRTC_LOG(kTraceDebug, "invalid windowRef");
+ UnlockAGLCntx();
+ return;
+ }
+
+ if(FALSE == IsValidWindowPtr(window))
+ {
+ //WEBRTC_LOG(kTraceError, "invalid windowRef");
+ UnlockAGLCntx();
+ return;
+ }
+
+ if (window == NULL)
+ {
+ //WEBRTC_LOG(kTraceError, "windowRef = NULL");
+ UnlockAGLCntx();
+ return;
+ }
+
+ if(FALSE == MacIsWindowVisible(window))
+ {
+ //WEBRTC_LOG(kTraceDebug, "MacIsWindowVisible = FALSE. Returning early.");
+ UnlockAGLCntx();
+ return;
+ }
+
+ Rect contentBounds =
+ { 0, 0, 0, 0};
+
+#if defined(USE_CONTENT_RGN)
+ GetWindowBounds(window, kWindowContentRgn, &contentBounds);
+#elif defined(USE_STRUCT_RGN)
+ GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
+#endif
+
+ //WEBRTC_LOG(kTraceDebug, "%s contentBounds t:%d r:%d b:%d l:%d", __FUNCTION__, contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
+
+ // update global vars
+ _currentParentWindowBounds.top = contentBounds.top;
+ _currentParentWindowBounds.left = contentBounds.left;
+ _currentParentWindowBounds.bottom = contentBounds.bottom;
+ _currentParentWindowBounds.right = contentBounds.right;
+
+ _currentParentWindowWidth = _currentParentWindowBounds.right - _currentParentWindowBounds.left;
+ _currentParentWindowHeight = _currentParentWindowBounds.bottom - _currentParentWindowBounds.top;
+
+ _windowHasResized = true;
+
+ // ********* update AGL offsets
+ HIRect viewBounds;
+ HIViewGetBounds(_hiviewRef, &viewBounds);
+ HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
+
+ const GLint offs[4] =
+ { (int)(0.5f + viewBounds.origin.x),
+ (int)(0.5f + _currentParentWindowHeight - (viewBounds.origin.y + viewBounds.size.height)),
+ viewBounds.size.width, viewBounds.size.height};
+ //WEBRTC_LOG(kTraceDebug, "%s _currentParentWindowHeight=%d", __FUNCTION__, _currentParentWindowHeight);
+ //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
+
+ aglSetCurrentContext(_aglContext);
+ aglSetDrawable (_aglContext, GetWindowPort(window));
+ aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
+ aglEnable(_aglContext, AGL_BUFFER_RECT);
+
+ // We need to change the viewport too if the HIView size has changed
+ glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
+
+ UnlockAGLCntx();
+
+ return;
+}
+
+int VideoRenderAGL::CreateMixingContext()
+{
+
+ LockAGLCntx();
+
+ //WEBRTC_LOG(kTraceDebug, "Entering CreateMixingContext()");
+
+ // Use both AGL_ACCELERATED and AGL_NO_RECOVERY to make sure
+ // a hardware renderer is used and not a software renderer.
+
+ GLint attributes[] =
+ {
+ AGL_DOUBLEBUFFER,
+ AGL_WINDOW,
+ AGL_RGBA,
+ AGL_NO_RECOVERY,
+ AGL_ACCELERATED,
+ AGL_RED_SIZE, 8,
+ AGL_GREEN_SIZE, 8,
+ AGL_BLUE_SIZE, 8,
+ AGL_ALPHA_SIZE, 8,
+ AGL_DEPTH_SIZE, 24,
+ AGL_NONE,
+ };
+
+ AGLPixelFormat aglPixelFormat;
+
+ // ***** Set up the OpenGL Context *****
+
+ // Get a pixel format for the attributes above
+ aglPixelFormat = aglChoosePixelFormat(NULL, 0, attributes);
+ if (NULL == aglPixelFormat)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not create pixel format");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // Create an AGL context
+ _aglContext = aglCreateContext(aglPixelFormat, NULL);
+ if (_aglContext == NULL)
+ {
+ //WEBRTC_LOG(kTraceError, "Could no create AGL context");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // Release the pixel format memory
+ aglDestroyPixelFormat(aglPixelFormat);
+
+ // Set the current AGL context for the rest of the settings
+ if (aglSetCurrentContext(_aglContext) == false)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set current context: %d", aglGetError());
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ if (_isHIViewRef)
+ {
+ //---------------------------
+ // BEGIN: new test code
+#if 0
+ // Don't use this one!
+ // There seems to be an OS X bug that can't handle
+ // movements and resizing of the parent window
+ // and or the HIView
+ if (aglSetHIViewRef(_aglContext,_hiviewRef) == false)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
+ UnlockAGLCntx();
+ return -1;
+ }
+#else
+
+ // Get the parent window for this control
+ WindowRef window = GetControlOwner(_hiviewRef);
+
+ Rect globalBounds =
+ { 0,0,0,0}; // The bounds for the parent window
+ HIRect viewBounds; // Placemnt in the parent window and size.
+ int windowHeight = 0;
+
+ // Rect titleBounds = {0,0,0,0};
+ // GetWindowBounds(window, kWindowTitleBarRgn, &titleBounds);
+ // _titleBarHeight = titleBounds.top - titleBounds.bottom;
+ // if(0 == _titleBarHeight)
+ // {
+ // //WEBRTC_LOG(kTraceError, "Titlebar height = 0");
+ // //return -1;
+ // }
+
+
+ // Get the bounds for the parent window
+#if defined(USE_CONTENT_RGN)
+ GetWindowBounds(window, kWindowContentRgn, &globalBounds);
+#elif defined(USE_STRUCT_RGN)
+ GetWindowBounds(window, kWindowStructureRgn, &globalBounds);
+#endif
+ windowHeight = globalBounds.bottom - globalBounds.top;
+
+ // Get the bounds for the HIView
+ HIViewGetBounds(_hiviewRef, &viewBounds);
+
+ HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
+
+ const GLint offs[4] =
+ { (int)(0.5f + viewBounds.origin.x),
+ (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
+ viewBounds.size.width, viewBounds.size.height};
+
+ //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
+
+
+ aglSetDrawable (_aglContext, GetWindowPort(window));
+ aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
+ aglEnable(_aglContext, AGL_BUFFER_RECT);
+
+ GLint surfaceOrder = 1; // 1: above window, -1 below.
+ //OSStatus status = aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
+ aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
+
+ glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
+#endif
+
+ }
+ else
+ {
+ if(GL_FALSE == aglSetDrawable (_aglContext, GetWindowPort(_windowRef)))
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
+ UnlockAGLCntx();
+ return -1;
+ }
+ }
+
+ _windowWidth = _windowRect.right - _windowRect.left;
+ _windowHeight = _windowRect.bottom - _windowRect.top;
+
+ // opaque surface
+ int surfaceOpacity = 1;
+ if (aglSetInteger(_aglContext, AGL_SURFACE_OPACITY, (const GLint *) &surfaceOpacity) == false)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set surface opacity: %d", aglGetError());
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // 1 -> sync to screen rat, slow...
+ //int swapInterval = 0; // 0 don't sync with vertical trace
+ int swapInterval = 0; // 1 sync with vertical trace
+ if (aglSetInteger(_aglContext, AGL_SWAP_INTERVAL, (const GLint *) &swapInterval) == false)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set swap interval: %d", aglGetError());
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // Update the rect with the current size
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not get window size");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // Disable not needed functionality to increase performance
+ glDisable(GL_DITHER);
+ glDisable(GL_ALPHA_TEST);
+ glDisable(GL_STENCIL_TEST);
+ glDisable(GL_FOG);
+ glDisable(GL_TEXTURE_2D);
+ glPixelZoom(1.0, 1.0);
+
+ glDisable(GL_BLEND);
+ glDisable(GL_DEPTH_TEST);
+ glDepthMask(GL_FALSE);
+ glDisable(GL_CULL_FACE);
+
+ glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
+ glClear(GL_COLOR_BUFFER_BIT);
+
+ GLenum glErr = glGetError();
+
+ if (glErr)
+ {
+ }
+
+ UpdateClipping();
+
+ //WEBRTC_LOG(kTraceDebug, "Leaving CreateMixingContext()");
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderAGL::RenderOffScreenBuffers()
+{
+ LockAGLCntx();
+
+ // Get the current window size, it might have changed since last render.
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not get window rect");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ if (aglSetCurrentContext(_aglContext) == false)
+ {
+ //WEBRTC_LOG(kTraceError, "Could not set current context for rendering");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ // HERE - onl if updated!
+ glClear(GL_COLOR_BUFFER_BIT);
+
+ // Loop through all channels starting highest zOrder ending with lowest.
+ for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
+ rIt != _zOrderToChannel.rend();
+ rIt++)
+ {
+ int channelId = rIt->second;
+ std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channelId);
+
+ VideoChannelAGL* aglChannel = it->second;
+
+ aglChannel->RenderOffScreenBuffer();
+ }
+
+ SwapAndDisplayBuffers();
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderAGL::SwapAndDisplayBuffers()
+{
+
+ LockAGLCntx();
+ if (_fullScreen)
+ {
+ // TODO:
+ // Swap front and back buffers, rendering taking care of in the same call
+ //aglSwapBuffers(_aglContext);
+ // Update buffer index to the idx for the next rendering!
+ //_textureIdx = (_textureIdx + 1) & 1;
+ }
+ else
+ {
+ // Single buffer rendering, only update context.
+ glFlush();
+ aglSwapBuffers(_aglContext);
+ HIViewSetNeedsDisplay(_hiviewRef, true);
+ }
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderAGL::GetWindowRect(Rect& rect)
+{
+
+ LockAGLCntx();
+
+ if (_isHIViewRef)
+ {
+ if (_hiviewRef)
+ {
+ HIRect HIViewRect1;
+ if(FALSE == HIViewIsValid(_hiviewRef))
+ {
+ rect.top = 0;
+ rect.left = 0;
+ rect.right = 0;
+ rect.bottom = 0;
+ //WEBRTC_LOG(kTraceError,"GetWindowRect() HIViewIsValid() returned false");
+ UnlockAGLCntx();
+ }
+ HIViewGetBounds(_hiviewRef,&HIViewRect1);
+ HIRectConvert(&HIViewRect1, 1, NULL, 2, NULL);
+ if(HIViewRect1.origin.x < 0)
+ {
+ rect.top = 0;
+ //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.top = 0");
+ }
+ else
+ {
+ rect.top = HIViewRect1.origin.x;
+ }
+
+ if(HIViewRect1.origin.y < 0)
+ {
+ rect.left = 0;
+ //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.left = 0");
+ }
+ else
+ {
+ rect.left = HIViewRect1.origin.y;
+ }
+
+ if(HIViewRect1.size.width < 0)
+ {
+ rect.right = 0;
+ //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.right = 0");
+ }
+ else
+ {
+ rect.right = HIViewRect1.size.width;
+ }
+
+ if(HIViewRect1.size.height < 0)
+ {
+ rect.bottom = 0;
+ //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.bottom = 0");
+ }
+ else
+ {
+ rect.bottom = HIViewRect1.size.height;
+ }
+
+ ////WEBRTC_LOG(kTraceDebug,"GetWindowRect() HIViewRef: rect.top = %d, rect.left = %d, rect.right = %d, rect.bottom =%d in GetWindowRect", rect.top,rect.left,rect.right,rect.bottom);
+ UnlockAGLCntx();
+ }
+ else
+ {
+ //WEBRTC_LOG(kTraceError, "invalid HIViewRef");
+ UnlockAGLCntx();
+ }
+ }
+ else
+ {
+ if (_windowRef)
+ {
+ GetWindowBounds(_windowRef, kWindowContentRgn, &rect);
+ UnlockAGLCntx();
+ }
+ else
+ {
+ //WEBRTC_LOG(kTraceError, "No WindowRef");
+ UnlockAGLCntx();
+ }
+ }
+}
+
+int VideoRenderAGL::UpdateClipping()
+{
+ //WEBRTC_LOG(kTraceDebug, "Entering UpdateClipping()");
+ LockAGLCntx();
+
+ if(_isHIViewRef)
+ {
+ if(FALSE == HIViewIsValid(_hiviewRef))
+ {
+ //WEBRTC_LOG(kTraceError, "UpdateClipping() _isHIViewRef is invalid. Returning -1");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ RgnHandle visibleRgn = NewRgn();
+ SetEmptyRgn (visibleRgn);
+
+ if(-1 == CalculateVisibleRegion((ControlRef)_hiviewRef, visibleRgn, true))
+ {
+ }
+
+ if(GL_FALSE == aglSetCurrentContext(_aglContext))
+ {
+ GLenum glErr = aglGetError();
+ //WEBRTC_LOG(kTraceError, "aglSetCurrentContext returned FALSE with error code %d at line %d", glErr, __LINE__);
+ }
+
+ if(GL_FALSE == aglEnable(_aglContext, AGL_CLIP_REGION))
+ {
+ GLenum glErr = aglGetError();
+ //WEBRTC_LOG(kTraceError, "aglEnable returned FALSE with error code %d at line %d\n", glErr, __LINE__);
+ }
+
+ if(GL_FALSE == aglSetInteger(_aglContext, AGL_CLIP_REGION, (const GLint*)visibleRgn))
+ {
+ GLenum glErr = aglGetError();
+ //WEBRTC_LOG(kTraceError, "aglSetInteger returned FALSE with error code %d at line %d\n", glErr, __LINE__);
+ }
+
+ DisposeRgn(visibleRgn);
+ }
+ else
+ {
+ //WEBRTC_LOG(kTraceDebug, "Not using a hiviewref!\n");
+ }
+
+ //WEBRTC_LOG(kTraceDebug, "Leaving UpdateClipping()");
+ UnlockAGLCntx();
+ return true;
+}
+
+int VideoRenderAGL::CalculateVisibleRegion(ControlRef control, RgnHandle &visibleRgn, bool clipChildren)
+{
+
+ // LockAGLCntx();
+
+ //WEBRTC_LOG(kTraceDebug, "Entering CalculateVisibleRegion()");
+ OSStatus osStatus = 0;
+ OSErr osErr = 0;
+
+ RgnHandle tempRgn = NewRgn();
+ if (IsControlVisible(control))
+ {
+ RgnHandle childRgn = NewRgn();
+ WindowRef window = GetControlOwner(control);
+ ControlRef rootControl;
+ GetRootControl(window, &rootControl); // 'wvnc'
+ ControlRef masterControl;
+ osStatus = GetSuperControl(rootControl, &masterControl);
+ // //WEBRTC_LOG(kTraceDebug, "IBM GetSuperControl=%d", osStatus);
+
+ if (masterControl != NULL)
+ {
+ CheckValidRegion(visibleRgn);
+ // init visibleRgn with region of 'wvnc'
+ osStatus = GetControlRegion(rootControl, kControlStructureMetaPart, visibleRgn);
+ // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
+ //GetSuperControl(rootControl, &rootControl);
+ ControlRef tempControl = control, lastControl = 0;
+ while (tempControl != masterControl) // current control != master
+
+ {
+ CheckValidRegion(tempRgn);
+
+ // //WEBRTC_LOG(kTraceDebug, "IBM tempControl=%d masterControl=%d", tempControl, masterControl);
+ ControlRef subControl;
+
+ osStatus = GetControlRegion(tempControl, kControlStructureMetaPart, tempRgn); // intersect the region of the current control with visibleRgn
+ // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
+ CheckValidRegion(tempRgn);
+
+ osErr = HIViewConvertRegion(tempRgn, tempControl, rootControl);
+ // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d : %d", osErr, __LINE__);
+ CheckValidRegion(tempRgn);
+
+ SectRgn(tempRgn, visibleRgn, visibleRgn);
+ CheckValidRegion(tempRgn);
+ CheckValidRegion(visibleRgn);
+ if (EmptyRgn(visibleRgn)) // if the region is empty, bail
+ break;
+
+ if (clipChildren || tempControl != control) // clip children if true, cut out the tempControl if it's not one passed to this function
+
+ {
+ UInt16 numChildren;
+ osStatus = CountSubControls(tempControl, &numChildren); // count the subcontrols
+ // //WEBRTC_LOG(kTraceDebug, "IBM CountSubControls=%d : %d", osStatus, __LINE__);
+
+ // //WEBRTC_LOG(kTraceDebug, "IBM numChildren=%d", numChildren);
+ for (int i = 0; i < numChildren; i++)
+ {
+ osErr = GetIndexedSubControl(tempControl, numChildren - i, &subControl); // retrieve the subcontrol in order by zorder
+ // //WEBRTC_LOG(kTraceDebug, "IBM GetIndexedSubControls=%d : %d", osErr, __LINE__);
+ if ( subControl == lastControl ) // break because of zorder
+
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM breaking because of zorder %d", __LINE__);
+ break;
+ }
+
+ if (!IsControlVisible(subControl)) // dont' clip invisible controls
+
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM continue. Control is not visible %d", __LINE__);
+ continue;
+ }
+
+ if(!subControl) continue;
+
+ osStatus = GetControlRegion(subControl, kControlStructureMetaPart, tempRgn); //get the region of the current control and union to childrg
+ // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d %d", osStatus, __LINE__);
+ CheckValidRegion(tempRgn);
+ if(osStatus != 0)
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
+ continue;
+ }
+ if(!tempRgn)
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !tempRgn %d", osStatus, __LINE__);
+ continue;
+ }
+
+ osStatus = HIViewConvertRegion(tempRgn, subControl, rootControl);
+ CheckValidRegion(tempRgn);
+ // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d %d", osStatus, __LINE__);
+ if(osStatus != 0)
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
+ continue;
+ }
+ if(!rootControl)
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !rootControl %d", osStatus, __LINE__);
+ continue;
+ }
+
+ UnionRgn(tempRgn, childRgn, childRgn);
+ CheckValidRegion(tempRgn);
+ CheckValidRegion(childRgn);
+ CheckValidRegion(visibleRgn);
+ if(!childRgn)
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !childRgn %d", osStatus, __LINE__);
+ continue;
+ }
+
+ } // next child control
+ }
+ lastControl = tempControl;
+ GetSuperControl(tempControl, &subControl);
+ tempControl = subControl;
+ }
+
+ DiffRgn(visibleRgn, childRgn, visibleRgn);
+ CheckValidRegion(visibleRgn);
+ CheckValidRegion(childRgn);
+ DisposeRgn(childRgn);
+ }
+ else
+ {
+ CopyRgn(tempRgn, visibleRgn);
+ CheckValidRegion(tempRgn);
+ CheckValidRegion(visibleRgn);
+ }
+ DisposeRgn(tempRgn);
+ }
+
+ //WEBRTC_LOG(kTraceDebug, "Leaving CalculateVisibleRegion()");
+ //_aglCritPtr->Leave();
+ return 0;
+}
+
+bool VideoRenderAGL::CheckValidRegion(RgnHandle rHandle)
+{
+
+ Handle hndSize = (Handle)rHandle;
+ long size = GetHandleSize(hndSize);
+ if(0 == size)
+ {
+
+ OSErr memErr = MemError();
+ if(noErr != memErr)
+ {
+ // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle. MemError() returned %d", memErr);
+ }
+ else
+ {
+ // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle yet MemError() returned noErr");
+ }
+
+ }
+ else
+ {
+ // //WEBRTC_LOG(kTraceDebug, "IBM handleSize = %d", size);
+ }
+
+ if(false == IsValidRgnHandle(rHandle))
+ {
+ // //WEBRTC_LOG(kTraceError, "IBM ERROR Invalid Region found : $%d", rHandle);
+ assert(false);
+ }
+
+ int err = QDError();
+ switch(err)
+ {
+ case 0:
+ break;
+ case -147:
+ //WEBRTC_LOG(kTraceError, "ERROR region too big");
+ assert(false);
+ break;
+
+ case -149:
+ //WEBRTC_LOG(kTraceError, "ERROR not enough stack");
+ assert(false);
+ break;
+
+ default:
+ //WEBRTC_LOG(kTraceError, "ERROR Unknown QDError %d", err);
+ assert(false);
+ break;
+ }
+
+ return true;
+}
+
+int VideoRenderAGL::ChangeWindow(void* newWindowRef)
+{
+
+ LockAGLCntx();
+
+ UnlockAGLCntx();
+ return -1;
+}
+WebRtc_Word32 VideoRenderAGL::ChangeUniqueID(WebRtc_Word32 id)
+{
+ LockAGLCntx();
+
+ UnlockAGLCntx();
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderAGL::StartRender()
+{
+
+ LockAGLCntx();
+ const unsigned int MONITOR_FREQ = 60;
+ if(TRUE == _renderingIsPaused)
+ {
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Rendering is paused. Restarting now", __FUNCTION__, __LINE__);
+
+ // we already have the thread. Most likely StopRender() was called and they were paused
+ if(FALSE == _screenUpdateThread->Start(_threadID))
+ {
+ //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return -1;
+ }
+ if(FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
+ {
+ //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateEvent", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ return 0;
+ }
+
+ _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
+ _screenUpdateEvent = EventWrapper::Create();
+
+ if (!_screenUpdateThread)
+ {
+ //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ _screenUpdateThread->Start(_threadID);
+ _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ);
+
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Started screenUpdateThread", __FUNCTION__, __LINE__);
+
+ UnlockAGLCntx();
+ return 0;
+
+}
+
+WebRtc_Word32 VideoRenderAGL::StopRender()
+{
+ LockAGLCntx();
+
+ if(!_screenUpdateThread || !_screenUpdateEvent)
+ {
+ _renderingIsPaused = TRUE;
+ UnlockAGLCntx();
+ return 0;
+ }
+
+ if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
+ {
+ _renderingIsPaused = FALSE;
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Could not stop either: screenUpdateThread or screenUpdateEvent", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ _renderingIsPaused = TRUE;
+
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Stopped screenUpdateThread", __FUNCTION__, __LINE__);
+ UnlockAGLCntx();
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderAGL::DeleteAGLChannel(const WebRtc_UWord32 streamID)
+{
+
+ LockAGLCntx();
+
+ std::map<int, VideoChannelAGL*>::iterator it;
+ it = _aglChannels.begin();
+
+ while (it != _aglChannels.end())
+ {
+ VideoChannelAGL* channel = it->second;
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Deleting channel %d", __FUNCTION__, __LINE__, streamID);
+ delete channel;
+ it++;
+ }
+ _aglChannels.clear();
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderAGL::GetChannelProperties(const WebRtc_UWord16 streamId,
+WebRtc_UWord32& zOrder,
+float& left,
+float& top,
+float& right,
+float& bottom)
+{
+
+ LockAGLCntx();
+ UnlockAGLCntx();
+ return -1;
+
+}
+
+void VideoRenderAGL::LockAGLCntx()
+{
+ _renderCritSec.Enter();
+}
+void VideoRenderAGL::UnlockAGLCntx()
+{
+ _renderCritSec.Leave();
+}
+
+} //namespace webrtc
+
+#endif // CARBON_RENDERING
+
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_agl.h b/trunk/src/modules/video_render/main/source/mac/video_render_agl.h
new file mode 100644
index 0000000..bdee619
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_agl.h
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(CARBON_RENDERING)
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
+
+
+#include "video_render_defines.h"
+
+
+#define NEW_HIVIEW_PARENT_EVENT_HANDLER 1
+#define NEW_HIVIEW_EVENT_HANDLER 1
+#define USE_STRUCT_RGN
+
+#include <AGL/agl.h>
+#include <Carbon/Carbon.h>
+#include <OpenGL/OpenGL.h>
+#include <OpenGL/glu.h>
+#include <OpenGL/glext.h>
+#include <list>
+#include <map>
+
+class VideoRenderAGL;
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+
+class VideoChannelAGL : public VideoRenderCallback
+{
+public:
+
+ VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner);
+ virtual ~VideoChannelAGL();
+ virtual int FrameSizeChange(int width, int height, int numberOfStreams);
+ virtual int DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int timeStame90kHz);
+ virtual int UpdateSize(int width, int height);
+ int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ int RenderOffScreenBuffer();
+ int IsUpdated(bool& isUpdated);
+ virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
+
+
+private:
+
+ AGLContext _aglContext;
+ int _id;
+ VideoRenderAGL* _owner;
+ int _width;
+ int _height;
+ int _stretchedWidth;
+ int _stretchedHeight;
+ float _startHeight;
+ float _startWidth;
+ float _stopWidth;
+ float _stopHeight;
+ int _xOldWidth;
+ int _yOldHeight;
+ int _oldStretchedHeight;
+ int _oldStretchedWidth;
+ unsigned char* _buffer;
+ int _bufferSize;
+ int _incommingBufferSize;
+ bool _bufferIsUpdated;
+ bool _sizeInitialized;
+ int _numberOfStreams;
+ bool _bVideoSizeStartedChanging;
+ GLenum _pixelFormat;
+ GLenum _pixelDataType;
+ unsigned int _texture;
+};
+
+
+
+
+class VideoRenderAGL
+{
+public:
+ VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId);
+ VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId);
+ ~VideoRenderAGL();
+
+ int Init();
+ VideoChannelAGL* CreateAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ VideoChannelAGL* ConfigureAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ int DeleteAGLChannel(int channel);
+ int DeleteAllAGLChannels();
+ int StopThread();
+ bool IsFullScreen();
+ bool HasChannels();
+ bool HasChannel(int channel);
+ int GetChannels(std::list<int>& channelList);
+ void LockAGLCntx();
+ void UnlockAGLCntx();
+
+ static int GetOpenGLVersion(int& aglMajor, int& aglMinor);
+
+ // ********** new module functions ************ //
+ int ChangeWindow(void* newWindowRef);
+ WebRtc_Word32 ChangeUniqueID(WebRtc_Word32 id);
+ WebRtc_Word32 StartRender();
+ WebRtc_Word32 StopRender();
+ WebRtc_Word32 DeleteAGLChannel(const WebRtc_UWord32 streamID);
+ WebRtc_Word32 GetChannelProperties(const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom);
+
+protected:
+ static bool ScreenUpdateThreadProc(void* obj);
+ bool ScreenUpdateProcess();
+ int GetWindowRect(Rect& rect);
+
+private:
+ int CreateMixingContext();
+ int RenderOffScreenBuffers();
+ int SwapAndDisplayBuffers();
+ int UpdateClipping();
+ int CalculateVisibleRegion(ControlRef control, RgnHandle &visibleRgn, bool clipChildren);
+ bool CheckValidRegion(RgnHandle rHandle);
+ void ParentWindowResized(WindowRef window);
+
+ // Carbon GUI event handlers
+ static pascal OSStatus sHandleWindowResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
+ static pascal OSStatus sHandleHiViewResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
+
+ HIViewRef _hiviewRef;
+ WindowRef _windowRef;
+ bool _fullScreen;
+ int _id;
+ webrtc::CriticalSectionWrapper& _renderCritSec;
+ webrtc::ThreadWrapper* _screenUpdateThread;
+ webrtc::EventWrapper* _screenUpdateEvent;
+ bool _isHIViewRef;
+ AGLContext _aglContext;
+ int _windowWidth;
+ int _windowHeight;
+ int _lastWindowWidth;
+ int _lastWindowHeight;
+ int _lastHiViewWidth;
+ int _lastHiViewHeight;
+ int _currentParentWindowHeight;
+ int _currentParentWindowWidth;
+ Rect _currentParentWindowBounds;
+ bool _windowHasResized;
+ Rect _lastParentWindowBounds;
+ Rect _currentHIViewBounds;
+ Rect _lastHIViewBounds;
+ Rect _windowRect;
+ std::map<int, VideoChannelAGL*> _aglChannels;
+ std::multimap<int, int> _zOrderToChannel;
+ EventHandlerRef _hiviewEventHandlerRef;
+ EventHandlerRef _windowEventHandlerRef;
+ HIRect _currentViewBounds;
+ HIRect _lastViewBounds;
+ bool _renderingIsPaused;
+ unsigned int _threadID;
+
+
+
+
+};
+
+} //namespace webrtc
+
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
+
+#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc
new file mode 100644
index 0000000..88dc1e6
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc
@@ -0,0 +1,297 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#if defined(CARBON_RENDERING)
+
+#include "video_render_mac_carbon_impl.h"
+#include "critical_section_wrapper.h"
+#include "video_render_agl.h"
+#include "trace.h"
+#include <AGL/agl.h>
+
+namespace webrtc {
+
+VideoRenderMacCarbonImpl::VideoRenderMacCarbonImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+_id(id),
+_renderMacCarbonCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+_fullScreen(fullscreen),
+_ptrWindow(window)
+{
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
+
+}
+
+VideoRenderMacCarbonImpl::~VideoRenderMacCarbonImpl()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
+ delete &_renderMacCarbonCritsect;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::Init()
+{
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
+
+ if (!_ptrWindow)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ // We don't know if the user passed us a WindowRef or a HIViewRef, so test.
+ bool referenceIsValid = false;
+
+ // Check if it's a valid WindowRef
+ //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef before WindowRef cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
+ WindowRef* windowRef = static_cast<WindowRef*>(_ptrWindow);
+ //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef after cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
+ if (IsValidWindowPtr(*windowRef))
+ {
+ _ptrCarbonRender = new VideoRenderAGL(*windowRef, _fullScreen, _id);
+ referenceIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with WindowRef:%x", __FUNCTION__, __LINE__, *windowRef);
+ }
+ else
+ {
+ HIViewRef* hiviewRef = static_cast<HIViewRef*>(_ptrWindow);
+ if (HIViewIsValid(*hiviewRef))
+ {
+ _ptrCarbonRender = new VideoRenderAGL(*hiviewRef, _fullScreen, _id);
+ referenceIsValid = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with HIViewRef:%x", __FUNCTION__, __LINE__, hiviewRef);
+ }
+ }
+
+ if(!referenceIsValid)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef/HIViewRef Returning -1", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ if(!_ptrCarbonRender)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to create an instance of VideoRenderAGL. Returning -1", __FUNCTION__, __LINE__);
+ }
+
+ int retVal = _ptrCarbonRender->Init();
+ if (retVal == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to init CarbonRenderer", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ return -1;
+
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ _id = id;
+
+ if(_ptrCarbonRender)
+ {
+ _ptrCarbonRender->ChangeUniqueID(_id);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::ChangeWindow(void* window)
+{
+ return -1;
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
+
+ if (window == NULL)
+ {
+ return -1;
+ }
+ _ptrWindow = window;
+
+
+ _ptrWindow = window;
+
+ return 0;
+}
+
+VideoRenderCallback*
+VideoRenderMacCarbonImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ VideoChannelAGL* AGLChannel = NULL;
+
+ if(!_ptrWindow)
+ {
+ }
+
+ if(!AGLChannel)
+ {
+ AGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
+ }
+
+ return AGLChannel;
+
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::DeleteIncomingRenderStream(const WebRtc_UWord32 streamId)
+{
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ _ptrCarbonRender->DeleteAGLChannel(streamId);
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ return -1;
+ return _ptrCarbonRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::StartRender()
+{
+ return _ptrCarbonRender->StartRender();
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::StopRender()
+{
+ return _ptrCarbonRender->StopRender();
+}
+
+VideoRenderType
+VideoRenderMacCarbonImpl::RenderType()
+{
+ return kRenderCarbon;
+}
+
+RawVideoType
+VideoRenderMacCarbonImpl::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool
+VideoRenderMacCarbonImpl::FullScreen()
+{
+ return false;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const
+{
+ totalGraphicsMemory = 0;
+ availableGraphicsMemory = 0;
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const
+{
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ //NSScreen* mainScreen = [NSScreen mainScreen];
+
+ //NSRect frame = [mainScreen frame];
+
+ //screenWidth = frame.size.width;
+ //screenHeight = frame.size.height;
+ return 0;
+}
+
+WebRtc_UWord32
+VideoRenderMacCarbonImpl::RenderFrameRate(const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_renderMacCarbonCritsect);
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCarbonImpl::ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCarbonImpl::SetTransparentBackground(const bool enable)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCarbonImpl::SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCarbonImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+
+} //namespace webrtc
+
+#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h
new file mode 100644
index 0000000..3ff3d26
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h
@@ -0,0 +1,148 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#if defined(CARBON_RENDERING)
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
+
+#include "i_video_render.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class VideoRenderAGL;
+
+// Class definitions
+class VideoRenderMacCarbonImpl : IVideoRender
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ VideoRenderMacCarbonImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen);
+
+ virtual ~VideoRenderMacCarbonImpl();
+
+ virtual WebRtc_Word32 Init();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+ virtual VideoRenderCallback* AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32 GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable)
+ {
+ // not supported in Carbon at this time
+ return -1;
+ }
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _renderMacCarbonCritsect;
+ bool _fullScreen;
+ void* _ptrWindow;
+ VideoRenderAGL* _ptrCarbonRender;
+
+};
+
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
+#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h
new file mode 100644
index 0000000..f9216bb
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(COCOA_RENDERING)
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
+
+#include "i_video_render.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class VideoRenderNSOpenGL;
+
+// Class definitions
+class VideoRenderMacCocoaImpl : IVideoRender
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ VideoRenderMacCocoaImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen);
+
+ virtual ~VideoRenderMacCocoaImpl();
+
+ virtual WebRtc_Word32 Init();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+ virtual VideoRenderCallback* AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32 GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable);
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _renderMacCocoaCritsect;
+ bool _fullScreen;
+ void* _ptrWindow;
+ VideoRenderNSOpenGL* _ptrCocoaRender;
+
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
+#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm
new file mode 100644
index 0000000..4698b6e
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm
@@ -0,0 +1,269 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#if defined(COCOA_RENDERING)
+
+#import "cocoa_render_view.h"
+
+#include "video_render_mac_cocoa_impl.h"
+#include "critical_section_wrapper.h"
+#include "video_render_nsopengl.h"
+#include "trace.h"
+
+namespace webrtc {
+
+VideoRenderMacCocoaImpl::VideoRenderMacCocoaImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+_id(id),
+_renderMacCocoaCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+_fullScreen(fullscreen),
+_ptrWindow(window)
+{
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
+}
+
+VideoRenderMacCocoaImpl::~VideoRenderMacCocoaImpl()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
+ delete &_renderMacCocoaCritsect;
+ if (_ptrCocoaRender)
+ {
+ delete _ptrCocoaRender;
+ _ptrCocoaRender = NULL;
+ }
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::Init()
+{
+
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
+
+ // cast ptrWindow from void* to CocoaRenderer. Void* was once NSOpenGLView, and CocoaRenderer is NSOpenGLView.
+ _ptrCocoaRender = new VideoRenderNSOpenGL((CocoaRenderView*)_ptrWindow, _fullScreen, _id);
+ if (!_ptrWindow)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
+ return -1;
+ }
+ int retVal = _ptrCocoaRender->Init();
+ if (retVal == -1)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Failed to init %s:%d", __FUNCTION__, __LINE__);
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ _id = id;
+
+ if(_ptrCocoaRender)
+ {
+ _ptrCocoaRender->ChangeUniqueID(_id);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::ChangeWindow(void* window)
+{
+
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
+
+ if (window == NULL)
+ {
+ return -1;
+ }
+ _ptrWindow = window;
+
+
+ _ptrWindow = window;
+ _ptrCocoaRender->ChangeWindow((CocoaRenderView*)_ptrWindow);
+
+ return 0;
+}
+
+VideoRenderCallback*
+VideoRenderMacCocoaImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
+ VideoChannelNSOpenGL* nsOpenGLChannel = NULL;
+
+ if(!_ptrWindow)
+ {
+ }
+
+ if(!nsOpenGLChannel)
+ {
+ nsOpenGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
+ }
+
+ return nsOpenGLChannel;
+
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::DeleteIncomingRenderStream(const WebRtc_UWord32 streamId)
+{
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ _ptrCocoaRender->DeleteNSGLChannel(streamId);
+
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ return _ptrCocoaRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::StartRender()
+{
+ return _ptrCocoaRender->StartRender();
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::StopRender()
+{
+ return _ptrCocoaRender->StopRender();
+}
+
+VideoRenderType
+VideoRenderMacCocoaImpl::RenderType()
+{
+ return kRenderCocoa;
+}
+
+RawVideoType
+VideoRenderMacCocoaImpl::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool
+VideoRenderMacCocoaImpl::FullScreen()
+{
+ return false;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const
+{
+ totalGraphicsMemory = 0;
+ availableGraphicsMemory = 0;
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const
+{
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ NSScreen* mainScreen = [NSScreen mainScreen];
+
+ NSRect frame = [mainScreen frame];
+
+ screenWidth = frame.size.width;
+ screenHeight = frame.size.height;
+ return 0;
+}
+
+WebRtc_UWord32
+VideoRenderMacCocoaImpl::RenderFrameRate(const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_renderMacCocoaCritsect);
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCocoaImpl::ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32
+VideoRenderMacCocoaImpl::SetTransparentBackground(const bool enable)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCocoaImpl::SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return _ptrCocoaRender->SetText(textId, text, textLength, textColorRef, backgroundColorRef, left, top, right, bottom);
+}
+
+WebRtc_Word32 VideoRenderMacCocoaImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderMacCocoaImpl::FullScreenRender(void* window, const bool enable)
+{
+ return -1;
+}
+
+} //namespace webrtc
+
+#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h
new file mode 100644
index 0000000..cdd3be7
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h
@@ -0,0 +1,193 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#if defined(COCOA_RENDERING)
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
+
+#import <Cocoa/Cocoa.h>
+#import <OpenGL/OpenGL.h>
+#import <OpenGL/glu.h>
+#import <OpenGL/glext.h>
+#include <QuickTime/QuickTime.h>
+#include <list>
+#include <map>
+
+#include "video_render_defines.h"
+
+#import "cocoa_render_view.h"
+#import "cocoa_full_screen_window.h"
+
+class Trace;
+
+namespace webrtc {
+class EventWrapper;
+class ThreadWrapper;
+class VideoRenderNSOpenGL;
+class CriticalSectionWrapper;
+
+class VideoChannelNSOpenGL : public VideoRenderCallback
+{
+
+public:
+
+ VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner);
+ virtual ~VideoChannelNSOpenGL();
+
+ // A new frame is delivered
+ virtual int DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int timeStame90kHz);
+
+ // Called when the incomming frame size and/or number of streams in mix changes
+ virtual int FrameSizeChange(int width, int height, int numberOfStreams);
+
+ virtual int UpdateSize(int width, int height);
+
+ // Setup
+ int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
+
+ // Called when it's time to render the last frame for the channel
+ int RenderOffScreenBuffer();
+
+ // Returns true if a new buffer has been delivered to the texture
+ int IsUpdated(bool& isUpdated);
+ virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
+
+ // ********** new module functions ************ //
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
+
+ // ********** new module helper functions ***** //
+ int ChangeContext(NSOpenGLContext *nsglContext);
+ WebRtc_Word32 GetChannelProperties(float& left,
+ float& top,
+ float& right,
+ float& bottom);
+
+private:
+
+ NSOpenGLContext* _nsglContext;
+ int _id;
+ VideoRenderNSOpenGL* _owner;
+ WebRtc_Word32 _width;
+ WebRtc_Word32 _height;
+ float _startWidth;
+ float _startHeight;
+ float _stopWidth;
+ float _stopHeight;
+ int _stretchedWidth;
+ int _stretchedHeight;
+ int _oldStretchedHeight;
+ int _oldStretchedWidth;
+ int _xOldWidth;
+ int _yOldHeight;
+ unsigned char* _buffer;
+ int _bufferSize;
+ int _incommingBufferSize;
+ bool _bufferIsUpdated;
+ int _numberOfStreams;
+ GLenum _pixelFormat;
+ GLenum _pixelDataType;
+ unsigned int _texture;
+ bool _bVideoSizeStartedChanging;
+};
+
+class VideoRenderNSOpenGL
+{
+
+public: // methods
+ VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId);
+ ~VideoRenderNSOpenGL();
+
+ static int GetOpenGLVersion(int& nsglMajor, int& nsglMinor);
+
+ // Allocates textures
+ int Init();
+ VideoChannelNSOpenGL* CreateNSGLChannel(int streamID, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ VideoChannelNSOpenGL* ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
+ int DeleteNSGLChannel(int channel);
+ int DeleteAllNSGLChannels();
+ int StopThread();
+ bool IsFullScreen();
+ bool HasChannels();
+ bool HasChannel(int channel);
+ int GetChannels(std::list<int>& channelList);
+ void LockAGLCntx();
+ void UnlockAGLCntx();
+
+ // ********** new module functions ************ //
+ int ChangeWindow(CocoaRenderView* newWindowRef);
+ WebRtc_Word32 ChangeUniqueID(WebRtc_Word32 id);
+ WebRtc_Word32 StartRender();
+ WebRtc_Word32 StopRender();
+ WebRtc_Word32 DeleteNSGLChannel(const WebRtc_UWord32 streamID);
+ WebRtc_Word32 GetChannelProperties(const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom);
+
+ WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ // ********** new module helper functions ***** //
+ int configureNSOpenGLEngine();
+ int configureNSOpenGLView();
+ int setRenderTargetWindow();
+ int setRenderTargetFullScreen();
+
+protected: // methods
+ static bool ScreenUpdateThreadProc(void* obj);
+ bool ScreenUpdateProcess();
+ int GetWindowRect(Rect& rect);
+
+private: // methods
+
+ int CreateMixingContext();
+ int RenderOffScreenBuffers();
+ int DisplayBuffers();
+
+private: // variables
+
+
+ CocoaRenderView* _windowRef;
+ bool _fullScreen;
+ int _id;
+ CriticalSectionWrapper& _nsglContextCritSec;
+ ThreadWrapper* _screenUpdateThread;
+ EventWrapper* _screenUpdateEvent;
+ NSOpenGLContext* _nsglContext;
+ NSOpenGLContext* _nsglFullScreenContext;
+ CocoaFullScreenWindow* _fullScreenWindow;
+ Rect _windowRect; // The size of the window
+ int _windowWidth;
+ int _windowHeight;
+ std::map<int, VideoChannelNSOpenGL*> _nsglChannels;
+ std::multimap<int, int> _zOrderToChannel;
+ unsigned int _threadID;
+ bool _renderingIsPaused;
+ NSView* _windowRefSuperView;
+ NSRect _windowRefSuperViewFrame;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
+#endif // COCOA_RENDERING
+
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm
new file mode 100644
index 0000000..4f1a6c5
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm
@@ -0,0 +1,1275 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#if defined(COCOA_RENDERING)
+
+#include "video_render_nsopengl.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc {
+
+VideoChannelNSOpenGL::VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner) :
+_nsglContext( nsglContext),
+_id( iId),
+_owner( owner),
+_width( 0),
+_height( 0),
+_startWidth( 0.0f),
+_startHeight( 0.0f),
+_stopWidth( 0.0f),
+_stopHeight( 0.0f),
+_stretchedWidth( 0),
+_stretchedHeight( 0),
+_oldStretchedHeight( 0),
+_oldStretchedWidth( 0),
+_xOldWidth( 0),
+_yOldHeight( 0),
+_buffer( 0),
+_bufferSize( 0),
+_incommingBufferSize( 0),
+_bufferIsUpdated( false),
+_numberOfStreams( 0),
+_pixelFormat( GL_RGBA),
+_pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
+_texture( 0),
+_bVideoSizeStartedChanging(false)
+
+{
+
+}
+
+VideoChannelNSOpenGL::~VideoChannelNSOpenGL()
+{
+ if (_buffer)
+ {
+ delete [] _buffer;
+ _buffer = NULL;
+ }
+
+ if (_texture != 0)
+ {
+ [_nsglContext makeCurrentContext];
+ glDeleteTextures(1, (const GLuint*) &_texture);
+ _texture = 0;
+ }
+}
+
+int VideoChannelNSOpenGL::ChangeContext(NSOpenGLContext *nsglContext)
+{
+ _owner->UnlockAGLCntx();
+
+ _nsglContext = nsglContext;
+ [_nsglContext makeCurrentContext];
+
+ _owner->UnlockAGLCntx();
+ return 0;
+
+}
+
+WebRtc_Word32 VideoChannelNSOpenGL::GetChannelProperties(float& left,
+ float& top,
+ float& right,
+ float& bottom)
+{
+
+ _owner->LockAGLCntx();
+
+ left = _startWidth;
+ top = _startHeight;
+ right = _stopWidth;
+ bottom = _stopHeight;
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+WebRtc_Word32 VideoChannelNSOpenGL::RenderFrame(const WebRtc_UWord32 /*streamId*/, VideoFrame& videoFrame)
+{
+
+ _owner->LockAGLCntx();
+
+ if(_width != (int)videoFrame.Width() ||
+ _height != (int)videoFrame.Height())
+ {
+ if(FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+ }
+
+ int ret = DeliverFrame(videoFrame.Buffer(), videoFrame.Length(), videoFrame.TimeStamp());
+
+ _owner->UnlockAGLCntx();
+ return ret;
+}
+
+int VideoChannelNSOpenGL::UpdateSize(int width, int height)
+{
+ _owner->LockAGLCntx();
+ _width = width;
+ _height = height;
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
+{
+
+ _owner->LockAGLCntx();
+ _stretchedHeight = stretchHeight;
+ _stretchedWidth = stretchWidth;
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::FrameSizeChange(int width, int height, int numberOfStreams)
+{
+ // We got a new frame size from VideoAPI, prepare the buffer
+
+ _owner->LockAGLCntx();
+
+ if (width == _width && _height == height)
+ {
+ // We already have a correct buffer size
+ _numberOfStreams = numberOfStreams;
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ _width = width;
+ _height = height;
+
+ // Delete the old buffer, create a new one with correct size.
+ if (_buffer)
+ {
+ delete [] _buffer;
+ _bufferSize = 0;
+ }
+
+ _incommingBufferSize = CalcBufferSize(kI420, _width, _height);
+ _bufferSize = CalcBufferSize(kARGB, _width, _height);//_width * _height * bytesPerPixel;
+ _buffer = new unsigned char [_bufferSize];
+ memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
+
+ [_nsglContext makeCurrentContext];
+
+ if(glIsTexture(_texture))
+ {
+ glDeleteTextures(1, (const GLuint*) &_texture);
+ _texture = 0;
+ }
+
+ // Create a new texture
+ glGenTextures(1, (GLuint *) &_texture);
+
+ GLenum glErr = glGetError();
+
+ if (glErr != GL_NO_ERROR)
+ {
+
+ }
+
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
+
+ GLint texSize;
+ glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
+
+ if (texSize < _width || texSize < _height)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ // Set up th texture type and size
+ glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
+ 0, // level
+ GL_RGBA, // internal format
+ _width, // width
+ _height, // height
+ 0, // border 0/1 = off/on
+ _pixelFormat, // format, GL_RGBA
+ _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
+ _buffer); // pixel data
+
+ glErr = glGetError();
+ if (glErr != GL_NO_ERROR)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int /*timeStamp90kHz*/)
+{
+
+ _owner->LockAGLCntx();
+
+ if (_texture == 0)
+ {
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ if (bufferSize != _incommingBufferSize)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ int rgbRet = ConvertFromYV12(buffer, _width,
+ kBGRA, 0, _width, _height,
+ _buffer);
+ if (rgbRet < 0)
+ {
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ [_nsglContext makeCurrentContext];
+
+
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture); // Make sure this texture is the active one
+ GLenum glErr = glGetError();
+ if (glErr != GL_NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "ERROR %d while calling glBindTexture", glErr);
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
+ 0, // Level, not use
+ 0, // start point x, (low left of pic)
+ 0, // start point y,
+ _width, // width
+ _height, // height
+ _pixelFormat, // pictue format for _buffer
+ _pixelDataType, // data type of _buffer
+ (const GLvoid*) _buffer); // the pixel data
+
+ glErr = glGetError();
+ if (glErr != GL_NO_ERROR)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "ERROR %d while calling glTexSubImage2d", glErr);
+ _owner->UnlockAGLCntx();
+ return -1;
+ }
+
+ _bufferIsUpdated = true;
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::RenderOffScreenBuffer()
+{
+
+ _owner->LockAGLCntx();
+
+ if (_texture == 0)
+ {
+ _owner->UnlockAGLCntx();
+ return 0;
+ }
+
+ // if(_fullscreen)
+ // {
+ // NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
+ // _width = mainDisplayRect.size.width;
+ // _height = mainDisplayRect.size.height;
+ // glViewport(0, 0, mainDisplayRect.size.width, mainDisplayRect.size.height);
+ // float newX = mainDisplayRect.size.width/_width;
+ // float newY = mainDisplayRect.size.height/_height;
+
+ // convert from 0.0 <= size <= 1.0 to
+ // open gl world -1.0 < size < 1.0
+ GLfloat xStart = 2.0f * _startWidth - 1.0f;
+ GLfloat xStop = 2.0f * _stopWidth - 1.0f;
+ GLfloat yStart = 1.0f - 2.0f * _stopHeight;
+ GLfloat yStop = 1.0f - 2.0f * _startHeight;
+
+ [_nsglContext makeCurrentContext];
+
+ glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
+ _oldStretchedHeight = _stretchedHeight;
+ _oldStretchedWidth = _stretchedWidth;
+
+ glLoadIdentity();
+ glEnable(GL_TEXTURE_RECTANGLE_EXT);
+ glBegin(GL_POLYGON);
+ {
+ glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
+ glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
+ glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
+ glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
+ }
+ glEnd();
+
+ glDisable(GL_TEXTURE_RECTANGLE_EXT);
+
+ _bufferIsUpdated = false;
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::IsUpdated(bool& isUpdated)
+{
+ _owner->LockAGLCntx();
+
+ isUpdated = _bufferIsUpdated;
+
+ _owner->UnlockAGLCntx();
+ return 0;
+}
+
+int VideoChannelNSOpenGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+ _owner->LockAGLCntx();
+
+ _startWidth = startWidth;
+ _stopWidth = stopWidth;
+ _startHeight = startHeight;
+ _stopHeight = stopHeight;
+
+ int oldWidth = _width;
+ int oldHeight = _height;
+ int oldNumberOfStreams = _numberOfStreams;
+
+ _width = 0;
+ _height = 0;
+
+ int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
+
+ _owner->UnlockAGLCntx();
+ return retVal;
+}
+
+int VideoChannelNSOpenGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
+{
+ return -1;
+}
+
+/*
+ *
+ * VideoRenderNSOpenGL
+ *
+ */
+
+VideoRenderNSOpenGL::VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId) :
+_windowRef( (CocoaRenderView*)windowRef),
+_fullScreen( fullScreen),
+_id( iId),
+_nsglContextCritSec( *CriticalSectionWrapper::CreateCriticalSection()),
+_screenUpdateThread( 0),
+_screenUpdateEvent( 0),
+_nsglContext( 0),
+_nsglFullScreenContext( 0),
+_fullScreenWindow( nil),
+_windowRect( ),
+_windowWidth( 0),
+_windowHeight( 0),
+_nsglChannels( ),
+_zOrderToChannel( ),
+_threadID (0),
+_renderingIsPaused (FALSE),
+_windowRefSuperView(NULL),
+_windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
+{
+ _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
+ _screenUpdateEvent = EventWrapper::Create();
+}
+
+int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
+{
+
+ LockAGLCntx();
+
+ _windowRef = newWindowRef;
+
+ if(CreateMixingContext() == -1)
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ int error = 0;
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
+ while (it!= _nsglChannels.end())
+ {
+ error |= (it->second)->ChangeContext(_nsglContext);
+ it++;
+ }
+ if(error != 0)
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+/* Check if the thread and event already exist.
+ * If so then they will simply be restarted
+ * If not then create them and continue
+ */
+WebRtc_Word32 VideoRenderNSOpenGL::StartRender()
+{
+
+ LockAGLCntx();
+
+ const unsigned int MONITOR_FREQ = 60;
+ if(TRUE == _renderingIsPaused)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
+
+ // we already have the thread. Most likely StopRender() was called and they were paused
+ if(FALSE == _screenUpdateThread->Start(_threadID) ||
+ FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ UnlockAGLCntx();
+ return 0;
+ }
+
+
+ if (!_screenUpdateThread)
+ {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "failed start screenUpdateThread");
+ UnlockAGLCntx();
+ return -1;
+ }
+
+
+ UnlockAGLCntx();
+ return 0;
+}
+WebRtc_Word32 VideoRenderNSOpenGL::StopRender()
+{
+
+ LockAGLCntx();
+
+ /* The code below is functional
+ * but it pauses for several seconds
+ */
+
+ // pause the update thread and the event timer
+ if(!_screenUpdateThread || !_screenUpdateEvent)
+ {
+ _renderingIsPaused = TRUE;
+
+ UnlockAGLCntx();
+ return 0;
+ }
+
+ if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
+ {
+ _renderingIsPaused = FALSE;
+
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ _renderingIsPaused = TRUE;
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderNSOpenGL::configureNSOpenGLView()
+{
+ return 0;
+
+}
+
+int VideoRenderNSOpenGL::configureNSOpenGLEngine()
+{
+
+ LockAGLCntx();
+
+ // Disable not needed functionality to increase performance
+ glDisable(GL_DITHER);
+ glDisable(GL_ALPHA_TEST);
+ glDisable(GL_STENCIL_TEST);
+ glDisable(GL_FOG);
+ glDisable(GL_TEXTURE_2D);
+ glPixelZoom(1.0, 1.0);
+ glDisable(GL_BLEND);
+ glDisable(GL_DEPTH_TEST);
+ glDepthMask(GL_FALSE);
+ glDisable(GL_CULL_FACE);
+
+ // Set texture parameters
+ glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+ glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
+ glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
+ glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
+
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+
+ if (_windowWidth != (_windowRect.right - _windowRect.left)
+ || _windowHeight != (_windowRect.bottom - _windowRect.top))
+ {
+ _windowWidth = _windowRect.right - _windowRect.left;
+ _windowHeight = _windowRect.bottom - _windowRect.top;
+ }
+ glViewport(0, 0, _windowWidth, _windowHeight);
+
+ // Synchronize buffer swaps with vertical refresh rate
+ GLint swapInt = 1;
+ [_nsglContext setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderNSOpenGL::setRenderTargetWindow()
+{
+ LockAGLCntx();
+
+
+ GLuint attribs[] =
+ {
+ NSOpenGLPFAColorSize, 24,
+ NSOpenGLPFAAlphaSize, 8,
+ NSOpenGLPFADepthSize, 16,
+ NSOpenGLPFAAccelerated,
+ 0
+ };
+
+ NSOpenGLPixelFormat* fmt = [[NSOpenGLPixelFormat alloc] initWithAttributes: (NSOpenGLPixelFormatAttribute*) attribs];
+
+ if(_windowRef)
+ {
+ [_windowRef initCocoaRenderView:fmt];
+ }
+ else
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ [fmt release];
+
+ _nsglContext = [_windowRef nsOpenGLContext];
+ [_nsglContext makeCurrentContext];
+
+ glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
+ glClear(GL_COLOR_BUFFER_BIT);
+
+
+ DisplayBuffers();
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderNSOpenGL::setRenderTargetFullScreen()
+{
+ LockAGLCntx();
+
+
+ GLuint attribs[] =
+ {
+ NSOpenGLPFAColorSize, 24,
+ NSOpenGLPFAAlphaSize, 8,
+ NSOpenGLPFADepthSize, 16,
+ NSOpenGLPFAAccelerated,
+ 0
+ };
+
+ NSOpenGLPixelFormat* fmt = [[NSOpenGLPixelFormat alloc] initWithAttributes: (NSOpenGLPixelFormatAttribute*) attribs];
+
+ // Store original superview and frame for use when exiting full screens
+ _windowRefSuperViewFrame = [_windowRef frame];
+ _windowRefSuperView = [_windowRef superview];
+
+
+ // create new fullscreen window
+ NSRect screenRect = [[NSScreen mainScreen]frame];
+ [_windowRef setFrame:screenRect];
+ [_windowRef setBounds:screenRect];
+
+
+ _fullScreenWindow = [[CocoaFullScreenWindow alloc]init];
+ [_fullScreenWindow grabFullScreen];
+ [[[_fullScreenWindow window] contentView] addSubview:_windowRef];
+
+ if(_windowRef)
+ {
+ [_windowRef initCocoaRenderViewFullScreen:fmt];
+ }
+ else
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ [fmt release];
+
+ _nsglContext = [_windowRef nsOpenGLContext];
+ [_nsglContext makeCurrentContext];
+
+ glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
+ glClear(GL_COLOR_BUFFER_BIT);
+
+ DisplayBuffers();
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
+{
+
+ if(_fullScreen)
+ {
+ if(_fullScreenWindow)
+ {
+ // Detach CocoaRenderView from full screen view back to
+ // it's original parent.
+ [_windowRef removeFromSuperview];
+ if(_windowRefSuperView)
+ {
+ [_windowRefSuperView addSubview:_windowRef];
+ [_windowRef setFrame:_windowRefSuperViewFrame];
+ }
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, 0, "%s:%d Attempting to release fullscreen window", __FUNCTION__, __LINE__);
+ [_fullScreenWindow releaseFullScreen];
+
+ }
+ }
+
+ // Signal event to exit thread, then delete it
+ ThreadWrapper* tmpPtr = _screenUpdateThread;
+ _screenUpdateThread = NULL;
+
+ if (tmpPtr)
+ {
+ tmpPtr->SetNotAlive();
+ _screenUpdateEvent->Set();
+ _screenUpdateEvent->StopTimer();
+
+ if (tmpPtr->Stop())
+ {
+ delete tmpPtr;
+ }
+ delete _screenUpdateEvent;
+ _screenUpdateEvent = NULL;
+ }
+
+ if (_nsglContext != 0)
+ {
+ [_nsglContext makeCurrentContext];
+ _nsglContext = nil;
+ }
+
+ // Delete all channels
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
+ while (it!= _nsglChannels.end())
+ {
+ delete it->second;
+ _nsglChannels.erase(it);
+ it = _nsglChannels.begin();
+ }
+ _nsglChannels.clear();
+
+ // Clean the zOrder map
+ std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
+ while(zIt != _zOrderToChannel.end())
+ {
+ _zOrderToChannel.erase(zIt);
+ zIt = _zOrderToChannel.begin();
+ }
+ _zOrderToChannel.clear();
+
+}
+
+/* static */
+int VideoRenderNSOpenGL::GetOpenGLVersion(int& /*nsglMajor*/, int& /*nsglMinor*/)
+{
+ return -1;
+}
+
+int VideoRenderNSOpenGL::Init()
+{
+
+ LockAGLCntx();
+ if (!_screenUpdateThread)
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ _screenUpdateThread->Start(_threadID);
+
+ // Start the event triggering the render process
+ unsigned int monitorFreq = 60;
+ _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
+
+ if (CreateMixingContext() == -1)
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+VideoChannelNSOpenGL* VideoRenderNSOpenGL::CreateNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ if (HasChannel(channel))
+ {
+ return NULL;
+ }
+
+ if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
+ {
+
+ }
+
+ VideoChannelNSOpenGL* newAGLChannel = new VideoChannelNSOpenGL(_nsglContext, _id, this);
+ if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
+ {
+ if (newAGLChannel)
+ {
+ delete newAGLChannel;
+ newAGLChannel = NULL;
+ }
+
+ return NULL;
+ }
+
+ _nsglChannels[channel] = newAGLChannel;
+ _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s successfully created NSGL channel number %d", __FUNCTION__, channel);
+
+ return newAGLChannel;
+}
+
+int VideoRenderNSOpenGL::DeleteAllNSGLChannels()
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator it;
+ it = _nsglChannels.begin();
+
+ while (it != _nsglChannels.end())
+ {
+ VideoChannelNSOpenGL* channel = it->second;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Deleting channel %d", __FUNCTION__, channel);
+ delete channel;
+ it++;
+ }
+ _nsglChannels.clear();
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderNSOpenGL::DeleteNSGLChannel(const WebRtc_UWord32 channel)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator it;
+ it = _nsglChannels.find(channel);
+ if (it != _nsglChannels.end())
+ {
+ delete it->second;
+ _nsglChannels.erase(it);
+ }
+ else
+ {
+ return -1;
+ }
+
+ std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
+ while( zIt != _zOrderToChannel.end())
+ {
+ if (zIt->second == (int)channel)
+ {
+ _zOrderToChannel.erase(zIt);
+ break;
+ }
+ zIt++;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderNSOpenGL::GetChannelProperties(const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ bool channelFound = false;
+
+ // Loop through all channels until we find a match.
+ // From that, get zorder.
+ // From that, get T, L, R, B
+ for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
+ rIt != _zOrderToChannel.rend();
+ rIt++)
+ {
+ if(streamId == rIt->second)
+ {
+ channelFound = true;
+
+ zOrder = rIt->second;
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator rIt = _nsglChannels.find(streamId);
+ VideoChannelNSOpenGL* tempChannel = rIt->second;
+
+ if(-1 == tempChannel->GetChannelProperties(left, top, right, bottom) )
+ {
+ return -1;
+ }
+ break;
+ }
+ }
+
+ if(false == channelFound)
+ {
+
+ return -1;
+ }
+
+ return 0;
+}
+
+int VideoRenderNSOpenGL::StopThread()
+{
+
+ ThreadWrapper* tmpPtr = _screenUpdateThread;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Stopping thread ", __FUNCTION__, _screenUpdateThread);
+ _screenUpdateThread = NULL;
+
+ if (tmpPtr)
+ {
+ tmpPtr->SetNotAlive();
+ _screenUpdateEvent->Set();
+ if (tmpPtr->Stop())
+ {
+ delete tmpPtr;
+ }
+ }
+
+ delete _screenUpdateEvent;
+ _screenUpdateEvent = NULL;
+
+ return 0;
+}
+
+bool VideoRenderNSOpenGL::IsFullScreen()
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+ return _fullScreen;
+}
+
+bool VideoRenderNSOpenGL::HasChannels()
+{
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ if (_nsglChannels.begin() != _nsglChannels.end())
+ {
+ return true;
+ }
+ return false;
+}
+
+bool VideoRenderNSOpenGL::HasChannel(int channel)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
+
+ if (it != _nsglChannels.end())
+ {
+ return true;
+ }
+ return false;
+}
+
+int VideoRenderNSOpenGL::GetChannels(std::list<int>& channelList)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
+
+ while (it != _nsglChannels.end())
+ {
+ channelList.push_back(it->first);
+ it++;
+ }
+
+ return 0;
+}
+
+VideoChannelNSOpenGL* VideoRenderNSOpenGL::ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
+
+ if (it != _nsglChannels.end())
+ {
+ VideoChannelNSOpenGL* aglChannel = it->second;
+ if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s failed to set stream settings: channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
+ __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
+ return NULL;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Configuring channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
+ __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
+
+ std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
+ while(it != _zOrderToChannel.end())
+ {
+ if (it->second == channel)
+ {
+ if (it->first != zOrder)
+ {
+ _zOrderToChannel.erase(it);
+ _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
+ }
+ break;
+ }
+ it++;
+ }
+ return aglChannel;
+ }
+
+ return NULL;
+}
+
+/*
+ *
+ * Rendering process
+ *
+ */
+
+bool VideoRenderNSOpenGL::ScreenUpdateThreadProc(void* obj)
+{
+ return static_cast<VideoRenderNSOpenGL*>(obj)->ScreenUpdateProcess();
+}
+
+bool VideoRenderNSOpenGL::ScreenUpdateProcess()
+{
+
+ _screenUpdateEvent->Wait(10);
+ LockAGLCntx();
+
+ if (!_screenUpdateThread)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s no screen update thread", __FUNCTION__);
+ UnlockAGLCntx();
+ return false;
+ }
+
+ [_nsglContext makeCurrentContext];
+
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+
+ if (_windowWidth != (_windowRect.right - _windowRect.left)
+ || _windowHeight != (_windowRect.bottom - _windowRect.top))
+ {
+ _windowWidth = _windowRect.right - _windowRect.left;
+ _windowHeight = _windowRect.bottom - _windowRect.top;
+ glViewport(0, 0, _windowWidth, _windowHeight);
+ }
+
+ // Check if there are any updated buffers
+ bool updated = false;
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
+ while (it != _nsglChannels.end())
+ {
+
+ VideoChannelNSOpenGL* aglChannel = it->second;
+ aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
+ aglChannel->IsUpdated(updated);
+ if (updated)
+ {
+ break;
+ }
+ it++;
+ }
+
+ if (updated)
+ {
+
+ // At least on buffers is updated, we need to repaint the texture
+ if (RenderOffScreenBuffers() != -1)
+ {
+ UnlockAGLCntx();
+ return true;
+ }
+ }
+ // }
+ UnlockAGLCntx();
+ return true;
+}
+
+/*
+ *
+ * Functions for creating mixing buffers and screen settings
+ *
+ */
+
+int VideoRenderNSOpenGL::CreateMixingContext()
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ if(_fullScreen)
+ {
+ if(-1 == setRenderTargetFullScreen())
+ {
+ return -1;
+ }
+ }
+ else
+ {
+
+ if(-1 == setRenderTargetWindow())
+ {
+ return -1;
+ }
+ }
+
+ configureNSOpenGLEngine();
+
+ DisplayBuffers();
+
+ GLenum glErr = glGetError();
+ if (glErr)
+ {
+ }
+
+ return 0;
+}
+
+/*
+ *
+ * Rendering functions
+ *
+ */
+
+int VideoRenderNSOpenGL::RenderOffScreenBuffers()
+{
+ LockAGLCntx();
+
+ // Get the current window size, it might have changed since last render.
+ if (GetWindowRect(_windowRect) == -1)
+ {
+ UnlockAGLCntx();
+ return -1;
+ }
+
+ [_nsglContext makeCurrentContext];
+ glClear(GL_COLOR_BUFFER_BIT);
+
+ // Loop through all channels starting highest zOrder ending with lowest.
+ for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
+ rIt != _zOrderToChannel.rend();
+ rIt++)
+ {
+ int channelId = rIt->second;
+ std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channelId);
+
+ VideoChannelNSOpenGL* aglChannel = it->second;
+
+ aglChannel->RenderOffScreenBuffer();
+ }
+
+ DisplayBuffers();
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+/*
+ *
+ * Help functions
+ *
+ * All help functions assumes external protections
+ *
+ */
+
+int VideoRenderNSOpenGL::DisplayBuffers()
+{
+
+ LockAGLCntx();
+
+ glFinish();
+ [_nsglContext flushBuffer];
+
+ WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s glFinish and [_nsglContext flushBuffer]", __FUNCTION__);
+
+ UnlockAGLCntx();
+ return 0;
+}
+
+int VideoRenderNSOpenGL::GetWindowRect(Rect& rect)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+
+ if (_windowRef)
+ {
+ if(_fullScreen)
+ {
+ NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
+ rect.bottom = 0;
+ rect.left = 0;
+ rect.right = mainDisplayRect.size.width;
+ rect.top = mainDisplayRect.size.height;
+ }
+ else
+ {
+ rect.top = [_windowRef frame].origin.y;
+ rect.left = [_windowRef frame].origin.x;
+ rect.bottom = [_windowRef frame].origin.y + [_windowRef frame].size.height;
+ rect.right = [_windowRef frame].origin.x + [_windowRef frame].size.width;
+ }
+
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+WebRtc_Word32 VideoRenderNSOpenGL::ChangeUniqueID(WebRtc_Word32 id)
+{
+
+ CriticalSectionScoped cs(_nsglContextCritSec);
+ _id = id;
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderNSOpenGL::SetText(const WebRtc_UWord8 /*textId*/,
+ const WebRtc_UWord8* /*text*/,
+ const WebRtc_Word32 /*textLength*/,
+ const WebRtc_UWord32 /*textColorRef*/,
+ const WebRtc_UWord32 /*backgroundColorRef*/,
+ const float /*left*/,
+ const float /*top*/,
+ const float /*right*/,
+ const float /*bottom*/)
+{
+
+ return 0;
+
+}
+
+void VideoRenderNSOpenGL::LockAGLCntx()
+{
+ _nsglContextCritSec.Enter();
+}
+void VideoRenderNSOpenGL::UnlockAGLCntx()
+{
+ _nsglContextCritSec.Leave();
+}
+
+/*
+
+ bool VideoRenderNSOpenGL::SetFullScreen(bool fullscreen)
+ {
+ NSRect mainDisplayRect, viewRect;
+
+ // Create a screen-sized window on the display you want to take over
+ // Note, mainDisplayRect has a non-zero origin if the key window is on a secondary display
+ mainDisplayRect = [[NSScreen mainScreen] frame];
+ fullScreenWindow = [[NSWindow alloc] initWithContentRect:mainDisplayRect styleMask:NSBorderlessWindowMask
+ backing:NSBackingStoreBuffered defer:YES];
+
+ // Set the window level to be above the menu bar
+ [fullScreenWindow setLevel:NSMainMenuWindowLevel+1];
+
+ // Perform any other window configuration you desire
+ [fullScreenWindow setOpaque:YES];
+ [fullScreenWindow setHidesOnDeactivate:YES];
+
+ // Create a view with a double-buffered OpenGL context and attach it to the window
+ // By specifying the non-fullscreen context as the shareContext, we automatically inherit the OpenGL objects (textures, etc) it has defined
+ viewRect = NSMakeRect(0.0, 0.0, mainDisplayRect.size.width, mainDisplayRect.size.height);
+ fullScreenView = [[MyOpenGLView alloc] initWithFrame:viewRect shareContext:[openGLView openGLContext]];
+ [fullScreenWindow setContentView:fullScreenView];
+
+ // Show the window
+ [fullScreenWindow makeKeyAndOrderFront:self];
+
+ // Set the scene with the full-screen viewport and viewing transformation
+ [scene setViewportRect:viewRect];
+
+ // Assign the view's MainController to self
+ [fullScreenView setMainController:self];
+
+ if (!isAnimating) {
+ // Mark the view as needing drawing to initalize its contents
+ [fullScreenView setNeedsDisplay:YES];
+ }
+ else {
+ // Start playing the animation
+ [fullScreenView startAnimation];
+ }
+
+ }
+
+
+
+ */
+
+
+} //namespace webrtc
+
+#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/video_render.gypi b/trunk/src/modules/video_render/main/source/video_render.gypi
new file mode 100644
index 0000000..7a7545e
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/video_render.gypi
@@ -0,0 +1,195 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'video_render_module',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'webrtc_utility',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '.',
+ '../interface',
+ '../../../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ '../../../interface',
+ ],
+ },
+ 'sources': [
+ # interfaces
+ '../interface/video_render.h',
+ '../interface/video_render_defines.h',
+
+ # headers
+ 'incoming_video_stream.h',
+ 'video_render_frames.h',
+ 'video_render_impl.h',
+ 'i_video_render.h',
+ # Linux
+ 'linux/video_render_linux_impl.h',
+ 'linux/video_x11_channel.h',
+ 'linux/video_x11_render.h',
+ # Mac
+ 'mac/cocoa_full_screen_window.h',
+ 'mac/cocoa_render_view.h',
+ 'mac/video_render_agl.h',
+ 'mac/video_render_mac_carbon_impl.h',
+ 'mac/video_render_mac_cocoa_impl.h',
+ 'mac/video_render_nsopengl.h',
+ # Windows
+ 'windows/i_video_render_win.h',
+ 'windows/video_render_direct3d9.h',
+ 'windows/video_render_directdraw.h',
+ 'windows/video_render_windows_impl.h',
+ # External
+ 'external/video_render_external_impl.h',
+
+ # PLATFORM INDEPENDENT SOURCE FILES
+ 'incoming_video_stream.cc',
+ 'video_render_frames.cc',
+ 'video_render_impl.cc',
+ # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
+ # Linux
+ 'linux/video_render_linux_impl.cc',
+ 'linux/video_x11_channel.cc',
+ 'linux/video_x11_render.cc',
+ # Mac
+ 'mac/video_render_nsopengl.mm',
+ 'mac/video_render_mac_cocoa_impl.mm',
+ 'mac/video_render_agl.cc',
+ 'mac/video_render_mac_carbon_impl.cc',
+ 'mac/cocoa_render_view.mm',
+ 'mac/cocoa_full_screen_window.mm',
+ # Windows
+ 'windows/video_render_direct3d9.cc',
+ 'windows/video_render_directdraw.cc',
+ 'windows/video_render_windows_impl.cc',
+ # External
+ 'external/video_render_external_impl.cc',
+ ],
+ # TODO(andrew): with the proper suffix, these files will be excluded
+ # automatically.
+ 'conditions': [
+ ['include_internal_video_render==1', {
+ 'defines': [
+ 'WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER',
+ ],
+ }],
+ ['OS!="linux" or include_internal_video_render==0', {
+ 'sources!': [
+ 'linux/video_render_linux_impl.h',
+ 'linux/video_x11_channel.h',
+ 'linux/video_x11_render.h',
+ 'linux/video_render_linux_impl.cc',
+ 'linux/video_x11_channel.cc',
+ 'linux/video_x11_render.cc',
+ ],
+ }],
+ ['OS!="mac" or include_internal_video_render==0', {
+ 'sources!': [
+ 'mac/cocoa_full_screen_window.h',
+ 'mac/cocoa_render_view.h',
+ 'mac/video_render_agl.h',
+ 'mac/video_render_mac_carbon_impl.h',
+ 'mac/video_render_mac_cocoa_impl.h',
+ 'mac/video_render_nsopengl.h',
+ 'mac/video_render_nsopengl.mm',
+ 'mac/video_render_mac_cocoa_impl.mm',
+ 'mac/video_render_agl.cc',
+ 'mac/video_render_mac_carbon_impl.cc',
+ 'mac/cocoa_render_view.mm',
+ 'mac/cocoa_full_screen_window.mm',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'mac',
+ ],
+ },
+ }],
+ ['OS!="win" or include_internal_video_render==0', {
+ 'sources!': [
+ 'windows/i_video_render_win.h',
+ 'windows/video_render_direct3d9.h',
+ 'windows/video_render_directdraw.h',
+ 'windows/video_render_windows_impl.h',
+ 'windows/video_render_direct3d9.cc',
+ 'windows/video_render_directdraw.cc',
+ 'windows/video_render_windows_impl.cc',
+ ],
+ }],
+ ] # conditions
+ }, # video_render_module
+ ], # targets
+
+ # Exclude the test target when building with chromium.
+ 'conditions': [
+ ['include_internal_video_render==1', {
+ 'defines': [
+ 'WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER',
+ ],
+ }],
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'video_render_module_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_render_module',
+ 'webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+ 'sources': [
+ '../test/testAPI/testAPI.cc',
+ '../test/testAPI/testAPI.h',
+ '../test/testAPI/testAPI_mac.mm',
+ ],
+ 'conditions': [
+ ['OS=="mac" or OS=="linux"', {
+ 'cflags': [
+ '-Wno-write-strings',
+ ],
+ 'ldflags': [
+ '-lpthread -lm',
+ ],
+ }],
+ ['OS=="linux"', {
+ 'libraries': [
+ '-lrt',
+ '-lXext',
+ '-lX11',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ 'OTHER_LDFLAGS': [
+ '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL',
+ ],
+ },
+ }],
+ ] # conditions
+ }, # video_render_module_test
+ ], # targets
+ }], # build_with_chromium==0
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_render/main/source/video_render_frames.cc b/trunk/src/modules/video_render/main/source/video_render_frames.cc
new file mode 100644
index 0000000..28078e3
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/video_render_frames.cc
@@ -0,0 +1,209 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_frames.h"
+#include "module_common_types.h"
+#include "tick_util.h"
+#include "trace.h"
+#include <cassert>
+
+namespace webrtc {
+
+VideoRenderFrames::VideoRenderFrames() :
+ _incomingFrames(), _renderDelayMs(10)
+{
+}
+
+VideoRenderFrames::~VideoRenderFrames()
+{
+ ReleaseAllFrames();
+}
+
+WebRtc_Word32 VideoRenderFrames::AddFrame(VideoFrame* ptrNewFrame)
+{
+ const WebRtc_Word64 timeNow = TickTime::MillisecondTimestamp();
+
+ if (ptrNewFrame->RenderTimeMs() + KOldRenderTimestampMS < timeNow)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
+ "%s: too old frame.", __FUNCTION__);
+ return -1;
+ }
+ if (ptrNewFrame->RenderTimeMs() > timeNow + KFutureRenderTimestampMS)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
+ "%s: frame too long into the future.", __FUNCTION__);
+ return -1;
+ }
+
+ // Get an empty frame
+ VideoFrame* ptrFrameToAdd = NULL;
+ if (!_emptyFrames.Empty())
+ {
+ ListItem* item = _emptyFrames.First();
+ if (item)
+ {
+ ptrFrameToAdd = static_cast<VideoFrame*> (item->GetItem());
+ _emptyFrames.Erase(item);
+ }
+ }
+ if (!ptrFrameToAdd)
+ {
+
+ if (_emptyFrames.GetSize() + _incomingFrames.GetSize()
+ > KMaxNumberOfFrames)
+ {
+ // Already allocated toom many frames...
+ WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer,
+ -1, "%s: too many frames, limit: %d", __FUNCTION__,
+ KMaxNumberOfFrames);
+ return -1;
+ }
+
+ // Allocate new memory
+ WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1,
+ "%s: allocating buffer %d", __FUNCTION__,
+ _emptyFrames.GetSize() + _incomingFrames.GetSize());
+
+ ptrFrameToAdd = new VideoFrame();
+ if (!ptrFrameToAdd)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "%s: could not create new frame for", __FUNCTION__);
+ return -1;
+ }
+ }
+
+ ptrFrameToAdd->VerifyAndAllocate(ptrNewFrame->Length());
+ ptrFrameToAdd->SwapFrame(const_cast<VideoFrame&> (*ptrNewFrame)); //remove const ness. Copying will be costly.
+ _incomingFrames.PushBack(ptrFrameToAdd);
+
+ return _incomingFrames.GetSize();
+}
+
+VideoFrame*
+VideoRenderFrames::FrameToRender()
+{
+ VideoFrame* ptrRenderFrame = NULL;
+ while (!_incomingFrames.Empty())
+ {
+ ListItem* item = _incomingFrames.First();
+ if (item)
+ {
+ VideoFrame* ptrOldestFrameInList =
+ static_cast<VideoFrame*> (item->GetItem());
+ if (ptrOldestFrameInList->RenderTimeMs()
+ <= TickTime::MillisecondTimestamp() + _renderDelayMs)
+ {
+ // This is the oldest one so far and it's ok to render
+ if (ptrRenderFrame)
+ {
+ // This one is older than the newly found frame, remove this one.
+ ptrRenderFrame->SetWidth(0);
+ ptrRenderFrame->SetHeight(0);
+ ptrRenderFrame->SetLength(0);
+ ptrRenderFrame->SetRenderTime(0);
+ ptrRenderFrame->SetTimeStamp(0);
+ _emptyFrames.PushFront(ptrRenderFrame);
+ }
+ ptrRenderFrame = ptrOldestFrameInList;
+ _incomingFrames.Erase(item);
+ }
+ else
+ {
+ // We can't release this one yet, we're done here.
+ break;
+ }
+ }
+ else
+ {
+ assert(false);
+ }
+ }
+ return ptrRenderFrame;
+}
+
+WebRtc_Word32 VideoRenderFrames::ReturnFrame(VideoFrame* ptrOldFrame)
+{
+ ptrOldFrame->SetWidth(0);
+ ptrOldFrame->SetHeight(0);
+ ptrOldFrame->SetRenderTime(0);
+ ptrOldFrame->SetLength(0);
+ _emptyFrames.PushBack(ptrOldFrame);
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderFrames::ReleaseAllFrames()
+{
+ while (!_incomingFrames.Empty())
+ {
+ ListItem* item = _incomingFrames.First();
+ if (item)
+ {
+ VideoFrame* ptrFrame =
+ static_cast<VideoFrame*> (item->GetItem());
+ assert(ptrFrame != NULL);
+ ptrFrame->Free();
+ delete ptrFrame;
+ }
+ _incomingFrames.Erase(item);
+ }
+ while (!_emptyFrames.Empty())
+ {
+ ListItem* item = _emptyFrames.First();
+ if (item)
+ {
+ VideoFrame* ptrFrame =
+ static_cast<VideoFrame*> (item->GetItem());
+ assert(ptrFrame != NULL);
+ ptrFrame->Free();
+ delete ptrFrame;
+ }
+ _emptyFrames.Erase(item);
+ }
+ return 0;
+}
+
+WebRtc_Word32 KEventMaxWaitTimeMs = 200;
+
+WebRtc_UWord32 VideoRenderFrames::TimeToNextFrameRelease()
+{
+ WebRtc_Word64 timeToRelease = 0;
+ ListItem* item = _incomingFrames.First();
+ if (item)
+ {
+ VideoFrame* oldestFrame =
+ static_cast<VideoFrame*> (item->GetItem());
+ timeToRelease = oldestFrame->RenderTimeMs() - _renderDelayMs
+ - TickTime::MillisecondTimestamp();
+ if (timeToRelease < 0)
+ {
+ timeToRelease = 0;
+ }
+ }
+ else
+ {
+ timeToRelease = KEventMaxWaitTimeMs;
+ }
+
+ return (WebRtc_UWord32) timeToRelease;
+}
+
+//
+WebRtc_Word32 VideoRenderFrames::SetRenderDelay(
+ const WebRtc_UWord32 renderDelay)
+{
+ _renderDelayMs = renderDelay;
+ return 0;
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/video_render_frames.h b/trunk/src/modules/video_render/main/source/video_render_frames.h
new file mode 100644
index 0000000..84a2a1c
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/video_render_frames.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
+
+#include "list_wrapper.h"
+#include "video_render.h"
+
+namespace webrtc {
+
+// Class definitions
+class VideoRenderFrames
+{
+public:
+ VideoRenderFrames();
+ ~VideoRenderFrames();
+
+ /*
+ * Add a frame to the render queue
+ */
+ WebRtc_Word32 AddFrame(VideoFrame* ptrNewFrame);
+
+ /*
+ * Get a frame for rendering, if it's time to render.
+ */
+ VideoFrame* FrameToRender();
+
+ /*
+ * Return an old frame
+ */
+ WebRtc_Word32 ReturnFrame(VideoFrame* ptrOldFrame);
+
+ /*
+ * Releases all frames
+ */
+ WebRtc_Word32 ReleaseAllFrames();
+
+ /*
+ * Returns the number of ms to next frame to render
+ */
+ WebRtc_UWord32 TimeToNextFrameRelease();
+
+ /*
+ * Sets estimates delay in renderer
+ */
+ WebRtc_Word32 SetRenderDelay(const WebRtc_UWord32 renderDelay);
+
+private:
+ enum
+ {
+ KMaxNumberOfFrames = 300
+ }; // 10 seconds for 30 fps.
+ enum
+ {
+ KOldRenderTimestampMS = 500
+ }; //Don't render frames with timestamp older than 500ms from now.
+ enum
+ {
+ KFutureRenderTimestampMS = 10000
+ }; //Don't render frames with timestamp more than 10s into the future.
+
+ ListWrapper _incomingFrames; // Sorted oldest video frame first
+ ListWrapper _emptyFrames; // Empty frames
+
+ WebRtc_UWord32 _renderDelayMs; // Set render delay
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
diff --git a/trunk/src/modules/video_render/main/source/video_render_impl.cc b/trunk/src/modules/video_render/main/source/video_render_impl.cc
new file mode 100644
index 0000000..616d03b
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/video_render_impl.cc
@@ -0,0 +1,981 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_impl.h"
+#include "engine_configurations.h"
+#include "critical_section_wrapper.h"
+#include "video_render_defines.h"
+#include "trace.h"
+#include "incoming_video_stream.h"
+#include "i_video_render.h"
+
+#include <cassert>
+
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+#if defined (_WIN32)
+#include "windows/video_render_windows_impl.h"
+#define STANDARD_RENDERING kRenderWindows
+
+#elif defined(MAC_IPHONE) // MAC_IPHONE should go before WEBRTC_MAC_INTEL because WEBRTC_MAC_INTEL gets defined if MAC_IPHONE is defined
+#if defined(IPHONE_GLES_RENDERING)
+#define STANDARD_RENDERING kRenderiPhone
+#include "iPhone/video_render_iphone_impl.h"
+#endif
+
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
+#if defined(COCOA_RENDERING)
+#define STANDARD_RENDERING kRenderCocoa
+#include "mac/video_render_mac_cocoa_impl.h"
+#elif defined(CARBON_RENDERING)
+#define STANDARD_RENDERING kRenderCarbon
+#include "mac/video_render_mac_carbon_impl.h"
+#endif
+
+#elif defined(WEBRTC_ANDROID)
+#include "Android/video_render_android_impl.h"
+#include "Android/video_render_android_surface_view.h"
+#include "Android/video_render_android_native_opengl2.h"
+#define STANDARD_RENDERING kRenderAndroid
+
+#elif defined(WEBRTC_LINUX)
+#include "linux/video_render_linux_impl.h"
+#define STANDARD_RENDERING kRenderX11
+
+#else
+//Other platforms
+#endif
+
+#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+// For external rendering
+#include "external/video_render_external_impl.h"
+#ifndef STANDARD_RENDERING
+#define STANDARD_RENDERING kRenderExternal
+#endif // STANDARD_RENDERING
+
+namespace webrtc {
+
+VideoRender*
+VideoRender::CreateVideoRender(const WebRtc_Word32 id,
+ void* window,
+ const bool fullscreen,
+ const VideoRenderType videoRenderType/*=kRenderDefault*/)
+{
+
+ VideoRenderType resultVideoRenderType = videoRenderType;
+ if (videoRenderType == kRenderDefault)
+ {
+ resultVideoRenderType = STANDARD_RENDERING;
+ }
+ return new ModuleVideoRenderImpl(id, resultVideoRenderType, window,
+ fullscreen);
+}
+
+void VideoRender::DestroyVideoRender(
+ VideoRender* module)
+{
+ if (module)
+ {
+ delete module;
+ }
+}
+
+WebRtc_Word32 VideoRender::SetAndroidObjects(void *javaVM)
+{
+#ifdef WEBRTC_ANDROID
+ return VideoRenderAndroid::SetAndroidEnvVariables(javaVM);
+#else
+ return -1;
+#endif
+}
+
+ModuleVideoRenderImpl::ModuleVideoRenderImpl(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+ _id(id), _moduleCrit(*CriticalSectionWrapper::CreateCriticalSection()),
+ _ptrWindow(window), _renderType(videoRenderType),
+ _fullScreen(fullscreen), _ptrRenderer(NULL),
+ _streamRenderMap(*(new MapWrapper()))
+{
+
+ // Create platform specific renderer
+ switch (videoRenderType)
+ {
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+#if defined(_WIN32)
+ case kRenderWindows:
+ {
+ VideoRenderWindowsImpl* ptrRenderer;
+ ptrRenderer = new VideoRenderWindowsImpl(_id, videoRenderType, window, _fullScreen);
+ if (ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ }
+ }
+ break;
+
+#elif defined(MAC_IPHONE)
+ case kRenderiPhone:
+ {
+ VideoRenderIPhoneImpl* ptrRenderer = new VideoRenderIPhoneImpl(_id, videoRenderType, window, _fullScreen);
+ if(ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ }
+ }
+ break;
+
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
+
+#if defined(COCOA_RENDERING)
+ case kRenderCocoa:
+ {
+ VideoRenderMacCocoaImpl* ptrRenderer = new VideoRenderMacCocoaImpl(_id, videoRenderType, window, _fullScreen);
+ if(ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ }
+ }
+
+ break;
+#elif defined(CARBON_RENDERING)
+ case kRenderCarbon:
+ {
+ VideoRenderMacCarbonImpl* ptrRenderer = new VideoRenderMacCarbonImpl(_id, videoRenderType, window, _fullScreen);
+ if(ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ }
+ }
+ break;
+#endif
+
+#elif defined(WEBRTC_ANDROID)
+ case kRenderAndroid:
+ {
+ if(AndroidNativeOpenGl2Renderer::UseOpenGL2(window))
+ {
+ AndroidNativeOpenGl2Renderer* ptrRenderer = NULL;
+ ptrRenderer = new AndroidNativeOpenGl2Renderer(_id, videoRenderType, window, _fullScreen);
+ if (ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
+ }
+ }
+ else
+ {
+ AndroidSurfaceViewRenderer* ptrRenderer = NULL;
+ ptrRenderer = new AndroidSurfaceViewRenderer(_id, videoRenderType, window, _fullScreen);
+ if (ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
+ }
+ }
+
+ }
+ break;
+#elif defined(WEBRTC_LINUX)
+ case kRenderX11:
+ {
+ VideoRenderLinuxImpl* ptrRenderer = NULL;
+ ptrRenderer = new VideoRenderLinuxImpl(_id, videoRenderType, window, _fullScreen);
+ if ( ptrRenderer )
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
+ }
+ }
+ break;
+
+#else
+ // Other platforms
+#endif
+
+#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+ case kRenderExternal:
+ {
+ VideoRenderExternalImpl* ptrRenderer(NULL);
+ ptrRenderer = new VideoRenderExternalImpl(_id, videoRenderType,
+ window, _fullScreen);
+ if (ptrRenderer)
+ {
+ _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
+ }
+ }
+ break;
+ default:
+ // Error...
+ break;
+ }
+ if (_ptrRenderer)
+ {
+ if (_ptrRenderer->Init() == -1)
+ {
+ }
+ }
+}
+
+ModuleVideoRenderImpl::~ModuleVideoRenderImpl()
+{
+ delete &_moduleCrit;
+
+ while (_streamRenderMap.Size() > 0)
+ {
+ MapItem* item = _streamRenderMap.First();
+ IncomingVideoStream* ptrIncomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ assert(ptrIncomingStream != NULL);
+ delete ptrIncomingStream;
+ _streamRenderMap.Erase(item);
+ }
+ delete &_streamRenderMap;
+
+ // Delete platform specific renderer
+ if (_ptrRenderer)
+ {
+ VideoRenderType videoRenderType = _ptrRenderer->RenderType();
+ switch (videoRenderType)
+ {
+ case kRenderExternal:
+ {
+ VideoRenderExternalImpl
+ * ptrRenderer =
+ reinterpret_cast<VideoRenderExternalImpl*> (_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+#if defined(_WIN32)
+ case kRenderWindows:
+ {
+ VideoRenderWindowsImpl* ptrRenderer = reinterpret_cast<VideoRenderWindowsImpl*>(_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
+
+#if defined(COCOA_RENDERING)
+ case kRenderCocoa:
+ {
+ VideoRenderMacCocoaImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCocoaImpl*> (_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+#elif defined(CARBON_RENDERING)
+ case kRenderCarbon:
+ {
+ VideoRenderMacCarbonImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCarbonImpl*> (_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+#endif
+
+#elif defined(MAC_IPHONE)
+ case kRenderiPhone:
+ break;
+
+#elif defined(WEBRTC_ANDROID)
+ case kRenderAndroid:
+ {
+ VideoRenderAndroid* ptrRenderer = reinterpret_cast<VideoRenderAndroid*> (_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+
+#elif defined(WEBRTC_LINUX)
+ case kRenderX11:
+ {
+ VideoRenderLinuxImpl* ptrRenderer = reinterpret_cast<VideoRenderLinuxImpl*> (_ptrRenderer);
+ _ptrRenderer = NULL;
+ delete ptrRenderer;
+ }
+ break;
+#else
+ //other platforms
+#endif
+
+#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+ default:
+ // Error...
+ break;
+ }
+ }
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+
+ CriticalSectionScoped cs(_moduleCrit);
+
+ _id = id;
+
+ if (_ptrRenderer)
+ {
+ _ptrRenderer->ChangeUniqueId(_id);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::TimeUntilNextProcess()
+{
+ // Not used
+ return 50;
+}
+WebRtc_Word32 ModuleVideoRenderImpl::Process()
+{
+ // Not used
+ return 0;
+}
+
+void*
+ModuleVideoRenderImpl::Window()
+{
+ CriticalSectionScoped cs(_moduleCrit);
+ return _ptrWindow;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::ChangeWindow(void* window)
+{
+
+ CriticalSectionScoped cs(_moduleCrit);
+
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+
+#if defined(MAC_IPHONE) // MAC_IPHONE must go before WEBRTC_MAC or WEBRTC_MAC_INTEL
+ _ptrRenderer = NULL;
+ delete _ptrRenderer;
+
+ VideoRenderIPhoneImpl* ptrRenderer;
+ ptrRenderer = new VideoRenderIPhoneImpl(_id, kRenderiPhone, window, _fullScreen);
+ if (!ptrRenderer)
+ {
+ return -1;
+ }
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ return _ptrRenderer->ChangeWindow(window);
+
+#elif defined(WEBRTC_MAC) | defined(WEBRTC_MAC_INTEL)
+
+ _ptrRenderer = NULL;
+ delete _ptrRenderer;
+
+#if defined(COCOA_RENDERING)
+ VideoRenderMacCocoaImpl* ptrRenderer;
+ ptrRenderer = new VideoRenderMacCocoaImpl(_id, kRenderCocoa, window, _fullScreen);
+#elif defined(CARBON_RENDERING)
+ VideoRenderMacCarbonImpl* ptrRenderer;
+ ptrRenderer = new VideoRenderMacCarbonImpl(_id, kRenderCarbon, window, _fullScreen);
+#endif
+ if (!ptrRenderer)
+ {
+ return -1;
+ }
+ _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
+ return _ptrRenderer->ChangeWindow(window);
+
+#else
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+ return _ptrRenderer->ChangeWindow(window);
+
+#endif
+
+#else // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::Id()
+{
+ CriticalSectionScoped cs(_moduleCrit);
+ return _id;
+}
+
+WebRtc_UWord32 ModuleVideoRenderImpl::GetIncomingFrameRate(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ MapItem* mapItem = _streamRenderMap.Find(streamId);
+ if (mapItem == NULL)
+ {
+ // This stream doesn't exist
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return 0;
+ }
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (mapItem->GetItem());
+ if (incomingStream == NULL)
+ {
+ // This should never happen
+ assert(false);
+ _streamRenderMap.Erase(mapItem);
+ return 0;
+ }
+ return incomingStream->IncomingRate();
+}
+
+VideoRenderCallback*
+ModuleVideoRenderImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return NULL;
+ }
+
+ if (_streamRenderMap.Find(streamId) != NULL)
+ {
+ // The stream already exists...
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream already exists", __FUNCTION__);
+ return NULL;
+ }
+
+ VideoRenderCallback* ptrRenderCallback =
+ _ptrRenderer->AddIncomingRenderStream(streamId, zOrder, left, top,
+ right, bottom);
+ if (ptrRenderCallback == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Can't create incoming stream in renderer",
+ __FUNCTION__);
+ return NULL;
+ }
+
+ // Create platform independant code
+ IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(_id,
+ streamId);
+ if (ptrIncomingStream == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Can't create incoming stream", __FUNCTION__);
+ return NULL;
+ }
+
+
+ if (ptrIncomingStream->SetRenderCallback(ptrRenderCallback) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: Can't set render callback", __FUNCTION__);
+ delete ptrIncomingStream;
+ _ptrRenderer->DeleteIncomingRenderStream(streamId);
+ return NULL;
+ }
+
+ VideoRenderCallback* moduleCallback =
+ ptrIncomingStream->ModuleCallback();
+
+ // Store the stream
+ _streamRenderMap.Insert(streamId, ptrIncomingStream);
+
+ return moduleCallback;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::DeleteIncomingRenderStream(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ MapItem* mapItem = _streamRenderMap.Find(streamId);
+ if (!mapItem)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return -1;
+ }
+
+ IncomingVideoStream* ptrIncomingStream =
+ static_cast<IncomingVideoStream*> (mapItem->GetItem());
+ delete ptrIncomingStream;
+ ptrIncomingStream = NULL;
+ _ptrRenderer->DeleteIncomingRenderStream(streamId);
+ _streamRenderMap.Erase(mapItem);
+
+ return 0;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::AddExternalRenderCallback(
+ const WebRtc_UWord32 streamId,
+ VideoRenderCallback* renderObject)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ MapItem* mapItem = _streamRenderMap.Find(streamId);
+ if (!mapItem)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return -1;
+ }
+
+ IncomingVideoStream* ptrIncomingStream =
+ static_cast<IncomingVideoStream*> (mapItem->GetItem());
+ if (!ptrIncomingStream) {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: could not get stream", __FUNCTION__);
+ return -1;
+ }
+ return ptrIncomingStream->SetExternalCallback(renderObject);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
+ const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ return _ptrRenderer->GetIncomingRenderStreamProperties(streamId, zOrder,
+ left, top, right,
+ bottom);
+}
+
+WebRtc_UWord32 ModuleVideoRenderImpl::GetNumIncomingRenderStreams() const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ return (WebRtc_UWord32) _streamRenderMap.Size();
+}
+
+bool ModuleVideoRenderImpl::HasIncomingRenderStream(
+ const WebRtc_UWord32 streamId) const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ bool hasStream = false;
+ if (_streamRenderMap.Find(streamId) != NULL)
+ {
+ hasStream = true;
+ }
+ return hasStream;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::RegisterRawFrameCallback(
+ const WebRtc_UWord32 streamId,
+ VideoRenderCallback* callbackObj)
+{
+ return -1;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::StartRender(const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ // Start the stream
+ MapItem* item = _streamRenderMap.Find(streamId);
+ if (item == NULL)
+ {
+ return -1;
+ }
+
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream->Start() == -1)
+ {
+ return -1;
+ }
+
+ // Start the HW renderer
+ if (_ptrRenderer->StartRender() == -1)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::StopRender(const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s(%d): No renderer", __FUNCTION__, streamId);
+ return -1;
+ }
+
+ // Stop the incoming stream
+ MapItem* item = _streamRenderMap.Find(streamId);
+ if (item == NULL)
+ {
+ return -1;
+ }
+
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream->Stop() == -1)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::ResetRender()
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ WebRtc_Word32 error = 0;
+
+ // Loop through all incoming streams and stop them
+ MapItem* item = _streamRenderMap.First();
+ while (item)
+ {
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream->Reset() == -1)
+ {
+ error = -1;
+ }
+ item = _streamRenderMap.Next(item);
+ }
+ return error;
+}
+
+RawVideoType ModuleVideoRenderImpl::PreferredVideoType() const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (_ptrRenderer == NULL)
+ {
+ return kVideoI420;
+ }
+
+ return _ptrRenderer->PerferedVideoType();
+}
+
+bool ModuleVideoRenderImpl::IsFullScreen()
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->FullScreen();
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::GetScreenResolution(
+ WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->GetScreenResolution(screenWidth, screenHeight);
+}
+
+WebRtc_UWord32 ModuleVideoRenderImpl::RenderFrameRate(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->RenderFrameRate(streamId);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetStreamCropping(
+ const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->SetStreamCropping(streamId, left, top, right, bottom);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetTransparentBackground(const bool enable)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->SetTransparentBackground(enable);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::FullScreenRender(void* window,
+ const bool enable)
+{
+ return -1;
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetText(
+ const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+ return _ptrRenderer->SetText(textId, text, textLength, textColorRef,
+ backgroundColorRef, left, top, right, bottom);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+ return _ptrRenderer->SetBitmap(bitMap, pictureId, colorKey, left, top,
+ right, bottom);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::GetLastRenderedFrame(
+ const WebRtc_UWord32 streamId,
+ VideoFrame &frame) const
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ MapItem *item = _streamRenderMap.Find(streamId);
+ if (item == NULL)
+ {
+ // This stream doesn't exist
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return 0;
+ }
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream == NULL)
+ {
+ // This should never happen
+ assert(false);
+ _streamRenderMap.Erase(item);
+ return 0;
+ }
+ return incomingStream->GetLastRenderedFrame(frame);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::ConfigureRenderer(
+ const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return false;
+ }
+ return _ptrRenderer->ConfigureRenderer(streamId, zOrder, left, top, right,
+ bottom);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetStartImage(
+ const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ MapItem *item = _streamRenderMap.Find(streamId);
+ if (item == NULL)
+ {
+ // This stream doesn't exist
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return -1;
+ }
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream == NULL)
+ {
+ // This should never happen
+ assert(false);
+ _streamRenderMap.Erase(item);
+ return 0;
+ }
+ return incomingStream->SetStartImage(videoFrame);
+
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::SetTimeoutImage(
+ const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame,
+ const WebRtc_UWord32 timeout)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ MapItem *item = _streamRenderMap.Find(streamId);
+ if (item == NULL)
+ {
+ // This stream doesn't exist
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return -1;
+ }
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream == NULL)
+ {
+ // This should never happen
+ assert(false);
+ _streamRenderMap.Erase(item);
+ return 0;
+ }
+ return incomingStream->SetTimeoutImage(videoFrame, timeout);
+}
+
+WebRtc_Word32 ModuleVideoRenderImpl::MirrorRenderStream(const int renderId,
+ const bool enable,
+ const bool mirrorXAxis,
+ const bool mirrorYAxis)
+{
+ CriticalSectionScoped cs(_moduleCrit);
+
+ if (!_ptrRenderer)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: No renderer", __FUNCTION__);
+ return -1;
+ }
+
+ MapItem *item = _streamRenderMap.Find(renderId);
+ if (item == NULL)
+ {
+ // This stream doesn't exist
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
+ "%s: stream doesn't exist", __FUNCTION__);
+ return 0;
+ }
+ IncomingVideoStream* incomingStream =
+ static_cast<IncomingVideoStream*> (item->GetItem());
+ if (incomingStream == NULL)
+ {
+ // This should never happen
+ assert(false);
+ _streamRenderMap.Erase(item);
+ return 0;
+ }
+
+ return incomingStream->EnableMirroring(enable, mirrorXAxis, mirrorYAxis);
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/video_render_impl.h b/trunk/src/modules/video_render/main/source/video_render_impl.h
new file mode 100644
index 0000000..16f934e
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/video_render_impl.h
@@ -0,0 +1,229 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
+
+#include "engine_configurations.h"
+#include "video_render.h"
+#include "map_wrapper.h"
+
+//#include "video_render_defines.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class IncomingVideoStream;
+class IVideoRender;
+class MapWrapper;
+
+// Class definitions
+class ModuleVideoRenderImpl: public VideoRender
+{
+public:
+ /*
+ * VideoRenderer constructor/destructor
+ */
+ ModuleVideoRenderImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window, const bool fullscreen);
+
+ virtual ~ModuleVideoRenderImpl();
+
+ /*
+ * Change the unique identifier of this object
+ */
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ /*
+ * Returns the render window
+ */
+ virtual void* Window();
+
+ /*
+ * Change render window
+ */
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /*
+ * Returns module id
+ */
+ WebRtc_Word32 Id();
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ /*
+ * Add incoming render stream
+ */
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom);
+ /*
+ * Delete incoming render stream
+ */
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ /*
+ * Add incoming render callback, used for external rendering
+ */
+ virtual WebRtc_Word32
+ AddExternalRenderCallback(const WebRtc_UWord32 streamId,
+ VideoRenderCallback* renderObject);
+
+ /*
+ * Get the porperties for an incoming render stream
+ */
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const;
+ /*
+ * Incoming frame rate for the specified stream.
+ */
+ virtual WebRtc_UWord32 GetIncomingFrameRate(const WebRtc_UWord32 streamId);
+
+ /*
+ * Returns the number of incoming streams added to this render module
+ */
+ virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const;
+
+ /*
+ * Returns true if this render module has the streamId added, false otherwise.
+ */
+ virtual bool HasIncomingRenderStream(const WebRtc_UWord32 streamId) const;
+
+ /*
+ *
+ */
+ virtual WebRtc_Word32
+ RegisterRawFrameCallback(const WebRtc_UWord32 streamId,
+ VideoRenderCallback* callbackObj);
+
+ virtual WebRtc_Word32 GetLastRenderedFrame(const WebRtc_UWord32 streamId,
+ VideoFrame &frame) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ /*
+ * Starts rendering the specified stream
+ */
+ virtual WebRtc_Word32 StartRender(const WebRtc_UWord32 streamId);
+
+ /*
+ * Stops the renderer
+ */
+ virtual WebRtc_Word32 StopRender(const WebRtc_UWord32 streamId);
+
+ /*
+ * Sets the renderer in start state, no streams removed.
+ */
+ virtual WebRtc_Word32 ResetRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ /*
+ * Returns the prefered render video type
+ */
+ virtual RawVideoType PreferredVideoType() const;
+
+ /*
+ * Returns true if the renderer is in fullscreen mode, otherwise false.
+ */
+ virtual bool IsFullScreen();
+
+ /*
+ * Gets screen resolution in pixels
+ */
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ /*
+ * Get the actual render rate for this stream. I.e rendered frame rate,
+ * not frames delivered to the renderer.
+ */
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ /*
+ * Set cropping of incoming stream
+ */
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32 SetStartImage(const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame);
+
+ virtual WebRtc_Word32 SetTimeoutImage(const WebRtc_UWord32 streamId,
+ const VideoFrame& videoFrame,
+ const WebRtc_UWord32 timeout);
+
+ virtual WebRtc_Word32 MirrorRenderStream(const int renderId,
+ const bool enable,
+ const bool mirrorXAxis,
+ const bool mirrorYAxis);
+
+private:
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _moduleCrit;
+ void* _ptrWindow;
+ VideoRenderType _renderType;
+ bool _fullScreen;
+
+ IVideoRender* _ptrRenderer;
+ MapWrapper& _streamRenderMap;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h b/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h
new file mode 100644
index 0000000..a765134
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h
@@ -0,0 +1,118 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
+
+#include "video_render.h"
+
+namespace webrtc {
+
+// Class definitions
+class IVideoRenderWin
+{
+public:
+ /**************************************************************************
+ *
+ * Constructor/destructor
+ *
+ ***************************************************************************/
+ virtual ~IVideoRenderWin()
+ {
+ };
+
+ virtual WebRtc_Word32 Init() = 0;
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderCallback
+ * CreateChannel(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId) = 0;
+
+ virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) = 0;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender() = 0;
+
+ virtual WebRtc_Word32 StopRender() = 0;
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual bool IsFullScreen() = 0;
+
+ virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 colorText,
+ const WebRtc_UWord32 colorBg,
+ const float left,
+ const float top,
+ const float rigth,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
+
+ virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
+ WebRtc_UWord64& availableMemory) = 0;
+
+};
+
+} //namespace webrtc
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc
new file mode 100644
index 0000000..00bc41b
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc
@@ -0,0 +1,1192 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Own include file
+#include "video_render_direct3d9.h"
+
+// System include files
+#include <windows.h>
+
+// WebRtc include files
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "thread_wrapper.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc {
+
+// A structure for our custom vertex type
+struct CUSTOMVERTEX
+{
+ FLOAT x, y, z;
+ DWORD color; // The vertex color
+ FLOAT u, v;
+};
+
+// Our custom FVF, which describes our custom vertex structure
+#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ|D3DFVF_DIFFUSE|D3DFVF_TEX1)
+
+/*
+ *
+ * D3D9Channel
+ *
+ */
+D3D9Channel::D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
+ CriticalSectionWrapper* critSect,
+ Trace* trace) :
+ _width(0),
+ _height(0),
+ _pd3dDevice(pd3DDevice),
+ _pTexture(NULL),
+ _bufferIsUpdated(false),
+ _critSect(critSect),
+ _streamId(0),
+ _zOrder(0),
+ _startWidth(0),
+ _startHeight(0),
+ _stopWidth(0),
+ _stopHeight(0)
+{
+
+}
+
+D3D9Channel::~D3D9Channel()
+{
+ //release the texture
+ if (_pTexture != NULL)
+ {
+ _pTexture->Release();
+ _pTexture = NULL;
+ }
+}
+
+void D3D9Channel::SetStreamSettings(WebRtc_UWord16 streamId,
+ WebRtc_UWord32 zOrder,
+ float startWidth,
+ float startHeight,
+ float stopWidth,
+ float stopHeight)
+{
+ _streamId = streamId;
+ _zOrder = zOrder;
+ _startWidth = startWidth;
+ _startHeight = startHeight;
+ _stopWidth = stopWidth;
+ _stopHeight = stopHeight;
+}
+
+int D3D9Channel::GetStreamSettings(WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& startWidth,
+ float& startHeight,
+ float& stopWidth,
+ float& stopHeight)
+{
+ streamId = _streamId;
+ zOrder = _zOrder;
+ startWidth = _startWidth;
+ startHeight = _startHeight;
+ stopWidth = _stopWidth;
+ stopHeight = _stopHeight;
+ return 0;
+}
+
+int D3D9Channel::GetTextureWidth()
+{
+ return _width;
+}
+
+int D3D9Channel::GetTextureHeight()
+{
+ return _height;
+}
+
+// Called from video engine when a the frame size changed
+int D3D9Channel::FrameSizeChange(int width, int height, int numberOfStreams)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "FrameSizeChange, wifth: %d, height: %d, streams: %d", width,
+ height, numberOfStreams);
+
+ CriticalSectionScoped cs(*_critSect);
+ _width = width;
+ _height = height;
+
+ //clean the previous texture
+ if (_pTexture != NULL)
+ {
+ _pTexture->Release();
+ _pTexture = NULL;
+ }
+
+ HRESULT ret = E_POINTER;
+
+ if (_pd3dDevice)
+ ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
+ D3DPOOL_MANAGED, &_pTexture, NULL);
+
+ if (FAILED(ret))
+ {
+ _pTexture = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 D3D9Channel::RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+{
+ CriticalSectionScoped cs(*_critSect);
+ if (_width != videoFrame.Width() || _height != videoFrame.Height())
+ {
+ if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
+ {
+ return -1;
+ }
+ }
+ return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
+ videoFrame.TimeStamp());
+}
+
+// Called from video engine when a new frame should be rendered.
+int D3D9Channel::DeliverFrame(unsigned char* buffer,
+ int bufferSize,
+ unsigned int timeStamp90kHz)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
+ "DeliverFrame to D3D9Channel");
+
+ CriticalSectionScoped cs(*_critSect);
+
+ //FIXME if _bufferIsUpdated is still true (not be renderred), do we what to update the texture?)
+ //probably not
+ if (_bufferIsUpdated)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
+ "Last frame hasn't been rendered yet. Drop this frame.");
+ return -1;
+ }
+
+ if (!_pd3dDevice)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "D3D for rendering not initialized.");
+ return -1;
+ }
+
+ if (!_pTexture)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Texture for rendering not initialized.");
+ return -1;
+ }
+
+ D3DLOCKED_RECT lr;
+
+ if (FAILED(_pTexture->LockRect(0, &lr, NULL, 0)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to lock a texture in D3D9 Channel.");
+ return -1;
+ }
+ UCHAR* pRect = (UCHAR*) lr.pBits;
+
+ ConvertFromI420(buffer, _width, kARGB, 0, _width, _height, pRect);
+
+ if (FAILED(_pTexture->UnlockRect(0)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to unlock a texture in D3D9 Channel.");
+ return -1;
+ }
+
+ _bufferIsUpdated = true;
+
+ return 0;
+}
+
+// Called by d3d channel owner to indicate the frame/texture has been rendered off
+int D3D9Channel::RenderOffFrame()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
+ "Frame has been rendered to the screen.");
+ CriticalSectionScoped cs(*_critSect);
+ _bufferIsUpdated = false;
+ return 0;
+}
+
+// Called by d3d channel owner to check if the texture is updated
+int D3D9Channel::IsUpdated(bool& isUpdated)
+{
+ CriticalSectionScoped cs(*_critSect);
+ isUpdated = _bufferIsUpdated;
+ return 0;
+}
+
+// Called by d3d channel owner to get the texture
+LPDIRECT3DTEXTURE9 D3D9Channel::GetTexture()
+{
+ CriticalSectionScoped cs(*_critSect);
+ return _pTexture;
+}
+
+int D3D9Channel::ReleaseTexture()
+{
+ CriticalSectionScoped cs(*_critSect);
+
+ //release the texture
+ if (_pTexture != NULL)
+ {
+ _pTexture->Release();
+ _pTexture = NULL;
+ }
+ _pd3dDevice = NULL;
+ return 0;
+}
+
+int D3D9Channel::RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice)
+{
+ CriticalSectionScoped cs(*_critSect);
+
+ _pd3dDevice = pd3DDevice;
+
+ if (_pTexture != NULL)
+ {
+ _pTexture->Release();
+ _pTexture = NULL;
+ }
+
+ HRESULT ret;
+
+ ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
+ D3DPOOL_MANAGED, &_pTexture, NULL);
+
+ if (FAILED(ret))
+ {
+ _pTexture = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+/*
+ *
+ * VideoRenderDirect3D9
+ *
+ */
+VideoRenderDirect3D9::VideoRenderDirect3D9(Trace* trace,
+ HWND hWnd,
+ bool fullScreen) :
+ _refD3DCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _trace(trace),
+ _hWnd(hWnd),
+ _fullScreen(fullScreen),
+ _pTextureLogo(NULL),
+ _pVB(NULL),
+ _pd3dDevice(NULL),
+ _pD3D(NULL),
+ _d3dChannels(),
+ _d3dZorder(),
+ _screenUpdateThread(NULL),
+ _screenUpdateEvent(NULL),
+ _logoLeft(0),
+ _logoTop(0),
+ _logoRight(0),
+ _logoBottom(0),
+ _pd3dSurface(NULL),
+ _totalMemory(-1),
+ _availableMemory(-1)
+{
+ _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
+ this, kRealtimePriority);
+ _screenUpdateEvent = EventWrapper::Create();
+ SetRect(&_originalHwndRect, 0, 0, 0, 0);
+}
+
+VideoRenderDirect3D9::~VideoRenderDirect3D9()
+{
+ //NOTE: we should not enter CriticalSection in here!
+
+ // Signal event to exit thread, then delete it
+ ThreadWrapper* tmpPtr = _screenUpdateThread;
+ _screenUpdateThread = NULL;
+ if (tmpPtr)
+ {
+ tmpPtr->SetNotAlive();
+ _screenUpdateEvent->Set();
+ _screenUpdateEvent->StopTimer();
+
+ if (tmpPtr->Stop())
+ {
+ delete tmpPtr;
+ }
+ }
+ delete _screenUpdateEvent;
+
+ //close d3d device
+ CloseDevice();
+
+ // Delete all channels
+ std::map<int, D3D9Channel*>::iterator it = _d3dChannels.begin();
+ while (it != _d3dChannels.end())
+ {
+ delete it->second;
+ it = _d3dChannels.erase(it);
+ }
+ // Clean the zOrder map
+ _d3dZorder.clear();
+
+ if (_fullScreen)
+ {
+ // restore hwnd to original size and position
+ ::SetWindowPos(_hWnd, HWND_NOTOPMOST, _originalHwndRect.left,
+ _originalHwndRect.top, _originalHwndRect.right
+ - _originalHwndRect.left,
+ _originalHwndRect.bottom - _originalHwndRect.top,
+ SWP_FRAMECHANGED);
+ ::RedrawWindow(_hWnd, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
+ | RDW_ERASE);
+ ::RedrawWindow(NULL, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
+ | RDW_ERASE);
+ }
+
+ delete &_refD3DCritsect;
+}
+
+DWORD VideoRenderDirect3D9::GetVertexProcessingCaps()
+{
+ D3DCAPS9 caps;
+ DWORD dwVertexProcessing = D3DCREATE_SOFTWARE_VERTEXPROCESSING;
+ if (SUCCEEDED(_pD3D->GetDeviceCaps(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL,
+ &caps)))
+ {
+ if ((caps.DevCaps & D3DDEVCAPS_HWTRANSFORMANDLIGHT)
+ == D3DDEVCAPS_HWTRANSFORMANDLIGHT)
+ {
+ dwVertexProcessing = D3DCREATE_HARDWARE_VERTEXPROCESSING;
+ }
+ }
+ return dwVertexProcessing;
+}
+
+int VideoRenderDirect3D9::InitializeD3D(HWND hWnd,
+ D3DPRESENT_PARAMETERS* pd3dpp)
+{
+ // initialize Direct3D
+ if (NULL == (_pD3D = Direct3DCreate9(D3D_SDK_VERSION)))
+ {
+ return -1;
+ }
+
+ // determine what type of vertex processing to use based on the device capabilities
+ DWORD dwVertexProcessing = GetVertexProcessingCaps();
+
+ // get the display mode
+ D3DDISPLAYMODE d3ddm;
+ _pD3D->GetAdapterDisplayMode(D3DADAPTER_DEFAULT, &d3ddm);
+ pd3dpp->BackBufferFormat = d3ddm.Format;
+
+ // create the D3D device
+ if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL, hWnd,
+ dwVertexProcessing | D3DCREATE_MULTITHREADED
+ | D3DCREATE_FPU_PRESERVE, pd3dpp,
+ &_pd3dDevice)))
+ {
+ //try the ref device
+ if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_REF,
+ hWnd, dwVertexProcessing
+ | D3DCREATE_MULTITHREADED
+ | D3DCREATE_FPU_PRESERVE,
+ pd3dpp, &_pd3dDevice)))
+ {
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+int VideoRenderDirect3D9::ResetDevice()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "VideoRenderDirect3D9::ResetDevice");
+
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+ //release the channel texture
+ std::map<int, D3D9Channel*>::iterator it;
+ it = _d3dChannels.begin();
+ while (it != _d3dChannels.end())
+ {
+ if (it->second)
+ {
+ it->second->ReleaseTexture();
+ }
+ it++;
+ }
+
+ //close d3d device
+ if (CloseDevice() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "VideoRenderDirect3D9::ResetDevice failed to CloseDevice");
+ return -1;
+ }
+
+ //reinit d3d device
+ if (InitDevice() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "VideoRenderDirect3D9::ResetDevice failed to InitDevice");
+ return -1;
+ }
+
+ //recreate channel texture
+ it = _d3dChannels.begin();
+ while (it != _d3dChannels.end())
+ {
+ if (it->second)
+ {
+ it->second->RecreateTexture(_pd3dDevice);
+ }
+ it++;
+ }
+
+ return 0;
+}
+
+int VideoRenderDirect3D9::InitDevice()
+{
+ // Set up the structure used to create the D3DDevice
+ ZeroMemory(&_d3dpp, sizeof(_d3dpp));
+ _d3dpp.SwapEffect = D3DSWAPEFFECT_DISCARD;
+ _d3dpp.BackBufferFormat = D3DFMT_A8R8G8B8;
+ if (GetWindowRect(_hWnd, &_originalHwndRect) == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "VideoRenderDirect3D9::InitDevice Could not get window size");
+ return -1;
+ }
+ if (!_fullScreen)
+ {
+ _winWidth = _originalHwndRect.right - _originalHwndRect.left;
+ _winHeight = _originalHwndRect.bottom - _originalHwndRect.top;
+ _d3dpp.Windowed = TRUE;
+ _d3dpp.BackBufferHeight = 0;
+ _d3dpp.BackBufferWidth = 0;
+ }
+ else
+ {
+ _winWidth = (LONG) ::GetSystemMetrics(SM_CXSCREEN);
+ _winHeight = (LONG) ::GetSystemMetrics(SM_CYSCREEN);
+ _d3dpp.Windowed = FALSE;
+ _d3dpp.BackBufferWidth = _winWidth;
+ _d3dpp.BackBufferHeight = _winHeight;
+ _d3dpp.PresentationInterval = D3DPRESENT_INTERVAL_IMMEDIATE;
+ }
+
+ if (InitializeD3D(_hWnd, &_d3dpp) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "VideoRenderDirect3D9::InitDevice failed in InitializeD3D");
+ return -1;
+ }
+
+ // Turn off culling, so we see the front and back of the triangle
+ _pd3dDevice->SetRenderState(D3DRS_CULLMODE, D3DCULL_NONE);
+
+ // Turn off D3D lighting, since we are providing our own vertex colors
+ _pd3dDevice->SetRenderState(D3DRS_LIGHTING, FALSE);
+
+ // Settings for alpha blending
+ _pd3dDevice->SetRenderState(D3DRS_ALPHABLENDENABLE, TRUE);
+ _pd3dDevice->SetRenderState(D3DRS_SRCBLEND, D3DBLEND_SRCALPHA);
+ _pd3dDevice->SetRenderState(D3DRS_DESTBLEND, D3DBLEND_INVSRCALPHA);
+ //_pd3dDevice->SetTextureStageState(0,D3DTSS_ALPHAOP,D3DTOP_SELECTARG1);
+ //_pd3dDevice->SetTextureStageState(0,D3DTSS_ALPHAARG1,D3DTA_TEXTURE);
+
+ // Initialize Vertices
+ CUSTOMVERTEX Vertices[] = {
+ //front
+ { -1.0f, -1.0f, 0.0f, 0xffffffff, 0, 1 }, { -1.0f, 1.0f, 0.0f,
+ 0xffffffff, 0, 0 },
+ { 1.0f, -1.0f, 0.0f, 0xffffffff, 1, 1 }, { 1.0f, 1.0f, 0.0f,
+ 0xffffffff, 1, 0 } };
+
+ // Create the vertex buffer.
+ if (FAILED(_pd3dDevice->CreateVertexBuffer(sizeof(Vertices), 0,
+ D3DFVF_CUSTOMVERTEX,
+ D3DPOOL_DEFAULT, &_pVB, NULL )))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to create the vertex buffer.");
+ return -1;
+ }
+
+ // Now we fill the vertex buffer.
+ VOID* pVertices;
+ if (FAILED(_pVB->Lock(0, sizeof(Vertices), (void**) &pVertices, 0)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to lock the vertex buffer.");
+ return -1;
+ }
+ memcpy(pVertices, Vertices, sizeof(Vertices));
+ _pVB->Unlock();
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::Init()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "VideoRenderDirect3D9::Init");
+
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+ // Start rendering thread...
+ if (!_screenUpdateThread)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Thread not created");
+ return -1;
+ }
+ unsigned int threadId;
+ _screenUpdateThread->Start(threadId);
+
+ // Start the event triggering the render process
+ unsigned int monitorFreq = 60;
+ DEVMODE dm;
+ // initialize the DEVMODE structure
+ ZeroMemory(&dm, sizeof(dm));
+ dm.dmSize = sizeof(dm);
+ if (0 != EnumDisplaySettings(NULL, ENUM_CURRENT_SETTINGS, &dm))
+ {
+ monitorFreq = dm.dmDisplayFrequency;
+ }
+ _screenUpdateEvent->StartTimer(true, 1000 / monitorFreq);
+
+ return InitDevice();
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::ChangeWindow(void* window)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return -1;
+}
+
+int VideoRenderDirect3D9::UpdateRenderSurface()
+{
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+ // Check if there are any updated buffers
+ bool updated = false;
+ std::map<int, D3D9Channel*>::iterator it;
+ it = _d3dChannels.begin();
+ while (it != _d3dChannels.end())
+ {
+
+ D3D9Channel* channel = it->second;
+ channel->IsUpdated(updated);
+ if (updated)
+ {
+ break;
+ }
+ it++;
+ }
+ //nothing is updated, continue
+ if (!updated)
+ return -1;
+
+ // Clear the backbuffer to a black color
+ _pd3dDevice->Clear(0, NULL, D3DCLEAR_TARGET, D3DCOLOR_XRGB(0, 0, 0), 1.0f,
+ 0);
+
+ // Begin the scene
+ if (SUCCEEDED(_pd3dDevice->BeginScene()))
+ {
+ _pd3dDevice->SetStreamSource(0, _pVB, 0, sizeof(CUSTOMVERTEX));
+ _pd3dDevice->SetFVF(D3DFVF_CUSTOMVERTEX);
+
+ D3DXMATRIX matWorld;
+ D3DXMATRIX matWorldTemp;
+
+ //draw all the channels
+ //get texture from the channels
+ LPDIRECT3DTEXTURE9 textureFromChannel = NULL;
+ DWORD textureWidth, textureHeight;
+
+ std::multimap<int, unsigned int>::reverse_iterator it;
+ it = _d3dZorder.rbegin();
+ while (it != _d3dZorder.rend())
+ {
+ // loop through all channels and streams in Z order
+ int channel = it->second & 0x0000ffff;
+
+ std::map<int, D3D9Channel*>::iterator ddIt;
+ ddIt = _d3dChannels.find(channel);
+ if (ddIt != _d3dChannels.end())
+ {
+ // found the channel
+ D3D9Channel* channelObj = ddIt->second;
+ if (channelObj)
+ {
+ textureFromChannel = channelObj->GetTexture();
+ textureWidth = channelObj->GetTextureWidth();
+ textureHeight = channelObj->GetTextureHeight();
+
+ WebRtc_UWord32 zOrder;
+ float startWidth, startHeight, stopWidth, stopHeight;
+ channelObj->GetStreamSettings(0, zOrder, startWidth,
+ startHeight, stopWidth,
+ stopHeight);
+
+ //draw the video stream
+ UpdateVerticeBuffer(_pVB, 0, startWidth, startHeight,
+ stopWidth, stopHeight);
+ _pd3dDevice->SetTexture(0, textureFromChannel);
+ _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
+
+ //Notice channel that this frame as been rendered
+ channelObj->RenderOffFrame();
+ }
+ }
+ it++;
+ }
+
+ //draw the logo
+ if (_pTextureLogo)
+ {
+ UpdateVerticeBuffer(_pVB, 0, _logoLeft, _logoTop, _logoRight,
+ _logoBottom);
+ _pd3dDevice->SetTexture(0, _pTextureLogo);
+ _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
+ }
+
+ // End the scene
+ _pd3dDevice->EndScene();
+ }
+
+ // Present the backbuffer contents to the display
+ _pd3dDevice->Present(NULL, NULL, NULL, NULL );
+
+ return 0;
+}
+
+//set the alpha value of the pixal with a particular colorkey as 0
+int VideoRenderDirect3D9::SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
+ DDCOLORKEY* transparentColorKey,
+ DWORD width,
+ DWORD height)
+{
+ D3DLOCKED_RECT lr;
+ if (!pTexture)
+ return -1;
+
+ CriticalSectionScoped cs(_refD3DCritsect);
+ if (SUCCEEDED(pTexture->LockRect(0, &lr, NULL, D3DLOCK_DISCARD)))
+ {
+ for (DWORD y = 0; y < height; y++)
+ {
+ DWORD dwOffset = y * width;
+
+ for (DWORD x = 0; x < width; x)
+ {
+ DWORD a = (DWORD) 0;
+
+ DWORD temp = ((DWORD*) lr.pBits)[dwOffset + x];
+ if ((temp & 0x00FFFFFF)
+ == transparentColorKey->dwColorSpaceLowValue)
+ {
+ temp &= 0x00FFFFFF;
+ }
+ else
+ {
+ temp |= 0xFF000000;
+ }
+ ((DWORD*) lr.pBits)[dwOffset + x] = temp;
+ x++;
+ }
+ }
+ pTexture->UnlockRect(0);
+ return 0;
+ }
+ return -1;
+}
+
+/*
+ *
+ * Rendering process
+ *
+ */
+bool VideoRenderDirect3D9::ScreenUpdateThreadProc(void* obj)
+{
+ return static_cast<VideoRenderDirect3D9*> (obj)->ScreenUpdateProcess();
+}
+
+bool VideoRenderDirect3D9::ScreenUpdateProcess()
+{
+ _screenUpdateEvent->Wait(100);
+
+ if (!_screenUpdateThread)
+ {
+ //stop the thread
+ return false;
+ }
+ if (!_pd3dDevice)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "d3dDevice not created.");
+ return true;
+ }
+
+ HRESULT hr = _pd3dDevice->TestCooperativeLevel();
+
+ if (SUCCEEDED(hr))
+ {
+ UpdateRenderSurface();
+ }
+
+ if (hr == D3DERR_DEVICELOST)
+ {
+ //Device is lost and cannot be reset yet
+
+ }
+ else if (hr == D3DERR_DEVICENOTRESET)
+ {
+ //Lost but we can reset it now
+ //Note: the standard way is to call Reset, however for some reason doesn't work here.
+ //so we will release the device and create it again.
+ ResetDevice();
+ }
+
+ return true;
+}
+
+int VideoRenderDirect3D9::CloseDevice()
+{
+ CriticalSectionScoped cs(_refD3DCritsect);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "VideoRenderDirect3D9::CloseDevice");
+
+ if (_pTextureLogo != NULL)
+ {
+ _pTextureLogo->Release();
+ _pTextureLogo = NULL;
+ }
+
+ if (_pVB != NULL)
+ {
+ _pVB->Release();
+ _pVB = NULL;
+ }
+
+ if (_pd3dDevice != NULL)
+ {
+ _pd3dDevice->Release();
+ _pd3dDevice = NULL;
+ }
+
+ if (_pD3D != NULL)
+ {
+ _pD3D->Release();
+ _pD3D = NULL;
+ }
+
+ if (_pd3dSurface != NULL)
+ _pd3dSurface->Release();
+ return 0;
+}
+
+D3D9Channel* VideoRenderDirect3D9::GetD3DChannel(int channel)
+{
+ std::map<int, D3D9Channel*>::iterator ddIt;
+ ddIt = _d3dChannels.find(channel & 0x0000ffff);
+ D3D9Channel* ddobj = NULL;
+ if (ddIt != _d3dChannels.end())
+ {
+ ddobj = ddIt->second;
+ }
+ if (ddobj == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D render failed to find channel");
+ return NULL;
+ }
+ return ddobj;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::DeleteChannel(const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+
+ std::multimap<int, unsigned int>::iterator it;
+ it = _d3dZorder.begin();
+ while (it != _d3dZorder.end())
+ {
+ if ((streamId & 0x0000ffff) == (it->second & 0x0000ffff))
+ {
+ it = _d3dZorder.erase(it);
+ break;
+ }
+ it++;
+ }
+
+ std::map<int, D3D9Channel*>::iterator ddIt;
+ ddIt = _d3dChannels.find(streamId & 0x0000ffff);
+ D3D9Channel* ddobj = NULL;
+ if (ddIt != _d3dChannels.end())
+ {
+ delete ddIt->second;
+ _d3dChannels.erase(ddIt);
+ return 0;
+ }
+ return -1;
+}
+
+VideoRenderCallback* VideoRenderDirect3D9::CreateChannel(const WebRtc_UWord32 channel,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+ //FIXME this should be done in VideoAPIWindows? stop the frame deliver first
+ //remove the old channel
+ DeleteChannel(channel);
+
+ D3D9Channel* d3dChannel = new D3D9Channel(_pd3dDevice,
+ &_refD3DCritsect, _trace);
+ d3dChannel->SetStreamSettings(0, zOrder, left, top, right, bottom);
+
+ // store channel
+ _d3dChannels[channel & 0x0000ffff] = d3dChannel;
+
+ // store Z order
+ // default streamID is 0
+ _d3dZorder.insert(
+ std::pair<int, unsigned int>(zOrder, channel & 0x0000ffff));
+
+ return d3dChannel;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::GetStreamSettings(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom)
+{
+ std::map<int, D3D9Channel*>::iterator ddIt;
+ ddIt = _d3dChannels.find(channel & 0x0000ffff);
+ D3D9Channel* ddobj = NULL;
+ if (ddIt != _d3dChannels.end())
+ {
+ ddobj = ddIt->second;
+ }
+ if (ddobj == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D render failed to find channel");
+ return -1;
+ }
+ // Only allow one stream per channel, demuxing is
+ return ddobj->GetStreamSettings(0, zOrder, left, top, right, bottom);
+ //return ddobj->GetStreamSettings(streamId, zOrder, left, top, right, bottom);
+}
+
+int VideoRenderDirect3D9::UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB,
+ int offset,
+ float startWidth,
+ float startHeight,
+ float stopWidth,
+ float stopHeight)
+{
+ if (pVB == NULL)
+ return -1;
+
+ float left, right, top, bottom;
+
+ //update the vertice buffer
+ //0,1 => -1,1
+ left = startWidth * 2 - 1;
+ right = stopWidth * 2 - 1;
+
+ //0,1 => 1,-1
+ top = 1 - startHeight * 2;
+ bottom = 1 - stopHeight * 2;
+
+ CUSTOMVERTEX newVertices[] = {
+ //logo
+ { left, bottom, 0.0f, 0xffffffff, 0, 1 }, { left, top, 0.0f,
+ 0xffffffff, 0, 0 },
+ { right, bottom, 0.0f, 0xffffffff, 1, 1 }, { right, top, 0.0f,
+ 0xffffffff, 1, 0 }, };
+ // Now we fill the vertex buffer.
+ VOID* pVertices;
+ if (FAILED(pVB->Lock(sizeof(CUSTOMVERTEX) * offset, sizeof(newVertices),
+ (void**) &pVertices, 0)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to lock the vertex buffer.");
+ return -1;
+ }
+ memcpy(pVertices, newVertices, sizeof(newVertices));
+ pVB->Unlock();
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::StartRender()
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::StopRender()
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+bool VideoRenderDirect3D9::IsFullScreen()
+{
+ return _fullScreen;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::SetCropping(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::SetTransparentBackground(
+ const bool enable)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 colorText,
+ const WebRtc_UWord32 colorBg,
+ const float left,
+ const float top,
+ const float rigth,
+ const float bottom)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ if (!bitMap)
+ {
+ if (_pTextureLogo != NULL)
+ {
+ _pTextureLogo->Release();
+ _pTextureLogo = NULL;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "Remove bitmap.");
+ return 0;
+ }
+
+ // sanity
+ if (left > 1.0f || left < 0.0f ||
+ top > 1.0f || top < 0.0f ||
+ right > 1.0f || right < 0.0f ||
+ bottom > 1.0f || bottom < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D SetBitmap invalid parameter");
+ return -1;
+ }
+
+ if ((bottom <= top) || (right <= left))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D SetBitmap invalid parameter");
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_refD3DCritsect);
+
+ unsigned char* srcPtr;
+ HGDIOBJ oldhand;
+ BITMAPINFO pbi;
+ BITMAP bmap;
+ HDC hdcNew;
+ hdcNew = CreateCompatibleDC(0);
+ // Fill out the BITMAP structure.
+ GetObject((HBITMAP)bitMap, sizeof(bmap), &bmap);
+ //Select the bitmap handle into the new device context.
+ oldhand = SelectObject(hdcNew, (HGDIOBJ) bitMap);
+ // we are done with this object
+ DeleteObject(oldhand);
+ pbi.bmiHeader.biSize = 40;
+ pbi.bmiHeader.biWidth = bmap.bmWidth;
+ pbi.bmiHeader.biHeight = bmap.bmHeight;
+ pbi.bmiHeader.biPlanes = 1;
+ pbi.bmiHeader.biBitCount = bmap.bmBitsPixel;
+ pbi.bmiHeader.biCompression = BI_RGB;
+ pbi.bmiHeader.biSizeImage = bmap.bmWidth * bmap.bmHeight * 3;
+ srcPtr = new unsigned char[bmap.bmWidth * bmap.bmHeight * 4];
+ // the original un-stretched image in RGB24
+ int pixelHeight = GetDIBits(hdcNew, (HBITMAP)bitMap, 0, bmap.bmHeight, srcPtr, &pbi,
+ DIB_RGB_COLORS);
+ if (pixelHeight == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D failed to GetDIBits in SetBitmap");
+ return -1;
+ }
+ DeleteDC(hdcNew);
+ if (pbi.bmiHeader.biBitCount != 24 && pbi.bmiHeader.biBitCount != 32)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D failed to SetBitmap invalid bit depth");
+ return -1;
+ }
+
+ HRESULT ret;
+ //release the previous logo texture
+ if (_pTextureLogo != NULL)
+ {
+ _pTextureLogo->Release();
+ _pTextureLogo = NULL;
+ }
+ ret = _pd3dDevice->CreateTexture(bmap.bmWidth, bmap.bmHeight, 1, 0,
+ D3DFMT_A8R8G8B8, D3DPOOL_MANAGED,
+ &_pTextureLogo, NULL);
+ if (FAILED(ret))
+ {
+ _pTextureLogo = NULL;
+ return -1;
+ }
+ if (!_pTextureLogo)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Texture for rendering not initialized.");
+ return -1;
+ }
+
+ D3DLOCKED_RECT lr;
+ if (FAILED(_pTextureLogo->LockRect(0, &lr, NULL, 0)))
+ {
+ return -1;
+ }
+ unsigned char* dstPtr = (UCHAR*) lr.pBits;
+ int pitch = bmap.bmWidth * 4;
+
+ if (pbi.bmiHeader.biBitCount == 24)
+ {
+ ConvertRGB24ToARGB(srcPtr, dstPtr, bmap.bmWidth, bmap.bmHeight, 0);
+ }
+ else
+ {
+ unsigned char* srcTmp = srcPtr + (bmap.bmWidth * 4) * (bmap.bmHeight - 1);
+ for (int i = 0; i < bmap.bmHeight; ++i)
+ {
+ memcpy(dstPtr, srcTmp, bmap.bmWidth * 4);
+ srcTmp -= bmap.bmWidth * 4;
+ dstPtr += pitch;
+ }
+ }
+
+ delete srcPtr;
+ if (FAILED(_pTextureLogo->UnlockRect(0)))
+ {
+ return -1;
+ }
+
+ if (colorKey)
+ {
+ DDCOLORKEY* ddColorKey =
+ static_cast<DDCOLORKEY*> (const_cast<void*> (colorKey));
+ SetTransparentColor(_pTextureLogo, ddColorKey, bmap.bmWidth,
+ bmap.bmHeight);
+ }
+
+ //update the vertice buffer
+ //0,1 => -1,1
+ _logoLeft = left;
+ _logoRight = right;
+
+ //0,1 => 1,-1
+ _logoTop = top;
+ _logoBottom = bottom;
+
+ return 0;
+
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::GetGraphicsMemory(WebRtc_UWord64& totalMemory,
+ WebRtc_UWord64& availableMemory)
+{
+ if (_totalMemory == -1 || _availableMemory == -1)
+ {
+ totalMemory = 0;
+ availableMemory = 0;
+ return -1;
+ }
+ totalMemory = _totalMemory;
+ availableMemory = _availableMemory;
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirect3D9::ConfigureRenderer(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ std::map<int, D3D9Channel*>::iterator ddIt;
+ ddIt = _d3dChannels.find(channel & 0x0000ffff);
+ D3D9Channel* ddobj = NULL;
+ if (ddIt != _d3dChannels.end())
+ {
+ ddobj = ddIt->second;
+ }
+ if (ddobj == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct3D render failed to find channel");
+ return -1;
+ }
+ // Only allow one stream per channel, demuxing is
+ ddobj->SetStreamSettings(0, zOrder, left, top, right, bottom);
+
+ return 0;
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h
new file mode 100644
index 0000000..6d6fef3
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h
@@ -0,0 +1,267 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
+
+// WebRtc includes
+#include "i_video_render_win.h"
+
+#include <d3d9.h>
+#include <d3dx9.h>
+#include "ddraw.h"
+
+#include <Map>
+
+// Added
+#include "video_render_defines.h"
+
+#pragma comment(lib, "d3d9.lib") // located in DirectX SDK
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class Trace;
+class ThreadWrapper;
+
+class D3D9Channel: public VideoRenderCallback
+{
+public:
+ D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
+ CriticalSectionWrapper* critSect, Trace* trace);
+
+ virtual ~D3D9Channel();
+
+ // Inherited from VideoRencerCallback, called from VideoAPI class.
+ // Called when the incomming frame size and/or number of streams in mix changes
+ virtual int FrameSizeChange(int width, int height, int numberOfStreams);
+
+ // A new frame is delivered
+ virtual int DeliverFrame(unsigned char* buffer,
+ int bufferSize,
+ unsigned int timeStame90kHz);
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+ // Called to check if the video frame is updated.
+ int IsUpdated(bool& isUpdated);
+ // Called after the video frame has been render to the screen
+ int RenderOffFrame();
+ // Called to get the texture that contains the video frame
+ LPDIRECT3DTEXTURE9 GetTexture();
+ // Called to get the texture(video frame) size
+ int GetTextureWidth();
+ int GetTextureHeight();
+ //
+ void SetStreamSettings(WebRtc_UWord16 streamId,
+ WebRtc_UWord32 zOrder,
+ float startWidth,
+ float startHeight,
+ float stopWidth,
+ float stopHeight);
+ int GetStreamSettings(WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& startWidth,
+ float& startHeight,
+ float& stopWidth,
+ float& stopHeight);
+
+ int ReleaseTexture();
+ int RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice);
+
+protected:
+
+private:
+ //critical section passed from the owner
+ CriticalSectionWrapper* _critSect;
+ LPDIRECT3DDEVICE9 _pd3dDevice;
+ LPDIRECT3DTEXTURE9 _pTexture;
+
+ bool _bufferIsUpdated;
+ // the frame size
+ int _width;
+ int _height;
+ //sream settings
+ //TODO support multiple streams in one channel
+ WebRtc_UWord16 _streamId;
+ WebRtc_UWord32 _zOrder;
+ float _startWidth;
+ float _startHeight;
+ float _stopWidth;
+ float _stopHeight;
+};
+
+class VideoRenderDirect3D9: IVideoRenderWin
+{
+public:
+ VideoRenderDirect3D9(Trace* trace, HWND hWnd, bool fullScreen);
+ ~VideoRenderDirect3D9();
+
+public:
+ //IVideoRenderWin
+
+ /**************************************************************************
+ *
+ * Init
+ *
+ ***************************************************************************/
+ virtual WebRtc_Word32 Init();
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+ virtual VideoRenderCallback
+ * CreateChannel(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom);
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual bool IsFullScreen();
+
+ virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
+ WebRtc_UWord64& availableMemory);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 colorText,
+ const WebRtc_UWord32 colorBg,
+ const float left,
+ const float top,
+ const float rigth,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+public:
+ // Get a channel by channel id
+ D3D9Channel* GetD3DChannel(int channel);
+ int UpdateRenderSurface();
+
+protected:
+ // The thread rendering the screen
+ static bool ScreenUpdateThreadProc(void* obj);
+ bool ScreenUpdateProcess();
+
+private:
+ // Init/close the d3d device
+ int InitDevice();
+ int CloseDevice();
+
+ // Transparent related functions
+ int SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
+ DDCOLORKEY* transparentColorKey,
+ DWORD width,
+ DWORD height);
+
+ CriticalSectionWrapper& _refD3DCritsect;
+ Trace* _trace;
+ ThreadWrapper* _screenUpdateThread;
+ EventWrapper* _screenUpdateEvent;
+
+ HWND _hWnd;
+ bool _fullScreen;
+ RECT _originalHwndRect;
+ //FIXME we probably don't need this since all the information can be get from _d3dChannels
+ int _channel;
+ //Window size
+ UINT _winWidth;
+ UINT _winHeight;
+
+ // Device
+ LPDIRECT3D9 _pD3D; // Used to create the D3DDevice
+ LPDIRECT3DDEVICE9 _pd3dDevice; // Our rendering device
+ LPDIRECT3DVERTEXBUFFER9 _pVB; // Buffer to hold Vertices
+ LPDIRECT3DTEXTURE9 _pTextureLogo;
+
+ std::map<int, D3D9Channel*> _d3dChannels;
+ std::multimap<int, unsigned int> _d3dZorder;
+
+ // The position where the logo will be placed
+ float _logoLeft;
+ float _logoTop;
+ float _logoRight;
+ float _logoBottom;
+
+ typedef HRESULT (WINAPI *DIRECT3DCREATE9EX)(UINT SDKVersion, IDirect3D9Ex**);
+ LPDIRECT3DSURFACE9 _pd3dSurface;
+
+ DWORD GetVertexProcessingCaps();
+ int InitializeD3D(HWND hWnd, D3DPRESENT_PARAMETERS* pd3dpp);
+
+ D3DPRESENT_PARAMETERS _d3dpp;
+ int ResetDevice();
+
+ int UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB, int offset,
+ float startWidth, float startHeight,
+ float stopWidth, float stopHeight);
+
+ //code for providing graphics settings
+ DWORD _totalMemory;
+ DWORD _availableMemory;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc
new file mode 100644
index 0000000..034f710
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc
@@ -0,0 +1,4012 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_render_directdraw.h"
+#include "video_render_windows_impl.h"
+#include "Windows.h"
+#include <ddraw.h>
+#include <assert.h>
+#include <initguid.h>
+#include <MMSystem.h> // timeGetTime
+DEFINE_GUID( IID_IDirectDraw7,0x15e65ec0,0x3b9c,0x11d2,0xb9,0x2f,0x00,0x60,0x97,0x97,0xea,0x5b );
+
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+#include "trace.h"
+#include "critical_section_wrapper.h"
+//#include "VideoErrors.h"
+
+// Added
+#include "module_common_types.h"
+
+#pragma warning(disable: 4355) // 'this' : used in base member initializer list
+// picture in picture do we need overlay? answer no we can blit directly
+// conference is easy since we can blt the quadrants seperatly
+
+// To determine if the driver supports DMA, retrieve the driver capabilities by calling the IDirectDraw::GetCaps method,
+// then look for DDBLTCAPS_READSYSMEM and/or DDBLTCAPS_WRITESYSMEM. If either of these flags is set, the device supports DMA.
+// Blt with SRCCOPY should do this can we use it?
+// investigate DDLOCK_NOSYSLOCK
+
+namespace webrtc {
+
+#define EXTRACT_BITS_RL(the_val, bits_start, bits_len) ((the_val >> (bits_start - 1)) & ((1 << bits_len) - 1))
+
+WindowsThreadCpuUsage::WindowsThreadCpuUsage() :
+ _lastGetCpuUsageTime(0),
+ _lastCpuUsageTime(0),
+ _hThread(::GetCurrentThread()),
+ _cores(0),
+ _lastCpuUsage(0)
+{
+
+ DWORD_PTR pmask, smask;
+ DWORD access = PROCESS_QUERY_INFORMATION;
+ if (GetProcessAffinityMask(
+ OpenProcess(access, false, GetCurrentProcessId()),
+ &pmask, &smask) != 0)
+ {
+
+ for (int i = 1; i < 33; i++)
+ {
+ if (EXTRACT_BITS_RL(pmask,i,1) == 0)
+ {
+ break;
+ }
+ _cores++;
+ }
+ //sanity
+ if (_cores > 32)
+ {
+ _cores = 32;
+ }
+ if (_cores < 1)
+ {
+ _cores = 1;
+ }
+ }
+ else
+ {
+ _cores = 1;
+ }
+ GetCpuUsage();
+}
+
+//in % since last call
+int WindowsThreadCpuUsage::GetCpuUsage()
+{
+ DWORD now = timeGetTime();
+
+ _int64 newTime = 0;
+ FILETIME creationTime;
+ FILETIME exitTime;
+ _int64 kernelTime = 0;
+ _int64 userTime = 0;
+ if (GetThreadTimes(_hThread, (FILETIME*) &creationTime, &exitTime,
+ (FILETIME*) &kernelTime, (FILETIME*) &userTime) != 0)
+ {
+ newTime = (kernelTime + userTime);
+ }
+ if (newTime == 0)
+ {
+ _lastGetCpuUsageTime = now;
+ return _lastCpuUsage;
+ }
+
+ // calculate the time difference since last call
+ const DWORD diffTime = (now - _lastGetCpuUsageTime);
+ _lastGetCpuUsageTime = now;
+
+ if (newTime < _lastCpuUsageTime)
+ {
+ _lastCpuUsageTime = newTime;
+ return _lastCpuUsage;
+ }
+ const int cpuDiff = (int) (newTime - _lastCpuUsageTime) / 10000;
+ _lastCpuUsageTime = newTime;
+
+ // calculate the CPU usage
+
+ _lastCpuUsage = (int) (float((cpuDiff * 100)) / (diffTime * _cores) + 0.5f);
+
+ if (_lastCpuUsage > 100)
+ {
+ _lastCpuUsage = 100;
+ }
+ return _lastCpuUsage;
+
+}
+
+DirectDrawStreamSettings::DirectDrawStreamSettings() :
+ _startWidth(0.0F),
+ _stopWidth(1.0F),
+ _startHeight(0.0F),
+ _stopHeight(1.0F),
+ _cropStartWidth(0.0F),
+ _cropStopWidth(1.0F),
+ _cropStartHeight(0.0F),
+ _cropStopHeight(1.0F)
+{
+}
+;
+
+DirectDrawBitmapSettings::DirectDrawBitmapSettings() :
+ _transparentBitMap(NULL),
+ _transparentBitmapLeft(0.0f),
+ _transparentBitmapRight(1.0f),
+ _transparentBitmapTop(0.0f),
+ _transparentBitmapBottom(1.0f),
+ _transparentBitmapWidth(0),
+ _transparentBitmapHeight(0),
+ _transparentBitmapColorKey(NULL),
+ _transparentBitmapSurface(NULL)
+{
+}
+;
+
+DirectDrawBitmapSettings::~DirectDrawBitmapSettings()
+{
+ if (_transparentBitmapColorKey)
+ {
+ delete _transparentBitmapColorKey;
+ }
+ if (_transparentBitmapSurface)
+ {
+ _transparentBitmapSurface->Release();
+ }
+ _transparentBitmapColorKey = NULL;
+ _transparentBitmapSurface = NULL;
+}
+;
+
+int DirectDrawBitmapSettings::SetBitmap(Trace* _trace,
+ DirectDraw* directDraw)
+{
+ VideoFrame tempVideoBuffer;
+ HGDIOBJ oldhand;
+ BITMAPINFO pbi;
+ BITMAP bmap;
+ HDC hdcNew;
+
+ hdcNew = CreateCompatibleDC(0);
+
+ // Fill out the BITMAP structure.
+ GetObject(_transparentBitMap, sizeof(bmap), &bmap);
+
+ //Select the bitmap handle into the new device context.
+ oldhand = SelectObject(hdcNew, (HGDIOBJ) _transparentBitMap);
+
+ // we are done with this object
+ DeleteObject(oldhand);
+
+ pbi.bmiHeader.biSize = 40;
+ pbi.bmiHeader.biWidth = bmap.bmWidth;
+ pbi.bmiHeader.biHeight = bmap.bmHeight;
+ pbi.bmiHeader.biPlanes = 1;
+ pbi.bmiHeader.biBitCount = bmap.bmBitsPixel;
+ pbi.bmiHeader.biCompression = BI_RGB;
+ pbi.bmiHeader.biSizeImage = bmap.bmWidth * bmap.bmHeight * 3;
+
+ tempVideoBuffer.VerifyAndAllocate(bmap.bmWidth * bmap.bmHeight * 4);
+
+ // the original un-stretched image in RGB24
+ // todo is there another struct for pbi purify reports read of 24 bytes larger than size
+ int pixelHeight = GetDIBits(hdcNew, _transparentBitMap, 0, bmap.bmHeight,
+ tempVideoBuffer.Buffer(), &pbi, DIB_RGB_COLORS);
+ if (pixelHeight == 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to GetDIBits in SetBitmap.");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ DeleteDC(hdcNew);
+
+ if (pbi.bmiHeader.biBitCount != 24 && pbi.bmiHeader.biBitCount != 32)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetBitmap invalid bit depth");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ DirectDrawSurfaceDesc ddsd;
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH | DDSD_PIXELFORMAT;
+ ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
+ ddsd.dwHeight = bmap.bmHeight;
+ ddsd.dwWidth = bmap.bmWidth;
+
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+
+ _transparentBitmapWidth = bmap.bmWidth;
+ _transparentBitmapHeight = bmap.bmHeight;
+
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 32;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+
+ if (_transparentBitmapSurface)
+ {
+ _transparentBitmapSurface->Release();
+ _transparentBitmapSurface = NULL;
+ }
+
+ HRESULT ddrval =
+ directDraw->CreateSurface(&ddsd, &_transparentBitmapSurface, NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _transparentBitmapSurface: 0x%x",
+ ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ memset(&ddsd, 0, sizeof(DDSURFACEDESC));
+ ddsd.dwSize = sizeof(DDSURFACEDESC);
+ ddrval = _transparentBitmapSurface->Lock(NULL, &ddsd, DDLOCK_WAIT, NULL);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ ddrval = _transparentBitmapSurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw failed to restore lost _transparentBitmapSurface");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw restored lost _transparentBitmapSurface");
+
+ ddrval
+ = _transparentBitmapSurface->Lock(NULL, &ddsd, DDLOCK_WAIT,
+ NULL);
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "DirectDraw lock error 0x%x _transparentBitmapSurface",
+ ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ }
+ unsigned char* dstPtr = (unsigned char*) ddsd.lpSurface;
+ unsigned char* srcPtr = (unsigned char*) tempVideoBuffer.Buffer();
+
+ int pitch = bmap.bmWidth * 4;
+ if (ddsd.dwFlags & DDSD_PITCH)
+ {
+ pitch = ddsd.lPitch;
+ }
+
+ if (pbi.bmiHeader.biBitCount == 24)
+ {
+ ConvertRGB24ToARGB(srcPtr, dstPtr, bmap.bmWidth, bmap.bmHeight,
+ 0);
+ }
+ else
+ {
+ srcPtr += (bmap.bmWidth * 4) * (bmap.bmHeight - 1);
+
+ for (int i = 0; i < bmap.bmHeight; ++i)
+ {
+ memcpy(dstPtr, srcPtr, bmap.bmWidth * 4);
+ srcPtr -= bmap.bmWidth * 4;
+ dstPtr += pitch;
+ }
+ }
+
+ _transparentBitmapSurface->Unlock(NULL);
+ return 0;
+}
+/**
+ *
+ * DirectDrawTextSettings
+ *
+ */
+DirectDrawTextSettings::DirectDrawTextSettings() :
+ _ptrText(NULL),
+ _textLength(0),
+ _colorRefText(RGB(255, 255, 255)), // white
+ _colorRefBackground(RGB(0, 0, 0)), // black
+ _textLeft(0.0f),
+ _textRight(0.0f),
+ _textTop(0.0f),
+ _textBottom(0.0f),
+ _transparent(true)
+{
+}
+
+DirectDrawTextSettings::~DirectDrawTextSettings()
+{
+ if (_ptrText)
+ {
+ delete[] _ptrText;
+ }
+}
+
+int DirectDrawTextSettings::SetText(const char* text, int textLength,
+ COLORREF colorText, COLORREF colorBg,
+ float left, float top, float right,
+ float bottom)
+{
+ if (_ptrText)
+ {
+ delete[] _ptrText;
+ }
+ _ptrText = new char[textLength];
+ memcpy(_ptrText, text, textLength);
+ _textLength = textLength;
+ _colorRefText = colorText;
+ _colorRefBackground = colorBg;
+ //_transparent = transparent;
+ _textLeft = left;
+ _textRight = right;
+ _textTop = top;
+ _textBottom = bottom;
+ return 0;
+}
+
+/**
+ *
+ * DirectDrawChannel
+ *
+ *
+ */
+
+// this need to have a refcount dueto multiple HWNDS demux
+DirectDrawChannel::DirectDrawChannel(DirectDraw* directDraw,
+ VideoType blitVideoType,
+ VideoType incomingVideoType,
+ VideoType screenVideoType,
+ VideoRenderDirectDraw* owner) :
+
+ _critSect(CriticalSectionWrapper::CreateCriticalSection()), _refCount(1),
+ _width(0), _height(0), _numberOfStreams(0), _doubleBuffer(false),
+ _directDraw(directDraw), _offScreenSurface(NULL),
+ _offScreenSurfaceNext(NULL), _incomingVideoType(incomingVideoType),
+ _blitVideoType(blitVideoType),
+ _originalBlitVideoType(blitVideoType),
+ _screenVideoType(screenVideoType), _deliverInScreenType(false),
+ _owner(owner)
+{
+ _directDraw->AddRef();
+}
+
+DirectDrawChannel::~DirectDrawChannel()
+{
+ if (_directDraw)
+ {
+ _directDraw->Release();
+ }
+ if (_offScreenSurface)
+ {
+ _offScreenSurface->Release();
+ }
+ if (_offScreenSurfaceNext)
+ {
+ _offScreenSurfaceNext->Release();
+ }
+ std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
+ _streamIdToSettings.begin();
+ while (it != _streamIdToSettings.end())
+ {
+ DirectDrawStreamSettings* streamSettings = it->second;
+ if (streamSettings)
+ {
+ delete streamSettings;
+ }
+ it = _streamIdToSettings.erase(it);
+ }
+ delete _critSect;
+}
+
+void DirectDrawChannel::AddRef()
+{
+ CriticalSectionScoped cs(*_critSect);
+ _refCount++;
+}
+
+void DirectDrawChannel::Release()
+{
+ bool deleteObj = false;
+ _critSect->Enter();
+ _refCount--;
+ if (_refCount == 0)
+ {
+ deleteObj = true;
+ }
+ _critSect->Leave();
+
+ if (deleteObj)
+ {
+ delete this;
+ }
+}
+
+void DirectDrawChannel::SetStreamSettings(VideoRenderDirectDraw* DDobj,
+ short streamId, float startWidth,
+ float startHeight,
+ float stopWidth, float stopHeight)
+{
+ // we can save 5 bits due to 16 byte alignment of the pointer
+ unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDobj);
+ lookupID &= 0xffffffffffffffe0;
+ lookupID <<= 11;
+ lookupID += streamId;
+
+ CriticalSectionScoped cs(*_critSect);
+
+ DirectDrawStreamSettings* streamSettings = NULL;
+
+ std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
+ _streamIdToSettings.find(lookupID);
+ if (it == _streamIdToSettings.end())
+ {
+ streamSettings = new DirectDrawStreamSettings();
+ _streamIdToSettings[lookupID] = streamSettings;
+ }
+ else
+ {
+ streamSettings = it->second;
+ }
+
+ streamSettings->_startHeight = startHeight;
+ streamSettings->_startWidth = startWidth;
+ streamSettings->_stopWidth = stopWidth;
+ streamSettings->_stopHeight = stopHeight;
+
+ _offScreenSurfaceUpdated = false;
+}
+
+void DirectDrawChannel::SetStreamCropSettings(VideoRenderDirectDraw* DDObj,
+ short streamId,
+ float startWidth,
+ float startHeight,
+ float stopWidth,
+ float stopHeight)
+{
+ unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDObj);
+ lookupID &= 0xffffffffffffffe0;
+ lookupID <<= 11;
+ lookupID += streamId;
+
+ CriticalSectionScoped cs(*_critSect);
+
+ DirectDrawStreamSettings* streamSettings = NULL;
+ std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
+ _streamIdToSettings.find(lookupID);
+ if (it == _streamIdToSettings.end())
+ {
+ streamSettings = new DirectDrawStreamSettings();
+ _streamIdToSettings[streamId] = streamSettings;
+ }
+ else
+ {
+ streamSettings = it->second;
+ }
+ streamSettings->_cropStartWidth = startWidth;
+ streamSettings->_cropStopWidth = stopWidth;
+ streamSettings->_cropStartHeight = startHeight;
+ streamSettings->_cropStopHeight = stopHeight;
+}
+
+int DirectDrawChannel::GetStreamSettings(VideoRenderDirectDraw* DDObj,
+ short streamId, float& startWidth,
+ float& startHeight,
+ float& stopWidth,
+ float& stopHeight)
+{
+ CriticalSectionScoped cs(*_critSect);
+
+ unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDObj);
+ lookupID &= 0xffffffffffffffe0;
+ lookupID <<= 11;
+ lookupID += streamId;
+
+ DirectDrawStreamSettings* streamSettings = NULL;
+ std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
+ _streamIdToSettings.find(lookupID);
+ if (it == _streamIdToSettings.end())
+ {
+ // Didn't find this stream...
+ return -1;
+ }
+ streamSettings = it->second;
+ startWidth = streamSettings->_startWidth;
+ startHeight = streamSettings->_startHeight;
+ stopWidth = streamSettings->_stopWidth;
+ stopHeight = streamSettings->_stopHeight;
+
+ return 0;
+}
+
+bool DirectDrawChannel::IsOffScreenSurfaceUpdated(VideoRenderDirectDraw* DDobj)
+{
+ CriticalSectionScoped cs(*_critSect);
+ return _offScreenSurfaceUpdated;
+}
+
+void DirectDrawChannel::GetLargestSize(RECT* mixingRect)
+{
+ CriticalSectionScoped cs(*_critSect);
+ if (mixingRect)
+ {
+ if (mixingRect->bottom < _height)
+ {
+ mixingRect->bottom = _height;
+ }
+ if (mixingRect->right < _width)
+ {
+ mixingRect->right = _width;
+ }
+ }
+}
+
+int DirectDrawChannel::ChangeDeliverColorFormat(bool useScreenType)
+{
+ _deliverInScreenType = useScreenType;
+ return FrameSizeChange(0, 0, 0);
+}
+
+WebRtc_Word32 DirectDrawChannel::RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+{
+ CriticalSectionScoped cs(*_critSect);
+ if (_width != videoFrame.Width() || _height != videoFrame.Height())
+ {
+ if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
+ {
+ return -1;
+ }
+ }
+ return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
+ videoFrame.TimeStamp());
+}
+
+int DirectDrawChannel::FrameSizeChange(int width, int height,
+ int numberOfStreams)
+{
+ CriticalSectionScoped cs(*_critSect);
+
+ if (_directDraw == NULL)
+ {
+ return -1; // signal that we are not ready for the change
+ }
+ if (_width == width && _height == height && _offScreenSurface
+ && _offScreenSurfaceNext)
+ {
+ _numberOfStreams = numberOfStreams;
+ return 0;
+ }
+ if (_offScreenSurface)
+ {
+ _offScreenSurface->Release();
+ _offScreenSurface = NULL;
+ }
+ if (_offScreenSurfaceNext)
+ {
+ _offScreenSurfaceNext->Release();
+ _offScreenSurfaceNext = NULL;
+ }
+ if (width && height)
+ {
+ _width = width;
+ _height = height;
+ _numberOfStreams = numberOfStreams;
+ }
+
+ // create this channels offscreen buffer
+ DirectDrawSurfaceDesc ddsd;
+ HRESULT ddrval = DD_OK;
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH | DDSD_PIXELFORMAT;
+ ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
+ ddsd.dwHeight = _height;
+ ddsd.dwWidth = _width;
+ /*
+ char logStr[256];
+ _snprintf(logStr,256, "offscreen H:%d W:%d \n",_height, _width);
+ OutputDebugString(logStr);
+ */
+ //Fix for bad video driver on HP Mini. If it takes to long time to deliver a frame - try to blit using the same pixel format as used by the screen.
+ if (_deliverInScreenType && _screenVideoType != kUnknown)
+ {
+ //The HP mini netbook, which this fix for, uses the VIA processor.
+ //The measuring shows that this fix will impact systems with Intel processor, including Atom.
+ //So let's disable it here. If we really need this for VIA processor, we should have additional logic to detect
+ //the processor model.
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDrawChannel changing to screen video type");
+ //_blitVideoType=_screenVideoType;
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "DirectDrawChannel changing to originial blit video type %d",
+ _originalBlitVideoType);
+ _blitVideoType = _originalBlitVideoType;
+ }
+
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "DirectDrawChannel::FrameSizeChange height %d, width %d, _blitVideoType %d",
+ ddsd.dwHeight, ddsd.dwWidth, _blitVideoType);
+ switch (_blitVideoType)
+ {
+ case kYV12:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
+ ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('Y', 'V', '1', '2');
+ }
+ break;
+ case kYUY2:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
+ ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('Y', 'U', 'Y', '2');
+ }
+ break;
+ case kUYVY:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
+ ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('U', 'Y', 'V', 'Y');
+ }
+ break;
+ case kIYUV:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
+ ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('I', 'Y', 'U', 'V');
+ }
+ break;
+ case kARGB:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 32;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+ }
+ break;
+ case kRGB24:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 24;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+ }
+ break;
+ case kRGB565:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0x0000F800;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0x000007e0;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0x0000001F;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+ }
+ break;
+ case kARGB4444:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0x00000f00;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0x000000f0;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0x0000000f;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+ break;
+ }
+ case kARGB1555:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
+ ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
+ ddsd.ddpfPixelFormat.dwRBitMask = 0x00007C00;
+ ddsd.ddpfPixelFormat.dwGBitMask = 0x3E0;
+ ddsd.ddpfPixelFormat.dwBBitMask = 0x1F;
+ ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
+ break;
+ }
+ case kI420:
+ {
+ ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
+ ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
+ ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('I', '4', '2', '0');
+ }
+ break;
+ default:
+ ddrval = S_FALSE;
+ }
+
+ if (ddrval == DD_OK)
+ {
+ if (!_owner->IsPrimaryOrMixingSurfaceOnSystem())
+ {
+ ddrval
+ = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "CreateSurface failed for _offScreenSurface on VideoMemory, trying on System Memory");
+
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
+
+ ddsd.dwHeight = _height;
+ ddsd.dwWidth = _width;
+
+ ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
+ _blitVideoType = kARGB;
+
+ ddrval = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _offScreenSurface using SystemMemory: 0x%x",
+ ddrval);
+ }
+ ddrval = _directDraw->CreateSurface(&ddsd,
+ &_offScreenSurfaceNext,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
+ ddrval);
+ }
+ }
+ else
+ {
+ ddrval = _directDraw->CreateSurface(&ddsd,
+ &_offScreenSurfaceNext,
+ NULL);
+ if (ddrval == DDERR_OUTOFVIDEOMEMORY)
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "CreateSurface failed for _offScreenSurfaceNext on VideoMemory, trying on System Memory");
+
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
+
+ ddsd.dwHeight = _height;
+ ddsd.dwWidth = _width;
+
+ ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
+ _blitVideoType = kARGB;
+
+ ddrval = _directDraw->CreateSurface(&ddsd,
+ &_offScreenSurfaceNext,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
+ ddrval);
+ }
+ }
+ }
+ }
+ else
+ {
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
+
+ ddsd.dwHeight = _height;
+ ddsd.dwWidth = _width;
+
+ ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
+ if (_owner->CanBltFourCC())
+ {
+ _blitVideoType = kARGB;
+ }
+ else
+ {
+ _blitVideoType = _originalBlitVideoType;
+ }
+
+ ddrval
+ = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _offScreenSurface using SystemMemory: 0x%x",
+ ddrval);
+ }
+
+ ddrval = _directDraw->CreateSurface(&ddsd, &_offScreenSurfaceNext,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
+ ddrval);
+ }
+ }
+ }
+
+ if (FAILED(ddrval))
+ {
+ // failed to change size
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to CreateSurface : 0x%x", ddrval);
+ return -1;
+ }
+
+ return 0;
+}
+
+int DirectDrawChannel::DeliverFrame(unsigned char* buffer, int bufferSize,
+ unsigned int /*timeStamp90KHz*/)
+{
+ CriticalSectionScoped cs(*_critSect);
+
+ if (CalcBufferSize(_incomingVideoType, _width, _height)
+ != bufferSize)
+ {
+ // sanity
+ return -1;
+ }
+ if (!_offScreenSurface || !_offScreenSurfaceNext)
+ {
+ if (_width && _height && _numberOfStreams)
+ {
+ // our surface was lost recreate it
+ FrameSizeChange(_width, _height, _numberOfStreams);
+ }
+ return -1;
+ }
+ if (_offScreenSurface->IsLost() == DDERR_SURFACELOST)
+ {
+ HRESULT ddrval = _offScreenSurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ // failed to restore our surface remove it and it will be re-created in next frame
+ _offScreenSurface->Release();
+ _offScreenSurface = NULL;
+ _offScreenSurfaceNext->Release();
+ _offScreenSurfaceNext = NULL;
+ return -1;
+ }
+ ddrval = _offScreenSurfaceNext->Restore();
+ if (ddrval != DD_OK)
+ {
+ // failed to restore our surface remove it and it will be re-created in next frame
+ _offScreenSurface->Release();
+ _offScreenSurface = NULL;
+ _offScreenSurfaceNext->Release();
+ _offScreenSurfaceNext = NULL;
+ return -1;
+ }
+ }
+ _doubleBuffer = false;
+
+ // check if _offScreenSurfaceUpdated is true
+ DirectDrawSurface* offScreenSurface = _offScreenSurface;
+ {
+
+ if (_offScreenSurfaceUpdated)
+ {
+ // this frame is not yet rendered
+ offScreenSurface = _offScreenSurfaceNext;
+ _doubleBuffer = true;
+ }
+ }
+
+ DirectDrawSurfaceDesc ddsd;
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ HRESULT ddrval = offScreenSurface->Lock(NULL, &ddsd, DDLOCK_WAIT, NULL);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDrawChannel::DeliverFrame offScreenSurface lost");
+ ddrval = offScreenSurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ // failed to restore our surface remove it and it will be re-created in next frame
+ _offScreenSurface->Release();
+ _offScreenSurface = NULL;
+ _offScreenSurfaceNext->Release();
+ _offScreenSurfaceNext = NULL;
+ return -1;
+ }
+ return 0;
+ }
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDrawChannel::DeliverFrame failed to lock");
+ // failed to lock our surface remove it and it will be re-created in next frame
+ _offScreenSurface->Release();
+ _offScreenSurface = NULL;
+ _offScreenSurfaceNext->Release();
+ _offScreenSurfaceNext = NULL;
+ return -1;
+ }
+
+ unsigned char* ptr = (unsigned char*) ddsd.lpSurface;
+ // ddsd.lPitch; distance in bytes
+
+
+ switch (_incomingVideoType)
+ {
+ case kI420:
+ {
+ switch (_blitVideoType)
+ {
+ case kYUY2:
+ case kUYVY:
+ case kIYUV: // same as kYV12
+ case kYV12:
+ ConvertFromI420(buffer, _width,
+ _blitVideoType, 0,
+ _width, _height,
+ ptr);
+ break;
+ case kRGB24:
+ {
+ _tempRenderBuffer.VerifyAndAllocate(_width * _height * 3);
+ unsigned char *ptrTempBuffer = _tempRenderBuffer.Buffer();
+ ConvertFromI420(buffer, _width, kRGB24, 0, _width, _height,
+ ptrTempBuffer);
+ for (int i = 0; i < _height; i++)
+ {
+ memcpy(ptr, ptrTempBuffer, _width * 3);
+ ptrTempBuffer += _width * 3;
+ ptr += ddsd.lPitch;
+ }
+ break;
+ }
+ case kARGB:
+ ConvertFromI420(buffer, ddsd.lPitch, kARGB, 0,
+ _width, _height, ptr);
+ break;
+ case kARGB4444:
+ ConvertI420ToARGB4444(buffer, ptr, _width, _height,
+ (ddsd.lPitch >> 1) - _width);
+ break;
+ case kARGB1555:
+ ConvertI420ToARGB1555(buffer, ptr, _width, _height,
+ (ddsd.lPitch >> 1) - _width);
+ break;
+ case kRGB565:
+ {
+ _tempRenderBuffer.VerifyAndAllocate(_width * _height * 2);
+ unsigned char *ptrTempBuffer = _tempRenderBuffer.Buffer();
+ ConvertI420ToRGB565(buffer, ptrTempBuffer, _width, _height);
+ ptr += ddsd.lPitch * (_height - 1);
+ for (int i = 0; i < _height; i++)
+ {
+ memcpy(ptr, ptrTempBuffer, _width * 2);
+ ptrTempBuffer += _width * 2;
+ ptr -= ddsd.lPitch;
+ }
+ break;
+ }
+ default:
+ assert(!"DirectDrawChannel::DeliverFrame unknown blitVideoType");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDrawChannel::DeliverFrame unknown blitVideoType %d",
+ _blitVideoType);
+ }
+ break;
+ }
+ default:
+ assert(!"DirectDrawChannel::DeliverFrame wrong incomming video type");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDrawChannel::DeliverFrame wrong incomming %d",
+ _incomingVideoType);
+ }
+
+ _offScreenSurfaceUpdated = true;
+ offScreenSurface->Unlock(NULL);
+ return 0;
+}
+
+int DirectDrawChannel::BlitFromOffscreenBufferToMixingBuffer(
+ VideoRenderDirectDraw* DDobj,
+ short streamID,
+ DirectDrawSurface* mixingSurface,
+ RECT &hwndRect,
+ bool demuxing)
+{
+ HRESULT ddrval;
+ RECT srcRect;
+ RECT dstRect;
+ DirectDrawStreamSettings* streamSettings = NULL;
+ unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDobj);
+ lookupID &= 0xffffffffffffffe0;
+ lookupID <<= 11;
+ lookupID += streamID;
+
+ CriticalSectionScoped cs(*_critSect);
+
+ if (_offScreenSurface == NULL)
+ {
+ // The offscreen surface has been deleted but not restored yet
+ return 0;
+ }
+ if (mixingSurface == NULL)
+ {
+ // Not a valid input argument
+ return 0;
+ }
+
+ std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
+ _streamIdToSettings.find(lookupID);
+ if (it == _streamIdToSettings.end())
+ {
+ // ignore this stream id
+ return 0;
+ }
+ streamSettings = it->second;
+
+ int numberOfStreams = _numberOfStreams;
+ if (!demuxing)
+ {
+ numberOfStreams = 1; // treat as one stream if we only have one config
+ }
+
+ switch (numberOfStreams)
+ {
+ case 0:
+ return 0;
+ case 1:
+ {
+ // no demux
+ if (streamID > 0)
+ return 0;
+
+ ::SetRect(&srcRect, int(_width * streamSettings->_cropStartWidth),
+ int(_height * streamSettings->_cropStartHeight),
+ int(_width * streamSettings->_cropStopWidth), int(_height
+ * streamSettings->_cropStopHeight));
+
+ ::SetRect(&dstRect, int(hwndRect.right
+ * streamSettings->_startWidth), int(hwndRect.bottom
+ * streamSettings->_startHeight), int(hwndRect.right
+ * streamSettings->_stopWidth), int(hwndRect.bottom
+ * streamSettings->_stopHeight));
+ }
+ break;
+ case 2:
+ case 3:
+ case 4:
+ // classic quadrant demux
+ {
+ int width = _width >> 1;
+ int height = _height >> 1;
+ ::SetRect(&srcRect, int(width * streamSettings->_cropStartWidth),
+ int(height * streamSettings->_cropStartHeight), int(width
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+
+ ::SetRect(&dstRect, int(hwndRect.right
+ * streamSettings->_startWidth), int(hwndRect.bottom
+ * streamSettings->_startHeight), int(hwndRect.right
+ * streamSettings->_stopWidth), int(hwndRect.bottom
+ * streamSettings->_stopHeight));
+
+ // stream id to select quadrant
+ if (streamID == 1)
+ {
+ ::OffsetRect(&srcRect, width, 0);
+ }
+ if (streamID == 2)
+ {
+ ::OffsetRect(&srcRect, 0, height);
+ }
+ if (streamID == 3)
+ {
+ ::OffsetRect(&srcRect, width, height);
+ }
+ }
+ break;
+ case 5:
+ case 6:
+ {
+ const int width = (_width / (3 * 16)) * 16;
+ const int widthMidCol = width + ((_width % (16 * 3)) / 16) * 16;
+ const int height = _height / (2 * 16) * 16;
+ if (streamID == 1 || streamID == 4)
+ {
+ ::SetRect(&srcRect, int(widthMidCol
+ * streamSettings->_cropStartWidth), int(height
+ * streamSettings->_cropStartHeight), int(widthMidCol
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+ }
+ else
+ {
+ ::SetRect(&srcRect,
+ int(width * streamSettings->_cropStartWidth),
+ int(height * streamSettings->_cropStartHeight),
+ int(width * streamSettings->_cropStopWidth),
+ int(height * streamSettings->_cropStopHeight));
+ }
+ ::SetRect(&dstRect, int(hwndRect.right
+ * streamSettings->_startWidth), int(hwndRect.bottom
+ * streamSettings->_startHeight), int(hwndRect.right
+ * streamSettings->_stopWidth), int(hwndRect.bottom
+ * streamSettings->_stopHeight));
+
+ // stream id to select quadrant
+ switch (streamID)
+ {
+ case 1:
+ ::OffsetRect(&srcRect, width, 0);
+ break;
+ case 2:
+ ::OffsetRect(&srcRect, width + widthMidCol, 0);
+ break;
+ case 3:
+ ::OffsetRect(&srcRect, 0, height);
+ break;
+ case 4:
+ ::OffsetRect(&srcRect, width, height);
+ break;
+ case 5:
+ ::OffsetRect(&srcRect, width + widthMidCol, height);
+ break;
+ }
+ }
+ break;
+ case 7:
+ case 8:
+ case 9:
+
+ {
+ const int width = (_width / (3 * 16)) * 16;
+ const int widthMidCol = width + ((_width % (16 * 3)) / 16) * 16;
+ const int height = _height / (3 * 16) * 16;
+ const int heightMidRow = height + ((_height % (16 * 3)) / 16) * 16;
+
+ ::SetRect(&dstRect, int(hwndRect.right
+ * streamSettings->_startWidth), int(hwndRect.bottom
+ * streamSettings->_startHeight), int(hwndRect.right
+ * streamSettings->_stopWidth), int(hwndRect.bottom
+ * streamSettings->_stopHeight));
+
+ switch (streamID)
+ {
+ case 0:
+ //Size
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth), int(height
+ * streamSettings->_cropStartHeight), int(width
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+ //Position
+ ::OffsetRect(&srcRect, 0, 0);
+ break;
+ case 1:
+ ::SetRect(
+ &srcRect,
+ int(widthMidCol * streamSettings->_cropStartWidth),
+ int(height * streamSettings->_cropStartHeight),
+ int(widthMidCol * streamSettings->_cropStopWidth),
+ int(height * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width, 0);
+ break;
+ case 2:
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth), int(height
+ * streamSettings->_cropStartHeight), int(width
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width + widthMidCol, 0);
+ break;
+ case 3:
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth),
+ int(heightMidRow
+ * streamSettings->_cropStartHeight),
+ int(width * streamSettings->_cropStopWidth),
+ int(heightMidRow
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, 0, height);
+ break;
+ case 4:
+ ::SetRect(
+ &srcRect,
+ int(widthMidCol * streamSettings->_cropStartWidth),
+ int(heightMidRow
+ * streamSettings->_cropStartHeight),
+ int(widthMidCol * streamSettings->_cropStopWidth),
+ int(heightMidRow
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width, height);
+
+ break;
+ case 5:
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth),
+ int(heightMidRow
+ * streamSettings->_cropStartHeight),
+ int(width * streamSettings->_cropStopWidth),
+ int(heightMidRow
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width + widthMidCol, height);
+ break;
+ case 6:
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth), int(height
+ * streamSettings->_cropStartHeight), int(width
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, 0, height + heightMidRow);
+ break;
+ case 7:
+ ::SetRect(
+ &srcRect,
+ int(widthMidCol * streamSettings->_cropStartWidth),
+ int(height * streamSettings->_cropStartHeight),
+ int(widthMidCol * streamSettings->_cropStopWidth),
+ int(height * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width, height + heightMidRow);
+ break;
+ case 8:
+ ::SetRect(&srcRect, int(width
+ * streamSettings->_cropStartWidth), int(height
+ * streamSettings->_cropStartHeight), int(width
+ * streamSettings->_cropStopWidth), int(height
+ * streamSettings->_cropStopHeight));
+ ::OffsetRect(&srcRect, width + widthMidCol, height
+ + heightMidRow);
+ break;
+ }
+ }
+ break;
+ case 10:
+ case 11:
+ case 12:
+ case 13:
+ case 14:
+ case 15:
+ case 16:
+ default:
+ {
+ ::SetRect(&srcRect, int(_width * streamSettings->_cropStartWidth),
+ int(_height * streamSettings->_cropStartHeight),
+ int(_width * streamSettings->_cropStopWidth), int(_height
+ * streamSettings->_cropStopHeight));
+
+ ::SetRect(&dstRect, int(hwndRect.right
+ * streamSettings->_startWidth), int(hwndRect.bottom
+ * streamSettings->_startHeight), int(hwndRect.right
+ * streamSettings->_stopWidth), int(hwndRect.bottom
+ * streamSettings->_stopHeight));
+ }
+ }
+
+ if (dstRect.right > hwndRect.right)
+ {
+ srcRect.right -= (int) ((float) (srcRect.right - srcRect.left)
+ * ((float) (dstRect.right - hwndRect.right)
+ / (float) (dstRect.right - dstRect.left)));
+ dstRect.right = hwndRect.right;
+ }
+ if (dstRect.left < hwndRect.left)
+ {
+ srcRect.left += (int) ((float) (srcRect.right - srcRect.left)
+ * ((float) (hwndRect.left - dstRect.left)
+ / (float) (dstRect.right - dstRect.left)));
+ dstRect.left = hwndRect.left;
+ }
+ if (dstRect.bottom > hwndRect.bottom)
+ {
+ srcRect.bottom -= (int) ((float) (srcRect.bottom - srcRect.top)
+ * ((float) (dstRect.bottom - hwndRect.bottom)
+ / (float) (dstRect.bottom - dstRect.top)));
+ dstRect.bottom = hwndRect.bottom;
+ }
+ if (dstRect.top < hwndRect.top)
+ {
+ srcRect.top += (int) ((float) (srcRect.bottom - srcRect.top)
+ * ((float) (hwndRect.top - dstRect.top)
+ / (float) (dstRect.bottom - dstRect.top)));
+ dstRect.top = hwndRect.top;
+ }
+
+ DDBLTFX ddbltfx;
+ ZeroMemory(&ddbltfx, sizeof(ddbltfx));
+ ddbltfx.dwSize = sizeof(ddbltfx);
+ ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
+
+ // wait for the _mixingSurface to be available
+ ddrval = mixingSurface->Blt(&dstRect, _offScreenSurface, &srcRect,
+ DDBLT_WAIT | DDBLT_DDFX, &ddbltfx);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "mixingSurface->Blt surface lost");
+ ddrval = mixingSurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ // we dont own the surface just report the error
+ return -1;
+ }
+ }
+ else if (ddrval == DDERR_INVALIDRECT)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "mixingSurface->Blt DDERR_INVALIDRECT");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "dstRect co-ordinates - top: %d left: %d bottom: %d right: %d",
+ dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "srcRect co-ordinates - top: %d left: %d bottom: %d right: %d",
+ srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
+
+ // ignore
+ }
+ else if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "mixingSurface->Blt !DD_OK");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw blt mixingSurface BlitFromOffscreenBufferToMixingBuffer error 0x%x ",
+ ddrval);
+
+ //logging the co-ordinates and hwnd
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "dstRect co-ordinates - top: %d left: %d bottom: %d right: %d",
+ dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "srcRect co-ordinates - top: %d left: %d bottom: %d right: %d",
+ srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
+
+ /* char logStr[256];
+ _snprintf(logStr,256, "srcRect T:%d L:%d B:%d R:%d\n",srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
+ OutputDebugString(logStr);
+ char logStr1[256];
+ _snprintf(logStr1,256, "dstRect T:%d L:%d B:%d R:%d\n",dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
+ OutputDebugString(logStr1);
+ char logStr2[256];
+ _snprintf(logStr2,256, "error 0x:%x \n",ddrval);
+ OutputDebugString(logStr2);
+ */
+ // we dont own the surface just report the error
+ return -1;
+ }
+ if (_doubleBuffer)
+ {
+ DirectDrawSurface* oldOffScreenSurface = _offScreenSurface;
+ _offScreenSurface = _offScreenSurfaceNext;
+ _offScreenSurfaceNext = oldOffScreenSurface;
+ _doubleBuffer = false;
+ }
+ else
+ {
+ _offScreenSurfaceUpdated = false;
+ }
+ return 0;
+}
+
+/**
+ *
+ * VideoRenderDirectDraw
+ *
+ *
+ */
+
+VideoRenderDirectDraw::VideoRenderDirectDraw(Trace* trace,
+ HWND hWnd, bool fullscreen) :
+ _trace(trace),
+ _confCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+ _fullscreen(fullscreen),
+ _demuxing(false),
+ _transparentBackground(false),
+ _supportTransparency(false),
+ _canStretch(false),
+ _canMirrorLeftRight(false),
+ _clearMixingSurface(false),
+ _deliverInScreenType(false),
+ _renderModeWaitForCorrectScanLine(false),
+ _deliverInHalfFrameRate(false),
+ _deliverInQuarterFrameRate(false),
+ _bCanBltFourcc(true),
+ _frameChanged(false),
+ _processCount(0),
+ _hWnd(hWnd),
+ _screenRect(),
+ _mixingRect(),
+
+ _incomingVideoType(kUnknown),
+ _blitVideoType(kUnknown),
+ _rgbVideoType(kUnknown),
+
+ _directDraw(NULL),
+ _primarySurface(NULL),
+ _backSurface(NULL),
+ _mixingSurface(NULL),
+ _bitmapSettings(),
+ _textSettings(),
+ _directDrawChannels(),
+ _directDrawZorder(),
+
+ _fullScreenWaitEvent(EventWrapper::Create()),
+ _screenEvent(EventWrapper::Create()),
+ _screenRenderThread(
+ ThreadWrapper::CreateThread(
+ RemoteRenderingThreadProc,
+ this,
+ kRealtimePriority,
+ "Video_directdraw_thread")),
+ _blit(true), _lastRenderModeCpuUsage(-1), _totalMemory(-1),
+ _availableMemory(-1), _systemCPUUsage(0), _maxAllowedRenderTime(0),
+ _nrOfTooLongRenderTimes(0),
+ _isPrimaryOrMixingSurfaceOnSystem(false)
+{
+ SetRect(&_screenRect, 0, 0, 0, 0);
+ SetRect(&_mixingRect, 0, 0, 0, 0);
+ SetRect(&_originalHwndRect, 0, 0, 0, 0);
+ ::GetClientRect(_hWnd, &_hwndRect);
+}
+
+VideoRenderDirectDraw::~VideoRenderDirectDraw()
+{
+ ThreadWrapper* temp = _screenRenderThread;
+ _screenRenderThread = NULL;
+ if (temp)
+ {
+ temp->SetNotAlive();
+ _screenEvent->Set();
+ _screenEvent->StopTimer();
+ _fullScreenWaitEvent->StopTimer();
+
+ if (temp->Stop())
+ {
+ delete temp;
+ }
+ }
+ delete _screenEvent;
+ delete _fullScreenWaitEvent;
+
+ std::map<int, DirectDrawChannel*>::iterator it;
+ it = _directDrawChannels.begin();
+ while (it != _directDrawChannels.end())
+ {
+ it->second->Release();
+ it = _directDrawChannels.erase(it);
+ }
+ if (_primarySurface)
+ {
+ _primarySurface->Release();
+ }
+ if (_mixingSurface)
+ {
+ _mixingSurface->Release();
+ }
+
+ std::map<unsigned char, DirectDrawBitmapSettings*>::iterator bitIt;
+
+ bitIt = _bitmapSettings.begin();
+ while (_bitmapSettings.end() != bitIt)
+ {
+ delete bitIt->second;
+ bitIt = _bitmapSettings.erase(bitIt);
+ }
+
+ std::map<unsigned char, DirectDrawTextSettings*>::iterator textIt;
+ textIt = _textSettings.begin();
+ while (_textSettings.end() != textIt)
+ {
+ delete textIt->second;
+ textIt = _textSettings.erase(textIt);
+ }
+ if (_directDraw)
+ {
+ _directDraw->Release();
+ if (_fullscreen)
+ {
+ // restore hwnd to original size and position
+ ::SetWindowPos(_hWnd, HWND_NOTOPMOST, _originalHwndRect.left,
+ _originalHwndRect.top, _originalHwndRect.right
+ - _originalHwndRect.left,
+ _originalHwndRect.bottom - _originalHwndRect.top,
+ SWP_FRAMECHANGED);
+ ::RedrawWindow(_hWnd, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
+ | RDW_ERASE);
+ ::RedrawWindow(NULL, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
+ | RDW_ERASE);
+ }
+ }
+ delete _confCritSect;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::Init()
+{
+ int retVal = 0;
+ HRESULT ddrval = DirectDrawCreateEx(NULL, (void**) &_directDraw,
+ IID_IDirectDraw7, NULL);
+ if (FAILED(ddrval) || NULL == _directDraw)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Failed to created DirectDraw7 object");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ retVal = CheckCapabilities();
+ if (retVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw CheckCapabilities failed");
+ return retVal;
+ }
+ if (_hWnd)
+ {
+ retVal = CreatePrimarySurface();
+ if (retVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to CreatePrimarySurface");
+ return retVal;
+ }
+ retVal = CreateMixingSurface();
+ if (retVal != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to CreateMixingSurface");
+ return retVal;
+ }
+ if (_screenRenderThread)
+ {
+ unsigned int tid;
+ _screenRenderThread->Start(tid);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Screen Render thread started, thread id: %d", tid);
+ }
+ DWORD freq = 0;
+ _directDraw->GetMonitorFrequency(&freq);
+ if (freq == 0)
+ {
+ freq = 60;
+ }
+ // Do this now to not do it in each render process loop
+ _maxAllowedRenderTime = (int) (1000 / freq * 0.8F);
+ _nrOfTooLongRenderTimes = 0;
+
+ _screenEvent->StartTimer(true, 1000 / freq);
+
+ _deliverInScreenType = false;
+ _renderModeWaitForCorrectScanLine = false;
+ _deliverInHalfFrameRate = false;
+ _deliverInQuarterFrameRate = false;
+
+ _lastRenderModeCpuUsage = -1;
+ if (_fullscreen)
+ {
+ _fullScreenWaitEvent->StartTimer(true, 1);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Screen freq %d", freq);
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Created DirectDraw object");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::GetGraphicsMemory(
+ WebRtc_UWord64& totalMemory,
+ WebRtc_UWord64& availableMemory)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ if (_totalMemory == -1 || _availableMemory == -1)
+ {
+ totalMemory = 0;
+ availableMemory = 0;
+ return -1;
+ }
+ totalMemory = _totalMemory;
+ availableMemory = _availableMemory;
+ return 0;
+}
+
+int VideoRenderDirectDraw::GetScreenResolution(int& screenWidth,
+ int& screenHeight)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ screenWidth = _screenRect.right - _screenRect.left;
+ screenHeight = _screenRect.bottom - _screenRect.top;
+ return 0;
+}
+
+int VideoRenderDirectDraw::UpdateSystemCPUUsage(int systemCPU)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+ if (systemCPU <= 100 && systemCPU >= 0)
+ {
+ _systemCPUUsage = systemCPU;
+ }
+ return 0;
+}
+
+int VideoRenderDirectDraw::CheckCapabilities()
+{
+ HRESULT ddrval = DD_OK;
+ DDCAPS ddcaps;
+ DDCAPS ddcapsEmul;
+ memset(&ddcaps, 0, sizeof(ddcaps));
+ memset(&ddcapsEmul, 0, sizeof(ddcapsEmul));
+ ddcaps.dwSize = sizeof(ddcaps);
+ ddcapsEmul.dwSize = sizeof(ddcapsEmul);
+ if (_directDraw == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw object not created");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (IsRectEmpty(&_screenRect))
+ {
+ ::GetWindowRect(GetDesktopWindow(), &_screenRect);
+ }
+ // Log Screen resolution
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "ScreenRect. Top: %d, left: %d, bottom: %d, right: %d",
+ _screenRect.top, _screenRect.left, _screenRect.bottom,
+ _screenRect.right);
+
+ bool fullAccelerationEnabled = false;
+ bool badDriver = false;
+ VideoRenderWindowsImpl::CheckHWDriver(badDriver, fullAccelerationEnabled);
+ if (!fullAccelerationEnabled)
+ {
+
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Direct draw Hardware acceleration is not enabled.");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_HWACC_NOT_ENABLED;
+
+ }
+
+ // ddcaps supported by the HW
+ // ddcapsEmul supported by the OS emulating the HW
+ ddrval = _directDraw->GetCaps(&ddcaps, &ddcapsEmul);
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw HW: could not get capabilities: %x", ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ unsigned int minVideoMemory = 3 * 4 * (_screenRect.right
+ * _screenRect.bottom); // assuming ARGB size (4 bytes)
+
+ // Store the memory for possible calls to GetMemory()
+ _totalMemory = ddcaps.dwVidMemTotal;
+ _availableMemory = ddcaps.dwVidMemFree;
+
+ if (ddcaps.dwVidMemFree < minVideoMemory)
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw HW does not have enough memory, freeMem:%d, requiredMem:%d",
+ ddcaps.dwVidMemFree, minVideoMemory);
+ // If memory is not available on the Video Card...allocate it on RAM
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw video memory, freeMem:%d, totalMem:%d",
+ ddcaps.dwVidMemFree, ddcaps.dwVidMemTotal);
+ }
+
+ /*
+ DirectDrawCaps ddsCaps ;
+ ZeroMemory(&ddsCaps, sizeof(ddsCaps)) ;
+ ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
+ DWORD memTotal=0;
+ DWORD memFree=0;
+ ddrval = _directDraw->GetAvailableVidMem(&ddsCaps, &memTotal, &memFree);
+ if(ddrval == DD_OK)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw video memory, freeMem:%d, totalMem:%d", memFree, memTotal);
+ }
+ */
+ // Determine if the hardware supports overlay deinterlacing
+ // bCanDeinterlace = (ddcaps.dwCaps2 & DDCAPS2_CANFLIPODDEVEN) ? 1 : 0;
+
+ // this fail since we check before we set the mode
+ // bool bCanFlip =(ddcaps.dwCaps & DDSCAPS_FLIP) ? 1 : 0;
+
+ // Determine if the hardware supports colorkeying
+ _supportTransparency = (ddcaps.dwCaps & DDCAPS_COLORKEY) ? 1 : 0;
+ if (_supportTransparency)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw support colorkey");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw don't support colorkey");
+ }
+
+ if (ddcaps.dwCaps2 & DDCAPS2_CANRENDERWINDOWED)
+ {
+ // required for _directDraw->FlipToGDISurface();
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw support CANRENDERWINDOWED");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw don't support CANRENDERWINDOWED");
+ }
+
+ // Determine if the hardware supports scaling during a blit
+ _canStretch = (ddcaps.dwCaps & DDCAPS_BLTSTRETCH) ? 1 : 0;
+ if (_canStretch)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw blit can stretch");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw blit can't stretch");
+ }
+
+ _canMirrorLeftRight = (ddcaps.dwFXAlphaCaps & DDBLTFX_MIRRORLEFTRIGHT) ? 1
+ : 0;
+ if (_canMirrorLeftRight)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw mirroring is supported");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw mirroring is not supported");
+ }
+
+ // Determine if the hardware supports color conversion during a blit
+ _bCanBltFourcc = (ddcaps.dwCaps & DDCAPS_BLTFOURCC) ? 1 : 0;
+ if (_bCanBltFourcc)
+ _bCanBltFourcc = (ddcaps.dwCKeyCaps & DDCKEYCAPS_DESTBLT) ? 1 : 0;
+
+ if (_bCanBltFourcc)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw can blit Fourcc");
+ DWORD i_codes;
+ ddrval = _directDraw->GetFourCCCodes(&i_codes, NULL);
+
+ if (i_codes > 0)
+ {
+ DWORD* pi_codes = new DWORD[i_codes];
+
+ ddrval = _directDraw->GetFourCCCodes(&i_codes, pi_codes);
+ for (unsigned int i = 0; i < i_codes && _blitVideoType
+ != kI420; i++)
+ {
+ DWORD w = pi_codes[i];
+ switch (w)
+ {
+ case MAKEFOURCC('I', '4', '2', '0'):
+ // _blitVideoType = kI420;
+ // not enabled since its not tested
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support I420");
+ break;
+ case MAKEFOURCC('I', 'Y', 'U', 'V'): // same as YV12
+ // _blitVideoType = kIYUV;
+ // not enabled since its not tested
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support IYUV");
+ break;
+ case MAKEFOURCC('U', 'Y', 'N', 'V'): // same shit different name
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support UYNV");
+ // not enabled since its not tested
+ break;
+ case MAKEFOURCC('Y', '4', '2', '2'): // same shit different name
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support Y422");
+ // not enabled since its not tested
+ break;
+ case MAKEFOURCC('Y', 'U', 'N', 'V'): // same shit different name
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support YUNV");
+ // not enabled since its not tested
+ break;
+ case MAKEFOURCC('Y', 'V', '1', '2'):
+ _blitVideoType = kYV12;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support YV12");
+ break;
+ case MAKEFOURCC('Y', 'U', 'Y', '2'):
+ if (_blitVideoType != kYV12)
+ {
+ _blitVideoType = kYUY2;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support YUY2");
+ break;
+ case MAKEFOURCC('U', 'Y', 'V', 'Y'):
+ if (_blitVideoType != kYV12)
+ {
+ _blitVideoType = kUYVY;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw support UYVY");
+ break;
+ default:
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ -1, "DirectDraw unknown blit type %x", w);
+ break;
+ }
+ }
+ delete[] pi_codes;
+ }
+ }
+ return 0;
+}
+
+int VideoRenderDirectDraw::Stop()
+{
+ _confCritSect->Enter();
+
+ _blit = false;
+
+ _confCritSect->Leave();
+ return 0;
+}
+
+bool VideoRenderDirectDraw::IsPrimaryOrMixingSurfaceOnSystem()
+{
+ return _isPrimaryOrMixingSurfaceOnSystem;
+}
+
+int VideoRenderDirectDraw::CreatePrimarySurface()
+{
+ // Create the primary surface
+ DirectDrawSurfaceDesc ddsd;
+ ZeroMemory(&ddsd, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ HRESULT ddrval = DD_OK;
+
+ if (_directDraw == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw object not created");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (_primarySurface)
+ {
+ _primarySurface->Release();
+ _primarySurface = NULL;
+ }
+
+ if (!_fullscreen)
+ {
+ // create a normal window
+ ddrval = _directDraw->SetCooperativeLevel(_hWnd, DDSCL_NORMAL);
+ if (FAILED(ddrval))
+ {
+ //******** Potential workaround for D#4608 *************** Ignore error.
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw failed to set SetCooperativeLevel %x, ddrval");
+ }
+ // we cant size the primary surface based on _hwndRect
+ ddsd.dwFlags = DDSD_CAPS;
+ ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_VIDEOMEMORY;
+
+#ifndef NOGRAPHICSCARD_MEMORY
+ ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _primarySurface using VideoMemory: 0x%x",
+ ddrval);
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
+ _hWnd, _hwndRect.top, _hwndRect.left,
+ _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
+ __LINE__);
+
+#endif
+ //allocate using System memory
+ ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_SYSTEMMEMORY;
+ ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _primarySurface using SystemMemory: 0x%x",
+ ddrval);
+ if (ddrval != 0x887600E1)
+ {
+ _directDraw->Release();
+ _directDraw = 0;
+ }
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ _isPrimaryOrMixingSurfaceOnSystem = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw _primarySurface on SystemMemory");
+
+#ifndef NOGRAPHICSCARD_MEMORY
+ }
+#endif
+
+ // Create a clipper to ensure that our drawing stays inside our window
+ LPDIRECTDRAWCLIPPER directDrawClipper;
+ ddrval = _directDraw->CreateClipper(0, &directDrawClipper, NULL );
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to CreateClipper");
+ _primarySurface->Release();
+ _directDraw->Release();
+ _primarySurface = 0;
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // setting it to our hwnd gives the clipper the coordinates from our window
+ // when using cliplist we run into problem with transparent HWNDs (such as REX)
+ ddrval = directDrawClipper->SetHWnd(0, _hWnd);
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetHWnd");
+ _primarySurface->Release();
+ _directDraw->Release();
+ _primarySurface = 0;
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // attach the clipper to the primary surface
+ ddrval = _primarySurface->SetClipper(directDrawClipper);
+ directDrawClipper->Release(); // no need to keep the clipper around
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetClipper");
+ _primarySurface->Release();
+ _directDraw->Release();
+ _primarySurface = 0;
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ }
+ else
+ {
+ /* The cooperative level determines how much control we have over the
+ * screen. This must at least be either DDSCL_EXCLUSIVE or DDSCL_NORMAL
+ *
+ * DDSCL_EXCLUSIVE allows us to change video modes, and requires
+ * the DDSCL_FULLSCREEN flag, which will cause the window to take over
+ * the fullscreen. This is the preferred DirectDraw mode because it allows
+ * us to have control of the whole screen without regard for GDI.
+ *
+ * DDSCL_NORMAL is used to allow the DirectDraw app to run windowed.
+ */
+
+ // Note: debuging in fullscreen mode does not work, thanks MS...
+ ::GetWindowRect(_hWnd, &_originalHwndRect);
+
+ // DDSCL_NOWINDOWCHANGES prevents DD to change the window but it give us trouble too, not using it
+ ddrval = _directDraw->SetCooperativeLevel(_hWnd, DDSCL_EXCLUSIVE
+ | DDSCL_FULLSCREEN | DDSCL_ALLOWREBOOT);
+
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetCooperativeLevel DDSCL_EXCLUSIVE");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
+ _hWnd, _hwndRect.top, _hwndRect.left,
+ _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
+ __LINE__);
+
+ _directDraw->Release();
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ ddsd.dwFlags = DDSD_CAPS | DDSD_BACKBUFFERCOUNT;
+ ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_FLIP
+ | DDSCAPS_COMPLEX | DDSCAPS_VIDEOMEMORY;
+ ddsd.dwBackBufferCount = 1;
+
+ ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _primarySurface, fullscreen mode: 0x%x",
+ ddrval);
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
+ _hWnd, _hwndRect.top, _hwndRect.left,
+ _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
+ __LINE__);
+
+ _directDraw->Release();
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // Get a pointer to the back buffer
+ DirectDrawCaps ddsCaps;
+ ZeroMemory(&ddsCaps, sizeof(ddsCaps));
+ ddsCaps.dwCaps = DDSCAPS_BACKBUFFER | DDSCAPS_VIDEOMEMORY;
+
+ ddrval = _primarySurface->GetAttachedSurface(&ddsCaps, &_backSurface);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to GetAttachedSurface, fullscreen mode ");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
+ _hWnd, _hwndRect.top, _hwndRect.left,
+ _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
+ __LINE__);
+
+ _primarySurface->Release();
+ _directDraw->Release();
+ _primarySurface = 0;
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // Get the screen size and save it as a rect
+ ZeroMemory(&ddsd, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ }
+
+ ZeroMemory(&ddsd, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+
+ // get our prinmary surface description
+ ddrval = _primarySurface->GetSurfaceDesc(&ddsd);
+ if (!(SUCCEEDED(ddrval) && (ddsd.dwFlags & DDSD_WIDTH) && (ddsd.dwFlags
+ & DDSD_HEIGHT)))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to GetSurfaceDesc _primarySurface");
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
+ _hWnd, _hwndRect.top, _hwndRect.left, _hwndRect.bottom,
+ _hwndRect.right, ddsd.dwFlags, __LINE__);
+
+ _primarySurface->Release();
+ _directDraw->Release();
+ _primarySurface = 0;
+ _directDraw = 0;
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // first we need to figure out the size of the primary surface
+
+ // store screen size
+ ::SetRect(&_screenRect, 0, 0, ddsd.dwWidth, ddsd.dwHeight);
+
+ // store RGB type
+ if (ddsd.ddpfPixelFormat.dwFlags & DDPF_RGB)
+ {
+ // RGB surface
+ switch (ddsd.ddpfPixelFormat.dwRGBBitCount)
+ {
+ case 16:
+ switch (ddsd.ddpfPixelFormat.dwGBitMask)
+ {
+ case 0x00e0:
+ _rgbVideoType = kARGB4444;
+ break;
+ case 0x03e0:
+ _rgbVideoType = kARGB1555;
+ break;
+ case 0x07e0:
+ _rgbVideoType = kRGB565;
+ break;
+ }
+ break;
+ case 24:
+ _rgbVideoType = kRGB24;
+ break;
+ case 32:
+ _rgbVideoType = kARGB;
+ break;
+ }
+ }
+ switch (_blitVideoType)
+ {
+ case kI420:
+ case kIYUV:
+ case kYUY2:
+ case kYV12:
+ case kUYVY:
+ _incomingVideoType = kI420;
+ break;
+ case kUnknown:
+ _blitVideoType = _rgbVideoType;
+ _incomingVideoType = kI420;
+ break;
+ default:
+ _blitVideoType = _rgbVideoType;
+ _incomingVideoType = kI420;
+ break;
+ }
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "DirectDraw created _primarySurface, _blitVideoType %d, _rgbvideoType %d",
+ _blitVideoType, _rgbVideoType);
+ return 0;
+}
+
+int VideoRenderDirectDraw::CreateMixingSurface()
+{
+ if (_directDraw == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw object not created");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ if (_fullscreen)
+ {
+ ::CopyRect(&_hwndRect, &_screenRect);
+ }
+ else
+ {
+ // update our _hWnd size
+ ::GetClientRect(_hWnd, &_hwndRect);
+ }
+
+ if (_mixingSurface)
+ {
+ _mixingSurface->Release();
+ _mixingSurface = NULL;
+ }
+ // create mixing surface
+ DirectDrawSurfaceDesc ddsd;
+ memset(&ddsd, 0, sizeof(ddsd));
+ ddsd.dwSize = sizeof(ddsd);
+ ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
+ ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
+ ddsd.dwHeight = _hwndRect.bottom;
+ ddsd.dwWidth = _hwndRect.right;
+
+ /* char logStr[256];
+ _snprintf(logStr,256, "CreateMixingSurface H:%d W:%d \n",_hwndRect.bottom, _hwndRect.right);
+ OutputDebugString(logStr);
+ */
+
+#ifndef NOGRAPHICSCARD_MEMORY
+ HRESULT ddrval = _directDraw->CreateSurface(&ddsd, &_mixingSurface, NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _mixingSurface using VideoMemory: 0x%x",
+ ddrval);
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d",
+ _hWnd, _hwndRect.top, _hwndRect.left, _hwndRect.bottom,
+ _hwndRect.right, ddsd.dwFlags);
+#endif
+
+ ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
+ HRESULT ddrval = _directDraw->CreateSurface(&ddsd, &_mixingSurface,
+ NULL);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to CreateSurface _mixingSurface on System Memory: 0x%x",
+ ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ _isPrimaryOrMixingSurfaceOnSystem = true;
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw CreateSurface _mixingSurface on SystemMemory");
+
+#ifndef NOGRAPHICSCARD_MEMORY
+ }
+#endif
+
+ _clearMixingSurface = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw _mixingSurface created");
+ return 0;
+}
+
+VideoRenderCallback* VideoRenderDirectDraw::CreateChannel(WebRtc_UWord32 channel,
+ WebRtc_UWord32 zOrder,
+ float startWidth,
+ float startHeight,
+ float stopWidth,
+ float stopHeight)
+{
+ if (!_canStretch)
+ {
+ if (startWidth != 0.0f || startHeight != 0.0f || stopWidth != 1.0f
+ || stopHeight != 1.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to CreateChannel HW don't support stretch");
+ return NULL;
+ }
+ }
+ DirectDrawChannel* ddobj =
+ new DirectDrawChannel(_directDraw, _blitVideoType,
+ _incomingVideoType, _rgbVideoType, this);
+ ddobj->SetStreamSettings(this, 0, startWidth, startHeight, stopWidth,
+ stopHeight);
+
+ // store channel
+ _directDrawChannels[channel & 0x0000ffff] = ddobj;
+
+ // store Z order
+ // default streamID is 0
+ _directDrawZorder.insert(ZorderPair(zOrder, channel & 0x0000ffff));
+ return ddobj;
+}
+
+int VideoRenderDirectDraw::AddDirectDrawChannel(int channel,
+ unsigned char streamID,
+ int zOrder,
+ DirectDrawChannel* ddObj)
+{
+ // Only allow one stream per channel, demuxing is done outside of DirectDraw...
+ streamID = 0;
+ unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
+
+ // store channel
+ _directDrawChannels[channel & 0x0000ffff] = ddObj;
+
+ _demuxing = true; // with this function it's always demux
+
+ // store Z order
+ _directDrawZorder.insert(ZorderPair(zOrder, streamChannel));
+ return 0;
+}
+
+DirectDrawChannel* VideoRenderDirectDraw::ShareDirectDrawChannel(
+ int channel)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ DirectDrawChannel* obj = NULL;
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ if (ddIt != _directDrawChannels.end())
+ {
+ obj = ddIt->second;
+ obj->AddRef();
+ }
+ return obj;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::DeleteChannel(const WebRtc_UWord32 channel)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ // Remove the old z order
+
+ //unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
+ std::multimap<int, unsigned int>::iterator it;
+ it = _directDrawZorder.begin();
+ while (it != _directDrawZorder.end())
+ {
+ //if(streamChannel == it->second )
+ if ((channel & 0x0000ffff) == (it->second & 0x0000ffff))
+ {
+ it = _directDrawZorder.erase(it);
+ break;
+ }
+ it++;
+ }
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ if (ddIt != _directDrawChannels.end())
+ {
+ ddIt->second->Release();
+ _directDrawChannels.erase(ddIt);
+ _clearMixingSurface = true;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::GetStreamSettings(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& startWidth,
+ float& startHeight,
+ float& stopWidth,
+ float& stopHeight)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ if (ddIt == _directDrawChannels.end())
+ {
+ // This channel doesn't exist.
+ return -1;
+ }
+
+ DirectDrawChannel* ptrChannel = ddIt->second;
+ // Only support one stream per channel, is demuxing done outside if DD.
+ //if (ptrChannel->GetStreamSettings(this, streamId, startWidth, startHeight, stopWidth, stopHeight) == -1)
+ if (ptrChannel->GetStreamSettings(this, 0, startWidth, startHeight,
+ stopWidth, stopHeight) == -1)
+ {
+ // Error for this stream
+ return -1;
+ }
+
+ // Get the zOrder
+ std::multimap<int, unsigned int>::iterator it;
+ it = _directDrawZorder.begin();
+ while (it != _directDrawZorder.end())
+ {
+ if ((channel & 0x0000ffff) == (it->second & 0x0000ffff))
+ {
+ // We found our channel zOrder
+ zOrder = (unsigned int) (it->first);
+ break;
+ }
+ it++;
+ }
+
+ return 0;
+}
+
+int VideoRenderDirectDraw::GetChannels(std::list<int>& channelList)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.begin();
+
+ while (ddIt != _directDrawChannels.end())
+ {
+ int channel = ddIt->first;
+ if (channel == 0x0000ffff)
+ {
+ channel = -1;
+ }
+ channelList.push_back(channel);
+ ddIt++;
+ }
+ return 0;
+}
+
+bool VideoRenderDirectDraw::HasChannel(int channel)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ if (ddIt != _directDrawChannels.end())
+ {
+ return true;
+ }
+ return false;
+}
+
+bool VideoRenderDirectDraw::HasChannels()
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ if (_directDrawChannels.begin() != _directDrawChannels.end())
+ {
+ return true;
+ }
+ return false;
+}
+
+bool VideoRenderDirectDraw::IsFullScreen()
+{
+ return _fullscreen;
+}
+
+VideoType VideoRenderDirectDraw::GetPerferedVideoFormat()
+{
+ return _incomingVideoType;
+}
+
+// this can be called rutime from another thread
+DirectDrawChannel* VideoRenderDirectDraw::ConfigureDirectDrawChannel(int channel,
+ unsigned char streamID,
+ int zOrder,
+ float left,
+ float top,
+ float right,
+ float bottom)
+{
+ // Only support one stream per channel, is demuxing done outside if DD.
+ streamID = 0;
+
+ CriticalSectionScoped cs(*_confCritSect);
+
+ if (!_canStretch)
+ {
+ if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to ConfigureDirectDrawChannel HW don't support stretch");
+ return NULL;
+ }
+ }
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ DirectDrawChannel* ddobj = NULL;
+ if (ddIt != _directDrawChannels.end())
+ {
+ ddobj = ddIt->second;
+ }
+ if (ddobj == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "DirectDraw failed to find channel");
+ return NULL;
+ }
+ unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
+ // remove the old z order
+ std::multimap<int, unsigned int>::iterator it;
+ it = _directDrawZorder.begin();
+ while (it != _directDrawZorder.end())
+ {
+ if (streamChannel == it->second)
+ {
+ it = _directDrawZorder.erase(it);
+ break;
+ }
+ it++;
+ }
+ // if this channel already are in the zOrder map it's demux
+ it = _directDrawZorder.begin();
+ while (it != _directDrawZorder.end())
+ {
+ if (channel == (it->second & 0x0000ffff))
+ {
+ _demuxing = true;
+ break;
+ }
+ it++;
+ }
+ if (it == _directDrawZorder.end())
+ {
+ _demuxing = false;
+ }
+
+ _clearMixingSurface = true;
+
+ if (left == 0.0f && top == 0.0f && right == 0.0f && bottom == 0.0f)
+ {
+ // remove
+ _directDrawChannels.erase(ddIt);
+ ddobj->Release();
+ return NULL;
+ }
+ ddobj->SetStreamSettings(this, streamID, left, top, right, bottom);
+
+ _directDrawZorder.insert(ZorderPair(zOrder, streamChannel));
+ return ddobj;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::SetCropping(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamID,
+ float left, float top,
+ float right, float bottom)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+ if (!_canStretch)
+ {
+ if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
+ "DirectDraw failed to SetCropping HW don't support stretch");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ }
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel & 0x0000ffff);
+ if (ddIt != _directDrawChannels.end())
+ {
+ DirectDrawChannel* ddobj = ddIt->second;
+ if (ddobj)
+ {
+ // Only support one stream per channel, is demuxing done outside if DD.
+ ddobj->SetStreamCropSettings(this, 0, left, top, right, bottom);
+ //ddobj->SetStreamCropSettings(this, streamID, left, top, right, bottom);
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::ConfigureRenderer(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ if (ConfigureDirectDrawChannel(channel, (unsigned char) streamId, zOrder,
+ left, top, right, bottom) == NULL)
+ {
+ if (left == 0.0f && top == 0.0f && right == 0.0f && bottom == 0.0f)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
+ "ConfigureRender, removed channel:%d streamId:%d",
+ channel, streamId);
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideoRenderer,
+ -1,
+ "DirectDraw failed to ConfigureRenderer for channel: %d",
+ channel);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+// this can be called runtime from another thread
+WebRtc_Word32 VideoRenderDirectDraw::SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 colorText,
+ const WebRtc_UWord32 colorBg,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ DirectDrawTextSettings* textSetting = NULL;
+
+ CriticalSectionScoped cs(*_confCritSect);
+
+ _frameChanged = true;
+
+ std::map<unsigned char, DirectDrawTextSettings*>::iterator it;
+ it = _textSettings.find(textId);
+ if (it != _textSettings.end())
+ {
+ if (it->second)
+ {
+ textSetting = it->second;
+ }
+ }
+ _clearMixingSurface = true;
+
+ if (text == NULL || textLength == 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw remove text textId:%d", textId);
+ if (textSetting)
+ {
+ delete textSetting;
+ _textSettings.erase(it);
+ }
+ return 0;
+ }
+
+ // sanity
+ if (left > 1.0f || left < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetText invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (top > 1.0f || top < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetText invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (right > 1.0f || right < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetText invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (bottom > 1.0f || bottom < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetText invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (textSetting == NULL)
+ {
+ textSetting = new DirectDrawTextSettings();
+ }
+ int retVal = textSetting->SetText((const char*) text, textLength,
+ (COLORREF) colorText, (COLORREF) colorBg,
+ left, top, right, bottom);
+ if (retVal != 0)
+ {
+ delete textSetting;
+ textSetting = NULL;
+ _textSettings.erase(textId);
+ return retVal;
+ }
+ if (textSetting)
+ {
+ _textSettings[textId] = textSetting;
+ }
+ return retVal;
+}
+
+// this can be called runtime from another thread
+WebRtc_Word32 VideoRenderDirectDraw::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ DirectDrawBitmapSettings* bitmapSetting = NULL;
+
+ CriticalSectionScoped cs(*_confCritSect);
+
+ _frameChanged = true;
+ std::map<unsigned char, DirectDrawBitmapSettings*>::iterator it;
+ it = _bitmapSettings.find(pictureId);
+ if (it != _bitmapSettings.end())
+ {
+ if (it->second)
+ {
+ bitmapSetting = it->second;
+ }
+ }
+ _clearMixingSurface = true;
+
+ if (bitMap == NULL)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw remove bitmap pictureId:%d", pictureId);
+ if (bitmapSetting)
+ {
+ delete bitmapSetting;
+ _bitmapSettings.erase(it);
+ }
+ return 0;
+ }
+
+ // sanity
+ if (left > 1.0f || left < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetBitmap invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (top > 1.0f || top < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetBitmap invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (right > 1.0f || right < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetBitmap invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (bottom > 1.0f || bottom < 0.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw SetBitmap invalid parameter");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (!_canStretch)
+ {
+ if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetBitmap HW don't support stretch");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ }
+ if (bitmapSetting == NULL)
+ {
+ bitmapSetting = new DirectDrawBitmapSettings();
+ }
+
+ bitmapSetting->_transparentBitMap = (HBITMAP) bitMap;
+ bitmapSetting->_transparentBitmapLeft = left;
+ bitmapSetting->_transparentBitmapRight = right;
+ bitmapSetting->_transparentBitmapTop = top;
+ bitmapSetting->_transparentBitmapBottom = bottom;
+
+ // colorKey == NULL equals no transparency
+ if (colorKey)
+ {
+ // first remove constness
+ DDCOLORKEY* ddColorKey =
+ static_cast<DDCOLORKEY*> (const_cast<void*> (colorKey));
+ if (!_supportTransparency)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to SetBitmap HW don't support transparency");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+ }
+ if (bitmapSetting->_transparentBitmapColorKey == NULL)
+ {
+ bitmapSetting->_transparentBitmapColorKey = new DDCOLORKEY();
+ }
+
+ if (ddColorKey)
+ {
+ bitmapSetting->_transparentBitmapColorKey->dwColorSpaceLowValue
+ = ddColorKey->dwColorSpaceLowValue;
+ bitmapSetting->_transparentBitmapColorKey->dwColorSpaceHighValue
+ = ddColorKey->dwColorSpaceHighValue;
+ }
+ }
+ int retval = bitmapSetting->SetBitmap(_trace, _directDraw);
+ if (retval != 0)
+ {
+ delete bitmapSetting;
+ bitmapSetting = NULL;
+ _bitmapSettings.erase(pictureId);
+ return retval;
+ }
+ if (bitmapSetting)
+ {
+ _bitmapSettings[pictureId] = bitmapSetting;
+ }
+ return retval;
+}
+
+// this can be called rutime from another thread
+WebRtc_Word32 VideoRenderDirectDraw::SetTransparentBackground(
+ const bool enable)
+{
+ CriticalSectionScoped cs(*_confCritSect);
+
+ if (_supportTransparency)
+ {
+ _transparentBackground = enable;
+ if (enable)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw enabled TransparentBackground");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw disabled TransparentBackground");
+ }
+ return 0;
+ }
+ WEBRTC_TRACE(
+ kTraceError,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to EnableTransparentBackground HW don't support transparency");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_INVALID_ARG;
+}
+
+int VideoRenderDirectDraw::FillSurface(DirectDrawSurface *pDDSurface,
+ RECT* rect)
+{
+ // sanity checks
+ if (NULL == pDDSurface)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (NULL == rect)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+
+ // Repaint the whole specified surface
+ HRESULT ddrval;
+ DDBLTFX ddFX;
+
+ ZeroMemory(&ddFX, sizeof(ddFX));
+ ddFX.dwSize = sizeof(ddFX);
+ ddFX.dwFillColor = RGB(0, 0, 0);
+
+ // Draw color key on the video area of given surface
+ ddrval = pDDSurface->Blt(rect, NULL, NULL, DDBLT_COLORFILL | DDBLT_WAIT,
+ &ddFX);
+ if (FAILED(ddrval))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw failed to fill surface");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ return 0;
+}
+
+// the real rendering thread
+bool VideoRenderDirectDraw::RemoteRenderingThreadProc(void *obj)
+{
+ return static_cast<VideoRenderDirectDraw*> (obj)->RemoteRenderingProcess();
+}
+
+bool VideoRenderDirectDraw::RemoteRenderingProcess()
+{
+ bool hwndChanged = false;
+ int waitTime = 0;
+
+ _screenEvent->Wait(100);
+
+ _confCritSect->Enter();
+
+ if (_blit == false)
+ {
+ _confCritSect->Leave();
+ return true;
+ }
+
+ if (!::GetForegroundWindow())
+ {
+ //no window, i.e the user have clicked CTRL+ALT+DEL, return true and wait
+ _confCritSect->Leave();
+ return true;
+ }
+
+ // Skip to blit if last render to primare surface took too long time.
+ _processCount++;
+ if (_deliverInQuarterFrameRate)
+ {
+ if (_processCount % 4 != 0)
+ {
+ _confCritSect->Leave();
+ return true;
+ }
+ }
+ else if (_deliverInHalfFrameRate)
+ {
+ if (_processCount % 2 != 0)
+ {
+ _confCritSect->Leave();
+ return true;
+ }
+ }
+
+ // Calculate th erender process time
+ unsigned int startProcessTime = timeGetTime();
+
+ hwndChanged = HasHWNDChanged();
+ if (hwndChanged)
+ {
+ _clearMixingSurface = true;
+ }
+
+ std::map<int, DirectDrawChannel*>::iterator it;
+ it = _directDrawChannels.begin();
+ while (it != _directDrawChannels.end() && !_frameChanged)
+ {
+ if (it->second)
+ {
+ int channel = it->first;
+ _frameChanged = it->second->IsOffScreenSurfaceUpdated(this);
+ }
+ it++;
+ }
+ if (_backSurface)
+ {
+ if (hwndChanged || _frameChanged)
+ {
+ BlitFromOffscreenBuffersToMixingBuffer();
+ BlitFromBitmapBuffersToMixingBuffer();
+ BlitFromTextToMixingBuffer();
+ }
+ BlitFromMixingBufferToBackBuffer();
+ WaitAndFlip(waitTime);
+ }
+ else
+ {
+ if (hwndChanged || _frameChanged)
+ {
+ BlitFromOffscreenBuffersToMixingBuffer();
+ BlitFromBitmapBuffersToMixingBuffer();
+ BlitFromTextToMixingBuffer();
+ }
+ BlitFromMixingBufferToFrontBuffer(hwndChanged, waitTime);
+
+ }
+ // Check the total time it took processing all rendering. Don't consider waitTime.
+ //const int totalRenderTime=GET_TIME_IN_MS()- startProcessTime-waitTime;
+ const int totalRenderTime = ::timeGetTime() - startProcessTime - waitTime;
+ DecideBestRenderingMode(hwndChanged, totalRenderTime);
+ _frameChanged = false;
+ _confCritSect->Leave();
+
+ return true;
+}
+void VideoRenderDirectDraw::DecideBestRenderingMode(bool hwndChanged,
+ int totalRenderTime)
+{
+ /* Apply variuos fixes for bad graphic drivers.
+ 1. If cpu to high- test wait fix
+ 2. If cpu still too high render in 1/2 display update period.
+ 3. If RemoteRenderingProcess take to long time reduce the blit period to 1/2 display update period.
+ 4. If RemoteRenderingProcess still take to long time try color conversion fix. It do color conversion in VieoRenderDirectDrawChannel::DeliverFrame
+ 5. If RemoteRenderingProcess still take to long time reduce the blit period to 1/4 display update period and disable color conversion fix.
+ 6 if RemoteRenderingProcess still take to long time reduce the blit period to 1/4 display update period and enable color conversion fix again.
+ */
+
+ const int timesSinceLastCPUCheck = timeGetTime()
+ - _screenRenderCpuUsage.LastGetCpuTime();
+ int cpu = 0;
+
+ if (hwndChanged) // Render window changed.
+ {
+ cpu = _screenRenderCpuUsage.GetCpuUsage(); // Get CPU usage for this thread. (Called if hwndCanged just to reset the GET CPU Usage function)
+ _nrOfTooLongRenderTimes = 0; // Reset count of too long render times.
+ return; // Return - nothing more to do since the window has changed.
+ }
+ // Check total rendering times
+ if (_maxAllowedRenderTime > 0 && totalRenderTime > _maxAllowedRenderTime)
+ {
+ if (!_deliverInHalfFrameRate || totalRenderTime > 2
+ * _maxAllowedRenderTime)
+ {
+ _nrOfTooLongRenderTimes += totalRenderTime / _maxAllowedRenderTime; //Weighted with the number of to long render times
+ }
+ }
+
+ // If we are not using back surface (ie full screen rendering) we might try to switch BlitFromMixingBufferToFrontBuffer mode.
+ if (timesSinceLastCPUCheck > WindowsThreadCpuUsage::CPU_CHECK_INTERVAL)
+ {
+ cpu = _screenRenderCpuUsage.GetCpuUsage(); // Get CPU usage for this thread. (Called if hwndCanged just to reset the GET CPU Usage function)
+ WEBRTC_TRACE(
+ kTraceStream,
+ kTraceVideo,
+ -1,
+ "Screen render thread cpu usage. (Tid %d), cpu usage %d processTime %d, no of too long render times %d",
+ GetCurrentThreadId(), cpu, totalRenderTime,
+ _nrOfTooLongRenderTimes);
+
+ // If this screen render thread uses more than 5% of the total CPU time and the
+ // 1. try waitFix
+ if (cpu >= 5 && _renderModeWaitForCorrectScanLine == false
+ && !_backSurface)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ -1,
+ "HIGH screen render thread cpu usage. (Tid %d), cpu usage %d, applying wait for scan line",
+ GetCurrentThreadId(), cpu);
+ _renderModeWaitForCorrectScanLine = true;
+ _fullScreenWaitEvent->StartTimer(true, 1);
+ }
+ else if (cpu >= 10 && _deliverInHalfFrameRate == false)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ -1,
+ "HIGH screen render thread cpu usage. (Tid %d), cpu usage %d, Render half rate",
+ GetCurrentThreadId(), cpu);
+ _deliverInHalfFrameRate = true;
+ }
+ else
+ {
+ // Check if rendering takes too long time
+ if (_nrOfTooLongRenderTimes > 15 || totalRenderTime
+ >= WindowsThreadCpuUsage::CPU_CHECK_INTERVAL)
+ {
+
+ // The rendering is taking too long time
+ if (_deliverInHalfFrameRate == false)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Render half rate, tid: %d",
+ GetCurrentThreadId());
+ _deliverInHalfFrameRate = true;
+ }
+ else if (_deliverInScreenType == false
+ && !_deliverInQuarterFrameRate)
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "Applying deliver in screen type format, tid: %d",
+ GetCurrentThreadId());
+ // 2. try RGB fix
+ std::map<int, DirectDrawChannel*>::iterator it;
+ it = _directDrawChannels.begin();
+ while (it != _directDrawChannels.end())
+ {
+ it->second->ChangeDeliverColorFormat(true);
+ it++;
+ }
+ _deliverInScreenType = true;
+ }
+ else if (_deliverInQuarterFrameRate == false)
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "Render quarter rate and disable deliver in screen type format, tid: %d",
+ GetCurrentThreadId());
+ _deliverInQuarterFrameRate = true;
+ if (_deliverInScreenType)
+ {
+ //Disable RGB fix
+ std::map<int, DirectDrawChannel*>::iterator it;
+ it = _directDrawChannels.begin();
+ while (it != _directDrawChannels.end())
+ {
+ it->second->ChangeDeliverColorFormat(false);
+ it++;
+ }
+ _deliverInScreenType = false;
+ }
+ }
+ else if (_deliverInQuarterFrameRate == true
+ && !_deliverInScreenType)
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "Render quarter rate and enable RGB fix, tid: %d",
+ GetCurrentThreadId());
+ _deliverInQuarterFrameRate = true;
+
+ //Enabe RGB fix
+ std::map<int, DirectDrawChannel*>::iterator it;
+ it = _directDrawChannels.begin();
+ while (it != _directDrawChannels.end())
+ {
+ it->second->ChangeDeliverColorFormat(true);
+ it++;
+ }
+ _deliverInScreenType = true;
+ }
+ }
+ }
+ _nrOfTooLongRenderTimes = 0; // Reset count of too long render times.
+ }
+}
+
+/*
+ * Internal help functions for blitting
+ */
+
+bool VideoRenderDirectDraw::HasHWNDChanged()
+{
+ // we check if the HWND has changed
+ if (!_fullscreen)
+ {
+ RECT currentRect;
+ ::GetClientRect(_hWnd, ¤tRect);
+ if (!EqualRect(¤tRect, &_hwndRect))
+ {
+ int retVal = CreateMixingSurface(); // this will delete the old mixing surface
+ if (retVal != 0)
+ {
+ return false;
+ }
+ return true;
+ }
+ }
+ return false;
+}
+
+int VideoRenderDirectDraw::BlitFromOffscreenBuffersToMixingBuffer()
+{
+ bool updateAll = false; // used to minimize the number of blt
+
+ DDBLTFX ddbltfx;
+ ZeroMemory(&ddbltfx, sizeof(ddbltfx));
+ ddbltfx.dwSize = sizeof(ddbltfx);
+ ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
+
+ if (_mixingSurface == NULL)
+ {
+ int retVal = CreateMixingSurface();
+ if (retVal != 0)
+ {
+ // trace done
+ return retVal;
+ }
+ }
+ RECT mixingRect;
+ ::SetRectEmpty(&mixingRect);
+
+ if (_fullscreen)
+ {
+ ::CopyRect(&mixingRect, &_screenRect);
+ }
+ else
+ {
+ ::CopyRect(&mixingRect, &_hwndRect);
+ // what if largest size is larger than screen
+ if (mixingRect.right > _screenRect.right)
+ {
+ mixingRect.right = _screenRect.right;
+ }
+ if (mixingRect.bottom > _screenRect.bottom)
+ {
+ mixingRect.bottom = _screenRect.bottom;
+ }
+ }
+ if (!EqualRect(&_mixingRect, &mixingRect))
+ {
+ // size changed
+ CopyRect(&_mixingRect, &mixingRect);
+ FillSurface(_mixingSurface, &mixingRect);
+ updateAll = true;
+ }
+
+ if (_clearMixingSurface)
+ {
+ FillSurface(_mixingSurface, &_mixingRect);
+ _clearMixingSurface = false;
+ updateAll = true;
+ }
+
+ std::multimap<int, unsigned int>::reverse_iterator it;
+ it = _directDrawZorder.rbegin();
+ while (it != _directDrawZorder.rend())
+ {
+ // loop through all channels and streams in Z order
+ short streamID = (it->second >> 16);
+ int channel = it->second & 0x0000ffff;
+
+ std::map<int, DirectDrawChannel*>::iterator ddIt;
+ ddIt = _directDrawChannels.find(channel);
+ if (ddIt != _directDrawChannels.end())
+ {
+ // found the channel
+ DirectDrawChannel* channelObj = ddIt->second;
+ if (channelObj && _mixingSurface)
+ {
+ if (updateAll || channelObj->IsOffScreenSurfaceUpdated(this))
+ {
+ updateAll = true;
+ if (channelObj->BlitFromOffscreenBufferToMixingBuffer(
+ this,
+ streamID,
+ _mixingSurface,
+ _mixingRect,
+ _demuxing)
+ != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ -1,
+ "DirectDraw error BlitFromOffscreenBufferToMixingBuffer ");
+ _mixingSurface->Release();
+ _mixingSurface = NULL;
+ }
+ }
+ }
+ }
+ it++;
+ }
+ return 0;
+}
+
+int VideoRenderDirectDraw::BlitFromTextToMixingBuffer()
+{
+ if (_directDraw == NULL)
+ {
+ return -1;
+ }
+ if (!_mixingSurface)
+ {
+ return -1;
+ }
+ if (_textSettings.empty())
+ {
+ return 0;
+ }
+
+ HDC hdcDDSurface;
+ HRESULT res = _mixingSurface->GetDC(&hdcDDSurface);
+ if (res != S_OK)
+ {
+ return -1;
+ }
+ //
+ std::map<unsigned char, DirectDrawTextSettings*>::reverse_iterator it;
+ it = _textSettings.rbegin();
+
+ while (it != _textSettings.rend())
+ {
+ DirectDrawTextSettings* settings = it->second;
+ it++;
+ if (settings == NULL)
+ {
+ continue;
+ }
+ SetTextColor(hdcDDSurface, settings->_colorRefText);
+ SetBkColor(hdcDDSurface, settings->_colorRefBackground);
+
+ if (settings->_transparent)
+ {
+ SetBkMode(hdcDDSurface, TRANSPARENT); // do we need to call this all the time?
+ }
+ else
+ {
+ SetBkMode(hdcDDSurface, OPAQUE); // do we need to call this all the time?
+ }
+ RECT textRect;
+ textRect.left = int(_mixingRect.right * settings->_textLeft);
+ textRect.right = int(_mixingRect.right * settings->_textRight);
+ textRect.top = int(_mixingRect.bottom * settings->_textTop);
+ textRect.bottom = int(_mixingRect.bottom * settings->_textBottom);
+
+ DrawTextA(hdcDDSurface, settings->_ptrText, settings->_textLength,
+ &textRect, DT_LEFT);
+ }
+ _mixingSurface->ReleaseDC(hdcDDSurface);
+ return 0;
+}
+
+int VideoRenderDirectDraw::BlitFromBitmapBuffersToMixingBuffer()
+{
+ HRESULT ddrval;
+ DDBLTFX ddbltfx;
+ ZeroMemory(&ddbltfx, sizeof(ddbltfx));
+ ddbltfx.dwSize = sizeof(ddbltfx);
+ ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
+
+ if (_directDraw == NULL)
+ {
+ return -1; // signal that we are not ready for the change
+ }
+
+ std::map<unsigned char, DirectDrawBitmapSettings*>::reverse_iterator it;
+ it = _bitmapSettings.rbegin();
+
+ while (it != _bitmapSettings.rend())
+ {
+ DirectDrawBitmapSettings* settings = it->second;
+ it++;
+ if (settings == NULL)
+ {
+ continue;
+ }
+
+ // Color keying lets you set colors on a surface to be completely transparent.
+ // always blit _transparentBitmapSurface last
+ if (_mixingSurface && settings->_transparentBitmapSurface
+ && settings->_transparentBitmapWidth
+ && settings->_transparentBitmapHeight)
+ {
+ DWORD signal = DDBLT_WAIT | DDBLT_DDFX;
+ // Set transparent color
+ if (settings->_transparentBitmapColorKey)
+ {
+ signal |= DDBLT_KEYSRC;
+ settings->_transparentBitmapSurface->SetColorKey(
+ DDCKEY_SRCBLT,
+ settings->_transparentBitmapColorKey);
+ }
+
+ // Now we can blt the transparent surface to another surface
+ RECT srcRect;
+ SetRect(&srcRect, 0, 0, settings->_transparentBitmapWidth,
+ settings->_transparentBitmapHeight);
+
+ RECT dstRect;
+ if (settings->_transparentBitmapLeft
+ != settings->_transparentBitmapRight
+ && settings->_transparentBitmapTop
+ != settings->_transparentBitmapBottom)
+ {
+ CopyRect(&dstRect, &_mixingRect);
+ dstRect.left = (int) (dstRect.right
+ * settings->_transparentBitmapLeft);
+ dstRect.right = (int) (dstRect.right
+ * settings->_transparentBitmapRight);
+ dstRect.top = (int) (dstRect.bottom
+ * settings->_transparentBitmapTop);
+ dstRect.bottom = (int) (dstRect.bottom
+ * settings->_transparentBitmapBottom);
+ }
+ else
+ {
+
+ // if left, right, top and bottom are describing one point use the original size
+ CopyRect(&dstRect, &srcRect);
+ POINT startp;
+ startp.x = (int) (_mixingRect.right
+ * settings->_transparentBitmapLeft);
+ startp.y = (int) (_mixingRect.bottom
+ * settings->_transparentBitmapTop);
+ OffsetRect(&dstRect, startp.x, startp.y);
+
+ // make sure that we blit inside our surface
+ if (dstRect.bottom > _mixingRect.bottom)
+ {
+ srcRect.bottom -= dstRect.bottom - _mixingRect.bottom;
+ // sanity
+ if (srcRect.bottom < 0)
+ {
+ srcRect.bottom = 0;
+ }
+ dstRect.bottom = _mixingRect.bottom;
+ }
+ if (dstRect.right > _mixingRect.right)
+ {
+ srcRect.right -= dstRect.right - _mixingRect.right;
+ // sanity
+ if (srcRect.right < 0)
+ {
+ srcRect.right = 0;
+ }
+ dstRect.right = _mixingRect.right;
+ }
+ }
+ // ddbltfx.dwDDFX |= DDBLTFX_MIRRORUPDOWN; //only for test requires hw support
+
+ // wait for the _mixingSurface to be available
+ ddrval = _mixingSurface->Blt(&dstRect,
+ settings->_transparentBitmapSurface,
+ &srcRect, signal, &ddbltfx);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ if (!::GetForegroundWindow())
+ {
+ // no window, i.e the user have clicked CTRL+ALT+DEL
+ return 0;
+ }
+ // always re-creted via the SetBitmap call
+ settings->_transparentBitmapSurface->Release();
+ settings->_transparentBitmapSurface = NULL;
+
+ _clearMixingSurface = true;
+
+ if (settings->_transparentBitMap)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw re-set transparent bitmap");
+ settings->SetBitmap(_trace, _directDraw);
+ }
+ }
+ else if (ddrval != DD_OK)
+ {
+ settings->_transparentBitmapSurface->Release();
+ settings->_transparentBitmapSurface = NULL;
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVideo,
+ -1,
+ "DirectDraw blt error 0x%x _transparentBitmapSurface",
+ ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * normal blitting
+ */
+int VideoRenderDirectDraw::BlitFromMixingBufferToFrontBuffer(
+ bool hwndChanged,
+ int& waitTime)
+{
+ DDBLTFX ddbltfx;
+ ZeroMemory(&ddbltfx, sizeof(ddbltfx));
+ ddbltfx.dwSize = sizeof(ddbltfx);
+ ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
+ RECT rcRectDest;
+
+ // test for changing mode
+ /* for(int i= 0; i< 6000000; i ++)
+ {
+ rcRectDest.left = i;
+ }
+ */
+
+ if (IsRectEmpty(&_mixingRect))
+ {
+ // no error just nothing to blit
+ return 0;
+ }
+ if (_mixingSurface == NULL)
+ {
+ // The mixing surface has probably been deleted
+ // and we haven't had time to restore it yet. Wait...
+ return 0;
+ }
+ if (_primarySurface == NULL)
+ {
+ int retVal = CreatePrimarySurface();
+ if (retVal != 0)
+ {
+ // tracing done
+ return retVal;
+ }
+ }
+
+ // first we need to figure out where on the primary surface our window lives
+ ::GetWindowRect(_hWnd, &rcRectDest);
+
+ DWORD signal = DDBLT_WAIT | DDBLT_DDFX;
+
+ // Set transparent color
+ if (_transparentBackground)
+ {
+ signal |= DDBLT_KEYSRC;
+ DDCOLORKEY ColorKey;
+ ColorKey.dwColorSpaceLowValue = RGB(0, 0, 0);
+ ColorKey.dwColorSpaceHighValue = RGB(0, 0, 0);
+ _mixingSurface->SetColorKey(DDCKEY_SRCBLT, &ColorKey);
+ }
+
+ if (_renderModeWaitForCorrectScanLine)
+ {
+ // wait for previus draw to complete
+ DWORD scanLines = 0;
+ DWORD screenLines = _screenRect.bottom - 1; // scanlines start on 0
+ DWORD screenLines90 = (screenLines * 9) / 10; // % of the screen is rendered
+ //waitTime=GET_TIME_IN_MS();
+ waitTime = ::timeGetTime();
+ HRESULT hr = _directDraw->GetScanLine(&scanLines);
+ while (screenLines90 > scanLines && hr == DD_OK)
+ {
+ _confCritSect->Leave();
+ _fullScreenWaitEvent->Wait(3);
+ _confCritSect->Enter();
+ if (_directDraw == NULL)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ hr = _directDraw->GetScanLine(&scanLines);
+ }
+ //waitTime=GET_TIME_IN_MS()-waitTime;
+ waitTime = ::timeGetTime() - waitTime;
+ }
+
+ HRESULT ddrval = _primarySurface->Blt(&rcRectDest, _mixingSurface,
+ &_mixingRect, signal, &ddbltfx);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ if (!::GetForegroundWindow())
+ {
+ // no window, i.e the user have clicked CTRL+ALT+DEL
+ return 0;
+ }
+ ddrval = _primarySurface->Restore();
+ if (ddrval == DD_OK) // Try again
+ {
+ ddrval = _primarySurface->Blt(&rcRectDest, _mixingSurface,
+ &_mixingRect, signal, &ddbltfx);
+ }
+ if (ddrval != DD_OK) // If restore failed or second time blt failed. Delete the surface. It will be recreated next time.
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to restore lost _primarySurface 0x%x",
+ ddrval);
+ _primarySurface->Release();
+ _primarySurface = NULL;
+ if (_mixingSurface)
+ {
+ _mixingSurface->Release();
+ _mixingSurface = NULL;
+ }
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw restored lost _primarySurface");
+ }
+ else if (ddrval == DDERR_EXCEPTION)
+ {
+ _primarySurface->Release();
+ _primarySurface = NULL;
+ if (_mixingSurface)
+ {
+ _mixingSurface->Release();
+ _mixingSurface = NULL;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw exception in _primarySurface");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (ddrval != DD_OK)
+ {
+ if (ddrval != 0x80004005) // Undefined error. Ignore
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw blt error 0x%x _primarySurface", ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ }
+ return 0;
+}
+
+/**
+ * fullscreen mode blitting
+ */
+
+int VideoRenderDirectDraw::WaitAndFlip(int& waitTime)
+{
+ if (_primarySurface == NULL)
+ {
+ // no trace, too much in the file
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (_directDraw == NULL)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ // wait for previus draw to complete
+ DWORD scanLines = 0;
+ DWORD screenLines = _screenRect.bottom - 1; // scanlines start on 0
+ DWORD screenLines90 = (screenLines * 9) / 10; // % of the screen is rendered
+
+ //waitTime=GET_TIME_IN_MS();
+ waitTime = ::timeGetTime();
+ HRESULT hr = _directDraw->GetScanLine(&scanLines);
+ while (screenLines90 > scanLines && hr == DD_OK)
+ {
+ _confCritSect->Leave();
+ _fullScreenWaitEvent->Wait(3);
+ _confCritSect->Enter();
+ if (_directDraw == NULL)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ hr = _directDraw->GetScanLine(&scanLines);
+ }
+ //waitTime=GET_TIME_IN_MS()-waitTime;
+ waitTime = ::timeGetTime() - waitTime;
+ if (screenLines > scanLines)
+ {
+ // this function sucks a lot of the CPU... but it's worth it
+ _directDraw->WaitForVerticalBlank(DDWAITVB_BLOCKBEGIN, NULL);
+ }
+
+ // schedule a flip
+ HRESULT ddrval = _primarySurface->Flip(NULL, DDFLIP_WAIT); // schedule flip DDFLIP_WAIT
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ if (!::GetForegroundWindow())
+ {
+ // no window, i.e the user have clicked CTRL+ALT+DEL
+ return 0;
+ }
+ //if(::IsIconic(_hWnd))
+ //{
+ // need to do this before Restore
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw our window is an icon maximize it ");
+ // When the full screen window is switched out by ALT-TAB or ALT-CTRL-DEL-TASKMANAGER,
+ // this call will hang the app. Remove it to fix the problem.
+ // FIXME:
+ // 1) Why we want to active and max the window when it was minimized?
+ // 2) Why this is needed before restore? We didn't do that in non full screen mode.
+ //::ShowWindow(_hWnd, SW_SHOWMAXIMIZED);
+ //}
+ ddrval = _primarySurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ -1,
+ "DirectDraw failed to restore _primarySurface, in flip, 0x%x",
+ ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw restore _primarySurface in flip");
+
+ }
+ else if (ddrval != DD_OK)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ return 0;
+}
+
+int VideoRenderDirectDraw::BlitFromMixingBufferToBackBuffer()
+{
+ if (_backSurface == NULL)
+ {
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ if (IsRectEmpty(&_mixingRect))
+ {
+ // nothing to blit
+ return 0;
+ }
+ DDBLTFX ddbltfx;
+ ZeroMemory(&ddbltfx, sizeof(ddbltfx));
+ ddbltfx.dwSize = sizeof(ddbltfx);
+ ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
+
+ // wait for the _backSurface to be available
+ HRESULT ddrval = _backSurface->Blt(&_screenRect, _mixingSurface,
+ &_mixingRect, DDBLT_WAIT | DDBLT_DDFX,
+ &ddbltfx);
+ if (ddrval == DDERR_SURFACELOST)
+ {
+ if (!::GetForegroundWindow())
+ {
+ // no window, i.e the user have clicked CTRL+ALT+DEL
+ return 0;
+ }
+ //if(::IsIconic(_hWnd))
+ //{
+ // need to do this before Restore
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw our window is an icon maximize it ");
+ //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw show our window is an icon maximize it ");
+ // When the full screen window is switch out by ALT-TAB or ALT-CTRL-DEL-TASKMANAGER,
+ // this call will hang the app. Remove it to fix the problem.
+ // FIXME:
+ // 1) Why we want to active and max the window when it was minimized?
+ // 2) Why this is needed before restore? We didn't do that in non full screen mode.
+ //::ShowWindow(_hWnd, SW_SHOWMAXIMIZED);
+ //}
+ ddrval = _primarySurface->Restore();
+ if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw failed to restore _primarySurface");
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "DirectDraw restored _primarySurface");
+
+ _clearMixingSurface = true;
+
+ }
+ else if (ddrval != DD_OK)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "DirectDraw blt error 0x%x _backSurface", ddrval);
+ return -1;
+ //return VIDEO_DIRECT_DRAW_FAILURE;
+ }
+ return 0;
+}
+
+/*
+ Saving the code for using a clip list instead of HWND, problem was that other transparent
+ HWNDs caused us not to update an area or that we painted in other HWNDs area.
+
+ RECT hWndRect;
+ ::GetWindowRect(_hWnd, &hWndRect);
+
+ LPRGNDATA lpClipList = (LPRGNDATA)malloc(sizeof(RGNDATAHEADER) + sizeof(RECT));
+
+ // now fill out all the structure fields
+ memcpy(lpClipList->Buffer, &hWndRect, sizeof(RECT));
+
+ ::CopyRect(&(lpClipList->rdh.rcBound), &hWndRect);
+ lpClipList->rdh.dwSize = sizeof(RGNDATAHEADER);
+ lpClipList->rdh.iType = RDH_RECTANGLES;
+ lpClipList->rdh.nCount = 1;
+ lpClipList->rdh.nRgnSize = sizeof(RECT) * lpClipList->rdh.nCount;
+ ddrval= _directDrawClipper->SetClipList(lpClipList, 0);
+
+ void Visible(HWND hwnd, HRGN &hRgn)
+ {
+ if (!IsWindowVisible(hwnd)) // If the window is visible
+ {
+ if(CombineRgn(hRgn, hRgn, hRgn, RGN_XOR) == NULLREGION)
+ {
+ return;
+ }
+ }
+ // Gets the topmost window
+ HWND hWnd=GetTopWindow(NULL);
+ while (hWnd != NULL && hWnd != hwnd) // If the window is above in Z-order
+ {
+ if (IsWindowVisible(hWnd)) // If the window is visible
+ {
+ RECT Rect;
+ // Gets window dimension
+ GetWindowRect(hWnd, &Rect);
+ // Creates a region corresponding to the window
+ if(Rect.left > 0) // test fo rnow
+ {
+ HRGN hrgnWnd = CreateRectRgn(Rect.left, Rect.top, Rect.right, Rect.bottom);
+ // int err = GetUpdateRgn(hWnd, hrgnWnd, FALSE);
+ // Creates a region corresponding to region not overlapped
+ if(CombineRgn(hRgn, hRgn, hrgnWnd, RGN_DIFF) == COMPLEXREGION)
+ {
+ int a = 0;
+ }
+ DeleteObject(hrgnWnd);
+ }
+ }
+ // Loops through all windows till the specified window
+ hWnd = GetWindow(hWnd, GW_HWNDNEXT);
+ }
+
+ HRGN region;
+ region = CreateRectRgn(0, 0, 500, 500);
+
+ // Get the affected region
+ // if (GetUpdateRgn(_hWnd, region, FALSE) != ERROR)
+ HDC dc = GetDC(_hWnd);
+ if(GetClipRgn(dc, region) > 0)
+ {
+ int buffsize;
+ UINT x;
+ RGNDATA *buff;
+ POINT TopLeft;
+
+ // Get the top-left point of the client area
+ TopLeft.x = 0;
+ TopLeft.y = 0;
+ if (!ClientToScreen(_hWnd, &TopLeft))
+ {
+ int a = 0;
+ }
+
+
+ // Get the size of buffer required
+ buffsize = GetRegionData(region, 0, 0);
+ if (buffsize != 0)
+ {
+ buff = (RGNDATA *) new BYTE [buffsize];
+ if (buff == NULL)
+ {
+ int a = 0;
+ }
+
+ // Now get the region data
+ if(GetRegionData(region, buffsize, buff))
+ {
+ if(buff->rdh.nCount > 0)
+ {
+ ::OffsetRect(&(buff->rdh.rcBound), TopLeft.x, TopLeft.y);
+ for (x=0; x<(buff->rdh.nCount); x++)
+ {
+ RECT *urect = (RECT *) (((BYTE *) buff) + sizeof(RGNDATAHEADER) + (x * sizeof(RECT)));
+ ::OffsetRect(urect, TopLeft.x, TopLeft.y);
+ char logStr[256];
+ _snprintf(logStr,256, "rect T:%d L:%d B:%d R:%d\n",urect->top, urect->left, urect->bottom, urect->right);
+ OutputDebugString(logStr);
+
+ }
+ OutputDebugString("\n");
+ _directDrawClipper->SetClipList(buff, 0);
+ }
+ LPRGNDATA lpClipList = (LPRGNDATA)malloc(sizeof(RGNDATAHEADER) + sizeof(RECT) * buff->rdh.nCount);
+ if(buff->rdh.nCount > 0)
+ {
+ _directDrawClipper->SetClipList(lpClipList, 0);
+
+ lpClipList->
+ DWORD size = sizeof(RGNDATAHEADER) + sizeof(RECT)* buff->rdh.nCount;
+ lpClipList->rdh.dwSize = sizeof(RGNDATAHEADER);
+ lpClipList->rdh.iType = RDH_RECTANGLES;
+ lpClipList->rdh.nCount = 1;
+
+ HRESULT ddrval1 = _directDrawClipper->GetClipList(NULL, lpClipList, &size);
+ memcpy(lpClipList->Buffer, &rcRectDest, sizeof(RECT));
+ ::CopyRect(&(lpClipList->rdh.rcBound), &rcRectDest);
+ _directDrawClipper->SetClipList(lpClipList, 0);
+ } }
+
+ for (x=0; x<(buff->rdh.nCount); x++)
+ {
+ // Obtain the rectangles from the list
+ RECT *urect = (RECT *) (((BYTE *) buff) + sizeof(RGNDATAHEADER) + (x * sizeof(RECT)));
+ int a = 0;
+
+ }
+ delete lpClipList;
+ }
+ delete buff;
+ }
+ }
+ */
+/*
+ void VideoRenderDirectDraw::Wait()
+ {
+ // wait for previus draw to complete
+ int count = 0;
+ DWORD scanLines = 0;
+ DWORD screenLines = _screenRect.bottom -1; // scanlines start on 0
+ DWORD screenLines75 = (screenLines*3)/4; // % of the screen is rendered
+ HRESULT hr = DD_OK;
+ if(_directDraw == NULL)
+ {
+ return;
+ }
+ hr =_directDraw->GetScanLine(&scanLines);
+ while ( screenLines75 > scanLines && hr == DD_OK)
+ {
+ // _confCritSect->Leave();
+ _screenEvent->Wait(10);
+ // _confCritSect->Enter();
+ if(_directDraw == NULL)
+ {
+ return;
+ }
+ hr = _directDraw->GetScanLine(&scanLines);
+ }
+ }
+ */
+
+WebRtc_Word32 VideoRenderDirectDraw::StartRender()
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::StopRender()
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderDirectDraw::ChangeWindow(void* window)
+{
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
+ return -1;
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h
new file mode 100644
index 0000000..e790661
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h
@@ -0,0 +1,399 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
+
+#include "typedefs.h"
+#include "i_video_render_win.h"
+#include "common_video/libyuv/include/libyuv.h"
+
+#include "ddraw.h"
+#include <Map>
+#include <List>
+
+// Added
+#include "video_render_defines.h"
+
+#pragma comment(lib, "ddraw.lib") // located in DirectX SDK
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class Trace;
+
+class VideoRenderDirectDraw;
+
+// some typedefs to make it easy to test different versions
+typedef IDirectDraw7 DirectDraw;
+typedef IDirectDrawSurface7 DirectDrawSurface;
+typedef DDSURFACEDESC2 DirectDrawSurfaceDesc;
+typedef DDSCAPS2 DirectDrawCaps;
+typedef std::pair<int, unsigned int> ZorderPair;
+
+class WindowsThreadCpuUsage
+{
+public:
+ WindowsThreadCpuUsage();
+ int GetCpuUsage(); //in % since last call
+ DWORD LastGetCpuTime()
+ {
+ return _lastGetCpuUsageTime;
+ }
+ const enum
+ {
+ CPU_CHECK_INTERVAL = 1000
+ };
+private:
+ _int64 _lastCpuUsageTime;
+ DWORD _lastGetCpuUsageTime;
+ int _lastCpuUsage;
+ HANDLE _hThread;
+ int _cores;
+};
+
+class DirectDrawStreamSettings
+{
+public:
+ DirectDrawStreamSettings();
+
+ float _startWidth;
+ float _stopWidth;
+ float _startHeight;
+ float _stopHeight;
+
+ float _cropStartWidth;
+ float _cropStopWidth;
+ float _cropStartHeight;
+ float _cropStopHeight;
+};
+
+class DirectDrawBitmapSettings
+{
+public:
+ DirectDrawBitmapSettings();
+ ~DirectDrawBitmapSettings();
+
+ int SetBitmap(Trace* trace, DirectDraw* directDraw);
+
+ HBITMAP _transparentBitMap;
+ float _transparentBitmapLeft;
+ float _transparentBitmapRight;
+ float _transparentBitmapTop;
+ float _transparentBitmapBottom;
+ int _transparentBitmapWidth;
+ int _transparentBitmapHeight;
+ DDCOLORKEY* _transparentBitmapColorKey;
+ DirectDrawSurface* _transparentBitmapSurface; // size of bitmap image
+};
+
+class DirectDrawTextSettings
+{
+public:
+ DirectDrawTextSettings();
+ ~DirectDrawTextSettings();
+
+ int SetText(const char* text, int textLength, COLORREF colorText,
+ COLORREF colorBg, float left, float top, float right,
+ float bottom);
+
+ char* _ptrText;
+ WebRtc_UWord32 _textLength;
+ COLORREF _colorRefText;
+ COLORREF _colorRefBackground;
+ float _textLeft;
+ float _textRight;
+ float _textTop;
+ float _textBottom;
+ bool _transparent;
+};
+
+class DirectDrawChannel: public VideoRenderCallback
+{
+public:
+ DirectDrawChannel(DirectDraw* directDraw,
+ VideoType blitVideoType,
+ VideoType incomingVideoType,
+ VideoType screenVideoType,
+ VideoRenderDirectDraw* owner);
+
+ int FrameSizeChange(int width, int height, int numberOfStreams);
+ int DeliverFrame(unsigned char* buffer, int buffeSize,
+ unsigned int timeStamp90KHz);
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame);
+
+ int ChangeDeliverColorFormat(bool useScreenType);
+
+ void AddRef();
+ void Release();
+
+ void SetStreamSettings(VideoRenderDirectDraw* DDObj, short streamId,
+ float startWidth, float startHeight,
+ float stopWidth, float stopHeight);
+ void SetStreamCropSettings(VideoRenderDirectDraw* DDObj,
+ short streamId, float startWidth,
+ float startHeight, float stopWidth,
+ float stopHeight);
+
+ int GetStreamSettings(VideoRenderDirectDraw* DDObj, short streamId,
+ float& startWidth, float& startHeight,
+ float& stopWidth, float& stopHeight);
+
+ void GetLargestSize(RECT* mixingRect);
+ int
+ BlitFromOffscreenBufferToMixingBuffer(
+ VideoRenderDirectDraw* DDObj,
+ short streamID,
+ DirectDrawSurface* mixingSurface,
+ RECT &dstRect, bool demuxing);
+ bool IsOffScreenSurfaceUpdated(VideoRenderDirectDraw* DDobj);
+
+protected:
+ virtual ~DirectDrawChannel();
+
+private:
+ CriticalSectionWrapper* _critSect; // protect members from change while using them
+ int _refCount;
+ int _width;
+ int _height;
+ int _numberOfStreams;
+ bool _deliverInScreenType;
+ bool _doubleBuffer;
+ DirectDraw* _directDraw;
+ DirectDrawSurface* _offScreenSurface; // size of incoming stream
+ DirectDrawSurface* _offScreenSurfaceNext; // size of incoming stream
+ VideoType _blitVideoType;
+ VideoType _originalBlitVideoType;
+ VideoType _incomingVideoType;
+ VideoType _screenVideoType;
+ enum
+ {
+ MAX_FRAMEDELIVER_TIME = 20
+ }; //Maximum time it might take to deliver a frame (process time in DeliverFrame)
+ enum
+ {
+ MAX_NO_OF_LATE_FRAMEDELIVER_TIME = 10
+ }; //No of times we allow DeliverFrame process time to exceed MAX_FRAMEDELIVER_TIME before we take action.
+ VideoFrame _tempRenderBuffer;
+
+ std::map<unsigned long long, DirectDrawStreamSettings*>
+ _streamIdToSettings;
+ bool _offScreenSurfaceUpdated;
+ VideoRenderDirectDraw* _owner;
+};
+
+class VideoRenderDirectDraw: IVideoRenderWin
+{
+public:
+ VideoRenderDirectDraw(Trace* trace, HWND hWnd, bool fullscreen);
+ ~VideoRenderDirectDraw();
+public:
+ //IVideoRenderWin
+
+ /**************************************************************************
+ *
+ * Init
+ *
+ ***************************************************************************/
+ virtual WebRtc_Word32 Init();
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+ virtual VideoRenderCallback
+ * CreateChannel(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder, const float left,
+ const float top, const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom);
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual bool IsFullScreen();
+
+ virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
+ WebRtc_UWord64& availableMemory);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 colorText,
+ const WebRtc_UWord32 colorBg,
+ const float left, const float top,
+ const float rigth, const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
+ const WebRtc_UWord16 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+public:
+
+ // Used for emergency stops...
+ int Stop();
+
+ DirectDrawChannel* ShareDirectDrawChannel(int channel);
+ DirectDrawChannel* ConfigureDirectDrawChannel(int channel,
+ unsigned char streamID,
+ int zOrder, float left,
+ float top, float right,
+ float bottom);
+
+ int AddDirectDrawChannel(int channel, unsigned char streamID, int zOrder,
+ DirectDrawChannel*);
+
+ VideoType GetPerferedVideoFormat();
+ bool HasChannels();
+ bool HasChannel(int channel);
+ bool DeliverInScreenType();
+ int GetChannels(std::list<int>& channelList);
+
+ // code for getting graphics settings
+ int GetScreenResolution(int& screenWidth, int& screenHeight);
+ int UpdateSystemCPUUsage(int systemCPU);
+
+ int SetBitmap(HBITMAP bitMap, unsigned char pictureId,
+ DDCOLORKEY* colorKey, float left, float top, float rigth,
+ float bottom);
+
+ bool IsPrimaryOrMixingSurfaceOnSystem();
+ bool CanBltFourCC()
+ {
+ return _bCanBltFourcc;
+ }
+
+protected:
+ static bool RemoteRenderingThreadProc(void* obj);
+ bool RemoteRenderingProcess();
+
+private:
+ int CheckCapabilities();
+ int CreateMixingSurface();
+ int CreatePrimarySurface();
+
+ int FillSurface(DirectDrawSurface *pDDSurface, RECT* rect);
+ int DrawOnSurface(unsigned char* buffer, int buffeSize);
+ int BlitFromOffscreenBuffersToMixingBuffer();
+ int BlitFromBitmapBuffersToMixingBuffer();
+ int BlitFromTextToMixingBuffer();
+
+ bool HasHWNDChanged();
+ void DecideBestRenderingMode(bool hwndChanged, int totalRenderTime);
+
+ // in fullscreen flip mode
+ int WaitAndFlip(int& waitTime);
+ int BlitFromMixingBufferToBackBuffer();
+
+ // in normal window mode
+ int BlitFromMixingBufferToFrontBuffer(bool hwndChanged, int& waitTime);
+
+ // private members
+ Trace* _trace;
+ CriticalSectionWrapper* _confCritSect; // protect members from change while using them
+
+ bool _fullscreen;
+ bool _demuxing;
+ bool _transparentBackground;
+ bool _supportTransparency;
+ bool _canStretch;
+ bool _canMirrorLeftRight;
+ bool _clearMixingSurface;
+ bool _deliverInScreenType;
+ bool _renderModeWaitForCorrectScanLine;
+ bool _deliverInHalfFrameRate;
+ bool _deliverInQuarterFrameRate;
+ bool _bCanBltFourcc;
+ bool _frameChanged; // True if a frame has changed or bitmap or text has changed.
+ int _processCount;
+ HWND _hWnd;
+ RECT _screenRect; // whole screen as a rect
+ RECT _mixingRect;
+ RECT _originalHwndRect;
+ RECT _hwndRect;
+
+ VideoType _incomingVideoType;
+ VideoType _blitVideoType;
+ VideoType _rgbVideoType;
+
+ DirectDraw* _directDraw;
+ DirectDrawSurface* _primarySurface; // size of screen
+ DirectDrawSurface* _backSurface; // size of screen
+ DirectDrawSurface* _mixingSurface; // size of screen
+
+ std::map<unsigned char, DirectDrawBitmapSettings*> _bitmapSettings;
+ std::map<unsigned char, DirectDrawTextSettings*> _textSettings;
+ std::map<int, DirectDrawChannel*> _directDrawChannels;
+ std::multimap<int, unsigned int> _directDrawZorder;
+
+ EventWrapper* _fullScreenWaitEvent;
+ EventWrapper* _screenEvent;
+ ThreadWrapper* _screenRenderThread;
+ WindowsThreadCpuUsage _screenRenderCpuUsage;
+
+ int _lastRenderModeCpuUsage;
+
+ // Used for emergency stop caused by OnDisplayChange
+ bool _blit;
+
+ //code for providing graphics settings
+ DWORD _totalMemory;
+ DWORD _availableMemory;
+ int _systemCPUUsage;
+
+ // Variables used for checking render time
+ int _maxAllowedRenderTime;
+ int _nrOfTooLongRenderTimes;
+ bool _isPrimaryOrMixingSurfaceOnSystem;
+};
+
+} //namespace webrtc
+
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc
new file mode 100644
index 0000000..6f2237f
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc
@@ -0,0 +1,989 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+#include "video_render_windows_impl.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#ifdef DIRECTDRAW_RENDERING
+#include "video_render_directdraw.h"
+#endif
+#ifdef DIRECT3D9_RENDERING
+#include "video_render_direct3d9.h"
+#endif
+
+#include <tchar.h>
+
+namespace webrtc {
+
+VideoRenderWindowsImpl::VideoRenderWindowsImpl(
+ const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window,
+ const bool fullscreen) :
+ _id(id),
+ _renderWindowsCritsect(
+ *CriticalSectionWrapper::CreateCriticalSection()),
+ _prtWindow(window), _fullscreen(fullscreen), _ptrRendererWin(NULL)
+{
+}
+
+VideoRenderWindowsImpl::~VideoRenderWindowsImpl()
+{
+ delete &_renderWindowsCritsect;
+ if (_ptrRendererWin)
+ {
+ delete _ptrRendererWin;
+ _ptrRendererWin = NULL;
+ }
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::Init()
+{
+ //LogOSAndHardwareDetails();
+ CheckHWAcceleration();
+
+ _renderMethod = kVideoRenderWinD3D9;
+
+ // Create the win renderer
+ switch (_renderMethod)
+ {
+ case kVideoRenderWinDd:
+ {
+#ifdef DIRECTDRAW_RENDERING
+ VideoRenderDirectDraw* ptrRenderer;
+ ptrRenderer = new VideoRenderDirectDraw(NULL, (HWND) _prtWindow, _fullscreen);
+ if (ptrRenderer == NULL)
+ {
+ break;
+ }
+ _ptrRendererWin = reinterpret_cast<IVideoRenderWin*>(ptrRenderer);
+#else
+ return NULL;
+#endif //DIRECTDRAW_RENDERING
+ }
+ break;
+ case kVideoRenderWinD3D9:
+ {
+#ifdef DIRECT3D9_RENDERING
+ VideoRenderDirect3D9* ptrRenderer;
+ ptrRenderer = new VideoRenderDirect3D9(NULL, (HWND) _prtWindow, _fullscreen);
+ if (ptrRenderer == NULL)
+ {
+ break;
+ }
+ _ptrRendererWin = reinterpret_cast<IVideoRenderWin*>(ptrRenderer);
+#else
+ return NULL;
+#endif //DIRECT3D9_RENDERING
+ }
+ break;
+ default:
+ break;
+ }
+
+ //Init renderer
+ if (_ptrRendererWin)
+ return _ptrRendererWin->Init();
+ else
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ _id = id;
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::ChangeWindow(void* window)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ if (!_ptrRendererWin)
+ {
+ return -1;
+ }
+ else
+ {
+ return _ptrRendererWin->ChangeWindow(window);
+ }
+}
+
+VideoRenderCallback*
+VideoRenderWindowsImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ VideoRenderCallback* renderCallback = NULL;
+
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ renderCallback = _ptrRendererWin->CreateChannel(streamId, zOrder, left,
+ top, right, bottom);
+ }
+
+ return renderCallback;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::DeleteIncomingRenderStream(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->DeleteChannel(streamId);
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::GetIncomingRenderStreamProperties(
+ const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left,
+ float& top,
+ float& right,
+ float& bottom) const
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ zOrder = 0;
+ left = 0;
+ top = 0;
+ right = 0;
+ bottom = 0;
+
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->GetStreamSettings(streamId, 0, zOrder, left,
+ top, right, bottom);
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::StartRender()
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->StartRender();
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::StopRender()
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->StopRender();
+ }
+ return error;
+}
+
+VideoRenderType VideoRenderWindowsImpl::RenderType()
+{
+ return kRenderWindows;
+}
+
+RawVideoType VideoRenderWindowsImpl::PerferedVideoType()
+{
+ return kVideoI420;
+}
+
+bool VideoRenderWindowsImpl::FullScreen()
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ bool fullscreen = false;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ fullscreen = _ptrRendererWin->IsFullScreen();
+ }
+ return fullscreen;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::GetGraphicsMemory(
+ WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const
+{
+ if (_ptrRendererWin)
+ {
+ return _ptrRendererWin->GetGraphicsMemory(totalGraphicsMemory,
+ availableGraphicsMemory);
+ }
+
+ totalGraphicsMemory = 0;
+ availableGraphicsMemory = 0;
+ return -1;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::GetScreenResolution(
+ WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ screenWidth = 0;
+ screenHeight = 0;
+ return 0;
+}
+
+WebRtc_UWord32 VideoRenderWindowsImpl::RenderFrameRate(
+ const WebRtc_UWord32 streamId)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ return 0;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::SetStreamCropping(
+ const WebRtc_UWord32 streamId,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->SetCropping(streamId, 0, left, top, right,
+ bottom);
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::ConfigureRenderer(
+ const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->ConfigureRenderer(streamId, 0, zOrder, left,
+ top, right, bottom);
+ }
+
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::SetTransparentBackground(
+ const bool enable)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->SetTransparentBackground(enable);
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::SetText(
+ const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->SetText(textId, text, textLength,
+ textColorRef, backgroundColorRef,
+ left, top, right, bottom);
+ }
+ return error;
+}
+
+WebRtc_Word32 VideoRenderWindowsImpl::SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom)
+{
+ CriticalSectionScoped cs(_renderWindowsCritsect);
+ WebRtc_Word32 error = -1;
+ if (!_ptrRendererWin)
+ {
+ }
+ else
+ {
+ error = _ptrRendererWin->SetBitmap(bitMap, pictureId, colorKey, left,
+ top, right, bottom);
+ }
+ return error;
+}
+
+void VideoRenderWindowsImpl::LogOSAndHardwareDetails()
+{
+ HRESULT hr;
+ IDxDiagProvider* m_pDxDiagProvider = NULL;
+ IDxDiagContainer* m_pDxDiagRoot = NULL;
+
+ hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
+ bool coUninitializeIsRequired = true;
+ if (FAILED(hr))
+ {
+ // Avoid calling CoUninitialize() since CoInitializeEx() failed.
+ coUninitializeIsRequired = false;
+ if (hr == RPC_E_CHANGED_MODE)
+ {
+ // Calling thread has already initialized COM to be used in a single-threaded
+ // apartment (STA). We are then prevented from using STA.
+ // Details: hr = 0x80010106 <=> "Cannot change thread mode after it is set".
+ //
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideoRenderer,
+ _id,
+ "VideoRenderWindowsImpl::LogOSAndHardwareDetails() CoInitializeEx(NULL, COINIT_APARTMENTTHREADED) => RPC_E_CHANGED_MODE, error 0x%x",
+ hr);
+ }
+ }
+
+ hr = CoCreateInstance(CLSID_DxDiagProvider, NULL, CLSCTX_INPROC_SERVER,
+ IID_IDxDiagProvider, (LPVOID*) &m_pDxDiagProvider);
+
+ if (FAILED(hr) || m_pDxDiagProvider == NULL)
+ {
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ // Fill out a DXDIAG_INIT_PARAMS struct and pass it to IDxDiagContainer::Initialize
+ // Passing in TRUE for bAllowWHQLChecks, allows dxdiag to check if drivers are
+ // digital signed as logo'd by WHQL which may connect via internet to update
+ // WHQL certificates.
+ DXDIAG_INIT_PARAMS dxDiagInitParam;
+ ZeroMemory(&dxDiagInitParam, sizeof(DXDIAG_INIT_PARAMS));
+
+ dxDiagInitParam.dwSize = sizeof(DXDIAG_INIT_PARAMS);
+ dxDiagInitParam.dwDxDiagHeaderVersion = DXDIAG_DX9_SDK_VERSION;
+ dxDiagInitParam.bAllowWHQLChecks = TRUE;
+ dxDiagInitParam.pReserved = NULL;
+
+ hr = m_pDxDiagProvider->Initialize(&dxDiagInitParam);
+ if (FAILED(hr))
+ {
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ hr = m_pDxDiagProvider->GetRootContainer(&m_pDxDiagRoot);
+ if (FAILED(hr) || m_pDxDiagRoot == NULL)
+ {
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ IDxDiagContainer* pObject = NULL;
+
+ hr = m_pDxDiagRoot->GetChildContainer(L"DxDiag_SystemInfo", &pObject);
+ if (FAILED(hr) || pObject == NULL)
+ {
+ m_pDxDiagRoot->Release();
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ TCHAR m_szDirectXVersionLongEnglish[100];
+ TCHAR m_szOSLocalized[100];
+ TCHAR m_szProcessorEnglish[200];
+ TCHAR m_szSystemManufacturerEnglish[200];
+
+ ZeroMemory(m_szDirectXVersionLongEnglish, sizeof(TCHAR) * 100);
+ ZeroMemory(m_szOSLocalized, sizeof(TCHAR) * 100);
+ ZeroMemory(m_szProcessorEnglish, sizeof(TCHAR) * 200);
+ ZeroMemory(m_szSystemManufacturerEnglish, sizeof(TCHAR) * 200);
+
+ GetStringValue( pObject, L"szDirectXVersionLongEnglish",
+ EXPAND(m_szDirectXVersionLongEnglish) );
+ GetStringValue(pObject, L"szOSLocalized", EXPAND(m_szOSLocalized) );
+ GetStringValue(pObject, L"szProcessorEnglish", EXPAND(m_szProcessorEnglish) );
+ GetStringValue( pObject, L"szSystemManufacturerEnglish",
+ EXPAND(m_szSystemManufacturerEnglish) );
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "System Manufacturer --- %s",
+ m_szSystemManufacturerEnglish);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Processor --- %s", m_szProcessorEnglish);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Operating System --- %s", m_szOSLocalized);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "DirectX Version --- %s",
+ m_szDirectXVersionLongEnglish);
+
+ if (pObject)
+ pObject->Release();
+
+ struct DisplayInfo
+ {
+ TCHAR m_szDescription[200];
+ TCHAR m_szManufacturer[200];
+ TCHAR m_szChipType[100];
+ TCHAR m_szDisplayMemoryEnglish[100];
+ TCHAR m_szDisplayModeEnglish[100];
+ TCHAR m_szDriverName[100];
+ TCHAR m_szDriverVersion[100];
+ TCHAR m_szDDStatusEnglish[100];
+ TCHAR m_szD3DStatusEnglish[100];
+ BOOL m_bDDAccelerationEnabled;
+ BOOL m_bNoHardware;
+ BOOL m_b3DAccelerationExists;
+ BOOL m_b3DAccelerationEnabled;
+ };
+
+ WCHAR wszContainer[256];
+ IDxDiagContainer* pContainer = NULL;
+
+ DWORD nInstanceCount = 0;
+ DWORD nItem = 0;
+ DWORD nCurCount = 0;
+
+ // Get the IDxDiagContainer object called "DxDiag_DisplayDevices".
+ // This call may take some time while dxdiag gathers the info.
+ if (FAILED(hr = m_pDxDiagRoot->GetChildContainer(L"DxDiag_DisplayDevices",
+ &pContainer)))
+ {
+ m_pDxDiagRoot->Release();
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ if (FAILED(hr = pContainer->GetNumberOfChildContainers(&nInstanceCount)))
+ {
+ pContainer->Release();
+ m_pDxDiagRoot->Release();
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ DisplayInfo *pDisplayInfo = new DisplayInfo;
+ if (pDisplayInfo == NULL)
+ return;
+ ZeroMemory(pDisplayInfo, sizeof(DisplayInfo));
+
+ hr = pContainer->EnumChildContainerNames(nItem, wszContainer, 256);
+ if (FAILED(hr))
+ {
+ delete pDisplayInfo;
+ pContainer->Release();
+ m_pDxDiagRoot->Release();
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ hr = pContainer->GetChildContainer(wszContainer, &pObject);
+ if (FAILED(hr) || pObject == NULL)
+ {
+ delete pDisplayInfo;
+ pContainer->Release();
+ m_pDxDiagRoot->Release();
+ m_pDxDiagProvider->Release();
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+ return;
+ }
+
+ GetStringValue( pObject, L"szDescription",
+ EXPAND(pDisplayInfo->m_szDescription) );
+ GetStringValue( pObject, L"szManufacturer",
+ EXPAND(pDisplayInfo->m_szManufacturer) );
+ GetStringValue(pObject, L"szChipType", EXPAND(pDisplayInfo->m_szChipType) );
+ GetStringValue( pObject, L"szDisplayMemoryEnglish",
+ EXPAND(pDisplayInfo->m_szDisplayMemoryEnglish) );
+ GetStringValue( pObject, L"szDisplayModeEnglish",
+ EXPAND(pDisplayInfo->m_szDisplayModeEnglish) );
+ GetStringValue( pObject, L"szDriverName",
+ EXPAND(pDisplayInfo->m_szDriverName) );
+ GetStringValue( pObject, L"szDriverVersion",
+ EXPAND(pDisplayInfo->m_szDriverVersion) );
+ GetBoolValue(pObject, L"bDDAccelerationEnabled",
+ &pDisplayInfo->m_bDDAccelerationEnabled);
+ GetBoolValue(pObject, L"bNoHardware", &pDisplayInfo->m_bNoHardware);
+ GetBoolValue(pObject, L"bDDAccelerationEnabled",
+ &pDisplayInfo->m_bDDAccelerationEnabled);
+ GetBoolValue(pObject, L"b3DAccelerationExists",
+ &pDisplayInfo->m_b3DAccelerationExists);
+ GetBoolValue(pObject, L"b3DAccelerationEnabled",
+ &pDisplayInfo->m_b3DAccelerationEnabled);
+ GetStringValue( pObject, L"szDDStatusEnglish",
+ EXPAND(pDisplayInfo->m_szDDStatusEnglish));
+ GetStringValue( pObject, L"szD3DStatusEnglish",
+ EXPAND(pDisplayInfo->m_szD3DStatusEnglish));
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Device Name --- %s",
+ pDisplayInfo->m_szDescription);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Device Manufacturer --- %s",
+ pDisplayInfo->m_szManufacturer);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Device ChipType --- %s",
+ pDisplayInfo->m_szChipType);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Approx. Total Device Memory --- %s",
+ pDisplayInfo->m_szDisplayMemoryEnglish);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Current Display Mode --- %s",
+ pDisplayInfo->m_szDisplayModeEnglish);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Device Driver Name --- %s",
+ pDisplayInfo->m_szDriverName);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "Device Driver Version --- %s",
+ pDisplayInfo->m_szDriverVersion);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "DirectDraw Acceleration Enabled --- %s",
+ pDisplayInfo->m_szDescription ? "Enabled" : "Disabled");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "bNoHardware --- %s",
+ pDisplayInfo->m_bNoHardware ? "Enabled" : "Disabled");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "b3DAccelerationExists Enabled --- %s",
+ pDisplayInfo->m_b3DAccelerationExists ? "Enabled" : "Disabled");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "b3DAccelerationEnabled Enabled --- %s",
+ pDisplayInfo->m_b3DAccelerationEnabled ? "Enabled"
+ : "Disabled");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "DDraw Status --- %s",
+ pDisplayInfo->m_szDDStatusEnglish);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
+ "D3D Status --- %s",
+ pDisplayInfo->m_szD3DStatusEnglish);
+
+ // Get OS version
+ OSVERSIONINFOEX osvie;
+ osvie.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+ GetVersionEx((LPOSVERSIONINFO) & osvie);
+ /*
+ Operating system Version number dwMajorVersion dwMinorVersion
+ Windows 7 6.1 6 1
+ Windows Server 2008 R2 6.1 6 1
+ Windows Server 2008 6.0 6 0
+ Windows Vista 6.0 6 0
+ Windows Server 2003 R2 5.2 5 2
+ Windows Server 2003 5.2 5 2
+ Windows XP 5.1 5 1
+ Windows 2000 5.0 5 0
+ */
+ //RDP problem exists only when XP is involved
+ if (osvie.dwMajorVersion < 6)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideoRenderer, _id,
+ "Checking for RDP driver");
+ if (_tcsncmp(pDisplayInfo->m_szDriverName, _T("RDPDD.dll"), 9) == 0)
+ {
+ //
+ }
+ }
+
+ if (pObject)
+ {
+ pObject->Release();
+ pObject = NULL;
+ }
+
+ if (pContainer)
+ pContainer->Release();
+
+ if (m_pDxDiagProvider)
+ m_pDxDiagProvider->Release();
+
+ if (m_pDxDiagRoot)
+ m_pDxDiagRoot->Release();
+
+ if (pDisplayInfo)
+ delete pDisplayInfo;
+
+ if (coUninitializeIsRequired)
+ CoUninitialize();
+
+ return;
+}
+
+//-----------------------------------------------------------------------------
+// Name: GetStringValue()
+// Desc: Get a string value from a IDxDiagContainer object
+//-----------------------------------------------------------------------------
+HRESULT VideoRenderWindowsImpl::GetStringValue(IDxDiagContainer* pObject,
+ WCHAR* wstrName,
+ TCHAR* strValue, int nStrLen)
+{
+ HRESULT hr;
+ VARIANT var;
+ VariantInit(&var);
+
+ if (FAILED(hr = pObject->GetProp(wstrName, &var)))
+ return hr;
+
+ if (var.vt != VT_BSTR)
+ return E_INVALIDARG;
+
+#ifdef _UNICODE
+ wcsncpy( strValue, var.bstrVal, nStrLen-1 );
+#else
+ wcstombs(strValue, var.bstrVal, nStrLen);
+#endif
+ strValue[nStrLen - 1] = TEXT('\0');
+ VariantClear(&var);
+
+ return S_OK;
+}
+
+//-----------------------------------------------------------------------------
+// Name: GetBoolValue()
+// Desc: Get a BOOL value from a IDxDiagContainer object
+//-----------------------------------------------------------------------------
+HRESULT VideoRenderWindowsImpl::GetBoolValue(IDxDiagContainer* pObject,
+ WCHAR* wstrName, BOOL* pbValue)
+{
+ HRESULT hr;
+ VARIANT var;
+ VariantInit(&var);
+
+ if (FAILED(hr = pObject->GetProp(wstrName, &var)))
+ return hr;
+
+ if (var.vt != VT_BOOL)
+ return E_INVALIDARG;
+
+ *pbValue = (var.boolVal != 0);
+ VariantClear(&var);
+
+ return S_OK;
+}
+
+int VideoRenderWindowsImpl::CheckHWAcceleration()
+{
+ // Read the registry to check if HW acceleration is enabled or not.
+ HKEY regKey;
+ DWORD value = 0;
+ DWORD valueLength = 4;
+
+ bool directDraw = true;
+ bool direct3D = true;
+ bool dci = true;
+
+ // DirectDraw
+ if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, _T("SOFTWARE\\Microsoft\\DirectDraw"),
+ 0, KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ // We have the registry key
+ value = 0;
+ if (RegQueryValueEx(regKey, _T("EmulationOnly"), NULL, NULL,
+ (BYTE*) &value, &valueLength) == ERROR_SUCCESS)
+ {
+ if (value == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw acceleration is disabled");
+ directDraw = false;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DirectDraw acceleration is enabled");
+ }
+ }
+ else
+ {
+ // Could not get the value for this one.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not find EmulationOnly key, DirectDraw acceleration is probably enabled");
+ }
+ RegCloseKey(regKey);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not open DirectDraw settings");
+ }
+
+ // Direct3D
+ if (RegOpenKeyEx(HKEY_LOCAL_MACHINE,
+ _T("SOFTWARE\\Microsoft\\Direct3D\\Drivers"), 0,
+ KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ // We have the registry key
+ value = 0;
+ if (RegQueryValueEx(regKey, _T("SoftwareOnly"), NULL, NULL,
+ (BYTE*) &value, &valueLength) == ERROR_SUCCESS)
+ {
+ if (value == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "Direct3D acceleration is disabled");
+ direct3D = false;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "Direct3D acceleration is enabled");
+ }
+ }
+ else
+ {
+ // Could not get the value for this one.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not find SoftwarOnly key, Direct3D acceleration is probably enabled");
+ }
+ RegCloseKey(regKey);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not open Direct3D settings");
+ }
+
+ // DCI
+ if (RegOpenKeyEx(
+ HKEY_LOCAL_MACHINE,
+ _T(
+ "SYSTEM\\CurrentControlSet\\Control\\GraphicsDrivers\\DCI"),
+ 0, KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ // We have found the registry key
+ value = 0;
+ if (RegQueryValueEx(regKey, _T("Timeout"), NULL, NULL, (BYTE*) &value,
+ &valueLength) == ERROR_SUCCESS)
+ {
+ if (value == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DCI - DirectDraw acceleration is disabled");
+ dci = false;
+ }
+ else if (value == 7)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
+ "DCI is fully enabled");
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVideo,
+ -1,
+ "DCI - DirectDraw acceleration is enabled, but short timeout: %d",
+ value);
+ }
+ }
+ else
+ {
+ // Could not get the value for this one.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not find Timeout key");
+ }
+ RegCloseKey(regKey);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Could not open DCI settings");
+ }
+
+ // We don't care about Direct3D right now...
+ if (dci == false || directDraw == false)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+void VideoRenderWindowsImpl::CheckHWDriver(bool& badDriver,
+ bool& fullAccelerationEnabled)
+{
+ // Read the registry to check if HW acceleration is enabled or not.
+ HKEY regKey;
+ DWORD value = 0;
+ DWORD valueLength = 4;
+
+ //Assume the best
+ badDriver = false;
+ fullAccelerationEnabled = true;
+
+ // Check the path to the currently used driver
+ if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, _T("HARDWARE\\DEVICEMAP\\VIDEO"), 0,
+ KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ // We have found the registry key containing the driver location
+ value = 0;
+ DWORD driverPathLen = 512;
+ TCHAR driverPath[512];
+ memset(driverPath, 0, driverPathLen * sizeof(TCHAR));
+ DWORD dwType = REG_SZ;
+
+ long retVal = RegQueryValueEx(regKey, _T("\\Device\\Video0"), NULL,
+ NULL, (BYTE*) driverPath, &driverPathLen);
+
+ // Close the key...
+ RegCloseKey(regKey);
+
+ if (retVal == ERROR_SUCCESS)
+ {
+ // We have the path to the currently used video card
+
+ // trueDriverPath = modified nameStr, from above, that works
+ // for RegOpenKeyEx
+ TCHAR trueDriverPath[512];
+ memset(trueDriverPath, 0, 512 * sizeof(TCHAR));
+
+ // Convert the path to correct format.
+ // - Remove \Registry\Machine\
+ // - Replace '\' with '\\'
+ // Should be something like this: System\\CurrentControlSet\\Control\\Video\\{F6987E15-F12C-4B15-8C84-0F635F3F09EA}\\0000"
+ int idx = 0;
+ for (DWORD i = 18; i < (driverPathLen / sizeof(TCHAR)); i++)
+ {
+ trueDriverPath[idx++] = driverPath[i];
+ if (driverPath[i] == _T('\\'))
+ {
+ trueDriverPath[idx++] = driverPath[i];
+ }
+ }
+
+ // Open the driver key
+ if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, trueDriverPath, 0,
+ KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ TCHAR driverName[64];
+ memset(driverName, 0, 64 * sizeof(TCHAR));
+ DWORD driverNameLength = 64;
+ retVal = RegQueryValueEx(regKey, _T("drv"), NULL, NULL,
+ (BYTE*) driverName, &driverNameLength);
+ if (retVal == ERROR_SUCCESS)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
+ "Graphics card driver name: %s", driverName);
+ }
+ DWORD accLevel = 0;
+ DWORD accLevelS = sizeof(accLevel);
+
+ RegQueryValueEx(regKey, _T("Acceleration.Level"), NULL, NULL,
+ (LPBYTE) & accLevel, &accLevelS);
+ //Don't care if the key is not found. It probably means that acceleration is enabled
+ if (accLevel != 0)
+ {
+ // Close the key...
+ RegCloseKey(regKey);
+
+ if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, trueDriverPath, 0,
+ KEY_SET_VALUE, ®Key) == ERROR_SUCCESS)
+ {
+ // try setting it to full
+ accLevel = 0;
+ LONG retVal;
+ retVal = RegSetValueEx(regKey,
+ _T("Acceleration.Level"), NULL,
+ REG_DWORD, (PBYTE) & accLevel,
+ sizeof(DWORD));
+ if (retVal != ERROR_SUCCESS)
+ {
+ fullAccelerationEnabled = false;
+ }
+ else
+ {
+ RegQueryValueEx(regKey, _T("Acceleration.Level"),
+ NULL, NULL, (LPBYTE) & accLevel,
+ &accLevelS);
+ if (accLevel != 0)
+ {
+ fullAccelerationEnabled = false;
+ }
+ else
+ {
+ fullAccelerationEnabled = true;
+ }
+ }
+ }
+ else
+ {
+ fullAccelerationEnabled = false;
+ }
+ }
+ else
+ {
+ fullAccelerationEnabled = true;
+ }
+
+ // Close the key...
+ RegCloseKey(regKey);
+ }
+ }
+ }
+}
+
+} //namespace webrtc
+
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h
new file mode 100644
index 0000000..bdc0a0a
--- /dev/null
+++ b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
+
+#include <Winerror.h>
+#include <dxdiag.h>
+
+#include "i_video_render.h"
+#include "i_video_render_win.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+#define EXPAND(x) x, sizeof(x)/sizeof(TCHAR)
+
+#pragma comment(lib, "dxguid.lib")
+
+enum VideoRenderWinMethod
+{
+ kVideoRenderWinDd = 0, kVideoRenderWinD3D9 = 1
+};
+
+// Class definitions
+class VideoRenderWindowsImpl: IVideoRender
+{
+public:
+ /*
+ * Constructor/destructor
+ */
+
+ VideoRenderWindowsImpl(const WebRtc_Word32 id,
+ const VideoRenderType videoRenderType,
+ void* window, const bool fullscreen);
+
+ virtual ~VideoRenderWindowsImpl();
+
+ virtual WebRtc_Word32 Init();
+
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ virtual WebRtc_Word32 ChangeWindow(void* window);
+
+ /**************************************************************************
+ *
+ * Incoming Streams
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderCallback
+ * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
+ const WebRtc_UWord32 zOrder,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32
+ DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32
+ GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
+ WebRtc_UWord32& zOrder,
+ float& left, float& top,
+ float& right, float& bottom) const;
+
+ /**************************************************************************
+ *
+ * Start/Stop
+ *
+ ***************************************************************************/
+
+ virtual WebRtc_Word32 StartRender();
+
+ virtual WebRtc_Word32 StopRender();
+
+ /**************************************************************************
+ *
+ * Properties
+ *
+ ***************************************************************************/
+
+ virtual VideoRenderType RenderType();
+
+ virtual RawVideoType PerferedVideoType();
+
+ virtual bool FullScreen();
+
+ virtual WebRtc_Word32
+ GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
+ WebRtc_UWord64& availableGraphicsMemory) const;
+
+ virtual WebRtc_Word32
+ GetScreenResolution(WebRtc_UWord32& screenWidth,
+ WebRtc_UWord32& screenHeight) const;
+
+ virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
+
+ virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
+ const unsigned int zOrder,
+ const float left, const float top,
+ const float right,
+ const float bottom);
+
+ virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
+
+ virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
+ const WebRtc_UWord8* text,
+ const WebRtc_Word32 textLength,
+ const WebRtc_UWord32 textColorRef,
+ const WebRtc_UWord32 backgroundColorRef,
+ const float left, const float top,
+ const float right, const float bottom);
+
+ virtual WebRtc_Word32 SetBitmap(const void* bitMap,
+ const WebRtc_UWord8 pictureId,
+ const void* colorKey, const float left,
+ const float top, const float right,
+ const float bottom);
+
+ static int CheckHWAcceleration();
+ static void CheckHWDriver(bool& badDriver, bool& fullAccelerationEnabled);
+
+private:
+
+ void LogOSAndHardwareDetails();
+ HRESULT GetBoolValue(IDxDiagContainer* pObject, WCHAR* wstrName,
+ BOOL* pbValue);
+ HRESULT GetStringValue(IDxDiagContainer* pObject, WCHAR* wstrName,
+ TCHAR* strValue, int nStrLen);
+
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _renderWindowsCritsect;
+
+ void* _prtWindow;
+ bool _fullscreen;
+
+ VideoRenderWinMethod _renderMethod;
+ IVideoRenderWin* _ptrRendererWin;
+};
+
+} //namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp b/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp
new file mode 100644
index 0000000..c443a58
--- /dev/null
+++ b/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp
Binary files differ
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc b/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc
new file mode 100644
index 0000000..607a866
--- /dev/null
+++ b/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc
@@ -0,0 +1,680 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testAPI.h"
+
+#include <stdio.h>
+
+#if defined(_WIN32)
+#include <tchar.h>
+#include <windows.h>
+#include <cassert>
+#include <fstream>
+#include <iostream>
+#include <string>
+#include <windows.h>
+#include <ddraw.h>
+
+#elif defined(WEBRTC_LINUX)
+
+#include <iostream>
+#include <X11/Xlib.h>
+#include <X11/Xutil.h>
+#include <sys/time.h>
+
+#endif
+
+#include "common_types.h"
+#include "process_thread.h"
+#include "module_common_types.h"
+#include "video_render_defines.h"
+#include "video_render.h"
+#include "tick_util.h"
+#include "trace.h"
+
+using namespace webrtc;
+
+void GetTestVideoFrame(WebRtc_UWord8* frame,
+ WebRtc_Word32 width,
+ WebRtc_Word32 height,
+ WebRtc_UWord8 startColor);
+int TestSingleStream(VideoRender* renderModule);
+int TestFullscreenStream(VideoRender* &renderModule,
+ void* window,
+ const VideoRenderType videoRenderType);
+int TestBitmapText(VideoRender* renderModule);
+int TestMultipleStreams(VideoRender* renderModule);
+int TestExternalRender(VideoRender* renderModule);
+
+#define TEST_FRAME_RATE 30
+#define TEST_TIME_SECOND 5
+#define TEST_FRAME_NUM (TEST_FRAME_RATE*TEST_TIME_SECOND)
+#define TEST_STREAM0_START_COLOR 0
+#define TEST_STREAM1_START_COLOR 64
+#define TEST_STREAM2_START_COLOR 128
+#define TEST_STREAM3_START_COLOR 192
+
+#if defined(_WIN32) && defined(_DEBUG)
+// #include "vld.h"
+#define SLEEP(x) ::Sleep(x)
+#elif defined(WEBRTC_LINUX)
+
+#define GET_TIME_IN_MS timeGetTime()
+#define SLEEP(x) Sleep(x)
+
+void Sleep(unsigned long x)
+{
+ timespec t;
+ t.tv_sec = x/1000;
+ t.tv_nsec = (x-(x/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+}
+
+unsigned long timeGetTime()
+{
+ struct timeval tv;
+ struct timezone tz;
+ unsigned long val;
+
+ gettimeofday(&tv, &tz);
+ val= tv.tv_sec*1000+ tv.tv_usec/1000;
+ return(val);
+}
+
+#elif defined(WEBRTC_MAC_INTEL)
+
+#include <unistd.h>
+
+#define GET_TIME_IN_MS timeGetTime()
+#define SLEEP(x) usleep(x * 1000)
+
+unsigned long timeGetTime()
+{
+ return 0;
+}
+
+#else
+
+#define GET_TIME_IN_MS ::timeGetTime()
+#define SLEEP(x) ::Sleep(x)
+
+#endif
+
+using namespace std;
+
+#if defined(_WIN32)
+LRESULT CALLBACK WebRtcWinProc( HWND hWnd,UINT uMsg,WPARAM wParam,LPARAM lParam)
+{
+ switch(uMsg)
+ {
+ case WM_DESTROY:
+ break;
+ case WM_COMMAND:
+ break;
+ }
+ return DefWindowProc(hWnd,uMsg,wParam,lParam);
+}
+
+int WebRtcCreateWindow(HWND &hwndMain,int winNum, int width, int height)
+{
+ HINSTANCE hinst = GetModuleHandle(0);
+ WNDCLASSEX wcx;
+ wcx.hInstance = hinst;
+ wcx.lpszClassName = TEXT("VideoRenderTest");
+ wcx.lpfnWndProc = (WNDPROC)WebRtcWinProc;
+ wcx.style = CS_DBLCLKS;
+ wcx.hIcon = LoadIcon (NULL, IDI_APPLICATION);
+ wcx.hIconSm = LoadIcon (NULL, IDI_APPLICATION);
+ wcx.hCursor = LoadCursor (NULL, IDC_ARROW);
+ wcx.lpszMenuName = NULL;
+ wcx.cbSize = sizeof (WNDCLASSEX);
+ wcx.cbClsExtra = 0;
+ wcx.cbWndExtra = 0;
+ wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
+
+ // Register our window class with the operating system.
+ // If there is an error, exit program.
+ if ( !RegisterClassEx (&wcx) )
+ {
+ MessageBox( 0, TEXT("Failed to register window class!"),TEXT("Error!"), MB_OK|MB_ICONERROR );
+ return 0;
+ }
+
+ // Create the main window.
+ hwndMain = CreateWindowEx(
+ 0, // no extended styles
+ TEXT("VideoRenderTest"), // class name
+ TEXT("VideoRenderTest Window"), // window name
+ WS_OVERLAPPED |WS_THICKFRAME, // overlapped window
+ 800, // horizontal position
+ 0, // vertical position
+ width, // width
+ height, // height
+ (HWND) NULL, // no parent or owner window
+ (HMENU) NULL, // class menu used
+ hinst, // instance handle
+ NULL); // no window creation data
+
+ if (!hwndMain)
+ {
+ int error = GetLastError();
+ return -1;
+ }
+
+ // Show the window using the flag specified by the program
+ // that started the application, and send the application
+ // a WM_PAINT message.
+
+ ShowWindow(hwndMain, SW_SHOWDEFAULT);
+ UpdateWindow(hwndMain);
+ return 0;
+}
+
+#elif defined(WEBRTC_LINUX)
+
+int WebRtcCreateWindow(Window *outWindow, Display **outDisplay, int winNum, int width, int height) // unsigned char* title, int titleLength)
+
+{
+ int screen, xpos = 10, ypos = 10;
+ XEvent evnt;
+ XSetWindowAttributes xswa; // window attribute struct
+ XVisualInfo vinfo; // screen visual info struct
+ unsigned long mask; // attribute mask
+
+ // get connection handle to xserver
+ Display* _display = XOpenDisplay( NULL );
+
+ // get screen number
+ screen = DefaultScreen(_display);
+
+ // put desired visual info for the screen in vinfo
+ if( XMatchVisualInfo(_display, screen, 24, TrueColor, &vinfo) != 0 )
+ {
+ //printf( "Screen visual info match!\n" );
+ }
+
+ // set window attributes
+ xswa.colormap = XCreateColormap(_display, DefaultRootWindow(_display), vinfo.visual, AllocNone);
+ xswa.event_mask = StructureNotifyMask | ExposureMask;
+ xswa.background_pixel = 0;
+ xswa.border_pixel = 0;
+
+ // value mask for attributes
+ mask = CWBackPixel | CWBorderPixel | CWColormap | CWEventMask;
+
+ switch( winNum )
+ {
+ case 0:
+ xpos = 200;
+ ypos = 200;
+ break;
+ case 1:
+ xpos = 300;
+ ypos = 200;
+ break;
+ default:
+ break;
+ }
+
+ // create a subwindow for parent (defroot)
+ Window _window = XCreateWindow(_display, DefaultRootWindow(_display),
+ xpos, ypos,
+ width,
+ height,
+ 0, vinfo.depth,
+ InputOutput,
+ vinfo.visual,
+ mask, &xswa);
+
+ // Set window name
+ if( winNum == 0 )
+ {
+ XStoreName(_display, _window, "VE MM Local Window");
+ XSetIconName(_display, _window, "VE MM Local Window");
+ }
+ else if( winNum == 1 )
+ {
+ XStoreName(_display, _window, "VE MM Remote Window");
+ XSetIconName(_display, _window, "VE MM Remote Window");
+ }
+
+ // make x report events for mask
+ XSelectInput(_display, _window, StructureNotifyMask);
+
+ // map the window to the display
+ XMapWindow(_display, _window);
+
+ // wait for map event
+ do
+ {
+ XNextEvent(_display, &evnt);
+ }
+ while (evnt.type != MapNotify || evnt.xmap.event != _window);
+
+ *outWindow = _window;
+ *outDisplay = _display;
+
+ return 0;
+}
+#endif // LINUX
+
+// Note: Mac code is in testApi_mac.mm.
+
+class MyRenderCallback: public VideoRenderCallback
+{
+public:
+ MyRenderCallback() :
+ _cnt(0)
+ {
+ }
+ ;
+ ~MyRenderCallback()
+ {
+ }
+ ;
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
+ VideoFrame& videoFrame)
+ {
+ _cnt++;
+ if (_cnt % 100 == 0)
+ {
+ printf("Render callback %d \n",_cnt);
+ }
+ return 0;
+ }
+ WebRtc_Word32 _cnt;
+};
+
+void GetTestVideoFrame(WebRtc_UWord8* frame,
+ WebRtc_Word32 width,
+ WebRtc_Word32 height,
+ WebRtc_UWord8 startColor) {
+ // changing color
+ static WebRtc_UWord8 color = startColor;
+
+ WebRtc_UWord8* destY = frame;
+ WebRtc_UWord8* destU = &frame[width*height];
+ WebRtc_UWord8* destV = &frame[width*height*5/4];
+ //Y
+ for (WebRtc_Word32 y=0; y<(width*height); y++)
+ {
+ destY[y] = color;
+ }
+ //U
+ for (WebRtc_Word32 u=0; u<(width*height/4); u++)
+ {
+ destU[u] = color;
+ }
+ //V
+ for (WebRtc_Word32 v=0; v<(width*height/4); v++)
+ {
+ destV[v] = color;
+ }
+
+ color++;
+}
+
+int TestSingleStream(VideoRender* renderModule) {
+ int error = 0;
+ // Add settings for a stream to render
+ printf("Add stream 0 to entire window\n");
+ const int streamId0 = 0;
+ VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
+ assert(renderCallback0 != NULL);
+
+#ifndef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+ MyRenderCallback externalRender;
+ renderModule->AddExternalRenderCallback(streamId0, &externalRender);
+#endif
+
+ printf("Start render\n");
+ error = renderModule->StartRender(streamId0);
+ if (error != 0) {
+ // TODO(phoglund): This test will not work if compiled in release mode.
+ // This rather silly construct here is to avoid compilation errors when
+ // compiling in release. Release => no asserts => unused 'error' variable.
+ assert(false);
+ }
+
+ // Loop through an I420 file and render each frame
+ const WebRtc_UWord32 width = 352;
+ const WebRtc_UWord32 height = 288;
+ const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
+
+ VideoFrame videoFrame0;
+ videoFrame0.VerifyAndAllocate(numBytes);
+
+ const WebRtc_UWord32 renderDelayMs = 500;
+
+ for (int i=0; i<TEST_FRAME_NUM; i++) {
+ GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
+ videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame0.SetWidth(width);
+ videoFrame0.SetHeight(height);
+ videoFrame0.SetLength(numBytes);
+ renderCallback0->RenderFrame(streamId0, videoFrame0);
+ SLEEP(1000/TEST_FRAME_RATE);
+ }
+
+ videoFrame0.Free();
+
+ // Shut down
+ printf("Closing...\n");
+ error = renderModule->StopRender(streamId0);
+ assert(error == 0);
+
+ error = renderModule->DeleteIncomingRenderStream(streamId0);
+ assert(error == 0);
+
+ return 0;
+}
+
+int TestFullscreenStream(VideoRender* &renderModule,
+ void* window,
+ const VideoRenderType videoRenderType) {
+ VideoRender::DestroyVideoRender(renderModule);
+ renderModule = VideoRender::CreateVideoRender(12345, window, true, videoRenderType);
+
+ TestSingleStream(renderModule);
+
+ VideoRender::DestroyVideoRender(renderModule);
+ renderModule = VideoRender::CreateVideoRender(12345, window, false, videoRenderType);
+
+ return 0;
+}
+
+int TestBitmapText(VideoRender* renderModule) {
+#if defined(WIN32)
+
+ int error = 0;
+ // Add settings for a stream to render
+ printf("Add stream 0 to entire window\n");
+ const int streamId0 = 0;
+ VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
+ assert(renderCallback0 != NULL);
+
+ printf("Adding Bitmap\n");
+ DDCOLORKEY ColorKey; // black
+ ColorKey.dwColorSpaceHighValue = RGB(0, 0, 0);
+ ColorKey.dwColorSpaceLowValue = RGB(0, 0, 0);
+ HBITMAP hbm = (HBITMAP)LoadImage(NULL,
+ (LPCTSTR)_T("renderStartImage.bmp"),
+ IMAGE_BITMAP, 0, 0, LR_LOADFROMFILE);
+ renderModule->SetBitmap(hbm, 0, &ColorKey, 0.0f, 0.0f, 0.3f,
+ 0.3f);
+
+ printf("Adding Text\n");
+ renderModule->SetText(1, (WebRtc_UWord8*) "WebRtc Render Demo App", 20,
+ RGB(255, 0, 0), RGB(0, 0, 0), 0.25f, 0.1f, 1.0f,
+ 1.0f);
+
+ printf("Start render\n");
+ error = renderModule->StartRender(streamId0);
+ assert(error == 0);
+
+ // Loop through an I420 file and render each frame
+ const WebRtc_UWord32 width = 352;
+ const WebRtc_UWord32 height = 288;
+ const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
+
+ VideoFrame videoFrame0;
+ videoFrame0.VerifyAndAllocate(numBytes);
+
+ const WebRtc_UWord32 renderDelayMs = 500;
+
+ for (int i=0; i<TEST_FRAME_NUM; i++) {
+ GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
+ videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame0.SetWidth(width);
+ videoFrame0.SetHeight(height);
+ videoFrame0.SetLength(numBytes);
+ renderCallback0->RenderFrame(streamId0, videoFrame0);
+ SLEEP(1000/TEST_FRAME_RATE);
+ }
+ videoFrame0.Free();
+ // Sleep and let all frames be rendered before closing
+ SLEEP(renderDelayMs*2);
+
+
+ // Shut down
+ printf("Closing...\n");
+ ColorKey.dwColorSpaceHighValue = RGB(0,0,0);
+ ColorKey.dwColorSpaceLowValue = RGB(0,0,0);
+ renderModule->SetBitmap(NULL, 0, &ColorKey, 0.0f, 0.0f, 0.0f, 0.0f);
+ renderModule->SetText(1, NULL, 20, RGB(255,255,255),
+ RGB(0,0,0), 0.0f, 0.0f, 0.0f, 0.0f);
+
+ error = renderModule->StopRender(streamId0);
+ assert(error == 0);
+
+ error = renderModule->DeleteIncomingRenderStream(streamId0);
+ assert(error == 0);
+#endif
+
+ return 0;
+}
+
+int TestMultipleStreams(VideoRender* renderModule) {
+ // Add settings for a stream to render
+ printf("Add stream 0\n");
+ const int streamId0 = 0;
+ VideoRenderCallback* renderCallback0 =
+ renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 0.45f, 0.45f);
+ assert(renderCallback0 != NULL);
+ printf("Add stream 1\n");
+ const int streamId1 = 1;
+ VideoRenderCallback* renderCallback1 =
+ renderModule->AddIncomingRenderStream(streamId1, 0, 0.55f, 0.0f, 1.0f, 0.45f);
+ assert(renderCallback1 != NULL);
+ printf("Add stream 2\n");
+ const int streamId2 = 2;
+ VideoRenderCallback* renderCallback2 =
+ renderModule->AddIncomingRenderStream(streamId2, 0, 0.0f, 0.55f, 0.45f, 1.0f);
+ assert(renderCallback2 != NULL);
+ printf("Add stream 3\n");
+ const int streamId3 = 3;
+ VideoRenderCallback* renderCallback3 =
+ renderModule->AddIncomingRenderStream(streamId3, 0, 0.55f, 0.55f, 1.0f, 1.0f);
+ assert(renderCallback3 != NULL);
+ assert(renderModule->StartRender(streamId0) == 0);
+ assert(renderModule->StartRender(streamId1) == 0);
+ assert(renderModule->StartRender(streamId2) == 0);
+ assert(renderModule->StartRender(streamId3) == 0);
+
+ // Loop through an I420 file and render each frame
+ const WebRtc_UWord32 width = 352;
+ const WebRtc_UWord32 height = 288;
+ const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
+
+ VideoFrame videoFrame0;
+ videoFrame0.VerifyAndAllocate(numBytes);
+ VideoFrame videoFrame1;
+ videoFrame1.VerifyAndAllocate(numBytes);
+ VideoFrame videoFrame2;
+ videoFrame2.VerifyAndAllocate(numBytes);
+ VideoFrame videoFrame3;
+ videoFrame3.VerifyAndAllocate(numBytes);
+
+ const WebRtc_UWord32 renderDelayMs = 500;
+
+ for (int i=0; i<TEST_FRAME_NUM; i++) {
+ GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
+ videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame0.SetWidth(width);
+ videoFrame0.SetHeight(height);
+ videoFrame0.SetLength(numBytes);
+ renderCallback0->RenderFrame(streamId0, videoFrame0);
+
+ GetTestVideoFrame(videoFrame1.Buffer(), width, height, TEST_STREAM1_START_COLOR);
+ videoFrame1.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame1.SetWidth(width);
+ videoFrame1.SetHeight(height);
+ videoFrame1.SetLength(numBytes);
+ renderCallback1->RenderFrame(streamId1, videoFrame1);
+
+ GetTestVideoFrame(videoFrame2.Buffer(), width, height, TEST_STREAM2_START_COLOR);
+ videoFrame2.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame2.SetWidth(width);
+ videoFrame2.SetHeight(height);
+ videoFrame2.SetLength(numBytes);
+ renderCallback2->RenderFrame(streamId2, videoFrame2);
+
+ GetTestVideoFrame(videoFrame3.Buffer(), width, height, TEST_STREAM3_START_COLOR);
+ videoFrame3.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
+ videoFrame3.SetWidth(width);
+ videoFrame3.SetHeight(height);
+ videoFrame3.SetLength(numBytes);
+ renderCallback3->RenderFrame(streamId3, videoFrame3);
+
+ SLEEP(1000/TEST_FRAME_RATE);
+ }
+
+ videoFrame0.Free();
+ videoFrame1.Free();
+ videoFrame2.Free();
+ videoFrame3.Free();
+
+ // Shut down
+ printf("Closing...\n");
+ assert(renderModule->StopRender(streamId0) == 0);
+ assert(renderModule->DeleteIncomingRenderStream(streamId0) == 0);
+ assert(renderModule->StopRender(streamId1) == 0);
+ assert(renderModule->DeleteIncomingRenderStream(streamId1) == 0);
+ assert(renderModule->StopRender(streamId2) == 0);
+ assert(renderModule->DeleteIncomingRenderStream(streamId2) == 0);
+ assert(renderModule->StopRender(streamId3) == 0);
+ assert(renderModule->DeleteIncomingRenderStream(streamId3) == 0);
+
+ return 0;
+}
+
+int TestExternalRender(VideoRender* renderModule) {
+ MyRenderCallback *externalRender = new MyRenderCallback();
+
+ const int streamId0 = 0;
+ VideoRenderCallback* renderCallback0 =
+ renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f,
+ 1.0f, 1.0f);
+ assert(renderCallback0 != NULL);
+ assert(renderModule->AddExternalRenderCallback(streamId0,
+ externalRender) == 0);
+
+ assert(renderModule->StartRender(streamId0) == 0);
+
+ const WebRtc_UWord32 width = 352;
+ const WebRtc_UWord32 height = 288;
+ const WebRtc_UWord32 numBytes = (WebRtc_UWord32) (1.5 * width * height);
+ VideoFrame videoFrame0;
+ videoFrame0.VerifyAndAllocate(numBytes);
+
+ const WebRtc_UWord32 renderDelayMs = 500;
+ int frameCount = TEST_FRAME_NUM;
+ for (int i=0; i<frameCount; i++) {
+ videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs);
+ videoFrame0.SetWidth(width);
+ videoFrame0.SetHeight(height);
+ renderCallback0->RenderFrame(streamId0, videoFrame0);
+ SLEEP(33);
+ }
+
+ // Sleep and let all frames be rendered before closing
+ SLEEP(2*renderDelayMs);
+ videoFrame0.Free();
+
+ assert(renderModule->StopRender(streamId0) == 0);
+ assert(renderModule->DeleteIncomingRenderStream(streamId0) == 0);
+ assert(frameCount == externalRender->_cnt);
+
+ delete externalRender;
+ externalRender = NULL;
+
+ return 0;
+}
+
+void RunVideoRenderTests(void* window, VideoRenderType windowType) {
+#ifndef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+ windowType = kRenderExternal;
+#endif
+
+ int myId = 12345;
+
+ // Create the render module
+ printf("Create render module\n");
+ VideoRender* renderModule = NULL;
+ renderModule = VideoRender::CreateVideoRender(myId,
+ window,
+ false,
+ windowType);
+ assert(renderModule != NULL);
+
+
+ // ##### Test single stream rendering ####
+ printf("#### TestSingleStream ####\n");
+ if (TestSingleStream(renderModule) != 0) {
+ printf ("TestSingleStream failed\n");
+ }
+
+ // ##### Test fullscreen rendering ####
+ printf("#### TestFullscreenStream ####\n");
+ if (TestFullscreenStream(renderModule, window, windowType) != 0) {
+ printf ("TestFullscreenStream failed\n");
+ }
+
+ // ##### Test bitmap and text ####
+ printf("#### TestBitmapText ####\n");
+ if (TestBitmapText(renderModule) != 0) {
+ printf ("TestBitmapText failed\n");
+ }
+
+ // ##### Test multiple streams ####
+ printf("#### TestMultipleStreams ####\n");
+ if (TestMultipleStreams(renderModule) != 0) {
+ printf ("TestMultipleStreams failed\n");
+ }
+
+ // ##### Test multiple streams ####
+ printf("#### TestExternalRender ####\n");
+ if (TestExternalRender(renderModule) != 0) {
+ printf ("TestExternalRender failed\n");
+ }
+
+ delete renderModule;
+ renderModule = NULL;
+
+ printf("VideoRender unit tests passed.\n");
+}
+
+// Note: The Mac main is implemented in testApi_mac.mm.
+#if defined(_WIN32)
+int _tmain(int argc, _TCHAR* argv[])
+#elif defined(WEBRTC_LINUX)
+int main(int argc, char* argv[])
+#endif
+#if !defined(WEBRTC_MAC)
+{
+ // Create a window for testing.
+ void* window = NULL;
+#if defined (_WIN32)
+ HWND testHwnd;
+ WebRtcCreateWindow(testHwnd, 0, 352, 288);
+ window = (void*)testHwnd;
+ VideoRenderType windowType = kRenderWindows;
+#elif defined(WEBRTC_LINUX)
+ Window testWindow;
+ Display* display;
+ WebRtcCreateWindow(&testWindow, &display, 0, 352, 288);
+ VideoRenderType windowType = kRenderX11;
+ window = (void*)testWindow;
+#endif // WEBRTC_LINUX
+
+ RunVideoRenderTests(window, windowType);
+ return 0;
+}
+#endif // !WEBRTC_MAC
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI.h b/trunk/src/modules/video_render/main/test/testAPI/testAPI.h
new file mode 100644
index 0000000..e0e0631
--- /dev/null
+++ b/trunk/src/modules/video_render/main/test/testAPI/testAPI.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
+#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
+
+#include "video_render_defines.h"
+
+void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType);
+
+#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm b/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm
new file mode 100644
index 0000000..2f836ab
--- /dev/null
+++ b/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testAPI.h"
+
+#include <iostream>
+
+#import <Foundation/Foundation.h>
+#import <Cocoa/Cocoa.h>
+#import <AppKit/AppKit.h>
+#import <QTKit/QTKit.h>
+#include <sys/time.h>
+
+#include "common_types.h"
+#import "modules/video_render/main/source/mac/cocoa_render_view.h"
+#include "module_common_types.h"
+#include "process_thread.h"
+#include "tick_util.h"
+#include "trace.h"
+#include "video_render_defines.h"
+#include "video_render.h"
+
+using namespace webrtc;
+
+int WebRtcCreateWindow(CocoaRenderView*& cocoaRenderer, int winNum, int width, int height)
+{
+ // In Cocoa, rendering is not done directly to a window like in Windows and Linux.
+ // It is rendererd to a Subclass of NSOpenGLView
+
+ // create cocoa container window
+ NSRect outWindowFrame = NSMakeRect(200, 800, width + 20, height + 20);
+ NSWindow* outWindow = [[NSWindow alloc] initWithContentRect:outWindowFrame
+ styleMask:NSTitledWindowMask
+ backing:NSBackingStoreBuffered
+ defer:NO];
+ [outWindow orderOut:nil];
+ [outWindow setTitle:@"Cocoa Renderer"];
+ [outWindow setBackgroundColor:[NSColor blueColor]];
+
+ // create renderer and attach to window
+ NSRect cocoaRendererFrame = NSMakeRect(10, 10, width, height);
+ cocoaRenderer = [[CocoaRenderView alloc] initWithFrame:cocoaRendererFrame];
+ [[outWindow contentView] addSubview:(NSView*)cocoaRenderer];
+
+ [outWindow makeKeyAndOrderFront:NSApp];
+
+ return 0;
+}
+
+int main (int argc, const char * argv[]) {
+ NSAutoreleasePool * pool = [[NSAutoreleasePool alloc] init];
+ [NSApplication sharedApplication];
+
+ CocoaRenderView* testWindow;
+ WebRtcCreateWindow(testWindow, 0, 352, 288);
+ VideoRenderType windowType = kRenderCocoa;
+ void* window = (void*)testWindow;
+
+ RunVideoRenderTests(window, windowType);
+
+ [pool release];
+}
diff --git a/trunk/src/system_wrappers/OWNERS b/trunk/src/system_wrappers/OWNERS
new file mode 100644
index 0000000..4091a93
--- /dev/null
+++ b/trunk/src/system_wrappers/OWNERS
@@ -0,0 +1,7 @@
+henrike@webrtc.org
+pwestin@webrtc.org
+perkj@webrtc.org
+henrika@webrtc.org
+henrikg@webrtc.org
+mflodman@webrtc.org
+niklas.enbom@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/system_wrappers/interface/aligned_malloc.h b/trunk/src/system_wrappers/interface/aligned_malloc.h
new file mode 100644
index 0000000..c229435
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/aligned_malloc.h
@@ -0,0 +1,25 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
+
+#include <stddef.h>
+
+namespace webrtc
+{
+ void* AlignedMalloc(
+ size_t size,
+ size_t alignment);
+ void AlignedFree(
+ void* memBlock);
+}
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
diff --git a/trunk/src/system_wrappers/interface/atomic32_wrapper.h b/trunk/src/system_wrappers/interface/atomic32_wrapper.h
new file mode 100644
index 0000000..40862fb
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/atomic32_wrapper.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Atomic system independant 32-bit integer.
+// Note: uses full memory barriers.
+// Note: assumes 32-bit (or higher) system
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+class Atomic32Impl;
+class Atomic32Wrapper
+{
+public:
+ Atomic32Wrapper(WebRtc_Word32 initialValue = 0);
+ ~Atomic32Wrapper();
+
+ // Prefix operator!
+ WebRtc_Word32 operator++();
+ WebRtc_Word32 operator--();
+
+ Atomic32Wrapper& operator=(const Atomic32Wrapper& rhs);
+ Atomic32Wrapper& operator=(WebRtc_Word32 rhs);
+
+ WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
+ WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
+
+ // Sets the value atomically to newValue if the value equals compare value.
+ // The function returns true if the exchange happened.
+ bool CompareExchange(WebRtc_Word32 newValue, WebRtc_Word32 compareValue);
+ WebRtc_Word32 Value() const;
+private:
+ // Disable the + and - operator since it's unclear what these operations
+ // should do.
+ Atomic32Wrapper operator+(const Atomic32Wrapper& rhs);
+ Atomic32Wrapper operator-(const Atomic32Wrapper& rhs);
+
+ WebRtc_Word32& operator++(int);
+ WebRtc_Word32& operator--(int);
+
+ // Cheshire cat to hide the implementation (faster than
+ // using virtual functions)
+ Atomic32Impl& _impl;
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/condition_variable_wrapper.h b/trunk/src/system_wrappers/interface/condition_variable_wrapper.h
new file mode 100644
index 0000000..c040fbf
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/condition_variable_wrapper.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class ConditionVariableWrapper
+{
+public:
+ // Factory method, constructor disabled.
+ static ConditionVariableWrapper* CreateConditionVariable();
+
+ virtual ~ConditionVariableWrapper() {}
+
+ // Calling thread will atomically release critSect and wait until next
+ // some other thread calls Wake() or WakeAll().
+ virtual void SleepCS(CriticalSectionWrapper& critSect) = 0;
+
+ // Same as above but with a timeout.
+ virtual bool SleepCS(CriticalSectionWrapper& critSect,
+ unsigned long maxTimeInMS) = 0;
+
+ // Wakes one thread calling SleepCS().
+ virtual void Wake() = 0;
+
+ // Wakes all threads calling SleepCS().
+ virtual void WakeAll() = 0;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/constructor_magic.h b/trunk/src/system_wrappers/interface/constructor_magic.h
new file mode 100644
index 0000000..b2aabc5
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/constructor_magic.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * WebRtc
+ * Copy from third_party/libjingle/source/talk/base/constructormagic.h
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
+
+#ifndef DISALLOW_ASSIGN
+#define DISALLOW_ASSIGN(TypeName) \
+ void operator=(const TypeName&)
+#endif
+
+#ifndef DISALLOW_COPY_AND_ASSIGN
+// A macro to disallow the evil copy constructor and operator= functions
+// This should be used in the private: declarations for a class
+#define DISALLOW_COPY_AND_ASSIGN(TypeName) \
+ TypeName(const TypeName&); \
+ DISALLOW_ASSIGN(TypeName)
+#endif
+
+#ifndef DISALLOW_EVIL_CONSTRUCTORS
+// Alternative, less-accurate legacy name.
+#define DISALLOW_EVIL_CONSTRUCTORS(TypeName) \
+ DISALLOW_COPY_AND_ASSIGN(TypeName)
+#endif
+
+#ifndef DISALLOW_IMPLICIT_CONSTRUCTORS
+// A macro to disallow all the implicit constructors, namely the
+// default constructor, copy constructor and operator= functions.
+//
+// This should be used in the private: declarations for a class
+// that wants to prevent anyone from instantiating it. This is
+// especially useful for classes containing only static methods.
+#define DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
+ TypeName(); \
+ DISALLOW_EVIL_CONSTRUCTORS(TypeName)
+#endif
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_features_wrapper.h b/trunk/src/system_wrappers/interface/cpu_features_wrapper.h
new file mode 100644
index 0000000..d949592
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/cpu_features_wrapper.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include <typedefs.h>
+
+// List of features in x86.
+typedef enum {
+ kSSE2,
+ kSSE3
+} CPUFeature;
+
+// List of features in ARM.
+enum {
+ kCPUFeatureARMv7 = (1 << 0),
+ kCPUFeatureVFPv3 = (1 << 1),
+ kCPUFeatureNEON = (1 << 2),
+ kCPUFeatureLDREXSTREX = (1 << 3)
+};
+
+typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
+// returns true if the CPU supports the feature.
+extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
+// No CPU feature is available => straight C path.
+extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
+
+// Return the features in an ARM device.
+// It detects the features in the hardware platform, and returns supported
+// values in the above enum definition as a bitmask.
+extern uint64_t WebRtc_GetCPUFeaturesARM(void);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+} // extern "C"
+#endif
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_info.h b/trunk/src/system_wrappers/interface/cpu_info.h
new file mode 100644
index 0000000..a6da29f
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/cpu_info.h
@@ -0,0 +1,27 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CpuInfo
+{
+public:
+ static WebRtc_UWord32 DetectNumberOfCores();
+
+private:
+ CpuInfo() {}
+ static WebRtc_UWord32 _numberOfCores;
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_wrapper.h b/trunk/src/system_wrappers/interface/cpu_wrapper.h
new file mode 100644
index 0000000..d938741
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/cpu_wrapper.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CpuWrapper
+{
+public:
+ static CpuWrapper* CreateCpu();
+ virtual ~CpuWrapper() {}
+
+ // Returns the average CPU usage for all processors. The CPU usage can be
+ // between and including 0 to 100 (%)
+ virtual WebRtc_Word32 CpuUsage() = 0;
+ virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* processName,
+ WebRtc_UWord32 length) = 0;
+ virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 dwProcessID) = 0;
+
+ // The CPU usage per core is returned in cpu_usage. The CPU can be between
+ // and including 0 to 100 (%)
+ // Note that the pointer passed as cpu_usage is redirected to a local member
+ // of the CPU wrapper.
+ // numCores is the number of cores in the cpu_usage array.
+ // The return value is -1 for failure or 0-100, indicating the average
+ // CPU usage across all cores.
+ // Note: on some OSs this class is initialized lazy. This means that it
+ // might not yet be possible to retrieve any CPU metrics. When this happens
+ // the return value will be zero (indicating that there is not a failure),
+ // numCores will be 0 and cpu_usage will be set to NULL (indicating that
+ // no metrics are available yet). Once the initialization is completed,
+ // which can take in the order of seconds, CPU metrics can be retrieved.
+ virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
+ WebRtc_UWord32*& cpu_usage) = 0;
+
+ virtual void Reset() = 0;
+ virtual void Stop() = 0;
+
+protected:
+ CpuWrapper() {}
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/critical_section_wrapper.h b/trunk/src/system_wrappers/interface/critical_section_wrapper.h
new file mode 100644
index 0000000..cfec9ae
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/critical_section_wrapper.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
+
+// If the critical section is heavily contended it may be beneficial to use
+// read/write locks instead.
+
+#include "common_types.h"
+
+namespace webrtc {
+class CriticalSectionWrapper
+{
+public:
+ // Factory method, constructor disabled
+ static CriticalSectionWrapper* CreateCriticalSection();
+
+ virtual ~CriticalSectionWrapper() {}
+
+ // Tries to grab lock, beginning of a critical section. Will wait for the
+ // lock to become available if the grab failed.
+ virtual void Enter() = 0;
+
+ // Returns a grabbed lock, end of critical section.
+ virtual void Leave() = 0;
+};
+
+// RAII extension of the critical section. Prevents Enter/Leave mismatches and
+// provides more compact critical section syntax.
+class CriticalSectionScoped
+{
+public:
+ // Deprecated, don't add more users of this constructor.
+ // TODO(mflodman) Remove this version of the constructor when no one is
+ // using it any longer.
+ explicit CriticalSectionScoped(CriticalSectionWrapper& critsec)
+ : _ptrCritSec(&critsec)
+ {
+ _ptrCritSec->Enter();
+ }
+
+ explicit CriticalSectionScoped(CriticalSectionWrapper* critsec)
+ : _ptrCritSec(critsec)
+ {
+ _ptrCritSec->Enter();
+ }
+
+ ~CriticalSectionScoped()
+ {
+ if (_ptrCritSec)
+ {
+ Leave();
+ }
+ }
+
+private:
+ void Leave()
+ {
+ _ptrCritSec->Leave();
+ _ptrCritSec = 0;
+ }
+
+ CriticalSectionWrapper* _ptrCritSec;
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/data_log.h b/trunk/src/system_wrappers/interface/data_log.h
new file mode 100644
index 0000000..6fc1d64
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/data_log.h
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This singleton can be used for logging data for offline processing. Data
+ * logged with it can conveniently be parsed and processed with e.g. Matlab.
+ *
+ * Following is an example of the log file format, starting with the header
+ * row at line 1, and the data rows following.
+ * col1,col2,col3,multi-value-col4[3],,,col5
+ * 123,10.2,-243,1,2,3,100
+ * 241,12.3,233,1,2,3,200
+ * 13,16.4,-13,1,2,3,300
+ *
+ * As can be seen in the example, a multi-value-column is specified with the
+ * name followed the number of elements it contains. This followed by
+ * number of elements - 1 empty columns.
+ *
+ * Without multi-value-columns this format can be natively by Matlab. With
+ * multi-value-columns a small Matlab script is needed, available at
+ * trunk/tools/matlab/parseLog.m.
+ *
+ * Table names and column names are case sensitive.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+
+#include <string>
+
+#include "data_log_impl.h"
+
+namespace webrtc {
+
+class DataLog {
+ public:
+ // Creates a log which uses a separate thread (referred to as the file
+ // writer thread) for writing log rows to file.
+ //
+ // Calls to this function after the log object has been created will only
+ // increment the reference counter.
+ static int CreateLog();
+
+ // Decrements the reference counter and deletes the log when the counter
+ // reaches 0. Should be called equal number of times as successful calls to
+ // CreateLog or memory leak will occur.
+ static void ReturnLog();
+
+ // Combines the string table_name and the integer table_id into a new string
+ // table_name + _ + table_id. The new string will be lower-case.
+ static std::string Combine(const std::string& table_name, int table_id);
+
+ // Adds a new table, with the name table_name, and creates the file, with the
+ // name table_name + ".txt", to which the table will be written.
+ // table_name is treated in a case sensitive way.
+ static int AddTable(const std::string& table_name);
+
+ // Adds a new column to a table. The column will be a multi-value-column
+ // if multi_value_length is greater than 1.
+ // table_name and column_name are treated in a case sensitive way.
+ static int AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length);
+
+ // Inserts a single value into a table with name table_name at the column with
+ // name column_name.
+ // Note that the ValueContainer makes use of the copy constructor,
+ // operator= and operator<< of the type T, and that the template type must
+ // implement a deep copy copy constructor and operator=.
+ // Copy constructor and operator= must not be disabled for the type T.
+ // table_name and column_name are treated in a case sensitive way.
+ template<class T>
+ static int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ T value) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->InsertCell(
+ table_name,
+ column_name,
+ new ValueContainer<T>(value));
+ }
+
+ // Inserts an array of values into a table with name table_name at the
+ // column specified by column_name, which must be a multi-value-column.
+ // Note that the MultiValueContainer makes use of the copy constructor,
+ // operator= and operator<< of the type T, and that the template type
+ // must implement a deep copy copy constructor and operator=.
+ // Copy constructor and operator= must not be disabled for the type T.
+ // table_name and column_name are treated in a case sensitive way.
+ template<class T>
+ static int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const T* array,
+ int length) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->InsertCell(
+ table_name,
+ column_name,
+ new MultiValueContainer<T>(array, length));
+ }
+
+ // For the table with name table_name: Writes the current row to file.
+ // Starts a new empty row.
+ // table_name is treated in a case-sensitive way.
+ static int NextRow(const std::string& table_name);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
diff --git a/trunk/src/system_wrappers/interface/data_log_c.h b/trunk/src/system_wrappers/interface/data_log_c.h
new file mode 100644
index 0000000..fffbb4f
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/data_log_c.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is a pure C wrapper of the DataLog class. The functions are directly
+ * mapped here except for InsertCell as C does not support templates.
+ * See data_log.h for a description of the functions.
+ */
+
+#ifndef SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
+#define SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
+
+#include <stddef.h> /* size_t */
+
+#include "typedefs.h" /* NOLINT(build/include) */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * All char* parameters in this file are expected to be null-terminated
+ * character sequences.
+ */
+int WebRtcDataLog_CreateLog();
+void WebRtcDataLog_ReturnLog();
+char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
+ const char* table_name, int table_id);
+int WebRtcDataLog_AddTable(const char* table_name);
+int WebRtcDataLog_AddColumn(const char* table_name, const char* column_name,
+ int multi_value_length);
+
+int WebRtcDataLog_InsertCell_int(const char* table_name,
+ const char* column_name,
+ int value);
+int WebRtcDataLog_InsertArray_int(const char* table_name,
+ const char* column_name,
+ const int* values,
+ int length);
+int WebRtcDataLog_InsertCell_float(const char* table_name,
+ const char* column_name,
+ float value);
+int WebRtcDataLog_InsertArray_float(const char* table_name,
+ const char* column_name,
+ const float* values,
+ int length);
+int WebRtcDataLog_InsertCell_double(const char* table_name,
+ const char* column_name,
+ double value);
+int WebRtcDataLog_InsertArray_double(const char* table_name,
+ const char* column_name,
+ const double* values,
+ int length);
+int WebRtcDataLog_InsertCell_int32(const char* table_name,
+ const char* column_name,
+ int32_t value);
+int WebRtcDataLog_InsertArray_int32(const char* table_name,
+ const char* column_name,
+ const int32_t* values,
+ int length);
+int WebRtcDataLog_InsertCell_uint32(const char* table_name,
+ const char* column_name,
+ uint32_t value);
+int WebRtcDataLog_InsertArray_uint32(const char* table_name,
+ const char* column_name,
+ const uint32_t* values,
+ int length);
+int WebRtcDataLog_InsertCell_int64(const char* table_name,
+ const char* column_name,
+ int64_t value);
+int WebRtcDataLog_InsertArray_int64(const char* table_name,
+ const char* column_name,
+ const int64_t* values,
+ int length);
+
+int WebRtcDataLog_NextRow(const char* table_name);
+
+#ifdef __cplusplus
+} /* end of extern "C" */
+#endif
+
+#endif /* SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_ */ /* NOLINT */
diff --git a/trunk/src/system_wrappers/interface/data_log_impl.h b/trunk/src/system_wrappers/interface/data_log_impl.h
new file mode 100644
index 0000000..cef4964
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/data_log_impl.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the helper classes for the DataLog APIs. See data_log.h
+ * for the APIs.
+ *
+ * These classes are helper classes used for logging data for offline
+ * processing. Data logged with these classes can conveniently be parsed and
+ * processed with e.g. Matlab.
+ */
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+
+#include <map>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "scoped_ptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class EventWrapper;
+class LogTable;
+class RWLockWrapper;
+class ThreadWrapper;
+
+// All container classes need to implement a ToString-function to be
+// writable to file. Enforce this via the Container interface.
+class Container {
+ public:
+ virtual ~Container() {}
+
+ virtual void ToString(std::string* container_string) const = 0;
+};
+
+template<class T>
+class ValueContainer : public Container {
+ public:
+ explicit ValueContainer(T data) : data_(data) {}
+
+ virtual void ToString(std::string* container_string) const {
+ *container_string = "";
+ std::stringstream ss;
+ ss << data_ << ",";
+ ss >> *container_string;
+ }
+
+ private:
+ T data_;
+};
+
+template<class T>
+class MultiValueContainer : public Container {
+ public:
+ MultiValueContainer(const T* data, int length)
+ : data_(data, data + length) {
+ }
+
+ virtual void ToString(std::string* container_string) const {
+ *container_string = "";
+ std::stringstream ss;
+ for (size_t i = 0; i < data_.size(); ++i)
+ ss << data_[i] << ",";
+ *container_string += ss.str();
+ }
+
+ private:
+ std::vector<T> data_;
+};
+
+class DataLogImpl {
+ public:
+ ~DataLogImpl();
+
+ // The implementation of the CreateLog() method declared in data_log.h.
+ // See data_log.h for a description.
+ static int CreateLog();
+
+ // The implementation of the StaticInstance() method declared in data_log.h.
+ // See data_log.h for a description.
+ static DataLogImpl* StaticInstance();
+
+ // The implementation of the ReturnLog() method declared in data_log.h. See
+ // data_log.h for a description.
+ static void ReturnLog();
+
+ // The implementation of the AddTable() method declared in data_log.h. See
+ // data_log.h for a description.
+ int AddTable(const std::string& table_name);
+
+ // The implementation of the AddColumn() method declared in data_log.h. See
+ // data_log.h for a description.
+ int AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length);
+
+ // Inserts a Container into a table with name table_name at the column
+ // with name column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const Container* value_container);
+
+ // The implementation of the NextRow() method declared in data_log.h. See
+ // data_log.h for a description.
+ int NextRow(const std::string& table_name);
+
+ private:
+ DataLogImpl();
+
+ // Initializes the DataLogImpl object, allocates and starts the
+ // thread file_writer_thread_.
+ int Init();
+
+ // Write all complete rows in every table to file.
+ // This function should only be called by the file_writer_thread_ if that
+ // thread is running to avoid race conditions.
+ void Flush();
+
+ // Run() is called by the thread file_writer_thread_.
+ static bool Run(void* obj);
+
+ // This function writes data to file. Note, it blocks if there is no data
+ // that should be written to file availble. Flush is the non-blocking
+ // version of this function.
+ void Process();
+
+ // Stops the continuous calling of Process().
+ void StopThread();
+
+ // Collection of tables indexed by the table name as std::string.
+ typedef std::map<std::string, LogTable*> TableMap;
+ typedef webrtc::scoped_ptr<CriticalSectionWrapper> CritSectScopedPtr;
+
+ static CritSectScopedPtr crit_sect_;
+ static DataLogImpl* instance_;
+ int counter_;
+ TableMap tables_;
+ EventWrapper* flush_event_;
+ ThreadWrapper* file_writer_thread_;
+ RWLockWrapper* tables_lock_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
diff --git a/trunk/src/system_wrappers/interface/event_wrapper.h b/trunk/src/system_wrappers/interface/event_wrapper.h
new file mode 100644
index 0000000..0c9a908
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/event_wrapper.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
+
+namespace webrtc {
+enum EventTypeWrapper
+{
+ kEventSignaled = 1,
+ kEventError = 2,
+ kEventTimeout = 3
+};
+
+#define WEBRTC_EVENT_10_SEC 10000
+#define WEBRTC_EVENT_INFINITE 0xffffffff
+
+class EventWrapper
+{
+public:
+ // Factory method. Constructor disabled.
+ static EventWrapper* Create();
+ virtual ~EventWrapper() {}
+
+ // Releases threads who are calling Wait() and has started waiting. Please
+ // note that a thread calling Wait() will not start waiting immediately.
+ // assumptions to the contrary is a very common source of issues in
+ // multithreaded programming.
+ // Set is sticky in the sense that it will release at least one thread
+ // either immediately or some time in the future.
+ virtual bool Set() = 0;
+
+ // Prevents future Wait() calls from finishing without a new Set() call.
+ virtual bool Reset() = 0;
+
+ // Puts the calling thread into a wait state. The thread may be released
+ // by a Set() call depending on if other threads are waiting and if so on
+ // timing. The thread that was released will call Reset() before leaving
+ // preventing more threads from being released. If multiple threads
+ // are waiting for the same Set(), only one (random) thread is guaranteed to
+ // be released. It is possible that multiple (random) threads are released
+ // Depending on timing.
+ virtual EventTypeWrapper Wait(unsigned long maxTime) = 0;
+
+ // Starts a timer that will call a non-sticky version of Set() either once
+ // or periodically. If the timer is periodic it ensures that there is no
+ // drift over time relative to the system clock.
+ virtual bool StartTimer(bool periodic, unsigned long time) = 0;
+
+ virtual bool StopTimer() = 0;
+
+ // Only implemented on Windows
+ // Returns 1 if a key has been pressed since last call to this function.
+ // -1 indicates failure
+ // 0 indicates no key has been pressed since last call
+ // TODO(hellner) this function does not seem to belong here
+ static int KeyPressed();
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/file_wrapper.h b/trunk/src/system_wrappers/interface/file_wrapper.h
new file mode 100644
index 0000000..4d17438
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/file_wrapper.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
+
+#include <stddef.h>
+
+#include "common_types.h"
+#include "typedefs.h"
+
+// Implementation of an InStream and OutStream that can read (exclusive) or
+// write from/to a file.
+
+namespace webrtc {
+
+class FileWrapper : public InStream, public OutStream
+{
+public:
+ static const size_t kMaxFileNameSize = 1024;
+
+ // Factory method. Constructor disabled.
+ static FileWrapper* Create();
+
+ // Returns true if a file has been opened.
+ virtual bool Open() const = 0;
+
+ // Opens a file in read or write mode, decided by the readOnly parameter.
+ virtual int OpenFile(const char* fileNameUTF8,
+ bool readOnly,
+ bool loop = false,
+ bool text = false) = 0;
+
+ virtual int CloseFile() = 0;
+
+ // Limits the file size to |bytes|. Writing will fail after the cap
+ // is hit. Pass zero to use an unlimited size.
+ virtual int SetMaxFileSize(size_t bytes) = 0;
+
+ // Flush any pending writes.
+ virtual int Flush() = 0;
+
+ // Returns the opened file's name in |fileNameUTF8|. Provide the size of
+ // the buffer in bytes in |size|. The name will be truncated if |size| is
+ // too small.
+ virtual int FileName(char* fileNameUTF8,
+ size_t size) const = 0;
+
+ // Write |format| to the opened file. Arguments are taken in the same manner
+ // as printf. That is, supply a format string containing text and
+ // specifiers. Returns the number of characters written or -1 on error.
+ virtual int WriteText(const char* format, ...) = 0;
+
+ // Inherited from Instream.
+ // Reads |length| bytes from file to |buf|. Returns the number of bytes read
+ // or -1 on error.
+ virtual int Read(void* buf, int length) = 0;
+
+ // Inherited from OutStream.
+ // Writes |length| bytes from |buf| to file. The actual writing may happen
+ // some time later. Call Flush() to force a write.
+ virtual bool Write(const void *buf, int length) = 0;
+
+ // Inherited from both Instream and OutStream.
+ // Rewinds the file to the start. Only available when OpenFile() has been
+ // called with |loop| == true or |readOnly| == true.
+ virtual int Rewind() = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h b/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h
new file mode 100644
index 0000000..d85c724
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h
@@ -0,0 +1,35 @@
+// Copyright (c) 2006-2008 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file under third_party_mods/chromium directory of
+// source tree or at
+// http://src.chromium.org/viewvc/chrome/trunk/src/LICENSE
+
+// Various inline functions and macros to fix compilation of 32 bit target
+// on MSVC with /Wp64 flag enabled.
+
+// The original code can be found here:
+// http://src.chromium.org/svn/trunk/src/base/fix_wp64.h
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
+
+#include <windows.h>
+
+// Platform SDK fixes when building with /Wp64 for a 32 bits target.
+#if !defined(_WIN64) && defined(_Wp64)
+
+#ifdef InterlockedExchangePointer
+#undef InterlockedExchangePointer
+// The problem is that the macro provided for InterlockedExchangePointer() is
+// doing a (LONG) C-style cast that triggers invariably the warning C4312 when
+// building on 32 bits.
+inline void* InterlockedExchangePointer(void* volatile* target, void* value) {
+ return reinterpret_cast<void*>(static_cast<LONG_PTR>(InterlockedExchange(
+ reinterpret_cast<volatile LONG*>(target),
+ static_cast<LONG>(reinterpret_cast<LONG_PTR>(value)))));
+}
+#endif // #ifdef InterlockedExchangePointer
+
+#endif // #if !defined(_WIN64) && defined(_Wp64)
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/interface/list_wrapper.h b/trunk/src/system_wrappers/interface/list_wrapper.h
new file mode 100644
index 0000000..3608ada
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/list_wrapper.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
+
+#include "constructor_magic.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class ListItem
+{
+friend class ListWrapper;
+
+public:
+ ListItem(const void* ptr);
+ ListItem(const unsigned int item);
+ virtual ~ListItem();
+ void* GetItem() const;
+ unsigned int GetUnsignedItem() const;
+
+protected:
+ ListItem* next_;
+ ListItem* prev_;
+
+private:
+ const void* item_ptr_;
+ const unsigned int item_;
+};
+
+class ListWrapper
+{
+public:
+ ListWrapper();
+ virtual ~ListWrapper();
+
+ // Returns the number of elements stored in the list.
+ unsigned int GetSize() const;
+
+ // Puts a pointer to anything last in the list.
+ int PushBack(const void* ptr);
+ // Puts a pointer to anything first in the list.
+ int PushFront(const void* ptr);
+
+ // Puts a copy of the specified integer last in the list.
+ int PushBack(const unsigned int item_id);
+ // Puts a copy of the specified integer first in the list.
+ int PushFront(const unsigned int item_id);
+
+ // Pops the first ListItem from the list
+ int PopFront();
+
+ // Pops the last ListItem from the list
+ int PopBack();
+
+ // Returns true if the list is empty
+ bool Empty() const;
+
+ // Returns a pointer to the first ListItem in the list.
+ ListItem* First() const;
+
+ // Returns a pointer to the last ListItem in the list.
+ ListItem* Last() const;
+
+ // Returns a pointer to the ListItem stored after item in the list.
+ ListItem* Next(ListItem* item) const;
+
+ // Returns a pointer to the ListItem stored before item in the list.
+ ListItem* Previous(ListItem* item) const;
+
+ // Removes item from the list.
+ int Erase(ListItem* item);
+
+ // Insert list item after existing_previous_item. Please note that new_item
+ // must be created using new ListItem(). The map will take ownership of
+ // new_item following a successfull insert. If insert fails new_item will
+ // not be released by the List
+ int Insert(ListItem* existing_previous_item,
+ ListItem* new_item);
+
+ // Insert list item before existing_next_item. Please note that new_item
+ // must be created using new ListItem(). The map will take ownership of
+ // new_item following a successfull insert. If insert fails new_item will
+ // not be released by the List
+ int InsertBefore(ListItem* existing_next_item,
+ ListItem* new_item);
+
+private:
+ void PushBackImpl(ListItem* item);
+ void PushFrontImpl(ListItem* item);
+
+ CriticalSectionWrapper* critical_section_;
+ ListItem* first_;
+ ListItem* last_;
+ unsigned int size_;
+};
+} //namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/map_wrapper.h b/trunk/src/system_wrappers/interface/map_wrapper.h
new file mode 100644
index 0000000..7d4e733
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/map_wrapper.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
+
+#include <map>
+
+#include "constructor_magic.h"
+
+namespace webrtc {
+class MapItem
+{
+friend class MapWrapper;
+
+public:
+ MapItem(int id, void* ptr);
+ virtual ~MapItem();
+ void* GetItem();
+ int GetId();
+ unsigned int GetUnsignedId();
+ void SetItem(void* ptr);
+
+private:
+ int item_id_;
+ void* item_pointer_;
+};
+
+class MapWrapper
+{
+public:
+ MapWrapper();
+ ~MapWrapper();
+
+ // Puts a pointer to anything in the map and associates it with id. Note, id
+ // needs to be unique for all items in the map.
+ int Insert(int id, void* ptr);
+
+ // Removes item from map.
+ int Erase(MapItem* item);
+
+ // Finds item with associated with id and removes it from the map.
+ int Erase(int id);
+
+ // Returns the number of elements stored in the map.
+ int Size() const;
+
+ // Returns a pointer to the first MapItem in the map.
+ MapItem* First() const;
+
+ // Returns a pointer to the last MapItem in the map.
+ MapItem* Last() const;
+
+ // Returns a pointer to the MapItem stored after item in the map.
+ MapItem* Next(MapItem* item) const;
+
+ // Returns a pointer to the MapItem stored before item in the map.
+ MapItem* Previous(MapItem* item) const;
+
+ // Returns a pointer to the MapItem associated with id from the map.
+ MapItem* Find(int id) const;
+
+private:
+ std::map<int, MapItem*> map_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/ref_count.h b/trunk/src/system_wrappers/interface/ref_count.h
new file mode 100644
index 0000000..f90b0b3
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/ref_count.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+#define SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+
+#include "system_wrappers/interface/atomic32_wrapper.h"
+
+namespace webrtc {
+
+// This class can be used for instantiating
+// reference counted objects.
+// int32_t AddRef() and int32_t Release().
+// Usage:
+// RefCountImpl<T>* implementation = new RefCountImpl<T>(p);
+//
+// Example:
+// class MyInterface {
+// public:
+// virtual void DoSomething() = 0;
+// virtual int32_t AddRef() = 0;
+// virtual int32_t Release() = 0:
+// private:
+// virtual ~MyInterface(){};
+// }
+// class MyImplementation : public MyInterface {
+// public:
+// virtual DoSomething() { printf("hello"); };
+// };
+// MyImplementation* CreateMyImplementation() {
+// RefCountImpl<MyImplementation>* implementation =
+// new RefCountImpl<MyImplementation>();
+// return implementation;
+// }
+
+template <class T>
+class RefCountImpl : public T {
+ public:
+ RefCountImpl() : ref_count_(0) {}
+
+ template<typename P>
+ explicit RefCountImpl(P p) : T(p), ref_count_(0) {}
+
+ template<typename P1, typename P2>
+ RefCountImpl(P1 p1, P2 p2) : T(p1, p2), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3>
+ RefCountImpl(P1 p1, P2 p2, P3 p3) : T(p1, p2, p3), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3, typename P4>
+ RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4) : T(p1, p2, p3, p4), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3, typename P4, typename P5>
+ RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5)
+ : T(p1, p2, p3, p4, p5), ref_count_(0) {}
+
+ virtual int32_t AddRef() {
+ return ++ref_count_;
+ }
+
+ virtual int32_t Release() {
+ int32_t ref_count;
+ ref_count = --ref_count_;
+ if (ref_count == 0)
+ delete this;
+ return ref_count;
+ }
+
+ protected:
+ Atomic32Wrapper ref_count_;
+};
+
+} // namespace webrtc
+
+#endif // SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
diff --git a/trunk/src/system_wrappers/interface/rw_lock_wrapper.h b/trunk/src/system_wrappers/interface/rw_lock_wrapper.h
new file mode 100644
index 0000000..f0842ac
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/rw_lock_wrapper.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
+
+// Note, Windows pre-Vista version of RW locks are not supported nativly. For
+// these OSs regular critical sections have been used to approximate RW lock
+// functionality and will therefore have worse performance.
+
+namespace webrtc {
+class RWLockWrapper
+{
+public:
+ static RWLockWrapper* CreateRWLock();
+ virtual ~RWLockWrapper();
+
+ virtual void AcquireLockExclusive() = 0;
+ virtual void ReleaseLockExclusive() = 0;
+
+ virtual void AcquireLockShared() = 0;
+ virtual void ReleaseLockShared() = 0;
+
+protected:
+ virtual int Init() = 0;
+};
+
+// RAII extensions of the RW lock. Prevents Acquire/Release missmatches and
+// provides more compact locking syntax.
+class ReadLockScoped
+{
+public:
+ ReadLockScoped(RWLockWrapper& rwLock)
+ :
+ _rwLock(rwLock)
+ {
+ _rwLock.AcquireLockShared();
+ }
+
+ ~ReadLockScoped()
+ {
+ _rwLock.ReleaseLockShared();
+ }
+
+private:
+ RWLockWrapper& _rwLock;
+};
+
+class WriteLockScoped
+{
+public:
+ WriteLockScoped(RWLockWrapper& rwLock)
+ :
+ _rwLock(rwLock)
+ {
+ _rwLock.AcquireLockExclusive();
+ }
+
+ ~WriteLockScoped()
+ {
+ _rwLock.ReleaseLockExclusive();
+ }
+
+private:
+ RWLockWrapper& _rwLock;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/scoped_ptr.h b/trunk/src/system_wrappers/interface/scoped_ptr.h
new file mode 100644
index 0000000..74b6ad3
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/scoped_ptr.h
@@ -0,0 +1,258 @@
+// (C) Copyright Greg Colvin and Beman Dawes 1998, 1999.
+// Copyright (c) 2001, 2002 Peter Dimov
+//
+// Permission to copy, use, modify, sell and distribute this software
+// is granted provided this copyright notice appears in all copies.
+// This software is provided "as is" without express or implied
+// warranty, and with no claim as to its suitability for any purpose.
+//
+// See http://www.boost.org/libs/smart_ptr/scoped_ptr.htm for documentation.
+//
+
+// scoped_ptr mimics a built-in pointer except that it guarantees deletion
+// of the object pointed to, either on destruction of the scoped_ptr or via
+// an explicit reset(). scoped_ptr is a simple solution for simple needs;
+// use shared_ptr or std::auto_ptr if your needs are more complex.
+
+// scoped_ptr_malloc added in by Google. When one of
+// these goes out of scope, instead of doing a delete or delete[], it
+// calls free(). scoped_ptr_malloc<char> is likely to see much more
+// use than any other specializations.
+
+// release() added in by Google. Use this to conditionally
+// transfer ownership of a heap-allocated object to the caller, usually on
+// method success.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
+
+#include <assert.h> // for assert
+#include <stdlib.h> // for free() decl
+
+#include <cstddef> // for std::ptrdiff_t
+
+#ifdef _WIN32
+namespace std { using ::ptrdiff_t; };
+#endif // _WIN32
+
+namespace webrtc {
+
+template <typename T>
+class scoped_ptr {
+ private:
+
+ T* ptr;
+
+ scoped_ptr(scoped_ptr const &);
+ scoped_ptr & operator=(scoped_ptr const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_ptr(T* p = NULL): ptr(p) {}
+
+ ~scoped_ptr() {
+ typedef char type_must_be_complete[sizeof(T)];
+ delete ptr;
+ }
+
+ void reset(T* p = NULL) {
+ typedef char type_must_be_complete[sizeof(T)];
+
+ if (ptr != p) {
+ T* obj = ptr;
+ ptr = p;
+ // Delete last, in case obj destructor indirectly results in ~scoped_ptr
+ delete obj;
+ }
+ }
+
+ T& operator*() const {
+ assert(ptr != NULL);
+ return *ptr;
+ }
+
+ T* operator->() const {
+ assert(ptr != NULL);
+ return ptr;
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_ptr & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = NULL;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ delete ptr;
+ ptr = NULL;
+ }
+ return &ptr;
+ }
+
+ T** use() {
+ return &ptr;
+ }
+};
+
+template<typename T> inline
+void swap(scoped_ptr<T>& a, scoped_ptr<T>& b) {
+ a.swap(b);
+}
+
+
+
+
+// scoped_array extends scoped_ptr to arrays. Deletion of the array pointed to
+// is guaranteed, either on destruction of the scoped_array or via an explicit
+// reset(). Use shared_array or std::vector if your needs are more complex.
+
+template<typename T>
+class scoped_array {
+ private:
+
+ T* ptr;
+
+ scoped_array(scoped_array const &);
+ scoped_array & operator=(scoped_array const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_array(T* p = NULL) : ptr(p) {}
+
+ ~scoped_array() {
+ typedef char type_must_be_complete[sizeof(T)];
+ delete[] ptr;
+ }
+
+ void reset(T* p = NULL) {
+ typedef char type_must_be_complete[sizeof(T)];
+
+ if (ptr != p) {
+ T* arr = ptr;
+ ptr = p;
+ // Delete last, in case arr destructor indirectly results in ~scoped_array
+ delete [] arr;
+ }
+ }
+
+ T& operator[](std::ptrdiff_t i) const {
+ assert(ptr != NULL);
+ assert(i >= 0);
+ return ptr[i];
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_array & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = NULL;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ delete [] ptr;
+ ptr = NULL;
+ }
+ return &ptr;
+ }
+};
+
+template<class T> inline
+void swap(scoped_array<T>& a, scoped_array<T>& b) {
+ a.swap(b);
+}
+
+// scoped_ptr_malloc<> is similar to scoped_ptr<>, but it accepts a
+// second template argument, the function used to free the object.
+
+template<typename T, void (*FF)(void*) = free> class scoped_ptr_malloc {
+ private:
+
+ T* ptr;
+
+ scoped_ptr_malloc(scoped_ptr_malloc const &);
+ scoped_ptr_malloc & operator=(scoped_ptr_malloc const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_ptr_malloc(T* p = 0): ptr(p) {}
+
+ ~scoped_ptr_malloc() {
+ FF(static_cast<void*>(ptr));
+ }
+
+ void reset(T* p = 0) {
+ if (ptr != p) {
+ FF(static_cast<void*>(ptr));
+ ptr = p;
+ }
+ }
+
+ T& operator*() const {
+ assert(ptr != 0);
+ return *ptr;
+ }
+
+ T* operator->() const {
+ assert(ptr != 0);
+ return ptr;
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_ptr_malloc & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = 0;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ FF(static_cast<void*>(ptr));
+ ptr = 0;
+ }
+ return &ptr;
+ }
+};
+
+template<typename T, void (*FF)(void*)> inline
+void swap(scoped_ptr_malloc<T,FF>& a, scoped_ptr_malloc<T,FF>& b) {
+ a.swap(b);
+}
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
diff --git a/trunk/src/system_wrappers/interface/scoped_refptr.h b/trunk/src/system_wrappers/interface/scoped_refptr.h
new file mode 100644
index 0000000..0df15be
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/scoped_refptr.h
@@ -0,0 +1,137 @@
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file under third_party_mods/chromium or at:
+// http://src.chromium.org/svn/trunk/src/LICENSE
+
+#ifndef SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
+#define SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
+
+namespace webrtc {
+
+// Extracted from Chromium's src/base/memory/ref_counted.h.
+
+//
+// A smart pointer class for reference counted objects. Use this class instead
+// of calling AddRef and Release manually on a reference counted object to
+// avoid common memory leaks caused by forgetting to Release an object
+// reference. Sample usage:
+//
+// class MyFoo : public RefCounted<MyFoo> {
+// ...
+// };
+//
+// void some_function() {
+// scoped_refptr<MyFoo> foo = new MyFoo();
+// foo->Method(param);
+// // |foo| is released when this function returns
+// }
+//
+// void some_other_function() {
+// scoped_refptr<MyFoo> foo = new MyFoo();
+// ...
+// foo = NULL; // explicitly releases |foo|
+// ...
+// if (foo)
+// foo->Method(param);
+// }
+//
+// The above examples show how scoped_refptr<T> acts like a pointer to T.
+// Given two scoped_refptr<T> classes, it is also possible to exchange
+// references between the two objects, like so:
+//
+// {
+// scoped_refptr<MyFoo> a = new MyFoo();
+// scoped_refptr<MyFoo> b;
+//
+// b.swap(a);
+// // now, |b| references the MyFoo object, and |a| references NULL.
+// }
+//
+// To make both |a| and |b| in the above example reference the same MyFoo
+// object, simply use the assignment operator:
+//
+// {
+// scoped_refptr<MyFoo> a = new MyFoo();
+// scoped_refptr<MyFoo> b;
+//
+// b = a;
+// // now, |a| and |b| each own a reference to the same MyFoo object.
+// }
+//
+template <class T>
+class scoped_refptr {
+ public:
+ scoped_refptr() : ptr_(NULL) {
+ }
+
+ scoped_refptr(T* p) : ptr_(p) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ template <typename U>
+ scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ ~scoped_refptr() {
+ if (ptr_)
+ ptr_->Release();
+ }
+
+ T* get() const { return ptr_; }
+ operator T*() const { return ptr_; }
+ T* operator->() const { return ptr_; }
+
+ // Release a pointer.
+ // The return value is the current pointer held by this object.
+ // If this object holds a NULL pointer, the return value is NULL.
+ // After this operation, this object will hold a NULL pointer,
+ // and will not own the object any more.
+ T* release() {
+ T* retVal = ptr_;
+ ptr_ = NULL;
+ return retVal;
+ }
+
+ scoped_refptr<T>& operator=(T* p) {
+ // AddRef first so that self assignment should work
+ if (p)
+ p->AddRef();
+ if (ptr_ )
+ ptr_->Release();
+ ptr_ = p;
+ return *this;
+ }
+
+ scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
+ return *this = r.ptr_;
+ }
+
+ template <typename U>
+ scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
+ return *this = r.get();
+ }
+
+ void swap(T** pp) {
+ T* p = ptr_;
+ ptr_ = *pp;
+ *pp = p;
+ }
+
+ void swap(scoped_refptr<T>& r) {
+ swap(&r.ptr_);
+ }
+
+ protected:
+ T* ptr_;
+};
+} // namespace webrtc
+
+#endif // SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
diff --git a/trunk/src/system_wrappers/interface/sort.h b/trunk/src/system_wrappers/interface/sort.h
new file mode 100644
index 0000000..fb25ecf
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/sort.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Generic unstable sorting routines.
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
+
+#include "typedefs.h"
+#include "common_types.h"
+
+namespace webrtc
+{
+ enum Type
+ {
+ TYPE_Word8,
+ TYPE_UWord8,
+ TYPE_Word16,
+ TYPE_UWord16,
+ TYPE_Word32,
+ TYPE_UWord32,
+ TYPE_Word64,
+ TYPE_UWord64,
+ TYPE_Float32,
+ TYPE_Float64
+ };
+ // Sorts intrinsic data types.
+ //
+ // data [in/out] A pointer to an array of intrinsic type.
+ // Upon return it will be sorted in ascending order.
+ // numOfElements The number of elements in the array.
+ // dataType Enum corresponding to the type of the array.
+ //
+ // returns 0 on success, -1 on failure.
+ WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type dataType);
+
+ // Sorts arbitrary data types. This requires an array of intrinsically typed
+ // key values which will be used to sort the data array. There must be a
+ // one-to-one correspondence between data elements and key elements, with
+ // corresponding elements sharing the same position in their respective
+ // arrays.
+ //
+ // data [in/out] A pointer to an array of arbitrary type.
+ // Upon return it will be sorted in ascending order.
+ // key [in] A pointer to an array of keys used to sort the
+ // data array.
+ // numOfElements The number of elements in the arrays.
+ // sizeOfElement The size, in bytes, of the data array.
+ // keyType Enum corresponding to the type of the key array.
+ //
+ // returns 0 on success, -1 on failure.
+ //
+ WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
+ WebRtc_UWord32 sizeOfElement, Type keyType);
+}
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
diff --git a/trunk/src/system_wrappers/interface/static_instance.h b/trunk/src/system_wrappers/interface/static_instance.h
new file mode 100644
index 0000000..8fe91cc
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/static_instance.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
+
+#include <assert.h>
+
+#include "critical_section_wrapper.h"
+#ifdef _WIN32
+#include "fix_interlocked_exchange_pointer_win.h"
+#endif
+
+namespace webrtc {
+
+enum CountOperation {
+ kRelease,
+ kAddRef,
+ kAddRefNoCreate
+};
+enum CreateOperation {
+ kInstanceExists,
+ kCreate,
+ kDestroy
+};
+
+template <class T>
+// Construct On First Use idiom. Avoids
+// "static initialization order fiasco".
+static T* GetStaticInstance(CountOperation count_operation) {
+ // TODO (hellner): use atomic wrapper instead.
+ static volatile long instance_count = 0;
+ static T* volatile instance = NULL;
+ CreateOperation state = kInstanceExists;
+#ifndef _WIN32
+ // This memory is staticly allocated once. The application does not try to
+ // free this memory. This approach is taken to avoid issues with
+ // destruction order for statically allocated memory. The memory will be
+ // reclaimed by the OS and memory leak tools will not recognize memory
+ // reachable from statics leaked so no noise is added by doing this.
+ static CriticalSectionWrapper* crit_sect(
+ CriticalSectionWrapper::CreateCriticalSection());
+ CriticalSectionScoped lock(crit_sect);
+
+ if (count_operation ==
+ kAddRefNoCreate && instance_count == 0) {
+ return NULL;
+ }
+ if (count_operation ==
+ kAddRef ||
+ count_operation == kAddRefNoCreate) {
+ instance_count++;
+ if (instance_count == 1) {
+ state = kCreate;
+ }
+ } else {
+ instance_count--;
+ if (instance_count == 0) {
+ state = kDestroy;
+ }
+ }
+ if (state == kCreate) {
+ instance = T::CreateInstance();
+ } else if (state == kDestroy) {
+ T* old_instance = instance;
+ instance = NULL;
+ // The state will not change past this point. Release the critical
+ // section while deleting the object in case it would be blocking on
+ // access back to this object. (This is the case for the tracing class
+ // since the thread owned by the tracing class also traces).
+ // TODO(hellner): this is a bit out of place but here goes, de-couple
+ // thread implementation with trace implementation.
+ crit_sect->Leave();
+ if (old_instance) {
+ delete old_instance;
+ }
+ // Re-acquire the lock since the scoped critical section will release
+ // it.
+ crit_sect->Enter();
+ return NULL;
+ }
+#else // _WIN32
+ if (count_operation ==
+ kAddRefNoCreate && instance_count == 0) {
+ return NULL;
+ }
+ if (count_operation == kAddRefNoCreate) {
+ if (1 == InterlockedIncrement(&instance_count)) {
+ // The instance has been destroyed by some other thread. Rollback.
+ InterlockedDecrement(&instance_count);
+ assert(false);
+ return NULL;
+ }
+ // Sanity to catch corrupt state.
+ if (instance == NULL) {
+ assert(false);
+ InterlockedDecrement(&instance_count);
+ return NULL;
+ }
+ } else if (count_operation == kAddRef) {
+ if (instance_count == 0) {
+ state = kCreate;
+ } else {
+ if (1 == InterlockedIncrement(&instance_count)) {
+ // InterlockedDecrement because reference count should not be
+ // updated just yet (that's done when the instance is created).
+ InterlockedDecrement(&instance_count);
+ state = kCreate;
+ }
+ }
+ } else {
+ int newValue = InterlockedDecrement(&instance_count);
+ if (newValue == 0) {
+ state = kDestroy;
+ }
+ }
+
+ if (state == kCreate) {
+ // Create instance and let whichever thread finishes first assign its
+ // local copy to the global instance. All other threads reclaim their
+ // local copy.
+ T* new_instance = T::CreateInstance();
+ if (1 == InterlockedIncrement(&instance_count)) {
+ T* old_value = static_cast<T*> (InterlockedExchangePointer(
+ reinterpret_cast<void* volatile*>(&instance), new_instance));
+ assert(old_value == NULL);
+ assert(instance);
+ } else {
+ InterlockedDecrement(&instance_count);
+ if (new_instance) {
+ delete static_cast<T*>(new_instance);
+ }
+ }
+ } else if (state == kDestroy) {
+ T* old_value = static_cast<T*> (InterlockedExchangePointer(
+ reinterpret_cast<void* volatile*>(&instance), NULL));
+ if (old_value) {
+ delete static_cast<T*>(old_value);
+ }
+ return NULL;
+ }
+#endif // #ifndef _WIN32
+ return instance;
+}
+
+} // namspace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
diff --git a/trunk/src/system_wrappers/interface/thread_wrapper.h b/trunk/src/system_wrappers/interface/thread_wrapper.h
new file mode 100644
index 0000000..030ac8a
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/thread_wrapper.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// System independant wrapper for spawning threads
+// Note: the spawned thread will loop over the callback function until stopped.
+// Note: The callback function is expected to return every 2 seconds or more
+// often.
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
+
+#include "common_types.h"
+#include "typedefs.h"
+
+namespace webrtc {
+// Object that will be passed by the spawned thread when it enters the callback
+// function.
+#define ThreadObj void*
+
+// Callback function that the spawned thread will enter once spawned.
+// A return value of false is interpreted as that the function has no
+// more work to do and that the thread can be released.
+typedef bool(*ThreadRunFunction)(ThreadObj);
+
+enum ThreadPriority
+{
+ kLowPriority = 1,
+ kNormalPriority = 2,
+ kHighPriority = 3,
+ kHighestPriority = 4,
+ kRealtimePriority = 5
+};
+
+class ThreadWrapper
+{
+public:
+ enum {kThreadMaxNameLength = 64};
+
+ virtual ~ThreadWrapper() {};
+
+ // Factory method. Constructor disabled.
+ //
+ // func Pointer to a, by user, specified callback function.
+ // obj Object associated with the thread. Passed in the callback
+ // function.
+ // prio Thread priority. May require root/admin rights.
+ // threadName NULL terminated thread name, will be visable in the Windows
+ // debugger.
+ static ThreadWrapper* CreateThread(ThreadRunFunction func = 0,
+ ThreadObj obj= 0,
+ ThreadPriority prio = kNormalPriority,
+ const char* threadName = 0);
+
+ // Get the current thread's kernel thread ID.
+ static uint32_t GetThreadId();
+
+ // Non blocking termination of the spawned thread. Note that it is not safe
+ // to delete this class until the spawned thread has been reclaimed.
+ virtual void SetNotAlive() = 0;
+
+ // Spawns the thread. This will start the triggering of the callback
+ // function.
+ virtual bool Start(unsigned int& id) = 0;
+
+ // Sets the threads CPU affinity. CPUs are listed 0 - (number of CPUs - 1).
+ // The numbers in processorNumbers specify which CPUs are allowed to run the
+ // thread. processorNumbers should not contain any duplicates and elements
+ // should be lower than (number of CPUs - 1). amountOfProcessors should be
+ // equal to the number of processors listed in processorNumbers
+ virtual bool SetAffinity(const int* /*processorNumbers*/,
+ const unsigned int /*amountOfProcessors*/) {
+ return false;
+ }
+
+ // Stops the spawned thread and waits for it to be reclaimed with a timeout
+ // of two seconds. Will return false if the thread was not reclaimed.
+ // Multiple tries to Stop are allowed (e.g. to wait longer than 2 seconds).
+ // It's ok to call Stop() even if the spawned thread has been reclaimed.
+ virtual bool Stop() = 0;
+
+ // Stops the spawned thread dead in its tracks. Will likely result in a
+ // corrupt state. There should be an extremely good reason for even looking
+ // at this function. Can cause many problems deadlock being one of them.
+ virtual bool Shutdown() {return false;}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/tick_util.h b/trunk/src/system_wrappers/interface/tick_util.h
new file mode 100644
index 0000000..e78e53d
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/tick_util.h
@@ -0,0 +1,325 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// System independant wrapper for polling elapsed time in ms and us.
+// The implementation works in the tick domain which can be mapped over to the
+// time domain.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
+
+#if _WIN32
+#include <windows.h>
+#include <mmsystem.h>
+#elif WEBRTC_LINUX
+#include <ctime>
+#else
+#include <sys/time.h>
+#include <time.h>
+#endif
+
+#include "typedefs.h"
+
+namespace webrtc {
+class TickInterval;
+
+class TickTime
+{
+public:
+ // Current time in the tick domain.
+ static TickTime Now();
+
+ // Now in the time domain in ms.
+ static WebRtc_Word64 MillisecondTimestamp();
+
+ // Now in the time domain in us.
+ static WebRtc_Word64 MicrosecondTimestamp();
+
+ WebRtc_Word64 Ticks() const;
+
+ static WebRtc_Word64 MillisecondsToTicks(const WebRtc_Word64 ms);
+
+ static WebRtc_Word64 TicksToMilliseconds(const WebRtc_Word64 ticks);
+
+ // Returns a TickTime that is ticks later than the passed TickTime
+ friend TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks);
+ TickTime& operator+=(const WebRtc_Word64& rhs);
+
+
+ // Returns a TickInterval that is the difference in ticks beween rhs and lhs
+ friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
+private:
+ WebRtc_Word64 _ticks;
+};
+
+class TickInterval
+{
+public:
+ TickInterval();
+
+ WebRtc_Word64 Milliseconds() const;
+ WebRtc_Word64 Microseconds() const;
+
+ // Returns the sum of two TickIntervals as a TickInterval
+ friend TickInterval operator+(const TickInterval& lhs,
+ const TickInterval& rhs);
+ TickInterval& operator-=(const TickInterval& rhs);
+
+ // Returns a TickInterval corresponding to rhs - lhs
+ friend TickInterval operator-(const TickInterval& lhs,
+ const TickInterval& rhs);
+ TickInterval& operator+=(const TickInterval& rhs);
+
+ friend bool operator>(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator<=(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator<(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
+
+private:
+ TickInterval(WebRtc_Word64 interval);
+
+ friend class TickTime;
+ friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
+
+private:
+ WebRtc_Word64 _interval;
+};
+
+inline TickInterval operator+(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return TickInterval(lhs._interval + rhs._interval);
+}
+
+inline TickInterval operator-(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return TickInterval(lhs._interval - rhs._interval);
+}
+
+inline TickInterval operator-(const TickTime& lhs,const TickTime& rhs)
+{
+ return TickInterval(lhs._ticks - rhs._ticks);
+}
+
+inline TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks)
+{
+ TickTime time = lhs;
+ time._ticks += ticks;
+ return time;
+}
+inline bool operator>(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval > rhs._interval;
+}
+inline bool operator<=(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval <= rhs._interval;
+}
+inline bool operator<(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval <= rhs._interval;
+}
+inline bool operator>=(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval >= rhs._interval;
+}
+
+inline TickTime TickTime::Now()
+{
+ TickTime result;
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ // QueryPerformanceCounter returns the value from the TSC which is
+ // incremented at the CPU frequency. The algorithm used requires
+ // the CPU frequency to be constant. Technology like speed stepping
+ // which has variable CPU frequency will therefore yield unpredictable,
+ // incorrect time estimations.
+ LARGE_INTEGER qpcnt;
+ QueryPerformanceCounter(&qpcnt);
+ result._ticks = qpcnt.QuadPart;
+ #else
+ static volatile LONG lastTimeGetTime = 0;
+ static volatile WebRtc_Word64 numWrapTimeGetTime = 0;
+ volatile LONG* lastTimeGetTimePtr = &lastTimeGetTime;
+ DWORD now = timeGetTime();
+ // Atomically update the last gotten time
+ DWORD old = InterlockedExchange(lastTimeGetTimePtr, now);
+ if(now < old)
+ {
+ // If now is earlier than old, there may have been a race between
+ // threads.
+ // 0x0fffffff ~3.1 days, the code will not take that long to execute
+ // so it must have been a wrap around.
+ if(old > 0xf0000000 && now < 0x0fffffff)
+ {
+ numWrapTimeGetTime++;
+ }
+ }
+ result._ticks = now + (numWrapTimeGetTime<<32);
+ #endif
+#elif defined(WEBRTC_LINUX)
+ struct timespec ts;
+ #ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ clock_gettime(CLOCK_REALTIME, &ts);
+ #else
+ clock_gettime(CLOCK_MONOTONIC, &ts);
+ #endif
+ result._ticks = 1000000000LL * static_cast<WebRtc_Word64>(ts.tv_sec) + static_cast<WebRtc_Word64>(ts.tv_nsec);
+#else
+ struct timeval tv;
+ gettimeofday(&tv, NULL);
+ result._ticks = 1000000LL * static_cast<WebRtc_Word64>(tv.tv_sec) + static_cast<WebRtc_Word64>(tv.tv_usec);
+#endif
+ return result;
+}
+
+inline WebRtc_Word64 TickTime::MillisecondTimestamp()
+{
+ TickTime now = TickTime::Now();
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (now._ticks * 1000) / qpfreq.QuadPart;
+ #else
+ return now._ticks;
+ #endif
+#elif WEBRTC_LINUX
+ return now._ticks / 1000000LL;
+#else
+ return now._ticks / 1000LL;
+#endif
+}
+
+inline WebRtc_Word64 TickTime::MicrosecondTimestamp()
+{
+ TickTime now = TickTime::Now();
+
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (now._ticks * 1000) / (qpfreq.QuadPart/1000);
+ #else
+ return now._ticks *1000LL;
+ #endif
+#elif WEBRTC_LINUX
+ return now._ticks / 1000LL;
+#else
+ return now._ticks;
+#endif
+}
+
+inline WebRtc_Word64 TickTime::Ticks() const
+{
+ return _ticks;
+}
+
+inline WebRtc_Word64 TickTime::MillisecondsToTicks(const WebRtc_Word64 ms)
+{
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (qpfreq.QuadPart * ms) / 1000;
+ #else
+ return ms;
+ #endif
+#elif WEBRTC_LINUX
+ return ms * 1000000LL;
+#else
+ return ms * 1000LL;
+#endif
+}
+
+inline WebRtc_Word64 TickTime::TicksToMilliseconds(const WebRtc_Word64 ticks)
+{
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (ticks * 1000) / qpfreq.QuadPart;
+ #else
+ return ticks;
+ #endif
+#elif WEBRTC_LINUX
+ return ticks / 1000000LL;
+#else
+ return ticks / 1000LL;
+#endif
+}
+
+inline TickTime& TickTime::operator+=(const WebRtc_Word64& ticks)
+{
+ _ticks += ticks;
+ return *this;
+}
+
+inline TickInterval::TickInterval() : _interval(0)
+{
+}
+
+inline TickInterval::TickInterval(const WebRtc_Word64 interval)
+ : _interval(interval)
+{
+}
+
+inline WebRtc_Word64 TickInterval::Milliseconds() const
+{
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (_interval * 1000) / qpfreq.QuadPart;
+ #else
+ // _interval is in ms
+ return _interval;
+ #endif
+#elif WEBRTC_LINUX
+ // _interval is in ns
+ return _interval / 1000000;
+#else
+ // _interval is usecs
+ return _interval / 1000;
+#endif
+}
+
+inline WebRtc_Word64 TickInterval::Microseconds() const
+{
+#if _WIN32
+ #ifdef USE_QUERY_PERFORMANCE_COUNTER
+ LARGE_INTEGER qpfreq;
+ QueryPerformanceFrequency(&qpfreq);
+ return (_interval * 1000000) / qpfreq.QuadPart;
+ #else
+ // _interval is in ms
+ return _interval *1000LL;
+ #endif
+#elif WEBRTC_LINUX
+ // _interval is in ns
+ return _interval / 1000;
+#else
+ // _interval is usecs
+ return _interval;
+#endif
+}
+
+inline TickInterval& TickInterval::operator+=(const TickInterval& rhs)
+{
+ _interval += rhs._interval;
+ return *this;
+}
+
+inline TickInterval& TickInterval::operator-=(const TickInterval& rhs)
+{
+ _interval -= rhs._interval;
+ return *this;
+}
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
diff --git a/trunk/src/system_wrappers/interface/trace.h b/trunk/src/system_wrappers/interface/trace.h
new file mode 100644
index 0000000..0333e76
--- /dev/null
+++ b/trunk/src/system_wrappers/interface/trace.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// System independant wrapper for logging runtime information to file.
+// Note: All log messages will be written to the same trace file.
+// Note: If to many messages are written to file there will be a build up of
+// messages. Apply filtering to avoid that.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
+
+#include "common_types.h"
+#include "typedefs.h"
+
+#define WEBRTC_TRACE Trace::Add
+
+namespace webrtc {
+class Trace
+{
+public:
+
+ // Increments the reference count to the trace.
+ static void CreateTrace();
+ // Decrements the reference count to the trace.
+ static void ReturnTrace();
+ // Note: any instance that writes to the trace file should increment and
+ // decrement the reference count on construction and destruction
+ // respectively
+
+ // Specifies what type of messages should be written to the trace file. The
+ // filter parameter is a bitmask where each message type is enumerated by
+ // the TraceLevel enumerator. TODO (hellner) why is the
+ // TraceLevel enumerator not defined in this file?
+ static WebRtc_Word32 SetLevelFilter(const WebRtc_UWord32 filter);
+
+ // Returns what type of messages are written to the trace file.
+ static WebRtc_Word32 LevelFilter(WebRtc_UWord32& filter);
+
+ // Sets the file name. If addFileCounter is false the same file will be
+ // reused when it fills up. If it's true a new file with incremented name
+ // will be used.
+ static WebRtc_Word32 SetTraceFile(const char* fileName,
+ const bool addFileCounter = false);
+
+ // Returns the name of the file that the trace is currently writing to.
+ static WebRtc_Word32 TraceFile(char fileName[1024]);
+
+ // Registers callback to receive trace messages. TODO (hellner)
+ // why not use OutStream instead? Why is TraceCallback not defined in this
+ // file
+ static WebRtc_Word32 SetTraceCallback(TraceCallback* callback);
+
+ // Adds a trace message for writing to file. The message is put in a queue
+ // for writing to file whenever possible for performance reasons. I.e. there
+ // is a crash it is possible that the last, vital logs are not logged yet.
+ // level is the the type of message to log. If that type of messages is
+ // filtered it will not be written to file. module is an identifier for what
+ // part of the code the message is comming.
+ // id is an identifier that should be unique for that set of classes that
+ // are associated (e.g. all instances owned by an engine).
+ // msg and the elipsis are the same as e.g. sprintf.
+ // TODO (hellner) Why is TraceModule not defined in this file?
+ static void Add(const TraceLevel level,
+ const TraceModule module,
+ const WebRtc_Word32 id,
+ const char* msg, ...);
+
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
diff --git a/trunk/src/system_wrappers/source/Android.mk b/trunk/src/system_wrappers/source/Android.mk
new file mode 100644
index 0000000..aea9b32
--- /dev/null
+++ b/trunk/src/system_wrappers/source/Android.mk
@@ -0,0 +1,60 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_system_wrappers
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ map.cc \
+ sort.cc \
+ aligned_malloc.cc \
+ atomic32.cc \
+ condition_variable.cc \
+ cpu_no_op.cc \
+ cpu_features.cc \
+ cpu_features_arm.c \
+ cpu_info.cc \
+ critical_section.cc \
+ event.cc \
+ file_impl.cc \
+ list_no_stl.cc \
+ rw_lock.cc \
+ thread.cc \
+ trace_impl.cc \
+ condition_variable_posix.cc \
+ cpu_linux.cc \
+ critical_section_posix.cc \
+ event_posix.cc \
+ thread_posix.cc \
+ trace_posix.cc \
+ rw_lock_posix.cc
+
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/spreadsortlib
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/system_wrappers/source/aligned_malloc.cc b/trunk/src/system_wrappers/source/aligned_malloc.cc
new file mode 100644
index 0000000..bb10c6b
--- /dev/null
+++ b/trunk/src/system_wrappers/source/aligned_malloc.cc
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aligned_malloc.h"
+
+#include <assert.h>
+#include <memory.h>
+
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#if WEBRTC_MAC
+ #include <malloc/malloc.h>
+#else
+ #include <malloc.h>
+#endif
+
+#if _WIN32
+ #include <windows.h>
+#else
+ #include <stdint.h>
+#endif
+
+#include "typedefs.h"
+
+// Ok reference on memory alignment:
+// http://stackoverflow.com/questions/227897/solve-the-memory-alignment-in-c-interview-question-that-stumped-me
+
+namespace webrtc
+{
+// TODO (hellner) better to create just one memory block and
+// interpret the first sizeof(AlignedMemory) bytes as
+// an AlignedMemory struct.
+struct AlignedMemory
+{
+ void* alignedBuffer;
+ void* memoryPointer;
+};
+
+void* AlignedMalloc(size_t size, size_t alignment)
+{
+ if(alignment == 0)
+ {
+ // Don't allow alignment 0 since it's undefined.
+ return NULL;
+ }
+ // Make sure that the alignment is an integer power of two or fail.
+ if(alignment & (alignment - 1))
+ {
+ return NULL;
+ }
+
+ AlignedMemory* returnValue = new AlignedMemory();
+ if(returnValue == NULL)
+ {
+ return NULL;
+ }
+
+ // The memory is aligned towards the lowest address that so only
+ // alignment - 1 bytes needs to be allocated.
+ // A pointer to AlignedMemory must be stored so that it can be retreived for
+ // deletion, ergo the sizeof(uintptr_t).
+ returnValue->memoryPointer = malloc(size + sizeof(uintptr_t) +
+ alignment - 1);
+ if(returnValue->memoryPointer == NULL)
+ {
+ delete returnValue;
+ return NULL;
+ }
+
+ // Alligning after the sizeof(header) bytes will leave room for the header
+ // in the same memory block.
+ uintptr_t alignStartPos = (uintptr_t)returnValue->memoryPointer;
+ alignStartPos += sizeof(uintptr_t);
+
+ // The buffer should be aligned with 'alignment' bytes. The - 1 guarantees
+ // that we align towards the lowest address.
+ uintptr_t alignedPos = (alignStartPos + alignment - 1) & ~(alignment - 1);
+
+ // alignedPos is the address sought for.
+ returnValue->alignedBuffer = (void*)alignedPos;
+
+ // Store the address to the AlignedMemory struct in the header so that a
+ // it's possible to reclaim all memory.
+ uintptr_t headerPos = alignedPos;
+ headerPos -= sizeof(uintptr_t);
+ void* headerPtr = (void*) headerPos;
+ uintptr_t headerValue = (uintptr_t)returnValue;
+ memcpy(headerPtr,&headerValue,sizeof(uintptr_t));
+
+ return returnValue->alignedBuffer;
+}
+
+void AlignedFree(void* memBlock)
+{
+ if(memBlock == NULL)
+ {
+ return;
+ }
+ uintptr_t alignedPos = (uintptr_t)memBlock;
+ uintptr_t headerPos = alignedPos - sizeof(uintptr_t);
+
+ // Read out the address of the AlignedMemory struct from the header.
+ uintptr_t* headerPtr = (uintptr_t*)headerPos;
+ AlignedMemory* deleteMemory = (AlignedMemory*) *headerPtr;
+
+ if(deleteMemory->memoryPointer != NULL)
+ {
+ free(deleteMemory->memoryPointer);
+ }
+ delete deleteMemory;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/atomic32.cc b/trunk/src/system_wrappers/source/atomic32.cc
new file mode 100644
index 0000000..588dd3e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/atomic32.cc
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "atomic32_wrapper.h"
+
+#if defined(_WIN32)
+ #include "atomic32_win.h"
+#elif defined(WEBRTC_LINUX)
+ #include "atomic32_linux.h"
+#elif defined(WEBRTC_MAC)
+ #include "atomic32_mac.h"
+#else
+ #error unsupported os!
+#endif
+
+namespace webrtc {
+Atomic32Wrapper::Atomic32Wrapper(WebRtc_Word32 initialValue)
+ : _impl(*new Atomic32Impl(initialValue))
+{
+}
+
+Atomic32Wrapper::~Atomic32Wrapper()
+{
+ delete &_impl;
+}
+
+WebRtc_Word32 Atomic32Wrapper::operator++()
+{
+ return ++_impl;
+}
+
+WebRtc_Word32 Atomic32Wrapper::operator--()
+{
+ return --_impl;
+}
+
+// Read and write to properly aligned variables are atomic operations.
+// Ex reference (for Windows): http://msdn.microsoft.com/en-us/library/ms684122(v=VS.85).aspx
+// TODO (hellner) operator= and Atomic32Wrapper::Value() can be fully
+// implemented here.
+Atomic32Wrapper& Atomic32Wrapper::operator=(const Atomic32Wrapper& rhs)
+{
+ if(this == &rhs)
+ {
+ return *this;
+ }
+ _impl = rhs._impl;
+ return *this;
+}
+
+Atomic32Wrapper& Atomic32Wrapper::operator=(WebRtc_Word32 rhs)
+{
+ _impl = rhs;
+ return *this;
+}
+
+WebRtc_Word32 Atomic32Wrapper::operator+=(WebRtc_Word32 rhs)
+{
+ return _impl += rhs;
+}
+
+WebRtc_Word32 Atomic32Wrapper::operator-=(WebRtc_Word32 rhs)
+{
+ return _impl -= rhs;
+}
+
+bool Atomic32Wrapper::CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue)
+{
+ return _impl.CompareExchange(newValue,compareValue);
+}
+
+WebRtc_Word32 Atomic32Wrapper::Value() const
+{
+ return _impl.Value();
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/atomic32_linux.h b/trunk/src/system_wrappers/source/atomic32_linux.h
new file mode 100644
index 0000000..f9f5650
--- /dev/null
+++ b/trunk/src/system_wrappers/source/atomic32_linux.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Atomic system independant 32-bit signed integer.
+// Linux implementation.
+// Note: Requires gcc 4.1.2 or later.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
+
+#include <inttypes.h>
+#include <malloc.h>
+
+#include "common_types.h"
+
+namespace webrtc {
+class Atomic32Impl
+{
+public:
+ inline Atomic32Impl(WebRtc_Word32 initialValue);
+ inline ~Atomic32Impl();
+
+ inline WebRtc_Word32 operator++();
+ inline WebRtc_Word32 operator--();
+
+ inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
+ inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
+
+ inline bool CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue);
+
+ inline WebRtc_Word32 Value() const;
+private:
+ void* _ptrMemory;
+ // Volatile ensures full memory barriers.
+ volatile WebRtc_Word32* _value;
+};
+
+// TODO (hellner) use aligned_malloc instead of doing it manually.
+inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
+ : _ptrMemory(NULL),
+ _value(NULL)
+{ // Align the memory associated with _value on a 32-bit boundary. This is a
+ // requirement for the used Linux APIs to be atomic.
+ // Keep _ptrMemory to be able to reclaim memory.
+ _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
+ _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
+ *_value = initialValue;
+}
+
+inline Atomic32Impl::~Atomic32Impl()
+{
+ if(_ptrMemory != NULL)
+ {
+ free(_ptrMemory);
+ }
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator++()
+{
+ WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,1);
+ returnValue++;
+ return returnValue;
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator--()
+{
+ WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,1);
+ returnValue--;
+ return returnValue;
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
+{
+ *_value = *rhs._value;
+ return *this;
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
+{
+ *_value = rhs;
+ return *this;
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
+{
+ WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,rhs);
+ returnValue += rhs;
+ return returnValue;
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
+{
+ WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,rhs);
+ returnValue -= rhs;
+ return returnValue;
+}
+
+inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue)
+{
+ return __sync_bool_compare_and_swap(_value,compareValue,newValue);
+}
+
+inline WebRtc_Word32 Atomic32Impl::Value() const
+{
+ return *_value;
+}
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
diff --git a/trunk/src/system_wrappers/source/atomic32_mac.h b/trunk/src/system_wrappers/source/atomic32_mac.h
new file mode 100644
index 0000000..bf8febc
--- /dev/null
+++ b/trunk/src/system_wrappers/source/atomic32_mac.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Atomic system independant 32-bit signed integer.
+// Mac implementation.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
+
+#include <stdlib.h>
+#include <libkern/OSAtomic.h>
+
+#include "common_types.h"
+
+namespace webrtc {
+class Atomic32Impl
+{
+public:
+ inline Atomic32Impl(WebRtc_Word32 initialValue);
+ inline ~Atomic32Impl();
+
+ inline WebRtc_Word32 operator++();
+ inline WebRtc_Word32 operator--();
+
+ inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
+ inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
+
+ inline bool CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue);
+
+ inline WebRtc_Word32 Value() const;
+private:
+ void* _ptrMemory;
+ // Volatile ensures full memory barriers.
+ volatile WebRtc_Word32* _value;
+};
+
+// TODO (hellner) use aligned_malloc instead of doing it manually.
+inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
+ :
+ _ptrMemory(NULL),
+ _value(NULL)
+{ // Align the memory associated with _value on a 32-bit boundary. This is a
+ // requirement for the used Mac APIs to be atomic.
+ // Keep _ptrMemory to be able to reclaim memory.
+ _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
+ _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
+ *_value = initialValue;
+}
+
+inline Atomic32Impl::~Atomic32Impl()
+{
+ if(_ptrMemory != NULL)
+ {
+ free(_ptrMemory);
+ }
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator++()
+{
+ return OSAtomicIncrement32Barrier(
+ reinterpret_cast<volatile int32_t*>(_value));
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator--()
+{
+ return OSAtomicDecrement32Barrier(
+ reinterpret_cast<volatile int32_t*>(_value));
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
+{
+ *_value = *rhs._value;
+ return *this;
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
+{
+ *_value = rhs;
+ return *this;
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
+{
+ return OSAtomicAdd32Barrier(rhs,
+ reinterpret_cast<volatile int32_t*>(_value));
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
+{
+ return OSAtomicAdd32Barrier(-rhs,
+ reinterpret_cast<volatile int32_t*>(_value));
+}
+
+inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue)
+{
+ return OSAtomicCompareAndSwap32Barrier(
+ compareValue,
+ newValue,
+ reinterpret_cast<volatile int32_t*>(_value));
+}
+
+inline WebRtc_Word32 Atomic32Impl::Value() const
+{
+ return *_value;
+}
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
diff --git a/trunk/src/system_wrappers/source/atomic32_win.h b/trunk/src/system_wrappers/source/atomic32_win.h
new file mode 100644
index 0000000..c27e48e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/atomic32_win.h
@@ -0,0 +1,113 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Atomic system independant 32-bit signed integer.
+// Windows implementation.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
+
+#include <malloc.h>
+#include <windows.h>
+
+#include "common_types.h"
+
+namespace webrtc {
+class Atomic32Impl
+{
+public:
+ inline Atomic32Impl(WebRtc_Word32 initialValue);
+ inline ~Atomic32Impl();
+
+ inline WebRtc_Word32 operator++();
+ inline WebRtc_Word32 operator--();
+
+ inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
+ inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
+ inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
+
+ inline bool CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue);
+
+ inline WebRtc_Word32 Value() const;
+private:
+ void* _ptrMemory;
+ // Volatile ensures full memory barriers.
+ volatile LONG* _value;
+};
+
+// TODO (hellner) use aligned_malloc instead of doing it manually.
+inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
+ : _ptrMemory(NULL),
+ _value(NULL)
+{ // Align the memory associated with _value on a 32-bit boundary. This is a
+ // requirement for the used Windows APIs to be atomic.
+ // Keep _ptrMemory to be able to reclaim memory.
+ _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
+ _value = reinterpret_cast<LONG*> (((uintptr_t)_ptrMemory+3)&(~0x3));
+ *_value = initialValue;
+}
+
+inline Atomic32Impl::~Atomic32Impl()
+{
+ if(_ptrMemory != NULL)
+ {
+ free(_ptrMemory);
+ }
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator++()
+{
+ return (WebRtc_Word32)InterlockedIncrement(_value);
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator--()
+{
+ return (WebRtc_Word32)InterlockedDecrement(_value);
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
+{
+ *_value = *rhs._value;
+ return *this;
+}
+
+inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
+{
+ *_value = rhs;
+ return *this;
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
+{
+ return InterlockedExchangeAdd(_value,rhs);
+}
+
+inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
+{
+ return InterlockedExchangeAdd(_value,-rhs);
+}
+
+inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
+ WebRtc_Word32 compareValue)
+{
+ const LONG oldValue = InterlockedCompareExchange(_value,newValue,
+ compareValue);
+ // If the old value and the compare value is the same an exchange happened.
+ return (oldValue == compareValue);
+}
+
+inline WebRtc_Word32 Atomic32Impl::Value() const
+{
+ return *_value;
+}
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/condition_variable.cc b/trunk/src/system_wrappers/source/condition_variable.cc
new file mode 100644
index 0000000..b37d037
--- /dev/null
+++ b/trunk/src/system_wrappers/source/condition_variable.cc
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(_WIN32)
+ #include <windows.h>
+ #include "condition_variable_wrapper.h"
+ #include "condition_variable_win.h"
+#elif defined(WEBRTC_LINUX)
+ #include <pthread.h>
+ #include "condition_variable_wrapper.h"
+ #include "condition_variable_posix.h"
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
+ #include <pthread.h>
+ #include "condition_variable_wrapper.h"
+ #include "condition_variable_posix.h"
+#endif
+
+namespace webrtc {
+ConditionVariableWrapper*
+ConditionVariableWrapper::CreateConditionVariable()
+{
+#if defined(_WIN32)
+ return new ConditionVariableWindows;
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
+ return ConditionVariablePosix::Create();
+#else
+ return NULL;
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_posix.cc b/trunk/src/system_wrappers/source/condition_variable_posix.cc
new file mode 100644
index 0000000..48835ab
--- /dev/null
+++ b/trunk/src/system_wrappers/source/condition_variable_posix.cc
@@ -0,0 +1,151 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "condition_variable_posix.h"
+
+#if defined(WEBRTC_LINUX)
+#include <ctime>
+#else
+#include <sys/time.h>
+#endif
+
+#include <errno.h>
+
+#include "critical_section_posix.h"
+
+namespace webrtc {
+ConditionVariableWrapper* ConditionVariablePosix::Create()
+{
+ ConditionVariablePosix* ptr = new ConditionVariablePosix;
+ if (!ptr)
+ {
+ return NULL;
+ }
+
+ const int error = ptr->Construct();
+ if (error)
+ {
+ delete ptr;
+ return NULL;
+ }
+
+ return ptr;
+}
+
+ConditionVariablePosix::ConditionVariablePosix()
+{
+}
+
+int ConditionVariablePosix::Construct()
+{
+ int result = 0;
+#ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ result = pthread_cond_init(&_cond, NULL);
+#else
+ pthread_condattr_t condAttr;
+ result = pthread_condattr_init(&condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_cond_init(&_cond, &condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_condattr_destroy(&condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+#endif
+ return 0;
+}
+
+ConditionVariablePosix::~ConditionVariablePosix()
+{
+ pthread_cond_destroy(&_cond);
+}
+
+void ConditionVariablePosix::SleepCS(CriticalSectionWrapper& critSect)
+{
+ CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
+ &critSect);
+ pthread_cond_wait(&_cond, &cs->_mutex);
+}
+
+
+bool
+ConditionVariablePosix::SleepCS(
+ CriticalSectionWrapper& critSect,
+ unsigned long maxTimeInMS)
+{
+ const unsigned long INFINITE = 0xFFFFFFFF;
+
+ const int MILLISECONDS_PER_SECOND = 1000;
+#ifndef WEBRTC_LINUX
+ const int MICROSECONDS_PER_MILLISECOND = 1000;
+#endif
+ const int NANOSECONDS_PER_SECOND = 1000000000;
+ const int NANOSECONDS_PER_MILLISECOND = 1000000;
+
+ CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
+ &critSect);
+
+ if (maxTimeInMS != INFINITE)
+ {
+ timespec ts;
+#ifndef WEBRTC_MAC
+#ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ clock_gettime(CLOCK_REALTIME, &ts);
+#else
+ clock_gettime(CLOCK_MONOTONIC, &ts);
+#endif
+#else
+ struct timeval tv;
+ gettimeofday(&tv, 0);
+ ts.tv_sec = tv.tv_sec;
+ ts.tv_nsec = tv.tv_usec * MICROSECONDS_PER_MILLISECOND;
+#endif
+
+ ts.tv_sec += maxTimeInMS / MILLISECONDS_PER_SECOND;
+ ts.tv_nsec += (maxTimeInMS - ((maxTimeInMS / MILLISECONDS_PER_SECOND)*
+ MILLISECONDS_PER_SECOND)) * NANOSECONDS_PER_MILLISECOND;
+
+ if (ts.tv_nsec >= NANOSECONDS_PER_SECOND)
+ {
+ ts.tv_sec += ts.tv_nsec / NANOSECONDS_PER_SECOND;
+ ts.tv_nsec %= NANOSECONDS_PER_SECOND;
+ }
+ const int res = pthread_cond_timedwait(&_cond, &cs->_mutex, &ts);
+ return (res == ETIMEDOUT) ? false : true;
+ }
+ else
+ {
+ pthread_cond_wait(&_cond, &cs->_mutex);
+ return true;
+ }
+}
+
+void ConditionVariablePosix::Wake()
+{
+ pthread_cond_signal(&_cond);
+}
+
+void ConditionVariablePosix::WakeAll()
+{
+ pthread_cond_broadcast(&_cond);
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_posix.h b/trunk/src/system_wrappers/source/condition_variable_posix.h
new file mode 100644
index 0000000..c239a47
--- /dev/null
+++ b/trunk/src/system_wrappers/source/condition_variable_posix.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
+
+#include "condition_variable_wrapper.h"
+
+#include <pthread.h>
+
+namespace webrtc {
+class ConditionVariablePosix : public ConditionVariableWrapper
+{
+public:
+ static ConditionVariableWrapper* Create();
+ ~ConditionVariablePosix();
+
+ void SleepCS(CriticalSectionWrapper& critSect);
+ bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
+ void Wake();
+ void WakeAll();
+
+private:
+ ConditionVariablePosix();
+ int Construct();
+
+private:
+ pthread_cond_t _cond;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/condition_variable_win.cc b/trunk/src/system_wrappers/source/condition_variable_win.cc
new file mode 100644
index 0000000..f4fae0b
--- /dev/null
+++ b/trunk/src/system_wrappers/source/condition_variable_win.cc
@@ -0,0 +1,224 @@
+/*
+ * Use of this source code is governed by the ACE copyright license which
+ * can be found in the LICENSE file in the third_party_mods/ace directory of
+ * the source tree or at http://www1.cse.wustl.edu/~schmidt/ACE-copying.html.
+ */
+/*
+ * This source code contain modifications to the original source code
+ * which can be found here:
+ * http://www.cs.wustl.edu/~schmidt/win32-cv-1.html (section 3.2).
+ * Modifications:
+ * 1) Dynamic detection of native support for condition variables.
+ * 2) Use of WebRTC defined types and classes. Renaming of some functions.
+ * 3) Introduction of a second event for wake all functionality. This prevents
+ * a thread from spinning on the same condition variable, preventing other
+ * threads from waking up.
+ */
+
+// TODO (hellner): probably nicer to split up native and generic
+// implementation into two different files
+
+#include "condition_variable_win.h"
+
+#include "critical_section_win.h"
+#include "trace.h"
+
+namespace webrtc {
+bool ConditionVariableWindows::_winSupportConditionVariablesPrimitive = false;
+static HMODULE library = NULL;
+
+PInitializeConditionVariable _PInitializeConditionVariable;
+PSleepConditionVariableCS _PSleepConditionVariableCS;
+PWakeConditionVariable _PWakeConditionVariable;
+PWakeAllConditionVariable _PWakeAllConditionVariable;
+
+typedef void (WINAPI *PInitializeConditionVariable)(PCONDITION_VARIABLE);
+typedef BOOL (WINAPI *PSleepConditionVariableCS)(PCONDITION_VARIABLE,
+ PCRITICAL_SECTION, DWORD);
+typedef void (WINAPI *PWakeConditionVariable)(PCONDITION_VARIABLE);
+typedef void (WINAPI *PWakeAllConditionVariable)(PCONDITION_VARIABLE);
+
+ConditionVariableWindows::ConditionVariableWindows()
+ : _eventID(WAKEALL_0)
+{
+ if (!library)
+ {
+ // Use native implementation if supported (i.e Vista+)
+ library = LoadLibrary(TEXT("Kernel32.dll"));
+ if (library)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Loaded Kernel.dll");
+
+ _PInitializeConditionVariable =
+ (PInitializeConditionVariable) GetProcAddress(
+ library,
+ "InitializeConditionVariable");
+ _PSleepConditionVariableCS =
+ (PSleepConditionVariableCS)GetProcAddress(
+ library,
+ "SleepConditionVariableCS");
+ _PWakeConditionVariable =
+ (PWakeConditionVariable)GetProcAddress(
+ library,
+ "WakeConditionVariable");
+ _PWakeAllConditionVariable =
+ (PWakeAllConditionVariable)GetProcAddress(
+ library,
+ "WakeAllConditionVariable");
+
+ if(_PInitializeConditionVariable &&
+ _PSleepConditionVariableCS &&
+ _PWakeConditionVariable &&
+ _PWakeAllConditionVariable)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Loaded native condition variables");
+ _winSupportConditionVariablesPrimitive = true;
+ }
+ }
+ }
+
+ if (_winSupportConditionVariablesPrimitive)
+ {
+ _PInitializeConditionVariable(&_conditionVariable);
+
+ _events[WAKEALL_0] = NULL;
+ _events[WAKEALL_1] = NULL;
+ _events[WAKE] = NULL;
+
+ } else {
+ memset(&_numWaiters[0],0,sizeof(_numWaiters));
+
+ InitializeCriticalSection(&_numWaitersCritSect);
+
+ _events[WAKEALL_0] = CreateEvent(NULL, // no security attributes
+ TRUE, // manual-reset, sticky event
+ FALSE, // initial state non-signaled
+ NULL); // no name for event
+
+ _events[WAKEALL_1] = CreateEvent(NULL, // no security attributes
+ TRUE, // manual-reset, sticky event
+ FALSE, // initial state non-signaled
+ NULL); // no name for event
+
+ _events[WAKE] = CreateEvent(NULL, // no security attributes
+ FALSE, // auto-reset, sticky event
+ FALSE, // initial state non-signaled
+ NULL); // no name for event
+ }
+}
+
+ConditionVariableWindows::~ConditionVariableWindows()
+{
+ if(!_winSupportConditionVariablesPrimitive)
+ {
+ CloseHandle(_events[WAKE]);
+ CloseHandle(_events[WAKEALL_1]);
+ CloseHandle(_events[WAKEALL_0]);
+
+ DeleteCriticalSection(&_numWaitersCritSect);
+ }
+}
+
+void ConditionVariableWindows::SleepCS(CriticalSectionWrapper& critSect)
+{
+ SleepCS(critSect, INFINITE);
+}
+
+bool ConditionVariableWindows::SleepCS(CriticalSectionWrapper& critSect,
+ unsigned long maxTimeInMS)
+{
+ CriticalSectionWindows* cs = reinterpret_cast<CriticalSectionWindows*>(
+ &critSect);
+
+ if(_winSupportConditionVariablesPrimitive)
+ {
+ BOOL retVal = _PSleepConditionVariableCS(&_conditionVariable,
+ &(cs->crit),maxTimeInMS);
+ return (retVal == 0) ? false : true;
+
+ }else
+ {
+ EnterCriticalSection(&_numWaitersCritSect);
+ // Get the eventID for the event that will be triggered by next
+ // WakeAll() call and start waiting for it.
+ const EventWakeUpType eventID = (WAKEALL_0 == _eventID) ?
+ WAKEALL_1 : WAKEALL_0;
+ ++(_numWaiters[eventID]);
+ LeaveCriticalSection(&_numWaitersCritSect);
+
+ LeaveCriticalSection(&cs->crit);
+ HANDLE events[2];
+ events[0] = _events[WAKE];
+ events[1] = _events[eventID];
+ const DWORD result = WaitForMultipleObjects(2, // Wait on 2 events.
+ events,
+ FALSE, // Wait for either.
+ maxTimeInMS);
+
+ const bool retVal = (result != WAIT_TIMEOUT);
+
+ EnterCriticalSection(&_numWaitersCritSect);
+ --(_numWaiters[eventID]);
+ // Last waiter should only be true for WakeAll(). WakeAll() correspond
+ // to position 1 in events[] -> (result == WAIT_OBJECT_0 + 1)
+ const bool lastWaiter = (result == WAIT_OBJECT_0 + 1) &&
+ (_numWaiters[eventID] == 0);
+ LeaveCriticalSection(&_numWaitersCritSect);
+
+ if (lastWaiter)
+ {
+ // Reset/unset the WakeAll() event since all threads have been
+ // released.
+ ResetEvent(_events[eventID]);
+ }
+
+ EnterCriticalSection(&cs->crit);
+ return retVal;
+ }
+}
+
+void
+ConditionVariableWindows::Wake()
+{
+ if(_winSupportConditionVariablesPrimitive)
+ {
+ _PWakeConditionVariable(&_conditionVariable);
+ }else
+ {
+ EnterCriticalSection(&_numWaitersCritSect);
+ const bool haveWaiters = (_numWaiters[WAKEALL_0] > 0) ||
+ (_numWaiters[WAKEALL_1] > 0);
+ LeaveCriticalSection(&_numWaitersCritSect);
+
+ if (haveWaiters)
+ {
+ SetEvent(_events[WAKE]);
+ }
+ }
+}
+
+void
+ConditionVariableWindows::WakeAll()
+{
+ if(_winSupportConditionVariablesPrimitive)
+ {
+ _PWakeAllConditionVariable(&_conditionVariable);
+ }else
+ {
+ EnterCriticalSection(&_numWaitersCritSect);
+ // Update current WakeAll() event
+ _eventID = (WAKEALL_0 == _eventID) ? WAKEALL_1 : WAKEALL_0;
+ // Trigger current event
+ const EventWakeUpType eventID = _eventID;
+ const bool haveWaiters = _numWaiters[eventID] > 0;
+ LeaveCriticalSection(&_numWaitersCritSect);
+
+ if (haveWaiters)
+ {
+ SetEvent(_events[eventID]);
+ }
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_win.h b/trunk/src/system_wrappers/source/condition_variable_win.h
new file mode 100644
index 0000000..aab2564
--- /dev/null
+++ b/trunk/src/system_wrappers/source/condition_variable_win.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
+
+#include "condition_variable_wrapper.h"
+
+#include <windows.h>
+
+namespace webrtc {
+#if !defined CONDITION_VARIABLE_INIT
+ typedef struct _RTL_CONDITION_VARIABLE
+ {
+ void* Ptr;
+ } RTL_CONDITION_VARIABLE, *PRTL_CONDITION_VARIABLE;
+
+ typedef RTL_CONDITION_VARIABLE CONDITION_VARIABLE, *PCONDITION_VARIABLE;
+#endif
+
+typedef void (WINAPI *PInitializeConditionVariable)(PCONDITION_VARIABLE);
+typedef BOOL (WINAPI *PSleepConditionVariableCS)(PCONDITION_VARIABLE,
+ PCRITICAL_SECTION, DWORD);
+typedef void (WINAPI *PWakeConditionVariable)(PCONDITION_VARIABLE);
+typedef void (WINAPI *PWakeAllConditionVariable)(PCONDITION_VARIABLE);
+
+
+class ConditionVariableWindows : public ConditionVariableWrapper
+{
+public:
+ ConditionVariableWindows();
+ ~ConditionVariableWindows();
+
+ void SleepCS(CriticalSectionWrapper& critSect);
+ bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
+ void Wake();
+ void WakeAll();
+
+private:
+ enum EventWakeUpType
+ {
+ WAKEALL_0 = 0,
+ WAKEALL_1 = 1,
+ WAKE = 2,
+ EVENT_COUNT = 3
+ };
+
+private:
+ // Native support for Windows Vista+
+ static bool _winSupportConditionVariablesPrimitive;
+ CONDITION_VARIABLE _conditionVariable;
+
+ unsigned int _numWaiters[2];
+ EventWakeUpType _eventID;
+ CRITICAL_SECTION _numWaitersCritSect;
+ HANDLE _events[EVENT_COUNT];
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/cpu.cc b/trunk/src/system_wrappers/source/cpu.cc
new file mode 100644
index 0000000..3df5d18
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu.cc
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_wrapper.h"
+
+#if defined(_WIN32)
+ #include "cpu_win.h"
+#elif defined(WEBRTC_MAC)
+ #include "cpu_mac.h"
+#elif defined(WEBRTC_MAC_INTEL)
+ #include "cpu_mac.h"
+#elif defined(WEBRTC_ANDROID)
+ // Not implemented yet, might be possible to use Linux implementation
+#else // defined(WEBRTC_LINUX)
+ #include "cpu_linux.h"
+#endif
+
+namespace webrtc {
+CpuWrapper* CpuWrapper::CreateCpu()
+{
+#if defined(_WIN32)
+ return new CpuWindows();
+#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
+ return new CpuWrapperMac();
+#elif defined(WEBRTC_ANDROID)
+ return 0;
+#else
+ return new CpuLinux();
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_features.cc b/trunk/src/system_wrappers/source/cpu_features.cc
new file mode 100644
index 0000000..41a86e3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_features.cc
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Parts of this file derived from Chromium's base/cpu.cc.
+
+#include "cpu_features_wrapper.h"
+
+#include "typedefs.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(_MSC_VER)
+#include <intrin.h>
+#endif
+#endif
+
+// No CPU feature is available => straight C path.
+int GetCPUInfoNoASM(CPUFeature feature) {
+ (void)feature;
+ return 0;
+}
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#ifndef _MSC_VER
+// Intrinsic for "cpuid".
+#if defined(__pic__) && defined(__i386__)
+static inline void __cpuid(int cpu_info[4], int info_type) {
+ __asm__ volatile (
+ "mov %%ebx, %%edi\n"
+ "cpuid\n"
+ "xchg %%edi, %%ebx\n"
+ : "=a"(cpu_info[0]), "=D"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
+ : "a"(info_type));
+}
+#else
+static inline void __cpuid(int cpu_info[4], int info_type) {
+ __asm__ volatile (
+ "cpuid\n"
+ : "=a"(cpu_info[0]), "=b"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
+ : "a"(info_type));
+}
+#endif
+#endif // _MSC_VER
+#endif // WEBRTC_ARCH_X86_FAMILY
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Actual feature detection for x86.
+static int GetCPUInfo(CPUFeature feature) {
+ int cpu_info[4];
+ __cpuid(cpu_info, 1);
+ if (feature == kSSE2) {
+ return 0 != (cpu_info[3] & 0x04000000);
+ }
+ if (feature == kSSE3) {
+ return 0 != (cpu_info[2] & 0x00000001);
+ }
+ return 0;
+}
+#else
+// Default to straight C for other platforms.
+static int GetCPUInfo(CPUFeature feature) {
+ (void)feature;
+ return 0;
+}
+#endif
+
+WebRtc_CPUInfo WebRtc_GetCPUInfo = GetCPUInfo;
+WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM = GetCPUInfoNoASM;
diff --git a/trunk/src/system_wrappers/source/cpu_features_arm.c b/trunk/src/system_wrappers/source/cpu_features_arm.c
new file mode 100644
index 0000000..1065118
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_features_arm.c
@@ -0,0 +1,333 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is derived from Android's NDK package r7, located at
+// <ndk>/sources/android/cpufeatures/ (downloadable from
+// http://developer.android.com/sdk/ndk/index.html).
+
+#include "cpu_features_wrapper.h"
+
+#include <fcntl.h>
+#include <errno.h>
+#include <pthread.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+// Define CPU family.
+typedef enum {
+ CPU_FAMILY_UNKNOWN = 0,
+ CPU_FAMILY_ARM,
+ CPU_FAMILY_X86,
+ CPU_FAMILY_MAX // Do not remove.
+} CpuFamily;
+
+static pthread_once_t g_once;
+static CpuFamily g_cpuFamily;
+static uint64_t g_cpuFeatures;
+static int g_cpuCount;
+
+static const int cpufeatures_debug = 0;
+
+#ifdef __arm__
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_ARM
+#elif defined __i386__
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_X86
+#else
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_UNKNOWN
+#endif
+
+#define D(...) \
+ do { \
+ if (cpufeatures_debug) { \
+ printf(__VA_ARGS__); fflush(stdout); \
+ } \
+ } while (0)
+
+/* Read the content of /proc/cpuinfo into a user-provided buffer.
+ * Return the length of the data, or -1 on error. Does *not*
+ * zero-terminate the content. Will not read more
+ * than 'buffsize' bytes.
+ */
+static int read_file(const char* pathname, char* buffer, size_t buffsize) {
+ int fd, len;
+
+ fd = open(pathname, O_RDONLY);
+ if (fd < 0)
+ return -1;
+
+ do {
+ len = read(fd, buffer, buffsize);
+ } while (len < 0 && errno == EINTR);
+
+ close(fd);
+
+ return len;
+}
+
+/* Extract the content of a the first occurence of a given field in
+ * the content of /proc/cpuinfo and return it as a heap-allocated
+ * string that must be freed by the caller.
+ *
+ * Return NULL if not found
+ */
+static char* extract_cpuinfo_field(char* buffer, int buflen, const char* field) {
+ int fieldlen = strlen(field);
+ char* bufend = buffer + buflen;
+ char* result = NULL;
+ int len, ignore;
+ const char* p, *q;
+
+ /* Look for first field occurence, and ensures it starts the line.
+ */
+ p = buffer;
+ bufend = buffer + buflen;
+ for (;;) {
+ p = memmem(p, bufend - p, field, fieldlen);
+ if (p == NULL)
+ goto EXIT;
+
+ if (p == buffer || p[-1] == '\n')
+ break;
+
+ p += fieldlen;
+ }
+
+ /* Skip to the first column followed by a space */
+ p += fieldlen;
+ p = memchr(p, ':', bufend - p);
+ if (p == NULL || p[1] != ' ')
+ goto EXIT;
+
+ /* Find the end of the line */
+ p += 2;
+ q = memchr(p, '\n', bufend - p);
+ if (q == NULL)
+ q = bufend;
+
+ /* Copy the line into a heap-allocated buffer */
+ len = q - p;
+ result = malloc(len + 1);
+ if (result == NULL)
+ goto EXIT;
+
+ memcpy(result, p, len);
+ result[len] = '\0';
+
+EXIT:
+ return result;
+}
+
+/* Count the number of occurences of a given field prefix in /proc/cpuinfo.
+ */
+static int count_cpuinfo_field(char* buffer, int buflen, const char* field) {
+ int fieldlen = strlen(field);
+ const char* p = buffer;
+ const char* bufend = buffer + buflen;
+ const char* q;
+ int count = 0;
+
+ for (;;) {
+ const char* q;
+
+ p = memmem(p, bufend - p, field, fieldlen);
+ if (p == NULL)
+ break;
+
+ /* Ensure that the field is at the start of a line */
+ if (p > buffer && p[-1] != '\n') {
+ p += fieldlen;
+ continue;
+ }
+
+
+ /* skip any whitespace */
+ q = p + fieldlen;
+ while (q < bufend && (*q == ' ' || *q == '\t'))
+ q++;
+
+ /* we must have a colon now */
+ if (q < bufend && *q == ':') {
+ count += 1;
+ q ++;
+ }
+ p = q;
+ }
+
+ return count;
+}
+
+/* Like strlen(), but for constant string literals */
+#define STRLEN_CONST(x) ((sizeof(x)-1)
+
+
+/* Checks that a space-separated list of items contains one given 'item'.
+ * Returns 1 if found, 0 otherwise.
+ */
+static int has_list_item(const char* list, const char* item) {
+ const char* p = list;
+ int itemlen = strlen(item);
+
+ if (list == NULL)
+ return 0;
+
+ while (*p) {
+ const char* q;
+
+ /* skip spaces */
+ while (*p == ' ' || *p == '\t')
+ p++;
+
+ /* find end of current list item */
+ q = p;
+ while (*q && *q != ' ' && *q != '\t')
+ q++;
+
+ if (itemlen == q - p && !memcmp(p, item, itemlen))
+ return 1;
+
+ /* skip to next item */
+ p = q;
+ }
+ return 0;
+}
+
+
+static void cpuInit(void) {
+ char cpuinfo[4096];
+ int cpuinfo_len;
+
+ g_cpuFamily = DEFAULT_CPU_FAMILY;
+ g_cpuFeatures = 0;
+ g_cpuCount = 1;
+
+ cpuinfo_len = read_file("/proc/cpuinfo", cpuinfo, sizeof cpuinfo);
+ D("cpuinfo_len is (%d):\n%.*s\n", cpuinfo_len,
+ cpuinfo_len >= 0 ? cpuinfo_len : 0, cpuinfo);
+
+ if (cpuinfo_len < 0) { /* should not happen */
+ return;
+ }
+
+ /* Count the CPU cores, the value may be 0 for single-core CPUs */
+ g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "processor");
+ if (g_cpuCount == 0) {
+ g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "Processor");
+ if (g_cpuCount == 0) {
+ g_cpuCount = 1;
+ }
+ }
+
+ D("found cpuCount = %d\n", g_cpuCount);
+
+#ifdef __arm__
+ {
+ char* features = NULL;
+ char* architecture = NULL;
+
+ /* Extract architecture from the "CPU Architecture" field.
+ * The list is well-known, unlike the the output of
+ * the 'Processor' field which can vary greatly.
+ *
+ * See the definition of the 'proc_arch' array in
+ * $KERNEL/arch/arm/kernel/setup.c and the 'c_show' function in
+ * same file.
+ */
+ char* cpuArch = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "CPU architecture");
+
+ if (cpuArch != NULL) {
+ char* end;
+ long archNumber;
+ int hasARMv7 = 0;
+
+ D("found cpuArch = '%s'\n", cpuArch);
+
+ /* read the initial decimal number, ignore the rest */
+ archNumber = strtol(cpuArch, &end, 10);
+
+ /* Here we assume that ARMv8 will be upwards compatible with v7
+ * in the future. Unfortunately, there is no 'Features' field to
+ * indicate that Thumb-2 is supported.
+ */
+ if (end > cpuArch && archNumber >= 7) {
+ hasARMv7 = 1;
+ }
+
+ /* Unfortunately, it seems that certain ARMv6-based CPUs
+ * report an incorrect architecture number of 7!
+ *
+ * We try to correct this by looking at the 'elf_format'
+ * field reported by the 'Processor' field, which is of the
+ * form of "(v7l)" for an ARMv7-based CPU, and "(v6l)" for
+ * an ARMv6-one.
+ */
+ if (hasARMv7) {
+ char* cpuProc = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "Processor");
+ if (cpuProc != NULL) {
+ D("found cpuProc = '%s'\n", cpuProc);
+ if (has_list_item(cpuProc, "(v6l)")) {
+ D("CPU processor and architecture mismatch!!\n");
+ hasARMv7 = 0;
+ }
+ free(cpuProc);
+ }
+ }
+
+ if (hasARMv7) {
+ g_cpuFeatures |= kCPUFeatureARMv7;
+ }
+
+ /* The LDREX / STREX instructions are available from ARMv6 */
+ if (archNumber >= 6) {
+ g_cpuFeatures |= kCPUFeatureLDREXSTREX;
+ }
+
+ free(cpuArch);
+ }
+
+ /* Extract the list of CPU features from 'Features' field */
+ char* cpuFeatures = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "Features");
+
+ if (cpuFeatures != NULL) {
+
+ D("found cpuFeatures = '%s'\n", cpuFeatures);
+
+ if (has_list_item(cpuFeatures, "vfpv3"))
+ g_cpuFeatures |= kCPUFeatureVFPv3;
+
+ else if (has_list_item(cpuFeatures, "vfpv3d16"))
+ g_cpuFeatures |= kCPUFeatureVFPv3;
+
+ if (has_list_item(cpuFeatures, "neon")) {
+ /* Note: Certain kernels only report neon but not vfpv3
+ * in their features list. However, ARM mandates
+ * that if Neon is implemented, so must be VFPv3
+ * so always set the flag.
+ */
+ g_cpuFeatures |= kCPUFeatureNEON |
+ kCPUFeatureVFPv3;
+ }
+ free(cpuFeatures);
+ }
+ }
+#endif // __arm__
+
+#ifdef __i386__
+ g_cpuFamily = CPU_FAMILY_X86;
+#endif
+}
+
+
+uint64_t WebRtc_GetCPUFeaturesARM(void) {
+ pthread_once(&g_once, cpuInit);
+ return g_cpuFeatures;
+}
diff --git a/trunk/src/system_wrappers/source/cpu_info.cc b/trunk/src/system_wrappers/source/cpu_info.cc
new file mode 100644
index 0000000..e367abf
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_info.cc
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_info.h"
+
+#if defined(_WIN32)
+#include <Windows.h>
+#elif defined(WEBRTC_MAC)
+#include <sys/types.h>
+#include <sys/sysctl.h>
+#elif defined(WEBRTC_MAC_INTEL)
+// Intentionally empty
+#elif defined(WEBRTC_ANDROID)
+// Not implemented yet, might be possible to use Linux implementation
+#else // defined(WEBRTC_LINUX)
+#include <sys/sysinfo.h>
+#endif
+
+#include "trace.h"
+
+namespace webrtc {
+
+WebRtc_UWord32 CpuInfo::_numberOfCores = 0;
+
+WebRtc_UWord32 CpuInfo::DetectNumberOfCores()
+{
+ if (!_numberOfCores)
+ {
+#if defined(_WIN32)
+ SYSTEM_INFO si;
+ GetSystemInfo(&si);
+ _numberOfCores = static_cast<WebRtc_UWord32>(si.dwNumberOfProcessors);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+
+#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+ _numberOfCores = get_nprocs();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+
+#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
+ int name[] = {CTL_HW, HW_AVAILCPU};
+ int ncpu;
+ size_t size = sizeof(ncpu);
+ if(0 == sysctl(name, 2, &ncpu, &size, NULL, 0))
+ {
+ _numberOfCores = static_cast<WebRtc_UWord32>(ncpu);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "Failed to get number of cores");
+ _numberOfCores = 1;
+ }
+#else
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
+ "No function to get number of cores");
+ _numberOfCores = 1;
+#endif
+ }
+ return _numberOfCores;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_linux.cc b/trunk/src/system_wrappers/source/cpu_linux.cc
new file mode 100644
index 0000000..8e8ecda
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_linux.cc
@@ -0,0 +1,202 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_linux.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+
+namespace webrtc {
+CpuLinux::CpuLinux()
+ : m_oldBusyTime(0),
+ m_oldIdleTime(0),
+ m_oldBusyTimeMulti(NULL),
+ m_oldIdleTimeMulti(NULL),
+ m_idleArray(NULL),
+ m_busyArray(NULL),
+ m_resultArray(NULL),
+ m_numCores(0) {
+ const int result = GetNumCores();
+ if (result != -1) {
+ m_numCores = result;
+ m_oldBusyTimeMulti = new long long[m_numCores];
+ memset(m_oldBusyTimeMulti, 0, sizeof(long long) * m_numCores);
+ m_oldIdleTimeMulti = new long long[m_numCores];
+ memset(m_oldIdleTimeMulti, 0, sizeof(long long) * m_numCores);
+ m_idleArray = new long long[m_numCores];
+ memset(m_idleArray, 0, sizeof(long long) * m_numCores);
+ m_busyArray = new long long[m_numCores];
+ memset(m_busyArray, 0, sizeof(long long) * m_numCores);
+ m_resultArray = new WebRtc_UWord32[m_numCores];
+
+ GetData(m_oldBusyTime, m_oldIdleTime, m_busyArray, m_idleArray);
+ }
+}
+
+CpuLinux::~CpuLinux()
+{
+ delete [] m_oldBusyTimeMulti;
+ delete [] m_oldIdleTimeMulti;
+ delete [] m_idleArray;
+ delete [] m_busyArray;
+ delete [] m_resultArray;
+}
+
+WebRtc_Word32 CpuLinux::CpuUsage()
+{
+ WebRtc_UWord32 dummy = 0;
+ WebRtc_UWord32* dummyArray = NULL;
+ return CpuUsageMultiCore(dummy, dummyArray);
+}
+
+WebRtc_Word32 CpuLinux::CpuUsageMultiCore(WebRtc_UWord32& numCores,
+ WebRtc_UWord32*& coreArray)
+{
+ coreArray = m_resultArray;
+ numCores = m_numCores;
+ long long busy = 0;
+ long long idle = 0;
+ if (GetData(busy, idle, m_busyArray, m_idleArray) != 0)
+ return -1;
+
+ long long deltaBusy = busy - m_oldBusyTime;
+ long long deltaIdle = idle - m_oldIdleTime;
+ m_oldBusyTime = busy;
+ m_oldIdleTime = idle;
+
+ int retVal = -1;
+ if (deltaBusy + deltaIdle == 0)
+ {
+ retVal = 0;
+ }
+ else
+ {
+ retVal = (int)(100 * (deltaBusy) / (deltaBusy + deltaIdle));
+ }
+
+ if (coreArray == NULL)
+ {
+ return retVal;
+ }
+
+ for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
+ {
+ deltaBusy = m_busyArray[i] - m_oldBusyTimeMulti[i];
+ deltaIdle = m_idleArray[i] - m_oldIdleTimeMulti[i];
+ m_oldBusyTimeMulti[i] = m_busyArray[i];
+ m_oldIdleTimeMulti[i] = m_idleArray[i];
+ if(deltaBusy + deltaIdle == 0)
+ {
+ coreArray[i] = 0;
+ }
+ else
+ {
+ coreArray[i] = (int)(100 * (deltaBusy) / (deltaBusy+deltaIdle));
+ }
+ }
+ return retVal;
+}
+
+
+int CpuLinux::GetData(long long& busy, long long& idle, long long*& busyArray,
+ long long*& idleArray)
+{
+ FILE* fp = fopen("/proc/stat", "r");
+ if (!fp)
+ {
+ return -1;
+ }
+
+ char line[100];
+ if (fgets(line, 100, fp) == NULL) {
+ fclose(fp);
+ return -1;
+ }
+ char firstWord[100];
+ if (sscanf(line, "%s ", firstWord) != 1) {
+ fclose(fp);
+ return -1;
+ }
+ if (strncmp(firstWord, "cpu", 3) != 0) {
+ fclose(fp);
+ return -1;
+ }
+ char sUser[100];
+ char sNice[100];
+ char sSystem[100];
+ char sIdle[100];
+ if (sscanf(line, "%s %s %s %s %s ",
+ firstWord, sUser, sNice, sSystem, sIdle) != 5) {
+ fclose(fp);
+ return -1;
+ }
+ long long luser = atoll(sUser);
+ long long lnice = atoll(sNice);
+ long long lsystem = atoll(sSystem);
+ long long lidle = atoll (sIdle);
+
+ busy = luser + lnice + lsystem;
+ idle = lidle;
+ for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
+ {
+ if (fgets(line, 100, fp) == NULL) {
+ fclose(fp);
+ return -1;
+ }
+ if (sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem,
+ sIdle) != 5) {
+ fclose(fp);
+ return -1;
+ }
+ luser = atoll(sUser);
+ lnice = atoll(sNice);
+ lsystem = atoll(sSystem);
+ lidle = atoll (sIdle);
+ busyArray[i] = luser + lnice + lsystem;
+ idleArray[i] = lidle;
+ }
+ fclose(fp);
+ return 0;
+}
+
+int CpuLinux::GetNumCores()
+{
+ FILE* fp = fopen("/proc/stat", "r");
+ if (!fp)
+ {
+ return -1;
+ }
+ // Skip first line
+ char line[100];
+ if (!fgets(line, 100, fp))
+ {
+ fclose(fp);
+ return -1;
+ }
+ int numCores = -1;
+ char firstWord[100];
+ do
+ {
+ numCores++;
+ if (fgets(line, 100, fp))
+ {
+ if (sscanf(line, "%s ", firstWord) != 1) {
+ firstWord[0] = '\0';
+ }
+ } else {
+ break;
+ }
+ } while (strncmp(firstWord, "cpu", 3) == 0);
+ fclose(fp);
+ return numCores;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_linux.h b/trunk/src/system_wrappers/source/cpu_linux.h
new file mode 100644
index 0000000..9b22e83
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_linux.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
+
+#include "cpu_wrapper.h"
+
+namespace webrtc {
+class CpuLinux : public CpuWrapper
+{
+public:
+ CpuLinux();
+ virtual ~CpuLinux();
+
+ virtual WebRtc_Word32 CpuUsage();
+ virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
+ WebRtc_UWord32 /*length*/) {return 0;}
+ virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return 0;}
+
+ virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
+ WebRtc_UWord32*& array);
+
+ virtual void Reset() {return;}
+ virtual void Stop() {return;}
+private:
+ int GetData(long long& busy, long long& idle, long long*& busyArray,
+ long long*& idleArray);
+ int GetNumCores();
+
+ long long m_oldBusyTime;
+ long long m_oldIdleTime;
+
+ long long* m_oldBusyTimeMulti;
+ long long* m_oldIdleTimeMulti;
+
+ long long* m_idleArray;
+ long long* m_busyArray;
+ WebRtc_UWord32* m_resultArray;
+ WebRtc_UWord32 m_numCores;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
diff --git a/trunk/src/system_wrappers/source/cpu_mac.cc b/trunk/src/system_wrappers/source/cpu_mac.cc
new file mode 100644
index 0000000..d82bf07
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_mac.cc
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_mac.h"
+
+#include <iostream>
+#include <mach/mach.h>
+#include <mach/mach_error.h>
+
+#include "tick_util.h"
+
+namespace webrtc {
+CpuWrapperMac::CpuWrapperMac()
+ : _cpuCount(0),
+ _cpuUsage(NULL),
+ _totalCpuUsage(0),
+ _lastTickCount(NULL),
+ _lastTime(0)
+{
+ natural_t cpuCount;
+ processor_info_array_t infoArray;
+ mach_msg_type_number_t infoCount;
+
+ kern_return_t error = host_processor_info(mach_host_self(),
+ PROCESSOR_CPU_LOAD_INFO,
+ &cpuCount,
+ &infoArray,
+ &infoCount);
+ if (error)
+ {
+ return;
+ }
+
+ _cpuCount = cpuCount;
+ _cpuUsage = new WebRtc_UWord32[cpuCount];
+ _lastTickCount = new WebRtc_Word64[cpuCount];
+ _lastTime = TickTime::MillisecondTimestamp();
+
+ processor_cpu_load_info_data_t* cpuLoadInfo =
+ (processor_cpu_load_info_data_t*) infoArray;
+ for (unsigned int cpu= 0; cpu < cpuCount; cpu++)
+ {
+ WebRtc_Word64 ticks = 0;
+ for (int state = 0; state < 2; state++)
+ {
+ ticks += cpuLoadInfo[cpu].cpu_ticks[state];
+ }
+ _lastTickCount[cpu] = ticks;
+ _cpuUsage[cpu] = 0;
+ }
+ vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
+}
+
+CpuWrapperMac::~CpuWrapperMac()
+{
+ delete[] _cpuUsage;
+ delete[] _lastTickCount;
+}
+
+WebRtc_Word32 CpuWrapperMac::CpuUsage()
+{
+ WebRtc_UWord32 numCores;
+ WebRtc_UWord32* array = NULL;
+ return CpuUsageMultiCore(numCores, array);
+}
+
+WebRtc_Word32
+CpuWrapperMac::CpuUsageMultiCore(WebRtc_UWord32& numCores,
+ WebRtc_UWord32*& array)
+{
+ // sanity check
+ if(_cpuUsage == NULL)
+ {
+ return -1;
+ }
+
+ WebRtc_Word64 now = TickTime::MillisecondTimestamp();
+ WebRtc_Word64 timeDiffMS = now - _lastTime;
+ if(timeDiffMS >= 500)
+ {
+ if(Update(timeDiffMS) != 0)
+ {
+ return -1;
+ }
+ _lastTime = now;
+ }
+
+ numCores = _cpuCount;
+ array = _cpuUsage;
+ return _totalCpuUsage / _cpuCount;
+}
+
+WebRtc_Word32 CpuWrapperMac::Update(WebRtc_Word64 timeDiffMS)
+{
+ natural_t cpuCount;
+ processor_info_array_t infoArray;
+ mach_msg_type_number_t infoCount;
+
+ kern_return_t error = host_processor_info(mach_host_self(),
+ PROCESSOR_CPU_LOAD_INFO,
+ &cpuCount,
+ &infoArray,
+ &infoCount);
+ if (error)
+ {
+ return -1;
+ }
+
+ processor_cpu_load_info_data_t* cpuLoadInfo =
+ (processor_cpu_load_info_data_t*) infoArray;
+
+ _totalCpuUsage = 0;
+ for (unsigned int cpu = 0; cpu < cpuCount; cpu++)
+ {
+ WebRtc_Word64 ticks = 0;
+ for (int state = 0; state < 2; state++)
+ {
+ ticks += cpuLoadInfo[cpu].cpu_ticks[state];
+ }
+ if(timeDiffMS <= 0)
+ {
+ _cpuUsage[cpu] = 0;
+ }else {
+ _cpuUsage[cpu] = (WebRtc_UWord32)((1000 *
+ (ticks - _lastTickCount[cpu])) /
+ timeDiffMS);
+ }
+ _lastTickCount[cpu] = ticks;
+ _totalCpuUsage += _cpuUsage[cpu];
+ }
+
+ vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
+
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_mac.h b/trunk/src/system_wrappers/source/cpu_mac.h
new file mode 100644
index 0000000..f9f8207
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_mac.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
+
+#include "cpu_wrapper.h"
+
+namespace webrtc {
+class CpuWrapperMac : public CpuWrapper
+{
+public:
+ CpuWrapperMac();
+ virtual ~CpuWrapperMac();
+
+ virtual WebRtc_Word32 CpuUsage();
+ virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
+ WebRtc_UWord32 /*length*/) {return -1;}
+ virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return -1;}
+
+ // Note: this class will block the call and sleep if called too fast
+ // This function blocks the calling thread if the thread is calling it more
+ // often than every 500 ms.
+ virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
+ WebRtc_UWord32*& array);
+
+ virtual void Reset() {}
+ virtual void Stop() {}
+
+private:
+ WebRtc_Word32 Update(WebRtc_Word64 timeDiffMS);
+
+ WebRtc_UWord32 _cpuCount;
+ WebRtc_UWord32* _cpuUsage;
+ WebRtc_Word32 _totalCpuUsage;
+ WebRtc_Word64* _lastTickCount;
+ WebRtc_Word64 _lastTime;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
diff --git a/trunk/src/system_wrappers/source/cpu_measurement_harness.cc b/trunk/src/system_wrappers/source/cpu_measurement_harness.cc
new file mode 100644
index 0000000..237e776
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_measurement_harness.cc
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/cpu_wrapper.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "system_wrappers/source/cpu_measurement_harness.h"
+
+const int kCpuCheckPeriodMs = 100;
+
+namespace webrtc {
+
+CpuMeasurementHarness* CpuMeasurementHarness::Create(
+ CpuTarget* target,
+ int work_period_ms,
+ int work_iterations_per_period,
+ int duration_ms) {
+ if (target == NULL) {
+ return NULL;
+ }
+ if (work_period_ms > duration_ms) {
+ return NULL;
+ }
+ if (work_period_ms < 0) {
+ return NULL;
+ }
+ if (duration_ms < 0) {
+ return NULL;
+ }
+ if (work_iterations_per_period < 1) {
+ return NULL;
+ }
+ return new CpuMeasurementHarness(target, work_period_ms,
+ work_iterations_per_period, duration_ms);
+}
+
+CpuMeasurementHarness::CpuMeasurementHarness(CpuTarget* target,
+ int work_period_ms,
+ int work_iterations_per_period,
+ int duration_ms)
+ : cpu_target_(target),
+ work_period_ms_(work_period_ms),
+ work_iterations_per_period_(work_iterations_per_period),
+ duration_ms_(duration_ms),
+ cpu_sum_(0),
+ cpu_iterations_(0),
+ cpu_(CpuWrapper::CreateCpu()),
+ event_(EventWrapper::Create()) {
+}
+
+CpuMeasurementHarness::~CpuMeasurementHarness() {
+}
+
+bool CpuMeasurementHarness::Run() {
+ if (!WaitForCpuInit()) {
+ return false;
+ }
+ // No need for precision. Run for approximately the asked for duration.
+ // TODO(hellner): very low prio if at all, the actual duration of the test
+ // will be longer if calling DoWork() is not negligable and/or called many
+ // times. It may make sense to compensate for drift here. This will,
+ // however, only add complexity with minimal gains. Perhaps renaming the
+ // duration_ms_ to something more fuzzy is a better idea. However, the name
+ // would be very convoluted if it is to be self documenting.
+ int elapsed_time_ms = 0;
+ int last_measured_time = 0;
+ while (elapsed_time_ms < duration_ms_) {
+ if (((elapsed_time_ms - last_measured_time) / kCpuCheckPeriodMs) >= 1) {
+ last_measured_time = elapsed_time_ms;
+ Measure();
+ }
+ if (!DoWork()) {
+ return false;
+ }
+ event_->Wait(work_period_ms_);
+ elapsed_time_ms += work_period_ms_;
+ }
+ return true;
+}
+
+int CpuMeasurementHarness::AverageCpu() {
+ if (cpu_iterations_ == 0) {
+ return 0;
+ }
+ assert(cpu_sum_ >= 0);
+ assert(cpu_iterations_ >= 0);
+ return cpu_sum_ / cpu_iterations_;
+}
+
+bool CpuMeasurementHarness::WaitForCpuInit() {
+ bool cpu_usage_available = false;
+ int num_iterations = 0;
+ // Initializing the CPU measurements may take a couple of seconds on Windows.
+ // Since the initialization is lazy we need to wait until it is completed.
+ // Should not take more than 10000 ms.
+ while (!cpu_usage_available && (++num_iterations < 10000)) {
+ event_->Wait(1);
+ cpu_usage_available = cpu_->CpuUsage() != -1;
+ }
+ return cpu_usage_available;
+}
+
+void CpuMeasurementHarness::Measure() {
+ WebRtc_UWord32 num_cores = 0;
+ WebRtc_UWord32* cores = NULL;
+ // Return the average CPU for now.
+ cpu_sum_ = cpu_->CpuUsageMultiCore(num_cores, cores);
+ ++cpu_iterations_;
+}
+
+bool CpuMeasurementHarness::DoWork() {
+ for (int i = 0; i < work_iterations_per_period_; ++i) {
+ if (!cpu_target_->DoWork()) {
+ return false;
+ }
+ }
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_measurement_harness.h b/trunk/src/system_wrappers/source/cpu_measurement_harness.h
new file mode 100644
index 0000000..3b87f27
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_measurement_harness.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
+#define SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
+
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CpuWrapper;
+class EventWrapper;
+class ThreadWrapper;
+
+// This abstract class provides an interface that should be passed to
+// CpuMeasurementHarness. CpuMeasurementHarness will call it with the
+// frequency requested and measure the CPU usage for all calls.
+class CpuTarget {
+ public:
+ // Callback function for which the CPU usage should be calculated.
+ virtual bool DoWork() = 0;
+
+ protected:
+ CpuTarget() {}
+ virtual ~CpuTarget() {}
+};
+
+class CpuMeasurementHarness {
+ public:
+ static CpuMeasurementHarness* Create(CpuTarget* target,
+ int work_period_ms,
+ int work_iterations_per_period,
+ int duration_ms);
+ ~CpuMeasurementHarness();
+ bool Run();
+ int AverageCpu();
+
+ protected:
+ CpuMeasurementHarness(CpuTarget* target, int work_period_ms,
+ int work_iterations_per_period, int duration_ms);
+
+ private:
+ bool WaitForCpuInit();
+ void Measure();
+ bool DoWork();
+
+ CpuTarget* cpu_target_;
+ const int work_period_ms_;
+ const int work_iterations_per_period_;
+ const int duration_ms_;
+ int cpu_sum_;
+ int cpu_iterations_;
+ scoped_ptr<CpuWrapper> cpu_;
+ scoped_ptr<EventWrapper> event_;
+};
+
+} // namespace webrtc
+
+#endif // SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
diff --git a/trunk/src/system_wrappers/source/cpu_no_op.cc b/trunk/src/system_wrappers/source/cpu_no_op.cc
new file mode 100644
index 0000000..e42ef91
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_no_op.cc
@@ -0,0 +1,22 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include "cpu_wrapper.h"
+
+namespace webrtc {
+
+CpuWrapper* CpuWrapper::CreateCpu()
+{
+ return NULL;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_win.cc b/trunk/src/system_wrappers/source/cpu_win.cc
new file mode 100644
index 0000000..5792023
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_win.cc
@@ -0,0 +1,534 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_win.h"
+
+#define _WIN32_DCOM
+
+#include <assert.h>
+#include <iostream>
+#include <Wbemidl.h>
+
+#pragma comment(lib, "wbemuuid.lib")
+
+#include "condition_variable_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+
+namespace webrtc {
+WebRtc_Word32 CpuWindows::CpuUsage()
+{
+ if (!has_initialized_)
+ {
+ return -1;
+ }
+ // Last element is the average
+ return cpu_usage_[number_of_objects_ - 1];
+}
+
+WebRtc_Word32 CpuWindows::CpuUsageMultiCore(WebRtc_UWord32& num_cores,
+ WebRtc_UWord32*& cpu_usage)
+{
+ if (has_terminated_) {
+ num_cores = 0;
+ cpu_usage = NULL;
+ return -1;
+ }
+ if (!has_initialized_)
+ {
+ num_cores = 0;
+ cpu_usage = NULL;
+ return -1;
+ }
+ num_cores = number_of_objects_ - 1;
+ cpu_usage = cpu_usage_;
+ return cpu_usage_[number_of_objects_-1];
+}
+
+CpuWindows::CpuWindows()
+ : cpu_polling_thread(NULL),
+ initialize_(true),
+ has_initialized_(false),
+ terminate_(false),
+ has_terminated_(false),
+ cpu_usage_(NULL),
+ wbem_enum_access_(NULL),
+ number_of_objects_(0),
+ cpu_usage_handle_(0),
+ previous_processor_timestamp_(NULL),
+ timestamp_sys_100_ns_handle_(0),
+ previous_100ns_timestamp_(NULL),
+ wbem_service_(NULL),
+ wbem_service_proxy_(NULL),
+ wbem_refresher_(NULL),
+ wbem_enum_(NULL)
+{
+ // All resources are allocated in PollingCpu().
+ if (AllocateComplexDataTypes())
+ {
+ StartPollingCpu();
+ }
+ else
+ {
+ assert(false);
+ }
+}
+
+CpuWindows::~CpuWindows()
+{
+ // All resources are reclaimed in StopPollingCpu().
+ const bool success = StopPollingCpu();
+ assert(success);
+ DeAllocateComplexDataTypes();
+}
+
+bool CpuWindows::AllocateComplexDataTypes()
+{
+ cpu_polling_thread = ThreadWrapper::CreateThread(
+ CpuWindows::Process,
+ reinterpret_cast<void*>(this),
+ kNormalPriority,
+ "CpuWindows");
+ init_crit_ = CriticalSectionWrapper::CreateCriticalSection();
+ init_cond_ = ConditionVariableWrapper::CreateConditionVariable();
+ terminate_crit_ = CriticalSectionWrapper::CreateCriticalSection();
+ terminate_cond_ = ConditionVariableWrapper::CreateConditionVariable();
+ sleep_event = EventWrapper::Create();
+ return (cpu_polling_thread != NULL) && (init_crit_ != NULL) &&
+ (init_cond_ != NULL) && (terminate_crit_ != NULL) &&
+ (terminate_cond_ != NULL) && (sleep_event != NULL);
+}
+
+void CpuWindows::DeAllocateComplexDataTypes()
+{
+ if (sleep_event != NULL)
+ {
+ delete sleep_event;
+ sleep_event = NULL;
+ }
+ if (terminate_cond_ != NULL)
+ {
+ delete terminate_cond_;
+ terminate_cond_ = NULL;
+ }
+ if (terminate_crit_ != NULL)
+ {
+ delete terminate_crit_;
+ terminate_crit_ = NULL;
+ }
+ if (init_cond_ != NULL)
+ {
+ delete init_cond_;
+ init_cond_ = NULL;
+ }
+ if (init_crit_ != NULL)
+ {
+ delete init_crit_;
+ init_crit_ = NULL;
+ }
+ if (cpu_polling_thread != NULL)
+ {
+ delete cpu_polling_thread;
+ cpu_polling_thread = NULL;
+ }
+}
+
+void CpuWindows::StartPollingCpu()
+{
+ unsigned int dummy_id = 0;
+ if (!cpu_polling_thread->Start(dummy_id))
+ {
+ initialize_ = false;
+ has_terminated_ = true;
+ assert(false);
+ }
+}
+
+bool CpuWindows::StopPollingCpu()
+{
+ {
+ // If StopPollingCpu is called immediately after StartPollingCpu() it is
+ // possible that cpu_polling_thread is in the process of initializing.
+ // Let initialization finish to avoid getting into a bad state.
+ CriticalSectionScoped cs(init_crit_);
+ while(initialize_)
+ {
+ init_cond_->SleepCS(*init_crit_);
+ }
+ }
+
+ CriticalSectionScoped cs(terminate_crit_);
+ terminate_ = true;
+ sleep_event->Set();
+ while (!has_terminated_)
+ {
+ terminate_cond_->SleepCS(*terminate_crit_);
+ }
+ cpu_polling_thread->Stop();
+ delete cpu_polling_thread;
+ cpu_polling_thread = NULL;
+ return true;
+}
+
+bool CpuWindows::Process(void* thread_object)
+{
+ return reinterpret_cast<CpuWindows*>(thread_object)->ProcessImpl();
+}
+
+bool CpuWindows::ProcessImpl()
+{
+ {
+ CriticalSectionScoped cs(terminate_crit_);
+ if (terminate_)
+ {
+ const bool success = Terminate();
+ assert(success);
+ terminate_cond_->WakeAll();
+ return false;
+ }
+ }
+ // Initialize on first iteration
+ if (initialize_)
+ {
+ CriticalSectionScoped cs(init_crit_);
+ initialize_ = false;
+ const bool success = Initialize();
+ init_cond_->WakeAll();
+ if (!success || !has_initialized_)
+ {
+ has_initialized_ = false;
+ terminate_ = true;
+ return true;
+ }
+ }
+ // Approximately one seconds sleep for each CPU measurement. Precision is
+ // not important. 1 second refresh rate is also used by Performance Monitor
+ // (perfmon).
+ if(kEventTimeout != sleep_event->Wait(1000))
+ {
+ // Terminating. No need to update CPU usage.
+ assert(terminate_);
+ return true;
+ }
+
+ // UpdateCpuUsage() returns false if a single (or more) CPU read(s) failed.
+ // Not a major problem if it happens but make sure it doesnt trigger in
+ // debug.
+ const bool success = UpdateCpuUsage();
+ assert(success);
+ return true;
+}
+
+bool CpuWindows::CreateWmiConnection()
+{
+ IWbemLocator* service_locator = NULL;
+ HRESULT hr = CoCreateInstance(CLSID_WbemLocator, NULL,
+ CLSCTX_INPROC_SERVER, IID_IWbemLocator,
+ reinterpret_cast<void**> (&service_locator));
+ if (FAILED(hr))
+ {
+ return false;
+ }
+ // To get the WMI service specify the WMI namespace.
+ BSTR wmi_namespace = SysAllocString(L"\\\\.\\root\\cimv2");
+ if (wmi_namespace == NULL)
+ {
+ // This type of failure signifies running out of memory.
+ service_locator->Release();
+ return false;
+ }
+ hr = service_locator->ConnectServer(wmi_namespace, NULL, NULL, NULL, 0L,
+ NULL, NULL, &wbem_service_);
+ SysFreeString(wmi_namespace);
+ service_locator->Release();
+ return !FAILED(hr);
+}
+
+// Sets up WMI refresher and enum
+bool CpuWindows::CreatePerfOsRefresher()
+{
+ // Create refresher.
+ HRESULT hr = CoCreateInstance(CLSID_WbemRefresher, NULL,
+ CLSCTX_INPROC_SERVER, IID_IWbemRefresher,
+ reinterpret_cast<void**> (&wbem_refresher_));
+ if (FAILED(hr))
+ {
+ return false;
+ }
+ // Create PerfOS_Processor enum.
+ IWbemConfigureRefresher* wbem_refresher_config = NULL;
+ hr = wbem_refresher_->QueryInterface(
+ IID_IWbemConfigureRefresher,
+ reinterpret_cast<void**> (&wbem_refresher_config));
+ if (FAILED(hr))
+ {
+ return false;
+ }
+
+ // Create a proxy to the IWbemServices so that a local authentication
+ // can be set up (this is needed to be able to successfully call
+ // IWbemConfigureRefresher::AddEnum). Setting authentication with
+ // CoInitializeSecurity is process-wide (which is too intrusive).
+ hr = CoCopyProxy(static_cast<IUnknown*> (wbem_service_),
+ reinterpret_cast<IUnknown**> (&wbem_service_proxy_));
+ if(FAILED(hr))
+ {
+ return false;
+ }
+ // Set local authentication.
+ // RPC_C_AUTHN_WINNT means using NTLM instead of Kerberos which is default.
+ hr = CoSetProxyBlanket(static_cast<IUnknown*> (wbem_service_proxy_),
+ RPC_C_AUTHN_WINNT, RPC_C_AUTHZ_NONE, NULL,
+ RPC_C_AUTHN_LEVEL_DEFAULT,
+ RPC_C_IMP_LEVEL_IMPERSONATE, NULL, EOAC_NONE);
+ if(FAILED(hr))
+ {
+ return false;
+ }
+
+ // Don't care about the particular id for the enum.
+ long enum_id = 0;
+ hr = wbem_refresher_config->AddEnum(wbem_service_proxy_,
+ L"Win32_PerfRawData_PerfOS_Processor",
+ 0, NULL, &wbem_enum_, &enum_id);
+ wbem_refresher_config->Release();
+ wbem_refresher_config = NULL;
+ return !FAILED(hr);
+}
+
+// Have to pull the first round of data to be able set the handles.
+bool CpuWindows::CreatePerfOsCpuHandles()
+{
+ // Update the refresher so that there is data available in wbem_enum_.
+ wbem_refresher_->Refresh(0L);
+
+ // The number of enumerators is the number of processor + 1 (the total).
+ // This is unknown at this point.
+ DWORD number_returned = 0;
+ HRESULT hr = wbem_enum_->GetObjects(0L, number_of_objects_,
+ wbem_enum_access_, &number_returned);
+ // number_returned indicates the number of enumerators that are needed.
+ if (hr == WBEM_E_BUFFER_TOO_SMALL &&
+ number_returned > number_of_objects_)
+ {
+ // Allocate the number IWbemObjectAccess asked for by the
+ // GetObjects(..) function.
+ wbem_enum_access_ = new IWbemObjectAccess*[number_returned];
+ cpu_usage_ = new WebRtc_UWord32[number_returned];
+ previous_processor_timestamp_ = new unsigned __int64[number_returned];
+ previous_100ns_timestamp_ = new unsigned __int64[number_returned];
+ if ((wbem_enum_access_ == NULL) || (cpu_usage_ == NULL) ||
+ (previous_processor_timestamp_ == NULL) ||
+ (previous_100ns_timestamp_ == NULL))
+ {
+ // Out of memory.
+ return false;
+ }
+
+ SecureZeroMemory(wbem_enum_access_, number_returned *
+ sizeof(IWbemObjectAccess*));
+ memset(cpu_usage_, 0, sizeof(int) * number_returned);
+ memset(previous_processor_timestamp_, 0, sizeof(unsigned __int64) *
+ number_returned);
+ memset(previous_100ns_timestamp_, 0, sizeof(unsigned __int64) *
+ number_returned);
+
+ number_of_objects_ = number_returned;
+ // Read should be successfull now that memory has been allocated.
+ hr = wbem_enum_->GetObjects(0L, number_of_objects_, wbem_enum_access_,
+ &number_returned);
+ if (FAILED(hr))
+ {
+ return false;
+ }
+ }
+ else
+ {
+ // 0 enumerators should not be enough. Something has gone wrong here.
+ return false;
+ }
+
+ // Get the enumerator handles that are needed for calculating CPU usage.
+ CIMTYPE cpu_usage_type;
+ hr = wbem_enum_access_[0]->GetPropertyHandle(L"PercentProcessorTime",
+ &cpu_usage_type,
+ &cpu_usage_handle_);
+ if (FAILED(hr))
+ {
+ return false;
+ }
+ CIMTYPE timestamp_sys_100_ns_type;
+ hr = wbem_enum_access_[0]->GetPropertyHandle(L"TimeStamp_Sys100NS",
+ ×tamp_sys_100_ns_type,
+ ×tamp_sys_100_ns_handle_);
+ return !FAILED(hr);
+}
+
+bool CpuWindows::Initialize()
+{
+ if (terminate_)
+ {
+ return false;
+ }
+ // Initialize COM library.
+ HRESULT hr = CoInitializeEx(NULL,COINIT_MULTITHREADED);
+ if (FAILED(hr))
+ {
+ return false;
+ }
+ if (FAILED(hr))
+ {
+ return false;
+ }
+
+ if (!CreateWmiConnection())
+ {
+ return false;
+ }
+ if (!CreatePerfOsRefresher())
+ {
+ return false;
+ }
+ if (!CreatePerfOsCpuHandles())
+ {
+ return false;
+ }
+ has_initialized_ = true;
+ return true;
+}
+
+bool CpuWindows::Terminate()
+{
+ if (has_terminated_)
+ {
+ return false;
+ }
+ // Reverse order of Initialize().
+ // Some compilers complain about deleting NULL though it's well defined
+ if (previous_100ns_timestamp_ != NULL)
+ {
+ delete[] previous_100ns_timestamp_;
+ previous_100ns_timestamp_ = NULL;
+ }
+ if (previous_processor_timestamp_ != NULL)
+ {
+ delete[] previous_processor_timestamp_;
+ previous_processor_timestamp_ = NULL;
+ }
+ if (cpu_usage_ != NULL)
+ {
+ delete[] cpu_usage_;
+ cpu_usage_ = NULL;
+ }
+ if (wbem_enum_access_ != NULL)
+ {
+ for (DWORD i = 0; i < number_of_objects_; i++)
+ {
+ if(wbem_enum_access_[i] != NULL)
+ {
+ wbem_enum_access_[i]->Release();
+ }
+ }
+ delete[] wbem_enum_access_;
+ wbem_enum_access_ = NULL;
+ }
+ if (wbem_enum_ != NULL)
+ {
+ wbem_enum_->Release();
+ wbem_enum_ = NULL;
+ }
+ if (wbem_refresher_ != NULL)
+ {
+ wbem_refresher_->Release();
+ wbem_refresher_ = NULL;
+ }
+ if (wbem_service_proxy_ != NULL)
+ {
+ wbem_service_proxy_->Release();
+ wbem_service_proxy_ = NULL;
+ }
+ if (wbem_service_ != NULL)
+ {
+ wbem_service_->Release();
+ wbem_service_ = NULL;
+ }
+ // CoUninitialized should be called once for every CoInitializeEx.
+ // Regardless if it failed or not.
+ CoUninitialize();
+ has_terminated_ = true;
+ return true;
+}
+
+bool CpuWindows::UpdateCpuUsage()
+{
+ wbem_refresher_->Refresh(0L);
+ DWORD number_returned = 0;
+ HRESULT hr = wbem_enum_->GetObjects(0L, number_of_objects_,
+ wbem_enum_access_,&number_returned);
+ if (FAILED(hr))
+ {
+ // wbem_enum_access_ has already been allocated. Unless the number of
+ // CPUs change runtime this should not happen.
+ return false;
+ }
+ unsigned __int64 cpu_usage = 0;
+ unsigned __int64 timestamp_100ns = 0;
+ bool returnValue = true;
+ for (DWORD i = 0; i < number_returned; i++)
+ {
+ hr = wbem_enum_access_[i]->ReadQWORD(cpu_usage_handle_,&cpu_usage);
+ if (FAILED(hr))
+ {
+ returnValue = false;
+ }
+ hr = wbem_enum_access_[i]->ReadQWORD(timestamp_sys_100_ns_handle_,
+ ×tamp_100ns);
+ if (FAILED(hr))
+ {
+ returnValue = false;
+ }
+ wbem_enum_access_[i]->Release();
+ wbem_enum_access_[i] = NULL;
+
+ const bool wrapparound =
+ (previous_processor_timestamp_[i] > cpu_usage) ||
+ (previous_100ns_timestamp_[i] > timestamp_100ns);
+ const bool first_time = (previous_processor_timestamp_[i] == 0) ||
+ (previous_100ns_timestamp_[i] == 0);
+ if (wrapparound || first_time)
+ {
+ previous_processor_timestamp_[i] = cpu_usage;
+ previous_100ns_timestamp_[i] = timestamp_100ns;
+ continue;
+ }
+ const unsigned __int64 processor_timestamp_delta =
+ cpu_usage - previous_processor_timestamp_[i];
+ const unsigned __int64 timestamp_100ns_delta =
+ timestamp_100ns - previous_100ns_timestamp_[i];
+
+ if (processor_timestamp_delta >= timestamp_100ns_delta)
+ {
+ cpu_usage_[i] = 0;
+ } else {
+ // Quotient must be float since the division is guaranteed to yield
+ // a value between 0 and 1 which is 0 in integer division.
+ const float delta_quotient =
+ static_cast<float>(processor_timestamp_delta) /
+ static_cast<float>(timestamp_100ns_delta);
+ cpu_usage_[i] = 100 - static_cast<WebRtc_UWord32>(delta_quotient *
+ 100);
+ }
+ previous_processor_timestamp_[i] = cpu_usage;
+ previous_100ns_timestamp_[i] = timestamp_100ns;
+ }
+ return returnValue;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_win.h b/trunk/src/system_wrappers/source/cpu_win.h
new file mode 100644
index 0000000..d15073c
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_win.h
@@ -0,0 +1,103 @@
+// This file contains a Windows implementation of CpuWrapper.
+// Note: Windows XP, Windows Server 2003 are the minimum requirements.
+// The requirements are due to the implementation being based on
+// WMI.
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
+
+#include "cpu_wrapper.h"
+
+#include <Wbemidl.h>
+
+namespace webrtc {
+class ConditionVariableWrapper;
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+
+class CpuWindows : public CpuWrapper
+{
+public:
+ virtual WebRtc_Word32 CpuUsage();
+ virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
+ WebRtc_UWord32 /*length*/) {return -1;}
+ virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return -1;}
+
+ virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& num_cores,
+ WebRtc_UWord32*& cpu_usage);
+
+ virtual void Reset() {}
+ virtual void Stop() {}
+
+ CpuWindows();
+ virtual ~CpuWindows();
+private:
+ bool AllocateComplexDataTypes();
+ void DeAllocateComplexDataTypes();
+
+ void StartPollingCpu();
+ bool StopPollingCpu();
+
+ static bool Process(void* thread_object);
+ bool ProcessImpl();
+
+ bool CreateWmiConnection();
+ bool CreatePerfOsRefresher();
+ bool CreatePerfOsCpuHandles();
+ bool Initialize();
+ bool Terminate();
+
+ bool UpdateCpuUsage();
+
+ ThreadWrapper* cpu_polling_thread;
+
+ bool initialize_;
+ bool has_initialized_;
+ CriticalSectionWrapper* init_crit_;
+ ConditionVariableWrapper* init_cond_;
+
+ bool terminate_;
+ bool has_terminated_;
+ CriticalSectionWrapper* terminate_crit_;
+ ConditionVariableWrapper* terminate_cond_;
+
+ // For sleep with wake-up functionality.
+ EventWrapper* sleep_event;
+
+ // Will be an array. Just care about CPU 0 for now.
+ WebRtc_UWord32* cpu_usage_;
+
+ // One IWbemObjectAccess for each processor and one for the total.
+ // 0-n-1 is the individual processors.
+ // n is the total.
+ IWbemObjectAccess** wbem_enum_access_;
+ DWORD number_of_objects_;
+
+ // Cpu timestamp
+ long cpu_usage_handle_;
+ unsigned __int64* previous_processor_timestamp_;
+
+ // Timestamp
+ long timestamp_sys_100_ns_handle_;
+ unsigned __int64* previous_100ns_timestamp_;
+
+ IWbemServices* wbem_service_;
+ IWbemServices* wbem_service_proxy_;
+
+ IWbemRefresher* wbem_refresher_;
+
+ IWbemHiPerfEnum* wbem_enum_;
+
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
diff --git a/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc b/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc
new file mode 100644
index 0000000..cd149dc
--- /dev/null
+++ b/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/cpu_wrapper.h"
+
+#include "gtest/gtest.h"
+#include "system_wrappers/interface/cpu_info.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "system_wrappers/interface/trace.h"
+#include "testsupport/fileutils.h"
+
+using webrtc::CpuInfo;
+using webrtc::CpuWrapper;
+using webrtc::EventWrapper;
+using webrtc::scoped_ptr;
+using webrtc::Trace;
+
+// This test is flaky on Windows/Release.
+// http://code.google.com/p/webrtc/issues/detail?id=290
+#ifdef _WIN32
+#define MAYBE_Usage DISABLED_Usage
+#else
+#define MAYBE_Usage Usage
+#endif
+TEST(CpuWrapperTest, MAYBE_Usage) {
+ Trace::CreateTrace();
+ std::string trace_file = webrtc::test::OutputPath() +
+ "cpu_wrapper_unittest.txt";
+ Trace::SetTraceFile(trace_file.c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ printf("Number of cores detected:%u\n", CpuInfo::DetectNumberOfCores());
+ scoped_ptr<CpuWrapper> cpu(CpuWrapper::CreateCpu());
+ ASSERT_TRUE(cpu.get() != NULL);
+ scoped_ptr<EventWrapper> sleep_event(EventWrapper::Create());
+ ASSERT_TRUE(sleep_event.get() != NULL);
+
+ int num_iterations = 0;
+ WebRtc_UWord32 num_cores = 0;
+ WebRtc_UWord32* cores = NULL;
+ bool cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
+ // Initializing the CPU measurements may take a couple of seconds on Windows.
+ // Since the initialization is lazy we need to wait until it is completed.
+ // Should not take more than 10000 ms.
+ while (!cpu_usage_available && (++num_iterations < 10000)) {
+ if (cores != NULL) {
+ ASSERT_GT(num_cores, 0u);
+ break;
+ }
+ sleep_event->Wait(1);
+ cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
+ }
+ ASSERT_TRUE(cpu_usage_available);
+
+ const WebRtc_Word32 average = cpu->CpuUsageMultiCore(num_cores, cores);
+ ASSERT_TRUE(cores != NULL);
+ EXPECT_GT(num_cores, 0u);
+ EXPECT_GE(average, 0);
+ EXPECT_LE(average, 100);
+
+ printf("\nNumber of cores:%d\n", num_cores);
+ printf("Average cpu:%d\n", average);
+ for (WebRtc_UWord32 i = 0; i < num_cores; i++) {
+ printf("Core:%u CPU:%u \n", i, cores[i]);
+ EXPECT_GE(cores[i], 0u);
+ EXPECT_LE(cores[i], 100u);
+ }
+
+ Trace::ReturnTrace();
+};
diff --git a/trunk/src/system_wrappers/source/critical_section.cc b/trunk/src/system_wrappers/source/critical_section.cc
new file mode 100644
index 0000000..d3f3f01
--- /dev/null
+++ b/trunk/src/system_wrappers/source/critical_section.cc
@@ -0,0 +1,27 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(_WIN32)
+ #include <windows.h>
+ #include "critical_section_win.h"
+#else
+ #include "critical_section_posix.h"
+#endif
+
+namespace webrtc {
+CriticalSectionWrapper* CriticalSectionWrapper::CreateCriticalSection()
+{
+#ifdef _WIN32
+ return new CriticalSectionWindows();
+#else
+ return new CriticalSectionPosix();
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_posix.cc b/trunk/src/system_wrappers/source/critical_section_posix.cc
new file mode 100644
index 0000000..70f85f9
--- /dev/null
+++ b/trunk/src/system_wrappers/source/critical_section_posix.cc
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// General note: return values for the various pthread synchronization APIs
+// are explicitly ignored here. In Chromium, the same thing is done for release.
+// However, in debugging, failure in these APIs are logged. There is currently
+// no equivalent to DCHECK_EQ in WebRTC code so this is the best we can do here.
+// TODO(henrike): add logging when pthread synchronization APIs are failing.
+
+#include "critical_section_posix.h"
+
+namespace webrtc {
+
+CriticalSectionPosix::CriticalSectionPosix()
+{
+ pthread_mutexattr_t attr;
+ (void) pthread_mutexattr_init(&attr);
+ (void) pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
+ (void) pthread_mutex_init(&_mutex, &attr);
+}
+
+CriticalSectionPosix::~CriticalSectionPosix()
+{
+ (void) pthread_mutex_destroy(&_mutex);
+}
+
+void
+CriticalSectionPosix::Enter()
+{
+ (void) pthread_mutex_lock(&_mutex);
+}
+
+void
+CriticalSectionPosix::Leave()
+{
+ (void) pthread_mutex_unlock(&_mutex);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_posix.h b/trunk/src/system_wrappers/source/critical_section_posix.h
new file mode 100644
index 0000000..40b7dc9
--- /dev/null
+++ b/trunk/src/system_wrappers/source/critical_section_posix.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
+
+#include "critical_section_wrapper.h"
+
+#include <pthread.h>
+
+namespace webrtc {
+class CriticalSectionPosix : public CriticalSectionWrapper
+{
+public:
+ CriticalSectionPosix();
+
+ virtual ~CriticalSectionPosix();
+
+ virtual void Enter();
+ virtual void Leave();
+
+private:
+ pthread_mutex_t _mutex;
+ friend class ConditionVariablePosix;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/critical_section_win.cc b/trunk/src/system_wrappers/source/critical_section_win.cc
new file mode 100644
index 0000000..bbc66e5
--- /dev/null
+++ b/trunk/src/system_wrappers/source/critical_section_win.cc
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "critical_section_win.h"
+
+namespace webrtc {
+CriticalSectionWindows::CriticalSectionWindows()
+{
+ InitializeCriticalSection(&crit);
+}
+
+CriticalSectionWindows::~CriticalSectionWindows()
+{
+ DeleteCriticalSection(&crit);
+}
+
+void
+CriticalSectionWindows::Enter()
+{
+ EnterCriticalSection(&crit);
+}
+
+void
+CriticalSectionWindows::Leave()
+{
+ LeaveCriticalSection(&crit);
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_win.h b/trunk/src/system_wrappers/source/critical_section_win.h
new file mode 100644
index 0000000..9556fa9
--- /dev/null
+++ b/trunk/src/system_wrappers/source/critical_section_win.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
+
+#include "typedefs.h"
+#include "critical_section_wrapper.h"
+#include <windows.h>
+
+namespace webrtc {
+class CriticalSectionWindows : public CriticalSectionWrapper
+{
+public:
+ CriticalSectionWindows();
+
+ virtual ~CriticalSectionWindows();
+
+ virtual void Enter();
+ virtual void Leave();
+
+private:
+ CRITICAL_SECTION crit;
+
+ friend class ConditionVariableWindows;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/data_log.cc b/trunk/src/system_wrappers/source/data_log.cc
new file mode 100644
index 0000000..f123896
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log.cc
@@ -0,0 +1,455 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "data_log.h"
+
+#include <assert.h>
+
+#include <algorithm>
+#include <list>
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "file_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "thread_wrapper.h"
+
+namespace webrtc {
+
+DataLogImpl::CritSectScopedPtr DataLogImpl::crit_sect_(
+ CriticalSectionWrapper::CreateCriticalSection());
+
+DataLogImpl* DataLogImpl::instance_ = NULL;
+
+// A Row contains cells, which are indexed by the column names as std::string.
+// The string index is treated in a case sensitive way.
+class Row {
+ public:
+ Row();
+ ~Row();
+
+ // Inserts a Container into the cell of the column specified with
+ // column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& column_name,
+ const Container* value_container);
+
+ // Converts the value at the column specified by column_name to a string
+ // stored in value_string.
+ // column_name is treated in a case sensitive way.
+ void ToString(const std::string& column_name, std::string* value_string);
+
+ private:
+ // Collection of containers indexed by column name as std::string
+ typedef std::map<std::string, const Container*> CellMap;
+
+ CellMap cells_;
+ CriticalSectionWrapper* cells_lock_;
+};
+
+// A LogTable contains multiple rows, where only the latest row is active for
+// editing. The rows are defined by the ColumnMap, which contains the name of
+// each column and the length of the column (1 for one-value-columns and greater
+// than 1 for multi-value-columns).
+class LogTable {
+ public:
+ LogTable();
+ ~LogTable();
+
+ // Adds the column with name column_name to the table. The column will be a
+ // multi-value-column if multi_value_length is greater than 1.
+ // column_name is treated in a case sensitive way.
+ int AddColumn(const std::string& column_name, int multi_value_length);
+
+ // Buffers the current row while it is waiting to be written to file,
+ // which is done by a call to Flush(). A new row is available when the
+ // function returns
+ void NextRow();
+
+ // Inserts a Container into the cell of the column specified with
+ // column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& column_name,
+ const Container* value_container);
+
+ // Creates a log file, named as specified in the string file_name, to
+ // where the table will be written when calling Flush().
+ int CreateLogFile(const std::string& file_name);
+
+ // Write all complete rows to file.
+ // May not be called by two threads simultaneously (doing so may result in
+ // a race condition). Will be called by the file_writer_thread_ when that
+ // thread is running.
+ void Flush();
+
+ private:
+ // Collection of multi_value_lengths indexed by column name as std::string
+ typedef std::map<std::string, int> ColumnMap;
+ typedef std::list<Row*> RowList;
+
+ ColumnMap columns_;
+ RowList rows_[2];
+ RowList* rows_history_;
+ RowList* rows_flush_;
+ Row* current_row_;
+ FileWrapper* file_;
+ bool write_header_;
+ CriticalSectionWrapper* table_lock_;
+};
+
+Row::Row()
+ : cells_(),
+ cells_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
+
+Row::~Row() {
+ for (CellMap::iterator it = cells_.begin(); it != cells_.end();) {
+ delete it->second;
+ // For maps all iterators (except the erased) are valid after an erase
+ cells_.erase(it++);
+ }
+ delete cells_lock_;
+}
+
+int Row::InsertCell(const std::string& column_name,
+ const Container* value_container) {
+ CriticalSectionScoped synchronize(cells_lock_);
+ assert(cells_.count(column_name) == 0);
+ if (cells_.count(column_name) > 0)
+ return -1;
+ cells_[column_name] = value_container;
+ return 0;
+}
+
+void Row::ToString(const std::string& column_name,
+ std::string* value_string) {
+ CriticalSectionScoped synchronize(cells_lock_);
+ const Container* container = cells_[column_name];
+ if (container == NULL) {
+ *value_string = "NaN,";
+ return;
+ }
+ container->ToString(value_string);
+}
+
+LogTable::LogTable()
+ : columns_(),
+ rows_(),
+ rows_history_(&rows_[0]),
+ rows_flush_(&rows_[1]),
+ current_row_(new Row),
+ file_(FileWrapper::Create()),
+ write_header_(true),
+ table_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
+
+LogTable::~LogTable() {
+ for (RowList::iterator row_it = rows_history_->begin();
+ row_it != rows_history_->end();) {
+ delete *row_it;
+ row_it = rows_history_->erase(row_it);
+ }
+ for (ColumnMap::iterator col_it = columns_.begin();
+ col_it != columns_.end();) {
+ // For maps all iterators (except the erased) are valid after an erase
+ columns_.erase(col_it++);
+ }
+ if (file_ != NULL) {
+ file_->Flush();
+ file_->CloseFile();
+ delete file_;
+ }
+ delete current_row_;
+ delete table_lock_;
+}
+
+int LogTable::AddColumn(const std::string& column_name,
+ int multi_value_length) {
+ assert(multi_value_length > 0);
+ if (!write_header_) {
+ // It's not allowed to add new columns after the header
+ // has been written.
+ assert(false);
+ return -1;
+ } else {
+ CriticalSectionScoped synchronize(table_lock_);
+ if (write_header_)
+ columns_[column_name] = multi_value_length;
+ else
+ return -1;
+ }
+ return 0;
+}
+
+void LogTable::NextRow() {
+ CriticalSectionScoped sync_rows(table_lock_);
+ rows_history_->push_back(current_row_);
+ current_row_ = new Row;
+}
+
+int LogTable::InsertCell(const std::string& column_name,
+ const Container* value_container) {
+ CriticalSectionScoped synchronize(table_lock_);
+ assert(columns_.count(column_name) > 0);
+ if (columns_.count(column_name) == 0)
+ return -1;
+ return current_row_->InsertCell(column_name, value_container);
+}
+
+int LogTable::CreateLogFile(const std::string& file_name) {
+ if (file_name.length() == 0)
+ return -1;
+ if (file_->Open())
+ return -1;
+ file_->OpenFile(file_name.c_str(),
+ false, // Open with read/write permissions
+ false, // Don't wraparound and write at the beginning when
+ // the file is full
+ true); // Open as a text file
+ if (file_ == NULL)
+ return -1;
+ return 0;
+}
+
+void LogTable::Flush() {
+ ColumnMap::iterator column_it;
+ bool commit_header = false;
+ if (write_header_) {
+ CriticalSectionScoped synchronize(table_lock_);
+ if (write_header_) {
+ commit_header = true;
+ write_header_ = false;
+ }
+ }
+ if (commit_header) {
+ for (column_it = columns_.begin();
+ column_it != columns_.end(); ++column_it) {
+ if (column_it->second > 1) {
+ file_->WriteText("%s[%u],", column_it->first.c_str(),
+ column_it->second);
+ for (int i = 1; i < column_it->second; ++i)
+ file_->WriteText(",");
+ } else {
+ file_->WriteText("%s,", column_it->first.c_str());
+ }
+ }
+ if (columns_.size() > 0)
+ file_->WriteText("\n");
+ }
+
+ // Swap the list used for flushing with the list containing the row history
+ // and clear the history. We also create a local pointer to the new
+ // list used for flushing to avoid race conditions if another thread
+ // calls this function while we are writing.
+ // We don't want to block the list while we're writing to file.
+ {
+ CriticalSectionScoped synchronize(table_lock_);
+ RowList* tmp = rows_flush_;
+ rows_flush_ = rows_history_;
+ rows_history_ = tmp;
+ rows_history_->clear();
+ }
+
+ // Write all complete rows to file and delete them
+ for (RowList::iterator row_it = rows_flush_->begin();
+ row_it != rows_flush_->end();) {
+ for (column_it = columns_.begin();
+ column_it != columns_.end(); ++column_it) {
+ std::string row_string;
+ (*row_it)->ToString(column_it->first, &row_string);
+ file_->WriteText("%s", row_string.c_str());
+ }
+ if (columns_.size() > 0)
+ file_->WriteText("\n");
+ delete *row_it;
+ row_it = rows_flush_->erase(row_it);
+ }
+}
+
+int DataLog::CreateLog() {
+ return DataLogImpl::CreateLog();
+}
+
+void DataLog::ReturnLog() {
+ return DataLogImpl::ReturnLog();
+}
+
+std::string DataLog::Combine(const std::string& table_name, int table_id) {
+ std::stringstream ss;
+ std::string combined_id = table_name;
+ std::string number_suffix;
+ ss << "_" << table_id;
+ ss >> number_suffix;
+ combined_id += number_suffix;
+ std::transform(combined_id.begin(), combined_id.end(), combined_id.begin(),
+ ::tolower);
+ return combined_id;
+}
+
+int DataLog::AddTable(const std::string& table_name) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->AddTable(table_name);
+}
+
+int DataLog::AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->DataLogImpl::StaticInstance()->AddColumn(table_name,
+ column_name,
+ multi_value_length);
+}
+
+int DataLog::NextRow(const std::string& table_name) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->DataLogImpl::StaticInstance()->NextRow(table_name);
+}
+
+DataLogImpl::DataLogImpl()
+ : counter_(1),
+ tables_(),
+ flush_event_(EventWrapper::Create()),
+ file_writer_thread_(NULL),
+ tables_lock_(RWLockWrapper::CreateRWLock()) {
+}
+
+DataLogImpl::~DataLogImpl() {
+ StopThread();
+ Flush(); // Write any remaining rows
+ delete file_writer_thread_;
+ delete flush_event_;
+ for (TableMap::iterator it = tables_.begin(); it != tables_.end();) {
+ delete static_cast<LogTable*>(it->second);
+ // For maps all iterators (except the erased) are valid after an erase
+ tables_.erase(it++);
+ }
+ delete tables_lock_;
+}
+
+int DataLogImpl::CreateLog() {
+ CriticalSectionScoped synchronize(crit_sect_.get());
+ if (instance_ == NULL) {
+ instance_ = new DataLogImpl();
+ return instance_->Init();
+ } else {
+ ++instance_->counter_;
+ }
+ return 0;
+}
+
+int DataLogImpl::Init() {
+ file_writer_thread_ = ThreadWrapper::CreateThread(
+ DataLogImpl::Run,
+ instance_,
+ kHighestPriority,
+ "DataLog");
+ if (file_writer_thread_ == NULL)
+ return -1;
+ unsigned int thread_id = 0;
+ bool success = file_writer_thread_->Start(thread_id);
+ if (!success)
+ return -1;
+ return 0;
+}
+
+DataLogImpl* DataLogImpl::StaticInstance() {
+ return instance_;
+}
+
+void DataLogImpl::ReturnLog() {
+ CriticalSectionScoped synchronize(crit_sect_.get());
+ if (instance_ && instance_->counter_ > 1) {
+ --instance_->counter_;
+ return;
+ }
+ delete instance_;
+ instance_ = NULL;
+}
+
+int DataLogImpl::AddTable(const std::string& table_name) {
+ WriteLockScoped synchronize(*tables_lock_);
+ // Make sure we don't add a table which already exists
+ if (tables_.count(table_name) > 0)
+ return -1;
+ tables_[table_name] = new LogTable();
+ if (tables_[table_name]->CreateLogFile(table_name + ".txt") == -1)
+ return -1;
+ return 0;
+}
+
+int DataLogImpl::AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length) {
+ ReadLockScoped synchronize(*tables_lock_);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ return tables_[table_name]->AddColumn(column_name, multi_value_length);
+}
+
+int DataLogImpl::InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const Container* value_container) {
+ ReadLockScoped synchronize(*tables_lock_);
+ assert(tables_.count(table_name) > 0);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ return tables_[table_name]->InsertCell(column_name, value_container);
+}
+
+int DataLogImpl::NextRow(const std::string& table_name) {
+ ReadLockScoped synchronize(*tables_lock_);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ tables_[table_name]->NextRow();
+ if (file_writer_thread_ == NULL) {
+ // Write every row to file as they get complete.
+ tables_[table_name]->Flush();
+ } else {
+ // Signal a complete row
+ flush_event_->Set();
+ }
+ return 0;
+}
+
+void DataLogImpl::Flush() {
+ ReadLockScoped synchronize(*tables_lock_);
+ for (TableMap::iterator it = tables_.begin(); it != tables_.end(); ++it) {
+ it->second->Flush();
+ }
+}
+
+bool DataLogImpl::Run(void* obj) {
+ static_cast<DataLogImpl*>(obj)->Process();
+ return true;
+}
+
+void DataLogImpl::Process() {
+ // Wait for a row to be complete
+ flush_event_->Wait(WEBRTC_EVENT_INFINITE);
+ Flush();
+}
+
+void DataLogImpl::StopThread() {
+ if (file_writer_thread_ != NULL) {
+ file_writer_thread_->SetNotAlive();
+ flush_event_->Set();
+ // Call Stop() repeatedly, waiting for the Flush() call in Process() to
+ // finish.
+ while (!file_writer_thread_->Stop()) continue;
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/data_log_c.cc b/trunk/src/system_wrappers/source/data_log_c.cc
new file mode 100644
index 0000000..f8d7efd
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_c.cc
@@ -0,0 +1,145 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is the pure C wrapper of the DataLog class.
+ */
+
+#include "system_wrappers/interface/data_log_c.h"
+
+#include <string>
+
+#include "system_wrappers/interface/data_log.h"
+
+extern "C" int WebRtcDataLog_CreateLog() {
+ return webrtc::DataLog::CreateLog();
+}
+
+extern "C" void WebRtcDataLog_ReturnLog() {
+ return webrtc::DataLog::ReturnLog();
+}
+
+extern "C" char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
+ const char* table_name, int table_id) {
+ if (!table_name) return NULL;
+ std::string combined = webrtc::DataLog::Combine(table_name, table_id);
+ if (combined.size() >= combined_len) return NULL;
+ std::copy(combined.begin(), combined.end(), combined_name);
+ combined_name[combined.size()] = '\0';
+ return combined_name;
+}
+
+extern "C" int WebRtcDataLog_AddTable(const char* table_name) {
+ if (!table_name) return -1;
+ return webrtc::DataLog::AddTable(table_name);
+}
+
+extern "C" int WebRtcDataLog_AddColumn(const char* table_name,
+ const char* column_name,
+ int multi_value_length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::AddColumn(table_name, column_name,
+ multi_value_length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int(const char* table_name,
+ const char* column_name,
+ int value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int(const char* table_name,
+ const char* column_name,
+ const int* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_float(const char* table_name,
+ const char* column_name,
+ float value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_float(const char* table_name,
+ const char* column_name,
+ const float* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_double(const char* table_name,
+ const char* column_name,
+ double value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_double(const char* table_name,
+ const char* column_name,
+ const double* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int32(const char* table_name,
+ const char* column_name,
+ int32_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int32(const char* table_name,
+ const char* column_name,
+ const int32_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_uint32(const char* table_name,
+ const char* column_name,
+ uint32_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_uint32(const char* table_name,
+ const char* column_name,
+ const uint32_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int64(const char* table_name,
+ const char* column_name,
+ int64_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int64(const char* table_name,
+ const char* column_name,
+ const int64_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_NextRow(const char* table_name) {
+ if (!table_name) return -1;
+ return webrtc::DataLog::NextRow(table_name);
+}
diff --git a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c
new file mode 100644
index 0000000..e78a0e3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/source/data_log_c_helpers_unittest.h"
+
+#include <assert.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "system_wrappers/interface/data_log_c.h"
+
+enum { kTestArrayLen = 4 };
+static const char kTableName[] = "c_wrapper_table";
+static const char kColumnName1[] = "Scalar";
+static const char kColumnName2[] = "Vector";
+
+int WebRtcDataLogCHelper_TestCreateLog() {
+ return WebRtcDataLog_CreateLog();
+}
+
+int WebRtcDataLogCHelper_TestReturnLog() {
+ WebRtcDataLog_ReturnLog();
+ return 0;
+}
+
+int WebRtcDataLogCHelper_TestCombine() {
+ const int kOutLen = strlen(kTableName) + 4; /* Room for "_17" + '\0' */
+ char* combined_name = malloc(kOutLen * sizeof(char));
+ char* out_ptr = WebRtcDataLog_Combine(combined_name, kOutLen, kTableName, 17);
+ int return_code = 0;
+ if (!out_ptr) {
+ return_code = -1;
+ }
+ if (strcmp(combined_name, "c_wrapper_table_17") != 0) {
+ return_code = -2;
+ }
+ free(combined_name);
+ return return_code;
+}
+
+int WebRtcDataLogCHelper_TestAddTable() {
+ return WebRtcDataLog_AddTable(kTableName);
+}
+
+int WebRtcDataLogCHelper_TestAddColumn() {
+ if (WebRtcDataLog_AddColumn(kTableName, kColumnName1, 1) != 0) {
+ return -1;
+ }
+ if (WebRtcDataLog_AddColumn(kTableName, kColumnName2, kTestArrayLen) != 0) {
+ return -2;
+ }
+ return 0;
+}
+
+int WebRtcDataLogCHelper_TestNextRow() {
+ return WebRtcDataLog_NextRow(kTableName);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int() {
+ return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int() {
+ int values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_float() {
+ return WebRtcDataLog_InsertCell_float(kTableName, kColumnName1, 17.0f);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_float() {
+ float values[kTestArrayLen] = {1.0f, 2.0f, 3.0f, 4.0f};
+ return WebRtcDataLog_InsertArray_float(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_double() {
+ return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17.0);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_double() {
+ double values[kTestArrayLen] = {1.0, 2.0, 3.0, 4.0};
+ return WebRtcDataLog_InsertArray_double(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int32() {
+ return WebRtcDataLog_InsertCell_int32(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int32() {
+ int32_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int32(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_uint32() {
+ return WebRtcDataLog_InsertCell_uint32(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_uint32() {
+ uint32_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_uint32(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int64() {
+ return WebRtcDataLog_InsertCell_int64(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int64() {
+ int64_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int64(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
diff --git a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h
new file mode 100644
index 0000000..ef86eae
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
+#define SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+int WebRtcDataLogCHelper_TestCreateLog();
+
+int WebRtcDataLogCHelper_TestReturnLog();
+
+int WebRtcDataLogCHelper_TestCombine();
+
+int WebRtcDataLogCHelper_TestAddTable();
+
+int WebRtcDataLogCHelper_TestAddColumn();
+
+int WebRtcDataLogCHelper_TestNextRow();
+
+int WebRtcDataLogCHelper_TestInsertCell_int();
+
+int WebRtcDataLogCHelper_TestInsertArray_int();
+
+int WebRtcDataLogCHelper_TestInsertCell_float();
+
+int WebRtcDataLogCHelper_TestInsertArray_float();
+
+int WebRtcDataLogCHelper_TestInsertCell_double();
+
+int WebRtcDataLogCHelper_TestInsertArray_double();
+
+int WebRtcDataLogCHelper_TestInsertCell_int32();
+
+int WebRtcDataLogCHelper_TestInsertArray_int32();
+
+int WebRtcDataLogCHelper_TestInsertCell_uint32();
+
+int WebRtcDataLogCHelper_TestInsertArray_uint32();
+
+int WebRtcDataLogCHelper_TestInsertCell_int64();
+
+int WebRtcDataLogCHelper_TestInsertArray_int64();
+
+#ifdef __cplusplus
+} // end of extern "C"
+#endif
+
+#endif // SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
diff --git a/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc b/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc
new file mode 100644
index 0000000..94b4d6e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+
+#include "data_log.h"
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+TEST(TestDataLog, IntContainers) {
+ int c = 5;
+ webrtc::ValueContainer<int> v1(c);
+ c = 10;
+ webrtc::ValueContainer<int> v2(c);
+ std::string s1, s2;
+ v1.ToString(&s1);
+ v2.ToString(&s2);
+ ASSERT_EQ(s1, "5,");
+ ASSERT_EQ(s2, "10,");
+ v1 = v2;
+ v1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+}
+
+TEST(TestDataLog, DoubleContainers) {
+ double c = 3.5;
+ webrtc::ValueContainer<double> v1(c);
+ c = 10.3;
+ webrtc::ValueContainer<double> v2(c);
+ std::string s1, s2;
+ v1.ToString(&s1);
+ v2.ToString(&s2);
+ ASSERT_EQ(s1, "3.5,");
+ ASSERT_EQ(s2, "10.3,");
+ v1 = v2;
+ v1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+}
+
+TEST(TestDataLog, MultiValueContainers) {
+ int a[3] = {1, 2, 3};
+ int b[3] = {4, 5, 6};
+ webrtc::MultiValueContainer<int> m1(a, 3);
+ webrtc::MultiValueContainer<int> m2(b, 3);
+ webrtc::MultiValueContainer<int> m3(a, 3);
+ std::string s1, s2, s3;
+ m1.ToString(&s1);
+ m2.ToString(&s2);
+ ASSERT_EQ(s1, "1,2,3,");
+ ASSERT_EQ(s2, "4,5,6,");
+ m1 = m2;
+ m1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+ m3.ToString(&s3);
+ ASSERT_EQ(s3, "1,2,3,");
+}
diff --git a/trunk/src/system_wrappers/source/data_log_no_op.cc b/trunk/src/system_wrappers/source/data_log_no_op.cc
new file mode 100644
index 0000000..bedc82a
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_no_op.cc
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "data_log.h"
+
+#include <string>
+
+namespace webrtc {
+
+int DataLog::CreateLog() {
+ return 0;
+}
+
+void DataLog::ReturnLog() {
+}
+
+std::string DataLog::Combine(const std::string& table_name, int table_id) {
+ return std::string();
+}
+
+int DataLog::AddTable(const std::string& /*table_name*/) {
+ return 0;
+}
+
+int DataLog::AddColumn(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ int /*multi_value_length*/) {
+ return 0;
+}
+
+int DataLog::NextRow(const std::string& /*table_name*/) {
+ return 0;
+}
+
+DataLogImpl::DataLogImpl() {
+}
+
+DataLogImpl::~DataLogImpl() {
+}
+
+DataLogImpl* DataLogImpl::StaticInstance() {
+ return NULL;
+}
+
+void DataLogImpl::ReturnLog() {
+}
+
+int DataLogImpl::AddTable(const std::string& /*table_name*/) {
+ return 0;
+}
+
+int DataLogImpl::AddColumn(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ int /*multi_value_length*/) {
+ return 0;
+}
+
+int DataLogImpl::InsertCell(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ const Container* /*value_container*/) {
+ return 0;
+}
+
+int DataLogImpl::NextRow(const std::string& /*table_name*/) {
+ return 0;
+}
+
+void DataLogImpl::Flush() {
+}
+
+bool DataLogImpl::Run(void* /*obj*/) {
+ return true;
+}
+
+void DataLogImpl::Process() {
+}
+
+void DataLogImpl::StopThread() {
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/data_log_unittest.cc b/trunk/src/system_wrappers/source/data_log_unittest.cc
new file mode 100644
index 0000000..c64ed94
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_unittest.cc
@@ -0,0 +1,310 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <map>
+#include <string>
+
+#include "system_wrappers/interface/data_log.h"
+#include "system_wrappers/interface/data_log_c.h"
+#include "system_wrappers/source/data_log_c_helpers_unittest.h"
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+// A class for storing the values expected from a log table column when
+// verifying a log table file.
+struct ExpectedValues {
+ public:
+ ExpectedValues()
+ : values(NULL),
+ multi_value_length(1) {
+ }
+
+ ExpectedValues(std::vector<std::string> expected_values,
+ int expected_multi_value_length)
+ : values(expected_values),
+ multi_value_length(expected_multi_value_length) {
+ }
+
+ std::vector<std::string> values;
+ int multi_value_length;
+};
+
+typedef std::map<std::string, ExpectedValues> ExpectedValuesMap;
+
+// A static class used for parsing and verifying data log files.
+class DataLogParser {
+ public:
+ // Verifies that the log table stored in the file "log_file" corresponds to
+ // the cells and columns specified in "columns".
+ static int VerifyTable(FILE* log_file, const ExpectedValuesMap& columns) {
+ int row = 0;
+ char line_buffer[kMaxLineLength];
+ char* ret = fgets(line_buffer, kMaxLineLength, log_file);
+ EXPECT_FALSE(ret == NULL);
+ if (ret == NULL)
+ return -1;
+
+ std::string line(line_buffer, kMaxLineLength);
+ VerifyHeader(line, columns);
+ while (fgets(line_buffer, kMaxLineLength, log_file) != NULL) {
+ line = std::string(line_buffer, kMaxLineLength);
+ size_t line_position = 0;
+
+ for (ExpectedValuesMap::const_iterator it = columns.begin();
+ it != columns.end(); ++it) {
+ std::string str = ParseElement(line, &line_position,
+ it->second.multi_value_length);
+ EXPECT_EQ(str, it->second.values[row]);
+ if (str != it->second.values[row])
+ return -1;
+ }
+ ++row;
+ }
+ return 0;
+ }
+
+ // Verifies the table header stored in "line" to correspond with the header
+ // specified in "columns".
+ static int VerifyHeader(const std::string& line,
+ const ExpectedValuesMap& columns) {
+ size_t line_position = 0;
+ for (ExpectedValuesMap::const_iterator it = columns.begin();
+ it != columns.end(); ++it) {
+ std::string str = ParseElement(line, &line_position,
+ it->second.multi_value_length);
+ EXPECT_EQ(str, it->first);
+ if (str != it->first)
+ return -1;
+ }
+ return 0;
+ }
+
+ // Parses out and returns one element from the string "line", which contains
+ // one line read from a log table file. An element can either be a column
+ // header or a cell of a row.
+ static std::string ParseElement(const std::string& line,
+ size_t* line_position,
+ int multi_value_length) {
+ std::string parsed_cell;
+ parsed_cell = "";
+ for (int i = 0; i < multi_value_length; ++i) {
+ size_t next_separator = line.find(',', *line_position);
+ EXPECT_NE(next_separator, std::string::npos);
+ if (next_separator == std::string::npos)
+ break;
+ parsed_cell += line.substr(*line_position,
+ next_separator - *line_position + 1);
+ *line_position = next_separator + 1;
+ }
+ return parsed_cell;
+ }
+
+ // This constant defines the maximum line length the DataLogParser can
+ // parse.
+ enum { kMaxLineLength = 100 };
+};
+
+TEST(TestDataLog, CreateReturnTest) {
+ for (int i = 0; i < 10; ++i)
+ ASSERT_EQ(DataLog::CreateLog(), 0);
+ ASSERT_EQ(DataLog::AddTable(DataLog::Combine("a proper table", 1)), 0);
+ for (int i = 0; i < 10; ++i)
+ DataLog::ReturnLog();
+ ASSERT_LT(DataLog::AddTable(DataLog::Combine("table failure", 1)), 0);
+}
+
+TEST(TestDataLog, VerifyCombineMethod) {
+ EXPECT_EQ(std::string("a proper table_1"),
+ DataLog::Combine("a proper table", 1));
+}
+
+TEST(TestDataLog, VerifySingleTable) {
+ DataLog::CreateLog();
+ DataLog::AddTable(DataLog::Combine("table", 1));
+ DataLog::AddColumn(DataLog::Combine("table", 1), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 1), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 1), "size", 5);
+ WebRtc_UWord32 sizes[5] = {1400, 1500, 1600, 1700, 1800};
+ for (int i = 0; i < 10; ++i) {
+ DataLog::InsertCell(DataLog::Combine("table", 1), "arrival",
+ static_cast<double>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 1), "timestamp",
+ static_cast<WebRtc_Word64>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 1), "size", sizes, 5);
+ DataLog::NextRow(DataLog::Combine("table", 1));
+ }
+ DataLog::ReturnLog();
+ // Verify file
+ FILE* table = fopen("table_1.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ // Read the column names and verify with the expected columns.
+ // Note that the columns are written to file in alphabetical order.
+ // Data expected from parsing the file
+ const int kNumberOfRows = 10;
+ std::string string_arrival[kNumberOfRows] = {
+ "0,", "1,", "2,", "3,", "4,",
+ "5,", "6,", "7,", "8,", "9,"
+ };
+ std::string string_timestamp[kNumberOfRows] = {
+ "4354,", "4355,", "4356,", "4357,",
+ "4358,", "4359,", "4360,", "4361,",
+ "4362,", "4363,"
+ };
+ std::string string_sizes = "1400,1500,1600,1700,1800,";
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["size[5],,,,,"] = ExpectedValues(
+ std::vector<std::string>(10, string_sizes), 5);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+}
+
+TEST(TestDataLog, VerifyMultipleTables) {
+ DataLog::CreateLog();
+ DataLog::AddTable(DataLog::Combine("table", 2));
+ DataLog::AddTable(DataLog::Combine("table", 3));
+ DataLog::AddColumn(DataLog::Combine("table", 2), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 2), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 2), "size", 1);
+ DataLog::AddTable(DataLog::Combine("table", 4));
+ DataLog::AddColumn(DataLog::Combine("table", 3), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 3), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 4), "size", 1);
+ for (WebRtc_Word32 i = 0; i < 10; ++i) {
+ DataLog::InsertCell(DataLog::Combine("table", 2), "arrival",
+ static_cast<WebRtc_Word32>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 2), "timestamp",
+ static_cast<WebRtc_Word32>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 2), "size",
+ static_cast<WebRtc_Word32>(1200 + 10 * i));
+ DataLog::InsertCell(DataLog::Combine("table", 3), "timestamp",
+ static_cast<WebRtc_Word32>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 3), "arrival",
+ static_cast<WebRtc_Word32>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 4), "size",
+ static_cast<WebRtc_Word32>(1200 + 10 * i));
+ DataLog::NextRow(DataLog::Combine("table", 4));
+ DataLog::NextRow(DataLog::Combine("table", 2));
+ DataLog::NextRow(DataLog::Combine("table", 3));
+ }
+ DataLog::ReturnLog();
+
+ // Data expected from parsing the file
+ const int kNumberOfRows = 10;
+ std::string string_arrival[kNumberOfRows] = {
+ "0,", "1,", "2,", "3,", "4,",
+ "5,", "6,", "7,", "8,", "9,"
+ };
+ std::string string_timestamp[kNumberOfRows] = {
+ "4354,", "4355,", "4356,", "4357,",
+ "4358,", "4359,", "4360,", "4361,",
+ "4362,", "4363,"
+ };
+ std::string string_size[kNumberOfRows] = {
+ "1200,", "1210,", "1220,", "1230,",
+ "1240,", "1250,", "1260,", "1270,",
+ "1280,", "1290,"
+ };
+
+ // Verify table 2
+ {
+ FILE* table = fopen("table_2.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["size,"] = ExpectedValues(
+ std::vector<std::string>(string_size,
+ string_size + kNumberOfRows),
+ 1);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+
+ // Verify table 3
+ {
+ FILE* table = fopen("table_3.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+
+ // Verify table 4
+ {
+ FILE* table = fopen("table_4.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["size,"] = ExpectedValues(
+ std::vector<std::string>(string_size,
+ string_size +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+}
+
+TEST(TestDataLogCWrapper, VerifyCWrapper) {
+ // Simply call all C wrapper log functions through the C helper unittests.
+ // Main purpose is to make sure that the linkage is correct.
+
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestCreateLog());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestCombine());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddTable());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddColumn());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_float());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_float());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_double());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_double());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_uint32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_uint32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int64());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int64());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestReturnLog());
+}
diff --git a/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc b/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc
new file mode 100644
index 0000000..9d630b6
--- /dev/null
+++ b/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/data_log.h"
+
+#include <cstdio>
+
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+const char* kDataLogFileName = "table_1.txt";
+
+void PerformLogging(std::string table_name) {
+ // Simulate normal DataTable logging behavior using this table name.
+ ASSERT_EQ(0, DataLog::AddTable(table_name));
+ ASSERT_EQ(0, DataLog::AddColumn(table_name, "test", 1));
+ for (int i = 0; i < 10; ++i) {
+ // TODO(kjellander): Check InsertCell result when the DataLog dummy is
+ // fixed.
+ DataLog::InsertCell(table_name, "test", static_cast<double>(i));
+ ASSERT_EQ(0, DataLog::NextRow(table_name));
+ }
+}
+
+// Simple test to verify DataLog is still working when the GYP variable
+// enable_data_logging==0 (the default case).
+TEST(TestDataLogDisabled, VerifyLoggingWorks) {
+ ASSERT_EQ(0, DataLog::CreateLog());
+ // Generate a table_name name and assure it's an empty string
+ // (dummy behavior).
+ std::string table_name = DataLog::Combine("table", 1);
+ ASSERT_EQ("", table_name);
+ PerformLogging(table_name);
+ DataLog::ReturnLog();
+}
+
+TEST(TestDataLogDisabled, EnsureNoFileIsWritten) {
+ // Remove any previous data files on disk:
+ std::remove(kDataLogFileName);
+ ASSERT_EQ(0, DataLog::CreateLog());
+ // Don't use the table name we would get from Combine on a disabled DataLog.
+ // Use "table_1" instead (which is what an enabled DataLog would give us).
+ PerformLogging("table_1");
+ DataLog::ReturnLog();
+ // Verify no data log file have been written:
+ ASSERT_EQ(NULL, fopen(kDataLogFileName, "r"));
+}
diff --git a/trunk/src/system_wrappers/source/event.cc b/trunk/src/system_wrappers/source/event.cc
new file mode 100644
index 0000000..608cd53
--- /dev/null
+++ b/trunk/src/system_wrappers/source/event.cc
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "event_wrapper.h"
+
+#if defined(_WIN32)
+ #include <windows.h>
+ #include "event_win.h"
+#elif defined(WEBRTC_MAC_INTEL)
+ #include <ApplicationServices/ApplicationServices.h>
+ #include <pthread.h>
+ #include "event_posix.h"
+#else
+ #include <pthread.h>
+ #include "event_posix.h"
+#endif
+
+namespace webrtc {
+EventWrapper* EventWrapper::Create()
+{
+#if defined(_WIN32)
+ return new EventWindows();
+#else
+ return EventPosix::Create();
+#endif
+}
+
+int EventWrapper::KeyPressed()
+{
+#if defined(_WIN32)
+ int keyDown = 0;
+ for(int key = 0x20; key < 0x90; key++)
+ {
+ short res = GetAsyncKeyState(key);
+ keyDown |= res%2; // Get the LSB
+ }
+ if(keyDown)
+ {
+ return 1;
+ }
+ else
+ {
+ return 0;
+ }
+#elif defined(WEBRTC_MAC_INTEL)
+ bool keyDown = false;
+ // loop through all Mac virtual key constant values
+ for(int keyIndex = 0; keyIndex <= 0x5C; keyIndex++)
+ {
+ keyDown |= CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, keyIndex);
+ }
+ if(keyDown)
+ {
+ return 1;
+ }
+ else
+ {
+ return 0;
+ }
+#else
+ return -1;
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_posix.cc b/trunk/src/system_wrappers/source/event_posix.cc
new file mode 100644
index 0000000..b77b902
--- /dev/null
+++ b/trunk/src/system_wrappers/source/event_posix.cc
@@ -0,0 +1,324 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "event_posix.h"
+
+#include <errno.h>
+#include <pthread.h>
+#include <signal.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/time.h>
+#include <unistd.h>
+
+namespace webrtc {
+const long int E6 = 1000000;
+const long int E9 = 1000 * E6;
+
+EventWrapper* EventPosix::Create()
+{
+ EventPosix* ptr = new EventPosix;
+ if (!ptr)
+ {
+ return NULL;
+ }
+
+ const int error = ptr->Construct();
+ if (error)
+ {
+ delete ptr;
+ return NULL;
+ }
+ return ptr;
+}
+
+
+EventPosix::EventPosix()
+ : _timerThread(0),
+ _timerEvent(0),
+ _periodic(false),
+ _time(0),
+ _count(0),
+ _state(kDown)
+{
+}
+
+int EventPosix::Construct()
+{
+ // Set start time to zero
+ memset(&_tCreate, 0, sizeof(_tCreate));
+
+ int result = pthread_mutex_init(&mutex, 0);
+ if (result != 0)
+ {
+ return -1;
+ }
+#ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ result = pthread_cond_init(&cond, 0);
+ if (result != 0)
+ {
+ return -1;
+ }
+#else
+ pthread_condattr_t condAttr;
+ result = pthread_condattr_init(&condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_cond_init(&cond, &condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_condattr_destroy(&condAttr);
+ if (result != 0)
+ {
+ return -1;
+ }
+#endif
+ return 0;
+}
+
+EventPosix::~EventPosix()
+{
+ StopTimer();
+ pthread_cond_destroy(&cond);
+ pthread_mutex_destroy(&mutex);
+}
+
+bool EventPosix::Reset()
+{
+ if (0 != pthread_mutex_lock(&mutex))
+ {
+ return false;
+ }
+ _state = kDown;
+ pthread_mutex_unlock(&mutex);
+ return true;
+}
+
+bool EventPosix::Set()
+{
+ if (0 != pthread_mutex_lock(&mutex))
+ {
+ return false;
+ }
+ _state = kUp;
+ // Release all waiting threads
+ pthread_cond_broadcast(&cond);
+ pthread_mutex_unlock(&mutex);
+ return true;
+}
+
+EventTypeWrapper EventPosix::Wait(unsigned long timeout)
+{
+ int retVal = 0;
+ if (0 != pthread_mutex_lock(&mutex))
+ {
+ return kEventError;
+ }
+
+ if (kDown == _state)
+ {
+ if (WEBRTC_EVENT_INFINITE != timeout)
+ {
+ timespec tEnd;
+#ifndef WEBRTC_MAC
+#ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ clock_gettime(CLOCK_REALTIME, &tEnd);
+#else
+ clock_gettime(CLOCK_MONOTONIC, &tEnd);
+#endif
+#else
+ timeval tVal;
+ struct timezone tZone;
+ tZone.tz_minuteswest = 0;
+ tZone.tz_dsttime = 0;
+ gettimeofday(&tVal,&tZone);
+ TIMEVAL_TO_TIMESPEC(&tVal,&tEnd);
+#endif
+ tEnd.tv_sec += timeout / 1000;
+ tEnd.tv_nsec += (timeout - (timeout / 1000) * 1000) * E6;
+
+ if (tEnd.tv_nsec >= E9)
+ {
+ tEnd.tv_sec++;
+ tEnd.tv_nsec -= E9;
+ }
+ retVal = pthread_cond_timedwait(&cond, &mutex, &tEnd);
+ } else {
+ retVal = pthread_cond_wait(&cond, &mutex);
+ }
+ }
+
+ _state = kDown;
+ pthread_mutex_unlock(&mutex);
+
+ switch(retVal)
+ {
+ case 0:
+ return kEventSignaled;
+ case ETIMEDOUT:
+ return kEventTimeout;
+ default:
+ return kEventError;
+ }
+}
+
+EventTypeWrapper EventPosix::Wait(timespec& tPulse)
+{
+ int retVal = 0;
+ if (0 != pthread_mutex_lock(&mutex))
+ {
+ return kEventError;
+ }
+
+ if (kUp != _state)
+ {
+ retVal = pthread_cond_timedwait(&cond, &mutex, &tPulse);
+ }
+ _state = kDown;
+
+ pthread_mutex_unlock(&mutex);
+
+ switch(retVal)
+ {
+ case 0:
+ return kEventSignaled;
+ case ETIMEDOUT:
+ return kEventTimeout;
+ default:
+ return kEventError;
+ }
+}
+
+bool EventPosix::StartTimer(bool periodic, unsigned long time)
+{
+ if (_timerThread)
+ {
+ if(_periodic)
+ {
+ // Timer already started.
+ return false;
+ } else {
+ // New one shot timer
+ _time = time;
+ _tCreate.tv_sec = 0;
+ _timerEvent->Set();
+ return true;
+ }
+ }
+
+ // Start the timer thread
+ _timerEvent = static_cast<EventPosix*>(EventWrapper::Create());
+ const char* threadName = "WebRtc_event_timer_thread";
+ _timerThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
+ threadName);
+ _periodic = periodic;
+ _time = time;
+ unsigned int id = 0;
+ if (_timerThread->Start(id))
+ {
+ return true;
+ }
+ return false;
+}
+
+bool EventPosix::Run(ThreadObj obj)
+{
+ return static_cast<EventPosix*>(obj)->Process();
+}
+
+bool EventPosix::Process()
+{
+ if (_tCreate.tv_sec == 0)
+ {
+#ifndef WEBRTC_MAC
+#ifdef WEBRTC_CLOCK_TYPE_REALTIME
+ clock_gettime(CLOCK_REALTIME, &_tCreate);
+#else
+ clock_gettime(CLOCK_MONOTONIC, &_tCreate);
+#endif
+#else
+ timeval tVal;
+ struct timezone tZone;
+ tZone.tz_minuteswest = 0;
+ tZone.tz_dsttime = 0;
+ gettimeofday(&tVal,&tZone);
+ TIMEVAL_TO_TIMESPEC(&tVal,&_tCreate);
+#endif
+ _count=0;
+ }
+
+ timespec tEnd;
+ unsigned long long time = _time * ++_count;
+ tEnd.tv_sec = _tCreate.tv_sec + time/1000;
+ tEnd.tv_nsec = _tCreate.tv_nsec + (time - (time/1000)*1000)*E6;
+
+ if ( tEnd.tv_nsec >= E9 )
+ {
+ tEnd.tv_sec++;
+ tEnd.tv_nsec -= E9;
+ }
+
+ switch(_timerEvent->Wait(tEnd))
+ {
+ case kEventSignaled:
+ return true;
+ case kEventError:
+ return false;
+ case kEventTimeout:
+ break;
+ }
+ if(_periodic || _count==1)
+ {
+ Set();
+ }
+ return true;
+}
+
+bool EventPosix::StopTimer()
+{
+ if(_timerThread)
+ {
+ _timerThread->SetNotAlive();
+ }
+ if (_timerEvent)
+ {
+ _timerEvent->Set();
+ }
+ if (_timerThread)
+ {
+ if(!_timerThread->Stop())
+ {
+ return false;
+ }
+
+ delete _timerThread;
+ _timerThread = 0;
+ }
+ if (_timerEvent)
+ {
+ delete _timerEvent;
+ _timerEvent = 0;
+ }
+
+ // Set time to zero to force new reference time for the timer.
+ memset(&_tCreate, 0, sizeof(_tCreate));
+ _count=0;
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_posix.h b/trunk/src/system_wrappers/source/event_posix.h
new file mode 100644
index 0000000..0e5893b
--- /dev/null
+++ b/trunk/src/system_wrappers/source/event_posix.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
+
+#include "event_wrapper.h"
+
+#include <pthread.h>
+#include <time.h>
+
+#include "thread_wrapper.h"
+
+namespace webrtc {
+enum State
+{
+ kUp = 1,
+ kDown = 2
+};
+
+class EventPosix : public EventWrapper
+{
+public:
+ static EventWrapper* Create();
+
+ virtual ~EventPosix();
+
+ virtual EventTypeWrapper Wait(unsigned long maxTime);
+ virtual bool Set();
+ virtual bool Reset();
+
+ virtual bool StartTimer(bool periodic, unsigned long time);
+ virtual bool StopTimer();
+
+private:
+ EventPosix();
+ int Construct();
+
+ static bool Run(ThreadObj obj);
+ bool Process();
+ EventTypeWrapper Wait(timespec& tPulse);
+
+
+private:
+ pthread_cond_t cond;
+ pthread_mutex_t mutex;
+
+ ThreadWrapper* _timerThread;
+ EventPosix* _timerEvent;
+ timespec _tCreate;
+
+ bool _periodic;
+ unsigned long _time; // In ms
+ unsigned long _count;
+ State _state;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/event_win.cc b/trunk/src/system_wrappers/source/event_win.cc
new file mode 100644
index 0000000..efcb5af
--- /dev/null
+++ b/trunk/src/system_wrappers/source/event_win.cc
@@ -0,0 +1,84 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "event_win.h"
+
+#include "Mmsystem.h"
+
+namespace webrtc {
+EventWindows::EventWindows()
+ : _event(::CreateEvent(NULL /* security attributes */,
+ FALSE /* manual reset */,
+ FALSE /* initial state */,
+ NULL /* name of event */)),
+ _timerID(NULL)
+{
+}
+
+EventWindows::~EventWindows()
+{
+ CloseHandle(_event);
+}
+
+bool EventWindows::Set()
+{
+ // Note: setting an event that is already set has no effect.
+ return SetEvent(_event) == 1 ? true : false;
+}
+
+bool EventWindows::Reset()
+{
+ return ResetEvent(_event) == 1 ? true : false;
+}
+
+EventTypeWrapper EventWindows::Wait(unsigned long maxTime)
+{
+ unsigned long res = WaitForSingleObject(_event, maxTime);
+ switch(res)
+ {
+ case WAIT_OBJECT_0:
+ return kEventSignaled;
+ case WAIT_TIMEOUT:
+ return kEventTimeout;
+ default:
+ return kEventError;
+ }
+}
+
+bool EventWindows::StartTimer(bool periodic, unsigned long time)
+{
+ if (_timerID != NULL)
+ {
+ timeKillEvent(_timerID);
+ _timerID=NULL;
+ }
+ if (periodic)
+ {
+ _timerID=timeSetEvent(time, 0,(LPTIMECALLBACK)HANDLE(_event),0,
+ TIME_PERIODIC|TIME_CALLBACK_EVENT_PULSE);
+ } else {
+ _timerID=timeSetEvent(time, 0,(LPTIMECALLBACK)HANDLE(_event),0,
+ TIME_ONESHOT|TIME_CALLBACK_EVENT_SET);
+ }
+
+ if (_timerID == NULL)
+ {
+ return false;
+ }
+ return true;
+}
+
+bool EventWindows::StopTimer()
+{
+ timeKillEvent(_timerID);
+ _timerID = NULL;
+ return true;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_win.h b/trunk/src/system_wrappers/source/event_win.h
new file mode 100644
index 0000000..8ca1360
--- /dev/null
+++ b/trunk/src/system_wrappers/source/event_win.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
+
+#include <windows.h>
+
+#include "event_wrapper.h"
+
+#include "typedefs.h"
+
+namespace webrtc {
+class EventWindows : public EventWrapper
+{
+public:
+ EventWindows();
+ virtual ~EventWindows();
+
+ virtual EventTypeWrapper Wait(unsigned long maxTime);
+ virtual bool Set();
+ virtual bool Reset();
+
+ virtual bool StartTimer(bool periodic, unsigned long time);
+ virtual bool StopTimer();
+
+private:
+ HANDLE _event;
+ WebRtc_UWord32 _timerID;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/file_impl.cc b/trunk/src/system_wrappers/source/file_impl.cc
new file mode 100644
index 0000000..4d06c54
--- /dev/null
+++ b/trunk/src/system_wrappers/source/file_impl.cc
@@ -0,0 +1,265 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "file_impl.h"
+
+#include <assert.h>
+
+#ifdef _WIN32
+#include <Windows.h>
+#else
+#include <stdarg.h>
+#include <string.h>
+#endif
+
+namespace webrtc {
+
+FileWrapper* FileWrapper::Create()
+{
+ return new FileWrapperImpl();
+}
+
+FileWrapperImpl::FileWrapperImpl()
+ : _id(NULL),
+ _open(false),
+ _looping(false),
+ _readOnly(false),
+ _maxSizeInBytes(0),
+ _sizeInBytes(0)
+{
+ memset(_fileNameUTF8, 0, kMaxFileNameSize);
+}
+
+FileWrapperImpl::~FileWrapperImpl()
+{
+ if (_id != NULL)
+ {
+ fclose(_id);
+ }
+}
+
+int FileWrapperImpl::CloseFile()
+{
+ if (_id != NULL)
+ {
+ fclose(_id);
+ _id = NULL;
+ }
+ memset(_fileNameUTF8, 0, kMaxFileNameSize);
+ _open = false;
+ return 0;
+}
+
+int FileWrapperImpl::Rewind()
+{
+ if(_looping || !_readOnly)
+ {
+ if (_id != NULL)
+ {
+ _sizeInBytes = 0;
+ return fseek(_id, 0, SEEK_SET);
+ }
+ }
+ return -1;
+}
+
+int FileWrapperImpl::SetMaxFileSize(size_t bytes)
+{
+ _maxSizeInBytes = bytes;
+ return 0;
+}
+
+int FileWrapperImpl::Flush()
+{
+ if (_id != NULL)
+ {
+ return fflush(_id);
+ }
+ return -1;
+}
+
+int FileWrapperImpl::FileName(char* fileNameUTF8,
+ size_t size) const
+{
+ size_t length = strlen(_fileNameUTF8);
+ if(length > kMaxFileNameSize)
+ {
+ assert(false);
+ return -1;
+ }
+ if(length < 1)
+ {
+ return -1;
+ }
+
+ // Make sure to NULL terminate
+ if(size < length)
+ {
+ length = size - 1;
+ }
+ memcpy(fileNameUTF8, _fileNameUTF8, length);
+ fileNameUTF8[length] = 0;
+ return 0;
+}
+
+bool FileWrapperImpl::Open() const
+{
+ return _open;
+}
+
+int FileWrapperImpl::OpenFile(const char *fileNameUTF8, bool readOnly,
+ bool loop, bool text)
+{
+ size_t length = strlen(fileNameUTF8);
+ if (length > kMaxFileNameSize - 1)
+ {
+ return -1;
+ }
+
+ _readOnly = readOnly;
+
+ FILE *tmpId = NULL;
+#if defined _WIN32
+ wchar_t wideFileName[kMaxFileNameSize];
+ wideFileName[0] = 0;
+
+ MultiByteToWideChar(CP_UTF8,
+ 0 /*UTF8 flag*/,
+ fileNameUTF8,
+ -1 /*Null terminated string*/,
+ wideFileName,
+ kMaxFileNameSize);
+ if(text)
+ {
+ if(readOnly)
+ {
+ tmpId = _wfopen(wideFileName, L"rt");
+ } else {
+ tmpId = _wfopen(wideFileName, L"wt");
+ }
+ } else {
+ if(readOnly)
+ {
+ tmpId = _wfopen(wideFileName, L"rb");
+ } else {
+ tmpId = _wfopen(wideFileName, L"wb");
+ }
+ }
+#else
+ if(text)
+ {
+ if(readOnly)
+ {
+ tmpId = fopen(fileNameUTF8, "rt");
+ } else {
+ tmpId = fopen(fileNameUTF8, "wt");
+ }
+ } else {
+ if(readOnly)
+ {
+ tmpId = fopen(fileNameUTF8, "rb");
+ } else {
+ tmpId = fopen(fileNameUTF8, "wb");
+ }
+ }
+#endif
+
+ if (tmpId != NULL)
+ {
+ // +1 comes from copying the NULL termination character.
+ memcpy(_fileNameUTF8, fileNameUTF8, length + 1);
+ if (_id != NULL)
+ {
+ fclose(_id);
+ }
+ _id = tmpId;
+ _looping = loop;
+ _open = true;
+ return 0;
+ }
+ return -1;
+}
+
+int FileWrapperImpl::Read(void* buf, int length)
+{
+ if (length < 0)
+ return -1;
+
+ if (_id == NULL)
+ return -1;
+
+ int bytes_read = static_cast<int>(fread(buf, 1, length, _id));
+ if (bytes_read != length && !_looping)
+ {
+ CloseFile();
+ }
+ return bytes_read;
+}
+
+int FileWrapperImpl::WriteText(const char* format, ...)
+{
+ if (format == NULL)
+ return -1;
+
+ if (_readOnly)
+ return -1;
+
+ if (_id == NULL)
+ return -1;
+
+ va_list args;
+ va_start(args, format);
+ int num_chars = vfprintf(_id, format, args);
+ va_end(args);
+
+ if (num_chars >= 0)
+ {
+ return num_chars;
+ }
+ else
+ {
+ CloseFile();
+ return -1;
+ }
+}
+
+bool FileWrapperImpl::Write(const void* buf, int length)
+{
+ if (buf == NULL)
+ return false;
+
+ if (length < 0)
+ return false;
+
+ if (_readOnly)
+ return false;
+
+ if (_id == NULL)
+ return false;
+
+ // Check if it's time to stop writing.
+ if (_maxSizeInBytes > 0 && (_sizeInBytes + length) > _maxSizeInBytes)
+ {
+ Flush();
+ return false;
+ }
+
+ size_t num_bytes = fwrite(buf, 1, length, _id);
+ if (num_bytes > 0)
+ {
+ _sizeInBytes += num_bytes;
+ return true;
+ }
+
+ CloseFile();
+ return false;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/file_impl.h b/trunk/src/system_wrappers/source/file_impl.h
new file mode 100644
index 0000000..31ab31e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/file_impl.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
+
+#include "file_wrapper.h"
+
+#include <stdio.h>
+
+namespace webrtc {
+
+class FileWrapperImpl : public FileWrapper
+{
+public:
+ FileWrapperImpl();
+ virtual ~FileWrapperImpl();
+
+ virtual int FileName(char* fileNameUTF8,
+ size_t size) const;
+
+ virtual bool Open() const;
+
+ virtual int OpenFile(const char* fileNameUTF8,
+ bool readOnly,
+ bool loop = false,
+ bool text = false);
+
+ virtual int CloseFile();
+ virtual int SetMaxFileSize(size_t bytes);
+ virtual int Flush();
+
+ virtual int Read(void* buf, int length);
+ virtual bool Write(const void *buf, int length);
+ virtual int WriteText(const char* format, ...);
+ virtual int Rewind();
+
+private:
+ FILE* _id;
+ bool _open;
+ bool _looping;
+ bool _readOnly;
+ size_t _maxSizeInBytes; // -1 indicates file size limitation is off
+ size_t _sizeInBytes;
+ char _fileNameUTF8[kMaxFileNameSize];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
diff --git a/trunk/src/system_wrappers/source/list_no_stl.cc b/trunk/src/system_wrappers/source/list_no_stl.cc
new file mode 100644
index 0000000..dbba571
--- /dev/null
+++ b/trunk/src/system_wrappers/source/list_no_stl.cc
@@ -0,0 +1,289 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "list_wrapper.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+ListItem::ListItem(const void* item)
+ : next_(0),
+ prev_(0),
+ item_ptr_(item),
+ item_(0)
+{
+}
+
+ListItem::ListItem(const unsigned int item)
+ : next_(0),
+ prev_(0),
+ item_ptr_(0),
+ item_(item)
+{
+}
+
+ListItem::~ListItem()
+{
+}
+
+void* ListItem::GetItem() const
+{
+ return const_cast<void*>(item_ptr_);
+}
+
+unsigned int ListItem::GetUnsignedItem() const
+{
+ return item_;
+}
+
+ListWrapper::ListWrapper()
+ : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
+ first_(0),
+ last_(0),
+ size_(0)
+{
+}
+
+ListWrapper::~ListWrapper()
+{
+ if (!Empty())
+ {
+ // TODO (hellner) I'm not sure this loggin is useful.
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
+ "Potential memory leak in ListWrapper");
+ // Remove all remaining list items.
+ while (Erase(First()) == 0)
+ {}
+ }
+ delete critical_section_;
+}
+
+bool ListWrapper::Empty() const
+{
+ return !first_ && !last_;
+}
+
+unsigned int ListWrapper::GetSize() const
+{
+ return size_;
+}
+
+int ListWrapper::PushBack(const void* ptr)
+{
+ ListItem* item = new ListItem(ptr);
+ CriticalSectionScoped lock(critical_section_);
+ PushBackImpl(item);
+ return 0;
+}
+
+int ListWrapper::PushBack(const unsigned int item_id)
+{
+ ListItem* item = new ListItem(item_id);
+ CriticalSectionScoped lock(critical_section_);
+ PushBackImpl(item);
+ return 0;
+}
+
+int ListWrapper::PushFront(const unsigned int item_id)
+{
+ ListItem* item = new ListItem(item_id);
+ CriticalSectionScoped lock(critical_section_);
+ PushFrontImpl(item);
+ return 0;
+}
+
+int ListWrapper::PushFront(const void* ptr)
+{
+ ListItem* item = new ListItem(ptr);
+ CriticalSectionScoped lock(critical_section_);
+ PushFrontImpl(item);
+ return 0;
+}
+
+int ListWrapper::PopFront()
+{
+ return Erase(first_);
+}
+
+int ListWrapper::PopBack()
+{
+ return Erase(last_);
+}
+
+ListItem* ListWrapper::First() const
+{
+ return first_;
+}
+
+ListItem* ListWrapper::Last() const
+{
+ return last_;
+}
+
+ListItem* ListWrapper::Next(ListItem* item) const
+{
+ if(!item)
+ {
+ return 0;
+ }
+ return item->next_;
+}
+
+ListItem* ListWrapper::Previous(ListItem* item) const
+{
+ if (!item)
+ {
+ return 0;
+ }
+ return item->prev_;
+}
+
+int ListWrapper::Insert(ListItem* existing_previous_item, ListItem* new_item)
+{
+ if (!new_item)
+ {
+ return -1;
+ }
+ // Allow existing_previous_item to be NULL if the list is empty.
+ // TODO (hellner) why allow this? Keep it as is for now to avoid
+ // breaking API contract.
+ if (!existing_previous_item && !Empty())
+ {
+ return -1;
+ }
+ CriticalSectionScoped lock(critical_section_);
+ if (!existing_previous_item)
+ {
+ PushBackImpl(new_item);
+ return 0;
+ }
+ ListItem* next_item = existing_previous_item->next_;
+ new_item->next_ = existing_previous_item->next_;
+ new_item->prev_ = existing_previous_item;
+ existing_previous_item->next_ = new_item;
+ if (next_item)
+ {
+ next_item->prev_ = new_item;
+ }
+ else
+ {
+ last_ = new_item;
+ }
+ size_++;
+ return 0;
+}
+
+int ListWrapper::InsertBefore(ListItem* existing_next_item,
+ ListItem* new_item)
+{
+ if (!new_item)
+ {
+ return -1;
+ }
+ // Allow existing_next_item to be NULL if the list is empty.
+ // Todo: why allow this? Keep it as is for now to avoid breaking API
+ // contract.
+ if (!existing_next_item && !Empty())
+ {
+ return -1;
+ }
+ CriticalSectionScoped lock(critical_section_);
+ if (!existing_next_item)
+ {
+ PushBackImpl(new_item);
+ return 0;
+ }
+
+ ListItem* previous_item = existing_next_item->prev_;
+ new_item->next_ = existing_next_item;
+ new_item->prev_ = previous_item;
+ existing_next_item->prev_ = new_item;
+ if (previous_item)
+ {
+ previous_item->next_ = new_item;
+ }
+ else
+ {
+ first_ = new_item;
+ }
+ size_++;
+ return 0;
+}
+
+int ListWrapper::Erase(ListItem* item)
+{
+ if (!item)
+ {
+ return -1;
+ }
+ size_--;
+ ListItem* previous_item = item->prev_;
+ ListItem* next_item = item->next_;
+ if (!previous_item)
+ {
+ if(next_item)
+ {
+ next_item->prev_ = 0;
+ }
+ first_ = next_item;
+ }
+ else
+ {
+ previous_item->next_ = next_item;
+ }
+ if (!next_item)
+ {
+ if(previous_item)
+ {
+ previous_item->next_ = 0;
+ }
+ last_ = previous_item;
+ }
+ else
+ {
+ next_item->prev_ = previous_item;
+ }
+ delete item;
+ return 0;
+}
+
+void ListWrapper::PushBackImpl(ListItem* item)
+{
+ if (Empty())
+ {
+ first_ = item;
+ last_ = item;
+ size_++;
+ return;
+ }
+
+ item->prev_ = last_;
+ last_->next_ = item;
+ last_ = item;
+ size_++;
+}
+
+void ListWrapper::PushFrontImpl(ListItem* item)
+{
+ if (Empty())
+ {
+ first_ = item;
+ last_ = item;
+ size_++;
+ return;
+ }
+
+ item->next_ = first_;
+ first_->prev_ = item;
+ first_ = item;
+ size_++;
+}
+} //namespace webrtc
diff --git a/trunk/src/system_wrappers/source/list_no_stl.h b/trunk/src/system_wrappers/source/list_no_stl.h
new file mode 100644
index 0000000..26d844c
--- /dev/null
+++ b/trunk/src/system_wrappers/source/list_no_stl.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
+
+#include "constructor_magic.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class ListNoStlItem
+{
+public:
+ ListNoStlItem(const void* ptr);
+ ListNoStlItem(const unsigned int item);
+ virtual ~ListNoStlItem();
+ void* GetItem() const;
+ unsigned int GetUnsignedItem() const;
+
+protected:
+ ListNoStlItem* next_;
+ ListNoStlItem* prev_;
+
+private:
+ friend class ListNoStl;
+
+ const void* item_ptr_;
+ const unsigned int item_;
+ DISALLOW_COPY_AND_ASSIGN(ListNoStlItem);
+};
+
+
+class ListNoStl
+{
+public:
+ ListNoStl();
+ virtual ~ListNoStl();
+
+ // ListWrapper functions
+ unsigned int GetSize() const;
+ int PushBack(const void* ptr);
+ int PushBack(const unsigned int item_id);
+ int PushFront(const void* ptr);
+ int PushFront(const unsigned int item_id);
+ int PopFront();
+ int PopBack();
+ bool Empty() const;
+ ListNoStlItem* First() const;
+ ListNoStlItem* Last() const;
+ ListNoStlItem* Next(ListNoStlItem* item) const;
+ ListNoStlItem* Previous(ListNoStlItem* item) const;
+ int Erase(ListNoStlItem* item);
+ int Insert(ListNoStlItem* existing_previous_item,
+ ListNoStlItem* new_item);
+
+ int InsertBefore(ListNoStlItem* existing_next_item,
+ ListNoStlItem* new_item);
+
+private:
+ void PushBack(ListNoStlItem* item);
+ void PushFront(ListNoStlItem* item);
+
+ CriticalSectionWrapper* critical_section_;
+ ListNoStlItem* first_;
+ ListNoStlItem* last_;
+ unsigned int size_;
+ DISALLOW_COPY_AND_ASSIGN(ListNoStl);
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
diff --git a/trunk/src/system_wrappers/source/list_stl.cc b/trunk/src/system_wrappers/source/list_stl.cc
new file mode 100644
index 0000000..dcc63c3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/list_stl.cc
@@ -0,0 +1,244 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "list_wrapper.h"
+
+#include "trace.h"
+
+namespace webrtc {
+ListItem::ListItem(const void* item)
+ : this_iter_(),
+ item_ptr_(item),
+ item_(0)
+{
+}
+
+ListItem::ListItem(const unsigned int item)
+ : this_iter_(),
+ item_ptr_(0),
+ item_(item)
+{
+}
+
+ListItem::~ListItem()
+{
+}
+
+void* ListItem::GetItem() const
+{
+ return const_cast<void*>(item_ptr_);
+}
+
+unsigned int ListItem::GetUnsignedItem() const
+{
+ return item_;
+}
+
+ListWrapper::ListWrapper() : list_()
+{
+}
+
+ListWrapper::~ListWrapper()
+{
+ if (!Empty())
+ {
+ // TODO (hellner) I'm not sure this loggin is useful.
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
+ "Potential memory leak in ListWrapper");
+ // Remove all remaining list items.
+ while (Erase(First()) == 0)
+ {}
+ }
+}
+
+bool ListWrapper::Empty() const
+{
+ return list_.empty();
+}
+
+unsigned int ListWrapper::GetSize() const
+{
+ return list_.size();
+}
+
+int ListWrapper::PushBack(const void* ptr)
+{
+ ListItem* item = new ListItem(ptr);
+ list_.push_back(item);
+ return 0;
+}
+
+int ListWrapper::PushBack(const unsigned int item_id)
+{
+ ListItem* item = new ListItem(item_id);
+ list_.push_back(item);
+ return 0;
+}
+
+int ListWrapper::PushFront(const unsigned int item_id)
+{
+ ListItem* item = new ListItem(item_id);
+ list_.push_front(item);
+ return 0;
+}
+
+int ListWrapper::PushFront(const void* ptr)
+{
+ ListItem* item = new ListItem(ptr);
+ list_.push_front(item);
+ return 0;
+}
+
+int ListWrapper::PopFront()
+{
+ if(list_.empty())
+ {
+ return -1;
+ }
+ list_.pop_front();
+ return 0;
+}
+
+int ListWrapper::PopBack()
+{
+ if(list_.empty())
+ {
+ return -1;
+ }
+ list_.pop_back();
+ return 0;
+}
+
+ListItem* ListWrapper::First() const
+{
+ if(list_.empty())
+ {
+ return NULL;
+ }
+ std::list<ListItem*>::iterator item_iter = list_.begin();
+ ListItem* return_item = (*item_iter);
+ return_item->this_iter_ = item_iter;
+ return return_item;
+}
+
+ListItem* ListWrapper::Last() const
+{
+ if(list_.empty())
+ {
+ return NULL;
+ }
+ // std::list::end() addresses the last item + 1. Decrement so that the
+ // actual last is accessed.
+ std::list<ListItem*>::iterator item_iter = list_.end();
+ --item_iter;
+ ListItem* return_item = (*item_iter);
+ return_item->this_iter_ = item_iter;
+ return return_item;
+}
+
+ListItem* ListWrapper::Next(ListItem* item) const
+{
+ if(item == NULL)
+ {
+ return NULL;
+ }
+ std::list<ListItem*>::iterator item_iter = item->this_iter_;
+ ++item_iter;
+ if (item_iter == list_.end())
+ {
+ return NULL;
+ }
+ ListItem* return_item = (*item_iter);
+ return_item->this_iter_ = item_iter;
+ return return_item;
+}
+
+ListItem* ListWrapper::Previous(ListItem* item) const
+{
+ if(item == NULL)
+ {
+ return NULL;
+ }
+ std::list<ListItem*>::iterator item_iter = item->this_iter_;
+ if (item_iter == list_.begin())
+ {
+ return NULL;
+ }
+ --item_iter;
+ ListItem* return_item = (*item_iter);
+ return_item->this_iter_ = item_iter;
+ return return_item;
+}
+
+int ListWrapper::Insert(ListItem* existing_previous_item,
+ ListItem* new_item)
+{
+ // Allow existingPreviousItem to be NULL if the list is empty.
+ // TODO (hellner) why allow this? Keep it as is for now to avoid
+ // breaking API contract.
+ if (!existing_previous_item && !Empty())
+ {
+ return -1;
+ }
+
+ if (!new_item)
+ {
+ return -1;
+ }
+
+ std::list<ListItem*>::iterator insert_location = list_.begin();
+ if (!Empty())
+ {
+ insert_location = existing_previous_item->this_iter_;
+ if(insert_location != list_.end())
+ {
+ ++insert_location;
+ }
+ }
+
+ list_.insert(insert_location,new_item);
+ return 0;
+}
+
+int ListWrapper::InsertBefore(ListItem* existing_next_item,
+ ListItem* new_item)
+{
+ // Allow existing_next_item to be NULL if the list is empty.
+ // Todo: why allow this? Keep it as is for now to avoid breaking API
+ // contract.
+ if (!existing_next_item && !Empty())
+ {
+ return -1;
+ }
+ if (!new_item)
+ {
+ return -1;
+ }
+
+ std::list<ListItem*>::iterator insert_location = list_.begin();
+ if (!Empty())
+ {
+ insert_location = existing_next_item->this_iter_;
+ }
+
+ list_.insert(insert_location,new_item);
+ return 0;
+}
+
+int ListWrapper::Erase(ListItem* item)
+{
+ if(item == NULL)
+ {
+ return -1;
+ }
+ list_.erase(item->this_iter_);
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/list_stl.h b/trunk/src/system_wrappers/source/list_stl.h
new file mode 100644
index 0000000..b83a664
--- /dev/null
+++ b/trunk/src/system_wrappers/source/list_stl.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
+
+#include <list>
+
+#include "constructor_magic.h"
+
+namespace webrtc {
+class ListItem
+{
+friend class ListWrapper;
+
+public:
+ ListItem(const void* ptr);
+ ListItem(const unsigned int item);
+ virtual ~ListItem();
+ void* GetItem() const;
+ unsigned int GetUnsignedItem() const;
+
+private:
+ mutable std::list<ListItem*>::iterator this_iter_;
+ const void* item_ptr_;
+ const unsigned int item_;
+ DISALLOW_COPY_AND_ASSIGN(ListItem);
+};
+
+class ListWrapper
+{
+public:
+ ListWrapper();
+ ~ListWrapper();
+
+ // ListWrapper functions
+ unsigned int GetSize() const;
+ int PushBack(const void* ptr);
+ int PushBack(const unsigned int item_id);
+ int PushFront(const void* ptr);
+ int PushFront(const unsigned int item_id);
+ int PopFront();
+ int PopBack();
+ bool Empty() const;
+ ListItem* First() const;
+ ListItem* Last() const;
+ ListItem* Next(ListItem* item) const;
+ ListItem* Previous(ListItem* item) const;
+ int Erase(ListItem* item);
+ int Insert(ListItem* existing_previous_item, ListItem* new_item);
+ int InsertBefore(ListItem* existing_next_item, ListItem* new_item);
+
+private:
+ mutable std::list<ListItem*> list_;
+ DISALLOW_COPY_AND_ASSIGN(ListWrapper);
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
diff --git a/trunk/src/system_wrappers/source/list_unittest.cc b/trunk/src/system_wrappers/source/list_unittest.cc
new file mode 100644
index 0000000..4d32f59
--- /dev/null
+++ b/trunk/src/system_wrappers/source/list_unittest.cc
@@ -0,0 +1,479 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+#include "system_wrappers/interface/list_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+
+using ::webrtc::ListWrapper;
+using ::webrtc::ListItem;
+using ::webrtc::scoped_ptr;
+
+// Note: kNumberOfElements needs to be even.
+const unsigned int kNumberOfElements = 10;
+
+// An opaque implementation of dynamic or statically allocated unsigned ints.
+// This class makes it possible to use the exact same code for testing of both
+// the dynamic and static implementation of ListWrapper.
+// Clarification: ListWrapper has two versions of PushBack(..). It takes an
+// unsigned integer or a void pointer. The integer implementation takes care
+// of memory management. The void pointer version expect the caller to manage
+// the memory associated with the void pointer.
+// This class works like the integer version but can be implemented on top of
+// either the integer version or void pointer version of ListWrapper.
+// Note: the non-virtual fuctions behave the same for both versions.
+class ListWrapperSimple {
+public:
+ static ListWrapperSimple* Create(bool static_allocation);
+ virtual ~ListWrapperSimple() {}
+
+ // These three functions should be used for manipulating ListItems so that
+ // they are the type corresponding to the underlying implementation.
+ virtual unsigned int GetUnsignedItem(
+ const ListItem* item) const = 0;
+ virtual ListItem* CreateListItem(unsigned int item_id) = 0;
+ unsigned int GetSize() const {
+ return list_.GetSize();
+ }
+ virtual int PushBack(const unsigned int item_id) = 0;
+ virtual int PushFront(const unsigned int item_id) = 0;
+ virtual int PopFront() = 0;
+ virtual int PopBack() = 0;
+ bool Empty() const {
+ return list_.Empty();
+ }
+ ListItem* First() const {
+ return list_.First();
+ }
+ ListItem* Last() const {
+ return list_.Last();
+ }
+ ListItem* Next(ListItem* item) const {
+ return list_.Next(item);
+ }
+ ListItem* Previous(ListItem* item) const {
+ return list_.Previous(item);
+ }
+ virtual int Erase(ListItem* item) = 0;
+ int Insert(ListItem* existing_previous_item,
+ ListItem* new_item) {
+ const int retval = list_.Insert(existing_previous_item, new_item);
+ if (retval != 0) {
+ EXPECT_TRUE(DestroyListItem(new_item));
+ }
+ return retval;
+ }
+
+ int InsertBefore(ListItem* existing_next_item,
+ ListItem* new_item) {
+ const int retval = list_.InsertBefore(existing_next_item, new_item);
+ if (retval != 0) {
+ EXPECT_TRUE(DestroyListItem(new_item));
+ }
+ return retval;
+ }
+protected:
+ ListWrapperSimple() {}
+
+ virtual bool DestroyListItemContent(ListItem* item) = 0;
+ bool DestroyListItem(ListItem* item) {
+ const bool retval = DestroyListItemContent(item);
+ delete item;
+ return retval;
+ }
+
+ ListWrapper list_;
+};
+
+void ClearList(ListWrapperSimple* list_wrapper) {
+ if (list_wrapper == NULL) {
+ return;
+ }
+ ListItem* list_item = list_wrapper->First();
+ while (list_item != NULL) {
+ EXPECT_EQ(list_wrapper->Erase(list_item), 0);
+ list_item = list_wrapper->First();
+ }
+}
+
+class ListWrapperStatic : public ListWrapperSimple {
+public:
+ ListWrapperStatic() {}
+ virtual ~ListWrapperStatic() {
+ ClearList(this);
+ }
+
+ virtual unsigned int GetUnsignedItem(const ListItem* item) const {
+ return item->GetUnsignedItem();
+ }
+ virtual ListItem* CreateListItem(unsigned int item_id) {
+ return new ListItem(item_id);
+ }
+ virtual bool DestroyListItemContent(ListItem* item) {
+ return true;
+ }
+ virtual int PushBack(const unsigned int item_id) {
+ return list_.PushBack(item_id);
+ }
+ virtual int PushFront(const unsigned int item_id) {
+ return list_.PushFront(item_id);
+ }
+ virtual int PopFront() {
+ return list_.PopFront();
+ }
+ virtual int PopBack() {
+ return list_.PopBack();
+ }
+ virtual int Erase(ListItem* item) {
+ return list_.Erase(item);
+ }
+};
+
+class ListWrapperDynamic : public ListWrapperSimple {
+public:
+ ListWrapperDynamic() {}
+ virtual ~ListWrapperDynamic() {
+ ClearList(this);
+ }
+
+ virtual unsigned int GetUnsignedItem(const ListItem* item) const {
+ const unsigned int* return_value_pointer =
+ reinterpret_cast<unsigned int*> (item->GetItem());
+ if (return_value_pointer == NULL) {
+ return -1;
+ }
+ return *return_value_pointer;
+ }
+ virtual ListItem* CreateListItem(unsigned int item_id) {
+ unsigned int* item_id_pointer = new unsigned int;
+ if (item_id_pointer == NULL) {
+ return NULL;
+ }
+ *item_id_pointer = item_id;
+ ListItem* return_value = new ListItem(
+ reinterpret_cast<void*>(item_id_pointer));
+ if (return_value == NULL) {
+ delete item_id_pointer;
+ return NULL;
+ }
+ return return_value;
+ }
+ virtual bool DestroyListItemContent(ListItem* item) {
+ if (item == NULL) {
+ return false;
+ }
+ bool return_value = false;
+ unsigned int* item_id_ptr = reinterpret_cast<unsigned int*>(
+ item->GetItem());
+ if (item_id_ptr != NULL) {
+ return_value = true;
+ delete item_id_ptr;
+ }
+ return return_value;
+ }
+ virtual int PushBack(const unsigned int item_id) {
+ unsigned int* item_id_ptr = new unsigned int;
+ if (item_id_ptr == NULL) {
+ return -1;
+ }
+ *item_id_ptr = item_id;
+ const int return_value = list_.PushBack(
+ reinterpret_cast<void*>(item_id_ptr));
+ if (return_value != 0) {
+ delete item_id_ptr;
+ }
+ return return_value;
+ }
+ virtual int PushFront(const unsigned int item_id) {
+ unsigned int* item_id_ptr = new unsigned int;
+ if (item_id_ptr == NULL) {
+ return -1;
+ }
+ *item_id_ptr = item_id;
+ const int return_value = list_.PushFront(
+ reinterpret_cast<void*>(item_id_ptr));
+ if (return_value != 0) {
+ delete item_id_ptr;
+ }
+ return return_value;
+ }
+ virtual int PopFront() {
+ return Erase(list_.First());
+ }
+ virtual int PopBack() {
+ return Erase(list_.Last());
+ }
+ virtual int Erase(ListItem* item) {
+ if (item == NULL) {
+ return -1;
+ }
+ int retval = 0;
+ if (!DestroyListItemContent(item)) {
+ retval = -1;
+ ADD_FAILURE();
+ }
+ if (list_.Erase(item) != 0) {
+ retval = -1;
+ }
+ return retval;
+ }
+};
+
+ListWrapperSimple* ListWrapperSimple::Create(bool static_allocation) {
+ if(static_allocation)
+ {
+ return new ListWrapperStatic();
+ }
+ return new ListWrapperDynamic();
+}
+
+ListWrapperSimple* CreateAscendingList(bool static_allocation) {
+ ListWrapperSimple* return_value = ListWrapperSimple::Create(
+ static_allocation);
+ if (return_value == NULL) {
+ return NULL;
+ }
+ for (unsigned int i = 0; i < kNumberOfElements; ++i) {
+ if (return_value->PushBack(i) == -1) {
+ ClearList(return_value);
+ delete return_value;
+ return NULL;
+ }
+ }
+ return return_value;
+}
+
+ListWrapperSimple* CreateDescendingList(bool static_allocation) {
+ ListWrapperSimple* return_value = ListWrapperSimple::Create(
+ static_allocation);
+ if (return_value == NULL) {
+ return NULL;
+ }
+ for (unsigned int i = 0; i < kNumberOfElements; ++i) {
+ if (return_value->PushBack(kNumberOfElements - i - 1) == -1) {
+ ClearList(return_value);
+ delete return_value;
+ return NULL;
+ }
+ }
+ return return_value;
+}
+
+// [0,kNumberOfElements - 1,1,kNumberOfElements - 2,...] (this is why
+// kNumberOfElements need to be even)
+ListWrapperSimple* CreateInterleavedList(bool static_allocation) {
+ ListWrapperSimple* return_value = ListWrapperSimple::Create(
+ static_allocation);
+ if (return_value == NULL) {
+ return NULL;
+ }
+ unsigned int uneven_count = 0;
+ unsigned int even_count = 0;
+ for (unsigned int i = 0; i < kNumberOfElements; i++) {
+ unsigned int push_value = 0;
+ if ((i % 2) == 0) {
+ push_value = even_count;
+ even_count++;
+ } else {
+ push_value = kNumberOfElements - uneven_count - 1;
+ uneven_count++;
+ }
+ if (return_value->PushBack(push_value) == -1) {
+ ClearList(return_value);
+ delete return_value;
+ return NULL;
+ }
+ }
+ return return_value;
+}
+
+void PrintList(const ListWrapperSimple* list) {
+ ListItem* list_item = list->First();
+ printf("[");
+ while (list_item != NULL)
+ {
+ printf("%3u", list->GetUnsignedItem(list_item));
+ list_item = list->Next(list_item);
+ }
+ printf("]\n");
+}
+
+bool CompareLists(const ListWrapperSimple* lhs, const ListWrapperSimple* rhs) {
+ const unsigned int list_size = lhs->GetSize();
+ if (lhs->GetSize() != rhs->GetSize()) {
+ return false;
+ }
+ if (lhs->Empty()) {
+ return rhs->Empty();
+ }
+ unsigned int i = 0;
+ ListItem* lhs_item = lhs->First();
+ ListItem* rhs_item = rhs->First();
+ while (i < list_size) {
+ if (lhs_item == NULL) {
+ return false;
+ }
+ if (rhs_item == NULL) {
+ return false;
+ }
+ if (lhs->GetUnsignedItem(lhs_item) != rhs->GetUnsignedItem(rhs_item)) {
+ return false;
+ }
+ i++;
+ lhs_item = lhs->Next(lhs_item);
+ rhs_item = rhs->Next(rhs_item);
+ }
+ return true;
+}
+
+TEST(ListWrapperTest,ReverseNewIntList) {
+ // Create a new temporary list with elements reversed those of
+ // new_int_list_
+ const scoped_ptr<ListWrapperSimple> descending_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(descending_list.get() == NULL);
+ ASSERT_FALSE(descending_list->Empty());
+ ASSERT_EQ(kNumberOfElements,descending_list->GetSize());
+
+ const scoped_ptr<ListWrapperSimple> ascending_list(
+ CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
+ ASSERT_FALSE(ascending_list->Empty());
+ ASSERT_EQ(kNumberOfElements,ascending_list->GetSize());
+
+ scoped_ptr<ListWrapperSimple> list_to_reverse(
+ ListWrapperSimple::Create(rand()%2));
+
+ // Reverse the list using PushBack and Previous.
+ for (ListItem* item = ascending_list->Last(); item != NULL;
+ item = ascending_list->Previous(item)) {
+ list_to_reverse->PushBack(ascending_list->GetUnsignedItem(item));
+ }
+
+ ASSERT_TRUE(CompareLists(descending_list.get(), list_to_reverse.get()));
+
+ scoped_ptr<ListWrapperSimple> list_to_un_reverse(
+ ListWrapperSimple::Create(rand()%2));
+ ASSERT_FALSE(list_to_un_reverse.get() == NULL);
+ // Reverse the reversed list using PushFront and Next.
+ for (ListItem* item = list_to_reverse->First(); item != NULL;
+ item = list_to_reverse->Next(item)) {
+ list_to_un_reverse->PushFront(list_to_reverse->GetUnsignedItem(item));
+ }
+ ASSERT_TRUE(CompareLists(ascending_list.get(), list_to_un_reverse.get()));
+}
+
+TEST(ListWrapperTest,PopTest) {
+ scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
+ ASSERT_FALSE(ascending_list->Empty());
+ EXPECT_EQ(0, ascending_list->PopFront());
+ EXPECT_EQ(1U, ascending_list->GetUnsignedItem(ascending_list->First()));
+
+ EXPECT_EQ(0, ascending_list->PopBack());
+ EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetUnsignedItem(
+ ascending_list->Last()));
+ EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetSize());
+}
+
+// Use Insert to interleave two lists.
+TEST(ListWrapperTest,InterLeaveTest) {
+ scoped_ptr<ListWrapperSimple> interleave_list(
+ CreateAscendingList(rand()%2));
+ ASSERT_FALSE(interleave_list.get() == NULL);
+ ASSERT_FALSE(interleave_list->Empty());
+
+ scoped_ptr<ListWrapperSimple> descending_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(descending_list.get() == NULL);
+
+ for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
+ ASSERT_EQ(0, interleave_list->PopBack());
+ ASSERT_EQ(0, descending_list->PopBack());
+ }
+ ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, descending_list->GetSize());
+
+ unsigned int insert_position = kNumberOfElements/2;
+ ASSERT_EQ(insert_position * 2, kNumberOfElements);
+ while (!descending_list->Empty())
+ {
+ ListItem* item = descending_list->Last();
+ ASSERT_FALSE(item == NULL);
+
+ const unsigned int item_id = descending_list->GetUnsignedItem(item);
+ ASSERT_EQ(0, descending_list->Erase(item));
+
+ ListItem* insert_item = interleave_list->CreateListItem(item_id);
+ ASSERT_FALSE(insert_item == NULL);
+ item = interleave_list->First();
+ ASSERT_FALSE(item == NULL);
+ for (unsigned int j = 0; j < insert_position - 1; ++j) {
+ item = interleave_list->Next(item);
+ ASSERT_FALSE(item == NULL);
+ }
+ EXPECT_EQ(0, interleave_list->Insert(item, insert_item));
+ --insert_position;
+ }
+
+ scoped_ptr<ListWrapperSimple> interleaved_list(
+ CreateInterleavedList(rand()%2));
+ ASSERT_FALSE(interleaved_list.get() == NULL);
+ ASSERT_FALSE(interleaved_list->Empty());
+ ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
+}
+
+// Use InsertBefore to interleave two lists.
+TEST(ListWrapperTest,InterLeaveTestII) {
+ scoped_ptr<ListWrapperSimple> interleave_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(interleave_list.get() == NULL);
+ ASSERT_FALSE(interleave_list->Empty());
+
+ scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
+
+ for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
+ ASSERT_EQ(0, interleave_list->PopBack());
+ ASSERT_EQ(0, ascending_list->PopBack());
+ }
+ ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, ascending_list->GetSize());
+
+ unsigned int insert_position = kNumberOfElements/2;
+ ASSERT_EQ(insert_position * 2, kNumberOfElements);
+ while (!ascending_list->Empty())
+ {
+ ListItem* item = ascending_list->Last();
+ ASSERT_FALSE(item == NULL);
+
+ const unsigned int item_id = ascending_list->GetUnsignedItem(item);
+ ASSERT_EQ(0,ascending_list->Erase(item));
+
+ ListItem* insert_item = interleave_list->CreateListItem(item_id);
+ ASSERT_FALSE(insert_item == NULL);
+ item = interleave_list->First();
+ ASSERT_FALSE(item == NULL);
+ for (unsigned int j = 0; j < insert_position - 1; ++j) {
+ item = interleave_list->Next(item);
+ ASSERT_FALSE(item == NULL);
+ }
+ EXPECT_EQ(interleave_list->InsertBefore(item, insert_item), 0);
+ --insert_position;
+ }
+
+ scoped_ptr<ListWrapperSimple> interleaved_list(
+ CreateInterleavedList(rand()%2));
+ ASSERT_FALSE(interleaved_list.get() == NULL);
+ ASSERT_FALSE(interleaved_list->Empty());
+
+ ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
+}
diff --git a/trunk/src/system_wrappers/source/map.cc b/trunk/src/system_wrappers/source/map.cc
new file mode 100644
index 0000000..331da32
--- /dev/null
+++ b/trunk/src/system_wrappers/source/map.cc
@@ -0,0 +1,166 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "map_wrapper.h"
+
+#include "trace.h"
+
+namespace webrtc {
+MapItem::MapItem(int id, void* item) : item_id_(id), item_pointer_(item)
+{
+}
+
+MapItem::~MapItem()
+{
+}
+
+void* MapItem::GetItem()
+{
+ return item_pointer_;
+}
+
+int MapItem::GetId()
+{
+ return item_id_;
+}
+
+unsigned int MapItem::GetUnsignedId()
+{
+ return static_cast<unsigned int>(item_id_);
+}
+
+void MapItem::SetItem(void* ptr)
+{
+ item_pointer_ = ptr;
+}
+
+MapWrapper::MapWrapper() : map_()
+{
+}
+
+MapWrapper::~MapWrapper()
+{
+ if (!map_.empty())
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
+ "Potential memory leak in MapWrapper");
+ // Remove all map items. Please note that std::map::clear() can't be
+ // used because each item has some dynamically allocated memory
+ // associated with it (i.e. using std::map::clear would introduce a
+ // memory leak).
+ while (Erase(First()) == 0)
+ {}
+ }
+}
+
+int MapWrapper::Size() const
+{
+ return (int)map_.size();
+}
+
+int MapWrapper::Insert(int id, void* ptr)
+{
+ map_[id] = new MapItem(id,ptr);
+ return 0;
+}
+
+MapItem* MapWrapper::First() const
+{
+ std::map<int, MapItem*>::const_iterator it = map_.begin();
+ if (it != map_.end())
+ {
+ return it->second;
+ }
+ return 0;
+}
+
+MapItem* MapWrapper::Last() const
+{
+ std::map<int, MapItem*>::const_reverse_iterator it = map_.rbegin();
+ if (it != map_.rend())
+ {
+ return it->second;
+ }
+ return 0;
+}
+
+MapItem* MapWrapper::Next(MapItem* item) const
+{
+ if (item == 0)
+ {
+ return 0;
+ }
+ std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
+ if (it != map_.end())
+ {
+ it++;
+ if (it != map_.end())
+ {
+ return it->second;
+ }
+ }
+ return 0;
+}
+
+MapItem* MapWrapper::Previous(MapItem* item) const
+{
+ if (item == 0)
+ {
+ return 0;
+ }
+
+ std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
+ if ((it != map_.end()) &&
+ (it != map_.begin()))
+ {
+ --it;
+ return it->second;
+ }
+ return 0;
+}
+
+MapItem* MapWrapper::Find(int id) const
+{
+ std::map<int, MapItem*>::const_iterator it = map_.find(id);
+ if (it != map_.end())
+ {
+ return it->second;
+ }
+ return 0;
+}
+
+int MapWrapper::Erase(MapItem* item)
+{
+ if (item == 0)
+ {
+ return -1;
+ }
+ std::map<int, MapItem*>::iterator it = map_.find(item->item_id_);
+ if (it != map_.end())
+ {
+ delete it->second;
+ map_.erase(it);
+ return 0;
+ }
+ return -1;
+}
+
+int MapWrapper::Erase(const int id)
+{
+ std::map<int, MapItem*>::iterator it = map_.find(id);
+ if (it != map_.end())
+ {
+ delete it->second;
+ map_.erase(it);
+ return 0;
+ }
+ return -1;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/map_no_stl.cc b/trunk/src/system_wrappers/source/map_no_stl.cc
new file mode 100644
index 0000000..ef93a1f
--- /dev/null
+++ b/trunk/src/system_wrappers/source/map_no_stl.cc
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "map_no_stl.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+MapNoStlItem::MapNoStlItem(int id, void* item)
+ : next_(0),
+ prev_(0),
+ item_id_(id),
+ item_ptr_(item)
+{
+}
+
+MapNoStlItem::~MapNoStlItem()
+{
+}
+
+void* MapNoStlItem::GetItem()
+{
+ return item_ptr_;
+}
+
+int MapNoStlItem::GetId()
+{
+ return item_id_;
+}
+
+unsigned int MapNoStlItem::GetUnsignedId()
+{
+ return static_cast<unsigned int>(item_id_);
+}
+
+void MapNoStlItem::SetItem(void* ptr)
+{
+ item_ptr_ = ptr;
+}
+
+MapNoStl::MapNoStl()
+ : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
+ first_(0),
+ last_(0),
+ size_(0)
+{
+}
+
+MapNoStl::~MapNoStl()
+{
+ if (First())
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
+ "Potential memory leak in MapNoStl");
+ while (Erase(First()) == 0)
+ {}
+ }
+ delete critical_section_;
+}
+
+int MapNoStl::Size() const
+{
+ return size_;
+}
+
+int MapNoStl::Insert(int id, void* ptr)
+{
+ MapNoStlItem* new_item = new MapNoStlItem(id, ptr);
+
+ CriticalSectionScoped lock(critical_section_);
+ MapNoStlItem* item = first_;
+ size_++;
+ if (!item)
+ {
+ first_ = new_item;
+ last_ = new_item;
+ return 0;
+ }
+ while(item->next_)
+ {
+ // Three scenarios
+ // 1. Item should be inserted first.
+ // 2. Item should be inserted between two items
+ // 3. Item should be inserted last
+ if (item->GetId() > id)
+ {
+ new_item->next_ = item;
+ item->prev_ = new_item;
+ if (item == first_)
+ {
+ first_ = new_item;
+ }
+ else
+ {
+ new_item->prev_ = item->prev_;
+ new_item->prev_->next_ = new_item;
+ }
+ return 0;
+ }
+ item = item->next_;
+ }
+ // 3
+ item->next_ = new_item;
+ new_item->prev_ = item;
+ last_ = new_item;
+ return 0;
+}
+
+MapNoStlItem* MapNoStl::First() const
+{
+ return first_;
+}
+
+MapNoStlItem* MapNoStl::Last() const
+{
+ return last_;
+}
+
+MapNoStlItem* MapNoStl::Next(MapNoStlItem* item) const
+{
+ if (!item)
+ {
+ return 0;
+ }
+ return item->next_;
+}
+
+MapNoStlItem* MapNoStl::Previous(MapNoStlItem* item) const
+{
+ if (!item)
+ {
+ return 0;
+ }
+ return item->prev_;
+}
+
+MapNoStlItem* MapNoStl::Find(int id) const
+{
+ CriticalSectionScoped lock(critical_section_);
+ MapNoStlItem* item = Locate(id);
+ return item;
+}
+
+int MapNoStl::Erase(MapNoStlItem* item)
+{
+ if(!item)
+ {
+ return -1;
+ }
+ CriticalSectionScoped lock(critical_section_);
+ return Remove(item);
+}
+
+int MapNoStl::Erase(const int id)
+{
+ CriticalSectionScoped lock(critical_section_);
+ MapNoStlItem* item = Locate(id);
+ if(!item)
+ {
+ return -1;
+ }
+ return Remove(item);
+}
+
+MapNoStlItem* MapNoStl::Locate(int id) const
+{
+ MapNoStlItem* item = first_;
+ while(item)
+ {
+ if (item->GetId() == id)
+ {
+ return item;
+ }
+ item = item->next_;
+ }
+ return 0;
+}
+
+int MapNoStl::Remove(MapNoStlItem* item)
+{
+ if (!item)
+ {
+ return -1;
+ }
+ size_--;
+ MapNoStlItem* previous_item = item->prev_;
+ MapNoStlItem* next_item = item->next_;
+ if (!previous_item)
+ {
+ next_item->prev_ = 0;
+ first_ = next_item;
+ }
+ else
+ {
+ previous_item->next_ = next_item;
+ }
+ if (!next_item)
+ {
+ previous_item->next_ = 0;
+ last_ = previous_item;
+ }
+ else
+ {
+ next_item->prev_ = previous_item;
+ }
+ delete item;
+ return 0;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/map_no_stl.h b/trunk/src/system_wrappers/source/map_no_stl.h
new file mode 100644
index 0000000..51bc011
--- /dev/null
+++ b/trunk/src/system_wrappers/source/map_no_stl.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
+
+#include "constructor_magic.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class MapNoStlItem
+{
+friend class Map;
+
+public:
+ MapNoStlItem(int id, void* ptr);
+ virtual ~MapNoStlItem();
+ void* GetItem();
+ int GetId();
+ unsigned int GetUnsignedId();
+ void SetItem(void* ptr);
+
+protected:
+ MapNoStlItem* next_;
+ MapNoStlItem* prev_;
+
+private:
+ int item_id_;
+ void* item_ptr_;
+ DISALLOW_COPY_AND_ASSIGN(MapNoStlItem);
+};
+
+class MapNoStl
+{
+public:
+ MapNoStl();
+ virtual ~MapNoStl();
+
+ // MapWrapper functions.
+ int Insert(int id, void* ptr);
+ int Erase(MapNoStlItem* item);
+ int Erase(int id);
+ int Size() const;
+ MapNoStlItem* First() const;
+ MapNoStlItem* Last() const;
+ MapNoStlItem* Next(MapNoStlItem* item) const;
+ MapNoStlItem* Previous(MapNoStlItem* item) const;
+ MapNoStlItem* Find(int id) const;
+
+private:
+ MapNoStlItem* Locate(int id) const;
+ int Remove(MapNoStlItem* item);
+
+ CriticalSection* critical_section_;
+ MapNoStlItem* first_;
+ MapNoStlItem* last_;
+ int size_;
+ DISALLOW_COPY_AND_ASSIGN(MapNoStl);
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
diff --git a/trunk/src/system_wrappers/source/map_unittest.cc b/trunk/src/system_wrappers/source/map_unittest.cc
new file mode 100644
index 0000000..1c85a92
--- /dev/null
+++ b/trunk/src/system_wrappers/source/map_unittest.cc
@@ -0,0 +1,231 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+#include "map_wrapper.h"
+
+using ::webrtc::MapWrapper;
+using ::webrtc::MapItem;
+
+const int kNumberOfElements = 10;
+
+int* ItemPointer(MapItem* item) {
+ if (item == NULL) {
+ return NULL;
+ }
+ return reinterpret_cast<int*>(item->GetItem());
+}
+
+bool DeleteItemContent(MapItem* item) {
+ if(item == NULL) {
+ return false;
+ }
+ int* value_ptr = ItemPointer(item);
+ delete value_ptr;
+ return true;
+}
+
+int ItemValue(MapItem* item) {
+ if (item == NULL) {
+ return -1;
+ }
+ const int* value_ptr = ItemPointer(item);
+ if (value_ptr == 0) {
+ return -1;
+ }
+ return *value_ptr;
+}
+
+void PrintToConsole(const char* message, bool supress) {
+ if (supress) {
+ return;
+ }
+ printf("%s", message);
+}
+
+bool CreateAscendingMap(MapWrapper* ascending_map) {
+ int* insert_value = NULL;
+ for (int i = 0; i < kNumberOfElements; ++i) {
+ insert_value = new int;
+ if (insert_value == NULL) {
+ return false;
+ }
+ *insert_value = i;
+ if (0 != ascending_map->Insert(
+ i,
+ reinterpret_cast<void*>(insert_value))) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool ClearMap(MapWrapper* clear_map) {
+ bool success = true;
+ while (clear_map->Size() != 0) {
+ MapItem* remove_item = clear_map->First();
+ if (remove_item == NULL) {
+ return false;
+ }
+ if (!DeleteItemContent(remove_item)) {
+ success = false;
+ }
+ if (clear_map->Erase(remove_item) != 0) {
+ return false;
+ }
+ }
+ return success;
+}
+
+void PrintMapItem(MapItem* item, bool supress) {
+ const int id = item->GetId();
+ const int value = ItemValue(item);
+ char print_buffer[16];
+ sprintf(print_buffer, "(%3i,%3i) ", id, value);
+ PrintToConsole(print_buffer, supress);
+}
+
+// Succeeds only if all the items were printed.
+bool PrintMap(const MapWrapper& print_map, bool supress) {
+ const int elements_to_print = print_map.Size();
+ int elements_printed = 0;
+ MapItem* item = print_map.First();
+ PrintToConsole("[", supress);
+ while (item != NULL) {
+ PrintMapItem(item, supress);
+ ++elements_printed;
+ item = print_map.Next(item);
+ }
+ PrintToConsole("]\n", supress);
+ return elements_printed == elements_to_print;
+}
+
+// Succeeds only if all the items were printed.
+bool ReversePrintMap(const MapWrapper& print_map, bool supress) {
+ const int elements_to_print = print_map.Size();
+ int elements_printed = 0;
+ MapItem* item = print_map.Last();
+ PrintToConsole("[", supress);
+ while (item != NULL) {
+ PrintMapItem(item, supress);
+ ++elements_printed;
+ item = print_map.Previous(item);
+ }
+ PrintToConsole("]\n", supress);
+ return elements_printed == elements_to_print;
+}
+
+// Returns true if the map items contain the same item.
+bool CompareItems(MapItem* lhs_item, MapItem* rhs_item) {
+ if ((lhs_item == NULL) || (rhs_item == NULL)) {
+ return false;
+ }
+ if (lhs_item->GetId() != rhs_item->GetId()) {
+ return false;
+ }
+ return lhs_item->GetItem() == rhs_item->GetItem();
+}
+
+// Returns true if the map contains the same items.
+bool CompareMaps(const MapWrapper& lhs, const MapWrapper& rhs) {
+ const int map_size = lhs.Size();
+ if (map_size != rhs.Size()) {
+ return false;
+ }
+ int item_count = 0;
+ MapItem* lhs_item = lhs.First();
+ while (lhs_item != NULL) {
+ MapItem* rhs_item = rhs.Find(lhs_item->GetId());
+ if (rhs_item == NULL) {
+ return false;
+ }
+ if (!CompareItems(lhs_item, rhs_item)) {
+ return false;
+ }
+ ++item_count;
+ lhs_item = lhs.Next(lhs_item);
+ }
+ return item_count == map_size;
+}
+
+class MapWrapperTest : public ::testing::Test {
+protected:
+ virtual void SetUp() {
+ ASSERT_TRUE(CreateAscendingMap(&ascending_map_));
+ }
+
+ virtual void TearDown() {
+ EXPECT_TRUE(ClearMap(&ascending_map_));
+ }
+ MapWrapper ascending_map_;
+};
+
+TEST_F(MapWrapperTest,RemoveTest) {
+ // Erase using int id
+ { // Create local scope to avoid accidental re-use
+ MapItem* item_first = ascending_map_.First();
+ ASSERT_FALSE(item_first == NULL);
+ const int first_value_id = item_first->GetId();
+ const int first_value = ItemValue(item_first);
+ EXPECT_TRUE(first_value == 0);
+ EXPECT_EQ(first_value_id,first_value);
+ EXPECT_FALSE(NULL == ascending_map_.Find(first_value_id));
+ EXPECT_TRUE(DeleteItemContent(item_first));
+ ascending_map_.Erase(first_value_id);
+ EXPECT_TRUE(NULL == ascending_map_.Find(first_value_id));
+ EXPECT_EQ(kNumberOfElements-1,ascending_map_.Size());
+ }
+
+ // Erase using MapItem* item
+ MapItem* item_last = ascending_map_.Last();
+ ASSERT_FALSE(item_last == NULL);
+ const int last_value_id = item_last->GetId();
+ const int last_value = ItemValue(item_last);
+ EXPECT_TRUE(last_value == kNumberOfElements - 1);
+ EXPECT_EQ(last_value_id, last_value);
+ EXPECT_FALSE(NULL == ascending_map_.Find(last_value_id));
+ EXPECT_TRUE(DeleteItemContent(item_last));
+ ascending_map_.Erase(last_value_id);
+ EXPECT_TRUE(NULL == ascending_map_.Find(last_value_id));
+ EXPECT_EQ(kNumberOfElements-2,ascending_map_.Size());
+}
+
+TEST_F(MapWrapperTest, PrintTest) {
+ const bool supress = true; // Don't spam the console
+
+ EXPECT_TRUE(PrintMap(ascending_map_, supress));
+ EXPECT_TRUE(ReversePrintMap(ascending_map_, supress));
+}
+
+TEST_F(MapWrapperTest, CopyTest) {
+ MapWrapper compare_map;
+ ASSERT_TRUE(CreateAscendingMap(&compare_map));
+ const int map_size = compare_map.Size();
+ ASSERT_EQ(ascending_map_.Size(), map_size);
+ // CompareMaps compare the pointers not value of the pointers.
+ // (the values are the same since both are ascending maps).
+ EXPECT_FALSE(CompareMaps(compare_map,ascending_map_));
+
+ int copy_count = 0;
+ MapItem* ascend_item = ascending_map_.First();
+ while (ascend_item != NULL) {
+ MapItem* compare_item = compare_map.Find(ascend_item->GetId());
+ ASSERT_FALSE(compare_item == NULL);
+ DeleteItemContent(compare_item);
+ compare_item->SetItem(ascend_item->GetItem());
+ ascend_item = ascending_map_.Next(ascend_item);
+ ++copy_count;
+ }
+ EXPECT_TRUE(CompareMaps(compare_map,ascending_map_));
+ while (compare_map.Erase(compare_map.First()) == 0) {
+ }
+ EXPECT_EQ(map_size, copy_count);
+}
diff --git a/trunk/src/system_wrappers/source/rw_lock.cc b/trunk/src/system_wrappers/source/rw_lock.cc
new file mode 100644
index 0000000..16da0e3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock.cc
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rw_lock_wrapper.h"
+
+#include <assert.h>
+
+#if defined(_WIN32)
+ #include "rw_lock_win.h"
+#else
+ #include "rw_lock_posix.h"
+#endif
+
+namespace webrtc {
+RWLockWrapper* RWLockWrapper::CreateRWLock()
+{
+#ifdef _WIN32
+ RWLockWrapper* lock = new RWLockWindows();
+#else
+ RWLockWrapper* lock = new RWLockPosix();
+#endif
+ if(lock->Init() != 0)
+ {
+ delete lock;
+ assert(false);
+ return NULL;
+ }
+ return lock;
+}
+
+RWLockWrapper::~RWLockWrapper()
+{
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_generic.cc b/trunk/src/system_wrappers/source/rw_lock_generic.cc
new file mode 100644
index 0000000..a468ef3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_generic.cc
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rw_lock_generic.h"
+
+#include "condition_variable_wrapper.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+RWLockWrapperGeneric::RWLockWrapperGeneric()
+ : _readersActive(0),
+ _writerActive(false),
+ _readersWaiting(0),
+ _writersWaiting(0)
+{
+ _critSectPtr = CriticalSectionWrapper::CreateCriticalSection();
+ _readCondPtr = ConditionVariableWrapper::CreateConditionVariable();
+ _writeCondPtr = ConditionVariableWrapper::CreateConditionVariable();
+}
+
+RWLockWrapperGeneric::~RWLockWrapperGeneric()
+{
+ delete _writeCondPtr;
+ delete _readCondPtr;
+ delete _critSectPtr;
+}
+
+int RWLockWrapperGeneric::Init()
+{
+ return 0;
+}
+
+void RWLockWrapperGeneric::AcquireLockExclusive()
+{
+ _critSectPtr->Enter();
+
+ if (_writerActive || _readersActive > 0)
+ {
+ ++_writersWaiting;
+
+ while (_writerActive || _readersActive > 0)
+ {
+ _writeCondPtr->SleepCS(*_critSectPtr);
+ }
+
+ --_writersWaiting;
+ }
+ _writerActive = true;
+ _critSectPtr->Leave();
+}
+
+void RWLockWrapperGeneric::ReleaseLockExclusive()
+{
+ _critSectPtr->Enter();
+
+ _writerActive = false;
+
+ if (_writersWaiting > 0)
+ {
+ _writeCondPtr->Wake();
+
+ }else if (_readersWaiting > 0)
+ {
+ _readCondPtr->WakeAll();
+ }
+ _critSectPtr->Leave();
+}
+
+void RWLockWrapperGeneric::AcquireLockShared()
+{
+ _critSectPtr->Enter();
+
+ if (_writerActive || _writersWaiting > 0)
+ {
+ ++_readersWaiting;
+
+ while (_writerActive || _writersWaiting > 0)
+ {
+ _readCondPtr->SleepCS(*_critSectPtr);
+ }
+ --_readersWaiting;
+ }
+ ++_readersActive;
+ _critSectPtr->Leave();
+}
+
+void RWLockWrapperGeneric::ReleaseLockShared()
+{
+ _critSectPtr->Enter();
+
+ --_readersActive;
+
+ if (_readersActive == 0 && _writersWaiting > 0)
+ {
+ _writeCondPtr->Wake();
+ }
+ _critSectPtr->Leave();
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_generic.h b/trunk/src/system_wrappers/source/rw_lock_generic.h
new file mode 100644
index 0000000..fff5e5d
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_generic.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
+
+#include "rw_lock_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class ConditionVariableWrapper;
+
+class RWLockWrapperGeneric : public RWLockWrapper
+{
+public:
+ RWLockWrapperGeneric();
+ virtual ~RWLockWrapperGeneric();
+
+ virtual void AcquireLockExclusive();
+ virtual void ReleaseLockExclusive();
+
+ virtual void AcquireLockShared();
+ virtual void ReleaseLockShared();
+
+protected:
+ virtual int Init();
+
+private:
+ CriticalSectionWrapper* _critSectPtr;
+ ConditionVariableWrapper* _readCondPtr;
+ ConditionVariableWrapper* _writeCondPtr;
+
+ int _readersActive;
+ bool _writerActive;
+ int _readersWaiting;
+ int _writersWaiting;
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
diff --git a/trunk/src/system_wrappers/source/rw_lock_posix.cc b/trunk/src/system_wrappers/source/rw_lock_posix.cc
new file mode 100644
index 0000000..81a161e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_posix.cc
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rw_lock_posix.h"
+
+namespace webrtc {
+RWLockPosix::RWLockPosix() : _lock()
+{
+}
+
+RWLockPosix::~RWLockPosix()
+{
+ pthread_rwlock_destroy(&_lock);
+}
+
+int RWLockPosix::Init()
+{
+ return pthread_rwlock_init(&_lock, 0);
+}
+
+void RWLockPosix::AcquireLockExclusive()
+{
+ pthread_rwlock_wrlock(&_lock);
+}
+
+void RWLockPosix::ReleaseLockExclusive()
+{
+ pthread_rwlock_unlock(&_lock);
+}
+
+void RWLockPosix::AcquireLockShared()
+{
+ pthread_rwlock_rdlock(&_lock);
+}
+
+void RWLockPosix::ReleaseLockShared()
+{
+ pthread_rwlock_unlock(&_lock);
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_posix.h b/trunk/src/system_wrappers/source/rw_lock_posix.h
new file mode 100644
index 0000000..929bbb8
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_posix.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
+
+#include "rw_lock_wrapper.h"
+
+#include <pthread.h>
+
+namespace webrtc {
+class RWLockPosix : public RWLockWrapper
+{
+public:
+ RWLockPosix();
+ virtual ~RWLockPosix();
+
+ virtual void AcquireLockExclusive();
+ virtual void ReleaseLockExclusive();
+
+ virtual void AcquireLockShared();
+ virtual void ReleaseLockShared();
+
+protected:
+ virtual int Init();
+
+private:
+ pthread_rwlock_t _lock;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/rw_lock_win.cc b/trunk/src/system_wrappers/source/rw_lock_win.cc
new file mode 100644
index 0000000..82cd0ac
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_win.cc
@@ -0,0 +1,186 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rw_lock_win.h"
+
+#include "critical_section_wrapper.h"
+#include "condition_variable_wrapper.h"
+#include "trace.h"
+
+// TODO (hellner) why not just use the rw_lock_generic.cc solution if
+// native is not supported? Unnecessary redundancy!
+
+namespace webrtc {
+bool RWLockWindows::_winSupportRWLockPrimitive = false;
+static HMODULE library = NULL;
+
+PInitializeSRWLock _PInitializeSRWLock;
+PAcquireSRWLockExclusive _PAcquireSRWLockExclusive;
+PAcquireSRWLockShared _PAcquireSRWLockShared;
+PReleaseSRWLockShared _PReleaseSRWLockShared;
+PReleaseSRWLockExclusive _PReleaseSRWLockExclusive;
+
+RWLockWindows::RWLockWindows()
+ : _critSectPtr(NULL),
+ _readCondPtr(NULL),
+ _writeCondPtr(NULL),
+ _readersActive(0),
+ _writerActive(false),
+ _readersWaiting(0),
+ _writersWaiting(0)
+{
+}
+
+RWLockWindows::~RWLockWindows()
+{
+ delete _writeCondPtr;
+ delete _readCondPtr;
+ delete _critSectPtr;
+}
+
+int RWLockWindows::Init()
+{
+ if(!library)
+ {
+ // Use native implementation if supported (i.e Vista+)
+ library = LoadLibrary(TEXT("Kernel32.dll"));
+ if(library)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Loaded Kernel.dll");
+
+ _PInitializeSRWLock =
+ (PInitializeSRWLock)GetProcAddress(
+ library,
+ "InitializeSRWLock");
+
+ _PAcquireSRWLockExclusive =
+ (PAcquireSRWLockExclusive)GetProcAddress(
+ library,
+ "AcquireSRWLockExclusive");
+ _PReleaseSRWLockExclusive =
+ (PReleaseSRWLockExclusive)GetProcAddress(
+ library,
+ "ReleaseSRWLockExclusive");
+ _PAcquireSRWLockShared =
+ (PAcquireSRWLockShared)GetProcAddress(
+ library,
+ "AcquireSRWLockShared");
+ _PReleaseSRWLockShared =
+ (PReleaseSRWLockShared)GetProcAddress(
+ library,
+ "ReleaseSRWLockShared");
+
+ if( _PInitializeSRWLock &&
+ _PAcquireSRWLockExclusive &&
+ _PReleaseSRWLockExclusive &&
+ _PAcquireSRWLockShared &&
+ _PReleaseSRWLockShared )
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Loaded Simple RW Lock");
+ _winSupportRWLockPrimitive = true;
+ }
+ }
+ }
+ if(_winSupportRWLockPrimitive)
+ {
+ _PInitializeSRWLock(&_lock);
+ } else {
+ _critSectPtr = CriticalSectionWrapper::CreateCriticalSection();
+ _readCondPtr = ConditionVariableWrapper::CreateConditionVariable();
+ _writeCondPtr = ConditionVariableWrapper::CreateConditionVariable();
+ }
+ return 0;
+}
+
+void RWLockWindows::AcquireLockExclusive()
+{
+ if (_winSupportRWLockPrimitive)
+ {
+ _PAcquireSRWLockExclusive(&_lock);
+ } else {
+ _critSectPtr->Enter();
+
+ if (_writerActive || _readersActive > 0)
+ {
+ ++_writersWaiting;
+ while (_writerActive || _readersActive > 0)
+ {
+ _writeCondPtr->SleepCS(*_critSectPtr);
+ }
+ --_writersWaiting;
+ }
+ _writerActive = true;
+ _critSectPtr->Leave();
+ }
+}
+
+void RWLockWindows::ReleaseLockExclusive()
+{
+ if(_winSupportRWLockPrimitive)
+ {
+ _PReleaseSRWLockExclusive(&_lock);
+ } else {
+ _critSectPtr->Enter();
+ _writerActive = false;
+ if (_writersWaiting > 0)
+ {
+ _writeCondPtr->Wake();
+
+ }else if (_readersWaiting > 0) {
+ _readCondPtr->WakeAll();
+ }
+ _critSectPtr->Leave();
+ }
+}
+
+void RWLockWindows::AcquireLockShared()
+{
+ if(_winSupportRWLockPrimitive)
+ {
+ _PAcquireSRWLockShared(&_lock);
+ } else
+ {
+ _critSectPtr->Enter();
+ if (_writerActive || _writersWaiting > 0)
+ {
+ ++_readersWaiting;
+
+ while (_writerActive || _writersWaiting > 0)
+ {
+ _readCondPtr->SleepCS(*_critSectPtr);
+ }
+ --_readersWaiting;
+ }
+ ++_readersActive;
+ _critSectPtr->Leave();
+ }
+}
+
+void RWLockWindows::ReleaseLockShared()
+{
+ if(_winSupportRWLockPrimitive)
+ {
+ _PReleaseSRWLockShared(&_lock);
+ } else
+ {
+ _critSectPtr->Enter();
+
+ --_readersActive;
+
+ if (_readersActive == 0 && _writersWaiting > 0)
+ {
+ _writeCondPtr->Wake();
+ }
+ _critSectPtr->Leave();
+ }
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_win.h b/trunk/src/system_wrappers/source/rw_lock_win.h
new file mode 100644
index 0000000..dc5355e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/rw_lock_win.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
+
+#include "rw_lock_wrapper.h"
+
+#include <Windows.h>
+
+#if !defined(RTL_SRWLOCK_INIT)
+ typedef struct _RTL_SRWLOCK
+ {
+ void* Ptr;
+ } RTL_SRWLOCK, *PRTL_SRWLOCK;
+ typedef RTL_SRWLOCK SRWLOCK, *PSRWLOCK;
+#endif
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class ConditionVariableWrapper;
+
+typedef void (WINAPI *PInitializeSRWLock)(PSRWLOCK);
+
+typedef void (WINAPI *PAcquireSRWLockExclusive)(PSRWLOCK);
+typedef void (WINAPI *PReleaseSRWLockExclusive)(PSRWLOCK);
+
+typedef void (WINAPI *PAcquireSRWLockShared)(PSRWLOCK);
+typedef void (WINAPI *PReleaseSRWLockShared)(PSRWLOCK);
+
+
+class RWLockWindows :public RWLockWrapper
+{
+public:
+ RWLockWindows();
+ virtual ~RWLockWindows();
+
+ virtual void AcquireLockExclusive();
+ virtual void ReleaseLockExclusive();
+
+ virtual void AcquireLockShared();
+ virtual void ReleaseLockShared();
+
+protected:
+ virtual int Init();
+
+private:
+ // For native implementation.
+ static bool _winSupportRWLockPrimitive;
+ SRWLOCK _lock;
+
+ // Genric implementation, fallback if native is not supported.
+ CriticalSectionWrapper* _critSectPtr;
+ ConditionVariableWrapper* _readCondPtr;
+ ConditionVariableWrapper* _writeCondPtr;
+
+ int _readersActive;
+ bool _writerActive;
+ int _readersWaiting;
+ int _writersWaiting;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/set_thread_name_win.h b/trunk/src/system_wrappers/source/set_thread_name_win.h
new file mode 100644
index 0000000..a46f4d6
--- /dev/null
+++ b/trunk/src/system_wrappers/source/set_thread_name_win.h
@@ -0,0 +1,43 @@
+/*
+ * Use of this source code is governed by the MICROSOFT LIMITED PUBLIC LICENSE
+ * copyright license which can be found in the LICENSE file in the
+ * third_party_mods/mslpl directory of the source tree or at
+ * http://msdn.microsoft.com/en-us/cc300389.aspx#P.
+ */
+/*
+ * The original code can be found here:
+ * http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
+
+namespace webrtc {
+
+struct THREADNAME_INFO
+{
+ DWORD dwType; // must be 0x1000
+ LPCSTR szName; // pointer to name (in user addr space)
+ DWORD dwThreadID; // thread ID (-1 = caller thread)
+ DWORD dwFlags; // reserved for future use, must be zero
+};
+
+void SetThreadName(DWORD dwThreadID, LPCSTR szThreadName)
+{
+ THREADNAME_INFO info;
+ info.dwType = 0x1000;
+ info.szName = szThreadName;
+ info.dwThreadID = dwThreadID;
+ info.dwFlags = 0;
+
+ __try
+ {
+ RaiseException(0x406D1388, 0, sizeof(info) / sizeof(DWORD),
+ (ULONG_PTR*)&info);
+ }
+ __except (EXCEPTION_CONTINUE_EXECUTION)
+ {
+ }
+}
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
diff --git a/trunk/src/system_wrappers/source/sort.cc b/trunk/src/system_wrappers/source/sort.cc
new file mode 100644
index 0000000..34aa437
--- /dev/null
+++ b/trunk/src/system_wrappers/source/sort.cc
@@ -0,0 +1,549 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// When the platform supports STL, the functions are implemented using a
+// templated spreadsort algorithm (http://sourceforge.net/projects/spreadsort/),
+// part of the Boost C++ library collection. Otherwise, the C standard library's
+// qsort() will be used.
+
+#include "sort.h"
+
+#include <cassert>
+#include <cstring> // memcpy
+#include <new> // nothrow new
+
+#ifdef NO_STL
+#include <cstdlib> // qsort
+#else
+#include <algorithm> // std::sort
+#include <vector>
+#include "spreadsort.hpp" // TODO (ajm) upgrade to spreadsortv2.
+#endif
+
+#ifdef NO_STL
+#define COMPARE_DEREFERENCED(XT, YT) \
+ do \
+ { \
+ if ((XT) > (YT)) \
+ { \
+ return 1; \
+ } \
+ else if ((XT) < (YT)) \
+ { \
+ return -1; \
+ } \
+ \
+ return 0; \
+ } \
+ while(0)
+
+#define COMPARE_FOR_QSORT(X, Y, TYPE) \
+ do \
+ { \
+ TYPE xT = static_cast<TYPE>(*static_cast<const TYPE*>(X)); \
+ TYPE yT = static_cast<TYPE>(*static_cast<const TYPE*>(Y)); \
+ COMPARE_DEREFERENCED(xT, yT); \
+ } \
+ while(0)
+
+#define COMPARE_KEY_FOR_QSORT(SORT_KEY_X, SORT_KEY_Y, TYPE) \
+ do \
+ { \
+ TYPE xT = static_cast<TYPE>(*static_cast<TYPE*> \
+ (static_cast<const SortKey*>(SORT_KEY_X)->key)); \
+ TYPE yT = static_cast<TYPE>(*static_cast<TYPE*> \
+ (static_cast<const SortKey*>(SORT_KEY_Y)->key)); \
+ COMPARE_DEREFERENCED(xT, yT); \
+ } \
+ while(0)
+
+#define KEY_QSORT(SORT_KEY, KEY, NUM_OF_ELEMENTS, KEY_TYPE, COMPARE_FUNC) \
+ do \
+ { \
+ KEY_TYPE* keyT = (KEY_TYPE*)(key); \
+ for (WebRtc_UWord32 i = 0; i < (NUM_OF_ELEMENTS); i++) \
+ { \
+ ptrSortKey[i].key = &keyT[i]; \
+ ptrSortKey[i].index = i; \
+ } \
+ \
+ qsort((SORT_KEY), (NUM_OF_ELEMENTS), sizeof(SortKey), (COMPARE_FUNC));\
+ } \
+ while(0)
+#endif
+
+namespace webrtc
+{
+#ifdef NO_STL
+ struct SortKey
+ {
+ void* key;
+ WebRtc_UWord32 index;
+ };
+#else
+ template<typename KeyType>
+ struct SortKey
+ {
+ KeyType key;
+ WebRtc_UWord32 index;
+ };
+#endif
+
+ namespace // Unnamed namespace provides internal linkage.
+ {
+#ifdef NO_STL
+ int CompareWord8(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_Word8);
+ }
+
+ int CompareUWord8(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_UWord8);
+ }
+
+ int CompareWord16(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_Word16);
+ }
+
+ int CompareUWord16(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_UWord16);
+ }
+
+ int CompareWord32(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_Word32);
+ }
+
+ int CompareUWord32(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_UWord32);
+ }
+
+ int CompareWord64(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_Word64);
+ }
+
+ int CompareUWord64(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, WebRtc_UWord64);
+ }
+
+ int CompareFloat32(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, float);
+ }
+
+ int CompareFloat64(const void* x, const void* y)
+ {
+ COMPARE_FOR_QSORT(x, y, double);
+ }
+
+ int CompareKeyWord8(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word8);
+ }
+
+ int CompareKeyUWord8(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord8);
+ }
+
+ int CompareKeyWord16(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word16);
+ }
+
+ int CompareKeyUWord16(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord16);
+ }
+
+ int CompareKeyWord32(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word32);
+ }
+
+ int CompareKeyUWord32(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord32);
+ }
+
+ int CompareKeyWord64(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word64);
+ }
+
+ int CompareKeyUWord64(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord64);
+ }
+
+ int CompareKeyFloat32(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, float);
+ }
+
+ int CompareKeyFloat64(const void* sortKeyX, const void* sortKeyY)
+ {
+ COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, double);
+ }
+#else
+ template <typename KeyType>
+ struct KeyLessThan
+ {
+ bool operator()(const SortKey<KeyType>& sortKeyX,
+ const SortKey<KeyType>& sortKeyY) const
+ {
+ return sortKeyX.key < sortKeyY.key;
+ }
+ };
+
+ template <typename KeyType>
+ struct KeyRightShift
+ {
+ KeyType operator()(const SortKey<KeyType>& sortKey,
+ const unsigned offset) const
+ {
+ return sortKey.key >> offset;
+ }
+ };
+
+ template <typename DataType>
+ inline void IntegerSort(void* data, WebRtc_UWord32 numOfElements)
+ {
+ DataType* dataT = static_cast<DataType*>(data);
+ boost::integer_sort(dataT, dataT + numOfElements);
+ }
+
+ template <typename DataType, typename IntegerType>
+ inline void FloatSort(void* data, WebRtc_UWord32 numOfElements)
+ {
+ DataType* dataT = static_cast<DataType*>(data);
+ IntegerType cVal = 0;
+ boost::float_sort_cast(dataT, dataT + numOfElements, cVal);
+ }
+
+ template <typename DataType>
+ inline void StdSort(void* data, WebRtc_UWord32 numOfElements)
+ {
+ DataType* dataT = static_cast<DataType*>(data);
+ std::sort(dataT, dataT + numOfElements);
+ }
+
+ template<typename KeyType>
+ inline WebRtc_Word32 SetupKeySort(void* key,
+ SortKey<KeyType>*& ptrSortKey,
+ WebRtc_UWord32 numOfElements)
+ {
+ ptrSortKey = new(std::nothrow) SortKey<KeyType>[numOfElements];
+ if (ptrSortKey == NULL)
+ {
+ return -1;
+ }
+
+ KeyType* keyT = static_cast<KeyType*>(key);
+ for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
+ {
+ ptrSortKey[i].key = keyT[i];
+ ptrSortKey[i].index = i;
+ }
+
+ return 0;
+ }
+
+ template<typename KeyType>
+ inline WebRtc_Word32 TeardownKeySort(void* data,
+ SortKey<KeyType>* ptrSortKey,
+ WebRtc_UWord32 numOfElements, WebRtc_UWord32 sizeOfElement)
+ {
+ WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
+ WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
+ [numOfElements * sizeOfElement];
+ if (ptrDataSorted == NULL)
+ {
+ return -1;
+ }
+
+ for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
+ {
+ memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
+ ptrSortKey[i].index * sizeOfElement, sizeOfElement);
+ }
+ memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
+ delete[] ptrSortKey;
+ delete[] ptrDataSorted;
+ return 0;
+ }
+
+ template<typename KeyType>
+ inline WebRtc_Word32 IntegerKeySort(void* data, void* key,
+ WebRtc_UWord32 numOfElements,
+ WebRtc_UWord32 sizeOfElement)
+ {
+ SortKey<KeyType>* ptrSortKey;
+ if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
+ {
+ return -1;
+ }
+
+ boost::integer_sort(ptrSortKey, ptrSortKey + numOfElements,
+ KeyRightShift<KeyType>(), KeyLessThan<KeyType>());
+
+ if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
+ sizeOfElement) != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+ }
+
+ template<typename KeyType>
+ inline WebRtc_Word32 StdKeySort(void* data, void* key,
+ WebRtc_UWord32 numOfElements,
+ WebRtc_UWord32 sizeOfElement)
+ {
+ SortKey<KeyType>* ptrSortKey;
+ if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
+ {
+ return -1;
+ }
+
+ std::sort(ptrSortKey, ptrSortKey + numOfElements,
+ KeyLessThan<KeyType>());
+
+ if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
+ sizeOfElement) != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+ }
+#endif
+ }
+
+ WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type type)
+ {
+ if (data == NULL)
+ {
+ return -1;
+ }
+
+#ifdef NO_STL
+ switch (type)
+ {
+ case TYPE_Word8:
+ qsort(data, numOfElements, sizeof(WebRtc_Word8), CompareWord8);
+ break;
+ case TYPE_UWord8:
+ qsort(data, numOfElements, sizeof(WebRtc_UWord8), CompareUWord8);
+ break;
+ case TYPE_Word16:
+ qsort(data, numOfElements, sizeof(WebRtc_Word16), CompareWord16);
+ break;
+ case TYPE_UWord16:
+ qsort(data, numOfElements, sizeof(WebRtc_UWord16), CompareUWord16);
+ break;
+ case TYPE_Word32:
+ qsort(data, numOfElements, sizeof(WebRtc_Word32), CompareWord32);
+ break;
+ case TYPE_UWord32:
+ qsort(data, numOfElements, sizeof(WebRtc_UWord32), CompareUWord32);
+ break;
+ case TYPE_Word64:
+ qsort(data, numOfElements, sizeof(WebRtc_Word64), CompareWord64);
+ break;
+ case TYPE_UWord64:
+ qsort(data, numOfElements, sizeof(WebRtc_UWord64), CompareUWord64);
+ break;
+ case TYPE_Float32:
+ qsort(data, numOfElements, sizeof(float), CompareFloat32);
+ break;
+ case TYPE_Float64:
+ qsort(data, numOfElements, sizeof(double), CompareFloat64);
+ break;
+ default:
+ return -1;
+ }
+#else
+ // Fall back to std::sort for 64-bit types and floats due to compiler
+ // warnings and VS 2003 build crashes respectively with spreadsort.
+ switch (type)
+ {
+ case TYPE_Word8:
+ IntegerSort<WebRtc_Word8>(data, numOfElements);
+ break;
+ case TYPE_UWord8:
+ IntegerSort<WebRtc_UWord8>(data, numOfElements);
+ break;
+ case TYPE_Word16:
+ IntegerSort<WebRtc_Word16>(data, numOfElements);
+ break;
+ case TYPE_UWord16:
+ IntegerSort<WebRtc_UWord16>(data, numOfElements);
+ break;
+ case TYPE_Word32:
+ IntegerSort<WebRtc_Word32>(data, numOfElements);
+ break;
+ case TYPE_UWord32:
+ IntegerSort<WebRtc_UWord32>(data, numOfElements);
+ break;
+ case TYPE_Word64:
+ StdSort<WebRtc_Word64>(data, numOfElements);
+ break;
+ case TYPE_UWord64:
+ StdSort<WebRtc_UWord64>(data, numOfElements);
+ break;
+ case TYPE_Float32:
+ StdSort<float>(data, numOfElements);
+ break;
+ case TYPE_Float64:
+ StdSort<double>(data, numOfElements);
+ break;
+ }
+#endif
+ return 0;
+ }
+
+ WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
+ WebRtc_UWord32 sizeOfElement, Type keyType)
+ {
+ if (data == NULL)
+ {
+ return -1;
+ }
+
+ if (key == NULL)
+ {
+ return -1;
+ }
+
+ if ((WebRtc_UWord64)numOfElements * sizeOfElement > 0xffffffff)
+ {
+ return -1;
+ }
+
+#ifdef NO_STL
+ SortKey* ptrSortKey = new(std::nothrow) SortKey[numOfElements];
+ if (ptrSortKey == NULL)
+ {
+ return -1;
+ }
+
+ switch (keyType)
+ {
+ case TYPE_Word8:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word8,
+ CompareKeyWord8);
+ break;
+ case TYPE_UWord8:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord8,
+ CompareKeyUWord8);
+ break;
+ case TYPE_Word16:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word16,
+ CompareKeyWord16);
+ break;
+ case TYPE_UWord16:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord16,
+ CompareKeyUWord16);
+ break;
+ case TYPE_Word32:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word32,
+ CompareKeyWord32);
+ break;
+ case TYPE_UWord32:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord32,
+ CompareKeyUWord32);
+ break;
+ case TYPE_Word64:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word64,
+ CompareKeyWord64);
+ break;
+ case TYPE_UWord64:
+ KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord64,
+ CompareKeyUWord64);
+ break;
+ case TYPE_Float32:
+ KEY_QSORT(ptrSortKey, key, numOfElements, float,
+ CompareKeyFloat32);
+ break;
+ case TYPE_Float64:
+ KEY_QSORT(ptrSortKey, key, numOfElements, double,
+ CompareKeyFloat64);
+ break;
+ default:
+ return -1;
+ }
+
+ // Shuffle into sorted position based on index map.
+ WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
+ WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
+ [numOfElements * sizeOfElement];
+ if (ptrDataSorted == NULL)
+ {
+ return -1;
+ }
+
+ for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
+ {
+ memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
+ ptrSortKey[i].index * sizeOfElement, sizeOfElement);
+ }
+ memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
+
+ delete[] ptrSortKey;
+ delete[] ptrDataSorted;
+
+ return 0;
+#else
+ // Fall back to std::sort for 64-bit types and floats due to compiler
+ // warnings and errors respectively with spreadsort.
+ switch (keyType)
+ {
+ case TYPE_Word8:
+ return IntegerKeySort<WebRtc_Word8>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_UWord8:
+ return IntegerKeySort<WebRtc_UWord8>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_Word16:
+ return IntegerKeySort<WebRtc_Word16>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_UWord16:
+ return IntegerKeySort<WebRtc_UWord16>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_Word32:
+ return IntegerKeySort<WebRtc_Word32>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_UWord32:
+ return IntegerKeySort<WebRtc_UWord32>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_Word64:
+ return StdKeySort<WebRtc_Word64>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_UWord64:
+ return StdKeySort<WebRtc_UWord64>(data, key, numOfElements,
+ sizeOfElement);
+ case TYPE_Float32:
+ return StdKeySort<float>(data, key, numOfElements, sizeOfElement);
+ case TYPE_Float64:
+ return StdKeySort<double>(data, key, numOfElements, sizeOfElement);
+ }
+ assert(false);
+ return -1;
+#endif
+ }
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp b/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp
new file mode 100644
index 0000000..fa81ece
--- /dev/null
+++ b/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp
@@ -0,0 +1,42 @@
+/*Boost Software License - Version 1.0 - August 17th, 2003
+
+Permission is hereby granted, free of charge, to any person or organization
+obtaining a copy of the software and accompanying documentation covered by
+this license (the "Software") to use, reproduce, display, distribute,
+execute, and transmit the Software, and to prepare derivative works of the
+Software, and to permit third-parties to whom the Software is furnished to
+do so, all subject to the following:
+
+The copyright notices in the Software and this entire statement, including
+the above license grant, this restriction and the following disclaimer,
+must be included in all copies of the Software, in whole or in part, and
+all derivative works of the Software, unless such copies or derivative
+works are solely in the form of machine-executable object code generated by
+a source language processor.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
+SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
+FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
+ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+DEALINGS IN THE SOFTWARE.*/
+#ifndef BOOST_SPREADSORT_CONSTANTS
+#define BOOST_SPREADSORT_CONSTANTS
+namespace boost {
+namespace detail {
+//Tuning constants
+//Sets the minimum number of items per bin.
+static const unsigned LOG_MEAN_BIN_SIZE = 2;
+//This should be tuned to your processor cache; if you go too large you get cache misses on bins
+//The smaller this number, the less worst-case memory usage. If too small, too many recursions slow down spreadsort
+static const unsigned MAX_SPLITS = 10;
+//Used to force a comparison-based sorting for small bins, if it's faster. Minimum value 0
+static const unsigned LOG_MIN_SPLIT_COUNT = 5;
+//There is a minimum size below which it is not worth using spreadsort
+static const long MIN_SORT_SIZE = 1000;
+//This is the constant on the log base n of m calculation; make this larger the faster std::sort is relative to spreadsort
+static const unsigned LOG_CONST = 2;
+}
+}
+#endif
diff --git a/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp b/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
new file mode 100644
index 0000000..2d1529a
--- /dev/null
+++ b/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
@@ -0,0 +1,1688 @@
+//Templated spread_sort library
+
+// Copyright Steven J. Ross 2001 - 2009.
+// Distributed under the Boost Software License, Version 1.0.
+// (See accompanying file LICENSE_1_0.txt or copy at
+// http://www.boost.org/LICENSE_1_0.txt)
+
+// See http://www.boost.org/ for updates, documentation, and revision history.
+
+/*
+Some improvements suggested by:
+Phil Endecott and Frank Gennari
+Cygwin fix provided by:
+Scott McMurray
+*/
+
+#ifndef BOOST_SPREAD_SORT_H
+#define BOOST_SPREAD_SORT_H
+#include <algorithm>
+#include <vector>
+#include "constants.hpp"
+#include <cstring>
+
+namespace boost {
+ namespace detail {
+ //This only works on unsigned data types
+ template <typename T>
+ inline unsigned
+ rough_log_2_size(const T& input)
+ {
+ unsigned result = 0;
+ //The && is necessary on some compilers to avoid infinite loops; it doesn't significantly impair performance
+ while((input >> result) && (result < (8*sizeof(T)))) ++result;
+ return result;
+ }
+
+ //Gets the maximum size which we'll call spread_sort on to control worst-case performance
+ //Maintains both a minimum size to recurse and a check of distribution size versus count
+ //This is called for a set of bins, instead of bin-by-bin, to avoid performance overhead
+ inline size_t
+ get_max_count(unsigned log_range, size_t count)
+ {
+ unsigned divisor = rough_log_2_size(count);
+ //Making sure the divisor is positive
+ if(divisor > LOG_MEAN_BIN_SIZE)
+ divisor -= LOG_MEAN_BIN_SIZE;
+ else
+ divisor = 1;
+ unsigned relative_width = (LOG_CONST * log_range)/((divisor > MAX_SPLITS) ? MAX_SPLITS : divisor);
+ //Don't try to bitshift more than the size of an element
+ if((8*sizeof(size_t)) <= relative_width)
+ relative_width = (8*sizeof(size_t)) - 1;
+ return (size_t)1 << ((relative_width < (LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT)) ?
+ (LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT) : relative_width);
+ }
+
+ //Find the minimum and maximum using <
+ template <class RandomAccessIter>
+ inline void
+ find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min)
+ {
+ min = max = current;
+ //Start from the second item, as max and min are initialized to the first
+ while(++current < last) {
+ if(*max < *current)
+ max = current;
+ else if(*current < *min)
+ min = current;
+ }
+ }
+
+ //Uses a user-defined comparison operator to find minimum and maximum
+ template <class RandomAccessIter, class compare>
+ inline void
+ find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min, compare comp)
+ {
+ min = max = current;
+ while(++current < last) {
+ if(comp(*max, *current))
+ max = current;
+ else if(comp(*current, *min))
+ min = current;
+ }
+ }
+
+ //Gets a non-negative right bit shift to operate as a logarithmic divisor
+ inline int
+ get_log_divisor(size_t count, unsigned log_range)
+ {
+ int log_divisor;
+ //If we can finish in one iteration without exceeding either (2 to the MAX_SPLITS) or n bins, do so
+ if((log_divisor = log_range - rough_log_2_size(count)) <= 0 && log_range < MAX_SPLITS)
+ log_divisor = 0;
+ else {
+ //otherwise divide the data into an optimized number of pieces
+ log_divisor += LOG_MEAN_BIN_SIZE;
+ if(log_divisor < 0)
+ log_divisor = 0;
+ //Cannot exceed MAX_SPLITS or cache misses slow down bin lookups dramatically
+ if((log_range - log_divisor) > MAX_SPLITS)
+ log_divisor = log_range - MAX_SPLITS;
+ }
+ return log_divisor;
+ }
+
+ template <class RandomAccessIter>
+ inline RandomAccessIter *
+ size_bins(std::vector<size_t> &bin_sizes, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset, unsigned &cache_end, unsigned bin_count)
+ {
+ //Assure space for the size of each bin, followed by initializing sizes
+ if(bin_count > bin_sizes.size())
+ bin_sizes.resize(bin_count);
+ for(size_t u = 0; u < bin_count; u++)
+ bin_sizes[u] = 0;
+ //Make sure there is space for the bins
+ cache_end = cache_offset + bin_count;
+ if(cache_end > bin_cache.size())
+ bin_cache.resize(cache_end);
+ return &(bin_cache[cache_offset]);
+ }
+
+ //Implementation for recursive integer sorting
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void
+ spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes)
+ {
+ //This step is roughly 10% of runtime, but it helps avoid worst-case behavior and improve behavior with real data
+ //If you know the maximum and minimum ahead of time, you can pass those values in and skip this step for the first iteration
+ RandomAccessIter max, min;
+ find_extremes(first, last, max, min);
+ //max and min will be the same (the first item) iff all values are equivalent
+ if(max == min)
+ return;
+ RandomAccessIter * target_bin;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(*max >> 0) - (*min >> 0)));
+ div_type div_min = *min >> log_divisor;
+ div_type div_max = *max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[(*(current++) >> log_divisor) - div_min]++;
+ //Assign the bin positions
+ bins[0] = first;
+ for(unsigned u = 0; u < bin_count - 1; u++)
+ bins[u + 1] = bins[u] + bin_sizes[u];
+
+ //Swap into place
+ //This dominates runtime, mostly in the swap and bin lookups
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count - 1; ++u) {
+ RandomAccessIter * local_bin = bins + u;
+ nextbinstart += bin_sizes[u];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = (bins + ((*current >> log_divisor) - div_min)); target_bin != local_bin;
+ target_bin = bins + ((*current >> log_divisor) - div_min)) {
+ //3-way swap; this is about 1% faster than a 2-way swap with integers
+ //The main advantage is less copies are involved per item put in the correct place
+ data_type tmp;
+ RandomAccessIter b = (*target_bin)++;
+ RandomAccessIter * b_bin = bins + ((*b >> log_divisor) - div_min);
+ if (b_bin != local_bin) {
+ RandomAccessIter c = (*b_bin)++;
+ tmp = *c;
+ *c = *b;
+ }
+ else
+ tmp = *b;
+ *b = *current;
+ *current = tmp;
+ }
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[bin_count - 1] = last;
+
+ //If we've bucketsorted, the array is sorted and we should skip recursion
+ if(!log_divisor)
+ return;
+
+ //Recursing; log_divisor is the remaining range
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ else
+ spread_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //Generic bitshift-based 3-way swapping code
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void inner_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
+ , const unsigned log_divisor, const div_type div_min)
+ {
+ RandomAccessIter * local_bin = bins + ii;
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ for(RandomAccessIter * target_bin = (bins + (shift(*current, log_divisor) - div_min)); target_bin != local_bin;
+ target_bin = bins + (shift(*current, log_divisor) - div_min)) {
+ data_type tmp;
+ RandomAccessIter b = (*target_bin)++;
+ RandomAccessIter * b_bin = bins + (shift(*b, log_divisor) - div_min);
+ //Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
+ if (b_bin != local_bin) {
+ RandomAccessIter c = (*b_bin)++;
+ tmp = *c;
+ *c = *b;
+ }
+ //Note: we could increment current once the swap is done in this case, but that seems to impair performance
+ else
+ tmp = *b;
+ *b = *current;
+ *current = tmp;
+ }
+ }
+ *local_bin = nextbinstart;
+ }
+
+ //Standard swapping wrapper for ascending values
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
+ , const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min)
+ {
+ nextbinstart += bin_sizes[ii];
+ inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, log_divisor, div_min);
+ }
+
+ //Functor implementation for recursive sorting
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
+ inline void
+ spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
+ {
+ RandomAccessIter max, min;
+ find_extremes(first, last, max, min, comp);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
+ div_type div_min = shift(*min, log_divisor);
+ div_type div_max = shift(*max, log_divisor);
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ bins[0] = first;
+ for(unsigned u = 0; u < bin_count - 1; u++)
+ bins[u + 1] = bins[u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count - 1; ++u)
+ swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, bin_sizes, log_divisor, div_min);
+ bins[bin_count - 1] = last;
+
+ //If we've bucketsorted, the array is sorted and we should skip recursion
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u], comp);
+ else
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
+ }
+ }
+
+ //Functor implementation for recursive sorting with only Shift overridden
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void
+ spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift)
+ {
+ RandomAccessIter max, min;
+ find_extremes(first, last, max, min);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
+ div_type div_min = shift(*min, log_divisor);
+ div_type div_max = shift(*max, log_divisor);
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ bins[0] = first;
+ for(unsigned u = 0; u < bin_count - 1; u++)
+ bins[u + 1] = bins[u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned ii = 0; ii < bin_count - 1; ++ii)
+ swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
+ bins[bin_count - 1] = last;
+
+ //If we've bucketsorted, the array is sorted and we should skip recursion
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ else
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
+ }
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void
+ spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ spread_sort_rec<RandomAccessIter, div_type, data_type>(first, last, bin_cache, 0, bin_sizes);
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
+ inline void
+ spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(first, last, bin_cache, 0, bin_sizes, shift, comp);
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void
+ spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
+ }
+ }
+
+ //Top-level sorting call for integers
+ template <class RandomAccessIter>
+ inline void integer_sort(RandomAccessIter first, RandomAccessIter last)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else
+ detail::spread_sort(first, last, *first >> 0, *first);
+ }
+
+ //integer_sort with functors
+ template <class RandomAccessIter, class right_shift, class compare>
+ inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
+ right_shift shift, compare comp) {
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last, comp);
+ else
+ detail::spread_sort(first, last, shift(*first, 0), *first, shift, comp);
+ }
+
+ //integer_sort with right_shift functor
+ template <class RandomAccessIter, class right_shift>
+ inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
+ right_shift shift) {
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else
+ detail::spread_sort(first, last, shift(*first, 0), *first, shift);
+ }
+
+ //------------------------------------------------------ float_sort source --------------------------------------
+ //Casts a RandomAccessIter to the specified data type
+ template<class cast_type, class RandomAccessIter>
+ inline cast_type
+ cast_float_iter(const RandomAccessIter & floatiter)
+ {
+ cast_type result;
+ std::memcpy(&result, &(*floatiter), sizeof(cast_type));
+ return result;
+ }
+
+ //Casts a data element to the specified datinner_float_a type
+ template<class data_type, class cast_type>
+ inline cast_type
+ mem_cast(const data_type & data)
+ {
+ cast_type result;
+ std::memcpy(&result, &data, sizeof(cast_type));
+ return result;
+ }
+
+ namespace detail {
+ template <class RandomAccessIter, class div_type, class right_shift>
+ inline void
+ find_extremes(RandomAccessIter current, RandomAccessIter last, div_type & max, div_type & min, right_shift shift)
+ {
+ min = max = shift(*current, 0);
+ while(++current < last) {
+ div_type value = shift(*current, 0);
+ if(max < value)
+ max = value;
+ else if(value < min)
+ min = value;
+ }
+ }
+
+ //Specialized swap loops for floating-point casting
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void inner_float_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii
+ , const unsigned log_divisor, const div_type div_min)
+ {
+ RandomAccessIter * local_bin = bins + ii;
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ for(RandomAccessIter * target_bin = (bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min)); target_bin != local_bin;
+ target_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min)) {
+ data_type tmp;
+ RandomAccessIter b = (*target_bin)++;
+ RandomAccessIter * b_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(b) >> log_divisor) - div_min);
+ //Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
+ if (b_bin != local_bin) {
+ RandomAccessIter c = (*b_bin)++;
+ tmp = *c;
+ *c = *b;
+ }
+ else
+ tmp = *b;
+ *b = *current;
+ *current = tmp;
+ }
+ }
+ *local_bin = nextbinstart;
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void float_swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii
+ , const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min)
+ {
+ nextbinstart += bin_sizes[ii];
+ inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, log_divisor, div_min);
+ }
+
+ template <class RandomAccessIter, class cast_type>
+ inline void
+ find_extremes(RandomAccessIter current, RandomAccessIter last, cast_type & max, cast_type & min)
+ {
+ min = max = cast_float_iter<cast_type, RandomAccessIter>(current);
+ while(++current < last) {
+ cast_type value = cast_float_iter<cast_type, RandomAccessIter>(current);
+ if(max < value)
+ max = value;
+ else if(value < min)
+ min = value;
+ }
+ }
+
+ //Special-case sorting of positive floats with casting instead of a right_shift
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void
+ positive_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
+ bins[0] = first;
+ for(unsigned u = 0; u < bin_count - 1; u++)
+ bins[u + 1] = bins[u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count - 1; ++u)
+ float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, bin_sizes, log_divisor, div_min);
+ bins[bin_count - 1] = last;
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ else
+ positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //Sorting negative_ float_s
+ //Note that bins are iterated in reverse order because max_neg_float = min_neg_int
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void
+ negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
+ bins[bin_count - 1] = first;
+ for(int ii = bin_count - 2; ii >= 0; --ii)
+ bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //The last bin will always have the correct elements in it
+ for(int ii = bin_count - 1; ii > 0; --ii)
+ float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, bin_sizes, log_divisor, div_min);
+ //Since we don't process the last bin, we need to update its end position
+ bin_cache[cache_offset] = last;
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii]);
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //Sorting negative_ float_s
+ //Note that bins are iterated in reverse order because max_neg_float = min_neg_int
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void
+ negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min, shift);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ bins[bin_count - 1] = first;
+ for(int ii = bin_count - 2; ii >= 0; --ii)
+ bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //The last bin will always have the correct elements in it
+ for(int ii = bin_count - 1; ii > 0; --ii)
+ swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
+ //Since we don't process the last bin, we need to update its end position
+ bin_cache[cache_offset] = last;
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii]);
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
+ }
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
+ inline void
+ negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min, shift);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ bins[bin_count - 1] = first;
+ for(int ii = bin_count - 2; ii >= 0; --ii)
+ bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //The last bin will always have the correct elements in it
+ for(int ii = bin_count - 1; ii > 0; --ii)
+ swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
+ //Since we don't process the last bin, we need to update its end position
+ bin_cache[cache_offset] = last;
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Recursing
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii], comp);
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
+ }
+ }
+
+ //Casting special-case for floating-point sorting
+ template <class RandomAccessIter, class div_type, class data_type>
+ inline void
+ float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
+ //The index of the first positive bin
+ div_type first_positive = (div_min < 0) ? -div_min : 0;
+ //Resetting if all bins are negative
+ if(cache_offset + first_positive > cache_end)
+ first_positive = cache_end - cache_offset;
+ //Reversing the order of the negative bins
+ //Note that because of the negative/positive ordering direction flip
+ //We can not depend upon bin order and positions matching up
+ //so bin_sizes must be reused to contain the end of the bin
+ if(first_positive > 0) {
+ bins[first_positive - 1] = first;
+ for(int ii = first_positive - 2; ii >= 0; --ii) {
+ bins[ii] = first + bin_sizes[ii + 1];
+ bin_sizes[ii] += bin_sizes[ii + 1];
+ }
+ //Handling positives following negatives
+ if((unsigned)first_positive < bin_count) {
+ bins[first_positive] = first + bin_sizes[0];
+ bin_sizes[first_positive] += bin_sizes[0];
+ }
+ }
+ else
+ bins[0] = first;
+ for(unsigned u = first_positive; u < bin_count - 1; u++) {
+ bins[u + 1] = first + bin_sizes[u];
+ bin_sizes[u + 1] += bin_sizes[u];
+ }
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count; ++u) {
+ nextbinstart = first + bin_sizes[u];
+ inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, log_divisor, div_min);
+ }
+
+ if(!log_divisor)
+ return;
+
+ //Handling negative values first
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii]);
+ //sort negative values using reversed-bin spread_sort
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
+ }
+
+ for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ //sort positive values using normal spread_sort
+ else
+ positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //Functor implementation for recursive sorting
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void
+ float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min, shift);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ //The index of the first positive bin
+ div_type first_positive = (div_min < 0) ? -div_min : 0;
+ //Resetting if all bins are negative
+ if(cache_offset + first_positive > cache_end)
+ first_positive = cache_end - cache_offset;
+ //Reversing the order of the negative bins
+ //Note that because of the negative/positive ordering direction flip
+ //We can not depend upon bin order and positions matching up
+ //so bin_sizes must be reused to contain the end of the bin
+ if(first_positive > 0) {
+ bins[first_positive - 1] = first;
+ for(int ii = first_positive - 2; ii >= 0; --ii) {
+ bins[ii] = first + bin_sizes[ii + 1];
+ bin_sizes[ii] += bin_sizes[ii + 1];
+ }
+ //Handling positives following negatives
+ if((unsigned)first_positive < bin_count) {
+ bins[first_positive] = first + bin_sizes[0];
+ bin_sizes[first_positive] += bin_sizes[0];
+ }
+ }
+ else
+ bins[0] = first;
+ for(unsigned u = first_positive; u < bin_count - 1; u++) {
+ bins[u + 1] = first + bin_sizes[u];
+ bin_sizes[u + 1] += bin_sizes[u];
+ }
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count; ++u) {
+ nextbinstart = first + bin_sizes[u];
+ inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
+ }
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Handling negative values first
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii]);
+ //sort negative values using reversed-bin spread_sort
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
+ }
+
+ for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ //sort positive values using normal spread_sort
+ else
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
+ }
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
+ inline void
+ float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
+ , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
+ {
+ div_type max, min;
+ find_extremes(first, last, max, min, shift);
+ if(max == min)
+ return;
+ unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
+ div_type div_min = min >> log_divisor;
+ div_type div_max = max >> log_divisor;
+ unsigned bin_count = div_max - div_min + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
+
+ //Calculating the size of each bin
+ for (RandomAccessIter current = first; current != last;)
+ bin_sizes[shift(*(current++), log_divisor) - div_min]++;
+ //The index of the first positive bin
+ div_type first_positive = (div_min < 0) ? -div_min : 0;
+ //Resetting if all bins are negative
+ if(cache_offset + first_positive > cache_end)
+ first_positive = cache_end - cache_offset;
+ //Reversing the order of the negative bins
+ //Note that because of the negative/positive ordering direction flip
+ //We can not depend upon bin order and positions matching up
+ //so bin_sizes must be reused to contain the end of the bin
+ if(first_positive > 0) {
+ bins[first_positive - 1] = first;
+ for(int ii = first_positive - 2; ii >= 0; --ii) {
+ bins[ii] = first + bin_sizes[ii + 1];
+ bin_sizes[ii] += bin_sizes[ii + 1];
+ }
+ //Handling positives following negatives
+ if((unsigned)first_positive < bin_count) {
+ bins[first_positive] = first + bin_sizes[0];
+ bin_sizes[first_positive] += bin_sizes[0];
+ }
+ }
+ else
+ bins[0] = first;
+ for(unsigned u = first_positive; u < bin_count - 1; u++) {
+ bins[u + 1] = first + bin_sizes[u];
+ bin_sizes[u + 1] += bin_sizes[u];
+ }
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ for(unsigned u = 0; u < bin_count; ++u) {
+ nextbinstart = first + bin_sizes[u];
+ inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
+ }
+
+ //Return if we've completed bucketsorting
+ if(!log_divisor)
+ return;
+
+ //Handling negative values first
+ size_t max_count = get_max_count(log_divisor, last - first);
+ RandomAccessIter lastPos = first;
+ for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
+ size_t count = bin_cache[ii] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[ii]);
+ //sort negative values using reversed-bin spread_sort
+ else
+ negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
+ }
+
+ for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ if(count < 2)
+ continue;
+ if(count < max_count)
+ std::sort(lastPos, bin_cache[u]);
+ //sort positive values using normal spread_sort
+ else
+ spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
+ }
+ }
+
+ template <class RandomAccessIter, class cast_type, class data_type>
+ inline void
+ float_Sort(RandomAccessIter first, RandomAccessIter last, cast_type, data_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ float_sort_rec<RandomAccessIter, cast_type, data_type>(first, last, bin_cache, 0, bin_sizes);
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift>
+ inline void
+ float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
+ }
+
+ template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
+ inline void
+ float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift, comp);
+ }
+ }
+
+ //float_sort with casting
+ //The cast_type must be equal in size to the data type, and must be a signed integer
+ template <class RandomAccessIter, class cast_type>
+ inline void float_sort_cast(RandomAccessIter first, RandomAccessIter last, cast_type cVal)
+ {
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else
+ detail::float_Sort(first, last, cVal, *first);
+ }
+
+ //float_sort with casting to an int
+ //Only use this with IEEE floating-point numbers
+ template <class RandomAccessIter>
+ inline void float_sort_cast_to_int(RandomAccessIter first, RandomAccessIter last)
+ {
+ int cVal = 0;
+ float_sort_cast(first, last, cVal);
+ }
+
+ //float_sort with functors
+ template <class RandomAccessIter, class right_shift>
+ inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift)
+ {
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else
+ detail::float_Sort(first, last, shift(*first, 0), *first, shift);
+ }
+
+ template <class RandomAccessIter, class right_shift, class compare>
+ inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift, compare comp)
+ {
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last, comp);
+ else
+ detail::float_Sort(first, last, shift(*first, 0), *first, shift, comp);
+ }
+
+ //------------------------------------------------- string_sort source ---------------------------------------------
+ namespace detail {
+ //Offsetting on identical characters. This function works a character at a time for optimal worst-case performance.
+ template<class RandomAccessIter>
+ inline void
+ update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset)
+ {
+ unsigned nextOffset = char_offset;
+ bool done = false;
+ while(!done) {
+ RandomAccessIter curr = first;
+ do {
+ //ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
+ if((*curr).size() > char_offset && ((*curr).size() <= (nextOffset + 1) || (*curr)[nextOffset] != (*first)[nextOffset])) {
+ done = true;
+ break;
+ }
+ } while(++curr != finish);
+ if(!done)
+ ++nextOffset;
+ }
+ char_offset = nextOffset;
+ }
+
+ //Offsetting on identical characters. This function works a character at a time for optimal worst-case performance.
+ template<class RandomAccessIter, class get_char, class get_length>
+ inline void
+ update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset, get_char getchar, get_length length)
+ {
+ unsigned nextOffset = char_offset;
+ bool done = false;
+ while(!done) {
+ RandomAccessIter curr = first;
+ do {
+ //ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
+ if(length(*curr) > char_offset && (length(*curr) <= (nextOffset + 1) || getchar((*curr), nextOffset) != getchar((*first), nextOffset))) {
+ done = true;
+ break;
+ }
+ } while(++curr != finish);
+ if(!done)
+ ++nextOffset;
+ }
+ char_offset = nextOffset;
+ }
+
+ //A comparison functor for strings that assumes they are identical up to char_offset
+ template<class data_type, class unsignedchar_type>
+ struct offset_lessthan {
+ offset_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
+ inline bool operator()(const data_type &x, const data_type &y) const
+ {
+ unsigned minSize = std::min(x.size(), y.size());
+ for(unsigned u = fchar_offset; u < minSize; ++u) {
+ if(static_cast<unsignedchar_type>(x[u]) < static_cast<unsignedchar_type>(y[u]))
+ return true;
+ else if(static_cast<unsignedchar_type>(y[u]) < static_cast<unsignedchar_type>(x[u]))
+ return false;
+ }
+ return x.size() < y.size();
+ }
+ unsigned fchar_offset;
+ };
+
+ //A comparison functor for strings that assumes they are identical up to char_offset
+ template<class data_type, class unsignedchar_type>
+ struct offset_greaterthan {
+ offset_greaterthan(unsigned char_offset) : fchar_offset(char_offset){}
+ inline bool operator()(const data_type &x, const data_type &y) const
+ {
+ unsigned minSize = std::min(x.size(), y.size());
+ for(unsigned u = fchar_offset; u < minSize; ++u) {
+ if(static_cast<unsignedchar_type>(x[u]) > static_cast<unsignedchar_type>(y[u]))
+ return true;
+ else if(static_cast<unsignedchar_type>(y[u]) > static_cast<unsignedchar_type>(x[u]))
+ return false;
+ }
+ return x.size() > y.size();
+ }
+ unsigned fchar_offset;
+ };
+
+ //A comparison functor for strings that assumes they are identical up to char_offset
+ template<class data_type, class get_char, class get_length>
+ struct offset_char_lessthan {
+ offset_char_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
+ inline bool operator()(const data_type &x, const data_type &y) const
+ {
+ unsigned minSize = std::min(length(x), length(y));
+ for(unsigned u = fchar_offset; u < minSize; ++u) {
+ if(getchar(x, u) < getchar(y, u))
+ return true;
+ else if(getchar(y, u) < getchar(x, u))
+ return false;
+ }
+ return length(x) < length(y);
+ }
+ unsigned fchar_offset;
+ get_char getchar;
+ get_length length;
+ };
+
+ //String sorting recursive implementation
+ template <class RandomAccessIter, class data_type, class unsignedchar_type>
+ inline void
+ string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
+ , unsigned cache_offset, std::vector<size_t> &bin_sizes)
+ {
+ //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
+ //Iterate to the end of the empties. If all empty, return
+ while((*first).size() <= char_offset) {
+ if(++first == last)
+ return;
+ }
+ RandomAccessIter finish = last - 1;
+ //Getting the last non-empty
+ for(;(*finish).size() <= char_offset; --finish) { }
+ ++finish;
+ //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
+ update_offset(first, finish, char_offset);
+
+ const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
+ //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
+ const unsigned max_size = bin_count;
+ const unsigned membin_count = bin_count + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last; ++current) {
+ if((*current).size() <= char_offset) {
+ bin_sizes[0]++;
+ }
+ else
+ bin_sizes[static_cast<unsignedchar_type>((*current)[char_offset]) + 1]++;
+ }
+ //Assign the bin positions
+ bin_cache[cache_offset] = first;
+ for(unsigned u = 0; u < membin_count - 1; u++)
+ bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //handling empty bins
+ RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
+ nextbinstart += bin_sizes[0];
+ RandomAccessIter * target_bin;
+ //Iterating over each element in the bin of empties
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //empties belong in this bin
+ while((*current).size() > char_offset) {
+ target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]);
+ iter_swap(current, (*target_bin)++);
+ }
+ }
+ *local_bin = nextbinstart;
+ //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
+ unsigned last_bin = bin_count - 1;
+ for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
+ //This dominates runtime, mostly in the swap and bin lookups
+ for(unsigned u = 0; u < last_bin; ++u) {
+ local_bin = bins + u;
+ nextbinstart += bin_sizes[u + 1];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]); target_bin != local_bin;
+ target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]))
+ iter_swap(current, (*target_bin)++);
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[last_bin] = last;
+ //Recursing
+ RandomAccessIter lastPos = bin_cache[cache_offset];
+ //Skip this loop for empties
+ for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_size)
+ std::sort(lastPos, bin_cache[u], offset_lessthan<data_type, unsignedchar_type>(char_offset + 1));
+ else
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //Sorts strings in reverse order, with empties at the end
+ template <class RandomAccessIter, class data_type, class unsignedchar_type>
+ inline void
+ reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
+ , unsigned cache_offset, std::vector<size_t> &bin_sizes)
+ {
+ //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
+ RandomAccessIter curr = first;
+ //Iterate to the end of the empties. If all empty, return
+ while((*curr).size() <= char_offset) {
+ if(++curr == last)
+ return;
+ }
+ //Getting the last non-empty
+ while((*(--last)).size() <= char_offset) { }
+ ++last;
+ //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
+ update_offset(curr, last, char_offset);
+ RandomAccessIter * target_bin;
+
+ const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
+ //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
+ const unsigned max_size = bin_count;
+ const unsigned membin_count = bin_count + 1;
+ const unsigned max_bin = bin_count - 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
+ RandomAccessIter * end_bin = &(bin_cache[cache_offset + max_bin]);
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last; ++current) {
+ if((*current).size() <= char_offset) {
+ bin_sizes[bin_count]++;
+ }
+ else
+ bin_sizes[max_bin - static_cast<unsignedchar_type>((*current)[char_offset])]++;
+ }
+ //Assign the bin positions
+ bin_cache[cache_offset] = first;
+ for(unsigned u = 0; u < membin_count - 1; u++)
+ bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = last;
+ //handling empty bins
+ RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
+ RandomAccessIter lastFull = *local_bin;
+ //Iterating over each element in the bin of empties
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //empties belong in this bin
+ while((*current).size() > char_offset) {
+ target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]);
+ iter_swap(current, (*target_bin)++);
+ }
+ }
+ *local_bin = nextbinstart;
+ nextbinstart = first;
+ //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
+ unsigned last_bin = max_bin;
+ for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
+ //This dominates runtime, mostly in the swap and bin lookups
+ for(unsigned u = 0; u < last_bin; ++u) {
+ local_bin = bins + u;
+ nextbinstart += bin_sizes[u];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]); target_bin != local_bin;
+ target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]))
+ iter_swap(current, (*target_bin)++);
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[last_bin] = lastFull;
+ //Recursing
+ RandomAccessIter lastPos = first;
+ //Skip this loop for empties
+ for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_size)
+ std::sort(lastPos, bin_cache[u], offset_greaterthan<data_type, unsignedchar_type>(char_offset + 1));
+ else
+ reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
+ }
+ }
+
+ //String sorting recursive implementation
+ template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length>
+ inline void
+ string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
+ , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length)
+ {
+ //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
+ //Iterate to the end of the empties. If all empty, return
+ while(length(*first) <= char_offset) {
+ if(++first == last)
+ return;
+ }
+ RandomAccessIter finish = last - 1;
+ //Getting the last non-empty
+ for(;length(*finish) <= char_offset; --finish) { }
+ ++finish;
+ update_offset(first, finish, char_offset, getchar, length);
+
+ const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
+ //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
+ const unsigned max_size = bin_count;
+ const unsigned membin_count = bin_count + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last; ++current) {
+ if(length(*current) <= char_offset) {
+ bin_sizes[0]++;
+ }
+ else
+ bin_sizes[getchar((*current), char_offset) + 1]++;
+ }
+ //Assign the bin positions
+ bin_cache[cache_offset] = first;
+ for(unsigned u = 0; u < membin_count - 1; u++)
+ bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //handling empty bins
+ RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
+ nextbinstart += bin_sizes[0];
+ RandomAccessIter * target_bin;
+ //Iterating over each element in the bin of empties
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //empties belong in this bin
+ while(length(*current) > char_offset) {
+ target_bin = bins + getchar((*current), char_offset);
+ iter_swap(current, (*target_bin)++);
+ }
+ }
+ *local_bin = nextbinstart;
+ //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
+ unsigned last_bin = bin_count - 1;
+ for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
+ //This dominates runtime, mostly in the swap and bin lookups
+ for(unsigned ii = 0; ii < last_bin; ++ii) {
+ local_bin = bins + ii;
+ nextbinstart += bin_sizes[ii + 1];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = bins + getchar((*current), char_offset); target_bin != local_bin;
+ target_bin = bins + getchar((*current), char_offset))
+ iter_swap(current, (*target_bin)++);
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[last_bin] = last;
+
+ //Recursing
+ RandomAccessIter lastPos = bin_cache[cache_offset];
+ //Skip this loop for empties
+ for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_size)
+ std::sort(lastPos, bin_cache[u], offset_char_lessthan<data_type, get_char, get_length>(char_offset + 1));
+ else
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length);
+ }
+ }
+
+ //String sorting recursive implementation
+ template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
+ inline void
+ string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
+ , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
+ {
+ //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
+ //Iterate to the end of the empties. If all empty, return
+ while(length(*first) <= char_offset) {
+ if(++first == last)
+ return;
+ }
+ RandomAccessIter finish = last - 1;
+ //Getting the last non-empty
+ for(;length(*finish) <= char_offset; --finish) { }
+ ++finish;
+ update_offset(first, finish, char_offset, getchar, length);
+
+ const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
+ //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
+ const unsigned max_size = bin_count;
+ const unsigned membin_count = bin_count + 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last; ++current) {
+ if(length(*current) <= char_offset) {
+ bin_sizes[0]++;
+ }
+ else
+ bin_sizes[getchar((*current), char_offset) + 1]++;
+ }
+ //Assign the bin positions
+ bin_cache[cache_offset] = first;
+ for(unsigned u = 0; u < membin_count - 1; u++)
+ bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = first;
+ //handling empty bins
+ RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
+ nextbinstart += bin_sizes[0];
+ RandomAccessIter * target_bin;
+ //Iterating over each element in the bin of empties
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //empties belong in this bin
+ while(length(*current) > char_offset) {
+ target_bin = bins + getchar((*current), char_offset);
+ iter_swap(current, (*target_bin)++);
+ }
+ }
+ *local_bin = nextbinstart;
+ //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
+ unsigned last_bin = bin_count - 1;
+ for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
+ //This dominates runtime, mostly in the swap and bin lookups
+ for(unsigned u = 0; u < last_bin; ++u) {
+ local_bin = bins + u;
+ nextbinstart += bin_sizes[u + 1];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = bins + getchar((*current), char_offset); target_bin != local_bin;
+ target_bin = bins + getchar((*current), char_offset))
+ iter_swap(current, (*target_bin)++);
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[last_bin] = last;
+
+ //Recursing
+ RandomAccessIter lastPos = bin_cache[cache_offset];
+ //Skip this loop for empties
+ for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_size)
+ std::sort(lastPos, bin_cache[u], comp);
+ else
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
+ , bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
+ }
+ }
+
+ //Sorts strings in reverse order, with empties at the end
+ template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
+ inline void
+ reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
+ , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
+ {
+ //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
+ RandomAccessIter curr = first;
+ //Iterate to the end of the empties. If all empty, return
+ while(length(*curr) <= char_offset) {
+ if(++curr == last)
+ return;
+ }
+ //Getting the last non-empty
+ while(length(*(--last)) <= char_offset) { }
+ ++last;
+ //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
+ update_offset(first, last, char_offset, getchar, length);
+
+ const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
+ //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
+ const unsigned max_size = bin_count;
+ const unsigned membin_count = bin_count + 1;
+ const unsigned max_bin = bin_count - 1;
+ unsigned cache_end;
+ RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
+ RandomAccessIter *end_bin = &(bin_cache[cache_offset + max_bin]);
+
+ //Calculating the size of each bin; this takes roughly 10% of runtime
+ for (RandomAccessIter current = first; current != last; ++current) {
+ if(length(*current) <= char_offset) {
+ bin_sizes[bin_count]++;
+ }
+ else
+ bin_sizes[max_bin - getchar((*current), char_offset)]++;
+ }
+ //Assign the bin positions
+ bin_cache[cache_offset] = first;
+ for(unsigned u = 0; u < membin_count - 1; u++)
+ bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
+
+ //Swap into place
+ RandomAccessIter nextbinstart = last;
+ //handling empty bins
+ RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
+ RandomAccessIter lastFull = *local_bin;
+ RandomAccessIter * target_bin;
+ //Iterating over each element in the bin of empties
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //empties belong in this bin
+ while(length(*current) > char_offset) {
+ target_bin = end_bin - getchar((*current), char_offset);
+ iter_swap(current, (*target_bin)++);
+ }
+ }
+ *local_bin = nextbinstart;
+ nextbinstart = first;
+ //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
+ unsigned last_bin = max_bin;
+ for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
+ //This dominates runtime, mostly in the swap and bin lookups
+ for(unsigned u = 0; u < last_bin; ++u) {
+ local_bin = bins + u;
+ nextbinstart += bin_sizes[u];
+ //Iterating over each element in this bin
+ for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
+ //Swapping elements in current into place until the correct element has been swapped in
+ for(target_bin = end_bin - getchar((*current), char_offset); target_bin != local_bin;
+ target_bin = end_bin - getchar((*current), char_offset))
+ iter_swap(current, (*target_bin)++);
+ }
+ *local_bin = nextbinstart;
+ }
+ bins[last_bin] = lastFull;
+ //Recursing
+ RandomAccessIter lastPos = first;
+ //Skip this loop for empties
+ for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
+ size_t count = bin_cache[u] - lastPos;
+ //don't sort unless there are at least two items to compare
+ if(count < 2)
+ continue;
+ //using std::sort if its worst-case is better
+ if(count < max_size)
+ std::sort(lastPos, bin_cache[u], comp);
+ else
+ reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
+ , bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
+ }
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class data_type, class unsignedchar_type>
+ inline void
+ string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class data_type, class unsignedchar_type>
+ inline void
+ reverse_string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class get_char, class get_length, class data_type, class unsignedchar_type>
+ inline void
+ string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, data_type, unsignedchar_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length);
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
+ inline void
+ string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
+ }
+
+ //Holds the bin vector and makes the initial recursive call
+ template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
+ inline void
+ reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
+ {
+ std::vector<size_t> bin_sizes;
+ std::vector<RandomAccessIter> bin_cache;
+ reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
+ }
+ }
+
+ //Allows character-type overloads
+ template <class RandomAccessIter, class unsignedchar_type>
+ inline void string_sort(RandomAccessIter first, RandomAccessIter last, unsignedchar_type unused)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else
+ detail::string_sort(first, last, *first, unused);
+ }
+
+ //Top-level sorting call; wraps using default of unsigned char
+ template <class RandomAccessIter>
+ inline void string_sort(RandomAccessIter first, RandomAccessIter last)
+ {
+ unsigned char unused = '\0';
+ string_sort(first, last, unused);
+ }
+
+ //Allows character-type overloads
+ template <class RandomAccessIter, class compare, class unsignedchar_type>
+ inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp, unsignedchar_type unused)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last, comp);
+ else
+ detail::reverse_string_sort(first, last, *first, unused);
+ }
+
+ //Top-level sorting call; wraps using default of unsigned char
+ template <class RandomAccessIter, class compare>
+ inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp)
+ {
+ unsigned char unused = '\0';
+ reverse_string_sort(first, last, comp, unused);
+ }
+
+ template <class RandomAccessIter, class get_char, class get_length>
+ inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last);
+ else {
+ //skipping past empties at the beginning, which allows us to get the character type
+ //.empty() is not used so as not to require a user declaration of it
+ while(!length(*first)) {
+ if(++first == last)
+ return;
+ }
+ detail::string_sort(first, last, getchar, length, *first, getchar((*first), 0));
+ }
+ }
+
+ template <class RandomAccessIter, class get_char, class get_length, class compare>
+ inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last, comp);
+ else {
+ //skipping past empties at the beginning, which allows us to get the character type
+ //.empty() is not used so as not to require a user declaration of it
+ while(!length(*first)) {
+ if(++first == last)
+ return;
+ }
+ detail::string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
+ }
+ }
+
+ template <class RandomAccessIter, class get_char, class get_length, class compare>
+ inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp)
+ {
+ //Don't sort if it's too small to optimize
+ if(last - first < detail::MIN_SORT_SIZE)
+ std::sort(first, last, comp);
+ else {
+ //skipping past empties at the beginning, which allows us to get the character type
+ //.empty() is not used so as not to require a user declaration of it
+ while(!length(*(--last))) {
+ //Note: if there is just one non-empty, and it's at the beginning, then it's already in sorted order
+ if(first == last)
+ return;
+ }
+ //making last just after the end of the non-empty part of the array
+ ++last;
+ detail::reverse_string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
+ }
+ }
+}
+
+#endif
diff --git a/trunk/src/system_wrappers/source/system_wrappers.gyp b/trunk/src/system_wrappers/source/system_wrappers.gyp
new file mode 100644
index 0000000..79820e9
--- /dev/null
+++ b/trunk/src/system_wrappers/source/system_wrappers.gyp
@@ -0,0 +1,190 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ '../../build/common.gypi', ],
+ 'targets': [
+ {
+ 'target_name': 'system_wrappers',
+ 'type': '<(library)',
+ 'include_dirs': [
+ 'spreadsortlib',
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../interface',
+ ],
+ },
+ 'sources': [
+ '../interface/aligned_malloc.h',
+ '../interface/atomic32_wrapper.h',
+ '../interface/condition_variable_wrapper.h',
+ '../interface/cpu_info.h',
+ '../interface/cpu_wrapper.h',
+ '../interface/cpu_features_wrapper.h',
+ '../interface/critical_section_wrapper.h',
+ '../interface/data_log.h',
+ '../interface/data_log_c.h',
+ '../interface/data_log_impl.h',
+ '../interface/event_wrapper.h',
+ '../interface/file_wrapper.h',
+ '../interface/fix_interlocked_exchange_pointer_win.h',
+ '../interface/list_wrapper.h',
+ '../interface/map_wrapper.h',
+ '../interface/ref_count.h',
+ '../interface/rw_lock_wrapper.h',
+ '../interface/scoped_ptr.h',
+ '../interface/scoped_refptr.h',
+ '../interface/sort.h',
+ '../interface/static_instance.h',
+ '../interface/thread_wrapper.h',
+ '../interface/tick_util.h',
+ '../interface/trace.h',
+ 'aligned_malloc.cc',
+ 'atomic32.cc',
+ 'atomic32_linux.h',
+ 'atomic32_mac.h',
+ 'atomic32_win.h',
+ 'condition_variable.cc',
+ 'condition_variable_posix.cc',
+ 'condition_variable_posix.h',
+ 'condition_variable_win.cc',
+ 'condition_variable_win.h',
+ 'cpu.cc',
+ 'cpu_no_op.cc',
+ 'cpu_info.cc',
+ 'cpu_linux.cc',
+ 'cpu_linux.h',
+ 'cpu_mac.cc',
+ 'cpu_mac.h',
+ 'cpu_win.cc',
+ 'cpu_win.h',
+ 'cpu_features.cc',
+ 'critical_section.cc',
+ 'critical_section_posix.cc',
+ 'critical_section_posix.h',
+ 'critical_section_win.cc',
+ 'critical_section_win.h',
+ 'data_log.cc',
+ 'data_log_c.cc',
+ 'data_log_no_op.cc',
+ 'event.cc',
+ 'event_posix.cc',
+ 'event_posix.h',
+ 'event_win.cc',
+ 'event_win.h',
+ 'file_impl.cc',
+ 'file_impl.h',
+ 'list_no_stl.cc',
+ 'map.cc',
+ 'rw_lock.cc',
+ 'rw_lock_posix.cc',
+ 'rw_lock_posix.h',
+ 'rw_lock_win.cc',
+ 'rw_lock_win.h',
+ 'sort.cc',
+ 'thread.cc',
+ 'thread_posix.cc',
+ 'thread_posix.h',
+ 'thread_win.cc',
+ 'thread_win.h',
+ 'set_thread_name_win.h',
+ 'trace_impl.cc',
+ 'trace_impl.h',
+ 'trace_impl_no_op.cc',
+ 'trace_posix.cc',
+ 'trace_posix.h',
+ 'trace_win.cc',
+ 'trace_win.h',
+ ],
+ 'conditions': [
+ ['enable_data_logging==1', {
+ 'sources!': [ 'data_log_no_op.cc', ],
+ },{
+ 'sources!': [ 'data_log.cc', ],
+ },],
+ ['OS=="linux"', {
+ 'link_settings': {
+ 'libraries': [ '-lrt', ],
+ },
+ }],
+ ['OS=="mac"', {
+ 'link_settings': {
+ 'libraries': [ '$(SDKROOT)/System/Library/Frameworks/ApplicationServices.framework', ],
+ },
+ }],
+ ['OS=="win"', {
+ 'link_settings': {
+ 'libraries': [ '-lwinmm.lib', ],
+ },
+ }],
+ ['build_with_chromium==1', {
+ 'sources!': [
+ 'cpu.cc',
+ 'cpu_linux.h',
+ 'cpu_mac.h',
+ 'cpu_win.h',
+ 'trace_impl.cc',
+ 'trace_impl.h',
+ 'trace_posix.cc',
+ 'trace_posix.h',
+ 'trace_win.cc',
+ 'trace_win.h',
+ ],
+ }, {
+ 'sources!': [
+ 'cpu_no_op.cc',
+ 'trace_impl_no_op.cc',
+ ],
+ }]
+ ] # conditions
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'system_wrappers_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'system_wrappers',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'cpu_wrapper_unittest.cc',
+ 'cpu_measurement_harness.h',
+ 'cpu_measurement_harness.cc',
+ 'list_unittest.cc',
+ 'map_unittest.cc',
+ 'data_log_unittest.cc',
+ 'data_log_unittest_disabled.cc',
+ 'data_log_helpers_unittest.cc',
+ 'data_log_c_helpers_unittest.c',
+ 'data_log_c_helpers_unittest.h',
+ 'trace_unittest.cc',
+ ],
+ 'conditions': [
+ ['enable_data_logging==1', {
+ 'sources!': [ 'data_log_unittest_disabled.cc', ],
+ }, {
+ 'sources!': [ 'data_log_unittest.cc', ],
+ }],
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/system_wrappers/source/thread.cc b/trunk/src/system_wrappers/source/thread.cc
new file mode 100644
index 0000000..32dcc63
--- /dev/null
+++ b/trunk/src/system_wrappers/source/thread.cc
@@ -0,0 +1,30 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "thread_wrapper.h"
+
+#if defined(_WIN32)
+ #include "thread_win.h"
+#else
+ #include "thread_posix.h"
+#endif
+
+namespace webrtc {
+ThreadWrapper* ThreadWrapper::CreateThread(ThreadRunFunction func,
+ ThreadObj obj, ThreadPriority prio,
+ const char* threadName)
+{
+#if defined(_WIN32)
+ return new ThreadWindows(func, obj, prio, threadName);
+#else
+ return ThreadPosix::Create(func, obj, prio, threadName);
+#endif
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_posix.cc b/trunk/src/system_wrappers/source/thread_posix.cc
new file mode 100644
index 0000000..05e7943
--- /dev/null
+++ b/trunk/src/system_wrappers/source/thread_posix.cc
@@ -0,0 +1,340 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "thread_posix.h"
+
+#include <errno.h>
+#include <string.h> // strncpy
+#include <time.h> // nanosleep
+#include <unistd.h>
+#ifdef WEBRTC_LINUX
+#include <sys/types.h>
+#include <sched.h>
+#include <sys/syscall.h>
+#include <linux/unistd.h>
+#include <sys/prctl.h>
+#endif
+
+#if defined(WEBRTC_MAC)
+#include <mach/mach.h>
+#endif
+
+#include "event_wrapper.h"
+#include "trace.h"
+
+namespace webrtc {
+extern "C"
+{
+ static void* StartThread(void* lpParameter)
+ {
+ static_cast<ThreadPosix*>(lpParameter)->Run();
+ return 0;
+ }
+}
+
+ThreadWrapper* ThreadPosix::Create(ThreadRunFunction func, ThreadObj obj,
+ ThreadPriority prio, const char* threadName)
+{
+ ThreadPosix* ptr = new ThreadPosix(func, obj, prio, threadName);
+ if (!ptr)
+ {
+ return NULL;
+ }
+ const int error = ptr->Construct();
+ if (error)
+ {
+ delete ptr;
+ return NULL;
+ }
+ return ptr;
+}
+
+ThreadPosix::ThreadPosix(ThreadRunFunction func, ThreadObj obj,
+ ThreadPriority prio, const char* threadName)
+ : _runFunction(func),
+ _obj(obj),
+ _alive(false),
+ _dead(true),
+ _prio(prio),
+ _event(EventWrapper::Create()),
+ _name(),
+ _setThreadName(false),
+#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
+ _pid(-1),
+#endif
+ _attr(),
+ _thread(0)
+{
+ if (threadName != NULL)
+ {
+ _setThreadName = true;
+ strncpy(_name, threadName, kThreadMaxNameLength);
+ _name[kThreadMaxNameLength - 1] = '\0';
+ }
+}
+
+uint32_t ThreadWrapper::GetThreadId() {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_LINUX)
+ return static_cast<uint32_t>(syscall(__NR_gettid));
+#elif defined(WEBRTC_MAC)
+ return static_cast<uint32_t>(mach_thread_self());
+#else
+ return reinterpret_cast<uint32_t>(pthread_self());
+#endif
+}
+
+int ThreadPosix::Construct()
+{
+ int result = 0;
+#if !defined(WEBRTC_ANDROID)
+ // Enable immediate cancellation if requested, see Shutdown()
+ result = pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
+ if (result != 0)
+ {
+ return -1;
+ }
+ result = pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL);
+ if (result != 0)
+ {
+ return -1;
+ }
+#endif
+ result = pthread_attr_init(&_attr);
+ if (result != 0)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+ThreadPosix::~ThreadPosix()
+{
+ pthread_attr_destroy(&_attr);
+ delete _event;
+}
+
+#define HAS_THREAD_ID !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM) && \
+ !defined(WEBRTC_MAC) && !defined(WEBRTC_MAC_INTEL) && \
+ !defined(MAC_DYLIB) && !defined(MAC_INTEL_DYLIB)
+#if HAS_THREAD_ID
+bool ThreadPosix::Start(unsigned int& threadID)
+#else
+bool ThreadPosix::Start(unsigned int& /*threadID*/)
+#endif
+{
+ if (!_runFunction)
+ {
+ return false;
+ }
+ int result = pthread_attr_setdetachstate(&_attr, PTHREAD_CREATE_DETACHED);
+ // Set the stack stack size to 1M.
+ result |= pthread_attr_setstacksize(&_attr, 1024*1024);
+#ifdef WEBRTC_THREAD_RR
+ const int policy = SCHED_RR;
+#else
+ const int policy = SCHED_FIFO;
+#endif
+ _event->Reset();
+ result |= pthread_create(&_thread, &_attr, &StartThread, this);
+ if (result != 0)
+ {
+ return false;
+ }
+
+ // Wait up to 10 seconds for the OS to call the callback function. Prevents
+ // race condition if Stop() is called too quickly after start.
+ if (kEventSignaled != _event->Wait(WEBRTC_EVENT_10_SEC))
+ {
+ // Timed out. Something went wrong.
+ _runFunction = NULL;
+ return false;
+ }
+
+#if HAS_THREAD_ID
+ threadID = static_cast<unsigned int>(_thread);
+#endif
+ sched_param param;
+
+ const int minPrio = sched_get_priority_min(policy);
+ const int maxPrio = sched_get_priority_max(policy);
+ if ((minPrio == EINVAL) || (maxPrio == EINVAL))
+ {
+ return false;
+ }
+
+ switch (_prio)
+ {
+ case kLowPriority:
+ param.sched_priority = minPrio + 1;
+ break;
+ case kNormalPriority:
+ param.sched_priority = (minPrio + maxPrio) / 2;
+ break;
+ case kHighPriority:
+ param.sched_priority = maxPrio - 3;
+ break;
+ case kHighestPriority:
+ param.sched_priority = maxPrio - 2;
+ break;
+ case kRealtimePriority:
+ param.sched_priority = maxPrio - 1;
+ break;
+ }
+ result = pthread_setschedparam(_thread, policy, ¶m);
+ if (result == EINVAL)
+ {
+ return false;
+ }
+ return true;
+}
+
+// CPU_ZERO and CPU_SET are not available in NDK r7, so disable
+// SetAffinity on Android for now.
+#if (defined(WEBRTC_LINUX) && (!defined(WEBRTC_ANDROID)))
+bool ThreadPosix::SetAffinity(const int* processorNumbers,
+ const unsigned int amountOfProcessors) {
+ if (!processorNumbers || (amountOfProcessors == 0)) {
+ return false;
+ }
+ cpu_set_t mask;
+ CPU_ZERO(&mask);
+
+ for (unsigned int processor = 0;
+ processor < amountOfProcessors;
+ processor++) {
+ CPU_SET(processorNumbers[processor], &mask);
+ }
+#if defined(WEBRTC_ANDROID)
+ // Android.
+ const int result = syscall(__NR_sched_setaffinity,
+ _pid,
+ sizeof(mask),
+ &mask);
+#else
+ // "Normal" Linux.
+ const int result = sched_setaffinity(_pid,
+ sizeof(mask),
+ &mask);
+#endif
+ if (result != 0) {
+ return false;
+ }
+ return true;
+}
+
+#else
+// NOTE: On Mac OS X, use the Thread affinity API in
+// /usr/include/mach/thread_policy.h: thread_policy_set and mach_thread_self()
+// instead of Linux gettid() syscall.
+bool ThreadPosix::SetAffinity(const int* , const unsigned int)
+{
+ return false;
+}
+#endif
+
+void ThreadPosix::SetNotAlive()
+{
+ _alive = false;
+}
+
+bool ThreadPosix::Shutdown()
+{
+#if !defined(WEBRTC_ANDROID)
+ if (_thread && (0 != pthread_cancel(_thread)))
+ {
+ return false;
+ }
+
+ return true;
+#else
+ return false;
+#endif
+}
+
+bool ThreadPosix::Stop()
+{
+ _alive = false;
+
+ // TODO (hellner) why not use an event here?
+ // Wait up to 10 seconds for the thread to terminate
+ for (int i = 0; i < 1000 && !_dead; i++)
+ {
+ timespec t;
+ t.tv_sec = 0;
+ t.tv_nsec = 10*1000*1000;
+ nanosleep(&t, NULL);
+ }
+ if (_dead)
+ {
+ return true;
+ }
+ else
+ {
+ return false;
+ }
+}
+
+void ThreadPosix::Run()
+{
+ _alive = true;
+ _dead = false;
+#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
+ _pid = GetThreadId();
+#endif
+ // The event the Start() is waiting for.
+ _event->Set();
+
+ if (_setThreadName)
+ {
+#ifdef WEBRTC_LINUX
+ prctl(PR_SET_NAME, (unsigned long)_name, 0, 0, 0);
+#endif
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
+ "Thread with name:%s started ", _name);
+ } else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Thread without name started");
+ }
+ do
+ {
+ if (_runFunction)
+ {
+ if (!_runFunction(_obj))
+ {
+ _alive = false;
+ }
+ }
+ else
+ {
+ _alive = false;
+ }
+ }
+ while (_alive);
+
+ if (_setThreadName)
+ {
+ // Don't set the name for the trace thread because it may cause a
+ // deadlock. TODO (hellner) there should be a better solution than
+ // coupling the thread and the trace class like this.
+ if (strcmp(_name, "Trace"))
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
+ "Thread with name:%s stopped", _name);
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
+ "Thread without name stopped");
+ }
+ _dead = true;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_posix.h b/trunk/src/system_wrappers/source/thread_posix.h
new file mode 100644
index 0000000..fc5757d
--- /dev/null
+++ b/trunk/src/system_wrappers/source/thread_posix.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
+
+#include "thread_wrapper.h"
+#include <pthread.h>
+
+namespace webrtc {
+class EventWrapper;
+
+class ThreadPosix : public ThreadWrapper
+{
+public:
+ static ThreadWrapper* Create(ThreadRunFunction func, ThreadObj obj,
+ ThreadPriority prio, const char* threadName);
+
+ ThreadPosix(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
+ const char* threadName);
+ ~ThreadPosix();
+
+ // From ThreadWrapper
+ virtual void SetNotAlive();
+ virtual bool Start(unsigned int& id);
+ // Not implemented on Mac
+ virtual bool SetAffinity(const int* processorNumbers,
+ unsigned int amountOfProcessors);
+ virtual bool Stop();
+ virtual bool Shutdown();
+
+ void Run();
+
+private:
+ int Construct();
+
+private:
+ // processing function
+ ThreadRunFunction _runFunction;
+ ThreadObj _obj;
+
+ // internal state
+ bool _alive;
+ bool _dead;
+ ThreadPriority _prio;
+ EventWrapper* _event;
+
+ // zero-terminated thread name string
+ char _name[kThreadMaxNameLength];
+ bool _setThreadName;
+
+ // handle to thread
+#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
+ pid_t _pid;
+#endif
+ pthread_attr_t _attr;
+ pthread_t _thread;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/thread_win.cc b/trunk/src/system_wrappers/source/thread_win.cc
new file mode 100644
index 0000000..4599bed
--- /dev/null
+++ b/trunk/src/system_wrappers/source/thread_win.cc
@@ -0,0 +1,238 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "thread_win.h"
+
+#include <assert.h>
+#include <process.h>
+#include <stdio.h>
+#include <windows.h>
+
+#include "set_thread_name_win.h"
+#include "trace.h"
+
+#if defined(_WIN32)
+// VS 2005: Disable warnings for default initialized arrays.
+#pragma warning(disable:4351)
+#endif
+
+namespace webrtc {
+ThreadWindows::ThreadWindows(ThreadRunFunction func, ThreadObj obj,
+ ThreadPriority prio, const char* threadName)
+ : ThreadWrapper(),
+ _runFunction(func),
+ _obj(obj),
+ _alive(false),
+ _dead(true),
+ _doNotCloseHandle(false),
+ _prio(prio),
+ _event(NULL),
+ _thread(NULL),
+ _id(0),
+ _name(),
+ _setThreadName(false)
+{
+ _event = EventWrapper::Create();
+ _critsectStop = CriticalSectionWrapper::CreateCriticalSection();
+ if (threadName != NULL)
+ {
+ // Set the thread name to appear in the VS debugger.
+ _setThreadName = true;
+ strncpy(_name, threadName, kThreadMaxNameLength);
+ }
+}
+
+ThreadWindows::~ThreadWindows()
+{
+#ifdef _DEBUG
+ assert(!_alive);
+#endif
+ if (_thread)
+ {
+ CloseHandle(_thread);
+ }
+ if(_event)
+ {
+ delete _event;
+ }
+ if(_critsectStop)
+ {
+ delete _critsectStop;
+ }
+}
+
+uint32_t ThreadWrapper::GetThreadId() {
+ return GetCurrentThreadId();
+}
+
+unsigned int WINAPI ThreadWindows::StartThread(LPVOID lpParameter)
+{
+ static_cast<ThreadWindows*>(lpParameter)->Run();
+ return 0;
+}
+
+bool ThreadWindows::Start(unsigned int& threadID)
+{
+ _doNotCloseHandle = false;
+
+ // Set stack size to 1M
+ _thread=(HANDLE)_beginthreadex(NULL, 1024*1024, StartThread, (void*)this, 0,
+ &threadID);
+ if(_thread == NULL)
+ {
+ return false;
+ }
+ _id = threadID;
+ _event->Wait(INFINITE);
+
+ switch(_prio)
+ {
+ case kLowPriority:
+ SetThreadPriority(_thread, THREAD_PRIORITY_BELOW_NORMAL);
+ break;
+ case kNormalPriority:
+ SetThreadPriority(_thread, THREAD_PRIORITY_NORMAL);
+ break;
+ case kHighPriority:
+ SetThreadPriority(_thread, THREAD_PRIORITY_ABOVE_NORMAL);
+ break;
+ case kHighestPriority:
+ SetThreadPriority(_thread, THREAD_PRIORITY_HIGHEST);
+ break;
+ case kRealtimePriority:
+ SetThreadPriority(_thread, THREAD_PRIORITY_TIME_CRITICAL);
+ break;
+ };
+ return true;
+}
+
+bool ThreadWindows::SetAffinity(const int* processorNumbers,
+ const unsigned int amountOfProcessors)
+{
+ DWORD_PTR processorBitMask = 0;
+ for(unsigned int processorIndex = 0;
+ processorIndex < amountOfProcessors;
+ processorIndex++)
+ {
+ // Convert from an array with processor numbers to a bitmask
+ // Processor numbers start at zero.
+ // TODO (hellner): this looks like a bug. Shouldn't the '=' be a '+='?
+ // Or even better |=
+ processorBitMask = 1 << processorNumbers[processorIndex];
+ }
+ return SetThreadAffinityMask(_thread,processorBitMask) != 0;
+}
+
+void ThreadWindows::SetNotAlive()
+{
+ _alive = false;
+}
+
+bool ThreadWindows::Shutdown()
+{
+ DWORD exitCode = 0;
+ BOOL ret = TRUE;
+ if (_thread)
+ {
+ ret = TerminateThread(_thread, exitCode);
+ _alive = false;
+ _dead = true;
+ _thread = NULL;
+ }
+ return ret == TRUE;
+}
+
+bool ThreadWindows::Stop()
+{
+ _critsectStop->Enter();
+ // Prevents the handle from being closed in ThreadWindows::Run()
+ _doNotCloseHandle = true;
+ _alive = false;
+ bool signaled = false;
+ if (_thread && !_dead)
+ {
+ _critsectStop->Leave();
+ // Wait up to 2 seconds for the thread to complete.
+ if( WAIT_OBJECT_0 == WaitForSingleObject(_thread, 2000))
+ {
+ signaled = true;
+ }
+ _critsectStop->Enter();
+ }
+ if (_thread)
+ {
+ CloseHandle(_thread);
+ _thread = NULL;
+ }
+ _critsectStop->Leave();
+
+ if (_dead || signaled)
+ {
+ return true;
+ }
+ else
+ {
+ return false;
+ }
+}
+
+void ThreadWindows::Run()
+{
+ _alive = true;
+ _dead = false;
+ _event->Set();
+
+ // All tracing must be after _event->Set to avoid deadlock in Trace.
+ if (_setThreadName)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
+ "Thread with name:%s started ", _name);
+ SetThreadName(-1, _name); // -1, set thread name for the calling thread.
+ }else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
+ "Thread without name started");
+ }
+
+ do
+ {
+ if (_runFunction)
+ {
+ if (!_runFunction(_obj))
+ {
+ _alive = false;
+ }
+ } else {
+ _alive = false;
+ }
+ } while(_alive);
+
+ if (_setThreadName)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
+ "Thread with name:%s stopped", _name);
+ } else {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,_id,
+ "Thread without name stopped");
+ }
+
+ _critsectStop->Enter();
+
+ if (_thread && !_doNotCloseHandle)
+ {
+ HANDLE thread = _thread;
+ _thread = NULL;
+ CloseHandle(thread);
+ }
+ _dead = true;
+
+ _critsectStop->Leave();
+};
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_win.h b/trunk/src/system_wrappers/source/thread_win.h
new file mode 100644
index 0000000..4fd7523
--- /dev/null
+++ b/trunk/src/system_wrappers/source/thread_win.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
+
+#include "thread_wrapper.h"
+#include "event_wrapper.h"
+#include "critical_section_wrapper.h"
+
+#include <windows.h>
+
+namespace webrtc {
+
+class ThreadWindows : public ThreadWrapper
+{
+public:
+ ThreadWindows(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
+ const char* threadName);
+ virtual ~ThreadWindows();
+
+ virtual bool Start(unsigned int& id);
+ bool SetAffinity(const int* processorNumbers,
+ const unsigned int amountOfProcessors);
+ virtual bool Stop();
+ virtual void SetNotAlive();
+
+ static unsigned int WINAPI StartThread(LPVOID lpParameter);
+
+ virtual bool Shutdown();
+
+protected:
+ virtual void Run();
+
+private:
+ ThreadRunFunction _runFunction;
+ ThreadObj _obj;
+
+ bool _alive;
+ bool _dead;
+
+ // TODO (hellner)
+ // _doNotCloseHandle member seem pretty redundant. Should be able to remove
+ // it. Basically it should be fine to reclaim the handle when calling stop
+ // and in the destructor.
+ bool _doNotCloseHandle;
+ ThreadPriority _prio;
+ EventWrapper* _event;
+ CriticalSectionWrapper* _critsectStop;
+
+ HANDLE _thread;
+ unsigned int _id;
+ char _name[kThreadMaxNameLength];
+ bool _setThreadName;
+
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/trace_impl.cc b/trunk/src/system_wrappers/source/trace_impl.cc
new file mode 100644
index 0000000..ad908a3
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_impl.cc
@@ -0,0 +1,829 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace_impl.h"
+
+#include <cassert>
+#include <string.h> // memset
+
+#ifdef _WIN32
+#include "trace_win.h"
+#else
+#include <stdio.h>
+#include <time.h>
+#include <stdarg.h>
+#include "trace_posix.h"
+#endif // _WIN32
+
+#define KEY_LEN_CHARS 31
+
+#ifdef _WIN32
+ #pragma warning(disable:4355)
+// VS 2005: Disable warnings for default initialized arrays.
+ #pragma warning(disable:4351)
+#endif // _WIN32
+
+namespace webrtc {
+static WebRtc_UWord32 levelFilter = kTraceDefault;
+
+// Construct On First Use idiom. Avoids "static initialization order fiasco".
+TraceImpl* TraceImpl::StaticInstance(CountOperation count_operation,
+ const TraceLevel level)
+{
+ // Sanities to avoid taking lock unless absolutely necessary (for
+ // performance reasons).
+ // count_operation == kAddRefNoCreate implies that a message will be
+ // written to file.
+ if((level != kTraceAll) && (count_operation == kAddRefNoCreate))
+ {
+ if(!(level & levelFilter))
+ {
+ return NULL;
+ }
+ }
+ TraceImpl* impl =
+ GetStaticInstance<TraceImpl>(count_operation);
+ return impl;
+}
+
+TraceImpl* TraceImpl::GetTrace(const TraceLevel level)
+{
+ return StaticInstance(kAddRefNoCreate, level);
+}
+
+TraceImpl* TraceImpl::CreateInstance()
+{
+#if defined(_WIN32)
+ return new TraceWindows();
+#else
+ return new TracePosix();
+#endif
+}
+
+TraceImpl::TraceImpl()
+ : _critsectInterface(CriticalSectionWrapper::CreateCriticalSection()),
+ _callback(NULL),
+ _rowCountText(0),
+ _fileCountText(0),
+ _traceFile(*FileWrapper::Create()),
+ _thread(*ThreadWrapper::CreateThread(TraceImpl::Run, this,
+ kHighestPriority, "Trace")),
+ _event(*EventWrapper::Create()),
+ _critsectArray(CriticalSectionWrapper::CreateCriticalSection()),
+ _nextFreeIdx(),
+ _level(),
+ _length(),
+ _messageQueue(),
+ _activeQueue(0)
+{
+ _nextFreeIdx[0] = 0;
+ _nextFreeIdx[1] = 0;
+
+ unsigned int tid = 0;
+ _thread.Start(tid);
+
+ for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
+ {
+ for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
+ {
+ _messageQueue[m][n] = new
+ char[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
+ }
+ }
+}
+
+bool TraceImpl::StopThread()
+{
+ // Release the worker thread so that it can flush any lingering messages.
+ _event.Set();
+
+ // Allow 10 ms for pending messages to be flushed out.
+ // TODO (hellner): why not use condition variables to do this? Or let the
+ // worker thread die and let this thread flush remaining
+ // messages?
+#ifdef _WIN32
+ Sleep(10);
+#else
+ timespec t;
+ t.tv_sec = 0;
+ t.tv_nsec = 10*1000000;
+ nanosleep(&t,NULL);
+#endif
+
+ _thread.SetNotAlive();
+ // Make sure the thread finishes as quickly as possible (instead of having
+ // to wait for the timeout).
+ _event.Set();
+ bool stopped = _thread.Stop();
+
+ CriticalSectionScoped lock(_critsectInterface);
+ _traceFile.Flush();
+ _traceFile.CloseFile();
+ return stopped;
+}
+
+TraceImpl::~TraceImpl()
+{
+ StopThread();
+ delete &_event;
+ delete &_traceFile;
+ delete &_thread;
+ delete _critsectInterface;
+ delete _critsectArray;
+
+ for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
+ {
+ for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
+ {
+ delete [] _messageQueue[m][n];
+ }
+ }
+}
+
+WebRtc_Word32 TraceImpl::AddThreadId(char* traceMessage) const {
+ WebRtc_UWord32 threadId = ThreadWrapper::GetThreadId();
+ // Messages is 12 characters.
+ return sprintf(traceMessage, "%10u; ", threadId);
+}
+
+WebRtc_Word32 TraceImpl::AddLevel(char* szMessage, const TraceLevel level) const
+{
+ switch (level)
+ {
+ case kTraceStateInfo:
+ sprintf (szMessage, "STATEINFO ; ");
+ break;
+ case kTraceWarning:
+ sprintf (szMessage, "WARNING ; ");
+ break;
+ case kTraceError:
+ sprintf (szMessage, "ERROR ; ");
+ break;
+ case kTraceCritical:
+ sprintf (szMessage, "CRITICAL ; ");
+ break;
+ case kTraceInfo:
+ sprintf (szMessage, "DEBUGINFO ; ");
+ break;
+ case kTraceModuleCall:
+ sprintf (szMessage, "MODULECALL; ");
+ break;
+ case kTraceMemory:
+ sprintf (szMessage, "MEMORY ; ");
+ break;
+ case kTraceTimer:
+ sprintf (szMessage, "TIMER ; ");
+ break;
+ case kTraceStream:
+ sprintf (szMessage, "STREAM ; ");
+ break;
+ case kTraceApiCall:
+ sprintf (szMessage, "APICALL ; ");
+ break;
+ case kTraceDebug:
+ sprintf (szMessage, "DEBUG ; ");
+ break;
+ default:
+ assert(false);
+ return 0;
+ }
+ // All messages are 12 characters.
+ return 12;
+}
+
+WebRtc_Word32 TraceImpl::AddModuleAndId(char* traceMessage,
+ const TraceModule module,
+ const WebRtc_Word32 id) const
+{
+ // Use long int to prevent problems with different definitions of
+ // WebRtc_Word32.
+ // TODO (hellner): is this actually a problem? If so, it should be better to
+ // clean up WebRtc_Word32
+ const long int idl = id;
+ if(idl != -1)
+ {
+ const unsigned long int idEngine = id>>16;
+ const unsigned long int idChannel = id & 0xffff;
+
+ switch (module)
+ {
+ case kTraceVoice:
+ sprintf(traceMessage, " VOICE:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideo:
+ sprintf(traceMessage, " VIDEO:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceUtility:
+ sprintf(traceMessage, " UTILITY:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceRtpRtcp:
+ sprintf(traceMessage, " RTP/RTCP:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceTransport:
+ sprintf(traceMessage, " TRANSPORT:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceAudioCoding:
+ sprintf(traceMessage, "AUDIO CODING:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceSrtp:
+ sprintf(traceMessage, " SRTP:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceAudioMixerServer:
+ sprintf(traceMessage, " AUDIO MIX/S:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceAudioMixerClient:
+ sprintf(traceMessage, " AUDIO MIX/C:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideoCoding:
+ sprintf(traceMessage, "VIDEO CODING:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideoMixer:
+ // Print sleep time and API call
+ sprintf(traceMessage, " VIDEO MIX:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceFile:
+ sprintf(traceMessage, " FILE:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceAudioProcessing:
+ sprintf(traceMessage, " AUDIO PROC:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceAudioDevice:
+ sprintf(traceMessage, "AUDIO DEVICE:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideoRenderer:
+ sprintf(traceMessage, "VIDEO RENDER:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideoCapture:
+ sprintf(traceMessage, "VIDEO CAPTUR:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ case kTraceVideoPreocessing:
+ sprintf(traceMessage, " VIDEO PROC:%5ld %5ld;", idEngine,
+ idChannel);
+ break;
+ }
+ } else {
+ switch (module)
+ {
+ case kTraceVoice:
+ sprintf (traceMessage, " VOICE:%11ld;", idl);
+ break;
+ case kTraceVideo:
+ sprintf (traceMessage, " VIDEO:%11ld;", idl);
+ break;
+ case kTraceUtility:
+ sprintf (traceMessage, " UTILITY:%11ld;", idl);
+ break;
+ case kTraceRtpRtcp:
+ sprintf (traceMessage, " RTP/RTCP:%11ld;", idl);
+ break;
+ case kTraceTransport:
+ sprintf (traceMessage, " TRANSPORT:%11ld;", idl);
+ break;
+ case kTraceAudioCoding:
+ sprintf (traceMessage, "AUDIO CODING:%11ld;", idl);
+ break;
+ case kTraceSrtp:
+ sprintf (traceMessage, " SRTP:%11ld;", idl);
+ break;
+ case kTraceAudioMixerServer:
+ sprintf (traceMessage, " AUDIO MIX/S:%11ld;", idl);
+ break;
+ case kTraceAudioMixerClient:
+ sprintf (traceMessage, " AUDIO MIX/C:%11ld;", idl);
+ break;
+ case kTraceVideoCoding:
+ sprintf (traceMessage, "VIDEO CODING:%11ld;", idl);
+ break;
+ case kTraceVideoMixer:
+ sprintf (traceMessage, " VIDEO MIX:%11ld;", idl);
+ break;
+ case kTraceFile:
+ sprintf (traceMessage, " FILE:%11ld;", idl);
+ break;
+ case kTraceAudioProcessing:
+ sprintf (traceMessage, " AUDIO PROC:%11ld;", idl);
+ break;
+ case kTraceAudioDevice:
+ sprintf (traceMessage, "AUDIO DEVICE:%11ld;", idl);
+ break;
+ case kTraceVideoRenderer:
+ sprintf (traceMessage, "VIDEO RENDER:%11ld;", idl);
+ break;
+ case kTraceVideoCapture:
+ sprintf (traceMessage, "VIDEO CAPTUR:%11ld;", idl);
+ break;
+ case kTraceVideoPreocessing:
+ sprintf (traceMessage, " VIDEO PROC:%11ld;", idl);
+ break;
+ }
+ }
+ // All messages are 25 characters.
+ return 25;
+}
+
+WebRtc_Word32 TraceImpl::SetTraceFileImpl(const char* fileNameUTF8,
+ const bool addFileCounter)
+{
+ CriticalSectionScoped lock(_critsectInterface);
+
+ _traceFile.Flush();
+ _traceFile.CloseFile();
+
+ if(fileNameUTF8)
+ {
+ if(addFileCounter)
+ {
+ _fileCountText = 1;
+
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize];
+ CreateFileName(fileNameUTF8, fileNameWithCounterUTF8,
+ _fileCountText);
+ if(_traceFile.OpenFile(fileNameWithCounterUTF8, false, false,
+ true) == -1)
+ {
+ return -1;
+ }
+ }else {
+ _fileCountText = 0;
+ if(_traceFile.OpenFile(fileNameUTF8, false, false, true) == -1)
+ {
+ return -1;
+ }
+ }
+ }
+ _rowCountText = 0;
+ return 0;
+}
+
+WebRtc_Word32 TraceImpl::TraceFileImpl(
+ char fileNameUTF8[FileWrapper::kMaxFileNameSize])
+{
+ CriticalSectionScoped lock(_critsectInterface);
+ return _traceFile.FileName(fileNameUTF8, FileWrapper::kMaxFileNameSize);
+}
+
+WebRtc_Word32 TraceImpl::SetTraceCallbackImpl(TraceCallback* callback)
+{
+ CriticalSectionScoped lock(_critsectInterface);
+ _callback = callback;
+ return 0;
+}
+
+WebRtc_Word32 TraceImpl::AddMessage(
+ char* traceMessage,
+ const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
+ const WebRtc_UWord16 writtenSoFar) const
+
+{
+ int length = 0;
+ if(writtenSoFar >= WEBRTC_TRACE_MAX_MESSAGE_SIZE)
+ {
+ return -1;
+ }
+ // - 2 to leave room for newline and NULL termination
+#ifdef _WIN32
+ length = _snprintf(traceMessage,
+ WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2,
+ "%s",msg);
+ if(length < 0)
+ {
+ length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
+ traceMessage[length] = 0;
+ }
+#else
+ length = snprintf(traceMessage,
+ WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar-2, "%s",msg);
+ if(length < 0 || length > WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar - 2)
+ {
+ length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
+ traceMessage[length] = 0;
+ }
+#endif
+ // Length with NULL termination.
+ return length+1;
+}
+
+void TraceImpl::AddMessageToList(
+ const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
+ const WebRtc_UWord16 length,
+ const TraceLevel level)
+{
+ CriticalSectionScoped lock(_critsectArray);
+
+ if(_nextFreeIdx[_activeQueue] >= WEBRTC_TRACE_MAX_QUEUE)
+ {
+ if( ! _traceFile.Open() &&
+ !_callback)
+ {
+ // Keep at least the last 1/4 of old messages when not logging.
+ // TODO (hellner): isn't this redundant. The user will make it known
+ // when to start logging. Why keep messages before
+ // that?
+ for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE/4; n++)
+ {
+ const int lastQuarterOffset = (3*WEBRTC_TRACE_MAX_QUEUE/4);
+ memcpy(_messageQueue[_activeQueue][n],
+ _messageQueue[_activeQueue][n + lastQuarterOffset],
+ WEBRTC_TRACE_MAX_MESSAGE_SIZE);
+ }
+ _nextFreeIdx[_activeQueue] = WEBRTC_TRACE_MAX_QUEUE/4;
+ } else {
+ // More messages are being written than there is room for in the
+ // buffer. Drop any new messages.
+ // TODO (hellner): its probably better to drop old messages instead
+ // of new ones. One step further: if this happens
+ // it's due to writing faster than what can be
+ // processed. Maybe modify the filter at this point.
+ // E.g. turn of STREAM.
+ return;
+ }
+ }
+
+ WebRtc_UWord16 idx = _nextFreeIdx[_activeQueue];
+ _nextFreeIdx[_activeQueue]++;
+
+ _level[_activeQueue][idx] = level;
+ _length[_activeQueue][idx] = length;
+ memcpy(_messageQueue[_activeQueue][idx], traceMessage, length);
+
+ if(_nextFreeIdx[_activeQueue] == WEBRTC_TRACE_MAX_QUEUE-1)
+ {
+ // Logging more messages than can be worked off. Log a warning.
+ const char warning_msg[] = "WARNING MISSING TRACE MESSAGES\n";
+ _level[_activeQueue][_nextFreeIdx[_activeQueue]] = kTraceWarning;
+ _length[_activeQueue][_nextFreeIdx[_activeQueue]] = strlen(warning_msg);
+ memcpy(_messageQueue[_activeQueue][_nextFreeIdx[_activeQueue]],
+ warning_msg, _length[_activeQueue][idx]);
+ _nextFreeIdx[_activeQueue]++;
+ }
+}
+
+bool TraceImpl::Run(void* obj)
+{
+ return static_cast<TraceImpl*>(obj)->Process();
+}
+
+bool TraceImpl::Process()
+{
+ if(_event.Wait(1000) == kEventSignaled)
+ {
+ if(_traceFile.Open() || _callback)
+ {
+ // File mode (not calback mode).
+ WriteToFile();
+ }
+ } else {
+ _traceFile.Flush();
+ }
+ return true;
+}
+
+void TraceImpl::WriteToFile()
+{
+ WebRtc_UWord8 localQueueActive = 0;
+ WebRtc_UWord16 localNextFreeIdx = 0;
+
+ // There are two buffer. One for reading (for writing to file) and one for
+ // writing (for storing new messages). Let new messages be posted to the
+ // unused buffer so that the current buffer can be flushed safely.
+ {
+ CriticalSectionScoped lock(_critsectArray);
+ localNextFreeIdx = _nextFreeIdx[_activeQueue];
+ _nextFreeIdx[_activeQueue] = 0;
+ localQueueActive = _activeQueue;
+ if(_activeQueue == 0)
+ {
+ _activeQueue = 1;
+ } else
+ {
+ _activeQueue = 0;
+ }
+ }
+ if(localNextFreeIdx == 0)
+ {
+ return;
+ }
+
+ CriticalSectionScoped lock(_critsectInterface);
+
+ for(WebRtc_UWord16 idx = 0; idx <localNextFreeIdx; idx++)
+ {
+ TraceLevel localLevel = _level[localQueueActive][idx];
+ if(_callback)
+ {
+ _callback->Print(localLevel, _messageQueue[localQueueActive][idx],
+ _length[localQueueActive][idx]);
+ }
+ if(_traceFile.Open())
+ {
+ if(_rowCountText > WEBRTC_TRACE_MAX_FILE_SIZE)
+ {
+ // wrap file
+ _rowCountText = 0;
+ _traceFile.Flush();
+
+ if(_fileCountText == 0)
+ {
+ _traceFile.Rewind();
+ } else
+ {
+ char oldFileName[FileWrapper::kMaxFileNameSize];
+ char newFileName[FileWrapper::kMaxFileNameSize];
+
+ // get current name
+ _traceFile.FileName(oldFileName,
+ FileWrapper::kMaxFileNameSize);
+ _traceFile.CloseFile();
+
+ _fileCountText++;
+
+ UpdateFileName(oldFileName, newFileName, _fileCountText);
+
+ if(_traceFile.OpenFile(newFileName, false, false,
+ true) == -1)
+ {
+ return;
+ }
+ }
+ }
+ if(_rowCountText == 0)
+ {
+ char message[WEBRTC_TRACE_MAX_MESSAGE_SIZE + 1];
+ WebRtc_Word32 length = AddDateTimeInfo(message);
+ if(length != -1)
+ {
+ message[length] = 0;
+ message[length-1] = '\n';
+ _traceFile.Write(message, length);
+ _rowCountText++;
+ }
+ length = AddBuildInfo(message);
+ if(length != -1)
+ {
+ message[length+1] = 0;
+ message[length] = '\n';
+ message[length-1] = '\n';
+ _traceFile.Write(message, length+1);
+ _rowCountText++;
+ _rowCountText++;
+ }
+ }
+ WebRtc_UWord16 length = _length[localQueueActive][idx];
+ _messageQueue[localQueueActive][idx][length] = 0;
+ _messageQueue[localQueueActive][idx][length-1] = '\n';
+ _traceFile.Write(_messageQueue[localQueueActive][idx], length);
+ _rowCountText++;
+ }
+ }
+}
+
+void TraceImpl::AddImpl(const TraceLevel level, const TraceModule module,
+ const WebRtc_Word32 id,
+ const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE])
+{
+ if (TraceCheck(level))
+ {
+ char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
+ char* meassagePtr = traceMessage;
+
+ WebRtc_Word32 len = 0;
+ WebRtc_Word32 ackLen = 0;
+
+ len = AddLevel(meassagePtr, level);
+ if(len == -1)
+ {
+ return;
+ }
+ meassagePtr += len;
+ ackLen += len;
+
+ len = AddTime(meassagePtr, level);
+ if(len == -1)
+ {
+ return;
+ }
+ meassagePtr += len;
+ ackLen += len;
+
+ len = AddModuleAndId(meassagePtr, module, id);
+ if(len == -1)
+ {
+ return;
+ }
+ meassagePtr += len;
+ ackLen += len;
+
+ len = AddThreadId(meassagePtr);
+ if(len < 0)
+ {
+ return;
+ }
+ meassagePtr += len;
+ ackLen += len;
+
+ len = AddMessage(meassagePtr, msg, (WebRtc_UWord16)ackLen);
+ if(len == -1)
+ {
+ return;
+ }
+ ackLen += len;
+ AddMessageToList(traceMessage,(WebRtc_UWord16)ackLen, level);
+
+ // Make sure that messages are written as soon as possible.
+ _event.Set();
+ }
+}
+
+bool TraceImpl::TraceCheck(const TraceLevel level) const
+{
+ return (level & levelFilter)? true:false;
+}
+
+bool TraceImpl::UpdateFileName(
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const WebRtc_UWord32 newCount) const
+{
+ WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
+ if(length < 0)
+ {
+ return false;
+ }
+
+ WebRtc_Word32 lengthWithoutFileEnding = length-1;
+ while(lengthWithoutFileEnding > 0)
+ {
+ if(fileNameUTF8[lengthWithoutFileEnding] == '.')
+ {
+ break;
+ } else {
+ lengthWithoutFileEnding--;
+ }
+ }
+ if(lengthWithoutFileEnding == 0)
+ {
+ lengthWithoutFileEnding = length;
+ }
+ WebRtc_Word32 lengthTo_ = lengthWithoutFileEnding - 1;
+ while(lengthTo_ > 0)
+ {
+ if(fileNameUTF8[lengthTo_] == '_')
+ {
+ break;
+ } else {
+ lengthTo_--;
+ }
+ }
+
+ memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthTo_);
+ sprintf(fileNameWithCounterUTF8+lengthTo_, "_%lu%s",
+ static_cast<long unsigned int> (newCount),
+ fileNameUTF8+lengthWithoutFileEnding);
+ return true;
+}
+
+bool TraceImpl::CreateFileName(
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const WebRtc_UWord32 newCount) const
+{
+ WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
+ if(length < 0)
+ {
+ return false;
+ }
+
+ WebRtc_Word32 lengthWithoutFileEnding = length-1;
+ while(lengthWithoutFileEnding > 0)
+ {
+ if(fileNameUTF8[lengthWithoutFileEnding] == '.')
+ {
+ break;
+ }else
+ {
+ lengthWithoutFileEnding--;
+ }
+ }
+ if(lengthWithoutFileEnding == 0)
+ {
+ lengthWithoutFileEnding = length;
+ }
+ memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthWithoutFileEnding);
+ sprintf(fileNameWithCounterUTF8+lengthWithoutFileEnding, "_%lu%s",
+ static_cast<long unsigned int> (newCount),
+ fileNameUTF8+lengthWithoutFileEnding);
+ return true;
+}
+
+void Trace::CreateTrace()
+{
+ TraceImpl::StaticInstance(kAddRef);
+}
+
+void Trace::ReturnTrace()
+{
+ TraceImpl::StaticInstance(kRelease);
+}
+
+WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 filter)
+{
+ levelFilter = filter;
+ return 0;
+}
+
+WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& filter)
+{
+ filter = levelFilter;
+ return 0;
+}
+
+WebRtc_Word32 Trace::TraceFile(char fileName[FileWrapper::kMaxFileNameSize])
+{
+ TraceImpl* trace = TraceImpl::GetTrace();
+ if(trace)
+ {
+ int retVal = trace->TraceFileImpl(fileName);
+ ReturnTrace();
+ return retVal;
+ }
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceFile(const char* fileName,
+ const bool addFileCounter)
+{
+ TraceImpl* trace = TraceImpl::GetTrace();
+ if(trace)
+ {
+ int retVal = trace->SetTraceFileImpl(fileName, addFileCounter);
+ ReturnTrace();
+ return retVal;
+ }
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* callback)
+{
+ TraceImpl* trace = TraceImpl::GetTrace();
+ if(trace)
+ {
+ int retVal = trace->SetTraceCallbackImpl(callback);
+ ReturnTrace();
+ return retVal;
+ }
+ return -1;
+}
+
+void Trace::Add(const TraceLevel level, const TraceModule module,
+ const WebRtc_Word32 id, const char* msg, ...)
+
+{
+ TraceImpl* trace = TraceImpl::GetTrace(level);
+ if(trace)
+ {
+ if(trace->TraceCheck(level))
+ {
+ char tempBuff[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
+ char* buff = 0;
+ if(msg)
+ {
+ va_list args;
+ va_start(args, msg);
+#ifdef _WIN32
+ _vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
+#else
+ vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
+#endif
+ va_end(args);
+ buff = tempBuff;
+ }
+ trace->AddImpl(level, module, id, buff);
+ }
+ ReturnTrace();
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_impl.h b/trunk/src/system_wrappers/source/trace_impl.h
new file mode 100644
index 0000000..2b85813
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_impl.h
@@ -0,0 +1,132 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
+
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/file_wrapper.h"
+#include "system_wrappers/interface/static_instance.h"
+#include "system_wrappers/interface/trace.h"
+#include "system_wrappers/interface/thread_wrapper.h"
+
+namespace webrtc {
+
+// TODO (pwestin) WEBRTC_TRACE_MAX_QUEUE needs to be tweaked
+// TODO (hellner) the buffer should be close to how much the system can write to
+// file. Increasing the buffer will not solve anything. Sooner or
+// later the buffer is going to fill up anyways.
+#if defined(MAC_IPHONE)
+ #define WEBRTC_TRACE_MAX_QUEUE 2000
+#else
+ #define WEBRTC_TRACE_MAX_QUEUE 8000
+#endif
+#define WEBRTC_TRACE_NUM_ARRAY 2
+#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 256
+// Total buffer size is WEBRTC_TRACE_NUM_ARRAY (number of buffer partitions) *
+// WEBRTC_TRACE_MAX_QUEUE (number of lines per buffer partition) *
+// WEBRTC_TRACE_MAX_MESSAGE_SIZE (number of 1 byte charachters per line) =
+// 1 or 4 Mbyte
+
+#define WEBRTC_TRACE_MAX_FILE_SIZE 100*1000
+// Number of rows that may be written to file. On average 110 bytes per row (max
+// 256 bytes per row). So on average 110*100*1000 = 11 Mbyte, max 256*100*1000 =
+// 25.6 Mbyte
+
+class TraceImpl : public Trace
+{
+public:
+ virtual ~TraceImpl();
+
+ static TraceImpl* CreateInstance();
+ static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
+
+ WebRtc_Word32 SetTraceFileImpl(const char* fileName,
+ const bool addFileCounter);
+ WebRtc_Word32 TraceFileImpl(
+ char fileName[FileWrapper::kMaxFileNameSize]);
+
+ WebRtc_Word32 SetTraceCallbackImpl(TraceCallback* callback);
+
+ void AddImpl(const TraceLevel level, const TraceModule module,
+ const WebRtc_Word32 id, const char* msg);
+
+ bool StopThread();
+
+ bool TraceCheck(const TraceLevel level) const;
+
+protected:
+ TraceImpl();
+
+ static TraceImpl* StaticInstance(CountOperation count_operation,
+ const TraceLevel level = kTraceAll);
+
+ WebRtc_Word32 AddThreadId(char* traceMessage) const;
+
+ // OS specific implementations
+ virtual WebRtc_Word32 AddTime(char* traceMessage,
+ const TraceLevel level) const = 0;
+
+ virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const = 0;
+ virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const = 0;
+
+ static bool Run(void* obj);
+ bool Process();
+
+private:
+ friend class Trace;
+
+ WebRtc_Word32 AddLevel(char* szMessage, const TraceLevel level) const;
+
+ WebRtc_Word32 AddModuleAndId(char* traceMessage, const TraceModule module,
+ const WebRtc_Word32 id) const;
+
+ WebRtc_Word32 AddMessage(char* traceMessage,
+ const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
+ const WebRtc_UWord16 writtenSoFar) const;
+
+ void AddMessageToList(
+ const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
+ const WebRtc_UWord16 length,
+ const TraceLevel level);
+
+ bool UpdateFileName(
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const WebRtc_UWord32 newCount) const;
+
+ bool CreateFileName(
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const WebRtc_UWord32 newCount) const;
+
+ void WriteToFile();
+
+ CriticalSectionWrapper* _critsectInterface;
+ TraceCallback* _callback;
+ WebRtc_UWord32 _rowCountText;
+ WebRtc_UWord32 _fileCountText;
+
+ FileWrapper& _traceFile;
+ ThreadWrapper& _thread;
+ EventWrapper& _event;
+
+ // _critsectArray protects _activeQueue
+ CriticalSectionWrapper* _critsectArray;
+ WebRtc_UWord16 _nextFreeIdx[WEBRTC_TRACE_NUM_ARRAY];
+ TraceLevel _level[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
+ WebRtc_UWord16 _length[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
+ char* _messageQueue[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
+ WebRtc_UWord8 _activeQueue;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
diff --git a/trunk/src/system_wrappers/source/trace_impl_no_op.cc b/trunk/src/system_wrappers/source/trace_impl_no_op.cc
new file mode 100644
index 0000000..1752871
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_impl_no_op.cc
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+
+namespace webrtc {
+
+void Trace::CreateTrace()
+{
+}
+
+void Trace::ReturnTrace()
+{
+}
+
+WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 /*filter*/)
+{
+ return 0;
+}
+
+WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& /*filter*/)
+{
+ return 0;
+}
+
+WebRtc_Word32 Trace::TraceFile(
+ WebRtc_Word8 /*fileName*/[1024])
+{
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceFile(const WebRtc_Word8* /*fileName*/,
+ const bool /*addFileCounter*/)
+{
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* /*callback*/)
+{
+ return -1;
+}
+
+void Trace::Add(const TraceLevel /*level*/, const TraceModule /*module*/,
+ const WebRtc_Word32 /*id*/, const char* /*msg*/, ...)
+
+{
+}
+
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_posix.cc b/trunk/src/system_wrappers/source/trace_posix.cc
new file mode 100644
index 0000000..caf1128
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_posix.cc
@@ -0,0 +1,118 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace_posix.h"
+
+#include <cassert>
+#include <stdarg.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/time.h>
+#include <time.h>
+#ifdef WEBRTC_ANDROID
+ #include <pthread.h>
+#else
+ #include <iostream>
+#endif
+
+#if defined(_DEBUG)
+ #define BUILDMODE "d"
+#elif defined(DEBUG)
+ #define BUILDMODE "d"
+#elif defined(NDEBUG)
+ #define BUILDMODE "r"
+#else
+ #define BUILDMODE "?"
+#endif
+#define BUILDTIME __TIME__
+#define BUILDDATE __DATE__
+// example: "Oct 10 2002 12:05:30 r"
+#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
+
+namespace webrtc {
+TracePosix::TracePosix()
+{
+ struct timeval systemTimeHighRes;
+ gettimeofday(&systemTimeHighRes, 0);
+ _prevAPITickCount = _prevTickCount = systemTimeHighRes.tv_sec;
+}
+
+TracePosix::~TracePosix()
+{
+ StopThread();
+}
+
+WebRtc_Word32 TracePosix::AddTime(char* traceMessage,
+ const TraceLevel level) const
+{
+ struct timeval systemTimeHighRes;
+ if (gettimeofday(&systemTimeHighRes, 0) == -1)
+ {
+ return -1;
+ }
+ const struct tm* systemTime =
+ localtime(&systemTimeHighRes.tv_sec);
+
+ const WebRtc_UWord32 ms_time = systemTimeHighRes.tv_usec / 1000;
+ WebRtc_UWord32 prevTickCount = 0;
+ if (level == kTraceApiCall)
+ {
+ prevTickCount = _prevTickCount;
+ _prevTickCount = ms_time;
+ } else {
+ prevTickCount = _prevAPITickCount;
+ _prevAPITickCount = ms_time;
+ }
+ WebRtc_UWord32 dwDeltaTime = ms_time - prevTickCount;
+ if (prevTickCount == 0)
+ {
+ dwDeltaTime = 0;
+ }
+ if (dwDeltaTime > 0x0fffffff)
+ {
+ // Either wraparound or data race.
+ dwDeltaTime = 0;
+ }
+ if(dwDeltaTime > 99999)
+ {
+ dwDeltaTime = 99999;
+ }
+
+ sprintf(traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime->tm_hour,
+ systemTime->tm_min, systemTime->tm_sec, ms_time,
+ static_cast<unsigned long>(dwDeltaTime));
+ // Messages are 22 characters.
+ return 22;
+}
+
+WebRtc_Word32 TracePosix::AddBuildInfo(char* traceMessage) const
+{
+ sprintf(traceMessage, "Build info: %s", BUILDINFO);
+ // Include NULL termination (hence + 1).
+ return strlen(traceMessage) + 1;
+}
+
+WebRtc_Word32 TracePosix::AddDateTimeInfo(char* traceMessage) const
+{
+ time_t t;
+ time(&t);
+ sprintf(traceMessage, "Local Date: %s", ctime(&t));
+ WebRtc_Word32 len = static_cast<WebRtc_Word32>(strlen(traceMessage));
+
+ if ('\n' == traceMessage[len - 1])
+ {
+ traceMessage[len - 1] = '\0';
+ --len;
+ }
+
+ // Messages is 12 characters.
+ return len + 1;
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_posix.h b/trunk/src/system_wrappers/source/trace_posix.h
new file mode 100644
index 0000000..8c37cd2
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_posix.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
+
+#include "critical_section_wrapper.h"
+#include "trace_impl.h"
+
+namespace webrtc {
+class TracePosix : public TraceImpl
+{
+public:
+ TracePosix();
+ virtual ~TracePosix();
+
+ virtual WebRtc_Word32 AddTime(char* traceMessage,
+ const TraceLevel level) const;
+
+ virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const;
+ virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const;
+
+private:
+ volatile mutable WebRtc_UWord32 _prevAPITickCount;
+ volatile mutable WebRtc_UWord32 _prevTickCount;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/trace_unittest.cc b/trunk/src/system_wrappers/source/trace_unittest.cc
new file mode 100644
index 0000000..982e715
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_unittest.cc
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/trace.h"
+
+#include "gtest/gtest.h"
+#include "system_wrappers/source/cpu_measurement_harness.h"
+#include "testsupport/fileutils.h"
+
+using webrtc::CpuMeasurementHarness;
+using webrtc::Trace;
+using webrtc::kTraceWarning;
+using webrtc::kTraceUtility;
+
+class Logger : public webrtc::CpuTarget {
+ public:
+ Logger() {
+ Trace::CreateTrace();
+ std::string trace_file = webrtc::test::OutputPath() +
+ "trace_unittest.txt";
+ Trace::SetTraceFile(trace_file.c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ }
+ virtual ~Logger() {
+ Trace::ReturnTrace();
+ }
+
+ virtual bool DoWork() {
+ // Use input paremeters to WEBRTC_TRACE that are not likely to be removed
+ // in future code. E.g. warnings will likely be kept and this file is in
+ // utility so it should use kTraceUtility.
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, 0, "Log line");
+ return true;
+ }
+};
+
+// This test is disabled because it measures CPU usage. This is flaky because
+// the CPU usage for a machine may spike due to OS or other application.
+TEST(TraceTest, DISABLED_CpuUsage) {
+ Logger logger;
+ const int periodicity_ms = 1;
+ const int iterations_per_period = 10;
+ const int duration_ms = 1000;
+ CpuMeasurementHarness* cpu_harness =
+ CpuMeasurementHarness::Create(&logger, periodicity_ms,
+ iterations_per_period, duration_ms);
+ cpu_harness->Run();
+ const int average_cpu = cpu_harness->AverageCpu();
+ EXPECT_GE(5, average_cpu);
+}
diff --git a/trunk/src/system_wrappers/source/trace_win.cc b/trunk/src/system_wrappers/source/trace_win.cc
new file mode 100644
index 0000000..db3da76
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_win.cc
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace_win.h"
+
+#include <cassert>
+#include <stdarg.h>
+
+#include "Mmsystem.h"
+
+#if defined(_DEBUG)
+ #define BUILDMODE "d"
+#elif defined(DEBUG)
+ #define BUILDMODE "d"
+#elif defined(NDEBUG)
+ #define BUILDMODE "r"
+#else
+ #define BUILDMODE "?"
+#endif
+#define BUILDTIME __TIME__
+#define BUILDDATE __DATE__
+// Example: "Oct 10 2002 12:05:30 r"
+#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
+
+namespace webrtc {
+TraceWindows::TraceWindows()
+ : _prevAPITickCount(0),
+ _prevTickCount(0)
+{
+}
+
+TraceWindows::~TraceWindows()
+{
+ StopThread();
+}
+
+WebRtc_Word32 TraceWindows::AddTime(char* traceMessage,
+ const TraceLevel level) const
+{
+ WebRtc_UWord32 dwCurrentTime = timeGetTime();
+ SYSTEMTIME systemTime;
+ GetSystemTime(&systemTime);
+
+ if(level == kTraceApiCall)
+ {
+ WebRtc_UWord32 dwDeltaTime = dwCurrentTime- _prevTickCount;
+ _prevTickCount = dwCurrentTime;
+
+ if(_prevTickCount == 0)
+ {
+ dwDeltaTime = 0;
+ }
+ if(dwDeltaTime > 0x0fffffff)
+ {
+ // Either wraparound or data race.
+ dwDeltaTime = 0;
+ }
+ if(dwDeltaTime > 99999)
+ {
+ dwDeltaTime = 99999;
+ }
+
+ sprintf (traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.wHour,
+ systemTime.wMinute, systemTime.wSecond,
+ systemTime.wMilliseconds, dwDeltaTime);
+ } else {
+ WebRtc_UWord32 dwDeltaTime = dwCurrentTime - _prevAPITickCount;
+ _prevAPITickCount = dwCurrentTime;
+
+ if(_prevAPITickCount == 0)
+ {
+ dwDeltaTime = 0;
+ }
+ if(dwDeltaTime > 0x0fffffff)
+ {
+ // Either wraparound or data race.
+ dwDeltaTime = 0;
+ }
+ if(dwDeltaTime > 99999)
+ {
+ dwDeltaTime = 99999;
+ }
+ sprintf (traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.wHour,
+ systemTime.wMinute, systemTime.wSecond,
+ systemTime.wMilliseconds, dwDeltaTime);
+ }
+ // Messages is 12 characters.
+ return 22;
+}
+
+WebRtc_Word32 TraceWindows::AddBuildInfo(char* traceMessage) const
+{
+ // write data and time to text file
+ sprintf(traceMessage, "Build info: %s", BUILDINFO);
+ // Include NULL termination (hence + 1).
+ return static_cast<WebRtc_Word32>(strlen(traceMessage)+1);
+}
+
+WebRtc_Word32 TraceWindows::AddDateTimeInfo(char* traceMessage) const
+{
+ _prevAPITickCount = timeGetTime();
+ _prevTickCount = _prevAPITickCount;
+
+ SYSTEMTIME sysTime;
+ GetLocalTime (&sysTime);
+
+ TCHAR szDateStr[20];
+ TCHAR szTimeStr[20];
+ TCHAR *pSzDateStr = szDateStr;
+ TCHAR *pSzTimeStr = szTimeStr;
+
+ // Create date string (e.g. Apr 04 2002)
+ GetDateFormat(LOCALE_SYSTEM_DEFAULT, 0, &sysTime, TEXT("MMM dd yyyy"),
+ szDateStr, 20);
+
+ // Create time string (e.g. 15:32:08)
+ GetTimeFormat(LOCALE_SYSTEM_DEFAULT, 0, &sysTime, TEXT("HH':'mm':'ss"),
+ szTimeStr, 20);
+
+ sprintf(traceMessage, "Local Date: %s Local Time: %s", szDateStr,
+ szTimeStr);
+
+ // Include NULL termination (hence + 1).
+ return static_cast<WebRtc_Word32>(strlen(traceMessage)+ 1);
+}
+} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_win.h b/trunk/src/system_wrappers/source/trace_win.h
new file mode 100644
index 0000000..803198e
--- /dev/null
+++ b/trunk/src/system_wrappers/source/trace_win.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
+
+#include "trace_impl.h"
+#include <stdio.h>
+#include <windows.h>
+
+namespace webrtc {
+class TraceWindows : public TraceImpl
+{
+public:
+ TraceWindows();
+ virtual ~TraceWindows();
+
+ virtual WebRtc_Word32 AddTime(char* traceMessage,
+ const TraceLevel level) const;
+
+ virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const;
+ virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const;
+private:
+ volatile mutable WebRtc_UWord32 _prevAPITickCount;
+ volatile mutable WebRtc_UWord32 _prevTickCount;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/test/TestSort/TestSort.cc b/trunk/src/system_wrappers/test/TestSort/TestSort.cc
new file mode 100644
index 0000000..6846a71
--- /dev/null
+++ b/trunk/src/system_wrappers/test/TestSort/TestSort.cc
@@ -0,0 +1,265 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+#include <algorithm>
+#include <cstring>
+
+#include "sort.h"
+#include "tick_util.h"
+
+// Excellent work polluting the global namespace Visual Studio...
+#undef max
+#undef min
+#include <limits>
+
+template<typename KeyType>
+struct LotsOfData
+{
+ KeyType key;
+ char data[64];
+};
+
+template<typename DataType>
+int Compare(const void* dataX, const void* dataY)
+{
+ DataType dataX = (DataType)*(const DataType*)dataX;
+ DataType dataY = (DataType)*(const DataType*)dataY;
+ if (dataX > dataY)
+ {
+ return 1;
+ }
+ else if (dataX < dataY)
+ {
+ return -1;
+ }
+
+ return 0;
+};
+
+template<typename DataType, typename KeyType>
+int CompareKey(const void* dataX, const void* dataY)
+{
+ KeyType keyX = ((const DataType*)dataX)->key;
+ KeyType keyY = ((const DataType*)dataY)->key;
+ if (keyX > keyY)
+ {
+ return 1;
+ }
+ else if (keyX < keyY)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+template<typename DataType>
+struct KeyLessThan
+{
+ bool operator()(const DataType &dataX, const DataType &dataY) const
+ {
+ return dataX.key < dataY.key;
+ }
+};
+
+const char* TypeEnumToString(webrtc::Type type)
+{
+ switch (type)
+ {
+ using namespace webrtc;
+ case TYPE_Word8:
+ return "Word8";
+ case TYPE_UWord8:
+ return "UWord8";
+ case TYPE_Word16:
+ return "Word16";
+ case TYPE_UWord16:
+ return "UWord16";
+ case TYPE_Word32:
+ return "Word32";
+ case TYPE_UWord32:
+ return "UWord32";
+ case TYPE_Word64:
+ return "Word64";
+ case TYPE_UWord64:
+ return "UWord64";
+ case TYPE_Float32:
+ return "Float32";
+ case TYPE_Float64:
+ return "Float64";
+ default:
+ return "Unrecognized";
+ }
+}
+
+template<typename Type>
+Type TypedRand()
+{
+ if (std::numeric_limits<Type>::is_integer)
+ {
+ double floatRand = static_cast<double>(rand()) / RAND_MAX;
+ if (std::numeric_limits<Type>::is_signed)
+ {
+ floatRand -= 0.5;
+ }
+
+ // Uniform [-max()/2, max()/2] for signed
+ // [0, max()] for unsigned
+ return static_cast<Type>(floatRand * std::numeric_limits<Type>::max());
+ }
+ else // Floating point
+ {
+ // Uniform [-0.5, 0.5]
+ // The outer cast is to remove template warnings.
+ return static_cast<Type>((static_cast<Type>(rand()) / RAND_MAX) - 0.5);
+ }
+}
+
+template<typename KeyType>
+void RunSortTest(webrtc::Type sortType, bool keySort)
+{
+ enum { DataLength = 1000 };
+ enum { NumOfTests = 10000 };
+ KeyType key[DataLength];
+ KeyType keyRef[DataLength];
+ LotsOfData<KeyType> data[DataLength];
+ LotsOfData<KeyType> dataRef[DataLength];
+ WebRtc_Word32 retVal = 0;
+
+ if (keySort)
+ {
+ printf("Running %s KeySort() tests...\n", TypeEnumToString(sortType));
+ }
+ else
+ {
+ printf("Running %s Sort() tests...\n", TypeEnumToString(sortType));
+ }
+
+ TickInterval accTicks;
+ for (int i = 0; i < NumOfTests; i++)
+ {
+ for (int j = 0; j < DataLength; j++)
+ {
+ key[j] = TypedRand<KeyType>();
+ data[j].key = key[j];
+ // Write index to payload. We use this later for verification.
+ sprintf(data[j].data, "%d", j);
+ }
+
+ memcpy(dataRef, data, sizeof(data));
+ memcpy(keyRef, key, sizeof(key));
+
+ retVal = 0;
+ TickTime t0 = TickTime::Now();
+ if (keySort)
+ {
+ retVal = webrtc::KeySort(data, key, DataLength, sizeof(LotsOfData<KeyType>),
+ sortType);
+
+ //std::sort(data, data + DataLength, KeyLessThan<KeyType>());
+ //qsort(data, DataLength, sizeof(LotsOfData<KeyType>),
+ // CompareKey<LotsOfData<KeyType>, KeyType>);
+ }
+ else
+ {
+ retVal = webrtc::Sort(key, DataLength, sortType);
+
+ //std::sort(key, key + DataLength);
+ //qsort(key, DataLength, sizeof(KeyType), Compare<KeyType>);
+ }
+ TickTime t1 = TickTime::Now();
+ accTicks += (t1 - t0);
+
+ if (retVal != 0)
+ {
+ printf("Test failed at iteration %d:\n", i);
+ printf("Sort returned an error. ");
+ printf("It likely does not support the requested type\nExiting...\n");
+ exit(0);
+ }
+
+ // Reference sort.
+ if (!keySort)
+ {
+ std::sort(keyRef, keyRef + DataLength);
+ }
+
+ if (keySort)
+ {
+ for (int j = 0; j < DataLength - 1; j++)
+ {
+ if (data[j].key > data[j + 1].key)
+ {
+ printf("Test failed at iteration %d:\n", i);
+ printf("Keys are not monotonically increasing\nExiting...\n");
+ exit(0);
+ }
+
+ int index = atoi(data[j].data);
+ if (index < 0 || index >= DataLength || data[j].key != dataRef[index].key)
+ {
+ printf("Test failed at iteration %d:\n", i);
+ printf("Payload data is corrupt\nExiting...\n");
+ exit(0);
+ }
+ }
+ }
+ else
+ {
+ for (int j = 0; j < DataLength - 1; j++)
+ {
+ if (key[j] > key[j + 1])
+ {
+ printf("Test failed at iteration %d:\n", i);
+ printf("Data is not monotonically increasing\nExiting...\n");
+ exit(0);
+ }
+ }
+
+ if (memcmp(key, keyRef, sizeof(key)) != 0)
+ {
+ printf("Test failed at iteration %d:\n", i);
+ printf("Sort data differs from std::sort reference\nExiting...\n");
+ exit(0);
+ }
+ }
+ }
+
+ printf("Compliance test passed over %d iterations\n", NumOfTests);
+
+ WebRtc_Word64 executeTime = accTicks.Milliseconds();
+ printf("Execute time: %.2f s\n\n", (float)executeTime / 1000);
+}
+
+int main()
+{
+ // Seed rand().
+ srand(42);
+ bool keySort = false;
+ for (int i = 0; i < 2; i++) {
+ RunSortTest<WebRtc_Word8>(webrtc::TYPE_Word8, keySort);
+ RunSortTest<WebRtc_UWord8>(webrtc::TYPE_UWord8, keySort);
+ RunSortTest<WebRtc_Word16>(webrtc::TYPE_Word16, keySort);
+ RunSortTest<WebRtc_UWord16>(webrtc::TYPE_UWord16, keySort);
+ RunSortTest<WebRtc_Word32>(webrtc::TYPE_Word32, keySort);
+ RunSortTest<WebRtc_UWord32>(webrtc::TYPE_UWord32, keySort);
+ RunSortTest<WebRtc_Word64>(webrtc::TYPE_Word64, keySort);
+ RunSortTest<WebRtc_UWord64>(webrtc::TYPE_UWord64, keySort);
+ RunSortTest<float>(webrtc::TYPE_Float32, keySort);
+ RunSortTest<double>(webrtc::TYPE_Float64, keySort);
+
+ keySort = !keySort;
+ }
+
+ printf("All tests passed\n");
+
+ return 0;
+}
diff --git a/trunk/src/system_wrappers/test/list/list.cc b/trunk/src/system_wrappers/test/list/list.cc
new file mode 100644
index 0000000..5c4f0b9
--- /dev/null
+++ b/trunk/src/system_wrappers/test/list/list.cc
@@ -0,0 +1,174 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "list_wrapper.h"
+
+const int kNumberOfElements = 10;
+
+void FailTest(bool failed)
+{
+ if (failed)
+ {
+ printf("Test failed!\n");
+ printf("Press enter to continue:");
+ getchar();
+ exit(0);
+ }
+}
+
+int GetStoredIntegerValue(ListItem* list_item)
+{
+ void* list_item_pointer = list_item->GetItem();
+ if (list_item_pointer != NULL)
+ {
+ return *(reinterpret_cast<int*>(list_item_pointer));
+ }
+ return static_cast<int>(list_item->GetUnsignedItem());
+}
+
+void PrintList(ListWrapper& list)
+{
+ ListItem* list_item = list.First();
+ printf("List: ");
+ while (list_item != NULL)
+ {
+ int item_value = GetStoredIntegerValue(list_item);
+ FailTest(item_value < 0);
+ printf(" %d",item_value);
+ list_item = list.Next(list_item);
+ }
+ printf("\n");
+}
+
+// The list should always be in ascending order
+void ListSanity(ListWrapper& list)
+{
+ if(list.Empty())
+ {
+ return;
+ }
+ ListItem* item_iter = list.First();
+ // Fake a previous value for the first iteration
+ int previous_value = GetStoredIntegerValue(item_iter) - 1;
+ while (item_iter != NULL)
+ {
+ const int value = GetStoredIntegerValue(item_iter);
+ FailTest(value != previous_value + 1);
+ previous_value = value;
+ item_iter = list.Next(item_iter);
+ }
+}
+
+int main(int /*argc*/, char* /*argv*/[])
+{
+ printf("List Test:\n");
+ int element_array[kNumberOfElements];
+ for (int i = 0; i < kNumberOfElements; i++)
+ {
+ element_array[i] = i;
+ }
+ // Test PushBack 1
+ ListWrapper test_list;
+ for (int i = 2; i < kNumberOfElements - 2; i++)
+ {
+ FailTest(test_list.PushBack((void*)&element_array[i]) != 0);
+ }
+ // Test PushBack 2
+ FailTest(test_list.PushBack(element_array[kNumberOfElements - 2]) != 0);
+ FailTest(test_list.PushBack(element_array[kNumberOfElements - 1]) != 0);
+ // Test PushFront 2
+ FailTest(test_list.PushFront(element_array[1]) != 0);
+ // Test PushFront 1
+ FailTest(test_list.PushFront((void*)&element_array[0]) != 0);
+ // Test GetSize
+ FailTest(test_list.GetSize() != kNumberOfElements);
+ PrintList(test_list);
+ //Test PopFront
+ FailTest(test_list.PopFront() != 0);
+ //Test PopBack
+ FailTest(test_list.PopBack() != 0);
+ // Test GetSize
+ FailTest(test_list.GetSize() != kNumberOfElements - 2);
+ // Test Empty
+ FailTest(test_list.Empty());
+ // Test First
+ ListItem* first_item = test_list.First();
+ FailTest(first_item == NULL);
+ // Test Last
+ ListItem* last_item = test_list.Last();
+ FailTest(last_item == NULL);
+ // Test Next
+ ListItem* second_item = test_list.Next(first_item);
+ FailTest(second_item == NULL);
+ FailTest(test_list.Next(last_item) != NULL);
+ FailTest(test_list.Next(NULL) != NULL);
+ // Test Previous
+ ListItem* second_to_last_item = test_list.Previous(last_item);
+ FailTest(second_to_last_item == NULL);
+ FailTest(test_list.Previous(first_item) != NULL);
+ FailTest(test_list.Previous(NULL) != NULL);
+ // Test GetUnsignedItem
+ FailTest(last_item->GetUnsignedItem() !=
+ kNumberOfElements - 2);
+ FailTest(last_item->GetItem() !=
+ NULL);
+ // Test GetItem
+ FailTest(GetStoredIntegerValue(second_to_last_item) !=
+ kNumberOfElements - 3);
+ FailTest(second_to_last_item->GetUnsignedItem() != 0);
+ // Pop last and first since they are pushed as unsigned items.
+ FailTest(test_list.PopFront() != 0);
+ FailTest(test_list.PopBack() != 0);
+ // Test Insert. Please note that old iterators are no longer valid at
+ // this point.
+ ListItem* insert_item_last = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
+ FailTest(test_list.Insert(test_list.Last(),insert_item_last) != 0);
+ FailTest(test_list.Insert(NULL,insert_item_last) == 0);
+ ListItem* insert_item_last2 = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
+ FailTest(test_list.Insert(insert_item_last2,NULL) == 0);
+ // test InsertBefore
+ ListItem* insert_item_first = new ListItem(reinterpret_cast<void*>(&element_array[1]));
+ FailTest(test_list.InsertBefore(test_list.First(),insert_item_first) != 0);
+ FailTest(test_list.InsertBefore(NULL,insert_item_first) == 0);
+ ListItem* insert_item_first2 = new ListItem(reinterpret_cast<void*>(&element_array[1]));
+ FailTest(test_list.InsertBefore(insert_item_first2,NULL) == 0);
+ PrintList(test_list);
+ ListSanity(test_list);
+ // Erase the whole list
+ int counter = 0;
+ while (test_list.PopFront() == 0)
+ {
+ FailTest(counter++ > kNumberOfElements);
+ }
+ PrintList(test_list);
+ // Test APIs when list is empty
+ FailTest(test_list.GetSize() != 0);
+ FailTest(test_list.PopFront() != -1);
+ FailTest(test_list.PopBack() != -1);
+ FailTest(!test_list.Empty());
+ FailTest(test_list.First() != NULL);
+ FailTest(test_list.Last() != NULL);
+ FailTest(test_list.Next(NULL) != NULL);
+ FailTest(test_list.Previous(NULL) != NULL);
+ FailTest(test_list.Erase(NULL) != -1);
+ // Test Insert APIs when list is empty
+ ListItem* new_item = new ListItem(reinterpret_cast<void*>(&element_array[0]));
+ FailTest(test_list.Insert(NULL,new_item) != 0);
+ FailTest(test_list.Empty());
+ FailTest(test_list.PopFront() != 0);
+ ListItem* new_item2 = new ListItem(reinterpret_cast<void*>(&element_array[0]));
+ FailTest(test_list.InsertBefore(NULL,new_item2) != 0);
+ FailTest(test_list.Empty());
+
+ printf("Tests passed successfully!\n");
+}
diff --git a/trunk/src/system_wrappers/test/map/map.cc b/trunk/src/system_wrappers/test/map/map.cc
new file mode 100644
index 0000000..8a0d3e3
--- /dev/null
+++ b/trunk/src/system_wrappers/test/map/map.cc
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "map_wrapper.h"
+
+const int kNumberOfElements = 10;
+
+void FailTest(bool failed)
+{
+ if (failed)
+ {
+ printf("Test failed!\n");
+ printf("Press enter to continue:");
+ getchar();
+ exit(0);
+ }
+}
+
+int GetStoredIntegerValue(MapItem* map_item)
+{
+ void* map_item_pointer = map_item->GetItem();
+ if (map_item_pointer != NULL)
+ {
+ return *(reinterpret_cast<int*>(map_item_pointer));
+ }
+ return static_cast<int>(map_item->GetUnsignedId());
+}
+
+void PrintMap(MapWrapper& map)
+{
+ MapItem* map_item = map.First();
+ printf("Map: ");
+ while (map_item != NULL)
+ {
+ int item_value = GetStoredIntegerValue(map_item);
+ FailTest(item_value < 0);
+ printf(" %d",item_value);
+ map_item = map.Next(map_item);
+ }
+ printf("\n");
+}
+
+int main(int /*argc*/, char* /*argv*/[])
+{
+ int element_array[kNumberOfElements];
+ for (int i = 0; i < kNumberOfElements; i++)
+ {
+ element_array[i] = i;
+ }
+ // Test insert
+ MapWrapper test_map;
+ for (int i = 0; i < kNumberOfElements; i++)
+ {
+ test_map.Insert(i,(void*)&element_array[i]);
+ }
+ // Test Erase1
+ MapItem* remove_item = test_map.Find(2);
+ FailTest(remove_item == NULL);
+ FailTest(test_map.Erase(remove_item) != 0);
+ FailTest(test_map.Find(2) != NULL);
+ remove_item = NULL;
+ FailTest(test_map.Erase(remove_item) != -1);
+ // Test Erase2
+ FailTest(test_map.Erase(1) != 0);
+ FailTest(test_map.Find(1) != NULL);
+ FailTest(test_map.Erase(1) != -1);
+ // Test Size
+ FailTest(test_map.Size() != kNumberOfElements - 2);
+ PrintMap(test_map);
+ // Test First
+ MapItem* first_item = test_map.First();
+ FailTest(first_item == NULL);
+ FailTest(GetStoredIntegerValue(first_item) != 0);
+ // Test Last
+ MapItem* last_item = test_map.Last();
+ FailTest(last_item == NULL);
+ FailTest(GetStoredIntegerValue(last_item) != 9);
+ // Test Next
+ MapItem* second_item = test_map.Next(first_item);
+ FailTest(second_item == NULL);
+ FailTest(GetStoredIntegerValue(second_item) != 3);
+ FailTest(test_map.Next(last_item) != NULL);
+ // Test Previous
+ MapItem* second_to_last_item = test_map.Previous(last_item);
+ FailTest(second_to_last_item == NULL);
+ FailTest(GetStoredIntegerValue(second_to_last_item) != 8);
+ FailTest(test_map.Previous(first_item) != NULL);
+ // Test Find (only improper usage untested)
+ FailTest(test_map.Find(kNumberOfElements + 2) != NULL);
+ // Test GetId
+ FailTest(*(reinterpret_cast<int*>(second_to_last_item->GetItem())) !=
+ second_to_last_item->GetId());
+ FailTest(second_to_last_item->GetUnsignedId() !=
+ static_cast<unsigned int>(second_to_last_item->GetId()));
+ // Test SetItem
+ int swapped_item = kNumberOfElements;
+ last_item->SetItem(reinterpret_cast<void*>(&swapped_item));
+ FailTest(GetStoredIntegerValue(last_item) !=
+ swapped_item);
+
+ printf("Tests passed successfully!\n");
+}
diff --git a/trunk/src/typedefs.h b/trunk/src/typedefs.h
new file mode 100644
index 0000000..ba87309
--- /dev/null
+++ b/trunk/src/typedefs.h
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains platform-specific typedefs and defines.
+
+#ifndef WEBRTC_TYPEDEFS_H_
+#define WEBRTC_TYPEDEFS_H_
+
+// Reserved words definitions
+// TODO(andrew): Look at removing these.
+#define WEBRTC_EXTERN extern
+#define G_CONST const
+#define WEBRTC_INLINE extern __inline
+
+// Define WebRTC preprocessor identifiers based on the current build platform.
+// TODO(andrew): Clean these up. We can probably remove everything in this
+// block.
+// - TARGET_MAC_INTEL and TARGET_MAC aren't used anywhere.
+// - In the few places where TARGET_PC is used, it should be replaced by
+// something more specific.
+// - Do we really support PowerPC? Probably not. Remove WEBRTC_MAC_INTEL
+// from build/common.gypi as well.
+#if defined(WIN32)
+ // Windows & Windows Mobile.
+ #if !defined(WEBRTC_TARGET_PC)
+ #define WEBRTC_TARGET_PC
+ #endif
+#elif defined(__APPLE__)
+ // Mac OS X.
+ #if defined(__LITTLE_ENDIAN__ )
+ #if !defined(WEBRTC_TARGET_MAC_INTEL)
+ #define WEBRTC_TARGET_MAC_INTEL
+ #endif
+ #else
+ #if !defined(WEBRTC_TARGET_MAC)
+ #define WEBRTC_TARGET_MAC
+ #endif
+ #endif
+#else
+ // Linux etc.
+ #if !defined(WEBRTC_TARGET_PC)
+ #define WEBRTC_TARGET_PC
+ #endif
+#endif
+
+// Derived from Chromium's build/build_config.h
+// Processor architecture detection. For more info on what's defined, see:
+// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+// http://www.agner.org/optimize/calling_conventions.pdf
+// or with gcc, run: "echo | gcc -E -dM -"
+// TODO(andrew): replace WEBRTC_LITTLE_ENDIAN with WEBRTC_ARCH_LITTLE_ENDIAN?
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+// TODO(andrew): We'd prefer to control platform defines here, but this is
+// currently provided by the Android makefiles. Commented to avoid duplicate
+// definition warnings.
+//#define WEBRTC_ARCH_ARM
+// TODO(andrew): Chromium uses the following two defines. Should we switch?
+//#define WEBRTC_ARCH_ARM_FAMILY
+//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in typedefs.h
+#endif
+
+#if defined(__SSE2__) || defined(_MSC_VER)
+#define WEBRTC_USE_SSE2
+#endif
+
+#if defined(WEBRTC_TARGET_PC)
+
+#if !defined(_MSC_VER)
+ #include <stdint.h>
+#else
+ // Define C99 equivalent types.
+ // Since MSVC doesn't include these headers, we have to write our own
+ // version to provide a compatibility layer between MSVC and the WebRTC
+ // headers.
+ typedef signed char int8_t;
+ typedef signed short int16_t;
+ typedef signed int int32_t;
+ typedef signed long long int64_t;
+ typedef unsigned char uint8_t;
+ typedef unsigned short uint16_t;
+ typedef unsigned int uint32_t;
+ typedef unsigned long long uint64_t;
+#endif
+
+#if defined(WIN32)
+ typedef __int64 WebRtc_Word64;
+ typedef unsigned __int64 WebRtc_UWord64;
+#else
+ typedef int64_t WebRtc_Word64;
+ typedef uint64_t WebRtc_UWord64;
+#endif
+ typedef int32_t WebRtc_Word32;
+ typedef uint32_t WebRtc_UWord32;
+ typedef int16_t WebRtc_Word16;
+ typedef uint16_t WebRtc_UWord16;
+ typedef char WebRtc_Word8;
+ typedef uint8_t WebRtc_UWord8;
+
+ // Define endian for the platform
+ #define WEBRTC_LITTLE_ENDIAN
+
+#elif defined(WEBRTC_TARGET_MAC_INTEL)
+ #include <stdint.h>
+
+ typedef int64_t WebRtc_Word64;
+ typedef uint64_t WebRtc_UWord64;
+ typedef int32_t WebRtc_Word32;
+ typedef uint32_t WebRtc_UWord32;
+ typedef int16_t WebRtc_Word16;
+ typedef char WebRtc_Word8;
+ typedef uint16_t WebRtc_UWord16;
+ typedef uint8_t WebRtc_UWord8;
+
+ // Define endian for the platform
+ #define WEBRTC_LITTLE_ENDIAN
+
+#else
+ #error "No platform defined for WebRTC type definitions (typedefs.h)"
+#endif
+
+#endif // WEBRTC_TYPEDEFS_H_
diff --git a/trunk/src/video_engine/Android.mk b/trunk/src/video_engine/Android.mk
new file mode 100644
index 0000000..164c289
--- /dev/null
+++ b/trunk/src/video_engine/Android.mk
@@ -0,0 +1,94 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_vie_core
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ vie_base_impl.cc \
+ vie_capture_impl.cc \
+ vie_codec_impl.cc \
+ vie_encryption_impl.cc \
+ vie_external_codec_impl.cc \
+ vie_file_impl.cc \
+ vie_image_process_impl.cc \
+ vie_impl.cc \
+ vie_network_impl.cc \
+ vie_ref_count.cc \
+ vie_render_impl.cc \
+ vie_rtp_rtcp_impl.cc \
+ vie_shared_data.cc \
+ vie_capturer.cc \
+ vie_channel.cc \
+ vie_channel_manager.cc \
+ vie_encoder.cc \
+ vie_file_image.cc \
+ vie_file_player.cc \
+ vie_file_recorder.cc \
+ vie_frame_provider_base.cc \
+ vie_input_manager.cc \
+ vie_manager_base.cc \
+ vie_performance_monitor.cc \
+ vie_receiver.cc \
+ vie_remb.cc \
+ vie_renderer.cc \
+ vie_render_manager.cc \
+ vie_sender.cc \
+ vie_sync_module.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include\
+ $(LOCAL_PATH)/.. \
+ $(LOCAL_PATH)/../common_video/interface \
+ $(LOCAL_PATH)/../common_video/jpeg/main/interface \
+ $(LOCAL_PATH)/../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../modules/interface \
+ $(LOCAL_PATH)/../modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/../modules/media_file/interface \
+ $(LOCAL_PATH)/../modules/rtp_rtcp/interface \
+ $(LOCAL_PATH)/../modules/udp_transport/interface \
+ $(LOCAL_PATH)/../modules/utility/interface \
+ $(LOCAL_PATH)/../modules/video_capture/main/interface \
+ $(LOCAL_PATH)/../modules/video_capture/main/source \
+ $(LOCAL_PATH)/../modules/video_capture/main/source/Android \
+ $(LOCAL_PATH)/../modules/video_coding/codecs/interface \
+ $(LOCAL_PATH)/../modules/video_coding/main/interface \
+ $(LOCAL_PATH)/../modules/video_mixer/main/interface \
+ $(LOCAL_PATH)/../modules/video_processing/main/interface \
+ $(LOCAL_PATH)/../modules/video_render/main/interface \
+ $(LOCAL_PATH)/../system_wrappers/interface \
+ $(LOCAL_PATH)/../voice_engine/main/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/video_engine/OWNERS b/trunk/src/video_engine/OWNERS
new file mode 100644
index 0000000..ac607bd
--- /dev/null
+++ b/trunk/src/video_engine/OWNERS
@@ -0,0 +1,4 @@
+mflodman@webrtc.org
+perkj@webrtc.org
+wu@webrtc.org
+mallinath@webrtc.org
diff --git a/trunk/src/video_engine/include/vie_base.h b/trunk/src/video_engine/include/vie_base.h
new file mode 100644
index 0000000..2757a78
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_base.h
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Creating and deleting VideoEngine instances.
+// - Creating and deleting channels.
+// - Connect a video channel with a corresponding voice channel for audio/video
+// synchronization.
+// - Start and stop sending and receiving.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// Class used for all callbacks from ViEBase.
+class WEBRTC_DLLEXPORT ViEBaseObserver {
+ public:
+ // This method will be called periodically if the average system CPU usage
+ // exceeds 75%.
+ virtual void PerformanceAlarm(const unsigned int cpu_load) = 0;
+
+ protected:
+ virtual ~ViEBaseObserver() {}
+};
+
+class WEBRTC_DLLEXPORT VideoEngine {
+ public:
+ // Creates a VideoEngine object, which can then be used to acquire sub‐APIs.
+ static VideoEngine* Create();
+
+ // Deletes a VideoEngine instance.
+ static bool Delete(VideoEngine*& video_engine);
+
+ // Specifies the amount and type of trace information, which will be created
+ // by the VideoEngine.
+ static int SetTraceFilter(const unsigned int filter);
+
+ // Sets the name of the trace file and enables non‐encrypted trace messages.
+ static int SetTraceFile(const char* file_nameUTF8,
+ const bool add_file_counter = false);
+
+ // Installs the TraceCallback implementation to ensure that the VideoEngine
+ // user receives callbacks for generated trace messages.
+ static int SetTraceCallback(TraceCallback* callback);
+
+ // Android specific.
+ // Provides VideoEngine with pointers to objects supplied by the Java
+ // applications JNI interface.
+ static int SetAndroidObjects(void* java_vm, void* java_context);
+
+ protected:
+ VideoEngine() {}
+ virtual ~VideoEngine() {}
+};
+
+class WEBRTC_DLLEXPORT ViEBase {
+ public:
+ // Factory for the ViEBase sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViEBase* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViEBase sub-API and decreases an internal reference counter.
+ // Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Initiates all common parts of the VideoEngine.
+ virtual int Init() = 0;
+
+ // Connects a VideoEngine instance to a VoiceEngine instance for audio video
+ // synchronization.
+ virtual int SetVoiceEngine(VoiceEngine* voice_engine) = 0;
+
+ // Creates a new channel, either with a new encoder instance or by sharing
+ // encoder instance with an already created channel.
+ virtual int CreateChannel(int& video_channel) = 0;
+ virtual int CreateChannel(int& video_channel, int original_channel) = 0;
+
+ // Deletes an existing channel and releases the utilized resources.
+ virtual int DeleteChannel(const int video_channel) = 0;
+
+ // Specifies the VoiceEngine and VideoEngine channel pair to use for
+ // audio/video synchronization.
+ virtual int ConnectAudioChannel(const int video_channel,
+ const int audio_channel) = 0;
+
+ // Disconnects a previously paired VideoEngine and VoiceEngine channel pair.
+ virtual int DisconnectAudioChannel(const int video_channel) = 0;
+
+ // Starts sending packets to an already specified IP address and port number
+ // for a specified channel.
+ virtual int StartSend(const int video_channel) = 0;
+
+ // Stops packets from being sent for a specified channel.
+ virtual int StopSend(const int video_channel) = 0;
+
+ // Prepares VideoEngine for receiving packets on the specified channel.
+ virtual int StartReceive(const int video_channel) = 0;
+
+ // Stops receiving incoming RTP and RTCP packets on the specified channel.
+ virtual int StopReceive(const int video_channel) = 0;
+
+ // Registers an instance of a user implementation of the ViEBase
+ // observer.
+ virtual int RegisterObserver(ViEBaseObserver& observer) = 0;
+
+ // Removes an already registered instance of ViEBaseObserver.
+ virtual int DeregisterObserver() = 0;
+
+ // Retrieves the version information for VideoEngine and its components.
+ virtual int GetVersion(char version[1024]) = 0;
+
+ // Returns the last VideoEngine error code.
+ virtual int LastError() = 0;
+
+ protected:
+ ViEBase() {}
+ virtual ~ViEBase() {}
+};
+
+} // namespace webrtc
+
+#endif // #define WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_BASE_H_
diff --git a/trunk/src/video_engine/include/vie_capture.h b/trunk/src/video_engine/include/vie_capture.h
new file mode 100644
index 0000000..45d62e5
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_capture.h
@@ -0,0 +1,255 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Allocating capture devices.
+// - Connect a capture device with one or more channels.
+// - Start and stop capture devices.
+// - Getting capture device capabilities.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+class VideoCaptureModule;
+
+// This structure describes one set of the supported capabilities for a capture
+// device.
+struct CaptureCapability {
+ unsigned int width;
+ unsigned int height;
+ unsigned int maxFPS;
+ RawVideoType rawType;
+ VideoCodecType codecType;
+ unsigned int expectedCaptureDelay;
+ bool interlaced;
+ CaptureCapability() {
+ width = 0;
+ height = 0;
+ maxFPS = 0;
+ rawType = kVideoI420;
+ codecType = kVideoCodecUnknown;
+ expectedCaptureDelay = 0;
+ interlaced = false;
+ }
+};
+
+// This enumerator tells the current brightness alarm mode.
+enum Brightness {
+ Normal = 0,
+ Bright = 1,
+ Dark = 2
+};
+
+// This enumerator describes the capture alarm mode.
+enum CaptureAlarm {
+ AlarmRaised = 0,
+ AlarmCleared = 1
+};
+
+enum RotateCapturedFrame {
+ RotateCapturedFrame_0 = 0,
+ RotateCapturedFrame_90 = 90,
+ RotateCapturedFrame_180 = 180,
+ RotateCapturedFrame_270 = 270
+};
+
+struct ViEVideoFrameI420 {
+ ViEVideoFrameI420() {
+ y_plane = NULL;
+ u_plane = NULL;
+ v_plane = NULL;
+ y_pitch = 0;
+ u_pitch = 0;
+ v_pitch = 0;
+ width = 0;
+ height = 0;
+ }
+
+ unsigned char* y_plane;
+ unsigned char* u_plane;
+ unsigned char* v_plane;
+
+ int y_pitch;
+ int u_pitch;
+ int v_pitch;
+
+ unsigned short width;
+ unsigned short height;
+};
+
+// This class declares an abstract interface to be used when implementing
+// a user-defined capture device. This interface is not meant to be
+// implemented by the user. Instead, the user should call AllocateCaptureDevice
+// in the ViECapture interface, which will create a suitable implementation.
+// The user should then call IncomingFrame in this interface to deliver
+// captured frames to the system.
+class WEBRTC_DLLEXPORT ViEExternalCapture {
+ public:
+ ViEExternalCapture() {}
+ virtual ~ViEExternalCapture() {}
+
+ // This method is called by the user to deliver a new captured frame to
+ // VideoEngine.
+ virtual int IncomingFrame(unsigned char* video_frame,
+ unsigned int video_frame_length,
+ unsigned short width,
+ unsigned short height,
+ RawVideoType video_type,
+ unsigned long long capture_time = 0) = 0;
+
+ // This method is specifically for delivering a new captured I420 frame to
+ // VideoEngine.
+ virtual int IncomingFrameI420(
+ const ViEVideoFrameI420& video_frame,
+ unsigned long long capture_time = 0) = 0;
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterObserver() and
+// deregistered using DeregisterObserver().
+class WEBRTC_DLLEXPORT ViECaptureObserver {
+ public:
+ // This method is called if a bright or dark captured image is detected.
+ virtual void BrightnessAlarm(const int capture_id,
+ const Brightness brightness) = 0;
+
+ // This method is called periodically telling the capture device frame rate.
+ virtual void CapturedFrameRate(const int capture_id,
+ const unsigned char frame_rate) = 0;
+
+ // This method is called if the capture device stops delivering images to
+ // VideoEngine.
+ virtual void NoPictureAlarm(const int capture_id,
+ const CaptureAlarm alarm) = 0;
+
+ protected:
+ virtual ~ViECaptureObserver() {}
+};
+
+class WEBRTC_DLLEXPORT ViECapture {
+ public:
+ // Factory for the ViECapture sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViECapture* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViECapture sub-API and decreases an internal reference
+ // counter.
+ // Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Gets the number of available capture devices.
+ virtual int NumberOfCaptureDevices() = 0;
+
+ // Gets the name and unique id of a capture device.
+ virtual int GetCaptureDevice(unsigned int list_number,
+ char* device_nameUTF8,
+ const unsigned int device_nameUTF8Length,
+ char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length) = 0;
+
+ // Allocates a capture device to be used in VideoEngine.
+ virtual int AllocateCaptureDevice(const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ int& capture_id) = 0;
+
+ // Registers an external capture device to be used in VideoEngine
+ virtual int AllocateExternalCaptureDevice(
+ int& capture_id,
+ ViEExternalCapture *&external_capture) = 0;
+
+ // Use capture device using external capture module.
+ virtual int AllocateCaptureDevice(VideoCaptureModule& capture_module,
+ int& capture_id) = 0;
+
+ // Releases a capture device and makes it available for other applications.
+ virtual int ReleaseCaptureDevice(const int capture_id) = 0;
+
+ // This function connects a capture device with a channel. Multiple channels
+ // can be connected to the same capture device.
+ virtual int ConnectCaptureDevice(const int capture_id,
+ const int video_channel) = 0;
+
+ // Disconnects a capture device as input for a specified channel.
+ virtual int DisconnectCaptureDevice(const int video_channel) = 0;
+
+ // Makes a capture device start capturing video frames.
+ virtual int StartCapture(
+ const int capture_id,
+ const CaptureCapability& capture_capability = CaptureCapability()) = 0;
+
+ // Stops a started capture device from capturing video frames.
+ virtual int StopCapture(const int capture_id) = 0;
+
+ // Rotates captured frames before encoding and sending.
+ // Used on mobile devices with rotates cameras.
+ virtual int SetRotateCapturedFrames(const int capture_id,
+ const RotateCapturedFrame rotation) = 0;
+
+ // This function sets the expected delay from when a video frame is captured
+ // to when that frame is delivered to VideoEngine.
+ virtual int SetCaptureDelay(const int capture_id,
+ const unsigned int capture_delay_ms) = 0;
+
+ // Returns the number of sets of capture capabilities the capture device
+ // supports.
+ virtual int NumberOfCapabilities(
+ const char* unique_id_utf8,
+ const unsigned int unique_id_utf8_length) = 0;
+
+ // Gets a set of capture capabilities for a specified capture device.
+ virtual int GetCaptureCapability(const char* unique_id_utf8,
+ const unsigned int unique_id_utf8_length,
+ const unsigned int capability_number,
+ CaptureCapability& capability) = 0;
+
+ // Displays the capture device property dialog box for the specified capture
+ // device. Windows only.
+ virtual int ShowCaptureSettingsDialogBox(
+ const char* unique_idUTF8,
+ const unsigned int unique_id_utf8_length,
+ const char* dialog_title,
+ void* parent_window = NULL,
+ const unsigned int x = 200,
+ const unsigned int y = 200) = 0;
+
+ // Gets the clockwise angle the frames from the camera must be rotated in
+ // order to display the frames correctly if the display is rotated in its
+ // natural orientation.
+ virtual int GetOrientation(const char* unique_id_utf8,
+ RotateCapturedFrame& orientation) = 0;
+
+ // Enables brightness alarm detection and the brightness alarm callback.
+ virtual int EnableBrightnessAlarm(const int capture_id,
+ const bool enable) = 0;
+
+ // Registers an instance of a user implementation of the ViECaptureObserver.
+ virtual int RegisterObserver(const int capture_id,
+ ViECaptureObserver& observer) = 0;
+
+ // Removes an already registered instance of ViECaptureObserver.
+ virtual int DeregisterObserver(const int capture_id) = 0;
+
+ protected:
+ ViECapture() {}
+ virtual ~ViECapture() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
diff --git a/trunk/src/video_engine/include/vie_codec.h b/trunk/src/video_engine/include/vie_codec.h
new file mode 100644
index 0000000..0535038
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_codec.h
@@ -0,0 +1,176 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - Setting send and receive codecs.
+// - Codec specific settings.
+// - Key frame signaling.
+// - Stream management settings.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+struct VideoCodec;
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterEncoderObserver()
+// and deregistered using DeregisterEncoderObserver().
+class WEBRTC_DLLEXPORT ViEEncoderObserver {
+ public:
+ // This method is called once per second with the current encoded frame rate
+ // and bit rate.
+ virtual void OutgoingRate(const int video_channel,
+ const unsigned int framerate,
+ const unsigned int bitrate) = 0;
+ protected:
+ virtual ~ViEEncoderObserver() {}
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterDecoderObserver()
+// and deregistered using DeregisterDecoderObserver().
+class WEBRTC_DLLEXPORT ViEDecoderObserver {
+ public:
+ // This method is called when a new incoming stream is detected, normally
+ // triggered by a new incoming SSRC or payload type.
+ virtual void IncomingCodecChanged(const int video_channel,
+ const VideoCodec& video_codec) = 0;
+
+ // This method is called once per second containing the frame rate and bit
+ // rate for the incoming stream
+ virtual void IncomingRate(const int video_channel,
+ const unsigned int framerate,
+ const unsigned int bitrate) = 0;
+
+ // This method is called when the decoder needs a new key frame from encoder
+ // on the sender.
+ virtual void RequestNewKeyFrame(const int video_channel) = 0;
+
+ protected:
+ virtual ~ViEDecoderObserver() {}
+};
+
+class WEBRTC_DLLEXPORT ViECodec {
+ public:
+ // Factory for the ViECodec sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViECodec* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViECodec sub-API and decreases an internal reference
+ // counter.
+ // Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Gets the number of available codecs for the VideoEngine build.
+ virtual int NumberOfCodecs() const = 0;
+
+ // Gets a VideoCodec struct for a codec containing the default configuration
+ // for that codec type.
+ virtual int GetCodec(const unsigned char list_number,
+ VideoCodec& video_codec) const = 0;
+
+ // Sets the send codec to use for a specified channel.
+ virtual int SetSendCodec(const int video_channel,
+ const VideoCodec& video_codec) = 0;
+
+ // Gets the current send codec settings.
+ virtual int GetSendCodec(const int video_channel,
+ VideoCodec& video_codec) const = 0;
+
+ // Prepares VideoEngine to receive a certain codec type and setting for a
+ // specified payload type.
+ virtual int SetReceiveCodec(const int video_channel,
+ const VideoCodec& video_codec) = 0;
+
+ // Gets the current receive codec.
+ virtual int GetReceiveCodec(const int video_channel,
+ VideoCodec& video_codec) const = 0;
+
+ // This function is used to get codec configuration parameters to be
+ // signaled from the encoder to the decoder in the call setup.
+ virtual int GetCodecConfigParameters(
+ const int video_channel,
+ unsigned char config_parameters[kConfigParameterSize],
+ unsigned char& config_parameters_size) const = 0;
+
+ // Enables advanced scaling of the captured video stream if the stream
+ // differs from the send codec settings.
+ virtual int SetImageScaleStatus(const int video_channel,
+ const bool enable) = 0;
+
+ // Gets the number of sent key frames and number of sent delta frames.
+ virtual int GetSendCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const = 0;
+
+ // Gets the number of decoded key frames and number of decoded delta frames.
+ virtual int GetReceiveCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const = 0;
+
+ // Gets the bitrate targeted by the video codec rate control in kbit/s.
+ virtual int GetCodecTargetBitrate(const int video_channel,
+ unsigned int* bitrate) const = 0;
+
+ // Gets the number of packets discarded by the jitter buffer because they
+ // arrived too late.
+ virtual unsigned int GetDiscardedPackets(const int video_channel) const = 0;
+
+ // Enables key frame request callback in ViEDecoderObserver.
+ virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
+ const bool enable) = 0;
+
+ // Enables key frame requests for detected lost packets.
+ virtual int SetSignalKeyPacketLossStatus(
+ const int video_channel,
+ const bool enable,
+ const bool only_key_frames = false) = 0;
+
+ // Registers an instance of a user implementation of the ViEEncoderObserver.
+ virtual int RegisterEncoderObserver(const int video_channel,
+ ViEEncoderObserver& observer) = 0;
+
+ // Removes an already registered instance of ViEEncoderObserver.
+ virtual int DeregisterEncoderObserver(const int video_channel) = 0;
+
+ // Registers an instance of a user implementation of the ViEDecoderObserver.
+ virtual int RegisterDecoderObserver(const int video_channel,
+ ViEDecoderObserver& observer) = 0;
+
+ // Removes an already registered instance of ViEDecoderObserver.
+ virtual int DeregisterDecoderObserver(const int video_channel) = 0;
+
+ // This function forces the next encoded frame to be a key frame. This is
+ // normally used when the remote endpoint only supports out‐band key frame
+ // request.
+ virtual int SendKeyFrame(const int video_channel) = 0;
+
+ // This function makes the decoder wait for a key frame before starting to
+ // decode the incoming video stream.
+ virtual int WaitForFirstKeyFrame(const int video_channel,
+ const bool wait) = 0;
+
+ protected:
+ ViECodec() {}
+ virtual ~ViECodec() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
diff --git a/trunk/src/video_engine/include/vie_encryption.h b/trunk/src/video_engine/include/vie_encryption.h
new file mode 100644
index 0000000..8e7c955
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_encryption.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - External encryption and decryption.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+class VideoEngine;
+
+class WEBRTC_DLLEXPORT ViEEncryption {
+ public:
+ // Factory for the ViEEncryption sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViEEncryption* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViEEncryption sub-API and decreases an internal reference
+ // counter.
+ // Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // This function registers a encryption derived instance and enables
+ // external encryption for the specified channel.
+ virtual int RegisterExternalEncryption(const int video_channel,
+ Encryption& encryption) = 0;
+
+ // This function deregisters a registered encryption derived instance
+ // and disables external encryption.
+ virtual int DeregisterExternalEncryption(const int video_channel) = 0;
+
+ protected:
+ ViEEncryption() {}
+ virtual ~ViEEncryption() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/include/vie_errors.h b/trunk/src/video_engine/include/vie_errors.h
new file mode 100644
index 0000000..63100b5
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_errors.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
+
+enum ViEErrors {
+ // ViEBase.
+ kViENotInitialized = 12000, // Init has not been called successfully.
+ kViEBaseVoEFailure, // SetVoiceEngine. ViE failed to use VE instance. Check VE instance pointer.ConnectAudioChannel failed to set voice channel. Have SetVoiceEngine been called? Is the voice channel correct.
+ kViEBaseChannelCreationFailed, // CreateChannel.
+ kViEBaseInvalidChannelId, // The channel does not exist.
+ kViEAPIDoesNotExist, // Release called on Interface that has not been created.
+ kViEBaseInvalidArgument,
+ kViEBaseAlreadySending, // StartSend called on channel that is already sending.
+ kViEBaseNotSending, // StopSend called on channel that is not sending.
+ kViEBaseAlreadyReceiving, // StartReceive called on channel that is already receiving.
+ kViEBaseObserverAlreadyRegistered, // RegisterObserver- an observer has already been set.
+ kViEBaseObserverNotRegistered, // DeregisterObserver - no observer has been registered.
+ kViEBaseUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViECodec.
+ kViECodecInvalidArgument = 12100, // Wrong input parameter to function.
+ kViECodecObserverAlreadyRegistered, // RegisterEncoderObserver, RegisterDecoderObserver.
+ kViECodecObserverNotRegistered, // DeregisterEncoderObserver, DeregisterDecoderObserver.
+ kViECodecInvalidCodec, // SetSendCodec,SetReceiveCodec- The codec structure is invalid.
+ kViECodecInvalidChannelId, // The channel does not exist.
+ kViECodecInUse, // SetSendCodec- Can't change codec size or type when multiple channels use the same encoder.
+ kViECodecUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViERender.
+ kViERenderInvalidRenderId = 12200, // No renderer with the ID exist. In AddRenderer - The render ID is invalid. No capture device, channel or file is allocated with that id.
+ kViERenderAlreadyExists, // AddRenderer: the renderer already exist.
+ kViERenderInvalidFrameFormat, // AddRender (external renderer). The user has requested a frame format that we don't support.
+ kViERenderUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViECapture.
+ kViECaptureDeviceAlreadyConnected = 12300, // ConnectCaptureDevice - A capture device has already been connected to this video channel.
+ kViECaptureDeviceDoesNotExist, // No capture device exist with the provided capture id or unique name.
+ kViECaptureDeviceInvalidChannelId, // ConnectCaptureDevice, DisconnectCaptureDevice- No Channel exist with the provided channel id.
+ kViECaptureDeviceNotConnected, // DisconnectCaptureDevice- No capture device is connected to the channel.
+ kViECaptureDeviceNotStarted, // Stop- The capture device is not started.
+ kViECaptureDeviceAlreadyStarted, // Start- The capture device is already started.
+ kViECaptureDeviceAlreadyAllocated, // AllocateCaptureDevice The device is already allocated.
+ kViECaptureDeviceMaxNoDevicesAllocated, // AllocateCaptureDevice Max number of devices already allocated.
+ kViECaptureObserverAlreadyRegistered, // RegisterObserver- An observer is already registered. Need to deregister first.
+ kViECaptureDeviceObserverNotRegistered, // DeregisterObserver- No observer is registered.
+ kViECaptureDeviceUnknownError, // An unknown error has occurred. Check the log file.
+ kViECaptureDeviceMacQtkitNotSupported, // QTKit handles the capture devices automatically. Thus querying capture capabilities is not supported.
+
+ // ViEFile.
+ kViEFileInvalidChannelId = 12400, // No Channel exist with the provided channel id.
+ kViEFileInvalidArgument, // Incorrect input argument
+ kViEFileAlreadyRecording, // StartRecordOutgoingVideo - already recording channel
+ kViEFileVoENotSet, // StartRecordOutgoingVideo. Failed to access voice engine. Has SetVoiceEngine been called?
+ kViEFileNotRecording, // StopRecordOutgoingVideo
+ kViEFileMaxNoOfFilesOpened, // StartPlayFile
+ kViEFileNotPlaying, // StopPlayFile. The file with the provided id is not playing.
+ kViEFileObserverAlreadyRegistered, // RegisterObserver
+ kViEFileObserverNotRegistered, // DeregisterObserver
+ kViEFileInputAlreadyConnected, // SendFileOnChannel- the video channel already have a connected input.
+ kViEFileNotConnected, // StopSendFileOnChannel- No file is being sent on the channel.
+ kViEFileVoEFailure, // SendFileOnChannel,StartPlayAudioLocally - failed to play audio stream
+ kViEFileInvalidRenderId, // SetRenderTimeoutImage and SetRenderStartImage: Renderer with the provided render id does not exist.
+ kViEFileInvalidFile, // Can't open the file with provided filename. Is the path and file format correct?
+ kViEFileInvalidCapture, // Can't use ViEPicture. Is the object correct?
+ kViEFileSetRenderTimeoutError, // SetRenderTimeoutImage- Please see log file.
+ kViEFileInvalidCaptureId, // SetCaptureDeviceImage capture id does not exist.
+ kViEFileSetCaptureImageError, // SetCaptureDeviceImage error. Please see log file.
+ kViEFileSetStartImageError, // SetRenderStartImage error. Please see log file.
+ kViEFileUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViENetwork.
+ kViENetworkInvalidChannelId = 12500, // No Channel exist with the provided channel id.
+ kViENetworkAlreadyReceiving, // SetLocalReceiver: Can not change ports while receiving.
+ kViENetworkLocalReceiverNotSet, // GetLocalReceiver: SetLocalReceiver not called.
+ kViENetworkAlreadySending, // SetSendDestination
+ kViENetworkDestinationNotSet, // GetSendDestination
+ kViENetworkInvalidArgument, // GetLocalIP- Check function arguments.
+ kViENetworkSendCodecNotSet, // SetSendGQoS- Need to set the send codec first.
+ kViENetworkServiceTypeNotSupported, // SetSendGQoS
+ kViENetworkNotSupported, // SetSendGQoS Not supported on this OS.
+ kViENetworkObserverAlreadyRegistered, // RegisterObserver
+ kViENetworkObserverNotRegistered, // SetPeriodicDeadOrAliveStatus - Need to call RegisterObserver first, DeregisterObserver if no observer is registered.
+ kViENetworkUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViERTP_RTCP.
+ kViERtpRtcpInvalidChannelId = 12600, // No Channel exist with the provided channel id.
+ kViERtpRtcpAlreadySending, // The channel is already sending. Need to stop send before calling this API.
+ kViERtpRtcpNotSending, // The channel needs to be sending in order for this function to work.
+ kViERtpRtcpRtcpDisabled, // Functions failed because RTCP is disabled.
+ kViERtpRtcpObserverAlreadyRegistered, // An observer is already registered. Need to deregister the old first.
+ kViERtpRtcpObserverNotRegistered, // No observer registered.
+ kViERtpRtcpUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViEEncryption.
+ kViEEncryptionInvalidChannelId = 12700, // Channel id does not exist.
+ kViEEncryptionInvalidSrtpParameter, // EnableSRTPSend, EnableSRTPReceive- Check the SRTP parameters.
+ kViEEncryptionSrtpNotSupported, // This build does not support SRTP.
+ kViEEncryptionUnknownError, // An unknown error has occurred. Check the log file.
+
+ // ViEImageProcess.
+ kViEImageProcessInvalidChannelId = 12800, // No Channel exist with the provided channel id.
+ kViEImageProcessInvalidCaptureId, // No capture device exist with the provided capture id.
+ kViEImageProcessFilterExists, // RegisterCaptureEffectFilter,RegisterSendEffectFilter,RegisterRenderEffectFilter - Effect filter already registered.
+ kViEImageProcessFilterDoesNotExist, // DeRegisterCaptureEffectFilter,DeRegisterSendEffectFilter,DeRegisterRenderEffectFilter - Effect filter not registered.
+ kViEImageProcessAlreadyEnabled, // EnableDeflickering,EnableDenoising,EnableColorEnhancement- Function already enabled.
+ kViEImageProcessAlreadyDisabled, // EnableDeflickering,EnableDenoising,EnableColorEnhancement- Function already disabled.
+ kViEImageProcessUnknownError // An unknown error has occurred. Check the log file.
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
diff --git a/trunk/src/video_engine/include/vie_external_codec.h b/trunk/src/video_engine/include/vie_external_codec.h
new file mode 100644
index 0000000..a2e686a
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_external_codec.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoDecoder;
+class VideoEncoder;
+class VideoEngine;
+
+class WEBRTC_DLLEXPORT ViEExternalCodec {
+ public:
+ static ViEExternalCodec* GetInterface(VideoEngine* video_engine);
+
+ virtual int Release() = 0;
+
+ virtual int RegisterExternalSendCodec(const int video_channel,
+ const unsigned char pl_type,
+ VideoEncoder* encoder) = 0;
+
+ virtual int DeRegisterExternalSendCodec(const int video_channel,
+ const unsigned char pl_type) = 0;
+
+ virtual int RegisterExternalReceiveCodec(const int video_channel,
+ const unsigned int pl_type,
+ VideoDecoder* decoder,
+ bool decoder_render = false,
+ int render_delay = 0) = 0;
+
+ virtual int DeRegisterExternalReceiveCodec(const int video_channel,
+ const unsigned char pl_type) = 0;
+
+ protected:
+ ViEExternalCodec() {}
+ virtual ~ViEExternalCodec() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
diff --git a/trunk/src/video_engine/include/vie_file.h b/trunk/src/video_engine/include/vie_file.h
new file mode 100644
index 0000000..fbc3d2b
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_file.h
@@ -0,0 +1,221 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - File recording and playing.
+// - Snapshots.
+// - Background images.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+struct VideoCodec;
+
+// This structure contains picture data and describes the picture type.
+struct ViEPicture {
+ unsigned char* data;
+ unsigned int size;
+ unsigned int width;
+ unsigned int height;
+ RawVideoType type;
+
+ ViEPicture() {
+ data = NULL;
+ size = 0;
+ width = 0;
+ height = 0;
+ type = kVideoI420;
+ }
+
+ // Call FreePicture to free data.
+ ~ViEPicture() {
+ data = NULL;
+ size = 0;
+ width = 0;
+ height = 0;
+ type = kVideoUnknown;
+ }
+};
+
+// This enumerator tells which audio source to use for media files.
+enum AudioSource {
+ NO_AUDIO,
+ MICROPHONE,
+ PLAYOUT,
+ VOICECALL
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterObserver() and
+// deregistered using DeregisterObserver().
+class WEBRTC_DLLEXPORT ViEFileObserver {
+ public:
+ // This method is called when the end is reached of a played file.
+ virtual void PlayFileEnded(const WebRtc_Word32 file_id) = 0;
+
+ protected:
+ virtual ~ViEFileObserver() {}
+};
+
+class WEBRTC_DLLEXPORT ViEFile {
+ public:
+ // Factory for the ViEFile sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViEFile* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViEFile sub-API and decreases an internal reference counter.
+ // Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Starts playing a video file.
+ virtual int StartPlayFile(
+ const char* file_name_utf8,
+ int& file_id,
+ const bool loop = false,
+ const FileFormats file_format = kFileFormatAviFile) = 0;
+
+ // Stops a file from being played.
+ virtual int StopPlayFile(const int file_id) = 0;
+
+ // Registers an instance of a user implementation of the ViEFileObserver.
+ virtual int RegisterObserver(int file_id, ViEFileObserver& observer) = 0;
+
+ // Removes an already registered instance of ViEFileObserver.
+ virtual int DeregisterObserver(int file_id, ViEFileObserver& observer) = 0;
+
+ // This function tells which channel, if any, the file should be sent on.
+ virtual int SendFileOnChannel(const int file_id, const int video_channel) = 0;
+
+ // Stops a file from being sent on a a channel.
+ virtual int StopSendFileOnChannel(const int video_channel) = 0;
+
+ // Starts playing the file audio as microphone input for the specified voice
+ // channel.
+ virtual int StartPlayFileAsMicrophone(const int file_id,
+ const int audio_channel,
+ bool mix_microphone = false,
+ float volume_scaling = 1) = 0;
+
+ // The function stop the audio from being played on a VoiceEngine channel.
+ virtual int StopPlayFileAsMicrophone(const int file_id,
+ const int audio_channel) = 0;
+
+ // The function plays and mixes the file audio with the local speaker signal
+ // for playout.
+ virtual int StartPlayAudioLocally(const int file_id, const int audio_channel,
+ float volume_scaling = 1) = 0;
+
+ // Stops the audio from a file from being played locally.
+ virtual int StopPlayAudioLocally(const int file_id,
+ const int audio_channel) = 0;
+
+ // This function starts recording the video transmitted to another endpoint.
+ virtual int StartRecordOutgoingVideo(
+ const int video_channel,
+ const char* file_name_utf8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format = kFileFormatAviFile) = 0;
+
+ // This function starts recording the incoming video stream on a channel.
+ virtual int StartRecordIncomingVideo(
+ const int video_channel,
+ const char* file_name_utf8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format = kFileFormatAviFile) = 0;
+
+ // Stops the file recording of the outgoing stream.
+ virtual int StopRecordOutgoingVideo(const int video_channel) = 0;
+
+ // Stops the file recording of the incoming stream.
+ virtual int StopRecordIncomingVideo(const int video_channel) = 0;
+
+ // Gets the audio codec, video codec and file format of a recorded file.
+ virtual int GetFileInformation(
+ const char* file_name,
+ VideoCodec& video_codec,
+ CodecInst& audio_codec,
+ const FileFormats file_format = kFileFormatAviFile) = 0;
+
+ // The function takes a snapshot of the last rendered image for a video
+ // channel.
+ virtual int GetRenderSnapshot(const int video_channel,
+ const char* file_name_utf8) = 0;
+
+ // The function takes a snapshot of the last rendered image for a video
+ // channel
+ virtual int GetRenderSnapshot(const int video_channel,
+ ViEPicture& picture) = 0;
+
+ // The function takes a snapshot of the last captured image by a specified
+ // capture device.
+ virtual int GetCaptureDeviceSnapshot(const int capture_id,
+ const char* file_name_utf8) = 0;
+
+ // The function takes a snapshot of the last captured image by a specified
+ // capture device.
+ virtual int GetCaptureDeviceSnapshot(const int capture_id,
+ ViEPicture& picture) = 0;
+
+ // This function sets a jpg image to show before the first frame is captured
+ // by the capture device. This frame will be encoded and transmitted to a
+ // possible receiver
+ virtual int SetCaptureDeviceImage(const int capture_id,
+ const char* file_name_utf8) = 0;
+
+ // This function sets an image to show before the first frame is captured by
+ // the capture device. This frame will be encoded and transmitted to a
+ // possible receiver
+ virtual int SetCaptureDeviceImage(const int capture_id,
+ const ViEPicture& picture) = 0;
+
+ virtual int FreePicture(ViEPicture& picture) = 0;
+
+ // This function sets a jpg image to render before the first received video
+ // frame is decoded for a specified channel.
+ virtual int SetRenderStartImage(const int video_channel,
+ const char* file_name_utf8) = 0;
+
+ // This function sets an image to render before the first received video
+ // frame is decoded for a specified channel.
+ virtual int SetRenderStartImage(const int video_channel,
+ const ViEPicture& picture) = 0;
+
+ // This function sets a jpg image to render if no frame is decoded for a
+ // specified time interval.
+ virtual int SetRenderTimeoutImage(const int video_channel,
+ const char* file_name_utf8,
+ const unsigned int timeout_ms = 1000) = 0;
+
+ // This function sets an image to render if no frame is decoded for a
+ // specified time interval.
+ virtual int SetRenderTimeoutImage(const int video_channel,
+ const ViEPicture& picture,
+ const unsigned int timeout_ms) = 0;
+
+ protected:
+ ViEFile() {}
+ virtual ~ViEFile() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
diff --git a/trunk/src/video_engine/include/vie_image_process.h b/trunk/src/video_engine/include/vie_image_process.h
new file mode 100644
index 0000000..dfad08d
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_image_process.h
@@ -0,0 +1,101 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - Effect filters
+// - Deflickering
+// - Denoising
+// - Color enhancement
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+
+// This class declares an abstract interface for a user defined effect filter.
+// The effect filter is registered using RegisterCaptureEffectFilter(),
+// RegisterSendEffectFilter() or RegisterRenderEffectFilter() and deregistered
+// with the corresponding deregister function.
+class WEBRTC_DLLEXPORT ViEEffectFilter {
+ public:
+ // This method is called with an I420 video frame allowing the user to
+ // modify the video frame.
+ virtual int Transform(int size,
+ unsigned char* frame_buffer,
+ unsigned int time_stamp90KHz,
+ unsigned int width,
+ unsigned int height) = 0;
+ protected:
+ ViEEffectFilter() {}
+ virtual ~ViEEffectFilter() {}
+};
+
+class WEBRTC_DLLEXPORT ViEImageProcess {
+ public:
+ // Factory for the ViEImageProcess sub‐API and increases an internal
+ // reference counter if successful. Returns NULL if the API is not supported
+ // or if construction fails.
+ static ViEImageProcess* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViEImageProcess sub-API and decreases an internal reference
+ // counter. Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // This function registers a EffectFilter to use for a specified capture
+ // device.
+ virtual int RegisterCaptureEffectFilter(const int capture_id,
+ ViEEffectFilter& capture_filter) = 0;
+
+ // This function deregisters a EffectFilter for a specified capture device.
+ virtual int DeregisterCaptureEffectFilter(const int capture_id) = 0;
+
+ // This function registers an EffectFilter to use for a specified channel.
+ virtual int RegisterSendEffectFilter(const int video_channel,
+ ViEEffectFilter& send_filter) = 0;
+
+ // This function deregisters a send effect filter for a specified channel.
+ virtual int DeregisterSendEffectFilter(const int video_channel) = 0;
+
+ // This function registers a EffectFilter to use for the rendered video
+ // stream on an incoming channel.
+ virtual int RegisterRenderEffectFilter(const int video_channel,
+ ViEEffectFilter& render_filter) = 0;
+
+ // This function deregisters a render effect filter for a specified channel.
+ virtual int DeregisterRenderEffectFilter(const int video_channel) = 0;
+
+ // All cameras run the risk of getting in almost perfect sync with
+ // florescent lamps, which will result in a very annoying flickering of the
+ // image. Most cameras have some type of filter to protect against this but
+ // not all of them succeed. Enabling this function will remove the flicker.
+ virtual int EnableDeflickering(const int capture_id, const bool enable) = 0;
+
+ // Some cameras produce very noisy captured images, especially in low‐light
+ // conditions. This functionality will reduce the camera noise.
+ virtual int EnableDenoising(const int capture_id, const bool enable) = 0;
+
+ // This function enhances the colors on the decoded video stream, enabled by
+ // default.
+ virtual int EnableColorEnhancement(const int video_channel,
+ const bool enable) = 0;
+
+ protected:
+ ViEImageProcess() {}
+ virtual ~ViEImageProcess() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
diff --git a/trunk/src/video_engine/include/vie_network.h b/trunk/src/video_engine/include/vie_network.h
new file mode 100644
index 0000000..9752008
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_network.h
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
+
+// This sub-API supports the following functionalities:
+// - Configuring send and receive addresses.
+// - External transport support.
+// - Port and address filters.
+// - Windows GQoS functions and ToS functions.
+// - Packet timeout notification.
+// - Dead‐or‐Alive connection observations.
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class Transport;
+class VideoEngine;
+
+// This enumerator describes VideoEngine packet timeout states.
+enum ViEPacketTimeout {
+ NoPacket = 0,
+ PacketReceived = 1
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterObserver() and
+// deregistered using DeregisterObserver().
+class WEBRTC_DLLEXPORT ViENetworkObserver {
+ public:
+ // This method will be called periodically delivering a dead‐or‐alive
+ // decision for a specified channel.
+ virtual void OnPeriodicDeadOrAlive(const int video_channel,
+ const bool alive) = 0;
+
+ // This method is called once if a packet timeout occurred.
+ virtual void PacketTimeout(const int video_channel,
+ const ViEPacketTimeout timeout) = 0;
+ protected:
+ virtual ~ViENetworkObserver() {}
+};
+
+class WEBRTC_DLLEXPORT ViENetwork {
+ public:
+ // Default values.
+ enum { KDefaultSampleTimeSeconds = 2 };
+
+ // Factory for the ViENetwork sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViENetwork* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViENetwork sub-API and decreases an internal reference
+ // counter.Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Specifies the ports to receive RTP packets on. It is also possible to set
+ // port for RTCP and local IP address.
+ virtual int SetLocalReceiver(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port = 0,
+ const char* ip_address = NULL) = 0;
+
+ // Gets the local receiver ports and address for a specified channel.
+ virtual int GetLocalReceiver(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port, char* ip_address) = 0;
+
+ // Specifies the destination port and IP address for a specified channel.
+ virtual int SetSendDestination(const int video_channel,
+ const char* ip_address,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port = 0,
+ const unsigned short source_rtp_port = 0,
+ const unsigned short source_rtcp_port = 0) = 0;
+
+ // Get the destination port and address for a specified channel.
+ virtual int GetSendDestination(const int video_channel,
+ char* ip_address,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ unsigned short& source_rtp_port,
+ unsigned short& source_rtcp_port) = 0;
+
+ // This function registers a user implementation of Transport to use for
+ // sending RTP and RTCP packets on this channel.
+ virtual int RegisterSendTransport(const int video_channel,
+ Transport& transport) = 0;
+
+ // This function deregisters a used Transport for a specified channel.
+ virtual int DeregisterSendTransport(const int video_channel) = 0;
+
+ // When using external transport for a channel, received RTP packets should
+ // be passed to VideoEngine using this function. The input should contain
+ // the RTP header and payload.
+ virtual int ReceivedRTPPacket(const int video_channel,
+ const void* data,
+ const int length) = 0;
+
+ // When using external transport for a channel, received RTCP packets should
+ // be passed to VideoEngine using this function.
+ virtual int ReceivedRTCPPacket(const int video_channel,
+ const void* data,
+ const int length) = 0;
+
+ // Gets the source ports and IP address of the incoming stream for a
+ // specified channel.
+ virtual int GetSourceInfo(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address,
+ unsigned int ip_address_length) = 0;
+
+ // Gets the local IP address, in string format.
+ virtual int GetLocalIP(char ip_address[64], bool ipv6 = false) = 0;
+
+ // Enables IPv6, instead of IPv4, for a specified channel.
+ virtual int EnableIPv6(int video_channel) = 0;
+
+ // The function returns true if IPv6 is enabled, false otherwise.
+ virtual bool IsIPv6Enabled(int video_channel) = 0;
+
+ // Enables a port and IP address filtering for incoming packets on a
+ // specific channel.
+ virtual int SetSourceFilter(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port = 0,
+ const char* ip_address = NULL) = 0;
+
+ // Gets current port and IP address filter for a specified channel.
+ virtual int GetSourceFilter(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address) = 0;
+
+ // This function sets the six‐bit Differentiated Services Code Point (DSCP)
+ // in the IP header of the outgoing stream for a specific channel.
+ // Windows and Linux only.
+ virtual int SetSendToS(const int video_channel,
+ const int DSCP,
+ const bool use_set_sockOpt = false) = 0;
+
+ // Retrieves the six‐bit Differentiated Services Code Point (DSCP) in the IP
+ // header of the outgoing stream for a specific channel.
+ virtual int GetSendToS(const int video_channel,
+ int& DSCP,
+ bool& use_set_sockOpt) = 0;
+
+ // This function sets the Generic Quality of Service (GQoS) service level.
+ // The Windows operating system then maps to a Differentiated Services Code
+ // Point (DSCP) and to an 802.1p setting. Windows only.
+ virtual int SetSendGQoS(const int video_channel, const bool enable,
+ const int service_type,
+ const int overrideDSCP = 0) = 0;
+
+ // This function retrieves the currently set GQoS service level for a
+ // specific channel.
+ virtual int GetSendGQoS(const int video_channel,
+ bool& enabled,
+ int& service_type,
+ int& overrideDSCP) = 0;
+
+ // This function sets the Maximum Transition Unit (MTU) for a channel. The
+ // RTP packet will be packetized based on this MTU to optimize performance
+ // over the network.
+ virtual int SetMTU(int video_channel, unsigned int mtu) = 0;
+
+ // This function enables or disables warning reports if packets have not
+ // been received for a specified time interval.
+ virtual int SetPacketTimeoutNotification(const int video_channel,
+ bool enable,
+ int timeout_seconds) = 0;
+
+ // Registers an instance of a user implementation of the ViENetwork
+ // observer.
+ virtual int RegisterObserver(const int video_channel,
+ ViENetworkObserver& observer) = 0;
+
+ // Removes a registered instance of ViENetworkObserver.
+ virtual int DeregisterObserver(const int video_channel) = 0;
+
+ // This function enables or disables the periodic dead‐or‐alive callback
+ // functionality for a specified channel.
+ virtual int SetPeriodicDeadOrAliveStatus(
+ const int video_channel,
+ const bool enable,
+ const unsigned int sample_time_seconds = KDefaultSampleTimeSeconds) = 0;
+
+ // This function handles sending a raw UDP data packet over an existing RTP
+ // or RTCP socket.
+ virtual int SendUDPPacket(const int video_channel,
+ const void* data,
+ const unsigned int length,
+ int& transmitted_bytes,
+ bool use_rtcp_socket = false) = 0;
+
+ protected:
+ ViENetwork() {}
+ virtual ~ViENetwork() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
diff --git a/trunk/src/video_engine/include/vie_render.h b/trunk/src/video_engine/include/vie_render.h
new file mode 100644
index 0000000..0b8328e
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_render.h
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - Specify render destinations for incoming video streams, capture devices
+// and files.
+// - Configuring render streams.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+class VideoRender;
+
+// This class declares an abstract interface to be used for external renderers.
+// The user implemented derived class is registered using AddRenderer().
+class WEBRTC_DLLEXPORT ExternalRenderer {
+ public:
+ // This method will be called when the stream to be rendered changes in
+ // resolution or number of streams mixed in the image.
+ virtual int FrameSizeChange(unsigned int width,
+ unsigned int height,
+ unsigned int number_of_streams) = 0;
+
+ // This method is called when a new frame should be rendered.
+ virtual int DeliverFrame(unsigned char* buffer,
+ int buffer_size,
+ // RTP timestamp in 90kHz.
+ uint32_t time_stamp,
+ // Wallclock render time in miliseconds
+ int64_t render_time) = 0;
+
+ protected:
+ virtual ~ExternalRenderer() {}
+};
+
+class WEBRTC_DLLEXPORT ViERender {
+ public:
+ // Factory for the ViERender sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViERender* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViERender sub-API and decreases an internal reference
+ // counter. Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Registers render module.
+ virtual int RegisterVideoRenderModule(VideoRender& render_module) = 0;
+
+ // Deregisters render module.
+ virtual int DeRegisterVideoRenderModule(VideoRender& render_module) = 0;
+
+ // Sets the render destination for a given render ID.
+ virtual int AddRenderer(const int render_id,
+ void* window,
+ const unsigned int z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ // Removes the renderer for a stream.
+ virtual int RemoveRenderer(const int render_id) = 0;
+
+ // Starts rendering a render stream.
+ virtual int StartRender(const int render_id) = 0;
+
+ // Stops rendering a render stream.
+ virtual int StopRender(const int render_id) = 0;
+
+ // Configures an already added render stream.
+ virtual int ConfigureRender(int render_id,
+ const unsigned int z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) = 0;
+
+ // This function mirrors the rendered stream left and right or up and down.
+ virtual int MirrorRenderStream(const int render_id,
+ const bool enable,
+ const bool mirror_xaxis,
+ const bool mirror_yaxis) = 0;
+
+ // External render.
+ virtual int AddRenderer(const int render_id,
+ RawVideoType video_input_format,
+ ExternalRenderer* renderer) = 0;
+
+ protected:
+ ViERender() {}
+ virtual ~ViERender() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
diff --git a/trunk/src/video_engine/include/vie_rtp_rtcp.h b/trunk/src/video_engine/include/vie_rtp_rtcp.h
new file mode 100644
index 0000000..1397222
--- /dev/null
+++ b/trunk/src/video_engine/include/vie_rtp_rtcp.h
@@ -0,0 +1,311 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
+// - SSRC handling.
+// - Transmission of RTCP reports.
+// - Obtaining RTCP data from incoming RTCP sender reports.
+// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
+// - Forward Error Correction (FEC).
+// - RTP Keep‐alive for maintaining the NAT mappings associated to RTP flows.
+// - Writing RTP and RTCP packets to binary files for off‐line analysis of the
+// call quality.
+// - Inserting extra RTP packets into active audio stream.
+
+#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
+#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VideoEngine;
+
+// This enumerator sets the RTCP mode.
+enum ViERTCPMode {
+ kRtcpNone = 0,
+ kRtcpCompound_RFC4585 = 1,
+ kRtcpNonCompound_RFC5506 = 2
+};
+
+// This enumerator describes the key frame request mode.
+enum ViEKeyFrameRequestMethod {
+ kViEKeyFrameRequestNone = 0,
+ kViEKeyFrameRequestPliRtcp = 1,
+ kViEKeyFrameRequestFirRtp = 2,
+ kViEKeyFrameRequestFirRtcp = 3
+};
+
+enum StreamType {
+ kViEStreamTypeNormal = 0, // Normal media stream
+ kViEStreamTypeRtx = 1 // Retransmission media stream
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterRTPObserver() and
+// deregistered using DeregisterRTPObserver().
+class WEBRTC_DLLEXPORT ViERTPObserver {
+ public:
+ // This method is called if SSRC of the incoming stream is changed.
+ virtual void IncomingSSRCChanged(const int video_channel,
+ const unsigned int SSRC) = 0;
+
+ // This method is called if a field in CSRC changes or if the number of
+ // CSRCs changes.
+ virtual void IncomingCSRCChanged(const int video_channel,
+ const unsigned int CSRC,
+ const bool added) = 0;
+ protected:
+ virtual ~ViERTPObserver() {}
+};
+
+// This class declares an abstract interface for a user defined observer. It is
+// up to the VideoEngine user to implement a derived class which implements the
+// observer class. The observer is registered using RegisterRTCPObserver() and
+// deregistered using DeregisterRTCPObserver().
+
+class WEBRTC_DLLEXPORT ViERTCPObserver {
+ public:
+ // This method is called if a application-defined RTCP packet has been
+ // received.
+ virtual void OnApplicationDataReceived(
+ const int video_channel,
+ const unsigned char sub_type,
+ const unsigned int name,
+ const char* data,
+ const unsigned short data_length_in_bytes) = 0;
+ protected:
+ virtual ~ViERTCPObserver() {}
+};
+
+class WEBRTC_DLLEXPORT ViERTP_RTCP {
+ public:
+ enum { KDefaultDeltaTransmitTimeSeconds = 15 };
+ enum { KMaxRTCPCNameLength = 256 };
+
+ // Factory for the ViERTP_RTCP sub‐API and increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static ViERTP_RTCP* GetInterface(VideoEngine* video_engine);
+
+ // Releases the ViERTP_RTCP sub-API and decreases an internal reference
+ // counter. Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VideoEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // This function enables you to specify the RTP synchronization source
+ // identifier (SSRC) explicitly.
+ virtual int SetLocalSSRC(const int video_channel,
+ const unsigned int SSRC,
+ const StreamType usage = kViEStreamTypeNormal,
+ const unsigned char simulcast_idx = 0) = 0;
+
+ // This function gets the SSRC for the outgoing RTP stream for the specified
+ // channel.
+ virtual int GetLocalSSRC(const int video_channel,
+ unsigned int& SSRC) const = 0;
+
+ // This function map a incoming SSRC to a StreamType so that the engine
+ // can know which is the normal stream and which is the RTX
+ virtual int SetRemoteSSRCType(const int video_channel,
+ const StreamType usage,
+ const unsigned int SSRC) const = 0;
+
+ // This function gets the SSRC for the incoming RTP stream for the specified
+ // channel.
+ virtual int GetRemoteSSRC(const int video_channel,
+ unsigned int& SSRC) const = 0;
+
+ // This function returns the CSRCs of the incoming RTP packets.
+ virtual int GetRemoteCSRCs(const int video_channel,
+ unsigned int CSRCs[kRtpCsrcSize]) const = 0;
+
+ // This function enables manual initialization of the sequence number. The
+ // start sequence number is normally a random number.
+ virtual int SetStartSequenceNumber(const int video_channel,
+ unsigned short sequence_number) = 0;
+
+ // This function sets the RTCP status for the specified channel.
+ // Default mode is kRtcpCompound_RFC4585.
+ virtual int SetRTCPStatus(const int video_channel,
+ const ViERTCPMode rtcp_mode) = 0;
+
+ // This function gets the RTCP status for the specified channel.
+ virtual int GetRTCPStatus(const int video_channel,
+ ViERTCPMode& rtcp_mode) const = 0;
+
+ // This function sets the RTCP canonical name (CNAME) for the RTCP reports
+ // on a specific channel.
+ virtual int SetRTCPCName(const int video_channel,
+ const char rtcp_cname[KMaxRTCPCNameLength]) = 0;
+
+ // This function gets the RTCP canonical name (CNAME) for the RTCP reports
+ // sent the specified channel.
+ virtual int GetRTCPCName(const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const = 0;
+
+ // This function gets the RTCP canonical name (CNAME) for the RTCP reports
+ // received on the specified channel.
+ virtual int GetRemoteRTCPCName(
+ const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const = 0;
+
+ // This function sends an RTCP APP packet on a specific channel.
+ virtual int SendApplicationDefinedRTCPPacket(
+ const int video_channel,
+ const unsigned char sub_type,
+ unsigned int name,
+ const char* data,
+ unsigned short data_length_in_bytes) = 0;
+
+ // This function enables Negative Acknowledgment (NACK) using RTCP,
+ // implemented based on RFC 4585. NACK retransmits RTP packets if lost on
+ // the network. This creates a lossless transport at the expense of delay.
+ // If using NACK, NACK should be enabled on both endpoints in a call.
+ virtual int SetNACKStatus(const int video_channel, const bool enable) = 0;
+
+ // This function enables Forward Error Correction (FEC) using RTCP,
+ // implemented based on RFC 5109, to improve packet loss robustness. Extra
+ // FEC packets are sent together with the usual media packets, hence
+ // part of the bitrate will be used for FEC packets.
+ virtual int SetFECStatus(const int video_channel,
+ const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) = 0;
+
+ // This function enables hybrid Negative Acknowledgment using RTCP
+ // and Forward Error Correction (FEC) implemented based on RFC 5109,
+ // to improve packet loss robustness. Extra
+ // FEC packets are sent together with the usual media packets, hence will
+ // part of the bitrate be used for FEC packets.
+ // The hybrid mode will choose between nack only, fec only and both based on
+ // network conditions. When both are applied, only packets that were not
+ // recovered by the FEC will be nacked.
+ virtual int SetHybridNACKFECStatus(const int video_channel,
+ const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) = 0;
+
+ // This function enables RTCP key frame requests.
+ virtual int SetKeyFrameRequestMethod(
+ const int video_channel, const ViEKeyFrameRequestMethod method) = 0;
+
+ // This function enables signaling of temporary bitrate constraints using
+ // RTCP, implemented based on RFC4585.
+ virtual int SetTMMBRStatus(const int video_channel, const bool enable) = 0;
+
+ // Enables and disables REMB packets for this channel. |sender| indicates
+ // this channel is encoding, |receiver| tells the bitrate estimate for
+ // this channel should be included in the REMB packet.
+ virtual int SetRembStatus(int video_channel,
+ bool sender,
+ bool receiver) = 0;
+
+ // This function returns our locally created statistics of the received RTP
+ // stream.
+ virtual int GetReceivedRTCPStatistics(
+ const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter,
+ int& rtt_ms) const = 0;
+
+ // This function returns statistics reported by the remote client in a RTCP
+ // packet.
+ virtual int GetSentRTCPStatistics(const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter,
+ int& rtt_ms) const = 0;
+
+ // The function gets statistics from the sent and received RTP streams.
+ virtual int GetRTPStatistics(const int video_channel,
+ unsigned int& bytes_sent,
+ unsigned int& packets_sent,
+ unsigned int& bytes_received,
+ unsigned int& packets_received) const = 0;
+
+ // The function gets bandwidth usage statistics from the sent RTP streams in
+ // bits/s.
+ virtual int GetBandwidthUsage(const int video_channel,
+ unsigned int& total_bitrate_sent,
+ unsigned int& video_bitrate_sent,
+ unsigned int& fec_bitrate_sent,
+ unsigned int& nackBitrateSent) const = 0;
+
+ // This function gets the send-side estimated bandwidth available for video,
+ // including overhead, in bits/s.
+ virtual int GetEstimatedSendBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const = 0;
+
+ // This function gets the receive-side estimated bandwidth available for
+ // video, including overhead, in bits/s.
+ // Returns -1 when no valid estimate is available.
+ virtual int GetEstimatedReceiveBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const = 0;
+
+ // This function enables or disables an RTP keep-alive mechanism which can
+ // be used to maintain an existing Network Address Translator (NAT) mapping
+ // while regular RTP is no longer transmitted.
+ virtual int SetRTPKeepAliveStatus(
+ const int video_channel,
+ bool enable,
+ const char unknown_payload_type,
+ const unsigned int delta_transmit_time_seconds =
+ KDefaultDeltaTransmitTimeSeconds) = 0;
+
+ // This function gets the RTP keep-alive status.
+ virtual int GetRTPKeepAliveStatus(
+ const int video_channel,
+ bool& enabled,
+ char& unkown_payload_type,
+ unsigned int& delta_transmit_time_seconds) const = 0;
+
+ // This function enables capturing of RTP packets to a binary file on a
+ // specific channel and for a given direction. The file can later be
+ // replayed using e.g. RTP Tools rtpplay since the binary file format is
+ // compatible with the rtpdump format.
+ virtual int StartRTPDump(const int video_channel,
+ const char file_nameUTF8[1024],
+ RTPDirections direction) = 0;
+
+ // This function disables capturing of RTP packets to a binary file on a
+ // specific channel and for a given direction.
+ virtual int StopRTPDump(const int video_channel,
+ RTPDirections direction) = 0;
+
+ // Registers an instance of a user implementation of the ViERTPObserver.
+ virtual int RegisterRTPObserver(const int video_channel,
+ ViERTPObserver& observer) = 0;
+
+ // Removes a registered instance of ViERTPObserver.
+ virtual int DeregisterRTPObserver(const int video_channel) = 0;
+
+ // Registers an instance of a user implementation of the ViERTCPObserver.
+ virtual int RegisterRTCPObserver(const int video_channel,
+ ViERTCPObserver& observer) = 0;
+
+ // Removes a registered instance of ViERTCPObserver.
+ virtual int DeregisterRTCPObserver(const int video_channel) = 0;
+
+ protected:
+ ViERTP_RTCP() {}
+ virtual ~ViERTP_RTCP() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h b/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h
new file mode 100644
index 0000000..8382844
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * GUI_Defines.h
+ *
+ */
+
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
+
+#define ViE_TEST(x) if(-1 == x){ \
+int errNum = _ptrViEBase->LastError(); \
+NSLog(@"ERROR: %d at %s:%d", errNum, __FUNCTION__, __LINE__); \
+}
+
+
+// Video Engine Related
+#define V_CAPTURE_DEVICE_INDEX 0
+#define V_VIE_CAPTURE_ID 747
+#define V_DEVICE_NAME_LENGTH 256
+#define V_CODEC_INDEX 2
+#define V_IP_ADDRESS "127.0.0.1"
+#define V_RTP_PORT 8000
+
+
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist
new file mode 100644
index 0000000..d0d3a18
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+ <key>CFBundleDevelopmentRegion</key>
+ <string>English</string>
+ <key>CFBundleExecutable</key>
+ <string>${EXECUTABLE_NAME}</string>
+ <key>CFBundleIconFile</key>
+ <string></string>
+ <key>CFBundleIdentifier</key>
+ <string>com.yourcompany.${PRODUCT_NAME:rfc1034identifier}</string>
+ <key>CFBundleInfoDictionaryVersion</key>
+ <string>6.0</string>
+ <key>CFBundleName</key>
+ <string>${PRODUCT_NAME}</string>
+ <key>CFBundlePackageType</key>
+ <string>APPL</string>
+ <key>CFBundleShortVersionString</key>
+ <string>1.0</string>
+ <key>CFBundleSignature</key>
+ <string>????</string>
+ <key>CFBundleVersion</key>
+ <string>1</string>
+ <key>LSMinimumSystemVersion</key>
+ <string>${MACOSX_DEPLOYMENT_TARGET}</string>
+ <key>NSMainNibFile</key>
+ <string>SimpleCocoaGUI</string>
+ <key>NSPrincipalClass</key>
+ <string>NSApplication</string>
+</dict>
+</plist>
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h
new file mode 100644
index 0000000..10d52fc
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// SimpleCocoaGUIAppDelegate.h
+//
+
+#import <Cocoa/Cocoa.h>
+#include <iostream>
+using namespace std;
+
+@class ViECocoaRenderView;
+
+#include "GUI_Defines.h"
+
+#include "common_types.h"
+#include "voe_base.h"
+
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_file.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+#include "vie_errors.h"
+
+
+
+@interface SimpleCocoaGUIAppDelegate : NSObject <NSApplicationDelegate> {
+ NSWindow* _window;
+ IBOutlet NSOpenGLView* _vieCocoaRenderView1;
+ IBOutlet NSOpenGLView* _vieCocoaRenderView2;
+ IBOutlet NSButton* _butRestartLoopback;
+ VideoEngine* _ptrViE;
+ ViEBase* _ptrViEBase;
+ ViECapture* _ptrViECapture;
+ ViERender* _ptrViERender;
+ ViECodec* _ptrViECodec;
+ ViENetwork* _ptrViENetwork;
+
+ bool _fullScreen;
+ int _videoChannel;
+
+ int _captureId;
+
+ VideoEngine* ptrViE;
+ ViEBase* ptrViEBase;
+ ViECapture* ptrViECapture;
+ ViERTP_RTCP* ptrViERtpRtcp;
+ ViERender* ptrViERender;
+ ViECodec* ptrViECodec;
+ ViENetwork* ptrViENetwork;
+}
+
+@property (assign) IBOutlet NSWindow* window;
+-(void)createUI:(bool)fullScreen;
+-(void)initViECocoaTest;
+-(void)initializeVariables;
+-(void)NSLogVideoCodecs;
+-(void)startViECocoaTest;
+-(int)initLoopback;
+-(int)ioLooback;
+-(int)startLoopback;
+-(int)stopLooback;
+
+-(IBAction)handleRestart:(id)sender;
+
+
+@end
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm
new file mode 100644
index 0000000..d594cfe
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm
@@ -0,0 +1,1075 @@
+//
+// SimpleCocoaGUIAppDelegate.m
+//
+
+#import "SimpleCocoaGUIAppDelegate.h"
+
+@implementation SimpleCocoaGUIAppDelegate
+
+@synthesize window = _window;
+
+- (void)applicationDidFinishLaunching:(NSNotification *)aNotification {
+
+// [self initializeVariables];
+ [self createUI];
+// [self initViECocoaTest];
+// [self NSLogVideoCodecs];
+// [self startViECocoaTest];
+
+// [self startLoopback];
+
+ [self ioLooback];
+}
+
+-(void)createUI{
+
+ NSRect outWindow1Frame = NSMakeRect(200, 200, 200, 200);
+ NSWindow* outWindow1 = [[NSWindow alloc] initWithContentRect:outWindow1Frame styleMask:NSTitledWindowMask backing:NSBackingStoreBuffered defer:NO];
+ [outWindow1 orderOut:nil];
+ NSRect vieAutotestCocoaRenderView1Frame = NSMakeRect(0, 0, 200, 200);
+ _vieCocoaRenderView1 = [[ViECocoaRenderView alloc] initWithFrame:vieAutotestCocoaRenderView1Frame];
+ [[outWindow1 contentView] addSubview:_vieCocoaRenderView1];
+ [outWindow1 setTitle:[NSString stringWithFormat:@"window1"]];
+ [outWindow1 makeKeyAndOrderFront:NSApp];
+
+
+ NSRect outWindow2Frame = NSMakeRect(400, 200, 200, 200);
+ NSWindow* outWindow2 = [[NSWindow alloc] initWithContentRect:outWindow2Frame styleMask:NSTitledWindowMask backing:NSBackingStoreBuffered defer:NO];
+ [outWindow2 orderOut:nil];
+ NSRect vieAutotestCocoaRenderView2Frame = NSMakeRect(0, 0, 200, 200);
+ _vieCocoaRenderView2 = [[ViECocoaRenderView alloc] initWithFrame:vieAutotestCocoaRenderView2Frame];
+ [[outWindow2 contentView] addSubview:_vieCocoaRenderView2];
+ [outWindow2 setTitle:[NSString stringWithFormat:@"window2"]];
+ [outWindow2 makeKeyAndOrderFront:NSApp];
+
+
+
+
+
+
+
+}
+
+-(void)initViECocoaTest{
+
+ int _error = 0;
+ _ptrViE = VideoEngine::Create();
+ _ptrViEBase = ViEBase::GetInterface(_ptrViE);
+ _error = _ptrViEBase->Init();
+
+ _ptrViECapture = ViECapture::GetInterface(_ptrViE);
+ _ptrViERender = ViERender::GetInterface(_ptrViE);
+ _ptrViECodec = ViECodec::GetInterface(_ptrViE);
+ _ptrViENetwork = ViENetwork::GetInterface(_ptrViE);
+
+
+ _error = _ptrViE->SetTraceFile("ViEBaseStandardTest.txt");
+ _error = _ptrViE->SetEncryptedTraceFile("ViEBaseStandardTestEncrypted.txt");
+
+
+}
+
+
+-(void)initializeVariables{
+ _fullScreen = YES;
+}
+
+-(void)NSLogVideoCodecs{
+ NSLog(@"Searching for video codecs.....");
+
+ VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(VideoCodec));
+ for(int index = 0; index < _ptrViECodec->NumberOfCodecs(); index++)
+ {
+ ViE_TEST(_ptrViECodec->GetCodec(index, videoCodec));
+ NSLog(@"Video codec found: %s", videoCodec.plName);
+ }
+
+}
+-(void)startViECocoaTest{
+
+
+
+
+ int error=0;
+
+ char deviceName[128];
+ char deviceUniqueName[512];
+ int captureId = 0;
+ int dummy = 0;
+
+ //ViE_TEST(_ptrViEBase->CreateChannel(_videoChannel));
+ //ViE_TEST(_ptrViECapture->GetCaptureDevice(0,deviceName,sizeof(deviceName),deviceUniqueName,sizeof(deviceUniqueName)));
+ //ViE_TEST(_ptrViECapture->AllocateCaptureDevice(deviceUniqueName,sizeof(deviceUniqueName),captureId));
+ //ViE_TEST(_ptrViECapture->AllocateCaptureDevice("dummydevicethatdoesnotexist",sizeof(deviceUniqueName),dummy));
+
+ char captureDeviceName[V_DEVICE_NAME_LENGTH] = "";
+ char captureDeviceUniqueId[V_DEVICE_NAME_LENGTH] = "";
+ int captureDeviceId = 0;
+
+
+
+ ViE_TEST(_ptrViE->SetTraceFilter(webrtc::TR_ALL));
+ ViE_TEST(_ptrViE->SetTraceFile("ViECocoaTrace.txt"));
+ ViE_TEST(_ptrViE->SetEncryptedTraceFile("ViECocoaEncryptedTrace.txt"));
+
+
+
+
+ // base
+ ViE_TEST(_ptrViEBase->CreateChannel(_videoChannel));
+
+ // capture device
+ ViE_TEST(_ptrViECapture->GetCaptureDevice(V_CAPTURE_DEVICE_INDEX, captureDeviceName, V_DEVICE_NAME_LENGTH, captureDeviceUniqueId, V_DEVICE_NAME_LENGTH));
+ ViE_TEST(_ptrViECapture->AllocateCaptureDevice(captureDeviceUniqueId, V_DEVICE_NAME_LENGTH, captureDeviceId));
+ ViE_TEST(_ptrViECapture->ConnectCaptureDevice(captureDeviceId, _videoChannel));
+ ViE_TEST(_ptrViECapture->StartCapture(captureDeviceId));
+
+ // renderer
+ ViE_TEST(_ptrViERender->AddRenderer(captureDeviceId, (void*)_vieCocoaRenderView1, 0, 0.0, 0.0, 1.0, 1.0));
+ ViE_TEST(_ptrViERender->StartRender(captureDeviceId));
+// usleep(3 * 1000);
+// ViE_TEST(_ptrViERender->RemoveRenderer(captureDeviceId));
+ //exit(0);
+
+
+// // codec
+// [self NSLogVideoCodecs];
+// VideoCodec videoCodec;
+// memset(&videoCodec, 0, sizeof(VideoCodec));
+// ViE_TEST(_ptrViECodec->GetCodec(V_CODEC_INDEX, videoCodec));
+// ViE_TEST(_ptrViECodec->SetReceiveCodec(_videoChannel, videoCodec));
+// ViE_TEST(_ptrViECodec->SetSendCodec(_videoChannel, videoCodec));
+//
+// // network + base
+// ViE_TEST(_ptrViENetwork->SetLocalReceiver(_videoChannel, V_RTP_PORT));
+// ViE_TEST(_ptrViEBase->StartReceive(_videoChannel));
+// ViE_TEST(_ptrViENetwork->SetSendDestination(_videoChannel, V_IP_ADDRESS, V_RTP_PORT));
+// ViE_TEST(_ptrViEBase->StartSend(_videoChannel));
+// ViE_TEST(_ptrViERender->MirrorRenderStream(captureDeviceId, true, false, true));
+
+
+}
+
+-(int)initLoopback
+{
+
+}
+-(int)startLoopback
+{
+ //********************************************************
+ // Begin create/initialize Video Engine for testing
+ //********************************************************
+
+ int error = 0;
+ bool succeeded = true;
+ int numberOfErrors = 0;
+ std::string str;
+
+ //
+ // Create a VideoEngine instance
+ //
+// VideoEngine* ptrViE = NULL;
+ ptrViE = VideoEngine::Create();
+ if (ptrViE == NULL)
+ {
+ printf("ERROR in VideoEngine::Create\n");
+ return -1;
+ }
+
+ error = ptrViE->SetTraceFilter(webrtc::TR_ALL);
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceLevel\n");
+ return -1;
+ }
+
+
+ error = ptrViE->SetTraceFile("ViETrace.txt");
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceFile\n");
+ return -1;
+ }
+
+ error = ptrViE->SetEncryptedTraceFile("ViEEncryptedTrace.txt");
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetEncryptedTraceFile\n");
+ return -1;
+ }
+
+ //
+ // Init VideoEngine and create a channel
+ //
+ ptrViEBase = ViEBase::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViEBase::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViEBase->Init();
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::Init\n");
+ return -1;
+ }
+
+ int videoChannel = -1;
+ error = ptrViEBase->CreateChannel(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::CreateChannel\n");
+ return -1;
+ }
+
+ //
+ // List available capture devices, allocate and connect.
+ //
+ ptrViECapture = ViECapture::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViECapture::GetInterface\n");
+ return -1;
+ }
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ char deviceName[KMaxDeviceNameLength];
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ char uniqueId[KMaxUniqueIdLength];
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ std::cout << std::endl;
+ std::cout << "Available capture devices:" << std::endl;
+ unsigned int captureIdx = 0;
+ for (captureIdx = 0;
+ captureIdx < ptrViECapture->NumberOfCaptureDevices();
+ captureIdx++)
+ {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ error = ptrViECapture->GetCaptureDevice(captureIdx,
+ deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+ std::cout << " " << captureIdx+1 << ". " << deviceName
+ << std::endl;
+ }
+ std::cout << std::endl;
+ std::cout << "Choose capture devices: ";
+// std::getline(std::cin, str);
+// captureIdx = atoi(str.c_str()) - 1;
+ captureIdx = 0;
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+
+ _captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId,
+ KMaxUniqueIdLength, _captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::AllocateCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ConnectCaptureDevice(_captureId,
+ _videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ConnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StartCapture(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StartCapture\n");
+ return -1;
+ }
+
+ //
+ // RTP/RTCP settings
+ //
+ ptrViERtpRtcp = ViERTP_RTCP::GetInterface(ptrViE);
+ if (ptrViERtpRtcp == NULL)
+ {
+ printf("ERROR in ViERTP_RTCP::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetRTCPStatus(_videoChannel,
+ kRtcpCompound_RFC4585);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetKeyFrameRequestMethod(_videoChannel,
+ kViEKeyFrameRequestPliRtcp);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetTMMBRStatus(_videoChannel, true);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
+ return -1;
+ }
+
+ //
+ // Set up rendering
+ //
+ ptrViERender = ViERender::GetInterface(ptrViE);
+ if (ptrViERender == NULL)
+ {
+ printf("ERROR in ViERender::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViERender->AddRenderer(_captureId, _vieCocoaRenderView1,
+ 0, 0.0, 0.0, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->AddRenderer(_videoChannel, _vieCocoaRenderView2,
+ 1, 0.0, 0.0, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ //
+ // Setup codecs
+ //
+ ptrViECodec = ViECodec::GetInterface(ptrViE);
+ if (ptrViECodec == NULL)
+ {
+ printf("ERROR in ViECodec::GetInterface\n");
+ return -1;
+ }
+
+ std::cout << std::endl;
+ std::cout << "Available codecs:" << std::endl;
+
+ // Check available codecs and prepare receive codecs
+ VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(VideoCodec));
+ unsigned int codecIdx = 0;
+ for (codecIdx = 0;
+ codecIdx < ptrViECodec->NumberOfCodecs();
+ codecIdx++)
+ {
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ error = ptrViECodec->SetReceiveCodec(_videoChannel,
+ videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetReceiveCodec\n");
+ return -1;
+ }
+ if (videoCodec.codecType != kVideoCodecRED &&
+ videoCodec.codecType != kVideoCodecULPFEC)
+ {
+ std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
+ << std::endl;
+ }
+ }
+// std::cout << std::endl;
+// std::cout << "Choose codec: ";
+// std::getline(std::cin, str);
+// codecIdx = atoi(str.c_str()) - 1;
+ codecIdx = 0;
+
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ error = ptrViECodec->SetSendCodec(_videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetSendCodec\n");
+ return -1;
+ }
+
+ //
+ // Address settings
+ //
+ ptrViENetwork = ViENetwork::GetInterface(ptrViE);
+ if (ptrViENetwork == NULL)
+ {
+ printf("ERROR in ViENetwork::GetInterface\n");
+ return -1;
+ }
+
+ const char* ipAddress = "127.0.0.1";
+ const unsigned short rtpPort = 6000;
+ error = ptrViENetwork->SetLocalReceiver(_videoChannel, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetLocalReceiver\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartReceive(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartReceive\n");
+ return -1;
+ }
+
+ error = ptrViENetwork->SetSendDestination(_videoChannel,
+ ipAddress, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetSendDestination\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartSend(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartSend\n");
+ return -1;
+ }
+
+
+ //********************************************************
+ // Engine started
+ //********************************************************
+
+
+ // Call started
+ std::cout << std::endl;
+ std::cout << "Loopback call started" << std::endl;
+// std::cout << std::endl << std::endl;
+// std::cout << "Press enter to stop...";
+// std::getline(std::cin, str);
+}
+
+-(int)stopLooback
+{
+ int error = 0;
+
+
+
+ //********************************************************
+ // Testing finished. Tear down Video Engine
+ //********************************************************
+
+ error = ptrViEBase->StopReceive(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StopSend(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopSend\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StopCapture(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StopCapture\n");
+ return -1;
+ }
+
+ error = ptrViECapture->DisconnectCaptureDevice(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ReleaseCaptureDevice(_captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViEBase->DeleteChannel(_videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::DeleteChannel\n");
+ return -1;
+ }
+
+ int remainingInterfaces = 0;
+ remainingInterfaces = ptrViECodec->Release();
+ remainingInterfaces += ptrViECapture->Release();
+ remainingInterfaces += ptrViERtpRtcp->Release();
+ remainingInterfaces += ptrViERender->Release();
+ remainingInterfaces += ptrViENetwork->Release();
+ remainingInterfaces += ptrViEBase->Release();
+ if (remainingInterfaces > 0)
+ {
+ printf("ERROR: Could not release all interfaces\n");
+ return -1;
+ }
+
+ bool deleted = VideoEngine::Delete(ptrViE);
+ if (deleted == false)
+ {
+ printf("ERROR in VideoEngine::Delete\n");
+ return -1;
+ }
+
+ return 0;
+
+ // ===================================================================
+ //
+ // END: VideoEngine 3.0 Sample Code
+ //
+ // ===================================================================
+
+
+}
+
+-(int)ioLooback
+{
+ //********************************************************
+ // Begin create/initialize Video Engine for testing
+ //********************************************************
+
+ int error = 0;
+ bool succeeded = true;
+ int numberOfErrors = 0;
+ std::string str;
+
+ //
+ // Create a VideoEngine instance
+ //
+ VideoEngine* ptrViE = NULL;
+ ptrViE = VideoEngine::Create();
+ if (ptrViE == NULL)
+ {
+ printf("ERROR in VideoEngine::Create\n");
+ return -1;
+ }
+
+ error = ptrViE->SetTraceFilter(webrtc::TR_ALL);
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceLevel\n");
+ return -1;
+ }
+
+
+ error = ptrViE->SetTraceFile("ViETrace.txt");
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceFile\n");
+ return -1;
+ }
+
+ error = ptrViE->SetEncryptedTraceFile("ViEEncryptedTrace.txt");
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetEncryptedTraceFile\n");
+ return -1;
+ }
+
+ //
+ // Init VideoEngine and create a channel
+ //
+ ViEBase* ptrViEBase = ViEBase::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViEBase::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViEBase->Init();
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::Init\n");
+ return -1;
+ }
+
+ int videoChannel = -1;
+ error = ptrViEBase->CreateChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::CreateChannel\n");
+ return -1;
+ }
+
+ //
+ // List available capture devices, allocate and connect.
+ //
+ ViECapture* ptrViECapture =
+ ViECapture::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViECapture::GetInterface\n");
+ return -1;
+ }
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ char deviceName[KMaxDeviceNameLength];
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ char uniqueId[KMaxUniqueIdLength];
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ std::cout << std::endl;
+ std::cout << "Available capture devices:" << std::endl;
+ unsigned int captureIdx = 0;
+ for (captureIdx = 0;
+ captureIdx < ptrViECapture->NumberOfCaptureDevices();
+ captureIdx++)
+ {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ error = ptrViECapture->GetCaptureDevice(captureIdx,
+ deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+ std::cout << " " << captureIdx+1 << ". " << deviceName
+ << std::endl;
+ }
+ std::cout << std::endl;
+ std::cout << "Choose capture devices: ";
+// std::getline(std::cin, str);
+// captureIdx = atoi(str.c_str()) - 1;
+ captureIdx = 0;
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+
+ int captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId,
+ KMaxUniqueIdLength, captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::AllocateCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ConnectCaptureDevice(captureId,
+ videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ConnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StartCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StartCapture\n");
+ return -1;
+ }
+
+ //
+ // RTP/RTCP settings
+ //
+ ViERTP_RTCP* ptrViERtpRtcp =
+ ViERTP_RTCP::GetInterface(ptrViE);
+ if (ptrViERtpRtcp == NULL)
+ {
+ printf("ERROR in ViERTP_RTCP::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
+ kRtcpCompound_RFC4585);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetKeyFrameRequestMethod(videoChannel,
+ kViEKeyFrameRequestPliRtcp);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
+ return -1;
+ }
+
+ //
+ // Set up rendering
+ //
+ ViERender* ptrViERender =
+ ViERender::GetInterface(ptrViE);
+ if (ptrViERender == NULL)
+ {
+ printf("ERROR in ViERender::GetInterface\n");
+ return -1;
+ }
+
+// error = ptrViERender->EnableFullScreenRender(_vieCocoaRenderView1);
+// if (error == -1)
+// {
+// printf("ERROR in ViERender::AddRenderer\n");
+// return -1;
+// }
+
+
+ error = ptrViERender->AddRenderer(captureId, _vieCocoaRenderView1,
+ 0, 0.5, 0.5, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->AddRenderer(videoChannel, _vieCocoaRenderView2,
+ 1, 0.0, 0.0, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ //
+ // Setup codecs
+ //
+ ViECodec* ptrViECodec = ViECodec::GetInterface(ptrViE);
+ if (ptrViECodec == NULL)
+ {
+ printf("ERROR in ViECodec::GetInterface\n");
+ return -1;
+ }
+
+ std::cout << std::endl;
+ std::cout << "Available codecs:" << std::endl;
+
+ // Check available codecs and prepare receive codecs
+ VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(VideoCodec));
+ unsigned int codecIdx = 0;
+ for (codecIdx = 0;
+ codecIdx < ptrViECodec->NumberOfCodecs();
+ codecIdx++)
+ {
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ error = ptrViECodec->SetReceiveCodec(videoChannel,
+ videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetReceiveCodec\n");
+ return -1;
+ }
+ if (videoCodec.codecType != kVideoCodecRED &&
+ videoCodec.codecType != kVideoCodecULPFEC)
+ {
+ std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
+ << std::endl;
+ }
+ }
+ std::cout << std::endl;
+ std::cout << "Choose codec: ";
+// std::getline(std::cin, str);
+// codecIdx = atoi(str.c_str()) - 1;
+
+
+ error = ptrViECapture->ShowCaptureSettingsDialogBox("unique",10, "mytitle");
+ codecIdx = 0;
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetSendCodec\n");
+ return -1;
+ }
+
+ //
+ // Address settings
+ //
+ ViENetwork* ptrViENetwork =
+ ViENetwork::GetInterface(ptrViE);
+ if (ptrViENetwork == NULL)
+ {
+ printf("ERROR in ViENetwork::GetInterface\n");
+ return -1;
+ }
+
+ const char* ipAddress = "127.0.0.1";
+ const unsigned short rtpPort = 6000;
+ error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetLocalReceiver\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartReceive\n");
+ return -1;
+ }
+
+ error = ptrViENetwork->SetSendDestination(videoChannel,
+ ipAddress, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetSendDestination\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartSend\n");
+ return -1;
+ }
+
+
+ //********************************************************
+ // Engine started
+ //********************************************************
+
+
+ // Call started
+ std::cout << std::endl;
+ std::cout << "Loopback call started" << std::endl;
+ std::cout << std::endl << std::endl;
+ std::cout << "Press enter to stop...";
+// [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1]];
+// std::getline(std::cin, str);
+ usleep(5 * 1000 * 1000);
+
+ //int i = 0;
+// while(1)
+// {
+// NSLog(@"app iteration %d", i);
+// i++;
+// [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1]];
+// std::getline(std::cin, str);
+// if(i > 3)
+// {
+// break;
+// }
+// }
+
+ //********************************************************
+ // Testing finished. Tear down Video Engine
+ //********************************************************
+
+ error = ptrViEBase->StopReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StopSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopSend\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StopCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StopCapture\n");
+ return -1;
+ }
+
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ReleaseCaptureDevice(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViEBase->DeleteChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::DeleteChannel\n");
+ return -1;
+ }
+
+ int remainingInterfaces = 0;
+ remainingInterfaces = ptrViECodec->Release();
+ remainingInterfaces += ptrViECapture->Release();
+ remainingInterfaces += ptrViERtpRtcp->Release();
+ remainingInterfaces += ptrViERender->Release();
+ remainingInterfaces += ptrViENetwork->Release();
+ remainingInterfaces += ptrViEBase->Release();
+ if (remainingInterfaces > 0)
+ {
+ printf("ERROR: Could not release all interfaces\n");
+ return -1;
+ }
+
+ bool deleted = VideoEngine::Delete(ptrViE);
+ if (deleted == false)
+ {
+ printf("ERROR in VideoEngine::Delete\n");
+ return -1;
+ }
+
+ NSLog(@"Finished function");
+ return 0;
+
+ //
+ // END: VideoEngine 3.0 Sample Code
+ //
+ // ===================================================================
+}
+
+
+
+
+-(IBAction)handleRestart:(id)sender
+{
+// [self stopLooback];
+// [self startLoopback];
+ [self ioLooback];
+}
+@end
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch
new file mode 100644
index 0000000..72b5870
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch
@@ -0,0 +1,7 @@
+//
+// Prefix header for all source files of the 'SimpleCocoaGUI' target in the 'SimpleCocoaGUI' project
+//
+
+#ifdef __OBJC__
+ #import <Cocoa/Cocoa.h>
+#endif
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m b/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m
new file mode 100644
index 0000000..9d52a1c
--- /dev/null
+++ b/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m
@@ -0,0 +1,12 @@
+//
+// main.m
+// SimpleCocoaGUI
+//
+//
+
+#import <Cocoa/Cocoa.h>
+
+int main(int argc, char *argv[])
+{
+ return NSApplicationMain(argc, (const char **) argv);
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/Capture.rc b/trunk/src/video_engine/main/test/WindowsTest/Capture.rc
new file mode 100644
index 0000000..962256c
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/Capture.rc
@@ -0,0 +1,255 @@
+// Microsoft Visual C++ generated resource script.
+//
+#include "resource.h"
+
+#define APSTUDIO_READONLY_SYMBOLS
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 2 resource.
+//
+#include "afxres.h"
+
+/////////////////////////////////////////////////////////////////////////////
+#undef APSTUDIO_READONLY_SYMBOLS
+
+/////////////////////////////////////////////////////////////////////////////
+// Korean resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)
+#ifdef _WIN32
+LANGUAGE LANG_KOREAN, SUBLANG_DEFAULT
+#pragma code_page(949)
+#endif //_WIN32
+
+#ifdef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// TEXTINCLUDE
+//
+
+1 TEXTINCLUDE
+BEGIN
+ "resource.h\0"
+END
+
+2 TEXTINCLUDE
+BEGIN
+ "#include ""afxres.h""\r\n"
+ "\0"
+END
+
+3 TEXTINCLUDE
+BEGIN
+ "#define _AFX_NO_SPLITTER_RESOURCES\r\n"
+ "#define _AFX_NO_OLE_RESOURCES\r\n"
+ "#define _AFX_NO_TRACKER_RESOURCES\r\n"
+ "#define _AFX_NO_PROPERTY_RESOURCES\r\n"
+ "\r\n"
+ "#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)\r\n"
+ "#ifdef _WIN32\r\n"
+ "LANGUAGE 18, 1\r\n"
+ "#pragma code_page(949)\r\n"
+ "#endif //_WIN32\r\n"
+ "#include ""res\\Capture.rc2"" // non-Microsoft Visual C++ edited resources\r\n"
+ "#include ""l.kor\\afxres.rc"" // Standard components\r\n"
+ "#endif\r\n"
+ "\0"
+END
+
+#endif // APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Version
+//
+
+VS_VERSION_INFO VERSIONINFO
+ FILEVERSION 1,0,0,1
+ PRODUCTVERSION 1,0,0,1
+ FILEFLAGSMASK 0x3fL
+#ifdef _DEBUG
+ FILEFLAGS 0x1L
+#else
+ FILEFLAGS 0x0L
+#endif
+ FILEOS 0x4L
+ FILETYPE 0x1L
+ FILESUBTYPE 0x0L
+BEGIN
+END
+
+#endif // Korean resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+/////////////////////////////////////////////////////////////////////////////
+// Swedish resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
+#ifdef _WIN32
+LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
+#pragma code_page(1252)
+#endif //_WIN32
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Dialog
+//
+
+IDD_SLAVE_CHANNEL DIALOGEX 0, 0, 677, 358
+STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
+CAPTION "Slave channel"
+FONT 8, "MS Shell Dlg", 400, 0, 0x1
+BEGIN
+ CONTROL "",IDC_IPADDRESS1,"SysIPAddress32",WS_TABSTOP,485,18,105,15
+ EDITTEXT IDC_LOCAL_PORT1,631,18,36,16,ES_AUTOHSCROLL
+ LTEXT "IP-address",IDC_STATIC,495,7,42,9
+ LTEXT "Local Port",IDC_STATIC,633,7,37,9
+ EDITTEXT IDC_REMOTE_PORT1,593,18,36,16,ES_AUTOHSCROLL
+ LTEXT "Port",IDC_STATIC,595,7,17,9
+ CONTROL "Ext",IDC_EXTTRANSPORT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,489,41,29,12
+ LTEXT "delay",IDC_STATIC,589,41,21,9
+ COMBOBOX IDC_PACKETLOSS,535,40,45,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
+ COMBOBOX IDC_DELAY,611,40,45,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
+END
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// DESIGNINFO
+//
+
+#ifdef APSTUDIO_INVOKED
+GUIDELINES DESIGNINFO
+BEGIN
+ IDD_SLAVE_CHANNEL, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 670
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 351
+ END
+END
+#endif // APSTUDIO_INVOKED
+
+#endif // Swedish resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+/////////////////////////////////////////////////////////////////////////////
+// Neutral (Sys. Default) resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_NEUSD)
+#ifdef _WIN32
+LANGUAGE LANG_NEUTRAL, SUBLANG_SYS_DEFAULT
+#pragma code_page(1252)
+#endif //_WIN32
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Dialog
+//
+
+IDD_DXQUALITY_DIALOG DIALOGEX 0, 0, 699, 385
+STYLE DS_ABSALIGN | DS_SETFONT | DS_MODALFRAME | DS_3DLOOK | WS_MINIMIZEBOX | WS_POPUP | WS_VISIBLE | WS_CAPTION | WS_SYSMENU
+EXSTYLE WS_EX_WINDOWEDGE | WS_EX_STATICEDGE | WS_EX_APPWINDOW | WS_EX_NOINHERITLAYOUT
+CAPTION "webrtc ViE test program"
+FONT 9, "Arial", 400, 0, 0x0
+BEGIN
+ PUSHBUTTON "Start Send",IDC_STARTSEND,589,270,50,19
+ PUSHBUTTON "Stop Send",IDC_STOPSend,639,270,50,19
+ PUSHBUTTON "Start Listen",IDC_STARTLISTEN,589,291,50,19
+ PUSHBUTTON "StopListen",IDC_STOPLISTEN,639,291,50,19
+ CONTROL "",IDC_LIVEVIDEO,"Static",SS_BITMAP | SS_CENTERIMAGE | SS_SUNKEN,450,179,139,101
+ COMBOBOX IDC_DEVICE,487,14,185,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ CTEXT "Select Capture Device",IDC_STATIC,485,7,78,8
+ COMBOBOX IDC_CODEC_LIST,490,90,58,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Codec",IDC_STATIC,490,82,21,8
+ COMBOBOX IDC_CODEC_SIZE,627,90,62,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Codec Size",IDC_STATIC,611,82,36,8
+ CONTROL "",IDC_IPADDRESS1,"SysIPAddress32",WS_TABSTOP,490,46,90,13
+ EDITTEXT IDC_LOCAL_PORT1,615,46,31,14,ES_AUTOHSCROLL
+ LTEXT "IP-address",IDC_STATIC,498,37,36,8
+ LTEXT "Local Port",IDC_STATIC,616,36,32,8
+ LTEXT "Start Bitrate",IDC_STATIC,553,80,37,8
+ COMBOBOX IDC_BITRATE,558,90,49,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ EDITTEXT IDC_REMOTE_PORT1,582,46,31,14,ES_AUTOHSCROLL
+ LTEXT "Port",IDC_STATIC,584,37,14,8
+ GROUPBOX "Remote client 1",IDC_STATIC,487,27,203,50
+ LTEXT "Max FrameRate",IDC_STATIC,488,106,50,8
+ COMBOBOX IDC_MIN_FRAME_RATE,488,115,48,82,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ CONTROL "",IDC_CAPTURE,"Static",SS_BITMAP | SS_CENTERIMAGE | SS_REALSIZEIMAGE,7,7,418,276
+ CONTROL "TMMBR",IDC_TMMBR,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,486,138,40,10
+ GROUPBOX "Standard Protection",IDC_STATIC,607,138,72,55
+ CONTROL "None",IDC_PROT_NONE,"Button",BS_AUTORADIOBUTTON | WS_GROUP,615,146,33,10
+ CONTROL "NACK",IDC_PROT_NACK,"Button",BS_AUTORADIOBUTTON,615,165,35,10
+ CONTROL "FEC",IDC_PROT_FEC,"Button",BS_AUTORADIOBUTTON,615,155,30,10
+ CONTROL "NACK & FEC",IDC_PROT_NACKFEC,"Button",BS_AUTORADIOBUTTON,615,174,52,10
+ COMBOBOX IDC_RTCPMODE,571,119,80,57,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "RTCP Mode",IDC_STATIC,571,110,39,8
+ LISTBOX IDC_INFORMATION,476,309,214,63,LBS_SORT | LBS_NOINTEGRALHEIGHT | WS_VSCROLL | WS_TABSTOP
+ COMBOBOX IDC_PACKETBURST,653,118,36,57,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Packet Burst",IDC_STATIC,649,109,40,8
+ CONTROL "Stop Log",IDC_FREEZELOG,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,465,292,44,10
+ PUSHBUTTON "Version",IDC_VERSION,530,291,55,16
+ CONTROL "Ext",IDC_EXTTRANSPORT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,493,66,25,10
+ LTEXT "loss",IDC_STATIC,519,66,15,8
+ LTEXT "delay",IDC_STATIC,578,66,18,8
+ COMBOBOX IDC_PACKETLOSS,533,65,38,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
+ COMBOBOX IDC_DELAY,598,65,38,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
+ PUSHBUTTON "Record Incoming",IDC_BTN_RECORD_INCOMING,587,198,69,14
+ PUSHBUTTON "Record outgoing",IDC_BTN_RECORD_OUTGOING,587,212,69,14
+ PUSHBUTTON "Create Slave",IDC_BTN_CREATE_SLAVE,586,231,50,14
+END
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// DESIGNINFO
+//
+
+#ifdef APSTUDIO_INVOKED
+GUIDELINES DESIGNINFO
+BEGIN
+ IDD_DXQUALITY_DIALOG, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 690
+ VERTGUIDE, 321
+ VERTGUIDE, 372
+ VERTGUIDE, 425
+ VERTGUIDE, 465
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 372
+ END
+END
+#endif // APSTUDIO_INVOKED
+
+#endif // Neutral (Sys. Default) resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+
+#ifndef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 3 resource.
+//
+#define _AFX_NO_SPLITTER_RESOURCES
+#define _AFX_NO_OLE_RESOURCES
+#define _AFX_NO_TRACKER_RESOURCES
+#define _AFX_NO_PROPERTY_RESOURCES
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)
+#ifdef _WIN32
+LANGUAGE 18, 1
+#pragma code_page(949)
+#endif //_WIN32
+#include "res\Capture.rc2" // non-Microsoft Visual C++ edited resources
+#include "l.kor\afxres.rc" // Standard components
+#endif
+
+/////////////////////////////////////////////////////////////////////////////
+#endif // not APSTUDIO_INVOKED
+
diff --git a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc
new file mode 100644
index 0000000..8f7c9a2
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc
@@ -0,0 +1,98 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "CaptureDevicePool.h"
+#include "map_wrapper.h"
+#include <string.h>
+#include <assert.h>
+#include "critical_section_wrapper.h"
+#include "vie_file.h"
+
+CaptureDevicePool::CaptureDevicePool(VideoEngine* videoEngine):
+_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_vieCapture(ViECapture::GetInterface(videoEngine)),
+_vieFile(ViEFile::GetInterface(videoEngine))
+{
+}
+
+CaptureDevicePool::~CaptureDevicePool(void)
+{
+ assert(_deviceMap.Size()==0);
+ _vieCapture->Release();
+ _vieFile->Release();
+ delete &_critSect;
+}
+
+WebRtc_Word32 CaptureDevicePool::GetCaptureDevice(int& captureId, const char* uniqeDeviceName)
+{
+ CriticalSectionScoped cs(_critSect);
+ DeviceItem* device=NULL;
+
+ for(MapItem* item=_deviceMap.First();
+ item!=NULL;
+ item=_deviceMap.Next(item))
+ {
+ //Found the device?
+ if(strcmp(uniqeDeviceName,(static_cast<DeviceItem*>( item->GetItem()))->uniqeDeviceName)==0)
+ {
+ device=static_cast<DeviceItem*>( item->GetItem());
+ device->refCount++;
+ captureId=device->captureId;
+ return 0;
+ }
+ }
+ device = new DeviceItem;
+ strncpy(device->uniqeDeviceName,uniqeDeviceName,255);
+
+
+ // Device does not exist. Create it.
+ WebRtc_Word32 result=_vieCapture->AllocateCaptureDevice(device->uniqeDeviceName,strlen(device->uniqeDeviceName),device->captureId);
+ if(result==0)
+ {
+ //CaptureCapability cap;
+ /*cap.height=1080;
+ cap.width=1920;
+ cap.maxFPS=25;
+ cap.interlaced=true;*/
+ // result=_vieCapture->StartCapture(device->captureId,cap);
+ result=_vieFile->SetCaptureDeviceImage(device->captureId,"captureDeviceImage.jpg");
+ }
+ captureId=device->captureId;
+ _deviceMap.Insert(captureId,device);
+ device->refCount++;
+
+ return result;
+
+
+}
+WebRtc_Word32 CaptureDevicePool::ReturnCaptureDevice(int captureId)
+{
+ CriticalSectionScoped cs(_critSect);
+
+ MapItem* mapItem=_deviceMap.Find(captureId);
+ if(!mapItem)
+ return -1;
+
+ DeviceItem* item=static_cast<DeviceItem*> (mapItem->GetItem());
+ if(!item)
+ return 0;
+ item->refCount--;
+ WebRtc_Word32 result=0;
+
+ if(item->refCount==0)
+ {
+ result=_vieCapture->ReleaseCaptureDevice(captureId);
+
+ _deviceMap.Erase(mapItem);
+ delete item;
+
+ }
+ return result;
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h
new file mode 100644
index 0000000..104b84f
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma once
+
+#include "common_types.h"
+
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_file.h"
+#include "map_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+}
+using namespace webrtc;
+class CaptureDevicePool
+{
+public:
+ CaptureDevicePool(VideoEngine* videoEngine);
+ ~CaptureDevicePool(void);
+ WebRtc_Word32 GetCaptureDevice(int& captureId, const char uniqeDeviceName[256]);
+ WebRtc_Word32 ReturnCaptureDevice(int captureId);
+
+ private:
+ struct DeviceItem
+ {
+ int captureId;
+ WebRtc_Word32 refCount;
+ char uniqeDeviceName[256];
+ DeviceItem()
+ {
+ captureId=-1;
+ refCount=0;
+ }
+ };
+ CriticalSectionWrapper& _critSect;
+ ViECapture* _vieCapture;
+ ViEFile* _vieFile;
+ MapWrapper _deviceMap;
+
+};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc
new file mode 100644
index 0000000..cccfa8e
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc
@@ -0,0 +1,1297 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ChannelDlg.h"
+#include "VideoSize.h"
+#include "CaptureDevicePool.h"
+#include "ChannelPool.h"
+
+#include <Mmsystem.h>
+#include <dbt.h>
+
+
+#include "assert.h"
+
+
+#include <process.h> // threads.
+
+#if defined _WIN32
+ #define SLEEP_10_SEC ::Sleep(10000)
+ #define GET_TIME_IN_MS timeGetTime
+#endif
+
+// Hack to convert char to TCHAR, using two buffers to be able to
+// call twice in the same statement
+TCHAR convertTemp1[256] = {0};
+TCHAR convertTemp2[256] = {0};
+bool convertBufferSwitch(false);
+TCHAR* CharToTchar(const char* str, int len)
+{
+#ifdef _UNICODE
+ TCHAR* temp = convertBufferSwitch ? convertTemp1 : convertTemp2;
+ convertBufferSwitch = !convertBufferSwitch;
+ memset(temp, 0, sizeof(convertTemp1));
+ MultiByteToWideChar(CP_UTF8, 0, str, len, temp, 256);
+ return temp;
+#else
+ return str;
+#endif
+}
+
+// Hack to convert TCHAR to char
+char convertTemp3[256] = {0};
+char* TcharToChar(TCHAR* str, int len)
+{
+#ifdef _UNICODE
+ memset(convertTemp3, 0, sizeof(convertTemp3));
+ WideCharToMultiByte(CP_UTF8, 0, str, len, convertTemp3, 256, 0, 0);
+ return convertTemp3;
+#else
+ return str;
+#endif
+}
+
+/////////////////////////////////////////////////////////////////////////////
+// CDXChannelDlg dialog
+
+CDXChannelDlg::CDXChannelDlg(VideoEngine* videoEngine,
+ CaptureDevicePool& captureDevicePool,
+ ChannelPool& channelPool,
+ void* voiceEngine
+ ,CWnd* pParent,CDXChannelDlgObserver* observer,
+ int parentChannel/*=-1*/)
+ : CDialog(CDXChannelDlg::IDD, pParent),
+ _canAddLog(true),
+ _dialogObserver(observer),
+ _videoEngine(videoEngine),
+ _captureDevicePool(captureDevicePool),
+ _channelPool(channelPool),
+ _parentChannel(parentChannel),
+#ifndef NO_VOICE_ENGINE
+ _voiceEngine((VoiceEngine*) voiceEngine),
+#endif
+ _callbackEvent(::CreateEvent( NULL, FALSE, FALSE, NULL)),
+ _externalTransport(NULL)
+{
+ strcpy(_logMsg,"");
+ _channelId = -1;
+ _audioChannel=-1;
+ _captureId=-1;
+
+ //_transport = NULL;
+
+
+ //{{AFX_DATA_INIT(CDXChannelDlg)
+ //}}AFX_DATA_INIT
+ // Note that LoadIcon does not require a subsequent DestroyIcon in Win32
+
+ InitializeCriticalSection(&_critCallback);
+ unsigned int threadID;
+ _callbackThread=(HANDLE)_beginthreadex(NULL,1024*1024,CallbackThread,(void*)this,0, &threadID);
+
+
+}
+
+void CDXChannelDlg::DoDataExchange(CDataExchange* pDX)
+{
+ CDialog::DoDataExchange(pDX);
+ //{{AFX_DATA_MAP(CDXChannelDlg)
+ DDX_Control(pDX, IDC_DEVICE, m_ctrlDevice);
+ DDX_Control(pDX, IDC_CODEC_LIST, m_ctrlCodec);
+ DDX_Control(pDX, IDC_CAPTURE, m_ctrlLiveRemoteVideo);
+ DDX_Control(pDX, IDC_LIVEVIDEO, m_ctrlLiveVideo);
+ DDX_Control(pDX, IDC_LOCAL_PORT1, m_localPort1);
+ DDX_Control(pDX, IDC_REMOTE_PORT1, m_remotePort1);
+ DDX_Control(pDX, IDC_IPADDRESS1, m_remoteIp1);
+ DDX_Control(pDX, IDC_CODEC_SIZE, m_ctrlCodecSize);
+ DDX_Control(pDX, IDC_RTCPMODE, m_ctrlRtcpMode);
+ DDX_Control(pDX, IDC_PACKETBURST, m_ctrlPacketBurst);
+ DDX_Control(pDX, IDC_BITRATE, m_ctrlBitrate);
+ DDX_Control(pDX, IDC_MIN_FRAME_RATE, m_ctrlMinFrameRate);
+ DDX_Control(pDX, IDC_TMMBR,m_cbTmmbr);
+ DDX_Control(pDX, IDC_EXTTRANSPORT,m_cbExternalTransport);
+ DDX_Control(pDX, IDC_PACKETLOSS,m_ctrlPacketLoss);
+ DDX_Control(pDX, IDC_DELAY,m_ctrlDelay);
+ DDX_Control(pDX, IDC_FREEZELOG,m_cbFreezeLog);
+ DDX_Control(pDX,IDC_INFORMATION,m_ctrlInfo);
+ //}}AFX_DATA_MAP
+}
+
+// ON_WM_SYSKEYDOWN ALT+key
+
+BEGIN_MESSAGE_MAP(CDXChannelDlg, CDialog)
+ //{{AFX_MSG_MAP(CDXChannelDlg)
+ ON_WM_SYSCOMMAND()
+ ON_WM_RBUTTONUP()
+ //ON_WM_DEVICECHANGE()
+ ON_WM_PAINT()
+ ON_WM_QUERYDRAGICON()
+ ON_BN_CLICKED(IDC_STARTSEND, OnStartSend)
+ ON_BN_CLICKED(IDC_STOPSend, OnStopSend)
+ //ON_WM_TIMER()
+ ON_WM_DESTROY()
+ //}}AFX_MSG_MAP
+ ON_CBN_SELCHANGE(IDC_CODEC_LIST, OnCbnSelchangeCodecList)
+ ON_CBN_SELCHANGE(IDC_DEVICE, OnCbnSelchangeDevice)
+ ON_CBN_SELCHANGE(IDC_CODEC_SIZE, OnCbnSelchangeSize)
+ ON_CBN_SELCHANGE(IDC_BITRATE, OnCbnSelchangeBitrate)
+ //ON_MESSAGE(WM_DISPLAYCHANGE, OnDisplayChange)
+ ON_CBN_SELCHANGE(IDC_MIN_FRAME_RATE, OnCbnSelchangeMinFrameRate)
+ ON_BN_CLICKED(IDC_STARTLISTEN, OnBnClickedStartlisten)
+ ON_BN_CLICKED(IDC_STOPLISTEN, OnBnClickedStoplisten)
+ ON_BN_CLICKED(IDC_TMMBR, &CDXChannelDlg::OnBnClickedTmmbr)
+ ON_CBN_SELCHANGE(IDC_RTCPMODE, &CDXChannelDlg::OnCbnSelchangeRtcpmode)
+ ON_BN_CLICKED(IDC_PROT_NACK, &CDXChannelDlg::OnBnClickedProtNack)
+ ON_BN_CLICKED(IDC_PROT_NONE, &CDXChannelDlg::OnBnClickedProtNone)
+ ON_BN_CLICKED(IDC_PROT_FEC, &CDXChannelDlg::OnBnClickedProtFec)
+ ON_BN_CLICKED(IDC_FREEZELOG, &CDXChannelDlg::OnBnClickedFreezelog)
+ ON_BN_CLICKED(IDC_VERSION, &CDXChannelDlg::OnBnClickedVersion)
+ ON_BN_CLICKED(IDC_EXTTRANSPORT, &CDXChannelDlg::OnBnClickedExttransport)
+ ON_CBN_SELCHANGE(IDC_PACKETLOSS, &CDXChannelDlg::OnCbnSelchangePacketloss)
+ ON_CBN_SELCHANGE(IDC_DELAY, &CDXChannelDlg::OnCbnSelchangeDelay)
+ ON_BN_CLICKED(IDC_BTN_RECORD_INCOMING, &CDXChannelDlg::OnBnClickedBtnRecordIncoming)
+ ON_BN_CLICKED(IDC_BTN_RECORD_OUTGOING, &CDXChannelDlg::OnBnClickedBtnRecordOutgoing)
+ ON_BN_CLICKED(IDC_BTN_CREATE_SLAVE, &CDXChannelDlg::OnBnClickedBtnCreateSlave)
+ ON_BN_CLICKED(IDC_PROT_NACKFEC, &CDXChannelDlg::OnBnClickedProtNackFec)
+END_MESSAGE_MAP()
+
+/////////////////////////////////////////////////////////////////////////////
+// CDXChannelDlg message handlers
+
+
+BOOL CDXChannelDlg::OnInitDialog()
+{
+ CDialog::OnInitDialog();
+
+ // Set the icon for this dialog. The framework does this automatically
+ // when the application's main window is not a dialog
+ SetIcon(m_hIcon, TRUE); // Set big icon
+ SetIcon(m_hIcon, FALSE); // Set small icon
+
+
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("5"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("7"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("8"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("9"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("11"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("12"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("13"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("14"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("15"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("16"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("17"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("18"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("19"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("21"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("22"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("23"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("24"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("25"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("26"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("27"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("28"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("29"));
+ ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
+ m_ctrlMinFrameRate.SetCurSel(25);
+
+ // Codec sizes
+ for(VideoSize i=UNDEFINED;i<NUMBER_OF_VIDEO_SIZE;i=VideoSize(i+1))
+ {
+ char sizeStr[64];
+ int width=0;
+ int height=0;
+ GetWidthHeight(i,width,height);
+ sprintf(sizeStr,"%d x %d",width,height);
+ ::SendMessage(m_ctrlCodecSize.m_hWnd, CB_ADDSTRING, 0,(LPARAM) CharToTchar(sizeStr,-1));
+ }
+ m_ctrlCodecSize.SetCurSel(8);
+
+ // RTCP mode
+ /*
+ kRtcpNone = 0,
+ kRtcpCompound_RFC4585 = 1,
+ kRtcpNonCompound_RFC5506 = 2 */
+ ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_NONE"));
+ ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_COMPOUND_RFC4585"));
+ ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_NON_COMPOUND_RFC5506"));
+ m_ctrlRtcpMode.SetCurSel(2);
+
+
+ //Packet Burst
+ ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
+ ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
+ ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
+ ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
+ m_ctrlPacketBurst.SetCurSel(0);
+
+
+ //Send Bitrate
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("50"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("100"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("200"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("300"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("500"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("1000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("2000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("3000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("4000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("5000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6000"));
+ ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("7000"));
+
+ m_ctrlBitrate.SetCurSel(3);
+
+ // External transport packet loss
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("2"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("4"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("8"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("12"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("14"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("16"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("18"));
+ ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
+ m_ctrlPacketLoss.SetCurSel(0);
+
+ // External transport delay
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("60"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("90"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("120"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("150"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("180"));
+ ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("210"));
+ m_ctrlDelay.SetCurSel(0);
+
+
+ _vieBase=ViEBase::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieBase==0,-5);
+
+ _vieCapture=ViECapture::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieCapture==0,-5);
+
+ _vieRTPRTCP=ViERTP_RTCP::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieRTPRTCP==0,-5);
+
+ _vieRender=ViERender::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieRender==0,-5);
+
+ _vieCodec=ViECodec::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieCodec==0,-5);
+ _vieNetwork=ViENetwork::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieNetwork==0,-5);
+
+ _vieFile=ViEFile::GetInterface(_videoEngine);
+ TEST_MUSTPASS(_vieFile==0,-5);
+
+#ifndef NO_VOICE_ENGINE
+
+ _veBase = VoEBase::GetInterface(_voiceEngine);
+ _veNetwork = VoENetwork::GetInterface(_voiceEngine);
+ _veCodec = VoECodec::GetInterface(_voiceEngine);
+ _veRTCP = VoERTP_RTCP::GetInterface(_voiceEngine);
+ TEST_MUSTPASS(_vieBase->SetVoiceEngine(_voiceEngine),-5);
+#endif
+
+ int err = 0;
+
+ char str[64];
+ bool found = false;
+
+
+ int captureIdx = 0;
+ while (-1 !=_vieCapture->GetCaptureDevice(captureIdx,str,sizeof(str),NULL,0))
+ {
+ char* tmp = strstr(str,"(VFW)");
+ if (!tmp)
+ {
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(str,-1));
+ found = true;
+ }
+ captureIdx++;
+ memset(str, 0, 64);
+ }
+ WIN32_FIND_DATA FindFileData;
+ HANDLE hFind;
+ //char fileSearch[256];
+ //strcpy(fileSearch,_T("*.avi"));
+ hFind = FindFirstFile(_T("*.avi"), &FindFileData);
+ if (hFind != INVALID_HANDLE_VALUE)
+ {
+
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)(FindFileData.cFileName));
+ while(FindNextFile(hFind,&FindFileData))
+ {
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)(FindFileData.cFileName));
+ }
+ FindClose(hFind);
+ }
+
+
+
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("Conference"));
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("None"));
+
+ if (!found)
+ {
+ strncpy(str,"N/A",64);
+ ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(str,-1));
+ }
+ m_ctrlDevice.SetCurSel(0);
+
+ //Codecs
+ int numOfCodecs = _vieCodec->NumberOfCodecs();
+ for(int i=0; i<numOfCodecs;++i)
+ {
+ VideoCodec codec;
+ if(-1 !=_vieCodec->GetCodec(i,codec))
+ {
+ ::SendMessage(m_ctrlCodec.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(codec.plName,-1));
+ }
+ }
+ m_ctrlCodec.SetCurSel(0);
+
+#ifndef NO_VOICE_ENGINE
+ CodecInst voiceCodec;
+ int numOfVeCodecs = _veCodec->NumOfCodecs();
+ for(int i=0; i<numOfVeCodecs;++i)
+ {
+ if(_veCodec->GetCodec(i,voiceCodec)!=-1)
+ {
+ if(strncmp(voiceCodec.plname,"ISAC",4)==0)
+ break;
+
+
+ }
+ }
+
+ _audioChannel = _veBase->CreateChannel();
+
+
+ TEST_MUSTPASS(_veRTCP->SetRTCPStatus(_audioChannel, true),-5);
+ TEST_MUSTPASS(_veCodec->SetSendCodec(_audioChannel, voiceCodec),-5);
+ TEST_MUSTPASS(_veBase->StartPlayout(_audioChannel),-5);
+
+#endif //NO_VOICE_ENGINE
+
+ if(_parentChannel==-1)
+ {
+ TEST_MUSTPASS(_vieBase->CreateChannel(_channelId),-5);
+ }
+ else // This is a slave channel
+ {
+ TEST_MUSTPASS(_vieBase->CreateChannel(_channelId,_parentChannel),-5);
+ }
+#ifndef NO_VOICE_ENGINE
+ TEST_MUSTPASS(_vieBase->ConnectAudioChannel(_channelId,_audioChannel),-5);
+#endif
+
+ _channelPool.AddChannel(_channelId);
+
+ //Set Receive codec
+ {
+ VideoCodec codec;
+ int numOfCodecs = _vieCodec->NumberOfCodecs();;
+ for(int i=0; i<numOfCodecs;++i)
+ {
+ if(-1 !=_vieCodec->GetCodec(i,codec))
+ {
+ if(codec.codecType == webrtc::kVideoCodecVP8)
+ {
+ codec.codecSpecific.VP8.feedbackModeOn = true;
+ codec.codecSpecific.VP8.pictureLossIndicationOn = true;
+ }
+ TEST_MUSTPASS(_vieCodec->SetReceiveCodec(_channelId,codec),-5);
+ }
+ }
+ }
+
+ //TMMBR
+ m_cbTmmbr.SetCheck(BST_CHECKED);
+ OnBnClickedTmmbr();
+
+ //Packet Burst
+ m_ctrlPacketBurst.SetCurSel(0);
+
+
+ //Protection method none
+ CButton *opProtection = (CButton *) GetDlgItem(IDC_PROT_NONE);
+ opProtection->SetCheck(BST_CHECKED);
+ OnBnClickedProtNone();
+
+
+ // Configure the renderer
+ ConfigureRender();
+
+ TEST_MUSTPASS(_vieCodec->RegisterEncoderObserver(_channelId,*this),kViECodecObserverAlreadyRegistered);
+ TEST_MUSTPASS(_vieCodec->RegisterDecoderObserver(_channelId,*this),-5);
+
+ TEST_MUSTPASS(_vieBase->RegisterObserver(*this),kViEBaseObserverAlreadyRegistered);
+
+
+
+
+ //Set captions based on channel id
+ m_remoteIp1.SetAddress(127,0,0,1);
+ CString port;
+ port.AppendFormat(_T("%d"),11111+_channelId*4);
+ m_remotePort1.SetWindowText(port);
+ m_localPort1.SetWindowText(port);
+
+ CString title;
+ this->GetWindowText(title);
+ if(_parentChannel==-1)
+ {
+ title.AppendFormat(_T("%s - channel %d"),title,_channelId);
+ }
+ else
+ {
+ title.AppendFormat(_T("%s - slave channel %d - parent %d"),title,_channelId,_parentChannel);
+ }
+ this->SetWindowText(title);
+
+ if(_parentChannel!=-1)
+ m_ctrlDevice.EnableWindow(FALSE); //Prevent from changing capture device
+
+ return TRUE; // return TRUE unless you set the focus to a control
+}
+
+
+
+void CDXChannelDlg::OnTimer(UINT nIDEvent)
+{
+ CDialog::OnTimer(nIDEvent);
+}
+
+void CDXChannelDlg::SetSendCodec()
+{
+ // Get the codec stucture
+ int codecSel= m_ctrlCodec.GetCurSel();
+ VideoCodec codec;
+ TEST_MUSTPASS(_vieCodec->GetCodec(codecSel,codec),-5);
+
+
+ // Set Codec Size
+ VideoSize sizeSel=VideoSize(m_ctrlCodecSize.GetCurSel());
+ int width, height;
+ GetWidthHeight(sizeSel, width, height);
+ codec.width=width;
+ codec.height=height;
+
+ //Set the codec bitrate
+ CString bitrateStr;
+ m_ctrlBitrate.GetLBText(m_ctrlBitrate.GetCurSel(), bitrateStr);
+ int bitrate = _ttoi(bitrateStr.GetBuffer(0));
+ if(codec.codecType!=kVideoCodecI420)
+ {
+ codec.startBitrate=bitrate;
+ codec.maxBitrate=bitrate*4;
+ }
+
+
+ //Set the codec frame rate
+ codec.maxFramerate = m_ctrlMinFrameRate.GetCurSel() +5;
+
+ if(strncmp(codec.plName, "VP8", 5) == 0)
+ {
+ codec.codecSpecific.VP8.feedbackModeOn = true;
+ codec.codecSpecific.VP8.pictureLossIndicationOn = true;
+ TEST_MUSTPASS(_vieRTPRTCP->SetKeyFrameRequestMethod(_channelId, kViEKeyFrameRequestPliRtcp),-5);
+ }else
+ {
+ TEST_MUSTPASS(_vieRTPRTCP->SetKeyFrameRequestMethod(_channelId, kViEKeyFrameRequestPliRtcp),-5);
+ }
+ TEST_MUSTPASS(_vieCodec->SetSendCodec(_channelId, codec),-5);
+
+ if (codec.codecType == webrtc::kVideoCodecI420)
+ { // Need to set the receive codec size
+ _vieCodec->SetReceiveCodec(_channelId, codec);
+ }
+}
+
+void CDXChannelDlg::SetSendDestination()
+{
+ if(_externalTransport)
+ return;
+
+ BYTE part1, part2, part3, part4;
+ char sendIP1[16];
+ m_remoteIp1.GetAddress(part1, part2, part3, part4);
+ sprintf(sendIP1,"%d.%d.%d.%d",part1,part2,part3,part4);
+
+ CString strPort;
+ m_remotePort1.GetWindowText(strPort);
+ int remotePort1 = _ttoi(strPort.GetString());
+
+#ifdef IPV6
+ char* recIP = "::0";
+#else
+ char* recIP = "0.0.0.0";
+#endif //IPV6
+
+ TEST_MUSTPASS(_vieNetwork->SetSendDestination(_channelId,sendIP1,remotePort1),kViENetworkAlreadySending);
+
+ #ifndef NO_VOICE_ENGINE
+ m_localPort1.GetWindowText(strPort);
+ int localPort1 = _ttoi(strPort.GetString());
+ int res=_veBase->SetLocalReceiver(_audioChannel,localPort1+2);
+
+ TEST_MUSTPASS(_veBase->SetSendDestination(_audioChannel, remotePort1+2, sendIP1),-5)
+ #endif
+
+}
+
+void CDXChannelDlg::SetLocalReceiver()
+{
+ if(_externalTransport)
+ return;
+
+ CString strPort;
+ m_localPort1.GetWindowText(strPort);
+ int localPort1 = _ttoi(strPort.GetString());
+
+
+
+ // May fail because we are sending
+ TEST_MUSTPASS(_vieNetwork->SetLocalReceiver(_channelId, localPort1),-5);
+
+ #ifndef NO_VOICE_ENGINE
+ int res=_veBase->SetLocalReceiver(_audioChannel,localPort1+2);
+ #endif
+
+
+
+}
+
+void CDXChannelDlg::SetCaptureDevice()
+{
+ if(_parentChannel!=-1) // don't accept changing input on slave channels.
+ return;
+
+ int camSel=-1;
+ camSel=m_ctrlDevice.GetCurSel();
+
+ CString captureStr;
+ //captureStr.Compare
+ m_ctrlDevice.GetLBText(camSel, captureStr);
+ if(captureStr!=_T("N/A") != 0)
+ {
+
+ TEST_MUSTPASS(_vieFile->StopPlayFile(_captureId),kViEFileNotPlaying);
+ TEST_MUSTPASS(_vieCapture->DisconnectCaptureDevice(_channelId),kViECaptureDeviceNotConnected);
+ TEST_MUSTPASS(_vieRender->RemoveRenderer(_captureId),kViERenderInvalidRenderId);
+
+ if(_captureId>=0x1001 && _captureId<0x10FF)// ID is a capture device
+ {
+ TEST_MUSTPASS(_captureDevicePool.ReturnCaptureDevice(_captureId),-5);
+ }
+
+ if(captureStr!=_T("None")==0)
+ {
+ _captureId=-1;
+ }
+ else if(_tcsstr(captureStr,_T(".avi"))!=NULL ) // Selected an AVI file
+ {
+ TEST_MUSTPASS(_vieFile->StartPlayFile(TcharToChar(captureStr.GetBuffer(),-1),_captureId,false,webrtc::kFileFormatAviFile),-5);
+ TEST_MUSTPASS(_vieRender->AddRenderer(_captureId,m_ctrlLiveVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
+ TEST_MUSTPASS(_vieRender->StartRender(_captureId),-5);
+ TEST_MUSTPASS(_vieFile->SendFileOnChannel(_captureId,_channelId),-5);
+ TEST_MUSTPASS(_vieFile->StartPlayFileAsMicrophone(_captureId,_channelId,true),-5);
+ //TEST_MUSTPASS(_vieFile->StartPlayAudioLocally(_captureId,_channelId),-5);
+ }
+ else
+ {
+
+ char captureName[256];
+ char uniqueCaptureName[256];
+
+ TEST_MUSTPASS(_vieCapture->GetCaptureDevice(camSel,captureName,256,uniqueCaptureName,256),-5);
+
+ TEST_MUSTPASS(_captureDevicePool.GetCaptureDevice(_captureId,uniqueCaptureName),-5);
+ TEST_MUSTPASS(_vieCapture->StartCapture(_captureId),kViECaptureDeviceAlreadyStarted);
+ TEST_MUSTPASS(_vieCapture->RegisterObserver(_captureId,*this),kViECaptureObserverAlreadyRegistered);
+
+ TEST_MUSTPASS(_vieRender->AddRenderer(_captureId,m_ctrlLiveVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
+ TEST_MUSTPASS(_vieCapture->ConnectCaptureDevice(_captureId,_channelId),-5);
+ TEST_MUSTPASS(_vieRender->StartRender(_captureId),-5);
+ }
+ }
+
+}
+
+
+
+void CDXChannelDlg::OnBnClickedStartlisten()
+{
+
+
+ // Configure the local ports
+ SetLocalReceiver();
+
+ //Configure the remote destination- needed in order to be able to respond to RTCP messages
+ SetSendDestination();
+
+
+ #ifndef NO_VOICE_ENGINE
+ TEST_MUSTPASS(_veBase->StartReceive(_audioChannel),-5);
+ #endif
+ TEST_MUSTPASS(_vieBase->StartReceive(_channelId),-5);
+
+
+}
+
+void CDXChannelDlg::OnStartSend()
+{
+
+ // Set the send destination
+ SetSendDestination();
+
+ // Configure the local ports (Needed to be able to receive RTCP
+ //SetLocalReceiver();
+
+
+ // Set the send codec
+ SetSendCodec();
+
+ if(_captureId==-1) // If no capture device has been set.
+ SetCaptureDevice(); //Set the capture device
+
+
+
+ //TEST_MUSTPASS(_vieRTPRTCP->SetStartSequenceNumber(_channelId,1),-5);
+
+ // Start sending
+ TEST_MUSTPASS(_vieBase->StartSend(_channelId),-5);
+
+
+ #ifndef NO_VOICE_ENGINE
+ TEST_MUSTPASS(_veBase->StartSend(_audioChannel),-5);
+ #endif
+
+
+}
+
+void CDXChannelDlg::ConfigureRender()
+{
+ TEST_MUSTPASS(_vieRender->AddRenderer(_channelId,m_ctrlLiveRemoteVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
+
+ TEST_MUSTPASS(_vieFile->SetRenderStartImage(_channelId,"renderStartImage.jpg"),-5);
+ TEST_MUSTPASS(_vieRender->StartRender(_channelId),-5);
+ TEST_MUSTPASS(_vieFile->SetRenderTimeoutImage(_channelId,"renderTimeoutImage.jpg"),-5);
+
+
+}
+
+
+void CDXChannelDlg::OnStopSend()
+{
+
+ #ifndef NO_VOICE_ENGINE
+ TEST_MUSTPASS(_veBase->StopSend(_audioChannel),-5);
+ #endif
+
+
+ TEST_MUSTPASS(_vieBase->StopSend(_channelId),kViEBaseNotSending); // Accept error Not sending
+
+
+}
+void CDXChannelDlg::OnBnClickedStoplisten()
+{
+
+
+ #ifndef NO_VOICE_ENGINE
+ TEST_MUSTPASS(_veBase->StopReceive(_audioChannel),-5);
+ #endif
+ TEST_MUSTPASS(_vieBase->StopReceive(_channelId),-5);
+}
+
+
+void CDXChannelDlg::OnDestroy()
+{
+
+ OnStopSend();
+ OnBnClickedStoplisten();
+
+ if(_vieCapture && _parentChannel==-1)
+ {
+ _vieCapture->DisconnectCaptureDevice(_channelId);
+ _captureDevicePool.ReturnCaptureDevice(_captureId);
+ }
+ if(_vieFile && _parentChannel!=-1)
+ {
+ TEST_MUSTPASS(_vieFile->StopPlayFile(_captureId),kViEFileNotPlaying);
+ }
+
+
+
+
+ if(_videoEngine)
+ {
+ if(_parentChannel==-1)
+ {
+ _vieCodec->DeregisterEncoderObserver(_channelId);
+ }
+ _vieBase->DeleteChannel(_channelId);
+ _channelPool.RemoveChannel(_channelId);
+ }
+
+ _videoEngine = NULL;
+#ifndef NO_VOICE_ENGINE
+
+ if (_voiceEngine)
+ {
+ _veBase->DeleteChannel(_audioChannel);
+ _veBase->Release();
+ _veNetwork->Release();
+ _veCodec->Release();
+ _veRTCP->Release();
+ }
+#endif
+
+
+ strcpy(_logMsg,"");
+ SetEvent(_callbackEvent);
+ MSG msg; // Wait until the callback thread exits. Need to handle messages since the callback thread can call SendMessage when updating UI
+ while(WaitForSingleObject(_callbackThread,10)==WAIT_TIMEOUT)
+ {
+ DWORD ret = PeekMessage( &msg, NULL, 0, 0,PM_REMOVE );
+ if (ret >0)
+ {
+ TranslateMessage(&msg);
+ DispatchMessage(&msg);
+ }
+ }
+
+ CloseHandle(_callbackThread);
+ CloseHandle(_callbackEvent);
+ DeleteCriticalSection(&_critCallback);
+
+ TEST_MUSTPASS(_vieCapture->Release()<0,-5);
+ TEST_MUSTPASS(_vieRTPRTCP->Release()<0,-5);
+ TEST_MUSTPASS(_vieRender->Release()<0,-5);
+ TEST_MUSTPASS(_vieCodec->Release()<0,-5);
+ TEST_MUSTPASS(_vieNetwork->Release()<0,-5);
+ TEST_MUSTPASS(_vieFile->Release()<0,-5);
+ TEST_MUSTPASS(_vieBase->Release()<0,-5);
+
+
+
+#ifdef TEST_EXTERNAL_TRANSPORT
+ if(_transport)
+ delete _transport;
+ _transport = NULL;
+#endif
+
+ delete _externalTransport;
+
+ CDialog::OnDestroy();
+ if(_dialogObserver)
+ {
+ _dialogObserver->ChannelDialogEnded(this);
+ }
+}
+
+void CDXChannelDlg::OnCancel()
+{
+ DestroyWindow();
+}
+// If you add a minimize button to your dialog, you will need the code below
+// to draw the icon. For MFC applications using the document/view model,
+// this is automatically done for you by the framework.
+
+void CDXChannelDlg::OnPaint()
+{
+ if (IsIconic())
+ {
+ CPaintDC dc(this); // device context for painting
+
+ SendMessage(WM_ICONERASEBKGND, (WPARAM) dc.GetSafeHdc(), 0);
+
+ // Center icon in client rectangle
+ int cxIcon = GetSystemMetrics(SM_CXICON);
+ int cyIcon = GetSystemMetrics(SM_CYICON);
+ CRect rect;
+ GetClientRect(&rect);
+ int x = (rect.Width() - cxIcon + 1) / 2;
+ int y = (rect.Height() - cyIcon + 1) / 2;
+
+ // Draw the icon
+ dc.DrawIcon(x, y, m_hIcon);
+ }
+ else
+ {
+ CDialog::OnPaint();
+ }
+}
+
+BOOL CDXChannelDlg::OnDeviceChange( UINT nID, DWORD lParam)
+{
+ if(nID == DBT_DEVNODES_CHANGED)
+ {
+ // SetCaptureDevice();
+ }
+ return CDialog::OnDeviceChange(nID, lParam);
+}
+
+
+void CDXChannelDlg::OnSysCommand(UINT nID, LPARAM lParam)
+{
+ if(SC_MAXIMIZE == nID)
+ {}
+ CDialog::OnSysCommand(nID, lParam);
+}
+
+
+static bool fullScreen = false;
+void CDXChannelDlg::OnRButtonUp( UINT nFlags, CPoint point)
+{
+ CDialog::OnRButtonUp( nFlags, point);
+}
+
+// The system calls this to obtain the cursor to display while the user drags
+// the minimized window.
+HCURSOR CDXChannelDlg::OnQueryDragIcon()
+{
+ return (HCURSOR) m_hIcon;
+}
+
+void CDXChannelDlg::OnCbnSelchangeCodecList()
+{
+ SetSendCodec();
+}
+
+
+void CDXChannelDlg::OnCbnSelchangeSize()
+{
+ SetSendCodec();
+}
+
+void CDXChannelDlg::OnCbnSelchangeDevice()
+{
+
+
+ SetCaptureDevice();
+
+}
+
+
+void CDXChannelDlg::OnCbnSelchangeBitrate()
+{
+
+ SetSendCodec();
+
+}
+
+void CDXChannelDlg::OnCbnSelchangeMinFrameRate()
+{
+
+ SetSendCodec();
+
+}
+
+
+void CDXChannelDlg::OnBnClickedTmmbr()
+{
+
+ TEST_MUSTPASS(_vieRTPRTCP->SetTMMBRStatus(_channelId,m_cbTmmbr.GetCheck()==BST_CHECKED),-5);
+
+}
+
+void CDXChannelDlg::OnCbnSelchangeRtcpmode()
+{
+
+ /*
+ kRtcpNone = 0,
+ kRtcpCompound_RFC4585 = 1,
+ kRtcpNonCompound_RFC5506 = 2 */
+ ViERTCPMode mode=ViERTCPMode(m_ctrlRtcpMode.GetCurSel());
+ TEST_MUSTPASS(_vieRTPRTCP->SetRTCPStatus(_channelId,mode),-5);
+
+}
+
+void CDXChannelDlg::OnBnClickedFreezelog()
+{
+ _canAddLog=m_cbFreezeLog.GetCheck()!=BST_CHECKED;
+}
+
+void CDXChannelDlg::OnBnClickedProtNack()
+{
+
+ TEST_MUSTPASS(_vieRTPRTCP->SetNACKStatus(_channelId,true),-5);
+
+}
+
+void CDXChannelDlg::OnBnClickedProtNone()
+{
+
+ TEST_MUSTPASS(_vieRTPRTCP->SetNACKStatus(_channelId,false),-5);
+ TEST_MUSTPASS(_vieRTPRTCP->SetFECStatus(_channelId,false,0,0),-5);
+ TEST_MUSTPASS(_vieRTPRTCP->SetHybridNACKFECStatus(_channelId,false,0,0),-5);
+}
+
+void CDXChannelDlg::OnBnClickedProtFec()
+{
+ int noCodec=_vieCodec->NumberOfCodecs();
+ int redPayloadType=0;
+ int fecPayloadType=0;
+ for(unsigned char i=0;i<noCodec;++i)
+ {
+ VideoCodec codec;
+ _vieCodec->GetCodec(i,codec);
+ if(codec.codecType==webrtc::kVideoCodecRED)
+ {
+ redPayloadType=codec.plType;
+ }
+ if(codec.codecType==webrtc::kVideoCodecULPFEC)
+ {
+ fecPayloadType=codec.plType;
+ }
+ }
+ TEST_MUSTPASS(_vieRTPRTCP->SetFECStatus(_channelId,true,redPayloadType,fecPayloadType),-5);
+}
+
+void CDXChannelDlg::OnBnClickedProtNackFec()
+{
+ int noCodec=_vieCodec->NumberOfCodecs();
+ int redPayloadType=0;
+ int fecPayloadType=0;
+ for(unsigned char i=0;i<noCodec;++i)
+ {
+ VideoCodec codec;
+ _vieCodec->GetCodec(i,codec);
+ if(codec.codecType==webrtc::kVideoCodecRED)
+ {
+ redPayloadType=codec.plType;
+ }
+ if(codec.codecType==webrtc::kVideoCodecULPFEC)
+ {
+ fecPayloadType=codec.plType;
+ }
+ }
+ TEST_MUSTPASS(_vieRTPRTCP->SetHybridNACKFECStatus(_channelId,true,
+ redPayloadType,
+ fecPayloadType),-5);
+
+}
+
+void CDXChannelDlg::OnBnClickedVersion()
+{
+ char version[1024];
+ _vieBase->GetVersion(version);
+ MessageBox(CharToTchar(version,-1));
+ int p=strlen(version);
+#ifndef NO_VOICE_ENGINE
+ _veBase->GetVersion(version);
+ MessageBox(CharToTchar(version,-1));
+#endif
+}
+
+unsigned int WINAPI CDXChannelDlg::CallbackThread(LPVOID lpParameter)
+{
+ static_cast<CDXChannelDlg*>(lpParameter)->CallbackThreadProcess();
+ return 0;
+}
+
+void CDXChannelDlg::CallbackThreadProcess()
+{
+ while(1)
+ {
+ if(WAIT_OBJECT_0==WaitForSingleObject(_callbackEvent,INFINITE))
+ {
+ char smsg[512];
+ EnterCriticalSection(&_critCallback);
+ strncpy(smsg,_logMsg,strlen(_logMsg)+1);
+ strcpy(_logMsg,"");
+
+
+ LeaveCriticalSection(&_critCallback);
+ if(strstr(smsg,"Send")!=NULL)
+ {
+ unsigned short fractionLost=0;
+ unsigned int cumulativeLost=0;
+ unsigned int extendedMax=0;
+ unsigned int jitter=0;
+ int rttMs=0;
+
+
+
+ _vieRTPRTCP->GetReceivedRTCPStatistics(_channelId,
+ fractionLost,
+ cumulativeLost,
+ extendedMax,
+ jitter,
+ rttMs);
+
+ //int bw=0;
+ //if(_vieCodec->GetAvailableBandwidth(_channelId,bw)==0)
+ //{
+ // sprintf(smsg,"%s, rtt %d, loss %d,bw %d", smsg,rttMs,fractionLost,bw);
+ //}
+ //else
+ //{
+ // _vieBase->LastError(); // Reset last error.
+ //}
+
+
+
+ }
+ if(strlen(smsg))
+ {
+ m_ctrlInfo.InsertString(0,(LPCTSTR) CharToTchar(smsg,-1));
+ while(m_ctrlInfo.GetCount()==151)
+ m_ctrlInfo.DeleteString(150);
+ }
+ else
+ {
+ break; // End the callback thread
+ }
+ }
+ }
+
+}
+void CDXChannelDlg::AddToInfo(const char* msg)
+{
+ if(!_canAddLog)
+ return;
+ EnterCriticalSection(&_critCallback);
+
+ SYSTEMTIME systemTime;
+ GetSystemTime(&systemTime);
+
+ if(strlen(_logMsg)==0)
+ {
+ SetEvent(_callbackEvent); // Notify of new
+ }
+
+ sprintf (_logMsg, "(%2u:%2u:%2u:%3u) %s", systemTime.wHour,
+ systemTime.wMinute,
+ systemTime.wSecond,
+ systemTime.wMilliseconds,
+ msg
+ );
+
+
+
+ LeaveCriticalSection(&_critCallback);
+
+
+}
+
+void CDXChannelDlg::IncomingRate(const int videoChannel,
+ unsigned int framerate,
+ unsigned int bitrate)
+{
+ char str[64];
+ sprintf(str,"Incoming Fr:%d br %d\n", framerate, bitrate);
+ AddToInfo(str);
+}
+
+void CDXChannelDlg::RequestNewKeyFrame(int channel)
+{
+ assert(!"(RequestNewKeyFrame why is it called");
+}
+void CDXChannelDlg::PerformanceAlarm(unsigned int cpuLoad)
+{
+ char str[64];
+ sprintf(str,"Performance alarm %d",cpuLoad);
+ AddToInfo(str);
+}
+void CDXChannelDlg::OutgoingRate(const int videoChannel,
+ unsigned int framerate,
+ unsigned int bitrate)
+ {
+ char str[64];
+ sprintf(str,"Send Fr:%d br %d", framerate, bitrate);
+ AddToInfo(str);
+ }
+void CDXChannelDlg::IncomingCodecChanged(const int videoChannel,
+ const VideoCodec& videoCodec)
+ {
+ char str[128];
+ sprintf(str,"Incoming codec channel:%d pltype:%d width:%d height:%d\n", videoChannel, videoCodec.plType, videoCodec.width,videoCodec.height);
+ AddToInfo(str);
+ }
+void CDXChannelDlg::BrightnessAlarm(const int captureId,
+ const Brightness brightness)
+{
+
+ switch(brightness)
+ {
+ case Normal:
+ AddToInfo("BrightnessAlarm - image ok.\n");
+ break;
+ case Bright:
+ AddToInfo("BrightnessAlarm - light image.\n");
+ break;
+ case Dark:
+ AddToInfo("BrightnessAlarm - dark image.\n");
+ break;
+ }
+}
+
+void CDXChannelDlg::CapturedFrameRate(const int captureId,
+ const unsigned char frameRate)
+{
+ char str[64];
+ sprintf(str,"Local Camera Frame rate:%d \n", frameRate);
+ AddToInfo(str);
+}
+
+void CDXChannelDlg::NoPictureAlarm(const int captureId,
+ const CaptureAlarm alarm)
+{
+ char str[64];
+ sprintf(str,"No Picture alarm\n");
+ AddToInfo(str);
+
+}
+
+
+void CDXChannelDlg::OnBnClickedExttransport()
+{
+ if(m_cbExternalTransport.GetCheck()==BST_CHECKED)
+ {
+ m_localPort1.EnableWindow(FALSE);
+ m_remotePort1.EnableWindow(FALSE);
+ m_remoteIp1.EnableWindow(FALSE);
+ m_ctrlPacketLoss.EnableWindow(TRUE);
+ m_ctrlDelay.EnableWindow(TRUE);
+ _externalTransport= new TbExternalTransport(*_vieNetwork);
+ _vieNetwork->RegisterSendTransport(_channelId,*_externalTransport);
+ }
+ else
+ {
+ _vieNetwork->DeregisterSendTransport(_channelId);
+
+ delete _externalTransport;
+ _externalTransport=NULL;
+ m_localPort1.EnableWindow(TRUE);
+ m_remotePort1.EnableWindow(TRUE);
+ m_remoteIp1.EnableWindow(TRUE);
+ m_ctrlPacketLoss.EnableWindow(FALSE);
+ m_ctrlDelay.EnableWindow(FALSE);
+ }
+}
+
+
+void CDXChannelDlg::OnCbnSelchangePacketloss()
+{
+ if(_externalTransport)
+ {
+ _externalTransport->SetPacketLoss(m_ctrlPacketLoss.GetCurSel()*2);
+ }
+}
+
+
+void CDXChannelDlg::OnCbnSelchangeDelay()
+{
+ if(_externalTransport)
+ {
+ _externalTransport->SetNetworkDelay(m_ctrlDelay.GetCurSel()*30);
+ }
+
+}
+
+void CDXChannelDlg::OnBnClickedBtnRecordIncoming()
+{
+
+ CButton *recordBtn = (CButton *) GetDlgItem(IDC_BTN_RECORD_INCOMING);
+
+ CString text;
+ recordBtn->GetWindowText(text);
+ if(text!=_T("Stop Rec Inc")!=0)
+ {
+ recordBtn->SetWindowText(_T("Stop Rec Inc"));
+ SYSTEMTIME time;
+ GetSystemTime(&time);
+ sprintf(_fileName,"IncomingChannel%d_%4d%2d%2d%2d%2d.avi",_channelId,time.wYear,time.wMonth,time.wDay,time.wHour,time.wMinute);
+
+ AudioSource audioSource=PLAYOUT;
+ webrtc::CodecInst audioCodec;
+ strcpy(audioCodec.plname,"L16");
+ audioCodec.rate = 256000;
+ audioCodec.plfreq = 16000;
+ audioCodec.pacsize = 160;
+
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec,0,sizeof(videoCodec));
+
+ strcpy(videoCodec.plName,"VP8");
+ videoCodec.maxBitrate=1000;
+ videoCodec.startBitrate=1000;
+ videoCodec.width=352;
+ videoCodec.height=288;
+ videoCodec.codecType=webrtc::kVideoCodecVP8;
+ videoCodec.maxFramerate=30;
+ TEST_MUSTPASS(_vieFile->StartRecordIncomingVideo(_channelId,_fileName,audioSource,audioCodec, videoCodec),-5);
+ }
+ else
+ {
+ recordBtn->SetWindowText(_T("Record Incoming"));
+ TEST_MUSTPASS(_vieFile->StopRecordIncomingVideo(_channelId),-5);
+ CString msg;
+ msg.AppendFormat(_T("Recorded file %s"),_fileName);
+ MessageBox(msg);
+ }
+}
+
+void CDXChannelDlg::OnBnClickedBtnRecordOutgoing()
+{
+
+ CButton *recordBtn = (CButton *) GetDlgItem(IDC_BTN_RECORD_OUTGOING);
+ CString text;
+ recordBtn->GetWindowText(text);
+ if(text!=_T("Stop Rec Out"))
+ {
+ recordBtn->SetWindowText(_T("Stop Rec Out"));
+ SYSTEMTIME time;
+ GetSystemTime(&time);
+ sprintf(_fileName,"OutgoingChannel%d_%4d%2d%2d%2d%2d.avi",_channelId,time.wYear,time.wMonth,time.wDay,time.wHour,time.wMinute);
+
+ AudioSource audioSource=MICROPHONE;
+ webrtc::CodecInst audioCodec;
+ strcpy(audioCodec.plname,"L16");
+ audioCodec.rate = 256000;
+ audioCodec.plfreq = 16000;
+ audioCodec.pacsize = 160;
+
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec,0,sizeof(videoCodec));
+
+ strcpy(videoCodec.plName,"VP8");
+ videoCodec.maxBitrate=1000;
+ videoCodec.startBitrate=1000;
+ videoCodec.width=352;
+ videoCodec.height=288;
+ videoCodec.codecType=webrtc::kVideoCodecVP8;
+ videoCodec.maxFramerate=30;
+ TEST_MUSTPASS(_vieFile->StartRecordOutgoingVideo(_channelId,_fileName,audioSource,audioCodec,videoCodec),-5);
+ }
+ else
+ {
+ recordBtn->SetWindowText(_T("Record Outgoing"));
+ TEST_MUSTPASS(_vieFile->StopRecordOutgoingVideo(_channelId),-5);
+ CString msg;
+ msg.AppendFormat(_T("Recorded file %s"),_fileName);
+ MessageBox(msg);
+ }
+}
+
+void CDXChannelDlg::OnBnClickedBtnCreateSlave()
+{
+ CDXChannelDlg* newSlave =new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,_dialogObserver,_channelId);
+ newSlave->Create(CDXChannelDlg::IDD,NULL);
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h
new file mode 100644
index 0000000..43aeb09
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h
@@ -0,0 +1,273 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
+
+#include "StdAfx.h"
+//#define NO_VOICE_ENGINE
+
+/////////////////////////////////////////////////////////////////////////////
+// CDXChannelDlg dialog
+// Include ViE headers
+
+#include "common_types.h"
+
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+#include "vie_errors.h"
+#include "vie_file.h"
+#include "tbExternalTransport.h"
+
+#include "resource.h" // main symbols
+
+
+#ifndef NO_VOICE_ENGINE
+
+#include "voe_base.h"
+#include "voe_errors.h"
+#include "voe_base.h"
+#include "voe_network.h"
+#include "voe_codec.h"
+#include "voe_rtp_rtcp.h"
+#endif
+
+using namespace webrtc;
+class CDXChannelDlg;
+class CaptureDevicePool;
+class ChannelPool;
+
+#define TEST_MUSTPASS(expr,oklasterror) \
+ { \
+ if ((expr)) \
+ { \
+ CString r_msg; \
+ int r_lastError=_vieBase->LastError(); \
+ CString exp; \
+ exp=#expr;\
+ r_msg.Format(_T("\nError at line:%i, %s \nError code: %i\n"),__LINE__, exp,r_lastError); \
+ if(r_lastError!=oklasterror) \
+ ::MessageBox (NULL, (LPCTSTR)r_msg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION); \
+ } \
+ }
+
+class CDXChannelDlgObserver
+{
+public:
+ virtual void ChannelDialogEnded(CDXChannelDlg* context)=0;
+
+protected:
+ virtual ~CDXChannelDlgObserver(){};
+
+};
+
+class CDXChannelDlg : public CDialog , public ViEEncoderObserver, public ViEDecoderObserver, public ViEBaseObserver, public ViECaptureObserver
+{
+// Construction
+public:
+ CDXChannelDlg(VideoEngine* videoEngine,
+ CaptureDevicePool& captureDevicePool,
+ ChannelPool& channelPool,
+ void* voiceEngine=NULL
+ ,CWnd* pParent = NULL,CDXChannelDlgObserver* observer=NULL,int parentChannel=-1); // standard constructor
+
+// Dialog Data
+ //{{AFX_DATA(CDXChannelDlg)
+ enum { IDD = IDD_DXQUALITY_DIALOG };
+ CComboBox m_ctrlDevice;
+ CComboBox m_ctrlCodec;
+ CComboBox m_ctrlBitrate;
+ CComboBox m_ctrlCodecSize;
+ CComboBox m_ctrlRtcpMode;
+ CComboBox m_ctrlPacketBurst;
+ CComboBox m_ctrlMinFrameRate;
+
+ CListBox m_ctrlInfo;
+
+ CStatic m_ctrlLiveRemoteVideo;
+ CStatic m_ctrlLiveVideo;
+ CEdit m_localPort1;
+ CEdit m_remotePort1;
+ CIPAddressCtrl m_remoteIp1;
+ CButton m_cbTmmbr;
+ CButton m_cbExternalTransport;
+ CButton m_cbFreezeLog;
+ CButton m_cbDefaultSendChannel;
+ CComboBox m_ctrlPacketLoss;
+ CComboBox m_ctrlDelay;
+
+
+ //}}AFX_DATA
+
+ // ClassWizard generated virtual function overrides
+ //{{AFX_VIRTUAL(CDXChannelDlg)
+ protected:
+ virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
+ //}}AFX_VIRTUAL
+
+
+
+public :
+ // Callback
+
+ //Capture observer
+ virtual void BrightnessAlarm(const int captureId,
+ const Brightness brightness);
+
+ virtual void CapturedFrameRate(const int captureId,
+ const unsigned char frameRate);
+
+ virtual void NoPictureAlarm(const int captureId,
+ const CaptureAlarm alarm);
+
+
+ // same callback method is being used to raise also to clear.
+ // true - raise, false - clear
+ // virtual void NoPictureAlarm(bool active = true);
+
+ // Encoder observer
+ virtual void OutgoingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate) ;
+
+ //Decoder observer
+ virtual void IncomingCodecChanged(const int videoChannel,
+ const VideoCodec& videoCodec);
+
+ virtual void IncomingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate);
+
+ virtual void RequestNewKeyFrame(const int videoChannel);
+
+ // Base observer
+ virtual void PerformanceAlarm(const unsigned int cpuLoad);
+
+
+ //virtual void IncomingCSRCChanged(int channel, unsigned int csrc, bool added);
+
+
+
+// Implementation
+protected:
+ HICON m_hIcon;
+ int _channelId;
+ int _parentChannel;
+ int _audioChannel;
+ bool _canAddLog;
+
+ // Thread and function for callbacks
+ CRITICAL_SECTION _critCallback;
+ HANDLE _callbackThread;
+ HANDLE _callbackEvent;
+ char _logMsg[512];
+ static unsigned int WINAPI CallbackThread(LPVOID lpParameter);
+ void CallbackThreadProcess();
+
+
+
+ //void GetSize(int sizeSel, int &width, int &height);
+ virtual void ConfigureRender();
+
+ virtual void SetCaptureDevice();
+ virtual void SetLocalReceiver();
+ virtual void SetSendDestination();
+ virtual void SetSendCodec();
+
+
+ void AddToInfo(const char* msg);
+
+ // afx_msg void Command(UINT nID, LPARAM lParam);
+
+ // Generated message map functions
+ //{{AFX_MSG(CDXChannelDlg)
+ virtual BOOL OnInitDialog();
+ afx_msg void OnSysCommand(UINT nID, LPARAM lParam);
+ afx_msg void OnRButtonUp( UINT nFlags, CPoint point);
+ afx_msg BOOL OnDeviceChange( UINT, DWORD );
+ afx_msg void OnPaint();
+ //afx_msg LRESULT OnDisplayChange(WPARAM, LPARAM);
+ afx_msg HCURSOR OnQueryDragIcon();
+ virtual afx_msg void OnStartSend();
+ virtual afx_msg void OnDestroy();
+ virtual afx_msg void OnStopSend();
+ virtual afx_msg void OnCancel();
+ afx_msg void OnTimer(UINT nIDEvent);
+
+ //}}AFX_MSG
+ DECLARE_MESSAGE_MAP()
+
+private:
+ CDXChannelDlgObserver* _dialogObserver;
+
+ VideoEngine* _videoEngine;
+ ViEBase* _vieBase;
+ ViECapture* _vieCapture;
+ ViERTP_RTCP* _vieRTPRTCP;
+ ViERender* _vieRender;
+ ViECodec* _vieCodec;
+ ViENetwork* _vieNetwork;
+ ViEFile* _vieFile;
+ TbExternalTransport* _externalTransport;
+ char _fileName[256];
+
+
+#ifndef NO_VOICE_ENGINE
+ VoiceEngine* _voiceEngine;
+ VoEBase* _veBase;
+ VoENetwork* _veNetwork;
+ VoECodec* _veCodec;
+ VoERTP_RTCP* _veRTCP;
+#else
+ void* _voiceEngine;
+
+#endif
+
+ VideoCodec _sendCodec;
+ int _captureId;
+ CaptureDevicePool& _captureDevicePool;
+ ChannelPool& _channelPool;
+
+
+ afx_msg void OnCbnSelchangeCodecList();
+ afx_msg void OnCbnSelchangeDevice();
+ afx_msg void OnCbnSelchangeSize();
+ afx_msg void OnCbnSelchangeBitrate();
+ afx_msg void OnCbnSelchangeWindowSize();
+ afx_msg void OnBnClickedversion();
+ afx_msg void OnCbnSelchangeMinFrameRate();
+ afx_msg void OnBnClickedStartlisten();
+ afx_msg void OnBnClickedStoplisten();
+ afx_msg void OnBnClickedStopsend();
+ afx_msg void OnBnClickedTmmbr();
+ afx_msg void OnCbnSelchangeRtcpmode();
+ afx_msg void OnBnClickedProtNack();
+ afx_msg void OnBnClickedProtNone();
+ afx_msg void OnBnClickedProtFec();
+ afx_msg void OnBnClickedProtNackFec();
+ afx_msg void OnBnClickedFreezelog();
+public:
+ afx_msg void OnBnClickedExttransport();
+ afx_msg void OnCbnSelchangePacketloss();
+ afx_msg void OnCbnSelchangeDelay();
+ afx_msg void OnBnClickedBtnRecordIncoming();
+ afx_msg void OnBnClickedBtnRecordOutgoing();
+ afx_msg void OnBnClickedBtnCreateSlave();
+ afx_msg void OnBnClickedVersion();
+};
+
+//{{AFX_INSERT_LOCATION}}
+// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc
new file mode 100644
index 0000000..dbd1644
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "ChannelPool.h"
+#include "map_wrapper.h"
+#include <string.h>
+#include <assert.h>
+#include "critical_section_wrapper.h"
+
+ChannelPool::ChannelPool():
+_critSect(*webrtc::CriticalSectionWrapper::CreateCriticalSection())
+{
+}
+
+ChannelPool::~ChannelPool(void)
+{
+ assert(_channelMap.Size()==0);
+ delete &_critSect;
+}
+
+WebRtc_Word32 ChannelPool::AddChannel(int channel)
+{
+ return _channelMap.Insert(channel,(void*) channel);
+}
+WebRtc_Word32 ChannelPool::RemoveChannel(int channel)
+{
+ return _channelMap.Erase(channel);
+}
+
+webrtc::MapWrapper& ChannelPool::ChannelMap()
+{
+ return _channelMap;
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h
new file mode 100644
index 0000000..374c676
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma once
+#include "StdAfx.h"
+#include "common_types.h"
+
+#include "vie_base.h"
+#include "map_wrapper.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+}
+
+class ChannelPool
+{
+public:
+ ChannelPool();
+ ~ChannelPool(void);
+ WebRtc_Word32 AddChannel(int channel);
+ WebRtc_Word32 RemoveChannel(int channel);
+
+ webrtc::MapWrapper& ChannelMap();
+
+ private:
+ webrtc::CriticalSectionWrapper& _critSect;
+ webrtc::MapWrapper _channelMap;
+
+};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h b/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h
new file mode 100644
index 0000000..78b1fbd
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// stdafx.h : include file for standard system include files,
+// or project specific include files that are used frequently, but
+// are changed infrequently
+//
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
+
+#if _MSC_VER > 1000
+#pragma once
+#endif // _MSC_VER > 1000
+
+#define VC_EXTRALEAN // Exclude rarely-used stuff from Windows headers
+
+#include <afxwin.h> // MFC core and standard components
+#include <afxext.h> // MFC extensions
+#include <afxdisp.h> // MFC Automation classes
+#include <afxdtctl.h> // MFC support for Internet Explorer 4 Common Controls
+#ifndef _AFX_NO_AFXCMN_SUPPORT
+#include <afxcmn.h> // MFC support for Windows Common Controls
+#endif // _AFX_NO_AFXCMN_SUPPORT
+
+
+//{{AFX_INSERT_LOCATION}}
+// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h b/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h
new file mode 100644
index 0000000..60e2bdd
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
+#include "StdAfx.h"
+enum VideoSize
+ {
+ UNDEFINED,
+ SQCIF, // 128*96 = 12 288
+ QQVGA, // 160*120 = 19 200
+ QCIF, // 176*144 = 25 344
+ CGA, // 320*200 = 64 000
+ QVGA, // 320*240 = 76 800
+ SIF, // 352*240 = 84 480
+ WQVGA, // 400*240 = 96 000
+ CIF, // 352*288 = 101 376
+ W288P, // 512*288 = 147 456 (WCIF)
+ W368P, // 640*368 = 235 520
+ S_448P, // 576*448 = 281 088
+ VGA, // 640*480 = 307 200
+ S_432P, // 720*432 = 311 040
+ W432P, // 768*432 = 331 776 (a.k.a WVGA 16:9)
+ S_4SIF, // 704*480 = 337 920
+ W448P, // 768*448 = 344 064
+ NTSC, // 720*480 = 345 600
+ FW448P, // 800*448 = 358 400
+ S_768x480P, // 768*480 = 368 640 (a.k.a WVGA 16:10)
+ WVGA, // 800*480 = 384 000
+ S_4CIF, // 704576 = 405 504
+ SVGA, // 800*600 = 480 000
+ W544P, // 960*544 = 522 240
+ W576P, // 1024*576 = 589 824 (W4CIF)
+ HD, // 960*720 = 691 200
+ XGA, // 1024*768 = 786 432
+ WHD, // 1280*720 = 921 600
+ FULL_HD, // 1440*1080 = 1 555 200
+ UXGA, // 1600*1200 = 1 920 000
+ WFULL_HD, // 1920*1080 = 2 073 600
+ NUMBER_OF_VIDEO_SIZE
+ };
+
+int GetWidthHeight(VideoSize size, int& width, int& height);
+
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc
new file mode 100644
index 0000000..ceda8a9
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "WindowsTest.h"
+#include "ChannelDlg.h"
+#include "WindowsTestMainDlg.h"
+#include "engine_configurations.h"
+
+#ifdef _DEBUG
+#define new DEBUG_NEW
+#undef THIS_FILE
+static char THIS_FILE[] = __FILE__;
+#endif
+
+// Check memory leaks id running debug
+#if (defined(_DEBUG) && defined(_WIN32))
+// #include "vld.h"
+#endif
+/////////////////////////////////////////////////////////////////////////////
+// CDXWindowsTestApp
+
+BEGIN_MESSAGE_MAP(CDXWindowsTestApp, CWinApp)
+ //{{AFX_MSG_MAP(CDXWindowsTestApp)
+ // NOTE - the ClassWizard will add and remove mapping macros here.
+ // DO NOT EDIT what you see in these blocks of generated code!
+ //}}AFX_MSG
+ ON_COMMAND(ID_HELP, CWinApp::OnHelp)
+END_MESSAGE_MAP()
+
+/////////////////////////////////////////////////////////////////////////////
+// CDXWindowsTestApp construction
+
+CDXWindowsTestApp::CDXWindowsTestApp()
+{
+
+}
+
+/////////////////////////////////////////////////////////////////////////////
+// The one and only object
+
+CDXWindowsTestApp theApp;
+
+/////////////////////////////////////////////////////////////////////////////
+// CDXWindowsTestApp initialization
+
+BOOL CDXWindowsTestApp::InitInstance()
+{
+ int result=0;
+ #ifndef NO_VOICE_ENGINE
+ _voiceEngine = VoiceEngine::Create();
+ _veBase = VoEBase::GetInterface(_voiceEngine);
+ result+=_veBase->Init();
+ #else
+ _voiceEngine=NULL;
+ #endif
+
+ _videoEngine = VideoEngine::Create();
+
+ _videoEngine->SetTraceFilter(webrtc::kTraceDefault);//webrtc::kTraceDebug | webrtc::kTraceError | webrtc::kTraceApiCall | webrtc::kTraceWarning | webrtc::kTraceCritical | webrtc::kTraceStateInfo | webrtc::kTraceInfo | webrtc::kTraceStream);
+ _videoEngine->SetTraceFile("trace.txt");
+
+ ViEBase* vieBase=ViEBase::GetInterface(_videoEngine);
+ result+=vieBase->Init();
+ if(result!=0)
+ {
+ ::MessageBox (NULL, (LPCTSTR)("failed to init VideoEngine"), TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
+ }
+
+ {
+ WindowsTestMainDlg dlg(_videoEngine,_voiceEngine);
+
+ m_pMainWnd = &dlg;
+ int nResponse = dlg.DoModal();
+ }
+
+ vieBase->Release();
+
+ if(!VideoEngine::Delete(_videoEngine))
+ {
+ char errorMsg[255];
+ sprintf(errorMsg,"All VideoEngine interfaces are not released properly!");
+ ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
+ }
+
+ #ifndef NO_VOICE_ENGINE
+
+ _veBase->Terminate();
+ if(_veBase->Release()!=0)
+ {
+ // ensure that no interface is still referenced
+ char errorMsg[256];
+ sprintf(errorMsg,"All VoiceEngine interfaces are not released properly!");
+ ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
+ }
+
+ if (false == VoiceEngine::Delete(_voiceEngine))
+ {
+ char errorMsg[256];
+ sprintf(errorMsg,"VoiceEngine::Delete() failed!");
+ ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
+ }
+ #endif
+
+ // Since the dialog has been closed, return FALSE so that we exit the
+ // application, rather than start the application's message pump.
+ return FALSE;
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h
new file mode 100644
index 0000000..dc3ee9d
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
+
+
+#include "StdAfx.h"
+#include "resource.h" // main symbols
+
+
+
+/////////////////////////////////////////////////////////////////////////////
+
+//Forward declarations
+namespace webrtc {
+ class VoiceEngine;
+ class VoEBase;
+ class VideoEngine;
+}
+using namespace webrtc;
+
+class CDXWindowsTestApp : public CWinApp
+{
+public:
+ CDXWindowsTestApp();
+
+// Overrides
+ // ClassWizard generated virtual function overrides
+ //{{AFX_VIRTUAL(CDXWindowsTestApp)
+ public:
+ virtual BOOL InitInstance();
+ //}}AFX_VIRTUAL
+
+// Implementation
+
+ //{{AFX_MSG(CDXWindowsTestApp)
+ // NOTE - the ClassWizard will add and remove member functions here.
+ // DO NOT EDIT what you see in these blocks of generated code !
+ //}}AFX_MSG
+ DECLARE_MESSAGE_MAP()
+
+ VideoEngine* _videoEngine;
+ VoiceEngine* _voiceEngine;
+ VoEBase* _veBase;
+};
+
+
+/////////////////////////////////////////////////////////////////////////////
+
+//{{AFX_INSERT_LOCATION}}
+// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc
new file mode 100644
index 0000000..fcc490d
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// WindowsTestMainDlg.cpp : implementation file
+//
+#include "WindowsTestMainDlg.h"
+#include "WindowsTest.h"
+#include "ChannelDlg.h"
+
+#include "voe_base.h"
+
+// WindowsTestMainDlg dialog
+
+IMPLEMENT_DYNAMIC(WindowsTestMainDlg, CDialog)
+
+WindowsTestMainDlg::WindowsTestMainDlg(VideoEngine* videoEngine,void* voiceEngine,CWnd* pParent /*=NULL*/)
+ : CDialog(WindowsTestMainDlg::IDD, pParent),
+ _videoEngine(videoEngine),
+ _voiceEngine((VoiceEngine*) voiceEngine),
+ _testDlg1(NULL),
+ _testDlg2(NULL),
+ _testDlg3(NULL),
+ _testDlg4(NULL),
+ _externalInWidth(0),
+ _externalInHeight(0),
+ _externalInVideoType(0),
+ _captureDevicePool(videoEngine)
+{
+
+}
+
+WindowsTestMainDlg::~WindowsTestMainDlg()
+{
+}
+
+void WindowsTestMainDlg::DoDataExchange(CDataExchange* pDX)
+{
+ CDialog::DoDataExchange(pDX);
+}
+
+
+BEGIN_MESSAGE_MAP(WindowsTestMainDlg, CDialog)
+ ON_BN_CLICKED(IDC_CHANNEL1, &WindowsTestMainDlg::OnBnClickedChannel1)
+ ON_BN_CLICKED(IDC_CHANNEL2, &WindowsTestMainDlg::OnBnClickedChannel2)
+ ON_BN_CLICKED(IDC_CHANNEL3, &WindowsTestMainDlg::OnBnClickedChannel3)
+ ON_BN_CLICKED(IDC_CHANNEL4, &WindowsTestMainDlg::OnBnClickedChannel4)
+END_MESSAGE_MAP()
+
+
+
+void WindowsTestMainDlg::OnBnClickedChannel1()
+{
+ if(!_testDlg1)
+ {
+ _testDlg1=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
+ _testDlg1->Create(CDXChannelDlg::IDD,this);
+ }
+ else
+ {
+ _testDlg1->SetActiveWindow();
+ }
+}
+
+void WindowsTestMainDlg::OnBnClickedChannel2()
+{
+ if(!_testDlg2)
+ {
+ _testDlg2=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
+ _testDlg2->Create(CDXChannelDlg::IDD,this);
+
+ }
+ else
+ {
+ _testDlg2->SetActiveWindow();
+ }
+}
+
+void WindowsTestMainDlg::ChannelDialogEnded(CDXChannelDlg* context)
+{
+ if(context==_testDlg4)
+ {
+ delete _testDlg4;
+ _testDlg4=NULL;
+ }
+ else if(context==_testDlg3)
+ {
+ delete _testDlg3;
+ _testDlg3=NULL;
+ }
+ else if(context==_testDlg2)
+ {
+ delete _testDlg2;
+ _testDlg2=NULL;
+ }
+ else if(context==_testDlg1)
+ {
+ delete _testDlg1;
+ _testDlg1=NULL;
+ }
+ else // Slave channel
+ {
+ delete context;
+ }
+
+}
+
+
+
+void WindowsTestMainDlg::OnBnClickedChannel3()
+{
+ if(!_testDlg3)
+ {
+ _testDlg3=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
+ _testDlg3->Create(CDXChannelDlg::IDD,this);
+
+ }
+ else
+ {
+ _testDlg3->SetActiveWindow();
+ }
+}
+
+void WindowsTestMainDlg::OnBnClickedChannel4()
+{
+ if(!_testDlg4)
+ {
+ _testDlg4=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
+ _testDlg4->Create(CDXChannelDlg::IDD,this);
+
+ }
+ else
+ {
+ _testDlg4->SetActiveWindow();
+ }
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h
new file mode 100644
index 0000000..8aae99a
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma once
+#include "StdAfx.h"
+#include "WindowsTestResource.h"
+
+#include "ChannelDlg.h"
+#include "CaptureDevicePool.h"
+#include "ChannelPool.h"
+
+//Forward declarations
+namespace webrtc {
+ class VideoEngine;
+ class VoiceEngine;
+}
+using namespace webrtc;
+class CDXCaptureDlg;
+
+
+class WindowsTestMainDlg : public CDialog, private CDXChannelDlgObserver
+{
+ DECLARE_DYNAMIC(WindowsTestMainDlg)
+
+public:
+ WindowsTestMainDlg(VideoEngine* videoEngine,void* voiceEngine=NULL,CWnd* pParent = NULL); // standard constructor
+ virtual ~WindowsTestMainDlg();
+
+// Dialog Data
+ enum { IDD = IDD_WINDOWSTEST_MAIN };
+
+protected:
+ virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
+
+ DECLARE_MESSAGE_MAP()
+public:
+ afx_msg void OnBnClickedChannel1();
+ afx_msg void OnBnClickedChannel2();
+ afx_msg void OnBnClickedChannel3();
+ afx_msg void OnBnClickedChannel4();
+
+
+ VideoEngine* _videoEngine;
+ VoiceEngine* _voiceEngine;
+ VoEBase* _veBase;
+
+ CDXChannelDlg* _testDlg1;
+ CDXChannelDlg* _testDlg2;
+ CDXChannelDlg* _testDlg3;
+ CDXChannelDlg* _testDlg4;
+
+ int _externalInWidth;
+ int _externalInHeight;
+ int _externalInVideoType;
+
+ CaptureDevicePool _captureDevicePool;
+ ChannelPool _channelPool;
+
+
+private:
+ virtual void ChannelDialogEnded(CDXChannelDlg* context);
+
+public:
+
+};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc
new file mode 100644
index 0000000..5e866ad
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc
@@ -0,0 +1,101 @@
+// Microsoft Visual C++ generated resource script.
+//
+#include "WindowsTestResource.h"
+
+#define APSTUDIO_READONLY_SYMBOLS
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 2 resource.
+//
+#include "afxres.h"
+
+/////////////////////////////////////////////////////////////////////////////
+#undef APSTUDIO_READONLY_SYMBOLS
+
+/////////////////////////////////////////////////////////////////////////////
+// Swedish resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
+#ifdef _WIN32
+LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
+#pragma code_page(1252)
+#endif //_WIN32
+
+#ifdef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// TEXTINCLUDE
+//
+
+1 TEXTINCLUDE
+BEGIN
+ "WindowsTestResource.h\0"
+END
+
+2 TEXTINCLUDE
+BEGIN
+ "#include ""afxres.h""\r\n"
+ "\0"
+END
+
+3 TEXTINCLUDE
+BEGIN
+ "\r\n"
+ "\0"
+END
+
+#endif // APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Dialog
+//
+
+IDD_WINDOWSTEST_MAIN DIALOGEX 0, 0, 186, 156
+STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
+CAPTION "Windows ViE Test"
+FONT 8, "MS Shell Dlg", 400, 0, 0x1
+BEGIN
+ DEFPUSHBUTTON "OK",IDOK,129,7,50,14
+ PUSHBUTTON "Cancel",IDCANCEL,129,24,50,14
+ PUSHBUTTON "Channel 1",IDC_CHANNEL1,129,45,50,14
+ PUSHBUTTON "Channel 2",IDC_CHANNEL2,129,62,50,14
+ PUSHBUTTON "Channel 3",IDC_CHANNEL3,129,79,50,14
+ PUSHBUTTON "Channel 4",IDC_CHANNEL4,129,96,50,14
+END
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// DESIGNINFO
+//
+
+#ifdef APSTUDIO_INVOKED
+GUIDELINES DESIGNINFO
+BEGIN
+ IDD_WINDOWSTEST_MAIN, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 179
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 149
+ END
+END
+#endif // APSTUDIO_INVOKED
+
+#endif // Swedish resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+
+#ifndef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 3 resource.
+//
+
+
+/////////////////////////////////////////////////////////////////////////////
+#endif // not APSTUDIO_INVOKED
+
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h
new file mode 100644
index 0000000..2d49c28
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h
@@ -0,0 +1,28 @@
+//{{NO_DEPENDENCIES}}
+// Microsoft Visual C++ generated include file.
+// Used by WindowsTestResouce.rc
+//
+#define IDD_WINDOWSTEST_MAIN 101
+#define IDC_CHANNEL1 1001
+#define IDC_CHANNEL2 1002
+#define IDC_CHANNEL3 1004
+#define IDC_CHANNEL4 1005
+#define IDC_POSITION 1009
+#define IDC_INFORMATION 1050
+#define IDC_CHECK_CHANNEL1 1070
+#define IDC_CHECK_CHANNEL2 1071
+#define IDC_CHECK_CHANNEL3 1072
+#define IDC_CHECK_CHANNEL4 1073
+#define IDC_COMBO1 1074
+#define IDC_BTN_CREATE2 1076
+
+// Next default values for new objects
+//
+#ifdef APSTUDIO_INVOKED
+#ifndef APSTUDIO_READONLY_SYMBOLS
+#define _APS_NEXT_RESOURCE_VALUE 106
+#define _APS_NEXT_COMMAND_VALUE 40001
+#define _APS_NEXT_CONTROL_VALUE 1076
+#define _APS_NEXT_SYMED_VALUE 107
+#endif
+#endif
diff --git a/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg
new file mode 100644
index 0000000..3bb3ba4
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg
new file mode 100644
index 0000000..b10a842
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg
new file mode 100644
index 0000000..cb34d67
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2 b/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2
new file mode 100644
index 0000000..d9acfd2
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2
@@ -0,0 +1,13 @@
+//
+// DXCAPTURE.RC2 - resources Microsoft Visual C++ does not edit directly
+//
+
+#ifdef APSTUDIO_INVOKED
+ #error this file is not editable by Microsoft Visual C++
+#endif //APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+// Add manually edited resources here...
+
+/////////////////////////////////////////////////////////////////////////////
diff --git a/trunk/src/video_engine/main/test/WindowsTest/resource.h b/trunk/src/video_engine/main/test/WindowsTest/resource.h
new file mode 100644
index 0000000..34e90eb
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/resource.h
@@ -0,0 +1,58 @@
+//{{NO_DEPENDENCIES}}
+// Microsoft Visual C++ generated include file.
+// Used by Capture.rc
+//
+#define IDM_ABOUTBOX 0x0010
+#define IDD_DXQUALITY_DIALOG 102
+#define IDI_ICON1 130
+#define IDD_SLAVE_CHANNEL 132
+#define IDC_LIVEVIDEO 1000
+#define IDC_CAPTURE 1001
+#define IDC_DEVICE 1003
+#define IDC_STARTSEND 1004
+#define IDC_STARTLISTEN 1006
+#define IDC_STOPLISTEN 1007
+#define IDC_STOPSend 1008
+#define IDC_CODEC_LIST 1010
+#define IDC_CODEC_SIZE 1011
+#define IDC_IPADDRESS1 1012
+#define IDC_LOCAL_PORT1 1013
+#define IDC_CHANGE_SIZE 1017
+#define IDC_QUALITY 1018
+#define IDC_BITRATE 1019
+#define IDC_WINDOW_SIZE 1022
+#define IDC_REMOTE_PORT1 1025
+#define IDC_START_REC 1030
+#define IDC_STOP_REC 1031
+#define IDC_CAM_PROPERTY 1033
+#define IDC_ONMODE 1034
+#define IDC_CAPTURECAP 1038
+#define IDC_RADIO1 1039
+#define IDC_MIN_FRAME_RATE 1040
+#define IDC_RTCPMODE 1042
+#define IDC_TMMBR 1043
+#define IDC_PACKETBURST 1044
+#define IDC_PROT_NONE 1045
+#define IDC_PROT_NACK 1046
+#define IDC_PROT_FEC 1047
+#define IDC_PROT_NACKFEC 1048
+#define IDC_INFORMATION 1050
+#define IDC_PACKETLOSS 1051
+#define IDC_FREEZELOG 1052
+#define IDC_VERSION 1053
+#define IDC_EXTTRANSPORT 1054
+#define IDC_DELAY 1055
+#define IDC_BTN_RECORD_INCOMING 1056
+#define IDC_BTN_RECORD_OUTGOING 1057
+#define IDC_BTN_CREATE_SLAVE 1058
+
+// Next default values for new objects
+//
+#ifdef APSTUDIO_INVOKED
+#ifndef APSTUDIO_READONLY_SYMBOLS
+#define _APS_NEXT_RESOURCE_VALUE 133
+#define _APS_NEXT_COMMAND_VALUE 32771
+#define _APS_NEXT_CONTROL_VALUE 1059
+#define _APS_NEXT_SYMED_VALUE 101
+#endif
+#endif
diff --git a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc
new file mode 100644
index 0000000..f0910ee
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc
@@ -0,0 +1,317 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// tbExternalTransport.cpp
+//
+
+#include "tbExternalTransport.h"
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "vie_network.h"
+#include "tick_util.h"
+
+using namespace webrtc;
+
+TbExternalTransport::TbExternalTransport(ViENetwork& vieNetwork)
+ :
+ _vieNetwork(vieNetwork),
+ _thread(*ThreadWrapper::CreateThread(ViEExternalTransportRun, this, kHighPriority, "AutotestTransport")),
+ _event(*EventWrapper::Create()),
+ _crit(*CriticalSectionWrapper::CreateCriticalSection()),
+ _statCrit(*CriticalSectionWrapper::CreateCriticalSection()),
+ _lossRate(0),
+ _networkDelayMs(0),
+ _rtpCount(0),
+ _dropCount(0),
+ _rtcpCount(0),
+ _rtpPackets(),
+ _rtcpPackets(),
+ _checkSSRC(false),
+ _lastSSRC(0),
+ _checkSequenceNumber(0),
+ _firstSequenceNumber(0),
+ _lastSeq(0)
+{
+ srand((int)TickTime::MicrosecondTimestamp());
+ unsigned int tId = 0;
+ _thread.Start(tId);
+}
+
+
+TbExternalTransport::~TbExternalTransport()
+{
+ // TODO: stop thread
+ _thread.SetNotAlive();
+ _event.Set();
+ if (_thread.Stop())
+ {
+ delete &_thread;
+ delete &_event;
+ }
+ delete &_crit;
+ delete &_statCrit;
+}
+
+
+
+
+
+int TbExternalTransport::SendPacket(int channel, const void *data, int len)
+{
+ _statCrit.Enter();
+ _rtpCount++;
+ _statCrit.Leave();
+
+
+ unsigned short sequenceNumber = (((unsigned char*) data)[2]) << 8;
+ sequenceNumber += (((unsigned char*) data)[3]);
+
+
+ int marker=((unsigned char*)data)[1] & 0x80;
+ unsigned int timestamp=((((unsigned char*)data)[4]) << 24) + ((((unsigned char*)data)[5])<<16) +((((unsigned char*)data)[6])<<8)+(((unsigned char*)data)[7]);
+
+
+ // Packet loss
+ int dropThis = rand() % 100;
+ bool nacked=false;
+ if(sequenceNumber<_lastSeq)
+ {
+ nacked=true;
+ }
+ else
+ {
+ _lastSeq=sequenceNumber;
+ }
+
+ if (dropThis < _lossRate)
+ {
+ _statCrit.Enter();
+ _dropCount++;
+ _statCrit.Leave();
+
+
+ /* char str[256];
+ sprintf(str,"Dropping seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
+ OutputDebugString(str);*/
+
+ return len;
+ }
+ else
+ {
+ if(nacked)
+ {
+ /*char str[256];
+ sprintf(str,"Resending seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
+ OutputDebugString(str);*/
+ }
+ else
+ {
+ /*char str[256];
+ sprintf(str,"Sending seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
+ OutputDebugString(str);*/
+
+ }
+ }
+
+
+ VideoPacket* newPacket = new VideoPacket();
+ memcpy(newPacket->packetBuffer, data, len);
+ newPacket->length = len;
+ newPacket->channel = channel;
+
+ _crit.Enter();
+ newPacket->receiveTime = NowMs() + _networkDelayMs;
+ _rtpPackets.push(newPacket);
+ _event.Set();
+ _crit.Leave();
+ return len;
+}
+
+int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
+{
+ _statCrit.Enter();
+ _rtcpCount++;
+ _statCrit.Leave();
+
+ VideoPacket* newPacket = new VideoPacket();
+ memcpy(newPacket->packetBuffer, data, len);
+ newPacket->length = len;
+ newPacket->channel = channel;
+
+ _crit.Enter();
+ newPacket->receiveTime = NowMs() + _networkDelayMs;
+ _rtcpPackets.push(newPacket);
+ _event.Set();
+ _crit.Leave();
+ return len;
+}
+
+WebRtc_Word32 TbExternalTransport::SetPacketLoss(WebRtc_Word32 lossRate)
+{
+ CriticalSectionScoped cs(_statCrit);
+ _lossRate = lossRate;
+ return 0;
+}
+
+void TbExternalTransport::SetNetworkDelay(WebRtc_Word64 delayMs)
+{
+ CriticalSectionScoped cs(_crit);
+ _networkDelayMs = delayMs;
+ return;
+}
+
+void TbExternalTransport::ClearStats()
+{
+ CriticalSectionScoped cs(_statCrit);
+ _rtpCount = 0;
+ _dropCount = 0;
+ _rtcpCount = 0;
+ return;
+}
+
+void TbExternalTransport::GetStats(WebRtc_Word32& numRtpPackets, WebRtc_Word32& numDroppedPackets, WebRtc_Word32& numRtcpPackets)
+{
+ CriticalSectionScoped cs(_statCrit);
+ numRtpPackets = _rtpCount;
+ numDroppedPackets = _dropCount;
+ numRtcpPackets = _rtcpCount;
+ return;
+}
+
+void TbExternalTransport::EnableSSRCCheck()
+{
+ CriticalSectionScoped cs(_statCrit);
+ _checkSSRC = true;
+}
+unsigned int TbExternalTransport::ReceivedSSRC()
+{
+ CriticalSectionScoped cs(_statCrit);
+ return _lastSSRC;
+}
+
+void TbExternalTransport::EnableSequenceNumberCheck()
+{
+ CriticalSectionScoped cs(_statCrit);
+ _checkSequenceNumber = true;
+}
+
+unsigned short TbExternalTransport::GetFirstSequenceNumber()
+{
+ CriticalSectionScoped cs(_statCrit);
+ return _firstSequenceNumber;
+}
+
+
+bool TbExternalTransport::ViEExternalTransportRun(void* object)
+{
+ return static_cast<TbExternalTransport*>(object)->ViEExternalTransportProcess();
+}
+bool TbExternalTransport::ViEExternalTransportProcess()
+{
+ unsigned int waitTime = KMaxWaitTimeMs;
+
+ VideoPacket* packet = NULL;
+
+ while (!_rtpPackets.empty())
+ {
+ // Take first packet in queue
+ _crit.Enter();
+ packet = _rtpPackets.front();
+ WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
+ if (timeToReceive > 0)
+ {
+ // No packets to receive yet
+ if (timeToReceive < waitTime &&
+ timeToReceive > 0)
+ {
+ waitTime = (unsigned int) timeToReceive;
+ }
+ _crit.Leave();
+ break;
+ }
+ _rtpPackets.pop();
+ _crit.Leave();
+
+ // Send to ViE
+ if (packet)
+ {
+ {
+ CriticalSectionScoped cs(_statCrit);
+ if (_checkSSRC)
+ {
+ _lastSSRC = ((packet->packetBuffer[8]) << 24);
+ _lastSSRC += (packet->packetBuffer[9] << 16);
+ _lastSSRC += (packet->packetBuffer[10] << 8);
+ _lastSSRC += packet->packetBuffer[11];
+ _checkSSRC = false;
+ }
+ if (_checkSequenceNumber)
+ {
+ _firstSequenceNumber = (unsigned char)packet->packetBuffer[2] << 8;
+ _firstSequenceNumber += (unsigned char)packet->packetBuffer[3];
+ _checkSequenceNumber = false;
+ }
+ }
+ /*
+ unsigned short sequenceNumber = (unsigned char)packet->packetBuffer[2] << 8;
+ sequenceNumber += (unsigned char)packet->packetBuffer[3];
+
+ int marker=packet->packetBuffer[1] & 0x80;
+ unsigned int timestamp=((((unsigned char*)packet->packetBuffer)[4]) << 24) + ((((unsigned char*)packet->packetBuffer)[5])<<16) +((((unsigned char*)packet->packetBuffer)[6])<<8)+(((unsigned char*)packet->packetBuffer)[7]);
+ char str[256];
+ sprintf(str,"Receiving seq %u length %d m %d, ts %u\n", sequenceNumber,packet->length,marker,timestamp) ;
+ OutputDebugString(str);*/
+
+ _vieNetwork.ReceivedRTPPacket(packet->channel, packet->packetBuffer, packet->length);
+ delete packet;
+ packet = NULL;
+ }
+ }
+ while (!_rtcpPackets.empty())
+ {
+ // Take first packet in queue
+ _crit.Enter();
+ packet = _rtcpPackets.front();
+ WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
+ if (timeToReceive > 0)
+ {
+ // No packets to receive yet
+ if (timeToReceive < waitTime &&
+ timeToReceive > 0)
+ {
+ waitTime = (unsigned int) timeToReceive;
+ }
+ _crit.Leave();
+ break;
+ }
+ packet = _rtcpPackets.front();
+ _rtcpPackets.pop();
+ _crit.Leave();
+
+ // Send to ViE
+ if (packet)
+ {
+ _vieNetwork.ReceivedRTCPPacket(packet->channel, packet->packetBuffer, packet->length);
+ delete packet;
+ packet = NULL;
+ }
+ }
+ _event.Wait(waitTime + 1); // Add 1 ms to not call to early...
+ return true;
+}
+
+WebRtc_Word64 TbExternalTransport::NowMs()
+{
+ return TickTime::MillisecondTimestamp();
+}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h
new file mode 100644
index 0000000..53226c6
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// tbExternalTransport.h
+//
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
+
+#include "common_types.h"
+#include <queue>
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class ViENetwork;
+}
+
+class TbExternalTransport : public webrtc::Transport
+{
+public:
+ TbExternalTransport(webrtc::ViENetwork& vieNetwork);
+ ~TbExternalTransport(void);
+
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+
+ WebRtc_Word32 SetPacketLoss(WebRtc_Word32 lossRate); // Rate in %
+ void SetNetworkDelay(WebRtc_Word64 delayMs);
+
+ void ClearStats();
+ void GetStats(WebRtc_Word32& numRtpPackets, WebRtc_Word32& numDroppedPackets, WebRtc_Word32& numRtcpPackets);
+
+ void EnableSSRCCheck();
+ unsigned int ReceivedSSRC();
+
+ void EnableSequenceNumberCheck();
+ unsigned short GetFirstSequenceNumber();
+
+
+protected:
+ static bool ViEExternalTransportRun(void* object);
+ bool ViEExternalTransportProcess();
+private:
+ WebRtc_Word64 NowMs();
+
+ enum { KMaxPacketSize = 1650};
+ enum { KMaxWaitTimeMs = 100};
+ typedef struct
+ {
+ WebRtc_Word8 packetBuffer[KMaxPacketSize];
+ WebRtc_Word32 length;
+ WebRtc_Word32 channel;
+ WebRtc_Word64 receiveTime;
+ } VideoPacket;
+
+ typedef std::queue<VideoPacket*> VideoPacketQueue;
+
+
+ webrtc::ViENetwork& _vieNetwork;
+ webrtc::ThreadWrapper& _thread;
+ webrtc::EventWrapper& _event;
+ webrtc::CriticalSectionWrapper& _crit;
+ webrtc::CriticalSectionWrapper& _statCrit;
+
+ WebRtc_Word32 _lossRate;
+ WebRtc_Word64 _networkDelayMs;
+ WebRtc_Word32 _rtpCount;
+ WebRtc_Word32 _rtcpCount;
+ WebRtc_Word32 _dropCount;
+
+ VideoPacketQueue _rtpPackets;
+ VideoPacketQueue _rtcpPackets;
+
+ bool _checkSSRC;
+ WebRtc_UWord32 _lastSSRC;
+ bool _checkSequenceNumber;
+ WebRtc_UWord16 _firstSequenceNumber;
+ WebRtc_Word32 _lastSeq;
+
+ //int& numberOfErrors;
+
+ //int _bits;
+ //int _lastTicks;
+ //int _dropCnt;
+ //int _sentCount;
+ //int _frameCount;
+ //int _packetLoss;
+
+ //VideoEngine* _video;
+
+ //ReceiveBufferQueue _videoBufferQueue;
+ //ReceiveBufferQueue _rtcpBufferQueue;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/videosize.cc b/trunk/src/video_engine/main/test/WindowsTest/videosize.cc
new file mode 100644
index 0000000..a675ec5
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/videosize.cc
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "VideoSize.h"
+int GetWidthHeight( VideoSize size, int& width, int& height)
+{
+ switch(size)
+ {
+ case SQCIF:
+ width = 128;
+ height = 96;
+ return 0;
+ case QQVGA:
+ width = 160;
+ height = 120;
+ return 0;
+ case QCIF:
+ width = 176;
+ height = 144;
+ return 0;
+ case CGA:
+ width = 320;
+ height = 200;
+ return 0;
+ case QVGA:
+ width = 320;
+ height = 240;
+ return 0;
+ case SIF:
+ width = 352;
+ height = 240;
+ return 0;
+ case WQVGA:
+ width = 400;
+ height = 240;
+ return 0;
+ case CIF:
+ width = 352;
+ height = 288;
+ return 0;
+ case W288P:
+ width = 512;
+ height = 288;
+ return 0;
+ case W368P:
+ width = 640;
+ height = 368;
+ return 0;
+ case S_448P:
+ width = 576;
+ height = 448;
+ return 0;
+ case VGA:
+ width = 640;
+ height = 480;
+ return 0;
+ case S_432P:
+ width = 720;
+ height = 432;
+ return 0;
+ case W432P:
+ width = 768;
+ height = 432;
+ return 0;
+ case S_4SIF:
+ width = 704;
+ height = 480;
+ return 0;
+ case W448P:
+ width = 768;
+ height = 448;
+ return 0;
+ case NTSC:
+ width = 720;
+ height = 480;
+ return 0;
+ case FW448P:
+ width = 800;
+ height = 448;
+ return 0;
+ case S_768x480P:
+ width = 768;
+ height = 480;
+ return 0;
+ case WVGA:
+ width = 800;
+ height = 480;
+ return 0;
+ case S_4CIF:
+ width = 704;
+ height = 576;
+ return 0;
+ case SVGA:
+ width = 800;
+ height = 600;
+ return 0;
+ case W544P:
+ width = 960;
+ height = 544;
+ return 0;
+ case W576P:
+ width = 1024;
+ height = 576;
+ return 0;
+ case HD:
+ width = 960;
+ height = 720;
+ return 0;
+ case XGA:
+ width = 1024;
+ height = 768;
+ return 0;
+ case FULL_HD:
+ width = 1440;
+ height = 1080;
+ return 0;
+ case WHD:
+ width = 1280;
+ height = 720;
+ return 0;
+ case UXGA:
+ width = 1600;
+ height = 1200;
+ return 0;
+ case WFULL_HD:
+ width = 1920;
+ height = 1080;
+ return 0;
+ default:
+ return -1;
+ }
+ return -1;
+}
\ No newline at end of file
diff --git a/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi b/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi
new file mode 100644
index 0000000..0bebd98
--- /dev/null
+++ b/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi
@@ -0,0 +1,84 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'conditions': [
+ ['OS=="win"', {
+ 'targets': [
+ # WinTest - GUI test for Windows
+ {
+ 'target_name': 'vie_win_test',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/modules/modules.gyp:video_render_module',
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ## VoiceEngine
+ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
+ ## VideoEngine
+ 'video_engine_core',
+ ],
+ 'include_dirs': [
+ './interface',
+ '../../../../', # common_types.h and typedefs.h
+ '../commonTestClasses/'
+ ],
+ 'sources': [
+ 'Capture.rc',
+ 'captureDeviceImage.jpg',
+ 'ChannelDlg.cc',
+ 'ChannelDlg.h',
+ 'ChannelPool.cc',
+ 'ChannelPool.h',
+ 'renderStartImage.jpg',
+ 'renderTimeoutImage.jpg',
+ 'res\Capture.rc2',
+ 'resource.h',
+ 'StdAfx.h',
+ 'videosize.cc',
+ 'VideoSize.h',
+ 'WindowsTest.cc',
+ 'WindowsTest.h',
+ 'WindowsTestMainDlg.cc',
+ 'WindowsTestMainDlg.h',
+ 'WindowsTestResouce.rc',
+ 'WindowsTestResource.h',
+ 'tbExternalTransport.cc',
+ 'CaptureDevicePool.cc',
+ 'tbExternalTransport.h',
+ 'CaptureDevicePool.h',
+
+ ],
+ 'configurations': {
+ 'Common_Base': {
+ 'msvs_configuration_attributes': {
+ 'conditions': [
+ ['component=="shared_library"', {
+ 'UseOfMFC': '2', # Shared DLL
+ },{
+ 'UseOfMFC': '1', # Static
+ }],
+ ],
+ },
+ },
+ },
+ 'msvs_settings': {
+ 'VCLinkerTool': {
+ 'SubSystem': '2', # Windows
+ },
+ },
+ },
+ ],
+ }],
+ ],
+}
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/main/test/android_test/.classpath b/trunk/src/video_engine/main/test/android_test/.classpath
new file mode 100644
index 0000000..f2adf55
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/.classpath
@@ -0,0 +1,9 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+ <classpathentry kind="src" path="src"/>
+ <classpathentry kind="src" path="gen"/>
+ <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+ <classpathentry kind="lib" path="libs/VideoEngine_android_java.jar"/>
+ <classpathentry kind="lib" path="libs/VoiceEngine_android_java.jar"/>
+ <classpathentry kind="output" path="bin"/>
+</classpath>
diff --git a/trunk/src/video_engine/main/test/android_test/Android.mk b/trunk/src/video_engine/main/test/android_test/Android.mk
new file mode 100644
index 0000000..3ee8627
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/Android.mk
@@ -0,0 +1,41 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+MY_CAPTURE_FOLDER := ../../../../modules/video_capture/main/source
+MY_CAPTURE_JAVA_FOLDER := Android/java/org/webrtc/videoengine
+MY_CAPTURE_PATH := $(MY_CAPTURE_FOLDER)/$(MY_CAPTURE_JAVA_FOLDER)
+
+MY_RENDER_FOLDER := ../../../../modules/video_render/main/source
+MY_RENDER_JAVA_FOLDER := Android/java/org/webrtc/videoengine
+MY_RENDER_PATH := $(MY_RENDER_FOLDER)/$(MY_RENDER_JAVA_FOLDER)
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ src/org/webrtc/videoengineapp/ViEAndroidDemo.java \
+ src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java \
+ src/org/webrtc/videoengineapp/IViEAndroidCallback.java \
+ $(MY_CAPTURE_PATH)/CaptureCapabilityAndroid.java \
+ $(MY_CAPTURE_PATH)/VideoCaptureAndroid.java \
+ $(MY_CAPTURE_PATH)/VideoCaptureDeviceInfoAndroid.java \
+ $(MY_RENDER_PATH)/ViEAndroidGLES20.java \
+ $(MY_RENDER_PATH)/ViERenderer.java \
+ $(MY_RENDER_PATH)/ViESurfaceRenderer.java
+
+LOCAL_PACKAGE_NAME := webrtc-video-demo
+LOCAL_CERTIFICATE := platform
+
+LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-video-demo-jni
+
+include $(BUILD_PACKAGE)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml b/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml
new file mode 100644
index 0000000..0714aed
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml
@@ -0,0 +1,28 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ android:versionCode="1" package="org.webrtc.videoengineapp" android:versionName="1.07">
+ <application android:icon="@drawable/logo"
+ android:label="@string/app_name"
+ android:debuggable="true">
+ <activity android:name=".ViEAndroidDemo"
+ android:label="@string/app_name"
+ android:configChanges="keyboardHidden|orientation"
+ >
+ <!--android:configChanges="keyboardHidden|orientation" -->
+ <!-- android:screenOrientation="landscape" -->
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+ </application>
+ <uses-sdk android:minSdkVersion="7" />
+ <uses-permission android:name="android.permission.CAMERA"></uses-permission>
+ <uses-feature android:name="android.hardware.camera" />
+ <uses-feature android:name="android.hardware.camera.autofocus" />
+ <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+ <uses-permission android:name="android.permission.RECORD_AUDIO" />
+ <uses-permission android:name="android.permission.INTERNET" />
+ <uses-permission android:name="android.permission.WAKE_LOCK" />
+ <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+</manifest>
diff --git a/trunk/src/video_engine/main/test/android_test/default.properties b/trunk/src/video_engine/main/test/android_test/default.properties
new file mode 100644
index 0000000..c206f58
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/default.properties
@@ -0,0 +1,13 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Indicates whether an apk should be generated for each density.
+split.density=false
+# Project target.
+target=android-9
diff --git a/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java b/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java
new file mode 100644
index 0000000..c35047b
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java
@@ -0,0 +1,68 @@
+/* AUTO-GENERATED FILE. DO NOT MODIFY.
+ *
+ * This class was automatically generated by the
+ * aapt tool from the resource data it found. It
+ * should not be modified by hand.
+ */
+
+package org.webrtc.videoengineapp;
+
+public final class R {
+ public static final class array {
+ public static final int codecSize=0x7f040001;
+ public static final int codectype=0x7f040000;
+ public static final int voiceCodecType=0x7f040002;
+ }
+ public static final class attr {
+ }
+ public static final class drawable {
+ public static final int bar=0x7f020000;
+ public static final int bg=0x7f020001;
+ public static final int logo=0x7f020002;
+ public static final int robot=0x7f020003;
+ public static final int video=0x7f020004;
+ }
+ public static final class id {
+ public static final int LinearLayout01=0x7f060010;
+ public static final int LinearLayout02=0x7f060006;
+ public static final int TextView01=0x7f060005;
+ public static final int TextView02=0x7f06000b;
+ public static final int TextView03=0x7f060004;
+ public static final int btStartBoth=0x7f060013;
+ public static final int btStartListen=0x7f060011;
+ public static final int btStartSend=0x7f060012;
+ public static final int cbLoopback=0x7f06000e;
+ public static final int cbVoice=0x7f06000d;
+ public static final int etRemoteIp=0x7f06000c;
+ public static final int ivPreview=0x7f060014;
+ public static final int ivTopBar=0x7f060002;
+ public static final int rlSurfaces=0x7f060000;
+ public static final int spCodecSize=0x7f06000a;
+ public static final int spCodecType=0x7f060007;
+ public static final int spVoiceCodecType=0x7f060008;
+ public static final int svLocal=0x7f060001;
+ public static final int tvCodecSize=0x7f060009;
+ public static final int tvLocalIp=0x7f06000f;
+ public static final int tvTitle=0x7f060003;
+ }
+ public static final class layout {
+ public static final int both=0x7f030000;
+ public static final int main=0x7f030001;
+ public static final int send=0x7f030002;
+ }
+ public static final class string {
+ public static final int app_name=0x7f050001;
+ public static final int codecSize=0x7f050007;
+ public static final int codecType=0x7f050006;
+ public static final int codectype_prompt=0x7f050004;
+ public static final int demoTitle=0x7f050005;
+ public static final int enableVoice=0x7f05000d;
+ public static final int error=0x7f050002;
+ public static final int errorCamera=0x7f050003;
+ public static final int loopback=0x7f050009;
+ public static final int remoteIp=0x7f050008;
+ public static final int startBoth=0x7f05000c;
+ public static final int startListen=0x7f05000a;
+ public static final int startSend=0x7f05000b;
+ }
+}
diff --git a/trunk/src/video_engine/main/test/android_test/jni/Android.mk b/trunk/src/video_engine/main/test/android_test/jni/Android.mk
new file mode 100644
index 0000000..4121cc6
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/jni/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../../build/interface
+LIBS_PATH := $(LOCAL_PATH)/../../../../../../../build/libraries
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_MODULE := libwebrtc-video-demo-jni
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := vie_android_java_api.cc
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_ANDROID'
+
+LOCAL_C_INCLUDES := \
+ external/gtest/include \
+ $(LOCAL_PATH)/../../../../.. \
+ $(LOCAL_PATH)/../../../../include \
+ $(LOCAL_PATH)/../../../../../voice_engine/main/interface
+
+LOCAL_PRELINK_MODULE := false
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libandroid \
+ libwebrtc \
+ libGLESv2
+LOCAL_LDLIBS := $(LIBS_PATH)/VideoEngine_android_gcc.a -llog -lgcc
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h b/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h
new file mode 100644
index 0000000..37a9c43
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h
@@ -0,0 +1,494 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* DO NOT EDIT THIS FILE - it is machine generated */
+#include <jni.h>
+/* Header for class org_webrtc_videoengineapp_ViEAndroidJavaAPI */
+
+#ifndef _Included_org_webrtc_videoengineapp_ViEAndroidJavaAPI
+#define _Included_org_webrtc_videoengineapp_ViEAndroidJavaAPI
+#ifdef __cplusplus
+extern "C" {
+#endif
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: NativeInit
+ * Signature: (Landroid/content/Context;)Z
+ */
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_NativeInit
+(JNIEnv *, jobject, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: GetVideoEngine
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetVideoEngine
+(JNIEnv *, jobject);
+
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: Init
+ * Signature: (IIIZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Init
+(JNIEnv *, jobject, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: Terminate
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Terminate
+(JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartSend
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSend
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StopRender
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopRender
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StopSend
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopSend
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartReceive
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartReceive
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StopReceive
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopReceive
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: CreateChannel
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_CreateChannel
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetLocalReceiver
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetLocalReceiver
+(JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetSendDestination
+ * Signature: (II[B)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendDestination
+(JNIEnv *, jobject, jint, jint, jbyteArray);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetReceiveCodec
+ * Signature: (IIIIIIZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetReceiveCodec
+(JNIEnv *, jobject, jint, jint, jint, jint, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetSendCodec
+ * Signature: (IIIIIIZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendCodec
+(JNIEnv *, jobject, jint, jint, jint, jint, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: AddRemoteRenderer
+ * Signature: (ILandroid/view/SurfaceView;)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_AddRemoteRenderer
+(JNIEnv *, jobject, jint, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: RemoveRemoteRenderer
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_RemoveRemoteRenderer
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartRender
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartRender
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartCamera
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartCamera
+(JNIEnv *, jobject,jint channel,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StopCamera
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopCamera
+(JNIEnv *, jobject,jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: GetCameraOrientation
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetCameraOrientation
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetRotation
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetRotation
+(JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: EnableNACK
+ * Signature: (IZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnableNACK
+(JNIEnv *, jobject, jint, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: EnablePLI
+ * Signature: (IZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnablePLI
+(JNIEnv *, jobject, jint, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartSendNative
+ * Signature: (III[BIIIIIILandroid/view/SurfaceView;III)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSendNative
+(JNIEnv *, jobject, jint, jint, jint, jbyteArray, jint, jint,
+jint, jint, jint, jint, jobject, jint, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StartListenNative
+ * Signature: (III[BIIIIIIILandroid/view/SurfaceView;)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartListenNative
+(JNIEnv *, jobject, jint, jint, jint, jbyteArray, jint, jint,
+jint, jint, jint, jint, jint, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: StopAllNative
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopAllNative
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: SetCallback
+ * Signature: (ILorg/webrtc/videoengineapp/IViEAndroidCallback;)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallback
+(JNIEnv *, jobject, jint, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_Create
+ * Signature: (Landroid/app/Activity;)Z
+ */
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Create
+(JNIEnv *, jobject, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_Delete
+ * Signature: ()Z
+ */
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Delete
+(JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_Init
+ * Signature: (IIIZZ)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Init
+(JNIEnv *, jobject, jint, jint, jint, jboolean, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_Terminate
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Terminate
+(JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_CreateChannel
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_CreateChannel
+(JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_DeleteChannel
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_DeleteChannel
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetLocalReceiver
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLocalReceiver
+(JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetSendDestination
+ * Signature: (IILjava/lang/String;)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendDestination
+(JNIEnv *, jobject, jint, jint, jstring);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StartListen
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartListen
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StartPlayout
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayout
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StartSend
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartSend
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StopListen
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopListen
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StopPlayout
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayout
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StopSend
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopSend
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetSpeakerVolume
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSpeakerVolume
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetLoudspeakerStatus
+ * Signature: (Z)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLoudspeakerStatus
+(JNIEnv *, jobject, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StartPlayingFileLocally
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileLocally
+(JNIEnv *, jobject, jint, jstring, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StopPlayingFileLocally
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileLocally
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StartPlayingFileAsMicrophone
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileAsMicrophone
+(JNIEnv *, jobject, jint, jstring, jboolean);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_StopPlayingFileAsMicrophone
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileAsMicrophone
+(JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_NumOfCodecs
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_NumOfCodecs
+(JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetSendCodec
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendCodec
+(JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetECStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetECStatus
+(JNIEnv *, jobject, jboolean, jint, jint, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetNSStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetNSStatus
+(JNIEnv *, jobject, jboolean, jint);
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: VE_SetAGCStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetAGCStatus
+(JNIEnv *, jobject, jboolean, jint);
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc b/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc
new file mode 100644
index 0000000..112667f
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc
@@ -0,0 +1,1563 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <android/log.h>
+
+#include "org_webrtc_videoengineapp_vie_android_java_api.h"
+
+#include "voe_base.h"
+#include "voe_codec.h"
+#include "voe_file.h"
+#include "voe_network.h"
+#include "voe_audio_processing.h"
+#include "voe_volume_control.h"
+#include "voe_hardware.h"
+
+#include "vie_base.h"
+#include "vie_codec.h"
+#include "vie_capture.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+
+#include "common_types.h"
+
+#define WEBRTC_LOG_TAG "*WEBRTCN*"
+#define VALIDATE_BASE_POINTER \
+ if (!veData.base) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Base pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_CODEC_POINTER \
+ if (!veData.codec) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Codec pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_FILE_POINTER \
+ if (!veData.file) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "File pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_APM_POINTER \
+ if (!veData.codec) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Apm pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_HARDWARE_POINTER \
+ if (!veData.hardware) \
+ { \
+ __android_log_write( \
+ ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Hardware pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_VOLUME_POINTER \
+ if (!veData.volume) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Volume pointer doesn't exist"); \
+ return -1; \
+ }
+
+using namespace webrtc;
+
+//Forward declaration.
+class VideoCallbackAndroid;
+
+// VoiceEngine data struct
+typedef struct
+{
+ // VoiceEngine
+ VoiceEngine* ve;
+ // Sub-APIs
+ VoEBase* base;
+ VoECodec* codec;
+ VoEFile* file;
+ VoENetwork* netw;
+ VoEAudioProcessing* apm;
+ VoEVolumeControl* volume;
+ VoEHardware* hardware;
+ JavaVM* jvm;
+} VoiceEngineData;
+
+class AndroidVideoRenderCallback;
+// VideoEngine data struct
+typedef struct
+{
+ VideoEngine* vie;
+ ViEBase* base;
+ ViECodec* codec;
+ ViENetwork* netw;
+ ViERTP_RTCP* rtp;
+ ViERender* render;
+ ViECapture* capture;
+ VideoCallbackAndroid* callback;
+
+} VideoEngineData;
+
+// Global variables
+JavaVM* webrtcGlobalVM;
+
+// Global variables visible in this file
+static VoiceEngineData veData;
+static VideoEngineData vieData;
+
+// "Local" functions (i.e. not Java accessible)
+#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 1024
+static bool VE_GetSubApis();
+static bool VE_ReleaseSubApis();
+
+#define CHECK_API_RETURN(ret) \
+ if (ret!=0) \
+ { \
+ __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Return error %d",ret); \
+ break; \
+ }
+
+class VideoCallbackAndroid: public ViEDecoderObserver,
+ public ViEEncoderObserver
+{
+
+ // Implements ViEDecoderObserver
+ virtual void IncomingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate)
+ {
+ // Let's print out the network statistics from this call back as well
+ unsigned short fraction_lost;
+ unsigned int dummy;
+ int intdummy;
+ _vieData.rtp->GetReceivedRTCPStatistics(videoChannel, fraction_lost,
+ dummy, dummy, dummy, intdummy);
+ unsigned short packetLossRate = 0;
+ if (fraction_lost > 0)
+ {
+ // Change from frac to %
+ packetLossRate = (fraction_lost * 100) >> 8;
+ }
+
+ JNIEnv* threadEnv = NULL;
+ int ret = webrtcGlobalVM->AttachCurrentThread(&threadEnv, NULL);
+ // Get the JNI env for this thread
+ if ((ret < 0) || !threadEnv)
+ {
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "Could not attach thread to JVM (%d, %p)", ret,
+ threadEnv);
+ return;
+ }
+ threadEnv->CallIntMethod(_callbackObj, _callbackId, framerate, bitrate,
+ packetLossRate, _frameRateO, _bitRateO);
+ webrtcGlobalVM->DetachCurrentThread();
+ }
+ ;
+
+ virtual void IncomingCodecChanged(const int videoChannel,
+ const webrtc::VideoCodec& videoCodec)
+ {
+ }
+ ;
+
+ virtual void RequestNewKeyFrame(const int videoChannel)
+ {
+ }
+ ;
+
+ virtual void OutgoingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate)
+ {
+ _frameRateO = framerate;
+ _bitRateO = bitrate;
+ //__android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ // "SendRate frameRate %d bitrate %d\n",frameRate,bitrate);
+ }
+ ;
+
+public:
+ VideoEngineData& _vieData;
+ JNIEnv * _env;
+ jobject _callbackObj;
+ jclass _callbackCls;
+ jmethodID _callbackId;
+ int _frameRateO, _bitRateO;
+ VideoCallbackAndroid(VideoEngineData& vieData, JNIEnv * env,
+ jobject callback) :
+ _vieData(vieData), _env(env), _callbackObj(callback),
+ _frameRateO(0), _bitRateO(0) {
+ _callbackCls = _env->GetObjectClass(_callbackObj);
+ _callbackId
+ = _env->GetMethodID(_callbackCls, "UpdateStats", "(IIIII)I");
+ if (_callbackId == NULL) {
+ __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to get jid");
+ }
+ _callbackObj = _env->NewGlobalRef(_callbackObj);
+ }
+};
+
+// JNI_OnLoad
+jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/) {
+ webrtcGlobalVM = vm;
+ if (!webrtcGlobalVM)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad did not receive a valid VM pointer");
+ return -1;
+ }
+
+ // Get JNI
+ JNIEnv* env;
+ if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
+ JNI_VERSION_1_4)) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad could not get JNI env");
+ return -1;
+ }
+
+ // Init VoiceEngine data
+ memset(&veData, 0, sizeof(veData));
+ // Store the JVM
+ veData.jvm = vm;
+
+ // Init VideoEngineData data
+ memset(&vieData, 0, sizeof(vieData));
+
+ return JNI_VERSION_1_4;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: NativeInit
+// Signature: (Landroid/content/Context;)Z
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_NativeInit(
+ JNIEnv * env,
+ jobject,
+ jobject context)
+{
+ return true;
+}
+
+// VideoEngine API wrapper functions
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: GetVideoEngine
+// Signature: ()I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetVideoEngine(
+ JNIEnv *,
+ jobject context) {
+
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "GetVideoEngine");
+
+ VideoEngine::SetAndroidObjects(webrtcGlobalVM, context);
+
+ // Check if already got
+ if (vieData.vie) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "ViE already got");
+ return -1;
+ }
+
+ // Create
+ vieData.vie = VideoEngine::Create();
+ if (!vieData.vie) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, "Get ViE failed");
+ return -1;
+ }
+ vieData.base = ViEBase::GetInterface(vieData.vie);
+ if (!vieData.base) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get base sub-API failed");
+ return -1;
+ }
+
+ vieData.codec = ViECodec::GetInterface(vieData.vie);
+ if (!vieData.codec) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get codec sub-API failed");
+ return -1;
+ }
+
+ vieData.netw = ViENetwork::GetInterface(vieData.vie);
+ if (!vieData.netw) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get network sub-API failed");
+ return -1;
+ }
+
+ vieData.rtp = ViERTP_RTCP::GetInterface(vieData.vie);
+ if (!vieData.rtp) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get RTP sub-API failed");
+ return -1;
+ }
+
+ vieData.render = ViERender::GetInterface(vieData.vie);
+ if (!vieData.render) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get Render sub-API failed");
+ return -1;
+ }
+
+ vieData.capture = ViECapture::GetInterface(vieData.vie);
+ if (!vieData.capture) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get Capture sub-API failed");
+ return -1;
+ }
+
+ return 0;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: Init
+// Signature: (IIIZ)I
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Init(
+ JNIEnv *,
+ jobject,
+ jboolean enableTrace)
+{
+ if (vieData.vie) {
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Init");
+
+ int ret = vieData.base->Init();
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "Init return %d", ret);
+ if (enableTrace)
+ {
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetTraceFile");
+ if (0 != vieData.vie->SetTraceFile(("/sdcard/trace.txt"), false))
+ {
+ __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Video Engine could not enable trace");
+ }
+
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetTraceFilter");
+ if (0 != vieData.vie->SetTraceFilter(webrtc::kTraceDefault))
+ {
+ __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
+ "Could not set trace filter");
+ }
+ }
+ else
+ {
+ if (0 != vieData.vie->SetTraceFilter(webrtc::kTraceNone))
+ {
+ __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
+ "Could not set trace filter");
+ }
+ }
+ if (veData.ve) // VoiceEngine is enabled
+ {
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetVoiceEngine");
+ if (0 != vieData.base->SetVoiceEngine(veData.ve))
+ {
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetVoiceEngine failed");
+ }
+ }
+ return ret;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+/*
+ * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+ * Method: Terminate
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Terminate(
+ JNIEnv *,
+ jobject)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Terminate");
+
+ if (vieData.vie) {
+ if (!vieData.rtp || vieData.rtp->Release() != 0) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release RTP sub-API");
+ }
+
+ if (!vieData.netw || vieData.netw->Release() != 0) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release Network sub-API");
+ }
+
+ if (!vieData.codec || vieData.codec->Release() != 0) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release Codec sub-API");
+ }
+
+ if (!vieData.render || vieData.render->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release Render sub-API");
+ }
+
+ if (!vieData.capture || vieData.capture->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release Capture sub-API");
+ }
+
+ if (!vieData.base || vieData.base->Release() != 0) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to release Base sub-API");
+ }
+
+ // Delete Vie
+ if (!VideoEngine::Delete(vieData.vie)) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to delete ViE ");
+ return -1;
+ }
+ memset(&vieData, 0, sizeof(vieData));
+ VideoEngine::SetAndroidObjects(NULL, NULL);
+ return 0;
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StartSend
+// Signature: (I)I
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartSend");
+
+ if (vieData.base) {
+ int ret = vieData.base->StartSend(channel);
+ return ret;
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StopRender
+// Signature: (I)I
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopRender(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopRender");
+
+ if (vieData.render) {
+ return vieData.render->StopRender(channel);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: Stop
+// Signature: ()I
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopSend");
+
+ if (vieData.base) {
+ return vieData.base->StopSend(channel);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StartReceive
+// Signature: ()I
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartReceive(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartReceive");
+
+ if (vieData.base) {
+ return vieData.base->StartReceive(channel);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StopReceive
+// Signature: ()I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopReceive(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopReceive");
+ if (vieData.base) {
+ return vieData.base->StopReceive(channel);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: CreateChannel
+// Signature: ()I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_CreateChannel(
+ JNIEnv *,
+ jobject,
+ jint voiceChannel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "CreateChannel");
+
+ if (vieData.vie) {
+ int channel = 0;
+ if (vieData.base->CreateChannel(channel) != 0) {
+ return -1;
+ }
+ if (voiceChannel >= 0) {
+ vieData.base->ConnectAudioChannel(channel, voiceChannel);
+ }
+
+ return channel;
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetLocalReceiver
+// Signature: (II)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetLocalReceiver(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint port)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
+
+ if (vieData.vie) {
+ int ret = vieData.netw->SetLocalReceiver(channel, port);
+ return ret;
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetSendDestination
+// Signature: (II[B)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendDestination(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jint port,
+ jbyteArray ipadr)
+{
+
+ if (NULL == vieData.vie)
+ return -1;
+
+ char ip[64];
+ jsize len = env->GetArrayLength(ipadr);
+ if ((len >= 64) || (len == 0))
+ return -1;
+ env->GetByteArrayRegion(ipadr, 0, len, (jbyte*) ip);
+ ip[len] = '\0';
+
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetSendDestination: channel=%d, port=%d, ip=%s\n",
+ channel, port, ip);
+
+ return vieData.netw->SetSendDestination(channel, ip, port);
+}
+
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetReceiveCodec
+// Signature: (IIIIII)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetReceiveCodec(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint codecNum,
+ jint intbitRate,
+ jint width,
+ jint height,
+ jint frameRate)
+{
+ if (NULL == vieData.codec)
+ return -1;
+
+ //Create codec
+ webrtc::VideoCodec codec;
+ vieData.codec->GetCodec(codecNum, codec);
+
+ __android_log_print(
+ ANDROID_LOG_DEBUG,
+ WEBRTC_LOG_TAG,
+ "SetReceiveCodec %s, pltype=%d, bitRate=%d, maxBitRate=%d,"
+ " width=%d, height=%d, frameRate=%d, codecSpecific=%d \n",
+ codec.plName, codec.plType, codec.startBitrate,
+ codec.maxBitrate, codec.width, codec.height,
+ codec.maxFramerate, codec.codecSpecific);
+ int ret = vieData.codec->SetReceiveCodec(channel, codec);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "SetReceiveCodec return %d", ret);
+ return ret;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetSendCodec
+// Signature: (IIIIII)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendCodec(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint codecNum,
+ jint intbitRate,
+ jint width,
+ jint height,
+ jint frameRate)
+{
+ if (NULL == vieData.codec)
+ return -1;
+
+ //Create codec
+ webrtc::VideoCodec codec;
+ vieData.codec->GetCodec(codecNum, codec);
+ codec.startBitrate = intbitRate;
+ codec.maxBitrate = 600;
+ codec.width = width;
+ codec.height = height;
+ codec.maxFramerate = frameRate;
+
+ for (int i = 0; i < vieData.codec->NumberOfCodecs(); ++i) {
+ webrtc::VideoCodec codecToList;
+ vieData.codec->GetCodec(i, codecToList);
+ __android_log_print(
+ ANDROID_LOG_DEBUG,
+ WEBRTC_LOG_TAG,
+ "Codec list %s, pltype=%d, bitRate=%d, maxBitRate=%d,"
+ " width=%d, height=%d, frameRate=%d\n",
+ codecToList.plName, codecToList.plType,
+ codecToList.startBitrate, codecToList.maxBitrate,
+ codecToList.width, codecToList.height,
+ codecToList.maxFramerate);
+ }
+ __android_log_print(
+ ANDROID_LOG_DEBUG,
+ WEBRTC_LOG_TAG,
+ "SetSendCodec %s, pltype=%d, bitRate=%d, maxBitRate=%d, "
+ "width=%d, height=%d, frameRate=%d\n",
+ codec.plName, codec.plType, codec.startBitrate,
+ codec.maxBitrate, codec.width, codec.height,
+ codec.maxFramerate);
+
+ return vieData.codec->SetSendCodec(channel, codec);
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: AddRemoteRenderer
+// Signature: (ILandroid/view/SurfaceView;)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_AddRemoteRenderer(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jobject glSurface)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "AddRemoteRenderer");
+ if (vieData.vie) {
+ return vieData.render->AddRenderer(channel, glSurface, 0, 0, 0, 1, 1);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: RemoveRemoteRenderer
+// Signature: (I)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_RemoveRemoteRenderer(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "RemoveRemoteRenderer");
+
+ if (vieData.vie) {
+ return vieData.render->RemoveRenderer(channel);
+ }
+ else {
+ return -1;
+ }
+ return 0;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StartRender
+// Signature: (I)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartRender(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartRender");
+
+ if (vieData.render) {
+ return vieData.render->StartRender(channel);
+ }
+ else {
+ return -1;
+ }
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StarteCamera
+// Signature: (II)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartCamera(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jint cameraNum)
+{
+ if (NULL == vieData.vie)
+ return -1;
+
+ int i = 0;
+ char deviceName[64];
+ char deviceUniqueName[64];
+ int re;
+ do {
+ re = vieData.capture->GetCaptureDevice(i, deviceName,
+ sizeof(deviceName),
+ deviceUniqueName,
+ sizeof(deviceUniqueName));
+ __android_log_print(
+ ANDROID_LOG_DEBUG,
+ WEBRTC_LOG_TAG,
+ "GetCaptureDevice ret %d devicenum %d deviceUniqueName %s",
+ re, i, deviceUniqueName);
+ i++;
+ } while (re == 0);
+
+ int ret;
+ int cameraId;
+ vieData.capture->GetCaptureDevice(cameraNum, deviceName,
+ sizeof(deviceName), deviceUniqueName,
+ sizeof(deviceUniqueName));
+ vieData.capture->AllocateCaptureDevice(deviceUniqueName,
+ sizeof(deviceUniqueName), cameraId);
+
+ if (cameraId >= 0) { //Connect the
+ ret = vieData.capture->ConnectCaptureDevice(cameraId, channel);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "ConnectCaptureDevice ret %d ", ret);
+
+ ret = vieData.capture->StartCapture(cameraId);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "StartCapture ret %d ", ret);
+ }
+
+ return cameraId;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: StopCamera
+// Signature: ()I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopCamera(
+ JNIEnv *,
+ jobject,
+ jint cameraId)
+{
+ if (NULL == vieData.capture)
+ return -1;
+
+ int ret = vieData.capture->StopCapture(cameraId);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "StopCapture ret %d ", ret);
+ ret = vieData.capture->ReleaseCaptureDevice(cameraId);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "ReleaseCaptureDevice ret %d ", ret);
+
+ return ret;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: GetCameraOrientation
+// Signature: (I)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetCameraOrientation(
+ JNIEnv *,
+ jobject,
+ jint cameraNum)
+{
+ char deviceName[64];
+ char deviceUniqueName[64];
+ int ret;
+
+ ret = vieData.capture->GetCaptureDevice(cameraNum, deviceName,
+ sizeof(deviceName),
+ deviceUniqueName,
+ sizeof(deviceUniqueName));
+ if (ret != 0) {
+ return -1;
+ }
+
+ RotateCapturedFrame orientation;
+ ret = vieData.capture->GetOrientation(deviceUniqueName, orientation);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "GetOrientation ret %d orientation %d", ret,
+ orientation);
+
+ return (jint) orientation;
+
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetRotation
+// Signature: (II)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetRotation(
+ JNIEnv *,
+ jobject,
+ jint captureId,
+ jint degrees)
+{
+
+ if (NULL == vieData.capture)
+ return -1;
+ RotateCapturedFrame rotation = RotateCapturedFrame_0;
+ if (degrees == 90)
+ rotation = RotateCapturedFrame_90;
+ else if (degrees == 180)
+ rotation = RotateCapturedFrame_180;
+ else if (degrees == 270)
+ rotation = RotateCapturedFrame_270;
+
+ int ret = vieData.capture->SetRotateCapturedFrames(captureId, rotation);
+ return ret;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: EnableNACK
+// Signature: (IZ)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnableNACK(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jboolean enable)
+{
+ if (NULL == vieData.rtp)
+ return -1;
+
+ if (enable)
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "EnableNACK enable");
+ else
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "EnableNACK disable");
+
+ int ret = vieData.rtp->SetNACKStatus(channel, enable);
+ return ret;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: EnablePLI
+// Signature: (IZ)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnablePLI(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jboolean enable)
+{
+ if (NULL == vieData.rtp)
+ return -1;
+
+ if (enable)
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "EnablePLI enable");
+ else
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "EnablePLI disable");
+
+ int ret = vieData.rtp->SetKeyFrameRequestMethod(channel,
+ kViEKeyFrameRequestPliRtcp);
+ return ret;
+}
+
+// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
+// Method: SetCallback
+// Signature: (ILorg/webrtc/videoengineapp/IViEAndroidCallback;)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallback(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jobject callback)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetCallback");
+
+ if (NULL == vieData.codec)
+ return -1;
+ if (vieData.callback == NULL) {
+ vieData.callback = new VideoCallbackAndroid(vieData, env, callback);
+ }
+ else if (vieData.codec) {
+ vieData.codec->DeregisterDecoderObserver(channel); // Wrong channel?
+ vieData.codec->DeregisterEncoderObserver(channel);
+ }
+
+ vieData.codec->RegisterDecoderObserver(channel, *vieData.callback);
+ vieData.codec->RegisterEncoderObserver(channel, *vieData.callback);
+
+ return 0;
+}
+
+//
+// VoiceEngine API wrapper functions
+//
+
+// Create VoiceEngine instance
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Create(
+ JNIEnv *env,
+ jobject,
+ jobject context)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Create");
+
+ // Check if already created
+ if (veData.ve) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "VoE already created");
+ return false;
+ }
+
+ // Init Android Object
+ VoiceEngine::SetAndroidObjects(veData.jvm, env, context);
+ // Create
+ veData.ve = VoiceEngine::Create();
+ if (!veData.ve) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Create VoE failed");
+ return false;
+ }
+
+ // Get sub-APIs
+ if (!VE_GetSubApis()) {
+ // If not OK, release all sub-APIs and delete VoE
+ VE_ReleaseSubApis();
+ if (!VoiceEngine::Delete(veData.ve)) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Delete VoE failed");
+ }
+ return false;
+ }
+
+ return true;
+}
+
+// Delete VoiceEngine instance
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Delete(
+ JNIEnv *,
+ jobject)
+{
+ // Check if exists
+ if (!veData.ve) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "VoE does not exist");
+ return false;
+ }
+
+ // Release sub-APIs
+ VE_ReleaseSubApis();
+
+ // Delete
+ if (!VoiceEngine::Delete(veData.ve)) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Delete VoE failed");
+ return false;
+ }
+
+ veData.ve = NULL;
+
+ // Clear instance independent Java objects
+ VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
+
+ return true;
+}
+
+// [Base] Initialize VoiceEngine
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Init(
+ JNIEnv *,
+ jobject,
+ jboolean enableTrace,
+ jboolean useExtTrans)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "VE_Init");
+
+ VALIDATE_BASE_POINTER;
+
+ if (useExtTrans) {
+ // Not implemented
+ return -1;
+ }
+
+ return veData.base->Init();
+}
+
+// [Base] Terminate VoiceEngine
+JNIEXPORT jint
+JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Terminate(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_BASE_POINTER;
+
+ jint retVal = veData.base->Terminate();
+ return retVal;
+}
+
+// [Base] Create channel
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_CreateChannel(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_BASE_POINTER;
+
+ webrtc::CodecInst voiceCodec;
+ int numOfVeCodecs = veData.codec->NumOfCodecs();
+
+ //enum all the supported codec
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "Supported Voice Codec:\n");
+ for (int i = 0; i < numOfVeCodecs; ++i) {
+ if (veData.codec->GetCodec(i, voiceCodec) != -1) {
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "num: %d name: %s\n", i, voiceCodec.plname);
+ }
+ }
+
+ jint channel = veData.base->CreateChannel();
+
+ return channel;
+}
+
+// [Base] Delete channel
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_DeleteChannel(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData.base->DeleteChannel(channel);
+}
+
+// [Base] SetLocalReceiver
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLocalReceiver(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint port)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
+ VALIDATE_BASE_POINTER;
+ return veData.base->SetLocalReceiver(channel, port);
+}
+
+// [Base] SetSendDestination
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendDestination(
+ JNIEnv *env,
+ jobject,
+ jint channel,
+ jint port,
+ jstring ipaddr)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetSendDestination");
+ VALIDATE_BASE_POINTER;
+
+ const char* ipaddrNative = env->GetStringUTFChars(ipaddr, NULL);
+ if (!ipaddrNative) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+ jint retVal = veData.base->SetSendDestination(channel, port, ipaddrNative);
+ env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
+ return retVal;
+}
+
+// [Base] StartListen
+JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartListen(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartListen");
+ VALIDATE_BASE_POINTER;
+ return veData.base->StartReceive(channel);
+}
+
+// [Base] Start playout
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayout(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartPlayout");
+ VALIDATE_BASE_POINTER;
+ return veData.base->StartPlayout(channel);
+}
+
+// [Base] Start send
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartSend");
+ VALIDATE_BASE_POINTER;
+ return veData.base->StartSend(channel);
+}
+
+// [Base] Stop listen
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopListen(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData.base->StartReceive(channel);
+}
+
+// [Base] Stop playout
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayout(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData.base->StopPlayout(channel);
+}
+
+// [Base] Stop send
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData.base->StopSend(channel);
+}
+
+// [codec] Number of codecs
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_NumOfCodecs(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_CODEC_POINTER;
+ return veData.codec->NumOfCodecs();
+}
+
+// [codec] Set send codec
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendCodec(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint index)
+{
+ VALIDATE_CODEC_POINTER;
+
+ webrtc::CodecInst codec;
+
+ for (int i = 0; i < veData.codec->NumOfCodecs(); ++i) {
+ webrtc::CodecInst codecToList;
+ veData.codec->GetCodec(i, codecToList);
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "VE Codec list %s, pltype=%d\n",
+ codecToList.plname, codecToList.pltype);
+ }
+
+ if (veData.codec->GetCodec(index, codec) != 0) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to get codec");
+ return -1;
+ }
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetSendCodec %s\n",
+ codec.plname);
+
+ return veData.codec->SetSendCodec(channel, codec);
+}
+
+// [audioprocessing] SetNSStatus
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetNSStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode)
+{
+ //TODO implement
+ return -1;
+}
+
+// [audioprocessing] SetAGCStatus
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetAGCStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode)
+{
+ //TODO implement
+ return -1;
+}
+
+// [audioprocessing] SetECStatus
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetECStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode,
+ jint AESmode,
+ jint AESattenuation)
+{
+ //TODO implement
+ return -1;
+}
+
+// [File] Start play file locally
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileLocally(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jstring fileName,
+ jboolean loop)
+{
+ VALIDATE_FILE_POINTER;
+
+ const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
+ if (!fileNameNative) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData.file->StartPlayingFileLocally(channel,
+ fileNameNative,
+ loop);
+
+ env->ReleaseStringUTFChars(fileName, fileNameNative);
+
+ return retVal;
+}
+
+// [File] Stop play file locally
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileLocally(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_FILE_POINTER;
+ return veData.file->StopPlayingFileLocally(channel);
+}
+
+// [File] Start playing file as microphone
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileAsMicrophone(
+ JNIEnv *env,
+ jobject,
+ jint channel,
+ jstring fileName,
+ jboolean loop)
+{
+ VALIDATE_FILE_POINTER;
+
+ const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
+ if (!fileNameNative) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData.file->StartPlayingFileAsMicrophone(channel,
+ fileNameNative,
+ loop);
+
+ env->ReleaseStringUTFChars(fileName, fileNameNative);
+
+ return retVal;
+}
+
+// [File] Stop playing file as microphone
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileAsMicrophone(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_FILE_POINTER;
+ return veData.file->StopPlayingFileAsMicrophone(channel);
+}
+
+// [Volume] Set speaker volume
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSpeakerVolume(
+ JNIEnv *,
+ jobject,
+ jint level)
+{
+ VALIDATE_VOLUME_POINTER;
+
+ if (veData.volume->SetSpeakerVolume(level) != 0) {
+ return -1;
+ }
+
+ unsigned int storedVolume = 0;
+ if (veData.volume->GetSpeakerVolume(storedVolume) != 0) {
+ return -1;
+ }
+
+ if (storedVolume != level) {
+ return -1;
+ }
+
+ return 0;
+}
+
+// [Hardware] Set speaker volume
+JNIEXPORT jint JNICALL
+Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLoudspeakerStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable)
+{
+ VALIDATE_HARDWARE_POINTER;
+
+ if (veData.hardware->SetLoudspeakerStatus(enable) != 0) {
+ return -1;
+ }
+
+ return 0;
+}
+
+//
+// local function
+//
+
+// Get all sub-APIs
+bool VE_GetSubApis() {
+ bool getOK = true;
+
+ // Base
+ veData.base = VoEBase::GetInterface(veData.ve);
+ if (!veData.base) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get base sub-API failed");
+ getOK = false;
+ }
+
+ // Codec
+ veData.codec = VoECodec::GetInterface(veData.ve);
+ if (!veData.codec) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get codec sub-API failed");
+ getOK = false;
+ }
+
+ // File
+ veData.file = VoEFile::GetInterface(veData.ve);
+ if (!veData.file) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get file sub-API failed");
+ getOK = false;
+ }
+
+ // Network
+ veData.netw = VoENetwork::GetInterface(veData.ve);
+ if (!veData.netw) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get network sub-API failed");
+ getOK = false;
+ }
+
+ // audioprocessing
+ veData.apm = VoEAudioProcessing::GetInterface(veData.ve);
+ if (!veData.apm) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get VoEAudioProcessing sub-API failed");
+ getOK = false;
+ }
+
+ // Volume
+ veData.volume = VoEVolumeControl::GetInterface(veData.ve);
+ if (!veData.volume) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get volume sub-API failed");
+ getOK = false;
+ }
+
+ // Hardware
+ veData.hardware = VoEHardware::GetInterface(veData.ve);
+ if (!veData.hardware) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get hardware sub-API failed");
+ getOK = false;
+ }
+
+ return getOK;
+}
+
+// Release all sub-APIs
+bool VE_ReleaseSubApis() {
+ bool releaseOK = true;
+
+ // Base
+ if (veData.base) {
+ if (0 != veData.base->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release base sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.base = NULL;
+ }
+ }
+
+ // Codec
+ if (veData.codec) {
+ if (0 != veData.codec->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release codec sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.codec = NULL;
+ }
+ }
+
+ // File
+ if (veData.file) {
+ if (0 != veData.file->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release file sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.file = NULL;
+ }
+ }
+
+ // Network
+ if (veData.netw) {
+ if (0 != veData.netw->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release network sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.netw = NULL;
+ }
+ }
+
+ // apm
+ if (veData.apm) {
+ if (0 != veData.apm->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release apm sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.apm = NULL;
+ }
+ }
+
+ // Volume
+ if (veData.volume) {
+ if (0 != veData.volume->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release volume sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.volume = NULL;
+ }
+ }
+
+ // Hardware
+ if (veData.hardware) {
+ if (0 != veData.hardware->Release()) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release hardware sub-API failed");
+ releaseOK = false;
+ }
+ else {
+ veData.hardware = NULL;
+ }
+ }
+
+ return releaseOK;
+}
diff --git a/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png b/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png
new file mode 100644
index 0000000..a07c69f
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png
Binary files differ
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml b/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml
new file mode 100644
index 0000000..e88b7c6
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml
@@ -0,0 +1,81 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout android:layout_width="fill_parent"
+ android:layout_height="fill_parent" android:orientation="vertical"
+ xmlns:android="http://schemas.android.com/apk/res/android">
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:textStyle="bold"
+ android:textSize="24dip"
+ android:text="Audio Settings"></TextView>
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+ <TextView android:id="@+id/TextView01"
+ android:layout_height="wrap_content"
+ android:layout_gravity="bottom"
+ android:layout_width="wrap_content"
+ android:text="@string/codecType">
+ </TextView>
+ <Spinner android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/spVoiceCodecType">
+ </Spinner>
+
+ <LinearLayout android:id="@+id/LinearLayout02"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:text="@string/aTxPort">
+ </TextView>
+ <EditText android:layout_height="wrap_content"
+ android:layout_width="wrap_content"
+ android:id="@+id/etATxPort">
+ </EditText>
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:text="@string/aRxPort">
+ </TextView>
+ <EditText android:layout_height="wrap_content"
+ android:layout_width="wrap_content"
+ android:id="@+id/etARxPort" >
+ </EditText>
+ </LinearLayout>
+
+ <LinearLayout android:id="@+id/LinearLayout02"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbSpeaker"
+ android:text="@string/speaker">
+ </CheckBox>
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbAutoGainControl"
+ android:text="@string/AutoGainControl">
+ </CheckBox>
+ </LinearLayout>
+
+ <LinearLayout android:id="@+id/LinearLayout02"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbAECM"
+ android:text="@string/AECM">
+ </CheckBox>
+ </LinearLayout>
+
+ <LinearLayout android:id="@+id/LinearLayout03"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbNoiseSuppression"
+ android:text="@string/NoiseSuppression">
+ </CheckBox>
+ </LinearLayout>
+
+</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/both.xml b/trunk/src/video_engine/main/test/android_test/res/layout/both.xml
new file mode 100644
index 0000000..d29d906
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/both.xml
@@ -0,0 +1,43 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout
+ xmlns:android="http://schemas.android.com/apk/res/android"
+ android:orientation="horizontal"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <LinearLayout
+ android:orientation="vertical"
+ android:layout_width="120dip"
+ android:layout_height="fill_parent">
+ <LinearLayout android:id="@+id/llLocalView"
+ android:layout_width="fill_parent"
+ android:layout_height="80dip">
+ </LinearLayout>
+ <TextView
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:layout_weight="1"
+ android:text="" />
+ <Button android:id="@+id/btSwitchCamera"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:text="@string/frontCamera"
+ android:layout_gravity="bottom"/>
+ <Button android:id="@+id/btStartStopCall"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:text="@string/startCall"
+ android:layout_gravity="bottom"/>
+ <Button android:id="@+id/btExit"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:layout_gravity="bottom"
+ android:text="@string/exit"/>
+ </LinearLayout>
+ <LinearLayout
+ android:id="@+id/llRemoteView"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:layout_weight="1">
+ </LinearLayout>
+ </LinearLayout >
+
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/main.xml b/trunk/src/video_engine/main/test/android_test/res/layout/main.xml
new file mode 100644
index 0000000..3b16cb7
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/main.xml
@@ -0,0 +1,54 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout android:layout_width="fill_parent"
+ android:layout_height="fill_parent" android:layout_gravity="right"
+ android:orientation="vertical"
+ xmlns:android="http://schemas.android.com/apk/res/android">
+
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/tvTitle"
+ android:textStyle="bold"
+ android:textSize="24dip"
+ android:text="@string/gSettings">
+ </TextView>
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+ <LinearLayout android:id="@+id/LinearLayout02"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbVideoReceive"
+ android:text="@string/enableVideoReceive">
+ </CheckBox>
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbVideoSend"
+ android:text="@string/enableVideoSend">
+ </CheckBox>
+ </LinearLayout>
+
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbVoice"
+ android:text="@string/enableVoice">
+ </CheckBox>
+
+ <TextView android:id="@+id/TextView02"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:text="@string/remoteIp">
+ </TextView>
+ <EditText android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/etRemoteIp" >
+ </EditText>
+
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbLoopback"
+ android:text="@string/loopback">
+ </CheckBox>
+</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/send.xml b/trunk/src/video_engine/main/test/android_test/res/layout/send.xml
new file mode 100644
index 0000000..ee230f5
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/send.xml
@@ -0,0 +1,17 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+
+
+ <SurfaceView android:id="@+id/svLocal"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </SurfaceView>
+ <ImageView android:id="@+id/ivPreview"
+ android:layout_height="fill_parent"
+ android:layout_width="fill_parent">
+ </ImageView>
+
+</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml b/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml
new file mode 100644
index 0000000..1612ec3
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml
@@ -0,0 +1,59 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout android:layout_width="fill_parent"
+ android:layout_height="fill_parent" android:orientation="vertical"
+ xmlns:android="http://schemas.android.com/apk/res/android">
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:textStyle="bold"
+ android:textSize="24dip"
+ android:text="Stats">
+ </TextView>
+
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvLocalIp"
+ android:textStyle="bold">
+ </TextView>
+
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvFrameRateI"
+ android:textStyle="bold">
+ </TextView>
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvBitRateI"
+ android:textStyle="bold">
+ </TextView>
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvPacketLoss"
+ android:textStyle="bold">
+ </TextView>
+
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvFrameRateO"
+ android:textStyle="bold">
+ </TextView>
+ <TextView android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/tvBitRateO"
+ android:textStyle="bold">
+ </TextView>
+</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml b/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml
new file mode 100644
index 0000000..9997113
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml
@@ -0,0 +1,25 @@
+<?xml version="1.0" encoding="utf-8"?>
+<TabHost xmlns:android="http://schemas.android.com/apk/res/android"
+ android:id="@android:id/tabhost"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <LinearLayout
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <TabWidget
+ android:id="@android:id/tabs"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content" />
+ <FrameLayout
+ android:id="@android:id/tabcontent"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <include android:id="@+id/tab_video" layout="@layout/both" />
+ <include android:id="@+id/tab_config" layout="@layout/main" />
+ <include android:id="@+id/tab_vconfig" layout="@layout/vconfig" />
+ <include android:id="@+id/tab_aconfig" layout="@layout/aconfig" />
+ <include android:id="@+id/tab_stats" layout="@layout/stats" />
+ </FrameLayout>
+ </LinearLayout>
+</TabHost>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml b/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml
new file mode 100644
index 0000000..5270c10
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml
@@ -0,0 +1,67 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout android:layout_width="fill_parent"
+ android:layout_height="fill_parent" android:orientation="vertical"
+ xmlns:android="http://schemas.android.com/apk/res/android">
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:textStyle="bold"
+ android:textSize="24dip"
+ android:text="@string/vSettings">
+ </TextView>
+
+ <TextView android:id="@+id/TextView03"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+
+ <TextView android:id="@+id/TextView01"
+ android:layout_height="wrap_content"
+ android:layout_width="wrap_content"
+ android:text="@string/codecType">
+ </TextView>
+ <Spinner android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/spCodecType">
+ </Spinner>
+
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/tvCodecSize"
+ android:text="@string/codecSize">
+ </TextView>
+ <Spinner android:layout_height="wrap_content"
+ android:layout_width="fill_parent"
+ android:id="@+id/spCodecSize">
+ </Spinner>
+
+ <LinearLayout android:id="@+id/LinearLayout02"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:text="@string/vTxPort">
+ </TextView>
+ <EditText android:layout_height="wrap_content"
+ android:layout_width="wrap_content"
+ android:id="@+id/etVTxPort" >
+ </EditText>
+ <TextView android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:text="@string/vRxPort">
+ </TextView>
+ <EditText android:layout_height="wrap_content"
+ android:layout_width="wrap_content"
+ android:id="@+id/etVRxPort" >
+ </EditText>
+ </LinearLayout>
+
+ <LinearLayout android:id="@+id/LinearLayout03"
+ android:layout_height="wrap_content"
+ android:layout_width="fill_parent">
+ <CheckBox android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:id="@+id/cbNack"
+ android:text="@string/nack">
+ </CheckBox>
+ </LinearLayout>
+</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml b/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml
new file mode 100644
index 0000000..77dd017
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml
@@ -0,0 +1,22 @@
+<resources>
+
+ <string-array name="codectype">
+ <item>vp8</item>
+ </string-array>
+
+<string-array name="codecSize"><item>176x144</item><item>320x240</item>
+<item>352x288</item><item>640x480</item>
+
+</string-array>
+
+<string-array name="voiceCodecType">
+<item>iPCM-wb</item>
+<item>iSAC</item>
+<item>iSAC-LC</item>
+<item>EG711U</item>
+<item>EG711A</item>
+<item>PCMU</item>
+<item>PCMA</item>
+<item>iLBC</item>
+</string-array>
+</resources>
diff --git a/trunk/src/video_engine/main/test/android_test/res/values/strings.xml b/trunk/src/video_engine/main/test/android_test/res/values/strings.xml
new file mode 100644
index 0000000..6597bec
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/res/values/strings.xml
@@ -0,0 +1,36 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+ <string name="app_name">ViEDemo</string>
+ <string name="error">Error</string>
+ <string name="errorCamera">Camera Error</string>
+<string name="codectype_prompt">Choose a codec type</string>
+<string name="demoTitle">Video Engine Android Demo</string>
+<string name="codecType">Codec Type</string>
+<string name="codecSize">Codec Size</string>
+<string name="remoteIp">Remote IP address</string>
+<string name="loopback">Loopback</string>
+<string name="startListen">Start Listen</string>
+<string name="startSend">Start Send</string>
+<string name="startBoth">Start Both</string>
+<string name="enableVoice">Voice</string>
+<string name="enableVideoReceive">Video Receive</string>
+<string name="enableVideoSend">Video Send</string>
+<string name="gSettings">Global Settings</string>
+<string name="vSettings">Video Settings</string>
+<string name="vTxPort">Video Tx Port</string>
+<string name="vRxPort">Video Rx Port</string>
+<string name="aTxPort">Audio Tx Port</string>
+<string name="aRxPort">Audio Rx Port</string>
+<string name="AutoGainControl">Auto Gain Control</string>
+<string name="VoiceActivityDetection">Voice Activity Detection</string>
+<string name="AECM">Acoustic Echo Control</string>
+<string name="NoiseSuppression">Noise Suppression</string>
+<string name="nack">NACK</string>
+<string name="frontCamera">SwitchToFront</string>
+<string name="backCamera">SwitchToBack</string>
+<string name="startCall">StartCall</string>
+<string name="stopCall">StopCall</string>
+<string name="exit">Exit</string>
+<string name="speaker">Speaker</string>
+<string name="h263DecSw">H263 SW Decoder</string>
+</resources>
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java
new file mode 100644
index 0000000..7a00d14
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengineapp;
+
+public interface IViEAndroidCallback {
+ public int UpdateStats(int frameRateI, int bitRateI,
+ int packetLoss, int frameRateO,
+ int bitRateO);
+}
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java
new file mode 100644
index 0000000..9301948
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java
@@ -0,0 +1,803 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengineapp;
+
+import java.net.InetAddress;
+import java.net.NetworkInterface;
+import java.net.SocketException;
+import java.util.Enumeration;
+
+import org.webrtc.videoengine.ViERenderer;
+
+
+import android.app.TabActivity;
+import android.content.Context;
+import android.content.res.Configuration;
+import android.hardware.SensorManager;
+import android.media.AudioManager;
+import android.os.Bundle;
+import android.os.PowerManager;
+import android.os.PowerManager.WakeLock;
+
+import android.util.Log;
+import android.view.KeyEvent;
+import android.view.Surface;
+import android.view.SurfaceView;
+import android.view.View;
+import android.view.Display;
+import android.view.Window;
+import android.view.WindowManager;
+
+import android.widget.AdapterView;
+import android.widget.ArrayAdapter;
+import android.widget.Button;
+import android.widget.CheckBox;
+
+import android.widget.EditText;
+import android.widget.LinearLayout;
+import android.widget.Spinner;
+import android.widget.TabHost;
+import android.widget.TextView;
+import android.widget.AdapterView.OnItemSelectedListener;
+import android.widget.TabHost.TabSpec;
+import android.view.OrientationEventListener;
+
+public class ViEAndroidDemo extends TabActivity implements IViEAndroidCallback,
+ View.OnClickListener,
+ OnItemSelectedListener {
+ private ViEAndroidJavaAPI ViEAndroidAPI = null;
+
+ // remote renderer
+ private SurfaceView remoteSurfaceView = null;
+
+ // local renderer and camera
+ private SurfaceView svLocal = null;
+
+ // channel number
+ private int channel;
+ private int cameraId;
+ private int voiceChannel = -1;
+
+ // flags
+ private boolean viERunning = false;
+ private boolean voERunning = false;
+
+ // debug
+ private boolean enableTrace = false;
+
+ // Constant
+ private static final String LOG_TAG = "*WEBRTCJ*";
+ private static final int RECEIVE_CODEC_FRAMERATE = 30;
+ private static final int SEND_CODEC_FRAMERATE = 15;
+ private static final int INIT_BITRATE = 400;
+
+ private static final int EXPIRARY_YEAR = 2010;
+ private static final int EXPIRARY_MONTH = 10;
+ private static final int EXPIRARY_DAY = 22;
+
+ private int volumeLevel = 204;
+
+ private TabHost mTabHost = null;
+
+ private TabSpec mTabSpecConfig;
+ private TabSpec mTabSpecVideo;
+
+ private LinearLayout mLlRemoteSurface = null;
+ private LinearLayout mLlLocalSurface = null;
+
+ private Button btStartStopCall;
+ private Button btSwitchCamera;
+
+ //Global Settings
+ private CheckBox cbVideoSend;
+ private boolean enableVideoSend = true;
+ private CheckBox cbVideoReceive;
+ private boolean enableVideoReceive = true;
+ private boolean enableVideo = true;
+ private CheckBox cbVoice;
+ private boolean enableVoice = false;
+ private EditText etRemoteIp;
+ private String remoteIp = "10.1.100.68";
+ private CheckBox cbLoopback;
+ private boolean loopbackMode = true;
+
+ //Video settings
+ private Spinner spCodecType;
+ private int codecType = 0;
+ private Spinner spCodecSize;
+ private int codecSizeWidth = 352;
+ private int codecSizeHeight = 288;
+ private TextView etVRxPort;
+ private int receivePortVideo = 11111;
+ private TextView etVTxPort;
+ private int destinationPortVideo = 11111;
+ private CheckBox cbEnableNack;
+ private boolean enableNack = false;
+
+ //Audio settings
+ private Spinner spVoiceCodecType;
+ private int voiceCodecType = 5; //PCMU = 5
+ private TextView etARxPort;
+ private int receivePortVoice = 11113;
+ private TextView etATxPort;
+ private int destinationPortVoice = 11113;
+ private CheckBox cbEnableSpeaker;
+ private boolean enableSpeaker = false;
+ private CheckBox cbEnableAGC;
+ private boolean enableAGC = false;
+ private CheckBox cbEnableAECM;
+ private boolean enableAECM = false;
+ private CheckBox cbEnableNS;
+ private boolean enableNS = false;
+
+ //Stats
+ private TextView tvFrameRateI;
+ private TextView tvBitRateI;
+ private TextView tvPacketLoss;
+ private TextView tvFrameRateO;
+ private TextView tvBitRateO;
+ private int frameRateI;
+ private int bitRateI;
+ private int packetLoss;
+ private int frameRateO;
+ private int bitRateO;
+
+ private WakeLock wakeLock;
+
+ private boolean usingFrontCamera = false;
+
+ private OrientationEventListener orientationListener;
+ int currentOrientation = OrientationEventListener.ORIENTATION_UNKNOWN;
+ int currentCameraOrientation = 0;
+
+ //Convert current display orientation to how much the camera should be rotated.
+ public int GetCameraOrientation(int cameraOrientation) {
+ Display display = this.getWindowManager().getDefaultDisplay();
+ int displatyRotation = display.getRotation();
+ int degrees = 0;
+ switch (displatyRotation) {
+ case Surface.ROTATION_0: degrees = 0; break;
+ case Surface.ROTATION_90: degrees = 90; break;
+ case Surface.ROTATION_180: degrees = 180; break;
+ case Surface.ROTATION_270: degrees = 270; break;
+ }
+ int result=0;
+ if(cameraOrientation>180) {
+ result=(cameraOrientation + degrees) % 360;
+ }
+ else {
+ result=(cameraOrientation - degrees+360) % 360;
+ }
+
+ return result;
+ }
+
+ public void onConfigurationChanged(Configuration newConfig) {
+ super.onConfigurationChanged(newConfig);
+ int newRotation = GetCameraOrientation(currentCameraOrientation);
+ if (viERunning){
+ ViEAndroidAPI.SetRotation(cameraId,newRotation);
+ }
+ }
+
+ // Called when the activity is first created.
+ @Override
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ requestWindowFeature(Window.FEATURE_NO_TITLE);
+ getWindow().setFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN,
+ WindowManager.LayoutParams.FLAG_FULLSCREEN);
+
+ PowerManager pm = (PowerManager)this.getSystemService(
+ Context.POWER_SERVICE);
+ wakeLock = pm.newWakeLock(
+ PowerManager.SCREEN_DIM_WAKE_LOCK, LOG_TAG);
+
+ setContentView(R.layout.tabhost);
+ mTabHost = getTabHost();
+
+ //Video tab
+ mTabSpecVideo = mTabHost.newTabSpec("tab_video");
+ mTabSpecVideo.setIndicator("Video");
+ mTabSpecVideo.setContent(R.id.tab_video);
+ mTabHost.addTab(mTabSpecVideo);
+
+ //Shared config tab
+ mTabHost = getTabHost();
+ mTabSpecConfig = mTabHost.newTabSpec("tab_config");
+ mTabSpecConfig.setIndicator("Config");
+ mTabSpecConfig.setContent(R.id.tab_config);
+ mTabHost.addTab(mTabSpecConfig);
+
+ TabSpec mTabv;
+ mTabv = mTabHost.newTabSpec("tab_vconfig");
+ mTabv.setIndicator("V. Config");
+ mTabv.setContent(R.id.tab_vconfig);
+ mTabHost.addTab(mTabv);
+ TabSpec mTaba;
+ mTaba = mTabHost.newTabSpec("tab_aconfig");
+ mTaba.setIndicator("A. Config");
+ mTaba.setContent(R.id.tab_aconfig);
+ mTabHost.addTab(mTaba);
+ TabSpec mTabs;
+ mTabs = mTabHost.newTabSpec("tab_stats");
+ mTabs.setIndicator("Stats");
+ mTabs.setContent(R.id.tab_stats);
+ mTabHost.addTab(mTabs);
+
+ int childCount = mTabHost.getTabWidget().getChildCount();
+ for (int i=0; i<childCount; i++)
+ mTabHost.getTabWidget().getChildAt(i).getLayoutParams().height = 50;
+
+ orientationListener =
+ new OrientationEventListener(this,SensorManager.SENSOR_DELAY_UI) {
+ public void onOrientationChanged (int orientation) {
+ if (orientation != ORIENTATION_UNKNOWN) {
+ currentOrientation = orientation;
+ }
+ }
+ };
+ orientationListener.enable ();
+
+ StartMain();
+ return;
+ }
+
+ private String GetLocalIpAddress() {
+ String localIPs = "";
+ try {
+ for (Enumeration<NetworkInterface> en = NetworkInterface
+ .getNetworkInterfaces(); en.hasMoreElements();) {
+ NetworkInterface intf = en.nextElement();
+ for (Enumeration<InetAddress> enumIpAddr = intf
+ .getInetAddresses(); enumIpAddr.hasMoreElements();) {
+ InetAddress inetAddress = enumIpAddr.nextElement();
+ if (!inetAddress.isLoopbackAddress()) {
+ localIPs += inetAddress.getHostAddress().toString() + " ";
+ //set the remote ip address the same as
+ // the local ip address of the last netif
+ remoteIp = inetAddress.getHostAddress().toString();
+ }
+ }
+ }
+ } catch (SocketException ex) {
+ Log.e(LOG_TAG, ex.toString());
+ }
+ return localIPs;
+ }
+
+ @Override
+ public boolean onKeyDown(int keyCode, KeyEvent event) {
+ if (keyCode == KeyEvent.KEYCODE_BACK) {
+ if (viERunning) {
+ StopAll();
+ StartMain();
+ }
+ finish();
+ return true;
+ }
+ return super.onKeyDown(keyCode, event);
+ }
+
+ private void StopAll() {
+ if (ViEAndroidAPI != null) {
+ if (voERunning) {
+ voERunning = false;
+ StopVoiceEngine();
+ }
+
+ if (viERunning) {
+ viERunning = false;
+ ViEAndroidAPI.StopRender(channel);
+ ViEAndroidAPI.StopReceive(channel);
+ ViEAndroidAPI.StopSend(channel);
+ ViEAndroidAPI.RemoveRemoteRenderer(channel);
+ // stop the camera
+ ViEAndroidAPI.StopCamera(cameraId);
+ ViEAndroidAPI.Terminate();
+ mLlRemoteSurface.removeView(remoteSurfaceView);
+ mLlLocalSurface.removeView(svLocal);
+ remoteSurfaceView = null;
+
+ svLocal = null;
+ }
+ }
+ }
+
+ private void StartMain() {
+ mTabHost.setCurrentTab(0);
+
+ mLlRemoteSurface = (LinearLayout) findViewById(R.id.llRemoteView);
+ mLlLocalSurface = (LinearLayout) findViewById(R.id.llLocalView);
+
+ if (null == ViEAndroidAPI)
+ ViEAndroidAPI = new ViEAndroidJavaAPI(this);
+
+ //setContentView(R.layout.main);
+ btSwitchCamera = (Button)findViewById(R.id.btSwitchCamera);
+ btSwitchCamera.setOnClickListener(this);
+ btStartStopCall = (Button)findViewById(R.id.btStartStopCall);
+ btStartStopCall.setOnClickListener(this);
+ findViewById(R.id.btExit).setOnClickListener(this);
+
+ // cleaning
+ remoteSurfaceView = null;
+ svLocal = null;
+
+ // init UI
+ ArrayAdapter<?> adapter;
+
+ int resource = android.R.layout.simple_spinner_item;
+ int dropdownRes = android.R.layout.simple_spinner_dropdown_item;
+
+ // video codec
+ spCodecType = (Spinner) findViewById(R.id.spCodecType);
+ adapter = ArrayAdapter.createFromResource(this,
+ R.array.codectype,
+ resource);
+ adapter.setDropDownViewResource(dropdownRes);
+ spCodecType.setAdapter(adapter);
+ spCodecType.setSelection(codecType);
+ spCodecType.setOnItemSelectedListener(this);
+
+ // voice codec
+ spVoiceCodecType = (Spinner) findViewById(R.id.spVoiceCodecType);
+ adapter = ArrayAdapter.createFromResource(this, R.array.voiceCodecType,
+ resource);
+ adapter.setDropDownViewResource(dropdownRes);
+ spVoiceCodecType.setAdapter(adapter);
+ spVoiceCodecType.setSelection(voiceCodecType);
+ spVoiceCodecType.setOnItemSelectedListener(this);
+
+ spCodecSize = (Spinner) findViewById(R.id.spCodecSize);
+ adapter = ArrayAdapter.createFromResource(this, R.array.codecSize,
+ resource);
+ adapter.setDropDownViewResource(dropdownRes);
+ spCodecSize.setAdapter(adapter);
+ spCodecSize.setOnItemSelectedListener(this);
+
+ String ip = GetLocalIpAddress();
+ TextView tvLocalIp = (TextView) findViewById(R.id.tvLocalIp);
+ tvLocalIp.setText("Local IP address - " + ip);
+
+ etRemoteIp = (EditText) findViewById(R.id.etRemoteIp);
+ etRemoteIp.setText(remoteIp);
+
+ cbLoopback = (CheckBox) findViewById(R.id.cbLoopback);
+ cbLoopback.setChecked(loopbackMode);
+
+ cbVoice = (CheckBox) findViewById(R.id.cbVoice);
+ cbVoice.setChecked(enableVoice);
+
+ cbVideoSend = (CheckBox) findViewById(R.id.cbVideoSend);
+ cbVideoSend.setChecked(enableVideoSend);
+ cbVideoReceive = (CheckBox) findViewById(R.id.cbVideoReceive);
+ cbVideoReceive.setChecked(enableVideoReceive);
+
+ etVTxPort = (EditText) findViewById(R.id.etVTxPort);
+ etVTxPort.setText(Integer.toString(destinationPortVideo));
+
+ etVRxPort = (EditText) findViewById(R.id.etVRxPort);
+ etVRxPort.setText(Integer.toString(receivePortVideo));
+
+ etATxPort = (EditText) findViewById(R.id.etATxPort);
+ etATxPort.setText(Integer.toString(destinationPortVoice));
+
+ etARxPort = (EditText) findViewById(R.id.etARxPort);
+ etARxPort.setText(Integer.toString(receivePortVoice));
+
+ cbEnableNack = (CheckBox) findViewById(R.id.cbNack);
+ cbEnableNack.setChecked(enableNack);
+
+ cbEnableSpeaker = (CheckBox) findViewById(R.id.cbSpeaker);
+ cbEnableSpeaker.setChecked(enableSpeaker);
+ cbEnableAGC = (CheckBox) findViewById(R.id.cbAutoGainControl);
+ cbEnableAGC.setChecked(enableAGC);
+ cbEnableAECM = (CheckBox) findViewById(R.id.cbAECM);
+ cbEnableAECM.setChecked(enableAECM);
+ cbEnableNS = (CheckBox) findViewById(R.id.cbNoiseSuppression);
+ cbEnableNS.setChecked(enableNS);
+
+ cbEnableNack.setOnClickListener(this);
+ cbEnableSpeaker.setOnClickListener(this);
+ cbEnableAECM.setOnClickListener(this);
+
+ cbEnableAGC.setOnClickListener(this);
+ cbEnableNS.setOnClickListener(this);
+
+ tvFrameRateI = (TextView) findViewById(R.id.tvFrameRateI);
+ tvBitRateI = (TextView) findViewById(R.id.tvBitRateI);
+ tvPacketLoss = (TextView) findViewById(R.id.tvPacketLoss);
+ tvFrameRateO = (TextView) findViewById(R.id.tvFrameRateO);
+ tvBitRateO = (TextView) findViewById(R.id.tvBitRateO);
+
+ }
+
+ @Override
+ protected void onPause() {
+ super.onPause();
+ // if (remoteSurfaceView != null)
+ // glSurfaceView.onPause();
+ }
+
+ @Override
+ protected void onResume() {
+ super.onResume();
+ // if (glSurfaceView != null)
+ // glSurfaceView.onResume();
+ }
+
+ private void StartCall() {
+ int ret = 0;
+
+ if (enableVoice) {
+ SetupVoE();
+ StartVoiceEngine();
+ }
+
+ if (enableVideo) {
+ if (enableVideoSend) {
+ // camera and preview surface
+ svLocal = ViERenderer.CreateLocalRenderer(this);
+ }
+
+ ret = ViEAndroidAPI.GetVideoEngine();
+ ret = ViEAndroidAPI.Init(enableTrace);
+ channel = ViEAndroidAPI.CreateChannel(voiceChannel);
+ ret = ViEAndroidAPI.SetLocalReceiver(channel,
+ receivePortVideo);
+ ret = ViEAndroidAPI.SetSendDestination(channel,
+ destinationPortVideo,
+ remoteIp.getBytes());
+
+ if (enableVideoReceive) {
+ if(android.os.Build.MANUFACTURER.equals("samsung")) {
+ // Create an Open GL renderer
+ remoteSurfaceView = ViERenderer.CreateRenderer(this, true);
+ ret = ViEAndroidAPI.AddRemoteRenderer(channel, remoteSurfaceView);
+ }
+ else {
+ remoteSurfaceView = ViERenderer.CreateRenderer(this, false);
+ ret = ViEAndroidAPI.AddRemoteRenderer(channel, remoteSurfaceView);
+ }
+
+ ret = ViEAndroidAPI.SetReceiveCodec(channel,
+ codecType,
+ INIT_BITRATE,
+ codecSizeWidth,
+ codecSizeHeight,
+ RECEIVE_CODEC_FRAMERATE);
+ ret = ViEAndroidAPI.StartRender(channel);
+ ret = ViEAndroidAPI.StartReceive(channel);
+ }
+
+ if (enableVideoSend) {
+ currentCameraOrientation =
+ ViEAndroidAPI.GetCameraOrientation(usingFrontCamera?1:0);
+ ret = ViEAndroidAPI.SetSendCodec(channel,
+ codecType,
+ INIT_BITRATE,
+ codecSizeWidth,
+ codecSizeHeight,
+ SEND_CODEC_FRAMERATE);
+ int cameraId = ViEAndroidAPI.StartCamera(channel, usingFrontCamera?1:0);
+
+ if(cameraId>0) {
+ cameraId = cameraId;
+ int neededRotation = GetCameraOrientation(currentCameraOrientation);
+ ViEAndroidAPI.SetRotation(cameraId,neededRotation);
+ }
+ else {
+ ret=cameraId;
+ }
+ ret = ViEAndroidAPI.StartSend(channel);
+ }
+
+ ret = ViEAndroidAPI.SetCallback(channel, this);
+
+ if (enableVideoSend) {
+ if (mLlLocalSurface != null)
+ mLlLocalSurface.addView(svLocal);
+ }
+
+ if (enableVideoReceive) {
+ if (mLlRemoteSurface != null)
+ mLlRemoteSurface.addView(remoteSurfaceView);
+ }
+
+ viERunning = true;
+ }
+
+ }
+
+ private void DemoLog(String msg) {
+ Log.d("*WEBRTC*", msg);
+ }
+
+ private void StopVoiceEngine() {
+ // Stop send
+ if (0 != ViEAndroidAPI.VoE_StopSend(voiceChannel)) {
+ DemoLog("VoE stop send failed");
+ }
+
+ // Stop listen
+ if (0 != ViEAndroidAPI.VoE_StopListen(voiceChannel)) {
+ DemoLog("VoE stop listen failed");
+ }
+
+ // Stop playout
+ if (0 != ViEAndroidAPI.VoE_StopPlayout(voiceChannel)) {
+ DemoLog("VoE stop playout failed");
+ }
+
+ if (0 != ViEAndroidAPI.VoE_DeleteChannel(voiceChannel)) {
+ DemoLog("VoE delete channel failed");
+ }
+ voiceChannel=-1;
+
+ // Terminate
+ if (0 != ViEAndroidAPI.VoE_Terminate()) {
+ DemoLog("VoE terminate failed");
+ }
+ }
+
+ private void SetupVoE() {
+ // Create VoiceEngine
+ // Error logging is done in native API wrapper
+ ViEAndroidAPI.VoE_Create(this);
+
+ // Initialize
+ if (0 != ViEAndroidAPI.VoE_Init(enableTrace)) {
+ DemoLog("VoE init failed");
+ }
+
+ // Create channel
+ voiceChannel = ViEAndroidAPI.VoE_CreateChannel();
+ if (0 != voiceChannel) {
+ DemoLog("VoE create channel failed");
+ }
+
+ // Suggest to use the voice call audio stream for hardware volume controls
+ setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
+ }
+
+ private int StartVoiceEngine() {
+ // Set local receiver
+ if (0 != ViEAndroidAPI.VoE_SetLocalReceiver(voiceChannel,
+ receivePortVoice)) {
+ DemoLog("VoE set local receiver failed");
+ }
+
+ if (0 != ViEAndroidAPI.VoE_StartListen(voiceChannel)) {
+ DemoLog("VoE start listen failed");
+ }
+
+ // Route audio
+ RouteAudio(enableSpeaker);
+
+ // set volume to default value
+ if (0 != ViEAndroidAPI.VoE_SetSpeakerVolume(volumeLevel)) {
+ DemoLog("VoE set speaker volume failed");
+ }
+
+ // Start playout
+ if (0 != ViEAndroidAPI.VoE_StartPlayout(voiceChannel)) {
+ DemoLog("VoE start playout failed");
+ }
+
+ if (0 != ViEAndroidAPI.VoE_SetSendDestination(voiceChannel,
+ destinationPortVoice,
+ remoteIp)) {
+ DemoLog("VoE set send destination failed");
+ }
+
+ // 0 = iPCM-wb, 5 = PCMU
+ if (0 != ViEAndroidAPI.VoE_SetSendCodec(voiceChannel, voiceCodecType)) {
+ DemoLog("VoE set send codec failed");
+ }
+
+ if (0 != ViEAndroidAPI.VoE_SetECStatus(enableAECM, 5, 0, 28)){
+ DemoLog("VoE set EC Status failed");
+ }
+
+ if (0 != ViEAndroidAPI.VoE_StartSend(voiceChannel)) {
+ DemoLog("VoE start send failed");
+ }
+
+ voERunning = true;
+ return 0;
+ }
+
+ private void RouteAudio(boolean enableSpeaker) {
+ int sdkVersion = Integer.parseInt(android.os.Build.VERSION.SDK);
+ if (sdkVersion >= 5) {
+ AudioManager am =
+ (AudioManager) this.getSystemService(Context.AUDIO_SERVICE);
+ am.setSpeakerphoneOn(enableSpeaker);
+ }
+ else {
+ if (0 != ViEAndroidAPI.VoE_SetLoudspeakerStatus(enableSpeaker)) {
+ DemoLog("VoE set louspeaker status failed");
+ }
+ }
+ }
+
+ public void onClick(View arg0) {
+ switch (arg0.getId()) {
+ case R.id.btSwitchCamera:
+ if (usingFrontCamera ){
+ btSwitchCamera.setText(R.string.frontCamera);
+ }
+ else {
+ btSwitchCamera.setText(R.string.backCamera);
+ }
+ usingFrontCamera = !usingFrontCamera;
+
+ if (viERunning) {
+ currentCameraOrientation =
+ ViEAndroidAPI.GetCameraOrientation(usingFrontCamera?1:0);
+ ViEAndroidAPI.StopCamera(cameraId);
+ mLlLocalSurface.removeView(svLocal);
+
+ ViEAndroidAPI.StartCamera(channel,usingFrontCamera?1:0);
+ mLlLocalSurface.addView(svLocal);
+ int neededRotation = GetCameraOrientation(currentCameraOrientation);
+ ViEAndroidAPI.SetRotation(cameraId, neededRotation);
+ }
+ break;
+ case R.id.btStartStopCall:
+ ReadSettings();
+ if (viERunning || voERunning) {
+ StopAll();
+ wakeLock.release();//release the wake lock
+ btStartStopCall.setText(R.string.startCall);
+ }
+ else if (enableVoice || enableVideo){
+ StartCall();
+ wakeLock.acquire();//screen stay on during the call
+ btStartStopCall.setText(R.string.stopCall);
+ }
+ break;
+ case R.id.btExit:
+ StopAll();
+ finish();
+ break;
+ case R.id.cbNack:
+ enableNack = cbEnableNack.isChecked();
+ if (viERunning) {
+ ViEAndroidAPI.EnableNACK(channel, enableNack);
+ }
+ break;
+ case R.id.cbSpeaker:
+ enableSpeaker = cbEnableSpeaker.isChecked();
+ if (voERunning){
+ RouteAudio(enableSpeaker);
+ }
+ break;
+ case R.id.cbAutoGainControl:
+ enableAGC=cbEnableAGC.isChecked();
+ if(voERunning) {
+ //Enable AGC default mode.
+ ViEAndroidAPI.VoE_SetAGCStatus(enableAGC,1);
+ }
+ break;
+ case R.id.cbNoiseSuppression:
+ enableNS=cbEnableNS.isChecked();
+ if(voERunning) {
+ //Enable NS default mode.
+ ViEAndroidAPI.VoE_SetNSStatus(enableNS, 1);
+ }
+ break;
+ case R.id.cbAECM:
+ enableAECM = cbEnableAECM.isChecked();
+ if (voERunning) {
+ //EC_AECM=5
+ //AECM_DEFAULT=0
+ ViEAndroidAPI.VoE_SetECStatus(enableAECM, 5, 0, 28);
+ }
+ break;
+ }
+ }
+
+ private void ReadSettings() {
+ codecType = spCodecType.getSelectedItemPosition();
+ voiceCodecType = spVoiceCodecType.getSelectedItemPosition();
+
+ String sCodecSize = spCodecSize.getSelectedItem().toString();
+ String[] aCodecSize = sCodecSize.split("x");
+ codecSizeWidth = Integer.parseInt(aCodecSize[0]);
+ codecSizeHeight = Integer.parseInt(aCodecSize[1]);
+
+ loopbackMode = cbLoopback.isChecked();
+ enableVoice = cbVoice.isChecked();
+ enableVideoSend = cbVideoSend.isChecked();
+ enableVideoReceive = cbVideoReceive.isChecked();
+ enableVideo = enableVideoSend || enableVideoReceive;
+
+ destinationPortVideo =
+ Integer.parseInt(etVTxPort.getText().toString());
+ receivePortVideo =
+ Integer.parseInt(etVRxPort.getText().toString());
+ destinationPortVoice =
+ Integer.parseInt(etATxPort.getText().toString());
+ receivePortVoice =
+ Integer.parseInt(etARxPort.getText().toString());
+
+ enableNack = cbEnableNack.isChecked();
+ enableSpeaker = cbEnableSpeaker.isChecked();
+ enableAGC = cbEnableAGC.isChecked();
+ enableAECM = cbEnableAECM.isChecked();
+ enableNS = cbEnableNS.isChecked();
+
+ if (loopbackMode)
+ remoteIp = "127.0.0.1";
+ else
+ remoteIp = etRemoteIp.getText().toString();
+ }
+
+ public void onItemSelected(AdapterView<?> adapterView, View view,
+ int position, long id) {
+ if ((adapterView==spCodecType || adapterView==spCodecSize) &&
+ viERunning) {
+ ReadSettings();
+ //change the codectype
+ if (enableVideoReceive) {
+ if (0 !=ViEAndroidAPI.SetReceiveCodec(channel, codecType,
+ INIT_BITRATE, codecSizeWidth,
+ codecSizeHeight,
+ RECEIVE_CODEC_FRAMERATE))
+ DemoLog("ViE set receive codec failed");
+ }
+ if (enableVideoSend) {
+ if (0!=ViEAndroidAPI.SetSendCodec(channel, codecType, INIT_BITRATE,
+ codecSizeWidth,
+ codecSizeHeight,
+ SEND_CODEC_FRAMERATE))
+ DemoLog("ViE set send codec failed");
+ }
+ }
+ else if ((adapterView==spVoiceCodecType) && voERunning) {
+ //change voice engine codec
+ ReadSettings();
+ if (0 != ViEAndroidAPI.VoE_SetSendCodec(voiceChannel, voiceCodecType)) {
+ DemoLog("VoE set send codec failed");
+ }
+ }
+ }
+
+ public void onNothingSelected(AdapterView<?> arg0) {
+ DemoLog("No setting selected");
+ }
+
+ public int UpdateStats(int in_frameRateI, int in_bitRateI, int in_packetLoss,
+ int in_frameRateO, int in_bitRateO) {
+ frameRateI = in_frameRateI;
+ bitRateI = in_bitRateI;
+ packetLoss = in_packetLoss;
+ frameRateO = in_frameRateO;
+ bitRateO = in_bitRateO;
+ runOnUiThread(new Runnable() {
+ public void run() {
+ tvFrameRateI.setText("Incoming FrameRate - " +
+ Integer.toString(frameRateI));
+ tvBitRateI.setText("Incoming BitRate - " +
+ Integer.toString(bitRateI));
+ tvPacketLoss.setText("Incoming Packet Loss - " +
+ Integer.toString(packetLoss));
+ tvFrameRateO.setText("Send FrameRate - " +
+ Integer.toString(frameRateO));
+ tvBitRateO.setText("Send BitRate - " +
+ Integer.toString(bitRateO));
+ }
+ });
+ return 0;
+ }
+}
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java
new file mode 100644
index 0000000..701f8bd
--- /dev/null
+++ b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java
@@ -0,0 +1,138 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.videoengineapp;
+
+import android.app.Activity;
+import android.content.Context;
+import android.util.Log;
+import android.view.SurfaceHolder;
+import android.view.SurfaceView;
+
+public class ViEAndroidJavaAPI {
+
+ public ViEAndroidJavaAPI(Context context) {
+ Log.d("*WEBRTCJ*", "Loading ViEAndroidJavaAPI...");
+ System.loadLibrary("webrtc-video-demo-jni");
+
+ Log.d("*WEBRTCJ*", "Calling native init...");
+ if (!NativeInit(context)) {
+ Log.e("*WEBRTCJ*", "Native init failed");
+ throw new RuntimeException("Native init failed");
+ }
+ else {
+ Log.d("*WEBRTCJ*", "Native init successful");
+ }
+ String a = "";
+ a.getBytes();
+ }
+
+ // API Native
+ private native boolean NativeInit(Context context);
+
+ // Video Engine API
+ // Initialization and Termination functions
+ public native int GetVideoEngine();
+ public native int Init(boolean enableTrace);
+ public native int Terminate();
+
+
+ public native int StartSend(int channel);
+ public native int StopRender(int channel);
+ public native int StopSend(int channel);
+ public native int StartReceive(int channel);
+ public native int StopReceive(int channel);
+ // Channel functions
+ public native int CreateChannel(int voiceChannel);
+ // Receiver & Destination functions
+ public native int SetLocalReceiver(int channel, int port);
+ public native int SetSendDestination(int channel, int port, byte ipadr[]);
+ // Codec
+ public native int SetReceiveCodec(int channel, int codecNum,
+ int intbitRate, int width,
+ int height, int frameRate);
+ public native int SetSendCodec(int channel, int codecNum,
+ int intbitRate, int width,
+ int height, int frameRate);
+ // Rendering
+ public native int AddRemoteRenderer(int channel, Object glSurface);
+ public native int RemoveRemoteRenderer(int channel);
+ public native int StartRender(int channel);
+
+ // Capture
+ public native int StartCamera(int channel, int cameraNum);
+ public native int StopCamera(int cameraId);
+ public native int GetCameraOrientation(int cameraNum);
+ public native int SetRotation(int cameraId,int degrees);
+
+ // NACK
+ public native int EnableNACK(int channel, boolean enable);
+
+ //PLI for H.264
+ public native int EnablePLI(int channel, boolean enable);
+
+ // Enable stats callback
+ public native int SetCallback(int channel, IViEAndroidCallback callback);
+
+ // Voice Engine API
+ // Create and Delete functions
+ public native boolean VoE_Create(Activity context);
+ public native boolean VoE_Delete();
+
+ // Initialization and Termination functions
+ public native int VoE_Authenticate(String key);
+ public native int VoE_Init(boolean enableTrace);
+ public native int VoE_Terminate();
+
+ // Channel functions
+ public native int VoE_CreateChannel();
+ public native int VoE_DeleteChannel(int channel);
+
+ // Receiver & Destination functions
+ public native int VoE_SetLocalReceiver(int channel, int port);
+ public native int VoE_SetSendDestination(int channel, int port,
+ String ipaddr);
+
+ // Media functions
+ public native int VoE_StartListen(int channel);
+ public native int VoE_StartPlayout(int channel);
+ public native int VoE_StartSend(int channel);
+ public native int VoE_StopListen(int channel);
+ public native int VoE_StopPlayout(int channel);
+ public native int VoE_StopSend(int channel);
+
+ // Volume
+ public native int VoE_SetSpeakerVolume(int volume);
+
+ // Hardware
+ public native int VoE_SetLoudspeakerStatus(boolean enable);
+
+ // Playout file locally
+ public native int VoE_StartPlayingFileLocally(int channel,
+ String fileName,
+ boolean loop);
+ public native int VoE_StopPlayingFileLocally(int channel);
+
+ // Play file as microphone
+ public native int VoE_StartPlayingFileAsMicrophone(int channel,
+ String fileName,
+ boolean loop);
+ public native int VoE_StopPlayingFileAsMicrophone(int channel);
+
+ // Codec-setting functions
+ public native int VoE_NumOfCodecs();
+ public native int VoE_SetSendCodec(int channel, int index);
+
+ //VE funtions
+ public native int VoE_SetECStatus(boolean enable, int mode,
+ int AESmode, int AESattenuation);
+ public native int VoE_SetAGCStatus(boolean enable, int mode);
+ public native int VoE_SetNSStatus(boolean enable, int mode);
+}
diff --git a/trunk/src/video_engine/test/auto_test/android/.classpath b/trunk/src/video_engine/test/auto_test/android/.classpath
new file mode 100644
index 0000000..f2adf55
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/.classpath
@@ -0,0 +1,9 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+ <classpathentry kind="src" path="src"/>
+ <classpathentry kind="src" path="gen"/>
+ <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+ <classpathentry kind="lib" path="libs/VideoEngine_android_java.jar"/>
+ <classpathentry kind="lib" path="libs/VoiceEngine_android_java.jar"/>
+ <classpathentry kind="output" path="bin"/>
+</classpath>
diff --git a/trunk/src/video_engine/test/auto_test/android/Android.mk b/trunk/src/video_engine/test/auto_test/android/Android.mk
new file mode 100644
index 0000000..2b20e80
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/Android.mk
@@ -0,0 +1,39 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+MY_CAPTURE_FOLDER := ../../../../modules/video_capture/main/source
+MY_CAPTURE_JAVA_FOLDER := Android/java/org/webrtc/videoengine
+MY_CAPTURE_PATH := $(MY_CAPTURE_FOLDER)/$(MY_CAPTURE_JAVA_FOLDER)
+
+MY_RENDER_FOLDER := ../../../../modules/video_render/main/source
+MY_RENDER_JAVA_FOLDER := Android/java/org/webrtc/videoengine
+MY_RENDER_PATH := $(MY_RENDER_FOLDER)/$(MY_RENDER_JAVA_FOLDER)
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ src/org/webrtc/vieautotest/ViEAutotest.java \
+ $(MY_CAPTURE_PATH)/CaptureCapabilityAndroid.java \
+ $(MY_CAPTURE_PATH)/VideoCaptureAndroid.java \
+ $(MY_CAPTURE_PATH)/VideoCaptureDeviceInfoAndroid.java \
+ $(MY_RENDER_PATH)/ViEAndroidGLES20.java \
+ $(MY_RENDER_PATH)/ViERenderer.java \
+ $(MY_RENDER_PATH)/ViESurfaceRenderer.java
+
+LOCAL_PACKAGE_NAME := webrtc-video-autotest
+LOCAL_CERTIFICATE := platform
+
+LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-video-autotest-jni
+
+include $(BUILD_PACKAGE)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml b/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml
new file mode 100644
index 0000000..11b3e27
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml
@@ -0,0 +1,26 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ android:versionCode="1"
+ android:versionName="1.0" package="org.webrtc.vieautotest">
+ <application android:label="@string/app_name"
+ android:debuggable="true"
+ android:icon="@drawable/logo">
+ <activity android:label="@string/app_name"
+ android:name="ViEAutotest">
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+
+ </application>
+ <uses-sdk android:minSdkVersion="3" android:targetSdkVersion="8" />
+ <uses-permission android:name="android.permission.CAMERA"></uses-permission>
+ <uses-feature android:name="android.hardware.camera" />
+ <uses-feature android:name="android.hardware.camera.autofocus" />
+ <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+ <uses-permission android:name="android.permission.RECORD_AUDIO" />
+ <uses-permission android:name="android.permission.INTERNET" />
+ <uses-permission android:name="android.permission.WAKE_LOCK" />
+ <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+</manifest>
diff --git a/trunk/src/video_engine/test/auto_test/android/default.properties b/trunk/src/video_engine/test/auto_test/android/default.properties
new file mode 100644
index 0000000..2ad44a4
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target.
+target=android-9
diff --git a/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java b/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
new file mode 100644
index 0000000..4b46020
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
@@ -0,0 +1,37 @@
+/* AUTO-GENERATED FILE. DO NOT MODIFY.
+ *
+ * This class was automatically generated by the
+ * aapt tool from the resource data it found. It
+ * should not be modified by hand.
+ */
+
+package org.webrtc.vieautotest;
+
+public final class R {
+ public static final class array {
+ public static final int subtest_array=0x7f050001;
+ public static final int test_array=0x7f050000;
+ }
+ public static final class attr {
+ }
+ public static final class drawable {
+ public static final int logo=0x7f020000;
+ }
+ public static final class id {
+ public static final int Button01=0x7f060004;
+ public static final int LocalView=0x7f060001;
+ public static final int RemoteView=0x7f060000;
+ public static final int subtestSpinner=0x7f060003;
+ public static final int testSpinner=0x7f060002;
+ }
+ public static final class layout {
+ public static final int main=0x7f030000;
+ }
+ public static final class string {
+ public static final int SpinnerSubtest=0x7f040004;
+ public static final int SpinnerTitle=0x7f040003;
+ public static final int TitleName=0x7f040001;
+ public static final int app_name=0x7f040000;
+ public static final int run_button=0x7f040002;
+ }
+}
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/Android.mk b/trunk/src/video_engine/test/auto_test/android/jni/Android.mk
new file mode 100644
index 0000000..ac43975
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/jni/Android.mk
@@ -0,0 +1,77 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+# the follow two lines are for NDK build
+INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../build/interface
+LIBS_PATH := $(LOCAL_PATH)/../../../../../../build/libraries
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_MODULE := libwebrtc-video-autotest-jni
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ vie_autotest_jni.cc \
+ ../../source/vie_autotest_android.cc \
+ ../../source/vie_autotest.cc \
+ ../../source/vie_autotest_base.cc \
+ ../../source/vie_autotest_capture.cc \
+ ../../source/vie_autotest_codec.cc \
+ ../../source/vie_autotest_encryption.cc \
+ ../../source/vie_autotest_file.cc \
+ ../../source/vie_autotest_image_process.cc \
+ ../../source/vie_autotest_loopback.cc \
+ ../../source/vie_autotest_network.cc \
+ ../../source/vie_autotest_render.cc \
+ ../../source/vie_autotest_rtp_rtcp.cc \
+ ../../source/tb_I420_codec.cc \
+ ../../source/tb_capture_device.cc \
+ ../../source/tb_external_transport.cc \
+ ../../source/tb_interfaces.cc \
+ ../../source/tb_video_channel.cc
+
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES'
+
+LOCAL_C_INCLUDES := \
+ external/gtest/include \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../../interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../../.. \
+ $(LOCAL_PATH)/../../../../../common_video/interface \
+ $(LOCAL_PATH)/../../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../../modules/interface \
+ $(LOCAL_PATH)/../../../../../modules/video_capture/main/interface \
+ $(LOCAL_PATH)/../../../../../modules/video_capture/main/source \
+ $(LOCAL_PATH)/../../../../../modules/video_coding/codecs/interface \
+ $(LOCAL_PATH)/../../../../../modules/video_render/main/interface \
+ $(LOCAL_PATH)/../../../../../voice_engine/main/interface \
+ $(LOCAL_PATH)/../../../../../system_wrappers/interface
+
+LOCAL_PRELINK_MODULE := false
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libandroid \
+ libwebrtc \
+ libGLESv2
+
+# the following line is for NDK build
+LOCAL_LDLIBS := $(LIBS_PATH)/VideoEngine_android_gcc.a -llog -lgcc
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h b/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
new file mode 100644
index 0000000..68ec601
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/* DO NOT EDIT THIS FILE - it is machine generated */
+#include <jni.h>
+/* Header for class org_webrtc_vieautotest_ViEAutotest */
+
+#ifndef _Included_org_webrtc_vieautotest_ViEAutotest
+#define _Included_org_webrtc_vieautotest_ViEAutotest
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+/*
+ * Class: org_webrtc_vieautotest_ViEAutotest
+ * Method: RunTest
+ * Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2
+(JNIEnv *, jobject, jint, jint, jobject, jobject);
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc b/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
new file mode 100644
index 0000000..2e55283
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <android/log.h>
+#include <pthread.h>
+#include <unistd.h>
+
+#include "org_webrtc_vieautotest_vie_autotest.h"
+
+#include "vie_autotest_android.h"
+
+#define WEBRTC_LOG_TAG "*WEBRTCN*"
+
+// VideoEngine data struct
+typedef struct
+{
+ JavaVM* jvm;
+} VideoEngineData;
+
+// Global variables
+JavaVM* webrtcGlobalVM;
+
+// Global variables visible in this file
+static VideoEngineData vieData;
+
+// "Local" functions (i.e. not Java accessible)
+#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 1024
+
+static bool GetSubAPIs(VideoEngineData& vieData);
+static bool ReleaseSubAPIs(VideoEngineData& vieData);
+
+//
+// General functions
+//
+
+// JNI_OnLoad
+jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/) {
+ if (!vm) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad did not receive a valid VM pointer");
+ return -1;
+ }
+
+ JNIEnv* env;
+ if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
+ JNI_VERSION_1_4)) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad could not get JNI env");
+ return -1;
+ }
+
+ // Init ViE data
+ vieData.jvm = vm;
+
+ return JNI_VERSION_1_4;
+}
+
+// Class: org_webrtc_vieautotest_ViEAutotest
+// Method: RunTest
+// Signature: (IILandroid/opengl/GLSurfaceView;Landroid/opengl/GLSurfaceView;)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_opengl_GLSurfaceView_2Landroid_opengl_GLSurfaceView_2(
+ JNIEnv* env,
+ jobject context,
+ jint testType,
+ jint subtestType,
+ jobject glView1,
+ jobject glView2)
+{
+ int numErrors = -1;
+ numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType, glView1,
+ glView2, vieData.jvm, env,
+ context);
+ return numErrors;
+}
+
+// Class: org_webrtc_vieautotest_ViEAutotest
+// Method: RunTest
+// Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
+JNIEXPORT jint JNICALL
+Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2(
+ JNIEnv* env,
+ jobject context,
+ jint testType,
+ jint subtestType,
+ jobject surfaceHolder1,
+ jobject surfaceHolder2)
+{
+ int numErrors = -1;
+ numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType,
+ surfaceHolder1, surfaceHolder2,
+ vieData.jvm, env, context);
+ return numErrors;
+}
+
+//
+//local function
+//
+
+bool GetSubAPIs(VideoEngineData& vieData) {
+ bool retVal = true;
+ //vieData.base = ViEBase::GetInterface(vieData.vie);
+ //if (vieData.base == NULL)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get Base API");
+ retVal = false;
+ }
+ return retVal;
+}
+
+bool ReleaseSubAPIs(VideoEngineData& vieData) {
+ bool releaseOk = true;
+ //if (vieData.base)
+ {
+ //if (vieData.base->Release() != 0)
+ if (false) {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release base sub-API failed");
+ releaseOk = false;
+ }
+ else {
+ //vieData.base = NULL;
+ }
+ }
+
+ return releaseOk;
+}
diff --git a/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png b/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png
new file mode 100644
index 0000000..c3e0a12
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml b/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml
new file mode 100644
index 0000000..1f2aaf9
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml
@@ -0,0 +1,64 @@
+<?xml version="1.0" encoding="utf-8"?>
+<FrameLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <RelativeLayout
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+ <LinearLayout
+ android:id="@+id/RemoteView"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:layout_weight="1">
+ <!-- log instead of video
+ <ImageView
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:scaleType="fitXY"
+ android:src="@drawable/logo" /> -->
+
+ </LinearLayout>
+ <LinearLayout
+ android:id="@+id/LocalView"
+ android:layout_width="120dip"
+ android:layout_height="120dip"
+ android:layout_weight="1">
+ <!-- <ImageView
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent"
+ android:scaleType="fitXY"
+ android:src="@drawable/logo" /> -->
+ </LinearLayout>
+ <LinearLayout
+ android:orientation="horizontal"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:layout_alignParentBottom="true">
+ <LinearLayout
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:layout_alignParentBottom="true">
+ <Spinner
+ android:id="@+id/testSpinner"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:prompt="@string/SpinnerTitle"
+ />
+ <Spinner
+ android:id="@+id/subtestSpinner"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:prompt="@string/SpinnerSubtest"
+ />
+ <Button
+ android:text="@string/run_button"
+ android:id="@+id/Button01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ </LinearLayout>
+
+ </LinearLayout>
+ </RelativeLayout>
+</FrameLayout>
diff --git a/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml b/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml
new file mode 100644
index 0000000..ba59c5e
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+
+<string name="app_name">ViEAutotest</string>
+<string name="TitleName">ViEAutotest</string>
+<string name="run_button">Run Test</string>
+<string name="SpinnerTitle">Test type...</string>
+<string-array name="test_array">
+ <item>Standard</item>
+ <item>API</item>
+ <item>Extended</item>
+ <item>Loopback</item>
+ <item>Custom</item>
+</string-array>
+<string name="SpinnerSubtest">Run...</string>
+<string-array name="subtest_array">
+ <item>All</item>
+ <item>Base</item>
+ <item>Capture</item>
+ <item>Codec</item>
+ <item>Mix</item>
+ <item>Encryption</item>
+ <item>External Codec</item>
+ <item>File</item>
+ <item>Image Process</item>
+ <item>Network</item>
+ <item>Render</item>
+ <item>RTP/RTCP</item>
+</string-array>
+
+</resources>
diff --git a/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java b/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
new file mode 100644
index 0000000..de228a8
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.vieautotest;
+
+import org.webrtc.vieautotest.R;
+
+import android.app.Activity;
+import android.os.Bundle;
+import android.util.Log;
+import android.widget.Button;
+import android.view.SurfaceView;
+import android.view.View;
+import android.view.SurfaceHolder;
+import android.widget.LinearLayout;
+import android.opengl.GLSurfaceView;
+import android.widget.Spinner;
+import android.widget.ArrayAdapter;
+import android.widget.AdapterView;
+
+public class ViEAutotest extends Activity
+ implements
+ AdapterView.OnItemSelectedListener,
+ View.OnClickListener {
+
+ private Thread testThread;
+ private Spinner testSpinner;
+ private Spinner subtestSpinner;
+ private int testSelection;
+ private int subTestSelection;
+
+ // View for remote video
+ private LinearLayout remoteSurface = null;
+ private GLSurfaceView glSurfaceView = null;
+ private SurfaceView surfaceView = null;
+
+ private LinearLayout localSurface = null;
+ private GLSurfaceView glLocalSurfaceView = null;
+ private SurfaceView localSurfaceView = null;
+
+ /** Called when the activity is first created. */
+ @Override
+ public void onCreate(Bundle savedInstanceState) {
+
+ Log.d("*WEBRTC*", "onCreate called");
+
+ super.onCreate(savedInstanceState);
+ setContentView(R.layout.main);
+
+ // Set the Start button action
+ final Button buttonStart = (Button) findViewById(R.id.Button01);
+ buttonStart.setOnClickListener(this);
+
+ // Set test spinner
+ testSpinner = (Spinner) findViewById(R.id.testSpinner);
+ ArrayAdapter<CharSequence> adapter =
+ ArrayAdapter.createFromResource(this, R.array.test_array,
+ android.R.layout.simple_spinner_item);
+
+ int resource = android.R.layout.simple_spinner_dropdown_item;
+ adapter.setDropDownViewResource(resource);
+ testSpinner.setAdapter(adapter);
+ testSpinner.setOnItemSelectedListener(this);
+
+ // Set sub test spinner
+ subtestSpinner = (Spinner) findViewById(R.id.subtestSpinner);
+ ArrayAdapter<CharSequence> subtestAdapter =
+ ArrayAdapter.createFromResource(this, R.array.subtest_array,
+ android.R.layout.simple_spinner_item);
+
+ subtestAdapter.setDropDownViewResource(resource);
+ subtestSpinner.setAdapter(subtestAdapter);
+ subtestSpinner.setOnItemSelectedListener(this);
+
+ remoteSurface = (LinearLayout) findViewById(R.id.RemoteView);
+ surfaceView = new SurfaceView(this);
+ remoteSurface.addView(surfaceView);
+
+ localSurface = (LinearLayout) findViewById(R.id.LocalView);
+ localSurfaceView = new SurfaceView(this);
+ localSurfaceView.setZOrderMediaOverlay(true);
+ localSurface.addView(localSurfaceView);
+
+ // Set members
+ testSelection = 0;
+ subTestSelection = 0;
+ }
+
+ public void onClick(View v) {
+ Log.d("*WEBRTC*", "Button clicked...");
+ switch (v.getId()) {
+ case R.id.Button01:
+ new Thread(new Runnable() {
+ public void run() {
+ Log.d("*WEBRTC*", "Calling RunTest...");
+ RunTest(testSelection, subTestSelection,
+ localSurfaceView, surfaceView);
+ Log.d("*WEBRTC*", "RunTest done");
+ }
+ }).start();
+ }
+ }
+
+ public void onItemSelected(AdapterView<?> parent, View v,
+ int position, long id) {
+
+ if (parent == (Spinner) findViewById(R.id.testSpinner)) {
+ testSelection = position;
+ } else {
+ subTestSelection = position;
+ }
+ }
+
+ public void onNothingSelected(AdapterView<?> parent) {
+ }
+
+ @Override
+ protected void onStart() {
+ super.onStart();
+ }
+
+ @Override
+ protected void onResume() {
+ super.onResume();
+ }
+
+ @Override
+ protected void onPause() {
+ super.onPause();
+ }
+
+ @Override
+ protected void onStop() {
+ super.onStop();
+ }
+
+ @Override
+ protected void onDestroy() {
+
+ super.onDestroy();
+ }
+
+ // C++ function performing the chosen test
+ // private native int RunTest(int testSelection, int subtestSelection,
+ // GLSurfaceView window1, GLSurfaceView window2);
+ private native int RunTest(int testSelection, int subtestSelection,
+ SurfaceView window1, SurfaceView window2);
+
+ // this is used to load the 'ViEAutotestJNIAPI' library on application
+ // startup.
+ static {
+ Log.d("*WEBRTC*", "Loading ViEAutotest...");
+ System.loadLibrary("webrtc-video-autotest-jni");
+ }
+}
diff --git a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc
new file mode 100644
index 0000000..591a567
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc
@@ -0,0 +1,28 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/test/auto_test/automated/legacy_fixture.h"
+
+#include "video_engine/test/auto_test/interface/vie_autotest.h"
+
+void LegacyFixture::SetUpTestCase() {
+ TwoWindowsFixture::SetUpTestCase();
+
+ // Create the test cases
+ tests_ = new ViEAutoTest(window_1_, window_2_);
+}
+
+void LegacyFixture::TearDownTestCase() {
+ delete tests_;
+
+ TwoWindowsFixture::TearDownTestCase();
+}
+
+ViEAutoTest* LegacyFixture::tests_ = NULL;
diff --git a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h
new file mode 100644
index 0000000..b452766
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
+#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
+
+#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
+
+// Inherited by old-style standard integration tests based on ViEAutoTest.
+class LegacyFixture : public TwoWindowsFixture {
+ public:
+ // Initializes ViEAutoTest in addition to the work done by ViEIntegrationTest.
+ static void SetUpTestCase();
+
+ // Releases anything allocated by SetupTestCase.
+ static void TearDownTestCase();
+
+ protected:
+ static ViEAutoTest* tests_;
+};
+
+#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
diff --git a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc
new file mode 100644
index 0000000..d181aa5
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
+
+#include "video_engine/test/auto_test/helpers/vie_window_creator.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
+
+void TwoWindowsFixture::SetUpTestCase() {
+ window_creator_ = new ViEWindowCreator();
+
+ ViEAutoTestWindowManagerInterface* window_manager =
+ window_creator_->CreateTwoWindows();
+
+ window_1_ = window_manager->GetWindow1();
+ window_2_ = window_manager->GetWindow2();
+}
+
+void TwoWindowsFixture::TearDownTestCase() {
+ window_creator_->TerminateWindows();
+ delete window_creator_;
+}
+
+ViEWindowCreator* TwoWindowsFixture::window_creator_ = NULL;
+void* TwoWindowsFixture::window_1_ = NULL;
+void* TwoWindowsFixture::window_2_ = NULL;
diff --git a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h
new file mode 100644
index 0000000..175a42d
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
+
+#include "gtest/gtest.h"
+
+class ViEWindowCreator;
+class ViEAutoTest;
+
+// Meant to be inherited by all standard test who require two windows.
+class TwoWindowsFixture : public testing::Test {
+ public:
+ // Launches two windows in a platform-dependent manner and stores the handles
+ // in the window_1_ and window_2_ fields.
+ static void SetUpTestCase();
+
+ // Releases anything allocated by SetupTestCase.
+ static void TearDownTestCase();
+
+ protected:
+ static void* window_1_;
+ static void* window_2_;
+ static ViEWindowCreator* window_creator_;
+};
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc
new file mode 100644
index 0000000..a2c65a2
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/**
+ * Runs "extended" integration tests.
+ */
+
+#include "gtest/gtest.h"
+#include "legacy_fixture.h"
+#include "vie_autotest.h"
+
+namespace {
+
+class ViEApiIntegrationTest : public LegacyFixture {
+};
+
+TEST_F(ViEApiIntegrationTest, RunsBaseTestWithoutErrors) {
+ tests_->ViEBaseAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsCaptureTestWithoutErrors) {
+ tests_->ViECaptureAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsCodecTestWithoutErrors) {
+ tests_->ViECodecAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsEncryptionTestWithoutErrors) {
+ tests_->ViEEncryptionAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsFileTestWithoutErrors) {
+ tests_->ViEFileAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsImageProcessTestWithoutErrors) {
+ tests_->ViEImageProcessAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsNetworkTestWithoutErrors) {
+ tests_->ViENetworkAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsRenderTestWithoutErrors) {
+ tests_->ViERenderAPITest();
+}
+
+TEST_F(ViEApiIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
+ tests_->ViERtpRtcpAPITest();
+}
+
+} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
new file mode 100644
index 0000000..facb659
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/**
+ * Runs "extended" integration tests.
+ */
+
+#include "gtest/gtest.h"
+#include "legacy_fixture.h"
+#include "vie_autotest.h"
+
+namespace {
+
+class ViEExtendedIntegrationTest : public LegacyFixture {
+};
+
+TEST_F(ViEExtendedIntegrationTest, RunsBaseTestWithoutErrors) {
+ tests_->ViEBaseExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsCaptureTestWithoutErrors) {
+ tests_->ViECaptureExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsCodecTestWithoutErrors) {
+ tests_->ViECodecExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsEncryptionTestWithoutErrors) {
+ tests_->ViEEncryptionExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsFileTestWithoutErrors) {
+ tests_->ViEFileExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsImageProcessTestWithoutErrors) {
+ tests_->ViEImageProcessExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsNetworkTestWithoutErrors) {
+ tests_->ViENetworkExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsRenderTestWithoutErrors) {
+ tests_->ViERenderExtendedTest();
+}
+
+TEST_F(ViEExtendedIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
+ tests_->ViERtpRtcpExtendedTest();
+}
+
+} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc
new file mode 100644
index 0000000..f8740be
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc
@@ -0,0 +1,150 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <ctime>
+
+#include "gtest/gtest.h"
+#include "gflags/gflags.h"
+#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
+#include "video_engine/test/auto_test/helpers/bit_flip_encryption.h"
+#include "video_engine/test/auto_test/helpers/random_encryption.h"
+#include "video_engine/test/auto_test/helpers/vie_window_creator.h"
+#include "video_engine/test/auto_test/interface/tb_capture_device.h"
+#include "video_engine/test/auto_test/interface/tb_interfaces.h"
+#include "video_engine/test/auto_test/interface/tb_video_channel.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
+#include "video_engine/test/auto_test/primitives/general_primitives.h"
+
+namespace {
+
+DEFINE_int32(rtp_fuzz_test_rand_seed, 0, "The rand seed to use for "
+ "the RTP fuzz test. Defaults to time(). 0 cannot be specified.");
+
+class ViERtpFuzzTest : public TwoWindowsFixture {
+ protected:
+ TbVideoChannel* video_channel_;
+ TbInterfaces* video_engine_;
+ TbCaptureDevice* capture_device_;
+
+ void SetUp() {
+ video_engine_ = new TbInterfaces(
+ "ViERtpTryInjectingRandomPacketsIntoRtpStream");
+ video_channel_ = new TbVideoChannel(
+ *video_engine_, webrtc::kVideoCodecVP8);
+ capture_device_ = new TbCaptureDevice(*video_engine_);
+
+ capture_device_->ConnectTo(video_channel_->videoChannel);
+
+ // Enable PLI RTCP, which will allow the video engine to recover better.
+ video_engine_->rtp_rtcp->SetKeyFrameRequestMethod(
+ video_channel_->videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
+
+ video_channel_->StartReceive();
+ video_channel_->StartSend();
+
+ RenderInWindow(
+ video_engine_->render, capture_device_->captureId, window_1_, 0);
+ RenderInWindow(
+ video_engine_->render, video_channel_->videoChannel, window_2_, 1);
+ }
+
+ void TearDown() {
+ delete capture_device_;
+ delete video_channel_;
+ delete video_engine_;
+ }
+
+ unsigned int FetchRandSeed() {
+ if (FLAGS_rtp_fuzz_test_rand_seed != 0) {
+ return FLAGS_rtp_fuzz_test_rand_seed;
+ }
+ return std::time(NULL);
+ }
+
+ // Pass in a number [0, 1] which will be the bit flip probability per byte.
+ void BitFlipFuzzTest(float flip_probability) {
+ unsigned int rand_seed = FetchRandSeed();
+ ViETest::Log("Running test with rand seed %d.", rand_seed);
+
+ ViETest::Log("Running as usual. You should see video output.");
+ AutoTestSleep(2000);
+
+ ViETest::Log("Starting to flip bits in packets (%f%% chance per byte).",
+ flip_probability * 100);
+ BitFlipEncryption bit_flip_encryption(rand_seed, flip_probability);
+ video_engine_->encryption->RegisterExternalEncryption(
+ video_channel_->videoChannel, bit_flip_encryption);
+
+ AutoTestSleep(5000);
+
+ ViETest::Log("Back to normal. Flipped %d bits.",
+ bit_flip_encryption.flip_count());
+ video_engine_->encryption->DeregisterExternalEncryption(
+ video_channel_->videoChannel);
+
+ AutoTestSleep(5000);
+ }
+};
+
+TEST_F(ViERtpFuzzTest, VideoEngineDealsWithASmallNumberOfTamperedPackets) {
+ // Try 0.005% bit flip chance per byte.
+ BitFlipFuzzTest(0.00005f);
+}
+
+TEST_F(ViERtpFuzzTest, VideoEngineDealsWithAMediumNumberOfTamperedPackets) {
+ // Try 0.05% bit flip chance per byte.
+ BitFlipFuzzTest(0.0005f);
+}
+
+TEST_F(ViERtpFuzzTest, VideoEngineDealsWithALargeNumberOfTamperedPackets) {
+ // Try 0.5% bit flip chance per byte.
+ BitFlipFuzzTest(0.005f);
+}
+
+TEST_F(ViERtpFuzzTest, VideoEngineDealsWithAVeryLargeNumberOfTamperedPackets) {
+ // Try 5% bit flip chance per byte.
+ BitFlipFuzzTest(0.05f);
+}
+
+TEST_F(ViERtpFuzzTest,
+ VideoEngineDealsWithAExtremelyLargeNumberOfTamperedPackets) {
+ // Try 25% bit flip chance per byte (madness!)
+ BitFlipFuzzTest(0.25f);
+}
+
+TEST_F(ViERtpFuzzTest, VideoEngineDealsWithSeveralPeriodsOfTamperedPackets) {
+ // Try 0.05% bit flip chance per byte.
+ BitFlipFuzzTest(0.0005f);
+ BitFlipFuzzTest(0.0005f);
+ BitFlipFuzzTest(0.0005f);
+}
+
+TEST_F(ViERtpFuzzTest, VideoEngineRecoversAfterSomeCompletelyRandomPackets) {
+ unsigned int rand_seed = FetchRandSeed();
+ ViETest::Log("Running test with rand seed %d.", rand_seed);
+
+ ViETest::Log("Running as usual. You should see video output.");
+ AutoTestSleep(2000);
+
+ ViETest::Log("Injecting completely random packets...");
+ RandomEncryption random_encryption(rand_seed);
+ video_engine_->encryption->RegisterExternalEncryption(
+ video_channel_->videoChannel, random_encryption);
+
+ AutoTestSleep(5000);
+
+ ViETest::Log("Back to normal.");
+ video_engine_->encryption->DeregisterExternalEncryption(
+ video_channel_->videoChannel);
+
+ AutoTestSleep(5000);
+}
+
+}
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
new file mode 100644
index 0000000..e85b6e4
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
@@ -0,0 +1,71 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the "standard" suite of integration tests, implemented
+ * as a GUnit test. This file is a part of the effort to try to automate all
+ * tests in this section of the code. Currently, this code makes no attempt
+ * to verify any video output - it only checks for direct errors.
+ */
+
+#include <cstdio>
+
+#include "gflags/gflags.h"
+#include "gtest/gtest.h"
+#include "legacy_fixture.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "vie_autotest.h"
+#include "vie_autotest_window_manager_interface.h"
+#include "vie_to_file_renderer.h"
+#include "vie_window_creator.h"
+#include "testsupport/metrics/video_metrics.h"
+
+namespace {
+
+class ViEStandardIntegrationTest : public LegacyFixture {
+};
+
+TEST_F(ViEStandardIntegrationTest, RunsBaseTestWithoutErrors) {
+ tests_->ViEBaseStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsCodecTestWithoutErrors) {
+ tests_->ViECodecStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsCaptureTestWithoutErrors) {
+ tests_->ViECaptureStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsEncryptionTestWithoutErrors) {
+ tests_->ViEEncryptionStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsFileTestWithoutErrors) {
+ tests_->ViEFileStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsImageProcessTestWithoutErrors) {
+ tests_->ViEImageProcessStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsNetworkTestWithoutErrors) {
+ tests_->ViENetworkStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsRenderTestWithoutErrors) {
+ tests_->ViERenderStandardTest();
+}
+
+TEST_F(ViEStandardIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
+ tests_->ViERtpRtcpStandardTest();
+}
+
+} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc
new file mode 100644
index 0000000..861af8a
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc
@@ -0,0 +1,253 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <vector>
+
+#include "gflags/gflags.h"
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/metrics/video_metrics.h"
+#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
+#include "video_engine/test/auto_test/interface/vie_autotest.h"
+#include "video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
+#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
+
+namespace {
+
+// The input file must be QCIF since I420 gets scaled to that in the tests
+// (it is so bandwidth-heavy we have no choice). Our comparison algorithms
+// wouldn't like scaling, so this will work when we compare with the original.
+const int kInputWidth = 176;
+const int kInputHeight = 144;
+const int kVerifyingTestMaxNumAttempts = 3;
+
+class ViEVideoVerificationTest : public testing::Test {
+ protected:
+ void SetUp() {
+ input_file_ = webrtc::test::ResourcePath("paris_qcif", "yuv");
+ }
+
+ void TearDown() {
+ TearDownFileRenderer(local_file_renderer_);
+ TearDownFileRenderer(remote_file_renderer_);
+ }
+
+ void InitializeFileRenderers() {
+ local_file_renderer_ = new ViEToFileRenderer();
+ remote_file_renderer_ = new ViEToFileRenderer();
+ SetUpLocalFileRenderer(local_file_renderer_);
+ SetUpRemoteFileRenderer(remote_file_renderer_);
+ }
+
+ void SetUpLocalFileRenderer(ViEToFileRenderer* file_renderer) {
+ SetUpFileRenderer(file_renderer, "-local-preview.yuv");
+ }
+
+ void SetUpRemoteFileRenderer(ViEToFileRenderer* file_renderer) {
+ SetUpFileRenderer(file_renderer, "-remote.yuv");
+ }
+
+ // Must be called manually inside the tests.
+ void StopRenderers() {
+ local_file_renderer_->StopRendering();
+ remote_file_renderer_->StopRendering();
+ }
+
+ void TearDownFileRenderer(ViEToFileRenderer* file_renderer) {
+ assert(file_renderer);
+ bool test_failed = ::testing::UnitTest::GetInstance()->
+ current_test_info()->result()->Failed();
+ if (test_failed) {
+ // Leave the files for analysis if the test failed.
+ file_renderer->SaveOutputFile("failed-");
+ } else {
+ // No reason to keep the files if we succeeded.
+ file_renderer->DeleteOutputFile();
+ }
+ delete file_renderer;
+ }
+
+ void CompareFiles(const std::string& reference_file,
+ const std::string& test_file,
+ double* psnr_result, double *ssim_result) {
+ webrtc::test::QualityMetricsResult psnr;
+ int error = I420PSNRFromFiles(reference_file.c_str(), test_file.c_str(),
+ kInputWidth, kInputHeight, &psnr);
+
+ EXPECT_EQ(0, error) << "PSNR routine failed - output files missing?";
+ *psnr_result = psnr.average;
+
+ webrtc::test::QualityMetricsResult ssim;
+ error = I420SSIMFromFiles(reference_file.c_str(), test_file.c_str(),
+ kInputWidth, kInputHeight, &ssim);
+ EXPECT_EQ(0, error) << "SSIM routine failed - output files missing?";
+ *ssim_result = ssim.average;
+
+ ViETest::Log("Results: PSNR is %f (dB), SSIM is %f (1 is perfect)",
+ psnr.average, ssim.average);
+ }
+
+ std::string input_file_;
+ ViEToFileRenderer* local_file_renderer_;
+ ViEToFileRenderer* remote_file_renderer_;
+ ViEFileBasedComparisonTests tests_;
+
+ private:
+ void SetUpFileRenderer(ViEToFileRenderer* file_renderer,
+ const std::string& suffix) {
+ std::string test_case_name =
+ ::testing::UnitTest::GetInstance()->current_test_info()->name();
+
+ std::string output_path = ViETest::GetResultOutputPath();
+ std::string filename = test_case_name + suffix;
+
+ if (!file_renderer->PrepareForRendering(output_path, filename)) {
+ FAIL() << "Could not open output file " << filename <<
+ " for writing.";
+ }
+ }
+};
+
+TEST_F(ViEVideoVerificationTest, RunsBaseStandardTestWithoutErrors) {
+ // The I420 test should give pretty good values since it's a lossless codec
+ // running on the default bitrate. It should average about 30 dB but there
+ // may be cases where it dips as low as 26 under adverse conditions. That's
+ // why we have a retrying mechanism in place for this test.
+ const double kExpectedMinimumPSNR = 30;
+ const double kExpectedMinimumSSIM = 0.95;
+
+ for (int attempt = 0; attempt < kVerifyingTestMaxNumAttempts; attempt++) {
+ InitializeFileRenderers();
+ ASSERT_TRUE(tests_.TestCallSetup(input_file_, kInputWidth, kInputHeight,
+ local_file_renderer_,
+ remote_file_renderer_));
+ std::string output_file = remote_file_renderer_->GetFullOutputPath();
+ StopRenderers();
+
+ double actual_psnr = 0;
+ double actual_ssim = 0;
+ CompareFiles(input_file_, output_file, &actual_psnr, &actual_ssim);
+
+ if (actual_psnr >= kExpectedMinimumPSNR &&
+ actual_ssim >= kExpectedMinimumSSIM) {
+ // Test succeeded!
+ return;
+ }
+ }
+
+ ADD_FAILURE() << "Failed to achieve PSNR " << kExpectedMinimumPSNR <<
+ " and SSIM " << kExpectedMinimumSSIM << " after " <<
+ kVerifyingTestMaxNumAttempts << " attempts.";
+}
+
+TEST_F(ViEVideoVerificationTest, RunsCodecTestWithoutErrors) {
+ // We compare the local and remote here instead of with the original.
+ // The reason is that it is hard to say when the three consecutive tests
+ // switch over into each other, at which point we would have to restart the
+ // original to get a fair comparison.
+ //
+ // The PSNR and SSIM values are quite low here, and they have to be since
+ // the codec switches will lead to lag in the output. This is considered
+ // acceptable, but it probably shouldn't get worse than this.
+ const double kExpectedMinimumPSNR = 20;
+ const double kExpectedMinimumSSIM = 0.7;
+
+ for (int attempt = 0; attempt < kVerifyingTestMaxNumAttempts; attempt++) {
+ InitializeFileRenderers();
+ ASSERT_TRUE(tests_.TestCodecs(input_file_, kInputWidth, kInputHeight,
+ local_file_renderer_,
+ remote_file_renderer_));
+ std::string reference_file = local_file_renderer_->GetFullOutputPath();
+ std::string output_file = remote_file_renderer_->GetFullOutputPath();
+ StopRenderers();
+
+ double actual_psnr = 0;
+ double actual_ssim = 0;
+ CompareFiles(reference_file, output_file, &actual_psnr, &actual_ssim);
+
+ if (actual_psnr >= kExpectedMinimumPSNR &&
+ actual_ssim >= kExpectedMinimumSSIM) {
+ // Test succeeded!
+ return;
+ }
+ }
+}
+
+// Runs a whole stack processing with tracking of which frames are dropped
+// in the encoder. The local and remote file will not be of equal size because
+// of unknown reasons. Tests show that they start at the same frame, which is
+// the important thing when doing frame-to-frame comparison with PSNR/SSIM.
+TEST_F(ViEVideoVerificationTest, RunsFullStackWithoutErrors) {
+ FrameDropDetector detector;
+ local_file_renderer_ = new ViEToFileRenderer();
+ remote_file_renderer_ = new FrameDropMonitoringRemoteFileRenderer(&detector);
+ SetUpLocalFileRenderer(local_file_renderer_);
+ SetUpRemoteFileRenderer(remote_file_renderer_);
+
+ // Set a low bit rate so the encoder budget will be tight, causing it to drop
+ // frames every now and then.
+ const int kBitRateKbps = 50;
+ const int kPacketLossPercent = 10;
+ const int kNetworkDelayMs = 100;
+ ViETest::Log("Bit rate : %5d kbps", kBitRateKbps);
+ ViETest::Log("Packet loss : %5d %%", kPacketLossPercent);
+ ViETest::Log("Network delay: %5d ms", kNetworkDelayMs);
+ tests_.TestFullStack(input_file_, kInputWidth, kInputHeight, kBitRateKbps,
+ kPacketLossPercent, kNetworkDelayMs,
+ local_file_renderer_, remote_file_renderer_, &detector);
+ const std::string reference_file = local_file_renderer_->GetFullOutputPath();
+ const std::string output_file = remote_file_renderer_->GetFullOutputPath();
+ StopRenderers();
+
+ detector.CalculateResults();
+ detector.PrintReport();
+
+ if (detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kRendered) !=
+ detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kDecoded)) {
+ detector.PrintDebugDump();
+ }
+
+ ASSERT_EQ(detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kRendered),
+ detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kDecoded))
+ << "The number of dropped frames on the decode and render steps are not "
+ "equal. This may be because we have a major problem in the buffers of "
+ "the ViEToFileRenderer?";
+
+ // We may have dropped frames during the processing, which means the output
+ // file does not contain all the frames that are present in the input file.
+ // To make the quality measurement correct, we must adjust the output file to
+ // that by copying the last successful frame into the place where the dropped
+ // frame would be, for all dropped frames.
+ const int frame_length_in_bytes = 3 * kInputHeight * kInputWidth / 2;
+ ViETest::Log("Frame length: %d bytes", frame_length_in_bytes);
+ std::vector<Frame*> all_frames = detector.GetAllFrames();
+ FixOutputFileForComparison(output_file, frame_length_in_bytes, all_frames);
+
+ // Verify all sent frames are present in the output file.
+ size_t output_file_size = webrtc::test::GetFileSize(output_file);
+ EXPECT_EQ(all_frames.size(), output_file_size / frame_length_in_bytes)
+ << "The output file size is incorrect. It should be equal to the number "
+ "of frames multiplied by the frame size. This will likely affect "
+ "PSNR/SSIM calculations in a bad way.";
+
+ // We are running on a lower bitrate here so we need to settle for somewhat
+ // lower PSNR and SSIM values.
+ double actual_psnr = 0;
+ double actual_ssim = 0;
+ CompareFiles(reference_file, output_file, &actual_psnr, &actual_ssim);
+
+ const double kExpectedMinimumPSNR = 24;
+ const double kExpectedMinimumSSIM = 0.7;
+
+ EXPECT_GE(actual_psnr, kExpectedMinimumPSNR);
+ EXPECT_GE(actual_ssim, kExpectedMinimumSSIM);
+}
+
+} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc
new file mode 100644
index 0000000..423ad5a
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/test/auto_test/helpers/bit_flip_encryption.h"
+
+#include <cstdlib>
+
+#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
+
+float NormalizedRand() {
+ return static_cast<float>(rand()) /
+ static_cast<float>(RAND_MAX);
+}
+
+BitFlipEncryption::BitFlipEncryption(unsigned int rand_seed,
+ float flip_probability)
+ : flip_probability_(flip_probability),
+ flip_count_(0) {
+ srand(rand_seed);
+}
+
+void BitFlipEncryption::FlipSomeBitsInData(const unsigned char* in_data,
+ unsigned char* out_data,
+ int bytes_in, int* bytes_out) {
+ for (int i = 0; i < bytes_in; i++) {
+ out_data[i] = in_data[i];
+
+ if (NormalizedRand() < flip_probability_) {
+ int bit_to_flip = rand() % 8;
+ out_data[i] ^= 1 << bit_to_flip;
+ flip_count_++;
+ }
+ }
+ *bytes_out = bytes_in;
+}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h
new file mode 100644
index 0000000..ccc6974
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
+#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
+
+#include "video_engine/include/vie_encryption.h"
+
+// This encryption scheme will randomly flip bits every now and then in the
+// input data.
+class BitFlipEncryption : public webrtc::Encryption {
+ public:
+ // Args:
+ // rand_seed: the seed to initialize the test's random generator with.
+ // flip_probability: A number [0, 1] which is the percentage chance a bit
+ // gets flipped in a particular byte.
+ BitFlipEncryption(unsigned int rand_seed, float flip_probability);
+
+ virtual void encrypt(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in, int* bytes_out) {
+ FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
+ }
+
+ virtual void decrypt(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in, int* bytes_out) {
+ FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
+ }
+
+ virtual void encrypt_rtcp(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in,
+ int* bytes_out) {
+ FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
+ }
+
+ virtual void decrypt_rtcp(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in,
+ int* bytes_out) {
+ FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
+ }
+
+ int64_t flip_count() const { return flip_count_; }
+
+ private:
+ // The flip probability ([0, 1]).
+ float flip_probability_;
+ // The number of bits we've flipped so far.
+ int64_t flip_count_;
+
+ // Flips some bits in the data at random.
+ void FlipSomeBitsInData(const unsigned char *in_data, unsigned char* out_data,
+ int bytes_in, int* bytes_out);
+};
+
+#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc
new file mode 100644
index 0000000..04904b5
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/test/auto_test/helpers/random_encryption.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <cmath>
+
+#include "video_engine/vie_defines.h"
+
+static int Saturate(int value, int min, int max) {
+ return std::min(std::max(value, min), max);
+}
+
+RandomEncryption::RandomEncryption(unsigned int rand_seed) {
+ srand(rand_seed);
+}
+
+// Generates some completely random data with roughly the right length.
+void RandomEncryption::GenerateRandomData(unsigned char* out_data, int bytes_in,
+ int* bytes_out) {
+ int out_length = MakeUpSimilarLength(bytes_in);
+ for (int i = 0; i < out_length; i++) {
+ // The modulo will skew the random distribution a bit, but I think it
+ // will be random enough.
+ out_data[i] = static_cast<unsigned char>(rand() % 256);
+ }
+ *bytes_out = out_length;
+}
+
+// Makes up a length within +- 50 of the original length, without
+// overstepping the contract for encrypt / decrypt.
+int RandomEncryption::MakeUpSimilarLength(int original_length) {
+ int sign = rand() - RAND_MAX / 2;
+ int length = original_length + sign * rand() % 50;
+
+ return Saturate(length, 0, static_cast<int>(webrtc::kViEMaxMtu));
+}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h
new file mode 100644
index 0000000..cbca844
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
+#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
+
+#include "video_engine/include/vie_encryption.h"
+
+// These algorithms attempt to create an uncrackable encryption
+// scheme by completely disregarding the input data.
+class RandomEncryption : public webrtc::Encryption {
+ public:
+ explicit RandomEncryption(unsigned int rand_seed);
+
+ virtual void encrypt(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in, int* bytes_out) {
+ GenerateRandomData(out_data, bytes_in, bytes_out);
+ }
+
+ virtual void decrypt(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in, int* bytes_out) {
+ GenerateRandomData(out_data, bytes_in, bytes_out);
+ }
+
+ virtual void encrypt_rtcp(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in,
+ int* bytes_out) {
+ GenerateRandomData(out_data, bytes_in, bytes_out);
+ }
+
+ virtual void decrypt_rtcp(int channel_no, unsigned char* in_data,
+ unsigned char* out_data, int bytes_in,
+ int* bytes_out) {
+ GenerateRandomData(out_data, bytes_in, bytes_out);
+ }
+
+ private:
+ // Generates some completely random data with roughly the right length.
+ void GenerateRandomData(unsigned char* out_data, int bytes_in,
+ int* bytes_out);
+
+ // Makes up a length within +- 50 of the original length, without
+ // overstepping the contract for encrypt / decrypt.
+ int MakeUpSimilarLength(int original_length);
+};
+
+#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc
new file mode 100644
index 0000000..f7dae2f
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "vie_fake_camera.h"
+
+#include <assert.h>
+
+#include "vie_capture.h"
+#include "vie_file_capture_device.h"
+#include "thread_wrapper.h"
+
+// This callback runs the camera thread:
+bool StreamVideoFileRepeatedlyIntoCaptureDevice(void* data) {
+ ViEFileCaptureDevice* file_capture_device =
+ reinterpret_cast<ViEFileCaptureDevice*>(data);
+
+ // We want to interrupt the camera feeding thread every now and then in order
+ // to follow the contract for the system_wrappers thread library. 1.5 seconds
+ // seems about right here.
+ uint64_t time_slice_ms = 1500;
+ uint32_t max_fps = 30;
+
+ file_capture_device->ReadFileFor(time_slice_ms, max_fps);
+
+ return true;
+}
+
+ViEFakeCamera::ViEFakeCamera(webrtc::ViECapture* capture_interface)
+ : capture_interface_(capture_interface),
+ capture_id_(-1),
+ camera_thread_(NULL),
+ file_capture_device_(NULL) {
+}
+
+ViEFakeCamera::~ViEFakeCamera() {
+}
+
+bool ViEFakeCamera::StartCameraInNewThread(
+ const std::string& i420_test_video_path, int width, int height) {
+
+ assert(file_capture_device_ == NULL && camera_thread_ == NULL);
+
+ webrtc::ViEExternalCapture* externalCapture;
+ int result = capture_interface_->
+ AllocateExternalCaptureDevice(capture_id_, externalCapture);
+ if (result != 0) {
+ return false;
+ }
+
+ file_capture_device_ = new ViEFileCaptureDevice(externalCapture);
+ if (!file_capture_device_->OpenI420File(i420_test_video_path,
+ width,
+ height)) {
+ return false;
+ }
+
+ // Set up a thread which runs the fake camera. The capturer object is
+ // thread-safe.
+ camera_thread_ = webrtc::ThreadWrapper::CreateThread(
+ StreamVideoFileRepeatedlyIntoCaptureDevice, file_capture_device_);
+ unsigned int id;
+ camera_thread_->Start(id);
+
+ return true;
+}
+
+bool ViEFakeCamera::StopCamera() {
+ assert(file_capture_device_ != NULL && camera_thread_ != NULL);
+
+ camera_thread_->Stop();
+ file_capture_device_->CloseFile();
+
+ int result = capture_interface_->ReleaseCaptureDevice(capture_id_);
+
+ delete camera_thread_;
+ delete file_capture_device_;
+ camera_thread_ = NULL;
+ file_capture_device_ = NULL;
+
+ return result == 0;
+}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h
new file mode 100644
index 0000000..0b9c2e7
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
+
+#include <string>
+
+namespace webrtc {
+class ViECapture;
+class ThreadWrapper;
+}
+
+class ViEFileCaptureDevice;
+
+// Registers an external capture device with the provided capture interface
+// and starts running a fake camera by reading frames from a file. The frame-
+// reading code runs in a separate thread which makes it possible to run tests
+// while the fake camera feeds data into the system. This class is not thread-
+// safe in itself (but handles its own thread in a safe manner).
+class ViEFakeCamera {
+ public:
+ // The argument is the capture interface to register with.
+ explicit ViEFakeCamera(webrtc::ViECapture* capture_interface);
+ virtual ~ViEFakeCamera();
+
+ // Runs the scenario in the class comments.
+ bool StartCameraInNewThread(const std::string& i420_test_video_path,
+ int width,
+ int height);
+ // Stops the camera and cleans up everything allocated by the start method.
+ bool StopCamera();
+
+ int capture_id() const { return capture_id_; }
+
+ private:
+ webrtc::ViECapture* capture_interface_;
+
+ int capture_id_;
+ webrtc::ThreadWrapper* camera_thread_;
+ ViEFileCaptureDevice* file_capture_device_;
+};
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc
new file mode 100644
index 0000000..e4caa2a
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc
@@ -0,0 +1,106 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "vie_file_capture_device.h"
+
+#include <assert.h>
+
+#include "common_types.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "module_common_types.h"
+#include "vie_autotest_defines.h"
+#include "vie_capture.h"
+#include "tick_util.h"
+
+// This class ensures we are not exceeding the max FPS.
+class FramePacemaker {
+ public:
+ explicit FramePacemaker(uint32_t max_fps)
+ : time_per_frame_ms_(1000 / max_fps) {
+ frame_start_ = webrtc::TickTime::MillisecondTimestamp();
+ }
+
+ void SleepIfNecessary(webrtc::EventWrapper* sleeper) {
+ uint64_t now = webrtc::TickTime::MillisecondTimestamp();
+ if (now - frame_start_ < time_per_frame_ms_) {
+ sleeper->Wait(time_per_frame_ms_ - (now - frame_start_));
+ }
+ }
+
+ private:
+ uint64_t frame_start_;
+ uint64_t time_per_frame_ms_;
+};
+
+ViEFileCaptureDevice::ViEFileCaptureDevice(
+ webrtc::ViEExternalCapture* input_sink)
+ : input_sink_(input_sink),
+ input_file_(NULL) {
+ mutex_ = webrtc::CriticalSectionWrapper::CreateCriticalSection();
+}
+
+ViEFileCaptureDevice::~ViEFileCaptureDevice() {
+ delete mutex_;
+}
+
+bool ViEFileCaptureDevice::OpenI420File(const std::string& path,
+ int width,
+ int height) {
+ webrtc::CriticalSectionScoped cs(*mutex_);
+ assert(input_file_ == NULL);
+
+ input_file_ = std::fopen(path.c_str(), "rb");
+ if (input_file_ == NULL) {
+ return false;
+ }
+
+ frame_length_ = 3 * width * height / 2;
+ width_ = width;
+ height_ = height;
+ return true;
+}
+
+void ViEFileCaptureDevice::ReadFileFor(uint64_t time_slice_ms,
+ uint32_t max_fps) {
+ webrtc::CriticalSectionScoped cs(*mutex_);
+ assert(input_file_ != NULL);
+
+ unsigned char* frame_buffer = new unsigned char[frame_length_];
+
+ webrtc::EventWrapper* sleeper = webrtc::EventWrapper::Create();
+
+ uint64_t start_time_ms = webrtc::TickTime::MillisecondTimestamp();
+ uint64_t elapsed_ms = 0;
+
+ while (elapsed_ms < time_slice_ms) {
+ FramePacemaker pacemaker(max_fps);
+ int read = std::fread(frame_buffer, 1, frame_length_, input_file_);
+
+ if (std::feof(input_file_)) {
+ std::rewind(input_file_);
+ }
+ input_sink_->IncomingFrame(frame_buffer, read, width_, height_,
+ webrtc::kVideoI420,
+ webrtc::TickTime::MillisecondTimestamp());
+
+ pacemaker.SleepIfNecessary(sleeper);
+ elapsed_ms = webrtc::TickTime::MillisecondTimestamp() - start_time_ms;
+ }
+
+ delete sleeper;
+ delete[] frame_buffer;
+}
+
+void ViEFileCaptureDevice::CloseFile() {
+ webrtc::CriticalSectionScoped cs(*mutex_);
+ assert(input_file_ != NULL);
+
+ std::fclose(input_file_);
+}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h
new file mode 100644
index 0000000..5e62c48
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
+
+#include <cstdio>
+
+#include <string>
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ViEExternalCapture;
+}
+
+// This class opens a i420 file and feeds it into a ExternalCapture instance,
+// thereby acting as a faked capture device with deterministic input.
+class ViEFileCaptureDevice {
+ public:
+ // The input sink is where to send the I420 video frames.
+ explicit ViEFileCaptureDevice(webrtc::ViEExternalCapture* input_sink);
+ virtual ~ViEFileCaptureDevice();
+
+ // Opens the provided I420 file and interprets it according to the provided
+ // width and height. Returns false if the file doesn't exist.
+ bool OpenI420File(const std::string& path, int width, int height);
+
+ // Reads the previously opened file for at most time_slice_ms milliseconds,
+ // after which it will return. It will make sure to sleep accordingly so we
+ // do not send more than max_fps cap (we may send less, though).
+ void ReadFileFor(uint64_t time_slice_ms, uint32_t max_fps);
+
+ // Closes the opened input file.
+ void CloseFile();
+
+ private:
+ webrtc::ViEExternalCapture* input_sink_;
+
+ std::FILE* input_file_;
+ webrtc::CriticalSectionWrapper* mutex_;
+
+ WebRtc_UWord32 frame_length_;
+ WebRtc_UWord8* frame_buffer_;
+ WebRtc_UWord32 width_;
+ WebRtc_UWord32 height_;
+};
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc
new file mode 100644
index 0000000..d9c8da3
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_to_file_renderer.h"
+
+#include <assert.h>
+
+ViEToFileRenderer::ViEToFileRenderer()
+ : output_file_(NULL) {
+}
+
+ViEToFileRenderer::~ViEToFileRenderer() {
+}
+
+bool ViEToFileRenderer::PrepareForRendering(
+ const std::string& output_path,
+ const std::string& output_filename) {
+
+ assert(output_file_ == NULL);
+
+ output_file_ = std::fopen((output_path + output_filename).c_str(), "wb");
+ if (output_file_ == NULL) {
+ return false;
+ }
+
+ output_filename_ = output_filename;
+ output_path_ = output_path;
+ return true;
+}
+
+void ViEToFileRenderer::StopRendering() {
+ assert(output_file_ != NULL);
+ std::fclose(output_file_);
+ output_file_ = NULL;
+}
+
+bool ViEToFileRenderer::SaveOutputFile(const std::string& prefix) {
+ assert(output_file_ == NULL && output_filename_ != "");
+ if (std::rename((output_path_ + output_filename_).c_str(),
+ (output_path_ + prefix + output_filename_).c_str()) != 0) {
+ std::perror("Failed to rename output file");
+ return false;
+ }
+ ForgetOutputFile();
+ return true;
+}
+
+bool ViEToFileRenderer::DeleteOutputFile() {
+ assert(output_file_ == NULL && output_filename_ != "");
+ if (std::remove((output_path_ + output_filename_).c_str()) != 0) {
+ std::perror("Failed to delete output file");
+ return false;
+ }
+ ForgetOutputFile();
+ return true;
+}
+
+const std::string ViEToFileRenderer::GetFullOutputPath() const {
+ return output_path_ + output_filename_;
+}
+
+void ViEToFileRenderer::ForgetOutputFile() {
+ output_filename_ = "";
+ output_path_ = "";
+}
+
+int ViEToFileRenderer::DeliverFrame(unsigned char *buffer,
+ int buffer_size,
+ uint32_t time_stamp,
+ int64_t render_time) {
+ assert(output_file_ != NULL);
+
+ int written = std::fwrite(buffer, sizeof(unsigned char),
+ buffer_size, output_file_);
+
+ if (written == buffer_size) {
+ return 0;
+ } else {
+ return -1;
+ }
+}
+
+int ViEToFileRenderer::FrameSizeChange(unsigned int width,
+ unsigned int height,
+ unsigned int number_of_streams) {
+ return 0;
+}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h
new file mode 100644
index 0000000..08559af
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
+#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
+
+#include <cstdio>
+#include <string>
+
+#include "video_engine/include/vie_render.h"
+
+class ViEToFileRenderer: public webrtc::ExternalRenderer {
+ public:
+ ViEToFileRenderer();
+ virtual ~ViEToFileRenderer();
+
+ // Returns false if we fail opening the output filename for writing.
+ bool PrepareForRendering(const std::string& output_path,
+ const std::string& output_filename);
+
+ // Closes the output file.
+ void StopRendering();
+
+ // Deletes the closed output file from the file system. This is one option
+ // after calling StopRendering, the other being KeepOutputFile. This file
+ // renderer will forget about the file after this call and can be used again.
+ bool DeleteOutputFile();
+
+ // Renames the closed output file to its previous name with the provided
+ // prefix prepended. This file renderer will forget about the file after this
+ // call and can be used again.
+ bool SaveOutputFile(const std::string& prefix);
+
+ // Implementation of ExternalRenderer:
+ int FrameSizeChange(unsigned int width, unsigned int height,
+ unsigned int number_of_streams);
+
+ int DeliverFrame(unsigned char* buffer, int buffer_size,
+ uint32_t time_stamp,
+ int64_t render_time);
+
+ const std::string GetFullOutputPath() const;
+
+ private:
+ void ForgetOutputFile();
+
+ std::FILE* output_file_;
+ std::string output_path_;
+ std::string output_filename_;
+};
+
+#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc
new file mode 100644
index 0000000..40ec0bb
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_window_creator.h"
+
+#include "vie_autotest_main.h"
+#include "vie_codec.h"
+#include "voe_codec.h"
+#include "vie_window_manager_factory.h"
+#include "vie_autotest_window_manager_interface.h"
+
+#if defined(WIN32)
+#include <tchar.h>
+#endif
+
+ViEWindowCreator::ViEWindowCreator() {
+ window_manager_ =
+ ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform();
+}
+
+ViEWindowCreator::~ViEWindowCreator() {
+ delete window_manager_;
+}
+
+ViEAutoTestWindowManagerInterface*
+ ViEWindowCreator::CreateTwoWindows() {
+#if defined(WIN32)
+ TCHAR window1Title[1024] = _T("ViE Autotest Window 1");
+ TCHAR window2Title[1024] = _T("ViE Autotest Window 2");
+#else
+ char window1Title[1024] = "ViE Autotest Window 1";
+ char window2Title[1024] = "ViE Autotest Window 2";
+#endif
+
+ AutoTestRect window1Size(352, 288, 600, 100);
+ AutoTestRect window2Size(352, 288, 1000, 100);
+ window_manager_->CreateWindows(window1Size, window2Size, window1Title,
+ window2Title);
+ window_manager_->SetTopmostWindow();
+
+ return window_manager_;
+}
+
+void ViEWindowCreator::TerminateWindows() {
+ window_manager_->TerminateWindows();
+}
+
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h
new file mode 100644
index 0000000..25c23a3
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
+
+class ViEAutoTestWindowManagerInterface;
+
+class ViEWindowCreator {
+ public:
+ ViEWindowCreator();
+ virtual ~ViEWindowCreator();
+
+ // The pointer returned here will still be owned by this object.
+ // Only use it to retrieve the created windows.
+ ViEAutoTestWindowManagerInterface* CreateTwoWindows();
+
+ // Terminates windows opened by CreateTwoWindows, which must
+ // have been called before this method.
+ void TerminateWindows();
+ private:
+ ViEAutoTestWindowManagerInterface* window_manager_;
+};
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h b/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h
new file mode 100644
index 0000000..d6be4b2
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * tb_I420_codec.h
+ *
+ * This file contains the interface to I420 "codec"
+ * This is a dummy wrapper to allow VCM deal with raw I420 sequences
+ *
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
+
+#include "video_codec_interface.h"
+
+class TbI420Encoder: public webrtc::VideoEncoder
+{
+public:
+ TbI420Encoder();
+ virtual ~TbI420Encoder();
+
+ static WebRtc_Word32 VersionStatic(WebRtc_Word8* version,
+ WebRtc_Word32 length);
+ virtual WebRtc_Word32 Version(WebRtc_Word8 *version,
+ WebRtc_Word32 length) const;
+
+ virtual WebRtc_Word32 InitEncode(const webrtc::VideoCodec* codecSettings,
+ WebRtc_Word32 numberOfCores,
+ WebRtc_UWord32 maxPayloadSize);
+
+ virtual WebRtc_Word32 Encode(
+ const webrtc::RawImage& inputImage,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo,
+ const webrtc::VideoFrameType* frameType);
+
+ virtual WebRtc_Word32 RegisterEncodeCompleteCallback(
+ webrtc::EncodedImageCallback* callback);
+
+ virtual WebRtc_Word32 Release();
+
+ virtual WebRtc_Word32 Reset();
+
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
+ int rtt);
+
+ virtual WebRtc_Word32 SetPacketLoss(WebRtc_UWord32 packetLoss);
+
+ virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate,
+ WebRtc_UWord32 frameRate);
+
+ virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
+
+ virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
+ WebRtc_Word32 /*size*/);
+
+ struct FunctionCalls
+ {
+ WebRtc_Word32 InitEncode;
+ WebRtc_Word32 Encode;
+ WebRtc_Word32 RegisterEncodeCompleteCallback;
+ WebRtc_Word32 Release;
+ WebRtc_Word32 Reset;
+ WebRtc_Word32 SetRates;
+ WebRtc_Word32 SetPacketLoss;
+ WebRtc_Word32 SetPeriodicKeyFrames;
+ WebRtc_Word32 CodecConfigParameters;
+
+ };
+
+ FunctionCalls GetFunctionCalls();
+private:
+ bool _inited;
+ webrtc::EncodedImage _encodedImage;
+ FunctionCalls _functionCalls;
+ webrtc::EncodedImageCallback* _encodedCompleteCallback;
+
+}; // end of tbI420Encoder class
+
+
+/***************************/
+/* tbI420Decoder class */
+/***************************/
+
+class TbI420Decoder: public webrtc::VideoDecoder
+{
+public:
+ TbI420Decoder();
+ virtual ~TbI420Decoder();
+
+ virtual WebRtc_Word32 InitDecode(const webrtc::VideoCodec* inst,
+ WebRtc_Word32 numberOfCores);
+ virtual WebRtc_Word32 Decode(
+ const webrtc::EncodedImage& inputImage,
+ bool missingFrames,
+ const webrtc::RTPFragmentationHeader* fragmentation,
+ const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
+ WebRtc_Word64 renderTimeMs = -1);
+
+ virtual WebRtc_Word32
+ RegisterDecodeCompleteCallback(webrtc::DecodedImageCallback* callback);
+ virtual WebRtc_Word32 Release();
+ virtual WebRtc_Word32 Reset();
+
+ struct FunctionCalls
+ {
+ WebRtc_Word32 InitDecode;
+ WebRtc_Word32 Decode;
+ WebRtc_Word32 RegisterDecodeCompleteCallback;
+ WebRtc_Word32 Release;
+ WebRtc_Word32 Reset;
+ };
+
+ FunctionCalls GetFunctionCalls();
+
+private:
+
+ webrtc::RawImage _decodedImage;
+ WebRtc_Word32 _width;
+ WebRtc_Word32 _height;
+ bool _inited;
+ FunctionCalls _functionCalls;
+ webrtc::DecodedImageCallback* _decodeCompleteCallback;
+
+}; // end of tbI420Decoder class
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h b/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h
new file mode 100644
index 0000000..3ade1e0
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
+
+#include "tb_interfaces.h"
+#include "video_capture_factory.h"
+
+class TbCaptureDevice
+{
+public:
+ TbCaptureDevice(TbInterfaces& Engine);
+ ~TbCaptureDevice(void);
+
+ int captureId;
+ void ConnectTo(int videoChannel);
+ void Disconnect(int videoChannel);
+private:
+ TbInterfaces& ViE;
+ webrtc::VideoCaptureModule* vcpm_;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h b/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h
new file mode 100644
index 0000000..6578811
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// tb_external_transport.h
+//
+
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
+
+#include <list>
+
+#include "common_types.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class ViENetwork;
+}
+
+// Allows to subscribe for callback when a frame is started being sent.
+class SendFrameCallback
+{
+public:
+ // Called once per frame (when a new RTP timestamp is detected) when the
+ // first data packet of the frame is being sent using the
+ // TbExternalTransport.SendPacket method.
+ virtual void FrameSent(unsigned int rtp_timestamp) = 0;
+protected:
+ SendFrameCallback() {}
+ virtual ~SendFrameCallback() {}
+};
+
+// Allows to subscribe for callback when the first packet of a frame is
+// received.
+class ReceiveFrameCallback
+{
+public:
+ // Called once per frame (when a new RTP timestamp is detected)
+ // during the processing of the RTP packet queue in
+ // TbExternalTransport::ViEExternalTransportProcess.
+ virtual void FrameReceived(unsigned int rtp_timestamp) = 0;
+protected:
+ ReceiveFrameCallback() {}
+ virtual ~ReceiveFrameCallback() {}
+};
+
+// External transport implementation for testing purposes.
+// A packet loss probability must be set in order to drop packets from the data
+// being sent to this class.
+// Will never drop packets from the first frame of a video sequence.
+class TbExternalTransport : public webrtc::Transport
+{
+public:
+ TbExternalTransport(webrtc::ViENetwork& vieNetwork);
+ ~TbExternalTransport(void);
+
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+
+ // Should only be called before/after traffic is being processed.
+ // Only one observer can be set (multiple calls will overwrite each other).
+ virtual void RegisterSendFrameCallback(SendFrameCallback* callback);
+
+ // Should only be called before/after traffic is being processed.
+ // Only one observer can be set (multiple calls will overwrite each other).
+ virtual void RegisterReceiveFrameCallback(ReceiveFrameCallback* callback);
+
+ // The probability of a packet of being dropped. Packets belonging to the
+ // first packet (same RTP timestamp) will never be dropped.
+ WebRtc_Word32 SetPacketLoss(WebRtc_Word32 lossRate); // Rate in %
+ void SetNetworkDelay(WebRtc_Word64 delayMs);
+ void SetSSRCFilter(WebRtc_UWord32 SSRC);
+
+ void ClearStats();
+ void GetStats(WebRtc_Word32& numRtpPackets,
+ WebRtc_Word32& numDroppedPackets,
+ WebRtc_Word32& numRtcpPackets);
+
+ void SetTemporalToggle(unsigned char layers);
+ void EnableSSRCCheck();
+ unsigned int ReceivedSSRC();
+
+ void EnableSequenceNumberCheck();
+ unsigned short GetFirstSequenceNumber();
+
+protected:
+ static bool ViEExternalTransportRun(void* object);
+ bool ViEExternalTransportProcess();
+private:
+ WebRtc_Word64 NowMs();
+
+ enum
+ {
+ KMaxPacketSize = 1650
+ };
+ enum
+ {
+ KMaxWaitTimeMs = 100
+ };
+ typedef struct
+ {
+ WebRtc_Word8 packetBuffer[KMaxPacketSize];
+ WebRtc_Word32 length;
+ WebRtc_Word32 channel;
+ WebRtc_Word64 receiveTime;
+ } VideoPacket;
+
+ webrtc::ViENetwork& _vieNetwork;
+ webrtc::ThreadWrapper& _thread;
+ webrtc::EventWrapper& _event;
+ webrtc::CriticalSectionWrapper& _crit;
+ webrtc::CriticalSectionWrapper& _statCrit;
+
+ WebRtc_Word32 _lossRate;
+ WebRtc_Word64 _networkDelayMs;
+ WebRtc_Word32 _rtpCount;
+ WebRtc_Word32 _rtcpCount;
+ WebRtc_Word32 _dropCount;
+
+ std::list<VideoPacket*> _rtpPackets;
+ std::list<VideoPacket*> _rtcpPackets;
+
+ SendFrameCallback* _send_frame_callback;
+ ReceiveFrameCallback* _receive_frame_callback;
+
+ unsigned char _temporalLayers;
+ unsigned short _seqNum;
+ unsigned short _sendPID;
+ unsigned char _receivedPID;
+ bool _switchLayer;
+ unsigned char _currentRelayLayer;
+ unsigned int _lastTimeMs;
+
+ bool _checkSSRC;
+ WebRtc_UWord32 _lastSSRC;
+ bool _filterSSRC;
+ WebRtc_UWord32 _SSRC;
+ bool _checkSequenceNumber;
+ WebRtc_UWord16 _firstSequenceNumber;
+
+ // Keep track of the first RTP timestamp so we don't do packet loss on
+ // the first frame.
+ WebRtc_UWord32 _firstRTPTimestamp;
+ // Track RTP timestamps so we invoke callbacks properly (if registered).
+ WebRtc_UWord32 _lastSendRTPTimestamp;
+ WebRtc_UWord32 _lastReceiveRTPTimestamp;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h b/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h
new file mode 100644
index 0000000..63e78a3
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
+
+#include "vie_autotest_defines.h"
+
+#include "common_types.h"
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_image_process.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+#include "vie_encryption.h"
+#include "vie_defines.h"
+
+// This class deals with all the tedium of setting up video engine interfaces.
+// It does its work in constructor and destructor, so keeping it in scope is
+// enough.
+class TbInterfaces
+{
+public:
+ TbInterfaces(const char* test_name);
+ ~TbInterfaces(void);
+
+ webrtc::VideoEngine* video_engine;
+ webrtc::ViEBase* base;
+ webrtc::ViECapture* capture;
+ webrtc::ViERender* render;
+ webrtc::ViERTP_RTCP* rtp_rtcp;
+ webrtc::ViECodec* codec;
+ webrtc::ViENetwork* network;
+ webrtc::ViEImageProcess* image_process;
+ webrtc::ViEEncryption* encryption;
+
+ int LastError() {
+ return base->LastError();
+ }
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h b/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h
new file mode 100644
index 0000000..5961e62
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
+
+#include "tb_interfaces.h"
+class TbVideoChannel
+{
+public:
+ TbVideoChannel(TbInterfaces& Engine,
+ webrtc::VideoCodecType sendCodec = webrtc::kVideoCodecVP8,
+ int width = 352, int height = 288, int frameRate = 30,
+ int startBitrate = 300);
+
+ ~TbVideoChannel(void);
+
+ void SetFrameSettings(int width, int height, int frameRate);
+
+ void StartSend(const unsigned short rtpPort = 11000,
+ const char* ipAddress = "127.0.0.1");
+
+ void StopSend();
+
+ void StartReceive(const unsigned short rtpPort = 11000);
+
+ void StopReceive();
+
+ int videoChannel;
+private:
+ TbInterfaces& ViE;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h
new file mode 100644
index 0000000..a508cd4
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest.h
+//
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
+
+#include "common_types.h"
+
+#include "voe_base.h"
+#include "voe_codec.h"
+#include "voe_hardware.h"
+#include "voe_audio_processing.h"
+
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_file.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+#include "vie_defines.h"
+#include "vie_errors.h"
+#include "video_render_defines.h"
+
+#include "vie_autotest_defines.h"
+
+#ifndef WEBRTC_ANDROID
+#include <string>
+#endif
+
+class TbCaptureDevice;
+class TbInterfaces;
+class TbVideoChannel;
+class ViEToFileRenderer;
+
+// This class provides a bunch of methods, implemented across several .cc
+// files, which runs tests on the video engine. All methods will report
+// errors using standard googletest macros, except when marked otherwise.
+class ViEAutoTest
+{
+public:
+ ViEAutoTest(void* window1, void* window2);
+ ~ViEAutoTest();
+
+ // These three are special and should not be run in a googletest harness.
+ // They keep track of their errors by themselves and return the number
+ // of errors.
+ int ViELoopbackCall();
+ int ViESimulcastCall();
+ int ViECustomCall();
+
+ // All functions except the three above are meant to run in a
+ // googletest harness.
+ void ViEStandardTest();
+ void ViEExtendedTest();
+ void ViEAPITest();
+
+ // vie_autotest_base.cc
+ void ViEBaseStandardTest();
+ void ViEBaseExtendedTest();
+ void ViEBaseAPITest();
+
+ // vie_autotest_capture.cc
+ void ViECaptureStandardTest();
+ void ViECaptureExtendedTest();
+ void ViECaptureAPITest();
+ void ViECaptureExternalCaptureTest();
+
+ // vie_autotest_codec.cc
+ void ViECodecStandardTest();
+ void ViECodecExtendedTest();
+ void ViECodecExternalCodecTest();
+ void ViECodecAPITest();
+
+ // vie_autotest_encryption.cc
+ void ViEEncryptionStandardTest();
+ void ViEEncryptionExtendedTest();
+ void ViEEncryptionAPITest();
+
+ // vie_autotest_file.ccs
+ void ViEFileStandardTest();
+ void ViEFileExtendedTest();
+ void ViEFileAPITest();
+
+ // vie_autotest_image_process.cc
+ void ViEImageProcessStandardTest();
+ void ViEImageProcessExtendedTest();
+ void ViEImageProcessAPITest();
+
+ // vie_autotest_network.cc
+ void ViENetworkStandardTest();
+ void ViENetworkExtendedTest();
+ void ViENetworkAPITest();
+
+ // vie_autotest_render.cc
+ void ViERenderStandardTest();
+ void ViERenderExtendedTest();
+ void ViERenderAPITest();
+
+ // vie_autotest_rtp_rtcp.cc
+ void ViERtpRtcpStandardTest();
+ void ViERtpRtcpExtendedTest();
+ void ViERtpRtcpAPITest();
+
+ // vie_autotest_rtp_fuzz.cc
+ void ViERtpTryInjectingRandomPacketsIntoRtpStream(long rand_seed);
+
+private:
+ void PrintAudioCodec(const webrtc::CodecInst audioCodec);
+ void PrintVideoCodec(const webrtc::VideoCodec videoCodec);
+
+ // Sets up rendering so the capture device output goes to window 1 and
+ // the video engine output goes to window 2.
+ void RenderCaptureDeviceAndOutputStream(TbInterfaces* video_engine,
+ TbVideoChannel* video_channel,
+ TbCaptureDevice* capture_device);
+
+ void* _window1;
+ void* _window2;
+
+ webrtc::VideoRenderType _renderType;
+ webrtc::VideoRender* _vrm1;
+ webrtc::VideoRender* _vrm2;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h
new file mode 100644
index 0000000..53b8cc3
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
+
+class ViEAutoTestAndroid
+{
+public:
+ static int RunAutotest(int testSelection,
+ int subTestSelection,
+ void* window1,
+ void* window2,
+ void* javaVM,
+ void* env,
+ void* context);
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h
new file mode 100644
index 0000000..34d489b
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_defines.h
+//
+
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
+
+#include <cassert>
+#include <stdarg.h>
+#include <stdio.h>
+#include <string>
+
+#include "engine_configurations.h"
+#include "gtest/gtest.h"
+
+#if defined(_WIN32)
+#include <windows.h>
+#elif defined (WEBRTC_ANDROID)
+#include <android/log.h>
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#include <string.h>
+#include <time.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#endif
+
+// Choose how to log
+//#define VIE_LOG_TO_FILE
+#define VIE_LOG_TO_STDOUT
+
+// Choose one way to test error
+#define VIE_ASSERT_ERROR
+
+#define VIE_LOG_FILE_NAME "ViEAutotestLog.txt"
+
+#undef RGB
+#define RGB(r,g,b) r|g<<8|b<<16
+
+enum {
+ KAutoTestSleepTimeMs = 5000
+};
+
+struct AutoTestSize {
+ unsigned int width;
+ unsigned int height;
+ AutoTestSize() :
+ width(0), height(0) {
+ }
+ AutoTestSize(unsigned int iWidth, unsigned int iHeight) :
+ width(iWidth), height(iHeight) {
+ }
+};
+
+struct AutoTestOrigin {
+ unsigned int x;
+ unsigned int y;
+ AutoTestOrigin() :
+ x(0), y(0) {
+ }
+ AutoTestOrigin(unsigned int iX, unsigned int iY) :
+ x(iX), y(iY) {
+ }
+};
+
+struct AutoTestRect {
+ AutoTestSize size;
+ AutoTestOrigin origin;
+ AutoTestRect() :
+ size(), origin() {
+ }
+
+ AutoTestRect(unsigned int iX, unsigned int iY, unsigned int iWidth, unsigned int iHeight) :
+ size(iX, iY), origin(iWidth, iHeight) {
+ }
+
+ void Copy(AutoTestRect iRect) {
+ origin.x = iRect.origin.x;
+ origin.y = iRect.origin.y;
+ size.width = iRect.size.width;
+ size.height = iRect.size.height;
+ }
+};
+
+// ============================================
+
+class ViETest {
+ public:
+ static int Init() {
+#ifdef VIE_LOG_TO_FILE
+ log_file_ = fopen(VIE_LOG_FILE_NAME, "w+t");
+#else
+ log_file_ = NULL;
+#endif
+ log_str_ = new char[kMaxLogSize];
+ memset(log_str_, 0, kMaxLogSize);
+ return 0;
+ }
+
+ static int Terminate() {
+ if (log_file_) {
+ fclose(log_file_);
+ log_file_ = NULL;
+ }
+ if (log_str_) {
+ delete[] log_str_;
+ log_str_ = NULL;
+ }
+ return 0;
+ }
+
+ static void Log(const char* fmt, ...) {
+ va_list va;
+ va_start(va, fmt);
+ memset(log_str_, 0, kMaxLogSize);
+ vsprintf(log_str_, fmt, va);
+ va_end(va);
+
+ WriteToSuitableOutput(log_str_);
+ }
+
+ // Writes to a suitable output, depending on platform and log mode.
+ static void WriteToSuitableOutput(const char* message) {
+#ifdef VIE_LOG_TO_FILE
+ if (log_file_)
+ {
+ fwrite(log_str_, 1, strlen(log_str_), log_file_);
+ fwrite("\n", 1, 1, log_file_);
+ fflush(log_file_);
+ }
+#endif
+#ifdef VIE_LOG_TO_STDOUT
+#if WEBRTC_ANDROID
+ __android_log_write(ANDROID_LOG_DEBUG, "*WebRTCN*", log_str_);
+#else
+ printf("%s\n", log_str_);
+#endif
+#endif
+ }
+
+ // Deprecated(phoglund): Prefer to use googletest macros in all cases
+ // except the custom call case.
+ static int TestError(bool expr, const char* fmt, ...) {
+ if (!expr) {
+ va_list va;
+ va_start(va, fmt);
+ memset(log_str_, 0, kMaxLogSize);
+ vsprintf(log_str_, fmt, va);
+#ifdef WEBRTC_ANDROID
+ __android_log_write(ANDROID_LOG_ERROR, "*WebRTCN*", log_str_);
+#endif
+ WriteToSuitableOutput(log_str_);
+ va_end(va);
+
+ AssertError(log_str_);
+ return 1;
+ }
+ return 0;
+ }
+
+ // Returns a suitable path to write trace and result files to.
+ // You should always use this when you want to write output files.
+ // The returned path is guaranteed to end with a path separator.
+ // This function may be run at any time during the program's execution.
+ // Implemented in vie_autotest.cc
+ static std::string GetResultOutputPath();
+
+private:
+ static void AssertError(const char* message) {
+#ifdef VIE_ASSERT_ERROR
+ assert(false);
+#endif
+ }
+
+ static FILE* log_file_;
+ enum {
+ kMaxLogSize = 512
+ };
+ static char* log_str_;
+};
+
+// milliseconds
+#if defined(_WIN32)
+#define AutoTestSleep ::Sleep
+#elif defined(WEBRTC_MAC_INTEL)
+#define AutoTestSleep(x) usleep(x * 1000)
+#elif defined(WEBRTC_LINUX)
+namespace {
+ void Sleep(unsigned long x) {
+ timespec t;
+ t.tv_sec = x/1000;
+ t.tv_nsec = (x-(x/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+ }
+}
+#define AutoTestSleep ::Sleep
+#endif
+
+#ifdef WEBRTC_ANDROID
+namespace {
+ void Sleep(unsigned long x) {
+ timespec t;
+ t.tv_sec = x/1000;
+ t.tv_nsec = (x-(x/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+ }
+}
+
+#define AutoTestSleep ::Sleep
+#endif
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h
new file mode 100644
index 0000000..fb1d319
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
+
+#include "vie_autotest_window_manager_interface.h"
+#include <X11/Xlib.h>
+#include <X11/Xutil.h>
+
+// Forward declaration
+
+class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
+{
+public:
+ ViEAutoTestWindowManager();
+ virtual ~ViEAutoTestWindowManager();
+ virtual void* GetWindow1();
+ virtual void* GetWindow2();
+ virtual int TerminateWindows();
+ virtual int CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size, void* window1Title,
+ void* window2Title);
+ virtual bool SetTopmostWindow();
+
+private:
+ int ViECreateWindow(Window *outWindow, Display **outDisplay, int xpos,
+ int ypos, int width, int height, char* title);
+ int ViEDestroyWindow(Window *window, Display *display);
+
+ Window _hwnd1;
+ Window _hwnd2;
+ Display* _hdsp1;
+ Display* _hdsp2;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h
new file mode 100644
index 0000000..6c35baa
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "EngineConfigurations.h"
+
+#if defined(CARBON_RENDERING)
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
+
+#include "vie_autotest_window_manager_interface.h"
+
+// #define HIVIEWREF_MODE 1
+
+#include <Carbon/Carbon.h>
+#import <Cocoa/Cocoa.h>
+
+class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
+{
+public:
+ ViEAutoTestWindowManager();
+ virtual ~ViEAutoTestWindowManager();
+ virtual void* GetWindow1();
+ virtual void* GetWindow2();
+ virtual int CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size, char* window1Title,
+ char* window2Title);
+ virtual int TerminateWindows();
+ virtual bool SetTopmostWindow();
+
+ // event handler static methods
+static pascal OSStatus HandleWindowEvent (EventHandlerCallRef nextHandler,
+ EventRef theEvent, void* userData);
+static pascal OSStatus HandleHIViewEvent (EventHandlerCallRef nextHandler,
+ EventRef theEvent, void* userData);
+private:
+ WindowRef* _carbonWindow1;
+ WindowRef* _carbonWindow2;
+ HIViewRef* _hiView1;
+ HIViewRef* _hiView2;
+
+ EventHandlerRef _carbonWindow1EventHandlerRef;
+ EventHandlerRef _carbonWindow2EventHandlerRef;
+ EventHandlerRef _carbonHIView1EventHandlerRef;
+ EventHandlerRef _carbonHIView2EventHandlerRef;
+
+};
+
+@interface AutoTestClass : NSObject
+{
+}
+
+-(void)autoTestWithArg:(NSString*)answerFile;
+@end
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
+#endif CARBON_RENDERING
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
new file mode 100644
index 0000000..745431d
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(COCOA_RENDERING)
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
+
+#include "vie_autotest_window_manager_interface.h"
+#define MAC_COCOA_USE_NSRUNLOOP 1
+
+@class CocoaRenderView;
+
+#import <Cocoa/Cocoa.h>
+
+class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface {
+ public:
+ ViEAutoTestWindowManager();
+ virtual ~ViEAutoTestWindowManager() {}
+ virtual void* GetWindow1();
+ virtual void* GetWindow2();
+ virtual int CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size,
+ void* window1Title,
+ void* window2Title);
+ virtual int TerminateWindows();
+ virtual bool SetTopmostWindow();
+
+ private:
+ CocoaRenderView* _cocoaRenderView1;
+ CocoaRenderView* _cocoaRenderView2;
+ NSWindow* outWindow1_;
+ NSWindow* outWindow2_;
+};
+
+@interface AutoTestClass : NSObject {
+ int argc_;
+ char** argv_;
+ int result_;
+}
+
+-(void)setArgc:(int)argc argv:(char**)argv;
+-(int) result;
+-(void)autoTestWithArg:(NSObject*)ignored;
+
+@end
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
+#endif // COCOA_RENDERING
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h
new file mode 100644
index 0000000..d24de5c
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
+
+#include <string>
+#include <map>
+
+class ViEAutoTestMain {
+ public:
+ ViEAutoTestMain();
+
+ // Runs the test according to the specified arguments.
+ // Pass in --automated to run in automated mode; interactive
+ // mode is default. All usual googletest flags also apply.
+ int RunTests(int argc, char** argv);
+
+ private:
+ std::map<int, std::string> index_to_test_method_map_;
+
+ static const int kInvalidChoice = -1;
+
+ // Starts interactive mode.
+ bool RunInteractiveMode();
+ // Prompts the user for a specific test method in the provided test case.
+ // Returns 0 on success, nonzero otherwise.
+ int RunSpecificTestCaseIn(const std::string test_case_name);
+ // Asks the user for a particular test case to run.
+ int AskUserForTestCase();
+ // Retrieves a number from the user in the interval
+ // [min_allowed, max_allowed]. Returns kInvalidChoice on failure.
+ int AskUserForNumber(int min_allowed, int max_allowed);
+ // Runs all tests matching the provided filter. * are wildcards.
+ // Returns the test runner result (0 == OK).
+ int RunTestMatching(const std::string test_case,
+ const std::string test_method);
+ // Runs a non-gtest test case. Choice must be [7,9]. Returns 0 on success.
+ int RunSpecialTestCase(int choice);
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
new file mode 100644
index 0000000..6dd043e
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * vie_autotest_window_manager_interface.h
+ */
+
+#include "vie_autotest_defines.h"
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
+
+class ViEAutoTestWindowManagerInterface
+{
+public:
+ virtual int CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size, void* window1Title,
+ void* window2Title) = 0;
+ virtual int TerminateWindows() = 0;
+ virtual void* GetWindow1() = 0;
+ virtual void* GetWindow2() = 0;
+ virtual bool SetTopmostWindow() = 0;
+ virtual ~ViEAutoTestWindowManagerInterface() {}
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h
new file mode 100644
index 0000000..314a121
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
+
+#include "vie_autotest_window_manager_interface.h"
+#include "engine_configurations.h"
+
+#include <windows.h>
+#define TITLE_LENGTH 1024
+
+// Forward declaration
+namespace webrtc {
+class ThreadWrapper;
+class CriticalSectionWrapper;
+}
+
+class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
+{
+public:
+ ViEAutoTestWindowManager();
+ virtual ~ViEAutoTestWindowManager();
+ virtual void* GetWindow1();
+ virtual void* GetWindow2();
+ virtual int CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size, void* window1Title,
+ void* window2Title);
+ virtual int TerminateWindows();
+ virtual bool SetTopmostWindow();
+protected:
+ static bool EventProcess(void* obj);
+ bool EventLoop();
+
+private:
+ int ViECreateWindow(HWND &hwndMain, int xPos, int yPos, int width,
+ int height, TCHAR* className);
+ int ViEDestroyWindow(HWND& hwnd);
+
+ void* _window1;
+ void* _window2;
+
+ bool _terminate;
+ webrtc::ThreadWrapper& _eventThread;
+ webrtc::CriticalSectionWrapper& _crit;
+ HWND _hwndMain;
+ HWND _hwnd1;
+ HWND _hwnd2;
+
+ AutoTestRect _hwnd1Size;
+ AutoTestRect _hwnd2Size;
+ TCHAR _hwnd1Title[TITLE_LENGTH];
+ TCHAR _hwnd2Title[TITLE_LENGTH];
+
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h b/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
new file mode 100644
index 0000000..a9560bc
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
+#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
+
+#include <string>
+
+class FrameDropDetector;
+class ViEToFileRenderer;
+
+// This class contains comparison tests, which will exercise video engine
+// functionality and then run comparison tests on the result using PSNR and
+// SSIM algorithms. These tests are intended mostly as sanity checks so that
+// we know we are outputting roughly the right thing and not random noise or
+// black screens.
+//
+// We will set up a fake ExternalCapture device which will pose as a webcam
+// and read the input from the provided raw YUV file. Output will be written
+// as a local preview in the local file renderer; the remote side output gets
+// written to the provided remote file renderer.
+//
+// The local preview is a straight, unaltered copy of the input. This can be
+// useful for comparisons if the test method contains several stages where the
+// input is restarted between stages.
+class ViEFileBasedComparisonTests {
+ public:
+ // Test a typical simple call setup. Returns false if the input file
+ // could not be opened; reports errors using googletest macros otherwise.
+ bool TestCallSetup(
+ const std::string& i420_test_video_path,
+ int width,
+ int height,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer);
+
+ // Tries testing the I420 and VP8 codecs in turn. Returns false if the
+ // input file could not be opened; reports errors using googletest macros
+ // otherwise.
+ bool TestCodecs(
+ const std::string& i420_video_file,
+ int width,
+ int height,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer);
+
+ // Runs a full stack test using the VP8 codec. Tests the full stack and uses
+ // RTP timestamps to sync frames between the endpoints.
+ void TestFullStack(
+ const std::string& i420_video_file,
+ int width,
+ int height,
+ int bit_rate_kbps,
+ int packet_loss_percent,
+ int network_delay_ms,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer,
+ FrameDropDetector* frame_drop_detector);
+};
+
+#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h b/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h
new file mode 100644
index 0000000..a85280d
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h
@@ -0,0 +1,25 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
+
+class ViEAutoTestWindowManagerInterface;
+
+class ViEWindowManagerFactory {
+ public:
+ // This method is implemented in different files depending on platform.
+ // The caller is responsible for freeing the resulting object using
+ // the delete operator.
+ static ViEAutoTestWindowManagerInterface*
+ CreateWindowManagerForCurrentPlatform();
+};
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
diff --git a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp
new file mode 100644
index 0000000..6cd34ba
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg
new file mode 100644
index 0000000..3bb3ba4
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp b/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp
new file mode 100644
index 0000000..c443a58
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg b/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg
new file mode 100644
index 0000000..b10a842
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp
new file mode 100644
index 0000000..8159bad
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg
new file mode 100644
index 0000000..cb34d67
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc
new file mode 100644
index 0000000..b8c827e
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "base_primitives.h"
+
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+#include "video_capture_factory.h"
+
+void TestI420CallSetup(webrtc::ViECodec* codec_interface,
+ webrtc::VideoEngine* video_engine,
+ webrtc::ViEBase* base_interface,
+ webrtc::ViENetwork* network_interface,
+ int video_channel,
+ const unsigned char *device_name) {
+ webrtc::VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
+
+ // Set up the codec interface with all known receive codecs and with
+ // I420 as the send codec.
+ for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
+ EXPECT_EQ(0, codec_interface->GetCodec(i, video_codec));
+
+ // Try to keep the test frame size small when I420.
+ if (video_codec.codecType == webrtc::kVideoCodecI420) {
+ video_codec.width = 176;
+ video_codec.height = 144;
+ EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, video_codec));
+ }
+
+ EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
+ }
+
+ // Verify that we really found the I420 codec.
+ EXPECT_EQ(0, codec_interface->GetSendCodec(video_channel, video_codec));
+ EXPECT_EQ(webrtc::kVideoCodecI420, video_codec.codecType);
+
+ // Set up senders and receivers.
+ char version[1024] = "";
+ EXPECT_EQ(0, base_interface->GetVersion(version));
+ ViETest::Log("\nUsing WebRTC Video Engine version: %s", version);
+
+ const char *ipAddress = "127.0.0.1";
+ WebRtc_UWord16 rtpPortListen = 6100;
+ WebRtc_UWord16 rtpPortSend = 6100;
+ EXPECT_EQ(0, network_interface->SetLocalReceiver(video_channel,
+ rtpPortListen));
+ EXPECT_EQ(0, base_interface->StartReceive(video_channel));
+ EXPECT_EQ(0, network_interface->SetSendDestination(video_channel, ipAddress,
+ rtpPortSend));
+ EXPECT_EQ(0, base_interface->StartSend(video_channel));
+
+ // Call started.
+ ViETest::Log("Call started");
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ // Done.
+ EXPECT_EQ(0, base_interface->StopSend(video_channel));
+}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h
new file mode 100644
index 0000000..19b6f23
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
+#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
+
+namespace webrtc {
+class VideoEngine;
+class ViEBase;
+class ViECodec;
+class ViENetwork;
+}
+
+// Tests a I420-to-I420 call. This test exercises the most basic WebRTC
+// functionality by training the codec interface to recognize the most common
+// codecs, and the initiating a I420 call. A video channel with a capture device
+// must be set up prior to this call.
+void TestI420CallSetup(webrtc::ViECodec* codec_interface,
+ webrtc::VideoEngine* video_engine,
+ webrtc::ViEBase* base_interface,
+ webrtc::ViENetwork* network_interface,
+ int video_channel,
+ const unsigned char *device_name);
+
+#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc
new file mode 100644
index 0000000..2d43440
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "codec_primitives.h"
+
+#include "general_primitives.h"
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+#include "vie_to_file_renderer.h"
+#include "video_capture_factory.h"
+#include "tb_interfaces.h"
+
+// Helper functions.
+
+void TestCodecImageProcess(webrtc::VideoCodec video_codec,
+ webrtc::ViECodec* codec_interface,
+ int video_channel,
+ webrtc::ViEImageProcess* image_process) {
+
+ EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, video_codec));
+ FrameCounterEffectFilter frame_counter;
+ EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
+ frame_counter));
+ AutoTestSleep (KAutoTestSleepTimeMs);
+
+ int max_number_of_rendered_frames = video_codec.maxFramerate *
+ KAutoTestSleepTimeMs / 1000;
+
+ if (video_codec.codecType == webrtc::kVideoCodecI420) {
+ // Due to that I420 needs a huge bandwidth, rate control can set
+ // frame rate very low. This happen since we use the same channel
+ // as we just tested with vp8.
+ EXPECT_GT(frame_counter.numFrames, 0);
+ } else {
+#ifdef WEBRTC_ANDROID
+ // Special case to get the autotest to pass on some slow devices
+ EXPECT_GT(frame_counter.numFrames, max_number_of_rendered_frames / 6);
+#else
+ EXPECT_GT(frame_counter.numFrames, max_number_of_rendered_frames / 4);
+#endif
+ }
+ EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(video_channel));
+}
+
+// Test switching from i420 to VP8 as send codec and make sure that
+// the codec observer gets called after the switch.
+void TestCodecCallbacks(webrtc::ViEBase *& base_interface,
+ webrtc::ViECodec *codec_interface,
+ int video_channel,
+ int forced_codec_width,
+ int forced_codec_height) {
+
+ // Set I420 as send codec so we don't make any assumptions about what
+ // we currently have as send codec:
+ SetSendCodec(webrtc::kVideoCodecI420, codec_interface, video_channel,
+ forced_codec_width, forced_codec_height);
+
+ // Register the observer:
+ ViEAutotestCodecObserver codec_observer;
+ EXPECT_EQ(0, codec_interface->RegisterEncoderObserver(video_channel,
+ codec_observer));
+ EXPECT_EQ(0, codec_interface->RegisterDecoderObserver(video_channel,
+ codec_observer));
+
+ // Make the switch.
+ ViETest::Log("Testing codec callbacks...");
+
+ SetSendCodec(webrtc::kVideoCodecVP8, codec_interface, video_channel,
+ forced_codec_width, forced_codec_height);
+
+ AutoTestSleep (KAutoTestSleepTimeMs);
+
+ // Verify that we got the right codec.
+ EXPECT_EQ(webrtc::kVideoCodecVP8, codec_observer.incomingCodec.codecType);
+
+ // Clean up.
+ EXPECT_EQ(0, codec_interface->DeregisterEncoderObserver(video_channel));
+ EXPECT_EQ(0, codec_interface->DeregisterDecoderObserver(video_channel));
+
+ // Verify results.
+ EXPECT_GT(codec_observer.incomingCodecCalled, 0);
+ EXPECT_GT(codec_observer.incomingRatecalled, 0);
+ EXPECT_GT(codec_observer.outgoingRatecalled, 0);
+}
+
+void TestCodecs(const TbInterfaces& interfaces,
+ int capture_id,
+ int video_channel,
+ int forced_codec_width,
+ int forced_codec_height) {
+ webrtc::VideoEngine *video_engine_interface = interfaces.video_engine;
+ webrtc::ViEBase *base_interface = interfaces.base;
+ webrtc::ViECapture *capture_interface = interfaces.capture;
+ webrtc::ViERender *render_interface = interfaces.render;
+ webrtc::ViECodec *codec_interface = interfaces.codec;
+ webrtc::ViENetwork *network_interface = interfaces.network;
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+ webrtc::VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof (webrtc::VideoCodec));
+
+ // Set up all receive codecs. This basically trains the codec interface
+ // to be able to recognize all receive codecs based on payload type.
+ for (int idx = 0; idx < codec_interface->NumberOfCodecs(); idx++) {
+ EXPECT_EQ(0, codec_interface->GetCodec(idx, video_codec));
+ SetSuitableResolution(&video_codec,
+ forced_codec_width,
+ forced_codec_height);
+
+ EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
+ }
+ const char *ip_address = "127.0.0.1";
+ const unsigned short rtp_port = 6000;
+ EXPECT_EQ(0, network_interface->SetLocalReceiver(video_channel, rtp_port));
+ EXPECT_EQ(0, base_interface->StartReceive(video_channel));
+ EXPECT_EQ(0, network_interface->SetSendDestination(video_channel, ip_address,
+ rtp_port));
+ EXPECT_EQ(0, base_interface->StartSend(video_channel));
+
+ // Run all found codecs
+ webrtc::ViEImageProcess *image_process =
+ webrtc::ViEImageProcess::GetInterface(video_engine_interface);
+ EXPECT_TRUE(image_process != NULL);
+
+ ViETest::Log("Loop through all codecs for %d seconds",
+ KAutoTestSleepTimeMs / 1000);
+ for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
+ EXPECT_EQ(0, codec_interface->GetCodec(i, video_codec));
+
+ if (video_codec.codecType == webrtc::kVideoCodecRED ||
+ video_codec.codecType == webrtc::kVideoCodecULPFEC) {
+ ViETest::Log("\t %d. %s not tested", i, video_codec.plName);
+ } else {
+ ViETest::Log("\t %d. %s", i, video_codec.plName);
+ SetSuitableResolution(&video_codec, forced_codec_width,
+ forced_codec_height);
+ TestCodecImageProcess(video_codec, codec_interface, video_channel,
+ image_process);
+ }
+ }
+ image_process->Release();
+
+ TestCodecCallbacks(base_interface, codec_interface, video_channel,
+ forced_codec_width, forced_codec_height);
+
+ ViETest::Log("Done!");
+
+ // ***************************************************************
+ // Testing finished. Tear down Video Engine
+ // ***************************************************************
+ EXPECT_EQ(0, base_interface->StopSend(video_channel));
+ EXPECT_EQ(0, base_interface->StopReceive(video_channel));
+ EXPECT_EQ(0, render_interface->StopRender(capture_id));
+ EXPECT_EQ(0, render_interface->StopRender(video_channel));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
+ EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
+ EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
+}
+
+void SetSendCodec(webrtc::VideoCodecType of_type,
+ webrtc::ViECodec* codec_interface,
+ int video_channel,
+ int forced_codec_width,
+ int forced_codec_height) {
+ webrtc::VideoCodec codec;
+ bool ok;
+ EXPECT_TRUE(ok = FindSpecificCodec(of_type, codec_interface, &codec));
+ if (!ok) {
+ return;
+ }
+
+ SetSuitableResolution(&codec, forced_codec_width, forced_codec_height);
+ EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, codec));
+}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h
new file mode 100644
index 0000000..e778044
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h
@@ -0,0 +1,126 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
+
+#include "video_engine/include/vie_codec.h"
+#include "video_engine/include/vie_image_process.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "video_engine/test/auto_test/primitives/general_primitives.h"
+
+class TbInterfaces;
+
+// Tests that a codec actually renders frames by registering a basic
+// render effect filter on the codec and then running it. This test is
+// quite lenient on the number of frames that get rendered, so it should not
+// be seen as a end-user-visible quality measure - it is more a sanity check
+// that the codec at least gets some frames through.
+
+// The codec resolution can be forced by specifying the forced* variables
+// (pass in kDoNotForceResolution if you don't care).
+void TestCodecs(const TbInterfaces& interfaces,
+ int capture_id,
+ int video_channel,
+ int forced_codec_width,
+ int forced_codec_height);
+
+// This helper function will set the send codec in the codec interface to a
+// codec of the specified type. It will generate a test failure if we do not
+// support the provided codec type.
+
+// The codec resolution can be forced by specifying the forced* variables
+// (pass in kDoNotForceResolution if you don't care).
+void SetSendCodec(webrtc::VideoCodecType of_type,
+ webrtc::ViECodec* codec_interface,
+ int video_channel,
+ int forced_codec_width,
+ int forced_codec_height);
+
+class ViEAutotestCodecObserver: public webrtc::ViEEncoderObserver,
+ public webrtc::ViEDecoderObserver {
+ public:
+ int incomingCodecCalled;
+ int incomingRatecalled;
+ int outgoingRatecalled;
+
+ unsigned char lastPayloadType;
+ unsigned short lastWidth;
+ unsigned short lastHeight;
+
+ unsigned int lastOutgoingFramerate;
+ unsigned int lastOutgoingBitrate;
+ unsigned int lastIncomingFramerate;
+ unsigned int lastIncomingBitrate;
+
+ webrtc::VideoCodec incomingCodec;
+
+ ViEAutotestCodecObserver() {
+ incomingCodecCalled = 0;
+ incomingRatecalled = 0;
+ outgoingRatecalled = 0;
+ lastPayloadType = 0;
+ lastWidth = 0;
+ lastHeight = 0;
+ lastOutgoingFramerate = 0;
+ lastOutgoingBitrate = 0;
+ lastIncomingFramerate = 0;
+ lastIncomingBitrate = 0;
+ memset(&incomingCodec, 0, sizeof(incomingCodec));
+ }
+ virtual void IncomingCodecChanged(const int videoChannel,
+ const webrtc::VideoCodec& videoCodec) {
+ incomingCodecCalled++;
+ lastPayloadType = videoCodec.plType;
+ lastWidth = videoCodec.width;
+ lastHeight = videoCodec.height;
+
+ memcpy(&incomingCodec, &videoCodec, sizeof(videoCodec));
+ }
+
+ virtual void IncomingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate) {
+ incomingRatecalled++;
+ lastIncomingFramerate += framerate;
+ lastIncomingBitrate += bitrate;
+ }
+
+ virtual void OutgoingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate) {
+ outgoingRatecalled++;
+ lastOutgoingFramerate += framerate;
+ lastOutgoingBitrate += bitrate;
+ }
+
+ virtual void RequestNewKeyFrame(const int videoChannel) {
+ }
+};
+
+class FrameCounterEffectFilter : public webrtc::ViEEffectFilter
+{
+ public:
+ int numFrames;
+ FrameCounterEffectFilter() {
+ numFrames = 0;
+ }
+ virtual ~FrameCounterEffectFilter() {
+ }
+
+ virtual int Transform(int size, unsigned char* frameBuffer,
+ unsigned int timeStamp90KHz, unsigned int width,
+ unsigned int height) {
+ numFrames++;
+ return 0;
+ }
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc
new file mode 100644
index 0000000..6767cc6
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc
@@ -0,0 +1,513 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <string>
+
+#include "modules/video_capture/main/interface/video_capture_factory.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/frame_reader.h"
+#include "testsupport/frame_writer.h"
+#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
+#include "video_engine/test/auto_test/interface/tb_interfaces.h"
+#include "video_engine/test/auto_test/interface/tb_external_transport.h"
+#include "video_engine/test/auto_test/interface/vie_autotest.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
+#include "video_engine/test/auto_test/primitives/general_primitives.h"
+
+// Tracks which frames are created on the local side and reports them to the
+// FrameDropDetector class.
+class CreatedTimestampEffectFilter : public webrtc::ViEEffectFilter {
+ public:
+ explicit CreatedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
+ : frame_drop_detector_(frame_drop_detector) {}
+ virtual ~CreatedTimestampEffectFilter() {}
+ virtual int Transform(int size, unsigned char* frameBuffer,
+ unsigned int timeStamp90KHz, unsigned int width,
+ unsigned int height) {
+ frame_drop_detector_->ReportFrameState(FrameDropDetector::kCreated,
+ timeStamp90KHz);
+ return 0;
+ }
+
+ private:
+ FrameDropDetector* frame_drop_detector_;
+};
+
+// Tracks which frames are sent in external transport on the local side
+// and reports them to the FrameDropDetector class.
+class FrameSentCallback : public SendFrameCallback {
+ public:
+ explicit FrameSentCallback(FrameDropDetector* frame_drop_detector)
+ : frame_drop_detector_(frame_drop_detector) {}
+ virtual ~FrameSentCallback() {}
+ virtual void FrameSent(unsigned int rtp_timestamp) {
+ frame_drop_detector_->ReportFrameState(FrameDropDetector::kSent,
+ rtp_timestamp);
+ }
+
+ private:
+ FrameDropDetector* frame_drop_detector_;
+};
+
+// Tracks which frames are received in external transport on the remote side
+// and reports them to the FrameDropDetector class.
+class FrameReceivedCallback : public ReceiveFrameCallback {
+ public:
+ explicit FrameReceivedCallback(FrameDropDetector* frame_drop_detector)
+ : frame_drop_detector_(frame_drop_detector) {}
+ virtual ~FrameReceivedCallback() {}
+ virtual void FrameReceived(unsigned int rtp_timestamp) {
+ frame_drop_detector_->ReportFrameState(FrameDropDetector::kReceived,
+ rtp_timestamp);
+ }
+
+ private:
+ FrameDropDetector* frame_drop_detector_;
+};
+
+// Tracks when frames are decoded on the remote side (received from the
+// jitter buffer) and reports them to the FrameDropDetector class.
+class DecodedTimestampEffectFilter : public webrtc::ViEEffectFilter {
+ public:
+ explicit DecodedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
+ : frame_drop_detector_(frame_drop_detector) {}
+ virtual ~DecodedTimestampEffectFilter() {}
+ virtual int Transform(int size, unsigned char* frameBuffer,
+ unsigned int timeStamp90KHz, unsigned int width,
+ unsigned int height) {
+ frame_drop_detector_->ReportFrameState(FrameDropDetector::kDecoded,
+ timeStamp90KHz);
+ return 0;
+ }
+
+ private:
+ FrameDropDetector* frame_drop_detector_;
+};
+
+void TestFullStack(const TbInterfaces& interfaces,
+ int capture_id,
+ int video_channel,
+ int width,
+ int height,
+ int bit_rate_kbps,
+ int packet_loss_percent,
+ int network_delay_ms,
+ FrameDropDetector* frame_drop_detector) {
+ webrtc::VideoEngine *video_engine_interface = interfaces.video_engine;
+ webrtc::ViEBase *base_interface = interfaces.base;
+ webrtc::ViECapture *capture_interface = interfaces.capture;
+ webrtc::ViERender *render_interface = interfaces.render;
+ webrtc::ViECodec *codec_interface = interfaces.codec;
+ webrtc::ViENetwork *network_interface = interfaces.network;
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+ webrtc::VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof (webrtc::VideoCodec));
+
+ // Set up all receive codecs. This basically setup the codec interface
+ // to be able to recognize all receive codecs based on payload type.
+ for (int idx = 0; idx < codec_interface->NumberOfCodecs(); idx++) {
+ EXPECT_EQ(0, codec_interface->GetCodec(idx, video_codec));
+ SetSuitableResolution(&video_codec, width, height);
+
+ EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
+ }
+
+ // Configure External transport to simulate network interference:
+ TbExternalTransport external_transport(*interfaces.network);
+ external_transport.SetPacketLoss(packet_loss_percent);
+ external_transport.SetNetworkDelay(network_delay_ms);
+
+ FrameSentCallback frame_sent_callback(frame_drop_detector);
+ FrameReceivedCallback frame_received_callback(frame_drop_detector);
+ external_transport.RegisterSendFrameCallback(&frame_sent_callback);
+ external_transport.RegisterReceiveFrameCallback(&frame_received_callback);
+ EXPECT_EQ(0, network_interface->RegisterSendTransport(video_channel,
+ external_transport));
+ EXPECT_EQ(0, base_interface->StartReceive(video_channel));
+
+ // Setup only the VP8 codec, which is what we'll use.
+ webrtc::VideoCodec codec;
+ EXPECT_TRUE(FindSpecificCodec(webrtc::kVideoCodecVP8, codec_interface,
+ &codec));
+ codec.startBitrate = bit_rate_kbps;
+ codec.maxBitrate = bit_rate_kbps;
+ codec.width = width;
+ codec.height = height;
+ EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, codec));
+
+ webrtc::ViEImageProcess *image_process =
+ webrtc::ViEImageProcess::GetInterface(video_engine_interface);
+ EXPECT_TRUE(image_process);
+
+ // Setup the effect filters
+ CreatedTimestampEffectFilter create_filter(frame_drop_detector);
+ EXPECT_EQ(0, image_process->RegisterSendEffectFilter(video_channel,
+ create_filter));
+ DecodedTimestampEffectFilter decode_filter(frame_drop_detector);
+ EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
+ decode_filter));
+ // Send video.
+ EXPECT_EQ(0, base_interface->StartSend(video_channel));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ // Cleanup.
+ EXPECT_EQ(0, image_process->DeregisterSendEffectFilter(video_channel));
+ EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(video_channel));
+ image_process->Release();
+ ViETest::Log("Done!");
+
+ WebRtc_Word32 num_rtp_packets = 0;
+ WebRtc_Word32 num_dropped_packets = 0;
+ WebRtc_Word32 num_rtcp_packets = 0;
+ external_transport.GetStats(num_rtp_packets, num_dropped_packets,
+ num_rtcp_packets);
+ ViETest::Log("RTP packets : %5d", num_rtp_packets);
+ ViETest::Log("Dropped packets: %5d", num_dropped_packets);
+ ViETest::Log("RTCP packets : %5d", num_rtcp_packets);
+
+ // ***************************************************************
+ // Testing finished. Tear down Video Engine
+ // ***************************************************************
+ EXPECT_EQ(0, base_interface->StopSend(video_channel));
+ EXPECT_EQ(0, base_interface->StopReceive(video_channel));
+ EXPECT_EQ(0, network_interface->DeregisterSendTransport(video_channel));
+ EXPECT_EQ(0, render_interface->StopRender(capture_id));
+ EXPECT_EQ(0, render_interface->StopRender(video_channel));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
+ EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
+ EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
+}
+
+void FixOutputFileForComparison(const std::string& output_file,
+ int frame_length_in_bytes,
+ const std::vector<Frame*>& frames) {
+ webrtc::test::FrameReaderImpl frame_reader(output_file,
+ frame_length_in_bytes);
+ const std::string temp_file = output_file + ".fixed";
+ webrtc::test::FrameWriterImpl frame_writer(temp_file, frame_length_in_bytes);
+ frame_reader.Init();
+ frame_writer.Init();
+
+ ASSERT_FALSE(frames.front()->dropped_at_render) << "It should not be "
+ "possible to drop the first frame. Both because we don't have anything "
+ "useful to fill that gap with and it is impossible to detect it without "
+ "any previous timestamps to compare with.";
+
+ WebRtc_UWord8* last_frame_data = new WebRtc_UWord8[frame_length_in_bytes];
+
+ // Process the file and write frame duplicates for all dropped frames.
+ for (std::vector<Frame*>::const_iterator it = frames.begin();
+ it != frames.end(); ++it) {
+ if ((*it)->dropped_at_render) {
+ // Write the previous frame to the output file:
+ EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
+ } else {
+ EXPECT_TRUE(frame_reader.ReadFrame(last_frame_data));
+ EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
+ }
+ }
+ delete[] last_frame_data;
+ frame_reader.Close();
+ frame_writer.Close();
+ ASSERT_EQ(0, std::remove(output_file.c_str()));
+ ASSERT_EQ(0, std::rename(temp_file.c_str(), output_file.c_str()));
+}
+
+void FrameDropDetector::ReportFrameState(State state, unsigned int timestamp) {
+ dirty_ = true;
+ switch (state) {
+ case kCreated: {
+ int number = created_frames_vector_.size();
+ Frame* frame = new Frame(number, timestamp);
+ frame->created_timestamp_in_us_ =
+ webrtc::TickTime::MicrosecondTimestamp();
+ created_frames_vector_.push_back(frame);
+ created_frames_[timestamp] = frame;
+ num_created_frames_++;
+ break;
+ }
+ case kSent:
+ sent_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
+ if (timestamp_diff_ == 0) {
+ // When the first created frame arrives we calculate the fixed
+ // difference between the timestamps of the frames entering and leaving
+ // the encoder. This diff is used to identify the frames from the
+ // created_frames_ map.
+ timestamp_diff_ =
+ timestamp - created_frames_vector_.front()->frame_timestamp_;
+ }
+ num_sent_frames_++;
+ break;
+ case kReceived:
+ received_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
+ num_received_frames_++;
+ break;
+ case kDecoded:
+ decoded_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
+ num_decoded_frames_++;
+ break;
+ case kRendered:
+ rendered_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
+ num_rendered_frames_++;
+ break;
+ }
+}
+
+void FrameDropDetector::CalculateResults() {
+ // Fill in all fields of the Frame objects in the created_frames_ map.
+ // Iterate over the maps from converted timestamps to the arrival timestamps.
+ std::map<unsigned int, int64_t>::const_iterator it;
+ for (it = sent_frames_.begin(); it != sent_frames_.end(); ++it) {
+ int created_timestamp = it->first - timestamp_diff_;
+ created_frames_[created_timestamp]->sent_timestamp_in_us_ = it->second;
+ }
+ for (it = received_frames_.begin(); it != received_frames_.end(); ++it) {
+ int created_timestamp = it->first - timestamp_diff_;
+ created_frames_[created_timestamp]->received_timestamp_in_us_ = it->second;
+ }
+ for (it = decoded_frames_.begin(); it != decoded_frames_.end(); ++it) {
+ int created_timestamp = it->first - timestamp_diff_;
+ created_frames_[created_timestamp]->decoded_timestamp_in_us_ =it->second;
+ }
+ for (it = rendered_frames_.begin(); it != rendered_frames_.end(); ++it) {
+ int created_timestamp = it->first - timestamp_diff_;
+ created_frames_[created_timestamp]->rendered_timestamp_in_us_ = it->second;
+ }
+ // Find out where the frames were not present in the different states.
+ dropped_frames_at_send_ = 0;
+ dropped_frames_at_receive_ = 0;
+ dropped_frames_at_decode_ = 0;
+ dropped_frames_at_render_ = 0;
+ for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
+ it != created_frames_vector_.end(); ++it) {
+ int encoded_timestamp = (*it)->frame_timestamp_ + timestamp_diff_;
+ if (sent_frames_.find(encoded_timestamp) == sent_frames_.end()) {
+ (*it)->dropped_at_send = true;
+ dropped_frames_at_send_++;
+ }
+ if (received_frames_.find(encoded_timestamp) == received_frames_.end()) {
+ (*it)->dropped_at_receive = true;
+ dropped_frames_at_receive_++;
+ }
+ if (decoded_frames_.find(encoded_timestamp) == decoded_frames_.end()) {
+ (*it)->dropped_at_decode = true;
+ dropped_frames_at_decode_++;
+ }
+ if (rendered_frames_.find(encoded_timestamp) == rendered_frames_.end()) {
+ (*it)->dropped_at_render = true;
+ dropped_frames_at_render_++;
+ }
+ }
+ dirty_ = false;
+}
+
+void FrameDropDetector::PrintReport() {
+ assert(!dirty_);
+ ViETest::Log("Frame Drop Detector report:");
+ ViETest::Log(" Created frames: %ld", created_frames_.size());
+ ViETest::Log(" Sent frames: %ld", sent_frames_.size());
+ ViETest::Log(" Received frames: %ld", received_frames_.size());
+ ViETest::Log(" Decoded frames: %ld", decoded_frames_.size());
+ ViETest::Log(" Rendered frames: %ld", rendered_frames_.size());
+
+ // Display all frames and stats for them:
+ long last_created = 0;
+ long last_sent = 0;
+ long last_received = 0;
+ long last_decoded = 0;
+ long last_rendered = 0;
+ ViETest::Log("\nDeltas between sent frames and drop status:");
+ ViETest::Log("Unit: Microseconds");
+ ViETest::Log("Frame Created Sent Received Decoded Rendered "
+ "Dropped at Dropped at Dropped at Dropped at");
+ ViETest::Log(" nbr delta delta delta delta delta "
+ " Send? Receive? Decode? Render?");
+ for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
+ it != created_frames_vector_.end(); ++it) {
+ int created_delta =
+ static_cast<int>((*it)->created_timestamp_in_us_ - last_created);
+ int sent_delta = (*it)->dropped_at_send ? -1 :
+ static_cast<int>((*it)->sent_timestamp_in_us_ - last_sent);
+ int received_delta = (*it)->dropped_at_receive ? -1 :
+ static_cast<int>((*it)->received_timestamp_in_us_ - last_received);
+ int decoded_delta = (*it)->dropped_at_decode ? -1 :
+ static_cast<int>((*it)->decoded_timestamp_in_us_ - last_decoded);
+ int rendered_delta = (*it)->dropped_at_render ? -1 :
+ static_cast<int>((*it)->rendered_timestamp_in_us_ - last_rendered);
+
+ // Set values to -1 for the first frame:
+ if ((*it)->number_ == 0) {
+ created_delta = -1;
+ sent_delta = -1;
+ received_delta = -1;
+ decoded_delta = -1;
+ rendered_delta = -1;
+ }
+ ViETest::Log("%5d %8d %8d %8d %8d %8d %10s %10s %10s %10s",
+ (*it)->number_,
+ created_delta,
+ sent_delta,
+ received_delta,
+ decoded_delta,
+ rendered_delta,
+ (*it)->dropped_at_send ? "DROPPED" : " ",
+ (*it)->dropped_at_receive ? "DROPPED" : " ",
+ (*it)->dropped_at_decode ? "DROPPED" : " ",
+ (*it)->dropped_at_render ? "DROPPED" : " ");
+ last_created = (*it)->created_timestamp_in_us_;
+ if (!(*it)->dropped_at_send) {
+ last_sent = (*it)->sent_timestamp_in_us_;
+ }
+ if (!(*it)->dropped_at_receive) {
+ last_received = (*it)->received_timestamp_in_us_;
+ }
+ if (!(*it)->dropped_at_decode) {
+ last_decoded = (*it)->decoded_timestamp_in_us_;
+ }
+ if (!(*it)->dropped_at_render) {
+ last_rendered = (*it)->rendered_timestamp_in_us_;
+ }
+ }
+ ViETest::Log("\nLatency between states (-1 means N/A because of drop):");
+ ViETest::Log("Unit: Microseconds");
+ ViETest::Log("Frame Created Sent Received Decoded Total "
+ " Total");
+ ViETest::Log(" nbr ->Sent ->Received ->Decoded ->Rendered latency "
+ " latency");
+ ViETest::Log(" (incl network)"
+ "(excl network)");
+ for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
+ it != created_frames_vector_.end(); ++it) {
+ int created_to_sent = (*it)->dropped_at_send ? -1 :
+ static_cast<int>((*it)->sent_timestamp_in_us_ -
+ (*it)->created_timestamp_in_us_);
+ int sent_to_received = (*it)->dropped_at_receive ? -1 :
+ static_cast<int>((*it)->received_timestamp_in_us_ -
+ (*it)->sent_timestamp_in_us_);
+ int received_to_decoded = (*it)->dropped_at_decode ? -1 :
+ static_cast<int>((*it)->decoded_timestamp_in_us_ -
+ (*it)->received_timestamp_in_us_);
+ int decoded_to_render = (*it)->dropped_at_render ? -1 :
+ static_cast<int>((*it)->rendered_timestamp_in_us_ -
+ (*it)->decoded_timestamp_in_us_);
+ int total_latency_incl_network = (*it)->dropped_at_render ? -1 :
+ static_cast<int>((*it)->rendered_timestamp_in_us_ -
+ (*it)->created_timestamp_in_us_);
+ int total_latency_excl_network = (*it)->dropped_at_render ? -1 :
+ static_cast<int>((*it)->rendered_timestamp_in_us_ -
+ (*it)->created_timestamp_in_us_ - sent_to_received);
+ ViETest::Log("%5d %9d %9d %9d %9d %12d %12d",
+ (*it)->number_,
+ created_to_sent,
+ sent_to_received,
+ received_to_decoded,
+ decoded_to_render,
+ total_latency_incl_network,
+ total_latency_excl_network);
+ }
+ // Find and print the dropped frames.
+ ViETest::Log("\nTotal # dropped frames at:");
+ ViETest::Log(" Send : %d", dropped_frames_at_send_);
+ ViETest::Log(" Receive: %d", dropped_frames_at_receive_);
+ ViETest::Log(" Decode : %d", dropped_frames_at_decode_);
+ ViETest::Log(" Render : %d", dropped_frames_at_render_);
+}
+
+void FrameDropDetector::PrintDebugDump() {
+ assert(!dirty_);
+ ViETest::Log("\nPrintDebugDump: Frame objects:");
+ ViETest::Log("Frame FrTimeStamp Created Sent Received Decoded"
+ " Rendered ");
+ for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
+ it != created_frames_vector_.end(); ++it) {
+ ViETest::Log("%5d %11d %11d %11d %11d %11d %11d",
+ (*it)->number_,
+ (*it)->frame_timestamp_,
+ (*it)->created_timestamp_in_us_,
+ (*it)->sent_timestamp_in_us_,
+ (*it)->received_timestamp_in_us_,
+ (*it)->decoded_timestamp_in_us_,
+ (*it)->rendered_timestamp_in_us_);
+ }
+ std::vector<int> mismatch_frame_num_list;
+ for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
+ it != created_frames_vector_.end(); ++it) {
+ if ((*it)->dropped_at_render != (*it)->dropped_at_decode) {
+ mismatch_frame_num_list.push_back((*it)->number_);
+ }
+ }
+ if (mismatch_frame_num_list.size() > 0) {
+ ViETest::Log("\nDecoded/Rendered mismatches:");
+ ViETest::Log("Frame FrTimeStamp Created Sent Received "
+ "Decoded Rendered ");
+ for (std::vector<int>::const_iterator it = mismatch_frame_num_list.begin();
+ it != mismatch_frame_num_list.end(); ++it) {
+ Frame* frame = created_frames_vector_[*it];
+ ViETest::Log("%5d %11d %11d %11d %11d %11d %11d",
+ frame->number_,
+ frame->frame_timestamp_,
+ frame->created_timestamp_in_us_,
+ frame->sent_timestamp_in_us_,
+ frame->received_timestamp_in_us_,
+ frame->decoded_timestamp_in_us_,
+ frame->rendered_timestamp_in_us_);
+ }
+ }
+
+ ViETest::Log("\nReportFrameState method invocations:");
+ ViETest::Log(" Created : %d", num_created_frames_);
+ ViETest::Log(" Send : %d", num_sent_frames_);
+ ViETest::Log(" Received: %d", num_received_frames_);
+ ViETest::Log(" Decoded : %d", num_decoded_frames_);
+ ViETest::Log(" Rendered: %d", num_rendered_frames_);
+}
+
+const std::vector<Frame*>& FrameDropDetector::GetAllFrames() {
+ assert(!dirty_);
+ return created_frames_vector_;
+}
+
+int FrameDropDetector::GetNumberOfFramesDroppedAt(State state) {
+ assert(!dirty_);
+ switch (state) {
+ case kSent:
+ return dropped_frames_at_send_;
+ case kReceived:
+ return dropped_frames_at_receive_;
+ case kDecoded:
+ return dropped_frames_at_decode_;
+ case kRendered:
+ return dropped_frames_at_render_;
+ default:
+ return 0;
+ }
+}
+
+int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
+ unsigned char *buffer, int buffer_size, uint32_t time_stamp,
+ int64_t render_time) {
+ // Register that this frame has been rendered:
+ frame_drop_detector_->ReportFrameState(FrameDropDetector::kRendered,
+ time_stamp);
+ return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
+ time_stamp, render_time);
+}
+
+int FrameDropMonitoringRemoteFileRenderer::FrameSizeChange(
+ unsigned int width, unsigned int height, unsigned int number_of_streams) {
+ return ViEToFileRenderer::FrameSizeChange(width, height, number_of_streams);
+}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h
new file mode 100644
index 0000000..80575c3
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h
@@ -0,0 +1,229 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
+
+#include <map>
+#include <vector>
+
+#include "video_engine/include/vie_codec.h"
+#include "video_engine/include/vie_image_process.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
+
+class FrameDropDetector;
+class TbInterfaces;
+
+// Initializes the Video engine and its components, runs video playback using
+// for KAutoTestSleepTimeMs milliseconds, then shuts down everything.
+// The bit rate and packet loss parameters should be configured so that
+// frames are dropped, in order to test the frame drop detection that is
+// performed by the FrameDropDetector class.
+void TestFullStack(const TbInterfaces& interfaces,
+ int capture_id,
+ int video_channel,
+ int width,
+ int height,
+ int bit_rate_kbps,
+ int packet_loss_percent,
+ int network_delay_ms,
+ FrameDropDetector* frame_drop_detector);
+
+// A frame in a video file. The four different points in the stack when
+// register the frame state are (in time order): created, transmitted, decoded,
+// rendered.
+class Frame {
+ public:
+ Frame(int number, unsigned int timestamp)
+ : number_(number),
+ frame_timestamp_(timestamp),
+ created_timestamp_in_us_(-1),
+ sent_timestamp_in_us_(-1),
+ received_timestamp_in_us_(-1),
+ decoded_timestamp_in_us_(-1),
+ rendered_timestamp_in_us_(-1),
+ dropped_at_send(false),
+ dropped_at_receive(false),
+ dropped_at_decode(false),
+ dropped_at_render(false) {}
+
+ // Frame number, starting at 0.
+ int number_;
+
+ // Frame timestamp, that is used by Video Engine and RTP headers and set when
+ // the frame is sent into the stack.
+ unsigned int frame_timestamp_;
+
+ // Timestamps for our measurements of when the frame is in different states.
+ int64_t created_timestamp_in_us_;
+ int64_t sent_timestamp_in_us_;
+ int64_t received_timestamp_in_us_;
+ int64_t decoded_timestamp_in_us_;
+ int64_t rendered_timestamp_in_us_;
+
+ // Where the frame was dropped (more than one may be true).
+ bool dropped_at_send;
+ bool dropped_at_receive;
+ bool dropped_at_decode;
+ bool dropped_at_render;
+};
+
+// Fixes the output file by copying the last successful frame into the place
+// where the dropped frame would be, for all dropped frames (if any).
+// This method will not be able to fix data for the first frame if that is
+// dropped, since there'll be no previous frame to copy. This case should never
+// happen because of encoder frame dropping at least.
+// Parameters:
+// output_file The output file to modify (pad with frame copies
+// for all dropped frames)
+// frame_length_in_bytes Byte length of each frame.
+// frames A vector of all Frame objects. Must be sorted by
+// frame number. If empty this method will do nothing.
+void FixOutputFileForComparison(const std::string& output_file,
+ int frame_length_in_bytes,
+ const std::vector<Frame*>& frames);
+
+// Handles statistics about dropped frames. Frames travel through the stack
+// with different timestamps. The frames created and sent to the encoder have
+// one timestamp on the sending side while the decoded/rendered frames have
+// another timestamp on the receiving side. The difference between these
+// timestamps is fixed, which we can use to identify the frames when they
+// arrive, since the FrameDropDetector class gets data reported from both sides.
+// The four different points in the stack when this class examines the frame
+// states are (in time order): created, sent, received, decoded, rendered.
+//
+// The flow can be visualized like this:
+//
+// Created Sent Received Decoded Rendered
+// +-------+ | +-------+ | +---------+ | +------+ +-------+ | +--------+
+// |Capture| | |Encoder| | | Ext. | | |Jitter| |Decoder| | | Ext. |
+// | device|---->| |-->|transport|-->|buffer|->| |---->|renderer|
+// +-------+ +-------+ +---------+ +------+ +-------+ +--------+
+//
+// This class has no intention of being thread-safe.
+class FrameDropDetector {
+ public:
+ enum State {
+ // A frame being created, i.e. sent to the encoder; the first step of
+ // a frame's life cycle. This timestamp becomes the frame timestamp in the
+ // Frame objects.
+ kCreated,
+ // A frame being sent in external transport (to the simulated network). This
+ // timestamp differs from the one in the Created state by a constant diff.
+ kSent,
+ // A frame being received in external transport (from the simulated
+ // network). This timestamp differs from the one in the Created state by a
+ // constant diff.
+ kReceived,
+ // A frame that has been decoded in the decoder. This timestamp differs
+ // from the one in the Created state by a constant diff.
+ kDecoded,
+ // A frame that has been rendered; the last step of a frame's life cycle.
+ // This timestamp differs from the one in the Created state by a constant
+ // diff.
+ kRendered
+ };
+
+ FrameDropDetector()
+ : dirty_(true),
+ dropped_frames_at_send_(0),
+ dropped_frames_at_receive_(0),
+ dropped_frames_at_decode_(0),
+ dropped_frames_at_render_(0),
+ num_created_frames_(0),
+ num_sent_frames_(0),
+ num_received_frames_(0),
+ num_decoded_frames_(0),
+ num_rendered_frames_(0),
+ timestamp_diff_(0) {}
+
+ // Reports a frame has reached a state in the frame life cycle.
+ void ReportFrameState(State state, unsigned int timestamp);
+
+ // Uses all the gathered timestamp information to calculate which frames have
+ // been dropped during the test and where they were dropped. Not until
+ // this method has been executed, the Frame objects will have all fields
+ // filled with the proper timestamp information.
+ void CalculateResults();
+
+ // Calculates the number of frames have been registered as dropped at the
+ // specified state of the frame life cycle.
+ // CalculateResults() must be called before calling this method.
+ int GetNumberOfFramesDroppedAt(State state);
+
+ // Gets a vector of all the created frames.
+ // CalculateResults() must be called before calling this method to have all
+ // fields of the Frame objects to represent the current state.
+ const std::vector<Frame*>& GetAllFrames();
+
+ // Prints a detailed report about all the different frame states and which
+ // ones are detected as dropped, using ViETest::Log.
+ // CalculateResults() must be called before calling this method.
+ void PrintReport();
+
+ // Prints all the timestamp maps. Mainly used for debugging purposes to find
+ // missing timestamps.
+ void PrintDebugDump();
+ private:
+ // Will be false until CalculateResults() is called. Switches to true
+ // as soon as new timestamps are reported using ReportFrameState().
+ bool dirty_;
+
+ // Map of frame creation timestamps to all Frame objects.
+ std::map<unsigned int, Frame*> created_frames_;
+
+ // Maps converted frame timestamps (differ from creation timestamp) to the
+ // time they arrived in the different states of the frame's life cycle.
+ std::map<unsigned int, int64_t> sent_frames_;
+ std::map<unsigned int, int64_t> received_frames_;
+ std::map<unsigned int, int64_t> decoded_frames_;
+ std::map<unsigned int, int64_t> rendered_frames_;
+
+ // A vector with the frames sorted in their created order.
+ std::vector<Frame*> created_frames_vector_;
+
+ // Statistics.
+ int dropped_frames_at_send_;
+ int dropped_frames_at_receive_;
+ int dropped_frames_at_decode_;
+ int dropped_frames_at_render_;
+
+ int num_created_frames_;
+ int num_sent_frames_;
+ int num_received_frames_;
+ int num_decoded_frames_;
+ int num_rendered_frames_;
+
+ // The constant diff between the created and transmitted frames, since their
+ // timestamps are converted.
+ unsigned int timestamp_diff_;
+};
+
+// Tracks which frames are received on the remote side and reports back to the
+// FrameDropDetector class when they are rendered.
+class FrameDropMonitoringRemoteFileRenderer : public ViEToFileRenderer {
+ public:
+ explicit FrameDropMonitoringRemoteFileRenderer(
+ FrameDropDetector* frame_drop_detector)
+ : frame_drop_detector_(frame_drop_detector) {}
+ virtual ~FrameDropMonitoringRemoteFileRenderer() {}
+
+ // Implementation of ExternalRenderer:
+ int FrameSizeChange(unsigned int width, unsigned int height,
+ unsigned int number_of_streams);
+ int DeliverFrame(unsigned char* buffer, int buffer_size,
+ uint32_t time_stamp,
+ int64_t render_time);
+ private:
+ FrameDropDetector* frame_drop_detector_;
+};
+
+#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
new file mode 100644
index 0000000..f4cc390
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "framedrop_primitives.h"
+
+#include <cstdio>
+#include <vector>
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+#include "testsupport/frame_reader.h"
+#include "testsupport/frame_writer.h"
+
+namespace webrtc {
+
+const std::string kOutputFilename = "temp_outputfile.tmp";
+const int kFrameLength = 1000;
+
+class FrameDropPrimitivesTest: public testing::Test {
+ protected:
+ FrameDropPrimitivesTest() {}
+ virtual ~FrameDropPrimitivesTest() {}
+ void SetUp() {
+ // Cleanup any previous output file.
+ std::remove(kOutputFilename.c_str());
+ }
+ void TearDown() {
+ // Cleanup the temporary file.
+ std::remove(kOutputFilename.c_str());
+ }
+};
+
+TEST_F(FrameDropPrimitivesTest, FixOutputFileForComparison) {
+ // Create test frame objects, where the second and fourth frame is marked
+ // as dropped at rendering.
+ std::vector<Frame*> frames;
+ Frame first_frame(0, kFrameLength);
+ Frame second_frame(0, kFrameLength);
+ Frame third_frame(0, kFrameLength);
+ Frame fourth_frame(0, kFrameLength);
+
+ second_frame.dropped_at_render = true;
+ fourth_frame.dropped_at_render = true;
+
+ frames.push_back(&first_frame);
+ frames.push_back(&second_frame);
+ frames.push_back(&third_frame);
+ frames.push_back(&fourth_frame);
+
+ // Prepare data for the first and third frames:
+ WebRtc_UWord8 first_frame_data[kFrameLength];
+ memset(first_frame_data, 5, kFrameLength); // Fill it with 5's to identify.
+ WebRtc_UWord8 third_frame_data[kFrameLength];
+ memset(third_frame_data, 7, kFrameLength); // Fill it with 7's to identify.
+
+ // Write the first and third frames to the temporary file. This means the fix
+ // method should add two frames of data by filling the file with data from
+ // the first and third frames after executing.
+ webrtc::test::FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
+ EXPECT_TRUE(frame_writer.Init());
+ EXPECT_TRUE(frame_writer.WriteFrame(first_frame_data));
+ EXPECT_TRUE(frame_writer.WriteFrame(third_frame_data));
+ frame_writer.Close();
+ EXPECT_EQ(2 * kFrameLength,
+ static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
+
+ FixOutputFileForComparison(kOutputFilename, kFrameLength, frames);
+
+ // Verify that the output file has correct size.
+ EXPECT_EQ(4 * kFrameLength,
+ static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
+
+ webrtc::test::FrameReaderImpl frame_reader(kOutputFilename, kFrameLength);
+ frame_reader.Init();
+ WebRtc_UWord8 read_buffer[kFrameLength];
+ EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
+ EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
+ EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
+ EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
+
+ EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
+ EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
+ EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
+ EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
+
+ frame_reader.Close();
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc
new file mode 100644
index 0000000..6198a64
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc
@@ -0,0 +1,130 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "general_primitives.h"
+
+#include "video_capture_factory.h"
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+#include "vie_to_file_renderer.h"
+
+void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
+ unsigned char* device_name,
+ unsigned int device_name_length,
+ int* device_id,
+ webrtc::VideoCaptureModule** device_video) {
+
+ bool capture_device_set = false;
+ webrtc::VideoCaptureModule::DeviceInfo *dev_info =
+ webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
+
+ const unsigned int kMaxUniqueIdLength = 256;
+ WebRtc_UWord8 unique_id[kMaxUniqueIdLength];
+ memset(unique_id, 0, kMaxUniqueIdLength);
+
+ for (unsigned int i = 0; i < dev_info->NumberOfDevices(); i++) {
+ EXPECT_EQ(0, dev_info->GetDeviceName(i, device_name, device_name_length,
+ unique_id, kMaxUniqueIdLength));
+
+ *device_video = webrtc::VideoCaptureFactory::Create(4571, unique_id);
+ EXPECT_TRUE(*device_video != NULL);
+
+ (*device_video)->AddRef();
+
+ int error = capture->AllocateCaptureDevice(**device_video, *device_id);
+ if (error == 0) {
+ ViETest::Log("Using capture device: %s, captureId: %d.",
+ device_name, *device_id);
+ capture_device_set = true;
+ break;
+ } else {
+ (*device_video)->Release();
+ (*device_video) = NULL;
+ }
+ }
+ delete dev_info;
+ EXPECT_TRUE(capture_device_set) << "Found no suitable camera on your system.";
+}
+
+void RenderInWindow(webrtc::ViERender* video_render_interface,
+ int frame_provider_id,
+ void* os_window,
+ float z_index) {
+ EXPECT_EQ(0,
+ video_render_interface->AddRenderer(frame_provider_id, os_window,
+ z_index, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, video_render_interface->StartRender(frame_provider_id));
+}
+
+void RenderToFile(webrtc::ViERender* renderer_interface,
+ int frame_provider_id,
+ ViEToFileRenderer *to_file_renderer) {
+ EXPECT_EQ(0, renderer_interface->AddRenderer(
+ frame_provider_id, webrtc::kVideoI420, to_file_renderer));
+ EXPECT_EQ(0, renderer_interface->StartRender(frame_provider_id));
+}
+
+void StopAndRemoveRenderers(webrtc::ViEBase* base_interface,
+ webrtc::ViERender* render_interface,
+ int channel_id,
+ int capture_id) {
+ EXPECT_EQ(0, render_interface->StopRender(channel_id));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(channel_id));
+ EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
+}
+
+void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
+ int video_channel) {
+ EXPECT_EQ(0, rtcp_interface->SetRTCPStatus(video_channel,
+ webrtc::kRtcpCompound_RFC4585));
+ EXPECT_EQ(0, rtcp_interface->SetKeyFrameRequestMethod(
+ video_channel, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, rtcp_interface->SetTMMBRStatus(video_channel, true));
+}
+
+bool FindSpecificCodec(webrtc::VideoCodecType of_type,
+ webrtc::ViECodec* codec_interface,
+ webrtc::VideoCodec* result) {
+
+ memset(result, 0, sizeof(webrtc::VideoCodec));
+
+ for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
+ webrtc::VideoCodec codec;
+ memset(&codec, 0, sizeof(webrtc::VideoCodec));
+ if (codec_interface->GetCodec(i, codec) != 0) {
+ return false;
+ }
+ if (codec.codecType == of_type) {
+ // Done
+ *result = codec;
+ return true;
+ }
+ }
+ // Didn't find it
+ return false;
+}
+
+void SetSuitableResolution(webrtc::VideoCodec* video_codec,
+ int forced_codec_width,
+ int forced_codec_height) {
+ if (forced_codec_width != kDoNotForceResolution &&
+ forced_codec_height != kDoNotForceResolution) {
+ video_codec->width = forced_codec_width;
+ video_codec->height = forced_codec_height;
+ } else if (video_codec->codecType == webrtc::kVideoCodecI420) {
+ // I420 is very bandwidth heavy, so limit it here.
+ video_codec->width = 176;
+ video_codec->height = 144;
+ } else {
+ // Otherwise go with 640x480.
+ video_codec->width = 640;
+ video_codec->height = 480;
+ }
+}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h
new file mode 100644
index 0000000..07eca0d
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
+
+class ViEToFileRenderer;
+
+#include "common_types.h"
+
+namespace webrtc {
+class VideoCaptureModule;
+class ViEBase;
+class ViECapture;
+class ViECodec;
+class ViERender;
+class ViERTP_RTCP;
+struct VideoCodec;
+}
+
+// This constant can be used as input to various functions to not force the
+// codec resolution.
+const int kDoNotForceResolution = 0;
+
+// Finds a suitable capture device (e.g. camera) on the current system
+// and allocates it. Details about the found device are filled into the out
+// parameters. If this operation fails, device_id is assigned a negative value
+// and number_of_errors is incremented.
+void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
+ unsigned char* device_name,
+ const unsigned int kDeviceNameLength,
+ int* device_id,
+ webrtc::VideoCaptureModule** device_video);
+
+// Sets up rendering in a window previously created using a Window Manager
+// (See vie_window_manager_factory.h for more details on how to make one of
+// those). The frame provider id is a source of video frames, for instance
+// a capture device or a video channel.
+void RenderInWindow(webrtc::ViERender* video_render_interface,
+ int frame_provider_id,
+ void* os_window,
+ float z_index);
+
+// Similar in function to RenderInWindow, this function instead renders to
+// a file using a to-file-renderer. The frame provider id is a source of
+// video frames, for instance a capture device or a video channel.
+void RenderToFile(webrtc::ViERender* renderer_interface,
+ int frame_provider_id,
+ ViEToFileRenderer* to_file_renderer);
+
+// Stops all rendering given the normal case that we have a capture device
+// and a video channel set up for rendering.
+void StopAndRemoveRenderers(webrtc::ViEBase* base_interface,
+ webrtc::ViERender* render_interface,
+ int channel_id,
+ int capture_id);
+
+// Configures RTP-RTCP.
+void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
+ int video_channel);
+
+// Finds a codec in the codec list. Returns true on success, false otherwise.
+// The resulting codec is filled into result on success but is zeroed out
+// on failure.
+bool FindSpecificCodec(webrtc::VideoCodecType of_type,
+ webrtc::ViECodec* codec_interface,
+ webrtc::VideoCodec* result);
+
+// Sets up the provided codec with a resolution that takes individual codec
+// quirks into account (except if the forced* variables are
+// != kDoNotForceResolution)
+void SetSuitableResolution(webrtc::VideoCodec* video_codec,
+ int forced_codec_width,
+ int forced_codec_height);
+
+#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/source/Android.mk b/trunk/src/video_engine/test/auto_test/source/Android.mk
new file mode 100644
index 0000000..9e19418
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/Android.mk
@@ -0,0 +1,65 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+# voice engine test app
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ vie_autotest.cc \
+ vie_autotest_android.cc \
+ vie_autotest_base.cc \
+ vie_autotest_capture.cc \
+ vie_autotest_codec.cc \
+ vie_autotest_encryption.cc \
+ vie_autotest_file.cc \
+ vie_autotest_image_process.cc \
+ vie_autotest_loopback.cc \
+ vie_autotest_network.cc \
+ vie_autotest_render.cc \
+ vie_autotest_rtp_rtcp.cc \
+ vie_comparison_tests.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../helpers \
+ $(LOCAL_PATH)/../primitives \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../include \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../../modules/video_coding/codecs/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface \
+ $(LOCAL_PATH)/../../../../modules/video_render/main/interface \
+ $(LOCAL_PATH)/../../../../modules/interface \
+ $(LOCAL_PATH)/../../../../modules/video_capture/main/interface \
+ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
+ $(LOCAL_PATH)/../../../../voice_engine/main/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libwebrtc
+
+LOCAL_MODULE:= webrtc_video_test
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_EXECUTABLE)
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc b/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc
new file mode 100644
index 0000000..df837c7
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc
@@ -0,0 +1,324 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * tb_I420_codec.cc
+ *
+ */
+
+#include "tb_I420_codec.h"
+#include <string.h>
+#include <stdio.h>
+#include <assert.h>
+
+TbI420Encoder::TbI420Encoder() :
+ _inited(false), _encodedImage(), _encodedCompleteCallback(NULL)
+{
+ //
+ memset(&_functionCalls, 0, sizeof(_functionCalls));
+}
+
+TbI420Encoder::~TbI420Encoder()
+{
+ _inited = false;
+ if (_encodedImage._buffer != NULL)
+ {
+ delete[] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ }
+}
+
+WebRtc_Word32 TbI420Encoder::VersionStatic(WebRtc_Word8* version,
+ WebRtc_Word32 length)
+{
+ const WebRtc_Word8* str = "I420 version 1.0.0\n";
+ WebRtc_Word32 verLen = (WebRtc_Word32) strlen(str);
+ if (verLen > length)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ strncpy(version, str, length);
+ return verLen;
+}
+
+WebRtc_Word32 TbI420Encoder::Version(WebRtc_Word8 *version,
+ WebRtc_Word32 length) const
+{
+ return VersionStatic(version, length);
+}
+
+WebRtc_Word32 TbI420Encoder::Release()
+{
+ _functionCalls.Release++;
+ // should allocate an encoded frame and then release it here, for that we
+ // actaully need an init flag
+ if (_encodedImage._buffer != NULL)
+ {
+ delete[] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ }
+ _inited = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::Reset()
+{
+ _functionCalls.Reset++;
+ if (!_inited)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+
+}
+
+WebRtc_Word32 TbI420Encoder::SetChannelParameters(WebRtc_UWord32 packetLoss,
+ int rtt) {
+ return 0;
+}
+
+WebRtc_Word32 TbI420Encoder::InitEncode(const webrtc::VideoCodec* inst,
+ WebRtc_Word32 /*numberOfCores*/,
+ WebRtc_UWord32 /*maxPayloadSize */)
+{
+ _functionCalls.InitEncode++;
+ if (inst == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (inst->width < 1 || inst->height < 1)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+
+ // allocating encoded memory
+ if (_encodedImage._buffer != NULL)
+ {
+ delete[] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ _encodedImage._size = 0;
+ }
+ const WebRtc_UWord32 newSize = (3 * inst->width * inst->height) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _encodedImage._size = newSize;
+ _encodedImage._buffer = newBuffer;
+
+ // if no memeory allocation, no point to init
+ _inited = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::Encode(
+ const webrtc::RawImage& inputImage,
+ const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
+ const webrtc::VideoFrameType* /*frameType*/)
+{
+ _functionCalls.Encode++;
+ if (!_inited)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (_encodedCompleteCallback == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+
+ _encodedImage._frameType = webrtc::kKeyFrame; // no coding
+ _encodedImage._timeStamp = inputImage._timeStamp;
+ _encodedImage._encodedHeight = inputImage._height;
+ _encodedImage._encodedWidth = inputImage._width;
+ if (inputImage._length > _encodedImage._size)
+ {
+
+ // allocating encoded memory
+ if (_encodedImage._buffer != NULL)
+ {
+ delete[] _encodedImage._buffer;
+ _encodedImage._buffer = NULL;
+ _encodedImage._size = 0;
+ }
+ const WebRtc_UWord32 newSize = (3 * _encodedImage._encodedWidth
+ * _encodedImage._encodedHeight) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _encodedImage._size = newSize;
+ _encodedImage._buffer = newBuffer;
+ }
+ assert(_encodedImage._size >= inputImage._length);
+ memcpy(_encodedImage._buffer, inputImage._buffer, inputImage._length);
+ _encodedImage._length = inputImage._length;
+ _encodedCompleteCallback->Encoded(_encodedImage);
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::RegisterEncodeCompleteCallback(
+ webrtc::EncodedImageCallback* callback)
+{
+ _functionCalls.RegisterEncodeCompleteCallback++;
+ _encodedCompleteCallback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::SetPacketLoss(WebRtc_UWord32 packetLoss)
+{
+ _functionCalls.SetPacketLoss++;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::SetRates(WebRtc_UWord32 newBitRate,
+ WebRtc_UWord32 frameRate)
+{
+ _functionCalls.SetRates++;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Encoder::SetPeriodicKeyFrames(bool enable)
+{
+ _functionCalls.SetPeriodicKeyFrames++;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+}
+
+WebRtc_Word32 TbI420Encoder::CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
+ WebRtc_Word32 /*size*/)
+{
+ _functionCalls.CodecConfigParameters++;
+ return WEBRTC_VIDEO_CODEC_ERROR;
+}
+TbI420Encoder::FunctionCalls TbI420Encoder::GetFunctionCalls()
+{
+ return _functionCalls;
+}
+
+TbI420Decoder::TbI420Decoder():
+ _decodedImage(), _width(0), _height(0), _inited(false),
+ _decodeCompleteCallback(NULL)
+{
+ memset(&_functionCalls, 0, sizeof(_functionCalls));
+}
+
+TbI420Decoder::~TbI420Decoder()
+{
+ Release();
+}
+
+WebRtc_Word32 TbI420Decoder::Reset()
+{
+ _functionCalls.Reset++;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Decoder::InitDecode(const webrtc::VideoCodec* inst,
+ WebRtc_Word32 /*numberOfCores */)
+{
+ _functionCalls.InitDecode++;
+ if (inst == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ else if (inst->width < 1 || inst->height < 1)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ _width = inst->width;
+ _height = inst->height;
+ _inited = true;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Decoder::Decode(
+ const webrtc::EncodedImage& inputImage,
+ bool /*missingFrames*/,
+ const webrtc::RTPFragmentationHeader* /*fragmentation*/,
+ const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
+ WebRtc_Word64 /*renderTimeMs*/)
+{
+ _functionCalls.Decode++;
+ if (inputImage._buffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (_decodeCompleteCallback == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (inputImage._length <= 0)
+ {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+ }
+ if (!_inited)
+ {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+
+ // Allocate memory for decoded image
+
+ if (_decodedImage._buffer != NULL)
+ {
+ delete[] _decodedImage._buffer;
+ _decodedImage._buffer = NULL;
+ _decodedImage._size = 0;
+ }
+ if (_decodedImage._buffer == NULL)
+ {
+ const WebRtc_UWord32 newSize = (3 * _width * _height) >> 1;
+ WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+ if (newBuffer == NULL)
+ {
+ return WEBRTC_VIDEO_CODEC_MEMORY;
+ }
+ _decodedImage._size = newSize;
+ _decodedImage._buffer = newBuffer;
+ }
+
+ // Set decoded image parameters
+ _decodedImage._height = _height;
+ _decodedImage._width = _width;
+ _decodedImage._timeStamp = inputImage._timeStamp;
+ assert(_decodedImage._size >= inputImage._length);
+ memcpy(_decodedImage._buffer, inputImage._buffer, inputImage._length);
+ _decodedImage._length = inputImage._length;
+ //_decodedImage._buffer = inputImage._buffer;
+
+ _decodeCompleteCallback->Decoded(_decodedImage);
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Decoder::RegisterDecodeCompleteCallback(
+ webrtc::DecodedImageCallback* callback)
+{
+ _functionCalls.RegisterDecodeCompleteCallback++;
+ _decodeCompleteCallback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32 TbI420Decoder::Release()
+{
+ _functionCalls.Release++;
+ if (_decodedImage._buffer != NULL)
+ {
+ delete[] _decodedImage._buffer;
+ _decodedImage._buffer = NULL;
+ }
+ _inited = false;
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+TbI420Decoder::FunctionCalls TbI420Decoder::GetFunctionCalls()
+{
+ return _functionCalls;
+}
+
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc b/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc
new file mode 100644
index 0000000..0d4e46a
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tb_capture_device.h"
+
+TbCaptureDevice::TbCaptureDevice(TbInterfaces& Engine) :
+ captureId(-1),
+ ViE(Engine),
+ vcpm_(NULL)
+{
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ WebRtc_UWord8 deviceName[KMaxDeviceNameLength];
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ WebRtc_UWord8 uniqueId[KMaxUniqueIdLength];
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ bool captureDeviceSet = false;
+
+ webrtc::VideoCaptureModule::DeviceInfo* devInfo =
+ webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
+ for (size_t captureIdx = 0;
+ captureIdx < devInfo->NumberOfDevices();
+ captureIdx++)
+ {
+ EXPECT_EQ(0, devInfo->GetDeviceName(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId,
+ KMaxUniqueIdLength));
+
+ vcpm_ = webrtc::VideoCaptureFactory::Create(
+ captureIdx, uniqueId);
+ if (vcpm_ == NULL) // Failed to open this device. Try next.
+ {
+ continue;
+ }
+ vcpm_->AddRef();
+
+ int error = ViE.capture->AllocateCaptureDevice(*vcpm_, captureId);
+ if (error == 0)
+ {
+ ViETest::Log("Using capture device: %s, captureId: %d", deviceName,
+ captureId);
+ captureDeviceSet = true;
+ break;
+ }
+ }
+ delete devInfo;
+ EXPECT_TRUE(captureDeviceSet);
+ if (!captureDeviceSet) {
+ return;
+ }
+
+ ViETest::Log("Starting capture device %s with captureId %d\n", deviceName,
+ captureId);
+ EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
+}
+
+TbCaptureDevice::~TbCaptureDevice(void)
+{
+ ViETest::Log("Stopping capture device with id %d\n", captureId);
+ EXPECT_EQ(0, ViE.capture->StopCapture(captureId));
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
+ vcpm_->Release();
+}
+
+void TbCaptureDevice::ConnectTo(int videoChannel)
+{
+ EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId, videoChannel));
+}
+
+void TbCaptureDevice::Disconnect(int videoChannel)
+{
+ EXPECT_EQ(0, ViE.capture->DisconnectCaptureDevice(videoChannel));
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc b/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc
new file mode 100644
index 0000000..8670c40
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc
@@ -0,0 +1,445 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tb_external_transport.h"
+
+#include <stdio.h> // printf
+#include <stdlib.h> // rand
+#include <cassert>
+
+#if defined(WEBRTC_LINUX) || defined(__linux__)
+#include <string.h>
+#endif
+#if defined(WEBRTC_MAC)
+#include <cstring>
+#endif
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "vie_network.h"
+
+#if defined(_WIN32)
+#pragma warning(disable: 4355) // 'this' : used in base member initializer list
+#endif
+
+TbExternalTransport::TbExternalTransport(webrtc::ViENetwork& vieNetwork) :
+ _vieNetwork(vieNetwork),
+ _thread(*webrtc::ThreadWrapper::CreateThread(
+ ViEExternalTransportRun, this, webrtc::kHighPriority,
+ "AutotestTransport")),
+ _event(*webrtc::EventWrapper::Create()),
+ _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+ _statCrit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+ _lossRate(0),
+ _networkDelayMs(0),
+ _rtpCount(0),
+ _rtcpCount(0),
+ _dropCount(0),
+ _rtpPackets(),
+ _rtcpPackets(),
+ _send_frame_callback(NULL),
+ _receive_frame_callback(NULL),
+ _temporalLayers(0),
+ _seqNum(0),
+ _sendPID(0),
+ _receivedPID(0),
+ _switchLayer(false),
+ _currentRelayLayer(0),
+ _lastTimeMs(webrtc::TickTime::MillisecondTimestamp()),
+ _checkSSRC(false),
+ _lastSSRC(0),
+ _filterSSRC(false),
+ _SSRC(0),
+ _checkSequenceNumber(0),
+ _firstSequenceNumber(0),
+ _firstRTPTimestamp(0),
+ _lastSendRTPTimestamp(0),
+ _lastReceiveRTPTimestamp(0)
+{
+ srand((int) webrtc::TickTime::MicrosecondTimestamp());
+ unsigned int tId = 0;
+ _thread.Start(tId);
+}
+
+TbExternalTransport::~TbExternalTransport()
+{
+ _thread.SetNotAlive();
+ _event.Set();
+ if (_thread.Stop())
+ {
+ delete &_thread;
+ delete &_event;
+ }
+ delete &_crit;
+ delete &_statCrit;
+}
+
+int TbExternalTransport::SendPacket(int channel, const void *data, int len)
+{
+ // Parse timestamp from RTP header according to RFC 3550, section 5.1.
+ WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
+ WebRtc_UWord32 rtp_timestamp = ptr[4] << 24;
+ rtp_timestamp += ptr[5] << 16;
+ rtp_timestamp += ptr[6] << 8;
+ rtp_timestamp += ptr[7];
+ _crit.Enter();
+ if (_firstRTPTimestamp == 0) {
+ _firstRTPTimestamp = rtp_timestamp;
+ }
+ _crit.Leave();
+ if (_send_frame_callback != NULL &&
+ _lastSendRTPTimestamp != rtp_timestamp) {
+ _send_frame_callback->FrameSent(rtp_timestamp);
+ }
+ _lastSendRTPTimestamp = rtp_timestamp;
+
+ if (_filterSSRC)
+ {
+ WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
+ WebRtc_UWord32 ssrc = ptr[8] << 24;
+ ssrc += ptr[9] << 16;
+ ssrc += ptr[10] << 8;
+ ssrc += ptr[11];
+ if (ssrc != _SSRC)
+ {
+ return len; // return len to avoid error in trace file
+ }
+ }
+ if (_temporalLayers) {
+ // parse out vp8 temporal layers
+ // 12 bytes RTP
+ WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
+
+ if (ptr[12] & 0x80 && // X-bit
+ ptr[13] & 0x20) // T-bit
+ {
+ int offset = 1;
+ if (ptr[13] & 0x80) // PID-bit
+ {
+ offset++;
+ if (ptr[14] & 0x80) // 2 byte PID
+ {
+ offset++;
+ }
+ }
+ if (ptr[13] & 0x40)
+ {
+ offset++;
+ }
+ unsigned char TID = (ptr[13 + offset] >> 5);
+ unsigned int timeMs = NowMs();
+
+ // Every 5 second switch layer
+ if (_lastTimeMs + 5000 < timeMs)
+ {
+ _lastTimeMs = timeMs;
+ _switchLayer = true;
+ }
+ // Switch at the non ref frame
+ if (_switchLayer && (ptr[12] & 0x20))
+ { // N-bit
+ _currentRelayLayer++;
+ if (_currentRelayLayer >= _temporalLayers)
+ _currentRelayLayer = 0;
+
+ _switchLayer = false;
+ printf("\t Switching to layer:%d\n", _currentRelayLayer);
+ }
+ if (_currentRelayLayer < TID)
+ {
+ return len; // return len to avoid error in trace file
+ }
+ if (ptr[14] & 0x80) // 2 byte PID
+ {
+ if(_receivedPID != ptr[15])
+ {
+ _sendPID++;
+ _receivedPID = ptr[15];
+ }
+ } else
+ {
+ if(_receivedPID != ptr[14])
+ {
+ _sendPID++;
+ _receivedPID = ptr[14];
+ }
+ }
+ }
+ }
+ _statCrit.Enter();
+ _rtpCount++;
+ _statCrit.Leave();
+
+ // Packet loss. Never drop packets from the first RTP timestamp, i.e. the
+ // first frame being transmitted.
+ int dropThis = rand() % 100;
+ if (dropThis < _lossRate && _firstRTPTimestamp != rtp_timestamp)
+ {
+ _statCrit.Enter();
+ _dropCount++;
+ _statCrit.Leave();
+ return 0;
+ }
+
+ VideoPacket* newPacket = new VideoPacket();
+ memcpy(newPacket->packetBuffer, data, len);
+
+ if (_temporalLayers)
+ {
+ // rewrite seqNum
+ newPacket->packetBuffer[2] = _seqNum >> 8;
+ newPacket->packetBuffer[3] = _seqNum;
+ _seqNum++;
+
+ // rewrite PID
+ if (newPacket->packetBuffer[14] & 0x80) // 2 byte PID
+ {
+ newPacket->packetBuffer[14] = (_sendPID >> 8) | 0x80;
+ newPacket->packetBuffer[15] = _sendPID;
+ } else
+ {
+ newPacket->packetBuffer[14] = (_sendPID & 0x7f);
+ }
+ }
+ newPacket->length = len;
+ newPacket->channel = channel;
+
+ _crit.Enter();
+ newPacket->receiveTime = NowMs() + _networkDelayMs;
+ _rtpPackets.push_back(newPacket);
+ _event.Set();
+ _crit.Leave();
+ return len;
+}
+
+void TbExternalTransport::RegisterSendFrameCallback(
+ SendFrameCallback* callback) {
+ _send_frame_callback = callback;
+}
+
+void TbExternalTransport::RegisterReceiveFrameCallback(
+ ReceiveFrameCallback* callback) {
+ _receive_frame_callback = callback;
+}
+
+// Set to 0 to disable.
+void TbExternalTransport::SetTemporalToggle(unsigned char layers)
+{
+ _temporalLayers = layers;
+}
+
+int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
+{
+ _statCrit.Enter();
+ _rtcpCount++;
+ _statCrit.Leave();
+
+ VideoPacket* newPacket = new VideoPacket();
+ memcpy(newPacket->packetBuffer, data, len);
+ newPacket->length = len;
+ newPacket->channel = channel;
+
+ _crit.Enter();
+ newPacket->receiveTime = NowMs() + _networkDelayMs;
+ _rtcpPackets.push_back(newPacket);
+ _event.Set();
+ _crit.Leave();
+ return len;
+}
+
+WebRtc_Word32 TbExternalTransport::SetPacketLoss(WebRtc_Word32 lossRate)
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ _lossRate = lossRate;
+ return 0;
+}
+
+void TbExternalTransport::SetNetworkDelay(WebRtc_Word64 delayMs)
+{
+ webrtc::CriticalSectionScoped cs(_crit);
+ _networkDelayMs = delayMs;
+}
+
+void TbExternalTransport::SetSSRCFilter(WebRtc_UWord32 ssrc)
+{
+ webrtc::CriticalSectionScoped cs(_crit);
+ _filterSSRC = true;
+ _SSRC = ssrc;
+}
+
+void TbExternalTransport::ClearStats()
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ _rtpCount = 0;
+ _dropCount = 0;
+ _rtcpCount = 0;
+}
+
+void TbExternalTransport::GetStats(WebRtc_Word32& numRtpPackets,
+ WebRtc_Word32& numDroppedPackets,
+ WebRtc_Word32& numRtcpPackets)
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ numRtpPackets = _rtpCount;
+ numDroppedPackets = _dropCount;
+ numRtcpPackets = _rtcpCount;
+}
+
+void TbExternalTransport::EnableSSRCCheck()
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ _checkSSRC = true;
+}
+
+unsigned int TbExternalTransport::ReceivedSSRC()
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ return _lastSSRC;
+}
+
+void TbExternalTransport::EnableSequenceNumberCheck()
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ _checkSequenceNumber = true;
+}
+
+unsigned short TbExternalTransport::GetFirstSequenceNumber()
+{
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ return _firstSequenceNumber;
+}
+
+bool TbExternalTransport::ViEExternalTransportRun(void* object)
+{
+ return static_cast<TbExternalTransport*>
+ (object)->ViEExternalTransportProcess();
+}
+bool TbExternalTransport::ViEExternalTransportProcess()
+{
+ unsigned int waitTime = KMaxWaitTimeMs;
+
+ VideoPacket* packet = NULL;
+
+ while (!_rtpPackets.empty())
+ {
+ // Take first packet in queue
+ _crit.Enter();
+ packet = _rtpPackets.front();
+ WebRtc_Word64 timeToReceive = 0;
+ if (packet)
+ {
+ timeToReceive = packet->receiveTime - NowMs();
+ }
+ else
+ {
+ // There should never be any empty packets in the list.
+ assert(false);
+ }
+ if (timeToReceive > 0)
+ {
+ // No packets to receive yet
+ if (timeToReceive < waitTime && timeToReceive > 0)
+ {
+ waitTime = (unsigned int) timeToReceive;
+ }
+ _crit.Leave();
+ break;
+ }
+ _rtpPackets.pop_front();
+ _crit.Leave();
+
+ // Send to ViE
+ if (packet)
+ {
+ {
+ webrtc::CriticalSectionScoped cs(_statCrit);
+ if (_checkSSRC)
+ {
+ _lastSSRC = ((packet->packetBuffer[8]) << 24);
+ _lastSSRC += (packet->packetBuffer[9] << 16);
+ _lastSSRC += (packet->packetBuffer[10] << 8);
+ _lastSSRC += packet->packetBuffer[11];
+ _checkSSRC = false;
+ }
+ if (_checkSequenceNumber)
+ {
+ _firstSequenceNumber
+ = (unsigned char) packet->packetBuffer[2] << 8;
+ _firstSequenceNumber
+ += (unsigned char) packet->packetBuffer[3];
+ _checkSequenceNumber = false;
+ }
+ }
+ // Signal received packet of frame
+ WebRtc_UWord8* ptr = (WebRtc_UWord8*)packet->packetBuffer;
+ WebRtc_UWord32 rtp_timestamp = ptr[4] << 24;
+ rtp_timestamp += ptr[5] << 16;
+ rtp_timestamp += ptr[6] << 8;
+ rtp_timestamp += ptr[7];
+ if (_receive_frame_callback != NULL &&
+ _lastReceiveRTPTimestamp != rtp_timestamp) {
+ _receive_frame_callback->FrameReceived(rtp_timestamp);
+ }
+ _lastReceiveRTPTimestamp = rtp_timestamp;
+
+ _vieNetwork.ReceivedRTPPacket(packet->channel,
+ packet->packetBuffer, packet->length);
+ delete packet;
+ packet = NULL;
+ }
+ }
+ while (!_rtcpPackets.empty())
+ {
+ // Take first packet in queue
+ _crit.Enter();
+ packet = _rtcpPackets.front();
+ WebRtc_Word64 timeToReceive = 0;
+ if (packet)
+ {
+ timeToReceive = packet->receiveTime - NowMs();
+ }
+ else
+ {
+ // There should never be any empty packets in the list.
+ assert(false);
+ }
+ if (timeToReceive > 0)
+ {
+ // No packets to receive yet
+ if (timeToReceive < waitTime && timeToReceive > 0)
+ {
+ waitTime = (unsigned int) timeToReceive;
+ }
+ _crit.Leave();
+ break;
+ }
+ _rtcpPackets.pop_front();
+ _crit.Leave();
+
+ // Send to ViE
+ if (packet)
+ {
+ _vieNetwork.ReceivedRTCPPacket(
+ packet->channel,
+ packet->packetBuffer, packet->length);
+ delete packet;
+ packet = NULL;
+ }
+ }
+ _event.Wait(waitTime + 1); // Add 1 ms to not call to early...
+ return true;
+}
+
+WebRtc_Word64 TbExternalTransport::NowMs()
+{
+ return webrtc::TickTime::MillisecondTimestamp();
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc b/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc
new file mode 100644
index 0000000..b6b13a2
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tb_interfaces.h"
+
+#include "gtest/gtest.h"
+
+TbInterfaces::TbInterfaces(const char* test_name) {
+ std::string trace_file_path =
+ (ViETest::GetResultOutputPath() + test_name) + "_trace.txt";
+
+ ViETest::Log("Creating ViE Interfaces for test %s\n", test_name);
+
+ video_engine = webrtc::VideoEngine::Create();
+ EXPECT_TRUE(video_engine != NULL);
+
+ EXPECT_EQ(0, video_engine->SetTraceFile(trace_file_path.c_str()));
+ EXPECT_EQ(0, video_engine->SetTraceFilter(webrtc::kTraceAll));
+
+ base = webrtc::ViEBase::GetInterface(video_engine);
+ EXPECT_TRUE(base != NULL);
+
+ EXPECT_EQ(0, base->Init());
+
+ capture = webrtc::ViECapture::GetInterface(video_engine);
+ EXPECT_TRUE(capture != NULL);
+
+ rtp_rtcp = webrtc::ViERTP_RTCP::GetInterface(video_engine);
+ EXPECT_TRUE(rtp_rtcp != NULL);
+
+ render = webrtc::ViERender::GetInterface(video_engine);
+ EXPECT_TRUE(render != NULL);
+
+ codec = webrtc::ViECodec::GetInterface(video_engine);
+ EXPECT_TRUE(codec != NULL);
+
+ network = webrtc::ViENetwork::GetInterface(video_engine);
+ EXPECT_TRUE(network != NULL);
+
+ image_process = webrtc::ViEImageProcess::GetInterface(video_engine);
+ EXPECT_TRUE(image_process != NULL);
+
+ encryption = webrtc::ViEEncryption::GetInterface(video_engine);
+ EXPECT_TRUE(encryption != NULL);
+}
+
+TbInterfaces::~TbInterfaces(void)
+{
+ EXPECT_EQ(0, encryption->Release());
+ EXPECT_EQ(0, image_process->Release());
+ EXPECT_EQ(0, codec->Release());
+ EXPECT_EQ(0, capture->Release());
+ EXPECT_EQ(0, render->Release());
+ EXPECT_EQ(0, rtp_rtcp->Release());
+ EXPECT_EQ(0, network->Release());
+ EXPECT_EQ(0, base->Release());
+ EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine)) <<
+ "Since we have released all interfaces at this point, deletion "
+ "should be successful.";
+
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc b/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc
new file mode 100644
index 0000000..1e62a6b
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tb_video_channel.h"
+
+TbVideoChannel::TbVideoChannel(TbInterfaces& Engine,
+ webrtc::VideoCodecType sendCodec, int width,
+ int height, int frameRate, int startBitrate) :
+ videoChannel(-1), ViE(Engine)
+{
+ EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel));
+
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ bool sendCodecSet = false;
+ for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
+ videoCodec.width = width;
+ videoCodec.height = height;
+ videoCodec.maxFramerate = frameRate;
+
+ if (videoCodec.codecType == sendCodec && sendCodecSet == false)
+ {
+ if(videoCodec.codecType != webrtc::kVideoCodecI420 )
+ {
+ videoCodec.startBitrate = startBitrate;
+ videoCodec.maxBitrate = startBitrate * 3;
+ }
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
+ sendCodecSet = true;
+ }
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8)
+ {
+ videoCodec.width = 352;
+ videoCodec.height = 288;
+ }
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
+ }
+ EXPECT_TRUE(sendCodecSet);
+}
+
+TbVideoChannel::~TbVideoChannel(void)
+{
+ EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel));
+}
+
+void TbVideoChannel::StartSend(const unsigned short rtpPort /*= 11000*/,
+ const char* ipAddress /*= "127.0.0.1"*/)
+{
+ EXPECT_EQ(0, ViE.network->SetSendDestination(videoChannel, ipAddress,
+ rtpPort));
+
+ EXPECT_EQ(0, ViE.base->StartSend(videoChannel));
+}
+
+void TbVideoChannel::SetFrameSettings(int width, int height, int frameRate)
+{
+ webrtc::VideoCodec videoCodec;
+ EXPECT_EQ(0, ViE.codec->GetSendCodec(videoChannel, videoCodec));
+ videoCodec.width = width;
+ videoCodec.height = height;
+ videoCodec.maxFramerate = frameRate;
+
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
+}
+
+void TbVideoChannel::StopSend()
+{
+ EXPECT_EQ(0, ViE.base->StopSend(videoChannel));
+}
+
+void TbVideoChannel::StartReceive(const unsigned short rtpPort /*= 11000*/)
+{
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(videoChannel, rtpPort));
+ EXPECT_EQ(0, ViE.base->StartReceive(videoChannel));
+}
+
+void TbVideoChannel::StopReceive()
+{
+ EXPECT_EQ(0, ViE.base->StopReceive(videoChannel));
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc
new file mode 100644
index 0000000..8281ea2
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc
@@ -0,0 +1,159 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest.cc
+//
+
+#include "vie_autotest.h"
+
+#include <stdio.h>
+
+#include "engine_configurations.h"
+#include "general_primitives.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "tb_capture_device.h"
+#include "testsupport/fileutils.h"
+#include "video_render.h"
+#include "vie_autotest_defines.h"
+
+// ViETest implementation
+FILE* ViETest::log_file_ = NULL;
+char* ViETest::log_str_ = NULL;
+
+std::string ViETest::GetResultOutputPath() {
+#ifdef WEBRTC_ANDROID
+ return "/sdcard/";
+#else
+ return webrtc::test::OutputPath();
+#endif
+}
+
+// ViEAutoTest implementation
+ViEAutoTest::ViEAutoTest(void* window1, void* window2) :
+ _window1(window1),
+ _window2(window2),
+ _renderType(webrtc::kRenderDefault),
+ _vrm1(webrtc::VideoRender::CreateVideoRender(
+ 4561, window1, false, _renderType)),
+ _vrm2(webrtc::VideoRender::CreateVideoRender(
+ 4562, window2, false, _renderType))
+{
+ assert(_vrm1);
+ assert(_vrm2);
+}
+
+ViEAutoTest::~ViEAutoTest()
+{
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ webrtc::VideoRender::DestroyVideoRender(_vrm2);
+ _vrm2 = NULL;
+}
+
+void ViEAutoTest::ViEStandardTest()
+{
+ ViEBaseStandardTest();
+ ViECaptureStandardTest();
+ ViECodecStandardTest();
+ ViEEncryptionStandardTest();
+ ViEFileStandardTest();
+ ViEImageProcessStandardTest();
+ ViENetworkStandardTest();
+ ViERenderStandardTest();
+ ViERtpRtcpStandardTest();
+}
+
+void ViEAutoTest::ViEExtendedTest()
+{
+ ViEBaseExtendedTest();
+ ViECaptureExtendedTest();
+ ViECodecExtendedTest();
+ ViEEncryptionExtendedTest();
+ ViEFileExtendedTest();
+ ViEImageProcessExtendedTest();
+ ViENetworkExtendedTest();
+ ViERenderExtendedTest();
+ ViERtpRtcpExtendedTest();
+}
+
+void ViEAutoTest::ViEAPITest()
+{
+ ViEBaseAPITest();
+ ViECaptureAPITest();
+ ViECodecAPITest();
+ ViEEncryptionAPITest();
+ ViEFileAPITest();
+ ViEImageProcessAPITest();
+ ViENetworkAPITest();
+ ViERenderAPITest();
+ ViERtpRtcpAPITest();
+}
+
+void ViEAutoTest::PrintVideoCodec(const webrtc::VideoCodec videoCodec)
+{
+ ViETest::Log("Video Codec Information:");
+
+ switch (videoCodec.codecType)
+ {
+ case webrtc::kVideoCodecVP8:
+ ViETest::Log("\tcodecType: VP8");
+ break;
+ // TODO(sh): keep or remove MPEG4?
+ // case webrtc::kVideoCodecMPEG4:
+ // ViETest::Log("\tcodecType: MPEG4");
+ // break;
+ case webrtc::kVideoCodecI420:
+ ViETest::Log("\tcodecType: I420");
+ break;
+ case webrtc::kVideoCodecRED:
+ ViETest::Log("\tcodecType: RED");
+ break;
+ case webrtc::kVideoCodecULPFEC:
+ ViETest::Log("\tcodecType: ULPFEC");
+ break;
+ case webrtc::kVideoCodecUnknown:
+ ViETest::Log("\tcodecType: ????");
+ break;
+ }
+
+ ViETest::Log("\theight: %u", videoCodec.height);
+ ViETest::Log("\tmaxBitrate: %u", videoCodec.maxBitrate);
+ ViETest::Log("\tmaxFramerate: %u", videoCodec.maxFramerate);
+ ViETest::Log("\tminBitrate: %u", videoCodec.minBitrate);
+ ViETest::Log("\tplName: %s", videoCodec.plName);
+ ViETest::Log("\tplType: %u", videoCodec.plType);
+ ViETest::Log("\tstartBitrate: %u", videoCodec.startBitrate);
+ ViETest::Log("\twidth: %u", videoCodec.width);
+ ViETest::Log("");
+}
+
+void ViEAutoTest::PrintAudioCodec(const webrtc::CodecInst audioCodec)
+{
+ ViETest::Log("Audio Codec Information:");
+ ViETest::Log("\tchannels: %u", audioCodec.channels);
+ ViETest::Log("\t: %u", audioCodec.pacsize);
+ ViETest::Log("\t: %u", audioCodec.plfreq);
+ ViETest::Log("\t: %s", audioCodec.plname);
+ ViETest::Log("\t: %u", audioCodec.pltype);
+ ViETest::Log("\t: %u", audioCodec.rate);
+ ViETest::Log("");
+}
+
+void ViEAutoTest::RenderCaptureDeviceAndOutputStream(
+ TbInterfaces* video_engine,
+ TbVideoChannel* video_channel,
+ TbCaptureDevice* capture_device) {
+ RenderInWindow(
+ video_engine->render, capture_device->captureId, _window1, 0);
+ RenderInWindow(
+ video_engine->render, video_channel->videoChannel, _window1, 1);
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc
new file mode 100644
index 0000000..81af69b
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc
@@ -0,0 +1,209 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../interface/vie_autotest_android.h"
+
+#include <android/log.h>
+#include <stdio.h>
+
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+
+int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
+ void* window1, void* window2,
+ void* javaVM, void* env, void* context) {
+ ViEAutoTest vieAutoTest(window1, window2);
+ ViETest::Log("RunAutoTest(%d, %d)", testSelection, subTestSelection);
+ webrtc::VideoEngine::SetAndroidObjects(javaVM, context);
+#ifndef WEBRTC_ANDROID_OPENSLES
+ // voice engine calls into ADM directly
+ webrtc::VoiceEngine::SetAndroidAudioDeviceObjects(javaVM, env, context);
+#endif
+ int testErrors = 0;
+
+ if (subTestSelection == 0) {
+ // Run all selected test
+ switch (testSelection) {
+ case 0:
+ testErrors += vieAutoTest.ViEStandardTest();
+ if (testErrors == 0) {
+ // No errors found in delivery test, create delivery
+ ViETest::Log("Standard/delivery passed. ");
+ }
+ else {
+ // Didn't pass
+ ViETest::Log("\nStandard/delivery test failed.");
+ }
+ break;
+ case 1:
+ testErrors += vieAutoTest.ViEAPITest();
+ break;
+ case 2:
+ testErrors += vieAutoTest.ViEExtendedTest();
+ break;
+ case 3:
+ testErrors += vieAutoTest.ViELoopbackCall();
+ break;
+ default:
+ break;
+ }
+ }
+
+ switch (testSelection) {
+ case 0: // Specific standard test
+ switch (subTestSelection) {
+ case 1: // base
+ testErrors += vieAutoTest.ViEBaseStandardTest();
+ break;
+
+ case 2: // capture
+ testErrors += vieAutoTest.ViECaptureStandardTest();
+ break;
+
+ case 3: // codec
+ testErrors += vieAutoTest.ViECodecStandardTest();
+ break;
+
+ case 5: //encryption
+ testErrors += vieAutoTest.ViEEncryptionStandardTest();
+ break;
+
+ case 6: // file
+ testErrors += vieAutoTest.ViEFileStandardTest();
+ break;
+
+ case 7: // image process
+ testErrors += vieAutoTest.ViEImageProcessStandardTest();
+ break;
+
+ case 8: // network
+ testErrors += vieAutoTest.ViENetworkStandardTest();
+ break;
+
+ case 9: // Render
+ testErrors += vieAutoTest.ViERenderStandardTest();
+ break;
+
+ case 10: // RTP/RTCP
+ testErrors += vieAutoTest.ViERtpRtcpStandardTest();
+ break;
+
+ default:
+ break;
+ }
+ break;
+
+ case 1:// specific API
+ switch (subTestSelection) {
+ case 1: // base
+ testErrors += vieAutoTest.ViEBaseAPITest();
+ break;
+
+ case 2: // capture
+ testErrors += vieAutoTest.ViECaptureAPITest();
+ break;
+
+ case 3: // codec
+ testErrors += vieAutoTest.ViECodecAPITest();
+ break;
+
+ case 5: //encryption
+ testErrors += vieAutoTest.ViEEncryptionAPITest();
+ break;
+
+ case 6: // file
+ testErrors += vieAutoTest.ViEFileAPITest();
+ break;
+
+ case 7: // image process
+ testErrors += vieAutoTest.ViEImageProcessAPITest();
+ break;
+
+ case 8: // network
+ testErrors += vieAutoTest.ViENetworkAPITest();
+ break;
+
+ case 9: // Render
+ testErrors += vieAutoTest.ViERenderAPITest();
+ break;
+
+ case 10: // RTP/RTCP
+ testErrors += vieAutoTest.ViERtpRtcpAPITest();
+ break;
+ case 11:
+ break;
+
+ default:
+ break;
+ }
+ break;
+
+ case 2:// specific extended
+ switch (subTestSelection) {
+ case 1: // base
+ testErrors += vieAutoTest.ViEBaseExtendedTest();
+ break;
+
+ case 2: // capture
+ testErrors += vieAutoTest.ViECaptureExtendedTest();
+ break;
+
+ case 3: // codec
+ testErrors += vieAutoTest.ViECodecExtendedTest();
+ break;
+
+ case 5: //encryption
+ testErrors += vieAutoTest.ViEEncryptionExtendedTest();
+ break;
+
+ case 6: // file
+ testErrors += vieAutoTest.ViEFileExtendedTest();
+ break;
+
+ case 7: // image process
+ testErrors += vieAutoTest.ViEImageProcessExtendedTest();
+ break;
+
+ case 8: // network
+ testErrors += vieAutoTest.ViENetworkExtendedTest();
+ break;
+
+ case 9: // Render
+ testErrors += vieAutoTest.ViERenderExtendedTest();
+ break;
+
+ case 10: // RTP/RTCP
+ testErrors += vieAutoTest.ViERtpRtcpExtendedTest();
+ break;
+
+ case 11:
+ break;
+
+ default:
+ break;
+ }
+ break;
+
+ case 3:
+ testErrors += vieAutoTest.ViELoopbackCall();
+ break;
+
+ default:
+ break;
+ }
+
+ if (testErrors) {
+ ViETest::Log("Test done with %d errors!\n", testErrors);
+ }
+ else {
+ ViETest::Log("Test passed!\n");
+ }
+ return testErrors;
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc
new file mode 100644
index 0000000..d2a29b2
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc
@@ -0,0 +1,208 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_autotest.h"
+
+#include "base_primitives.h"
+#include "general_primitives.h"
+#include "tb_interfaces.h"
+#include "vie_autotest_defines.h"
+#include "video_capture_factory.h"
+
+class BaseObserver : public webrtc::ViEBaseObserver {
+ public:
+ BaseObserver()
+ : cpu_load_(0) {}
+
+ virtual void PerformanceAlarm(const unsigned int cpu_load) {
+ cpu_load_ = cpu_load;
+ }
+ unsigned int cpu_load_;
+};
+
+void ViEAutoTest::ViEBaseStandardTest() {
+ // ***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ // ***************************************************************
+
+ TbInterfaces interfaces("ViEBaseStandardTest");
+
+ // ***************************************************************
+ // Engine ready. Set up the test case:
+ // ***************************************************************
+ int video_channel = -1;
+ EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
+
+ webrtc::VideoCaptureModule* video_capture_module(NULL);
+ const unsigned int kMaxDeviceNameLength = 128;
+ WebRtc_UWord8 device_name[kMaxDeviceNameLength];
+ memset(device_name, 0, kMaxDeviceNameLength);
+ int capture_id;
+
+ webrtc::ViEBase *base_interface = interfaces.base;
+ webrtc::ViERender *render_interface = interfaces.render;
+ webrtc::ViECapture *capture_interface = interfaces.capture;
+
+ FindCaptureDeviceOnSystem(capture_interface,
+ device_name,
+ kMaxDeviceNameLength,
+ &capture_id,
+ &video_capture_module);
+
+ EXPECT_EQ(0, capture_interface->ConnectCaptureDevice(capture_id,
+ video_channel));
+ EXPECT_EQ(0, capture_interface->StartCapture(capture_id));
+
+ ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
+
+ EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm2));
+
+ RenderInWindow(render_interface, capture_id, _window1, 0);
+ RenderInWindow(render_interface, video_channel, _window2, 1);
+
+ // ***************************************************************
+ // Run the actual test:
+ // ***************************************************************
+ ViETest::Log("You should shortly see a local preview from camera %s"
+ " in window 1 and the remote video in window 2.", device_name);
+ ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
+ base_interface, interfaces.network, video_channel,
+ device_name);
+
+ // ***************************************************************
+ // Testing finished. Tear down Video Engine
+ // ***************************************************************
+ EXPECT_EQ(0, capture_interface->StopCapture(capture_id));
+ EXPECT_EQ(0, base_interface->StopReceive(video_channel));
+
+ StopAndRemoveRenderers(base_interface, render_interface, video_channel,
+ capture_id);
+
+ EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm2));
+
+ EXPECT_EQ(0, capture_interface->ReleaseCaptureDevice(capture_id));
+
+ video_capture_module->Release();
+ video_capture_module = NULL;
+
+ EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
+}
+
+void ViEAutoTest::ViEBaseExtendedTest() {
+ // Start with standard test
+ ViEBaseAPITest();
+ ViEBaseStandardTest();
+
+ // ***************************************************************
+ // Test BaseObserver
+ // ***************************************************************
+ // TODO(mflodman) Add test for base observer. Cpu load must be over 75%.
+// BaseObserver base_observer;
+// EXPECT_EQ(ptrViEBase->RegisterObserver(base_observer), 0);
+//
+// AutoTestSleep(KAutoTestSleepTimeMs);
+//
+// EXPECT_EQ(ptrViEBase->DeregisterObserver(), 0);
+// EXPECT_GT(base_observer.cpu_load, 0);
+}
+
+void ViEAutoTest::ViEBaseAPITest() {
+ // ***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ // ***************************************************************
+ // Get the ViEBase API
+ webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(NULL);
+ EXPECT_EQ(NULL, ptrViEBase) << "Should return null for a bad ViE pointer";
+
+ webrtc::VideoEngine* ptrViE = webrtc::VideoEngine::Create();
+ EXPECT_TRUE(NULL != ptrViE);
+
+ std::string trace_file_path =
+ ViETest::GetResultOutputPath() + "ViEBaseAPI_trace.txt";
+ EXPECT_EQ(0, ptrViE->SetTraceFile(trace_file_path.c_str()));
+
+ ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
+ EXPECT_TRUE(NULL != ptrViEBase);
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+ char version[1024] = "";
+ EXPECT_EQ(0, ptrViEBase->GetVersion(version));
+ EXPECT_EQ(0, ptrViEBase->LastError());
+
+ // Create without init
+ int videoChannel = -1;
+ EXPECT_NE(0, ptrViEBase->CreateChannel(videoChannel)) <<
+ "Should fail since Init has not been called yet";
+ EXPECT_EQ(0, ptrViEBase->Init());
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
+
+ int videoChannel2 = -1;
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel2));
+ EXPECT_NE(videoChannel, videoChannel2) <<
+ "Should allocate new number for independent channel";
+
+ EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel2));
+
+ EXPECT_EQ(-1, ptrViEBase->CreateChannel(videoChannel2, videoChannel + 1)) <<
+ "Should fail since neither channel exists (the second must)";
+
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel2, videoChannel));
+
+ // Test Voice Engine integration with Video Engine.
+ webrtc::VoiceEngine* ptrVoE = NULL;
+ webrtc::VoEBase* ptrVoEBase = NULL;
+ int audioChannel = -1;
+
+ ptrVoE = webrtc::VoiceEngine::Create();
+ EXPECT_TRUE(NULL != ptrVoE);
+
+ ptrVoEBase = webrtc::VoEBase::GetInterface(ptrVoE);
+ EXPECT_TRUE(NULL != ptrVoEBase);
+ EXPECT_EQ(0, ptrVoEBase->Init());
+
+ audioChannel = ptrVoEBase->CreateChannel();
+ EXPECT_NE(-1, audioChannel);
+
+ // Connect before setting VoE.
+ EXPECT_NE(0, ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel)) <<
+ "Should fail since Voice Engine is not set yet.";
+
+ // Then do it right.
+ EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(ptrVoE));
+ EXPECT_EQ(0, ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel));
+
+ // ***************************************************************
+ // Testing finished. Tear down Video Engine
+ // ***************************************************************
+ EXPECT_NE(0, ptrViEBase->DisconnectAudioChannel(videoChannel + 5)) <<
+ "Should fail: disconnecting bogus channel";
+
+ EXPECT_EQ(0, ptrViEBase->DisconnectAudioChannel(videoChannel));
+
+ // Clean up voice engine
+ EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(NULL));
+ EXPECT_EQ(0, ptrVoEBase->Release());
+ EXPECT_TRUE(webrtc::VoiceEngine::Delete(ptrVoE));
+
+ webrtc::ViEBase* ptrViEBase2 = webrtc::ViEBase::GetInterface(ptrViE);
+ EXPECT_TRUE(NULL != ptrViEBase2);
+
+ EXPECT_EQ(1, ptrViEBase->Release()) << "There should be one interface left.";
+
+ EXPECT_FALSE(webrtc::VideoEngine::Delete(ptrViE)) <<
+ "Should fail since there are interfaces left.";
+
+ EXPECT_EQ(0, ptrViEBase->Release());
+ EXPECT_TRUE(webrtc::VideoEngine::Delete(ptrViE));
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc
new file mode 100644
index 0000000..cbb8b03
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc
@@ -0,0 +1,559 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_autotest.h"
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "gflags/gflags.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "tick_util.h"
+#include "vie_autotest_defines.h"
+#include "video_capture_factory.h"
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+#include "voe_base.h"
+
+DEFINE_bool(capture_test_ensure_resolution_alignment_in_capture_device, true,
+ "If true, we will give resolutions slightly below a reasonable "
+ "value to test the camera's ability to choose a good resolution. "
+ "If false, we will provide reasonable resolutions instead.");
+
+class CaptureObserver: public webrtc::ViECaptureObserver
+{
+public:
+ CaptureObserver() :
+ _brightness(webrtc::Normal),
+ _alarm(webrtc::AlarmCleared),
+ _frameRate(0) {}
+
+ virtual void BrightnessAlarm(const int captureId,
+ const webrtc::Brightness brightness)
+ {
+ _brightness = brightness;
+ switch (brightness)
+ {
+ case webrtc::Normal:
+ ViETest::Log(" BrightnessAlarm Normal");
+ break;
+ case webrtc::Bright:
+ ViETest::Log(" BrightnessAlarm Bright");
+ break;
+ case webrtc::Dark:
+ ViETest::Log(" BrightnessAlarm Dark");
+ break;
+ }
+ }
+
+ virtual void CapturedFrameRate(const int captureId,
+ const unsigned char frameRate)
+ {
+ ViETest::Log(" CapturedFrameRate %u", frameRate);
+ _frameRate = frameRate;
+ }
+
+ virtual void NoPictureAlarm(const int captureId,
+ const webrtc::CaptureAlarm alarm)
+ {
+ _alarm = alarm;
+ if (alarm == webrtc::AlarmRaised)
+ {
+ ViETest::Log("NoPictureAlarm CARaised.");
+ }
+ else
+ {
+ ViETest::Log("NoPictureAlarm CACleared.");
+ }
+ }
+
+ webrtc::Brightness _brightness;
+ webrtc::CaptureAlarm _alarm;
+ unsigned char _frameRate;
+};
+
+class CaptureEffectFilter: public webrtc::ViEEffectFilter {
+ public:
+ CaptureEffectFilter(unsigned int expected_width, unsigned int expected_height)
+ : number_of_captured_frames_(0),
+ expected_width_(expected_width),
+ expected_height_(expected_height) {
+ }
+
+ // Implements ViEEffectFilter
+ virtual int Transform(int size, unsigned char* frameBuffer,
+ unsigned int timeStamp90KHz, unsigned int width,
+ unsigned int height) {
+ EXPECT_TRUE(frameBuffer != NULL);
+ EXPECT_EQ(expected_width_, width);
+ EXPECT_EQ(expected_height_, height);
+ ++number_of_captured_frames_;
+ return 0;
+ }
+
+ int number_of_captured_frames_;
+
+ protected:
+ unsigned int expected_width_;
+ unsigned int expected_height_;
+ };
+
+void ViEAutoTest::ViECaptureStandardTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ TbInterfaces ViE("ViECaptureStandardTest");
+
+ webrtc::VideoCaptureModule::DeviceInfo* devInfo =
+ webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
+
+ int numberOfCaptureDevices = devInfo->NumberOfDevices();
+ ViETest::Log("Number of capture devices %d", numberOfCaptureDevices);
+ EXPECT_GT(numberOfCaptureDevices, 0);
+
+ int captureDeviceId[10];
+ memset(captureDeviceId, 0, sizeof(captureDeviceId));
+ webrtc::VideoCaptureModule* vcpms[10];
+ memset(vcpms, 0, sizeof(vcpms));
+
+ // Check capabilities
+ for (int deviceIndex = 0;
+ deviceIndex < numberOfCaptureDevices;
+ ++deviceIndex)
+ {
+ WebRtc_UWord8 deviceName[128];
+ WebRtc_UWord8 deviceUniqueName[512];
+
+ EXPECT_EQ(0, devInfo->GetDeviceName(deviceIndex,
+ deviceName,
+ sizeof(deviceName),
+ deviceUniqueName,
+ sizeof(deviceUniqueName)));
+ ViETest::Log("Found capture device %s\nUnique name %s", deviceName,
+ deviceUniqueName);
+
+#if !defined(WEBRTC_MAC_INTEL) // these functions will return -1
+ int numberOfCapabilities =
+ devInfo->NumberOfCapabilities(deviceUniqueName);
+ EXPECT_GT(numberOfCapabilities, 0);
+
+ for (int capIndex = 0; capIndex < numberOfCapabilities; ++capIndex)
+ {
+ webrtc::VideoCaptureCapability capability;
+ EXPECT_EQ(0, devInfo->GetCapability(deviceUniqueName, capIndex,
+ capability));
+ ViETest::Log("Capture capability %d (of %u)", capIndex + 1,
+ numberOfCapabilities);
+ ViETest::Log("witdh %d, height %d, frame rate %d",
+ capability.width, capability.height, capability.maxFPS);
+ ViETest::Log("expected delay %d, color type %d, encoding %d",
+ capability.expectedCaptureDelay, capability.rawType,
+ capability.codecType);
+
+ EXPECT_GT(capability.width, 0);
+ EXPECT_GT(capability.height, 0);
+ EXPECT_GT(capability.maxFPS, -1); // >= 0
+ EXPECT_GT(capability.expectedCaptureDelay, 0);
+ }
+#endif
+ }
+ // Capture Capability Functions are not supported on WEBRTC_MAC_INTEL.
+#if !defined(WEBRTC_MAC_INTEL)
+
+ // Check allocation. Try to allocate them all after each other.
+ for (int deviceIndex = 0;
+ deviceIndex < numberOfCaptureDevices;
+ ++deviceIndex)
+ {
+ WebRtc_UWord8 deviceName[128];
+ WebRtc_UWord8 deviceUniqueName[512];
+
+ EXPECT_EQ(0, devInfo->GetDeviceName(deviceIndex,
+ deviceName,
+ sizeof(deviceName),
+ deviceUniqueName,
+ sizeof(deviceUniqueName)));
+
+ webrtc::VideoCaptureModule* vcpm =
+ webrtc::VideoCaptureFactory::Create(
+ deviceIndex, deviceUniqueName);
+ EXPECT_TRUE(vcpm != NULL);
+ vcpm->AddRef();
+ vcpms[deviceIndex] = vcpm;
+
+ EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(
+ *vcpm, captureDeviceId[deviceIndex]));
+
+ webrtc::VideoCaptureCapability capability;
+ EXPECT_EQ(0, devInfo->GetCapability(deviceUniqueName, 0, capability));
+
+ // Test that the camera select the closest capability to the selected
+ // width and height.
+ CaptureEffectFilter filter(capability.width, capability.height);
+ EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
+ captureDeviceId[deviceIndex], filter));
+
+ ViETest::Log("Testing Device %s capability width %d height %d",
+ deviceUniqueName, capability.width, capability.height);
+
+ if (FLAGS_capture_test_ensure_resolution_alignment_in_capture_device) {
+ // This tests that the capture device properly aligns to a
+ // multiple of 16 (or at least 8).
+ capability.height = capability.height - 2;
+ capability.width = capability.width - 2;
+ }
+
+ webrtc::CaptureCapability vieCapability;
+ vieCapability.width = capability.width;
+ vieCapability.height = capability.height;
+ vieCapability.codecType = capability.codecType;
+ vieCapability.maxFPS = capability.maxFPS;
+ vieCapability.rawType = capability.rawType;
+
+ EXPECT_EQ(0, ViE.capture->StartCapture(captureDeviceId[deviceIndex],
+ vieCapability));
+ webrtc::TickTime startTime = webrtc::TickTime::Now();
+
+ while (filter.number_of_captured_frames_ < 10
+ && (webrtc::TickTime::Now() - startTime).Milliseconds() < 10000)
+ {
+ AutoTestSleep(100);
+ }
+
+ EXPECT_GT(filter.number_of_captured_frames_, 9) <<
+ "Should capture at least some frames";
+
+ EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
+ captureDeviceId[deviceIndex]));
+
+#ifdef WEBRTC_ANDROID // Can only allocate one camera at the time on Android.
+ EXPECT_EQ(0, ViE.capture->StopCapture(captureDeviceId[deviceIndex]));
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(
+ captureDeviceId[deviceIndex]));
+#endif
+ }
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+
+
+ // Stop all started capture devices.
+ for (int deviceIndex = 0;
+ deviceIndex < numberOfCaptureDevices;
+ ++deviceIndex) {
+#if !defined(WEBRTC_ANDROID)
+ // Don't stop on Android since we can only allocate one camera.
+ EXPECT_EQ(0, ViE.capture->StopCapture(
+ captureDeviceId[deviceIndex]));
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(
+ captureDeviceId[deviceIndex]));
+#endif // !WEBRTC_ANDROID
+ vcpms[deviceIndex]->Release();
+ }
+#endif // !WEBRTC_MAC_INTEL
+}
+
+void ViEAutoTest::ViECaptureExtendedTest() {
+ ViECaptureStandardTest();
+ ViECaptureAPITest();
+ ViECaptureExternalCaptureTest();
+}
+
+void ViEAutoTest::ViECaptureAPITest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+ TbInterfaces ViE("ViECaptureAPITest");
+
+ ViE.capture->NumberOfCaptureDevices();
+
+ WebRtc_UWord8 deviceName[128];
+ WebRtc_UWord8 deviceUniqueName[512];
+ int captureId = 0;
+
+ webrtc::VideoCaptureModule::DeviceInfo* devInfo =
+ webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
+ EXPECT_TRUE(devInfo != NULL);
+
+ // Get the first capture device
+ EXPECT_EQ(0, devInfo->GetDeviceName(0, deviceName,
+ sizeof(deviceName),
+ deviceUniqueName,
+ sizeof(deviceUniqueName)));
+
+ webrtc::VideoCaptureModule* vcpm =
+ webrtc::VideoCaptureFactory::Create(0, deviceUniqueName);
+ vcpm->AddRef();
+ EXPECT_TRUE(vcpm != NULL);
+
+ // Allocate capture device.
+ EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
+
+ // Start the capture device.
+ EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
+
+ // Start again. Should fail.
+ EXPECT_NE(0, ViE.capture->StartCapture(captureId));
+ EXPECT_EQ(kViECaptureDeviceAlreadyStarted, ViE.LastError());
+
+ // Start invalid capture device.
+ EXPECT_NE(0, ViE.capture->StartCapture(captureId + 1));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Stop invalid capture device.
+ EXPECT_NE(0, ViE.capture->StopCapture(captureId + 1));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Stop the capture device.
+ EXPECT_EQ(0, ViE.capture->StopCapture(captureId));
+
+ // Stop the capture device again.
+ EXPECT_NE(0, ViE.capture->StopCapture(captureId));
+ EXPECT_EQ(kViECaptureDeviceNotStarted, ViE.LastError());
+
+ // Connect to invalid channel.
+ EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId, 0));
+ EXPECT_EQ(kViECaptureDeviceInvalidChannelId, ViE.LastError());
+
+ TbVideoChannel channel(ViE);
+
+ // Connect invalid captureId.
+ EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId + 1,
+ channel.videoChannel));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Connect the capture device to the channel.
+ EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId,
+ channel.videoChannel));
+
+ // Connect the channel again.
+ EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId,
+ channel.videoChannel));
+ EXPECT_EQ(kViECaptureDeviceAlreadyConnected, ViE.LastError());
+
+ // Start the capture device.
+ EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
+
+ // Release invalid capture device.
+ EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId + 1));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Release the capture device.
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
+
+ // Release the capture device again.
+ EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Test GetOrientation.
+ webrtc::VideoCaptureRotation orientation;
+ WebRtc_UWord8 dummy_name[5];
+ EXPECT_NE(0, devInfo->GetOrientation(dummy_name, orientation));
+
+ // Test SetRotation.
+ EXPECT_NE(0, ViE.capture->SetRotateCapturedFrames(
+ captureId, webrtc::RotateCapturedFrame_90));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+
+ // Allocate capture device.
+ EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
+
+ EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
+ captureId, webrtc::RotateCapturedFrame_0));
+ EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
+ captureId, webrtc::RotateCapturedFrame_90));
+ EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
+ captureId, webrtc::RotateCapturedFrame_180));
+ EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
+ captureId, webrtc::RotateCapturedFrame_270));
+
+ // Release the capture device
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+ delete devInfo;
+ vcpm->Release();
+}
+
+void ViEAutoTest::ViECaptureExternalCaptureTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ TbInterfaces ViE("ViECaptureExternalCaptureTest");
+ TbVideoChannel channel(ViE);
+ channel.StartReceive();
+ channel.StartSend();
+
+ webrtc::VideoCaptureExternal* externalCapture;
+ int captureId = 0;
+
+ // Allocate the external capture device.
+ webrtc::VideoCaptureModule* vcpm = webrtc::VideoCaptureFactory::Create(
+ 0, externalCapture);
+ EXPECT_TRUE(vcpm != NULL);
+ EXPECT_TRUE(externalCapture != NULL);
+ vcpm->AddRef();
+
+ EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
+
+ // Connect the capture device to the channel.
+ EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId,
+ channel.videoChannel));
+
+ // Render the local capture.
+ EXPECT_EQ(0, ViE.render->AddRenderer(captureId, _window1, 1, 0.0, 0.0,
+ 1.0, 1.0));
+
+ // Render the remote capture.
+ EXPECT_EQ(0, ViE.render->AddRenderer(channel.videoChannel, _window2, 1,
+ 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(captureId));
+ EXPECT_EQ(0, ViE.render->StartRender(channel.videoChannel));
+
+ // Register observer.
+ CaptureObserver observer;
+ EXPECT_EQ(0, ViE.capture->RegisterObserver(captureId, observer));
+
+ // Enable brightness alarm.
+ EXPECT_EQ(0, ViE.capture->EnableBrightnessAlarm(captureId, true));
+
+ CaptureEffectFilter effectFilter(176, 144);
+ EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(captureId,
+ effectFilter));
+
+ // Call started.
+ ViETest::Log("You should see local preview from external capture\n"
+ "in window 1 and the remote video in window 2.\n");
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+ const unsigned int videoFrameLength = (176 * 144 * 3) / 2;
+ unsigned char* videoFrame = new unsigned char[videoFrameLength];
+ memset(videoFrame, 128, 176 * 144);
+
+ int frameCount = 0;
+ webrtc::VideoCaptureCapability capability;
+ capability.width = 176;
+ capability.height = 144;
+ capability.rawType = webrtc::kVideoI420;
+
+ ViETest::Log("Testing external capturing and frame rate callbacks.");
+ // TODO: Change when using a real file!
+ // while (fread(videoFrame, videoFrameLength, 1, foreman) == 1)
+ while (frameCount < 120)
+ {
+ externalCapture->IncomingFrame(
+ videoFrame, videoFrameLength, capability,
+ webrtc::TickTime::Now().MillisecondTimestamp());
+ AutoTestSleep(33);
+
+ if (effectFilter.number_of_captured_frames_ > 2)
+ {
+ EXPECT_EQ(webrtc::Normal, observer._brightness) <<
+ "Brightness or picture alarm should not have been called yet.";
+ EXPECT_EQ(webrtc::AlarmCleared, observer._alarm) <<
+ "Brightness or picture alarm should not have been called yet.";
+ }
+ frameCount++;
+ }
+
+ // Test brightness alarm
+ // Test bright image
+ for (int i = 0; i < 176 * 144; ++i)
+ {
+ if (videoFrame[i] <= 155)
+ videoFrame[i] = videoFrame[i] + 100;
+ else
+ videoFrame[i] = 255;
+ }
+ ViETest::Log("Testing Brighness alarm");
+ for (int frame = 0; frame < 30; ++frame)
+ {
+ externalCapture->IncomingFrame(
+ videoFrame, videoFrameLength, capability,
+ webrtc::TickTime::Now().MillisecondTimestamp());
+ AutoTestSleep(33);
+ }
+ EXPECT_EQ(webrtc::Bright, observer._brightness) <<
+ "Should be bright at this point since we are using a bright image.";
+
+ // Test Dark image
+ for (int i = 0; i < 176 * 144; ++i)
+ {
+ videoFrame[i] = videoFrame[i] > 200 ? videoFrame[i] - 200 : 0;
+ }
+ for (int frame = 0; frame < 30; ++frame)
+ {
+ externalCapture->IncomingFrame(
+ videoFrame, videoFrameLength, capability,
+ webrtc::TickTime::Now().MillisecondTimestamp());
+ AutoTestSleep(33);
+ }
+ EXPECT_EQ(webrtc::Dark, observer._brightness) <<
+ "Should be dark at this point since we are using a dark image.";
+ EXPECT_GT(effectFilter.number_of_captured_frames_, 150) <<
+ "Frames should have been played.";
+
+ EXPECT_GE(observer._frameRate, 29) <<
+ "Frame rate callback should be approximately correct.";
+ EXPECT_LE(observer._frameRate, 30) <<
+ "Frame rate callback should be approximately correct.";
+
+ // Test no picture alarm
+ ViETest::Log("Testing NoPictureAlarm.");
+ AutoTestSleep(1050);
+
+ EXPECT_EQ(webrtc::AlarmRaised, observer._alarm) <<
+ "No picture alarm should be raised.";
+ for (int frame = 0; frame < 10; ++frame)
+ {
+ externalCapture->IncomingFrame(
+ videoFrame, videoFrameLength, capability,
+ webrtc::TickTime::Now().MillisecondTimestamp());
+ AutoTestSleep(33);
+ }
+ EXPECT_EQ(webrtc::AlarmCleared, observer._alarm) <<
+ "Alarm should be cleared since ge just got some data.";
+
+ delete videoFrame;
+
+ // Release the capture device
+ EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
+
+ // Release the capture device again
+ EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId));
+ EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
+ vcpm->Release();
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc
new file mode 100644
index 0000000..38d9c4e
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc
@@ -0,0 +1,355 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(CARBON_RENDERING)
+#include "vie_autotest_mac_carbon.h"
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "vie_autotest_main.h"
+
+ViEAutoTestWindowManager::ViEAutoTestWindowManager() :
+ _carbonWindow1(new WindowRef()),
+ _carbonWindow2(new WindowRef()),
+ _hiView1(new HIViewRef()),
+ _hiView2(new HIViewRef())
+{
+}
+
+ViEAutoTestWindowManager::~ViEAutoTestWindowManager()
+{
+ if (_carbonWindow1EventHandlerRef)
+ RemoveEventHandler(_carbonWindow1EventHandlerRef);
+
+ if (_carbonWindow2EventHandlerRef)
+ RemoveEventHandler(_carbonWindow2EventHandlerRef);
+
+ if (_carbonHIView1EventHandlerRef)
+ RemoveEventHandler(_carbonHIView1EventHandlerRef);
+
+ if (_carbonHIView2EventHandlerRef)
+ RemoveEventHandler(_carbonHIView2EventHandlerRef);
+
+ delete _carbonWindow1;
+ delete _carbonWindow2;
+ delete _hiView1;
+ delete _hiView2;
+}
+
+int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size,
+ char* window1Title,
+ char* window2Title)
+{
+
+ WindowAttributes windowAttributes = kWindowStandardDocumentAttributes
+ | kWindowStandardHandlerAttribute | kWindowCompositingAttribute;
+ Rect windowContentRect;
+ static const EventTypeSpec
+ windowEventTypes[] = { kEventClassWindow, kEventWindowBoundsChanged,
+ kEventClassWindow, kEventWindowBoundsChanging, kEventClassWindow,
+ kEventWindowZoomed, kEventClassWindow, kEventWindowExpanded,
+ kEventClassWindow, kEventWindowClickResizeRgn, kEventClassWindow,
+ kEventWindowClickDragRgn };
+
+ // ************* Window 1 and Event Handler ***********************
+
+ SetRect(&windowContentRect, window1Size.origin.x, window1Size.origin.y,
+ window1Size.origin.x + window1Size.size.width, window1Size.origin.y
+ + window1Size.size.height);
+
+ CreateNewWindow(kDocumentWindowClass, windowAttributes, &windowContentRect,
+ _carbonWindow1);
+ SetWindowTitleWithCFString(*_carbonWindow1, CFSTR("Carbon Window 1"));
+ ShowWindow(*_carbonWindow1);
+ InitCursor();
+ InstallWindowEventHandler(*_carbonWindow1,
+ NewEventHandlerUPP(HandleWindowEvent),
+ GetEventTypeCount(windowEventTypes),
+ windowEventTypes, (void*) this,
+ &_carbonWindow1EventHandlerRef);
+
+ // ************* Window 2 and Event Handler ***********************
+
+ SetRect(&windowContentRect, window2Size.origin.x, window2Size.origin.y,
+ window2Size.origin.x + window2Size.size.width, window2Size.origin.y
+ + window2Size.size.height);
+
+ CreateNewWindow(kDocumentWindowClass, windowAttributes, &windowContentRect,
+ _carbonWindow2);
+ SetWindowTitleWithCFString(*_carbonWindow2, CFSTR("Carbon Window 2"));
+ ShowWindow(*_carbonWindow2);
+ InitCursor();
+ InstallWindowEventHandler(*_carbonWindow2,
+ NewEventHandlerUPP(HandleWindowEvent),
+ GetEventTypeCount(windowEventTypes),
+ windowEventTypes, (void*) this,
+ &_carbonWindow2EventHandlerRef);
+
+#if defined(HIVIEWREF_MODE)
+ OSStatus status;
+ static const EventTypeSpec hiviewEventTypes[] = { kEventClassControl,
+ kEventControlBoundsChanged, kEventClassControl, kEventControlDraw };
+
+ HIRect hiView1Rect = { 10, 10, 200, 200 };
+ status = HICreateCustomView(&hiView1Rect, &_hiView1);
+ status = HIViewAddSubview(&_carbonWindow1, _hiView1);
+ HIViewSetZOrder(_hiView1, kHIViewZOrderAbove, NULL);
+ HIViewSetVisible(_hiView1, true);
+
+ HIViewInstallEventHandler(_hiView1, NewEventHandlerUPP(HandleHIViewEvent),
+ GetEventTypeCount(hiviewEventTypes),
+ hiviewEventTypes, (void *) this,
+ &_carbonHIView1EventHandlerRef);
+
+ HIRect hiView2Rect = { 10, 10, 200, 200 };
+ status = HICreateCustomView(&hiView2Rect, &_hiView2);
+ status = HIViewAddSubview(&_carbonWindow2, _hiView2);
+ HIViewSetZOrder(_hiView2, kHIViewZOrderAbove, NULL);
+ HIViewSetVisible(_hiView2, true);
+
+ HIViewInstallEventHandler(_hiView2, NewEventHandlerUPP(HandleHIViewEvent),
+ GetEventTypeCount(hiviewEventTypes),
+ hiviewEventTypes, (void *) this,
+ &_carbonHIView2EventHandlerRef);
+#endif
+
+ return 0;
+}
+
+pascal OSStatus ViEAutoTestWindowManager::HandleWindowEvent(
+ EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
+{
+
+ WindowRef windowRef = NULL;
+
+ int eventType = GetEventKind(theEvent);
+
+ // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/
+ // i686-win32/include/quicktime/CarbonEvents.h for a list of codes
+ GetEventParameter(theEvent, kEventParamDirectObject, typeWindowRef, NULL,
+ sizeof(WindowRef), NULL, &windowRef);
+
+ ViEAutoTestWindowManager* obj = (ViEAutoTestWindowManager*) (userData);
+
+ if (windowRef == obj->GetWindow1())
+ {
+ // event was triggered on window 1
+ }
+ else if (windowRef == obj->GetWindow2())
+ {
+ // event was triggered on window 2
+ }
+
+ if (kEventWindowBoundsChanged == eventType)
+ {
+ }
+ else if (kEventWindowBoundsChanging == eventType)
+ {
+ }
+ else if (kEventWindowZoomed == eventType)
+ {
+ }
+ else if (kEventWindowExpanding == eventType)
+ {
+ }
+ else if (kEventWindowExpanded == eventType)
+ {
+ }
+ else if (kEventWindowClickResizeRgn == eventType)
+ {
+ }
+ else if (kEventWindowClickDragRgn == eventType)
+ {
+ }
+ else
+ {
+ }
+
+ return noErr;
+}
+
+pascal OSStatus ViEAutoTestWindowManager::HandleHIViewEvent(
+ EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
+{
+ HIViewRef hiviewRef = NULL;
+
+ // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/
+ // i686-win32/include/quicktime/CarbonEvents.h for a list of codes
+ int eventType = GetEventKind(theEvent);
+ OSStatus status = noErr;
+ status = GetEventParameter(theEvent, kEventParamDirectObject,
+ typeControlRef, NULL, sizeof(ControlRef), NULL,
+ &hiviewRef);
+
+ if (GetEventClass(theEvent) == kEventClassControl)
+ {
+ if (GetEventKind(theEvent) == kEventControlDraw)
+ {
+ ViEAutoTestWindowManager* obj =
+ (ViEAutoTestWindowManager*) (userData);
+
+ CGContextRef context;
+ status = GetEventParameter(theEvent, kEventParamCGContextRef,
+ typeCGContextRef, NULL, sizeof(context),
+ NULL, &context);
+ HIRect viewBounds;
+
+ HIViewRef* ptrHIViewRef =
+ static_cast<HIViewRef*> (obj->GetWindow1());
+ if (hiviewRef == *ptrHIViewRef)
+ {
+ // color hiview1
+ CGContextSetRGBFillColor(context, 1, 0, 0, 1);
+ HIViewGetBounds(hiviewRef, &viewBounds);
+ CGContextFillRect(context, viewBounds);
+ }
+
+ ptrHIViewRef = static_cast<HIViewRef*> (obj->GetWindow1());
+ if (hiviewRef == *ptrHIViewRef)
+ {
+ CGContextSetRGBFillColor(context, 0, 1, 0, 1);
+ HIViewGetBounds(hiviewRef, &viewBounds);
+ CGContextFillRect(context, viewBounds);
+ }
+
+ }
+ }
+
+ /*
+
+
+ VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
+ WindowRef parentWindow = HIViewGetWindow(hiviewRef);
+ bool updateUI = true;
+
+ if(kEventControlBoundsChanged == eventType){
+ }
+ else if(kEventControlDraw == eventType){
+ }
+ else{
+ updateUI = false;
+ }
+
+ if(true == updateUI){
+ obj->ParentWindowResized(parentWindow);
+ obj->UpdateClipping();
+ obj->RenderOffScreenBuffers();
+ }
+ */
+
+ return status;
+}
+
+int ViEAutoTestWindowManager::TerminateWindows()
+{
+ return 0;
+}
+
+void* ViEAutoTestWindowManager::GetWindow1()
+{
+#if defined(HIVIEWREF_MODE)
+ return (void*)_hiView1;
+#else
+ return (void*) _carbonWindow1;
+#endif
+
+}
+void* ViEAutoTestWindowManager::GetWindow2()
+{
+#if defined(HIVIEWREF_MODE)
+ return (void*)_hiView2;
+#else
+ return (void*) _carbonWindow2;
+#endif
+
+}
+
+bool ViEAutoTestWindowManager::SetTopmostWindow()
+{
+ return true;
+}
+
+/*
+
+ int main (int argc, const char * argv[])
+ {
+ ViEAutoTestMain autoTest;
+
+ if(argc > 1){
+ autoTest.UseAnswerFile(argv[1]);
+ }
+
+ int success = autoTest.BeginOSIndependentTesting();
+
+ }
+
+ */
+
+int main(int argc, const char * argv[])
+{
+ NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
+
+ [NSApplication sharedApplication];
+
+ // We have to run the test in a secondary thread because we need to run a
+ // runloop, which blocks.
+ if (argc > 1)
+ {
+AutoTestClass * autoTestClass = [[AutoTestClass alloc]init];
+ [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
+ toTarget:autoTestClass withObject:[NSString stringWithFormat:@"%s",
+ argv[1]]];
+}
+else
+{
+ AutoTestClass* autoTestClass = [[AutoTestClass alloc]init];
+ [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
+ toTarget:autoTestClass withObject:nil];
+}
+
+// process OS events. Blocking call
+[[NSRunLoop currentRunLoop]run];
+[pool release];
+}
+
+@implementation AutoTestClass
+
+-(void)autoTestWithArg:(NSString*)answerFile;
+{
+ // TODO(phoglund): Rewrite this file to work with the new way of running
+ // vie_auto_test. The file doesn't seem to be used at the moment though.
+ NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
+
+ ViEAutoTestMain autoTest;
+
+ if(NSOrderedSame != [answerFile compare:@""])
+ {
+ char answerFileUTF8[1024] = "";
+ strcpy(answerFileUTF8, (char*)[answerFileUTF8 UTF8]);
+ autoTest.UseAnswerFile(answerFileUTF8);
+ }
+
+ int success = autoTest.BeginOSIndependentTesting();
+
+ [pool release];
+ return;
+}
+// TODO: move window creation to Obj-c class so GUI commands can be run on the
+// main NSThread
+// -(void)createWindow1:(AutoTestRect)window1Size
+// AndWindow2:(AutoTestRect)window2Size WithTitle1:(char*)window1Title
+// AndTitle2:(char*)window2Title{
+
+@end
+
+#endif
+
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm b/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
new file mode 100644
index 0000000..3f9ef6b
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
@@ -0,0 +1,132 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#if defined(COCOA_RENDERING)
+#import "cocoa_render_view.h"
+#include "vie_autotest_mac_cocoa.h"
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "vie_autotest_main.h"
+
+ViEAutoTestWindowManager::ViEAutoTestWindowManager()
+ : _cocoaRenderView1(nil), _cocoaRenderView2(nil) {
+}
+
+int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size,
+ void* window1Title,
+ void* window2Title) {
+ NSRect outWindow1Frame = NSMakeRect(window1Size.origin.x,
+ window1Size.origin.y,
+ window1Size.size.width,
+ window1Size.size.height);
+ outWindow1_ = [[NSWindow alloc] initWithContentRect:outWindow1Frame
+ styleMask:NSTitledWindowMask
+ backing:NSBackingStoreBuffered defer:NO];
+ [outWindow1_ orderOut:nil];
+ NSRect cocoaRenderView1Frame = NSMakeRect(0, 0, window1Size.size.width,
+ window1Size.size.height);
+ _cocoaRenderView1 = [[CocoaRenderView alloc]
+ initWithFrame:cocoaRenderView1Frame];
+ [[outWindow1_ contentView] addSubview:(NSView*)_cocoaRenderView1];
+ [outWindow1_ setTitle:[NSString stringWithFormat:@"%s", window1Title]];
+ [outWindow1_ makeKeyAndOrderFront:NSApp];
+
+ NSRect outWindow2Frame = NSMakeRect(window2Size.origin.x,
+ window2Size.origin.y,
+ window2Size.size.width,
+ window2Size.size.height);
+ outWindow2_ = [[NSWindow alloc] initWithContentRect:outWindow2Frame
+ styleMask:NSTitledWindowMask
+ backing:NSBackingStoreBuffered defer:NO];
+ [outWindow2_ orderOut:nil];
+ NSRect cocoaRenderView2Frame = NSMakeRect(0, 0, window2Size.size.width,
+ window2Size.size.height);
+ _cocoaRenderView2 = [[CocoaRenderView alloc]
+ initWithFrame:cocoaRenderView2Frame];
+ [[outWindow2_ contentView] addSubview:(NSView*)_cocoaRenderView2];
+ [outWindow2_ setTitle:[NSString stringWithFormat:@"%s", window2Title]];
+ [outWindow2_ makeKeyAndOrderFront:NSApp];
+
+ return 0;
+}
+
+int ViEAutoTestWindowManager::TerminateWindows() {
+ [outWindow1_ close];
+ [outWindow2_ close];
+ return 0;
+}
+
+void* ViEAutoTestWindowManager::GetWindow1() {
+ return _cocoaRenderView1;
+}
+
+void* ViEAutoTestWindowManager::GetWindow2() {
+ return _cocoaRenderView2;
+}
+
+bool ViEAutoTestWindowManager::SetTopmostWindow() {
+ return true;
+}
+
+int main(int argc, char * argv[]) {
+ NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
+
+ [NSApplication sharedApplication];
+
+ int result = 0;
+#if defined(MAC_COCOA_USE_NSRUNLOOP)
+ AutoTestClass* tests = [[AutoTestClass alloc] init];
+
+ [tests setArgc:argc argv:argv];
+ [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
+ toTarget:tests withObject:nil];
+ // Process OS events. Blocking call.
+ [[NSRunLoop mainRunLoop]run];
+
+ result = [tests result];
+
+#else
+ ViEAutoTestMain autoTest;
+ result = autoTest.RunTests(argc, argv);
+
+#endif
+ [pool release];
+ return result;
+}
+
+@implementation AutoTestClass
+
+- (void)setArgc:(int)argc argv:(char**)argv {
+ argc_ = argc;
+ argv_ = argv;
+}
+
+- (void)autoTestWithArg:(NSObject*)ignored {
+ NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
+
+ ViEAutoTestMain auto_test;
+
+ result_ = auto_test.RunTests(argc_, argv_);
+
+ [pool release];
+ return;
+}
+
+- (int)result {
+ return result_;
+}
+
+@end
+
+#endif
+
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc
new file mode 100644
index 0000000..ed8f20b
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -0,0 +1,455 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_autotest.h"
+
+#include "codec_primitives.h"
+#include "common_types.h"
+#include "general_primitives.h"
+#include "tb_capture_device.h"
+#include "tb_I420_codec.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "vie_autotest_defines.h"
+
+class RenderFilter : public webrtc::ViEEffectFilter {
+ public:
+ RenderFilter()
+ : last_render_width_(0),
+ last_render_height_(0) {}
+
+ ~RenderFilter() {}
+
+ virtual int Transform(int size, unsigned char* frame_buffer,
+ unsigned int time_stamp, unsigned int width,
+ unsigned int height) {
+ last_render_width_ = width;
+ last_render_height_ = height;
+ return 0;
+ }
+ unsigned int last_render_width_;
+ unsigned int last_render_height_;
+};
+
+void ViEAutoTest::ViECodecStandardTest()
+{
+ TbInterfaces interfaces = TbInterfaces("ViECodecStandardTest");
+ TbCaptureDevice capture_device = TbCaptureDevice(interfaces);
+
+ int video_channel = -1;
+
+ EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
+ EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
+ capture_device.captureId, video_channel));
+
+ ConfigureRtpRtcp(interfaces.rtp_rtcp,
+ video_channel);
+
+ RenderInWindow(interfaces.render, capture_device.captureId, _window1, 0);
+ RenderInWindow(interfaces.render, video_channel, _window2, 1);
+
+ TestCodecs(interfaces, capture_device.captureId, video_channel,
+ kDoNotForceResolution, kDoNotForceResolution);
+}
+
+void ViEAutoTest::ViECodecExtendedTest()
+{
+ {
+ ViECodecAPITest();
+ ViECodecStandardTest();
+ ViECodecExternalCodecTest();
+
+ TbInterfaces interfaces = TbInterfaces("ViECodecExtendedTest");
+ webrtc::ViEBase* ptrViEBase = interfaces.base;
+ webrtc::ViECapture* ptrViECapture = interfaces.capture;
+ webrtc::ViERender* ptrViERender = interfaces.render;
+ webrtc::ViECodec* ptrViECodec = interfaces.codec;
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp = interfaces.rtp_rtcp;
+ webrtc::ViENetwork* ptrViENetwork = interfaces.network;
+
+ TbCaptureDevice captureDevice = TbCaptureDevice(interfaces);
+ int captureId = captureDevice.captureId;
+
+ int videoChannel = -1;
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
+ EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
+ captureId, videoChannel));
+
+ EXPECT_EQ(0, ptrViERtpRtcp->SetRTCPStatus(
+ videoChannel, webrtc::kRtcpCompound_RFC4585));
+ EXPECT_EQ(0, ptrViERtpRtcp->SetKeyFrameRequestMethod(
+ videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true));
+ EXPECT_EQ(0, ptrViERender->AddRenderer(
+ captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ptrViERender->AddRenderer(
+ videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ptrViERender->StartRender(captureId));
+ EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
+
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
+
+ if (videoCodec.codecType != webrtc::kVideoCodecI420)
+ {
+ videoCodec.width = 640;
+ videoCodec.height = 480;
+ }
+ EXPECT_EQ(0, ptrViECodec->SetReceiveCodec(
+ videoChannel, videoCodec));
+ }
+
+ const char* ipAddress = "127.0.0.1";
+ const unsigned short rtpPort = 6000;
+ EXPECT_EQ(0, ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort));
+ EXPECT_EQ(0, ptrViEBase->StartReceive(videoChannel));
+ EXPECT_EQ(0, ptrViENetwork->SetSendDestination(
+ videoChannel, ipAddress, rtpPort));
+ EXPECT_EQ(0, ptrViEBase->StartSend(videoChannel));
+
+ //
+ // Codec specific tests
+ //
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ EXPECT_EQ(0, ptrViEBase->StopSend(videoChannel));
+ ViEAutotestCodecObserver codecObserver;
+ EXPECT_EQ(0, ptrViECodec->RegisterEncoderObserver(
+ videoChannel, codecObserver));
+ EXPECT_EQ(0, ptrViECodec->RegisterDecoderObserver(
+ videoChannel, codecObserver));
+
+ EXPECT_EQ(0, ptrViEBase->StopReceive(videoChannel));
+ EXPECT_NE(0, ptrViEBase->StopSend(videoChannel)); // Already stopped
+
+ EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
+ EXPECT_EQ(0, ptrViERender->RemoveRenderer(captureId));
+ EXPECT_EQ(0, ptrViERender->RemoveRenderer(videoChannel));
+ EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
+ EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
+ }
+
+ //
+ // Default channel
+ //
+ {
+ // Create VIE
+ TbInterfaces ViE("ViECodecExtendedTest2");
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+
+ // Create channel 1
+ int videoChannel1 = -1;
+ EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel1));
+
+ unsigned short rtpPort1 = 12000;
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ videoChannel1, rtpPort1));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ videoChannel1, "127.0.0.1", rtpPort1));
+ tbCapture.ConnectTo(videoChannel1);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ videoChannel1, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ videoChannel1, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(videoChannel1));
+
+ ViEAutotestCodecObserver codecObserver1;
+ EXPECT_EQ(0, ViE.codec->RegisterEncoderObserver(
+ videoChannel1, codecObserver1));
+ EXPECT_EQ(0, ViE.codec->RegisterDecoderObserver(
+ videoChannel1, codecObserver1));
+
+ // Set Send codec
+ unsigned short codecWidth = 176;
+ unsigned short codecHeight = 144;
+ bool codecSet = false;
+ webrtc::VideoCodec videoCodec;
+ for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel1, videoCodec));
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8)
+ {
+ videoCodec.width = codecWidth;
+ videoCodec.height = codecHeight;
+ videoCodec.startBitrate = 200;
+ videoCodec.maxBitrate = 300;
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(
+ videoChannel1, videoCodec));
+ codecSet = true;
+ break;
+ }
+ }
+ EXPECT_TRUE(codecSet);
+ webrtc::VideoCodec send_codec;
+ memcpy(&send_codec, &videoCodec, sizeof(videoCodec));
+
+ EXPECT_EQ(0, ViE.base->StartSend(videoChannel1));
+ EXPECT_EQ(0, ViE.base->StartReceive(videoChannel1));
+
+ // Create channel 2, based on channel 1
+ int videoChannel2 = -1;
+ EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel2, videoChannel1));
+ EXPECT_NE(videoChannel1, videoChannel2) <<
+ "Channel 2 should be seop";
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ videoChannel2, webrtc::kViEKeyFrameRequestPliRtcp));
+
+ // Prepare receive codecs
+ for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel2, videoCodec));
+ }
+
+ ViEAutotestCodecObserver codecObserver2;
+ EXPECT_EQ(0, ViE.codec->RegisterDecoderObserver(
+ videoChannel2, codecObserver2));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ videoChannel2, _window2, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(videoChannel2));
+
+ unsigned short rtpPort2 = 13000;
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(videoChannel2, rtpPort2));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ videoChannel2, "127.0.0.1", rtpPort2));
+
+ EXPECT_EQ(0, ViE.base->StartReceive(videoChannel2));
+ EXPECT_EQ(-1, ViE.base->StartSend(videoChannel2));
+
+ EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel1));
+ EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel2));
+
+ }
+}
+
+void ViEAutoTest::ViECodecAPITest()
+{
+ // ***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ // ***************************************************************
+ webrtc::VideoEngine* ptrViE = NULL;
+ ptrViE = webrtc::VideoEngine::Create();
+ EXPECT_TRUE(ptrViE != NULL);
+
+ webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
+ EXPECT_TRUE(ptrViEBase != NULL);
+
+ EXPECT_EQ(0, ptrViEBase->Init());
+
+ int videoChannel = -1;
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
+
+ webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
+ EXPECT_TRUE(ptrViECodec != NULL);
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ //
+ // SendCodec
+ //
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+
+ const int numberOfCodecs = ptrViECodec->NumberOfCodecs();
+ EXPECT_GT(numberOfCodecs, 0);
+
+ SetSendCodec(webrtc::kVideoCodecVP8, ptrViECodec, videoChannel,
+ kDoNotForceResolution, kDoNotForceResolution);
+
+ memset(&videoCodec, 0, sizeof(videoCodec));
+ EXPECT_EQ(0, ptrViECodec->GetSendCodec(videoChannel, videoCodec));
+ EXPECT_EQ(webrtc::kVideoCodecVP8, videoCodec.codecType);
+ // Verify that the target bit rate is equal to the start bitrate.
+ unsigned int target_bitrate = 0;
+ EXPECT_EQ(0, ptrViECodec->GetCodecTargetBitrate(videoChannel,
+ &target_bitrate));
+ EXPECT_EQ(videoCodec.startBitrate, target_bitrate);
+
+ SetSendCodec(webrtc::kVideoCodecI420, ptrViECodec, videoChannel,
+ kDoNotForceResolution, kDoNotForceResolution);
+ memset(&videoCodec, 0, sizeof(videoCodec));
+ EXPECT_EQ(0, ptrViECodec->GetSendCodec(videoChannel, videoCodec));
+ EXPECT_EQ(webrtc::kVideoCodecI420, videoCodec.codecType);
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+
+ EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
+
+ EXPECT_EQ(0, ptrViECodec->Release());
+ EXPECT_EQ(0, ptrViEBase->Release());
+ EXPECT_TRUE(webrtc::VideoEngine::Delete(ptrViE));
+}
+
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+#include "vie_external_codec.h"
+#endif
+void ViEAutoTest::ViECodecExternalCodecTest()
+{
+ // ***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ // ***************************************************************
+
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+ {
+ TbInterfaces ViE("ViEExternalCodec");
+ TbCaptureDevice captureDevice(ViE);
+ TbVideoChannel channel(
+ ViE, webrtc::kVideoCodecI420, 352,288,30,(352*288*3*8*30)/(2*1000));
+
+ captureDevice.ConnectTo(channel.videoChannel);
+
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ channel.videoChannel, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(channel.videoChannel));
+
+ channel.StartReceive();
+ channel.StartSend();
+
+ ViETest::Log("Using internal I420 codec");
+ AutoTestSleep(KAutoTestSleepTimeMs/2);
+
+ webrtc::ViEExternalCodec* ptrViEExtCodec =
+ webrtc::ViEExternalCodec::GetInterface(ViE.video_engine);
+ EXPECT_TRUE(ptrViEExtCodec != NULL);
+
+ webrtc::VideoCodec codecStruct;
+
+ EXPECT_EQ(0, ViE.codec->GetSendCodec(
+ channel.videoChannel, codecStruct));
+
+ // Use external encoder instead
+ {
+ TbI420Encoder extEncoder;
+
+ // Test to register on wrong channel
+ EXPECT_NE(0, ptrViEExtCodec->RegisterExternalSendCodec(
+ channel.videoChannel+5,codecStruct.plType,&extEncoder));
+ EXPECT_EQ(kViECodecInvalidArgument, ViE.LastError());
+
+ EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalSendCodec(
+ channel.videoChannel,codecStruct.plType,&extEncoder));
+
+ // Use new external encoder
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(
+ channel.videoChannel, codecStruct));
+
+ TbI420Decoder extDecoder;
+ EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalReceiveCodec(
+ channel.videoChannel,codecStruct.plType,&extDecoder));
+
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(
+ channel.videoChannel, codecStruct));
+
+ ViETest::Log("Using external I420 codec");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ // Test to deregister on wrong channel
+ EXPECT_NE(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
+ channel.videoChannel+5,codecStruct.plType));
+ EXPECT_EQ(kViECodecInvalidArgument, ViE.LastError());
+
+ // Test to deregister wrong payload type.
+ EXPECT_NE(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
+ channel.videoChannel,codecStruct.plType-1));
+
+ // Deregister external send codec
+ EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
+ channel.videoChannel,codecStruct.plType));
+
+ EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalReceiveCodec(
+ channel.videoChannel,codecStruct.plType));
+
+ // Verify that the encoder and decoder has been used
+ TbI420Encoder::FunctionCalls encodeCalls =
+ extEncoder.GetFunctionCalls();
+ EXPECT_EQ(1, encodeCalls.InitEncode);
+ EXPECT_EQ(1, encodeCalls.Release);
+ EXPECT_EQ(1, encodeCalls.RegisterEncodeCompleteCallback);
+ EXPECT_GT(encodeCalls.Encode, 30);
+ EXPECT_GT(encodeCalls.SetRates, 1);
+ EXPECT_GT(encodeCalls.SetPacketLoss, 1);
+
+ TbI420Decoder::FunctionCalls decodeCalls =
+ extDecoder.GetFunctionCalls();
+ EXPECT_EQ(1, decodeCalls.InitDecode);
+ EXPECT_EQ(1, decodeCalls.Release);
+ EXPECT_EQ(1, decodeCalls.RegisterDecodeCompleteCallback);
+ EXPECT_GT(decodeCalls.Decode, 30);
+
+ ViETest::Log("Changing payload type Using external I420 codec");
+
+ codecStruct.plType = codecStruct.plType - 1;
+ EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalReceiveCodec(
+ channel.videoChannel, codecStruct.plType, &extDecoder));
+
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(
+ channel.videoChannel, codecStruct));
+
+ EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalSendCodec(
+ channel.videoChannel, codecStruct.plType, &extEncoder));
+
+ // Use new external encoder
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(
+ channel.videoChannel, codecStruct));
+
+ AutoTestSleep(KAutoTestSleepTimeMs/2);
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+
+ EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
+ channel.videoChannel,codecStruct.plType));
+ EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalReceiveCodec(
+ channel.videoChannel,codecStruct.plType));
+
+ // Verify that the encoder and decoder has been used
+ encodeCalls = extEncoder.GetFunctionCalls();
+ EXPECT_EQ(2, encodeCalls.InitEncode);
+ EXPECT_EQ(2, encodeCalls.Release);
+ EXPECT_EQ(2, encodeCalls.RegisterEncodeCompleteCallback);
+ EXPECT_GT(encodeCalls.Encode, 30);
+ EXPECT_GT(encodeCalls.SetRates, 1);
+ EXPECT_GT(encodeCalls.SetPacketLoss, 1);
+
+ decodeCalls = extDecoder.GetFunctionCalls();
+
+ EXPECT_EQ(2, decodeCalls.InitDecode);
+ EXPECT_EQ(2, decodeCalls.Release);
+ EXPECT_EQ(2, decodeCalls.RegisterDecodeCompleteCallback);
+ EXPECT_GT(decodeCalls.Decode, 30);
+
+ EXPECT_EQ(0, ptrViEExtCodec->Release());
+ } // tbI420Encoder and extDecoder goes out of scope
+
+ ViETest::Log("Using internal I420 codec");
+ AutoTestSleep(KAutoTestSleepTimeMs/2);
+
+ }
+
+#else
+ ViETest::Log(" ViEExternalCodec not enabled\n");
+#endif
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
new file mode 100644
index 0000000..fa8bf76
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
@@ -0,0 +1,2058 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * vie_autotest_custom_call.cc
+ *
+ */
+
+#include <iostream>
+
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+
+#define VCM_RED_PAYLOAD_TYPE 96
+#define VCM_ULPFEC_PAYLOAD_TYPE 97
+#define DEFAULT_SEND_IP "127.0.0.1"
+#define DEFAULT_VIDEO_PORT 11111
+#define DEFAULT_VIDEO_CODEC "vp8"
+#define DEFAULT_VIDEO_CODEC_WIDTH 640
+#define DEFAULT_VIDEO_CODEC_HEIGHT 480
+#define DEFAULT_VIDEO_CODEC_BITRATE 300
+#define DEFAULT_VIDEO_CODEC_MIN_BITRATE 100
+#define DEFAULT_VIDEO_CODEC_MAX_BITRATE 1000
+#define DEFAULT_AUDIO_PORT 11113
+#define DEFAULT_AUDIO_CODEC "ISAC"
+#define DEFAULT_INCOMING_FILE_NAME "IncomingFile.avi"
+#define DEFAULT_OUTGOING_FILE_NAME "OutgoingFile.avi"
+#define DEFAULT_VIDEO_CODEC_MAX_FRAMERATE 30
+#define DEFAULT_VIDEO_PROTECTION_METHOD 0
+#define DEFAULT_TEMPORAL_LAYER 0
+
+enum StatisticsType {
+ kSendStatistic,
+ kReceivedStatistic
+};
+
+class ViEAutotestFileObserver: public webrtc::ViEFileObserver {
+ public:
+ ViEAutotestFileObserver() {};
+ ~ViEAutotestFileObserver() {};
+
+ void PlayFileEnded(const WebRtc_Word32 fileId) {
+ ViETest::Log("PlayFile ended");
+ }
+};
+
+class ViEAutotestEncoderObserver: public webrtc::ViEEncoderObserver {
+ public:
+ ViEAutotestEncoderObserver() {};
+ ~ViEAutotestEncoderObserver() {};
+
+ void OutgoingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate) {
+ std::cout << "Send FR: " << framerate
+ << " BR: " << bitrate << std::endl;
+ }
+};
+
+class ViEAutotestDecoderObserver: public webrtc::ViEDecoderObserver {
+ public:
+ ViEAutotestDecoderObserver() {};
+ ~ViEAutotestDecoderObserver() {};
+
+ void IncomingRate(const int videoChannel,
+ const unsigned int framerate,
+ const unsigned int bitrate) {
+ std::cout << "Received FR: " << framerate
+ << " BR: " << bitrate << std::endl;
+ }
+ void IncomingCodecChanged(const int videoChannel,
+ const webrtc::VideoCodec& codec) {}
+ void RequestNewKeyFrame(const int videoChannel) {
+ std::cout << "Decoder requesting a new key frame." << std::endl;
+ }
+};
+
+// The following are general helper functions.
+bool GetVideoDevice(webrtc::ViEBase* ptrViEBase,
+ webrtc::ViECapture* ptrViECapture,
+ char* captureDeviceName, char* captureDeviceUniqueId);
+bool GetIPAddress(char* IP);
+#ifndef WEBRTC_ANDROID
+bool ValidateIP(std::string iStr);
+#endif
+// The following are Print to stdout functions.
+void PrintCallInformation(char* IP, char* videoCaptureDeviceName,
+ char* videoCaptureUniqueId,
+ webrtc::VideoCodec videoCodec, int videoTxPort,
+ int videoRxPort, char* audioCaptureDeviceName,
+ char* audioPlaybackDeviceName,
+ webrtc::CodecInst audioCodec, int audioTxPort,
+ int audioRxPort, int protectionMethod);
+void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel, StatisticsType statType);
+void PrintRTPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel);
+void PrintBandwidthUsage(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel);
+void PrintCodecStatistics(webrtc::ViECodec* ptrViECodec, int videoChannel,
+ StatisticsType statType);
+void PrintGetDiscardedPackets(webrtc::ViECodec* ptrViECodec, int videoChannel);
+void PrintVideoStreamInformation(webrtc::ViECodec* ptrViECodec,
+ int videoChannel);
+void PrintVideoCodec(webrtc::VideoCodec videoCodec);
+
+// The following are video functions.
+// TODO(amyfong): change to pointers as input arguments
+// instead of references
+bool SetVideoPorts(int* txPort, int* rxPort);
+bool SetVideoCodecType(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecResolution(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecSize(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecMinBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecMaxBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecMaxFramerate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec);
+bool SetVideoCodecTemporalLayer(webrtc::VideoCodec& videoCodec);
+int GetVideoProtection();
+bool SetVideoProtection(webrtc::ViECodec* ptrViECodec,
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel, int protectionMethod);
+bool GetBitrateSignaling();
+
+// The following are audio helper functions.
+bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
+ webrtc::VoEHardware* ptrVEHardware,
+ char* recordingDeviceName, int& recordingDeviceIndex,
+ char* playbackDeviceName, int& playbackDeviceIndex);
+bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
+ webrtc::VoEHardware* ptrVEHardware,
+ int& recordingDeviceIndex, int& playbackDeviceIndex);
+bool GetAudioPorts(int* txPort, int* rxPort);
+bool GetAudioCodec(webrtc::VoECodec* ptrVeCodec,
+ webrtc::CodecInst& audioCodec);
+
+int ViEAutoTest::ViECustomCall()
+{
+ ViETest::Log(" ");
+ ViETest::Log("========================================");
+ ViETest::Log(" Enter values to use custom settings\n");
+
+ int error = 0;
+ int numberOfErrors = 0;
+ std::string str;
+
+ // Create the VoE and get the VoE interfaces.
+ webrtc::VoiceEngine* ptrVE = webrtc::VoiceEngine::Create();
+ numberOfErrors += ViETest::TestError(ptrVE != NULL, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ webrtc::VoEBase* ptrVEBase = webrtc::VoEBase::GetInterface(ptrVE);
+ numberOfErrors += ViETest::TestError(ptrVEBase != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ error = ptrVEBase->Init();
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ webrtc::VoECodec* ptrVECodec = webrtc::VoECodec::GetInterface(ptrVE);
+ numberOfErrors += ViETest::TestError(ptrVECodec != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::VoEHardware* ptrVEHardware =
+ webrtc::VoEHardware::GetInterface(ptrVE);
+ numberOfErrors += ViETest::TestError(ptrVEHardware != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::VoEAudioProcessing* ptrVEAPM =
+ webrtc::VoEAudioProcessing::GetInterface(ptrVE);
+ numberOfErrors += ViETest::TestError(ptrVEAPM != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ // Create the ViE and get the ViE Interfaces.
+ webrtc::VideoEngine* ptrViE = webrtc::VideoEngine::Create();
+ numberOfErrors += ViETest::TestError(ptrViE != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViEBase != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ error = ptrViEBase->Init();
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ webrtc::ViECapture* ptrViECapture =
+ webrtc::ViECapture::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViECapture != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViERender != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViECodec != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::ViENetwork* ptrViENetwork =
+ webrtc::ViENetwork::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViENetwork != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ webrtc::ViEFile* ptrViEFile = webrtc::ViEFile::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViEFile != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
+ bool startCall = false;
+ const unsigned int kMaxIPLength = 16;
+ char ipAddress[kMaxIPLength] = "";
+ const unsigned int KMaxUniqueIdLength = 256;
+ char uniqueId[KMaxUniqueIdLength] = "";
+ char deviceName[KMaxUniqueIdLength] = "";
+ int videoTxPort = 0;
+ int videoRxPort = 0;
+ int videoChannel = -1;
+ webrtc::VideoCodec videoSendCodec;
+ char audioCaptureDeviceName[KMaxUniqueIdLength] = "";
+ char audioPlaybackDeviceName[KMaxUniqueIdLength] = "";
+ int audioCaptureDeviceIndex = -1;
+ int audioPlaybackDeviceIndex = -1;
+ int audioTxPort = 0;
+ int audioRxPort = 0;
+ webrtc::CodecInst audioCodec;
+ int audioChannel = -1;
+ bool isImageScaleEnabled = false;
+ int protectionMethod = DEFAULT_VIDEO_PROTECTION_METHOD;
+ bool remb = true;
+
+ while (!startCall) {
+ // Get the IP address to use from call.
+ memset(ipAddress, 0, kMaxIPLength);
+ GetIPAddress(ipAddress);
+
+ // Get the video device to use for call.
+ memset(deviceName, 0, KMaxUniqueIdLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+ GetVideoDevice(ptrViEBase, ptrViECapture, deviceName, uniqueId);
+
+ // Get and set the video ports for the call.
+ videoTxPort = 0;
+ videoRxPort = 0;
+ SetVideoPorts(&videoTxPort, &videoRxPort);
+
+ // Get and set the video codec parameters for the call.
+ memset((void*)&videoSendCodec, 0, sizeof(webrtc::VideoCodec));
+ SetVideoCodecType(ptrViECodec, videoSendCodec);
+ SetVideoCodecSize(ptrViECodec, videoSendCodec);
+ SetVideoCodecBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMinBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMaxBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMaxFramerate(ptrViECodec, videoSendCodec);
+ SetVideoCodecTemporalLayer(videoSendCodec);
+ remb = GetBitrateSignaling();
+
+ // Get the video protection method for the call.
+ protectionMethod = GetVideoProtection();
+
+ // Get the audio device for the call.
+ memset(audioCaptureDeviceName, 0, KMaxUniqueIdLength);
+ memset(audioPlaybackDeviceName, 0, KMaxUniqueIdLength);
+ GetAudioDevices(ptrVEBase, ptrVEHardware, audioCaptureDeviceName,
+ audioCaptureDeviceIndex, audioPlaybackDeviceName,
+ audioPlaybackDeviceIndex);
+
+ // Get the audio port for the call.
+ audioTxPort = 0;
+ audioRxPort = 0;
+ GetAudioPorts(&audioTxPort, &audioRxPort);
+
+ // Get the audio codec for the call.
+ memset((void*)&audioCodec, 0, sizeof(audioCodec));
+ GetAudioCodec(ptrVECodec, audioCodec);
+
+ // Now ready to start the call. Check user wants to continue.
+ PrintCallInformation(ipAddress, deviceName, uniqueId, videoSendCodec,
+ videoTxPort, videoRxPort, audioCaptureDeviceName,
+ audioPlaybackDeviceName, audioCodec, audioTxPort,
+ audioRxPort, protectionMethod);
+
+ std::cout << std::endl;
+ std::cout << "1. Start the call" << std::endl;
+ std::cout << "2. Reconfigure call settings" << std::endl;
+ std::cout << "What do you want to do? Press enter for default "
+ << "(Start the call): ";
+
+ std::getline(std::cin, str);
+ int selection = 0;
+ selection = atoi(str.c_str());
+
+ switch (selection) {
+ case 0:
+ startCall = true;
+ break;
+ case 1:
+ startCall = true;
+ break;
+ case 2:
+ startCall = false;
+ break;
+ default:
+ // Invalid selection gets error mesage.
+ std::cout << "ERROR: Code=" << error
+ << " Invalid selection" << std::endl;
+ continue;
+ }
+ }
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing.
+ //***************************************************************
+ if (startCall == true) {
+ // Configure audio channel first.
+ audioChannel = ptrVEBase->CreateChannel();
+ error = ptrVEBase->SetSendDestination(audioChannel, audioTxPort,
+ ipAddress);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEBase->SetLocalReceiver(audioChannel, audioRxPort);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEHardware->SetRecordingDevice(audioCaptureDeviceIndex);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEHardware->SetPlayoutDevice(audioPlaybackDeviceIndex);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVECodec->SetSendCodec(audioChannel, audioCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEAPM->SetAgcStatus(true, webrtc::kAgcDefault);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEAPM->SetNsStatus(true, webrtc::kNsHighSuppression);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ // Now configure the video channel.
+ error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ std::string trace_file =
+ ViETest::GetResultOutputPath() + "ViECustomCall_trace.txt";
+ error = ptrViE->SetTraceFile(trace_file.c_str());
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->SetVoiceEngine(ptrVE);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->CreateChannel(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ int captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId,
+ KMaxUniqueIdLength,
+ captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECapture->StartCapture(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp =
+ webrtc::ViERTP_RTCP::GetInterface(ptrViE);
+ numberOfErrors += ViETest::TestError(ptrViE != NULL,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
+ webrtc::kRtcpCompound_RFC4585);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERtpRtcp->SetKeyFrameRequestMethod(videoChannel,
+ webrtc::kViEKeyFrameRequestPliRtcp);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ if (remb) {
+ error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ } else {
+ error = ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true);
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ }
+
+ error = ptrViERender->AddRenderer(captureId, _window1, 0, 0.0, 0.0,
+ 1.0, 1.0);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->AddRenderer(videoChannel, _window2, 1, 0.0, 0.0,
+ 1.0, 1.0);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViENetwork->SetSendDestination(videoChannel, ipAddress,
+ videoTxPort);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViENetwork->SetLocalReceiver(videoChannel, videoRxPort);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ // Set the Video Protection before start send and receive.
+ SetVideoProtection(ptrViECodec, ptrViERtpRtcp,
+ videoChannel, protectionMethod);
+
+ // Start Voice Playout and Receive.
+ error = ptrVEBase->StartReceive(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEBase->StartPlayout(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEBase->StartSend(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ // Now start the Video Send & Receive.
+ error = ptrViEBase->StartSend(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->StartReceive(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->StartRender(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->StartRender(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ ViEAutotestFileObserver fileObserver;
+ int fileId;
+
+ ViEAutotestEncoderObserver* codecEncoderObserver = NULL;
+ ViEAutotestDecoderObserver* codecDecoderObserver = NULL;
+
+ //***************************************************************
+ // Engine ready. Wait for input
+ //***************************************************************
+
+ // Call started.
+ std::cout << std::endl;
+ std::cout << "Custom call started" << std::endl;
+ std::cout << std::endl << std::endl;
+
+ // Modify call or stop call.
+
+ std::cout << "Custom call in progress, would you like do?" << std::endl;
+ std::cout << " 0. Stop the call" << std::endl;
+ std::cout << " 1. Modify the call" << std::endl;
+ std::cout << "What do you want to do? "
+ << "Press enter for default (Stop the call): ";
+
+ std::getline(std::cin, str);
+ int selection = 0;
+ selection = atoi(str.c_str());
+
+ // Keep on modifying the call until user selects finish modify call.
+ bool modify_call = false;
+
+ while (selection == 1) {
+ std::cout << "Modify Custom Call" << std::endl;
+ std::cout << " 0. Finished modifying custom call" << std::endl;
+ std::cout << " 1. Change Video Send Codec" << std::endl;
+ std::cout << " 2. Change Video Send Size by Common Resolutions"
+ << std::endl;
+ std::cout << " 3. Change Video Send Size by Width & Height" << std::endl;
+ std::cout << " 4. Change Video Capture Device" << std::endl;
+ std::cout << " 5. Record Incoming Call" << std::endl;
+ std::cout << " 6. Record Outgoing Call" << std::endl;
+ std::cout << " 7. Play File on Video Channel"
+ << "(Assumes you recorded incoming & outgoing call)"
+ << std::endl;
+ std::cout << " 8. Change Video Protection Method" << std::endl;
+ std::cout << " 9. Toggle Encoder Observer" << std::endl;
+ std::cout << " 10. Toggle Decoder Observer" << std::endl;
+ std::cout << " 11. Print Call Information" << std::endl;
+ std::cout << " 12. Print Call Statistics" << std::endl;
+ std::cout << " 13. Toggle Image Scaling "
+ << "(Warning high CPU usage when enabled)"
+ << std::endl;
+ std::cout << "What do you want to do? ";
+ std::cout << "Press enter for default "
+ << "(Finished modifying custom call): ";
+
+ std::getline(std::cin, str);
+ int modify_selection = 0;
+ int file_selection = 0;
+
+ modify_selection = atoi(str.c_str());
+
+ switch (modify_selection) {
+ case 0:
+ std::cout << "Finished modifying custom call." << std::endl;
+ modify_call = false;
+ break;
+ case 1:
+ // Change video Codec
+ SetVideoCodecType(ptrViECodec, videoSendCodec);
+ SetVideoCodecSize(ptrViECodec, videoSendCodec);
+ SetVideoCodecBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMinBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMaxBitrate(ptrViECodec, videoSendCodec);
+ SetVideoCodecMaxFramerate(ptrViECodec, videoSendCodec);
+ SetVideoCodecTemporalLayer(videoSendCodec);
+ PrintCallInformation(ipAddress, deviceName,
+ uniqueId, videoSendCodec,
+ videoTxPort, videoRxPort,
+ audioCaptureDeviceName,
+ audioPlaybackDeviceName, audioCodec,
+ audioTxPort, audioRxPort, protectionMethod);
+ error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 2:
+ // Change Video codec size by common resolution
+ SetVideoCodecResolution(ptrViECodec, videoSendCodec);
+ PrintCallInformation(ipAddress, deviceName,
+ uniqueId, videoSendCodec,
+ videoTxPort, videoRxPort,
+ audioCaptureDeviceName,
+ audioPlaybackDeviceName, audioCodec,
+ audioTxPort, audioRxPort, protectionMethod);
+ error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 3:
+ // Change Video codec by size height and width
+ SetVideoCodecSize(ptrViECodec, videoSendCodec);
+ PrintCallInformation(ipAddress, deviceName,
+ uniqueId, videoSendCodec,
+ videoTxPort, videoRxPort,
+ audioCaptureDeviceName,
+ audioPlaybackDeviceName, audioCodec,
+ audioTxPort, audioRxPort, protectionMethod);
+ error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 4:
+ error = ptrViERender->StopRender(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViERender->RemoveRenderer(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->StopCapture(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->ReleaseCaptureDevice(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ memset(deviceName, 0, KMaxUniqueIdLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+ GetVideoDevice(ptrViEBase, ptrViECapture, deviceName, uniqueId);
+ captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId,
+ KMaxUniqueIdLength,
+ captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->ConnectCaptureDevice(captureId,
+ videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->StartCapture(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViERender->AddRenderer(
+ captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViERender->StartRender(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 5:
+ // Record the incoming call
+ std::cout << "Start Recording Incoming Video "
+ << DEFAULT_INCOMING_FILE_NAME << std::endl;
+ error = ptrViEFile->StartRecordIncomingVideo(
+ videoChannel, DEFAULT_INCOMING_FILE_NAME,
+ webrtc::NO_AUDIO, audioCodec, videoSendCodec);
+ std::cout << "Press enter to stop...";
+ std::getline(std::cin, str);
+ error = ptrViEFile->StopRecordIncomingVideo(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 6:
+ // Record the outgoing call
+ std::cout << "Start Recording Outgoing Video "
+ << DEFAULT_OUTGOING_FILE_NAME << std::endl;
+ error = ptrViEFile->StartRecordOutgoingVideo(
+ videoChannel, DEFAULT_OUTGOING_FILE_NAME,
+ webrtc::NO_AUDIO, audioCodec, videoSendCodec);
+ std::cout << "Press enter to stop...";
+ std::getline(std::cin, str);
+ error = ptrViEFile->StopRecordOutgoingVideo(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 7:
+ // Send the file on the videoChannel
+ file_selection = 0;
+ std::cout << "Available files to play" << std::endl;
+ std::cout << " 0. " << DEFAULT_INCOMING_FILE_NAME << std::endl;
+ std::cout << " 1. " << DEFAULT_OUTGOING_FILE_NAME << std::endl;
+ std::cout << "Press enter for default ("
+ << DEFAULT_INCOMING_FILE_NAME << "): ";
+ std::getline(std::cin, str);
+ file_selection = atoi(str.c_str());
+ // Disconnect the camera first
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ if (file_selection == 1)
+ error = ptrViEFile->StartPlayFile(DEFAULT_OUTGOING_FILE_NAME,
+ fileId, true);
+ else
+ error = ptrViEFile->StartPlayFile(DEFAULT_INCOMING_FILE_NAME,
+ fileId, true);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ ViETest::Log("Registering file observer");
+ error = ptrViEFile->RegisterObserver(fileId, fileObserver);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ std::cout << std::endl;
+ std::cout << "Start sending the file that is played in a loop "
+ << std::endl;
+ error = ptrViEFile->SendFileOnChannel(fileId, videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ std::cout << "Press enter to stop...";
+ std::getline(std::cin, str);
+ ViETest::Log("Stopped sending video on channel");
+ error = ptrViEFile->StopSendFileOnChannel(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ ViETest::Log("Stop playing the file.");
+ error = ptrViEFile->StopPlayFile(fileId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ error = ptrViECapture->ConnectCaptureDevice(captureId,
+ videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ error = ptrViEFile->DeregisterObserver(fileId, fileObserver);
+ numberOfErrors += ViETest::TestError(error == -1,
+ "ERROR:%d %s at line %d",
+ ptrViEBase->LastError(),
+ __FUNCTION__, __LINE__);
+ modify_call = true;
+ break;
+ case 8:
+ // Change the Video Protection
+ protectionMethod = GetVideoProtection();
+ SetVideoProtection(ptrViECodec, ptrViERtpRtcp,
+ videoChannel, protectionMethod);
+ modify_call = true;
+ break;
+ case 9:
+ // Toggle Encoder Observer
+ if (!codecEncoderObserver) {
+ std::cout << "Registering Encoder Observer" << std::endl;
+ codecEncoderObserver = new ViEAutotestEncoderObserver();
+ error = ptrViECodec->RegisterEncoderObserver(videoChannel,
+ *codecEncoderObserver);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ } else {
+ std::cout << "Deregistering Encoder Observer" << std::endl;
+ error = ptrViECodec->DeregisterEncoderObserver(videoChannel);
+ delete codecEncoderObserver;
+ codecEncoderObserver = NULL;
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ }
+ modify_call = true;
+ break;
+ case 10:
+ // Toggle Decoder Observer
+ if (!codecDecoderObserver) {
+ std::cout << "Registering Decoder Observer" << std::endl;
+ codecDecoderObserver = new ViEAutotestDecoderObserver();
+ error = ptrViECodec->RegisterDecoderObserver(videoChannel,
+ *codecDecoderObserver);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ } else {
+ std::cout << "Deregistering Decoder Observer" << std::endl;
+ error = ptrViECodec->DeregisterDecoderObserver(videoChannel);
+ delete codecDecoderObserver;
+ codecDecoderObserver = NULL;
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ }
+ modify_call = true;
+ break;
+ case 11:
+ // Print Call information
+ PrintCallInformation(ipAddress, deviceName,
+ uniqueId, videoSendCodec,
+ videoTxPort, videoRxPort,
+ audioCaptureDeviceName,
+ audioPlaybackDeviceName,
+ audioCodec, audioTxPort,
+ audioRxPort, protectionMethod);
+ PrintVideoStreamInformation(ptrViECodec,
+ videoChannel);
+ modify_call = true;
+ break;
+ case 12:
+ // Print Call statistics
+ PrintRTCCPStatistics(ptrViERtpRtcp, videoChannel,
+ kSendStatistic);
+ PrintRTCCPStatistics(ptrViERtpRtcp, videoChannel,
+ kReceivedStatistic);
+ PrintRTPStatistics(ptrViERtpRtcp, videoChannel);
+ PrintBandwidthUsage(ptrViERtpRtcp, videoChannel);
+ PrintCodecStatistics(ptrViECodec, videoChannel,
+ kSendStatistic);
+ PrintCodecStatistics(ptrViECodec, videoChannel,
+ kReceivedStatistic);
+ PrintGetDiscardedPackets(ptrViECodec, videoChannel);
+ modify_call = true;
+ break;
+ case 13:
+ isImageScaleEnabled = !isImageScaleEnabled;
+ ptrViECodec->SetImageScaleStatus(videoChannel, isImageScaleEnabled);
+ if (isImageScaleEnabled) {
+ std::cout << "Image Scale is now enabled" << std::endl;
+ } else {
+ std::cout << "Image Scale is now disabled" << std::endl;
+ }
+ modify_call = true;
+ break;
+ default:
+ // invalid selection, shows options menu again
+ std::cout << "Invalid selection. Select Again." << std::endl;
+ break;
+ }
+ // modify_call is false if user does not select one of the
+ // modify options
+ if (modify_call == false) {
+ selection = 0;
+ }
+ }
+ // Stop the Call
+ std::cout << "Press enter to stop...";
+ std::getline(std::cin, str);
+ //***************************************************************
+ // Testing finished. Tear down Voice and Video Engine.
+ //***************************************************************
+ // Tear down the VE first.
+ error = ptrVEBase->StopReceive(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEBase->StopPlayout(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrVEBase->DeleteChannel(audioChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ // Now tear down the ViE engine.
+ error = ptrViEBase->DisconnectAudioChannel(videoChannel);
+
+ // If Encoder/Decoder Observer is running, delete them.
+ if (codecEncoderObserver) {
+ error = ptrViECodec->DeregisterEncoderObserver(videoChannel);
+ delete codecEncoderObserver;
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ }
+ if (codecDecoderObserver) {
+ error = ptrViECodec->DeregisterDecoderObserver(videoChannel);
+ delete codecDecoderObserver;
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ }
+
+ error = ptrViEBase->StopReceive(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->StopSend(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->StopRender(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->StopRender(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->RemoveRenderer(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViERender->RemoveRenderer(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECapture->StopCapture(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViECapture->ReleaseCaptureDevice(captureId);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ error = ptrViEBase->DeleteChannel(videoChannel);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ int remainingInterfaces = 0;
+ remainingInterfaces = ptrViEFile->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ remainingInterfaces = ptrViECodec->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ remainingInterfaces = ptrViECapture->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ remainingInterfaces = ptrViERtpRtcp->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ remainingInterfaces = ptrViERender->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ remainingInterfaces = ptrViENetwork->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ remainingInterfaces = ptrViEBase->Release();
+ numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ bool deleted = webrtc::VideoEngine::Delete(ptrViE);
+ numberOfErrors += ViETest::TestError(deleted == true,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ ViETest::Log(" ");
+ ViETest::Log(" ViE Autotest Custom Call Started");
+ ViETest::Log("========================================");
+ ViETest::Log(" ");
+ }
+ return numberOfErrors;
+}
+
+bool GetVideoDevice(webrtc::ViEBase* ptrViEBase,
+ webrtc::ViECapture* ptrViECapture,
+ char* captureDeviceName,
+ char* captureDeviceUniqueId) {
+ int error = 0;
+ int numberOfErrors = 0;
+ int captureDeviceIndex = 0;
+ std::string str;
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ char deviceName[KMaxDeviceNameLength];
+ char uniqueId[KMaxUniqueIdLength];
+
+ while (1) {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ std::cout << std::endl;
+ std::cout << "Available video capture devices:" << std::endl;
+ int captureIdx = 0;
+ for (captureIdx = 0;
+ captureIdx < ptrViECapture->NumberOfCaptureDevices();
+ captureIdx++) {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength,
+ uniqueId,
+ KMaxUniqueIdLength);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << " " << captureIdx+1 << ". " << deviceName
+ << "/" << uniqueId
+ << std::endl;
+ }
+ // Get the devName of the default (or first) camera for display.
+ error = ptrViECapture->GetCaptureDevice(0, deviceName,
+ KMaxDeviceNameLength,
+ uniqueId,
+ KMaxUniqueIdLength);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ std::cout << "Choose a video capture device. Press enter for default ("
+ << deviceName << "/" << uniqueId << "): ";
+ std::getline(std::cin, str);
+ captureDeviceIndex = atoi(str.c_str());
+
+ if (captureDeviceIndex == 0) {
+ // Use the default (or first) camera.
+ error = ptrViECapture->GetCaptureDevice(0, deviceName,
+ KMaxDeviceNameLength,
+ uniqueId,
+ KMaxUniqueIdLength);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ strcpy(captureDeviceUniqueId, uniqueId);
+ strcpy(captureDeviceName, deviceName);
+ return true;
+ } else if (captureDeviceIndex < 0
+ || (captureDeviceIndex >
+ (int)ptrViECapture->NumberOfCaptureDevices())) {
+ // invalid selection
+ continue;
+ } else {
+ error = ptrViECapture->GetCaptureDevice(captureDeviceIndex - 1,
+ deviceName,
+ KMaxDeviceNameLength,
+ uniqueId,
+ KMaxUniqueIdLength);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ strcpy(captureDeviceUniqueId, uniqueId);
+ strcpy(captureDeviceName, deviceName);
+ return true;
+ }
+ }
+}
+
+bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
+ webrtc::VoEHardware* ptrVEHardware,
+ char* recordingDeviceName,
+ int& recordingDeviceIndex,
+ char* playbackDeviceName,
+ int& playbackDeviceIndex) {
+ int error = 0;
+ int numberOfErrors = 0;
+ std::string str;
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 128;
+ char recordingDeviceUniqueName[KMaxDeviceNameLength];
+ char playbackDeviceUniqueName[KMaxUniqueIdLength];
+
+ int numberOfRecordingDevices = -1;
+ error = ptrVEHardware->GetNumOfRecordingDevices(numberOfRecordingDevices);
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ while (1) {
+ recordingDeviceIndex = -1;
+ std::cout << std::endl;
+ std::cout << "Available audio capture devices:" << std::endl;
+ int captureIdx = 0;
+
+ for (captureIdx = 0; captureIdx < numberOfRecordingDevices;
+ captureIdx++) {
+ memset(recordingDeviceName, 0, KMaxDeviceNameLength);
+ memset(recordingDeviceUniqueName, 0, KMaxDeviceNameLength);
+ error = ptrVEHardware->GetRecordingDeviceName(
+ captureIdx, recordingDeviceName, recordingDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << " " << captureIdx+1 << ". " << recordingDeviceName
+ << std::endl;
+ }
+
+ std::cout << "Choose an audio capture device. Press enter for default("
+ << recordingDeviceName << "): ";
+ std::getline(std::cin, str);
+ int captureDeviceIndex = atoi(str.c_str());
+
+ if (captureDeviceIndex == 0) {
+ // Use the default (or first) recording device.
+ recordingDeviceIndex = 0;
+ error = ptrVEHardware->GetRecordingDeviceName(
+ recordingDeviceIndex, recordingDeviceName,
+ recordingDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ } else if (captureDeviceIndex < 0
+ || captureDeviceIndex > numberOfRecordingDevices) {
+ // invalid selection
+ continue;
+ } else {
+ recordingDeviceIndex = captureDeviceIndex - 1;
+ error = ptrVEHardware->GetRecordingDeviceName(
+ recordingDeviceIndex, recordingDeviceName,
+ recordingDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+ }
+
+ int numberOfPlaybackDevices = -1;
+ error = ptrVEHardware->GetNumOfPlayoutDevices(numberOfPlaybackDevices);
+ numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ while (1) {
+ playbackDeviceIndex = -1;
+ std::cout << std::endl;
+ std::cout << "Available audio playout devices:" << std::endl;
+ int captureIdx = 0;
+
+ for (captureIdx = 0; captureIdx < numberOfPlaybackDevices;
+ captureIdx++) {
+ memset(playbackDeviceName, 0, KMaxDeviceNameLength);
+ memset(playbackDeviceUniqueName, 0, KMaxDeviceNameLength);
+ error = ptrVEHardware->GetPlayoutDeviceName(
+ captureIdx, playbackDeviceName, playbackDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << " " << captureIdx+1 << ". " << playbackDeviceName
+ << std::endl;
+ }
+
+ std::cout <<
+ "Choose an audio playback device. Press enter for default ("
+ << playbackDeviceName << "): ";
+ std::getline(std::cin, str);
+ int captureDeviceIndex = atoi(str.c_str());
+
+ if (captureDeviceIndex == 0) {
+ // Use the default (or first) playout device.
+ playbackDeviceIndex = 0;
+ error = ptrVEHardware->GetPlayoutDeviceName(
+ playbackDeviceIndex, playbackDeviceName,
+ playbackDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ return true;
+ } else if (captureDeviceIndex < 0
+ || captureDeviceIndex > numberOfPlaybackDevices) {
+ // invalid selection
+ continue;
+ } else {
+ playbackDeviceIndex = captureDeviceIndex - 1;
+ error = ptrVEHardware->GetPlayoutDeviceName(
+ playbackDeviceIndex, playbackDeviceName,
+ playbackDeviceUniqueName);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ return true;
+ }
+ }
+}
+
+// General helper functions.
+
+bool GetIPAddress(char* iIP) {
+ char oIP[16] = DEFAULT_SEND_IP;
+ std::string str;
+
+ while (1) {
+ std::cout << std::endl;
+ std::cout << "Enter destination IP. Press enter for default ("
+ << oIP << "): ";
+ std::getline(std::cin, str);
+
+ if (str.compare("") == 0) {
+ // use default value;
+ strcpy(iIP, oIP);
+ return true;
+ }
+ if (ValidateIP(str) == false) {
+ std::cout << "Invalid entry. Try again." << std::endl;
+ continue;
+ }
+ // done. Copy std::string to c_string and return
+ strcpy(iIP, str.c_str());
+ return true;
+ }
+ assert(false);
+ return false;
+}
+
+bool ValidateIP(std::string iStr) {
+ if (0 == iStr.compare("")) {
+ return false;
+ }
+ return true;
+}
+
+// Video settings functions.
+
+bool SetVideoPorts(int* txPort, int* rxPort) {
+ std::string str;
+ int port = 0;
+
+ // set to default values
+ *txPort = DEFAULT_VIDEO_PORT;
+ *rxPort = DEFAULT_VIDEO_PORT;
+
+ while (1) {
+ std::cout << "Enter video send port. Press enter for default ("
+ << *txPort << "): ";
+ std::getline(std::cin, str);
+ port = atoi(str.c_str());
+
+ if (port == 0) {
+ // default value
+ break;
+ } else {
+ // user selection
+ if (port <= 0 || port > 63556) {
+ // invalid selection
+ continue;
+ } else {
+ *txPort = port;
+ break; // move on to rxport
+ }
+ }
+ }
+
+ while (1) {
+ std::cout << "Enter video receive port. Press enter for default ("
+ << *rxPort << "): ";
+ std::getline(std::cin, str);
+ port = atoi(str.c_str());
+
+ if (port == 0) {
+ // default value
+ return true;
+ } else {
+ // user selection
+ if (port <= 0 || port > 63556) {
+ // invalid selection
+ continue;
+ } else {
+ *rxPort = port;
+ return true;
+ }
+ }
+ }
+ assert(false);
+ return false;
+}
+
+// Audio settings functions.
+
+bool GetAudioPorts(int* txPort, int* rxPort) {
+ int port = 0;
+ std::string str;
+
+ // set to default values
+ *txPort = DEFAULT_AUDIO_PORT;
+ *rxPort = DEFAULT_AUDIO_PORT;
+
+ while (1) {
+ std::cout << "Enter audio send port. Press enter for default ("
+ << *txPort << "): ";
+ std::getline(std::cin, str);
+ port = atoi(str.c_str());
+
+ if (port == 0) {
+ // default value
+ break;
+ } else {
+ // user selection
+ if (port <= 0 || port > 63556) {
+ // invalid selection
+ continue;
+ } else {
+ *txPort = port;
+ break; // move on to rxport
+ }
+ }
+ }
+
+ while (1) {
+ std::cout << "Enter audio receive port. Press enter for default ("
+ << *rxPort << "): ";
+ std::getline(std::cin, str);
+ port = atoi(str.c_str());
+
+ if (port == 0) {
+ // default value
+ return true;
+ } else {
+ // user selection
+ if (port <= 0 || port > 63556) {
+ // invalid selection
+ continue;
+ } else {
+ *rxPort = port;
+ return true;
+ }
+ }
+ }
+ assert(false);
+ return false;
+}
+
+bool GetAudioCodec(webrtc::VoECodec* ptrVeCodec,
+ webrtc::CodecInst& audioCodec) {
+ int error = 0;
+ int numberOfErrors = 0;
+ int codecSelection = 0;
+ std::string str;
+ memset(&audioCodec, 0, sizeof(webrtc::CodecInst));
+
+ while (1) {
+ std::cout << std::endl;
+ std::cout << "Available audio codecs:" << std::endl;
+ int codecIdx = 0;
+ int defaultCodecIdx = 0;
+ for (codecIdx = 0; codecIdx < ptrVeCodec->NumOfCodecs(); codecIdx++) {
+ error = ptrVeCodec->GetCodec(codecIdx, audioCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+
+ // test for default codec index
+ if (strcmp(audioCodec.plname, DEFAULT_AUDIO_CODEC) == 0) {
+ defaultCodecIdx = codecIdx;
+ }
+ std::cout << " " << codecIdx+1 << ". " << audioCodec.plname
+ << std::endl;
+ }
+ std::cout << std::endl;
+ std::cout << "Choose audio codec. Press enter for default ("
+ << DEFAULT_AUDIO_CODEC << "): ";
+ std::getline(std::cin, str);
+ codecSelection = atoi(str.c_str());
+
+ if (codecSelection == 0) {
+ // use default
+ error = ptrVeCodec->GetCodec(defaultCodecIdx, audioCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ return true;
+ } else {
+ // user selection
+ codecSelection = atoi(str.c_str())-1;
+ error = ptrVeCodec->GetCodec(codecSelection, audioCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ if (error != 0) {
+ std::cout << "ERROR: Code = " << error << " Invalid selection"
+ << std::endl;
+ continue;
+ }
+ return true;
+ }
+ }
+ assert(false);
+ return false;
+}
+
+void PrintCallInformation(char* IP, char* videoCaptureDeviceName,
+ char* videoCaptureUniqueId,
+ webrtc::VideoCodec videoCodec,
+ int videoTxPort, int videoRxPort,
+ char* audioCaptureDeviceName,
+ char* audioPlaybackDeviceName,
+ webrtc::CodecInst audioCodec,
+ int audioTxPort, int audioRxPort,
+ int protectionMethod) {
+ std::string str;
+
+ std::cout << "************************************************"
+ << std::endl;
+ std::cout << "The call has the following settings: " << std::endl;
+ std::cout << "\tIP: " << IP << std::endl;
+ std::cout << "\tVideo Capture Device: " << videoCaptureDeviceName
+ << std::endl;
+ std::cout << "\t\tName: " << videoCaptureDeviceName << std::endl;
+ std::cout << "\t\tUniqueId: " << videoCaptureUniqueId << std::endl;
+ PrintVideoCodec(videoCodec);
+ std::cout << "\t Video Tx Port: " << videoTxPort << std::endl;
+ std::cout << "\t Video Rx Port: " << videoRxPort << std::endl;
+ std::cout << "\t Video Protection Method: " << protectionMethod
+ << std::endl;
+ std::cout << "\tAudio Capture Device: " << audioCaptureDeviceName
+ << std::endl;
+ std::cout << "\tAudio Playback Device: " << audioPlaybackDeviceName
+ << std::endl;
+ std::cout << "\tAudio Codec: " << std::endl;
+ std::cout << "\t\tplname: " << audioCodec.plname << std::endl;
+ std::cout << "\t\tpltype: " << (int)audioCodec.pltype << std::endl;
+ std::cout << "\t Audio Tx Port: " << audioTxPort << std::endl;
+ std::cout << "\t Audio Rx Port: " << audioRxPort << std::endl;
+ std::cout << "************************************************"
+ << std::endl;
+}
+
+bool SetVideoCodecType(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ int error = 0;
+ int numberOfErrors = 0;
+ int codecSelection = 0;
+ std::string str;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+
+ bool exitLoop=false;
+ while (!exitLoop) {
+ std::cout << std::endl;
+ std::cout << "Available video codecs:" << std::endl;
+ int codecIdx = 0;
+ int defaultCodecIdx = 0;
+ // Print out all the codecs available to set Codec to.
+ for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++) {
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ // test for default codec index
+ if (strcmp(videoCodec.plName, DEFAULT_VIDEO_CODEC) == 0) {
+ defaultCodecIdx = codecIdx;
+ }
+ std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
+ << std::endl;
+ }
+ std::cout << std::endl;
+ std::cout << "Choose video codec. Press enter for default ("
+ << DEFAULT_VIDEO_CODEC << "): ";
+ std::getline(std::cin, str);
+ codecSelection = atoi(str.c_str());
+ if (codecSelection == 0) {
+ // use default
+ error = ptrViECodec->GetCodec(defaultCodecIdx, videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ exitLoop=true;
+ } else {
+ // user selection
+ codecSelection = atoi(str.c_str())-1;
+ error = ptrViECodec->GetCodec(codecSelection, videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ if (error != 0) {
+ std::cout << "ERROR: Code=" << error << " Invalid selection"
+ << std::endl;
+ continue;
+ }
+ exitLoop=true;
+ }
+ }
+ if (videoCodec.codecType == webrtc::kVideoCodecI420) {
+ videoCodec.width = 176;
+ videoCodec.height = 144;
+ }
+ return true;
+}
+
+bool SetVideoCodecResolution(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ std::string str;
+ int sizeOption = 5;
+
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
+ std::cout << std::endl;
+ std::cout << "Available Common Resolutions : " << std::endl;
+ std::cout << " 1. SQCIF (128X96) " << std::endl;
+ std::cout << " 2. QQVGA (160X120) " << std::endl;
+ std::cout << " 3. QCIF (176X144) " << std::endl;
+ std::cout << " 4. CIF (352X288) " << std::endl;
+ std::cout << " 5. VGA (640X480) " << std::endl;
+ std::cout << " 6. WVGA (800x480) " << std::endl;
+ std::cout << " 7. 4CIF (704X576) " << std::endl;
+ std::cout << " 8. SVGA (800X600) " << std::endl;
+ std::cout << " 9. HD (1280X720) " << std::endl;
+ std::cout << " 10. XGA (1024x768) " << std::endl;
+ std::cout << "Enter frame size option: " << std::endl;
+
+ std::getline(std::cin, str);
+ sizeOption = atoi(str.c_str());
+
+ switch (sizeOption) {
+ case 1:
+ videoCodec.width = 128;
+ videoCodec.height = 96;
+ break;
+ case 2:
+ videoCodec.width = 160;
+ videoCodec.height = 120;
+ break;
+ case 3:
+ videoCodec.width = 176;
+ videoCodec.height = 144;
+ break;
+ case 4:
+ videoCodec.width = 352;
+ videoCodec.height = 288;
+ break;
+ case 5:
+ videoCodec.width = 640;
+ videoCodec.height = 480;
+ break;
+ case 6:
+ videoCodec.width = 800;
+ videoCodec.height = 480;
+ break;
+ case 7:
+ videoCodec.width = 704;
+ videoCodec.height = 576;
+ break;
+ case 8:
+ videoCodec.width = 800;
+ videoCodec.height = 600;
+ break;
+ case 9:
+ videoCodec.width = 1280;
+ videoCodec.height = 720;
+ break;
+ case 10:
+ videoCodec.width = 1024;
+ videoCodec.height = 768;
+ break;
+ }
+ } else {
+ std::cout << "Can Only change codec size if it's VP8" << std::endl;
+ }
+ return true;
+}
+
+bool SetVideoCodecSize(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
+ std::string str;
+ videoCodec.width = DEFAULT_VIDEO_CODEC_WIDTH;
+ videoCodec.height = DEFAULT_VIDEO_CODEC_HEIGHT;
+ std::cout << "Choose video width. Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_WIDTH << "): ";
+ std::getline(std::cin, str);
+ int sizeSelection = atoi(str.c_str());
+ if (sizeSelection!=0) {
+ videoCodec.width=sizeSelection;
+ }
+ std::cout << "Choose video height. Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_HEIGHT << "): ";
+ std::getline(std::cin, str);
+ sizeSelection = atoi(str.c_str());
+ if (sizeSelection!=0) {
+ videoCodec.height=sizeSelection;
+ }
+ } else {
+ std::cout << "Can Only change codec size if it's VP8" << std::endl;
+ }
+ return true;
+}
+
+bool SetVideoCodecBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose start rate (in kbps). Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_BITRATE << "): ";
+ std::getline(std::cin, str);
+ int startRate = atoi(str.c_str());
+ videoCodec.startBitrate = DEFAULT_VIDEO_CODEC_BITRATE;
+ if (startRate != 0) {
+ videoCodec.startBitrate = startRate;
+ }
+ return true;
+}
+
+bool SetVideoCodecMaxBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose max bitrate (in kbps). Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_MAX_BITRATE << "): ";
+ std::getline(std::cin, str);
+ int maxRate = atoi(str.c_str());
+ videoCodec.maxBitrate = DEFAULT_VIDEO_CODEC_MAX_BITRATE;
+ if (maxRate != 0) {
+ videoCodec.maxBitrate = maxRate;
+ }
+ return true;
+}
+
+bool SetVideoCodecMinBitrate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose min bitrate (in fps). Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_MIN_BITRATE << "): ";
+ std::getline(std::cin, str);
+ char minBitRate = atoi(str.c_str());
+ videoCodec.minBitrate = DEFAULT_VIDEO_CODEC_MIN_BITRATE;
+ if (minBitRate != 0) {
+ videoCodec.minBitrate = minBitRate;
+ }
+ return true;
+}
+
+bool SetVideoCodecMaxFramerate(webrtc::ViECodec* ptrViECodec,
+ webrtc::VideoCodec& videoCodec) {
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose max framerate (in fps). Press enter for default ("
+ << DEFAULT_VIDEO_CODEC_MAX_FRAMERATE << "): ";
+ std::getline(std::cin, str);
+ char maxFrameRate = atoi(str.c_str());
+ videoCodec.maxFramerate = DEFAULT_VIDEO_CODEC_MAX_FRAMERATE;
+ if (maxFrameRate != 0) {
+ videoCodec.maxFramerate = maxFrameRate;
+ }
+ return true;
+}
+
+bool SetVideoCodecTemporalLayer(webrtc::VideoCodec& videoCodec) {
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose number of temporal layers (1 to 4). "
+ << "Press enter for default ("
+ << DEFAULT_TEMPORAL_LAYER << "): ";
+ std::getline(std::cin, str);
+ char numTemporalLayers = atoi(str.c_str());
+ videoCodec.codecSpecific.VP8.numberOfTemporalLayers
+ = DEFAULT_TEMPORAL_LAYER;
+ if(numTemporalLayers != 0) {
+ videoCodec.codecSpecific.VP8.numberOfTemporalLayers
+ = numTemporalLayers;
+ }
+ }
+ return true;
+}
+// GetVideoProtection only prints the prompt to get a number
+// that SetVideoProtection method uses
+// 0 = None
+// 1 = FEC
+// 2 = NACK
+// 3 = NACK + FEC (aka Hybrid)
+// Default = DEFAULT_VIDEO_PROTECTION METHOD
+int GetVideoProtection() {
+ int protectionMethod = DEFAULT_VIDEO_PROTECTION_METHOD;
+
+ std::cout << "Available Video Protection Method." << std::endl;
+ std::cout << " 0. None" << std::endl;
+ std::cout << " 1. FEC" << std::endl;
+ std::cout << " 2. NACK" << std::endl;
+ std::cout << " 3. NACK+FEC" << std::endl;
+ std::cout << "Enter Video Protection Method. "
+ << "Press enter for default (" << protectionMethod << "):"
+ << std::endl;
+ std::string method;
+ std::getline(std::cin, method);
+ protectionMethod = atoi(method.c_str());
+
+ return protectionMethod;
+}
+
+bool SetVideoProtection(webrtc::ViECodec* ptrViECodec,
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel, int protectionMethod) {
+ int error = 0;
+ int numberOfErrors = 0;
+ webrtc::VideoCodec videoCodec;
+
+ memset((void*)&videoCodec, 0, sizeof(webrtc::VideoCodec));
+
+ // Set all video protection to false initially
+ error = ptrViERtpRtcp->SetHybridNACKFECStatus(videoChannel, false,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViERtpRtcp->SetFECStatus(videoChannel, false,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ error = ptrViERtpRtcp->SetNACKStatus(videoChannel, false);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ // Set video protection for FEC, NACK or Hybrid
+ switch (protectionMethod) {
+ case 0: // None
+ // No protection selected, all protection already at false
+ std::cout << "Call using None protection Method"
+ << std::endl;
+ break;
+ case 1: // FEC only
+ std::cout << "Call using FEC protection Method"
+ << std::endl;
+ error = ptrViERtpRtcp->SetFECStatus(videoChannel, true,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ case 2: // NACK only
+ std::cout << "Call using NACK protection Method"
+ << std::endl;
+ error = ptrViERtpRtcp->SetNACKStatus(videoChannel, true);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ case 3: // Hybrid NACK and FEC
+ std::cout << "Call using Hybrid NACK and FEC protection Method"
+ << std::endl;
+ error = ptrViERtpRtcp->SetHybridNACKFECStatus(videoChannel, true,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+
+ // Set receive codecs for FEC and hybrid NACK/FEC
+ if (protectionMethod == 1 || protectionMethod == 3) {
+ // RED
+ error = ptrViECodec->GetCodec(ptrViECodec->NumberOfCodecs() - 2,
+ videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << "RED Codec Information:" << std::endl;
+ PrintVideoCodec(videoCodec);
+ // ULPFEC
+ error = ptrViECodec->GetCodec(ptrViECodec->NumberOfCodecs() - 1,
+ videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << "ULPFEC Codec Information:" << std::endl;
+ PrintVideoCodec(videoCodec);
+ }
+
+
+
+ return true;
+}
+
+bool GetBitrateSignaling() {
+ std::cout << std::endl;
+ std::cout << "Available bitrate signaling methods." << std::endl;
+ std::cout << " 0. REMB" << std::endl;
+ std::cout << " 1. TMMBR" << std::endl;
+ std::cout << "Enter bitrate signaling methods. "
+ << "Press enter for default (REMB): " << std::endl;
+ std::string method;
+ std::getline(std::cin, method);
+ if (atoi(method.c_str()) == 1) {
+ return false;
+ }
+ return true;
+}
+
+void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel, StatisticsType statType) {
+ int error = 0;
+ int numberOfErrors =0;
+ unsigned short fractionLost = 0;
+ unsigned int cumulativeLost = 0;
+ unsigned int extendedMax = 0;
+ unsigned int jitter = 0;
+ int rttMS = 0;
+
+ switch (statType) {
+ case kReceivedStatistic:
+ std::cout << "RTCP Received statistics"
+ << std::endl;
+ // Get and print the Received RTCP Statistics
+ error = ptrViERtpRtcp->GetReceivedRTCPStatistics(videoChannel,
+ fractionLost,
+ cumulativeLost,
+ extendedMax,
+ jitter, rttMS);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ case kSendStatistic:
+ std::cout << "RTCP Sent statistics"
+ << std::endl;
+ // Get and print the Sent RTCP Statistics
+ error = ptrViERtpRtcp->GetSentRTCPStatistics(videoChannel, fractionLost,
+ cumulativeLost, extendedMax,
+ jitter, rttMS);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+ std::cout << "\tRTCP fraction of lost packets: "
+ << fractionLost << std::endl;
+ std::cout << "\tRTCP cumulative number of lost packets: "
+ << cumulativeLost << std::endl;
+ std::cout << "\tRTCP max received sequence number "
+ << extendedMax << std::endl;
+ std::cout << "\tRTCP jitter: "
+ << jitter << std::endl;
+ std::cout << "\tRTCP round trip (ms): "
+ << rttMS<< std::endl;
+}
+
+void PrintRTPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel) {
+ int error = 0;
+ int numberOfErrors =0;
+ unsigned int bytesSent = 0;
+ unsigned int packetsSent= 0;
+ unsigned int bytesReceived = 0;
+ unsigned int packetsReceived = 0;
+
+ std::cout << "RTP statistics"
+ << std::endl;
+
+ // Get and print the RTP Statistics
+ error = ptrViERtpRtcp->GetRTPStatistics(videoChannel, bytesSent, packetsSent,
+ bytesReceived, packetsReceived);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << "\tRTP bytes sent: "
+ << bytesSent << std::endl;
+ std::cout << "\tRTP packets sent: "
+ << packetsSent << std::endl;
+ std::cout << "\tRTP bytes received: "
+ << bytesReceived << std::endl;
+ std::cout << "\tRTP packets received: "
+ << packetsReceived << std::endl;
+}
+
+void PrintBandwidthUsage(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
+ int videoChannel) {
+ int error = 0;
+ int numberOfErrors = 0;
+ unsigned int totalBitrateSent = 0;
+ unsigned int videoBitrateSent = 0;
+ unsigned int fecBitrateSent = 0;
+ unsigned int nackBitrateSent = 0;
+ double percentageFEC = 0;
+ double percentageNACK = 0;
+
+ std::cout << "Bandwidth Usage"
+ << std::endl;
+
+ // Get and print Bandwidth usage
+ error = ptrViERtpRtcp->GetBandwidthUsage(videoChannel, totalBitrateSent,
+ videoBitrateSent, fecBitrateSent,
+ nackBitrateSent);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ std::cout << "\tTotal bitrate sent (Kbit/s): "
+ << totalBitrateSent << std::endl;
+ std::cout << "\tVideo bitrate sent (Kbit/s): "
+ << videoBitrateSent << std::endl;
+ std::cout << "\tFEC bitrate sent (Kbit/s): "
+ << fecBitrateSent << std::endl;
+ percentageFEC = ((double)fecBitrateSent/(double)totalBitrateSent) * 100;
+ std::cout << "\tPercentage FEC bitrate sent from total bitrate: "
+ << percentageFEC << std::endl;
+ std::cout << "\tNACK bitrate sent (Kbit/s): "
+ << nackBitrateSent << std::endl;
+ percentageNACK = ((double)nackBitrateSent/(double)totalBitrateSent) * 100;
+ std::cout << "\tPercentage NACK bitrate sent from total bitrate: "
+ << percentageNACK << std::endl;
+}
+
+void PrintCodecStatistics(webrtc::ViECodec* ptrViECodec, int videoChannel,
+ StatisticsType statType) {
+ int error = 0;
+ int numberOfErrors = 0;
+ unsigned int keyFrames = 0;
+ unsigned int deltaFrames = 0;
+ switch(statType) {
+ case kReceivedStatistic:
+ std::cout << "Codec Receive statistics"
+ << std::endl;
+ // Get and print the Receive Codec Statistics
+ error = ptrViECodec->GetReceiveCodecStastistics(videoChannel, keyFrames,
+ deltaFrames);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ case kSendStatistic:
+ std::cout << "Codec Send statistics"
+ << std::endl;
+ // Get and print the Send Codec Statistics
+ error = ptrViECodec->GetSendCodecStastistics(videoChannel, keyFrames,
+ deltaFrames);
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);
+ break;
+ }
+ std::cout << "\tNumber of encoded key frames: "
+ << keyFrames << std::endl;
+ std::cout << "\tNumber of encoded delta frames: "
+ << deltaFrames << std::endl;
+}
+
+void PrintGetDiscardedPackets(webrtc::ViECodec* ptrViECodec, int videoChannel) {
+ std::cout << "Discarded Packets"
+ << std::endl;
+ int discardedPackets = 0;
+ discardedPackets = ptrViECodec->GetDiscardedPackets(videoChannel);
+ std::cout << "\tNumber of discarded packets: "
+ << discardedPackets << std::endl;
+}
+
+void PrintVideoStreamInformation(webrtc::ViECodec* ptrViECodec,
+ int videoChannel) {
+ webrtc::VideoCodec outgoingCodec;
+ webrtc::VideoCodec incomingCodec;
+
+ memset((void*)&outgoingCodec, 0, sizeof(webrtc::VideoCodec));
+ memset((void*)&incomingCodec, 0, sizeof(webrtc::VideoCodec));
+
+ ptrViECodec->GetSendCodec(videoChannel, outgoingCodec);
+ ptrViECodec->GetReceiveCodec(videoChannel, incomingCodec);
+
+ std::cout << "************************************************"
+ << std::endl;
+ std::cout << "ChannelId: " << videoChannel << std::endl;
+ std::cout << "Outgoing Stream information:" <<std::endl;
+ PrintVideoCodec(outgoingCodec);
+ std::cout << "Incoming Stream information:" <<std::endl;
+ PrintVideoCodec(incomingCodec);
+ std::cout << "************************************************"
+ << std::endl;
+}
+
+void PrintVideoCodec(webrtc::VideoCodec videoCodec) {
+ std::cout << "\t\tplName: " << videoCodec.plName << std::endl;
+ std::cout << "\t\tplType: " << (int)videoCodec.plType << std::endl;
+ std::cout << "\t\twidth: " << videoCodec.width << std::endl;
+ std::cout << "\t\theight: " << videoCodec.height << std::endl;
+ std::cout << "\t\tstartBitrate: " << videoCodec.startBitrate
+ << std::endl;
+ std::cout << "\t\tminBitrate: " << videoCodec.minBitrate
+ << std::endl;
+ std::cout << "\t\tmaxBitrate: " << videoCodec.maxBitrate
+ << std::endl;
+ std::cout << "\t\tmaxFramerate: " << (int)videoCodec.maxFramerate
+ << std::endl;
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
+ std::cout << "\t\tVP8 Temporal Layer: "
+ << (int)videoCodec.codecSpecific.VP8.numberOfTemporalLayers
+ << std::endl;
+ }
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc
new file mode 100644
index 0000000..0c06f32
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc
@@ -0,0 +1,567 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_encryption.cc
+//
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "engine_configurations.h"
+
+#include "tb_capture_device.h"
+#include "tb_external_transport.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+
+class ViEAutotestEncryption: public webrtc::Encryption
+{
+public:
+ ViEAutotestEncryption()
+ {
+ }
+ ~ViEAutotestEncryption()
+ {
+ }
+
+ virtual void encrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int* bytes_out)
+ {
+ for (int i = 0; i < bytes_in; i++)
+ {
+ out_data[i] = ~in_data[i];
+ }
+ *bytes_out = bytes_in + 2;
+ }
+
+ virtual void decrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int* bytes_out)
+ {
+ for (int i = 0; i < bytes_in - 2; i++)
+ {
+ out_data[i] = ~in_data[i];
+ }
+ *bytes_out = bytes_in - 2;
+ }
+
+ virtual void encrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int* bytes_out)
+ {
+ for (int i = 0; i < bytes_in; i++)
+ {
+ out_data[i] = ~in_data[i];
+ }
+ *bytes_out = bytes_in + 2;
+ }
+
+ virtual void decrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int* bytes_out)
+ {
+ for (int i = 0; i < bytes_in - 2; i++)
+ {
+ out_data[i] = ~in_data[i];
+ }
+ *bytes_out = bytes_in - 2;
+ }
+};
+
+void ViEAutoTest::ViEEncryptionStandardTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ // Create VIE
+ TbInterfaces ViE("ViEEncryptionStandardTest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ tbChannel.StartReceive();
+
+ tbChannel.StartSend();
+
+ RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
+
+#ifdef WEBRTC_SRTP
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ //
+ // SRTP
+ //
+ unsigned char srtpKey1[30] =
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
+ 4, 5, 6, 7, 8, 9};
+
+ // Encryption only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
+ ViETest::Log("SRTP encryption only");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Authentication only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kAuthentication, srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kAuthentication, srtpKey1));
+
+ ViETest::Log("SRTP authentication only");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Full protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey1));
+
+ ViETest::Log("SRTP full protection");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+#endif // WEBRTC_SRTP
+
+ //
+ // External encryption
+ //
+ ViEAutotestEncryption testEncryption;
+ // Note(qhogpat): StartSend fails, not sure if this is intentional.
+ EXPECT_NE(0, ViE.base->StartSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
+ tbChannel.videoChannel, testEncryption));
+ ViETest::Log(
+ "External encryption/decryption added, you should still see video");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
+ tbChannel.videoChannel));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
+
+void ViEAutoTest::ViEEncryptionExtendedTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ // Create VIE
+ TbInterfaces ViE("ViEEncryptionExtendedTest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ tbChannel.StartReceive();
+ tbChannel.StartSend();
+
+ RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+#ifdef WEBRTC_SRTP
+
+ //
+ // SRTP
+ //
+ unsigned char srtpKey1[30] =
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
+ 4, 5, 6, 7, 8, 9};
+ unsigned char srtpKey2[30] =
+ { 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, 9, 8, 7, 6,
+ 5, 4, 3, 2, 1, 0};
+ // NULL
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
+ webrtc::kNoProtection, srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
+ webrtc::kNoProtection, srtpKey1));
+
+ ViETest::Log("SRTP NULL encryption/authentication");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Encryption only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
+
+ ViETest::Log("SRTP encryption only");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Authentication only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kAuthentication, srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kAuthentication, srtpKey1));
+
+ ViETest::Log("SRTP authentication only");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Full protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey1));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey1));
+
+ ViETest::Log("SRTP full protection");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Change receive key, but not send key...
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey2));
+
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey1));
+
+ ViETest::Log(
+ "\nSRTP receive key changed, you should not see any remote images");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ // Change send key too
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey2));
+
+ ViETest::Log("\nSRTP send key changed too, you should see remote video "
+ "again with some decoding artefacts at start");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+
+ // Disable receive, keep send
+ ViETest::Log("SRTP receive disabled , you shouldn't see any video");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+#endif //WEBRTC_SRTP
+ //
+ // External encryption
+ //
+ ViEAutotestEncryption testEncryption;
+ EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
+ tbChannel.videoChannel, testEncryption));
+ ViETest::Log(
+ "External encryption/decryption added, you should still see video");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
+ tbChannel.videoChannel));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
+
+void ViEAutoTest::ViEEncryptionAPITest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ // Create VIE
+ TbInterfaces ViE("ViEEncryptionAPITest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ // Connect to channel
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+#ifdef WEBRTC_SRTP
+ unsigned char srtpKey[30] =
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
+ 4, 5, 6, 7, 8, 9};
+
+ //
+ // EnableSRTPSend and DisableSRTPSend
+ //
+
+ // Incorrect input argument, complete protection not enabled
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kNoProtection, srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryption, srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kAuthentication, srtpKey));
+
+ // Incorrect cipher key length
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 15,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 257,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 15, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
+
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 257, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
+
+ // Incorrect auth key length
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode,
+ 30, webrtc::kAuthHmacSha1, 21, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 257, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 21, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 20, 13, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+
+ // NULL input
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ NULL));
+
+ // Double enable and disable
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+
+ // Note(qhogpat): the second check is likely incorrect.
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // No protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
+ webrtc::kNoProtection, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Authentication only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 1, 4, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 20, 20, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 1, 1, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Encryption only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 16,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // Full protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ //
+ // EnableSRTPReceive and DisableSRTPReceive
+ //
+
+ // Incorrect input argument, complete protection not enabled
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kNoProtection, srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryption, srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kAuthentication, srtpKey));
+
+ // Incorrect cipher key length
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 15,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 257,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 15, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 257, webrtc::kAuthHmacSha1,
+ 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
+
+ // Incorrect auth key length
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 21, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 257, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 21, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 20, 13, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+
+ // NULL input
+ EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ NULL));
+
+ // Double enable and disable
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
+
+ // No protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
+ webrtc::kNoProtection, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+
+ // Authentication only
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 1, 4, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0,
+ webrtc::kAuthHmacSha1, 20, 20, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
+ 1, 1, webrtc::kAuthentication, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+
+ // Encryption only
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 16,
+ webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+
+ // Full protection
+ EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
+ tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
+ webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
+ srtpKey));
+ EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
+#endif //WEBRTC_SRTP
+ //
+ // External encryption
+ //
+
+ ViEAutotestEncryption testEncryption;
+ EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
+ tbChannel.videoChannel, testEncryption));
+ EXPECT_NE(0, ViE.encryption->RegisterExternalEncryption(
+ tbChannel.videoChannel, testEncryption));
+ EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
+ tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
+ tbChannel.videoChannel));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc
new file mode 100644
index 0000000..83ace52
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc
@@ -0,0 +1,493 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "engine_configurations.h"
+
+#include "testsupport/fileutils.h"
+#include "tb_interfaces.h"
+#include "tb_capture_device.h"
+
+#include "voe_codec.h"
+
+class ViEAutotestFileObserver: public webrtc::ViEFileObserver
+{
+public:
+ ViEAutotestFileObserver() {};
+ ~ViEAutotestFileObserver() {};
+
+ void PlayFileEnded(const WebRtc_Word32 fileId)
+ {
+ ViETest::Log("PlayFile ended");
+ }
+};
+
+void ViEAutoTest::ViEFileStandardTest()
+{
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+ {
+ ViETest::Log("Starting a loopback call...");
+
+ TbInterfaces interfaces = TbInterfaces("ViEFileStandardTest");
+
+ webrtc::VideoEngine* ptrViE = interfaces.video_engine;
+ webrtc::ViEBase* ptrViEBase = interfaces.base;
+ webrtc::ViECapture* ptrViECapture = interfaces.capture;
+ webrtc::ViERender* ptrViERender = interfaces.render;
+ webrtc::ViECodec* ptrViECodec = interfaces.codec;
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp = interfaces.rtp_rtcp;
+ webrtc::ViENetwork* ptrViENetwork = interfaces.network;
+
+ TbCaptureDevice captureDevice = TbCaptureDevice(interfaces);
+ int captureId = captureDevice.captureId;
+
+ int videoChannel = -1;
+ EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
+ EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
+ captureId, videoChannel));
+
+ EXPECT_EQ(0, ptrViERtpRtcp->SetRTCPStatus(
+ videoChannel, webrtc::kRtcpCompound_RFC4585));
+ EXPECT_EQ(0, ptrViERtpRtcp->SetKeyFrameRequestMethod(
+ videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true));
+
+ EXPECT_EQ(0, ptrViERender->AddRenderer(
+ captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ptrViERender->AddRenderer(
+ videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ptrViERender->StartRender(captureId));
+ EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
+
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
+ EXPECT_EQ(0, ptrViECodec->SetReceiveCodec(videoChannel,
+ videoCodec));
+ }
+
+ // Find the codec used for encoding the channel
+ for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8)
+ {
+ EXPECT_EQ(0, ptrViECodec->SetSendCodec(videoChannel, videoCodec));
+ break;
+ }
+ }
+ // Find the codec used for recording.
+ for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
+ {
+ EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
+ if (videoCodec.codecType == webrtc::kVideoCodecI420)
+ {
+ break;
+ }
+ }
+
+
+ const char* ipAddress = "127.0.0.1";
+ const unsigned short rtpPort = 6000;
+ EXPECT_EQ(0, ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort));
+ EXPECT_EQ(0, ptrViEBase->StartReceive(videoChannel));
+ EXPECT_EQ(0, ptrViENetwork->SetSendDestination(
+ videoChannel, ipAddress, rtpPort));
+ EXPECT_EQ(0, ptrViEBase->StartSend(videoChannel));
+ webrtc::ViEFile* ptrViEFile = webrtc::ViEFile::GetInterface(ptrViE);
+ EXPECT_TRUE(ptrViEFile != NULL);
+
+ webrtc::VoiceEngine* ptrVEEngine = webrtc::VoiceEngine::Create();
+ webrtc::VoEBase* ptrVEBase = webrtc::VoEBase::GetInterface(ptrVEEngine);
+ ptrVEBase->Init();
+
+ int audioChannel = ptrVEBase->CreateChannel();
+ ptrViEBase->SetVoiceEngine(ptrVEEngine);
+ ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel);
+
+ webrtc::CodecInst audioCodec;
+ webrtc::VoECodec* ptrVECodec =
+ webrtc::VoECodec::GetInterface(ptrVEEngine);
+ for (int index = 0; index < ptrVECodec->NumOfCodecs(); index++)
+ {
+ ptrVECodec->GetCodec(index, audioCodec);
+ if (0 == strcmp(audioCodec.plname, "PCMU") || 0
+ == strcmp(audioCodec.plname, "PCMA"))
+ {
+ break; // these two types are allowed as avi recording formats
+ }
+ }
+
+ webrtc::CodecInst audioCodec2;
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ // Call started
+ ViETest::Log("Call started\nYou should see local preview from camera\n"
+ "in window 1 and the remote video in window 2.");
+ AutoTestSleep(2000);
+
+ const int RENDER_TIMEOUT = 1000;
+ const int TEST_SPACING = 1000;
+ const int VIDEO_LENGTH = 5000;
+
+ const std::string root = webrtc::test::ProjectRootPath() +
+ "src/video_engine/test/auto_test/media/";
+ const std::string renderStartImage = root + "renderStartImage.jpg";
+ const std::string captureDeviceImage = root + "captureDeviceImage.jpg";
+ const std::string renderTimeoutFile = root + "renderTimeoutImage.jpg";
+
+ const std::string output = webrtc::test::OutputPath();
+ const std::string snapshotCaptureDeviceFileName =
+ output + "snapshotCaptureDevice.jpg";
+ const std::string incomingVideo = output + "incomingVideo.avi";
+ const std::string outgoingVideo = output + "outgoingVideo.avi";
+ const std::string snapshotRenderFileName =
+ output + "snapshotRenderer.jpg";
+
+ webrtc::ViEPicture capturePicture;
+ webrtc::ViEPicture renderPicture;
+ webrtc::ViEPicture renderTimeoutPicture; // TODO: init with and image
+
+ ViEAutotestFileObserver fileObserver;
+ int fileId;
+
+ AutoTestSleep(TEST_SPACING);
+
+ // testing StartRecordIncomingVideo and StopRecordIncomingVideo
+ {
+ ViETest::Log("Recording incoming video (currently no audio) for %d "
+ "seconds", VIDEO_LENGTH);
+
+ EXPECT_EQ(0, ptrViEFile->StartRecordIncomingVideo(
+ videoChannel, incomingVideo.c_str(), webrtc::NO_AUDIO,
+ audioCodec2, videoCodec));
+
+ AutoTestSleep(VIDEO_LENGTH);
+ ViETest::Log("Stop recording incoming video");
+
+ EXPECT_EQ(0, ptrViEFile->StopRecordIncomingVideo(videoChannel));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // testing GetFileInformation
+ {
+ webrtc::VideoCodec fileVideoCodec;
+ webrtc::CodecInst fileAudioCodec;
+ ViETest::Log("Reading video file information");
+
+ EXPECT_EQ(0, ptrViEFile->GetFileInformation(
+ incomingVideo.c_str(), fileVideoCodec, fileAudioCodec));
+ PrintAudioCodec(fileAudioCodec);
+ PrintVideoCodec(fileVideoCodec);
+ }
+
+ // testing StartPlayFile and RegisterObserver
+ {
+ ViETest::Log("Start playing file: %s with observer",
+ incomingVideo.c_str());
+ EXPECT_EQ(0, ptrViEFile->StartPlayFile(incomingVideo.c_str(),
+ fileId));
+
+ ViETest::Log("Registering file observer");
+ EXPECT_EQ(0, ptrViEFile->RegisterObserver(fileId, fileObserver));
+ ViETest::Log("Done\n");
+ }
+
+ // testing SendFileOnChannel and StopSendFileOnChannel
+ {
+ ViETest::Log("Sending video on channel");
+ // should fail since we are sending the capture device.
+ EXPECT_NE(0, ptrViEFile->SendFileOnChannel(fileId, videoChannel));
+
+ // Disconnect the camera
+ EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
+
+ // And try playing the file again.
+ EXPECT_EQ(0, ptrViEFile->SendFileOnChannel(fileId, videoChannel));
+
+ AutoTestSleep(VIDEO_LENGTH);
+ ViETest::Log("Stopped sending video on channel");
+ EXPECT_EQ(0, ptrViEFile->StopSendFileOnChannel(videoChannel));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // stop playing the file
+ {
+ ViETest::Log("Stop playing the file.");
+ EXPECT_EQ(0, ptrViEFile->StopPlayFile(fileId));
+ ViETest::Log("Done\n");
+ }
+
+ // testing StartRecordOutgoingVideo and StopRecordOutgoingVideo
+ {
+ // connect the camera to the output.
+ EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
+ captureId, videoChannel));
+
+ ViETest::Log("Recording outgoing video (currently no audio) for %d "
+ "seconds", VIDEO_LENGTH);
+ EXPECT_EQ(0, ptrViEFile->StartRecordOutgoingVideo(
+ videoChannel, outgoingVideo.c_str(), webrtc::NO_AUDIO,
+ audioCodec2, videoCodec));
+
+ AutoTestSleep(VIDEO_LENGTH);
+ ViETest::Log("Stop recording outgoing video");
+ EXPECT_EQ(0, ptrViEFile->StopRecordOutgoingVideo(videoChannel));
+ ViETest::Log("Done\n");
+ }
+
+ // again testing GetFileInformation
+ {
+ EXPECT_EQ(0, ptrViEFile->GetFileInformation(
+ incomingVideo.c_str(), videoCodec, audioCodec2));
+ PrintAudioCodec(audioCodec2);
+ PrintVideoCodec(videoCodec);
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // GetCaptureDeviceSnapshot
+ {
+ ViETest::Log("Testing GetCaptureDeviceSnapshot(int, ViEPicture)");
+ ViETest::Log("Taking a picture to use for displaying ViEPictures "
+ "for the rest of file test");
+ ViETest::Log("Hold an object to the camera. Ready?...");
+ AutoTestSleep(1000);
+ ViETest::Log("3");
+ AutoTestSleep(1000);
+ ViETest::Log("...2");
+ AutoTestSleep(1000);
+ ViETest::Log("...1");
+ AutoTestSleep(1000);
+ ViETest::Log("...Taking picture!");
+ EXPECT_EQ(0, ptrViEFile->GetCaptureDeviceSnapshot(
+ captureId, capturePicture));
+ ViETest::Log("Picture has been taken.");
+ AutoTestSleep(TEST_SPACING);
+
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // GetRenderSnapshot
+ {
+ ViETest::Log("Testing GetRenderSnapshot(int, char*)");
+
+ ViETest::Log("Taking snapshot of videoChannel %d", captureId);
+ EXPECT_EQ(0, ptrViEFile->GetRenderSnapshot(
+ captureId, snapshotRenderFileName.c_str()));
+ ViETest::Log("Wrote image to file %s",
+ snapshotRenderFileName.c_str());
+ ViETest::Log("Done\n");
+ AutoTestSleep(TEST_SPACING);
+ }
+
+ // GetRenderSnapshot
+ {
+ ViETest::Log("Testing GetRenderSnapshot(int, ViEPicture)");
+ EXPECT_EQ(0, ptrViEFile->GetRenderSnapshot(
+ captureId, renderPicture));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // GetCaptureDeviceSnapshot
+ {
+ ViETest::Log("Testing GetCaptureDeviceSnapshot(int, char*)");
+ ViETest::Log("Taking snapshot from capture device %d", captureId);
+ EXPECT_EQ(0, ptrViEFile->GetCaptureDeviceSnapshot(
+ captureId, snapshotCaptureDeviceFileName.c_str()));
+ ViETest::Log("Wrote image to file %s",
+ snapshotCaptureDeviceFileName.c_str());
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // Testing: SetCaptureDeviceImage
+ {
+ ViETest::Log("Testing SetCaptureDeviceImage(int, char*)");
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ EXPECT_EQ(0, ptrViEFile->SetCaptureDeviceImage(
+ captureId, captureDeviceImage.c_str()));
+
+ ViETest::Log("you should see the capture device image now");
+ AutoTestSleep(2 * RENDER_TIMEOUT);
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // Testing: SetCaptureDeviceImage
+ {
+ ViETest::Log("Testing SetCaptureDeviceImage(int, ViEPicture)");
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ EXPECT_EQ(0, ptrViEFile->SetCaptureDeviceImage(
+ captureId, capturePicture));
+
+ ViETest::Log("you should see the capture device image now");
+ AutoTestSleep(2 * RENDER_TIMEOUT);
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // testing SetRenderStartImage(videoChannel, renderStartImage);
+ {
+ ViETest::Log("Testing SetRenderStartImage(int, char*)");
+ // set render image, then stop capture and stop render to display it
+ ViETest::Log("Stoping renderer, setting start image, then "
+ "restarting");
+ EXPECT_EQ(0, ptrViEFile->SetRenderStartImage(
+ videoChannel, renderStartImage.c_str()));
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
+
+ ViETest::Log("Render start image should be displayed.");
+ AutoTestSleep(RENDER_TIMEOUT);
+
+ // restarting capture and render
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // testing SetRenderStartImage(videoChannel, renderStartImage);
+ {
+ ViETest::Log("Testing SetRenderStartImage(int, ViEPicture)");
+ // set render image, then stop capture and stop render to display it
+ ViETest::Log("Stoping renderer, setting start image, then "
+ "restarting");
+ EXPECT_EQ(0, ptrViEFile->SetRenderStartImage(
+ videoChannel, capturePicture));
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
+
+ ViETest::Log("Render start image should be displayed.");
+ AutoTestSleep(RENDER_TIMEOUT);
+
+ // restarting capture and render
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // testing SetRenderTimeoutImage(videoChannel, renderTimeoutFile,
+ // RENDER_TIMEOUT);
+ {
+ ViETest::Log("Testing SetRenderTimeoutImage(int, char*)");
+ ViETest::Log("Stopping capture device to induce timeout of %d ms",
+ RENDER_TIMEOUT);
+ EXPECT_EQ(0, ptrViEFile->SetRenderTimeoutImage(
+ videoChannel, renderTimeoutFile.c_str(), RENDER_TIMEOUT));
+
+ // now stop sending frames to the remote renderer and wait for
+ // timeout
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ AutoTestSleep(RENDER_TIMEOUT);
+ ViETest::Log("Timeout image should be displayed now for %d ms",
+ RENDER_TIMEOUT * 2);
+ AutoTestSleep(RENDER_TIMEOUT * 2);
+
+ // restart the capture device to undo the timeout
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ ViETest::Log("Restarting capture device");
+ AutoTestSleep(RENDER_TIMEOUT);
+ ViETest::Log("Done\n");
+ }
+
+ AutoTestSleep(TEST_SPACING);
+
+ // Need to create a ViEPicture object to pass into this function.
+ // SetRenderTimeoutImage(videoChannel, renderTimeoutFile,
+ // RENDER_TIMEOUT);
+ {
+ ViETest::Log("Testing SetRenderTimeoutImage(int, ViEPicture)");
+ ViETest::Log("Stopping capture device to induce timeout of %d",
+ RENDER_TIMEOUT);
+ EXPECT_EQ(0, ptrViEFile->SetRenderTimeoutImage(
+ videoChannel, capturePicture, RENDER_TIMEOUT));
+
+ // now stop sending frames to the remote renderer and wait for
+ // timeout
+ EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
+ AutoTestSleep(RENDER_TIMEOUT);
+ ViETest::Log("Timeout image should be displayed now for %d",
+ RENDER_TIMEOUT * 2);
+ AutoTestSleep(RENDER_TIMEOUT * 2);
+
+ // restart the capture device to undo the timeout
+ EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
+ ViETest::Log("Restarting capture device");
+ ViETest::Log("Done\n");
+ }
+
+ // testing DeregisterObserver
+ {
+ ViETest::Log("Deregistering file observer");
+ // Should fail since we don't observe this file.
+ EXPECT_NE(0, ptrViEFile->DeregisterObserver(fileId, fileObserver));
+ }
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+
+ EXPECT_EQ(0, ptrViEBase->DisconnectAudioChannel(videoChannel));
+ EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(NULL));
+ EXPECT_EQ(0, ptrVEBase->DeleteChannel(audioChannel));
+ EXPECT_EQ(0, ptrVEBase->Release());
+ EXPECT_EQ(0, ptrVECodec->Release());
+ EXPECT_TRUE(webrtc::VoiceEngine::Delete(ptrVEEngine));
+
+ EXPECT_EQ(0, ptrViEBase->StopReceive(videoChannel));
+ EXPECT_EQ(0, ptrViEBase->StopSend(videoChannel));
+ EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
+ EXPECT_EQ(0, ptrViERender->RemoveRenderer(captureId));
+ EXPECT_EQ(0, ptrViERender->RemoveRenderer(videoChannel));
+ EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
+ EXPECT_EQ(0, ptrViEFile->FreePicture(capturePicture));
+ EXPECT_EQ(0, ptrViEFile->FreePicture(renderPicture));
+ EXPECT_EQ(0, ptrViEFile->FreePicture(renderTimeoutPicture));
+ EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
+
+ EXPECT_EQ(0, ptrViEFile->Release());
+ }
+#endif
+}
+
+void ViEAutoTest::ViEFileExtendedTest()
+{
+}
+
+void ViEAutoTest::ViEFileAPITest()
+{
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc
new file mode 100644
index 0000000..887fe6e
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc
@@ -0,0 +1,240 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_image_process.cc
+//
+
+// Settings
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "engine_configurations.h"
+
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "tb_capture_device.h"
+
+class MyEffectFilter: public webrtc::ViEEffectFilter
+{
+public:
+ MyEffectFilter() {}
+
+ ~MyEffectFilter() {}
+
+ virtual int Transform(int size, unsigned char* frameBuffer,
+ unsigned int timeStamp90KHz, unsigned int width,
+ unsigned int height)
+ {
+ // Black and white
+ memset(frameBuffer + (2 * size) / 3, 0x7f, size / 3);
+ return 0;
+ }
+};
+
+void ViEAutoTest::ViEImageProcessStandardTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+ int rtpPort = 6000;
+ // Create VIE
+ TbInterfaces ViE("ViEImageProcessAPITest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+ tbChannel.StartReceive(rtpPort);
+ tbChannel.StartSend(rtpPort);
+
+ MyEffectFilter effectFilter;
+
+ RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
+
+ ViETest::Log("Capture device is renderered in Window 1");
+ ViETest::Log("Remote stream is renderered in Window 2");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+
+ EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
+ tbCapture.captureId, effectFilter));
+
+ ViETest::Log("Black and white filter registered for capture device, "
+ "affects both windows");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
+ tbCapture.captureId));
+
+ EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+
+ ViETest::Log("Remove capture effect filter, adding filter for incoming "
+ "stream");
+ ViETest::Log("Only Window 2 should be black and white");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+
+ int rtpPort2 = rtpPort + 100;
+ // Create a video channel
+ TbVideoChannel tbChannel2(ViE, webrtc::kVideoCodecVP8);
+
+ tbCapture.ConnectTo(tbChannel2.videoChannel);
+ tbChannel2.StartReceive(rtpPort2);
+ tbChannel2.StartSend(rtpPort2);
+
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbChannel2.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbChannel2.videoChannel));
+ EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
+ tbChannel.videoChannel));
+
+ ViETest::Log("Local renderer removed, added new channel and rendering in "
+ "Window1.");
+
+ EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
+ tbCapture.captureId, effectFilter));
+
+ ViETest::Log("Black and white filter registered for capture device, "
+ "affects both windows");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
+ tbCapture.captureId));
+
+ EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+
+ ViETest::Log("Capture filter removed.");
+ ViETest::Log("Black and white filter registered for one channel, Window2 "
+ "should be black and white");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
+ tbChannel.videoChannel));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
+
+void ViEAutoTest::ViEImageProcessExtendedTest()
+{
+ ViEImageProcessStandardTest();
+}
+
+void ViEAutoTest::ViEImageProcessAPITest()
+{
+ TbInterfaces ViE("ViEImageProcessAPITest");
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ TbCaptureDevice tbCapture(ViE);
+
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ MyEffectFilter effectFilter;
+
+ //
+ // Capture effect filter
+ //
+ // Add effect filter
+ EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
+ tbCapture.captureId, effectFilter));
+ // Add again -> error
+ EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
+ tbCapture.captureId, effectFilter));
+ EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
+ tbCapture.captureId));
+
+ // Double deregister
+ EXPECT_NE(0, ViE.image_process->DeregisterCaptureEffectFilter(
+ tbCapture.captureId));
+ // Non-existing capture device
+ EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+
+ //
+ // Render effect filter
+ //
+ EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+ EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+ EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.image_process->DeregisterRenderEffectFilter(
+ tbChannel.videoChannel));
+
+ // Non-existing channel id
+ EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
+ tbCapture.captureId, effectFilter));
+
+ //
+ // Send effect filter
+ //
+ EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+ EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
+ tbChannel.videoChannel, effectFilter));
+ EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.image_process->DeregisterSendEffectFilter(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
+ tbCapture.captureId, effectFilter));
+
+ //
+ // Denoising
+ //
+ EXPECT_EQ(0, ViE.image_process->EnableDenoising(tbCapture.captureId, true));
+ EXPECT_NE(0, ViE.image_process->EnableDenoising(tbCapture.captureId, true));
+ EXPECT_EQ(0, ViE.image_process->EnableDenoising(
+ tbCapture.captureId, false));
+ EXPECT_NE(0, ViE.image_process->EnableDenoising(
+ tbCapture.captureId, false));
+ EXPECT_NE(0, ViE.image_process->EnableDenoising(
+ tbChannel.videoChannel, true));
+
+ //
+ // Deflickering
+ //
+ EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
+ tbCapture.captureId, true));
+ EXPECT_NE(0, ViE.image_process->EnableDeflickering(
+ tbCapture.captureId, true));
+ EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
+ tbCapture.captureId, false));
+ EXPECT_NE(0, ViE.image_process->EnableDeflickering(
+ tbCapture.captureId, false));
+ EXPECT_NE(0, ViE.image_process->EnableDeflickering(
+ tbChannel.videoChannel, true));
+
+ //
+ // Color enhancement
+ //
+ EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
+ tbChannel.videoChannel, false));
+ EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
+ tbChannel.videoChannel, true));
+ EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
+ tbChannel.videoChannel, true));
+ EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
+ tbChannel.videoChannel, false));
+ EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
+ tbChannel.videoChannel, false));
+ EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
+ tbCapture.captureId, true));
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc
new file mode 100644
index 0000000..aca96bd
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_linux.cc
+//
+#include "vie_autotest_linux.h"
+
+#include <string>
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest_main.h"
+#include "engine_configurations.h"
+#include "critical_section_wrapper.h"
+#include "thread_wrapper.h"
+
+ViEAutoTestWindowManager::ViEAutoTestWindowManager()
+ : _hdsp1(NULL),
+ _hdsp2(NULL) {
+}
+
+ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
+ TerminateWindows();
+}
+
+void* ViEAutoTestWindowManager::GetWindow1() {
+ return reinterpret_cast<void*>(_hwnd1);
+}
+
+void* ViEAutoTestWindowManager::GetWindow2() {
+ return reinterpret_cast<void*>(_hwnd2);
+}
+
+int ViEAutoTestWindowManager::TerminateWindows() {
+ if (_hdsp1) {
+ ViEDestroyWindow(&_hwnd1, _hdsp1);
+ _hdsp1 = NULL;
+ }
+ if (_hdsp2) {
+ ViEDestroyWindow(&_hwnd2, _hdsp2);
+ _hdsp2 = NULL;
+ }
+ return 0;
+}
+
+int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size,
+ void* window1Title,
+ void* window2Title) {
+ ViECreateWindow(&_hwnd1, &_hdsp1, window1Size.origin.x,
+ window1Size.origin.y, window1Size.size.width,
+ window1Size.size.height,
+ reinterpret_cast<char*>(window1Title));
+ ViECreateWindow(&_hwnd2, &_hdsp2, window2Size.origin.x,
+ window2Size.origin.y, window2Size.size.width,
+ window2Size.size.height,
+ reinterpret_cast<char*>(window2Title));
+
+ return 0;
+}
+
+int ViEAutoTestWindowManager::ViECreateWindow(Window *out_window,
+ Display **out_display, int x_pos,
+ int y_pos, int width, int height,
+ char* title) {
+ Display* display = XOpenDisplay(NULL);
+ if (display == NULL) {
+ // There's no point to continue if this happens: nothing will work anyway.
+ printf("Failed to connect to X server: X environment likely broken\n");
+ exit(-1);
+ }
+
+ int screen = DefaultScreen(display);
+
+ // Try to establish a 24-bit TrueColor display
+ // (our environment must allow this).
+ XVisualInfo visual_info;
+ if (XMatchVisualInfo(display, screen, 24, TrueColor, &visual_info) == 0) {
+ printf("Failed to establish 24-bit TrueColor in X environment.\n");
+ exit(-1);
+ }
+
+ // Create suitable window attributes.
+ XSetWindowAttributes window_attributes;
+ window_attributes.colormap = XCreateColormap(
+ display, DefaultRootWindow(display), visual_info.visual, AllocNone);
+ window_attributes.event_mask = StructureNotifyMask | ExposureMask;
+ window_attributes.background_pixel = 0;
+ window_attributes.border_pixel = 0;
+
+ unsigned long attribute_mask = CWBackPixel | CWBorderPixel | CWColormap |
+ CWEventMask;
+
+ Window _window = XCreateWindow(display, DefaultRootWindow(display), x_pos,
+ y_pos, width, height, 0, visual_info.depth,
+ InputOutput, visual_info.visual,
+ attribute_mask, &window_attributes);
+
+ // Set window name.
+ XStoreName(display, _window, title);
+ XSetIconName(display, _window, title);
+
+ // Make x report events for mask.
+ XSelectInput(display, _window, StructureNotifyMask);
+
+ // Map the window to the display.
+ XMapWindow(display, _window);
+
+ // Wait for map event.
+ XEvent event;
+ do {
+ XNextEvent(display, &event);
+ } while (event.type != MapNotify || event.xmap.event != _window);
+
+ *out_window = _window;
+ *out_display = display;
+ return 0;
+}
+
+int ViEAutoTestWindowManager::ViEDestroyWindow(Window *window,
+ Display *display) {
+ XUnmapWindow(display, *window);
+ XDestroyWindow(display, *window);
+ XSync(display, false);
+ return 0;
+}
+
+bool ViEAutoTestWindowManager::SetTopmostWindow() {
+ return 0;
+}
+
+int main(int argc, char** argv) {
+ ViEAutoTestMain auto_test;
+ return auto_test.RunTests(argc, argv);
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc
new file mode 100644
index 0000000..d660745
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc
@@ -0,0 +1,663 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_loopback.cc
+//
+// This code is also used as sample code for ViE 3.0
+//
+
+// ===================================================================
+//
+// BEGIN: VideoEngine 3.0 Sample Code
+//
+
+#include <iostream>
+
+#include "common_types.h"
+#include "tb_external_transport.h"
+#include "voe_base.h"
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+
+#define VCM_RED_PAYLOAD_TYPE 96
+#define VCM_ULPFEC_PAYLOAD_TYPE 97
+
+int VideoEngineSampleCode(void* window1, void* window2)
+{
+ //********************************************************
+ // Begin create/initialize Video Engine for testing
+ //********************************************************
+
+ int error = 0;
+
+ //
+ // Create a VideoEngine instance
+ //
+ webrtc::VideoEngine* ptrViE = NULL;
+ ptrViE = webrtc::VideoEngine::Create();
+ if (ptrViE == NULL)
+ {
+ printf("ERROR in VideoEngine::Create\n");
+ return -1;
+ }
+
+ error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceLevel\n");
+ return -1;
+ }
+
+ std::string trace_file =
+ ViETest::GetResultOutputPath() + "ViELoopbackCall_trace.txt";
+ error = ptrViE->SetTraceFile(trace_file.c_str());
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceFile\n");
+ return -1;
+ }
+
+ //
+ // Init VideoEngine and create a channel
+ //
+ webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViEBase::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViEBase->Init();
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::Init\n");
+ return -1;
+ }
+
+ int videoChannel = -1;
+ error = ptrViEBase->CreateChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::CreateChannel\n");
+ return -1;
+ }
+
+ //
+ // List available capture devices, allocate and connect.
+ //
+ webrtc::ViECapture* ptrViECapture =
+ webrtc::ViECapture::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViECapture::GetInterface\n");
+ return -1;
+ }
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ char deviceName[KMaxDeviceNameLength];
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ char uniqueId[KMaxUniqueIdLength];
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ printf("Available capture devices:\n");
+ int captureIdx = 0;
+ for (captureIdx = 0;
+ captureIdx < ptrViECapture->NumberOfCaptureDevices();
+ captureIdx++)
+ {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId,
+ KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+ printf("\t %d. %s\n", captureIdx + 1, deviceName);
+ }
+ printf("\nChoose capture device: ");
+#ifdef WEBRTC_ANDROID
+ captureIdx = 0;
+ printf("0\n");
+#else
+ if (scanf("%d", &captureIdx) != 1)
+ {
+ printf("Error in scanf()\n");
+ return -1;
+ }
+ getchar();
+ captureIdx = captureIdx - 1; // Compensate for idx start at 1.
+#endif
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId,
+ KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+
+ int captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
+ captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::AllocateCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ConnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StartCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StartCapture\n");
+ return -1;
+ }
+
+ //
+ // RTP/RTCP settings
+ //
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp =
+ webrtc::ViERTP_RTCP::GetInterface(ptrViE);
+ if (ptrViERtpRtcp == NULL)
+ {
+ printf("ERROR in ViERTP_RTCP::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
+ webrtc::kRtcpCompound_RFC4585);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
+ videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
+ return -1;
+ }
+
+ //
+ // Set up rendering
+ //
+ webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
+ if (ptrViERender == NULL)
+ {
+ printf("ERROR in ViERender::GetInterface\n");
+ return -1;
+ }
+
+ error
+ = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
+ 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ //
+ // Setup codecs
+ //
+ webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
+ if (ptrViECodec == NULL)
+ {
+ printf("ERROR in ViECodec::GetInterface\n");
+ return -1;
+ }
+
+ // Check available codecs and prepare receive codecs
+ printf("\nAvailable codecs:\n");
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ int codecIdx = 0;
+ for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++)
+ {
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ // try to keep the test frame size small when I420
+ if (videoCodec.codecType == webrtc::kVideoCodecI420)
+ {
+ videoCodec.width = 176;
+ videoCodec.height = 144;
+ }
+
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetReceiveCodec\n");
+ return -1;
+ }
+ if (videoCodec.codecType != webrtc::kVideoCodecRED
+ && videoCodec.codecType != webrtc::kVideoCodecULPFEC)
+ {
+ printf("\t %d. %s\n", codecIdx + 1, videoCodec.plName);
+ }
+ }
+ printf("Choose codec: ");
+#ifdef WEBRTC_ANDROID
+ codecIdx = 0;
+ printf("0\n");
+#else
+ if (scanf("%d", &codecIdx) != 1)
+ {
+ printf("Error in scanf()\n");
+ return -1;
+ }
+ getchar();
+ codecIdx = codecIdx - 1; // Compensate for idx start at 1.
+#endif
+
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ // Set spatial resolution option
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Enter frame size option (default is CIF):" << std::endl;
+ std::cout << "1. QCIF (176X144) " << std::endl;
+ std::cout << "2. CIF (352X288) " << std::endl;
+ std::cout << "3. VGA (640X480) " << std::endl;
+ std::cout << "4. 4CIF (704X576) " << std::endl;
+ std::cout << "5. WHD (1280X720) " << std::endl;
+ std::getline(std::cin, str);
+ int resolnOption = atoi(str.c_str());
+ // Try to keep the test frame size small when I420
+ if (videoCodec.codecType == webrtc::kVideoCodecI420)
+ {
+ resolnOption = 1;
+ }
+ switch (resolnOption)
+ {
+ case 1:
+ videoCodec.width = 176;
+ videoCodec.height = 144;
+ break;
+ case 2:
+ videoCodec.width = 352;
+ videoCodec.height = 288;
+ break;
+ case 3:
+ videoCodec.width = 640;
+ videoCodec.height = 480;
+ break;
+ case 4:
+ videoCodec.width = 704;
+ videoCodec.height = 576;
+ break;
+ case 5:
+ videoCodec.width = 1280;
+ videoCodec.height = 720;
+ break;
+ }
+
+ // Set number of temporal layers.
+ std::cout << std::endl;
+ std::cout << "Choose number of temporal layers (1 to 4).";
+ std::cout << "Press enter for default: \n";
+ std::getline(std::cin, str);
+ int numTemporalLayers = atoi(str.c_str());
+ if(numTemporalLayers != 0)
+ {
+ videoCodec.codecSpecific.VP8.numberOfTemporalLayers = numTemporalLayers;
+ }
+
+ // Set start bit rate
+ std::cout << std::endl;
+ std::cout << "Choose start rate (in kbps). Press enter for default: ";
+ std::getline(std::cin, str);
+ int startRate = atoi(str.c_str());
+ if(startRate != 0)
+ {
+ videoCodec.startBitrate=startRate;
+ }
+
+ error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetSendCodec\n");
+ return -1;
+ }
+
+ //
+ // Choose Protection Mode
+ //
+ std::cout << std::endl;
+ std::cout << "Enter Protection Method:" << std::endl;
+ std::cout << "0. None" << std::endl;
+ std::cout << "1. FEC" << std::endl;
+ std::cout << "2. NACK" << std::endl;
+ std::cout << "3. NACK+FEC" << std::endl;
+ std::getline(std::cin, str);
+ int protectionMethod = atoi(str.c_str());
+ error = 0;
+ bool temporalToggling = true;
+ switch (protectionMethod)
+ {
+ case 0: // None: default is no protection
+ break;
+
+ case 1: // FEC only
+ error = ptrViERtpRtcp->SetFECStatus(videoChannel,
+ true,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ temporalToggling = false;
+ break;
+
+ case 2: // Nack only
+ error = ptrViERtpRtcp->SetNACKStatus(videoChannel, true);
+
+ break;
+
+ case 3: // Hybrid NAck and FEC
+ error = ptrViERtpRtcp->SetHybridNACKFECStatus(
+ videoChannel,
+ true,
+ VCM_RED_PAYLOAD_TYPE,
+ VCM_ULPFEC_PAYLOAD_TYPE);
+ temporalToggling = false;
+ break;
+ }
+
+ if (error < 0)
+ {
+ printf("ERROR in ViERTP_RTCP::SetProtectionStatus\n");
+ }
+
+
+ //
+ // Address settings
+ //
+ webrtc::ViENetwork* ptrViENetwork =
+ webrtc::ViENetwork::GetInterface(ptrViE);
+ if (ptrViENetwork == NULL)
+ {
+ printf("ERROR in ViENetwork::GetInterface\n");
+ return -1;
+ }
+
+ // Setting External transport
+ TbExternalTransport extTransport(*(ptrViENetwork));
+
+ int testMode = 0;
+ std::cout << std::endl;
+ std::cout << "Enter 1 for testing packet loss and delay with "
+ "external transport: ";
+ std::string test_str;
+ std::getline(std::cin, test_str);
+ testMode = atoi(test_str.c_str());
+ if (testMode == 1)
+ {
+ // Avoid changing SSRC due to collision.
+ error = ptrViERtpRtcp->SetLocalSSRC(videoChannel, 1);
+
+ error = ptrViENetwork->RegisterSendTransport(videoChannel,
+ extTransport);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::RegisterSendTransport \n");
+ return -1;
+ }
+
+ // Set up packet loss value
+ std::cout << "Enter Packet Loss Percentage" << std::endl;
+ std::string rate_str;
+ std::getline(std::cin, rate_str);
+ int rate = atoi(rate_str.c_str());
+ extTransport.SetPacketLoss(rate);
+ if (rate) {
+ temporalToggling = false;
+ }
+
+ // Set network delay value
+ std::cout << "Enter network delay value [mS]" << std::endl;
+ std::string delay_str;
+ std::getline(std::cin, delay_str);
+ int delayMs = atoi(delay_str.c_str());
+ extTransport.SetNetworkDelay(delayMs);
+
+ if (numTemporalLayers > 1 && temporalToggling) {
+ extTransport.SetTemporalToggle(numTemporalLayers);
+ } else {
+ // Disabled
+ extTransport.SetTemporalToggle(0);
+ }
+ }
+ else
+ {
+ const char* ipAddress = "127.0.0.1";
+ const unsigned short rtpPort = 6000;
+ std::cout << std::endl;
+ std::cout << "Using rtp port: " << rtpPort << std::endl;
+ std::cout << std::endl;
+ error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetLocalReceiver\n");
+ return -1;
+ }
+ error = ptrViENetwork->SetSendDestination(videoChannel,
+ ipAddress, rtpPort);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::SetSendDestination\n");
+ return -1;
+ }
+ }
+
+ error = ptrViEBase->StartReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartSend\n");
+ return -1;
+ }
+
+ //********************************************************
+ // Engine started
+ //********************************************************
+
+
+ // Call started
+ printf("\nLoopback call started\n\n");
+ printf("Press enter to stop...");
+ while ((getchar()) != '\n')
+ ;
+
+ //********************************************************
+ // Testing finished. Tear down Video Engine
+ //********************************************************
+
+ error = ptrViEBase->StopReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StopSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopSend\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StopCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StopCapture\n");
+ return -1;
+ }
+
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ReleaseCaptureDevice(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViEBase->DeleteChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::DeleteChannel\n");
+ return -1;
+ }
+
+ int remainingInterfaces = 0;
+ remainingInterfaces = ptrViECodec->Release();
+ remainingInterfaces += ptrViECapture->Release();
+ remainingInterfaces += ptrViERtpRtcp->Release();
+ remainingInterfaces += ptrViERender->Release();
+ remainingInterfaces += ptrViENetwork->Release();
+ remainingInterfaces += ptrViEBase->Release();
+ if (remainingInterfaces > 0)
+ {
+ printf("ERROR: Could not release all interfaces\n");
+ return -1;
+ }
+
+ bool deleted = webrtc::VideoEngine::Delete(ptrViE);
+ if (deleted == false)
+ {
+ printf("ERROR in VideoEngine::Delete\n");
+ return -1;
+ }
+
+ return 0;
+
+ //
+ // END: VideoEngine 3.0 Sample Code
+ //
+ // ===================================================================
+}
+
+int ViEAutoTest::ViELoopbackCall()
+{
+ ViETest::Log(" ");
+ ViETest::Log("========================================");
+ ViETest::Log(" ViE Autotest Loopback Call\n");
+
+ if (VideoEngineSampleCode(_window1, _window2) == 0)
+ {
+ ViETest::Log(" ");
+ ViETest::Log(" ViE Autotest Loopback Call Done");
+ ViETest::Log("========================================");
+ ViETest::Log(" ");
+
+ return 0;
+ }
+
+ ViETest::Log(" ");
+ ViETest::Log(" ViE Autotest Loopback Call Failed");
+ ViETest::Log("========================================");
+ ViETest::Log(" ");
+ return 1;
+
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc
new file mode 100644
index 0000000..b9f6958
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_autotest_main.h"
+
+#include "gflags/gflags.h"
+#include "gtest/gtest.h"
+#include "vie_autotest.h"
+#include "vie_autotest_window_manager_interface.h"
+#include "vie_window_creator.h"
+
+DEFINE_bool(automated, false, "Run Video engine tests in noninteractive mode.");
+
+static const std::string kStandardTest = "ViEStandardIntegrationTest";
+static const std::string kExtendedTest = "ViEExtendedIntegrationTest";
+static const std::string kApiTest = "ViEApiIntegrationTest";
+
+ViEAutoTestMain::ViEAutoTestMain() {
+ index_to_test_method_map_[1] = "RunsBaseTestWithoutErrors";
+ index_to_test_method_map_[2] = "RunsCaptureTestWithoutErrors";
+ index_to_test_method_map_[3] = "RunsCodecTestWithoutErrors";
+ index_to_test_method_map_[4] = "RunsEncryptionTestWithoutErrors";
+ index_to_test_method_map_[5] = "RunsFileTestWithoutErrors";
+ index_to_test_method_map_[6] = "RunsImageProcessTestWithoutErrors";
+ index_to_test_method_map_[7] = "RunsNetworkTestWithoutErrors";
+ index_to_test_method_map_[8] = "RunsRenderTestWithoutErrors";
+ index_to_test_method_map_[9] = "RunsRtpRtcpTestWithoutErrors";
+}
+
+int ViEAutoTestMain::RunTests(int argc, char** argv) {
+ // Initialize logging.
+ ViETest::Init();
+ // Initialize the testing framework.
+ testing::InitGoogleTest(&argc, argv);
+ // Parse remaining flags:
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ int result;
+ if (FLAGS_automated) {
+ // Run in automated mode.
+ result = RUN_ALL_TESTS();
+ } else {
+ // Run in interactive mode.
+ result = RunInteractiveMode();
+ }
+
+ ViETest::Terminate();
+ return result;
+}
+
+int ViEAutoTestMain::AskUserForTestCase() {
+ int choice;
+ std::string answer;
+
+ do {
+ ViETest::Log("\nSpecific tests:");
+ ViETest::Log("\t 0. Go back to previous menu.");
+
+ // Print all test method choices. Assumes that map sorts on its key.
+ int last_valid_choice = 0;
+ std::map<int, std::string>::const_iterator iterator;
+ for (iterator = index_to_test_method_map_.begin();
+ iterator != index_to_test_method_map_.end();
+ ++iterator) {
+ ViETest::Log("\t %d. %s", iterator->first, iterator->second.c_str());
+ last_valid_choice = iterator->first;
+ }
+
+ ViETest::Log("Choose specific test:");
+ choice = AskUserForNumber(0, last_valid_choice);
+ } while (choice == kInvalidChoice);
+
+ return choice;
+}
+
+int ViEAutoTestMain::AskUserForNumber(int min_allowed, int max_allowed) {
+ int result;
+ if (scanf("%d", &result) <= 0) {
+ ViETest::Log("\nPlease enter a number instead, then hit enter.");
+ getchar();
+ return kInvalidChoice;
+ }
+ getchar(); // Consume enter key.
+
+ if (result < min_allowed || result > max_allowed) {
+ ViETest::Log("%d-%d are valid choices. Please try again.", min_allowed,
+ max_allowed);
+ return kInvalidChoice;
+ }
+
+ return result;
+}
+
+int ViEAutoTestMain::RunTestMatching(const std::string test_case,
+ const std::string test_method) {
+ testing::FLAGS_gtest_filter = test_case + "." + test_method;
+ return RUN_ALL_TESTS();
+}
+
+int ViEAutoTestMain::RunSpecificTestCaseIn(const std::string test_case_name)
+{
+ // If user says 0, it means don't run anything.
+ int specific_choice = AskUserForTestCase();
+ if (specific_choice != 0){
+ return RunTestMatching(test_case_name,
+ index_to_test_method_map_[specific_choice]);
+ }
+ return 0;
+}
+
+int ViEAutoTestMain::RunSpecialTestCase(int choice) {
+ // 7-9 don't run in GTest and need to initialize by themselves.
+ assert(choice >= 7 && choice <= 9);
+
+ // Create the windows
+ ViEWindowCreator windowCreator;
+ ViEAutoTestWindowManagerInterface* windowManager =
+ windowCreator.CreateTwoWindows();
+
+ // Create the test cases
+ ViEAutoTest vieAutoTest(windowManager->GetWindow1(),
+ windowManager->GetWindow2());
+
+ int errors = 0;
+ switch (choice) {
+ case 7: errors = vieAutoTest.ViELoopbackCall(); break;
+ case 8: errors = vieAutoTest.ViECustomCall(); break;
+ case 9: errors = vieAutoTest.ViESimulcastCall(); break;
+ }
+
+ windowCreator.TerminateWindows();
+ return errors;
+}
+
+bool ViEAutoTestMain::RunInteractiveMode() {
+ ViETest::Log(" ============================== ");
+ ViETest::Log(" WebRTC ViE 3.x Autotest ");
+ ViETest::Log(" ============================== \n");
+
+ int choice = 0;
+ int errors = 0;
+ do {
+ ViETest::Log("Test types: ");
+ ViETest::Log("\t 0. Quit");
+ ViETest::Log("\t 1. All standard tests (delivery test)");
+ ViETest::Log("\t 2. All API tests");
+ ViETest::Log("\t 3. All extended test");
+ ViETest::Log("\t 4. Specific standard test");
+ ViETest::Log("\t 5. Specific API test");
+ ViETest::Log("\t 6. Specific extended test");
+ ViETest::Log("\t 7. Simple loopback call");
+ ViETest::Log("\t 8. Custom configure a call");
+ ViETest::Log("\t 9. Simulcast in loopback");
+ ViETest::Log("Select type of test:");
+
+ choice = AskUserForNumber(0, 9);
+ if (choice == kInvalidChoice) {
+ continue;
+ }
+ switch (choice) {
+ case 0: break;
+ case 1: errors = RunTestMatching(kStandardTest, "*"); break;
+ case 2: errors = RunTestMatching(kApiTest, "*"); break;
+ case 3: errors = RunTestMatching(kExtendedTest, "*"); break;
+ case 4: errors = RunSpecificTestCaseIn(kStandardTest); break;
+ case 5: errors = RunSpecificTestCaseIn(kApiTest); break;
+ case 6: errors = RunSpecificTestCaseIn(kExtendedTest); break;
+ default: errors = RunSpecialTestCase(choice); break;
+ }
+ } while (choice != 0);
+
+ if (errors) {
+ ViETest::Log("Test done with errors, see ViEAutotestLog.txt for test "
+ "result.\n");
+ } else {
+ ViETest::Log("Test done without errors, see ViEAutotestLog.txt for "
+ "test result.\n");
+ }
+ return true;
+}
+
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc
new file mode 100644
index 0000000..2a84644
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc
@@ -0,0 +1,563 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_network.cc
+//
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "engine_configurations.h"
+
+#include "tb_capture_device.h"
+#include "tb_external_transport.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+
+#if defined(_WIN32)
+#include <qos.h>
+#elif defined(WEBRTC_MAC_INTEL)
+
+#endif
+
+class ViEAutoTestNetworkObserver: public webrtc::ViENetworkObserver
+{
+public:
+ ViEAutoTestNetworkObserver()
+ {
+ }
+ virtual ~ViEAutoTestNetworkObserver()
+ {
+ }
+ virtual void OnPeriodicDeadOrAlive(const int videoChannel, const bool alive)
+ {
+ }
+ virtual void PacketTimeout(const int videoChannel,
+ const webrtc::ViEPacketTimeout timeout)
+ {
+ }
+};
+
+void ViEAutoTest::ViENetworkStandardTest()
+{
+ TbInterfaces ViE("ViENetworkStandardTest"); // Create VIE
+ TbCaptureDevice tbCapture(ViE);
+ {
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+
+ //
+ // Transport
+ //
+ TbExternalTransport testTransport(*ViE.network);
+ EXPECT_EQ(0, ViE.network->RegisterSendTransport(
+ tbChannel.videoChannel, testTransport));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
+
+ ViETest::Log("Call started using external transport, video should "
+ "see video in both windows\n");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
+ tbChannel.videoChannel));
+
+ char myIpAddress[64];
+ memset(myIpAddress, 0, 64);
+ unsigned short rtpPort = 1234;
+ memcpy(myIpAddress, "127.0.0.1", sizeof("127.0.0.1"));
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, myIpAddress, rtpPort,
+ rtpPort + 1, rtpPort));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ ViETest::Log("Changed to WebRTC SocketTransport, you should still see "
+ "video in both windows\n");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, rtpPort + 10, rtpPort + 11, myIpAddress));
+ ViETest::Log("Added UDP port filter for incorrect ports, you should "
+ "not see video in Window2");
+ AutoTestSleep(2000);
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, rtpPort, rtpPort + 1, "123.1.1.0"));
+ ViETest::Log("Added IP filter for incorrect IP address, you should not "
+ "see video in Window2");
+ AutoTestSleep(2000);
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
+ ViETest::Log("Added IP filter for this computer, you should see video "
+ "in Window2 again\n");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ tbCapture.Disconnect(tbChannel.videoChannel);
+ }
+}
+
+void ViEAutoTest::ViENetworkExtendedTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ TbInterfaces ViE("ViENetworkExtendedTest"); // Create VIE
+ TbCaptureDevice tbCapture(ViE);
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+
+ {
+ //
+ // ToS
+ //
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+ const char* remoteIp = "192.168.200.1";
+ int DSCP = 0;
+ bool useSetSockOpt = false;
+
+ webrtc::VideoCodec videoCodec;
+ EXPECT_EQ(0, ViE.codec->GetSendCodec(
+ tbChannel.videoChannel, videoCodec));
+ videoCodec.maxFramerate = 5;
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(
+ tbChannel.videoChannel, videoCodec));
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ char myIpAddress[64];
+ memset(myIpAddress, 0, 64);
+ unsigned short rtpPort = 9000;
+ EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, remoteIp, rtpPort, rtpPort + 1, rtpPort));
+
+ // ToS
+ int tos_result = ViE.network->SetSendToS(tbChannel.videoChannel, 2);
+ EXPECT_EQ(0, tos_result);
+ if (tos_result != 0)
+ {
+ ViETest::Log("ViESetSendToS error!.");
+ ViETest::Log("You must be admin to run these tests.");
+ ViETest::Log("On Win7 and late Vista, you need to right click the "
+ "exe and choose");
+ ViETest::Log("\"Run as administrator\"\n");
+ getchar();
+ }
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ ViETest::Log("Use Wireshark to capture the outgoing video stream and "
+ "verify ToS settings\n");
+ ViETest::Log(" DSCP set to 0x%x\n", DSCP);
+ AutoTestSleep(1000);
+
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+ ViETest::Log(" DSCP set to 0x%x\n", DSCP);
+ AutoTestSleep(1000);
+
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0));
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 2, true));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+ ViETest::Log(" DSCP set to 0x%x\n", DSCP);
+ AutoTestSleep(1000);
+
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63, true));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+ ViETest::Log(" DSCP set to 0x%x\n", DSCP);
+ AutoTestSleep(1000);
+
+ tbCapture.Disconnect(tbChannel.videoChannel);
+ }
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
+
+void ViEAutoTest::ViENetworkAPITest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+
+ TbInterfaces ViE("ViENetworkAPITest"); // Create VIE
+ {
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ //
+ // External transport
+ //
+ TbExternalTransport testTransport(*ViE.network);
+ EXPECT_EQ(0, ViE.network->RegisterSendTransport(
+ tbChannel.videoChannel, testTransport));
+ EXPECT_NE(0, ViE.network->RegisterSendTransport(
+ tbChannel.videoChannel, testTransport));
+
+ unsigned char packet[1500];
+ packet[0] = 0x80; // V=2, P=0, X=0, CC=0
+ packet[1] = 0x78; // M=0, PT = 120 (VP8)
+ EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 1500));
+ EXPECT_NE(0, ViE.network->ReceivedRTCPPacket(
+ tbChannel.videoChannel, packet, 1500));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 1500));
+ EXPECT_EQ(0, ViE.network->ReceivedRTCPPacket(
+ tbChannel.videoChannel, packet, 1500));
+ EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 11));
+ EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 11));
+ EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 3000));
+ EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
+ tbChannel.videoChannel, packet, 3000));
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.network->DeregisterSendTransport(
+ tbChannel.videoChannel)); // Sending
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.network->DeregisterSendTransport(
+ tbChannel.videoChannel)); // Already deregistered
+
+ //
+ // Local receiver
+ //
+ // TODO (perkj) change when B 4239431 is fixed.
+ /*error = ViE.ptrViENetwork->SetLocalReceiver(tbChannel.videoChannel,
+ 1234, 1234, "127.0.0.1");
+ numberOfErrors += ViETest::TestError(error == 0,
+ "ERROR: %s at line %d",
+ __FUNCTION__, __LINE__);*/
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
+
+ unsigned short rtpPort = 0;
+ unsigned short rtcpPort = 0;
+ char ipAddress[64];
+ memset(ipAddress, 0, 64);
+ EXPECT_EQ(0, ViE.network->GetLocalReceiver(
+ tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
+ EXPECT_EQ(0, ViE.network->GetLocalReceiver(
+ tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+
+ //
+ // Send destination
+ //
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, "127.0.0.1", 1236, 1237, 1234, 1235));
+
+ unsigned short sourceRtpPort = 0;
+ unsigned short sourceRtcpPort = 0;
+ EXPECT_EQ(0, ViE.network->GetSendDestination(
+ tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
+ sourceRtpPort, sourceRtcpPort));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ // Not allowed while sending
+ EXPECT_NE(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
+ EXPECT_EQ(kViENetworkAlreadySending, ViE.base->LastError());
+
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.network->GetSendDestination(
+ tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
+ sourceRtpPort, sourceRtcpPort));
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ //
+ // Address information
+ //
+
+ // GetSourceInfo: Tested in functional test
+ EXPECT_EQ(0, ViE.network->GetLocalIP(ipAddress, false));
+
+ // TODO(unknown): IPv6
+
+ //
+ // Filter
+ //
+ EXPECT_NE(0, ViE.network->GetSourceFilter(
+ tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, 1234, 1235, "10.10.10.10"));
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
+ EXPECT_EQ(0, ViE.network->GetSourceFilter(
+ tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
+ EXPECT_EQ(0, ViE.network->SetSourceFilter(
+ tbChannel.videoChannel, 0, 0, NULL));
+ EXPECT_NE(0, ViE.network->GetSourceFilter(
+ tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
+ }
+ {
+ TbVideoChannel tbChannel(ViE); // Create a video channel
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234));
+
+ int DSCP = 0;
+ bool useSetSockOpt = false;
+ // SetSockOpt should work without a locally bind socket
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+ EXPECT_EQ(0, DSCP);
+
+ // Invalid input
+ EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, -1, true));
+
+ // Invalid input
+ EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, 64, true));
+
+ // Valid
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 20, true));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt));
+
+ EXPECT_EQ(20, DSCP);
+ EXPECT_TRUE(useSetSockOpt);
+
+ // Disable
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, true));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt));
+ EXPECT_EQ(0, DSCP);
+
+ char myIpAddress[64];
+ memset(myIpAddress, 0, 64);
+ // Get local ip to be able to set ToS withtou setSockOpt
+ EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234, 1235, myIpAddress));
+
+ // Invalid input
+ EXPECT_NE(0, ViE.network->SetSendToS(
+ tbChannel.videoChannel, -1, false));
+ EXPECT_NE(0, ViE.network->SetSendToS(
+ tbChannel.videoChannel, 64, false)); // Invalid input
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
+ EXPECT_EQ(0, DSCP);
+ int tos_result = ViE.network->SetSendToS(
+ tbChannel.videoChannel, 20, false); // Valid
+ EXPECT_EQ(0, tos_result);
+ if (tos_result != 0)
+ {
+ ViETest::Log("ViESetSendToS error!.");
+ ViETest::Log("You must be admin to run these tests.");
+ ViETest::Log("On Win7 and late Vista, you need to right click the "
+ "exe and choose");
+ ViETest::Log("\"Run as administrator\"\n");
+ getchar();
+ }
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt));
+ EXPECT_EQ(20, DSCP);
+#ifdef _WIN32
+ EXPECT_FALSE(useSetSockOpt);
+#else // useSetSockOpt is true on Linux and Mac
+ EXPECT_TRUE(useSetSockOpt);
+#endif
+ EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, false));
+ EXPECT_EQ(0, ViE.network->GetSendToS(
+ tbChannel.videoChannel, DSCP, useSetSockOpt));
+ EXPECT_EQ(0, DSCP);
+ }
+ {
+ // From qos.h. (*) -> supported by ViE
+ //
+ // #define SERVICETYPE_NOTRAFFIC 0x00000000
+ // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
+ // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
+ // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
+ // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
+ // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
+ // #define SERVICETYPE_NOCHANGE 0x00000006
+ // #define SERVICETYPE_NONCONFORMING 0x00000009
+ // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
+ // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
+ //
+ // #define SERVICE_BESTEFFORT 0x80010000
+ // #define SERVICE_CONTROLLEDLOAD 0x80020000
+ // #define SERVICE_GUARANTEED 0x80040000
+ // #define SERVICE_QUALITATIVE 0x80200000
+
+ TbVideoChannel tbChannel(ViE); // Create a video channel
+
+
+#if defined(_WIN32)
+ // No socket
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
+
+ EXPECT_EQ(0, ViE.network->SetLocalReceiver(
+ tbChannel.videoChannel, 1234));
+
+ // Sender not initialized
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
+ EXPECT_EQ(0, ViE.network->SetSendDestination(
+ tbChannel.videoChannel, "127.0.0.1", 12345));
+
+ // Try to set all non-supported service types
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NETWORK_UNAVAILABLE));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_GENERAL_INFORMATION));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NOCHANGE));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NONCONFORMING));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_NETWORK_CONTROL));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICE_BESTEFFORT));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICE_CONTROLLEDLOAD));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICE_GUARANTEED));
+ EXPECT_NE(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICE_QUALITATIVE));
+
+ // Loop through valid service settings
+ bool enabled = false;
+ int serviceType = 0;
+ int overrideDSCP = 0;
+
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_FALSE(enabled);
+ EXPECT_EQ(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_TRUE(enabled);
+ EXPECT_EQ(SERVICETYPE_BESTEFFORT, serviceType);
+ EXPECT_FALSE(overrideDSCP);
+
+ EXPECT_EQ(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_CONTROLLEDLOAD));
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_TRUE(enabled);
+ EXPECT_EQ(SERVICETYPE_CONTROLLEDLOAD, serviceType);
+ EXPECT_FALSE(overrideDSCP);
+
+ EXPECT_EQ(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_GUARANTEED));
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_TRUE(enabled);
+ EXPECT_EQ(SERVICETYPE_GUARANTEED, serviceType);
+ EXPECT_FALSE(overrideDSCP);
+
+ EXPECT_EQ(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, true, SERVICETYPE_QUALITATIVE));
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_TRUE(enabled);
+ EXPECT_EQ(SERVICETYPE_QUALITATIVE, serviceType);
+ EXPECT_FALSE(overrideDSCP);
+
+ EXPECT_EQ(0, ViE.network->SetSendGQoS(
+ tbChannel.videoChannel, false, SERVICETYPE_QUALITATIVE));
+ EXPECT_EQ(0, ViE.network->GetSendGQoS(
+ tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
+ EXPECT_FALSE(enabled);
+#endif
+ }
+ {
+ //
+ // MTU and packet burst
+ //
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE);
+ // Invalid input
+ EXPECT_NE(0, ViE.network->SetMTU(tbChannel.videoChannel, 1600));
+ // Valid input
+ EXPECT_EQ(0, ViE.network->SetMTU(tbChannel.videoChannel, 800));
+
+ //
+ // Observer and timeout
+ //
+ ViEAutoTestNetworkObserver vieTestObserver;
+ EXPECT_EQ(0, ViE.network->RegisterObserver(
+ tbChannel.videoChannel, vieTestObserver));
+ EXPECT_NE(0, ViE.network->RegisterObserver(
+ tbChannel.videoChannel, vieTestObserver));
+ EXPECT_EQ(0, ViE.network->SetPeriodicDeadOrAliveStatus(
+ tbChannel.videoChannel, true)); // No observer
+ EXPECT_EQ(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
+
+ EXPECT_NE(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.network->SetPeriodicDeadOrAliveStatus(
+ tbChannel.videoChannel, true)); // No observer
+
+ // Packet timout notification
+ EXPECT_EQ(0, ViE.network->SetPacketTimeoutNotification(
+ tbChannel.videoChannel, true, 10));
+ }
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc
new file mode 100644
index 0000000..2a73a6c
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc
@@ -0,0 +1,296 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_render.cc
+//
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "engine_configurations.h"
+
+#include "video_render.h"
+
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "tb_capture_device.h"
+
+#if defined(WIN32)
+#include <windows.h>
+#include <ddraw.h>
+#include <tchar.h>
+#elif defined(WEBRTC_LINUX)
+ //From windgi.h
+ #undef RGB
+ #define RGB(r,g,b) ((unsigned long)(((unsigned char)(r)|((unsigned short)((unsigned char)(g))<<8))|(((unsigned long)(unsigned char)(b))<<16)))
+ //From ddraw.h
+/* typedef struct _DDCOLORKEY
+ {
+ DWORD dwColorSpaceLowValue; // low boundary of color space that is to
+ DWORD dwColorSpaceHighValue; // high boundary of color space that is
+ } DDCOLORKEY;*/
+#elif defined(WEBRTC_MAC)
+#endif
+
+class ViEAutoTestExternalRenderer: public webrtc::ExternalRenderer
+{
+public:
+ ViEAutoTestExternalRenderer() :
+ _width(0),
+ _height(0)
+ {
+ }
+ virtual int FrameSizeChange(unsigned int width, unsigned int height,
+ unsigned int numberOfStreams)
+ {
+ _width = width;
+ _height = height;
+ return 0;
+ }
+
+ virtual int DeliverFrame(unsigned char* buffer, int bufferSize,
+ uint32_t time_stamp,
+ int64_t render_time)
+ {
+ if (bufferSize != _width * _height * 3 / 2)
+ {
+ ViETest::Log("incorrect render buffer received, of length = %d\n",
+ bufferSize);
+ return 0;
+ }
+ ViETest::Log("callback DeliverFrame is good\n");
+ return 0;
+ }
+
+public:
+ virtual ~ViEAutoTestExternalRenderer()
+ {
+ }
+private:
+ int _width, _height;
+};
+
+void ViEAutoTest::ViERenderStandardTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+ int rtpPort = 6000;
+
+ TbInterfaces ViE("ViERenderStandardTest");
+
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ TbCaptureDevice tbCapture(ViE); // Create a capture device
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+ tbChannel.StartReceive(rtpPort);
+ tbChannel.StartSend(rtpPort);
+
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
+
+ ViETest::Log("\nCapture device is renderered in Window 1");
+ ViETest::Log("Remote stream is renderered in Window 2");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+
+ // PIP and full screen rendering is not supported on Android
+#ifndef WEBRTC_ANDROID
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window2, 0, 0.75, 0.75, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+
+ ViETest::Log("\nCapture device is now rendered in Window 2, PiP.");
+ ViETest::Log("Switching to full screen rendering in %d seconds.\n",
+ KAutoTestSleepTimeMs / 1000);
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
+
+ // Destroy render module and create new in full screen mode
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ _vrm1 = webrtc::VideoRender::CreateVideoRender(
+ 4563, _window1, true, _renderType);
+ EXPECT_TRUE(_vrm1 != NULL);
+
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window1, 0, 0.75f, 0.75f, 1.0f, 1.0f));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbChannel.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
+
+ // Destroy full screen render module and create new in normal mode
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ _vrm1 = webrtc::VideoRender::CreateVideoRender(
+ 4561, _window1, false, _renderType);
+ EXPECT_TRUE(_vrm1 != NULL);
+#endif
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+ tbCapture.Disconnect(tbChannel.videoChannel);
+}
+
+void ViEAutoTest::ViERenderExtendedTest()
+{
+ int rtpPort = 6000;
+
+ TbInterfaces ViE("ViERenderExtendedTest");
+
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ TbCaptureDevice tbCapture(ViE); // Create a capture device
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+ tbChannel.StartReceive(rtpPort);
+ tbChannel.StartSend(rtpPort);
+
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
+ EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
+
+ ViETest::Log("\nCapture device is renderered in Window 1");
+ ViETest::Log("Remote stream is renderered in Window 2");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+#ifdef _WIN32
+ ViETest::Log("\nConfiguring Window2");
+ ViETest::Log("you will see video only in first quadrant");
+ EXPECT_EQ(0, ViE.render->ConfigureRender(
+ tbChannel.videoChannel, 0, 0.0f, 0.0f, 0.5f, 0.5f));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("you will see video only in fourth quadrant");
+ EXPECT_EQ(0, ViE.render->ConfigureRender(
+ tbChannel.videoChannel, 0, 0.5f, 0.5f, 1.0f, 1.0f));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("normal video on Window2");
+ EXPECT_EQ(0, ViE.render->ConfigureRender(
+ tbChannel.videoChannel, 0, 0.0f, 0.0f, 1.0f, 1.0f));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+#endif
+
+ ViETest::Log("Mirroring Local Preview (Window1) Left-Right");
+ EXPECT_EQ(0, ViE.render->MirrorRenderStream(
+ tbCapture.captureId, true, false, true));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("\nMirroring Local Preview (Window1) Left-Right and Up-Down");
+ EXPECT_EQ(0, ViE.render->MirrorRenderStream(
+ tbCapture.captureId, true, true, true));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("\nMirroring Remote Window(Window2) Up-Down");
+ EXPECT_EQ(0, ViE.render->MirrorRenderStream(
+ tbChannel.videoChannel, true, true, false));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("Disabling Mirroing on Window1 and Window2");
+ EXPECT_EQ(0, ViE.render->MirrorRenderStream(
+ tbCapture.captureId, false, false, false));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+ EXPECT_EQ(0, ViE.render->MirrorRenderStream(
+ tbChannel.videoChannel, false, false, false));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("\nEnabling Full Screen render in 5 sec");
+
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
+
+ // Destroy render module and create new in full screen mode
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ _vrm1 = webrtc::VideoRender::CreateVideoRender(
+ 4563, _window1, true, _renderType);
+ EXPECT_TRUE(_vrm1 != NULL);
+
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, _window1, 0, 0.0f, 0.0f, 1.0f, 1.0f));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ ViETest::Log("\nStop renderer");
+ EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
+ ViETest::Log("\nRemove renderer");
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
+
+ // Destroy full screen render module and create new for external rendering
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ _vrm1 = webrtc::VideoRender::CreateVideoRender(4564, NULL, false,
+ _renderType);
+ EXPECT_TRUE(_vrm1 != NULL);
+
+ EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
+
+ ViETest::Log("\nExternal Render Test");
+ ViEAutoTestExternalRenderer externalRenderObj;
+ EXPECT_EQ(0, ViE.render->AddRenderer(
+ tbCapture.captureId, webrtc::kVideoI420, &externalRenderObj));
+ EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+ EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
+
+ // Destroy render module for external rendering and create new in normal
+ // mode
+ webrtc::VideoRender::DestroyVideoRender(_vrm1);
+ _vrm1 = NULL;
+ _vrm1 = webrtc::VideoRender::CreateVideoRender(
+ 4561, _window1, false, _renderType);
+ EXPECT_TRUE(_vrm1 != NULL);
+ tbCapture.Disconnect(tbChannel.videoChannel);
+}
+
+void ViEAutoTest::ViERenderAPITest()
+{
+ // TODO(unknown): add the real tests cases
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
new file mode 100644
index 0000000..4ea8383
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -0,0 +1,682 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_rtp_rtcp.cc
+//
+#include <iostream>
+
+#include "engine_configurations.h"
+#include "tb_capture_device.h"
+#include "tb_external_transport.h"
+#include "tb_interfaces.h"
+#include "tb_video_channel.h"
+#include "testsupport/fileutils.h"
+#include "vie_autotest.h"
+#include "vie_autotest_defines.h"
+
+class ViERtpObserver: public webrtc::ViERTPObserver
+{
+public:
+ ViERtpObserver()
+ {
+ }
+ virtual ~ViERtpObserver()
+ {
+ }
+
+ virtual void IncomingSSRCChanged(const int videoChannel,
+ const unsigned int SSRC)
+ {
+ }
+ virtual void IncomingCSRCChanged(const int videoChannel,
+ const unsigned int CSRC, const bool added)
+ {
+ }
+};
+
+class ViERtcpObserver: public webrtc::ViERTCPObserver
+{
+public:
+ int _channel;
+ unsigned char _subType;
+ unsigned int _name;
+ char* _data;
+ unsigned short _dataLength;
+
+ ViERtcpObserver() :
+ _channel(-1),
+ _subType(0),
+ _name(-1),
+ _data(NULL),
+ _dataLength(0)
+ {
+ }
+ ~ViERtcpObserver()
+ {
+ if (_data)
+ {
+ delete[] _data;
+ }
+ }
+ virtual void OnApplicationDataReceived(
+ const int videoChannel, const unsigned char subType,
+ const unsigned int name, const char* data,
+ const unsigned short dataLengthInBytes)
+ {
+ _channel = videoChannel;
+ _subType = subType;
+ _name = name;
+ if (dataLengthInBytes > _dataLength)
+ {
+ delete[] _data;
+ _data = NULL;
+ }
+ if (_data == NULL)
+ {
+ _data = new char[dataLengthInBytes];
+ }
+ memcpy(_data, data, dataLengthInBytes);
+ _dataLength = dataLengthInBytes;
+ }
+};
+
+void ViEAutoTest::ViERtpRtcpStandardTest()
+{
+ // ***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ // ***************************************************************
+
+ // Create VIE
+ TbInterfaces ViE("ViERtpRtcpStandardTest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ ViETest::Log("\n");
+ TbExternalTransport myTransport(*(ViE.network));
+
+ EXPECT_EQ(0, ViE.network->RegisterSendTransport(
+ tbChannel.videoChannel, myTransport));
+
+ // ***************************************************************
+ // Engine ready. Begin testing class
+ // ***************************************************************
+
+ unsigned short startSequenceNumber = 12345;
+ ViETest::Log("Set start sequence number: %u", startSequenceNumber);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, startSequenceNumber));
+
+ myTransport.EnableSequenceNumberCheck();
+
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ AutoTestSleep(2000);
+
+ unsigned short receivedSequenceNumber =
+ myTransport.GetFirstSequenceNumber();
+ ViETest::Log("First received sequence number: %u\n",
+ receivedSequenceNumber);
+ EXPECT_EQ(startSequenceNumber, receivedSequenceNumber);
+
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ //
+ // RTCP CName
+ //
+ ViETest::Log("Testing CName\n");
+ const char* sendCName = "ViEAutoTestCName\0";
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(tbChannel.videoChannel, sendCName));
+
+ char returnCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
+ memset(returnCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
+ tbChannel.videoChannel, returnCName));
+ EXPECT_STRCASEEQ(sendCName, returnCName);
+
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ AutoTestSleep(1000);
+
+ char remoteCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
+ memset(remoteCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteRTCPCName(
+ tbChannel.videoChannel, remoteCName));
+ EXPECT_STRCASEEQ(sendCName, remoteCName);
+
+ //
+ // Statistics
+ //
+ // Stop and restart to clear stats
+ ViETest::Log("Testing statistics\n");
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ myTransport.ClearStats();
+ int rate = 20;
+ myTransport.SetPacketLoss(rate);
+
+ // Start send to verify sending stats
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, startSequenceNumber));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ unsigned short sentFractionsLost = 0;
+ unsigned int sentCumulativeLost = 0;
+ unsigned int sentExtendedMax = 0;
+ unsigned int sentJitter = 0;
+ int sentRttMs = 0;
+ unsigned short recFractionsLost = 0;
+ unsigned int recCumulativeLost = 0;
+ unsigned int recExtendedMax = 0;
+ unsigned int recJitter = 0;
+ int recRttMs = 0;
+
+ unsigned int sentTotalBitrate = 0;
+ unsigned int sentVideoBitrate = 0;
+ unsigned int sentFecBitrate = 0;
+ unsigned int sentNackBitrate = 0;
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
+ tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
+ sentFecBitrate, sentNackBitrate));
+
+ EXPECT_GT(sentTotalBitrate, 0u);
+ EXPECT_EQ(sentFecBitrate, 0u);
+ EXPECT_EQ(sentNackBitrate, 0u);
+
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+
+ AutoTestSleep(2000);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetSentRTCPStatistics(
+ tbChannel.videoChannel, sentFractionsLost, sentCumulativeLost,
+ sentExtendedMax, sentJitter, sentRttMs));
+ EXPECT_GT(sentCumulativeLost, 0u);
+ EXPECT_GT(sentExtendedMax, startSequenceNumber);
+ EXPECT_GT(sentJitter, 0u);
+ EXPECT_GT(sentRttMs, 0);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetReceivedRTCPStatistics(
+ tbChannel.videoChannel, recFractionsLost, recCumulativeLost,
+ recExtendedMax, recJitter, recRttMs));
+
+ EXPECT_GT(recCumulativeLost, 0u);
+ EXPECT_GT(recExtendedMax, startSequenceNumber);
+ EXPECT_GT(recJitter, 0u);
+ EXPECT_GT(recRttMs, 0);
+
+ unsigned int estimated_bandwidth = 0;
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedSendBandwidth(
+ tbChannel.videoChannel,
+ &estimated_bandwidth));
+ EXPECT_GT(estimated_bandwidth, 0u);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedReceiveBandwidth(
+ tbChannel.videoChannel,
+ &estimated_bandwidth));
+ EXPECT_GT(estimated_bandwidth, 0u);
+
+ // Check that rec stats extended max is greater than what we've sent.
+ EXPECT_GE(recExtendedMax, sentExtendedMax);
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ //
+ // Test bandwidth statistics with NACK and FEC separately
+ //
+
+ myTransport.ClearStats();
+ myTransport.SetPacketLoss(rate);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
+ tbChannel.videoChannel, true, 96, 97));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
+ tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
+ sentFecBitrate, sentNackBitrate));
+
+ EXPECT_GT(sentTotalBitrate, 0u);
+ EXPECT_GE(sentFecBitrate, 10u);
+ EXPECT_EQ(sentNackBitrate, 0u);
+
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
+ tbChannel.videoChannel, false, 96, 97));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
+ tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
+ sentFecBitrate, sentNackBitrate));
+
+ // TODO(holmer): Write a non-flaky verification of this API.
+ // numberOfErrors += ViETest::TestError(sentTotalBitrate > 0 &&
+ // sentFecBitrate == 0 &&
+ // sentNackBitrate > 0,
+ // "ERROR: %s at line %d",
+ // __FUNCTION__, __LINE__);
+
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, false));
+
+ //
+ // Keepalive
+ //
+ ViETest::Log("Testing RTP keep alive...\n");
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+
+ myTransport.SetPacketLoss(0);
+ myTransport.ClearStats();
+
+ const char keepAlivePT = 109;
+ unsigned int deltaTimeSeconds = 2;
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, keepAlivePT, deltaTimeSeconds));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, false, keepAlivePT, deltaTimeSeconds));
+
+ WebRtc_Word32 numRtpPackets = 0;
+ WebRtc_Word32 numDroppedPackets = 0;
+ WebRtc_Word32 numRtcpPackets = 0;
+ myTransport.GetStats(numRtpPackets, numDroppedPackets, numRtcpPackets);
+ WebRtc_Word32 expectedPackets = KAutoTestSleepTimeMs / (1000 *
+ static_cast<WebRtc_Word32>(deltaTimeSeconds));
+ EXPECT_EQ(expectedPackets, numRtpPackets);
+
+ // Test to set SSRC
+ unsigned int setSSRC = 0x01234567;
+ ViETest::Log("Set SSRC %u", setSSRC);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, setSSRC));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ myTransport.EnableSSRCCheck();
+
+ AutoTestSleep(2000);
+ unsigned int receivedSSRC = myTransport.ReceivedSSRC();
+ ViETest::Log("Received SSRC %u\n", receivedSSRC);
+ EXPECT_EQ(setSSRC, receivedSSRC);
+
+ unsigned int localSSRC = 0;
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, localSSRC));
+ EXPECT_EQ(setSSRC, localSSRC);
+
+ unsigned int remoteSSRC = 0;
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteSSRC(
+ tbChannel.videoChannel, remoteSSRC));
+ EXPECT_EQ(setSSRC, remoteSSRC);
+
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ ViETest::Log("Testing RTP dump...\n");
+
+ std::string inDumpName =
+ ViETest::GetResultOutputPath() + "IncomingRTPDump.rtp";
+ std::string outDumpName =
+ ViETest::GetResultOutputPath() + "OutgoingRTPDump.rtp";
+ EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
+ tbChannel.videoChannel, inDumpName.c_str(), webrtc::kRtpIncoming));
+ EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
+ tbChannel.videoChannel, outDumpName.c_str(), webrtc::kRtpOutgoing));
+
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ AutoTestSleep(1000);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpIncoming));
+ EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpOutgoing));
+
+ // Make sure data was actually saved to the file and we stored the same
+ // amount of data in both files
+ FILE* inDump = fopen(inDumpName.c_str(), "r");
+ fseek(inDump, 0L, SEEK_END);
+ long inEndPos = ftell(inDump);
+ fclose(inDump);
+ FILE* outDump = fopen(outDumpName.c_str(), "r");
+ fseek(outDump, 0L, SEEK_END);
+ long outEndPos = ftell(outDump);
+ fclose(outDump);
+
+ EXPECT_GT(inEndPos, 0);
+ EXPECT_LT(inEndPos, outEndPos + 100);
+
+ // Deregister external transport
+ EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
+
+void ViEAutoTest::ViERtpRtcpExtendedTest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+ ViERtpRtcpStandardTest();
+
+ // Create VIE
+ TbInterfaces ViE("ViERtpRtcpExtendedTest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ //tbChannel.StartReceive(rtpPort);
+ //tbChannel.StartSend(rtpPort);
+ TbExternalTransport myTransport(*(ViE.network));
+
+ EXPECT_EQ(0, ViE.network->RegisterSendTransport(
+ tbChannel.videoChannel, myTransport));
+ EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ //
+ // Application specific RTCP
+ //
+ //
+
+ ViERtcpObserver rtcpObserver;
+ EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
+ tbChannel.videoChannel, rtcpObserver));
+
+ unsigned char subType = 3;
+ unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
+ const char* data = "ViEAutoTest Data of length 32 -\0";
+ const unsigned short numBytes = 32;
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, data, numBytes));
+
+ ViETest::Log("Sending RTCP application data...\n");
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(subType, rtcpObserver._subType);
+ EXPECT_STRCASEEQ(data, rtcpObserver._data);
+ EXPECT_EQ(name, rtcpObserver._name);
+ EXPECT_EQ(numBytes, rtcpObserver._dataLength);
+
+ ViETest::Log("\t RTCP application data received\n");
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+ EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
+ EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
+
+ EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
+}
+
+void ViEAutoTest::ViERtpRtcpAPITest()
+{
+ //***************************************************************
+ // Begin create/initialize WebRTC Video Engine for testing
+ //***************************************************************
+ // Create VIE
+ TbInterfaces ViE("ViERtpRtcpAPITest");
+ // Create a video channel
+ TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
+ // Create a capture device
+ TbCaptureDevice tbCapture(ViE);
+ tbCapture.ConnectTo(tbChannel.videoChannel);
+
+ //***************************************************************
+ // Engine ready. Begin testing class
+ //***************************************************************
+
+ //
+ // Check different RTCP modes
+ //
+ webrtc::ViERTCPMode rtcpMode = webrtc::kRtcpNone;
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
+ tbChannel.videoChannel, rtcpMode));
+ EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
+ tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
+ tbChannel.videoChannel, rtcpMode));
+ EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
+ tbChannel.videoChannel, webrtc::kRtcpNonCompound_RFC5506));
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
+ tbChannel.videoChannel, rtcpMode));
+ EXPECT_EQ(webrtc::kRtcpNonCompound_RFC5506, rtcpMode);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
+ tbChannel.videoChannel, webrtc::kRtcpNone));
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
+ tbChannel.videoChannel, rtcpMode));
+ EXPECT_EQ(webrtc::kRtcpNone, rtcpMode);
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
+ tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
+
+ //
+ // CName is testedn in SimpleTest
+ // Start sequence number is tested in SimplTEst
+ //
+ const char* testCName = "ViEAutotestCName";
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(
+ tbChannel.videoChannel, testCName));
+
+ char returnCName[256];
+ memset(returnCName, 0, 256);
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
+ tbChannel.videoChannel, returnCName));
+ EXPECT_STRCASEEQ(testCName, returnCName);
+
+ //
+ // SSRC
+ //
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
+ tbChannel.videoChannel, 0x01234567));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
+ tbChannel.videoChannel, 0x76543210));
+
+ unsigned int ssrc = 0;
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, ssrc));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, 1000));
+ tbChannel.StartSend();
+ EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, 12345));
+ tbChannel.StopSend();
+
+ //
+ // Start sequence number
+ //
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, 12345));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, 1000));
+ tbChannel.StartSend();
+ EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
+ tbChannel.videoChannel, 12345));
+ tbChannel.StopSend();
+
+ //
+ // Application specific RTCP
+ //
+ {
+ unsigned char subType = 3;
+ unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
+ const char* data = "ViEAutoTest Data of length 32 --";
+ const unsigned short numBytes = 32;
+
+ tbChannel.StartSend();
+ EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, data, numBytes));
+ EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, NULL, numBytes)) <<
+ "Should fail on NULL input.";
+ EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, data, numBytes - 1)) <<
+ "Should fail on incorrect length.";
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
+ tbChannel.videoChannel, rtcpMode));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, data, numBytes));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
+ tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
+ tbChannel.StopSend();
+ EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ tbChannel.videoChannel, subType, name, data, numBytes));
+ }
+
+ //
+ // Statistics
+ //
+ // Tested in SimpleTest(), we'll get errors if we haven't received a RTCP
+ // packet.
+
+ //
+ // RTP Keepalive
+ //
+ {
+ char setPT = 123;
+ unsigned int setDeltaTime = 10;
+ bool enabled = false;
+ char getPT = 0;
+ unsigned int getDeltaTime = 0;
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, 119));
+ EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, setPT, setDeltaTime));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, false, setPT, setDeltaTime));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, setPT, setDeltaTime));
+ EXPECT_EQ(0, ViE.rtp_rtcp->GetRTPKeepAliveStatus(
+ tbChannel.videoChannel, enabled, getPT, getDeltaTime));
+
+ EXPECT_TRUE(enabled);
+ EXPECT_EQ(setPT, getPT);
+ EXPECT_EQ(setDeltaTime, getDeltaTime);
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, false, setPT, setDeltaTime));
+
+ EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
+
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, setPT, setDeltaTime));
+
+ EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, true, setPT, setDeltaTime));
+
+ tbChannel.StopSend();
+ EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, enabled, getPT, 0));
+ EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
+ tbChannel.videoChannel, enabled, getPT, 61));
+ }
+ //
+ // RTP Dump
+ //
+ {
+#ifdef WEBRTC_ANDROID
+ const char* dumpName = "/sdcard/DumpFileName.rtp";
+#else
+ std::string output_file = webrtc::test::OutputPath() +
+ "DumpFileName.rtp";
+ const char* dumpName = output_file.c_str();
+#endif
+ EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
+ tbChannel.videoChannel, dumpName, webrtc::kRtpIncoming));
+ EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpIncoming));
+ EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpIncoming));
+ EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
+ tbChannel.videoChannel, dumpName, webrtc::kRtpOutgoing));
+ EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpOutgoing));
+ EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
+ tbChannel.videoChannel, webrtc::kRtpOutgoing));
+ EXPECT_NE(0, ViE.rtp_rtcp->StartRTPDump(
+ tbChannel.videoChannel, dumpName, (webrtc::RTPDirections) 3));
+ }
+ //
+ // RTP/RTCP Observers
+ //
+ {
+ ViERtpObserver rtpObserver;
+ EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTPObserver(
+ tbChannel.videoChannel, rtpObserver));
+ EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTPObserver(
+ tbChannel.videoChannel, rtpObserver));
+ EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTPObserver(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTPObserver(
+ tbChannel.videoChannel));
+
+ ViERtcpObserver rtcpObserver;
+ EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
+ tbChannel.videoChannel, rtcpObserver));
+ EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTCPObserver(
+ tbChannel.videoChannel, rtcpObserver));
+ EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
+ tbChannel.videoChannel));
+ EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
+ tbChannel.videoChannel));
+ }
+ //
+ // PLI
+ //
+ {
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
+ tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
+ }
+ //
+ // NACK
+ //
+ {
+ EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
+ }
+
+ //***************************************************************
+ // Testing finished. Tear down Video Engine
+ //***************************************************************
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
new file mode 100644
index 0000000..6d77115
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
@@ -0,0 +1,531 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iostream>
+
+#include "common_types.h"
+#include "tb_external_transport.h"
+#include "voe_base.h"
+#include "vie_autotest_defines.h"
+#include "vie_autotest.h"
+#include "vie_base.h"
+#include "vie_capture.h"
+#include "vie_codec.h"
+#include "vie_network.h"
+#include "vie_render.h"
+#include "vie_rtp_rtcp.h"
+
+#define VCM_RED_PAYLOAD_TYPE 96
+#define VCM_ULPFEC_PAYLOAD_TYPE 97
+
+int VideoEngineSimulcastTest(void* window1, void* window2)
+{
+ //********************************************************
+ // Begin create/initialize Video Engine for testing
+ //********************************************************
+
+ int error = 0;
+
+ //
+ // Create a VideoEngine instance
+ //
+ webrtc::VideoEngine* ptrViE = NULL;
+ ptrViE = webrtc::VideoEngine::Create();
+ if (ptrViE == NULL)
+ {
+ printf("ERROR in VideoEngine::Create\n");
+ return -1;
+ }
+
+ error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceLevel\n");
+ return -1;
+ }
+
+
+ std::string trace_file =
+ ViETest::GetResultOutputPath() + "ViESimulcast_trace.txt";
+ error = ptrViE->SetTraceFile(trace_file.c_str());
+ if (error == -1)
+ {
+ printf("ERROR in VideoEngine::SetTraceFile\n");
+ return -1;
+ }
+
+ //
+ // Init VideoEngine and create a channel
+ //
+ webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViEBase::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViEBase->Init();
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::Init\n");
+ return -1;
+ }
+
+ int videoChannel = -1;
+ error = ptrViEBase->CreateChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::CreateChannel\n");
+ return -1;
+ }
+
+ //
+ // List available capture devices, allocate and connect.
+ //
+ webrtc::ViECapture* ptrViECapture =
+ webrtc::ViECapture::GetInterface(ptrViE);
+ if (ptrViEBase == NULL)
+ {
+ printf("ERROR in ViECapture::GetInterface\n");
+ return -1;
+ }
+
+ const unsigned int KMaxDeviceNameLength = 128;
+ const unsigned int KMaxUniqueIdLength = 256;
+ char deviceName[KMaxDeviceNameLength];
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ char uniqueId[KMaxUniqueIdLength];
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ printf("Available capture devices:\n");
+ int captureIdx = 0;
+ for (captureIdx = 0;
+ captureIdx < ptrViECapture->NumberOfCaptureDevices();
+ captureIdx++)
+ {
+ memset(deviceName, 0, KMaxDeviceNameLength);
+ memset(uniqueId, 0, KMaxUniqueIdLength);
+
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId,
+ KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+ printf("\t %d. %s\n", captureIdx + 1, deviceName);
+ }
+ printf("\nChoose capture device: ");
+#ifdef WEBRTC_ANDROID
+ captureIdx = 0;
+ printf("0\n");
+#else
+ if (scanf("%d", &captureIdx) != 1)
+ {
+ printf("Error in scanf()\n");
+ return -1;
+ }
+ getchar();
+ captureIdx = captureIdx - 1; // Compensate for idx start at 1.
+#endif
+ error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
+ KMaxDeviceNameLength, uniqueId,
+ KMaxUniqueIdLength);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::GetCaptureDevice\n");
+ return -1;
+ }
+
+ int captureId = 0;
+ error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
+ captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::AllocateCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ConnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StartCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StartCapture\n");
+ return -1;
+ }
+
+ //
+ // RTP/RTCP settings
+ //
+ webrtc::ViERTP_RTCP* ptrViERtpRtcp =
+ webrtc::ViERTP_RTCP::GetInterface(ptrViE);
+ if (ptrViERtpRtcp == NULL)
+ {
+ printf("ERROR in ViERTP_RTCP::GetInterface\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
+ webrtc::kRtcpCompound_RFC4585);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
+ return -1;
+ }
+
+ error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
+ videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
+ return -1;
+ }
+
+ //
+ // Set up rendering
+ //
+ webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
+ if (ptrViERender == NULL)
+ {
+ printf("ERROR in ViERender::GetInterface\n");
+ return -1;
+ }
+
+ error
+ = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
+ 1.0);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::AddRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StartRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StartRender\n");
+ return -1;
+ }
+
+ //
+ // Setup codecs
+ //
+ webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
+ if (ptrViECodec == NULL)
+ {
+ printf("ERROR in ViECodec::GetInterface\n");
+ return -1;
+ }
+
+ // Check available codecs and prepare receive codecs
+ printf("\nAvailable codecs:\n");
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ int codecIdx = 0;
+ for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++)
+ {
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+ // try to keep the test frame size small when I420
+ if (videoCodec.codecType != webrtc::kVideoCodecVP8)
+ {
+ continue;
+ }
+ error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetReceiveCodec\n");
+ return -1;
+ }
+ if (videoCodec.codecType != webrtc::kVideoCodecRED
+ && videoCodec.codecType != webrtc::kVideoCodecULPFEC)
+ {
+ printf("\t %d. %s\n", codecIdx + 1, videoCodec.plName);
+ }
+ break;
+ }
+ error = ptrViECodec->GetCodec(codecIdx, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::GetCodec\n");
+ return -1;
+ }
+
+ // Set spatial resolution option
+ videoCodec.width = 1280;
+ videoCodec.height = 720;
+
+ // simulcast settings
+ videoCodec.numberOfSimulcastStreams = 3;
+ videoCodec.simulcastStream[0].width = 320;
+ videoCodec.simulcastStream[0].height = 180;
+ videoCodec.simulcastStream[0].numberOfTemporalLayers = 0;
+ videoCodec.simulcastStream[0].maxBitrate = 100;
+ videoCodec.simulcastStream[0].qpMax = videoCodec.qpMax;
+
+ videoCodec.simulcastStream[1].width = 640;
+ videoCodec.simulcastStream[1].height = 360;
+ videoCodec.simulcastStream[1].numberOfTemporalLayers = 0;
+ videoCodec.simulcastStream[1].maxBitrate = 500;
+ videoCodec.simulcastStream[1].qpMax = videoCodec.qpMax;
+
+ videoCodec.simulcastStream[2].width = 1280;
+ videoCodec.simulcastStream[2].height = 720;
+ videoCodec.simulcastStream[2].numberOfTemporalLayers = 0;
+ videoCodec.simulcastStream[2].maxBitrate = 1200;
+ videoCodec.simulcastStream[2].qpMax = videoCodec.qpMax;
+
+ // Set start bit rate
+ std::string str;
+ std::cout << std::endl;
+ std::cout << "Choose start rate (in kbps). Press enter for default: ";
+ std::getline(std::cin, str);
+ int startRate = atoi(str.c_str());
+ if(startRate != 0)
+ {
+ videoCodec.startBitrate=startRate;
+ }
+
+ error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::SetSendCodec\n");
+ return -1;
+ }
+ //
+ // Address settings
+ //
+ webrtc::ViENetwork* ptrViENetwork =
+ webrtc::ViENetwork::GetInterface(ptrViE);
+ if (ptrViENetwork == NULL)
+ {
+ printf("ERROR in ViENetwork::GetInterface\n");
+ return -1;
+ }
+
+ // Setting External transport
+ TbExternalTransport extTransport(*(ptrViENetwork));
+
+ error = ptrViENetwork->RegisterSendTransport(videoChannel,
+ extTransport);
+ if (error == -1)
+ {
+ printf("ERROR in ViECodec::RegisterSendTransport \n");
+ return -1;
+ }
+
+ extTransport.SetPacketLoss(0);
+
+ // Set network delay value
+ extTransport.SetNetworkDelay(10);
+
+ extTransport.SetSSRCFilter(3);
+
+ for (int idx = 0; idx < 3; idx++)
+ {
+ error = ptrViERtpRtcp->SetLocalSSRC(videoChannel,
+ idx+1, // SSRC
+ webrtc::kViEStreamTypeNormal,
+ idx);
+ if (error == -1)
+ {
+ printf("ERROR in ViERTP_RTCP::SetLocalSSRC(idx:%d)\n", idx);
+ return -1;
+ }
+ }
+
+ error = ptrViEBase->StartReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StartSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViENetwork::StartSend\n");
+ return -1;
+ }
+
+ //********************************************************
+ // Engine started
+ //********************************************************
+
+ printf("\nSimulcast call started\n\n");
+ do
+ {
+ printf("Enter new SSRC filter 1,2 or 3\n");
+ printf("Press enter to stop...");
+ str.clear();
+ std::getline(std::cin, str);
+ if (!str.empty())
+ {
+ int ssrc = atoi(str.c_str());
+ if (ssrc > 0 && ssrc < 4)
+ {
+ extTransport.SetSSRCFilter(ssrc);
+ } else
+ {
+ printf("Invalid SSRC\n");
+ }
+ } else
+ {
+ break;
+ }
+ } while (true);
+
+ //********************************************************
+ // Testing finished. Tear down Video Engine
+ //********************************************************
+
+ error = ptrViEBase->StopReceive(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopReceive\n");
+ return -1;
+ }
+
+ error = ptrViEBase->StopSend(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::StopSend\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViERender->StopRender(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::StopRender\n");
+ return -1;
+ }
+
+ error = ptrViERender->RemoveRenderer(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViERender::RemoveRenderer\n");
+ return -1;
+ }
+
+ error = ptrViECapture->StopCapture(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::StopCapture\n");
+ return -1;
+ }
+
+ error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViECapture->ReleaseCaptureDevice(captureId);
+ if (error == -1)
+ {
+ printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
+ return -1;
+ }
+
+ error = ptrViEBase->DeleteChannel(videoChannel);
+ if (error == -1)
+ {
+ printf("ERROR in ViEBase::DeleteChannel\n");
+ return -1;
+ }
+
+ int remainingInterfaces = 0;
+ remainingInterfaces = ptrViECodec->Release();
+ remainingInterfaces += ptrViECapture->Release();
+ remainingInterfaces += ptrViERtpRtcp->Release();
+ remainingInterfaces += ptrViERender->Release();
+ remainingInterfaces += ptrViENetwork->Release();
+ remainingInterfaces += ptrViEBase->Release();
+ if (remainingInterfaces > 0)
+ {
+ printf("ERROR: Could not release all interfaces\n");
+ return -1;
+ }
+
+ bool deleted = webrtc::VideoEngine::Delete(ptrViE);
+ if (deleted == false)
+ {
+ printf("ERROR in VideoEngine::Delete\n");
+ return -1;
+ }
+ return 0;
+
+ //
+ // END: VideoEngine 3.0 Sample Code
+ //
+ // ===================================================================
+}
+
+int ViEAutoTest::ViESimulcastCall()
+{
+ ViETest::Log(" ");
+ ViETest::Log("========================================");
+ ViETest::Log(" ViE Autotest Simulcast Call\n");
+
+ if (VideoEngineSimulcastTest(_window1, _window2) == 0)
+ {
+ ViETest::Log(" ");
+ ViETest::Log(" ViE Autotest Simulcast Call Done");
+ ViETest::Log("========================================");
+ ViETest::Log(" ");
+
+ return 0;
+ }
+ ViETest::Log(" ");
+ ViETest::Log(" ViE Autotest Simulcast Call Failed");
+ ViETest::Log("========================================");
+ ViETest::Log(" ");
+ return 1;
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc
new file mode 100755
index 0000000..7887767
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc
@@ -0,0 +1,214 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// vie_autotest_windows.cc
+//
+
+#include "vie_autotest_windows.h"
+
+#include "vie_autotest_defines.h"
+#include "vie_autotest_main.h"
+
+#include "engine_configurations.h"
+#include "critical_section_wrapper.h"
+#include "thread_wrapper.h"
+
+#include <windows.h>
+
+#ifdef _DEBUG
+//#include "vld.h"
+#endif
+
+// Disable Visual studio warnings
+// 'this' : used in base member initializer list
+#pragma warning(disable: 4355)
+// new behavior: elements of array 'XXX' will be default initialized
+#pragma warning(disable: 4351)
+
+LRESULT CALLBACK ViEAutoTestWinProc(HWND hWnd, UINT uMsg, WPARAM wParam,
+ LPARAM lParam) {
+ switch (uMsg) {
+ case WM_DESTROY:
+ PostQuitMessage( WM_QUIT);
+ break;
+ case WM_COMMAND:
+ break;
+ }
+ return DefWindowProc(hWnd, uMsg, wParam, lParam);
+}
+
+ViEAutoTestWindowManager::ViEAutoTestWindowManager()
+ : _window1(NULL),
+ _window2(NULL),
+ _terminate(false),
+ _eventThread(*webrtc::ThreadWrapper::CreateThread(
+ EventProcess, this, webrtc::kNormalPriority,
+ "ViEAutotestEventThread")),
+ _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+ _hwnd1(NULL),
+ _hwnd2(NULL),
+ _hwnd1Size(),
+ _hwnd2Size(),
+ _hwnd1Title(),
+ _hwnd2Title() {
+}
+
+ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
+ if (_hwnd1) {
+ ViEDestroyWindow(_hwnd1);
+ }
+ if (_hwnd2) {
+ ViEDestroyWindow(_hwnd1);
+ }
+ delete &_crit;
+}
+
+void* ViEAutoTestWindowManager::GetWindow1() {
+ return _window1;
+}
+
+void* ViEAutoTestWindowManager::GetWindow2() {
+ return _window2;
+}
+
+int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
+ AutoTestRect window2Size,
+ void* window1Title,
+ void* window2Title) {
+ _hwnd1Size.Copy(window1Size);
+ _hwnd2Size.Copy(window2Size);
+ memcpy(_hwnd1Title, window1Title, TITLE_LENGTH);
+ memcpy(_hwnd2Title, window2Title, TITLE_LENGTH);
+
+ unsigned int tId = 0;
+ _eventThread.Start(tId);
+
+ do {
+ _crit.Enter();
+ if (_window1 != NULL) {
+ break;
+ }
+ _crit.Leave();
+ AutoTestSleep(10);
+ } while (true);
+ _crit.Leave();
+ return 0;
+}
+
+int ViEAutoTestWindowManager::TerminateWindows() {
+ _eventThread.SetNotAlive();
+
+ _terminate = true;
+ if (_eventThread.Stop()) {
+ _crit.Enter();
+ delete &_eventThread;
+ _crit.Leave();
+ }
+
+ return 0;
+}
+
+bool ViEAutoTestWindowManager::EventProcess(void* obj) {
+ return static_cast<ViEAutoTestWindowManager*> (obj)->EventLoop();
+}
+
+bool ViEAutoTestWindowManager::EventLoop() {
+ _crit.Enter();
+
+ ViECreateWindow(_hwnd1, _hwnd1Size.origin.x, _hwnd1Size.origin.y,
+ _hwnd1Size.size.width, _hwnd1Size.size.height, _hwnd1Title);
+ ViECreateWindow(_hwnd2, _hwnd2Size.origin.x, _hwnd2Size.origin.y,
+ _hwnd2Size.size.width, _hwnd2Size.size.height, _hwnd2Title);
+
+ _window1 = (void*) _hwnd1;
+ _window2 = (void*) _hwnd2;
+ MSG msg;
+ while (!_terminate) {
+ if (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
+ TranslateMessage(&msg);
+ DispatchMessage(&msg);
+ }
+ _crit.Leave();
+ AutoTestSleep(10);
+ _crit.Enter();
+ }
+ ViEDestroyWindow(_hwnd1);
+ ViEDestroyWindow(_hwnd2);
+ _crit.Leave();
+
+ return false;
+}
+
+int ViEAutoTestWindowManager::ViECreateWindow(HWND &hwndMain, int xPos,
+ int yPos, int width, int height,
+ TCHAR* className) {
+ HINSTANCE hinst = GetModuleHandle(0);
+ WNDCLASSEX wcx;
+ wcx.hInstance = hinst;
+ wcx.lpszClassName = className;
+ wcx.lpfnWndProc = (WNDPROC) ViEAutoTestWinProc;
+ wcx.style = CS_DBLCLKS;
+ wcx.hIcon = LoadIcon(NULL, IDI_APPLICATION);
+ wcx.hIconSm = LoadIcon(NULL, IDI_APPLICATION);
+ wcx.hCursor = LoadCursor(NULL, IDC_ARROW);
+ wcx.lpszMenuName = NULL;
+ wcx.cbSize = sizeof(WNDCLASSEX);
+ wcx.cbClsExtra = 0;
+ wcx.cbWndExtra = 0;
+ wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
+
+ RegisterClassEx(&wcx);
+
+ // Create the main window.
+ hwndMain = CreateWindowEx(0, // no extended styles
+ className, // class name
+ className, // window name
+ WS_OVERLAPPED | WS_THICKFRAME, // overlapped window
+ xPos, // horizontal position
+ yPos, // vertical position
+ width, // width
+ height, // height
+ (HWND) NULL, // no parent or owner window
+ (HMENU) NULL, // class menu used
+ hinst, // instance handle
+ NULL); // no window creation data
+
+ if (!hwndMain) {
+ int error = GetLastError();
+ return -1;
+ }
+
+ // Show the window using the flag specified by the program
+ // that started the application, and send the application
+ // a WM_PAINT message.
+ ShowWindow(hwndMain, SW_SHOWDEFAULT);
+ UpdateWindow(hwndMain);
+
+ ::SetWindowPos(hwndMain, HWND_TOP, xPos, yPos, width, height,
+ SWP_FRAMECHANGED);
+
+ return 0;
+}
+
+int ViEAutoTestWindowManager::ViEDestroyWindow(HWND& hwnd) {
+ ::DestroyWindow(hwnd);
+ return 0;
+}
+
+bool ViEAutoTestWindowManager::SetTopmostWindow() {
+ // Meant to put terminal window on top
+ return true;
+}
+
+int main(int argc, char* argv[]) {
+ ViEAutoTestMain auto_test;
+ return auto_test.RunTests(argc, argv);
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc b/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
new file mode 100644
index 0000000..aff5118
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
@@ -0,0 +1,160 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
+
+#include "video_engine/test/auto_test/interface/tb_interfaces.h"
+#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "video_engine/test/auto_test/helpers/vie_fake_camera.h"
+#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
+#include "video_engine/test/auto_test/primitives/base_primitives.h"
+#include "video_engine/test/auto_test/primitives/codec_primitives.h"
+#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
+#include "video_engine/test/auto_test/primitives/general_primitives.h"
+
+bool ViEFileBasedComparisonTests::TestCallSetup(
+ const std::string& i420_video_file,
+ int width,
+ int height,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer) {
+
+ TbInterfaces interfaces("TestCallSetup");
+
+ int video_channel = -1;
+ EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
+
+ ViEFakeCamera fake_camera(interfaces.capture);
+ if (!fake_camera.StartCameraInNewThread(i420_video_file,
+ width,
+ height)) {
+ // No point in continuing if we have no proper video source
+ ADD_FAILURE() << "Could not open input video " << i420_video_file <<
+ ": aborting test...";
+ return false;
+ }
+ int capture_id = fake_camera.capture_id();
+
+ // Apparently, we need to connect external capture devices, but we should
+ // not start them since the external device is not a proper device.
+ EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
+ capture_id, video_channel));
+
+ ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
+
+ webrtc::ViERender *render_interface = interfaces.render;
+
+ RenderToFile(render_interface, capture_id, local_file_renderer);
+ RenderToFile(render_interface, video_channel, remote_file_renderer);
+
+ // Run the test itself:
+ const WebRtc_UWord8* device_name =
+ reinterpret_cast<const WebRtc_UWord8*>("Fake Capture Device");
+
+ ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
+ interfaces.base, interfaces.network, video_channel,
+ device_name);
+
+ AutoTestSleep(KAutoTestSleepTimeMs);
+
+ EXPECT_EQ(0, interfaces.base->StopReceive(video_channel));
+
+ StopAndRemoveRenderers(interfaces.base, render_interface, video_channel,
+ capture_id);
+
+ interfaces.capture->DisconnectCaptureDevice(video_channel);
+
+ // Stop sending data, clean up the camera thread and release the capture
+ // device. Note that this all happens after StopEverything, so this
+ // tests that the system doesn't mind that the external capture device sends
+ // data after rendering has been stopped.
+ fake_camera.StopCamera();
+
+ EXPECT_EQ(0, interfaces.base->DeleteChannel(video_channel));
+ return true;
+}
+
+bool ViEFileBasedComparisonTests::TestCodecs(
+ const std::string& i420_video_file,
+ int width,
+ int height,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer) {
+
+ TbInterfaces interfaces = TbInterfaces("TestCodecs");
+
+ ViEFakeCamera fake_camera(interfaces.capture);
+ if (!fake_camera.StartCameraInNewThread(i420_video_file, width, height)) {
+ // No point in continuing if we have no proper video source
+ ADD_FAILURE() << "Could not open input video " << i420_video_file <<
+ ": aborting test...";
+ return false;
+ }
+
+ int video_channel = -1;
+ int capture_id = fake_camera.capture_id();
+
+ EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
+ EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
+ capture_id, video_channel));
+
+ ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
+
+ RenderToFile(interfaces.render, capture_id, local_file_renderer);
+ RenderToFile(interfaces.render, video_channel, remote_file_renderer);
+
+ // Force the codec resolution to what our input video is so we can make
+ // comparisons later. Our comparison algorithms wouldn't like scaling.
+ ::TestCodecs(interfaces, capture_id, video_channel, width, height);
+
+ fake_camera.StopCamera();
+ return true;
+}
+
+void ViEFileBasedComparisonTests::TestFullStack(
+ const std::string& i420_video_file,
+ int width,
+ int height,
+ int bit_rate_kbps,
+ int packet_loss_percent,
+ int network_delay_ms,
+ ViEToFileRenderer* local_file_renderer,
+ ViEToFileRenderer* remote_file_renderer,
+ FrameDropDetector* frame_drop_detector) {
+ TbInterfaces interfaces = TbInterfaces("TestFullStack");
+
+ // Setup camera capturing from file.
+ ViEFakeCamera fake_camera(interfaces.capture);
+ if (!fake_camera.StartCameraInNewThread(i420_video_file, width, height)) {
+ // No point in continuing if we have no proper video source
+ ADD_FAILURE() << "Could not open input video " << i420_video_file <<
+ ": aborting test...";
+ return;
+ }
+ int video_channel = -1;
+ int capture_id = fake_camera.capture_id();
+ EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
+
+ // Must set SSRC to avoid SSRC collision detection since we're sending and
+ // receiving from the same machine (that would cause frames being discarded
+ // and decoder reset).
+ EXPECT_EQ(0, interfaces.rtp_rtcp->SetLocalSSRC(video_channel, 12345));
+
+ EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
+ capture_id, video_channel));
+ ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
+ RenderToFile(interfaces.render, capture_id, local_file_renderer);
+ RenderToFile(interfaces.render, video_channel, remote_file_renderer);
+
+ ::TestFullStack(interfaces, capture_id, video_channel, width, height,
+ bit_rate_kbps, packet_loss_percent, network_delay_ms,
+ frame_drop_detector);
+ EXPECT_TRUE(fake_camera.StopCamera());
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
new file mode 100644
index 0000000..1b16878
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_window_manager_factory.h"
+
+#include "vie_autotest_linux.h"
+
+ViEAutoTestWindowManagerInterface*
+ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
+ return new ViEAutoTestWindowManager();
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
new file mode 100644
index 0000000..806d10f
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
@@ -0,0 +1,23 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_window_manager_factory.h"
+
+#include "engine_configurations.h"
+#if defined(COCOA_RENDERING)
+#include "vie_autotest_mac_cocoa.h"
+#elif defined(CARBON_RENDERING)
+#include "vie_autotest_mac_carbon.h"
+#endif
+
+ViEAutoTestWindowManagerInterface*
+ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
+ return new ViEAutoTestWindowManager();
+}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
new file mode 100644
index 0000000..11114fd
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "vie_window_manager_factory.h"
+
+#include "vie_autotest_windows.h"
+
+ViEAutoTestWindowManagerInterface*
+ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
+ return new ViEAutoTestWindowManager();
+}
diff --git a/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi b/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi
new file mode 100644
index 0000000..cf403b7
--- /dev/null
+++ b/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi
@@ -0,0 +1,151 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'vie_auto_test',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/modules/modules.gyp:video_render_module',
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module',
+ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
+ '<(webrtc_root)/../test/metrics.gyp:metrics',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ 'video_engine_core',
+ ],
+ 'include_dirs': [
+ 'interface/',
+ 'helpers/',
+ 'primitives',
+ '../../include',
+ '../..',
+ '../../../modules/video_coding/codecs/interface',
+ '../../../common_video/interface',
+ ],
+ 'sources': [
+ 'interface/tb_capture_device.h',
+ 'interface/tb_external_transport.h',
+ 'interface/tb_I420_codec.h',
+ 'interface/tb_interfaces.h',
+ 'interface/tb_video_channel.h',
+ 'interface/vie_autotest.h',
+ 'interface/vie_autotest_defines.h',
+ 'interface/vie_autotest_linux.h',
+ 'interface/vie_autotest_mac_carbon.h',
+ 'interface/vie_autotest_mac_cocoa.h',
+ 'interface/vie_autotest_main.h',
+ 'interface/vie_autotest_window_manager_interface.h',
+ 'interface/vie_autotest_windows.h',
+ 'interface/vie_file_based_comparison_tests.h',
+ 'interface/vie_window_manager_factory.h',
+
+ # Helper classes
+ 'helpers/bit_flip_encryption.cc',
+ 'helpers/bit_flip_encryption.h',
+ 'helpers/random_encryption.cc',
+ 'helpers/random_encryption.h',
+ 'helpers/vie_fake_camera.cc',
+ 'helpers/vie_fake_camera.h',
+ 'helpers/vie_file_capture_device.cc',
+ 'helpers/vie_file_capture_device.h',
+ 'helpers/vie_to_file_renderer.cc',
+ 'helpers/vie_to_file_renderer.h',
+ 'helpers/vie_window_creator.cc',
+ 'helpers/vie_window_creator.h',
+
+ # New, fully automated tests
+ 'automated/legacy_fixture.cc',
+ 'automated/two_windows_fixture.cc',
+ 'automated/vie_api_integration_test.cc',
+ 'automated/vie_extended_integration_test.cc',
+ 'automated/vie_rtp_fuzz_test.cc',
+ 'automated/vie_standard_integration_test.cc',
+ 'automated/vie_video_verification_test.cc',
+
+ # Test primitives
+ 'primitives/base_primitives.cc',
+ 'primitives/base_primitives.h',
+ 'primitives/codec_primitives.cc',
+ 'primitives/codec_primitives.h',
+ 'primitives/framedrop_primitives.h',
+ 'primitives/framedrop_primitives.cc',
+ 'primitives/framedrop_primitives_unittest.cc',
+ 'primitives/general_primitives.cc',
+ 'primitives/general_primitives.h',
+
+ # Platform independent
+ 'source/tb_capture_device.cc',
+ 'source/tb_external_transport.cc',
+ 'source/tb_I420_codec.cc',
+ 'source/tb_interfaces.cc',
+ 'source/tb_video_channel.cc',
+ 'source/vie_autotest.cc',
+ 'source/vie_autotest_base.cc',
+ 'source/vie_autotest_capture.cc',
+ 'source/vie_autotest_codec.cc',
+ 'source/vie_autotest_encryption.cc',
+ 'source/vie_autotest_file.cc',
+ 'source/vie_autotest_image_process.cc',
+ 'source/vie_autotest_loopback.cc',
+ 'source/vie_autotest_main.cc',
+ 'source/vie_autotest_network.cc',
+ 'source/vie_autotest_render.cc',
+ 'source/vie_autotest_rtp_rtcp.cc',
+ 'source/vie_autotest_custom_call.cc',
+ 'source/vie_autotest_simulcast.cc',
+ 'source/vie_file_based_comparison_tests.cc',
+
+ # Platform dependent
+ # Linux
+ 'source/vie_autotest_linux.cc',
+ 'source/vie_window_manager_factory_linux.cc',
+ # Mac
+ 'source/vie_autotest_cocoa_mac.mm',
+ 'source/vie_autotest_carbon_mac.cc',
+ 'source/vie_window_manager_factory_mac.mm',
+ # Windows
+ 'source/vie_autotest_win.cc',
+ 'source/vie_window_manager_factory_win.cc',
+ ],
+ 'conditions': [
+ # TODO(andrew): this likely isn't an actual dependency. It should be
+ # included in webrtc.gyp or video_engine.gyp instead.
+ ['OS=="win"', {
+ 'dependencies': [
+ 'vie_win_test',
+ ],
+ }],
+ ['OS=="linux"', {
+ # TODO(andrew): these should be provided directly by the projects
+ # # which require them instead.
+ 'libraries': [
+ '-lXext',
+ '-lX11',
+ ],
+ }],
+ ['OS=="mac"', {
+ 'xcode_settings': {
+ 'OTHER_LDFLAGS': [
+ '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL -framework CoreVideo -framework CoreAudio -framework AudioToolbox',
+ ],
+ },
+ }],
+ ], # conditions
+ },
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/video_engine.gyp b/trunk/src/video_engine/video_engine.gyp
new file mode 100644
index 0000000..261292c
--- /dev/null
+++ b/trunk/src/video_engine/video_engine.gyp
@@ -0,0 +1,25 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ './video_engine_core.gypi',
+ ],
+
+ # Test targets, excluded when building with Chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'includes': [
+ 'test/auto_test/vie_auto_test.gypi',
+ 'main/test/WindowsTest/windowstest.gypi',
+ ],
+ }],
+ ],
+}
+
diff --git a/trunk/src/video_engine/video_engine_core.gypi b/trunk/src/video_engine/video_engine_core.gypi
new file mode 100644
index 0000000..38015af
--- /dev/null
+++ b/trunk/src/video_engine/video_engine_core.gypi
@@ -0,0 +1,160 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'video_engine_core',
+ 'type': '<(library)',
+ 'dependencies': [
+
+ # common_video
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_jpeg',
+
+ # ModulesShared
+ '<(webrtc_root)/modules/modules.gyp:media_file',
+ '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+ '<(webrtc_root)/modules/modules.gyp:udp_transport',
+ '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+
+ # ModulesVideo
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module',
+ '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
+ '<(webrtc_root)/modules/modules.gyp:video_processing',
+ '<(webrtc_root)/modules/modules.gyp:video_render_module',
+
+ # VoiceEngine
+ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
+
+ # system_wrappers
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ 'include',
+ '../common_video/interface',
+ '../modules/video_capture/main/interface',
+ '../modules/video_render/main/interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ # interface
+ 'include/vie_base.h',
+ 'include/vie_capture.h',
+ 'include/vie_codec.h',
+ 'include/vie_encryption.h',
+ 'include/vie_errors.h',
+ 'include/vie_external_codec.h',
+ 'include/vie_file.h',
+ 'include/vie_image_process.h',
+ 'include/vie_network.h',
+ 'include/vie_render.h',
+ 'include/vie_rtp_rtcp.h',
+
+ # headers
+ 'vie_base_impl.h',
+ 'vie_capture_impl.h',
+ 'vie_codec_impl.h',
+ 'vie_defines.h',
+ 'vie_encryption_impl.h',
+ 'vie_external_codec_impl.h',
+ 'vie_file_impl.h',
+ 'vie_image_process_impl.h',
+ 'vie_impl.h',
+ 'vie_network_impl.h',
+ 'vie_ref_count.h',
+ 'vie_remb.h',
+ 'vie_render_impl.h',
+ 'vie_rtp_rtcp_impl.h',
+ 'vie_shared_data.h',
+ 'vie_capturer.h',
+ 'vie_channel.h',
+ 'vie_channel_manager.h',
+ 'vie_encoder.h',
+ 'vie_file_image.h',
+ 'vie_file_player.h',
+ 'vie_file_recorder.h',
+ 'vie_frame_provider_base.h',
+ 'vie_input_manager.h',
+ 'vie_manager_base.h',
+ 'vie_performance_monitor.h',
+ 'vie_receiver.h',
+ 'vie_renderer.h',
+ 'vie_render_manager.h',
+ 'vie_sender.h',
+ 'vie_sync_module.h',
+
+ # ViE
+ 'vie_base_impl.cc',
+ 'vie_capture_impl.cc',
+ 'vie_codec_impl.cc',
+ 'vie_encryption_impl.cc',
+ 'vie_external_codec_impl.cc',
+ 'vie_file_impl.cc',
+ 'vie_image_process_impl.cc',
+ 'vie_impl.cc',
+ 'vie_network_impl.cc',
+ 'vie_ref_count.cc',
+ 'vie_render_impl.cc',
+ 'vie_rtp_rtcp_impl.cc',
+ 'vie_shared_data.cc',
+ 'vie_capturer.cc',
+ 'vie_channel.cc',
+ 'vie_channel_manager.cc',
+ 'vie_encoder.cc',
+ 'vie_file_image.cc',
+ 'vie_file_player.cc',
+ 'vie_file_recorder.cc',
+ 'vie_frame_provider_base.cc',
+ 'vie_input_manager.cc',
+ 'vie_manager_base.cc',
+ 'vie_performance_monitor.cc',
+ 'vie_receiver.cc',
+ 'vie_remb.cc',
+ 'vie_renderer.cc',
+ 'vie_render_manager.cc',
+ 'vie_sender.cc',
+ 'vie_sync_module.cc',
+ ], # source
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'video_engine_core_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'video_engine_core',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'include_dirs': [
+ '..',
+ '../modules/interface',
+ '../modules/rtp_rtcp/interface',
+ ],
+ 'sources': [
+ 'vie_remb_unittest.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/vie_base_impl.cc b/trunk/src/video_engine/vie_base_impl.cc
new file mode 100644
index 0000000..4f0eae2
--- /dev/null
+++ b/trunk/src/video_engine/vie_base_impl.cc
@@ -0,0 +1,494 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "critical_section_wrapper.h"
+#include "engine_configurations.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+#include "video_coding.h"
+#include "video_processing.h"
+#include "video_render.h"
+#include "vie_base_impl.h"
+#include "vie_channel.h"
+#include "vie_channel_manager.h"
+#include "vie_defines.h"
+#include "vie_encoder.h"
+#include "vie_errors.h"
+#include "vie_impl.h"
+#include "vie_input_manager.h"
+#include "vie_performance_monitor.h"
+#include "vie_shared_data.h"
+
+namespace webrtc {
+
+ViEBase* ViEBase::GetInterface(VideoEngine* video_engine) {
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViEBaseImpl* vie_base_impl = vie_impl;
+ (*vie_base_impl)++; // Increase ref count.
+
+ return vie_base_impl;
+}
+
+int ViEBaseImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_.instance_id(),
+ "ViEBase::Release()");
+ (*this)--; // Decrease ref count.
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_.instance_id(),
+ "ViEBase release too many times");
+ shared_data_.SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_.instance_id(),
+ "ViEBase reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViEBaseImpl::ViEBaseImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_.instance_id(),
+ "ViEBaseImpl::ViEBaseImpl() Ctor");
+}
+
+ViEBaseImpl::~ViEBaseImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_.instance_id(),
+ "ViEBaseImpl::ViEBaseImpl() Dtor");
+}
+
+int ViEBaseImpl::Init() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_.instance_id(),
+ "Init");
+ if (shared_data_.Initialized()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_.instance_id(),
+ "Init called twice");
+ return 0;
+ }
+
+ shared_data_.SetInitialized();
+ return 0;
+}
+
+int ViEBaseImpl::SetVoiceEngine(VoiceEngine* voice_engine) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s", __FUNCTION__);
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+
+ if (shared_data_.channel_manager()->SetVoiceEngine(voice_engine) != 0) {
+ shared_data_.SetLastError(kViEBaseVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::CreateChannel(int& video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s", __FUNCTION__);
+
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+
+ if (shared_data_.channel_manager()->CreateChannel(video_channel) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: Could not create channel", __FUNCTION__);
+ video_channel = -1;
+ shared_data_.SetLastError(kViEBaseChannelCreationFailed);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: channel created: %d", __FUNCTION__, video_channel);
+ return 0;
+}
+
+int ViEBaseImpl::CreateChannel(int& video_channel, int original_channel) {
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ if (!cs.Channel(original_channel)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - original_channel does not exist.", __FUNCTION__,
+ shared_data_.instance_id());
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ if (shared_data_.channel_manager()->CreateChannel(video_channel,
+ original_channel) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: Could not create channel", __FUNCTION__);
+ video_channel = -1;
+ shared_data_.SetLastError(kViEBaseChannelCreationFailed);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: channel created: %d", __FUNCTION__, video_channel);
+ return 0;
+}
+
+int ViEBaseImpl::DeleteChannel(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s(%d)", __FUNCTION__, video_channel);
+
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+
+ {
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id()),
+ "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ // Deregister the ViEEncoder if no other channel is using it.
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (cs.ChannelUsingViEEncoder(video_channel) == false) {
+ ViEInputManagerScoped is(*(shared_data_.input_manager()));
+ ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
+ if (provider) {
+ provider->DeregisterFrameCallback(vie_encoder);
+ }
+ }
+ }
+
+ if (shared_data_.channel_manager()->DeleteChannel(video_channel) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: Could not delete channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: channel deleted: %d", __FUNCTION__, video_channel);
+ return 0;
+}
+
+int ViEBaseImpl::ConnectAudioChannel(const int video_channel,
+ const int audio_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s(%d)", __FUNCTION__, video_channel);
+
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ if (!cs.Channel(video_channel)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ if (shared_data_.channel_manager()->ConnectVoiceChannel(video_channel,
+ audio_channel) != 0) {
+ shared_data_.SetLastError(kViEBaseVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::DisconnectAudioChannel(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s(%d)", __FUNCTION__, video_channel);
+ if (!(shared_data_.Initialized())) {
+ shared_data_.SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_.instance_id());
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ if (!cs.Channel(video_channel)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ if (shared_data_.channel_manager()->DisconnectVoiceChannel(
+ video_channel) != 0) {
+ shared_data_.SetLastError(kViEBaseVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::StartSend(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Channel %d does not exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ // Verify no other channel using the same encoder is sending.
+ ChannelList channels;
+ cs.ChannelsUsingViEEncoder(video_channel, &channels);
+ for (ChannelList::iterator it = channels.begin(); it != channels.end();
+ ++it) {
+ if ((*it)->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "A channel using this encoder is already synding");
+ shared_data_.SetLastError(kViEBaseAlreadySending);
+ return -1;
+ }
+ }
+
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ assert(false);
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Could not find encoder for channel %d", __FUNCTION__,
+ video_channel);
+ return -1;
+ }
+
+ // Pause and trigger a key frame.
+ vie_encoder->Pause();
+ WebRtc_Word32 error = vie_channel->StartSend();
+ if (error != 0) {
+ vie_encoder->Restart();
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Could not start sending on channel %d", __FUNCTION__,
+ video_channel);
+ if (error == kViEBaseAlreadySending) {
+ shared_data_.SetLastError(kViEBaseAlreadySending);
+ }
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ vie_encoder->SendKeyFrame();
+ vie_encoder->Restart();
+ return 0;
+}
+
+int ViEBaseImpl::StopSend(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Channel %d does not exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+
+ WebRtc_Word32 error = vie_channel->StopSend();
+ if (error != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Could not stop sending on channel %d", __FUNCTION__,
+ video_channel);
+ if (error == kViEBaseNotSending) {
+ shared_data_.SetLastError(kViEBaseNotSending);
+ } else {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ }
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::StartReceive(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Channel %d does not exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Receiving()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Channel %d already receive.", __FUNCTION__,
+ video_channel);
+ shared_data_.SetLastError(kViEBaseAlreadyReceiving);
+ return -1;
+ }
+ if (vie_channel->StartReceive() != 0) {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::StopReceive(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_.instance_id(), video_channel),
+ "%s: Channel %d does not exist", __FUNCTION__, video_channel);
+ shared_data_.SetLastError(kViEBaseInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->StopReceive() != 0) {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEBaseImpl::RegisterObserver(ViEBaseObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s", __FUNCTION__);
+ if (shared_data_.vie_performance_monitor()->ViEBaseObserverRegistered()) {
+ shared_data_.SetLastError(kViEBaseObserverAlreadyRegistered);
+ return -1;
+ }
+ return shared_data_.vie_performance_monitor()->Init(&observer);
+}
+
+int ViEBaseImpl::DeregisterObserver() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "%s", __FUNCTION__);
+
+ if (!shared_data_.vie_performance_monitor()->ViEBaseObserverRegistered()) {
+ shared_data_.SetLastError(kViEBaseObserverNotRegistered);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_.instance_id(),
+ "%s No observer registered.", __FUNCTION__);
+ return -1;
+ }
+ shared_data_.vie_performance_monitor()->Terminate();
+ return 0;
+}
+
+int ViEBaseImpl::GetVersion(char version[1024]) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
+ "GetVersion(version=?)");
+ assert(kViEVersionMaxMessageSize == 1024);
+
+ if (!version) {
+ shared_data_.SetLastError(kViEBaseInvalidArgument);
+ return -1;
+ }
+
+ char version_buf[kViEVersionMaxMessageSize];
+ char* version_ptr = version_buf;
+
+ WebRtc_Word32 len = 0; // Does not include NULL termination.
+ WebRtc_Word32 acc_len = 0;
+
+ len = AddViEVersion(version_ptr);
+ if (len == -1) {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ version_ptr += len;
+ acc_len += len;
+ assert(acc_len < kViEVersionMaxMessageSize);
+
+ len = AddBuildInfo(version_ptr);
+ if (len == -1) {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ version_ptr += len;
+ acc_len += len;
+ assert(acc_len < kViEVersionMaxMessageSize);
+
+ len = AddExternalTransportBuild(version_ptr);
+ if (len == -1) {
+ shared_data_.SetLastError(kViEBaseUnknownError);
+ return -1;
+ }
+ version_ptr += len;
+ acc_len += len;
+ assert(acc_len < kViEVersionMaxMessageSize);
+
+ memcpy(version, version_buf, acc_len);
+ version[acc_len] = '\0';
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
+ ViEId(shared_data_.instance_id()), "GetVersion() => %s",
+ version);
+ return 0;
+}
+
+int ViEBaseImpl::LastError() {
+ return shared_data_.LastErrorInternal();
+}
+
+WebRtc_Word32 ViEBaseImpl::AddBuildInfo(char* str) const {
+ return sprintf(str, "Build: %s\n", BUILDINFO);
+}
+
+WebRtc_Word32 ViEBaseImpl::AddViEVersion(char* str) const {
+ return sprintf(str, "VideoEngine 3.2.0\n");
+}
+
+WebRtc_Word32 ViEBaseImpl::AddExternalTransportBuild(char* str) const {
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ return sprintf(str, "External transport build\n");
+#else
+ return 0;
+#endif
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_base_impl.h b/trunk/src/video_engine/vie_base_impl.h
new file mode 100644
index 0000000..a950300
--- /dev/null
+++ b/trunk/src/video_engine/vie_base_impl.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
+
+#include "vie_base.h"
+#include "vie_defines.h"
+#include "vie_ref_count.h"
+#include "vie_shared_data.h"
+
+namespace webrtc {
+
+class Module;
+class VoiceEngine;
+
+class ViEBaseImpl
+ : public ViEBase,
+ public ViERefCount {
+ public:
+ virtual int Release();
+
+ // Initializes VideoEngine and must be called before any other API is called.
+ virtual int Init();
+
+ // Connects ViE to a VoE instance. Pass in NULL to forget about a previously
+ // set voice engine and release all resources we allocated from it.
+ virtual int SetVoiceEngine(VoiceEngine* voice_engine);
+
+ // Creates a new ViE channel.
+ virtual int CreateChannel(int& video_channel);
+
+ // Creates a new ViE channel that will use the same capture device and encoder
+ // as |original_channel|.
+ virtual int CreateChannel(int& video_channel, int original_channel);
+
+ // Deletes a ViE channel.
+ virtual int DeleteChannel(const int video_channel);
+
+ // Connects a ViE channel with a VoE channel.
+ virtual int ConnectAudioChannel(const int video_channel,
+ const int audio_channel);
+
+ // Disconnects a video/voice channel pair.
+ virtual int DisconnectAudioChannel(const int video_channel);
+
+ // Starts sending on video_channel and also starts the encoder.
+ virtual int StartSend(const int video_channel);
+
+ // Stops sending on the specified channel.
+ virtual int StopSend(const int video_channel);
+
+ // Starts receiving on the channel and also start decoding.
+ virtual int StartReceive(const int video_channel);
+
+ // Stops receiving on the specified channel.
+ virtual int StopReceive(const int video_channel);
+
+ // Registers a customer implemented observer.
+ virtual int RegisterObserver(ViEBaseObserver& observer);
+
+ // Deregisters the observer.
+ virtual int DeregisterObserver();
+
+ // Prints version information into |version|.
+ virtual int GetVersion(char version[1024]);
+
+ // Returns the error code for the last registered error.
+ virtual int LastError();
+
+ protected:
+ ViEBaseImpl();
+ virtual ~ViEBaseImpl();
+
+ ViESharedData* shared_data() { return &shared_data_; }
+
+ private:
+ // Version functions.
+ WebRtc_Word32 AddViEVersion(char* str) const;
+ WebRtc_Word32 AddBuildInfo(char* str) const;
+ WebRtc_Word32 AddExternalTransportBuild(char* str) const;
+
+ // ViEBaseImpl owns ViESharedData used by all interface implementations.
+ ViESharedData shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_capture_impl.cc b/trunk/src/video_engine/vie_capture_impl.cc
new file mode 100644
index 0000000..a94c29c
--- /dev/null
+++ b/trunk/src/video_engine/vie_capture_impl.cc
@@ -0,0 +1,590 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_capture_impl.h"
+
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_input_manager.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViECapture* ViECapture::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViECaptureImpl* vie_capture_impl = vie_impl;
+ // Increase ref count.
+ (*vie_capture_impl)++;
+ return vie_capture_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViECaptureImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViECapture::Release()");
+ // Decrease ref count
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViECapture release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViECapture reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViECaptureImpl::ViECaptureImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViECaptureImpl::ViECaptureImpl() Ctor");
+}
+
+ViECaptureImpl::~ViECaptureImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViECaptureImpl::~ViECaptureImpl() Dtor");
+}
+
+int ViECaptureImpl::NumberOfCaptureDevices() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ return shared_data_->input_manager()->NumberOfCaptureDevices();
+}
+
+
+int ViECaptureImpl::GetCaptureDevice(unsigned int list_number,
+ char* device_nameUTF8,
+ unsigned int device_nameUTF8Length,
+ char* unique_idUTF8,
+ unsigned int unique_idUTF8Length) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(list_number: %d)", __FUNCTION__, list_number);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ return shared_data_->input_manager()->GetDeviceName(
+ list_number,
+ reinterpret_cast<WebRtc_UWord8*>(device_nameUTF8), device_nameUTF8Length,
+ reinterpret_cast<WebRtc_UWord8*>(unique_idUTF8), unique_idUTF8Length);
+}
+
+int ViECaptureImpl::AllocateCaptureDevice(
+ const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ int& capture_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(unique_idUTF8: %s)", __FUNCTION__, unique_idUTF8);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ const WebRtc_Word32 result =
+ shared_data_->input_manager()->CreateCaptureDevice(
+ reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
+ static_cast<const WebRtc_UWord32>(unique_idUTF8Length), capture_id);
+ if (result != 0) {
+ shared_data_->SetLastError(result);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::AllocateExternalCaptureDevice(
+ int& capture_id, ViEExternalCapture*& external_capture) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ const WebRtc_Word32 result =
+ shared_data_->input_manager()->CreateExternalCaptureDevice(
+ external_capture, capture_id);
+
+ if (result != 0) {
+ shared_data_->SetLastError(result);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::AllocateCaptureDevice(VideoCaptureModule& capture_module,
+ int& capture_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ const WebRtc_Word32 result =
+ shared_data_->input_manager()->CreateCaptureDevice(capture_module,
+ capture_id);
+ if (result != 0) {
+ shared_data_->SetLastError(result);
+ return -1;
+ }
+ return 0;
+}
+
+
+int ViECaptureImpl::ReleaseCaptureDevice(const int capture_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+ {
+ ViEInputManagerScoped is((*(shared_data_->input_manager())));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ }
+
+ // Destroy the capture device.
+ return shared_data_->input_manager()->DestroyCaptureDevice(capture_id);
+}
+
+int ViECaptureImpl::ConnectCaptureDevice(const int capture_id,
+ const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(capture_id: %d, video_channel: %d)", __FUNCTION__,
+ capture_id, video_channel);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
+ return -1;
+ }
+ // Check if the encoder already has a connected frame provider
+ if (is.FrameProvider(vie_encoder) != NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d already connected to a capture device.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECaptureDeviceAlreadyConnected);
+ return -1;
+ }
+ VideoCodec codec;
+ bool use_hardware_encoder = false;
+ if (vie_encoder->GetEncoder(codec) == 0) {
+ // Try to provide the encoder with pre-encoded frames if possible.
+ if (vie_capture->PreEncodeToViEEncoder(codec, *vie_encoder,
+ video_channel) == 0) {
+ use_hardware_encoder = true;
+ }
+ }
+ // If we don't use the camera as hardware encoder, we register the vie_encoder
+ // for callbacks.
+ if (!use_hardware_encoder &&
+ vie_capture->RegisterFrameCallback(video_channel, vie_encoder) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+
+int ViECaptureImpl::DisconnectCaptureDevice(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
+ return -1;
+ }
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
+ if (!frame_provider) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s: No capture device connected to channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECaptureDeviceNotConnected);
+ return -1;
+ }
+ if (frame_provider->Id() < kViECaptureIdBase ||
+ frame_provider->Id() > kViECaptureIdMax) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s: No capture device connected to channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECaptureDeviceNotConnected);
+ return -1;
+ }
+
+ if (frame_provider->DeregisterFrameCallback(vie_encoder) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+
+ return 0;
+}
+
+int ViECaptureImpl::StartCapture(const int capture_id,
+ const CaptureCapability& capture_capability) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (vie_capture->Started()) {
+ shared_data_->SetLastError(kViECaptureDeviceAlreadyStarted);
+ return -1;
+ }
+ if (vie_capture->Start(capture_capability) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::StopCapture(const int capture_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (!vie_capture->Started()) {
+ shared_data_->SetLastError(kViECaptureDeviceNotStarted);
+ return -1;
+ }
+ if (vie_capture->Stop() != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+
+ return 0;
+}
+
+int ViECaptureImpl::SetRotateCapturedFrames(
+ const int capture_id,
+ const RotateCapturedFrame rotation) {
+ int i_rotation = -1;
+ switch (rotation) {
+ case RotateCapturedFrame_0:
+ i_rotation = 0;
+ break;
+ case RotateCapturedFrame_90:
+ i_rotation = 90;
+ break;
+ case RotateCapturedFrame_180:
+ i_rotation = 180;
+ break;
+ case RotateCapturedFrame_270:
+ i_rotation = 270;
+ break;
+ }
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(rotation: %d)", __FUNCTION__, i_rotation);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (vie_capture->SetRotateCapturedFrames(rotation) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::SetCaptureDelay(const int capture_id,
+ const unsigned int capture_delay_ms) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d, capture_delay_ms %u)", __FUNCTION__,
+ capture_id, capture_delay_ms);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+
+ if (vie_capture->SetCaptureDelay(capture_delay_ms) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::NumberOfCapabilities(
+ const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
+
+#if defined(WEBRTC_MAC_INTEL)
+ // TODO(mflodman) Move to capture module!
+ // QTKit framework handles all capabilities and capture settings
+ // automatically (mandatory).
+ // Thus this function cannot be supported on the Mac platform.
+ shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s This API is not supported on Mac OS", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+#endif
+
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ return shared_data_->input_manager()->NumberOfCaptureCapabilities(
+ reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8));
+}
+
+
+int ViECaptureImpl::GetCaptureCapability(const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ const unsigned int capability_number,
+ CaptureCapability& capability) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
+
+#if defined(WEBRTC_MAC_INTEL)
+ // TODO(mflodman) Move to capture module!
+ // QTKit framework handles all capabilities and capture settings
+ // automatically (mandatory).
+ // Thus this function cannot be supported on the Mac platform.
+ shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s This API is not supported on Mac OS", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+#endif
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ if (shared_data_->input_manager()->GetCaptureCapability(
+ reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
+ capability_number, capability) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::ShowCaptureSettingsDialogBox(
+ const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ const char* dialog_title,
+ void* parent_window,
+ const unsigned int x,
+ const unsigned int y) {
+#if defined(WEBRTC_MAC_INTEL)
+ // TODO(mflodman) Move to capture module
+ // QTKit framework handles all capabilities and capture settings
+ // automatically (mandatory).
+ // Thus this function cannot be supported on the Mac platform.
+ shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s This API is not supported on Mac OS", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+#endif
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s capture_id (capture_device_name: %s)", __FUNCTION__,
+ unique_idUTF8);
+
+ return shared_data_->input_manager()->DisplayCaptureSettingsDialogBox(
+ reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
+ reinterpret_cast<const WebRtc_UWord8*>(dialog_title),
+ parent_window, x, y);
+}
+
+int ViECaptureImpl::GetOrientation(const char* unique_idUTF8,
+ RotateCapturedFrame& orientation) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s (capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
+
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ if (shared_data_->input_manager()->GetOrientation(
+ reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
+ orientation) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+
+int ViECaptureImpl::EnableBrightnessAlarm(const int capture_id,
+ const bool enable) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (vie_capture->EnableBrightnessAlarm(enable) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::RegisterObserver(const int capture_id,
+ ViECaptureObserver& observer) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (vie_capture->IsObserverRegistered()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Observer already registered", __FUNCTION__);
+ shared_data_->SetLastError(kViECaptureObserverAlreadyRegistered);
+ return -1;
+ }
+ if (vie_capture->RegisterObserver(observer) != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECaptureImpl::DeregisterObserver(const int capture_id) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+ return -1;
+ }
+ if (!vie_capture->IsObserverRegistered()) {
+ shared_data_->SetLastError(kViECaptureDeviceObserverNotRegistered);
+ return -1;
+ }
+
+ if (vie_capture->DeRegisterObserver() != 0) {
+ shared_data_->SetLastError(kViECaptureDeviceUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_capture_impl.h b/trunk/src/video_engine/vie_capture_impl.h
new file mode 100644
index 0000000..ca88a06
--- /dev/null
+++ b/trunk/src/video_engine/vie_capture_impl.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
+
+#include "typedefs.h"
+#include "video_engine/include/vie_capture.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViECaptureImpl
+ : public ViECapture,
+ public ViERefCount {
+ public:
+ // Implements ViECapture.
+ virtual int Release();
+ virtual int NumberOfCaptureDevices();
+ virtual int GetCaptureDevice(unsigned int list_number, char* device_nameUTF8,
+ const unsigned int device_nameUTF8Length,
+ char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length);
+ virtual int AllocateCaptureDevice(const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ int& capture_id);
+ virtual int AllocateCaptureDevice(VideoCaptureModule& capture_module,
+ int& capture_id);
+ virtual int AllocateExternalCaptureDevice(
+ int& capture_id, ViEExternalCapture *&external_capture);
+ virtual int ReleaseCaptureDevice(const int capture_id);
+
+ virtual int ConnectCaptureDevice(const int capture_id,
+ const int video_channel);
+ virtual int DisconnectCaptureDevice(const int video_channel);
+ virtual int StartCapture(
+ const int capture_id,
+ const CaptureCapability& capture_capability = CaptureCapability());
+ virtual int StopCapture(const int capture_id);
+ virtual int SetRotateCapturedFrames(const int capture_id,
+ const RotateCapturedFrame rotation);
+ virtual int SetCaptureDelay(const int capture_id,
+ const unsigned int capture_delay_ms);
+ virtual int NumberOfCapabilities(const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length);
+ virtual int GetCaptureCapability(const char* unique_idUTF8,
+ const unsigned int unique_idUTF8Length,
+ const unsigned int capability_number,
+ CaptureCapability& capability);
+ virtual int ShowCaptureSettingsDialogBox(
+ const char* unique_idUTF8, const unsigned int unique_idUTF8Length,
+ const char* dialog_title, void* parent_window = NULL,
+ const unsigned int x = 200, const unsigned int y = 200);
+ virtual int GetOrientation(const char* unique_idUTF8,
+ RotateCapturedFrame& orientation);
+ virtual int EnableBrightnessAlarm(const int capture_id, const bool enable);
+ virtual int RegisterObserver(const int capture_id,
+ ViECaptureObserver& observer);
+ virtual int DeregisterObserver(const int capture_id);
+
+ protected:
+ ViECaptureImpl(ViESharedData* shared_data);
+ virtual ~ViECaptureImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_capturer.cc b/trunk/src/video_engine/vie_capturer.cc
new file mode 100644
index 0000000..cd9bd0c
--- /dev/null
+++ b/trunk/src/video_engine/vie_capturer.cc
@@ -0,0 +1,900 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_capturer.h"
+
+#include "modules/interface/module_common_types.h"
+#include "modules/utility/interface/process_thread.h"
+#include "modules/video_capture/main/interface/video_capture_factory.h"
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/thread_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_image_process.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+
+namespace webrtc {
+
+const int kThreadWaitTimeMs = 100;
+const int kMaxDeliverWaitTime = 500;
+
+ViECapturer::ViECapturer(int capture_id,
+ int engine_id,
+ ProcessThread& module_process_thread)
+ : ViEFrameProviderBase(capture_id, engine_id),
+ capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ deliver_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ capture_module_(NULL),
+ external_capture_module_(NULL),
+ module_process_thread_(module_process_thread),
+ capture_id_(capture_id),
+ capture_thread_(*ThreadWrapper::CreateThread(ViECaptureThreadFunction,
+ this, kHighPriority,
+ "ViECaptureThread")),
+ capture_event_(*EventWrapper::Create()),
+ deliver_event_(*EventWrapper::Create()),
+ effect_filter_(NULL),
+ image_proc_module_(NULL),
+ image_proc_module_ref_counter_(0),
+ deflicker_frame_stats_(NULL),
+ brightness_frame_stats_(NULL),
+ current_brightness_level_(Normal),
+ reported_brightness_level_(Normal),
+ denoising_enabled_(false),
+ observer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ observer_(NULL),
+ encoding_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ capture_encoder_(NULL),
+ encode_complete_callback_(NULL),
+ vie_encoder_(NULL),
+ vcm_(NULL),
+ decoder_initialized_(false) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, capture_id),
+ "ViECapturer::ViECapturer(capture_id: %d, engine_id: %d)",
+ capture_id, engine_id);
+ unsigned int t_id = 0;
+ if (capture_thread_.Start(t_id)) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id, capture_id),
+ "%s: thread started: %u", __FUNCTION__, t_id);
+ } else {
+ assert(false);
+ }
+}
+
+ViECapturer::~ViECapturer() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "ViECapturer::~ViECapturer() - capture_id: %d, engine_id: %d",
+ capture_id_, engine_id_);
+
+ // Stop the thread.
+ deliver_cs_->Enter();
+ capture_cs_->Enter();
+ capture_thread_.SetNotAlive();
+ capture_event_.Set();
+ capture_cs_->Leave();
+ deliver_cs_->Leave();
+
+ provider_cs_->Enter();
+ if (vie_encoder_) {
+ vie_encoder_->DeRegisterExternalEncoder(codec_.plType);
+ }
+ provider_cs_->Leave();
+
+ // Stop the camera input.
+ if (capture_module_) {
+ module_process_thread_.DeRegisterModule(capture_module_);
+ capture_module_->DeRegisterCaptureDataCallback();
+ capture_module_->Release();
+ capture_module_ = NULL;
+ }
+ if (capture_thread_.Stop()) {
+ // Thread stopped.
+ delete &capture_thread_;
+ delete &capture_event_;
+ delete &deliver_event_;
+ } else {
+ assert(false);
+ WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer,
+ ViEId(engine_id_, capture_id_),
+ "%s: Not able to stop capture thread for device %d, leaking",
+ __FUNCTION__, capture_id_);
+ }
+
+ if (image_proc_module_) {
+ VideoProcessingModule::Destroy(image_proc_module_);
+ }
+ if (deflicker_frame_stats_) {
+ delete deflicker_frame_stats_;
+ deflicker_frame_stats_ = NULL;
+ }
+ delete brightness_frame_stats_;
+ if (vcm_) {
+ delete vcm_;
+ }
+}
+
+ViECapturer* ViECapturer::CreateViECapture(
+ int capture_id,
+ int engine_id,
+ VideoCaptureModule& capture_module,
+ ProcessThread& module_process_thread) {
+ ViECapturer* capture = new ViECapturer(capture_id, engine_id,
+ module_process_thread);
+ if (!capture || capture->Init(capture_module) != 0) {
+ delete capture;
+ capture = NULL;
+ }
+ return capture;
+}
+
+WebRtc_Word32 ViECapturer::Init(VideoCaptureModule& capture_module) {
+ assert(capture_module_ == NULL);
+ capture_module_ = &capture_module;
+ capture_module_->RegisterCaptureDataCallback(*this);
+ capture_module_->AddRef();
+ if (module_process_thread_.RegisterModule(capture_module_) != 0) {
+ return -1;
+ }
+
+ return 0;
+}
+
+ViECapturer* ViECapturer::CreateViECapture(
+ int capture_id,
+ int engine_id,
+ const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_unique_idUTF8Length,
+ ProcessThread& module_process_thread) {
+ ViECapturer* capture = new ViECapturer(capture_id, engine_id,
+ module_process_thread);
+ if (!capture ||
+ capture->Init(device_unique_idUTF8, device_unique_idUTF8Length) != 0) {
+ delete capture;
+ capture = NULL;
+ }
+ return capture;
+}
+
+WebRtc_Word32 ViECapturer::Init(
+ const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_unique_idUTF8Length) {
+ assert(capture_module_ == NULL);
+ if (device_unique_idUTF8 == NULL) {
+ capture_module_ = VideoCaptureFactory::Create(
+ ViEModuleId(engine_id_, capture_id_), external_capture_module_);
+ } else {
+ capture_module_ = VideoCaptureFactory::Create(
+ ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8);
+ }
+ if (!capture_module_) {
+ return -1;
+ }
+ capture_module_->AddRef();
+ capture_module_->RegisterCaptureDataCallback(*this);
+ if (module_process_thread_.RegisterModule(capture_module_) != 0) {
+ return -1;
+ }
+
+ return 0;
+}
+
+int ViECapturer::FrameCallbackChanged() {
+ if (Started() && !EncoderActive() && !CaptureCapabilityFixed()) {
+ // Reconfigure the camera if a new size is required and the capture device
+ // does not provide encoded frames.
+ int best_width;
+ int best_height;
+ int best_frame_rate;
+ VideoCaptureCapability capture_settings;
+ capture_module_->CaptureSettings(capture_settings);
+ GetBestFormat(best_width, best_height, best_frame_rate);
+ if (best_width != 0 && best_height != 0 && best_frame_rate != 0) {
+ if (best_width != capture_settings.width ||
+ best_height != capture_settings.height ||
+ best_frame_rate != capture_settings.maxFPS ||
+ capture_settings.codecType != kVideoCodecUnknown) {
+ Stop();
+ Start(requested_capability_);
+ }
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::Start(const CaptureCapability& capture_capability) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
+ __FUNCTION__);
+ int width;
+ int height;
+ int frame_rate;
+ VideoCaptureCapability capability;
+ requested_capability_ = capture_capability;
+ if (EncoderActive()) {
+ CriticalSectionScoped cs(encoding_cs_.get());
+ capability.width = codec_.width;
+ capability.height = codec_.height;
+ capability.maxFPS = codec_.maxFramerate;
+ capability.codecType = codec_.codecType;
+ capability.rawType = kVideoI420;
+
+ } else if (!CaptureCapabilityFixed()) {
+ // Ask the observers for best size.
+ GetBestFormat(width, height, frame_rate);
+ if (width == 0) {
+ width = kViECaptureDefaultWidth;
+ }
+ if (height == 0) {
+ height = kViECaptureDefaultHeight;
+ }
+ if (frame_rate == 0) {
+ frame_rate = kViECaptureDefaultFramerate;
+ }
+ capability.height = height;
+ capability.width = width;
+ capability.maxFPS = frame_rate;
+ capability.rawType = kVideoI420;
+ capability.codecType = kVideoCodecUnknown;
+ } else {
+ // Width, height and type specified with call to Start, not set by
+ // observers.
+ capability.width = requested_capability_.width;
+ capability.height = requested_capability_.height;
+ capability.maxFPS = requested_capability_.maxFPS;
+ capability.rawType = requested_capability_.rawType;
+ capability.interlaced = requested_capability_.interlaced;
+ }
+ return capture_module_->StartCapture(capability);
+}
+
+WebRtc_Word32 ViECapturer::Stop() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
+ __FUNCTION__);
+ requested_capability_ = CaptureCapability();
+ return capture_module_->StopCapture();
+}
+
+bool ViECapturer::Started() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
+ __FUNCTION__);
+ return capture_module_->CaptureStarted();
+}
+
+const WebRtc_UWord8* ViECapturer::CurrentDeviceName() const {
+ return capture_module_->CurrentDeviceName();
+}
+
+WebRtc_Word32 ViECapturer::SetCaptureDelay(WebRtc_Word32 delay_ms) {
+ return capture_module_->SetCaptureDelay(delay_ms);
+}
+
+WebRtc_Word32 ViECapturer::SetRotateCapturedFrames(
+ const RotateCapturedFrame rotation) {
+ VideoCaptureRotation converted_rotation = kCameraRotate0;
+ switch (rotation) {
+ case RotateCapturedFrame_0:
+ converted_rotation = kCameraRotate0;
+ break;
+ case RotateCapturedFrame_90:
+ converted_rotation = kCameraRotate90;
+ break;
+ case RotateCapturedFrame_180:
+ converted_rotation = kCameraRotate180;
+ break;
+ case RotateCapturedFrame_270:
+ converted_rotation = kCameraRotate270;
+ break;
+ }
+ return capture_module_->SetCaptureRotation(converted_rotation);
+}
+
+int ViECapturer::IncomingFrame(unsigned char* video_frame,
+ unsigned int video_frame_length,
+ unsigned short width,
+ unsigned short height,
+ RawVideoType video_type,
+ unsigned long long capture_time) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "ExternalCapture::IncomingFrame width %d, height %d, "
+ "capture_time %u", width, height, capture_time);
+
+ if (!external_capture_module_) {
+ return -1;
+ }
+ VideoCaptureCapability capability;
+ capability.width = width;
+ capability.height = height;
+ capability.rawType = video_type;
+ return external_capture_module_->IncomingFrame(video_frame,
+ video_frame_length,
+ capability, capture_time);
+}
+
+int ViECapturer::IncomingFrameI420(const ViEVideoFrameI420& video_frame,
+ unsigned long long capture_time) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "ExternalCapture::IncomingFrame width %d, height %d, "
+ " capture_time %u", video_frame.width, video_frame.height,
+ capture_time);
+
+ if (!external_capture_module_) {
+ return -1;
+ }
+
+ VideoFrameI420 frame;
+ frame.width = video_frame.width;
+ frame.height = video_frame.height;
+ frame.y_plane = video_frame.y_plane;
+ frame.u_plane = video_frame.u_plane;
+ frame.v_plane = video_frame.v_plane;
+ frame.y_pitch = video_frame.y_pitch;
+ frame.u_pitch = video_frame.u_pitch;
+ frame.v_pitch = video_frame.v_pitch;
+
+ return external_capture_module_->IncomingFrameI420(frame, capture_time);
+}
+
+void ViECapturer::OnIncomingCapturedFrame(const WebRtc_Word32 capture_id,
+ VideoFrame& video_frame,
+ VideoCodecType codec_type) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ CriticalSectionScoped cs(capture_cs_.get());
+ if (codec_type != kVideoCodecUnknown) {
+ if (encoded_frame_.Length() != 0) {
+ // The last encoded frame has not been sent yet. Need to wait.
+ deliver_event_.Reset();
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_id: %d) Last encoded frame not yet delivered.",
+ __FUNCTION__, capture_id);
+ capture_cs_->Leave();
+ // Wait for the coded frame to be sent before unblocking this.
+ deliver_event_.Wait(kMaxDeliverWaitTime);
+ assert(encoded_frame_.Length() == 0);
+ capture_cs_->Enter();
+ }
+ encoded_frame_.SwapFrame(video_frame);
+ } else {
+ captured_frame_.SwapFrame(video_frame);
+ }
+ capture_event_.Set();
+ return;
+}
+
+void ViECapturer::OnCaptureDelayChanged(const WebRtc_Word32 id,
+ const WebRtc_Word32 delay) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_id: %d) delay %d", __FUNCTION__, capture_id_,
+ delay);
+
+ // Deliver the network delay to all registered callbacks.
+ ViEFrameProviderBase::SetFrameDelay(delay);
+ CriticalSectionScoped cs(encoding_cs_.get());
+ if (vie_encoder_) {
+ vie_encoder_->DelayChanged(id, delay);
+ }
+}
+
+WebRtc_Word32 ViECapturer::RegisterEffectFilter(
+ ViEEffectFilter* effect_filter) {
+ CriticalSectionScoped cs(deliver_cs_.get());
+
+ if (!effect_filter) {
+ if (!effect_filter_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: no effect filter added for capture device %d",
+ __FUNCTION__, capture_id_);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: deregister effect filter for device %d", __FUNCTION__,
+ capture_id_);
+ } else {
+ if (effect_filter_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: effect filter already added for capture device %d",
+ __FUNCTION__, capture_id_);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: register effect filter for device %d", __FUNCTION__,
+ capture_id_);
+ }
+ effect_filter_ = effect_filter;
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::IncImageProcRefCount() {
+ if (!image_proc_module_) {
+ assert(image_proc_module_ref_counter_ == 0);
+ image_proc_module_ = VideoProcessingModule::Create(
+ ViEModuleId(engine_id_, capture_id_));
+ if (!image_proc_module_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: could not create video processing module",
+ __FUNCTION__);
+ return -1;
+ }
+ }
+ image_proc_module_ref_counter_++;
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::DecImageProcRefCount() {
+ image_proc_module_ref_counter_--;
+ if (image_proc_module_ref_counter_ == 0) {
+ // Destroy module.
+ VideoProcessingModule::Destroy(image_proc_module_);
+ image_proc_module_ = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::EnableDenoising(bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
+ capture_id_, enable);
+
+ CriticalSectionScoped cs(deliver_cs_.get());
+ if (enable) {
+ if (denoising_enabled_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: denoising already enabled", __FUNCTION__);
+ return -1;
+ }
+ denoising_enabled_ = true;
+ if (IncImageProcRefCount() != 0) {
+ return -1;
+ }
+ } else {
+ if (denoising_enabled_ == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: denoising not enabled", __FUNCTION__);
+ return -1;
+ }
+ denoising_enabled_ = false;
+ DecImageProcRefCount();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::EnableDeflickering(bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
+ capture_id_, enable);
+
+ CriticalSectionScoped cs(deliver_cs_.get());
+ if (enable) {
+ if (deflicker_frame_stats_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: deflickering already enabled", __FUNCTION__);
+ return -1;
+ }
+ if (IncImageProcRefCount() != 0) {
+ return -1;
+ }
+ deflicker_frame_stats_ = new VideoProcessingModule::FrameStats();
+ } else {
+ if (deflicker_frame_stats_ == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: deflickering not enabled", __FUNCTION__);
+ return -1;
+ }
+ DecImageProcRefCount();
+ delete deflicker_frame_stats_;
+ deflicker_frame_stats_ = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::EnableBrightnessAlarm(bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
+ capture_id_, enable);
+
+ CriticalSectionScoped cs(deliver_cs_.get());
+ if (enable) {
+ if (brightness_frame_stats_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: BrightnessAlarm already enabled", __FUNCTION__);
+ return -1;
+ }
+ if (IncImageProcRefCount() != 0) {
+ return -1;
+ }
+ brightness_frame_stats_ = new VideoProcessingModule::FrameStats();
+ } else {
+ DecImageProcRefCount();
+ if (brightness_frame_stats_ == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: deflickering not enabled", __FUNCTION__);
+ return -1;
+ }
+ delete brightness_frame_stats_;
+ brightness_frame_stats_ = NULL;
+ }
+ return 0;
+}
+
+bool ViECapturer::ViECaptureThreadFunction(void* obj) {
+ return static_cast<ViECapturer*>(obj)->ViECaptureProcess();
+}
+
+bool ViECapturer::ViECaptureProcess() {
+ if (capture_event_.Wait(kThreadWaitTimeMs) == kEventSignaled) {
+ deliver_cs_->Enter();
+ if (captured_frame_.Length() > 0) {
+ // New I420 frame.
+ capture_cs_->Enter();
+ deliver_frame_.SwapFrame(captured_frame_);
+ captured_frame_.SetLength(0);
+ capture_cs_->Leave();
+ DeliverI420Frame(deliver_frame_);
+ }
+ if (encoded_frame_.Length() > 0) {
+ capture_cs_->Enter();
+ deliver_frame_.SwapFrame(encoded_frame_);
+ encoded_frame_.SetLength(0);
+ deliver_event_.Set();
+ capture_cs_->Leave();
+ DeliverCodedFrame(deliver_frame_);
+ }
+ deliver_cs_->Leave();
+ if (current_brightness_level_ != reported_brightness_level_) {
+ CriticalSectionScoped cs(observer_cs_.get());
+ if (observer_) {
+ observer_->BrightnessAlarm(id_, current_brightness_level_);
+ reported_brightness_level_ = current_brightness_level_;
+ }
+ }
+ }
+ // We're done!
+ return true;
+}
+
+void ViECapturer::DeliverI420Frame(VideoFrame& video_frame) {
+ // Apply image enhancement and effect filter.
+ if (deflicker_frame_stats_) {
+ if (image_proc_module_->GetFrameStats(*deflicker_frame_stats_,
+ video_frame) == 0) {
+ image_proc_module_->Deflickering(video_frame, *deflicker_frame_stats_);
+ } else {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: could not get frame stats for captured frame",
+ __FUNCTION__);
+ }
+ }
+ if (denoising_enabled_) {
+ image_proc_module_->Denoising(video_frame);
+ }
+ if (brightness_frame_stats_) {
+ if (image_proc_module_->GetFrameStats(*brightness_frame_stats_,
+ video_frame) == 0) {
+ WebRtc_Word32 brightness = image_proc_module_->BrightnessDetection(
+ video_frame, *brightness_frame_stats_);
+
+ switch (brightness) {
+ case VideoProcessingModule::kNoWarning:
+ current_brightness_level_ = Normal;
+ break;
+ case VideoProcessingModule::kDarkWarning:
+ current_brightness_level_ = Dark;
+ break;
+ case VideoProcessingModule::kBrightWarning:
+ current_brightness_level_ = Bright;
+ break;
+ default:
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s: Brightness detection failed", __FUNCTION__);
+ }
+ }
+ }
+ if (effect_filter_) {
+ effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
+ video_frame.TimeStamp(), video_frame.Width(),
+ video_frame.Height());
+ }
+ // Deliver the captured frame to all observers (channels, renderer or file).
+ ViEFrameProviderBase::DeliverFrame(video_frame);
+}
+
+void ViECapturer::DeliverCodedFrame(VideoFrame& video_frame) {
+ if (encode_complete_callback_) {
+ EncodedImage encoded_image(video_frame.Buffer(), video_frame.Length(),
+ video_frame.Size());
+ encoded_image._timeStamp = 90 * (WebRtc_UWord32) video_frame.RenderTimeMs();
+ encode_complete_callback_->Encoded(encoded_image);
+ }
+
+ if (NumberOfRegisteredFrameCallbacks() > 0 && decoder_initialized_) {
+ video_frame.Swap(decode_buffer_.payloadData, decode_buffer_.bufferSize,
+ decode_buffer_.payloadSize);
+ decode_buffer_.encodedHeight = video_frame.Height();
+ decode_buffer_.encodedWidth = video_frame.Width();
+ decode_buffer_.renderTimeMs = video_frame.RenderTimeMs();
+ decode_buffer_.timeStamp = 90 * (WebRtc_UWord32) video_frame.RenderTimeMs();
+ decode_buffer_.payloadType = codec_.plType;
+ vcm_->DecodeFromStorage(decode_buffer_);
+ }
+}
+
+int ViECapturer::DeregisterFrameCallback(
+ const ViEFrameCallback* callbackObject) {
+ provider_cs_->Enter();
+ if (callbackObject == vie_encoder_) {
+ // Don't use this camera as encoder anymore. Need to tell the ViEEncoder.
+ ViEEncoder* vie_encoder = NULL;
+ vie_encoder = vie_encoder_;
+ vie_encoder_ = NULL;
+ provider_cs_->Leave();
+
+ // Need to take this here in order to avoid deadlock with VCM. The reason is
+ // that VCM will call ::Release and a deadlock can occur.
+ deliver_cs_->Enter();
+ vie_encoder->DeRegisterExternalEncoder(codec_.plType);
+ deliver_cs_->Leave();
+ return 0;
+ }
+ provider_cs_->Leave();
+ return ViEFrameProviderBase::DeregisterFrameCallback(callbackObject);
+}
+
+bool ViECapturer::IsFrameCallbackRegistered(
+ const ViEFrameCallback* callbackObject) {
+ CriticalSectionScoped cs(provider_cs_.get());
+ if (callbackObject == vie_encoder_) {
+ return true;
+ }
+ return ViEFrameProviderBase::IsFrameCallbackRegistered(callbackObject);
+}
+
+WebRtc_Word32 ViECapturer::PreEncodeToViEEncoder(const VideoCodec& codec,
+ ViEEncoder& vie_encoder,
+ WebRtc_Word32 vie_encoder_id) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+ if (vie_encoder_ && &vie_encoder != vie_encoder_) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d Capture device already encoding)",
+ __FUNCTION__, capture_id_);
+ return -1;
+ }
+
+ CriticalSectionScoped cs(encoding_cs_.get());
+ VideoCaptureModule::VideoCaptureEncodeInterface* capture_encoder =
+ capture_module_->GetEncodeInterface(codec);
+ if (!capture_encoder) {
+ // Encoding not supported?
+ return -1;
+ }
+ capture_encoder_ = capture_encoder;
+
+ // Create VCM module used for decoding frames if needed.
+ if (!vcm_) {
+ vcm_ = VideoCodingModule::Create(capture_id_);
+ }
+
+ if (vie_encoder.RegisterExternalEncoder(this, codec.plType) != 0) {
+ return -1;
+ }
+ if (vie_encoder.SetEncoder(codec) != 0) {
+ vie_encoder.DeRegisterExternalEncoder(codec.plType);
+ return -1;
+ }
+
+ // Make sure the encoder is not an I420 observer.
+ ViEFrameProviderBase::DeregisterFrameCallback(&vie_encoder);
+ // Store the vie_encoder using this capture device.
+ vie_encoder_ = &vie_encoder;
+ vie_encoder_id_ = vie_encoder_id;
+ memcpy(&codec_, &codec, sizeof(VideoCodec));
+ return 0;
+}
+
+bool ViECapturer::EncoderActive() {
+ return vie_encoder_ != NULL;
+}
+
+bool ViECapturer::CaptureCapabilityFixed() {
+ return requested_capability_.width != 0 &&
+ requested_capability_.height != 0 &&
+ requested_capability_.maxFPS != 0;
+}
+
+WebRtc_Word32 ViECapturer::Version(WebRtc_Word8* version,
+ WebRtc_Word32 length) const {
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::InitEncode(const VideoCodec* codec_settings,
+ WebRtc_Word32 number_of_cores,
+ WebRtc_UWord32 max_payload_size) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+
+ CriticalSectionScoped cs(encoding_cs_.get());
+ if (!capture_encoder_ || !codec_settings) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+
+ if (vcm_) {
+ // Initialize VCM to be able to decode frames if needed.
+ if (vcm_->InitializeReceiver() == 0) {
+ if (vcm_->RegisterReceiveCallback(this) == 0) {
+ if (vcm_->RegisterReceiveCodec(codec_settings, number_of_cores,
+ false) == 0) {
+ decoder_initialized_ = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d) VCM Decoder initialized",
+ __FUNCTION__, capture_id_);
+ }
+ }
+ }
+ }
+ return capture_encoder_->ConfigureEncoder(*codec_settings, max_payload_size);
+}
+
+WebRtc_Word32 ViECapturer::Encode(const RawImage& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const VideoFrameType* frame_types) {
+ CriticalSectionScoped cs(encoding_cs_.get());
+ if (!capture_encoder_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ if (*frame_types == kKeyFrame) {
+ return capture_encoder_->EncodeFrameType(kVideoFrameKey);
+ }
+ if (*frame_types == kSkipFrame) {
+ return capture_encoder_->EncodeFrameType(kFrameEmpty);
+ }
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+}
+
+WebRtc_Word32 ViECapturer::RegisterEncodeCompleteCallback(
+ EncodedImageCallback* callback) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+
+ CriticalSectionScoped cs(deliver_cs_.get());
+ if (!capture_encoder_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ encode_complete_callback_ = callback;
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::Release() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+ {
+ CriticalSectionScoped cs(deliver_cs_.get());
+ encode_complete_callback_ = NULL;
+ }
+
+ {
+ CriticalSectionScoped cs(encoding_cs_.get());
+
+ decoder_initialized_ = false;
+ codec_.codecType = kVideoCodecUnknown;
+ // Reset the camera to output I420.
+ capture_encoder_->ConfigureEncoder(codec_, 0);
+
+ if (vie_encoder_) {
+ // Need to add the encoder as an observer of I420.
+ ViEFrameProviderBase::RegisterFrameCallback(vie_encoder_id_,
+ vie_encoder_);
+ }
+ vie_encoder_ = NULL;
+ }
+ return 0;
+}
+
+// Should reset the capture device to the state it was in after the InitEncode
+// function. Current implementation do nothing.
+WebRtc_Word32 ViECapturer::Reset() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::SetChannelParameters(WebRtc_UWord32 packet_loss,
+ int rtt) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+
+ CriticalSectionScoped cs(encoding_cs_.get());
+ if (!capture_encoder_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ return capture_encoder_->SetChannelParameters(packet_loss, rtt);
+}
+
+WebRtc_Word32 ViECapturer::SetRates(WebRtc_UWord32 new_bit_rate,
+ WebRtc_UWord32 frame_rate) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
+
+ CriticalSectionScoped cs(encoding_cs_.get());
+ if (!capture_encoder_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ return capture_encoder_->SetRates(new_bit_rate, frame_rate);
+}
+
+WebRtc_Word32 ViECapturer::FrameToRender(VideoFrame& video_frame) {
+ deliver_cs_->Enter();
+ DeliverI420Frame(video_frame);
+ deliver_cs_->Leave();
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::RegisterObserver(ViECaptureObserver& observer) {
+ if (observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s Observer already registered", __FUNCTION__, capture_id_);
+ return -1;
+ }
+ if (capture_module_->RegisterCaptureCallback(*this) != 0) {
+ return -1;
+ }
+ capture_module_->EnableFrameRateCallback(true);
+ capture_module_->EnableNoPictureAlarm(true);
+ observer_ = &observer;
+ return 0;
+}
+
+WebRtc_Word32 ViECapturer::DeRegisterObserver() {
+ CriticalSectionScoped cs(observer_cs_.get());
+ if (!observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "%s No observer registered", __FUNCTION__, capture_id_);
+ return -1;
+ }
+ capture_module_->EnableFrameRateCallback(false);
+ capture_module_->EnableNoPictureAlarm(false);
+ capture_module_->DeRegisterCaptureCallback();
+ observer_ = NULL;
+ return 0;
+}
+
+bool ViECapturer::IsObserverRegistered() {
+ CriticalSectionScoped cs(observer_cs_.get());
+ return observer_ != NULL;
+}
+
+void ViECapturer::OnCaptureFrameRate(const WebRtc_Word32 id,
+ const WebRtc_UWord32 frame_rate) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "OnCaptureFrameRate %d", frame_rate);
+
+ CriticalSectionScoped cs(observer_cs_.get());
+ observer_->CapturedFrameRate(id_, (WebRtc_UWord8) frame_rate);
+}
+
+void ViECapturer::OnNoPictureAlarm(const WebRtc_Word32 id,
+ const VideoCaptureAlarm alarm) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
+ "OnNoPictureAlarm %d", alarm);
+
+ CriticalSectionScoped cs(observer_cs_.get());
+ CaptureAlarm vie_alarm = (alarm == Raised) ? AlarmRaised : AlarmCleared;
+ observer_->NoPictureAlarm(id, vie_alarm);
+}
+
+WebRtc_Word32 ViECapturer::SetCaptureDeviceImage(
+ const VideoFrame& capture_device_image) {
+ return capture_module_->StartSendImage(capture_device_image, 10);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_capturer.h b/trunk/src/video_engine/vie_capturer.h
new file mode 100644
index 0000000..da68278
--- /dev/null
+++ b/trunk/src/video_engine/vie_capturer.h
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "modules/video_capture/main/interface/video_capture.h"
+#include "modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_capture.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_frame_provider_base.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class EventWrapper;
+class ProcessThread;
+class ThreadWrapper;
+class ViEEffectFilter;
+class ViEEncoder;
+struct ViEPicture;
+
+class ViECapturer
+ : public ViEFrameProviderBase,
+ public ViEExternalCapture,
+ protected VCMReceiveCallback,
+ protected VideoCaptureDataCallback,
+ protected VideoCaptureFeedBack,
+ protected VideoEncoder {
+ public:
+ static ViECapturer* CreateViECapture(int capture_id,
+ int engine_id,
+ VideoCaptureModule& capture_module,
+ ProcessThread& module_process_thread);
+
+ static ViECapturer* CreateViECapture(
+ int capture_id,
+ int engine_id,
+ const WebRtc_UWord8* device_unique_idUTF8,
+ WebRtc_UWord32 device_unique_idUTF8Length,
+ ProcessThread& module_process_thread);
+
+ ~ViECapturer();
+
+ // Implements ViEFrameProviderBase.
+ int FrameCallbackChanged();
+ virtual int DeregisterFrameCallback(const ViEFrameCallback* callbackObject);
+ bool IsFrameCallbackRegistered(const ViEFrameCallback* callbackObject);
+
+ // Implements ExternalCapture.
+ virtual int IncomingFrame(unsigned char* video_frame,
+ unsigned int video_frame_length,
+ unsigned short width, unsigned short height,
+ RawVideoType video_type,
+ unsigned long long capture_time = 0);
+
+ virtual int IncomingFrameI420(const ViEVideoFrameI420& video_frame,
+ unsigned long long capture_time = 0);
+
+ // Use this capture device as encoder.
+ // Returns 0 if the codec is supported by this capture device.
+ virtual WebRtc_Word32 PreEncodeToViEEncoder(const VideoCodec& codec,
+ ViEEncoder& vie_encoder,
+ WebRtc_Word32 vie_encoder_id);
+
+ // Start/Stop.
+ WebRtc_Word32 Start(
+ const CaptureCapability& capture_capability = CaptureCapability());
+ WebRtc_Word32 Stop();
+ bool Started();
+
+ // Overrides the capture delay.
+ WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delay_ms);
+
+ // Sets rotation of the incoming captured frame.
+ WebRtc_Word32 SetRotateCapturedFrames(const RotateCapturedFrame rotation);
+
+ // Effect filter.
+ WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
+ WebRtc_Word32 EnableDenoising(bool enable);
+ WebRtc_Word32 EnableDeflickering(bool enable);
+ WebRtc_Word32 EnableBrightnessAlarm(bool enable);
+
+ // Statistics observer.
+ WebRtc_Word32 RegisterObserver(ViECaptureObserver& observer);
+ WebRtc_Word32 DeRegisterObserver();
+ bool IsObserverRegistered();
+
+ // Information.
+ const WebRtc_UWord8* CurrentDeviceName() const;
+
+ // Set device image.
+ WebRtc_Word32 SetCaptureDeviceImage(const VideoFrame& capture_device_image);
+
+ protected:
+ ViECapturer(int capture_id,
+ int engine_id,
+ ProcessThread& module_process_thread);
+
+ WebRtc_Word32 Init(VideoCaptureModule& capture_module);
+ WebRtc_Word32 Init(const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_unique_idUTF8Length);
+
+ // Implements VideoCaptureDataCallback.
+ virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
+ VideoFrame& video_frame,
+ VideoCodecType codec_type);
+ virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
+ const WebRtc_Word32 delay);
+
+ bool EncoderActive();
+
+ // Returns true if the capture capability has been set in |StartCapture|
+ // function and may not be changed.
+ bool CaptureCapabilityFixed();
+
+ // Help function used for keeping track of VideoImageProcesingModule.
+ // Creates the module if it is needed, returns 0 on success and guarantees
+ // that the image proc module exist.
+ WebRtc_Word32 IncImageProcRefCount();
+ WebRtc_Word32 DecImageProcRefCount();
+
+ // Implements VideoEncoder.
+ virtual WebRtc_Word32 Version(WebRtc_Word8* version,
+ WebRtc_Word32 length) const;
+ virtual WebRtc_Word32 InitEncode(const VideoCodec* codec_settings,
+ WebRtc_Word32 number_of_cores,
+ WebRtc_UWord32 max_payload_size);
+ virtual WebRtc_Word32 Encode(const RawImage& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const VideoFrameType* frame_types);
+ virtual WebRtc_Word32 RegisterEncodeCompleteCallback(
+ EncodedImageCallback* callback);
+ virtual WebRtc_Word32 Release();
+ virtual WebRtc_Word32 Reset();
+ virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packet_loss,
+ int rtt);
+ virtual WebRtc_Word32 SetRates(WebRtc_UWord32 new_bit_rate,
+ WebRtc_UWord32 frame_rate);
+
+ // Implements VCMReceiveCallback.
+ virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
+
+ // Implements VideoCaptureFeedBack
+ virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
+ const WebRtc_UWord32 frame_rate);
+ virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
+ const VideoCaptureAlarm alarm);
+
+ // Thread functions for deliver captured frames to receivers.
+ static bool ViECaptureThreadFunction(void* obj);
+ bool ViECaptureProcess();
+
+ void DeliverI420Frame(VideoFrame& video_frame);
+ void DeliverCodedFrame(VideoFrame& video_frame);
+
+ private:
+ // Never take capture_cs_ before deliver_cs_!
+ scoped_ptr<CriticalSectionWrapper> capture_cs_;
+ scoped_ptr<CriticalSectionWrapper> deliver_cs_;
+ VideoCaptureModule* capture_module_;
+ VideoCaptureExternal* external_capture_module_;
+ ProcessThread& module_process_thread_;
+ const int capture_id_;
+
+ // Capture thread.
+ ThreadWrapper& capture_thread_;
+ EventWrapper& capture_event_;
+ EventWrapper& deliver_event_;
+
+ VideoFrame captured_frame_;
+ VideoFrame deliver_frame_;
+ VideoFrame encoded_frame_;
+
+ // Image processing.
+ ViEEffectFilter* effect_filter_;
+ VideoProcessingModule* image_proc_module_;
+ int image_proc_module_ref_counter_;
+ VideoProcessingModule::FrameStats* deflicker_frame_stats_;
+ VideoProcessingModule::FrameStats* brightness_frame_stats_;
+ Brightness current_brightness_level_;
+ Brightness reported_brightness_level_;
+ bool denoising_enabled_;
+
+ // Statistics observer.
+ scoped_ptr<CriticalSectionWrapper> observer_cs_;
+ ViECaptureObserver* observer_;
+
+ // Encoding using encoding capable cameras.
+ scoped_ptr<CriticalSectionWrapper> encoding_cs_;
+ VideoCaptureModule::VideoCaptureEncodeInterface* capture_encoder_;
+ EncodedImageCallback* encode_complete_callback_;
+ VideoCodec codec_;
+ // The ViEEncoder we are encoding for.
+ ViEEncoder* vie_encoder_;
+ // ViEEncoder id we are encoding for.
+ WebRtc_Word32 vie_encoder_id_;
+ // Used for decoding preencoded frames.
+ VideoCodingModule* vcm_;
+ EncodedVideoData decode_buffer_;
+ bool decoder_initialized_;
+ CaptureCapability requested_capability_;
+
+ VideoFrame capture_device_image_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
diff --git a/trunk/src/video_engine/vie_channel.cc b/trunk/src/video_engine/vie_channel.cc
new file mode 100644
index 0000000..e6f7a0a
--- /dev/null
+++ b/trunk/src/video_engine/vie_channel.cc
@@ -0,0 +1,2529 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_channel.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "modules/udp_transport/interface/udp_transport.h"
+#include "modules/utility/interface/process_thread.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_processing/main/interface/video_processing.h"
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/thread_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_codec.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/include/vie_image_process.h"
+#include "video_engine/include/vie_rtp_rtcp.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_receiver.h"
+#include "video_engine/vie_sender.h"
+#include "video_engine/vie_sync_module.h"
+
+namespace webrtc {
+
+const int kMaxDecodeWaitTimeMs = 50;
+
+ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
+ WebRtc_Word32 engine_id,
+ WebRtc_UWord32 number_of_cores,
+ ProcessThread& module_process_thread)
+ : ViEFrameProviderBase(channel_id, engine_id),
+ channel_id_(channel_id),
+ engine_id_(engine_id),
+ number_of_cores_(number_of_cores),
+ num_socket_threads_(kViESocketThreads),
+ callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ rtp_rtcp_(*RtpRtcp::CreateRtpRtcp(ViEModuleId(engine_id, channel_id),
+ false)),
+ default_rtp_rtcp_(NULL),
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ socket_transport_(*UdpTransport::Create(
+ ViEModuleId(engine_id, channel_id), num_socket_threads_)),
+#endif
+ vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
+ vie_receiver_(*(new ViEReceiver(engine_id, channel_id, rtp_rtcp_, vcm_))),
+ vie_sender_(*(new ViESender(engine_id, channel_id))),
+ vie_sync_(*(new ViESyncModule(ViEId(engine_id, channel_id), vcm_,
+ rtp_rtcp_))),
+ module_process_thread_(module_process_thread),
+ codec_observer_(NULL),
+ do_key_frame_callbackRequest_(false),
+ rtp_observer_(NULL),
+ rtcp_observer_(NULL),
+ networkObserver_(NULL),
+ rtp_packet_timeout_(false),
+ using_packet_spread_(false),
+ external_transport_(NULL),
+ decoder_reset_(true),
+ wait_for_key_frame_(false),
+ decode_thread_(NULL),
+ external_encryption_(NULL),
+ effect_filter_(NULL),
+ color_enhancement_(true),
+ vcm_rttreported_(TickTime::Now()),
+ file_recorder_(channel_id) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, channel_id),
+ "ViEChannel::ViEChannel(channel_id: %d, engine_id: %d)",
+ channel_id, engine_id);
+}
+
+WebRtc_Word32 ViEChannel::Init() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: channel_id: %d, engine_id: %d)", __FUNCTION__, channel_id_,
+ engine_id_);
+ // RTP/RTCP initialization.
+ if (rtp_rtcp_.InitSender() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::InitSender failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.SetSendingMediaStatus(false) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::SetSendingMediaStatus failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.InitReceiver() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::InitReceiver failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.RegisterIncomingDataCallback(
+ static_cast<RtpData*>(&vie_receiver_)) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterIncomingDataCallback failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.RegisterSendTransport(
+ static_cast<Transport*>(&vie_sender_)) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterSendTransport failure", __FUNCTION__);
+ return -1;
+ }
+ if (module_process_thread_.RegisterModule(&rtp_rtcp_) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterModule failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.SetKeyFrameRequestMethod(kKeyFrameReqFirRtp) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::SetKeyFrameRequestMethod failure", __FUNCTION__);
+ }
+ if (rtp_rtcp_.SetRTCPStatus(kRtcpCompound) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
+ }
+ if (rtp_rtcp_.RegisterIncomingRTPCallback(this) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterIncomingRTPCallback failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.RegisterIncomingRTCPCallback(this) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterIncomingRTCPCallback failure", __FUNCTION__);
+ return -1;
+ }
+
+ // VCM initialization
+ if (vcm_.InitializeReceiver() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: VCM::InitializeReceiver failure", __FUNCTION__);
+ return -1;
+ }
+ if (vcm_.RegisterReceiveCallback(this) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM::RegisterReceiveCallback failure", __FUNCTION__);
+ return -1;
+ }
+ if (vcm_.RegisterFrameTypeCallback(this) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM::RegisterFrameTypeCallback failure", __FUNCTION__);
+ }
+ if (vcm_.RegisterReceiveStatisticsCallback(this) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM::RegisterReceiveStatisticsCallback failure",
+ __FUNCTION__);
+ }
+ if (vcm_.SetRenderDelay(kViEDefaultRenderDelayMs) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM::SetRenderDelay failure", __FUNCTION__);
+ }
+ if (module_process_thread_.RegisterModule(&vcm_) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM::RegisterModule(vcm) failure", __FUNCTION__);
+ return -1;
+ }
+#ifdef VIDEOCODEC_VP8
+ VideoCodec video_codec;
+ if (vcm_.Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
+ rtp_rtcp_.RegisterSendPayload(video_codec);
+ rtp_rtcp_.RegisterReceivePayload(video_codec);
+ vcm_.RegisterReceiveCodec(&video_codec, number_of_cores_);
+ vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+ rtp_rtcp_.MaxDataPayloadLength());
+ } else {
+ assert(false);
+ }
+#endif
+
+ return 0;
+}
+
+ViEChannel::~ViEChannel() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "ViEChannel Destructor, channel_id: %d, engine_id: %d",
+ channel_id_, engine_id_);
+
+ // Make sure we don't get more callbacks from the RTP module.
+ rtp_rtcp_.RegisterIncomingRTPCallback(NULL);
+ rtp_rtcp_.RegisterSendTransport(NULL);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ socket_transport_.StopReceiving();
+#endif
+ module_process_thread_.DeRegisterModule(&rtp_rtcp_);
+ module_process_thread_.DeRegisterModule(&vcm_);
+ module_process_thread_.DeRegisterModule(&vie_sync_);
+ while (simulcast_rtp_rtcp_.size() > 0) {
+ std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
+ rtp_rtcp->RegisterSendTransport(NULL);
+ module_process_thread_.DeRegisterModule(rtp_rtcp);
+ RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
+ simulcast_rtp_rtcp_.erase(it);
+ }
+ if (decode_thread_) {
+ StopDecodeThread();
+ }
+
+ delete &vie_receiver_;
+ delete &vie_sender_;
+ delete &vie_sync_;
+
+ // Release modules.
+ RtpRtcp::DestroyRtpRtcp(&rtp_rtcp_);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ UdpTransport::Destroy(&socket_transport_);
+#endif
+ VideoCodingModule::Destroy(&vcm_);
+}
+
+WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
+ bool new_stream) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: codec_type: %d", __FUNCTION__, video_codec.codecType);
+
+ if (video_codec.codecType == kVideoCodecRED ||
+ video_codec.codecType == kVideoCodecULPFEC) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: codec_type: %d is not a valid send codec.", __FUNCTION__,
+ video_codec.codecType);
+ return -1;
+ }
+ if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Too many simulcast streams", __FUNCTION__);
+ return -1;
+ }
+ // Update the RTP module with the settings.
+ // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
+ // set explicitly.
+ bool restart_rtp = false;
+ if (rtp_rtcp_.Sending() && new_stream) {
+ restart_rtp = true;
+ rtp_rtcp_.SetSendingStatus(false);
+ }
+ if (video_codec.numberOfSimulcastStreams > 0) {
+ WebRtc_UWord32 start_bitrate = video_codec.startBitrate * 1000;
+ WebRtc_UWord32 stream_bitrate =
+ std::min(start_bitrate, video_codec.simulcastStream[0].maxBitrate);
+ start_bitrate -= stream_bitrate;
+ // Set correct bitrate to base layer.
+ rtp_rtcp_.SetSendBitrate(stream_bitrate, video_codec.minBitrate,
+ video_codec.simulcastStream[0].maxBitrate);
+ // Create our simulcast RTP modules.
+ for (int i = simulcast_rtp_rtcp_.size();
+ i < video_codec.numberOfSimulcastStreams - 1;
+ i++) {
+ RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(
+ ViEModuleId(engine_id_, channel_id_), false);
+ if (rtp_rtcp->RegisterDefaultModule(default_rtp_rtcp_)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not register default module", __FUNCTION__);
+ return -1;
+ }
+ simulcast_rtp_rtcp_.push_back(rtp_rtcp);
+ }
+ // Remove last in list if we have too many.
+ for (int j = simulcast_rtp_rtcp_.size();
+ j > (video_codec.numberOfSimulcastStreams - 1);
+ j--) {
+ RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
+ rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
+ rtp_rtcp->RegisterSendTransport(NULL);
+ module_process_thread_.DeRegisterModule(rtp_rtcp);
+ RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
+ simulcast_rtp_rtcp_.pop_back();
+ }
+ VideoCodec video_codec;
+ if (vcm_.Codec(kVideoCodecVP8, &video_codec) != VCM_OK) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: VCM: failure geting default VP8 pl_type", __FUNCTION__);
+ return -1;
+ }
+ WebRtc_UWord8 idx = 0;
+ // Configure all simulcast modules.
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ idx++;
+ RtpRtcp* rtp_rtcp = *it;
+ if (rtp_rtcp->InitSender() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::InitSender failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp->InitReceiver() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::InitReceiver failure", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp->RegisterSendTransport(
+ static_cast<Transport*>(&vie_sender_)) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::RegisterSendTransport failure", __FUNCTION__);
+ return -1;
+ }
+ // Silently ignore error.
+ module_process_thread_.RegisterModule(rtp_rtcp);
+ if (rtp_rtcp->SetRTCPStatus(rtp_rtcp_.RTCP()) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
+ }
+ rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
+ if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not register payload type", __FUNCTION__);
+ return -1;
+ }
+ if (restart_rtp) {
+ rtp_rtcp->SetSendingStatus(true);
+ }
+ // Configure all simulcast streams min and max bitrates
+ const WebRtc_UWord32 stream_bitrate =
+ std::min(start_bitrate, video_codec.simulcastStream[idx].maxBitrate);
+ start_bitrate -= stream_bitrate;
+ rtp_rtcp->SetSendBitrate(stream_bitrate, video_codec.minBitrate,
+ video_codec.simulcastStream[idx].maxBitrate);
+ }
+ vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
+ } else {
+ if (!simulcast_rtp_rtcp_.empty()) {
+ // Delete all simulcast rtp modules.
+ while (!simulcast_rtp_rtcp_.empty()) {
+ RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
+ rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
+ rtp_rtcp->RegisterSendTransport(NULL);
+ module_process_thread_.DeRegisterModule(rtp_rtcp);
+ RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
+ simulcast_rtp_rtcp_.pop_back();
+ }
+ }
+ // Clear any previous modules.
+ vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
+
+ rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
+ video_codec.minBitrate,
+ video_codec.maxBitrate);
+ }
+ // Enable this if H264 is available.
+ // This sets the wanted packetization mode.
+ // if (video_codec.plType == kVideoCodecH264) {
+ // if (video_codec.codecSpecific.H264.packetization == kH264SingleMode) {
+ // rtp_rtcp_.SetH264PacketizationMode(H264_SINGLE_NAL_MODE);
+ // } else {
+ // rtp_rtcp_.SetH264PacketizationMode(H264_NON_INTERLEAVED_MODE);
+ // }
+ // if (video_codec.codecSpecific.H264.configParametersSize > 0) {
+ // rtp_rtcp_.SetH264SendModeNALU_PPS_SPS(true);
+ // }
+ // }
+
+ // Don't log this error, no way to check in advance if this pl_type is
+ // registered or not...
+ rtp_rtcp_.DeRegisterSendPayload(video_codec.plType);
+ if (rtp_rtcp_.RegisterSendPayload(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not register payload type", __FUNCTION__);
+ return -1;
+ }
+ if (restart_rtp) {
+ rtp_rtcp_.SetSendingStatus(true);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
+ // We will not receive simulcast streams, so no need to handle that use case.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ WebRtc_Word8 old_pltype = -1;
+ if (rtp_rtcp_.ReceivePayloadType(video_codec, &old_pltype) != -1) {
+ rtp_rtcp_.DeRegisterReceivePayload(old_pltype);
+ }
+
+ if (rtp_rtcp_.RegisterReceivePayload(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not register receive payload type", __FUNCTION__);
+ return -1;
+ }
+
+ if (video_codec.codecType != kVideoCodecRED &&
+ video_codec.codecType != kVideoCodecULPFEC) {
+ // Register codec type with VCM, but do not register RED or ULPFEC.
+ if (vcm_.RegisterReceiveCodec(&video_codec, number_of_cores_,
+ wait_for_key_frame_) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not register decoder", __FUNCTION__);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetReceiveCodec(VideoCodec& video_codec) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ if (vcm_.ReceiveCodec(&video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get receive codec", __FUNCTION__);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterCodecObserver(ViEDecoderObserver* observer) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (observer) {
+ if (codec_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: already added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer added", __FUNCTION__);
+ codec_observer_ = observer;
+ } else {
+ if (!codec_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no observer added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer removed", __FUNCTION__);
+ codec_observer_ = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
+ VideoDecoder* decoder,
+ bool decoder_render,
+ WebRtc_Word32 render_delay) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ WebRtc_Word32 result = 0;
+ result = vcm_.RegisterExternalDecoder(decoder, pl_type, decoder_render);
+ if (decoder_render && result == 0) {
+ // Let VCM know how long before the actual render time the decoder needs
+ // to get a frame for decoding.
+ result = vcm_.SetRenderDelay(render_delay);
+ }
+ return result;
+}
+
+WebRtc_Word32 ViEChannel::DeRegisterExternalDecoder(
+ const WebRtc_UWord8 pl_type) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s pl_type", __FUNCTION__, pl_type);
+
+ VideoCodec current_receive_codec;
+ WebRtc_Word32 result = 0;
+ result = vcm_.ReceiveCodec(¤t_receive_codec);
+ if (vcm_.RegisterExternalDecoder(NULL, pl_type, false) != VCM_OK) {
+ return -1;
+ }
+
+ if (result == 0 && current_receive_codec.plType == pl_type) {
+ result = vcm_.RegisterReceiveCodec(¤t_receive_codec, number_of_cores_,
+ wait_for_key_frame_);
+ }
+ return result;
+}
+
+WebRtc_Word32 ViEChannel::ReceiveCodecStatistics(
+ WebRtc_UWord32& num_key_frames, WebRtc_UWord32& num_delta_frames) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ VCMFrameCount received_frames;
+ if (vcm_.ReceivedFrameCount(received_frames) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get received frame information", __FUNCTION__);
+ return -1;
+ }
+ num_key_frames = received_frames.numKeyFrames;
+ num_delta_frames = received_frames.numDeltaFrames;
+ return 0;
+}
+
+WebRtc_UWord32 ViEChannel::DiscardedPackets() const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ return vcm_.DiscardedPackets();
+}
+
+WebRtc_Word32 ViEChannel::WaitForKeyFrame(bool wait) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(wait: %d)", __FUNCTION__, wait);
+ wait_for_key_frame_ = wait;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetSignalPacketLossStatus(bool enable,
+ bool only_key_frames) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(enable: %d)", __FUNCTION__, enable);
+ if (enable) {
+ if (only_key_frames) {
+ vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
+ if (vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, true) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s failed %d", __FUNCTION__, enable);
+ return -1;
+ }
+ } else {
+ vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
+ if (vcm_.SetVideoProtection(kProtectionKeyOnLoss, true) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s failed %d", __FUNCTION__, enable);
+ return -1;
+ }
+ }
+ } else {
+ vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
+ vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %d", __FUNCTION__, rtcp_mode);
+
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetRTCPStatus(rtcp_mode);
+ }
+ return rtp_rtcp_.SetRTCPStatus(rtcp_mode);
+}
+
+WebRtc_Word32 ViEChannel::GetRTCPMode(RTCPMethod& rtcp_mode) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ rtcp_mode = rtp_rtcp_.RTCP();
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetNACKStatus(const bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(enable: %d)", __FUNCTION__, enable);
+
+ // Update the decoding VCM.
+ if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
+ enable);
+ return -1;
+ }
+ if (enable) {
+ // Disable possible FEC.
+ SetFECStatus(false, 0, 0);
+ }
+ // Update the decoding VCM.
+ if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
+ enable);
+ return -1;
+ }
+ return ProcessNACKRequest(enable);
+}
+
+WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(enable: %d)", __FUNCTION__, enable);
+
+ if (enable) {
+ // Turn on NACK.
+ NACKMethod nackMethod = kNackRtcp;
+ if (rtp_rtcp_.RTCP() == kRtcpOff) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not enable NACK, RTPC not on ", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.SetNACKStatus(nackMethod) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set NACK method %d", __FUNCTION__,
+ nackMethod);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Using NACK method %d", __FUNCTION__, nackMethod);
+ rtp_rtcp_.SetStorePacketsStatus(true, kNackHistorySize);
+
+ vcm_.RegisterPacketRequestCallback(this);
+
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
+ }
+ } else {
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetStorePacketsStatus(false);
+ }
+ rtp_rtcp_.SetStorePacketsStatus(false);
+ vcm_.RegisterPacketRequestCallback(NULL);
+ if (rtp_rtcp_.SetNACKStatus(kNackOff) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not turn off NACK", __FUNCTION__);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetFECStatus(const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) {
+ // Disable possible NACK.
+ if (enable) {
+ SetNACKStatus(false);
+ }
+
+ return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
+}
+
+WebRtc_Word32 ViEChannel::ProcessFECRequest(
+ const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(enable: %d, payload_typeRED: %u, payload_typeFEC: %u)",
+ __FUNCTION__, enable, payload_typeRED, payload_typeFEC);
+
+ if (rtp_rtcp_.SetGenericFECStatus(enable, payload_typeRED,
+ payload_typeFEC) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not change FEC status to %d", __FUNCTION__,
+ enable);
+ return -1;
+ }
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetGenericFECStatus(enable, payload_typeRED, payload_typeFEC);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetHybridNACKFECStatus(
+ const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) {
+ // Update the decoding VCM with hybrid mode.
+ if (vcm_.SetVideoProtection(kProtectionNackFEC, enable) != VCM_OK) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
+ enable);
+ return -1;
+ }
+
+ WebRtc_Word32 ret_val = 0;
+ ret_val = ProcessNACKRequest(enable);
+ if (ret_val < 0) {
+ return ret_val;
+ }
+ return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
+}
+
+WebRtc_Word32 ViEChannel::SetKeyFrameRequestMethod(
+ const KeyFrameRequestMethod method) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %d", __FUNCTION__, method);
+ return rtp_rtcp_.SetKeyFrameRequestMethod(method);
+}
+
+bool ViEChannel::EnableRemb(bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "ViEChannel::EnableRemb: %d", enable);
+ if (rtp_rtcp_.SetREMBStatus(enable) != 0)
+ return false;
+ return true;
+}
+
+WebRtc_Word32 ViEChannel::EnableTMMBR(const bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %d", __FUNCTION__, enable);
+ return rtp_rtcp_.SetTMMBRStatus(enable);
+}
+
+WebRtc_Word32 ViEChannel::EnableKeyFrameRequestCallback(const bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %d", __FUNCTION__, enable);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (enable && !codec_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: No ViECodecObserver set", __FUNCTION__, enable);
+ return -1;
+ }
+ do_key_frame_callbackRequest_ = enable;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetSSRC(const WebRtc_UWord32 SSRC,
+ const StreamType usage,
+ const uint8_t simulcast_idx) {
+ WEBRTC_TRACE(webrtc::kTraceInfo,
+ webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s(usage:%d, SSRC: 0x%x, idx:%u)",
+ __FUNCTION__, usage, SSRC, simulcast_idx);
+ if (simulcast_idx == 0) {
+ return rtp_rtcp_.SetSSRC(SSRC);
+ }
+ std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ for (int i = 1; i < simulcast_idx; ++i, ++it) {
+ if (it == simulcast_rtp_rtcp_.end()) {
+ return -1;
+ }
+ }
+ RtpRtcp* rtp_rtcp = *it;
+ if (usage == kViEStreamTypeRtx) {
+ return rtp_rtcp->SetRTXSendStatus(true, true, SSRC);
+ }
+ return rtp_rtcp->SetSSRC(SSRC);
+}
+
+WebRtc_Word32 ViEChannel::SetRemoteSSRCType(const StreamType usage,
+ const uint32_t SSRC) const {
+ WEBRTC_TRACE(webrtc::kTraceInfo,
+ webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s(usage:%d, SSRC: 0x%x)",
+ __FUNCTION__, usage, SSRC);
+
+ return rtp_rtcp_.SetRTXReceiveStatus(true, SSRC);
+}
+
+WebRtc_Word32 ViEChannel::GetLocalSSRC(WebRtc_UWord32& SSRC) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ SSRC = rtp_rtcp_.SSRC();
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetRemoteSSRC(WebRtc_UWord32& SSRC) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ SSRC = rtp_rtcp_.RemoteSSRC();
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ WebRtc_UWord32 arrayCSRC[kRtpCsrcSize];
+ memset(arrayCSRC, 0, sizeof(arrayCSRC));
+
+ WebRtc_Word32 num_csrcs = rtp_rtcp_.RemoteCSRCs(arrayCSRC);
+ if (num_csrcs > 0) {
+ memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(WebRtc_UWord32));
+ for (int idx = 0; idx < num_csrcs; idx++) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "\tCSRC[%d] = %lu", idx, CSRCs[idx]);
+ }
+ } else {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: CSRC list is empty", __FUNCTION__);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetStartSequenceNumber(
+ WebRtc_UWord16 sequence_number) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: already sending", __FUNCTION__);
+ return -1;
+ }
+ return rtp_rtcp_.SetSequenceNumber(sequence_number);
+}
+
+WebRtc_Word32 ViEChannel::SetRTCPCName(const WebRtc_Word8 rtcp_cname[]) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ if (rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: already sending", __FUNCTION__);
+ return -1;
+ }
+ return rtp_rtcp_.SetCNAME(rtcp_cname);
+}
+
+WebRtc_Word32 ViEChannel::GetRTCPCName(WebRtc_Word8 rtcp_cname[]) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ return rtp_rtcp_.CNAME(rtcp_cname);
+}
+
+WebRtc_Word32 ViEChannel::GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
+ return rtp_rtcp_.RemoteCNAME(remoteSSRC, rtcp_cname);
+}
+
+WebRtc_Word32 ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (observer) {
+ if (rtp_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer alread added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer added", __FUNCTION__);
+ rtp_observer_ = observer;
+ } else {
+ if (!rtp_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no observer added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer removed", __FUNCTION__);
+ rtp_observer_ = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterRtcpObserver(ViERTCPObserver* observer) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (observer) {
+ if (rtcp_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer alread added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer added", __FUNCTION__);
+ rtcp_observer_ = observer;
+ } else {
+ if (!rtcp_observer_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no observer added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer removed", __FUNCTION__);
+ rtcp_observer_ = NULL;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SendApplicationDefinedRTCPPacket(
+ const WebRtc_UWord8 sub_type,
+ WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ WebRtc_UWord16 data_length_in_bytes) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (!rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not sending", __FUNCTION__);
+ return -1;
+ }
+ if (!data) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no input argument", __FUNCTION__);
+ return -1;
+ }
+ if (data_length_in_bytes % 4 != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: input length error", __FUNCTION__);
+ return -1;
+ }
+ RTCPMethod rtcp_method = rtp_rtcp_.RTCP();
+ if (rtcp_method == kRtcpOff) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTCP not enabled", __FUNCTION__);
+ return -1;
+ }
+ // Create and send packet.
+ if (rtp_rtcp_.SetRTCPApplicationSpecificData(sub_type, name, data,
+ data_length_in_bytes) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not send RTCP application data", __FUNCTION__);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
+ WebRtc_UWord32& cumulative_lost,
+ WebRtc_UWord32& extended_max,
+ WebRtc_UWord32& jitter_samples,
+ WebRtc_Word32& rtt_ms) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ // TODO(pwestin) how do we do this for simulcast ? average for all
+ // except cumulative_lost that is the sum ?
+ // for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ // it != simulcast_rtp_rtcp_.end();
+ // it++) {
+ // RtpRtcp* rtp_rtcp = *it;
+ // }
+ WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
+
+ // Get all RTCP receiver report blocks that have been received on this
+ // channel. If we receive RTP packets from a remote source we know the
+ // remote SSRC and use the report block from him.
+ // Otherwise use the first report block.
+ std::vector<RTCPReportBlock> remote_stats;
+ if (rtp_rtcp_.RemoteRTCPStat(&remote_stats) != 0 || remote_stats.empty()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get remote stats", __FUNCTION__);
+ return -1;
+ }
+ std::vector<RTCPReportBlock>::const_iterator statistics =
+ remote_stats.begin();
+ for (; statistics != remote_stats.end(); ++statistics) {
+ if (statistics->remoteSSRC == remoteSSRC)
+ break;
+ }
+
+ if (statistics == remote_stats.end()) {
+ // If we have not received any RTCP packets from this SSRC it probably means
+ // we have not received any RTP packets.
+ // Use the first received report block instead.
+ statistics = remote_stats.begin();
+ remoteSSRC = statistics->remoteSSRC;
+ }
+
+ fraction_lost = statistics->fractionLost;
+ cumulative_lost = statistics->cumulativeLost;
+ extended_max = statistics->extendedHighSeqNum;
+ jitter_samples = statistics->jitter;
+
+ WebRtc_UWord16 dummy;
+ WebRtc_UWord16 rtt = 0;
+ if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get RTT", __FUNCTION__);
+ return -1;
+ }
+ rtt_ms = rtt;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetReceivedRtcpStatistics(
+ WebRtc_UWord16& fraction_lost,
+ WebRtc_UWord32& cumulative_lost,
+ WebRtc_UWord32& extended_max,
+ WebRtc_UWord32& jitter_samples,
+ WebRtc_Word32& rtt_ms) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ WebRtc_UWord8 frac_lost = 0;
+ if (rtp_rtcp_.StatisticsRTP(&frac_lost, &cumulative_lost, &extended_max,
+ &jitter_samples) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get received RTP statistics", __FUNCTION__);
+ return -1;
+ }
+ fraction_lost = frac_lost;
+
+ WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
+ WebRtc_UWord16 dummy = 0;
+ WebRtc_UWord16 rtt = 0;
+ if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get RTT", __FUNCTION__);
+ }
+ rtt_ms = rtt;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetRtpStatistics(
+ WebRtc_UWord32& bytes_sent,
+ WebRtc_UWord32& packets_sent,
+ WebRtc_UWord32& bytes_received,
+ WebRtc_UWord32& packets_received) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (rtp_rtcp_.DataCountersRTP(&bytes_sent,
+ &packets_sent,
+ &bytes_received,
+ &packets_received) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get counters", __FUNCTION__);
+ return -1;
+ }
+ for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ WebRtc_UWord32 bytes_sent_temp = 0;
+ WebRtc_UWord32 packets_sent_temp = 0;
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp, NULL, NULL);
+ bytes_sent += bytes_sent_temp;
+ packets_sent += packets_sent_temp;
+ }
+ return 0;
+}
+
+void ViEChannel::GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
+ WebRtc_UWord32& video_bitrate_sent,
+ WebRtc_UWord32& fec_bitrate_sent,
+ WebRtc_UWord32& nackBitrateSent) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ rtp_rtcp_.BitrateSent(&total_bitrate_sent,
+ &video_bitrate_sent,
+ &fec_bitrate_sent,
+ &nackBitrateSent);
+ for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end(); it++) {
+ WebRtc_UWord32 stream_rate = 0;
+ WebRtc_UWord32 video_rate = 0;
+ WebRtc_UWord32 fec_rate = 0;
+ WebRtc_UWord32 nackRate = 0;
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->BitrateSent(&stream_rate, &video_rate, &fec_rate, &nackRate);
+ total_bitrate_sent += stream_rate;
+ fec_bitrate_sent += fec_rate;
+ nackBitrateSent += nackRate;
+ }
+}
+
+int ViEChannel::GetEstimatedReceiveBandwidth(
+ WebRtc_UWord32* estimated_bandwidth) const {
+ return rtp_rtcp_.EstimatedReceiveBandwidth(estimated_bandwidth);
+}
+
+WebRtc_Word32 ViEChannel::SetKeepAliveStatus(
+ const bool enable,
+ const WebRtc_Word8 unknown_payload_type,
+ const WebRtc_UWord16 delta_transmit_timeMS) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ if (enable && rtp_rtcp_.RTPKeepalive()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP keepalive already enabled", __FUNCTION__);
+ return -1;
+ } else if (!enable && !rtp_rtcp_.RTPKeepalive()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: RTP keepalive already disabled", __FUNCTION__);
+ return -1;
+ }
+
+ if (rtp_rtcp_.SetRTPKeepaliveStatus(enable, unknown_payload_type,
+ delta_transmit_timeMS) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set RTP keepalive status %d", __FUNCTION__,
+ enable);
+ if (enable == false && !rtp_rtcp_.DefaultModuleRegistered()) {
+ // Not sending media and we try to disable keep alive
+ rtp_rtcp_.ResetSendDataCountersRTP();
+ rtp_rtcp_.SetSendingStatus(false);
+ }
+ return -1;
+ }
+
+ if (enable && !rtp_rtcp_.Sending()) {
+ // Enable sending to start sending Sender reports instead of receive
+ // reports.
+ if (rtp_rtcp_.SetSendingStatus(true) != 0) {
+ rtp_rtcp_.SetRTPKeepaliveStatus(false, 0, 0);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not start sending", __FUNCTION__);
+ return -1;
+ }
+ } else if (!enable && !rtp_rtcp_.SendingMedia()) {
+ // Not sending media and we're disabling keep alive.
+ rtp_rtcp_.ResetSendDataCountersRTP();
+ if (rtp_rtcp_.SetSendingStatus(false) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not stop sending", __FUNCTION__);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::GetKeepAliveStatus(
+ bool& enabled,
+ WebRtc_Word8& unknown_payload_type,
+ WebRtc_UWord16& delta_transmit_time_ms) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (rtp_rtcp_.RTPKeepaliveStatus(&enabled, &unknown_payload_type,
+ &delta_transmit_time_ms) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get RTP keepalive status", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: enabled = %d, unknown_payload_type = %d, "
+ "delta_transmit_time_ms = %ul",
+ __FUNCTION__, enabled, (WebRtc_Word32) unknown_payload_type,
+ delta_transmit_time_ms);
+
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
+ RTPDirections direction) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (direction != kRtpIncoming && direction != kRtpOutgoing) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: invalid input", __FUNCTION__);
+ return -1;
+ }
+
+ if (direction == kRtpIncoming) {
+ return vie_receiver_.StartRTPDump(file_nameUTF8);
+ } else {
+ return vie_sender_.StartRTPDump(file_nameUTF8);
+ }
+}
+
+WebRtc_Word32 ViEChannel::StopRTPDump(RTPDirections direction) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ if (direction != kRtpIncoming && direction != kRtpOutgoing) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: invalid input", __FUNCTION__);
+ return -1;
+ }
+
+ if (direction == kRtpIncoming) {
+ return vie_receiver_.StopRTPDump();
+ } else {
+ return vie_sender_.StopRTPDump();
+ }
+}
+
+WebRtc_Word32 ViEChannel::SetLocalReceiver(const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_Word8* ip_address) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ callback_cs_->Enter();
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.Receiving()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: already receiving", __FUNCTION__);
+ return -1;
+ }
+
+ const WebRtc_Word8* multicast_ip_address = NULL;
+ if (socket_transport_.InitializeReceiveSockets(&vie_receiver_, rtp_port,
+ ip_address,
+ multicast_ip_address,
+ rtcp_port) != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not initialize receive sockets. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetLocalReceiver(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ callback_cs_->Enter();
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.ReceiveSocketsInitialized() == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: receive sockets not initialized", __FUNCTION__);
+ return -1;
+ }
+
+ WebRtc_Word8 multicast_ip_address[UdpTransport::kIpAddressVersion6Length];
+ if (socket_transport_.ReceiveSocketInformation(ip_address, rtp_port,
+ rtcp_port,
+ multicast_ip_address) != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not get receive socket information. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::SetSendDestination(
+ const WebRtc_Word8* ip_address,
+ const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_UWord16 source_rtp_port,
+ const WebRtc_UWord16 source_rtcp_port) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ callback_cs_->Enter();
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ const bool is_ipv6 = socket_transport_.IpV6Enabled();
+ if (UdpTransport::IsIpAddressValid(ip_address, is_ipv6) == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Not a valid IP address: %s", __FUNCTION__, ip_address);
+ return -1;
+ }
+ if (socket_transport_.InitializeSendSockets(ip_address, rtp_port,
+ rtcp_port)!= 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not initialize send socket. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+
+ if (source_rtp_port != 0) {
+ WebRtc_UWord16 receive_rtp_port = 0;
+ WebRtc_UWord16 receive_rtcp_port = 0;
+ if (socket_transport_.ReceiveSocketInformation(NULL, receive_rtp_port,
+ receive_rtcp_port,
+ NULL) != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not get receive port information. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ // Initialize an extra socket only if send port differs from receive
+ // port.
+ if (source_rtp_port != receive_rtp_port) {
+ if (socket_transport_.InitializeSourcePorts(source_rtp_port,
+ source_rtcp_port) != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not set source ports. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ }
+ }
+ vie_sender_.RegisterSendTransport(&socket_transport_);
+
+ // Workaround to avoid SSRC colision detection in loppback tests.
+ if (!is_ipv6) {
+ WebRtc_UWord32 local_host_address = 0;
+ const WebRtc_UWord32 current_ip_address =
+ UdpTransport::InetAddrIPV4(ip_address);
+
+ if ((UdpTransport::LocalHostAddress(local_host_address) == 0 &&
+ local_host_address == current_ip_address) ||
+ strncmp("127.0.0.1", ip_address, 9) == 0) {
+ rtp_rtcp_.SetSSRC(0xFFFFFFFF);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "Running in loopback. Forcing fixed SSRC");
+ }
+ } else {
+ char local_host_address[16];
+ char current_ip_address[16];
+
+ WebRtc_Word32 conv_result =
+ UdpTransport::LocalHostAddressIPV6(local_host_address);
+ conv_result += socket_transport_.InetPresentationToNumeric(
+ 23, ip_address, current_ip_address);
+ if (conv_result == 0) {
+ bool local_host = true;
+ for (WebRtc_Word32 i = 0; i < 16; i++) {
+ if (local_host_address[i] != current_ip_address[i]) {
+ local_host = false;
+ break;
+ }
+ }
+ if (!local_host) {
+ local_host = true;
+ for (WebRtc_Word32 i = 0; i < 15; i++) {
+ if (current_ip_address[i] != 0) {
+ local_host = false;
+ break;
+ }
+ }
+ if (local_host == true && current_ip_address[15] != 1) {
+ local_host = false;
+ }
+ }
+ if (local_host) {
+ rtp_rtcp_.SetSSRC(0xFFFFFFFF);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Running in loopback. Forcing fixed SSRC");
+ }
+ }
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetSendDestination(
+ WebRtc_Word8* ip_address,
+ WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_UWord16& source_rtp_port,
+ WebRtc_UWord16& source_rtcp_port) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ callback_cs_->Enter();
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.SendSocketsInitialized() == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: send sockets not initialized", __FUNCTION__);
+ return -1;
+ }
+ if (socket_transport_.SendSocketInformation(ip_address, rtp_port, rtcp_port)
+ != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not get send socket information. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ source_rtp_port = 0;
+ source_rtcp_port = 0;
+ if (socket_transport_.SourcePortsInitialized()) {
+ socket_transport_.SourcePorts(source_rtp_port, source_rtcp_port);
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::StartSend() {
+ CriticalSectionScoped cs(callback_cs_.get());
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!external_transport_) {
+ if (socket_transport_.SendSocketsInitialized() == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: send sockets not initialized", __FUNCTION__);
+ return -1;
+ }
+ }
+#endif
+ rtp_rtcp_.SetSendingMediaStatus(true);
+
+ if (rtp_rtcp_.Sending() && !rtp_rtcp_.RTPKeepalive()) {
+ if (rtp_rtcp_.RTPKeepalive()) {
+ // Sending Keep alive, don't trigger an error.
+ return 0;
+ }
+ // Already sending.
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Already sending", __FUNCTION__);
+ return kViEBaseAlreadySending;
+ }
+ if (rtp_rtcp_.SetSendingStatus(true) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not start sending RTP", __FUNCTION__);
+ return -1;
+ }
+ for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetSendingMediaStatus(true);
+ rtp_rtcp->SetSendingStatus(true);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::StopSend() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ rtp_rtcp_.SetSendingMediaStatus(false);
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetSendingMediaStatus(false);
+ }
+ if (rtp_rtcp_.RTPKeepalive()) {
+ // Don't turn off sending since we'll send keep alive packets.
+ return 0;
+ }
+ if (!rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Not sending", __FUNCTION__);
+ return kViEBaseNotSending;
+ }
+
+ // Reset.
+ rtp_rtcp_.ResetSendDataCountersRTP();
+ if (rtp_rtcp_.SetSendingStatus(false) != 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not stop RTP sending", __FUNCTION__);
+ return -1;
+ }
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->ResetSendDataCountersRTP();
+ rtp_rtcp->SetSendingStatus(false);
+ }
+ return 0;
+}
+
+bool ViEChannel::Sending() {
+ return rtp_rtcp_.Sending();
+}
+
+WebRtc_Word32 ViEChannel::StartReceive() {
+ CriticalSectionScoped cs(callback_cs_.get());
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!external_transport_) {
+ if (socket_transport_.Receiving()) {
+ // Warning, don't return error.
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: already receiving", __FUNCTION__);
+ return 0;
+ }
+ if (socket_transport_.ReceiveSocketsInitialized() == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: receive sockets not initialized", __FUNCTION__);
+ return -1;
+ }
+ if (socket_transport_.StartReceiving(kViENumReceiveSocketBuffers) != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not get receive socket information. Socket error:%d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ }
+#endif
+ if (StartDecodeThread() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not start decoder thread", __FUNCTION__);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ socket_transport_.StopReceiving();
+#endif
+ vie_receiver_.StopReceive();
+ return -1;
+ }
+ vie_receiver_.StartReceive();
+
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::StopReceive() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ vie_receiver_.StopReceive();
+ StopDecodeThread();
+ vcm_.ResetDecoder();
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ return 0;
+ }
+ }
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.Receiving() == false) {
+ // Warning, don't return error
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: not receiving",
+ __FUNCTION__);
+ return 0;
+ }
+ if (socket_transport_.StopReceiving() != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Socket error: %d", __FUNCTION__, socket_error);
+ return -1;
+ }
+#endif
+
+ return 0;
+}
+
+bool ViEChannel::Receiving() {
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ return socket_transport_.Receiving();
+#else
+ return false;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetSourceInfo(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address,
+ WebRtc_UWord32 ip_address_length) {
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.IpV6Enabled() &&
+ ip_address_length < UdpTransport::kIpAddressVersion6Length) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: IP address length is too small for IPv6", __FUNCTION__);
+ return -1;
+ } else if (ip_address_length < UdpTransport::kIpAddressVersion4Length) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: IP address length is too small for IPv4", __FUNCTION__);
+ return -1;
+ }
+
+ if (socket_transport_.RemoteSocketInformation(ip_address, rtp_port, rtcp_port)
+ != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Error getting source ports. Socket error: %d",
+ __FUNCTION__, socket_error);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+WebRtc_Word32 ViEChannel::RegisterSendTransport(Transport& transport) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.SendSocketsInitialized() ||
+ socket_transport_.ReceiveSocketsInitialized()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: socket transport already initialized", __FUNCTION__);
+ return -1;
+ }
+#endif
+ if (rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Sending", __FUNCTION__);
+ return -1;
+ }
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: transport already registered", __FUNCTION__);
+ return -1;
+ }
+ external_transport_ = &transport;
+ vie_sender_.RegisterSendTransport(&transport);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Transport registered: 0x%p", __FUNCTION__,
+ &external_transport_);
+
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::DeregisterSendTransport() {
+ CriticalSectionScoped cs(callback_cs_.get());
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (!external_transport_) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no transport registered", __FUNCTION__);
+ return -1;
+ }
+ if (rtp_rtcp_.Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Sending", __FUNCTION__);
+ return -1;
+ }
+ external_transport_ = NULL;
+ vie_sender_.DeregisterSendTransport();
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::ReceivedRTPPacket(
+ const void* rtp_packet, const WebRtc_Word32 rtp_packet_length) {
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!external_transport_) {
+ return -1;
+ }
+ }
+ return vie_receiver_.ReceivedRTPPacket(rtp_packet, rtp_packet_length);
+}
+
+WebRtc_Word32 ViEChannel::ReceivedRTCPPacket(
+ const void* rtcp_packet, const WebRtc_Word32 rtcp_packet_length) {
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!external_transport_) {
+ return -1;
+ }
+ }
+ return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
+}
+
+WebRtc_Word32 ViEChannel::EnableIPv6() {
+ callback_cs_->Enter();
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.IpV6Enabled()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: IPv6 already enabled", __FUNCTION__);
+ return -1;
+ }
+
+ if (socket_transport_.EnableIpV6() != 0) {
+ WebRtc_Word32 socket_error = socket_transport_.LastError();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not enable IPv6. Socket error: %d", __FUNCTION__,
+ socket_error);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+bool ViEChannel::IsIPv6Enabled() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return false;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ return socket_transport_.IpV6Enabled();
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return false;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::SetSourceFilter(const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_Word8* ip_address) {
+ callback_cs_->Enter();
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.SetFilterIP(ip_address) != 0) {
+ // Logging done in module.
+ return -1;
+ }
+ if (socket_transport_.SetFilterPorts(rtp_port, rtcp_port) != 0) {
+ // Logging done.
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetSourceFilter(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address) const {
+ callback_cs_->Enter();
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ if (external_transport_) {
+ callback_cs_->Leave();
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ callback_cs_->Leave();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.FilterIP(ip_address) != 0) {
+ // Logging done in module.
+ return -1;
+ }
+ if (socket_transport_.FilterPorts(rtp_port, rtcp_port) != 0) {
+ // Logging done in module.
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::SetToS(const WebRtc_Word32 DSCP,
+ const bool use_set_sockOpt) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.SetToS(DSCP, use_set_sockOpt) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Socket error: %d", __FUNCTION__,
+ socket_transport_.LastError());
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetToS(WebRtc_Word32& DSCP,
+ bool& use_set_sockOpt) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.ToS(DSCP, use_set_sockOpt) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Socket error: %d", __FUNCTION__,
+ socket_transport_.LastError());
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::SetSendGQoS(const bool enable,
+ const WebRtc_Word32 service_type,
+ const WebRtc_UWord32 max_bitrate,
+ const WebRtc_Word32 overrideDSCP) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.SetQoS(enable, service_type, max_bitrate, overrideDSCP,
+ false) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Socket error: %d", __FUNCTION__,
+ socket_transport_.LastError());
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::GetSendGQoS(bool& enabled,
+ WebRtc_Word32& service_type,
+ WebRtc_Word32& overrideDSCP) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.QoS(enabled, service_type, overrideDSCP) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Socket error: %d", __FUNCTION__,
+ socket_transport_.LastError());
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::SetMTU(WebRtc_UWord16 mtu) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (rtp_rtcp_.SetMaxTransferUnit(mtu) != 0) {
+ // Logging done.
+ return -1;
+ }
+ for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->SetMaxTransferUnit(mtu);
+ }
+ return 0;
+}
+
+WebRtc_UWord16 ViEChannel::MaxDataPayloadLength() const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ return rtp_rtcp_.MaxDataPayloadLength();
+}
+
+WebRtc_Word32 ViEChannel::SetPacketTimeoutNotification(
+ bool enable, WebRtc_UWord32 timeout_seconds) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (enable) {
+ WebRtc_UWord32 timeout_ms = 1000 * timeout_seconds;
+ if (rtp_rtcp_.SetPacketTimeout(timeout_ms, 0) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ return -1;
+ }
+ } else {
+ if (rtp_rtcp_.SetPacketTimeout(0, 0) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterNetworkObserver(
+ ViENetworkObserver* observer) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (observer) {
+ if (networkObserver_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer alread added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer added", __FUNCTION__);
+ networkObserver_ = observer;
+ } else {
+ if (!networkObserver_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no observer added", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: observer removed", __FUNCTION__);
+ networkObserver_ = NULL;
+ }
+ return 0;
+}
+
+bool ViEChannel::NetworkObserverRegistered() {
+ CriticalSectionScoped cs(callback_cs_.get());
+ return networkObserver_ != NULL;
+}
+
+WebRtc_Word32 ViEChannel::SetPeriodicDeadOrAliveStatus(
+ const bool enable, const WebRtc_UWord32 sample_time_seconds) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!networkObserver_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no observer added", __FUNCTION__);
+ return -1;
+ }
+
+ bool enabled = false;
+ WebRtc_UWord8 current_sampletime_seconds = 0;
+
+ // Get old settings.
+ rtp_rtcp_.PeriodicDeadOrAliveStatus(enabled, current_sampletime_seconds);
+ // Set new settings.
+ if (rtp_rtcp_.SetPeriodicDeadOrAliveStatus(
+ enable, static_cast<WebRtc_UWord8>(sample_time_seconds)) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not set periodic dead-or-alive status",
+ __FUNCTION__);
+ return -1;
+ }
+ if (!enable) {
+ // Restore last utilized sample time.
+ // Without this trick, the sample time would always be reset to default
+ // (2 sec), each time dead-or-alive was disabled without sample-time
+ // parameter.
+ rtp_rtcp_.SetPeriodicDeadOrAliveStatus(enable, current_sampletime_seconds);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SendUDPPacket(const WebRtc_Word8* data,
+ const WebRtc_UWord32 length,
+ WebRtc_Word32& transmitted_bytes,
+ bool use_rtcp_socket) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: External transport registered", __FUNCTION__);
+ return -1;
+ }
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ transmitted_bytes = socket_transport_.SendRaw(data, length, use_rtcp_socket);
+ if (transmitted_bytes == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not available for external transport", __FUNCTION__);
+ return -1;
+#endif
+}
+
+WebRtc_Word32 ViEChannel::EnableColorEnhancement(bool enable) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(enable: %d)", __FUNCTION__, enable);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (enable && color_enhancement_) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Already enabled", __FUNCTION__);
+ return -1;
+ } else if (!enable && !color_enhancement_) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: not enabled", __FUNCTION__);
+ return -1;
+ }
+ color_enhancement_ = enable;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterSendRtpRtcpModule(
+ RtpRtcp& send_rtp_rtcp_module) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ WebRtc_Word32 ret_val = rtp_rtcp_.RegisterDefaultModule(
+ &send_rtp_rtcp_module);
+ if (ret_val == 0) {
+ // We need to store this for the SetSendCodec call.
+ default_rtp_rtcp_ = &send_rtp_rtcp_module;
+ }
+ return ret_val;
+}
+
+WebRtc_Word32 ViEChannel::DeregisterSendRtpRtcpModule() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ default_rtp_rtcp_ = NULL;
+
+ for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+ it != simulcast_rtp_rtcp_.end();
+ it++) {
+ RtpRtcp* rtp_rtcp = *it;
+ rtp_rtcp->DeRegisterDefaultModule();
+ }
+ return rtp_rtcp_.DeRegisterDefaultModule();
+}
+
+RtpRtcp* ViEChannel::rtp_rtcp() {
+ return &rtp_rtcp_;
+}
+
+
+WebRtc_Word32 ViEChannel::FrameToRender(VideoFrame& video_frame) {
+ CriticalSectionScoped cs(callback_cs_.get());
+
+ if (decoder_reset_) {
+ // Trigger a callback to the user if the incoming codec has changed.
+ if (codec_observer_) {
+ VideoCodec decoder;
+ memset(&decoder, 0, sizeof(decoder));
+ if (vcm_.ReceiveCodec(&decoder) == VCM_OK) {
+ // VCM::ReceiveCodec returns the codec set by
+ // RegisterReceiveCodec, which might not be the size we're
+ // actually decoding.
+ decoder.width = static_cast<unsigned short>(video_frame.Width());
+ decoder.height = static_cast<unsigned short>(video_frame.Height());
+ codec_observer_->IncomingCodecChanged(channel_id_, decoder);
+ } else {
+ assert(false);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: Could not get receive codec", __FUNCTION__);
+ }
+ }
+ decoder_reset_ = false;
+ }
+ if (effect_filter_) {
+ effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
+ video_frame.TimeStamp(), video_frame.Width(),
+ video_frame.Height());
+ }
+ if (color_enhancement_) {
+ VideoProcessingModule::ColorEnhancement(video_frame);
+ }
+
+ // Record videoframe.
+ file_recorder_.RecordVideoFrame(video_frame);
+
+ WebRtc_UWord32 arr_ofCSRC[kRtpCsrcSize];
+ WebRtc_Word32 no_of_csrcs = rtp_rtcp_.RemoteCSRCs(arr_ofCSRC);
+ if (no_of_csrcs <= 0) {
+ arr_ofCSRC[0] = rtp_rtcp_.RemoteSSRC();
+ no_of_csrcs = 1;
+ }
+ DeliverFrame(video_frame, no_of_csrcs, arr_ofCSRC);
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 picture_id) {
+ return rtp_rtcp_.SendRTCPReferencePictureSelection(picture_id);
+}
+
+WebRtc_Word32 ViEChannel::StoreReceivedFrame(
+ const EncodedVideoData& frame_to_store) {
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::ReceiveStatistics(const WebRtc_UWord32 bit_rate,
+ const WebRtc_UWord32 frame_rate) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (codec_observer_) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: bitrate %u, framerate %u", __FUNCTION__, bit_rate,
+ frame_rate);
+ codec_observer_->IncomingRate(channel_id_, frame_rate, bit_rate);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RequestKeyFrame() {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (codec_observer_ && do_key_frame_callbackRequest_) {
+ codec_observer_->RequestNewKeyFrame(channel_id_);
+ }
+ }
+ return rtp_rtcp_.RequestKeyFrame();
+}
+
+WebRtc_Word32 ViEChannel::SliceLossIndicationRequest(
+ const WebRtc_UWord64 picture_id) {
+ return rtp_rtcp_.SendRTCPSliceLossIndication((WebRtc_UWord8) picture_id);
+}
+
+WebRtc_Word32 ViEChannel::ResendPackets(const WebRtc_UWord16* sequence_numbers,
+ WebRtc_UWord16 length) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(length: %d)", __FUNCTION__, length);
+ return rtp_rtcp_.SendNACK(sequence_numbers, length);
+}
+
+bool ViEChannel::ChannelDecodeThreadFunction(void* obj) {
+ return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess();
+}
+
+bool ViEChannel::ChannelDecodeProcess() {
+ // Decode is blocking, but sleep some time anyway to not get a spin.
+ vcm_.Decode(kMaxDecodeWaitTimeMs);
+
+ if ((TickTime::Now() - vcm_rttreported_).Milliseconds() > 1000) {
+ WebRtc_UWord16 RTT;
+ WebRtc_UWord16 avgRTT;
+ WebRtc_UWord16 minRTT;
+ WebRtc_UWord16 maxRTT;
+
+ if (rtp_rtcp_.RTT(rtp_rtcp_.RemoteSSRC(), &RTT, &avgRTT, &minRTT, &maxRTT)
+ == 0) {
+ vcm_.SetReceiveChannelParameters(RTT);
+ vcm_rttreported_ = TickTime::Now();
+ } else if (!rtp_rtcp_.Sending() &&
+ (TickTime::Now() - vcm_rttreported_).Milliseconds() > 5000) {
+ // Wait at least 5 seconds before faking a 200 ms RTT. This is to
+ // make sure we have a chance to start sending before we decide to fake.
+ vcm_.SetReceiveChannelParameters(200);
+ vcm_rttreported_ = TickTime::Now();
+ }
+ }
+ return true;
+}
+
+WebRtc_Word32 ViEChannel::StartDecodeThread() {
+ // Start the decode thread
+ if (decode_thread_) {
+ // Already started.
+ return 0;
+ }
+ decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction,
+ this, kHighestPriority,
+ "DecodingThread");
+ if (!decode_thread_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not create decode thread", __FUNCTION__);
+ return -1;
+ }
+
+ unsigned int thread_id;
+ if (decode_thread_->Start(thread_id) == false) {
+ delete decode_thread_;
+ decode_thread_ = NULL;
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not start decode thread", __FUNCTION__);
+ return -1;
+ }
+
+ // Used to make sure that we don't give the VCM a faked RTT
+ // too early.
+ vcm_rttreported_ = TickTime::Now();
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: decode thread with id %u started", __FUNCTION__);
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::StopDecodeThread() {
+ if (!decode_thread_) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: decode thread not running", __FUNCTION__);
+ return 0;
+ }
+
+ decode_thread_->SetNotAlive();
+ if (decode_thread_->Stop()) {
+ delete decode_thread_;
+ } else {
+ // Couldn't stop the thread, leak instead of crash.
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: could not stop decode thread", __FUNCTION__);
+ assert(!"could not stop decode thread");
+ }
+ decode_thread_ = NULL;
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::RegisterExternalEncryption(Encryption* encryption) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (external_encryption_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external encryption already registered", __FUNCTION__);
+ return -1;
+ }
+
+ external_encryption_ = encryption;
+
+ vie_receiver_.RegisterExternalDecryption(encryption);
+ vie_sender_.RegisterExternalEncryption(encryption);
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s", "external encryption object registerd with channel=%d",
+ channel_id_);
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::DeRegisterExternalEncryption() {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!external_encryption_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: external encryption is not registered", __FUNCTION__);
+ return -1;
+ }
+
+ external_transport_ = NULL;
+ vie_receiver_.DeregisterExternalDecryption();
+ vie_sender_.DeregisterExternalEncryption();
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s external encryption object de-registerd with channel=%d",
+ __FUNCTION__, channel_id_);
+ return 0;
+}
+
+WebRtc_Word32 ViEChannel::SetVoiceChannel(WebRtc_Word32 ve_channel_id,
+ VoEVideoSync* ve_sync_interface) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s, audio channel %d, video channel %d", __FUNCTION__,
+ ve_channel_id, channel_id_);
+
+ if (ve_sync_interface) {
+ // Register lip sync
+ module_process_thread_.RegisterModule(&vie_sync_);
+ } else {
+ module_process_thread_.DeRegisterModule(&vie_sync_);
+ }
+ return vie_sync_.SetVoiceChannel(ve_channel_id, ve_sync_interface);
+}
+
+WebRtc_Word32 ViEChannel::VoiceChannel() {
+ return vie_sync_.VoiceChannel();
+}
+
+WebRtc_Word32 ViEChannel::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!effect_filter) {
+ if (!effect_filter_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: no effect filter added for channel %d",
+ __FUNCTION__, channel_id_);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: deregister effect filter for device %d", __FUNCTION__,
+ channel_id_);
+ } else {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: register effect filter for device %d", __FUNCTION__,
+ channel_id_);
+ if (effect_filter_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: effect filter already added for channel %d",
+ __FUNCTION__, channel_id_);
+ return -1;
+ }
+ }
+ effect_filter_ = effect_filter;
+ return 0;
+}
+
+ViEFileRecorder& ViEChannel::GetIncomingFileRecorder() {
+ // Start getting callback of all frames before they are decoded.
+ vcm_.RegisterFrameStorageCallback(this);
+ return file_recorder_;
+}
+
+void ViEChannel::ReleaseIncomingFileRecorder() {
+ // Stop getting callback of all frames before they are decoded.
+ vcm_.RegisterFrameStorageCallback(NULL);
+}
+
+void ViEChannel::OnLipSyncUpdate(const WebRtc_Word32 id,
+ const WebRtc_Word32 audio_video_offset) {
+ if (channel_id_ != ChannelId(id)) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s, incorrect id", __FUNCTION__, id);
+ return;
+ }
+ vie_sync_.SetNetworkDelay(audio_video_offset);
+}
+
+void ViEChannel::OnApplicationDataReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 sub_type,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8* data) {
+ if (channel_id_ != ChannelId(id)) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s, incorrect id", __FUNCTION__, id);
+ return;
+ }
+ CriticalSectionScoped cs(callback_cs_.get());
+ {
+ if (rtcp_observer_) {
+ rtcp_observer_->OnApplicationDataReceived(
+ channel_id_, sub_type, name, reinterpret_cast<const char*>(data),
+ length);
+ }
+ }
+}
+
+WebRtc_Word32 ViEChannel::OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payload_type,
+ const WebRtc_Word8 payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: payload_type %d, payload_name %s", __FUNCTION__,
+ payload_type, payload_name);
+ vcm_.ResetDecoder();
+
+ callback_cs_->Enter();
+ decoder_reset_ = true;
+ callback_cs_->Leave();
+ return 0;
+}
+
+void ViEChannel::OnPacketTimeout(const WebRtc_Word32 id) {
+ assert(ChannelId(id) == channel_id_);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (networkObserver_) {
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (socket_transport_.Receiving() || external_transport_) {
+#else
+ if (external_transport_) {
+#endif
+ networkObserver_->PacketTimeout(channel_id_, NoPacket);
+ rtp_packet_timeout_ = true;
+ }
+ }
+}
+
+void ViEChannel::OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packet_type) {
+ assert(ChannelId(id) == channel_id_);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (rtp_packet_timeout_ && packet_type == kPacketRtp) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (networkObserver_) {
+ networkObserver_->PacketTimeout(channel_id_, PacketReceived);
+ }
+
+ // Reset even if no observer set, might have been removed during timeout.
+ rtp_packet_timeout_ = false;
+ }
+}
+
+void ViEChannel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive) {
+ assert(ChannelId(id) == channel_id_);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s(id=%d, alive=%d)", __FUNCTION__, id, alive);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (!networkObserver_) {
+ return;
+ }
+ bool is_alive = true;
+ if (alive == kRtpDead) {
+ is_alive = false;
+ }
+ networkObserver_->OnPeriodicDeadOrAlive(channel_id_, is_alive);
+ return;
+}
+
+void ViEChannel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC) {
+ if (channel_id_ != ChannelId(id)) {
+ assert(false);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s, incorrect id", __FUNCTION__, id);
+ return;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %u", __FUNCTION__, SSRC);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ {
+ if (rtp_observer_) {
+ rtp_observer_->IncomingSSRCChanged(channel_id_, SSRC);
+ }
+ }
+}
+
+void ViEChannel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %u added: %d", __FUNCTION__, CSRC, added);
+
+ if (channel_id_ != ChannelId(id)) {
+ assert(false);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s, incorrect id", __FUNCTION__, id);
+ return;
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: %u", __FUNCTION__, CSRC);
+
+ CriticalSectionScoped cs(callback_cs_.get());
+ {
+ if (rtp_observer_) {
+ rtp_observer_->IncomingCSRCChanged(channel_id_, CSRC, added);
+ }
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_channel.h b/trunk/src/video_engine/vie_channel.h
new file mode 100644
index 0000000..dac268f
--- /dev/null
+++ b/trunk/src/video_engine/vie_channel.h
@@ -0,0 +1,404 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// vie_channel.h
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+
+#include <list>
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "modules/udp_transport/interface/udp_transport.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_network.h"
+#include "video_engine/include/vie_rtp_rtcp.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_file_recorder.h"
+#include "video_engine/vie_frame_provider_base.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class Encryption;
+class ProcessThread;
+class RtpRtcp;
+class ThreadWrapper;
+class VideoCodingModule;
+class VideoDecoder;
+class VideoRenderCallback;
+class ViEDecoderObserver;
+class ViEEffectFilter;
+class ViENetworkObserver;
+class ViEReceiver;
+class ViERTCPObserver;
+class ViERTPObserver;
+class ViESender;
+class ViESyncModule;
+class VoEVideoSync;
+
+class ViEChannel
+ : public VCMFrameTypeCallback,
+ public VCMReceiveCallback,
+ public VCMReceiveStatisticsCallback,
+ public VCMPacketRequestCallback,
+ public VCMFrameStorageCallback,
+ public RtcpFeedback,
+ public RtpFeedback,
+ public ViEFrameProviderBase {
+ public:
+ ViEChannel(WebRtc_Word32 channel_id,
+ WebRtc_Word32 engine_id,
+ WebRtc_UWord32 number_of_cores,
+ ProcessThread& module_process_thread);
+ ~ViEChannel();
+
+ WebRtc_Word32 Init();
+
+ // Sets the encoder to use for the channel. |new_stream| indicates the encoder
+ // type has changed and we should start a new RTP stream.
+ WebRtc_Word32 SetSendCodec(const VideoCodec& video_codec,
+ bool new_stream = true);
+ WebRtc_Word32 SetReceiveCodec(const VideoCodec& video_codec);
+ WebRtc_Word32 GetReceiveCodec(VideoCodec& video_codec);
+ WebRtc_Word32 RegisterCodecObserver(ViEDecoderObserver* observer);
+ // Registers an external decoder. |decoder_render| is set to true if the
+ // decoder will do the rendering. If |decoder_render| is set,|render_delay|
+ // indicates the time needed to decode and render a frame.
+ WebRtc_Word32 RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
+ VideoDecoder* decoder,
+ bool decoder_render,
+ WebRtc_Word32 render_delay);
+ WebRtc_Word32 DeRegisterExternalDecoder(const WebRtc_UWord8 pl_type);
+ WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& num_key_frames,
+ WebRtc_UWord32& num_delta_frames);
+ WebRtc_UWord32 DiscardedPackets() const;
+
+ // Only affects calls to SetReceiveCodec done after this call.
+ WebRtc_Word32 WaitForKeyFrame(bool wait);
+
+ // If enabled, a key frame request will be sent as soon as there are lost
+ // packets. If |only_key_frames| are set, requests are only sent for loss in
+ // key frames.
+ WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool only_key_frames);
+
+ WebRtc_Word32 SetRTCPMode(const RTCPMethod rtcp_mode);
+ WebRtc_Word32 GetRTCPMode(RTCPMethod& rtcp_mode);
+ WebRtc_Word32 SetNACKStatus(const bool enable);
+ WebRtc_Word32 SetFECStatus(const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC);
+ WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC);
+ WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
+ bool EnableRemb(bool enable);
+ WebRtc_Word32 EnableTMMBR(const bool enable);
+ WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);
+
+ // Sets SSRC for outgoing stream.
+ WebRtc_Word32 SetSSRC(const WebRtc_UWord32 SSRC,
+ const StreamType usage,
+ const unsigned char simulcast_idx);
+
+ // Gets SSRC for outgoing stream.
+ WebRtc_Word32 GetLocalSSRC(WebRtc_UWord32& SSRC);
+
+ // Gets SSRC for the incoming stream.
+ WebRtc_Word32 GetRemoteSSRC(WebRtc_UWord32& SSRC);
+
+ // Gets the CSRC for the incoming stream.
+ WebRtc_Word32 GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]);
+
+ // Sets the starting sequence number, must be called before StartSend.
+ WebRtc_Word32 SetStartSequenceNumber(WebRtc_UWord16 sequence_number);
+
+ // Sets the CName for the outgoing stream on the channel.
+ WebRtc_Word32 SetRTCPCName(const WebRtc_Word8 rtcp_cname[]);
+
+ // Gets the CName for the outgoing stream on the channel.
+ WebRtc_Word32 GetRTCPCName(WebRtc_Word8 rtcp_cname[]);
+
+ // Gets the CName of the incoming stream.
+ WebRtc_Word32 GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]);
+ WebRtc_Word32 RegisterRtpObserver(ViERTPObserver* observer);
+ WebRtc_Word32 RegisterRtcpObserver(ViERTCPObserver* observer);
+ WebRtc_Word32 SendApplicationDefinedRTCPPacket(
+ const WebRtc_UWord8 sub_type,
+ WebRtc_UWord32 name,
+ const WebRtc_UWord8* data,
+ WebRtc_UWord16 data_length_in_bytes);
+
+ // Returns statistics reported by the remote client in an RTCP packet.
+ WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
+ WebRtc_UWord32& cumulative_lost,
+ WebRtc_UWord32& extended_max,
+ WebRtc_UWord32& jitter_samples,
+ WebRtc_Word32& rtt_ms);
+
+ // Returns our localy created statistics of the received RTP stream.
+ WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fraction_lost,
+ WebRtc_UWord32& cumulative_lost,
+ WebRtc_UWord32& extended_max,
+ WebRtc_UWord32& jitter_samples,
+ WebRtc_Word32& rtt_ms);
+
+ // Gets sent/received packets statistics.
+ WebRtc_Word32 GetRtpStatistics(WebRtc_UWord32& bytes_sent,
+ WebRtc_UWord32& packets_sent,
+ WebRtc_UWord32& bytes_received,
+ WebRtc_UWord32& packets_received) const;
+ void GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
+ WebRtc_UWord32& video_bitrate_sent,
+ WebRtc_UWord32& fec_bitrate_sent,
+ WebRtc_UWord32& nackBitrateSent) const;
+ int GetEstimatedReceiveBandwidth(WebRtc_UWord32* estimated_bandwidth) const;
+ WebRtc_Word32 SetKeepAliveStatus(const bool enable,
+ const WebRtc_Word8 unknown_payload_type,
+ const WebRtc_UWord16 delta_transmit_timeMS);
+ WebRtc_Word32 GetKeepAliveStatus(bool& enable,
+ WebRtc_Word8& unknown_payload_type,
+ WebRtc_UWord16& delta_transmit_timeMS);
+ WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
+ RTPDirections direction);
+ WebRtc_Word32 StopRTPDump(RTPDirections direction);
+
+ // Implements RtcpFeedback.
+ virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
+ const WebRtc_Word32 audio_video_offset);
+ virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 sub_type,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8* data);
+
+ // Implements RtpFeedback.
+ virtual WebRtc_Word32 OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payload_type,
+ const WebRtc_Word8 payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
+ virtual void OnPacketTimeout(const WebRtc_Word32 id);
+ virtual void OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packet_type);
+ virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive);
+ virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC);
+ virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added);
+
+ WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_Word8* ip_address);
+ WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address) const;
+ WebRtc_Word32 SetSendDestination(const WebRtc_Word8* ip_address,
+ const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_UWord16 source_rtp_port,
+ const WebRtc_UWord16 source_rtcp_port);
+ WebRtc_Word32 GetSendDestination(WebRtc_Word8* ip_address,
+ WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_UWord16& source_rtp_port,
+ WebRtc_UWord16& source_rtcp_port) const;
+ WebRtc_Word32 GetSourceInfo(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address,
+ WebRtc_UWord32 ip_address_length);
+
+ WebRtc_Word32 SetRemoteSSRCType(const StreamType usage,
+ const uint32_t SSRC) const;
+
+ WebRtc_Word32 StartSend();
+ WebRtc_Word32 StopSend();
+ bool Sending();
+ WebRtc_Word32 StartReceive();
+ WebRtc_Word32 StopReceive();
+ bool Receiving();
+
+ WebRtc_Word32 RegisterSendTransport(Transport& transport);
+ WebRtc_Word32 DeregisterSendTransport();
+
+ // Incoming packet from external transport.
+ WebRtc_Word32 ReceivedRTPPacket(const void* rtp_packet,
+ const WebRtc_Word32 rtp_packet_length);
+
+ // Incoming packet from external transport.
+ WebRtc_Word32 ReceivedRTCPPacket(const void* rtcp_packet,
+ const WebRtc_Word32 rtcp_packet_length);
+
+ WebRtc_Word32 EnableIPv6();
+ bool IsIPv6Enabled();
+ WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtp_port,
+ const WebRtc_UWord16 rtcp_port,
+ const WebRtc_Word8* ip_address);
+ WebRtc_Word32 GetSourceFilter(WebRtc_UWord16& rtp_port,
+ WebRtc_UWord16& rtcp_port,
+ WebRtc_Word8* ip_address) const;
+
+ WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool use_set_sockOpt);
+ WebRtc_Word32 GetToS(WebRtc_Word32& DSCP, bool& use_set_sockOpt) const;
+ WebRtc_Word32 SetSendGQoS(const bool enable,
+ const WebRtc_Word32 service_type,
+ const WebRtc_UWord32 max_bitrate,
+ const WebRtc_Word32 overrideDSCP);
+ WebRtc_Word32 GetSendGQoS(bool& enabled, WebRtc_Word32& service_type,
+ WebRtc_Word32& overrideDSCP) const;
+
+ // Sets the maximum transfer unit size for the network link, i.e. including
+ // IP, UDP and RTP headers.
+ WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
+
+ // Returns maximum allowed payload size, i.e. the maximum allowed size of
+ // encoded data in each packet.
+ WebRtc_UWord16 MaxDataPayloadLength() const;
+ WebRtc_Word32 SetMaxPacketBurstSize(WebRtc_UWord16 max_number_of_packets);
+ WebRtc_Word32 SetPacketBurstSpreadState(bool enable,
+ const WebRtc_UWord16 frame_periodMS);
+
+ WebRtc_Word32 SetPacketTimeoutNotification(bool enable,
+ WebRtc_UWord32 timeout_seconds);
+ WebRtc_Word32 RegisterNetworkObserver(ViENetworkObserver* observer);
+ bool NetworkObserverRegistered();
+ WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
+ const bool enable, const WebRtc_UWord32 sample_time_seconds);
+
+ WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
+ const WebRtc_UWord32 length,
+ WebRtc_Word32& transmitted_bytes,
+ bool use_rtcp_socket);
+
+ WebRtc_Word32 EnableColorEnhancement(bool enable);
+
+ // Register send RTP RTCP module, which will deliver encoded frames to the
+ // to the channel RTP module.
+ WebRtc_Word32 RegisterSendRtpRtcpModule(RtpRtcp& send_rtp_rtcp_module);
+
+ // Deregisters the send RTP RTCP module, which will stop the encoder input to
+ // the channel.
+ WebRtc_Word32 DeregisterSendRtpRtcpModule();
+
+ // Gets the modules used by the channel.
+ RtpRtcp* rtp_rtcp();
+
+ // Implements VCMReceiveCallback.
+ virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
+
+ // Implements VCMReceiveCallback.
+ virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
+ const WebRtc_UWord64 picture_id);
+
+ // Implements VCM.
+ virtual WebRtc_Word32 StoreReceivedFrame(
+ const EncodedVideoData& frame_to_store);
+
+ // Implements VideoReceiveStatisticsCallback.
+ virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bit_rate,
+ const WebRtc_UWord32 frame_rate);
+
+ // Implements VideoFrameTypeCallback.
+ virtual WebRtc_Word32 RequestKeyFrame();
+
+ // Implements VideoFrameTypeCallback.
+ virtual WebRtc_Word32 SliceLossIndicationRequest(
+ const WebRtc_UWord64 picture_id);
+
+ // Implements VideoPacketRequestCallback.
+ virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
+ WebRtc_UWord16 length);
+
+ WebRtc_Word32 RegisterExternalEncryption(Encryption* encryption);
+ WebRtc_Word32 DeRegisterExternalEncryption();
+
+ WebRtc_Word32 SetVoiceChannel(WebRtc_Word32 ve_channel_id,
+ VoEVideoSync* ve_sync_interface);
+ WebRtc_Word32 VoiceChannel();
+
+ // Implements ViEFrameProviderBase.
+ virtual int FrameCallbackChanged() {return -1;}
+
+ WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
+
+ ViEFileRecorder& GetIncomingFileRecorder();
+ void ReleaseIncomingFileRecorder();
+
+ protected:
+ static bool ChannelDecodeThreadFunction(void* obj);
+ bool ChannelDecodeProcess();
+
+ private:
+ // Assumed to be protected.
+ WebRtc_Word32 StartDecodeThread();
+ WebRtc_Word32 StopDecodeThread();
+
+ WebRtc_Word32 ProcessNACKRequest(const bool enable);
+ WebRtc_Word32 ProcessFECRequest(const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC);
+
+ WebRtc_Word32 channel_id_;
+ WebRtc_Word32 engine_id_;
+ WebRtc_UWord32 number_of_cores_;
+ WebRtc_UWord8 num_socket_threads_;
+
+ // Used for all registered callbacks except rendering.
+ scoped_ptr<CriticalSectionWrapper> callback_cs_;
+
+ // Owned modules/classes.
+ RtpRtcp& rtp_rtcp_;
+ RtpRtcp* default_rtp_rtcp_;
+ std::list<RtpRtcp*> simulcast_rtp_rtcp_;
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ UdpTransport& socket_transport_;
+#endif
+ VideoCodingModule& vcm_;
+ ViEReceiver& vie_receiver_;
+ ViESender& vie_sender_;
+ ViESyncModule& vie_sync_;
+
+ // Not owned.
+ ProcessThread& module_process_thread_;
+ ViEDecoderObserver* codec_observer_;
+ bool do_key_frame_callbackRequest_;
+ ViERTPObserver* rtp_observer_;
+ ViERTCPObserver* rtcp_observer_;
+ ViENetworkObserver* networkObserver_;
+ bool rtp_packet_timeout_;
+ bool using_packet_spread_;
+
+ Transport* external_transport_;
+
+ bool decoder_reset_;
+ bool wait_for_key_frame_;
+ ThreadWrapper* decode_thread_;
+
+ Encryption* external_encryption_;
+
+ ViEEffectFilter* effect_filter_;
+ bool color_enhancement_;
+
+ // Time when RTT time was last reported to VCM JB.
+ TickTime vcm_rttreported_;
+
+ ViEFileRecorder file_recorder_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
diff --git a/trunk/src/video_engine/vie_channel_manager.cc b/trunk/src/video_engine/vie_channel_manager.cc
new file mode 100644
index 0000000..268bc18
--- /dev/null
+++ b/trunk/src/video_engine/vie_channel_manager.cc
@@ -0,0 +1,484 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_channel_manager.h"
+
+#include "engine_configurations.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "modules/utility/interface/process_thread.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/map_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_remb.h"
+#include "voice_engine/main/interface/voe_video_sync.h"
+
+namespace webrtc {
+
+ViEChannelManager::ViEChannelManager(
+ int engine_id,
+ int number_of_cores,
+ ViEPerformanceMonitor& vie_performance_monitor)
+ : channel_id_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ engine_id_(engine_id),
+ number_of_cores_(number_of_cores),
+ vie_performance_monitor_(vie_performance_monitor),
+ free_channel_ids_(new bool[kViEMaxNumberOfChannels]),
+ free_channel_ids_size_(kViEMaxNumberOfChannels),
+ voice_sync_interface_(NULL),
+ remb_(new VieRemb(engine_id)),
+ voice_engine_(NULL),
+ module_process_thread_(NULL) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id),
+ "ViEChannelManager::ViEChannelManager(engine_id: %d)",
+ engine_id);
+ for (int idx = 0; idx < free_channel_ids_size_; idx++) {
+ free_channel_ids_[idx] = true;
+ }
+}
+
+ViEChannelManager::~ViEChannelManager() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_),
+ "ViEChannelManager Destructor, engine_id: %d", engine_id_);
+
+ module_process_thread_->DeRegisterModule(remb_.get());
+ while (channel_map_.size() > 0) {
+ ChannelMap::iterator it = channel_map_.begin();
+ // DeleteChannel will erase this channel from the map and invalidate |it|.
+ DeleteChannel(it->first);
+ }
+
+ if (voice_sync_interface_) {
+ voice_sync_interface_->Release();
+ }
+ if (channel_id_critsect_) {
+ delete channel_id_critsect_;
+ channel_id_critsect_ = NULL;
+ }
+ if (free_channel_ids_) {
+ delete[] free_channel_ids_;
+ free_channel_ids_ = NULL;
+ free_channel_ids_size_ = 0;
+ }
+}
+
+void ViEChannelManager::SetModuleProcessThread(
+ ProcessThread& module_process_thread) {
+ assert(!module_process_thread_);
+ module_process_thread_ = &module_process_thread;
+ module_process_thread_->RegisterModule(remb_.get());
+}
+
+int ViEChannelManager::CreateChannel(int& channel_id) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+
+ // Get a free id for the new channel.
+ if (!GetFreeChannelId(channel_id)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "Max number of channels reached: %d", channel_map_.size());
+ return -1;
+ }
+
+ ViEChannel* vie_channel = new ViEChannel(channel_id, engine_id_,
+ number_of_cores_,
+ *module_process_thread_);
+ if (!vie_channel) {
+ ReturnChannelId(channel_id);
+ return -1;
+ }
+ if (vie_channel->Init() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s could not init channel", __FUNCTION__, channel_id);
+ ReturnChannelId(channel_id);
+ delete vie_channel;
+ vie_channel = NULL;
+ return -1;
+ }
+
+ // There is no ViEEncoder for this channel, create one with default settings.
+ ViEEncoder* vie_encoder = new ViEEncoder(engine_id_, channel_id,
+ number_of_cores_,
+ *module_process_thread_);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s(video_channel_id: %d) - Could not create a new encoder",
+ __FUNCTION__, channel_id);
+ delete vie_channel;
+ return -1;
+ }
+
+ vie_encoder_map_[channel_id] = vie_encoder;
+ channel_map_[channel_id] = vie_channel;
+
+ // Register the channel at the encoder.
+ RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
+ if (vie_channel->RegisterSendRtpRtcpModule(*send_rtp_rtcp_module) != 0) {
+ assert(false);
+ vie_encoder_map_.erase(channel_id);
+ channel_map_.erase(channel_id);
+ ReturnChannelId(channel_id);
+ delete vie_channel;
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
+ "%s: Could not register rtp module %d", __FUNCTION__,
+ channel_id);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEChannelManager::CreateChannel(int& channel_id, int original_channel) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+
+ // Check that original_channel already exists.
+ ViEEncoder* vie_encoder = ViEEncoderPtr(original_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s: Original channel doesn't exist", __FUNCTION__,
+ original_channel);
+ return -1;
+ }
+ VideoCodec video_codec;
+ vie_encoder->GetEncoder(video_codec);
+
+ // Get a free id for the new channel.
+ if (GetFreeChannelId(channel_id) == false) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "Max number of channels reached: %d", channel_map_.size());
+ return -1;
+ }
+ ViEChannel* vie_channel = new ViEChannel(channel_id, engine_id_,
+ number_of_cores_,
+ *module_process_thread_);
+ if (!vie_channel) {
+ ReturnChannelId(channel_id);
+ return -1;
+ }
+ if (vie_channel->Init() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s could not init channel", __FUNCTION__, channel_id);
+ ReturnChannelId(channel_id);
+ delete vie_channel;
+ vie_channel = NULL;
+ return -1;
+ }
+ vie_encoder_map_[channel_id] = vie_encoder;
+
+ // Set the same encoder settings for the channel as used by the master
+ // channel. Do this before attaching rtp module to ensure all rtp children has
+ // the same codec type.
+ VideoCodec encoder;
+ if (vie_encoder->GetEncoder(encoder) == 0) {
+ vie_channel->SetSendCodec(encoder);
+ }
+ channel_map_[channel_id] = vie_channel;
+
+ // Register the channel at the encoder.
+ RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
+ if (vie_channel->RegisterSendRtpRtcpModule(*send_rtp_rtcp_module) != 0) {
+ assert(false);
+ vie_encoder_map_.erase(channel_id);
+ channel_map_.erase(channel_id);
+ ReturnChannelId(channel_id);
+ delete vie_channel;
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
+ "%s: Could not register rtp module %d", __FUNCTION__,
+ channel_id);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEChannelManager::DeleteChannel(int channel_id) {
+ ViEChannel* vie_channel = NULL;
+ ViEEncoder* vie_encoder = NULL;
+ {
+ // Write lock to make sure no one is using the channel.
+ ViEManagerWriteScoped wl(*this);
+
+ // Protect the map.
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ ChannelMap::iterator c_it = channel_map_.find(channel_id);
+ if (c_it == channel_map_.end()) {
+ // No such channel.
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s Channel doesn't exist: %d", __FUNCTION__, channel_id);
+ return -1;
+ }
+ vie_channel = c_it->second;
+ channel_map_.erase(c_it);
+
+ // Deregister possible remb modules.
+ RtpRtcp* rtp_module = vie_channel->rtp_rtcp();
+ remb_->RemoveRembSender(rtp_module);
+ remb_->RemoveReceiveChannel(rtp_module);
+
+ // Deregister the channel from the ViEEncoder to stop the media flow.
+ vie_channel->DeregisterSendRtpRtcpModule();
+ ReturnChannelId(channel_id);
+
+ // Find the encoder object.
+ EncoderMap::iterator e_it = vie_encoder_map_.find(channel_id);
+ assert(e_it != vie_encoder_map_.end());
+ vie_encoder = e_it->second;
+
+ remb_->RemoveSendChannel(vie_encoder->SendRtpRtcpModule());
+
+ // Check if other channels are using the same encoder.
+ if (ChannelUsingViEEncoder(channel_id)) {
+ // Don't delete the ViEEncoder, at least one other channel is using it.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
+ "%s ViEEncoder removed from map for channel %d, not deleted",
+ __FUNCTION__, channel_id);
+ vie_encoder = NULL;
+ } else {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
+ "%s ViEEncoder deleted for channel %d", __FUNCTION__,
+ channel_id);
+ // Delete later when we've released the critsect.
+ }
+
+ // We can't erase the item before we've checked for other channels using
+ // same ViEEncoder.
+ vie_encoder_map_.erase(e_it);
+ }
+
+ // Leave the write critsect before deleting the objects.
+ // Deleting a channel can cause other objects, such as renderers, to be
+ // deleted, which might take time.
+ if (vie_encoder) {
+ delete vie_encoder;
+ }
+ delete vie_channel;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
+ "%s Channel %d deleted", __FUNCTION__, channel_id);
+ return 0;
+}
+
+int ViEChannelManager::SetVoiceEngine(VoiceEngine* voice_engine) {
+ // Write lock to make sure no one is using the channel.
+ ViEManagerWriteScoped wl(*this);
+
+ CriticalSectionScoped cs(*channel_id_critsect_);
+
+ VoEVideoSync* sync_interface = NULL;
+ if (voice_engine) {
+ // Get new sync interface.
+ sync_interface = VoEVideoSync::GetInterface(voice_engine);
+ if (!sync_interface) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s Can't get audio sync interface from VoiceEngine.",
+ __FUNCTION__);
+ return -1;
+ }
+ }
+
+ for (ChannelMap::iterator it = channel_map_.begin(); it != channel_map_.end();
+ ++it) {
+ it->second->SetVoiceChannel(-1, sync_interface);
+ }
+ if (voice_sync_interface_) {
+ voice_sync_interface_->Release();
+ }
+ voice_engine_ = voice_engine;
+ voice_sync_interface_ = sync_interface;
+ return 0;
+}
+
+int ViEChannelManager::ConnectVoiceChannel(int channel_id,
+ int audio_channel_id) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ if (!voice_sync_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
+ "No VoE set");
+ return -1;
+ }
+ ViEChannel* channel = ViEChannelPtr(channel_id);
+ if (!channel) {
+ return -1;
+ }
+ return channel->SetVoiceChannel(audio_channel_id, voice_sync_interface_);
+}
+
+int ViEChannelManager::DisconnectVoiceChannel(int channel_id) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ ViEChannel* channel = ViEChannelPtr(channel_id);
+ if (channel) {
+ channel->SetVoiceChannel(-1, NULL);
+ return 0;
+ }
+ return -1;
+}
+
+VoiceEngine* ViEChannelManager::GetVoiceEngine() {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ return voice_engine_;
+}
+
+bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
+ bool receiver) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ ViEChannel* channel = ViEChannelPtr(channel_id);
+ if (!channel) {
+ return false;
+ }
+
+ ViEEncoder* encoder = ViEEncoderPtr(channel_id);
+
+ if (sender || receiver) {
+ if (!channel->EnableRemb(true)) {
+ return false;
+ }
+ } else {
+ channel->EnableRemb(false);
+ }
+ RtpRtcp* rtp_module = channel->rtp_rtcp();
+ if (sender) {
+ remb_->AddRembSender(rtp_module);
+ remb_->AddSendChannel(encoder->SendRtpRtcpModule());
+ } else {
+ remb_->RemoveRembSender(rtp_module);
+ remb_->RemoveSendChannel(encoder->SendRtpRtcpModule());
+ }
+ if (receiver) {
+ remb_->AddReceiveChannel(rtp_module);
+ } else {
+ remb_->RemoveReceiveChannel(rtp_module);
+ }
+ if (sender || receiver) {
+ rtp_module->SetRemoteBitrateObserver(remb_.get());
+ } else {
+ rtp_module->SetRemoteBitrateObserver(NULL);
+ }
+ return true;
+}
+
+ViEChannel* ViEChannelManager::ViEChannelPtr(int channel_id) const {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ ChannelMap::const_iterator it = channel_map_.find(channel_id);
+ if (it == channel_map_.end()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
+ "%s Channel doesn't exist: %d", __FUNCTION__, channel_id);
+ return NULL;
+ }
+ return it->second;
+}
+
+void ViEChannelManager::GetViEChannels(MapWrapper& channel_map) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ if (channel_map.Size() == 0) {
+ return;
+ }
+
+ // Add all items to 'channelMap'.
+ for (ChannelMap::iterator it = channel_map_.begin(); it != channel_map_.end();
+ ++it) {
+ channel_map.Insert(it->first, it->second);
+ }
+}
+
+ViEEncoder* ViEChannelManager::ViEEncoderPtr(int video_channel_id) const {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ EncoderMap::const_iterator it = vie_encoder_map_.find(video_channel_id);
+ if (it == vie_encoder_map_.end()) {
+ return NULL;
+ }
+ return it->second;
+}
+
+bool ViEChannelManager::GetFreeChannelId(int& free_channel_id) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ int idx = 0;
+ while (idx < free_channel_ids_size_) {
+ if (free_channel_ids_[idx] == true) {
+ // We've found a free id, allocate it and return.
+ free_channel_ids_[idx] = false;
+ free_channel_id = idx + kViEChannelIdBase;
+ return true;
+ }
+ idx++;
+ }
+ // No free channel id.
+ free_channel_id = -1;
+ return false;
+}
+
+void ViEChannelManager::ReturnChannelId(int channel_id) {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ assert(channel_id < kViEMaxNumberOfChannels + kViEChannelIdBase &&
+ channel_id >= kViEChannelIdBase);
+ free_channel_ids_[channel_id - kViEChannelIdBase] = true;
+}
+
+bool ViEChannelManager::ChannelUsingViEEncoder(int channel_id) const {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
+ if(orig_it == vie_encoder_map_.end()) {
+ // No ViEEncoder for this channel.
+ return false;
+ }
+
+ // Loop through all other channels to see if anyone points at the same
+ // ViEEncoder.
+ for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
+ comp_it != vie_encoder_map_.end(); ++comp_it) {
+ // Make sure we're not comparing the same channel with itself.
+ if (comp_it->first != channel_id) {
+ if (comp_it->second == orig_it->second) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+void ViEChannelManager::ChannelsUsingViEEncoder(int channel_id,
+ ChannelList* channels) const {
+ CriticalSectionScoped cs(*channel_id_critsect_);
+ EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
+
+ for (ChannelMap::const_iterator c_it = channel_map_.begin();
+ c_it != channel_map_.end(); ++c_it) {
+ EncoderMap::const_iterator comp_it = vie_encoder_map_.find(c_it->first);
+ assert(comp_it != vie_encoder_map_.end());
+ if (comp_it->second == orig_it->second) {
+ channels->push_back(c_it->second);
+ }
+ }
+}
+
+ViEChannelManagerScoped::ViEChannelManagerScoped(
+ const ViEChannelManager& vie_channel_manager)
+ : ViEManagerScopedBase(vie_channel_manager) {
+}
+
+ViEChannel* ViEChannelManagerScoped::Channel(int vie_channel_id) const {
+ return static_cast<const ViEChannelManager*>(vie_manager_)->ViEChannelPtr(
+ vie_channel_id);
+}
+ViEEncoder* ViEChannelManagerScoped::Encoder(int vie_channel_id) const {
+ return static_cast<const ViEChannelManager*>(vie_manager_)->ViEEncoderPtr(
+ vie_channel_id);
+}
+
+bool ViEChannelManagerScoped::ChannelUsingViEEncoder(int channel_id) const {
+ return (static_cast<const ViEChannelManager*>(vie_manager_))->
+ ChannelUsingViEEncoder(channel_id);
+}
+
+void ViEChannelManagerScoped::ChannelsUsingViEEncoder(
+ int channel_id, ChannelList* channels) const {
+ (static_cast<const ViEChannelManager*>(vie_manager_))->
+ ChannelsUsingViEEncoder(channel_id, channels);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_channel_manager.h b/trunk/src/video_engine/vie_channel_manager.h
new file mode 100644
index 0000000..f0fa348
--- /dev/null
+++ b/trunk/src/video_engine/vie_channel_manager.h
@@ -0,0 +1,131 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
+
+#include <list>
+#include <map>
+
+#include "engine_configurations.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_manager_base.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class MapWrapper;
+class ProcessThread;
+class ViEChannel;
+class ViEEncoder;
+class ViEPerformanceMonitor;
+class VieRemb;
+class VoEVideoSync;
+class VoiceEngine;
+
+typedef std::list<ViEChannel*> ChannelList;
+typedef std::map<int, ViEChannel*> ChannelMap;
+typedef std::map<int, ViEEncoder*> EncoderMap;
+
+class ViEChannelManager: private ViEManagerBase {
+ friend class ViEChannelManagerScoped;
+ public:
+ ViEChannelManager(int engine_id,
+ int number_of_cores,
+ ViEPerformanceMonitor& vie_performance_monitor);
+ ~ViEChannelManager();
+
+ void SetModuleProcessThread(ProcessThread& module_process_thread);
+
+ // Creates a new channel. 'channelId' will be the id of the created channel.
+ int CreateChannel(int& channel_id);
+
+ // Creates a channel and attaches to an already existing ViEEncoder.
+ int CreateChannel(int& channel_id, int original_channel);
+
+ // Deletes a channel.
+ int DeleteChannel(int channel_id);
+
+ // Set the voice engine instance to be used by all video channels.
+ int SetVoiceEngine(VoiceEngine* voice_engine);
+
+ // Enables lip sync of the channel.
+ int ConnectVoiceChannel(int channel_id, int audio_channel_id);
+
+ // Disables lip sync of the channel.
+ int DisconnectVoiceChannel(int channel_id);
+
+ VoiceEngine* GetVoiceEngine();
+
+ // Adds a channel to include when sending REMB.
+ bool SetRembStatus(int channel_id, bool sender, bool receiver);
+
+ private:
+ // Used by ViEChannelScoped, forcing a manager user to use scoped.
+ // Returns a pointer to the channel with id 'channelId'.
+ ViEChannel* ViEChannelPtr(int channel_id) const;
+
+ // Adds all channels to channel_map.
+ void GetViEChannels(MapWrapper& channel_map);
+
+ // Methods used by ViECaptureScoped and ViEEncoderScoped.
+ // Gets the ViEEncoder used as input for video_channel_id
+ ViEEncoder* ViEEncoderPtr(int video_channel_id) const;
+
+ // Returns true if we found a new channel id, free_channel_id, false
+ // otherwise.
+ bool GetFreeChannelId(int& free_channel_id);
+
+ // Returns a previously allocated channel id.
+ void ReturnChannelId(int channel_id);
+
+ // Returns true if at least one other channels uses the same ViEEncoder as
+ // channel_id.
+ bool ChannelUsingViEEncoder(int channel_id) const;
+ void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
+
+ // Protects channel_map_ and free_channel_ids_.
+ CriticalSectionWrapper* channel_id_critsect_;
+ int engine_id_;
+ int number_of_cores_;
+ ViEPerformanceMonitor& vie_performance_monitor_;
+ ChannelMap channel_map_;
+ bool* free_channel_ids_;
+ int free_channel_ids_size_;
+
+ // Maps Channel id -> ViEEncoder.
+ EncoderMap vie_encoder_map_;
+ VoEVideoSync* voice_sync_interface_;
+ scoped_ptr<VieRemb> remb_;
+ VoiceEngine* voice_engine_;
+ ProcessThread* module_process_thread_;
+};
+
+class ViEChannelManagerScoped: private ViEManagerScopedBase {
+ public:
+ explicit ViEChannelManagerScoped(
+ const ViEChannelManager& vie_channel_manager);
+ ViEChannel* Channel(int vie_channel_id) const;
+ ViEEncoder* Encoder(int vie_channel_id) const;
+
+ // Returns true if at least one other channels uses the same ViEEncoder as
+ // channel_id.
+ bool ChannelUsingViEEncoder(int channel_id) const;
+
+ // Returns a list with pointers to all channels using the same encoder as the
+ // channel with |channel_id|, including the one with the specified id.
+ void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_codec_impl.cc b/trunk/src/video_engine/vie_codec_impl.cc
new file mode 100644
index 0000000..f995117
--- /dev/null
+++ b/trunk/src/video_engine/vie_codec_impl.cc
@@ -0,0 +1,748 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_codec_impl.h"
+
+#include "engine_configurations.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_input_manager.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViECodec* ViECodec::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViECodecImpl* vie_codec_impl = vie_impl;
+ // Increase ref count.
+ (*vie_codec_impl)++;
+ return vie_codec_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViECodecImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViECodecImpl::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViECodec released too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViECodec reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViECodecImpl::ViECodecImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViECodecImpl::ViECodecImpl() Ctor");
+}
+
+ViECodecImpl::~ViECodecImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViECodecImpl::~ViECodecImpl() Dtor");
+}
+
+int ViECodecImpl::NumberOfCodecs() const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+ // +2 because of FEC(RED and ULPFEC)
+ return static_cast<int>((VideoCodingModule::NumberOfCodecs() + 2));
+}
+
+int ViECodecImpl::GetCodec(const unsigned char list_number,
+ VideoCodec& video_codec) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(list_number: %d, codec_type: %d)", __FUNCTION__,
+ list_number, video_codec.codecType);
+ if (list_number == VideoCodingModule::NumberOfCodecs()) {
+ memset(&video_codec, 0, sizeof(VideoCodec));
+ strcpy(video_codec.plName, "red");
+ video_codec.codecType = kVideoCodecRED;
+ video_codec.plType = VCM_RED_PAYLOAD_TYPE;
+ } else if (list_number == VideoCodingModule::NumberOfCodecs() + 1) {
+ memset(&video_codec, 0, sizeof(VideoCodec));
+ strcpy(video_codec.plName, "ulpfec");
+ video_codec.codecType = kVideoCodecULPFEC;
+ video_codec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
+ } else if (VideoCodingModule::Codec(list_number, &video_codec) != VCM_OK) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Could not get codec for list_number: %u", __FUNCTION__,
+ list_number);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::SetSendCodec(const int video_channel,
+ const VideoCodec& video_codec) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(),video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel, video_codec.codecType);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: codec: %d, pl_type: %d, width: %d, height: %d, bitrate: %d"
+ "maxBr: %d, min_br: %d, frame_rate: %d, qpMax: %u,"
+ "numberOfSimulcastStreams: %u )", __FUNCTION__,
+ video_codec.codecType, video_codec.plType, video_codec.width,
+ video_codec.height, video_codec.startBitrate,
+ video_codec.maxBitrate, video_codec.minBitrate,
+ video_codec.maxFramerate, video_codec.qpMax,
+ video_codec.numberOfSimulcastStreams);
+ if (video_codec.codecType == kVideoCodecVP8) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "pictureLossIndicationOn: %d, feedbackModeOn: %d, "
+ "complexity: %d, resilience: %d, numberOfTemporalLayers: %u",
+ video_codec.codecSpecific.VP8.pictureLossIndicationOn,
+ video_codec.codecSpecific.VP8.feedbackModeOn,
+ video_codec.codecSpecific.VP8.complexity,
+ video_codec.codecSpecific.VP8.resilience,
+ video_codec.codecSpecific.VP8.numberOfTemporalLayers);
+ }
+ if (!CodecValid(video_codec)) {
+ // Error logged.
+ shared_data_->SetLastError(kViECodecInvalidCodec);
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ // Set a max_bitrate if the user hasn't set one.
+ VideoCodec video_codec_internal;
+ memcpy(&video_codec_internal, &video_codec, sizeof(VideoCodec));
+ if (video_codec_internal.maxBitrate == 0) {
+ // Max is one bit per pixel.
+ video_codec_internal.maxBitrate = (video_codec_internal.width *
+ video_codec_internal.height *
+ video_codec_internal.maxFramerate)
+ / 1000;
+ if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
+ // Don't limit the set start bitrate.
+ video_codec_internal.maxBitrate = video_codec_internal.startBitrate;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: New max bitrate set to %d kbps", __FUNCTION__,
+ video_codec_internal.maxBitrate);
+ }
+
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ assert(false);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No encoder found for channel %d", __FUNCTION__);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ VideoCodec encoder;
+ vie_encoder->GetEncoder(encoder);
+ if (encoder.codecType != video_codec_internal.codecType &&
+ cs.ChannelUsingViEEncoder(video_channel)) {
+ // We don't allow changing codec type when several channels share encoder.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Settings differs from other channels using encoder",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViECodecInUse);
+ return -1;
+ }
+ // Make sure to generate a new SSRC if the codec type and/or resolution has
+ // changed. This won't have any effect if the user has set an SSRC.
+ bool new_rtp_stream = false;
+ if (encoder.codecType != video_codec_internal.codecType ||
+ encoder.width != video_codec_internal.width ||
+ encoder.height != video_codec_internal.height) {
+ new_rtp_stream = true;
+ }
+ if (video_codec_internal.numberOfSimulcastStreams > 1) {
+ if (cs.ChannelUsingViEEncoder(video_channel)) {
+ // We don't allow simulcast channels to share encoder.
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Can't share simulcast encoder",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViECodecInUse);
+ return -1;
+ }
+ }
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* frame_provider = NULL;
+
+ // Stop the media flow while reconfiguring.
+ vie_encoder->Pause();
+
+ // Check if we have a frame provider that is a camera and can provide this
+ // codec for us.
+ bool use_capture_device_as_encoder = false;
+ frame_provider = is.FrameProvider(vie_encoder);
+ if (frame_provider) {
+ if (frame_provider->Id() >= kViECaptureIdBase &&
+ frame_provider->Id() <= kViECaptureIdMax) {
+ ViECapturer* vie_capture = static_cast<ViECapturer*>(frame_provider);
+ // Try to get preencoded. Nothing to do if it is not supported.
+ if (vie_capture && vie_capture->PreEncodeToViEEncoder(
+ video_codec_internal,
+ *vie_encoder,
+ video_channel) == 0) {
+ use_capture_device_as_encoder = true;
+ }
+ }
+ }
+
+ // Update the encoder settings if we are not using a capture device capable
+ // of this codec.
+ if (!use_capture_device_as_encoder &&
+ vie_encoder->SetEncoder(video_codec_internal) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not change encoder for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+
+ // Give the channel(s) the new information.
+ ChannelList channels;
+ cs.ChannelsUsingViEEncoder(video_channel, &channels);
+ for (ChannelList::iterator it = channels.begin(); it != channels.end();
+ ++it) {
+ bool ret = true;
+ if ((*it)->SetSendCodec(video_codec_internal, new_rtp_stream) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not set send codec for channel %d", __FUNCTION__,
+ video_channel);
+ ret = false;
+ }
+ if (!ret) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ }
+
+ // Update the protection mode, we might be switching NACK/FEC.
+ vie_encoder->UpdateProtectionMethod();
+
+ // Get new best format for frame provider.
+ if (frame_provider) {
+ frame_provider->FrameCallbackChanged();
+ }
+ // Restart the media flow
+ if (new_rtp_stream) {
+ // Stream settings changed, make sure we get a key frame.
+ vie_encoder->SendKeyFrame();
+ }
+ vie_encoder->Restart();
+ return 0;
+}
+
+int ViECodecImpl::GetSendCodec(const int video_channel,
+ VideoCodec& video_codec) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No encoder for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ return vie_encoder->GetEncoder(video_codec);
+}
+
+int ViECodecImpl::SetReceiveCodec(const int video_channel,
+ const VideoCodec& video_codec) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel, video_codec.codecType);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: codec: %d, pl_type: %d, width: %d, height: %d, bitrate: %d,"
+ "maxBr: %d, min_br: %d, frame_rate: %d", __FUNCTION__,
+ video_codec.codecType, video_codec.plType, video_codec.width,
+ video_codec.height, video_codec.startBitrate,
+ video_codec.maxBitrate, video_codec.minBitrate,
+ video_codec.maxFramerate);
+
+ if (CodecValid(video_codec) == false) {
+ // Error logged.
+ shared_data_->SetLastError(kViECodecInvalidCodec);
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->SetReceiveCodec(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not set receive codec for channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::GetReceiveCodec(const int video_channel,
+ VideoCodec& video_codec) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel, video_codec.codecType);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->GetReceiveCodec(video_codec) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::GetCodecConfigParameters(
+ const int video_channel,
+ unsigned char config_parameters[kConfigParameterSize],
+ unsigned char& config_parameters_size) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No encoder for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_encoder->GetCodecConfigParameters(config_parameters,
+ config_parameters_size) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::SetImageScaleStatus(const int video_channel,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, enable: %d)", __FUNCTION__, video_channel,
+ enable);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_encoder->ScaleInputImage(enable) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::GetSendCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No send codec for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_encoder->SendCodecStatistics(key_frames, delta_frames) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::GetReceiveCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->ReceiveCodecStatistics(key_frames, delta_frames) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::GetCodecTargetBitrate(const int video_channel,
+ unsigned int* bitrate) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No send codec for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ return vie_encoder->CodecTargetBitrate(static_cast<WebRtc_UWord32*>(bitrate));
+}
+
+unsigned int ViECodecImpl::GetDiscardedPackets(const int video_channel) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
+ video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ return vie_channel->DiscardedPackets();
+}
+
+int ViECodecImpl::SetKeyFrameRequestCallbackStatus(const int video_channel,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->EnableKeyFrameRequestCallback(enable) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::SetSignalKeyPacketLossStatus(const int video_channel,
+ const bool enable,
+ const bool only_key_frames) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d, enable: %d, only_key_frames: %d)",
+ __FUNCTION__, video_channel, enable);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetSignalPacketLossStatus(enable, only_key_frames) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::RegisterEncoderObserver(const int video_channel,
+ ViEEncoderObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No encoder for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_encoder->RegisterCodecObserver(&observer) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not register codec observer at channel",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::DeregisterEncoderObserver(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No encoder for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_encoder->RegisterCodecObserver(NULL) != 0) {
+ shared_data_->SetLastError(kViECodecObserverNotRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::RegisterDecoderObserver(const int video_channel,
+ ViEDecoderObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterCodecObserver(&observer) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not register codec observer at channel",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::DeregisterDecoderObserver(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id()), "%s",
+ __FUNCTION__);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterCodecObserver(NULL) != 0) {
+ shared_data_->SetLastError(kViECodecObserverNotRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::SendKeyFrame(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_encoder->SendKeyFrame() != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViECodecImpl::WaitForFirstKeyFrame(const int video_channel,
+ const bool wait) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d, wait: %d)", __FUNCTION__, video_channel,
+ wait);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->WaitForKeyFrame(wait) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
+ // Check pl_name matches codec_type.
+ if (video_codec.codecType == kVideoCodecRED) {
+#if defined(WIN32)
+ if (_strnicmp(video_codec.plName, "red", 3) == 0) {
+#else
+ if (strncasecmp(video_codec.plName, "red", 3) == 0) {
+#endif
+ // We only care about the type and name for red.
+ return true;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Codec type doesn't match pl_name", video_codec.plType);
+ return false;
+ } else if (video_codec.codecType == kVideoCodecULPFEC) {
+#if defined(WIN32)
+ if (_strnicmp(video_codec.plName, "ULPFEC", 6) == 0) {
+#else
+ if (strncasecmp(video_codec.plName, "ULPFEC", 6) == 0) {
+#endif
+ // We only care about the type and name for ULPFEC.
+ return true;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Codec type doesn't match pl_name", video_codec.plType);
+ return false;
+ } else if ((video_codec.codecType == kVideoCodecVP8 &&
+ strncmp(video_codec.plName, "VP8", 4) == 0) ||
+ (video_codec.codecType == kVideoCodecI420 &&
+ strncmp(video_codec.plName, "I420", 4) == 0)) {
+ // OK.
+ } else {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Codec type doesn't match pl_name", video_codec.plType);
+ return false;
+ }
+
+ if (video_codec.plType == 0 && video_codec.plType > 127) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Invalid codec payload type: %d", video_codec.plType);
+ return false;
+ }
+
+ if (video_codec.width > kViEMaxCodecWidth ||
+ video_codec.height > kViEMaxCodecHeight) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid codec size: %u x %u",
+ video_codec.width, video_codec.height);
+ return false;
+ }
+
+ if (video_codec.startBitrate < kViEMinCodecBitrate) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid start_bitrate: %u",
+ video_codec.startBitrate);
+ return false;
+ }
+ if (video_codec.minBitrate < kViEMinCodecBitrate) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid min_bitrate: %u",
+ video_codec.minBitrate);
+ return false;
+ }
+ if (video_codec.numberOfSimulcastStreams == 1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
+ "Number of Simulcast streams can not be 1");
+ return false;
+ }
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_codec_impl.h b/trunk/src/video_engine/vie_codec_impl.h
new file mode 100644
index 0000000..e0e5e94
--- /dev/null
+++ b/trunk/src/video_engine/vie_codec_impl.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
+
+#include "typedefs.h"
+#include "video_engine/include/vie_codec.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViECodecImpl
+ : public ViECodec,
+ public ViERefCount {
+ public:
+ virtual int Release();
+
+ // Implements ViECodec.
+ virtual int NumberOfCodecs() const;
+ virtual int GetCodec(const unsigned char list_number,
+ VideoCodec& video_codec) const;
+ virtual int SetSendCodec(const int video_channel,
+ const VideoCodec& video_codec);
+ virtual int GetSendCodec(const int video_channel,
+ VideoCodec& video_codec) const;
+ virtual int SetReceiveCodec(const int video_channel,
+ const VideoCodec& video_codec);
+ virtual int GetReceiveCodec(const int video_channel,
+ VideoCodec& video_codec) const;
+ virtual int GetCodecConfigParameters(
+ const int video_channel,
+ unsigned char config_parameters[kConfigParameterSize],
+ unsigned char& config_parameters_size) const;
+ virtual int SetImageScaleStatus(const int video_channel, const bool enable);
+ virtual int GetSendCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const;
+ virtual int GetReceiveCodecStastistics(const int video_channel,
+ unsigned int& key_frames,
+ unsigned int& delta_frames) const;
+ virtual int GetCodecTargetBitrate(const int video_channel,
+ unsigned int* bitrate) const;
+ virtual unsigned int GetDiscardedPackets(const int video_channel) const;
+ virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
+ const bool enable);
+ virtual int SetSignalKeyPacketLossStatus(const int video_channel,
+ const bool enable,
+ const bool only_key_frames = false);
+ virtual int RegisterEncoderObserver(const int video_channel,
+ ViEEncoderObserver& observer);
+ virtual int DeregisterEncoderObserver(const int video_channel);
+ virtual int RegisterDecoderObserver(const int video_channel,
+ ViEDecoderObserver& observer);
+ virtual int DeregisterDecoderObserver(const int video_channel);
+ virtual int SendKeyFrame(const int video_channel);
+ virtual int WaitForFirstKeyFrame(const int video_channel, const bool wait);
+
+ protected:
+ ViECodecImpl(ViESharedData* shared_data);
+ virtual ~ViECodecImpl();
+
+ private:
+ bool CodecValid(const VideoCodec& video_codec);
+
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
diff --git a/trunk/src/video_engine/vie_defines.h b/trunk/src/video_engine/vie_defines.h
new file mode 100644
index 0000000..73c9008
--- /dev/null
+++ b/trunk/src/video_engine/vie_defines.h
@@ -0,0 +1,261 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * vie_defines.h
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
+
+#include "engine_configurations.h"
+
+#ifdef WEBRTC_MAC_INTEL
+#include <stdio.h>
+#include <unistd.h>
+#endif
+
+#ifdef WEBRTC_ANDROID
+#include <pthread.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <linux/net.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <sys/time.h>
+#endif
+
+namespace webrtc
+{
+// ===================================================
+// ViE Defines
+// ===================================================
+
+// General
+enum { kViEMinKeyRequestIntervalMs = 300};
+
+// ViEBase
+enum { kViEMaxNumberOfChannels = 32};
+enum { kViEVersionMaxMessageSize = 1024 };
+enum { kViEMaxModuleVersionSize = 960 };
+
+// ViECapture
+enum { kViEMaxCaptureDevices=10};
+// Width used if no send codec has been set when a capture device is started
+enum { kViECaptureDefaultWidth = 352};
+// Height used if no send codec has been set when a capture device is started
+enum { kViECaptureDefaultHeight = 288};
+enum { kViECaptureDefaultFramerate = 30};
+enum { kViECaptureMaxSnapshotWaitTimeMs = 500 };
+
+// ViECodec
+enum { kViEMaxCodecWidth = 1920};
+enum { kViEMaxCodecHeight = 1200};
+enum { kViEMaxCodecFramerate = 60};
+enum { kViEMinCodecBitrate = 30};
+
+// ViEEncryption
+enum { kViEMaxSrtpKeyLength = 30};
+enum { kViEMinSrtpEncryptLength = 16};
+enum { kViEMaxSrtpEncryptLength = 256};
+enum { kViEMaxSrtpAuthSh1Length = 20};
+enum { kViEMaxSrtpTagAuthNullLength = 12};
+enum { kViEMaxSrtpKeyAuthNullLength = 256};
+
+// ViEExternalCodec
+
+// ViEFile
+enum { kViEMaxFilePlayers = 3};
+
+// ViEImageProcess
+
+// ViENetwork
+enum { kViEMaxMtu = 1500};
+enum { kViESocketThreads = 1};
+enum { kViENumReceiveSocketBuffers = 500};
+
+// ViERender
+// Max valid time set in SetRenderTimeoutImage
+enum { kViEMaxRenderTimeoutTimeMs = 10000};
+// Min valid time set in SetRenderTimeoutImage
+enum { kViEMinRenderTimeoutTimeMs = 33};
+enum { kViEDefaultRenderDelayMs = 10};
+
+// ViERTP_RTCP
+enum { kNackHistorySize = 400};
+
+// Id definitions
+enum {
+ kViEChannelIdBase=0x0,
+ kViEChannelIdMax=0xFF,
+ kViECaptureIdBase=0x1001,
+ kViECaptureIdMax=0x10FF,
+ kViEFileIdBase=0x2000,
+ kViEFileIdMax=0x200F,
+ kViEDummyChannelId=0xFFFF
+};
+
+// Module id
+// Create a unique id based on the ViE instance id and the
+// channel id. ViE id > 0 and 0 <= channel id <= 255
+
+inline int ViEId(const int vieId, const int channelId = -1)
+{
+ if (channelId == -1)
+ {
+ return (int) ((vieId << 16) + kViEDummyChannelId);
+ }
+ return (int) ((vieId << 16) + channelId);
+}
+
+inline int ViEModuleId(const int vieId, const int channelId = -1)
+{
+ if (channelId == -1)
+ {
+ return (int) ((vieId << 16) + kViEDummyChannelId);
+ }
+ return (int) ((vieId << 16) + channelId);
+}
+
+inline int ChannelId(const int moduleId)
+{
+ return (int) (moduleId & 0xffff);
+}
+
+
+// ============================================================================
+// Platform specifics
+// ============================================================================
+
+//-------------------------------------
+// Windows
+//-------------------------------------
+
+#if defined(_WIN32)
+// Build information macros
+ #if defined(_DEBUG)
+ #define BUILDMODE TEXT("d")
+ #elif defined(DEBUG)
+ #define BUILDMODE TEXT("d")
+ #elif defined(NDEBUG)
+ #define BUILDMODE TEXT("r")
+ #else
+ #define BUILDMODE TEXT("?")
+ #endif
+
+ #define BUILDTIME TEXT(__TIME__)
+ #define BUILDDATE TEXT(__DATE__)
+
+ // example: "Oct 10 2002 12:05:30 r"
+ #define BUILDINFO BUILDDATE TEXT(" ") BUILDTIME TEXT(" ") BUILDMODE
+
+
+ #define RENDER_MODULE_TYPE kRenderWindows
+ // Warning pragmas
+ // new behavior: elements of array 'XXX' will be default initialized
+ #pragma warning(disable: 4351)
+ // 'this' : used in base member initializer list
+ #pragma warning(disable: 4355)
+ // frame pointer register 'ebp' modified by inline assembly code
+ #pragma warning(disable: 4731)
+
+ // Include libraries
+ #pragma comment( lib, "winmm.lib" )
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ #pragma comment( lib, "ws2_32.lib" )
+ #pragma comment( lib, "Iphlpapi.lib" ) // _GetAdaptersAddresses
+#endif
+#endif
+
+
+//-------------------------------------
+// Mac
+//-------------------------------------
+
+#ifdef WEBRTC_MAC_INTEL
+ #define SLEEP(x) usleep(x * 1000)
+
+ // Build information macros
+ #define TEXT(x) x
+ #if defined(_DEBUG)
+ #define BUILDMODE TEXT("d")
+ #elif defined(DEBUG)
+ #define BUILDMODE TEXT("d")
+ #elif defined(NDEBUG)
+ #define BUILDMODE TEXT("r")
+ #else
+ #define BUILDMODE TEXT("?")
+ #endif
+
+ #define BUILDTIME TEXT(__TIME__)
+ #define BUILDDATE TEXT(__DATE__)
+
+ // example: "Oct 10 2002 12:05:30 r"
+ #define BUILDINFO BUILDDATE TEXT(" ") BUILDTIME TEXT(" ") BUILDMODE
+
+ #define RENDER_MODULE_TYPE kRenderWindows
+#endif
+
+//-------------------------------------
+// Linux
+//-------------------------------------
+
+#ifndef WEBRTC_ANDROID
+#ifdef WEBRTC_LINUX
+
+// Build information macros
+#if defined(_DEBUG)
+ #define BUILDMODE "d"
+#elif defined(DEBUG)
+ #define BUILDMODE "d"
+#elif defined(NDEBUG)
+ #define BUILDMODE "r"
+#else
+ #define BUILDMODE "?"
+#endif
+
+#define BUILDTIME __TIME__
+#define BUILDDATE __DATE__
+
+// example: "Oct 10 2002 12:05:30 r"
+#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
+
+#endif // ifdef WEBRTC_LINUX
+#endif // ifndef WEBRTC_ANDROID
+
+#ifdef WEBRTC_ANDROID
+
+ #define FAR
+ #define __cdecl
+
+ #if defined(_DEBUG)
+ #define BUILDMODE "d"
+ #elif defined(DEBUG)
+ #define BUILDMODE "d"
+ #elif defined(NDEBUG)
+ #define BUILDMODE "r"
+ #else
+ #define BUILDMODE "?"
+ #endif
+
+ #define BUILDTIME __TIME__
+ #define BUILDDATE __DATE__
+
+ // example: "Oct 10 2002 12:05:30 r"
+ #define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
+
+#endif // #ifdef WEBRTC_ANDROID
+
+} //namespace webrtc
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
diff --git a/trunk/src/video_engine/vie_encoder.cc b/trunk/src/video_engine/vie_encoder.cc
new file mode 100644
index 0000000..cf3587b
--- /dev/null
+++ b/trunk/src/video_engine/vie_encoder.cc
@@ -0,0 +1,887 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_encoder.h"
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#include "process_thread.h"
+#include "rtp_rtcp.h"
+#include "tick_util.h"
+#include "trace.h"
+#include "video_codec_interface.h"
+#include "video_coding.h"
+#include "video_coding_defines.h"
+#include "vie_codec.h"
+#include "vie_defines.h"
+#include "vie_image_process.h"
+
+namespace webrtc {
+
+class QMVideoSettingsCallback : public VCMQMSettingsCallback {
+ public:
+ QMVideoSettingsCallback(WebRtc_Word32 engine_id,
+ WebRtc_Word32 channel_id,
+ VideoProcessingModule* vpm,
+ VideoCodingModule* vcm,
+ WebRtc_Word32 num_of_cores,
+ WebRtc_Word32 max_payload_length);
+ ~QMVideoSettingsCallback();
+
+ // Update VPM with QM (quality modes: frame size & frame rate) settings.
+ WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frame_rate,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height);
+
+ void SetMaxPayloadLength(WebRtc_Word32 max_payload_length);
+
+ private:
+ WebRtc_Word32 engine_id_;
+ WebRtc_Word32 channel_id_;
+ VideoProcessingModule* vpm_;
+ VideoCodingModule* vcm_;
+ WebRtc_Word32 num_cores_;
+ WebRtc_Word32 max_payload_length_;
+};
+
+
+ViEEncoder::ViEEncoder(WebRtc_Word32 engine_id, WebRtc_Word32 channel_id,
+ WebRtc_UWord32 number_of_cores,
+ ProcessThread& module_process_thread)
+ : engine_id_(engine_id),
+ channel_id_(channel_id),
+ number_of_cores_(number_of_cores),
+ vcm_(*webrtc::VideoCodingModule::Create(ViEModuleId(engine_id,
+ channel_id))),
+ vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
+ channel_id))),
+ default_rtp_rtcp_(*RtpRtcp::CreateRtpRtcp(
+ ViEModuleId(engine_id, channel_id), false)),
+ callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ paused_(false),
+ channels_dropping_delta_frames_(0),
+ drop_next_frame_(false),
+ fec_enabled_(false),
+ nack_enabled_(false),
+ codec_observer_(NULL),
+ effect_filter_(NULL),
+ module_process_thread_(module_process_thread),
+ has_received_sli_(false),
+ picture_id_sli_(0),
+ has_received_rpsi_(false),
+ picture_id_rpsi_(0),
+ file_recorder_(channel_id) {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo,
+ ViEId(engine_id, channel_id),
+ "%s(engine_id: %d) 0x%p - Constructor", __FUNCTION__, engine_id,
+ this);
+ for (int i = 0; i < kMaxSimulcastStreams; i++) {
+ time_last_intra_request_ms_[i] = 0;
+ }
+ vcm_.InitializeSender();
+ vpm_.EnableTemporalDecimation(true);
+
+ // Enable/disable content analysis: off by default for now.
+ vpm_.EnableContentAnalysis(false);
+
+ module_process_thread_.RegisterModule(&vcm_);
+ if (default_rtp_rtcp_.InitSender() != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViEEncoder: RTP::InitSender failure");
+ assert(false);
+ }
+ default_rtp_rtcp_.RegisterIncomingVideoCallback(this);
+ default_rtp_rtcp_.RegisterIncomingRTCPCallback(this);
+ module_process_thread_.RegisterModule(&default_rtp_rtcp_);
+
+ qm_callback_ = new QMVideoSettingsCallback(
+ engine_id_,
+ channel_id_,
+ &vpm_,
+ &vcm_,
+ number_of_cores,
+ default_rtp_rtcp_.MaxDataPayloadLength());
+
+#ifdef VIDEOCODEC_VP8
+ VideoCodec video_codec;
+ if (vcm_.Codec(webrtc::kVideoCodecVP8, &video_codec) == VCM_OK) {
+ vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+ default_rtp_rtcp_.MaxDataPayloadLength());
+ default_rtp_rtcp_.RegisterSendPayload(video_codec);
+ } else {
+ assert(false);
+ }
+#else
+ VideoCodec video_codec;
+ if (vcm_.Codec(webrtc::kVideoCodecI420, &video_codec) == VCM_OK) {
+ vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+ default_rtp_rtcp_.MaxDataPayloadLength());
+ default_rtp_rtcp_.RegisterSendPayload(video_codec);
+ } else {
+ assert(false);
+ }
+#endif
+
+ if (vcm_.RegisterTransportCallback(this) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViEEncoder: VCM::RegisterTransportCallback failure");
+ }
+ if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViEEncoder: VCM::RegisterSendStatisticsCallback failure");
+ }
+
+ if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "VCM::RegisterQMCallback failure");
+ }
+}
+
+ViEEncoder::~ViEEncoder() {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViEEncoder Destructor 0x%p, engine_id: %d", this, engine_id_);
+
+ if (default_rtp_rtcp_.NumberChildModules() > 0) {
+ assert(false);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Channels still attached %d, leaking memory",
+ default_rtp_rtcp_.NumberChildModules());
+ return;
+ }
+ module_process_thread_.DeRegisterModule(&vcm_);
+ module_process_thread_.DeRegisterModule(&vpm_);
+ module_process_thread_.DeRegisterModule(&default_rtp_rtcp_);
+ delete &vcm_;
+ delete &vpm_;
+ delete &default_rtp_rtcp_;
+ delete qm_callback_;
+}
+
+void ViEEncoder::Pause() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ CriticalSectionScoped cs(data_cs_.get());
+ paused_ = true;
+}
+
+void ViEEncoder::Restart() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s", __FUNCTION__);
+ CriticalSectionScoped cs(data_cs_.get());
+ paused_ = false;
+}
+
+WebRtc_Word32 ViEEncoder::DropDeltaAfterKey(bool enable) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s(%d)", __FUNCTION__, enable);
+ CriticalSectionScoped cs(data_cs_.get());
+
+ if (enable) {
+ channels_dropping_delta_frames_++;
+ } else {
+ channels_dropping_delta_frames_--;
+ if (channels_dropping_delta_frames_ < 0) {
+ channels_dropping_delta_frames_ = 0;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Called too many times", __FUNCTION__);
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_UWord8 ViEEncoder::NumberOfCodecs() {
+ return vcm_.NumberOfCodecs();
+}
+
+WebRtc_Word32 ViEEncoder::GetCodec(WebRtc_UWord8 list_index,
+ webrtc::VideoCodec& video_codec) {
+ if (vcm_.Codec(list_index, &video_codec) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: Could not get codec",
+ __FUNCTION__);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
+ WebRtc_UWord8 pl_type) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: pltype %u", __FUNCTION__,
+ pl_type);
+
+ if (encoder == NULL)
+ return -1;
+
+ if (vcm_.RegisterExternalEncoder(encoder, pl_type) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not register external encoder");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::DeRegisterExternalEncoder(WebRtc_UWord8 pl_type) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: pltype %u", __FUNCTION__, pl_type);
+
+ webrtc::VideoCodec current_send_codec;
+ if (vcm_.SendCodec(¤t_send_codec) == VCM_OK) {
+ if (vcm_.Bitrate(¤t_send_codec.startBitrate) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Failed to get the current encoder target bitrate.");
+ }
+ }
+
+ if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not deregister external encoder");
+ return -1;
+ }
+
+ // If the external encoder is the current send codeci, use vcm internal
+ // encoder.
+ if (current_send_codec.plType == pl_type) {
+ WebRtc_UWord16 max_data_payload_length =
+ default_rtp_rtcp_.MaxDataPayloadLength();
+ if (vcm_.RegisterSendCodec(¤t_send_codec, number_of_cores_,
+ max_data_payload_length) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not use internal encoder");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: CodecType: %d, width: %u, height: %u", __FUNCTION__,
+ video_codec.codecType, video_codec.width, video_codec.height);
+
+ // Convert from kbps to bps.
+ default_rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
+ video_codec.minBitrate,
+ video_codec.maxBitrate);
+
+ // Setting target width and height for VPM.
+ if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
+ video_codec.maxFramerate) != VPM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not set VPM target dimensions");
+ return -1;
+ }
+
+ if (default_rtp_rtcp_.RegisterSendPayload(video_codec) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could register RTP module video payload");
+ return -1;
+ }
+
+ WebRtc_UWord16 max_data_payload_length =
+ default_rtp_rtcp_.MaxDataPayloadLength();
+
+ qm_callback_->SetMaxPayloadLength(max_data_payload_length);
+
+ if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+ max_data_payload_length) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not register send codec");
+ return -1;
+ }
+
+ data_cs_->Enter();
+ memcpy(&send_codec_, &video_codec, sizeof(send_codec_));
+ data_cs_->Leave();
+
+ // Set this module as sending right away, let the slave module in the channel
+ // start and stop sending.
+ if (default_rtp_rtcp_.Sending() == false) {
+ if (default_rtp_rtcp_.SetSendingStatus(true) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could start RTP module sending");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::GetEncoder(webrtc::VideoCodec& video_codec) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ if (vcm_.SendCodec(&video_codec) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not get VCM send codec");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::GetCodecConfigParameters(
+ unsigned char config_parameters[kConfigParameterSize],
+ unsigned char& config_parameters_size) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ WebRtc_Word32 num_parameters =
+ vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
+ if (num_parameters <= 0) {
+ config_parameters_size = 0;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not get config parameters");
+ return -1;
+ }
+ config_parameters_size = static_cast<unsigned char>(num_parameters);
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::ScaleInputImage(bool enable) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s(enable %d)", __FUNCTION__,
+ enable);
+
+ VideoFrameResampling resampling_mode = kFastRescaling;
+ if (enable == true) {
+ // kInterpolation is currently not supported.
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s not supported",
+ __FUNCTION__, enable);
+ return -1;
+ }
+ vpm_.SetInputFrameResampleMode(resampling_mode);
+
+ return 0;
+}
+
+RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ return &default_rtp_rtcp_;
+}
+
+void ViEEncoder::DeliverFrame(int id, webrtc::VideoFrame& video_frame,
+ int num_csrcs,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: %llu", __FUNCTION__,
+ video_frame.TimeStamp());
+
+ {
+ CriticalSectionScoped cs(data_cs_.get());
+ if (paused_ || default_rtp_rtcp_.SendingMedia() == false) {
+ // We've paused or we have no channels attached, don't encode.
+ return;
+ }
+ if (drop_next_frame_) {
+ // Drop this frame.
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Dropping frame %llu after a key fame", __FUNCTION__,
+ video_frame.TimeStamp());
+ drop_next_frame_ = false;
+ return;
+ }
+ }
+
+ // Convert render time, in ms, to RTP timestamp.
+ const WebRtc_UWord32 time_stamp =
+ 90 * static_cast<WebRtc_UWord32>(video_frame.RenderTimeMs());
+ video_frame.SetTimeStamp(time_stamp);
+ {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (effect_filter_) {
+ effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
+ video_frame.TimeStamp(),
+ video_frame.Width(), video_frame.Height());
+ }
+ }
+ // Record raw frame.
+ file_recorder_.RecordVideoFrame(video_frame);
+
+ // Make sure the CSRC list is correct.
+ if (num_csrcs > 0) {
+ WebRtc_UWord32 tempCSRC[kRtpCsrcSize];
+ for (int i = 0; i < num_csrcs; i++) {
+ if (CSRC[i] == 1) {
+ tempCSRC[i] = default_rtp_rtcp_.SSRC();
+ } else {
+ tempCSRC[i] = CSRC[i];
+ }
+ }
+ default_rtp_rtcp_.SetCSRCs(tempCSRC, (WebRtc_UWord8) num_csrcs);
+ }
+
+#ifdef VIDEOCODEC_VP8
+ if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
+ webrtc::CodecSpecificInfo codec_specific_info;
+ codec_specific_info.codecType = webrtc::kVideoCodecVP8;
+ if (has_received_sli_ || has_received_rpsi_) {
+ {
+ codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
+ has_received_rpsi_;
+ codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
+ has_received_sli_;
+ codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
+ picture_id_rpsi_;
+ codec_specific_info.codecSpecific.VP8.pictureIdSLI =
+ picture_id_sli_;
+ }
+ has_received_sli_ = false;
+ has_received_rpsi_ = false;
+ }
+ VideoFrame* decimated_frame = NULL;
+ const int ret = vpm_.PreprocessFrame(&video_frame, &decimated_frame);
+ if (ret == 1) {
+ // Drop this frame.
+ return;
+ } else if (ret != VPM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Error preprocessing frame %u", __FUNCTION__,
+ video_frame.TimeStamp());
+ return;
+ }
+
+ VideoContentMetrics* content_metrics = NULL;
+ content_metrics = vpm_.ContentMetrics();
+
+ // Frame was not re-sampled => use original.
+ if (decimated_frame == NULL) {
+ decimated_frame = &video_frame;
+ }
+
+ if (vcm_.AddVideoFrame(*decimated_frame, content_metrics,
+ &codec_specific_info) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Error encoding frame %u", __FUNCTION__,
+ video_frame.TimeStamp());
+ }
+ return;
+ }
+#endif
+ // TODO(mflodman) Rewrite this to use code common to VP8 case.
+ // Pass frame via preprocessor.
+ VideoFrame* decimated_frame = NULL;
+ const int ret = vpm_.PreprocessFrame(&video_frame, &decimated_frame);
+ if (ret == 1) {
+ // Drop this frame.
+ return;
+ } else if (ret != VPM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Error preprocessing frame %u", __FUNCTION__,
+ video_frame.TimeStamp());
+ return;
+ }
+
+ // Frame was not sampled => use original.
+ if (decimated_frame == NULL) {
+ decimated_frame = &video_frame;
+ }
+ if (vcm_.AddVideoFrame(*decimated_frame) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: Error encoding frame %u",
+ __FUNCTION__, video_frame.TimeStamp());
+ }
+}
+
+void ViEEncoder::DelayChanged(int id, int frame_delay) {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: %u", __FUNCTION__,
+ frame_delay);
+
+ default_rtp_rtcp_.SetCameraDelay(frame_delay);
+ file_recorder_.SetFrameDelay(frame_delay);
+}
+
+int ViEEncoder::GetPreferedFrameSettings(int& width,
+ int& height,
+ int& frame_rate) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ webrtc::VideoCodec video_codec;
+ memset(&video_codec, 0, sizeof(video_codec));
+ if (vcm_.SendCodec(&video_codec) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Could not get VCM send codec");
+ return -1;
+ }
+
+ width = video_codec.width;
+ height = video_codec.height;
+ frame_rate = video_codec.maxFramerate;
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::SendKeyFrame() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+ return vcm_.FrameTypeRequest(kVideoFrameKey, 0); // Simulcast idx = 0.
+}
+
+WebRtc_Word32 ViEEncoder::SendCodecStatistics(
+ WebRtc_UWord32& num_key_frames, WebRtc_UWord32& num_delta_frames) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ webrtc::VCMFrameCount sent_frames;
+ if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Could not get sent frame information", __FUNCTION__);
+ return -1;
+ }
+ num_key_frames = sent_frames.numKeyFrames;
+ num_delta_frames = sent_frames.numDeltaFrames;
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::EstimatedSendBandwidth(
+ WebRtc_UWord32* available_bandwidth) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ return default_rtp_rtcp_.EstimatedSendBandwidth(available_bandwidth);
+}
+
+int ViEEncoder::CodecTargetBitrate(WebRtc_UWord32* bitrate) const {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+ __FUNCTION__);
+ if (vcm_.Bitrate(bitrate) != 0)
+ return -1;
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::UpdateProtectionMethod() {
+ bool fec_enabled = false;
+ WebRtc_UWord8 dummy_ptype_red = 0;
+ WebRtc_UWord8 dummy_ptypeFEC = 0;
+
+ // Updated protection method to VCM to get correct packetization sizes.
+ // FEC has larger overhead than NACK -> set FEC if used.
+ WebRtc_Word32 error = default_rtp_rtcp_.GenericFECStatus(fec_enabled,
+ dummy_ptype_red,
+ dummy_ptypeFEC);
+ if (error) {
+ return -1;
+ }
+
+ bool nack_enabled = (default_rtp_rtcp_.NACK() == kNackOff) ? false : true;
+ if (fec_enabled_ == fec_enabled && nack_enabled_ == nack_enabled) {
+ // No change needed, we're already in correct state.
+ return 0;
+ }
+ fec_enabled_ = fec_enabled;
+ nack_enabled_ = nack_enabled;
+
+ // Set Video Protection for VCM.
+ if (fec_enabled && nack_enabled) {
+ vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
+ } else {
+ vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
+ vcm_.SetVideoProtection(webrtc::kProtectionNack, nack_enabled_);
+ vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
+ }
+
+ if (fec_enabled || nack_enabled) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: FEC status ",
+ __FUNCTION__, fec_enabled);
+ vcm_.RegisterProtectionCallback(this);
+ // The send codec must be registered to set correct MTU.
+ webrtc::VideoCodec codec;
+ if (vcm_.SendCodec(&codec) == 0) {
+ WebRtc_UWord16 max_pay_load = default_rtp_rtcp_.MaxDataPayloadLength();
+ if (vcm_.Bitrate(&codec.startBitrate) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Failed to get the current encoder target bitrate.");
+ }
+ if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Failed to update Sendcodec when enabling FEC",
+ __FUNCTION__, fec_enabled);
+ return -1;
+ }
+ }
+ return 0;
+ } else {
+ // FEC and NACK are disabled.
+ vcm_.RegisterProtectionCallback(NULL);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::SendData(
+ const FrameType frame_type,
+ const WebRtc_UWord8 payload_type,
+ const WebRtc_UWord32 time_stamp,
+ const WebRtc_UWord8* payload_data,
+ const WebRtc_UWord32 payload_size,
+ const webrtc::RTPFragmentationHeader& fragmentation_header,
+ const RTPVideoHeader* rtp_video_hdr) {
+ {
+ CriticalSectionScoped cs(data_cs_.get());
+ if (paused_) {
+ // Paused, don't send this packet.
+ return 0;
+ }
+ if (channels_dropping_delta_frames_ &&
+ frame_type == webrtc::kVideoFrameKey) {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Sending key frame, drop next frame", __FUNCTION__);
+ drop_next_frame_ = true;
+ }
+ }
+
+ // New encoded data, hand over to the rtp module.
+ return default_rtp_rtcp_.SendOutgoingData(frame_type, payload_type,
+ time_stamp, payload_data,
+ payload_size, &fragmentation_header,
+ rtp_video_hdr);
+}
+
+WebRtc_Word32 ViEEncoder::ProtectionRequest(
+ WebRtc_UWord8 delta_fecrate,
+ WebRtc_UWord8 key_fecrate,
+ bool delta_use_uep_protection,
+ bool key_use_uep_protection,
+ bool nack_enabled,
+ WebRtc_UWord32* sent_video_rate_bps,
+ WebRtc_UWord32* sent_nack_rate_bps,
+ WebRtc_UWord32* sent_fec_rate_bps) {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s, deltaFECRate: %u, key_fecrate: %u, "
+ "delta_use_uep_protection: %d, key_use_uep_protection: %d, ",
+ __FUNCTION__, delta_fecrate, key_fecrate,
+ delta_use_uep_protection, key_use_uep_protection);
+
+ if (default_rtp_rtcp_.SetFECCodeRate(key_fecrate, delta_fecrate) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Could not update FEC code rate", __FUNCTION__);
+ }
+ if (default_rtp_rtcp_.SetFECUepProtection(key_use_uep_protection,
+ delta_use_uep_protection) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Could not update FEC-UEP protection", __FUNCTION__);
+ }
+ default_rtp_rtcp_.BitrateSent(NULL,
+ sent_video_rate_bps,
+ sent_fec_rate_bps,
+ sent_nack_rate_bps);
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::SendStatistics(const WebRtc_UWord32 bit_rate,
+ const WebRtc_UWord32 frame_rate) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (codec_observer_) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: bitrate %u, framerate %u",
+ __FUNCTION__, bit_rate, frame_rate);
+ codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (observer) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: observer added",
+ __FUNCTION__);
+ if (codec_observer_) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: observer already set.",
+ __FUNCTION__);
+ return -1;
+ }
+ codec_observer_ = observer;
+ } else {
+ if (codec_observer_ == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: observer does not exist.", __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: observer removed",
+ __FUNCTION__);
+ codec_observer_ = NULL;
+ }
+ return 0;
+}
+
+void ViEEncoder::OnSLIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 picture_id) {
+ picture_id_sli_ = picture_id;
+ has_received_sli_ = true;
+}
+
+void ViEEncoder::OnRPSIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord64 picture_id) {
+ picture_id_rpsi_ = picture_id;
+ has_received_rpsi_ = true;
+}
+
+void ViEEncoder::OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+ const FrameType type,
+ const WebRtc_UWord8 stream_idx) {
+ assert(stream_idx < kMaxSimulcastStreams);
+
+ // Key frame request from remote side, signal to VCM.
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
+
+ WebRtc_Word64 now = TickTime::MillisecondTimestamp();
+ if (time_last_intra_request_ms_[stream_idx] + kViEMinKeyRequestIntervalMs >
+ now) {
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: Not not encoding new intra due to timing", __FUNCTION__);
+ return;
+ }
+ vcm_.FrameTypeRequest(type, stream_idx);
+ time_last_intra_request_ms_[stream_idx] = now;
+}
+
+void ViEEncoder::OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 bitrate_bps,
+ const WebRtc_UWord8 fraction_lost,
+ const WebRtc_UWord16 round_trip_time_ms) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s(bitrate_bps: %u, fraction_lost: %u, rtt_ms: %u",
+ __FUNCTION__, bitrate_bps, fraction_lost, round_trip_time_ms);
+
+ vcm_.SetChannelParameters(bitrate_bps / 1000, fraction_lost,
+ round_trip_time_ms);
+}
+
+WebRtc_Word32 ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
+ CriticalSectionScoped cs(callback_cs_.get());
+ if (effect_filter == NULL) {
+ if (effect_filter_ == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: no effect filter added",
+ __FUNCTION__);
+ return -1;
+ }
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: deregister effect filter",
+ __FUNCTION__);
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "%s: register effect",
+ __FUNCTION__);
+ if (effect_filter_) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "%s: effect filter already added ", __FUNCTION__);
+ return -1;
+ }
+ }
+ effect_filter_ = effect_filter;
+ return 0;
+}
+
+ViEFileRecorder& ViEEncoder::GetOutgoingFileRecorder() {
+ return file_recorder_;
+}
+
+QMVideoSettingsCallback::QMVideoSettingsCallback(
+ WebRtc_Word32 engine_id,
+ WebRtc_Word32 channel_id,
+ VideoProcessingModule* vpm,
+ VideoCodingModule* vcm,
+ WebRtc_Word32 num_cores,
+ WebRtc_Word32 max_payload_length)
+ : engine_id_(engine_id),
+ channel_id_(channel_id),
+ vpm_(vpm),
+ vcm_(vcm),
+ num_cores_(num_cores),
+ max_payload_length_(max_payload_length) {
+}
+
+QMVideoSettingsCallback::~QMVideoSettingsCallback() {
+}
+
+WebRtc_Word32 QMVideoSettingsCallback::SetVideoQMSettings(
+ const WebRtc_UWord32 frame_rate,
+ const WebRtc_UWord32 width,
+ const WebRtc_UWord32 height) {
+ WebRtc_Word32 ret_val = 0;
+ ret_val = vpm_->SetTargetResolution(width, height, frame_rate);
+
+ if (!ret_val) {
+ // Get current settings.
+ VideoCodec current_codec;
+ vcm_->SendCodec(¤t_codec);
+ WebRtc_UWord32 current_bit_rate;
+ if (vcm_->Bitrate(¤t_bit_rate) != 0) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "Failed to get the current encoder target bitrate.");
+ }
+
+ // Set the new values.
+ current_codec.height = static_cast<WebRtc_UWord16>(height);
+ current_codec.width = static_cast<WebRtc_UWord16>(width);
+ current_codec.maxFramerate = static_cast<WebRtc_UWord8>(frame_rate);
+ current_codec.startBitrate = current_bit_rate;
+
+ // Re-register encoder with the updated settings.
+ ret_val = vcm_->RegisterSendCodec(¤t_codec, num_cores_,
+ max_payload_length_);
+ }
+ return ret_val;
+}
+
+void QMVideoSettingsCallback::SetMaxPayloadLength(
+ WebRtc_Word32 max_payload_length) {
+ max_payload_length_ = max_payload_length;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_encoder.h b/trunk/src/video_engine/vie_encoder.h
new file mode 100644
index 0000000..1b20491
--- /dev/null
+++ b/trunk/src/video_engine/vie_encoder.h
@@ -0,0 +1,182 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
+
+#include "common_types.h"
+#include "rtp_rtcp_defines.h"
+#include "typedefs.h"
+#include "video_coding_defines.h"
+#include "video_processing.h"
+#include "vie_defines.h"
+#include "vie_file_recorder.h"
+#include "vie_frame_provider_base.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class ProcessThread;
+class QMVideoSettingsCallback;
+class RtpRtcp;
+class VideoCodingModule;
+class ViEEffectFilter;
+class ViEEncoderObserver;
+
+class ViEEncoder
+ : public RtpVideoFeedback,
+ public RtcpFeedback,
+ public VCMPacketizationCallback,
+ public VCMProtectionCallback,
+ public VCMSendStatisticsCallback,
+ public ViEFrameCallback {
+ public:
+ ViEEncoder(WebRtc_Word32 engine_id,
+ WebRtc_Word32 channel_id,
+ WebRtc_UWord32 number_of_cores,
+ ProcessThread& module_process_thread);
+ ~ViEEncoder();
+
+ // Drops incoming packets before they get to the encoder.
+ void Pause();
+ void Restart();
+
+ WebRtc_Word32 DropDeltaAfterKey(bool enable);
+
+ // Codec settings.
+ WebRtc_UWord8 NumberOfCodecs();
+ WebRtc_Word32 GetCodec(WebRtc_UWord8 list_index, VideoCodec& video_codec);
+ WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* encoder,
+ WebRtc_UWord8 pl_type);
+ WebRtc_Word32 DeRegisterExternalEncoder(WebRtc_UWord8 pl_type);
+ WebRtc_Word32 SetEncoder(const VideoCodec& video_codec);
+ WebRtc_Word32 GetEncoder(VideoCodec& video_codec);
+
+ WebRtc_Word32 GetCodecConfigParameters(
+ unsigned char config_parameters[kConfigParameterSize],
+ unsigned char& config_parameters_size);
+
+ // Scale or crop/pad image.
+ WebRtc_Word32 ScaleInputImage(bool enable);
+
+ // RTP settings.
+ RtpRtcp* SendRtpRtcpModule();
+
+ // Implementing ViEFrameCallback.
+ virtual void DeliverFrame(int id,
+ VideoFrame& video_frame,
+ int num_csrcs = 0,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
+ virtual void DelayChanged(int id, int frame_delay);
+ virtual int GetPreferedFrameSettings(int& width,
+ int& height,
+ int& frame_rate);
+
+ virtual void ProviderDestroyed(int id) {
+ return;
+ }
+
+ WebRtc_Word32 EncodeFrame(VideoFrame& video_frame);
+ WebRtc_Word32 SendKeyFrame();
+ WebRtc_Word32 SendCodecStatistics(WebRtc_UWord32& num_key_frames,
+ WebRtc_UWord32& num_delta_frames);
+ WebRtc_Word32 EstimatedSendBandwidth(
+ WebRtc_UWord32* available_bandwidth) const;
+ int CodecTargetBitrate(WebRtc_UWord32* bitrate) const;
+ // Loss protection.
+ WebRtc_Word32 UpdateProtectionMethod();
+
+ // Implements VCMPacketizationCallback.
+ virtual WebRtc_Word32 SendData(
+ const FrameType frame_type,
+ const WebRtc_UWord8 payload_type,
+ const WebRtc_UWord32 time_stamp,
+ const WebRtc_UWord8* payload_data,
+ const WebRtc_UWord32 payload_size,
+ const RTPFragmentationHeader& fragmentation_header,
+ const RTPVideoHeader* rtp_video_hdr);
+
+ // Implements VideoProtectionCallback.
+ virtual WebRtc_Word32 ProtectionRequest(
+ WebRtc_UWord8 delta_fecrate,
+ WebRtc_UWord8 key_fecrate,
+ bool delta_use_uep_protection,
+ bool key_use_uep_protection,
+ bool nack_enabled,
+ WebRtc_UWord32* sent_video_rate_bps,
+ WebRtc_UWord32* sent_nack_rate_bps,
+ WebRtc_UWord32* sent_fec_rate_bps);
+
+ // Implements VideoSendStatisticsCallback.
+ virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bit_rate,
+ const WebRtc_UWord32 frame_rate);
+ WebRtc_Word32 RegisterCodecObserver(ViEEncoderObserver* observer);
+
+ // Implements RtcpFeedback.
+ virtual void OnSLIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 picture_id);
+ virtual void OnRPSIReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord64 picture_id);
+
+ // Implements RtpVideoFeedback.
+ virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+ const FrameType type,
+ const WebRtc_UWord8 stream_idx);
+
+ virtual void OnNetworkChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 bitrate_bps,
+ const WebRtc_UWord8 fraction_lost,
+ const WebRtc_UWord16 round_trip_time_ms);
+
+ // Effect filter.
+ WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
+
+ // Recording.
+ ViEFileRecorder& GetOutgoingFileRecorder();
+
+ private:
+ WebRtc_Word32 engine_id_;
+ WebRtc_Word32 channel_id_;
+ const WebRtc_UWord32 number_of_cores_;
+
+ VideoCodingModule& vcm_;
+ VideoProcessingModule& vpm_;
+ RtpRtcp& default_rtp_rtcp_;
+ scoped_ptr<CriticalSectionWrapper> callback_cs_;
+ scoped_ptr<CriticalSectionWrapper> data_cs_;
+ VideoCodec send_codec_;
+
+ bool paused_;
+ WebRtc_Word64 time_last_intra_request_ms_[kMaxSimulcastStreams];
+ WebRtc_Word32 channels_dropping_delta_frames_;
+ bool drop_next_frame_;
+
+ bool fec_enabled_;
+ bool nack_enabled_;
+
+ ViEEncoderObserver* codec_observer_;
+ ViEEffectFilter* effect_filter_;
+ ProcessThread& module_process_thread_;
+
+ bool has_received_sli_;
+ WebRtc_UWord8 picture_id_sli_;
+ bool has_received_rpsi_;
+ WebRtc_UWord64 picture_id_rpsi_;
+
+ ViEFileRecorder file_recorder_;
+
+ // Quality modes callback
+ QMVideoSettingsCallback* qm_callback_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
diff --git a/trunk/src/video_engine/vie_encryption_impl.cc b/trunk/src/video_engine/vie_encryption_impl.cc
new file mode 100644
index 0000000..adbda42
--- /dev/null
+++ b/trunk/src/video_engine/vie_encryption_impl.cc
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_encryption_impl.h"
+
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViEEncryption* ViEEncryption::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+ if (video_engine == NULL) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViEEncryptionImpl* vie_encryption_impl = vie_impl;
+ // Increase ref count.
+ (*vie_encryption_impl)++;
+ return vie_encryption_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViEEncryptionImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViEEncryptionImpl::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViEEncryptionImpl release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViEEncryptionImpl reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViEEncryptionImpl::ViEEncryptionImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEEncryptionImpl::ViEEncryptionImpl() Ctor");
+}
+
+ViEEncryptionImpl::~ViEEncryptionImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEEncryptionImpl::~ViEEncryptionImpl() Dtor");
+}
+
+int ViEEncryptionImpl::RegisterExternalEncryption(const int video_channel,
+ Encryption& encryption) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "RegisterExternalEncryption(video_channel=%d)", video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (vie_channel == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEEncryptionInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterExternalEncryption(&encryption) != 0) {
+ shared_data_->SetLastError(kViEEncryptionUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEEncryptionImpl::DeregisterExternalEncryption(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "RegisterExternalEncryption(video_channel=%d)", video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (vie_channel == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEEncryptionInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->DeRegisterExternalEncryption() != 0) {
+ shared_data_->SetLastError(kViEEncryptionUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_encryption_impl.h b/trunk/src/video_engine/vie_encryption_impl.h
new file mode 100644
index 0000000..f398461
--- /dev/null
+++ b/trunk/src/video_engine/vie_encryption_impl.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
+
+#include "typedefs.h"
+#include "video_engine/include/vie_encryption.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViEEncryptionImpl
+ : public ViEEncryption,
+ public ViERefCount {
+ public:
+ virtual int Release();
+
+ // Implements ViEEncryption.
+ virtual int RegisterExternalEncryption(const int video_channel,
+ Encryption& encryption);
+ virtual int DeregisterExternalEncryption(const int video_channel);
+
+ protected:
+ ViEEncryptionImpl(ViESharedData* shared_data);
+ virtual ~ViEEncryptionImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
diff --git a/trunk/src/video_engine/vie_external_codec_impl.cc b/trunk/src/video_engine/vie_external_codec_impl.cc
new file mode 100644
index 0000000..92e7dd6
--- /dev/null
+++ b/trunk/src/video_engine/vie_external_codec_impl.cc
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_external_codec_impl.h"
+
+#include "engine_configurations.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViEExternalCodec* ViEExternalCodec::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+ if (video_engine == NULL) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViEExternalCodecImpl* vie_external_codec_impl = vie_impl;
+ // Increase ref count.
+ (*vie_external_codec_impl)++;
+ return vie_external_codec_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViEExternalCodecImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViEExternalCodec::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViEExternalCodec release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViEExternalCodec reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViEExternalCodecImpl::ViEExternalCodecImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEExternalCodecImpl::ViEExternalCodecImpl() Ctor");
+}
+
+ViEExternalCodecImpl::~ViEExternalCodecImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEExternalCodecImpl::~ViEExternalCodecImpl() Dtor");
+}
+
+int ViEExternalCodecImpl::RegisterExternalSendCodec(const int video_channel,
+ const unsigned char pl_type,
+ VideoEncoder* encoder) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s channel %d pl_type %d encoder 0x%x", __FUNCTION__,
+ video_channel, pl_type, encoder);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument video_channel %u. Does it exist?",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+ if (!encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument Encoder 0x%x.", __FUNCTION__, encoder);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+
+ if (vie_encoder->RegisterExternalEncoder(encoder, pl_type) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEExternalCodecImpl::DeRegisterExternalSendCodec(
+ const int video_channel, const unsigned char pl_type) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s channel %d pl_type %d", __FUNCTION__, video_channel,
+ pl_type);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument video_channel %u. Does it exist?",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+
+ if (vie_encoder->DeRegisterExternalEncoder(pl_type) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEExternalCodecImpl::RegisterExternalReceiveCodec(
+ const int video_channel,
+ const unsigned int pl_type,
+ VideoDecoder* decoder,
+ bool decoder_render,
+ int render_delay) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s channel %d pl_type %d decoder 0x%x, decoder_render %d, "
+ "renderDelay %d", __FUNCTION__, video_channel, pl_type, decoder,
+ decoder_render, render_delay);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument video_channel %u. Does it exist?",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+ if (!decoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument decoder 0x%x.", __FUNCTION__, decoder);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+
+ if (vie_channel->RegisterExternalDecoder(pl_type, decoder, decoder_render,
+ render_delay) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEExternalCodecImpl::DeRegisterExternalReceiveCodec(
+const int video_channel, const unsigned char pl_type) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s channel %d pl_type %u", __FUNCTION__, video_channel,
+ pl_type);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Invalid argument video_channel %u. Does it exist?",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViECodecInvalidArgument);
+ return -1;
+ }
+ if (vie_channel->DeRegisterExternalDecoder(pl_type) != 0) {
+ shared_data_->SetLastError(kViECodecUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_external_codec_impl.h b/trunk/src/video_engine/vie_external_codec_impl.h
new file mode 100644
index 0000000..67d6a78
--- /dev/null
+++ b/trunk/src/video_engine/vie_external_codec_impl.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
+
+#include "video_engine/include/vie_external_codec.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViEExternalCodecImpl
+ : public ViEExternalCodec,
+ public ViERefCount {
+ public:
+ // Implements ViEExternalCodec.
+ virtual int Release();
+ virtual int RegisterExternalSendCodec(const int video_channel,
+ const unsigned char pl_type,
+ VideoEncoder* encoder);
+ virtual int DeRegisterExternalSendCodec(const int video_channel,
+ const unsigned char pl_type);
+ virtual int RegisterExternalReceiveCodec(const int video_channel,
+ const unsigned int pl_type,
+ VideoDecoder* decoder,
+ bool decoder_render = false,
+ int render_delay = 0);
+ virtual int DeRegisterExternalReceiveCodec(const int video_channel,
+ const unsigned char pl_type);
+
+ protected:
+ ViEExternalCodecImpl(ViESharedData* shared_data);
+ virtual ~ViEExternalCodecImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
diff --git a/trunk/src/video_engine/vie_file_image.cc b/trunk/src/video_engine/vie_file_image.cc
new file mode 100644
index 0000000..2472a3a
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_image.cc
@@ -0,0 +1,125 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Placed first to get WEBRTC_VIDEO_ENGINE_FILE_API.
+#include "engine_configurations.h"
+
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+
+#include "video_engine/vie_file_image.h"
+
+#include <stdio.h>
+
+#include "common_video/interface/video_image.h"
+#include "common_video/jpeg/include/jpeg.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+int ViEFileImage::ConvertJPEGToVideoFrame(int engine_id,
+ const char* file_nameUTF8,
+ VideoFrame& video_frame) {
+ // Read jpeg file into temporary buffer.
+ EncodedImage image_buffer;
+
+ FILE* image_file = fopen(file_nameUTF8, "rb");
+ if (!image_file) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s could not open file %s", __FUNCTION__, file_nameUTF8);
+ return -1;
+ }
+ if (fseek(image_file, 0, SEEK_END) != 0) {
+ fclose(image_file);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "ConvertJPEGToVideoFrame fseek SEEK_END error for file %s",
+ file_nameUTF8);
+ return -1;
+ }
+ int buffer_size = ftell(image_file);
+ if (buffer_size == -1) {
+ fclose(image_file);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "ConvertJPEGToVideoFrame could tell file size for file %s",
+ file_nameUTF8);
+ return -1;
+ }
+ image_buffer._size = buffer_size;
+ if (fseek(image_file, 0, SEEK_SET) != 0) {
+ fclose(image_file);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "ConvertJPEGToVideoFrame fseek SEEK_SET error for file %s",
+ file_nameUTF8);
+ return -1;
+ }
+ image_buffer._buffer = new WebRtc_UWord8[ image_buffer._size + 1];
+ if (image_buffer._size != fread(image_buffer._buffer, sizeof(WebRtc_UWord8),
+ image_buffer._size, image_file)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s could not read file %s", __FUNCTION__, file_nameUTF8);
+ fclose(image_file);
+ delete [] image_buffer._buffer;
+ return -1;
+ }
+ fclose(image_file);
+
+ JpegDecoder decoder;
+ RawImage decoded_image;
+ int ret = decoder.Decode(image_buffer, decoded_image);
+
+ delete [] image_buffer._buffer;
+ image_buffer._buffer = NULL;
+
+ if (ret == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s could decode file %s from jpeg format", __FUNCTION__,
+ file_nameUTF8);
+ return -1;
+ } else if (ret == -3) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s could not convert jpeg's data to i420 format",
+ __FUNCTION__, file_nameUTF8);
+ }
+
+ // Image length in I420.
+ WebRtc_UWord32 image_length = (WebRtc_UWord32)(decoded_image._width *
+ decoded_image._height * 1.5);
+ if (-1 == video_frame.Swap(decoded_image._buffer, image_length,
+ image_length)) {
+ WEBRTC_TRACE(kTraceDebug, kTraceVideo, engine_id,
+ "%s could not copy frame image_decoded_buffer to video_frame ",
+ __FUNCTION__, file_nameUTF8);
+ return -1;
+ }
+
+ if (decoded_image._buffer) {
+ delete [] decoded_image._buffer;
+ decoded_image._buffer = NULL;
+ }
+
+ video_frame.SetWidth(decoded_image._width);
+ video_frame.SetHeight(decoded_image._height);
+ return 0;
+}
+
+int ViEFileImage::ConvertPictureToVideoFrame(int engine_id,
+ const ViEPicture& picture,
+ VideoFrame& video_frame) {
+ WebRtc_UWord32 picture_length = (WebRtc_UWord32)(picture.width *
+ picture.height * 1.5);
+ video_frame.CopyFrame(picture_length, picture.data);
+ video_frame.SetWidth(picture.width);
+ video_frame.SetHeight(picture.height);
+ video_frame.SetLength(picture_length);
+ return 0;
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_FILE_API
diff --git a/trunk/src/video_engine/vie_file_image.h b/trunk/src/video_engine/vie_file_image.h
new file mode 100644
index 0000000..e79b115
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_image.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
+
+#include "modules/interface/module_common_types.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_file.h"
+
+namespace webrtc {
+
+class ViEFileImage {
+ public:
+ static int ConvertJPEGToVideoFrame(int engine_id,
+ const char* file_nameUTF8,
+ VideoFrame& video_frame);
+ static int ConvertPictureToVideoFrame(int engine_id,
+ const ViEPicture& picture,
+ VideoFrame& video_frame);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
diff --git a/trunk/src/video_engine/vie_file_impl.cc b/trunk/src/video_engine/vie_file_impl.cc
new file mode 100644
index 0000000..28f9854
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_impl.cc
@@ -0,0 +1,1016 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "engine_configurations.h"
+
+#include "video_engine/vie_file_impl.h"
+
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+#include "common_video/jpeg/include/jpeg.h"
+#include "system_wrappers/interface/condition_variable_wrapper.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_file_image.h"
+#include "video_engine/vie_file_player.h"
+#include "video_engine/vie_file_recorder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_input_manager.h"
+#include "video_engine/vie_render_manager.h"
+#include "video_engine/vie_renderer.h"
+#endif
+
+namespace webrtc {
+
+ViEFile* ViEFile::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViEFileImpl* vie_file_impl = vie_impl;
+ // Increase ref count.
+ (*vie_file_impl)++;
+ return vie_file_impl;
+#else
+ return NULL;
+#endif
+}
+
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+
+int ViEFileImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViEFile::Release()");
+ // Decrease ref count.
+ (*this)--;
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViEFile release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViEFile reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViEFileImpl::ViEFileImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEFileImpl::ViEFileImpl() Ctor");
+}
+
+ViEFileImpl::~ViEFileImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEFileImpl::~ViEFileImpl() Dtor");
+}
+
+int ViEFileImpl::StartPlayFile(const char* file_nameUTF8,
+ int& file_id,
+ const bool loop,
+ const FileFormats file_format) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s", __FUNCTION__);
+
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ VoiceEngine* voice = shared_data_->channel_manager()->GetVoiceEngine();
+ const WebRtc_Word32 result = shared_data_->input_manager()->CreateFilePlayer(
+ file_nameUTF8, loop, file_format, voice, file_id);
+ if (result != 0) {
+ shared_data_->SetLastError(result);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StopPlayFile(const int file_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(file_id: %d)", __FUNCTION__, file_id);
+ {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ }
+ // Destroy the capture device.
+ return shared_data_->input_manager()->DestroyFilePlayer(file_id);
+}
+
+int ViEFileImpl::RegisterObserver(int file_id, ViEFileObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(file_id: %d)", __FUNCTION__, file_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ if (vie_file_player->IsObserverRegistered()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), file_id),
+ "%s: Observer already registered", __FUNCTION__);
+ shared_data_->SetLastError(kViEFileObserverAlreadyRegistered);
+ return -1;
+ }
+ if (vie_file_player->RegisterObserver(observer) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), file_id),
+ "%s: Failed to register observer", __FUNCTION__, file_id);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::DeregisterObserver(int file_id, ViEFileObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(file_id: %d)", __FUNCTION__, file_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ if (!vie_file_player->IsObserverRegistered()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), file_id),
+ "%s: No Observer registered", __FUNCTION__);
+ shared_data_->SetLastError(kViEFileObserverNotRegistered);
+ return -1;
+ }
+ if (vie_file_player->DeRegisterObserver() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), file_id),
+ "%s: Failed to deregister observer", __FUNCTION__, file_id);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::SendFileOnChannel(const int file_id, const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(file_id: %d)", __FUNCTION__, file_id);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ if (is.FrameProvider(vie_encoder) != NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d already connected to a capture device or "
+ "file.", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInputAlreadyConnected);
+ return -1;
+ }
+
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+
+ if (vie_file_player->RegisterFrameCallback(video_channel, vie_encoder)
+ != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Failed to register frame callback.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StopSendFileOnChannel(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
+ if (!frame_provider ||
+ frame_provider->Id() < kViEFileIdBase ||
+ frame_provider->Id() > kViEFileIdMax) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: No file connected to Channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileNotConnected);
+ return -1;
+ }
+ if (frame_provider->DeregisterFrameCallback(vie_encoder) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Failed to deregister file from channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ }
+ return 0;
+}
+
+int ViEFileImpl::StartPlayFileAsMicrophone(const int file_id,
+ const int audio_channel,
+ bool mix_microphone,
+ float volume_scaling) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ if (vie_file_player->SendAudioOnChannel(audio_channel, mix_microphone,
+ volume_scaling) != 0) {
+ shared_data_->SetLastError(kViEFileVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StopPlayFileAsMicrophone(const int file_id,
+ const int audio_channel) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+
+ if (vie_file_player->StopSendAudioOnChannel(audio_channel) != 0) {
+ shared_data_->SetLastError(kViEFileVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StartPlayAudioLocally(const int file_id,
+ const int audio_channel,
+ float volume_scaling) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ if (vie_file_player->PlayAudioLocally(audio_channel, volume_scaling) != 0) {
+ shared_data_->SetLastError(kViEFileVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StopPlayAudioLocally(const int file_id,
+ const int audio_channel) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+
+ ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: File with id %d is not playing.", __FUNCTION__,
+ file_id);
+ shared_data_->SetLastError(kViEFileNotPlaying);
+ return -1;
+ }
+ if (vie_file_player->StopPlayAudioLocally(audio_channel) != 0) {
+ shared_data_->SetLastError(kViEFileVoEFailure);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StartRecordOutgoingVideo(const int video_channel,
+ const char* file_nameUTF8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+ ViEFileRecorder& file_recorder = vie_encoder->GetOutgoingFileRecorder();
+ if (file_recorder.RecordingStarted()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Already recording outgoing video on channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileAlreadyRecording);
+ return -1;
+ }
+
+ WebRtc_Word32 ve_channel_id = -1;
+ VoiceEngine* ve_ptr = NULL;
+ if (audio_source != NO_AUDIO) {
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ // Channel should exists since we have a ViEEncoder above.
+ assert(false);
+ return -1;
+ }
+ ve_channel_id = vie_channel->VoiceChannel();
+ ve_ptr = shared_data_->channel_manager()->GetVoiceEngine();
+ if (!ve_ptr) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Can't access voice engine. Have SetVoiceEngine "
+ "been called?", __FUNCTION__);
+ shared_data_->SetLastError(kViEFileVoENotSet);
+ return -1;
+ }
+ }
+ if (file_recorder.StartRecording(file_nameUTF8, video_codec, audio_source,
+ ve_channel_id, audio_codec, ve_ptr,
+ file_format) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Failed to start recording. Check arguments.",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+
+ return 0;
+}
+
+int ViEFileImpl::StopRecordOutgoingVideo(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+ ViEFileRecorder& file_recorder = vie_encoder->GetOutgoingFileRecorder();
+ if (!file_recorder.RecordingStarted()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d is not recording.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileNotRecording);
+ return -1;
+ }
+ if (file_recorder.StopRecording() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Failed to stop recording of channel %d.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::StopRecordIncomingVideo(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+ ViEFileRecorder& file_recorder = vie_channel->GetIncomingFileRecorder();
+ if (!file_recorder.RecordingStarted()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d is not recording.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileNotRecording);
+ vie_channel->ReleaseIncomingFileRecorder();
+ return -1;
+ }
+ if (file_recorder.StopRecording() != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Failed to stop recording of channel %d.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ vie_channel->ReleaseIncomingFileRecorder();
+ return -1;
+ }
+ // Let the channel know we are no longer recording.
+ vie_channel->ReleaseIncomingFileRecorder();
+ return 0;
+}
+
+int ViEFileImpl::StartRecordIncomingVideo(const int video_channel,
+ const char* file_nameUTF8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileInvalidChannelId);
+ return -1;
+ }
+ ViEFileRecorder& file_recorder = vie_channel->GetIncomingFileRecorder();
+ if (file_recorder.RecordingStarted()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Already recording outgoing video on channel %d",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileAlreadyRecording);
+ return -1;
+ }
+
+ WebRtc_Word32 ve_channel_id = -1;
+ VoiceEngine* ve_ptr = NULL;
+ if (audio_source != NO_AUDIO) {
+ ve_channel_id = vie_channel->VoiceChannel();
+ ve_ptr = shared_data_->channel_manager()->GetVoiceEngine();
+
+ if (!ve_ptr) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Can't access voice engine. Have SetVoiceEngine "
+ "been called?", __FUNCTION__);
+ shared_data_->SetLastError(kViEFileVoENotSet);
+ return -1;
+ }
+ }
+ if (file_recorder.StartRecording(file_nameUTF8, video_codec, audio_source,
+ ve_channel_id, audio_codec, ve_ptr,
+ file_format) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Failed to start recording. Check arguments.",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViEFileUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::GetFileInformation(const char* file_name,
+ VideoCodec& video_codec,
+ CodecInst& audio_codec,
+ const FileFormats file_format) {
+ return ViEFilePlayer::GetFileInformation(
+ shared_data_->instance_id(),
+ static_cast<const WebRtc_Word8*>(file_name),
+ video_codec, audio_codec, file_format);
+}
+
+int ViEFileImpl::GetRenderSnapshot(const int video_channel,
+ const char* file_nameUTF8) {
+ // Gain access to the renderer for the specified channel and get it's
+ // current frame.
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ return -1;
+ }
+
+ VideoFrame video_frame;
+ if (renderer->GetLastRenderedFrame(video_channel, video_frame) == -1) {
+ return -1;
+ }
+
+ // JPEGEncoder writes the jpeg file for you (no control over it) and does
+ // not return you the buffer. Thus, we are not going to be writing to the
+ // disk here.
+ JpegEncoder jpeg_encoder;
+ RawImage input_image;
+ if (jpeg_encoder.SetFileName(file_nameUTF8) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "\tCould not open output file '%s' for writing!",
+ file_nameUTF8);
+ return -1;
+ }
+
+ input_image._width = video_frame.Width();
+ input_image._height = video_frame.Height();
+ video_frame.Swap(input_image._buffer, input_image._length,
+ input_image._size);
+
+ if (jpeg_encoder.Encode(input_image) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "\tCould not encode i420 -> jpeg file '%s' for writing!",
+ file_nameUTF8);
+ if (input_image._buffer) {
+ delete [] input_image._buffer;
+ input_image._buffer = NULL;
+ }
+ return -1;
+ }
+ delete [] input_image._buffer;
+ input_image._buffer = NULL;
+ return 0;
+}
+
+int ViEFileImpl::GetRenderSnapshot(const int video_channel,
+ ViEPicture& picture) {
+ // Gain access to the renderer for the specified channel and get it's
+ // current frame.
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ return -1;
+ }
+
+ VideoFrame video_frame;
+ if (renderer->GetLastRenderedFrame(video_channel, video_frame) == -1) {
+ return -1;
+ }
+
+ // Copy from VideoFrame class to ViEPicture struct.
+ int buffer_length =
+ static_cast<int>(video_frame.Width() * video_frame.Height() * 1.5);
+ picture.data = static_cast<WebRtc_UWord8*>(malloc(
+ buffer_length * sizeof(WebRtc_UWord8)));
+ memcpy(picture.data, video_frame.Buffer(), buffer_length);
+ picture.size = buffer_length;
+ picture.width = video_frame.Width();
+ picture.height = video_frame.Height();
+ picture.type = kVideoI420;
+ return 0;
+}
+
+int ViEFileImpl::GetCaptureDeviceSnapshot(const int capture_id,
+ const char* file_nameUTF8) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* capturer = is.Capture(capture_id);
+ if (!capturer) {
+ return -1;
+ }
+
+ VideoFrame video_frame;
+ if (GetNextCapturedFrame(capture_id, video_frame) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "Could not gain acces to capture device %d video frame "
+ "%s:%d", capture_id, __FUNCTION__);
+ return -1;
+ }
+
+ // JPEGEncoder writes the jpeg file for you (no control over it) and does
+ // not return you the buffer Thusly, we are not going to be writing to the
+ // disk here.
+ JpegEncoder jpeg_encoder;
+ RawImage input_image;
+ input_image._width = video_frame.Width();
+ input_image._height = video_frame.Height();
+ video_frame.Swap(input_image._buffer, input_image._length,
+ input_image._size);
+
+ if (jpeg_encoder.SetFileName(file_nameUTF8) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "\tCould not open output file '%s' for writing!",
+ file_nameUTF8);
+
+ if (input_image._buffer) {
+ delete [] input_image._buffer;
+ }
+ return -1;
+ }
+ if (jpeg_encoder.Encode(input_image) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "\tCould not encode i420 -> jpeg file '%s' for "
+ "writing!", file_nameUTF8);
+ if (input_image._buffer) {
+ delete [] input_image._buffer;
+ }
+ return -1;
+ }
+ delete [] input_image._buffer;
+ input_image._buffer = NULL;
+ return 0;
+}
+
+int ViEFileImpl::GetCaptureDeviceSnapshot(const int capture_id,
+ ViEPicture& picture) {
+ VideoFrame video_frame;
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* capturer = is.Capture(capture_id);
+ if (!capturer) {
+ return -1;
+ }
+ if (GetNextCapturedFrame(capture_id, video_frame) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
+ "Could not gain acces to capture device %d video frame "
+ "%s:%d", capture_id, __FUNCTION__);
+ return -1;
+ }
+
+ // Copy from VideoFrame class to ViEPicture struct.
+ int buffer_length =
+ static_cast<int>(video_frame.Width() * video_frame.Height() * 1.5);
+ picture.data = static_cast<WebRtc_UWord8*>(malloc(
+ buffer_length * sizeof(WebRtc_UWord8)));
+ memcpy(picture.data, video_frame.Buffer(), buffer_length);
+ picture.size = buffer_length;
+ picture.width = video_frame.Width();
+ picture.height = video_frame.Height();
+ picture.type = kVideoI420;
+ return 0;
+}
+
+int ViEFileImpl::FreePicture(ViEPicture& picture) {
+ if (picture.data) {
+ free(picture.data);
+ }
+
+ picture.data = NULL;
+ picture.size = 0;
+ picture.width = 0;
+ picture.height = 0;
+ picture.type = kVideoUnknown;
+ return 0;
+}
+int ViEFileImpl::SetCaptureDeviceImage(const int capture_id,
+ const char* file_nameUTF8) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* capturer = is.Capture(capture_id);
+ if (!capturer) {
+ shared_data_->SetLastError(kViEFileInvalidCaptureId);
+ return -1;
+ }
+
+ VideoFrame capture_image;
+ if (ViEFileImage::ConvertJPEGToVideoFrame(
+ ViEId(shared_data_->instance_id(), capture_id), file_nameUTF8,
+ capture_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s(capture_id: %d) Failed to open file.", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEFileInvalidFile);
+ return -1;
+ }
+ if (capturer->SetCaptureDeviceImage(capture_image)) {
+ shared_data_->SetLastError(kViEFileSetCaptureImageError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::SetCaptureDeviceImage(const int capture_id,
+const ViEPicture& picture) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ if (picture.type != kVideoI420) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s(capture_id: %d) Not a valid picture type.",
+ __FUNCTION__, capture_id);
+ shared_data_->SetLastError(kViEFileInvalidArgument);
+ return -1;
+ }
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* capturer = is.Capture(capture_id);
+ if (!capturer) {
+ shared_data_->SetLastError(kViEFileSetCaptureImageError);
+ return -1;
+ }
+
+ VideoFrame capture_image;
+ if (ViEFileImage::ConvertPictureToVideoFrame(
+ ViEId(shared_data_->instance_id(), capture_id), picture, capture_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), capture_id),
+ "%s(capture_id: %d) Failed to use picture.", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEFileInvalidFile);
+ return -1;
+ }
+ if (capturer->SetCaptureDeviceImage(capture_image)) {
+ shared_data_->SetLastError(kViEFileInvalidCapture);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::SetRenderStartImage(const int video_channel,
+const char* file_nameUTF8) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ shared_data_->SetLastError(kViEFileInvalidRenderId);
+ return -1;
+ }
+
+ VideoFrame start_image;
+ if (ViEFileImage::ConvertJPEGToVideoFrame(
+ ViEId(shared_data_->instance_id(), video_channel), file_nameUTF8, start_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Failed to open file.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileInvalidFile);
+ return -1;
+ }
+ if (renderer->SetRenderStartImage(start_image) != 0) {
+ shared_data_->SetLastError(kViEFileSetStartImageError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::SetRenderStartImage(const int video_channel,
+ const ViEPicture& picture) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+ if (picture.type != kVideoI420) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Not a valid picture type.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidArgument);
+ return -1;
+ }
+
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ shared_data_->SetLastError(kViEFileInvalidRenderId);
+ return -1;
+ }
+
+ VideoFrame start_image;
+ if (ViEFileImage::ConvertPictureToVideoFrame(
+ ViEId(shared_data_->instance_id(), video_channel), picture, start_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Failed to use picture.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidCapture);
+ return -1;
+ }
+ if (renderer->SetRenderStartImage(start_image) != 0) {
+ shared_data_->SetLastError(kViEFileSetStartImageError);
+ return -1;
+ }
+ return 0;
+}
+int ViEFileImpl::SetRenderTimeoutImage(const int video_channel,
+ const char* file_nameUTF8,
+ const unsigned int timeout_ms) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ shared_data_->SetLastError(kViEFileInvalidRenderId);
+ return -1;
+ }
+ VideoFrame timeout_image;
+ if (ViEFileImage::ConvertJPEGToVideoFrame(
+ ViEId(shared_data_->instance_id(), video_channel), file_nameUTF8,
+ timeout_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Failed to open file.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViEFileInvalidFile);
+ return -1;
+ }
+ WebRtc_Word32 timeout_time = timeout_ms;
+ if (timeout_ms < kViEMinRenderTimeoutTimeMs) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Invalid timeout_ms, using %d.",
+ __FUNCTION__, video_channel, kViEMinRenderTimeoutTimeMs);
+ timeout_time = kViEMinRenderTimeoutTimeMs;
+ }
+ if (timeout_ms > kViEMaxRenderTimeoutTimeMs) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Invalid timeout_ms, using %d.",
+ __FUNCTION__, video_channel, kViEMaxRenderTimeoutTimeMs);
+ timeout_time = kViEMaxRenderTimeoutTimeMs;
+ }
+ if (renderer->SetTimeoutImage(timeout_image, timeout_time) != 0) {
+ shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEFileImpl::SetRenderTimeoutImage(const int video_channel,
+ const ViEPicture& picture,
+const unsigned int timeout_ms) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ if (picture.type != kVideoI420) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Not a valid picture type.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidArgument);
+ return -1;
+ }
+
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(video_channel);
+ if (!renderer) {
+ shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
+ return -1;
+ }
+ VideoFrame timeout_image;
+ if (ViEFileImage::ConvertPictureToVideoFrame(
+ ViEId(shared_data_->instance_id(), video_channel), picture,
+ timeout_image) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Failed to use picture.",
+ __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEFileInvalidCapture);
+ return -1;
+ }
+ WebRtc_Word32 timeout_time = timeout_ms;
+ if (timeout_ms < kViEMinRenderTimeoutTimeMs) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Invalid timeout_ms, using %d.",
+ __FUNCTION__, video_channel, kViEMinRenderTimeoutTimeMs);
+ timeout_time = kViEMinRenderTimeoutTimeMs;
+ }
+ if (timeout_ms > kViEMaxRenderTimeoutTimeMs) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(video_channel: %d) Invalid timeout_ms, using %d.",
+ __FUNCTION__, video_channel, kViEMaxRenderTimeoutTimeMs);
+ timeout_time = kViEMaxRenderTimeoutTimeMs;
+ }
+ if (renderer->SetTimeoutImage(timeout_image, timeout_time) != 0) {
+ shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 ViEFileImpl::GetNextCapturedFrame(WebRtc_Word32 capture_id,
+VideoFrame& video_frame) {
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* capturer = is.Capture(capture_id);
+ if (!capturer) {
+ return -1;
+ }
+
+ ViECaptureSnapshot* snap_shot = new ViECaptureSnapshot();
+ capturer->RegisterFrameCallback(-1, snap_shot);
+ bool snapshot_taken = snap_shot->GetSnapshot(
+ video_frame, kViECaptureMaxSnapshotWaitTimeMs);
+
+ // Check once again if it has been destroyed.
+ capturer->DeregisterFrameCallback(snap_shot);
+ delete snap_shot;
+ snap_shot = NULL;
+
+ if (snapshot_taken) {
+ return 0;
+ }
+ return -1;
+}
+
+ViECaptureSnapshot::ViECaptureSnapshot()
+ : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ condition_varaible_(ConditionVariableWrapper::CreateConditionVariable()),
+ video_frame_(NULL) {
+}
+
+ViECaptureSnapshot::~ViECaptureSnapshot() {
+ if (video_frame_) {
+ delete video_frame_;
+ video_frame_ = NULL;
+ }
+}
+
+bool ViECaptureSnapshot::GetSnapshot(VideoFrame& video_frame,
+ unsigned int max_wait_time) {
+ crit_->Enter();
+ video_frame_ = new VideoFrame();
+ if (condition_varaible_->SleepCS(*(crit_.get()), max_wait_time)) {
+ // Snapshot taken.
+ video_frame.SwapFrame(*video_frame_);
+ delete video_frame_;
+ video_frame_ = NULL;
+ crit_->Leave();
+ return true;
+ }
+ crit_->Leave();
+ return false;
+}
+
+void ViECaptureSnapshot::DeliverFrame(int id, VideoFrame& video_frame,
+ int num_csrcs,
+const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
+ CriticalSectionScoped cs(crit_.get());
+ if (!video_frame_) {
+ return;
+ }
+ video_frame_->SwapFrame(video_frame);
+ condition_varaible_->WakeAll();
+ return;
+}
+
+#endif
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_impl.h b/trunk/src/video_engine/vie_file_impl.h
new file mode 100644
index 0000000..63e2d4a
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_impl.h
@@ -0,0 +1,131 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
+
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_file.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_frame_provider_base.h"
+#include "video_engine/vie_ref_count.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+class ConditionVariableWrapper;
+class CriticalSectionWrapper;
+class ViESharedData;
+
+class ViECaptureSnapshot : public ViEFrameCallback {
+ public:
+ ViECaptureSnapshot();
+ ~ViECaptureSnapshot();
+
+ bool GetSnapshot(VideoFrame& video_frame, unsigned int max_wait_time);
+
+ // Implements ViEFrameCallback.
+ virtual void DeliverFrame(int id, VideoFrame& video_frame, int num_csrcs = 0,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
+ virtual void DelayChanged(int id, int frame_delay) {}
+ virtual int GetPreferedFrameSettings(int& width, int& height,
+ int& frame_rate) {
+ return -1;
+ }
+ virtual void ProviderDestroyed(int id) {}
+
+ private:
+ scoped_ptr<CriticalSectionWrapper> crit_;
+ scoped_ptr<ConditionVariableWrapper> condition_varaible_;
+ VideoFrame* video_frame_;
+};
+
+class ViEFileImpl
+ : public ViEFile,
+ public ViERefCount {
+ public:
+ // Implements ViEFile.
+ virtual int Release();
+ virtual int StartPlayFile(const char* file_nameUTF8, int& file_id,
+ const bool loop = false,
+ const FileFormats file_format = kFileFormatAviFile);
+ virtual int StopPlayFile(const int file_id);
+ virtual int RegisterObserver(int file_id, ViEFileObserver& observer);
+ virtual int DeregisterObserver(int file_id, ViEFileObserver& observer);
+ virtual int SendFileOnChannel(const int file_id, const int video_channel);
+ virtual int StopSendFileOnChannel(const int video_channel);
+ virtual int StartPlayFileAsMicrophone(const int file_id,
+ const int audio_channel,
+ bool mix_microphone = false,
+ float volume_scaling = 1);
+ virtual int StopPlayFileAsMicrophone(const int file_id,
+ const int audio_channel);
+ virtual int StartPlayAudioLocally(const int file_id, const int audio_channel,
+ float volume_scaling = 1);
+ virtual int StopPlayAudioLocally(const int file_id, const int audio_channel);
+ virtual int StartRecordOutgoingVideo(
+ const int video_channel,
+ const char* file_nameUTF8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format = kFileFormatAviFile);
+ virtual int StartRecordIncomingVideo(
+ const int video_channel,
+ const char* file_nameUTF8,
+ AudioSource audio_source,
+ const CodecInst& audio_codec,
+ const VideoCodec& video_codec,
+ const FileFormats file_format = kFileFormatAviFile);
+ virtual int StopRecordOutgoingVideo(const int video_channel);
+ virtual int StopRecordIncomingVideo(const int video_channel);
+ virtual int GetFileInformation(
+ const char* file_name,
+ VideoCodec& video_codec,
+ CodecInst& audio_codec,
+ const FileFormats file_format = kFileFormatAviFile);
+ virtual int GetRenderSnapshot(const int video_channel,
+ const char* file_nameUTF8);
+ virtual int GetRenderSnapshot(const int video_channel, ViEPicture& picture);
+ virtual int FreePicture(ViEPicture& picture);
+ virtual int GetCaptureDeviceSnapshot(const int capture_id,
+ const char* file_nameUTF8);
+ virtual int GetCaptureDeviceSnapshot(const int capture_id,
+ ViEPicture& picture);
+ virtual int SetCaptureDeviceImage(const int capture_id,
+ const char* file_nameUTF8);
+ virtual int SetCaptureDeviceImage(const int capture_id,
+ const ViEPicture& picture);
+ virtual int SetRenderStartImage(const int video_channel,
+ const char* file_nameUTF8);
+ virtual int SetRenderStartImage(const int video_channel,
+ const ViEPicture& picture);
+ virtual int SetRenderTimeoutImage(const int video_channel,
+ const char* file_nameUTF8,
+ const unsigned int timeout_ms);
+ virtual int SetRenderTimeoutImage(const int video_channel,
+ const ViEPicture& picture,
+ const unsigned int timeout_ms);
+
+ protected:
+ ViEFileImpl(ViESharedData* shared_data);
+ virtual ~ViEFileImpl();
+
+ private:
+ WebRtc_Word32 GetNextCapturedFrame(WebRtc_Word32 capture_id,
+ VideoFrame& video_frame);
+
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_file_player.cc b/trunk/src/video_engine/vie_file_player.cc
new file mode 100644
index 0000000..e40f337
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_player.cc
@@ -0,0 +1,508 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_file_player.h"
+
+#include "modules/utility/interface/file_player.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/thread_wrapper.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_file.h"
+#include "video_engine/vie_input_manager.h"
+#include "voice_engine/main/interface/voe_base.h"
+#include "voice_engine/main/interface/voe_file.h"
+#include "voice_engine/main/interface/voe_video_sync.h"
+
+namespace webrtc {
+
+const int kThreadWaitTimeMs = 100;
+
+ViEFilePlayer* ViEFilePlayer::CreateViEFilePlayer(
+ int file_id,
+ int engine_id,
+ const char* file_nameUTF8,
+ const bool loop,
+ const FileFormats file_format,
+ ViEInputManager& input_manager,
+ VoiceEngine* voe_ptr) {
+ ViEFilePlayer* self = new ViEFilePlayer(file_id, engine_id, input_manager);
+ if (!self || self->Init(file_nameUTF8, loop, file_format, voe_ptr) != 0) {
+ delete self;
+ self = NULL;
+ }
+ return self;
+}
+
+ViEFilePlayer::ViEFilePlayer(int Id,
+ int engine_id,
+ ViEInputManager& input_manager)
+ : ViEFrameProviderBase(Id, engine_id),
+ play_back_started_(false),
+ input_manager_(input_manager),
+ feedback_cs_(NULL),
+ audio_cs_(NULL),
+ file_player_(NULL),
+ audio_stream_(false),
+ video_clients_(0),
+ audio_clients_(0),
+ local_audio_channel_(-1),
+ observer_(NULL),
+ voe_file_interface_(NULL),
+ voe_video_sync_(NULL),
+ decode_thread_(NULL),
+ decode_event_(NULL),
+ decoded_audio_length_(0) {
+ memset(file_name_, 0, FileWrapper::kMaxFileNameSize);
+ memset(decoded_audio_, 0, kMaxDecodedAudioLength);
+}
+
+ViEFilePlayer::~ViEFilePlayer() {
+ // StopPlay deletes decode_thread_.
+ StopPlay();
+ delete decode_event_;
+ delete audio_cs_;
+ delete feedback_cs_;
+}
+
+int ViEFilePlayer::Init(const char* file_nameUTF8,
+ const bool loop,
+ const FileFormats file_format,
+ VoiceEngine* voice_engine) {
+ feedback_cs_ = CriticalSectionWrapper::CreateCriticalSection();
+ if (!feedback_cs_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to allocate critsect");
+ return -1;
+ }
+
+ audio_cs_ = CriticalSectionWrapper::CreateCriticalSection();
+ if (!audio_cs_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to allocate critsect");
+ return -1;
+ }
+
+ decode_event_ = EventWrapper::Create();
+ if (!decode_event_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to allocate event");
+ return -1;
+ }
+ if (strlen(file_nameUTF8) > FileWrapper::kMaxFileNameSize) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() Too long filename");
+ return -1;
+ }
+ strncpy(file_name_, file_nameUTF8, strlen(file_nameUTF8) + 1);
+
+ file_player_ = FilePlayer::CreateFilePlayer(ViEId(engine_id_, id_),
+ file_format);
+ if (!file_player_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to create file player");
+ return -1;
+ }
+ if (file_player_->RegisterModuleFileCallback(this) == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to "
+ "RegisterModuleFileCallback");
+ file_player_ = NULL;
+ return -1;
+ }
+ decode_thread_ = ThreadWrapper::CreateThread(FilePlayDecodeThreadFunction,
+ this, kHighestPriority,
+ "ViEFilePlayThread");
+ if (!decode_thread_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to start decode thread.");
+ file_player_ = NULL;
+ return -1;
+ }
+
+ // Always try to open with Audio since we don't know on what channels the
+ // audio should be played on.
+ WebRtc_Word32 error = file_player_->StartPlayingVideoFile(file_name_, loop,
+ false);
+ if (error) {
+ // Failed to open the file with audio, try without.
+ error = file_player_->StartPlayingVideoFile(file_name_, loop, true);
+ audio_stream_ = false;
+ if (error) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to Start play video "
+ "file");
+ return -1;
+ }
+
+ } else {
+ audio_stream_ = true;
+ }
+
+ if (audio_stream_) {
+ if (voice_engine) {
+ // A VoiceEngine has been provided and we want to play audio on local
+ // a channel.
+ voe_file_interface_ = VoEFile::GetInterface(voice_engine);
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to get VEFile "
+ "interface");
+ return -1;
+ }
+ voe_video_sync_ = VoEVideoSync::GetInterface(voice_engine);
+ if (!voe_video_sync_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() failed to get "
+ "VoEVideoSync interface");
+ return -1;
+ }
+ }
+ }
+
+ // Read audio /(or just video) every 10ms.
+ decode_event_->StartTimer(true, 10);
+ return 0;
+}
+
+int ViEFilePlayer::FrameCallbackChanged() {
+ // Starts the decode thread when someone cares.
+ if (ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks() >
+ video_clients_) {
+ if (!play_back_started_) {
+ play_back_started_ = true;
+ unsigned int thread_id;
+ if (decode_thread_->Start(thread_id)) {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::FrameCallbackChanged() Started file decode"
+ " thread %u", thread_id);
+ } else {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::FrameCallbackChanged() Failed to start "
+ "file decode thread.");
+ }
+ } else if (!file_player_->IsPlayingFile()) {
+ if (file_player_->StartPlayingVideoFile(file_name_, false,
+ !audio_stream_) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::FrameCallbackChanged(), Failed to restart "
+ "the file player.");
+ }
+ }
+ }
+ video_clients_ = ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks();
+ return 0;
+}
+
+bool ViEFilePlayer::FilePlayDecodeThreadFunction(void* obj) {
+ return static_cast<ViEFilePlayer*>(obj)->FilePlayDecodeProcess();
+}
+
+bool ViEFilePlayer::FilePlayDecodeProcess() {
+ if (decode_event_->Wait(kThreadWaitTimeMs) == kEventSignaled) {
+ if (audio_stream_ && audio_clients_ == 0) {
+ // There is audio but no one cares, read the audio here.
+ Read(NULL, 0);
+ }
+ if (file_player_->TimeUntilNextVideoFrame() < 10) {
+ // Less than 10ms to next videoframe.
+ if (file_player_->GetVideoFromFile(decoded_video_) != 0) {
+ }
+ }
+ if (decoded_video_.Length() > 0) {
+ if (local_audio_channel_ != -1 && voe_video_sync_) {
+ // We are playing audio locally.
+ int audio_delay = 0;
+ if (voe_video_sync_->GetPlayoutBufferSize(audio_delay) == 0) {
+ decoded_video_.SetRenderTime(decoded_video_.RenderTimeMs() +
+ audio_delay);
+ }
+ }
+ DeliverFrame(decoded_video_);
+ decoded_video_.SetLength(0);
+ }
+ }
+ return true;
+}
+
+int ViEFilePlayer::StopPlay() {
+ // Only called from destructor.
+ if (decode_thread_) {
+ decode_thread_->SetNotAlive();
+ if (decode_thread_->Stop()) {
+ delete decode_thread_;
+ } else {
+ assert(!"ViEFilePlayer::StopPlay() Failed to stop decode thread");
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StartPlay() Failed to stop file decode "
+ "thread.");
+ }
+ }
+ decode_thread_ = NULL;
+ if (decode_event_) {
+ decode_event_->StopTimer();
+ }
+ StopPlayAudio();
+
+ if (voe_file_interface_) {
+ voe_file_interface_->Release();
+ voe_file_interface_ = NULL;
+ }
+ if (voe_video_sync_) {
+ voe_video_sync_->Release();
+ voe_video_sync_ = NULL;
+ }
+
+ if (file_player_) {
+ file_player_->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(file_player_);
+ file_player_ = NULL;
+ }
+ return 0;
+}
+
+int ViEFilePlayer::StopPlayAudio() {
+ // Stop sending audio.
+
+ std::set<int>::iterator it = audio_channels_sending_.begin();
+ while (it != audio_channels_sending_.end()) {
+ StopSendAudioOnChannel(*it);
+ // StopSendAudioOnChannel erases the item from the map.
+ it = audio_channels_sending_.begin();
+ }
+
+ // Stop local audio playback.
+ if (local_audio_channel_ != -1) {
+ StopPlayAudioLocally(local_audio_channel_);
+ }
+ local_audio_channel_ = -1;
+ audio_channel_buffers_.clear();
+ audio_clients_ = 0;
+ return 0;
+}
+
+int ViEFilePlayer::Read(void* buf, int len) {
+ // Protect from simultaneous reading from multiple channels.
+ CriticalSectionScoped lock(*audio_cs_);
+ if (NeedsAudioFromFile(buf)) {
+ // We will run the VoE in 16KHz.
+ if (file_player_->Get10msAudioFromFile(decoded_audio_,
+ decoded_audio_length_, 16000) != 0) {
+ // No data.
+ decoded_audio_length_ = 0;
+ return 0;
+ }
+ // 2 bytes per sample.
+ decoded_audio_length_ *= 2;
+ if (buf) {
+ audio_channel_buffers_.push_back(buf);
+ }
+ } else {
+ // No need for new audiobuffer from file, ie the buffer read from file has
+ // not been played on this channel.
+ }
+ if (buf) {
+ memcpy(buf, decoded_audio_, decoded_audio_length_);
+ }
+ return decoded_audio_length_;
+}
+
+bool ViEFilePlayer::NeedsAudioFromFile(void* buf) {
+ bool needs_new_audio = false;
+ if (audio_channel_buffers_.size() == 0) {
+ return true;
+ }
+
+ // Check if we the buf already have read the current audio.
+ for (std::list<void*>::iterator it = audio_channel_buffers_.begin();
+ it != audio_channel_buffers_.end(); ++it) {
+ if (*it == buf) {
+ needs_new_audio = true;
+ audio_channel_buffers_.erase(it);
+ break;
+ }
+ }
+ return needs_new_audio;
+}
+
+void ViEFilePlayer::PlayFileEnded(const WebRtc_Word32 id) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id),
+ "%s: file_id %d", __FUNCTION__, id_);
+ file_player_->StopPlayingFile();
+
+ CriticalSectionScoped lock(*feedback_cs_);
+ if (observer_) {
+ observer_->PlayFileEnded(id_);
+ }
+}
+
+bool ViEFilePlayer::IsObserverRegistered() {
+ CriticalSectionScoped lock(*feedback_cs_);
+ return observer_ != NULL;
+}
+
+int ViEFilePlayer::RegisterObserver(ViEFileObserver& observer) {
+ CriticalSectionScoped lock(*feedback_cs_);
+ if (observer_) {
+ return -1;
+ }
+ observer_ = &observer;
+ return 0;
+}
+
+int ViEFilePlayer::DeRegisterObserver() {
+ CriticalSectionScoped lock(*feedback_cs_);
+ observer_ = NULL;
+ return 0;
+}
+
+int ViEFilePlayer::SendAudioOnChannel(const int audio_channel,
+ bool mix_microphone,
+ float volume_scaling) {
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "%s No VEFile interface.", __FUNCTION__);
+ return -1;
+ }
+ if (voe_file_interface_->StartPlayingFileAsMicrophone(audio_channel, this,
+ mix_microphone,
+ kFileFormatPcm16kHzFile,
+ volume_scaling) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::SendAudioOnChannel() "
+ "VE_StartPlayingFileAsMicrophone failed. audio_channel %d, "
+ " mix_microphone %d, volume_scaling %.2f",
+ audio_channel, mix_microphone, volume_scaling);
+ return -1;
+ }
+ audio_channels_sending_.insert(audio_channel);
+
+ CriticalSectionScoped lock(*audio_cs_);
+ audio_clients_++;
+ return 0;
+}
+
+int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) {
+ int result = 0;
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StopSendAudioOnChannel() - no VoE interface");
+ return -1;
+ }
+ std::set<int>::iterator it = audio_channels_sending_.find(audio_channel);
+ if (it == audio_channels_sending_.end()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StopSendAudioOnChannel AudioChannel %d not "
+ "sending", audio_channel);
+ return -1;
+ }
+ result = voe_file_interface_->StopPlayingFileAsMicrophone(audio_channel);
+ if (result != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "ViEFilePlayer::StopSendAudioOnChannel() "
+ "VE_StopPlayingFileAsMicrophone failed. audio_channel %d",
+ audio_channel);
+ }
+ audio_channels_sending_.erase(audio_channel);
+ CriticalSectionScoped lock(*audio_cs_);
+ audio_clients_--;
+ assert(audio_clients_ >= 0);
+ return 0;
+}
+
+int ViEFilePlayer::PlayAudioLocally(const int audio_channel,
+ float volume_scaling) {
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "%s No VEFile interface.", __FUNCTION__);
+ return -1;
+ }
+ if (voe_file_interface_->StartPlayingFileLocally(audio_channel, this,
+ kFileFormatPcm16kHzFile,
+ volume_scaling) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "%s VE_StartPlayingFileAsMicrophone failed. audio_channel %d,"
+ " mix_microphone %d, volume_scaling %.2f",
+ __FUNCTION__, audio_channel, volume_scaling);
+ return -1;
+ }
+
+ CriticalSectionScoped lock(*audio_cs_);
+ local_audio_channel_ = audio_channel;
+ audio_clients_++;
+ return 0;
+}
+
+int ViEFilePlayer::StopPlayAudioLocally(const int audio_channel) {
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "%s No VEFile interface.", __FUNCTION__);
+ return -1;
+ }
+ if (voe_file_interface_->StopPlayingFileLocally(audio_channel) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
+ "%s VE_StopPlayingFileLocally failed. audio_channel %d.",
+ __FUNCTION__, audio_channel);
+ return -1;
+ }
+
+ CriticalSectionScoped lock(*audio_cs_);
+ local_audio_channel_ = -1;
+ audio_clients_--;
+ return 0;
+}
+
+int ViEFilePlayer::GetFileInformation(int engine_id,
+ const char* file_name,
+ VideoCodec& video_codec,
+ CodecInst& audio_codec,
+ const FileFormats file_format) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, engine_id, "%s ", __FUNCTION__);
+
+ FilePlayer* file_player = FilePlayer::CreateFilePlayer(engine_id,
+ file_format);
+ if (!file_player) {
+ return -1;
+ }
+
+ bool video_only = false;
+
+ memset(&video_codec, 0, sizeof(video_codec));
+ memset(&audio_codec, 0, sizeof(audio_codec));
+
+ if (file_player->StartPlayingVideoFile(file_name, false, false) != 0) {
+ video_only = true;
+ if (file_player->StartPlayingVideoFile(file_name, false, true) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s Failed to open file.", __FUNCTION__);
+ FilePlayer::DestroyFilePlayer(file_player);
+ return -1;
+ }
+ }
+
+ if (!video_only && file_player->AudioCodec(audio_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s Failed to get audio codec.", __FUNCTION__);
+ FilePlayer::DestroyFilePlayer(file_player);
+ return -1;
+ }
+ if (file_player->video_codec_info(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
+ "%s Failed to get video codec.", __FUNCTION__);
+ FilePlayer::DestroyFilePlayer(file_player);
+ return -1;
+ }
+ FilePlayer::DestroyFilePlayer(file_player);
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_player.h b/trunk/src/video_engine/vie_file_player.h
new file mode 100644
index 0000000..640b37c
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_player.h
@@ -0,0 +1,141 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
+
+#include <list>
+#include <set>
+
+#include "common_types.h"
+#include "modules/media_file/interface/media_file_defines.h"
+#include "system_wrappers/interface/file_wrapper.h"
+#include "typedefs.h"
+#include "video_engine/vie_frame_provider_base.h"
+
+namespace webrtc {
+
+class EventWrapper;
+class FilePlayer;
+class ThreadWrapper;
+class ViEFileObserver;
+class ViEInputManager;
+class VoEFile;
+class VoEVideoSync;
+class VoiceEngine;
+
+class ViEFilePlayer
+ : public ViEFrameProviderBase,
+ protected FileCallback,
+ protected InStream {
+ public:
+ static ViEFilePlayer* CreateViEFilePlayer(int file_id,
+ int engine_id,
+ const char* file_nameUTF8,
+ const bool loop,
+ const FileFormats file_format,
+ ViEInputManager& input_manager,
+ VoiceEngine* voe_ptr);
+
+ static int GetFileInformation(const int engine_id,
+ const char* file_name,
+ VideoCodec& video_codec,
+ CodecInst& audio_codec,
+ const FileFormats file_format);
+ ~ViEFilePlayer();
+
+ bool IsObserverRegistered();
+ int RegisterObserver(ViEFileObserver& observer);
+ int DeRegisterObserver();
+ int SendAudioOnChannel(const int audio_channel,
+ bool mix_microphone,
+ float volume_scaling);
+ int StopSendAudioOnChannel(const int audio_channel);
+ int PlayAudioLocally(const int audio_channel, float volume_scaling);
+ int StopPlayAudioLocally(const int audio_channel);
+
+ // Implements ViEFrameProviderBase.
+ virtual int FrameCallbackChanged();
+
+ protected:
+ ViEFilePlayer(int Id, int engine_id, ViEInputManager& input_manager);
+ int Init(const WebRtc_Word8* file_nameUTF8,
+ const bool loop,
+ const FileFormats file_format,
+ VoiceEngine* voe_ptr);
+ int StopPlay();
+ int StopPlayAudio();
+
+ // File play decode function.
+ static bool FilePlayDecodeThreadFunction(void* obj);
+ bool FilePlayDecodeProcess();
+ bool NeedsAudioFromFile(void* buf);
+
+ // Implements webrtc::InStream.
+ virtual int Read(void* buf, int len);
+ virtual int Rewind() {
+ return 0;
+ }
+
+ // Implements FileCallback.
+ virtual void PlayNotification(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord32 /*notification_ms*/) {}
+ virtual void RecordNotification(const WebRtc_Word32 /*id*/,
+ const WebRtc_UWord32 /*notification_ms*/) {}
+ virtual void PlayFileEnded(const WebRtc_Word32 id);
+ virtual void RecordFileEnded(const WebRtc_Word32 /*id*/) {}
+
+ private:
+ static const int kMaxDecodedAudioLength = 320;
+ bool play_back_started_;
+ ViEInputManager& input_manager_;
+
+ CriticalSectionWrapper* feedback_cs_;
+ CriticalSectionWrapper* audio_cs_;
+
+ FilePlayer* file_player_;
+ bool audio_stream_;
+
+ // Number of active video clients.
+ int video_clients_;
+
+ // Number of audio channels sending this audio.
+ int audio_clients_;
+
+ // Local audio channel playing this video. Sync video against this.
+ int local_audio_channel_;
+
+ ViEFileObserver* observer_;
+ WebRtc_Word8 file_name_[FileWrapper::kMaxFileNameSize];
+
+ // VoE Interface.
+ VoEFile* voe_file_interface_;
+ VoEVideoSync* voe_video_sync_;
+
+ // Thread for decoding video (and audio if no audio clients connected).
+ ThreadWrapper* decode_thread_;
+ EventWrapper* decode_event_;
+ WebRtc_Word16 decoded_audio_[kMaxDecodedAudioLength];
+ WebRtc_UWord32 decoded_audio_length_;
+
+ // Trick - list containing VoE buffer reading this file. Used if multiple
+ // audio channels are sending.
+ std::list<void*> audio_channel_buffers_;
+
+ // AudioChannels sending audio from this file.
+ std::set<int> audio_channels_sending_;
+
+ // Frame receiving decoded video from file.
+ VideoFrame decoded_video_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
diff --git a/trunk/src/video_engine/vie_file_recorder.cc b/trunk/src/video_engine/vie_file_recorder.cc
new file mode 100644
index 0000000..c2f8fa0
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_recorder.cc
@@ -0,0 +1,239 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_file_recorder.h"
+
+#include "modules/utility/interface/file_player.h"
+#include "modules/utility/interface/file_recorder.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/vie_defines.h"
+
+namespace webrtc {
+
+ViEFileRecorder::ViEFileRecorder(int instanceID)
+ : recorder_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ file_recorder_(NULL),
+ is_first_frame_recorded_(false),
+ is_out_stream_started_(false),
+ instance_id_(instanceID),
+ frame_delay_(0),
+ audio_channel_(-1),
+ audio_source_(NO_AUDIO),
+ voe_file_interface_(NULL) {
+}
+
+ViEFileRecorder::~ViEFileRecorder() {
+ StopRecording();
+ delete recorder_cs_;
+}
+
+int ViEFileRecorder::StartRecording(const char* file_nameUTF8,
+ const VideoCodec& codec_inst,
+ AudioSource audio_source,
+ int audio_channel,
+ const CodecInst& audio_codec_inst,
+ VoiceEngine* voe_ptr,
+ const FileFormats file_format) {
+ CriticalSectionScoped lock(*recorder_cs_);
+
+ if (file_recorder_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StartRecording() - already recording.");
+ return -1;
+ }
+ file_recorder_ = FileRecorder::CreateFileRecorder(instance_id_, file_format);
+ if (!file_recorder_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StartRecording() failed to create recoder.");
+ return -1;
+ }
+
+ int error = file_recorder_->StartRecordingVideoFile(file_nameUTF8,
+ audio_codec_inst,
+ codec_inst,
+ AMRFileStorage,
+ audio_source == NO_AUDIO);
+ if (error) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StartRecording() failed to "
+ "StartRecordingVideoFile.");
+ FileRecorder::DestroyFileRecorder(file_recorder_);
+ file_recorder_ = NULL;
+ return -1;
+ }
+
+ audio_source_ = audio_source;
+ if (voe_ptr && audio_source != NO_AUDIO) {
+ // VoE interface has been provided and we want to record audio.
+ voe_file_interface_ = VoEFile::GetInterface(voe_ptr);
+ if (!voe_file_interface_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StartRecording() failed to get VEFile "
+ "interface");
+ return -1;
+ }
+
+ // Always L16.
+ CodecInst engine_audio_codec_inst = {96, "L16", audio_codec_inst.plfreq,
+ audio_codec_inst.plfreq / 100, 1,
+ audio_codec_inst.plfreq * 16 };
+
+ switch (audio_source) {
+ // case NO_AUDIO is checked above.
+ case MICROPHONE:
+ error = voe_file_interface_->StartRecordingMicrophone(
+ this, &engine_audio_codec_inst);
+ break;
+ case PLAYOUT:
+ error = voe_file_interface_->StartRecordingPlayout(
+ audio_channel, this, &engine_audio_codec_inst);
+ break;
+ default:
+ assert(!"Unknown audio_source");
+ }
+ if (error != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StartRecording() failed to start recording"
+ " audio");
+ FileRecorder::DestroyFileRecorder(file_recorder_);
+ file_recorder_ = NULL;
+ return -1;
+ }
+ is_out_stream_started_ = true;
+ audio_channel_ = audio_channel;
+ }
+ is_first_frame_recorded_ = false;
+ return 0;
+}
+
+int ViEFileRecorder::StopRecording() {
+ int error = 0;
+ // We can not hold the ptr_cs_ while accessing VoE functions. It might cause
+ // deadlock in Write.
+ if (voe_file_interface_) {
+ switch (audio_source_) {
+ case MICROPHONE:
+ error = voe_file_interface_->StopRecordingMicrophone();
+ break;
+ case PLAYOUT:
+ error = voe_file_interface_->StopRecordingPlayout(audio_channel_);
+ break;
+ case NO_AUDIO:
+ break;
+ default:
+ assert(!"Unknown audio_source");
+ }
+ if (error != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
+ "ViEFileRecorder::StopRecording() failed to stop recording "
+ "audio");
+ }
+ }
+ CriticalSectionScoped lock(*recorder_cs_);
+ if (voe_file_interface_) {
+ voe_file_interface_->Release();
+ voe_file_interface_ = NULL;
+ }
+
+ if (file_recorder_) {
+ if (file_recorder_->IsRecording()) {
+ int error = file_recorder_->StopRecording();
+ if (error) {
+ return -1;
+ }
+ }
+ FileRecorder::DestroyFileRecorder(file_recorder_);
+ file_recorder_ = NULL;
+ }
+ is_first_frame_recorded_ = false;
+ is_out_stream_started_ = false;
+ return 0;
+}
+
+void ViEFileRecorder::SetFrameDelay(int frame_delay) {
+ CriticalSectionScoped lock(*recorder_cs_);
+ frame_delay_ = frame_delay;
+}
+
+bool ViEFileRecorder::RecordingStarted() {
+ CriticalSectionScoped lock(*recorder_cs_);
+ return file_recorder_ && file_recorder_->IsRecording();
+}
+
+bool ViEFileRecorder::FirstFrameRecorded() {
+ CriticalSectionScoped lock(*recorder_cs_);
+ return is_first_frame_recorded_;
+}
+
+bool ViEFileRecorder::IsRecordingFileFormat(const FileFormats file_format) {
+ CriticalSectionScoped lock(*recorder_cs_);
+ return (file_recorder_->RecordingFileFormat() == file_format) ? true : false;
+}
+
+void ViEFileRecorder::RecordVideoFrame(const VideoFrame& video_frame) {
+ CriticalSectionScoped lock(*recorder_cs_);
+
+ if (file_recorder_ && file_recorder_->IsRecording()) {
+ if (!IsRecordingFileFormat(kFileFormatAviFile))
+ return;
+
+ // Compensate for frame delay in order to get audio/video sync when
+ // recording local video.
+ const WebRtc_UWord32 time_stamp = video_frame.TimeStamp();
+ const WebRtc_Word64 render_time_stamp = video_frame.RenderTimeMs();
+ VideoFrame& unconst_video_frame = const_cast<VideoFrame&>(video_frame);
+ unconst_video_frame.SetTimeStamp(time_stamp - 90 * frame_delay_);
+ unconst_video_frame.SetRenderTime(render_time_stamp - frame_delay_);
+
+ file_recorder_->RecordVideoToFile(unconst_video_frame);
+
+ unconst_video_frame.SetRenderTime(render_time_stamp);
+ unconst_video_frame.SetTimeStamp(time_stamp);
+ }
+}
+
+bool ViEFileRecorder::Write(const void* buf, int len) {
+ if (!is_out_stream_started_)
+ return true;
+
+ // Always 10 ms L16 from VoE.
+ if (len % (2 * 80)) {
+ // Not 2 bytes 80 samples.
+ WEBRTC_TRACE(kTraceError, kTraceVideo, audio_channel_,
+ "Audio length not supported: %d.", len);
+ return true;
+ }
+
+ AudioFrame audio_frame;
+ WebRtc_UWord16 length_in_samples = len / 2;
+ audio_frame.UpdateFrame(audio_channel_, 0,
+ static_cast<const WebRtc_Word16*>(buf),
+ length_in_samples, length_in_samples * 100,
+ AudioFrame::kUndefined,
+ AudioFrame::kVadUnknown);
+
+ CriticalSectionScoped lock(*recorder_cs_);
+ if (file_recorder_ && file_recorder_->IsRecording()) {
+ TickTime tick_time = TickTime::Now();
+ file_recorder_->RecordAudioToFile(audio_frame, &tick_time);
+ }
+
+ // Always return true to continue recording.
+ return true;
+}
+
+int ViEFileRecorder::Rewind() {
+ // Not supported!
+ return -1;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_recorder.h b/trunk/src/video_engine/vie_file_recorder.h
new file mode 100644
index 0000000..1d34ad3
--- /dev/null
+++ b/trunk/src/video_engine/vie_file_recorder.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
+
+#include "modules/utility/interface/file_recorder.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_file.h"
+#include "voice_engine/main/interface/voe_file.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+
+class ViEFileRecorder : protected OutStream {
+ public:
+ explicit ViEFileRecorder(int channel_id);
+ ~ViEFileRecorder();
+
+ int StartRecording(const char* file_nameUTF8,
+ const VideoCodec& codec_inst,
+ AudioSource audio_source, int audio_channel,
+ const CodecInst& audio_codec_inst,
+ VoiceEngine* voe_ptr,
+ const FileFormats file_format = kFileFormatAviFile);
+ int StopRecording();
+
+ void SetFrameDelay(int frame_delay);
+ bool RecordingStarted();
+
+ // Records incoming decoded video frame to file.
+ void RecordVideoFrame(const VideoFrame& video_frame);
+
+ protected:
+ bool FirstFrameRecorded();
+ bool IsRecordingFileFormat(const FileFormats file_format);
+
+ // Implements OutStream.
+ bool Write(const void* buf, int len);
+ int Rewind();
+
+ private:
+ CriticalSectionWrapper* recorder_cs_;
+
+ FileRecorder* file_recorder_;
+ bool is_first_frame_recorded_;
+ bool is_out_stream_started_;
+ int instance_id_;
+ int frame_delay_;
+ int audio_channel_;
+ AudioSource audio_source_;
+ VoEFile* voe_file_interface_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
diff --git a/trunk/src/video_engine/vie_frame_provider_base.cc b/trunk/src/video_engine/vie_frame_provider_base.cc
new file mode 100644
index 0000000..8b7bf1f
--- /dev/null
+++ b/trunk/src/video_engine/vie_frame_provider_base.cc
@@ -0,0 +1,197 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_frame_provider_base.h"
+
+#include <algorithm>
+
+#include "modules/interface/module_common_types.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/vie_defines.h"
+
+namespace webrtc {
+
+ViEFrameProviderBase::ViEFrameProviderBase(int Id, int engine_id)
+ : id_(Id),
+ engine_id_(engine_id),
+ provider_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ frame_delay_(0) {
+}
+
+ViEFrameProviderBase::~ViEFrameProviderBase() {
+ if (frame_callbacks_.size() > 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
+ "FrameCallbacks still exist when Provider deleted %d",
+ frame_callbacks_.size());
+ }
+
+ for (FrameCallbacks::iterator it = frame_callbacks_.begin();
+ it != frame_callbacks_.end(); ++it) {
+ (*it)->ProviderDestroyed(id_);
+ }
+ frame_callbacks_.clear();
+}
+
+int ViEFrameProviderBase::Id() {
+ return id_;
+}
+
+void ViEFrameProviderBase::DeliverFrame(
+ VideoFrame& video_frame,
+ int num_csrcs,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
+#ifdef DEBUG_
+ const TickTime start_process_time = TickTime::Now();
+#endif
+ CriticalSectionScoped cs(provider_cs_.get());
+
+ // Deliver the frame to all registered callbacks.
+ if (frame_callbacks_.size() > 0) {
+ if (frame_callbacks_.size() == 1) {
+ // We don't have to copy the frame.
+ frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC);
+ } else {
+ // Make a copy of the frame for all callbacks.callback
+ for (FrameCallbacks::iterator it = frame_callbacks_.begin();
+ it != frame_callbacks_.end(); ++it) {
+ if (!extra_frame_.get()) {
+ extra_frame_.reset(new VideoFrame());
+ }
+ extra_frame_->CopyFrame(video_frame);
+ (*it)->DeliverFrame(id_, *(extra_frame_.get()), num_csrcs, CSRC);
+ }
+ }
+ }
+#ifdef DEBUG_
+ const int process_time =
+ static_cast<int>((TickTime::Now() - start_process_time).Milliseconds());
+ if (process_time > 25) {
+ // Warn if the delivery time is too long.
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
+ "%s Too long time: %ums", __FUNCTION__, process_time);
+ }
+#endif
+}
+
+void ViEFrameProviderBase::SetFrameDelay(int frame_delay) {
+ CriticalSectionScoped cs(provider_cs_.get());
+ frame_delay_ = frame_delay;
+
+ for (FrameCallbacks::iterator it = frame_callbacks_.begin();
+ it != frame_callbacks_.end(); ++it) {
+ (*it)->DelayChanged(id_, frame_delay);
+ }
+}
+
+int ViEFrameProviderBase::FrameDelay() {
+ return frame_delay_;
+}
+
+int ViEFrameProviderBase::GetBestFormat(int& best_width,
+ int& best_height,
+ int& best_frame_rate) {
+ int largest_width = 0;
+ int largest_height = 0;
+ int highest_frame_rate = 0;
+
+ CriticalSectionScoped cs(provider_cs_.get());
+ for (FrameCallbacks::iterator it = frame_callbacks_.begin();
+ it != frame_callbacks_.end(); ++it) {
+ int prefered_width = 0;
+ int prefered_height = 0;
+ int prefered_frame_rate = 0;
+ if ((*it)->GetPreferedFrameSettings(prefered_width, prefered_height,
+ prefered_frame_rate) == 0) {
+ if (prefered_width > largest_width) {
+ largest_width = prefered_width;
+ }
+ if (prefered_height > largest_height) {
+ largest_height = prefered_height;
+ }
+ if (prefered_frame_rate > highest_frame_rate) {
+ highest_frame_rate = prefered_frame_rate;
+ }
+ }
+ }
+ best_width = largest_width;
+ best_height = largest_height;
+ best_frame_rate = highest_frame_rate;
+ return 0;
+}
+
+int ViEFrameProviderBase::RegisterFrameCallback(
+ int observer_id, ViEFrameCallback* callback_object) {
+ assert(callback_object);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_), "%s(0x%p)",
+ __FUNCTION__, callback_object);
+ {
+ CriticalSectionScoped cs(provider_cs_.get());
+ if (std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
+ callback_object) != frame_callbacks_.end()) {
+ // This object is already registered.
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
+ "%s 0x%p already registered", __FUNCTION__,
+ callback_object);
+ assert("!frameObserver already registered");
+ return -1;
+ }
+ frame_callbacks_.push_back(callback_object);
+ }
+ // Report current capture delay.
+ callback_object->DelayChanged(id_, frame_delay_);
+
+ // Notify implementer of this class that the callback list have changed.
+ FrameCallbackChanged();
+ return 0;
+}
+
+int ViEFrameProviderBase::DeregisterFrameCallback(
+ const ViEFrameCallback* callback_object) {
+ assert(callback_object);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_), "%s(0x%p)",
+ __FUNCTION__, callback_object);
+ CriticalSectionScoped cs(provider_cs_.get());
+
+ FrameCallbacks::iterator it = std::find(frame_callbacks_.begin(),
+ frame_callbacks_.end(),
+ callback_object);
+ if (it == frame_callbacks_.end()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
+ "%s 0x%p not found", __FUNCTION__, callback_object);
+ return -1;
+ }
+ frame_callbacks_.erase(it);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_),
+ "%s 0x%p deregistered", __FUNCTION__, callback_object);
+
+ // Notify implementer of this class that the callback list have changed.
+ FrameCallbackChanged();
+ return 0;
+}
+
+bool ViEFrameProviderBase::IsFrameCallbackRegistered(
+ const ViEFrameCallback* callback_object) {
+ assert(callback_object);
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_),
+ "%s(0x%p)", __FUNCTION__, callback_object);
+
+ CriticalSectionScoped cs(provider_cs_.get());
+ return std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
+ callback_object) != frame_callbacks_.end();
+}
+
+int ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks() {
+ CriticalSectionScoped cs(provider_cs_.get());
+ return frame_callbacks_.size();
+}
+
+} // namespac webrtc
diff --git a/trunk/src/video_engine/vie_frame_provider_base.h b/trunk/src/video_engine/vie_frame_provider_base.h
new file mode 100644
index 0000000..466e12e
--- /dev/null
+++ b/trunk/src/video_engine/vie_frame_provider_base.h
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
+
+#include <vector>
+
+#include "common_types.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class VideoEncoder;
+class VideoFrame;
+
+// ViEFrameCallback shall be implemented by all classes receiving frames from a
+// frame provider.
+class ViEFrameCallback {
+ public:
+ virtual void DeliverFrame(int id,
+ VideoFrame& video_frame,
+ int num_csrcs = 0,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL) = 0;
+
+ // The capture delay has changed from the provider. |frame_delay| is given in
+ // ms.
+ virtual void DelayChanged(int id, int frame_delay) = 0;
+
+ // Get the width, height and frame rate preferred by this observer.
+ virtual int GetPreferedFrameSettings(int& width,
+ int& height,
+ int& frame_rate) = 0;
+
+ // ProviderDestroyed is called when the frame is about to be destroyed. There
+ // must not be any more calls to the frame provider after this.
+ virtual void ProviderDestroyed(int id) = 0;
+
+ virtual ~ViEFrameCallback() {}
+};
+
+// ViEFrameProviderBase is a base class that will deliver frames to all
+// registered ViEFrameCallbacks.
+class ViEFrameProviderBase {
+ public:
+ ViEFrameProviderBase(int Id, int engine_id);
+ virtual ~ViEFrameProviderBase();
+
+ // Returns the frame provider id.
+ int Id();
+
+ // Register frame callbacks, i.e. a receiver of the captured frame.
+ virtual int RegisterFrameCallback(int observer_id,
+ ViEFrameCallback* callback_object);
+
+ virtual int DeregisterFrameCallback(const ViEFrameCallback* callback_object);
+
+ virtual bool IsFrameCallbackRegistered(
+ const ViEFrameCallback* callback_object);
+
+ int NumberOfRegisteredFrameCallbacks();
+
+ // FrameCallbackChanged
+ // Inherited classes should check for new frame_settings and reconfigure
+ // output if possible.
+ virtual int FrameCallbackChanged() = 0;
+
+ protected:
+ void DeliverFrame(VideoFrame& video_frame,
+ int num_csrcs = 0,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
+ void SetFrameDelay(int frame_delay);
+ int FrameDelay();
+ int GetBestFormat(int& best_width,
+ int& best_height,
+ int& best_frame_rate);
+
+ int id_;
+ int engine_id_;
+
+ // Frame callbacks.
+ typedef std::vector<ViEFrameCallback*> FrameCallbacks;
+ FrameCallbacks frame_callbacks_;
+ scoped_ptr<CriticalSectionWrapper> provider_cs_;
+
+ private:
+ scoped_ptr<VideoFrame> extra_frame_;
+ int frame_delay_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
diff --git a/trunk/src/video_engine/vie_image_process_impl.cc b/trunk/src/video_engine/vie_image_process_impl.cc
new file mode 100644
index 0000000..d9dfcd4
--- /dev/null
+++ b/trunk/src/video_engine/vie_image_process_impl.cc
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_image_process_impl.h"
+
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_input_manager.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViEImageProcess* ViEImageProcess::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViEImageProcessImpl* vie_image_process_impl = vie_impl;
+ // Increase ref count.
+ (*vie_image_process_impl)++;
+ return vie_image_process_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViEImageProcessImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViEImageProcess::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViEImageProcess release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViEImageProcess reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViEImageProcessImpl::ViEImageProcessImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEImageProcessImpl::ViEImageProcessImpl() Ctor");
+}
+
+ViEImageProcessImpl::~ViEImageProcessImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViEImageProcessImpl::~ViEImageProcessImpl() Dtor");
+}
+
+int ViEImageProcessImpl::RegisterCaptureEffectFilter(
+ const int capture_id,
+ ViEEffectFilter& capture_filter) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
+ return -1;
+ }
+ if (vie_capture->RegisterEffectFilter(&capture_filter) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterExists);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::DeregisterCaptureEffectFilter(const int capture_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
+ return -1;
+ }
+ if (vie_capture->RegisterEffectFilter(NULL) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::RegisterSendEffectFilter(
+ const int video_channel,
+ ViEEffectFilter& send_filter) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (vie_encoder == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_encoder->RegisterEffectFilter(&send_filter) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterExists);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::DeregisterSendEffectFilter(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (vie_encoder == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+ if (vie_encoder->RegisterEffectFilter(NULL) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::RegisterRenderEffectFilter(
+ const int video_channel,
+ ViEEffectFilter& render_filter) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterEffectFilter(&render_filter) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterExists);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::DeregisterRenderEffectFilter(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->RegisterEffectFilter(NULL) != 0) {
+ shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::EnableDeflickering(const int capture_id,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d, enable: %d)", __FUNCTION__, capture_id,
+ enable);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_capture->EnableDeflickering(enable) != 0) {
+ if (enable) {
+ shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
+ } else {
+ shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
+ }
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::EnableDenoising(const int capture_id,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(capture_id: %d, enable: %d)", __FUNCTION__, capture_id,
+ enable);
+
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViECapturer* vie_capture = is.Capture(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Capture device %d doesn't exist", __FUNCTION__,
+ capture_id);
+ shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
+ return -1;
+ }
+
+ if (vie_capture->EnableDenoising(enable) != 0) {
+ if (enable) {
+ shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
+ } else {
+ shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
+ }
+ return -1;
+ }
+ return 0;
+}
+
+int ViEImageProcessImpl::EnableColorEnhancement(const int video_channel,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(video_channel: %d, enable: %d)", __FUNCTION__, video_channel,
+ enable);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->EnableColorEnhancement(enable) != 0) {
+ if (enable) {
+ shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
+ } else {
+ shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
+ }
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_image_process_impl.h b/trunk/src/video_engine/vie_image_process_impl.h
new file mode 100644
index 0000000..169c9e7
--- /dev/null
+++ b/trunk/src/video_engine/vie_image_process_impl.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
+
+#include "typedefs.h"
+#include "video_engine/include/vie_image_process.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViEImageProcessImpl
+ : public ViEImageProcess,
+ public ViERefCount {
+ public:
+ // Implements ViEImageProcess.
+ virtual int Release();
+ virtual int RegisterCaptureEffectFilter(const int capture_id,
+ ViEEffectFilter& capture_filter);
+ virtual int DeregisterCaptureEffectFilter(const int capture_id);
+ virtual int RegisterSendEffectFilter(const int video_channel,
+ ViEEffectFilter& send_filter);
+ virtual int DeregisterSendEffectFilter(const int video_channel);
+ virtual int RegisterRenderEffectFilter(const int video_channel,
+ ViEEffectFilter& render_filter);
+ virtual int DeregisterRenderEffectFilter(const int video_channel);
+ virtual int EnableDeflickering(const int capture_id, const bool enable);
+ virtual int EnableDenoising(const int capture_id, const bool enable);
+ virtual int EnableColorEnhancement(const int video_channel,
+ const bool enable);
+
+ protected:
+ ViEImageProcessImpl(ViESharedData* shared_data);
+ virtual ~ViEImageProcessImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
diff --git a/trunk/src/video_engine/vie_impl.cc b/trunk/src/video_engine/vie_impl.cc
new file mode 100644
index 0000000..d8a421b
--- /dev/null
+++ b/trunk/src/video_engine/vie_impl.cc
@@ -0,0 +1,232 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_impl.h"
+
+#if (defined(WIN32_) || defined(WIN64_))
+#include <Windows.h> // For LoadLibrary.
+#include <tchar.h> // For T_.
+#endif
+
+#include "system_wrappers/interface/trace.h"
+
+#ifdef WEBRTC_ANDROID
+#include "modules/video_capture/main/interface/video_capture_factory.h"
+#include "modules/video_render/main/interface/video_render.h"
+#endif
+
+// Global counter to get an id for each new ViE instance.
+static WebRtc_Word32 g_vie_active_instance_counter = 0;
+
+namespace webrtc {
+
+// extern "C" ensures that GetProcAddress() can find the function address.
+extern "C" {
+ VideoEngine* GetVideoEngine() {
+ VideoEngineImpl* self = new VideoEngineImpl();
+ if (!self) {
+ return NULL;
+ }
+ g_vie_active_instance_counter++;
+ VideoEngine* vie = reinterpret_cast<VideoEngine*>(self);
+ return vie;
+ }
+}
+
+VideoEngine* VideoEngine::Create() {
+#if (defined(WIN32_) || defined(WIN64_))
+ // Load a debug dll, if there is one.
+ HMODULE hmod_ = LoadLibrary(TEXT("VideoEngineTestingDLL.dll"));
+ if (hmod_) {
+ typedef VideoEngine* (*PFNGetVideoEngineLib)(void);
+ PFNGetVideoEngineLib pfn =
+ (PFNGetVideoEngineLib)GetProcAddress(hmod_, "GetVideoEngine");
+ if (pfn) {
+ VideoEngine* self = pfn();
+ return self;
+ } else {
+ assert(!"Failed to open test dll VideoEngineTestingDLL.dll");
+ return NULL;
+ }
+ }
+#endif
+
+ return GetVideoEngine();
+}
+
+bool VideoEngine::Delete(VideoEngine*& video_engine) {
+ if (!video_engine) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "VideoEngine::Delete - No argument");
+ return false;
+ }
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "VideoEngine::Delete(vie = 0x%p)", video_engine);
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+
+ // Check all reference counters.
+ ViEBaseImpl* vie_base = vie_impl;
+ if (vie_base->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViEBase ref count: %d", vie_base->GetCount());
+ return false;
+ }
+#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
+ ViECaptureImpl* vie_capture = vie_impl;
+ if (vie_capture->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViECapture ref count: %d", vie_capture->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
+ ViECodecImpl* vie_codec = vie_impl;
+ if (vie_codec->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViECodec ref count: %d", vie_codec->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+ ViEEncryptionImpl* vie_encryption = vie_impl;
+ if (vie_encryption->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViEEncryption ref count: %d", vie_encryption->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+ ViEExternalCodecImpl* vie_external_codec = vie_impl;
+ if (vie_external_codec->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViEEncryption ref count: %d", vie_encryption->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+ ViEFileImpl* vie_file = vie_impl;
+ if (vie_file->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViEFile ref count: %d", vie_file->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+ ViEImageProcessImpl* vie_image_process = vie_impl;
+ if (vie_image_process->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViEImageProcess ref count: %d",
+ vie_image_process->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
+ ViENetworkImpl* vie_network = vie_impl;
+ if (vie_network->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViENetwork ref count: %d", vie_network->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
+ ViERenderImpl* vie_render = vie_impl;
+ if (vie_render->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViERender ref count: %d", vie_render->GetCount());
+ return false;
+ }
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+ ViERTP_RTCPImpl* vie_rtp_rtcp = vie_impl;
+ if (vie_rtp_rtcp->GetCount() > 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "ViERTP_RTCP ref count: %d", vie_rtp_rtcp->GetCount());
+ return false;
+ }
+#endif
+
+ delete vie_impl;
+ vie_impl = NULL;
+ video_engine = NULL;
+
+ // Decrease the number of instances.
+ g_vie_active_instance_counter--;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, g_vie_active_instance_counter,
+ "%s: instance deleted. Remaining instances: %d", __FUNCTION__,
+ g_vie_active_instance_counter);
+ return true;
+}
+
+int VideoEngine::SetTraceFile(const char* file_nameUTF8,
+ const bool add_file_counter) {
+ if (!file_nameUTF8) {
+ return -1;
+ }
+ if (Trace::SetTraceFile(file_nameUTF8, add_file_counter) == -1) {
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "SetTraceFileName(file_nameUTF8 = %s, add_file_counter = %d",
+ file_nameUTF8, add_file_counter);
+ return 0;
+}
+
+int VideoEngine::SetTraceFilter(const unsigned int filter) {
+ WebRtc_UWord32 old_filter = 0;
+ Trace::LevelFilter(old_filter);
+
+ if (filter == kTraceNone && old_filter != kTraceNone) {
+ // Do the logging before turning it off.
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "SetTraceFilter(filter = 0x%x)", filter);
+ }
+
+ WebRtc_Word32 error = Trace::SetLevelFilter(filter);
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "SetTraceFilter(filter = 0x%x)", filter);
+ if (error != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "SetTraceFilter error: %d", error);
+ return -1;
+ }
+ return 0;
+}
+
+int VideoEngine::SetTraceCallback(TraceCallback* callback) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "SetTraceCallback(TraceCallback = 0x%p)", callback);
+ return Trace::SetTraceCallback(callback);
+}
+
+int VideoEngine::SetAndroidObjects(void* javaVM, void* javaContext) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
+ "SetAndroidObjects()");
+
+#ifdef WEBRTC_ANDROID
+ if (VideoCaptureFactory::SetAndroidObjects(javaVM, javaContext) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "Could not set capture module Android objects");
+ return -1;
+ }
+ if (VideoRender::SetAndroidObjects(javaVM) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "Could not set render module Android objects");
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
+ "WEBRTC_ANDROID not defined for VideoEngine::SetAndroidObjects");
+ return -1;
+#endif
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_impl.h b/trunk/src/video_engine/vie_impl.h
new file mode 100644
index 0000000..3141868
--- /dev/null
+++ b/trunk/src/video_engine/vie_impl.h
@@ -0,0 +1,115 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
+
+#include "engine_configurations.h"
+#include "video_engine/vie_defines.h"
+
+#include "video_engine/vie_base_impl.h"
+
+#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
+#include "video_engine/vie_capture_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
+#include "video_engine/vie_codec_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+#include "video_engine/vie_encryption_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+#include "video_engine/vie_file_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+#include "video_engine/vie_image_process_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
+#include "video_engine/vie_network_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
+#include "video_engine/vie_render_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+#include "video_engine/vie_rtp_rtcp_impl.h"
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+#include "video_engine/vie_external_codec_impl.h"
+#endif
+
+namespace webrtc {
+
+class VideoEngineImpl
+ : public ViEBaseImpl
+#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
+ , public ViECodecImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
+ , public ViECaptureImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+ , public ViEEncryptionImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+ , public ViEFileImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+ , public ViEImageProcessImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
+ , public ViENetworkImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
+ , public ViERenderImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+ , public ViERTP_RTCPImpl
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+ , public ViEExternalCodecImpl
+#endif
+{ // NOLINT
+ public:
+ VideoEngineImpl()
+ :
+#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
+ ViECodecImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
+ , ViECaptureImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
+ , ViEEncryptionImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
+ , ViEFileImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+ , ViEImageProcessImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
+ , ViENetworkImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
+ , ViERenderImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+ , ViERTP_RTCPImpl(ViEBaseImpl::shared_data())
+#endif
+#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
+ , ViEExternalCodecImpl(ViEBaseImpl::shared_data())
+#endif
+ {}
+ virtual ~VideoEngineImpl() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_input_manager.cc b/trunk/src/video_engine/vie_input_manager.cc
new file mode 100644
index 0000000..98c5535
--- /dev/null
+++ b/trunk/src/video_engine/vie_input_manager.cc
@@ -0,0 +1,600 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_input_manager.h"
+
+#include <cassert>
+
+#include "common_types.h"
+#include "modules/video_capture/main/interface/video_capture_factory.h"
+#include "modules/video_coding/main/interface/video_coding.h"
+#include "modules/video_coding/main/interface/video_coding_defines.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/rw_lock_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_file_player.h"
+
+namespace webrtc {
+
+ViEInputManager::ViEInputManager(const int engine_id)
+ : engine_id_(engine_id),
+ map_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ vie_frame_provider_map_(),
+ capture_device_info_(NULL),
+ module_process_thread_(NULL) {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s", __FUNCTION__);
+
+ for (int idx = 0; idx < kViEMaxCaptureDevices; idx++) {
+ free_capture_device_id_[idx] = true;
+ }
+ capture_device_info_ = VideoCaptureFactory::CreateDeviceInfo(
+ ViEModuleId(engine_id_));
+ for (int idx = 0; idx < kViEMaxFilePlayers; idx++) {
+ free_file_id_[idx] = true;
+ }
+}
+
+ViEInputManager::~ViEInputManager() {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s", __FUNCTION__);
+ while (vie_frame_provider_map_.Size() != 0) {
+ MapItem* item = vie_frame_provider_map_.First();
+ assert(item);
+ ViEFrameProviderBase* frame_provider =
+ static_cast<ViEFrameProviderBase*>(item->GetItem());
+ vie_frame_provider_map_.Erase(item);
+ delete frame_provider;
+ }
+
+ if (capture_device_info_) {
+ delete capture_device_info_;
+ capture_device_info_ = NULL;
+ }
+}
+void ViEInputManager::SetModuleProcessThread(
+ ProcessThread& module_process_thread) {
+ assert(!module_process_thread_);
+ module_process_thread_ = &module_process_thread;
+}
+
+int ViEInputManager::NumberOfCaptureDevices() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+ assert(capture_device_info_);
+ return capture_device_info_->NumberOfDevices();
+}
+
+int ViEInputManager::GetDeviceName(WebRtc_UWord32 device_number,
+ WebRtc_UWord8* device_nameUTF8,
+ WebRtc_UWord32 device_name_length,
+ WebRtc_UWord8* device_unique_idUTF8,
+ WebRtc_UWord32 device_unique_idUTF8Length) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_number: %d)", __FUNCTION__, device_number);
+ assert(capture_device_info_);
+ return capture_device_info_->GetDeviceName(device_number, device_nameUTF8,
+ device_name_length,
+ device_unique_idUTF8,
+ device_unique_idUTF8Length);
+}
+
+int ViEInputManager::NumberOfCaptureCapabilities(
+ const WebRtc_UWord8* device_unique_idUTF8) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+ assert(capture_device_info_);
+ return capture_device_info_->NumberOfCapabilities(device_unique_idUTF8);
+}
+
+int ViEInputManager::GetCaptureCapability(
+ const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_capability_number,
+ CaptureCapability& capability) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_unique_idUTF8: %s, device_capability_number: %d)",
+ __FUNCTION__, device_unique_idUTF8, device_capability_number);
+ assert(capture_device_info_);
+ VideoCaptureCapability module_capability;
+ int result = capture_device_info_->GetCapability(device_unique_idUTF8,
+ device_capability_number,
+ module_capability);
+ if (result != 0)
+ return result;
+
+ // Copy from module type to public type.
+ capability.expectedCaptureDelay = module_capability.expectedCaptureDelay;
+ capability.height = module_capability.height;
+ capability.width = module_capability.width;
+ capability.interlaced = module_capability.interlaced;
+ capability.rawType = module_capability.rawType;
+ capability.codecType = module_capability.codecType;
+ capability.maxFPS = module_capability.maxFPS;
+ return result;
+}
+
+int ViEInputManager::GetOrientation(const WebRtc_UWord8* device_unique_idUTF8,
+ RotateCapturedFrame& orientation) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_unique_idUTF8: %s,)", __FUNCTION__,
+ device_unique_idUTF8);
+ assert(capture_device_info_);
+ VideoCaptureRotation module_orientation;
+ int result = capture_device_info_->GetOrientation(device_unique_idUTF8,
+ module_orientation);
+ // Copy from module type to public type.
+ switch (module_orientation) {
+ case kCameraRotate0:
+ orientation = RotateCapturedFrame_0;
+ break;
+ case kCameraRotate90:
+ orientation = RotateCapturedFrame_90;
+ break;
+ case kCameraRotate180:
+ orientation = RotateCapturedFrame_180;
+ break;
+ case kCameraRotate270:
+ orientation = RotateCapturedFrame_270;
+ break;
+ }
+ return result;
+}
+
+int ViEInputManager::DisplayCaptureSettingsDialogBox(
+ const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord8* dialog_titleUTF8,
+ void* parent_window,
+ WebRtc_UWord32 positionX,
+ WebRtc_UWord32 positionY) {
+ assert(capture_device_info_);
+ return capture_device_info_->DisplayCaptureSettingsDialogBox(
+ device_unique_idUTF8, dialog_titleUTF8, parent_window, positionX,
+ positionY);
+}
+
+int ViEInputManager::CreateCaptureDevice(
+ const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_unique_idUTF8Length,
+ int& capture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_unique_id: %s)", __FUNCTION__, device_unique_idUTF8);
+ CriticalSectionScoped cs(map_cs_.get());
+
+ // Make sure the device is not already allocated.
+ for (MapItem* item = vie_frame_provider_map_.First(); item != NULL;
+ item = vie_frame_provider_map_.Next(item)) {
+ // Make sure this is a capture device.
+ if (item->GetId() >= kViECaptureIdBase &&
+ item->GetId() <= kViECaptureIdMax) {
+ ViECapturer* vie_capture = static_cast<ViECapturer*>(item->GetItem());
+ assert(vie_capture);
+ // TODO(mflodman) Can we change input to avoid this cast?
+ const char* device_name =
+ reinterpret_cast<const char*>(vie_capture->CurrentDeviceName());
+ if (strncmp(device_name,
+ reinterpret_cast<const char*>(device_unique_idUTF8),
+ strlen(device_name)) == 0) {
+ return kViECaptureDeviceAlreadyAllocated;
+ }
+ }
+ }
+
+ // Make sure the device name is valid.
+ bool found_device = false;
+ for (WebRtc_UWord32 device_index = 0;
+ device_index < capture_device_info_->NumberOfDevices(); ++device_index) {
+ if (device_unique_idUTF8Length > kVideoCaptureUniqueNameLength) {
+ // User's string length is longer than the max.
+ return -1;
+ }
+
+ WebRtc_UWord8 found_name[kVideoCaptureDeviceNameLength] = "";
+ WebRtc_UWord8 found_unique_name[kVideoCaptureUniqueNameLength] = "";
+ capture_device_info_->GetDeviceName(device_index, found_name,
+ kVideoCaptureDeviceNameLength,
+ found_unique_name,
+ kVideoCaptureUniqueNameLength);
+
+ // TODO(mflodman) Can we change input to avoid this cast?
+ const char* cast_id = reinterpret_cast<const char*>(device_unique_idUTF8);
+ if (strncmp(cast_id, reinterpret_cast<const char*>(found_unique_name),
+ strlen(cast_id)) == 0) {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s:%d Capture device was found by unique ID: %s. Returning",
+ __FUNCTION__, __LINE__, device_unique_idUTF8);
+ found_device = true;
+ break;
+ }
+ }
+ if (!found_device) {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s:%d Capture device NOT found by unique ID: %s. Returning",
+ __FUNCTION__, __LINE__, device_unique_idUTF8);
+ return kViECaptureDeviceDoesNotExist;
+ }
+
+ int newcapture_id = 0;
+ if (GetFreeCaptureId(newcapture_id) == false) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Maximum supported number of capture devices already in "
+ "use", __FUNCTION__);
+ return kViECaptureDeviceMaxNoDevicesAllocated;
+ }
+ ViECapturer* vie_capture = ViECapturer::CreateViECapture(
+ newcapture_id, engine_id_, device_unique_idUTF8,
+ device_unique_idUTF8Length, *module_process_thread_);
+ if (!vie_capture) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not create capture module for %s", __FUNCTION__,
+ device_unique_idUTF8);
+ return kViECaptureDeviceUnknownError;
+ }
+
+ if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not insert capture module for %s", __FUNCTION__,
+ device_unique_idUTF8);
+ return kViECaptureDeviceUnknownError;
+ }
+ capture_id = newcapture_id;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_unique_id: %s, capture_id: %d)", __FUNCTION__,
+ device_unique_idUTF8, capture_id);
+ return 0;
+}
+
+int ViEInputManager::CreateCaptureDevice(VideoCaptureModule& capture_module,
+ int& capture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+
+ CriticalSectionScoped cs(map_cs_.get());
+ int newcapture_id = 0;
+ if (!GetFreeCaptureId(newcapture_id)) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Maximum supported number of capture devices already in "
+ "use", __FUNCTION__);
+ return kViECaptureDeviceMaxNoDevicesAllocated;
+ }
+
+ ViECapturer* vie_capture = ViECapturer::CreateViECapture(
+ newcapture_id, engine_id_, capture_module, *module_process_thread_);
+ if (!vie_capture) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could attach capture module.", __FUNCTION__);
+ return kViECaptureDeviceUnknownError;
+ }
+ if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not insert capture module", __FUNCTION__);
+ return kViECaptureDeviceUnknownError;
+ }
+ capture_id = newcapture_id;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s, capture_id: %d", __FUNCTION__, capture_id);
+ return 0;
+}
+
+int ViEInputManager::DestroyCaptureDevice(const int capture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(capture_id: %d)", __FUNCTION__, capture_id);
+ ViECapturer* vie_capture = NULL;
+ {
+ // We need exclusive access to the object to delete it.
+ // Take this write lock first since the read lock is taken before map_cs_.
+ ViEManagerWriteScoped wl(*this);
+ CriticalSectionScoped cs(map_cs_.get());
+
+ vie_capture = ViECapturePtr(capture_id);
+ if (!vie_capture) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(capture_id: %d) - No such capture device id",
+ __FUNCTION__, capture_id);
+ return -1;
+ }
+ WebRtc_UWord32 num_callbacks =
+ vie_capture->NumberOfRegisteredFrameCallbacks();
+ if (num_callbacks > 0) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_), "%s(capture_id: %d) - %u registered "
+ "callbacks when destroying capture device",
+ __FUNCTION__, capture_id, num_callbacks);
+ }
+ vie_frame_provider_map_.Erase(capture_id);
+ ReturnCaptureId(capture_id);
+ // Leave cs before deleting the capture object. This is because deleting the
+ // object might cause deletions of renderers so we prefer to not have a lock
+ // at that time.
+ }
+ delete vie_capture;
+ return 0;
+}
+
+int ViEInputManager::CreateExternalCaptureDevice(
+ ViEExternalCapture*& external_capture,
+ int& capture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+ CriticalSectionScoped cs(map_cs_.get());
+
+ int newcapture_id = 0;
+ if (GetFreeCaptureId(newcapture_id) == false) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Maximum supported number of capture devices already in "
+ "use", __FUNCTION__);
+ return kViECaptureDeviceMaxNoDevicesAllocated;
+ }
+
+ ViECapturer* vie_capture = ViECapturer::CreateViECapture(
+ newcapture_id, engine_id_, NULL, 0, *module_process_thread_);
+ if (!vie_capture) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not create capture module for external capture.",
+ __FUNCTION__);
+ return kViECaptureDeviceUnknownError;
+ }
+
+ if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
+ ReturnCaptureId(newcapture_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not insert capture module for external capture.",
+ __FUNCTION__);
+ return kViECaptureDeviceUnknownError;
+ }
+ capture_id = newcapture_id;
+ external_capture = vie_capture;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s, capture_id: %d)", __FUNCTION__, capture_id);
+ return 0;
+}
+
+int ViEInputManager::CreateFilePlayer(const WebRtc_Word8* file_nameUTF8,
+ const bool loop,
+ const webrtc::FileFormats file_format,
+ VoiceEngine* voe_ptr, int& file_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(device_unique_id: %s)", __FUNCTION__, file_nameUTF8);
+
+ CriticalSectionScoped cs(map_cs_.get());
+ int new_file_id = 0;
+ if (GetFreeFileId(new_file_id) == false) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Maximum supported number of file players already in use",
+ __FUNCTION__);
+ return kViEFileMaxNoOfFilesOpened;
+ }
+
+ ViEFilePlayer* vie_file_player = ViEFilePlayer::CreateViEFilePlayer(
+ new_file_id, engine_id_, file_nameUTF8, loop, file_format, *this,
+ voe_ptr);
+ if (!vie_file_player) {
+ ReturnFileId(new_file_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not open file %s for playback", __FUNCTION__,
+ file_nameUTF8);
+ return kViEFileUnknownError;
+ }
+
+ if (vie_frame_provider_map_.Insert(new_file_id, vie_file_player) != 0) {
+ ReturnCaptureId(new_file_id);
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not insert file player for %s", __FUNCTION__,
+ file_nameUTF8);
+ delete vie_file_player;
+ return kViEFileUnknownError;
+ }
+
+ file_id = new_file_id;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(filename: %s, file_id: %d)", __FUNCTION__, file_nameUTF8,
+ new_file_id);
+ return 0;
+}
+
+int ViEInputManager::DestroyFilePlayer(int file_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(file_id: %d)", __FUNCTION__, file_id);
+
+ ViEFilePlayer* vie_file_player = NULL;
+ {
+ // We need exclusive access to the object to delete it.
+ // Take this write lock first since the read lock is taken before map_cs_.
+ ViEManagerWriteScoped wl(*this);
+
+ CriticalSectionScoped cs(map_cs_.get());
+ vie_file_player = ViEFilePlayerPtr(file_id);
+ if (!vie_file_player) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(file_id: %d) - No such file player", __FUNCTION__,
+ file_id);
+ return -1;
+ }
+ int num_callbacks = vie_file_player->NumberOfRegisteredFrameCallbacks();
+ if (num_callbacks > 0) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_), "%s(file_id: %d) - %u registered "
+ "callbacks when destroying file player", __FUNCTION__,
+ file_id, num_callbacks);
+ }
+ vie_frame_provider_map_.Erase(file_id);
+ ReturnFileId(file_id);
+ // Leave cs before deleting the file object. This is because deleting the
+ // object might cause deletions of renderers so we prefer to not have a lock
+ // at that time.
+ }
+ delete vie_file_player;
+ return 0;
+}
+
+bool ViEInputManager::GetFreeCaptureId(int& freecapture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+ for (int id = 0; id < kViEMaxCaptureDevices; id++) {
+ if (free_capture_device_id_[id]) {
+ // We found a free capture device id.
+ free_capture_device_id_[id] = false;
+ freecapture_id = id + kViECaptureIdBase;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: new id: %d", __FUNCTION__, freecapture_id);
+ return true;
+ }
+ }
+ return false;
+}
+
+void ViEInputManager::ReturnCaptureId(int capture_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(%d)", __FUNCTION__, capture_id);
+ CriticalSectionScoped cs(map_cs_.get());
+ if (capture_id >= kViECaptureIdBase &&
+ capture_id < kViEMaxCaptureDevices + kViECaptureIdBase) {
+ free_capture_device_id_[capture_id - kViECaptureIdBase] = true;
+ }
+ return;
+}
+
+bool ViEInputManager::GetFreeFileId(int& free_file_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
+ __FUNCTION__);
+
+ for (int id = 0; id < kViEMaxFilePlayers; id++) {
+ if (free_file_id_[id]) {
+ // We found a free capture device id.
+ free_file_id_[id] = false;
+ free_file_id = id + kViEFileIdBase;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: new id: %d", __FUNCTION__, free_file_id);
+ return true;
+ }
+ }
+ return false;
+}
+
+void ViEInputManager::ReturnFileId(int file_id) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s(%d)", __FUNCTION__, file_id);
+
+ CriticalSectionScoped cs(map_cs_.get());
+ if (file_id >= kViEFileIdBase &&
+ file_id < kViEMaxFilePlayers + kViEFileIdBase) {
+ free_file_id_[file_id - kViEFileIdBase] = true;
+ }
+ return;
+}
+
+ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(
+ const ViEFrameCallback* capture_observer) const {
+ assert(capture_observer);
+ CriticalSectionScoped cs(map_cs_.get());
+
+ for (MapItem* provider_item = vie_frame_provider_map_.First(); provider_item
+ != NULL; provider_item = vie_frame_provider_map_.Next(provider_item)) {
+ ViEFrameProviderBase* vie_frame_provider =
+ static_cast<ViEFrameProviderBase*>(provider_item->GetItem());
+ assert(vie_frame_provider != NULL);
+
+ if (vie_frame_provider->IsFrameCallbackRegistered(capture_observer)) {
+ // We found it.
+ return vie_frame_provider;
+ }
+ }
+ // No capture device set for this channel.
+ return NULL;
+}
+
+ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(int provider_id) const {
+ CriticalSectionScoped cs(map_cs_.get());
+ MapItem* map_item = vie_frame_provider_map_.Find(provider_id);
+ if (!map_item) {
+ return NULL;
+ }
+ ViEFrameProviderBase* vie_frame_provider =
+ static_cast<ViEFrameProviderBase*>(map_item->GetItem());
+ return vie_frame_provider;
+}
+
+ViECapturer* ViEInputManager::ViECapturePtr(int capture_id) const {
+ if (!(capture_id >= kViECaptureIdBase &&
+ capture_id <= kViECaptureIdBase + kViEMaxCaptureDevices))
+ return NULL;
+
+ CriticalSectionScoped cs(map_cs_.get());
+ MapItem* map_item = vie_frame_provider_map_.Find(capture_id);
+ if (!map_item) {
+ return NULL;
+ }
+ ViECapturer* vie_capture = static_cast<ViECapturer*>(map_item->GetItem());
+ return vie_capture;
+}
+
+void ViEInputManager::GetViECaptures(MapWrapper& vie_capture_map) {
+ CriticalSectionScoped cs(map_cs_.get());
+
+ if (vie_frame_provider_map_.Size() == 0) {
+ return;
+ }
+ // Add all items to the map.
+ for (MapItem* item = vie_frame_provider_map_.First(); item != NULL;
+ item = vie_frame_provider_map_.Next(item)) {
+ vie_capture_map.Insert(item->GetId(), item->GetItem());
+ }
+ return;
+}
+
+ViEFilePlayer* ViEInputManager::ViEFilePlayerPtr(int file_id) const {
+ if (file_id < kViEFileIdBase || file_id > kViEFileIdMax) {
+ return NULL;
+ }
+ CriticalSectionScoped cs(map_cs_.get());
+ MapItem* map_item = vie_frame_provider_map_.Find(file_id);
+ if (!map_item) {
+ return NULL;
+ }
+ ViEFilePlayer* vie_file_player =
+ static_cast<ViEFilePlayer*>(map_item->GetItem());
+ return vie_file_player;
+}
+
+ViEInputManagerScoped::ViEInputManagerScoped(
+ const ViEInputManager& vie_input_manager)
+ : ViEManagerScopedBase(vie_input_manager) {
+}
+
+ViECapturer* ViEInputManagerScoped::Capture(int capture_id) const {
+ return static_cast<const ViEInputManager*>(vie_manager_)->ViECapturePtr(
+ capture_id);
+}
+
+ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
+ const ViEFrameCallback* capture_observer) const {
+ return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
+ capture_observer);
+}
+
+ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
+ int provider_id) const {
+ return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
+ provider_id);
+}
+
+ViEFilePlayer* ViEInputManagerScoped::FilePlayer(int file_id) const {
+ return static_cast<const ViEInputManager*>(vie_manager_)->ViEFilePlayerPtr(
+ file_id);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_input_manager.h b/trunk/src/video_engine/vie_input_manager.h
new file mode 100644
index 0000000..c70a2ee
--- /dev/null
+++ b/trunk/src/video_engine/vie_input_manager.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
+
+#include "modules/video_capture/main/interface/video_capture.h"
+#include "system_wrappers/interface/map_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_capture.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_frame_provider_base.h"
+#include "video_engine/vie_manager_base.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class ProcessThread;
+class RWLockWrapper;
+class ViECapturer;
+class ViEExternalCapture;
+class ViEFilePlayer;
+class VoiceEngine;
+
+class ViEInputManager : private ViEManagerBase {
+ friend class ViEInputManagerScoped;
+ public:
+ explicit ViEInputManager(int engine_id);
+ ~ViEInputManager();
+
+ void SetModuleProcessThread(ProcessThread& module_process_thread);
+
+ // Returns number of capture devices.
+ int NumberOfCaptureDevices();
+
+ // Gets name and id for a capture device.
+ int GetDeviceName(WebRtc_UWord32 device_number,
+ WebRtc_UWord8* device_nameUTF8,
+ WebRtc_UWord32 device_name_length,
+ WebRtc_UWord8* device_unique_idUTF8,
+ WebRtc_UWord32 device_unique_idUTF8Length);
+
+ // Returns the number of capture capabilities for a specified device.
+ int NumberOfCaptureCapabilities(const WebRtc_UWord8* device_unique_idUTF8);
+
+ // Gets a specific capability for a capture device.
+ int GetCaptureCapability(const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_capability_number,
+ CaptureCapability& capability);
+
+ // Show OS specific Capture settings.
+ int DisplayCaptureSettingsDialogBox(const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord8* dialog_titleUTF8,
+ void* parent_window,
+ WebRtc_UWord32 positionX,
+ WebRtc_UWord32 positionY);
+ int GetOrientation(const WebRtc_UWord8* device_unique_idUTF8,
+ RotateCapturedFrame& orientation);
+
+ // Creates a capture module for the specified capture device and assigns
+ // a capture device id for the device.
+ // Return zero on success, ViEError on failure.
+ int CreateCaptureDevice(const WebRtc_UWord8* device_unique_idUTF8,
+ const WebRtc_UWord32 device_unique_idUTF8Length,
+ int& capture_id);
+ int CreateCaptureDevice(VideoCaptureModule& capture_module,
+ int& capture_id);
+ int CreateExternalCaptureDevice(ViEExternalCapture*& external_capture,
+ int& capture_id);
+ int DestroyCaptureDevice(int capture_id);
+
+ int CreateFilePlayer(const WebRtc_Word8* file_nameUTF8, const bool loop,
+ const FileFormats file_format,
+ VoiceEngine* voe_ptr,
+ int& file_id);
+ int DestroyFilePlayer(int file_id);
+
+ private:
+ // Gets and allocates a free capture device id. Assumed protected by caller.
+ bool GetFreeCaptureId(int& freecapture_id);
+
+ // Frees a capture id assigned in GetFreeCaptureId.
+ void ReturnCaptureId(int capture_id);
+
+ // Gets and allocates a free file id. Assumed protected by caller.
+ bool GetFreeFileId(int& free_file_id);
+
+ // Frees a file id assigned in GetFreeFileId.
+ void ReturnFileId(int file_id);
+
+ // Gets the ViEFrameProvider for this capture observer.
+ ViEFrameProviderBase* ViEFrameProvider(
+ const ViEFrameCallback* capture_observer) const;
+
+ // Gets the ViEFrameProvider for this capture observer.
+ ViEFrameProviderBase* ViEFrameProvider(int provider_id) const;
+
+ // Gets the ViECapturer for the capture device id.
+ ViECapturer* ViECapturePtr(int capture_id) const;
+
+ // Gets the the entire map with GetViECaptures.
+ void GetViECaptures(MapWrapper& vie_capture_map);
+
+ // Gets the ViEFilePlayer for this file_id.
+ ViEFilePlayer* ViEFilePlayerPtr(int file_id) const;
+
+ int engine_id_;
+ scoped_ptr<CriticalSectionWrapper> map_cs_;
+ MapWrapper vie_frame_provider_map_;
+
+ // Capture devices.
+ VideoCaptureModule::DeviceInfo* capture_device_info_;
+ int free_capture_device_id_[kViEMaxCaptureDevices];
+
+ // File Players.
+ int free_file_id_[kViEMaxFilePlayers];
+
+ ProcessThread* module_process_thread_;
+};
+
+// Provides protected access to ViEInputManater.
+class ViEInputManagerScoped: private ViEManagerScopedBase {
+ public:
+ explicit ViEInputManagerScoped(const ViEInputManager& vie_input_manager);
+
+ ViECapturer* Capture(int capture_id) const;
+ ViEFilePlayer* FilePlayer(int file_id) const;
+ ViEFrameProviderBase* FrameProvider(int provider_id) const;
+ ViEFrameProviderBase* FrameProvider(const ViEFrameCallback*
+ capture_observer) const;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_manager_base.cc b/trunk/src/video_engine/vie_manager_base.cc
new file mode 100644
index 0000000..7e6695c
--- /dev/null
+++ b/trunk/src/video_engine/vie_manager_base.cc
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "assert.h"
+
+#include "system_wrappers/interface/rw_lock_wrapper.h"
+#include "video_engine/vie_manager_base.h"
+
+namespace webrtc {
+
+ViEManagerBase::ViEManagerBase()
+ : instance_rwlock_(*RWLockWrapper::CreateRWLock()) {
+}
+
+ViEManagerBase::~ViEManagerBase() {
+ delete &instance_rwlock_;
+}
+
+void ViEManagerBase::ReadLockManager() const {
+ instance_rwlock_.AcquireLockShared();
+}
+
+void ViEManagerBase::ReleaseLockManager() const {
+ instance_rwlock_.ReleaseLockShared();
+}
+
+void ViEManagerBase::WriteLockManager() {
+ instance_rwlock_.AcquireLockExclusive();
+}
+
+void ViEManagerBase::ReleaseWriteLockManager() {
+ instance_rwlock_.ReleaseLockExclusive();
+}
+
+ViEManagerScopedBase::ViEManagerScopedBase(const ViEManagerBase& ViEManagerBase)
+ : vie_manager_(&ViEManagerBase),
+ ref_count_(0) {
+ vie_manager_->ReadLockManager();
+}
+
+ViEManagerScopedBase::~ViEManagerScopedBase() {
+ assert(ref_count_ == 0);
+ vie_manager_->ReleaseLockManager();
+}
+
+ViEManagerWriteScoped::ViEManagerWriteScoped(ViEManagerBase& vie_manager)
+ : vie_manager_(&vie_manager) {
+ vie_manager_->WriteLockManager();
+}
+
+ViEManagerWriteScoped::~ViEManagerWriteScoped() {
+ vie_manager_->ReleaseWriteLockManager();
+}
+
+ViEManagedItemScopedBase::ViEManagedItemScopedBase(
+ ViEManagerScopedBase& vie_scoped_manager)
+ : vie_scoped_manager_(vie_scoped_manager) {
+ vie_scoped_manager_.ref_count_++;
+}
+
+ViEManagedItemScopedBase::~ViEManagedItemScopedBase() {
+ vie_scoped_manager_.ref_count_--;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_manager_base.h b/trunk/src/video_engine/vie_manager_base.h
new file mode 100644
index 0000000..44cf40f
--- /dev/null
+++ b/trunk/src/video_engine/vie_manager_base.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
+
+namespace webrtc {
+
+class RWLockWrapper;
+
+class ViEManagerBase {
+ friend class ViEManagerScopedBase;
+ friend class ViEManagedItemScopedBase;
+ friend class ViEManagerWriteScoped;
+ public:
+ ViEManagerBase();
+ ~ViEManagerBase();
+
+ private:
+ // Exclusive lock, used by ViEManagerWriteScoped
+ void WriteLockManager();
+
+ // Releases exclusive lock, used by ViEManagerWriteScoped.
+ void ReleaseWriteLockManager();
+
+ // Increases lock count, used by ViEManagerScopedBase.
+ void ReadLockManager() const;
+
+ // Releases the lock count, used by ViEManagerScopedBase.
+ void ReleaseLockManager() const;
+
+ RWLockWrapper& instance_rwlock_;
+};
+
+class ViEManagerWriteScoped {
+ public:
+ explicit ViEManagerWriteScoped(ViEManagerBase& vie_manager);
+ ~ViEManagerWriteScoped();
+
+ private:
+ ViEManagerBase* vie_manager_;
+};
+
+class ViEManagerScopedBase {
+ friend class ViEManagedItemScopedBase;
+ public:
+ explicit ViEManagerScopedBase(const ViEManagerBase& vie_manager);
+ ~ViEManagerScopedBase();
+
+ protected:
+ const ViEManagerBase* vie_manager_;
+
+ private:
+ int ref_count_;
+};
+
+class ViEManagedItemScopedBase {
+ public:
+ explicit ViEManagedItemScopedBase(ViEManagerScopedBase& vie_scoped_manager);
+ ~ViEManagedItemScopedBase();
+ protected:
+ ViEManagerScopedBase& vie_scoped_manager_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
diff --git a/trunk/src/video_engine/vie_network_impl.cc b/trunk/src/video_engine/vie_network_impl.cc
new file mode 100644
index 0000000..b4bb0e9
--- /dev/null
+++ b/trunk/src/video_engine/vie_network_impl.cc
@@ -0,0 +1,798 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_network_impl.h"
+
+#include <stdio.h>
+#if (defined(WIN32_) || defined(WIN64_))
+#include <qos.h>
+#endif
+
+#include "engine_configurations.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViENetwork* ViENetwork::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViENetworkImpl* vie_networkImpl = vie_impl;
+ // Increase ref count.
+ (*vie_networkImpl)++;
+ return vie_networkImpl;
+#else
+ return NULL;
+#endif
+}
+
+int ViENetworkImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViENetwork::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViENetwork release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViENetwork reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViENetworkImpl::ViENetworkImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViENetworkImpl::ViENetworkImpl() Ctor");
+}
+
+ViENetworkImpl::~ViENetworkImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViENetworkImpl::~ViENetworkImpl() Dtor");
+}
+
+int ViENetworkImpl::SetLocalReceiver(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const char* ip_address) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
+ __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ // The channel doesn't exists.
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->Receiving()) {
+ shared_data_->SetLastError(kViENetworkAlreadyReceiving);
+ return -1;
+ }
+ if (vie_channel->SetLocalReceiver(rtp_port, rtcp_port, ip_address) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::GetLocalReceiver(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetLocalReceiver(rtp_port, rtcp_port, ip_address) != 0) {
+ shared_data_->SetLastError(kViENetworkLocalReceiverNotSet);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SetSendDestination(const int video_channel,
+ const char* ip_address,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const unsigned short source_rtp_port,
+ const unsigned short source_rtcp_port) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, ip_address: %s, rtp_port: %u, rtcp_port: %u, "
+ "sourceRtpPort: %u, source_rtcp_port: %u)",
+ __FUNCTION__, video_channel, ip_address, rtp_port, rtcp_port,
+ source_rtp_port, source_rtcp_port);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel doesn't exist", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel already sending.", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkAlreadySending);
+ return -1;
+ }
+ if (vie_channel->SetSendDestination(ip_address, rtp_port, rtcp_port,
+ source_rtp_port,
+ source_rtcp_port) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::GetSendDestination(const int video_channel,
+ char* ip_address,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ unsigned short& source_rtp_port,
+ unsigned short& source_rtcp_port) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetSendDestination(ip_address, rtp_port, rtcp_port,
+ source_rtp_port,
+ source_rtcp_port) != 0) {
+ shared_data_->SetLastError(kViENetworkDestinationNotSet);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::RegisterSendTransport(const int video_channel,
+ Transport& transport) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel doesn't exist", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel already sending.", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkAlreadySending);
+ return -1;
+ }
+ if (vie_channel->RegisterSendTransport(transport) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::DeregisterSendTransport(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel doesn't exist", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s Channel already sending", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkAlreadySending);
+ return -1;
+ }
+ if (vie_channel->DeregisterSendTransport() != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::ReceivedRTPPacket(const int video_channel, const void* data,
+ const int length) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, data: -, length: %d)", __FUNCTION__,
+ video_channel, length);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ // The channel doesn't exists
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ return vie_channel->ReceivedRTPPacket(data, length);
+}
+
+int ViENetworkImpl::ReceivedRTCPPacket(const int video_channel,
+ const void* data, const int length) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, data: -, length: %d)", __FUNCTION__,
+ video_channel, length);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ return vie_channel->ReceivedRTCPPacket(data, length);
+}
+
+int ViENetworkImpl::GetSourceInfo(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port, char* ip_address,
+ unsigned int ip_address_length) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetSourceInfo(rtp_port, rtcp_port, ip_address,
+ ip_address_length) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::GetLocalIP(char ip_address[64], bool ipv6) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s( ip_address, ipV6: %d)", __FUNCTION__, ipv6);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ if (!ip_address) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: No argument", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkInvalidArgument);
+ return -1;
+ }
+
+ WebRtc_UWord8 num_socket_threads = 1;
+ UdpTransport* socket_transport = UdpTransport::Create(
+ ViEModuleId(shared_data_->instance_id(), -1), num_socket_threads);
+
+ if (!socket_transport) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Could not create socket module", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+
+ char local_ip_address[64];
+ if (ipv6) {
+ char local_ip[16];
+ if (socket_transport->LocalHostAddressIPV6(local_ip) != 0) {
+ UdpTransport::Destroy(socket_transport);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Could not get local IP", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ // Convert 128-bit address to character string (a:b:c:d:e:f:g:h).
+ sprintf(local_ip_address,
+ "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:"
+ "%.2x%.2x",
+ local_ip[0], local_ip[1], local_ip[2], local_ip[3], local_ip[4],
+ local_ip[5], local_ip[6], local_ip[7], local_ip[8], local_ip[9],
+ local_ip[10], local_ip[11], local_ip[12], local_ip[13],
+ local_ip[14], local_ip[15]);
+ } else {
+ WebRtc_UWord32 local_ip = 0;
+ if (socket_transport->LocalHostAddress(local_ip) != 0) {
+ UdpTransport::Destroy(socket_transport);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: Could not get local IP", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ // Convert 32-bit address to character string (x.y.z.w).
+ sprintf(local_ip_address, "%d.%d.%d.%d",
+ static_cast<int>((local_ip >> 24) & 0x0ff),
+ static_cast<int>((local_ip >> 16) & 0x0ff),
+ static_cast<int>((local_ip >> 8) & 0x0ff),
+ static_cast<int>(local_ip & 0x0ff));
+ }
+ strcpy(ip_address, local_ip_address);
+ UdpTransport::Destroy(socket_transport);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: local ip = %s", __FUNCTION__, local_ip_address);
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: not available for external transport", __FUNCTION__);
+
+ return -1;
+#endif
+}
+
+int ViENetworkImpl::EnableIPv6(int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->EnableIPv6() != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+bool ViENetworkImpl::IsIPv6Enabled(int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return false;
+ }
+ return vie_channel->IsIPv6Enabled();
+}
+
+int ViENetworkImpl::SetSourceFilter(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const char* ip_address) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
+ __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetSourceFilter(rtp_port, rtcp_port, ip_address) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::GetSourceFilter(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetSourceFilter(rtp_port, rtcp_port, ip_address) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SetSendToS(const int video_channel, const int DSCP,
+ const bool use_set_sockOpt = false) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, DSCP: %d, use_set_sockOpt: %d)", __FUNCTION__,
+ video_channel, DSCP, use_set_sockOpt);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ " force use_set_sockopt=true since there is no alternative"
+ " implementation");
+ if (vie_channel->SetToS(DSCP, true) != 0) {
+#else
+ if (vie_channel->SetToS(DSCP, use_set_sockOpt) != 0) {
+#endif
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::GetSendToS(const int video_channel, int& DSCP,
+ bool& use_set_sockOpt) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetToS((WebRtc_Word32&) DSCP, use_set_sockOpt) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SetSendGQoS(const int video_channel, const bool enable,
+ const int service_type,
+ const int overrideDSCP) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, service_type: %d, "
+ "overrideDSCP: %d)", __FUNCTION__, video_channel, enable,
+ service_type, overrideDSCP);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+#if (defined(WIN32_) || defined(WIN64_))
+ // Sanity check. We might crash if testing and relying on an OS socket error.
+ if (enable &&
+ (service_type != SERVICETYPE_BESTEFFORT) &&
+ (service_type != SERVICETYPE_CONTROLLEDLOAD) &&
+ (service_type != SERVICETYPE_GUARANTEED) &&
+ (service_type != SERVICETYPE_QUALITATIVE)) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: service type %d not supported", __FUNCTION__,
+ video_channel, service_type);
+ shared_data_->SetLastError(kViENetworkServiceTypeNotSupported);
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ VideoCodec video_codec;
+ if (vie_encoder->GetEncoder(video_codec) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get max bitrate for the channel",
+ __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkSendCodecNotSet);
+ return -1;
+ }
+ if (vie_channel->SetSendGQoS(enable, service_type, video_codec.maxBitrate,
+ overrideDSCP) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+#else
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Not supported", __FUNCTION__);
+ shared_data_->SetLastError(kViENetworkNotSupported);
+ return -1;
+#endif
+}
+
+int ViENetworkImpl::GetSendGQoS(const int video_channel, bool& enabled,
+ int& service_type, int& overrideDSCP) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetSendGQoS(enabled, service_type, overrideDSCP) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SetMTU(int video_channel, unsigned int mtu) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, mtu: %u)", __FUNCTION__, video_channel, mtu);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetMTU(mtu) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SetPacketTimeoutNotification(const int video_channel,
+ bool enable,
+ int timeout_seconds) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, timeout_seconds: %u)",
+ __FUNCTION__, video_channel, enable, timeout_seconds);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetPacketTimeoutNotification(enable,
+ timeout_seconds) != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::RegisterObserver(const int video_channel,
+ ViENetworkObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterNetworkObserver(&observer) != 0) {
+ shared_data_->SetLastError(kViENetworkObserverAlreadyRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::DeregisterObserver(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (!vie_channel->NetworkObserverRegistered()) {
+ shared_data_->SetLastError(kViENetworkObserverNotRegistered);
+ return -1;
+ }
+ return vie_channel->RegisterNetworkObserver(NULL);
+}
+
+int ViENetworkImpl::SetPeriodicDeadOrAliveStatus(
+ const int video_channel,
+ bool enable,
+ unsigned int sample_time_seconds) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, sample_time_seconds: %ul)",
+ __FUNCTION__, video_channel, enable, sample_time_seconds);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (!vie_channel->NetworkObserverRegistered()) {
+ shared_data_->SetLastError(kViENetworkObserverNotRegistered);
+ return -1;
+ }
+ if (vie_channel->SetPeriodicDeadOrAliveStatus(enable, sample_time_seconds)
+ != 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViENetworkImpl::SendUDPPacket(const int video_channel, const void* data,
+ const unsigned int length,
+ int& transmitted_bytes,
+ bool use_rtcp_socket = false) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, data: -, length: %d, transmitter_bytes: -, "
+ "useRtcpSocket: %d)", __FUNCTION__, video_channel, length,
+ use_rtcp_socket);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "Channel doesn't exist");
+ shared_data_->SetLastError(kViENetworkInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SendUDPPacket((const WebRtc_Word8*) data, length,
+ (WebRtc_Word32&) transmitted_bytes,
+ use_rtcp_socket) < 0) {
+ shared_data_->SetLastError(kViENetworkUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_network_impl.h b/trunk/src/video_engine/vie_network_impl.h
new file mode 100644
index 0000000..8eff523
--- /dev/null
+++ b/trunk/src/video_engine/vie_network_impl.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
+
+#include "typedefs.h"
+#include "video_engine/include/vie_network.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViENetworkImpl
+ : public ViENetwork,
+ public ViERefCount {
+ public:
+ // Implements ViENetwork.
+ virtual int Release();
+ virtual int SetLocalReceiver(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const char* ip_address);
+ virtual int GetLocalReceiver(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address);
+ virtual int SetSendDestination(const int video_channel,
+ const char* ip_address,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const unsigned short source_rtp_port,
+ const unsigned short source_rtcp_port);
+ virtual int GetSendDestination(const int video_channel,
+ char* ip_address,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ unsigned short& source_rtp_port,
+ unsigned short& source_rtcp_port);
+ virtual int RegisterSendTransport(const int video_channel,
+ Transport& transport);
+ virtual int DeregisterSendTransport(const int video_channel);
+ virtual int ReceivedRTPPacket(const int video_channel,
+ const void* data,
+ const int length);
+ virtual int ReceivedRTCPPacket(const int video_channel,
+ const void* data,
+ const int length);
+ virtual int GetSourceInfo(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address,
+ unsigned int ip_address_length);
+ virtual int GetLocalIP(char ip_address[64], bool ipv6);
+ virtual int EnableIPv6(int video_channel);
+ virtual bool IsIPv6Enabled(int video_channel);
+ virtual int SetSourceFilter(const int video_channel,
+ const unsigned short rtp_port,
+ const unsigned short rtcp_port,
+ const char* ip_address);
+ virtual int GetSourceFilter(const int video_channel,
+ unsigned short& rtp_port,
+ unsigned short& rtcp_port,
+ char* ip_address);
+ virtual int SetSendToS(const int video_channel,
+ const int DSCP,
+ const bool use_set_sockOpt);
+ virtual int GetSendToS(const int video_channel,
+ int& DSCP,
+ bool& use_set_sockOpt);
+ virtual int SetSendGQoS(const int video_channel,
+ const bool enable,
+ const int service_type,
+ const int overrideDSCP);
+ virtual int GetSendGQoS(const int video_channel,
+ bool& enabled,
+ int& service_type,
+ int& overrideDSCP);
+ virtual int SetMTU(int video_channel, unsigned int mtu);
+ virtual int SetPacketTimeoutNotification(const int video_channel,
+ bool enable,
+ int timeout_seconds);
+ virtual int RegisterObserver(const int video_channel,
+ ViENetworkObserver& observer);
+ virtual int DeregisterObserver(const int video_channel);
+ virtual int SetPeriodicDeadOrAliveStatus(
+ const int video_channel,
+ const bool enable,
+ const unsigned int sample_time_seconds);
+ virtual int SendUDPPacket(const int video_channel,
+ const void* data,
+ const unsigned int length,
+ int& transmitted_bytes,
+ bool use_rtcp_socket);
+
+ protected:
+ ViENetworkImpl(ViESharedData* shared_data);
+ virtual ~ViENetworkImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
diff --git a/trunk/src/video_engine/vie_performance_monitor.cc b/trunk/src/video_engine/vie_performance_monitor.cc
new file mode 100644
index 0000000..43ae056
--- /dev/null
+++ b/trunk/src/video_engine/vie_performance_monitor.cc
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_performance_monitor.h"
+
+#include "cpu_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "trace.h"
+#include "vie_base.h"
+#include "vie_defines.h"
+
+namespace webrtc {
+
+enum { kVieMonitorPeriodMs = 975 };
+enum { kVieCpuStartValue = 75 };
+
+ViEPerformanceMonitor::ViEPerformanceMonitor(int engine_id)
+ : engine_id_(engine_id),
+ pointer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ monitor_thread_(NULL),
+ monitor_event_(*EventWrapper::Create()),
+ average_application_cpu_(kVieCpuStartValue),
+ average_system_cpu_(kVieCpuStartValue),
+ cpu_(NULL),
+ vie_base_observer_(NULL) {
+}
+
+ViEPerformanceMonitor::~ViEPerformanceMonitor() {
+ Terminate();
+ delete pointer_cs_;
+ delete &monitor_event_;
+}
+
+int ViEPerformanceMonitor::Init(ViEBaseObserver* vie_base_observer) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s", __FUNCTION__);
+
+ CriticalSectionScoped cs(pointer_cs_);
+ if (!vie_base_observer || vie_base_observer_) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Bad input argument or observer already set",
+ __FUNCTION__);
+ return -1;
+ }
+
+ cpu_ = CpuWrapper::CreateCpu();
+ if (cpu_ == NULL) {
+ // Performance monitoring not supported
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_), "%s: Not supported", __FUNCTION__);
+ return 0;
+ }
+
+ if (monitor_thread_ == NULL) {
+ monitor_event_.StartTimer(true, kVieMonitorPeriodMs);
+ monitor_thread_ = ThreadWrapper::CreateThread(ViEMonitorThreadFunction,
+ this, kNormalPriority,
+ "ViEPerformanceMonitor");
+ unsigned int t_id = 0;
+ if (monitor_thread_->Start(t_id)) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Performance monitor thread started %u",
+ __FUNCTION__, t_id);
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s: Could not start performance monitor", __FUNCTION__);
+ monitor_event_.StopTimer();
+ return -1;
+ }
+ }
+ vie_base_observer_ = vie_base_observer;
+ return 0;
+}
+
+void ViEPerformanceMonitor::Terminate() {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
+ "%s", __FUNCTION__);
+
+ pointer_cs_->Enter();
+ if (!vie_base_observer_) {
+ pointer_cs_->Leave();
+ return;
+ }
+
+ vie_base_observer_ = NULL;
+ monitor_event_.StopTimer();
+ if (monitor_thread_) {
+ ThreadWrapper* tmp_thread = monitor_thread_;
+ monitor_thread_ = NULL;
+ monitor_event_.Set();
+ pointer_cs_->Leave();
+ if (tmp_thread->Stop()) {
+ pointer_cs_->Enter();
+ delete tmp_thread;
+ tmp_thread = NULL;
+ delete cpu_;
+ }
+ cpu_ = NULL;
+ }
+ pointer_cs_->Leave();
+}
+
+bool ViEPerformanceMonitor::ViEBaseObserverRegistered() {
+ CriticalSectionScoped cs(pointer_cs_);
+ return vie_base_observer_ != NULL;
+}
+
+bool ViEPerformanceMonitor::ViEMonitorThreadFunction(void* obj) {
+ return static_cast<ViEPerformanceMonitor*>(obj)->ViEMonitorProcess();
+}
+
+bool ViEPerformanceMonitor::ViEMonitorProcess() {
+ // Periodically triggered with time KViEMonitorPeriodMs
+ monitor_event_.Wait(kVieMonitorPeriodMs);
+ if (monitor_thread_ == NULL) {
+ // Thread removed, exit
+ return false;
+ }
+
+ CriticalSectionScoped cs(pointer_cs_);
+ if (cpu_) {
+ int cpu_load = cpu_->CpuUsage();
+ if (cpu_load > 75) {
+ if (vie_base_observer_) {
+ vie_base_observer_->PerformanceAlarm(cpu_load);
+ }
+ }
+ }
+ return true;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_performance_monitor.h b/trunk/src/video_engine/vie_performance_monitor.h
new file mode 100644
index 0000000..c3ea931
--- /dev/null
+++ b/trunk/src/video_engine/vie_performance_monitor.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// ViEPerformanceMonitor is used to check the current CPU usage and triggers a
+// callback when getting over a specified threshold.
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
+
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "vie_defines.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class CpuWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class ViEBaseObserver;
+
+class ViEPerformanceMonitor {
+ public:
+ explicit ViEPerformanceMonitor(int engine_id);
+ ~ViEPerformanceMonitor();
+
+ int Init(ViEBaseObserver* vie_base_observer);
+ void Terminate();
+ bool ViEBaseObserverRegistered();
+
+ protected:
+ static bool ViEMonitorThreadFunction(void* obj);
+ bool ViEMonitorProcess();
+
+ private:
+ const int engine_id_;
+ // TODO(mfldoman) Make this one scoped_ptr.
+ CriticalSectionWrapper* pointer_cs_;
+ ThreadWrapper* monitor_thread_;
+ EventWrapper& monitor_event_;
+ int average_application_cpu_;
+ int average_system_cpu_;
+ CpuWrapper* cpu_;
+ ViEBaseObserver* vie_base_observer_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
diff --git a/trunk/src/video_engine/vie_receiver.cc b/trunk/src/video_engine/vie_receiver.cc
new file mode 100644
index 0000000..af0dc75
--- /dev/null
+++ b/trunk/src/video_engine/vie_receiver.cc
@@ -0,0 +1,267 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_receiver.h"
+
+#include "critical_section_wrapper.h"
+#include "rtp_dump.h"
+#include "rtp_rtcp.h"
+#include "video_coding.h"
+#include "trace.h"
+
+namespace webrtc {
+
+ViEReceiver::ViEReceiver(int engine_id, int channel_id,
+ RtpRtcp& rtp_rtcp,
+ VideoCodingModule& module_vcm)
+ : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ engine_id_(engine_id),
+ channel_id_(channel_id),
+ rtp_rtcp_(rtp_rtcp),
+ vcm_(module_vcm),
+ external_decryption_(NULL),
+ decryption_buffer_(NULL),
+ rtp_dump_(NULL),
+ receiving_(false) {
+}
+
+ViEReceiver::~ViEReceiver() {
+ if (decryption_buffer_) {
+ delete[] decryption_buffer_;
+ decryption_buffer_ = NULL;
+ }
+ if (rtp_dump_) {
+ rtp_dump_->Stop();
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ }
+}
+
+int ViEReceiver::RegisterExternalDecryption(Encryption* decryption) {
+ CriticalSectionScoped cs(receive_cs_.get());
+ if (external_decryption_) {
+ return -1;
+ }
+ decryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu];
+ if (decryption_buffer_ == NULL) {
+ return -1;
+ }
+ external_decryption_ = decryption;
+ return 0;
+}
+
+int ViEReceiver::DeregisterExternalDecryption() {
+ CriticalSectionScoped cs(receive_cs_.get());
+ if (external_decryption_ == NULL) {
+ return -1;
+ }
+ external_decryption_ = NULL;
+ return 0;
+}
+
+void ViEReceiver::RegisterSimulcastRtpRtcpModules(
+ const std::list<RtpRtcp*>& rtp_modules) {
+ CriticalSectionScoped cs(receive_cs_.get());
+ rtp_rtcp_simulcast_.clear();
+
+ if (!rtp_modules.empty()) {
+ rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(),
+ rtp_modules.begin(),
+ rtp_modules.end());
+ }
+}
+
+void ViEReceiver::IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
+ const WebRtc_Word32 rtp_packet_length,
+ const WebRtc_Word8* from_ip,
+ const WebRtc_UWord16 from_port) {
+ InsertRTPPacket(rtp_packet, rtp_packet_length);
+}
+
+void ViEReceiver::IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
+ const WebRtc_Word32 rtcp_packet_length,
+ const WebRtc_Word8* from_ip,
+ const WebRtc_UWord16 from_port) {
+ InsertRTCPPacket(rtcp_packet, rtcp_packet_length);
+}
+
+int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
+ int rtp_packet_length) {
+ if (!receiving_) {
+ return -1;
+ }
+ return InsertRTPPacket((const WebRtc_Word8*) rtp_packet, rtp_packet_length);
+}
+
+int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
+ int rtcp_packet_length) {
+ if (!receiving_) {
+ return -1;
+ }
+ return InsertRTCPPacket((const WebRtc_Word8*) rtcp_packet,
+ rtcp_packet_length);
+}
+
+WebRtc_Word32 ViEReceiver::OnReceivedPayloadData(
+ const WebRtc_UWord8* payload_data, const WebRtc_UWord16 payload_size,
+ const WebRtcRTPHeader* rtp_header) {
+ if (rtp_header == NULL) {
+ return 0;
+ }
+
+ if (vcm_.IncomingPacket(payload_data, payload_size, *rtp_header) != 0) {
+ // Check this...
+ return -1;
+ }
+ return 0;
+}
+
+int ViEReceiver::InsertRTPPacket(const WebRtc_Word8* rtp_packet,
+ int rtp_packet_length) {
+ // TODO(mflodman) Change decrypt to get rid of this cast.
+ WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtp_packet);
+ unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr);
+ int received_packet_length = rtp_packet_length;
+
+ {
+ CriticalSectionScoped cs(receive_cs_.get());
+
+ if (external_decryption_) {
+ int decrypted_length = 0;
+ external_decryption_->decrypt(channel_id_, received_packet,
+ decryption_buffer_, received_packet_length,
+ &decrypted_length);
+ if (decrypted_length <= 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "RTP decryption failed");
+ return -1;
+ } else if (decrypted_length > kViEMaxMtu) {
+ WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "InsertRTPPacket: %d bytes is allocated as RTP decrytption"
+ " output, external decryption used %d bytes. => memory is "
+ " now corrupted", kViEMaxMtu, decrypted_length);
+ return -1;
+ }
+ received_packet = decryption_buffer_;
+ received_packet_length = decrypted_length;
+ }
+
+ if (rtp_dump_) {
+ rtp_dump_->DumpPacket(received_packet,
+ static_cast<WebRtc_UWord16>(received_packet_length));
+ }
+ }
+ return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length);
+}
+
+int ViEReceiver::InsertRTCPPacket(const WebRtc_Word8* rtcp_packet,
+ int rtcp_packet_length) {
+ // TODO(mflodman) Change decrypt to get rid of this cast.
+ WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtcp_packet);
+ unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr);
+ int received_packet_length = rtcp_packet_length;
+ {
+ CriticalSectionScoped cs(receive_cs_.get());
+
+ if (external_decryption_) {
+ int decrypted_length = 0;
+ external_decryption_->decrypt_rtcp(channel_id_, received_packet,
+ decryption_buffer_,
+ received_packet_length,
+ &decrypted_length);
+ if (decrypted_length <= 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_), "RTP decryption failed");
+ return -1;
+ } else if (decrypted_length > kViEMaxMtu) {
+ WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "InsertRTCPPacket: %d bytes is allocated as RTP "
+ " decrytption output, external decryption used %d bytes. "
+ " => memory is now corrupted",
+ kViEMaxMtu, decrypted_length);
+ return -1;
+ }
+ received_packet = decryption_buffer_;
+ received_packet_length = decrypted_length;
+ }
+
+ if (rtp_dump_) {
+ rtp_dump_->DumpPacket(
+ received_packet, static_cast<WebRtc_UWord16>(received_packet_length));
+ }
+ }
+ {
+ CriticalSectionScoped cs(receive_cs_.get());
+ std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
+ while (it != rtp_rtcp_simulcast_.end()) {
+ RtpRtcp* rtp_rtcp = *it++;
+ rtp_rtcp->IncomingPacket(received_packet, received_packet_length);
+ }
+ }
+ return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length);
+}
+
+void ViEReceiver::StartReceive() {
+ receiving_ = true;
+}
+
+void ViEReceiver::StopReceive() {
+ receiving_ = false;
+}
+
+int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) {
+ CriticalSectionScoped cs(receive_cs_.get());
+ if (rtp_dump_) {
+ // Restart it if it already exists and is started
+ rtp_dump_->Stop();
+ } else {
+ rtp_dump_ = RtpDump::CreateRtpDump();
+ if (rtp_dump_ == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StartRTPDump: Failed to create RTP dump");
+ return -1;
+ }
+ }
+ if (rtp_dump_->Start(file_nameUTF8) != 0) {
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StartRTPDump: Failed to start RTP dump");
+ return -1;
+ }
+ return 0;
+}
+
+int ViEReceiver::StopRTPDump() {
+ CriticalSectionScoped cs(receive_cs_.get());
+ if (rtp_dump_) {
+ if (rtp_dump_->IsActive()) {
+ rtp_dump_->Stop();
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StopRTPDump: Dump not active");
+ }
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StopRTPDump: RTP dump not started");
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_receiver.h b/trunk/src/video_engine/vie_receiver.h
new file mode 100644
index 0000000..a99d388
--- /dev/null
+++ b/trunk/src/video_engine/vie_receiver.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
+
+#include <list>
+
+#include "engine_configurations.h"
+#include "rtp_rtcp_defines.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "udp_transport.h"
+#include "vie_defines.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class Encryption;
+class RtpDump;
+class RtpRtcp;
+class VideoCodingModule;
+
+class ViEReceiver : public UdpTransportData, public RtpData {
+ public:
+ ViEReceiver(int engine_id, int channel_id, RtpRtcp& rtp_rtcp,
+ VideoCodingModule& module_vcm);
+ ~ViEReceiver();
+
+ int RegisterExternalDecryption(Encryption* decryption);
+ int DeregisterExternalDecryption();
+
+ void RegisterSimulcastRtpRtcpModules(const std::list<RtpRtcp*>& rtp_modules);
+
+ void StartReceive();
+ void StopReceive();
+
+ int StartRTPDump(const char file_nameUTF8[1024]);
+ int StopRTPDump();
+
+ // Implements UdpTransportData.
+ virtual void IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
+ const WebRtc_Word32 rtp_packet_length,
+ const WebRtc_Word8* from_ip,
+ const WebRtc_UWord16 from_port);
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
+ const WebRtc_Word32 rtcp_packet_length,
+ const WebRtc_Word8* from_ip,
+ const WebRtc_UWord16 from_port);
+
+ // Receives packets from external transport.
+ int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length);
+ int ReceivedRTCPPacket(const void* rtcp_packet, int rtcp_packet_length);
+
+ // Implements RtpData.
+ virtual WebRtc_Word32 OnReceivedPayloadData(
+ const WebRtc_UWord8* payload_data,
+ const WebRtc_UWord16 payload_size,
+ const WebRtcRTPHeader* rtp_header);
+
+ private:
+ int InsertRTPPacket(const WebRtc_Word8* rtp_packet, int rtp_packet_length);
+ int InsertRTCPPacket(const WebRtc_Word8* rtcp_packet, int rtcp_packet_length);
+
+ scoped_ptr<CriticalSectionWrapper> receive_cs_;
+ int engine_id_;
+ int channel_id_;
+ RtpRtcp& rtp_rtcp_;
+ std::list<RtpRtcp*> rtp_rtcp_simulcast_;
+ VideoCodingModule& vcm_;
+
+ Encryption* external_decryption_;
+ WebRtc_UWord8* decryption_buffer_;
+ RtpDump* rtp_dump_;
+ bool receiving_;
+};
+
+} // namespace webrt
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
diff --git a/trunk/src/video_engine/vie_ref_count.cc b/trunk/src/video_engine/vie_ref_count.cc
new file mode 100644
index 0000000..2ab285c
--- /dev/null
+++ b/trunk/src/video_engine/vie_ref_count.cc
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_ref_count.h"
+
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+
+ViERefCount::ViERefCount()
+ : count_(0),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
+
+ViERefCount::~ViERefCount() {
+}
+
+ViERefCount& ViERefCount::operator++(int) {
+ CriticalSectionScoped lock(crit_.get());
+ count_++;
+ return *this;
+}
+
+ViERefCount& ViERefCount::operator--(int) {
+ CriticalSectionScoped lock(crit_.get());
+ count_--;
+ return *this;
+}
+
+void ViERefCount::Reset() {
+ CriticalSectionScoped lock(crit_.get());
+ count_ = 0;
+}
+
+int ViERefCount::GetCount() const {
+ return count_;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_ref_count.h b/trunk/src/video_engine/vie_ref_count.h
new file mode 100644
index 0000000..65efe36
--- /dev/null
+++ b/trunk/src/video_engine/vie_ref_count.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// TODO(mflodman) Use ref count class in system_wrappers.
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
+
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+
+class ViERefCount {
+ public:
+ ViERefCount();
+ ~ViERefCount();
+
+ ViERefCount& operator++(int);
+ ViERefCount& operator--(int);
+
+ void Reset();
+ int GetCount() const;
+
+ private:
+ volatile int count_;
+ scoped_ptr<CriticalSectionWrapper> crit_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
diff --git a/trunk/src/video_engine/vie_remb.cc b/trunk/src/video_engine/vie_remb.cc
new file mode 100644
index 0000000..7732408
--- /dev/null
+++ b/trunk/src/video_engine/vie_remb.cc
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_remb.h"
+
+#include <algorithm>
+#include <cassert>
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/tick_util.h"
+#include "system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+const int kRembSendIntervallMs = 1000;
+
+// % threshold for if we should send a new REMB asap.
+const int kSendThresholdPercent = 97;
+
+VieRemb::VieRemb(int engine_id)
+ : engine_id_(engine_id),
+ list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ last_remb_time_(TickTime::MillisecondTimestamp()),
+ last_send_bitrate_(0) {
+}
+
+VieRemb::~VieRemb() {
+}
+
+void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::AddReceiveChannel(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+ if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
+ receive_modules_.end())
+ return;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, engine_id_, "AddRembChannel");
+ // The module probably doesn't have a remote SSRC yet, so don't add it to the
+ // map.
+ receive_modules_.push_back(rtp_rtcp);
+}
+
+void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::RemoveReceiveChannel(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+ unsigned int ssrc = rtp_rtcp->RemoteSSRC();
+ for (RtpModules::iterator it = receive_modules_.begin();
+ it != receive_modules_.end(); ++it) {
+ if ((*it) == rtp_rtcp) {
+ receive_modules_.erase(it);
+ break;
+ }
+ }
+ bitrates_.erase(ssrc);
+}
+
+void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::AddRembSender(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+
+ // Verify this module hasn't been added earlier.
+ if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
+ rtcp_sender_.end())
+ return;
+ rtcp_sender_.push_back(rtp_rtcp);
+}
+
+void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::RemoveRembSender(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+ for (RtpModules::iterator it = rtcp_sender_.begin();
+ it != rtcp_sender_.end(); ++it) {
+ if ((*it) == rtp_rtcp) {
+ rtcp_sender_.erase(it);
+ return;
+ }
+ }
+}
+
+void VieRemb::AddSendChannel(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::AddSendChannel(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+
+ // Verify this module hasn't been added earlier.
+ if (std::find(send_modules_.begin(), send_modules_.end(), rtp_rtcp) !=
+ send_modules_.end())
+ return;
+ send_modules_.push_back(rtp_rtcp);
+}
+
+void VieRemb::RemoveSendChannel(RtpRtcp* rtp_rtcp) {
+ assert(rtp_rtcp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
+ "VieRemb::RemoveSendChannel(%p)", rtp_rtcp);
+
+ CriticalSectionScoped cs(list_crit_.get());
+ for (RtpModules::iterator it = send_modules_.begin();
+ it != send_modules_.end(); ++it) {
+ if ((*it) == rtp_rtcp) {
+ send_modules_.erase(it);
+ return;
+ }
+ }
+}
+
+void VieRemb::OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, engine_id_,
+ "VieRemb::UpdateBitrateEstimate(ssrc: %u, bitrate: %u)",
+ ssrc, bitrate);
+ CriticalSectionScoped cs(list_crit_.get());
+
+ // Check if this is a new ssrc and add it to the map if it is.
+ if (bitrates_.find(ssrc) == bitrates_.end()) {
+ bitrates_[ssrc] = bitrate;
+ }
+
+ int new_remb_bitrate = last_send_bitrate_ - bitrates_[ssrc] + bitrate;
+ if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
+ // The new bitrate estimate is less than kSendThresholdPercent % of the last
+ // report. Send a REMB asap.
+ last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervallMs;
+ }
+ bitrates_[ssrc] = bitrate;
+}
+
+void VieRemb::OnReceivedRemb(unsigned int bitrate) {
+ WEBRTC_TRACE(kTraceStream, kTraceVideo, engine_id_,
+ "VieRemb::OnReceivedRemb(bitrate: %u)", bitrate);
+ // TODO(mflodman) Should be extended to allow different split of bitrate.
+ // TODO(mflodman) Do we want to call |SetMaximumBitrateEstimate| from
+ // |Process| instead?
+
+ // Split the bitrate estimate between all sending channels.
+ CriticalSectionScoped cs(list_crit_.get());
+ for (RtpModules::iterator it = send_modules_.begin();
+ it != send_modules_.end(); ++it) {
+ (*it)->SetMaximumBitrateEstimate(bitrate / send_modules_.size());
+ }
+}
+
+WebRtc_Word32 VieRemb::ChangeUniqueId(const WebRtc_Word32 id) {
+ return 0;
+}
+
+WebRtc_Word32 VieRemb::TimeUntilNextProcess() {
+ return kRembSendIntervallMs -
+ (TickTime::MillisecondTimestamp() - last_remb_time_);
+}
+
+WebRtc_Word32 VieRemb::Process() {
+ int64_t now = TickTime::MillisecondTimestamp();
+ if (now - last_remb_time_ < kRembSendIntervallMs)
+ return 0;
+
+ last_remb_time_ = now;
+
+ // Calculate total receive bitrate estimate.
+ list_crit_->Enter();
+ int total_bitrate = 0;
+ int num_bitrates = bitrates_.size();
+
+ if (num_bitrates == 0) {
+ list_crit_->Leave();
+ return 0;
+ }
+
+ // TODO(mflodman) Use std::vector and change RTP module API.
+ unsigned int* ssrcs = new unsigned int[num_bitrates];
+
+ int idx = 0;
+ for (SsrcBitrate::iterator it = bitrates_.begin(); it != bitrates_.end();
+ ++it, ++idx) {
+ total_bitrate += it->second;
+ ssrcs[idx] = it->first;
+ }
+
+ // Send a REMB packet.
+ RtpRtcp* sender = NULL;
+ if (!rtcp_sender_.empty()) {
+ sender = rtcp_sender_.front();
+ }
+ last_send_bitrate_ = total_bitrate;
+ list_crit_->Leave();
+
+ if (sender) {
+ sender->SetREMBData(total_bitrate, num_bitrates, ssrcs);
+ }
+ delete [] ssrcs;
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_remb.h b/trunk/src/video_engine/vie_remb.h
new file mode 100644
index 0000000..32d58a7
--- /dev/null
+++ b/trunk/src/video_engine/vie_remb.h
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// 1. Register a RtpRtcp module to include in the REMB packet.
+// 2. When UpdateBitrateEstimate is called for the first time for a SSRC, add it
+// to the map.
+// 3. Send a new REMB every kRembSendIntervallMs or if a lower bitrate estimate
+// for a specified SSRC.
+
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
+
+#include <list>
+#include <map>
+
+#include "modules/interface/module.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RtpRtcp;
+
+class VieRemb : public RtpRemoteBitrateObserver, public Module {
+ public:
+ explicit VieRemb(int engine_id);
+ ~VieRemb();
+
+ // Called to add a receive channel to include in the REMB packet.
+ void AddReceiveChannel(RtpRtcp* rtp_rtcp);
+
+ // Removes the specified channel from REMB estimate.
+ void RemoveReceiveChannel(RtpRtcp* rtp_rtcp);
+
+ // Called to add a module that can generate and send REMB RTCP.
+ void AddRembSender(RtpRtcp* rtp_rtcp);
+
+ // Removes a REMB RTCP sender.
+ void RemoveRembSender(RtpRtcp* rtp_rtcp);
+
+ // Called to add a send channel encoding and sending data, affected by
+ // received REMB packets.
+ void AddSendChannel(RtpRtcp* rtp_rtcp);
+
+ // Removes the specified channel from receiving REMB packet estimates.
+ void RemoveSendChannel(RtpRtcp* rtp_rtcp);
+
+ // Called every time there is a new bitrate estimate for the received stream
+ // with given SSRC. This call will trigger a new RTCP REMB packet if the
+ // bitrate estimate has decreased or if no RTCP REMB packet has been sent for
+ // a certain time interval.
+ // Implements RtpReceiveBitrateUpdate.
+ virtual void OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate);
+
+ // Called for every new receive REMB packet and distributes the estmate
+ // between all sending modules.
+ virtual void OnReceivedRemb(unsigned int bitrate);
+
+ // Implements Module.
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ private:
+ typedef std::list<RtpRtcp*> RtpModules;
+ typedef std::map<unsigned int, unsigned int> SsrcBitrate;
+
+ int engine_id_;
+ scoped_ptr<CriticalSectionWrapper> list_crit_;
+
+ // The last time a REMB was sent.
+ int64_t last_remb_time_;
+ int last_send_bitrate_;
+
+ // All RtpRtcp modules to include in the REMB packet.
+ RtpModules receive_modules_;
+
+ // All modules encoding and sending data.
+ RtpModules send_modules_;
+
+ // All modules that can send REMB RTCP.
+ RtpModules rtcp_sender_;
+
+ // The last bitrate update for each SSRC.
+ SsrcBitrate bitrates_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
diff --git a/trunk/src/video_engine/vie_remb_unittest.cc b/trunk/src/video_engine/vie_remb_unittest.cc
new file mode 100644
index 0000000..a842103
--- /dev/null
+++ b/trunk/src/video_engine/vie_remb_unittest.cc
@@ -0,0 +1,335 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This file includes unit tests for ViERemb.
+
+#include <gmock/gmock.h>
+#include <gtest/gtest.h>
+#if defined(_WIN32)
+#include <windows.h>
+#endif
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "video_engine/vie_remb.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Return;
+
+namespace webrtc {
+
+class ViERembTest : public ::testing::Test {
+ protected:
+ virtual void SetUp() {
+ vie_remb_.reset(new VieRemb(1234));
+ }
+ scoped_ptr<VieRemb> vie_remb_;
+
+ void TestSleep(unsigned int time_ms) {
+#if defined(_WIN32)
+ ::Sleep(time_ms);
+#else
+ usleep(time_ms * 1000);
+#endif
+ }
+};
+
+TEST_F(ViERembTest, OneModuleTestForSendingRemb)
+{
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddSendChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+
+ const unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ EXPECT_CALL(rtp, RemoteSSRC())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ // TODO(mflodman) Add fake clock and remove the lowered bitrate below.
+ TestSleep(1010);
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+ .Times(1);
+ vie_remb_->Process();
+
+ // Lower bitrate to send another REMB packet.
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate - 100);
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, 1, _))
+ .Times(1);
+ vie_remb_->Process();
+
+ vie_remb_->RemoveReceiveChannel(&rtp);
+ vie_remb_->RemoveSendChannel(&rtp);
+ vie_remb_->RemoveRembSender(&rtp);
+}
+
+TEST_F(ViERembTest, LowerEstimateToSendRemb)
+{
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddSendChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ EXPECT_CALL(rtp, RemoteSSRC())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ // Lower the estimate with more than 3% to trigger a call to SetREMBData right
+ // away.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ vie_remb_->Process();
+}
+
+TEST_F(ViERembTest, VerifyCombinedBitrateEstimate)
+{
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddSendChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate[] = { 456, 789 };
+ unsigned int ssrc[] = { 1234, 5678 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
+ EXPECT_CALL(rtp_0, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1] + 100);
+ EXPECT_CALL(rtp_1, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[1]));
+
+ // Lower the estimate to trigger a callback.
+ int total_bitrate = bitrate_estimate[0] + bitrate_estimate[1];
+ EXPECT_CALL(rtp_0, SetREMBData(total_bitrate, 2, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
+ vie_remb_->Process();
+
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveSendChannel(&rtp_0);
+ vie_remb_->RemoveRembSender(&rtp_0);
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+}
+
+// Add two senders, and insert a received REMB estimate. Both sending channels
+// should get half of the received value.
+TEST_F(ViERembTest, IncomingRemb)
+{
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddSendChannel(&rtp_0);
+ vie_remb_->AddSendChannel(&rtp_1);
+
+ const unsigned int bitrate_estimate = 1200;
+
+ // Fake received REMB and verify both modules get half of the bitrate.
+ EXPECT_CALL(rtp_0, SetMaximumBitrateEstimate(bitrate_estimate/2))
+ .Times(1);
+ EXPECT_CALL(rtp_1, SetMaximumBitrateEstimate(bitrate_estimate/2))
+ .Times(1);
+ vie_remb_->OnReceivedRemb(bitrate_estimate);
+
+ // Remove one of the modules and verify the other module get the entire
+ // bitrate.
+ vie_remb_->RemoveSendChannel(&rtp_0);
+ EXPECT_CALL(rtp_0, SetMaximumBitrateEstimate(_))
+ .Times(0);
+ EXPECT_CALL(rtp_1, SetMaximumBitrateEstimate(bitrate_estimate))
+ .Times(1);
+ vie_remb_->OnReceivedRemb(bitrate_estimate);
+
+ vie_remb_->RemoveSendChannel(&rtp_1);
+}
+
+TEST_F(ViERembTest, NoRembForIncreasedBitrate)
+{
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddSendChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate[] = { 456, 789 };
+ unsigned int ssrc[] = { 1234, 5678 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
+ EXPECT_CALL(rtp_0, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
+ EXPECT_CALL(rtp_1, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[1]));
+
+ // Trigger a first call to have a running state.
+ // TODO(mflodman) Add fake clock.
+ TestSleep(1010);
+ EXPECT_CALL(rtp_0,
+ SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2, _))
+ .Times(1);
+ vie_remb_->Process();
+
+ // Increased estimate shouldn't trigger a callback right away.
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0] + 1);
+ EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
+ .Times(0);
+
+ // Decresing the estimate less than 3% shouldn't trigger a new callback.
+ int lower_estimate = bitrate_estimate[0] * 98 / 100;
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], lower_estimate);
+ EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
+ .Times(0);
+
+ vie_remb_->Process();
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveSendChannel(&rtp_0);
+ vie_remb_->RemoveRembSender(&rtp_0);
+}
+
+TEST_F(ViERembTest, ChangeSendRtpModule)
+{
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddSendChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate[] = { 456, 789 };
+ unsigned int ssrc[] = { 1234, 5678 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
+ EXPECT_CALL(rtp_0, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
+ EXPECT_CALL(rtp_1, RemoteSSRC())
+ .Times(AnyNumber())
+ .WillRepeatedly(Return(ssrc[1]));
+
+ // Decrease estimate to trigger a REMB.
+ bitrate_estimate[0] = bitrate_estimate[0] - 100;
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2,
+ _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
+ vie_remb_->Process();
+
+ // Remove the sending module, add it again -> should get remb on the second
+ // module.
+ vie_remb_->RemoveSendChannel(&rtp_0);
+ vie_remb_->RemoveRembSender(&rtp_0);
+ vie_remb_->AddSendChannel(&rtp_1);
+ vie_remb_->AddRembSender(&rtp_1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
+
+ bitrate_estimate[1] = bitrate_estimate[1] - 100;
+ EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2,
+ _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
+ vie_remb_->Process();
+
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+ vie_remb_->RemoveSendChannel(&rtp_1);
+}
+
+TEST_F(ViERembTest, OnlyOneRembForDoubleProcess)
+{
+ MockRtpRtcp rtp;
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234 };
+
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddSendChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ EXPECT_CALL(rtp, RemoteSSRC())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ // Lower the estimate, should trigger a call to SetREMBData right away.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ vie_remb_->Process();
+
+ // Call Process again, this should not trigger a new callback.
+ EXPECT_CALL(rtp, SetREMBData(_, _, _))
+ .Times(0);
+ vie_remb_->Process();
+ vie_remb_->RemoveReceiveChannel(&rtp);
+ vie_remb_->RemoveSendChannel(&rtp);
+ vie_remb_->RemoveRembSender(&rtp);
+}
+
+TEST_F(ViERembTest, NoOnReceivedBitrateChangedCall)
+{
+ MockRtpRtcp rtp;
+ EXPECT_CALL(rtp, RemoteSSRC())
+ .WillRepeatedly(Return(1234));
+
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddSendChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+ // TODO(mflodman) Add fake clock.
+ TestSleep(1010);
+ // No bitrate estimate given, no callback expected.
+ EXPECT_CALL(rtp, SetREMBData(_, _, _))
+ .Times(0);
+ vie_remb_->Process();
+
+ vie_remb_->RemoveReceiveChannel(&rtp);
+ vie_remb_->RemoveSendChannel(&rtp);
+ vie_remb_->RemoveRembSender(&rtp);
+}
+
+TEST_F(ViERembTest, NoSendingRtpModule)
+{
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234 };
+
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ EXPECT_CALL(rtp, RemoteSSRC())
+ .WillRepeatedly(Return(ssrc[0]));
+
+ // Lower the estimate. This should normally trigger a callback, but not now
+ // since we have no sending module.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(_, _, _))
+ .Times(0);
+ vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
+ vie_remb_->Process();
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_impl.cc b/trunk/src/video_engine/vie_render_impl.cc
new file mode 100644
index 0000000..e37e5d4
--- /dev/null
+++ b/trunk/src/video_engine/vie_render_impl.cc
@@ -0,0 +1,398 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_render_impl.h"
+
+#include "engine_configurations.h"
+#include "modules/video_render/main/interface/video_render.h"
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_capturer.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_frame_provider_base.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_input_manager.h"
+#include "video_engine/vie_render_manager.h"
+#include "video_engine/vie_renderer.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+ViERender* ViERender::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViERenderImpl* vie_render_impl = vie_impl;
+ // Increase ref count.
+ (*vie_render_impl)++;
+ return vie_render_impl;
+#else
+ return NULL;
+#endif
+}
+
+int ViERenderImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViERender::Release()");
+ // Decrease ref count
+ (*this)--;
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViERender release too many times");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViERender reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViERenderImpl::ViERenderImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViERenderImpl::ViERenderImpl() Ctor");
+}
+
+ViERenderImpl::~ViERenderImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViERenderImpl::~ViERenderImpl() Dtor");
+}
+
+int ViERenderImpl::RegisterVideoRenderModule(
+ VideoRender& render_module) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s (&render_module: %p)", __FUNCTION__, &render_module);
+ if (shared_data_->render_manager()->RegisterVideoRenderModule(
+ render_module) != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::DeRegisterVideoRenderModule(
+ VideoRender& render_module) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s (&render_module: %p)", __FUNCTION__, &render_module);
+ if (shared_data_->render_manager()->DeRegisterVideoRenderModule(
+ render_module) != 0) {
+ // Error logging is done in ViERenderManager::DeRegisterVideoRenderModule.
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::AddRenderer(const int render_id, void* window,
+ const unsigned int z_order, const float left,
+ const float top, const float right,
+ const float bottom) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s (render_id: %d, window: 0x%p, z_order: %u, left: %f, "
+ "top: %f, right: %f, bottom: %f)",
+ __FUNCTION__, render_id, window, z_order, left, top, right,
+ bottom);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ {
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ if (rs.Renderer(render_id)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - Renderer already exist %d.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderAlreadyExists);
+ return -1;
+ }
+ }
+ if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
+ // This is a channel.
+ ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
+ ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
+ if (!frame_provider) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
+ render_id, window, z_order, left, top, right, bottom);
+ if (!renderer) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return frame_provider->RegisterFrameCallback(render_id, renderer);
+ } else {
+ // Camera or file.
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
+ if (!frame_provider) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
+ render_id, window, z_order, left, top, right, bottom);
+ if (!renderer) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return frame_provider->RegisterFrameCallback(render_id, renderer);
+ }
+}
+
+int ViERenderImpl::RemoveRenderer(const int render_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s(render_id: %d)", __FUNCTION__, render_id);
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+
+ ViERenderer* renderer = NULL;
+ {
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ renderer = rs.Renderer(render_id);
+ if (!renderer) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s No render exist with render_id: %d", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ // Leave the scope lock since we don't want to lock two managers
+ // simultanousely.
+ }
+ if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
+ // This is a channel.
+ ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
+ ViEChannel* channel = cm.Channel(render_id);
+ if (!channel) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s: no channel with id %d exists ", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ channel->DeregisterFrameCallback(renderer);
+ } else {
+ // Provider owned by inputmanager, i.e. file or capture device.
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* provider = is.FrameProvider(render_id);
+ if (!provider) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo,
+ ViEId(shared_data_->instance_id()),
+ "%s: no provider with id %d exists ", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ provider->DeregisterFrameCallback(renderer);
+ }
+ if (shared_data_->render_manager()->RemoveRenderStream(render_id) != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::StartRender(const int render_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s(channel: %d)", __FUNCTION__, render_id);
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(render_id);
+ if (!renderer) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s: No renderer with render Id %d exist.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ if (renderer->StartRender() != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::StopRender(const int render_id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s(channel: %d)", __FUNCTION__, render_id);
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(render_id);
+ if (!renderer) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s: No renderer with render_id %d exist.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ if (renderer->StopRender() != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::ConfigureRender(int render_id, const unsigned int z_order,
+ const float left, const float top,
+ const float right, const float bottom) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s(channel: %d)", __FUNCTION__, render_id);
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(render_id);
+ if (!renderer) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s: No renderer with render_id %d exist.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+
+ if (renderer->ConfigureRenderer(z_order, left, top, right, bottom) != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::MirrorRenderStream(const int render_id, const bool enable,
+ const bool mirror_xaxis,
+ const bool mirror_yaxis) {
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ ViERenderer* renderer = rs.Renderer(render_id);
+ if (!renderer) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s: No renderer with render_id %d exist.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ if (renderer->EnableMirroring(render_id, enable, mirror_xaxis, mirror_yaxis)
+ != 0) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERenderImpl::AddRenderer(const int render_id,
+ RawVideoType video_input_format,
+ ExternalRenderer* external_renderer) {
+ // Check if the client requested a format that we can convert the frames to.
+ if (video_input_format != kVideoI420 &&
+ video_input_format != kVideoYV12 &&
+ video_input_format != kVideoYUY2 &&
+ video_input_format != kVideoUYVY &&
+ video_input_format != kVideoARGB &&
+ video_input_format != kVideoRGB24 &&
+ video_input_format != kVideoRGB565 &&
+ video_input_format != kVideoARGB4444 &&
+ video_input_format != kVideoARGB1555) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), render_id),
+ "%s: Unsupported video frame format requested",
+ __FUNCTION__, render_id);
+ shared_data_->SetLastError(kViERenderInvalidFrameFormat);
+ return -1;
+ }
+ if (!shared_data_->Initialized()) {
+ shared_data_->SetLastError(kViENotInitialized);
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - ViE instance %d not initialized", __FUNCTION__,
+ shared_data_->instance_id());
+ return -1;
+ }
+ {
+ // Verify the renderer doesn't exist.
+ ViERenderManagerScoped rs(*(shared_data_->render_manager()));
+ if (rs.Renderer(render_id)) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s - Renderer already exist %d.", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderAlreadyExists);
+ return -1;
+ }
+ }
+ if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
+ // This is a channel.
+ ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
+ ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
+ if (!frame_provider) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
+ render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
+ if (!renderer) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ if (renderer->SetExternalRenderer(render_id, video_input_format,
+ external_renderer) == -1) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+
+ return frame_provider->RegisterFrameCallback(render_id, renderer);
+ } else {
+ // Camera or file.
+ ViEInputManagerScoped is(*(shared_data_->input_manager()));
+ ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
+ if (!frame_provider) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+ "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
+ render_id);
+ shared_data_->SetLastError(kViERenderInvalidRenderId);
+ return -1;
+ }
+ ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
+ render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
+ if (!renderer) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ if (renderer->SetExternalRenderer(render_id, video_input_format,
+ external_renderer) == -1) {
+ shared_data_->SetLastError(kViERenderUnknownError);
+ return -1;
+ }
+ return frame_provider->RegisterFrameCallback(render_id, renderer);
+ }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_impl.h b/trunk/src/video_engine/vie_render_impl.h
new file mode 100644
index 0000000..0c74b3d
--- /dev/null
+++ b/trunk/src/video_engine/vie_render_impl.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
+
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_render.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViERenderImpl
+ : public ViERender,
+ public ViERefCount {
+ public:
+ // Implements ViERender
+ virtual int Release();
+ virtual int RegisterVideoRenderModule(VideoRender& render_module);
+ virtual int DeRegisterVideoRenderModule(VideoRender& render_module);
+ virtual int AddRenderer(const int render_id, void* window,
+ const unsigned int z_order, const float left,
+ const float top, const float right,
+ const float bottom);
+ virtual int RemoveRenderer(const int render_id);
+ virtual int StartRender(const int render_id);
+ virtual int StopRender(const int render_id);
+ virtual int ConfigureRender(int render_id, const unsigned int z_order,
+ const float left, const float top,
+ const float right, const float bottom);
+ virtual int MirrorRenderStream(const int render_id, const bool enable,
+ const bool mirror_xaxis,
+ const bool mirror_yaxis);
+ virtual int AddRenderer(const int render_id, RawVideoType video_input_format,
+ ExternalRenderer* renderer);
+
+ protected:
+ ViERenderImpl(ViESharedData* shared_data);
+ virtual ~ViERenderImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
diff --git a/trunk/src/video_engine/vie_render_manager.cc b/trunk/src/video_engine/vie_render_manager.cc
new file mode 100644
index 0000000..810c079
--- /dev/null
+++ b/trunk/src/video_engine/vie_render_manager.cc
@@ -0,0 +1,230 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_render_manager.h"
+
+#include "engine_configurations.h"
+#include "modules/video_render/main/interface/video_render.h"
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "system_wrappers/interface/rw_lock_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_renderer.h"
+
+namespace webrtc {
+
+ViERenderManagerScoped::ViERenderManagerScoped(
+ const ViERenderManager& vie_render_manager)
+ : ViEManagerScopedBase(vie_render_manager) {
+}
+
+ViERenderer* ViERenderManagerScoped::Renderer(WebRtc_Word32 render_id) const {
+ return static_cast<const ViERenderManager*>(vie_manager_)->ViERenderPtr(
+ render_id);
+}
+
+ViERenderManager::ViERenderManager(WebRtc_Word32 engine_id)
+ : list_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ engine_id_(engine_id),
+ use_external_render_module_(false) {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id),
+ "ViERenderManager::ViERenderManager(engine_id: %d) - "
+ "Constructor", engine_id);
+}
+
+ViERenderManager::~ViERenderManager() {
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
+ "ViERenderManager Destructor, engine_id: %d", engine_id_);
+
+ while (stream_to_vie_renderer_.Size() != 0) {
+ MapItem* item = stream_to_vie_renderer_.First();
+ assert(item);
+ const WebRtc_Word32 render_id = item->GetId();
+ // The renderer is delete in RemoveRenderStream.
+ item = NULL;
+ RemoveRenderStream(render_id);
+ }
+}
+
+WebRtc_Word32 ViERenderManager::RegisterVideoRenderModule(
+ VideoRender& render_module) {
+ // See if there is already a render module registered for the window that
+ // the registrant render module is associated with.
+ VideoRender* current_module = FindRenderModule(render_module.Window());
+ if (current_module) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "A module is already registered for this window (window=%p, "
+ "current module=%p, registrant module=%p.",
+ render_module.Window(), current_module, &render_module);
+ return -1;
+ }
+
+ // Register module.
+ render_list_.PushBack(static_cast<void*>(&render_module));
+ use_external_render_module_ = true;
+ return 0;
+}
+
+WebRtc_Word32 ViERenderManager::DeRegisterVideoRenderModule(
+ VideoRender& render_module) {
+ // Check if there are streams in the module.
+ WebRtc_UWord32 n_streams = render_module.GetNumIncomingRenderStreams();
+ if (n_streams != 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "There are still %d streams in this module, cannot "
+ "de-register", n_streams);
+ return -1;
+ }
+
+ // Erase the render module from the map.
+ ListItem* list_item = NULL;
+ bool found = false;
+ for (list_item = render_list_.First(); list_item != NULL;
+ list_item = render_list_.Next(list_item)) {
+ if (&render_module == static_cast<VideoRender*>(list_item->GetItem())) {
+ // We've found our renderer.
+ render_list_.Erase(list_item);
+ found = true;
+ break;
+ }
+ }
+ if (!found) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "Module not registered");
+ return -1;
+ }
+ return 0;
+}
+
+ViERenderer* ViERenderManager::AddRenderStream(const WebRtc_Word32 render_id,
+ void* window,
+ const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) {
+ CriticalSectionScoped cs(list_cs_.get());
+
+ if (stream_to_vie_renderer_.Find(render_id) != NULL) {
+ // This stream is already added to a renderer, not allowed!
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "Render stream already exists");
+ return NULL;
+ }
+
+ // Get the render module for this window.
+ VideoRender* render_module = FindRenderModule(window);
+ if (render_module == NULL) {
+ // No render module for this window, create a new one.
+ render_module = VideoRender::CreateVideoRender(ViEModuleId(engine_id_, -1),
+ window, false);
+ if (!render_module) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
+ "Could not create new render module");
+ return NULL;
+ }
+ render_list_.PushBack(static_cast<void*>(render_module));
+ }
+
+ ViERenderer* vie_renderer = ViERenderer::CreateViERenderer(render_id,
+ engine_id_,
+ *render_module,
+ *this, z_order,
+ left, top, right,
+ bottom);
+ if (!vie_renderer) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, render_id),
+ "Could not create new render stream");
+ return NULL;
+ }
+ stream_to_vie_renderer_.Insert(render_id, vie_renderer);
+ return vie_renderer;
+}
+
+WebRtc_Word32 ViERenderManager::RemoveRenderStream(
+ const WebRtc_Word32 render_id) {
+ // We need exclusive right to the items in the render manager to delete a
+ // stream.
+ ViEManagerWriteScoped(*this);
+
+ CriticalSectionScoped cs(list_cs_.get());
+ MapItem* map_item = stream_to_vie_renderer_.Find(render_id);
+ if (!map_item) {
+ // No such stream
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, ViEId(engine_id_),
+ "No renderer for this stream found, channel_id");
+ return 0;
+ }
+
+ ViERenderer* vie_renderer = static_cast<ViERenderer*>(map_item->GetItem());
+ assert(vie_renderer);
+
+ // Get the render module pointer for this vie_render object.
+ VideoRender& renderer = vie_renderer->RenderModule();
+
+ // Delete the vie_render.
+ // This deletes the stream in the render module.
+ delete vie_renderer;
+
+ // Remove from the stream map.
+ stream_to_vie_renderer_.Erase(map_item);
+
+ // Check if there are other streams in the module.
+ if (!use_external_render_module_ &&
+ renderer.GetNumIncomingRenderStreams() == 0) {
+ // Erase the render module from the map.
+ ListItem* list_item = NULL;
+ for (list_item = render_list_.First(); list_item != NULL;
+ list_item = render_list_.Next(list_item)) {
+ if (&renderer == static_cast<VideoRender*>(list_item->GetItem())) {
+ // We've found our renderer.
+ render_list_.Erase(list_item);
+ break;
+ }
+ }
+ // Destroy the module.
+ VideoRender::DestroyVideoRender(&renderer);
+ }
+ return 0;
+}
+
+VideoRender* ViERenderManager::FindRenderModule(void* window) {
+ VideoRender* renderer = NULL;
+ ListItem* list_item = NULL;
+ for (list_item = render_list_.First(); list_item != NULL;
+ list_item = render_list_.Next(list_item)) {
+ renderer = static_cast<VideoRender*>(list_item->GetItem());
+ if (renderer == NULL) {
+ break;
+ }
+ if (renderer->Window() == window) {
+ // We've found the render module.
+ break;
+ }
+ renderer = NULL;
+ }
+ return renderer;
+}
+
+ViERenderer* ViERenderManager::ViERenderPtr(WebRtc_Word32 render_id) const {
+ ViERenderer* renderer = NULL;
+ MapItem* map_item = stream_to_vie_renderer_.Find(render_id);
+ if (!map_item) {
+ // No such stream in any renderer.
+ return NULL;
+ }
+ renderer = static_cast<ViERenderer*>(map_item->GetItem());
+
+ return renderer;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_manager.h b/trunk/src/video_engine/vie_render_manager.h
new file mode 100644
index 0000000..242b63a
--- /dev/null
+++ b/trunk/src/video_engine/vie_render_manager.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
+
+#include "system_wrappers/interface/list_wrapper.h"
+#include "system_wrappers/interface/map_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "video_engine/vie_manager_base.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RWLockWrapper;
+class VideoRender;
+class VideoRenderCallback;
+class ViERenderer;
+
+class ViERenderManager : private ViEManagerBase {
+ friend class ViERenderManagerScoped;
+ public:
+ explicit ViERenderManager(WebRtc_Word32 engine_id);
+ ~ViERenderManager();
+
+ WebRtc_Word32 RegisterVideoRenderModule(VideoRender& render_module);
+ WebRtc_Word32 DeRegisterVideoRenderModule(VideoRender& render_module);
+
+ ViERenderer* AddRenderStream(const WebRtc_Word32 render_id,
+ void* window,
+ const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ WebRtc_Word32 RemoveRenderStream(WebRtc_Word32 render_id);
+
+ private:
+ // Returns a pointer to the render module if it exists in the render list.
+ // Assumed protected.
+ VideoRender* FindRenderModule(void* window);
+
+ // Methods used by ViERenderScoped.
+ ViERenderer* ViERenderPtr(WebRtc_Word32 render_id) const;
+
+ scoped_ptr<CriticalSectionWrapper> list_cs_;
+ WebRtc_Word32 engine_id_;
+ MapWrapper stream_to_vie_renderer_; // Protected by ViEManagerBase.
+ ListWrapper render_list_;
+ bool use_external_render_module_;
+};
+
+class ViERenderManagerScoped: private ViEManagerScopedBase {
+ public:
+ explicit ViERenderManagerScoped(const ViERenderManager& vie_render_manager);
+
+ // Returns a pointer to the ViERender object.
+ ViERenderer* Renderer(WebRtc_Word32 render_id) const;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_renderer.cc b/trunk/src/video_engine/vie_renderer.cc
new file mode 100644
index 0000000..7e56b7a
--- /dev/null
+++ b/trunk/src/video_engine/vie_renderer.cc
@@ -0,0 +1,235 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_renderer.h"
+
+#include "common_video/libyuv/include/libyuv.h"
+#include "modules/video_render/main/interface/video_render.h"
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "video_engine/vie_render_manager.h"
+
+namespace webrtc {
+
+ViERenderer* ViERenderer::CreateViERenderer(const WebRtc_Word32 render_id,
+ const WebRtc_Word32 engine_id,
+ VideoRender& render_module,
+ ViERenderManager& render_manager,
+ const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) {
+ ViERenderer* self = new ViERenderer(render_id, engine_id, render_module,
+ render_manager);
+ if (!self || self->Init(z_order, left, top, right, bottom) != 0) {
+ delete self;
+ self = NULL;
+ }
+ return self;
+}
+
+ViERenderer::~ViERenderer(void) {
+ if (render_callback_)
+ render_module_.DeleteIncomingRenderStream(render_id_);
+
+ if (incoming_external_callback_)
+ delete incoming_external_callback_;
+}
+
+WebRtc_Word32 ViERenderer::StartRender() {
+ return render_module_.StartRender(render_id_);
+}
+WebRtc_Word32 ViERenderer::StopRender() {
+ return render_module_.StopRender(render_id_);
+}
+
+WebRtc_Word32 ViERenderer::GetLastRenderedFrame(const WebRtc_Word32 renderID,
+ VideoFrame& video_frame) {
+ return render_module_.GetLastRenderedFrame(renderID, video_frame);
+}
+
+WebRtc_Word32 ViERenderer::ConfigureRenderer(const unsigned int z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) {
+ return render_module_.ConfigureRenderer(render_id_, z_order, left, top, right,
+ bottom);
+}
+
+VideoRender& ViERenderer::RenderModule() {
+ return render_module_;
+}
+
+WebRtc_Word32 ViERenderer::EnableMirroring(const WebRtc_Word32 render_id,
+ const bool enable,
+ const bool mirror_xaxis,
+ const bool mirror_yaxis) {
+ return render_module_.MirrorRenderStream(render_id, enable, mirror_xaxis,
+ mirror_yaxis);
+}
+
+WebRtc_Word32 ViERenderer::SetTimeoutImage(const VideoFrame& timeout_image,
+ const WebRtc_Word32 timeout_value) {
+ return render_module_.SetTimeoutImage(render_id_, timeout_image,
+ timeout_value);
+}
+
+WebRtc_Word32 ViERenderer::SetRenderStartImage(const VideoFrame& start_image) {
+ return render_module_.SetStartImage(render_id_, start_image);
+}
+
+WebRtc_Word32 ViERenderer::SetExternalRenderer(
+ const WebRtc_Word32 render_id,
+ RawVideoType video_input_format,
+ ExternalRenderer* external_renderer) {
+ if (!incoming_external_callback_)
+ return -1;
+
+ incoming_external_callback_->SetViEExternalRenderer(external_renderer,
+ video_input_format);
+ return render_module_.AddExternalRenderCallback(render_id,
+ incoming_external_callback_);
+}
+
+ViERenderer::ViERenderer(const WebRtc_Word32 render_id,
+ const WebRtc_Word32 engine_id,
+ VideoRender& render_module,
+ ViERenderManager& render_manager)
+ : render_id_(render_id),
+ engine_id_(engine_id),
+ render_module_(render_module),
+ render_manager_(render_manager),
+ render_callback_(NULL),
+ incoming_external_callback_(new ViEExternalRendererImpl()) {
+}
+
+WebRtc_Word32 ViERenderer::Init(const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom) {
+ render_callback_ =
+ static_cast<VideoRenderCallback*>(render_module_.AddIncomingRenderStream(
+ render_id_, z_order, left, top, right, bottom));
+ if (!render_callback_) {
+ // Logging done.
+ return -1;
+ }
+ return 0;
+}
+
+void ViERenderer::DeliverFrame(int id,
+ VideoFrame& video_frame,
+ int num_csrcs,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
+ render_callback_->RenderFrame(render_id_, video_frame);
+}
+
+void ViERenderer::DelayChanged(int id, int frame_delay) {}
+
+int ViERenderer::GetPreferedFrameSettings(int& width,
+ int& height,
+ int& frame_rate) {
+ return -1;
+}
+
+void ViERenderer::ProviderDestroyed(int id) {
+ // Remove the render stream since the provider is destroyed.
+ render_manager_.RemoveRenderStream(render_id_);
+}
+
+ViEExternalRendererImpl::ViEExternalRendererImpl()
+ : external_renderer_(NULL),
+ external_renderer_format_(kVideoUnknown),
+ external_renderer_width_(0),
+ external_renderer_height_(0),
+ converted_frame_(new VideoFrame()){
+}
+
+int ViEExternalRendererImpl::SetViEExternalRenderer(
+ ExternalRenderer* external_renderer,
+ RawVideoType video_input_format) {
+ external_renderer_ = external_renderer;
+ external_renderer_format_ = video_input_format;
+ return 0;
+}
+
+WebRtc_Word32 ViEExternalRendererImpl::RenderFrame(
+ const WebRtc_UWord32 stream_id,
+ VideoFrame& video_frame) {
+ VideoFrame* out_frame = converted_frame_.get();
+
+ // Convert to requested format.
+ VideoType type =
+ RawVideoTypeToCommonVideoVideoType(external_renderer_format_);
+ int buffer_size = CalcBufferSize(type, video_frame.Width(),
+ video_frame.Height());
+ if (buffer_size <= 0) {
+ // Unsupported video format.
+ assert(false);
+ return -1;
+ }
+ converted_frame_->VerifyAndAllocate(buffer_size);
+
+ switch (external_renderer_format_) {
+ case kVideoI420:
+ out_frame = &video_frame;
+ break;
+ case kVideoYV12:
+ case kVideoYUY2:
+ case kVideoUYVY:
+ case kVideoARGB:
+ case kVideoRGB24:
+ {
+ ConvertFromI420(video_frame.Buffer(), video_frame.Width(), type, 0,
+ video_frame.Width(), video_frame.Height(),
+ converted_frame_->Buffer());
+ }
+ break;
+ case kVideoIYUV:
+ // no conversion available
+ break;
+ case kVideoRGB565:
+ ConvertI420ToRGB565(video_frame.Buffer(), converted_frame_->Buffer(),
+ video_frame.Width(), video_frame.Height());
+ break;
+ case kVideoARGB4444:
+ ConvertI420ToARGB4444(video_frame.Buffer(), converted_frame_->Buffer(),
+ video_frame.Width(), video_frame.Height(), 0);
+ break;
+ case kVideoARGB1555 :
+ ConvertI420ToARGB1555(video_frame.Buffer(), converted_frame_->Buffer(),
+ video_frame.Width(), video_frame.Height(), 0);
+ break;
+ default:
+ assert(false);
+ out_frame = NULL;
+ break;
+ }
+
+ if (external_renderer_width_ != video_frame.Width() ||
+ external_renderer_height_ != video_frame.Height()) {
+ external_renderer_width_ = video_frame.Width();
+ external_renderer_height_ = video_frame.Height();
+ external_renderer_->FrameSizeChange(external_renderer_width_,
+ external_renderer_height_, stream_id);
+ }
+
+ if (out_frame) {
+ external_renderer_->DeliverFrame(out_frame->Buffer(),
+ out_frame->Length(),
+ video_frame.TimeStamp(),
+ video_frame.RenderTimeMs());
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_renderer.h b/trunk/src/video_engine/vie_renderer.h
new file mode 100644
index 0000000..028d72b
--- /dev/null
+++ b/trunk/src/video_engine/vie_renderer.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
+
+#include "modules/video_render/main/interface/video_render_defines.h"
+#include "system_wrappers/interface/map_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "video_engine/include/vie_render.h"
+#include "video_engine/vie_frame_provider_base.h"
+
+namespace webrtc {
+
+class VideoRender;
+class VideoRenderCallback;
+class ViERenderManager;
+
+class ViEExternalRendererImpl : public VideoRenderCallback {
+ public:
+ ViEExternalRendererImpl();
+ virtual ~ViEExternalRendererImpl() {}
+
+ int SetViEExternalRenderer(ExternalRenderer* external_renderer,
+ RawVideoType video_input_format);
+
+ // Implements VideoRenderCallback.
+ virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 stream_id,
+ VideoFrame& video_frame);
+
+ private:
+ ExternalRenderer* external_renderer_;
+ RawVideoType external_renderer_format_;
+ WebRtc_UWord32 external_renderer_width_;
+ WebRtc_UWord32 external_renderer_height_;
+ scoped_ptr<VideoFrame> converted_frame_;
+};
+
+class ViERenderer: public ViEFrameCallback {
+ public:
+ static ViERenderer* CreateViERenderer(const WebRtc_Word32 render_id,
+ const WebRtc_Word32 engine_id,
+ VideoRender& render_module,
+ ViERenderManager& render_manager,
+ const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+ ~ViERenderer(void);
+
+ WebRtc_Word32 StartRender();
+ WebRtc_Word32 StopRender();
+
+ WebRtc_Word32 GetLastRenderedFrame(const WebRtc_Word32 renderID,
+ VideoFrame& video_frame);
+
+ WebRtc_Word32 ConfigureRenderer(const unsigned int z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ VideoRender& RenderModule();
+
+ WebRtc_Word32 EnableMirroring(const WebRtc_Word32 render_id,
+ const bool enable,
+ const bool mirror_xaxis,
+ const bool mirror_yaxis);
+
+ WebRtc_Word32 SetTimeoutImage(const VideoFrame& timeout_image,
+ const WebRtc_Word32 timeout_value);
+ WebRtc_Word32 SetRenderStartImage(const VideoFrame& start_image);
+ WebRtc_Word32 SetExternalRenderer(const WebRtc_Word32 render_id,
+ RawVideoType video_input_format,
+ ExternalRenderer* external_renderer);
+
+ private:
+ ViERenderer(const WebRtc_Word32 render_id, const WebRtc_Word32 engine_id,
+ VideoRender& render_module,
+ ViERenderManager& render_manager);
+
+ WebRtc_Word32 Init(const WebRtc_UWord32 z_order,
+ const float left,
+ const float top,
+ const float right,
+ const float bottom);
+
+ // Implement ViEFrameCallback
+ virtual void DeliverFrame(int id,
+ VideoFrame& video_frame,
+ int num_csrcs = 0,
+ const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
+ virtual void DelayChanged(int id, int frame_delay);
+ virtual int GetPreferedFrameSettings(int& width,
+ int& height,
+ int& frame_rate);
+ virtual void ProviderDestroyed(int id);
+
+ WebRtc_UWord32 render_id_;
+ WebRtc_Word32 engine_id_;
+ VideoRender& render_module_;
+ ViERenderManager& render_manager_;
+ VideoRenderCallback* render_callback_;
+ ViEExternalRendererImpl* incoming_external_callback_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
diff --git a/trunk/src/video_engine/vie_rtp_rtcp_impl.cc b/trunk/src/video_engine/vie_rtp_rtcp_impl.cc
new file mode 100644
index 0000000..497481c
--- /dev/null
+++ b/trunk/src/video_engine/vie_rtp_rtcp_impl.cc
@@ -0,0 +1,957 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_engine/vie_rtp_rtcp_impl.h"
+
+#include "engine_configurations.h"
+#include "system_wrappers/interface/file_wrapper.h"
+#include "system_wrappers/interface/trace.h"
+#include "video_engine/include/vie_errors.h"
+#include "video_engine/vie_channel.h"
+#include "video_engine/vie_channel_manager.h"
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_encoder.h"
+#include "video_engine/vie_impl.h"
+#include "video_engine/vie_shared_data.h"
+
+namespace webrtc {
+
+// Helper methods for converting between module format and ViE API format.
+
+static RTCPMethod ViERTCPModeToRTCPMethod(ViERTCPMode api_mode) {
+ switch (api_mode) {
+ case kRtcpNone:
+ return kRtcpOff;
+
+ case kRtcpCompound_RFC4585:
+ return kRtcpCompound;
+
+ case kRtcpNonCompound_RFC5506:
+ return kRtcpNonCompound;
+ }
+ assert(false);
+ return kRtcpOff;
+}
+
+static ViERTCPMode RTCPMethodToViERTCPMode(RTCPMethod module_method) {
+ switch (module_method) {
+ case kRtcpOff:
+ return kRtcpNone;
+
+ case kRtcpCompound:
+ return kRtcpCompound_RFC4585;
+
+ case kRtcpNonCompound:
+ return kRtcpNonCompound_RFC5506;
+ }
+ assert(false);
+ return kRtcpNone;
+}
+
+static KeyFrameRequestMethod APIRequestToModuleRequest(
+ ViEKeyFrameRequestMethod api_method) {
+ switch (api_method) {
+ case kViEKeyFrameRequestNone:
+ return kKeyFrameReqFirRtp;
+
+ case kViEKeyFrameRequestPliRtcp:
+ return kKeyFrameReqPliRtcp;
+
+ case kViEKeyFrameRequestFirRtp:
+ return kKeyFrameReqFirRtp;
+
+ case kViEKeyFrameRequestFirRtcp:
+ return kKeyFrameReqFirRtcp;
+ }
+ assert(false);
+ return kKeyFrameReqFirRtp;
+}
+
+ViERTP_RTCP* ViERTP_RTCP::GetInterface(VideoEngine* video_engine) {
+#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
+ if (!video_engine) {
+ return NULL;
+ }
+ VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
+ ViERTP_RTCPImpl* vie_rtpimpl = vie_impl;
+ // Increase ref count.
+ (*vie_rtpimpl)++;
+ return vie_rtpimpl;
+#else
+ return NULL;
+#endif
+}
+
+int ViERTP_RTCPImpl::Release() {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
+ "ViERTP_RTCP::Release()");
+ // Decrease ref count.
+ (*this)--;
+
+ WebRtc_Word32 ref_count = GetCount();
+ if (ref_count < 0) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
+ "ViERTP_RTCP release too many times");
+ shared_data_->SetLastError(kViEAPIDoesNotExist);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
+ "ViERTP_RTCP reference count: %d", ref_count);
+ return ref_count;
+}
+
+ViERTP_RTCPImpl::ViERTP_RTCPImpl(ViESharedData* shared_data)
+ : shared_data_(shared_data) {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViERTP_RTCPImpl::ViERTP_RTCPImpl() Ctor");
+}
+
+ViERTP_RTCPImpl::~ViERTP_RTCPImpl() {
+ WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
+ "ViERTP_RTCPImpl::~ViERTP_RTCPImpl() Dtor");
+}
+
+int ViERTP_RTCPImpl::SetLocalSSRC(const int video_channel,
+ const unsigned int SSRC,
+ const StreamType usage,
+ const unsigned char simulcast_idx) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, SSRC: %d)", __FUNCTION__, video_channel, SSRC);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ // The channel doesn't exists
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetSSRC(SSRC, usage, simulcast_idx) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetRemoteSSRCType(const int videoChannel,
+ const StreamType usage,
+ const unsigned int SSRC) const {
+ WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideo,
+ ViEId(shared_data_->instance_id(), videoChannel),
+ "%s(channel: %d, usage:%d SSRC: 0x%x)",
+ __FUNCTION__, usage, videoChannel, SSRC);
+
+ // Get the channel
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* ptrViEChannel = cs.Channel(videoChannel);
+ if (ptrViEChannel == NULL) {
+ // The channel doesn't exists
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(shared_data_->instance_id(), videoChannel),
+ "%s: Channel %d doesn't exist",
+ __FUNCTION__, videoChannel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (ptrViEChannel->SetRemoteSSRCType(usage, SSRC) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetLocalSSRC(const int video_channel,
+ unsigned int& SSRC) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, SSRC: %d)", __FUNCTION__, video_channel, SSRC);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetLocalSSRC((WebRtc_UWord32&) SSRC) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRemoteSSRC(const int video_channel,
+ unsigned int& SSRC) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel, SSRC);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetRemoteSSRC(SSRC) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRemoteCSRCs(const int video_channel,
+ unsigned int CSRCs[kRtpCsrcSize]) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetRemoteCSRC(CSRCs) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetStartSequenceNumber(const int video_channel,
+ unsigned short sequence_number) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, sequence_number: %u)", __FUNCTION__,
+ video_channel, sequence_number);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d already sending.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpAlreadySending);
+ return -1;
+ }
+ if (vie_channel->SetStartSequenceNumber(sequence_number) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetRTCPStatus(const int video_channel,
+ const ViERTCPMode rtcp_mode) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, mode: %d)", __FUNCTION__, video_channel,
+ rtcp_mode);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+
+ RTCPMethod module_mode = ViERTCPModeToRTCPMethod(rtcp_mode);
+ if (vie_channel->SetRTCPMode(module_mode) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRTCPStatus(const int video_channel,
+ ViERTCPMode& rtcp_mode) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel, rtcp_mode);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ RTCPMethod module_mode = kRtcpOff;
+ if (vie_channel->GetRTCPMode(module_mode) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: could not get current RTCP mode", __FUNCTION__);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ rtcp_mode = RTCPMethodToViERTCPMode(module_mode);
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetRTCPCName(const int video_channel,
+ const char rtcp_cname[KMaxRTCPCNameLength]) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, name: %s)", __FUNCTION__, video_channel,
+ rtcp_cname);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d already sending.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpAlreadySending);
+ return -1;
+ }
+ if (vie_channel->SetRTCPCName(rtcp_cname) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRTCPCName(const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetRTCPCName(rtcp_cname) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRemoteRTCPCName(
+ const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetRemoteRTCPCName(rtcp_cname) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
+ const int video_channel,
+ const unsigned char sub_type,
+ unsigned int name,
+ const char* data,
+ unsigned short data_length_in_bytes) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, sub_type: %c, name: %d, data: x, length: %u)",
+ __FUNCTION__, video_channel, sub_type, name,
+ data_length_in_bytes);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (!vie_channel->Sending()) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d not sending", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpNotSending);
+ return -1;
+ }
+ RTCPMethod method;
+ if (vie_channel->GetRTCPMode(method) != 0 || method == kRtcpOff) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: RTCP disabled on channel %d.", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpRtcpDisabled);
+ return -1;
+ }
+ if (vie_channel->SendApplicationDefinedRTCPPacket(
+ sub_type, name, reinterpret_cast<const WebRtc_UWord8*>(data),
+ data_length_in_bytes) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetNACKStatus(const int video_channel, const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d)", __FUNCTION__, video_channel,
+ enable);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetNACKStatus(enable) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: failed for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+
+ // Update the encoder
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get encoder for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ vie_encoder->UpdateProtectionMethod();
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetFECStatus(const int video_channel, const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, payload_typeRED: %u, "
+ "payloadTypeFEC: %u)",
+ __FUNCTION__, video_channel, enable, payload_typeRED,
+ payload_typeFEC);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->SetFECStatus(enable, payload_typeRED,
+ payload_typeFEC) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: failed for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ // Update the encoder.
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get encoder for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ vie_encoder->UpdateProtectionMethod();
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetHybridNACKFECStatus(
+ const int video_channel,
+ const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, payload_typeRED: %u, "
+ "payloadTypeFEC: %u)",
+ __FUNCTION__, video_channel, enable, payload_typeRED,
+ payload_typeFEC);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+
+ // Update the channel status with hybrid NACK FEC mode.
+ if (vie_channel->SetHybridNACKFECStatus(enable, payload_typeRED,
+ payload_typeFEC) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: failed for channel %d", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+
+ // Update the encoder.
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get encoder for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ vie_encoder->UpdateProtectionMethod();
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetKeyFrameRequestMethod(
+ const int video_channel,
+ const ViEKeyFrameRequestMethod method) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, method: %d)", __FUNCTION__, video_channel,
+ method);
+
+ // Get the channel.
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ KeyFrameRequestMethod module_method = APIRequestToModuleRequest(method);
+ if (vie_channel->SetKeyFrameRequestMethod(module_method) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetTMMBRStatus(const int video_channel,
+ const bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d)", __FUNCTION__, video_channel,
+ enable);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->EnableTMMBR(enable) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::SetRembStatus(int video_channel, bool sender,
+ bool receiver) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "ViERTP_RTCPImpl::SetRembStatus(%d, %d, %d)", video_channel,
+ sender, receiver);
+ if (!shared_data_->channel_manager()->SetRembStatus(video_channel, sender,
+ receiver)) {
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetReceivedRTCPStatistics(const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter,
+ int& rtt_ms) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetReceivedRtcpStatistics(
+ static_cast<WebRtc_UWord16&>(fraction_lost),
+ static_cast<WebRtc_UWord32&>(cumulative_lost),
+ static_cast<WebRtc_UWord32&>(extended_max),
+ static_cast<WebRtc_UWord32&>(jitter),
+ static_cast<WebRtc_Word32&>(rtt_ms)) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetSentRTCPStatistics(const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter,
+ int& rtt_ms) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+
+ if (vie_channel->GetSendRtcpStatistics(
+ static_cast<WebRtc_UWord16&>(fraction_lost),
+ static_cast<WebRtc_UWord32&>(cumulative_lost),
+ static_cast<WebRtc_UWord32&>(extended_max),
+ static_cast<WebRtc_UWord32&>(jitter),
+ static_cast<WebRtc_Word32&>(rtt_ms)) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRTPStatistics(const int video_channel,
+ unsigned int& bytes_sent,
+ unsigned int& packets_sent,
+ unsigned int& bytes_received,
+ unsigned int& packets_received) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->GetRtpStatistics(
+ static_cast<WebRtc_UWord32&>(bytes_sent),
+ static_cast<WebRtc_UWord32&>(packets_sent),
+ static_cast<WebRtc_UWord32&>(bytes_received),
+ static_cast<WebRtc_UWord32&>(packets_received)) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetBandwidthUsage(const int video_channel,
+ unsigned int& total_bitrate_sent,
+ unsigned int& video_bitrate_sent,
+ unsigned int& fec_bitrate_sent,
+ unsigned int& nackBitrateSent) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ vie_channel->GetBandwidthUsage(
+ static_cast<WebRtc_UWord32&>(total_bitrate_sent),
+ static_cast<WebRtc_UWord32&>(video_bitrate_sent),
+ static_cast<WebRtc_UWord32&>(fec_bitrate_sent),
+ static_cast<WebRtc_UWord32&>(nackBitrateSent));
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetEstimatedSendBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+ if (!vie_encoder) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get encoder for channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ return vie_encoder->EstimatedSendBandwidth(
+ static_cast<WebRtc_UWord32*>(estimated_bandwidth));
+}
+
+int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Could not get channel %d", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ return vie_channel->GetEstimatedReceiveBandwidth(
+ static_cast<WebRtc_UWord32*>(estimated_bandwidth));
+}
+
+int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
+ const int video_channel,
+ bool enable,
+ const char unknown_payload_type,
+ const unsigned int delta_transmit_time_seconds) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, enable: %d, unknown_payload_type: %d, "
+ "deltaTransmitTimeMS: %ul)",
+ __FUNCTION__, video_channel, enable,
+ static_cast<int>(unknown_payload_type),
+ delta_transmit_time_seconds);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__,
+ video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ WebRtc_UWord16 delta_transmit_time_ms = 1000 * delta_transmit_time_seconds;
+ if (vie_channel->SetKeepAliveStatus(enable, unknown_payload_type,
+ delta_transmit_time_ms) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::GetRTPKeepAliveStatus(
+ const int video_channel,
+ bool& enabled,
+ char& unknown_payload_type,
+ unsigned int& delta_transmit_time_seconds) const {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+
+ WebRtc_UWord16 delta_time_ms = 0;
+ int ret_val = vie_channel->GetKeepAliveStatus(enabled, unknown_payload_type,
+ delta_time_ms);
+ delta_transmit_time_seconds = delta_time_ms / 1000;
+ if (ret_val != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ }
+ return ret_val;
+}
+
+int ViERTP_RTCPImpl::StartRTPDump(const int video_channel,
+ const char file_nameUTF8[1024],
+ RTPDirections direction) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, file_name: %s, direction: %d)", __FUNCTION__,
+ video_channel, file_nameUTF8, direction);
+ assert(FileWrapper::kMaxFileNameSize == 1024);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->StartRTPDump(file_nameUTF8, direction) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::StopRTPDump(const int video_channel,
+ RTPDirections direction) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d, direction: %d)", __FUNCTION__, video_channel,
+ direction);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->StopRTPDump(direction) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpUnknownError);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::RegisterRTPObserver(const int video_channel,
+ ViERTPObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterRtpObserver(&observer) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::DeregisterRTPObserver(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterRtpObserver(NULL) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::RegisterRTCPObserver(const int video_channel,
+ ViERTCPObserver& observer) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterRtcpObserver(&observer) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+int ViERTP_RTCPImpl::DeregisterRTCPObserver(const int video_channel) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s(channel: %d)", __FUNCTION__, video_channel);
+ ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+ ViEChannel* vie_channel = cs.Channel(video_channel);
+ if (!vie_channel) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo,
+ ViEId(shared_data_->instance_id(), video_channel),
+ "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
+ shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+ return -1;
+ }
+ if (vie_channel->RegisterRtcpObserver(NULL) != 0) {
+ shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
+ return -1;
+ }
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_rtp_rtcp_impl.h b/trunk/src/video_engine/vie_rtp_rtcp_impl.h
new file mode 100644
index 0000000..72e8131
--- /dev/null
+++ b/trunk/src/video_engine/vie_rtp_rtcp_impl.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
+
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "typedefs.h"
+#include "video_engine/include/vie_rtp_rtcp.h"
+#include "video_engine/vie_ref_count.h"
+
+namespace webrtc {
+
+class ViESharedData;
+
+class ViERTP_RTCPImpl
+ : public ViERTP_RTCP,
+ public ViERefCount {
+ public:
+ // Implements ViERTP_RTCP.
+ virtual int Release();
+ virtual int SetLocalSSRC(const int video_channel,
+ const unsigned int SSRC,
+ const StreamType usage,
+ const unsigned char simulcast_idx);
+ virtual int GetLocalSSRC(const int video_channel, unsigned int& SSRC) const;
+ virtual int SetRemoteSSRCType(const int video_channel,
+ const StreamType usage,
+ const unsigned int SSRC) const;
+ virtual int GetRemoteSSRC(const int video_channel, unsigned int& SSRC) const;
+ virtual int GetRemoteCSRCs(const int video_channel,
+ unsigned int CSRCs[kRtpCsrcSize]) const;
+ virtual int SetStartSequenceNumber(const int video_channel,
+ unsigned short sequence_number);
+ virtual int SetRTCPStatus(const int video_channel,
+ const ViERTCPMode rtcp_mode);
+ virtual int GetRTCPStatus(const int video_channel,
+ ViERTCPMode& rtcp_mode) const;
+ virtual int SetRTCPCName(const int video_channel,
+ const char rtcp_cname[KMaxRTCPCNameLength]);
+ virtual int GetRTCPCName(const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const;
+ virtual int GetRemoteRTCPCName(const int video_channel,
+ char rtcp_cname[KMaxRTCPCNameLength]) const;
+ virtual int SendApplicationDefinedRTCPPacket(
+ const int video_channel,
+ const unsigned char sub_type,
+ unsigned int name,
+ const char* data,
+ unsigned short data_length_in_bytes);
+ virtual int SetNACKStatus(const int video_channel, const bool enable);
+ virtual int SetFECStatus(const int video_channel, const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC);
+ virtual int SetHybridNACKFECStatus(const int video_channel, const bool enable,
+ const unsigned char payload_typeRED,
+ const unsigned char payload_typeFEC);
+ virtual int SetKeyFrameRequestMethod(const int video_channel,
+ const ViEKeyFrameRequestMethod method);
+ virtual int SetTMMBRStatus(const int video_channel, const bool enable);
+ virtual int SetRembStatus(int video_channel, bool sender, bool receiver);
+ virtual int GetReceivedRTCPStatistics(const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter,
+ int& rtt_ms) const;
+ virtual int GetSentRTCPStatistics(const int video_channel,
+ unsigned short& fraction_lost,
+ unsigned int& cumulative_lost,
+ unsigned int& extended_max,
+ unsigned int& jitter, int& rtt_ms) const;
+ virtual int GetRTPStatistics(const int video_channel,
+ unsigned int& bytes_sent,
+ unsigned int& packets_sent,
+ unsigned int& bytes_received,
+ unsigned int& packets_received) const;
+ virtual int GetBandwidthUsage(const int video_channel,
+ unsigned int& total_bitrate_sent,
+ unsigned int& video_bitrate_sent,
+ unsigned int& fec_bitrate_sent,
+ unsigned int& nackBitrateSent) const;
+ virtual int GetEstimatedSendBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const;
+ virtual int GetEstimatedReceiveBandwidth(
+ const int video_channel,
+ unsigned int* estimated_bandwidth) const;
+ virtual int SetRTPKeepAliveStatus(
+ const int video_channel,
+ bool enable,
+ const char unknown_payload_type,
+ const unsigned int delta_transmit_time_seconds);
+ virtual int GetRTPKeepAliveStatus(
+ const int video_channel,
+ bool& enabled,
+ char& unkown_payload_type,
+ unsigned int& delta_transmit_time_seconds) const;
+ virtual int StartRTPDump(const int video_channel,
+ const char file_nameUTF8[1024],
+ RTPDirections direction);
+ virtual int StopRTPDump(const int video_channel, RTPDirections direction);
+ virtual int RegisterRTPObserver(const int video_channel,
+ ViERTPObserver& observer);
+ virtual int DeregisterRTPObserver(const int video_channel);
+ virtual int RegisterRTCPObserver(const int video_channel,
+ ViERTCPObserver& observer);
+ virtual int DeregisterRTCPObserver(const int video_channel);
+
+ protected:
+ ViERTP_RTCPImpl(ViESharedData* shared_data);
+ virtual ~ViERTP_RTCPImpl();
+
+ private:
+ ViESharedData* shared_data_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
diff --git a/trunk/src/video_engine/vie_sender.cc b/trunk/src/video_engine/vie_sender.cc
new file mode 100644
index 0000000..e0fe29b
--- /dev/null
+++ b/trunk/src/video_engine/vie_sender.cc
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+
+#include "critical_section_wrapper.h"
+#include "rtp_dump.h"
+#include "vie_sender.h"
+#include "trace.h"
+
+namespace webrtc {
+
+ViESender::ViESender(int engine_id, int channel_id)
+ : engine_id_(engine_id),
+ channel_id_(channel_id),
+ critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ external_encryption_(NULL),
+ encryption_buffer_(NULL),
+ transport_(NULL),
+ rtp_dump_(NULL) {
+}
+
+ViESender::~ViESender() {
+ if (encryption_buffer_) {
+ delete[] encryption_buffer_;
+ encryption_buffer_ = NULL;
+ }
+
+ if (rtp_dump_) {
+ rtp_dump_->Stop();
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ }
+}
+
+int ViESender::RegisterExternalEncryption(Encryption* encryption) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (external_encryption_) {
+ return -1;
+ }
+ encryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu];
+ if (encryption_buffer_ == NULL) {
+ return -1;
+ }
+ external_encryption_ = encryption;
+ return 0;
+}
+
+int ViESender::DeregisterExternalEncryption() {
+ CriticalSectionScoped cs(critsect_.get());
+ if (external_encryption_ == NULL) {
+ return -1;
+ }
+ if (encryption_buffer_) {
+ delete encryption_buffer_;
+ encryption_buffer_ = NULL;
+ }
+ external_encryption_ = NULL;
+ return 0;
+}
+
+int ViESender::RegisterSendTransport(Transport* transport) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (transport_) {
+ return -1;
+ }
+ transport_ = transport;
+ return 0;
+}
+
+int ViESender::DeregisterSendTransport() {
+ CriticalSectionScoped cs(critsect_.get());
+ if (transport_ == NULL) {
+ return -1;
+ }
+ transport_ = NULL;
+ return 0;
+}
+
+int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (rtp_dump_) {
+ // Packet dump is already started, restart it.
+ rtp_dump_->Stop();
+ } else {
+ rtp_dump_ = RtpDump::CreateRtpDump();
+ if (rtp_dump_ == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StartSRTPDump: Failed to create RTP dump");
+ return -1;
+ }
+ }
+ if (rtp_dump_->Start(file_nameUTF8) != 0) {
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StartRTPDump: Failed to start RTP dump");
+ return -1;
+ }
+ return 0;
+}
+
+int ViESender::StopRTPDump() {
+ CriticalSectionScoped cs(critsect_.get());
+ if (rtp_dump_) {
+ if (rtp_dump_->IsActive()) {
+ rtp_dump_->Stop();
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StopRTPDump: Dump not active");
+ }
+ RtpDump::DestroyRtpDump(rtp_dump_);
+ rtp_dump_ = NULL;
+ } else {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "StopRTPDump: RTP dump not started");
+ return -1;
+ }
+ return 0;
+}
+
+int ViESender::SendPacket(int vie_id, const void* data, int len) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (!transport_) {
+ // No transport
+ return -1;
+ }
+
+ assert(ChannelId(vie_id) == channel_id_);
+
+ // TODO(mflodman) Change decrypt to get rid of this cast.
+ void* tmp_ptr = const_cast<void*>(data);
+ unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
+ int send_packet_length = len;
+
+ if (rtp_dump_) {
+ rtp_dump_->DumpPacket(send_packet, send_packet_length);
+ }
+
+ if (external_encryption_) {
+ external_encryption_->encrypt(channel_id_, send_packet,
+ encryption_buffer_, send_packet_length,
+ static_cast<int*>(&send_packet_length));
+ send_packet = encryption_buffer_;
+ }
+
+ const int bytes_sent = transport_->SendPacket(channel_id_, send_packet,
+ send_packet_length);
+ if (bytes_sent != send_packet_length) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViESender::SendPacket - Transport failed to send RTP packet");
+ }
+ return bytes_sent;
+}
+
+int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
+ CriticalSectionScoped cs(critsect_.get());
+
+ if (!transport_) {
+ return -1;
+ }
+
+ assert(ChannelId(vie_id) == channel_id_);
+
+ // Prepare for possible encryption and sending.
+ // TODO(mflodman) Change decrypt to get rid of this cast.
+ void* tmp_ptr = const_cast<void*>(data);
+ unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
+ int send_packet_length = len;
+
+ if (rtp_dump_) {
+ rtp_dump_->DumpPacket(send_packet, send_packet_length);
+ }
+
+ if (external_encryption_) {
+ external_encryption_->encrypt_rtcp(
+ channel_id_, send_packet, encryption_buffer_, send_packet_length,
+ static_cast<int*>(&send_packet_length));
+ send_packet = encryption_buffer_;
+ }
+
+ const int bytes_sent = transport_->SendRTCPPacket(channel_id_, send_packet,
+ send_packet_length);
+ if (bytes_sent != send_packet_length) {
+ WEBRTC_TRACE(
+ webrtc::kTraceWarning, webrtc::kTraceVideo,
+ ViEId(engine_id_, channel_id_),
+ "ViESender::SendRTCPPacket - Transport failed to send RTCP packet");
+ }
+ return bytes_sent;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_sender.h b/trunk/src/video_engine/vie_sender.h
new file mode 100644
index 0000000..a60446d
--- /dev/null
+++ b/trunk/src/video_engine/vie_sender.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// ViESender is responsible for encrypting, if enabled, packets and send to
+// network.
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "typedefs.h"
+#include "vie_defines.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RtpDump;
+class Transport;
+class VideoCodingModule;
+
+class ViESender: public Transport {
+ public:
+ ViESender(int engine_id, int channel_id);
+ ~ViESender();
+
+ // Registers an encryption class to use before sending packets.
+ int RegisterExternalEncryption(Encryption* encryption);
+ int DeregisterExternalEncryption();
+
+ // Registers transport to use for sending RTP and RTCP.
+ int RegisterSendTransport(Transport* transport);
+ int DeregisterSendTransport();
+
+ // Stores all incoming packets to file.
+ int StartRTPDump(const char file_nameUTF8[1024]);
+ int StopRTPDump();
+
+ // Implements Transport.
+ virtual int SendPacket(int vie_id, const void* data, int len);
+ virtual int SendRTCPPacket(int vie_id, const void* data, int len);
+
+ private:
+ int engine_id_;
+ int channel_id_;
+
+ scoped_ptr<CriticalSectionWrapper> critsect_;
+
+ Encryption* external_encryption_;
+ WebRtc_UWord8* encryption_buffer_;
+ Transport* transport_;
+ RtpDump* rtp_dump_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
diff --git a/trunk/src/video_engine/vie_shared_data.cc b/trunk/src/video_engine/vie_shared_data.cc
new file mode 100644
index 0000000..915bcfd
--- /dev/null
+++ b/trunk/src/video_engine/vie_shared_data.cc
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_info.h"
+#include "process_thread.h"
+#include "trace.h"
+#include "vie_channel_manager.h"
+#include "vie_defines.h"
+#include "vie_input_manager.h"
+#include "vie_render_manager.h"
+#include "vie_shared_data.h"
+
+namespace webrtc {
+
+// Active instance counter
+int ViESharedData::instance_counter_ = 0;
+
+ViESharedData::ViESharedData()
+ : instance_id_(++instance_counter_),
+ initialized_(false),
+ number_cores_(CpuInfo::DetectNumberOfCores()),
+ vie_performance_monitor_(ViEPerformanceMonitor(instance_id_)),
+ channel_manager_(*new ViEChannelManager(instance_id_, number_cores_,
+ vie_performance_monitor_)),
+ input_manager_(*new ViEInputManager(instance_id_)),
+ render_manager_(*new ViERenderManager(instance_id_)),
+ module_process_thread_(ProcessThread::CreateProcessThread()),
+ last_error_(0) {
+ Trace::CreateTrace();
+ channel_manager_.SetModuleProcessThread(*module_process_thread_);
+ input_manager_.SetModuleProcessThread(*module_process_thread_);
+ module_process_thread_->Start();
+}
+
+ViESharedData::~ViESharedData() {
+ delete &input_manager_;
+ delete &channel_manager_;
+ delete &render_manager_;
+
+ module_process_thread_->Stop();
+ ProcessThread::DestroyProcessThread(module_process_thread_);
+ Trace::ReturnTrace();
+}
+
+bool ViESharedData::Initialized() const {
+ return initialized_;
+}
+
+int ViESharedData::SetInitialized() {
+ initialized_ = true;
+ return 0;
+}
+
+int ViESharedData::SetUnInitialized() {
+ initialized_ = false;
+ return 0;
+}
+
+void ViESharedData::SetLastError(const int error) const {
+ last_error_ = error;
+}
+
+int ViESharedData::LastErrorInternal() const {
+ int error = last_error_;
+ last_error_ = 0;
+ return error;
+}
+
+int ViESharedData::NumberOfCores() const {
+ return number_cores_;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_shared_data.h b/trunk/src/video_engine/vie_shared_data.h
new file mode 100644
index 0000000..7f755fa
--- /dev/null
+++ b/trunk/src/video_engine/vie_shared_data.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// ViESharedData contains data and instances common to all interface
+// implementations.
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
+
+#include "video_engine/vie_defines.h"
+#include "video_engine/vie_performance_monitor.h"
+
+namespace webrtc {
+
+class ProcessThread;
+class ViEChannelManager;
+class ViEInputManager;
+class ViERenderManager;
+
+class ViESharedData {
+ public:
+ ViESharedData();
+ ~ViESharedData();
+
+ bool Initialized() const;
+ int SetInitialized();
+ int SetUnInitialized();
+ void SetLastError(const int error) const;
+ int LastErrorInternal() const;
+
+ int NumberOfCores() const;
+
+ int instance_id() { return instance_id_;}
+ ViEPerformanceMonitor* vie_performance_monitor() {
+ return &vie_performance_monitor_; }
+ ViEChannelManager* channel_manager() { return &channel_manager_; }
+ ViEInputManager* input_manager() { return &input_manager_; }
+ ViERenderManager* render_manager() { return &render_manager_; }
+
+ private:
+ static int instance_counter_;
+ const int instance_id_;
+ bool initialized_;
+ const int number_cores_;
+
+ ViEPerformanceMonitor vie_performance_monitor_;
+ ViEChannelManager& channel_manager_;
+ ViEInputManager& input_manager_;
+ ViERenderManager& render_manager_;
+ ProcessThread* module_process_thread_;
+ mutable int last_error_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
diff --git a/trunk/src/video_engine/vie_sync_module.cc b/trunk/src/video_engine/vie_sync_module.cc
new file mode 100644
index 0000000..d9b60ac
--- /dev/null
+++ b/trunk/src/video_engine/vie_sync_module.cc
@@ -0,0 +1,297 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vie_sync_module.h"
+
+#include "critical_section_wrapper.h"
+#include "rtp_rtcp.h"
+#include "trace.h"
+#include "video_coding.h"
+#include "voe_video_sync.h"
+
+namespace webrtc {
+
+enum { kSyncInterval = 1000};
+enum { kMaxVideoDiffMs = 80 };
+enum { kMaxAudioDiffMs = 80 };
+enum { kMaxDelay = 1500 };
+
+ViESyncModule::ViESyncModule(int id, VideoCodingModule& vcm,
+ RtpRtcp& rtcp_module)
+ : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+ id_(id),
+ vcm_(vcm),
+ rtcp_module_(rtcp_module),
+ voe_channel_id_(-1),
+ voe_sync_interface_(NULL),
+ last_sync_time_(TickTime::Now()) {
+}
+
+ViESyncModule::~ViESyncModule() {
+}
+
+int ViESyncModule::SetVoiceChannel(int voe_channel_id,
+ VoEVideoSync* voe_sync_interface) {
+ CriticalSectionScoped cs(data_cs_.get());
+ voe_channel_id_ = voe_channel_id;
+ voe_sync_interface_ = voe_sync_interface;
+ rtcp_module_.DeRegisterSyncModule();
+
+ if (!voe_sync_interface) {
+ voe_channel_id_ = -1;
+ if (voe_channel_id >= 0) {
+ // Trying to set a voice channel but no interface exist.
+ return -1;
+ }
+ return 0;
+ }
+ RtpRtcp* voe_rtp_rtcp = NULL;
+ voe_sync_interface->GetRtpRtcp(voe_channel_id_, voe_rtp_rtcp);
+ return rtcp_module_.RegisterSyncModule(voe_rtp_rtcp);
+}
+
+int ViESyncModule::VoiceChannel() {
+ return voe_channel_id_;
+}
+
+void ViESyncModule::SetNetworkDelay(int network_delay) {
+ channel_delay_.network_delay = network_delay;
+}
+
+WebRtc_Word32 ViESyncModule::Version(WebRtc_Word8* version,
+ WebRtc_UWord32& remaining_buffer_in_bytes,
+ WebRtc_UWord32& position) const {
+ if (version == NULL) {
+ WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, -1,
+ "Invalid in argument to ViESyncModule Version()");
+ return -1;
+ }
+ WebRtc_Word8 our_version[] = "ViESyncModule 1.1.0";
+ WebRtc_UWord32 our_length = (WebRtc_UWord32) strlen(our_version);
+ if (remaining_buffer_in_bytes < our_length + 1) {
+ return -1;
+ }
+ memcpy(version, our_version, our_length);
+ version[our_length] = '\0';
+ remaining_buffer_in_bytes -= (our_length + 1);
+ position += (our_length + 1);
+ return 0;
+}
+
+WebRtc_Word32 ViESyncModule::ChangeUniqueId(const WebRtc_Word32 id) {
+ id_ = id;
+ return 0;
+}
+
+WebRtc_Word32 ViESyncModule::TimeUntilNextProcess() {
+ return (WebRtc_Word32)(kSyncInterval -
+ (TickTime::Now() - last_sync_time_).Milliseconds());
+}
+
+WebRtc_Word32 ViESyncModule::Process() {
+ CriticalSectionScoped cs(data_cs_.get());
+ last_sync_time_ = TickTime::Now();
+
+ int total_video_delay_target_ms = vcm_.Delay();
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "Video delay (JB + decoder) is %d ms",
+ total_video_delay_target_ms);
+
+ if (voe_channel_id_ == -1) {
+ return 0;
+ }
+
+ int current_audio_delay_ms = 0;
+ if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
+ current_audio_delay_ms) != 0) {
+ // Could not get VoE delay value, probably not a valid channel Id.
+ WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, id_,
+ "%s: VE_GetDelayEstimate error for voice_channel %d",
+ __FUNCTION__, total_video_delay_target_ms, voe_channel_id_);
+ return 0;
+ }
+
+ int current_diff_ms = 0;
+ // Total video delay.
+ int video_delay_ms = 0;
+ // VoiceEngine report delay estimates even when not started, ignore if the
+ // reported value is lower than 40 ms.
+ if (current_audio_delay_ms < 40) {
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "A/V Sync: Audio delay < 40, skipping.");
+ return 0;
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "Audio delay is: %d for voice channel: %d",
+ current_audio_delay_ms, voe_channel_id_);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "Network delay diff is: %d for voice channel: %d",
+ channel_delay_.network_delay, voe_channel_id_);
+ // Calculate the difference between the lowest possible video delay and
+ // the current audio delay.
+ current_diff_ms = total_video_delay_target_ms - current_audio_delay_ms +
+ channel_delay_.network_delay;
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "Current diff is: %d for audio channel: %d",
+ current_diff_ms, voe_channel_id_);
+
+ if (current_diff_ms > 0) {
+ // The minimum video delay is longer than the current audio delay.
+ // We need to decrease extra video delay, if we have added extra delay
+ // earlier, or add extra audio delay.
+ if (channel_delay_.extra_video_delay_ms > 0) {
+ // We have extra delay added to ViE. Reduce this delay before adding
+ // extra delay to VoE.
+
+ // This is the desired delay, we can't reduce more than this.
+ video_delay_ms = total_video_delay_target_ms;
+
+ // Check that we don't reduce the delay more than what is allowed.
+ if (video_delay_ms <
+ channel_delay_.last_video_delay_ms - kMaxVideoDiffMs) {
+ video_delay_ms =
+ channel_delay_.last_video_delay_ms - kMaxVideoDiffMs;
+ channel_delay_.extra_video_delay_ms =
+ video_delay_ms - total_video_delay_target_ms;
+ } else {
+ channel_delay_.extra_video_delay_ms = 0;
+ }
+ channel_delay_.last_video_delay_ms = video_delay_ms;
+ channel_delay_.last_sync_delay = -1;
+ channel_delay_.extra_audio_delay_ms = 0;
+ } else { // channel_delay_.extra_video_delay_ms > 0
+ // We have no extra video delay to remove, increase the audio delay.
+ if (channel_delay_.last_sync_delay >= 0) {
+ // We have increased the audio delay earlier, increase it even more.
+ int audio_diff_ms = current_diff_ms / 2;
+ if (audio_diff_ms > kMaxAudioDiffMs) {
+ // We only allow a maximum change of KMaxAudioDiffMS for audio
+ // due to NetEQ maximum changes.
+ audio_diff_ms = kMaxAudioDiffMs;
+ }
+ // Increase the audio delay
+ channel_delay_.extra_audio_delay_ms += audio_diff_ms;
+
+ // Don't set a too high delay.
+ if (channel_delay_.extra_audio_delay_ms > kMaxDelay) {
+ channel_delay_.extra_audio_delay_ms = kMaxDelay;
+ }
+
+ // Don't add any extra video delay.
+ video_delay_ms = total_video_delay_target_ms;
+ channel_delay_.extra_video_delay_ms = 0;
+ channel_delay_.last_video_delay_ms = video_delay_ms;
+ channel_delay_.last_sync_delay = 1;
+ } else { // channel_delay_.last_sync_delay >= 0
+ // First time after a delay change, don't add any extra delay.
+ // This is to not toggle back and forth too much.
+ channel_delay_.extra_audio_delay_ms = 0;
+ // Set minimum video delay
+ video_delay_ms = total_video_delay_target_ms;
+ channel_delay_.extra_video_delay_ms = 0;
+ channel_delay_.last_video_delay_ms = video_delay_ms;
+ channel_delay_.last_sync_delay = 0;
+ }
+ }
+ } else { // if (current_diffMS > 0)
+ // The minimum video delay is lower than the current audio delay.
+ // We need to decrease possible extra audio delay, or
+ // add extra video delay.
+
+ if (channel_delay_.extra_audio_delay_ms > 0) {
+ // We have extra delay in VoiceEngine
+ // Start with decreasing the voice delay
+ int audio_diff_ms = current_diff_ms / 2;
+ if (audio_diff_ms < -1 * kMaxAudioDiffMs) {
+ // Don't change the delay too much at once.
+ audio_diff_ms = -1 * kMaxAudioDiffMs;
+ }
+ // Add the negative difference.
+ channel_delay_.extra_audio_delay_ms += audio_diff_ms;
+
+ if (channel_delay_.extra_audio_delay_ms < 0) {
+ // Negative values not allowed.
+ channel_delay_.extra_audio_delay_ms = 0;
+ channel_delay_.last_sync_delay = 0;
+ } else {
+ // There is more audio delay to use for the next round.
+ channel_delay_.last_sync_delay = 1;
+ }
+
+ // Keep the video delay at the minimum values.
+ video_delay_ms = total_video_delay_target_ms;
+ channel_delay_.extra_video_delay_ms = 0;
+ channel_delay_.last_video_delay_ms = video_delay_ms;
+ } else { // channel_delay_.extra_audio_delay_ms > 0
+ // We have no extra delay in VoiceEngine, increase the video delay.
+ channel_delay_.extra_audio_delay_ms = 0;
+
+ // Make the difference positive.
+ int video_diff_ms = -1 * current_diff_ms;
+
+ // This is the desired delay.
+ video_delay_ms = total_video_delay_target_ms + video_diff_ms;
+ if (video_delay_ms > channel_delay_.last_video_delay_ms) {
+ if (video_delay_ms >
+ channel_delay_.last_video_delay_ms + kMaxVideoDiffMs) {
+ // Don't increase the delay too much at once
+ video_delay_ms =
+ channel_delay_.last_video_delay_ms + kMaxVideoDiffMs;
+ }
+ // Verify we don't go above the maximum allowed delay
+ if (video_delay_ms > kMaxDelay) {
+ video_delay_ms = kMaxDelay;
+ }
+ } else {
+ if (video_delay_ms <
+ channel_delay_.last_video_delay_ms - kMaxVideoDiffMs) {
+ // Don't decrease the delay too much at once
+ video_delay_ms =
+ channel_delay_.last_video_delay_ms - kMaxVideoDiffMs;
+ }
+ // Verify we don't go below the minimum delay
+ if (video_delay_ms < total_video_delay_target_ms) {
+ video_delay_ms = total_video_delay_target_ms;
+ }
+ }
+ // Store the values
+ channel_delay_.extra_video_delay_ms =
+ video_delay_ms - total_video_delay_target_ms;
+ channel_delay_.last_video_delay_ms = video_delay_ms;
+ channel_delay_.last_sync_delay = -1;
+ }
+ }
+
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "Sync video delay %d ms for video channel and audio delay %d for audio "
+ "channel %d",
+ video_delay_ms, channel_delay_.extra_audio_delay_ms, voe_channel_id_);
+
+ // Set the extra audio delay.synchronization
+ if (voe_sync_interface_->SetMinimumPlayoutDelay(
+ voe_channel_id_, channel_delay_.extra_audio_delay_ms) == -1) {
+ WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, id_,
+ "Error setting voice delay");
+ }
+
+ if (video_delay_ms < 0) {
+ video_delay_ms = 0;
+ }
+ total_video_delay_target_ms =
+ (total_video_delay_target_ms > video_delay_ms) ?
+ total_video_delay_target_ms : video_delay_ms;
+ vcm_.SetMinimumPlayoutDelay(total_video_delay_target_ms);
+ WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
+ "New Video delay target is: %d", total_video_delay_target_ms);
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_sync_module.h b/trunk/src/video_engine/vie_sync_module.h
new file mode 100644
index 0000000..01373a9
--- /dev/null
+++ b/trunk/src/video_engine/vie_sync_module.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// ViESyncModule is responsible for synchronization audio and video for a given
+// VoE and ViE channel couple.
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
+
+#include "module.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "tick_util.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RtpRtcp;
+class VideoCodingModule;
+class VoEVideoSync;
+
+class ViESyncModule : public Module {
+ public:
+ ViESyncModule(int id, VideoCodingModule& vcm, RtpRtcp& rtcp_module);
+ ~ViESyncModule();
+
+ int SetVoiceChannel(int voe_channel_id, VoEVideoSync* voe_sync_interface);
+ int VoiceChannel();
+
+ // Set how long time, in ms, voice is ahead of video when received on the
+ // network. Positive value means audio is ahead of video.
+ void SetNetworkDelay(int network_delay);
+
+ // Implements Module.
+ virtual WebRtc_Word32 Version(WebRtc_Word8* version,
+ WebRtc_UWord32& remaining_buffer_in_bytes,
+ WebRtc_UWord32& position) const;
+ virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+ virtual WebRtc_Word32 TimeUntilNextProcess();
+ virtual WebRtc_Word32 Process();
+
+ private:
+ scoped_ptr<CriticalSectionWrapper> data_cs_;
+ int id_;
+ VideoCodingModule& vcm_;
+ RtpRtcp& rtcp_module_;
+ int voe_channel_id_;
+ VoEVideoSync* voe_sync_interface_;
+ TickTime last_sync_time_;
+
+ struct ViESyncDelay {
+ ViESyncDelay() {
+ extra_video_delay_ms = 0;
+ last_video_delay_ms = 0;
+ extra_audio_delay_ms = 0;
+ last_sync_delay = 0;
+ network_delay = 120;
+ }
+ int extra_video_delay_ms;
+ int last_video_delay_ms;
+ int extra_audio_delay_ms;
+ int last_sync_delay;
+ int network_delay;
+ };
+ ViESyncDelay channel_delay_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
diff --git a/trunk/src/voice_engine/OWNERS b/trunk/src/voice_engine/OWNERS
new file mode 100644
index 0000000..a07ced3
--- /dev/null
+++ b/trunk/src/voice_engine/OWNERS
@@ -0,0 +1,4 @@
+henrikg@webrtc.org
+henrika@webrtc.org
+niklas.enbom@webrtc.org
+xians@webrtc.org
diff --git a/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h b/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h
new file mode 100644
index 0000000..9eb9271
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MOCK_VOE_CONNECTION_OBSERVER_H_
+#define MOCK_VOE_CONNECTION_OBSERVER_H_
+
+#include "voe_network.h"
+
+namespace webrtc {
+
+class MockVoeConnectionObserver : public VoEConnectionObserver {
+ public:
+ MOCK_METHOD2(OnPeriodicDeadOrAlive, void(const int channel,
+ const bool alive));
+};
+
+}
+
+#endif // MOCK_VOE_CONNECTION_OBSERVER_H_
diff --git a/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h b/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h
new file mode 100644
index 0000000..70d1c13
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
+#define WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
+
+#include "voe_base.h"
+#include "gmock/gmock.h"
+
+namespace webrtc {
+
+class MockVoEObserver: public VoiceEngineObserver {
+ public:
+ MockVoEObserver() {}
+ virtual ~MockVoEObserver() {}
+
+ MOCK_METHOD2(CallbackOnError, void(const int channel, const int error_code));
+};
+
+}
+
+#endif // WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
diff --git a/trunk/src/voice_engine/main/interface/voe_audio_processing.h b/trunk/src/voice_engine/main/interface/voe_audio_processing.h
new file mode 100644
index 0000000..541807b
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_audio_processing.h
@@ -0,0 +1,189 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Noise Suppression (NS).
+// - Automatic Gain Control (AGC).
+// - Echo Control (EC).
+// - Receiving side VAD, NS and AGC.
+// - Measurements of instantaneous speech, noise and echo levels.
+// - Generation of AP debug recordings.
+// - Detection of keyboard typing which can disrupt a voice conversation.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface();
+// VoEAudioProcessing* ap = VoEAudioProcessing::GetInterface(voe);
+// base->Init();
+// ap->SetEcStatus(true, kAgcAdaptiveAnalog);
+// ...
+// base->Terminate();
+// base->Release();
+// ap->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
+#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// VoERxVadCallback
+class WEBRTC_DLLEXPORT VoERxVadCallback
+{
+public:
+ virtual void OnRxVad(int channel, int vadDecision) = 0;
+
+protected:
+ virtual ~VoERxVadCallback() {}
+};
+
+// VoEAudioProcessing
+class WEBRTC_DLLEXPORT VoEAudioProcessing
+{
+public:
+ // Factory for the VoEAudioProcessing sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEAudioProcessing sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Sets Noise Suppression (NS) status and mode.
+ // The NS reduces noise in the microphone signal.
+ virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;
+
+ // Gets the NS status and mode.
+ virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;
+
+ // Sets the Automatic Gain Control (AGC) status and mode.
+ // The AGC adjusts the microphone signal to an appropriate level.
+ virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;
+
+ // Gets the AGC status and mode.
+ virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;
+
+ // Sets the AGC configuration.
+ // Should only be used in situations where the working environment
+ // is well known.
+ virtual int SetAgcConfig(const AgcConfig config) = 0;
+
+ // Gets the AGC configuration.
+ virtual int GetAgcConfig(AgcConfig& config) = 0;
+
+ // Sets the Echo Control (EC) status and mode.
+ // The EC mitigates acoustic echo where a user can hear their own
+ // speech repeated back due to an acoustic coupling between the
+ // speaker and the microphone at the remote end.
+ virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;
+
+ // Gets the EC status and mode.
+ virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;
+
+ // Modifies settings for the AEC designed for mobile devices (AECM).
+ virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
+ bool enableCNG = true) = 0;
+
+ // Gets settings for the AECM.
+ virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;
+
+ // Sets status and mode of the receiving-side (Rx) NS.
+ // The Rx NS reduces noise in the received signal for the specified
+ // |channel|. Intended for advanced usage only.
+ virtual int SetRxNsStatus(int channel,
+ bool enable,
+ NsModes mode = kNsUnchanged) = 0;
+
+ // Gets status and mode of the receiving-side NS.
+ virtual int GetRxNsStatus(int channel,
+ bool& enabled,
+ NsModes& mode) = 0;
+
+ // Sets status and mode of the receiving-side (Rx) AGC.
+ // The Rx AGC adjusts the received signal to an appropriate level
+ // for the specified |channel|. Intended for advanced usage only.
+ virtual int SetRxAgcStatus(int channel,
+ bool enable,
+ AgcModes mode = kAgcUnchanged) = 0;
+
+ // Gets status and mode of the receiving-side AGC.
+ virtual int GetRxAgcStatus(int channel,
+ bool& enabled,
+ AgcModes& mode) = 0;
+
+ // Modifies the AGC configuration on the receiving side for the
+ // specified |channel|.
+ virtual int SetRxAgcConfig(int channel, const AgcConfig config) = 0;
+
+ // Gets the AGC configuration on the receiving side.
+ virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
+
+ // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
+ // notifications for the specified |channel|.
+ virtual int RegisterRxVadObserver(int channel,
+ VoERxVadCallback &observer) = 0;
+
+ // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
+ // notifications for the specified |channel|.
+ virtual int DeRegisterRxVadObserver(int channel) = 0;
+
+ // Gets the VAD/DTX activity for the specified |channel|.
+ // The returned value is 1 if frames of audio contains speech
+ // and 0 if silence. The output is always 1 if VAD is disabled.
+ virtual int VoiceActivityIndicator(int channel) = 0;
+
+ // Enables or disables the possibility to retrieve echo metrics and delay
+ // logging values during an active call. The metrics are only supported in
+ // AEC.
+ virtual int SetEcMetricsStatus(bool enable) = 0;
+
+ // Gets the current EC metric status.
+ virtual int GetEcMetricsStatus(bool& enabled) = 0;
+
+ // Gets the instantaneous echo level metrics.
+ virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
+
+ // Gets the EC internal |delay_median| and |delay_std| in ms between
+ // near-end and far-end. The values are calculated over the time period
+ // since the last GetEcDelayMetrics() call.
+ virtual int GetEcDelayMetrics(int& delay_median, int& delay_std) = 0;
+
+ // Enables recording of Audio Processing (AP) debugging information.
+ // The file can later be used for off-line analysis of the AP performance.
+ virtual int StartDebugRecording(const char* fileNameUTF8) = 0;
+
+ // Disables recording of AP debugging information.
+ virtual int StopDebugRecording() = 0;
+
+ // Enables or disables detection of disturbing keyboard typing.
+ // An error notification will be given as a callback upon detection.
+ virtual int SetTypingDetectionStatus(bool enable) = 0;
+
+ // Gets the current typing detection status.
+ virtual int GetTypingDetectionStatus(bool& enabled) = 0;
+
+protected:
+ VoEAudioProcessing() {}
+ virtual ~VoEAudioProcessing() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
diff --git a/trunk/src/voice_engine/main/interface/voe_base.h b/trunk/src/voice_engine/main/interface/voe_base.h
new file mode 100644
index 0000000..be6849c
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_base.h
@@ -0,0 +1,213 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law).
+// - Initialization and termination.
+// - Trace information on text files or via callbacks.
+// - Multi-channel support (mixing, sending to multiple destinations etc.).
+// - Call setup (port and address) for receiving and sending sides.
+//
+// To support other codecs than G.711, the VoECodec sub-API must be utilized.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// base->StartPlayout(ch);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_H
+#define WEBRTC_VOICE_ENGINE_VOE_BASE_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class AudioDeviceModule;
+
+const int kVoEDefault = -1;
+
+// VoiceEngineObserver
+class WEBRTC_DLLEXPORT VoiceEngineObserver
+{
+public:
+ // This method will be called after the occurrence of any runtime error
+ // code, or warning notification, when the observer interface has been
+ // installed using VoEBase::RegisterVoiceEngineObserver().
+ virtual void CallbackOnError(const int channel, const int errCode) = 0;
+
+protected:
+ virtual ~VoiceEngineObserver() {}
+};
+
+// VoiceEngine
+class WEBRTC_DLLEXPORT VoiceEngine
+{
+public:
+ // Creates a VoiceEngine object, which can then be used to acquire
+ // sub-APIs. Returns NULL on failure.
+ static VoiceEngine* Create();
+
+ // Deletes a created VoiceEngine object and releases the utilized resources.
+ // If |ignoreRefCounters| is set to false, all reference counters must be
+ // zero to enable a valid release of the allocated resources. When set to
+ // true, a release of all resources allocated by the VoE is performed
+ // without checking the reference counter state.
+ static bool Delete(VoiceEngine*& voiceEngine,
+ bool ignoreRefCounters = false);
+
+ // Specifies the amount and type of trace information which will be
+ // created by the VoiceEngine.
+ static int SetTraceFilter(const unsigned int filter);
+
+ // Sets the name of the trace file and enables non-encrypted trace messages.
+ static int SetTraceFile(const char* fileNameUTF8,
+ const bool addFileCounter = false);
+
+ // Installs the TraceCallback implementation to ensure that the user
+ // receives callbacks for generated trace messages.
+ static int SetTraceCallback(TraceCallback* callback);
+
+ static int SetAndroidObjects(void* javaVM, void* env, void* context);
+
+protected:
+ VoiceEngine() {}
+ virtual ~VoiceEngine() {}
+};
+
+// VoEBase
+class WEBRTC_DLLEXPORT VoEBase
+{
+public:
+ // Factory for the VoEBase sub-API. Increases an internal reference
+ // counter if successful. Returns NULL if the API is not supported or if
+ // construction fails.
+ static VoEBase* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEBase sub-API and decreases an internal reference
+ // counter. Returns the new reference count. This value should be zero
+ // for all sub-API:s before the VoiceEngine object can be safely deleted.
+ virtual int Release() = 0;
+
+ // Installs the observer class to enable runtime error control and
+ // warning notifications.
+ virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;
+
+ // Removes and disables the observer class for runtime error control
+ // and warning notifications.
+ virtual int DeRegisterVoiceEngineObserver() = 0;
+
+ // Initiates all common parts of the VoiceEngine; e.g. all
+ // encoders/decoders, the sound card and core receiving components.
+ // This method also makes it possible to install a user-defined
+ // external Audio Device Module (ADM) which implements all the audio
+ // layer functionality in a separate (reference counted) module.
+ virtual int Init(AudioDeviceModule* external_adm = NULL) = 0;
+
+ // Terminates all VoiceEngine functions and releses allocated resources.
+ virtual int Terminate() = 0;
+
+ // Retrieves the maximum number of channels that can be created.
+ virtual int MaxNumOfChannels() = 0;
+
+ // Creates a new channel and allocates the required resources for it.
+ virtual int CreateChannel() = 0;
+
+ // Deletes an existing channel and releases the utilized resources.
+ virtual int DeleteChannel(int channel) = 0;
+
+ // Sets the local receiver port and address for a specified
+ // |channel| number.
+ virtual int SetLocalReceiver(int channel, int port,
+ int RTCPport = kVoEDefault,
+ const char ipAddr[64] = NULL,
+ const char multiCastAddr[64] = NULL) = 0;
+
+ // Gets the local receiver port and address for a specified
+ // |channel| number.
+ virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
+ char ipAddr[64]) = 0;
+
+ // Sets the destination port and address for a specified |channel| number.
+ virtual int SetSendDestination(int channel, int port,
+ const char ipAddr[64],
+ int sourcePort = kVoEDefault,
+ int RTCPport = kVoEDefault) = 0;
+
+ // Gets the destination port and address for a specified |channel| number.
+ virtual int GetSendDestination(int channel, int& port, char ipAddr[64],
+ int& sourcePort, int& RTCPport) = 0;
+
+ // Prepares and initiates the VoiceEngine for reception of
+ // incoming RTP/RTCP packets on the specified |channel|.
+ virtual int StartReceive(int channel) = 0;
+
+ // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
+ virtual int StopReceive(int channel) = 0;
+
+ // Starts forwarding the packets to the mixer/soundcard for a
+ // specified |channel|.
+ virtual int StartPlayout(int channel) = 0;
+
+ // Stops forwarding the packets to the mixer/soundcard for a
+ // specified |channel|.
+ virtual int StopPlayout(int channel) = 0;
+
+ // Starts sending packets to an already specified IP address and
+ // port number for a specified |channel|.
+ virtual int StartSend(int channel) = 0;
+
+ // Stops sending packets from a specified |channel|.
+ virtual int StopSend(int channel) = 0;
+
+ // Gets the version information for VoiceEngine and its components.
+ virtual int GetVersion(char version[1024]) = 0;
+
+ // Gets the last VoiceEngine error code.
+ virtual int LastError() = 0;
+
+
+ // Stops or resumes playout and transmission on a temporary basis.
+ virtual int SetOnHoldStatus(int channel, bool enable,
+ OnHoldModes mode = kHoldSendAndPlay) = 0;
+
+ // Gets the current playout and transmission status.
+ virtual int GetOnHoldStatus(int channel, bool& enabled,
+ OnHoldModes& mode) = 0;
+
+ // Sets the NetEQ playout mode for a specified |channel| number.
+ virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode) = 0;
+
+ // Gets the NetEQ playout mode for a specified |channel| number.
+ virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode) = 0;
+
+ // Sets the NetEQ background noise mode for a specified |channel| number.
+ virtual int SetNetEQBGNMode(int channel, NetEqBgnModes mode) = 0;
+
+ // Gets the NetEQ background noise mode for a specified |channel| number.
+ virtual int GetNetEQBGNMode(int channel, NetEqBgnModes& mode) = 0;
+
+protected:
+ VoEBase() {}
+ virtual ~VoEBase() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_call_report.h b/trunk/src/voice_engine/main/interface/voe_call_report.h
new file mode 100644
index 0000000..c4d3abd
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_call_report.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Long-term speech and noise level metrics.
+// - Long-term echo metric statistics.
+// - Round Trip Time (RTT) statistics.
+// - Dead-or-Alive connection summary.
+// - Generation of call reports to text files.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoECallReport report = VoECallReport::GetInterface(voe);
+// base->Init();
+// LevelStatistics stats;
+// report->GetSpeechAndNoiseSummary(stats);
+// ...
+// base->Terminate();
+// base->Release();
+// report->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
+#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// VoECallReport
+class WEBRTC_DLLEXPORT VoECallReport
+{
+public:
+ // Factory for the VoECallReport sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoECallReport* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoECallReport sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Performs a combined reset of all components involved in generating
+ // the call report for a specified |channel|. Pass in -1 to reset
+ // all channels.
+ virtual int ResetCallReportStatistics(int channel) = 0;
+
+ // Gets minimum, maximum and average levels for long-term echo metrics.
+ virtual int GetEchoMetricSummary(EchoStatistics& stats) = 0;
+
+ // Gets minimum, maximum and average levels for Round Trip Time (RTT)
+ // measurements.
+ virtual int GetRoundTripTimeSummary(int channel,
+ StatVal& delaysMs) = 0;
+
+ // Gets the total amount of dead and alive connection detections
+ // during a VoIP session.
+ virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections,
+ int& numOfAliveDetections) = 0;
+
+ // Creates a text file in ASCII format, which contains a summary
+ // of all the statistics that can be obtained by the call report sub-API.
+ virtual int WriteReportToFile(const char* fileNameUTF8) = 0;
+
+protected:
+ VoECallReport() { }
+ virtual ~VoECallReport() { }
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
diff --git a/trunk/src/voice_engine/main/interface/voe_codec.h b/trunk/src/voice_engine/main/interface/voe_codec.h
new file mode 100644
index 0000000..413bc90
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_codec.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Support of non-default codecs (e.g. iLBC, iSAC, etc.).
+// - Voice Activity Detection (VAD) on a per channel basis.
+// - Possibility to specify how to map received payload types to codecs.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoECodec* codec = VoECodec::GetInterface(voe);
+// base->Init();
+// int num_of_codecs = codec->NumOfCodecs()
+// ...
+// base->Terminate();
+// base->Release();
+// codec->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_H
+#define WEBRTC_VOICE_ENGINE_VOE_CODEC_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoECodec
+{
+public:
+ // Factory for the VoECodec sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoECodec* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoECodec sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Gets the number of supported codecs.
+ virtual int NumOfCodecs() = 0;
+
+ // Get the |codec| information for a specified list |index|.
+ virtual int GetCodec(int index, CodecInst& codec) = 0;
+
+ // Sets the |codec| for the |channel| to be used for sending.
+ virtual int SetSendCodec(int channel, const CodecInst& codec) = 0;
+
+ // Gets the |codec| parameters for the sending codec on a specified
+ // |channel|.
+ virtual int GetSendCodec(int channel, CodecInst& codec) = 0;
+
+ // Gets the currently received |codec| for a specific |channel|.
+ virtual int GetRecCodec(int channel, CodecInst& codec) = 0;
+
+ // Sets the initial values of target rate and frame size for iSAC
+ // for a specified |channel|. This API is only valid if iSAC is setup
+ // to run in channel-adaptive mode
+ virtual int SetISACInitTargetRate(int channel, int rateBps,
+ bool useFixedFrameSize = false) = 0;
+
+ // Sets the maximum allowed iSAC rate which the codec may not exceed
+ // for a single packet for the specified |channel|. The maximum rate is
+ // defined as payload size per frame size in bits per second.
+ virtual int SetISACMaxRate(int channel, int rateBps) = 0;
+
+ // Sets the maximum allowed iSAC payload size for a specified |channel|.
+ // The maximum value is set independently of the frame size, i.e.
+ // 30 ms and 60 ms packets have the same limit.
+ virtual int SetISACMaxPayloadSize(int channel, int sizeBytes) = 0;
+
+ // Sets the dynamic payload type number for a particular |codec| or
+ // disables (ignores) a codec for receiving. For instance, when receiving
+ // an invite from a SIP-based client, this function can be used to change
+ // the dynamic payload type number to match that in the INVITE SDP-
+ // message. The utilized parameters in the |codec| structure are:
+ // plname, plfreq, pltype and channels.
+ virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0;
+
+ // Gets the actual payload type that is set for receiving a |codec| on a
+ // |channel|. The value it retrieves will either be the default payload
+ // type, or a value earlier set with SetRecPayloadType().
+ virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0;
+
+ // Sets the payload |type| for the sending of SID-frames with background
+ // noise estimation during silence periods detected by the VAD.
+ virtual int SetSendCNPayloadType(
+ int channel, int type, PayloadFrequencies frequency = kFreq16000Hz) = 0;
+
+
+ // Sets the VAD/DTX (silence suppression) status and |mode| for a
+ // specified |channel|.
+ virtual int SetVADStatus(int channel, bool enable,
+ VadModes mode = kVadConventional,
+ bool disableDTX = false) = 0;
+
+ // Gets the VAD/DTX status and |mode| for a specified |channel|.
+ virtual int GetVADStatus(int channel, bool& enabled, VadModes& mode,
+ bool& disabledDTX) = 0;
+
+ // Not supported
+ virtual int SetAMREncFormat(int channel, AmrMode mode) = 0;
+
+ // Not supported
+ virtual int SetAMRDecFormat(int channel, AmrMode mode) = 0;
+
+ // Not supported
+ virtual int SetAMRWbEncFormat(int channel, AmrMode mode) = 0;
+
+ // Not supported
+ virtual int SetAMRWbDecFormat(int channel, AmrMode mode) = 0;
+
+protected:
+ VoECodec() {}
+ virtual ~VoECodec() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H
diff --git a/trunk/src/voice_engine/main/interface/voe_dtmf.h b/trunk/src/voice_engine/main/interface/voe_dtmf.h
new file mode 100644
index 0000000..3ed1749
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_dtmf.h
@@ -0,0 +1,148 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Telephone event transmission.
+// - DTMF tone generation.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEDtmf* dtmf = VoEDtmf::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// ...
+// dtmf->SendTelephoneEvent(ch, 7);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// dtmf->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_H
+#define WEBRTC_VOICE_ENGINE_VOE_DTMF_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// VoETelephoneEventObserver
+class WEBRTC_DLLEXPORT VoETelephoneEventObserver
+{
+public:
+ // This method will be called after the detection of an inband
+ // telephone event. The event code is given as output in the
+ // |eventCode| parameter.
+ virtual void OnReceivedTelephoneEventInband(int channel,
+ int eventCode,
+ bool endOfEvent) = 0;
+
+ // This method will be called after the detection of an out-of-band
+ // telephone event. The event code is given as output in the
+ // |eventCode| parameter.
+ virtual void OnReceivedTelephoneEventOutOfBand(
+ int channel,
+ int eventCode,
+ bool endOfEvent) = 0;
+
+protected:
+ virtual ~VoETelephoneEventObserver() {}
+};
+
+// VoEDtmf
+class WEBRTC_DLLEXPORT VoEDtmf
+{
+public:
+
+ // Factory for the VoEDtmf sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEDtmf* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEDtmf sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Sends telephone events either in-band or out-of-band.
+ virtual int SendTelephoneEvent(int channel, int eventCode,
+ bool outOfBand = true, int lengthMs = 160,
+ int attenuationDb = 10) = 0;
+
+
+ // Sets the dynamic payload |type| that should be used for telephone
+ // events.
+ virtual int SetSendTelephoneEventPayloadType(int channel,
+ unsigned char type) = 0;
+
+
+ // Gets the currently set dynamic payload |type| for telephone events.
+ virtual int GetSendTelephoneEventPayloadType(int channel,
+ unsigned char& type) = 0;
+
+ // Enables or disables local tone playout for received DTMF events
+ // out-of-band.
+ virtual int SetDtmfPlayoutStatus(int channel, bool enable) = 0;
+
+ // Gets the DTMF playout status.
+ virtual int GetDtmfPlayoutStatus(int channel, bool& enabled) = 0;
+
+ // Toogles DTMF feedback state: when a DTMF tone is sent, the same tone
+ // is played out on the speaker.
+ virtual int SetDtmfFeedbackStatus(bool enable,
+ bool directFeedback = false) = 0;
+
+ // Gets the DTMF feedback status.
+ virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) = 0;
+
+ // Plays a DTMF feedback tone (only locally).
+ virtual int PlayDtmfTone(int eventCode, int lengthMs = 200,
+ int attenuationDb = 10) = 0;
+
+ // Starts playing out a DTMF feedback tone locally.
+ // The tone will be played out until the corresponding stop function
+ // is called.
+ virtual int StartPlayingDtmfTone(int eventCode,
+ int attenuationDb = 10) = 0;
+
+ // Stops playing out a DTMF feedback tone locally.
+ virtual int StopPlayingDtmfTone() = 0;
+
+ // Installs an instance of a VoETelephoneEventObserver derived class and
+ // activates detection of telephone events for the specified |channel|.
+ virtual int RegisterTelephoneEventDetection(
+ int channel, TelephoneEventDetectionMethods detectionMethod,
+ VoETelephoneEventObserver& observer) = 0;
+
+ // Removes an instance of a VoETelephoneEventObserver derived class and
+ // disables detection of telephone events for the specified |channel|.
+ virtual int DeRegisterTelephoneEventDetection(int channel) = 0;
+
+ // Gets the current telephone-event detection status for a specified
+ // |channel|.
+ virtual int GetTelephoneEventDetectionStatus(
+ int channel, bool& enabled,
+ TelephoneEventDetectionMethods& detectionMethod) = 0;
+
+protected:
+ VoEDtmf() {}
+ virtual ~VoEDtmf() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_DTMF_H
diff --git a/trunk/src/voice_engine/main/interface/voe_encryption.h b/trunk/src/voice_engine/main/interface/voe_encryption.h
new file mode 100644
index 0000000..ae3f373
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_encryption.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - External encryption and decryption.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEEncryption* encrypt = VoEEncryption::GetInterface(voe);
+// ...
+// encrypt->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
+#define WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEEncryption
+{
+public:
+ // Factory for the VoEEncryption sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEEncryption* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEEncryption sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Installs an Encryption instance and enables external encryption
+ // for the selected |channel|.
+ virtual int RegisterExternalEncryption(
+ int channel, Encryption& encryption) = 0;
+
+ // Removes an Encryption instance and disables external encryption
+ // for the selected |channel|.
+ virtual int DeRegisterExternalEncryption(int channel) = 0;
+
+ // Not supported
+ virtual int EnableSRTPSend(int channel, CipherTypes cipherType,
+ int cipherKeyLength, AuthenticationTypes authType, int authKeyLength,
+ int authTagLength, SecurityLevels level, const unsigned char key[30],
+ bool useForRTCP = false) = 0;
+
+ // Not supported
+ virtual int DisableSRTPSend(int channel) = 0;
+
+ // Not supported
+ virtual int EnableSRTPReceive(int channel, CipherTypes cipherType,
+ int cipherKeyLength, AuthenticationTypes authType, int authKeyLength,
+ int authTagLength, SecurityLevels level, const unsigned char key[30],
+ bool useForRTCP = false) = 0;
+
+ // Not supported
+ virtual int DisableSRTPReceive(int channel) = 0;
+
+protected:
+ VoEEncryption() {}
+ virtual ~VoEEncryption() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
diff --git a/trunk/src/voice_engine/main/interface/voe_errors.h b/trunk/src/voice_engine/main/interface/voe_errors.h
new file mode 100644
index 0000000..cc05970
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_errors.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
+#define WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
+
+// Warnings
+#define VE_PORT_NOT_DEFINED 8001
+#define VE_CHANNEL_NOT_VALID 8002
+#define VE_FUNC_NOT_SUPPORTED 8003
+#define VE_INVALID_LISTNR 8004
+#define VE_INVALID_ARGUMENT 8005
+#define VE_INVALID_PORT_NMBR 8006
+#define VE_INVALID_PLNAME 8007
+#define VE_INVALID_PLFREQ 8008
+#define VE_INVALID_PLTYPE 8009
+#define VE_INVALID_PACSIZE 8010
+#define VE_NOT_SUPPORTED 8011
+#define VE_ALREADY_LISTENING 8012
+#define VE_CHANNEL_NOT_CREATED 8013
+#define VE_MAX_ACTIVE_CHANNELS_REACHED 8014
+#define VE_REC_CANNOT_PREPARE_HEADER 8015
+#define VE_REC_CANNOT_ADD_BUFFER 8016
+#define VE_PLAY_CANNOT_PREPARE_HEADER 8017
+#define VE_ALREADY_SENDING 8018
+#define VE_INVALID_IP_ADDRESS 8019
+#define VE_ALREADY_PLAYING 8020
+#define VE_NOT_ALL_VERSION_INFO 8021
+#define VE_DTMF_OUTOF_RANGE 8022
+#define VE_INVALID_CHANNELS 8023
+#define VE_SET_PLTYPE_FAILED 8024
+#define VE_ENCRYPT_NOT_INITED 8025
+#define VE_NOT_INITED 8026
+#define VE_NOT_SENDING 8027
+#define VE_EXT_TRANSPORT_NOT_SUPPORTED 8028
+#define VE_EXTERNAL_TRANSPORT_ENABLED 8029
+#define VE_STOP_RECORDING_FAILED 8030
+#define VE_INVALID_RATE 8031
+#define VE_INVALID_PACKET 8032
+#define VE_NO_GQOS 8033
+#define VE_INVALID_TIMESTAMP 8034
+#define VE_RECEIVE_PACKET_TIMEOUT 8035
+#define VE_STILL_PLAYING_PREV_DTMF 8036
+#define VE_INIT_FAILED_WRONG_EXPIRY 8037
+#define VE_SENDING 8038
+#define VE_ENABLE_IPV6_FAILED 8039
+#define VE_FUNC_NO_STEREO 8040
+// Range 8041-8080 is not used
+#define VE_FW_TRAVERSAL_ALREADY_INITIALIZED 8081
+#define VE_PACKET_RECEIPT_RESTARTED 8082
+#define VE_NOT_ALL_INFO 8083
+#define VE_CANNOT_SET_SEND_CODEC 8084
+#define VE_CODEC_ERROR 8085
+#define VE_NETEQ_ERROR 8086
+#define VE_RTCP_ERROR 8087
+#define VE_INVALID_OPERATION 8088
+#define VE_CPU_INFO_ERROR 8089
+#define VE_SOUNDCARD_ERROR 8090
+#define VE_SPEECH_LEVEL_ERROR 8091
+#define VE_SEND_ERROR 8092
+#define VE_CANNOT_REMOVE_CONF_CHANNEL 8093
+#define VE_PLTYPE_ERROR 8094
+#define VE_SET_FEC_FAILED 8095
+#define VE_CANNOT_GET_PLAY_DATA 8096
+#define VE_APM_ERROR 8097
+#define VE_RUNTIME_PLAY_WARNING 8098
+#define VE_RUNTIME_REC_WARNING 8099
+#define VE_NOT_PLAYING 8100
+#define VE_SOCKETS_NOT_INITED 8101
+#define VE_CANNOT_GET_SOCKET_INFO 8102
+#define VE_INVALID_MULTICAST_ADDRESS 8103
+#define VE_DESTINATION_NOT_INITED 8104
+#define VE_RECEIVE_SOCKETS_CONFLICT 8105
+#define VE_SEND_SOCKETS_CONFLICT 8106
+#define VE_TYPING_NOISE_WARNING 8107
+#define VE_SATURATION_WARNING 8108
+#define VE_NOISE_WARNING 8109
+#define VE_CANNOT_GET_SEND_CODEC 8110
+#define VE_CANNOT_GET_REC_CODEC 8111
+#define VE_ALREADY_INITED 8112
+
+// Errors causing limited functionality
+#define VE_RTCP_SOCKET_ERROR 9001
+#define VE_MIC_VOL_ERROR 9002
+#define VE_SPEAKER_VOL_ERROR 9003
+#define VE_CANNOT_ACCESS_MIC_VOL 9004
+#define VE_CANNOT_ACCESS_SPEAKER_VOL 9005
+#define VE_GET_MIC_VOL_ERROR 9006
+#define VE_GET_SPEAKER_VOL_ERROR 9007
+#define VE_THREAD_RTCP_ERROR 9008
+#define VE_CANNOT_INIT_APM 9009
+#define VE_SEND_SOCKET_TOS_ERROR 9010
+#define VE_CANNOT_RETRIEVE_DEVICE_NAME 9013
+#define VE_SRTP_ERROR 9014
+// 9015 is not used
+#define VE_INTERFACE_NOT_FOUND 9016
+#define VE_TOS_GQOS_CONFLICT 9017
+#define VE_CANNOT_ADD_CONF_CHANNEL 9018
+#define VE_BUFFER_TOO_SMALL 9019
+#define VE_CANNOT_EXECUTE_SETTING 9020
+#define VE_CANNOT_RETRIEVE_SETTING 9021
+// 9022 is not used
+#define VE_RTP_KEEPALIVE_FAILED 9023
+#define VE_SEND_DTMF_FAILED 9024
+#define VE_CANNOT_RETRIEVE_CNAME 9025
+#define VE_DECRYPTION_FAILED 9026
+#define VE_ENCRYPTION_FAILED 9027
+#define VE_CANNOT_RETRIEVE_RTP_STAT 9028
+#define VE_GQOS_ERROR 9029
+#define VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED 9030
+#define VE_TOS_INVALID 9031
+#define VE_TOS_ERROR 9032
+#define VE_CANNOT_RETRIEVE_VALUE 9033
+
+// Critical errors that stops voice functionality
+#define VE_PLAY_UNDEFINED_SC_ERR 10001
+#define VE_REC_CANNOT_OPEN_SC 10002
+#define VE_SOCKET_ERROR 10003
+#define VE_MMSYSERR_INVALHANDLE 10004
+#define VE_MMSYSERR_NODRIVER 10005
+#define VE_MMSYSERR_NOMEM 10006
+#define VE_WAVERR_UNPREPARED 10007
+#define VE_WAVERR_STILLPLAYING 10008
+#define VE_UNDEFINED_SC_ERR 10009
+#define VE_UNDEFINED_SC_REC_ERR 10010
+#define VE_THREAD_ERROR 10011
+#define VE_CANNOT_START_RECORDING 10012
+#define VE_PLAY_CANNOT_OPEN_SC 10013
+#define VE_NO_WINSOCK_2 10014
+#define VE_SEND_SOCKET_ERROR 10015
+#define VE_BAD_FILE 10016
+#define VE_EXPIRED_COPY 10017
+#define VE_NOT_AUTHORISED 10018
+#define VE_RUNTIME_PLAY_ERROR 10019
+#define VE_RUNTIME_REC_ERROR 10020
+#define VE_BAD_ARGUMENT 10021
+#define VE_LINUX_API_ONLY 10022
+#define VE_REC_DEVICE_REMOVED 10023
+#define VE_NO_MEMORY 10024
+#define VE_BAD_HANDLE 10025
+#define VE_RTP_RTCP_MODULE_ERROR 10026
+#define VE_AUDIO_CODING_MODULE_ERROR 10027
+#define VE_AUDIO_DEVICE_MODULE_ERROR 10028
+#define VE_CANNOT_START_PLAYOUT 10029
+#define VE_CANNOT_STOP_RECORDING 10030
+#define VE_CANNOT_STOP_PLAYOUT 10031
+#define VE_CANNOT_INIT_CHANNEL 10032
+#define VE_RECV_SOCKET_ERROR 10033
+#define VE_SOCKET_TRANSPORT_MODULE_ERROR 10034
+#define VE_AUDIO_CONF_MIX_MODULE_ERROR 10035
+
+// Warnings for other platforms (reserved range 8061-8080)
+#define VE_IGNORED_FUNCTION 8061
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
diff --git a/trunk/src/voice_engine/main/interface/voe_external_media.h b/trunk/src/voice_engine/main/interface/voe_external_media.h
new file mode 100644
index 0000000..50d2d38
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_external_media.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// In some cases it is desirable to use an audio source or sink which may
+// not be available to the VoiceEngine, such as a DV camera. This sub-API
+// contains functions that allow for the use of such external recording
+// sources and playout sinks. It also describes how recorded data, or data
+// to be played out, can be modified outside the VoiceEngine.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEMediaProcess media = VoEMediaProcess::GetInterface(voe);
+// base->Init();
+// ...
+// media->SetExternalRecordingStatus(true);
+// ...
+// base->Terminate();
+// base->Release();
+// media->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
+#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEMediaProcess
+{
+public:
+ // The VoiceEngine user should override the Process() method in a
+ // derived class. Process() will be called when audio is ready to
+ // be processed. The audio can be accessed in several different modes
+ // given by the |type| parameter. The function should modify the
+ // original data and ensure that it is copied back to the |audio10ms|
+ // array. The number of samples in the frame cannot be changed.
+ // The sampling frequency will depend upon the codec used.
+ // If |isStereo| is true, audio10ms will contain 16-bit PCM data
+ // samples in interleaved stereo format (L0,R0,L1,R1,
):
+ virtual void Process(const int channel, const ProcessingTypes type,
+ WebRtc_Word16 audio10ms[], const int length,
+ const int samplingFreq, const bool isStereo) = 0;
+
+protected:
+ virtual ~VoEMediaProcess() {}
+};
+
+class WEBRTC_DLLEXPORT VoEExternalMedia
+{
+public:
+ // Factory for the VoEExternalMedia sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEExternalMedia sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Installs a VoEMediaProcess derived instance and activates external
+ // media for the specified |channel| and |type|.
+ virtual int RegisterExternalMediaProcessing(
+ int channel, ProcessingTypes type, VoEMediaProcess& processObject) = 0;
+
+ // Removes the VoEMediaProcess derived instance and deactivates external
+ // media for the specified |channel| and |type|.
+ virtual int DeRegisterExternalMediaProcessing(
+ int channel, ProcessingTypes type) = 0;
+
+ // Toogles state of external recording.
+ virtual int SetExternalRecordingStatus(bool enable) = 0;
+
+ // Toogles state of external playout.
+ virtual int SetExternalPlayoutStatus(bool enable) = 0;
+
+ // This function accepts externally recorded audio. During transmission,
+ // this method should be called at as regular an interval as possible
+ // with frames of corresponding size.
+ virtual int ExternalRecordingInsertData(
+ const WebRtc_Word16 speechData10ms[], int lengthSamples,
+ int samplingFreqHz, int current_delay_ms) = 0;
+
+ // This function gets audio for an external playout sink.
+ // During transmission, this function should be called every ~10 ms
+ // to obtain a new 10 ms frame of audio. The length of the block will
+ // be 160, 320, 440 or 480 samples (for 16, 32, 44 or 48 kHz sampling
+ // rates respectively).
+ virtual int ExternalPlayoutGetData(
+ WebRtc_Word16 speechData10ms[], int samplingFreqHz,
+ int current_delay_ms, int& lengthSamples) = 0;
+
+protected:
+ VoEExternalMedia() {}
+ virtual ~VoEExternalMedia() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
diff --git a/trunk/src/voice_engine/main/interface/voe_file.h b/trunk/src/voice_engine/main/interface/voe_file.h
new file mode 100644
index 0000000..d968dcf
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_file.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - File playback.
+// - File recording.
+// - File conversion.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEFile* file = VoEFile::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// ...
+// base->StartPlayout(ch);
+// file->StartPlayingFileAsMicrophone(ch, "data_file_16kHz.pcm", true);
+// ...
+// file->StopPlayingFileAsMicrophone(ch);
+// base->StopPlayout(ch);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// file->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_FILE_H
+#define WEBRTC_VOICE_ENGINE_VOE_FILE_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEFile
+{
+public:
+ // Factory for the VoEFile sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEFile* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEFile sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Starts playing and mixing files with the local speaker signal for
+ // playout.
+ virtual int StartPlayingFileLocally(
+ int channel,
+ const char fileNameUTF8[1024],
+ bool loop = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0,
+ int startPointMs = 0,
+ int stopPointMs = 0) = 0;
+
+ // Starts playing and mixing streams with the local speaker signal for
+ // playout.
+ virtual int StartPlayingFileLocally(
+ int channel,
+ InStream* stream,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0,
+ int startPointMs = 0, int stopPointMs = 0) = 0;
+
+ // Stops playback of a file on a specific |channel|.
+ virtual int StopPlayingFileLocally(int channel) = 0;
+
+ // Returns the current file playing state for a specific |channel|.
+ virtual int IsPlayingFileLocally(int channel) = 0;
+
+ // Sets the volume scaling for a speaker file that is already playing.
+ virtual int ScaleLocalFilePlayout(int channel, float scale) = 0;
+
+ // Starts reading data from a file and transmits the data either
+ // mixed with or instead of the microphone signal.
+ virtual int StartPlayingFileAsMicrophone(
+ int channel,
+ const char fileNameUTF8[1024],
+ bool loop = false ,
+ bool mixWithMicrophone = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0) = 0;
+
+ // Starts reading data from a stream and transmits the data either
+ // mixed with or instead of the microphone signal.
+ virtual int StartPlayingFileAsMicrophone(
+ int channel,
+ InStream* stream,
+ bool mixWithMicrophone = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0) = 0;
+
+ // Stops playing of a file as microphone signal for a specific |channel|.
+ virtual int StopPlayingFileAsMicrophone(int channel) = 0;
+
+ // Returns whether the |channel| is currently playing a file as microphone.
+ virtual int IsPlayingFileAsMicrophone(int channel) = 0;
+
+ // Sets the volume scaling for a microphone file that is already playing.
+ virtual int ScaleFileAsMicrophonePlayout(int channel, float scale) = 0;
+
+ // Starts recording the mixed playout audio.
+ virtual int StartRecordingPlayout(int channel,
+ const char* fileNameUTF8,
+ CodecInst* compression = NULL,
+ int maxSizeBytes = -1) = 0;
+
+ // Stops recording the mixed playout audio.
+ virtual int StopRecordingPlayout(int channel) = 0;
+
+ virtual int StartRecordingPlayout(int channel,
+ OutStream* stream,
+ CodecInst* compression = NULL) = 0;
+
+ // Starts recording the microphone signal to a file.
+ virtual int StartRecordingMicrophone(const char* fileNameUTF8,
+ CodecInst* compression = NULL,
+ int maxSizeBytes = -1) = 0;
+
+ // Starts recording the microphone signal to a stream.
+ virtual int StartRecordingMicrophone(OutStream* stream,
+ CodecInst* compression = NULL) = 0;
+
+ // Stops recording the microphone signal.
+ virtual int StopRecordingMicrophone() = 0;
+
+
+ // Gets the duration of a file.
+ virtual int GetFileDuration(const char* fileNameUTF8, int& durationMs,
+ FileFormats format = kFileFormatPcm16kHzFile) = 0;
+
+ // Gets the current played position of a file on a specific |channel|.
+ virtual int GetPlaybackPosition(int channel, int& positionMs) = 0;
+
+ virtual int ConvertPCMToWAV(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8) = 0;
+
+ virtual int ConvertPCMToWAV(InStream* streamIn,
+ OutStream* streamOut) = 0;
+
+ virtual int ConvertWAVToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8) = 0;
+
+ virtual int ConvertWAVToPCM(InStream* streamIn,
+ OutStream* streamOut) = 0;
+
+ virtual int ConvertPCMToCompressed(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8,
+ CodecInst* compression) = 0;
+
+ virtual int ConvertPCMToCompressed(InStream* streamIn,
+ OutStream* streamOut,
+ CodecInst* compression) = 0;
+
+ virtual int ConvertCompressedToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8) = 0;
+
+ virtual int ConvertCompressedToPCM(InStream* streamIn,
+ OutStream* streamOut) = 0;
+
+protected:
+ VoEFile() {}
+ virtual ~VoEFile() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_FILE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_hardware.h b/trunk/src/voice_engine/main/interface/voe_hardware.h
new file mode 100644
index 0000000..24ed1ff
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_hardware.h
@@ -0,0 +1,142 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Audio device handling.
+// - Device information.
+// - CPU load monitoring.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEHardware* hardware = VoEHardware::GetInterface(voe);
+// base->Init();
+// ...
+// int n_devices = hardware->GetNumOfPlayoutDevices();
+// ...
+// base->Terminate();
+// base->Release();
+// hardware->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
+#define WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEHardware
+{
+public:
+ // Factory for the VoEHardware sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEHardware* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEHardware sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Gets the number of audio devices available for recording.
+ virtual int GetNumOfRecordingDevices(int& devices) = 0;
+
+ // Gets the number of audio devices available for playout.
+ virtual int GetNumOfPlayoutDevices(int& devices) = 0;
+
+ // Gets the name of a specific recording device given by an |index|.
+ // On Windows Vista/7, it also retrieves an additional unique ID
+ // (GUID) for the recording device.
+ virtual int GetRecordingDeviceName(int index, char strNameUTF8[128],
+ char strGuidUTF8[128]) = 0;
+
+ // Gets the name of a specific playout device given by an |index|.
+ // On Windows Vista/7, it also retrieves an additional unique ID
+ // (GUID) for the playout device.
+ virtual int GetPlayoutDeviceName(int index, char strNameUTF8[128],
+ char strGuidUTF8[128]) = 0;
+
+ // Checks if the sound card is available to be opened for recording.
+ virtual int GetRecordingDeviceStatus(bool& isAvailable) = 0;
+
+ // Checks if the sound card is available to be opened for playout.
+ virtual int GetPlayoutDeviceStatus(bool& isAvailable) = 0;
+
+ // Sets the audio device used for recording.
+ virtual int SetRecordingDevice(
+ int index, StereoChannel recordingChannel = kStereoBoth) = 0;
+
+ // Sets the audio device used for playout.
+ virtual int SetPlayoutDevice(int index) = 0;
+
+ // Sets the type of audio device layer to use.
+ virtual int SetAudioDeviceLayer(AudioLayers audioLayer) = 0;
+
+ // Gets the currently used (active) audio device layer.
+ virtual int GetAudioDeviceLayer(AudioLayers& audioLayer) = 0;
+
+ // Gets the VoiceEngine's current CPU consumption in terms of the percent
+ // of total CPU availability. [Windows only]
+ virtual int GetCPULoad(int& loadPercent) = 0;
+
+ // Gets the computer's current CPU consumption in terms of the percent
+ // of the total CPU availability. This method may fail a few times on
+ // Windows because it needs a certain warm-up time before reporting the
+ // result. You should check the return value and either try again or
+ // give up when it fails.
+ virtual int GetSystemCPULoad(int& loadPercent) = 0;
+
+ // Not supported
+ virtual int ResetAudioDevice() = 0;
+
+ // Not supported
+ virtual int AudioDeviceControl(
+ unsigned int par1, unsigned int par2, unsigned int par3) = 0;
+
+ // Not supported
+ virtual int SetLoudspeakerStatus(bool enable) = 0;
+
+ // Not supported
+ virtual int GetLoudspeakerStatus(bool& enabled) = 0;
+
+ // *Experimental - not recommended for use.*
+ // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
+ //
+ // Currently incompatible with the standard VoE AEC and AGC; don't attempt
+ // to enable them while this is active.
+ //
+ // Must be called before VoEBase::StartSend(). When enabled:
+ // 1. VoEBase::StartPlayout() must be called before VoEBase::StartSend().
+ // 2. VoEBase::StopSend() should be called before VoEBase::StopPlayout().
+ // The reverse order may cause garbage audio to be rendered or the
+ // capture side to halt until StopSend() is called.
+ //
+ // As a consequence, SetPlayoutDevice() should be used with caution
+ // during a call. It will function, but may cause the above issues for
+ // the duration it takes to complete. (In practice, it should complete
+ // fast enough to avoid audible degradation).
+ virtual int EnableBuiltInAEC(bool enable) = 0;
+ virtual bool BuiltInAECIsEnabled() const = 0;
+
+protected:
+ VoEHardware() {}
+ virtual ~VoEHardware() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_neteq_stats.h b/trunk/src/voice_engine/main/interface/voe_neteq_stats.h
new file mode 100644
index 0000000..4940bed
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_neteq_stats.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
+#define WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoENetEqStats
+{
+public:
+ // Factory for the VoENetEqStats sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoENetEqStats* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoENetEqStats sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Get the "in-call" statistics from NetEQ.
+ // The statistics are reset after the query.
+ virtual int GetNetworkStatistics(int channel, NetworkStatistics& stats) = 0;
+
+protected:
+ VoENetEqStats() {}
+ virtual ~VoENetEqStats() {}
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
diff --git a/trunk/src/voice_engine/main/interface/voe_network.h b/trunk/src/voice_engine/main/interface/voe_network.h
new file mode 100644
index 0000000..10acf1c
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_network.h
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - External protocol support.
+// - Extended port and address APIs.
+// - Port and address filters.
+// - Windows GQoS functions.
+// - Packet timeout notification.
+// - Dead-or-Alive connection observations.
+// - Transmission of raw RTP/RTCP packets into existing channels.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoENetwork* netw = VoENetwork::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// ...
+// netw->SetPeriodicDeadOrAliveStatus(ch, true);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// netw->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
+#define WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// VoEConnectionObserver
+class WEBRTC_DLLEXPORT VoEConnectionObserver
+{
+public:
+ // This method will be called peridically and deliver dead-or-alive
+ // notifications for a specified |channel| when the observer interface
+ // has been installed and activated.
+ virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive) = 0;
+
+protected:
+ virtual ~VoEConnectionObserver() {}
+};
+
+// VoENetwork
+class WEBRTC_DLLEXPORT VoENetwork
+{
+public:
+ // Factory for the VoENetwork sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoENetwork* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoENetwork sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Installs and enables a user-defined external transport protocol for a
+ // specified |channel|.
+ virtual int RegisterExternalTransport(
+ int channel, Transport& transport) = 0;
+
+ // Removes and disables a user-defined external transport protocol for a
+ // specified |channel|.
+ virtual int DeRegisterExternalTransport(int channel) = 0;
+
+ // The packets received from the network should be passed to this
+ // function when external transport is enabled. Note that the data
+ // including the RTP-header must also be given to the VoiceEngine.
+ virtual int ReceivedRTPPacket(
+ int channel, const void* data, unsigned int length) = 0;
+
+ // The packets received from the network should be passed to this
+ // function when external transport is enabled. Note that the data
+ // including the RTCP-header must also be given to the VoiceEngine.
+ virtual int ReceivedRTCPPacket(
+ int channel, const void* data, unsigned int length) = 0;
+
+ // Gets the source ports and IP address of incoming packets on a
+ // specific |channel|.
+ virtual int GetSourceInfo(
+ int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
+
+ // Gets the local (host) IP address.
+ virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false) = 0;
+
+ // Enables IPv6 for a specified |channel|.
+ virtual int EnableIPv6(int channel) = 0;
+
+ // Gets the current IPv6 staus for a specified |channel|.
+ virtual bool IPv6IsEnabled(int channel) = 0;
+
+ // Enables a port and IP address filter for incoming packets on a
+ // specific |channel|.
+ virtual int SetSourceFilter(int channel,
+ int rtpPort, int rtcpPort = 0, const char ipAddr[64] = 0) = 0;
+
+ // Gets the current port and IP-address filter for a specified |channel|.
+ virtual int GetSourceFilter(
+ int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
+
+ // Sets the six-bit Differentiated Services Code Point (DSCP) in the
+ // IP header of the outgoing stream for a specific |channel|.
+ virtual int SetSendTOS(int channel,
+ int DSCP, int priority = -1, bool useSetSockopt = false) = 0;
+
+ // Gets the six-bit DSCP in the IP header of the outgoing stream for
+ // a specific channel.
+ virtual int GetSendTOS(
+ int channel, int& DSCP, int& priority, bool& useSetSockopt) = 0;
+
+ // Sets the Generic Quality of Service (GQoS) service level.
+ // The Windows operating system then maps to a Differentiated Services
+ // Code Point (DSCP) and to an 802.1p setting. [Windows only]
+ virtual int SetSendGQoS(
+ int channel, bool enable, int serviceType, int overrideDSCP = 0) = 0;
+
+ // Gets the Generic Quality of Service (GQoS) service level.
+ virtual int GetSendGQoS(
+ int channel, bool& enabled, int& serviceType, int& overrideDSCP) = 0;
+
+ // Enables or disables warnings that report if packets have not been
+ // received in |timeoutSeconds| seconds for a specific |channel|.
+ virtual int SetPacketTimeoutNotification(
+ int channel, bool enable, int timeoutSeconds = 2) = 0;
+
+ // Gets the current time-out notification status.
+ virtual int GetPacketTimeoutNotification(
+ int channel, bool& enabled, int& timeoutSeconds) = 0;
+
+ // Installs the observer class implementation for a specified |channel|.
+ virtual int RegisterDeadOrAliveObserver(
+ int channel, VoEConnectionObserver& observer) = 0;
+
+ // Removes the observer class implementation for a specified |channel|.
+ virtual int DeRegisterDeadOrAliveObserver(int channel) = 0;
+
+ // Enables or disables the periodic dead-or-alive callback functionality
+ // for a specified |channel|.
+ virtual int SetPeriodicDeadOrAliveStatus(
+ int channel, bool enable, int sampleTimeSeconds = 2) = 0;
+
+ // Gets the current dead-or-alive notification status.
+ virtual int GetPeriodicDeadOrAliveStatus(
+ int channel, bool& enabled, int& sampleTimeSeconds) = 0;
+
+ // Handles sending a raw UDP data packet over an existing RTP or RTCP
+ // socket.
+ virtual int SendUDPPacket(
+ int channel, const void* data, unsigned int length,
+ int& transmittedBytes, bool useRtcpSocket = false) = 0;
+
+protected:
+ VoENetwork() {}
+ virtual ~VoENetwork() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
diff --git a/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h b/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h
new file mode 100644
index 0000000..e26d85f
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h
@@ -0,0 +1,234 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
+// - SSRC handling.
+// - Transmission of RTCP sender reports.
+// - Obtaining RTCP data from incoming RTCP sender reports.
+// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
+// - Forward Error Correction (FEC).
+// - RTP Keepalive for maintaining the NAT mappings associated to RTP flows.
+// - Writing RTP and RTCP packets to binary files for off-line analysis of
+// the call quality.
+// - Inserting extra RTP packets into active audio stream.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// ...
+// rtp_rtcp->SetLocalSSRC(ch, 12345);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// rtp_rtcp->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
+#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+// VoERTPObserver
+class WEBRTC_DLLEXPORT VoERTPObserver
+{
+public:
+ virtual void OnIncomingCSRCChanged(
+ const int channel, const unsigned int CSRC, const bool added) = 0;
+
+ virtual void OnIncomingSSRCChanged(
+ const int channel, const unsigned int SSRC) = 0;
+
+protected:
+ virtual ~VoERTPObserver() {}
+};
+
+// VoERTCPObserver
+class WEBRTC_DLLEXPORT VoERTCPObserver
+{
+public:
+ virtual void OnApplicationDataReceived(
+ const int channel, const unsigned char subType,
+ const unsigned int name, const unsigned char* data,
+ const unsigned short dataLengthInBytes) = 0;
+
+protected:
+ virtual ~VoERTCPObserver() {}
+};
+
+// CallStatistics
+struct CallStatistics
+{
+ unsigned short fractionLost;
+ unsigned int cumulativeLost;
+ unsigned int extendedMax;
+ unsigned int jitterSamples;
+ int rttMs;
+ int bytesSent;
+ int packetsSent;
+ int bytesReceived;
+ int packetsReceived;
+};
+
+// VoERTP_RTCP
+class WEBRTC_DLLEXPORT VoERTP_RTCP
+{
+public:
+
+ // Factory for the VoERTP_RTCP sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoERTP_RTCP sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Registers an instance of a VoERTPObserver derived class for a specified
+ // |channel|. It will allow the user to observe callbacks related to the
+ // RTP protocol such as changes in the incoming SSRC.
+ virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer) = 0;
+
+ // Deregisters an instance of a VoERTPObserver derived class for a
+ // specified |channel|.
+ virtual int DeRegisterRTPObserver(int channel) = 0;
+
+ // Registers an instance of a VoERTCPObserver derived class for a specified
+ // |channel|.
+ virtual int RegisterRTCPObserver(
+ int channel, VoERTCPObserver& observer) = 0;
+
+ // Deregisters an instance of a VoERTCPObserver derived class for a
+ // specified |channel|.
+ virtual int DeRegisterRTCPObserver(int channel) = 0;
+
+ // Sets the local RTP synchronization source identifier (SSRC) explicitly.
+ virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
+
+ // Gets the local RTP SSRC of a specified |channel|.
+ virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
+
+ // Gets the SSRC of the incoming RTP packets.
+ virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
+
+ // Sets the status of rtp-audio-level-indication on a specific |channel|.
+ virtual int SetRTPAudioLevelIndicationStatus(
+ int channel, bool enable, unsigned char ID = 1) = 0;
+
+ // Sets the status of rtp-audio-level-indication on a specific |channel|.
+ virtual int GetRTPAudioLevelIndicationStatus(
+ int channel, bool& enabled, unsigned char& ID) = 0;
+
+ // Gets the CSRCs of the incoming RTP packets.
+ virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]) = 0;
+
+ // Sets the RTCP status on a specific |channel|.
+ virtual int SetRTCPStatus(int channel, bool enable) = 0;
+
+ // Gets the RTCP status on a specific |channel|.
+ virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
+
+ // Sets the canonical name (CNAME) parameter for RTCP reports on a
+ // specific |channel|.
+ virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
+
+ // Gets the canonical name (CNAME) parameter for RTCP reports on a
+ // specific |channel|.
+ virtual int GetRTCP_CNAME(int channel, char cName[256]) = 0;
+
+ // Gets the canonical name (CNAME) parameter for incoming RTCP reports
+ // on a specific channel.
+ virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
+
+ // Gets RTCP data from incoming RTCP Sender Reports.
+ virtual int GetRemoteRTCPData(
+ int channel, unsigned int& NTPHigh, unsigned int& NTPLow,
+ unsigned int& timestamp, unsigned int& playoutTimestamp,
+ unsigned int* jitter = NULL, unsigned short* fractionLost = NULL) = 0;
+
+ // Gets RTP statistics for a specific |channel|.
+ virtual int GetRTPStatistics(
+ int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs,
+ unsigned int& discardedPackets) = 0;
+
+ // Gets RTCP statistics for a specific |channel|.
+ virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
+
+ // Sends an RTCP APP packet on a specific |channel|.
+ virtual int SendApplicationDefinedRTCPPacket(
+ int channel, const unsigned char subType, unsigned int name,
+ const char* data, unsigned short dataLengthInBytes) = 0;
+
+ // Sets the Forward Error Correction (FEC) status on a specific |channel|.
+ virtual int SetFECStatus(
+ int channel, bool enable, int redPayloadtype = -1) = 0;
+
+ // Gets the FEC status on a specific |channel|.
+ virtual int GetFECStatus(
+ int channel, bool& enabled, int& redPayloadtype) = 0;
+
+ // Sets the RTP keepalive mechanism status.
+ // This functionality can maintain an existing Network Address Translator
+ // (NAT) mapping while regular RTP is no longer transmitted.
+ virtual int SetRTPKeepaliveStatus(
+ int channel, bool enable, unsigned char unknownPayloadType,
+ int deltaTransmitTimeSeconds = 15) = 0;
+
+ // Gets the RTP keepalive mechanism status.
+ virtual int GetRTPKeepaliveStatus(
+ int channel, bool& enabled, unsigned char& unknownPayloadType,
+ int& deltaTransmitTimeSeconds) = 0;
+
+ // Enables capturing of RTP packets to a binary file on a specific
+ // |channel| and for a given |direction|. The file can later be replayed
+ // using e.g. RTP Tools rtpplay since the binary file format is
+ // compatible with the rtpdump format.
+ virtual int StartRTPDump(
+ int channel, const char fileNameUTF8[1024],
+ RTPDirections direction = kRtpIncoming) = 0;
+
+ // Disables capturing of RTP packets to a binary file on a specific
+ // |channel| and for a given |direction|.
+ virtual int StopRTPDump(
+ int channel, RTPDirections direction = kRtpIncoming) = 0;
+
+ // Gets the the current RTP capturing state for the specified
+ // |channel| and |direction|.
+ virtual int RTPDumpIsActive(
+ int channel, RTPDirections direction = kRtpIncoming) = 0;
+
+ // Sends an extra RTP packet using an existing/active RTP session.
+ // It is possible to set the payload type, marker bit and payload
+ // of the extra RTP
+ virtual int InsertExtraRTPPacket(
+ int channel, unsigned char payloadType, bool markerBit,
+ const char* payloadData, unsigned short payloadSize) = 0;
+
+protected:
+ VoERTP_RTCP() {}
+ virtual ~VoERTP_RTCP() {}
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
diff --git a/trunk/src/voice_engine/main/interface/voe_video_sync.h b/trunk/src/voice_engine/main/interface/voe_video_sync.h
new file mode 100644
index 0000000..ac3b84a
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_video_sync.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - RTP header modification (time stamp and sequence number fields).
+// - Playout delay tuning to synchronize the voice with video.
+// - Playout delay monitoring.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEVideoSync* vsync = VoEVideoSync::GetInterface(voe);
+// base->Init();
+// ...
+// int buffer_ms(0);
+// vsync->GetPlayoutBufferSize(buffer_ms);
+// ...
+// base->Terminate();
+// base->Release();
+// vsync->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
+#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class RtpRtcp;
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEVideoSync
+{
+public:
+ // Factory for the VoEVideoSync sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEVideoSync sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Gets the current sound card buffer size (playout delay).
+ virtual int GetPlayoutBufferSize(int& bufferMs) = 0;
+
+ // Sets an additional delay for the playout jitter buffer.
+ virtual int SetMinimumPlayoutDelay(int channel, int delayMs) = 0;
+
+ // Gets the sum of the algorithmic delay, jitter buffer delay, and the
+ // playout buffer delay for a specified |channel|.
+ virtual int GetDelayEstimate(int channel, int& delayMs) = 0;
+
+ // Manual initialization of the RTP timestamp.
+ virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
+
+ // Manual initialization of the RTP sequence number.
+ virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
+
+ // Get the received RTP timestamp
+ virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
+
+ virtual int GetRtpRtcp (int channel, RtpRtcp* &rtpRtcpModule) = 0;
+
+protected:
+ VoEVideoSync() { }
+ virtual ~VoEVideoSync() { }
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
diff --git a/trunk/src/voice_engine/main/interface/voe_volume_control.h b/trunk/src/voice_engine/main/interface/voe_volume_control.h
new file mode 100644
index 0000000..6d64e96
--- /dev/null
+++ b/trunk/src/voice_engine/main/interface/voe_volume_control.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This sub-API supports the following functionalities:
+//
+// - Speaker volume controls.
+// - Microphone volume control.
+// - Non-linear speech level control.
+// - Mute functions.
+// - Additional stereo scaling methods.
+//
+// Usage example, omitting error checking:
+//
+// using namespace webrtc;
+// VoiceEngine* voe = VoiceEngine::Create();
+// VoEBase* base = VoEBase::GetInterface(voe);
+// VoEVolumeControl* volume = VoEVolumeControl::GetInterface(voe);
+// base->Init();
+// int ch = base->CreateChannel();
+// ...
+// volume->SetInputMute(ch, true);
+// ...
+// base->DeleteChannel(ch);
+// base->Terminate();
+// base->Release();
+// volume->Release();
+// VoiceEngine::Delete(voe);
+//
+#ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
+#define WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
+
+#include "common_types.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class WEBRTC_DLLEXPORT VoEVolumeControl
+{
+public:
+ // Factory for the VoEVolumeControl sub-API. Increases an internal
+ // reference counter if successful. Returns NULL if the API is not
+ // supported or if construction fails.
+ static VoEVolumeControl* GetInterface(VoiceEngine* voiceEngine);
+
+ // Releases the VoEVolumeControl sub-API and decreases an internal
+ // reference counter. Returns the new reference count. This value should
+ // be zero for all sub-API:s before the VoiceEngine object can be safely
+ // deleted.
+ virtual int Release() = 0;
+
+ // Sets the speaker |volume| level. Valid range is [0,255].
+ virtual int SetSpeakerVolume(unsigned int volume) = 0;
+
+ // Gets the speaker |volume| level.
+ virtual int GetSpeakerVolume(unsigned int& volume) = 0;
+
+ // Mutes the speaker device completely in the operating system.
+ virtual int SetSystemOutputMute(bool enable) = 0;
+
+ // Gets the output device mute state in the operating system.
+ virtual int GetSystemOutputMute(bool &enabled) = 0;
+
+ // Sets the microphone volume level. Valid range is [0,255].
+ virtual int SetMicVolume(unsigned int volume) = 0;
+
+ // Gets the microphone volume level.
+ virtual int GetMicVolume(unsigned int& volume) = 0;
+
+ // Mutes the microphone input signal completely without affecting
+ // the audio device volume.
+ virtual int SetInputMute(int channel, bool enable) = 0;
+
+ // Gets the current microphone input mute state.
+ virtual int GetInputMute(int channel, bool& enabled) = 0;
+
+ // Mutes the microphone device completely in the operating system.
+ virtual int SetSystemInputMute(bool enable) = 0;
+
+ // Gets the mute state of the input device in the operating system.
+ virtual int GetSystemInputMute(bool& enabled) = 0;
+
+ // Gets the microphone speech |level|, mapped non-linearly to the range
+ // [0,9].
+ virtual int GetSpeechInputLevel(unsigned int& level) = 0;
+
+ // Gets the speaker speech |level|, mapped non-linearly to the range
+ // [0,9].
+ virtual int GetSpeechOutputLevel(int channel, unsigned int& level) = 0;
+
+ // Gets the microphone speech |level|, mapped linearly to the range
+ // [0,32768].
+ virtual int GetSpeechInputLevelFullRange(unsigned int& level) = 0;
+
+ // Gets the speaker speech |level|, mapped linearly to the range [0,32768].
+ virtual int GetSpeechOutputLevelFullRange(
+ int channel, unsigned int& level) = 0;
+
+ // Sets a volume |scaling| applied to the outgoing signal of a specific
+ // channel. Valid scale range is [0.0, 10.0].
+ virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0;
+
+ // Gets the current volume scaling for a specified |channel|.
+ virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0;
+
+ // Scales volume of the |left| and |right| channels independently.
+ // Valid scale range is [0.0, 1.0].
+ virtual int SetOutputVolumePan(int channel, float left, float right) = 0;
+
+ // Gets the current left and right scaling factors.
+ virtual int GetOutputVolumePan(int channel, float& left, float& right) = 0;
+
+protected:
+ VoEVolumeControl() {};
+ virtual ~VoEVolumeControl() {};
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
diff --git a/trunk/src/voice_engine/main/source/Android.mk b/trunk/src/voice_engine/main/source/Android.mk
new file mode 100644
index 0000000..5040f17
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/Android.mk
@@ -0,0 +1,88 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_voe_core
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ audio_frame_operations.cc \
+ channel.cc \
+ channel_manager.cc \
+ channel_manager_base.cc \
+ dtmf_inband.cc \
+ dtmf_inband_queue.cc \
+ level_indicator.cc \
+ monitor_module.cc \
+ output_mixer.cc \
+ ref_count.cc \
+ shared_data.cc \
+ statistics.cc \
+ transmit_mixer.cc \
+ utility.cc \
+ voe_audio_processing_impl.cc \
+ voe_base_impl.cc \
+ voe_call_report_impl.cc \
+ voe_codec_impl.cc \
+ voe_dtmf_impl.cc \
+ voe_encryption_impl.cc \
+ voe_external_media_impl.cc \
+ voe_file_impl.cc \
+ voe_hardware_impl.cc \
+ voe_neteq_stats_impl.cc \
+ voe_network_impl.cc \
+ voe_rtp_rtcp_impl.cc \
+ voe_video_sync_impl.cc \
+ voe_volume_control_impl.cc \
+ voice_engine_impl.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_ANDROID_OPENSLES'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/resampler/include \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../modules/interface \
+ $(LOCAL_PATH)/../../../modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/../../../modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/../../../modules/audio_device/main/interface \
+ $(LOCAL_PATH)/../../../modules/audio_device/main/source \
+ $(LOCAL_PATH)/../../../modules/audio_processing/include \
+ $(LOCAL_PATH)/../../../modules/media_file/interface \
+ $(LOCAL_PATH)/../../../modules/rtp_rtcp/interface \
+ $(LOCAL_PATH)/../../../modules/udp_transport/interface \
+ $(LOCAL_PATH)/../../../modules/utility/interface \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/voice_engine/main/source/audio_frame_operations.cc b/trunk/src/voice_engine/main/source/audio_frame_operations.cc
new file mode 100644
index 0000000..e08d0a2
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/audio_frame_operations.cc
@@ -0,0 +1,129 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_frame_operations.h"
+#include "module_common_types.h"
+
+namespace webrtc {
+
+namespace voe {
+
+WebRtc_Word32
+AudioFrameOperations::MonoToStereo(AudioFrame& audioFrame)
+{
+ if (audioFrame._audioChannel != 1)
+ {
+ return -1;
+ }
+ if ((audioFrame._payloadDataLengthInSamples << 1) >=
+ AudioFrame::kMaxAudioFrameSizeSamples)
+ {
+ // not enough memory to expand from mono to stereo
+ return -1;
+ }
+
+ WebRtc_Word16* payloadCopy =
+ new WebRtc_Word16[audioFrame._payloadDataLengthInSamples];
+ memcpy(payloadCopy, audioFrame._payloadData,
+ sizeof(WebRtc_Word16)*audioFrame._payloadDataLengthInSamples);
+
+ for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
+ {
+ audioFrame._payloadData[2*i] = payloadCopy[i];
+ audioFrame._payloadData[2*i+1] = payloadCopy[i];
+ }
+
+ audioFrame._audioChannel = 2;
+
+ delete [] payloadCopy;
+ return 0;
+}
+
+WebRtc_Word32
+AudioFrameOperations::StereoToMono(AudioFrame& audioFrame)
+{
+ if (audioFrame._audioChannel != 2)
+ {
+ return -1;
+ }
+
+ for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
+ {
+ audioFrame._payloadData[i] = (audioFrame._payloadData[2*i] >> 1) +
+ (audioFrame._payloadData[2*i+1] >> 1);
+ }
+
+ audioFrame._audioChannel = 1;
+
+ return 0;
+}
+
+WebRtc_Word32
+AudioFrameOperations::Mute(AudioFrame& audioFrame)
+{
+ const int sizeInBytes = sizeof(WebRtc_Word16) *
+ audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel;
+ memset(audioFrame._payloadData, 0, sizeInBytes);
+ audioFrame._energy = 0;
+ return 0;
+}
+
+WebRtc_Word32
+AudioFrameOperations::Scale(const float left,
+ const float right,
+ AudioFrame& audioFrame)
+{
+ if (audioFrame._audioChannel == 1)
+ {
+ assert(false);
+ return -1;
+ }
+
+ for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
+ {
+ audioFrame._payloadData[2*i] =
+ (WebRtc_Word16)(left*audioFrame._payloadData[2*i]);
+ audioFrame._payloadData[2*i+1] =
+ (WebRtc_Word16)(right*audioFrame._payloadData[2*i+1]);
+ }
+ return 0;
+}
+
+WebRtc_Word32
+AudioFrameOperations::ScaleWithSat(const float scale, AudioFrame& audioFrame)
+{
+ WebRtc_Word32 tmp(0);
+
+ // Ensure that the output result is saturated [-32768, +32768].
+ for (int i = 0;
+ i < audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel;
+ i++)
+ {
+ tmp = static_cast<WebRtc_Word32> (scale * audioFrame._payloadData[i]);
+ if (tmp < -32768)
+ {
+ audioFrame._payloadData[i] = -32768;
+ }
+ else if (tmp > 32767)
+ {
+ audioFrame._payloadData[i] = 32767;
+ }
+ else
+ {
+ audioFrame._payloadData[i] = static_cast<WebRtc_Word16> (tmp);
+ }
+ }
+ return 0;
+}
+
+} // namespace voe
+
+} // namespace webrtc
+
diff --git a/trunk/src/voice_engine/main/source/audio_frame_operations.h b/trunk/src/voice_engine/main/source/audio_frame_operations.h
new file mode 100644
index 0000000..368850b
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/audio_frame_operations.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
+#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+namespace voe {
+
+class AudioFrameOperations
+{
+public:
+ static WebRtc_Word32 MonoToStereo(AudioFrame& audioFrame);
+
+ static WebRtc_Word32 StereoToMono(AudioFrame& audioFrame);
+
+ static WebRtc_Word32 Mute(AudioFrame& audioFrame);
+
+ static WebRtc_Word32 Scale(const float left,
+ const float right,
+ AudioFrame& audioFrame);
+
+ static WebRtc_Word32 ScaleWithSat(const float scale,
+ AudioFrame& audioFrame);
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
diff --git a/trunk/src/voice_engine/main/source/channel.cc b/trunk/src/voice_engine/main/source/channel.cc
new file mode 100644
index 0000000..4fc1c8e
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel.cc
@@ -0,0 +1,6684 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "channel.h"
+
+#include "audio_device.h"
+#include "audio_frame_operations.h"
+#include "audio_processing.h"
+#include "critical_section_wrapper.h"
+#include "output_mixer.h"
+#include "process_thread.h"
+#include "rtp_dump.h"
+#include "statistics.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "utility.h"
+#include "voe_base.h"
+#include "voe_external_media.h"
+#include "voe_rtp_rtcp.h"
+
+#if defined(_WIN32)
+#include <Qos.h>
+#endif
+
+namespace webrtc
+{
+
+namespace voe
+{
+
+WebRtc_Word32
+Channel::SendData(FrameType frameType,
+ WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
+ " payloadSize=%u, fragmentation=0x%x)",
+ frameType, payloadType, timeStamp, payloadSize, fragmentation);
+
+ if (_includeAudioLevelIndication)
+ {
+ assert(_rtpAudioProc.get() != NULL);
+ // Store current audio level in the RTP/RTCP module.
+ // The level will be used in combination with voice-activity state
+ // (frameType) to add an RTP header extension
+ _rtpRtcpModule.SetAudioLevel(_rtpAudioProc->level_estimator()->RMS());
+ }
+
+ // Push data from ACM to RTP/RTCP-module to deliver audio frame for
+ // packetization.
+ // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
+ if (_rtpRtcpModule.SendOutgoingData((FrameType&)frameType,
+ payloadType,
+ timeStamp,
+ payloadData,
+ payloadSize,
+ fragmentation) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
+ "Channel::SendData() failed to send data to RTP/RTCP module");
+ return -1;
+ }
+
+ _lastLocalTimeStamp = timeStamp;
+ _lastPayloadType = payloadType;
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::InFrameType(WebRtc_Word16 frameType)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::InFrameType(frameType=%d)", frameType);
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ // 1 indicates speech
+ _sendFrameType = (frameType == 1) ? 1 : 0;
+ return 0;
+}
+
+#ifdef WEBRTC_DTMF_DETECTION
+int
+Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::IncomingDtmf(digitDtmf=%u, end=%d)",
+ digitDtmf, end);
+
+ if (digitDtmf != 999)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_telephoneEventDetectionPtr)
+ {
+ _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband(
+ _channelId, digitDtmf, end);
+ }
+ }
+
+ return 0;
+}
+#endif
+
+WebRtc_Word32
+Channel::OnRxVadDetected(const int vadDecision)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_rxVadObserverPtr)
+ {
+ _rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
+ }
+
+ return 0;
+}
+
+int
+Channel::SendPacket(int channel, const void *data, int len)
+{
+ channel = VoEChannelId(channel);
+ assert(channel == _channelId);
+
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SendPacket(channel=%d, len=%d)", channel, len);
+
+ if (_transportPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() failed to send RTP packet due to"
+ " invalid transport object");
+ return -1;
+ }
+
+ // Insert extra RTP packet using if user has called the InsertExtraRTPPacket
+ // API
+ if (_insertExtraRTPPacket)
+ {
+ WebRtc_UWord8* rtpHdr = (WebRtc_UWord8*)data;
+ WebRtc_UWord8 M_PT(0);
+ if (_extraMarkerBit)
+ {
+ M_PT = 0x80; // set the M-bit
+ }
+ M_PT += _extraPayloadType; // set the payload type
+ *(++rtpHdr) = M_PT; // modify the M|PT-byte within the RTP header
+ _insertExtraRTPPacket = false; // insert one packet only
+ }
+
+ WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
+ WebRtc_Word32 bufferLength = len;
+
+ // Dump the RTP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTP dump to output file failed");
+ }
+
+ // SRTP or External encryption
+ if (_encrypting)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encryptionPtr)
+ {
+ if (!_encryptionRTPBufferPtr)
+ {
+ // Allocate memory for encryption buffer one time only
+ _encryptionRTPBufferPtr =
+ new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
+ }
+
+ // Perform encryption (SRTP or external)
+ WebRtc_Word32 encryptedBufferLength = 0;
+ _encryptionPtr->encrypt(_channelId,
+ bufferToSendPtr,
+ _encryptionRTPBufferPtr,
+ bufferLength,
+ (int*)&encryptedBufferLength);
+ if (encryptedBufferLength <= 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ENCRYPTION_FAILED,
+ kTraceError, "Channel::SendPacket() encryption failed");
+ return -1;
+ }
+
+ // Replace default data buffer with encrypted buffer
+ bufferToSendPtr = _encryptionRTPBufferPtr;
+ bufferLength = encryptedBufferLength;
+ }
+ }
+
+ // Packet transmission using WebRtc socket transport
+ if (!_externalTransport)
+ {
+ int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
+ bufferLength);
+ if (n < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTP transmission using WebRtc"
+ " sockets failed");
+ return -1;
+ }
+ return n;
+ }
+
+ // Packet transmission using external transport transport
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ int n = _transportPtr->SendPacket(channel,
+ bufferToSendPtr,
+ bufferLength);
+ if (n < 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTP transmission using external"
+ " transport failed");
+ return -1;
+ }
+ return n;
+ }
+}
+
+int
+Channel::SendRTCPPacket(int channel, const void *data, int len)
+{
+ channel = VoEChannelId(channel);
+ assert(channel == _channelId);
+
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
+
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_transportPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendRTCPPacket() failed to send RTCP packet"
+ " due to invalid transport object");
+ return -1;
+ }
+ }
+
+ WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
+ WebRtc_Word32 bufferLength = len;
+
+ // Dump the RTCP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTCP dump to output file failed");
+ }
+
+ // SRTP or External encryption
+ if (_encrypting)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encryptionPtr)
+ {
+ if (!_encryptionRTCPBufferPtr)
+ {
+ // Allocate memory for encryption buffer one time only
+ _encryptionRTCPBufferPtr =
+ new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
+ }
+
+ // Perform encryption (SRTP or external).
+ WebRtc_Word32 encryptedBufferLength = 0;
+ _encryptionPtr->encrypt_rtcp(_channelId,
+ bufferToSendPtr,
+ _encryptionRTCPBufferPtr,
+ bufferLength,
+ (int*)&encryptedBufferLength);
+ if (encryptedBufferLength <= 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ENCRYPTION_FAILED, kTraceError,
+ "Channel::SendRTCPPacket() encryption failed");
+ return -1;
+ }
+
+ // Replace default data buffer with encrypted buffer
+ bufferToSendPtr = _encryptionRTCPBufferPtr;
+ bufferLength = encryptedBufferLength;
+ }
+ }
+
+ // Packet transmission using WebRtc socket transport
+ if (!_externalTransport)
+ {
+ int n = _transportPtr->SendRTCPPacket(channel,
+ bufferToSendPtr,
+ bufferLength);
+ if (n < 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendRTCPPacket() transmission using WebRtc"
+ " sockets failed");
+ return -1;
+ }
+ return n;
+ }
+
+ // Packet transmission using external transport transport
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ int n = _transportPtr->SendRTCPPacket(channel,
+ bufferToSendPtr,
+ bufferLength);
+ if (n < 0)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendRTCPPacket() transmission using external"
+ " transport failed");
+ return -1;
+ }
+ return n;
+ }
+
+ return len;
+}
+
+void
+Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::IncomingRTPPacket(rtpPacketLength=%d,"
+ " fromIP=%s, fromPort=%u)",
+ rtpPacketLength, fromIP, fromPort);
+
+ // Store playout timestamp for the received RTP packet
+ // to be used for upcoming delay estimations
+ WebRtc_UWord32 playoutTimestamp(0);
+ if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
+ {
+ _playoutTimeStampRTP = playoutTimestamp;
+ }
+
+ WebRtc_UWord8* rtpBufferPtr = (WebRtc_UWord8*)incomingRtpPacket;
+ WebRtc_Word32 rtpBufferLength = rtpPacketLength;
+
+ // SRTP or External decryption
+ if (_decrypting)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encryptionPtr)
+ {
+ if (!_decryptionRTPBufferPtr)
+ {
+ // Allocate memory for decryption buffer one time only
+ _decryptionRTPBufferPtr =
+ new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
+ }
+
+ // Perform decryption (SRTP or external)
+ WebRtc_Word32 decryptedBufferLength = 0;
+ _encryptionPtr->decrypt(_channelId,
+ rtpBufferPtr,
+ _decryptionRTPBufferPtr,
+ rtpBufferLength,
+ (int*)&decryptedBufferLength);
+ if (decryptedBufferLength <= 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_DECRYPTION_FAILED, kTraceError,
+ "Channel::IncomingRTPPacket() decryption failed");
+ return;
+ }
+
+ // Replace default data buffer with decrypted buffer
+ rtpBufferPtr = _decryptionRTPBufferPtr;
+ rtpBufferLength = decryptedBufferLength;
+ }
+ }
+
+ // Dump the RTP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpIn.DumpPacket(rtpBufferPtr,
+ (WebRtc_UWord16)rtpBufferLength) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTP dump to input file failed");
+ }
+
+ // Deliver RTP packet to RTP/RTCP module for parsing
+ // The packet will be pushed back to the channel thru the
+ // OnReceivedPayloadData callback so we don't push it to the ACM here
+ if (_rtpRtcpModule.IncomingPacket((const WebRtc_UWord8*)rtpBufferPtr,
+ (WebRtc_UWord16)rtpBufferLength) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
+ "Channel::IncomingRTPPacket() RTP packet is invalid");
+ return;
+ }
+}
+
+void
+Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::IncomingRTCPPacket(rtcpPacketLength=%d, fromIP=%s,"
+ " fromPort=%u)",
+ rtcpPacketLength, fromIP, fromPort);
+
+ // Temporary buffer pointer and size for decryption
+ WebRtc_UWord8* rtcpBufferPtr = (WebRtc_UWord8*)incomingRtcpPacket;
+ WebRtc_Word32 rtcpBufferLength = rtcpPacketLength;
+
+ // Store playout timestamp for the received RTCP packet
+ // which will be read by the GetRemoteRTCPData API
+ WebRtc_UWord32 playoutTimestamp(0);
+ if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
+ {
+ _playoutTimeStampRTCP = playoutTimestamp;
+ }
+
+ // SRTP or External decryption
+ if (_decrypting)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encryptionPtr)
+ {
+ if (!_decryptionRTCPBufferPtr)
+ {
+ // Allocate memory for decryption buffer one time only
+ _decryptionRTCPBufferPtr =
+ new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
+ }
+
+ // Perform decryption (SRTP or external).
+ WebRtc_Word32 decryptedBufferLength = 0;
+ _encryptionPtr->decrypt_rtcp(_channelId,
+ rtcpBufferPtr,
+ _decryptionRTCPBufferPtr,
+ rtcpBufferLength,
+ (int*)&decryptedBufferLength);
+ if (decryptedBufferLength <= 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_DECRYPTION_FAILED, kTraceError,
+ "Channel::IncomingRTCPPacket() decryption failed");
+ return;
+ }
+
+ // Replace default data buffer with decrypted buffer
+ rtcpBufferPtr = _decryptionRTCPBufferPtr;
+ rtcpBufferLength = decryptedBufferLength;
+ }
+ }
+
+ // Dump the RTCP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpIn.DumpPacket(rtcpBufferPtr,
+ (WebRtc_UWord16)rtcpBufferLength) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTCP dump to input file failed");
+ }
+
+ // Deliver RTCP packet to RTP/RTCP module for parsing
+ if (_rtpRtcpModule.IncomingPacket((const WebRtc_UWord8*)rtcpBufferPtr,
+ (WebRtc_UWord16)rtcpBufferLength) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
+ "Channel::IncomingRTPPacket() RTCP packet is invalid");
+ return;
+ }
+}
+
+void
+Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const bool endOfEvent)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnReceivedTelephoneEvent(id=%d, event=%u,"
+ " endOfEvent=%d)", id, event, endOfEvent);
+
+#ifdef WEBRTC_DTMF_DETECTION
+ if (_outOfBandTelephoneEventDetecion)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_telephoneEventDetectionPtr)
+ {
+ _telephoneEventDetectionPtr->OnReceivedTelephoneEventOutOfBand(
+ _channelId, event, endOfEvent);
+ }
+ }
+#endif
+}
+
+void
+Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const WebRtc_UWord16 lengthMs,
+ const WebRtc_UWord8 volume)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
+ " volume=%u)", id, event, lengthMs, volume);
+
+ if (!_playOutbandDtmfEvent || (event > 15))
+ {
+ // Ignore callback since feedback is disabled or event is not a
+ // Dtmf tone event.
+ return;
+ }
+
+ assert(_outputMixerPtr != NULL);
+
+ // Start playing out the Dtmf tone (if playout is enabled).
+ // Reduce length of tone with 80ms to the reduce risk of echo.
+ _outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume);
+}
+
+void
+Channel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
+ id, SSRC);
+
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+
+ // Reset RTP-module counters since a new incoming RTP stream is detected
+ _rtpRtcpModule.ResetReceiveDataCountersRTP();
+ _rtpRtcpModule.ResetStatisticsRTP();
+
+ if (_rtpObserver)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rtpObserverPtr)
+ {
+ // Send new SSRC to registered observer using callback
+ _rtpObserverPtr->OnIncomingSSRCChanged(channel, SSRC);
+ }
+ }
+}
+
+void Channel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC,
+ const bool added)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
+ id, CSRC, added);
+
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+
+ if (_rtpObserver)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rtpObserverPtr)
+ {
+ _rtpObserverPtr->OnIncomingCSRCChanged(channel, CSRC, added);
+ }
+ }
+}
+
+void
+Channel::OnApplicationDataReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8* data)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnApplicationDataReceived(id=%d, subType=%u,"
+ " name=%u, length=%u)",
+ id, subType, name, length);
+
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+
+ if (_rtcpObserver)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rtcpObserverPtr)
+ {
+ _rtcpObserverPtr->OnApplicationDataReceived(channel,
+ subType,
+ name,
+ data,
+ length);
+ }
+ }
+}
+
+WebRtc_Word32
+Channel::OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
+ "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
+ id, payloadType, payloadName, frequency, channels, rate);
+
+ assert(VoEChannelId(id) == _channelId);
+
+ CodecInst receiveCodec = {0};
+ CodecInst dummyCodec = {0};
+
+ receiveCodec.pltype = payloadType;
+ receiveCodec.plfreq = frequency;
+ receiveCodec.channels = channels;
+ receiveCodec.rate = rate;
+ strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+
+ _audioCodingModule.Codec(payloadName, dummyCodec, frequency);
+ receiveCodec.pacsize = dummyCodec.pacsize;
+
+ // Register the new codec to the ACM
+ if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::OnInitializeDecoder() invalid codec ("
+ "pt=%d, name=%s) received - 1", payloadType, payloadName);
+ _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
+ return -1;
+ }
+
+ return 0;
+}
+
+void
+Channel::OnPacketTimeout(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnPacketTimeout(id=%d)", id);
+
+ CriticalSectionScoped cs(*_callbackCritSectPtr);
+ if (_voiceEngineObserverPtr)
+ {
+ if (_receiving || _externalTransport)
+ {
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+ // Ensure that next OnReceivedPacket() callback will trigger
+ // a VE_PACKET_RECEIPT_RESTARTED callback.
+ _rtpPacketTimedOut = true;
+ // Deliver callback to the observer
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::OnPacketTimeout() => "
+ "CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)");
+ _voiceEngineObserverPtr->CallbackOnError(channel,
+ VE_RECEIVE_PACKET_TIMEOUT);
+ }
+ }
+}
+
+void
+Channel::OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnReceivedPacket(id=%d, packetType=%d)",
+ id, packetType);
+
+ assert(VoEChannelId(id) == _channelId);
+
+ // Notify only for the case when we have restarted an RTP session.
+ if (_rtpPacketTimedOut && (kPacketRtp == packetType))
+ {
+ CriticalSectionScoped cs(*_callbackCritSectPtr);
+ if (_voiceEngineObserverPtr)
+ {
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+ // Reset timeout mechanism
+ _rtpPacketTimedOut = false;
+ // Deliver callback to the observer
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::OnPacketTimeout() =>"
+ " CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)");
+ _voiceEngineObserverPtr->CallbackOnError(
+ channel,
+ VE_PACKET_RECEIPT_RESTARTED);
+ }
+ }
+}
+
+void
+Channel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive);
+
+ if (!_connectionObserver)
+ return;
+
+ WebRtc_Word32 channel = VoEChannelId(id);
+ assert(channel == _channelId);
+
+ // Use Alive as default to limit risk of false Dead detections
+ bool isAlive(true);
+
+ // Always mark the connection as Dead when the module reports kRtpDead
+ if (kRtpDead == alive)
+ {
+ isAlive = false;
+ }
+
+ // It is possible that the connection is alive even if no RTP packet has
+ // been received for a long time since the other side might use VAD/DTX
+ // and a low SID-packet update rate.
+ if ((kRtpNoRtp == alive) && _playing)
+ {
+ // Detect Alive for all NetEQ states except for the case when we are
+ // in PLC_CNG state.
+ // PLC_CNG <=> background noise only due to long expand or error.
+ // Note that, the case where the other side stops sending during CNG
+ // state will be detected as Alive. Dead is is not set until after
+ // missing RTCP packets for at least twelve seconds (handled
+ // internally by the RTP/RTCP module).
+ isAlive = (_outputSpeechType != AudioFrame::kPLCCNG);
+ }
+
+ UpdateDeadOrAliveCounters(isAlive);
+
+ // Send callback to the registered observer
+ if (_connectionObserver)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_connectionObserverPtr)
+ {
+ _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive);
+ }
+ }
+}
+
+WebRtc_Word32
+Channel::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::OnReceivedPayloadData(payloadSize=%d,"
+ " payloadType=%u, audioChannel=%u)",
+ payloadSize,
+ rtpHeader->header.payloadType,
+ rtpHeader->type.Audio.channel);
+
+ if (!_playing)
+ {
+ // Avoid inserting into NetEQ when we are not playing. Count the
+ // packet as discarded.
+ WEBRTC_TRACE(kTraceStream, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "received packet is discarded since playing is not"
+ " activated");
+ _numberOfDiscardedPackets++;
+ return 0;
+ }
+
+ // Push the incoming payload (parsed and ready for decoding) into the ACM
+ if (_audioCodingModule.IncomingPacket((const WebRtc_Word8*) payloadData,
+ payloadSize,
+ *rtpHeader) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
+ "Channel::OnReceivedPayloadData() unable to push data to the ACM");
+ return -1;
+ }
+
+ // Update the packet delay
+ UpdatePacketDelay(rtpHeader->header.timestamp,
+ rtpHeader->header.sequenceNumber);
+
+ return 0;
+}
+
+WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
+ AudioFrame& audioFrame)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetAudioFrame(id=%d)", id);
+
+ // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
+ if (_audioCodingModule.PlayoutData10Ms(audioFrame._frequencyInHz,
+ audioFrame) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
+ // In all likelihood, the audio in this frame is garbage. We return an
+ // error so that the audio mixer module doesn't add it to the mix. As
+ // a result, it won't be played out and the actions skipped here are
+ // irrelevant.
+ return -1;
+ }
+
+ if (_RxVadDetection)
+ {
+ UpdateRxVadDetection(audioFrame);
+ }
+
+ // Convert module ID to internal VoE channel ID
+ audioFrame._id = VoEChannelId(audioFrame._id);
+ // Store speech type for dead-or-alive detection
+ _outputSpeechType = audioFrame._speechType;
+
+ // Perform far-end AudioProcessing module processing on the received signal
+ if (_rxApmIsEnabled)
+ {
+ ApmProcessRx(audioFrame);
+ }
+
+ // Output volume scaling
+ if (_outputGain < 0.99f || _outputGain > 1.01f)
+ {
+ AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame);
+ }
+
+ // Scale left and/or right channel(s) if stereo and master balance is
+ // active
+
+ if (_panLeft != 1.0f || _panRight != 1.0f)
+ {
+ if (audioFrame._audioChannel == 1)
+ {
+ // Emulate stereo mode since panning is active.
+ // The mono signal is copied to both left and right channels here.
+ AudioFrameOperations::MonoToStereo(audioFrame);
+ }
+ // For true stereo mode (when we are receiving a stereo signal), no
+ // action is needed.
+
+ // Do the panning operation (the audio frame contains stereo at this
+ // stage)
+ AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame);
+ }
+
+ // Mix decoded PCM output with file if file mixing is enabled
+ if (_outputFilePlaying)
+ {
+ MixAudioWithFile(audioFrame, audioFrame._frequencyInHz);
+ }
+
+ // Place channel in on-hold state (~muted) if on-hold is activated
+ if (_outputIsOnHold)
+ {
+ AudioFrameOperations::Mute(audioFrame);
+ }
+
+ // External media
+ if (_outputExternalMedia)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ const bool isStereo = (audioFrame._audioChannel == 2);
+ if (_outputExternalMediaCallbackPtr)
+ {
+ _outputExternalMediaCallbackPtr->Process(
+ _channelId,
+ kPlaybackPerChannel,
+ (WebRtc_Word16*)audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples,
+ audioFrame._frequencyInHz,
+ isStereo);
+ }
+ }
+
+ // Record playout if enabled
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFileRecording && _outputFileRecorderPtr)
+ {
+ if(audioFrame._audioChannel == 2)
+ {
+ AudioFrame temp = audioFrame;
+ AudioFrameOperations::StereoToMono (temp);
+ _outputFileRecorderPtr->RecordAudioToFile(temp);
+ }
+ else if(audioFrame._audioChannel == 1)
+ {
+ _outputFileRecorderPtr->RecordAudioToFile(audioFrame);
+ }
+ else
+ {
+ assert(false);
+ }
+ }
+ }
+
+ // Measure audio level (0-9)
+ _outputAudioLevel.ComputeLevel(audioFrame);
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::NeededFrequency(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::NeededFrequency(id=%d)", id);
+
+ int highestNeeded = 0;
+
+ // Determine highest needed receive frequency
+ WebRtc_Word32 receiveFrequency = _audioCodingModule.ReceiveFrequency();
+
+ // Return the bigger of playout and receive frequency in the ACM.
+ if (_audioCodingModule.PlayoutFrequency() > receiveFrequency)
+ {
+ highestNeeded = _audioCodingModule.PlayoutFrequency();
+ }
+ else
+ {
+ highestNeeded = receiveFrequency;
+ }
+
+ // Special case, if we're playing a file on the playout side
+ // we take that frequency into consideration as well
+ // This is not needed on sending side, since the codec will
+ // limit the spectrum anyway.
+ if (_outputFilePlaying)
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ if (_outputFilePlayerPtr && _outputFilePlaying)
+ {
+ if(_outputFilePlayerPtr->Frequency()>highestNeeded)
+ {
+ highestNeeded=_outputFilePlayerPtr->Frequency();
+ }
+ }
+ }
+
+ return(highestNeeded);
+}
+
+WebRtc_Word32
+Channel::CreateChannel(Channel*& channel,
+ const WebRtc_Word32 channelId,
+ const WebRtc_UWord32 instanceId)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
+ "Channel::CreateChannel(channelId=%d, instanceId=%d)",
+ channelId, instanceId);
+
+ channel = new Channel(channelId, instanceId);
+ if (channel == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice,
+ VoEId(instanceId,channelId),
+ "Channel::CreateChannel() unable to allocate memory for"
+ " channel");
+ return -1;
+ }
+ return 0;
+}
+
+void
+Channel::PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::PlayNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void
+Channel::RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RecordNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void
+Channel::PlayFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::PlayFileEnded(id=%d)", id);
+
+ if (id == _inputFilePlayerId)
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ _inputFilePlaying = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::PlayFileEnded() => input file player module is"
+ " shutdown");
+ }
+ else if (id == _outputFilePlayerId)
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ _outputFilePlaying = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::PlayFileEnded() => output file player module is"
+ " shutdown");
+ }
+}
+
+void
+Channel::RecordFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RecordFileEnded(id=%d)", id);
+
+ assert(id == _outputFileRecorderId);
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ _outputFileRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::RecordFileEnded() => output file recorder module is"
+ " shutdown");
+}
+
+Channel::Channel(const WebRtc_Word32 channelId,
+ const WebRtc_UWord32 instanceId) :
+ _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _transmitCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _instanceId(instanceId),
+ _channelId(channelId),
+ _rtpRtcpModule(*RtpRtcp::CreateRtpRtcp(VoEModuleId(
+ instanceId, channelId), true)),
+ _audioCodingModule(*AudioCodingModule::Create(
+ VoEModuleId(instanceId, channelId))),
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ _numSocketThreads(KNumSocketThreads),
+ _socketTransportModule(*UdpTransport::Create(
+ VoEModuleId(instanceId, channelId), _numSocketThreads)),
+#endif
+#ifdef WEBRTC_SRTP
+ _srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId,
+ channelId))),
+#endif
+ _rtpDumpIn(*RtpDump::CreateRtpDump()),
+ _rtpDumpOut(*RtpDump::CreateRtpDump()),
+ _outputAudioLevel(),
+ _externalTransport(false),
+ _inputFilePlayerPtr(NULL),
+ _outputFilePlayerPtr(NULL),
+ _outputFileRecorderPtr(NULL),
+ // Avoid conflict with other channels by adding 1024 - 1026,
+ // won't use as much as 1024 channels.
+ _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
+ _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
+ _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
+ _inputFilePlaying(false),
+ _outputFilePlaying(false),
+ _outputFileRecording(false),
+ _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+ _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
+ _inputExternalMedia(false),
+ _outputExternalMedia(false),
+ _inputExternalMediaCallbackPtr(NULL),
+ _outputExternalMediaCallbackPtr(NULL),
+ _encryptionRTPBufferPtr(NULL),
+ _decryptionRTPBufferPtr(NULL),
+ _encryptionRTCPBufferPtr(NULL),
+ _decryptionRTCPBufferPtr(NULL),
+ _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
+ _sendTelephoneEventPayloadType(106),
+ _playoutTimeStampRTP(0),
+ _playoutTimeStampRTCP(0),
+ _numberOfDiscardedPackets(0),
+ _engineStatisticsPtr(NULL),
+ _outputMixerPtr(NULL),
+ _transmitMixerPtr(NULL),
+ _moduleProcessThreadPtr(NULL),
+ _audioDeviceModulePtr(NULL),
+ _voiceEngineObserverPtr(NULL),
+ _callbackCritSectPtr(NULL),
+ _transportPtr(NULL),
+ _encryptionPtr(NULL),
+ _rtpAudioProc(NULL),
+ _rxAudioProcessingModulePtr(NULL),
+#ifdef WEBRTC_DTMF_DETECTION
+ _telephoneEventDetectionPtr(NULL),
+#endif
+ _rxVadObserverPtr(NULL),
+ _oldVadDecision(-1),
+ _sendFrameType(0),
+ _rtpObserverPtr(NULL),
+ _rtcpObserverPtr(NULL),
+ _outputIsOnHold(false),
+ _externalPlayout(false),
+ _inputIsOnHold(false),
+ _playing(false),
+ _sending(false),
+ _receiving(false),
+ _mixFileWithMicrophone(false),
+ _rtpObserver(false),
+ _rtcpObserver(false),
+ _mute(false),
+ _panLeft(1.0f),
+ _panRight(1.0f),
+ _outputGain(1.0f),
+ _encrypting(false),
+ _decrypting(false),
+ _playOutbandDtmfEvent(false),
+ _playInbandDtmfEvent(false),
+ _inbandTelephoneEventDetection(false),
+ _outOfBandTelephoneEventDetecion(false),
+ _extraPayloadType(0),
+ _insertExtraRTPPacket(false),
+ _extraMarkerBit(false),
+ _lastLocalTimeStamp(0),
+ _lastPayloadType(0),
+ _includeAudioLevelIndication(false),
+ _rtpPacketTimedOut(false),
+ _rtpPacketTimeOutIsEnabled(false),
+ _rtpTimeOutSeconds(0),
+ _connectionObserver(false),
+ _connectionObserverPtr(NULL),
+ _countAliveDetections(0),
+ _countDeadDetections(0),
+ _outputSpeechType(AudioFrame::kNormalSpeech),
+ _averageDelayMs(0),
+ _previousSequenceNumber(0),
+ _previousTimestamp(0),
+ _recPacketDelayMs(20),
+ _RxVadDetection(false),
+ _rxApmIsEnabled(false),
+ _rxAgcIsEnabled(false),
+ _rxNsIsEnabled(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::Channel() - ctor");
+ _inbandDtmfQueue.ResetDtmf();
+ _inbandDtmfGenerator.Init();
+ _outputAudioLevel.Clear();
+
+ // Create far end AudioProcessing Module
+ _rxAudioProcessingModulePtr = AudioProcessing::Create(
+ VoEModuleId(instanceId, channelId));
+}
+
+Channel::~Channel()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::~Channel() - dtor");
+
+ if (_outputExternalMedia)
+ {
+ DeRegisterExternalMediaProcessing(kPlaybackPerChannel);
+ }
+ if (_inputExternalMedia)
+ {
+ DeRegisterExternalMediaProcessing(kRecordingPerChannel);
+ }
+ StopSend();
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ StopReceiving();
+ // De-register packet callback to ensure we're not in a callback when
+ // deleting channel state, avoids race condition and deadlock.
+ if (_socketTransportModule.InitializeReceiveSockets(NULL, 0, NULL, NULL, 0)
+ != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "~Channel() failed to de-register receive callback");
+ }
+#endif
+ StopPlayout();
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ if (_inputFilePlayerPtr)
+ {
+ _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ _inputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ }
+ if (_outputFilePlayerPtr)
+ {
+ _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ _outputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ }
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+ }
+
+ // The order to safely shutdown modules in a channel is:
+ // 1. De-register callbacks in modules
+ // 2. De-register modules in process thread
+ // 3. Destroy modules
+
+ // De-register all RTP module callbacks to ensure geting no callbacks
+ // (Receive socket callback was de-registered above)
+ if (_rtpRtcpModule.RegisterIncomingDataCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register incoming data callback"
+ " (RTP module)");
+ }
+ if (_rtpRtcpModule.RegisterSendTransport(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register send transport "
+ "(RTP module)");
+ }
+ if (_rtpRtcpModule.RegisterIncomingRTPCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register incoming RTP"
+ " callback (RTP module)");
+ }
+ if (_rtpRtcpModule.RegisterIncomingRTCPCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register incoming RTCP "
+ "callback (RTP module)");
+ }
+ if (_rtpRtcpModule.RegisterAudioCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register audio callback "
+ "(RTP module)");
+ }
+ if (_audioCodingModule.RegisterTransportCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register transport callback"
+ " (Audio coding module)");
+ }
+ if (_audioCodingModule.RegisterVADCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register VAD callback"
+ " (Audio coding module)");
+ }
+#ifdef WEBRTC_DTMF_DETECTION
+ if (_audioCodingModule.RegisterIncomingMessagesCallback(NULL) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to de-register incoming messages "
+ "callback (Audio coding module)");
+ }
+#endif
+ // De-register modules in process thread
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule)
+ == -1)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to deregister socket module");
+ }
+#endif
+ if (_moduleProcessThreadPtr->DeRegisterModule(&_rtpRtcpModule) == -1)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "~Channel() failed to deregister RTP/RTCP module");
+ }
+
+ // Destroy modules
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ UdpTransport::Destroy(
+ &_socketTransportModule);
+#endif
+ RtpRtcp::DestroyRtpRtcp(&_rtpRtcpModule);
+ AudioCodingModule::Destroy(&_audioCodingModule);
+#ifdef WEBRTC_SRTP
+ SrtpModule::DestroySrtpModule(&_srtpModule);
+#endif
+ if (_rxAudioProcessingModulePtr != NULL)
+ {
+ AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM
+ _rxAudioProcessingModulePtr = NULL;
+ }
+
+ // End of modules shutdown
+
+ // Delete other objects
+ RtpDump::DestroyRtpDump(&_rtpDumpIn);
+ RtpDump::DestroyRtpDump(&_rtpDumpOut);
+ delete [] _encryptionRTPBufferPtr;
+ delete [] _decryptionRTPBufferPtr;
+ delete [] _encryptionRTCPBufferPtr;
+ delete [] _decryptionRTCPBufferPtr;
+ delete &_callbackCritSect;
+ delete &_transmitCritSect;
+ delete &_fileCritSect;
+}
+
+WebRtc_Word32
+Channel::Init()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::Init()");
+
+ // --- Initial sanity
+
+ if ((_engineStatisticsPtr == NULL) ||
+ (_moduleProcessThreadPtr == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() must call SetEngineInformation() first");
+ return -1;
+ }
+
+ // --- Add modules to process thread (for periodic schedulation)
+
+ const bool processThreadFail =
+ ((_moduleProcessThreadPtr->RegisterModule(&_rtpRtcpModule) != 0) ||
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ (_moduleProcessThreadPtr->RegisterModule(
+ &_socketTransportModule) != 0));
+#else
+ false);
+#endif
+ if (processThreadFail)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_INIT_CHANNEL, kTraceError,
+ "Channel::Init() modules not registered");
+ return -1;
+ }
+ // --- ACM initialization
+
+ if ((_audioCodingModule.InitializeReceiver() == -1) ||
+#ifdef WEBRTC_CODEC_AVT
+ // out-of-band Dtmf tones are played out by default
+ (_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) ||
+#endif
+ (_audioCodingModule.InitializeSender() == -1))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "Channel::Init() unable to initialize the ACM - 1");
+ return -1;
+ }
+
+ // --- RTP/RTCP module initialization
+
+ // Ensure that RTCP is enabled by default for the created channel.
+ // Note that, the module will keep generating RTCP until it is explicitly
+ // disabled by the user.
+ // After StopListen (when no sockets exists), RTCP packets will no longer
+ // be transmitted since the Transport object will then be invalid.
+
+ const bool rtpRtcpFail =
+ ((_rtpRtcpModule.InitReceiver() == -1) ||
+ (_rtpRtcpModule.InitSender() == -1) ||
+ (_rtpRtcpModule.SetTelephoneEventStatus(false, true, true) == -1) ||
+ // RTCP is enabled by default
+ (_rtpRtcpModule.SetRTCPStatus(kRtcpCompound) == -1));
+ if (rtpRtcpFail)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "Channel::Init() RTP/RTCP module not initialized");
+ return -1;
+ }
+
+ // --- Register all permanent callbacks
+
+ const bool fail =
+ (_rtpRtcpModule.RegisterIncomingDataCallback(this) == -1) ||
+ (_rtpRtcpModule.RegisterIncomingRTPCallback(this) == -1) ||
+ (_rtpRtcpModule.RegisterIncomingRTCPCallback(this) == -1) ||
+ (_rtpRtcpModule.RegisterSendTransport(this) == -1) ||
+ (_rtpRtcpModule.RegisterAudioCallback(this) == -1) ||
+ (_audioCodingModule.RegisterTransportCallback(this) == -1) ||
+ (_audioCodingModule.RegisterVADCallback(this) == -1);
+
+ if (fail)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_INIT_CHANNEL, kTraceError,
+ "Channel::Init() callbacks not registered");
+ return -1;
+ }
+
+ // --- Register all supported codecs to the receiving side of the
+ // RTP/RTCP module
+
+ CodecInst codec;
+ const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
+
+ for (int idx = 0; idx < nSupportedCodecs; idx++)
+ {
+ // Open up the RTP/RTCP receiver for all supported codecs
+ if ((_audioCodingModule.Codec(idx, codec) == -1) ||
+ (_rtpRtcpModule.RegisterReceivePayload(codec) == -1))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() unable to register %s (%d/%d/%d/%d) "
+ "to RTP/RTCP receiver",
+ codec.plname, codec.pltype, codec.plfreq,
+ codec.channels, codec.rate);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() %s (%d/%d/%d/%d) has been added to "
+ "the RTP/RTCP receiver",
+ codec.plname, codec.pltype, codec.plfreq,
+ codec.channels, codec.rate);
+ }
+
+ // Ensure that PCMU is used as default codec on the sending side
+ if (!STR_CASE_CMP(codec.plname, "PCMU"))
+ {
+ SetSendCodec(codec);
+ }
+
+ // Register default PT for outband 'telephone-event'
+ if (!STR_CASE_CMP(codec.plname, "telephone-event"))
+ {
+ if ((_rtpRtcpModule.RegisterSendPayload(codec) == -1) ||
+ (_audioCodingModule.RegisterReceiveCodec(codec) == -1))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() failed to register outband "
+ "'telephone-event' (%d/%d) correctly",
+ codec.pltype, codec.plfreq);
+ }
+ }
+
+ if (!STR_CASE_CMP(codec.plname, "CN"))
+ {
+ if ((_audioCodingModule.RegisterSendCodec(codec) == -1) ||
+ (_audioCodingModule.RegisterReceiveCodec(codec) == -1) ||
+ (_rtpRtcpModule.RegisterSendPayload(codec) == -1))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() failed to register CN (%d/%d) "
+ "correctly - 1",
+ codec.pltype, codec.plfreq);
+ }
+ }
+#ifdef WEBRTC_CODEC_RED
+ // Register RED to the receiving side of the ACM.
+ // We will not receive an OnInitializeDecoder() callback for RED.
+ if (!STR_CASE_CMP(codec.plname, "RED"))
+ {
+ if (_audioCodingModule.RegisterReceiveCodec(codec) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::Init() failed to register RED (%d/%d) "
+ "correctly",
+ codec.pltype, codec.plfreq);
+ }
+ }
+#endif
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ // Ensure that the WebRtcSocketTransport implementation is used as
+ // Transport on the sending side
+ {
+ // A lock is needed here since users can call
+ // RegisterExternalTransport() at the same time.
+ CriticalSectionScoped cs(_callbackCritSect);
+ _transportPtr = &_socketTransportModule;
+ }
+#endif
+
+ // Initialize the far end AP module
+ // Using 8 kHz as initial Fs, the same as in transmission. Might be
+ // changed at the first receiving audio.
+ if (_rxAudioProcessingModulePtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_NO_MEMORY, kTraceCritical,
+ "Channel::Init() failed to create the far-end AudioProcessing"
+ " module");
+ return -1;
+ }
+
+ if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Channel::Init() failed to set the sample rate to 8K for"
+ " far-end AP module");
+ }
+
+ if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOUNDCARD_ERROR, kTraceWarning,
+ "Init() failed to set channels for the primary audio stream");
+ }
+
+ if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable(
+ WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Channel::Init() failed to set the high-pass filter for"
+ " far-end AP module");
+ }
+
+ if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(
+ (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Init() failed to set noise reduction level for far-end"
+ " AP module");
+ }
+ if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(
+ WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Init() failed to set noise reduction state for far-end"
+ " AP module");
+ }
+
+ if (_rxAudioProcessingModulePtr->gain_control()->set_mode(
+ (GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Init() failed to set AGC mode for far-end AP module");
+ }
+ if (_rxAudioProcessingModulePtr->gain_control()->Enable(
+ WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "Init() failed to set AGC state for far-end AP module");
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetEngineInformation(Statistics& engineStatistics,
+ OutputMixer& outputMixer,
+ voe::TransmitMixer& transmitMixer,
+ ProcessThread& moduleProcessThread,
+ AudioDeviceModule& audioDeviceModule,
+ VoiceEngineObserver* voiceEngineObserver,
+ CriticalSectionWrapper* callbackCritSect)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetEngineInformation()");
+ _engineStatisticsPtr = &engineStatistics;
+ _outputMixerPtr = &outputMixer;
+ _transmitMixerPtr = &transmitMixer,
+ _moduleProcessThreadPtr = &moduleProcessThread;
+ _audioDeviceModulePtr = &audioDeviceModule;
+ _voiceEngineObserverPtr = voiceEngineObserver;
+ _callbackCritSectPtr = callbackCritSect;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::UpdateLocalTimeStamp()
+{
+
+ _timeStamp += _audioFrame._payloadDataLengthInSamples;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StartPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartPlayout()");
+ if (_playing)
+ {
+ return 0;
+ }
+ // Add participant as candidates for mixing.
+ if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
+ "StartPlayout() failed to add participant to mixer");
+ return -1;
+ }
+
+ _playing = true;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StopPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StopPlayout()");
+ if (!_playing)
+ {
+ return 0;
+ }
+ // Remove participant as candidates for mixing
+ if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
+ "StartPlayout() failed to remove participant from mixer");
+ return -1;
+ }
+
+ _playing = false;
+ _outputAudioLevel.Clear();
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StartSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartSend()");
+ {
+ // A lock is needed because |_sending| can be accessed or modified by
+ // another thread at the same time.
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_sending)
+ {
+ return 0;
+ }
+ _sending = true;
+ }
+
+ if (_rtpRtcpModule.SetSendingStatus(true) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "StartSend() RTP/RTCP failed to start sending");
+ CriticalSectionScoped cs(_callbackCritSect);
+ _sending = false;
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StopSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StopSend()");
+ {
+ // A lock is needed because |_sending| can be accessed or modified by
+ // another thread at the same time.
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_sending)
+ {
+ return 0;
+ }
+ _sending = false;
+ }
+
+ // Reset sending SSRC and sequence number and triggers direct transmission
+ // of RTCP BYE
+ if (_rtpRtcpModule.SetSendingStatus(false) == -1 ||
+ _rtpRtcpModule.ResetSendDataCountersRTP() == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
+ "StartSend() RTP/RTCP failed to stop sending");
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StartReceiving()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartReceiving()");
+ if (_receiving)
+ {
+ return 0;
+ }
+ // If external transport is used, we will only initialize/set the variables
+ // after this section, since we are not using the WebRtc transport but
+ // still need to keep track of e.g. if we are receiving.
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_externalTransport)
+ {
+ if (!_socketTransportModule.ReceiveSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKETS_NOT_INITED, kTraceError,
+ "StartReceive() must set local receiver first");
+ return -1;
+ }
+ if (_socketTransportModule.StartReceiving(KNumberOfSocketBuffers) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "StartReceiving() failed to start receiving");
+ return -1;
+ }
+ }
+#endif
+ _receiving = true;
+ _numberOfDiscardedPackets = 0;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::StopReceiving()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StopReceiving()");
+ if (!_receiving)
+ {
+ return 0;
+ }
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_externalTransport &&
+ _socketTransportModule.ReceiveSocketsInitialized())
+ {
+ if (_socketTransportModule.StopReceiving() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "StopReceiving() failed to stop receiving.");
+ return -1;
+ }
+ }
+#endif
+ bool dtmfDetection = _rtpRtcpModule.TelephoneEvent();
+ WebRtc_Word32 ret = _rtpRtcpModule.InitReceiver();
+ if (ret != 0) {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "StopReceiving() failed to reinitialize the RTP receiver.");
+ return -1;
+ }
+ // Recover DTMF detection status.
+ ret = _rtpRtcpModule.SetTelephoneEventStatus(dtmfDetection, true, true);
+ if (ret != 0) {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "StopReceiving() failed to restore telephone-event status.");
+ }
+ RegisterReceiveCodecsToRTPModule();
+ _receiving = false;
+ return 0;
+}
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32
+Channel::SetLocalReceiver(const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort,
+ const WebRtc_Word8 ipAddr[64],
+ const WebRtc_Word8 multicastIpAddr[64])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetLocalReceiver()");
+
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetLocalReceiver() conflict with external transport");
+ return -1;
+ }
+
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_SENDING, kTraceError,
+ "SetLocalReceiver() already sending");
+ return -1;
+ }
+ if (_receiving)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_LISTENING, kTraceError,
+ "SetLocalReceiver() already receiving");
+ return -1;
+ }
+
+ if (_socketTransportModule.InitializeReceiveSockets(this,
+ rtpPort,
+ ipAddr,
+ multicastIpAddr,
+ rtcpPort) != 0)
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kIpAddressInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "SetLocalReceiver() invalid IP address");
+ break;
+ case UdpTransport::kSocketInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetLocalReceiver() invalid socket");
+ break;
+ case UdpTransport::kPortInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_PORT_NMBR, kTraceError,
+ "SetLocalReceiver() invalid port");
+ break;
+ case UdpTransport::kFailedToBindPort:
+ _engineStatisticsPtr->SetLastError(
+ VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED, kTraceError,
+ "SetLocalReceiver() binding failed");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetLocalReceiver() undefined socket error");
+ break;
+ }
+ return -1;
+ }
+ return 0;
+}
+#endif
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32
+Channel::GetLocalReceiver(int& port, int& RTCPport, char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetLocalReceiver()");
+
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetLocalReceiver() conflict with external transport");
+ return -1;
+ }
+
+ WebRtc_Word8 ipAddrTmp[UdpTransport::
+ kIpAddressVersion6Length] = {0};
+ WebRtc_UWord16 rtpPort(0);
+ WebRtc_UWord16 rtcpPort(0);
+ WebRtc_Word8 multicastIpAddr[UdpTransport::
+ kIpAddressVersion6Length] = {0};
+
+ // Acquire socket information from the socket module
+ if (_socketTransportModule.ReceiveSocketInformation(ipAddrTmp,
+ rtpPort,
+ rtcpPort,
+ multicastIpAddr) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_GET_SOCKET_INFO, kTraceError,
+ "GetLocalReceiver() unable to retrieve socket information");
+ return -1;
+ }
+
+ // Deliver valid results to the user
+ port = static_cast<int> (rtpPort);
+ RTCPport = static_cast<int> (rtcpPort);
+ if (ipAddr != NULL)
+ {
+ strcpy(ipAddr, ipAddrTmp);
+ }
+ return 0;
+}
+#endif
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32
+Channel::SetSendDestination(const WebRtc_UWord16 rtpPort,
+ const WebRtc_Word8 ipAddr[64],
+ const int sourcePort,
+ const WebRtc_UWord16 rtcpPort)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendDestination()");
+
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetSendDestination() conflict with external transport");
+ return -1;
+ }
+
+ // Initialize ports and IP address for the remote (destination) side.
+ // By default, the sockets used for receiving are used for transmission as
+ // well, hence the source ports for outgoing packets are the same as the
+ // receiving ports specified in SetLocalReceiver.
+ // If an extra send socket has been created, it will be utilized until a
+ // new source port is specified or until the channel has been deleted and
+ // recreated. If no socket exists, sockets will be created when the first
+ // RTP and RTCP packets shall be transmitted (see e.g.
+ // UdpTransportImpl::SendPacket()).
+ //
+ // NOTE: this function does not require that sockets exists; all it does is
+ // to build send structures to be used with the sockets when they exist.
+ // It is therefore possible to call this method before SetLocalReceiver.
+ // However, sockets must exist if a multi-cast address is given as input.
+
+ // Build send structures and enable QoS (if enabled and supported)
+ if (_socketTransportModule.InitializeSendSockets(
+ ipAddr, rtpPort, rtcpPort) != UdpTransport::kNoSocketError)
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kIpAddressInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "SetSendDestination() invalid IP address 1");
+ break;
+ case UdpTransport::kSocketInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetSendDestination() invalid socket 1");
+ break;
+ case UdpTransport::kQosError:
+ _engineStatisticsPtr->SetLastError(
+ VE_GQOS_ERROR, kTraceError,
+ "SetSendDestination() failed to set QoS");
+ break;
+ case UdpTransport::kMulticastAddressInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_MULTICAST_ADDRESS, kTraceError,
+ "SetSendDestination() invalid multicast address");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetSendDestination() undefined socket error 1");
+ break;
+ }
+ return -1;
+ }
+
+ // Check if the user has specified a non-default source port different from
+ // the local receive port.
+ // If so, an extra local socket will be created unless the source port is
+ // not unique.
+ if (sourcePort != kVoEDefault)
+ {
+ WebRtc_UWord16 receiverRtpPort(0);
+ WebRtc_UWord16 rtcpNA(0);
+ if (_socketTransportModule.ReceiveSocketInformation(NULL,
+ receiverRtpPort,
+ rtcpNA,
+ NULL) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_GET_SOCKET_INFO, kTraceError,
+ "SetSendDestination() failed to retrieve socket information");
+ return -1;
+ }
+
+ WebRtc_UWord16 sourcePortUW16 =
+ static_cast<WebRtc_UWord16> (sourcePort);
+
+ // An extra socket will only be created if the specified source port
+ // differs from the local receive port.
+ if (sourcePortUW16 != receiverRtpPort)
+ {
+ // Initialize extra local socket to get a different source port
+ // than the local
+ // receiver port. Always use default source for RTCP.
+ // Note that, this calls UdpTransport::CloseSendSockets().
+ if (_socketTransportModule.InitializeSourcePorts(
+ sourcePortUW16,
+ sourcePortUW16+1) != 0)
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kIpAddressInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "SetSendDestination() invalid IP address 2");
+ break;
+ case UdpTransport::kSocketInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetSendDestination() invalid socket 2");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetSendDestination() undefined socket error 2");
+ break;
+ }
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "SetSendDestination() extra local socket is created"
+ " to facilitate unique source port");
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "SetSendDestination() sourcePort equals the local"
+ " receive port => no extra socket is created");
+ }
+ }
+
+ return 0;
+}
+#endif
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32
+Channel::GetSendDestination(int& port,
+ char ipAddr[64],
+ int& sourcePort,
+ int& RTCPport)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSendDestination()");
+
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "GetSendDestination() conflict with external transport");
+ return -1;
+ }
+
+ WebRtc_Word8 ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
+ WebRtc_UWord16 rtpPort(0);
+ WebRtc_UWord16 rtcpPort(0);
+ WebRtc_UWord16 rtpSourcePort(0);
+ WebRtc_UWord16 rtcpSourcePort(0);
+
+ // Acquire sending socket information from the socket module
+ _socketTransportModule.SendSocketInformation(ipAddrTmp, rtpPort, rtcpPort);
+ _socketTransportModule.SourcePorts(rtpSourcePort, rtcpSourcePort);
+
+ // Deliver valid results to the user
+ port = static_cast<int> (rtpPort);
+ RTCPport = static_cast<int> (rtcpPort);
+ sourcePort = static_cast<int> (rtpSourcePort);
+ if (ipAddr != NULL)
+ {
+ strcpy(ipAddr, ipAddrTmp);
+ }
+
+ return 0;
+}
+#endif
+
+
+WebRtc_Word32
+Channel::SetNetEQPlayoutMode(NetEqModes mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetNetEQPlayoutMode()");
+ AudioPlayoutMode playoutMode(voice);
+ switch (mode)
+ {
+ case kNetEqDefault:
+ playoutMode = voice;
+ break;
+ case kNetEqStreaming:
+ playoutMode = streaming;
+ break;
+ case kNetEqFax:
+ playoutMode = fax;
+ break;
+ }
+ if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetNetEQPlayoutMode() failed to set playout mode");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetNetEQPlayoutMode(NetEqModes& mode)
+{
+ const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode();
+ switch (playoutMode)
+ {
+ case voice:
+ mode = kNetEqDefault;
+ break;
+ case streaming:
+ mode = kNetEqStreaming;
+ break;
+ case fax:
+ mode = kNetEqFax;
+ break;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::GetNetEQPlayoutMode() => mode=%u", mode);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetNetEQBGNMode(NetEqBgnModes mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetNetEQPlayoutMode()");
+ ACMBackgroundNoiseMode noiseMode(On);
+ switch (mode)
+ {
+ case kBgnOn:
+ noiseMode = On;
+ break;
+ case kBgnFade:
+ noiseMode = Fade;
+ break;
+ case kBgnOff:
+ noiseMode = Off;
+ break;
+ }
+ if (_audioCodingModule.SetBackgroundNoiseMode(noiseMode) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetBackgroundNoiseMode() failed to set noise mode");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetOnHoldStatus(bool enable, OnHoldModes mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetOnHoldStatus()");
+ if (mode == kHoldSendAndPlay)
+ {
+ _outputIsOnHold = enable;
+ _inputIsOnHold = enable;
+ }
+ else if (mode == kHoldPlayOnly)
+ {
+ _outputIsOnHold = enable;
+ }
+ if (mode == kHoldSendOnly)
+ {
+ _inputIsOnHold = enable;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetOnHoldStatus()");
+ enabled = (_outputIsOnHold || _inputIsOnHold);
+ if (_outputIsOnHold && _inputIsOnHold)
+ {
+ mode = kHoldSendAndPlay;
+ }
+ else if (_outputIsOnHold && !_inputIsOnHold)
+ {
+ mode = kHoldPlayOnly;
+ }
+ else if (!_outputIsOnHold && _inputIsOnHold)
+ {
+ mode = kHoldSendOnly;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetOnHoldStatus() => enabled=%d, mode=%d",
+ enabled, mode);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterVoiceEngineObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_voiceEngineObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterVoiceEngineObserver() observer already enabled");
+ return -1;
+ }
+ _voiceEngineObserverPtr = &observer;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::DeRegisterVoiceEngineObserver()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterVoiceEngineObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_voiceEngineObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterVoiceEngineObserver() observer already disabled");
+ return 0;
+ }
+ _voiceEngineObserverPtr = NULL;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetNetEQBGNMode(NetEqBgnModes& mode)
+{
+ ACMBackgroundNoiseMode noiseMode(On);
+ _audioCodingModule.BackgroundNoiseMode(noiseMode);
+ switch (noiseMode)
+ {
+ case On:
+ mode = kBgnOn;
+ break;
+ case Fade:
+ mode = kBgnFade;
+ break;
+ case Off:
+ mode = kBgnOff;
+ break;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetNetEQBGNMode() => mode=%u", mode);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetSendCodec(CodecInst& codec)
+{
+ return (_audioCodingModule.SendCodec(codec));
+}
+
+WebRtc_Word32
+Channel::GetRecCodec(CodecInst& codec)
+{
+ return (_audioCodingModule.ReceiveCodec(codec));
+}
+
+WebRtc_Word32
+Channel::SetSendCodec(const CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendCodec()");
+
+ if (_audioCodingModule.RegisterSendCodec(codec) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "SetSendCodec() failed to register codec to ACM");
+ return -1;
+ }
+
+ if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
+ {
+ _rtpRtcpModule.DeRegisterSendPayload(codec.pltype);
+ if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "SetSendCodec() failed to register codec to"
+ " RTP/RTCP module");
+ return -1;
+ }
+ }
+
+ if (_rtpRtcpModule.SetAudioPacketSize(codec.pacsize) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "SetSendCodec() failed to set audio packet size");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetVADStatus(mode=%d)", mode);
+ // To disable VAD, DTX must be disabled too
+ disableDTX = ((enableVAD == false) ? true : disableDTX);
+ if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetVADStatus() failed to set VAD");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetVADStatus");
+ if (_audioCodingModule.VAD(disabledDTX, enabledVAD, mode) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "GetVADStatus() failed to get VAD status");
+ return -1;
+ }
+ disabledDTX = !disabledDTX;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetRecPayloadType(const CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetRecPayloadType()");
+
+ if (_playing)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceError,
+ "SetRecPayloadType() unable to set PT while playing");
+ return -1;
+ }
+ if (_receiving)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_LISTENING, kTraceError,
+ "SetRecPayloadType() unable to set PT while listening");
+ return -1;
+ }
+
+ if (codec.pltype == -1)
+ {
+ // De-register the selected codec (RTP/RTCP module and ACM)
+
+ WebRtc_Word8 pltype(-1);
+ CodecInst rxCodec = codec;
+
+ // Get payload type for the given codec
+ _rtpRtcpModule.ReceivePayloadType(rxCodec, &pltype);
+ rxCodec.pltype = pltype;
+
+ if (_rtpRtcpModule.DeRegisterReceivePayload(pltype) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR,
+ kTraceError,
+ "SetRecPayloadType() RTP/RTCP-module deregistration "
+ "failed");
+ return -1;
+ }
+ if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetRecPayloadType() ACM deregistration failed - 1");
+ return -1;
+ }
+ return 0;
+ }
+
+ if (_rtpRtcpModule.RegisterReceivePayload(codec) != 0)
+ {
+ // First attempt to register failed => de-register and try again
+ _rtpRtcpModule.DeRegisterReceivePayload(codec.pltype);
+ if (_rtpRtcpModule.RegisterReceivePayload(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetRecPayloadType() RTP/RTCP-module registration failed");
+ return -1;
+ }
+ }
+ if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
+ {
+ _audioCodingModule.UnregisterReceiveCodec(codec.pltype);
+ if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetRecPayloadType() ACM registration failed - 1");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetRecPayloadType(CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRecPayloadType()");
+ WebRtc_Word8 payloadType(-1);
+ if (_rtpRtcpModule.ReceivePayloadType(codec, &payloadType) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "GetRecPayloadType() failed to retrieve RX payload type");
+ return -1;
+ }
+ codec.pltype = payloadType;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRecPayloadType() => pltype=%u", codec.pltype);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetAMREncFormat(AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetAMREncFormat()");
+
+ // ACM doesn't support AMR
+ return -1;
+}
+
+WebRtc_Word32
+Channel::SetAMRDecFormat(AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetAMRDecFormat()");
+
+ // ACM doesn't support AMR
+ return -1;
+}
+
+WebRtc_Word32
+Channel::SetAMRWbEncFormat(AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetAMRWbEncFormat()");
+
+ // ACM doesn't support AMR
+ return -1;
+
+}
+
+WebRtc_Word32
+Channel::SetAMRWbDecFormat(AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetAMRWbDecFormat()");
+
+ // ACM doesn't support AMR
+ return -1;
+}
+
+WebRtc_Word32
+Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendCNPayloadType()");
+
+ CodecInst codec;
+ WebRtc_Word32 samplingFreqHz(-1);
+ if (frequency == kFreq32000Hz)
+ samplingFreqHz = 32000;
+ else if (frequency == kFreq16000Hz)
+ samplingFreqHz = 16000;
+
+ if (_audioCodingModule.Codec("CN", codec, samplingFreqHz) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetSendCNPayloadType() failed to retrieve default CN codec "
+ "settings");
+ return -1;
+ }
+
+ // Modify the payload type (must be set to dynamic range)
+ codec.pltype = type;
+
+ if (_audioCodingModule.RegisterSendCodec(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetSendCNPayloadType() failed to register CN to ACM");
+ return -1;
+ }
+
+ if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
+ {
+ _rtpRtcpModule.DeRegisterSendPayload(codec.pltype);
+ if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetSendCNPayloadType() failed to register CN to RTP/RTCP "
+ "module");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetISACInitTargetRate()");
+
+ CodecInst sendCodec;
+ if (_audioCodingModule.SendCodec(sendCodec) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACInitTargetRate() failed to retrieve send codec");
+ return -1;
+ }
+ if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
+ {
+ // This API is only valid if iSAC is setup to run in channel-adaptive
+ // mode.
+ // We do not validate the adaptive mode here. It is done later in the
+ // ConfigISACBandwidthEstimator() API.
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACInitTargetRate() send codec is not iSAC");
+ return -1;
+ }
+
+ WebRtc_UWord8 initFrameSizeMsec(0);
+ if (16000 == sendCodec.plfreq)
+ {
+ // Note that 0 is a valid and corresponds to "use default
+ if ((rateBps != 0 &&
+ rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) ||
+ (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACInitTargetRate() invalid target rate - 1");
+ return -1;
+ }
+ // 30 or 60ms
+ initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 16);
+ }
+ else if (32000 == sendCodec.plfreq)
+ {
+ if ((rateBps != 0 &&
+ rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) ||
+ (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACInitTargetRate() invalid target rate - 2");
+ return -1;
+ }
+ initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 32); // 30ms
+ }
+
+ if (_audioCodingModule.ConfigISACBandwidthEstimator(
+ initFrameSizeMsec, rateBps, useFixedFrameSize) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetISACInitTargetRate() iSAC BWE config failed");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetISACMaxRate(int rateBps)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetISACMaxRate()");
+
+ CodecInst sendCodec;
+ if (_audioCodingModule.SendCodec(sendCodec) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACMaxRate() failed to retrieve send codec");
+ return -1;
+ }
+ if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
+ {
+ // This API is only valid if iSAC is selected as sending codec.
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACMaxRate() send codec is not iSAC");
+ return -1;
+ }
+ if (16000 == sendCodec.plfreq)
+ {
+ if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) ||
+ (rateBps > kVoiceEngineMaxIsacMaxRateBpsWb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACMaxRate() invalid max rate - 1");
+ return -1;
+ }
+ }
+ else if (32000 == sendCodec.plfreq)
+ {
+ if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) ||
+ (rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACMaxRate() invalid max rate - 2");
+ return -1;
+ }
+ }
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SENDING, kTraceError,
+ "SetISACMaxRate() unable to set max rate while sending");
+ return -1;
+ }
+
+ // Set the maximum instantaneous rate of iSAC (works for both adaptive
+ // and non-adaptive mode)
+ if (_audioCodingModule.SetISACMaxRate(rateBps) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetISACMaxRate() failed to set max rate");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetISACMaxPayloadSize(int sizeBytes)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetISACMaxPayloadSize()");
+ CodecInst sendCodec;
+ if (_audioCodingModule.SendCodec(sendCodec) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACMaxPayloadSize() failed to retrieve send codec");
+ return -1;
+ }
+ if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetISACMaxPayloadSize() send codec is not iSAC");
+ return -1;
+ }
+ if (16000 == sendCodec.plfreq)
+ {
+ if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) ||
+ (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACMaxPayloadSize() invalid max payload - 1");
+ return -1;
+ }
+ }
+ else if (32000 == sendCodec.plfreq)
+ {
+ if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) ||
+ (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetISACMaxPayloadSize() invalid max payload - 2");
+ return -1;
+ }
+ }
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SENDING, kTraceError,
+ "SetISACMaxPayloadSize() unable to set max rate while sending");
+ return -1;
+ }
+
+ if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetISACMaxPayloadSize() failed to set max payload size");
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 Channel::RegisterExternalTransport(Transport& transport)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::RegisterExternalTransport()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ // Sanity checks for default (non external transport) to avoid conflict with
+ // WebRtc sockets.
+ if (_socketTransportModule.SendSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(VE_SEND_SOCKETS_CONFLICT,
+ kTraceError,
+ "RegisterExternalTransport() send sockets already initialized");
+ return -1;
+ }
+ if (_socketTransportModule.ReceiveSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(VE_RECEIVE_SOCKETS_CONFLICT,
+ kTraceError,
+ "RegisterExternalTransport() receive sockets already initialized");
+ return -1;
+ }
+#endif
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION,
+ kTraceError,
+ "RegisterExternalTransport() external transport already enabled");
+ return -1;
+ }
+ _externalTransport = true;
+ _transportPtr = &transport;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::DeRegisterExternalTransport()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterExternalTransport()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_transportPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterExternalTransport() external transport already "
+ "disabled");
+ return 0;
+ }
+ _externalTransport = false;
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ _transportPtr = NULL;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "DeRegisterExternalTransport() all transport is disabled");
+#else
+ _transportPtr = &_socketTransportModule;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "DeRegisterExternalTransport() internal Transport is enabled");
+#endif
+ return 0;
+}
+
+WebRtc_Word32
+Channel::ReceivedRTPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ReceivedRTPPacket()");
+ const WebRtc_Word8 dummyIP[] = "127.0.0.1";
+ IncomingRTPPacket(data, length, dummyIP, 0);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::ReceivedRTCPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ReceivedRTCPPacket()");
+ const WebRtc_Word8 dummyIP[] = "127.0.0.1";
+ IncomingRTCPPacket(data, length, dummyIP, 0);
+ return 0;
+}
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32
+Channel::GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSourceInfo()");
+
+ WebRtc_UWord16 rtpPortModule;
+ WebRtc_UWord16 rtcpPortModule;
+ WebRtc_Word8 ipaddr[UdpTransport::kIpAddressVersion6Length] = {0};
+
+ if (_socketTransportModule.RemoteSocketInformation(ipaddr,
+ rtpPortModule,
+ rtcpPortModule) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "GetSourceInfo() failed to retrieve remote socket information");
+ return -1;
+ }
+ strcpy(ipAddr, ipaddr);
+ rtpPort = rtpPortModule;
+ rtcpPort = rtcpPortModule;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "GetSourceInfo() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
+ rtpPort, rtcpPort, ipAddr);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::EnableIPv6()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EnableIPv6()");
+ if (_socketTransportModule.ReceiveSocketsInitialized() ||
+ _socketTransportModule.SendSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "EnableIPv6() socket layer is already initialized");
+ return -1;
+ }
+ if (_socketTransportModule.EnableIpV6() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "EnableIPv6() failed to enable IPv6");
+ const UdpTransport::ErrorCode lastError =
+ _socketTransportModule.LastError();
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "UdpTransport::LastError() => %d", lastError);
+ return -1;
+ }
+ return 0;
+}
+
+bool
+Channel::IPv6IsEnabled() const
+{
+ bool isEnabled = _socketTransportModule.IpV6Enabled();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "IPv6IsEnabled() => %d", isEnabled);
+ return isEnabled;
+}
+
+WebRtc_Word32
+Channel::SetSourceFilter(int rtpPort, int rtcpPort, const char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSourceFilter()");
+ if (_socketTransportModule.SetFilterPorts(
+ static_cast<WebRtc_UWord16>(rtpPort),
+ static_cast<WebRtc_UWord16>(rtcpPort)) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "SetSourceFilter() failed to set filter ports");
+ const UdpTransport::ErrorCode lastError =
+ _socketTransportModule.LastError();
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "UdpTransport::LastError() => %d",
+ lastError);
+ return -1;
+ }
+ const WebRtc_Word8* filterIpAddress =
+ static_cast<const WebRtc_Word8*> (ipAddr);
+ if (_socketTransportModule.SetFilterIP(filterIpAddress) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "SetSourceFilter() failed to set filter IP address");
+ const UdpTransport::ErrorCode lastError =
+ _socketTransportModule.LastError();
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "UdpTransport::LastError() => %d", lastError);
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSourceFilter()");
+ WebRtc_UWord16 rtpFilterPort(0);
+ WebRtc_UWord16 rtcpFilterPort(0);
+ if (_socketTransportModule.FilterPorts(rtpFilterPort, rtcpFilterPort) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
+ "GetSourceFilter() failed to retrieve filter ports");
+ }
+ WebRtc_Word8 ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
+ if (_socketTransportModule.FilterIP(ipAddrTmp) != 0)
+ {
+ // no filter has been configured (not seen as an error)
+ memset(ipAddrTmp,
+ 0, UdpTransport::kIpAddressVersion6Length);
+ }
+ rtpPort = static_cast<int> (rtpFilterPort);
+ rtcpPort = static_cast<int> (rtcpFilterPort);
+ strcpy(ipAddr, ipAddrTmp);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "GetSourceFilter() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
+ rtpPort, rtcpPort, ipAddr);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetSendTOS(int DSCP, int priority, bool useSetSockopt)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendTOS(DSCP=%d, useSetSockopt=%d)",
+ DSCP, (int)useSetSockopt);
+
+ // Set TOS value and possibly try to force usage of setsockopt()
+ if (_socketTransportModule.SetToS(DSCP, useSetSockopt) != 0)
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kTosError:
+ _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
+ "SetSendTOS() TOS error");
+ break;
+ case UdpTransport::kQosError:
+ _engineStatisticsPtr->SetLastError(
+ VE_TOS_GQOS_CONFLICT, kTraceError,
+ "SetSendTOS() GQOS error");
+ break;
+ case UdpTransport::kTosInvalid:
+ // can't switch SetSockOpt method without disabling TOS first, or
+ // SetSockopt() call failed
+ _engineStatisticsPtr->SetLastError(VE_TOS_INVALID, kTraceError,
+ "SetSendTOS() invalid TOS");
+ break;
+ case UdpTransport::kSocketInvalid:
+ _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
+ "SetSendTOS() invalid Socket");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
+ "SetSendTOS() TOS error");
+ break;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "UdpTransport => lastError = %d",
+ lastSockError);
+ return -1;
+ }
+
+ // Set priority (PCP) value, -1 means don't change
+ if (-1 != priority)
+ {
+ if (_socketTransportModule.SetPCP(priority) != 0)
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kPcpError:
+ _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
+ "SetSendTOS() PCP error");
+ break;
+ case UdpTransport::kQosError:
+ _engineStatisticsPtr->SetLastError(
+ VE_TOS_GQOS_CONFLICT, kTraceError,
+ "SetSendTOS() GQOS conflict");
+ break;
+ case UdpTransport::kSocketInvalid:
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_ERROR, kTraceError,
+ "SetSendTOS() invalid Socket");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
+ "SetSendTOS() PCP error");
+ break;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "UdpTransport => lastError = %d",
+ lastSockError);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSendTOS(DSCP=?, useSetSockopt=?)");
+ WebRtc_Word32 dscp(0), prio(0);
+ bool setSockopt(false);
+ if (_socketTransportModule.ToS(dscp, setSockopt) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "GetSendTOS() failed to get TOS info");
+ return -1;
+ }
+ if (_socketTransportModule.PCP(prio) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "GetSendTOS() failed to get PCP info");
+ return -1;
+ }
+ DSCP = static_cast<int> (dscp);
+ priority = static_cast<int> (prio);
+ useSetSockopt = setSockopt;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSendTOS() => DSCP=%d, priority=%d, useSetSockopt=%d",
+ DSCP, priority, (int)useSetSockopt);
+ return 0;
+}
+
+#if defined(_WIN32)
+WebRtc_Word32
+Channel::SetSendGQoS(bool enable, int serviceType, int overrideDSCP)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendGQoS(enable=%d, serviceType=%d, "
+ "overrideDSCP=%d)",
+ (int)enable, serviceType, overrideDSCP);
+ if(!_socketTransportModule.ReceiveSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKETS_NOT_INITED, kTraceError,
+ "SetSendGQoS() GQoS state must be set after sockets are created");
+ return -1;
+ }
+ if(!_socketTransportModule.SendSocketsInitialized())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_DESTINATION_NOT_INITED, kTraceError,
+ "SetSendGQoS() GQoS state must be set after sending side is "
+ "initialized");
+ return -1;
+ }
+ if (enable &&
+ (serviceType != SERVICETYPE_BESTEFFORT) &&
+ (serviceType != SERVICETYPE_CONTROLLEDLOAD) &&
+ (serviceType != SERVICETYPE_GUARANTEED) &&
+ (serviceType != SERVICETYPE_QUALITATIVE))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendGQoS() Invalid service type");
+ return -1;
+ }
+ if (enable && ((overrideDSCP < 0) || (overrideDSCP > 63)))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendGQoS() Invalid overrideDSCP value");
+ return -1;
+ }
+
+ // Avoid GQoS/ToS conflict when user wants to override the default DSCP
+ // mapping
+ bool QoS(false);
+ WebRtc_Word32 sType(0);
+ WebRtc_Word32 ovrDSCP(0);
+ if (_socketTransportModule.QoS(QoS, sType, ovrDSCP))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "SetSendGQoS() failed to get QOS info");
+ return -1;
+ }
+ if (QoS && ovrDSCP == 0 && overrideDSCP != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_TOS_GQOS_CONFLICT, kTraceError,
+ "SetSendGQoS() QOS is already enabled and overrideDSCP differs,"
+ " not allowed");
+ return -1;
+ }
+ const WebRtc_Word32 maxBitrate(0);
+ if (_socketTransportModule.SetQoS(enable,
+ static_cast<WebRtc_Word32>(serviceType),
+ maxBitrate,
+ static_cast<WebRtc_Word32>(overrideDSCP),
+ true))
+ {
+ UdpTransport::ErrorCode lastSockError(
+ _socketTransportModule.LastError());
+ switch (lastSockError)
+ {
+ case UdpTransport::kQosError:
+ _engineStatisticsPtr->SetLastError(VE_GQOS_ERROR, kTraceError,
+ "SetSendGQoS() QOS error");
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
+ "SetSendGQoS() Socket error");
+ break;
+ }
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "UdpTransport() => lastError = %d",
+ lastSockError);
+ return -1;
+ }
+ return 0;
+}
+#endif
+
+#if defined(_WIN32)
+WebRtc_Word32
+Channel::GetSendGQoS(bool &enabled, int &serviceType, int &overrideDSCP)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSendGQoS(enable=?, serviceType=?, "
+ "overrideDSCP=?)");
+
+ bool QoS(false);
+ WebRtc_Word32 serviceTypeModule(0);
+ WebRtc_Word32 overrideDSCPModule(0);
+ _socketTransportModule.QoS(QoS, serviceTypeModule, overrideDSCPModule);
+
+ enabled = QoS;
+ serviceType = static_cast<int> (serviceTypeModule);
+ overrideDSCP = static_cast<int> (overrideDSCPModule);
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "GetSendGQoS() => enabled=%d, serviceType=%d, overrideDSCP=%d",
+ (int)enabled, serviceType, overrideDSCP);
+ return 0;
+}
+#endif
+#endif
+
+WebRtc_Word32
+Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetPacketTimeoutNotification()");
+ if (enable)
+ {
+ const WebRtc_UWord32 RTPtimeoutMS = 1000*timeoutSeconds;
+ const WebRtc_UWord32 RTCPtimeoutMS = 0;
+ _rtpRtcpModule.SetPacketTimeout(RTPtimeoutMS, RTCPtimeoutMS);
+ _rtpPacketTimeOutIsEnabled = true;
+ _rtpTimeOutSeconds = timeoutSeconds;
+ }
+ else
+ {
+ _rtpRtcpModule.SetPacketTimeout(0, 0);
+ _rtpPacketTimeOutIsEnabled = false;
+ _rtpTimeOutSeconds = 0;
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetPacketTimeoutNotification()");
+ enabled = _rtpPacketTimeOutIsEnabled;
+ if (enabled)
+ {
+ timeoutSeconds = _rtpTimeOutSeconds;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetPacketTimeoutNotification() => enabled=%d,"
+ " timeoutSeconds=%d",
+ enabled, timeoutSeconds);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterDeadOrAliveObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_connectionObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, kTraceError,
+ "RegisterDeadOrAliveObserver() observer already enabled");
+ return -1;
+ }
+
+ _connectionObserverPtr = &observer;
+ _connectionObserver = true;
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::DeRegisterDeadOrAliveObserver()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterDeadOrAliveObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_connectionObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterDeadOrAliveObserver() observer already disabled");
+ return 0;
+ }
+
+ _connectionObserver = false;
+ _connectionObserverPtr = NULL;
+
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetPeriodicDeadOrAliveStatus()");
+ if (!_connectionObserverPtr)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "SetPeriodicDeadOrAliveStatus() connection observer has"
+ " not been registered");
+ }
+ if (enable)
+ {
+ ResetDeadOrAliveCounters();
+ }
+ bool enabled(false);
+ WebRtc_UWord8 currentSampleTimeSec(0);
+ // Store last state (will be used later if dead-or-alive is disabled).
+ _rtpRtcpModule.PeriodicDeadOrAliveStatus(enabled, currentSampleTimeSec);
+ // Update the dead-or-alive state.
+ if (_rtpRtcpModule.SetPeriodicDeadOrAliveStatus(
+ enable, (WebRtc_UWord8)sampleTimeSeconds) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR,
+ kTraceError,
+ "SetPeriodicDeadOrAliveStatus() failed to set dead-or-alive "
+ "status");
+ return -1;
+ }
+ if (!enable)
+ {
+ // Restore last utilized sample time.
+ // Without this, the sample time would always be reset to default
+ // (2 sec), each time dead-or-alived was disabled without sample-time
+ // parameter.
+ _rtpRtcpModule.SetPeriodicDeadOrAliveStatus(enable,
+ currentSampleTimeSec);
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds)
+{
+ _rtpRtcpModule.PeriodicDeadOrAliveStatus(
+ enabled,
+ (WebRtc_UWord8&)sampleTimeSeconds);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetPeriodicDeadOrAliveStatus() => enabled=%d,"
+ " sampleTimeSeconds=%d",
+ enabled, sampleTimeSeconds);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SendUDPPacket(const void* data,
+ unsigned int length,
+ int& transmittedBytes,
+ bool useRtcpSocket)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SendUDPPacket()");
+ if (_externalTransport)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SendUDPPacket() external transport is enabled");
+ return -1;
+ }
+ if (useRtcpSocket && !_rtpRtcpModule.RTCP())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTCP_ERROR, kTraceError,
+ "SendUDPPacket() RTCP is disabled");
+ return -1;
+ }
+ if (!_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_NOT_SENDING, kTraceError,
+ "SendUDPPacket() not sending");
+ return -1;
+ }
+
+ char* dataC = new char[length];
+ if (NULL == dataC)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_NO_MEMORY, kTraceError,
+ "SendUDPPacket() memory allocation failed");
+ return -1;
+ }
+ memcpy(dataC, data, length);
+
+ transmittedBytes = SendPacketRaw(dataC, length, useRtcpSocket);
+
+ delete [] dataC;
+ dataC = NULL;
+
+ if (transmittedBytes <= 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SEND_ERROR, kTraceError,
+ "SendUDPPacket() transmission failed");
+ transmittedBytes = 0;
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "SendUDPPacket() => transmittedBytes=%d", transmittedBytes);
+ return 0;
+}
+
+
+int Channel::StartPlayingFileLocally(const char* fileName,
+ const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
+ " format=%d, volumeScaling=%5.3f, startPosition=%d, "
+ "stopPosition=%d)", fileName, loop, format, volumeScaling,
+ startPosition, stopPosition);
+
+ if (_outputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceError,
+ "StartPlayingFileLocally() is already playing");
+ return -1;
+ }
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFilePlayerPtr)
+ {
+ _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ }
+
+ _outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
+ _outputFilePlayerId, (const FileFormats)format);
+
+ if (_outputFilePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingFileLocally() filePlayer format is not correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_outputFilePlayerPtr->StartPlayingFile(
+ fileName,
+ loop,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*)codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _outputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ return -1;
+ }
+ _outputFilePlayerPtr->RegisterModuleFileCallback(this);
+ _outputFilePlaying = true;
+ }
+ // _fileCritSect cannot be taken while calling
+ // SetAnonymousMixabilityStatus() since as soon as the participant is added
+ // frames can be pulled by the mixer. Since the frames are generated from
+ // the file, _fileCritSect will be taken. This would result in a deadlock.
+ if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ _outputFilePlaying = false;
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
+ "StartPlayingFile() failed to add participant as file to mixer");
+ _outputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+int Channel::StartPlayingFileLocally(InStream* stream,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartPlayingFileLocally(format=%d,"
+ " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
+ format, volumeScaling, startPosition, stopPosition);
+
+ if(stream == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFileLocally() NULL as input stream");
+ return -1;
+ }
+
+
+ if (_outputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceError,
+ "StartPlayingFileLocally() is already playing");
+ return -1;
+ }
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_outputFilePlayerPtr)
+ {
+ _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ }
+
+ // Create the instance
+ _outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
+ _outputFilePlayerId,
+ (const FileFormats)format);
+
+ if (_outputFilePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingFileLocally() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition, codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to "
+ "start file playout");
+ _outputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ return -1;
+ }
+ _outputFilePlayerPtr->RegisterModuleFileCallback(this);
+ _outputFilePlaying = true;
+ }
+ // _fileCritSect cannot be taken while calling
+ // SetAnonymousMixibilityStatus. Refer to comments in
+ // StartPlayingFileLocally(const char* ...) for more details.
+ if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ _outputFilePlaying = false;
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
+ "StartPlayingFile() failed to add participant as file to mixer");
+ _outputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+int Channel::StopPlayingFileLocally()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StopPlayingFileLocally()");
+
+ if (!_outputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "StopPlayingFileLocally() isnot playing");
+ return 0;
+ }
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFilePlayerPtr->StopPlayingFile() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopPlayingFile() could not stop playing");
+ return -1;
+ }
+ _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
+ _outputFilePlayerPtr = NULL;
+ _outputFilePlaying = false;
+ }
+ // _fileCritSect cannot be taken while calling
+ // SetAnonymousMixibilityStatus. Refer to comments in
+ // StartPlayingFileLocally(const char* ...) for more details.
+ if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
+ "StopPlayingFile() failed to stop participant from playing as"
+ "file in the mixer");
+ return -1;
+ }
+
+ return 0;
+}
+
+int Channel::IsPlayingFileLocally() const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::IsPlayingFileLocally()");
+
+ return (WebRtc_Word32)_outputFilePlaying;
+}
+
+int Channel::ScaleLocalFilePlayout(const float scale)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale);
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (!_outputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ScaleLocalFilePlayout() isnot playing");
+ return -1;
+ }
+ if ((_outputFilePlayerPtr == NULL) ||
+ (_outputFilePlayerPtr->SetAudioScaling(scale) != 0))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "SetAudioScaling() failed to scale the playout");
+ return -1;
+ }
+
+ return 0;
+}
+
+int Channel::GetLocalPlayoutPosition(int& positionMs)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetLocalPlayoutPosition(position=?)");
+
+ WebRtc_UWord32 position;
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFilePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "GetLocalPlayoutPosition() filePlayer instance doesnot exist");
+ return -1;
+ }
+
+ if (_outputFilePlayerPtr->GetPlayoutPosition(position) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "GetLocalPlayoutPosition() failed");
+ return -1;
+ }
+ positionMs = position;
+
+ return 0;
+}
+
+int Channel::StartPlayingFileAsMicrophone(const char* fileName,
+ const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
+ "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
+ "stopPosition=%d)", fileName, loop, format, volumeScaling,
+ startPosition, stopPosition);
+
+ if (_inputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() filePlayer is playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_inputFilePlayerPtr)
+ {
+ _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ }
+
+ // Create the instance
+ _inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
+ _inputFilePlayerId, (const FileFormats)format);
+
+ if (_inputFilePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_inputFilePlayerPtr->StartPlayingFile(
+ fileName,
+ loop,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*)codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _inputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ return -1;
+ }
+ _inputFilePlayerPtr->RegisterModuleFileCallback(this);
+ _inputFilePlaying = true;
+
+ return 0;
+}
+
+int Channel::StartPlayingFileAsMicrophone(InStream* stream,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartPlayingFileAsMicrophone(format=%d, "
+ "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
+ format, volumeScaling, startPosition, stopPosition);
+
+ if(stream == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFileAsMicrophone NULL as input stream");
+ return -1;
+ }
+
+ if (_inputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() is playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_inputFilePlayerPtr)
+ {
+ _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ }
+
+ // Create the instance
+ _inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
+ _inputFilePlayerId, (const FileFormats)format);
+
+ if (_inputFilePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingInputFile() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
+ volumeScaling, notificationTime,
+ stopPosition, codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start "
+ "file playout");
+ _inputFilePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ return -1;
+ }
+
+ _inputFilePlayerPtr->RegisterModuleFileCallback(this);
+ _inputFilePlaying = true;
+
+ return 0;
+}
+
+int Channel::StopPlayingFileAsMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StopPlayingFileAsMicrophone()");
+
+ if (!_inputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "StopPlayingFileAsMicrophone() isnot playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+ if (_inputFilePlayerPtr->StopPlayingFile() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopPlayingFile() could not stop playing");
+ return -1;
+ }
+ _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
+ _inputFilePlayerPtr = NULL;
+ _inputFilePlaying = false;
+
+ return 0;
+}
+
+int Channel::IsPlayingFileAsMicrophone() const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::IsPlayingFileAsMicrophone()");
+
+ return _inputFilePlaying;
+}
+
+int Channel::ScaleFileAsMicrophonePlayout(const float scale)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale);
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (!_inputFilePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ScaleFileAsMicrophonePlayout() isnot playing");
+ return -1;
+ }
+
+ if ((_inputFilePlayerPtr == NULL) ||
+ (_inputFilePlayerPtr->SetAudioScaling(scale) != 0))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "SetAudioScaling() failed to scale playout");
+ return -1;
+ }
+
+ return 0;
+}
+
+int Channel::StartRecordingPlayout(const WebRtc_Word8* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartRecordingPlayout(fileName=%s)", fileName);
+
+ if (_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec={100,"L16",16000,320,1,320000};
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() invalid compression");
+ return(-1);
+ }
+ if(codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst=&dummyCodec;
+ }
+ else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ }
+ else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+
+ _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
+ _outputFileRecorderId, (const FileFormats)format);
+ if (_outputFileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_outputFileRecorderPtr->StartRecordingAudioFile(
+ fileName, (const CodecInst&)*codecInst, notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ return -1;
+ }
+ _outputFileRecorderPtr->RegisterModuleFileCallback(this);
+ _outputFileRecording = true;
+
+ return 0;
+}
+
+int Channel::StartRecordingPlayout(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::StartRecordingPlayout()");
+
+ if (_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec={100,"L16",16000,320,1,320000};
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() invalid compression");
+ return(-1);
+ }
+ if(codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst=&dummyCodec;
+ }
+ else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ }
+ else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+
+ _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
+ _outputFileRecorderId, (const FileFormats)format);
+ if (_outputFileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingPlayout() failed to "
+ "start file recording");
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ return -1;
+ }
+
+ _outputFileRecorderPtr->RegisterModuleFileCallback(this);
+ _outputFileRecording = true;
+
+ return 0;
+}
+
+int Channel::StopRecordingPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "Channel::StopRecordingPlayout()");
+
+ if (!_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StopRecordingPlayout() isnot recording");
+ return -1;
+ }
+
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFileRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording() could not stop recording");
+ return(-1);
+ }
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ _outputFileRecording = false;
+
+ return 0;
+}
+
+void
+Channel::SetMixWithMicStatus(bool mix)
+{
+ _mixFileWithMicrophone=mix;
+}
+
+int
+Channel::GetSpeechOutputLevel(WebRtc_UWord32& level) const
+{
+ WebRtc_Word8 currentLevel = _outputAudioLevel.Level();
+ level = static_cast<WebRtc_Word32> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetSpeechOutputLevel() => level=%u", level);
+ return 0;
+}
+
+int
+Channel::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const
+{
+ WebRtc_Word16 currentLevel = _outputAudioLevel.LevelFullRange();
+ level = static_cast<WebRtc_Word32> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetSpeechOutputLevelFullRange() => level=%u", level);
+ return 0;
+}
+
+int
+Channel::SetMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetMute(enable=%d)", enable);
+ _mute = enable;
+ return 0;
+}
+
+bool
+Channel::Mute() const
+{
+ return _mute;
+}
+
+int
+Channel::SetOutputVolumePan(float left, float right)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetOutputVolumePan()");
+ _panLeft = left;
+ _panRight = right;
+ return 0;
+}
+
+int
+Channel::GetOutputVolumePan(float& left, float& right) const
+{
+ left = _panLeft;
+ right = _panRight;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right);
+ return 0;
+}
+
+int
+Channel::SetChannelOutputVolumeScaling(float scaling)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetChannelOutputVolumeScaling()");
+ _outputGain = scaling;
+ return 0;
+}
+
+int
+Channel::GetChannelOutputVolumeScaling(float& scaling) const
+{
+ scaling = _outputGain;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling);
+ return 0;
+}
+
+#ifdef WEBRTC_SRTP
+
+int
+Channel::EnableSRTPSend(
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EnableSRTPSend()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encrypting)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "EnableSRTPSend() encryption already enabled");
+ return -1;
+ }
+
+ if (key == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceWarning,
+ "EnableSRTPSend() invalid key string");
+ return -1;
+ }
+
+ if (((kEncryption == level ||
+ kEncryptionAndAuthentication == level) &&
+ (cipherKeyLength < kVoiceEngineMinSrtpEncryptLength ||
+ cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength)) ||
+ ((kAuthentication == level ||
+ kEncryptionAndAuthentication == level) &&
+ kAuthHmacSha1 == authType &&
+ (authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length ||
+ authTagLength > kVoiceEngineMaxSrtpAuthSha1Length)) ||
+ ((kAuthentication == level ||
+ kEncryptionAndAuthentication == level) &&
+ kAuthNull == authType &&
+ (authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength ||
+ authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength)))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "EnableSRTPSend() invalid key length(s)");
+ return -1;
+ }
+
+
+ if (_srtpModule.EnableSRTPEncrypt(
+ !useForRTCP,
+ (SrtpModule::CipherTypes)cipherType,
+ cipherKeyLength,
+ (SrtpModule::AuthenticationTypes)authType,
+ authKeyLength, authTagLength,
+ (SrtpModule::SecurityLevels)level,
+ key) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SRTP_ERROR, kTraceError,
+ "EnableSRTPSend() failed to enable SRTP encryption");
+ return -1;
+ }
+
+ if (_encryptionPtr == NULL)
+ {
+ _encryptionPtr = &_srtpModule;
+ }
+ _encrypting = true;
+
+ return 0;
+}
+
+int
+Channel::DisableSRTPSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DisableSRTPSend()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_encrypting)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DisableSRTPSend() SRTP encryption already disabled");
+ return 0;
+ }
+
+ _encrypting = false;
+
+ if (_srtpModule.DisableSRTPEncrypt() == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SRTP_ERROR, kTraceError,
+ "DisableSRTPSend() failed to disable SRTP encryption");
+ return -1;
+ }
+
+ if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
+ {
+ // Both directions are disabled
+ _encryptionPtr = NULL;
+ }
+
+ return 0;
+}
+
+int
+Channel::EnableSRTPReceive(
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EnableSRTPReceive()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_decrypting)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "EnableSRTPReceive() SRTP decryption already enabled");
+ return -1;
+ }
+
+ if (key == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceWarning,
+ "EnableSRTPReceive() invalid key string");
+ return -1;
+ }
+
+ if ((((kEncryption == level) ||
+ (kEncryptionAndAuthentication == level)) &&
+ ((cipherKeyLength < kVoiceEngineMinSrtpEncryptLength) ||
+ (cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength))) ||
+ (((kAuthentication == level) ||
+ (kEncryptionAndAuthentication == level)) &&
+ (kAuthHmacSha1 == authType) &&
+ ((authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length) ||
+ (authTagLength > kVoiceEngineMaxSrtpAuthSha1Length))) ||
+ (((kAuthentication == level) ||
+ (kEncryptionAndAuthentication == level)) &&
+ (kAuthNull == authType) &&
+ ((authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength) ||
+ (authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength))))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "EnableSRTPReceive() invalid key length(s)");
+ return -1;
+ }
+
+ if (_srtpModule.EnableSRTPDecrypt(
+ !useForRTCP,
+ (SrtpModule::CipherTypes)cipherType,
+ cipherKeyLength,
+ (SrtpModule::AuthenticationTypes)authType,
+ authKeyLength,
+ authTagLength,
+ (SrtpModule::SecurityLevels)level,
+ key) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SRTP_ERROR, kTraceError,
+ "EnableSRTPReceive() failed to enable SRTP decryption");
+ return -1;
+ }
+
+ if (_encryptionPtr == NULL)
+ {
+ _encryptionPtr = &_srtpModule;
+ }
+
+ _decrypting = true;
+
+ return 0;
+}
+
+int
+Channel::DisableSRTPReceive()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DisableSRTPReceive()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_decrypting)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DisableSRTPReceive() SRTP decryption already disabled");
+ return 0;
+ }
+
+ _decrypting = false;
+
+ if (_srtpModule.DisableSRTPDecrypt() == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SRTP_ERROR, kTraceError,
+ "DisableSRTPReceive() failed to disable SRTP decryption");
+ return -1;
+ }
+
+ if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
+ {
+ _encryptionPtr = NULL;
+ }
+
+ return 0;
+}
+
+#endif
+
+int
+Channel::RegisterExternalEncryption(Encryption& encryption)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterExternalEncryption()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_encryptionPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterExternalEncryption() encryption already enabled");
+ return -1;
+ }
+
+ _encryptionPtr = &encryption;
+
+ _decrypting = true;
+ _encrypting = true;
+
+ return 0;
+}
+
+int
+Channel::DeRegisterExternalEncryption()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterExternalEncryption()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_encryptionPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterExternalEncryption() encryption already disabled");
+ return 0;
+ }
+
+ _decrypting = false;
+ _encrypting = false;
+
+ _encryptionPtr = NULL;
+
+ return 0;
+}
+
+int Channel::SendTelephoneEventOutband(unsigned char eventCode,
+ int lengthMs, int attenuationDb,
+ bool playDtmfEvent)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
+ playDtmfEvent);
+
+ _playOutbandDtmfEvent = playDtmfEvent;
+
+ if (_rtpRtcpModule.SendTelephoneEventOutband(eventCode, lengthMs,
+ attenuationDb) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SEND_DTMF_FAILED,
+ kTraceWarning,
+ "SendTelephoneEventOutband() failed to send event");
+ return -1;
+ }
+ return 0;
+}
+
+int Channel::SendTelephoneEventInband(unsigned char eventCode,
+ int lengthMs,
+ int attenuationDb,
+ bool playDtmfEvent)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
+ playDtmfEvent);
+
+ _playInbandDtmfEvent = playDtmfEvent;
+ _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
+
+ return 0;
+}
+
+int
+Channel::SetDtmfPlayoutStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetDtmfPlayoutStatus()");
+ if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
+ "SetDtmfPlayoutStatus() failed to set Dtmf playout");
+ return -1;
+ }
+ return 0;
+}
+
+bool
+Channel::DtmfPlayoutStatus() const
+{
+ return _audioCodingModule.DtmfPlayoutStatus();
+}
+
+int
+Channel::SetSendTelephoneEventPayloadType(unsigned char type)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetSendTelephoneEventPayloadType()");
+ if (type > 127)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendTelephoneEventPayloadType() invalid type");
+ return -1;
+ }
+ CodecInst codec;
+ codec.plfreq = 8000;
+ codec.pltype = type;
+ memcpy(codec.plname, "telephone-event", 16);
+ if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetSendTelephoneEventPayloadType() failed to register send"
+ "payload type");
+ return -1;
+ }
+ _sendTelephoneEventPayloadType = type;
+ return 0;
+}
+
+int
+Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetSendTelephoneEventPayloadType()");
+ type = _sendTelephoneEventPayloadType;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetSendTelephoneEventPayloadType() => type=%u", type);
+ return 0;
+}
+
+#ifdef WEBRTC_DTMF_DETECTION
+
+WebRtc_Word32
+Channel::RegisterTelephoneEventDetection(
+ TelephoneEventDetectionMethods detectionMethod,
+ VoETelephoneEventObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterTelephoneEventDetection()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_telephoneEventDetectionPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterTelephoneEventDetection() detection already enabled");
+ return -1;
+ }
+
+ _telephoneEventDetectionPtr = &observer;
+
+ switch (detectionMethod)
+ {
+ case kInBand:
+ _inbandTelephoneEventDetection = true;
+ _outOfBandTelephoneEventDetecion = false;
+ break;
+ case kOutOfBand:
+ _inbandTelephoneEventDetection = false;
+ _outOfBandTelephoneEventDetecion = true;
+ break;
+ case kInAndOutOfBand:
+ _inbandTelephoneEventDetection = true;
+ _outOfBandTelephoneEventDetecion = true;
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "RegisterTelephoneEventDetection() invalid detection method");
+ return -1;
+ }
+
+ if (_inbandTelephoneEventDetection)
+ {
+ // Enable in-band Dtmf detectin in the ACM.
+ if (_audioCodingModule.RegisterIncomingMessagesCallback(this) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "RegisterTelephoneEventDetection() failed to enable Dtmf "
+ "detection");
+ }
+ }
+
+ // Enable/disable out-of-band detection of received telephone-events.
+ // When enabled, RtpAudioFeedback::OnReceivedTelephoneEvent() will be
+ // called two times by the RTP/RTCP module (start & end).
+ const bool forwardToDecoder =
+ _rtpRtcpModule.TelephoneEventForwardToDecoder();
+ const bool detectEndOfTone = true;
+ _rtpRtcpModule.SetTelephoneEventStatus(_outOfBandTelephoneEventDetecion,
+ forwardToDecoder,
+ detectEndOfTone);
+
+ return 0;
+}
+
+int
+Channel::DeRegisterTelephoneEventDetection()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::DeRegisterTelephoneEventDetection()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_telephoneEventDetectionPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION,
+ kTraceWarning,
+ "DeRegisterTelephoneEventDetection() detection already disabled");
+ return 0;
+ }
+
+ // Disable out-of-band event detection
+ const bool forwardToDecoder =
+ _rtpRtcpModule.TelephoneEventForwardToDecoder();
+ _rtpRtcpModule.SetTelephoneEventStatus(false, forwardToDecoder);
+
+ // Disable in-band Dtmf detection
+ _audioCodingModule.RegisterIncomingMessagesCallback(NULL);
+
+ _inbandTelephoneEventDetection = false;
+ _outOfBandTelephoneEventDetecion = false;
+ _telephoneEventDetectionPtr = NULL;
+
+ return 0;
+}
+
+int
+Channel::GetTelephoneEventDetectionStatus(
+ bool& enabled,
+ TelephoneEventDetectionMethods& detectionMethod)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::GetTelephoneEventDetectionStatus()");
+
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ enabled = (_telephoneEventDetectionPtr != NULL);
+ }
+
+ if (enabled)
+ {
+ if (_inbandTelephoneEventDetection && !_outOfBandTelephoneEventDetecion)
+ detectionMethod = kInBand;
+ else if (!_inbandTelephoneEventDetection
+ && _outOfBandTelephoneEventDetecion)
+ detectionMethod = kOutOfBand;
+ else if (_inbandTelephoneEventDetection
+ && _outOfBandTelephoneEventDetecion)
+ detectionMethod = kInAndOutOfBand;
+ else
+ {
+ assert(false);
+ return -1;
+ }
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetTelephoneEventDetectionStatus() => enabled=%d,"
+ "detectionMethod=%d", enabled, detectionMethod);
+ return 0;
+}
+
+#endif // #ifdef WEBRTC_DTMF_DETECTION
+
+int
+Channel::UpdateRxVadDetection(AudioFrame& audioFrame)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::UpdateRxVadDetection()");
+
+ int vadDecision = 1;
+
+ vadDecision = (audioFrame._vadActivity == AudioFrame::kVadActive)? 1 : 0;
+
+ if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr)
+ {
+ OnRxVadDetected(vadDecision);
+ _oldVadDecision = vadDecision;
+ }
+
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::UpdateRxVadDetection() => vadDecision=%d",
+ vadDecision);
+ return 0;
+}
+
+int
+Channel::RegisterRxVadObserver(VoERxVadCallback &observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterRxVadObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rxVadObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterRxVadObserver() observer already enabled");
+ return -1;
+ }
+ _rxVadObserverPtr = &observer;
+ _RxVadDetection = true;
+ return 0;
+}
+
+int
+Channel::DeRegisterRxVadObserver()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterRxVadObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_rxVadObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterRxVadObserver() observer already disabled");
+ return 0;
+ }
+ _rxVadObserverPtr = NULL;
+ _RxVadDetection = false;
+ return 0;
+}
+
+int
+Channel::VoiceActivityIndicator(int &activity)
+{
+ activity = _sendFrameType;
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::VoiceActivityIndicator(indicator=%d)", activity);
+ return 0;
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+
+int
+Channel::SetRxAgcStatus(const bool enable, const AgcModes mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetRxAgcStatus(enable=%d, mode=%d)",
+ (int)enable, (int)mode);
+
+ GainControl::Mode agcMode(GainControl::kFixedDigital);
+ switch (mode)
+ {
+ case kAgcDefault:
+ agcMode = GainControl::kAdaptiveDigital;
+ break;
+ case kAgcUnchanged:
+ agcMode = _rxAudioProcessingModulePtr->gain_control()->mode();
+ break;
+ case kAgcFixedDigital:
+ agcMode = GainControl::kFixedDigital;
+ break;
+ case kAgcAdaptiveDigital:
+ agcMode =GainControl::kAdaptiveDigital;
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetRxAgcStatus() invalid Agc mode");
+ return -1;
+ }
+
+ if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcStatus() failed to set Agc mode");
+ return -1;
+ }
+ if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcStatus() failed to set Agc state");
+ return -1;
+ }
+
+ _rxAgcIsEnabled = enable;
+
+ _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
+
+ return 0;
+}
+
+int
+Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRxAgcStatus(enable=?, mode=?)");
+
+ bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled();
+ GainControl::Mode agcMode =
+ _rxAudioProcessingModulePtr->gain_control()->mode();
+
+ enabled = enable;
+
+ switch (agcMode)
+ {
+ case GainControl::kFixedDigital:
+ mode = kAgcFixedDigital;
+ break;
+ case GainControl::kAdaptiveDigital:
+ mode = kAgcAdaptiveDigital;
+ break;
+ default:
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "GetRxAgcStatus() invalid Agc mode");
+ return -1;
+ }
+
+ return 0;
+}
+
+int
+Channel::SetRxAgcConfig(const AgcConfig config)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetRxAgcConfig()");
+
+ if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs(
+ config.targetLeveldBOv) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcConfig() failed to set target peak |level|"
+ "(or envelope) of the Agc");
+ return -1;
+ }
+ if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db(
+ config.digitalCompressionGaindB) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcConfig() failed to set the range in |gain| the"
+ " digital compression stage may apply");
+ return -1;
+ }
+ if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter(
+ config.limiterEnable) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcConfig() failed to set hard limiter to the signal");
+ return -1;
+ }
+
+ return 0;
+}
+
+int
+Channel::GetRxAgcConfig(AgcConfig& config)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRxAgcConfig(config=%?)");
+
+ config.targetLeveldBOv =
+ _rxAudioProcessingModulePtr->gain_control()->target_level_dbfs();
+ config.digitalCompressionGaindB =
+ _rxAudioProcessingModulePtr->gain_control()->compression_gain_db();
+ config.limiterEnable =
+ _rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
+ "targetLeveldBOv=%u, digitalCompressionGaindB=%u,"
+ " limiterEnable=%d",
+ config.targetLeveldBOv,
+ config.digitalCompressionGaindB,
+ config.limiterEnable);
+
+ return 0;
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_AGC
+
+#ifdef WEBRTC_VOICE_ENGINE_NR
+
+int
+Channel::SetRxNsStatus(const bool enable, const NsModes mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetRxNsStatus(enable=%d, mode=%d)",
+ (int)enable, (int)mode);
+
+ NoiseSuppression::Level nsLevel(
+ (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE);
+ switch (mode)
+ {
+
+ case kNsDefault:
+ nsLevel = (NoiseSuppression::Level)
+ WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE;
+ break;
+ case kNsUnchanged:
+ nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level();
+ break;
+ case kNsConference:
+ nsLevel = NoiseSuppression::kHigh;
+ break;
+ case kNsLowSuppression:
+ nsLevel = NoiseSuppression::kLow;
+ break;
+ case kNsModerateSuppression:
+ nsLevel = NoiseSuppression::kModerate;
+ break;
+ case kNsHighSuppression:
+ nsLevel = NoiseSuppression::kHigh;
+ break;
+ case kNsVeryHighSuppression:
+ nsLevel = NoiseSuppression::kVeryHigh;
+ break;
+ }
+
+ if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel)
+ != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcStatus() failed to set Ns level");
+ return -1;
+ }
+ if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetRxAgcStatus() failed to set Agc state");
+ return -1;
+ }
+
+ _rxNsIsEnabled = enable;
+ _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
+
+ return 0;
+}
+
+int
+Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRxNsStatus(enable=?, mode=?)");
+
+ bool enable =
+ _rxAudioProcessingModulePtr->noise_suppression()->is_enabled();
+ NoiseSuppression::Level ncLevel =
+ _rxAudioProcessingModulePtr->noise_suppression()->level();
+
+ enabled = enable;
+
+ switch (ncLevel)
+ {
+ case NoiseSuppression::kLow:
+ mode = kNsLowSuppression;
+ break;
+ case NoiseSuppression::kModerate:
+ mode = kNsModerateSuppression;
+ break;
+ case NoiseSuppression::kHigh:
+ mode = kNsHighSuppression;
+ break;
+ case NoiseSuppression::kVeryHigh:
+ mode = kNsVeryHighSuppression;
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode);
+ return 0;
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_NR
+
+int
+Channel::RegisterRTPObserver(VoERTPObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::RegisterRTPObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rtpObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterRTPObserver() observer already enabled");
+ return -1;
+ }
+
+ _rtpObserverPtr = &observer;
+ _rtpObserver = true;
+
+ return 0;
+}
+
+int
+Channel::DeRegisterRTPObserver()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterRTPObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_rtpObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterRTPObserver() observer already disabled");
+ return 0;
+ }
+
+ _rtpObserver = false;
+ _rtpObserverPtr = NULL;
+
+ return 0;
+}
+
+int
+Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterRTCPObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_rtcpObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterRTCPObserver() observer already enabled");
+ return -1;
+ }
+
+ _rtcpObserverPtr = &observer;
+ _rtcpObserver = true;
+
+ return 0;
+}
+
+int
+Channel::DeRegisterRTCPObserver()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::DeRegisterRTCPObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (!_rtcpObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "DeRegisterRTCPObserver() observer already disabled");
+ return 0;
+ }
+
+ _rtcpObserver = false;
+ _rtcpObserverPtr = NULL;
+
+ return 0;
+}
+
+int
+Channel::SetLocalSSRC(unsigned int ssrc)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SetLocalSSRC()");
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_SENDING, kTraceError,
+ "SetLocalSSRC() already sending");
+ return -1;
+ }
+ if (_rtpRtcpModule.SetSSRC(ssrc) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetLocalSSRC() failed to set SSRC");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetLocalSSRC(unsigned int& ssrc)
+{
+ ssrc = _rtpRtcpModule.SSRC();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetLocalSSRC() => ssrc=%lu", ssrc);
+ return 0;
+}
+
+int
+Channel::GetRemoteSSRC(unsigned int& ssrc)
+{
+ ssrc = _rtpRtcpModule.RemoteSSRC();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetRemoteSSRC() => ssrc=%lu", ssrc);
+ return 0;
+}
+
+int
+Channel::GetRemoteCSRCs(unsigned int arrCSRC[15])
+{
+ if (arrCSRC == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetRemoteCSRCs() invalid array argument");
+ return -1;
+ }
+ WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize];
+ WebRtc_Word32 CSRCs(0);
+ CSRCs = _rtpRtcpModule.CSRCs(arrOfCSRC);
+ if (CSRCs > 0)
+ {
+ memcpy(arrCSRC, arrOfCSRC, CSRCs * sizeof(WebRtc_UWord32));
+ for (int i = 0; i < (int) CSRCs; i++)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteCSRCs() => arrCSRC[%d]=%lu", i, arrCSRC[i]);
+ }
+ } else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteCSRCs() => list is empty!");
+ }
+ return CSRCs;
+}
+
+int
+Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID)
+{
+ if (_rtpAudioProc.get() == NULL)
+ {
+ _rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
+ _channelId)));
+ if (_rtpAudioProc.get() == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical,
+ "Failed to create AudioProcessing");
+ return -1;
+ }
+ }
+
+ if (_rtpAudioProc->level_estimator()->Enable(enable) !=
+ AudioProcessing::kNoError)
+ {
+ _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning,
+ "Failed to enable AudioProcessing::level_estimator()");
+ }
+
+ _includeAudioLevelIndication = enable;
+ return _rtpRtcpModule.SetRTPAudioLevelIndicationStatus(enable, ID);
+}
+int
+Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID)
+{
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetRTPAudioLevelIndicationStatus() => enabled=%d, ID=%u",
+ enabled, ID);
+ return _rtpRtcpModule.GetRTPAudioLevelIndicationStatus(enabled, ID);
+}
+
+int
+Channel::SetRTCPStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetRTCPStatus()");
+ if (_rtpRtcpModule.SetRTCPStatus(enable ?
+ kRtcpCompound : kRtcpOff) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetRTCPStatus() failed to set RTCP status");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetRTCPStatus(bool& enabled)
+{
+ RTCPMethod method = _rtpRtcpModule.RTCP();
+ enabled = (method != kRtcpOff);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetRTCPStatus() => enabled=%d", enabled);
+ return 0;
+}
+
+int
+Channel::SetRTCP_CNAME(const char cName[256])
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SetRTCP_CNAME()");
+ if (_rtpRtcpModule.SetCNAME(cName) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetRTCP_CNAME() failed to set RTCP CNAME");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetRTCP_CNAME(char cName[256])
+{
+ if (_rtpRtcpModule.CNAME(cName) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "GetRTCP_CNAME() failed to retrieve RTCP CNAME");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTCP_CNAME() => cName=%s", cName);
+ return 0;
+}
+
+int
+Channel::GetRemoteRTCP_CNAME(char cName[256])
+{
+ if (cName == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetRemoteRTCP_CNAME() invalid CNAME input buffer");
+ return -1;
+ }
+ WebRtc_Word8 cname[RTCP_CNAME_SIZE];
+ const WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
+ if (_rtpRtcpModule.RemoteCNAME(remoteSSRC, cname) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_RETRIEVE_CNAME, kTraceError,
+ "GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
+ return -1;
+ }
+ strcpy(cName, cname);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCP_CNAME() => cName=%s", cName);
+ return 0;
+}
+
+int
+Channel::GetRemoteRTCPData(
+ unsigned int& NTPHigh,
+ unsigned int& NTPLow,
+ unsigned int& timestamp,
+ unsigned int& playoutTimestamp,
+ unsigned int* jitter,
+ unsigned short* fractionLost)
+{
+ // --- Information from sender info in received Sender Reports
+
+ RTCPSenderInfo senderInfo;
+ if (_rtpRtcpModule.RemoteRTCPStat(&senderInfo) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "GetRemoteRTCPData() failed to retrieve sender info for remote "
+ "side");
+ return -1;
+ }
+
+ // We only utilize 12 out of 20 bytes in the sender info (ignores packet
+ // and octet count)
+ NTPHigh = senderInfo.NTPseconds;
+ NTPLow = senderInfo.NTPfraction;
+ timestamp = senderInfo.RTPtimeStamp;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, "
+ "timestamp=%lu",
+ NTPHigh, NTPLow, timestamp);
+
+ // --- Locally derived information
+
+ // This value is updated on each incoming RTCP packet (0 when no packet
+ // has been received)
+ playoutTimestamp = _playoutTimeStampRTCP;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCPData() => playoutTimestamp=%lu",
+ _playoutTimeStampRTCP);
+
+ if (NULL != jitter || NULL != fractionLost)
+ {
+ // Get all RTCP receiver report blocks that have been received on this
+ // channel. If we receive RTP packets from a remote source we know the
+ // remote SSRC and use the report block from him.
+ // Otherwise use the first report block.
+ std::vector<RTCPReportBlock> remote_stats;
+ if (_rtpRtcpModule.RemoteRTCPStat(&remote_stats) != 0 ||
+ remote_stats.empty()) {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCPData() failed to measure statistics due"
+ " to lack of received RTP and/or RTCP packets");
+ return -1;
+ }
+
+ WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
+ std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin();
+ for (; it != remote_stats.end(); ++it) {
+ if (it->remoteSSRC == remoteSSRC)
+ break;
+ }
+
+ if (it == remote_stats.end()) {
+ // If we have not received any RTCP packets from this SSRC it probably
+ // means that we have not received any RTP packets.
+ // Use the first received report block instead.
+ it = remote_stats.begin();
+ remoteSSRC = it->remoteSSRC;
+ }
+
+ if (jitter) {
+ *jitter = it->jitter;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCPData() => jitter = %lu", *jitter);
+ }
+
+ if (fractionLost) {
+ *fractionLost = it->fractionLost;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRemoteRTCPData() => fractionLost = %lu",
+ *fractionLost);
+ }
+ }
+ return 0;
+}
+
+int
+Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType,
+ unsigned int name,
+ const char* data,
+ unsigned short dataLengthInBytes)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SendApplicationDefinedRTCPPacket()");
+ if (!_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_NOT_SENDING, kTraceError,
+ "SendApplicationDefinedRTCPPacket() not sending");
+ return -1;
+ }
+ if (NULL == data)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SendApplicationDefinedRTCPPacket() invalid data value");
+ return -1;
+ }
+ if (dataLengthInBytes % 4 != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SendApplicationDefinedRTCPPacket() invalid length value");
+ return -1;
+ }
+ RTCPMethod status = _rtpRtcpModule.RTCP();
+ if (status == kRtcpOff)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTCP_ERROR, kTraceError,
+ "SendApplicationDefinedRTCPPacket() RTCP is disabled");
+ return -1;
+ }
+
+ // Create and schedule the RTCP APP packet for transmission
+ if (_rtpRtcpModule.SetRTCPApplicationSpecificData(
+ subType,
+ name,
+ (const unsigned char*) data,
+ dataLengthInBytes) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SEND_ERROR, kTraceError,
+ "SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetRTPStatistics(
+ unsigned int& averageJitterMs,
+ unsigned int& maxJitterMs,
+ unsigned int& discardedPackets)
+{
+ WebRtc_UWord8 fraction_lost(0);
+ WebRtc_UWord32 cum_lost(0);
+ WebRtc_UWord32 ext_max(0);
+ WebRtc_UWord32 jitter(0);
+ WebRtc_UWord32 max_jitter(0);
+
+ // The jitter statistics is updated for each received RTP packet and is
+ // based on received packets.
+ if (_rtpRtcpModule.StatisticsRTP(&fraction_lost,
+ &cum_lost,
+ &ext_max,
+ &jitter,
+ &max_jitter) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
+ "GetRTPStatistics() failed to read RTP statistics from the "
+ "RTP/RTCP module");
+ }
+
+ const WebRtc_Word32 playoutFrequency =
+ _audioCodingModule.PlayoutFrequency();
+ if (playoutFrequency > 0)
+ {
+ // Scale RTP statistics given the current playout frequency
+ maxJitterMs = max_jitter / (playoutFrequency / 1000);
+ averageJitterMs = jitter / (playoutFrequency / 1000);
+ }
+
+ discardedPackets = _numberOfDiscardedPackets;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu,"
+ " discardedPackets = %lu)",
+ averageJitterMs, maxJitterMs, discardedPackets);
+ return 0;
+}
+
+int
+Channel::GetRTPStatistics(CallStatistics& stats)
+{
+ WebRtc_UWord8 fraction_lost(0);
+ WebRtc_UWord32 cum_lost(0);
+ WebRtc_UWord32 ext_max(0);
+ WebRtc_UWord32 jitter(0);
+ WebRtc_UWord32 max_jitter(0);
+
+ // --- Part one of the final structure (four values)
+
+ // The jitter statistics is updated for each received RTP packet and is
+ // based on received packets.
+ if (_rtpRtcpModule.StatisticsRTP(&fraction_lost,
+ &cum_lost,
+ &ext_max,
+ &jitter,
+ &max_jitter) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
+ "GetRTPStatistics() failed to read RTP statistics from the "
+ "RTP/RTCP module");
+ }
+
+ stats.fractionLost = fraction_lost;
+ stats.cumulativeLost = cum_lost;
+ stats.extendedMax = ext_max;
+ stats.jitterSamples = jitter;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu,"
+ " extendedMax=%lu, jitterSamples=%li)",
+ stats.fractionLost, stats.cumulativeLost, stats.extendedMax,
+ stats.jitterSamples);
+
+ // --- Part two of the final structure (one value)
+
+ WebRtc_UWord16 RTT(0);
+ RTCPMethod method = _rtpRtcpModule.RTCP();
+ if (method == kRtcpOff)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() RTCP is disabled => valid RTT "
+ "measurements cannot be retrieved");
+ } else
+ {
+ // The remote SSRC will be zero if no RTP packet has been received.
+ WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
+ if (remoteSSRC > 0)
+ {
+ WebRtc_UWord16 avgRTT(0);
+ WebRtc_UWord16 maxRTT(0);
+ WebRtc_UWord16 minRTT(0);
+
+ if (_rtpRtcpModule.RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT)
+ != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() failed to retrieve RTT from "
+ "the RTP/RTCP module");
+ }
+ } else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() failed to measure RTT since no "
+ "RTP packets have been received yet");
+ }
+ }
+
+ stats.rttMs = static_cast<int> (RTT);
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() => rttMs=%d", stats.rttMs);
+
+ // --- Part three of the final structure (four values)
+
+ WebRtc_UWord32 bytesSent(0);
+ WebRtc_UWord32 packetsSent(0);
+ WebRtc_UWord32 bytesReceived(0);
+ WebRtc_UWord32 packetsReceived(0);
+
+ if (_rtpRtcpModule.DataCountersRTP(&bytesSent,
+ &packetsSent,
+ &bytesReceived,
+ &packetsReceived) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() failed to retrieve RTP datacounters =>"
+ " output will not be complete");
+ }
+
+ stats.bytesSent = bytesSent;
+ stats.packetsSent = packetsSent;
+ stats.bytesReceived = bytesReceived;
+ stats.packetsReceived = packetsReceived;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPStatistics() => bytesSent=%d, packetsSent=%d,"
+ " bytesReceived=%d, packetsReceived=%d)",
+ stats.bytesSent, stats.packetsSent, stats.bytesReceived,
+ stats.packetsReceived);
+
+ return 0;
+}
+
+int
+Channel::SetFECStatus(bool enable, int redPayloadtype)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SetFECStatus()");
+
+ CodecInst codec;
+
+ // Get default RED settings from the ACM database
+ bool foundRED(false);
+ const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
+ for (int idx = 0; (!foundRED && idx < nSupportedCodecs); idx++)
+ {
+ _audioCodingModule.Codec(idx, codec);
+ if (!STR_CASE_CMP(codec.plname, "RED"))
+ {
+ foundRED = true;
+ }
+ }
+ if (!foundRED)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CODEC_ERROR, kTraceError,
+ "SetFECStatus() RED is not supported");
+ return -1;
+ }
+
+ if (redPayloadtype != -1)
+ {
+ codec.pltype = redPayloadtype;
+ }
+
+ if (_audioCodingModule.RegisterSendCodec(codec) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetFECStatus() RED registration in ACM module failed");
+ return -1;
+ }
+ if (_rtpRtcpModule.SetSendREDPayloadType(codec.pltype) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetFECStatus() RED registration in RTP/RTCP module failed");
+ return -1;
+ }
+ if (_audioCodingModule.SetFECStatus(enable) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetFECStatus() failed to set FEC state in the ACM");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetFECStatus(bool& enabled, int& redPayloadtype)
+{
+ enabled = _audioCodingModule.FECStatus();
+ if (enabled)
+ {
+ WebRtc_Word8 payloadType(0);
+ if (_rtpRtcpModule.SendREDPayloadType(payloadType) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "GetFECStatus() failed to retrieve RED PT from RTP/RTCP "
+ "module");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetFECStatus() => enabled=%d, redPayloadtype=%d",
+ enabled, redPayloadtype);
+ return 0;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetFECStatus() => enabled=%d", enabled);
+ return 0;
+}
+
+int
+Channel::SetRTPKeepaliveStatus(bool enable,
+ unsigned char unknownPayloadType,
+ int deltaTransmitTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::SetRTPKeepaliveStatus()");
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_SENDING, kTraceError,
+ "SetRTPKeepaliveStatus() already sending");
+ return -1;
+ }
+ if (_rtpRtcpModule.SetRTPKeepaliveStatus(
+ enable,
+ unknownPayloadType,
+ 1000 * deltaTransmitTimeSeconds) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetRTPKeepaliveStatus() failed to set RTP keepalive status");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetRTPKeepaliveStatus(bool& enabled,
+ unsigned char& unknownPayloadType,
+ int& deltaTransmitTimeSeconds)
+{
+ bool onOff(false);
+ WebRtc_Word8 payloadType(0);
+ WebRtc_UWord16 deltaTransmitTimeMS(0);
+ if (_rtpRtcpModule.RTPKeepaliveStatus(&onOff, &payloadType,
+ &deltaTransmitTimeMS) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "GetRTPKeepaliveStatus() failed to retrieve RTP keepalive status");
+ return -1;
+ }
+ enabled = onOff;
+ unknownPayloadType = payloadType;
+ deltaTransmitTimeSeconds = static_cast<int> (deltaTransmitTimeMS / 1000);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "GetRTPKeepaliveStatus() => enabled=%d, "
+ "unknownPayloadType=%u, deltaTransmitTimeSeconds=%d",
+ enabled, unknownPayloadType, deltaTransmitTimeSeconds);
+ return 0;
+}
+
+int
+Channel::StartRTPDump(const char fileNameUTF8[1024],
+ RTPDirections direction)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::StartRTPDump()");
+ if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRTPDump() invalid RTP direction");
+ return -1;
+ }
+ RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
+ &_rtpDumpIn : &_rtpDumpOut;
+ if (rtpDumpPtr == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ if (rtpDumpPtr->IsActive())
+ {
+ rtpDumpPtr->Stop();
+ }
+ if (rtpDumpPtr->Start(fileNameUTF8) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRTPDump() failed to create file");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::StopRTPDump(RTPDirections direction)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::StopRTPDump()");
+ if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StopRTPDump() invalid RTP direction");
+ return -1;
+ }
+ RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
+ &_rtpDumpIn : &_rtpDumpOut;
+ if (rtpDumpPtr == NULL)
+ {
+ assert(false);
+ return -1;
+ }
+ if (!rtpDumpPtr->IsActive())
+ {
+ return 0;
+ }
+ return rtpDumpPtr->Stop();
+}
+
+bool
+Channel::RTPDumpIsActive(RTPDirections direction)
+{
+ if ((direction != kRtpIncoming) &&
+ (direction != kRtpOutgoing))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "RTPDumpIsActive() invalid RTP direction");
+ return false;
+ }
+ RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
+ &_rtpDumpIn : &_rtpDumpOut;
+ return rtpDumpPtr->IsActive();
+}
+
+int
+Channel::InsertExtraRTPPacket(unsigned char payloadType,
+ bool markerBit,
+ const char* payloadData,
+ unsigned short payloadSize)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::InsertExtraRTPPacket()");
+ if (payloadType > 127)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_PLTYPE, kTraceError,
+ "InsertExtraRTPPacket() invalid payload type");
+ return -1;
+ }
+ if (payloadData == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "InsertExtraRTPPacket() invalid payload data");
+ return -1;
+ }
+ if (payloadSize > _rtpRtcpModule.MaxDataPayloadLength())
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "InsertExtraRTPPacket() invalid payload size");
+ return -1;
+ }
+ if (!_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_NOT_SENDING, kTraceError,
+ "InsertExtraRTPPacket() not sending");
+ return -1;
+ }
+
+ // Create extra RTP packet by calling RtpRtcp::SendOutgoingData().
+ // Transport::SendPacket() will be called by the module when the RTP packet
+ // is created.
+ // The call to SendOutgoingData() does *not* modify the timestamp and
+ // payloadtype to ensure that the RTP module generates a valid RTP packet
+ // (user might utilize a non-registered payload type).
+ // The marker bit and payload type will be replaced just before the actual
+ // transmission, i.e., the actual modification is done *after* the RTP
+ // module has delivered its RTP packet back to the VoE.
+ // We will use the stored values above when the packet is modified
+ // (see Channel::SendPacket()).
+
+ _extraPayloadType = payloadType;
+ _extraMarkerBit = markerBit;
+ _insertExtraRTPPacket = true;
+
+ if (_rtpRtcpModule.SendOutgoingData(kAudioFrameSpeech,
+ _lastPayloadType,
+ _lastLocalTimeStamp,
+ (const WebRtc_UWord8*) payloadData,
+ payloadSize) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "InsertExtraRTPPacket() failed to send extra RTP packet");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_UWord32
+Channel::Demultiplex(const AudioFrame& audioFrame)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::Demultiplex()");
+ _audioFrame = audioFrame;
+ _audioFrame._id = _channelId;
+ return 0;
+}
+
+WebRtc_UWord32
+Channel::PrepareEncodeAndSend(int mixingFrequency)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::PrepareEncodeAndSend()");
+
+ if (_audioFrame._payloadDataLengthInSamples == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::PrepareEncodeAndSend() invalid audio frame");
+ return -1;
+ }
+
+ if (_inputFilePlaying)
+ {
+ MixOrReplaceAudioWithFile(mixingFrequency);
+ }
+
+ if (_mute)
+ {
+ AudioFrameOperations::Mute(_audioFrame);
+ }
+
+ if (_inputExternalMedia)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ const bool isStereo = (_audioFrame._audioChannel == 2);
+ if (_inputExternalMediaCallbackPtr)
+ {
+ _inputExternalMediaCallbackPtr->Process(
+ _channelId,
+ kRecordingPerChannel,
+ (WebRtc_Word16*)_audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples,
+ _audioFrame._frequencyInHz,
+ isStereo);
+ }
+ }
+
+ InsertInbandDtmfTone();
+
+ if (_includeAudioLevelIndication)
+ {
+ assert(_rtpAudioProc.get() != NULL);
+
+ // Check if settings need to be updated.
+ if (_rtpAudioProc->sample_rate_hz() != _audioFrame._frequencyInHz)
+ {
+ if (_rtpAudioProc->set_sample_rate_hz(_audioFrame._frequencyInHz) !=
+ AudioProcessing::kNoError)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Error setting AudioProcessing sample rate");
+ return -1;
+ }
+ }
+
+ if (_rtpAudioProc->num_input_channels() != _audioFrame._audioChannel)
+ {
+ if (_rtpAudioProc->set_num_channels(_audioFrame._audioChannel,
+ _audioFrame._audioChannel)
+ != AudioProcessing::kNoError)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Error setting AudioProcessing channels");
+ return -1;
+ }
+ }
+
+ // Performs level analysis only; does not affect the signal.
+ _rtpAudioProc->ProcessStream(&_audioFrame);
+ }
+
+ return 0;
+}
+
+WebRtc_UWord32
+Channel::EncodeAndSend()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EncodeAndSend()");
+
+ assert(_audioFrame._audioChannel <= 2);
+ if (_audioFrame._payloadDataLengthInSamples == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EncodeAndSend() invalid audio frame");
+ return -1;
+ }
+
+ _audioFrame._id = _channelId;
+
+ // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
+
+ // The ACM resamples internally.
+ _audioFrame._timeStamp = _timeStamp;
+ if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::EncodeAndSend() ACM encoding failed");
+ return -1;
+ }
+
+ _timeStamp += _audioFrame._payloadDataLengthInSamples;
+
+ // --- Encode if complete frame is ready
+
+ // This call will trigger AudioPacketizationCallback::SendData if encoding
+ // is done and payload is ready for packetization and transmission.
+ return _audioCodingModule.Process();
+}
+
+int Channel::RegisterExternalMediaProcessing(
+ ProcessingTypes type,
+ VoEMediaProcess& processObject)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (kPlaybackPerChannel == type)
+ {
+ if (_outputExternalMediaCallbackPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "Channel::RegisterExternalMediaProcessing() "
+ "output external media already enabled");
+ return -1;
+ }
+ _outputExternalMediaCallbackPtr = &processObject;
+ _outputExternalMedia = true;
+ }
+ else if (kRecordingPerChannel == type)
+ {
+ if (_inputExternalMediaCallbackPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "Channel::RegisterExternalMediaProcessing() "
+ "output external media already enabled");
+ return -1;
+ }
+ _inputExternalMediaCallbackPtr = &processObject;
+ _inputExternalMedia = true;
+ }
+ return 0;
+}
+
+int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::DeRegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (kPlaybackPerChannel == type)
+ {
+ if (!_outputExternalMediaCallbackPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "Channel::DeRegisterExternalMediaProcessing() "
+ "output external media already disabled");
+ return 0;
+ }
+ _outputExternalMedia = false;
+ _outputExternalMediaCallbackPtr = NULL;
+ }
+ else if (kRecordingPerChannel == type)
+ {
+ if (!_inputExternalMediaCallbackPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "Channel::DeRegisterExternalMediaProcessing() "
+ "input external media already disabled");
+ return 0;
+ }
+ _inputExternalMedia = false;
+ _inputExternalMediaCallbackPtr = NULL;
+ }
+
+ return 0;
+}
+
+int
+Channel::ResetRTCPStatistics()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ResetRTCPStatistics()");
+ WebRtc_UWord32 remoteSSRC(0);
+ remoteSSRC = _rtpRtcpModule.RemoteSSRC();
+ return _rtpRtcpModule.ResetRTT(remoteSSRC);
+}
+
+int
+Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRoundTripTimeSummary()");
+ // Override default module outputs for the case when RTCP is disabled.
+ // This is done to ensure that we are backward compatible with the
+ // VoiceEngine where we did not use RTP/RTCP module.
+ if (!_rtpRtcpModule.RTCP())
+ {
+ delaysMs.min = -1;
+ delaysMs.max = -1;
+ delaysMs.average = -1;
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRoundTripTimeSummary() RTCP is disabled =>"
+ " valid RTT measurements cannot be retrieved");
+ return 0;
+ }
+
+ WebRtc_UWord32 remoteSSRC;
+ WebRtc_UWord16 RTT;
+ WebRtc_UWord16 avgRTT;
+ WebRtc_UWord16 maxRTT;
+ WebRtc_UWord16 minRTT;
+ // The remote SSRC will be zero if no RTP packet has been received.
+ remoteSSRC = _rtpRtcpModule.RemoteSSRC();
+ if (remoteSSRC == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRoundTripTimeSummary() unable to measure RTT"
+ " since no RTP packet has been received yet");
+ }
+
+ // Retrieve RTT statistics from the RTP/RTCP module for the specified
+ // channel and SSRC. The SSRC is required to parse out the correct source
+ // in conference scenarios.
+ if (_rtpRtcpModule.RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "GetRoundTripTimeSummary unable to retrieve RTT values"
+ " from the RTCP layer");
+ delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1;
+ }
+ else
+ {
+ delaysMs.min = minRTT;
+ delaysMs.max = maxRTT;
+ delaysMs.average = avgRTT;
+ }
+ return 0;
+}
+
+int
+Channel::GetNetworkStatistics(NetworkStatistics& stats)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetNetworkStatistics()");
+ return _audioCodingModule.NetworkStatistics(
+ (ACMNetworkStatistics &)stats);
+}
+
+int
+Channel::GetDelayEstimate(int& delayMs) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetDelayEstimate()");
+ delayMs = (_averageDelayMs + 5) / 10 + _recPacketDelayMs;
+ return 0;
+}
+
+int
+Channel::SetMinimumPlayoutDelay(int delayMs)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetMinimumPlayoutDelay()");
+ if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
+ (delayMs > kVoiceEngineMaxMinPlayoutDelayMs))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetMinimumPlayoutDelay() invalid min delay");
+ return -1;
+ }
+ if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
+ "SetMinimumPlayoutDelay() failed to set min playout delay");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetPlayoutTimestamp(unsigned int& timestamp)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetPlayoutTimestamp()");
+ WebRtc_UWord32 playoutTimestamp(0);
+ if (GetPlayoutTimeStamp(playoutTimestamp) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_RETRIEVE_VALUE, kTraceError,
+ "GetPlayoutTimestamp() failed to retrieve timestamp");
+ return -1;
+ }
+ timestamp = playoutTimestamp;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "GetPlayoutTimestamp() => timestamp=%u", timestamp);
+ return 0;
+}
+
+int
+Channel::SetInitTimestamp(unsigned int timestamp)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetInitTimestamp()");
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SENDING, kTraceError, "SetInitTimestamp() already sending");
+ return -1;
+ }
+ if (_rtpRtcpModule.SetStartTimestamp(timestamp) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetInitTimestamp() failed to set timestamp");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::SetInitSequenceNumber(short sequenceNumber)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::SetInitSequenceNumber()");
+ if (_sending)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SENDING, kTraceError,
+ "SetInitSequenceNumber() already sending");
+ return -1;
+ }
+ if (_rtpRtcpModule.SetSequenceNumber(sequenceNumber) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+ "SetInitSequenceNumber() failed to set sequence number");
+ return -1;
+ }
+ return 0;
+}
+
+int
+Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetRtpRtcp()");
+ rtpRtcpModule = &_rtpRtcpModule;
+ return 0;
+}
+
+WebRtc_Word32
+Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
+{
+ WebRtc_Word16 fileBuffer[320];
+ WebRtc_UWord32 fileSamples(0);
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_inputFilePlayerPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::MixOrReplaceAudioWithFile() fileplayer"
+ " doesnt exist");
+ return -1;
+ }
+
+ if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer,
+ fileSamples,
+ mixingFrequency) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::MixOrReplaceAudioWithFile() file mixing "
+ "failed");
+ return -1;
+ }
+ if (fileSamples == 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::MixOrReplaceAudioWithFile() file is ended");
+ return 0;
+ }
+ }
+
+ assert(_audioFrame._payloadDataLengthInSamples == fileSamples);
+
+ if (_mixFileWithMicrophone)
+ {
+ Utility::MixWithSat(_audioFrame._payloadData,
+ fileBuffer,
+ (WebRtc_UWord16)fileSamples);
+ }
+ else
+ {
+ // replace ACM audio with file
+ _audioFrame.UpdateFrame(_channelId,
+ -1,
+ fileBuffer,
+ (WebRtc_UWord16)fileSamples,
+ mixingFrequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadUnknown,
+ 1);
+
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::MixAudioWithFile(AudioFrame& audioFrame,
+ const int mixingFrequency)
+{
+ assert(mixingFrequency <= 32000);
+
+ WebRtc_Word16 fileBuffer[640];
+ WebRtc_UWord32 fileSamples(0);
+
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFilePlayerPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::MixAudioWithFile() file mixing failed");
+ return -1;
+ }
+
+ // We should get the frequency we ask for.
+ if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer,
+ fileSamples,
+ mixingFrequency) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::MixAudioWithFile() file mixing failed");
+ return -1;
+ }
+ }
+
+ if (audioFrame._payloadDataLengthInSamples == fileSamples)
+ {
+ // In case the incoming stream is stereo and file stream is mono,
+ // turn the file stream into stereo.
+ // TODO(xians): remove the code when FilePlayer supports real stereo.
+ if (audioFrame._audioChannel == 2)
+ {
+ // The mono file stream is copied to be stereo.
+ WebRtc_Word16* FileBufferCopy = new WebRtc_Word16[fileSamples];
+ memcpy(FileBufferCopy, fileBuffer,
+ sizeof(WebRtc_Word16) * fileSamples);
+ for (unsigned int i = 0; i < fileSamples; i++)
+ {
+ fileBuffer[2*i] = FileBufferCopy[i];
+ fileBuffer[2*i+1] = FileBufferCopy[i];
+ }
+ fileSamples = 2*fileSamples;
+ delete [] FileBufferCopy;
+ }
+
+ // Mix the incoming stream and file stream.
+ Utility::MixWithSat(audioFrame._payloadData,
+ fileBuffer,
+ (WebRtc_UWord16)fileSamples);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::MixAudioWithFile() _payloadDataLengthInSamples(%d) != "
+ "fileSamples(%d)",
+ audioFrame._payloadDataLengthInSamples, fileSamples);
+ return -1;
+ }
+
+ return 0;
+}
+
+int
+Channel::InsertInbandDtmfTone()
+{
+ // Check if we should start a new tone.
+ if (_inbandDtmfQueue.PendingDtmf() &&
+ !_inbandDtmfGenerator.IsAddingTone() &&
+ _inbandDtmfGenerator.DelaySinceLastTone() >
+ kMinTelephoneEventSeparationMs)
+ {
+ WebRtc_Word8 eventCode(0);
+ WebRtc_UWord16 lengthMs(0);
+ WebRtc_UWord8 attenuationDb(0);
+
+ eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
+ _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
+ if (_playInbandDtmfEvent)
+ {
+ // Add tone to output mixer using a reduced length to minimize
+ // risk of echo.
+ _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80,
+ attenuationDb);
+ }
+ }
+
+ if (_inbandDtmfGenerator.IsAddingTone())
+ {
+ WebRtc_UWord16 frequency(0);
+ _inbandDtmfGenerator.GetSampleRate(frequency);
+
+ if (frequency != _audioFrame._frequencyInHz)
+ {
+ // Update sample rate of Dtmf tone since the mixing frequency
+ // has changed.
+ _inbandDtmfGenerator.SetSampleRate(
+ (WebRtc_UWord16) (_audioFrame._frequencyInHz));
+ // Reset the tone to be added taking the new sample rate into
+ // account.
+ _inbandDtmfGenerator.ResetTone();
+ }
+
+ WebRtc_Word16 toneBuffer[320];
+ WebRtc_UWord16 toneSamples(0);
+ // Get 10ms tone segment and set time since last tone to zero
+ if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::EncodeAndSend() inserting Dtmf failed");
+ return -1;
+ }
+
+ // Replace mixed audio with DTMF tone.
+ for (int sample = 0;
+ sample < _audioFrame._payloadDataLengthInSamples;
+ sample++)
+ {
+ for (int channel = 0;
+ channel < _audioFrame._audioChannel;
+ channel++)
+ {
+ _audioFrame._payloadData[sample * _audioFrame._audioChannel + channel] =
+ toneBuffer[sample];
+ }
+ }
+
+ assert(_audioFrame._payloadDataLengthInSamples == toneSamples);
+ } else
+ {
+ // Add 10ms to "delay-since-last-tone" counter
+ _inbandDtmfGenerator.UpdateDelaySinceLastTone();
+ }
+ return 0;
+}
+
+WebRtc_Word32
+Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp)
+{
+ WebRtc_UWord32 timestamp(0);
+ CodecInst currRecCodec;
+
+ if (_audioCodingModule.PlayoutTimestamp(timestamp) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetPlayoutTimeStamp() failed to read playout"
+ " timestamp from the ACM");
+ return -1;
+ }
+
+ WebRtc_UWord16 delayMS(0);
+ if (_audioDeviceModulePtr->PlayoutDelay(&delayMS) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetPlayoutTimeStamp() failed to read playout"
+ " delay from the ADM");
+ return -1;
+ }
+
+ WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency();
+ if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
+ {
+ if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
+ {
+ playoutFrequency = 8000;
+ }
+ }
+ timestamp -= (delayMS * (playoutFrequency/1000));
+
+ playoutTimestamp = timestamp;
+
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::GetPlayoutTimeStamp() => playoutTimestamp = %lu",
+ playoutTimestamp);
+ return 0;
+}
+
+void
+Channel::ResetDeadOrAliveCounters()
+{
+ _countDeadDetections = 0;
+ _countAliveDetections = 0;
+}
+
+void
+Channel::UpdateDeadOrAliveCounters(bool alive)
+{
+ if (alive)
+ _countAliveDetections++;
+ else
+ _countDeadDetections++;
+}
+
+int
+Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const
+{
+ bool enabled;
+ WebRtc_UWord8 timeSec;
+
+ _rtpRtcpModule.PeriodicDeadOrAliveStatus(enabled, timeSec);
+ if (!enabled)
+ return (-1);
+
+ countDead = static_cast<int> (_countDeadDetections);
+ countAlive = static_cast<int> (_countAliveDetections);
+ return 0;
+}
+
+WebRtc_Word32
+Channel::SendPacketRaw(const void *data, int len, bool RTCP)
+{
+ if (_transportPtr == NULL)
+ {
+ return -1;
+ }
+ if (!RTCP)
+ {
+ return _transportPtr->SendPacket(_channelId, data, len);
+ }
+ else
+ {
+ return _transportPtr->SendRTCPPacket(_channelId, data, len);
+ }
+}
+
+WebRtc_Word32
+Channel::UpdatePacketDelay(const WebRtc_UWord32 timestamp,
+ const WebRtc_UWord16 sequenceNumber)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
+ timestamp, sequenceNumber);
+
+ WebRtc_Word32 rtpReceiveFrequency(0);
+
+ // Get frequency of last received payload
+ rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency();
+
+ CodecInst currRecCodec;
+ if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
+ {
+ if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
+ {
+ // Even though the actual sampling rate for G.722 audio is
+ // 16,000 Hz, the RTP clock rate for the G722 payload format is
+ // 8,000 Hz because that value was erroneously assigned in
+ // RFC 1890 and must remain unchanged for backward compatibility.
+ rtpReceiveFrequency = 8000;
+ }
+ }
+
+ const WebRtc_UWord32 timeStampDiff = timestamp - _playoutTimeStampRTP;
+ WebRtc_UWord32 timeStampDiffMs(0);
+
+ if (timeStampDiff > 0)
+ {
+ switch (rtpReceiveFrequency)
+ {
+ case 8000:
+ timeStampDiffMs = timeStampDiff >> 3;
+ break;
+ case 16000:
+ timeStampDiffMs = timeStampDiff >> 4;
+ break;
+ case 32000:
+ timeStampDiffMs = timeStampDiff >> 5;
+ break;
+ default:
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::UpdatePacketDelay() invalid sample "
+ "rate");
+ timeStampDiffMs = 0;
+ return -1;
+ }
+ if (timeStampDiffMs > 5000)
+ {
+ timeStampDiffMs = 0;
+ }
+
+ if (_averageDelayMs == 0)
+ {
+ _averageDelayMs = timeStampDiffMs;
+ }
+ else
+ {
+ // Filter average delay value using exponential filter (alpha is
+ // 7/8). We derive 10*_averageDelayMs here (reduces risk of
+ // rounding error) and compensate for it in GetDelayEstimate()
+ // later. Adding 4/8 results in correct rounding.
+ _averageDelayMs = ((_averageDelayMs*7 + 10*timeStampDiffMs + 4)>>3);
+ }
+
+ if (sequenceNumber - _previousSequenceNumber == 1)
+ {
+ WebRtc_UWord16 packetDelayMs = 0;
+ switch (rtpReceiveFrequency)
+ {
+ case 8000:
+ packetDelayMs = (WebRtc_UWord16)(
+ (timestamp - _previousTimestamp) >> 3);
+ break;
+ case 16000:
+ packetDelayMs = (WebRtc_UWord16)(
+ (timestamp - _previousTimestamp) >> 4);
+ break;
+ case 32000:
+ packetDelayMs = (WebRtc_UWord16)(
+ (timestamp - _previousTimestamp) >> 5);
+ break;
+ }
+
+ if (packetDelayMs >= 10 && packetDelayMs <= 60)
+ _recPacketDelayMs = packetDelayMs;
+ }
+ }
+
+ _previousSequenceNumber = sequenceNumber;
+ _previousTimestamp = timestamp;
+
+ return 0;
+}
+
+void
+Channel::RegisterReceiveCodecsToRTPModule()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::RegisterReceiveCodecsToRTPModule()");
+
+
+ CodecInst codec;
+ const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
+
+ for (int idx = 0; idx < nSupportedCodecs; idx++)
+ {
+ // Open up the RTP/RTCP receiver for all supported codecs
+ if ((_audioCodingModule.Codec(idx, codec) == -1) ||
+ (_rtpRtcpModule.RegisterReceivePayload(codec) == -1))
+ {
+ WEBRTC_TRACE(
+ kTraceWarning,
+ kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::RegisterReceiveCodecsToRTPModule() unable"
+ " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
+ codec.plname, codec.pltype, codec.plfreq,
+ codec.channels, codec.rate);
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::RegisterReceiveCodecsToRTPModule() %s "
+ "(%d/%d/%d/%d) has been added to the RTP/RTCP "
+ "receiver",
+ codec.plname, codec.pltype, codec.plfreq,
+ codec.channels, codec.rate);
+ }
+ }
+}
+
+int
+Channel::ApmProcessRx(AudioFrame& audioFrame)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ApmProcessRx()");
+
+ // Reset the APM frequency if the frequency has changed
+ if (_rxAudioProcessingModulePtr->sample_rate_hz() !=
+ audioFrame._frequencyInHz)
+ {
+ if (_rxAudioProcessingModulePtr->set_sample_rate_hz(
+ audioFrame._frequencyInHz) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "AudioProcessingModule::set_sample_rate_hz("
+ "_frequencyInHz=%u) => error",
+ _audioFrame._frequencyInHz);
+ }
+ }
+
+ if (_rxAudioProcessingModulePtr->ProcessStream(&audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "AudioProcessingModule::ProcessStream() => error");
+ }
+
+ return 0;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel.h b/trunk/src/voice_engine/main/source/channel.h
new file mode 100644
index 0000000..c9d5b82
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel.h
@@ -0,0 +1,659 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H
+#define WEBRTC_VOICE_ENGINE_CHANNEL_H
+
+#include "audio_coding_module.h"
+#include "audio_conference_mixer_defines.h"
+#include "common_types.h"
+#include "dtmf_inband.h"
+#include "dtmf_inband_queue.h"
+#include "file_player.h"
+#include "file_recorder.h"
+#include "level_indicator.h"
+#include "resampler.h"
+#include "rtp_rtcp.h"
+#include "scoped_ptr.h"
+#include "shared_data.h"
+#include "voe_audio_processing.h"
+#include "voe_network.h"
+#include "voice_engine_defines.h"
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+#include "udp_transport.h"
+#endif
+#ifdef WEBRTC_SRTP
+#include "SrtpModule.h"
+#endif
+#ifdef WEBRTC_DTMF_DETECTION
+#include "voe_dtmf.h" // TelephoneEventDetectionMethods, TelephoneEventObserver
+#endif
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class ProcessThread;
+class AudioDeviceModule;
+class RtpRtcp;
+class FileWrapper;
+class RtpDump;
+class VoiceEngineObserver;
+class VoEMediaProcess;
+class VoERTPObserver;
+class VoERTCPObserver;
+
+struct CallStatistics;
+
+namespace voe
+{
+class Statistics;
+class TransmitMixer;
+class OutputMixer;
+
+
+class Channel:
+ public RtpData,
+ public RtpFeedback,
+ public RtcpFeedback,
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ public UdpTransportData, // receiving packet from sockets
+#endif
+ public FileCallback, // receiving notification from file player & recorder
+ public Transport,
+ public RtpAudioFeedback,
+ public AudioPacketizationCallback, // receive encoded packets from the ACM
+ public ACMVADCallback, // receive voice activity from the ACM
+#ifdef WEBRTC_DTMF_DETECTION
+ public AudioCodingFeedback, // inband Dtmf detection in the ACM
+#endif
+ public MixerParticipant // supplies output mixer with audio frames
+{
+public:
+ enum {KNumSocketThreads = 1};
+ enum {KNumberOfSocketBuffers = 8};
+public:
+ virtual ~Channel();
+ static WebRtc_Word32 CreateChannel(Channel*& channel,
+ const WebRtc_Word32 channelId,
+ const WebRtc_UWord32 instanceId);
+ Channel(const WebRtc_Word32 channelId, const WebRtc_UWord32 instanceId);
+ WebRtc_Word32 Init();
+ WebRtc_Word32 SetEngineInformation(
+ Statistics& engineStatistics,
+ OutputMixer& outputMixer,
+ TransmitMixer& transmitMixer,
+ ProcessThread& moduleProcessThread,
+ AudioDeviceModule& audioDeviceModule,
+ VoiceEngineObserver* voiceEngineObserver,
+ CriticalSectionWrapper* callbackCritSect);
+ WebRtc_Word32 UpdateLocalTimeStamp();
+
+public:
+ // API methods
+
+ // VoEBase
+ WebRtc_Word32 StartPlayout();
+ WebRtc_Word32 StopPlayout();
+ WebRtc_Word32 StartSend();
+ WebRtc_Word32 StopSend();
+ WebRtc_Word32 StartReceiving();
+ WebRtc_Word32 StopReceiving();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtpPort,
+ const WebRtc_UWord16 rtcpPort,
+ const WebRtc_Word8 ipAddr[64],
+ const WebRtc_Word8 multicastIpAddr[64]);
+ WebRtc_Word32 GetLocalReceiver(int& port, int& RTCPport, char ipAddr[]);
+ WebRtc_Word32 SetSendDestination(const WebRtc_UWord16 rtpPort,
+ const WebRtc_Word8 ipAddr[64],
+ const int sourcePort,
+ const WebRtc_UWord16 rtcpPort);
+ WebRtc_Word32 GetSendDestination(int& port, char ipAddr[64],
+ int& sourcePort, int& RTCPport);
+#endif
+ WebRtc_Word32 SetNetEQPlayoutMode(NetEqModes mode);
+ WebRtc_Word32 GetNetEQPlayoutMode(NetEqModes& mode);
+ WebRtc_Word32 SetNetEQBGNMode(NetEqBgnModes mode);
+ WebRtc_Word32 GetNetEQBGNMode(NetEqBgnModes& mode);
+ WebRtc_Word32 SetOnHoldStatus(bool enable, OnHoldModes mode);
+ WebRtc_Word32 GetOnHoldStatus(bool& enabled, OnHoldModes& mode);
+ WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
+ WebRtc_Word32 DeRegisterVoiceEngineObserver();
+
+ // VoECodec
+ WebRtc_Word32 GetSendCodec(CodecInst& codec);
+ WebRtc_Word32 GetRecCodec(CodecInst& codec);
+ WebRtc_Word32 SetSendCodec(const CodecInst& codec);
+ WebRtc_Word32 SetVADStatus(bool enableVAD, ACMVADMode mode,
+ bool disableDTX);
+ WebRtc_Word32 GetVADStatus(bool& enabledVAD, ACMVADMode& mode,
+ bool& disabledDTX);
+ WebRtc_Word32 SetRecPayloadType(const CodecInst& codec);
+ WebRtc_Word32 GetRecPayloadType(CodecInst& codec);
+ WebRtc_Word32 SetAMREncFormat(AmrMode mode);
+ WebRtc_Word32 SetAMRDecFormat(AmrMode mode);
+ WebRtc_Word32 SetAMRWbEncFormat(AmrMode mode);
+ WebRtc_Word32 SetAMRWbDecFormat(AmrMode mode);
+ WebRtc_Word32 SetSendCNPayloadType(int type, PayloadFrequencies frequency);
+ WebRtc_Word32 SetISACInitTargetRate(int rateBps, bool useFixedFrameSize);
+ WebRtc_Word32 SetISACMaxRate(int rateBps);
+ WebRtc_Word32 SetISACMaxPayloadSize(int sizeBytes);
+
+ // VoENetwork
+ WebRtc_Word32 RegisterExternalTransport(Transport& transport);
+ WebRtc_Word32 DeRegisterExternalTransport();
+ WebRtc_Word32 ReceivedRTPPacket(const WebRtc_Word8* data,
+ WebRtc_Word32 length);
+ WebRtc_Word32 ReceivedRTCPPacket(const WebRtc_Word8* data,
+ WebRtc_Word32 length);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ WebRtc_Word32 GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64]);
+ WebRtc_Word32 EnableIPv6();
+ bool IPv6IsEnabled() const;
+ WebRtc_Word32 SetSourceFilter(int rtpPort, int rtcpPort,
+ const char ipAddr[64]);
+ WebRtc_Word32 GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64]);
+ WebRtc_Word32 SetSendTOS(int DSCP, int priority, bool useSetSockopt);
+ WebRtc_Word32 GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt);
+#if defined(_WIN32)
+ WebRtc_Word32 SetSendGQoS(bool enable, int serviceType, int overrideDSCP);
+ WebRtc_Word32 GetSendGQoS(bool &enabled, int &serviceType,
+ int &overrideDSCP);
+#endif
+#endif
+ WebRtc_Word32 SetPacketTimeoutNotification(bool enable, int timeoutSeconds);
+ WebRtc_Word32 GetPacketTimeoutNotification(bool& enabled,
+ int& timeoutSeconds);
+ WebRtc_Word32 RegisterDeadOrAliveObserver(VoEConnectionObserver& observer);
+ WebRtc_Word32 DeRegisterDeadOrAliveObserver();
+ WebRtc_Word32 SetPeriodicDeadOrAliveStatus(bool enable,
+ int sampleTimeSeconds);
+ WebRtc_Word32 GetPeriodicDeadOrAliveStatus(bool& enabled,
+ int& sampleTimeSeconds);
+ WebRtc_Word32 SendUDPPacket(const void* data, unsigned int length,
+ int& transmittedBytes, bool useRtcpSocket);
+
+ // VoEFile
+ int StartPlayingFileLocally(const char* fileName, const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+ int StartPlayingFileLocally(InStream* stream, const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+ int StopPlayingFileLocally();
+ int IsPlayingFileLocally() const;
+ int ScaleLocalFilePlayout(const float scale);
+ int GetLocalPlayoutPosition(int& positionMs);
+ int StartPlayingFileAsMicrophone(const char* fileName, const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+ int StartPlayingFileAsMicrophone(InStream* stream,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+ int StopPlayingFileAsMicrophone();
+ int IsPlayingFileAsMicrophone() const;
+ int ScaleFileAsMicrophonePlayout(const float scale);
+ int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
+ int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
+ int StopRecordingPlayout();
+
+ void SetMixWithMicStatus(bool mix);
+
+ // VoEExternalMediaProcessing
+ int RegisterExternalMediaProcessing(ProcessingTypes type,
+ VoEMediaProcess& processObject);
+ int DeRegisterExternalMediaProcessing(ProcessingTypes type);
+
+ // VoEVolumeControl
+ int GetSpeechOutputLevel(WebRtc_UWord32& level) const;
+ int GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const;
+ int SetMute(const bool enable);
+ bool Mute() const;
+ int SetOutputVolumePan(float left, float right);
+ int GetOutputVolumePan(float& left, float& right) const;
+ int SetChannelOutputVolumeScaling(float scaling);
+ int GetChannelOutputVolumeScaling(float& scaling) const;
+
+ // VoECallReport
+ void ResetDeadOrAliveCounters();
+ int ResetRTCPStatistics();
+ int GetRoundTripTimeSummary(StatVal& delaysMs) const;
+ int GetDeadOrAliveCounters(int& countDead, int& countAlive) const;
+
+ // VoENetEqStats
+ int GetNetworkStatistics(NetworkStatistics& stats);
+
+ // VoEVideoSync
+ int GetDelayEstimate(int& delayMs) const;
+ int SetMinimumPlayoutDelay(int delayMs);
+ int GetPlayoutTimestamp(unsigned int& timestamp);
+ int SetInitTimestamp(unsigned int timestamp);
+ int SetInitSequenceNumber(short sequenceNumber);
+
+ // VoEVideoSyncExtended
+ int GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const;
+
+ // VoEEncryption
+#ifdef WEBRTC_SRTP
+ int EnableSRTPSend(
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP);
+ int DisableSRTPSend();
+ int EnableSRTPReceive(
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP);
+ int DisableSRTPReceive();
+#endif
+ int RegisterExternalEncryption(Encryption& encryption);
+ int DeRegisterExternalEncryption();
+
+ // VoEDtmf
+ int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs,
+ int attenuationDb, bool playDtmfEvent);
+ int SendTelephoneEventInband(unsigned char eventCode, int lengthMs,
+ int attenuationDb, bool playDtmfEvent);
+ int SetDtmfPlayoutStatus(bool enable);
+ bool DtmfPlayoutStatus() const;
+ int SetSendTelephoneEventPayloadType(unsigned char type);
+ int GetSendTelephoneEventPayloadType(unsigned char& type);
+#ifdef WEBRTC_DTMF_DETECTION
+ int RegisterTelephoneEventDetection(
+ TelephoneEventDetectionMethods detectionMethod,
+ VoETelephoneEventObserver& observer);
+ int DeRegisterTelephoneEventDetection();
+ int GetTelephoneEventDetectionStatus(
+ bool& enabled,
+ TelephoneEventDetectionMethods& detectionMethod);
+#endif
+
+ // VoEAudioProcessingImpl
+ int UpdateRxVadDetection(AudioFrame& audioFrame);
+ int RegisterRxVadObserver(VoERxVadCallback &observer);
+ int DeRegisterRxVadObserver();
+ int VoiceActivityIndicator(int &activity);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ int SetRxAgcStatus(const bool enable, const AgcModes mode);
+ int GetRxAgcStatus(bool& enabled, AgcModes& mode);
+ int SetRxAgcConfig(const AgcConfig config);
+ int GetRxAgcConfig(AgcConfig& config);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NR
+ int SetRxNsStatus(const bool enable, const NsModes mode);
+ int GetRxNsStatus(bool& enabled, NsModes& mode);
+#endif
+
+ // VoERTP_RTCP
+ int RegisterRTPObserver(VoERTPObserver& observer);
+ int DeRegisterRTPObserver();
+ int RegisterRTCPObserver(VoERTCPObserver& observer);
+ int DeRegisterRTCPObserver();
+ int SetLocalSSRC(unsigned int ssrc);
+ int GetLocalSSRC(unsigned int& ssrc);
+ int GetRemoteSSRC(unsigned int& ssrc);
+ int GetRemoteCSRCs(unsigned int arrCSRC[15]);
+ int SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID);
+ int GetRTPAudioLevelIndicationStatus(bool& enable, unsigned char& ID);
+ int SetRTCPStatus(bool enable);
+ int GetRTCPStatus(bool& enabled);
+ int SetRTCP_CNAME(const char cName[256]);
+ int GetRTCP_CNAME(char cName[256]);
+ int GetRemoteRTCP_CNAME(char cName[256]);
+ int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow,
+ unsigned int& timestamp,
+ unsigned int& playoutTimestamp, unsigned int* jitter,
+ unsigned short* fractionLost);
+ int SendApplicationDefinedRTCPPacket(const unsigned char subType,
+ unsigned int name, const char* data,
+ unsigned short dataLengthInBytes);
+ int GetRTPStatistics(unsigned int& averageJitterMs,
+ unsigned int& maxJitterMs,
+ unsigned int& discardedPackets);
+ int GetRTPStatistics(CallStatistics& stats);
+ int SetFECStatus(bool enable, int redPayloadtype);
+ int GetFECStatus(bool& enabled, int& redPayloadtype);
+ int SetRTPKeepaliveStatus(bool enable, unsigned char unknownPayloadType,
+ int deltaTransmitTimeSeconds);
+ int GetRTPKeepaliveStatus(bool& enabled, unsigned char& unknownPayloadType,
+ int& deltaTransmitTimeSeconds);
+ int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
+ int StopRTPDump(RTPDirections direction);
+ bool RTPDumpIsActive(RTPDirections direction);
+ int InsertExtraRTPPacket(unsigned char payloadType, bool markerBit,
+ const char* payloadData,
+ unsigned short payloadSize);
+
+public:
+ // From AudioPacketizationCallback in the ACM
+ WebRtc_Word32 SendData(FrameType frameType,
+ WebRtc_UWord8 payloadType,
+ WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
+ // From ACMVADCallback in the ACM
+ WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
+
+#ifdef WEBRTC_DTMF_DETECTION
+public: // From AudioCodingFeedback in the ACM
+ int IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end);
+#endif
+
+public:
+ WebRtc_Word32 OnRxVadDetected(const int vadDecision);
+
+public:
+ // From RtpData in the RTP/RTCP module
+ WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const WebRtcRTPHeader* rtpHeader);
+
+public:
+ // From RtpFeedback in the RTP/RTCP module
+ WebRtc_Word32 OnInitializeDecoder(
+ const WebRtc_Word32 id,
+ const WebRtc_Word8 payloadType,
+ const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
+
+ void OnPacketTimeout(const WebRtc_Word32 id);
+
+ void OnReceivedPacket(const WebRtc_Word32 id,
+ const RtpRtcpPacketType packetType);
+
+ void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+ const RTPAliveType alive);
+
+ void OnIncomingSSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 SSRC);
+
+ void OnIncomingCSRCChanged(const WebRtc_Word32 id,
+ const WebRtc_UWord32 CSRC, const bool added);
+
+public:
+ // From RtcpFeedback in the RTP/RTCP module
+ void OnApplicationDataReceived(const WebRtc_Word32 id,
+ const WebRtc_UWord8 subType,
+ const WebRtc_UWord32 name,
+ const WebRtc_UWord16 length,
+ const WebRtc_UWord8* data);
+
+public:
+ // From RtpAudioFeedback in the RTP/RTCP module
+ void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const bool endOfEvent);
+
+ void OnPlayTelephoneEvent(const WebRtc_Word32 id,
+ const WebRtc_UWord8 event,
+ const WebRtc_UWord16 lengthMs,
+ const WebRtc_UWord8 volume);
+
+public:
+ // From UdpTransportData in the Socket Transport module
+ void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+ void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const WebRtc_Word8* fromIP,
+ const WebRtc_UWord16 fromPort);
+
+public:
+ // From Transport (called by the RTP/RTCP module)
+ int SendPacket(int /*channel*/, const void *data, int len);
+ int SendRTCPPacket(int /*channel*/, const void *data, int len);
+
+public:
+ // From MixerParticipant
+ WebRtc_Word32 GetAudioFrame(const WebRtc_Word32 id,
+ AudioFrame& audioFrame);
+ WebRtc_Word32 NeededFrequency(const WebRtc_Word32 id);
+
+public:
+ // From MonitorObserver
+ void OnPeriodicProcess();
+
+public:
+ // From FileCallback
+ void PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+ void RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+ void PlayFileEnded(const WebRtc_Word32 id);
+ void RecordFileEnded(const WebRtc_Word32 id);
+
+public:
+ WebRtc_UWord32 InstanceId() const
+ {
+ return _instanceId;
+ }
+ WebRtc_Word32 ChannelId() const
+ {
+ return _channelId;
+ }
+ bool Playing() const
+ {
+ return _playing;
+ }
+ bool Sending() const
+ {
+ // A lock is needed because |_sending| is accessed by both
+ // TransmitMixer::PrepareDemux() and StartSend()/StopSend(), which
+ // are called by different threads.
+ CriticalSectionScoped cs(_callbackCritSect);
+ return _sending;
+ }
+ bool Receiving() const
+ {
+ return _receiving;
+ }
+ bool ExternalTransport() const
+ {
+ return _externalTransport;
+ }
+ bool OutputIsOnHold() const
+ {
+ return _outputIsOnHold;
+ }
+ bool InputIsOnHold() const
+ {
+ return _inputIsOnHold;
+ }
+ RtpRtcp* RtpRtcpModulePtr() const
+ {
+ return &_rtpRtcpModule;
+ }
+ WebRtc_Word8 OutputEnergyLevel() const
+ {
+ return _outputAudioLevel.Level();
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ bool SendSocketsInitialized() const
+ {
+ return _socketTransportModule.SendSocketsInitialized();
+ }
+ bool ReceiveSocketsInitialized() const
+ {
+ return _socketTransportModule.ReceiveSocketsInitialized();
+ }
+#endif
+ WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame);
+ WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);
+ WebRtc_UWord32 EncodeAndSend();
+
+private:
+ int InsertInbandDtmfTone();
+ WebRtc_Word32
+ MixOrReplaceAudioWithFile(const int mixingFrequency);
+ WebRtc_Word32 MixAudioWithFile(AudioFrame& audioFrame,
+ const int mixingFrequency);
+ WebRtc_Word32 GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp);
+ void UpdateDeadOrAliveCounters(bool alive);
+ WebRtc_Word32 SendPacketRaw(const void *data, int len, bool RTCP);
+ WebRtc_Word32 UpdatePacketDelay(const WebRtc_UWord32 timestamp,
+ const WebRtc_UWord16 sequenceNumber);
+ void RegisterReceiveCodecsToRTPModule();
+ int ApmProcessRx(AudioFrame& audioFrame);
+
+private:
+ CriticalSectionWrapper& _fileCritSect;
+ CriticalSectionWrapper& _callbackCritSect;
+ CriticalSectionWrapper& _transmitCritSect;
+ WebRtc_UWord32 _instanceId;
+ WebRtc_Word32 _channelId;
+
+private:
+ RtpRtcp& _rtpRtcpModule;
+ AudioCodingModule& _audioCodingModule;
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ WebRtc_UWord8 _numSocketThreads;
+ UdpTransport& _socketTransportModule;
+#endif
+#ifdef WEBRTC_SRTP
+ SrtpModule& _srtpModule;
+#endif
+ RtpDump& _rtpDumpIn;
+ RtpDump& _rtpDumpOut;
+private:
+ AudioLevel _outputAudioLevel;
+ bool _externalTransport;
+ AudioFrame _audioFrame;
+ WebRtc_UWord8 _audioLevel_dBov;
+ FilePlayer* _inputFilePlayerPtr;
+ FilePlayer* _outputFilePlayerPtr;
+ FileRecorder* _outputFileRecorderPtr;
+ int _inputFilePlayerId;
+ int _outputFilePlayerId;
+ int _outputFileRecorderId;
+ bool _inputFilePlaying;
+ bool _outputFilePlaying;
+ bool _outputFileRecording;
+ DtmfInbandQueue _inbandDtmfQueue;
+ DtmfInband _inbandDtmfGenerator;
+ bool _inputExternalMedia;
+ bool _outputExternalMedia;
+ VoEMediaProcess* _inputExternalMediaCallbackPtr;
+ VoEMediaProcess* _outputExternalMediaCallbackPtr;
+ WebRtc_UWord8* _encryptionRTPBufferPtr;
+ WebRtc_UWord8* _decryptionRTPBufferPtr;
+ WebRtc_UWord8* _encryptionRTCPBufferPtr;
+ WebRtc_UWord8* _decryptionRTCPBufferPtr;
+ WebRtc_UWord32 _timeStamp;
+ WebRtc_UWord8 _sendTelephoneEventPayloadType;
+ WebRtc_UWord32 _playoutTimeStampRTP;
+ WebRtc_UWord32 _playoutTimeStampRTCP;
+ WebRtc_UWord32 _numberOfDiscardedPackets;
+private:
+ // uses
+ Statistics* _engineStatisticsPtr;
+ OutputMixer* _outputMixerPtr;
+ TransmitMixer* _transmitMixerPtr;
+ ProcessThread* _moduleProcessThreadPtr;
+ AudioDeviceModule* _audioDeviceModulePtr;
+ VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
+ CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
+ Transport* _transportPtr; // WebRtc socket or external transport
+ Encryption* _encryptionPtr; // WebRtc SRTP or external encryption
+ scoped_ptr<AudioProcessing> _rtpAudioProc;
+ AudioProcessing* _rxAudioProcessingModulePtr; // far end AudioProcessing
+#ifdef WEBRTC_DTMF_DETECTION
+ VoETelephoneEventObserver* _telephoneEventDetectionPtr;
+#endif
+ VoERxVadCallback* _rxVadObserverPtr;
+ WebRtc_Word32 _oldVadDecision;
+ WebRtc_Word32 _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
+ VoERTPObserver* _rtpObserverPtr;
+ VoERTCPObserver* _rtcpObserverPtr;
+private:
+ // VoEBase
+ bool _outputIsOnHold;
+ bool _externalPlayout;
+ bool _inputIsOnHold;
+ bool _playing;
+ bool _sending;
+ bool _receiving;
+ bool _mixFileWithMicrophone;
+ bool _rtpObserver;
+ bool _rtcpObserver;
+ // VoEVolumeControl
+ bool _mute;
+ float _panLeft;
+ float _panRight;
+ float _outputGain;
+ // VoEEncryption
+ bool _encrypting;
+ bool _decrypting;
+ // VoEDtmf
+ bool _playOutbandDtmfEvent;
+ bool _playInbandDtmfEvent;
+ bool _inbandTelephoneEventDetection;
+ bool _outOfBandTelephoneEventDetecion;
+ // VoeRTP_RTCP
+ WebRtc_UWord8 _extraPayloadType;
+ bool _insertExtraRTPPacket;
+ bool _extraMarkerBit;
+ WebRtc_UWord32 _lastLocalTimeStamp;
+ WebRtc_Word8 _lastPayloadType;
+ bool _includeAudioLevelIndication;
+ // VoENetwork
+ bool _rtpPacketTimedOut;
+ bool _rtpPacketTimeOutIsEnabled;
+ WebRtc_UWord32 _rtpTimeOutSeconds;
+ bool _connectionObserver;
+ VoEConnectionObserver* _connectionObserverPtr;
+ WebRtc_UWord32 _countAliveDetections;
+ WebRtc_UWord32 _countDeadDetections;
+ AudioFrame::SpeechType _outputSpeechType;
+ // VoEVideoSync
+ WebRtc_UWord32 _averageDelayMs;
+ WebRtc_UWord16 _previousSequenceNumber;
+ WebRtc_UWord32 _previousTimestamp;
+ WebRtc_UWord16 _recPacketDelayMs;
+ // VoEAudioProcessing
+ bool _RxVadDetection;
+ bool _rxApmIsEnabled;
+ bool _rxAgcIsEnabled;
+ bool _rxNsIsEnabled;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_CHANNEL_H
diff --git a/trunk/src/voice_engine/main/source/channel_manager.cc b/trunk/src/voice_engine/main/source/channel_manager.cc
new file mode 100644
index 0000000..47cec4a
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel_manager.cc
@@ -0,0 +1,161 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "channel.h"
+#include "channel_manager.h"
+
+namespace webrtc
+{
+
+namespace voe
+{
+
+ChannelManager::ChannelManager(const WebRtc_UWord32 instanceId) :
+ ChannelManagerBase(),
+ _instanceId(instanceId)
+{
+}
+
+ChannelManager::~ChannelManager()
+{
+ ChannelManagerBase::DestroyAllItems();
+}
+
+bool ChannelManager::CreateChannel(WebRtc_Word32& channelId)
+{
+ return ChannelManagerBase::CreateItem(channelId);
+}
+
+WebRtc_Word32 ChannelManager::DestroyChannel(const WebRtc_Word32 channelId)
+{
+ Channel* deleteChannel =
+ static_cast<Channel*> (ChannelManagerBase::RemoveItem(channelId));
+ if (!deleteChannel)
+ {
+ return -1;
+ }
+ delete deleteChannel;
+ return 0;
+}
+
+WebRtc_Word32 ChannelManager::NumOfChannels() const
+{
+ return ChannelManagerBase::NumOfItems();
+}
+
+WebRtc_Word32 ChannelManager::MaxNumOfChannels() const
+{
+ return ChannelManagerBase::MaxNumOfItems();
+}
+
+void* ChannelManager::NewItem(WebRtc_Word32 itemID)
+{
+ Channel* channel;
+ if (Channel::CreateChannel(channel, itemID, _instanceId) == -1)
+ {
+ return NULL;
+ }
+ return static_cast<void*> (channel);
+}
+
+void ChannelManager::DeleteItem(void* item)
+{
+ Channel* deleteItem = static_cast<Channel*> (item);
+ delete deleteItem;
+}
+
+Channel* ChannelManager::GetChannel(const WebRtc_Word32 channelId) const
+{
+ return static_cast<Channel*> (ChannelManagerBase::GetItem(channelId));
+}
+
+void ChannelManager::ReleaseChannel()
+{
+ ChannelManagerBase::ReleaseItem();
+}
+
+void ChannelManager::GetChannelIds(WebRtc_Word32* channelsArray,
+ WebRtc_Word32& numOfChannels) const
+{
+ ChannelManagerBase::GetItemIds(channelsArray, numOfChannels);
+}
+
+void ChannelManager::GetChannels(MapWrapper& channels) const
+{
+ ChannelManagerBase::GetChannels(channels);
+}
+
+ScopedChannel::ScopedChannel(ChannelManager& chManager) :
+ _chManager(chManager),
+ _channelPtr(NULL)
+{
+ // Copy all existing channels to the local map.
+ // It is not possible to utilize the ChannelPtr() API after
+ // this constructor. The intention is that this constructor
+ // is used in combination with the scoped iterator.
+ _chManager.GetChannels(_channels);
+}
+
+ScopedChannel::ScopedChannel(ChannelManager& chManager,
+ WebRtc_Word32 channelId) :
+ _chManager(chManager),
+ _channelPtr(NULL)
+{
+ _channelPtr = _chManager.GetChannel(channelId);
+}
+
+ScopedChannel::~ScopedChannel()
+{
+ if (_channelPtr != NULL || _channels.Size() != 0)
+ {
+ _chManager.ReleaseChannel();
+ }
+
+ // Delete the map
+ while (_channels.Erase(_channels.First()) == 0)
+ ;
+}
+
+Channel* ScopedChannel::ChannelPtr()
+{
+ return _channelPtr;
+}
+
+Channel* ScopedChannel::GetFirstChannel(void*& iterator) const
+{
+ MapItem* it = _channels.First();
+ iterator = (void*) it;
+ if (!it)
+ {
+ return NULL;
+ }
+ return static_cast<Channel*> (it->GetItem());
+}
+
+Channel* ScopedChannel::GetNextChannel(void*& iterator) const
+{
+ MapItem* it = (MapItem*) iterator;
+ if (!it)
+ {
+ iterator = NULL;
+ return NULL;
+ }
+ it = _channels.Next(it);
+ iterator = (void*) it;
+ if (!it)
+ {
+ return NULL;
+ }
+ return static_cast<Channel*> (it->GetItem());
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel_manager.h b/trunk/src/voice_engine/main/source/channel_manager.h
new file mode 100644
index 0000000..6c40ef1
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel_manager.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
+#define WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
+
+#include "channel_manager_base.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+namespace voe
+{
+
+class ScopedChannel;
+class Channel;
+
+class ChannelManager: private ChannelManagerBase
+{
+ friend class ScopedChannel;
+
+public:
+ bool CreateChannel(WebRtc_Word32& channelId);
+
+ WebRtc_Word32 DestroyChannel(const WebRtc_Word32 channelId);
+
+ WebRtc_Word32 MaxNumOfChannels() const;
+
+ WebRtc_Word32 NumOfChannels() const;
+
+ void GetChannelIds(WebRtc_Word32* channelsArray,
+ WebRtc_Word32& numOfChannels) const;
+
+ ChannelManager(const WebRtc_UWord32 instanceId);
+
+ ~ChannelManager();
+
+private:
+ ChannelManager(const ChannelManager&);
+
+ ChannelManager& operator=(const ChannelManager&);
+
+ Channel* GetChannel(const WebRtc_Word32 channelId) const;
+
+ void GetChannels(MapWrapper& channels) const;
+
+ void ReleaseChannel();
+
+ virtual void* NewItem(WebRtc_Word32 itemID);
+
+ virtual void DeleteItem(void* item);
+
+ WebRtc_UWord32 _instanceId;
+};
+
+class ScopedChannel
+{
+public:
+ // Can only be created by the channel manager
+ ScopedChannel(ChannelManager& chManager);
+
+ ScopedChannel(ChannelManager& chManager, WebRtc_Word32 channelId);
+
+ Channel* ChannelPtr();
+
+ Channel* GetFirstChannel(void*& iterator) const;
+
+ Channel* GetNextChannel(void*& iterator) const;
+
+ ~ScopedChannel();
+private:
+ ChannelManager& _chManager;
+ Channel* _channelPtr;
+ MapWrapper _channels;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
diff --git a/trunk/src/voice_engine/main/source/channel_manager_base.cc b/trunk/src/voice_engine/main/source/channel_manager_base.cc
new file mode 100644
index 0000000..ea9938f
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel_manager_base.cc
@@ -0,0 +1,227 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "channel_manager_base.h"
+
+#include "critical_section_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include <cassert>
+
+namespace webrtc
+{
+
+namespace voe
+{
+
+ChannelManagerBase::ChannelManagerBase() :
+ _itemsCritSectPtr(CriticalSectionWrapper::CreateCriticalSection()),
+ _itemsRWLockPtr(RWLockWrapper::CreateRWLock())
+{
+ for (int i = 0; i < KMaxNumberOfItems; i++)
+ {
+ _freeItemIds[i] = true;
+ }
+}
+
+ChannelManagerBase::~ChannelManagerBase()
+{
+ if (_itemsRWLockPtr)
+ {
+ delete _itemsRWLockPtr;
+ _itemsRWLockPtr = NULL;
+ }
+ if (_itemsCritSectPtr)
+ {
+ delete _itemsCritSectPtr;
+ _itemsCritSectPtr = NULL;
+ }
+}
+
+bool ChannelManagerBase::GetFreeItemId(WebRtc_Word32& itemId)
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ WebRtc_Word32 i(0);
+ while (i < KMaxNumberOfItems)
+ {
+ if (_freeItemIds[i])
+ {
+ itemId = i;
+ _freeItemIds[i] = false;
+ return true;
+ }
+ i++;
+ }
+ return false;
+}
+
+void ChannelManagerBase::AddFreeItemId(WebRtc_Word32 itemId)
+{
+ assert(itemId < KMaxNumberOfItems);
+ _freeItemIds[itemId] = true;
+}
+
+void ChannelManagerBase::RemoveFreeItemIds()
+{
+ for (int i = 0; i < KMaxNumberOfItems; i++)
+ {
+ _freeItemIds[i] = false;
+ }
+}
+
+bool ChannelManagerBase::CreateItem(WebRtc_Word32& itemId)
+{
+ _itemsCritSectPtr->Enter();
+ void* itemPtr;
+ itemId = -1;
+ const bool success = GetFreeItemId(itemId);
+ if (!success)
+ {
+ _itemsCritSectPtr->Leave();
+ return false;
+ }
+ itemPtr = NewItem(itemId);
+ if (!itemPtr)
+ {
+ _itemsCritSectPtr->Leave();
+ return false;
+ }
+ _itemsCritSectPtr->Leave();
+ InsertItem(itemId, itemPtr);
+
+ return true;
+}
+
+void ChannelManagerBase::InsertItem(WebRtc_Word32 itemId, void* item)
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ assert(!_items.Find(itemId));
+ _items.Insert(itemId, item);
+}
+
+void*
+ChannelManagerBase::RemoveItem(WebRtc_Word32 itemId)
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ WriteLockScoped wlock(*_itemsRWLockPtr);
+ MapItem* it = _items.Find(itemId);
+ if (!it)
+ {
+ return 0;
+ }
+ void* returnItem = it->GetItem();
+ _items.Erase(it);
+ AddFreeItemId(itemId);
+
+ return returnItem;
+}
+
+void ChannelManagerBase::DestroyAllItems()
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ MapItem* it = _items.First();
+ while (it)
+ {
+ DeleteItem(it->GetItem());
+ _items.Erase(it);
+ it = _items.First();
+ }
+ RemoveFreeItemIds();
+}
+
+WebRtc_Word32 ChannelManagerBase::NumOfItems() const
+{
+ return _items.Size();
+}
+
+WebRtc_Word32 ChannelManagerBase::MaxNumOfItems() const
+{
+ return static_cast<WebRtc_Word32> (KMaxNumberOfItems);
+}
+
+void*
+ChannelManagerBase::GetItem(WebRtc_Word32 itemId) const
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ MapItem* it = _items.Find(itemId);
+ if (!it)
+ {
+ return 0;
+ }
+ _itemsRWLockPtr->AcquireLockShared();
+ return it->GetItem();
+}
+
+void*
+ChannelManagerBase::GetFirstItem(void*& iterator) const
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ MapItem* it = _items.First();
+ iterator = (void*) it;
+ if (!it)
+ {
+ return 0;
+ }
+ return it->GetItem();
+}
+
+void*
+ChannelManagerBase::GetNextItem(void*& iterator) const
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ MapItem* it = (MapItem*) iterator;
+ if (!it)
+ {
+ iterator = 0;
+ return 0;
+ }
+ it = _items.Next(it);
+ iterator = (void*) it;
+ if (!it)
+ {
+ return 0;
+ }
+ return it->GetItem();
+}
+
+void ChannelManagerBase::ReleaseItem()
+{
+ _itemsRWLockPtr->ReleaseLockShared();
+}
+
+void ChannelManagerBase::GetItemIds(WebRtc_Word32* channelsArray,
+ WebRtc_Word32& numOfChannels) const
+{
+ MapItem* it = _items.First();
+ numOfChannels = (numOfChannels <= _items.Size()) ?
+ numOfChannels : _items.Size();
+ for (int i = 0; i < numOfChannels && it != NULL; i++)
+ {
+ channelsArray[i] = it->GetId();
+ it = _items.Next(it);
+ }
+}
+
+void ChannelManagerBase::GetChannels(MapWrapper& channels) const
+{
+ CriticalSectionScoped cs(*_itemsCritSectPtr);
+ if (_items.Size() == 0)
+ {
+ return;
+ }
+ _itemsRWLockPtr->AcquireLockShared();
+ for (MapItem* it = _items.First(); it != NULL; it = _items.Next(it))
+ {
+ channels.Insert(it->GetId(), it->GetItem());
+ }
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel_manager_base.h b/trunk/src/voice_engine/main/source/channel_manager_base.h
new file mode 100644
index 0000000..0831e43
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel_manager_base.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
+#define WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
+
+#include "typedefs.h"
+#include "map_wrapper.h"
+#include "voice_engine_defines.h"
+
+namespace webrtc
+{
+class CriticalSectionWrapper;
+class RWLockWrapper;
+
+namespace voe
+{
+
+class ScopedChannel;
+class Channel;
+
+class ChannelManagerBase
+{
+public:
+ enum {KMaxNumberOfItems = kVoiceEngineMaxNumOfChannels};
+
+protected:
+ bool CreateItem(WebRtc_Word32& itemId);
+
+ void InsertItem(WebRtc_Word32 itemId, void* item);
+
+ void* RemoveItem(WebRtc_Word32 itemId);
+
+ void* GetItem(WebRtc_Word32 itemId) const;
+
+ void* GetFirstItem(void*& iterator) const ;
+
+ void* GetNextItem(void*& iterator) const;
+
+ void ReleaseItem();
+
+ void AddFreeItemId(WebRtc_Word32 itemId);
+
+ bool GetFreeItemId(WebRtc_Word32& itemId);
+
+ void RemoveFreeItemIds();
+
+ void DestroyAllItems();
+
+ WebRtc_Word32 NumOfItems() const;
+
+ WebRtc_Word32 MaxNumOfItems() const;
+
+ void GetItemIds(WebRtc_Word32* channelsArray,
+ WebRtc_Word32& numOfChannels) const;
+
+ void GetChannels(MapWrapper& channels) const;
+
+ virtual void* NewItem(WebRtc_Word32 itemId) = 0;
+
+ virtual void DeleteItem(void* item) = 0;
+
+ ChannelManagerBase();
+
+ virtual ~ChannelManagerBase();
+
+private:
+ // Protects _items and _freeItemIds
+ CriticalSectionWrapper* _itemsCritSectPtr;
+
+ MapWrapper _items;
+
+ bool _freeItemIds[KMaxNumberOfItems];
+
+ // Protects channels from being destroyed while being used
+ RWLockWrapper* _itemsRWLockPtr;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
diff --git a/trunk/src/voice_engine/main/source/channel_unittest.cc b/trunk/src/voice_engine/main/source/channel_unittest.cc
new file mode 100644
index 0000000..fc78679
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/channel_unittest.cc
@@ -0,0 +1,15 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "channel.h"
+#include "gtest/gtest.h"
+
+// Empty test just to get coverage metrics.
+TEST(ChannelTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband.cc b/trunk/src/voice_engine/main/source/dtmf_inband.cc
new file mode 100644
index 0000000..473af10
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/dtmf_inband.cc
@@ -0,0 +1,389 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dtmf_inband.h"
+
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include <cassert>
+
+namespace webrtc {
+
+const WebRtc_Word16 Dtmf_a_times2Tab8Khz[8]=
+{
+ 27978, 26956, 25701, 24219,
+ 19073, 16325, 13085, 9314
+};
+
+const WebRtc_Word16 Dtmf_a_times2Tab16Khz[8]=
+{
+ 31548, 31281, 30951, 30556,
+ 29144, 28361, 27409, 26258
+};
+
+const WebRtc_Word16 Dtmf_a_times2Tab32Khz[8]=
+{
+ 32462,32394, 32311, 32210, 31849, 31647, 31400, 31098
+};
+
+// Second table is sin(2*pi*f/fs) in Q14
+
+const WebRtc_Word16 Dtmf_ym2Tab8Khz[8]=
+{
+ 8527, 9315, 10163, 11036,
+ 13322, 14206, 15021, 15708
+};
+
+const WebRtc_Word16 Dtmf_ym2Tab16Khz[8]=
+{
+ 4429, 4879, 5380, 5918,
+ 7490, 8207, 8979, 9801
+};
+
+const WebRtc_Word16 Dtmf_ym2Tab32Khz[8]=
+{
+ 2235, 2468, 2728, 3010, 3853, 4249, 4685, 5164
+};
+
+const WebRtc_Word16 Dtmf_dBm0kHz[37]=
+{
+ 16141, 14386, 12821, 11427, 10184, 9077,
+ 8090, 7210, 6426, 5727, 5104, 4549,
+ 4054, 3614, 3221, 2870, 2558, 2280,
+ 2032, 1811, 1614, 1439, 1282, 1143,
+ 1018, 908, 809, 721, 643, 573,
+ 510, 455, 405, 361, 322, 287,
+ 256
+};
+
+
+DtmfInband::DtmfInband(const WebRtc_Word32 id) :
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _id(id),
+ _outputFrequencyHz(8000),
+ _frameLengthSamples(0),
+ _remainingSamples(0),
+ _eventCode(0),
+ _attenuationDb(0),
+ _lengthMs(0),
+ _reinit(true),
+ _playing(false),
+ _delaySinceLastToneMS(1000)
+{
+ memset(_oldOutputLow, 0, sizeof(_oldOutputLow));
+ memset(_oldOutputHigh, 0, sizeof(_oldOutputHigh));
+}
+
+DtmfInband::~DtmfInband()
+{
+ delete &_critSect;
+}
+
+int
+DtmfInband::SetSampleRate(const WebRtc_UWord16 frequency)
+{
+ if (frequency != 8000 &&
+ frequency != 16000 &&
+ frequency != 32000)
+ {
+ // invalid sample rate
+ assert(false);
+ return -1;
+ }
+ _outputFrequencyHz = frequency;
+ return 0;
+}
+
+int
+DtmfInband::GetSampleRate(WebRtc_UWord16& frequency)
+{
+ frequency = _outputFrequencyHz;
+ return 0;
+}
+
+void
+DtmfInband::Init()
+{
+ _remainingSamples = 0;
+ _frameLengthSamples = 0;
+ _eventCode = 0;
+ _attenuationDb = 0;
+ _lengthMs = 0;
+ _reinit = true;
+ _oldOutputLow[0] = 0;
+ _oldOutputLow[1] = 0;
+ _oldOutputHigh[0] = 0;
+ _oldOutputHigh[1] = 0;
+ _delaySinceLastToneMS = 1000;
+}
+
+int
+DtmfInband::AddTone(const WebRtc_UWord8 eventCode,
+ WebRtc_Word32 lengthMs,
+ WebRtc_Word32 attenuationDb)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (attenuationDb > 36 || eventCode > 15)
+ {
+ assert(false);
+ return -1;
+ }
+
+ if (IsAddingTone())
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
+ "DtmfInband::AddTone() new tone interrupts ongoing tone");
+ }
+
+ ReInit();
+
+ _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
+ _eventCode = static_cast<WebRtc_Word16> (eventCode);
+ _attenuationDb = static_cast<WebRtc_Word16> (attenuationDb);
+ _remainingSamples = static_cast<WebRtc_Word32>
+ (lengthMs * (_outputFrequencyHz / 1000));
+ _lengthMs = lengthMs;
+
+ return 0;
+}
+
+int
+DtmfInband::ResetTone()
+{
+ CriticalSectionScoped lock(_critSect);
+
+ ReInit();
+
+ _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
+ _remainingSamples = static_cast<WebRtc_Word32>
+ (_lengthMs * (_outputFrequencyHz / 1000));
+
+ return 0;
+}
+
+int
+DtmfInband::StartTone(const WebRtc_UWord8 eventCode,
+ WebRtc_Word32 attenuationDb)
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (attenuationDb > 36 || eventCode > 15)
+ {
+ assert(false);
+ return -1;
+ }
+
+ if (IsAddingTone())
+ {
+ return -1;
+ }
+
+ ReInit();
+
+ _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
+ _eventCode = static_cast<WebRtc_Word16> (eventCode);
+ _attenuationDb = static_cast<WebRtc_Word16> (attenuationDb);
+ _playing = true;
+
+ return 0;
+}
+
+int
+DtmfInband::StopTone()
+{
+ CriticalSectionScoped lock(_critSect);
+
+ if (!_playing)
+ {
+ return 0;
+ }
+
+ _playing = false;
+
+ return 0;
+}
+
+// Shall be called between tones
+void
+DtmfInband::ReInit()
+{
+ _reinit = true;
+}
+
+bool
+DtmfInband::IsAddingTone()
+{
+ CriticalSectionScoped lock(_critSect);
+ return (_remainingSamples > 0 || _playing);
+}
+
+int
+DtmfInband::Get10msTone(WebRtc_Word16 output[320],
+ WebRtc_UWord16& outputSizeInSamples)
+{
+ CriticalSectionScoped lock(_critSect);
+ if (DtmfFix_generate(output,
+ _eventCode,
+ _attenuationDb,
+ _frameLengthSamples,
+ _outputFrequencyHz) == -1)
+ {
+ return -1;
+ }
+ _remainingSamples -= _frameLengthSamples;
+ outputSizeInSamples = _frameLengthSamples;
+ _delaySinceLastToneMS = 0;
+ return 0;
+}
+
+void
+DtmfInband::UpdateDelaySinceLastTone()
+{
+ _delaySinceLastToneMS += kDtmfFrameSizeMs;
+ // avoid wraparound
+ if (_delaySinceLastToneMS > (1<<30))
+ {
+ _delaySinceLastToneMS = 1000;
+ }
+}
+
+WebRtc_UWord32
+DtmfInband::DelaySinceLastTone() const
+{
+ return _delaySinceLastToneMS;
+}
+
+WebRtc_Word16
+DtmfInband::DtmfFix_generate(WebRtc_Word16 *decoded,
+ const WebRtc_Word16 value,
+ const WebRtc_Word16 volume,
+ const WebRtc_Word16 frameLen,
+ const WebRtc_Word16 fs)
+{
+ const WebRtc_Word16 *a_times2Tbl;
+ const WebRtc_Word16 *y2_Table;
+ WebRtc_Word16 a1_times2 = 0, a2_times2 = 0;
+
+ if (fs==8000) {
+ a_times2Tbl=Dtmf_a_times2Tab8Khz;
+ y2_Table=Dtmf_ym2Tab8Khz;
+ } else if (fs==16000) {
+ a_times2Tbl=Dtmf_a_times2Tab16Khz;
+ y2_Table=Dtmf_ym2Tab16Khz;
+ } else if (fs==32000) {
+ a_times2Tbl=Dtmf_a_times2Tab32Khz;
+ y2_Table=Dtmf_ym2Tab32Khz;
+ } else {
+ return(-1);
+ }
+
+ if ((value==1)||(value==2)||(value==3)||(value==12)) {
+ a1_times2=a_times2Tbl[0];
+ if (_reinit) {
+ _oldOutputLow[0]=y2_Table[0];
+ _oldOutputLow[1]=0;
+ }
+ } else if ((value==4)||(value==5)||(value==6)||(value==13)) {
+ a1_times2=a_times2Tbl[1];
+ if (_reinit) {
+ _oldOutputLow[0]=y2_Table[1];
+ _oldOutputLow[1]=0;
+ }
+ } else if ((value==7)||(value==8)||(value==9)||(value==14)) {
+ a1_times2=a_times2Tbl[2];
+ if (_reinit) {
+ _oldOutputLow[0]=y2_Table[2];
+ _oldOutputLow[1]=0;
+ }
+ } else if ((value==10)||(value==0)||(value==11)||(value==15)) {
+ a1_times2=a_times2Tbl[3];
+ if (_reinit) {
+ _oldOutputLow[0]=y2_Table[3];
+ _oldOutputLow[1]=0;
+ }
+ }
+ if ((value==1)||(value==4)||(value==7)||(value==10)) {
+ a2_times2=a_times2Tbl[4];
+ if (_reinit) {
+ _oldOutputHigh[0]=y2_Table[4];
+ _oldOutputHigh[1]=0;
+ _reinit=false;
+ }
+ } else if ((value==2)||(value==5)||(value==8)||(value==0)) {
+ a2_times2=a_times2Tbl[5];
+ if (_reinit) {
+ _oldOutputHigh[0]=y2_Table[5];
+ _oldOutputHigh[1]=0;
+ _reinit=false;
+ }
+ } else if ((value==3)||(value==6)||(value==9)||(value==11)) {
+ a2_times2=a_times2Tbl[6];
+ if (_reinit) {
+ _oldOutputHigh[0]=y2_Table[6];
+ _oldOutputHigh[1]=0;
+ _reinit=false;
+ }
+ } else if ((value==12)||(value==13)||(value==14)||(value==15)) {
+ a2_times2=a_times2Tbl[7];
+ if (_reinit) {
+ _oldOutputHigh[0]=y2_Table[7];
+ _oldOutputHigh[1]=0;
+ _reinit=false;
+ }
+ }
+
+ return (DtmfFix_generateSignal(a1_times2,
+ a2_times2,
+ volume,
+ decoded,
+ frameLen));
+}
+
+WebRtc_Word16
+DtmfInband::DtmfFix_generateSignal(const WebRtc_Word16 a1_times2,
+ const WebRtc_Word16 a2_times2,
+ const WebRtc_Word16 volume,
+ WebRtc_Word16 *signal,
+ const WebRtc_Word16 length)
+{
+ int i;
+
+ /* Generate Signal */
+ for (i=0;i<length;i++) {
+ WebRtc_Word32 tempVal;
+ WebRtc_Word16 tempValLow, tempValHigh;
+
+ /* Use recursion formula y[n] = a*2*y[n-1] - y[n-2] */
+ tempValLow = (WebRtc_Word16)(((( (WebRtc_Word32)(a1_times2 *
+ _oldOutputLow[1])) + 8192) >> 14) - _oldOutputLow[0]);
+ tempValHigh = (WebRtc_Word16)(((( (WebRtc_Word32)(a2_times2 *
+ _oldOutputHigh[1])) + 8192) >> 14) - _oldOutputHigh[0]);
+
+ /* Update memory */
+ _oldOutputLow[0]=_oldOutputLow[1];
+ _oldOutputLow[1]=tempValLow;
+ _oldOutputHigh[0]=_oldOutputHigh[1];
+ _oldOutputHigh[1]=tempValHigh;
+
+ tempVal = (WebRtc_Word32)(kDtmfAmpLow * tempValLow) +
+ (WebRtc_Word32)(kDtmfAmpHigh * tempValHigh);
+
+ /* Norm the signal to Q14 */
+ tempVal=(tempVal+16384)>>15;
+
+ /* Scale the signal to correct dbM0 value */
+ signal[i]=(WebRtc_Word16)((tempVal*Dtmf_dBm0kHz[volume]+8192)>>14);
+ }
+
+ return(0);
+}
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband.h b/trunk/src/voice_engine/main/source/dtmf_inband.h
new file mode 100644
index 0000000..806fff0
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/dtmf_inband.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
+#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
+
+#if _MSC_VER > 1000
+#pragma once
+#endif
+
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class DtmfInband
+{
+public:
+ DtmfInband(const WebRtc_Word32 id);
+
+ virtual ~DtmfInband();
+
+ void Init();
+
+ int SetSampleRate(const WebRtc_UWord16 frequency);
+
+ int GetSampleRate(WebRtc_UWord16& frequency);
+
+ int AddTone(const WebRtc_UWord8 eventCode,
+ WebRtc_Word32 lengthMs,
+ WebRtc_Word32 attenuationDb);
+
+ int ResetTone();
+ int StartTone(const WebRtc_UWord8 eventCode,
+ WebRtc_Word32 attenuationDb);
+
+ int StopTone();
+
+ bool IsAddingTone();
+
+ int Get10msTone(WebRtc_Word16 output[320],
+ WebRtc_UWord16& outputSizeInSamples);
+
+ WebRtc_UWord32 DelaySinceLastTone() const;
+
+ void UpdateDelaySinceLastTone();
+
+private:
+ void ReInit();
+ WebRtc_Word16 DtmfFix_generate(WebRtc_Word16* decoded,
+ const WebRtc_Word16 value,
+ const WebRtc_Word16 volume,
+ const WebRtc_Word16 frameLen,
+ const WebRtc_Word16 fs);
+
+private:
+ enum {kDtmfFrameSizeMs = 10};
+ enum {kDtmfAmpHigh = 32768};
+ enum {kDtmfAmpLow = 23171}; // 3 dB lower than the high frequency
+
+ WebRtc_Word16 DtmfFix_generateSignal(const WebRtc_Word16 a1_times2,
+ const WebRtc_Word16 a2_times2,
+ const WebRtc_Word16 volume,
+ WebRtc_Word16* signal,
+ const WebRtc_Word16 length);
+
+private:
+ CriticalSectionWrapper& _critSect;
+ WebRtc_Word32 _id;
+ WebRtc_UWord16 _outputFrequencyHz; // {8000, 16000, 32000}
+ WebRtc_Word16 _oldOutputLow[2]; // Data needed for oscillator model
+ WebRtc_Word16 _oldOutputHigh[2]; // Data needed for oscillator model
+ WebRtc_Word16 _frameLengthSamples; // {80, 160, 320}
+ WebRtc_Word32 _remainingSamples;
+ WebRtc_Word16 _eventCode; // [0, 15]
+ WebRtc_Word16 _attenuationDb; // [0, 36]
+ WebRtc_Word32 _lengthMs;
+ bool _reinit; // 'true' if the oscillator should be reinit for next event
+ bool _playing;
+ WebRtc_UWord32 _delaySinceLastToneMS; // time since last generated tone [ms]
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc b/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc
new file mode 100644
index 0000000..080ef3e
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dtmf_inband_queue.h"
+#include "trace.h"
+
+namespace webrtc {
+
+DtmfInbandQueue::DtmfInbandQueue(const WebRtc_Word32 id):
+ _id(id),
+ _DtmfCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _nextEmptyIndex(0)
+{
+ memset(_DtmfKey,0, sizeof(_DtmfKey));
+ memset(_DtmfLen,0, sizeof(_DtmfLen));
+ memset(_DtmfLevel,0, sizeof(_DtmfLevel));
+}
+
+DtmfInbandQueue::~DtmfInbandQueue()
+{
+ delete &_DtmfCritsect;
+}
+
+int
+DtmfInbandQueue::AddDtmf(WebRtc_UWord8 key,
+ WebRtc_UWord16 len,
+ WebRtc_UWord8 level)
+{
+ CriticalSectionScoped lock(_DtmfCritsect);
+
+ if (_nextEmptyIndex >= kDtmfInbandMax)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
+ "DtmfInbandQueue::AddDtmf() unable to add Dtmf tone");
+ return -1;
+ }
+ WebRtc_Word32 index = _nextEmptyIndex;
+ _DtmfKey[index] = key;
+ _DtmfLen[index] = len;
+ _DtmfLevel[index] = level;
+ _nextEmptyIndex++;
+ return 0;
+}
+
+WebRtc_Word8
+DtmfInbandQueue::NextDtmf(WebRtc_UWord16* len, WebRtc_UWord8* level)
+{
+ CriticalSectionScoped lock(_DtmfCritsect);
+
+ if(!PendingDtmf())
+ {
+ return -1;
+ }
+ WebRtc_Word8 nextDtmf = _DtmfKey[0];
+ *len=_DtmfLen[0];
+ *level=_DtmfLevel[0];
+
+ memmove(&(_DtmfKey[0]), &(_DtmfKey[1]),
+ _nextEmptyIndex*sizeof(WebRtc_UWord8));
+ memmove(&(_DtmfLen[0]), &(_DtmfLen[1]),
+ _nextEmptyIndex*sizeof(WebRtc_UWord16));
+ memmove(&(_DtmfLevel[0]), &(_DtmfLevel[1]),
+ _nextEmptyIndex*sizeof(WebRtc_UWord8));
+
+ _nextEmptyIndex--;
+ return nextDtmf;
+}
+
+bool
+DtmfInbandQueue::PendingDtmf()
+{
+ return(_nextEmptyIndex>0);
+}
+
+void
+DtmfInbandQueue::ResetDtmf()
+{
+ _nextEmptyIndex = 0;
+}
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband_queue.h b/trunk/src/voice_engine/main/source/dtmf_inband_queue.h
new file mode 100644
index 0000000..b3bd39e
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/dtmf_inband_queue.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
+#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
+
+#include "critical_section_wrapper.h"
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+
+
+namespace webrtc {
+
+class DtmfInbandQueue
+{
+public:
+
+ DtmfInbandQueue(const WebRtc_Word32 id);
+
+ virtual ~DtmfInbandQueue();
+
+ int AddDtmf(WebRtc_UWord8 DtmfKey,
+ WebRtc_UWord16 len,
+ WebRtc_UWord8 level);
+
+ WebRtc_Word8 NextDtmf(WebRtc_UWord16* len, WebRtc_UWord8* level);
+
+ bool PendingDtmf();
+
+ void ResetDtmf();
+
+private:
+ enum {kDtmfInbandMax = 20};
+
+ WebRtc_Word32 _id;
+ CriticalSectionWrapper& _DtmfCritsect;
+ WebRtc_UWord8 _nextEmptyIndex;
+ WebRtc_UWord8 _DtmfKey[kDtmfInbandMax];
+ WebRtc_UWord16 _DtmfLen[kDtmfInbandMax];
+ WebRtc_UWord8 _DtmfLevel[kDtmfInbandMax];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
diff --git a/trunk/src/voice_engine/main/source/level_indicator.cc b/trunk/src/voice_engine/main/source/level_indicator.cc
new file mode 100644
index 0000000..89004a5
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/level_indicator.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "level_indicator.h"
+#include "module_common_types.h"
+#include "signal_processing_library.h"
+
+namespace webrtc {
+
+namespace voe {
+
+
+// Number of bars on the indicator.
+// Note that the number of elements is specified because we are indexing it
+// in the range of 0-32
+const WebRtc_Word8 permutation[33] =
+ {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
+
+
+AudioLevel::AudioLevel() :
+ _absMax(0),
+ _count(0),
+ _currentLevel(0),
+ _currentLevelFullRange(0)
+{
+}
+
+AudioLevel::~AudioLevel()
+{
+}
+
+void
+AudioLevel::Clear()
+{
+ _absMax = 0;
+ _count = 0;
+ _currentLevel = 0;
+ _currentLevelFullRange = 0;
+}
+
+void
+AudioLevel::ComputeLevel(const AudioFrame& audioFrame)
+{
+ WebRtc_Word16 absValue(0);
+
+ // Check speech level (works for 2 channels as well)
+ absValue = WebRtcSpl_MaxAbsValueW16(
+ audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples*audioFrame._audioChannel);
+ if (absValue > _absMax)
+ _absMax = absValue;
+
+ // Update level approximately 10 times per second
+ if (_count++ == kUpdateFrequency)
+ {
+ _currentLevelFullRange = _absMax;
+
+ _count = 0;
+
+ // Highest value for a WebRtc_Word16 is 0x7fff = 32767
+ // Divide with 1000 to get in the range of 0-32 which is the range of
+ // the permutation vector
+ WebRtc_Word32 position = _absMax/1000;
+
+ // Make it less likely that the bar stays at position 0. I.e. only if
+ // its in the range 0-250 (instead of 0-1000)
+ if ((position == 0) && (_absMax > 250))
+ {
+ position = 1;
+ }
+ _currentLevel = permutation[position];
+
+ // Decay the absolute maximum (divide by 4)
+ _absMax >>= 2;
+ }
+}
+
+WebRtc_Word8
+AudioLevel::Level() const
+{
+ return _currentLevel;
+}
+
+WebRtc_Word16
+AudioLevel::LevelFullRange() const
+{
+ return _currentLevelFullRange;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/level_indicator.h b/trunk/src/voice_engine/main/source/level_indicator.h
new file mode 100644
index 0000000..564b068
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/level_indicator.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
+#define WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
+
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+
+namespace webrtc {
+
+class AudioFrame;
+namespace voe {
+
+class AudioLevel
+{
+public:
+ AudioLevel();
+ virtual ~AudioLevel();
+
+ void ComputeLevel(const AudioFrame& audioFrame);
+
+ WebRtc_Word8 Level() const;
+
+ WebRtc_Word16 LevelFullRange() const;
+
+ void Clear();
+
+private:
+ enum { kUpdateFrequency = 10};
+
+ WebRtc_Word16 _absMax;
+ WebRtc_Word16 _count;
+ WebRtc_Word8 _currentLevel;
+ WebRtc_Word16 _currentLevelFullRange;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
diff --git a/trunk/src/voice_engine/main/source/monitor_module.cc b/trunk/src/voice_engine/main/source/monitor_module.cc
new file mode 100644
index 0000000..6314f38
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/monitor_module.cc
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "critical_section_wrapper.h"
+#include "monitor_module.h"
+
+namespace webrtc {
+
+namespace voe {
+
+MonitorModule::MonitorModule() :
+ _observerPtr(NULL),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _lastProcessTime(GET_TIME_IN_MS())
+{
+}
+
+MonitorModule::~MonitorModule()
+{
+ delete &_callbackCritSect;
+}
+
+WebRtc_Word32
+MonitorModule::RegisterObserver(MonitorObserver& observer)
+{
+ CriticalSectionScoped lock(_callbackCritSect);
+ if (_observerPtr)
+ {
+ return -1;
+ }
+ _observerPtr = &observer;
+ return 0;
+}
+
+WebRtc_Word32
+MonitorModule::DeRegisterObserver()
+{
+ CriticalSectionScoped lock(_callbackCritSect);
+ if (!_observerPtr)
+ {
+ return 0;
+ }
+ _observerPtr = NULL;
+ return 0;
+}
+
+WebRtc_Word32
+MonitorModule::Version(WebRtc_Word8* version,
+ WebRtc_UWord32& remainingBufferInBytes,
+ WebRtc_UWord32& position) const
+{
+ return 0;
+}
+
+WebRtc_Word32
+MonitorModule::ChangeUniqueId(const WebRtc_Word32 id)
+{
+ return 0;
+}
+
+WebRtc_Word32
+MonitorModule::TimeUntilNextProcess()
+{
+ WebRtc_UWord32 now = GET_TIME_IN_MS();
+ WebRtc_Word32 timeToNext =
+ kAverageProcessUpdateTimeMs - (now - _lastProcessTime);
+ return (timeToNext);
+}
+
+WebRtc_Word32
+MonitorModule::Process()
+{
+ _lastProcessTime = GET_TIME_IN_MS();
+ if (_observerPtr)
+ {
+ CriticalSectionScoped lock(_callbackCritSect);
+ _observerPtr->OnPeriodicProcess();
+ }
+ return 0;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/monitor_module.h b/trunk/src/voice_engine/main/source/monitor_module.h
new file mode 100644
index 0000000..45cf228
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/monitor_module.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
+#define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
+
+#include "module.h"
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+
+class MonitorObserver
+{
+public:
+ virtual void OnPeriodicProcess() = 0;
+protected:
+ virtual ~MonitorObserver() {}
+};
+
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace voe {
+
+class MonitorModule : public Module
+{
+public:
+ WebRtc_Word32 RegisterObserver(MonitorObserver& observer);
+
+ WebRtc_Word32 DeRegisterObserver();
+
+ MonitorModule();
+
+ virtual ~MonitorModule();
+public: // module
+ WebRtc_Word32 Version(WebRtc_Word8* version,
+ WebRtc_UWord32& remainingBufferInBytes,
+ WebRtc_UWord32& position) const;
+
+ WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ WebRtc_Word32 TimeUntilNextProcess();
+
+ WebRtc_Word32 Process();
+private:
+ enum { kAverageProcessUpdateTimeMs = 1000 };
+ MonitorObserver* _observerPtr;
+ CriticalSectionWrapper& _callbackCritSect;
+ WebRtc_Word32 _lastProcessTime;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // VOICE_ENGINE_MONITOR_MODULE
diff --git a/trunk/src/voice_engine/main/source/output_mixer.cc b/trunk/src/voice_engine/main/source/output_mixer.cc
new file mode 100644
index 0000000..9dfe0ad
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/output_mixer.cc
@@ -0,0 +1,754 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "output_mixer.h"
+
+#include "audio_processing.h"
+#include "audio_frame_operations.h"
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "trace.h"
+#include "statistics.h"
+#include "voe_external_media.h"
+
+namespace webrtc {
+
+namespace voe {
+
+void
+OutputMixer::NewMixedAudio(const WebRtc_Word32 id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const WebRtc_UWord32 size)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
+
+ _audioFrame = generalAudioFrame;
+ _audioFrame._id = id;
+}
+
+void OutputMixer::MixedParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::MixedParticipants(id=%d, size=%u)", id, size);
+}
+
+void OutputMixer::VADPositiveParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::VADPositiveParticipants(id=%d, size=%u)",
+ id, size);
+}
+
+void OutputMixer::MixedAudioLevel(const WebRtc_Word32 id,
+ const WebRtc_UWord32 level)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level);
+}
+
+void OutputMixer::PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::PlayNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+ // Not implement yet
+}
+
+void OutputMixer::RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::RecordNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void OutputMixer::PlayFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::PlayFileEnded(id=%d)", id);
+
+ // not needed
+}
+
+void OutputMixer::RecordFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::RecordFileEnded(id=%d)", id);
+ assert(id == _instanceId);
+
+ CriticalSectionScoped cs(_fileCritSect);
+ _outputFileRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::RecordFileEnded() =>"
+ "output file recorder module is shutdown");
+}
+
+WebRtc_Word32
+OutputMixer::Create(OutputMixer*& mixer, const WebRtc_UWord32 instanceId)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
+ "OutputMixer::Create(instanceId=%d)", instanceId);
+ mixer = new OutputMixer(instanceId);
+ if (mixer == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
+ "OutputMixer::Create() unable to allocate memory for"
+ "mixer");
+ return -1;
+ }
+ return 0;
+}
+
+OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _mixerModule(*AudioConferenceMixer::Create(instanceId)),
+ _audioLevel(),
+ _dtmfGenerator(instanceId),
+ _instanceId(instanceId),
+ _externalMediaCallbackPtr(NULL),
+ _externalMedia(false),
+ _panLeft(1.0f),
+ _panRight(1.0f),
+ _mixingFrequencyHz(8000),
+ _outputFileRecorderPtr(NULL),
+ _outputFileRecording(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::OutputMixer() - ctor");
+
+ if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) ||
+ (_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::OutputMixer() failed to register mixer"
+ "callbacks");
+ }
+
+ _dtmfGenerator.Init();
+}
+
+void
+OutputMixer::Destroy(OutputMixer*& mixer)
+{
+ if (mixer)
+ {
+ delete mixer;
+ mixer = NULL;
+ }
+}
+
+OutputMixer::~OutputMixer()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::~OutputMixer() - dtor");
+ if (_externalMedia)
+ {
+ DeRegisterExternalMediaProcessing();
+ }
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+ }
+ _mixerModule.UnRegisterMixerStatusCallback();
+ _mixerModule.UnRegisterMixedStreamCallback();
+ delete &_mixerModule;
+ delete &_callbackCritSect;
+ delete &_fileCritSect;
+}
+
+WebRtc_Word32
+OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::SetEngineInformation()");
+ _engineStatisticsPtr = &engineStatistics;
+ return 0;
+}
+
+WebRtc_Word32
+OutputMixer::SetAudioProcessingModule(
+ AudioProcessing* audioProcessingModule)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::SetAudioProcessingModule("
+ "audioProcessingModule=0x%x)", audioProcessingModule);
+ _audioProcessingModulePtr = audioProcessingModule;
+ return 0;
+}
+
+int OutputMixer::RegisterExternalMediaProcessing(
+ VoEMediaProcess& proccess_object)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::RegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ _externalMediaCallbackPtr = &proccess_object;
+ _externalMedia = true;
+
+ return 0;
+}
+
+int OutputMixer::DeRegisterExternalMediaProcessing()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::DeRegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ _externalMedia = false;
+ _externalMediaCallbackPtr = NULL;
+
+ return 0;
+}
+
+int OutputMixer::PlayDtmfTone(WebRtc_UWord8 eventCode, int lengthMs,
+ int attenuationDb)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "OutputMixer::PlayDtmfTone()");
+ if (_dtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_STILL_PLAYING_PREV_DTMF,
+ kTraceError,
+ "OutputMixer::PlayDtmfTone()");
+ return -1;
+ }
+ return 0;
+}
+
+int OutputMixer::StartPlayingDtmfTone(WebRtc_UWord8 eventCode,
+ int attenuationDb)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "OutputMixer::StartPlayingDtmfTone()");
+ if (_dtmfGenerator.StartTone(eventCode, attenuationDb) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STILL_PLAYING_PREV_DTMF,
+ kTraceError,
+ "OutputMixer::StartPlayingDtmfTone())");
+ return -1;
+ }
+ return 0;
+}
+
+int OutputMixer::StopPlayingDtmfTone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "OutputMixer::StopPlayingDtmfTone()");
+ return (_dtmfGenerator.StopTone());
+}
+
+WebRtc_Word32
+OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
+ const bool mixable)
+{
+ return _mixerModule.SetMixabilityStatus(participant, mixable);
+}
+
+WebRtc_Word32
+OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
+ const bool mixable)
+{
+ return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable);
+}
+
+WebRtc_Word32
+OutputMixer::MixActiveChannels()
+{
+ return _mixerModule.Process();
+}
+
+int
+OutputMixer::GetSpeechOutputLevel(WebRtc_UWord32& level)
+{
+ WebRtc_Word8 currentLevel = _audioLevel.Level();
+ level = static_cast<WebRtc_UWord32> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechOutputLevel() => level=%u", level);
+ return 0;
+}
+
+int
+OutputMixer::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level)
+{
+ WebRtc_Word16 currentLevel = _audioLevel.LevelFullRange();
+ level = static_cast<WebRtc_UWord32> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechOutputLevelFullRange() => level=%u", level);
+ return 0;
+}
+
+int
+OutputMixer::SetOutputVolumePan(float left, float right)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::SetOutputVolumePan()");
+ _panLeft = left;
+ _panRight = right;
+ return 0;
+}
+
+int
+OutputMixer::GetOutputVolumePan(float& left, float& right)
+{
+ left = _panLeft;
+ right = _panRight;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetOutputVolumePan() => left=%2.1f, right=%2.1f",
+ left, right);
+ return 0;
+}
+
+int OutputMixer::StartRecordingPlayout(const char* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::StartRecordingPlayout(fileName=%s)", fileName);
+
+ if (_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0);
+ CodecInst dummyCodec={100,"L16",16000,320,1,320000};
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() invalid compression");
+ return(-1);
+ }
+ if(codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst=&dummyCodec;
+ }
+ else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ }
+ else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+
+ _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
+ _instanceId,
+ (const FileFormats)format);
+ if (_outputFileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_outputFileRecorderPtr->StartRecordingAudioFile(
+ fileName,
+ (const CodecInst&)*codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ return -1;
+ }
+ _outputFileRecorderPtr->RegisterModuleFileCallback(this);
+ _outputFileRecording = true;
+
+ return 0;
+}
+
+int OutputMixer::StartRecordingPlayout(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::StartRecordingPlayout()");
+
+ if (_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0);
+ CodecInst dummyCodec={100,"L16",16000,320,1,320000};
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() invalid compression");
+ return(-1);
+ }
+ if(codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst=&dummyCodec;
+ }
+ else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ }
+ else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ // Destroy the old instance
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ }
+
+ _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
+ _instanceId,
+ (const FileFormats)format);
+ if (_outputFileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingPlayout() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream,
+ *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _outputFileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ return -1;
+ }
+
+ _outputFileRecorderPtr->RegisterModuleFileCallback(this);
+ _outputFileRecording = true;
+
+ return 0;
+}
+
+int OutputMixer::StopRecordingPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::StopRecordingPlayout()");
+
+ if (!_outputFileRecording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StopRecordingPlayout() file isnot recording");
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_fileCritSect);
+
+ if (_outputFileRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording(), could not stop recording");
+ return -1;
+ }
+ _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
+ _outputFileRecorderPtr = NULL;
+ _outputFileRecording = false;
+
+ return 0;
+}
+
+WebRtc_Word32
+OutputMixer::GetMixedAudio(const WebRtc_Word32 desiredFreqHz,
+ const WebRtc_UWord8 channels,
+ AudioFrame& audioFrame)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::GetMixedAudio(desiredFreqHz=%d, channels=&d)",
+ desiredFreqHz, channels);
+
+ audioFrame = _audioFrame;
+
+ // --- Record playout if enabled
+ {
+ CriticalSectionScoped cs(_fileCritSect);
+ if (_outputFileRecording)
+ {
+ assert(audioFrame._audioChannel == 1);
+
+ if (_outputFileRecorderPtr)
+ {
+ _outputFileRecorderPtr->RecordAudioToFile(audioFrame);
+ }
+ }
+ }
+
+ int outLen(0);
+
+ if (audioFrame._audioChannel == 1)
+ {
+ if (_resampler.ResetIfNeeded(audioFrame._frequencyInHz,
+ desiredFreqHz,
+ kResamplerSynchronous) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::GetMixedAudio() unable to resample - 1");
+ return -1;
+ }
+ }
+ else
+ {
+ if (_resampler.ResetIfNeeded(audioFrame._frequencyInHz,
+ desiredFreqHz,
+ kResamplerSynchronousStereo) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::GetMixedAudio() unable to resample - 2");
+ return -1;
+ }
+ }
+ if (_resampler.Push(
+ _audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples*_audioFrame._audioChannel,
+ audioFrame._payloadData,
+ AudioFrame::kMaxAudioFrameSizeSamples,
+ outLen) == 0)
+ {
+ // Ensure that output from resampler matches the audio-frame format.
+ // Example: 10ms stereo output at 48kHz => outLen = 960 =>
+ // convert _payloadDataLengthInSamples to 480
+ audioFrame._payloadDataLengthInSamples =
+ (outLen / _audioFrame._audioChannel);
+ audioFrame._frequencyInHz = desiredFreqHz;
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::GetMixedAudio() resampling failed");
+ return -1;
+ }
+
+ if ((channels == 2) && (audioFrame._audioChannel == 1))
+ {
+ AudioFrameOperations::MonoToStereo(audioFrame);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+OutputMixer::DoOperationsOnCombinedSignal()
+{
+ if (_audioFrame._frequencyInHz != _mixingFrequencyHz)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "OutputMixer::DoOperationsOnCombinedSignal() => "
+ "mixing frequency = %d", _audioFrame._frequencyInHz);
+ _mixingFrequencyHz = _audioFrame._frequencyInHz;
+ }
+
+ // --- Insert inband Dtmf tone
+ if (_dtmfGenerator.IsAddingTone())
+ {
+ InsertInbandDtmfTone();
+ }
+
+ // Scale left and/or right channel(s) if balance is active
+ if (_panLeft != 1.0 || _panRight != 1.0)
+ {
+ if (_audioFrame._audioChannel == 1)
+ {
+ AudioFrameOperations::MonoToStereo(_audioFrame);
+ }
+ else
+ {
+ // Pure stereo mode (we are receiving a stereo signal).
+ }
+
+ assert(_audioFrame._audioChannel == 2);
+ AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
+ }
+
+ // --- Far-end Voice Quality Enhancement (AudioProcessing Module)
+
+ APMAnalyzeReverseStream();
+
+ // --- External media processing
+
+ if (_externalMedia)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ const bool isStereo = (_audioFrame._audioChannel == 2);
+ if (_externalMediaCallbackPtr)
+ {
+ _externalMediaCallbackPtr->Process(
+ -1,
+ kPlaybackAllChannelsMixed,
+ (WebRtc_Word16*)_audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples,
+ _audioFrame._frequencyInHz,
+ isStereo);
+ }
+ }
+
+ // --- Measure audio level (0-9) for the combined signal
+ _audioLevel.ComputeLevel(_audioFrame);
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// Private methods
+// ----------------------------------------------------------------------------
+
+int
+OutputMixer::APMAnalyzeReverseStream()
+{
+ int outLen(0);
+ AudioFrame audioFrame = _audioFrame;
+
+ // Convert from mixing frequency to APM frequency.
+ // Sending side determines APM frequency.
+
+ if (audioFrame._audioChannel == 1)
+ {
+ _apmResampler.ResetIfNeeded(_audioFrame._frequencyInHz,
+ _audioProcessingModulePtr->sample_rate_hz(),
+ kResamplerSynchronous);
+ }
+ else
+ {
+ _apmResampler.ResetIfNeeded(_audioFrame._frequencyInHz,
+ _audioProcessingModulePtr->sample_rate_hz(),
+ kResamplerSynchronousStereo);
+ }
+ if (_apmResampler.Push(
+ _audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples*_audioFrame._audioChannel,
+ audioFrame._payloadData,
+ AudioFrame::kMaxAudioFrameSizeSamples,
+ outLen) == 0)
+ {
+ audioFrame._payloadDataLengthInSamples =
+ (outLen / _audioFrame._audioChannel);
+ audioFrame._frequencyInHz = _audioProcessingModulePtr->sample_rate_hz();
+ }
+
+ if (audioFrame._audioChannel == 2)
+ {
+ AudioFrameOperations::StereoToMono(audioFrame);
+ }
+
+ // Perform far-end APM analyze
+
+ if (_audioProcessingModulePtr->AnalyzeReverseStream(&audioFrame) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
+ "AudioProcessingModule::AnalyzeReverseStream() => error");
+ }
+
+ return 0;
+}
+
+int
+OutputMixer::InsertInbandDtmfTone()
+{
+ WebRtc_UWord16 sampleRate(0);
+ _dtmfGenerator.GetSampleRate(sampleRate);
+ if (sampleRate != _audioFrame._frequencyInHz)
+ {
+ // Update sample rate of Dtmf tone since the mixing frequency changed.
+ _dtmfGenerator.SetSampleRate(
+ (WebRtc_UWord16)(_audioFrame._frequencyInHz));
+ // Reset the tone to be added taking the new sample rate into account.
+ _dtmfGenerator.ResetTone();
+ }
+
+ WebRtc_Word16 toneBuffer[320];
+ WebRtc_UWord16 toneSamples(0);
+ if (_dtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "OutputMixer::InsertInbandDtmfTone() inserting Dtmf"
+ "tone failed");
+ return -1;
+ }
+
+ // replace mixed audio with Dtmf tone
+ if (_audioFrame._audioChannel == 1)
+ {
+ // mono
+ memcpy(_audioFrame._payloadData, toneBuffer, sizeof(WebRtc_Word16)
+ * toneSamples);
+ } else
+ {
+ // stereo
+ for (int i = 0; i < _audioFrame._payloadDataLengthInSamples; i++)
+ {
+ _audioFrame._payloadData[2 * i] = toneBuffer[i];
+ _audioFrame._payloadData[2 * i + 1] = 0;
+ }
+ }
+ assert(_audioFrame._payloadDataLengthInSamples == toneSamples);
+
+ return 0;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/output_mixer.h b/trunk/src/voice_engine/main/source/output_mixer.h
new file mode 100644
index 0000000..a5df185
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/output_mixer.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H
+#define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H
+
+#include "audio_conference_mixer.h"
+#include "audio_conference_mixer_defines.h"
+#include "common_types.h"
+#include "dtmf_inband.h"
+#include "file_recorder.h"
+#include "level_indicator.h"
+#include "resampler.h"
+#include "voice_engine_defines.h"
+
+namespace webrtc {
+
+class AudioProcessing;
+class CriticalSectionWrapper;
+class FileWrapper;
+class VoEMediaProcess;
+
+namespace voe {
+
+class Statistics;
+
+class OutputMixer : public AudioMixerOutputReceiver,
+ public AudioMixerStatusReceiver,
+ public FileCallback
+{
+public:
+ static WebRtc_Word32 Create(OutputMixer*& mixer,
+ const WebRtc_UWord32 instanceId);
+
+ static void Destroy(OutputMixer*& mixer);
+
+ WebRtc_Word32 SetEngineInformation(Statistics& engineStatistics);
+
+ WebRtc_Word32 SetAudioProcessingModule(
+ AudioProcessing* audioProcessingModule);
+
+ // VoEExternalMedia
+ int RegisterExternalMediaProcessing(
+ VoEMediaProcess& proccess_object);
+
+ int DeRegisterExternalMediaProcessing();
+
+ // VoEDtmf
+ int PlayDtmfTone(WebRtc_UWord8 eventCode,
+ int lengthMs,
+ int attenuationDb);
+
+ int StartPlayingDtmfTone(WebRtc_UWord8 eventCode,
+ int attenuationDb);
+
+ int StopPlayingDtmfTone();
+
+ WebRtc_Word32 MixActiveChannels();
+
+ WebRtc_Word32 DoOperationsOnCombinedSignal();
+
+ WebRtc_Word32 SetMixabilityStatus(MixerParticipant& participant,
+ const bool mixable);
+
+ WebRtc_Word32 SetAnonymousMixabilityStatus(MixerParticipant& participant,
+ const bool mixable);
+
+ WebRtc_Word32 GetMixedAudio(const WebRtc_Word32 desiredFreqHz,
+ const WebRtc_UWord8 channels,
+ AudioFrame& audioFrame);
+
+ // VoEVolumeControl
+ int GetSpeechOutputLevel(WebRtc_UWord32& level);
+
+ int GetSpeechOutputLevelFullRange(WebRtc_UWord32& level);
+
+ int SetOutputVolumePan(float left, float right);
+
+ int GetOutputVolumePan(float& left, float& right);
+
+ // VoEFile
+ int StartRecordingPlayout(const char* fileName,
+ const CodecInst* codecInst);
+
+ int StartRecordingPlayout(OutStream* stream,
+ const CodecInst* codecInst);
+ int StopRecordingPlayout();
+
+ virtual ~OutputMixer();
+
+public: // from AudioMixerOutputReceiver
+ virtual void NewMixedAudio(
+ const WebRtc_Word32 id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const WebRtc_UWord32 size);
+
+public: // from AudioMixerStatusReceiver
+ virtual void MixedParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size);
+
+ virtual void VADPositiveParticipants(
+ const WebRtc_Word32 id,
+ const ParticipantStatistics* participantStatistics,
+ const WebRtc_UWord32 size);
+
+ virtual void MixedAudioLevel(const WebRtc_Word32 id,
+ const WebRtc_UWord32 level);
+
+public: // For file recording
+ void PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+
+ void RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+
+ void PlayFileEnded(const WebRtc_Word32 id);
+ void RecordFileEnded(const WebRtc_Word32 id);
+
+private:
+ OutputMixer(const WebRtc_UWord32 instanceId);
+ int APMAnalyzeReverseStream();
+ int InsertInbandDtmfTone();
+
+private: // uses
+ Statistics* _engineStatisticsPtr;
+ AudioProcessing* _audioProcessingModulePtr;
+
+private: // owns
+ CriticalSectionWrapper& _callbackCritSect;
+ // protect the _outputFileRecorderPtr and _outputFileRecording
+ CriticalSectionWrapper& _fileCritSect;
+ AudioConferenceMixer& _mixerModule;
+ AudioFrame _audioFrame;
+ Resampler _resampler; // converts mixed audio to fit ADM format
+ Resampler _apmResampler; // converts mixed audio to fit APM rate
+ AudioLevel _audioLevel; // measures audio level for the combined signal
+ DtmfInband _dtmfGenerator;
+ int _instanceId;
+ VoEMediaProcess* _externalMediaCallbackPtr;
+ bool _externalMedia;
+ float _panLeft;
+ float _panRight;
+ int _mixingFrequencyHz;
+ FileRecorder* _outputFileRecorderPtr;
+ bool _outputFileRecording;
+};
+
+} // namespace voe
+
+} // namespace werbtc
+
+#endif // VOICE_ENGINE_OUTPUT_MIXER_H
diff --git a/trunk/src/voice_engine/main/source/ref_count.cc b/trunk/src/voice_engine/main/source/ref_count.cc
new file mode 100644
index 0000000..f1ed0be
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/ref_count.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "critical_section_wrapper.h"
+#include "ref_count.h"
+
+namespace webrtc {
+
+namespace voe {
+
+RefCount::RefCount() :
+ _count(0),
+ _crit(*CriticalSectionWrapper::CreateCriticalSection())
+{
+}
+
+RefCount::~RefCount()
+{
+ delete &_crit;
+}
+
+RefCount&
+RefCount::operator++(int)
+{
+ CriticalSectionScoped lock(_crit);
+ _count++;
+ return *this;
+}
+
+RefCount&
+RefCount::operator--(int)
+{
+ CriticalSectionScoped lock(_crit);
+ _count--;
+ return *this;
+}
+
+void
+RefCount::Reset()
+{
+ CriticalSectionScoped lock(_crit);
+ _count = 0;
+}
+
+int
+RefCount::GetCount() const
+{
+ return _count;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/ref_count.h b/trunk/src/voice_engine/main/source/ref_count.h
new file mode 100644
index 0000000..e8c0a81
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/ref_count.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_REF_COUNT_H
+#define WEBRTC_VOICE_ENGINE_REF_COUNT_H
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace voe {
+
+class RefCount
+{
+public:
+ RefCount();
+ ~RefCount();
+ RefCount& operator++(int);
+ RefCount& operator--(int);
+ void Reset();
+ int GetCount() const;
+private:
+ volatile int _count;
+ CriticalSectionWrapper& _crit;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+#endif // #ifndef WEBRTC_VOICE_ENGINE_REF_COUNT_H
diff --git a/trunk/src/voice_engine/main/source/shared_data.cc b/trunk/src/voice_engine/main/source/shared_data.cc
new file mode 100644
index 0000000..2362da5
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/shared_data.cc
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "shared_data.h"
+
+#include "audio_processing.h"
+#include "critical_section_wrapper.h"
+#include "channel.h"
+#include "output_mixer.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+
+namespace webrtc {
+
+namespace voe {
+
+static WebRtc_Word32 _gInstanceCounter = 0;
+
+SharedData::SharedData() :
+ _instanceId(++_gInstanceCounter),
+ _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
+ _channelManager(_gInstanceCounter),
+ _engineStatistics(_gInstanceCounter),
+ _audioDevicePtr(NULL),
+ _audioProcessingModulePtr(NULL),
+ _moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
+ _externalRecording(false),
+ _externalPlayout(false)
+{
+ Trace::CreateTrace();
+ Trace::SetLevelFilter(WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER);
+ if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
+ {
+ _outputMixerPtr->SetEngineInformation(_engineStatistics);
+ }
+ if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0)
+ {
+ _transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
+ _engineStatistics,
+ _channelManager);
+ }
+ _audioDeviceLayer = AudioDeviceModule::kPlatformDefaultAudio;
+}
+
+SharedData::~SharedData()
+{
+ OutputMixer::Destroy(_outputMixerPtr);
+ TransmitMixer::Destroy(_transmitMixerPtr);
+ if (_audioDevicePtr) {
+ _audioDevicePtr->Release();
+ }
+ AudioProcessing::Destroy(_audioProcessingModulePtr);
+ delete _apiCritPtr;
+ ProcessThread::DestroyProcessThread(_moduleProcessThreadPtr);
+ Trace::ReturnTrace();
+}
+
+WebRtc_UWord16
+SharedData::NumOfSendingChannels()
+{
+ WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
+ if (numOfChannels <= 0)
+ {
+ return 0;
+ }
+
+ WebRtc_UWord16 nChannelsSending(0);
+ WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
+
+ _channelManager.GetChannelIds(channelsArray, numOfChannels);
+ for (int i = 0; i < numOfChannels; i++)
+ {
+ voe::ScopedChannel sc(_channelManager, channelsArray[i]);
+ Channel* chPtr = sc.ChannelPtr();
+ if (chPtr)
+ {
+ if (chPtr->Sending())
+ {
+ nChannelsSending++;
+ }
+ }
+ }
+ delete [] channelsArray;
+ return nChannelsSending;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/shared_data.h b/trunk/src/voice_engine/main/source/shared_data.h
new file mode 100644
index 0000000..afab37b
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/shared_data.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
+#define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
+
+#include "voice_engine_defines.h"
+
+#include "channel_manager.h"
+#include "statistics.h"
+#include "process_thread.h"
+
+#include "audio_device.h"
+#include "audio_processing.h"
+
+class ProcessThread;
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace voe {
+
+class TransmitMixer;
+class OutputMixer;
+
+class SharedData
+{
+public:
+ // Public accessors.
+ WebRtc_UWord32 instance_id() const { return _instanceId; }
+ Statistics& statistics() { return _engineStatistics; }
+ ChannelManager& channel_manager() { return _channelManager; }
+ AudioDeviceModule* audio_device() { return _audioDevicePtr; }
+
+protected:
+ WebRtc_UWord16 NumOfSendingChannels();
+
+protected:
+ const WebRtc_UWord32 _instanceId;
+ CriticalSectionWrapper* _apiCritPtr;
+ ChannelManager _channelManager;
+ Statistics _engineStatistics;
+ AudioDeviceModule* _audioDevicePtr;
+ OutputMixer* _outputMixerPtr;
+ TransmitMixer* _transmitMixerPtr;
+ AudioProcessing* _audioProcessingModulePtr;
+ ProcessThread* _moduleProcessThreadPtr;
+
+protected:
+ bool _externalRecording;
+ bool _externalPlayout;
+
+ AudioDeviceModule::AudioLayer _audioDeviceLayer;
+
+protected:
+ SharedData();
+ virtual ~SharedData();
+};
+
+} // namespace voe
+
+} // namespace webrtc
+#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H
diff --git a/trunk/src/voice_engine/main/source/statistics.cc b/trunk/src/voice_engine/main/source/statistics.cc
new file mode 100644
index 0000000..a534030
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/statistics.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cassert>
+#include <stdio.h>
+
+#include "statistics.h"
+
+#include "trace.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+
+namespace voe {
+
+Statistics::Statistics(const WebRtc_UWord32 instanceId) :
+ _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
+ _instanceId(instanceId),
+ _lastError(0),
+ _isInitialized(false)
+{
+}
+
+Statistics::~Statistics()
+{
+ if (_critPtr)
+ {
+ delete _critPtr;
+ _critPtr = NULL;
+ }
+}
+
+WebRtc_Word32 Statistics::SetInitialized()
+{
+ _isInitialized = true;
+ return 0;
+}
+
+WebRtc_Word32 Statistics::SetUnInitialized()
+{
+ _isInitialized = false;
+ return 0;
+}
+
+bool Statistics::Initialized() const
+{
+ return _isInitialized;
+}
+
+WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error) const
+{
+ CriticalSectionScoped cs(*_critPtr);
+ _lastError = error;
+ return 0;
+}
+
+WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error,
+ const TraceLevel level) const
+{
+ CriticalSectionScoped cs(*_critPtr);
+ _lastError = error;
+ WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
+ "error code is set to %d",
+ _lastError);
+ return 0;
+}
+
+WebRtc_Word32 Statistics::SetLastError(
+ const WebRtc_Word32 error,
+ const TraceLevel level, const char* msg) const
+{
+ CriticalSectionScoped cs(*_critPtr);
+ char traceMessage[KTraceMaxMessageSize];
+ assert(strlen(msg) < KTraceMaxMessageSize);
+ _lastError = error;
+ sprintf(traceMessage, "%s (error=%d)", msg, error);
+ WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1), "%s",
+ traceMessage);
+ return 0;
+}
+
+WebRtc_Word32 Statistics::LastError() const
+{
+ CriticalSectionScoped cs(*_critPtr);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "LastError() => %d", _lastError);
+ return _lastError;
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/statistics.h b/trunk/src/voice_engine/main/source/statistics.h
new file mode 100644
index 0000000..fc0bf8c
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/statistics.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_STATISTICS_H
+#define WEBRTC_VOICE_ENGINE_STATISTICS_H
+
+#include "common_types.h"
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+#include "voe_errors.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace voe {
+
+class Statistics
+{
+ public:
+ enum {KTraceMaxMessageSize = 256};
+ public:
+ Statistics(const WebRtc_UWord32 instanceId);
+ ~Statistics();
+
+ WebRtc_Word32 SetInitialized();
+ WebRtc_Word32 SetUnInitialized();
+ bool Initialized() const;
+ WebRtc_Word32 SetLastError(const WebRtc_Word32 error) const;
+ WebRtc_Word32 SetLastError(const WebRtc_Word32 error,
+ const TraceLevel level) const;
+ WebRtc_Word32 SetLastError(const WebRtc_Word32 error,
+ const TraceLevel level,
+ const char* msg) const;
+ WebRtc_Word32 LastError() const;
+
+ private:
+ CriticalSectionWrapper* _critPtr;
+ const WebRtc_UWord32 _instanceId;
+ mutable WebRtc_Word32 _lastError;
+ bool _isInitialized;
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_STATISTICS_H
diff --git a/trunk/src/voice_engine/main/source/transmit_mixer.cc b/trunk/src/voice_engine/main/source/transmit_mixer.cc
new file mode 100644
index 0000000..d131a85
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/transmit_mixer.cc
@@ -0,0 +1,1416 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "transmit_mixer.h"
+
+#include "audio_frame_operations.h"
+#include "channel.h"
+#include "channel_manager.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "statistics.h"
+#include "trace.h"
+#include "utility.h"
+#include "voe_base_impl.h"
+#include "voe_external_media.h"
+
+#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
+
+namespace webrtc {
+
+namespace voe {
+
+void
+TransmitMixer::OnPeriodicProcess()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess()");
+
+#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
+ if (_typingNoiseWarning > 0)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() => "
+ "CallbackOnError(VE_TYPING_NOISE_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(-1,
+ VE_TYPING_NOISE_WARNING);
+ }
+ _typingNoiseWarning = 0;
+ }
+#endif
+
+ if (_saturationWarning > 0)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() =>"
+ " CallbackOnError(VE_SATURATION_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
+ }
+ _saturationWarning = 0;
+ }
+
+ if (_noiseWarning > 0)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() =>"
+ "CallbackOnError(VE_NOISE_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(-1, VE_NOISE_WARNING);
+ }
+ _noiseWarning = 0;
+ }
+}
+
+
+void TransmitMixer::PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void TransmitMixer::RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void TransmitMixer::PlayFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayFileEnded(id=%d)", id);
+
+ assert(id == _filePlayerId);
+
+ CriticalSectionScoped cs(_critSect);
+
+ _filePlaying = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayFileEnded() =>"
+ "file player module is shutdown");
+}
+
+void
+TransmitMixer::RecordFileEnded(const WebRtc_Word32 id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded(id=%d)", id);
+
+ if (id == _fileRecorderId)
+ {
+ CriticalSectionScoped cs(_critSect);
+ _fileRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded() => fileRecorder module"
+ "is shutdown");
+ } else if (id == _fileCallRecorderId)
+ {
+ CriticalSectionScoped cs(_critSect);
+ _fileCallRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded() => fileCallRecorder"
+ "module is shutdown");
+ }
+}
+
+WebRtc_Word32
+TransmitMixer::Create(TransmitMixer*& mixer, const WebRtc_UWord32 instanceId)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
+ "TransmitMixer::Create(instanceId=%d)", instanceId);
+ mixer = new TransmitMixer(instanceId);
+ if (mixer == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
+ "TransmitMixer::Create() unable to allocate memory"
+ "for mixer");
+ return -1;
+ }
+ return 0;
+}
+
+void
+TransmitMixer::Destroy(TransmitMixer*& mixer)
+{
+ if (mixer)
+ {
+ delete mixer;
+ mixer = NULL;
+ }
+}
+
+TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
+ _engineStatisticsPtr(NULL),
+ _channelManagerPtr(NULL),
+ _audioProcessingModulePtr(NULL),
+ _voiceEngineObserverPtr(NULL),
+ _processThreadPtr(NULL),
+ _filePlayerPtr(NULL),
+ _fileRecorderPtr(NULL),
+ _fileCallRecorderPtr(NULL),
+ // Avoid conflict with other channels by adding 1024 - 1026,
+ // won't use as much as 1024 channels.
+ _filePlayerId(instanceId + 1024),
+ _fileRecorderId(instanceId + 1025),
+ _fileCallRecorderId(instanceId + 1026),
+ _filePlaying(false),
+ _fileRecording(false),
+ _fileCallRecording(false),
+ _audioLevel(),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ _timeActive(0),
+ _penaltyCounter(0),
+ _typingNoiseWarning(0),
+#endif
+ _saturationWarning(0),
+ _noiseWarning(0),
+ _instanceId(instanceId),
+ _mixFileWithMicrophone(false),
+ _captureLevel(0),
+ _externalMedia(false),
+ _externalMediaCallbackPtr(NULL),
+ _mute(false),
+ _remainingMuteMicTimeMs(0),
+ _mixingFrequency(0),
+ _includeAudioLevelIndication(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::TransmitMixer() - ctor");
+}
+
+TransmitMixer::~TransmitMixer()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::~TransmitMixer() - dtor");
+ _monitorModule.DeRegisterObserver();
+ if (_processThreadPtr)
+ {
+ _processThreadPtr->DeRegisterModule(&_monitorModule);
+ }
+ if (_externalMedia)
+ {
+ DeRegisterExternalMediaProcessing();
+ }
+ {
+ CriticalSectionScoped cs(_critSect);
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+ }
+ delete &_critSect;
+ delete &_callbackCritSect;
+}
+
+WebRtc_Word32
+TransmitMixer::SetEngineInformation(ProcessThread& processThread,
+ Statistics& engineStatistics,
+ ChannelManager& channelManager)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetEngineInformation()");
+
+ _processThreadPtr = &processThread;
+ _engineStatisticsPtr = &engineStatistics;
+ _channelManagerPtr = &channelManager;
+
+ if (_processThreadPtr->RegisterModule(&_monitorModule) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetEngineInformation() failed to"
+ "register the monitor module");
+ } else
+ {
+ _monitorModule.RegisterObserver(*this);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RegisterVoiceEngineObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+
+ if (_voiceEngineObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterVoiceEngineObserver() observer already enabled");
+ return -1;
+ }
+ _voiceEngineObserverPtr = &observer;
+ return 0;
+}
+
+WebRtc_Word32
+TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetAudioProcessingModule("
+ "audioProcessingModule=0x%x)",
+ audioProcessingModule);
+ _audioProcessingModulePtr = audioProcessingModule;
+ return 0;
+}
+
+WebRtc_Word32
+TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord16 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord16 currentMicLevel)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
+ "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%u,"
+ "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
+ totalDelayMS, clockDrift, currentMicLevel);
+
+
+ const int mixingFrequency = _mixingFrequency;
+
+ ScopedChannel sc(*_channelManagerPtr);
+ void* iterator(NULL);
+ Channel* channelPtr = sc.GetFirstChannel(iterator);
+ _mixingFrequency = 8000;
+ while (channelPtr != NULL)
+ {
+ if (channelPtr->Sending())
+ {
+ CodecInst tmpCdc;
+ channelPtr->GetSendCodec(tmpCdc);
+ if (tmpCdc.plfreq > _mixingFrequency)
+ _mixingFrequency = tmpCdc.plfreq;
+ }
+ channelPtr = sc.GetNextChannel(iterator);
+ }
+
+
+ // --- Resample input audio and create/store the initial audio frame
+
+ if (GenerateAudioFrame((const WebRtc_Word16*) audioSamples,
+ nSamples,
+ nChannels,
+ samplesPerSec,
+ _mixingFrequency) == -1)
+ {
+ return -1;
+ }
+
+ // --- Near-end Voice Quality Enhancement (APM) processing
+
+ APMProcessStream(totalDelayMS, clockDrift, currentMicLevel);
+
+ // --- Annoying typing detection (utilizes the APM/VAD decision)
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ TypingDetection();
+#endif
+
+ // --- Mute during DTMF tone if direct feedback is enabled
+
+ if (_remainingMuteMicTimeMs > 0)
+ {
+ AudioFrameOperations::Mute(_audioFrame);
+ _remainingMuteMicTimeMs -= 10;
+ if (_remainingMuteMicTimeMs < 0)
+ {
+ _remainingMuteMicTimeMs = 0;
+ }
+ }
+
+ // --- Mute signal
+
+ if (_mute)
+ {
+ AudioFrameOperations::Mute(_audioFrame);
+ }
+
+ // --- Measure audio level of speech after APM processing
+
+ _audioLevel.ComputeLevel(_audioFrame);
+
+ // --- Mix with file (does not affect the mixing frequency)
+
+ if (_filePlaying)
+ {
+ MixOrReplaceAudioWithFile(_mixingFrequency);
+ }
+
+ // --- Record to file
+
+ if (_fileRecording)
+ {
+ RecordAudioToFile(_mixingFrequency);
+ }
+
+ // --- External media processing
+
+ if (_externalMedia)
+ {
+ CriticalSectionScoped cs(_callbackCritSect);
+ const bool isStereo = (_audioFrame._audioChannel == 2);
+ if (_externalMediaCallbackPtr)
+ {
+ _externalMediaCallbackPtr->Process(
+ -1,
+ kRecordingAllChannelsMixed,
+ (WebRtc_Word16*) _audioFrame._payloadData,
+ _audioFrame._payloadDataLengthInSamples,
+ _audioFrame._frequencyInHz,
+ isStereo);
+ }
+ }
+
+ if (_mixingFrequency != mixingFrequency)
+ {
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::TransmitMixer::PrepareDemux() => "
+ "mixing frequency = %d",
+ _mixingFrequency);
+ }
+
+ return 0;
+}
+
+
+
+WebRtc_Word32
+TransmitMixer::DemuxAndMix()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::DemuxAndMix()");
+
+ ScopedChannel sc(*_channelManagerPtr);
+ void* iterator(NULL);
+ Channel* channelPtr = sc.GetFirstChannel(iterator);
+ while (channelPtr != NULL)
+ {
+ if (channelPtr->InputIsOnHold())
+ {
+ channelPtr->UpdateLocalTimeStamp();
+ } else if (channelPtr->Sending())
+ {
+ // load temporary audioframe with current (mixed) microphone signal
+ AudioFrame tmpAudioFrame = _audioFrame;
+
+ channelPtr->Demultiplex(tmpAudioFrame);
+ channelPtr->PrepareEncodeAndSend(_mixingFrequency);
+ }
+ channelPtr = sc.GetNextChannel(iterator);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32
+TransmitMixer::EncodeAndSend()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::EncodeAndSend()");
+
+ ScopedChannel sc(*_channelManagerPtr);
+ void* iterator(NULL);
+ Channel* channelPtr = sc.GetFirstChannel(iterator);
+ while (channelPtr != NULL)
+ {
+ if (channelPtr->Sending() && !channelPtr->InputIsOnHold())
+ {
+ channelPtr->EncodeAndSend();
+ }
+ channelPtr = sc.GetNextChannel(iterator);
+ }
+ return 0;
+}
+
+WebRtc_UWord32 TransmitMixer::CaptureLevel() const
+{
+ return _captureLevel;
+}
+
+void
+TransmitMixer::UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
+ lengthMs);
+ _remainingMuteMicTimeMs = lengthMs;
+}
+
+WebRtc_Word32
+TransmitMixer::StopSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopSend()");
+ _audioLevel.Clear();
+ return 0;
+}
+
+int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
+ const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartPlayingFileAsMicrophone("
+ "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f,"
+ " startPosition=%d, stopPosition=%d)", fileName, loop,
+ format, volumeScaling, startPosition, stopPosition);
+
+ if (_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() is already playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+
+ // Dynamically create the instance
+ _filePlayerPtr
+ = FilePlayer::CreateFilePlayer(_filePlayerId,
+ (const FileFormats) format);
+
+ if (_filePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_filePlayerPtr->StartPlayingFile(
+ fileName,
+ loop,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*) codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ return -1;
+ }
+
+ _filePlayerPtr->RegisterModuleFileCallback(this);
+ _filePlaying = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::StartPlayingFileAsMicrophone(format=%d,"
+ " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
+ format, volumeScaling, startPosition, stopPosition);
+
+ if (stream == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFileAsMicrophone() NULL as input stream");
+ return -1;
+ }
+
+ if (_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() is already playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+
+ // Dynamically create the instance
+ _filePlayerPtr
+ = FilePlayer::CreateFilePlayer(_filePlayerId,
+ (const FileFormats) format);
+
+ if (_filePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceWarning,
+ "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const WebRtc_UWord32 notificationTime(0);
+
+ if (_filePlayerPtr->StartPlayingFile(
+ (InStream&) *stream,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*) codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ return -1;
+ }
+ _filePlayerPtr->RegisterModuleFileCallback(this);
+ _filePlaying = true;
+
+ return 0;
+}
+
+int TransmitMixer::StopPlayingFileAsMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::StopPlayingFileAsMicrophone()");
+
+ if (!_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "StopPlayingFileAsMicrophone() isnot playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ if (_filePlayerPtr->StopPlayingFile() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_STOP_PLAYOUT, kTraceError,
+ "StopPlayingFile() couldnot stop playing file");
+ return -1;
+ }
+
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ _filePlaying = false;
+
+ return 0;
+}
+
+int TransmitMixer::IsPlayingFileAsMicrophone() const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::IsPlayingFileAsMicrophone()");
+ return _filePlaying;
+}
+
+int TransmitMixer::ScaleFileAsMicrophonePlayout(const float scale)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
+ scale);
+
+ CriticalSectionScoped cs(_critSect);
+
+ if (!_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ScaleFileAsMicrophonePlayout() isnot playing file");
+ return -1;
+ }
+
+ if ((_filePlayerPtr == NULL) ||
+ (_filePlayerPtr->SetAudioScaling(scale) != 0))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "SetAudioScaling() failed to scale playout");
+ return -1;
+ }
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingMicrophone(const WebRtc_Word8* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
+ fileName);
+
+ if (_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingMicrophone() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
+
+ if (codecInst != NULL &&
+ (codecInst->channels < 0 || codecInst->channels > 2))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() invalid compression");
+ return (-1);
+ }
+ if (codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst = &dummyCodec;
+ } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ } else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+
+ _fileRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileRecorderId,
+ (const FileFormats) format);
+ if (_fileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->StartRecordingAudioFile(
+ fileName,
+ (const CodecInst&) *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ return -1;
+ }
+ _fileRecorderPtr->RegisterModuleFileCallback(this);
+ _fileRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingMicrophone()");
+
+ if (_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingMicrophone() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() invalid compression");
+ return (-1);
+ }
+ if (codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst = &dummyCodec;
+ } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ } else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+
+ _fileRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileRecorderId,
+ (const FileFormats) format);
+ if (_fileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->StartRecordingAudioFile(*stream,
+ *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ return -1;
+ }
+
+ _fileRecorderPtr->RegisterModuleFileCallback(this);
+ _fileRecording = true;
+
+ return 0;
+}
+
+
+int TransmitMixer::StopRecordingMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopRecordingMicrophone()");
+
+ if (!_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StopRecordingMicrophone() isnot recording");
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ if (_fileRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording(), could not stop recording");
+ return -1;
+ }
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ _fileRecording = false;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingCall(const WebRtc_Word8* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingCall(fileName=%s)", fileName);
+
+ if (_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingCall() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingCall() invalid compression");
+ return (-1);
+ }
+ if (codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst = &dummyCodec;
+ } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ } else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+
+ _fileCallRecorderPtr
+ = FileRecorder::CreateFileRecorder(_fileCallRecorderId,
+ (const FileFormats) format);
+ if (_fileCallRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingCall() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileCallRecorderPtr->StartRecordingAudioFile(
+ fileName,
+ (const CodecInst&) *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ return -1;
+ }
+ _fileCallRecorderPtr->RegisterModuleFileCallback(this);
+ _fileCallRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingCall(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingCall()");
+
+ if (_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingCall() is already recording");
+ return 0;
+ }
+
+ FileFormats format;
+ const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
+ CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
+
+ if (codecInst != NULL && codecInst->channels != 1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingCall() invalid compression");
+ return (-1);
+ }
+ if (codecInst == NULL)
+ {
+ format = kFileFormatPcm16kHzFile;
+ codecInst = &dummyCodec;
+ } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
+ (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
+ {
+ format = kFileFormatWavFile;
+ } else
+ {
+ format = kFileFormatCompressedFile;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ // Destroy the old instance
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+
+ _fileCallRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileCallRecorderId,
+ (const FileFormats) format);
+ if (_fileCallRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingCall() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream,
+ *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ return -1;
+ }
+
+ _fileCallRecorderPtr->RegisterModuleFileCallback(this);
+ _fileCallRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StopRecordingCall()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopRecordingCall()");
+
+ if (!_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StopRecordingCall() file isnot recording");
+ return -1;
+ }
+
+ CriticalSectionScoped cs(_critSect);
+
+ if (_fileCallRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording(), could not stop recording");
+ return -1;
+ }
+
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ _fileCallRecording = false;
+
+ return 0;
+}
+
+void
+TransmitMixer::SetMixWithMicStatus(bool mix)
+{
+ _mixFileWithMicrophone = mix;
+}
+
+int TransmitMixer::RegisterExternalMediaProcessing(
+ VoEMediaProcess& proccess_object)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ _externalMediaCallbackPtr = &proccess_object;
+ _externalMedia = true;
+
+ return 0;
+}
+
+int TransmitMixer::DeRegisterExternalMediaProcessing()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::DeRegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(_callbackCritSect);
+ _externalMedia = false;
+ _externalMediaCallbackPtr = NULL;
+
+ return 0;
+}
+
+int
+TransmitMixer::SetMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetMute(enable=%d)", enable);
+ _mute = enable;
+ return 0;
+}
+
+bool
+TransmitMixer::Mute() const
+{
+ return _mute;
+}
+
+WebRtc_Word8 TransmitMixer::AudioLevel() const
+{
+ // Speech + file level [0,9]
+ return _audioLevel.Level();
+}
+
+WebRtc_Word16 TransmitMixer::AudioLevelFullRange() const
+{
+ // Speech + file level [0,32767]
+ return _audioLevel.LevelFullRange();
+}
+
+bool TransmitMixer::IsRecordingCall()
+{
+ return _fileCallRecording;
+}
+
+bool TransmitMixer::IsRecordingMic()
+{
+
+ return _fileRecording;
+}
+
+WebRtc_Word32
+TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const int mixingFrequency)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::GenerateAudioFrame(nSamples=%u,"
+ "samplesPerSec=%u, mixingFrequency=%u)",
+ nSamples, samplesPerSec, mixingFrequency);
+
+ ResamplerType resampType = (nChannels == 1) ?
+ kResamplerSynchronous : kResamplerSynchronousStereo;
+
+
+ if (_audioResampler.ResetIfNeeded(samplesPerSec,
+ mixingFrequency,
+ resampType) != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::GenerateAudioFrame() unable to resample");
+ return -1;
+ }
+ if (_audioResampler.Push(
+ (WebRtc_Word16*) audioSamples,
+ nSamples * nChannels,
+ _audioFrame._payloadData,
+ AudioFrame::kMaxAudioFrameSizeSamples,
+ (int&) _audioFrame._payloadDataLengthInSamples) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::GenerateAudioFrame() resampling failed");
+ return -1;
+ }
+
+ _audioFrame._payloadDataLengthInSamples /= nChannels;
+ _audioFrame._id = _instanceId;
+ _audioFrame._timeStamp = -1;
+ _audioFrame._frequencyInHz = mixingFrequency;
+ _audioFrame._speechType = AudioFrame::kNormalSpeech;
+ _audioFrame._vadActivity = AudioFrame::kVadUnknown;
+ _audioFrame._audioChannel = nChannels;
+
+ return 0;
+}
+
+WebRtc_Word32 TransmitMixer::RecordAudioToFile(
+ const WebRtc_UWord32 mixingFrequency)
+{
+ CriticalSectionScoped cs(_critSect);
+ if (_fileRecorderPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordAudioToFile() filerecorder doesnot"
+ "exist");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordAudioToFile() file recording"
+ "failed");
+ return -1;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
+ const int mixingFrequency)
+{
+ WebRtc_Word16 fileBuffer[320];
+
+ WebRtc_UWord32 fileSamples(0);
+
+ {
+ CriticalSectionScoped cs(_critSect);
+ if (_filePlayerPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "TransmitMixer::MixOrReplaceAudioWithFile()"
+ "fileplayer doesnot exist");
+ return -1;
+ }
+
+ if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer,
+ fileSamples,
+ mixingFrequency) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::MixOrReplaceAudioWithFile() file"
+ " mixing failed");
+ return -1;
+ }
+ }
+
+ if (_mixFileWithMicrophone)
+ {
+ Utility::MixWithSat(_audioFrame._payloadData,
+ fileBuffer,
+ (WebRtc_UWord16) fileSamples);
+ assert(_audioFrame._payloadDataLengthInSamples == fileSamples);
+ } else
+ {
+ // replace ACM audio with file
+ _audioFrame.UpdateFrame(-1,
+ -1,
+ fileBuffer,
+ (WebRtc_UWord16) fileSamples, mixingFrequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadUnknown,
+ 1);
+
+ }
+ return 0;
+}
+
+WebRtc_Word32 TransmitMixer::APMProcessStream(
+ const WebRtc_UWord16 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord16 currentMicLevel)
+{
+ WebRtc_UWord16 captureLevel(currentMicLevel);
+
+ // Check if the number of input channels has changed. Retain the number
+ // of output channels.
+ if (_audioFrame._audioChannel !=
+ _audioProcessingModulePtr->num_input_channels())
+ {
+ if (_audioProcessingModulePtr->set_num_channels(
+ _audioFrame._audioChannel,
+ _audioProcessingModulePtr->num_output_channels()))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::set_num_channels(%d, %d) => error",
+ _audioFrame._frequencyInHz,
+ _audioProcessingModulePtr->num_output_channels());
+ }
+ }
+
+ // If the frequency has changed we need to change APM settings
+ // Sending side is "master"
+ if (_audioProcessingModulePtr->sample_rate_hz() !=
+ _audioFrame._frequencyInHz)
+ {
+ if (_audioProcessingModulePtr->set_sample_rate_hz(
+ _audioFrame._frequencyInHz))
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::set_sample_rate_hz(%u) => error",
+ _audioFrame._frequencyInHz);
+ }
+ }
+
+ if (_audioProcessingModulePtr->set_stream_delay_ms(totalDelayMS) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::set_stream_delay_ms(%u) => error",
+ totalDelayMS);
+ }
+ if (_audioProcessingModulePtr->gain_control()->set_stream_analog_level(
+ captureLevel) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::set_stream_analog_level(%u) => error",
+ captureLevel);
+ }
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ is_drift_compensation_enabled())
+ {
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ set_stream_drift_samples(clockDrift) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::set_stream_drift_samples(%u) => error",
+ clockDrift);
+ }
+ }
+ if (_audioProcessingModulePtr->ProcessStream(&_audioFrame) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioProcessing::ProcessStream() => error");
+ }
+ captureLevel =
+ _audioProcessingModulePtr->gain_control()->stream_analog_level();
+
+ // Store new capture level (only updated when analog AGC is enabled)
+ _captureLevel = captureLevel;
+
+ // Log notifications
+ if (_audioProcessingModulePtr->gain_control()->stream_is_saturated())
+ {
+ if (_saturationWarning == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::APMProcessStream() pending "
+ "saturation warning exists");
+ }
+ _saturationWarning = 1; // triggers callback from moduleprocess thread
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::APMProcessStream() VE_SATURATION_WARNING "
+ "message has been posted for callback");
+ }
+
+ return 0;
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+int TransmitMixer::TypingDetection()
+{
+ // We let the VAD determine if we're using this feature or not.
+ if (_audioFrame._vadActivity == AudioFrame::kVadUnknown)
+ {
+ return (0);
+ }
+
+ int keyPressed = EventWrapper::KeyPressed();
+
+ if (keyPressed < 0)
+ {
+ return (-1);
+ }
+
+ if (_audioFrame._vadActivity == AudioFrame::kVadActive)
+ _timeActive++;
+ else
+ _timeActive = 0;
+
+ if (keyPressed && (_audioFrame._vadActivity == AudioFrame::kVadActive)
+ && (_timeActive < 10))
+ {
+ _penaltyCounter += 100;
+ if (_penaltyCounter > 300)
+ {
+ if (_typingNoiseWarning == 1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "TransmitMixer::TypingDetection() pending "
+ "noise-saturation warning exists");
+ }
+ // triggers callback from the module process thread
+ _typingNoiseWarning = 1;
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::TypingDetection() "
+ "VE_TYPING_NOISE_WARNING message has been posted for"
+ "callback");
+ }
+ }
+
+ if (_penaltyCounter > 0)
+ _penaltyCounter--;
+
+ return (0);
+}
+#endif
+
+int TransmitMixer::GetMixingFrequency()
+{
+ assert(_mixingFrequency!=0);
+ return (_mixingFrequency);
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/transmit_mixer.h b/trunk/src/voice_engine/main/source/transmit_mixer.h
new file mode 100644
index 0000000..469b288
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/transmit_mixer.h
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
+#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
+
+#include "common_types.h"
+#include "voe_base.h"
+#include "file_player.h"
+#include "file_recorder.h"
+#include "level_indicator.h"
+#include "module_common_types.h"
+#include "monitor_module.h"
+#include "resampler.h"
+#include "voice_engine_defines.h"
+
+
+namespace webrtc {
+
+class AudioProcessing;
+class ProcessThread;
+class VoEExternalMedia;
+class VoEMediaProcess;
+
+namespace voe {
+
+class ChannelManager;
+class MixedAudio;
+class Statistics;
+
+class TransmitMixer : public MonitorObserver,
+ public FileCallback
+
+{
+public:
+ static WebRtc_Word32 Create(TransmitMixer*& mixer,
+ const WebRtc_UWord32 instanceId);
+
+ static void Destroy(TransmitMixer*& mixer);
+
+ WebRtc_Word32 SetEngineInformation(ProcessThread& processThread,
+ Statistics& engineStatistics,
+ ChannelManager& channelManager);
+
+ WebRtc_Word32 SetAudioProcessingModule(
+ AudioProcessing* audioProcessingModule);
+
+ WebRtc_Word32 PrepareDemux(const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord16 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord16 currentMicLevel);
+
+
+ WebRtc_Word32 DemuxAndMix();
+
+ WebRtc_Word32 EncodeAndSend();
+
+ WebRtc_UWord32 CaptureLevel() const;
+
+ WebRtc_Word32 StopSend();
+
+ // VoEDtmf
+ void UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs);
+
+ // VoEExternalMedia
+ int RegisterExternalMediaProcessing(VoEMediaProcess& proccess_object);
+
+ int DeRegisterExternalMediaProcessing();
+
+ int GetMixingFrequency();
+
+ // VoEVolumeControl
+ int SetMute(const bool enable);
+
+ bool Mute() const;
+
+ WebRtc_Word8 AudioLevel() const;
+
+ WebRtc_Word16 AudioLevelFullRange() const;
+
+ bool IsRecordingCall();
+
+ bool IsRecordingMic();
+
+ int StartPlayingFileAsMicrophone(const char* fileName,
+ const bool loop,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+
+ int StartPlayingFileAsMicrophone(InStream* stream,
+ const FileFormats format,
+ const int startPosition,
+ const float volumeScaling,
+ const int stopPosition,
+ const CodecInst* codecInst);
+
+ int StopPlayingFileAsMicrophone();
+
+ int IsPlayingFileAsMicrophone() const;
+
+ int ScaleFileAsMicrophonePlayout(const float scale);
+
+ int StartRecordingMicrophone(const char* fileName,
+ const CodecInst* codecInst);
+
+ int StartRecordingMicrophone(OutStream* stream,
+ const CodecInst* codecInst);
+
+ int StopRecordingMicrophone();
+
+ int StartRecordingCall(const char* fileName, const CodecInst* codecInst);
+
+ int StartRecordingCall(OutStream* stream, const CodecInst* codecInst);
+
+ int StopRecordingCall();
+
+ void SetMixWithMicStatus(bool mix);
+
+ WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
+
+ virtual ~TransmitMixer();
+
+public: // MonitorObserver
+ void OnPeriodicProcess();
+
+
+public: // FileCallback
+ void PlayNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+
+ void RecordNotification(const WebRtc_Word32 id,
+ const WebRtc_UWord32 durationMs);
+
+ void PlayFileEnded(const WebRtc_Word32 id);
+
+ void RecordFileEnded(const WebRtc_Word32 id);
+
+private:
+ TransmitMixer(const WebRtc_UWord32 instanceId);
+
+private:
+ WebRtc_Word32 GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const int mixingFrequency);
+ WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency);
+
+ WebRtc_Word32 MixOrReplaceAudioWithFile(
+ const int mixingFrequency);
+
+ WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord16 currentMicLevel);
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ int TypingDetection();
+#endif
+
+private: // uses
+ Statistics* _engineStatisticsPtr;
+ ChannelManager* _channelManagerPtr;
+ AudioProcessing* _audioProcessingModulePtr;
+ VoiceEngineObserver* _voiceEngineObserverPtr;
+ ProcessThread* _processThreadPtr;
+
+private: // owns
+ MonitorModule _monitorModule;
+ AudioFrame _audioFrame;
+ Resampler _audioResampler; // ADM sample rate -> mixing rate
+ FilePlayer* _filePlayerPtr;
+ FileRecorder* _fileRecorderPtr;
+ FileRecorder* _fileCallRecorderPtr;
+ int _filePlayerId;
+ int _fileRecorderId;
+ int _fileCallRecorderId;
+ bool _filePlaying;
+ bool _fileRecording;
+ bool _fileCallRecording;
+ voe::AudioLevel _audioLevel;
+ // protect file instances and their variables in MixedParticipants()
+ CriticalSectionWrapper& _critSect;
+ CriticalSectionWrapper& _callbackCritSect;
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ WebRtc_Word32 _timeActive;
+ WebRtc_Word32 _penaltyCounter;
+ WebRtc_UWord32 _typingNoiseWarning;
+#endif
+ WebRtc_UWord32 _saturationWarning;
+ WebRtc_UWord32 _noiseWarning;
+
+private:
+ int _instanceId;
+ bool _mixFileWithMicrophone;
+ WebRtc_UWord32 _captureLevel;
+ bool _externalMedia;
+ VoEMediaProcess* _externalMediaCallbackPtr;
+ bool _mute;
+ WebRtc_Word32 _remainingMuteMicTimeMs;
+ int _mixingFrequency;
+ bool _includeAudioLevelIndication;
+};
+
+#endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/utility.cc b/trunk/src/voice_engine/main/source/utility.cc
new file mode 100644
index 0000000..6e70156
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/utility.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "utility.h"
+
+#include "module.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+namespace voe
+{
+
+void Utility::MixWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[],
+ WebRtc_UWord16 len)
+{
+ WebRtc_Word32 temp(0);
+ for (int i = 0; i < len; i++)
+ {
+ temp = source[i] + target[i];
+ if (temp > 32767)
+ target[i] = 32767;
+ else if (temp < -32768)
+ target[i] = -32768;
+ else
+ target[i] = (WebRtc_Word16) temp;
+ }
+}
+
+void Utility::MixSubtractWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[],
+ WebRtc_UWord16 len)
+{
+ WebRtc_Word32 temp(0);
+ for (int i = 0; i < len; i++)
+ {
+ temp = target[i] - source[i];
+ if (temp > 32767)
+ target[i] = 32767;
+ else if (temp < -32768)
+ target[i] = -32768;
+ else
+ target[i] = (WebRtc_Word16) temp;
+ }
+}
+
+void Utility::MixAndScaleWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[], float scale,
+ WebRtc_UWord16 len)
+{
+ WebRtc_Word32 temp(0);
+ for (int i = 0; i < len; i++)
+ {
+ temp = (WebRtc_Word32) (target[i] + scale * source[i]);
+ if (temp > 32767)
+ target[i] = 32767;
+ else if (temp < -32768)
+ target[i] = -32768;
+ else
+ target[i] = (WebRtc_Word16) temp;
+ }
+}
+
+void Utility::Scale(WebRtc_Word16 vector[], float scale, WebRtc_UWord16 len)
+{
+ for (int i = 0; i < len; i++)
+ {
+ vector[i] = (WebRtc_Word16) (scale * vector[i]);
+ }
+}
+
+void Utility::ScaleWithSat(WebRtc_Word16 vector[], float scale,
+ WebRtc_UWord16 len)
+{
+ WebRtc_Word32 temp(0);
+ for (int i = 0; i < len; i++)
+ {
+ temp = (WebRtc_Word32) (scale * vector[i]);
+ if (temp > 32767)
+ vector[i] = 32767;
+ else if (temp < -32768)
+ vector[i] = -32768;
+ else
+ vector[i] = (WebRtc_Word16) temp;
+ }
+}
+
+} // namespace voe
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/utility.h b/trunk/src/voice_engine/main/source/utility.h
new file mode 100644
index 0000000..084ddf4
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/utility.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Contains functions often used by different parts of VoiceEngine.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H
+#define WEBRTC_VOICE_ENGINE_UTILITY_H
+
+#include "typedefs.h"
+#include "voice_engine_defines.h"
+
+namespace webrtc
+{
+
+class Module;
+
+namespace voe
+{
+
+class Utility
+{
+public:
+ static void MixWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[],
+ WebRtc_UWord16 len);
+
+ static void MixSubtractWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[],
+ WebRtc_UWord16 len);
+
+ static void MixAndScaleWithSat(WebRtc_Word16 target[],
+ const WebRtc_Word16 source[],
+ float scale,
+ WebRtc_UWord16 len);
+
+ static void Scale(WebRtc_Word16 vector[], float scale, WebRtc_UWord16 len);
+
+ static void ScaleWithSat(WebRtc_Word16 vector[],
+ float scale,
+ WebRtc_UWord16 len);
+};
+
+} // namespace voe
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_UTILITY_H
diff --git a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc
new file mode 100644
index 0000000..95800f4
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc
@@ -0,0 +1,1176 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_audio_processing_impl.h"
+
+#include "audio_processing.h"
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEAudioProcessing* VoEAudioProcessing::GetInterface(
+ VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEAudioProcessingImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+VoEAudioProcessingImpl::VoEAudioProcessingImpl():
+ _isAecMode(WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE == EcAec?
+ true : false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEAudioProcessingImpl::VoEAudioProcessingImpl() - ctor");
+}
+
+VoEAudioProcessingImpl::~VoEAudioProcessingImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEAudioProcessingImpl::~VoEAudioProcessingImpl() - dtor");
+}
+
+int VoEAudioProcessingImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEAudioProcessing::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _engineStatistics.SetLastError(
+ VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEAudioProcessing reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEAudioProcessingImpl::SetNsStatus(bool enable, NsModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetNsStatus(enable=%d, mode=%d)", enable, mode);
+#ifdef WEBRTC_VOICE_ENGINE_NR
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ NoiseSuppression::Level nsLevel(
+ (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE);
+ switch (mode)
+ {
+ case kNsDefault:
+ nsLevel = (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE;
+ break;
+ case kNsUnchanged:
+ nsLevel = _audioProcessingModulePtr->noise_suppression()->level();
+ break;
+ case kNsConference:
+ nsLevel = NoiseSuppression::kHigh;
+ break;
+ case kNsLowSuppression:
+ nsLevel = NoiseSuppression::kLow;
+ break;
+ case kNsModerateSuppression:
+ nsLevel = NoiseSuppression::kModerate;
+ break;
+ case kNsHighSuppression:
+ nsLevel = NoiseSuppression::kHigh;
+ break;
+ case kNsVeryHighSuppression:
+ nsLevel = NoiseSuppression::kVeryHigh;
+ break;
+ }
+
+ if (_audioProcessingModulePtr->noise_suppression()->set_level(nsLevel) != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "SetNsStatus() failed to set Ns mode");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetNsStatus() failed to set Ns state");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetNsStatus() Ns is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetNsStatus(bool& enabled, NsModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetNsStatus(enabled=?, mode=?)");
+#ifdef WEBRTC_VOICE_ENGINE_NR
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool enable(false);
+ NoiseSuppression::Level nsLevel(
+ (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE);
+
+ enable = _audioProcessingModulePtr->noise_suppression()->is_enabled();
+ nsLevel = _audioProcessingModulePtr->noise_suppression()->level();
+
+ enabled = enable;
+
+ switch (nsLevel)
+ {
+ case NoiseSuppression::kLow:
+ mode = kNsLowSuppression;
+ break;
+ case NoiseSuppression::kModerate:
+ mode = kNsModerateSuppression;
+ break;
+ case NoiseSuppression::kHigh:
+ mode = kNsHighSuppression;
+ break;
+ case NoiseSuppression::kVeryHigh:
+ mode = kNsVeryHighSuppression;
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetNsStatus() => enabled=% d, mode=%d",enabled, mode);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetNsStatus() Ns is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetAgcStatus(enable=%d, mode=%d)", enable, mode);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+#if defined(MAC_IPHONE) || defined(ATA) || defined(WEBRTC_ANDROID)
+ if (mode == kAgcAdaptiveAnalog)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetAgcStatus() invalid Agc mode for mobile device");
+ return -1;
+ }
+#endif
+
+ GainControl::Mode agcMode(
+ (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE);
+ switch (mode)
+ {
+ case kAgcDefault:
+ agcMode = (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE;
+ break;
+ case kAgcUnchanged:
+ agcMode = _audioProcessingModulePtr->gain_control()->mode();;
+ break;
+ case kAgcFixedDigital:
+ agcMode = GainControl::kFixedDigital;
+ break;
+ case kAgcAdaptiveAnalog:
+ agcMode = GainControl::kAdaptiveAnalog;
+ break;
+ case kAgcAdaptiveDigital:
+ agcMode = GainControl::kAdaptiveDigital;
+ break;
+ }
+
+ if (_audioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAgcStatus() failed to set Agc mode");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->gain_control()->Enable(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAgcStatus() failed to set Agc state");
+ return -1;
+ }
+
+ if (agcMode != GainControl::kFixedDigital)
+ {
+ // Set Agc state in the ADM when adaptive Agc mode has been selected.
+ // Note that we also enable the ADM Agc when Adaptive Digital mode is
+ // used since we want to be able to provide the APM with updated mic
+ // levels when the user modifies the mic level manually.
+ if (_audioDevicePtr->SetAGC(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+ "SetAgcStatus() failed to set Agc mode");
+ }
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAgcStatus() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetAgcStatus(enabled=?, mode=?)");
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool enable(false);
+ GainControl::Mode agcMode(
+ (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE);
+
+ enable = _audioProcessingModulePtr->gain_control()->is_enabled();
+ agcMode = _audioProcessingModulePtr->gain_control()->mode();
+
+ enabled = enable;
+
+ switch (agcMode)
+ {
+ case GainControl::kFixedDigital:
+ mode = kAgcFixedDigital;
+ break;
+ case GainControl::kAdaptiveAnalog:
+ mode = kAgcAdaptiveAnalog;
+ break;
+ case GainControl::kAdaptiveDigital:
+ mode = kAgcAdaptiveDigital;
+ break;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetAgcStatus() => enabled=%d, mode=%d", enabled, mode);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetAgcStatus() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetAgcConfig(const AgcConfig config)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetAgcConfig()");
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioProcessingModulePtr->gain_control()->set_target_level_dbfs(
+ config.targetLeveldBOv) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAgcConfig() failed to set target peak |level|"
+ " (or envelope) of the Agc");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->gain_control()->set_compression_gain_db(
+ config.digitalCompressionGaindB) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAgcConfig() failed to set the range in |gain|"
+ "the digital compression stage may apply");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->gain_control()->enable_limiter(
+ config.limiterEnable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAgcConfig() failed to set hard limiter to the signal");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAgcConfig() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig &config)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetAgcConfig(config=?)");
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ config.targetLeveldBOv =
+ _audioProcessingModulePtr->gain_control()->target_level_dbfs();
+ config.digitalCompressionGaindB =
+ _audioProcessingModulePtr->gain_control()->compression_gain_db();
+ config.limiterEnable =
+ _audioProcessingModulePtr->gain_control()->is_limiter_enabled();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetAgcConfig() => targetLeveldBOv=%u, "
+ "digitalCompressionGaindB=%u, limiterEnable=%d",
+ config.targetLeveldBOv,
+ config.digitalCompressionGaindB,
+ config.limiterEnable);
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetAgcConfig() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
+ bool enable,
+ NsModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRxNsStatus(channel=%d, enable=%d, mode=%d)",
+ channel, (int)enable, (int)mode);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRxNsStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRxNsStatus(enable, mode);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetRxNsStatus() AGC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetRxNsStatus(int channel,
+ bool& enabled,
+ NsModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRxNsStatus(channel=%d, enable=?, mode=?)", channel);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRxNsStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRxNsStatus(enabled, mode);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetRxNsStatus() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetRxAgcStatus(int channel,
+ bool enable,
+ AgcModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRxAgcStatus(channel=%d, enable=%d, mode=%d)",
+ channel, (int)enable, (int)mode);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRxAgcStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRxAgcStatus(enable, mode);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetRxAgcStatus() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetRxAgcStatus(int channel,
+ bool& enabled,
+ AgcModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRxAgcStatus(channel=%d, enable=?, mode=?)", channel);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRxAgcStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRxAgcStatus(enabled, mode);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetRxAgcStatus() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetRxAgcConfig(int channel, const AgcConfig config)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRxAgcConfig(channel=%d)", channel);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRxAgcConfig() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRxAgcConfig(config);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetRxAgcConfig() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetRxAgcConfig(int channel, AgcConfig& config)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRxAgcConfig(channel=%d)", channel);
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRxAgcConfig() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRxAgcConfig(config);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetRxAgcConfig() Agc is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetEcStatus(bool enable, EcModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetEcStatus(enable=%d, mode=%d)", enable, mode);
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // AEC mode
+ if ((mode == kEcDefault) ||
+ (mode == kEcConference) ||
+ (mode == kEcAec) ||
+ ((mode == kEcUnchanged) &&
+ (_isAecMode == true)))
+ {
+ if (enable)
+ {
+ // Disable the AECM before enable the AEC
+ if (_audioProcessingModulePtr->echo_control_mobile()->is_enabled())
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "SetEcStatus() disable AECM before enabling AEC");
+ if (_audioProcessingModulePtr->echo_control_mobile()->
+ Enable(false) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to disable AECM");
+ return -1;
+ }
+ }
+ }
+ if (_audioProcessingModulePtr->echo_cancellation()->Enable(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to set AEC state");
+ return -1;
+ }
+#ifdef CLOCK_SKEW_COMP
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ enable_drift_compensation(true) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to enable drift compensation");
+ return -1;
+ }
+#else
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ enable_drift_compensation(false) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to disable drift compensation");
+ return -1;
+ }
+#endif
+ if (mode == kEcConference)
+ {
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ set_suppression_level(EchoCancellation::kHighSuppression) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to set aggressiveness to high");
+ return -1;
+ }
+ }
+ else
+ {
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ set_suppression_level(
+ EchoCancellation::kModerateSuppression) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to set aggressiveness to moderate");
+ return -1;
+ }
+ }
+
+ _isAecMode = true;
+ }
+ else if ((mode == kEcAecm) ||
+ ((mode == kEcUnchanged) &&
+ (_isAecMode == false)))
+ {
+ if (enable)
+ {
+ // Disable the AEC before enable the AECM
+ if (_audioProcessingModulePtr->echo_cancellation()->is_enabled())
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "SetEcStatus() disable AEC before enabling AECM");
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ Enable(false) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to disable AEC");
+ return -1;
+ }
+ }
+ }
+ if (_audioProcessingModulePtr->echo_control_mobile()->
+ Enable(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcStatus() failed to set AECM state");
+ return -1;
+ }
+ _isAecMode = false;
+ }
+ else
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetEcStatus() invalid EC mode");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetEcStatus(bool& enabled, EcModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEcStatus()");
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_isAecMode == true)
+ {
+ mode = kEcAec;
+ enabled = _audioProcessingModulePtr->echo_cancellation()->is_enabled();
+ }
+ else
+ {
+ mode = kEcAecm;
+ enabled = _audioProcessingModulePtr->echo_control_mobile()->
+ is_enabled();
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEcStatus() => enabled=%i, mode=%i",
+ enabled, (int)mode);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::SetAecmMode(AecmModes mode, bool enableCNG)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetAECMMode(mode = %d)", mode);
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ EchoControlMobile::RoutingMode aecmMode(
+ EchoControlMobile::kQuietEarpieceOrHeadset);
+
+ switch (mode)
+ {
+ case kAecmQuietEarpieceOrHeadset:
+ aecmMode = EchoControlMobile::kQuietEarpieceOrHeadset;
+ break;
+ case kAecmEarpiece:
+ aecmMode = EchoControlMobile::kEarpiece;
+ break;
+ case kAecmLoudEarpiece:
+ aecmMode = EchoControlMobile::kLoudEarpiece;
+ break;
+ case kAecmSpeakerphone:
+ aecmMode = EchoControlMobile::kSpeakerphone;
+ break;
+ case kAecmLoudSpeakerphone:
+ aecmMode = EchoControlMobile::kLoudSpeakerphone;
+ break;
+ }
+
+
+ if (_audioProcessingModulePtr->echo_control_mobile()->
+ set_routing_mode(aecmMode) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAECMMode() failed to set AECM routing mode");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->echo_control_mobile()->
+ enable_comfort_noise(enableCNG) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetAECMMode() failed to set comfort noise state for AECM");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAECMMode() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetAecmMode(AecmModes& mode, bool& enabledCNG)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetAECMMode(mode=?)");
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ enabledCNG = false;
+
+ EchoControlMobile::RoutingMode aecmMode =
+ _audioProcessingModulePtr->echo_control_mobile()->routing_mode();
+ enabledCNG = _audioProcessingModulePtr->echo_control_mobile()->
+ is_comfort_noise_enabled();
+
+ switch (aecmMode)
+ {
+ case EchoControlMobile::kQuietEarpieceOrHeadset:
+ mode = kAecmQuietEarpieceOrHeadset;
+ break;
+ case EchoControlMobile::kEarpiece:
+ mode = kAecmEarpiece;
+ break;
+ case EchoControlMobile::kLoudEarpiece:
+ mode = kAecmLoudEarpiece;
+ break;
+ case EchoControlMobile::kSpeakerphone:
+ mode = kAecmSpeakerphone;
+ break;
+ case EchoControlMobile::kLoudSpeakerphone:
+ mode = kAecmLoudSpeakerphone;
+ break;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetAECMMode() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::RegisterRxVadObserver(
+ int channel,
+ VoERxVadCallback &observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterRxVadObserver()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterRxVadObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterRxVadObserver(observer);
+}
+
+int VoEAudioProcessingImpl::DeRegisterRxVadObserver(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterRxVadObserver()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterRxVadObserver() failed to locate channel");
+ return -1;
+ }
+
+ return channelPtr->DeRegisterRxVadObserver();
+}
+
+int VoEAudioProcessingImpl::VoiceActivityIndicator(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoiceActivityIndicator(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterRxVadObserver() failed to locate channel");
+ return -1;
+ }
+ int activity(-1);
+ channelPtr->VoiceActivityIndicator(activity);
+
+ return activity;
+}
+
+int VoEAudioProcessingImpl::SetEcMetricsStatus(bool enable) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetEcMetricsStatus(enable=%d)", enable);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized()) {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if ((_audioProcessingModulePtr->echo_cancellation()->enable_metrics(enable)
+ != 0) ||
+ (_audioProcessingModulePtr->echo_cancellation()->enable_delay_logging(
+ enable) != 0)) {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "SetEcMetricsStatus() unable to set EC metrics mode");
+ return -1;
+ }
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetEcMetricsStatus(bool& enabled) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetEcMetricsStatus(enabled=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized()) {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool echo_mode =
+ _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
+ bool delay_mode =
+ _audioProcessingModulePtr->echo_cancellation()->is_delay_logging_enabled();
+
+ if (echo_mode != delay_mode) {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceError,
+ "GetEcMetricsStatus() delay logging and echo mode are not the same");
+ return -1;
+ }
+
+ enabled = echo_mode;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetEcMetricsStatus() => enabled=%d", enabled);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
+ int& ERLE,
+ int& RERL,
+ int& A_NLP) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetEchoMetrics(ERL=?, ERLE=?, RERL=?, A_NLP=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized()) {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_audioProcessingModulePtr->echo_cancellation()->is_enabled()) {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "GetEchoMetrics() AudioProcessingModule AEC is not enabled");
+ return -1;
+ }
+
+ // Get Echo Metrics from Audio Processing Module.
+ EchoCancellation::Metrics echoMetrics;
+ if (_audioProcessingModulePtr->echo_cancellation()->GetMetrics(&echoMetrics))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEchoMetrics(), AudioProcessingModule metrics error");
+ return -1;
+ }
+
+ // Echo quality metrics.
+ ERL = echoMetrics.echo_return_loss.instant;
+ ERLE = echoMetrics.echo_return_loss_enhancement.instant;
+ RERL = echoMetrics.residual_echo_return_loss.instant;
+ A_NLP = echoMetrics.a_nlp.instant;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEchoMetrics() => ERL=%d, ERLE=%d, RERL=%d, A_NLP=%d",
+ ERL, ERLE, RERL, A_NLP);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
+ int& delay_std) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetEcDelayMetrics(median=?, std=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ if (!_engineStatistics.Initialized()) {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_audioProcessingModulePtr->echo_cancellation()->is_enabled()) {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "GetEcDelayMetrics() AudioProcessingModule AEC is not enabled");
+ return -1;
+ }
+
+ int median = 0;
+ int std = 0;
+ // Get delay-logging values from Audio Processing Module.
+ if (_audioProcessingModulePtr->echo_cancellation()->GetDelayMetrics(
+ &median, &std)) {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEcDelayMetrics(), AudioProcessingModule delay-logging "
+ "error");
+ return -1;
+ }
+
+ // EC delay-logging metrics
+ delay_median = median;
+ delay_std = std;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetEcDelayMetrics() => delay_median=%d, delay_std=%d",
+ delay_median, delay_std);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::StartDebugRecording(const char* fileNameUTF8)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartDebugRecording()");
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ return _audioProcessingModulePtr->StartDebugRecording(fileNameUTF8);
+
+}
+
+int VoEAudioProcessingImpl::StopDebugRecording()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopDebugRecording()");
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ return _audioProcessingModulePtr->StopDebugRecording();
+}
+
+int VoEAudioProcessingImpl::SetTypingDetectionStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetTypingDetectionStatus()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // Just use the VAD state to determine if we should enable typing detection
+ // or not
+
+ if (_audioProcessingModulePtr->voice_detection()->Enable(enable))
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "SetTypingDetectionStatus() failed to set VAD state");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->voice_detection()->set_likelihood(
+ VoiceDetection::kVeryLowLikelihood))
+ {
+ _engineStatistics.SetLastError(
+ VE_APM_ERROR, kTraceWarning,
+ "SetTypingDetectionStatus() failed to set VAD likelihood to low");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetTypingDetectionStatus is not supported");
+ return -1;
+#endif
+}
+
+int VoEAudioProcessingImpl::GetTypingDetectionStatus(bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetTypingDetectionStatus()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ // Just use the VAD state to determine if we should enable typing
+ // detection or not
+
+ enabled = _audioProcessingModulePtr->voice_detection()->is_enabled();
+
+ return(0);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetTypingDetectionStatus is not supported");
+ return(-1);
+#endif
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h
new file mode 100644
index 0000000..277731e
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
+
+#include "voe_audio_processing.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+
+namespace webrtc {
+
+class VoEAudioProcessingImpl : public virtual voe::SharedData,
+ public VoEAudioProcessing,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged);
+
+ virtual int GetNsStatus(bool& enabled, NsModes& mode);
+
+ virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged);
+
+ virtual int GetAgcStatus(bool& enabled, AgcModes& mode);
+
+ virtual int SetAgcConfig(const AgcConfig config);
+
+ virtual int GetAgcConfig(AgcConfig& config);
+
+ virtual int SetRxNsStatus(int channel,
+ bool enable,
+ NsModes mode = kNsUnchanged);
+
+ virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode);
+
+ virtual int SetRxAgcStatus(int channel,
+ bool enable,
+ AgcModes mode = kAgcUnchanged);
+
+ virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode);
+
+ virtual int SetRxAgcConfig(int channel, const AgcConfig config);
+
+ virtual int GetRxAgcConfig(int channel, AgcConfig& config);
+
+ virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged);
+
+ virtual int GetEcStatus(bool& enabled, EcModes& mode);
+
+ virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
+ bool enableCNG = true);
+
+ virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG);
+
+ virtual int RegisterRxVadObserver(int channel,
+ VoERxVadCallback& observer);
+
+ virtual int DeRegisterRxVadObserver(int channel);
+
+ virtual int VoiceActivityIndicator(int channel);
+
+ virtual int SetEcMetricsStatus(bool enable);
+
+ virtual int GetEcMetricsStatus(bool& enabled);
+
+ virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP);
+
+ virtual int GetEcDelayMetrics(int& delay_median, int& delay_std);
+
+ virtual int StartDebugRecording(const char* fileNameUTF8);
+
+ virtual int StopDebugRecording();
+
+ virtual int SetTypingDetectionStatus(bool enable);
+
+ virtual int GetTypingDetectionStatus(bool& enabled);
+
+protected:
+ VoEAudioProcessingImpl();
+ virtual ~VoEAudioProcessingImpl();
+
+private:
+ bool _isAecMode;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
+
diff --git a/trunk/src/voice_engine/main/source/voe_base_impl.cc b/trunk/src/voice_engine/main/source/voe_base_impl.cc
new file mode 100644
index 0000000..cdc0ee3
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_base_impl.cc
@@ -0,0 +1,1679 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_base_impl.h"
+
+#include "audio_coding_module.h"
+#include "audio_device_impl.h"
+#include "audio_processing.h"
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "output_mixer.h"
+#include "signal_processing_library.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "utility.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+#if (defined(_WIN32) && defined(_DLL) && (_MSC_VER == 1400))
+// Fix for VS 2005 MD/MDd link problem
+#include <stdio.h>
+extern "C"
+ { FILE _iob[3] = { __iob_func()[0], __iob_func()[1], __iob_func()[2]}; }
+#endif
+
+namespace webrtc
+{
+
+VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine)
+{
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEBaseImpl* d = s;
+ (*d)++;
+ return (d);
+}
+
+VoEBaseImpl::VoEBaseImpl() :
+ _voiceEngineObserverPtr(NULL),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _voiceEngineObserver(false), _oldVoEMicLevel(0), _oldMicLevel(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl() - ctor");
+}
+
+VoEBaseImpl::~VoEBaseImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "~VoEBaseImpl() - dtor");
+
+ TerminateInternal();
+
+ delete &_callbackCritSect;
+}
+
+int VoEBaseImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl reference counter = %d", refCount);
+ return (refCount);
+}
+
+void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
+{
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserver)
+ {
+ if (_voiceEngineObserverPtr)
+ {
+ int errCode(0);
+ if (error == AudioDeviceObserver::kRecordingError)
+ {
+ errCode = VE_RUNTIME_REC_ERROR;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::OnErrorIsReported() => "
+ "VE_RUNTIME_REC_ERROR");
+ }
+ else if (error == AudioDeviceObserver::kPlayoutError)
+ {
+ errCode = VE_RUNTIME_PLAY_ERROR;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::OnErrorIsReported() => "
+ "VE_RUNTIME_PLAY_ERROR");
+ }
+ // Deliver callback (-1 <=> no channel dependency)
+ _voiceEngineObserverPtr->CallbackOnError(-1, errCode);
+ }
+ }
+}
+
+void VoEBaseImpl::OnWarningIsReported(const WarningCode warning)
+{
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserver)
+ {
+ if (_voiceEngineObserverPtr)
+ {
+ int warningCode(0);
+ if (warning == AudioDeviceObserver::kRecordingWarning)
+ {
+ warningCode = VE_RUNTIME_REC_WARNING;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::OnErrorIsReported() => "
+ "VE_RUNTIME_REC_WARNING");
+ }
+ else if (warning == AudioDeviceObserver::kPlayoutWarning)
+ {
+ warningCode = VE_RUNTIME_PLAY_WARNING;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::OnErrorIsReported() => "
+ "VE_RUNTIME_PLAY_WARNING");
+ }
+ // Deliver callback (-1 <=> no channel dependency)
+ _voiceEngineObserverPtr->CallbackOnError(-1, warningCode);
+ }
+ }
+}
+
+WebRtc_Word32 VoEBaseImpl::RecordedDataIsAvailable(
+ const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord32 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord32 currentMicLevel,
+ WebRtc_UWord32& newMicLevel)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, "
+ "nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, "
+ "totalDelayMS=%u, clockDrift=%d, currentMicLevel=%u)",
+ nSamples, nBytesPerSample, nChannels, samplesPerSec,
+ totalDelayMS, clockDrift, currentMicLevel);
+
+ assert(_transmitMixerPtr != NULL);
+ assert(_audioDevicePtr != NULL);
+
+ bool isAnalogAGC(false);
+ WebRtc_UWord32 maxVolume(0);
+ WebRtc_UWord16 currentVoEMicLevel(0);
+ WebRtc_UWord32 newVoEMicLevel(0);
+
+ if (_audioProcessingModulePtr
+ && (_audioProcessingModulePtr->gain_control()->mode()
+ == GainControl::kAdaptiveAnalog))
+ {
+ isAnalogAGC = true;
+ }
+
+ // Will only deal with the volume in adaptive analog mode
+ if (isAnalogAGC)
+ {
+ // Scale from ADM to VoE level range
+ if (_audioDevicePtr->MaxMicrophoneVolume(&maxVolume) == 0)
+ {
+ if (0 != maxVolume)
+ {
+ currentVoEMicLevel = (WebRtc_UWord16) ((currentMicLevel
+ * kMaxVolumeLevel + (int) (maxVolume / 2))
+ / (maxVolume));
+ }
+ }
+ // We learned that on certain systems (e.g Linux) the currentVoEMicLevel
+ // can be greater than the maxVolumeLevel therefore
+ // we are going to cap the currentVoEMicLevel to the maxVolumeLevel
+ // and change the maxVolume to currentMicLevel if it turns out that
+ // the currentVoEMicLevel is indeed greater than the maxVolumeLevel.
+ if (currentVoEMicLevel > kMaxVolumeLevel)
+ {
+ currentVoEMicLevel = kMaxVolumeLevel;
+ maxVolume = currentMicLevel;
+ }
+ }
+
+ // Keep track if the MicLevel has been changed by the AGC, if not,
+ // use the old value AGC returns to let AGC continue its trend,
+ // so eventually the AGC is able to change the mic level. This handles
+ // issues with truncation introduced by the scaling.
+ if (_oldMicLevel == currentMicLevel)
+ {
+ currentVoEMicLevel = (WebRtc_UWord16) _oldVoEMicLevel;
+ }
+
+ // Perform channel-independent operations
+ // (APM, mix with file, record to file, mute, etc.)
+ _transmitMixerPtr->PrepareDemux(audioSamples, nSamples, nChannels,
+ samplesPerSec,
+ (WebRtc_UWord16) totalDelayMS, clockDrift,
+ currentVoEMicLevel);
+
+ // Copy the audio frame to each sending channel and perform
+ // channel-dependent operations (file mixing, mute, etc.) to prepare
+ // for encoding.
+ _transmitMixerPtr->DemuxAndMix();
+ // Do the encoding and packetize+transmit the RTP packet when encoding
+ // is done.
+ _transmitMixerPtr->EncodeAndSend();
+
+ // Will only deal with the volume in adaptive analog mode
+ if (isAnalogAGC)
+ {
+ // Scale from VoE to ADM level range
+ newVoEMicLevel = _transmitMixerPtr->CaptureLevel();
+ if (newVoEMicLevel != currentVoEMicLevel)
+ {
+ // Add (kMaxVolumeLevel/2) to round the value
+ newMicLevel = (WebRtc_UWord32) ((newVoEMicLevel * maxVolume
+ + (int) (kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
+ }
+ else
+ {
+ // Pass zero if the level is unchanged
+ newMicLevel = 0;
+ }
+
+ // Keep track of the value AGC returns
+ _oldVoEMicLevel = newVoEMicLevel;
+ _oldMicLevel = currentMicLevel;
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ WebRtc_Word8* audioSamples,
+ WebRtc_UWord32& nSamplesOut)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
+ "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
+ nSamples, nBytesPerSample, nChannels, samplesPerSec);
+
+ assert(_outputMixerPtr != NULL);
+
+ // Perform mixing of all active participants (channel-based mixing)
+ _outputMixerPtr->MixActiveChannels();
+
+ // Additional operations on the combined signal
+ _outputMixerPtr->DoOperationsOnCombinedSignal();
+
+ // Retrieve the final output mix (resampled to match the ADM)
+ _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, _audioFrame);
+
+ assert(nSamples == _audioFrame._payloadDataLengthInSamples);
+ assert(samplesPerSec ==
+ static_cast<WebRtc_UWord32>(_audioFrame._frequencyInHz));
+
+ // Deliver audio (PCM) samples to the ADM
+ memcpy(
+ (WebRtc_Word16*) audioSamples,
+ (const WebRtc_Word16*) _audioFrame._payloadData,
+ sizeof(WebRtc_Word16) * (_audioFrame._payloadDataLengthInSamples
+ * _audioFrame._audioChannel));
+
+ nSamplesOut = _audioFrame._payloadDataLengthInSamples;
+
+ return 0;
+}
+
+int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "RegisterVoiceEngineObserver(observer=0x%d)", &observer);
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
+ "RegisterVoiceEngineObserver() observer"
+ " already enabled");
+ return -1;
+ }
+
+ // Register the observer in all active channels
+ voe::ScopedChannel sc(_channelManager);
+ void* iterator(NULL);
+ voe::Channel* channelPtr = sc.GetFirstChannel(iterator);
+ while (channelPtr != NULL)
+ {
+ channelPtr->RegisterVoiceEngineObserver(observer);
+ channelPtr = sc.GetNextChannel(iterator);
+ }
+ _transmitMixerPtr->RegisterVoiceEngineObserver(observer);
+
+ _voiceEngineObserverPtr = &observer;
+ _voiceEngineObserver = true;
+
+ return 0;
+}
+
+int VoEBaseImpl::DeRegisterVoiceEngineObserver()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "DeRegisterVoiceEngineObserver()");
+ CriticalSectionScoped cs(_callbackCritSect);
+ if (!_voiceEngineObserverPtr)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
+ "DeRegisterVoiceEngineObserver() observer already disabled");
+ return 0;
+ }
+
+ _voiceEngineObserver = false;
+ _voiceEngineObserverPtr = NULL;
+
+ // Deregister the observer in all active channels
+ voe::ScopedChannel sc(_channelManager);
+ void* iterator(NULL);
+ voe::Channel* channelPtr = sc.GetFirstChannel(iterator);
+ while (channelPtr != NULL)
+ {
+ channelPtr->DeRegisterVoiceEngineObserver();
+ channelPtr = sc.GetNextChannel(iterator);
+ }
+
+ return 0;
+}
+
+int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "Init(external_adm=0x%p)", external_adm);
+ CriticalSectionScoped cs(*_apiCritPtr);
+
+ if (_engineStatistics.Initialized())
+ {
+ return 0;
+ }
+
+ if (_moduleProcessThreadPtr)
+ {
+ if (_moduleProcessThreadPtr->Start() != 0)
+ {
+ _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
+ "Init() failed to start module process thread");
+ return -1;
+ }
+ }
+
+ // Create an internal ADM if the user has not added an external
+ // ADM implementation as input to Init().
+ if (external_adm == NULL)
+ {
+ // Create the internal ADM implementation.
+ _audioDevicePtr = AudioDeviceModuleImpl::Create(
+ VoEId(_instanceId, -1), _audioDeviceLayer);
+
+ if (_audioDevicePtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
+ "Init() failed to create the ADM");
+ return -1;
+ }
+ }
+ else
+ {
+ // Use the already existing external ADM implementation.
+ _audioDevicePtr = external_adm;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "An external ADM implementation will be used in VoiceEngine");
+ }
+
+ // Increase the reference counter for both external and internal usage.
+ _audioDevicePtr->AddRef();
+
+ // Register the ADM to the process thread, which will drive the error
+ // callback mechanism
+ if (_moduleProcessThreadPtr &&
+ _moduleProcessThreadPtr->RegisterModule(_audioDevicePtr) != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError,
+ "Init() failed to register the ADM");
+ return -1;
+ }
+
+ bool available(false);
+
+ // --------------------
+ // Reinitialize the ADM
+
+ // Register the AudioObserver implementation
+ if (_audioDevicePtr->RegisterEventObserver(this) != 0) {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceWarning,
+ "Init() failed to register event observer "
+ "for the ADM");
+ }
+
+ // Register the AudioTransport implementation
+ if (_audioDevicePtr->RegisterAudioCallback(this) != 0) {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceWarning,
+ "Init() failed to register audio callback "
+ "for the ADM");
+ }
+
+ // ADM initialization
+ if (_audioDevicePtr->Init() != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError,
+ "Init() failed to initialize the ADM");
+ return -1;
+ }
+
+ // Initialize the default speaker
+ if (_audioDevicePtr->SetPlayoutDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
+ "Init() failed to set the default output device");
+ }
+ if (_audioDevicePtr->SpeakerIsAvailable(&available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
+ "Init() failed to check speaker availability, trying to "
+ "initialize speaker anyway");
+ }
+ else if (!available)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
+ "Init() speaker not available, trying to initialize speaker "
+ "anyway");
+ }
+ if (_audioDevicePtr->InitSpeaker() != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
+ "Init() failed to initialize the speaker");
+ }
+
+ // Initialize the default microphone
+ if (_audioDevicePtr->SetRecordingDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
+ "Init() failed to set the default input device");
+ }
+ if (_audioDevicePtr->MicrophoneIsAvailable(&available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
+ "Init() failed to check microphone availability, trying to "
+ "initialize microphone anyway");
+ }
+ else if (!available)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
+ "Init() microphone not available, trying to initialize "
+ "microphone anyway");
+ }
+ if (_audioDevicePtr->InitMicrophone() != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
+ "Init() failed to initialize the microphone");
+ }
+
+ // Set number of channels
+ if (_audioDevicePtr->StereoPlayoutIsAvailable(&available) != 0) {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+ "Init() failed to query stereo playout "
+ "mode");
+ }
+ if (_audioDevicePtr->SetStereoPlayout(available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+ "Init() failed to set mono/stereo playout mode");
+ }
+
+ // TODO(andrew): These functions don't tell us whether stereo recording
+ // is truly available. We simply set the AudioProcessing input to stereo
+ // here, because we have to wait until receiving the first frame to
+ // determine the actual number of channels anyway.
+ //
+ // These functions may be changed; tracked here:
+ // http://code.google.com/p/webrtc/issues/detail?id=204
+ _audioDevicePtr->StereoRecordingIsAvailable(&available);
+ if (_audioDevicePtr->SetStereoRecording(available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+ "Init() failed to set mono/stereo recording mode");
+ }
+
+ // APM initialization done after sound card since we need
+ // to know if we support stereo recording or not.
+
+ // Create the AudioProcessing Module if it does not exist.
+
+ if (_audioProcessingModulePtr == NULL)
+ {
+ _audioProcessingModulePtr = AudioProcessing::Create(
+ VoEId(_instanceId, -1));
+ if (_audioProcessingModulePtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
+ "Init() failed to create the AP module");
+ return -1;
+ }
+ // Ensure that mixers in both directions has access to the created APM
+ _transmitMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
+ _outputMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
+
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ set_device_sample_rate_hz(
+ kVoiceEngineAudioProcessingDeviceSampleRateHz))
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set the device sample rate to 48K for AP "
+ " module");
+ return -1;
+ }
+ // Using 8 kHz as inital Fs. Might be changed already at first call.
+ if (_audioProcessingModulePtr->set_sample_rate_hz(8000))
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set the sample rate to 8K for AP module");
+ return -1;
+ }
+
+ // Assume mono output until a send codec is set, and stereo input until
+ // we receive the first captured frame. We set stereo input here to
+ // avoid triggering a possible error in SetSendCodec when a stereo
+ // codec is selected.
+ if (_audioProcessingModulePtr->set_num_channels(2, 1) != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
+ "Init() failed to set channels for the primary audio stream");
+ return -1;
+ }
+
+ if (_audioProcessingModulePtr->set_num_reverse_channels(1) != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
+ "Init() failed to set channels for the primary audio stream");
+ return -1;
+ }
+ // high-pass filter
+ if (_audioProcessingModulePtr->high_pass_filter()->Enable(
+ WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE) != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set the high-pass filter for AP module");
+ return -1;
+ }
+ // Echo Cancellation
+ if (_audioProcessingModulePtr->echo_cancellation()->
+ enable_drift_compensation(false) != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set drift compensation for AP module");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->echo_cancellation()->Enable(
+ WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE))
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set echo cancellation state for AP module");
+ return -1;
+ }
+ // Noise Reduction
+ if (_audioProcessingModulePtr->noise_suppression()->set_level(
+ (NoiseSuppression::Level) WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set noise reduction level for AP module");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->noise_suppression()->Enable(
+ WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE) != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set noise reduction state for AP module");
+ return -1;
+ }
+ // Automatic Gain control
+ if (_audioProcessingModulePtr->gain_control()->set_analog_level_limits(
+ kMinVolumeLevel,kMaxVolumeLevel) != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set AGC analog level for AP module");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->gain_control()->set_mode(
+ (GainControl::Mode) WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set AGC mode for AP module");
+ return -1;
+ }
+ if (_audioProcessingModulePtr->gain_control()->Enable(
+ WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set AGC state for AP module");
+ return -1;
+ }
+ // VAD
+ if (_audioProcessingModulePtr->voice_detection()->Enable(
+ WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "Init() failed to set VAD state for AP module");
+ return -1;
+ }
+ }
+
+ // Set default AGC mode for the ADM
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ bool enable(false);
+ if (_audioProcessingModulePtr->gain_control()->mode()
+ != GainControl::kFixedDigital)
+ {
+ enable = _audioProcessingModulePtr->gain_control()->is_enabled();
+ // Only set the AGC mode for the ADM when Adaptive AGC mode is selected
+ if (_audioDevicePtr->SetAGC(enable) != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError, "Init() failed to set default AGC mode in ADM 0");
+ }
+ }
+#endif
+
+ return _engineStatistics.SetInitialized();
+}
+
+int VoEBaseImpl::Terminate()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "Terminate()");
+ CriticalSectionScoped cs(*_apiCritPtr);
+ return TerminateInternal();
+}
+
+int VoEBaseImpl::MaxNumOfChannels()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "MaxNumOfChannels()");
+ WebRtc_Word32 maxNumOfChannels = _channelManager.MaxNumOfChannels();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "MaxNumOfChannels() => %d", maxNumOfChannels);
+ return (maxNumOfChannels);
+}
+
+int VoEBaseImpl::CreateChannel()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "CreateChannel()");
+ CriticalSectionScoped cs(*_apiCritPtr);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ WebRtc_Word32 channelId = -1;
+
+ if (!_channelManager.CreateChannel(channelId))
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
+ "CreateChannel() failed to allocate "
+ "memory for channel");
+ return -1;
+ }
+
+ bool destroyChannel(false);
+ {
+ voe::ScopedChannel sc(_channelManager, channelId);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
+ "CreateChannel() failed to allocate"
+ " memory for channel");
+ return -1;
+ }
+ else if (channelPtr->SetEngineInformation(_engineStatistics,
+ *_outputMixerPtr,
+ *_transmitMixerPtr,
+ *_moduleProcessThreadPtr,
+ *_audioDevicePtr,
+ _voiceEngineObserverPtr,
+ &_callbackCritSect) != 0)
+ {
+ destroyChannel = true;
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
+ "CreateChannel() failed to "
+ "associate engine and channel."
+ " Destroying channel.");
+ }
+ else if (channelPtr->Init() != 0)
+ {
+ destroyChannel = true;
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
+ "CreateChannel() failed to "
+ "initialize channel. Destroying"
+ " channel.");
+ }
+ }
+ if (destroyChannel)
+ {
+ _channelManager.DestroyChannel(channelId);
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "CreateChannel() => %d", channelId);
+ return channelId;
+}
+
+int VoEBaseImpl::DeleteChannel(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "DeleteChannel(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeleteChannel() failed to locate "
+ "channel");
+ return -1;
+ }
+ }
+
+ if (_channelManager.DestroyChannel(channel) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeleteChannel() failed to destroy "
+ "channel");
+ return -1;
+ }
+
+ if (StopSend() != 0)
+ {
+ return -1;
+ }
+
+ if (StopPlayout() != 0)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+int VoEBaseImpl::SetLocalReceiver(int channel, int port, int RTCPport,
+ const char ipAddr[64],
+ const char multiCastAddr[64])
+{
+ // Inititialize local receive sockets (RTP and RTCP).
+ //
+ // The sockets are always first closed and then created again by this
+ // function call. The created sockets are by default also used for
+ // transmission (unless source port is set in SetSendDestination).
+ //
+ // Note that, sockets can also be created automatically if a user calls
+ // SetSendDestination and StartSend without having called SetLocalReceiver
+ // first. The sockets are then created at the first packet transmission.
+
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (ipAddr == NULL && multiCastAddr == NULL)
+ {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d)",
+ channel, port, RTCPport);
+ }
+ else if (ipAddr != NULL && multiCastAddr == NULL)
+ {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
+ "ipAddr=%s)", channel, port, RTCPport, ipAddr);
+ }
+ else if (ipAddr == NULL && multiCastAddr != NULL)
+ {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
+ "multiCastAddr=%s)", channel, port, RTCPport,
+ multiCastAddr);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
+ "ipAddr=%s, multiCastAddr=%s)", channel, port,
+ RTCPport, ipAddr, multiCastAddr);
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if ((port < 0) || (port > 65535))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
+ "SetLocalReceiver() invalid RTP port");
+ return -1;
+ }
+ if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
+ != kVoEDefault) && (RTCPport > 65535)))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
+ "SetLocalReceiver() invalid RTCP port");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetLocalReceiver() failed to locate "
+ "channel");
+ return -1;
+ }
+
+ // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in
+ // the module, which is the default.
+ WebRtc_UWord16 rtcpPortUW16(0);
+ if (RTCPport != kVoEDefault)
+ {
+ rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
+ }
+
+ return channelPtr->SetLocalReceiver(port, rtcpPortUW16, ipAddr,
+ multiCastAddr);
+#else
+ _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
+ kTraceWarning,
+ "SetLocalReceiver() VoE is built for "
+ "external transport");
+ return -1;
+#endif
+}
+
+int VoEBaseImpl::GetLocalReceiver(int channel, int& port, int& RTCPport,
+ char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLocalReceiver(channel=%d, ipAddr[]=?)", channel);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetLocalReceiver() failed to locate "
+ "channel");
+ return -1;
+ }
+ WebRtc_Word32 ret = channelPtr->GetLocalReceiver(port, RTCPport, ipAddr);
+ if (ipAddr != NULL)
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLocalReceiver() => port=%d, RTCPport=%d, ipAddr=%s",
+ port, RTCPport, ipAddr);
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLocalReceiver() => port=%d, RTCPport=%d", port,
+ RTCPport);
+ }
+ return ret;
+#else
+ _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
+ kTraceWarning,
+ "SetLocalReceiver() VoE is built for "
+ "external transport");
+ return -1;
+#endif
+}
+
+int VoEBaseImpl::SetSendDestination(int channel, int port, const char* ipaddr,
+ int sourcePort, int RTCPport)
+{
+ WEBRTC_TRACE(
+ kTraceApiCall,
+ kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetSendDestination(channel=%d, port=%d, ipaddr=%s,"
+ "sourcePort=%d, RTCPport=%d)",
+ channel, port, ipaddr, sourcePort, RTCPport);
+ CriticalSectionScoped cs(*_apiCritPtr);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSendDestination() failed to locate "
+ "channel");
+ return -1;
+ }
+ if ((port < 0) || (port > 65535))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
+ "SetSendDestination() invalid RTP port");
+ return -1;
+ }
+ if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
+ != kVoEDefault) && (RTCPport > 65535)))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
+ "SetSendDestination() invalid RTCP "
+ "port");
+ return -1;
+ }
+ if (((sourcePort != kVoEDefault) && (sourcePort < 0)) || ((sourcePort
+ != kVoEDefault) && (sourcePort > 65535)))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
+ "SetSendDestination() invalid source "
+ "port");
+ return -1;
+ }
+
+ // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in the
+ // module, which is the default.
+ WebRtc_UWord16 rtcpPortUW16(0);
+ if (RTCPport != kVoEDefault)
+ {
+ rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
+ WEBRTC_TRACE(
+ kTraceInfo,
+ kTraceVoice,
+ VoEId(_instanceId, channel),
+ "SetSendDestination() non default RTCP port %u will be "
+ "utilized",
+ rtcpPortUW16);
+ }
+
+ return channelPtr->SetSendDestination(port, ipaddr, sourcePort,
+ rtcpPortUW16);
+#else
+ _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
+ kTraceWarning,
+ "SetSendDestination() VoE is built for "
+ "external transport");
+ return -1;
+#endif
+}
+
+int VoEBaseImpl::GetSendDestination(int channel, int& port, char ipAddr[64],
+ int& sourcePort, int& RTCPport)
+{
+ WEBRTC_TRACE(
+ kTraceApiCall,
+ kTraceVoice,
+ VoEId(_instanceId, -1),
+ "GetSendDestination(channel=%d, ipAddr[]=?, sourcePort=?,"
+ "RTCPport=?)",
+ channel);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendDestination() failed to locate "
+ "channel");
+ return -1;
+ }
+ WebRtc_Word32 ret = channelPtr->GetSendDestination(port, ipAddr,
+ sourcePort, RTCPport);
+ if (ipAddr != NULL)
+ {
+ WEBRTC_TRACE(
+ kTraceStateInfo,
+ kTraceVoice,
+ VoEId(_instanceId, -1),
+ "GetSendDestination() => port=%d, RTCPport=%d, ipAddr=%s, "
+ "sourcePort=%d, RTCPport=%d",
+ port, RTCPport, ipAddr, sourcePort, RTCPport);
+ }
+ else
+ {
+ WEBRTC_TRACE(
+ kTraceStateInfo,
+ kTraceVoice,
+ VoEId(_instanceId, -1),
+ "GetSendDestination() => port=%d, RTCPport=%d, "
+ "sourcePort=%d, RTCPport=%d",
+ port, RTCPport, sourcePort, RTCPport);
+ }
+ return ret;
+#else
+ _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
+ kTraceWarning,
+ "GetSendDestination() VoE is built for "
+ "external transport");
+ return -1;
+#endif
+}
+
+int VoEBaseImpl::StartReceive(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StartReceive(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartReceive() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->StartReceiving();
+}
+
+int VoEBaseImpl::StopReceive(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StopListen(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetLocalReceiver() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->StopReceiving();
+}
+
+int VoEBaseImpl::StartPlayout(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StartPlayout(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartPlayout() failed to locate "
+ "channel");
+ return -1;
+ }
+ if (channelPtr->Playing())
+ {
+ return 0;
+ }
+ if (StartPlayout() != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError,
+ "StartPlayout() failed to start "
+ "playout");
+ return -1;
+ }
+ return channelPtr->StartPlayout();
+}
+
+int VoEBaseImpl::StopPlayout(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StopPlayout(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopPlayout() failed to locate "
+ "channel");
+ return -1;
+ }
+ if (channelPtr->StopPlayout() != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StopPlayout() failed to stop playout for channel %d",
+ channel);
+ }
+ return StopPlayout();
+}
+
+int VoEBaseImpl::StartSend(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StartSend(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartSend() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->Sending())
+ {
+ return 0;
+ }
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!channelPtr->ExternalTransport()
+ && !channelPtr->SendSocketsInitialized())
+ {
+ _engineStatistics.SetLastError(VE_DESTINATION_NOT_INITED, kTraceError,
+ "StartSend() must set send destination "
+ "first");
+ return -1;
+ }
+#endif
+ if (StartSend() != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError,
+ "StartSend() failed to start recording");
+ return -1;
+ }
+ return channelPtr->StartSend();
+}
+
+int VoEBaseImpl::StopSend(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "StopSend(channel=%d)", channel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopSend() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->StopSend() != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StopSend() failed to stop sending for channel %d",
+ channel);
+ }
+ return StopSend();
+}
+
+int VoEBaseImpl::GetVersion(char version[1024])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetVersion(version=?)");
+ assert(kVoiceEngineVersionMaxMessageSize == 1024);
+
+ if (version == NULL)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError);
+ return (-1);
+ }
+
+ char versionBuf[kVoiceEngineVersionMaxMessageSize];
+ char* versionPtr = versionBuf;
+
+ WebRtc_Word32 len = 0;
+ WebRtc_Word32 accLen = 0;
+
+ len = AddVoEVersion(versionPtr);
+ if (len == -1)
+ {
+ return -1;
+ }
+ versionPtr += len;
+ accLen += len;
+ assert(accLen < kVoiceEngineVersionMaxMessageSize);
+
+ len = AddBuildInfo(versionPtr);
+ if (len == -1)
+ {
+ return -1;
+ }
+ versionPtr += len;
+ accLen += len;
+ assert(accLen < kVoiceEngineVersionMaxMessageSize);
+
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ len = AddExternalTransportBuild(versionPtr);
+ if (len == -1)
+ {
+ return -1;
+ }
+ versionPtr += len;
+ accLen += len;
+ assert(accLen < kVoiceEngineVersionMaxMessageSize);
+#endif
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ len = AddExternalRecAndPlayoutBuild(versionPtr);
+ if (len == -1)
+ {
+ return -1;
+ }
+ versionPtr += len;
+ accLen += len;
+ assert(accLen < kVoiceEngineVersionMaxMessageSize);
+ #endif
+
+ memcpy(version, versionBuf, accLen);
+ version[accLen] = '\0';
+
+ // to avoid the truncation in the trace, split the string into parts
+ char partOfVersion[256];
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetVersion() =>");
+ for (int partStart = 0; partStart < accLen;)
+ {
+ memset(partOfVersion, 0, sizeof(partOfVersion));
+ int partEnd = partStart + 180;
+ while (version[partEnd] != '\n' && version[partEnd] != '\0')
+ {
+ partEnd--;
+ }
+ if (partEnd < accLen)
+ {
+ memcpy(partOfVersion, &version[partStart], partEnd - partStart);
+ }
+ else
+ {
+ memcpy(partOfVersion, &version[partStart], accLen - partStart);
+ }
+ partStart = partEnd;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "%s", partOfVersion);
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::AddBuildInfo(char* str) const
+{
+ return sprintf(str, "Build: %s\n", BUILDINFO);
+}
+
+WebRtc_Word32 VoEBaseImpl::AddVoEVersion(char* str) const
+{
+ return sprintf(str, "VoiceEngine 4.1.0\n");
+}
+
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+WebRtc_Word32 VoEBaseImpl::AddExternalTransportBuild(char* str) const
+{
+ return sprintf(str, "External transport build\n");
+}
+#endif
+
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+WebRtc_Word32 VoEBaseImpl::AddExternalRecAndPlayoutBuild(char* str) const
+{
+ return sprintf(str, "External recording and playout build\n");
+}
+#endif
+
+int VoEBaseImpl::LastError()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "LastError()");
+ return (_engineStatistics.LastError());
+}
+
+
+int VoEBaseImpl::SetNetEQPlayoutMode(int channel, NetEqModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetNetEQPlayoutMode(channel=%i, mode=%i)", channel, mode);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetNetEQPlayoutMode() failed to locate"
+ " channel");
+ return -1;
+ }
+ return channelPtr->SetNetEQPlayoutMode(mode);
+}
+
+int VoEBaseImpl::GetNetEQPlayoutMode(int channel, NetEqModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetNetEQPlayoutMode(channel=%i, mode=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetNetEQPlayoutMode() failed to locate"
+ " channel");
+ return -1;
+ }
+ return channelPtr->GetNetEQPlayoutMode(mode);
+}
+
+int VoEBaseImpl::SetNetEQBGNMode(int channel, NetEqBgnModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetNetEQBGNMode(channel=%i, mode=%i)", channel, mode);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetNetEQBGNMode() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetNetEQBGNMode(mode);
+}
+
+int VoEBaseImpl::GetNetEQBGNMode(int channel, NetEqBgnModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetNetEQBGNMode(channel=%i, mode=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetNetEQBGNMode() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->GetNetEQBGNMode(mode);
+}
+
+int VoEBaseImpl::SetOnHoldStatus(int channel, bool enable, OnHoldModes mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetOnHoldStatus(channel=%d, enable=%d, mode=%d)", channel,
+ enable, mode);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetOnHoldStatus() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetOnHoldStatus(enable, mode);
+}
+
+int VoEBaseImpl::GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetOnHoldStatus(channel=%d, enabled=?, mode=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetOnHoldStatus() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->GetOnHoldStatus(enabled, mode);
+}
+
+WebRtc_Word32 VoEBaseImpl::StartPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::StartPlayout()");
+ if (_audioDevicePtr->Playing())
+ {
+ return 0;
+ }
+ if (!_externalPlayout)
+ {
+ if (_audioDevicePtr->InitPlayout() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StartPlayout() failed to initialize playout");
+ return -1;
+ }
+ if (_audioDevicePtr->StartPlayout() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StartPlayout() failed to start playout");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::StopPlayout()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::StopPlayout()");
+
+ WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
+ if (numOfChannels <= 0)
+ {
+ return 0;
+ }
+
+ WebRtc_UWord16 nChannelsPlaying(0);
+ WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
+
+ // Get number of playing channels
+ _channelManager.GetChannelIds(channelsArray, numOfChannels);
+ for (int i = 0; i < numOfChannels; i++)
+ {
+ voe::ScopedChannel sc(_channelManager, channelsArray[i]);
+ voe::Channel* chPtr = sc.ChannelPtr();
+ if (chPtr)
+ {
+ if (chPtr->Playing())
+ {
+ nChannelsPlaying++;
+ }
+ }
+ }
+ delete[] channelsArray;
+
+ // Stop audio-device playing if no channel is playing out
+ if (nChannelsPlaying == 0)
+ {
+ if (_audioDevicePtr->StopPlayout() != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError,
+ "StopPlayout() failed to stop "
+ "playout");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::StartSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::StartSend()");
+ if (_audioDevicePtr->Recording())
+ {
+ return 0;
+ }
+ if (!_externalRecording)
+ {
+ if (_audioDevicePtr->InitRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StartSend() failed to initialize recording");
+ return -1;
+ }
+ if (_audioDevicePtr->StartRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StartSend() failed to start recording");
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::StopSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::StopSend()");
+
+ if ((NumOfSendingChannels() == 0) && !_transmitMixerPtr->IsRecordingMic())
+ {
+ // Stop audio-device recording if no channel is recording
+ if (_audioDevicePtr->StopRecording() != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_STOP_RECORDING,
+ kTraceError,
+ "StopSend() failed to stop "
+ "recording");
+ return -1;
+ }
+ _transmitMixerPtr->StopSend();
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 VoEBaseImpl::TerminateInternal()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEBaseImpl::TerminateInternal()");
+
+ // Delete any remaining channel objects
+ WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
+ if (numOfChannels > 0)
+ {
+ WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
+ _channelManager.GetChannelIds(channelsArray, numOfChannels);
+ for (int i = 0; i < numOfChannels; i++)
+ {
+ DeleteChannel(channelsArray[i]);
+ }
+ delete[] channelsArray;
+ }
+
+ if (_moduleProcessThreadPtr)
+ {
+ if (_audioDevicePtr)
+ {
+ if (_moduleProcessThreadPtr->DeRegisterModule(_audioDevicePtr) != 0)
+ {
+ _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
+ "TerminateInternal() failed to "
+ "deregister ADM");
+ }
+ }
+ if (_moduleProcessThreadPtr->Stop() != 0)
+ {
+ _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
+ "TerminateInternal() failed to stop "
+ "module process thread");
+ }
+ }
+
+ // Audio Device Module
+
+ if (_audioDevicePtr != NULL)
+ {
+ if (_audioDevicePtr->StopPlayout() != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+ "TerminateInternal() failed to stop "
+ "playout");
+ }
+ if (_audioDevicePtr->StopRecording() != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+ "TerminateInternal() failed to stop "
+ "recording");
+ }
+ if (_audioDevicePtr->RegisterEventObserver(NULL) != 0) {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceWarning,
+ "TerminateInternal() failed to de-"
+ "register event observer for the ADM");
+ }
+ if (_audioDevicePtr->RegisterAudioCallback(NULL) != 0) {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceWarning,
+ "TerminateInternal() failed to de-"
+ "register audio callback for the ADM");
+ }
+ if (_audioDevicePtr->Terminate() != 0)
+ {
+ _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
+ kTraceError,
+ "TerminateInternal() failed to "
+ "terminate the ADM");
+ }
+
+ _audioDevicePtr->Release();
+ _audioDevicePtr = NULL;
+ }
+
+ // AP module
+
+ if (_audioProcessingModulePtr != NULL)
+ {
+ _transmitMixerPtr->SetAudioProcessingModule(NULL);
+ AudioProcessing::Destroy(_audioProcessingModulePtr);
+ _audioProcessingModulePtr = NULL;
+ }
+
+ return _engineStatistics.SetUnInitialized();
+}
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_base_impl.h b/trunk/src/voice_engine/main/source/voe_base_impl.h
new file mode 100644
index 0000000..6d470a0
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_base_impl.h
@@ -0,0 +1,152 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
+
+#include "voe_base.h"
+
+#include "module_common_types.h"
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc
+{
+
+class ProcessThread;
+
+class VoEBaseImpl: public virtual voe::SharedData,
+ public VoEBase,
+ public voe::RefCount,
+ public AudioTransport,
+ public AudioDeviceObserver
+{
+public:
+ virtual int Release();
+
+ virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
+
+ virtual int DeRegisterVoiceEngineObserver();
+
+ virtual int Init(AudioDeviceModule* external_adm = NULL);
+
+ virtual int Terminate();
+
+ virtual int MaxNumOfChannels();
+
+ virtual int CreateChannel();
+
+ virtual int DeleteChannel(int channel);
+
+ virtual int SetLocalReceiver(int channel, int port,
+ int RTCPport = kVoEDefault,
+ const char ipAddr[64] = NULL,
+ const char multiCastAddr[64] = NULL);
+
+ virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
+ char ipAddr[64]);
+
+ virtual int SetSendDestination(int channel, int port,
+ const char ipAddr[64],
+ int sourcePort = kVoEDefault,
+ int RTCPport = kVoEDefault);
+
+ virtual int GetSendDestination(int channel,
+ int& port,
+ char ipAddr[64],
+ int& sourcePort,
+ int& RTCPport);
+
+ virtual int StartReceive(int channel);
+
+ virtual int StartPlayout(int channel);
+
+ virtual int StartSend(int channel);
+
+ virtual int StopReceive(int channel);
+
+ virtual int StopPlayout(int channel);
+
+ virtual int StopSend(int channel);
+
+ virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode);
+
+ virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode);
+
+ virtual int SetNetEQBGNMode(int channel, NetEqBgnModes mode);
+
+ virtual int GetNetEQBGNMode(int channel, NetEqBgnModes& mode);
+
+
+ virtual int SetOnHoldStatus(int channel,
+ bool enable,
+ OnHoldModes mode = kHoldSendAndPlay);
+
+ virtual int GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode);
+
+ virtual int GetVersion(char version[1024]);
+
+ virtual int LastError();
+
+ // AudioTransport
+ virtual WebRtc_Word32
+ RecordedDataIsAvailable(const WebRtc_Word8* audioSamples,
+ const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ const WebRtc_UWord32 totalDelayMS,
+ const WebRtc_Word32 clockDrift,
+ const WebRtc_UWord32 currentMicLevel,
+ WebRtc_UWord32& newMicLevel);
+
+ virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
+ const WebRtc_UWord8 nBytesPerSample,
+ const WebRtc_UWord8 nChannels,
+ const WebRtc_UWord32 samplesPerSec,
+ WebRtc_Word8* audioSamples,
+ WebRtc_UWord32& nSamplesOut);
+
+ // AudioDeviceObserver
+ virtual void OnErrorIsReported(const ErrorCode error);
+ virtual void OnWarningIsReported(const WarningCode warning);
+
+protected:
+ VoEBaseImpl();
+ virtual ~VoEBaseImpl();
+
+private:
+ WebRtc_Word32 StartPlayout();
+ WebRtc_Word32 StopPlayout();
+ WebRtc_Word32 StartSend();
+ WebRtc_Word32 StopSend();
+ WebRtc_Word32 TerminateInternal();
+
+ WebRtc_Word32 AddBuildInfo(char* str) const;
+ WebRtc_Word32 AddVoEVersion(char* str) const;
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ WebRtc_Word32 AddExternalTransportBuild(char* str) const;
+#endif
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ WebRtc_Word32 AddExternalRecAndPlayoutBuild(char* str) const;
+#endif
+ VoiceEngineObserver* _voiceEngineObserverPtr;
+ CriticalSectionWrapper& _callbackCritSect;
+
+ bool _voiceEngineObserver;
+ WebRtc_UWord32 _oldVoEMicLevel;
+ WebRtc_UWord32 _oldMicLevel;
+ AudioFrame _audioFrame;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_call_report_impl.cc b/trunk/src/voice_engine/main/source/voe_call_report_impl.cc
new file mode 100644
index 0000000..8be8e01
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_call_report_impl.cc
@@ -0,0 +1,432 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_call_report_impl.h"
+
+#include "audio_processing.h"
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc
+{
+
+VoECallReport* VoECallReport::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoECallReportImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+
+VoECallReportImpl::VoECallReportImpl() :
+ _file(*FileWrapper::Create())
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECallReportImpl() - ctor");
+}
+
+VoECallReportImpl::~VoECallReportImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "~VoECallReportImpl() - dtor");
+ delete &_file;
+}
+
+int VoECallReportImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECallReportImpl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECallReportImpl reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoECallReportImpl::ResetCallReportStatistics(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "ResetCallReportStatistics(channel=%d)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ assert(_audioProcessingModulePtr != NULL);
+
+ bool echoMode =
+ _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " current AudioProcessingModule echo metric state %d)",
+ echoMode);
+ // Reset the APM statistics
+ if (_audioProcessingModulePtr->echo_cancellation()->enable_metrics(true)
+ != 0)
+ {
+ _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
+ "ResetCallReportStatistics() unable to "
+ "set the AudioProcessingModule echo "
+ "metrics state");
+ return -1;
+ }
+ // Restore metric states
+ _audioProcessingModulePtr->echo_cancellation()->enable_metrics(echoMode);
+
+ // Reset channel dependent statistics
+ if (channel != -1)
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "ResetCallReportStatistics() failed "
+ "to locate channel");
+ return -1;
+ }
+ channelPtr->ResetDeadOrAliveCounters();
+ channelPtr->ResetRTCPStatistics();
+ }
+ else
+ {
+ WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
+ if (numOfChannels <= 0)
+ {
+ return 0;
+ }
+ WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
+ _channelManager.GetChannelIds(channelsArray, numOfChannels);
+ for (int i = 0; i < numOfChannels; i++)
+ {
+ voe::ScopedChannel sc(_channelManager, channelsArray[i]);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr)
+ {
+ channelPtr->ResetDeadOrAliveCounters();
+ channelPtr->ResetRTCPStatistics();
+ }
+ }
+ delete[] channelsArray;
+ }
+
+ return 0;
+}
+
+int VoECallReportImpl::GetEchoMetricSummary(EchoStatistics& stats)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetEchoMetricSummary()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ assert(_audioProcessingModulePtr != NULL);
+
+ return (GetEchoMetricSummaryInternal(stats));
+}
+
+int VoECallReportImpl::GetEchoMetricSummaryInternal(EchoStatistics& stats)
+{
+ // Retrieve echo metrics from the AudioProcessingModule
+ int ret(0);
+ bool mode(false);
+ EchoCancellation::Metrics metrics;
+
+ // Ensure that echo metrics is enabled
+
+ mode =
+ _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
+ if (mode != false)
+ {
+ ret =
+ _audioProcessingModulePtr->echo_cancellation()->GetMetrics(&metrics);
+ if (ret != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ " AudioProcessingModule GetMetrics() => error");
+ }
+ }
+ else
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ " AudioProcessingModule echo metrics is not enabled");
+ }
+
+ if ((ret != 0) || (mode == false))
+ {
+ // Mark complete struct as invalid (-100 dB)
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ " unable to retrieve echo metrics from the "
+ "AudioProcessingModule");
+ stats.erl.min = -100;
+ stats.erl.max = -100;
+ stats.erl.average = -100;
+ stats.erle.min = -100;
+ stats.erle.max = -100;
+ stats.erle.average = -100;
+ stats.rerl.min = -100;
+ stats.rerl.max = -100;
+ stats.rerl.average = -100;
+ stats.a_nlp.min = -100;
+ stats.a_nlp.max = -100;
+ stats.a_nlp.average = -100;
+ }
+ else
+ {
+
+ // Deliver output results to user
+ stats.erl.min = metrics.echo_return_loss.minimum;
+ stats.erl.max = metrics.echo_return_loss.maximum;
+ stats.erl.average = metrics.echo_return_loss.average;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " erl: min=%d, max=%d, avg=%d", stats.erl.min,
+ stats.erl.max, stats.erl.average);
+
+ stats.erle.min = metrics.echo_return_loss_enhancement.minimum;
+ stats.erle.max = metrics.echo_return_loss_enhancement.maximum;
+ stats.erle.average = metrics.echo_return_loss_enhancement.average;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " erle: min=%d, max=%d, avg=%d", stats.erle.min,
+ stats.erle.max, stats.erle.average);
+
+ stats.rerl.min = metrics.residual_echo_return_loss.minimum;
+ stats.rerl.max = metrics.residual_echo_return_loss.maximum;
+ stats.rerl.average = metrics.residual_echo_return_loss.average;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " rerl: min=%d, max=%d, avg=%d", stats.rerl.min,
+ stats.rerl.max, stats.rerl.average);
+
+ stats.a_nlp.min = metrics.a_nlp.minimum;
+ stats.a_nlp.max = metrics.a_nlp.maximum;
+ stats.a_nlp.average = metrics.a_nlp.average;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " a_nlp: min=%d, max=%d, avg=%d", stats.a_nlp.min,
+ stats.a_nlp.max, stats.a_nlp.average);
+ }
+ return 0;
+}
+
+int VoECallReportImpl::GetRoundTripTimeSummary(int channel, StatVal& delaysMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRoundTripTimeSummary()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRoundTripTimeSummary() failed to "
+ "locate channel");
+ return -1;
+ }
+
+ return channelPtr->GetRoundTripTimeSummary(delaysMs);
+}
+
+int VoECallReportImpl::GetDeadOrAliveSummary(int channel,
+ int& numOfDeadDetections,
+ int& numOfAliveDetections)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetDeadOrAliveSummary(channel=%d)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ return (GetDeadOrAliveSummaryInternal(channel, numOfDeadDetections,
+ numOfAliveDetections));
+}
+
+int VoECallReportImpl::GetDeadOrAliveSummaryInternal(int channel,
+ int& numOfDeadDetections,
+ int& numOfAliveDetections)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetDeadOrAliveSummary(channel=%d)", channel);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRoundTripTimeSummary() failed to "
+ "locate channel");
+ return -1;
+ }
+
+ return channelPtr->GetDeadOrAliveCounters(numOfDeadDetections,
+ numOfAliveDetections);
+}
+
+int VoECallReportImpl::WriteReportToFile(const char* fileNameUTF8)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "WriteReportToFile(fileNameUTF8=%s)", fileNameUTF8);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (NULL == fileNameUTF8)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "WriteReportToFile() invalid filename");
+ return -1;
+ }
+
+ if (_file.Open())
+ {
+ _file.CloseFile();
+ }
+
+ // Open text file in write mode
+ if (_file.OpenFile(fileNameUTF8, false, false, true) != 0)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "WriteReportToFile() unable to open the "
+ "file");
+ return -1;
+ }
+
+ // Summarize information and add it to the open file
+ //
+ _file.WriteText("WebRtc VoiceEngine Call Report\n");
+ _file.WriteText("==============================\n");
+ _file.WriteText("\nNetwork Packet Round Trip Time (RTT)\n");
+ _file.WriteText("------------------------------------\n\n");
+
+ WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
+ if (numOfChannels <= 0)
+ {
+ return 0;
+ }
+ WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
+ _channelManager.GetChannelIds(channelsArray, numOfChannels);
+ for (int ch = 0; ch < numOfChannels; ch++)
+ {
+ voe::ScopedChannel sc(_channelManager, channelsArray[ch]);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr)
+ {
+ StatVal delaysMs;
+ _file.WriteText("channel %d:\n", ch);
+ channelPtr->GetRoundTripTimeSummary(delaysMs);
+ _file.WriteText(" min:%5d [ms]\n", delaysMs.min);
+ _file.WriteText(" max:%5d [ms]\n", delaysMs.max);
+ _file.WriteText(" avg:%5d [ms]\n", delaysMs.average);
+ }
+ }
+
+ _file.WriteText("\nDead-or-Alive Connection Detections\n");
+ _file.WriteText("------------------------------------\n\n");
+
+ for (int ch = 0; ch < numOfChannels; ch++)
+ {
+ voe::ScopedChannel sc(_channelManager, channelsArray[ch]);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr)
+ {
+ int nDead(0);
+ int nAlive(0);
+ _file.WriteText("channel %d:\n", ch);
+ GetDeadOrAliveSummary(ch, nDead, nAlive);
+ _file.WriteText(" #dead :%6d\n", nDead);
+ _file.WriteText(" #alive:%6d\n", nAlive);
+ }
+ }
+
+ delete[] channelsArray;
+
+ EchoStatistics echo;
+ GetEchoMetricSummary(echo);
+
+ _file.WriteText("\nEcho Metrics\n");
+ _file.WriteText("------------\n\n");
+
+ _file.WriteText("erl:\n");
+ _file.WriteText(" min:%5d [dB]\n", echo.erl.min);
+ _file.WriteText(" max:%5d [dB]\n", echo.erl.max);
+ _file.WriteText(" avg:%5d [dB]\n", echo.erl.average);
+ _file.WriteText("\nerle:\n");
+ _file.WriteText(" min:%5d [dB]\n", echo.erle.min);
+ _file.WriteText(" max:%5d [dB]\n", echo.erle.max);
+ _file.WriteText(" avg:%5d [dB]\n", echo.erle.average);
+ _file.WriteText("rerl:\n");
+ _file.WriteText(" min:%5d [dB]\n", echo.rerl.min);
+ _file.WriteText(" max:%5d [dB]\n", echo.rerl.max);
+ _file.WriteText(" avg:%5d [dB]\n", echo.rerl.average);
+ _file.WriteText("a_nlp:\n");
+ _file.WriteText(" min:%5d [dB]\n", echo.a_nlp.min);
+ _file.WriteText(" max:%5d [dB]\n", echo.a_nlp.max);
+ _file.WriteText(" avg:%5d [dB]\n", echo.a_nlp.average);
+
+ _file.WriteText("\n<END>");
+
+ _file.Flush();
+ _file.CloseFile();
+
+ return 0;
+}
+
+#endif // WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_call_report_impl.h b/trunk/src/voice_engine/main/source/voe_call_report_impl.h
new file mode 100644
index 0000000..f5852b4
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_call_report_impl.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
+
+#include "voe_call_report.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+
+namespace webrtc
+{
+class FileWrapper;
+
+class VoECallReportImpl: public virtual voe::SharedData,
+ public VoECallReport,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int ResetCallReportStatistics(int channel);
+
+ virtual int GetEchoMetricSummary(EchoStatistics& stats);
+
+ virtual int GetRoundTripTimeSummary(int channel,
+ StatVal& delaysMs);
+
+ virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections,
+ int& numOfAliveDetections);
+
+ virtual int WriteReportToFile(const char* fileNameUTF8);
+
+protected:
+ VoECallReportImpl();
+ virtual ~VoECallReportImpl();
+
+private:
+ int GetDeadOrAliveSummaryInternal(int channel,
+ int& numOfDeadDetections,
+ int& numOfAliveDetections);
+
+ int GetEchoMetricSummaryInternal(EchoStatistics& stats);
+
+ int GetSpeechAndNoiseSummaryInternal(LevelStatistics& stats);
+
+ FileWrapper& _file;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_codec_impl.cc b/trunk/src/voice_engine/main/source/voe_codec_impl.cc
new file mode 100644
index 0000000..6b308cc
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_codec_impl.cc
@@ -0,0 +1,755 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_codec_impl.h"
+
+#include "audio_coding_module.h"
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc
+{
+
+VoECodec* VoECodec::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_CODEC_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoECodecImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+
+VoECodecImpl::VoECodecImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECodecImpl() - ctor");
+}
+
+VoECodecImpl::~VoECodecImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "~VoECodecImpl() - dtor");
+}
+
+int VoECodecImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECodecImpl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoECodecImpl reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoECodecImpl::NumOfCodecs()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "NumOfCodecs()");
+
+ // Number of supported codecs in the ACM
+ WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "NumOfCodecs() => %u", nSupportedCodecs);
+ return (nSupportedCodecs);
+}
+
+int VoECodecImpl::GetCodec(int index, CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetCodec(index=%d, codec=?)", index);
+ CodecInst acmCodec;
+ if (AudioCodingModule::Codec(index, (CodecInst&) acmCodec)
+ == -1)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_LISTNR, kTraceError,
+ "GetCodec() invalid index");
+ return -1;
+ }
+ ACMToExternalCodecRepresentation(codec, acmCodec);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetCodec() => plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
+ "channels=%d, rate=%d", codec.plname, codec.pacsize,
+ codec.plfreq, codec.pltype, codec.channels, codec.rate);
+ return 0;
+}
+
+int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec)
+{
+ CodecInst copyCodec;
+ ExternalToACMCodecRepresentation(copyCodec, codec);
+
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetSendCodec(channel=%d, codec)", channel);
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
+ "channels=%d, rate=%d", codec.plname, codec.pacsize,
+ codec.plfreq, codec.pltype, codec.channels, codec.rate);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ // External sanity checks performed outside the ACM
+ if ((STR_CASE_CMP(copyCodec.plname, "L16") == 0) &&
+ (copyCodec.pacsize >= 960))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendCodec() invalid L16 packet "
+ "size");
+ return -1;
+ }
+ if (!STR_CASE_CMP(copyCodec.plname, "CN")
+ || !STR_CASE_CMP(copyCodec.plname, "TELEPHONE-EVENT")
+ || !STR_CASE_CMP(copyCodec.plname, "RED"))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendCodec() invalid codec name");
+ return -1;
+ }
+ if ((copyCodec.channels != 1) && (copyCodec.channels != 2))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendCodec() invalid number of "
+ "channels");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendCodec() failed to locate "
+ "channel");
+ return -1;
+ }
+ if (!AudioCodingModule::IsCodecValid(
+ (CodecInst&) copyCodec))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendCodec() invalid codec");
+ return -1;
+ }
+ if (channelPtr->SetSendCodec(copyCodec) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_SET_SEND_CODEC,
+ kTraceError,
+ "SetSendCodec() failed to set send "
+ "codec");
+ return -1;
+ }
+
+ // Need to check if we should change APM settings for mono/stereo.
+ // We'll check all channels (sending or not), so we don't have to
+ // check this again when starting/stopping sending.
+
+ voe::ScopedChannel sc2(_channelManager);
+ void* iterator(NULL);
+ channelPtr = sc2.GetFirstChannel(iterator);
+ int maxNumChannels = 1;
+ while (channelPtr != NULL)
+ {
+ CodecInst tmpCdc;
+ channelPtr->GetSendCodec(tmpCdc);
+ if (tmpCdc.channels > maxNumChannels)
+ maxNumChannels = tmpCdc.channels;
+
+ channelPtr = sc2.GetNextChannel(iterator);
+ }
+
+ // Reuse the currently set number of capture channels. We need to wait
+ // until receiving a frame to determine the true number.
+ //
+ // TODO(andrew): AudioProcessing will return an error if there are more
+ // output than input channels (it doesn't want to produce fake channels).
+ // This will happen with a stereo codec and a device which doesn't support
+ // stereo. AudioCoding should probably do the faking; look into how to
+ // handle this case properly.
+ //
+ // Check if the number of channels has changed to avoid an unnecessary
+ // reset.
+ // TODO(andrew): look at handling this logic in AudioProcessing.
+ if (_audioProcessingModulePtr->num_output_channels() != maxNumChannels)
+ {
+ if (_audioProcessingModulePtr->set_num_channels(
+ _audioProcessingModulePtr->num_input_channels(),
+ maxNumChannels) != 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
+ "Init() failed to set APM channels for the send audio stream");
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+int VoECodecImpl::GetSendCodec(int channel, CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSendCodec(channel=%d, codec=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendCodec() failed to locate "
+ "channel");
+ return -1;
+ }
+ CodecInst acmCodec;
+ if (channelPtr->GetSendCodec(acmCodec) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_GET_SEND_CODEC, kTraceError,
+ "GetSendCodec() failed to get send "
+ "codec");
+ return -1;
+ }
+ ACMToExternalCodecRepresentation(codec, acmCodec);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSendCodec() => plname=%s, pacsize=%d, plfreq=%d, "
+ "channels=%d, rate=%d", codec.plname, codec.pacsize,
+ codec.plfreq, codec.channels, codec.rate);
+ return 0;
+}
+
+int VoECodecImpl::GetRecCodec(int channel, CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRecCodec(channel=%d, codec=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRecCodec() failed to locate "
+ "channel");
+ return -1;
+ }
+ CodecInst acmCodec;
+ if (channelPtr->GetRecCodec(acmCodec) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CANNOT_GET_REC_CODEC, kTraceError,
+ "GetRecCodec() failed to get received "
+ "codec");
+ return -1;
+ }
+ ACMToExternalCodecRepresentation(codec, acmCodec);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRecCodec() => plname=%s, pacsize=%d, plfreq=%d, "
+ "channels=%d, rate=%d", codec.plname, codec.pacsize,
+ codec.plfreq, codec.channels, codec.rate);
+ return 0;
+}
+
+int VoECodecImpl::SetAMREncFormat(int channel, AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetAMREncFormat(channel=%d, mode=%d)", channel, mode);
+#ifdef WEBRTC_CODEC_GSMAMR
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetAMREncFormat() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetAMREncFormat(mode);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAMREncFormat() AMR codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetAMRDecFormat(int channel, AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetAMRDecFormat(channel=%i, mode=%i)", channel, mode);
+#ifdef WEBRTC_CODEC_GSMAMR
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetAMRDecFormat() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetAMRDecFormat(mode);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAMRDecFormat() AMR codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetAMRWbEncFormat(int channel, AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetAMRWbEncFormat(channel=%d, mode=%d)", channel, mode);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetAMRWbEncFormat() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetAMRWbEncFormat(mode);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAMRWbEncFormat() AMR-wb codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetAMRWbDecFormat(int channel, AmrMode mode)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetAMRWbDecFormat(channel=%i, mode=%i)", channel, mode);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetAMRWbDecFormat() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetAMRWbDecFormat(mode);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetAMRWbDecFormat() AMR-wb codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetRecPayloadType(int channel, const CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetRecPayloadType(channel=%d, codec)", channel);
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "codec: plname=%s, plfreq=%d, pltype=%d, channels=%u, "
+ "pacsize=%d, rate=%d", codec.plname, codec.plfreq, codec.pltype,
+ codec.channels, codec.pacsize, codec.rate);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRecPayloadType() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetRecPayloadType(codec);
+}
+
+int VoECodecImpl::GetRecPayloadType(int channel, CodecInst& codec)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRecPayloadType(channel=%d, codec)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRecPayloadType() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->GetRecPayloadType(codec);
+}
+
+int VoECodecImpl::SetSendCNPayloadType(int channel, int type,
+ PayloadFrequencies frequency)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetSendCNPayloadType(channel=%d, type=%d, frequency=%d)",
+ channel, type, frequency);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (type < 96 || type > 127)
+ {
+ // Only allow dynamic range: 96 to 127
+ _engineStatistics.SetLastError(VE_INVALID_PLTYPE, kTraceError,
+ "SetSendCNPayloadType() invalid payload "
+ "type");
+ return -1;
+ }
+ if ((frequency != kFreq16000Hz) && (frequency != kFreq32000Hz))
+ {
+ // It is not possible to modify the payload type for CN/8000.
+ // We only allow modification of the CN payload type for CN/16000
+ // and CN/32000.
+ _engineStatistics.SetLastError(VE_INVALID_PLFREQ, kTraceError,
+ "SetSendCNPayloadType() invalid payload"
+ " frequency");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSendCNPayloadType() failed to "
+ "locate channel");
+ return -1;
+ }
+ if (channelPtr->Sending())
+ {
+ _engineStatistics.SetLastError(VE_SENDING, kTraceError,
+ "SetSendCNPayloadType unable so set "
+ "payload type while sending");
+ return -1;
+ }
+ return channelPtr->SetSendCNPayloadType(type, frequency);
+}
+
+int VoECodecImpl::SetISACInitTargetRate(int channel, int rateBps,
+ bool useFixedFrameSize)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetISACInitTargetRate(channel=%d, rateBps=%d, "
+ "useFixedFrameSize=%d)", channel, rateBps, useFixedFrameSize);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_CODEC_ISAC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetISACInitTargetRate() failed to "
+ "locate channel");
+ return -1;
+ }
+ return channelPtr->SetISACInitTargetRate(rateBps, useFixedFrameSize);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetISACInitTargetRate() iSAC codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetISACMaxRate(int channel, int rateBps)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetISACMaxRate(channel=%d, rateBps=%d)", channel, rateBps);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_CODEC_ISAC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetISACMaxRate() failed to locate "
+ "channel");
+ return -1;
+ }
+ return channelPtr->SetISACMaxRate(rateBps);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetISACMaxRate() iSAC codec is not "
+ "supported");
+ return -1;
+#endif
+}
+
+int VoECodecImpl::SetISACMaxPayloadSize(int channel, int sizeBytes)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetISACMaxPayloadSize(channel=%d, sizeBytes=%d)", channel,
+ sizeBytes);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_CODEC_ISAC
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetISACMaxPayloadSize() failed to "
+ "locate channel");
+ return -1;
+ }
+ return channelPtr->SetISACMaxPayloadSize(sizeBytes);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetISACMaxPayloadSize() iSAC codec is not "
+ "supported");
+ return -1;
+#endif
+ return 0;
+}
+
+int VoECodecImpl::SetVADStatus(int channel, bool enable, VadModes mode,
+ bool disableDTX)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetVADStatus(channel=%i, enable=%i, mode=%i, disableDTX=%i)",
+ channel, enable, mode, disableDTX);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetVADStatus failed to locate channel");
+ return -1;
+ }
+
+ ACMVADMode vadMode(VADNormal);
+ switch (mode)
+ {
+ case kVadConventional:
+ vadMode = VADNormal;
+ break;
+ case kVadAggressiveLow:
+ vadMode = VADLowBitrate;
+ break;
+ case kVadAggressiveMid:
+ vadMode = VADAggr;
+ break;
+ case kVadAggressiveHigh:
+ vadMode = VADVeryAggr;
+ break;
+ }
+ return channelPtr->SetVADStatus(enable, vadMode, disableDTX);
+}
+
+int VoECodecImpl::GetVADStatus(int channel, bool& enabled, VadModes& mode,
+ bool& disabledDTX)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetVADStatus(channel=%i)", channel);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetVADStatus failed to locate channel");
+ return -1;
+ }
+
+ ACMVADMode vadMode;
+ int ret = channelPtr->GetVADStatus(enabled, vadMode, disabledDTX);
+
+ if (ret != 0)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
+ "GetVADStatus failed to get VAD mode");
+ return -1;
+ }
+ switch (vadMode)
+ {
+ case VADNormal:
+ mode = kVadConventional;
+ break;
+ case VADLowBitrate:
+ mode = kVadAggressiveLow;
+ break;
+ case VADAggr:
+ mode = kVadAggressiveMid;
+ break;
+ case VADVeryAggr:
+ mode = kVadAggressiveHigh;
+ break;
+ }
+
+ return 0;
+}
+
+void VoECodecImpl::ACMToExternalCodecRepresentation(CodecInst& toInst,
+ const CodecInst& fromInst)
+{
+ toInst = fromInst;
+ if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
+ {
+ if (fromInst.plfreq == 12000)
+ {
+ if (fromInst.pacsize == 320)
+ {
+ toInst.pacsize = 240;
+ }
+ else if (fromInst.pacsize == 640)
+ {
+ toInst.pacsize = 480;
+ }
+ else if (fromInst.pacsize == 960)
+ {
+ toInst.pacsize = 720;
+ }
+ }
+ else if (fromInst.plfreq == 24000)
+ {
+ if (fromInst.pacsize == 640)
+ {
+ toInst.pacsize = 480;
+ }
+ else if (fromInst.pacsize == 1280)
+ {
+ toInst.pacsize = 960;
+ }
+ else if (fromInst.pacsize == 1920)
+ {
+ toInst.pacsize = 1440;
+ }
+ }
+ }
+}
+
+void VoECodecImpl::ExternalToACMCodecRepresentation(CodecInst& toInst,
+ const CodecInst& fromInst)
+{
+ toInst = fromInst;
+ if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
+ {
+ if (fromInst.plfreq == 12000)
+ {
+ if (fromInst.pacsize == 240)
+ {
+ toInst.pacsize = 320;
+ }
+ else if (fromInst.pacsize == 480)
+ {
+ toInst.pacsize = 640;
+ }
+ else if (fromInst.pacsize == 720)
+ {
+ toInst.pacsize = 960;
+ }
+ }
+ else if (fromInst.plfreq == 24000)
+ {
+ if (fromInst.pacsize == 480)
+ {
+ toInst.pacsize = 640;
+ }
+ else if (fromInst.pacsize == 960)
+ {
+ toInst.pacsize = 1280;
+ }
+ else if (fromInst.pacsize == 1440)
+ {
+ toInst.pacsize = 1920;
+ }
+ }
+ }
+}
+
+#endif // WEBRTC_VOICE_ENGINE_CODEC_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_codec_impl.h b/trunk/src/voice_engine/main/source/voe_codec_impl.h
new file mode 100644
index 0000000..fde1d41
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_codec_impl.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
+
+#include "voe_codec.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc
+{
+
+class VoECodecImpl: public virtual voe::SharedData,
+ public VoECodec,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int NumOfCodecs();
+
+ virtual int GetCodec(int index, CodecInst& codec);
+
+ virtual int SetSendCodec(int channel, const CodecInst& codec);
+
+ virtual int GetSendCodec(int channel, CodecInst& codec);
+
+ virtual int GetRecCodec(int channel, CodecInst& codec);
+
+ virtual int SetAMREncFormat(int channel,
+ AmrMode mode = kRfc3267BwEfficient);
+
+ virtual int SetAMRDecFormat(int channel,
+ AmrMode mode = kRfc3267BwEfficient);
+
+ virtual int SetAMRWbEncFormat(int channel,
+ AmrMode mode = kRfc3267BwEfficient);
+
+ virtual int SetAMRWbDecFormat(int channel,
+ AmrMode mode = kRfc3267BwEfficient);
+
+ virtual int SetSendCNPayloadType(
+ int channel, int type,
+ PayloadFrequencies frequency = kFreq16000Hz);
+
+ virtual int SetRecPayloadType(int channel,
+ const CodecInst& codec);
+
+ virtual int GetRecPayloadType(int channel, CodecInst& codec);
+
+ virtual int SetISACInitTargetRate(int channel,
+ int rateBps,
+ bool useFixedFrameSize = false);
+
+ virtual int SetISACMaxRate(int channel, int rateBps);
+
+ virtual int SetISACMaxPayloadSize(int channel, int sizeBytes);
+
+ virtual int SetVADStatus(int channel,
+ bool enable,
+ VadModes mode = kVadConventional,
+ bool disableDTX = false);
+
+ virtual int GetVADStatus(int channel,
+ bool& enabled,
+ VadModes& mode,
+ bool& disabledDTX);
+
+protected:
+ VoECodecImpl();
+ virtual ~VoECodecImpl();
+
+private:
+ void ACMToExternalCodecRepresentation(CodecInst& toInst,
+ const CodecInst& fromInst);
+
+ void ExternalToACMCodecRepresentation(CodecInst& toInst,
+ const CodecInst& fromInst);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc b/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc
new file mode 100644
index 0000000..67f4c46
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc
@@ -0,0 +1,473 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_dtmf_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "output_mixer.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEDtmf* VoEDtmf::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_DTMF_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEDtmfImpl* d = s;
+ ( *d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+
+VoEDtmfImpl::VoEDtmfImpl() :
+ _dtmfFeedback(true),
+ _dtmfDirectFeedback(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1 ),
+ "VoEDtmfImpl::VoEDtmfImpl() - ctor");
+}
+
+VoEDtmfImpl::~VoEDtmfImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEDtmfImpl::~VoEDtmfImpl() - dtor");
+}
+
+int VoEDtmfImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEDtmf::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _engineStatistics.SetLastError(
+ VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEDtmf reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEDtmfImpl::SendTelephoneEvent(int channel,
+ int eventCode,
+ bool outOfBand,
+ int lengthMs,
+ int attenuationDb)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SendTelephoneEvent(channel=%d, eventCode=%d, outOfBand=%d,"
+ "length=%d, attenuationDb=%d)",
+ channel, eventCode, (int)outOfBand, lengthMs, attenuationDb);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SendTelephoneEvent() failed to locate channel");
+ return -1;
+ }
+ if (!channelPtr->Sending())
+ {
+ _engineStatistics.SetLastError(
+ VE_NOT_SENDING, kTraceError,
+ "SendTelephoneEvent() sending is not active");
+ return -1;
+ }
+
+ // Sanity check
+ const int maxEventCode = outOfBand ?
+ static_cast<int>(kMaxTelephoneEventCode) :
+ static_cast<int>(kMaxDtmfEventCode);
+ const bool testFailed = ((eventCode < 0) ||
+ (eventCode > maxEventCode) ||
+ (lengthMs < kMinTelephoneEventDuration) ||
+ (lengthMs > kMaxTelephoneEventDuration) ||
+ (attenuationDb < kMinTelephoneEventAttenuation) ||
+ (attenuationDb > kMaxTelephoneEventAttenuation));
+ if (testFailed)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SendTelephoneEvent() invalid parameter(s)");
+ return -1;
+ }
+
+ const bool isDtmf =
+ (eventCode >= 0) && (eventCode <= kMaxDtmfEventCode);
+ const bool playDtmfToneDirect =
+ isDtmf && (_dtmfFeedback && _dtmfDirectFeedback);
+
+ if (playDtmfToneDirect)
+ {
+ // Mute the microphone signal while playing back the tone directly.
+ // This is to reduce the risk of introducing echo from the added output.
+ _transmitMixerPtr->UpdateMuteMicrophoneTime(lengthMs);
+
+ // Play out local feedback tone directly (same approach for both inband
+ // and outband).
+ // Reduce the length of the the tone with 80ms to reduce risk of echo.
+ // For non-direct feedback, outband and inband cases are handled
+ // differently.
+ _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs-80, attenuationDb);
+ }
+
+ if (outOfBand)
+ {
+ // The RTP/RTCP module will always deliver OnPlayTelephoneEvent when
+ // an event is transmitted. It is up to the VoE to utilize it or not.
+ // This flag ensures that feedback/playout is enabled; however, the
+ // channel object must still parse out the Dtmf events (0-15) from
+ // all possible events (0-255).
+ const bool playDTFMEvent = (_dtmfFeedback && !_dtmfDirectFeedback);
+
+ return channelPtr->SendTelephoneEventOutband(eventCode,
+ lengthMs,
+ attenuationDb,
+ playDTFMEvent);
+ }
+ else
+ {
+ // For Dtmf tones, we want to ensure that inband tones are played out
+ // in sync with the transmitted audio. This flag is utilized by the
+ // channel object to determine if the queued Dtmf e vent shall also
+ // be fed to the output mixer in the same step as input audio is
+ // replaced by inband Dtmf tones.
+ const bool playDTFMEvent =
+ (isDtmf && _dtmfFeedback && !_dtmfDirectFeedback);
+
+ return channelPtr->SendTelephoneEventInband(eventCode,
+ lengthMs,
+ attenuationDb,
+ playDTFMEvent);
+ }
+}
+
+int VoEDtmfImpl::SetSendTelephoneEventPayloadType(int channel,
+ unsigned char type)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetSendTelephoneEventPayloadType(channel=%d, type=%u)",
+ channel, type);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSendTelephoneEventPayloadType() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetSendTelephoneEventPayloadType(type);
+}
+
+int VoEDtmfImpl::GetSendTelephoneEventPayloadType(int channel,
+ unsigned char& type)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSendTelephoneEventPayloadType(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendTelephoneEventPayloadType() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetSendTelephoneEventPayloadType(type);
+}
+
+int VoEDtmfImpl::PlayDtmfTone(int eventCode,
+ int lengthMs,
+ int attenuationDb)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "PlayDtmfTone(eventCode=%d, lengthMs=%d, attenuationDb=%d)",
+ eventCode, lengthMs, attenuationDb);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_audioDevicePtr->Playing())
+ {
+ _engineStatistics.SetLastError(
+ VE_NOT_PLAYING, kTraceError,
+ "PlayDtmfTone() no channel is playing out");
+ return -1;
+ }
+ if ((eventCode < kMinDtmfEventCode) ||
+ (eventCode > kMaxDtmfEventCode) ||
+ (lengthMs < kMinTelephoneEventDuration) ||
+ (lengthMs > kMaxTelephoneEventDuration) ||
+ (attenuationDb <kMinTelephoneEventAttenuation) ||
+ (attenuationDb > kMaxTelephoneEventAttenuation))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "PlayDtmfTone() invalid tone parameter(s)");
+ return -1;
+ }
+ return _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb);
+}
+
+int VoEDtmfImpl::StartPlayingDtmfTone(int eventCode,
+ int attenuationDb)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingDtmfTone(eventCode=%d, attenuationDb=%d)",
+ eventCode, attenuationDb);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_audioDevicePtr->Playing())
+ {
+ _engineStatistics.SetLastError(
+ VE_NOT_PLAYING, kTraceError,
+ "StartPlayingDtmfTone() no channel is playing out");
+ return -1;
+ }
+ if ((eventCode < kMinDtmfEventCode) ||
+ (eventCode > kMaxDtmfEventCode) ||
+ (attenuationDb < kMinTelephoneEventAttenuation) ||
+ (attenuationDb > kMaxTelephoneEventAttenuation))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingDtmfTone() invalid tone parameter(s)");
+ return -1;
+ }
+ return _outputMixerPtr->StartPlayingDtmfTone(eventCode, attenuationDb);
+}
+
+int VoEDtmfImpl::StopPlayingDtmfTone()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopPlayingDtmfTone()");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ return _outputMixerPtr->StopPlayingDtmfTone();
+}
+
+int VoEDtmfImpl::RegisterTelephoneEventDetection(
+ int channel,
+ TelephoneEventDetectionMethods detectionMethod,
+ VoETelephoneEventObserver& observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d,"
+ "observer=0x%x)", channel, detectionMethod, &observer);
+#ifdef WEBRTC_DTMF_DETECTION
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterTelephoneEventDetection() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterTelephoneEventDetection(detectionMethod,
+ observer);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetTelephoneEventDetectionStatus() Dtmf detection is not supported");
+ return -1;
+#endif
+}
+
+int VoEDtmfImpl::DeRegisterTelephoneEventDetection(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterTelephoneEventDetection(channel=%d)", channel);
+#ifdef WEBRTC_DTMF_DETECTION
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterTelephoneEventDe tection() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterTelephoneEventDetection();
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "DeRegisterTelephoneEventDetection() Dtmf detection is not supported");
+ return -1;
+#endif
+}
+
+
+int VoEDtmfImpl::GetTelephoneEventDetectionStatus(
+ int channel,
+ bool& enabled,
+ TelephoneEventDetectionMethods& detectionMethod)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetTelephoneEventDetectionStatus(channel=%d)", channel);
+#ifdef WEBRTC_DTMF_DETECTION
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetTelephoneEventDetectionStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetTelephoneEventDetectionStatus(enabled, detectionMethod);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetTelephoneEventDetectionStatus() Dtmf detection is not supported");
+ return -1;
+#endif
+}
+
+int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
+ (int)enable, (int)directFeedback);
+
+ CriticalSectionScoped sc(*_apiCritPtr);
+
+ _dtmfFeedback = enable;
+ _dtmfDirectFeedback = directFeedback;
+
+ return 0;
+}
+
+int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetDtmfFeedbackStatus()");
+
+ CriticalSectionScoped sc(*_apiCritPtr);
+
+ enabled = _dtmfFeedback;
+ directFeedback = _dtmfDirectFeedback;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetDtmfFeedbackStatus() => enabled=%d, directFeedback=%d",
+ enabled, directFeedback);
+ return 0;
+}
+
+int VoEDtmfImpl::SetDtmfPlayoutStatus(int channel, bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetDtmfPlayoutStatus(channel=%d, enable=%d)",
+ channel, enable);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetDtmfPlayoutStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetDtmfPlayoutStatus(enable);
+}
+
+int VoEDtmfImpl::GetDtmfPlayoutStatus(int channel, bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetDtmfPlayoutStatus(channel=%d, enabled=?)", channel);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetDtmfPlayoutStatus() failed to locate channel");
+ return -1;
+ }
+ enabled = channelPtr->DtmfPlayoutStatus();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetDtmfPlayoutStatus() => enabled=%d", enabled);
+ return 0;
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_dtmf_impl.h b/trunk/src/voice_engine/main/source/voe_dtmf_impl.h
new file mode 100644
index 0000000..5b53969
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_dtmf_impl.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
+
+#include "voe_dtmf.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc
+{
+
+class VoEDtmfImpl : public virtual voe::SharedData,
+ public VoEDtmf,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int SendTelephoneEvent(
+ int channel,
+ int eventCode,
+ bool outOfBand = true,
+ int lengthMs = 160,
+ int attenuationDb = 10);
+
+ virtual int SetSendTelephoneEventPayloadType(int channel,
+ unsigned char type);
+
+ virtual int GetSendTelephoneEventPayloadType(int channel,
+ unsigned char& type);
+
+ virtual int SetDtmfFeedbackStatus(bool enable,
+ bool directFeedback = false);
+
+ virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback);
+
+ virtual int PlayDtmfTone(int eventCode,
+ int lengthMs = 200,
+ int attenuationDb = 10);
+
+ virtual int StartPlayingDtmfTone(int eventCode,
+ int attenuationDb = 10);
+
+ virtual int StopPlayingDtmfTone();
+
+ virtual int RegisterTelephoneEventDetection(
+ int channel,
+ TelephoneEventDetectionMethods detectionMethod,
+ VoETelephoneEventObserver& observer);
+
+ virtual int DeRegisterTelephoneEventDetection(int channel);
+
+ virtual int GetTelephoneEventDetectionStatus(
+ int channel,
+ bool& enabled,
+ TelephoneEventDetectionMethods& detectionMethod);
+
+ virtual int SetDtmfPlayoutStatus(int channel, bool enable);
+
+ virtual int GetDtmfPlayoutStatus(int channel, bool& enabled);
+
+protected:
+ VoEDtmfImpl();
+ virtual ~VoEDtmfImpl();
+
+private:
+ bool _dtmfFeedback;
+ bool _dtmfDirectFeedback;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_encryption_impl.cc b/trunk/src/voice_engine/main/source/voe_encryption_impl.cc
new file mode 100644
index 0000000..5ba944b
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_encryption_impl.cc
@@ -0,0 +1,275 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_encryption_impl.h"
+
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEEncryption* VoEEncryption::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEEncryptionImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+
+VoEEncryptionImpl::VoEEncryptionImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEEncryptionImpl::VoEEncryptionImpl() - ctor");
+}
+
+VoEEncryptionImpl::~VoEEncryptionImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEEncryptionImpl::~VoEEncryptionImpl() - dtor");
+}
+
+int VoEEncryptionImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEEncryption::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ // reset reference counter to zero => OK to delete VE
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEEncryption reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEEncryptionImpl::EnableSRTPSend(
+ int channel,
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "EnableSRTPSend(channel=%i, cipherType=%i, cipherKeyLength=%i,"
+ " authType=%i, authKeyLength=%i, authTagLength=%i, level=%i, "
+ "key=?, useForRTCP=%d)",
+ channel, cipherType, cipherKeyLength, authType,
+ authKeyLength, authTagLength, level, useForRTCP);
+#ifdef WEBRTC_SRTP
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "EnableSRTPSend() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->EnableSRTPSend(cipherType,
+ cipherKeyLength,
+ authType,
+ authKeyLength,
+ authTagLength,
+ level,
+ key,
+ useForRTCP);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "EnableSRTPSend() SRTP is not supported");
+ return -1;
+#endif
+}
+
+int VoEEncryptionImpl::DisableSRTPSend(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DisableSRTPSend(channel=%i)",channel);
+#ifdef WEBRTC_SRTP
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DisableSRTPSend() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DisableSRTPSend();
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "DisableSRTPSend() SRTP is not supported");
+ return -1;
+#endif
+}
+
+int VoEEncryptionImpl::EnableSRTPReceive(
+ int channel,
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "EnableSRTPReceive(channel=%i, cipherType=%i, "
+ "cipherKeyLength=%i, authType=%i, authKeyLength=%i, "
+ "authTagLength=%i, level=%i, key=?, useForRTCP=%d)",
+ channel, cipherType, cipherKeyLength, authType,
+ authKeyLength, authTagLength, level, useForRTCP);
+#ifdef WEBRTC_SRTP
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "EnableSRTPReceive() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->EnableSRTPReceive(cipherType,
+ cipherKeyLength,
+ authType,
+ authKeyLength,
+ authTagLength,
+ level,
+ key,
+ useForRTCP);
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "EnableSRTPReceive() SRTP is not supported");
+ return -1;
+#endif
+}
+
+int VoEEncryptionImpl::DisableSRTPReceive(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DisableSRTPReceive(channel=%i)", channel);
+#ifdef WEBRTC_SRTP
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DisableSRTPReceive() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DisableSRTPReceive();
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "DisableSRTPReceive() SRTP is not supported");
+ return -1;
+#endif
+}
+
+int VoEEncryptionImpl::RegisterExternalEncryption(int channel,
+ Encryption& encryption)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterExternalEncryption(channel=%d, encryption=0x%x)",
+ channel, &encryption);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterExternalEncryption() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterExternalEncryption(encryption);
+}
+
+int VoEEncryptionImpl::DeRegisterExternalEncryption(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterExternalEncryption(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterExternalEncryption() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterExternalEncryption();
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+
+// EOF
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_encryption_impl.h b/trunk/src/voice_engine/main/source/voe_encryption_impl.h
new file mode 100644
index 0000000..050dd88
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_encryption_impl.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
+
+#include "voe_encryption.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoEEncryptionImpl : public virtual voe::SharedData,
+ public VoEEncryption,
+ public voe::RefCount
+{
+public:
+
+ virtual int Release();
+
+ // SRTP
+ virtual int EnableSRTPSend(
+ int channel,
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP = false);
+
+ virtual int DisableSRTPSend(int channel);
+
+ virtual int EnableSRTPReceive(
+ int channel,
+ CipherTypes cipherType,
+ int cipherKeyLength,
+ AuthenticationTypes authType,
+ int authKeyLength,
+ int authTagLength,
+ SecurityLevels level,
+ const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
+ bool useForRTCP = false);
+
+ virtual int DisableSRTPReceive(int channel);
+
+ // External encryption
+ virtual int RegisterExternalEncryption(
+ int channel,
+ Encryption& encryption);
+
+ virtual int DeRegisterExternalEncryption(int channel);
+
+protected:
+ VoEEncryptionImpl();
+ virtual ~VoEEncryptionImpl();
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_external_media_impl.cc b/trunk/src/voice_engine/main/source/voe_external_media_impl.cc
new file mode 100644
index 0000000..3fc3239
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_external_media_impl.cc
@@ -0,0 +1,397 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_external_media_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "output_mixer.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "voice_engine_impl.h"
+#include "voe_errors.h"
+
+namespace webrtc {
+
+VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEExternalMediaImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+
+VoEExternalMediaImpl::VoEExternalMediaImpl()
+ : playout_delay_ms_(0)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEExternalMediaImpl() - ctor");
+}
+
+VoEExternalMediaImpl::~VoEExternalMediaImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "~VoEExternalMediaImpl() - dtor");
+}
+
+int VoEExternalMediaImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEExternalMedia::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEExternalMedia reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEExternalMediaImpl::RegisterExternalMediaProcessing(
+ int channel,
+ ProcessingTypes type,
+ VoEMediaProcess& processObject)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterExternalMediaProcessing(channel=%d, type=%d, "
+ "processObject=0x%x)", channel, type, &processObject);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ switch (type)
+ {
+ case kPlaybackPerChannel:
+ case kRecordingPerChannel:
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterExternalMediaProcessing() "
+ "failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterExternalMediaProcessing(type,
+ processObject);
+ }
+ case kPlaybackAllChannelsMixed:
+ {
+ return _outputMixerPtr->RegisterExternalMediaProcessing(
+ processObject);
+ }
+ case kRecordingAllChannelsMixed:
+ {
+ return _transmitMixerPtr->RegisterExternalMediaProcessing(
+ processObject);
+ }
+ }
+ return -1;
+}
+
+int VoEExternalMediaImpl::DeRegisterExternalMediaProcessing(
+ int channel,
+ ProcessingTypes type)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterExternalMediaProcessing(channel=%d)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ switch (type)
+ {
+ case kPlaybackPerChannel:
+ case kRecordingPerChannel:
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterExternalMediaProcessing() "
+ "failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterExternalMediaProcessing(type);
+ }
+ case kPlaybackAllChannelsMixed:
+ {
+ return _outputMixerPtr->DeRegisterExternalMediaProcessing();
+ }
+ case kRecordingAllChannelsMixed:
+ {
+ return _transmitMixerPtr->DeRegisterExternalMediaProcessing();
+ }
+ }
+ return -1;
+}
+
+int VoEExternalMediaImpl::SetExternalRecordingStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetExternalRecordingStatus(enable=%d)", enable);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ if (_audioDevicePtr->Recording())
+ {
+ _engineStatistics.SetLastError(
+ VE_ALREADY_SENDING,
+ kTraceError,
+ "SetExternalRecordingStatus() cannot set state while sending");
+ return -1;
+ }
+ _externalRecording = enable;
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED,
+ kTraceError,
+ "SetExternalRecordingStatus() external recording is not supported");
+ return -1;
+#endif
+}
+
+int VoEExternalMediaImpl::ExternalRecordingInsertData(
+ const WebRtc_Word16 speechData10ms[],
+ int lengthSamples,
+ int samplingFreqHz,
+ int current_delay_ms)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "ExternalRecordingInsertData(speechData10ms=0x%x,"
+ " lengthSamples=%u, samplingFreqHz=%d, current_delay_ms=%d)",
+ &speechData10ms[0], lengthSamples, samplingFreqHz,
+ current_delay_ms);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_externalRecording)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_OPERATION,
+ kTraceError,
+ "ExternalRecordingInsertData() external recording is not enabled");
+ return -1;
+ }
+ if (NumOfSendingChannels() == 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_ALREADY_SENDING,
+ kTraceError,
+ "SetExternalRecordingStatus() no channel is sending");
+ return -1;
+ }
+ if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
+ (48000 != samplingFreqHz) && (44000 != samplingFreqHz))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT,
+ kTraceError,
+ "SetExternalRecordingStatus() invalid sample rate");
+ return -1;
+ }
+ if ((0 == lengthSamples) ||
+ ((lengthSamples % (samplingFreqHz / 100)) != 0))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT,
+ kTraceError,
+ "SetExternalRecordingStatus() invalid buffer size");
+ return -1;
+ }
+ if (current_delay_ms < 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT,
+ kTraceError,
+ "SetExternalRecordingStatus() invalid delay)");
+ return -1;
+ }
+
+ WebRtc_UWord16 blockSize = samplingFreqHz / 100;
+ WebRtc_UWord32 nBlocks = lengthSamples / blockSize;
+ WebRtc_Word16 totalDelayMS = 0;
+ WebRtc_UWord16 playoutDelayMS = 0;
+
+ for (WebRtc_UWord32 i = 0; i < nBlocks; i++)
+ {
+ if (!_externalPlayout)
+ {
+ // Use real playout delay if external playout is not enabled.
+ if (_audioDevicePtr->PlayoutDelay(&playoutDelayMS) != 0) {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+ "PlayoutDelay() unable to get the playout delay");
+ }
+ totalDelayMS = current_delay_ms + playoutDelayMS;
+ }
+ else
+ {
+ // Use stored delay value given the last call
+ // to ExternalPlayoutGetData.
+ totalDelayMS = current_delay_ms + playout_delay_ms_;
+ // Compensate for block sizes larger than 10ms
+ totalDelayMS -= (WebRtc_Word16)(i*10);
+ if (totalDelayMS < 0)
+ totalDelayMS = 0;
+ }
+ _transmitMixerPtr->PrepareDemux(
+ (const WebRtc_Word8*)(&speechData10ms[i*blockSize]),
+ blockSize,
+ 1,
+ samplingFreqHz,
+ totalDelayMS,
+ 0,
+ 0);
+
+ _transmitMixerPtr->DemuxAndMix();
+ _transmitMixerPtr->EncodeAndSend();
+ }
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED,
+ kTraceError,
+ "ExternalRecordingInsertData() external recording is not supported");
+ return -1;
+#endif
+}
+
+int VoEExternalMediaImpl::SetExternalPlayoutStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetExternalPlayoutStatus(enable=%d)", enable);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ if (_audioDevicePtr->Playing())
+ {
+ _engineStatistics.SetLastError(
+ VE_ALREADY_SENDING,
+ kTraceError,
+ "SetExternalPlayoutStatus() cannot set state while playing");
+ return -1;
+ }
+ _externalPlayout = enable;
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED,
+ kTraceError,
+ "SetExternalPlayoutStatus() external playout is not supported");
+ return -1;
+#endif
+}
+
+int VoEExternalMediaImpl::ExternalPlayoutGetData(
+ WebRtc_Word16 speechData10ms[],
+ int samplingFreqHz,
+ int current_delay_ms,
+ int& lengthSamples)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "ExternalPlayoutGetData(speechData10ms=0x%x, samplingFreqHz=%d"
+ ", current_delay_ms=%d)", &speechData10ms[0], samplingFreqHz,
+ current_delay_ms);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (!_externalPlayout)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_OPERATION,
+ kTraceError,
+ "ExternalPlayoutGetData() external playout is not enabled");
+ return -1;
+ }
+ if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
+ (48000 != samplingFreqHz) && (44000 != samplingFreqHz))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT,
+ kTraceError,
+ "ExternalPlayoutGetData() invalid sample rate");
+ return -1;
+ }
+ if (current_delay_ms < 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT,
+ kTraceError,
+ "ExternalPlayoutGetData() invalid delay)");
+ return -1;
+ }
+
+ AudioFrame audioFrame;
+
+ // Retrieve mixed output at the specified rate
+ _outputMixerPtr->MixActiveChannels();
+ _outputMixerPtr->DoOperationsOnCombinedSignal();
+ _outputMixerPtr->GetMixedAudio(samplingFreqHz, 1, audioFrame);
+
+ // Deliver audio (PCM) samples to the external sink
+ memcpy(speechData10ms,
+ audioFrame._payloadData,
+ sizeof(WebRtc_Word16)*(audioFrame._payloadDataLengthInSamples));
+ lengthSamples = audioFrame._payloadDataLengthInSamples;
+
+ // Store current playout delay (to be used by ExternalRecordingInsertData).
+ playout_delay_ms_ = current_delay_ms;
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED,
+ kTraceError,
+ "ExternalPlayoutGetData() external playout is not supported");
+ return -1;
+#endif
+}
+
+#endif // WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_external_media_impl.h b/trunk/src/voice_engine/main/source/voe_external_media_impl.h
new file mode 100644
index 0000000..fa1ff8a
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_external_media_impl.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
+
+#include "voe_external_media.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoEExternalMediaImpl : public virtual voe::SharedData,
+ public VoEExternalMedia,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int RegisterExternalMediaProcessing(
+ int channel,
+ ProcessingTypes type,
+ VoEMediaProcess& processObject);
+
+ virtual int DeRegisterExternalMediaProcessing(
+ int channel,
+ ProcessingTypes type);
+
+ virtual int SetExternalRecordingStatus(bool enable);
+
+ virtual int SetExternalPlayoutStatus(bool enable);
+
+ virtual int ExternalRecordingInsertData(
+ const WebRtc_Word16 speechData10ms[],
+ int lengthSamples,
+ int samplingFreqHz,
+ int current_delay_ms);
+
+ virtual int ExternalPlayoutGetData(WebRtc_Word16 speechData10ms[],
+ int samplingFreqHz,
+ int current_delay_ms,
+ int& lengthSamples);
+
+protected:
+ VoEExternalMediaImpl();
+ virtual ~VoEExternalMediaImpl();
+
+private:
+ int playout_delay_ms_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_file_impl.cc b/trunk/src/voice_engine/main/source/voe_file_impl.cc
new file mode 100644
index 0000000..cd3193f
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_file_impl.cc
@@ -0,0 +1,1423 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_file_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "file_wrapper.h"
+#include "media_file.h"
+#include "output_mixer.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEFile* VoEFile::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_FILE_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEFileImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+
+VoEFileImpl::VoEFileImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEFileImpl::VoEFileImpl() - ctor");
+}
+
+VoEFileImpl::~VoEFileImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEFileImpl::~VoEFileImpl() - dtor");
+}
+
+int VoEFileImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEFile::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEFile reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEFileImpl::StartPlayingFileLocally(
+ int channel,
+ const char fileNameUTF8[1024],
+ bool loop, FileFormats format,
+ float volumeScaling,
+ int startPointMs,
+ int stopPointMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileLocally(channel=%d, fileNameUTF8[]=%s, "
+ "loop=%d, format=%d, volumeScaling=%5.3f, startPointMs=%d,"
+ " stopPointMs=%d)",
+ channel, fileNameUTF8, loop, format, volumeScaling,
+ startPointMs, stopPointMs);
+ assert(1024 == FileWrapper::kMaxFileNameSize);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartPlayingFileLocally() failed to locate channel");
+ return -1;
+ }
+
+ return channelPtr->StartPlayingFileLocally(fileNameUTF8,
+ loop,
+ format,
+ startPointMs,
+ volumeScaling,
+ stopPointMs,
+ NULL);
+}
+
+int VoEFileImpl::StartPlayingFileLocally(int channel,
+ InStream* stream,
+ FileFormats format,
+ float volumeScaling,
+ int startPointMs,
+ int stopPointMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileLocally(channel=%d, stream, format=%d, "
+ "volumeScaling=%5.3f, startPointMs=%d, stopPointMs=%d)",
+ channel, format, volumeScaling, startPointMs, stopPointMs);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartPlayingFileLocally() failed to locate channel");
+ return -1;
+ }
+
+ return channelPtr->StartPlayingFileLocally(stream,
+ format,
+ startPointMs,
+ volumeScaling,
+ stopPointMs,
+ NULL);
+}
+
+int VoEFileImpl::StopPlayingFileLocally(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopPlayingFileLocally()");
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopPlayingFileLocally() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StopPlayingFileLocally();
+}
+
+int VoEFileImpl::IsPlayingFileLocally(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "IsPlayingFileLocally(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopPlayingFileLocally() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->IsPlayingFileLocally();
+}
+
+int VoEFileImpl::ScaleLocalFilePlayout(int channel, float scale)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ScaleLocalFilePlayout(channel=%d, scale=%5.3f)",
+ channel, scale);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopPlayingFileLocally() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->ScaleLocalFilePlayout(scale);
+}
+
+int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
+ const char fileNameUTF8[1024],
+ bool loop,
+ bool mixWithMicrophone,
+ FileFormats format,
+ float volumeScaling)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone(channel=%d, fileNameUTF8=%s, "
+ "loop=%d, mixWithMicrophone=%d, format=%d, "
+ "volumeScaling=%5.3f)",
+ channel, fileNameUTF8, loop, mixWithMicrophone, format,
+ volumeScaling);
+ assert(1024 == FileWrapper::kMaxFileNameSize);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ const WebRtc_UWord32 startPointMs(0);
+ const WebRtc_UWord32 stopPointMs(0);
+
+ if (channel == -1)
+ {
+ int res = _transmitMixerPtr->StartPlayingFileAsMicrophone(
+ fileNameUTF8,
+ loop,
+ format,
+ startPointMs,
+ volumeScaling,
+ stopPointMs,
+ NULL);
+ if (res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone() failed to start"
+ " playing file");
+ return(-1);
+ }
+ else
+ {
+ _transmitMixerPtr->SetMixWithMicStatus(mixWithMicrophone);
+ return(0);
+ }
+ }
+ else
+ {
+ // Add file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartPlayingFileAsMicrophone() failed to locate channel");
+ return -1;
+ }
+
+ int res = channelPtr->StartPlayingFileAsMicrophone(fileNameUTF8,
+ loop,
+ format,
+ startPointMs,
+ volumeScaling,
+ stopPointMs,
+ NULL);
+ if (res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone() failed to start "
+ "playing file");
+ return -1;
+ }
+ else
+ {
+ channelPtr->SetMixWithMicStatus(mixWithMicrophone);
+ return 0;
+ }
+ }
+}
+
+int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
+ InStream* stream,
+ bool mixWithMicrophone,
+ FileFormats format,
+ float volumeScaling)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone(channel=%d, stream,"
+ " mixWithMicrophone=%d, format=%d, volumeScaling=%5.3f)",
+ channel, mixWithMicrophone, format, volumeScaling);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ const WebRtc_UWord32 startPointMs(0);
+ const WebRtc_UWord32 stopPointMs(0);
+
+ if (channel == -1)
+ {
+ int res = _transmitMixerPtr->StartPlayingFileAsMicrophone(
+ stream,
+ format,
+ startPointMs,
+ volumeScaling,
+ stopPointMs,
+ NULL);
+ if (res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone() failed to start"
+ " playing stream");
+ return(-1);
+ }
+ else
+ {
+ _transmitMixerPtr->SetMixWithMicStatus(mixWithMicrophone);
+ return(0);
+ }
+ }
+ else
+ {
+ // Add file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartPlayingFileAsMicrophone() failed to locate channel");
+ return -1;
+ }
+
+ int res = channelPtr->StartPlayingFileAsMicrophone(
+ stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
+ if (res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartPlayingFileAsMicrophone() failed to start"
+ " playing stream");
+ return -1;
+ }
+ else
+ {
+ channelPtr->SetMixWithMicStatus(mixWithMicrophone);
+ return 0;
+ }
+ }
+}
+
+int VoEFileImpl::StopPlayingFileAsMicrophone(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopPlayingFileAsMicrophone(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ // Stop adding file before demultiplexing <=> affects all channels
+ return _transmitMixerPtr->StopPlayingFileAsMicrophone();
+ }
+ else
+ {
+ // Stop adding file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopPlayingFileAsMicrophone() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StopPlayingFileAsMicrophone();
+ }
+}
+
+int VoEFileImpl::IsPlayingFileAsMicrophone(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "IsPlayingFileAsMicrophone(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _transmitMixerPtr->IsPlayingFileAsMicrophone();
+ }
+ else
+ {
+ // Stop adding file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "IsPlayingFileAsMicrophone() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->IsPlayingFileAsMicrophone();
+ }
+}
+
+int VoEFileImpl::ScaleFileAsMicrophonePlayout(int channel, float scale)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ScaleFileAsMicrophonePlayout(channel=%d, scale=%5.3f)",
+ channel, scale);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _transmitMixerPtr->ScaleFileAsMicrophonePlayout(scale);
+ }
+ else
+ {
+ // Stop adding file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "IsPlayingFileAsMicrophone() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->ScaleFileAsMicrophonePlayout(scale);
+ }
+}
+
+int VoEFileImpl::StartRecordingPlayout(
+ int channel, const char* fileNameUTF8, CodecInst* compression,
+ int maxSizeBytes)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout(channel=%d, fileNameUTF8=%s, "
+ "compression, maxSizeBytes=%d)",
+ channel, fileNameUTF8, maxSizeBytes);
+ assert(1024 == FileWrapper::kMaxFileNameSize);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ _outputMixerPtr->StartRecordingPlayout(fileNameUTF8, compression);
+ return 0;
+ }
+ else
+ {
+ // Add file after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartRecordingPlayout() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StartRecordingPlayout(fileNameUTF8, compression);
+ }
+}
+
+int VoEFileImpl::StartRecordingPlayout(
+ int channel, OutStream* stream, CodecInst* compression)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingPlayout(channel=%d, stream, compression)",
+ channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _outputMixerPtr->StartRecordingPlayout(stream, compression);
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartRecordingPlayout() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StartRecordingPlayout(stream, compression);
+ }
+}
+
+int VoEFileImpl::StopRecordingPlayout(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopRecordingPlayout(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _outputMixerPtr->StopRecordingPlayout();
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopRecordingPlayout() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StopRecordingPlayout();
+ }
+}
+
+int VoEFileImpl::StartRecordingMicrophone(
+ const char* fileNameUTF8, CodecInst* compression, int maxSizeBytes)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone(fileNameUTF8=%s, compression, "
+ "maxSizeBytes=%d)", fileNameUTF8, maxSizeBytes);
+ assert(1024 == FileWrapper::kMaxFileNameSize);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (_transmitMixerPtr->StartRecordingMicrophone(fileNameUTF8, compression))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to start recording");
+ return -1;
+ }
+ if (_audioDevicePtr->Recording())
+ {
+ return 0;
+ }
+ if (!_externalRecording)
+ {
+ if (_audioDevicePtr->InitRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to initialize"
+ " recording");
+ return -1;
+ }
+ if (_audioDevicePtr->StartRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to start recording");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+int VoEFileImpl::StartRecordingMicrophone(
+ OutStream* stream, CodecInst* compression)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone(stream, compression)");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (_transmitMixerPtr->StartRecordingMicrophone(stream, compression) == -1)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to start recording");
+ return -1;
+ }
+ if (_audioDevicePtr->Recording())
+ {
+ return 0;
+ }
+ if (!_externalRecording)
+ {
+ if (_audioDevicePtr->InitRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to initialize "
+ "recording");
+ return -1;
+ }
+ if (_audioDevicePtr->StartRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRecordingMicrophone() failed to start"
+ " recording");
+ return -1;
+ }
+ }
+ return 0;
+}
+
+int VoEFileImpl::StopRecordingMicrophone()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopRecordingMicrophone()");
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if ((NumOfSendingChannels() == 0)&&_audioDevicePtr->Recording())
+ {
+ // Stop audio-device recording if no channel is recording
+ if (_audioDevicePtr->StopRecording() != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_CANNOT_STOP_RECORDING, kTraceError,
+ "StopRecordingMicrophone() failed to stop recording");
+ return -1;
+ }
+ }
+ return _transmitMixerPtr->StopRecordingMicrophone();
+}
+
+int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV(fileNameInUTF8=%s, fileNameOutUTF8=%s)",
+ fileNameInUTF8, fileNameOutUTF8);
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
+ -1,
+ kFileFormatPcm16kHzFile));
+
+ int res=playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0, NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "ConvertPCMToWAV failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatWavFile));
+
+ CodecInst codecInst;
+ strncpy(codecInst.plname,"L16",32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+
+ res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "ConvertPCMToWAV failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV failed during conversion "
+ "(audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertPCMToWAV(InStream* streamIn, OutStream* streamOut)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV(streamIn, streamOut)");
+
+ if ((streamIn == NULL) || (streamOut == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "invalid stream handles");
+ return (-1);
+ }
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
+ kFileFormatPcm16kHzFile));
+ int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToWAV failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(-1,
+ kFileFormatWavFile));
+ CodecInst codecInst;
+ strncpy(codecInst.plname, "L16", 32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+ res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToWAV failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength, frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV failed during conversion "
+ "(create audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToWAV failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertWAVToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM(fileNameInUTF8=%s, fileNameOutUTF8=%s)",
+ fileNameInUTF8, fileNameOutUTF8);
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
+ kFileFormatWavFile));
+ int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "ConvertWAVToPCM failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatPcm16kHzFile));
+
+ CodecInst codecInst;
+ strncpy(codecInst.plname,"L16",32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+
+ res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "ConvertWAVToPCM failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM failed during conversion "
+ "(audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertWAVToPCM(InStream* streamIn, OutStream* streamOut)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM(streamIn, streamOut)");
+
+ if ((streamIn == NULL) || (streamOut == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "invalid stream handles");
+ return (-1);
+ }
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
+ kFileFormatWavFile));
+ int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertWAVToPCM failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatPcm16kHzFile));
+
+ CodecInst codecInst;
+ strncpy(codecInst.plname,"L16",32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+
+ res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertWAVToPCM failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength, frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM failed during conversion "
+ "(audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertWAVToPCM failed during converstion"
+ " (write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertPCMToCompressed(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8,
+ CodecInst* compression)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed(fileNameInUTF8=%s, fileNameOutUTF8=%s"
+ ", compression)", fileNameInUTF8, fileNameOutUTF8);
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ " compression: plname=%s, plfreq=%d, pacsize=%d",
+ compression->plname, compression->plfreq,
+ compression->pacsize);
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
+ -1,
+ kFileFormatPcm16kHzFile));
+ int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0, NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToCompressed failed to create player object");
+ // Clean up and shutdown the file player
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1,
+ kFileFormatCompressedFile));
+ res = recObj.StartRecordingAudioFile(fileNameOutUTF8, *compression,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToCompressed failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed failed during conversion "
+ "(audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertPCMToCompressed(InStream* streamIn,
+ OutStream* streamOut,
+ CodecInst* compression)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed(streamIn, streamOut, compression)");
+
+ if ((streamIn == NULL) || (streamOut == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "invalid stream handles");
+ return (-1);
+ }
+
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ " compression: plname=%s, plfreq=%d, pacsize=%d",
+ compression->plname, compression->plfreq,
+ compression->pacsize);
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
+ -1, kFileFormatPcm16kHzFile));
+
+ int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToCompressed failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatCompressedFile));
+ res = recObj.StartRecordingAudioFile(*streamOut,*compression,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertPCMToCompressed failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency, AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed failed during conversion"
+ " (audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertPCMToCompressed failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertCompressedToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM(fileNameInUTF8=%s,"
+ " fileNameOutUTF8=%s)",
+ fileNameInUTF8, fileNameOutUTF8);
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
+ -1, kFileFormatCompressedFile));
+
+ int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertCompressedToPCM failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatPcm16kHzFile));
+
+ CodecInst codecInst;
+ strncpy(codecInst.plname,"L16",32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+
+ res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertCompressedToPCM failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM failed during conversion "
+ "(create audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM failed during converstion "
+ "(write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+int VoEFileImpl::ConvertCompressedToPCM(InStream* streamIn,
+ OutStream* streamOut)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM(file, file);");
+
+ if ((streamIn == NULL) || (streamOut == NULL))
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "invalid stream handles");
+ return (-1);
+ }
+
+ // Create file player object
+ FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
+ -1, kFileFormatCompressedFile));
+ int res;
+
+ res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertCompressedToPCM failed to create player object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ return -1;
+ }
+
+ // Create file recorder object
+ FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
+ -1, kFileFormatPcm16kHzFile));
+
+ CodecInst codecInst;
+ strncpy(codecInst.plname,"L16",32);
+ codecInst.channels = 1;
+ codecInst.rate = 256000;
+ codecInst.plfreq = 16000;
+ codecInst.pltype = 94;
+ codecInst.pacsize = 160;
+
+ res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "ConvertCompressedToPCM failed to create recorder object");
+ playerObj.StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ recObj.StopRecording();
+ FileRecorder::DestroyFileRecorder(&recObj);
+ return -1;
+ }
+
+ // Run throught the file
+ AudioFrame audioFrame;
+ WebRtc_Word16 decodedData[160];
+ WebRtc_UWord32 decLength=0;
+ const WebRtc_UWord32 frequency = 16000;
+
+ while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
+ {
+ if(decLength!=frequency/100)
+ {
+ // This is an OK way to end
+ break;
+ }
+ res=audioFrame.UpdateFrame(-1, 0, decodedData,
+ (WebRtc_UWord16)decLength,
+ frequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadActive);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM failed during conversion"
+ " (audio frame)");
+ break;
+ }
+
+ res=recObj.RecordAudioToFile(audioFrame);
+ if(res)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
+ "ConvertCompressedToPCM failed during converstion"
+ " (write frame)");
+ }
+ }
+
+ playerObj.StopPlayingFile();
+ recObj.StopRecording();
+ FilePlayer::DestroyFilePlayer(&playerObj);
+ FileRecorder::DestroyFileRecorder(&recObj);
+
+ return res;
+}
+
+
+int VoEFileImpl::GetFileDuration(const char* fileNameUTF8,
+ int& durationMs,
+ FileFormats format)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetFileDuration(fileNameUTF8=%s, format=%d)",
+ fileNameUTF8, format);
+
+ // Create a dummy file module for this
+ MediaFile * fileModule=MediaFile::CreateMediaFile(-1);
+
+ // Temp container of the right format
+ WebRtc_UWord32 duration;
+ int res=fileModule->FileDurationMs(fileNameUTF8,duration,format);
+ if (res)
+ {
+ _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
+ "GetFileDuration() failed measure file duration");
+ return -1;
+ }
+ durationMs = duration;
+ MediaFile::DestroyMediaFile(fileModule);
+ fileModule = NULL;
+
+ return(res);
+}
+
+int VoEFileImpl::GetPlaybackPosition(int channel, int& positionMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetPlaybackPosition(channel=%d)", channel);
+
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetPlaybackPosition() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetLocalPlayoutPosition(positionMs);
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_FILE_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_file_impl.h b/trunk/src/voice_engine/main/source/voe_file_impl.h
new file mode 100644
index 0000000..2d93876
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_file_impl.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
+
+#include "voe_file.h"
+#include "shared_data.h"
+#include "ref_count.h"
+
+namespace webrtc {
+
+class VoEFileImpl : public virtual voe::SharedData,
+ public VoEFile, public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ // Playout file locally
+
+ virtual int StartPlayingFileLocally(
+ int channel,
+ const char fileNameUTF8[1024],
+ bool loop = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0,
+ int startPointMs = 0,
+ int stopPointMs = 0);
+
+ virtual int StartPlayingFileLocally(
+ int channel,
+ InStream* stream,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0,
+ int startPointMs = 0, int stopPointMs = 0);
+
+ virtual int StopPlayingFileLocally(int channel);
+
+ virtual int IsPlayingFileLocally(int channel);
+
+ virtual int ScaleLocalFilePlayout(int channel, float scale);
+
+ // Use file as microphone input
+
+ virtual int StartPlayingFileAsMicrophone(
+ int channel,
+ const char fileNameUTF8[1024],
+ bool loop = false ,
+ bool mixWithMicrophone = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0);
+
+ virtual int StartPlayingFileAsMicrophone(
+ int channel,
+ InStream* stream,
+ bool mixWithMicrophone = false,
+ FileFormats format = kFileFormatPcm16kHzFile,
+ float volumeScaling = 1.0);
+
+ virtual int StopPlayingFileAsMicrophone(int channel);
+
+ virtual int IsPlayingFileAsMicrophone(int channel);
+
+ virtual int ScaleFileAsMicrophonePlayout(int channel, float scale);
+
+ // Record speaker signal to file
+
+ virtual int StartRecordingPlayout(int channel,
+ const char* fileNameUTF8,
+ CodecInst* compression = NULL,
+ int maxSizeBytes = -1);
+
+ virtual int StartRecordingPlayout(int channel,
+ OutStream* stream,
+ CodecInst* compression = NULL);
+
+ virtual int StopRecordingPlayout(int channel);
+
+ // Record microphone signal to file
+
+ virtual int StartRecordingMicrophone(const char* fileNameUTF8,
+ CodecInst* compression = NULL,
+ int maxSizeBytes = -1);
+
+ virtual int StartRecordingMicrophone(OutStream* stream,
+ CodecInst* compression = NULL);
+
+ virtual int StopRecordingMicrophone();
+
+ // Conversion between different file formats
+
+ virtual int ConvertPCMToWAV(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8);
+
+ virtual int ConvertPCMToWAV(InStream* streamIn,
+ OutStream* streamOut);
+
+ virtual int ConvertWAVToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8);
+
+ virtual int ConvertWAVToPCM(InStream* streamIn,
+ OutStream* streamOut);
+
+ virtual int ConvertPCMToCompressed(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8,
+ CodecInst* compression);
+
+ virtual int ConvertPCMToCompressed(InStream* streamIn,
+ OutStream* streamOut,
+ CodecInst* compression);
+
+ virtual int ConvertCompressedToPCM(const char* fileNameInUTF8,
+ const char* fileNameOutUTF8);
+
+ virtual int ConvertCompressedToPCM(InStream* streamIn,
+ OutStream* streamOut);
+
+ // Misc file functions
+
+ virtual int GetFileDuration(
+ const char* fileNameUTF8,
+ int& durationMs,
+ FileFormats format = kFileFormatPcm16kHzFile);
+
+ virtual int GetPlaybackPosition(int channel, int& positionMs);
+
+protected:
+ VoEFileImpl();
+ virtual ~VoEFileImpl();
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
+
diff --git a/trunk/src/voice_engine/main/source/voe_hardware_impl.cc b/trunk/src/voice_engine/main/source/voe_hardware_impl.cc
new file mode 100644
index 0000000..7e676c0
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_hardware_impl.cc
@@ -0,0 +1,850 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_hardware_impl.h"
+
+#include <cassert>
+
+#include "cpu_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc
+{
+
+VoEHardware* VoEHardware::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_HARDWARE_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEHardwareImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+
+VoEHardwareImpl::VoEHardwareImpl() :
+ _cpu(NULL)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEHardwareImpl() - ctor");
+
+ _cpu = CpuWrapper::CreateCpu();
+ if (_cpu)
+ {
+ _cpu->CpuUsage(); // init cpu usage
+ }
+}
+
+VoEHardwareImpl::~VoEHardwareImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "~VoEHardwareImpl() - dtor");
+
+ if (_cpu)
+ {
+ delete _cpu;
+ _cpu = NULL;
+ }
+}
+
+int VoEHardwareImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEHardwareImpl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoEHardwareImpl reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEHardwareImpl::SetAudioDeviceLayer(AudioLayers audioLayer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetAudioDeviceLayer(audioLayer=%d)", audioLayer);
+
+ // Don't allow a change if VoE is initialized
+ if (_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_ALREADY_INITED, kTraceError);
+ return -1;
+ }
+
+ // Map to AudioDeviceModule::AudioLayer
+ AudioDeviceModule::AudioLayer
+ wantedLayer(AudioDeviceModule::kPlatformDefaultAudio);
+ switch (audioLayer)
+ {
+ case kAudioPlatformDefault:
+ // already set above
+ break;
+ case kAudioWindowsCore:
+ wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
+ break;
+ case kAudioWindowsWave:
+ wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
+ break;
+ case kAudioLinuxAlsa:
+ wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
+ break;
+ case kAudioLinuxPulse:
+ wantedLayer = AudioDeviceModule::kLinuxPulseAudio;
+ break;
+ }
+
+ // Save the audio device layer for Init()
+ _audioDeviceLayer = wantedLayer;
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetAudioDeviceLayer(AudioLayers& audioLayer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetAudioDeviceLayer(devices=?)");
+
+ // Can always be called regardless of VoE state
+
+ AudioDeviceModule::AudioLayer
+ activeLayer(AudioDeviceModule::kPlatformDefaultAudio);
+
+ if (_audioDevicePtr)
+ {
+ // Get active audio layer from ADM
+ if (_audioDevicePtr->ActiveAudioLayer(&activeLayer) != 0)
+ {
+ _engineStatistics.SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
+ " Audio Device error");
+ return -1;
+ }
+ }
+ else
+ {
+ // Return VoE's internal layer setting
+ activeLayer = _audioDeviceLayer;
+ }
+
+ // Map to AudioLayers
+ switch (activeLayer)
+ {
+ case AudioDeviceModule::kPlatformDefaultAudio:
+ audioLayer = kAudioPlatformDefault;
+ break;
+ case AudioDeviceModule::kWindowsCoreAudio:
+ audioLayer = kAudioWindowsCore;
+ break;
+ case AudioDeviceModule::kWindowsWaveAudio:
+ audioLayer = kAudioWindowsWave;
+ break;
+ case AudioDeviceModule::kLinuxAlsaAudio:
+ audioLayer = kAudioLinuxAlsa;
+ break;
+ case AudioDeviceModule::kLinuxPulseAudio:
+ audioLayer = kAudioLinuxPulse;
+ break;
+ default:
+ _engineStatistics.SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
+ " unknown audio layer");
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: audioLayer=%d", audioLayer);
+
+ return 0;
+}
+int VoEHardwareImpl::GetNumOfRecordingDevices(int& devices)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetNumOfRecordingDevices(devices=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ devices = static_cast<int> (_audioDevicePtr->RecordingDevices());
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: devices=%d", devices);
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetNumOfPlayoutDevices(int& devices)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetNumOfPlayoutDevices(devices=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ devices = static_cast<int> (_audioDevicePtr->PlayoutDevices());
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: devices=%d", devices);
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetRecordingDeviceName(int index,
+ char strNameUTF8[128],
+ char strGuidUTF8[128])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRecordingDeviceName(index=%d)", index);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (strNameUTF8 == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetRecordingDeviceName() invalid argument");
+ return -1;
+ }
+
+ // Note that strGuidUTF8 is allowed to be NULL
+
+ // Init len variable to length of supplied vectors
+ const WebRtc_UWord16 strLen = 128;
+
+ // Check if length has been changed in module
+ assert(strLen == kAdmMaxDeviceNameSize);
+ assert(strLen == kAdmMaxGuidSize);
+
+ WebRtc_Word8 name[strLen];
+ WebRtc_Word8 guid[strLen];
+
+ // Get names from module
+ if (_audioDevicePtr->RecordingDeviceName(index, name, guid) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
+ "GetRecordingDeviceName() failed to get device name");
+ return -1;
+ }
+
+ // Copy to vectors supplied by user
+ strncpy(strNameUTF8, name, strLen);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: strNameUTF8=%s", strNameUTF8);
+
+ if (strGuidUTF8 != NULL)
+ {
+ strncpy(strGuidUTF8, guid, strLen);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: strGuidUTF8=%s", strGuidUTF8);
+ }
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetPlayoutDeviceName(int index,
+ char strNameUTF8[128],
+ char strGuidUTF8[128])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetPlayoutDeviceName(index=%d)", index);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (strNameUTF8 == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetPlayoutDeviceName() invalid argument");
+ return -1;
+ }
+
+ // Note that strGuidUTF8 is allowed to be NULL
+
+ // Init len variable to length of supplied vectors
+ const WebRtc_UWord16 strLen = 128;
+
+ // Check if length has been changed in module
+ assert(strLen == kAdmMaxDeviceNameSize);
+ assert(strLen == kAdmMaxGuidSize);
+
+ WebRtc_Word8 name[strLen];
+ WebRtc_Word8 guid[strLen];
+
+ // Get names from module
+ if (_audioDevicePtr->PlayoutDeviceName(index, name, guid) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
+ "GetPlayoutDeviceName() failed to get device name");
+ return -1;
+ }
+
+ // Copy to vectors supplied by user
+ strncpy(strNameUTF8, name, strLen);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: strNameUTF8=%s", strNameUTF8);
+
+ if (strGuidUTF8 != NULL)
+ {
+ strncpy(strGuidUTF8, guid, strLen);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: strGuidUTF8=%s", strGuidUTF8);
+ }
+
+ return 0;
+}
+
+int VoEHardwareImpl::SetRecordingDevice(int index,
+ StereoChannel recordingChannel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetRecordingDevice(index=%d, recordingChannel=%d)",
+ index, (int) recordingChannel);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool isRecording(false);
+
+ // Store state about activated recording to be able to restore it after the
+ // recording device has been modified.
+ if (_audioDevicePtr->Recording())
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "SetRecordingDevice() device is modified while recording"
+ " is active...");
+ isRecording = true;
+ if (_audioDevicePtr->StopRecording() == -1)
+ {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+ "SetRecordingDevice() unable to stop recording");
+ return -1;
+ }
+ }
+
+ // We let the module do the index sanity
+
+ // Set recording channel
+ AudioDeviceModule::ChannelType recCh =
+ AudioDeviceModule::kChannelBoth;
+ switch (recordingChannel)
+ {
+ case kStereoLeft:
+ recCh = AudioDeviceModule::kChannelLeft;
+ break;
+ case kStereoRight:
+ recCh = AudioDeviceModule::kChannelRight;
+ break;
+ case kStereoBoth:
+ // default setting kChannelBoth (<=> mono)
+ break;
+ }
+
+ if (_audioDevicePtr->SetRecordingChannel(recCh) != 0) {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+ "SetRecordingChannel() unable to set the recording channel");
+ }
+
+ // Map indices to unsigned since underlying functions need that
+ WebRtc_UWord16 indexU = static_cast<WebRtc_UWord16> (index);
+
+ WebRtc_Word32 res(0);
+
+ if (index == -1)
+ {
+ res = _audioDevicePtr->SetRecordingDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice);
+ }
+ else if (index == -2)
+ {
+ res = _audioDevicePtr->SetRecordingDevice(
+ AudioDeviceModule::kDefaultDevice);
+ }
+ else
+ {
+ res = _audioDevicePtr->SetRecordingDevice(indexU);
+ }
+
+ if (res != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+ "SetRecordingDevice() unable to set the recording device");
+ return -1;
+ }
+
+ // Init microphone, so user can do volume settings etc
+ if (_audioDevicePtr->InitMicrophone() == -1)
+ {
+ _engineStatistics.SetLastError(
+ VE_CANNOT_ACCESS_MIC_VOL, kTraceWarning,
+ "SetRecordingDevice() cannot access microphone");
+ }
+
+ // Set number of channels
+ bool available(false);
+ if (_audioDevicePtr->StereoRecordingIsAvailable(&available) != 0) {
+ _engineStatistics.SetLastError(
+ VE_SOUNDCARD_ERROR, kTraceWarning,
+ "StereoRecordingIsAvailable() failed to query stereo recording");
+ }
+
+ if (_audioDevicePtr->SetStereoRecording(available ? true : false) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_SOUNDCARD_ERROR, kTraceWarning,
+ "SetRecordingDevice() failed to set mono recording mode");
+ }
+
+ // Restore recording if it was enabled already when calling this function.
+ if (isRecording)
+ {
+ if (!_externalRecording)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "SetRecordingDevice() recording is now being "
+ "restored...");
+ if (_audioDevicePtr->InitRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetRecordingDevice() failed to initialize "
+ "recording");
+ return -1;
+ }
+ if (_audioDevicePtr->StartRecording() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetRecordingDevice() failed to start recording");
+ return -1;
+ }
+ }
+ }
+
+ return 0;
+}
+
+int VoEHardwareImpl::SetPlayoutDevice(int index)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetPlayoutDevice(index=%d)", index);
+ CriticalSectionScoped cs(*_apiCritPtr);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool isPlaying(false);
+
+ // Store state about activated playout to be able to restore it after the
+ // playout device has been modified.
+ if (_audioDevicePtr->Playing())
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "SetPlayoutDevice() device is modified while playout is "
+ "active...");
+ isPlaying = true;
+ if (_audioDevicePtr->StopPlayout() == -1)
+ {
+ _engineStatistics.SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+ "SetPlayoutDevice() unable to stop playout");
+ return -1;
+ }
+ }
+
+ // We let the module do the index sanity
+
+ // Map indices to unsigned since underlying functions need that
+ WebRtc_UWord16 indexU = static_cast<WebRtc_UWord16> (index);
+
+ WebRtc_Word32 res(0);
+
+ if (index == -1)
+ {
+ res = _audioDevicePtr->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultCommunicationDevice);
+ }
+ else if (index == -2)
+ {
+ res = _audioDevicePtr->SetPlayoutDevice(
+ AudioDeviceModule::kDefaultDevice);
+ }
+ else
+ {
+ res = _audioDevicePtr->SetPlayoutDevice(indexU);
+ }
+
+ if (res != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_SOUNDCARD_ERROR, kTraceError,
+ "SetPlayoutDevice() unable to set the playout device");
+ return -1;
+ }
+
+ // Init speaker, so user can do volume settings etc
+ if (_audioDevicePtr->InitSpeaker() == -1)
+ {
+ _engineStatistics.SetLastError(
+ VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceWarning,
+ "SetPlayoutDevice() cannot access speaker");
+ }
+
+ // Set number of channels
+ bool available(false);
+ _audioDevicePtr->StereoPlayoutIsAvailable(&available);
+ if (_audioDevicePtr->SetStereoPlayout(available ? true : false) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_SOUNDCARD_ERROR, kTraceWarning,
+ "SetPlayoutDevice() failed to set stereo playout mode");
+ }
+
+ // Restore playout if it was enabled already when calling this function.
+ if (isPlaying)
+ {
+ if (!_externalPlayout)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "SetPlayoutDevice() playout is now being restored...");
+ if (_audioDevicePtr->InitPlayout() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetPlayoutDevice() failed to initialize playout");
+ return -1;
+ }
+ if (_audioDevicePtr->StartPlayout() != 0)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetPlayoutDevice() failed to start playout");
+ return -1;
+ }
+ }
+ }
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetRecordingDeviceStatus(bool& isAvailable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetRecordingDeviceStatus()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // We let the module do isRecording sanity
+
+ bool available(false);
+
+ // Check availability
+ if (_audioDevicePtr->RecordingIsAvailable(&available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_UNDEFINED_SC_REC_ERR, kTraceError,
+ " Audio Device error");
+ return -1;
+ }
+
+ isAvailable = available;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: isAvailable = %d)", (int) isAvailable);
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetPlayoutDeviceStatus(bool& isAvailable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetPlayoutDeviceStatus()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // We let the module do isPlaying sanity
+
+ bool available(false);
+
+ // Check availability
+ if (_audioDevicePtr->PlayoutIsAvailable(&available) != 0)
+ {
+ _engineStatistics.SetLastError(VE_PLAY_UNDEFINED_SC_ERR,
+ kTraceError, " Audio Device error");
+ return -1;
+ }
+
+ isAvailable = available;
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: isAvailable = %d)", (int) isAvailable);
+
+ return 0;
+}
+
+int VoEHardwareImpl::ResetAudioDevice()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "ResetAudioDevice()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+#if defined(MAC_IPHONE)
+ if (_audioDevicePtr->ResetAudioDevice() < 0)
+ {
+ _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
+ " Failed to reset sound device");
+ return -1;
+ }
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ " no support for resetting sound device");
+ return -1;
+#endif
+
+ return 0;
+}
+
+int VoEHardwareImpl::AudioDeviceControl(unsigned int par1, unsigned int par2,
+ unsigned int par3)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "AudioDeviceControl(%i, %i, %i)", par1, par2, par3);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ " no support for resetting sound device");
+ return -1;
+}
+
+int VoEHardwareImpl::SetLoudspeakerStatus(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetLoudspeakerStatus(enable=%i)", (int) enable);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+#if defined(WEBRTC_ANDROID)
+ if (_audioDevicePtr->SetLoudspeakerStatus(enable) < 0)
+ {
+ _engineStatistics.SetLastError(VE_IGNORED_FUNCTION, kTraceError,
+ " Failed to set loudspeaker status");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ " no support for setting loudspeaker"
+ " status");
+ return -1;
+#endif
+}
+
+int VoEHardwareImpl::GetLoudspeakerStatus(bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLoudspeakerStatus()");
+ IPHONE_NOT_SUPPORTED();
+
+#if defined(WEBRTC_ANDROID)
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioDevicePtr->GetLoudspeakerStatus(&enabled) < 0)
+ {
+ _engineStatistics.SetLastError(VE_IGNORED_FUNCTION, kTraceError,
+ " Failed to get loudspeaker status");
+ return -1;
+ }
+
+ return 0;
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ " no support for setting loudspeaker "
+ "status");
+ return -1;
+#endif
+}
+
+int VoEHardwareImpl::GetCPULoad(int& loadPercent)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetCPULoad()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // Get CPU load from ADM
+ WebRtc_UWord16 load(0);
+ if (_audioDevicePtr->CPULoad(&load) != 0)
+ {
+ _engineStatistics.SetLastError(VE_CPU_INFO_ERROR, kTraceError,
+ " error getting system CPU load");
+ return -1;
+ }
+
+ loadPercent = static_cast<int> (load);
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: loadPercent = %d", loadPercent);
+
+ return 0;
+}
+
+int VoEHardwareImpl::GetSystemCPULoad(int& loadPercent)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSystemCPULoad(loadPercent=?)");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ // Check if implemented for this platform
+ if (!_cpu)
+ {
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ " no support for getting system CPU "
+ "load");
+ return -1;
+ }
+
+ // Get CPU load
+ WebRtc_Word32 load = _cpu->CpuUsage();
+ if (load < 0)
+ {
+ _engineStatistics.SetLastError(VE_CPU_INFO_ERROR, kTraceError,
+ " error getting system CPU load");
+ return -1;
+ }
+
+ loadPercent = static_cast<int> (load);
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ " Output: loadPercent = %d", loadPercent);
+
+ return 0;
+}
+
+int VoEHardwareImpl::EnableBuiltInAEC(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "%s", __FUNCTION__);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ return _audioDevicePtr->EnableBuiltInAEC(enable);
+}
+
+bool VoEHardwareImpl::BuiltInAECIsEnabled() const
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "%s", __FUNCTION__);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return false;
+ }
+
+ return _audioDevicePtr->BuiltInAECIsEnabled();
+}
+
+#endif // WEBRTC_VOICE_ENGINE_HARDWARE_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_hardware_impl.h b/trunk/src/voice_engine/main/source/voe_hardware_impl.h
new file mode 100644
index 0000000..af934ba
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_hardware_impl.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
+
+#include "voe_hardware.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc
+{
+class CpuWrapper;
+
+class VoEHardwareImpl: public virtual voe::SharedData,
+ public VoEHardware,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int GetNumOfRecordingDevices(int& devices);
+
+ virtual int GetNumOfPlayoutDevices(int& devices);
+
+ virtual int GetRecordingDeviceName(int index,
+ char strNameUTF8[128],
+ char strGuidUTF8[128]);
+
+ virtual int GetPlayoutDeviceName(int index,
+ char strNameUTF8[128],
+ char strGuidUTF8[128]);
+
+ virtual int GetRecordingDeviceStatus(bool& isAvailable);
+
+ virtual int GetPlayoutDeviceStatus(bool& isAvailable);
+
+ virtual int SetRecordingDevice(
+ int index,
+ StereoChannel recordingChannel = kStereoBoth);
+
+ virtual int SetPlayoutDevice(int index);
+
+ virtual int SetAudioDeviceLayer(AudioLayers audioLayer);
+
+ virtual int GetAudioDeviceLayer(AudioLayers& audioLayer);
+
+ virtual int GetCPULoad(int& loadPercent);
+
+ virtual int GetSystemCPULoad(int& loadPercent);
+
+ virtual int ResetAudioDevice();
+
+ virtual int AudioDeviceControl(unsigned int par1,
+ unsigned int par2,
+ unsigned int par3);
+
+ virtual int SetLoudspeakerStatus(bool enable);
+
+ virtual int GetLoudspeakerStatus(bool& enabled);
+
+ virtual int EnableBuiltInAEC(bool enable);
+ virtual bool BuiltInAECIsEnabled() const;
+
+protected:
+ VoEHardwareImpl();
+ virtual ~VoEHardwareImpl();
+
+private:
+ CpuWrapper* _cpu;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc
new file mode 100644
index 0000000..fd87a97
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_neteq_stats_impl.h"
+
+#include "audio_coding_module.h"
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+
+namespace webrtc {
+
+VoENetEqStats* VoENetEqStats::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoENetEqStatsImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+
+VoENetEqStatsImpl::VoENetEqStatsImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoENetEqStatsImpl::VoENetEqStatsImpl() - ctor");
+}
+
+VoENetEqStatsImpl::~VoENetEqStatsImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoENetEqStatsImpl::~VoENetEqStatsImpl() - dtor");
+}
+
+int VoENetEqStatsImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoENetEqStats::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _engineStatistics.SetLastError(
+ VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoENetEqStats reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
+ NetworkStatistics& stats)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetNetworkStatistics(channel=%d, stats=?)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetNetworkStatistics() failed to locate channel");
+ return -1;
+ }
+
+ return channelPtr->GetNetworkStatistics(stats);
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h
new file mode 100644
index 0000000..47dd2f6
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
+
+#include "voe_neteq_stats.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoENetEqStatsImpl : public virtual voe::SharedData,
+ public VoENetEqStats,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int GetNetworkStatistics(int channel,
+ NetworkStatistics& stats);
+
+protected:
+ VoENetEqStatsImpl();
+ virtual ~VoENetEqStatsImpl();
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_network_impl.cc b/trunk/src/voice_engine/main/source/voe_network_impl.cc
new file mode 100644
index 0000000..0bebbf5
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_network_impl.cc
@@ -0,0 +1,947 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_network_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc
+{
+
+VoENetwork* VoENetwork::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_NETWORK_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoENetworkImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+
+VoENetworkImpl::VoENetworkImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "VoENetworkImpl() - ctor");
+}
+
+VoENetworkImpl::~VoENetworkImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "~VoENetworkImpl() - dtor");
+}
+
+int VoENetworkImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "VoENetworkImpl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset();
+ _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "VoENetworkImpl reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoENetworkImpl::RegisterExternalTransport(int channel,
+ Transport& transport)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetExternalTransport(channel=%d, transport=0x%x)",
+ channel, &transport);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetExternalTransport() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterExternalTransport(transport);
+}
+
+int VoENetworkImpl::DeRegisterExternalTransport(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "DeRegisterExternalTransport(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterExternalTransport() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterExternalTransport();
+}
+
+int VoENetworkImpl::ReceivedRTPPacket(int channel,
+ const void* data,
+ unsigned int length)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "ReceivedRTPPacket(channel=%d, length=%u)", channel, length);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if ((length < 12) || (length > 807))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_PACKET, kTraceError,
+ "ReceivedRTPPacket() invalid packet length");
+ return -1;
+ }
+ if (NULL == data)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "ReceivedRTPPacket() invalid data vector");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "ReceivedRTPPacket() failed to locate channel");
+ return -1;
+ }
+
+ if (!channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ReceivedRTPPacket() external transport is not enabled");
+ return -1;
+ }
+ return channelPtr->ReceivedRTPPacket((const WebRtc_Word8*) data, length);
+}
+
+int VoENetworkImpl::ReceivedRTCPPacket(int channel, const void* data,
+ unsigned int length)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "ReceivedRTCPPacket(channel=%d, length=%u)", channel, length);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (length < 4)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_PACKET, kTraceError,
+ "ReceivedRTCPPacket() invalid packet length");
+ return -1;
+ }
+ if (NULL == data)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "ReceivedRTCPPacket() invalid data vector");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "ReceivedRTCPPacket() failed to locate channel");
+ return -1;
+ }
+ if (!channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ReceivedRTCPPacket() external transport is not enabled");
+ return -1;
+ }
+ return channelPtr->ReceivedRTCPPacket((const WebRtc_Word8*) data, length);
+}
+
+int VoENetworkImpl::GetSourceInfo(int channel,
+ int& rtpPort,
+ int& rtcpPort,
+ char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSourceInfo(channel=%d, rtpPort=?, rtcpPort=?, ipAddr[]=?)",
+ channel);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (NULL == ipAddr)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetSourceInfo() invalid IP-address buffer");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSourceInfo() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "GetSourceInfo() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->GetSourceInfo(rtpPort, rtcpPort, ipAddr);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "GetSourceInfo() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::GetLocalIP(char ipAddr[64], bool ipv6)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLocalIP(ipAddr[]=?, ipv6=%d)", ipv6);
+ IPHONE_NOT_SUPPORTED();
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (NULL == ipAddr)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetLocalIP() invalid IP-address buffer");
+ return -1;
+ }
+
+ // Create a temporary socket module to ensure that this method can be
+ // called also when no channels are created.
+ WebRtc_UWord8 numSockThreads(1);
+ UdpTransport* socketPtr =
+ UdpTransport::Create(
+ -1,
+ numSockThreads);
+ if (NULL == socketPtr)
+ {
+ _engineStatistics.SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
+ "GetLocalIP() failed to create socket module");
+ return -1;
+ }
+
+ char localIPAddr[64];
+
+ if (ipv6)
+ {
+ char localIP[16];
+ if (socketPtr->LocalHostAddressIPV6(localIP) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "GetLocalIP() failed to retrieve local IP - 1");
+ UdpTransport::Destroy(socketPtr);
+ return -1;
+ }
+ // Convert 128-bit address to character string (a:b:c:d:e:f:g:h)
+ sprintf(localIPAddr,
+ "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x"
+ "%.2x:%.2x%.2x",
+ localIP[0], localIP[1], localIP[2], localIP[3], localIP[4],
+ localIP[5], localIP[6], localIP[7], localIP[8], localIP[9],
+ localIP[10], localIP[11], localIP[12], localIP[13],
+ localIP[14], localIP[15]);
+ }
+ else
+ {
+ WebRtc_UWord32 localIP(0);
+ // Read local IP (as 32-bit address) from the socket module
+ if (socketPtr->LocalHostAddress(localIP) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_IP_ADDRESS, kTraceError,
+ "GetLocalIP() failed to retrieve local IP - 2");
+ UdpTransport::Destroy(socketPtr);
+ return -1;
+ }
+ // Convert 32-bit address to character string (x.y.z.w)
+ sprintf(localIPAddr, "%d.%d.%d.%d", (int) ((localIP >> 24) & 0x0ff),
+ (int) ((localIP >> 16) & 0x0ff),
+ (int) ((localIP >> 8) & 0x0ff),
+ (int) (localIP & 0x0ff));
+ }
+
+ strcpy(ipAddr, localIPAddr);
+
+ UdpTransport::Destroy(socketPtr);
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "GetLocalIP() => ipAddr=%s", ipAddr);
+ return 0;
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "GetLocalIP() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::EnableIPv6(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "EnableIPv6(channel=%d)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "EnableIPv6() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "EnableIPv6() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->EnableIPv6();
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "EnableIPv6() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+bool VoENetworkImpl::IPv6IsEnabled(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "IPv6IsEnabled(channel=%d)", channel);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return false;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "IPv6IsEnabled() failed to locate channel");
+ return false;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "IPv6IsEnabled() external transport is enabled");
+ return false;
+ }
+ return channelPtr->IPv6IsEnabled();
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "IPv6IsEnabled() VoE is built for external transport");
+ return false;
+#endif
+}
+
+int VoENetworkImpl::SetSourceFilter(int channel,
+ int rtpPort,
+ int rtcpPort,
+ const char ipAddr[64])
+{
+ (ipAddr == NULL) ? WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetSourceFilter(channel=%d, rtpPort=%d,"
+ " rtcpPort=%d)",
+ channel, rtpPort, rtcpPort)
+ : WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "SetSourceFilter(channel=%d, rtpPort=%d,"
+ " rtcpPort=%d, ipAddr=%s)",
+ channel, rtpPort, rtcpPort, ipAddr);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if ((rtpPort < 0) || (rtpPort > 65535))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_PORT_NMBR, kTraceError,
+ "SetSourceFilter() invalid RTP port");
+ return -1;
+ }
+ if ((rtcpPort < 0) || (rtcpPort > 65535))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_PORT_NMBR, kTraceError,
+ "SetSourceFilter() invalid RTCP port");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSourceFilter() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetSourceFilter() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->SetSourceFilter(rtpPort, rtcpPort, ipAddr);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "SetSourceFilter() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::GetSourceFilter(int channel,
+ int& rtpPort,
+ int& rtcpPort,
+ char ipAddr[64])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSourceFilter(channel=%d, rtpPort=?, rtcpPort=?, "
+ "ipAddr[]=?)",
+ channel);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (NULL == ipAddr)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "GetSourceFilter() invalid IP-address buffer");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSourceFilter() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "GetSourceFilter() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->GetSourceFilter(rtpPort, rtcpPort, ipAddr);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "GetSourceFilter() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::SetSendTOS(int channel,
+ int DSCP,
+ int priority,
+ bool useSetSockopt)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetSendTOS(channel=%d, DSCP=%d, useSetSockopt=%d)",
+ channel, DSCP, useSetSockopt);
+
+#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceWarning,
+ "SetSendTOS() is not supported on this platform");
+ return -1;
+#endif
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if ((DSCP < 0) || (DSCP > 63))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendTOS() Invalid DSCP value");
+ return -1;
+ }
+#if defined(_WIN32) || defined(WEBRTC_LINUX)
+ if ((priority < -1) || (priority > 7))
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendTOS() Invalid priority value");
+ return -1;
+ }
+#else
+ if (-1 != priority)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendTOS() priority not supported");
+ return -1;
+ }
+#endif
+#if defined(_WIN32)
+ if ((priority >= 0) && useSetSockopt)
+ {
+ // On Windows, priority and useSetSockopt cannot be combined
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetSendTOS() priority and useSetSockopt conflict");
+ return -1;
+ }
+#endif
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSendTOS() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetSendTOS() external transport is enabled");
+ return -1;
+ }
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+ useSetSockopt = true;
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ " force useSetSockopt=true since there is no alternative"
+ " implementation");
+#endif
+
+ return channelPtr->SetSendTOS(DSCP, priority, useSetSockopt);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "SetSendTOS() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::GetSendTOS(int channel,
+ int& DSCP,
+ int& priority,
+ bool& useSetSockopt)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSendTOS(channel=%d)", channel);
+
+#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceWarning,
+ "GetSendTOS() is not supported on this platform");
+ return -1;
+#endif
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendTOS() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "GetSendTOS() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->GetSendTOS(DSCP, priority, useSetSockopt);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "GetSendTOS() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::SetSendGQoS(int channel,
+ bool enable,
+ int serviceType,
+ int overrideDSCP)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetSendGQOS(channel=%d, enable=%d, serviceType=%d,"
+ " overrideDSCP=%d)",
+ channel, (int) enable, serviceType, overrideDSCP);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#if !defined(_WIN32)
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceWarning,
+ "SetSendGQOS() is not supported on this platform");
+ return -1;
+#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetSendGQOS() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "SetSendGQOS() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->SetSendGQoS(enable, serviceType, overrideDSCP);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "SetSendGQOS() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::GetSendGQoS(int channel,
+ bool& enabled,
+ int& serviceType,
+ int& overrideDSCP)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetSendGQOS(channel=%d)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+#if !defined(_WIN32)
+ _engineStatistics.SetLastError(
+ VE_FUNC_NOT_SUPPORTED, kTraceWarning,
+ "GetSendGQOS() is not supported on this platform");
+ return -1;
+#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSendGQOS() failed to locate channel");
+ return -1;
+ }
+ if (channelPtr->ExternalTransport())
+ {
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
+ "GetSendGQOS() external transport is enabled");
+ return -1;
+ }
+ return channelPtr->GetSendGQoS(enabled, serviceType, overrideDSCP);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "GetSendGQOS() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+int VoENetworkImpl::SetPacketTimeoutNotification(int channel,
+ bool enable,
+ int timeoutSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetPacketTimeoutNotification(channel=%d, enable=%d, "
+ "timeoutSeconds=%d)",
+ channel, (int) enable, timeoutSeconds);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (enable &&
+ ((timeoutSeconds < kVoiceEngineMinPacketTimeoutSec) ||
+ (timeoutSeconds > kVoiceEngineMaxPacketTimeoutSec)))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetPacketTimeoutNotification() invalid timeout size");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetPacketTimeoutNotification() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetPacketTimeoutNotification(enable, timeoutSeconds);
+}
+
+int VoENetworkImpl::GetPacketTimeoutNotification(int channel,
+ bool& enabled,
+ int& timeoutSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetPacketTimeoutNotification(channel=%d, enabled=?,"
+ " timeoutSeconds=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetPacketTimeoutNotification() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetPacketTimeoutNotification(enabled, timeoutSeconds);
+}
+
+int VoENetworkImpl::RegisterDeadOrAliveObserver(int channel,
+ VoEConnectionObserver&
+ observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "RegisterDeadOrAliveObserver(channel=%d, observer=0x%x)",
+ channel, &observer);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterDeadOrAliveObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterDeadOrAliveObserver(observer);
+}
+
+int VoENetworkImpl::DeRegisterDeadOrAliveObserver(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "DeRegisterDeadOrAliveObserver(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterDeadOrAliveObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterDeadOrAliveObserver();
+}
+
+int VoENetworkImpl::SetPeriodicDeadOrAliveStatus(int channel, bool enable,
+ int sampleTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SetPeriodicDeadOrAliveStatus(channel=%d, enable=%d,"
+ " sampleTimeSeconds=%d)",
+ channel, enable, sampleTimeSeconds);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (enable &&
+ ((sampleTimeSeconds < kVoiceEngineMinSampleTimeSec) ||
+ (sampleTimeSeconds > kVoiceEngineMaxSampleTimeSec)))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetPeriodicDeadOrAliveStatus() invalid sample time");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetPeriodicDeadOrAliveStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetPeriodicDeadOrAliveStatus(enable, sampleTimeSeconds);
+}
+
+int VoENetworkImpl::GetPeriodicDeadOrAliveStatus(int channel,
+ bool& enabled,
+ int& sampleTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "GetPeriodicDeadOrAliveStatus(channel=%d, enabled=?,"
+ " sampleTimeSeconds=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetPeriodicDeadOrAliveStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetPeriodicDeadOrAliveStatus(enabled,
+ sampleTimeSeconds);
+}
+
+int VoENetworkImpl::SendUDPPacket(int channel,
+ const void* data,
+ unsigned int length,
+ int& transmittedBytes,
+ bool useRtcpSocket)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
+ "SendUDPPacket(channel=%d, data=0x%x, length=%u, useRTCP=%d)",
+ channel, data, length, useRtcpSocket);
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (NULL == data)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SendUDPPacket() invalid data buffer");
+ return -1;
+ }
+ if (0 == length)
+ {
+ _engineStatistics.SetLastError(VE_INVALID_PACKET, kTraceError,
+ "SendUDPPacket() invalid packet size");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SendUDPPacket() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SendUDPPacket(data,
+ length,
+ transmittedBytes,
+ useRtcpSocket);
+#else
+ _engineStatistics.SetLastError(
+ VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
+ "SendUDPPacket() VoE is built for external transport");
+ return -1;
+#endif
+}
+
+#endif // WEBRTC_VOICE_ENGINE_NETWORK_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_network_impl.h b/trunk/src/voice_engine/main/source/voe_network_impl.h
new file mode 100644
index 0000000..cc300ac
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_network_impl.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
+
+#include "voe_network.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+
+namespace webrtc
+{
+
+class VoENetworkImpl: public virtual voe::SharedData,
+ public VoENetwork,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int RegisterExternalTransport(int channel, Transport& transport);
+
+ virtual int DeRegisterExternalTransport(int channel);
+
+ virtual int ReceivedRTPPacket(int channel,
+ const void* data,
+ unsigned int length);
+
+ virtual int ReceivedRTCPPacket(int channel,
+ const void* data,
+ unsigned int length);
+
+ virtual int GetSourceInfo(int channel,
+ int& rtpPort,
+ int& rtcpPort,
+ char ipAddr[64]);
+
+ virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false);
+
+ virtual int EnableIPv6(int channel);
+
+ virtual bool IPv6IsEnabled(int channel);
+
+ virtual int SetSourceFilter(int channel,
+ int rtpPort,
+ int rtcpPort,
+ const char ipAddr[64] = 0);
+
+ virtual int GetSourceFilter(int channel,
+ int& rtpPort,
+ int& rtcpPort,
+ char ipAddr[64]);
+
+ virtual int SetSendTOS(int channel,
+ int DSCP,
+ int priority = -1,
+ bool useSetSockopt = false);
+
+ virtual int GetSendTOS(int channel,
+ int& DSCP,
+ int& priority,
+ bool& useSetSockopt);
+
+ virtual int SetSendGQoS(int channel,
+ bool enable,
+ int serviceType,
+ int overrideDSCP);
+
+ virtual int GetSendGQoS(int channel,
+ bool& enabled,
+ int& serviceType,
+ int& overrideDSCP);
+
+ virtual int SetPacketTimeoutNotification(int channel,
+ bool enable,
+ int timeoutSeconds = 2);
+
+ virtual int GetPacketTimeoutNotification(int channel,
+ bool& enabled,
+ int& timeoutSeconds);
+
+ virtual int RegisterDeadOrAliveObserver(int channel,
+ VoEConnectionObserver& observer);
+
+ virtual int DeRegisterDeadOrAliveObserver(int channel);
+
+ virtual int SetPeriodicDeadOrAliveStatus(int channel,
+ bool enable,
+ int sampleTimeSeconds = 2);
+
+ virtual int GetPeriodicDeadOrAliveStatus(int channel,
+ bool& enabled,
+ int& sampleTimeSeconds);
+
+ virtual int SendUDPPacket(int channel,
+ const void* data,
+ unsigned int length,
+ int& transmittedBytes,
+ bool useRtcpSocket = false);
+
+protected:
+ VoENetworkImpl();
+ virtual ~VoENetworkImpl();
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
new file mode 100644
index 0000000..cbc4d0d
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
@@ -0,0 +1,730 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_rtp_rtcp_impl.h"
+#include "trace.h"
+#include "file_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "voice_engine_impl.h"
+#include "voe_errors.h"
+
+#include "channel.h"
+#include "transmit_mixer.h"
+
+namespace webrtc {
+
+VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoERTP_RTCPImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+
+VoERTP_RTCPImpl::VoERTP_RTCPImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
+}
+
+VoERTP_RTCPImpl::~VoERTP_RTCPImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
+}
+
+int VoERTP_RTCPImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoERTP_RTCP::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _engineStatistics.SetLastError(
+ VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoERTP_RTCP reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoERTP_RTCPImpl::RegisterRTPObserver(int channel, VoERTPObserver& observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterRTPObserver(channel=%d observer=0x%x)",
+ channel, &observer);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterRTPObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterRTPObserver(observer);
+}
+
+int VoERTP_RTCPImpl::DeRegisterRTPObserver(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterRTPObserver(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterRTPObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterRTPObserver();
+}
+
+int VoERTP_RTCPImpl::RegisterRTCPObserver(int channel, VoERTCPObserver& observer)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RegisterRTCPObserver(channel=%d observer=0x%x)",
+ channel, &observer);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "RegisterRTPObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RegisterRTCPObserver(observer);
+}
+
+int VoERTP_RTCPImpl::DeRegisterRTCPObserver(int channel)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "DeRegisterRTCPObserver(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "DeRegisterRTCPObserver() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->DeRegisterRTCPObserver();
+}
+
+int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetLocalSSRC() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetLocalSSRC(ssrc);
+}
+
+int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetLocalSSRC(channel=%d, ssrc=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetLocalSSRC() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetLocalSSRC(ssrc);
+}
+
+int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRemoteSSRC(channel=%d, ssrc=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRemoteSSRC() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRemoteSSRC(ssrc);
+}
+
+int VoERTP_RTCPImpl::GetRemoteCSRCs(int channel, unsigned int arrCSRC[15])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRemoteCSRCs(channel=%d, arrCSRC=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRemoteCSRCs() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRemoteCSRCs(arrCSRC);
+}
+
+
+int VoERTP_RTCPImpl::SetRTPAudioLevelIndicationStatus(int channel,
+ bool enable,
+ unsigned char ID)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRTPAudioLevelIndicationStatus(channel=%d, enable=%d,"
+ " ID=%u)", channel, enable, ID);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (ID < kVoiceEngineMinRtpExtensionId ||
+ ID > kVoiceEngineMaxRtpExtensionId)
+ {
+ // [RFC5285] The 4-bit ID is the local identifier of this element in
+ // the range 1-14 inclusive.
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetRTPAudioLevelIndicationStatus() invalid ID parameter");
+ return -1;
+ }
+
+ // Set state and ID for the specified channel.
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRTPAudioLevelIndicationStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRTPAudioLevelIndicationStatus(enable, ID);
+}
+
+int VoERTP_RTCPImpl::GetRTPAudioLevelIndicationStatus(int channel,
+ bool& enabled,
+ unsigned char& ID)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTPAudioLevelIndicationStatus(channel=%d, enable=?, ID=?)",
+ channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTPAudioLevelIndicationStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTPAudioLevelIndicationStatus(enabled, ID);
+}
+
+int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRTCPStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRTCPStatus(enable);
+}
+
+int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTCPStatus(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTCPStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTCPStatus(enabled);
+}
+
+int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRTCP_CNAME() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRTCP_CNAME(cName);
+}
+
+int VoERTP_RTCPImpl::GetRTCP_CNAME(int channel, char cName[256])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTCP_CNAME(channel=%d, cName=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTCP_CNAME() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTCP_CNAME(cName);
+}
+
+int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256])
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRemoteRTCP_CNAME(channel=%d, cName=?)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRemoteRTCP_CNAME() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRemoteRTCP_CNAME(cName);
+}
+
+int VoERTP_RTCPImpl::GetRemoteRTCPData(
+ int channel,
+ unsigned int& NTPHigh, // from sender info in SR
+ unsigned int& NTPLow, // from sender info in SR
+ unsigned int& timestamp, // from sender info in SR
+ unsigned int& playoutTimestamp, // derived locally
+ unsigned int* jitter, // from report block 1 in SR/RR
+ unsigned short* fractionLost) // from report block 1 in SR/RR
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRemoteRTCPData(channel=%d,...)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRemoteRTCP_CNAME() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRemoteRTCPData(NTPHigh,
+ NTPLow,
+ timestamp,
+ playoutTimestamp,
+ jitter,
+ fractionLost);
+}
+
+int VoERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
+ int channel,
+ const unsigned char subType,
+ unsigned int name,
+ const char* data,
+ unsigned short dataLengthInBytes)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SendApplicationDefinedRTCPPacket(channel=%d, subType=%u,"
+ "name=%u, data=?, dataLengthInBytes=%u)",
+ channel, subType, name, dataLengthInBytes);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SendApplicationDefinedRTCPPacket() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SendApplicationDefinedRTCPPacket(subType,
+ name,
+ data,
+ dataLengthInBytes);
+}
+
+int VoERTP_RTCPImpl::GetRTPStatistics(int channel,
+ unsigned int& averageJitterMs,
+ unsigned int& maxJitterMs,
+ unsigned int& discardedPackets)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTPStatistics(channel=%d,....)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTPStatistics() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTPStatistics(averageJitterMs,
+ maxJitterMs,
+ discardedPackets);
+}
+
+int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTCPStatistics(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTPStatistics() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTPStatistics(stats);
+}
+
+int VoERTP_RTCPImpl::SetFECStatus(int channel, bool enable, int redPayloadtype)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetFECStatus(channel=%d, enable=%d, redPayloadtype=%d)",
+ channel, enable, redPayloadtype);
+#ifdef WEBRTC_CODEC_RED
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetFECStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetFECStatus(enable, redPayloadtype);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "SetFECStatus() RED is not supported");
+ return -1;
+#endif
+}
+
+int VoERTP_RTCPImpl::GetFECStatus(int channel,
+ bool& enabled,
+ int& redPayloadtype)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetFECStatus(channel=%d, enabled=?, redPayloadtype=?)",
+ channel);
+#ifdef WEBRTC_CODEC_RED
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetFECStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetFECStatus(enabled, redPayloadtype);
+#else
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+ "GetFECStatus() RED is not supported");
+ return -1;
+#endif
+}
+
+int VoERTP_RTCPImpl::SetRTPKeepaliveStatus(int channel,
+ bool enable,
+ unsigned char unknownPayloadType,
+ int deltaTransmitTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetRTPKeepaliveStatus(channel=%d, enable=%d,"
+ " unknownPayloadType=%u, deltaTransmitTimeSeconds=%d)",
+ channel, enable, unknownPayloadType, deltaTransmitTimeSeconds);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetRTPKeepaliveStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetRTPKeepaliveStatus(enable,
+ unknownPayloadType,
+ deltaTransmitTimeSeconds);
+}
+
+int VoERTP_RTCPImpl::GetRTPKeepaliveStatus(int channel,
+ bool& enabled,
+ unsigned char& unknownPayloadType,
+ int& deltaTransmitTimeSeconds)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetRTPKeepaliveStatus(channel=%d)", channel);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetRTPKeepaliveStatus() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRTPKeepaliveStatus(enabled,
+ unknownPayloadType,
+ deltaTransmitTimeSeconds);
+}
+
+int VoERTP_RTCPImpl::StartRTPDump(int channel,
+ const char fileNameUTF8[1024],
+ RTPDirections direction)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StartRTPDump(channel=%d, fileNameUTF8=%s, direction=%d)",
+ channel, fileNameUTF8, direction);
+ assert(1024 == FileWrapper::kMaxFileNameSize);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StartRTPDump() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StartRTPDump(fileNameUTF8, direction);
+}
+
+int VoERTP_RTCPImpl::StopRTPDump(int channel, RTPDirections direction)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "StopRTPDump(channel=%d, direction=%d)", channel, direction);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopRTPDump() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->StopRTPDump(direction);
+}
+
+int VoERTP_RTCPImpl::RTPDumpIsActive(int channel, RTPDirections direction)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "RTPDumpIsActive(channel=%d, direction=%d)",
+ channel, direction);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopRTPDump() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->RTPDumpIsActive(direction);
+}
+
+int VoERTP_RTCPImpl::InsertExtraRTPPacket(int channel,
+ unsigned char payloadType,
+ bool markerBit,
+ const char* payloadData,
+ unsigned short payloadSize)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "InsertExtraRTPPacket(channel=%d, payloadType=%u,"
+ " markerBit=%u, payloadSize=%u)",
+ channel, payloadType, markerBit, payloadSize);
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "StopRTPDump() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->InsertExtraRTPPacket(payloadType,
+ markerBit,
+ payloadData,
+ payloadSize);
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
new file mode 100644
index 0000000..3cdf162
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
+
+#include "voe_rtp_rtcp.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoERTP_RTCPImpl : public virtual voe::SharedData,
+ public VoERTP_RTCP,
+ public voe::RefCount
+{
+public:
+
+ virtual int Release();
+ // Registration of observers for RTP and RTCP callbacks
+ virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer);
+
+ virtual int DeRegisterRTPObserver(int channel);
+
+ virtual int RegisterRTCPObserver(int channel, VoERTCPObserver& observer);
+
+ virtual int DeRegisterRTCPObserver(int channel);
+
+ // RTCP
+ virtual int SetRTCPStatus(int channel, bool enable);
+
+ virtual int GetRTCPStatus(int channel, bool& enabled);
+
+ virtual int SetRTCP_CNAME(int channel, const char cName[256]);
+
+ virtual int GetRTCP_CNAME(int channel, char cName[256]);
+
+ virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]);
+
+ virtual int GetRemoteRTCPData(int channel,
+ unsigned int& NTPHigh,
+ unsigned int& NTPLow,
+ unsigned int& timestamp,
+ unsigned int& playoutTimestamp,
+ unsigned int* jitter = NULL,
+ unsigned short* fractionLost = NULL);
+
+ virtual int SendApplicationDefinedRTCPPacket(
+ int channel,
+ const unsigned char subType,
+ unsigned int name,
+ const char* data,
+ unsigned short dataLengthInBytes);
+
+ // SSRC
+ virtual int SetLocalSSRC(int channel, unsigned int ssrc);
+
+ virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
+
+ virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
+
+ // RTP Header Extension for Client-to-Mixer Audio Level Indication
+ virtual int SetRTPAudioLevelIndicationStatus(int channel,
+ bool enable,
+ unsigned char ID);
+
+ virtual int GetRTPAudioLevelIndicationStatus(int channel,
+ bool& enabled,
+ unsigned char& ID);
+
+ // CSRC
+ virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]);
+
+ // Statistics
+ virtual int GetRTPStatistics(int channel,
+ unsigned int& averageJitterMs,
+ unsigned int& maxJitterMs,
+ unsigned int& discardedPackets);
+
+ virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
+
+ // RTP keepalive mechanism (maintains NAT mappings associated to RTP flows)
+ virtual int SetRTPKeepaliveStatus(int channel,
+ bool enable,
+ unsigned char unknownPayloadType,
+ int deltaTransmitTimeSeconds = 15);
+
+ virtual int GetRTPKeepaliveStatus(int channel,
+ bool& enabled,
+ unsigned char& unknownPayloadType,
+ int& deltaTransmitTimeSeconds);
+
+ // FEC
+ virtual int SetFECStatus(int channel,
+ bool enable,
+ int redPayloadtype = -1);
+
+ virtual int GetFECStatus(int channel, bool& enabled, int& redPayloadtype);
+
+ // Store RTP and RTCP packets and dump to file (compatible with rtpplay)
+ virtual int StartRTPDump(int channel,
+ const char fileNameUTF8[1024],
+ RTPDirections direction = kRtpIncoming);
+
+ virtual int StopRTPDump(int channel,
+ RTPDirections direction = kRtpIncoming);
+
+ virtual int RTPDumpIsActive(int channel,
+ RTPDirections direction = kRtpIncoming);
+
+ // Insert (and transmits) extra RTP packet into active RTP audio stream
+ virtual int InsertExtraRTPPacket(int channel,
+ unsigned char payloadType,
+ bool markerBit,
+ const char* payloadData,
+ unsigned short payloadSize);
+
+protected:
+ VoERTP_RTCPImpl();
+ virtual ~VoERTP_RTCPImpl();
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
+
diff --git a/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc b/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc
new file mode 100644
index 0000000..8ee6296
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc
@@ -0,0 +1,248 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_video_sync_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "trace.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEVideoSync* VoEVideoSync::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEVideoSyncImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+
+VoEVideoSyncImpl::VoEVideoSyncImpl(voe::SharedData* data) : _data(data)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "VoEVideoSyncImpl::VoEVideoSyncImpl() - ctor");
+}
+
+VoEVideoSyncImpl::~VoEVideoSyncImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "VoEVideoSyncImpl::~VoEVideoSyncImpl() - dtor");
+}
+
+int VoEVideoSyncImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "VoEVideoSync::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _data->statistics().SetLastError(VE_INTERFACE_NOT_FOUND,
+ kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "VoEVideoSync reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel, unsigned int& timestamp)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "GetPlayoutTimestamp(channel=%d, timestamp=?)", channel);
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetPlayoutTimestamp() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetPlayoutTimestamp(timestamp);
+}
+
+int VoEVideoSyncImpl::SetInitTimestamp(int channel,
+ unsigned int timestamp)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "SetInitTimestamp(channel=%d, timestamp=%lu)",
+ channel, timestamp);
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetInitTimestamp() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetInitTimestamp(timestamp);
+}
+
+int VoEVideoSyncImpl::SetInitSequenceNumber(int channel,
+ short sequenceNumber)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)",
+ channel, sequenceNumber);
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetInitSequenceNumber() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetInitSequenceNumber(sequenceNumber);
+}
+
+int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel,int delayMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)",
+ channel, delayMs);
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetMinimumPlayoutDelay() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetMinimumPlayoutDelay(delayMs);
+}
+
+int VoEVideoSyncImpl::GetDelayEstimate(int channel, int& delayMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "GetDelayEstimate(channel=%d, delayMs=?)", channel);
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetDelayEstimate() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetDelayEstimate(delayMs);
+}
+
+int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "GetPlayoutBufferSize(bufferMs=?)");
+ ANDROID_NOT_SUPPORTED(_data->statistics());
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ AudioDeviceModule::BufferType type
+ (AudioDeviceModule::kFixedBufferSize);
+ WebRtc_UWord16 sizeMS(0);
+ if (_data->audio_device()->PlayoutBuffer(&type, &sizeMS) != 0)
+ {
+ _data->statistics().SetLastError(
+ VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+ "GetPlayoutBufferSize() failed to read buffer size");
+ return -1;
+ }
+ bufferMs = sizeMS;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "GetPlayoutBufferSize() => bufferMs=%d", bufferMs);
+ return 0;
+}
+
+int VoEVideoSyncImpl::GetRtpRtcp(int channel, RtpRtcp* &rtpRtcpModule)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
+ "GetRtpRtcp(channel=%i)", channel);
+
+ if (!_data->statistics().Initialized())
+ {
+ _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_data->channel_manager(), channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _data->statistics().SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetPlayoutTimestamp() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetRtpRtcp(rtpRtcpModule);
+}
+
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_video_sync_impl.h b/trunk/src/voice_engine/main/source/voe_video_sync_impl.h
new file mode 100644
index 0000000..02c85d1
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_video_sync_impl.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
+
+#include "voe_video_sync.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoEVideoSyncImpl : public VoEVideoSync,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int GetPlayoutBufferSize(int& bufferMs);
+
+ virtual int SetMinimumPlayoutDelay(int channel, int delayMs);
+
+ virtual int GetDelayEstimate(int channel, int& delayMs);
+
+ virtual int SetInitTimestamp(int channel, unsigned int timestamp);
+
+ virtual int SetInitSequenceNumber(int channel, short sequenceNumber);
+
+ virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp);
+
+ virtual int GetRtpRtcp(int channel, RtpRtcp* &rtpRtcpModule);
+
+protected:
+ VoEVideoSyncImpl(voe::SharedData* data);
+ virtual ~VoEVideoSyncImpl();
+
+private:
+ voe::SharedData* _data;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc b/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc
new file mode 100644
index 0000000..dd3d54d
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc
@@ -0,0 +1,677 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voe_volume_control_impl.h"
+
+#include "channel.h"
+#include "critical_section_wrapper.h"
+#include "output_mixer.h"
+#include "trace.h"
+#include "transmit_mixer.h"
+#include "voe_errors.h"
+#include "voice_engine_impl.h"
+
+namespace webrtc {
+
+VoEVolumeControl* VoEVolumeControl::GetInterface(VoiceEngine* voiceEngine)
+{
+#ifndef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+ return NULL;
+#else
+ if (NULL == voiceEngine)
+ {
+ return NULL;
+ }
+ VoiceEngineImpl* s =
+ reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEVolumeControlImpl* d = s;
+ (*d)++;
+ return (d);
+#endif
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+
+VoEVolumeControlImpl::VoEVolumeControlImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEVolumeControlImpl::VoEVolumeControlImpl() - ctor");
+}
+
+VoEVolumeControlImpl::~VoEVolumeControlImpl()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEVolumeControlImpl::~VoEVolumeControlImpl() - dtor");
+}
+
+int VoEVolumeControlImpl::Release()
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEVolumeControl::Release()");
+ (*this)--;
+ int refCount = GetCount();
+ if (refCount < 0)
+ {
+ Reset(); // reset reference counter to zero => OK to delete VE
+ _engineStatistics.SetLastError(
+ VE_INTERFACE_NOT_FOUND, kTraceWarning);
+ return (-1);
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "VoEVolumeControl reference counter = %d", refCount);
+ return (refCount);
+}
+
+int VoEVolumeControlImpl::SetSpeakerVolume(unsigned int volume)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetSpeakerVolume(volume=%u)", volume);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (volume > kMaxVolumeLevel)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetSpeakerVolume() invalid argument");
+ return -1;
+ }
+
+ WebRtc_UWord32 maxVol(0);
+ WebRtc_UWord32 spkrVol(0);
+
+ // scale: [0,kMaxVolumeLevel] -> [0,MaxSpeakerVolume]
+ if (_audioDevicePtr->MaxSpeakerVolume(&maxVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_MIC_VOL_ERROR, kTraceError,
+ "SetSpeakerVolume() failed to get max volume");
+ return -1;
+ }
+ // Round the value and avoid floating computation.
+ spkrVol = (WebRtc_UWord32)((volume * maxVol +
+ (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
+
+ // set the actual volume using the audio mixer
+ if (_audioDevicePtr->SetSpeakerVolume(spkrVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_MIC_VOL_ERROR, kTraceError,
+ "SetSpeakerVolume() failed to set speaker volume");
+ return -1;
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSpeakerVolume(unsigned int& volume)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeakerVolume()");
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ WebRtc_UWord32 spkrVol(0);
+ WebRtc_UWord32 maxVol(0);
+
+ if (_audioDevicePtr->SpeakerVolume(&spkrVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "GetSpeakerVolume() unable to get speaker volume");
+ return -1;
+ }
+
+ // scale: [0, MaxSpeakerVolume] -> [0, kMaxVolumeLevel]
+ if (_audioDevicePtr->MaxSpeakerVolume(&maxVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "GetSpeakerVolume() unable to get max speaker volume");
+ return -1;
+ }
+ // Round the value and avoid floating computation.
+ volume = (WebRtc_UWord32) ((spkrVol * kMaxVolumeLevel +
+ (int)(maxVol / 2)) / (maxVol));
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeakerVolume() => volume=%d", volume);
+ return 0;
+}
+
+int VoEVolumeControlImpl::SetSystemOutputMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSystemOutputMute(enabled=%d)", enable);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioDevicePtr->SetSpeakerMute(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "SpeakerMute() unable to Set speaker mute");
+ return -1;
+ }
+
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSystemOutputMute(bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSystemOutputMute(enabled=?)");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioDevicePtr->SpeakerMute(&enabled) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "SpeakerMute() unable to get speaker mute state");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSystemOutputMute() => %d", enabled);
+ return 0;
+}
+
+int VoEVolumeControlImpl::SetMicVolume(unsigned int volume)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetMicVolume(volume=%u)", volume);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (volume > kMaxVolumeLevel)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetMicVolume() invalid argument");
+ return -1;
+ }
+
+ WebRtc_UWord32 maxVol(0);
+ WebRtc_UWord32 micVol(0);
+
+ // scale: [0, kMaxVolumeLevel] -> [0,MaxMicrophoneVolume]
+ if (_audioDevicePtr->MaxMicrophoneVolume(&maxVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_MIC_VOL_ERROR, kTraceError,
+ "SetMicVolume() failed to get max volume");
+ return -1;
+ }
+
+ if (volume == kMaxVolumeLevel) {
+ // On Linux running pulse, users are able to set the volume above 100%
+ // through the volume control panel, where the +100% range is digital
+ // scaling. WebRTC does not support setting the volume above 100%, and
+ // simply ignores changing the volume if the user tries to set it to
+ // |kMaxVolumeLevel| while the current volume is higher than |maxVol|.
+ if (_audioDevicePtr->MicrophoneVolume(&micVol) != 0) {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "SetMicVolume() unable to get microphone volume");
+ return -1;
+ }
+ if (micVol >= maxVol)
+ return 0;
+ }
+
+ // Round the value and avoid floating point computation.
+ micVol = (WebRtc_UWord32) ((volume * maxVol +
+ (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
+
+ // set the actual volume using the audio mixer
+ if (_audioDevicePtr->SetMicrophoneVolume(micVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_MIC_VOL_ERROR, kTraceError,
+ "SetMicVolume() failed to set mic volume");
+ return -1;
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetMicVolume(unsigned int& volume)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetMicVolume()");
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ WebRtc_UWord32 micVol(0);
+ WebRtc_UWord32 maxVol(0);
+
+ if (_audioDevicePtr->MicrophoneVolume(&micVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "GetMicVolume() unable to get microphone volume");
+ return -1;
+ }
+
+ // scale: [0, MaxMicrophoneVolume] -> [0, kMaxVolumeLevel]
+ if (_audioDevicePtr->MaxMicrophoneVolume(&maxVol) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "GetMicVolume() unable to get max microphone volume");
+ return -1;
+ }
+ if (micVol < maxVol) {
+ // Round the value and avoid floating point calculation.
+ volume = (WebRtc_UWord32) ((micVol * kMaxVolumeLevel +
+ (int)(maxVol / 2)) / (maxVol));
+ } else {
+ // Truncate the value to the kMaxVolumeLevel.
+ volume = kMaxVolumeLevel;
+ }
+
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetMicVolume() => volume=%d", volume);
+ return 0;
+}
+
+int VoEVolumeControlImpl::SetInputMute(int channel, bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetInputMute(channel=%d, enable=%d)", channel, enable);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ // Mute before demultiplexing <=> affects all channels
+ return _transmitMixerPtr->SetMute(enable);
+ }
+ else
+ {
+ // Mute after demultiplexing <=> affects one channel only
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetInputMute() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetMute(enable);
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetInputMute(int channel, bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetInputMute(channel=%d)", channel);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ enabled = _transmitMixerPtr->Mute();
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetInputMute() failed to locate channel");
+ return -1;
+ }
+ enabled = channelPtr->Mute();
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetInputMute() => enabled = %d", (int)enabled);
+ return 0;
+}
+
+int VoEVolumeControlImpl::SetSystemInputMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetSystemInputMute(enabled=%d)", enable);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioDevicePtr->SetMicrophoneMute(enable) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "MicrophoneMute() unable to set microphone mute state");
+ return -1;
+ }
+
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSystemInputMute(bool& enabled)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSystemInputMute(enabled=?)");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ if (_audioDevicePtr->MicrophoneMute(&enabled) != 0)
+ {
+ _engineStatistics.SetLastError(
+ VE_GET_MIC_VOL_ERROR, kTraceError,
+ "MicrophoneMute() unable to get microphone mute state");
+ return -1;
+ }
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSystemInputMute() => %d", enabled);
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSpeechInputLevel(unsigned int& level)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechInputLevel()");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ WebRtc_Word8 currentLevel = _transmitMixerPtr->AudioLevel();
+ level = static_cast<unsigned int> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechInputLevel() => %d", level);
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSpeechOutputLevel(int channel,
+ unsigned int& level)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechOutputLevel(channel=%d, level=?)", channel);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _outputMixerPtr->GetSpeechOutputLevel((WebRtc_UWord32&)level);
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSpeechOutputLevel() failed to locate channel");
+ return -1;
+ }
+ channelPtr->GetSpeechOutputLevel((WebRtc_UWord32&)level);
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSpeechInputLevelFullRange(unsigned int& level)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechInputLevelFullRange(level=?)");
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ WebRtc_Word16 currentLevel = _transmitMixerPtr->AudioLevelFullRange();
+ level = static_cast<unsigned int> (currentLevel);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechInputLevelFullRange() => %d", level);
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetSpeechOutputLevelFullRange(int channel,
+ unsigned int& level)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetSpeechOutputLevelFullRange(channel=%d, level=?)", channel);
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (channel == -1)
+ {
+ return _outputMixerPtr->GetSpeechOutputLevelFullRange(
+ (WebRtc_UWord32&)level);
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetSpeechOutputLevelFullRange() failed to locate channel");
+ return -1;
+ }
+ channelPtr->GetSpeechOutputLevelFullRange((WebRtc_UWord32&)level);
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::SetChannelOutputVolumeScaling(int channel,
+ float scaling)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetChannelOutputVolumeScaling(channel=%d, scaling=%3.2f)",
+ channel, scaling);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (scaling < kMinOutputVolumeScaling ||
+ scaling > kMaxOutputVolumeScaling)
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetChannelOutputVolumeScaling() invalid parameter");
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetChannelOutputVolumeScaling() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetChannelOutputVolumeScaling(scaling);
+}
+
+int VoEVolumeControlImpl::GetChannelOutputVolumeScaling(int channel,
+ float& scaling)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetChannelOutputVolumeScaling(channel=%d, scaling=?)", channel);
+ IPHONE_NOT_SUPPORTED();
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetChannelOutputVolumeScaling() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetChannelOutputVolumeScaling(scaling);
+}
+
+int VoEVolumeControlImpl::SetOutputVolumePan(int channel,
+ float left,
+ float right)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)",
+ channel, left, right);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool available(false);
+ _audioDevicePtr->StereoPlayoutIsAvailable(&available);
+ if (!available)
+ {
+ _engineStatistics.SetLastError(
+ VE_FUNC_NO_STEREO, kTraceError,
+ "SetOutputVolumePan() stereo playout not supported");
+ return -1;
+ }
+ if ((left < kMinOutputVolumePanning) ||
+ (left > kMaxOutputVolumePanning) ||
+ (right < kMinOutputVolumePanning) ||
+ (right > kMaxOutputVolumePanning))
+ {
+ _engineStatistics.SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "SetOutputVolumePan() invalid parameter");
+ return -1;
+ }
+
+ if (channel == -1)
+ {
+ // Master balance (affectes the signal after output mixing)
+ return _outputMixerPtr->SetOutputVolumePan(left, right);
+ }
+ else
+ {
+ // Per-channel balance (affects the signal before output mixing)
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetOutputVolumePan() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->SetOutputVolumePan(left, right);
+ }
+ return 0;
+}
+
+int VoEVolumeControlImpl::GetOutputVolumePan(int channel,
+ float& left,
+ float& right)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
+ "GetOutputVolumePan(channel=%d, left=?, right=?)", channel);
+ ANDROID_NOT_SUPPORTED(_engineStatistics);
+ IPHONE_NOT_SUPPORTED();
+
+ if (!_engineStatistics.Initialized())
+ {
+ _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+
+ bool available(false);
+ _audioDevicePtr->StereoPlayoutIsAvailable(&available);
+ if (!available)
+ {
+ _engineStatistics.SetLastError(
+ VE_FUNC_NO_STEREO, kTraceError,
+ "GetOutputVolumePan() stereo playout not supported");
+ return -1;
+ }
+
+ if (channel == -1)
+ {
+ return _outputMixerPtr->GetOutputVolumePan(left, right);
+ }
+ else
+ {
+ voe::ScopedChannel sc(_channelManager, channel);
+ voe::Channel* channelPtr = sc.ChannelPtr();
+ if (channelPtr == NULL)
+ {
+ _engineStatistics.SetLastError(
+ VE_CHANNEL_NOT_VALID, kTraceError,
+ "GetOutputVolumePan() failed to locate channel");
+ return -1;
+ }
+ return channelPtr->GetOutputVolumePan(left, right);
+ }
+ return 0;
+}
+
+#endif // #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+
+} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_volume_control_impl.h b/trunk/src/voice_engine/main/source/voe_volume_control_impl.h
new file mode 100644
index 0000000..18f4a1b
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voe_volume_control_impl.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
+
+#include "voe_volume_control.h"
+
+#include "ref_count.h"
+#include "shared_data.h"
+
+namespace webrtc {
+
+class VoEVolumeControlImpl : public virtual voe::SharedData,
+ public VoEVolumeControl,
+ public voe::RefCount
+{
+public:
+ virtual int Release();
+
+ virtual int SetSpeakerVolume(unsigned int volume);
+
+ virtual int GetSpeakerVolume(unsigned int& volume);
+
+ virtual int SetSystemOutputMute(bool enable);
+
+ virtual int GetSystemOutputMute(bool& enabled);
+
+ virtual int SetMicVolume(unsigned int volume);
+
+ virtual int GetMicVolume(unsigned int& volume);
+
+ virtual int SetInputMute(int channel, bool enable);
+
+ virtual int GetInputMute(int channel, bool& enabled);
+
+ virtual int SetSystemInputMute(bool enable);
+
+ virtual int GetSystemInputMute(bool& enabled);
+
+ virtual int GetSpeechInputLevel(unsigned int& level);
+
+ virtual int GetSpeechOutputLevel(int channel, unsigned int& level);
+
+ virtual int GetSpeechInputLevelFullRange(unsigned int& level);
+
+ virtual int GetSpeechOutputLevelFullRange(int channel,
+ unsigned int& level);
+
+ virtual int SetChannelOutputVolumeScaling(int channel, float scaling);
+
+ virtual int GetChannelOutputVolumeScaling(int channel, float& scaling);
+
+ virtual int SetOutputVolumePan(int channel, float left, float right);
+
+ virtual int GetOutputVolumePan(int channel, float& left, float& right);
+
+
+protected:
+ VoEVolumeControlImpl();
+ virtual ~VoEVolumeControlImpl();
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
+
diff --git a/trunk/src/voice_engine/main/source/voice_engine_core.gypi b/trunk/src/voice_engine/main/source/voice_engine_core.gypi
new file mode 100644
index 0000000..135d78b
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voice_engine_core.gypi
@@ -0,0 +1,158 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'voice_engine_core',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+ '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
+ '<(webrtc_root)/modules/modules.gyp:audio_device',
+ '<(webrtc_root)/modules/modules.gyp:audio_processing',
+ '<(webrtc_root)/modules/modules.gyp:media_file',
+ '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+ '<(webrtc_root)/modules/modules.gyp:udp_transport',
+ '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ '../../..',
+ '../interface',
+ '<(webrtc_root)/modules/audio_device/main/source',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '../../..',
+ '../interface',
+ ],
+ },
+ 'sources': [
+ '../../../common_types.h',
+ '../../../engine_configurations.h',
+ '../../../typedefs.h',
+ '../interface/voe_audio_processing.h',
+ '../interface/voe_base.h',
+ '../interface/voe_call_report.h',
+ '../interface/voe_codec.h',
+ '../interface/voe_dtmf.h',
+ '../interface/voe_encryption.h',
+ '../interface/voe_errors.h',
+ '../interface/voe_external_media.h',
+ '../interface/voe_file.h',
+ '../interface/voe_hardware.h',
+ '../interface/voe_neteq_stats.h',
+ '../interface/voe_network.h',
+ '../interface/voe_rtp_rtcp.h',
+ '../interface/voe_video_sync.h',
+ '../interface/voe_volume_control.h',
+ 'audio_frame_operations.cc',
+ 'audio_frame_operations.h',
+ 'channel.cc',
+ 'channel.h',
+ 'channel_manager.cc',
+ 'channel_manager.h',
+ 'channel_manager_base.cc',
+ 'channel_manager_base.h',
+ 'dtmf_inband.cc',
+ 'dtmf_inband.h',
+ 'dtmf_inband_queue.cc',
+ 'dtmf_inband_queue.h',
+ 'level_indicator.cc',
+ 'level_indicator.h',
+ 'monitor_module.cc',
+ 'monitor_module.h',
+ 'output_mixer.cc',
+ 'output_mixer.h',
+ 'ref_count.cc',
+ 'ref_count.h',
+ 'shared_data.cc',
+ 'shared_data.h',
+ 'statistics.cc',
+ 'statistics.h',
+ 'transmit_mixer.cc',
+ 'transmit_mixer.h',
+ 'utility.cc',
+ 'utility.h',
+ 'voe_audio_processing_impl.cc',
+ 'voe_audio_processing_impl.h',
+ 'voe_base_impl.cc',
+ 'voe_base_impl.h',
+ 'voe_call_report_impl.cc',
+ 'voe_call_report_impl.h',
+ 'voe_codec_impl.cc',
+ 'voe_codec_impl.h',
+ 'voe_dtmf_impl.cc',
+ 'voe_dtmf_impl.h',
+ 'voe_encryption_impl.cc',
+ 'voe_encryption_impl.h',
+ 'voe_external_media_impl.cc',
+ 'voe_external_media_impl.h',
+ 'voe_file_impl.cc',
+ 'voe_file_impl.h',
+ 'voe_hardware_impl.cc',
+ 'voe_hardware_impl.h',
+ 'voe_neteq_stats_impl.cc',
+ 'voe_neteq_stats_impl.h',
+ 'voe_network_impl.cc',
+ 'voe_network_impl.h',
+ 'voe_rtp_rtcp_impl.cc',
+ 'voe_rtp_rtcp_impl.h',
+ 'voe_video_sync_impl.cc',
+ 'voe_video_sync_impl.h',
+ 'voe_volume_control_impl.cc',
+ 'voe_volume_control_impl.h',
+ 'voice_engine_defines.h',
+ 'voice_engine_impl.cc',
+ 'voice_engine_impl.h',
+ ],
+ },
+ ],
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'voice_engine_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'voice_engine_core',
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+ '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
+ '<(webrtc_root)/modules/modules.gyp:audio_device',
+ '<(webrtc_root)/modules/modules.gyp:audio_processing',
+ '<(webrtc_root)/modules/modules.gyp:media_file',
+ '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+ '<(webrtc_root)/modules/modules.gyp:udp_transport',
+ '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'include_dirs': [
+ '../../..',
+ '../interface',
+ ],
+ 'sources': [
+ 'channel_unittest.cc',
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/voice_engine/main/source/voice_engine_defines.h b/trunk/src/voice_engine/main/source/voice_engine_defines.h
new file mode 100644
index 0000000..c652e6a
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voice_engine_defines.h
@@ -0,0 +1,597 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains common constants for VoiceEngine, as well as
+ * platform specific settings and include files.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
+#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
+
+#include "engine_configurations.h"
+
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+
+// VolumeControl
+enum { kMinVolumeLevel = 0 };
+enum { kMaxVolumeLevel = 255 };
+// Min scale factor for per-channel volume scaling
+const float kMinOutputVolumeScaling = 0.0f;
+// Max scale factor for per-channel volume scaling
+const float kMaxOutputVolumeScaling = 10.0f;
+// Min scale factor for output volume panning
+const float kMinOutputVolumePanning = 0.0f;
+// Max scale factor for output volume panning
+const float kMaxOutputVolumePanning = 1.0f;
+
+// DTMF
+enum { kMinDtmfEventCode = 0 }; // DTMF digit "0"
+enum { kMaxDtmfEventCode = 15 }; // DTMF digit "D"
+enum { kMinTelephoneEventCode = 0 }; // RFC4733 (Section 2.3.1)
+enum { kMaxTelephoneEventCode = 255 }; // RFC4733 (Section 2.3.1)
+enum { kMinTelephoneEventDuration = 100 };
+enum { kMaxTelephoneEventDuration = 60000 }; // Actual limit is 2^16
+enum { kMinTelephoneEventAttenuation = 0 }; // 0 dBm0
+enum { kMaxTelephoneEventAttenuation = 36 }; // -36 dBm0
+enum { kMinTelephoneEventSeparationMs = 100 }; // Min delta time between two
+ // telephone events
+
+enum { EcAec = 0 }; // AEC mode
+enum { EcAecm = 1 }; // AECM mode
+
+enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 }; // assumes Ethernet
+
+enum { kVoiceEngineMaxModuleVersionSize = 960 };
+
+// Base
+enum { kVoiceEngineVersionMaxMessageSize = 1024 };
+
+// Encryption
+// SRTP uses 30 bytes key length
+enum { kVoiceEngineMaxSrtpKeyLength = 30 };
+// SRTP minimum key/tag length for encryption level
+enum { kVoiceEngineMinSrtpEncryptLength = 16 };
+// SRTP maximum key/tag length for encryption level
+enum { kVoiceEngineMaxSrtpEncryptLength = 256 };
+// SRTP maximum key/tag length for authentication level,
+// HMAC SHA1 authentication type
+enum { kVoiceEngineMaxSrtpAuthSha1Length = 20 };
+// SRTP maximum tag length for authentication level,
+// null authentication type
+enum { kVoiceEngineMaxSrtpTagAuthNullLength = 12 };
+// SRTP maximum key length for authentication level,
+// null authentication type
+enum { kVoiceEngineMaxSrtpKeyAuthNullLength = 256 };
+
+// Audio processing
+enum { kVoiceEngineAudioProcessingDeviceSampleRateHz = 48000 };
+
+// Codec
+// Min init target rate for iSAC-wb
+enum { kVoiceEngineMinIsacInitTargetRateBpsWb = 10000 };
+// Max init target rate for iSAC-wb
+enum { kVoiceEngineMaxIsacInitTargetRateBpsWb = 32000 };
+// Min init target rate for iSAC-swb
+enum { kVoiceEngineMinIsacInitTargetRateBpsSwb = 10000 };
+// Max init target rate for iSAC-swb
+enum { kVoiceEngineMaxIsacInitTargetRateBpsSwb = 56000 };
+// Lowest max rate for iSAC-wb
+enum { kVoiceEngineMinIsacMaxRateBpsWb = 32000 };
+// Highest max rate for iSAC-wb
+enum { kVoiceEngineMaxIsacMaxRateBpsWb = 53400 };
+// Lowest max rate for iSAC-swb
+enum { kVoiceEngineMinIsacMaxRateBpsSwb = 32000 };
+// Highest max rate for iSAC-swb
+enum { kVoiceEngineMaxIsacMaxRateBpsSwb = 107000 };
+// Lowest max payload size for iSAC-wb
+enum { kVoiceEngineMinIsacMaxPayloadSizeBytesWb = 120 };
+// Highest max payload size for iSAC-wb
+enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesWb = 400 };
+// Lowest max payload size for iSAC-swb
+enum { kVoiceEngineMinIsacMaxPayloadSizeBytesSwb = 120 };
+// Highest max payload size for iSAC-swb
+enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb = 600 };
+
+// VideoSync
+// Lowest minimum playout delay
+enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
+// Highest minimum playout delay
+enum { kVoiceEngineMaxMinPlayoutDelayMs = 1000 };
+
+// Network
+// Min packet-timeout time for received RTP packets
+enum { kVoiceEngineMinPacketTimeoutSec = 1 };
+// Max packet-timeout time for received RTP packets
+enum { kVoiceEngineMaxPacketTimeoutSec = 150 };
+// Min sample time for dead-or-alive detection
+enum { kVoiceEngineMinSampleTimeSec = 1 };
+// Max sample time for dead-or-alive detection
+enum { kVoiceEngineMaxSampleTimeSec = 150 };
+
+// RTP/RTCP
+// Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
+enum { kVoiceEngineMinRtpExtensionId = 1 };
+// Max 4-bit ID for RTP extension
+enum { kVoiceEngineMaxRtpExtensionId = 14 };
+
+} // namespace webrtc
+
+// TODO(andrew): we shouldn't be using the precompiler for this.
+// Use enums or bools as appropriate.
+#define WEBRTC_AUDIO_PROCESSING_OFF false
+
+#define WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE true
+ // AudioProcessing HP is ON
+#define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing NS off
+#define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE true
+ // AudioProcessing AGC on
+#define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing EC off
+#define WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing off
+#define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing RX AGC off
+#define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing RX NS off
+#define WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ // AudioProcessing RX High Pass Filter off
+
+#define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE NoiseSuppression::kModerate
+ // AudioProcessing NS moderate suppression
+#define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE GainControl::kAdaptiveAnalog
+ // AudioProcessing AGC analog digital combined
+#define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAec
+ // AudioProcessing EC AEC
+#define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE GainControl::kAdaptiveDigital
+ // AudioProcessing AGC mode
+#define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE NoiseSuppression::kModerate
+ // AudioProcessing RX NS mode
+
+// Macros
+// Comparison of two strings without regard to case
+#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+// Compares characters of two strings without regard to case
+#define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n)
+
+// ----------------------------------------------------------------------------
+// Build information macros
+// ----------------------------------------------------------------------------
+
+#if defined(_DEBUG)
+#define BUILDMODE "d"
+#elif defined(DEBUG)
+#define BUILDMODE "d"
+#elif defined(NDEBUG)
+#define BUILDMODE "r"
+#else
+#define BUILDMODE "?"
+#endif
+
+#define BUILDTIME __TIME__
+#define BUILDDATE __DATE__
+
+// Example: "Oct 10 2002 12:05:30 r"
+#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
+
+// ----------------------------------------------------------------------------
+// Macros
+// ----------------------------------------------------------------------------
+
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+ #include <windows.h>
+ #include <stdio.h>
+ #define DEBUG_PRINT(...) \
+ { \
+ char msg[256]; \
+ sprintf(msg, __VA_ARGS__); \
+ OutputDebugStringA(msg); \
+ }
+#else
+ // special fix for visual 2003
+ #define DEBUG_PRINT(exp) ((void)0)
+#endif // defined(_DEBUG) && defined(_WIN32)
+
+#define CHECK_CHANNEL(channel) if (CheckChannel(channel) == -1) return -1;
+
+// ----------------------------------------------------------------------------
+// Default Trace filter
+// ----------------------------------------------------------------------------
+
+#define WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER \
+ kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | \
+ kTraceApiCall
+
+// ----------------------------------------------------------------------------
+// Inline functions
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+
+inline int VoEId(const int veId, const int chId)
+{
+ if (chId == -1)
+ {
+ const int dummyChannel(99);
+ return (int) ((veId << 16) + dummyChannel);
+ }
+ return (int) ((veId << 16) + chId);
+}
+
+inline int VoEModuleId(const int veId, const int chId)
+{
+ return (int) ((veId << 16) + chId);
+}
+
+// Convert module ID to internal VoE channel ID
+inline int VoEChannelId(const int moduleId)
+{
+ return (int) (moduleId & 0xffff);
+}
+
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Platform settings
+// ----------------------------------------------------------------------------
+
+// *** WINDOWS ***
+
+#if defined(_WIN32)
+
+ #pragma comment( lib, "winmm.lib" )
+
+ #ifndef WEBRTC_EXTERNAL_TRANSPORT
+ #pragma comment( lib, "ws2_32.lib" )
+ #endif
+
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+// Max number of supported channels
+enum { kVoiceEngineMaxNumOfChannels = 32 };
+// Max number of channels which can be played out simultaneously
+enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Defines
+// ----------------------------------------------------------------------------
+
+ #include <windows.h>
+ #include <mmsystem.h> // timeGetTime
+
+ #define GET_TIME_IN_MS() ::timeGetTime()
+ #define SLEEP(x) ::Sleep(x)
+ // Comparison of two strings without regard to case
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+ // Compares characters of two strings without regard to case
+ #define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n)
+
+// Default device for Windows PC
+ #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
+ AudioDeviceModule::kDefaultCommunicationDevice
+
+#endif // #if (defined(_WIN32)
+
+// *** LINUX ***
+
+#ifdef WEBRTC_LINUX
+
+#include <pthread.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#ifndef QNX
+ #include <linux/net.h>
+#ifndef ANDROID
+ #include <sys/soundcard.h>
+#endif // ANDROID
+#endif // QNX
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sched.h>
+#include <time.h>
+#include <sys/time.h>
+
+#define DWORD unsigned long int
+#define WINAPI
+#define LPVOID void *
+#define FALSE 0
+#define TRUE 1
+#define UINT unsigned int
+#define UCHAR unsigned char
+#define TCHAR char
+#ifdef QNX
+#define _stricmp stricmp
+#else
+#define _stricmp strcasecmp
+#endif
+#define GetLastError() errno
+#define WSAGetLastError() errno
+#define LPCTSTR const char*
+#define LPCSTR const char*
+#define wsprintf sprintf
+#define TEXT(a) a
+#define _ftprintf fprintf
+#define _tcslen strlen
+#define FAR
+#define __cdecl
+#define LPSOCKADDR struct sockaddr *
+
+namespace
+{
+ void Sleep(unsigned long x)
+ {
+ timespec t;
+ t.tv_sec = x/1000;
+ t.tv_nsec = (x-(x/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+ }
+
+ DWORD timeGetTime()
+ {
+ struct timeval tv;
+ struct timezone tz;
+ unsigned long val;
+
+ gettimeofday(&tv, &tz);
+ val= tv.tv_sec*1000+ tv.tv_usec/1000;
+ return(val);
+ }
+}
+
+#define SLEEP(x) ::Sleep(x)
+#define GET_TIME_IN_MS timeGetTime
+
+// Default device for Linux and Android
+#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
+
+#ifdef ANDROID
+
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+ // Max number of supported channels
+ enum { kVoiceEngineMaxNumOfChannels = 2 };
+ // Max number of channels which can be played out simultaneously
+ enum { kVoiceEngineMaxNumOfActiveChannels = 2 };
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Defines
+// ----------------------------------------------------------------------------
+
+ // Always excluded for Android builds
+ #undef WEBRTC_CODEC_ISAC
+ #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ #undef WEBRTC_CONFERENCING
+ #undef WEBRTC_TYPING_DETECTION
+
+ // Default audio processing states
+ #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE
+ #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE
+ #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE
+ #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+
+ // Default audio processing modes
+ #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE
+ #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE
+ #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE
+ #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \
+ NoiseSuppression::kModerate
+ #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \
+ GainControl::kAdaptiveDigital
+ #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm
+
+ #define ANDROID_NOT_SUPPORTED(stat) \
+ stat.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \
+ "API call not supported"); \
+ return -1;
+
+#else // LINUX PC
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+ // Max number of supported channels
+ enum { kVoiceEngineMaxNumOfChannels = 32 };
+ // Max number of channels which can be played out simultaneously
+ enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Defines
+// ----------------------------------------------------------------------------
+
+ #define ANDROID_NOT_SUPPORTED(stat)
+
+#endif // ANDROID - LINUX PC
+
+#else
+#define ANDROID_NOT_SUPPORTED(stat)
+#endif // #ifdef WEBRTC_LINUX
+
+// *** WEBRTC_MAC ***
+// including iPhone
+
+#ifdef WEBRTC_MAC
+
+#include <pthread.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <sys/stat.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sched.h>
+#include <sys/time.h>
+#include <time.h>
+#include <AudioUnit/AudioUnit.h>
+#if !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM)
+ #include <CoreServices/CoreServices.h>
+ #include <CoreAudio/CoreAudio.h>
+ #include <AudioToolbox/DefaultAudioOutput.h>
+ #include <AudioToolbox/AudioConverter.h>
+ #include <CoreAudio/HostTime.h>
+#endif
+
+#define DWORD unsigned long int
+#define WINAPI
+#define LPVOID void *
+#define FALSE 0
+#define TRUE 1
+#define SOCKADDR_IN struct sockaddr_in
+#define UINT unsigned int
+#define UCHAR unsigned char
+#define TCHAR char
+#define _stricmp strcasecmp
+#define GetLastError() errno
+#define WSAGetLastError() errno
+#define LPCTSTR const char*
+#define wsprintf sprintf
+#define TEXT(a) a
+#define _ftprintf fprintf
+#define _tcslen strlen
+#define FAR
+#define __cdecl
+#define LPSOCKADDR struct sockaddr *
+#define LPCSTR const char*
+#define ULONG unsigned long
+
+namespace
+{
+ void Sleep(unsigned long x)
+ {
+ timespec t;
+ t.tv_sec = x/1000;
+ t.tv_nsec = (x-(x/1000)*1000)*1000000;
+ nanosleep(&t,NULL);
+ }
+
+ DWORD WebRtcTimeGetTime()
+ {
+ struct timeval tv;
+ struct timezone tz;
+ unsigned long val;
+
+ gettimeofday(&tv, &tz);
+ val= tv.tv_sec*1000+ tv.tv_usec/1000;
+ return(val);
+ }
+}
+
+#define SLEEP(x) ::Sleep(x)
+#define GET_TIME_IN_MS WebRtcTimeGetTime
+
+// Default device for Mac and iPhone
+#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
+
+// iPhone specific
+#if defined(MAC_IPHONE) || defined(MAC_IPHONE_SIM)
+
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+ // Max number of supported channels
+ enum { kVoiceEngineMaxNumOfChannels = 2 };
+ // Max number of channels which can be played out simultaneously
+ enum { kVoiceEngineMaxNumOfActiveChannels = 2 };
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Defines
+// ----------------------------------------------------------------------------
+
+ // Always excluded for iPhone builds
+ #undef WEBRTC_CODEC_ISAC
+ #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+
+ #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE
+ #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE
+ #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE
+ #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+ #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
+
+ #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE
+ #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE
+ #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE
+ #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \
+ NoiseSuppression::kModerate
+ #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \
+ GainControl::kAdaptiveDigital
+ #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm
+
+ #define IPHONE_NOT_SUPPORTED() \
+ _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \
+ "API call not supported"); \
+ return -1;
+
+#else // Non-iPhone
+
+// ----------------------------------------------------------------------------
+// Enumerators
+// ----------------------------------------------------------------------------
+
+namespace webrtc
+{
+ // Max number of supported channels
+ enum { kVoiceEngineMaxNumOfChannels = 32 };
+ // Max number of channels which can be played out simultaneously
+ enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
+} // namespace webrtc
+
+// ----------------------------------------------------------------------------
+// Defines
+// ----------------------------------------------------------------------------
+
+ #define IPHONE_NOT_SUPPORTED()
+#endif
+
+#else
+#define IPHONE_NOT_SUPPORTED()
+#endif // #ifdef WEBRTC_MAC
+
+
+
+#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
diff --git a/trunk/src/voice_engine/main/source/voice_engine_impl.cc b/trunk/src/voice_engine/main/source/voice_engine_impl.cc
new file mode 100644
index 0000000..07af08e
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voice_engine_impl.cc
@@ -0,0 +1,317 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "voice_engine_impl.h"
+#include "trace.h"
+
+#ifdef WEBRTC_ANDROID
+extern "C"
+{
+extern WebRtc_Word32 SetAndroidAudioDeviceObjects(
+ void* javaVM, void* env, void* context);
+} // extern "C"
+#endif
+
+namespace webrtc
+{
+
+// Counter to be ensure that we can add a correct ID in all static trace
+// methods. It is not the nicest solution, especially not since we already
+// have a counter in VoEBaseImpl. In other words, there is room for
+// improvement here.
+static WebRtc_Word32 gVoiceEngineInstanceCounter = 0;
+
+extern "C"
+{
+WEBRTC_DLLEXPORT VoiceEngine* GetVoiceEngine();
+
+VoiceEngine* GetVoiceEngine()
+{
+ VoiceEngineImpl* self = new VoiceEngineImpl();
+ VoiceEngine* ve = reinterpret_cast<VoiceEngine*> (self);
+ if (ve != NULL)
+ {
+ gVoiceEngineInstanceCounter++;
+ }
+ return ve;
+}
+} // extern "C"
+
+VoiceEngine* VoiceEngine::Create()
+{
+#if (defined _WIN32)
+ HMODULE hmod_ = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
+
+ if (hmod_)
+ {
+ typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
+ PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
+ hmod_,"GetVoiceEngine");
+ if (pfn)
+ {
+ VoiceEngine* self = pfn();
+ return (self);
+ }
+ }
+#endif
+
+ return GetVoiceEngine();
+}
+
+int VoiceEngine::SetTraceFilter(const unsigned int filter)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+ VoEId(gVoiceEngineInstanceCounter, -1),
+ "SetTraceFilter(filter=0x%x)", filter);
+
+ // Remember old filter
+ WebRtc_UWord32 oldFilter = 0;
+ Trace::LevelFilter(oldFilter);
+
+ // Set new filter
+ WebRtc_Word32 ret = Trace::SetLevelFilter(filter);
+
+ // If previous log was ignored, log again after changing filter
+ if (kTraceNone == oldFilter)
+ {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
+ "SetTraceFilter(filter=0x%x)", filter);
+ }
+
+ return (ret);
+}
+
+int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
+ const bool addFileCounter)
+{
+ int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+ VoEId(gVoiceEngineInstanceCounter, -1),
+ "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)",
+ fileNameUTF8, addFileCounter);
+ return (ret);
+}
+
+int VoiceEngine::SetTraceCallback(TraceCallback* callback)
+{
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+ VoEId(gVoiceEngineInstanceCounter, -1),
+ "SetTraceCallback(callback=0x%x)", callback);
+ return (Trace::SetTraceCallback(callback));
+}
+
+bool VoiceEngine::Delete(VoiceEngine*& voiceEngine, bool ignoreRefCounters)
+{
+ if (voiceEngine == NULL)
+ {
+ return false;
+ }
+
+ VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
+ VoEBaseImpl* base = s;
+
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
+ "VoiceEngine::Delete(voiceEngine=0x%p, ignoreRefCounters=%d)",
+ voiceEngine, ignoreRefCounters);
+
+ if (!ignoreRefCounters)
+ {
+ if (base->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEBase reference counter is %d => memory will not "
+ "be released properly!", base->GetCount());
+ return false;
+ }
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+ VoECodecImpl* codec = s;
+ if (codec->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoECodec reference counter is %d => memory will not "
+ "be released properly!", codec->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+ VoEDtmfImpl* dtmf = s;
+ if (dtmf->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEDtmf reference counter is %d =>"
+ "memory will not be released properly!",
+ dtmf->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+ VoEEncryptionImpl* encrypt = s;
+ if (encrypt->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEEncryption reference counter is %d => "
+ "memory will not be released properly!",
+ encrypt->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+ VoEExternalMediaImpl* extmedia = s;
+ if (extmedia->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEExternalMedia reference counter is %d => "
+ "memory will not be released properly!",
+ extmedia->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+ VoECallReportImpl* report = s;
+ if (report->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoECallReport reference counter is %d => memory "
+ "will not be released properly!",
+ report->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+ VoEFileImpl* file = s;
+ if (file->GetCount() != 0)
+ {
+ WEBRTC_TRACE(
+ kTraceCritical,
+ kTraceVoice,
+ -1,
+ "VoEFile reference counter is %d => memory will not "
+ "be released properly!",
+ file->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+ VoEHardwareImpl* hware = s;
+ if (hware->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEHardware reference counter is %d => memory will "
+ "not be released properly!", hware->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ VoENetEqStatsImpl* neteqst = s;
+ if (neteqst->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoENetEqStats reference counter is %d => "
+ "memory will not be released properly!",
+ neteqst->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+ VoENetworkImpl* netw = s;
+ if (netw->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoENetworkImpl reference counter is %d => memory "
+ "will not be released properly!", netw->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+ VoERTP_RTCPImpl* rtcp = s;
+ if (rtcp->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoERTP_RTCP reference counter is %d =>"
+ "memory will not be released properly!",
+ rtcp->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ VoEVideoSyncImpl* vsync = s;
+ if (vsync->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEVideoSync reference counter is %d => "
+ "memory will not be released properly!",
+ vsync->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+ VoEVolumeControlImpl* volume = s;
+ if (volume->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEVolumeControl reference counter is %d =>"
+ "memory will not be released properly!",
+ volume->GetCount());
+ return false;
+ }
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+ VoEAudioProcessingImpl* apm = s;
+ if (apm->GetCount() != 0)
+ {
+ WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
+ "VoEAudioProcessing reference counter is %d => "
+ "memory will not be released properly!",
+ apm->GetCount());
+ return false;
+ }
+#endif
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, -1,
+ "all reference counters are zero => deleting the "
+ "VoiceEngine instance...");
+
+ } // if (!ignoreRefCounters)
+ else
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, -1,
+ "reference counters are ignored => deleting the "
+ "VoiceEngine instance...");
+ }
+
+ delete s;
+ voiceEngine = NULL;
+
+ return true;
+}
+
+int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
+{
+#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_ANDROID_OPENSLES)
+ // modules/audio_device/main/source/android/audio_device_android_jni.cc
+ // contains the actual implementation.
+ return SetAndroidAudioDeviceObjects(javaVM, env, context);
+#else
+ return -1;
+#endif
+}
+
+} //namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voice_engine_impl.h b/trunk/src/voice_engine/main/source/voice_engine_impl.h
new file mode 100644
index 0000000..708ac5d
--- /dev/null
+++ b/trunk/src/voice_engine/main/source/voice_engine_impl.h
@@ -0,0 +1,118 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
+#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
+
+#include "engine_configurations.h"
+#include "voe_base_impl.h"
+
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+#include "voe_audio_processing_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+#include "voe_call_report_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+#include "voe_codec_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+#include "voe_dtmf_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+#include "voe_encryption_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+#include "voe_external_media_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+#include "voe_file_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+#include "voe_hardware_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+#include "voe_neteq_stats_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+#include "voe_network_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+#include "voe_rtp_rtcp_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+#include "voe_video_sync_impl.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+#include "voe_volume_control_impl.h"
+#endif
+
+namespace webrtc
+{
+
+class VoiceEngineImpl :
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+ public VoEAudioProcessingImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+ public VoECallReportImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+ public VoECodecImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+ public VoEDtmfImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+ public VoEEncryptionImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+ public VoEExternalMediaImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+ public VoEFileImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+ public VoEHardwareImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ public VoENetEqStatsImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+ public VoENetworkImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+ public VoERTP_RTCPImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ public VoEVideoSyncImpl,
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+ public VoEVolumeControlImpl,
+#endif
+ public VoEBaseImpl
+{
+public:
+ VoiceEngineImpl() :
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ VoEVideoSyncImpl(this),
+#endif
+ VoEBaseImpl() // Included in initializer list to satisfy condition when
+ // none of the WEBRTC_VOICE_XXX defines are set.
+ {
+ }
+ virtual ~VoiceEngineImpl()
+ {
+ }
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
diff --git a/trunk/src/voice_engine/main/test/android/android_test/.classpath b/trunk/src/voice_engine/main/test/android/android_test/.classpath
new file mode 100644
index 0000000..86a15c9
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/.classpath
@@ -0,0 +1,7 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+ <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+ <classpathentry kind="src" path="src"/>
+ <classpathentry kind="src" path="gen"/>
+ <classpathentry kind="output" path="bin"/>
+</classpath>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/Android.mk b/trunk/src/voice_engine/main/test/android/android_test/Android.mk
new file mode 100644
index 0000000..a3f5ce6
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/Android.mk
@@ -0,0 +1,25 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ src/org/webrtc/voiceengine/test/AndroidTest.java
+
+LOCAL_PACKAGE_NAME := webrtc-voice-demo
+LOCAL_CERTIFICATE := platform
+
+LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-voice-demo-jni
+
+include $(BUILD_PACKAGE)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml b/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml
new file mode 100644
index 0000000..a614f8d
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
+<!-- -->
+<!-- Use of this source code is governed by a BSD-style license -->
+<!-- that can be found in the LICENSE file in the root of the source -->
+<!-- tree. An additional intellectual property rights grant can be found -->
+<!-- in the file PATENTS. All contributing project authors may -->
+<!-- be found in the AUTHORS file in the root of the source tree. -->
+
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ android:versionCode="1"
+ android:versionName="1.0" package="org.webrtc.voiceengine.test">
+ <application android:icon="@drawable/icon"
+ android:label="@string/app_name"
+ android:debuggable="true">
+ <activity android:name=".AndroidTest"
+ android:label="@string/app_name"
+ android:screenOrientation="portrait">
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN" />
+ <category android:name="android.intent.category.LAUNCHER" />
+ </intent-filter>
+ </activity>
+
+ </application>
+ <uses-sdk android:minSdkVersion="3" />
+
+ <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+ <uses-permission android:name="android.permission.RECORD_AUDIO" />
+ <uses-permission android:name="android.permission.INTERNET" />
+
+</manifest>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/default.properties b/trunk/src/voice_engine/main/test/android/android_test/default.properties
new file mode 100644
index 0000000..6ed608e
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target, OpenSL ES requires API level 9
+target=android-9
diff --git a/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java b/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java
new file mode 100644
index 0000000..ec8f5b4
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java
@@ -0,0 +1,30 @@
+/* AUTO-GENERATED FILE. DO NOT MODIFY.
+ *
+ * This class was automatically generated by the
+ * aapt tool from the resource data it found. It
+ * should not be modified by hand.
+ */
+
+package org.webrtc.voiceengine.test;
+
+public final class R {
+ public static final class attr {
+ }
+ public static final class drawable {
+ public static final int icon=0x7f020000;
+ }
+ public static final class id {
+ public static final int Button01=0x7f050002;
+ public static final int Button02=0x7f050005;
+ public static final int EditText01=0x7f050001;
+ public static final int Spinner01=0x7f050003;
+ public static final int Spinner02=0x7f050004;
+ public static final int TextView01=0x7f050000;
+ }
+ public static final class layout {
+ public static final int main=0x7f030000;
+ }
+ public static final class string {
+ public static final int app_name=0x7f040000;
+ }
+}
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk b/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk
new file mode 100644
index 0000000..9f9f761
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk
@@ -0,0 +1,152 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+ifdef NDK_ROOT
+
+MY_WEBRTC_ROOT_PATH := $(call my-dir)
+
+MY_WEBRTC_SRC_PATH := ../../../../../../..
+
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/resampler/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/signal_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/vad/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/neteq/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/cng/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/g711/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/g722/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/pcm16b/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/ilbc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_conference_mixer/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_device/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/aec/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/aecm/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/agc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/ns/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/utility/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/media_file/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/rtp_rtcp/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/udp_transport/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/utility/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/system_wrappers/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/voice_engine/main/source/Android.mk
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_audio_preprocessing
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_spl \
+ libwebrtc_resampler \
+ libwebrtc_apm \
+ libwebrtc_apm_utility \
+ libwebrtc_vad \
+ libwebrtc_ns \
+ libwebrtc_agc \
+ libwebrtc_aec \
+ libwebrtc_aecm \
+ libwebrtc_system_wrappers \
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libstlport_shared
+
+LOCAL_LDLIBS := \
+ -lgcc \
+ -llog
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
+
+###
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc-voice-jni
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_system_wrappers \
+ libwebrtc_audio_device \
+ libwebrtc_pcm16b \
+ libwebrtc_cng \
+ libwebrtc_audio_coding \
+ libwebrtc_rtp_rtcp \
+ libwebrtc_media_file \
+ libwebrtc_udp_transport \
+ libwebrtc_utility \
+ libwebrtc_neteq \
+ libwebrtc_audio_conference_mixer \
+ libwebrtc_isac \
+ libwebrtc_ilbc \
+ libwebrtc_isacfix \
+ libwebrtc_g722 \
+ libwebrtc_g711 \
+ libwebrtc_voe_core
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libstlport_shared \
+ libwebrtc_audio_preprocessing
+
+LOCAL_LDLIBS := \
+ -lgcc \
+ -llog \
+ -lOpenSLES
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
+
+###
+
+include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
+
+else
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_MODULE := libwebrtc-voice-demo-jni
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := android_test.cc
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_ANDROID'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../../../auto_test \
+ $(LOCAL_PATH)/../../../../interface \
+ $(LOCAL_PATH)/../../../../../.. \
+ $(LOCAL_PATH)/../../../../../../system_wrappers/interface
+
+LOCAL_PRELINK_MODULE := false
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libandroid \
+ libwebrtc \
+ libGLESv2
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk b/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk
new file mode 100644
index 0000000..03c35ac
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk
@@ -0,0 +1,11 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Build both ARMv5TE and ARMv7-A machine code.
+APP_ABI := armeabi armeabi-v7a x86
+APP_STL := stlport_shared
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc b/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc
new file mode 100644
index 0000000..8c5fdff
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc
@@ -0,0 +1,1554 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <android/log.h>
+
+#include "org_webrtc_voiceengine_test_AndroidTest.h"
+
+#include "thread_wrapper.h"
+
+#include "voe_base.h"
+#include "voe_codec.h"
+#include "voe_file.h"
+#include "voe_network.h"
+#include "voe_audio_processing.h"
+#include "voe_volume_control.h"
+#include "voe_hardware.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_encryption.h"
+
+#include "voe_test_interface.h"
+
+//#define USE_SRTP
+//#define INIT_FROM_THREAD
+//#define START_CALL_FROM_THREAD
+
+#define WEBRTC_LOG_TAG "*WEBRTCN*" // As in WEBRTC Native...
+#define VALIDATE_BASE_POINTER \
+ if (!veData1.base) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Base pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_CODEC_POINTER \
+ if (!veData1.codec) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Codec pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_FILE_POINTER \
+ if (!veData1.file) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "File pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_NETWORK_POINTER \
+ if (!veData1.netw) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Network pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_APM_POINTER \
+ if (!veData1.codec) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Apm pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_VOLUME_POINTER \
+ if (!veData1.volume) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Volume pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_HARDWARE_POINTER \
+ if (!veData1.hardware) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Hardware pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_RTP_RTCP_POINTER \
+ if (!veData1.rtp_rtcp) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "RTP / RTCP pointer doesn't exist"); \
+ return -1; \
+ }
+#define VALIDATE_ENCRYPT_POINTER \
+ if (!veData1.encrypt) \
+ { \
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
+ "Encrypt pointer doesn't exist"); \
+ return -1; \
+ }
+
+// Register functions in JNI_OnLoad()
+// How do we ensure that VoE is deleted? JNI_OnUnload?
+// What happens if class is unloaded? When loaded again, NativeInit will be
+// called again. Keep what we have?
+// Should we do something in JNI_OnUnload?
+// General design: create a class or keep global struct with "C" functions?
+// Otherwise make sure symbols are as unique as possible.
+
+// TestType enumerator
+enum TestType
+{
+ Invalid = -1,
+ Standard = 0,
+ Extended = 1,
+ Stress = 2,
+ Unit = 3,
+ CPU = 4
+};
+
+// ExtendedSelection enumerator
+enum ExtendedSelection
+{
+ XSEL_Invalid = -1,
+ XSEL_None = 0,
+ XSEL_All,
+ XSEL_Base,
+ XSEL_CallReport,
+ XSEL_Codec,
+ XSEL_DTMF,
+ XSEL_Encryption,
+ XSEL_ExternalMedia,
+ XSEL_File,
+ XSEL_Hardware,
+ XSEL_NetEqStats,
+ XSEL_Network,
+ XSEL_PTT,
+ XSEL_RTP_RTCP,
+ XSEL_VideoSync,
+ XSEL_VideoSyncExtended,
+ XSEL_VolumeControl,
+ XSEL_VQE,
+ XSEL_APM,
+ XSEL_VQMon
+};
+
+using namespace webrtc;
+
+class my_transportation;
+
+// VoiceEngine data struct
+typedef struct
+{
+ // VoiceEngine
+ VoiceEngine* ve;
+ // Sub-APIs
+ VoEBase* base;
+ VoECodec* codec;
+ VoEFile* file;
+ VoENetwork* netw;
+ VoEAudioProcessing* apm;
+ VoEVolumeControl* volume;
+ VoEHardware* hardware;
+ VoERTP_RTCP* rtp_rtcp;
+ VoEEncryption* encrypt;
+ // Other
+ my_transportation* extTrans;
+ JavaVM* jvm;
+} VoiceEngineData;
+
+// my_transportation is used when useExtTrans is enabled
+class my_transportation : public Transport
+{
+ public:
+ my_transportation(VoENetwork * network) :
+ netw(network) {
+ }
+
+ int SendPacket(int channel,const void *data,int len);
+ int SendRTCPPacket(int channel, const void *data, int len);
+ private:
+ VoENetwork * netw;
+};
+
+int my_transportation::SendPacket(int channel,const void *data,int len)
+{
+ netw->ReceivedRTPPacket(channel, data, len);
+ return len;
+}
+
+int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
+{
+ netw->ReceivedRTCPPacket(channel, data, len);
+ return len;
+}
+
+//Global variables visible in this file
+static VoiceEngineData veData1;
+static VoiceEngineData veData2;
+
+// "Local" functions (i.e. not Java accessible)
+static bool GetSubApis(VoiceEngineData &veData);
+static bool ReleaseSubApis(VoiceEngineData &veData);
+
+class ThreadTest
+{
+public:
+ ThreadTest();
+ ~ThreadTest();
+ int RunTest();
+ int CloseTest();
+private:
+ static bool Run(void* ptr);
+ bool Process();
+private:
+ ThreadWrapper* _thread;
+};
+
+ThreadTest::~ThreadTest()
+{
+ if (_thread)
+ {
+ _thread->SetNotAlive();
+ if (_thread->Stop())
+ {
+ delete _thread;
+ _thread = NULL;
+ }
+ }
+}
+
+ThreadTest::ThreadTest() :
+ _thread(NULL)
+{
+ _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
+ "ThreadTest thread");
+}
+
+bool ThreadTest::Run(void* ptr)
+{
+ return static_cast<ThreadTest*> (ptr)->Process();
+}
+
+bool ThreadTest::Process()
+{
+ // Attach this thread to JVM
+ /*JNIEnv* env = NULL;
+ jint res = veData1.jvm->AttachCurrentThread(&env, NULL);
+ char msg[32];
+ sprintf(msg, "res=%d, env=%d", res, env);
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, msg);*/
+
+#ifdef INIT_FROM_THREAD
+ VALIDATE_BASE_POINTER;
+ veData1.base->Init();
+#endif
+
+#ifdef START_CALL_FROM_THREAD
+ // receiving instance
+ veData2.ve = VoiceEngine::Create();
+ GetSubApis(veData2);
+ veData2.base->Init();
+ veData2.base->CreateChannel();
+ if(veData2.base->SetLocalReceiver(0, 1234) < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "set local receiver 2 failed");
+ }
+ veData2.hardware->SetLoudspeakerStatus(false);
+ veData2.volume->SetSpeakerVolume(204);
+ veData2.base->StartReceive(0);
+ if(veData2.base->StartPlayout(0) < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "start playout failed");
+ }
+
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "receiving instance started from thread");
+
+ // sending instance
+ veData1.ve = VoiceEngine::Create();
+ GetSubApis(veData1);
+ veData1.base->Init();
+ if(veData1.base->CreateChannel() < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "create channel failed");
+ }
+ if(veData1.base->SetLocalReceiver(0, 1256) < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "set local receiver failed");
+ }
+ if(veData1.base->SetSendDestination(0, 1234, "127.0.0.1") < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "set send destination failed");
+ }
+ if(veData1.base->StartSend(0) < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "start send failed");
+ }
+
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "sending instance started from thread");
+#endif
+
+ _thread->SetNotAlive();
+ _thread->Stop();
+
+ //res = veData1.jvm->DetachCurrentThread();
+
+ return true;
+}
+
+int ThreadTest::RunTest()
+{
+ if (_thread)
+ {
+ unsigned int id;
+ _thread->Start(id);
+ }
+ return 0;
+}
+
+int ThreadTest::CloseTest()
+{
+ VALIDATE_BASE_POINTER
+
+ veData1.base->DeleteChannel(0);
+ veData2.base->DeleteChannel(0);
+ veData1.base->Terminate();
+ veData2.base->Terminate();
+
+ // Release sub-APIs
+ ReleaseSubApis(veData1);
+ ReleaseSubApis(veData2);
+
+ // Delete
+ VoiceEngine::Delete(veData1.ve);
+ VoiceEngine::Delete(veData2.ve);
+ veData2.ve = NULL;
+ veData2.ve = NULL;
+
+ return 0;
+}
+
+ThreadTest threadTest;
+
+//////////////////////////////////////////////////////////////////
+// General functions
+//////////////////////////////////////////////////////////////////
+
+/////////////////////////////////////////////
+// JNI_OnLoad
+//
+jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
+{
+ if (!vm)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad did not receive a valid VM pointer");
+ return -1;
+ }
+
+ // Get JNI
+ JNIEnv* env;
+ if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
+ JNI_VERSION_1_4))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "JNI_OnLoad could not get JNI env");
+ return -1;
+ }
+
+ // Get class to register the native functions with
+ // jclass regClass = env->FindClass("webrtc/android/AndroidTest");
+ // if (!regClass) {
+ // return -1; // Exception thrown
+ // }
+
+ // Register native functions
+ // JNINativeMethod methods[1];
+ // methods[0].name = NULL;
+ // methods[0].signature = NULL;
+ // methods[0].fnPtr = NULL;
+ // if (JNI_OK != env->RegisterNatives(regClass, methods, 1))
+ // {
+ // return -1;
+ // }
+
+ // Init VoiceEngine data
+ memset(&veData1, 0, sizeof(veData1));
+ memset(&veData2, 0, sizeof(veData2));
+
+ // Store the JVM
+ veData1.jvm = vm;
+ veData2.jvm = vm;
+
+ return JNI_VERSION_1_4;
+}
+
+/////////////////////////////////////////////
+// Native initialization
+//
+JNIEXPORT jboolean JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_NativeInit(
+ JNIEnv * env,
+ jclass)
+{
+ // Look up and cache any interesting class, field and method IDs for
+ // any used java class here
+
+ return true;
+}
+
+/////////////////////////////////////////////
+// Run auto standard test
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_RunAutoTest(
+ JNIEnv *env,
+ jobject context,
+ jint testType,
+ jint extendedSel)
+{
+ TestType tType(Invalid);
+
+ switch (testType)
+ {
+ case 0:
+ return 0;
+ case 1:
+ tType = Standard;
+ break;
+ case 2:
+ tType = Extended;
+ break;
+ case 3:
+ tType = Stress;
+ break;
+ case 4:
+ tType = Unit;
+ break;
+ default:
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "RunAutoTest - Invalid TestType");
+ return -1;
+ }
+
+ ExtendedSelection xsel(XSEL_Invalid);
+
+ switch (extendedSel)
+ {
+ case 0:
+ xsel = XSEL_None;
+ break;
+ case 1:
+ xsel = XSEL_All;
+ break;
+ case 2:
+ xsel = XSEL_Base;
+ break;
+ case 3:
+ xsel = XSEL_CallReport;
+ break;
+ case 4:
+ xsel = XSEL_Codec;
+ break;
+ case 5:
+ xsel = XSEL_DTMF;
+ break;
+ case 6:
+ xsel = XSEL_Encryption;
+ break;
+ case 7:
+ xsel = XSEL_ExternalMedia;
+ break;
+ case 8:
+ xsel = XSEL_File;
+ break;
+ case 9:
+ xsel = XSEL_Hardware;
+ break;
+ case 10:
+ xsel = XSEL_NetEqStats;
+ break;
+ case 11:
+ xsel = XSEL_Network;
+ break;
+ case 12:
+ xsel = XSEL_PTT;
+ break;
+ case 13:
+ xsel = XSEL_RTP_RTCP;
+ break;
+ case 14:
+ xsel = XSEL_VideoSync;
+ break;
+ case 15:
+ xsel = XSEL_VideoSyncExtended;
+ break;
+ case 16:
+ xsel = XSEL_VolumeControl;
+ break;
+ case 17:
+ xsel = XSEL_APM;
+ break;
+ case 18:
+ xsel = XSEL_VQMon;
+ break;
+ default:
+ xsel = XSEL_Invalid;
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "RunAutoTest - Invalid extendedType");
+ return -1;
+ }
+
+ // Set instance independent Java objects
+ VoiceEngine::SetAndroidObjects(veData1.jvm, env, context);
+
+ // Call voe test interface function
+ // TODO(leozwang) add autotest setAndroidObjects(veData1.jvm, context);
+ // jint retVal = runAutoTest(tType, xsel);
+
+ // Clear instance independent Java objects
+ VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
+
+ return 0;
+}
+
+//////////////////////////////////////////////////////////////////
+// VoiceEngine API wrapper functions
+//////////////////////////////////////////////////////////////////
+
+/////////////////////////////////////////////
+// Create VoiceEngine instance
+//
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Create(
+ JNIEnv *env,
+ jobject context)
+{
+ // Check if already created
+ if (veData1.ve)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "VoE already created");
+ return false;
+ }
+
+ // Set instance independent Java objects
+ VoiceEngine::SetAndroidObjects(veData1.jvm, env, context);
+
+#ifdef START_CALL_FROM_THREAD
+ threadTest.RunTest();
+#else
+ // Create
+ veData1.ve = VoiceEngine::Create();
+ if (!veData1.ve)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Create VoE failed");
+ return false;
+ }
+
+ // Get sub-APIs
+ if (!GetSubApis(veData1))
+ {
+ // If not OK, release all sub-APIs and delete VoE
+ ReleaseSubApis(veData1);
+ if (!VoiceEngine::Delete(veData1.ve))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Delete VoE failed");
+ }
+ return false;
+ }
+#endif
+
+ return true;
+}
+
+/////////////////////////////////////////////
+// Delete VoiceEngine instance
+//
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Delete(
+ JNIEnv *,
+ jobject)
+{
+#ifdef START_CALL_FROM_THREAD
+ threadTest.CloseTest();
+#else
+ // Check if exists
+ if (!veData1.ve)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "VoE does not exist");
+ return false;
+ }
+
+ // Release sub-APIs
+ ReleaseSubApis(veData1);
+
+ // Delete
+ if (!VoiceEngine::Delete(veData1.ve))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Delete VoE failed");
+ return false;
+ }
+
+ veData1.ve = NULL;
+#endif
+
+ // Clear instance independent Java objects
+ VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
+
+ return true;
+}
+
+/////////////////////////////////////////////
+// [Base] Initialize VoiceEngine
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Init(
+ JNIEnv *,
+ jobject,
+ jboolean enableTrace,
+ jboolean useExtTrans)
+{
+ VALIDATE_BASE_POINTER;
+
+ if (enableTrace)
+ {
+ if (0 != VoiceEngine::SetTraceFile("/sdcard/trace.txt"))
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not enable trace");
+ }
+ if (0 != VoiceEngine::SetTraceFilter(kTraceAll))
+ {
+ __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
+ "Could not set trace filter");
+ }
+ }
+
+ if (useExtTrans)
+ {
+ VALIDATE_NETWORK_POINTER;
+ veData1.extTrans = new my_transportation(veData1.netw);
+ }
+
+ int retVal = 0;
+#ifdef INIT_FROM_THREAD
+ threadTest.RunTest();
+ usleep(200000);
+#else
+ retVal = veData1.base->Init();
+#endif
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] Terminate VoiceEngine
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Terminate(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_BASE_POINTER;
+
+ jint retVal = veData1.base->Terminate();
+
+ delete veData1.extTrans;
+ veData1.extTrans = NULL;
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] Create channel
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_CreateChannel(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_BASE_POINTER;
+ jint channel = veData1.base->CreateChannel();
+
+ if (veData1.extTrans)
+ {
+ VALIDATE_NETWORK_POINTER;
+ __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
+ "Enabling external transport on channel %d",
+ channel);
+ if (veData1.netw->RegisterExternalTransport(channel, *veData1.extTrans)
+ < 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not set external transport");
+ return -1;
+ }
+ }
+
+ return channel;
+}
+
+/////////////////////////////////////////////
+// [Base] Delete channel
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_DeleteChannel(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData1.base->DeleteChannel(channel);
+}
+
+/////////////////////////////////////////////
+// [Base] SetLocalReceiver
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetLocalReceiver(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint port)
+{
+ VALIDATE_BASE_POINTER;
+ return veData1.base->SetLocalReceiver(channel, port);
+}
+
+/////////////////////////////////////////////
+// [Base] SetSendDestination
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetSendDestination(
+ JNIEnv *env,
+ jobject,
+ jint channel,
+ jint port,
+ jstring ipaddr)
+{
+ VALIDATE_BASE_POINTER;
+
+ const char* ipaddrNative = env->GetStringUTFChars(ipaddr, NULL);
+ if (!ipaddrNative)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData1.base->SetSendDestination(channel, port, ipaddrNative);
+
+ env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] StartListen
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartListen(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+#ifdef USE_SRTP
+ VALIDATE_ENCRYPT_POINTER;
+ bool useForRTCP = false;
+ if (veData1.encrypt->EnableSRTPReceive(
+ channel,CIPHER_AES_128_COUNTER_MODE,30,AUTH_HMAC_SHA1,
+ 16,4, ENCRYPTION_AND_AUTHENTICATION,
+ (unsigned char*)nikkey, useForRTCP) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to enable SRTP receive");
+ return -1;
+ }
+#endif
+
+ VALIDATE_BASE_POINTER;
+ int retVal = veData1.base->StartReceive(channel);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] Start playout
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayout(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ int retVal = veData1.base->StartPlayout(channel);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] Start send
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ /* int dscp(0), serviceType(-1), overrideDscp(0), res(0);
+ bool gqosEnabled(false), useSetSockOpt(false);
+
+ if (veData1.netw->SetSendTOS(channel, 13, useSetSockOpt) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to set TOS");
+ return -1;
+ }
+
+ res = veData1.netw->GetSendTOS(channel, dscp, useSetSockOpt);
+ if (res != 0 || dscp != 13 || useSetSockOpt != true)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to get TOS");
+ return -1;
+ } */
+
+ /* if (veData1.rtp_rtcp->SetFECStatus(channel, 1) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to enable FEC");
+ return -1;
+ } */
+#ifdef USE_SRTP
+ VALIDATE_ENCRYPT_POINTER;
+ bool useForRTCP = false;
+ if (veData1.encrypt->EnableSRTPSend(
+ channel,CIPHER_AES_128_COUNTER_MODE,30,AUTH_HMAC_SHA1,
+ 16,4, ENCRYPTION_AND_AUTHENTICATION,
+ (unsigned char*)nikkey, useForRTCP) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to enable SRTP send");
+ return -1;
+ }
+#endif
+
+ VALIDATE_BASE_POINTER;
+ int retVal = veData1.base->StartSend(channel);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [Base] Stop listen
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopListen(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+#ifdef USE_SRTP
+ VALIDATE_ENCRYPT_POINTER;
+ if (veData1.encrypt->DisableSRTPReceive(channel) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to disable SRTP receive");
+ return -1;
+ }
+#endif
+
+ VALIDATE_BASE_POINTER;
+ return veData1.base->StopReceive(channel);
+}
+
+/////////////////////////////////////////////
+// [Base] Stop playout
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayout(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_BASE_POINTER;
+ return veData1.base->StopPlayout(channel);
+}
+
+/////////////////////////////////////////////
+// [Base] Stop send
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopSend(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ /* if (veData1.rtp_rtcp->SetFECStatus(channel, 0) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to disable FEC");
+ return -1;
+ } */
+
+#ifdef USE_SRTP
+ VALIDATE_ENCRYPT_POINTER;
+ if (veData1.encrypt->DisableSRTPSend(channel) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to disable SRTP send");
+ return -1;
+ }
+#endif
+
+ VALIDATE_BASE_POINTER;
+ return veData1.base->StopSend(channel);
+}
+
+/////////////////////////////////////////////
+// [codec] Number of codecs
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NumOfCodecs(
+ JNIEnv *,
+ jobject)
+{
+ VALIDATE_CODEC_POINTER;
+ return veData1.codec->NumOfCodecs();
+}
+
+/////////////////////////////////////////////
+// [codec] Set send codec
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetSendCodec(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jint index)
+{
+ VALIDATE_CODEC_POINTER;
+
+ CodecInst codec;
+
+ if (veData1.codec->GetCodec(index, codec) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to get codec");
+ return -1;
+ }
+
+ return veData1.codec->SetSendCodec(channel, codec);
+}
+
+/////////////////////////////////////////////
+// [codec] Set VAD status
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetVADStatus(
+ JNIEnv *,
+ jobject,
+ jint channel,
+ jboolean enable,
+ jint mode)
+{
+ VALIDATE_CODEC_POINTER;
+
+ VadModes VADmode = kVadConventional;
+
+ switch (mode)
+ {
+ case 0:
+ break; // already set
+ case 1:
+ VADmode = kVadAggressiveLow;
+ break;
+ case 2:
+ VADmode = kVadAggressiveMid;
+ break;
+ case 3:
+ VADmode = kVadAggressiveHigh;
+ break;
+ default:
+ VADmode = (VadModes) 17; // force error
+ break;
+ }
+
+ return veData1.codec->SetVADStatus(channel, enable, VADmode);
+}
+
+/////////////////////////////////////////////
+// [apm] SetNSStatus
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetNSStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode)
+{
+ VALIDATE_APM_POINTER;
+
+ NsModes NSmode = kNsDefault;
+
+ switch (mode)
+ {
+ case 0:
+ NSmode = kNsUnchanged;
+ break;
+ case 1:
+ break; // already set
+ case 2:
+ NSmode = kNsConference;
+ break;
+ case 3:
+ NSmode = kNsLowSuppression;
+ break;
+ case 4:
+ NSmode = kNsModerateSuppression;
+ break;
+ case 5:
+ NSmode = kNsHighSuppression;
+ break;
+ case 6:
+ NSmode = kNsVeryHighSuppression;
+ break;
+ default:
+ NSmode = (NsModes) 17; // force error
+ break;
+ }
+
+ return veData1.apm->SetNsStatus(enable, NSmode);
+}
+
+/////////////////////////////////////////////
+// [apm] SetAGCStatus
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetAGCStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode)
+{
+ VALIDATE_APM_POINTER;
+
+ AgcModes AGCmode = kAgcDefault;
+
+ switch (mode)
+ {
+ case 0:
+ AGCmode = kAgcUnchanged;
+ break;
+ case 1:
+ break; // already set
+ case 2:
+ AGCmode = kAgcAdaptiveAnalog;
+ break;
+ case 3:
+ AGCmode = kAgcAdaptiveDigital;
+ break;
+ case 4:
+ AGCmode = kAgcFixedDigital;
+ break;
+ default:
+ AGCmode = (AgcModes) 17; // force error
+ break;
+ }
+
+ /* AgcConfig agcConfig;
+ agcConfig.targetLeveldBOv = 3;
+ agcConfig.digitalCompressionGaindB = 50;
+ agcConfig.limiterEnable = 0;
+
+ if (veData1.apm->SetAGCConfig(agcConfig) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Failed to set AGC config");
+ return -1;
+ } */
+
+ return veData1.apm->SetAgcStatus(enable, AGCmode);
+}
+
+/////////////////////////////////////////////
+// [apm] SetECStatus
+//
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable,
+ jint mode)
+{
+ VALIDATE_APM_POINTER;
+
+ EcModes ECmode = kEcDefault;
+
+ switch (mode)
+ {
+ case 0:
+ ECmode = kEcDefault;
+ break;
+ case 1:
+ break; // already set
+ case 2:
+ ECmode = kEcConference;
+ break;
+ case 3:
+ ECmode = kEcAec;
+ break;
+ case 4:
+ ECmode = kEcAecm;
+ break;
+ default:
+ ECmode = (EcModes) 17; // force error
+ break;
+ }
+
+ return veData1.apm->SetEcStatus(enable, ECmode);
+}
+
+/////////////////////////////////////////////
+// [File] Start play file locally
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileLocally(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jstring fileName,
+ jboolean loop)
+{
+ VALIDATE_FILE_POINTER;
+
+ const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
+ if (!fileNameNative)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData1.file->StartPlayingFileLocally(channel,
+ fileNameNative, loop);
+
+ env->ReleaseStringUTFChars(fileName, fileNameNative);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [File] Stop play file locally
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileLocally(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_FILE_POINTER;
+ return veData1.file->StopPlayingFileLocally(channel);
+}
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartRecordingPlayout
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StartRecordingPlayout(
+ JNIEnv * env,
+ jobject,
+ jint channel,
+ jstring fileName,
+ jboolean)
+{
+ VALIDATE_FILE_POINTER;
+
+ const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
+ if (!fileNameNative)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData1.file->StartRecordingPlayout(channel, fileNameNative,
+ 0);
+
+ env->ReleaseStringUTFChars(fileName, fileNameNative);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [File] Stop Recording Playout
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StopRecordingPlayout(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_FILE_POINTER;
+ return veData1.file->StopRecordingPlayout(channel);
+}
+
+/////////////////////////////////////////////
+// [File] Start playing file as microphone
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileAsMicrophone(
+ JNIEnv *env,
+ jobject,
+ jint channel,
+ jstring fileName,
+ jboolean loop)
+{
+ VALIDATE_FILE_POINTER;
+
+ const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
+ if (!fileNameNative)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not get UTF string");
+ return -1;
+ }
+
+ jint retVal = veData1.file->StartPlayingFileAsMicrophone(channel,
+ fileNameNative,
+ loop);
+
+ env->ReleaseStringUTFChars(fileName, fileNameNative);
+
+ return retVal;
+}
+
+/////////////////////////////////////////////
+// [File] Stop playing file as microphone
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileAsMicrophone(
+ JNIEnv *,
+ jobject,
+ jint channel)
+{
+ VALIDATE_FILE_POINTER;
+ return veData1.file->StopPlayingFileAsMicrophone(channel);
+}
+
+/////////////////////////////////////////////
+// [Volume] Set speaker volume
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume(
+ JNIEnv *,
+ jobject,
+ jint level)
+{
+ VALIDATE_VOLUME_POINTER;
+ if (veData1.volume->SetSpeakerVolume(level) != 0)
+ {
+ return -1;
+ }
+
+ unsigned int storedVolume = 0;
+ if (veData1.volume->GetSpeakerVolume(storedVolume) != 0)
+ {
+ return -1;
+ }
+
+ if (storedVolume != level)
+ {
+ return -1;
+ }
+
+ return 0;
+}
+
+/////////////////////////////////////////////
+// [Hardware] Set loudspeaker status
+//
+JNIEXPORT jint JNICALL
+Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus(
+ JNIEnv *,
+ jobject,
+ jboolean enable)
+{
+ VALIDATE_HARDWARE_POINTER;
+ if (veData1.hardware->SetLoudspeakerStatus(enable) != 0)
+ {
+ return -1;
+ }
+
+ /*VALIDATE_RTP_RTCP_POINTER;
+
+ if (veData1.rtp_rtcp->SetFECStatus(0, enable, -1) != 0)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could not set FEC");
+ return -1;
+ }
+ else if(enable)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could enable FEC");
+ }
+ else
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Could disable FEC");
+ }*/
+
+ return 0;
+}
+
+//////////////////////////////////////////////////////////////////
+// "Local" functions (i.e. not Java accessible)
+//////////////////////////////////////////////////////////////////
+
+/////////////////////////////////////////////
+// Get all sub-APIs
+//
+bool GetSubApis(VoiceEngineData &veData)
+{
+ bool getOK = true;
+
+ // Base
+ veData.base = VoEBase::GetInterface(veData.ve);
+ if (!veData.base)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get base sub-API failed");
+ getOK = false;
+ }
+
+ // Codec
+ veData.codec = VoECodec::GetInterface(veData.ve);
+ if (!veData.codec)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get codec sub-API failed");
+ getOK = false;
+ }
+
+ // File
+ veData.file = VoEFile::GetInterface(veData.ve);
+ if (!veData.file)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get file sub-API failed");
+ getOK = false;
+ }
+
+ // Network
+ veData.netw = VoENetwork::GetInterface(veData.ve);
+ if (!veData.netw)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get network sub-API failed");
+ getOK = false;
+ }
+
+ // AudioProcessing module
+ veData.apm = VoEAudioProcessing::GetInterface(veData.ve);
+ if (!veData.apm)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get apm sub-API failed");
+ getOK = false;
+ }
+
+ // Volume
+ veData.volume = VoEVolumeControl::GetInterface(veData.ve);
+ if (!veData.volume)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get volume sub-API failed");
+ getOK = false;
+ }
+
+ // Hardware
+ veData.hardware = VoEHardware::GetInterface(veData.ve);
+ if (!veData.hardware)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get hardware sub-API failed");
+ getOK = false;
+ }
+
+ // RTP / RTCP
+ veData.rtp_rtcp = VoERTP_RTCP::GetInterface(veData.ve);
+ if (!veData.rtp_rtcp)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get rtp_rtcp sub-API failed");
+ getOK = false;
+ }
+
+ // Encrypt
+ veData.encrypt = VoEEncryption::GetInterface(veData.ve);
+ if (!veData.encrypt)
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Get encrypt sub-API failed");
+ getOK = false;
+ }
+
+ return getOK;
+}
+
+/////////////////////////////////////////////
+// Release all sub-APIs
+//
+bool ReleaseSubApis(VoiceEngineData &veData)
+{
+ bool releaseOK = true;
+
+ // Base
+ if (veData.base)
+ {
+ if (0 != veData.base->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release base sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.base = NULL;
+ }
+ }
+
+ // Codec
+ if (veData.codec)
+ {
+ if (0 != veData.codec->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release codec sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.codec = NULL;
+ }
+ }
+
+ // File
+ if (veData.file)
+ {
+ if (0 != veData.file->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release file sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.file = NULL;
+ }
+ }
+
+ // Network
+ if (veData.netw)
+ {
+ if (0 != veData.netw->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release network sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.netw = NULL;
+ }
+ }
+
+ // apm
+ if (veData.apm)
+ {
+ if (0 != veData.apm->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release apm sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.apm = NULL;
+ }
+ }
+
+ // Volume
+ if (veData.volume)
+ {
+ if (0 != veData.volume->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release volume sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.volume = NULL;
+ }
+ }
+
+ // Hardware
+ if (veData.hardware)
+ {
+ if (0 != veData.hardware->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release hardware sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.hardware = NULL;
+ }
+ }
+
+ // RTP RTCP
+ if (veData.rtp_rtcp)
+ {
+ if (0 != veData.rtp_rtcp->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release rtp_rtcp sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.rtp_rtcp = NULL;
+ }
+ }
+
+ // Encrypt
+ if (veData.encrypt)
+ {
+ if (0 != veData.encrypt->Release())
+ {
+ __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
+ "Release encrypt sub-API failed");
+ releaseOK = false;
+ }
+ else
+ {
+ veData.encrypt = NULL;
+ }
+ }
+
+ return releaseOK;
+}
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h b/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
new file mode 100644
index 0000000..60fe839
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
@@ -0,0 +1,253 @@
+/* DO NOT EDIT THIS FILE - it is machine generated */
+#include <jni.h>
+/* Header for class org_webrtc_voiceengine_test_AndroidTest */
+
+#ifndef _Included_org_webrtc_voiceengine_test_AndroidTest
+#define _Included_org_webrtc_voiceengine_test_AndroidTest
+#ifdef __cplusplus
+extern "C" {
+#endif
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: NativeInit
+ * Signature: ()Z
+ */
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NativeInit
+ (JNIEnv *, jclass);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: RunAutoTest
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_RunAutoTest
+ (JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: Create
+ * Signature: ()Z
+ */
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Create
+ (JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: Delete
+ * Signature: ()Z
+ */
+JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Delete
+ (JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: Init
+ * Signature: (IIIZZ)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Init
+ (JNIEnv *, jobject, jboolean, jboolean);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: Terminate
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Terminate
+ (JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: CreateChannel
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_CreateChannel
+ (JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: DeleteChannel
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_DeleteChannel
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetLocalReceiver
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLocalReceiver
+ (JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetSendDestination
+ * Signature: (IILjava/lang/String;)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendDestination
+ (JNIEnv *, jobject, jint, jint, jstring);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartListen
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartListen
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartPlayout
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayout
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartSend
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartSend
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopListen
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopListen
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopPlayout
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayout
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopSend
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopSend
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartPlayingFileLocally
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileLocally
+ (JNIEnv *, jobject, jint, jstring, jboolean);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopPlayingFileLocally
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileLocally
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartRecordingPlayout
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartRecordingPlayout
+ (JNIEnv *, jobject, jint, jstring, jboolean);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopRecordingPlayout
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopRecordingPlayout
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StartPlayingFileAsMicrophone
+ * Signature: (ILjava/lang/String;Z)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileAsMicrophone
+ (JNIEnv *, jobject, jint, jstring, jboolean);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: StopPlayingFileAsMicrophone
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileAsMicrophone
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: NumOfCodecs
+ * Signature: ()I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NumOfCodecs
+ (JNIEnv *, jobject);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetSendCodec
+ * Signature: (II)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendCodec
+ (JNIEnv *, jobject, jint, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetVADStatus
+ * Signature: (IZI)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetVADStatus
+ (JNIEnv *, jobject, jint, jboolean, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetNSStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetNSStatus
+ (JNIEnv *, jobject, jboolean, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetAGCStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetAGCStatus
+ (JNIEnv *, jobject, jboolean, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetECStatus
+ * Signature: (ZI)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus
+ (JNIEnv *, jobject, jboolean, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetSpeakerVolume
+ * Signature: (I)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume
+ (JNIEnv *, jobject, jint);
+
+/*
+ * Class: org_webrtc_voiceengine_test_AndroidTest
+ * Method: SetLoudspeakerStatus
+ * Signature: (Z)I
+ */
+JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus
+ (JNIEnv *, jobject, jboolean);
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png b/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png
new file mode 100644
index 0000000..a07c69f
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml b/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml
new file mode 100644
index 0000000..4165a07
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml
@@ -0,0 +1,43 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
+<!-- -->
+<!-- Use of this source code is governed by a BSD-style license -->
+<!-- that can be found in the LICENSE file in the root of the source -->
+<!-- tree. An additional intellectual property rights grant can be found -->
+<!-- in the file PATENTS. All contributing project authors may -->
+<!-- be found in the AUTHORS file in the root of the source tree. -->
+
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ android:orientation="vertical"
+ android:layout_width="fill_parent"
+ android:layout_height="fill_parent">
+
+ <TextView android:text="@+id/TextView01"
+ android:id="@+id/TextView01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </TextView>
+ <EditText android:text="@+id/EditText01"
+ android:id="@+id/EditText01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </EditText>
+ <Button android:text="@+id/Button01"
+ android:id="@+id/Button01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+ <Spinner android:id="@+id/Spinner01"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Spinner>
+ <Spinner android:id="@+id/Spinner02"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Spinner>
+ <Button android:text="@+id/Button02"
+ android:id="@+id/Button02"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content">
+ </Button>
+</LinearLayout>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml b/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml
new file mode 100644
index 0000000..29ec4ee
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml
@@ -0,0 +1,13 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
+<!-- -->
+<!-- Use of this source code is governed by a BSD-style license -->
+<!-- that can be found in the LICENSE file in the root of the source -->
+<!-- tree. An additional intellectual property rights grant can be found -->
+<!-- in the file PATENTS. All contributing project authors may -->
+<!-- be found in the AUTHORS file in the root of the source tree. -->
+
+<resources>
+
+ <string name="app_name">WebRtc VoE</string>
+</resources>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java b/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java
new file mode 100644
index 0000000..71b22b0
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java
@@ -0,0 +1,1190 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license that can be found
+ * in the LICENSE file in the root of the source tree. An additional
+ * intellectual property rights grant can be found in the file PATENTS. All
+ * contributing project authors may be found in the AUTHORS file in the root of
+ * the source tree.
+ */
+
+/*
+ * VoiceEngine Android test application. It starts either auto test or acts like
+ * a GUI test.
+ */
+
+package org.webrtc.voiceengine.test;
+
+import java.io.File;
+import java.io.FileInputStream;
+import java.io.FileNotFoundException;
+import java.io.FileOutputStream;
+import java.io.FileReader;
+import java.io.IOException;
+
+import android.app.Activity;
+import android.content.Context;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.media.MediaRecorder;
+import android.os.Bundle;
+import android.util.Log;
+import android.view.View;
+import android.widget.AdapterView;
+import android.widget.ArrayAdapter;
+import android.widget.Button;
+import android.widget.EditText;
+import android.widget.Spinner;
+import android.widget.TextView;
+
+public class AndroidTest extends Activity {
+ private byte[] _playBuffer = null;
+ private short[] _circBuffer = new short[8000]; // can hold 50 frames
+
+ private int _recIndex = 0;
+ private int _playIndex = 0;
+ // private int _streamVolume = 4;
+ private int _maxVolume = 0; // Android max level (commonly 5)
+ // VoE level (0-255), corresponds to level 4 out of 5
+ private int _volumeLevel = 204;
+
+ private Thread _playThread;
+ private Thread _recThread;
+ private Thread _autotestThread;
+
+ private static AudioTrack _at;
+ private static AudioRecord _ar;
+
+ private File _fr = null;
+ private FileInputStream _in = null;
+
+ private boolean _isRunningPlay = false;
+ private boolean _isRunningRec = false;
+ private boolean _settingSet = true;
+ private boolean _isCallActive = false;
+ private boolean _runAutotest = false; // ENABLE AUTOTEST HERE!
+
+ private int _channel = -1;
+ private int _codecIndex = 0;
+ private int _ecIndex = 0;
+ private int _nsIndex = 0;
+ private int _agcIndex = 0;
+ private int _vadIndex = 0;
+ private int _audioIndex = 3;
+ private int _settingMenu = 0;
+ private int _receivePort = 1234;
+ private int _destinationPort = 1234;
+ private String _destinationIP = "127.0.0.1";
+
+ // "Build" settings
+ private final boolean _playFromFile = false;
+ // Set to true to send data to native code and back
+ private final boolean _runThroughNativeLayer = true;
+ private final boolean enableSend = true;
+ private final boolean enableReceive = true;
+ private final boolean useNativeThread = false;
+
+ /** Called when the activity is first created. */
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ setContentView(R.layout.main);
+
+ TextView tv = (TextView) findViewById(R.id.TextView01);
+ tv.setText("");
+
+ final EditText ed = (EditText) findViewById(R.id.EditText01);
+ ed.setWidth(200);
+ ed.setText(_destinationIP);
+
+ final Button buttonStart = (Button) findViewById(R.id.Button01);
+ buttonStart.setWidth(200);
+ if (_runAutotest) {
+ buttonStart.setText("Run test");
+ } else {
+ buttonStart.setText("Start Call");
+ }
+ // button.layout(50, 50, 100, 40);
+ buttonStart.setOnClickListener(new View.OnClickListener() {
+ public void onClick(View v) {
+
+ if (_runAutotest) {
+ startAutoTest();
+ } else {
+ if (_isCallActive) {
+
+ if (stopCall() != -1) {
+ _isCallActive = false;
+ buttonStart.setText("Start Call");
+ }
+ } else {
+
+ _destinationIP = ed.getText().toString();
+ if (startCall() != -1) {
+ _isCallActive = true;
+ buttonStart.setText("Stop Call");
+ }
+ }
+ }
+
+ // displayTextFromFile();
+ // recordAudioToFile();
+ // if(!_playFromFile)
+ // {
+ // recAudioInThread();
+ // }
+ // playAudioInThread();
+ }
+ });
+
+ final Button buttonStop = (Button) findViewById(R.id.Button02);
+ buttonStop.setWidth(200);
+ buttonStop.setText("Close app");
+ buttonStop.setOnClickListener(new View.OnClickListener() {
+ public void onClick(View v) {
+
+ if (!_runAutotest) {
+ ShutdownVoE();
+ }
+
+ // This call terminates and should close the activity
+ finish();
+
+ // playAudioFromFile();
+ // if(!_playFromFile)
+ // {
+ // stopRecAudio();
+ // }
+ // stopPlayAudio();
+ }
+ });
+
+
+ String ap1[] = {"EC off", "AECM"};
+ final ArrayAdapter<String> adapterAp1 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ ap1);
+ String ap2[] =
+ {"NS off", "NS low", "NS moderate", "NS high",
+ "NS very high"};
+ final ArrayAdapter<String> adapterAp2 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ ap2);
+ String ap3[] = {"AGC off", "AGC adaptive", "AGC fixed"};
+ final ArrayAdapter<String> adapterAp3 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ ap3);
+ String ap4[] =
+ {"VAD off", "VAD conventional", "VAD high rate",
+ "VAD mid rate", "VAD low rate"};
+ final ArrayAdapter<String> adapterAp4 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ ap4);
+ String codecs[] = {"iSAC", "PCMU", "PCMA", "iLBC"};
+ final ArrayAdapter<String> adapterCodecs = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ codecs);
+
+ final Spinner spinnerSettings1 = (Spinner) findViewById(R.id.Spinner01);
+ final Spinner spinnerSettings2 = (Spinner) findViewById(R.id.Spinner02);
+ spinnerSettings1.setMinimumWidth(200);
+ String settings[] =
+ {"Codec", "Echo Control", "Noise Suppression",
+ "Automatic Gain Control",
+ "Voice Activity Detection"};
+ ArrayAdapter<String> adapterSettings1 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ settings);
+ spinnerSettings1.setAdapter(adapterSettings1);
+ spinnerSettings1.setOnItemSelectedListener(
+ new AdapterView.OnItemSelectedListener() {
+ public void onItemSelected(AdapterView adapterView, View view,
+ int position, long id) {
+
+ _settingMenu = position;
+ _settingSet = false;
+ if (position == 0) {
+ spinnerSettings2.setAdapter(adapterCodecs);
+ spinnerSettings2.setSelection(_codecIndex);
+ }
+ if (position == 1) {
+ spinnerSettings2.setAdapter(adapterAp1);
+ spinnerSettings2.setSelection(_ecIndex);
+ }
+ if (position == 2) {
+ spinnerSettings2.setAdapter(adapterAp2);
+ spinnerSettings2.setSelection(_nsIndex);
+ }
+ if (position == 3) {
+ spinnerSettings2.setAdapter(adapterAp3);
+ spinnerSettings2.setSelection(_agcIndex);
+ }
+ if (position == 4) {
+ spinnerSettings2.setAdapter(adapterAp4);
+ spinnerSettings2.setSelection(_vadIndex);
+ }
+ }
+
+ public void onNothingSelected(AdapterView adapterView) {
+ WebrtcLog("No setting1 selected");
+ }
+ });
+
+ spinnerSettings2.setMinimumWidth(200);
+ ArrayAdapter<String> adapterSettings2 = new ArrayAdapter<String>(
+ this,
+ android.R.layout.simple_spinner_dropdown_item,
+ codecs);
+ spinnerSettings2.setAdapter(adapterSettings2);
+ spinnerSettings2.setOnItemSelectedListener(
+ new AdapterView.OnItemSelectedListener() {
+ public void onItemSelected(AdapterView adapterView, View view,
+ int position, long id) {
+
+ // avoid unintentional setting
+ if (_settingSet == false) {
+ _settingSet = true;
+ return;
+ }
+
+ // Change volume
+ if (_settingMenu == 0) {
+ WebrtcLog("Selected audio " + position);
+ setAudioProperties(position);
+ spinnerSettings2.setSelection(_audioIndex);
+ }
+
+ // Change codec
+ if (_settingMenu == 1) {
+ _codecIndex = position;
+ WebrtcLog("Selected codec " + position);
+ if (0 != SetSendCodec(_channel, _codecIndex)) {
+ WebrtcLog("VoE set send codec failed");
+ }
+ }
+
+ // Change EC
+ if (_settingMenu == 2) {
+ boolean enable = true;
+ int ECmode = 5; // AECM
+ int AESmode = 0;
+
+ _ecIndex = position;
+ WebrtcLog("Selected EC " + position);
+
+ if (position == 0) {
+ enable = false;
+ }
+ if (position > 1) {
+ ECmode = 4; // AES
+ AESmode = position - 1;
+ }
+
+ if (0 != SetECStatus(enable, ECmode)) {
+ WebrtcLog("VoE set EC status failed");
+ }
+ }
+
+ // Change NS
+ if (_settingMenu == 3) {
+ boolean enable = true;
+
+ _nsIndex = position;
+ WebrtcLog("Selected NS " + position);
+
+ if (position == 0) {
+ enable = false;
+ }
+ if (0 != SetNSStatus(enable, position + 2)) {
+ WebrtcLog("VoE set NS status failed");
+ }
+ }
+
+ // Change AGC
+ if (_settingMenu == 4) {
+ boolean enable = true;
+
+ _agcIndex = position;
+ WebrtcLog("Selected AGC " + position);
+
+ if (position == 0) {
+ enable = false;
+ position = 1; // default
+ }
+ if (0 != SetAGCStatus(enable, position + 2)) {
+ WebrtcLog("VoE set AGC status failed");
+ }
+ }
+
+ // Change VAD
+ if (_settingMenu == 5) {
+ boolean enable = true;
+
+ _vadIndex = position;
+ WebrtcLog("Selected VAD " + position);
+
+ if (position == 0) {
+ enable = false;
+ position++;
+ }
+ if (0 != SetVADStatus(_channel, enable, position - 1)) {
+ WebrtcLog("VoE set VAD status failed");
+ }
+ }
+ }
+
+ public void onNothingSelected(AdapterView adapterView) {
+ }
+ });
+
+ // Setup VoiceEngine
+ if (!_runAutotest && !useNativeThread) SetupVoE();
+
+ // Suggest to use the voice call audio stream for hardware volume
+ // controls
+ setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
+
+ // Get max Android volume and adjust default volume to map exactly to an
+ // Android level
+ AudioManager am =
+ (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+ _maxVolume = am.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
+ if (_maxVolume <= 0) {
+ WebrtcLog("Could not get max volume!");
+ } else {
+ int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
+ _volumeLevel = (androidVolumeLevel * 255) / _maxVolume;
+ }
+
+ WebrtcLog("Started Webrtc Android Test");
+ }
+
+ // Will be called when activity is shutdown.
+ // NOTE: Activity may be killed without this function being called,
+ // but then we should not need to clean up.
+ protected void onDestroy() {
+ super.onDestroy();
+ // ShutdownVoE();
+ }
+
+ private void SetupVoE() {
+ // Create VoiceEngine
+ Create(); // Error logging is done in native API wrapper
+
+ // Initialize
+ if (0 != Init(false, false)) {
+ WebrtcLog("VoE init failed");
+ }
+
+ // Create channel
+ _channel = CreateChannel();
+ if (0 != _channel) {
+ WebrtcLog("VoE create channel failed");
+ }
+
+ }
+
+ private void ShutdownVoE() {
+ // Delete channel
+ if (0 != DeleteChannel(_channel)) {
+ WebrtcLog("VoE delete channel failed");
+ }
+
+ // Terminate
+ if (0 != Terminate()) {
+ WebrtcLog("VoE terminate failed");
+ }
+
+ // Delete VoiceEngine
+ Delete(); // Error logging is done in native API wrapper
+ }
+
+ int startCall() {
+
+ if (useNativeThread == true) {
+
+ Create();
+ return 0;
+ }
+
+ if (enableReceive == true) {
+ // Set local receiver
+ if (0 != SetLocalReceiver(_channel, _receivePort)) {
+ WebrtcLog("VoE set local receiver failed");
+ }
+
+ if (0 != StartListen(_channel)) {
+ WebrtcLog("VoE start listen failed");
+ return -1;
+ }
+
+ // Route audio to earpiece
+ if (0 != SetLoudspeakerStatus(false)) {
+ WebrtcLog("VoE set louspeaker status failed");
+ return -1;
+ }
+
+ /*
+ * WebrtcLog("VoE start record now"); if (0 !=
+ * StartRecordingPlayout(_channel, "/sdcard/singleUserDemoOut.pcm",
+ * false)) { WebrtcLog("VoE Recording Playout failed"); }
+ * WebrtcLog("VoE start Recording Playout end");
+ */
+ // Start playout
+ if (0 != StartPlayout(_channel)) {
+ WebrtcLog("VoE start playout failed");
+ return -1;
+ }
+
+ // Start playout file
+ // if (0 != StartPlayingFileLocally(_channel,
+ // "/sdcard/singleUserDemo.pcm", true)) {
+ // WebrtcLog("VoE start playout file failed");
+ // return -1;
+ // }
+ }
+
+ if (enableSend == true) {
+ if (0 != SetSendDestination(_channel, _destinationPort,
+ _destinationIP)) {
+ WebrtcLog("VoE set send destination failed");
+ return -1;
+ }
+
+ if (0 != SetSendCodec(_channel, _codecIndex)) {
+ WebrtcLog("VoE set send codec failed");
+ return -1;
+ }
+
+ /*
+ * if (0 != StartPlayingFileAsMicrophone(_channel,
+ * "/sdcard/singleUserDemo.pcm", true)) {
+ * WebrtcLog("VoE start playing file as microphone failed"); }
+ */
+ if (0 != StartSend(_channel)) {
+ WebrtcLog("VoE start send failed");
+ return -1;
+ }
+
+ // if (0 != StartPlayingFileAsMicrophone(_channel,
+ // "/sdcard/singleUserDemo.pcm", true)) {
+ // WebrtcLog("VoE start playing file as microphone failed");
+ // return -1;
+ // }
+ }
+
+ return 0;
+ }
+
+ int stopCall() {
+
+ if (useNativeThread == true) {
+
+ Delete();
+ return 0;
+ }
+
+ if (enableSend == true) {
+ // Stop playing file as microphone
+ /*
+ * if (0 != StopPlayingFileAsMicrophone(_channel)) {
+ * WebrtcLog("VoE stop playing file as microphone failed"); return
+ * -1; }
+ */
+ // Stop send
+ if (0 != StopSend(_channel)) {
+ WebrtcLog("VoE stop send failed");
+ return -1;
+ }
+ }
+
+ if (enableReceive == true) {
+ // if (0 != StopRecordingPlayout(_channel)) {
+ // WebrtcLog("VoE stop Recording Playout failed");
+ // }
+ // WebrtcLog("VoE stop Recording Playout ended");
+
+ // Stop listen
+ if (0 != StopListen(_channel)) {
+ WebrtcLog("VoE stop listen failed");
+ return -1;
+ }
+
+ // Stop playout file
+ // if (0 != StopPlayingFileLocally(_channel)) {
+ // WebrtcLog("VoE stop playout file failed");
+ // return -1;
+ // }
+
+ // Stop playout
+ if (0 != StopPlayout(_channel)) {
+ WebrtcLog("VoE stop playout failed");
+ return -1;
+ }
+
+ // Route audio to loudspeaker
+ if (0 != SetLoudspeakerStatus(true)) {
+ WebrtcLog("VoE set louspeaker status failed");
+ return -1;
+ }
+ }
+
+ return 0;
+ }
+
+ int startAutoTest() {
+
+ _autotestThread = new Thread(_autotestProc);
+ _autotestThread.start();
+
+ return 0;
+ }
+
+ private Runnable _autotestProc = new Runnable() {
+ public void run() {
+ // TODO(xians): choose test from GUI
+ // 1 = standard, not used
+ // 2 = extended, 2 = base
+ RunAutoTest(1, 2);
+ }
+ };
+
+ int setAudioProperties(int val) {
+
+ // AudioManager am = (AudioManager)
+ // getSystemService(Context.AUDIO_SERVICE);
+
+ if (val == 0) {
+ // _streamVolume =
+ // am.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
+ // am.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
+ // (_streamVolume+1), 0);
+
+ int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
+ if (androidVolumeLevel < _maxVolume) {
+ _volumeLevel = ((androidVolumeLevel + 1) * 255) / _maxVolume;
+ if (0 != SetSpeakerVolume(_volumeLevel)) {
+ WebrtcLog("VoE set speaker volume failed");
+ }
+ }
+ } else if (val == 1) {
+ // _streamVolume =
+ // am.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
+ // am.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
+ // (_streamVolume-1), 0);
+
+ int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
+ if (androidVolumeLevel > 0) {
+ _volumeLevel = ((androidVolumeLevel - 1) * 255) / _maxVolume;
+ if (0 != SetSpeakerVolume(_volumeLevel)) {
+ WebrtcLog("VoE set speaker volume failed");
+ }
+ }
+ } else if (val == 2) {
+ // route audio to back speaker
+ if (0 != SetLoudspeakerStatus(true)) {
+ WebrtcLog("VoE set loudspeaker status failed");
+ }
+ _audioIndex = 2;
+ } else if (val == 3) {
+ // route audio to earpiece
+ if (0 != SetLoudspeakerStatus(false)) {
+ WebrtcLog("VoE set loudspeaker status failed");
+ }
+ _audioIndex = 3;
+ }
+
+ return 0;
+ }
+
+ int displayTextFromFile() {
+
+ TextView tv = (TextView) findViewById(R.id.TextView01);
+ FileReader fr = null;
+ char[] fileBuffer = new char[64];
+
+ try {
+ fr = new FileReader("/sdcard/test.txt");
+ } catch (FileNotFoundException e) {
+ e.printStackTrace();
+ tv.setText("File not found!");
+ }
+
+ try {
+ fr.read(fileBuffer);
+ } catch (IOException e) {
+ e.printStackTrace();
+ }
+
+ String readString = new String(fileBuffer);
+ tv.setText(readString);
+ // setContentView(tv);
+
+ return 0;
+ }
+
+ int recordAudioToFile() {
+ File fr = null;
+ // final to be reachable within onPeriodicNotification
+ byte[] recBuffer = new byte[320];
+
+ int recBufSize =
+ AudioRecord.getMinBufferSize(16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+ AudioRecord rec =
+ new AudioRecord(MediaRecorder.AudioSource.MIC, 16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ recBufSize);
+
+ fr = new File("/sdcard/record.pcm");
+ FileOutputStream out = null;
+ try {
+ out = new FileOutputStream(fr);
+ } catch (FileNotFoundException e1) {
+ e1.printStackTrace();
+ }
+
+ // start recording
+ try {
+ rec.startRecording();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ for (int i = 0; i < 550; i++) {
+ // note, there is a short version of write as well!
+ int wrBytes = rec.read(recBuffer, 0, 320);
+
+ try {
+ out.write(recBuffer);
+ } catch (IOException e) {
+ e.printStackTrace();
+ }
+ }
+
+ // stop playout
+ try {
+ rec.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ return 0;
+ }
+
+ int playAudioFromFile() {
+
+ File fr = null;
+ // final to be reachable within onPeriodicNotification
+ // final byte[] playBuffer = new byte [320000];
+ // final to be reachable within onPeriodicNotification
+ final byte[] playBuffer = new byte[320];
+
+ final int playBufSize =
+ AudioTrack.getMinBufferSize(16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+ // final int playBufSize = 1920; // 100 ms buffer
+ // byte[] playBuffer = new byte [playBufSize];
+ final AudioTrack play =
+ new AudioTrack(AudioManager.STREAM_VOICE_CALL, 16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ playBufSize, AudioTrack.MODE_STREAM);
+
+ // implementation of the playpos callback functions
+ play.setPlaybackPositionUpdateListener(
+ new AudioTrack.OnPlaybackPositionUpdateListener() {
+
+ int count = 0;
+
+ public void onPeriodicNotification(AudioTrack track) {
+ // int wrBytes = play.write(playBuffer, count, 320);
+ count += 320;
+ }
+
+ public void onMarkerReached(AudioTrack track) {
+
+ }
+ });
+
+ // set the notification period = 160 samples
+ // int ret = play.setPositionNotificationPeriod(160);
+
+ fr = new File("/sdcard/record.pcm");
+ FileInputStream in = null;
+ try {
+ in = new FileInputStream(fr);
+ } catch (FileNotFoundException e1) {
+ e1.printStackTrace();
+ }
+
+ // try {
+ // in.read(playBuffer);
+ // } catch (IOException e) {
+ // e.printStackTrace();
+ // }
+
+ // play all at once
+ // int wrBytes = play.write(playBuffer, 0, 320000);
+
+
+ // start playout
+ try {
+ play.play();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ // returns the number of samples that has been written
+ // int headPos = play.getPlaybackHeadPosition();
+
+ // play with multiple writes
+ for (int i = 0; i < 500; i++) {
+ try {
+ in.read(playBuffer);
+ } catch (IOException e) {
+ e.printStackTrace();
+ }
+
+
+ // note, there is a short version of write as well!
+ int wrBytes = play.write(playBuffer, 0, 320);
+
+ Log.d("testWrite", "wrote");
+ }
+
+ // stop playout
+ try {
+ play.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ return 0;
+ }
+
+ int playAudioInThread() {
+
+ if (_isRunningPlay) {
+ return 0;
+ }
+
+ // File fr = null;
+ // final byte[] playBuffer = new byte[320];
+ if (_playFromFile) {
+ _playBuffer = new byte[320];
+ } else {
+ // reset index
+ _playIndex = 0;
+ }
+ // within
+ // onPeriodicNotification
+
+ // Log some info (static)
+ WebrtcLog("Creating AudioTrack object");
+ final int minPlayBufSize =
+ AudioTrack.getMinBufferSize(16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+ WebrtcLog("Min play buf size = " + minPlayBufSize);
+ WebrtcLog("Min volume = " + AudioTrack.getMinVolume());
+ WebrtcLog("Max volume = " + AudioTrack.getMaxVolume());
+ WebrtcLog("Native sample rate = "
+ + AudioTrack.getNativeOutputSampleRate(
+ AudioManager.STREAM_VOICE_CALL));
+
+ final int playBufSize = minPlayBufSize; // 3200; // 100 ms buffer
+ // byte[] playBuffer = new byte [playBufSize];
+ try {
+ _at = new AudioTrack(
+ AudioManager.STREAM_VOICE_CALL,
+ 16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ playBufSize, AudioTrack.MODE_STREAM);
+ } catch (Exception e) {
+ WebrtcLog(e.getMessage());
+ }
+
+ // Log some info (non-static)
+ WebrtcLog("Notification marker pos = "
+ + _at.getNotificationMarkerPosition());
+ WebrtcLog("Play head pos = " + _at.getPlaybackHeadPosition());
+ WebrtcLog("Pos notification dt = "
+ + _at.getPositionNotificationPeriod());
+ WebrtcLog("Playback rate = " + _at.getPlaybackRate());
+ WebrtcLog("Sample rate = " + _at.getSampleRate());
+
+ // implementation of the playpos callback functions
+ // _at.setPlaybackPositionUpdateListener(
+ // new AudioTrack.OnPlaybackPositionUpdateListener() {
+ //
+ // int count = 3200;
+ //
+ // public void onPeriodicNotification(AudioTrack track) {
+ // // int wrBytes = play.write(playBuffer, count, 320);
+ // count += 320;
+ // }
+ //
+ // public void onMarkerReached(AudioTrack track) {
+ // }
+ // });
+
+ // set the notification period = 160 samples
+ // int ret = _at.setPositionNotificationPeriod(160);
+
+ if (_playFromFile) {
+ _fr = new File("/sdcard/singleUserDemo.pcm");
+ try {
+ _in = new FileInputStream(_fr);
+ } catch (FileNotFoundException e1) {
+ e1.printStackTrace();
+ }
+ }
+
+ // try {
+ // in.read(playBuffer);
+ // } catch (IOException e) {
+ // e.printStackTrace();
+ // }
+
+ _isRunningPlay = true;
+
+ // buffer = new byte[3200];
+ _playThread = new Thread(_playProc);
+ // ar.startRecording();
+ // bytesRead = 3200;
+ // recording = true;
+ _playThread.start();
+
+ return 0;
+ }
+
+ int stopPlayAudio() {
+ if (!_isRunningPlay) {
+ return 0;
+ }
+
+ _isRunningPlay = false;
+
+ return 0;
+ }
+
+ private Runnable _playProc = new Runnable() {
+ public void run() {
+
+ // set high thread priority
+ android.os.Process.setThreadPriority(
+ android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+
+ // play all at once
+ // int wrBytes = play.write(playBuffer, 0, 320000);
+
+ // fill the buffer
+ // play.write(playBuffer, 0, 3200);
+
+ // play.flush();
+
+ // start playout
+ try {
+ _at.play();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ // play with multiple writes
+ int i = 0;
+ for (; i < 3000 && _isRunningPlay; i++) {
+
+ if (_playFromFile) {
+ try {
+ _in.read(_playBuffer);
+ } catch (IOException e) {
+ e.printStackTrace();
+ }
+
+ int wrBytes = _at.write(_playBuffer, 0 /* i * 320 */, 320);
+ } else {
+ int wrSamples =
+ _at.write(_circBuffer, _playIndex * 160,
+ 160);
+
+ // WebrtcLog("Played 10 ms from buffer, _playIndex = " +
+ // _playIndex);
+ // WebrtcLog("Diff = " + (_recIndex - _playIndex));
+
+ if (_playIndex == 49) {
+ _playIndex = 0;
+ } else {
+ _playIndex += 1;
+ }
+ }
+
+ // WebrtcLog("Wrote 10 ms to buffer, head = "
+ // + _at.getPlaybackHeadPosition());
+ }
+
+ // stop playout
+ try {
+ _at.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ // returns the number of samples that has been written
+ WebrtcLog("Test stopped, i = " + i + ", head = "
+ + _at.getPlaybackHeadPosition());
+ int headPos = _at.getPlaybackHeadPosition();
+
+ // flush the buffers
+ _at.flush();
+
+ // release the object
+ _at.release();
+ _at = null;
+
+ // try {
+ // Thread.sleep() must be within a try - catch block
+ // Thread.sleep(3000);
+ // }catch (Exception e){
+ // System.out.println(e.getMessage());
+ // }
+
+ _isRunningPlay = false;
+
+ }
+ };
+
+ int recAudioInThread() {
+
+ if (_isRunningRec) {
+ return 0;
+ }
+
+ // within
+ // onPeriodicNotification
+
+ // reset index
+ _recIndex = 20;
+
+ // Log some info (static)
+ WebrtcLog("Creating AudioRecord object");
+ final int minRecBufSize = AudioRecord.getMinBufferSize(16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT);
+ WebrtcLog("Min rec buf size = " + minRecBufSize);
+ // WebrtcLog("Min volume = " + AudioTrack.getMinVolume());
+ // WebrtcLog("Max volume = " + AudioTrack.getMaxVolume());
+ // WebrtcLog("Native sample rate = "
+ // + AudioRecord
+ // .getNativeInputSampleRate(AudioManager.STREAM_VOICE_CALL));
+
+ final int recBufSize = minRecBufSize; // 3200; // 100 ms buffer
+ try {
+ _ar = new AudioRecord(
+ MediaRecorder.AudioSource.MIC,
+ 16000,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT,
+ recBufSize);
+ } catch (Exception e) {
+ WebrtcLog(e.getMessage());
+ }
+
+ // Log some info (non-static)
+ WebrtcLog("Notification marker pos = "
+ + _ar.getNotificationMarkerPosition());
+ // WebrtcLog("Play head pos = " + _ar.getRecordHeadPosition());
+ WebrtcLog("Pos notification dt rec= "
+ + _ar.getPositionNotificationPeriod());
+ // WebrtcLog("Playback rate = " + _ar.getRecordRate());
+ // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
+ WebrtcLog("Sample rate = " + _ar.getSampleRate());
+ // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
+ // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
+
+ _isRunningRec = true;
+
+ _recThread = new Thread(_recProc);
+
+ _recThread.start();
+
+ return 0;
+ }
+
+ int stopRecAudio() {
+ if (!_isRunningRec) {
+ return 0;
+ }
+
+ _isRunningRec = false;
+
+ return 0;
+ }
+
+ private Runnable _recProc = new Runnable() {
+ public void run() {
+
+ // set high thread priority
+ android.os.Process.setThreadPriority(
+ android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+
+ // start recording
+ try {
+ _ar.startRecording();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ // keep recording to circular buffer
+ // for a while
+ int i = 0;
+ int rdSamples = 0;
+ short[] tempBuffer = new short[160]; // Only used for native case
+
+ for (; i < 3000 && _isRunningRec; i++) {
+ if (_runThroughNativeLayer) {
+ rdSamples = _ar.read(tempBuffer, 0, 160);
+ // audioLoop(tempBuffer, 160); // Insert into native layer
+ } else {
+ rdSamples = _ar.read(_circBuffer, _recIndex * 160, 160);
+
+ // WebrtcLog("Recorded 10 ms to buffer, _recIndex = " +
+ // _recIndex);
+ // WebrtcLog("rdSamples = " + rdSamples);
+
+ if (_recIndex == 49) {
+ _recIndex = 0;
+ } else {
+ _recIndex += 1;
+ }
+ }
+ }
+
+ // stop recording
+ try {
+ _ar.stop();
+ } catch (IllegalStateException e) {
+ e.printStackTrace();
+ }
+
+ // release the object
+ _ar.release();
+ _ar = null;
+
+ // try {
+ // Thread.sleep() must be within a try - catch block
+ // Thread.sleep(3000);
+ // }catch (Exception e){
+ // System.out.println(e.getMessage());
+ // }
+
+ _isRunningRec = false;
+
+ // returns the number of samples that has been written
+ // WebrtcLog("Test stopped, i = " + i + ", head = "
+ // + _at.getPlaybackHeadPosition());
+ // int headPos = _at.getPlaybackHeadPosition();
+ }
+ };
+
+ private void WebrtcLog(String msg) {
+ Log.d("*Webrtc*", msg);
+ }
+
+ // //////////////// Native function prototypes ////////////////////
+
+ private native static boolean NativeInit();
+
+ private native int RunAutoTest(int testType, int extendedSel);
+
+ private native boolean Create();
+
+ private native boolean Delete();
+
+ private native int Init(boolean enableTrace, boolean useExtTrans);
+
+ private native int Terminate();
+
+ private native int CreateChannel();
+
+ private native int DeleteChannel(int channel);
+
+ private native int SetLocalReceiver(int channel, int port);
+
+ private native int SetSendDestination(int channel, int port,
+ String ipaddr);
+
+ private native int StartListen(int channel);
+
+ private native int StartPlayout(int channel);
+
+ private native int StartSend(int channel);
+
+ private native int StopListen(int channel);
+
+ private native int StopPlayout(int channel);
+
+ private native int StopSend(int channel);
+
+ private native int StartPlayingFileLocally(int channel, String fileName,
+ boolean loop);
+
+ private native int StopPlayingFileLocally(int channel);
+
+ private native int StartRecordingPlayout(int channel, String fileName,
+ boolean loop);
+
+ private native int StopRecordingPlayout(int channel);
+
+ private native int StartPlayingFileAsMicrophone(int channel,
+ String fileName, boolean loop);
+
+ private native int StopPlayingFileAsMicrophone(int channel);
+
+ private native int NumOfCodecs();
+
+ private native int SetSendCodec(int channel, int index);
+
+ private native int SetVADStatus(int channel, boolean enable, int mode);
+
+ private native int SetNSStatus(boolean enable, int mode);
+
+ private native int SetAGCStatus(boolean enable, int mode);
+
+ private native int SetECStatus(boolean enable, int mode);
+
+ private native int SetSpeakerVolume(int volume);
+
+ private native int SetLoudspeakerStatus(boolean enable);
+
+ /*
+ * this is used to load the 'webrtc-voice-demo-jni'
+ * library on application startup.
+ * The library has already been unpacked into
+ * /data/data/webrtc.android.AndroidTest/lib/libwebrtc-voice-demo-jni.so
+ * at installation time by the package manager.
+ */
+ static {
+ Log.d("*Webrtc*", "Loading webrtc-voice-demo-jni...");
+ System.loadLibrary("webrtc-voice-demo-jni");
+
+ Log.d("*Webrtc*", "Calling native init...");
+ if (!NativeInit()) {
+ Log.e("*Webrtc*", "Native init failed");
+ throw new RuntimeException("Native init failed");
+ } else {
+ Log.d("*Webrtc*", "Native init successful");
+ }
+ }
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/Android.mk b/trunk/src/voice_engine/main/test/auto_test/Android.mk
new file mode 100644
index 0000000..cfb9d67
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/Android.mk
@@ -0,0 +1,56 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+# voice engine test app
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ automated_mode.cc \
+ voe_cpu_test.cc \
+ voe_standard_test.cc \
+ voe_stress_test.cc \
+ voe_unit_test.cc \
+ voe_extended_test.cc \
+ voe_standard_integration_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../modules/audio_device/main/interface \
+ $(LOCAL_PATH)/../../../../modules/interface \
+ $(LOCAL_PATH)/../../../../system_wrappers/interface \
+ $(LOCAL_PATH)/../../../../../test \
+ external/gtest/include \
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libwebrtc
+
+LOCAL_MODULE:= webrtc_voe_autotest
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
+
diff --git a/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc b/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc
new file mode 100644
index 0000000..13fa257
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+int RunInAutomatedMode(int argc, char** argv) {
+ testing::InitGoogleTest(&argc, argv);
+
+ return RUN_ALL_TESTS();
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/automated_mode.h b/trunk/src/voice_engine/main/test/auto_test/automated_mode.h
new file mode 100644
index 0000000..cd7ab9e
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/automated_mode.h
@@ -0,0 +1,16 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
+#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
+
+int RunInAutomatedMode(int argc, char** argv);
+
+#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc
new file mode 100644
index 0000000..f0e665b
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_initialization_fixture.h"
+
+class TestErrorObserver : public webrtc::VoiceEngineObserver {
+ public:
+ TestErrorObserver() {}
+ virtual ~TestErrorObserver() {}
+ void CallbackOnError(const int channel, const int error_code) {
+ ADD_FAILURE() << "Unexpected error on channel " << channel <<
+ ": error code " << error_code;
+ }
+};
+
+AfterInitializationFixture::AfterInitializationFixture()
+ : error_observer_(new TestErrorObserver()) {
+ EXPECT_EQ(0, voe_base_->Init());
+
+#if defined(WEBRTC_ANDROID)
+ EXPECT_EQ(0, voe_hardware_->SetLoudspeakerStatus(false));
+#endif
+
+ EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(*error_observer_));
+}
+
+AfterInitializationFixture::~AfterInitializationFixture() {
+ EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h
new file mode 100644
index 0000000..bbdd64d
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
+#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
+
+#include "before_initialization_fixture.h"
+#include "scoped_ptr.h"
+
+class TestErrorObserver;
+
+// This fixture initializes the voice engine in addition to the work
+// done by the before-initialization fixture. It also registers an error
+// observer which will fail tests on error callbacks. This fixture is
+// useful to tests that want to run before we have started any form of
+// streaming through the voice engine.
+class AfterInitializationFixture : public BeforeInitializationFixture {
+ public:
+ AfterInitializationFixture();
+ virtual ~AfterInitializationFixture();
+ protected:
+ webrtc::scoped_ptr<TestErrorObserver> error_observer_;
+};
+
+#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc
new file mode 100644
index 0000000..e1a4820
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+
+#include <cstring>
+
+static const char* kLoopbackIp = "127.0.0.1";
+
+AfterStreamingFixture::AfterStreamingFixture()
+ : channel_(voe_base_->CreateChannel()) {
+ EXPECT_GE(channel_, 0);
+
+ fake_microphone_input_file_ = resource_manager_.long_audio_file_path();
+ EXPECT_FALSE(fake_microphone_input_file_.empty());
+
+ SetUpLocalPlayback();
+ StartPlaying();
+}
+
+AfterStreamingFixture::~AfterStreamingFixture() {
+ voe_file_->StopPlayingFileAsMicrophone(channel_);
+ voe_base_->StopSend(channel_);
+ voe_base_->StopPlayout(channel_);
+ voe_base_->StopReceive(channel_);
+
+ voe_base_->DeleteChannel(channel_);
+}
+
+void AfterStreamingFixture::SwitchToManualMicrophone() {
+ EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
+
+ TEST_LOG("You need to speak manually into the microphone for this test.\n");
+ TEST_LOG("Please start speaking now.\n");
+ Sleep(2000);
+}
+
+void AfterStreamingFixture::RestartFakeMicrophone() {
+ EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
+ channel_, fake_microphone_input_file_.c_str(), true, true));
+}
+
+void AfterStreamingFixture::SetUpLocalPlayback() {
+ EXPECT_EQ(0, voe_base_->SetSendDestination(channel_, 8000, kLoopbackIp));
+ EXPECT_EQ(0, voe_base_->SetLocalReceiver(0, 8000));
+
+ webrtc::CodecInst codec;
+ codec.channels = 1;
+ codec.pacsize = 160;
+ codec.plfreq = 8000;
+ codec.pltype = 0;
+ codec.rate = 64000;
+ strcpy(codec.plname, "PCMU");
+
+ voe_codec_->SetSendCodec(channel_, codec);
+}
+
+void AfterStreamingFixture::StartPlaying() {
+ EXPECT_EQ(0, voe_base_->StartReceive(channel_));
+ EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ RestartFakeMicrophone();
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h
new file mode 100644
index 0000000..c589a4c
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
+#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
+
+#include "after_initialization_fixture.h"
+#include "resource_manager.h"
+
+// This fixture will, in addition to the work done by its superclasses,
+// create a channel and start playing a file through the fake microphone
+// to simulate microphone input. The purpose is to make it convenient
+// to write tests that require microphone input.
+class AfterStreamingFixture : public AfterInitializationFixture {
+ public:
+ AfterStreamingFixture();
+ virtual ~AfterStreamingFixture();
+
+ protected:
+ int channel_;
+ ResourceManager resource_manager_;
+ std::string fake_microphone_input_file_;
+
+ // Shuts off the fake microphone for this test.
+ void SwitchToManualMicrophone();
+
+ // Restarts the fake microphone if it's been shut off earlier.
+ void RestartFakeMicrophone();
+
+ private:
+ void SetUpLocalPlayback();
+ void StartPlaying();
+};
+
+
+#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc
new file mode 100644
index 0000000..408141b
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "before_initialization_fixture.h"
+
+#include "voice_engine_defines.h"
+
+BeforeInitializationFixture::BeforeInitializationFixture()
+ : voice_engine_(webrtc::VoiceEngine::Create()) {
+ EXPECT_TRUE(voice_engine_ != NULL);
+
+ voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_);
+ voe_codec_ = webrtc::VoECodec::GetInterface(voice_engine_);
+ voe_volume_control_ = webrtc::VoEVolumeControl::GetInterface(voice_engine_);
+ voe_dtmf_ = webrtc::VoEDtmf::GetInterface(voice_engine_);
+ voe_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(voice_engine_);
+ voe_apm_ = webrtc::VoEAudioProcessing::GetInterface(voice_engine_);
+ voe_network_ = webrtc::VoENetwork::GetInterface(voice_engine_);
+ voe_file_ = webrtc::VoEFile::GetInterface(voice_engine_);
+ voe_vsync_ = webrtc::VoEVideoSync::GetInterface(voice_engine_);
+ voe_encrypt_ = webrtc::VoEEncryption::GetInterface(voice_engine_);
+ voe_hardware_ = webrtc::VoEHardware::GetInterface(voice_engine_);
+ voe_xmedia_ = webrtc::VoEExternalMedia::GetInterface(voice_engine_);
+ voe_call_report_ = webrtc::VoECallReport::GetInterface(voice_engine_);
+ voe_neteq_stats_ = webrtc::VoENetEqStats::GetInterface(voice_engine_);
+}
+
+BeforeInitializationFixture::~BeforeInitializationFixture() {
+ EXPECT_EQ(0, voe_base_->Release());
+ EXPECT_EQ(0, voe_codec_->Release());
+ EXPECT_EQ(0, voe_volume_control_->Release());
+ EXPECT_EQ(0, voe_dtmf_->Release());
+ EXPECT_EQ(0, voe_rtp_rtcp_->Release());
+ EXPECT_EQ(0, voe_apm_->Release());
+ EXPECT_EQ(0, voe_network_->Release());
+ EXPECT_EQ(0, voe_file_->Release());
+ EXPECT_EQ(0, voe_vsync_->Release());
+ EXPECT_EQ(0, voe_encrypt_->Release());
+ EXPECT_EQ(0, voe_hardware_->Release());
+ EXPECT_EQ(0, voe_xmedia_->Release());
+ EXPECT_EQ(0, voe_call_report_->Release());
+ EXPECT_EQ(0, voe_neteq_stats_->Release());
+
+ EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine_));
+}
+
+void BeforeInitializationFixture::Sleep(long milliseconds) {
+ // Implementation note: This method is used to reduce usage of the macro and
+ // avoid ugly errors in Eclipse (its parser can't deal with the sleep macro).
+ SLEEP(milliseconds);
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h
new file mode 100644
index 0000000..ef1636b
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
+#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
+
+#include <assert.h>
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "voe_audio_processing.h"
+#include "voe_base.h"
+#include "voe_call_report.h"
+#include "voe_codec.h"
+#include "voe_dtmf.h"
+#include "voe_encryption.h"
+#include "voe_errors.h"
+#include "voe_external_media.h"
+#include "voe_file.h"
+#include "voe_hardware.h"
+#include "voe_neteq_stats.h"
+#include "voe_network.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_test_defines.h"
+#include "voe_video_sync.h"
+#include "voe_volume_control.h"
+
+// TODO(qhogpat): Remove these undefs once the clashing macros are gone.
+#undef TEST
+#undef ASSERT_TRUE
+#undef ASSERT_FALSE
+#include "gtest/gtest.h"
+#include "gmock/gmock.h"
+
+// This convenient fixture sets up all voice engine interfaces automatically for
+// use by testing subclasses. It allocates each interface and releases it once
+// which means that if a tests allocates additional interfaces from the voice
+// engine and forgets to release it, this test will fail in the destructor.
+// It will not call any init methods.
+//
+// Implementation note:
+// The interface fetching is done in the constructor and not SetUp() since
+// this relieves our subclasses from calling SetUp in the superclass if they
+// choose to override SetUp() themselves. This is fine as googletest will
+// construct new test objects for each method.
+class BeforeInitializationFixture : public testing::Test {
+ public:
+ BeforeInitializationFixture();
+ virtual ~BeforeInitializationFixture();
+
+ protected:
+ // Use this sleep function to sleep in test (avoid sleep macro).
+ void Sleep(long milliseconds);
+
+ webrtc::VoiceEngine* voice_engine_;
+ webrtc::VoEBase* voe_base_;
+ webrtc::VoECodec* voe_codec_;
+ webrtc::VoEVolumeControl* voe_volume_control_;
+ webrtc::VoEDtmf* voe_dtmf_;
+ webrtc::VoERTP_RTCP* voe_rtp_rtcp_;
+ webrtc::VoEAudioProcessing* voe_apm_;
+ webrtc::VoENetwork* voe_network_;
+ webrtc::VoEFile* voe_file_;
+ webrtc::VoEVideoSync* voe_vsync_;
+ webrtc::VoEEncryption* voe_encrypt_;
+ webrtc::VoEHardware* voe_hardware_;
+ webrtc::VoEExternalMedia* voe_xmedia_;
+ webrtc::VoECallReport* voe_call_report_;
+ webrtc::VoENetEqStats* voe_neteq_stats_;
+};
+
+#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc b/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc
new file mode 100644
index 0000000..65669ed
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "resource_manager.h"
+
+#include "testsupport/fileutils.h"
+
+ResourceManager::ResourceManager() {
+ std::string filename = "audio_long16.pcm";
+#if defined(WEBRTC_ANDROID)
+ long_audio_file_path_ = "/sdcard/" + filename;
+#else
+ std::string resource_path = webrtc::test::ProjectRootPath();
+ if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
+ long_audio_file_path_ = "";
+ } else {
+ long_audio_file_path_ =
+ resource_path + "test/data/voice_engine/" + filename;
+ }
+#endif
+}
+
diff --git a/trunk/src/voice_engine/main/test/auto_test/resource_manager.h b/trunk/src/voice_engine/main/test/auto_test/resource_manager.h
new file mode 100644
index 0000000..1bb91cf
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/resource_manager.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
+#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
+
+#include <string>
+
+class ResourceManager {
+ public:
+ ResourceManager();
+
+ // Returns the full path to a long audio file.
+ // Returns the empty string on failure.
+ const std::string& long_audio_file_path() const {
+ return long_audio_file_path_;
+ }
+
+ private:
+ std::string long_audio_file_path_;
+};
+
+#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc
new file mode 100644
index 0000000..bfe7f49
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc
@@ -0,0 +1,364 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "voe_standard_test.h"
+
+class AudioProcessingTest : public AfterStreamingFixture {
+ protected:
+ // Note: Be careful with this one, it is used in the
+ // Android / iPhone part too.
+ void TryEnablingAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
+ EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, agc_mode_to_set));
+
+ bool agc_enabled = false;
+ webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
+ EXPECT_TRUE(agc_enabled);
+ EXPECT_EQ(agc_mode_to_set, agc_mode);
+ }
+
+ void TryEnablingRxAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
+ EXPECT_EQ(0, voe_apm_->SetRxAgcStatus(channel_, true, agc_mode_to_set));
+
+ bool rx_agc_enabled = false;
+ webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetRxAgcStatus(channel_, rx_agc_enabled, agc_mode));
+ EXPECT_TRUE(rx_agc_enabled);
+ EXPECT_EQ(agc_mode_to_set, agc_mode);
+ }
+
+ // EC modes can map to other EC modes, so we have a separate parameter
+ // for what we expect the EC mode to be set to.
+ void TryEnablingEcWithMode(webrtc::EcModes ec_mode_to_set,
+ webrtc::EcModes expected_mode) {
+ EXPECT_EQ(0, voe_apm_->SetEcStatus(true, ec_mode_to_set));
+
+ bool ec_enabled = true;
+ webrtc::EcModes ec_mode = webrtc::kEcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
+
+ EXPECT_EQ(expected_mode, ec_mode);
+ }
+
+ // Here, the CNG mode will be expected to be on or off depending on the mode.
+ void TryEnablingAecmWithMode(webrtc::AecmModes aecm_mode_to_set,
+ bool cng_enabled_to_set) {
+ EXPECT_EQ(0, voe_apm_->SetAecmMode(aecm_mode_to_set, cng_enabled_to_set));
+
+ bool cng_enabled = false;
+ webrtc::AecmModes aecm_mode = webrtc::kAecmEarpiece;
+
+ voe_apm_->GetAecmMode(aecm_mode, cng_enabled);
+
+ EXPECT_EQ(cng_enabled_to_set, cng_enabled);
+ EXPECT_EQ(aecm_mode_to_set, aecm_mode);
+ }
+
+ void TryEnablingNsWithMode(webrtc::NsModes ns_mode_to_set,
+ webrtc::NsModes expected_ns_mode) {
+ EXPECT_EQ(0, voe_apm_->SetNsStatus(true, ns_mode_to_set));
+
+ bool ns_status = true;
+ webrtc::NsModes ns_mode = webrtc::kNsDefault;
+ EXPECT_EQ(0, voe_apm_->GetNsStatus(ns_status, ns_mode));
+
+ EXPECT_TRUE(ns_status);
+ EXPECT_EQ(expected_ns_mode, ns_mode);
+ }
+
+ void TryEnablingRxNsWithMode(webrtc::NsModes ns_mode_to_set,
+ webrtc::NsModes expected_ns_mode) {
+ EXPECT_EQ(0, voe_apm_->SetRxNsStatus(channel_, true, ns_mode_to_set));
+
+ bool ns_status = true;
+ webrtc::NsModes ns_mode = webrtc::kNsDefault;
+ EXPECT_EQ(0, voe_apm_->GetRxNsStatus(channel_, ns_status, ns_mode));
+
+ EXPECT_TRUE(ns_status);
+ EXPECT_EQ(expected_ns_mode, ns_mode);
+ }
+
+ void TryDetectingSilence() {
+ // Here, speech is running. Shut down speech.
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
+ EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, true));
+ EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
+
+ // We should detect the silence after a short time.
+ Sleep(50);
+ for (int i = 0; i < 25; i++) {
+ EXPECT_EQ(0, voe_apm_->VoiceActivityIndicator(channel_));
+ Sleep(10);
+ }
+ }
+
+ void TryDetectingSpeechAfterSilence() {
+ // Re-enable speech.
+ RestartFakeMicrophone();
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, false));
+ EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, false));
+
+ // We should detect the speech after a short time.
+ for (int i = 0; i < 50; i++) {
+ if (voe_apm_->VoiceActivityIndicator(channel_) == 1) {
+ return;
+ }
+ Sleep(10);
+ }
+
+ ADD_FAILURE() << "Failed to detect speech within 500 ms.";
+ }
+};
+
+#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+
+TEST_F(AudioProcessingTest, AgcIsOnByDefault) {
+ bool agc_enabled = false;
+ webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
+
+ EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
+ EXPECT_TRUE(agc_enabled);
+ EXPECT_EQ(webrtc::kAgcAdaptiveAnalog, agc_mode);
+}
+
+TEST_F(AudioProcessingTest, CanEnableAgcWithAllModes) {
+ TryEnablingAgcWithMode(webrtc::kAgcAdaptiveDigital);
+ TryEnablingAgcWithMode(webrtc::kAgcAdaptiveAnalog);
+ TryEnablingAgcWithMode(webrtc::kAgcFixedDigital);
+}
+
+TEST_F(AudioProcessingTest, EcIsDisabledAndAecIsDefaultEcMode) {
+ bool ec_enabled = true;
+ webrtc::EcModes ec_mode = webrtc::kEcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
+ EXPECT_FALSE(ec_enabled);
+ EXPECT_EQ(webrtc::kEcAec, ec_mode);
+}
+
+TEST_F(AudioProcessingTest, EnablingEcAecShouldEnableEcAec) {
+ TryEnablingEcWithMode(webrtc::kEcAec, webrtc::kEcAec);
+}
+
+TEST_F(AudioProcessingTest, EnablingEcConferenceShouldEnableEcAec) {
+ TryEnablingEcWithMode(webrtc::kEcConference, webrtc::kEcAec);
+}
+
+TEST_F(AudioProcessingTest, EcMetricsAreOffByDefault) {
+ bool enabled = true;
+ EXPECT_EQ(0, voe_apm_->GetEcMetricsStatus(enabled));
+ EXPECT_FALSE(enabled);
+}
+
+TEST_F(AudioProcessingTest, ManualTestEcMetrics) {
+ SwitchToManualMicrophone();
+
+ EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true));
+
+ // Must enable AEC to get valid echo metrics.
+ EXPECT_EQ(0, voe_apm_->SetEcStatus(true, webrtc::kEcAec));
+
+ TEST_LOG("Speak into microphone and check metrics for 10 seconds...\n");
+ int erl, erle, rerl, a_nlp;
+ int delay_median = 0;
+ int delay_std = 0;
+
+ for (int i = 0; i < 5; i++) {
+ Sleep(2000);
+ EXPECT_EQ(0, voe_apm_->GetEchoMetrics(erl, erle, rerl, a_nlp));
+ EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay_median, delay_std));
+ TEST_LOG(" Echo : ERL=%5d, ERLE=%5d, RERL=%5d, A_NLP=%5d [dB], "
+ " delay median=%3d, delay std=%3d [ms]\n", erl, erle, rerl, a_nlp,
+ delay_median, delay_std);
+ }
+
+ EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(false));
+}
+
+// TODO(phoglund): Reenable below test when it's no longer flaky.
+TEST_F(AudioProcessingTest, DISABLED_TestVoiceActivityDetectionWithObserver) {
+ voetest::RxCallback rx_callback;
+ EXPECT_EQ(0, voe_apm_->RegisterRxVadObserver(channel_, rx_callback));
+
+ // The extra sleeps are to allow decisions some time to propagate to the
+ // observer.
+ TryDetectingSilence();
+ Sleep(100);
+
+ EXPECT_EQ(0, rx_callback._vadDecision);
+
+ TryDetectingSpeechAfterSilence();
+ Sleep(100);
+
+ EXPECT_EQ(1, rx_callback._vadDecision);
+
+ EXPECT_EQ(0, voe_apm_->DeRegisterRxVadObserver(channel_));
+}
+
+#endif // !MAC_IPHONE && !WEBRTC_ANDROID
+
+TEST_F(AudioProcessingTest, EnablingEcAecmShouldEnableEcAecm) {
+ // This one apparently applies to Android and iPhone as well.
+ TryEnablingEcWithMode(webrtc::kEcAecm, webrtc::kEcAecm);
+}
+
+TEST_F(AudioProcessingTest, EcAecmModeIsEnabledAndSpeakerphoneByDefault) {
+ bool cng_enabled = false;
+ webrtc::AecmModes aecm_mode = webrtc::kAecmEarpiece;
+
+ voe_apm_->GetAecmMode(aecm_mode, cng_enabled);
+
+ EXPECT_TRUE(cng_enabled);
+ EXPECT_EQ(webrtc::kAecmSpeakerphone, aecm_mode);
+}
+
+TEST_F(AudioProcessingTest, CanSetAecmMode) {
+ EXPECT_EQ(0, voe_apm_->SetEcStatus(true, webrtc::kEcAecm));
+
+ // Try some AECM mode - CNG enabled combinations.
+ TryEnablingAecmWithMode(webrtc::kAecmEarpiece, true);
+ TryEnablingAecmWithMode(webrtc::kAecmEarpiece, false);
+ TryEnablingAecmWithMode(webrtc::kAecmLoudEarpiece, true);
+ TryEnablingAecmWithMode(webrtc::kAecmLoudSpeakerphone, false);
+ TryEnablingAecmWithMode(webrtc::kAecmQuietEarpieceOrHeadset, true);
+ TryEnablingAecmWithMode(webrtc::kAecmSpeakerphone, false);
+}
+
+TEST_F(AudioProcessingTest, RxAgcShouldBeOffByDefault) {
+ bool rx_agc_enabled = true;
+ webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetRxAgcStatus(channel_, rx_agc_enabled, agc_mode));
+ EXPECT_FALSE(rx_agc_enabled);
+ EXPECT_EQ(webrtc::kAgcAdaptiveDigital, agc_mode);
+}
+
+TEST_F(AudioProcessingTest, CanTurnOnDigitalRxAcg) {
+ TryEnablingRxAgcWithMode(webrtc::kAgcAdaptiveDigital);
+ TryEnablingRxAgcWithMode(webrtc::kAgcFixedDigital);
+}
+
+TEST_F(AudioProcessingTest, CannotTurnOnAdaptiveAnalogRxAgc) {
+ EXPECT_EQ(-1, voe_apm_->SetRxAgcStatus(
+ channel_, true, webrtc::kAgcAdaptiveAnalog));
+}
+
+TEST_F(AudioProcessingTest, NsIsOffWithModerateSuppressionByDefault) {
+ bool ns_status = true;
+ webrtc::NsModes ns_mode = webrtc::kNsDefault;
+ EXPECT_EQ(0, voe_apm_->GetNsStatus(ns_status, ns_mode));
+
+ EXPECT_FALSE(ns_status);
+ EXPECT_EQ(webrtc::kNsModerateSuppression, ns_mode);
+}
+
+TEST_F(AudioProcessingTest, CanSetNsMode) {
+ // Concrete suppression values map to themselves.
+ TryEnablingNsWithMode(webrtc::kNsHighSuppression,
+ webrtc::kNsHighSuppression);
+ TryEnablingNsWithMode(webrtc::kNsLowSuppression,
+ webrtc::kNsLowSuppression);
+ TryEnablingNsWithMode(webrtc::kNsModerateSuppression,
+ webrtc::kNsModerateSuppression);
+ TryEnablingNsWithMode(webrtc::kNsVeryHighSuppression,
+ webrtc::kNsVeryHighSuppression);
+
+ // Conference and Default map to concrete values.
+ TryEnablingNsWithMode(webrtc::kNsConference,
+ webrtc::kNsHighSuppression);
+ TryEnablingNsWithMode(webrtc::kNsDefault,
+ webrtc::kNsModerateSuppression);
+}
+
+TEST_F(AudioProcessingTest, RxNsIsOffWithModerateSuppressionByDefault) {
+ bool ns_status = true;
+ webrtc::NsModes ns_mode = webrtc::kNsDefault;
+ EXPECT_EQ(0, voe_apm_->GetRxNsStatus(channel_, ns_status, ns_mode));
+
+ EXPECT_FALSE(ns_status);
+ EXPECT_EQ(webrtc::kNsModerateSuppression, ns_mode);
+}
+
+TEST_F(AudioProcessingTest, CanSetRxNsMode) {
+ EXPECT_EQ(0, voe_apm_->SetRxNsStatus(channel_, true));
+
+ // See comments on the regular NS test above.
+ TryEnablingRxNsWithMode(webrtc::kNsHighSuppression,
+ webrtc::kNsHighSuppression);
+ TryEnablingRxNsWithMode(webrtc::kNsLowSuppression,
+ webrtc::kNsLowSuppression);
+ TryEnablingRxNsWithMode(webrtc::kNsModerateSuppression,
+ webrtc::kNsModerateSuppression);
+ TryEnablingRxNsWithMode(webrtc::kNsVeryHighSuppression,
+ webrtc::kNsVeryHighSuppression);
+ TryEnablingRxNsWithMode(webrtc::kNsConference,
+ webrtc::kNsHighSuppression);
+ TryEnablingRxNsWithMode(webrtc::kNsDefault,
+ webrtc::kNsModerateSuppression);
+}
+
+TEST_F(AudioProcessingTest, VadIsDisabledByDefault) {
+ bool vad_enabled;
+ bool disabled_dtx;
+ webrtc::VadModes vad_mode;
+
+ EXPECT_EQ(0, voe_codec_->GetVADStatus(
+ channel_, vad_enabled, vad_mode, disabled_dtx));
+
+ EXPECT_FALSE(vad_enabled);
+}
+
+TEST_F(AudioProcessingTest, VoiceActivityIndicatorReturns1WithSpeechOn) {
+ // This sleep is necessary since the voice detection algorithm needs some
+ // time to detect the speech from the fake microphone.
+ Sleep(500);
+ EXPECT_EQ(1, voe_apm_->VoiceActivityIndicator(channel_));
+}
+
+#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+
+TEST_F(AudioProcessingTest, AgcIsOffByDefaultAndDigital) {
+ bool agc_enabled = true;
+ webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
+
+ EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
+ EXPECT_FALSE(agc_enabled);
+ EXPECT_EQ(webrtc::kAgcAdaptiveDigital, agc_mode);
+}
+
+TEST_F(AudioProcessingTest, CanEnableAgcInAdaptiveDigitalMode) {
+ TryEnablingAgcWithMode(webrtc::kAgcAdaptiveDigital);
+}
+
+TEST_F(AudioProcessingTest, AgcIsPossibleExceptInAdaptiveAnalogMode) {
+ EXPECT_EQ(-1, voe_apm_->SetAgcStatus(true, webrtc::kAgcAdaptiveAnalog));
+ EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, webrtc::kAgcFixedDigital));
+ EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, webrtc::kAgcAdaptiveDigital));
+}
+
+TEST_F(AudioProcessingTest, EcIsDisabledAndAecmIsDefaultEcMode) {
+ bool ec_enabled = true;
+ webrtc::EcModes ec_mode = webrtc::kEcDefault;
+
+ EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
+ EXPECT_FALSE(ec_enabled);
+ EXPECT_EQ(webrtc::kEcAecm, ec_mode);
+}
+
+TEST_F(AudioProcessingTest, TestVoiceActivityDetection) {
+ TryDetectingSilence();
+ TryDetectingSpeechAfterSilence();
+}
+
+#endif // MAC_IPHONE || WEBRTC_ANDROID
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc
new file mode 100644
index 0000000..c96b14d
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "testsupport/fileutils.h"
+
+class CallReportTest : public AfterStreamingFixture {
+};
+
+TEST_F(CallReportTest, ResetCallReportStatisticsFailsForBadInput) {
+ EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(-2));
+ EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(1));
+}
+
+TEST_F(CallReportTest, ResetCallReportStatisticsSucceedsWithCorrectInput) {
+ EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(channel_));
+ EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(-1));
+}
+
+TEST_F(CallReportTest, EchoMetricSummarySucceeds) {
+ EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true));
+ Sleep(1000);
+
+ webrtc::EchoStatistics echo_statistics;
+ EXPECT_EQ(0, voe_call_report_->GetEchoMetricSummary(echo_statistics));
+}
+
+TEST_F(CallReportTest, GetRoundTripTimeSummaryReturnsAllMinusOnesIfRtcpIsOff) {
+ voe_rtp_rtcp_->SetRTCPStatus(channel_, false);
+
+ webrtc::StatVal delays;
+ EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays));
+ EXPECT_EQ(-1, delays.average);
+ EXPECT_EQ(-1, delays.min);
+ EXPECT_EQ(-1, delays.max);
+}
+
+TEST_F(CallReportTest, GetRoundTripTimesReturnsValuesIfRtcpIsOn) {
+ voe_rtp_rtcp_->SetRTCPStatus(channel_, true);
+ Sleep(1000);
+
+ webrtc::StatVal delays;
+ EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays));
+ EXPECT_NE(-1, delays.average);
+ EXPECT_NE(-1, delays.min);
+ EXPECT_NE(-1, delays.max);
+}
+
+TEST_F(CallReportTest, DeadOrAliveSummaryFailsIfDeadOrAliveTrackingNotActive) {
+ int count_the_dead;
+ int count_the_living;
+ EXPECT_EQ(-1, voe_call_report_->GetDeadOrAliveSummary(channel_,
+ count_the_dead,
+ count_the_living));
+}
+
+TEST_F(CallReportTest,
+ DeadOrAliveSummarySucceedsIfDeadOrAliveTrackingIsActive) {
+ EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
+ Sleep(1200);
+
+ int count_the_dead;
+ int count_the_living;
+ EXPECT_EQ(0, voe_call_report_->GetDeadOrAliveSummary(channel_,
+ count_the_dead,
+ count_the_living));
+
+ EXPECT_GE(count_the_dead, 0);
+ EXPECT_GE(count_the_living, 0);
+}
+
+TEST_F(CallReportTest, WriteReportToFileFailsOnBadInput) {
+ EXPECT_EQ(-1, voe_call_report_->WriteReportToFile(NULL));
+}
+
+TEST_F(CallReportTest, WriteReportToFileSucceedsWithCorrectFilename) {
+ std::string output_path = webrtc::test::OutputPath();
+ std::string report_filename = output_path + "call_report.txt";
+
+ EXPECT_EQ(0, voe_call_report_->WriteReportToFile(report_filename.c_str()));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc
new file mode 100644
index 0000000..6d902ef
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_initialization_fixture.h"
+
+class CodecBeforeStreamingTest : public AfterInitializationFixture {
+ protected:
+ void SetUp() {
+ memset(&codec_instance_, 0, sizeof(codec_instance_));
+ codec_instance_.channels = 1;
+ codec_instance_.plfreq = 16000;
+ codec_instance_.pacsize = 480;
+
+ channel_ = voe_base_->CreateChannel();
+ }
+
+ void TearDown() {
+ voe_base_->DeleteChannel(channel_);
+ }
+
+ webrtc::CodecInst codec_instance_;
+ int channel_;
+};
+
+// TODO(phoglund): add test which verifies default pltypes for various codecs.
+
+TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeFailsForInvalidCodecName) {
+ strcpy(codec_instance_.plname, "SomeInvalidCodecName");
+
+ // Should fail since the codec name is invalid.
+ EXPECT_NE(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+}
+
+TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeRecognizesISAC) {
+ strcpy(codec_instance_.plname, "iSAC");
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+ strcpy(codec_instance_.plname, "ISAC");
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+}
+
+TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeISACPayloadType) {
+ strcpy(codec_instance_.plname, "ISAC");
+
+ codec_instance_.pltype = 123;
+ EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+ EXPECT_EQ(123, codec_instance_.pltype);
+
+ codec_instance_.pltype = 104;
+ EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+
+ EXPECT_EQ(104, codec_instance_.pltype);
+}
+
+TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeILBCPayloadType) {
+ strcpy(codec_instance_.plname, "iLBC");
+ codec_instance_.plfreq = 8000;
+ codec_instance_.pacsize = 240;
+ codec_instance_.rate = 13300;
+
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+ int original_pltype = codec_instance_.pltype;
+ codec_instance_.pltype = 123;
+ EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+
+ EXPECT_EQ(123, codec_instance_.pltype);
+
+ codec_instance_.pltype = original_pltype;
+ EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
+ EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
+
+ EXPECT_EQ(original_pltype, codec_instance_.pltype);
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc
new file mode 100644
index 0000000..dc5558b
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc
@@ -0,0 +1,231 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "voe_test_defines.h"
+#include "voice_engine_defines.h"
+
+class CodecTest : public AfterStreamingFixture {
+ protected:
+ void SetUp() {
+ memset(&codec_instance_, 0, sizeof(codec_instance_));
+ }
+
+ void SetArbitrarySendCodec() {
+ // Just grab the first codec.
+ EXPECT_EQ(0, voe_codec_->GetCodec(0, codec_instance_));
+ EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
+ }
+
+ webrtc::CodecInst codec_instance_;
+};
+
+static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
+ if (!_stricmp(codec_instance->plname, "ilbc")) {
+ if (packet_size == 160 || packet_size == 320) {
+ codec_instance->rate = 15200;
+ } else {
+ codec_instance->rate = 13300;
+ }
+ }
+}
+
+static bool IsNotViableSendCodec(const char* codec_name) {
+ return !_stricmp(codec_name, "CN") ||
+ !_stricmp(codec_name, "telephone-event") ||
+ !_stricmp(codec_name, "red");
+}
+
+TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
+ EXPECT_EQ(0, voe_codec_->GetSendCodec(channel_, codec_instance_));
+ EXPECT_EQ(1, codec_instance_.channels);
+ EXPECT_EQ(160, codec_instance_.pacsize);
+ EXPECT_EQ(8000, codec_instance_.plfreq);
+ EXPECT_EQ(0, codec_instance_.pltype);
+ EXPECT_EQ(64000, codec_instance_.rate);
+ EXPECT_STRCASEEQ("PCMU", codec_instance_.plname);
+}
+
+TEST_F(CodecTest, VoiceActivityDetectionIsOffByDefault) {
+ bool vad_enabled = false;
+ bool dtx_disabled = false;
+ webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
+
+ voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
+
+ EXPECT_FALSE(vad_enabled);
+ EXPECT_TRUE(dtx_disabled);
+ EXPECT_EQ(webrtc::kVadConventional, vad_mode);
+}
+
+TEST_F(CodecTest, VoiceActivityDetectionCanBeEnabled) {
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
+
+ bool vad_enabled = false;
+ bool dtx_disabled = false;
+ webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
+
+ voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
+
+ EXPECT_TRUE(vad_enabled);
+ EXPECT_EQ(webrtc::kVadConventional, vad_mode);
+ EXPECT_FALSE(dtx_disabled);
+}
+
+TEST_F(CodecTest, VoiceActivityDetectionTypeSettingsCanBeChanged) {
+ bool vad_enabled = false;
+ bool dtx_disabled = false;
+ webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
+
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(
+ channel_, true, webrtc::kVadAggressiveLow, false));
+ EXPECT_EQ(0, voe_codec_->GetVADStatus(
+ channel_, vad_enabled, vad_mode, dtx_disabled));
+ EXPECT_EQ(vad_mode, webrtc::kVadAggressiveLow);
+ EXPECT_FALSE(dtx_disabled);
+
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(
+ channel_, true, webrtc::kVadAggressiveMid, false));
+ EXPECT_EQ(0, voe_codec_->GetVADStatus(
+ channel_, vad_enabled, vad_mode, dtx_disabled));
+ EXPECT_EQ(vad_mode, webrtc::kVadAggressiveMid);
+ EXPECT_FALSE(dtx_disabled);
+
+ // The fourth argument is the DTX disable flag.
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(
+ channel_, true, webrtc::kVadAggressiveHigh, true));
+ EXPECT_EQ(0, voe_codec_->GetVADStatus(
+ channel_, vad_enabled, vad_mode, dtx_disabled));
+ EXPECT_EQ(vad_mode, webrtc::kVadAggressiveHigh);
+ EXPECT_TRUE(dtx_disabled);
+
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(
+ channel_, true, webrtc::kVadConventional, true));
+ EXPECT_EQ(0, voe_codec_->GetVADStatus(
+ channel_, vad_enabled, vad_mode, dtx_disabled));
+ EXPECT_EQ(vad_mode, webrtc::kVadConventional);
+}
+
+TEST_F(CodecTest, VoiceActivityDetectionCanBeTurnedOff) {
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
+
+ // VAD is always on when DTX is on, so we need to turn off DTX too.
+ EXPECT_EQ(0, voe_codec_->SetVADStatus(
+ channel_, false, webrtc::kVadConventional, true));
+
+ bool vad_enabled = false;
+ bool dtx_disabled = false;
+ webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
+
+ voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
+
+ EXPECT_FALSE(vad_enabled);
+ EXPECT_TRUE(dtx_disabled);
+ EXPECT_EQ(webrtc::kVadConventional, vad_mode);
+}
+
+// Tests requiring manual verification (although they do have some value
+// without the manual verification):
+TEST_F(CodecTest, ManualExtendedISACApisBehaveAsExpected) {
+ strcpy(codec_instance_.plname, "isac");
+ codec_instance_.pltype = 103;
+ codec_instance_.plfreq = 16000;
+ codec_instance_.channels = 1;
+ // -1 here means "adaptive rate".
+ codec_instance_.rate = -1;
+ codec_instance_.pacsize = 480;
+
+ EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
+
+ EXPECT_NE(0, voe_codec_->SetISACInitTargetRate(channel_, 5000)) <<
+ "iSAC should reject rate 5000.";
+ EXPECT_NE(0, voe_codec_->SetISACInitTargetRate(channel_, 33000)) <<
+ "iSAC should reject rate 33000.";
+ EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 32000));
+
+ TEST_LOG("Ensure that the sound is good (iSAC, target = 32kbps)...\n");
+ Sleep(3000);
+
+ EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000));
+ TEST_LOG("Ensure that the sound is good (iSAC, target = 10kbps)...\n");
+ Sleep(3000);
+
+ EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000, true));
+ EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000, false));
+ EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 0));
+ TEST_LOG("Ensure that the sound is good (iSAC, target = default)...\n");
+ Sleep(3000);
+
+ TEST_LOG(" Testing SetISACMaxPayloadSize:\n");
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_NE(0, voe_codec_->SetISACMaxPayloadSize(channel_, 50));
+ EXPECT_NE(0, voe_codec_->SetISACMaxPayloadSize(channel_, 650));
+ EXPECT_EQ(0, voe_codec_->SetISACMaxPayloadSize(channel_, 120));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ TEST_LOG("Ensure that the sound is good (iSAC, "
+ "max payload size = 100 bytes)...\n");
+ Sleep(3000);
+
+ TEST_LOG(" Testing SetISACMaxRate:\n");
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_codec_->SetISACMaxPayloadSize(channel_, 400));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_NE(0, voe_codec_->SetISACMaxRate(channel_, 31900));
+ EXPECT_NE(0, voe_codec_->SetISACMaxRate(channel_, 53500));
+ EXPECT_EQ(0, voe_codec_->SetISACMaxRate(channel_, 32000));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ TEST_LOG("Ensure that the sound is good (iSAC, max rate = 32 kbps)...\n");
+ Sleep(3000);
+
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+
+ // Restore "no limitation". No, no limit, we reach for the sky.
+ EXPECT_EQ(0, voe_codec_->SetISACMaxRate(channel_, 53400));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+}
+
+TEST_F(CodecTest, ManualVerifySendCodecsForAllPacketSizes) {
+ for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
+ voe_codec_->GetCodec(i, codec_instance_);
+ if (IsNotViableSendCodec(codec_instance_.plname)) {
+ TEST_LOG("Skipping %s.\n", codec_instance_.plname);
+ continue;
+ }
+ EXPECT_NE(-1, codec_instance_.pltype) <<
+ "The codec database should suggest a payload type.";
+
+ // Test with default packet size:
+ TEST_LOG("%s (pt=%d): default packet size(%d), accepts sizes ",
+ codec_instance_.plname, codec_instance_.pltype,
+ codec_instance_.pacsize);
+ voe_codec_->SetSendCodec(channel_, codec_instance_);
+ Sleep(CODEC_TEST_TIME);
+
+ // Now test other reasonable packet sizes:
+ bool at_least_one_succeeded = false;
+ for (int packet_size = 80; packet_size < 1000; packet_size += 80) {
+ SetRateIfILBC(&codec_instance_, packet_size);
+ codec_instance_.pacsize = packet_size;
+
+ if (voe_codec_->SetSendCodec(channel_, codec_instance_) != -1) {
+ // Note that it's fine for SetSendCodec to fail - what packet sizes
+ // it accepts depends on the codec. It should accept one at minimum.
+ TEST_LOG("%d ", packet_size);
+ TEST_LOG_FLUSH;
+ at_least_one_succeeded = true;
+ Sleep(CODEC_TEST_TIME);
+ }
+ }
+ TEST_LOG("\n");
+ EXPECT_TRUE(at_least_one_succeeded);
+ }
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc
new file mode 100644
index 0000000..6aa4625
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc
@@ -0,0 +1,101 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "voice_engine_defines.h"
+
+class DtmfTest : public AfterStreamingFixture {
+ protected:
+ void RunSixteenDtmfEvents(bool out_of_band) {
+ TEST_LOG("Sending telephone events:\n");
+ EXPECT_EQ(0, voe_dtmf_->SetDtmfFeedbackStatus(false));
+
+ for (int i = 0; i < 16; i++) {
+ TEST_LOG("%d ", i);
+ TEST_LOG_FLUSH;
+ EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(
+ channel_, i, out_of_band, 160, 10));
+ Sleep(500);
+ }
+ TEST_LOG("\n");
+ }
+};
+
+TEST_F(DtmfTest, DtmfFeedbackIsEnabledByDefaultButNotDirectFeedback) {
+ bool dtmf_feedback = false;
+ bool dtmf_direct_feedback = false;
+
+ EXPECT_EQ(0, voe_dtmf_->GetDtmfFeedbackStatus(dtmf_feedback,
+ dtmf_direct_feedback));
+
+ EXPECT_TRUE(dtmf_feedback);
+ EXPECT_FALSE(dtmf_direct_feedback);
+}
+
+TEST_F(DtmfTest, ManualSuccessfullySendsInBandTelephoneEvents) {
+ RunSixteenDtmfEvents(false);
+}
+
+TEST_F(DtmfTest, ManualSuccessfullySendsOutOfBandTelephoneEvents) {
+ RunSixteenDtmfEvents(true);
+}
+
+TEST_F(DtmfTest, TestTwoNonDtmfEvents) {
+ EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 32, true));
+ EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 110, true));
+}
+
+#ifndef MAC_IPHONE
+TEST_F(DtmfTest, ManualCanDisableDtmfPlayoutExceptOnIphone) {
+ TEST_LOG("Disabling DTMF playout (no tone should be heard) \n");
+ EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, false));
+ EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 0, true));
+ Sleep(500);
+
+ TEST_LOG("Enabling DTMF playout (tone should be heard) \n");
+ EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, true));
+ EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 0, true));
+ Sleep(500);
+}
+#endif
+
+// This test modifies the DTMF payload type from the default 106 to 88
+// and then runs through 16 DTMF out.of-band events.
+TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) {
+ webrtc::CodecInst codec_instance;
+
+ TEST_LOG("Changing DTMF payload type.\n");
+
+ // Start by modifying the receiving side.
+ for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) {
+ EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance));
+ if (!_stricmp("telephone-event", codec_instance.plname)) {
+ codec_instance.pltype = 88; // Use 88 instead of default 106.
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+ EXPECT_EQ(0, voe_base_->StopReceive(channel_));
+ EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance));
+ EXPECT_EQ(0, voe_base_->StartReceive(channel_));
+ EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ break;
+ }
+ }
+
+ Sleep(500);
+
+ // Next, we must modify the sending side as well.
+ EXPECT_EQ(0, voe_dtmf_->SetSendTelephoneEventPayloadType(
+ channel_, codec_instance.pltype));
+
+ RunSixteenDtmfEvents(true);
+
+ EXPECT_EQ(0, voe_dtmf_->SetDtmfFeedbackStatus(true, false));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc
new file mode 100644
index 0000000..63b1600
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "testsupport/fileutils.h"
+
+class FileTest : public AfterStreamingFixture {
+ protected:
+ // Creates the string åäö.pcm.
+ std::string CreateTrickyFilenameInUtf8() {
+ char filename[16] = { (char)0xc3, (char)0xa5,
+ (char)0xc3, (char)0xa4,
+ (char)0xc3, (char)0xb6,
+ static_cast<char>(0) };
+ return std::string(filename) + ".pcm";
+ }
+};
+
+TEST_F(FileTest, ManualRecordToFileForThreeSecondsAndPlayback) {
+ SwitchToManualMicrophone();
+
+ std::string recording_filename =
+ webrtc::test::OutputPath() + CreateTrickyFilenameInUtf8();
+
+ TEST_LOG("Recording to %s for 3 seconds.\n", recording_filename.c_str());
+ EXPECT_EQ(0, voe_file_->StartRecordingMicrophone(recording_filename.c_str()));
+ Sleep(3000);
+ EXPECT_EQ(0, voe_file_->StopRecordingMicrophone());
+
+ TEST_LOG("Playing back %s.\n", recording_filename.c_str());
+ EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
+ channel_, recording_filename.c_str()));
+
+ // Play the file to the user and ensure the is-playing-locally
+ // and scaling methods also work. The clip is 3 seconds long.
+ Sleep(250);
+ EXPECT_EQ(1, voe_file_->IsPlayingFileLocally(channel_));
+ Sleep(1500);
+ TEST_LOG("Decreasing level by 50%%.\n");
+ EXPECT_EQ(0, voe_file_->ScaleLocalFilePlayout(channel_, 0.5f));
+ Sleep(1500);
+ EXPECT_EQ(0, voe_file_->IsPlayingFileLocally(channel_));
+}
+
+TEST_F(FileTest, ManualRecordPlayoutToWavFileForThreeSecondsAndPlayback) {
+ webrtc::CodecInst send_codec;
+ voe_codec_->GetSendCodec(channel_, send_codec);
+
+ std::string recording_filename =
+ webrtc::test::OutputPath() + "playout.wav";
+
+ TEST_LOG("Recording playout to %s.\n", recording_filename.c_str());
+ EXPECT_EQ(0, voe_file_->StartRecordingPlayout(
+ channel_, recording_filename.c_str(), &send_codec));
+ Sleep(3000);
+ EXPECT_EQ(0, voe_file_->StopRecordingPlayout(channel_));
+
+ TEST_LOG("Playing back the recording in looping mode.\n");
+ EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
+ channel_, recording_filename.c_str(), true, false,
+ webrtc::kFileFormatWavFile));
+
+ Sleep(2000);
+ EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
+ Sleep(2000);
+ // We should still be playing since we're looping.
+ EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
+
+ // Try scaling as well.
+ TEST_LOG("Decreasing level by 50%%.\n");
+ EXPECT_EQ(0, voe_file_->ScaleFileAsMicrophonePlayout(channel_, 0.5f));
+ Sleep(1000);
+
+ EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc
new file mode 100644
index 0000000..540614e
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc
@@ -0,0 +1,27 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_types.h"
+#include "before_initialization_fixture.h"
+
+using namespace webrtc;
+
+class HardwareBeforeInitializingTest : public BeforeInitializationFixture {
+};
+
+TEST_F(HardwareBeforeInitializingTest,
+ SetAudioDeviceLayerAcceptsPlatformDefaultBeforeInitializing) {
+ AudioLayers wanted_layer = kAudioPlatformDefault;
+ AudioLayers given_layer;
+ EXPECT_EQ(0, voe_hardware_->SetAudioDeviceLayer(wanted_layer));
+ EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer));
+ EXPECT_EQ(wanted_layer, given_layer) <<
+ "These should be the same before initializing.";
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc
new file mode 100644
index 0000000..6e56347
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc
@@ -0,0 +1,166 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstring>
+
+#include "after_initialization_fixture.h"
+
+using namespace webrtc;
+
+static const char* kNoDevicesErrorMessage =
+ "Either you have no recording / playout device "
+ "on your system, or the method failed.";
+
+class HardwareBeforeStreamingTest : public AfterInitializationFixture {
+};
+
+// Tests that apply to both mobile and desktop:
+
+TEST_F(HardwareBeforeStreamingTest,
+ SetAudioDeviceLayerFailsSinceTheVoiceEngineHasBeenInitialized) {
+ EXPECT_NE(0, voe_hardware_->SetAudioDeviceLayer(kAudioPlatformDefault));
+ EXPECT_EQ(VE_ALREADY_INITED, voe_base_->LastError());
+}
+
+TEST_F(HardwareBeforeStreamingTest,
+ GetCPULoadSucceedsOnWindowsButNotOtherPlatforms) {
+ int load_percent;
+#if defined(_WIN32)
+ EXPECT_EQ(0, voe_hardware_->GetCPULoad(load_percent));
+#else
+ EXPECT_NE(0, voe_hardware_->GetCPULoad(load_percent)) <<
+ "Should fail on non-Windows platforms.";
+#endif
+}
+
+// Tests that only apply to mobile:
+
+#ifdef MAC_IPHONE
+TEST_F(HardwareBeforeStreamingTest, ResetsAudioDeviceOnIphone) {
+ EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
+}
+#endif
+
+// Tests that only apply to desktop:
+#if !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
+
+TEST_F(HardwareBeforeStreamingTest, GetSystemCpuLoadSucceeds) {
+#ifdef _WIN32
+ // This method needs some warm-up time on Windows. We sleep a good amount
+ // of time instead of retrying to make the test simpler.
+ Sleep(2000);
+#endif
+
+ int load_percent;
+ EXPECT_EQ(0, voe_hardware_->GetSystemCPULoad(load_percent));
+}
+
+TEST_F(HardwareBeforeStreamingTest, GetPlayoutDeviceStatusReturnsTrue) {
+ bool play_available = false;
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceStatus(play_available));
+ ASSERT_TRUE(play_available) <<
+ "Ensures that the method works and that hardware is in the right state.";
+}
+
+TEST_F(HardwareBeforeStreamingTest, GetRecordingDeviceStatusReturnsTrue) {
+ bool recording_available = false;
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceStatus(recording_available));
+ EXPECT_TRUE(recording_available) <<
+ "Ensures that the method works and that hardware is in the right state.";
+}
+
+ // Win, Mac and Linux sound device tests.
+TEST_F(HardwareBeforeStreamingTest,
+ GetRecordingDeviceNameRetrievesDeviceNames) {
+ char device_name[128] = {0};
+ char guid_name[128] = {0};
+
+#ifdef _WIN32
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
+ -1, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
+ device_name[0] = '\0';
+
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
+ -1, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
+
+#else
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
+ 0, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
+ device_name[0] = '\0';
+
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
+ 0, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
+#endif // !WIN32
+}
+
+TEST_F(HardwareBeforeStreamingTest,
+ AllEnumeratedRecordingDevicesCanBeSetAsRecordingDevice) {
+ // Check recording side.
+ // Extended Win32 enumeration tests: unique GUID outputs on Vista and up:
+ // Win XP and below : device_name is copied to guid_name.
+ // Win Vista and up : device_name is the friendly name and GUID is a unique
+ // identifier.
+ // Other : guid_name is left unchanged.
+ int num_of_recording_devices = 0;
+ EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(
+ num_of_recording_devices));
+ EXPECT_GT(num_of_recording_devices, 0) << kNoDevicesErrorMessage;
+
+ char device_name[128] = {0};
+ char guid_name[128] = {0};
+
+ for (int i = 0; i < num_of_recording_devices; i++) {
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
+ i, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) <<
+ "There should be no empty device names "
+ "among the ones the system gives us.";
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(i));
+ }
+}
+
+TEST_F(HardwareBeforeStreamingTest,
+ AllEnumeratedPlayoutDevicesCanBeSetAsPlayoutDevice) {
+ // Check playout side (see recording side test for more info on GUIDs).
+ int num_of_playout_devices = 0;
+ EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(
+ num_of_playout_devices));
+ EXPECT_GT(num_of_playout_devices, 0) << kNoDevicesErrorMessage;
+
+ char device_name[128] = {0};
+ char guid_name[128] = {0};
+
+ for (int i = 0; i < num_of_playout_devices; ++i) {
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
+ i, device_name, guid_name));
+ EXPECT_GT(strlen(device_name), 0u) <<
+ "There should be no empty device names "
+ "among the ones the system gives us.";
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(i));
+ }
+}
+
+TEST_F(HardwareBeforeStreamingTest,
+ SetDeviceWithMagicalArgumentsSetsDefaultSoundDevices) {
+#ifdef _WIN32
+ // -1 means "default device" on Windows.
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
+#else
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(0));
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(0));
+#endif
+}
+
+#endif // !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc
new file mode 100644
index 0000000..41145e1
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc
@@ -0,0 +1,168 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "audio_device.h"
+#include "voe_test_defines.h"
+
+class HardwareTest : public AfterStreamingFixture {
+};
+
+#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+TEST_F(HardwareTest, AbleToQueryForDevices) {
+ int num_recording_devices = 0;
+ int num_playout_devices = 0;
+ EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(num_recording_devices));
+ EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(num_playout_devices));
+
+ ASSERT_GT(num_recording_devices, 0) <<
+ "There seem to be no recording devices on your system, "
+ "and this test really doesn't make sense then.";
+ ASSERT_GT(num_playout_devices, 0) <<
+ "There seem to be no playout devices on your system, "
+ "and this test really doesn't make sense then.";
+
+ // Recording devices are handled a bit differently on Windows - we can
+ // just tell it to set the 'default' communication device there.
+#ifdef _WIN32
+ // Should also work while already recording.
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(
+ webrtc::AudioDeviceModule::kDefaultCommunicationDevice));
+ // Should also work while already playing.
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(
+ webrtc::AudioDeviceModule::kDefaultCommunicationDevice));
+#else
+ // For other platforms, just use the first device encountered.
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(0));
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(0));
+#endif
+
+ // It's hard to know what names this will return (it's system-dependent),
+ // so just check that it's possible to do it.
+ char device_name[128] = {0};
+ char guid_name[128] = {0};
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
+ 0, device_name, guid_name));
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
+ 0, device_name, guid_name));
+}
+#endif
+
+#ifdef _WIN32
+TEST_F(HardwareTest, GetCpuLoadWorksOnWindows) {
+ int load = -1;
+ EXPECT_EQ(0, voe_hardware_->GetCPULoad(load));
+ EXPECT_GE(0, load);
+ TEST_LOG("Voice engine CPU load = %d%%\n", load);
+}
+#else
+TEST_F(HardwareTest, GetCpuLoadReturnsErrorOnNonWindowsPlatform) {
+ int load = -1;
+ EXPECT_EQ(-1, voe_hardware_->GetCPULoad(load));
+}
+#endif
+
+#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
+TEST_F(HardwareTest, GetSystemCpuLoadWorksExceptOnMacAndAndroid) {
+#ifdef _WIN32
+ // This method needs some warm-up time on Windows. We sleep a good amount
+ // of time instead of retrying to make the test simpler.
+ Sleep(2000);
+#endif
+ int load = -1;
+ EXPECT_EQ(0, voe_hardware_->GetSystemCPULoad(load));
+ EXPECT_GE(load, 0);
+ TEST_LOG("System CPU load = %d%%\n", load);
+}
+#endif
+
+TEST_F(HardwareTest, BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer) {
+#ifdef MAC_IPHONE
+ // Ensure the sound device is reset on iPhone.
+ EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
+ Sleep(2000);
+#endif
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+
+ webrtc::AudioLayers given_layer;
+ EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer));
+ if (given_layer != webrtc::kAudioWindowsCore) {
+ // Not Windows Audio Core - then it shouldn't work.
+ EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
+ EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
+ return;
+ }
+
+ TEST_LOG("Testing AEC for Audio Windows Core.\n");
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+ // Can't be set after StartSend().
+ EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
+ EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
+
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(true));
+
+ // Can't be called before StartPlayout().
+ EXPECT_EQ(-1, voe_base_->StartSend(channel_));
+
+ EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ TEST_LOG("Processing capture data with built-in AEC...\n");
+ Sleep(2000);
+
+ TEST_LOG("Looping through capture devices...\n");
+ int num_devs = 0;
+ char dev_name[128] = { 0 };
+ char guid_name[128] = { 0 };
+ EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(num_devs));
+ for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
+ EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(dev_index,
+ dev_name,
+ guid_name));
+ TEST_LOG("%d: %s\n", dev_index, dev_name);
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(dev_index));
+ Sleep(2000);
+ }
+
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
+
+ TEST_LOG("Looping through render devices, restarting for each "
+ "device...\n");
+ EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(num_devs));
+ for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
+ EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(dev_index,
+ dev_name,
+ guid_name));
+ TEST_LOG("%d: %s\n", dev_index, dev_name);
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(dev_index));
+ Sleep(2000);
+ }
+
+ TEST_LOG("Using default devices...\n");
+ EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
+ EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
+ Sleep(2000);
+
+ // Possible, but not recommended before StopSend().
+ EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+ Sleep(2000); // To verify that there is no garbage audio.
+
+ TEST_LOG("Disabling built-in AEC.\n");
+ EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(false));
+
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc
new file mode 100644
index 0000000..68f28b4
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+
+// Note: This class includes sleeps and requires manual verification.
+class ManualHoldTest : public AfterStreamingFixture {
+};
+
+TEST_F(ManualHoldTest, SetOnHoldStatusBlockAudio) {
+ TEST_LOG("Channel not on hold => should hear audio.\n");
+ Sleep(2000);
+ TEST_LOG("Put channel on hold => should *not* hear audio.\n");
+ EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true));
+ Sleep(2000);
+ TEST_LOG("Remove on hold => should hear audio again.\n");
+ EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, false));
+ Sleep(2000);
+ TEST_LOG("Put sending on hold => should *not* hear audio.\n");
+ EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true, webrtc::kHoldSendOnly));
+ Sleep(2000);
+}
+
+TEST_F(ManualHoldTest, SetOnHoldStatusBlocksLocalFileAudio) {
+ TEST_LOG("Start playing a file locally => "
+ "you should now hear this file being played out.\n");
+ voe_file_->StopPlayingFileAsMicrophone(channel_);
+ EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
+ channel_, resource_manager_.long_audio_file_path().c_str(), true));
+ Sleep(2000);
+
+ TEST_LOG("Put playing on hold => should *not* hear audio.\n");
+ EXPECT_EQ(0, voe_base_->SetOnHoldStatus(
+ channel_, true, webrtc::kHoldPlayOnly));
+ Sleep(2000);
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc
new file mode 100644
index 0000000..8184535
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+
+class NetEQTest : public AfterStreamingFixture {
+ protected:
+ void SetUp() {
+ additional_channel_ = voe_base_->CreateChannel();
+ }
+
+ void TearDown() {
+ voe_base_->DeleteChannel(additional_channel_);
+ }
+
+ int additional_channel_;
+};
+
+TEST_F(NetEQTest, GetNetEQPlayoutModeReturnsDefaultModeByDefault) {
+ webrtc::NetEqModes mode;
+ EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
+ EXPECT_EQ(webrtc::kNetEqDefault, mode);
+}
+
+TEST_F(NetEQTest, SetNetEQPlayoutModeActuallySetsTheModeForTheChannel) {
+ webrtc::NetEqModes mode;
+ // Set for the first channel but leave the second.
+ EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqFax));
+ EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
+ EXPECT_EQ(webrtc::kNetEqFax, mode);
+
+ EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(additional_channel_, mode));
+ EXPECT_EQ(webrtc::kNetEqDefault, mode);
+
+ // Set the second channel, leave the first.
+ EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(
+ additional_channel_, webrtc::kNetEqStreaming));
+ EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(additional_channel_, mode));
+ EXPECT_EQ(webrtc::kNetEqStreaming, mode);
+
+ EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
+ EXPECT_EQ(webrtc::kNetEqFax, mode);
+}
+
+TEST_F(NetEQTest, GetNetEQBgnModeReturnsBgnOnByDefault) {
+ webrtc::NetEqBgnModes bgn_mode;
+ EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
+ EXPECT_EQ(webrtc::kBgnOn, bgn_mode);
+}
+
+TEST_F(NetEQTest, SetNetEQBgnModeActuallySetsTheBgnMode) {
+ webrtc::NetEqBgnModes bgn_mode;
+ EXPECT_EQ(0, voe_base_->SetNetEQBGNMode(channel_, webrtc::kBgnOff));
+ EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
+ EXPECT_EQ(webrtc::kBgnOff, bgn_mode);
+
+ EXPECT_EQ(0, voe_base_->SetNetEQBGNMode(channel_, webrtc::kBgnFade));
+ EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
+ EXPECT_EQ(webrtc::kBgnFade, bgn_mode);
+}
+
+TEST_F(NetEQTest, ManualSetEQPlayoutModeStillProducesOkAudio) {
+ EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqDefault));
+ TEST_LOG("NetEQ default playout mode enabled => should hear OK audio.\n");
+ Sleep(2000);
+
+ EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(
+ channel_, webrtc::kNetEqStreaming));
+ TEST_LOG("NetEQ streaming playout mode enabled => should hear OK audio.\n");
+ Sleep(2000);
+
+ EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqFax));
+ TEST_LOG("NetEQ fax playout mode enabled => should hear OK audio.\n");
+ Sleep(2000);
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc
new file mode 100644
index 0000000..7a41e80
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_initialization_fixture.h"
+
+class NetworkBeforeStreamingTest : public AfterInitializationFixture {
+ protected:
+ void SetUp() {
+ channel_ = voe_base_->CreateChannel();
+ }
+
+ void TearDown() {
+ voe_base_->DeleteChannel(channel_);
+ }
+
+ int channel_;
+};
+
+TEST_F(NetworkBeforeStreamingTest,
+ GetSourceInfoReturnsEmptyValuesForUnconfiguredChannel) {
+ char src_ip[32] = "0.0.0.0";
+ int src_rtp_port = 1234;
+ int src_rtcp_port = 1235;
+
+ EXPECT_EQ(0, voe_network_->GetSourceInfo(
+ channel_, src_rtp_port, src_rtcp_port, src_ip));
+ EXPECT_EQ(0, src_rtp_port);
+ EXPECT_EQ(0, src_rtcp_port);
+ EXPECT_STRCASEEQ("", src_ip);
+}
+
+TEST_F(NetworkBeforeStreamingTest,
+ GetSourceFilterReturnsEmptyValuesForUnconfiguredChannel) {
+ int filter_port = -1;
+ int filter_port_rtcp = -1;
+ char filter_ip[32] = "0.0.0.0";
+
+ EXPECT_EQ(0, voe_network_->GetSourceFilter(
+ channel_, filter_port, filter_port_rtcp, filter_ip));
+
+ EXPECT_EQ(0, filter_port);
+ EXPECT_EQ(0, filter_port_rtcp);
+ EXPECT_STRCASEEQ("", filter_ip);
+}
+
+TEST_F(NetworkBeforeStreamingTest, SetSourceFilterSucceeds) {
+ EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc
new file mode 100644
index 0000000..335fba6
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc
@@ -0,0 +1,199 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "mock/mock_voe_observer.h"
+#include "mock/mock_voe_connection_observer.h"
+#include "voe_test_interface.h"
+
+static const int kDefaultRtpPort = 8000;
+static const int kDefaultRtcpPort = 8001;
+
+class NetworkTest : public AfterStreamingFixture {
+};
+
+using ::testing::Between;
+
+TEST_F(NetworkTest, GetSourceInfoReturnsPortsAndIpAfterReceivingPackets) {
+ // Give some time to send speech packets.
+ Sleep(200);
+
+ int rtp_port = 0;
+ int rtcp_port = 0;
+ char source_ip[32] = "127.0.0.1";
+
+ EXPECT_EQ(0, voe_network_->GetSourceInfo(channel_, rtp_port, rtcp_port,
+ source_ip));
+
+ EXPECT_EQ(kDefaultRtpPort, rtp_port);
+ EXPECT_EQ(kDefaultRtcpPort, rtcp_port);
+}
+
+TEST_F(NetworkTest, NoFilterIsEnabledByDefault) {
+ int filter_rtp_port = -1;
+ int filter_rtcp_port = -1;
+ char filter_ip[64] = { 0 };
+
+ EXPECT_EQ(0, voe_network_->GetSourceFilter(
+ channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
+
+ EXPECT_EQ(0, filter_rtp_port);
+ EXPECT_EQ(0, filter_rtcp_port);
+ EXPECT_STREQ("", filter_ip);
+}
+
+TEST_F(NetworkTest, ManualCanFilterRtpPort) {
+ TEST_LOG("No filter, should hear audio.\n");
+ Sleep(1000);
+
+ int port_to_block = kDefaultRtpPort + 10;
+ EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, port_to_block));
+
+ // Changes should take effect immediately.
+ int filter_rtp_port = -1;
+ int filter_rtcp_port = -1;
+ char filter_ip[64] = { 0 };
+
+ EXPECT_EQ(0, voe_network_->GetSourceFilter(
+ channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
+
+ EXPECT_EQ(port_to_block, filter_rtp_port);
+
+ TEST_LOG("Now filtering port %d, should not hear audio.\n", port_to_block);
+ Sleep(1000);
+
+ TEST_LOG("Removing filter, should hear audio.\n");
+ EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
+ Sleep(1000);
+}
+
+TEST_F(NetworkTest, ManualCanFilterIp) {
+ TEST_LOG("You should hear audio.\n");
+ Sleep(1000);
+
+ int rtcp_port_to_block = kDefaultRtcpPort + 10;
+ TEST_LOG("Filtering IP 10.10.10.10, should not hear audio.\n");
+ EXPECT_EQ(0, voe_network_->SetSourceFilter(
+ channel_, 0, rtcp_port_to_block, "10.10.10.10"));
+
+ int filter_rtp_port = -1;
+ int filter_rtcp_port = -1;
+ char filter_ip[64] = { 0 };
+ EXPECT_EQ(0, voe_network_->GetSourceFilter(
+ channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
+
+ EXPECT_EQ(0, filter_rtp_port);
+ EXPECT_EQ(rtcp_port_to_block, filter_rtcp_port);
+ EXPECT_STREQ("10.10.10.10", filter_ip);
+}
+
+TEST_F(NetworkTest,
+ CallsObserverOnTimeoutAndRestartWhenPacketTimeoutNotificationIsEnabled) {
+ // First, get rid of the default, asserting observer and install our observer.
+ EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
+ webrtc::MockVoEObserver mock_observer;
+ EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(mock_observer));
+
+ // Define expectations.
+ int expected_error = VE_RECEIVE_PACKET_TIMEOUT;
+ EXPECT_CALL(mock_observer, CallbackOnError(channel_, expected_error))
+ .Times(1);
+ expected_error = VE_PACKET_RECEIPT_RESTARTED;
+ EXPECT_CALL(mock_observer, CallbackOnError(channel_, expected_error))
+ .Times(1);
+
+ // Get some speech going.
+ Sleep(500);
+
+ // Enable packet timeout.
+ EXPECT_EQ(0, voe_network_->SetPacketTimeoutNotification(channel_, true, 1));
+
+ // Trigger a timeout.
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ Sleep(1500);
+
+ // Trigger a restart event.
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ Sleep(500);
+}
+
+TEST_F(NetworkTest, DoesNotCallDeRegisteredObserver) {
+ // De-register the default observer. This test will fail if the observer gets
+ // called for any reason, so if this de-register doesn't work the test will
+ // fail.
+ EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
+
+ // Get some speech going.
+ Sleep(500);
+
+ // Enable packet timeout.
+ EXPECT_EQ(0, voe_network_->SetPacketTimeoutNotification(channel_, true, 1));
+
+ // Trigger a timeout.
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ Sleep(1500);
+}
+
+TEST_F(NetworkTest, DeadOrAliveObserverSeesAliveMessagesIfEnabled) {
+ webrtc::MockVoeConnectionObserver mock_observer;
+ EXPECT_EQ(0, voe_network_->RegisterDeadOrAliveObserver(
+ channel_, mock_observer));
+
+ // We should be called about 4 times in four seconds, but 3 is OK too.
+ EXPECT_CALL(mock_observer, OnPeriodicDeadOrAlive(channel_, true))
+ .Times(Between(3, 4));
+
+ EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
+ Sleep(4000);
+
+ EXPECT_EQ(0, voe_network_->DeRegisterDeadOrAliveObserver(channel_));
+}
+
+TEST_F(NetworkTest, DeadOrAliveObserverSeesDeadMessagesIfEnabled) {
+ // "When do you see them?" - "All the time!"
+ webrtc::MockVoeConnectionObserver mock_observer;
+ EXPECT_EQ(0, voe_network_->RegisterDeadOrAliveObserver(
+ channel_, mock_observer));
+
+ Sleep(500);
+
+ // We should be called about 4 times in four seconds, but 3 is OK too.
+ EXPECT_CALL(mock_observer, OnPeriodicDeadOrAlive(channel_, false))
+ .Times(Between(3, 4));
+
+ EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, false));
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ Sleep(4000);
+
+ EXPECT_EQ(0, voe_network_->DeRegisterDeadOrAliveObserver(channel_));
+}
+
+TEST_F(NetworkTest, CanSwitchToExternalTransport) {
+ EXPECT_EQ(0, voe_base_->StopReceive(channel_));
+ EXPECT_EQ(0, voe_base_->DeleteChannel(channel_));
+ channel_ = voe_base_->CreateChannel();
+
+ voetest::FakeExternalTransport external_transport(voe_network_);
+ EXPECT_EQ(0, voe_network_->RegisterExternalTransport(
+ channel_, external_transport));
+
+ EXPECT_EQ(0, voe_base_->StartReceive(channel_));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+ EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+
+ Sleep(1000);
+
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+ EXPECT_EQ(0, voe_base_->StopReceive(channel_));
+
+ EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
new file mode 100644
index 0000000..afd0820
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_initialization_fixture.h"
+
+using namespace webrtc;
+using namespace testing;
+
+class RtpRtcpBeforeStreamingTest : public AfterInitializationFixture {
+ protected:
+ void SetUp();
+ void TearDown();
+
+ int channel_;
+};
+
+void RtpRtcpBeforeStreamingTest::SetUp() {
+ EXPECT_THAT(channel_ = voe_base_->CreateChannel(), Not(Lt(0)));
+}
+
+void RtpRtcpBeforeStreamingTest::TearDown() {
+ EXPECT_EQ(0, voe_base_->DeleteChannel(channel_));
+}
+
+TEST_F(RtpRtcpBeforeStreamingTest,
+ GetRtcpStatusReturnsTrueByDefaultAndObeysSetRtcpStatus) {
+ bool on = false;
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
+ EXPECT_TRUE(on);
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, false));
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
+ EXPECT_FALSE(on);
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, true));
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
+ EXPECT_TRUE(on);
+}
+
+TEST_F(RtpRtcpBeforeStreamingTest, RtpKeepAliveStatusIsOffByDefault) {
+ unsigned char payload_type;
+ int delta_seconds;
+ bool on;
+
+ // Should be off by default.
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
+ channel_, on, payload_type, delta_seconds));
+ EXPECT_FALSE(on);
+ EXPECT_EQ(255, payload_type);
+ EXPECT_EQ(0, delta_seconds);
+}
+
+TEST_F(RtpRtcpBeforeStreamingTest, SetRtpKeepAliveDealsWithInvalidParameters) {
+ unsigned char payload_type;
+ int delta_seconds;
+ bool on;
+
+ // Verify invalid input parameters.
+ EXPECT_NE(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
+ -1, on, payload_type, delta_seconds)) <<
+ "Should fail for invalid channel -1.";
+ EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ -1, true, 0, 15)) <<
+ "Should fail for invalid channel -1.";
+ EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, true, -1, 15)) <<
+ "Should fail for invalid payload -1.";
+ EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, true, 0, 61)) <<
+ "The delta time must be [1, 60] seconds.";
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
+ channel_, on, payload_type, delta_seconds));
+ EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, true, 0));
+
+ // Should still be off, default 0 used by PCMU.
+ EXPECT_FALSE(on);
+}
+
+TEST_F(RtpRtcpBeforeStreamingTest,
+ GetRtpKeepaliveStatusObeysSetRtpKeepaliveStatus) {
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCP_CNAME(channel_, "SomeName"));
+
+ // Try valid settings.
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, true, 1));
+
+ unsigned char payload_type;
+ int delta_seconds;
+ bool on;
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
+ 0, on, payload_type, delta_seconds));
+ EXPECT_TRUE(on);
+ EXPECT_EQ(1, payload_type);
+ EXPECT_EQ(15, delta_seconds) << "15 seconds delta is default.";
+
+ // Set the keep-alive payload to 60, which the codecs can't use.
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, true, 60, 3));
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
+ channel_, on, payload_type, delta_seconds));
+ EXPECT_TRUE(on);
+ EXPECT_EQ(60, payload_type);
+ EXPECT_EQ(3, delta_seconds);
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
+ channel_, false, 60));
+}
+
+TEST_F(RtpRtcpBeforeStreamingTest, GetLocalSsrcObeysSetLocalSsrc) {
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, 1234));
+ unsigned int result = 0;
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, result));
+ EXPECT_EQ(1234u, result);
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc
new file mode 100644
index 0000000..42c1b4e
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc
@@ -0,0 +1,195 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+#include "voe_standard_test.h"
+#include "testsupport/fileutils.h"
+
+static const char* const RTCP_CNAME = "Whatever";
+
+class RtpRtcpTest : public AfterStreamingFixture {
+ protected:
+ void SetUp() {
+ // We need a second channel for this test, so set it up.
+ second_channel_ = voe_base_->CreateChannel();
+ EXPECT_GE(second_channel_, 0);
+
+ EXPECT_EQ(0, voe_base_->SetSendDestination(
+ second_channel_, 8002, "127.0.0.1"));
+ EXPECT_EQ(0, voe_base_->SetLocalReceiver(
+ second_channel_, 8002));
+ EXPECT_EQ(0, voe_base_->StartReceive(second_channel_));
+ EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
+ EXPECT_EQ(0, voe_base_->StartSend(second_channel_));
+
+ // We'll set up the RTCP CNAME and SSRC to something arbitrary here.
+ voe_rtp_rtcp_->SetRTCP_CNAME(channel_, RTCP_CNAME);
+ }
+
+ void TearDown() {
+ voe_base_->DeleteChannel(second_channel_);
+ }
+
+ int second_channel_;
+};
+
+TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
+ // We need to sleep a bit here for the name to propagate. For instance,
+ // 200 milliseconds is not enough, so we'll go with one second here.
+ Sleep(1000);
+
+ char char_buffer[256];
+ voe_rtp_rtcp_->GetRemoteRTCP_CNAME(channel_, char_buffer);
+ EXPECT_STREQ(RTCP_CNAME, char_buffer);
+}
+
+TEST_F(RtpRtcpTest, SSRCPropagatesCorrectly) {
+ unsigned int local_ssrc = 1234;
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, local_ssrc));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+ Sleep(1000);
+
+ unsigned int ssrc;
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
+ EXPECT_EQ(local_ssrc, ssrc);
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
+ EXPECT_EQ(local_ssrc, ssrc);
+}
+
+TEST_F(RtpRtcpTest, RtcpApplicationDefinedPacketsCanBeSentAndReceived) {
+ voetest::RtcpAppHandler rtcp_app_handler;
+ EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTCPObserver(
+ channel_, rtcp_app_handler));
+
+ // Send data aligned to 32 bytes.
+ const char* data = "application-dependent data------";
+ unsigned short data_length = strlen(data);
+ unsigned int data_name = 0x41424344; // 'ABCD' in ascii
+ unsigned char data_subtype = 1;
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->SendApplicationDefinedRTCPPacket(
+ channel_, data_subtype, data_name, data, data_length));
+
+ // Ensure the RTP-RTCP process gets scheduled.
+ Sleep(1000);
+
+ // Ensure we received the data in the callback.
+ EXPECT_EQ(data_length, rtcp_app_handler.length_in_bytes_);
+ EXPECT_EQ(0, memcmp(data, rtcp_app_handler.data_, data_length));
+ EXPECT_EQ(data_name, rtcp_app_handler.name_);
+ EXPECT_EQ(data_subtype, rtcp_app_handler.sub_type_);
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTCPObserver(channel_));
+}
+
+TEST_F(RtpRtcpTest, DisabledRtcpObserverDoesNotReceiveData) {
+ voetest::RtcpAppHandler rtcp_app_handler;
+ EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTCPObserver(
+ channel_, rtcp_app_handler));
+
+ // Put observer in a known state before de-registering.
+ rtcp_app_handler.Reset();
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTCPObserver(channel_));
+
+ const char* data = "whatever";
+ EXPECT_EQ(0, voe_rtp_rtcp_->SendApplicationDefinedRTCPPacket(
+ channel_, 1, 0x41424344, data, strlen(data)));
+
+ // Ensure the RTP-RTCP process gets scheduled.
+ Sleep(1000);
+
+ // Ensure we received no data.
+ EXPECT_EQ(0u, rtcp_app_handler.name_);
+ EXPECT_EQ(0u, rtcp_app_handler.sub_type_);
+}
+
+TEST_F(RtpRtcpTest, InsertExtraRTPPacketDealsWithInvalidArguments) {
+ const char payload_data[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
+
+ EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ -1, 0, false, payload_data, 8)) <<
+ "Should reject: invalid channel.";
+ EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, -1, false, payload_data, 8)) <<
+ "Should reject: invalid payload type.";
+ EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, 128, false, payload_data, 8)) <<
+ "Should reject: invalid payload type.";
+ EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, 99, false, NULL, 8)) <<
+ "Should reject: bad pointer.";
+ EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, 99, false, payload_data, 1500 - 28 + 1)) <<
+ "Should reject: invalid size.";
+}
+
+TEST_F(RtpRtcpTest, CanTransmitExtraRtpPacketsWithoutError) {
+ const char payload_data[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
+
+ for (int i = 0; i < 128; ++i) {
+ // Try both with and without the marker bit set
+ EXPECT_EQ(0, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, i, false, payload_data, 8));
+ EXPECT_EQ(0, voe_rtp_rtcp_->InsertExtraRTPPacket(
+ channel_, i, true, payload_data, 8));
+ }
+}
+
+TEST_F(RtpRtcpTest, CanCreateRtpDumpFilesWithoutError) {
+ // Create two RTP dump files (3 seconds long). You can verify these after
+ // the test using rtpplay or NetEqRTPplay if you like.
+ std::string output_path = webrtc::test::OutputPath();
+ std::string incoming_filename = output_path + "dump_in_3sec.rtp";
+ std::string outgoing_filename = output_path + "dump_out_3sec.rtp";
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
+ channel_, incoming_filename.c_str(), webrtc::kRtpIncoming));
+ EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
+ channel_, outgoing_filename.c_str(), webrtc::kRtpOutgoing));
+
+ Sleep(3000);
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpIncoming));
+ EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpOutgoing));
+}
+
+TEST_F(RtpRtcpTest, ObserverGetsNotifiedOnSsrcChange) {
+ voetest::TestRtpObserver rtcp_observer;
+ EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTPObserver(
+ channel_, rtcp_observer));
+ rtcp_observer.Reset();
+
+ unsigned int new_ssrc = 7777;
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, new_ssrc));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+ Sleep(500);
+
+ // Verify we got the new SSRC.
+ EXPECT_EQ(new_ssrc, rtcp_observer.ssrc_[0]);
+
+ // Now try another SSRC.
+ unsigned int newer_ssrc = 1717;
+ EXPECT_EQ(0, voe_base_->StopSend(channel_));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, newer_ssrc));
+ EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+ Sleep(500);
+
+ EXPECT_EQ(newer_ssrc, rtcp_observer.ssrc_[0]);
+
+ EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTPObserver(channel_));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc
new file mode 100644
index 0000000..0388025
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "before_initialization_fixture.h"
+
+class VoeBaseMiscTest : public BeforeInitializationFixture {
+};
+
+using namespace testing;
+
+TEST_F(VoeBaseMiscTest, MaxNumChannelsIs32) {
+ EXPECT_EQ(32, voe_base_->MaxNumOfChannels());
+}
+
+TEST_F(VoeBaseMiscTest, GetVersionPrintsSomeUsefulInformation) {
+ char char_buffer[1024];
+ EXPECT_EQ(0, voe_base_->GetVersion(char_buffer));
+ EXPECT_THAT(char_buffer, ContainsRegex("VoiceEngine"));
+}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc
new file mode 100644
index 0000000..44887b7
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc
@@ -0,0 +1,238 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "after_streaming_fixture.h"
+
+class VolumeTest : public AfterStreamingFixture {
+};
+
+TEST_F(VolumeTest, DefaultSpeakerVolumeIsAtMost255) {
+ unsigned int volume = 1000;
+ EXPECT_EQ(0, voe_volume_control_->GetSpeakerVolume(volume));
+ EXPECT_LE(volume, 255u);
+}
+
+TEST_F(VolumeTest, ManualSetVolumeWorks) {
+ unsigned int original_volume = 0;
+ EXPECT_EQ(0, voe_volume_control_->GetSpeakerVolume(original_volume));
+ Sleep(1000);
+
+ TEST_LOG("Setting speaker volume to 0 out of 255.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(0));
+ Sleep(1000);
+
+ TEST_LOG("Setting speaker volume to 100 out of 255.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(100));
+ Sleep(1000);
+
+ // Set the volume to 255 very briefly so we don't blast the poor user
+ // listening to this. This is just to test the call succeeds.
+ EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(255));
+
+ TEST_LOG("Setting speaker volume to the original %d out of 255.\n",
+ original_volume);
+ EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(original_volume));
+ Sleep(1000);
+}
+
+#if !defined(MAC_IPHONE)
+
+// NOTE(phoglund): This test is flaky because of how the OS works, and is hence
+// disabled by default.
+TEST_F(VolumeTest, DISABLED_DefaultMicrophoneVolumeIsAtMost255) {
+ unsigned int volume = 1000;
+ EXPECT_EQ(0, voe_volume_control_->GetMicVolume(volume));
+ EXPECT_LE(volume, 255u);
+}
+
+TEST_F(VolumeTest, ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAcgOff) {
+ SwitchToManualMicrophone();
+ EXPECT_EQ(0, voe_apm_->SetAgcStatus(false));
+
+ unsigned int original_volume = 0;
+ EXPECT_EQ(0, voe_volume_control_->GetMicVolume(original_volume));
+
+ TEST_LOG("Setting microphone volume to 0.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetMicVolume(channel_));
+ Sleep(1000);
+ TEST_LOG("Setting microphone volume to 255.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetMicVolume(255));
+ Sleep(1000);
+ TEST_LOG("Setting microphone volume back to saved value.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetMicVolume(original_volume));
+ Sleep(1000);
+}
+
+TEST_F(VolumeTest, ChannelScalingIsOneByDefault) {
+ float scaling = -1.0f;
+
+ EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
+ channel_, scaling));
+ EXPECT_FLOAT_EQ(1.0f, scaling);
+}
+
+TEST_F(VolumeTest, ManualCanSetChannelScaling) {
+ EXPECT_EQ(0, voe_volume_control_->SetChannelOutputVolumeScaling(
+ channel_, 0.1f));
+
+ float scaling = 1.0f;
+ EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
+ channel_, scaling));
+
+ EXPECT_FLOAT_EQ(0.1f, scaling);
+
+ TEST_LOG("Channel scaling set to 0.1: audio should be barely audible.\n");
+ Sleep(2000);
+}
+
+#endif // !MAC_IPHONE
+
+#if !defined(WEBRTC_ANDROID) && !defined(MAC_IPHONE)
+
+TEST_F(VolumeTest, InputMutingIsNotEnabledByDefault) {
+ bool is_muted = true;
+ EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
+ EXPECT_FALSE(is_muted);
+}
+
+TEST_F(VolumeTest, ManualInputMutingMutesMicrophone) {
+ SwitchToManualMicrophone();
+
+ // Enable muting.
+ EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, true));
+ bool is_muted = false;
+ EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
+ EXPECT_TRUE(is_muted);
+
+ TEST_LOG("Muted: talk into microphone and verify you can't hear yourself.\n");
+ Sleep(2000);
+
+ // Test that we can disable muting.
+ EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, false));
+ EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
+ EXPECT_FALSE(is_muted);
+
+ TEST_LOG("Unmuted: talk into microphone and verify you can hear yourself.\n");
+ Sleep(2000);
+}
+
+// NOTE(phoglund): This test is flaky because of how the OS works, and is hence
+// disabled by default.
+TEST_F(VolumeTest, DISABLED_SystemInputMutingIsNotEnabledByDefault) {
+ bool is_muted = true;
+ EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
+ EXPECT_FALSE(is_muted);
+}
+
+TEST_F(VolumeTest, ManualSystemInputMutingMutesMicrophone) {
+ SwitchToManualMicrophone();
+
+ // Enable system input muting.
+ EXPECT_EQ(0, voe_volume_control_->SetSystemInputMute(true));
+ bool is_muted = false;
+ EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
+ EXPECT_TRUE(is_muted);
+
+ TEST_LOG("Muted: talk into microphone and verify you can't hear yourself.\n");
+ Sleep(2000);
+
+ // Test that we can disable system input muting.
+ EXPECT_EQ(0, voe_volume_control_->SetSystemInputMute(false));
+ EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
+ EXPECT_FALSE(is_muted);
+
+ TEST_LOG("Unmuted: talk into microphone and verify you can hear yourself.\n");
+ Sleep(2000);
+}
+
+TEST_F(VolumeTest, SystemOutputMutingIsNotEnabledByDefault) {
+ bool is_muted = true;
+ EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
+ EXPECT_FALSE(is_muted);
+}
+
+TEST_F(VolumeTest, ManualSystemOutputMutingMutesOutput) {
+ // Enable muting.
+ EXPECT_EQ(0, voe_volume_control_->SetSystemOutputMute(true));
+ bool is_muted = false;
+ EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
+ EXPECT_TRUE(is_muted);
+
+ TEST_LOG("Muted: you should hear no audio.\n");
+ Sleep(2000);
+
+ // Test that we can disable muting.
+ EXPECT_EQ(0, voe_volume_control_->SetSystemOutputMute(false));
+ EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
+ EXPECT_FALSE(is_muted);
+
+ TEST_LOG("Unmuted: you should hear audio.\n");
+ Sleep(2000);
+}
+
+TEST_F(VolumeTest, ManualTestInputAndOutputLevels) {
+ SwitchToManualMicrophone();
+
+ TEST_LOG("Speak and verify that the following levels look right:\n");
+ for (int i = 0; i < 5; i++) {
+ Sleep(1000);
+ unsigned int input_level = 0;
+ unsigned int output_level = 0;
+ unsigned int input_level_full_range = 0;
+ unsigned int output_level_full_range = 0;
+
+ EXPECT_EQ(0, voe_volume_control_->GetSpeechInputLevel(
+ input_level));
+ EXPECT_EQ(0, voe_volume_control_->GetSpeechOutputLevel(
+ channel_, output_level));
+ EXPECT_EQ(0, voe_volume_control_->GetSpeechInputLevelFullRange(
+ input_level_full_range));
+ EXPECT_EQ(0, voe_volume_control_->GetSpeechOutputLevelFullRange(
+ channel_, output_level_full_range));
+
+ TEST_LOG(" warped levels (0-9) : in=%5d, out=%5d\n",
+ input_level, output_level);
+ TEST_LOG(" linear levels (0-32768): in=%5d, out=%5d\n",
+ input_level_full_range, output_level_full_range);
+ }
+}
+
+TEST_F(VolumeTest, ChannelsAreNotPannedByDefault) {
+ float left = -1.0;
+ float right = -1.0;
+
+ EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
+ EXPECT_FLOAT_EQ(1.0, left);
+ EXPECT_FLOAT_EQ(1.0, right);
+}
+
+TEST_F(VolumeTest, ManualTestChannelPanning) {
+ TEST_LOG("Panning left.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.8f, 0.1f));
+ Sleep(1000);
+
+ TEST_LOG("Back to center.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 1.0f, 1.0f));
+ Sleep(1000);
+
+ TEST_LOG("Panning right.\n");
+ EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.1f, 0.8f));
+ Sleep(1000);
+
+ // To finish, verify that the getter works.
+ float left = 0.0f;
+ float right = 0.0f;
+
+ EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
+ EXPECT_FLOAT_EQ(0.1f, left);
+ EXPECT_FLOAT_EQ(0.8f, right);
+}
+
+#endif // !WEBRTC_ANDROID && !MAC_IPHONE
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc
new file mode 100644
index 0000000..051bce9
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include <cassert>
+#if defined(_WIN32)
+#include <conio.h>
+#endif
+
+#include "voe_cpu_test.h"
+
+using namespace webrtc;
+
+namespace voetest {
+
+#define CHECK(expr) \
+ if (expr) \
+ { \
+ printf("Error at line: %i, %s \n", __LINE__, #expr); \
+ printf("Error code: %i \n", base->LastError()); \
+ PAUSE \
+ return -1; \
+ }
+
+extern char* GetFilename(char* filename);
+extern const char* GetFilename(const char* filename);
+extern int GetResource(char* resource, char* dest, int destLen);
+extern char* GetResource(char* resource);
+extern const char* GetResource(const char* resource);
+
+VoECpuTest::VoECpuTest(VoETestManager& mgr)
+ : _mgr(mgr) {
+
+}
+
+int VoECpuTest::DoTest() {
+ printf("------------------------------------------------\n");
+ printf(" CPU Reference Test\n");
+ printf("------------------------------------------------\n");
+
+ VoEBase* base = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoECodec* codec = _mgr.CodecPtr();
+ VoEAudioProcessing* apm = _mgr.APMPtr();
+
+ int channel(-1);
+ CodecInst isac;
+
+ isac.pltype = 104;
+ strcpy(isac.plname, "ISAC");
+ isac.pacsize = 960;
+ isac.plfreq = 32000;
+ isac.channels = 1;
+ isac.rate = -1;
+
+ CHECK(base->Init());
+ channel = base->CreateChannel();
+
+ CHECK(base->SetLocalReceiver(channel, 5566));
+ CHECK(base->SetSendDestination(channel, 5566, "127.0.0.1"));
+ CHECK(codec->SetRecPayloadType(channel, isac));
+ CHECK(codec->SetSendCodec(channel, isac));
+
+ CHECK(base->StartReceive(channel));
+ CHECK(base->StartPlayout(channel));
+ CHECK(base->StartSend(channel));
+ CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(),
+ true, true));
+
+ CHECK(codec->SetVADStatus(channel, true));
+ CHECK(apm->SetAgcStatus(true, kAgcAdaptiveAnalog));
+ CHECK(apm->SetNsStatus(true, kNsModerateSuppression));
+ CHECK(apm->SetEcStatus(true, kEcAec));
+
+ TEST_LOG("\nMeasure CPU and memory while running a full-duplex"
+ " iSAC-swb call.\n\n");
+
+ PAUSE
+
+ CHECK(base->StopSend(channel));
+ CHECK(base->StopPlayout(channel));
+ CHECK(base->StopReceive(channel));
+
+ base->DeleteChannel(channel);
+ CHECK(base->Terminate());
+
+ return 0;
+}
+
+} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h
new file mode 100644
index 0000000..f883075
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
+#define WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
+
+#include "voe_standard_test.h"
+
+namespace voetest {
+
+class VoETestManager;
+
+class VoECpuTest {
+ public:
+ VoECpuTest(VoETestManager& mgr);
+ ~VoECpuTest() {}
+ int DoTest();
+ private:
+ VoETestManager& _mgr;
+};
+
+} // namespace voetest
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc
new file mode 100644
index 0000000..8b4ad45
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc
@@ -0,0 +1,8421 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include <vector>
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include "voe_extended_test.h"
+#include "../../source/voice_engine_defines.h" // defines build macros
+#include "system_wrappers/interface/ref_count.h"
+
+#if defined(_WIN32)
+#include <conio.h>
+#include <winsock2.h>
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#include <netdb.h>
+#endif
+
+using namespace webrtc;
+
+namespace voetest {
+
+// Set this flag to ensure that test packets are transmitted to
+// RemoteIP::RemotePort during tests of SetSendToS and SetSendGQos. Requires
+// receiver at the remote side and Wireshark with a proper ip.src filter.
+#define _SEND_TO_REMOTE_IP_
+
+#ifdef _SEND_TO_REMOTE_IP_
+const int RemotePort = 12345; // transmit to this UDP port
+const char* RemoteIP = "192.168.200.1"; // transmit to this IP address
+#endif
+
+#ifdef MAC_IPHONE
+#define SLEEP_IF_IPHONE(x) SLEEP(x)
+#else
+#define SLEEP_IF_IPHONE(x)
+#endif
+
+#ifdef WEBRTC_ANDROID
+// Global pointers
+extern void* globalJavaVM;
+extern void* globalContext;
+#endif
+
+extern char* GetFilename(char* filename);
+extern const char* GetFilename(const char* filename);
+extern int GetResource(char* resource, char* dest, int destLen);
+extern char* GetResource(char* resource);
+extern const char* GetResource(const char* resource);
+
+// ----------------------------------------------------------------------------
+// External AudioDeviceModule implementation
+// ----------------------------------------------------------------------------
+
+// static
+AudioDeviceModuleImpl* AudioDeviceModuleImpl::Create() {
+ AudioDeviceModuleImpl* xADM = new AudioDeviceModuleImpl();
+ if (xADM)
+ xADM->AddRef();
+ return xADM;
+}
+
+// static
+bool AudioDeviceModuleImpl::Destroy(AudioDeviceModuleImpl* adm) {
+ if (!adm)
+ return false;
+ int32_t count = adm->Release();
+ if (count != 0) {
+ return false;
+ } else {
+ delete adm;
+ return true;
+ }
+}
+
+AudioDeviceModuleImpl::AudioDeviceModuleImpl() :
+ _ref_count(0) {
+}
+
+AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
+}
+
+int32_t AudioDeviceModuleImpl::AddRef() {
+ return ++_ref_count;
+}
+
+int32_t AudioDeviceModuleImpl::Release() {
+ // Avoid self destruction in this mock implementation.
+ // Ensures that we can always check the reference counter while alive.
+ return --_ref_count;
+}
+
+// ----------------------------------------------------------------------------
+// External transport (Transport) implementations:
+// ----------------------------------------------------------------------------
+
+ExtendedTestTransport::ExtendedTestTransport(VoENetwork* ptr) :
+ myNetw(ptr), _thread(NULL), _lock(NULL), _event(NULL), _length(0), _channel(0) {
+ const char* threadName = "voe_extended_test_external_thread";
+ _lock = CriticalSectionWrapper::CreateCriticalSection();
+ _event = EventWrapper::Create();
+ _thread = ThreadWrapper::CreateThread(Run, this, kHighPriority, threadName);
+ if (_thread) {
+ unsigned int id;
+ _thread->Start(id);
+ }
+}
+
+ExtendedTestTransport::~ExtendedTestTransport() {
+ if (_thread) {
+ _thread->SetNotAlive();
+ _event->Set();
+ if (_thread->Stop()) {
+ delete _thread;
+ _thread = NULL;
+ delete _event;
+ _event = NULL;
+ delete _lock;
+ _lock = NULL;
+ }
+ }
+}
+
+bool ExtendedTestTransport::Run(void* ptr) {
+ return static_cast<ExtendedTestTransport*> (ptr)->Process();
+}
+
+bool ExtendedTestTransport::Process() {
+ switch (_event->Wait(500)) {
+ case kEventSignaled:
+ _lock->Enter();
+ myNetw->ReceivedRTPPacket(_channel, _packetBuffer, _length);
+ _lock->Leave();
+ return true;
+ case kEventTimeout:
+ return true;
+ case kEventError:
+ break;
+ }
+ return true;
+}
+
+int ExtendedTestTransport::SendPacket(int channel, const void *data, int len) {
+ _lock->Enter();
+ if (len < 1612) {
+ memcpy(_packetBuffer, (const unsigned char*) data, len);
+ _length = len;
+ _channel = channel;
+ }
+ _lock->Leave();
+ _event->Set(); // triggers ReceivedRTPPacket() from worker thread
+ return len;
+}
+
+int ExtendedTestTransport::SendRTCPPacket(int channel, const void *data, int len) {
+ myNetw->ReceivedRTCPPacket(channel, data, len);
+ return len;
+}
+
+XTransport::XTransport(VoENetwork* netw, VoEFile* file) :
+ _netw(netw), _file(file) {
+}
+
+int XTransport::SendPacket(int channel, const void *data, int len) {
+ // loopback
+ // _netw->ReceivedRTPPacket(channel, data, len);
+
+ return 0;
+}
+
+int XTransport::SendRTCPPacket(int, const void *, int) {
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoERTPObserver
+// ----------------------------------------------------------------------------
+
+XRTPObserver::XRTPObserver() :
+ _SSRC(0) {
+}
+
+XRTPObserver::~XRTPObserver() {
+}
+
+void XRTPObserver::OnIncomingCSRCChanged(const int /*channel*/, const unsigned int /*CSRC*/,
+ const bool /*added*/) {
+}
+
+void XRTPObserver::OnIncomingSSRCChanged(const int /*channel*/, const unsigned int SSRC) {
+ // char msg[128];
+ // sprintf(msg, "OnIncomingSSRCChanged(channel=%d, SSRC=%lu)\n",
+ // channel, SSRC);
+ // TEST_LOG(msg);
+
+ _SSRC = SSRC; // skip channel dependency for simplicty
+
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::PrepareTest(const char* str) const {
+ TEST_LOG("\n\n================================================\n");
+ TEST_LOG("\tExtended *%s* Test\n", str);
+ TEST_LOG("================================================\n\n");
+
+ return 0;
+}
+
+int VoEExtendedTest::TestPassed(const char* str) const {
+ TEST_LOG("\n\n------------------------------------------------\n");
+ TEST_LOG("\tExtended *%s* test passed!\n", str);
+ TEST_LOG("------------------------------------------------\n\n");
+
+ return 0;
+}
+
+void VoEExtendedTest::OnPeriodicDeadOrAlive(const int /*channel*/, const bool alive) {
+ _alive = alive;
+ if (alive) {
+ TEST_LOG("=> ALIVE ");
+ } else {
+ TEST_LOG("=> DEAD ");
+ }
+ fflush(NULL);
+}
+
+void VoEExtendedTest::CallbackOnError(const int errCode, int) {
+ _errCode = errCode;
+ TEST_LOG("\n************************\n");
+ TEST_LOG(" RUNTIME ERROR: %d \n", errCode);
+ TEST_LOG("************************\n");
+}
+
+VoEExtendedTest::VoEExtendedTest(VoETestManager& mgr) :
+ _mgr(mgr) {
+ for (int i = 0; i < 32; i++) {
+ _listening[i] = false;
+ _playing[i] = false;
+ _sending[i] = false;
+ }
+}
+
+VoEExtendedTest::~VoEExtendedTest() {
+}
+
+void VoEExtendedTest::StartMedia(int channel, int rtpPort, bool listen,
+ bool playout, bool send) {
+ VoEBase* voe_base_ = _mgr.BasePtr();
+
+ _listening[channel] = false;
+ _playing[channel] = false;
+ _sending[channel] = false;
+
+ voe_base_->SetLocalReceiver(channel, rtpPort);
+ voe_base_->SetSendDestination(channel, rtpPort, "127.0.0.1");
+ if (listen) {
+ _listening[channel] = true;
+ voe_base_->StartReceive(channel);
+ }
+ if (playout) {
+ _playing[channel] = true;
+ voe_base_->StartPlayout(channel);
+ }
+ if (send) {
+ _sending[channel] = true;
+ voe_base_->StartSend(channel);
+ }
+}
+
+void VoEExtendedTest::StopMedia(int channel) {
+ VoEBase* voe_base_ = _mgr.BasePtr();
+
+ if (_listening[channel]) {
+ _listening[channel] = false;
+ voe_base_->StopReceive(channel);
+ }
+ if (_playing[channel]) {
+ _playing[channel] = false;
+ voe_base_->StopPlayout(channel);
+ }
+ if (_sending[channel]) {
+ _sending[channel] = false;
+ voe_base_->StopSend(channel);
+ }
+}
+
+void VoEExtendedTest::Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone,
+ bool addTimeMarker) {
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+
+ voe_base_->StartPlayout(channel);
+ TEST_LOG("[playing]");
+ fflush(NULL);
+ if (addFileAsMicrophone) {
+ file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(), true, true);
+ TEST_LOG("[file as mic]");
+ fflush(NULL);
+ }
+ if (addTimeMarker) {
+ float dtSec = (float) ((float) timeMillisec / 1000.0);
+ TEST_LOG("[dT=%.1f]", dtSec);
+ fflush(NULL); // print sleep time in seconds
+ }
+ SLEEP(timeMillisec);
+ voe_base_->StopPlayout(channel);
+ file->StopPlayingFileAsMicrophone(channel);
+}
+
+void VoEExtendedTest::Sleep(unsigned int timeMillisec, bool addMarker) {
+ if (addMarker) {
+ float dtSec = (float) ((float) timeMillisec / 1000.0);
+ TEST_LOG("[dT=%.1f]", dtSec); // print sleep time in seconds
+ }
+ ::Sleep(timeMillisec);
+}
+
+int VoEExtendedTest::TestBase() {
+#ifndef _WIN32
+ // Sleep a bit instead when pause not supported
+#undef PAUSE
+#define PAUSE SLEEP(2000);
+#endif
+
+ PrepareTest("Base");
+
+ // TODO(qhogpat): make this an actual instance variable. I think the
+ // macro black magic will make more sense then. This is named like an
+ // instance variable since it is required in order to appease the
+ // gods of darkness.
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoENetwork* netw = _mgr.NetworkPtr();
+#ifdef _TEST_RTP_RTCP_
+ VoERTP_RTCP* rtp = _mgr.RTP_RTCPPtr();
+#endif
+
+ //////////////////////////
+ // SetTraceFileName
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST(SetTraceFileName - SetDebugTraceFileName); ANL();
+
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(NULL)); MARK();
+ // don't use these files
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
+ "VoEBase_trace_dont_use.txt"))); MARK();
+ // use these instead
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
+ "VoEBase_trace.txt"))); MARK();
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStream |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo)); MARK();
+
+ ANL(); AOK(); ANL(); ANL();
+#endif
+
+ ///////////////////////////////////////
+ // RegisterVoiceEngineObserver
+ // DeRegisterVoiceEngineObserver
+ TEST(SetObserver);
+ ANL();
+
+ TEST_MUSTPASS(voe_base_->RegisterVoiceEngineObserver(*this));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->DeRegisterVoiceEngineObserver());
+ MARK();
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ /////////////////////
+ // GetVersion
+ TEST(GetVersion);
+ ANL();
+
+ char version[1024];
+ // audio device module and AudioProcessing fail to getversion when they
+ // are not initiliazed
+ TEST_MUSTPASS(voe_base_->GetVersion(version));
+ MARK();
+ TEST_LOG("\n-----\n%s\n-----\n", version);
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ ///////////////
+ // Init
+ TEST(Init);
+ ANL();
+
+ TEST_MUSTPASS(voe_base_->Init());
+ MARK();
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ TEST_MUSTPASS(voe_base_->Init());
+ MARK();
+ // ensure that no new memory is allocated at the second call (check
+ // trace file)
+ TEST_MUSTPASS(voe_base_->Init());
+ MARK();
+ TEST_MUSTPASS(voe_base_->Terminate());
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // verify AEC recording
+ TEST_MUSTPASS(voe_base_->Init());
+ MARK(); // verify output dat-files
+ TEST_MUSTPASS(voe_base_->Terminate());
+#endif
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ ////////////////////
+ // Terminate
+ TEST(Terminate);
+ ANL();
+ TEST_MUSTPASS(voe_base_->Terminate());
+ MARK(); // should be ignored
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->Terminate());
+ MARK(); // should terminate
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // ------------------------------------------------------------------------
+ // >> Init(AudioDeviceModule)
+ //
+ // Note that our mock implementation of the ADM also mocks the
+ // reference counting part. This approach enables us to keep track
+ // of the internal reference counter without checking return values
+ // from the ADM and we also avoid the built-in self destruction.
+ //
+ // TODO(henrika): this test does not verify that external ADM methods
+ // are called by the VoiceEngine once registered. We could extend
+ // the mock implementation and add counters for each ADM API to ensure
+ // that they are called in the correct sequence and the correct number
+ // of times.
+ TEST_LOG("\nTesting: Init in combination with an external ADM\n");
+
+ // Create the ADM and call AddRef within the factory method.
+ AudioDeviceModuleImpl* xADM = AudioDeviceModuleImpl::Create();
+ ASSERT_FALSE(xADM == NULL);
+ ASSERT_TRUE(xADM->ReferenceCounter() == 1);
+
+ // Verify default usage case for external ADM.
+ TEST_MUSTPASS(voe_base_->Init(xADM));MARK();
+ ASSERT_TRUE(xADM->ReferenceCounter() == 2);
+ TEST_MUSTPASS(voe_base_->Terminate());
+ ASSERT_TRUE(xADM->ReferenceCounter() == 1);
+
+ // Our reference-count implementation does not self destruct.
+ // We do it manually here instead by calling Release followed by delete.
+ ASSERT_TRUE(AudioDeviceModuleImpl::Destroy(xADM));
+ ANL();
+ AOK();ANL();
+
+ // >> end of Init(AudioDeviceModule)
+ // ------------------------------------------------------------------------
+
+ ///////////////////////////
+ // MaxNumOfChannels
+ TEST(MaxNumOfChannels);
+ ANL();
+ TEST_MUSTPASS(voe_base_->MaxNumOfChannels() < 0);
+ MARK();
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ ////////////////////////
+ // CreateChannel
+ // DeleteChannel
+
+ int i;
+ int channel;
+ int nChannels(voe_base_->MaxNumOfChannels());
+
+ TEST(CreateChannel);
+ ANL();
+ TEST(DeleteChannel);
+ ANL();
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ channel = voe_base_->CreateChannel();
+ MARK();
+ TEST_MUSTPASS(channel != 0);
+ channel = voe_base_->CreateChannel();
+ MARK();
+ TEST_MUSTPASS(channel != 1);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ MARK();
+ TEST_MUSTPASS(voe_base_->DeleteChannel(1));
+ MARK();
+
+ // create and delete one channel many times
+ for (i = 0; i < 10; i++) {
+ channel = voe_base_->CreateChannel();
+ MARK();
+ TEST_MUSTPASS(channel != 0); // should be 0 each time
+ TEST_MUSTPASS(voe_base_->DeleteChannel(channel));
+ MARK();
+ }
+ // create max number of channels
+ for (i = 0; i < nChannels; i++) {
+ channel = voe_base_->CreateChannel();
+ MARK();
+ TEST_MUSTPASS(channel != i);
+ }
+ channel = voe_base_->CreateChannel();
+ MARK(); // should fail since no more channels can now be created
+ TEST_MUSTPASS(channel != -1);
+
+ int aChannel = (((nChannels - 17) > 0) ? (nChannels - 17) : 0);
+ TEST_MUSTPASS(voe_base_->DeleteChannel(aChannel));
+ MARK();
+ channel = voe_base_->CreateChannel();
+ MARK(); // should reuse channel
+ TEST_MUSTPASS(channel != aChannel);
+
+ // delete all created channels
+ for (i = 0; i < nChannels; i++) {
+ TEST_MUSTPASS(voe_base_->DeleteChannel(i));
+ MARK();
+ }
+
+ // try to delete a non-existing channel
+ TEST_MUSTPASS(-1 != voe_base_->DeleteChannel(aChannel));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // ------------------------------------------------------------------------
+ // >> SetLocalReceiver
+ //
+ // State: VE not initialized, no existing channels
+ TEST_MUSTPASS(voe_base_->Init());
+
+ int ch;
+
+ TEST(SetLocalReceiver);
+ ANL();
+
+ // no channel created yet => should fail
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 100));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ch = voe_base_->CreateChannel();
+
+#ifdef MAC_IPHONE
+ printf("\nNOTE: Local IP must be set in source code (line %d) \n",
+ __LINE__ + 1);
+ char* localIp = "127.0.0.1";
+#else
+ char localIp[64] = { 0 };
+ TEST_MUSTPASS(netw->GetLocalIP(localIp));
+ MARK();
+ // NOTE: This API is supported on Win, Mac and Linux and may fail or not
+ // return local IP for other platforms.
+#endif
+
+ // trivial invalid function calls
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch+1, 12345));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, -1));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR);
+
+ // check conflict with ongoing receiving
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ TEST_ERROR(VE_ALREADY_LISTENING);
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // check conflict with ongoing transmission
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ TEST_ERROR(VE_ALREADY_SENDING);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+
+ // valid function calls
+ // Need to sleep between, otherwise it may fail for unknown reason
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, NULL,
+ "230.1.2.3"));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp,
+ "230.1.2.3"));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 5555, NULL));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ SLEEP(100);
+
+ // STATE: no media but sockets exists and are binded to 12345 and 12346
+ // respectively
+
+ // Add some dynamic tests as well:
+
+ // ensure that last setting is used (cancels old settings)
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
+ MARK();
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 54321));
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 54321, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ Play(ch, 1000, true, true);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetLocalReceiver
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> GetLocalReceiver
+ //
+ // State: VE initialized, no existing channels
+ TEST(GetLocalReceiver);
+ ANL();
+
+ int port;
+ char ipaddr[64];
+ int RTCPport;
+
+ ch = voe_base_->CreateChannel();
+
+ // verify non-configured (blank) local receiver
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
+ MARK();
+ TEST_MUSTPASS(port != 0);
+ TEST_MUSTPASS(RTCPport != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ // check some trivial set/get combinations
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345))
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
+ MARK();
+ TEST_MUSTPASS(port != 12345);
+ TEST_MUSTPASS(RTCPport != 12346);
+ TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0); // now binded to "any" IP
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 55555))
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
+ MARK();
+ TEST_MUSTPASS(port != 12345);
+ TEST_MUSTPASS(RTCPport != 55555);
+ TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp))
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
+ MARK();
+ TEST_MUSTPASS(port != 12345);
+ TEST_MUSTPASS(RTCPport != 12346);
+ TEST_MUSTPASS(strcmp(ipaddr, localIp) != 0);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of GetLocalReceiver
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetSendDestination
+ //
+ // State: VE initialized, no existing channels
+ TEST(SetSendDestination);
+ ANL();
+
+ // call without existing channel
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ch = voe_base_->CreateChannel();
+
+ // trivial fail tests
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 65536, "127.0.0.1"));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTP port
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 65536));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid source port
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
+ 65536));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTCP port
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.300"));
+ MARK();
+ TEST_ERROR(VE_INVALID_IP_ADDRESS); // invalid IP address
+
+ // sockets must be created first to support multi-cast (not required
+ // otherwise)
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
+ MARK();
+ TEST_ERROR(VE_SOCKET_ERROR);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555)); // create sockets
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
+ MARK(); // should work now
+
+ voe_base_->DeleteChannel(0);
+ ch = voe_base_->CreateChannel();
+
+ // STATE: one channel created, no sockets exist
+
+ // valid function calls
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", kVoEDefault,
+ 55555));
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444,
+ 55555));
+ MARK();
+
+ voe_base_->DeleteChannel(0);
+ ch = voe_base_->CreateChannel();
+
+ // create receive sockets first and then an extra pair of send sockets
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1", 11111));
+ MARK(); // binds to 11111
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetSendDestination
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> GetSendDestination
+ //
+ // State: VE initialized, no existing channels
+ TEST(GetSendDestination);
+ ANL();
+
+ int sourcePort;
+
+ ch = voe_base_->CreateChannel();
+
+ // verify non-configured (blank) local receiver
+ TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
+ RTCPport));
+ MARK();
+ TEST_MUSTPASS(port != 0);
+ TEST_MUSTPASS(sourcePort != 0);
+ TEST_MUSTPASS(RTCPport != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ // check some trivial set/get combinations
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
+ RTCPport));
+ MARK();
+ TEST_MUSTPASS(port != 44444);
+ TEST_MUSTPASS(sourcePort != 0); // should be 0 since no local receiver has
+ // NOT been defined yet
+ TEST_MUSTPASS(RTCPport != 44445);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
+ TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
+ RTCPport));
+ MARK();
+ TEST_MUSTPASS(port != 44444);
+ TEST_MUSTPASS(sourcePort != 55555); // should be equal to local port
+ TEST_MUSTPASS(RTCPport != 44445);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ voe_base_->DeleteChannel(0);
+ ch = voe_base_->CreateChannel();
+
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
+ // NULL as IP-address input should work as well
+ TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, NULL, sourcePort,
+ RTCPport));
+ MARK();
+ TEST_MUSTPASS(port != 44444);
+ TEST_MUSTPASS(sourcePort != 0);
+ TEST_MUSTPASS(RTCPport != 44445);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of GetLocalReceiver
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> StartReceive
+ // >> StopReceive
+ //
+ // State: VE initialized, no existing channels
+ TEST(StartReceive);
+ ANL();
+ TEST(StopReceive);
+ ANL();
+
+ // call without existing channel
+ TEST_MUSTPASS(!voe_base_->StartReceive(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(!voe_base_->StopReceive(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ch = voe_base_->CreateChannel();
+
+ // sockets must be created first
+ TEST_MUSTPASS(!voe_base_->StartReceive(0));
+ MARK();
+ TEST_ERROR(VE_SOCKETS_NOT_INITED);
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ MARK(); // should work this time
+
+ // enable again (should work)
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ MARK();
+
+ // Stop/Start (should work)
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ MARK();
+
+ // Verify in loopback
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ Play(ch, 1000, true, true);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ MARK();
+
+ voe_base_->DeleteChannel(0);
+ ch = voe_base_->CreateChannel();
+
+ // Ensure that it is OK to add delay between SetLocalReceiver and StarListen
+ TEST_LOG("\nspeak after 2 seconds and ensure that no delay is added:\n");
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
+
+ Sleep(2000, true); // adding emulated delay here
+
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ Play(ch, 2000, true, true);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+ ANL();
+
+ // Multi-channel tests
+
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ MARK();
+ }
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ TEST_MUSTPASS(voe_base_->StopReceive(i));
+ MARK();
+ voe_base_->DeleteChannel(i);
+ }
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+ MARK();
+ voe_base_->DeleteChannel(ch);
+ }
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of StartReceive/StopReceive
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> StartPlayout
+ // >> StopPlayout
+ //
+ // State: VE initialized, no existing channels
+ TEST(StartPlayout);
+ ANL();
+ TEST(StopPlayout);
+ ANL();
+
+ // call without existing channel
+ TEST_MUSTPASS(!voe_base_->StartPlayout(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(!voe_base_->StopPlayout(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ch = voe_base_->CreateChannel();
+
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ MARK();
+
+ voe_base_->DeleteChannel(ch);
+
+ // Multi-channel tests
+ const int MaxNumberOfPlayingChannels(kVoiceEngineMaxNumOfActiveChannels);
+
+ for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ MARK();
+ }
+ for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
+ TEST_MUSTPASS(voe_base_->StopPlayout(i));
+ MARK();
+ voe_base_->DeleteChannel(i);
+ }
+ for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ MARK();
+ voe_base_->DeleteChannel(ch);
+ }
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of StartPlayout/StopPlayout
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> StartSend
+ // >> StopSend
+ //
+ // State: VE initialized, no existing channels
+ TEST(StartSend);
+ ANL();
+ TEST(StopSend);
+ ANL();
+
+ // call without existing channel
+ TEST_MUSTPASS(!voe_base_->StartSend(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(!voe_base_->StopSend(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ ch = voe_base_->CreateChannel();
+
+ // call without initialized destination
+ TEST_MUSTPASS(!voe_base_->StartSend(ch));
+ MARK();
+ TEST_ERROR(VE_DESTINATION_NOT_INITED);
+
+ // initialize destination and try again (should work even without existing
+ // sockets)
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ MARK();
+ SLEEP(100);
+
+ // STATE: sockets should now have been created automatically at the first
+ // transmitted packet should be binded to 33333 and "0.0.0.0"
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ MARK();
+
+ voe_base_->DeleteChannel(ch);
+ ch = voe_base_->CreateChannel();
+
+ // try loopback with unique send sockets (closed when channel is deleted or
+ // new source is set)
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ Play(ch, 2000, true, true);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ voe_base_->DeleteChannel(ch);
+ ANL();
+
+ // Multi-channel tests
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333 + 2*i));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333 + 2*i, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ MARK();
+ }
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ TEST_MUSTPASS(voe_base_->StopSend(i));
+ MARK();
+ voe_base_->DeleteChannel(i);
+ }
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 45633 + 2*i));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 45633 + 2*i, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ MARK();
+ voe_base_->DeleteChannel(ch);
+ }
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of StartSend/StopSend
+ // ------------------------------------------------------------------------
+
+ //////////////////////////////
+ // SetNetEQPlayoutMode
+ // GetNetEQPlayoutMode
+ TEST(SetNetEQPlayoutMode);
+ ANL();
+ TEST(GetNetEQPlayoutMode);
+ ANL();
+
+ NetEqModes mode;
+
+ ch = voe_base_->CreateChannel();
+
+ // invalid function calls (should fail)
+ TEST_MUSTPASS(!voe_base_->GetNetEQPlayoutMode(ch+1, mode));
+ MARK();
+ TEST_MUSTPASS(!voe_base_->SetNetEQPlayoutMode(ch+1, kNetEqDefault));
+ MARK();
+
+ // verify default mode (should be kNetEqDefault)
+ TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(ch, mode));
+ MARK();
+ TEST_MUSTPASS(mode != kNetEqDefault);
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqStreaming));
+ MARK();
+ voe_base_->DeleteChannel(ch);
+
+ // ensure that default mode is set as soon as new channel is created
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(ch, mode));
+ MARK();
+ TEST_MUSTPASS(mode != kNetEqDefault);
+ voe_base_->DeleteChannel(ch);
+
+ // verify Set/Get for all supported modes and max number of channels
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+
+ // verify Set/Get for all supported modes
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqDefault));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
+ MARK();
+ TEST_MUSTPASS(mode != kNetEqDefault);
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqStreaming));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
+ MARK();
+ TEST_MUSTPASS(mode != kNetEqStreaming);
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqFax));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
+ MARK();
+ TEST_MUSTPASS(mode != kNetEqFax);
+ SLEEP(50);
+ }
+
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ voe_base_->DeleteChannel(i);
+ }
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ //////////////////////////////
+ // SetNetEQBGNMode
+ // GetNetEQBGNMode
+ TEST(SetNetEQBGNMode);
+ ANL();
+ TEST(GetNetEQBGNMode);
+ ANL();
+
+ NetEqBgnModes bgnMode;
+
+ ch = voe_base_->CreateChannel();
+
+ // invalid function calls (should fail)
+ TEST_MUSTPASS(!voe_base_->GetNetEQBGNMode(ch+1, bgnMode));
+ MARK();
+ TEST_MUSTPASS(!voe_base_->SetNetEQBGNMode(ch+1, kBgnOn));
+ MARK();
+
+ // verify default mode (should be kBgnOn)
+ TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(ch, bgnMode));
+ MARK();
+ TEST_MUSTPASS(bgnMode != kBgnOn);
+ voe_base_->DeleteChannel(ch);
+
+ // ensure that default mode is set as soon as new channel is created
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(ch, bgnMode));
+ MARK();
+ TEST_MUSTPASS(bgnMode != kBgnOn);
+ voe_base_->DeleteChannel(ch);
+
+ // verify Set/Get for all supported modes and max number of channels
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ ch = voe_base_->CreateChannel();
+
+ // verify Set/Get for all supported modes
+ TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnOn));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
+ MARK();
+ TEST_MUSTPASS(bgnMode != kBgnOn);
+ TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnFade));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
+ MARK();
+ TEST_MUSTPASS(bgnMode != kBgnFade);
+ TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnOff));
+ MARK();
+ TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
+ MARK();
+ TEST_MUSTPASS(bgnMode != kBgnOff);
+ SLEEP(50);
+ }
+
+ for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
+ voe_base_->DeleteChannel(i);
+ }
+
+ // Verify real-time performance for all playout modes in full duplex
+
+ ch = voe_base_->CreateChannel();
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqDefault));
+ MARK();
+ TEST_LOG("\nenjoy full duplex using kNetEqDefault playout mode...\n");
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqStreaming));
+ MARK();
+ TEST_LOG("\nenjoy full duplex using kNetEqStreaming playout mode...\n");
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqFax));
+ MARK();
+ TEST_LOG("\nenjoy full duplex using kNetEqFax playout mode...\n");
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ voe_base_->DeleteChannel(ch);
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ /////////////////////
+ // Full duplex tests
+
+ ch = voe_base_->CreateChannel(); // We must delete this channel first to be able
+ // to reuse port 12345
+
+ // start with default case, also test non-default RTCP port
+#ifdef _TEST_RTP_RTCP_
+ TEST_MUSTPASS(rtp->SetRTCP_CNAME(ch, "Johnny"));
+#endif
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 12349));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
+ 12349));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_LOG("full duplex is now activated (1)\n");
+ TEST_LOG("waiting for RTCP packet...\n");
+
+ SLEEP(7000); // Make sure we get RTCP packet
+ PAUSE;
+
+ // Verify that we got RTCP packet from correct source port
+#ifdef _TEST_RTP_RTCP_
+ char tmpStr[64] = { 0 };
+ TEST_MUSTPASS(rtp->GetRemoteRTCP_CNAME(ch, tmpStr));
+ TEST_MUSTPASS(_stricmp("Johnny", tmpStr));
+#endif
+ int rtpPort(0), rtcpPort(0);
+ char ipAddr[64] = { 0 };
+ TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
+ TEST_MUSTPASS(12349 != rtcpPort);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // Call StartSend before StartReceive
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_LOG("\nfull duplex is now activated (2)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // Try again using same ports
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_LOG("\nfull duplex is now activated (3)\n");
+ TEST_LOG("waiting for RTCP packet...\n");
+
+ SLEEP(7000); // Make sure we get RTCP packet
+ PAUSE
+
+ // Verify correct RTCP source port
+ TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
+ TEST_MUSTPASS(12345+1 != rtcpPort);
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ voe_base_->DeleteChannel(ch);
+ ch = voe_base_->CreateChannel();
+
+ // Try with extra send socket
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 22222));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 22222, "127.0.0.1", 11111));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_LOG("\nfull duplex is now activated (4)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // repeat default case starting with a fresh channel
+
+ voe_base_->DeleteChannel(ch);
+ ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+
+ TEST_LOG("\nfull duplex is now activated (5)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // restart call again
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+
+ TEST_LOG("\nfull duplex is now activated (6)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // force sending from new socket
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 12350,
+ 12359));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+ TEST_LOG("\nfull duplex is now activated (7)\n");
+
+ PAUSE
+
+ // Test getting send settings
+ TEST_MUSTPASS(voe_base_->GetSendDestination(ch, rtpPort, ipAddr, sourcePort,
+ rtcpPort));
+ TEST_MUSTPASS(12345 != rtpPort);
+ TEST_MUSTPASS(_stricmp("127.0.0.1", ipAddr));
+ TEST_MUSTPASS(12350 != sourcePort);
+ TEST_MUSTPASS(12359 != rtcpPort);
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ // new channel and new port
+ ch = voe_base_->CreateChannel();
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 33221));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, "127.0.0.1"));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+
+ TEST_LOG("\nfull duplex is now activated (8)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+
+ voe_base_->DeleteChannel(ch);
+ ch = voe_base_->CreateChannel();
+
+#ifndef MAC_IPHONE
+ // bind to local IP and try again
+ strcpy(localIp, "127.0.0.1");
+#else
+ localIp = "127.0.0.1";
+#endif
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33221, 12349, localIp));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, localIp));
+
+ TEST_MUSTPASS(voe_base_->StartReceive(ch));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StartSend(ch));
+
+ TEST_LOG("\nfull duplex is now activated (9)\n");
+
+ PAUSE
+
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, rtpPort, rtcpPort, ipAddr));
+ TEST_MUSTPASS(33221 != rtpPort);
+ TEST_MUSTPASS(_stricmp(localIp, ipAddr));
+ TEST_MUSTPASS(12349 != rtcpPort);
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ //////////////////////
+ // Trace filter tests
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST(SetTraceFilter); ANL();
+
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
+ "VoEBase_trace_filter.txt"))); MARK();
+ SLEEP(100);
+
+ // Test a few different filters, verify in trace file
+ // Each SetTraceFilter calls should be seen once, no more, no less
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceNone)); MARK();
+ SLEEP(300);
+ // API call and info should NOT be seen in log
+ TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true)); MARK();
+ // API call and error should NOT be seen in log
+ TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
+
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceApiCall |
+ kTraceCritical |
+ kTraceError |
+ kTraceWarning)); MARK();
+ SLEEP(300);
+ // API call should and info should NOT be seen in log
+ TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, false)); MARK();
+ // API call and error should be seen in log
+ TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
+
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceApiCall | kTraceInfo));
+ MARK();
+ SLEEP(300);
+ // API call and info should be seen in log
+ TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true)); MARK();
+ // API call should and error should NOT be seen in log
+ TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
+
+ // Back to default
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceAll)); MARK();
+ SLEEP(300);
+
+ AOK(); ANL();
+#endif
+
+ // ------------------------------------------------------------------------
+ // >> Multiple instance testing
+ //
+ // We should support 8 instances simultaneously
+ // and at least one should be able to have a call running
+
+ // One instance is already created
+ VoiceEngine* instVE[7];
+ VoEBase* baseVE[7];
+ for (int instNum = 0; instNum < 7; instNum++) {
+ instVE[instNum] = VoiceEngine::Create();
+ baseVE[instNum] = VoEBase::GetInterface(instVE[instNum]);
+ TEST_MUSTPASS(baseVE[instNum]->Init());
+ TEST_MUSTPASS(baseVE[instNum]->CreateChannel());
+ }
+
+ TEST_LOG("Created 7 more instances of VE, make sure audio is ok...\n\n");
+ PAUSE
+
+ for (int instNum = 0; instNum < 7; instNum++) {
+ TEST_MUSTPASS(baseVE[instNum]->DeleteChannel(0));
+ TEST_MUSTPASS(baseVE[instNum]->Terminate());
+ TEST_MUSTPASS(baseVE[instNum]->Release());
+ VoiceEngine::Delete(instVE[instNum]);
+ }
+
+ AOK();
+ ANL();
+
+ //////////////
+ // Close down
+ TEST_MUSTPASS(voe_base_->StopSend(ch));
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ TEST_MUSTPASS(voe_base_->StopReceive(ch));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
+
+ voe_base_->DeleteChannel(0);
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestCallReport
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestCallReport() {
+ // Get required sub-API pointers
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoECallReport* report = _mgr.CallReportPtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoEAudioProcessing* apm = _mgr.APMPtr();
+ VoENetwork* netw = _mgr.NetworkPtr();
+
+ PrepareTest("CallReport");
+
+ // check if this interface is supported
+ if (!report) {
+ TEST_LOG("VoECallReport is not supported!");
+ return -1;
+ }
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoECallReport_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+
+ ///////////////////////////
+ // Actual test starts here
+ TEST(ResetCallReportStatistics);
+ ANL();
+ TEST_MUSTPASS(!report->ResetCallReportStatistics(-2));
+ MARK(); // not OK
+ TEST_MUSTPASS(!report->ResetCallReportStatistics(1));
+ MARK(); // not OK
+ TEST_MUSTPASS(report->ResetCallReportStatistics(0));
+ MARK(); // OK
+ TEST_MUSTPASS(report->ResetCallReportStatistics(-1));
+ MARK(); // OK
+ AOK();
+ ANL();
+
+ bool enabled = false;
+ EchoStatistics echo;
+ TEST(GetEchoMetricSummary);
+ ANL();
+ TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
+ TEST_MUSTPASS(report->GetEchoMetricSummary(echo)); // all outputs will be
+ // -100 in loopback (skip further tests)
+ AOK();
+ ANL();
+
+ // TODO(xians): investigate the cause of test failure before enabling.
+ /*
+ StatVal delays;
+ TEST(GetRoundTripTimeSummary);
+ ANL();
+ // All values should be >=0 since RTCP is now on
+ TEST_MUSTPASS(report->GetRoundTripTimeSummary(0, delays));
+ MARK();
+ TEST_MUSTPASS(delays.min == -1);
+ TEST_MUSTPASS(delays.max == -1);
+ TEST_MUSTPASS(delays.average == -1);
+ rtp_rtcp->SetRTCPStatus(0, false);
+ // All values should be -1 since RTCP is off
+ TEST_MUSTPASS(report->GetRoundTripTimeSummary(0, delays));
+ MARK();
+ TEST_MUSTPASS(delays.min != -1);
+ TEST_MUSTPASS(delays.max != -1);
+ TEST_MUSTPASS(delays.average != -1);
+ rtp_rtcp->SetRTCPStatus(0, true);
+ AOK();
+ ANL();
+ */
+
+ int nDead = 0;
+ int nAlive = 0;
+ TEST(GetDeadOrAliveSummary);
+ ANL();
+ // All results should be -1 since dead-or-alive is not active
+ TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive) != -1);
+ MARK();
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ SLEEP(2000);
+ // All results should be >= 0 since dead-or-alive is active
+ TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive));
+ MARK();
+ TEST_MUSTPASS(nDead == -1);
+ TEST_MUSTPASS(nAlive == -1)
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ AOK();
+ ANL();
+
+ TEST(WriteReportToFile);
+ ANL();
+
+ // Greek and Coptic (see http://www.utf8-chartable.de/unicode-utf8-table.pl)
+ char fileNameUTF8[64];
+
+ fileNameUTF8[0] = (char) 0xce;
+ fileNameUTF8[1] = (char) 0xba;
+ fileNameUTF8[2] = (char) 0xce;
+ fileNameUTF8[3] = (char) 0xbb;
+ fileNameUTF8[4] = (char) 0xce;
+ fileNameUTF8[5] = (char) 0xbd;
+ fileNameUTF8[6] = (char) 0xce;
+ fileNameUTF8[7] = (char) 0xbe;
+ fileNameUTF8[8] = '.';
+ fileNameUTF8[9] = 't';
+ fileNameUTF8[10] = 'x';
+ fileNameUTF8[11] = 't';
+ fileNameUTF8[12] = 0;
+
+ TEST_MUSTPASS(!report->WriteReportToFile(NULL));
+ MARK();
+ TEST_MUSTPASS(report->WriteReportToFile("call_report.txt"));
+ MARK();
+ TEST_MUSTPASS(report->WriteReportToFile(fileNameUTF8));
+ MARK(); // should work with UTF-8 as well (κλνξ.txt)
+ AOK();
+ ANL();
+
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestCodec
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestCodec() {
+ PrepareTest("Codec");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoECodec* codec = _mgr.CodecPtr();
+ VoEFile* file = _mgr.FilePtr();
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoECodec_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ ExtendedTestTransport* ptrTransport(NULL);
+ ptrTransport = new ExtendedTestTransport(netw);
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+#else
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+#endif
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ int i;
+ int err;
+
+ CodecInst cinst;
+
+ /////////////////////////
+ // GetNumOfCodecs
+
+ int nCodecs;
+
+ TEST(GetNumOfCodecs);
+ ANL();
+ // validate #codecs
+ nCodecs = codec->NumOfCodecs();
+ MARK();
+ TEST_MUSTPASS(nCodecs < 0);
+ AOK();
+ ANL();
+
+ ///////////////////
+ // GetCodec
+ TEST(GetCodec);
+ ANL();
+ // scan all supported codecs
+ nCodecs = codec->NumOfCodecs();
+ for (int index = 0; index < nCodecs; index++) {
+ TEST_MUSTPASS(codec->GetCodec(index, cinst));
+ TEST_LOG("[%2d] %16s: fs=%6d, pt=%4d, rate=%7d, ch=%2d, size=%5d", index, cinst.plname,
+ cinst.plfreq, cinst.pltype, cinst.rate, cinst.channels, cinst.pacsize);
+ if (cinst.pltype == -1) {
+ TEST_LOG(" <= NOTE pt=-1\n");
+ } else {
+ ANL();
+ }
+ }
+
+ // ensure that an invalid index parameter is detected
+ TEST_MUSTPASS(-1 != codec->GetCodec(-1, cinst));
+ nCodecs = codec->NumOfCodecs();
+ TEST_MUSTPASS(-1 != codec->GetCodec(nCodecs, cinst));
+ MARK();
+ // ensure that error code is VE_INVALID_LISTNR
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_LISTNR);
+ AOK();
+ ANL();
+
+ ///////////////////////
+ // GetSendCodec
+ TEST(GetSendCodec);
+ ANL();
+
+ CodecInst defaultCodec;
+
+ // check the channel parameter
+ int nMaxChannels(voe_base_->MaxNumOfChannels());
+ TEST_MUSTPASS(-1 != codec->GetSendCodec(nMaxChannels-1, cinst));
+ MARK(); // not created
+ TEST_MUSTPASS(-1 != codec->GetSendCodec(nMaxChannels, cinst));
+ MARK(); // out of range
+ TEST_MUSTPASS(-1 != codec->GetSendCodec(-1, cinst));
+ MARK(); // out of range
+ TEST_MUSTPASS(codec->GetSendCodec(0, cinst));
+ MARK(); // OK
+
+ nCodecs = codec->NumOfCodecs();
+ for (int index = 0; index < nCodecs; index++) {
+ TEST_MUSTPASS(codec->GetCodec(index, defaultCodec));
+ if (codec->SetSendCodec(0, defaultCodec) == 0) {
+ TEST_MUSTPASS(codec->GetSendCodec(0, cinst));
+ MARK();
+ //TEST_LOG("[%2d] %s: fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
+ // index, cinst.plname, cinst.plfreq, cinst.pltype, cinst.rate,
+ // cinst.channels, cinst.pacsize);
+ TEST_MUSTPASS(cinst.pacsize != defaultCodec.pacsize);
+ TEST_MUSTPASS(cinst.plfreq != defaultCodec.plfreq);
+ TEST_MUSTPASS(cinst.pltype != defaultCodec.pltype);
+ TEST_MUSTPASS(cinst.rate != defaultCodec.rate);
+ TEST_MUSTPASS(cinst.channels != defaultCodec.channels);
+ }
+ }
+
+ ANL();
+ AOK();
+ ANL();
+
+ ///////////////////////
+ // SetSendCodec
+ TEST(SetSendCodec);
+ ANL();
+
+ // --- Scan all supported codecs and set default parameters
+
+ nCodecs = codec->NumOfCodecs();
+ for (int index = 0; index < nCodecs; index++) {
+ // Get default (ACM) settings
+ TEST_MUSTPASS(codec->GetCodec(index, cinst));
+ defaultCodec = cinst;
+ TEST_LOG("[%2d] %s (default): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
+ index, cinst.plname, cinst.plfreq, cinst.pltype, cinst.rate,
+ cinst.channels, cinst.pacsize);
+
+ // Verify invalid codec names
+ if (!_stricmp("CN", cinst.plname) || !_stricmp("telephone-event",
+ cinst.plname)
+ || !_stricmp("red", cinst.plname)) {
+ // default settings for invalid payload names (should give
+ // VE_INVALID_PLNAME)
+ TEST_MUSTPASS(!codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ continue;
+ }
+
+ // If we build the ACM with more codecs than we have payload types,
+ // some codecs will be given -1 as default payload type. This is a fix
+ // to ensure that we can complete these tests also for this case.
+ if (cinst.pltype == -1) {
+ cinst.pltype = 97;
+ }
+
+ // --- Default settings
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ // --- Packet size
+ TEST_LOG("\npacsize : ");
+
+ for (int pacsize = 80; pacsize < 1440; pacsize += 80) {
+ cinst.pacsize = pacsize;
+ if (-1 != codec->SetSendCodec(0, cinst)) {
+ // log valid packet size
+ TEST_LOG("%d ", pacsize);
+ } else {
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ }
+ cinst.pacsize = defaultCodec.pacsize;
+
+ // --- Audio channels (1/mono or 2/stereo)
+ TEST_LOG("\nchannels: ");
+ for (int channels = 1; channels < 4; channels++) {
+ cinst.channels = channels;
+ if (-1 != codec->SetSendCodec(0, cinst)) {
+ // valid channels (only 1 should be OK)
+ TEST_LOG("%d ", channels);
+ } else {
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ }
+ cinst.channels = defaultCodec.channels;
+
+ // --- Payload frequency
+ TEST_LOG("\nplfreq : ");
+ cinst.plfreq = defaultCodec.plfreq;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.plfreq);
+
+ // --- Payload name
+
+ strcpy(cinst.plname, "INVALID");
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ // ensure that error code is VE_INVALID_PLNAME
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+
+ // restore default plname
+ strcpy(cinst.plname, defaultCodec.plname);
+
+ // --- Payload type (dynamic range is 96-127)
+ TEST_LOG("\npltype : ");
+ // All PT should be OK, test a few different
+ cinst.pltype = defaultCodec.pltype;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.pltype);
+ cinst.pltype = defaultCodec.pltype + 1;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.pltype);
+ const int valid_pltypes[4] = { 0, 96, 117, 127 };
+ for (i = 0; i < static_cast<int> (sizeof(valid_pltypes) / sizeof(int)); i++) {
+ cinst.pltype = valid_pltypes[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.pltype);
+ }
+ // Restore default
+ cinst.pltype = defaultCodec.pltype;
+
+ // --- Codec rate
+ TEST_LOG("\nrate : ");
+ if (_stricmp("isac", cinst.plname) == 0) {
+ // ISAC
+ if (cinst.plfreq == 16000) {
+ int valid_rates[3] = { -1, 10000, 32000 };
+ // failed in RegisterPayload when rate is 32000
+ for (i = 0; i < static_cast<int> (sizeof(valid_rates) / sizeof(int)); i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ cinst.rate = 0; // invalid
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ // ensure that error code is VE_CANNOT_SET_SEND_CODEC
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ ANL();
+ } else //ISACSWB
+ {
+ // rate changing fails in RegisterPayload
+ int valid_rates[8] = { -1, 10000, 25000, 32000, 35000, 45000, 50000, 52000 };
+ for (i = 0; i < static_cast<int> (sizeof(valid_rates) / sizeof(int)); i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ int invalid_rates[3] = { 0, 5000, 57000 }; // invalid
+ for (i = 0; i < static_cast<int> (sizeof(invalid_rates) / sizeof(int)); i++) {
+ cinst.rate = invalid_rates[i];
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ // ensure that error code is VE_CANNOT_SET_SEND_CODEC
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ }
+ ANL();
+ }
+ } else if (_stricmp("amr", cinst.plname) == 0) {
+ int valid_rates[8] = { 4750, 5150, 5900, 6700, 7400, 7950, 10200, 12200 };
+ for (i = 0;
+ i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
+ i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ ANL();
+ } else if (_stricmp("g7291", cinst.plname) == 0) {
+ int valid_rates[12] = { 8000, 12000, 14000, 16000, 18000, 20000, 22000,
+ 24000, 26000, 28000, 30000, 32000 };
+ for (i = 0;
+ i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
+ i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ ANL();
+ } else if (_stricmp("amr-wb", cinst.plname) == 0) {
+ int valid_rates[9] = { 7000, 9000, 12000, 14000, 16000, 18000, 20000,
+ 23000, 24000 };
+ for (i = 0;
+ i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
+ i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ TEST_LOG(" <=> ");
+ ANL();
+ } else if (_stricmp("speex", cinst.plname) == 0) {
+ // Valid speex rates are > 2000, testing some of them here
+ int valid_rates[9] = { 2001, 4000, 7000, 11000, 15000, 20000, 25000,
+ 33000, 46000 };
+ for (i = 0;
+ i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
+ i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ cinst.rate = 2000; // invalid
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ ANL();
+ } else if (_stricmp("silk", cinst.plname) == 0) {
+ // Valid Silk rates are 6000 - 40000, listing some of them here
+ int valid_rates[7] = { 6000, 10000, 15000, 20000, 25000, 32000, 40000 };
+ for (i = 0;
+ i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
+ i++) {
+ cinst.rate = valid_rates[i];
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ }
+ cinst.rate = 5999; // invalid
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ cinst.rate = 40001; // invalid
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
+ {
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ ANL();
+ } else {
+ // Use default rate for all other codecs.
+ cinst.rate = defaultCodec.rate;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_LOG("%d ", cinst.rate);
+ cinst.rate = defaultCodec.rate + 17;
+ TEST_MUSTPASS(!codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ ANL();
+ }
+ cinst.rate = defaultCodec.rate;
+
+ // run some extra tests for L16
+ if (_stricmp("l16", cinst.plname) == 0) {
+ if (8000 == cinst.plfreq) {
+ // valid pacsizes: 80, 160, 240, 320
+ cinst.pacsize = 480; // only supported in combination with 16kHz
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ cinst.pacsize = 640; // only supported in combination with 16kHz
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ } else {
+ // valid pacsizes: 160, 320, 480, 640
+ cinst.pacsize = 80; // only supported in combination with 8kHz
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ cinst.pacsize = 240; // only supported in combination with 8kHz
+ TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ }
+ }
+ ANL();
+ } // for (int index = 0; index < nCodecs; index++)
+
+ // restore PCMU
+ const CodecInst tmp = { 0, "PCMU", 8000, 160, 1, 64000 };
+ TEST_MUSTPASS(codec->SetSendCodec(0, tmp));
+
+ ANL();
+ AOK();
+ ANL();
+
+ ///////
+ // VAD
+
+ const int VADSleep = 0;
+
+ bool disabledDTX;
+ VadModes mode;
+ bool enabled;
+
+ // verify default settings (should be OFF, kVadConventional and DTX enabled)
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(mode != kVadConventional);
+ TEST_MUSTPASS(disabledDTX != true);
+
+ // enable default VAD settings
+ TEST_MUSTPASS(codec->SetVADStatus(0, true));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(mode != kVadConventional);
+ TEST_MUSTPASS(disabledDTX != false);
+ SLEEP(VADSleep);
+
+ // set kVadConventional mode
+ TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadConventional));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(mode != kVadConventional);
+ SLEEP(VADSleep);
+
+ // set kVadAggressiveLow mode
+ TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveLow));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(mode != kVadAggressiveLow);
+ SLEEP(VADSleep);
+
+ // set kVadAggressiveMid mode
+ TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveMid));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(mode != kVadAggressiveMid);
+ SLEEP(VADSleep);
+
+ // set kVadAggressiveMid mode
+ TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveHigh));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(mode != kVadAggressiveHigh);
+ SLEEP(VADSleep);
+
+ // turn DTX OFF (audio should not be affected by VAD decisions)
+ TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadConventional, true));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(disabledDTX != true);
+ SLEEP(VADSleep);
+
+ // try to enable DTX again (should fail since VAD is disabled)
+ TEST_MUSTPASS(codec->SetVADStatus(0, false, kVadConventional, false));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(disabledDTX == false);
+ SLEEP(VADSleep);
+
+ // disable VAD
+ TEST_MUSTPASS(codec->SetVADStatus(0, false));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(enabled != false);
+ SLEEP(VADSleep);
+
+ // restore default VAD
+ TEST_MUSTPASS(codec->SetVADStatus(0, true));
+ TEST_MUSTPASS(codec->SetVADStatus(0, false));
+ TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
+ TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
+ disabledDTX);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(mode != kVadConventional);
+ TEST_MUSTPASS(disabledDTX != true);
+ SLEEP(VADSleep);
+
+ AOK();
+ ANL();
+ ANL();
+
+ //////////////////////
+ // GetRecCodec
+ TEST(GetRecCodec);
+ ANL();
+
+ // stop all streaming first
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ // start loopback streaming (PCMU is default)
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0,8000,"127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
+#endif
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(100); // ensure that at least one packets is received
+
+ // scan all supported and valid codecs
+ CodecInst newCodec;
+ for (i = 0; i < codec->NumOfCodecs(); i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, newCodec));
+ // test all valid send codecs
+ if (!_stricmp("red", newCodec.plname) || !_stricmp("cn", newCodec.plname)
+ || !_stricmp("telephone-event", newCodec.plname)) {
+ continue; // Ignore these
+ }
+ if (-1 != codec->SetSendCodec(0, newCodec)) {
+ SLEEP(150);
+ // verify correct detection
+ TEST_MUSTPASS(codec->GetRecCodec(0, cinst));
+ TEST_LOG("%s %s ", newCodec.plname, cinst.plname);
+ TEST_MUSTPASS(_stricmp(newCodec.plname, cinst.plname) != 0);
+ TEST_MUSTPASS(cinst.pltype != newCodec.pltype);
+ TEST_MUSTPASS(cinst.plfreq != newCodec.plfreq);
+ }
+ }
+
+ // stop streaming
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+#ifdef WEBRTC_CODEC_GSMAMR
+ //////////////////////////
+ // SetAMREncFormat
+
+ // Fresh channel
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ TEST(SetAMREncFormat); ANL();
+
+ //set another codec which is not AMR
+ TEST_MUSTPASS(codec->GetCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ //try to change the encode format, tests should fail
+ TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267BwEfficient));
+ MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267OctetAligned));
+ MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267FileStorage));
+ MARK();
+
+ //set AMR as encoder
+ strcpy(cinst.plname,"AMR");
+ cinst.channels=1; cinst.plfreq=8000; cinst.rate=12200; cinst.pltype=112;
+ cinst.pacsize=160;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ //try to change the encode format, tests should pass
+ TEST_MUSTPASS(codec->SetAMREncFormat(0)); MARK();
+ TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267BwEfficient)); MARK();
+ TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267OctetAligned)); MARK();
+ TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267FileStorage)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMREncFormat(-1)); MARK();
+ TEST_MUSTPASS(codec->SetAMREncFormat(0)); MARK(); // restore default
+
+ ANL();
+ AOK();
+ ANL();
+
+ //////////////////////////
+ // SetAMRDecFormat
+
+ TEST(SetAMRDecFormat); ANL();
+
+ // It should not be possible to set AMR dec format before valid AMR decoder
+ // is registered
+ TEST_MUSTPASS(!codec->SetAMRDecFormat(0)); MARK();
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
+
+ // Ensure that ACM::RegisterReceiveCodec(AMR) is called
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+
+ // All these tests should now pass
+ TEST_MUSTPASS(codec->SetAMRDecFormat(0)); MARK();
+ TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267BwEfficient)); MARK();
+ TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267OctetAligned)); MARK();
+ TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267FileStorage)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRDecFormat(-1)); MARK();
+ TEST_MUSTPASS(codec->SetAMRDecFormat(0)); MARK(); // restore default
+
+ ANL();
+ AOK();
+ ANL();
+#endif // #ifdef WEBRTC_CODEC_GSMAMR
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ //////////////////////////
+ // SetAMRWbEncFormat
+
+ // Fresh channel
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ TEST(SetAMRWbEncFormat); ANL();
+
+ //set another codec which is not AMR-wb
+ TEST_MUSTPASS(codec->GetCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ //try to change the encode format, tests should fail
+ TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267BwEfficient));
+ MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267OctetAligned));
+ MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267FileStorage));
+ MARK();
+
+ //set AMR-wb as encoder
+ strcpy(cinst.plname,"AMR-WB");
+ cinst.channels=1; cinst.plfreq=16000; cinst.rate=20000;
+ cinst.pltype=112; cinst.pacsize=320;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ //try to change the encode format, tests should pass
+ TEST_MUSTPASS(codec->SetAMRWbEncFormat(0)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267BwEfficient)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267OctetAligned)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267FileStorage)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(-1)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbEncFormat(0)); MARK(); // restore default
+
+ ANL();
+ AOK();
+ ANL();
+
+ //////////////////////////
+ // SetAMRDecFormat
+
+ TEST(SetAMRWbDecFormat); ANL();
+
+ // It should not be possible to set AMR dec format before valid AMR decoder
+ // is registered
+ TEST_MUSTPASS(!codec->SetAMRWbDecFormat(0)); MARK();
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
+
+ // Ensure that ACM::RegisterReceiveCodec(AMR) is called
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+
+ // All these tests should now pass
+ TEST_MUSTPASS(codec->SetAMRWbDecFormat(0)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267BwEfficient)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267OctetAligned)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267FileStorage)); MARK();
+ TEST_MUSTPASS(-1 != codec->SetAMRWbDecFormat(-1)); MARK();
+ TEST_MUSTPASS(codec->SetAMRWbDecFormat(0)); MARK(); // restore default
+
+ ANL();
+ AOK();
+ ANL();
+#endif // #ifdef WEBRTC_CODEC_GSMAMRWB
+ ///////////////////////////////
+ // SetSendCNPayloadType
+ TEST(SetSendCNPayloadType);
+ ANL();
+
+ TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(-1, 0));
+ MARK(); // invalid channel
+
+ // Invalid payload range (only dynamic range [96,127]
+ TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 0));
+ MARK(); // invalid PT
+ TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 95));
+ MARK(); // invalid PT
+ TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 128));
+ MARK(); // invalid PT
+ TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, -1));
+ MARK(); // invalid PT
+
+ // Not possible to change PT for 8000
+ TEST_MUSTPASS(!codec->SetSendCNPayloadType(0, 96, kFreq8000Hz));
+ MARK();
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_PLFREQ);
+
+ // Try some dynamic for 16000 and 32000 as well
+ TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 96, kFreq16000Hz));
+ MARK();
+ TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 96, kFreq32000Hz));
+ MARK(); // same should work
+ TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 127, kFreq16000Hz));
+ MARK();
+ TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 127, kFreq32000Hz));
+ MARK();
+ TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 100, kFreq32000Hz));
+ MARK();
+
+ ANL();
+ AOK();
+ ANL();
+
+ /////////////////////////////
+ // SetRecPayloadType
+ TEST(SetRecPayloadType);
+ ANL();
+
+ // scan all supported and valid codecs without changing payloads
+ nCodecs = codec->NumOfCodecs();
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, newCodec));
+ // If no default payload type is defined, we use 127
+ if (-1 == newCodec.pltype) {
+ newCodec.pltype = 127;
+ }
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
+ MARK(); // use default
+ newCodec.pltype = 99;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
+ MARK(); // use same PT on all
+ newCodec.pltype = -1;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
+ MARK(); // deregister all PTs
+ }
+
+ ANL();
+ AOK();
+ ANL();
+
+ /////////////////////////////
+ // GetRecPayloadType
+ TEST(GetRecPayloadType);
+ ANL();
+
+ CodecInst extraCodec;
+ for (i = 0; i < nCodecs; i++) {
+ // Set defaults
+ TEST_MUSTPASS(codec->GetCodec(i, newCodec));
+ // If no default payload type is defined, we use 127
+ if (-1 == newCodec.pltype) {
+ newCodec.pltype = 127;
+ }
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
+ //TEST_LOG("[%2d] %s (SetRec): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
+ // i, newCodec.plname, newCodec.plfreq, newCodec.pltype, newCodec.rate,
+ // newCodec.channels, newCodec.pacsize);
+ extraCodec.pltype = -1; // don't know this yet
+ extraCodec.plfreq = newCodec.plfreq;
+ extraCodec.rate = newCodec.rate;
+ extraCodec.channels = newCodec.channels;
+ strcpy(extraCodec.plname, newCodec.plname);
+ // Verfify that setting is OK
+ TEST_MUSTPASS(codec->GetRecPayloadType(0, extraCodec));
+ //TEST_LOG("[%2d] %s (GetRec): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
+ // i, extraCodec.plname, extraCodec.plfreq, extraCodec.pltype,
+ // extraCodec.rate, extraCodec.channels, extraCodec.pacsize);
+ TEST_MUSTPASS(newCodec.pltype != extraCodec.pltype);
+ TEST_MUSTPASS(newCodec.plfreq != extraCodec.plfreq);
+ TEST_MUSTPASS(newCodec.channels != extraCodec.channels);
+ }
+
+ AOK();
+ ANL();
+
+ ////////////////////////////////////////////////////
+ // SetRecPayloadType - remove receive codecs
+ TEST(SetRecPayloadType - removing receive codecs);
+ ANL();
+
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+#endif
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ if (file) {
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0,
+ _mgr.AudioFilename(),
+ true,
+ true));
+ }
+
+ // Scan all supported and valid codecs and remove from receiving db, then
+ // restore
+ nCodecs = codec->NumOfCodecs();
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, cinst));
+ if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
+ || !_stricmp("telephone-event", cinst.plname)) {
+ continue; // Ignore these
+ }
+ TEST_LOG("Testing codec: %s", cinst.plname);
+ fflush(NULL);
+
+ if (-1 == cinst.pltype) {
+ // If no default payload type is defined, we use 127,
+ // codec is not registered for receiving
+ cinst.pltype = 127;
+ } else {
+ // Remove codec
+ memcpy(&extraCodec, &cinst, sizeof(CodecInst));
+ extraCodec.pltype = -1;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, extraCodec));
+ }
+
+ // Set send codec
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ // Verify no audio
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_LOG(" silence");
+ fflush(NULL);
+ SLEEP(800);
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ // Restore codec
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+
+ // Verify audio
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_LOG(" audio");
+ fflush(NULL);
+ SLEEP(800);
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ if (127 == cinst.pltype) {
+ // If no default payload type is defined, i.e. we have set pt to
+ //127 above,
+ // make sure we remove codec from receiving
+ cinst.pltype = -1;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ }
+
+ ANL();
+ }
+
+ // Remove certain codecs
+ TEST_LOG("Removing receive codecs:");
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, cinst));
+ if (!_stricmp("ipcmwb", cinst.plname) || !_stricmp("pcmu", cinst.plname)
+ || !_stricmp("eg711a", cinst.plname)) {
+ TEST_LOG(" %s", cinst.plname);
+ memcpy(&extraCodec, &cinst, sizeof(CodecInst));
+ extraCodec.pltype = -1;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, extraCodec));
+ }
+ }
+ ANL();
+
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ // Test sending all codecs - verify audio/no audio depending on codec
+ TEST_LOG("Looping through send codecs \n");
+ TEST_LOG("Verify that removed codecs are not audible and the other are \n");
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, cinst));
+ if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
+ || !_stricmp("telephone-event", cinst.plname)) {
+ continue; // Ignore these
+ }
+ TEST_LOG("Testing codec: %s \n", cinst.plname);
+
+ // If no default payload type is defined, we use 127 and set receive
+ // payload type
+ if (-1 == cinst.pltype) {
+ cinst.pltype = 127;
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ }
+
+ // Set send codec
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ // Verify audio/no audio
+ SLEEP(800);
+ }
+
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ // Restore codecs
+ TEST_LOG("Restoring receive codecs:");
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, cinst));
+ if (!_stricmp("ipcmwb", cinst.plname) || !_stricmp("pcmu", cinst.plname)
+ || !_stricmp("eg711a", cinst.plname)) {
+ TEST_LOG(" %s", cinst.plname);
+ memcpy(&extraCodec, &cinst, sizeof(CodecInst));
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ }
+ }
+ ANL();
+
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ // Test sending all codecs - verify audio
+ TEST_LOG("Looping through send codecs \n");
+ TEST_LOG("Verify that all codecs are audible \n");
+ for (i = 0; i < nCodecs; i++) {
+ TEST_MUSTPASS(codec->GetCodec(i, cinst));
+ if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
+ || !_stricmp("telephone-event", cinst.plname)) {
+ continue; // Ignore these
+ }
+ TEST_LOG("Testing codec: %s \n", cinst.plname);
+
+ // If no default payload type is defined, we use 127 and set receive
+ // payload type
+ if (-1 == cinst.pltype) {
+ cinst.pltype = 127;
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ }
+
+ // Set send codec
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ // Verify audio/no audio
+ SLEEP(800);
+ }
+
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ // Fresh channel
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+#if defined(WEBRTC_CODEC_ISAC)
+
+ /////////////////////////////////////
+ // SetISACInitTargetRate - wb
+ TEST(SetISACInitTargetRate);
+ ANL();
+
+ // set PCMU as sending codec
+ cinst.channels = 1;
+ cinst.pacsize = 160;
+ cinst.plfreq = 8000;
+ strcpy(cinst.plname, "PCMU");
+ cinst.pltype = 0;
+ cinst.rate = 64000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 10000));
+ MARK(); // should fail since iSAC is not active
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CODEC_ERROR);
+
+ // set iSAC as sending codec (16kHz)
+ cinst.channels = 1;
+ cinst.plfreq = 16000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 103;
+ cinst.rate = -1; // adaptive rate
+ cinst.pacsize = 480; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(1, 10000));
+ MARK(); // invalid channel
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 500));
+ MARK(); // invalid target rates (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 33000));
+ MARK(); // invalid target rates (too large)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 10000));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 0));
+ MARK(); // 0 is a valid rate
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000));
+ MARK(); // try max as well
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, true));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, false));
+ MARK();
+
+ cinst.pacsize = 960; // 60ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, false));
+ MARK();
+
+ cinst.rate = 20000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 32000));
+ MARK(); // only works in adaptive mode
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
+
+ cinst.rate = -1;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000));
+ MARK(); // back to adaptive mode
+
+ ANL();
+ AOK();
+ ANL();
+
+ /////////////////////////////////////
+ // SetISACInitTargetRate - swb
+ TEST(ISACSWB SetISACInitTargetRate);
+ ANL();
+
+ // set iSAC as sending codec
+ cinst.channels = 1;
+ cinst.plfreq = 32000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 104;
+ cinst.rate = -1; // default rate
+ cinst.pacsize = 960; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(1, 10000));
+ MARK(); // invalid channel
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, -1));
+ MARK(); // invalid target rates (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, -1));
+ MARK(); // invalid target rates (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 500));
+ MARK(); // invalid target rates (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 57000));
+ MARK(); // invalid target rates (valid range is [10000, 56000])
+
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 10000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 0));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000));
+ MARK(); // try max as well
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000, true));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000, false));
+ MARK();
+
+ ANL();
+ AOK();
+ ANL();
+
+ ////////////////////////////////
+ // SetISACMaxRate
+ TEST(SetISACMaxRate);
+ ANL();
+
+ // set PCMU as sending codec
+ cinst.channels = 1;
+ cinst.pacsize = 160;
+ cinst.plfreq = 8000;
+ strcpy(cinst.plname, "PCMU");
+ cinst.pltype = 0;
+ cinst.rate = 64000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 48000));
+ MARK(); // should fail since iSAC is not active
+ TEST_MUSTPASS(voe_base_->LastError() != VE_CODEC_ERROR);
+
+ // set iSAC as sending codec
+ cinst.channels = 1;
+ cinst.plfreq = 16000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 103;
+ cinst.rate = -1; // adaptive rate
+ cinst.pacsize = 480; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
+ MARK(); // invalid channel
+ TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
+ MARK(); // invalid target rates (too small)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 53500));
+ MARK(); // invalid target rates (too large)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 48000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 53400));
+ MARK(); // try max as well (default)
+
+ cinst.pacsize = 960; // 60ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 48000));
+ MARK();
+
+ cinst.rate = 20000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
+ MARK(); // also works in non-adaptive mode
+
+ ANL();
+ AOK();
+ ANL();
+
+ TEST(ISACSWB SetISACMaxRate);
+ ANL();
+ // set iSAC as sending codec
+ cinst.channels = 1;
+ cinst.plfreq = 32000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 104;
+ cinst.rate = 45000; // instantaneous mode
+ cinst.pacsize = 960; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
+ MARK(); // invalid channel
+ TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
+ MARK(); // invalid target rates (too small)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 107500));
+ MARK(); // invalid target rates (too large)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 55000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 80000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 107000));
+ MARK(); // try max as well (default)
+
+
+ cinst.rate = -1; // adaptive mode
+ cinst.pacsize = 960; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
+ MARK(); // invalid channel
+ TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
+ MARK(); // invalid target rates (too small)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACMaxRate(0, 107500));
+ MARK(); // invalid target rates (too large)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 55000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 80000));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxRate(0, 107000));
+ MARK(); // try max as well (default)
+
+ ANL();
+ AOK();
+ ANL();
+
+ ////////////////////////////////
+ // SetISACMaxPayloadSize
+ TEST(SetISACMaxPayloadSize);
+ ANL();
+
+ // set PCMU as sending codec
+ cinst.channels = 1;
+ cinst.pacsize = 160;
+ cinst.plfreq = 8000;
+ strcpy(cinst.plname, "PCMU");
+ cinst.pltype = 0;
+ cinst.rate = 64000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 120));
+ MARK(); // should fail since iSAC is not active
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CODEC_ERROR);
+
+ // set iSAC as sending codec
+ cinst.channels = 1;
+ cinst.plfreq = 16000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 103;
+ cinst.rate = -1; // adaptive rate
+ cinst.pacsize = 480; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(1, 120));
+ MARK(); // invalid channel
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 100));
+ MARK(); // invalid size (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 410));
+ MARK(); // invalid size (too large)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 200));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 120));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 400));
+ MARK();
+
+ ANL();
+ AOK();
+ ANL();
+
+ TEST(ISACSWB SetISACMaxPayloadSize);
+ ANL();
+ // set iSAC as sending codec
+ cinst.channels = 1;
+ cinst.plfreq = 32000;
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 104;
+ cinst.rate = 45000; // default rate
+ cinst.pacsize = 960; // 30ms
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(1, 100));
+ MARK(); // invalid channel
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 100));
+ MARK(); // invalid size (too small)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 610));
+ MARK(); // invalid size (too large)
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
+
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 200));
+ MARK(); // life is good now
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 120));
+ MARK();
+ TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 600));
+ MARK();
+
+ ANL();
+ AOK();
+ ANL();
+
+ // set iSAC as sending codec
+ // set iSAC-wb as sending codec
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+#else
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8001, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8001));
+#endif
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ 0, GetFilename("audio_long16.pcm"), true , true));
+ cinst.channels = 1;
+ TEST_LOG("Testing codec: Switch between iSAC-wb and iSAC-swb \n");
+ TEST_LOG("Testing codec: iSAC wideband \n");
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 103;
+ cinst.rate = -1; // default rate
+ cinst.pacsize = 480; // 30ms
+ cinst.plfreq = 16000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ SLEEP(2000);
+ TEST_LOG(" : iSAC superwideband \n");
+ cinst.pltype = 104;
+ cinst.rate = -1; // default rate
+ cinst.pacsize = 960; // 30ms
+ cinst.plfreq = 32000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ SLEEP(2000);
+ TEST_LOG(" : iSAC wideband \n");
+ strcpy(cinst.plname, "ISAC");
+ cinst.pltype = 103;
+ cinst.rate = -1; // default rate
+ cinst.pacsize = 480; // 30ms
+ cinst.plfreq = 16000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ SLEEP(2000);
+ TEST_LOG(" : iSAC superwideband \n");
+ cinst.pltype = 104;
+ cinst.rate = -1; // default rate
+ cinst.pacsize = 960; // 30ms
+ cinst.plfreq = 32000;
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ SLEEP(2000);
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+#else
+ TEST_LOG("Skipping extended iSAC API tests - "
+ "WEBRTC_CODEC_ISAC not defined\n");
+#endif // #if defined(WEBRTC_CODEC_ISAC)
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ delete ptrTransport;
+#endif
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestDtmf
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestDtmf() {
+ PrepareTest("Dtmf");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEDtmf* dtmf = _mgr.DtmfPtr();
+ VoECodec* codec = _mgr.CodecPtr();
+ VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+
+ //#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoEDtmf_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+ //#endif
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ // SetDtmfFeedbackStatus
+ TEST(SetDtmfFeedbackStatus & GetDtmfFeedbackStatus);
+ ANL();
+ bool dtmfFeedback = false, dtmfDirectFeedback = true;
+ TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
+ dtmfDirectFeedback));
+ TEST_MUSTPASS(!dtmfFeedback);
+ TEST_MUSTPASS(dtmfDirectFeedback);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
+ MARK();
+ SLEEP(500);
+
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, false));
+ TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
+ dtmfDirectFeedback));
+ TEST_MUSTPASS(dtmfFeedback);
+ TEST_MUSTPASS(dtmfDirectFeedback);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
+ MARK();
+ SLEEP(500);
+
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, true));
+ TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
+ dtmfDirectFeedback));
+ TEST_MUSTPASS(dtmfFeedback);
+ TEST_MUSTPASS(!dtmfDirectFeedback);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
+ MARK();
+ SLEEP(500);
+
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, false));
+ TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
+ dtmfDirectFeedback));
+ TEST_MUSTPASS(!dtmfFeedback);
+ TEST_MUSTPASS(dtmfDirectFeedback);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
+ MARK();
+ SLEEP(500);
+
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, true));
+ TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
+ dtmfDirectFeedback));
+ TEST_MUSTPASS(!dtmfFeedback);
+ TEST_MUSTPASS(!dtmfDirectFeedback);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, false));
+
+ AOK();
+ ANL();
+
+ // SendDtmf
+ TEST(SendDtmf);
+ ANL();
+
+ // Fail tests
+ // Event
+ // the eventcode is changed to unsigned char, so -1 will be interpreted as
+ // 255, 256->0
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, -1, false, 160, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 16, false, 160, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ // Length
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 99, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 60001, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 20, true, -1, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ // Volume
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 160, -1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 160, 37));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ // Without sending
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true));
+ MARK();
+ TEST_MUSTPASS(VE_NOT_SENDING != voe_base_->LastError());
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ // Testing Dtmf out-of-band: event, length and volume
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 16, true));
+ MARK();
+ SLEEP(500); // Flash, not audible
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 100, 10));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400, 10));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 160, 0));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 160, 36));
+ MARK();
+ SLEEP(500);
+
+ // Testing Dtmf inband: event, length and volume
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 15, false));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 100, 10));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 400, 10));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 160, 0));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 160, 36));
+ MARK();
+ SLEEP(500);
+
+ // Testing other events out-of-band: event and length
+ // These are not audible
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 17, true, 100, 10));
+ MARK();
+ SLEEP(200);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 100, 10));
+ MARK();
+ SLEEP(200);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 78, true, 100, 10));
+ MARK();
+ SLEEP(200);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 255, true, 100, 10));
+ MARK();
+ SLEEP(200);
+ // the minimum length is 100 for the telephoneevent
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 100, 10));
+ MARK();
+ SLEEP(200);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 1000, 10));
+ MARK();
+ SLEEP(1200);
+
+ AOK();
+ ANL();
+
+ // PlayDtmfTone
+ TEST(PlayDtmfTone);
+ ANL();
+ TEST_MUSTPASS(!dtmf->PlayDtmfTone(-1, 200, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->PlayDtmfTone(16, 200, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 9, 10));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 200, -1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 200, 37));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+
+ TEST_MUSTPASS(dtmf->PlayDtmfTone(0));
+ MARK();
+ SLEEP(500);
+ // the minimum length fo the DtmfTone is 100
+ TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 100, 10));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 2000, 10));
+ MARK();
+ SLEEP(2300);
+ TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 200, 0));
+ MARK();
+ SLEEP(500);
+ TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 200, 36));
+ MARK();
+ SLEEP(500);
+
+ AOK();
+ ANL();
+
+ // SetTelephoneEventDetection
+ TEST(SetTelephoneEventDetection);
+ ANL();
+ AOK();
+ ANL();
+
+ // Testing sending Dtmf under VAD/CN
+ TEST(SendDtmf - with VAD enabled);
+ ANL();
+ // Mute mic
+ TEST_MUSTPASS(volume->SetInputMute(0, true));
+ MARK();
+ // Enable VAD
+ TEST_MUSTPASS(codec->SetVADStatus(0, true));
+ MARK();
+ // Send Dtmf
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
+ MARK();
+ SLEEP(1000);
+ // Switch codec
+ CodecInst ci;
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ ci.channels = 1;
+ ci.pacsize = 480;
+ ci.plfreq = 16000;
+ strcpy(ci.plname, "ISAC");
+ ci.pltype = 103;
+ ci.rate = -1;
+#else
+ ci.pltype = 119;
+ strcpy(ci.plname, "isaclc");
+ ci.plfreq = 16000;
+ ci.pacsize = 320;
+ ci.channels = 1;
+ ci.rate = 40000;
+#endif
+ TEST_MUSTPASS(codec->SetSendCodec(0, ci));
+ MARK();
+ // Send Dtmf
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
+ MARK();
+ SLEEP(1000);
+ SLEEP(4000);
+ // Disable VAD
+ TEST_MUSTPASS(codec->SetVADStatus(0, false));
+ MARK();
+ // Unmute
+ TEST_MUSTPASS(volume->SetInputMute(0, false));
+ MARK();
+
+ AOK();
+ ANL();
+
+ // SetSendTelephoneEventPayloadType
+ TEST(SetSendTelephoneEventPayloadType);
+ ANL();
+ TEST_MUSTPASS(!dtmf->SetSendTelephoneEventPayloadType(0, 128));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+
+ TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 96));
+ MARK();
+ TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 127));
+ MARK();
+ TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 106));
+ MARK(); // restore default
+
+ AOK();
+ ANL();
+
+#ifdef WEBRTC_DTMF_DETECTION
+ TEST(RegisterTelephoneEventDetection - several channels); ANL();
+
+ ci.channels = 1;
+ ci.pacsize = 160;
+ ci.plfreq = 8000;
+ ci.pltype = 0;
+ ci.rate = 64000;
+ strcpy(ci.plname, "PCMU");
+ TEST_MUSTPASS(codec->SetSendCodec(0, ci));
+
+ int ch2 = voe_base_->CreateChannel();
+ TEST_MUSTPASS(voe_base_->SetSendDestination(ch2, 8002, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch2, 8002));
+ TEST_MUSTPASS(voe_base_->StartReceive(ch2));
+ TEST_MUSTPASS(codec->SetSendCodec(ch2, ci));
+ TEST_MUSTPASS(voe_base_->StartPlayout(ch2));
+ TEST_MUSTPASS(voe_base_->StartSend(ch2));
+ MARK();
+
+ DtmfCallback *d = new DtmfCallback();
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false));
+
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ // In-band
+ TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kInBand, *d));
+ TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kInBand, *d));
+ TEST_LOG("\nSending in-band telephone events:");
+ for(int i = 0; i < 16; i++)
+ {
+ TEST_LOG("\n %d ", i); fflush(NULL);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, false, 160, 10));
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, false, 160, 10));
+ SLEEP(500);
+ }
+ TEST_LOG("\nDetected %d events \n", d->counter);
+ TEST_MUSTPASS(d->counter != 32);
+ TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
+ TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
+
+ // Out-of-band
+ d->counter = 0;
+ TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kOutOfBand, *d));
+ TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kOutOfBand, *d));
+ TEST_LOG("\nSending out-band telephone events:");
+ for(int i = 0; i < 16; i++)
+ {
+ TEST_LOG("\n %d ", i); fflush(NULL);
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, true, 160, 10));
+ TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, true, 160, 10));
+ SLEEP(500);
+ }
+ TEST_LOG("\nDetected %d events \n", d->counter);
+ TEST_MUSTPASS(d->counter != 32);
+ TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
+ TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
+ delete d;
+
+ AOK(); ANL();
+#endif
+
+ TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, false));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestEncryption
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestEncryption() {
+ PrepareTest("Encryption");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoEEncryption* encrypt = _mgr.EncryptionPtr();
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(
+ GetFilename("VoEEncryption_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ unsigned char key1[30] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6,
+ 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 };
+
+#ifdef WEBRTC_SRTP
+ unsigned char key2[30]; // Different than key1 in first position
+ memcpy(key2, key1, 30);
+ key2[0] = 99;
+ unsigned char key3[30]; // Different than key1 in last position
+ memcpy(key3, key1, 30);
+ key3[29] = 99;
+ unsigned char key4[29]; // Same as key1 but shorter
+ memcpy(key4, key1, 29);
+
+ TEST(SRTP - Fail tests); ANL();
+
+ // Send
+ // Incorrect parameters when not all protection is enabled
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kNoProtection, key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kEncryption key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kAuthentication, key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ // Incorrect cipher key length
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 15,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 257,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 15, kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 257, kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // Incorrect auth key length
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 21, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 257, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // Incorrect auth tag length
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 21,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 20, 13,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+
+ // key NULL pointer
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, NULL));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+
+ // Same for receive
+ // Incorrect parameters when not all protection is enabled
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kNoProtection, key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kEncryption key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kAuthentication, key1));
+ TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
+ MARK();
+ // Incorrect cipher key length
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 15,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 257,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 15,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 257,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // Incorrect auth key length
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode,
+ 30, kAuthHmacSha1, 21, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // it crashed the application
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 257, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // Incorrect auth tag length
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 21,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // it crashed the application
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 20, 13,
+ kEncryptionAndAuthentication,
+ key1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ // key NULL pointer
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ NULL));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ ANL();
+
+ TEST(SRTP - Should hear audio at all time); ANL();
+
+ // Authentication only
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
+ 4, kAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+ ANL();
+
+ // No protection
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthNull, 0, 0,
+ kNoProtection, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthNull, 0, 0,
+ kNoProtection, key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+
+ // Encryption only
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0, kEncryption key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0,
+ kEncryption key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+
+ // Authentication only
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
+ 4, kAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+ ANL();
+
+ // Switching between keys
+ TEST(SRTP - Different keys - should hear audio at all time); ANL();
+
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key2));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key2));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key2));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key2));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+ ANL();
+
+ // Testing different keys that should be silent
+ TEST(SRTP - Should be silent or garbage); ANL();
+
+ // key1 and key2
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key2));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key2));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0, kEncryption key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0,
+ kEncryption key2));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key2));
+ MARK(); SLEEP(2000);
+
+ // key1 and key3
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key3));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key3));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0, kEncryption key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0,
+ kEncryption key3));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
+ 4, kAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key3));
+ MARK(); SLEEP(2000);
+
+ // key1 and key4
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key4));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication, key4));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1, 20, 4,
+ kEncryptionAndAuthentication,
+ key1));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0, kEncryption key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthNull, 0, 0,
+ kEncryption key4));
+ MARK(); SLEEP(2000);
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
+ 4, kAuthentication, key1));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
+ 20, 4, kAuthentication, key4));
+ MARK(); SLEEP(2000);
+ ANL();
+
+ // Back to normal
+ TEST(SRTP - Back to normal - should hear audio); ANL();
+
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ MARK(); SLEEP(2000);
+ ANL();
+
+ // SRTCP tests
+ TEST(SRTCP - Ignore voice or not); ANL();
+ VoERTP_RTCP* rtp_rtcp = _mgr.RTP_RTCPPtr();
+ char tmpStr[32];
+
+ // First test that RTCP packet is received and OK without encryption
+
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik1"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik1", tmpStr));
+
+ // Enable SRTP and SRTCP send and receive
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1, true));
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1, true));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik2"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik2", tmpStr));
+
+ // Disable SRTP and SRTCP send
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik3"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
+
+ // Enable SRTP send, but disable SRTCP send
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik4"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
+
+ // Enable SRTP and SRTCP send, disable SRTP and SRTCP receive
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1, true));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik5"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
+
+ // Enable SRTP receive, but disable SRTCP receive
+ TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik6"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
+
+ // Disable all
+ TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik7"));
+ MARK(); SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
+ TEST_MUSTPASS(_stricmp("Henrik7", tmpStr));
+ ANL();
+
+#else
+ TEST(SRTP disabled - Fail tests);
+ ANL();
+
+ TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, key1));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!encrypt->DisableSRTPSend(0));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!encrypt->DisableSRTPReceive(0));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ ANL();
+#endif
+ AOK();
+
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestExternalMedia
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestExternalMedia() {
+ PrepareTest("VoEExternalMedia");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEExternalMedia* xmedia = _mgr.ExternalMediaPtr();
+
+ // check if this interface is supported
+ if (!xmedia) {
+ TEST_LOG("VoEExternalMedia is not supported!");
+ return -1;
+ }
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(
+ GetFilename("VoEExternalMedia_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(
+ kTraceStateInfo | kTraceStateInfo | kTraceWarning |
+ kTraceError | kTraceCritical | kTraceApiCall |
+ kTraceMemory | kTraceInfo));
+#endif
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ int getLen = 0;
+ WebRtc_Word16 vector[32000];
+ memset(vector, 0, 32000 * sizeof(short));
+
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+
+ // ExternalPlayoutGetData
+ TEST(ExternalPlayoutGetData);
+ ANL();
+
+ TEST_MUSTPASS(!xmedia->SetExternalPlayoutStatus(true));
+ TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, 100, getLen));
+ TEST_MUSTPASS(VE_INVALID_OPERATION != voe_base_->LastError());
+
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(xmedia->SetExternalPlayoutStatus(true));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ TEST_MUSTPASS(xmedia->ExternalPlayoutGetData(vector, 48000, 0, getLen));
+ TEST_MUSTPASS(480 != getLen);
+ SLEEP(10);
+ TEST_MUSTPASS(xmedia->ExternalPlayoutGetData(vector, 16000, 3000, getLen));
+ TEST_MUSTPASS(160 != getLen);
+ SLEEP(10);
+
+ TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 8000, 100, getLen));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, -1, getLen));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(xmedia->SetExternalPlayoutStatus(false));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ // SetExternalRecording
+ TEST(SetExternalRecording);
+ ANL();
+
+ TEST_MUSTPASS(!xmedia->SetExternalRecordingStatus(true));
+ TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, 20));
+ TEST_MUSTPASS(VE_INVALID_OPERATION != voe_base_->LastError());
+
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(xmedia->SetExternalRecordingStatus(true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ TEST_MUSTPASS(xmedia->ExternalRecordingInsertData(vector, 480, 48000, 0));
+ SLEEP(10);
+ TEST_MUSTPASS(xmedia->ExternalRecordingInsertData(vector, 640, 16000, 0));
+ SLEEP(40);
+
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, -1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 80, 8000, 20));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 0, 16000, 20));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 80, 16000, 20));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 500, 16000, 20));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(xmedia->SetExternalRecordingStatus(false));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+#else // #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ TEST_MUSTPASS(!xmedia->SetExternalPlayoutStatus(true));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, 100, getLen));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->SetExternalRecordingStatus(true));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+ TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, 20));
+ TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
+
+#endif // #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ ANL();
+ AOK();
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestFile
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestFile() {
+ PrepareTest("File");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoECodec* codec = _mgr.CodecPtr();
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(
+ GetFilename("VoEFile_trace.txt"))); MARK();
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ const int dT(100);
+
+ TEST(StartPlayingFileLocally);
+ ANL();
+ TEST(StopPlayingFileLocally);
+ ANL();
+
+ voe_base_->StopPlayout(0);
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));MARK();
+ voe_base_->StartPlayout(0);
+ MARK(); // file should be mixed in and played out
+ SLEEP(dT);
+ TEST_MUSTPASS(!file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));
+ MARK(); // should fail (must stop first)
+ TEST_MUSTPASS(voe_base_->LastError() != VE_ALREADY_PLAYING);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));
+ MARK(); // should work again (restarts file)
+ SLEEP(dT);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm"), false, kFileFormatPcm16kHzFile));
+ MARK();
+ SLEEP(dT);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long8.pcm"), false, kFileFormatPcm8kHzFile));
+ MARK();
+ SLEEP(dT);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.wav"), false, kFileFormatPcm8kHzFile));
+ MARK();
+ SLEEP(dT);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long8mulaw.wav"), false,
+ kFileFormatPcm8kHzFile));
+ MARK();
+ SLEEP(dT);
+
+ // add compressed tests here...
+
+ // TEST_MUSTPASS(file->StopPlayingFileLocally(0)); MARK();
+ // TEST_MUSTPASS(file->StartPlayingFileLocally(
+ // 0, GetResource("audio_short16.pcm"), true,
+ // kFileFormatPcm16kHzFile)); MARK(); // loop
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_short16.pcm"), false,
+ kFileFormatPcm16kHzFile, 1.0, 0, 2000));
+ MARK(); // play segment
+ SLEEP(2500);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(!file->StartPlayingFileLocally(
+ 0, GetResource("audio_short16.pcm"), false,
+ kFileFormatPcm16kHzFile, 1.0, 2000, 1000));
+ MARK(); // invalid segment
+ TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
+ TEST_MUSTPASS(!file->StartPlayingFileLocally(
+ 0, GetResource("audio_short16.pcm"), false,
+ kFileFormatPcm16kHzFile, 1.0, 21000, 30000));
+ MARK(); // start > file size
+ TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
+ TEST_MUSTPASS(!file->StartPlayingFileLocally(
+ 0, GetResource("audio_short16.pcm"), false,
+ kFileFormatPcm16kHzFile, 1.0, 100, 100));
+ MARK(); // invalid segment
+ TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));
+ MARK(); // should work again (restarts file)
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ TEST_MUSTPASS(!file->StartPlayingFileLocally(0, (InStream*)NULL));
+ MARK(); // just do it
+ TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
+
+ AOK();
+ ANL();
+
+ TEST(IsPlayingFileLocally);
+ ANL();
+
+ TEST_MUSTPASS(0 != file->IsPlayingFileLocally(0));
+ MARK(); // inactive
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));
+ MARK();
+ TEST_MUSTPASS(1 != file->IsPlayingFileLocally(0));
+ MARK(); // active
+ AOK();
+ ANL();
+
+ TEST(ScaleLocalFilePlayout);
+ ANL();
+ TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 1.0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.5));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.25));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ MARK();
+ AOK();
+ ANL();
+
+ // Replace microphone with file and play out on remote side
+ // All channels, per channel
+ // Different mixing frequencies
+ TEST(StartPlayingFileAsMicrophone);
+ ANL();
+ TEST(IsPlayingFileAsMicrophone);
+ ANL();
+ TEST(ScaleFileAsMicrophonePlayout);
+ ANL();
+ CodecInst tempCodec;
+ for (int ch = -1; ch < 1; ++ch) // Channel -1 and 0
+ {
+ TEST_LOG("Testing channel = %d \n", ch);
+ for (int fs = 1; fs < 4; ++fs) // nb, wb and swb codecs
+ {
+ switch (fs) {
+ case 1: // nb
+ TEST_LOG("Testing with nb codec \n");
+ tempCodec.channels = 1;
+ tempCodec.pacsize = 160;
+ tempCodec.plfreq = 8000;
+ strcpy(tempCodec.plname, "PCMU");
+ tempCodec.pltype = 0;
+ tempCodec.rate = 64000;
+ break;
+ case 2: // wb
+#ifdef WEBRTC_CODEC_ISAC
+ TEST_LOG("Testing with wb codec \n");
+ tempCodec.channels = 1;
+ tempCodec.pacsize = 480;
+ tempCodec.plfreq = 16000;
+ strcpy(tempCodec.plname, "ISAC");
+ tempCodec.pltype = 103;
+ tempCodec.rate = 32000;
+ break;
+#else
+ TEST_LOG("NOT testing with wb codec - "
+ "WEBRTC_CODEC_ISAC not defined \n");
+ continue;
+#endif
+ case 3: // swb
+#ifdef WEBRTC_CODEC_PCM16
+ TEST_LOG("Testing with swb codec \n");
+ tempCodec.channels = 1;
+ tempCodec.pacsize = 640;
+ tempCodec.plfreq = 32000;
+ strcpy(tempCodec.plname, "L16");
+ tempCodec.pltype = 125;
+ tempCodec.rate = 512000;
+ break;
+#else
+ TEST_LOG("NOT testing with swb codec -"
+ " WEBRTC_CODEC_PCM16 not defined \n");
+ continue;
+#endif
+ }
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, tempCodec));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(codec->SetSendCodec(0, tempCodec));
+
+ TEST_LOG("File 1 in 16 kHz no mix, 2 in 16 kHz mix,"
+ " 3 in 8 kHz no mix, 4 in 8 kHz mix \n");
+
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long16.pcm")));
+ MARK(); // don't mix
+ SLEEP(2000);
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long16.wav"), false, true,
+ kFileFormatWavFile));
+ MARK(); // mix
+ SLEEP(2000);
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long8.pcm"), false, false,
+ kFileFormatPcm8kHzFile));
+ MARK(); // don't mix
+ SLEEP(2000);
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ MARK();
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long8.pcm"), false, true,
+ kFileFormatPcm8kHzFile));
+ MARK(); // mix
+ SLEEP(2000);
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ MARK();
+ TEST_MUSTPASS(!file->StartPlayingFileAsMicrophone(
+ ch, (InStream*)NULL));
+ MARK(); // force error
+ AOK();
+ ANL();
+
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long16.pcm")));
+ TEST_MUSTPASS(1 != file->IsPlayingFileAsMicrophone(ch));
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ TEST_MUSTPASS(0 != file->IsPlayingFileAsMicrophone(ch));
+ AOK();
+ ANL();
+
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ ch, GetResource("audio_long16.pcm")));
+ TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 1.0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.5));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.25));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
+ MARK();
+ AOK();
+ ANL();
+ }
+ }
+
+ // Record speaker signal to file
+
+ CodecInst fcomp = { 0, "L16", 8000, 80, 1, 128000 };
+
+ TEST(StartRecordingPlayout);
+ ANL();
+ TEST(StopRecordingPlayout);
+ ANL();
+
+ TEST_MUSTPASS(file->StartRecordingPlayout(0,
+ GetFilename("rec_play16.pcm")));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "L16");
+ TEST_MUSTPASS(file->StartRecordingPlayout(0, GetFilename("rec_play8.wav"),
+ &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ fcomp.plfreq = 16000;
+ strcpy(fcomp.plname, "L16");
+ TEST_MUSTPASS(file->StartRecordingPlayout(0, GetFilename("rec_play16.wav"),
+ &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ fcomp.pltype = 0;
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "PCMU");
+ fcomp.rate = 64000;
+ fcomp.pacsize = 160;
+ fcomp.channels = 1;
+
+ TEST_MUSTPASS(file->StartRecordingPlayout(0,
+ GetFilename("rec_play_pcmu.wav"),
+ &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ fcomp.pltype = 8;
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "PCMA");
+ TEST_MUSTPASS(file->StartRecordingPlayout(0,
+ GetFilename("rec_play_pcma.wav"),
+ &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ fcomp.pltype = 97;
+ fcomp.pacsize = 240;
+ fcomp.rate = 13300;
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "ILBC");
+ TEST_MUSTPASS(file->StartRecordingPlayout(0,
+ GetFilename("rec_play.ilbc"),
+ &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(0));
+ MARK();
+
+ TEST_MUSTPASS(file->StartRecordingPlayout(
+ -1, GetFilename("rec_play16_mixed.pcm")));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(-1));
+ MARK();
+
+ // TEST_MUSTPASS(file->StopPlayingFileLocally(0)); // Why should this work?
+ TEST_LOG("\nplaying out...\n");
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetFilename("rec_play.ilbc"), false, kFileFormatCompressedFile));
+ MARK();
+ SLEEP(2000);
+
+ AOK();
+ ANL();
+
+ // Record microphone signal to file
+ TEST(StartRecordingMicrophone);
+ ANL();
+ TEST(StopRecordingMicrophone);
+ ANL();
+
+ TEST_MUSTPASS(file->StartRecordingMicrophone(GetFilename("rec_mic16.pcm")));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+
+ voe_base_->StopSend(0);
+ TEST_MUSTPASS(file->StartRecordingMicrophone(GetFilename("rec_mic16.pcm")));
+ MARK(); // record without sending as well
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+ voe_base_->StartSend(0); // restore sending
+
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "L16");
+ TEST_MUSTPASS(file->StartRecordingMicrophone(
+ GetFilename("rec_play8.wav"), &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+
+ fcomp.plfreq = 16000;
+ strcpy(fcomp.plname, "L16");
+ TEST_MUSTPASS(file->StartRecordingMicrophone(
+ GetFilename("rec_play16.wav"), &fcomp));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+
+ // FT#1810, the following test is to make sure StartRecordingCall will
+ // record both mic and file
+ TEST_LOG("StartRecordingCall, record both mic and file in specific"
+ " channels \n");
+ TEST_LOG("Create maxnumofchannels \n");
+ for (int i = 1; i < voe_base_->MaxNumOfChannels(); i++) {
+ int ch = voe_base_->CreateChannel();
+ TEST_MUSTPASS(ch == -1);
+ TEST_MUSTPASS(voe_base_->StopPlayout(ch));
+ }
+
+ TEST_MUSTPASS(voe_base_->SetSendDestination(1, 12356, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(1, 12356));
+ TEST_MUSTPASS(voe_base_->StartReceive(1));
+ TEST_MUSTPASS(voe_base_->StopPlayout(1));
+ TEST_MUSTPASS(voe_base_->StartSend(1));
+ TEST_MUSTPASS(voe_base_->StartPlayout(1));
+
+ TEST_LOG("ALways playing audio_long16.pcm for "
+ "channel 0 in background \n");
+ fcomp.plfreq = 16000;
+ strcpy(fcomp.plname, "L16");
+ TEST_LOG("Recording microphone to L16, please speak \n");
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ 0, GetResource("audio_long16.pcm"), true , true));
+ TEST_MUSTPASS(file->StartRecordingMicrophone(
+ GetFilename("rec_play_ch.wav"), &fcomp));
+ MARK();
+ SLEEP(3000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_LOG("Playing recording file, you should only hear what you said \n");
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetFilename("rec_play_ch.wav"), false, kFileFormatWavFile));
+ SLEEP(2500);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ TEST_LOG("Recording microphone 0 to L16, please speak \n");
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ -1, GetResource("audio_long16.pcm"), true , true));
+ TEST_MUSTPASS(file->StartRecordingMicrophone(
+ GetFilename("rec_play_ch_0.wav"), &fcomp));
+ MARK();
+ SLEEP(3000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(-1));
+ TEST_LOG("Playing recording file, you should hear what you said and"
+ " audio_long16.pcm \n");
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetFilename("rec_play_ch_0.wav"), false, kFileFormatWavFile));
+ SLEEP(2500);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ TEST_LOG("Recording microphone to ilbc, please speak \n");
+ strcpy(fcomp.plname, "ilbc");
+ fcomp.plfreq = 8000;
+ fcomp.pacsize = 160;
+ fcomp.rate = 15200;
+ fcomp.channels = 1;
+ fcomp.pltype = 97;
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
+ 0, GetResource("audio_long16.pcm"), true , true));
+ TEST_MUSTPASS(file->StartRecordingMicrophone(
+ GetFilename("rec_play_ch_0.ilbc"), &fcomp));
+ MARK();
+ SLEEP(3000);
+ TEST_MUSTPASS(file->StopRecordingMicrophone());
+ MARK();
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_LOG("Playing recording file, you should only hear what you said \n");
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetFilename("rec_play_ch_0.ilbc"), false,
+ kFileFormatCompressedFile));
+ SLEEP(2500);
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ for (int i = 1; i < voe_base_->MaxNumOfChannels(); i++) {
+ TEST_MUSTPASS(voe_base_->DeleteChannel(i));
+ }
+
+ AOK();
+ ANL();
+
+ // Record mixed (speaker + microphone) signal to file
+
+
+#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+ TEST(StartRecordingSpeakerStereo);
+ ANL();
+ TEST(StopRecordingSpeakerStereo);
+ ANL();
+
+ VoEHardware* hardware = _mgr.HardwarePtr();
+ TEST_MUSTPASS(NULL == hardware);
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+#if defined(_WIN32)
+ TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
+#else
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
+#endif
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ MARK();
+
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+#if defined(_WIN32)
+ TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
+#else
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
+#endif
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ AOK();
+ ANL();
+#else
+ TEST_LOG("Skipping stereo record tests -"
+ " MAC_IPHONE or WEBRTC_ANDROID is defined \n");
+#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+ // Conversion between different file formats
+
+#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+#endif
+
+ TEST(ConvertPCMToWAV);
+ ANL();
+
+ TEST_MUSTPASS(file->ConvertPCMToWAV(
+ GetResource("audio_long16.pcm"),
+ GetFilename("singleUserDemoConv.wav")));
+ MARK();
+ TEST_MUSTPASS(!file->ConvertPCMToWAV((InStream*)NULL,
+ (OutStream*)NULL));MARK(); // invalid stream handles
+ AOK();
+ ANL();
+
+ TEST(ConvertWAVToPCM);
+ ANL();
+
+ TEST_MUSTPASS(file->ConvertWAVToPCM(
+ GetResource("audio_long16.wav"),
+ GetFilename("singleUserDemoConv.pcm")));
+ MARK();
+ TEST_MUSTPASS(!file->ConvertWAVToPCM((InStream*)NULL, (OutStream*)NULL));
+ MARK(); // invalid stream handles
+ AOK();
+ ANL();
+
+ TEST(ConvertPCMToCompressed);
+ ANL();
+
+ fcomp.plfreq = 16000;
+ strcpy(fcomp.plname, "L16");
+ TEST_MUSTPASS(!file->ConvertPCMToCompressed(
+ GetResource("audio_long16.pcm"),
+ GetFilename("singleUserDemoConv16_dummy.wav"), &fcomp));
+ MARK(); // should not be supported
+
+ fcomp.plfreq = 8000;
+ strcpy(fcomp.plname, "ilbc");
+ fcomp.pacsize = 160;
+ fcomp.rate = 15200;
+ fcomp.pltype = 97;
+ fcomp.channels = 1;
+ TEST_MUSTPASS(file->ConvertPCMToCompressed(
+ GetResource("audio_long16.pcm"),
+ GetFilename("singleUserDemoConv.ilbc"), &fcomp));MARK();
+ AOK();ANL();
+
+ TEST(ConvertCompressedToPCM);
+ ANL();
+
+ TEST_MUSTPASS(file->ConvertCompressedToPCM(
+ GetFilename("singleUserDemoConv.ilbc"),
+ GetFilename("singleUserDemoConv_ilbc.pcm")));MARK();
+ TEST_MUSTPASS(!file->ConvertCompressedToPCM(
+ GetResource("audio_long16.pcm"),
+ GetFilename("singleUserDemoConv_dummy.pcm")));MARK();
+ AOK();ANL();
+
+#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+#endif
+
+ // Misc file functions
+ TEST(GetFileDuration);
+ ANL();
+
+ int dur;
+
+ TEST_MUSTPASS(file->GetFileDuration(
+ GetResource("audio_long16.pcm"), dur));
+ TEST_MUSTPASS(file->GetFileDuration(
+ GetResource("audio_long8.pcm"), dur, kFileFormatPcm8kHzFile));
+ TEST_MUSTPASS(file->GetFileDuration(
+ GetResource("audio_long16.pcm"), dur, kFileFormatPcm16kHzFile));
+ TEST_MUSTPASS(file->GetFileDuration(
+ GetResource("audio_long16.wav"), dur, kFileFormatPcm8kHzFile));
+ TEST_MUSTPASS(file->GetFileDuration(
+ GetFilename("singleUserDemoConv.ilbc"), dur,
+ kFileFormatCompressedFile));
+
+ AOK();
+ ANL();
+
+ TEST(GetPlaybackPosition);
+ ANL();
+
+ int pos;
+
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource("audio_long16.pcm")));
+ SLEEP(1000);
+ TEST_MUSTPASS(file->GetPlaybackPosition(0, pos));
+ MARK(); // position should be ~1000
+ SLEEP(1000);
+ TEST_MUSTPASS(file->GetPlaybackPosition(0, pos));
+ MARK(); // position should be ~2000
+ // SLEEP(70*1000);
+ // file is no longer playing
+ // TEST_MUSTPASS(file->GetPlaybackPosition(0, pos)); MARK();
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ AOK();
+ ANL();
+
+ // These tests are related to defect 5136
+ // They play .wav files with different sample freq for 5s
+ char localFiles[7][50] = { "audio_tiny8.wav", "audio_tiny11.wav",
+ "audio_tiny16.wav", "audio_tiny22.wav", "audio_tiny32.wav",
+ "audio_tiny44.wav", "audio_tiny48.wav" };
+ char freq[7][5] = { "8", "11", "16", "22", "32", "44.1", "48" };
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ for (int i = 0; i < 7; i++) {
+ TEST_LOG("Playing file %s, in %s KHz \n", localFiles[i], freq[i]);
+ TEST_MUSTPASS(file->StartPlayingFileLocally(
+ 0, GetResource(localFiles[i]),false, kFileFormatWavFile, 1));
+ SLEEP(4500); // The file should not end
+ TEST_MUSTPASS(file->StopPlayingFileLocally(0));
+ }
+
+ // TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0)); // Should not work
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ AOK();
+ ANL();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestMixing
+// ----------------------------------------------------------------------------
+
+// Creates and mixes |num_remote_channels| which play a file "as microphone"
+// with |num_local_channels| which play a file "locally", using a constant
+// amplitude of |input_value|.
+//
+// The mixed output is verified to always fall between |max_output_value| and
+// |min_output_value|, after a startup phase.
+int VoEExtendedTest::RunMixingTest(int num_remote_channels,
+ int num_local_channels,
+ int16_t input_value,
+ int16_t max_output_value,
+ int16_t min_output_value) {
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoECodec* codec = _mgr.CodecPtr();
+ VoEAudioProcessing* apm = _mgr.APMPtr();
+
+ // Use L16 at 16kHz to minimize distortion (file recording is 16kHz
+ // and resampling will cause large distortions).
+ CodecInst codec_inst;
+ strcpy(codec_inst.plname, "L16");
+ codec_inst.channels = 1;
+ codec_inst.rate = 256000;
+ codec_inst.plfreq = 16000;
+ codec_inst.pltype = 105;
+ codec_inst.pacsize = 160;
+
+ apm->SetNsStatus(false);
+ apm->SetAgcStatus(false);
+ apm->SetEcStatus(false);
+
+ const char file_to_generate_name[] = "dc_file.pcm";
+ const char* input_filename = file_to_generate_name;
+ FILE* file_to_generate = fopen(file_to_generate_name, "wb");
+ ASSERT_TRUE(file_to_generate != NULL);
+ for (int i = 0; i < 160 * 100 * 5; i++) {
+ fwrite(&input_value, sizeof(input_value), 1, file_to_generate);
+ }
+ fclose(file_to_generate);
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ std::vector<int> local_channels(num_local_channels);
+ for (int i = 0; i < num_local_channels; ++i) {
+ local_channels[i] = voe_base_->CreateChannel();
+ ASSERT_TRUE(local_channels[i] != -1);
+ TEST_MUSTPASS(voe_base_->StartPlayout(local_channels[i]));
+ TEST_MUSTPASS(file->StartPlayingFileLocally(local_channels[i],
+ input_filename,
+ true));
+ }
+
+ std::vector<int> remote_channels(num_remote_channels);
+ for (int i = 0; i < num_remote_channels; ++i) {
+ remote_channels[i] = voe_base_->CreateChannel();
+ ASSERT_TRUE(remote_channels[i] != -1);
+ TEST_MUSTPASS(codec->SetRecPayloadType(remote_channels[i], codec_inst));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(remote_channels[i],
+ 1234 + 2 * i));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(remote_channels[i],
+ 1234 + 2 * i,
+ "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(remote_channels[i]));
+ TEST_MUSTPASS(voe_base_->StartPlayout(remote_channels[i]));
+ TEST_MUSTPASS(codec->SetSendCodec(remote_channels[i], codec_inst));
+ TEST_MUSTPASS(voe_base_->StartSend(remote_channels[i]));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(remote_channels[i],
+ input_filename,
+ true));
+ }
+
+ const char mix_result[] = "mix_result.pcm";
+ TEST_MUSTPASS(file->StartRecordingPlayout(-1/*record meeting*/,
+ mix_result));
+ TEST_LOG("Playing %d remote channels.\n", num_remote_channels);
+ TEST_LOG("Playing %d local channels.\n", num_local_channels);
+ SLEEP(5000);
+ TEST_MUSTPASS(file->StopRecordingPlayout(-1));
+ TEST_LOG("Stopping\n");
+
+ for (int i = 0; i < num_local_channels; ++i) {
+ TEST_MUSTPASS(voe_base_->StopPlayout(local_channels[i]));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(local_channels[i]));
+ }
+
+ for (int i = 0; i < num_remote_channels; ++i) {
+ TEST_MUSTPASS(voe_base_->StopSend(remote_channels[i]));
+ TEST_MUSTPASS(voe_base_->StopPlayout(remote_channels[i]));
+ TEST_MUSTPASS(voe_base_->StopReceive(remote_channels[i]));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(remote_channels[i]));
+ }
+
+ FILE* verification_file = fopen(mix_result, "rb");
+ ASSERT_TRUE(verification_file != NULL);
+ int16_t mix_value = 0;
+ // Skip the first 100 ms to avoid initialization and ramping-in effects.
+ ASSERT_TRUE(fseek(verification_file, sizeof(int16_t) * 1600, SEEK_SET) == 0);
+ while (fread(&mix_value, sizeof(mix_value), 1, verification_file)) {
+ ASSERT_TRUE(mix_value <= max_output_value)
+ ASSERT_TRUE(mix_value >= min_output_value);
+ }
+ fclose(verification_file);
+
+ return 0;
+}
+
+// TODO(andrew): move or copy these to the mixer module test when possible.
+int VoEExtendedTest::TestMixing() {
+ // These tests assume a maxmium of three mixed participants. We allow a
+ // +/- 10% range around the expected output level to accout for distortion
+ // from coding and processing in the loopback chain.
+
+ // Create four channels and make sure that only three are mixed.
+ TEST_LOG("Test max-three-participant mixing.\n");
+ int16_t input_value = 1000;
+ int16_t expected_output = input_value * 3;
+ if (RunMixingTest(4, 0, input_value, 1.1 * expected_output,
+ 0.9 * expected_output) != 0) {
+ return -1;
+ }
+
+ // Ensure the mixing saturation protection is working. We can do this because
+ // the mixing limiter is given some headroom, so the expected output is less
+ // than full scale.
+ TEST_LOG("Test mixing saturation protection.\n");
+ input_value = 20000;
+ expected_output = 29204; // = -1 dBFS, the limiter headroom.
+ // If this isn't satisfied, we're not testing anything.
+ assert(input_value * 3 > 32767);
+ assert(1.1 * expected_output < 32767);
+ if (RunMixingTest(3, 0, input_value, 1.1 * expected_output,
+ 0.9 * expected_output) != 0) {
+ return -1;
+ }
+
+ // Ensure the mixing saturation protection is not applied when only using a
+ // single channel.
+ TEST_LOG("Test saturation protection has no effect on one channel.\n");
+ input_value = 32767;
+ expected_output = 32767;
+ // If this isn't satisfied, we're not testing anything.
+ assert(0.95 * expected_output > 29204); // = -1 dBFS, the limiter headroom.
+ if (RunMixingTest(1, 0, input_value, expected_output,
+ 0.95 * expected_output) != 0) {
+ return -1;
+ }
+
+ TEST_LOG("Test combinations of 'anonymous' participants and regular "
+ "participants.\n");
+ input_value = 1000;
+ expected_output = input_value * 2;
+ if (RunMixingTest(1, 1, input_value, 1.1 * expected_output,
+ 0.9 * expected_output) != 0) {
+
+ return -1;
+ }
+
+ expected_output = input_value * 4;
+ if (RunMixingTest(3, 1, input_value, 1.1 * expected_output,
+ 0.9 * expected_output) != 0) {
+
+ return -1;
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestHardware
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestHardware() {
+ PrepareTest("Hardware");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEHardware* hardware = _mgr.HardwarePtr();
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
+ "VoEHardware_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ // Set/GetAudioDeviceLayer
+ TEST(Set/GetAudioDeviceLayer);
+ ANL();
+ AudioLayers wantedLayer = kAudioPlatformDefault;
+ AudioLayers givenLayer;
+
+#if defined(_WIN32)
+ wantedLayer = kAudioWindowsCore;
+ hardware->SetAudioDeviceLayer(wantedLayer);
+ TEST_LOG("If you run on XP or below, CoreAudio "
+ "should not be able to set.\n");
+ TEST_LOG("If you run on Vista or above, CoreAudio "
+ "should be able to set.\n");
+ TEST_LOG("Verify that this is the case.\n");
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ if(givenLayer == kAudioWindowsCore)
+ {
+ TEST_LOG("CoreAudio was set\n");
+ }
+ else
+ {
+ TEST_LOG("CoreAudio was *not* set\n");
+ }
+
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ wantedLayer = kAudioWindowsWave;
+ TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_LOG("Wave audio should always be able to set.\n");
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ if(givenLayer == kAudioWindowsWave)
+ {
+ TEST_LOG("Wave audio was set\n");
+ }
+ else
+ {
+ TEST_LOG("Wave audio was not set\n");
+ }
+
+ TEST_MUSTPASS(voe_base_->Terminate());
+ // end _WIN32
+#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+ wantedLayer = kAudioLinuxPulse;
+ TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_LOG("If you run on Linux with no/unsupported PA version, PulseAudio "
+ "7should not be able to set.\n");
+ TEST_LOG("If you run on Linux with supported PA version running, PulseAudio"
+ " should be able to set.\n");
+ TEST_LOG("Verify that this is the case.\n");
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ if(givenLayer == kAudioLinuxPulse)
+ {
+ TEST_LOG("\nPulseAudio was set\n");
+ }
+ else
+ {
+ TEST_LOG("\nPulseAudio was not set\n");
+ }
+
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ wantedLayer = kAudioLinuxAlsa;
+ TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_LOG("ALSA audio should always be able to set.\n");
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ if(givenLayer == kAudioLinuxAlsa)
+ {
+ TEST_LOG("\nALSA audio was set\n");
+ }
+ else
+ {
+ TEST_LOG("\nALSA audio was not set\n");
+ }
+
+ TEST_MUSTPASS(voe_base_->Terminate());
+#endif // defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+ // Invalid arguments
+ wantedLayer = (AudioLayers) 17;
+ TEST_MUSTPASS(-1 != hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+
+ // Basic usage
+ wantedLayer = kAudioPlatformDefault;
+ TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ TEST_MUSTPASS(givenLayer != wantedLayer);
+ MARK();
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ wantedLayer = kAudioPlatformDefault;
+ TEST_MUSTPASS(-1 != hardware->SetAudioDeviceLayer(wantedLayer));
+ TEST_MUSTPASS(VE_ALREADY_INITED != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
+ MARK();
+ switch (givenLayer) {
+ case kAudioPlatformDefault:
+ // already set above
+ break;
+ case kAudioWindowsCore:
+ TEST_LOG("\nRunning kAudioWindowsCore\n");
+ break;
+ case kAudioWindowsWave:
+ TEST_LOG("\nRunning kAudioWindowsWave\n");
+ break;
+ case kAudioLinuxAlsa:
+ TEST_LOG("\nRunning kAudioLinuxAlsa\n");
+ break;
+ case kAudioLinuxPulse:
+ TEST_LOG("\nRunning kAudioLinuxPulse\n");
+ break;
+ default:
+ TEST_LOG("\nERROR: Running unknown audio layer!!\n");
+ return -1;
+ }
+ ANL();
+
+#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+ // GetRecording/PlayoutDeviceStatus
+ TEST(Getrecording/PlayoutDeviceStatus);
+ ANL();
+ bool isRecAvailable = false;
+ bool isPlayAvailable = false;
+ TEST_MUSTPASS(hardware->GetRecordingDeviceStatus(isRecAvailable));
+ TEST_MUSTPASS(!isRecAvailable);
+ MARK();
+ TEST_MUSTPASS(hardware->GetPlayoutDeviceStatus(isPlayAvailable));
+ TEST_MUSTPASS(!isPlayAvailable);
+ MARK();
+
+ ANL();
+
+ int nRec = 0, nPlay = 0;
+ char devName[128];
+ char guidName[128];
+ int idx;
+
+ TEST_MUSTPASS(hardware->GetNumOfPlayoutDevices(nPlay));
+
+ // GetPlayoutDeviceName
+ TEST(GetPlayoutDeviceName);
+ ANL();
+ TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(nPlay, devName,
+ guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(-2, devName, guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(nPlay+1, devName,
+ guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(0, NULL, guidName));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(hardware->GetPlayoutDeviceName(0, devName, NULL));
+
+ // default tests
+ for (idx = 0; idx < nPlay; idx++) {
+ TEST_MUSTPASS(hardware->GetPlayoutDeviceName(idx, devName, guidName));
+ MARK();
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(idx));
+ }
+
+ ANL();
+
+ TEST_MUSTPASS(hardware->GetNumOfRecordingDevices(nRec));
+
+ // GetRecordingDeviceName
+ TEST(GetRecordingDeviceName);
+ ANL();
+ TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(nRec, devName,
+ guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(-2, devName, guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(nRec+1, devName,
+ guidName));
+ TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(0, NULL, guidName));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(hardware->GetRecordingDeviceName(0, devName, NULL));
+
+ // default tests
+ for (idx = 0; idx < nRec; idx++) {
+ TEST_MUSTPASS(hardware->GetRecordingDeviceName(idx, devName, guidName));
+ MARK();
+ TEST_MUSTPASS(hardware->SetRecordingDevice(idx));
+ }
+ ANL();
+
+ // // SetRecordingDevice
+ TEST(SetRecordingDevice);
+ ANL();
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0));
+ MARK();
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0, kStereoLeft));
+ MARK();
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0, kStereoRight));
+ MARK();
+ ANL();
+
+ // SetPlayoutDevice
+ TEST(SetPlayoutDevice);
+ ANL();
+#if defined(_WIN32)
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(-1)); MARK();
+#else
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
+ MARK();
+#endif
+ ANL();
+#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
+#if defined(MAC_IPHONE)
+ TEST(ResetSoundDevice); ANL();
+
+ for (int p=0; p<=60; p+=20)
+ {
+ TEST_LOG("Resetting sound device several times with pause %d ms\n", p);
+ for (int l=0; l<50; ++l)
+ {
+ TEST_MUSTPASS(hardware->ResetAudioDevice()); MARK();
+ SLEEP(p);
+ }
+ ANL();
+ }
+
+ TEST_LOG("Start streaming - verify the audio after each batch of resets \n");
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(2000);
+
+ SLEEP(2000);
+ for (int p=0; p<=60; p+=20)
+ {
+ TEST_LOG("Resetting sound device several time with pause %d ms\n", p);
+ for (int l=0; l<20; ++l)
+ {
+ TEST_MUSTPASS(hardware->ResetAudioDevice()); MARK();
+ SLEEP(p);
+ }
+ ANL();
+ SLEEP(2000);
+ }
+
+ TEST_LOG("Stop streaming \n");
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+#endif // defined(MAC_IPHONE))
+#ifdef MAC_IPHONE
+ TEST_LOG("\nNOTE: Always run hardware tests also without extended tests "
+ "enabled,\nsince the extended tests are pre-streaming tests only.\n");
+#endif
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ ANL();
+ AOK();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestNetEqStats
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestNetEqStats() {
+ PrepareTest("NetEqStats (!EMPTY!)");
+
+ AOK();
+ ANL();
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestNetwork
+//
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestNetwork() {
+ PrepareTest("Network");
+
+#ifdef WEBRTC_ANDROID
+ int sleepTime = 200;
+ int sleepTime2 = 250;
+#elif defined(MAC_IPHONE) // MAC_IPHONE needs more delay for getSourceInfo()
+ int sleepTime = 150;
+ int sleepTime2 = 200;
+#else
+ int sleepTime = 100;
+ int sleepTime2 = 200;
+#endif
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoENetwork* netw = _mgr.NetworkPtr();
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
+ "VoENetwork_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+ // ------------------------------------------------------------------------
+ // >> GetLocalIP
+ //
+ // State: VE initialized, no existing channels
+ TEST(GetLocalIP);
+ ANL();
+
+#ifdef MAC_IPHONE
+ // Should fail
+ TEST_MUSTPASS(!netw->GetLocalIP(NULL, 0)); MARK();
+ TEST_ERROR(VE_FUNC_NOT_SUPPORTED);
+
+ ANL();
+ printf("NOTE: Local IP must be set in source code (line %d) \n",
+ __LINE__ + 1);
+ const char* localIP = "192.168.1.4";
+
+#else
+ char localIP[64];
+
+ // invalid parameter
+ TEST_MUSTPASS(!netw->GetLocalIP(NULL));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+
+ // default function calls (IPv4)
+ TEST_MUSTPASS(netw->GetLocalIP(localIP));
+ MARK();
+ TEST_LOG("[local IPv4: %s]\n", localIP);
+ TEST_MUSTPASS(netw->GetLocalIP(localIP));
+ MARK();
+
+#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
+ // default function calls (IPv6)
+ TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
+ MARK();
+ TEST_LOG("[local IPv6: %s]\n", localIP);
+ TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
+ MARK();
+#endif
+
+ // one last call to ensure that local
+ TEST_MUSTPASS(netw->GetLocalIP(localIP));
+ MARK();
+#endif
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of GetLocalIP
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> GetSourceInfo
+ //
+ // - VE initialized
+ // - no existing channels
+ TEST(GetSourceInfo);
+ ANL();
+
+ int rtpPort(0);
+ int rtcpPort(0);
+ char ipaddr[64] = { 0 };
+ ExtendedTestTransport* ptrTransport(NULL);
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // NULL as input string
+ TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, NULL));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+
+ // call when external transport is enabled
+ ptrTransport = new ExtendedTestTransport(netw);
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ delete ptrTransport;
+
+ // call when external transport is disabled (no packet received yet)
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(rtcpPort != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+ // send and receive packets with default settings for a while
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime2); // does not guarantee RTCP
+
+ // verify remote parameters (exclude RTCP)
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 8000);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // ensure that valid results are maintained after StopListen
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 8000);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // verify that results are maintained after new call to SetLocalReceiver
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 8000);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // STATE: not listening, not sending
+ // send and receive packets with other settings for a while
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ // STATE: listening, sending
+
+ // verify new remote parameters
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 9005);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // restart sending to and from local IP
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ // verify new remote parameters
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 9005);
+ TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0); // should not be "127.0.0.1"
+
+ // use non-default source port in outgoing packets
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9010));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ // verify new remote parameters
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 9010);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // STATE: listening and sending using an extra local socket
+
+ // stop/start sending
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ // verify that the unique source port is maintained for the extra socket
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 9010);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // set new source port for outgoing packets (9010 -> 9020)
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9020));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+#ifdef MAC_IPHONE
+ SLEEP(500); // Need extra pause for some reason
+#endif
+
+ // verify that the unique source port is set for the new extra socket
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 9020);
+ // STATE: listening and sending using an extra local socket
+
+ // remove extra send socket and restart call again
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // delete channel => destroys the
+ // extra socket
+ TEST_MUSTPASS(voe_base_->CreateChannel()); // new channel uses one socket only
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000)); // use new port as well
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ // verify that remote info is correct
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 8000);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // STATE: listening and sending using shared socket
+
+ // use non-default source port in outgoing packets to create extra send
+ // socket
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 7000));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 7000, "127.0.0.1", 7010));
+ // RTP src is 7010 => RTCP src = 7011
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ // verify new remote parameters
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 7010);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+
+ // check RTCP port as well (should be 7010 + 1 = 7011)
+ Sleep(8000, true);
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 7010);
+ TEST_MUSTPASS(rtcpPort != 7011);
+ TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of GetSourceInfo
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetExternalTransport
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ //
+ TEST(SetExternalTransport);
+ ANL();
+
+ ptrTransport = new ExtendedTestTransport(netw);
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->DeRegisterExternalTransport(0));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // different valid call combinations
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK(); // must deregister first
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+
+ // STATE: external transport is disabled
+
+ // initialize sending and ensure that external transport can't be enabled
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 1234, "127.0.0.2"));
+ TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_ERROR(VE_SEND_SOCKETS_CONFLICT);
+
+ // restart channel to ensure that "initialized sender" state is cleared
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // initialize receiving and ensure that external transport can't be enabled
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 5678));
+ TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_ERROR(VE_RECEIVE_SOCKETS_CONFLICT);
+
+ // restart channel to ensure that "initialized receiver" state is cleared
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // enable external transport and verify that "emulated loopback" works
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartSend(0)); // should only start recording
+ TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK(); // should fail
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ Play(0, 2000, true, true); // play file as mic and verify loopback audio
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+
+ // STATE: external transport is disabled
+#if defined(WEBRTC_ANDROID) || defined(MAC_IPHONE)
+ int testError = VE_FUNC_NOT_SUPPORTED;
+#else
+ int testError = VE_EXTERNAL_TRANSPORT_ENABLED;
+#endif
+
+ // check all APIs that should fail when external transport is enabled
+ int DSCP, priority, serviceType, overrideDSCP, nBytes(0);
+ bool useSetSockopt, enabled;
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ MARK();
+ TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 12345));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!voe_base_->GetLocalReceiver(0, rtpPort, rtcpPort, ipaddr));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!voe_base_->GetSendDestination(0, rtpPort, ipaddr, rtpPort,
+ rtcpPort));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!netw->EnableIPv6(0))
+ TEST_ERROR(testError);
+ TEST_MUSTPASS(netw->IPv6IsEnabled(0) != false)
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+
+ // modified i VoE 3.4 (can be called also for external transport)
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+#if (!defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)) || \
+ defined(WEBRTC_EXTERNAL_TRANSPORT)
+ testError = VE_FUNC_NOT_SUPPORTED;
+#else
+ testError = VE_EXTERNAL_TRANSPORT_ENABLED;
+#endif
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 0));
+ TEST_ERROR(testError);
+ TEST_MUSTPASS(!netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_ERROR(testError);
+#if !defined(_WIN32) || defined(WEBRTC_EXTERNAL_TRANSPORT)
+ testError = VE_FUNC_NOT_SUPPORTED;
+#else
+ testError = VE_EXTERNAL_TRANSPORT_ENABLED;
+#endif
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0));
+ TEST_ERROR(testError);
+ TEST_MUSTPASS(!netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_ERROR(testError);
+ char dummy[1] = { 'a' };
+ TEST_MUSTPASS(!netw->SendUDPPacket(0, dummy, 1, nBytes));
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+
+ // always disable external transport before deleting the Transport object;
+ // will lead to crash for RTCP transmission otherwise
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ MARK();
+ delete ptrTransport;
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetExternalTransport
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> EnableIPv6
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ // - NOTE: set _ENABLE_IPV6_TESTS_ to include these tests
+ // - http://www.microsoft.com/resources/documentation/windows/xp/all/
+ // proddocs/en-us/sag_ip_v6_pro_rt_enable.mspx?mfr=true
+ // >> ipv6 install
+ // >> ipv6 [-v] if [IfIndex]
+ // >> ping6 ::1
+ // >> ping6 fe80::1
+
+#ifdef _ENABLE_IPV6_TESTS_
+
+ TEST(EnableIPv6); ANL();
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // call with enabled external transport
+ ptrTransport = new ExtendedTestTransport(netw);
+ TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
+ TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
+ TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ delete ptrTransport;
+
+ // Test "locking" to IPv4
+ TEST_MUSTPASS(netw->IPv6IsEnabled(0)); MARK(); // After this call we cannot
+ // enable IPv6
+ TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
+
+ // Check that IPv6 address is invalid
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "::1")); MARK(); // fail
+
+ // New channel
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // valid default call
+ TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
+ TEST_MUSTPASS(netw->GetLocalIP(localIP)); MARK(); // should still read IPv4
+ TEST_LOG("[local IPv4: %s]", localIP);
+
+ // ensure that Ipv6 is enabled
+ TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
+
+ // check that IPv4 address is invalid
+ TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_ERROR(VE_INVALID_IP_ADDRESS);
+
+ // verify usage of IPv6 loopback address
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ // IPv6 loopback address is 0:0:0:0:0:0:0:1
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true);
+ ANL();
+
+ // Restart channel
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
+ // ensure that Ipv6 is enabled
+ TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ // IPv6 loopback address is 0:0:0:0:0:0:0:1
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(), true,
+ true);
+ SLEEP(500); // ensure that we receieve some packets
+
+ // SetSourceFilter and GetSourceFilter
+ TEST(SetSourceFilter and GetSourceFilter for IPv6); ANL();
+ char sourceIp[64] =
+ { 0};
+ char filterIp[64] =
+ { 0};
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, sourceIp));
+ TEST_LOG("Source port: %d \n", rtpPort);
+ TEST_LOG("Source RTCP port: %d \n", rtcpPort);
+ TEST_LOG("Source IP: %s \n", sourceIp);
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_LOG("Filter port RTP: %d \n", rtpPort);
+ TEST_LOG("Filter port RTCP: %d \n", rtcpPort);
+ TEST_LOG("Filter IP: %s \n", filterIp);
+ TEST_MUSTPASS(0 != rtpPort);
+ TEST_MUSTPASS(0 != rtcpPort);
+ TEST_MUSTPASS(filterIp[0] != '\0');
+ TEST_LOG("Set filter IP to %s => should hear audio\n", sourceIp);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, sourceIp));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_MUSTPASS(0 != rtpPort);
+ TEST_MUSTPASS(0 != rtcpPort);
+ TEST_MUSTPASS(_stricmp(filterIp, sourceIp));
+ SLEEP(1500);
+ TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_MUSTPASS(_stricmp(filterIp, "::10:10:10"));
+ SLEEP(1500);
+ TEST_LOG("Disable IP filter => should hear audio again\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::0"));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_MUSTPASS(_stricmp(filterIp, "::"));
+ SLEEP(1500);
+ TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
+ SLEEP(1500);
+ TEST_LOG("Disable IP filter => should hear audio again\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_MUSTPASS(filterIp[0] != '\0');
+ SLEEP(1500);
+ TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
+ SLEEP(1500);
+ TEST_LOG("Disable IP filter => should hear audio again\n");
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::"));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
+ TEST_MUSTPASS(_stricmp(filterIp, "::"));
+ SLEEP(1500);
+
+ file->StopPlayingFileAsMicrophone(0);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+#endif // #ifdef _ENABLE_IPV6_TESTS_
+ // >> end of EnableIPv6
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetSourceFilter
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ TEST(SetSourceFilter);
+ ANL();
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // invalid parameters
+ TEST_MUSTPASS(!netw->SetSourceFilter(0, 65536));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR);
+ TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 65536));
+ MARK();
+ TEST_ERROR(VE_INVALID_PORT_NMBR);
+ TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346, "300.300.300.300"));
+ MARK();
+ TEST_ERROR(VE_INVALID_IP_ADDRESS);
+
+ // STATE: RTP filter port is 12345, RTCP filter port is 12346
+
+ // disable all filters and ensure that media is received
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 2000);
+ TEST_MUSTPASS(rtcpPort != 2001);
+ TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
+
+ // clear states and restart loopback session
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // set RTP filter to port 2002 and verify that source 2000 is blocked
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 2002, 0, NULL));;
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ // ensure that received packets originates from 2002 and that they now pass
+ // the filter
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ // RTP source is 2002
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2002, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2002, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 2002);
+ TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
+
+ // clear states and restart loopback session
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // set IP filter to local IP and verify that default loopback stream is
+ // blocked
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, localIP));;
+ MARK();
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ // ensure that received packets originates from the local IP and that they
+ // now pass the filter
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ // should pass the filter
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 2000);
+ TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
+
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+
+ // STATE: no active media, IP filter is active
+
+ // disable all filters
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));;
+ MARK();
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(rtcpPort != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetSourceFilter
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> GetSourceFilter
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ TEST(GetSourceFilter);
+ ANL();
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // invalid input parameters
+ TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, NULL));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+
+ // valid call without any filter set
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(rtcpPort != 0);
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ // STATE: no active media and no enabled filters
+
+ // set different filters and verify that they "bite"
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 54321, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 54321);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtpPort != 0);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 15425, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtcpPort != 15425);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(rtcpPort != 0);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "192.168.199.19"));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(strcmp(ipaddr, "192.168.199.19") != 0);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "0.0.0.0"));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
+ TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
+ TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
+ MARK();
+ TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of GetSourceFilter
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> RegisterDeadOrAliveObserver
+ // >> DeRegisterDeadOrAliveObserver
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ TEST(RegisterDeadOrAliveObserver);
+ ANL();
+ TEST(DeRegisterDeadOrAliveObserver);
+ ANL();
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ MARK();
+ TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
+ MARK(); // already registered
+ TEST_ERROR(VE_INVALID_OPERATION);
+ TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ MARK();
+ TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ MARK(); // OK to do it again
+ TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ MARK();
+ TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ MARK();
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+ // STATE: dead-or-alive observer is disabled
+
+ // >> end of RegisterDeadOrAliveObserver
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetPeriodicDeadOrAliveStatus
+ // >> GetPeriodicDeadOrAliveStatus
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+
+ // call without valid channel
+ TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, false));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // Invalid paramters
+ TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 0));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 151));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(1, true, 10));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ int sampleTime(0);
+
+ // Valid parameters
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ MARK();
+ TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(sampleTime != 1);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 150));
+ MARK();
+ TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(sampleTime != 150);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ MARK();
+ TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(sampleTime != 150); // ensure last set time isnt modified
+
+ StartMedia(0, 2000, true, true, true);
+
+ // STATE: full duplex media is active
+
+ // test the dead-or-alive mechanism
+ TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ MARK();
+ TEST_LOG("\nVerify that Alive callbacks are received (dT=2sec): ");
+ fflush(NULL);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
+ SLEEP(6000);
+ TEST_LOG("\nChange dT to 1 second: ");
+ fflush(NULL);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ SLEEP(6000);
+ TEST_LOG("\nDisable dead-or-alive callbacks: ");
+ fflush(NULL);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ SLEEP(6000);
+ TEST_LOG("\nStop sending and enable callbacks again.\n");
+ TEST_LOG("Verify that Dead callbacks are received (dT=2sec): ");
+ fflush(NULL);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
+ SLEEP(6000);
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_LOG("\nRestart sending.\n");
+ TEST_LOG("Verify that Alive callbacks are received again (dT=2sec): ");
+ fflush(NULL);
+ SLEEP(6000);
+ TEST_LOG("\nDisable dead-or-alive callbacks.");
+ fflush(NULL);
+ TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ MARK();
+
+ StopMedia(0);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetPeriodicDeadOrAliveStatus
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetPacketTimeoutNotification
+ // >> GetPacketTimeoutNotification
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ // - NOTE: dynamic tests are performed in standard test
+
+ int timeOut(0);
+
+ TEST(SetPacketTimeoutNotification);
+ ANL();
+ TEST(GetPacketTimeoutNotification);
+ ANL();
+
+ // call without existing valid channel
+ TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, false));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // invalid function calls
+ TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 0));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 151));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+
+ // valid function calls (no active media)
+ TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
+ MARK();
+ TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(timeOut != 2);
+ TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
+ MARK();
+ TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ MARK();
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 10));
+ MARK();
+ TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(timeOut != 10);
+ TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
+ MARK();
+ TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(timeOut != 2);
+ TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
+ MARK();
+ TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ MARK();
+ TEST_MUSTPASS(enabled != false);
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL();
+ AOK();
+ ANL();
+ ANL();
+
+ // >> end of SetPacketTimeoutNotification
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SendUDPPacket
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+
+
+ // >> end of SendUDPPacket
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetSendTOS
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ TEST(SetSendTOS);
+ ANL();
+#if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+
+ // call without existing valid channel
+
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 0)); MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // trivial invalid function calls
+ TEST_MUSTPASS(!netw->SetSendTOS(0, -1)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 64)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -2)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 1, 8)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 1)); MARK();
+ TEST_ERROR(VE_SOCKET_ERROR); // must create sockets first
+
+#ifdef _WIN32
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 3000));
+
+ // enable ToS using SetSockopt (should work without local binding)
+ TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, true)); MARK();
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
+ TEST_MUSTPASS(DSCP != 1);
+ TEST_MUSTPASS(priority != 0);
+ TEST_MUSTPASS(useSetSockopt != true);
+
+ // try to disable SetSockopt while ToS is enabled (should fail)
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
+ TEST_ERROR(VE_TOS_INVALID); // must disable ToS before enabling SetSockopt
+
+ // disable ToS to be able to stop using SetSockopt
+ TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true)); MARK(); // disable ToS
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
+ TEST_MUSTPASS(DSCP != 0);
+ TEST_MUSTPASS(priority != 0);
+ TEST_MUSTPASS(useSetSockopt != true);
+
+ // to use the "non-SetSockopt" method, local binding is required,
+ // trying without it should fail
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
+ TEST_ERROR(VE_TOS_ERROR); // must bind to local IP first
+
+ // bind to local IP and try again (should work this time)
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
+ TEST_LOG("\nThis test needs to be run as administrator\n");
+ TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, false)); MARK();
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
+ TEST_MUSTPASS(DSCP != 1);
+ TEST_MUSTPASS(priority != 0);
+ TEST_MUSTPASS(useSetSockopt != false);
+
+ // STATE: binded to local IP, local port is 12345 and DSCP is 1 (not using
+ // SetSockopt)
+
+ // verify loopback audio with the current settings
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+
+#ifdef _SEND_TO_REMOTE_IP_
+ // Send to remote destination and verify the DSCP using Wireshark.
+ // Use filter ip.src == "RemoteIP".
+ TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the "
+ "remote side!\n");
+ TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
+ RemoteIP, RemotePort);
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
+ TEST_LOG(" DSCP is set to 0x%02x\n", 1);
+ SLEEP(100);
+
+ // Change the DSCP while sending is active and verify on remote side.
+ TEST_MUSTPASS(netw->SetSendTOS(0, 2));
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
+ SLEEP(100);
+
+ // Change the DSCP while sending is active and verify on remote side.
+ TEST_MUSTPASS(netw->SetSendTOS(0, 63));
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
+ SLEEP(100);
+
+ // stop and resume sending
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 0));
+#endif // _SEND_TO_REMOTE_IP_
+ // Windows priority tests (priority cannot be set using setsockopt on Win)
+ TEST_LOG("Testing priority\n");
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, true)); // Should fail
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, false));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, false));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+#endif // _WIN32
+ // STATE: no media, disabled ToS, no defined receiver
+
+ // Repeat tests above but using setsockopt() this time.
+ // Binding to local IP should not be required.
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 10, -1, true)); MARK();
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
+ TEST_MUSTPASS(DSCP != 10);
+ TEST_MUSTPASS(priority != 0);
+ TEST_MUSTPASS(useSetSockopt != true);
+
+ // STATE: *not* binded to local IP, local port is 12345 and DSCP is 10
+ // (using SetSockopt)
+
+ // verify loopback audio with the current settings
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+
+#ifdef _SEND_TO_REMOTE_IP_
+ // Send to remote destination and verify the DSCP using Wireshark.
+ // Use filter ip.src == "RemoteIP".
+ TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the"
+ " remote side!\n");
+ TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
+ RemoteIP, RemotePort);
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
+ SLEEP(100);
+
+ // Change the DSCP while sending is active and verify on remote side.
+ TEST_MUSTPASS(netw->SetSendTOS(0, 20, -1, true)); // use setsockopt()
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
+ SLEEP(100);
+
+ // Change the DSCP while sending is active and verify on remote side.
+ TEST_MUSTPASS(netw->SetSendTOS(0, 61, -1, true)); // use setsockopt()
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
+ SLEEP(100);
+
+ // stop and resume sending
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
+ TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true));
+#endif // _SEND_TO_REMOTE_IP_
+#if defined(WEBRTC_LINUX)
+ // Linux priority tests (using setsockopt)
+ TEST_LOG("Testing priority\n");
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ Play(0, 2000, true, true); // file should be played out here...
+#endif // #if defined(WEBRTC_LINUX)
+#if !defined(_WIN32) && !defined(WEBRTC_LINUX)
+ // Fail tests for other than Wind and Linux
+ TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, false)); // Should fail
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+#endif // #if !defined(_WIN32) && !defined(WEBRTC_LINUX)
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL(); AOK(); ANL(); ANL();
+
+ // END #if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#else
+ TEST_LOG("Skipping ToS tests - _WIN32, LINUX, MAC is not defined or "
+ "WEBRTC_ANDROID is defined");
+#endif
+
+ // >> end of SetSendTOS
+ // ------------------------------------------------------------------------
+
+ // ------------------------------------------------------------------------
+ // >> SetSendGQoS (Windows only)
+ //
+ // - VE initialized
+ // - no existing channels
+ // - no media
+ //
+ // From qos.h:
+ //
+ // #define SERVICETYPE_NOTRAFFIC 0x00000000
+ // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
+ // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
+ // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
+ // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
+ // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
+ // #define SERVICETYPE_NOCHANGE 0x00000006
+ // #define SERVICETYPE_NONCONFORMING 0x00000009
+ // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
+ // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
+ //
+ // #define SERVICE_BESTEFFORT 0x80010000
+ // #define SERVICE_CONTROLLEDLOAD 0x80020000
+ // #define SERVICE_GUARANTEED 0x80040000
+ // #define SERVICE_QUALITATIVE 0x80200000
+ //
+ // (*) supported in WEBRTC VoE
+ TEST(SetSendGQoS);
+ ANL();
+#ifdef _WIN32
+
+ // call without existing valid channel
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0)); MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ // supported service type but no sockets
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
+ TEST_ERROR(VE_SOCKETS_NOT_INITED);
+
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+
+ // supported service type but sender is not initialized
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
+ TEST_ERROR(VE_DESTINATION_NOT_INITED);
+
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+
+ // invalid service types
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOTRAFFIC)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_UNAVAILABLE));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_GENERAL_INFORMATION));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOCHANGE)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NONCONFORMING));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_CONTROL));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_BESTEFFORT)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_CONTROLLEDLOAD)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_GUARANTEED)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_QUALITATIVE)); MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+
+ // Is ToS enabled here?
+
+ // Settings which don't require binding to local IP
+
+ // set SERVICETYPE_BESTEFFORT
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(serviceType != SERVICETYPE_BESTEFFORT);
+ TEST_MUSTPASS(overrideDSCP != false);
+
+ // set SERVICETYPE_CONTROLLEDLOAD
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
+ MARK();
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(serviceType != SERVICETYPE_CONTROLLEDLOAD);
+ TEST_MUSTPASS(overrideDSCP != false);
+
+ // set SERVICETYPE_GUARANTEED
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(serviceType != SERVICETYPE_GUARANTEED);
+ TEST_MUSTPASS(overrideDSCP != false);
+
+ // set SERVICETYPE_QUALITATIVE
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
+ TEST_MUSTPASS(overrideDSCP != false);
+
+ // disable GQoS
+ TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0)); MARK();
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ MARK();
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
+ TEST_MUSTPASS(overrideDSCP != false);
+
+ // STATE: diabled QGoS, sockets exists, sending side is initialized, no media
+
+ // Loopback tests using the four different GQoS settings
+
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ ANL();
+ TEST_LOG("[SERVICETYPE_BESTEFFORT]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD)); MARK();
+ ANL();
+ TEST_LOG("[SERVICETYPE_CONTROLLEDLOAD]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
+ ANL();
+ TEST_LOG("[SERVICETYPE_GUARANTEED]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
+ ANL();
+ TEST_LOG("[SERVICETYPE_QUALITATIVE]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+#ifdef _SEND_TO_REMOTE_IP_
+ // Send to remote destination and verify the DSCP mapping using Wireshark.
+ // Use filter ip.src == "RemoteIP".
+
+ // Modify the send destination on the fly
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
+
+ TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
+ " the remote side!\n");
+ TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
+ RemoteIP, RemotePort);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT (0x%02x), should "
+ "be mapped to DSCP = 0x00\n", serviceType);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_CONTROLLEDLOAD (0x%02x), "
+ "should be mapped to DSCP = 0x18\n", serviceType);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
+ TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_GUARANTEED (0x%02x), should "
+ "be mapped to DSCP = 0x28\n", serviceType);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_QUALITATIVE (0x%02x), should"
+ " be mapped to DSCP = 0x00\n", serviceType);
+ SLEEP(100);
+#endif // _SEND_TO_REMOTE_IP_
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+
+ // STATE: sockets exists, sending side is initialized, no media
+
+ // Repeat test above but this time using overrideDSCP.
+
+ // Some initial loopack tests.
+ // NOTE - override DSCP requres binding to local IP.
+
+ // should not work since QoS is enabled
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
+ MARK();
+ TEST_ERROR(VE_TOS_GQOS_CONFLICT);
+
+ // disble QoS and try to override again (should fail again since local
+ // binding is not done yet)
+ TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
+ TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
+ MARK();
+ TEST_ERROR(VE_GQOS_ERROR);
+
+ // make proper settings and try again (should work this time)
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
+ MARK();
+
+ // Now, let's try some loopback tests using override DSCP
+
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ ANL();
+ TEST_LOG("[overrideDSCP=3]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 17));
+ MARK();
+ ANL();
+ TEST_LOG("[overrideDSCP=17]");
+ Play(0, 2000, true, true); // file should be played out here...
+
+ // And finally, send to remote side as well to verify that the new mapping
+ // works as it should.
+
+#ifdef _SEND_TO_REMOTE_IP_
+ // Modify the send destination on the fly
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
+
+ TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
+ " the remote side!\n");
+ TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
+ RemoteIP, RemotePort);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 18));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
+ "overrided to DSCP = 0x%02x\n", overrideDSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 62));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
+ "overrided to DSCP = 0x%02x\n", overrideDSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 32));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
+ "overrided to DSCP = 0x%02x\n", overrideDSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 1));
+ TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
+ TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
+ "overrided to DSCP = 0x%02x\n", overrideDSCP);
+ SLEEP(100);
+ TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
+ TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
+ SLEEP(100);
+#endif // _SEND_TO_REMOTE_IP_
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ ANL(); AOK(); ANL(); ANL();
+
+#else
+ TEST_LOG("Skipping GQoS tests - _WIN32 is not defined");
+#endif // #ifdef _WIN32
+ // >> end of SetSendGQoS
+ // ------------------------------------------------------------------------
+
+ if (file) {
+ file->StopPlayingFileAsMicrophone(0);
+ }
+ voe_base_->StopSend(0);
+ voe_base_->StopPlayout(0);
+ voe_base_->StopReceive(0);
+ voe_base_->DeleteChannel(0);
+ voe_base_->Terminate();
+
+ ANL();
+ AOK();
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestRTP_RTCP
+// ----------------------------------------------------------------------------
+
+// Used to validate packets during the RTP audio level indication test.
+class RTPAudioTransport: public Transport {
+public:
+
+ RTPAudioTransport() :
+ mute_(false) {
+ }
+
+ virtual ~RTPAudioTransport() {
+ }
+
+ void set_mute(bool mute) {
+ mute_ = mute;
+ }
+ bool mute() const {
+ return mute_;
+ }
+
+ // TODO(andrew): use proper error checks here rather than asserts.
+ virtual int SendPacket(int channel, const void* data, int length) {
+ const uint8_t* packet = static_cast<const uint8_t*> (data);
+
+ // Extension bit.
+ assert(packet[0] & 0x10);
+ int index = 12; // Assume standard RTP header.
+ // Header extension ID
+ assert(packet[index++] == 0xBE);
+ assert(packet[index++] == 0xDE);
+ // Header extension length
+ assert(packet[index++] == 0x00);
+ assert(packet[index++] == 0x01);
+
+ // User-defined ID.
+ assert(((packet[index] & 0xf0) >> 4) == 1);
+ // Length
+ assert((packet[index++] & 0x0f) == 0);
+
+ int vad = packet[index] >> 7;
+ int level = packet[index] & 0x7f;
+ if (channel == 0) {
+ printf("%d -%d\n", vad, level);
+ } else if (channel == 1) {
+ printf(" %d -%d\n", vad, level);
+ } else {
+ assert(false);
+ }
+
+ if (mute_) {
+ assert(vad == 0);
+ assert(level == 127);
+ } else {
+ assert(vad == 0 || vad == 1);
+ assert(level >= 0 && level <= 127);
+ }
+
+ return 0;
+ }
+
+ virtual int SendRTCPPacket(int /*channel*/, const void* /*data*/,
+ int /*length*/) {
+ return 0;
+ }
+
+private:
+ bool mute_;
+};
+
+int VoEExtendedTest::TestRTP_RTCP() {
+ PrepareTest("RTP_RTCP");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoERTP_RTCP* rtp_rtcp = _mgr.RTP_RTCPPtr();
+ VoENetwork* network = _mgr.NetworkPtr();
+ VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+ VoECodec* codec = _mgr.CodecPtr();
+
+ XRTPObserver rtpObserver;
+
+#ifdef WEBRTC_ANDROID
+ int sleepTime = 200;
+#else
+ int sleepTime = 100;
+#endif
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
+ "VoERTP_RTCP_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ // ------------------------------------------------------------------------
+ // >> Set/GetRTPAudioLevelIndicationStatus
+ TEST(SetRTPAudioLevelIndicationStatus);
+ ANL();
+ TEST(GetRTPAudioLevelIndicationStatus);
+
+ // test invalid input parameters
+ TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, 0));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, 15));
+ MARK();
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, false, 15));
+ MARK();
+ TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(1, true, 5));
+ MARK();
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+
+ // test complete valid input range [1,14]
+ bool audioLevelEnabled(false);
+ unsigned char ID(0);
+ for (int id = 1; id < 15; id++) {
+ TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, id));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTPAudioLevelIndicationStatus(
+ 0, audioLevelEnabled, ID));
+ MARK();
+ TEST_MUSTPASS(audioLevelEnabled != true);
+ TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, false, id));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTPAudioLevelIndicationStatus(
+ 0, audioLevelEnabled, ID));
+ MARK();
+ TEST_MUSTPASS(audioLevelEnabled != false);
+ TEST_MUSTPASS(ID != id);
+ }
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+ RTPAudioTransport rtpAudioTransport;
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(network->RegisterExternalTransport(0, rtpAudioTransport));
+ TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true));
+ TEST_MUSTPASS(codec->SetVADStatus(0, true));
+
+ printf("\n\nReceving muted packets (expect VAD = 0, Level = -127)...\n");
+ printf("VAD Level [dbFS]\n");
+ SLEEP(2000);
+ rtpAudioTransport.set_mute(true);
+ TEST_MUSTPASS(volume->SetInputMute(0, true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(5000);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ rtpAudioTransport.set_mute(false);
+ TEST_MUSTPASS(volume->SetInputMute(0, false));
+
+ printf("\nReceiving packets from mic (should respond to mic level)...\n");
+ printf("VAD Level [dbFS]\n");
+ SLEEP(2000);
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(5000);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+
+ printf("\nReceiving packets from file (expect mostly VAD = 1)...\n");
+ printf("VAD Level [dbFS]\n");
+ SLEEP(2000);
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(5000);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+
+ printf("\nMuted and mic on independent channels...\n");
+ printf("Muted Mic\n");
+ SLEEP(2000);
+ ASSERT_TRUE(1 == voe_base_->CreateChannel());
+ TEST_MUSTPASS(network->RegisterExternalTransport(1, rtpAudioTransport));
+ TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(1, true));
+ TEST_MUSTPASS(codec->SetVADStatus(1, true));
+ TEST_MUSTPASS(volume->SetInputMute(0, true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartSend(1));
+ SLEEP(5000);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopSend(1));
+
+ TEST_MUSTPASS(network->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(network->DeRegisterExternalTransport(1));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(1));
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ MARK();
+ ANL();
+
+ // ------------------------------------------------------------------------
+ // >> InsertExtraRTPPacket
+
+ int i(0);
+
+ TEST(SetLocalSSRC);
+ TEST_MUSTPASS(!rtp_rtcp->SetLocalSSRC(0, 5678));
+ MARK();
+ TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(rtp_rtcp->SetLocalSSRC(0, 5678)); // force send SSRC to 5678
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ MARK();
+ ANL();
+
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+
+ // ------------------------------------------------------------------------
+ // >> InsertExtraRTPPacket
+ TEST(InsertExtraRTPPacket);
+ ANL();
+
+ const char payloadData[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
+
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(-1, 0, false,
+ payloadData, 8));
+ MARK(); // invalid channel
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, -1, false,
+ payloadData, 8));
+ MARK(); // invalid payload type
+ TEST_ERROR(VE_INVALID_PLTYPE);
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 128, false,
+ payloadData, 8));
+ MARK(); // invalid payload type
+ TEST_ERROR(VE_INVALID_PLTYPE);
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
+ NULL, 8));
+ MARK(); // invalid pointer
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
+ payloadData, 1500-28+1));
+ MARK(); // invalid size
+ TEST_ERROR(VE_INVALID_ARGUMENT);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
+ payloadData, 8));
+ MARK(); // not sending
+ TEST_ERROR(VE_NOT_SENDING);
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+
+ SLEEP(1000);
+ for (int p = 0; p < 128; p++) {
+ TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, false,
+ payloadData, 8));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, true,
+ payloadData, 8));
+ MARK();
+ }
+
+ // Ensure we have sent all extra packets before we move forward to avoid
+ //incorrect error code
+ SLEEP(1000);
+
+ ANL();
+
+ // ------------------------------------------------------------------------
+ // >> RTP dump APIs
+ TEST(Start/StopRtpDump);
+ ANL();
+ TEST(Start/RTPDumpIsActive);
+
+ TEST_MUSTPASS(-1 != rtp_rtcp->RTPDumpIsActive(-1, kRtpIncoming));
+ MARK(); // invalid channel
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(false != rtp_rtcp->RTPDumpIsActive(0, kRtpIncoming));
+ MARK(); // should be off by default
+ TEST_MUSTPASS(false != rtp_rtcp->RTPDumpIsActive(0, kRtpOutgoing));
+ MARK(); // should be off by default
+
+ TEST_MUSTPASS(-1 != rtp_rtcp->StartRTPDump(-1, NULL));
+ MARK(); // invalid channel
+ TEST_ERROR(VE_CHANNEL_NOT_VALID);
+ TEST_MUSTPASS(-1 != rtp_rtcp->StartRTPDump(0, NULL));
+ MARK(); // invalid file name
+ TEST_ERROR(VE_BAD_FILE);
+
+ // Create two RTP dump files:
+
+ // - dump_in_1sec.rtp <=> ~1 sec recording of input side
+ // - dump_in_2sec.rtp <=> ~2 sec recording of output side
+ //
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpIncoming));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpOutgoing));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0, GetFilename("dump_in_1sec.rtp"),
+ kRtpIncoming));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0, GetFilename("dump_out_2sec.rtp"),
+ kRtpOutgoing));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpIncoming));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpOutgoing));
+ MARK();
+
+ // Start/Stop tests:
+ //
+ // - only one file (called dump_in_200ms.rtp) should exist after this test
+ //
+ for (i = 0; i < 10; i++) {
+ TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0,
+ GetFilename("dump_in_200ms.rtp")));
+ MARK();
+ SLEEP(200);
+ TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0));
+ MARK();
+ }
+
+ // >> end of RTP dump APIs
+ // ------------------------------------------------------------------------
+ ANL();
+
+ TEST(GetRTCPStatus);
+ bool enabled;
+ TEST_MUSTPASS(!rtp_rtcp->GetRTCPStatus(-1, enabled));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
+ MARK(); // should be on by default
+ TEST_MUSTPASS(enabled != true);
+ ANL();
+
+ TEST(SetRTCPStatus);
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
+ TEST_MUSTPASS(enabled != false);
+ MARK();
+ SLEEP(2000);
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
+ TEST_MUSTPASS(enabled != true);
+ MARK();
+ SLEEP(6000); // Make sure we get an RTCP packet
+ ANL();
+
+ TEST(CNAME);
+ TEST_MUSTPASS(!rtp_rtcp->SetRTCP_CNAME(0, NULL));
+ MARK();
+ TEST_MUSTPASS(VE_RTP_RTCP_MODULE_ERROR != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!rtp_rtcp->GetRemoteRTCP_CNAME(0, NULL));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ ANL();
+
+ TEST(GetRemoteSSRC);
+ unsigned int ssrc(0);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteSSRC(0, ssrc));
+ MARK();
+ TEST_MUSTPASS(ssrc != 5678);
+ ANL();
+
+ TEST(GetRemoteCSRC); // only trivial tests added
+ unsigned int csrcs[2];
+ int n(0);
+ TEST_MUSTPASS(!rtp_rtcp->GetRemoteCSRCs(1, csrcs));
+ MARK();
+ n = rtp_rtcp->GetRemoteCSRCs(0, csrcs);
+ MARK();
+ TEST_MUSTPASS(n != 0); // should be empty
+ ANL();
+
+ TEST(SetRTPObserver);
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(rtp_rtcp->RegisterRTPObserver(0, rtpObserver));
+ TEST_MUSTPASS(rtp_rtcp->DeRegisterRTPObserver(0));
+ TEST_MUSTPASS(rtp_rtcp->RegisterRTPObserver(0, rtpObserver));
+ TEST_MUSTPASS(rtp_rtcp->SetLocalSSRC(0, 7777)); // force send SSRC to 7777
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(sleepTime);
+ // verify that the new SSRC has been detected by the observer
+ TEST_MUSTPASS(rtpObserver._SSRC != 7777);
+ TEST_MUSTPASS(rtp_rtcp->DeRegisterRTPObserver(0));
+ ANL();
+
+ TEST(GetRTPKeepaliveStatus);
+ unsigned char pt;
+ int dT;
+ TEST_MUSTPASS(!rtp_rtcp->GetRTPKeepaliveStatus(-1, enabled, pt, dT));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
+ MARK(); // should be off by default
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(pt != 255);
+ TEST_MUSTPASS(dT != 0);
+ ANL();
+
+ TEST(SetRTPKeepaliveStatus);
+ // stop send before changing the settings
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ // verify invalid input parameters
+ TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(-1, true, 0, 15));
+ MARK();
+ TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(0, true, -1, 15));
+ MARK();
+ TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(0, true, 0, 61));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
+ MARK(); // should still be off
+ TEST_MUSTPASS(enabled != false);
+ // try valid settings
+ TEST_MUSTPASS(rtp_rtcp->SetRTPKeepaliveStatus(0, true, 117));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
+ MARK(); // should be on now
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(pt != 117);
+ TEST_MUSTPASS(dT != 15);
+ // change from PT 99 to 121, as 99 is occupied
+ TEST_MUSTPASS(rtp_rtcp->SetRTPKeepaliveStatus(0, true, 121, 3));
+ MARK(); // on, PT=99, dT=3
+ TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(pt != 121);
+ TEST_MUSTPASS(dT != 3);
+ ANL();
+
+ // Make fresh restart (ensures that SSRC is randomized)
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+
+ SLEEP(100);
+
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+
+ SLEEP(8000);
+
+ TEST(GetRemoteRTCPData);
+ // Statistics based on received RTCP reports (i.e. statistics on the remote
+ // side sent to us).
+ unsigned int NTPHigh(0), NTPLow(0), timestamp(0), playoutTimestamp(0),
+ jitter(0);
+ unsigned short fractionLost(0);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh, NTPLow,
+ timestamp, playoutTimestamp));
+ TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
+ " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
+ NTPHigh, NTPLow, timestamp, playoutTimestamp, jitter, fractionLost);
+
+ unsigned int NTPHigh2(0), NTPLow2(0), timestamp2(0);
+ unsigned int playoutTimestamp2(0), jitter2(0);
+ unsigned short fractionLost2(0);
+
+ TEST_LOG("take a new sample and ensure that the playout timestamp is "
+ "maintained");
+ SLEEP(100);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2, timestamp2,
+ playoutTimestamp2, &jitter2,
+ &fractionLost2));
+ TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
+ " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
+ NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
+ TEST_MUSTPASS(playoutTimestamp != playoutTimestamp2);
+
+ TEST_LOG("wait for 8 seconds and ensure that the RTCP statistics is"
+ " updated...");
+ SLEEP(8000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2,
+ timestamp2, playoutTimestamp2,
+ &jitter2, &fractionLost2));
+ TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
+ " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
+ NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
+ TEST_MUSTPASS((NTPHigh == NTPHigh2) && (NTPLow == NTPLow2));
+ TEST_MUSTPASS(timestamp == timestamp2);
+ TEST_MUSTPASS(playoutTimestamp == playoutTimestamp2);
+
+#ifdef WEBRTC_CODEC_RED
+ //The following test is related to defect 4985 and 4986
+ TEST_LOG("Turn FEC and VAD on and wait for 4 seconds and ensure that "
+ "the jitter is still small...");
+ CodecInst cinst;
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ cinst.pltype = 104;
+ strcpy(cinst.plname, "isac");
+ cinst.plfreq = 32000;
+ cinst.pacsize = 960;
+ cinst.channels = 1;
+ cinst.rate = 45000;
+#else
+ cinst.pltype = 119;
+ strcpy(cinst.plname, "isaclc");
+ cinst.plfreq = 16000;
+ cinst.pacsize = 320;
+ cinst.channels = 1;
+ cinst.rate = 40000;
+#endif
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, -1));
+ MARK();
+ TEST_MUSTPASS(codec->SetVADStatus(0,true));
+ SLEEP(4000);
+ TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2, timestamp2,
+ playoutTimestamp2, &jitter2,
+ &fractionLost2));
+ TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
+ " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
+ NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
+ TEST_MUSTPASS(jitter2 > 1000)
+ TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, false));
+ MARK();
+ //4985 and 4986 end
+#endif // #ifdef WEBRTC_CODEC_RED
+ TEST(GetRTPStatistics);
+ ANL();
+ // Statistics summarized on local side based on received RTP packets.
+ CallStatistics stats;
+ // Call GetRTPStatistics over a longer period than 7.5 seconds
+ // (=dT RTCP transmissions).
+ unsigned int averageJitterMs, maxJitterMs, discardedPackets;
+ SLEEP(1000);
+ for (i = 0; i < 8; i++) {
+ TEST_MUSTPASS(rtp_rtcp->GetRTPStatistics(0, averageJitterMs,
+ maxJitterMs,
+ discardedPackets));
+ TEST_LOG(" %i) averageJitterMs = %u \n maxJitterMs = %u \n "
+ " discardedPackets = %u \n", i, averageJitterMs, maxJitterMs,
+ discardedPackets);
+ SLEEP(1000);
+ }
+
+ TEST(RTCPStatistics #1);
+ ANL();
+ unsigned int packetsSent(0);
+ unsigned int packetsReceived(0);
+ for (i = 0; i < 8; i++)
+ {
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
+ TEST_LOG(" %i) fractionLost = %hu \n cumulativeLost = %u \n "
+ " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
+ i, stats.fractionLost, stats.cumulativeLost,
+ stats.extendedMax, stats.jitterSamples, stats.rttMs);
+ TEST_LOG( " bytesSent = %d \n packetsSent = %d \n "
+ " bytesReceived = %d \n packetsReceived = %d \n",
+ stats.bytesSent, stats.packetsSent, stats.bytesReceived,
+ stats.packetsReceived);
+ if (i > 0)
+ {
+ TEST_LOG(" diff sent packets : %u (~50)\n",
+ stats.packetsSent - packetsSent);
+ TEST_LOG(" diff received packets: %u (~50)\n",
+ stats.packetsReceived - packetsReceived);
+ }
+ packetsSent = stats.packetsSent;
+ packetsReceived = stats.packetsReceived;
+ SLEEP(1000);
+ }
+
+ TEST(RTCPStatistics #2);
+ ANL();
+ TEST_LOG("restart sending and ensure that the statistics is reset");
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ SLEEP(50); // ensures approx. two received packets
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
+ TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
+ " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
+ stats.fractionLost, stats.cumulativeLost,
+ stats.extendedMax, stats.jitterSamples, stats.rttMs);
+ TEST_LOG( " bytesSent = %d \n packetsSent = %d \n "
+ " bytesReceived = %d \n packetsReceived = %d \n",
+ stats.bytesSent, stats.packetsSent, stats.bytesReceived,
+ stats.packetsReceived);
+
+ TEST(RTCPStatistics #3);
+ ANL();
+ TEST_LOG("disable RTCP and verify that statistics is not corrupt");
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
+ SLEEP(250);
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
+ TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
+ " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
+ stats.fractionLost, stats.cumulativeLost,
+ stats.extendedMax, stats.jitterSamples, stats.rttMs);
+ TEST_LOG(" bytesSent = %d \n packetsSent = %d \n "
+ "bytesReceived = %d \n packetsReceived = %d \n",
+ stats.bytesSent, stats.packetsSent,
+ stats.bytesReceived, stats.packetsReceived);
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
+
+ TEST(RTCPStatistics #4);
+ ANL();
+ TEST_LOG("restart receiving and check RX statistics");
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ SLEEP(50); // ensures approx. two received packets
+ TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
+ TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
+ " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
+ stats.fractionLost, stats.cumulativeLost,
+ stats.extendedMax, stats.jitterSamples,
+ stats.rttMs);
+ TEST_LOG(" bytesSent = %d \n packetsSent = %d \n "
+ " bytesReceived = %d \n packetsReceived = %d \n",
+ stats.bytesSent, stats.packetsSent,
+ stats.bytesReceived, stats.packetsReceived);
+
+ TEST(SendApplicationDefinedRTCPPacket);
+ // just do some fail tests here
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ // should fail since sending is off
+ TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
+ MARK();
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_MUSTPASS(rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
+ // should fail since RTCP is off
+ TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
+ MARK();
+ TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
+ TEST_MUSTPASS(rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
+ MARK();
+ // invalid data length
+ TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
+ 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabc", 31));
+ MARK();
+ // invalid data vector
+ TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(0, 0, 0, NULL, 0));
+ MARK();
+ ANL();
+
+#ifdef WEBRTC_CODEC_RED
+ TEST(SetFECStatus);
+ ANL();
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ cinst.pltype = 126;
+ strcpy(cinst.plname, "red");
+ cinst.plfreq = 8000;
+ cinst.pacsize = 0;
+ cinst.channels = 1;
+ cinst.rate = 0;
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ cinst.pltype = 104;
+ strcpy(cinst.plname, "isac");
+ cinst.plfreq = 32000;
+ cinst.pacsize = 960;
+ cinst.channels = 1;
+ cinst.rate = 45000;
+#else
+ cinst.pltype = 119;
+ strcpy(cinst.plname, "isaclc");
+ cinst.plfreq = 16000;
+ cinst.pacsize = 320;
+ cinst.channels = 1;
+ cinst.rate = 40000;
+#endif
+ // We have to re-register the audio codec payload type as stopReceive will
+ // clean the database
+ TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_LOG("Start playing a file as microphone again \n");
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+ TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, 126));
+ MARK();
+ TEST_LOG("Should sound OK with FEC enabled\n");
+ SLEEP(4000);
+ TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, false));
+ MARK();
+#endif // #ifdef WEBRTC_CODEC_RED
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ ANL();
+ AOK();
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestVideoSync
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestVideoSync()
+{
+ PrepareTest("VideoSync");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEVideoSync* vsync = _mgr.VideoSyncPtr();
+
+ // check if this interface is supported
+ if (!vsync)
+ {
+ TEST_LOG("VoEVideoSync is not supported!");
+ return -1;
+ }
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
+ "VoEVideoSync_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ ///////////////////////////
+ // Actual test starts here
+
+ TEST(SetInitTimestamp);
+ ANL();
+ TEST_MUSTPASS(!vsync->SetInitTimestamp(0, 12345));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(vsync->SetInitTimestamp(0, 12345));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ MARK();
+ SLEEP(1000);
+ AOK();
+ ANL();
+
+ TEST(SetInitSequenceNumber);
+ ANL();
+ TEST_MUSTPASS(!vsync->SetInitSequenceNumber(0, 123));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ MARK();
+ SLEEP(1000);
+ TEST_MUSTPASS(vsync->SetInitSequenceNumber(0, 123));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ MARK();
+ SLEEP(1000);
+ AOK();
+ ANL();
+
+ unsigned int timeStamp;
+ TEST(GetPlayoutTimestamp);
+ ANL();
+ TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
+ TEST_LOG("GetPlayoutTimestamp: %u", timeStamp);
+ SLEEP(1000);
+ TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
+ TEST_LOG(" %u", timeStamp);
+ SLEEP(1000);
+ TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
+ TEST_LOG(" %u\n", timeStamp);
+ AOK();
+ ANL();
+
+ TEST(SetMinimumPlayoutDelay);
+ ANL();
+ TEST_MUSTPASS(!vsync->SetMinimumPlayoutDelay(0, -1));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+ TEST_MUSTPASS(!vsync->SetMinimumPlayoutDelay(0, 5000));
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ MARK();
+
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ AOK();
+ ANL();
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestVolumeControl
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestVolumeControl()
+{
+ PrepareTest("TestVolumeControl");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+#ifdef _TEST_FILE_
+ VoEFile* file = _mgr.FilePtr();
+#endif
+#ifdef _TEST_HARDWARE_
+ VoEHardware* hardware = _mgr.HardwarePtr();
+#endif
+
+#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(
+ GetFilename("VoEVolumeControl_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE) && \
+ !defined(WEBRTC_ANDROID)))
+#if defined(_WIN32)
+ TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
+#else
+ TEST_MUSTPASS(hardware->SetRecordingDevice(0));
+ TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
+#endif
+#endif
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+#ifdef _TEST_FILE_
+ TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
+ true, true));
+#endif
+
+ ////////////////////////////
+ // Actual test starts here
+
+#if !defined(MAC_IPHONE)
+ TEST(SetSpeakerVolume);
+ ANL();
+ TEST_MUSTPASS(-1 != volume->SetSpeakerVolume(256));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ ANL();
+#endif // #if !defined(MAC_IPHONE)
+
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ TEST(SetMicVolume); ANL();
+ TEST_MUSTPASS(-1 != volume->SetMicVolume(256)); MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ ANL();
+#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+
+#if !defined(MAC_IPHONE)
+ TEST(SetChannelOutputVolumeScaling);
+ ANL();
+ TEST_MUSTPASS(-1 != volume->SetChannelOutputVolumeScaling(0, (float)-0.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetChannelOutputVolumeScaling(0, (float)10.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ ANL();
+#endif // #if !defined(MAC_IPHONE)
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ TEST(SetOutputVolumePan);
+ ANL();
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)-0.1,
+ (float)1.0));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.1,
+ (float)1.0));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.0,
+ (float)-0.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.0,
+ (float)1.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ ANL();
+
+ TEST(SetChannelOutputVolumePan);
+ ANL();
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)-0.1,
+ (float)1.0));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.1,
+ (float)1.0));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.0,
+ (float)-0.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.0,
+ (float)1.1));
+ MARK();
+ TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
+ ANL();
+#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+#ifdef _TEST_FILE_
+ TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
+#endif
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ AOK();
+ ANL();
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest::TestAPM
+// ----------------------------------------------------------------------------
+
+int VoEExtendedTest::TestAPM() {
+ PrepareTest("AudioProcessing");
+
+ VoEBase* voe_base_ = _mgr.BasePtr();
+ VoEAudioProcessing* apm = _mgr.APMPtr();
+
+ //#ifdef _USE_EXTENDED_TRACE_
+ TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("apm_trace.txt")));
+ TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
+ kTraceStateInfo |
+ kTraceWarning |
+ kTraceError |
+ kTraceCritical |
+ kTraceApiCall |
+ kTraceMemory |
+ kTraceInfo));
+ //#endif
+
+ TEST_MUSTPASS(voe_base_->Init());
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ ///////////////////////////
+ // Actual test starts here
+
+ int i;
+ bool enabled;
+
+ //////
+ // EC
+
+ const int ECSleep = 0;
+ const int ECIterations = 10;
+
+ EcModes ECmode(kEcAec);
+ AecmModes AECMmode(kAecmSpeakerphone);
+ bool enabledCNG(false);
+
+#if (defined(MAC_IPHONE) || defined(WEBRTC_ANDROID))
+ const EcModes ECmodeDefault(kEcAecm);
+#else
+ const EcModes ECmodeDefault(kEcAec);
+#endif
+
+ // verify default settings (should be OFF and mode as above)
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(ECmode != ECmodeDefault);
+
+ // set EC defaults
+ TEST_MUSTPASS(apm->SetEcStatus(false, kEcDefault));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(ECmode != ECmodeDefault);
+ SLEEP(ECSleep);
+
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // set kEcAec mode
+ TEST_MUSTPASS(apm->SetEcStatus(true, kEcAec));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(ECmode != kEcAec);
+ SLEEP(ECSleep);
+
+ // set kEcConference mode
+ TEST_MUSTPASS(apm->SetEcStatus(true, kEcConference));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(ECmode != kEcAec);
+ SLEEP(ECSleep);
+#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // get default AECM mode, should the kAecmSpeakerphone as default
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_MUSTPASS(AECMmode != kAecmSpeakerphone);
+ TEST_MUSTPASS(enabledCNG != true);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmQuietEarpieceOrHeadset, false));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmQuietEarpieceOrHeadset);
+ TEST_MUSTPASS(enabledCNG != false);
+
+ // set kEcAecm mode
+ TEST_MUSTPASS(apm->SetEcStatus(true, kEcAecm));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != true);
+ TEST_MUSTPASS(ECmode != kEcAecm);
+ SLEEP(ECSleep);
+
+ // AECM mode, get and set
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_MUSTPASS(AECMmode != kAecmQuietEarpieceOrHeadset);
+ TEST_MUSTPASS(enabledCNG != false);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmEarpiece, true));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmEarpiece);
+ TEST_MUSTPASS(enabledCNG != true);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmEarpiece, false));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmEarpiece);
+ TEST_MUSTPASS(enabledCNG != false);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmLoudEarpiece, true));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmLoudEarpiece);
+ TEST_MUSTPASS(enabledCNG != true);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmSpeakerphone, false));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmSpeakerphone);
+ TEST_MUSTPASS(enabledCNG != false);
+ TEST_MUSTPASS(apm->SetAecmMode(kAecmLoudSpeakerphone, true));
+ TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
+ TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
+ enabledCNG);
+ TEST_MUSTPASS(AECMmode != kAecmLoudSpeakerphone);
+ TEST_MUSTPASS(enabledCNG != true);
+
+ // verify that all modes are maintained when EC is disabled
+ TEST_MUSTPASS(apm->SetEcStatus(false));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(ECmode != kEcAecm);
+ SLEEP(ECSleep);
+
+ // restore defaults
+ TEST_MUSTPASS(apm->SetEcStatus(true, kEcDefault));
+ TEST_MUSTPASS(apm->SetEcStatus(false, kEcUnchanged));
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(ECmode != ECmodeDefault);
+ SLEEP(ECSleep);
+
+ // enable/disable many times in a row
+ for (i = 0; i < ECIterations; i++) {
+ TEST_MUSTPASS(apm->SetEcStatus(true));
+ TEST_MUSTPASS(apm->SetEcStatus(false));
+ }
+ TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
+ TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(ECmode != ECmodeDefault);
+ SLEEP(ECSleep);
+
+ ///////
+ // AGC
+
+ const int AGCSleep = 0;
+ const int AGCIterations = 10;
+
+ AgcModes AGCmode(kAgcAdaptiveAnalog);
+
+#if (defined(MAC_IPHONE) || defined(WEBRTC_ANDROID))
+ bool enabledDefault = false;
+ AgcModes AGCmodeDefault(kAgcAdaptiveDigital);
+#else
+ bool enabledDefault = true;
+ AgcModes AGCmodeDefault(kAgcAdaptiveAnalog);
+#endif
+
+ // verify default settings (should be as above)
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(enabled != enabledDefault);
+ TEST_MUSTPASS(AGCmode != AGCmodeDefault);
+
+ // set default AGC mode
+ TEST_MUSTPASS(apm->SetAgcStatus(false, kAgcDefault));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(AGCmode != AGCmodeDefault);
+ SLEEP(AGCSleep);
+
+ // set kAgcFixedDigital mode
+ TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcFixedDigital));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(AGCmode != kAgcFixedDigital);
+ SLEEP(AGCSleep);
+
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // set kAgcAdaptiveAnalog mode
+ TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcAdaptiveAnalog));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(AGCmode != kAgcAdaptiveAnalog);
+ SLEEP(AGCSleep);
+#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // set kAgcAdaptiveDigital mode
+ TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcAdaptiveDigital));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(AGCmode != kAgcAdaptiveDigital);
+ SLEEP(AGCSleep);
+
+ // verify that mode is maintained when AGC is disabled
+ TEST_MUSTPASS(apm->SetAgcStatus(false));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(AGCmode != kAgcAdaptiveDigital);
+ SLEEP(AGCSleep);
+
+ // restore default AGC
+ TEST_MUSTPASS(apm->SetAgcStatus(enabledDefault, kAgcDefault));
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(enabled != enabledDefault);
+ TEST_MUSTPASS(AGCmode != AGCmodeDefault);
+ SLEEP(AGCSleep);
+
+ // enable/disable many times in a row
+ for (i = 0; i < AGCIterations; i++)
+ {
+ TEST_MUSTPASS(apm->SetAgcStatus(true));
+ TEST_MUSTPASS(apm->SetAgcStatus(false));
+ }
+ TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
+ TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(AGCmode != AGCmodeDefault);
+
+ // --- Set/AGCConfig --
+
+ //
+ // targetLeveldBOv : [0, 31] (default 3)
+ // digitalCompressionGaindB: [0, 90] (default 9)
+ // limiterEnable : 0: Off, 1: On (default)
+
+ AgcConfig agcConfig;
+ AgcConfig agcConfigDefault;
+
+ const unsigned short targetLeveldBOvDefault = 3;
+ const unsigned short digitalCompressionGaindBDefault = 9;
+ const bool limiterEnableDefault = true;
+
+ const unsigned short targetLeveldBOvMax = 31;
+ const unsigned short digitalCompressionGaindBMax = 90;
+
+ // verify default configuration
+ TEST_MUSTPASS(apm->GetAgcConfig(agcConfigDefault));
+ TEST_LOG("AGC: targetLeveldBOv=%d, digitalCompressionGaindB=%d, "
+ "limiterEnable=%d\n",
+ agcConfigDefault.targetLeveldBOv,
+ agcConfigDefault.digitalCompressionGaindB,
+ agcConfigDefault.limiterEnable);
+ TEST_MUSTPASS(agcConfigDefault.targetLeveldBOv != targetLeveldBOvDefault);
+ TEST_MUSTPASS(agcConfigDefault.digitalCompressionGaindB !=
+ digitalCompressionGaindBDefault);
+ TEST_MUSTPASS(agcConfigDefault.limiterEnable != limiterEnableDefault);
+
+ // verify that invalid (out-of-range) parameters are detected
+ agcConfig = agcConfigDefault;
+ agcConfig.targetLeveldBOv = targetLeveldBOvMax + 1;
+ TEST_MUSTPASS(!apm->SetAgcConfig(agcConfig));
+ int err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_APM_ERROR);
+ agcConfig = agcConfigDefault;
+ agcConfig.digitalCompressionGaindB
+ = digitalCompressionGaindBMax + 1;
+ TEST_MUSTPASS(!apm->SetAgcConfig(agcConfig));
+
+ AgcConfig agcConfigSet;
+ agcConfigSet.digitalCompressionGaindB = 17;
+ agcConfigSet.targetLeveldBOv = 11;
+ agcConfigSet.limiterEnable = false;
+
+ // try some set/get operations using valid settings
+ TEST_MUSTPASS(apm->SetAgcConfig(agcConfigDefault));
+ TEST_MUSTPASS(apm->GetAgcConfig(agcConfig));
+ TEST_MUSTPASS(agcConfig.targetLeveldBOv != targetLeveldBOvDefault);
+ TEST_MUSTPASS(agcConfig.digitalCompressionGaindB !=
+ digitalCompressionGaindBDefault);
+ TEST_MUSTPASS(agcConfig.limiterEnable != limiterEnableDefault);
+
+ TEST_MUSTPASS(apm->SetAgcConfig(agcConfigSet));
+ TEST_MUSTPASS(apm->GetAgcConfig(agcConfig));
+ TEST_MUSTPASS(agcConfig.targetLeveldBOv != agcConfigSet.targetLeveldBOv);
+ TEST_MUSTPASS(agcConfig.digitalCompressionGaindB !=
+ agcConfigSet.digitalCompressionGaindB);
+ TEST_MUSTPASS(agcConfig.limiterEnable != agcConfigSet.limiterEnable);
+
+ // restore default AGC config
+ TEST_MUSTPASS(apm->SetAgcConfig(agcConfigDefault));
+ SLEEP(AGCSleep);
+
+ //////
+ // NS
+
+ const int NSSleep = 0;
+ const int NSIterations = 10;
+
+ NsModes NSmode(kNsHighSuppression);
+ NsModes NSmodeDefault(kNsModerateSuppression);
+
+ // verify default settings (should be OFF and mode as above)
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(NSmode != NSmodeDefault);
+
+ // enable default NS settings
+ // must set a value first time!
+ TEST_MUSTPASS(apm->SetNsStatus(false, kNsDefault));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(NSmode != NSmodeDefault);
+ SLEEP(NSSleep);
+
+ // set kNsLowSuppression mode
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsLowSuppression));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(NSmode != kNsLowSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsModerateSuppression mode
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsModerateSuppression));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(NSmode != kNsModerateSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsHighSuppression mode
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsHighSuppression));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(NSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsVeryHighSuppression mode
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsVeryHighSuppression));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(NSmode != kNsVeryHighSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsVeryHighSuppression mode
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsConference));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(NSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // verify that mode is maintained when NS is disabled
+ TEST_MUSTPASS(apm->SetNsStatus(false));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(NSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // restore default NS
+ TEST_MUSTPASS(apm->SetNsStatus(true, kNsDefault));
+ TEST_MUSTPASS(apm->SetNsStatus(false));
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(NSmode != NSmodeDefault);
+ SLEEP(NSSleep);
+
+ // enable/disable many times in a row
+ for (i = 0; i < NSIterations; i++) {
+ TEST_MUSTPASS(apm->SetNsStatus(true));
+ TEST_MUSTPASS(apm->SetNsStatus(false));
+ }
+ TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
+ TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(NSmode != NSmodeDefault);
+ SLEEP(NSSleep);
+
+ //////////////////////////////////
+ // Ec Metrics
+
+#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ TEST(GetEcMetricsStatus);
+ ANL();
+ TEST(SetEcMetricsStatus);
+ ANL();
+ TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
+ MARK();
+ TEST_MUSTPASS(enabled != false);
+ MARK(); // should be OFF by default
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
+ MARK();
+ TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
+ MARK();
+ TEST_MUSTPASS(enabled != true);
+ MARK();
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
+ MARK();
+ TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
+ MARK();
+ TEST_MUSTPASS(enabled != false);
+ MARK();
+ AOK();
+ ANL();
+
+ TEST(GetEchoMetrics);
+ ANL();
+
+ int ERL, ERLE, RERL, A_NLP;
+ TEST_MUSTPASS(-1 != apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
+ MARK(); // Should fail since not activated.
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_APM_ERROR);
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
+ TEST_MUSTPASS(-1 != apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
+ MARK(); // Should fail since AEC is off.
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_APM_ERROR);
+ TEST_MUSTPASS(apm->SetEcStatus(true));
+ TEST_MUSTPASS(apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
+ MARK(); // Should work now.
+ TEST_LOG("\nEcho: ERL=%d, ERLE=%d, RERL=%d, A_NLP=%d [dB]\n",
+ ERL, ERLE, RERL, A_NLP);
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
+ TEST_MUSTPASS(apm->SetEcStatus(false));
+ AOK();
+ ANL();
+
+ TEST(GetEcDelayMetrics);
+ ANL();
+
+ int delay_median = 0;
+ int delay_std = 0;
+ TEST_MUSTPASS(-1 != apm->GetEcDelayMetrics(delay_median, delay_std));
+ MARK(); // Should fail since not activated.
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_APM_ERROR);
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
+ TEST_MUSTPASS(-1 != apm->GetEcDelayMetrics(delay_median, delay_std));
+ MARK(); // Should fail since AEC is off.
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_APM_ERROR);
+ TEST_MUSTPASS(apm->SetEcStatus(true));
+ TEST_MUSTPASS(apm->GetEcDelayMetrics(delay_median, delay_std));
+ MARK(); // Should work now.
+ TEST_LOG("\nEC Delay: median=%d, std=%d [ms]\n", delay_median, delay_std);
+ TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
+ TEST_MUSTPASS(apm->SetEcStatus(false));
+ AOK();
+ ANL();
+
+#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
+ // far-end AudioProcessing
+ ///////
+ // AGC
+
+ AgcModes rxAGCmode(kAgcAdaptiveDigital);
+ AgcModes rxAGCmodeDefault(kAgcAdaptiveDigital);
+ bool rxEnabledDefault = false;
+
+ // verify default settings (should be as above)
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(enabled != rxEnabledDefault);
+ TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
+
+ // set default AGC mode
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, false, kAgcDefault));
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
+ SLEEP(AGCSleep);
+
+ // set kAgcAdaptiveAnalog mode, should fail
+ TEST_MUSTPASS(!apm->SetRxAgcStatus(0, true, kAgcAdaptiveAnalog));
+
+ // set kAgcFixedDigital mode
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, true, kAgcFixedDigital));
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(rxAGCmode != kAgcFixedDigital);
+ SLEEP(AGCSleep);
+
+ // set kAgcAdaptiveDigital mode
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, true, kAgcAdaptiveDigital));
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(rxAGCmode != kAgcAdaptiveDigital);
+ SLEEP(AGCSleep);
+
+ // verify that mode is maintained when AGC is disabled
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, false));
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxAGCmode != kAgcAdaptiveDigital);
+ SLEEP(AGCSleep);
+
+ // restore default AGC
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, enabledDefault, kAgcDefault));
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(enabled != enabledDefault);
+ TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
+ SLEEP(AGCSleep);
+
+ // enable/disable many times in a row
+ for (i = 0; i < AGCIterations; i++) {
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, true));
+ TEST_MUSTPASS(apm->SetRxAgcStatus(0, false));
+ }
+ TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
+ TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
+ rxAGCmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
+
+ // --- Set/GetAgcConfig --
+
+
+ // targetLeveldBOv : [0, 31] (default 3)
+ // digitalCompressionGaindB: [0, 90] (default 9)
+ // limiterEnable : 0: Off, 1: On (default)
+
+ AgcConfig rxAGCConfig;
+ AgcConfig rxAGCConfigDefault;
+
+ const unsigned short rxTargetLeveldBOvDefault = 3;
+ const unsigned short rxDigitalCompressionGaindBDefault = 9;
+ const bool rxLimiterEnableDefault = true;
+
+ const unsigned short rxTargetLeveldBOvMax = 31;
+ const unsigned short rxDigitalCompressionGaindBMax = 90;
+
+ // verify default configuration
+ TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfigDefault));
+ TEST_LOG(
+ "rxAGC: targetLeveldBOv=%u, digitalCompressionGaindB=%u, "
+ "limiterEnable=%d\n",
+ rxAGCConfigDefault.targetLeveldBOv,
+ rxAGCConfigDefault.digitalCompressionGaindB,
+ rxAGCConfigDefault.limiterEnable);
+ TEST_MUSTPASS(rxAGCConfigDefault.targetLeveldBOv !=
+ rxTargetLeveldBOvDefault);
+ TEST_MUSTPASS(rxAGCConfigDefault.digitalCompressionGaindB !=
+ rxDigitalCompressionGaindBDefault);
+ TEST_MUSTPASS(rxAGCConfigDefault.limiterEnable != rxLimiterEnableDefault);
+
+ // verify that invalid (out-of-range) parameters are detected
+ rxAGCConfig = rxAGCConfigDefault;
+ rxAGCConfig.targetLeveldBOv = rxTargetLeveldBOvMax + 1;
+ TEST_MUSTPASS(!apm->SetRxAgcConfig(0, rxAGCConfig));
+ int rxErr = voe_base_->LastError();
+ TEST_MUSTPASS(rxErr != VE_APM_ERROR);
+ rxAGCConfig = rxAGCConfigDefault;
+ rxAGCConfig.digitalCompressionGaindB
+ = rxDigitalCompressionGaindBMax + 1;
+ TEST_MUSTPASS(!apm->SetRxAgcConfig(0, rxAGCConfig));
+
+ AgcConfig rxAGCConfigSet;
+ rxAGCConfigSet.digitalCompressionGaindB = 17;
+ rxAGCConfigSet.targetLeveldBOv = 11;
+ rxAGCConfigSet.limiterEnable = false;
+
+ // try some set/get operations using valid settings
+ TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigDefault));
+ TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfig));
+ TEST_MUSTPASS(rxAGCConfig.targetLeveldBOv != rxTargetLeveldBOvDefault);
+ TEST_MUSTPASS(rxAGCConfig.digitalCompressionGaindB !=
+ rxDigitalCompressionGaindBDefault);
+ TEST_MUSTPASS(rxAGCConfig.limiterEnable != rxLimiterEnableDefault);
+
+ TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigSet));
+ TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfig));
+ TEST_MUSTPASS(rxAGCConfig.targetLeveldBOv !=
+ rxAGCConfigSet.targetLeveldBOv);
+ TEST_MUSTPASS(rxAGCConfig.digitalCompressionGaindB !=
+ rxAGCConfigSet.digitalCompressionGaindB);
+ TEST_MUSTPASS(rxAGCConfig.limiterEnable != rxAGCConfigSet.limiterEnable);
+
+ // restore default AGC config
+ TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigDefault));
+ SLEEP(AGCSleep);
+
+ //////
+ // NS
+
+ NsModes rxNSmode(kNsHighSuppression);
+ NsModes rxNSmodeDefault(kNsModerateSuppression);
+
+ // verify default settings (should be OFF and mode as above)
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxNSmode != rxNSmodeDefault);
+
+ // enable default NS settings
+ // must set a value first time!
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, false, kNsDefault));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxNSmode != rxNSmodeDefault);
+ SLEEP(NSSleep);
+
+ // set kNsLowSuppression mode
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsLowSuppression));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(rxNSmode != kNsLowSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsModerateSuppression mode
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsModerateSuppression));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(rxNSmode != kNsModerateSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsHighSuppression mode
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsHighSuppression));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsVeryHighSuppression mode
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsVeryHighSuppression));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(rxNSmode != kNsVeryHighSuppression);
+ SLEEP(NSSleep);
+
+ // set kNsVeryHighSuppression mode
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsConference));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // verify that mode is maintained when NS is disabled
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
+ SLEEP(NSSleep);
+
+ // restore default NS
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsDefault));
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxNSmode != NSmodeDefault);
+ SLEEP(NSSleep);
+
+ // enable/disable many times in a row
+ for (i = 0; i < NSIterations; i++)
+ {
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, true));
+ TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
+ }
+ TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
+ TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
+ TEST_MUSTPASS(enabled != false);
+ TEST_MUSTPASS(rxNSmode != NSmodeDefault);
+ SLEEP(NSSleep);
+
+ /////////////////////////////
+ // StartDebugRecording
+ ////////////////////////////
+ // StopDebugRecording
+ TEST_LOG("StartDebugRecording");
+ TEST_MUSTPASS(apm->StartDebugRecording(GetFilename("apm_debug.txt")));
+ SLEEP(1000);
+ TEST_LOG("StopDebugRecording");
+ TEST_MUSTPASS(apm->StopDebugRecording());
+
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h
new file mode 100644
index 0000000..9a10037
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h
@@ -0,0 +1,463 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
+#define WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
+
+#include "voe_standard_test.h"
+#include "audio_device.h"
+
+namespace voetest {
+
+class VoETestManager;
+
+// ----------------------------------------------------------------------------
+// AudioDeviceModule
+//
+// Implementation of the ADM to be used as external ADM in VoiceEngine.
+// This implementation is only a mock class, i.e., it does not provide
+// any real audio support.
+// ----------------------------------------------------------------------------
+
+class AudioDeviceModuleImpl : public AudioDeviceModule {
+ public:
+ // Factory methods
+ static AudioDeviceModuleImpl* Create();
+ static bool Destroy(AudioDeviceModuleImpl* adm);
+
+ // Helper methods which allows us to get some handy information about
+ // this mock implementation.
+ int32_t ReferenceCounter() const {
+ return _ref_count;
+ }
+
+ // RefCountedModule implementation (mocks default implementation)
+ virtual int32_t AddRef();
+ virtual int32_t Release();
+
+ // Module implementation
+ virtual int32_t Version(char* version,
+ uint32_t& remaining_buffer_in_bytes,
+ uint32_t& position) const {
+ return 0;
+ }
+ virtual int32_t ChangeUniqueId(const int32_t id) {
+ return 0;
+ }
+ virtual int32_t TimeUntilNextProcess() {
+ return -1;
+ }
+ virtual int32_t Process() {
+ return 0;
+ }
+
+ // AudioDeviceModule implementation
+ virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const {
+ return 0;
+ }
+
+ virtual ErrorCode LastError() const {
+ return static_cast<ErrorCode> (0);
+ }
+ virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) {
+ return 0;
+ }
+
+ virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) {
+ return 0;
+ }
+
+ virtual int32_t Init() {
+ return 0;
+ }
+ virtual int32_t Terminate() {
+ return 0;
+ }
+ virtual bool Initialized() const {
+ return true;
+ }
+
+ virtual int16_t PlayoutDevices() {
+ return -1;
+ }
+ virtual int16_t RecordingDevices() {
+ return -1;
+ }
+ virtual int32_t PlayoutDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
+ return -1;
+ }
+ virtual int32_t RecordingDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
+ return -1;
+ }
+
+ virtual int32_t SetPlayoutDevice(uint16_t index) {
+ return 0;
+ }
+ virtual int32_t SetPlayoutDevice(WindowsDeviceType device) {
+ return 0;
+ }
+ virtual int32_t SetRecordingDevice(uint16_t index) {
+ return 0;
+ }
+ virtual int32_t SetRecordingDevice(WindowsDeviceType device) {
+ return 0;
+ }
+
+ virtual int32_t PlayoutIsAvailable(bool* available) {
+ *available = true;
+ return 0;
+ }
+ virtual int32_t InitPlayout() {
+ return 0;
+ }
+ virtual bool PlayoutIsInitialized() const {
+ return true;
+ }
+ virtual int32_t RecordingIsAvailable(bool* available) {
+ *available = true;
+ return 0;
+ }
+ virtual int32_t InitRecording() {
+ return 0;
+ }
+ virtual bool RecordingIsInitialized() const {
+ return true;
+ }
+
+ virtual int32_t StartPlayout() {
+ return 0;
+ }
+ virtual int32_t StopPlayout() {
+ return 0;
+ }
+ virtual bool Playing() const {
+ return true;
+ }
+ virtual int32_t StartRecording() {
+ return 0;
+ }
+ virtual int32_t StopRecording() {
+ return 0;
+ }
+ virtual bool Recording() const {
+ return true;
+ }
+
+ virtual int32_t SetAGC(bool enable) {
+ return -1;
+ }
+ virtual bool AGC() const {
+ return false;
+ }
+
+ virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
+ uint16_t volumeRight) {
+ return -1;
+ }
+ virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
+ uint16_t* volumeRight) const {
+ return -1;
+ }
+
+ virtual int32_t SpeakerIsAvailable(bool* available) {
+ *available = true;
+ return 0;
+ }
+ virtual int32_t InitSpeaker() {
+ return 0;
+ }
+ virtual bool SpeakerIsInitialized() const {
+ return true;
+ }
+ virtual int32_t MicrophoneIsAvailable(bool* available) {
+ *available = true;
+ return 0;
+ }
+ virtual int32_t InitMicrophone() {
+ return 0;
+ }
+ virtual bool MicrophoneIsInitialized() const {
+ return true;
+ }
+
+ virtual int32_t SpeakerVolumeIsAvailable(bool* available) {
+ return -1;
+ }
+ virtual int32_t SetSpeakerVolume(uint32_t volume) {
+ return -1;
+ }
+ virtual int32_t SpeakerVolume(uint32_t* volume) const {
+ return -1;
+ }
+ virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const {
+ return -1;
+ }
+ virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const {
+ return -1;
+ }
+ virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const {
+ return -1;
+ }
+
+ virtual int32_t MicrophoneVolumeIsAvailable(bool* available) {
+ return -1;
+ }
+ virtual int32_t SetMicrophoneVolume(uint32_t volume) {
+ return -1;
+ }
+ virtual int32_t MicrophoneVolume(uint32_t* volume) const {
+ return -1;
+ }
+ virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const {
+ return -1;
+ }
+ virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const {
+ return -1;
+ }
+ virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const {
+ return -1;
+ }
+
+ virtual int32_t SpeakerMuteIsAvailable(bool* available) {
+ return -1;
+ }
+ virtual int32_t SetSpeakerMute(bool enable) {
+ return -1;
+ }
+ virtual int32_t SpeakerMute(bool* enabled) const {
+ return -1;
+ }
+
+ virtual int32_t MicrophoneMuteIsAvailable(bool* available) {
+ return -1;
+ }
+ virtual int32_t SetMicrophoneMute(bool enable) {
+ return -1;
+ }
+ virtual int32_t MicrophoneMute(bool* enabled) const {
+ return -1;
+ }
+
+ virtual int32_t MicrophoneBoostIsAvailable(bool* available) {
+ return -1;
+ }
+ virtual int32_t SetMicrophoneBoost(bool enable) {
+ return -1;
+ }
+ virtual int32_t MicrophoneBoost(bool* enabled) const {
+ return -1;
+ }
+
+ virtual int32_t StereoPlayoutIsAvailable(bool* available) const {
+ return -1;
+ }
+ virtual int32_t SetStereoPlayout(bool enable) {
+ return -1;
+ }
+ virtual int32_t StereoPlayout(bool* enabled) const {
+ return -1;
+ }
+ virtual int32_t StereoRecordingIsAvailable(bool* available) const {
+ return -1;
+ }
+ virtual int32_t SetStereoRecording(bool enable) {
+ return -1;
+ }
+ virtual int32_t StereoRecording(bool* enabled) const {
+ return -1;
+ }
+ virtual int32_t SetRecordingChannel(const ChannelType channel) {
+ return -1;
+ }
+ virtual int32_t RecordingChannel(ChannelType* channel) const {
+ return -1;
+ }
+
+ virtual int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0) {
+ return -1;
+ }
+ virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const {
+ return -1;
+ }
+ virtual int32_t PlayoutDelay(uint16_t* delayMS) const {
+ return -1;
+ }
+ virtual int32_t RecordingDelay(uint16_t* delayMS) const {
+ return -1;
+ }
+
+ virtual int32_t CPULoad(uint16_t* load) const {
+ return -1;
+ }
+
+ virtual int32_t StartRawOutputFileRecording(
+ const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
+ return -1;
+ }
+ virtual int32_t StopRawOutputFileRecording() {
+ return -1;
+ }
+ virtual int32_t StartRawInputFileRecording(
+ const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
+ return -1;
+ }
+ virtual int32_t StopRawInputFileRecording() {
+ return -1;
+ }
+
+ virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
+ return -1;
+ }
+ virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
+ return -1;
+ }
+ virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
+ return -1;
+ }
+ virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
+ return -1;
+ }
+
+ virtual int32_t ResetAudioDevice() {
+ return -1;
+ }
+ virtual int32_t SetLoudspeakerStatus(bool enable) {
+ return -1;
+ }
+ virtual int32_t GetLoudspeakerStatus(bool* enabled) const {
+ return -1;
+ }
+
+ protected:
+ AudioDeviceModuleImpl();
+ ~AudioDeviceModuleImpl();
+
+ private:
+ volatile int32_t _ref_count;
+};
+
+// ----------------------------------------------------------------------------
+// Transport
+// ----------------------------------------------------------------------------
+
+class ExtendedTestTransport : public Transport {
+ public:
+ ExtendedTestTransport(VoENetwork* ptr);
+ ~ExtendedTestTransport();
+ VoENetwork* myNetw;
+
+ protected:
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+
+ private:
+ static bool Run(void* ptr);
+ bool Process();
+
+ private:
+ ThreadWrapper* _thread;
+ CriticalSectionWrapper* _lock;
+ EventWrapper* _event;
+
+ private:
+ unsigned char _packetBuffer[1612];
+ int _length;
+ int _channel;
+};
+
+class XTransport : public Transport {
+ public:
+ XTransport(VoENetwork* netw, VoEFile* file);
+ VoENetwork* _netw;
+ VoEFile* _file;
+
+ public:
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+};
+
+class XRTPObserver : public VoERTPObserver {
+ public:
+ XRTPObserver();
+ ~XRTPObserver();
+ virtual void OnIncomingCSRCChanged(const int channel,
+ const unsigned int CSRC,
+ const bool added);
+ virtual void OnIncomingSSRCChanged(const int channel,
+ const unsigned int SSRC);
+ public:
+ unsigned int _SSRC;
+};
+
+// ----------------------------------------------------------------------------
+// VoEExtendedTest
+// ----------------------------------------------------------------------------
+
+class VoEExtendedTest : public VoiceEngineObserver,
+ public VoEConnectionObserver {
+ public:
+ VoEExtendedTest(VoETestManager& mgr);
+ ~VoEExtendedTest();
+ int PrepareTest(const char* str) const;
+ int TestPassed(const char* str) const;
+ int TestBase();
+ int TestCallReport();
+ int TestCodec();
+ int TestDtmf();
+ int TestEncryption();
+ int TestExternalMedia();
+ int TestFile();
+ int TestMixing();
+ int TestHardware();
+ int TestNetEqStats();
+ int TestNetwork();
+ int TestRTP_RTCP();
+ int TestVideoSync();
+ int TestVolumeControl();
+ int TestAPM();
+ public:
+ int ErrorCode() const {
+ return _errCode;
+ }
+ void ClearErrorCode() {
+ _errCode = 0;
+ }
+ protected:
+ // from VoiceEngineObserver
+ void CallbackOnError(const int errCode, const int channel);
+ void CallbackOnTrace(const TraceLevel level, const char* message, const int length);
+ protected:
+ // from VoEConnectionObserver
+ void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+ private:
+ void Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone = false,
+ bool addTimeMarker = false);
+ void Sleep(unsigned int timeMillisec, bool addMarker = false);
+ void StartMedia(int channel, int rtpPort, bool listen, bool playout, bool send);
+ void StopMedia(int channel);
+ int RunMixingTest(int num_remote_channels, int num_local_channels,
+ int16_t input_value, int16_t max_output_value,
+ int16_t min_output_value);
+ private:
+ VoETestManager& _mgr;
+ private:
+ int _errCode;
+ bool _alive;
+ bool _listening[32];
+ bool _playing[32];
+ bool _sending[32];
+};
+
+} // namespace voetest
+#endif // WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc
new file mode 100644
index 0000000..a6af7fa
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc
@@ -0,0 +1,1851 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "engine_configurations.h"
+#if defined(_WIN32)
+#include <conio.h> // exists only on windows
+#include <tchar.h>
+#endif
+
+#include "voe_standard_test.h"
+
+#if defined (_ENABLE_VISUAL_LEAK_DETECTOR_) && defined(_DEBUG) && \
+ defined(_WIN32) && !defined(_INSTRUMENTATION_TESTING_)
+#include "vld.h"
+#endif
+
+#ifdef MAC_IPHONE
+#include "../../source/voice_engine_defines.h" // defines build macros
+#else
+#include "../../source/voice_engine_defines.h" // defines build macros
+#endif
+
+#include "automated_mode.h"
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+
+#ifdef _TEST_NETEQ_STATS_
+#include "../../interface/voe_neteq_stats.h" // Not available in delivery folder
+#endif
+
+#include "voe_extended_test.h"
+#include "voe_stress_test.h"
+#include "voe_unit_test.h"
+#include "voe_cpu_test.h"
+
+using namespace webrtc;
+
+namespace voetest {
+
+#ifdef MAC_IPHONE
+// Defined in iPhone specific test file
+int GetDocumentsDir(char* buf, int bufLen);
+char* GetFilename(char* filename);
+const char* GetFilename(const char* filename);
+int GetResource(char* resource, char* dest, int destLen);
+char* GetResource(char* resource);
+const char* GetResource(const char* resource);
+// #ifdef MAC_IPHONE
+#elif defined(WEBRTC_ANDROID)
+char filenameStr[2][256];
+int currentStr = 0;
+
+char* GetFilename(char* filename) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/%s", filename);
+ return filenameStr[currentStr];
+}
+
+const char* GetFilename(const char* filename) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/%s", filename);
+ return filenameStr[currentStr];
+}
+
+int GetResource(char* resource, char* dest, int destLen) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
+ strncpy(dest, filenameStr[currentStr], destLen-1);
+ return 0;
+}
+
+char* GetResource(char* resource) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
+ return filenameStr[currentStr];
+}
+
+const char* GetResource(const char* resource) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
+ return filenameStr[currentStr];
+}
+
+#else
+char filenameStr[2][256];
+int currentStr = 0;
+
+char* GetFilename(char* filename) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/tmp/%s", filename);
+ return filenameStr[currentStr];
+}
+const char* GetFilename(const char* filename) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/tmp/%s", filename);
+ return filenameStr[currentStr];
+}
+int GetResource(char* resource, char* dest, int destLen) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/tmp/%s", resource);
+ strncpy(dest, filenameStr[currentStr], destLen - 1);
+ return 0;
+}
+char* GetResource(char* resource) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/tmp/%s", resource);
+ return filenameStr[currentStr];
+}
+const char* GetResource(const char* resource) {
+ currentStr = !currentStr;
+ sprintf(filenameStr[currentStr], "/tmp/%s", resource);
+ return filenameStr[currentStr];
+}
+#endif
+
+#if !defined(MAC_IPHONE)
+const char* summaryFilename = "/tmp/VoiceEngineSummary.txt";
+#endif
+// For iPhone the summary filename is created in createSummary
+
+int dummy = 0; // Dummy used in different functions to avoid warnings
+
+TestRtpObserver::TestRtpObserver() {
+ Reset();
+}
+
+TestRtpObserver::~TestRtpObserver() {
+}
+
+void TestRtpObserver::Reset() {
+ for (int i = 0; i < 2; i++) {
+ ssrc_[i] = 0;
+ csrc_[i][0] = 0;
+ csrc_[i][1] = 0;
+ added_[i][0] = false;
+ added_[i][1] = false;
+ size_[i] = 0;
+ }
+}
+
+void TestRtpObserver::OnIncomingCSRCChanged(const int channel,
+ const unsigned int CSRC,
+ const bool added) {
+ char msg[128];
+ sprintf(msg, "=> OnIncomingCSRCChanged(channel=%d, CSRC=%u, added=%d)\n",
+ channel, CSRC, added);
+ TEST_LOG("%s", msg);
+
+ if (channel > 1)
+ return; // Not enough memory.
+
+ csrc_[channel][size_[channel]] = CSRC;
+ added_[channel][size_[channel]] = added;
+
+ size_[channel]++;
+ if (size_[channel] == 2)
+ size_[channel] = 0;
+}
+
+void TestRtpObserver::OnIncomingSSRCChanged(const int channel,
+ const unsigned int SSRC) {
+ char msg[128];
+ sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
+ SSRC);
+ TEST_LOG("%s", msg);
+
+ ssrc_[channel] = SSRC;
+}
+
+void MyDeadOrAlive::OnPeriodicDeadOrAlive(const int /*channel*/,
+ const bool alive) {
+ if (alive) {
+ TEST_LOG("ALIVE\n");
+ } else {
+ TEST_LOG("DEAD\n");
+ }
+ fflush(NULL);
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+void MyMedia::Process(const int channel,
+ const ProcessingTypes type,
+ WebRtc_Word16 audio_10ms[],
+ const int length,
+ const int samplingFreqHz,
+ const bool stereo) {
+ for (int i = 0; i < length; i++) {
+ if (!stereo) {
+ audio_10ms[i] = (WebRtc_Word16) (audio_10ms[i] *
+ sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
+ } else {
+ // interleaved stereo
+ audio_10ms[2 * i] = (WebRtc_Word16) (audio_10ms[2 * i] *
+ sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
+ audio_10ms[2 * i + 1] = (WebRtc_Word16) (audio_10ms[2 * i + 1] *
+ sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
+ }
+ f++;
+ }
+}
+#endif
+
+MyMedia mobj;
+
+FakeExternalTransport::FakeExternalTransport(VoENetwork* ptr)
+ : my_network_(ptr),
+ thread_(NULL),
+ lock_(NULL),
+ event_(NULL),
+ length_(0),
+ channel_(0),
+ delay_is_enabled_(0),
+ delay_time_in_ms_(0) {
+ const char* threadName = "external_thread";
+ lock_ = CriticalSectionWrapper::CreateCriticalSection();
+ event_ = EventWrapper::Create();
+ thread_ = ThreadWrapper::CreateThread(Run, this, kHighPriority, threadName);
+ if (thread_) {
+ unsigned int id;
+ thread_->Start(id);
+ }
+}
+
+FakeExternalTransport::~FakeExternalTransport() {
+ if (thread_) {
+ thread_->SetNotAlive();
+ event_->Set();
+ if (thread_->Stop()) {
+ delete thread_;
+ thread_ = NULL;
+ delete event_;
+ event_ = NULL;
+ delete lock_;
+ lock_ = NULL;
+ }
+ }
+}
+
+bool FakeExternalTransport::Run(void* ptr) {
+ return static_cast<FakeExternalTransport*> (ptr)->Process();
+}
+
+bool FakeExternalTransport::Process() {
+ switch (event_->Wait(500)) {
+ case kEventSignaled:
+ lock_->Enter();
+ my_network_->ReceivedRTPPacket(channel_, packet_buffer_, length_);
+ lock_->Leave();
+ return true;
+ case kEventTimeout:
+ return true;
+ case kEventError:
+ break;
+ }
+ return true;
+}
+
+int FakeExternalTransport::SendPacket(int channel, const void *data, int len) {
+ lock_->Enter();
+ if (len < 1612) {
+ memcpy(packet_buffer_, (const unsigned char*) data, len);
+ length_ = len;
+ channel_ = channel;
+ }
+ lock_->Leave();
+ event_->Set(); // triggers ReceivedRTPPacket() from worker thread
+ return len;
+}
+
+int FakeExternalTransport::SendRTCPPacket(int channel, const void *data, int len) {
+ if (delay_is_enabled_) {
+ Sleep(delay_time_in_ms_);
+ }
+ my_network_->ReceivedRTCPPacket(channel, data, len);
+ return len;
+}
+
+void FakeExternalTransport::SetDelayStatus(bool enable, unsigned int delayInMs) {
+ delay_is_enabled_ = enable;
+ delay_time_in_ms_ = delayInMs;
+}
+
+ErrorObserver::ErrorObserver() {
+ code = -1;
+}
+void ErrorObserver::CallbackOnError(const int channel, const int errCode) {
+ code = errCode;
+#ifndef _INSTRUMENTATION_TESTING_
+ TEST_LOG("\n************************\n");
+ TEST_LOG(" RUNTIME ERROR: %d \n", errCode);
+ TEST_LOG("************************\n");
+#endif
+}
+
+void MyTraceCallback::Print(const TraceLevel level,
+ const char *traceString,
+ const int length) {
+ if (traceString) {
+ char* tmp = new char[length];
+ memcpy(tmp, traceString, length);
+ TEST_LOG("%s", tmp);
+ TEST_LOG("\n");
+ delete[] tmp;
+ }
+}
+
+void RtcpAppHandler::OnApplicationDataReceived(
+ const int /*channel*/, const unsigned char sub_type,
+ const unsigned int name, const unsigned char* data,
+ const unsigned short length_in_bytes) {
+ length_in_bytes_ = length_in_bytes;
+ memcpy(data_, &data[0], length_in_bytes);
+ sub_type_ = sub_type;
+ name_ = name;
+}
+
+void RtcpAppHandler::Reset() {
+ length_in_bytes_ = 0;
+ memset(data_, 0, sizeof(data_));
+ sub_type_ = 0;
+ name_ = 0;
+}
+
+void my_encryption::encrypt(int, unsigned char * in_data,
+ unsigned char * out_data,
+ int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = ~in_data[i];
+ *bytes_out = bytes_in + 2; // length is increased by 2
+}
+
+void my_encryption::decrypt(int, unsigned char * in_data,
+ unsigned char * out_data,
+ int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = ~in_data[i];
+ *bytes_out = bytes_in - 2; // length is decreased by 2
+}
+
+void my_encryption::encrypt_rtcp(int,
+ unsigned char * in_data,
+ unsigned char * out_data,
+ int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = ~in_data[i];
+ *bytes_out = bytes_in + 2;
+}
+
+void my_encryption::decrypt_rtcp(int, unsigned char * in_data,
+ unsigned char * out_data,
+ int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = ~in_data[i];
+ *bytes_out = bytes_in + 2;
+}
+
+void SubAPIManager::DisplayStatus() const {
+ TEST_LOG("Supported sub APIs:\n\n");
+ if (_base)
+ TEST_LOG(" Base\n");
+ if (_callReport)
+ TEST_LOG(" CallReport\n");
+ if (_codec)
+ TEST_LOG(" Codec\n");
+ if (_dtmf)
+ TEST_LOG(" Dtmf\n");
+ if (_encryption)
+ TEST_LOG(" Encryption\n");
+ if (_externalMedia)
+ TEST_LOG(" ExternalMedia\n");
+ if (_file)
+ TEST_LOG(" File\n");
+ if (_hardware)
+ TEST_LOG(" Hardware\n");
+ if (_netEqStats)
+ TEST_LOG(" NetEqStats\n");
+ if (_network)
+ TEST_LOG(" Network\n");
+ if (_rtp_rtcp)
+ TEST_LOG(" RTP_RTCP\n");
+ if (_videoSync)
+ TEST_LOG(" VideoSync\n");
+ if (_volumeControl)
+ TEST_LOG(" VolumeControl\n");
+ if (_apm)
+ TEST_LOG(" AudioProcessing\n");
+ ANL();
+ TEST_LOG("Excluded sub APIs:\n\n");
+ if (!_base)
+ TEST_LOG(" Base\n");
+ if (!_callReport)
+ TEST_LOG(" CallReport\n");
+ if (!_codec)
+ TEST_LOG(" Codec\n");
+ if (!_dtmf)
+ TEST_LOG(" Dtmf\n");
+ if (!_encryption)
+ TEST_LOG(" Encryption\n");
+ if (!_externalMedia)
+ TEST_LOG(" ExternamMedia\n");
+ if (!_file)
+ TEST_LOG(" File\n");
+ if (!_hardware)
+ TEST_LOG(" Hardware\n");
+ if (!_netEqStats)
+ TEST_LOG(" NetEqStats\n");
+ if (!_network)
+ TEST_LOG(" Network\n");
+ if (!_rtp_rtcp)
+ TEST_LOG(" RTP_RTCP\n");
+ if (!_videoSync)
+ TEST_LOG(" VideoSync\n");
+ if (!_volumeControl)
+ TEST_LOG(" VolumeControl\n");
+ if (!_apm)
+ TEST_LOG(" AudioProcessing\n");
+ ANL();
+}
+
+bool SubAPIManager::GetExtendedMenuSelection(ExtendedSelection& sel) {
+ printf("------------------------------------------------\n");
+ printf("Select extended test\n\n");
+ printf(" (0) None\n");
+ printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
+ printf(" (1) Base");
+ if (_base)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (2) CallReport");
+ if (_callReport)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (3) Codec");
+ if (_codec)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (4) Dtmf");
+ if (_dtmf)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (5) Encryption");
+ if (_encryption)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (6) VoEExternalMedia");
+ if (_externalMedia)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (7) File");
+ if (_file)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (8) Mixing");
+ if (_file)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (9) Hardware");
+ if (_hardware)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (10) NetEqStats");
+ if (_netEqStats)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (11) Network");
+ if (_network)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (12) RTP_RTCP");
+ if (_rtp_rtcp)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (13) VideoSync");
+ if (_videoSync)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (14) VolumeControl");
+ if (_volumeControl)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf(" (15) AudioProcessing");
+ if (_apm)
+ printf("\n");
+ else
+ printf(" (NA)\n");
+ printf("\n: ");
+
+ ExtendedSelection xsel(XSEL_Invalid);
+ int selection(0);
+ dummy = scanf("%d", &selection);
+
+ switch (selection) {
+ case 0:
+ xsel = XSEL_None;
+ break;
+ case 1:
+ if (_base)
+ xsel = XSEL_Base;
+ break;
+ case 2:
+ if (_callReport)
+ xsel = XSEL_CallReport;
+ break;
+ case 3:
+ if (_codec)
+ xsel = XSEL_Codec;
+ break;
+ case 4:
+ if (_dtmf)
+ xsel = XSEL_DTMF;
+ break;
+ case 5:
+ if (_encryption)
+ xsel = XSEL_Encryption;
+ break;
+ case 6:
+ if (_externalMedia)
+ xsel = XSEL_ExternalMedia;
+ break;
+ case 7:
+ if (_file)
+ xsel = XSEL_File;
+ break;
+ case 8:
+ if (_file)
+ xsel = XSEL_Mixing;
+ break;
+ case 9:
+ if (_hardware)
+ xsel = XSEL_Hardware;
+ break;
+ case 10:
+ if (_netEqStats)
+ xsel = XSEL_NetEqStats;
+ break;
+ case 11:
+ if (_network)
+ xsel = XSEL_Network;
+ break;
+ case 12:
+ if (_rtp_rtcp)
+ xsel = XSEL_RTP_RTCP;
+ break;
+ case 13:
+ if (_videoSync)
+ xsel = XSEL_VideoSync;
+ break;
+ case 14:
+ if (_volumeControl)
+ xsel = XSEL_VolumeControl;
+ break;
+ case 15:
+ if (_apm)
+ xsel = XSEL_AudioProcessing;
+ break;
+ default:
+ xsel = XSEL_Invalid;
+ break;
+ }
+ if (xsel == XSEL_Invalid)
+ printf("Invalid selection!\n");
+
+ sel = xsel;
+ _xsel = xsel;
+
+ return (xsel != XSEL_Invalid);
+}
+
+VoETestManager::VoETestManager()
+ : initialized_(false),
+ voice_engine_(NULL),
+ voe_base_(0),
+ voe_call_report_(0),
+ voe_codec_(0),
+ voe_dtmf_(0),
+ voe_encrypt_(0),
+ voe_xmedia_(0),
+ voe_file_(0),
+ voe_hardware_(0),
+ voe_network_(0),
+#ifdef _TEST_NETEQ_STATS_
+ voe_neteq_stats_(NULL),
+#endif
+ voe_rtp_rtcp_(0),
+ voe_vsync_(0),
+ voe_volume_control_(0),
+ voe_apm_(0)
+{
+}
+
+VoETestManager::~VoETestManager() {
+}
+
+bool VoETestManager::Init() {
+ if (initialized_)
+ return true;
+
+ if (VoiceEngine::SetTraceFile(NULL) != -1) {
+ // should not be possible to call a Trace method before the VoE is
+ // created
+ TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
+ "should fail)!\n", __LINE__);
+ return false;
+ }
+
+ voice_engine_ = VoiceEngine::Create();
+ if (!voice_engine_) {
+ TEST_LOG("Failed to create VoiceEngine\n");
+ return false;
+ }
+
+ return true;
+}
+
+void VoETestManager::GetInterfaces() {
+ if (voice_engine_) {
+ voe_base_ = VoEBase::GetInterface(voice_engine_);
+ voe_codec_ = VoECodec::GetInterface(voice_engine_);
+ voe_volume_control_ = VoEVolumeControl::GetInterface(voice_engine_);
+ voe_dtmf_ = VoEDtmf::GetInterface(voice_engine_);
+ voe_rtp_rtcp_ = VoERTP_RTCP::GetInterface(voice_engine_);
+ voe_apm_ = VoEAudioProcessing::GetInterface(voice_engine_);
+ voe_network_ = VoENetwork::GetInterface(voice_engine_);
+ voe_file_ = VoEFile::GetInterface(voice_engine_);
+#ifdef _TEST_VIDEO_SYNC_
+ voe_vsync_ = VoEVideoSync::GetInterface(voice_engine_);
+#endif
+ voe_encrypt_ = VoEEncryption::GetInterface(voice_engine_);
+ voe_hardware_ = VoEHardware::GetInterface(voice_engine_);
+ // Set the audio layer to use in all tests
+ if (voe_hardware_) {
+ int res = voe_hardware_->SetAudioDeviceLayer(TESTED_AUDIO_LAYER);
+ if (res < 0) {
+ printf("\nERROR: failed to set audio layer to use in "
+ "testing\n");
+ } else {
+ printf("\nAudio layer %d will be used in testing\n",
+ TESTED_AUDIO_LAYER);
+ }
+ }
+#ifdef _TEST_XMEDIA_
+ voe_xmedia_ = VoEExternalMedia::GetInterface(voice_engine_);
+#endif
+#ifdef _TEST_CALL_REPORT_
+ voe_call_report_ = VoECallReport::GetInterface(voice_engine_);
+#endif
+#ifdef _TEST_NETEQ_STATS_
+ voe_neteq_stats_ = VoENetEqStats::GetInterface(voice_engine_);
+#endif
+ }
+}
+
+int VoETestManager::ReleaseInterfaces() {
+ int err(0), remInt(1), j(0);
+ bool releaseOK(true);
+
+ if (voe_base_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_base_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d base interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ // try to release one addition time (should fail)
+ TEST_MUSTPASS(-1 != voe_base_->Release());
+ err = voe_base_->LastError();
+ // it is considered safe to delete even if Release has been called
+ // too many times
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_codec_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_codec_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d codec interfaces"
+ " (should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_codec_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_volume_control_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_volume_control_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d volume interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_volume_control_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_dtmf_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_dtmf_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d dtmf interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_dtmf_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_rtp_rtcp_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_rtp_rtcp_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d rtp/rtcp interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_rtp_rtcp_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_apm_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_apm_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d apm interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_apm_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_network_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_network_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d network interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_network_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_file_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_file_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d file interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_file_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#ifdef _TEST_VIDEO_SYNC_
+ if (voe_vsync_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_vsync_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d video sync interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_vsync_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#endif
+ if (voe_encrypt_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_encrypt_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d encryption interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_encrypt_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+ if (voe_hardware_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_hardware_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d hardware interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_hardware_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#ifdef _TEST_XMEDIA_
+ if (voe_xmedia_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_xmedia_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d external media interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_xmedia_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#endif
+#ifdef _TEST_CALL_REPORT_
+ if (voe_call_report_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_call_report_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d call report interfaces"
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_call_report_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#endif
+#ifdef _TEST_NETEQ_STATS_
+ if (voe_neteq_stats_) {
+ for (remInt = 1, j = 0; remInt > 0; j++)
+ TEST_MUSTPASS(-1 == (remInt = voe_neteq_stats_->Release()));
+ if (j > 1) {
+ TEST_LOG("\n\n*** Error: released %d neteq stat interfaces "
+ "(should only be 1) \n", j);
+ releaseOK = false;
+ }
+ TEST_MUSTPASS(-1 != voe_neteq_stats_->Release());
+ err = voe_base_->LastError();
+ TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
+ }
+#endif
+ if (false == VoiceEngine::Delete(voice_engine_)) {
+ TEST_LOG("\n\nVoiceEngine::Delete() failed. \n");
+ releaseOK = false;
+ }
+
+ if (VoiceEngine::SetTraceFile(NULL) != -1) {
+ TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
+ "should fail)!\n", __LINE__);
+ }
+
+ return (releaseOK == true) ? 0 : -1;
+}
+
+int VoETestManager::SetUp(ErrorObserver* error_observer) {
+ char char_buffer[1024];
+
+ TEST_MUSTPASS(voe_base_->Init());
+
+#if defined(WEBRTC_ANDROID)
+ TEST_MUSTPASS(voe_hardware_->SetLoudspeakerStatus(false));
+#endif
+
+ TEST_MUSTPASS(voe_base_->RegisterVoiceEngineObserver(*error_observer));
+
+ TEST_LOG("Get version \n");
+ TEST_MUSTPASS(voe_base_->GetVersion(char_buffer));
+ TEST_LOG("--------------------\n%s\n--------------------\n", char_buffer);
+
+ TEST_LOG("Create channel \n");
+ int nChannels = voe_base_->MaxNumOfChannels();
+ TEST_MUSTPASS(!(nChannels > 0));
+ TEST_LOG("Max number of channels = %d \n", nChannels);
+ TEST_MUSTPASS(voe_base_->CreateChannel());
+
+ return 0;
+}
+
+int VoETestManager::TestStartStreaming(FakeExternalTransport& channel0_transport) {
+ TEST_LOG("\n\n+++ Starting streaming +++\n\n");
+
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ TEST_LOG("Enabling external transport \n");
+ TEST_MUSTPASS(voe_network_->RegisterExternalTransport(0, channel0_transport));
+#else
+ TEST_LOG("Setting send and receive parameters \n");
+ TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+ // No IP specified => "0.0.0.0" will be stored.
+ TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
+
+ CodecInst Jing_inst;
+ Jing_inst.channels = 1;
+ Jing_inst.pacsize = 160;
+ Jing_inst.plfreq = 8000;
+ Jing_inst.pltype = 0;
+ Jing_inst.rate = 64000;
+ strcpy(Jing_inst.plname, "PCMU");
+ TEST_MUSTPASS(voe_codec_->SetSendCodec(0, Jing_inst));
+
+ int port = -1;
+ int src_port = -1;
+ int rtcp_port = -1;
+ char ip_address[64] = { 0 };
+ strcpy(ip_address, "10.10.10.10");
+ TEST_MUSTPASS(voe_base_->GetSendDestination(0, port, ip_address, src_port,
+ rtcp_port));
+ TEST_MUSTPASS(8000 != port);
+ TEST_MUSTPASS(8000 != src_port);
+ TEST_MUSTPASS(8001 != rtcp_port);
+ TEST_MUSTPASS(_stricmp(ip_address, "127.0.0.1"));
+
+ port = -1;
+ rtcp_port = -1;
+ TEST_MUSTPASS(voe_base_->GetLocalReceiver(0, port, rtcp_port, ip_address));
+ TEST_MUSTPASS(8000 != port);
+ TEST_MUSTPASS(8001 != rtcp_port);
+ TEST_MUSTPASS(_stricmp(ip_address, "0.0.0.0"));
+#endif
+ return 0;
+}
+
+int VoETestManager::TestStartPlaying() {
+ TEST_LOG("Start listening, playout and sending \n");
+ TEST_MUSTPASS(voe_base_->StartReceive(0));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ // Run in full duplex.
+ TEST_LOG("You should now hear yourself, running default codec (PCMU)\n");
+ SLEEP(2000);
+
+ if (voe_file_) {
+ TEST_LOG("Start playing a file as microphone, so you don't need to"
+ " speak all the time\n");
+ TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0,
+ AudioFilename(),
+ true,
+ true));
+ SLEEP(1000);
+ }
+ return 0;
+}
+
+int VoETestManager::DoStandardTest() {
+ // Ensure we have all input files:
+ TEST_MUSTPASS(!strcmp("", AudioFilename()));
+
+ TEST_LOG("\n\n+++ Base tests +++\n\n");
+
+ ErrorObserver error_observer;
+ if (SetUp(&error_observer) != 0) return -1;
+
+ voe_network_->SetSourceFilter(0, 0);
+
+ FakeExternalTransport channel0_transport(voe_network_);
+ if (TestStartStreaming(channel0_transport) != 0) return -1;
+ if (TestStartPlaying() != 0) return -1;
+
+ //////////////
+ // Video Sync
+
+#ifdef _TEST_VIDEO_SYNC_
+ TEST_LOG("\n\n+++ Video sync tests +++\n\n");
+
+ unsigned int val;
+ TEST_MUSTPASS(voe_vsync_->GetPlayoutTimestamp(0, val));
+ TEST_LOG("Playout timestamp = %lu\n", (long unsigned int) val);
+
+ TEST_LOG("Init timestamp and sequence number manually\n");
+ TEST_MUSTPASS(!voe_vsync_->SetInitTimestamp(0, 12345));
+ TEST_MUSTPASS(!voe_vsync_->SetInitSequenceNumber(0, 123));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_vsync_->SetInitTimestamp(0, 12345));
+ TEST_MUSTPASS(voe_vsync_->SetInitSequenceNumber(0, 123));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+ if (voe_file_) {
+ TEST_LOG("Start playing a file as microphone again \n");
+ TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0,
+ AudioFilename(),
+ true,
+ true));
+ }
+ SLEEP(3000);
+
+ TEST_LOG("Check delay estimates during 15 seconds, verify that "
+ "they stabilize during this time\n");
+ int valInt = -1;
+ for (int i = 0; i < 15; i++) {
+ TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
+ TEST_LOG("Delay estimate = %d ms\n", valInt);
+#if defined(MAC_IPHONE)
+ TEST_MUSTPASS(valInt <= 30);
+#else
+ TEST_MUSTPASS(valInt <= 45); // 45=20+25 => can't be this low
+#endif
+ SLEEP(1000);
+ }
+
+ TEST_LOG("Setting NetEQ min delay to 500 milliseconds and repeat "
+ "the test above\n");
+ TEST_MUSTPASS(voe_vsync_->SetMinimumPlayoutDelay(0, 500));
+ for (int i = 0; i < 15; i++) {
+ TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
+ TEST_LOG("Delay estimate = %d ms\n", valInt);
+ TEST_MUSTPASS(valInt <= 45);
+ SLEEP(1000);
+ }
+
+ TEST_LOG("Setting NetEQ min delay to 0 milliseconds and repeat"
+ " the test above\n");
+ TEST_MUSTPASS(voe_vsync_->SetMinimumPlayoutDelay(0, 0));
+ for (int i = 0; i < 15; i++) {
+ TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
+ TEST_LOG("Delay estimate = %d ms\n", valInt);
+ TEST_MUSTPASS(valInt <= 45);
+ SLEEP(1000);
+ }
+
+#if (defined (_WIN32) || (defined(WEBRTC_LINUX)) && !defined(WEBRTC_ANDROID))
+ valInt = -1;
+ TEST_MUSTPASS(voe_vsync_->GetPlayoutBufferSize(valInt));
+ TEST_LOG("Soundcard buffer size = %d ms\n", valInt);
+#endif
+#else
+ TEST_LOG("\n\n+++ Video sync tests NOT ENABLED +++\n");
+#endif // #ifdef _TEST_VIDEO_SYNC_
+ //////////////
+ // Encryption
+
+#ifdef _TEST_ENCRYPT_
+ TEST_LOG("\n\n+++ Encryption tests +++\n\n");
+
+#ifdef WEBRTC_SRTP
+ TEST_LOG("SRTP tests:\n");
+
+ unsigned char encrKey[30] = {1, 2, 3, 4, 5, 6, 7, 8, 9, 0,
+ 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0};
+
+ TEST_LOG("Enable SRTP encryption and decryption, you should still hear"
+ " the voice\n");
+ TEST_MUSTPASS(voe_encrypt_->EnableSRTPSend(0,
+ kCipherAes128CounterMode,
+ 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, encrKey));
+ TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
+ kCipherAes128CounterMode,
+ 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, encrKey));
+ SLEEP(2000);
+
+ TEST_LOG("Disabling decryption, you should hear nothing or garbage\n");
+ TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
+ SLEEP(2000);
+
+ TEST_LOG("Enable decryption again, you should hear the voice again\n");
+ TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
+ kCipherAes128CounterMode,
+ 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, encrKey));
+ SLEEP(2000);
+
+ TEST_LOG("Disabling encryption and enabling decryption, you should"
+ " hear nothing\n");
+ TEST_MUSTPASS(voe_encrypt_->DisableSRTPSend(0));
+ SLEEP(2000);
+
+ TEST_LOG("Back to normal\n");
+ // both SRTP sides are now inactive
+ TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
+ SLEEP(2000);
+
+ TEST_LOG("Enable SRTP and SRTCP encryption and decryption,"
+ " you should still hear the voice\n");
+ TEST_MUSTPASS(voe_encrypt_->EnableSRTPSend(0,
+ kCipherAes128CounterMode,
+ 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, encrKey, true));
+ TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
+ kCipherAes128CounterMode,
+ 30,
+ kAuthHmacSha1,
+ 20, 4, kEncryptionAndAuthentication, encrKey, true));
+ SLEEP(2000);
+
+ TEST_LOG("Back to normal\n");
+ TEST_MUSTPASS(voe_encrypt_->DisableSRTPSend(0));
+ // both SRTP sides are now inactive
+ TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
+ SLEEP(2000);
+
+#else
+ TEST_LOG("Skipping SRTP tests - WEBRTC_SRTP not defined \n");
+#endif // #ifdef WEBRTC_SRTP
+ TEST_LOG("\nExternal encryption tests:\n");
+ my_encryption * encObj = new my_encryption;
+ TEST_MUSTPASS(voe_encrypt_->RegisterExternalEncryption(0, *encObj));
+ TEST_LOG("Encryption enabled but you should still hear the voice\n");
+ SLEEP(2000);
+ TEST_LOG("Removing encryption object and deleting it\n");
+ TEST_MUSTPASS(voe_encrypt_->DeRegisterExternalEncryption(0));
+ delete encObj;
+ SLEEP(2000);
+#else
+ TEST_LOG("\n\n+++ Encryption tests NOT ENABLED +++\n");
+#endif // #ifdef _TEST_ENCRYPT_
+ //////////////////
+ // External media
+
+#ifdef _TEST_XMEDIA_
+ TEST_LOG("\n\n+++ External media tests +++\n\n");
+
+#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ TEST_LOG("Stop playing file as microphone \n");
+ TEST_LOG("==> Talk into the microphone \n");
+ TEST_MUSTPASS(voe_file_->StopPlayingFileAsMicrophone(0));
+
+ TEST_LOG("Enabling external playout\n");
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_xmedia_->SetExternalPlayoutStatus(true));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ TEST_LOG("Writing 2 secs of play data to vector\n");
+ int getLen;
+ WebRtc_Word16 speechData[32000];
+ for (int i = 0; i < 200; i++) {
+ TEST_MUSTPASS(voe_xmedia_->ExternalPlayoutGetData(speechData+i*160,
+ 16000,
+ 100,
+ getLen));
+ TEST_MUSTPASS(160 != getLen);
+ SLEEP(10);
+ }
+
+ TEST_LOG("Disabling external playout\n");
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_xmedia_->SetExternalPlayoutStatus(false));
+ TEST_MUSTPASS(voe_base_->StartPlayout(0));
+
+ TEST_LOG("Enabling external recording\n");
+ TEST_MUSTPASS(voe_xmedia_->SetExternalRecordingStatus(true));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ TEST_LOG("Inserting record data from vector\n");
+ for (int i = 0; i < 200; i++) {
+ TEST_MUSTPASS(voe_xmedia_->ExternalRecordingInsertData(speechData+i*160,
+ 160,
+ 16000,
+ 20));
+ SLEEP(10);
+ }
+
+ TEST_LOG("Disabling external recording\n");
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_xmedia_->SetExternalRecordingStatus(false));
+ TEST_MUSTPASS(voe_base_->StartSend(0));
+
+ TEST_LOG("==> Start playing a file as microphone again \n");
+ TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0, AudioFilename(),
+ true, true));
+#else
+ TEST_LOG("Skipping external rec and playout tests - \
+ WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT not defined \n");
+#endif // WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+ TEST_LOG("Enabling playout external media processing => "
+ "played audio should now be affected \n");
+ TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
+ -1, kPlaybackAllChannelsMixed, mobj));
+ SLEEP(2000);
+ TEST_LOG("Back to normal again \n");
+ TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
+ -1, kPlaybackAllChannelsMixed));
+ SLEEP(2000);
+ // Note that we must do per channel here because PlayFileAsMicrophone
+ // is only done on ch 0.
+ TEST_LOG("Enabling recording external media processing => "
+ "played audio should now be affected \n");
+ TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
+ 0, kRecordingPerChannel, mobj));
+ SLEEP(2000);
+ TEST_LOG("Back to normal again \n");
+ TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
+ 0, kRecordingPerChannel));
+ SLEEP(2000);
+ TEST_LOG("Enabling recording external media processing => "
+ "speak and make sure that voice is affected \n");
+ TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
+ -1, kRecordingAllChannelsMixed, mobj));
+ SLEEP(2000);
+ TEST_LOG("Back to normal again \n");
+ TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
+ -1, kRecordingAllChannelsMixed));
+ SLEEP(2000);
+#else
+ TEST_LOG("\n\n+++ External media tests NOT ENABLED +++\n");
+#endif // #ifdef _TEST_XMEDIA_
+ /////////////////////
+ // NetEQ statistics
+
+#ifdef _TEST_NETEQ_STATS_
+ TEST_LOG("\n\n+++ NetEQ statistics tests +++\n\n");
+
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ NetworkStatistics nStats;
+ TEST_MUSTPASS(voe_neteq_stats_->GetNetworkStatistics(0, nStats));
+ TEST_LOG("\nNetwork statistics: \n");
+ TEST_LOG(" currentAccelerateRate = %hu \n",
+ nStats.currentAccelerateRate);
+ TEST_LOG(" currentBufferSize = %hu \n",
+ nStats.currentBufferSize);
+ TEST_LOG(" currentDiscardRate = %hu \n",
+ nStats.currentDiscardRate);
+ TEST_LOG(" currentExpandRate = %hu \n",
+ nStats.currentExpandRate);
+ TEST_LOG(" currentPacketLossRate = %hu \n",
+ nStats.currentPacketLossRate);
+ TEST_LOG(" currentPreemptiveRate = %hu \n",
+ nStats.currentPreemptiveRate);
+ TEST_LOG(" preferredBufferSize = %hu \n",
+ nStats.preferredBufferSize);
+ TEST_LOG(" jitterPeaksFound = %i \n",
+ nStats.jitterPeaksFound);
+ TEST_LOG(" clockDriftPPM = %i \n",
+ nStats.clockDriftPPM);
+ TEST_LOG(" meanWaitingTimeMs = %i \n",
+ nStats.meanWaitingTimeMs);
+ TEST_LOG(" medianWaitingTimeMs = %i \n",
+ nStats.medianWaitingTimeMs);
+ TEST_LOG(" minWaitingTimeMs = %i \n",
+ nStats.minWaitingTimeMs);
+ TEST_LOG(" maxWaitingTimeMs = %i \n",
+ nStats.maxWaitingTimeMs);
+#else
+ TEST_LOG("Skipping NetEQ statistics tests - "
+ "WEBRTC_VOICE_ENGINE_NETEQ_STATS_API not defined \n");
+#endif // #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+#else
+ TEST_LOG("\n\n+++ NetEQ statistics tests NOT ENABLED +++\n");
+#endif // #ifdef _TEST_NETEQ_STATS_
+ //////////////////
+ // Stop streaming
+ TEST_LOG("\n\n+++ Stop streaming +++\n\n");
+
+ TEST_LOG("Stop playout, sending and listening \n");
+ TEST_MUSTPASS(voe_base_->StopPlayout(0));
+ TEST_MUSTPASS(voe_base_->StopSend(0));
+ TEST_MUSTPASS(voe_base_->StopReceive(0));
+
+ // Exit:
+ TEST_LOG("Delete channel and terminate VE \n");
+ TEST_MUSTPASS(voe_base_->DeleteChannel(0));
+ TEST_MUSTPASS(voe_base_->Terminate());
+
+ return 0;
+}
+
+int runAutoTest(TestType testType, ExtendedSelection extendedSel) {
+ SubAPIManager apiMgr;
+ apiMgr.DisplayStatus();
+
+ ////////////////////////////////////
+ // Create VoiceEngine and sub API:s
+
+ voetest::VoETestManager tm;
+ if (!tm.Init()) {
+ return -1;
+ }
+ tm.GetInterfaces();
+
+ //////////////////////
+ // Run standard tests
+
+ int mainRet(-1);
+ if (testType == Standard) {
+ mainRet = tm.DoStandardTest();
+
+ ////////////////////////////////
+ // Create configuration summary
+ TEST_LOG("\n\n+++ Creating configuration summary file +++\n");
+ createSummary(tm.VoiceEnginePtr());
+ } else if (testType == Extended) {
+ VoEExtendedTest xtend(tm);
+
+ mainRet = 0;
+ while (extendedSel != XSEL_None) {
+ if (extendedSel == XSEL_Base || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestBase()) == -1)
+ break;
+ xtend.TestPassed("Base");
+ }
+ if (extendedSel == XSEL_CallReport || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestCallReport()) == -1)
+ break;
+ xtend.TestPassed("CallReport");
+ }
+ if (extendedSel == XSEL_Codec || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestCodec()) == -1)
+ break;
+ xtend.TestPassed("Codec");
+ }
+ if (extendedSel == XSEL_DTMF || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestDtmf()) == -1)
+ break;
+ xtend.TestPassed("Dtmf");
+ }
+ if (extendedSel == XSEL_Encryption || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestEncryption()) == -1)
+ break;
+ xtend.TestPassed("Encryption");
+ }
+ if (extendedSel == XSEL_ExternalMedia || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestExternalMedia()) == -1)
+ break;
+ xtend.TestPassed("ExternalMedia");
+ }
+ if (extendedSel == XSEL_File || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestFile()) == -1)
+ break;
+ xtend.TestPassed("File");
+ }
+ if (extendedSel == XSEL_Mixing || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestMixing()) == -1)
+ break;
+ xtend.TestPassed("Mixing");
+ }
+ if (extendedSel == XSEL_Hardware || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestHardware()) == -1)
+ break;
+ xtend.TestPassed("Hardware");
+ }
+ if (extendedSel == XSEL_NetEqStats || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestNetEqStats()) == -1)
+ break;
+ xtend.TestPassed("NetEqStats");
+ }
+ if (extendedSel == XSEL_Network || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestNetwork()) == -1)
+ break;
+ xtend.TestPassed("Network");
+ }
+ if (extendedSel == XSEL_RTP_RTCP || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestRTP_RTCP()) == -1)
+ break;
+ xtend.TestPassed("RTP_RTCP");
+ }
+ if (extendedSel == XSEL_VideoSync || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestVideoSync()) == -1)
+ break;
+ xtend.TestPassed("VideoSync");
+ }
+ if (extendedSel == XSEL_VolumeControl || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestVolumeControl()) == -1)
+ break;
+ xtend.TestPassed("VolumeControl");
+ }
+ if (extendedSel == XSEL_AudioProcessing || extendedSel == XSEL_All) {
+ if ((mainRet = xtend.TestAPM()) == -1)
+ break;
+ xtend.TestPassed("AudioProcessing");
+ }
+ apiMgr.GetExtendedMenuSelection(extendedSel);
+ } // while (extendedSel != XSEL_None)
+ } else if (testType == Stress) {
+ VoEStressTest stressTest(tm);
+ mainRet = stressTest.DoTest();
+ } else if (testType == Unit) {
+ VoEUnitTest unitTest(tm);
+ mainRet = unitTest.DoTest();
+ } else if (testType == CPU) {
+ VoECpuTest cpuTest(tm);
+ mainRet = cpuTest.DoTest();
+ } else {
+ // Should never end up here
+ TEST_LOG("INVALID SELECTION \n");
+ }
+
+ //////////////////
+ // Release/Delete
+
+ int releaseOK = tm.ReleaseInterfaces();
+
+ if ((0 == mainRet) && (releaseOK != -1)) {
+ TEST_LOG("\n\n*** All tests passed *** \n\n");
+ } else {
+ TEST_LOG("\n\n*** Test failed! *** \n");
+ }
+
+ return 0;
+}
+
+void createSummary(VoiceEngine* ve) {
+ int len;
+ char str[256];
+
+#ifdef MAC_IPHONE
+ char summaryFilename[256];
+ GetDocumentsDir(summaryFilename, 256);
+ strcat(summaryFilename, "/summary.txt");
+#endif
+
+ VoEBase* voe_base_ = VoEBase::GetInterface(ve);
+ FILE* stream = fopen(summaryFilename, "wt");
+
+ sprintf(str, "WebRTc VoiceEngine ");
+#if defined(_WIN32)
+ strcat(str, "Win");
+#elif defined(WEBRTC_LINUX) && defined(WEBRTC_TARGET_PC) && \
+ !defined(WEBRTC_ANDROID)
+ strcat(str, "Linux");
+#elif defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
+ strcat(str, "Mac");
+#elif defined(WEBRTC_ANDROID)
+ strcat(str, "Android");
+#elif defined(MAC_IPHONE)
+ strcat(str, "iPhone");
+#endif
+ // Add for other platforms as needed
+
+ fprintf(stream, "%s\n", str);
+ len = (int) strlen(str);
+ for (int i = 0; i < len; i++) {
+ fprintf(stream, "=");
+ }
+ fprintf(stream, "\n\n");
+
+ char version[1024];
+ char veVersion[24];
+ voe_base_->GetVersion(version);
+ // find first NL <=> end of VoiceEngine version string
+ int pos = (int) strcspn(version, "\n");
+ strncpy(veVersion, version, pos);
+ veVersion[pos] = '\0';
+ sprintf(str, "Version: %s\n", veVersion);
+ fprintf(stream, "%s\n", str);
+
+ sprintf(str, "Build date & time: %s\n", BUILDDATE " " BUILDTIME);
+ fprintf(stream, "%s\n", str);
+
+ strcpy(str, "G.711 A-law");
+ fprintf(stream, "\nSupported codecs: %s\n", str);
+ strcpy(str, " G.711 mu-law");
+ fprintf(stream, "%s\n", str);
+#ifdef WEBRTC_CODEC_EG711
+ strcpy(str, " Enhanced G.711 A-law");
+ fprintf(stream, "%s\n", str);
+ strcpy(str, " Enhanced G.711 mu-law");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_IPCMWB
+ strcpy(str, " iPCM-wb");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ strcpy(str, " iLBC");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+ strcpy(str, " iSAC");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_ISACLC
+ strcpy(str, " iSAC-LC");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G722
+ strcpy(str, " G.722");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G722_1
+ strcpy(str, " G.722.1");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G722_1C
+ strcpy(str, " G.722.1C");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G723
+ strcpy(str, " G.723");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G726
+ strcpy(str, " G.726");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G729
+ strcpy(str, " G.729");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_G729_1
+ strcpy(str, " G.729.1");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_GSMFR
+ strcpy(str, " GSM-FR");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_GSMAMR
+ strcpy(str, " AMR");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_GSMAMRWB
+ strcpy(str, " AMR-WB");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_GSMEFR
+ strcpy(str, " GSM-EFR");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_SPEEX
+ strcpy(str, " Speex");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_SILK
+ strcpy(str, " Silk");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ strcpy(str, " L16");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef NETEQFIX_VOXWARE_SC3
+ strcpy(str, " Voxware SC3");
+ fprintf(stream, "%s\n", str);
+#endif
+ // Always included
+ strcpy(str, " AVT (RFC2833)");
+ fprintf(stream, "%s\n", str);
+#ifdef WEBRTC_CODEC_RED
+ strcpy(str, " RED (forward error correction)");
+ fprintf(stream, "%s\n", str);
+#endif
+
+ fprintf(stream, "\nEcho Control: ");
+#ifdef WEBRTC_VOICE_ENGINE_ECHO
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ fprintf(stream, "\nAutomatic Gain Control: ");
+#ifdef WEBRTC_VOICE_ENGINE_AGC
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ fprintf(stream, "\nNoise Reduction: ");
+#ifdef WEBRTC_VOICE_ENGINE_NR
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ fprintf(stream, "\nSRTP: ");
+#ifdef WEBRTC_SRTP
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ fprintf(stream, "\nExternal transport only: ");
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ fprintf(stream, "\nTelephone event detection: ");
+#ifdef WEBRTC_DTMF_DETECTION
+ fprintf(stream, "Yes\n");
+#else
+ fprintf(stream, "No\n");
+#endif
+
+ strcpy(str, "VoEBase");
+ fprintf(stream, "\nSupported sub-APIs: %s\n", str);
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+ strcpy(str, " VoECodec");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+ strcpy(str, " VoEDtmf");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+ strcpy(str, " VoEFile");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+ strcpy(str, " VoEHardware");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+ strcpy(str, " VoENetwork");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+ strcpy(str, " VoERTP_RTCP");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+ strcpy(str, " VoEVolumeControl");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+ strcpy(str, " VoEAudioProcessing");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+ strcpy(str, " VoeExternalMedia");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ strcpy(str, " VoENetEqStats");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+ strcpy(str, " VoEEncryption");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+ strcpy(str, " VoECallReport");
+ fprintf(stream, "%s\n", str);
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ strcpy(str, " VoEVideoSync");
+ fprintf(stream, "%s\n", str);
+#endif
+
+ fclose(stream);
+ voe_base_->Release();
+}
+
+/*********Knowledge Base******************/
+
+//An example for creating threads and calling VE API's from that thread.
+// Using thread. A generic API/Class for all platforms.
+#ifdef THEADTEST // find first NL <=> end of VoiceEngine version string
+//Definition of Thread Class
+class ThreadTest
+{
+public:
+ ThreadTest(
+ VoEBase* voe_base_);
+ ~ThreadTest()
+ {
+ delete _myThread;
+ }
+ void Stop();
+private:
+ static bool StartSend(
+ void* obj);
+ bool StartSend();
+
+ ThreadWrapper* _myThread;
+ VoEBase* _base;
+
+ bool _stopped;
+};
+
+//Main function from where StartSend is invoked as a seperate thread.
+ThreadTest::ThreadTest(
+ VoEBase* voe_base_)
+:
+_stopped(false),
+_base(voe_base_)
+{
+ //Thread Creation
+ _myThread = ThreadWrapper::CreateThread(StartSend, this, kLowPriority);
+ unsigned int id = 0;
+ //Starting the thread
+ _myThread->Start(id);
+}
+
+//Calls the StartSend. This is to avoid the static declaration issue.
+bool
+ThreadTest::StartSend(
+ void* obj)
+{
+ return ((ThreadTest*)obj)->StartSend();
+}
+
+bool
+ThreadTest::StartSend()
+{
+ _myThread->SetNotAlive(); //Ensures this function is called only once.
+ _base->StartSend(0);
+ return true;
+}
+
+void ThreadTest::Stop()
+{
+ _stopped = true;
+}
+
+// Use the following to invoke ThreatTest from the main function.
+// ThreadTest* threadtest = new ThreadTest(voe_base_);
+#endif
+
+// An example to create a thread and call VE API's call from that thread.
+// Specific to Windows Platform
+#ifdef THREAD_TEST_WINDOWS
+//Thread Declaration. Need to be added in the class controlling/dictating
+// the main code.
+/**
+ private:
+ static unsigned int WINAPI StartSend(void* obj);
+ unsigned int WINAPI StartSend();
+ **/
+
+//Thread Definition
+unsigned int WINAPI mainTest::StartSend(void *obj)
+{
+ return ((mainTest*)obj)->StartSend();
+}
+unsigned int WINAPI mainTest::StartSend()
+{
+ //base
+ voe_base_->StartSend(0);
+
+ // TEST_MUSTPASS(voe_base_->StartSend(0));
+ TEST_LOG("hi hi hi");
+ return 0;
+}
+
+//Thread invoking. From the main code
+/*****
+ unsigned int threadID=0;
+ if ((HANDLE)_beginthreadex(NULL,
+ 0,
+ StartSend,
+ (void*)this,
+ 0,
+ &threadID) == NULL)
+ return false;
+ ****/
+
+#endif
+
+} // namespace voetest
+
+int RunInManualMode(int argc, char** argv) {
+ using namespace voetest;
+
+ SubAPIManager apiMgr;
+ apiMgr.DisplayStatus();
+
+ printf("----------------------------\n");
+ printf("Select type of test\n\n");
+ printf(" (0) Quit\n");
+ printf(" (1) Standard test\n");
+ printf(" (2) Extended test(s)...\n");
+ printf(" (3) Stress test(s)...\n");
+ printf(" (4) Unit test(s)...\n");
+ printf(" (5) CPU & memory reference test [Windows]...\n");
+ printf("\n: ");
+
+ int selection(0);
+
+ dummy = scanf("%d", &selection);
+
+ ExtendedSelection extendedSel(XSEL_Invalid);
+
+ enum TestType testType(Invalid);
+
+ switch (selection) {
+ case 0:
+ return 0;
+ case 1:
+ testType = Standard;
+ break;
+ case 2:
+ testType = Extended;
+ while (!apiMgr.GetExtendedMenuSelection(extendedSel))
+ continue;
+ break;
+ case 3:
+ testType = Stress;
+ break;
+ case 4:
+ testType = Unit;
+ break;
+ case 5:
+ testType = CPU;
+ break;
+ default:
+ TEST_LOG("Invalid selection!\n");
+ return 0;
+ }
+
+ if (testType == Standard) {
+ TEST_LOG("\n\n+++ Running gtest-rewritten standard tests first +++\n\n");
+
+ // Run the automated tests too in standard mode since we are gradually
+ // rewriting the standard test to be automated. Running this will give
+ // the standard suite the same completeness.
+ RunInAutomatedMode(argc, argv);
+ }
+
+ // Function that can be called from other entry functions.
+ return runAutoTest(testType, extendedSel);
+}
+
+// ----------------------------------------------------------------------------
+// main
+// ----------------------------------------------------------------------------
+
+#if !defined(MAC_IPHONE)
+int main(int argc, char** argv) {
+ if (argc > 1 && std::string(argv[1]) == "--automated") {
+ // This function is defined in automated_mode.cc to avoid macro clashes
+ // with googletest (for instance the ASSERT_TRUE macro).
+ return RunInAutomatedMode(argc, argv);
+ }
+
+ return RunInManualMode(argc, argv);
+}
+#endif //#if !defined(MAC_IPHONE)
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h
new file mode 100644
index 0000000..ae0fb24
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h
@@ -0,0 +1,370 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
+#define WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
+
+#include <stdio.h>
+#include <string>
+
+#include "resource_manager.h"
+#include "voe_audio_processing.h"
+#include "voe_base.h"
+#include "voe_dtmf.h"
+#include "voe_errors.h"
+#include "voe_file.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_test_defines.h"
+#include "voe_test_interface.h"
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+#include "voe_call_report.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+#include "voe_codec.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+#include "voe_encryption.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+#include "voe_external_media.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+#include "voe_hardware.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+#include "voe_network.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+#include "voe_video_sync.h"
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+#include "voe_volume_control.h"
+#endif
+
+#ifdef _TEST_NETEQ_STATS_
+namespace webrtc {
+class CriticalSectionWrapper;
+class ThreadWrapper;
+class VoENetEqStats;
+}
+#endif
+
+#if defined(WEBRTC_ANDROID)
+extern char mobileLogMsg[640];
+#endif
+
+namespace voetest {
+
+void createSummary(VoiceEngine* ve);
+void prepareDelivery();
+
+class TestRtpObserver : public VoERTPObserver {
+ public:
+ TestRtpObserver();
+ virtual ~TestRtpObserver();
+ virtual void OnIncomingCSRCChanged(const int channel,
+ const unsigned int CSRC,
+ const bool added);
+ virtual void OnIncomingSSRCChanged(const int channel,
+ const unsigned int SSRC);
+ void Reset();
+ public:
+ unsigned int ssrc_[2];
+ unsigned int csrc_[2][2]; // Stores 2 CSRCs for each channel.
+ bool added_[2][2];
+ int size_[2];
+};
+
+class MyTraceCallback : public TraceCallback {
+ public:
+ void Print(const TraceLevel level, const char *traceString, const int length);
+};
+
+class MyDeadOrAlive : public VoEConnectionObserver {
+ public:
+ void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+};
+
+class ErrorObserver : public VoiceEngineObserver {
+ public:
+ ErrorObserver();
+ void CallbackOnError(const int channel, const int errCode);
+ public:
+ int code;
+};
+
+class RtcpAppHandler : public VoERTCPObserver {
+ public:
+ void OnApplicationDataReceived(const int channel,
+ const unsigned char sub_type,
+ const unsigned int name,
+ const unsigned char* data,
+ const unsigned short length_in_bytes);
+ void Reset();
+ ~RtcpAppHandler() {}
+ unsigned short length_in_bytes_;
+ unsigned char data_[256];
+ unsigned char sub_type_;
+ unsigned int name_;
+};
+
+class DtmfCallback : public VoETelephoneEventObserver {
+ public:
+ int counter;
+ DtmfCallback() {
+ counter = 0;
+ }
+ virtual void OnReceivedTelephoneEventInband(int channel,
+ int eventCode,
+ bool endOfEvent) {
+ char msg[128];
+ if (endOfEvent)
+ sprintf(msg, "(event=%d, [END])", eventCode);
+ else
+ sprintf(msg, "(event=%d, [START])", eventCode);
+ TEST_LOG("%s", msg);
+ if (!endOfEvent)
+ counter++; // cound start of event only
+ fflush(NULL);
+ }
+
+ virtual void OnReceivedTelephoneEventOutOfBand(int channel,
+ int eventCode,
+ bool endOfEvent) {
+ char msg[128];
+ if (endOfEvent)
+ sprintf(msg, "(event=%d, [END])", eventCode);
+ else
+ sprintf(msg, "(event=%d, [START])", eventCode);
+ TEST_LOG("%s", msg);
+ if (!endOfEvent)
+ counter++; // cound start of event only
+ fflush(NULL);
+ }
+};
+
+class my_encryption : public Encryption {
+ void encrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void decrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void encrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void decrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+};
+
+class RxCallback : public VoERxVadCallback {
+ public:
+ RxCallback() :
+ _vadDecision(-1) {
+ }
+
+ virtual void OnRxVad(int, int vadDecision) {
+ char msg[128];
+ sprintf(msg, "RX VAD detected decision %d \n", vadDecision);
+ TEST_LOG("%s", msg);
+ _vadDecision = vadDecision;
+ }
+
+ int _vadDecision;
+};
+
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+class MyMedia : public VoEMediaProcess {
+ public:
+ virtual void Process(const int channel,
+ const ProcessingTypes type,
+ WebRtc_Word16 audio_10ms[],
+ const int length,
+ const int samplingFreqHz,
+ const bool stereo);
+ private:
+ int f;
+};
+#endif
+
+class SubAPIManager {
+ public:
+ SubAPIManager()
+ : _base(true),
+ _callReport(false),
+ _codec(false),
+ _dtmf(false),
+ _encryption(false),
+ _externalMedia(false),
+ _file(false),
+ _hardware(false),
+ _netEqStats(false),
+ _network(false),
+ _rtp_rtcp(false),
+ _videoSync(false),
+ _volumeControl(false),
+ _apm(false),
+ _xsel(XSEL_Invalid) {
+#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+ _callReport = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
+ _codec = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
+ _dtmf = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+ _encryption = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+ _externalMedia = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_FILE_API
+ _file = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
+ _hardware = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+ _netEqStats = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
+ _network = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+ _rtp_rtcp = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+ _videoSync = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+ _volumeControl = true;
+#endif
+#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+ _apm = true;
+#endif
+ }
+
+ void DisplayStatus() const;
+ bool GetExtendedMenuSelection(ExtendedSelection& sel);
+
+ private:
+ bool _base, _callReport, _codec, _dtmf, _encryption;
+ bool _externalMedia, _file, _hardware;
+ bool _netEqStats, _network, _rtp_rtcp, _videoSync, _volumeControl, _apm;
+ ExtendedSelection _xsel;
+};
+
+class VoETestManager {
+ public:
+ VoETestManager();
+ ~VoETestManager();
+
+ // Must be called after construction.
+ bool Init();
+
+ void GetInterfaces();
+ int ReleaseInterfaces();
+ int DoStandardTest();
+
+ const char* AudioFilename() const {
+ const std::string& result = resource_manager_.long_audio_file_path();
+ if (result.length() == 0) {
+ TEST_LOG("ERROR: Failed to open input file!");
+ }
+ return result.c_str();
+ }
+
+ VoiceEngine* VoiceEnginePtr() const {
+ return voice_engine_;
+ }
+ VoEBase* BasePtr() const {
+ return voe_base_;
+ }
+ VoECodec* CodecPtr() const {
+ return voe_codec_;
+ }
+ VoEVolumeControl* VolumeControlPtr() const {
+ return voe_volume_control_;
+ }
+ VoEDtmf* DtmfPtr() const {
+ return voe_dtmf_;
+ }
+ VoERTP_RTCP* RTP_RTCPPtr() const {
+ return voe_rtp_rtcp_;
+ }
+ VoEAudioProcessing* APMPtr() const {
+ return voe_apm_;
+ }
+
+ VoENetwork* NetworkPtr() const {
+ return voe_network_;
+ }
+
+ VoEFile* FilePtr() const {
+ return voe_file_;
+ }
+
+ VoEHardware* HardwarePtr() const {
+ return voe_hardware_;
+ }
+
+ VoEVideoSync* VideoSyncPtr() const {
+ return voe_vsync_;
+ }
+
+ VoEEncryption* EncryptionPtr() const {
+ return voe_encrypt_;
+ }
+
+ VoEExternalMedia* ExternalMediaPtr() const {
+ return voe_xmedia_;
+ }
+
+ VoECallReport* CallReportPtr() const {
+ return voe_call_report_;
+ }
+
+#ifdef _TEST_NETEQ_STATS_
+ VoENetEqStats* NetEqStatsPtr() const {
+ return voe_neteq_stats_;
+ }
+
+#endif
+
+ private:
+ int SetUp(ErrorObserver* error_observer);
+ int TestStartStreaming(FakeExternalTransport& channel0_transport);
+ int TestStartPlaying();
+
+ bool initialized_;
+
+ VoiceEngine* voice_engine_;
+ VoEBase* voe_base_;
+ VoECallReport* voe_call_report_;
+ VoECodec* voe_codec_;
+ VoEDtmf* voe_dtmf_;
+ VoEEncryption* voe_encrypt_;
+ VoEExternalMedia* voe_xmedia_;
+ VoEFile* voe_file_;
+ VoEHardware* voe_hardware_;
+ VoENetwork* voe_network_;
+#ifdef _TEST_NETEQ_STATS_
+ VoENetEqStats* voe_neteq_stats_;
+#endif
+ VoERTP_RTCP* voe_rtp_rtcp_;
+ VoEVideoSync* voe_vsync_;
+ VoEVolumeControl* voe_volume_control_;
+ VoEAudioProcessing* voe_apm_;
+
+ ResourceManager resource_manager_;
+};
+
+} // namespace voetest
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc
new file mode 100644
index 0000000..b7098ec
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc
@@ -0,0 +1,415 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Some ideas of improvements:
+// Break out common init and maybe terminate to separate function(s).
+// How much trace should we have enabled?
+// API error counter, to print info and return -1 if any error.
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include <cassert>
+#if defined(_WIN32)
+#include <conio.h>
+#endif
+
+#include "voe_stress_test.h"
+#include "voe_standard_test.h"
+
+#include "../../source/voice_engine_defines.h" // defines build macros
+#include "thread_wrapper.h"
+
+using namespace webrtc;
+
+namespace voetest {
+
+#define VALIDATE_STRESS(expr) \
+ if (expr) \
+ { \
+ printf("Error at line: %i, %s \n", __LINE__, #expr); \
+ printf("Error code: %i \n", base->LastError()); \
+ }
+
+#ifdef _WIN32
+// Pause if supported
+#define PAUSE_OR_SLEEP(x) PAUSE;
+#else
+// Sleep a bit instead if pause not supported
+#define PAUSE_OR_SLEEP(x) SLEEP(x);
+#endif
+
+extern char* GetFilename(char* filename);
+extern const char* GetFilename(const char* filename);
+extern int GetResource(char* resource, char* dest, int destLen);
+extern char* GetResource(char* resource);
+extern const char* GetResource(const char* resource);
+
+const char* VoEStressTest::_key = "====YUtFWRAAAAADBtIHgAAAAAEAAAAcAAAAAQBHU0ds"
+ "b2JhbCBJUCBTb3VuZAAC\nAAAAIwAAAExpY2Vuc2VkIHRvIE5vcnRlbCBOZXR3cm9rcwAAAAA"
+ "xAAAAZxZ7/u0M\niFYyTwSwko5Uutf7mh8S0O4rYZYTFidbzQeuGonuL17F/2oD/2pfDp3jL4"
+ "Rf3z/A\nnlJsEJgEtASkDNFuwLILjGY0pzjjAYQp3pCl6z6k2MtE06AirdjGLYCjENpq/opX"
+ "\nOrs3sIuwdYK5va/aFcsjBDmlsGCUM48RDYG9s23bIHYafXUC4ofOaubbZPWiPTmL\nEVJ8WH"
+ "4F9pgNjALc14oJXfON7r/3\n=EsLx";
+
+int VoEStressTest::DoTest() {
+ int test(-1);
+ while (test != 0) {
+ test = MenuSelection();
+ switch (test) {
+ case 0:
+ // Quit stress test
+ break;
+ case 1:
+ // All tests
+ StartStopTest();
+ CreateDeleteChannelsTest();
+ MultipleThreadsTest();
+ break;
+ case 2:
+ StartStopTest();
+ break;
+ case 3:
+ CreateDeleteChannelsTest();
+ break;
+ case 4:
+ MultipleThreadsTest();
+ break;
+ default:
+ // Should not be possible
+ printf("Invalid selection! (Test code error)\n");
+ assert(false);
+ } // switch
+ } // while
+
+ return 0;
+}
+
+int VoEStressTest::MenuSelection() {
+ printf("------------------------------------------------\n");
+ printf("Select stress test\n\n");
+ printf(" (0) Quit\n");
+ printf(" (1) All\n");
+ printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
+ printf(" (2) Start/stop\n");
+ printf(" (3) Create/delete channels\n");
+ printf(" (4) Multiple threads\n");
+
+ const int maxMenuSelection = 4;
+ int selection(-1);
+
+ while ((selection < 0) || (selection > maxMenuSelection)) {
+ printf("\n: ");
+ int retval = scanf("%d", &selection);
+ if ((retval != 1) || (selection < 0) || (selection > maxMenuSelection)) {
+ printf("Invalid selection!\n");
+ }
+ }
+
+ return selection;
+}
+
+int VoEStressTest::StartStopTest() {
+ printf("------------------------------------------------\n");
+ printf("Running start/stop test\n");
+ printf("------------------------------------------------\n");
+
+ printf("\nNOTE: this thest will fail after a while if Core audio is used\n");
+ printf("because MS returns AUDCLNT_E_CPUUSAGE_EXCEEDED (VoE Error 10013).\n");
+
+ // Get sub-API pointers
+ VoEBase* base = _mgr.BasePtr();
+
+ // Set trace
+ // VALIDATE_STRESS(base->SetTraceFileName(
+ // GetFilename("VoEStressTest_StartStop_trace.txt")));
+ // VALIDATE_STRESS(base->SetDebugTraceFileName(
+ // GetFilename("VoEStressTest_StartStop_trace_debug.txt")));
+ // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
+ // kTraceWarning | kTraceError |
+ // kTraceCritical | kTraceApiCall |
+ // kTraceMemory | kTraceInfo));
+ VALIDATE_STRESS(base->Init());
+ VALIDATE_STRESS(base->CreateChannel());
+
+ ///////////// Start test /////////////
+
+ int numberOfLoops(2000);
+ int loopSleep(200);
+ int i(0);
+ int markInterval(20);
+
+ printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
+ numberOfLoops, loopSleep, markInterval);
+ printf("Test will take approximately %d minutes. \n",
+ numberOfLoops * loopSleep / 1000 / 60 + 1);
+
+ for (i = 0; i < numberOfLoops; ++i) {
+ VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
+ VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+ VALIDATE_STRESS(base->StartReceive(0));
+ VALIDATE_STRESS(base->StartPlayout(0));
+ VALIDATE_STRESS(base->StartSend(0));
+ if (!(i % markInterval))
+ MARK();
+ SLEEP(loopSleep);
+ VALIDATE_STRESS(base->StopSend(0));
+ VALIDATE_STRESS(base->StopPlayout(0));
+ VALIDATE_STRESS(base->StopReceive(0));
+ }
+ ANL();
+
+ VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
+ VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+ VALIDATE_STRESS(base->StartReceive(0));
+ VALIDATE_STRESS(base->StartPlayout(0));
+ VALIDATE_STRESS(base->StartSend(0));
+ printf("Verify that audio is good. \n");
+ PAUSE_OR_SLEEP(20000);
+ VALIDATE_STRESS(base->StopSend(0));
+ VALIDATE_STRESS(base->StopPlayout(0));
+ VALIDATE_STRESS(base->StopReceive(0));
+
+ ///////////// End test /////////////
+
+
+ // Terminate
+ VALIDATE_STRESS(base->DeleteChannel(0));
+ VALIDATE_STRESS(base->Terminate());
+
+ printf("Test finished \n");
+
+ return 0;
+}
+
+int VoEStressTest::CreateDeleteChannelsTest() {
+ printf("------------------------------------------------\n");
+ printf("Running create/delete channels test\n");
+ printf("------------------------------------------------\n");
+
+ // Get sub-API pointers
+ VoEBase* base = _mgr.BasePtr();
+
+ // Set trace
+ // VALIDATE_STRESS(base->SetTraceFileName(
+ // GetFilename("VoEStressTest_CreateChannels_trace.txt")));
+ // VALIDATE_STRESS(base->SetDebugTraceFileName(
+ // GetFilename("VoEStressTest_CreateChannels_trace_debug.txt")));
+ // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
+ // kTraceWarning | kTraceError |
+ // kTraceCritical | kTraceApiCall |
+ // kTraceMemory | kTraceInfo));
+ VALIDATE_STRESS(base->Init());
+
+ ///////////// Start test /////////////
+
+ int numberOfLoops(10000);
+ int loopSleep(10);
+ int i(0);
+ int markInterval(200);
+
+ printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
+ numberOfLoops, loopSleep, markInterval);
+ printf("Test will take approximately %d minutes. \n",
+ numberOfLoops * loopSleep / 1000 / 60 + 1);
+
+ // Some possible extensions include:
+ // Different sleep times (fixed or random) or zero.
+ // Start call on all or some channels.
+ // Two parts: first have a slight overweight to creating channels,
+ // then to deleting. (To ensure we hit max channels and go to zero.)
+ // Make sure audio is OK after test has finished.
+
+ // Set up, start with maxChannels/2 channels
+ const int maxChannels = base->MaxNumOfChannels();
+ VALIDATE_STRESS(maxChannels < 1); // Should always have at least one channel
+ bool* channelState = new bool[maxChannels];
+ memset(channelState, 0, maxChannels * sizeof(bool));
+ int channel(0);
+ int noOfActiveChannels(0);
+ for (i = 0; i < (maxChannels / 2); ++i) {
+ channel = base->CreateChannel();
+ VALIDATE_STRESS(channel < 0);
+ if (channel >= 0) {
+ channelState[channel] = true;
+ ++noOfActiveChannels;
+ }
+ }
+ srand((unsigned int) time(NULL));
+ bool action(false);
+ double rnd(0.0);
+ int res(0);
+
+ // Create/delete channels with slight
+ for (i = 0; i < numberOfLoops; ++i) {
+ // Randomize action (create or delete channel)
+ action = rand() <= (RAND_MAX / 2);
+ if (action) {
+ if (noOfActiveChannels < maxChannels) {
+ // Create new channel
+ channel = base->CreateChannel();
+ VALIDATE_STRESS(channel < 0);
+ if (channel >= 0) {
+ channelState[channel] = true;
+ ++noOfActiveChannels;
+ }
+ }
+ } else {
+ if (noOfActiveChannels > 0) {
+ // Delete random channel that's created [0, maxChannels - 1]
+ do {
+ rnd = static_cast<double> (rand());
+ channel = static_cast<int> (rnd /
+ (static_cast<double> (RAND_MAX) + 1.0f) *
+ maxChannels);
+ } while (!channelState[channel]); // Must find a created channel
+
+ res = base->DeleteChannel(channel);
+ VALIDATE_STRESS(0 != res);
+ if (0 == res) {
+ channelState[channel] = false;
+ --noOfActiveChannels;
+ }
+ }
+ }
+
+ if (!(i % markInterval))
+ MARK();
+ SLEEP(loopSleep);
+ }
+ ANL();
+
+ delete[] channelState;
+
+ ///////////// End test /////////////
+
+
+ // Terminate
+ VALIDATE_STRESS(base->Terminate()); // Deletes all channels
+
+ printf("Test finished \n");
+
+ return 0;
+}
+
+int VoEStressTest::MultipleThreadsTest() {
+ printf("------------------------------------------------\n");
+ printf("Running multiple threads test\n");
+ printf("------------------------------------------------\n");
+
+ // Get sub-API pointers
+ VoEBase* base = _mgr.BasePtr();
+
+ // Set trace
+ // VALIDATE_STRESS(base->SetTraceFileName(
+ // GetFilename("VoEStressTest_MultipleThreads_trace.txt")));
+ // VALIDATE_STRESS(base->SetDebugTraceFileName(
+ // GetFilename("VoEStressTest_MultipleThreads_trace_debug.txt")));
+ // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
+ // kTraceWarning | kTraceError |
+ // kTraceCritical | kTraceApiCall |
+ // kTraceMemory | kTraceInfo));
+
+ // Init
+ VALIDATE_STRESS(base->Init());
+ VALIDATE_STRESS(base->CreateChannel());
+
+ ///////////// Start test /////////////
+
+ int numberOfLoops(10000);
+ int loopSleep(0);
+ int i(0);
+ int markInterval(1000);
+
+ printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
+ numberOfLoops, loopSleep, markInterval);
+ printf("Test will take approximately %d minutes. \n",
+ numberOfLoops * loopSleep / 1000 / 60 + 1);
+
+ srand((unsigned int) time(NULL));
+ int rnd(0);
+
+ // Start extra thread
+ const char* threadName = "StressTest Extra API Thread";
+ _ptrExtraApiThread = ThreadWrapper::CreateThread(RunExtraApi, this,
+ kNormalPriority, threadName);
+ unsigned int id(0);
+ VALIDATE_STRESS(!_ptrExtraApiThread->Start(id));
+
+ // Some possible extensions include:
+ // Add more API calls to randomize
+ // More threads
+ // Different sleep times (fixed or random).
+ // Make sure audio is OK after test has finished.
+
+ // Call random API functions here and in extra thread, ignore any error
+ for (i = 0; i < numberOfLoops; ++i) {
+ // This part should be equal to the marked part in the extra thread
+ // --- BEGIN ---
+ rnd = rand();
+ if (rnd < (RAND_MAX / 2)) {
+ // Start playout
+ base->StartPlayout(0);
+ } else {
+ // Stop playout
+ base->StopPlayout(0);
+ }
+ // --- END ---
+
+ if (!(i % markInterval))
+ MARK();
+ SLEEP(loopSleep);
+ }
+ ANL();
+
+ // Stop extra thread
+ VALIDATE_STRESS(!_ptrExtraApiThread->Stop());
+ delete _ptrExtraApiThread;
+
+ ///////////// End test /////////////
+
+ // Terminate
+ VALIDATE_STRESS(base->Terminate()); // Deletes all channels
+
+ printf("Test finished \n");
+
+ return 0;
+}
+
+// Thread functions
+
+bool VoEStressTest::RunExtraApi(void* ptr) {
+ return static_cast<VoEStressTest*> (ptr)->ProcessExtraApi();
+}
+
+bool VoEStressTest::ProcessExtraApi() {
+ // Prepare
+ VoEBase* base = _mgr.BasePtr();
+ int rnd(0);
+
+ // Call random API function, ignore any error
+
+ // This part should be equal to the marked part in the main thread
+ // --- BEGIN ---
+ rnd = rand();
+ if (rnd < (RAND_MAX / 2)) {
+ // Start playout
+ base->StartPlayout(0);
+ } else {
+ // Stop playout
+ base->StopPlayout(0);
+ }
+ // --- END ---
+
+ return true;
+}
+
+} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h
new file mode 100644
index 0000000..b3a418c
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
+#define WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
+
+namespace webrtc {
+class ThreadWrapper;
+}
+
+namespace voetest {
+// TODO(andrew): using directives are not permitted.
+using namespace webrtc;
+
+class VoETestManager;
+
+class VoEStressTest {
+ public:
+ VoEStressTest(VoETestManager& mgr) :
+ _mgr(mgr), _ptrExtraApiThread(NULL) {
+ }
+ ~VoEStressTest() {
+ }
+ int DoTest();
+
+ private:
+ int MenuSelection();
+ int StartStopTest();
+ int CreateDeleteChannelsTest();
+ int MultipleThreadsTest();
+
+ static bool RunExtraApi(void* ptr);
+ bool ProcessExtraApi();
+
+ VoETestManager& _mgr;
+ static const char* _key;
+
+ ThreadWrapper* _ptrExtraApiThread;
+};
+
+} // namespace voetest
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h b/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h
new file mode 100644
index 0000000..c9c103a
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
+#define WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
+
+// Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags
+#include "engine_configurations.h"
+
+#ifdef WEBRTC_ANDROID
+#include <android/log.h>
+#define ANDROID_LOG_TAG "VoiceEngine Auto Test"
+#define TEST_LOG(...) \
+ __android_log_print(ANDROID_LOG_DEBUG, ANDROID_LOG_TAG, __VA_ARGS__)
+#define TEST_LOG_ERROR(...) \
+ __android_log_print(ANDROID_LOG_ERROR, ANDROID_LOG_TAG, __VA_ARGS__)
+#else
+#define TEST_LOG printf
+#define TEST_LOG_ERROR printf
+#define TEST_LOG_FLUSH fflush(NULL)
+#endif
+
+// Select the tests to execute, list order below is same as they will be
+// executed. Note that, all settings below will be overriden by sub-API
+// settings in engine_configurations.h.
+#define _TEST_BASE_
+#define _TEST_RTP_RTCP_
+#define _TEST_HARDWARE_
+#define _TEST_CODEC_
+#define _TEST_DTMF_
+#define _TEST_VOLUME_
+#define _TEST_AUDIO_PROCESSING_
+#define _TEST_FILE_
+#define _TEST_NETWORK_
+#define _TEST_CALL_REPORT_
+#define _TEST_VIDEO_SYNC_
+#define _TEST_ENCRYPT_
+#define _TEST_NETEQ_STATS_
+#define _TEST_XMEDIA_
+
+#define TESTED_AUDIO_LAYER kAudioPlatformDefault
+//#define TESTED_AUDIO_LAYER kAudioLinuxPulse
+
+// #define _ENABLE_VISUAL_LEAK_DETECTOR_ // Enables VLD to find memory leaks
+// #define _ENABLE_IPV6_TESTS_ // Enables IPv6 tests in network xtest
+// #define _USE_EXTENDED_TRACE_ // Adds unique trace files for extended test
+// #define _MEMORY_TEST_
+
+// Enable this when running instrumentation of some kind to exclude tests
+// that will not pass due to slowed down execution.
+// #define _INSTRUMENTATION_TESTING_
+
+// Exclude (override) API tests given preprocessor settings in
+// engine_configurations.h
+#ifndef WEBRTC_VOICE_ENGINE_CODEC_API
+#undef _TEST_CODEC_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+#undef _TEST_VOLUME_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_DTMF_API
+#undef _TEST_DTMF_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+#undef _TEST_RTP_RTCP_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
+#undef _TEST_AUDIO_PROCESSING_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_FILE_API
+#undef _TEST_FILE_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+#undef _TEST_VIDEO_SYNC_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+#undef _TEST_ENCRYPT_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_HARDWARE_API
+#undef _TEST_HARDWARE_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
+#undef _TEST_XMEDIA_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_NETWORK_API
+#undef _TEST_NETWORK_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
+#undef _TEST_NETEQ_STATS_
+#endif
+#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+#undef _TEST_CALL_REPORT_
+#endif
+
+// Some parts can cause problems while running Insure
+#ifdef __INSURE__
+#define _INSTRUMENTATION_TESTING_
+#undef WEBRTC_SRTP
+#endif
+
+// Time in ms to test each packet size for each codec
+#define CODEC_TEST_TIME 400
+
+#define MARK() TEST_LOG("."); fflush(NULL); // Add test marker
+#define ANL() TEST_LOG("\n") // Add New Line
+#define AOK() TEST_LOG("[Test is OK]"); fflush(NULL); // Add OK
+#if defined(_WIN32)
+#define PAUSE \
+ { \
+ TEST_LOG("Press any key to continue..."); \
+ _getch(); \
+ TEST_LOG("\n"); \
+ }
+#else
+#define PAUSE \
+ { \
+ TEST_LOG("Continuing (pause not supported)\n"); \
+ }
+#endif
+
+#define TEST(s) \
+ { \
+ TEST_LOG("Testing: %s", #s); \
+ } \
+
+#ifdef _INSTRUMENTATION_TESTING_
+// Don't stop execution if error occurs
+#define TEST_MUSTPASS(expr) \
+ { \
+ if ((expr)) \
+ { \
+ TEST_LOG_ERROR("Error at line:%i, %s \n",__LINE__, #expr); \
+ TEST_LOG_ERROR("Error code: %i\n",voe_base_->LastError()); \
+ } \
+ }
+#define TEST_ERROR(code) \
+ { \
+ int err = voe_base_->LastError(); \
+ if (err != code) \
+ { \
+ TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n",
+ code, err, __LINE__);
+}
+}
+#else
+#define ASSERT_TRUE(expr) TEST_MUSTPASS(!(expr))
+#define ASSERT_FALSE(expr) TEST_MUSTPASS(expr)
+#define TEST_MUSTFAIL(expr) TEST_MUSTPASS(!((expr) == -1))
+#define TEST_MUSTPASS(expr) \
+ { \
+ if ((expr)) \
+ { \
+ TEST_LOG_ERROR("\nError at line:%i, %s \n",__LINE__, #expr); \
+ TEST_LOG_ERROR("Error code: %i\n", voe_base_->LastError()); \
+ PAUSE \
+ return -1; \
+ } \
+ }
+#define TEST_ERROR(code) \
+ { \
+ int err = voe_base_->LastError(); \
+ if (err != code) \
+ { \
+ TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n", \
+ err, code, __LINE__); \
+ PAUSE \
+ return -1; \
+ } \
+ }
+#endif // #ifdef _INSTRUMENTATION_TESTING_
+#define EXCLUDE() \
+ { \
+ TEST_LOG("\n>>> Excluding test at line: %i <<<\n\n",__LINE__); \
+ }
+
+#define INCOMPLETE() \
+ { \
+ TEST_LOG("\n>>> Incomplete test at line: %i <<<\n\n",__LINE__); \
+ }
+
+#endif // WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h b/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h
new file mode 100644
index 0000000..87b37a4
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Interface for starting test
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
+#define WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
+
+#include "common_types.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventWrapper;
+class ThreadWrapper;
+class VoENetwork;
+}
+
+namespace voetest {
+// TODO(andrew): using directives are not permitted.
+using namespace webrtc;
+
+// TestType enumerator
+enum TestType {
+ Invalid = -1, Standard = 0, Extended = 1, Stress = 2, Unit = 3, CPU = 4
+};
+
+// ExtendedSelection enumerator
+enum ExtendedSelection {
+ XSEL_Invalid = -1,
+ XSEL_None = 0,
+ XSEL_All,
+ XSEL_Base,
+ XSEL_CallReport,
+ XSEL_Codec,
+ XSEL_DTMF,
+ XSEL_Encryption,
+ XSEL_ExternalMedia,
+ XSEL_File,
+ XSEL_Mixing,
+ XSEL_Hardware,
+ XSEL_NetEqStats,
+ XSEL_Network,
+ XSEL_RTP_RTCP,
+ XSEL_VideoSync,
+ XSEL_VolumeControl,
+ XSEL_AudioProcessing,
+};
+
+// ----------------------------------------------------------------------------
+// External transport (Transport)
+// ----------------------------------------------------------------------------
+
+class FakeExternalTransport : public Transport {
+ public:
+ FakeExternalTransport(VoENetwork* ptr);
+ virtual ~FakeExternalTransport();
+ VoENetwork* my_network_;
+ int SendPacket(int channel, const void *data, int len);
+ int SendRTCPPacket(int channel, const void *data, int len);
+ void SetDelayStatus(bool enabled, unsigned int delayInMs = 100);
+ private:
+ static bool Run(void* ptr);
+ bool Process();
+ private:
+ ThreadWrapper* thread_;
+ CriticalSectionWrapper* lock_;
+ EventWrapper* event_;
+ private:
+ unsigned char packet_buffer_[1612];
+ int length_;
+ int channel_;
+ bool delay_is_enabled_;
+ int delay_time_in_ms_;
+};
+
+// Main test function
+int runAutoTest(TestType testType, ExtendedSelection extendedSel);
+
+} // namespace voetest
+#endif // WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc
new file mode 100644
index 0000000..1aa9e10
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc
@@ -0,0 +1,1083 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include <cassert>
+#if defined(_WIN32)
+#include <conio.h>
+#endif
+
+#include "voe_unit_test.h"
+
+#include "../../source/voice_engine_defines.h"
+#include "thread_wrapper.h"
+
+using namespace webrtc;
+
+namespace voetest {
+
+#define CHECK(expr) \
+ if (expr) \
+ { \
+ printf("Error at line: %i, %s \n", __LINE__, #expr); \
+ printf("Error code: %i \n", base->LastError()); \
+ PAUSE \
+ return -1; \
+ }
+
+extern char* GetFilename(char* filename);
+extern const char* GetFilename(const char* filename);
+extern int GetResource(char* resource, char* dest, int destLen);
+extern char* GetResource(char* resource);
+extern const char* GetResource(const char* resource);
+
+const char* VoEUnitTest::_key = "====YUtFWRAAAAADBtIHgAAAAAEAAAAcAAAAAQBHU0dsb2"
+ "JhbCBJUCBTb3VuZAAC\nAAAAIwAAAExpY2Vuc2VkIHRvIE5vcnRlbCBOZXR3cm9rcwAAAAAxA"
+ "AAAZxZ7/u0M\niFYyTwSwko5Uutf7mh8S0O4rYZYTFidbzQeuGonuL17F/2oD/2pfDp3jL4Rf"
+ "3z/A\nnlJsEJgEtASkDNFuwLILjGY0pzjjAYQp3pCl6z6k2MtE06AirdjGLYCjENpq/opX\nO"
+ "rs3sIuwdYK5va/aFcsjBDmlsGCUM48RDYG9s23bIHYafXUC4ofOaubbZPWiPTmL\nEVJ8WH4F"
+ "9pgNjALc14oJXfON7r/3\n=EsLx";
+
+// ----------------------------------------------------------------------------
+// >>> R E A D M E F I R S T <<<
+// ----------------------------------------------------------------------------
+
+// 1) The user must ensure that the following codecs are included in VoE:
+//
+// - L16
+// - G.729
+// - G.722.1C
+
+// 2) It is also possible to modify the simulation time for each individual test
+//
+const int dTBetweenEachTest = 4000;
+
+// ----------------------------------------------------------------------------
+// Encrypt
+// ----------------------------------------------------------------------------
+
+void VoEUnitTest::encrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int * bytes_out) {
+ int i;
+
+ if (!_extOnOff) {
+ // no stereo emulation <=> pure bypass
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = in_data[i];
+ *bytes_out = bytes_in;
+ } else if (_extOnOff && (_extBitsPerSample == 16)) {
+ // stereo emulation (sample based, 2 bytes per sample)
+
+ const int nBytesPayload = bytes_in - 12;
+
+ // RTP header (first 12 bytes)
+ memcpy(out_data, in_data, 12);
+
+ // skip RTP header
+ short* ptrIn = (short*) &in_data[12];
+ short* ptrOut = (short*) &out_data[12];
+
+ // network byte order
+ for (i = 0; i < nBytesPayload / 2; i++) {
+ // produce two output samples for each input sample
+ *ptrOut++ = *ptrIn; // left sample
+ *ptrOut++ = *ptrIn; // right sample
+ ptrIn++;
+ }
+
+ *bytes_out = 12 + 2 * nBytesPayload;
+ } else if (_extOnOff && (_extBitsPerSample == 8)) {
+ // stereo emulation (sample based, 1 bytes per sample)
+
+ const int nBytesPayload = bytes_in - 12;
+
+ // RTP header (first 12 bytes)
+ memcpy(out_data, in_data, 12);
+
+ // skip RTP header
+ unsigned char* ptrIn = (unsigned char*) &in_data[12];
+ unsigned char* ptrOut = (unsigned char*) &out_data[12];
+
+ // network byte order
+ for (i = 0; i < nBytesPayload; i++) {
+ // produce two output samples for each input sample
+ *ptrOut++ = *ptrIn; // left sample
+ *ptrOut++ = *ptrIn; // right sample
+ ptrIn++;
+ }
+
+ *bytes_out = 12 + 2 * nBytesPayload;
+ } else if (_extOnOff && (_extBitsPerSample == -1)) {
+ // stereo emulation (frame based)
+
+ const int nBytesPayload = bytes_in - 12;
+
+ // RTP header (first 12 bytes)
+ memcpy(out_data, in_data, 12);
+
+ // skip RTP header
+ unsigned char* ptrIn = (unsigned char*) &in_data[12];
+ unsigned char* ptrOut = (unsigned char*) &out_data[12];
+
+ // left channel
+ for (i = 0; i < nBytesPayload; i++) {
+ *ptrOut++ = *ptrIn++;
+ }
+
+ ptrIn = (unsigned char*) &in_data[12];
+
+ // right channel
+ for (i = 0; i < nBytesPayload; i++) {
+ *ptrOut++ = *ptrIn++;
+ }
+
+ *bytes_out = 12 + 2 * nBytesPayload;
+ }
+}
+
+void VoEUnitTest::decrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = in_data[i];
+ *bytes_out = bytes_in;
+}
+
+void VoEUnitTest::encrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = in_data[i];
+ *bytes_out = bytes_in;
+}
+
+void VoEUnitTest::decrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in,
+ int * bytes_out) {
+ int i;
+ for (i = 0; i < bytes_in; i++)
+ out_data[i] = in_data[i];
+ *bytes_out = bytes_in;
+}
+
+void VoEUnitTest::SetStereoExternalEncryption(int channel, bool onOff,
+ int bitsPerSample) {
+ _extOnOff = onOff;
+ _extChannel = channel;
+ _extBitsPerSample = bitsPerSample;
+}
+
+// VoEVEMediaProcess
+MyMedia mpobj;
+
+// ----------------------------------------------------------------------------
+// VoEUnitTest
+// ----------------------------------------------------------------------------
+
+VoEUnitTest::VoEUnitTest(VoETestManager& mgr) :
+ _mgr(mgr), _extOnOff(false), _extBitsPerSample(-1) {
+ for (int i = 0; i < 32; i++) {
+ _listening[i] = false;
+ _playing[i] = false;
+ _sending[i] = false;
+ }
+}
+
+// ----------------------------------------------------------------------------
+// DoTest
+// ----------------------------------------------------------------------------
+
+int VoEUnitTest::DoTest() {
+ int test(-1);
+ int ret(0);
+ while ((test != 0) && (ret != -1)) {
+ test = MenuSelection();
+ switch (test) {
+ case 0:
+ // Quit stress test
+ break;
+ case 1:
+ ret = MixerTest();
+ break;
+ case 2:
+ ret = MixerTest();
+ break;
+ default:
+ // Should not be possible
+ printf("Invalid selection! (Test code error)\n");
+ assert(false);
+ } // switch
+ } // while
+
+ return ret;
+}
+
+// ----------------------------------------------------------------------------
+// MenuSelection
+// ----------------------------------------------------------------------------
+
+int VoEUnitTest::MenuSelection() {
+ printf("------------------------------------------------\n");
+ printf("Select unit test\n\n");
+ printf(" (0) Quit\n");
+ printf(" (1) All\n");
+ printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
+ printf(" (2) Mixer\n");
+
+ const int maxMenuSelection = 2;
+ int selection(-1);
+
+ while ((selection < 0) || (selection > maxMenuSelection)) {
+ printf("\n: ");
+ int retval = scanf("%d", &selection);
+ if ((retval != 1) || (selection < 0) || (selection > maxMenuSelection)) {
+ printf("Invalid selection!\n");
+ }
+ }
+
+ return selection;
+}
+
+// ----------------------------------------------------------------------------
+// StartMedia
+// ----------------------------------------------------------------------------
+
+int VoEUnitTest::StartMedia(int channel, int rtpPort, bool listen, bool playout,
+ bool send, bool fileAsMic, bool localFile) {
+ VoEBase* base = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+
+ _listening[channel] = false;
+ _playing[channel] = false;
+ _sending[channel] = false;
+
+ CHECK(base->SetLocalReceiver(channel, rtpPort));
+ CHECK(base->SetSendDestination(channel, rtpPort, "127.0.0.1"));
+ if (listen) {
+ _listening[channel] = true;
+ CHECK(base->StartReceive(channel));
+ }
+ if (playout) {
+ _playing[channel] = true;
+ CHECK(base->StartPlayout(channel));
+ }
+ if (send) {
+ _sending[channel] = true;
+ CHECK(base->StartSend(channel));
+ }
+ if (fileAsMic) {
+ // play mic as file, mix with microphone to ensure that SWB can be
+ //tested as well
+ const bool mixWithMic(true);
+ CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(),
+ true, mixWithMic));
+ }
+ if (localFile) {
+ CHECK(file->StartPlayingFileLocally(channel,
+ GetResource("audio_short16.pcm"),
+ false,
+ kFileFormatPcm16kHzFile));
+ }
+
+ return 0;
+}
+
+// ----------------------------------------------------------------------------
+// StopMedia
+// ----------------------------------------------------------------------------
+
+int VoEUnitTest::StopMedia(int channel) {
+ VoEBase* base = _mgr.BasePtr();
+ VoEFile* file = _mgr.FilePtr();
+
+ if (file->IsPlayingFileAsMicrophone(channel)) {
+ CHECK(file->StopPlayingFileAsMicrophone(channel));
+ }
+ if (file->IsPlayingFileLocally(channel)) {
+ CHECK(file->StopPlayingFileLocally(channel));
+ }
+ if (_listening[channel]) {
+ _listening[channel] = false;
+ CHECK(base->StopReceive(channel));
+ }
+ if (_playing[channel]) {
+ _playing[channel] = false;
+ CHECK(base->StopPlayout(channel));
+ }
+ if (_sending[channel]) {
+ _sending[channel] = false;
+ CHECK(base->StopSend(channel));
+ }
+
+ return 0;
+}
+
+void VoEUnitTest::Sleep(unsigned int timeMillisec, bool addMarker) {
+ if (addMarker) {
+ float dtSec = (float) ((float) timeMillisec / 1000.0);
+ printf("[dT=%.1f]", dtSec);
+ fflush(NULL);
+ }
+ ::Sleep(timeMillisec);
+}
+
+void VoEUnitTest::Wait() {
+#if defined(_WIN32)
+ printf("\npress any key..."); fflush(NULL);
+ _getch();
+#endif
+}
+
+void VoEUnitTest::Test(const char* msg) {
+ printf("%s", msg);
+ fflush(NULL);
+ printf("\n");
+ fflush(NULL);
+}
+
+int VoEUnitTest::MixerTest() {
+ // Set up test parameters first
+ //
+ const int testTime(dTBetweenEachTest);
+
+ printf("\n\n================================================\n");
+ printf(" Mixer Unit Test\n");
+ printf("================================================\n\n");
+
+ // Get sub-API pointers
+ //
+ VoEBase* base = _mgr.BasePtr();
+ VoECodec* codec = _mgr.CodecPtr();
+ VoEFile* file = _mgr.FilePtr();
+ VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+ VoEEncryption* encrypt = _mgr.EncryptionPtr();
+ VoEDtmf* dtmf = _mgr.DtmfPtr();
+ VoEExternalMedia* xmedia = _mgr.ExternalMediaPtr();
+
+ // Set trace
+ //
+ VoiceEngine::SetTraceFile(GetFilename("UnitTest_Mixer_trace.txt"));
+ VoiceEngine::SetTraceFilter(kTraceStateInfo | kTraceWarning | kTraceError |
+ kTraceCritical | kTraceApiCall | kTraceMemory |
+ kTraceInfo);
+
+ // Init
+ //
+ CHECK(base->Init());
+
+ // 8 kHz
+ // CodecInst l16_8 = { 123, "L16", 8000, 160, 1, 128000 };
+ CodecInst pcmu_8 = { 0, "pcmu", 8000, 160, 1, 64000 };
+ // CodecInst g729_8 = { 18, "g729", 8000, 160, 1, 8000 };
+
+ // 16 kHz
+ CodecInst ipcmwb_16 = { 97, "ipcmwb", 16000, 320, 1, 80000 };
+ CodecInst l16_16 = { 124, "L16", 16000, 320, 1, 256000 };
+
+ // 32 kHz
+ CodecInst l16_32 = { 125, "L16", 32000, 320, 1, 512000 };
+ CodecInst g722_1c_32 = { 126, "G7221", 32000, 640, 1, 32000 };// 20ms@32kHz
+
+ // ------------------------
+ // Verify mixing frequency
+ // ------------------------
+
+ base->CreateChannel();
+
+ Test(">> Verify correct mixing frequency:\n");
+
+ Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending file at 16kHz <=> mixing at 16kHz...");
+ CHECK(codec->SetSendCodec(0, ipcmwb_16));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending speech at 32kHz <=> mixing at 32Hz...");
+ CHECK(codec->SetSendCodec(0, l16_32));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
+ CHECK(codec->SetSendCodec(0, pcmu_8));
+ Sleep(testTime);
+
+ Test("(ch 0) Playing 16kHz file locally <=> mixing at 16kHz...");
+ CHECK(file->StartPlayingFileLocally(0, GetResource("audio_long16.pcm"),
+ false, kFileFormatPcm16kHzFile));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+
+ base->CreateChannel();
+
+ Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
+ CHECK(codec->SetSendCodec(0, pcmu_8));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending speech at 32kHz <=> mixing at 32Hz...");
+ CHECK(codec->SetSendCodec(0, l16_32));
+ Sleep(testTime);
+
+ Test("(ch 1) Playing 16kHz file locally <=> mixing at 32kHz...");
+ CHECK(StartMedia(1, 54321, false, true, false, false, true));
+ Sleep(testTime);
+
+ CHECK(StopMedia(1));
+ CHECK(StopMedia(0));
+
+ base->DeleteChannel(1);
+ base->DeleteChannel(0);
+ ANL();
+
+ // -------------------------
+ // Verify stereo mode mixing
+ // -------------------------
+
+ base->CreateChannel();
+ base->CreateChannel();
+
+ // SetOutputVolumePan
+ //
+ // Ensure that all cases sound OK and that the mixer changes state between
+ // mono and stereo as it should. A debugger is required to trace the state
+ // transitions.
+
+ Test(">> Verify correct mixing in stereo using SetOutputVolumePan():\n");
+
+ Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
+ CHECK(StartMedia(0, 12345, false, true, false, false, true));
+ Sleep(testTime);
+ Test("Panning volume to the left <=> mixing in stereo @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 0.0));
+ Sleep(testTime);
+ Test("Panning volume to the right <=> mixing in stereo @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
+ Sleep(testTime);
+ Test("Back to center volume again <=> mixing in mono @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
+ Sleep(testTime);
+ Test("(ch 1) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
+ CHECK(StartMedia(1, 54321, false, true, false, false, true));
+ Sleep(testTime);
+ Test("Panning volume to the left <=> mixing in stereo @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 0.0));
+ Sleep(testTime);
+ Test("Back to center volume again <=> mixing in mono @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
+ Sleep(testTime);
+ Test("(ch 1) Stopped playing file <=> mixing in mono @ 16kHz...");
+ CHECK(StopMedia(1));
+ Sleep(testTime);
+ CHECK(StopMedia(0));
+ Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+ Test("(ch 0) Sending speech at 32kHz <=> mixing at 32kHz...");
+ CHECK(codec->SetSendCodec(0, l16_32));
+ Sleep(testTime);
+ Test("Panning volume to the right <=> mixing in stereo @ 32kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
+ Sleep(testTime);
+ Test("Back to center volume again <=> mixing in mono @ 32kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
+ Sleep(testTime);
+ CHECK(StopMedia(0));
+ ANL();
+
+ base->DeleteChannel(0);
+ base->DeleteChannel(1);
+
+ // SetChannelOutputVolumePan
+ //
+ // Ensure that all cases sound OK and that the mixer changes state between
+ // mono and stereo as it should. A debugger is required to trace the state
+ // transitions.
+
+ base->CreateChannel();
+ base->CreateChannel();
+
+ Test(">> Verify correct mixing in stereo using"
+ " SetChannelOutputVolumePan():\n");
+
+ Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
+ CHECK(StartMedia(0, 12345, false, true, false, false, true));
+ Sleep(testTime);
+ Test("(ch 0) Panning channel volume to the left <=> mixing in stereo @ "
+ "16kHz...");
+ CHECK(volume->SetOutputVolumePan(0, 1.0, 0.0));
+ Sleep(testTime);
+ Test("(ch 0) Panning channel volume to the right <=> mixing in stereo"
+ " @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(0, 0.0, 1.0));
+ Sleep(testTime);
+ Test("(ch 0) Back to center volume again <=> mixing in mono @"
+ " 16kHz...");
+ CHECK(volume->SetOutputVolumePan(0, 1.0, 1.0));
+ Sleep(testTime);
+ Test("(ch 1) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
+ CHECK(StartMedia(1, 54321, false, true, false, false, true));
+ Sleep(testTime);
+ Test("(ch 1) Panning channel volume to the left <=> mixing in stereo "
+ "@ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(1, 1.0, 0.0));
+ Sleep(testTime);
+ Test("(ch 1) Back to center volume again <=> mixing in mono @ 16kHz...");
+ CHECK(volume->SetOutputVolumePan(1, 1.0, 1.0));
+ Sleep(testTime);
+ Test("(ch 1) Stopped playing file <=> mixing in mono @ 16kHz...");
+ CHECK(StopMedia(1));
+ Sleep(testTime);
+ CHECK(StopMedia(0));
+ ANL();
+
+ base->DeleteChannel(0);
+ base->DeleteChannel(1);
+
+ // Emulate stereo-encoding using Encryption
+ //
+ // Modify the transmitted RTP stream by using external encryption.
+ // Supports frame based and sample based "stereo-encoding schemes".
+
+ base->CreateChannel();
+
+ Test(">> Verify correct mixing in stereo using emulated stereo input:\n");
+
+ // enable external encryption
+ CHECK(encrypt->RegisterExternalEncryption(0, *this));
+ Test("(ch 0) External Encryption is now enabled:");
+
+ Test("(ch 0) Sending file at 8kHz <=> mixing in mono @ 8kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ // switch to 16kHz (L16) sending codec
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Test("(ch 0) Sending file at 16kHz (L16) <=> mixing in mono @ 16kHz...");
+ Sleep(testTime);
+
+ // register L16 as 2-channel codec on receiving side =>
+ // should sound bad since RTP module splits all received packets in half
+ // (sample based)
+ CHECK(base->StopPlayout(0));
+ CHECK(base->StopReceive(0));
+ l16_16.channels = 2;
+ CHECK(codec->SetRecPayloadType(0, l16_16));
+ CHECK(base->StartReceive(0));
+ CHECK(base->StartPlayout(0));
+ Test("(ch 0) 16kHz L16 is now registered as 2-channel codec on RX side => "
+ "should sound bad...");
+ Sleep(testTime);
+
+ // emulate sample-based stereo encoding
+ Test("(ch 0) Emulate sample-based stereo encoding on sending side => "
+ "should sound OK...");
+ SetStereoExternalEncryption(0, true, 16);
+ Sleep(testTime);
+ Test("(ch 0) Stop emulating sample-based stereo encoding on sending side =>"
+ " should sound bad...");
+ SetStereoExternalEncryption(0, false, 16);
+ Sleep(testTime);
+ Test("(ch 0) Emulate sample-based stereo encoding on sending side => "
+ "should sound OK...");
+ SetStereoExternalEncryption(0, true, 16);
+ Sleep(testTime);
+
+ // switch to 32kHz (L16) sending codec and disable stereo encoding
+ CHECK(codec->SetSendCodec(0, l16_32));
+ SetStereoExternalEncryption(0, false, 16);
+ Test("(ch 0) Sending file and spech at 32kHz (L16) <=> mixing in mono @ "
+ "32kHz...");
+ Sleep(testTime);
+
+ // register L16 32kHz as 2-channel codec on receiving side
+ CHECK(base->StopPlayout(0));
+ CHECK(base->StopReceive(0));
+ l16_32.channels = 2;
+ CHECK(codec->SetRecPayloadType(0, l16_32));
+ CHECK(base->StartReceive(0));
+ CHECK(base->StartPlayout(0));
+ Test("(ch 0) 32kHz L16 is now registered as 2-channel codec on RX side =>"
+ " should sound bad...");
+ Sleep(testTime);
+
+ // emulate sample-based stereo encoding
+ Test("(ch 0) Emulate sample-based stereo encoding on sending side =>"
+ " should sound OK...");
+ SetStereoExternalEncryption(0, true, 16);
+ Sleep(testTime);
+
+ StopMedia(0);
+ l16_32.channels = 1;
+
+ // disable external encryption
+ CHECK(encrypt->DeRegisterExternalEncryption(0));
+ ANL();
+
+ base->DeleteChannel(0);
+
+ // ------------------
+ // Verify put-on-hold
+ // ------------------
+
+ base->CreateChannel();
+ base->CreateChannel();
+
+ Test(">> Verify put-on-hold functionality:\n");
+
+ Test("(ch 0) Sending at 8kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
+ Test("(ch 0) Playout is now on hold...");
+ Sleep(testTime);
+ CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
+ Test("(ch 0) Playout is now enabled again...");
+ Sleep(testTime);
+
+ Test("(ch 0) Sending at 16kHz...");
+ l16_16.channels = 1;
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Sleep(testTime);
+
+ CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
+ Test("(ch 0) Playout is now on hold...");
+ Sleep(testTime);
+ CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
+ Test("(ch 0) Playout is now enabled again...");
+ Sleep(testTime);
+
+ Test("(ch 0) Perform minor panning to the left to force mixing in"
+ " stereo...");
+ CHECK(volume->SetOutputVolumePan(0, (float)1.0, (float)0.7));
+ Sleep(testTime);
+
+ CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
+ Test("(ch 0) Playout is now on hold...");
+ Sleep(testTime);
+ CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
+ Test("(ch 0) Playout is now enabled again...");
+ Sleep(testTime);
+
+ Test("(ch 0) Back to center volume again...");
+ CHECK(volume->SetOutputVolumePan(0, 1.0, 1.0));
+ Sleep(testTime);
+
+ Test("(ch 1) Add 16kHz local file to the mixer...");
+ CHECK(StartMedia(1, 54321, false, true, false, false, true));
+ Sleep(testTime);
+
+ CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
+ Test("(ch 0) Playout is now on hold...");
+ Sleep(testTime);
+ CHECK(base->SetOnHoldStatus(1, true, kHoldPlayOnly));
+ Test("(ch 1) Playout is now on hold => should be silent...");
+ Sleep(testTime);
+ CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
+ Test("(ch 0) Playout is now enabled again...");
+ CHECK(base->SetOnHoldStatus(1, false, kHoldPlayOnly));
+ Test("(ch 1) Playout is now enabled again...");
+ Sleep(testTime);
+ StopMedia(1);
+ Test("(ch 1) Stopped playing file...");
+ Sleep(testTime);
+ StopMedia(0);
+ ANL();
+
+ base->DeleteChannel(0);
+ base->DeleteChannel(1);
+
+ // -----------------------------------
+ // Verify recording of playout to file
+ // -----------------------------------
+
+ // StartRecordingPlayout
+ //
+ // Verify that the correct set of signals is recorded in the mixer.
+ // Record each channel and all channels (-1) to ensure that post and pre
+ // mixing recording works.
+
+ base->CreateChannel();
+ base->CreateChannel();
+
+ Test(">> Verify file-recording functionality:\n");
+
+ Test("(ch 0) Sending at 8kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ Test("(ch 0) Recording of playout to 16kHz PCM file...");
+ CHECK(file->StartRecordingPlayout(
+ 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
+ Sleep(testTime);
+ CHECK(file->StopRecordingPlayout(0));
+
+ Test("(ch 0) Playing out the recorded file...");
+ CHECK(volume->SetInputMute(0, true));
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+ CHECK(volume->SetInputMute(0, false));
+
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Test("(ch 0) Sending at 16kHz (L16)...");
+ Sleep(testTime);
+
+ Test("(ch 0) Recording of playout to 16kHz PCM file...");
+ CHECK(file->StartRecordingPlayout(
+ 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
+ Sleep(testTime);
+ CHECK(file->StopRecordingPlayout(0));
+
+ Test("(ch 0) Playing out the recorded file...");
+ CHECK(volume->SetInputMute(0, true));
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+ CHECK(volume->SetInputMute(0, false));
+
+ CHECK(codec->SetSendCodec(0, l16_32));
+ Test("(ch 0) Sending at 32kHz (L16)...");
+ Sleep(testTime);
+
+ Test("(ch 0) Recording of playout to 16kHz PCM file...");
+ CHECK(file->StartRecordingPlayout(
+ 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
+ Sleep(testTime);
+ CHECK(file->StopRecordingPlayout(0));
+
+ Test("(ch 0) Playing out the recorded file...");
+ CHECK(volume->SetInputMute(0, true));
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+ CHECK(volume->SetInputMute(0, false));
+
+ Test("(ch 0) Sending at 16kHz without file as mic but file added on the"
+ " playout side instead...");
+ CHECK(StopMedia(0));
+ CHECK(StartMedia(0, 12345, false, true, false, false, true));
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Sleep(testTime);
+
+ Test("(ch 0) Recording of playout to 16kHz PCM file...");
+ CHECK(file->StartRecordingPlayout(
+ 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
+ Sleep(testTime);
+ CHECK(file->StopRecordingPlayout(0));
+ CHECK(file->StopPlayingFileLocally(0));
+
+ Test("(ch 0) Playing out the recorded file...");
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+
+ CHECK(StopMedia(0));
+ CHECK(StopMedia(1));
+
+ Test("(ch 0) Sending at 16kHz...");
+ CHECK(StartMedia(0, 12345, true, true, true, false, false));
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Test("(ch 1) Adding playout file...");
+ CHECK(StartMedia(1, 33333, false, true, false, false, true));
+ Sleep(testTime);
+
+ Test("(ch -1) Speak while recording all channels to add mixer input on "
+ "channel 0...");
+ CHECK(file->StartRecordingPlayout(
+ -1, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
+ Sleep(testTime);
+ CHECK(file->StopRecordingPlayout(-1));
+ CHECK(file->StopPlayingFileLocally(1));
+
+ Test("(ch 0) Playing out the recorded file...");
+ CHECK(volume->SetInputMute(0, true));
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+ CHECK(volume->SetInputMute(0, false));
+
+ CHECK(StopMedia(0));
+ CHECK(StopMedia(1));
+ ANL();
+
+ // StartRecordingPlayoutStereo
+
+ Test(">> Verify recording of playout in stereo:\n");
+
+ Test("(ch 0) Sending at 32kHz...");
+ CHECK(codec->SetSendCodec(0, l16_16));
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ Test("Modified master balance (L=10%%, R=100%%) to force stereo mixing...");
+ CHECK(volume->SetOutputVolumePan(-1, (float)0.1, (float)1.0));
+ Sleep(testTime);
+
+ /*
+ Test("Recording of left and right channel playout to two 16kHz PCM "
+ "files...");
+ file->StartRecordingPlayoutStereo(
+ GetFilename("RecordedPlayout_Left_16kHz.pcm"),
+ GetFilename("RecordedPlayout_Right_16kHz.pcm"), StereoBoth);
+ Sleep(testTime);
+ Test("Back to center volume again...");
+ CHECK(volume->SetOutputVolumePan(-1, (float)1.0, (float)1.0));
+ */
+
+ Test("(ch 0) Playing out the recorded file for the left channel (10%%)...");
+ CHECK(volume->SetInputMute(0, true));
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout_Left_16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+
+ Test("(ch 0) Playing out the recorded file for the right channel (100%%) =>"
+ " should sound louder than the left channel...");
+ CHECK(file->StartPlayingFileLocally(
+ 0, GetFilename("RecordedPlayout_Right_16kHz.pcm")));
+ Sleep(testTime);
+ CHECK(file->StopPlayingFileLocally(0));
+ CHECK(volume->SetInputMute(0, false));
+
+ base->DeleteChannel(0);
+ base->DeleteChannel(1);
+ ANL();
+
+ // ---------------------------
+ // Verify inserted Dtmf tones
+ // ---------------------------
+
+ Test(">> Verify Dtmf feedback functionality:\n");
+
+ base->CreateChannel();
+
+ for (int i = 0; i < 2; i++) {
+ if (i == 0)
+ Test("Dtmf direct feedback is now enabled...");
+ else
+ Test("Dtmf direct feedback is now disabled...");
+
+ CHECK(dtmf->SetDtmfFeedbackStatus(true, (i==0)));
+
+ Test("(ch 0) Sending at 32kHz using G.722.1C...");
+ CHECK(codec->SetRecPayloadType(0, g722_1c_32));
+ CHECK(codec->SetSendCodec(0, g722_1c_32));
+ CHECK(StartMedia(0, 12345, true, true, true, false, false));
+ Sleep(500);
+
+ Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
+ " to the mixer...");
+ // ensure that receiver will not play out outband Dtmf
+ CHECK(dtmf->SetSendTelephoneEventPayloadType(0, 118));
+ CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
+ Sleep(500);
+ Sleep(testTime - 1500);
+
+ Test("(ch 0) Changing codec to 8kHz PCMU...");
+ CHECK(codec->SetSendCodec(0, pcmu_8));
+ Sleep(500);
+
+ Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
+ " to the mixer...");
+ CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
+ Sleep(500);
+ Sleep(testTime - 1500);
+
+ Test("(ch 0) Changing codec to 16kHz L16...");
+ CHECK(codec->SetSendCodec(0, l16_16));
+ Sleep(500);
+
+ Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
+ " to the mixer...");
+ CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
+ Sleep(500);
+ CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
+ Sleep(500);
+ Sleep(testTime - 1500);
+
+ StopMedia(0);
+ ANL();
+ }
+
+ base->DeleteChannel(0);
+
+ // ---------------------------
+ // Verify external processing
+ // --------------------------
+
+ base->CreateChannel();
+
+ Test(">> Verify external media processing:\n");
+
+ Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
+ CHECK(StartMedia(0, 12345, false, true, false, false, true));
+ Sleep(testTime);
+ Test("Enabling playout external media processing => played audio should "
+ "now be affected");
+ CHECK(xmedia->RegisterExternalMediaProcessing(
+ 0, kPlaybackAllChannelsMixed, mpobj));
+ Sleep(testTime);
+ Test("(ch 0) Sending speech at 32kHz <=> mixing at 32kHz...");
+ CHECK(codec->SetSendCodec(0, l16_32));
+ Sleep(testTime);
+ printf("Back to normal again\n");
+ CHECK(xmedia->DeRegisterExternalMediaProcessing(0,
+ kPlaybackAllChannelsMixed));
+ Sleep(testTime);
+ printf("Enabling playout external media processing on ch 0 => "
+ "played audio should now be affected\n");
+ CHECK(xmedia->RegisterExternalMediaProcessing(0, kPlaybackPerChannel,
+ mpobj));
+ Sleep(testTime);
+ Test("Panning volume to the right <=> mixing in stereo @ 32kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
+ Sleep(testTime);
+ Test("Back to center volume again <=> mixing in mono @ 32kHz...");
+ CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
+ Sleep(testTime);
+ printf("Back to normal again\n");
+ CHECK(xmedia->DeRegisterExternalMediaProcessing(0, kPlaybackPerChannel));
+ Sleep(testTime);
+ CHECK(StopMedia(0));
+ ANL();
+
+ base->DeleteChannel(0);
+
+ // --------------------------------------------------
+ // Extended tests of emulated stereo encoding schemes
+ // --------------------------------------------------
+
+ CodecInst PCMU;
+ CodecInst G729;
+ CodecInst L16_8;
+ CodecInst L16_16;
+ CodecInst L16_32;
+
+ base->CreateChannel();
+
+ Test(">> Verify emulated stereo encoding for differenct codecs:\n");
+
+ // enable external encryption
+ CHECK(encrypt->RegisterExternalEncryption(0, *this));
+ Test("(ch 0) External Encryption is now enabled:");
+
+ // register all codecs on the receiving side
+ strcpy(PCMU.plname, "PCMU");
+ PCMU.channels = 2;
+ PCMU.pacsize = 160;
+ PCMU.plfreq = 8000;
+ PCMU.pltype = 125;
+ PCMU.rate = 64000;
+ CHECK(codec->SetRecPayloadType(0, PCMU));
+
+ strcpy(G729.plname, "G729");
+ G729.channels = 2;
+ G729.pacsize = 160;
+ G729.plfreq = 8000;
+ G729.pltype = 18;
+ G729.rate = 8000;
+ CHECK(codec->SetRecPayloadType(0, G729));
+
+ strcpy(L16_8.plname, "L16");
+ L16_8.channels = 2;
+ L16_8.pacsize = 160;
+ L16_8.plfreq = 8000;
+ L16_8.pltype = 120;
+ L16_8.rate = 128000;
+ CHECK(codec->SetRecPayloadType(0, L16_8));
+
+ strcpy(L16_16.plname, "L16");
+ L16_16.channels = 2;
+ L16_16.pacsize = 320;
+ L16_16.plfreq = 16000;
+ L16_16.pltype = 121;
+ L16_16.rate = 256000;
+ CHECK(codec->SetRecPayloadType(0, L16_16));
+
+ // NOTE - we cannot send larger than 1500 bytes per RTP packet
+ strcpy(L16_32.plname, "L16");
+ L16_32.channels = 2;
+ L16_32.pacsize = 320;
+ L16_32.plfreq = 32000;
+ L16_32.pltype = 122;
+ L16_32.rate = 512000;
+ CHECK(codec->SetRecPayloadType(0, L16_32));
+
+ // sample-based, 8-bits per sample
+
+ Test("(ch 0) Sending using G.711 (sample based, 8 bits/sample)...");
+ PCMU.channels = 1;
+ CHECK(codec->SetSendCodec(0, PCMU));
+ SetStereoExternalEncryption(0, true, 8);
+ CHECK(StartMedia(0, 12345, true, true, true, true, false));
+ Sleep(testTime);
+
+ // sample-based, 16-bits per sample
+
+ Test("(ch 0) Sending using L16 8kHz (sample based, 16 bits/sample)...");
+ L16_8.channels = 1;
+ CHECK(codec->SetSendCodec(0, L16_8));
+ SetStereoExternalEncryption(0, true, 16);
+ Sleep(testTime);
+
+ Test("(ch 0) Sending using L16 16kHz (sample based, 16 bits/sample)...");
+ L16_16.channels = 1;
+ CHECK(codec->SetSendCodec(0, L16_16));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending using L16 32kHz (sample based, 16 bits/sample)...");
+ L16_32.channels = 1;
+ CHECK(codec->SetSendCodec(0, L16_32));
+ Sleep(testTime);
+
+ Test("(ch 0) Sending using G.729 (frame based)...");
+ G729.channels = 1;
+ CHECK(codec->SetSendCodec(0, G729));
+ Sleep(testTime);
+
+ StopMedia(0);
+
+ // disable external encryption
+ CHECK(encrypt->DeRegisterExternalEncryption(0));
+
+ base->DeleteChannel(0);
+
+ // ------------------------------------------------------------------------
+ CHECK(base->Terminate());
+
+ printf("\n\n------------------------------------------------\n");
+ printf(" Test passed!\n");
+ printf("------------------------------------------------\n\n");
+
+ return 0;
+}
+
+} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h
new file mode 100644
index 0000000..ce53241
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
+#define WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
+
+#include "voe_standard_test.h"
+
+namespace voetest {
+
+class VoETestManager;
+
+class VoEUnitTest : public Encryption {
+ public:
+ VoEUnitTest(VoETestManager& mgr);
+ ~VoEUnitTest() {}
+ int DoTest();
+
+ protected:
+ // Encryption
+ void encrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void decrypt(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void encrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+ void decrypt_rtcp(int channel_no, unsigned char * in_data,
+ unsigned char * out_data, int bytes_in, int * bytes_out);
+
+ private:
+ int MenuSelection();
+ int MixerTest();
+ void Sleep(unsigned int timeMillisec, bool addMarker = false);
+ void Wait();
+ int StartMedia(int channel,
+ int rtpPort,
+ bool listen,
+ bool playout,
+ bool send,
+ bool fileAsMic,
+ bool localFile);
+ int StopMedia(int channel);
+ void Test(const char* msg);
+ void SetStereoExternalEncryption(int channel, bool onOff, int bitsPerSample);
+
+ private:
+ VoETestManager& _mgr;
+ static const char* _key;
+
+ private:
+ bool _listening[32];
+ bool _playing[32];
+ bool _sending[32];
+
+ private:
+ bool _extOnOff;
+ int _extBitsPerSample;
+ int _extChannel;
+};
+
+} // namespace voetest
+#endif // WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
diff --git a/trunk/src/voice_engine/main/test/cmd_test/Android.mk b/trunk/src/voice_engine/main/test/cmd_test/Android.mk
new file mode 100644
index 0000000..f1a4f1a
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/cmd_test/Android.mk
@@ -0,0 +1,60 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH:= $(call my-dir)
+
+# voice engine test app
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ voe_cmd_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_ANDROID' \
+ '-DDEBUG'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../../interface \
+ $(LOCAL_PATH)/../../../.. \
+ $(LOCAL_PATH)/../../../../.. \
+ external/gtest/include \
+ frameworks/base/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libmedia \
+ libcamera_client \
+ libgui \
+ libhardware \
+ libandroid_runtime \
+ libbinder
+
+#libwilhelm.so libDunDef-Android.so libbinder.so libsystem_server.so
+
+LOCAL_MODULE:= webrtc_voe_cmd
+
+ifdef NDK_ROOT
+LOCAL_SHARED_LIBRARIES += \
+ libstlport_shared \
+ libwebrtc-voice-jni \
+ libwebrtc_audio_preprocessing
+include $(BUILD_EXECUTABLE)
+else
+LOCAL_SHARED_LIBRARIES += \
+ libstlport \
+ libwebrtc
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
diff --git a/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc b/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc
new file mode 100644
index 0000000..99e88d2
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc
@@ -0,0 +1,922 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#ifndef _WIN32
+#include <unistd.h>
+#endif
+
+#include <vector>
+
+#include "gtest/gtest.h"
+#include "test/testsupport/fileutils.h"
+
+#include "voe_errors.h"
+#include "voe_base.h"
+#include "voe_codec.h"
+#include "voe_volume_control.h"
+#include "voe_dtmf.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_audio_processing.h"
+#include "voe_file.h"
+#include "voe_video_sync.h"
+#include "voe_encryption.h"
+#include "voe_hardware.h"
+#include "voe_external_media.h"
+#include "voe_network.h"
+#include "voe_neteq_stats.h"
+#include "engine_configurations.h"
+
+// Enable this this flag to run this test with hard coded
+// IP/Port/codec and start test automatically with key input
+// it could be useful in repeat tests.
+//#define DEBUG
+
+// #define EXTERNAL_TRANSPORT
+
+using namespace webrtc;
+
+#define VALIDATE \
+ if (res != 0) \
+ { \
+ printf("*** Error at position %i / line %i \n", cnt, __LINE__); \
+ printf("*** Error code = %i \n", base1->LastError()); \
+ } \
+ cnt++;
+
+VoiceEngine* m_voe = NULL;
+VoEBase* base1 = NULL;
+VoECodec* codec = NULL;
+VoEVolumeControl* volume = NULL;
+VoEDtmf* dtmf = NULL;
+VoERTP_RTCP* rtp_rtcp = NULL;
+VoEAudioProcessing* apm = NULL;
+VoENetwork* netw = NULL;
+VoEFile* file = NULL;
+VoEVideoSync* vsync = NULL;
+VoEEncryption* encr = NULL;
+VoEHardware* hardware = NULL;
+VoEExternalMedia* xmedia = NULL;
+VoENetEqStats* neteqst = NULL;
+
+void RunTest(std::string out_path);
+
+#ifdef EXTERNAL_TRANSPORT
+
+class my_transportation : public Transport
+{
+ int SendPacket(int channel,const void *data,int len);
+ int SendRTCPPacket(int channel, const void *data, int len);
+};
+
+int my_transportation::SendPacket(int channel,const void *data,int len)
+{
+ netw->ReceivedRTPPacket(channel, data, len);
+ return 0;
+}
+
+int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
+{
+ netw->ReceivedRTCPPacket(channel, data, len);
+ return 0;
+}
+
+my_transportation my_transport;
+#endif
+
+class MyObserver : public VoiceEngineObserver {
+ public:
+ virtual void CallbackOnError(const int channel, const int err_code);
+};
+
+void MyObserver::CallbackOnError(const int channel, const int err_code) {
+ // Add printf for other error codes here
+ if (err_code == VE_TYPING_NOISE_WARNING) {
+ printf(" TYPING NOISE DETECTED \n");
+ } else if (err_code == VE_RECEIVE_PACKET_TIMEOUT) {
+ printf(" RECEIVE PACKET TIMEOUT \n");
+ } else if (err_code == VE_PACKET_RECEIPT_RESTARTED) {
+ printf(" PACKET RECEIPT RESTARTED \n");
+ } else if (err_code == VE_RUNTIME_PLAY_WARNING) {
+ printf(" RUNTIME PLAY WARNING \n");
+ } else if (err_code == VE_RUNTIME_REC_WARNING) {
+ printf(" RUNTIME RECORD WARNING \n");
+ } else if (err_code == VE_SATURATION_WARNING) {
+ printf(" SATURATION WARNING \n");
+ } else if (err_code == VE_RUNTIME_PLAY_ERROR) {
+ printf(" RUNTIME PLAY ERROR \n");
+ } else if (err_code == VE_RUNTIME_REC_ERROR) {
+ printf(" RUNTIME RECORD ERROR \n");
+ } else if (err_code == VE_REC_DEVICE_REMOVED) {
+ printf(" RECORD DEVICE REMOVED \n");
+ }
+}
+
+int main() {
+ int res = 0;
+ int cnt = 0;
+
+ printf("Test started \n");
+
+ m_voe = VoiceEngine::Create();
+ base1 = VoEBase::GetInterface(m_voe);
+ codec = VoECodec::GetInterface(m_voe);
+ apm = VoEAudioProcessing::GetInterface(m_voe);
+ volume = VoEVolumeControl::GetInterface(m_voe);
+ dtmf = VoEDtmf::GetInterface(m_voe);
+ rtp_rtcp = VoERTP_RTCP::GetInterface(m_voe);
+ netw = VoENetwork::GetInterface(m_voe);
+ file = VoEFile::GetInterface(m_voe);
+ vsync = VoEVideoSync::GetInterface(m_voe);
+ encr = VoEEncryption::GetInterface(m_voe);
+ hardware = VoEHardware::GetInterface(m_voe);
+ xmedia = VoEExternalMedia::GetInterface(m_voe);
+ neteqst = VoENetEqStats::GetInterface(m_voe);
+
+ MyObserver my_observer;
+
+#if defined(WEBRTC_ANDROID)
+ const std::string out_path = "/sdcard/";
+#else
+ const std::string out_path = webrtc::test::OutputPath();
+#endif
+ const std::string trace_filename = out_path + "webrtc_trace.txt";
+
+ printf("Set trace filenames (enable trace)\n");
+ VoiceEngine::SetTraceFilter(kTraceAll);
+ res = VoiceEngine::SetTraceFile(trace_filename.c_str());
+ VALIDATE;
+
+ res = VoiceEngine::SetTraceCallback(NULL);
+ VALIDATE;
+
+ printf("Init\n");
+ res = base1->Init();
+ if (res != 0) {
+ printf("\nError calling Init: %d\n", base1->LastError());
+ fflush(NULL);
+ exit(1);
+ }
+
+ res = base1->RegisterVoiceEngineObserver(my_observer);
+ VALIDATE;
+
+ cnt++;
+ printf("Version\n");
+ char tmp[1024];
+ res = base1->GetVersion(tmp);
+ VALIDATE;
+ cnt++;
+ printf("%s\n", tmp);
+
+ RunTest(out_path);
+
+ printf("Terminate \n");
+
+ base1->DeRegisterVoiceEngineObserver();
+
+ res = base1->Terminate();
+ VALIDATE;
+
+ if (base1)
+ base1->Release();
+
+ if (codec)
+ codec->Release();
+
+ if (volume)
+ volume->Release();
+
+ if (dtmf)
+ dtmf->Release();
+
+ if (rtp_rtcp)
+ rtp_rtcp->Release();
+
+ if (apm)
+ apm->Release();
+
+ if (netw)
+ netw->Release();
+
+ if (file)
+ file->Release();
+
+ if (vsync)
+ vsync->Release();
+
+ if (encr)
+ encr->Release();
+
+ if (hardware)
+ hardware->Release();
+
+ if (xmedia)
+ xmedia->Release();
+
+ if (neteqst)
+ neteqst->Release();
+
+ VoiceEngine::Delete(m_voe);
+
+ return 0;
+}
+
+void RunTest(std::string out_path) {
+ int chan, cnt, res;
+ CodecInst cinst;
+ cnt = 0;
+ int i;
+ int codecinput;
+ bool AEC = false;
+ bool AGC = true;
+ bool AGC1 = false;
+ bool VAD = false;
+ bool NS = false;
+ bool NS1 = false;
+ bool typing_detection = false;
+ bool muted = false;
+ bool on_hold = false;
+
+#if defined(WEBRTC_ANDROID)
+ std::string resource_path = "/sdcard/";
+#else
+ std::string resource_path = webrtc::test::ProjectRootPath();
+ if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
+ printf("*** Unable to get project root directory. "
+ "File playing may fail. ***\n");
+ // Fall back to the current directory.
+ resource_path = "./";
+ } else {
+ resource_path += "test/data/voice_engine/";
+ }
+#endif
+ const std::string audio_filename = resource_path + "audio_long16.pcm";
+
+ const std::string play_filename = out_path + "recorded_playout.pcm";
+ const std::string mic_filename = out_path + "recorded_mic.pcm";
+
+ chan = base1->CreateChannel();
+ if (chan < 0) {
+ printf("Error at position %i\n", cnt);
+ printf("************ Error code = %i\n", base1->LastError());
+ fflush(NULL);
+ }
+ cnt++;
+
+ int j = 0;
+#ifdef EXTERNAL_TRANSPORT
+ my_transportation ch0transport;
+ printf("Enabling external transport \n");
+ netw->RegisterExternalTransport(0, ch0transport);
+#else
+ char ip[64];
+#ifdef DEBUG
+ strcpy(ip, "127.0.0.1");
+#else
+ char localip[64];
+ netw->GetLocalIP(localip);
+ printf("local IP:%s\n", localip);
+
+ printf("1. 127.0.0.1 \n");
+ printf("2. Specify IP \n");
+ ASSERT_EQ(1, scanf("%i", &i));
+
+ if (1 == i)
+ strcpy(ip, "127.0.0.1");
+ else {
+ printf("Specify remote IP: ");
+ ASSERT_EQ(1, scanf("%s", ip));
+ }
+#endif
+
+ int colons(0);
+ while (ip[j] != '\0' && j < 64 && !(colons = (ip[j++] == ':')))
+ ;
+ if (colons) {
+ printf("Enabling IPv6\n");
+ res = netw->EnableIPv6(0);
+ VALIDATE;
+ }
+
+ int rPort;
+#ifdef DEBUG
+ rPort=8500;
+#else
+ printf("Specify remote port (1=1234): ");
+ ASSERT_EQ(1, scanf("%i", &rPort));
+ if (1 == rPort)
+ rPort = 1234;
+ printf("Set Send port \n");
+#endif
+
+ printf("Set Send IP \n");
+ res = base1->SetSendDestination(chan, rPort, ip);
+ VALIDATE;
+
+ int lPort;
+#ifdef DEBUG
+ lPort=8500;
+#else
+ printf("Specify local port (1=1234): ");
+ ASSERT_EQ(1, scanf("%i", &lPort));
+ if (1 == lPort)
+ lPort = 1234;
+ printf("Set Rec Port \n");
+#endif
+ res = base1->SetLocalReceiver(chan, lPort);
+ VALIDATE;
+#endif
+
+ printf("\n");
+ for (i = 0; i < codec->NumOfCodecs(); i++) {
+ res = codec->GetCodec(i, cinst);
+ VALIDATE;
+ if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq == 32000) {
+ printf("%i. ISAC-swb pltype:%i plfreqi:%i\n", i, cinst.pltype,
+ cinst.plfreq);
+ }
+ else {
+ printf("%i. %s pltype:%i plfreq:%i\n", i, cinst.plname,
+ cinst.pltype, cinst.plfreq);
+ }
+ }
+#ifdef DEBUG
+ codecinput=0;
+#else
+ printf("Select send codec: ");
+ ASSERT_EQ(1, scanf("%i", &codecinput));
+#endif
+ codec->GetCodec(codecinput, cinst);
+
+ printf("Set primary codec\n");
+ res = codec->SetSendCodec(chan, cinst);
+ VALIDATE;
+
+ const int kMaxNumChannels = 8;
+ int channel_index = 0;
+ std::vector<int> channels(kMaxNumChannels);
+ for (i = 0; i < kMaxNumChannels; ++i) {
+ channels[i] = base1->CreateChannel();
+ int port = rPort + (i + 1) * 2;
+ res = base1->SetSendDestination(channels[i], port, ip);
+ VALIDATE;
+ res = base1->SetLocalReceiver(channels[i], port);
+ VALIDATE;
+ res = codec->SetSendCodec(channels[i], cinst);
+ VALIDATE;
+ }
+
+ // Call loop
+ bool newcall = true;
+ while (newcall) {
+
+#ifdef WEBRTC_LINUX
+ int rd(-1), pd(-1);
+ res = hardware->GetNumOfRecordingDevices(rd);
+ VALIDATE;
+ res = hardware->GetNumOfPlayoutDevices(pd);
+ VALIDATE;
+
+ char dn[128] = { 0 };
+ char guid[128] = { 0 };
+ printf("\nPlayout devices (%d): \n", pd);
+ for (j=0; j<pd; ++j) {
+ res = hardware->GetPlayoutDeviceName(j, dn, guid);
+ VALIDATE;
+ printf(" %d: %s \n", j, dn);
+ }
+
+ printf("Recording devices (%d): \n", rd);
+ for (j=0; j<rd; ++j) {
+ res = hardware->GetRecordingDeviceName(j, dn, guid);
+ VALIDATE;
+ printf(" %d: %s \n", j, dn);
+ }
+
+ printf("Select playout device: ");
+ ASSERT_EQ(1, scanf("%d", &pd));
+ res = hardware->SetPlayoutDevice(pd);
+ VALIDATE;
+ printf("Select recording device: ");
+ ASSERT_EQ(1, scanf("%d", &rd));
+ printf("Setting sound devices \n");
+ res = hardware->SetRecordingDevice(rd);
+ VALIDATE;
+
+#endif // WEBRTC_LINUX
+ res = codec->SetVADStatus(0, VAD);
+ VALIDATE;
+
+ res = apm->SetAgcStatus(AGC);
+ VALIDATE;
+
+ res = apm->SetEcStatus(AEC);
+ VALIDATE;
+
+ res = apm->SetNsStatus(NS);
+ VALIDATE;
+
+#ifdef DEBUG
+ i = 1;
+#else
+ printf("\n1. Send, listen and playout \n");
+ printf("2. Send only \n");
+ printf("3. Listen and playout only \n");
+ printf("Select transfer mode: ");
+ ASSERT_EQ(1, scanf("%i", &i));
+#endif
+ const bool send = !(3 == i);
+ const bool receive = !(2 == i);
+
+ if (receive) {
+#ifndef EXTERNAL_TRANSPORT
+ printf("Start Listen \n");
+ res = base1->StartReceive(chan);
+ VALIDATE;
+#endif
+
+ printf("Start Playout \n");
+ res = base1->StartPlayout(chan);
+ VALIDATE;
+ }
+
+ if (send) {
+ printf("Start Send \n");
+ res = base1->StartSend(chan);
+ VALIDATE;
+ }
+
+ printf("Getting mic volume \n");
+ unsigned int vol = 999;
+ res = volume->GetMicVolume(vol);
+ VALIDATE;
+ if ((vol > 255) || (vol < 1)) {
+ printf("\n****ERROR in GetMicVolume");
+ }
+
+ int forever = 1;
+ while (forever) {
+ printf("\nActions\n");
+
+ printf("Codec Changes\n");
+ for (i = 0; i < codec->NumOfCodecs(); i++) {
+ res = codec->GetCodec(i, cinst);
+ VALIDATE;
+ if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq
+ == 32000) {
+ printf("\t%i. ISAC-swb pltype:%i plfreq:%i\n", i,
+ cinst.pltype, cinst.plfreq);
+ }
+ else {
+ printf("\t%i. %s pltype:%i plfreq:%i\n", i, cinst.plname,
+ cinst.pltype, cinst.plfreq);
+ }
+ }
+ printf("Other\n");
+ const int noCodecs = i - 1;
+ printf("\t%i. Toggle VAD\n", i);
+ i++;
+ printf("\t%i. Toggle AGC\n", i);
+ i++;
+ printf("\t%i. Toggle NS\n", i);
+ i++;
+ printf("\t%i. Toggle EC\n", i);
+ i++;
+ printf("\t%i. Select AEC\n", i);
+ i++;
+ printf("\t%i. Select AECM\n", i);
+ i++;
+ printf("\t%i. Get speaker volume\n", i);
+ i++;
+ printf("\t%i. Set speaker volume\n", i);
+ i++;
+ printf("\t%i. Get microphone volume\n", i);
+ i++;
+ printf("\t%i. Set microphone volume\n", i);
+ i++;
+ printf("\t%i. Play local file (audio_long16.pcm) \n", i);
+ i++;
+ printf("\t%i. Change playout device \n", i);
+ i++;
+ printf("\t%i. Change recording device \n", i);
+ i++;
+ printf("\t%i. Toggle receive-side AGC \n", i);
+ i++;
+ printf("\t%i. Toggle receive-side NS \n", i);
+ i++;
+ printf("\t%i. AGC status \n", i);
+ i++;
+ printf("\t%i. Toggle microphone mute \n", i);
+ i++;
+ printf("\t%i. Toggle on hold status \n", i);
+ i++;
+ printf("\t%i. Get last error code \n", i);
+ i++;
+ printf("\t%i. Toggle typing detection (for Mac/Windows only) \n", i);
+ i++;
+ printf("\t%i. Record a PCM file \n", i);
+ i++;
+ printf("\t%i. Play a previously recorded PCM file locally \n", i);
+ i++;
+ printf("\t%i. Play a previously recorded PCM file as microphone \n", i);
+ i++;
+ printf("\t%i. Add an additional file-playing channel \n", i);
+ i++;
+ printf("\t%i. Remove a file-playing channel \n", i);
+ i++;
+
+ printf("Select action or %i to stop the call: ", i);
+ ASSERT_EQ(1, scanf("%i", &codecinput));
+
+ if (codecinput < codec->NumOfCodecs()) {
+ res = codec->GetCodec(codecinput, cinst);
+ VALIDATE;
+
+ printf("Set primary codec\n");
+ res = codec->SetSendCodec(chan, cinst);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 1)) {
+ VAD = !VAD;
+ res = codec->SetVADStatus(0, VAD);
+ VALIDATE;
+ if (VAD)
+ printf("\n VAD is now on! \n");
+ else
+ printf("\n VAD is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 2)) {
+ AGC = !AGC;
+ res = apm->SetAgcStatus(AGC);
+ VALIDATE;
+ if (AGC)
+ printf("\n AGC is now on! \n");
+ else
+ printf("\n AGC is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 3)) {
+ NS = !NS;
+ res = apm->SetNsStatus(NS);
+ VALIDATE;
+ if (NS)
+ printf("\n NS is now on! \n");
+ else
+ printf("\n NS is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 4)) {
+ AEC = !AEC;
+ res = apm->SetEcStatus(AEC, kEcUnchanged);
+ VALIDATE;
+ if (AEC)
+ printf("\n Echo control is now on! \n");
+ else
+ printf("\n Echo control is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 5)) {
+ res = apm->SetEcStatus(AEC, kEcAec);
+ VALIDATE;
+ printf("\n AEC selected! \n");
+ if (AEC)
+ printf(" (Echo control is on)\n");
+ else
+ printf(" (Echo control is off)\n");
+ }
+ else if (codecinput == (noCodecs + 6)) {
+ res = apm->SetEcStatus(AEC, kEcAecm);
+ VALIDATE;
+ printf("\n AECM selected! \n");
+ if (AEC)
+ printf(" (Echo control is on)\n");
+ else
+ printf(" (Echo control is off)\n");
+ }
+ else if (codecinput == (noCodecs + 7)) {
+ unsigned vol(0);
+ res = volume->GetSpeakerVolume(vol);
+ VALIDATE;
+ printf("\n Speaker Volume is %d \n", vol);
+ }
+ else if (codecinput == (noCodecs + 8)) {
+ printf("Level: ");
+ ASSERT_EQ(1, scanf("%i", &i));
+ res = volume->SetSpeakerVolume(i);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 9)) {
+ unsigned vol(0);
+ res = volume->GetMicVolume(vol);
+ VALIDATE;
+ printf("\n Microphone Volume is %d \n", vol);
+ }
+ else if (codecinput == (noCodecs + 10)) {
+ printf("Level: ");
+ ASSERT_EQ(1, scanf("%i", &i));
+ res = volume->SetMicVolume(i);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 11)) {
+ res = file->StartPlayingFileLocally(0, audio_filename.c_str());
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 12)) {
+ // change the playout device with current call
+ int num_pd(-1);
+ res = hardware->GetNumOfPlayoutDevices(num_pd);
+ VALIDATE;
+
+ char dn[128] = { 0 };
+ char guid[128] = { 0 };
+
+ printf("\nPlayout devices (%d): \n", num_pd);
+ for (j = 0; j < num_pd; ++j) {
+ res = hardware->GetPlayoutDeviceName(j, dn, guid);
+ VALIDATE;
+ printf(" %d: %s \n", j, dn);
+ }
+ printf("Select playout device: ");
+ ASSERT_EQ(1, scanf("%d", &num_pd));
+ // Will use plughw for hardware devices
+ res = hardware->SetPlayoutDevice(num_pd);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 13)) {
+ // change the recording device with current call
+ int num_rd(-1);
+
+ res = hardware->GetNumOfRecordingDevices(num_rd);
+ VALIDATE;
+
+ char dn[128] = { 0 };
+ char guid[128] = { 0 };
+
+ printf("Recording devices (%d): \n", num_rd);
+ for (j = 0; j < num_rd; ++j) {
+ res = hardware->GetRecordingDeviceName(j, dn, guid);
+ VALIDATE;
+ printf(" %d: %s \n", j, dn);
+ }
+
+ printf("Select recording device: ");
+ ASSERT_EQ(1, scanf("%d", &num_rd));
+ printf("Setting sound devices \n");
+ // Will use plughw for hardware devices
+ res = hardware->SetRecordingDevice(num_rd);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 14)) {
+ // Remote AGC
+ AGC1 = !AGC1;
+ res = apm->SetRxAgcStatus(chan, AGC1);
+ VALIDATE;
+ if (AGC1)
+ printf("\n Receive-side AGC is now on! \n");
+ else
+ printf("\n Receive-side AGC is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 15)) {
+ // Remote NS
+ NS1 = !NS1;
+ res = apm->SetRxNsStatus(chan, NS);
+ VALIDATE;
+ if (NS1)
+ printf("\n Receive-side NS is now on! \n");
+ else
+ printf("\n Receive-side NS is now off! \n");
+ }
+ else if (codecinput == (noCodecs + 16)) {
+ AgcModes agcmode;
+ bool enable;
+ res = apm->GetAgcStatus(enable, agcmode);
+ VALIDATE
+ printf("\n AGC enable is %d, mode is %d \n", enable, agcmode);
+ }
+ else if (codecinput == (noCodecs + 17)) {
+ // Toggle Mute on Microphone
+ res = volume->GetInputMute(chan, muted);
+ VALIDATE;
+ muted = !muted;
+ res = volume->SetInputMute(chan, muted);
+ VALIDATE;
+ if (muted)
+ printf("\n Microphone is now on mute! \n");
+ else
+ printf("\n Microphone is no longer on mute! \n");
+
+ }
+ else if (codecinput == (noCodecs + 18)) {
+ // Toggle the call on hold
+ OnHoldModes mode;
+ res = base1->GetOnHoldStatus(chan, on_hold, mode);
+ VALIDATE;
+ on_hold = !on_hold;
+ mode = kHoldSendAndPlay;
+ res = base1->SetOnHoldStatus(chan, on_hold, mode);
+ VALIDATE;
+ if (on_hold)
+ printf("\n Call now on hold! \n");
+ else
+ printf("\n Call now not on hold! \n");
+ }
+
+ else if (codecinput == (noCodecs + 19)) {
+ // Get the last error code and print to screen
+ int err_code = 0;
+ err_code = base1->LastError();
+ if (err_code != -1)
+ printf("\n The last error code was %i.\n", err_code);
+ }
+ else if (codecinput == (noCodecs + 20)) {
+ typing_detection= !typing_detection;
+ res = apm->SetTypingDetectionStatus(typing_detection);
+ VALIDATE;
+ if (typing_detection)
+ printf("\n Typing detection is now on!\n");
+ else
+ printf("\n Typing detection is now off!\n");
+ }
+ else if (codecinput == (noCodecs + 21)) {
+ int stop_record = 1;
+ int file_source = 1;
+ printf("\n Select source of recorded file. ");
+ printf("\n 1. Record from microphone to file ");
+ printf("\n 2. Record from playout to file ");
+ printf("\n Enter your selection: \n");
+ ASSERT_EQ(1, scanf("%i", &file_source));
+ if (file_source == 1) {
+ printf("\n Start recording microphone as %s \n",
+ mic_filename.c_str());
+ res = file->StartRecordingMicrophone(mic_filename.c_str());
+ VALIDATE;
+ }
+ else {
+ printf("\n Start recording playout as %s \n", play_filename.c_str());
+ res = file->StartRecordingPlayout(chan, play_filename.c_str());
+ VALIDATE;
+ }
+ while (stop_record != 0) {
+ printf("\n Type 0 to stop recording file \n");
+ ASSERT_EQ(1, scanf("%i", &stop_record));
+ }
+ if (file_source == 1) {
+ res = file->StopRecordingMicrophone();
+ VALIDATE;
+ }
+ else {
+ res = file->StopRecordingPlayout(chan);
+ VALIDATE;
+ }
+ printf("\n File finished recording \n");
+ }
+ else if (codecinput == (noCodecs + 22)) {
+ int file_type = 1;
+ int stop_play = 1;
+ printf("\n Select a file to play locally in a loop.");
+ printf("\n 1. Play %s", mic_filename.c_str());
+ printf("\n 2. Play %s", play_filename.c_str());
+ printf("\n Enter your selection\n");
+ ASSERT_EQ(1, scanf("%i", &file_type));
+ if (file_type == 1) {
+ printf("\n Start playing %s locally in a loop\n",
+ mic_filename.c_str());
+ res = file->StartPlayingFileLocally(chan, mic_filename.c_str(), true);
+ VALIDATE;
+ }
+ else {
+ printf("\n Start playing %s locally in a loop\n",
+ play_filename.c_str());
+ res = file->StartPlayingFileLocally(chan, play_filename.c_str(),
+ true);
+ VALIDATE;
+ }
+ while (stop_play != 0) {
+ printf("\n Type 0 to stop playing file\n");
+ ASSERT_EQ(1, scanf("%i", &stop_play));
+ }
+ res = file->StopPlayingFileLocally(chan);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 23)) {
+ int file_type = 1;
+ int stop_play = 1;
+ printf("\n Select a file to play as microphone in a loop.");
+ printf("\n 1. Play %s", mic_filename.c_str());
+ printf("\n 2. Play %s", play_filename.c_str());
+ printf("\n Enter your selection\n");
+ ASSERT_EQ(1, scanf("%i", &file_type));
+ if (file_type == 1) {
+ printf("\n Start playing %s as mic in a loop\n",
+ mic_filename.c_str());
+ res = file->StartPlayingFileAsMicrophone(chan, mic_filename.c_str(),
+ true);
+ VALIDATE;
+ }
+ else {
+ printf("\n Start playing %s as mic in a loop\n",
+ play_filename.c_str());
+ res = file->StartPlayingFileAsMicrophone(chan, play_filename.c_str(),
+ true);
+ VALIDATE;
+ }
+ while (stop_play != 0) {
+ printf("\n Type 0 to stop playing file\n");
+ ASSERT_EQ(1, scanf("%i", &stop_play));
+ }
+ res = file->StopPlayingFileAsMicrophone(chan);
+ VALIDATE;
+ }
+ else if (codecinput == (noCodecs + 24)) {
+ if (channel_index < kMaxNumChannels) {
+ res = base1->StartReceive(channels[channel_index]);
+ VALIDATE;
+ res = base1->StartPlayout(channels[channel_index]);
+ VALIDATE;
+ res = base1->StartSend(channels[channel_index]);
+ VALIDATE;
+ res = file->StartPlayingFileAsMicrophone(channels[channel_index],
+ audio_filename.c_str(),
+ true,
+ false);
+ VALIDATE;
+ channel_index++;
+ printf("Using %d additional channels\n", channel_index);
+ } else {
+ printf("Max number of channels reached\n");
+ }
+ }
+ else if (codecinput == (noCodecs + 25)) {
+ if (channel_index > 0) {
+ channel_index--;
+ res = file->StopPlayingFileAsMicrophone(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopSend(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopPlayout(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopReceive(channels[channel_index]);
+ VALIDATE;
+ printf("Using %d additional channels\n", channel_index);
+ } else {
+ printf("All additional channels stopped\n");
+ }
+ }
+ else
+ break;
+ }
+
+ if (send) {
+ printf("Stop Send \n");
+ res = base1->StopSend(chan);
+ VALIDATE;
+ }
+
+ if (receive) {
+ printf("Stop Playout \n");
+ res = base1->StopPlayout(chan);
+ VALIDATE;
+
+#ifndef EXTERNAL_TRANSPORT
+ printf("Stop Listen \n");
+ res = base1->StopReceive(chan);
+ VALIDATE;
+#endif
+ }
+
+ while (channel_index > 0) {
+ --channel_index;
+ res = file->StopPlayingFileAsMicrophone(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopSend(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopPlayout(channels[channel_index]);
+ VALIDATE;
+ res = base1->StopReceive(channels[channel_index]);
+ VALIDATE;
+ }
+
+ printf("\n1. New call \n");
+ printf("2. Quit \n");
+ printf("Select action: ");
+ ASSERT_EQ(1, scanf("%i", &i));
+ newcall = (1 == i);
+ // Call loop
+ }
+
+ printf("Delete channels \n");
+ res = base1->DeleteChannel(chan);
+ VALIDATE;
+
+ for (i = 0; i < kMaxNumChannels; ++i) {
+ channels[i] = base1->DeleteChannel(channels[i]);
+ VALIDATE;
+ }
+}
diff --git a/trunk/src/voice_engine/main/test/voice_engine_tests.gypi b/trunk/src/voice_engine/main/test/voice_engine_tests.gypi
new file mode 100644
index 0000000..ad78912
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/voice_engine_tests.gypi
@@ -0,0 +1,140 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ # Auto test - command line test for all platforms
+ {
+ 'target_name': 'voe_auto_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'voice_engine_core',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ ],
+ 'include_dirs': [
+ 'auto_test',
+ 'auto_test/fixtures',
+ '<(webrtc_root)/modules/interface',
+ # TODO(phoglund): We only depend on voice_engine_defines.h here -
+ # move that file to interface and then remove this dependency.
+ '<(webrtc_root)/voice_engine/main/source',
+ '<(webrtc_root)/modules/audio_device/main/interface',
+ ],
+ 'sources': [
+ 'auto_test/automated_mode.cc',
+ 'auto_test/fixtures/after_initialization_fixture.cc',
+ 'auto_test/fixtures/after_initialization_fixture.h',
+ 'auto_test/fixtures/after_streaming_fixture.cc',
+ 'auto_test/fixtures/after_streaming_fixture.h',
+ 'auto_test/fixtures/before_initialization_fixture.cc',
+ 'auto_test/fixtures/before_initialization_fixture.h',
+ 'auto_test/standard/audio_processing_test.cc',
+ 'auto_test/standard/call_report_test.cc',
+ 'auto_test/standard/codec_before_streaming_test.cc',
+ 'auto_test/standard/codec_test.cc',
+ 'auto_test/standard/dtmf_test.cc',
+ 'auto_test/standard/file_test.cc',
+ 'auto_test/standard/hardware_before_initializing_test.cc',
+ 'auto_test/standard/hardware_before_streaming_test.cc',
+ 'auto_test/standard/hardware_test.cc',
+ 'auto_test/standard/manual_hold_test.cc',
+ 'auto_test/standard/neteq_test.cc',
+ 'auto_test/standard/network_before_streaming_test.cc',
+ 'auto_test/standard/network_test.cc',
+ 'auto_test/standard/rtp_rtcp_before_streaming_test.cc',
+ 'auto_test/standard/rtp_rtcp_test.cc',
+ 'auto_test/standard/voe_base_misc_test.cc',
+ 'auto_test/standard/volume_test.cc',
+ 'auto_test/resource_manager.cc',
+ 'auto_test/voe_cpu_test.cc',
+ 'auto_test/voe_cpu_test.h',
+ 'auto_test/voe_extended_test.cc',
+ 'auto_test/voe_extended_test.h',
+ 'auto_test/voe_standard_test.cc',
+ 'auto_test/voe_standard_test.h',
+ 'auto_test/voe_stress_test.cc',
+ 'auto_test/voe_stress_test.h',
+ 'auto_test/voe_test_defines.h',
+ 'auto_test/voe_test_interface.h',
+ 'auto_test/voe_unit_test.cc',
+ 'auto_test/voe_unit_test.h',
+ ],
+ },
+ {
+ # command line test that should work on linux/mac/win
+ 'target_name': 'voe_cmd_test',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ 'voice_engine_core',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'sources': [
+ 'cmd_test/voe_cmd_test.cc',
+ ],
+ },
+ ],
+ 'conditions': [
+ ['OS=="win"', {
+ 'targets': [
+ # WinTest - GUI test for Windows
+ {
+ 'target_name': 'voe_ui_win_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'voice_engine_core',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ 'win_test',
+ ],
+ 'sources': [
+ 'win_test/Resource.h',
+ 'win_test/WinTest.cc',
+ 'win_test/WinTest.h',
+ 'win_test/WinTest.rc',
+ 'win_test/WinTestDlg.cc',
+ 'win_test/WinTestDlg.h',
+ 'win_test/res/WinTest.ico',
+ 'win_test/res/WinTest.rc2',
+ 'win_test/stdafx.cc',
+ 'win_test/stdafx.h',
+ ],
+ 'configurations': {
+ 'Common_Base': {
+ 'msvs_configuration_attributes': {
+ 'conditions': [
+ ['component=="shared_library"', {
+ 'UseOfMFC': '2', # Shared DLL
+ },{
+ 'UseOfMFC': '1', # Static
+ }],
+ ],
+ },
+ },
+ },
+ 'msvs_settings': {
+ 'VCLinkerTool': {
+ 'SubSystem': '2', # Windows
+ },
+ },
+ },
+ ],
+ }],
+ ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/voice_engine/main/test/win_test/Resource.h b/trunk/src/voice_engine/main/test/win_test/Resource.h
new file mode 100644
index 0000000..5ae9c5f
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/Resource.h
@@ -0,0 +1,241 @@
+//{{NO_DEPENDENCIES}}
+// Microsoft Visual C++ generated include file.
+// Used by WinTest.rc
+//
+#define IDM_ABOUTBOX 0x0010
+#define IDD_ABOUTBOX 100
+#define IDS_ABOUTBOX 101
+#define IDD_WINTEST_DIALOG 102
+#define IDR_MAINFRAME 128
+#define IDD_DTMF_DIALOG 129
+#define IDC_BUTTON_CREATE_1 1000
+#define IDC_BUTTON_DELETE_1 1001
+#define IDC_EDIT_1 1002
+#define IDC_BUTTON_CREATE_2 1003
+#define IDC_BUTTON_DELETE_2 1004
+#define IDC_EDIT_2 1005
+#define IDC_EDIT_MESSAGE 1006
+#define IDC_BUTTON_START_LISTEN_1 1007
+#define IDC_COMBO_IP_1 1008
+#define IDC_EDIT_TX_PORT_1 1009
+#define IDC_EDIT_RX_PORT_1 1010
+#define IDC_COMBO_CODEC_1 1011
+#define IDC_BUTTON_STOP_LISTEN_1 1012
+#define IDC_STATIC_LISTEN 1013
+#define IDC_BUTTON_START_PLAYOUT_1 1014
+#define IDC_BUTTON_STOP_PLAYOUT_1 1015
+#define IDC_STATIC_PLAY 1016
+#define IDC_BUTTON_START_SEND_1 1017
+#define IDC_BUTTON_STOP_SEND_1 1018
+#define IDC_STATIC_SEND 1019
+#define IDC_COMBO_IP_2 1020
+#define IDC_STATIC_IP 1021
+#define IDC_STATIC_PORTS 1022
+#define IDC_STATIC_CODEC 1023
+#define IDC_STATIC_CHANNEL 1024
+#define IDC_STATIC_ID 1025
+#define IDC_EDIT_TX_PORT_2 1026
+#define IDC_EDIT_RX_PORT_2 1027
+#define IDC_COMBO_CODEC_2 1028
+#define IDC_BUTTON_START_LISTEN_2 1029
+#define IDC_BUTTON_STOP_LISTEN_2 1030
+#define IDC_BUTTON_START_PLAYOUT_2 1031
+#define IDC_BUTTON_STOP_PLAYOUT_2 1032
+#define IDC_BUTTON_START_SEND_2 1033
+#define IDC_BUTTON_STOP_SEND_2 1034
+#define IDC_BUTTON_START_SEND_3 1035
+#define IDC_BUTTON_TEST_1_1 1035
+#define IDC_BUTTON_TEST_1 1035
+#define IDC_EDIT_RESULT 1036
+#define IDC_EDIT_N_FAILS 1037
+#define IDC_STATIC_ERROR 1038
+#define IDC_EDIT_LAST_ERROR 1039
+#define IDC_STATIC_LAST_ERROR 1040
+#define IDC_STATIC_PLAY_FILE 1041
+#define IDC_STATIC_EXTERNAL 1042
+#define IDC_CHECK_EXT_TRANS_1 1043
+#define IDC_CHECK2 1044
+#define IDC_CHECK_PLAY_FILE_IN_1 1044
+#define IDC_CHECK_PLAY_FILE_OUT_1 1045
+#define IDC_CHECK_PLAY_FILE_IN_2 1046
+#define IDC_CHECK_PLAY_FILE_OUT_2 1047
+#define IDC_CHECK_EXT_TRANS_2 1048
+#define IDC_STATIC_ALL_CHANNELS 1049
+#define IDC_CHECK_PLAY_FILE_IN 1050
+#define IDC_CHECK_PLAY_FILE_OUT 1051
+#define IDC_CHECK_EXT_MEDIA_IN_1 1051
+#define IDC_COMBO_REC_DEVICE 1052
+#define IDC_STATIC_REC_DEVICE 1053
+#define IDC_COMBO_PLAY_DEVICE2 1054
+#define IDC_COMBO_PLAY_DEVICE 1054
+#define IDC_STATIC_PLAY_DEVICE 1055
+#define IDC_CHECK_EXT_MEDIA_PLAY_1 1056
+#define IDC_CHECK_EXT_MEDIA_OUT_1 1056
+#define IDC_STATIC_PLAY_FILE2 1057
+#define IDC_SLIDER_INPUT_VOLUME 1058
+#define IDC_STATIC_MIC_VOLUME 1059
+#define IDC_SLIDER_OUTPUT_VOLUME 1060
+#define IDC_STATIC_SPK_VOLUME2 1061
+#define IDC_STATIC_SPK_VOLUME 1061
+#define IDC_CHECK_PLAY_FILE_IN2 1062
+#define IDC_CHECK_AGC 1062
+#define IDC_STATIC_MIC_VOLUME2 1063
+#define IDC_STATIC_AUDIO_LEVEL_IN 1063
+#define IDC_PROGRESS_AUDIO_LEVEL_IN 1064
+#define IDC_CHECK_AGC2 1065
+#define IDC_CHECK_NS 1065
+#define IDC_BUTTON_1 1065
+#define IDC_CHECK_VAD 1066
+#define IDC_CHECK_EXT_MEDIA_IN_2 1066
+#define IDC_BUTTON_2 1066
+#define IDC_CHECK_VAD2 1067
+#define IDC_CHECK_EC 1067
+#define IDC_BUTTON_3 1067
+#define IDC_CHECK_VAD_1 1068
+#define IDC_BUTTON_4 1068
+#define IDC_CHECK_VAD_2 1069
+#define IDC_CHECK_EXT_MEDIA_OUT_2 1069
+#define IDC_BUTTON_5 1069
+#define IDC_CHECK_VAD_3 1070
+#define IDC_BUTTON_6 1070
+#define IDC_CHECK_MUTE_IN 1071
+#define IDC_BUTTON_7 1071
+#define IDC_CHECK_MUTE_IN_1 1072
+#define IDC_BUTTON_8 1072
+#define IDC_CHECK_MUTE_IN_2 1073
+#define IDC_BUTTON_9 1073
+#define IDC_CHECK_SRTP_TX_1 1074
+#define IDC_BUTTON_10 1074
+#define IDC_CHECK_SRTP_RX_1 1075
+#define IDC_BUTTON_11 1075
+#define IDC_STATIC_PLAY_FILE3 1076
+#define IDC_STATIC_SRTP 1076
+#define IDC_BUTTON_12 1076
+#define IDC_CHECK_SRTP_TX_2 1077
+#define IDC_BUTTON_13 1077
+#define IDC_CHECK_SRTP_RX_2 1078
+#define IDC_BUTTON_14 1078
+#define IDC_CHECK_EXT_ENCRYPTION_1 1079
+#define IDC_BUTTON_15 1079
+#define IDC_STATIC_PLAY_FILE4 1080
+#define IDC_BUTTON_16 1080
+#define IDC_CHECK_EXT_ENCRYPTION_2 1081
+#define IDC_BUTTON_17 1081
+#define IDC_BUTTON_DTMF_1 1082
+#define IDC_BUTTON_18 1082
+#define IDC_EDIT_DTMF_EVENT 1083
+#define IDC_CHECK_REC_ 1083
+#define IDC_CHECK_REC_MIC 1083
+#define IDC_STATIC_DTMF_EVENT 1084
+#define IDC_BUTTON_DTMF_2 1084
+#define IDC_STATIC_GROUP_DTMF 1085
+#define IDC_CHECK_CONFERENCE_1 1085
+#define IDC_BUTTON_19 1086
+#define IDC_CHECK_CONFERENCE_2 1086
+#define IDC_BUTTON_20 1087
+#define IDC_CHECK_ON_HOLD_1 1087
+#define IDC_BUTTON_21 1088
+#define IDC_CHECK_ON_HOLD_2 1088
+#define IDC_BUTTON_22 1089
+#define IDC_CHECK_DTMF_PLAYOUT_RX 1089
+#define IDC_CHECK_EXT_MEDIA_IN 1089
+#define IDC_STATIC_PLAYOUT_RX 1090
+#define IDC_EDIT_GET_OUTPUT 1090
+#define IDC_CHECK_DTMF_PLAY_TONE 1091
+#define IDC_STATIC_LAST_ERROR2 1091
+#define IDC_STATIC_GET 1091
+#define IDC_STATIC_PLAY_TONE 1092
+#define IDC_CHECK_EXT_MEDIA_OUT 1092
+#define IDC_CHECK_START_STOP_MODE 1093
+#define IDC_BUTTON_SET_TX_TELEPHONE_PT 1093
+#define IDC_PROGRESS_AUDIO_LEVEL_IN2 1093
+#define IDC_PROGRESS_AUDIO_LEVEL_OUT 1093
+#define IDC_EDIT_EVENT_LENGTH 1094
+#define IDC_EDIT_RX_PORT_3 1094
+#define IDC_EDIT_DELAY_ESTIMATE_1 1094
+#define IDC_STATIC_EVENT_LENGTH 1095
+#define IDC_EDIT_PLAYOUT_BUFFER_SIZE 1095
+#define IDC_STATIC_START_STOP_MODE 1096
+#define IDC_EDIT_EVENT_RX_PT 1096
+#define IDC_CHECK_DELAY_ESTIMATE_1 1096
+#define IDC_EDIT_EVENT_ATTENUATION 1097
+#define IDC_CHECK_AGC_1 1097
+#define IDC_CHECK_EVENT_INBAND 1098
+#define IDC_CHECK_NS_1 1098
+#define IDC_STATIC_EVENT_ATTENUATION 1099
+#define IDC_STATIC_SRTP2 1099
+#define IDC_STATIC_RX_VQE 1099
+#define IDC_EDIT_EVENT_TX_PT 1100
+#define IDC_CHECK_REC_MIC2 1100
+#define IDC_CHECK_REC_CALL 1100
+#define IDC_CHECK_DTMF_FEEDBACK 1101
+#define IDC_CHECK_REC_CALL2 1101
+#define IDC_CHECK_TYPING_DETECTION 1101
+#define IDC_CHECK_START_STOP_MODE2 1102
+#define IDC_CHECK_DIRECT_FEEDBACK 1102
+#define IDC_CHECK_FEC 1102
+#define IDC_BUTTON_SET_RX_TELEPHONE_PT_TYPE 1103
+#define IDC_BUTTON_SET_RX_TELEPHONE_PT 1103
+#define IDC_BUTTON_CLEAR_ERROR_CALLBACK 1103
+#define IDC_EDIT_EVENT_CODE 1104
+#define IDC_STATIC_DIRECT_FEEDBACK 1105
+#define IDC_RADIO_SINGLE 1106
+#define IDC_RADIO_MULTI 1107
+#define IDC_RADIO_START_STOP 1108
+#define IDC_STATIC_MODE 1109
+#define IDC_STATIC_EVENT_RX_PT 1110
+#define IDC_STATIC_EVENT_TX_PT 1111
+#define IDC_STATIC_PT 1112
+#define IDC_BUTTON_SEND_TELEPHONE_EVENT 1113
+#define IDC_STATIC_EVENT_CODE 1114
+#define IDC_CHECK_EVENT_DETECTION 1115
+#define IDC_CHECK_DETECT_INBAND 1116
+#define IDC_CHECK_DETECT_OUT_OF_BAND 1117
+#define IDC_STATIC_INBAND_DETECTION 1118
+#define IDC_STATIC_OUT_OF_BAND_DETECTION 1119
+#define IDC_STATIC_EVENT_DETECTION 1120
+#define IDC_STATIC_TELEPHONE_EVENTS 1121
+#define IDC_EDIT_EVENT_CODE2 1122
+#define IDC_EDIT_ON_EVENT 1122
+#define IDC_EDIT_ON_EVENT_OUT_OF_BAND 1122
+#define IDC_STATIC_ON_EVENT 1123
+#define IDC_EDIT_ON_EVENT_INBAND 1123
+#define IDC_STATIC_EVEN 1124
+#define IDC_STATIC_LINE 1125
+#define IDC_LIST_CODEC_1 1128
+#define IDC_EDIT2 1129
+#define IDC_EDIT_CODEC_1 1129
+#define IDC_STATIC_PANNING 1131
+#define IDC_SLIDER_PAN_LEFT 1132
+#define IDC_SLIDER_PAN_RIGHT 1133
+#define IDC_STATIC_LEFT 1134
+#define IDC_STATIC_LEFT2 1135
+#define IDC_STATIC_RIGHT 1135
+#define IDC_BUTTON_VERSION 1136
+#define IDC_STATIC_PLAYOUT_BUFFER 1137
+#define IDC_CHECK_RXVAD 1138
+#define IDC_EDIT1 1139
+#define IDC_EDIT_RXVAD 1139
+#define IDC_STATIC_RX_PORT 1140
+#define IDC_STATIC_RX_PORT2 1141
+#define IDC_EDIT3 1142
+#define IDC_EDIT_AUDIO_LAYER 1142
+#define IDC_EDIT_AUDIO_LAYER2 1143
+#define IDC_EDIT_CPU_LOAD 1143
+#define IDC_STATIC_ERROR_CALLBACK 1144
+#define IDC_EDIT_ERROR_CALLBACK 1145
+#define IDC_EDIT_RX_CODEC_1 1146
+#define IDC_STATIC_BYTES_SENT_TEXT 1147
+#define IDC_EDIT_RTCP_STAT 1147
+#define IDC_EDIT_RTCP_STAT_1 1147
+
+// Next default values for new objects
+//
+#ifdef APSTUDIO_INVOKED
+#ifndef APSTUDIO_READONLY_SYMBOLS
+#define _APS_NEXT_RESOURCE_VALUE 130
+#define _APS_NEXT_COMMAND_VALUE 32771
+#define _APS_NEXT_CONTROL_VALUE 1148
+#define _APS_NEXT_SYMED_VALUE 101
+#endif
+#endif
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.aps b/trunk/src/voice_engine/main/test/win_test/WinTest.aps
new file mode 100644
index 0000000..499db5f
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTest.aps
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.cc b/trunk/src/voice_engine/main/test/win_test/WinTest.cc
new file mode 100644
index 0000000..e0e0248
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTest.cc
@@ -0,0 +1,75 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "stdafx.h"
+#include "WinTest.h"
+#include "WinTestDlg.h"
+
+#ifdef _DEBUG
+#define new DEBUG_NEW
+#endif
+
+
+// CWinTestApp
+
+BEGIN_MESSAGE_MAP(CWinTestApp, CWinApp)
+ ON_COMMAND(ID_HELP, &CWinApp::OnHelp)
+END_MESSAGE_MAP()
+
+
+// CWinTestApp construction
+
+CWinTestApp::CWinTestApp()
+{
+}
+
+
+// The one and only CWinTestApp object
+
+CWinTestApp theApp;
+
+
+// CWinTestApp initialization
+
+BOOL CWinTestApp::InitInstance()
+{
+ // InitCommonControlsEx() is required on Windows XP if an application
+ // manifest specifies use of ComCtl32.dll version 6 or later to enable
+ // visual styles. Otherwise, any window creation will fail.
+ INITCOMMONCONTROLSEX InitCtrls;
+ InitCtrls.dwSize = sizeof(InitCtrls);
+ // Set this to include all the common control classes you want to use
+ // in your application.
+ InitCtrls.dwICC = ICC_WIN95_CLASSES;
+ InitCommonControlsEx(&InitCtrls);
+
+ CWinApp::InitInstance();
+
+ // Standard initialization
+ // If you are not using these features and wish to reduce the size
+ // of your final executable, you should remove from the following
+ // the specific initialization routines you do not need
+ // Change the registry key under which our settings are stored
+ SetRegistryKey(_T("Local AppWizard-Generated Applications"));
+
+ CWinTestDlg dlg;
+ m_pMainWnd = &dlg;
+ INT_PTR nResponse = dlg.DoModal();
+ if (nResponse == IDOK)
+ {
+ }
+ else if (nResponse == IDCANCEL)
+ {
+ }
+
+ // Since the dialog has been closed, return FALSE so that we exit the
+ // application, rather than start the application's message pump.
+ return FALSE;
+}
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.h b/trunk/src/voice_engine/main/test/win_test/WinTest.h
new file mode 100644
index 0000000..d012ce6
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTest.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma once
+
+#ifndef __AFXWIN_H__
+ #error "include 'stdafx.h' before including this file for PCH"
+#endif
+
+#include "resource.h" // main symbols
+
+
+// CWinTestApp:
+// See WinTest.cpp for the implementation of this class
+//
+
+class CWinTestApp : public CWinApp
+{
+public:
+ CWinTestApp();
+
+// Overrides
+ public:
+ virtual BOOL InitInstance();
+
+// Implementation
+
+ DECLARE_MESSAGE_MAP()
+};
+
+extern CWinTestApp theApp;
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.rc b/trunk/src/voice_engine/main/test/win_test/WinTest.rc
new file mode 100644
index 0000000..dfe503f
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTest.rc
@@ -0,0 +1,394 @@
+// Microsoft Visual C++ generated resource script.
+//
+#include "resource.h"
+
+#define APSTUDIO_READONLY_SYMBOLS
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 2 resource.
+//
+#include "afxres.h"
+
+/////////////////////////////////////////////////////////////////////////////
+#undef APSTUDIO_READONLY_SYMBOLS
+
+/////////////////////////////////////////////////////////////////////////////
+// Swedish resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
+#ifdef _WIN32
+LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
+#pragma code_page(1252)
+#endif //_WIN32
+
+#ifdef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// TEXTINCLUDE
+//
+
+1 TEXTINCLUDE
+BEGIN
+ "resource.h\0"
+END
+
+2 TEXTINCLUDE
+BEGIN
+ "#include ""afxres.h""\r\n"
+ "\0"
+END
+
+3 TEXTINCLUDE
+BEGIN
+ "#define _AFX_NO_SPLITTER_RESOURCES\r\n"
+ "#define _AFX_NO_OLE_RESOURCES\r\n"
+ "#define _AFX_NO_TRACKER_RESOURCES\r\n"
+ "#define _AFX_NO_PROPERTY_RESOURCES\r\n"
+ "\r\n"
+ "#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)\r\n"
+ "LANGUAGE 29, 1\r\n"
+ "#pragma code_page(1252)\r\n"
+ "#include ""res\\WinTest.rc2"" // non-Microsoft Visual C++ edited resources\r\n"
+ "#include ""afxres.rc"" // Standard components\r\n"
+ "#endif\r\n"
+ "\0"
+END
+
+#endif // APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Icon
+//
+
+// Icon with lowest ID value placed first to ensure application icon
+// remains consistent on all systems.
+IDR_MAINFRAME ICON "res\\WinTest.ico"
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Dialog
+//
+
+IDD_ABOUTBOX DIALOGEX 0, 0, 235, 55
+STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
+CAPTION "About WinTest"
+FONT 8, "MS Shell Dlg", 0, 0, 0x1
+BEGIN
+ ICON IDR_MAINFRAME,IDC_STATIC,11,17,20,20
+ LTEXT "WinTest Version 1.0",IDC_STATIC,40,10,119,8,SS_NOPREFIX
+ LTEXT "Copyright (C) 2010",IDC_STATIC,40,25,119,8
+ DEFPUSHBUTTON "OK",IDOK,178,7,50,16,WS_GROUP
+END
+
+IDD_WINTEST_DIALOG DIALOGEX 0, 0, 796, 278
+STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_MINIMIZEBOX | WS_POPUP | WS_VISIBLE | WS_CAPTION | WS_SYSMENU
+EXSTYLE WS_EX_APPWINDOW
+CAPTION "WinTest"
+FONT 8, "MS Shell Dlg", 0, 0, 0x1
+BEGIN
+ PUSHBUTTON "Create",IDC_BUTTON_CREATE_1,28,24,32,14
+ PUSHBUTTON "Delete",IDC_BUTTON_DELETE_1,28,40,32,14
+ EDITTEXT IDC_EDIT_1,6,32,18,14,ES_AUTOHSCROLL | ES_READONLY
+ PUSHBUTTON "Create",IDC_BUTTON_CREATE_2,28,72,32,14
+ PUSHBUTTON "Delete",IDC_BUTTON_DELETE_2,28,88,32,14
+ EDITTEXT IDC_EDIT_2,6,82,18,14,ES_AUTOHSCROLL | ES_READONLY
+ EDITTEXT IDC_EDIT_MESSAGE,28,244,764,12,ES_AUTOHSCROLL
+ COMBOBOX IDC_COMBO_IP_1,64,24,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
+ EDITTEXT IDC_EDIT_TX_PORT_1,144,24,28,14,ES_AUTOHSCROLL
+ EDITTEXT IDC_EDIT_RX_PORT_1,144,40,28,14,ES_AUTOHSCROLL
+ COMBOBOX IDC_COMBO_CODEC_1,176,24,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ PUSHBUTTON "Start",IDC_BUTTON_START_LISTEN_1,256,24,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_LISTEN_1,256,40,32,14
+ LTEXT "Receive",IDC_STATIC_LISTEN,262,8,26,8
+ PUSHBUTTON "Start",IDC_BUTTON_START_PLAYOUT_1,292,24,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_PLAYOUT_1,292,40,32,14
+ LTEXT "Playout",IDC_STATIC_PLAY,295,8,25,8
+ PUSHBUTTON "Start",IDC_BUTTON_START_SEND_1,328,24,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_SEND_1,328,40,32,14
+ LTEXT "Send",IDC_STATIC_SEND,335,8,17,8
+ COMBOBOX IDC_COMBO_IP_2,64,72,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Destination IP address",IDC_STATIC_IP,64,8,73,8
+ LTEXT "Ports",IDC_STATIC_PORTS,145,8,18,8
+ LTEXT "Codec",IDC_STATIC_CODEC,177,8,21,8
+ LTEXT "Channel",IDC_STATIC_CHANNEL,30,8,27,8
+ LTEXT "ID",IDC_STATIC_ID,12,8,8,8
+ EDITTEXT IDC_EDIT_TX_PORT_2,144,72,28,14,ES_AUTOHSCROLL
+ EDITTEXT IDC_EDIT_RX_PORT_2,144,88,28,14,ES_AUTOHSCROLL
+ COMBOBOX IDC_COMBO_CODEC_2,176,72,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ PUSHBUTTON "Start",IDC_BUTTON_START_LISTEN_2,256,72,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_LISTEN_2,256,88,32,14
+ PUSHBUTTON "Start",IDC_BUTTON_START_PLAYOUT_2,292,72,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_PLAYOUT_2,292,88,32,14
+ PUSHBUTTON "Start",IDC_BUTTON_START_SEND_2,328,72,32,14
+ PUSHBUTTON "Stop",IDC_BUTTON_STOP_SEND_2,328,88,32,14
+ PUSHBUTTON "TEST 1",IDC_BUTTON_TEST_1,756,224,36,14
+ LTEXT "API",IDC_STATIC,4,247,12,8
+ EDITTEXT IDC_EDIT_RESULT,28,260,96,12,ES_AUTOHSCROLL
+ LTEXT "Result",IDC_STATIC,3,263,21,8
+ EDITTEXT IDC_EDIT_N_FAILS,156,260,30,12,ES_AUTOHSCROLL
+ LTEXT "#Fails",IDC_STATIC_ERROR,132,263,20,8
+ EDITTEXT IDC_EDIT_LAST_ERROR,228,260,36,12,ES_AUTOHSCROLL
+ LTEXT "Last Error",IDC_STATIC_LAST_ERROR,192,262,32,8
+ LTEXT "Ext. Trans.",IDC_STATIC_EXTERNAL,361,8,37,8
+ CONTROL "",IDC_CHECK_EXT_TRANS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,33,16,10
+ CONTROL "In",IDC_CHECK_PLAY_FILE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,24,36,14,WS_EX_DLGMODALFRAME
+ LTEXT "Play File",IDC_STATIC_PLAY_FILE,401,8,27,8
+ CONTROL "Out",IDC_CHECK_PLAY_FILE_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,40,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "In",IDC_CHECK_PLAY_FILE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,72,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Out",IDC_CHECK_PLAY_FILE_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,88,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "",IDC_CHECK_EXT_TRANS_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,82,16,10
+ GROUPBOX "",IDC_STATIC_ALL_CHANNELS,6,107,662,113
+ CONTROL "PlayFileAsMic",IDC_CHECK_PLAY_FILE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,122,60,14,WS_EX_DLGMODALFRAME
+ COMBOBOX IDC_COMBO_REC_DEVICE,12,132,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Recording device",IDC_STATIC_REC_DEVICE,12,120,56,8
+ COMBOBOX IDC_COMBO_PLAY_DEVICE,12,180,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
+ LTEXT "Playout device",IDC_STATIC_PLAY_DEVICE,12,167,56,8
+ CONTROL "In",IDC_CHECK_EXT_MEDIA_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,24,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Out",IDC_CHECK_EXT_MEDIA_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,40,36,14,WS_EX_DLGMODALFRAME
+ LTEXT "Ext. Media",IDC_STATIC_PLAY_FILE2,437,8,35,8
+ CONTROL "",IDC_SLIDER_INPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,130,72,15
+ LTEXT "Microphone Volume",IDC_STATIC_MIC_VOLUME,202,120,62,8
+ CONTROL "",IDC_SLIDER_OUTPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,179,72,15
+ LTEXT "Speaker Volume",IDC_STATIC_SPK_VOLUME,202,167,52,8
+ CONTROL "AGC",IDC_CHECK_AGC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,122,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "",IDC_PROGRESS_AUDIO_LEVEL_IN,"msctls_progress32",WS_BORDER,268,135,42,6
+ LTEXT "Audio Level",IDC_STATIC_AUDIO_LEVEL_IN,271,120,38,8,NOT WS_GROUP
+ CONTROL "NS",IDC_CHECK_NS,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,142,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "EC",IDC_CHECK_EC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,122,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "VAD",IDC_CHECK_VAD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,24,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "In",IDC_CHECK_EXT_MEDIA_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,72,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Out",IDC_CHECK_EXT_MEDIA_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,88,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "VAD",IDC_CHECK_VAD_3,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,72,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Mute",IDC_CHECK_MUTE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,142,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Mute",IDC_CHECK_MUTE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,40,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Mute",IDC_CHECK_MUTE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,88,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "TX",IDC_CHECK_SRTP_TX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,24,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "RX",IDC_CHECK_SRTP_RX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,40,36,14,WS_EX_DLGMODALFRAME
+ LTEXT "SRTP",IDC_STATIC_SRTP,525,8,18,8
+ CONTROL "TX",IDC_CHECK_SRTP_TX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,72,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "RX",IDC_CHECK_SRTP_RX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,88,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "",IDC_CHECK_EXT_ENCRYPTION_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,33,16,10
+ LTEXT "Encrypt",IDC_STATIC_PLAY_FILE4,556,8,26,8
+ CONTROL "",IDC_CHECK_EXT_ENCRYPTION_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,82,16,10
+ PUSHBUTTON "DTMF>>",IDC_BUTTON_DTMF_1,584,24,36,14
+ CONTROL "RecMicToFile",IDC_CHECK_REC_MIC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,142,60,14,WS_EX_DLGMODALFRAME
+ PUSHBUTTON "DTMF>>",IDC_BUTTON_DTMF_2,584,72,36,14
+ CONTROL "Conf",IDC_CHECK_CONFERENCE_1,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,40,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Conf",IDC_CHECK_CONFERENCE_2,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,88,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Hold",IDC_CHECK_ON_HOLD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,24,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "Hold",IDC_CHECK_ON_HOLD_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,72,36,14,WS_EX_DLGMODALFRAME
+ EDITTEXT IDC_EDIT_GET_OUTPUT,292,260,500,12,ES_AUTOHSCROLL
+ LTEXT "Get",IDC_STATIC_GET,276,262,12,8
+ CONTROL "Ext. Media",IDC_CHECK_EXT_MEDIA_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,122,52,14,WS_EX_DLGMODALFRAME
+ CONTROL "Ext. Media",IDC_CHECK_EXT_MEDIA_OUT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,180,52,14,WS_EX_DLGMODALFRAME
+ LISTBOX IDC_LIST_CODEC_1,208,40,44,28,LBS_NOINTEGRALHEIGHT | NOT WS_BORDER | WS_VSCROLL | WS_TABSTOP,WS_EX_CLIENTEDGE
+ EDITTEXT IDC_EDIT_CODEC_1,176,40,28,14,ES_AUTOHSCROLL
+ CONTROL "",IDC_PROGRESS_AUDIO_LEVEL_OUT,"msctls_progress32",WS_BORDER,268,184,42,6
+ LTEXT "Panning",IDC_STATIC_PANNING,328,167,26,8
+ CONTROL "",IDC_SLIDER_PAN_LEFT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,328,175,12,28
+ CONTROL "",IDC_SLIDER_PAN_RIGHT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,344,175,12,28
+ LTEXT "L",IDC_STATIC_LEFT,332,200,8,8
+ LTEXT "R",IDC_STATIC_RIGHT,347,201,8,8
+ PUSHBUTTON "Version",IDC_BUTTON_VERSION,624,200,36,14
+ EDITTEXT IDC_EDIT_PLAYOUT_BUFFER_SIZE,363,181,28,12,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
+ LTEXT "Buffer Size",IDC_STATIC_PLAYOUT_BUFFER,361,167,36,8
+ CONTROL "Delay",IDC_CHECK_DELAY_ESTIMATE_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,624,24,36,14,WS_EX_DLGMODALFRAME
+ EDITTEXT IDC_EDIT_DELAY_ESTIMATE_1,631,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
+ CONTROL "RxVAD",IDC_CHECK_RXVAD,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,664,24,40,14,WS_EX_DLGMODALFRAME
+ EDITTEXT IDC_EDIT_RXVAD,671,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY
+ CONTROL "AGC",IDC_CHECK_AGC_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,24,36,14,WS_EX_DLGMODALFRAME
+ CONTROL "NS",IDC_CHECK_NS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,40,36,14,WS_EX_DLGMODALFRAME
+ LTEXT "RX VQE",IDC_STATIC_RX_VQE,753,8,25,8
+ CONTROL "RecordCall",IDC_CHECK_REC_CALL,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,517,156,52,14,WS_EX_DLGMODALFRAME
+ LTEXT "RX",IDC_STATIC_RX_PORT,133,42,10,8
+ LTEXT "RX",IDC_STATIC_RX_PORT2,133,91,10,8
+ CONTROL "TypingDetect",IDC_CHECK_TYPING_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,572,156,60,14,WS_EX_DLGMODALFRAME
+ EDITTEXT IDC_EDIT_AUDIO_LAYER,28,224,116,14,ES_AUTOHSCROLL | ES_READONLY
+ EDITTEXT IDC_EDIT_CPU_LOAD,152,224,116,14,ES_AUTOHSCROLL | ES_READONLY
+ CONTROL "FEC",IDC_CHECK_FEC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,176,55,28,14,WS_EX_DLGMODALFRAME
+ LTEXT "=> Callbacks",IDC_STATIC_ERROR_CALLBACK,283,226,43,8
+ EDITTEXT IDC_EDIT_ERROR_CALLBACK,328,224,312,14,ES_AUTOHSCROLL
+ PUSHBUTTON "Clear",IDC_BUTTON_CLEAR_ERROR_CALLBACK,644,224,24,14
+ EDITTEXT IDC_EDIT_RX_CODEC_1,256,56,216,12,ES_AUTOHSCROLL | ES_READONLY
+ EDITTEXT IDC_EDIT_RTCP_STAT_1,476,56,316,12,ES_AUTOHSCROLL | ES_READONLY
+END
+
+IDD_DTMF_DIALOG DIALOGEX 0, 0, 316, 212
+STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
+CAPTION "Telehone Events"
+FONT 8, "MS Shell Dlg", 400, 0, 0x1
+BEGIN
+ DEFPUSHBUTTON "OK",IDOK,260,192,50,14
+ PUSHBUTTON "1",IDC_BUTTON_1,16,20,16,14
+ PUSHBUTTON "2",IDC_BUTTON_2,36,20,16,14
+ PUSHBUTTON "3",IDC_BUTTON_3,56,20,16,14
+ PUSHBUTTON "4",IDC_BUTTON_4,16,36,16,14
+ PUSHBUTTON "5",IDC_BUTTON_5,36,36,16,14
+ PUSHBUTTON "6",IDC_BUTTON_6,56,36,16,14
+ PUSHBUTTON "7",IDC_BUTTON_7,16,52,16,14
+ PUSHBUTTON "8",IDC_BUTTON_8,36,52,16,14
+ PUSHBUTTON "9",IDC_BUTTON_9,56,52,16,14
+ PUSHBUTTON "*",IDC_BUTTON_10,16,68,16,14
+ PUSHBUTTON "0",IDC_BUTTON_11,36,68,16,14
+ PUSHBUTTON "#",IDC_BUTTON_12,56,68,16,14
+ PUSHBUTTON "A",IDC_BUTTON_13,76,20,16,14
+ PUSHBUTTON "B",IDC_BUTTON_14,76,36,16,14
+ PUSHBUTTON "C",IDC_BUTTON_15,76,52,16,14
+ PUSHBUTTON "D",IDC_BUTTON_16,76,68,16,14
+ EDITTEXT IDC_EDIT_DTMF_EVENT,56,90,16,12,ES_AUTOHSCROLL | ES_READONLY
+ LTEXT "Event code",IDC_STATIC_DTMF_EVENT,17,91,37,8
+ PUSHBUTTON "1",IDC_BUTTON_17,16,20,16,14
+ PUSHBUTTON "2",IDC_BUTTON_18,36,20,16,14
+ PUSHBUTTON "3",IDC_BUTTON_19,56,20,16,14
+ PUSHBUTTON "4",IDC_BUTTON_20,16,36,16,14
+ PUSHBUTTON "A",IDC_BUTTON_21,76,20,16,14
+ GROUPBOX "DTMF Events",IDC_STATIC_GROUP_DTMF,4,4,188,132
+ CONTROL "",IDC_CHECK_DTMF_PLAYOUT_RX,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,21,12,14
+ LTEXT "Play out-band RX",IDC_STATIC_PLAYOUT_RX,101,24,56,8
+ CONTROL "",IDC_CHECK_DTMF_PLAY_TONE,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,39,12,14
+ LTEXT "Play tone locally",IDC_STATIC_PLAY_TONE,101,41,52,8
+ EDITTEXT IDC_EDIT_EVENT_LENGTH,44,163,28,14,ES_AUTOHSCROLL
+ LTEXT "Duration",IDC_STATIC_EVENT_LENGTH,12,165,28,8
+ EDITTEXT IDC_EDIT_EVENT_ATTENUATION,44,183,28,14,ES_AUTOHSCROLL
+ LTEXT "Volume",IDC_STATIC_EVENT_ATTENUATION,12,186,24,8
+ CONTROL "Inband",IDC_CHECK_EVENT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,84,163,40,14,WS_EX_DLGMODALFRAME
+ CONTROL "Feedback",IDC_CHECK_DTMF_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,16,112,48,14,WS_EX_DLGMODALFRAME
+ CONTROL "",IDC_CHECK_DIRECT_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,96,112,12,14
+ LTEXT "Direct",IDC_STATIC_DIRECT_FEEDBACK,72,115,20,8
+ CONTROL "Single",IDC_RADIO_SINGLE,"Button",BS_AUTORADIOBUTTON | WS_GROUP,112,68,35,10
+ CONTROL "Sequence",IDC_RADIO_MULTI,"Button",BS_AUTORADIOBUTTON,112,80,47,10
+ CONTROL "Start/Stop",IDC_RADIO_START_STOP,"Button",BS_AUTORADIOBUTTON,112,92,49,10
+ GROUPBOX "Mode",IDC_STATIC_MODE,100,56,68,52
+ EDITTEXT IDC_EDIT_EVENT_RX_PT,220,20,24,14,ES_AUTOHSCROLL
+ EDITTEXT IDC_EDIT_EVENT_TX_PT,220,41,24,14,ES_AUTOHSCROLL
+ LTEXT "RX",IDC_STATIC_EVENT_RX_PT,208,22,10,8
+ LTEXT "TX",IDC_STATIC_EVENT_TX_PT,208,42,9,8
+ PUSHBUTTON "Set",IDC_BUTTON_SET_TX_TELEPHONE_PT,248,41,24,14
+ PUSHBUTTON "Set",IDC_BUTTON_SET_RX_TELEPHONE_PT,248,20,24,14
+ GROUPBOX "Payload Type",IDC_STATIC_PT,200,4,80,56
+ EDITTEXT IDC_EDIT_EVENT_CODE,128,163,28,14,ES_AUTOHSCROLL
+ LTEXT "Event code",IDC_STATIC_EVENT_CODE,125,152,37,8
+ PUSHBUTTON "Send",IDC_BUTTON_SEND_TELEPHONE_EVENT,160,163,24,14
+ CONTROL "On/Off",IDC_CHECK_EVENT_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,80,40,14,WS_EX_DLGMODALFRAME
+ CONTROL "",IDC_CHECK_DETECT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,100,12,14
+ CONTROL "",IDC_CHECK_DETECT_OUT_OF_BAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,116,12,14
+ LTEXT "Inband",IDC_STATIC_INBAND_DETECTION,220,103,24,8
+ LTEXT "Outband",IDC_STATIC_OUT_OF_BAND_DETECTION,220,120,29,8
+ GROUPBOX "Event Detection",IDC_STATIC_EVENT_DETECTION,200,68,108,68
+ GROUPBOX "Telephone Events",IDC_STATIC_TELEPHONE_EVENTS,4,140,188,64
+ EDITTEXT IDC_EDIT_ON_EVENT_OUT_OF_BAND,252,117,48,14,ES_AUTOHSCROLL
+ EDITTEXT IDC_EDIT_ON_EVENT_INBAND,252,101,48,14,ES_AUTOHSCROLL
+ LTEXT "=> Detections",IDC_STATIC_EVEN,253,90,48,8
+END
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Version
+//
+
+VS_VERSION_INFO VERSIONINFO
+ FILEVERSION 1,0,0,0
+ PRODUCTVERSION 1,0,0,0
+ FILEFLAGSMASK 0x3fL
+#ifdef _DEBUG
+ FILEFLAGS 0x1L
+#else
+ FILEFLAGS 0x0L
+#endif
+ FILEOS 0x4L
+ FILETYPE 0x1L
+ FILESUBTYPE 0x0L
+BEGIN
+ BLOCK "StringFileInfo"
+ BEGIN
+ BLOCK "040904e4"
+ BEGIN
+ VALUE "FileDescription", "WebRTC VoiceEngine Test"
+ VALUE "FileVersion", "1.0.0.0"
+ VALUE "InternalName", "WinTest.exe"
+ VALUE "LegalCopyright", "Copyright (c) 2011 The WebRTC project authors. All Rights Reserved."
+ VALUE "OriginalFilename", "WinTest.exe"
+ VALUE "ProductName", "WebRTC VoiceEngine"
+ VALUE "ProductVersion", "1.0.0.0"
+ END
+ END
+ BLOCK "VarFileInfo"
+ BEGIN
+ VALUE "Translation", 0x409, 1252
+ END
+END
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// DESIGNINFO
+//
+
+#ifdef APSTUDIO_INVOKED
+GUIDELINES DESIGNINFO
+BEGIN
+ IDD_ABOUTBOX, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 228
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 48
+ END
+
+ IDD_WINTEST_DIALOG, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 789
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 271
+ END
+
+ IDD_DTMF_DIALOG, DIALOG
+ BEGIN
+ LEFTMARGIN, 7
+ RIGHTMARGIN, 309
+ TOPMARGIN, 7
+ BOTTOMMARGIN, 205
+ END
+END
+#endif // APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// String Table
+//
+
+STRINGTABLE
+BEGIN
+ IDS_ABOUTBOX "&About WinTest..."
+END
+
+#endif // Swedish resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+
+#ifndef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 3 resource.
+//
+#define _AFX_NO_SPLITTER_RESOURCES
+#define _AFX_NO_OLE_RESOURCES
+#define _AFX_NO_TRACKER_RESOURCES
+#define _AFX_NO_PROPERTY_RESOURCES
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
+LANGUAGE 29, 1
+#pragma code_page(1252)
+#include "res\WinTest.rc2" // non-Microsoft Visual C++ edited resources
+#include "afxres.rc" // Standard components
+#endif
+
+/////////////////////////////////////////////////////////////////////////////
+#endif // not APSTUDIO_INVOKED
+
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc
new file mode 100644
index 0000000..de54802
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc
@@ -0,0 +1,3573 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include "stdafx.h"
+#include "WinTest.h"
+#include "WinTestDlg.h"
+
+#ifdef _DEBUG
+#define new DEBUG_NEW
+#endif
+
+using namespace webrtc;
+
+unsigned char key[30] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9};
+
+// Hack to convert char to TCHAR, using two buffers to be able to
+// call twice in the same statement
+TCHAR convertTemp1[256] = {0};
+TCHAR convertTemp2[256] = {0};
+bool convertBufferSwitch(false);
+TCHAR* CharToTchar(const char* str, int len)
+{
+#ifdef _UNICODE
+ TCHAR* temp = convertBufferSwitch ? convertTemp1 : convertTemp2;
+ convertBufferSwitch = !convertBufferSwitch;
+ memset(temp, 0, sizeof(convertTemp1));
+ MultiByteToWideChar(CP_UTF8, 0, str, len, temp, 256);
+ return temp;
+#else
+ return str;
+#endif
+}
+
+// Hack to convert TCHAR to char
+char convertTemp3[256] = {0};
+char* TcharToChar(TCHAR* str, int len)
+{
+#ifdef _UNICODE
+ memset(convertTemp3, 0, sizeof(convertTemp3));
+ WideCharToMultiByte(CP_UTF8, 0, str, len, convertTemp3, 256, 0, 0);
+ return convertTemp3;
+#else
+ return str;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+// VoEConnectionObserver
+// ----------------------------------------------------------------------------
+
+class ConnectionObserver : public VoEConnectionObserver
+{
+public:
+ ConnectionObserver();
+ virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+};
+
+ConnectionObserver::ConnectionObserver()
+{
+}
+
+void ConnectionObserver::OnPeriodicDeadOrAlive(const int channel, const bool alive)
+{
+ CString str;
+ str.Format(_T("OnPeriodicDeadOrAlive(channel=%d) => alive=%d"), channel, alive);
+ OutputDebugString(str);
+}
+
+// ----------------------------------------------------------------------------
+// VoiceEngineObserver
+// ----------------------------------------------------------------------------
+
+void CWinTestDlg::CallbackOnError(const int channel, const int errCode)
+{
+ _nErrorCallbacks++;
+
+ CString str;
+ str.Format(_T("[#%d] CallbackOnError(channel=%d) => errCode = %d"), _nErrorCallbacks, channel, errCode);
+ if (errCode == VE_RECEIVE_PACKET_TIMEOUT)
+ {
+ str += _T(" <=> VE_RECEIVE_PACKET_TIMEOUT");
+ }
+ else if (errCode == VE_PACKET_RECEIPT_RESTARTED)
+ {
+ str += _T(" <=> VE_PACKET_RECEIPT_RESTARTED");
+ }
+ else if (errCode == VE_RUNTIME_PLAY_WARNING)
+ {
+ str += _T(" <=> VE_RUNTIME_PLAY_WARNING");
+ }
+ else if (errCode == VE_RUNTIME_REC_WARNING)
+ {
+ str += _T(" <=> VE_RUNTIME_REC_WARNING");
+ }
+ else if (errCode == VE_RUNTIME_PLAY_ERROR)
+ {
+ str += _T(" <=> VE_RUNTIME_PLAY_ERROR");
+ }
+ else if (errCode == VE_RUNTIME_REC_ERROR)
+ {
+ str += _T(" <=> VE_RUNTIME_REC_ERROR");
+ }
+ else if (errCode == VE_SATURATION_WARNING)
+ {
+ str += _T(" <=> VE_SATURATION_WARNING");
+ }
+ else if (errCode == VE_TYPING_NOISE_WARNING)
+ {
+ str += _T(" <=> VE_TYPING_NOISE_WARNING");
+ }
+ else if (errCode == VE_REC_DEVICE_REMOVED)
+ {
+ str += _T(" <=> VE_REC_DEVICE_REMOVED");
+ }
+ // AfxMessageBox((LPCTSTR)str, MB_OK);
+ SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
+}
+
+// ----------------------------------------------------------------------------
+// VoERTPObserver
+// ----------------------------------------------------------------------------
+
+void CWinTestDlg::OnIncomingCSRCChanged(const int channel, const unsigned int CSRC, const bool added)
+{
+ CString str;
+ str.Format(_T("OnIncomingCSRCChanged(channel=%d) => CSRC=%u, added=%d"), channel, CSRC, added);
+ SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
+}
+
+void CWinTestDlg::OnIncomingSSRCChanged(const int channel, const unsigned int SSRC)
+{
+ CString str;
+ str.Format(_T("OnIncomingSSRCChanged(channel=%d) => SSRC=%u"), channel, SSRC);
+ SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
+}
+
+// ----------------------------------------------------------------------------
+// Transport
+// ----------------------------------------------------------------------------
+
+class MyTransport : public Transport
+{
+public:
+ MyTransport(VoENetwork* veNetwork);
+ virtual int SendPacket(int channel, const void *data, int len);
+ virtual int SendRTCPPacket(int channel, const void *data, int len);
+private:
+ VoENetwork* _veNetworkPtr;
+};
+
+MyTransport::MyTransport(VoENetwork* veNetwork) :
+ _veNetworkPtr(veNetwork)
+{
+}
+
+int
+MyTransport::SendPacket(int channel, const void *data, int len)
+{
+ _veNetworkPtr->ReceivedRTPPacket(channel, data, len);
+ return len;
+}
+
+int
+MyTransport::SendRTCPPacket(int channel, const void *data, int len)
+{
+ _veNetworkPtr->ReceivedRTCPPacket(channel, data, len);
+ return len;
+}
+
+// ----------------------------------------------------------------------------
+// VoEMediaProcess
+// ----------------------------------------------------------------------------
+
+class MediaProcessImpl : public VoEMediaProcess
+{
+public:
+ MediaProcessImpl();
+ virtual void Process(const int channel,
+ const ProcessingTypes type,
+ WebRtc_Word16 audio_10ms[],
+ const int length,
+ const int samplingFreqHz,
+ const bool stereo);
+};
+
+MediaProcessImpl::MediaProcessImpl()
+{
+}
+
+void MediaProcessImpl::Process(const int channel,
+ const ProcessingTypes type,
+ WebRtc_Word16 audio_10ms[],
+ const int length,
+ const int samplingFreqHz,
+ const bool stereo)
+{
+ int x = rand() % 100;
+
+ for (int i = 0; i < length; i++)
+ {
+ if (channel == -1)
+ {
+ if (type == kPlaybackAllChannelsMixed)
+ {
+ // playout: scale up
+ if (!stereo)
+ {
+ audio_10ms[i] = (audio_10ms[i] << 2);
+ }
+ else
+ {
+ audio_10ms[2*i] = (audio_10ms[2*i] << 2);
+ audio_10ms[2*i+1] = (audio_10ms[2*i+1] << 2);
+ }
+ }
+ else
+ {
+ // recording: emulate packet loss by "dropping" 10% of the packets
+ if (x >= 0 && x < 10)
+ {
+ if (!stereo)
+ {
+ audio_10ms[i] = 0;
+ }
+ else
+ {
+ audio_10ms[2*i] = 0;
+ audio_10ms[2*i+1] = 0;
+ }
+ }
+ }
+ }
+ else
+ {
+ if (type == kPlaybackPerChannel)
+ {
+ // playout: mute
+ if (!stereo)
+ {
+ audio_10ms[i] = 0;
+ }
+ else
+ {
+ audio_10ms[2*i] = 0;
+ audio_10ms[2*i+1] = 0;
+ }
+ }
+ else
+ {
+ // recording: emulate packet loss by "dropping" 50% of the packets
+ if (x >= 0 && x < 50)
+ {
+ if (!stereo)
+ {
+ audio_10ms[i] = 0;
+ }
+ else
+ {
+ audio_10ms[2*i] = 0;
+ audio_10ms[2*i+1] = 0;
+ }
+ }
+ }
+ }
+ }
+}
+
+// ----------------------------------------------------------------------------
+// Encryptionen
+// ----------------------------------------------------------------------------
+
+class MyEncryption : public Encryption
+{
+public:
+ void encrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
+ void decrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
+ void encrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
+ void decrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
+};
+
+void MyEncryption::encrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
+{
+ // --- Stereo emulation (sample based, 2 bytes per sample)
+
+ const int nBytesPayload = bytes_in-12;
+
+ // RTP header (first 12 bytes)
+ memcpy(out_data, in_data, 12);
+
+ // skip RTP header
+ short* ptrIn = (short*) &in_data[12];
+ short* ptrOut = (short*) &out_data[12];
+
+ // network byte order
+ for (int i = 0; i < nBytesPayload/2; i++)
+ {
+ // produce two output samples for each input sample
+ *ptrOut++ = *ptrIn; // left sample
+ *ptrOut++ = *ptrIn; // right sample
+ ptrIn++;
+ }
+
+ *bytes_out = 12 + 2*nBytesPayload;
+
+ /*
+ for(int i = 0; i < bytes_in; i++)
+ out_data[i] =~ in_data[i];
+ *bytes_out = bytes_in;
+ */
+}
+
+void MyEncryption::decrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
+{
+ // Do nothing (<=> memcpy)
+ for(int i = 0; i < bytes_in; i++)
+ out_data[i] = in_data[i];
+ *bytes_out = bytes_in;
+}
+
+void MyEncryption::encrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
+{
+ for(int i = 0; i < bytes_in; i++)
+ out_data[i] =~ in_data[i];
+ *bytes_out = bytes_in;
+}
+
+void MyEncryption::decrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
+{
+ for(int i = 0; i < bytes_in; i++)
+ out_data[i] =~ in_data[i];
+ *bytes_out = bytes_in;
+}
+
+// ----------------------------------------------------------------------------
+// TelephoneEventObserver
+// ----------------------------------------------------------------------------
+
+class TelephoneEventObserver: public VoETelephoneEventObserver
+{
+public:
+ TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn);
+ virtual void OnReceivedTelephoneEventInband(int channel, int eventCode,
+ bool endOfEvent);
+ virtual void OnReceivedTelephoneEventOutOfBand(int channel, int eventCode,
+ bool endOfEvent);
+private:
+ CWnd* _editControlOutPtr;
+ CWnd* _editControlInPtr;
+};
+
+TelephoneEventObserver::TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn) :
+ _editControlOutPtr(editControlOut),
+ _editControlInPtr(editControlIn)
+{
+}
+
+void TelephoneEventObserver::OnReceivedTelephoneEventInband(int channel,
+ int eventCode,
+ bool endOfEvent)
+{
+ CString msg;
+ if (endOfEvent)
+ {
+ msg.AppendFormat(_T("%d [END]"), eventCode);
+ _editControlInPtr->SetWindowText((LPCTSTR)msg);
+ }
+ else
+ {
+ msg.AppendFormat(_T("%d [START]"), eventCode);
+ _editControlInPtr->SetWindowText((LPCTSTR)msg);
+ }
+}
+
+void TelephoneEventObserver::OnReceivedTelephoneEventOutOfBand(int channel,
+ int eventCode,
+ bool endOfEvent)
+{
+ CString msg;
+ if (endOfEvent)
+ {
+ msg.AppendFormat(_T("%d [END]"), eventCode);
+ _editControlOutPtr->SetWindowText((LPCTSTR)msg);
+ }
+ else
+ {
+ msg.AppendFormat(_T("%d [START]"), eventCode);
+ _editControlOutPtr->SetWindowText((LPCTSTR)msg);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// RxVadCallback
+// ----------------------------------------------------------------------------
+
+class RxCallback : public VoERxVadCallback
+{
+public:
+ RxCallback() : _vadDecision(-1) {};
+
+ virtual void OnRxVad(int , int vadDecision)
+ {
+ _vadDecision = vadDecision;
+ }
+
+ int _vadDecision;
+};
+
+// ----------------------------------------------------------------------------
+// CAboutDlg dialog
+// ----------------------------------------------------------------------------
+
+class CAboutDlg : public CDialog
+{
+public:
+ CAboutDlg();
+
+// Dialog Data
+ enum { IDD = IDD_ABOUTBOX };
+
+ protected:
+ virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
+
+// Implementation
+protected:
+ DECLARE_MESSAGE_MAP()
+};
+
+CAboutDlg::CAboutDlg() : CDialog(CAboutDlg::IDD)
+{
+}
+
+void CAboutDlg::DoDataExchange(CDataExchange* pDX)
+{
+ CDialog::DoDataExchange(pDX);
+}
+
+BEGIN_MESSAGE_MAP(CAboutDlg, CDialog)
+END_MESSAGE_MAP()
+
+// ----------------------------------------------------------------------------
+// CTelephonyEvent dialog
+// ----------------------------------------------------------------------------
+
+class CTelephonyEvent : public CDialog
+{
+ DECLARE_DYNAMIC(CTelephonyEvent)
+
+public:
+ CTelephonyEvent(VoiceEngine* voiceEngine, int channel, CDialog* pParentDialog, CWnd* pParent = NULL); // standard constructor
+ virtual ~CTelephonyEvent();
+
+// Dialog Data
+ enum { IDD = IDD_DTMF_DIALOG };
+
+protected:
+ virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
+ virtual BOOL OnInitDialog();
+
+ DECLARE_MESSAGE_MAP()
+public:
+ afx_msg void OnBnClickedButton1();
+ afx_msg void OnBnClickedButton2();
+ afx_msg void OnBnClickedButton3();
+ afx_msg void OnBnClickedButton4();
+ afx_msg void OnBnClickedButton5();
+ afx_msg void OnBnClickedButton6();
+ afx_msg void OnBnClickedButton7();
+ afx_msg void OnBnClickedButton8();
+ afx_msg void OnBnClickedButton9();
+ afx_msg void OnBnClickedButton10();
+ afx_msg void OnBnClickedButton11();
+ afx_msg void OnBnClickedButton12();
+ afx_msg void OnBnClickedButtonA();
+ afx_msg void OnBnClickedButtonB();
+ afx_msg void OnBnClickedButtonC();
+ afx_msg void OnBnClickedButtonD();
+ afx_msg void OnBnClickedCheckDtmfPlayoutRx();
+ afx_msg void OnBnClickedCheckDtmfPlayTone();
+ afx_msg void OnBnClickedCheckStartStopMode();
+ afx_msg void OnBnClickedCheckEventInband();
+ afx_msg void OnBnClickedCheckDtmfFeedback();
+ afx_msg void OnBnClickedCheckDirectFeedback();
+ afx_msg void OnBnClickedRadioSingle();
+ afx_msg void OnBnClickedRadioMulti();
+ afx_msg void OnBnClickedRadioStartStop();
+ afx_msg void OnBnClickedButtonSetRxTelephonePt();
+ afx_msg void OnBnClickedButtonSetTxTelephonePt();
+ afx_msg void OnBnClickedButtonSendTelephoneEvent();
+ afx_msg void OnBnClickedCheckDetectInband();
+ afx_msg void OnBnClickedCheckDetectOutOfBand();
+ afx_msg void OnBnClickedCheckEventDetection();
+
+private:
+ void SendTelephoneEvent(unsigned char eventCode);
+
+private:
+ VoiceEngine* _vePtr;
+ VoEBase* _veBasePtr;
+ VoEDtmf* _veDTMFPtr;
+ VoECodec* _veCodecPtr;
+ int _channel;
+ CString _strMsg;
+ CDialog* _parentDialogPtr;
+ TelephoneEventObserver* _telephoneEventObserverPtr;
+ bool _PlayDtmfToneLocally;
+ bool _modeStartStop;
+ bool _modeSingle;
+ bool _modeSequence;
+ bool _playingDTMFTone;
+ bool _outOfBandEventDetection;
+ bool _inbandEventDetection;
+};
+
+IMPLEMENT_DYNAMIC(CTelephonyEvent, CDialog)
+
+CTelephonyEvent::CTelephonyEvent(VoiceEngine* voiceEngine,
+ int channel,
+ CDialog* pParentDialog,
+ CWnd* pParent /*=NULL*/)
+ : _vePtr(voiceEngine),
+ _channel(channel),
+ _PlayDtmfToneLocally(false),
+ _modeStartStop(false),
+ _modeSingle(true),
+ _modeSequence(false),
+ _playingDTMFTone(false),
+ _outOfBandEventDetection(true),
+ _inbandEventDetection(false),
+ _parentDialogPtr(pParentDialog),
+ _telephoneEventObserverPtr(NULL),
+ CDialog(CTelephonyEvent::IDD, pParent)
+{
+ _veBasePtr = VoEBase::GetInterface(_vePtr);
+ _veDTMFPtr = VoEDtmf::GetInterface(_vePtr);
+ _veCodecPtr = VoECodec::GetInterface(_vePtr);
+}
+
+CTelephonyEvent::~CTelephonyEvent()
+{
+ _veDTMFPtr->Release();
+ _veCodecPtr->Release();
+ _veBasePtr->Release();
+
+ if (_telephoneEventObserverPtr)
+ {
+ _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
+ delete _telephoneEventObserverPtr;
+ _telephoneEventObserverPtr = NULL;
+ }
+}
+
+void CTelephonyEvent::DoDataExchange(CDataExchange* pDX)
+{
+ CDialog::DoDataExchange(pDX);
+}
+
+
+BEGIN_MESSAGE_MAP(CTelephonyEvent, CDialog)
+ ON_BN_CLICKED(IDC_BUTTON_1, &CTelephonyEvent::OnBnClickedButton1)
+ ON_BN_CLICKED(IDC_BUTTON_2, &CTelephonyEvent::OnBnClickedButton2)
+ ON_BN_CLICKED(IDC_BUTTON_3, &CTelephonyEvent::OnBnClickedButton3)
+ ON_BN_CLICKED(IDC_BUTTON_4, &CTelephonyEvent::OnBnClickedButton4)
+ ON_BN_CLICKED(IDC_BUTTON_5, &CTelephonyEvent::OnBnClickedButton5)
+ ON_BN_CLICKED(IDC_BUTTON_6, &CTelephonyEvent::OnBnClickedButton6)
+ ON_BN_CLICKED(IDC_BUTTON_7, &CTelephonyEvent::OnBnClickedButton7)
+ ON_BN_CLICKED(IDC_BUTTON_8, &CTelephonyEvent::OnBnClickedButton8)
+ ON_BN_CLICKED(IDC_BUTTON_9, &CTelephonyEvent::OnBnClickedButton9)
+ ON_BN_CLICKED(IDC_BUTTON_10, &CTelephonyEvent::OnBnClickedButton10)
+ ON_BN_CLICKED(IDC_BUTTON_11, &CTelephonyEvent::OnBnClickedButton11)
+ ON_BN_CLICKED(IDC_BUTTON_12, &CTelephonyEvent::OnBnClickedButton12)
+ ON_BN_CLICKED(IDC_BUTTON_13, &CTelephonyEvent::OnBnClickedButtonA)
+ ON_BN_CLICKED(IDC_BUTTON_14, &CTelephonyEvent::OnBnClickedButtonB)
+ ON_BN_CLICKED(IDC_BUTTON_15, &CTelephonyEvent::OnBnClickedButtonC)
+ ON_BN_CLICKED(IDC_BUTTON_16, &CTelephonyEvent::OnBnClickedButtonD)
+ ON_BN_CLICKED(IDC_CHECK_DTMF_PLAYOUT_RX, &CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx)
+ ON_BN_CLICKED(IDC_CHECK_DTMF_PLAY_TONE, &CTelephonyEvent::OnBnClickedCheckDtmfPlayTone)
+ ON_BN_CLICKED(IDC_CHECK_EVENT_INBAND, &CTelephonyEvent::OnBnClickedCheckEventInband)
+ ON_BN_CLICKED(IDC_CHECK_DTMF_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDtmfFeedback)
+ ON_BN_CLICKED(IDC_CHECK_DIRECT_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDirectFeedback)
+ ON_BN_CLICKED(IDC_RADIO_SINGLE, &CTelephonyEvent::OnBnClickedRadioSingle)
+ ON_BN_CLICKED(IDC_RADIO_MULTI, &CTelephonyEvent::OnBnClickedRadioMulti)
+ ON_BN_CLICKED(IDC_RADIO_START_STOP, &CTelephonyEvent::OnBnClickedRadioStartStop)
+ ON_BN_CLICKED(IDC_BUTTON_SET_RX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt)
+ ON_BN_CLICKED(IDC_BUTTON_SET_TX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt)
+ ON_BN_CLICKED(IDC_BUTTON_SEND_TELEPHONE_EVENT, &CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent)
+ ON_BN_CLICKED(IDC_CHECK_DETECT_INBAND, &CTelephonyEvent::OnBnClickedCheckDetectInband)
+ ON_BN_CLICKED(IDC_CHECK_DETECT_OUT_OF_BAND, &CTelephonyEvent::OnBnClickedCheckDetectOutOfBand)
+ ON_BN_CLICKED(IDC_CHECK_EVENT_DETECTION, &CTelephonyEvent::OnBnClickedCheckEventDetection)
+END_MESSAGE_MAP()
+
+
+// CTelephonyEvent message handlers
+
+BOOL CTelephonyEvent::OnInitDialog()
+{
+ CDialog::OnInitDialog();
+
+ CString str;
+ GetWindowText(str);
+ str.AppendFormat(_T(" [channel = %d]"), _channel);
+ SetWindowText(str);
+
+ // Update dialog with latest playout state
+ bool enabled(false);
+ _veDTMFPtr->GetDtmfPlayoutStatus(_channel, enabled);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
+ button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
+
+ // Update dialog with latest feedback state
+ bool directFeedback(false);
+ _veDTMFPtr->GetDtmfFeedbackStatus(enabled, directFeedback);
+ button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
+ button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
+ button->SetCheck(directFeedback ? BST_CHECKED : BST_UNCHECKED);
+
+ // Default event length is 160 ms
+ SetDlgItemInt(IDC_EDIT_EVENT_LENGTH, 160);
+
+ // Default event attenuation is 10 (<-> -10dBm0)
+ SetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, 10);
+
+ // Current event-detection status
+ TelephoneEventDetectionMethods detectionMethod(kOutOfBand);
+ if (_veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod) == 0)
+ {
+ // DTMF detection is supported
+ if (enabled)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
+ button->SetCheck(BST_CHECKED);
+ }
+ if (detectionMethod == kOutOfBand || detectionMethod == kInAndOutOfBand)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
+ button->SetCheck(BST_CHECKED);
+ }
+ if (detectionMethod == kInBand || detectionMethod == kInAndOutOfBand)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
+ button->SetCheck(BST_CHECKED);
+ }
+ }
+ else
+ {
+ // DTMF detection is not supported
+ GetDlgItem(IDC_CHECK_EVENT_DETECTION)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_DETECT_INBAND)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_ON_EVENT_INBAND)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND)->EnableWindow(FALSE);
+ }
+
+ // Telephone-event PTs
+ unsigned char pt(0);
+ _veDTMFPtr->GetSendTelephoneEventPayloadType(_channel, pt);
+ SetDlgItemInt(IDC_EDIT_EVENT_TX_PT, pt);
+
+ CodecInst codec;
+ strcpy_s(codec.plname, 32, "telephone-event"); codec.channels = 1; codec.plfreq = 8000;
+ _veCodecPtr->GetRecPayloadType(_channel, codec);
+ SetDlgItemInt(IDC_EDIT_EVENT_RX_PT, codec.pltype);
+
+ if (_modeSingle)
+ {
+ ((CButton*)GetDlgItem(IDC_RADIO_SINGLE))->SetCheck(BST_CHECKED);
+ }
+ else if (_modeStartStop)
+ {
+ ((CButton*)GetDlgItem(IDC_RADIO_START_STOP))->SetCheck(BST_CHECKED);
+ }
+ else if (_modeSequence)
+ {
+ ((CButton*)GetDlgItem(IDC_RADIO_MULTI))->SetCheck(BST_CHECKED);
+ }
+
+ return TRUE; // return TRUE unless you set the focus to a control
+}
+void CTelephonyEvent::SendTelephoneEvent(unsigned char eventCode)
+{
+ BOOL ret;
+ int lengthMs(0);
+ int attenuationDb(0);
+ bool outBand(false);
+ int res(0);
+
+ // tone length
+ if (!_modeStartStop)
+ {
+ lengthMs = GetDlgItemInt(IDC_EDIT_EVENT_LENGTH, &ret);
+ if (ret == FALSE)
+ {
+ // use default length if edit field is empty
+ lengthMs = 160;
+ }
+ }
+
+ // attenuation
+ attenuationDb = GetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, &ret);
+ if (ret == FALSE)
+ {
+ // use default length if edit field is empty
+ attenuationDb = 10;
+ }
+
+ // out-band or in-band
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
+ int check = button->GetCheck();
+ outBand = (check == BST_UNCHECKED);
+
+ if (eventCode < 16)
+ SetDlgItemInt(IDC_EDIT_DTMF_EVENT, eventCode);
+
+ if (_PlayDtmfToneLocally)
+ {
+ // --- PlayDtmfTone
+
+ if (_modeSingle)
+ {
+ TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
+ _T("PlayDtmfTone(eventCode=%u, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
+ }
+ else if (_modeStartStop)
+ {
+ if (!_playingDTMFTone)
+ {
+ TEST2((res = _veDTMFPtr->StartPlayingDtmfTone(eventCode, attenuationDb)) == 0,
+ _T("StartPlayingDtmfTone(eventCode=%u, attenuationDb=%d)"), eventCode, attenuationDb);
+ }
+ else
+ {
+ TEST2((res = _veDTMFPtr->StopPlayingDtmfTone()) == 0,
+ _T("StopPlayingDTMFTone()"));
+ }
+ if (res == 0)
+ _playingDTMFTone = !_playingDTMFTone;
+ }
+ else if (_modeSequence)
+ {
+ int nTones(1);
+ int sleepMs(0);
+ int lenMult(1);
+ if (eventCode == 1)
+ {
+ nTones = 2;
+ sleepMs = lengthMs;
+ lenMult = 1;
+ }
+ else if (eventCode == 2)
+ {
+ nTones = 2;
+ sleepMs = lengthMs/2;
+ lenMult = 2;
+ }
+ else if (eventCode == 3)
+ {
+ nTones = 3;
+ sleepMs = 0;
+ lenMult = 1;
+ }
+ for (int i = 0; i < nTones; i++)
+ {
+ TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
+ _T("PlayDtmfTone(eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
+ Sleep(sleepMs);
+ lengthMs = lenMult*lengthMs;
+ eventCode++;
+ }
+ }
+ }
+ else
+ {
+ // --- SendTelephoneEvent
+
+ if (_modeSingle)
+ {
+ TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
+ _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
+ }
+ else if (_modeStartStop)
+ {
+ TEST2(false, _T("*** NOT IMPLEMENTED ***"));
+ }
+ else if (_modeSequence)
+ {
+ int nTones(1);
+ int sleepMs(0);
+ int lenMult(1);
+ if (eventCode == 1)
+ {
+ nTones = 2;
+ sleepMs = lengthMs;
+ lenMult = 1;
+ }
+ else if (eventCode == 2)
+ {
+ eventCode = 1;
+ nTones = 2;
+ sleepMs = lengthMs/2;
+ lenMult = 2;
+ }
+ else if (eventCode == 3)
+ {
+ eventCode = 1;
+ nTones = 3;
+ sleepMs = 0;
+ lenMult = 1;
+ }
+ for (int i = 0; i < nTones; i++)
+ {
+ TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
+ _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
+ Sleep(sleepMs);
+ lengthMs = lenMult*lengthMs;
+ eventCode++;
+ }
+ }
+ }
+}
+
+void CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent()
+{
+ BOOL ret;
+ unsigned char eventCode(0);
+
+ eventCode = (unsigned char)GetDlgItemInt(IDC_EDIT_EVENT_CODE, &ret);
+ if (ret == FALSE)
+ {
+ return;
+ }
+ SendTelephoneEvent(eventCode);
+}
+
+void CTelephonyEvent::OnBnClickedButton1()
+{
+ SendTelephoneEvent(1);
+}
+
+void CTelephonyEvent::OnBnClickedButton2()
+{
+ SendTelephoneEvent(2);
+}
+
+void CTelephonyEvent::OnBnClickedButton3()
+{
+ SendTelephoneEvent(3);
+}
+
+void CTelephonyEvent::OnBnClickedButton4()
+{
+ SendTelephoneEvent(4);
+}
+
+void CTelephonyEvent::OnBnClickedButton5()
+{
+ SendTelephoneEvent(5);
+}
+
+void CTelephonyEvent::OnBnClickedButton6()
+{
+ SendTelephoneEvent(6);
+}
+
+void CTelephonyEvent::OnBnClickedButton7()
+{
+ SendTelephoneEvent(7);
+}
+
+void CTelephonyEvent::OnBnClickedButton8()
+{
+ SendTelephoneEvent(8);
+}
+
+void CTelephonyEvent::OnBnClickedButton9()
+{
+ SendTelephoneEvent(9);
+}
+
+void CTelephonyEvent::OnBnClickedButton10()
+{
+ // *
+ SendTelephoneEvent(10);
+}
+
+void CTelephonyEvent::OnBnClickedButton11()
+{
+ SendTelephoneEvent(0);
+}
+
+void CTelephonyEvent::OnBnClickedButton12()
+{
+ // #
+ SendTelephoneEvent(11);
+}
+
+void CTelephonyEvent::OnBnClickedButtonA()
+{
+ SendTelephoneEvent(12);
+}
+
+void CTelephonyEvent::OnBnClickedButtonB()
+{
+ SendTelephoneEvent(13);
+}
+
+void CTelephonyEvent::OnBnClickedButtonC()
+{
+ SendTelephoneEvent(14);
+}
+
+void CTelephonyEvent::OnBnClickedButtonD()
+{
+ SendTelephoneEvent(15);
+}
+
+void CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ TEST2(_veDTMFPtr->SetDtmfPlayoutStatus(_channel, enable) == 0, _T("SetDtmfPlayoutStatus(channel=%d, enable=%d)"), _channel, enable);
+}
+
+void CTelephonyEvent::OnBnClickedCheckDtmfPlayTone()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAY_TONE);
+ int check = button->GetCheck();
+ _PlayDtmfToneLocally = (check == BST_CHECKED);
+}
+
+void CTelephonyEvent::OnBnClickedRadioSingle()
+{
+ _modeStartStop = false;
+ _modeSingle = true;
+ _modeSequence = false;
+}
+
+void CTelephonyEvent::OnBnClickedRadioMulti()
+{
+ _modeStartStop = false;
+ _modeSingle = false;
+ _modeSequence = true;
+}
+
+void CTelephonyEvent::OnBnClickedRadioStartStop()
+{
+ // CButton* button = (CButton*)GetDlgItem(IDC_RADIO_START_STOP);
+ // int check = button->GetCheck();
+ _modeStartStop = true;
+ _modeSingle = false;
+ _modeSequence = false;
+ // GetDlgItem(IDC_EDIT_EVENT_LENGTH)->EnableWindow();
+}
+
+void CTelephonyEvent::OnBnClickedCheckEventInband()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
+ int check = button->GetCheck();
+ GetDlgItem(IDC_EDIT_EVENT_CODE)->EnableWindow(check?FALSE:TRUE);
+ GetDlgItem(IDC_BUTTON_SEND_TELEPHONE_EVENT)->EnableWindow(check?FALSE:TRUE);
+}
+
+void CTelephonyEvent::OnBnClickedCheckDtmfFeedback()
+{
+ CButton* button(NULL);
+
+ // Retrieve feedback state
+ button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ // Retrieve direct-feedback setting
+ button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
+ check = button->GetCheck();
+ const bool directFeedback = (check == BST_CHECKED);
+
+ // GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK)->EnableWindow(enable ? TRUE : FALSE);
+
+ TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
+ _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
+}
+
+void CTelephonyEvent::OnBnClickedCheckDirectFeedback()
+{
+ CButton* button(NULL);
+
+ // Retrieve feedback state
+ button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ // Retrieve new direct-feedback setting
+ button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
+ check = button->GetCheck();
+ const bool directFeedback = (check == BST_CHECKED);
+
+ TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
+ _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
+}
+
+void CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt()
+{
+ BOOL ret;
+ int pt = GetDlgItemInt(IDC_EDIT_EVENT_RX_PT, &ret);
+ if (ret == FALSE)
+ return;
+ CodecInst codec;
+ strcpy_s(codec.plname, 32, "telephone-event");
+ codec.pltype = pt; codec.channels = 1; codec.plfreq = 8000;
+ TEST2(_veCodecPtr->SetRecPayloadType(_channel, codec) == 0,
+ _T("SetSendTelephoneEventPayloadType(channel=%d, codec.pltype=%u)"), _channel, codec.pltype);
+}
+
+void CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt()
+{
+ BOOL ret;
+ int pt = GetDlgItemInt(IDC_EDIT_EVENT_TX_PT, &ret);
+ if (ret == FALSE)
+ return;
+ TEST2(_veDTMFPtr->SetSendTelephoneEventPayloadType(_channel, pt) == 0,
+ _T("SetSendTelephoneEventPayloadType(channel=%d, type=%u)"), _channel, pt);
+}
+
+void CTelephonyEvent::OnBnClickedCheckDetectInband()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
+ int check = button->GetCheck();
+ _inbandEventDetection = (check == BST_CHECKED);
+
+ bool enabled(false);
+ TelephoneEventDetectionMethods detectionMethod;
+ _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
+ if (enabled)
+ {
+ // deregister
+ _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
+ delete _telephoneEventObserverPtr;
+ _telephoneEventObserverPtr = NULL;
+ SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
+ SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
+ }
+ OnBnClickedCheckEventDetection();
+}
+
+void CTelephonyEvent::OnBnClickedCheckDetectOutOfBand()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
+ int check = button->GetCheck();
+ _outOfBandEventDetection = (check == BST_CHECKED);
+
+ bool enabled(false);
+ TelephoneEventDetectionMethods detectionMethod;
+ _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
+ if (enabled)
+ {
+ // deregister
+ _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
+ delete _telephoneEventObserverPtr;
+ _telephoneEventObserverPtr = NULL;
+ SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
+ SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
+ }
+ OnBnClickedCheckEventDetection();
+}
+
+void CTelephonyEvent::OnBnClickedCheckEventDetection()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ if (enable)
+ {
+ TelephoneEventDetectionMethods method(kInBand);
+ if (_inbandEventDetection && !_outOfBandEventDetection)
+ method = kInBand;
+ else if (!_inbandEventDetection && _outOfBandEventDetection)
+ method = kOutOfBand;
+ else if (_inbandEventDetection && _outOfBandEventDetection)
+ method = kInAndOutOfBand;
+
+ CWnd* wndOut = GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND);
+ CWnd* wndIn = GetDlgItem(IDC_EDIT_ON_EVENT_INBAND);
+ _telephoneEventObserverPtr = new TelephoneEventObserver(wndOut, wndIn);
+
+ TEST2(_veDTMFPtr->RegisterTelephoneEventDetection(_channel, method, *_telephoneEventObserverPtr) == 0,
+ _T("RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d)"), _channel, method);
+ }
+ else
+ {
+ TEST2(_veDTMFPtr->DeRegisterTelephoneEventDetection(_channel) == 0,
+ _T("DeRegisterTelephoneEventDetection(channel=%d)"), _channel);
+ delete _telephoneEventObserverPtr;
+ _telephoneEventObserverPtr = NULL;
+ SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
+ SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
+ }
+}
+
+// ============================================================================
+// CWinTestDlg dialog
+// ============================================================================
+
+CWinTestDlg::CWinTestDlg(CWnd* pParent /*=NULL*/)
+ : CDialog(CWinTestDlg::IDD, pParent),
+ _failCount(0),
+ _vePtr(NULL),
+ _veBasePtr(NULL),
+ _veCodecPtr(NULL),
+ _veNetworkPtr(NULL),
+ _veFilePtr(NULL),
+ _veHardwarePtr(NULL),
+ _veExternalMediaPtr(NULL),
+ _veApmPtr(NULL),
+ _veEncryptionPtr(NULL),
+ _veRtpRtcpPtr(NULL),
+ _transportPtr(NULL),
+ _encryptionPtr(NULL),
+ _externalMediaPtr(NULL),
+ _externalTransport(false),
+ _externalTransportBuild(false),
+ _checkPlayFileIn(0),
+ _checkPlayFileIn1(0),
+ _checkPlayFileIn2(0),
+ _checkPlayFileOut1(0),
+ _checkPlayFileOut2(0),
+ _checkAGC(0),
+ _checkAGC1(0),
+ _checkNS(0),
+ _checkNS1(0),
+ _checkEC(0),
+ _checkVAD1(0),
+ _checkVAD2(0),
+ _checkSrtpTx1(0),
+ _checkSrtpTx2(0),
+ _checkSrtpRx1(0),
+ _checkSrtpRx2(0),
+ _checkConference1(0),
+ _checkConference2(0),
+ _checkOnHold1(0),
+ _checkOnHold2(0),
+ _strComboIp1(_T("")),
+ _strComboIp2(_T("")),
+ _delayEstimate1(false),
+ _delayEstimate2(false),
+ _rxVad(false),
+ _nErrorCallbacks(0),
+ _timerTicks(0)
+{
+ m_hIcon = AfxGetApp()->LoadIcon(IDR_MAINFRAME);
+
+ _vePtr = VoiceEngine::Create();
+
+ VoiceEngine::SetTraceFilter(kTraceNone);
+ // VoiceEngine::SetTraceFilter(kTraceAll);
+ // VoiceEngine::SetTraceFilter(kTraceStream | kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceDebug | kTraceInfo);
+ // VoiceEngine::SetTraceFilter(kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceInfo);
+
+ VoiceEngine::SetTraceFile("ve_win_test.txt");
+ VoiceEngine::SetTraceCallback(NULL);
+
+ if (_vePtr)
+ {
+ _veExternalMediaPtr = VoEExternalMedia::GetInterface(_vePtr);
+ _veVolumeControlPtr = VoEVolumeControl::GetInterface(_vePtr);
+ _veEncryptionPtr = VoEEncryption::GetInterface(_vePtr);
+ _veVideoSyncPtr = VoEVideoSync::GetInterface(_vePtr);
+ _veNetworkPtr = VoENetwork::GetInterface(_vePtr);
+ _veFilePtr = VoEFile::GetInterface(_vePtr);
+ _veApmPtr = VoEAudioProcessing::GetInterface(_vePtr);
+
+ _veBasePtr = VoEBase::GetInterface(_vePtr);
+ _veCodecPtr = VoECodec::GetInterface(_vePtr);
+ _veHardwarePtr = VoEHardware::GetInterface(_vePtr);
+ _veRtpRtcpPtr = VoERTP_RTCP::GetInterface(_vePtr);
+ _transportPtr = new MyTransport(_veNetworkPtr);
+ _encryptionPtr = new MyEncryption();
+ _externalMediaPtr = new MediaProcessImpl();
+ _connectionObserverPtr = new ConnectionObserver();
+ _rxVadObserverPtr = new RxCallback();
+ }
+
+ _veBasePtr->RegisterVoiceEngineObserver(*this);
+}
+
+CWinTestDlg::~CWinTestDlg()
+{
+ if (_connectionObserverPtr) delete _connectionObserverPtr;
+ if (_externalMediaPtr) delete _externalMediaPtr;
+ if (_transportPtr) delete _transportPtr;
+ if (_encryptionPtr) delete _encryptionPtr;
+ if (_rxVadObserverPtr) delete _rxVadObserverPtr;
+
+ if (_veExternalMediaPtr) _veExternalMediaPtr->Release();
+ if (_veEncryptionPtr) _veEncryptionPtr->Release();
+ if (_veVideoSyncPtr) _veVideoSyncPtr->Release();
+ if (_veVolumeControlPtr) _veVolumeControlPtr->Release();
+
+ if (_veBasePtr) _veBasePtr->Terminate();
+ if (_veBasePtr) _veBasePtr->Release();
+
+ if (_veCodecPtr) _veCodecPtr->Release();
+ if (_veNetworkPtr) _veNetworkPtr->Release();
+ if (_veFilePtr) _veFilePtr->Release();
+ if (_veHardwarePtr) _veHardwarePtr->Release();
+ if (_veApmPtr) _veApmPtr->Release();
+ if (_veRtpRtcpPtr) _veRtpRtcpPtr->Release();
+ if (_vePtr)
+ {
+ bool ret = VoiceEngine::Delete(_vePtr);
+ ASSERT(ret == true);
+ }
+ VoiceEngine::SetTraceFilter(kTraceNone);
+}
+
+void CWinTestDlg::DoDataExchange(CDataExchange* pDX)
+{
+ CDialog::DoDataExchange(pDX);
+ DDX_CBString(pDX, IDC_COMBO_IP_1, _strComboIp1);
+ DDX_CBString(pDX, IDC_COMBO_IP_2, _strComboIp2);
+}
+
+BEGIN_MESSAGE_MAP(CWinTestDlg, CDialog)
+ ON_WM_SYSCOMMAND()
+ ON_WM_PAINT()
+ ON_WM_QUERYDRAGICON()
+ ON_WM_TIMER()
+ //}}AFX_MSG_MAP
+ ON_BN_CLICKED(IDC_BUTTON_CREATE_1, &CWinTestDlg::OnBnClickedButtonCreate1)
+ ON_BN_CLICKED(IDC_BUTTON_DELETE_1, &CWinTestDlg::OnBnClickedButtonDelete1)
+ ON_BN_CLICKED(IDC_BUTTON_CREATE_2, &CWinTestDlg::OnBnClickedButtonCreate2)
+ ON_BN_CLICKED(IDC_BUTTON_DELETE_2, &CWinTestDlg::OnBnClickedButtonDelete2)
+ ON_CBN_SELCHANGE(IDC_COMBO_CODEC_1, &CWinTestDlg::OnCbnSelchangeComboCodec1)
+ ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStartListen1)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStopListen1)
+ ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStartPlayout1)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStopPlayout1)
+ ON_BN_CLICKED(IDC_BUTTON_START_SEND_1, &CWinTestDlg::OnBnClickedButtonStartSend1)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_1, &CWinTestDlg::OnBnClickedButtonStopSend1)
+ ON_CBN_SELCHANGE(IDC_COMBO_IP_2, &CWinTestDlg::OnCbnSelchangeComboIp2)
+ ON_CBN_SELCHANGE(IDC_COMBO_IP_1, &CWinTestDlg::OnCbnSelchangeComboIp1)
+ ON_CBN_SELCHANGE(IDC_COMBO_CODEC_2, &CWinTestDlg::OnCbnSelchangeComboCodec2)
+ ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStartListen2)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStopListen2)
+ ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStartPlayout2)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStopPlayout2)
+ ON_BN_CLICKED(IDC_BUTTON_START_SEND_2, &CWinTestDlg::OnBnClickedButtonStartSend2)
+ ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_2, &CWinTestDlg::OnBnClickedButtonStopSend2)
+ ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_1, &CWinTestDlg::OnBnClickedCheckExtTrans1)
+ ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_1, &CWinTestDlg::OnBnClickedCheckPlayFileIn1)
+ ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_1, &CWinTestDlg::OnBnClickedCheckPlayFileOut1)
+ ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_2, &CWinTestDlg::OnBnClickedCheckExtTrans2)
+ ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_2, &CWinTestDlg::OnBnClickedCheckPlayFileIn2)
+ ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_2, &CWinTestDlg::OnBnClickedCheckPlayFileOut2)
+ ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN, &CWinTestDlg::OnBnClickedCheckPlayFileIn)
+ ON_CBN_SELCHANGE(IDC_COMBO_REC_DEVICE, &CWinTestDlg::OnCbnSelchangeComboRecDevice)
+ ON_CBN_SELCHANGE(IDC_COMBO_PLAY_DEVICE, &CWinTestDlg::OnCbnSelchangeComboPlayDevice)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_1, &CWinTestDlg::OnBnClickedCheckExtMediaIn1)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_1, &CWinTestDlg::OnBnClickedCheckExtMediaOut1)
+ ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_INPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderInputVolume)
+ ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_OUTPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume)
+ ON_BN_CLICKED(IDC_CHECK_AGC, &CWinTestDlg::OnBnClickedCheckAgc)
+ ON_BN_CLICKED(IDC_CHECK_NS, &CWinTestDlg::OnBnClickedCheckNs)
+ ON_BN_CLICKED(IDC_CHECK_EC, &CWinTestDlg::OnBnClickedCheckEc)
+ ON_BN_CLICKED(IDC_CHECK_VAD_1, &CWinTestDlg::OnBnClickedCheckVad1)
+ ON_BN_CLICKED(IDC_CHECK_VAD_3, &CWinTestDlg::OnBnClickedCheckVad2)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_2, &CWinTestDlg::OnBnClickedCheckExtMediaIn2)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_2, &CWinTestDlg::OnBnClickedCheckExtMediaOut2)
+ ON_BN_CLICKED(IDC_CHECK_MUTE_IN, &CWinTestDlg::OnBnClickedCheckMuteIn)
+ ON_BN_CLICKED(IDC_CHECK_MUTE_IN_1, &CWinTestDlg::OnBnClickedCheckMuteIn1)
+ ON_BN_CLICKED(IDC_CHECK_MUTE_IN_2, &CWinTestDlg::OnBnClickedCheckMuteIn2)
+ ON_BN_CLICKED(IDC_CHECK_SRTP_TX_1, &CWinTestDlg::OnBnClickedCheckSrtpTx1)
+ ON_BN_CLICKED(IDC_CHECK_SRTP_RX_1, &CWinTestDlg::OnBnClickedCheckSrtpRx1)
+ ON_BN_CLICKED(IDC_CHECK_SRTP_TX_2, &CWinTestDlg::OnBnClickedCheckSrtpTx2)
+ ON_BN_CLICKED(IDC_CHECK_SRTP_RX_2, &CWinTestDlg::OnBnClickedCheckSrtpRx2)
+ ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_1, &CWinTestDlg::OnBnClickedCheckExtEncryption1)
+ ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_2, &CWinTestDlg::OnBnClickedCheckExtEncryption2)
+ ON_BN_CLICKED(IDC_BUTTON_DTMF_1, &CWinTestDlg::OnBnClickedButtonDtmf1)
+ ON_BN_CLICKED(IDC_CHECK_REC_MIC, &CWinTestDlg::OnBnClickedCheckRecMic)
+ ON_BN_CLICKED(IDC_BUTTON_DTMF_2, &CWinTestDlg::OnBnClickedButtonDtmf2)
+ ON_BN_CLICKED(IDC_BUTTON_TEST_1, &CWinTestDlg::OnBnClickedButtonTest1)
+ ON_BN_CLICKED(IDC_CHECK_CONFERENCE_1, &CWinTestDlg::OnBnClickedCheckConference1)
+ ON_BN_CLICKED(IDC_CHECK_CONFERENCE_2, &CWinTestDlg::OnBnClickedCheckConference2)
+ ON_BN_CLICKED(IDC_CHECK_ON_HOLD_1, &CWinTestDlg::OnBnClickedCheckOnHold1)
+ ON_BN_CLICKED(IDC_CHECK_ON_HOLD_2, &CWinTestDlg::OnBnClickedCheckOnHold2)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN, &CWinTestDlg::OnBnClickedCheckExtMediaIn)
+ ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT, &CWinTestDlg::OnBnClickedCheckExtMediaOut)
+ ON_LBN_SELCHANGE(IDC_LIST_CODEC_1, &CWinTestDlg::OnLbnSelchangeListCodec1)
+ ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_LEFT, &CWinTestDlg::OnNMReleasedcaptureSliderPanLeft)
+ ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_RIGHT, &CWinTestDlg::OnNMReleasedcaptureSliderPanRight)
+ ON_BN_CLICKED(IDC_BUTTON_VERSION, &CWinTestDlg::OnBnClickedButtonVersion)
+ ON_BN_CLICKED(IDC_CHECK_DELAY_ESTIMATE_1, &CWinTestDlg::OnBnClickedCheckDelayEstimate1)
+ ON_BN_CLICKED(IDC_CHECK_RXVAD, &CWinTestDlg::OnBnClickedCheckRxvad)
+ ON_BN_CLICKED(IDC_CHECK_AGC_1, &CWinTestDlg::OnBnClickedCheckAgc1)
+ ON_BN_CLICKED(IDC_CHECK_NS_1, &CWinTestDlg::OnBnClickedCheckNs1)
+ ON_BN_CLICKED(IDC_CHECK_REC_CALL, &CWinTestDlg::OnBnClickedCheckRecCall)
+ ON_BN_CLICKED(IDC_CHECK_TYPING_DETECTION, &CWinTestDlg::OnBnClickedCheckTypingDetection)
+ ON_BN_CLICKED(IDC_CHECK_FEC, &CWinTestDlg::OnBnClickedCheckFEC)
+ ON_BN_CLICKED(IDC_BUTTON_CLEAR_ERROR_CALLBACK, &CWinTestDlg::OnBnClickedButtonClearErrorCallback)
+END_MESSAGE_MAP()
+
+BOOL CWinTestDlg::UpdateTest(bool failed, const CString& strMsg)
+{
+ if (failed)
+ {
+ SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
+ _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError());
+ SetDlgItemText(IDC_EDIT_RESULT, _strErr);
+ _failCount++;
+ SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount);
+ SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError());
+ }
+ else
+ {
+ SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
+ SetDlgItemText(IDC_EDIT_RESULT, _T("OK"));
+ }
+ return TRUE;
+}
+
+
+// CWinTestDlg message handlers
+
+BOOL CWinTestDlg::OnInitDialog()
+{
+ CDialog::OnInitDialog();
+
+ // Add "About..." menu item to system menu.
+
+ // IDM_ABOUTBOX must be in the system command range.
+ ASSERT((IDM_ABOUTBOX & 0xFFF0) == IDM_ABOUTBOX);
+ ASSERT(IDM_ABOUTBOX < 0xF000);
+
+ CMenu* pSysMenu = GetSystemMenu(FALSE);
+ if (pSysMenu != NULL)
+ {
+ CString strAboutMenu;
+ strAboutMenu.LoadString(IDS_ABOUTBOX);
+ if (!strAboutMenu.IsEmpty())
+ {
+ pSysMenu->AppendMenu(MF_SEPARATOR);
+ pSysMenu->AppendMenu(MF_STRING, IDM_ABOUTBOX, strAboutMenu);
+ }
+ }
+
+ // Set the icon for this dialog. The framework does this automatically
+ // when the application's main window is not a dialog
+ SetIcon(m_hIcon, TRUE); // Set big icon
+ SetIcon(m_hIcon, FALSE); // Set small icon
+
+ // char version[1024];
+ // _veBasePtr->GetVersion(version);
+ // AfxMessageBox(version, MB_OK);
+
+ if (_veBasePtr->Init() != 0)
+ {
+ AfxMessageBox(_T("Init() failed "), MB_OKCANCEL);
+ }
+
+ int ch = _veBasePtr->CreateChannel();
+ if (_veBasePtr->SetSendDestination(ch, 1234, "127.0.0.1") == -1)
+ {
+ if (_veBasePtr->LastError() == VE_EXTERNAL_TRANSPORT_ENABLED)
+ {
+ _strMsg.Format(_T("*** External transport build ***"));
+ SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg);
+ _externalTransportBuild = true;
+ }
+ }
+ _veBasePtr->DeleteChannel(ch);
+
+ // --- Add (preferred) local IPv4 address in title
+
+ if (_veNetworkPtr)
+ {
+ char localIP[64];
+ _veNetworkPtr->GetLocalIP(localIP);
+ CString str;
+ GetWindowText(str);
+ str.AppendFormat(_T(" [Local IPv4 address: %s]"), CharToTchar(localIP, 64));
+ SetWindowText(str);
+ }
+
+ // --- Volume sliders
+
+ if (_veVolumeControlPtr)
+ {
+ unsigned int volume(0);
+ CSliderCtrl* slider(NULL);
+
+ slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
+ slider->SetRangeMin(0);
+ slider->SetRangeMax(255);
+ _veVolumeControlPtr->GetMicVolume(volume);
+ slider->SetPos(volume);
+
+ slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
+ slider->SetRangeMin(0);
+ slider->SetRangeMax(255);
+ _veVolumeControlPtr->GetSpeakerVolume(volume);
+ slider->SetPos(volume);
+ }
+
+ // --- Panning sliders
+
+ if (_veVolumeControlPtr)
+ {
+ float lVol(0.0);
+ float rVol(0.0);
+ int leftVol, rightVol;
+ unsigned int volumePan(0);
+ CSliderCtrl* slider(NULL);
+
+ _veVolumeControlPtr->GetOutputVolumePan(-1, lVol, rVol);
+
+ leftVol = (int)(lVol*10.0f); // [0,10]
+ rightVol = (int)(rVol*10.0f); // [0,10]
+
+ slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
+ slider->SetRange(0,10);
+ slider->SetPos(10-leftVol); // pos 0 <=> max pan 1.0 (top of slider)
+
+ slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
+ slider->SetRange(0,10);
+ slider->SetPos(10-rightVol);
+ }
+
+ // --- APM settings
+
+ bool enable(false);
+ CButton* button(NULL);
+
+ AgcModes agcMode(kAgcDefault);
+ if (_veApmPtr->GetAgcStatus(enable, agcMode) == 0)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_AGC);
+ enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
+ }
+ else
+ {
+ // AGC is not supported
+ GetDlgItem(IDC_CHECK_AGC)->EnableWindow(FALSE);
+ }
+
+ NsModes nsMode(kNsDefault);
+ if (_veApmPtr->GetNsStatus(enable, nsMode) == 0)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_NS);
+ enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
+ }
+ else
+ {
+ // NS is not supported
+ GetDlgItem(IDC_CHECK_NS)->EnableWindow(FALSE);
+ }
+
+ EcModes ecMode(kEcDefault);
+ if (_veApmPtr->GetEcStatus(enable, ecMode) == 0)
+ {
+ button = (CButton*)GetDlgItem(IDC_CHECK_EC);
+ enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
+ }
+ else
+ {
+ // EC is not supported
+ GetDlgItem(IDC_CHECK_EC)->EnableWindow(FALSE);
+ }
+
+ // --- First channel section
+
+ GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_FEC)->EnableWindow(FALSE);
+
+ CComboBox* comboIP(NULL);
+ comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
+ comboIP->AddString(_T("127.0.0.1"));
+ comboIP->SetCurSel(0);
+
+ SetDlgItemInt(IDC_EDIT_TX_PORT_1, 1111);
+ SetDlgItemInt(IDC_EDIT_RX_PORT_1, 1111);
+
+ // --- Add supported codecs to the codec combo box
+
+ CComboBox* comboCodec(NULL);
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
+ comboCodec->ResetContent();
+
+ int numCodecs = _veCodecPtr->NumOfCodecs();
+ for (int idx = 0; idx < numCodecs; idx++)
+ {
+ CodecInst codec;
+ _veCodecPtr->GetCodec(idx, codec);
+ if ((_stricmp(codec.plname, "CNNB") != 0) &&
+ (_stricmp(codec.plname, "CNWB") != 0))
+ {
+ CString strCodec;
+ if (_stricmp(codec.plname, "G7221") == 0)
+ strCodec.Format(_T("%s (%d/%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000, codec.rate/1000);
+ else
+ strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
+ comboCodec->AddString(strCodec);
+ }
+ if (idx == 0)
+ {
+ SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
+ }
+ }
+ comboCodec->SetCurSel(0);
+
+ CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
+ list->AddString(_T("pltype"));
+ list->AddString(_T("plfreq"));
+ list->AddString(_T("pacsize"));
+ list->AddString(_T("channels"));
+ list->AddString(_T("rate"));
+ list->SetCurSel(0);
+
+ // --- Add available audio devices to the combo boxes
+
+ CComboBox* comboRecDevice(NULL);
+ CComboBox* comboPlayDevice(NULL);
+ comboRecDevice = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
+ comboPlayDevice = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
+ comboRecDevice->ResetContent();
+ comboPlayDevice->ResetContent();
+
+ if (_veHardwarePtr)
+ {
+ int numPlayout(0);
+ int numRecording(0);
+ char nameStr[128];
+ char guidStr[128];
+ CString strDevice;
+ AudioLayers audioLayer;
+
+ _veHardwarePtr->GetAudioDeviceLayer(audioLayer);
+ if (kAudioWindowsWave == audioLayer)
+ {
+ strDevice.FormatMessage(_T("Audio Layer: Windows Wave API"));
+ }
+ else if (kAudioWindowsCore == audioLayer)
+ {
+ strDevice.FormatMessage(_T("Audio Layer: Windows Core API"));
+ }
+ else
+ {
+ strDevice.FormatMessage(_T("Audio Layer: ** UNKNOWN **"));
+ }
+ SetDlgItemText(IDC_EDIT_AUDIO_LAYER, (LPCTSTR)strDevice);
+
+ _veHardwarePtr->GetNumOfRecordingDevices(numRecording);
+
+ for (int idx = 0; idx < numRecording; idx++)
+ {
+ _veHardwarePtr->GetRecordingDeviceName(idx, nameStr, guidStr);
+ strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
+ comboRecDevice->AddString(strDevice);
+ }
+ // Select default (communication) device in the combo box
+ _veHardwarePtr->GetRecordingDeviceName(-1, nameStr, guidStr);
+ CString tmp = CString(nameStr);
+ int nIndex = comboRecDevice->SelectString(-1, tmp);
+ ASSERT(nIndex != CB_ERR);
+
+ _veHardwarePtr->GetNumOfPlayoutDevices(numPlayout);
+
+ for (int idx = 0; idx < numPlayout; idx++)
+ {
+ _veHardwarePtr->GetPlayoutDeviceName(idx, nameStr, guidStr);
+ strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
+ comboPlayDevice->AddString(strDevice);
+ }
+ // Select default (communication) device in the combo box
+ _veHardwarePtr->GetPlayoutDeviceName(-1, nameStr, guidStr);
+ nIndex = comboPlayDevice->SelectString(-1, CString(nameStr));
+ ASSERT(nIndex != CB_ERR);
+ }
+
+ // --- Second channel section
+
+ GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
+
+ comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
+ comboIP->AddString(_T("127.0.0.1"));
+ comboIP->SetCurSel(0);
+
+ SetDlgItemInt(IDC_EDIT_TX_PORT_2, 2222);
+ SetDlgItemInt(IDC_EDIT_RX_PORT_2, 2222);
+
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
+ comboCodec->ResetContent();
+
+ if (_veCodecPtr)
+ {
+ numCodecs = _veCodecPtr->NumOfCodecs();
+ for (int idx = 0; idx < numCodecs; idx++)
+ {
+ CodecInst codec;
+ _veCodecPtr->GetCodec(idx, codec);
+ CString strCodec;
+ strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
+ comboCodec->AddString(strCodec);
+ }
+ comboCodec->SetCurSel(0);
+ }
+
+ // --- Start windows timer
+
+ SetTimer(0, 1000, NULL);
+
+ return TRUE; // return TRUE unless you set the focus to a control
+}
+
+void CWinTestDlg::OnSysCommand(UINT nID, LPARAM lParam)
+{
+ if ((nID & 0xFFF0) == IDM_ABOUTBOX)
+ {
+ CAboutDlg dlgAbout;
+ dlgAbout.DoModal();
+ }
+ else if (nID == SC_CLOSE)
+ {
+ BOOL ret;
+ int channel(0);
+ channel = GetDlgItemInt(IDC_EDIT_1, &ret);
+ if (ret == TRUE)
+ {
+ _veBasePtr->DeleteChannel(channel);
+ }
+ channel = GetDlgItemInt(IDC_EDIT_2, &ret);
+ if (ret == TRUE)
+ {
+ _veBasePtr->DeleteChannel(channel);
+ }
+
+ CDialog::OnSysCommand(nID, lParam);
+ }
+ else
+ {
+ CDialog::OnSysCommand(nID, lParam);
+ }
+
+}
+
+// If you add a minimize button to your dialog, you will need the code below
+// to draw the icon. For MFC applications using the document/view model,
+// this is automatically done for you by the framework.
+
+void CWinTestDlg::OnPaint()
+{
+ if (IsIconic())
+ {
+ CPaintDC dc(this); // device context for painting
+
+ SendMessage(WM_ICONERASEBKGND, reinterpret_cast<WPARAM>(dc.GetSafeHdc()), 0);
+
+ // Center icon in client rectangle
+ int cxIcon = GetSystemMetrics(SM_CXICON);
+ int cyIcon = GetSystemMetrics(SM_CYICON);
+ CRect rect;
+ GetClientRect(&rect);
+ int x = (rect.Width() - cxIcon + 1) / 2;
+ int y = (rect.Height() - cyIcon + 1) / 2;
+
+ // Draw the icon
+ dc.DrawIcon(x, y, m_hIcon);
+ }
+ else
+ {
+ CDialog::OnPaint();
+ }
+}
+
+// The system calls this function to obtain the cursor to display while the user drags
+// the minimized window.
+HCURSOR CWinTestDlg::OnQueryDragIcon()
+{
+ return static_cast<HCURSOR>(m_hIcon);
+}
+
+
+void CWinTestDlg::OnBnClickedButtonCreate1()
+{
+ int channel(0);
+ TEST((channel = _veBasePtr->CreateChannel()) >= 0, _T("CreateChannel(channel=%d)"), channel);
+ if (channel >= 0)
+ {
+ _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
+
+ SetDlgItemInt(IDC_EDIT_1, channel);
+ GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_FEC)->EnableWindow(TRUE);
+
+ bool enabled(false);
+ bool includeCSRCs(false);
+
+ // Always set send codec to default codec <=> index 0.
+ CodecInst codec;
+ _veCodecPtr->GetCodec(0, codec);
+ _veCodecPtr->SetSendCodec(channel, codec);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonCreate2()
+{
+ int channel(0);
+ TEST((channel = _veBasePtr->CreateChannel()) >=0 , _T("CreateChannel(%d)"), channel);
+ if (channel >= 0)
+ {
+ _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
+
+ SetDlgItemInt(IDC_EDIT_2, channel);
+ GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(TRUE);
+
+ bool enabled(false);
+ bool includeCSRCs(false);
+
+
+ // Always set send codec to default codec <=> index 0.
+ CodecInst codec;
+ _veCodecPtr->GetCodec(0, codec);
+ _veCodecPtr->SetSendCodec(channel, codec);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonDelete1()
+{
+ BOOL ret;
+ int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
+ if (ret == TRUE)
+ {
+ _delayEstimate1 = false;
+ _rxVad = false;
+ _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
+ TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(channel=%d)"), channel);
+ SetDlgItemText(IDC_EDIT_1, _T(""));
+ GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_FEC)->EnableWindow(FALSE);
+ SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
+ GetDlgItem(IDC_EDIT_RXVAD)->EnableWindow(FALSE);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_FEC);
+ button->SetCheck(BST_UNCHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonDelete2()
+{
+ BOOL ret;
+ int channel = GetDlgItemInt(IDC_EDIT_2, &ret);
+ if (ret == TRUE)
+ {
+ _delayEstimate2 = false;
+ _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
+ TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(%d)"), channel);
+ SetDlgItemText(IDC_EDIT_2, _T(""));
+ GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_VAD_3);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_2);
+ button->SetCheck(BST_UNCHECKED);
+ button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
+ button->SetCheck(BST_UNCHECKED);
+ }
+}
+
+void CWinTestDlg::OnCbnSelchangeComboIp1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CString str;
+ int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
+ CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
+ int n = comboIP->GetLBTextLen(0);
+ comboIP->GetLBText(0, str.GetBuffer(n));
+ TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
+ _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
+ str.ReleaseBuffer();
+}
+
+void CWinTestDlg::OnCbnSelchangeComboIp2()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CString str;
+ int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
+ CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
+ int n = comboIP->GetLBTextLen(0);
+ comboIP->GetLBText(0, str.GetBuffer(n));
+ TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
+ _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
+ str.ReleaseBuffer();
+}
+
+void CWinTestDlg::OnCbnSelchangeComboCodec1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+
+ CodecInst codec;
+ CComboBox* comboCodec(NULL);
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
+ int index = comboCodec->GetCurSel();
+ _veCodecPtr->GetCodec(index, codec);
+ if (strncmp(codec.plname, "ISAC", 4) == 0)
+ {
+ // Set iSAC to adaptive mode by default.
+ codec.rate = -1;
+ }
+ TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
+ _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
+ channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
+
+ CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
+ list->SetCurSel(0);
+ SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
+}
+
+void CWinTestDlg::OnLbnSelchangeListCodec1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+
+ CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
+ int listIdx = list->GetCurSel();
+ if (listIdx < 0)
+ return;
+ CString str;
+ list->GetText(listIdx, str);
+
+ CodecInst codec;
+ _veCodecPtr->GetSendCodec(channel, codec);
+
+ int value = GetDlgItemInt(IDC_EDIT_CODEC_1);
+ if (str == _T("pltype"))
+ {
+ codec.pltype = value;
+ }
+ else if (str == _T("plfreq"))
+ {
+ codec.plfreq = value;
+ }
+ else if (str == _T("pacsize"))
+ {
+ codec.pacsize = value;
+ }
+ else if (str == _T("channels"))
+ {
+ codec.channels = value;
+ }
+ else if (str == _T("rate"))
+ {
+ codec.rate = value;
+ }
+ TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
+ _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
+ channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
+}
+
+void CWinTestDlg::OnCbnSelchangeComboCodec2()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+
+ CodecInst codec;
+ CComboBox* comboCodec(NULL);
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
+ int index = comboCodec->GetCurSel();
+ _veCodecPtr->GetCodec(index, codec);
+ TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
+ _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
+ channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
+}
+
+void CWinTestDlg::OnBnClickedButtonStartListen1()
+{
+ int ret1(0);
+ int ret2(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ int port = GetDlgItemInt(IDC_EDIT_RX_PORT_1);
+ TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
+ TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
+ if (ret1 == 0 && ret2 == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStartListen2()
+{
+ int ret1(0);
+ int ret2(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ int port = GetDlgItemInt(IDC_EDIT_RX_PORT_2);
+ TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
+ TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
+ if (ret1 == 0 && ret2 == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopListen1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopListen2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStartPlayout1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStartPlayout2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopPlayout1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopPlayout2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"));
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStartSend1()
+{
+ UpdateData(TRUE); // update IP address
+
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ if (!_externalTransport)
+ {
+ CString str;
+ int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
+ TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp1.GetBuffer(7), -1)) == 0,
+ _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp1.GetBuffer(7));
+ str.ReleaseBuffer();
+ }
+
+ //_veVideoSyncPtr->SetInitTimestamp(0,0);
+ // OnCbnSelchangeComboCodec1();
+
+ TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStartSend2()
+{
+ UpdateData(TRUE); // update IP address
+
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ if (!_externalTransport)
+ {
+ CString str;
+ int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
+ TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp2.GetBuffer(7), -1)) == 0,
+ _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp2.GetBuffer(7));
+ str.ReleaseBuffer();
+ }
+
+ // OnCbnSelchangeComboCodec2();
+
+ TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(TRUE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopSend1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonStopSend2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
+ if (ret == 0)
+ {
+ GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
+ GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtTrans1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
+ _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
+ }
+ else
+ {
+ TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
+ _T("DeRegisterExternalTransport(channel=%d)"), channel);
+ }
+ if (ret == 0)
+ {
+ _externalTransport = enable;
+ }
+ else
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtTrans2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
+ _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
+ }
+ else
+ {
+ TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
+ _T("DeRegisterExternalTransport(channel=%d)"), channel);
+ }
+ if (ret == 0)
+ {
+ _externalTransport = enable;
+ }
+ else
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckPlayFileIn1()
+{
+ // File path is relative to the location of 'voice_engine.gyp'.
+ const char micFile[] = "../test/data/voice_engine/audio_short16.pcm";
+
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ bool mix;
+ const bool loop(true);
+ const FileFormats format = kFileFormatPcm16kHzFile;
+ const float scale(1.0);
+
+ (_checkPlayFileIn1 %2 == 0) ? mix = true : mix = false;
+ TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0),
+ _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
+ channel, CharToTchar(micFile, -1), loop, mix, format, scale);
+ _checkPlayFileIn1++;
+ }
+ else
+ {
+ TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
+ _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckPlayFileIn2()
+{
+ // File path is relative to the location of 'voice_engine.gyp'.
+ const char micFile[] = "../test/data/voice_engine/audio_long16.pcm";
+
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ bool mix;
+ const bool loop(true);
+ const FileFormats format = kFileFormatPcm16kHzFile;
+ const float scale(1.0);
+
+ (_checkPlayFileIn2 %2 == 0) ? mix = true : mix = false;
+ TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0),
+ _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
+ channel, CharToTchar(micFile, -1), loop, mix, format, scale);
+ _checkPlayFileIn2++;
+ }
+ else
+ {
+ TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
+ _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckPlayFileOut1()
+{
+ const FileFormats formats[8] = {{kFileFormatPcm16kHzFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile},
+ {kFileFormatWavFile}};
+ // File path is relative to the location of 'voice_engine.gyp'.
+ const char spkrFiles[8][64] = {{"../../test/data/voice_engine/audio_short16.pcm"},
+ {"../../test/data/voice_engine/audio_tiny8.wav"},
+ {"../../test/data/voice_engine/audio_tiny11.wav"},
+ {"../../test/data/voice_engine/audio_tiny16.wav"},
+ {"../../test/data/voice_engine/audio_tiny22.wav"},
+ {"../../test/data/voice_engine/audio_tiny32.wav"},
+ {"../../test/data/voice_engine/audio_tiny44.wav"},
+ {"../../test/data/voice_engine/audio_tiny48.wav"}};
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ const bool loop(true);
+ const float volumeScaling(1.0);
+ const int startPointMs(0);
+ const int stopPointMs(0);
+ const FileFormats format = formats[_checkPlayFileOut1 % 8];
+ const char* spkrFile = spkrFiles[_checkPlayFileOut1 % 8];
+
+ CString str;
+ if (_checkPlayFileOut1 % 8 == 0)
+ {
+ str = _T("kFileFormatPcm16kHzFile");
+ }
+ else
+ {
+ str = _T("kFileFormatWavFile");
+ }
+ // (_checkPlayFileOut1 %2 == 0) ? mix = true : mix = false;
+ TEST((ret = _veFilePtr->StartPlayingFileLocally(channel, spkrFile, loop, format, volumeScaling, startPointMs,stopPointMs) == 0),
+ _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, format=%s, scale=%2.1f, start=%d, stop=%d)"),
+ channel, CharToTchar(spkrFile, -1), loop, str, volumeScaling, startPointMs, stopPointMs);
+ _checkPlayFileOut1++;
+ }
+ else
+ {
+ TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
+ _T("StopPlayingFileLocally(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckPlayFileOut2()
+{
+ // File path is relative to the location of 'voice_engine.gyp'.
+ const char spkrFile[] = "../test/data/voice_engine/audio_long16.pcm";
+
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ const bool loop(true);
+ const FileFormats format = kFileFormatPcm16kHzFile;
+ const float volumeScaling(1.0);
+ const int startPointMs(0);
+ const int stopPointMs(0);
+
+ // (_checkPlayFileOut2 %2 == 0) ? mix = true : mix = false;
+ TEST((ret = _veFilePtr->StartPlayingFileLocally(channel, spkrFile, loop, format, volumeScaling, startPointMs,stopPointMs) == 0),
+ _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, format=%d, scale=%2.1f, start=%d, stop=%d)"),
+ channel, CharToTchar(spkrFile, -1), loop, format, volumeScaling, startPointMs, stopPointMs);
+ // _checkPlayFileIn2++;
+ }
+ else
+ {
+ TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
+ _T("StopPlayingFileLocally(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaIn1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
+ int check = buttonExtTrans->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaIn2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
+ int check = buttonExtTrans->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaOut1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
+ int check = buttonExtTrans->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaOut2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
+ int check = buttonExtTrans->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckVad1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ VadModes mode(kVadConventional);
+ if (_checkVAD1 % 4 == 0)
+ {
+ mode = kVadConventional;
+ str = _T("kVadConventional");
+ }
+ else if (_checkVAD1 % 4 == 1)
+ {
+ mode = kVadAggressiveLow;
+ str = _T("kVadAggressiveLow");
+ }
+ else if (_checkVAD1 % 4 == 2)
+ {
+ mode = kVadAggressiveMid;
+ str = _T("kVadAggressiveMid");
+ }
+ else if (_checkVAD1 % 4 == 3)
+ {
+ mode = kVadAggressiveHigh;
+ str = _T("kVadAggressiveHigh");
+ }
+ const bool disableDTX(false);
+ TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX) == 0),
+ _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
+ _checkVAD1++;
+ }
+ else
+ {
+ TEST((ret = _veCodecPtr->SetVADStatus(channel, false)) == 0, _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckVad2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ VadModes mode(kVadConventional);
+ if (_checkVAD2 % 4 == 0)
+ {
+ mode = kVadConventional;
+ str = _T("kVadConventional");
+ }
+ else if (_checkVAD2 % 4 == 1)
+ {
+ mode = kVadAggressiveLow;
+ str = _T("kVadAggressiveLow");
+ }
+ else if (_checkVAD2 % 4 == 2)
+ {
+ mode = kVadAggressiveMid;
+ str = _T("kVadAggressiveMid");
+ }
+ else if (_checkVAD2 % 4 == 3)
+ {
+ mode = kVadAggressiveHigh;
+ str = _T("kVadAggressiveHigh");
+ }
+ const bool disableDTX(false);
+ TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX)) == 0,
+ _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
+ _checkVAD2++;
+ }
+ else
+ {
+ TEST((ret = _veCodecPtr->SetVADStatus(channel, false) == 0), _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckMuteIn1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
+ int check = buttonMute->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
+ _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
+}
+
+void CWinTestDlg::OnBnClickedCheckMuteIn2()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
+ int check = buttonMute->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
+ _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
+}
+
+void CWinTestDlg::OnBnClickedCheckSrtpTx1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ bool useForRTCP = false;
+ if (enable)
+ {
+ (_checkSrtpTx1++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
+ TEST((ret = _veEncryptionPtr->EnableSRTPSend(channel,
+ kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
+ _T("EnableSRTPSend(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
+ channel, useForRTCP);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DisableSRTPSend(channel) == 0), _T("DisableSRTPSend(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckSrtpTx2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ bool useForRTCP = false;
+ if (enable)
+ {
+ (_checkSrtpTx2++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
+ TEST((ret = _veEncryptionPtr->EnableSRTPSend(channel,
+ kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
+ _T("EnableSRTPSend(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
+ channel, useForRTCP);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DisableSRTPSend(channel) == 0), _T("DisableSRTPSend(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckSrtpRx1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ bool useForRTCP(false);
+ if (enable)
+ {
+ (_checkSrtpRx1++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
+ TEST((ret = _veEncryptionPtr->EnableSRTPReceive(channel,
+ kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
+ _T("EnableSRTPReceive(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
+ channel, useForRTCP);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DisableSRTPReceive(channel) == 0), _T("DisableSRTPReceive(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckSrtpRx2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ bool useForRTCP(false);
+ if (enable)
+ {
+ (_checkSrtpRx2++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
+ TEST((ret = _veEncryptionPtr->EnableSRTPReceive(channel,
+ kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
+ _T("EnableSRTPReceive(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
+ channel, useForRTCP);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DisableSRTPReceive(channel)) == 0, _T("DisableSRTPReceive(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtEncryption1()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST((ret = _veEncryptionPtr->RegisterExternalEncryption(channel, *_encryptionPtr)) == 0,
+ _T("RegisterExternalEncryption(channel=%d, encryption=0x%x)"), channel, _encryptionPtr);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DeRegisterExternalEncryption(channel)) == 0,
+ _T("DeRegisterExternalEncryption(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtEncryption2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST((ret = _veEncryptionPtr->RegisterExternalEncryption(channel, *_encryptionPtr)) == 0,
+ _T("RegisterExternalEncryption(channel=%d, encryption=0x%x)"), channel, _encryptionPtr);
+ }
+ else
+ {
+ TEST((ret = _veEncryptionPtr->DeRegisterExternalEncryption(channel)) == 0,
+ _T("DeRegisterExternalEncryption(channel=%d)"), channel);
+ }
+ if (ret == -1)
+ {
+ // restore inital state since API call failed
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+ }
+}
+
+void CWinTestDlg::OnBnClickedButtonDtmf1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
+ dlgTelephoneEvent.DoModal();
+}
+
+void CWinTestDlg::OnBnClickedButtonDtmf2()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
+ dlgTelephoneEvent.DoModal();
+}
+
+void CWinTestDlg::OnBnClickedCheckConference1()
+{
+ // Not supported yet
+}
+
+void CWinTestDlg::OnBnClickedCheckConference2()
+{
+ // Not supported yet
+}
+
+void CWinTestDlg::OnBnClickedCheckOnHold1()
+{
+ SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
+
+ CString str;
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
+ int check = button->GetCheck();
+
+ if (shiftKeyIsPressed)
+ {
+ bool enabled(false);
+ OnHoldModes mode(kHoldSendAndPlay);
+ TEST(_veBasePtr->GetOnHoldStatus(channel, enabled, mode) == 0,
+ _T("GetOnHoldStatus(channel=%d, enabled=?, mode=?)"), channel);
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+
+ switch (mode)
+ {
+ case kHoldSendAndPlay:
+ str = _T("kHoldSendAndPlay");
+ break;
+ case kHoldSendOnly:
+ str = _T("kHoldSendOnly");
+ break;
+ case kHoldPlayOnly:
+ str = _T("kHoldPlayOnly");
+ break;
+ default:
+ break;
+ }
+ PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
+ return;
+ }
+
+ int ret(0);
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ OnHoldModes mode(kHoldSendAndPlay);
+ if (_checkOnHold1 % 3 == 0)
+ {
+ mode = kHoldSendAndPlay;
+ str = _T("kHoldSendAndPlay");
+ }
+ else if (_checkOnHold1 % 3 == 1)
+ {
+ mode = kHoldSendOnly;
+ str = _T("kHoldSendOnly");
+ }
+ else if (_checkOnHold1 % 3 == 2)
+ {
+ mode = kHoldPlayOnly;
+ str = _T("kHoldPlayOnly");
+ }
+ TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
+ _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
+ _checkOnHold1++;
+ }
+ else
+ {
+ TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
+ _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckOnHold2()
+{
+ int ret(0);
+ int channel = GetDlgItemInt(IDC_EDIT_2);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ OnHoldModes mode(kHoldSendAndPlay);
+ if (_checkOnHold1 % 3 == 0)
+ {
+ mode = kHoldSendAndPlay;
+ str = _T("kHoldSendAndPlay");
+ }
+ else if (_checkOnHold1 % 3 == 1)
+ {
+ mode = kHoldSendOnly;
+ str = _T("kHoldSendOnly");
+ }
+ else if (_checkOnHold1 % 3 == 2)
+ {
+ mode = kHoldPlayOnly;
+ str = _T("kHoldPlayOnly");
+ }
+ TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
+ _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
+ _checkOnHold1++;
+ }
+ else
+ {
+ TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
+ _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckDelayEstimate1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ if (enable)
+ {
+ _delayEstimate1 = true;
+ SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, 0);
+ }
+ else
+ {
+ _delayEstimate1 = false;
+ SetDlgItemText(IDC_EDIT_DELAY_ESTIMATE_1, _T(""));
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckRxvad()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ if (enable)
+ {
+ _rxVad = true;
+ _veApmPtr->RegisterRxVadObserver(channel, *_rxVadObserverPtr);
+ SetDlgItemInt(IDC_EDIT_RXVAD, 0);
+ }
+ else
+ {
+ _rxVad = false;
+ _veApmPtr->DeRegisterRxVadObserver(channel);
+ SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckAgc1()
+{
+ SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
+
+ CString str;
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+
+ if (shiftKeyIsPressed)
+ {
+ bool enabled(false);
+ AgcModes mode(kAgcAdaptiveDigital);
+ TEST(_veApmPtr->GetRxAgcStatus(channel, enabled, mode) == 0,
+ _T("GetRxAgcStatus(channel=%d, enabled=?, mode=?)"), channel);
+ button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
+
+ switch (mode)
+ {
+ case kAgcAdaptiveAnalog:
+ str = _T("kAgcAdaptiveAnalog");
+ break;
+ case kAgcAdaptiveDigital:
+ str = _T("kAgcAdaptiveDigital");
+ break;
+ case kAgcFixedDigital:
+ str = _T("kAgcFixedDigital");
+ break;
+ default:
+ break;
+ }
+ PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
+ return;
+ }
+
+ if (enable)
+ {
+ CString str;
+ AgcModes mode(kAgcDefault);
+ if (_checkAGC1 % 3 == 0)
+ {
+ mode = kAgcDefault;
+ str = _T("kAgcDefault");
+ }
+ else if (_checkAGC1 % 3 == 1)
+ {
+ mode = kAgcAdaptiveDigital;
+ str = _T("kAgcAdaptiveDigital");
+ }
+ else if (_checkAGC1 % 3 == 2)
+ {
+ mode = kAgcFixedDigital;
+ str = _T("kAgcFixedDigital");
+ }
+ TEST(_veApmPtr->SetRxAgcStatus(channel, true, mode) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
+ _checkAGC1++;
+ }
+ else
+ {
+ TEST(_veApmPtr->SetRxAgcStatus(channel, false, kAgcUnchanged) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d)"), channel, enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckNs1()
+{
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
+ int check = buttonNS->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ NsModes mode(kNsDefault);
+ if (_checkNS1 % 6 == 0)
+ {
+ mode = kNsDefault;
+ str = _T("kNsDefault");
+ }
+ else if (_checkNS1 % 6 == 1)
+ {
+ mode = kNsConference;
+ str = _T("kNsConference");
+ }
+ else if (_checkNS1 % 6 == 2)
+ {
+ mode = kNsLowSuppression;
+ str = _T("kNsLowSuppression");
+ }
+ else if (_checkNS1 % 6 == 3)
+ {
+ mode = kNsModerateSuppression;
+ str = _T("kNsModerateSuppression");
+ }
+ else if (_checkNS1 % 6 == 4)
+ {
+ mode = kNsHighSuppression;
+ str = _T("kNsHighSuppression");
+ }
+ else if (_checkNS1 % 6 == 5)
+ {
+ mode = kNsVeryHighSuppression;
+ str = _T("kNsVeryHighSuppression");
+ }
+ TEST(_veApmPtr->SetRxNsStatus(channel, true, mode) == 0, _T("SetRxNsStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
+ _checkNS1++;
+ }
+ else
+ {
+ TEST(_veApmPtr->SetRxNsStatus(channel, false, kNsUnchanged) == 0, _T("SetRxNsStatus(channel=%d, enable=%d)"), enable, channel);
+ }
+}
+
+// ----------------------------------------------------------------------------
+// Channel-independent Operations
+// ----------------------------------------------------------------------------
+
+void CWinTestDlg::OnBnClickedCheckPlayFileIn()
+{
+ // File path is relative to the location of 'voice_engine.gyp'.
+ const char micFile[] = "../../test/data/voice_engine/audio_short16.pcm";
+ // const char micFile[] = "../../test/data/voice_engine/audio_long16noise.pcm";
+
+ int ret(0);
+ int channel(-1);
+ CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN);
+ int check = buttonExtTrans->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ bool mix;
+ const bool loop(true);
+ const FileFormats format = kFileFormatPcm16kHzFile;
+ const float scale(1.0);
+
+ (_checkPlayFileIn %2 == 0) ? mix = true : mix = false;
+ TEST(_veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0,
+ _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
+ channel, CharToTchar(micFile, -1), loop, mix, format, scale);
+ _checkPlayFileIn++;
+ }
+ else
+ {
+ TEST(_veFilePtr->StopPlayingFileAsMicrophone(channel) == 0,
+ _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckRecMic()
+{
+ const char micFile[] = "/tmp/rec_mic_mono_16kHz.pcm";
+
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_REC_MIC);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veFilePtr->StartRecordingMicrophone(micFile, NULL) == 0, _T("StartRecordingMicrophone(file=%s)"), CharToTchar(micFile, -1));
+ }
+ else
+ {
+ TEST(_veFilePtr->StopRecordingMicrophone() == 0, _T("StopRecordingMicrophone()"));
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckAgc()
+{
+ CButton* buttonAGC = (CButton*)GetDlgItem(IDC_CHECK_AGC);
+ int check = buttonAGC->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ AgcModes mode(kAgcDefault);
+ if (_checkAGC % 4 == 0)
+ {
+ mode = kAgcDefault;
+ str = _T("kAgcDefault");
+ }
+ else if (_checkAGC % 4 == 1)
+ {
+ mode = kAgcAdaptiveAnalog;
+ str = _T("kAgcAdaptiveAnalog");
+ }
+ else if (_checkAGC % 4 == 2)
+ {
+ mode = kAgcAdaptiveDigital;
+ str = _T("kAgcAdaptiveDigital");
+ }
+ else if (_checkAGC % 4 == 3)
+ {
+ mode = kAgcFixedDigital;
+ str = _T("kAgcFixedDigital");
+ }
+ TEST(_veApmPtr->SetAgcStatus(true, mode) == 0, _T("SetAgcStatus(enable=%d, %s)"), enable, str);
+ _checkAGC++;
+ }
+ else
+ {
+ TEST(_veApmPtr->SetAgcStatus(false, kAgcUnchanged) == 0, _T("SetAgcStatus(enable=%d)"), enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckNs()
+{
+ CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS);
+ int check = buttonNS->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ NsModes mode(kNsDefault);
+ if (_checkNS % 6 == 0)
+ {
+ mode = kNsDefault;
+ str = _T("kNsDefault");
+ }
+ else if (_checkNS % 6 == 1)
+ {
+ mode = kNsConference;
+ str = _T("kNsConference");
+ }
+ else if (_checkNS % 6 == 2)
+ {
+ mode = kNsLowSuppression;
+ str = _T("kNsLowSuppression");
+ }
+ else if (_checkNS % 6 == 3)
+ {
+ mode = kNsModerateSuppression;
+ str = _T("kNsModerateSuppression");
+ }
+ else if (_checkNS % 6 == 4)
+ {
+ mode = kNsHighSuppression;
+ str = _T("kNsHighSuppression");
+ }
+ else if (_checkNS % 6 == 5)
+ {
+ mode = kNsVeryHighSuppression;
+ str = _T("kNsVeryHighSuppression");
+ }
+ TEST(_veApmPtr->SetNsStatus(true, mode) == 0, _T("SetNsStatus(enable=%d, %s)"), enable, str);
+ _checkNS++;
+ }
+ else
+ {
+ TEST(_veApmPtr->SetNsStatus(false, kNsUnchanged) == 0, _T("SetNsStatus(enable=%d)"), enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckEc()
+{
+ CButton* buttonEC = (CButton*)GetDlgItem(IDC_CHECK_EC);
+ int check = buttonEC->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ CString str;
+ EcModes mode(kEcDefault);
+ if (_checkEC % 4 == 0)
+ {
+ mode = kEcDefault;
+ str = _T("kEcDefault");
+ }
+ else if (_checkEC % 4 == 1)
+ {
+ mode = kEcConference;
+ str = _T("kEcConference");
+ }
+ else if (_checkEC % 4 == 2)
+ {
+ mode = kEcAec;
+ str = _T("kEcAec");
+ }
+ else if (_checkEC % 4 == 3)
+ {
+ mode = kEcAecm;
+ str = _T("kEcAecm");
+ }
+ TEST(_veApmPtr->SetEcStatus(true, mode) == 0, _T("SetEcStatus(enable=%d, %s)"), enable, str);
+ _checkEC++;
+ }
+ else
+ {
+ TEST(_veApmPtr->SetEcStatus(false, kEcUnchanged) == 0, _T("SetEcStatus(enable=%d)"), enable);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckMuteIn()
+{
+ CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN);
+ int check = buttonMute->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ const int channel(-1);
+ TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
+ _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaIn()
+{
+ const int channel(-1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed)"), channel);
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckExtMediaOut()
+{
+ const int channel(-1);
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ if (enable)
+ {
+ TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed, *_externalMediaPtr) == 0,
+ _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
+ }
+ else
+ {
+ TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed) == 0,
+ _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed)"), channel);
+ }
+}
+
+void CWinTestDlg::OnCbnSelchangeComboRecDevice()
+{
+ CComboBox* comboCodec(NULL);
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
+ int index = comboCodec->GetCurSel();
+ TEST(_veHardwarePtr->SetRecordingDevice(index) == 0,
+ _T("SetRecordingDevice(index=%d)"), index);
+}
+
+void CWinTestDlg::OnCbnSelchangeComboPlayDevice()
+{
+ CComboBox* comboCodec(NULL);
+ comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
+ int index = comboCodec->GetCurSel();
+ TEST(_veHardwarePtr->SetPlayoutDevice(index) == 0,
+ _T("SetPlayoutDevice(index=%d)"), index);
+}
+
+void CWinTestDlg::OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult)
+{
+ CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
+ slider->SetRangeMin(0);
+ slider->SetRangeMax(255);
+ int pos = slider->GetPos();
+
+ TEST(_veVolumeControlPtr->SetMicVolume(pos) == 0, _T("SetMicVolume(volume=%d)"), pos);
+
+ *pResult = 0;
+}
+
+void CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult)
+{
+ CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
+ slider->SetRangeMin(0);
+ slider->SetRangeMax(255);
+ int pos = slider->GetPos();
+
+ TEST(_veVolumeControlPtr->SetSpeakerVolume(pos) == 0, _T("SetSpeakerVolume(volume=%d)"), pos);
+
+ *pResult = 0;
+}
+
+void CWinTestDlg::OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult)
+{
+ CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
+ slider->SetRange(0,10);
+ int pos = 10 - slider->GetPos(); // 0 <=> lower end, 10 <=> upper end
+
+ float left(0.0);
+ float right(0.0);
+ const int channel(-1);
+
+ // Only left channel will be modified
+ _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
+
+ left = (float)((float)pos/10.0f);
+
+ TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
+ _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
+
+ *pResult = 0;
+}
+
+void CWinTestDlg::OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult)
+{
+ CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
+ slider->SetRange(0,10);
+ int pos = 10 - slider->GetPos(); // 0 <=> lower end, 10 <=> upper end
+
+ float left(0.0);
+ float right(0.0);
+ const int channel(-1);
+
+ // Only right channel will be modified
+ _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
+
+ right = (float)((float)pos/10.0f);
+
+ TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
+ _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
+
+ *pResult = 0;
+}
+
+void CWinTestDlg::OnBnClickedButtonVersion()
+{
+ if (_veBasePtr)
+ {
+ char version[1024];
+ if (_veBasePtr->GetVersion(version) == 0)
+ {
+ AfxMessageBox(CString(version), MB_OK);
+ }
+ else
+ {
+ AfxMessageBox(_T("FAILED!"), MB_OK);
+ }
+ }
+}
+
+void CWinTestDlg::OnBnClickedCheckRecCall()
+{
+ // Not supported
+}
+
+void CWinTestDlg::OnBnClickedCheckTypingDetection()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_TYPING_DETECTION);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ TEST(_veApmPtr->SetTypingDetectionStatus(enable) == 0, _T("SetTypingDetectionStatus(enable=%d)"), enable);
+}
+
+void CWinTestDlg::OnBnClickedCheckFEC()
+{
+ CButton* button = (CButton*)GetDlgItem(IDC_CHECK_FEC);
+ int channel = GetDlgItemInt(IDC_EDIT_1);
+ int check = button->GetCheck();
+ const bool enable = (check == BST_CHECKED);
+ TEST(_veRtpRtcpPtr->SetFECStatus(channel, enable) == 0, _T("SetFECStatus(enable=%d)"), enable);
+}
+
+// ----------------------------------------------------------------------------
+// Message Handlers
+// ----------------------------------------------------------------------------
+
+void CWinTestDlg::OnTimer(UINT_PTR nIDEvent)
+{
+ CString str;
+
+ unsigned int svol(0);
+ unsigned int mvol(0);
+
+ _timerTicks++;
+
+ // Get speaker and microphone volumes
+ _veVolumeControlPtr->GetSpeakerVolume(svol);
+ _veVolumeControlPtr->GetMicVolume(mvol);
+
+ // Update speaker volume slider
+ CSliderCtrl* sliderSpkr = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
+ sliderSpkr->SetRangeMin(0);
+ sliderSpkr->SetRangeMax(255);
+ sliderSpkr->SetPos(svol);
+
+ // Update microphone volume slider
+ CSliderCtrl* sliderMic = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
+ sliderMic->SetRangeMin(0);
+ sliderMic->SetRangeMax(255);
+ sliderMic->SetPos(mvol);
+
+ unsigned int micLevel;
+ unsigned int combinedOutputLevel;
+
+ // Get audio levels
+ _veVolumeControlPtr->GetSpeechInputLevel(micLevel);
+ _veVolumeControlPtr->GetSpeechOutputLevel(-1, combinedOutputLevel);
+
+ // Update audio level controls
+ CProgressCtrl* progressMic = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_IN);
+ progressMic->SetRange(0,9);
+ progressMic->SetStep(1);
+ progressMic->SetPos(micLevel);
+ CProgressCtrl* progressOut = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_OUT);
+ progressOut->SetRange(0,9);
+ progressOut->SetStep(1);
+ progressOut->SetPos(combinedOutputLevel);
+
+ // Update playout delay (buffer size)
+ if (_veVideoSyncPtr)
+ {
+ int bufferMs(0);
+ _veVideoSyncPtr->GetPlayoutBufferSize(bufferMs);
+ SetDlgItemInt(IDC_EDIT_PLAYOUT_BUFFER_SIZE, bufferMs);
+ }
+
+ if (_delayEstimate1 && _veVideoSyncPtr)
+ {
+ const int channel = GetDlgItemInt(IDC_EDIT_1);
+ int delayMs(0);
+ _veVideoSyncPtr->GetDelayEstimate(channel, delayMs);
+ SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, delayMs);
+ }
+
+ if (_rxVad && _veApmPtr && _rxVadObserverPtr)
+ {
+ SetDlgItemInt(IDC_EDIT_RXVAD, _rxVadObserverPtr->_vadDecision);
+ }
+
+ if (_veHardwarePtr)
+ {
+ int load1, load2;
+ _veHardwarePtr->GetSystemCPULoad(load1);
+ _veHardwarePtr->GetCPULoad(load2);
+ str.Format(_T("CPU load (system/VoE): %d/%d [%%]"), load1, load2);
+ SetDlgItemText(IDC_EDIT_CPU_LOAD, (LPCTSTR)str);
+ }
+
+ BOOL ret;
+ int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
+
+ if (_veCodecPtr)
+ {
+ if (ret == TRUE)
+ {
+ CodecInst codec;
+ if (_veCodecPtr->GetRecCodec(channel, codec) == 0)
+ {
+ str.Format(_T("RX codec: %s, freq=%d, pt=%d, rate=%d, size=%d"), CharToTchar(codec.plname, 32), codec.plfreq, codec.pltype, codec.rate, codec.pacsize);
+ SetDlgItemText(IDC_EDIT_RX_CODEC_1, (LPCTSTR)str);
+ }
+ }
+ }
+
+ if (_veRtpRtcpPtr)
+ {
+ if (ret == TRUE)
+ {
+ CallStatistics stats;
+ if (_veRtpRtcpPtr->GetRTCPStatistics(channel, stats) == 0)
+ {
+ str.Format(_T("RTCP | RTP: cum=%u, ext=%d, frac=%u, jitter=%u | TX=%d, RX=%d, RTT=%d"),
+ stats.cumulativeLost, stats.extendedMax, stats.fractionLost, stats.jitterSamples, stats.packetsSent, stats.packetsReceived, stats.rttMs);
+ SetDlgItemText(IDC_EDIT_RTCP_STAT_1, (LPCTSTR)str);
+ }
+ }
+ }
+
+ SetTimer(0, 1000, NULL);
+ CDialog::OnTimer(nIDEvent);
+}
+
+void CWinTestDlg::OnBnClickedButtonClearErrorCallback()
+{
+ _nErrorCallbacks = 0;
+ SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, _T(""));
+}
+
+// ----------------------------------------------------------------------------
+// TEST
+// ----------------------------------------------------------------------------
+
+void CWinTestDlg::OnBnClickedButtonTest1()
+{
+ // add tests here...
+}
+
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h
new file mode 100644
index 0000000..6452775
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h
@@ -0,0 +1,278 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma once
+
+#if (_MSC_VER >= 1400)
+#define PRINT_GET_RESULT(...) \
+ { \
+ _strMsg.Format(__VA_ARGS__); \
+ SetDlgItemText(IDC_EDIT_GET_OUTPUT, _strMsg); \
+ } \
+
+#define TEST(x, ...) \
+ if (!(x)) \
+ { \
+ _strMsg.Format(__VA_ARGS__); \
+ SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
+ _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError()); \
+ SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
+ _failCount++; \
+ SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
+ SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
+ } \
+ else \
+ { \
+ _strMsg.Format(__VA_ARGS__); \
+ SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
+ SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
+ } \
+
+#define TEST2(x, ...) \
+ if (!(x)) \
+ { \
+ _strMsg.Format(__VA_ARGS__); \
+ ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
+ } \
+ else \
+ { \
+ _strMsg.Format(__VA_ARGS__); \
+ ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
+ }
+#else
+#define TEST(x, exp) \
+ if (!(x)) \
+ { \
+ _strMsg.Format(exp); \
+ SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
+ _strErr.Format("FAILED (error=%d)", _veBasePtr->LastError()); \
+ SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
+ _failCount++; \
+ SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
+ SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
+ } \
+ else \
+ { \
+ _strMsg.Format(exp); \
+ SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
+ SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
+ } \
+
+#define TEST2(x, exp) \
+ if (!(x)) \
+ { \
+ _strMsg.Format(exp); \
+ ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
+ } \
+ else \
+ { \
+ _strMsg.Format(exp); \
+ ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
+ }
+#endif
+
+#include "voe_base.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_codec.h"
+#include "voe_dtmf.h"
+#include "voe_encryption.h"
+#include "voe_external_media.h"
+#include "voe_file.h"
+#include "voe_hardware.h"
+#include "voe_network.h"
+#include "voe_video_sync.h"
+#include "voe_volume_control.h"
+
+#include "voe_audio_processing.h"
+#include "voe_rtp_rtcp.h"
+#include "voe_errors.h"
+
+class MediaProcessImpl;
+class ConnectionObserver;
+class MyEncryption;
+class RxCallback;
+class MyTransport;
+
+using namespace webrtc;
+
+#define MAX_NUM_OF_CHANNELS 10
+
+// CWinTestDlg dialog
+class CWinTestDlg : public CDialog,
+ public VoiceEngineObserver,
+ public VoERTPObserver
+{
+// Construction
+public:
+ CWinTestDlg(CWnd* pParent = NULL); // standard constructor
+ virtual ~CWinTestDlg();
+
+// Dialog Data
+ enum { IDD = IDD_WINTEST_DIALOG };
+
+ BOOL UpdateTest(bool failed, const CString& strMsg);
+
+protected:
+ virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
+
+protected: // VoiceEngineObserver
+ virtual void CallbackOnError(const int channel, const int errCode);
+
+protected: // VoERTPObserver
+ virtual void OnIncomingCSRCChanged(
+ const int channel, const unsigned int CSRC, const bool added);
+ virtual void OnIncomingSSRCChanged(
+ const int channel, const unsigned int SSRC);
+
+// Implementation
+protected:
+ HICON m_hIcon;
+
+ // Generated message map functions
+ virtual BOOL OnInitDialog();
+ afx_msg void OnSysCommand(UINT nID, LPARAM lParam);
+ afx_msg void OnPaint();
+ afx_msg HCURSOR OnQueryDragIcon();
+ DECLARE_MESSAGE_MAP()
+public:
+ afx_msg void OnBnClickedButtonCreate1();
+ afx_msg void OnBnClickedButtonDelete1();
+
+private:
+ VoiceEngine* _vePtr;
+
+ VoECodec* _veCodecPtr;
+ VoEExternalMedia* _veExternalMediaPtr;
+ VoEVolumeControl* _veVolumeControlPtr;
+ VoEEncryption* _veEncryptionPtr;
+ VoEHardware* _veHardwarePtr;
+ VoEVideoSync* _veVideoSyncPtr;
+ VoENetwork* _veNetworkPtr;
+ VoEFile* _veFilePtr;
+ VoEAudioProcessing* _veApmPtr;
+ VoEBase* _veBasePtr;
+ VoERTP_RTCP* _veRtpRtcpPtr;
+
+ MyTransport* _transportPtr;
+ MediaProcessImpl* _externalMediaPtr;
+ ConnectionObserver* _connectionObserverPtr;
+ MyEncryption* _encryptionPtr;
+ RxCallback* _rxVadObserverPtr;
+
+private:
+ int _failCount;
+ CString _strMsg;
+ CString _strErr;
+ bool _externalTransport;
+ bool _externalTransportBuild;
+ int _checkPlayFileIn;
+ int _checkPlayFileIn1;
+ int _checkPlayFileIn2;
+ int _checkPlayFileOut1;
+ int _checkPlayFileOut2;
+ int _checkAGC;
+ int _checkAGC1;
+ int _checkNS;
+ int _checkNS1;
+ int _checkEC;
+ int _checkVAD1;
+ int _checkVAD2;
+ int _checkSrtpTx1;
+ int _checkSrtpTx2;
+ int _checkSrtpRx1;
+ int _checkSrtpRx2;
+ int _checkConference1;
+ int _checkConference2;
+ int _checkOnHold1;
+ int _checkOnHold2;
+ bool _delayEstimate1;
+ bool _delayEstimate2;
+ bool _rxVad;
+ int _nErrorCallbacks;
+ int _timerTicks;
+
+public:
+ afx_msg void OnBnClickedButtonCreate2();
+ afx_msg void OnBnClickedButtonDelete2();
+ afx_msg void OnCbnSelchangeComboCodec1();
+ afx_msg void OnBnClickedButtonStartListen1();
+ afx_msg void OnBnClickedButtonStopListen1();
+ afx_msg void OnBnClickedButtonStartPlayout1();
+ afx_msg void OnBnClickedButtonStopPlayout1();
+ afx_msg void OnBnClickedButtonStartSend1();
+ afx_msg void OnBnClickedButtonStopSend1();
+ afx_msg void OnCbnSelchangeComboIp2();
+ afx_msg void OnCbnSelchangeComboIp1();
+ afx_msg void OnCbnSelchangeComboCodec2();
+ afx_msg void OnBnClickedButtonStartListen2();
+ afx_msg void OnBnClickedButtonStopListen2();
+ afx_msg void OnBnClickedButtonStartPlayout2();
+ afx_msg void OnBnClickedButtonStopPlayout2();
+ afx_msg void OnBnClickedButtonStartSend2();
+ afx_msg void OnBnClickedButtonStopSend2();
+ afx_msg void OnBnClickedButtonTest11();
+ afx_msg void OnBnClickedCheckExtTrans1();
+ afx_msg void OnBnClickedCheckPlayFileIn1();
+ afx_msg void OnBnClickedCheckPlayFileOut1();
+ afx_msg void OnBnClickedCheckExtTrans2();
+ afx_msg void OnBnClickedCheckPlayFileIn2();
+ afx_msg void OnBnClickedCheckPlayFileOut2();
+ afx_msg void OnBnClickedCheckPlayFileIn();
+ afx_msg void OnBnClickedCheckPlayFileOut();
+ afx_msg void OnCbnSelchangeComboRecDevice();
+ afx_msg void OnCbnSelchangeComboPlayDevice();
+ afx_msg void OnBnClickedCheckExtMediaIn1();
+ afx_msg void OnBnClickedCheckExtMediaOut1();
+ afx_msg void OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult);
+ afx_msg void OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult);
+ afx_msg void OnTimer(UINT_PTR nIDEvent);
+ afx_msg void OnBnClickedCheckAgc();
+ CString _strComboIp1;
+ CString _strComboIp2;
+ afx_msg void OnBnClickedCheckNs();
+ afx_msg void OnBnClickedCheckEc();
+ afx_msg void OnBnClickedCheckVad1();
+ afx_msg void OnBnClickedCheckVad2();
+ afx_msg void OnBnClickedCheckExtMediaIn2();
+ afx_msg void OnBnClickedCheckExtMediaOut2();
+ afx_msg void OnBnClickedCheckMuteIn();
+ afx_msg void OnBnClickedCheckMuteIn1();
+ afx_msg void OnBnClickedCheckMuteIn2();
+ afx_msg void OnBnClickedCheckSrtpTx1();
+ afx_msg void OnBnClickedCheckSrtpRx1();
+ afx_msg void OnBnClickedCheckSrtpTx2();
+ afx_msg void OnBnClickedCheckSrtpRx2();
+ afx_msg void OnBnClickedCheckExtEncryption1();
+ afx_msg void OnBnClickedCheckExtEncryption2();
+ afx_msg void OnBnClickedButtonDtmf1();
+ afx_msg void OnBnClickedCheckRecMic();
+ afx_msg void OnBnClickedButtonDtmf2();
+ afx_msg void OnBnClickedButtonTest1();
+ afx_msg void OnBnClickedCheckConference1();
+ afx_msg void OnBnClickedCheckConference2();
+ afx_msg void OnBnClickedCheckOnHold1();
+ afx_msg void OnBnClickedCheckOnHold2();
+ afx_msg void OnBnClickedCheckExtMediaIn();
+ afx_msg void OnBnClickedCheckExtMediaOut();
+ afx_msg void OnLbnSelchangeListCodec1();
+ afx_msg void OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult);
+ afx_msg void OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult);
+ afx_msg void OnBnClickedButtonVersion();
+ afx_msg void OnBnClickedCheckDelayEstimate1();
+ afx_msg void OnBnClickedCheckRxvad();
+ afx_msg void OnBnClickedCheckAgc1();
+ afx_msg void OnBnClickedCheckNs1();
+ afx_msg void OnBnClickedCheckRecCall();
+ afx_msg void OnBnClickedCheckTypingDetection();
+ afx_msg void OnBnClickedCheckFEC();
+ afx_msg void OnBnClickedButtonClearErrorCallback();
+ afx_msg void OnBnClickedCheckBwe1();
+};
+#pragma once
diff --git a/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico b/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico
new file mode 100644
index 0000000..8a84ca3
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2 b/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2
new file mode 100644
index 0000000..044bf7e
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2
@@ -0,0 +1,13 @@
+//
+// WinTest.RC2 - resources Microsoft Visual C++ does not edit directly
+//
+
+#ifdef APSTUDIO_INVOKED
+#error this file is not editable by Microsoft Visual C++
+#endif //APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+// Add manually edited resources here...
+
+/////////////////////////////////////////////////////////////////////////////
diff --git a/trunk/src/voice_engine/main/test/win_test/stdafx.cc b/trunk/src/voice_engine/main/test/win_test/stdafx.cc
new file mode 100644
index 0000000..6cdb906
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/stdafx.cc
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// stdafx.cpp : source file that includes just the standard includes
+// WinTest.pch will be the pre-compiled header
+// stdafx.obj will contain the pre-compiled type information
+
+#include "stdafx.h"
+
+
diff --git a/trunk/src/voice_engine/main/test/win_test/stdafx.h b/trunk/src/voice_engine/main/test/win_test/stdafx.h
new file mode 100644
index 0000000..b4d875c
--- /dev/null
+++ b/trunk/src/voice_engine/main/test/win_test/stdafx.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// stdafx.h : include file for standard system include files,
+// or project specific include files that are used frequently,
+// but are changed infrequently
+
+#pragma once
+
+#ifndef _SECURE_ATL
+#define _SECURE_ATL 1
+#endif
+
+#ifndef VC_EXTRALEAN
+#define VC_EXTRALEAN // Exclude rarely-used stuff from Windows headers
+#endif
+
+// Modify the following defines if you have to target a platform prior to the ones specified below.
+// Refer to MSDN for the latest info on corresponding values for different platforms.
+#ifndef WINVER // Allow use of features specific to Windows XP or later.
+#define WINVER 0x0501 // Change this to the appropriate value to target other versions of Windows.
+#endif
+
+#ifndef _WIN32_WINNT // Allow use of features specific to Windows XP or later.
+#define _WIN32_WINNT 0x0501 // Change this to the appropriate value to target other versions of Windows.
+#endif
+
+#ifndef _WIN32_WINDOWS // Allow use of features specific to Windows 98 or later.
+#define _WIN32_WINDOWS 0x0410 // Change this to the appropriate value to target Windows Me or later.
+#endif
+
+#ifndef _WIN32_IE // Allow use of features specific to IE 6.0 or later.
+#define _WIN32_IE 0x0600 // Change this to the appropriate value to target other versions of IE.
+#endif
+
+#define _ATL_CSTRING_EXPLICIT_CONSTRUCTORS // some CString constructors will be explicit
+
+// turns off MFC's hiding of some common and often safely ignored warning messages
+#define _AFX_ALL_WARNINGS
+
+#include <afxwin.h> // MFC core and standard components
+#include <afxext.h> // MFC extensions
+
+
+
+
+
+#ifndef _AFX_NO_OLE_SUPPORT
+#include <afxdtctl.h> // MFC support for Internet Explorer 4 Common Controls
+#endif
+#ifndef _AFX_NO_AFXCMN_SUPPORT
+#include <afxcmn.h> // MFC support for Windows Common Controls
+#endif // _AFX_NO_AFXCMN_SUPPORT
+
+
+
+
+
+
+
+
+
+#ifdef _UNICODE
+#if defined _M_IX86
+#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='x86' publicKeyToken='6595b64144ccf1df' language='*'\"")
+#elif defined _M_IA64
+#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='ia64' publicKeyToken='6595b64144ccf1df' language='*'\"")
+#elif defined _M_X64
+#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='amd64' publicKeyToken='6595b64144ccf1df' language='*'\"")
+#else
+#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='*' publicKeyToken='6595b64144ccf1df' language='*'\"")
+#endif
+#endif
+
+
diff --git a/trunk/src/voice_engine/voice_engine.gyp b/trunk/src/voice_engine/voice_engine.gyp
new file mode 100644
index 0000000..f45a269
--- /dev/null
+++ b/trunk/src/voice_engine/voice_engine.gyp
@@ -0,0 +1,23 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ 'main/source/voice_engine_core.gypi',
+ ],
+
+ # Test targets, excluded when building with Chromium.
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'includes': [
+ 'main/test/voice_engine_tests.gypi',
+ ],
+ }],
+ ],
+}
diff --git a/trunk/test/OWNERS b/trunk/test/OWNERS
new file mode 100644
index 0000000..fec9caa
--- /dev/null
+++ b/trunk/test/OWNERS
@@ -0,0 +1,4 @@
+phoglund@webrtc.org
+kjellander@webrtc.org
+ivinnichenko@webrtc.org
+
diff --git a/trunk/test/data/audio_coding/F00.BIT20 b/trunk/test/data/audio_coding/F00.BIT20
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+++ b/trunk/test/data/audio_coding/F06.INP
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diff --git a/trunk/test/data/audio_coding/READ.ME b/trunk/test/data/audio_coding/READ.ME
new file mode 100644
index 0000000..e42b9b3
--- /dev/null
+++ b/trunk/test/data/audio_coding/READ.ME
@@ -0,0 +1,18 @@
+Test files for Audio Coding Module
+
+testfile32kHz.pcm - mono speech file samples at 32 kHz
+teststereo32kHz.pcm - stereo speech file samples at 32 kHz
+
+Test and reference vectors to verify correct execution of PacketCable
+iLBC Fixed Point Reference Code
+
+Version 1.0.6
+Format: all .INP and .OUT files contain 16 bit sampled data using the
+Intel (PC) format. The .BIT files are stored in the appropriate byte
+sequence (big-endian format).
+
+*.INP - input files
+*.BIT20 - bit stream files 20 ms mode
+*.OUT20 - output files 20 ms mode (on a channel without packet loss)
+*.BIT30 - bit stream files 30 ms mode
+*.OUT30 - output files 30 ms mode (on a channel without packet loss)
diff --git a/trunk/test/data/audio_coding/clean.chn b/trunk/test/data/audio_coding/clean.chn
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+++ b/trunk/test/data/audio_coding/clean.chn
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diff --git a/trunk/test/data/audio_processing/output_data_float.pb b/trunk/test/data/audio_processing/output_data_float.pb
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diff --git a/trunk/test/data/common_video/jpeg/webrtc_logo.jpg b/trunk/test/data/common_video/jpeg/webrtc_logo.jpg
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index 0000000..ddb6192
--- /dev/null
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diff --git a/trunk/test/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin b/trunk/test/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin
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diff --git a/trunk/test/data/rtp_rtcp/H263_QCIF_IFRAME.bin b/trunk/test/data/rtp_rtcp/H263_QCIF_IFRAME.bin
new file mode 100644
index 0000000..0fa144c
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/H263_QCIF_IFRAME.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR0.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR0.bin
new file mode 100644
index 0000000..19df13c
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR0.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR1.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR1.bin
new file mode 100644
index 0000000..b7b7c94
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR1.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR2.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR2.bin
new file mode 100644
index 0000000..257835c
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR2.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR3.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR3.bin
new file mode 100644
index 0000000..4a8e375
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR3.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4.bin
new file mode 100644
index 0000000..28cd99c
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin
new file mode 100644
index 0000000..5080b88
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin
new file mode 100644
index 0000000..2c2f288
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin
Binary files differ
diff --git a/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR5.bin b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR5.bin
new file mode 100644
index 0000000..da7235a
--- /dev/null
+++ b/trunk/test/data/rtp_rtcp/RTCPPacketTMMBR5.bin
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long16.pcm b/trunk/test/data/voice_engine/audio_long16.pcm
new file mode 100644
index 0000000..853e0df
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long16.pcm
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long16.wav b/trunk/test/data/voice_engine/audio_long16.wav
new file mode 100644
index 0000000..ebe91c4
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long16.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long16big_endian.pcm b/trunk/test/data/voice_engine/audio_long16big_endian.pcm
new file mode 100644
index 0000000..563e4e9
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long16big_endian.pcm
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long16noise.pcm b/trunk/test/data/voice_engine/audio_long16noise.pcm
new file mode 100644
index 0000000..a7be537
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long16noise.pcm
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long8.pcm b/trunk/test/data/voice_engine/audio_long8.pcm
new file mode 100644
index 0000000..85d17e5
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long8.pcm
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_long8mulaw.wav b/trunk/test/data/voice_engine/audio_long8mulaw.wav
new file mode 100644
index 0000000..2d3d8b3
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_long8mulaw.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_short16.pcm b/trunk/test/data/voice_engine/audio_short16.pcm
new file mode 100644
index 0000000..15a0f18
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_short16.pcm
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny11.wav b/trunk/test/data/voice_engine/audio_tiny11.wav
new file mode 100644
index 0000000..6db80d5
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny11.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny16.wav b/trunk/test/data/voice_engine/audio_tiny16.wav
new file mode 100644
index 0000000..baab0ac
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny16.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny22.wav b/trunk/test/data/voice_engine/audio_tiny22.wav
new file mode 100644
index 0000000..b421867
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny22.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny32.wav b/trunk/test/data/voice_engine/audio_tiny32.wav
new file mode 100644
index 0000000..773ac23
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny32.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny44.wav b/trunk/test/data/voice_engine/audio_tiny44.wav
new file mode 100644
index 0000000..c9faa45
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny44.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny48.wav b/trunk/test/data/voice_engine/audio_tiny48.wav
new file mode 100644
index 0000000..8ebf11a
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny48.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/audio_tiny8.wav b/trunk/test/data/voice_engine/audio_tiny8.wav
new file mode 100644
index 0000000..d71c65e
--- /dev/null
+++ b/trunk/test/data/voice_engine/audio_tiny8.wav
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp
new file mode 100644
index 0000000..02abbc2
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp
new file mode 100644
index 0000000..4ed110b
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/README.txt b/trunk/test/data/voice_engine/stereo_rtp_files/README.txt
new file mode 100644
index 0000000..976ac56
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/README.txt
@@ -0,0 +1,4 @@
+Use RTP Play tool with command 'rtpplay.exe -v -T -f <path>\<file.rtp> 127.0.0.1/1236'
+Example: rtpplay.exe -v -T -f hrtf_g722_1C_48.rtp 127.0.0.1/1236.
+This sends the stereo rtp file to port 1236.
+You can hear the voice getting panned from left, right and center.
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp
new file mode 100644
index 0000000..b96d59b
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp
new file mode 100644
index 0000000..527a50a
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/rtpplay.exe b/trunk/test/data/voice_engine/stereo_rtp_files/rtpplay.exe
new file mode 100755
index 0000000..6f938c8
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/rtpplay.exe
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729.rtp
new file mode 100644
index 0000000..3c36e30
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp
new file mode 100644
index 0000000..913226c
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp
new file mode 100644
index 0000000..729b565
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp
new file mode 100644
index 0000000..efa2800
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp
new file mode 100644
index 0000000..bb2d93c
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp
new file mode 100644
index 0000000..fb79378
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp
new file mode 100644
index 0000000..eebcf34
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp
new file mode 100644
index 0000000..5c368b4
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp
Binary files differ
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp
new file mode 100644
index 0000000..1f713f6
--- /dev/null
+++ b/trunk/test/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp
Binary files differ
diff --git a/trunk/test/functional_test/README b/trunk/test/functional_test/README
new file mode 100644
index 0000000..093f759
--- /dev/null
+++ b/trunk/test/functional_test/README
@@ -0,0 +1,46 @@
+This test client is a simple functional test for a WebRTC enabled browser. It
+has only been tested with Chrome, and is most likely only working with Chrome at
+the moment. The following instructions are in part Chrome specific.
+
+The following is necessary to run the test:
+- A WebRTC enabled Chrome binary. (Available in dev or canary channel, 18.0.1008
+or newer.)
+- A peerconnection_server binary (make peerconnection_server).
+
+It can be used in two scenarios:
+1. Single client calling itself with the server test page in loopback mode as a
+fake client.
+2. Call between two clients.
+
+To start the test for scenario (1):
+1. Start peerconnection_server.
+2. Start the WebRTC Chrome build:
+$ <path_to_chrome_binary>/chrome --enable-media-stream
+The --enable-media-stream flag is required for the time being.
+3. Open the server test page, ensure loopback is enabled, choose a name (for
+example "loopback") and connect to the server.
+For version 18.0.1008 to 18.0.1020, use:
+http://libjingle.googlecode.com/svn-history/r103/trunk/talk/examples/peerconnection/server/server_test.html
+For version 18.0.1021 and later, use:
+http://libjingle.googlecode.com/svn/trunk/talk/examples/peerconnection/server/server_test.html
+4. Open the test page, connect to the server, select the loopback peer, click
+call.
+
+To start the test for scenario (2):
+1. Start peerconnection_server.
+2. Start the WebRTC Chrome build, see scenario (1).
+3. Open the test page, connect to the server.
+4. Open a new new tab, open the test page, connect to the server.
+ OR
+ On another machine, repeat 2 and 3.
+5. Select the other peer, click call.
+
+Note: The web page must normally be on a web server to be able to access the
+camera for security reasons.
+See http://blog.chromium.org/2008/12/security-in-depth-local-web-pages.html
+for more details on this topic. This can be overridden with the flag
+--allow-file-access-from-files, in which case running it over the file://
+URI scheme works.
+
+Note: It's possible to specify the server and name in the url:
+.../webrtc.html?server=my_server&name=my_name
diff --git a/trunk/test/functional_test/webrtc_test.html b/trunk/test/functional_test/webrtc_test.html
new file mode 100644
index 0000000..300884c
--- /dev/null
+++ b/trunk/test/functional_test/webrtc_test.html
@@ -0,0 +1,602 @@
+<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
+
+<!--
+Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+
+Use of this source code is governed by a BSD-style license
+that can be found in the LICENSE file in the root of the source
+tree. An additional intellectual property rights grant can be found
+in the file PATENTS. All contributing project authors may
+be found in the AUTHORS file in the root of the source tree.
+-->
+
+<html>
+
+<head>
+<title>WebRTC Test</title>
+
+<style type="text/css">
+body, input, button, select, table {
+ font-family:"Lucida Grande", "Lucida Sans", Verdana, Arial, sans-serif;
+ font-size: 13 px;
+}
+body, input:enable, button:enable, select:enable, table {
+ color: rgb(51, 51, 51);
+}
+h1 {font-size: 40 px;}
+</style>
+
+<script type="text/javascript">
+
+// TODO: Catch more exceptions
+
+var server;
+var myId = -1;
+var myName;
+var remoteId = -1;
+var remoteName;
+var request = null;
+var hangingGet = null;
+var pc = null;
+var localStream = null;
+var disconnecting = false;
+var callState = 0; // 0 - Not started, 1 - Call ongoing
+
+
+// General
+
+function setElementValuesFromURL() {
+ window.location.href.replace(/[?&]+([^=&]+)=([^&]*)/gi,
+ function(m, key, value) {
+ document.getElementById(key).value = unescape(value);
+ });
+}
+
+function trace(txt) {
+ var elem = document.getElementById("debug");
+ elem.innerHTML += txt + "<br>";
+}
+
+function trace_warning(txt) {
+ var wtxt = "<b>" + txt + "</b>";
+ trace(wtxt);
+}
+
+function trace_exception(e, txt) {
+ var etxt = "<b>" + txt + "</b> (" + e.name + " / " + e.message + ")";
+ trace(etxt);
+}
+
+function setCallState(state) {
+ trace("Changing call state: " + callState + " -> " + state);
+ callState = state;
+}
+
+function checkPeerConnection() {
+ if (!pc) {
+ trace_warning("No PeerConnection object exists");
+ return 0;
+ }
+ return 1;
+}
+
+
+// Local stream generation
+
+function gotStream(s) {
+ var url = webkitURL.createObjectURL(s);
+ document.getElementById("localView").src = url;
+ trace("User has granted access to local media. url = " + url);
+ localStream = s;
+}
+
+function gotStreamFailed(error) {
+ alert("Failed to get access to local media. Error code was " + error.code +
+ ".");
+ trace_warning("Failed to get access to local media. Error code was " +
+ error.code);
+}
+
+function getUserMedia() {
+ try {
+ navigator.webkitGetUserMedia("video,audio", gotStream, gotStreamFailed);
+ trace("Requested access to local media");
+ } catch (e) {
+ trace_exception(e, "getUserMedia error");
+ }
+}
+
+
+// Peer list and remote peer handling
+
+function peerExists(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id)
+ return true;
+ }
+ } catch (e) {
+ trace_exception(e, "Error searching for peer");
+ }
+ return false;
+}
+
+function addPeer(id, pname) {
+ try {
+ var peerList = document.getElementById("peers");
+ var option = document.createElement("option");
+ option.text = pname;
+ option.value = id;
+ peerList.add(option, null);
+ } catch (e) {
+ trace_exception(e, "Error adding peer");
+ }
+}
+
+function removePeer(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ peerList.remove(i);
+ break;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error removing peer");
+ }
+}
+
+function clearPeerList() {
+ var peerList = document.getElementById("peers");
+ while (peerList.length > 0)
+ peerList.remove(0);
+}
+
+function setSelectedPeer(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ peerList.options[i].selected = true;
+ return true;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error setting selected peer");
+ }
+ return false;
+}
+
+function getPeerName(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ return peerList.options[i].text;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error finding peer name");
+ return;
+ }
+ return;
+}
+
+function storeRemoteInfo() {
+ try {
+ var peerList = document.getElementById("peers");
+ if (peerList.selectedIndex < 0) {
+ alert("Please select a peer.");
+ return false;
+ } else
+ remoteId = parseInt(peerList.options[peerList.selectedIndex].value);
+ remoteName = peerList.options[peerList.selectedIndex].text;
+ } catch (e) {
+ trace_exception(e, "Error storing remote peer info");
+ return false;
+ }
+ return true;
+}
+
+
+// Call control
+
+function createPeerConnection() {
+ if (pc) {
+ trace_warning("PeerConnection object already exists");
+ }
+ trace("Creating PeerConnection object");
+ try {
+ pc = new webkitPeerConnection("STUN stun.l.google.com:19302",
+ onSignalingMessage);
+ pc.onaddstream = onAddStream;
+ pc.onremovestream = onRemoveStream;
+ } catch (e) {
+ trace_exception(e, "Create PeerConnection error");
+ }
+}
+
+function doCall() {
+ if (!storeRemoteInfo())
+ return;
+ document.getElementById("call").disabled = true;
+ document.getElementById("peers").disabled = true;
+ createPeerConnection();
+ trace("Adding stream");
+ pc.addStream(localStream);
+ document.getElementById("hangup").disabled = false;
+ setCallState(1);
+}
+
+function hangUp() {
+ document.getElementById("hangup").disabled = true;
+ trace("Sending BYE to " + remoteName + " (ID " + remoteId + ")");
+ sendToPeer(remoteId, "BYE");
+ closeCall();
+}
+
+function closeCall() {
+ trace("Stopping showing remote stream");
+ document.getElementById("remoteView").src = "dummy";
+ if (pc) {
+ trace("Stopping call [pc.close()]");
+ pc.close();
+ pc = null;
+ } else
+ trace("No pc object to close");
+ remoteId = -1;
+ document.getElementById("call").disabled = false;
+ document.getElementById("peers").disabled = false;
+ setCallState(0);
+}
+
+
+// PeerConnection callbacks
+
+function onAddStream(e) {
+ var stream = e.stream;
+ var url = webkitURL.createObjectURL(stream);
+ document.getElementById("remoteView").src = url;
+ trace("Started showing remote stream. url = " + url);
+}
+
+function onRemoveStream(e) {
+ // Currently if we get this callback, call has ended.
+ document.getElementById("remoteView").src = "";
+ trace("Stopped showing remote stream");
+}
+
+function onSignalingMessage(msg) {
+ trace("Sending message to " + remoteName + " (ID " + remoteId + "):\n" + msg);
+ sendToPeer(remoteId, msg);
+}
+
+// TODO: Add callbacks onconnecting, onopen and onstatechange.
+
+
+// Server interaction
+
+function handleServerNotification(data) {
+ trace("Server notification: " + data);
+ var parsed = data.split(",");
+ if (parseInt(parsed[2]) == 1) { // New peer
+ var peerId = parseInt(parsed[1]);
+ if (!peerExists(peerId)) {
+ var peerList = document.getElementById("peers");
+ if (peerList.length == 1 && peerList.options[0].value == -1)
+ clearPeerList();
+ addPeer(peerId, parsed[0]);
+ document.getElementById("peers").disabled = false;
+ document.getElementById("call").disabled = false;
+ }
+ } else if (parseInt(parsed[2]) == 0) { // Removed peer
+ removePeer(parseInt(parsed[1]));
+ if (document.getElementById("peers").length == 0) {
+ document.getElementById("peers").disabled = true;
+ addPeer(-1, "No other peer connected");
+ }
+ }
+}
+
+function handlePeerMessage(peer_id, msg) {
+ var peerName = getPeerName(peer_id);
+ if (peerName == undefined) {
+ trace_warning("Received message from unknown peer (ID " + peer_id +
+ "), ignoring message:");
+ trace(msg);
+ return;
+ }
+ trace("Received message from " + peerName + " (ID " + peer_id + "):\n" + msg);
+ // Assuming we receive the message from the peer we want to communicate with.
+ // TODO: Only accept messages from peer we communicate with with if call is
+ // ongoing.
+ if (msg.search("BYE") == 0) {
+ // Other side has hung up.
+ document.getElementById("hangup").disabled = true;
+ closeCall()
+ } else {
+ if (!pc) {
+ // Other side is calling us, startup
+ if (!setSelectedPeer(peer_id)) {
+ trace_warning("Recevied message from unknown peer, ignoring");
+ return;
+ }
+ if (!storeRemoteInfo())
+ return;
+ document.getElementById("call").disabled = true;
+ document.getElementById("peers").disabled = true;
+ createPeerConnection();
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_exception(e, "Process signaling message error");
+ }
+ trace("Adding stream");
+ pc.addStream(localStream);
+ document.getElementById("hangup").disabled = false;
+ } else {
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_exception(e, "Process signaling message error");
+ }
+ }
+ }
+}
+
+function getIntHeader(r, name) {
+ var val = r.getResponseHeader(name);
+ trace("header value: " + val);
+ return val != null && val.length ? parseInt(val) : -1;
+}
+
+function hangingGetCallback() {
+ try {
+ if (hangingGet.readyState != 4 || disconnecting)
+ return;
+ if (hangingGet.status != 200) {
+ trace_warning("server error, status: " + hangingGet.status + ", text: " +
+ hangingGet.statusText);
+ disconnect();
+ } else {
+ var peer_id = getIntHeader(hangingGet, "Pragma");
+ if (peer_id == myId) {
+ handleServerNotification(hangingGet.responseText);
+ } else {
+ handlePeerMessage(peer_id, hangingGet.responseText);
+ }
+ }
+
+ if (hangingGet) {
+ hangingGet.abort();
+ hangingGet = null;
+ }
+
+ if (myId != -1)
+ window.setTimeout(startHangingGet, 0);
+ } catch (e) {
+ trace_exception(e, "Hanging get error");
+ }
+}
+
+function onHangingGetTimeout() {
+ trace("hanging get timeout. issuing again");
+ hangingGet.abort();
+ hangingGet = null;
+ if (myId != -1)
+ window.setTimeout(startHangingGet, 0);
+}
+
+function startHangingGet() {
+ try {
+ hangingGet = new XMLHttpRequest();
+ hangingGet.onreadystatechange = hangingGetCallback;
+ hangingGet.ontimeout = onHangingGetTimeout;
+ hangingGet.open("GET", server + "/wait?peer_id=" + myId, true);
+ hangingGet.send();
+ } catch (e) {
+ trace_exception(e, "Start hanging get error");
+ }
+}
+
+function sendToPeer(peer_id, data) {
+ if (myId == -1) {
+ alert("Not connected.");
+ return;
+ }
+ if (peer_id == myId) {
+ alert("Can't send a message to oneself.");
+ return;
+ }
+ var r = new XMLHttpRequest();
+ r.open("POST", server + "/message?peer_id=" + myId + "&to=" + peer_id, false);
+ r.setRequestHeader("Content-Type", "text/plain");
+ r.send(data);
+ r = null;
+}
+
+function signInCallback() {
+ try {
+ if (request.readyState == 4) {
+ if (request.status == 200) {
+ var peers = request.responseText.split("\n");
+ myId = parseInt(peers[0].split(",")[1]);
+ trace("My id: " + myId);
+ clearPeerList();
+ var added = 0;
+ for (var i = 1; i < peers.length; ++i) {
+ if (peers[i].length > 0) {
+ trace("Peer " + i + ": " + peers[i]);
+ var parsed = peers[i].split(",");
+ addPeer(parseInt(parsed[1]), parsed[0]);
+ ++added;
+ }
+ }
+ if (added == 0)
+ addPeer(-1, "No other peer connected");
+ else {
+ document.getElementById("peers").disabled = false;
+ document.getElementById("call").disabled = false;
+ }
+ startHangingGet();
+ request = null;
+ document.getElementById("connect").disabled = true;
+ document.getElementById("disconnect").disabled = false;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Sign in error");
+ document.getElementById("connect").disabled = false;
+ }
+}
+
+function signIn() {
+ try {
+ request = new XMLHttpRequest();
+ request.onreadystatechange = signInCallback;
+ request.open("GET", server + "/sign_in?" + myName, true);
+ request.send();
+ } catch (e) {
+ trace_exception(e, "Start sign in error");
+ document.getElementById("connect").disabled = false;
+ }
+}
+
+function connect() {
+ myName = document.getElementById("name").value.toLowerCase();
+ server = document.getElementById("server").value.toLowerCase();
+ if (myName.length == 0) {
+ alert("I need a name please.");
+ document.getElementById("local").focus();
+ } else {
+ // TODO: Disable connect button here, but we need a timeout and check if we
+ // have connected, if so enable it again.
+ signIn();
+ }
+}
+
+function disconnect() {
+ if (callState == 1)
+ hangUp();
+
+ disconnecting = true;
+
+ if (request) {
+ request.abort();
+ request = null;
+ }
+
+ if (hangingGet) {
+ hangingGet.abort();
+ hangingGet = null;
+ }
+
+ if (myId != -1) {
+ request = new XMLHttpRequest();
+ request.open("GET", server + "/sign_out?peer_id=" + myId, false);
+ request.send();
+ request = null;
+ myId = -1;
+ }
+
+ clearPeerList();
+ addPeer(-1, "Not connected");
+ document.getElementById("connect").disabled = false;
+ document.getElementById("disconnect").disabled = true;
+ document.getElementById("peers").disabled = true;
+ document.getElementById("call").disabled = true;
+
+ disconnecting = false;
+}
+
+
+// Window event handling
+
+window.onload = function() {
+ if (navigator.webkitGetUserMedia) {
+ document.getElementById('testApp').hidden = false;
+ setElementValuesFromURL();
+ getUserMedia();
+ } else {
+ document.getElementById('errorText').hidden = false;
+ }
+}
+
+window.onbeforeunload = disconnect;
+
+</script>
+</head>
+
+<body>
+<h1>WebRTC</h1>
+<section id="errorText" hidden="true">
+Could not detect WebRTC support.<p>
+You must have WebRTC enabled Chrome browser to use this test page. The browser
+must be started with the --enable-media-stream command line flag. For more
+information, please see
+<a href="https://sites.google.com/site/webrtc/blog/webrtcnowavailableinthechromedevchannel">
+this blog post</a>.
+</section>
+
+<section id="testApp" hidden="true">
+<table border="0">
+<tr>
+ <td>Local Preview</td>
+ <td>Remote Video</td>
+</tr>
+<tr>
+ <td>
+ <video width="320" height="240" id="localView" autoplay="autoplay"></video>
+ </td>
+ <td>
+ <video width="640" height="480" id="remoteView" autoplay="autoplay"></video>
+ </td>
+</tr>
+</table>
+
+<table border="0">
+<tr>
+ <td valign="top">
+ <table border="0" cellpaddning="0" cellspacing="0">
+ <tr>
+ <td>Server:</td>
+ <td>
+ <input type="text" id="server" size="30" value="http://localhost:8888"/>
+ </td>
+ </tr>
+ <tr>
+ <td>Name:</td><td><input type="text" id="name" size="30" value="name"/></td>
+ </tr>
+ </table>
+ </td>
+ <td valign="top">
+ <button id="connect" onclick="connect();">Connect</button><br>
+ <button id="disconnect" onclick="disconnect();" disabled="true">Disconnect
+ </button>
+ </td>
+ <td> </td>
+ <td valign="top">
+ Connected peers:<br>
+ <select id="peers" size="5" disabled="true">
+ <option value="-1">Not connected</option>
+ </select>
+ </td>
+ <td valign="top">
+ <!--input type="text" id="peer_id" size="3" value="1"/><br-->
+ <button id="call" onclick="doCall();" disabled="true">Call</button><br>
+ <button id="hangup" onclick="hangUp();" disabled="true">Hang up</button><br>
+ </td>
+</tr>
+</table>
+
+<button onclick="document.getElementById('debug').innerHTML='';">Clear log
+</button>
+<pre id="debug"></pre>
+</section>
+
+</body>
+
+</html>
+
diff --git a/trunk/test/metrics.gyp b/trunk/test/metrics.gyp
new file mode 100644
index 0000000..70483f9
--- /dev/null
+++ b/trunk/test/metrics.gyp
@@ -0,0 +1,46 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../src/build/common.gypi',
+ ],
+ 'targets': [
+ {
+ # The metrics code must be kept in its own GYP file in order to
+ # avoid a circular dependency error due to the dependency on libyuv.
+ # If the code would be put in test.gyp a circular dependency error during
+ # GYP generation would occur, because the libyuv.gypi unittest target
+ # depends on test_support_main. See issue #160 for more info.
+ 'target_name': 'metrics',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+ 'include_dirs': [
+ '.',
+ ],
+ 'sources': [
+ 'testsupport/metrics/video_metrics.h',
+ 'testsupport/metrics/video_metrics.cc',
+ ],
+ },
+ {
+ 'target_name': 'metrics_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'metrics',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'testsupport/metrics/video_metrics_unittest.cc',
+ ],
+ },
+ ],
+}
diff --git a/trunk/test/run_all_unittests.cc b/trunk/test/run_all_unittests.cc
new file mode 100644
index 0000000..0cdf0cd
--- /dev/null
+++ b/trunk/test/run_all_unittests.cc
@@ -0,0 +1,16 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_suite.h"
+
+int main(int argc, char** argv) {
+ webrtc::test::TestSuite test_suite(argc, argv);
+ return test_suite.Run();
+}
diff --git a/trunk/test/sanity_check/README b/trunk/test/sanity_check/README
new file mode 100644
index 0000000..7b14aeb
--- /dev/null
+++ b/trunk/test/sanity_check/README
@@ -0,0 +1,13 @@
+This test will pop up a browser with two tabs, and will run video from
+one tab to the other tab, across the local network interface.
+
+In order to run this test, you need to have:
+
+- a WebRTC Chrome binary in
+ $PROJECT_ROOT/../../chromium/src/out/Debug/chrome
+- an installed lighttpd in your $PATH
+- a built peerconnection_server (make peerconnection_server)
+
+You need to run the script from the test/sanity_check directory.
+
+The first time you run it, you will be asked to choose a search engine :-(
diff --git a/trunk/test/sanity_check/lighttpd.conf b/trunk/test/sanity_check/lighttpd.conf
new file mode 100644
index 0000000..4dae92d
--- /dev/null
+++ b/trunk/test/sanity_check/lighttpd.conf
@@ -0,0 +1,5 @@
+server.document-root = "./www/html/"
+server.port = 3000
+mimetype.assign = (
+ ".html" => "text/html"
+)
diff --git a/trunk/test/sanity_check/run_sanity_check b/trunk/test/sanity_check/run_sanity_check
new file mode 100644
index 0000000..cf38e98
--- /dev/null
+++ b/trunk/test/sanity_check/run_sanity_check
@@ -0,0 +1,72 @@
+#!/bin/bash
+#
+# Run a test with the WebRTC Chromium build.
+# Should work on any machine with a camera.
+#
+# Method:
+# - Start server
+# - Start 2 browser tabs
+# - Browser tab 1 captures camera
+# - Both browsers sign in
+# - Browser 1 calls browser 2
+# - Browser 2 displays camera feed from browser 1
+#
+
+# Feel free to tweak this locally if your chrome build is somewhere else.
+# The default assumes that it is in a folder called chromium two levels
+# up from the project root ('trunk').
+
+set -e
+
+if [ ! -x run_sanity_check ]; then
+ echo "Error: This script must run from its own directory"
+ exit 1
+fi
+
+URLBASE=127.0.0.1:3000
+CALLER=$URLBASE/call_client.html
+CALLEE=$URLBASE/call_responder.html
+FLAGS="--enable-media-stream --enable-p2papi"
+PROJECT_ROOT=../..
+CHROME_BINARY=$PROJECT_ROOT/../../chromium/src/out/Debug/chrome
+
+if which lighttpd; then
+ LOCAL_WEB_SERVER_BINARY=$(which lighttpd)
+else
+ echo "Error: You must install lighttpd first (sudo apt-get install lighttpd)"
+ exit 1
+fi
+
+SERVER_BINARY=${PROJECT_ROOT}/out/Debug/peerconnection_server
+if [ ! -e "$SERVER_BINARY" ]; then
+ echo "Error: You must build peerconnection_server first."
+ exit 1
+fi
+
+CHROME_BINARY=$PROJECT_ROOT/../../chromium/src/out/Debug/chrome
+if [ ! -e "$CHROME_BINARY" ]; then
+ echo "Error: You must build chrome (could not open $CHROME_BINARY)."
+ exit 1
+fi
+
+# Launch the web server and make it serve the local www/html directory
+${LOCAL_WEB_SERVER_BINARY} -D -f lighttpd.conf &
+LOCAL_WEB_SERVER=$!
+
+${SERVER_BINARY} &
+SERVER=$!
+echo "Started server as $SERVER"
+
+# We can make 2 browsers
+#${CHROME_BINARY} $CALLER $FLAGS --user-data-dir=temp/user1 &
+#USER1=$!
+#echo "Started first user as $USER1"
+#${CHROME_BINARY} $CALLEE $FLAGS --user-data-dir=temp/user2
+
+# But it also works with separate tabs in one browser.
+${CHROME_BINARY} $CALLER $CALLEE $FLAGS --user-data-dir=temp/user1
+
+echo "Test finished, cleaning up"
+
+kill $SERVER || echo "No server"
+kill $LOCAL_WEB_SERVER || echo "No local web server"
diff --git a/trunk/test/sanity_check/www/html/call_client.html b/trunk/test/sanity_check/www/html/call_client.html
new file mode 100644
index 0000000..789bc9f
--- /dev/null
+++ b/trunk/test/sanity_check/www/html/call_client.html
@@ -0,0 +1,394 @@
+<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
+
+<html>
+
+<head>
+<title>WebRTC Autocaller</title>
+<script type="text/javascript">
+
+var local_name;
+var server;
+var my_id = -1;
+var other_peers = {};
+var request = null;
+var hanging_get = null;
+var pc = null;
+var local_stream = null;
+var call_state = 0;
+// 0 - Not started
+// 1 - Call ongoing
+// 2 - We have initiated closing the call
+// 3 - Other side has initiated closing the call
+
+// General
+function toggleExtraButtons() {
+ document.getElementById("createPcBtn").hidden =
+ !document.getElementById("createPcBtn").hidden;
+ document.getElementById("addStreamBtn").hidden =
+ !document.getElementById("addStreamBtn").hidden;
+}
+
+function trace(txt) {
+ var elem = document.getElementById("debug");
+ elem.innerHTML += txt + "<br>";
+}
+
+function trace_warning(txt) {
+ var wtxt = "<b>" + txt + "</b>";
+ trace(wtxt);
+}
+
+function setCallState(state) {
+ trace("Changing call state: " + call_state + " -> " + state);
+ call_state = state;
+}
+
+function checkPeerConnection() {
+ if (!pc) {
+ trace_warning("No PeerConnection object exists.");
+ return 0;
+ }
+ return 1;
+}
+
+// Local stream generation
+function gotStream(s) {
+ var url = webkitURL.createObjectURL(s);
+ document.getElementById("localView").src = url;
+ trace("User has granted access to webcam. url = " + url);
+ local_stream = s;
+}
+
+function gotStreamFailed(error) {
+ alert("Failed to get access to webcam. Error code was " + error.code);
+ trace("Failed to get access to webcam. Error code was " + error.code);
+}
+
+// Call control
+function createPeerConnection() {
+ if (pc) {
+ trace_warning("PeerConnection object already exists.");
+ }
+ trace("Creating PeerConnection object.");
+ // CRASH TEST
+ pc = new webkitPeerConnection("STUN 1.2.3.4:43", onSignalingMessage);
+ pc.onaddstream = onAddStream;
+ pc.onremovestream = onRemoveStream;
+}
+
+function startCall() {
+ document.getElementById("startcall").disabled = true;
+ createPeerConnection();
+ addStream();
+}
+
+function addStream() {
+ trace("Starting call. [pc.addStream(local_stream)]");
+ pc.addStream(local_stream);
+ document.getElementById("stopcall").disabled = false;
+}
+
+function stopCall() {
+ document.getElementById("stopcall").disabled = true;
+ trace("Stopping call [pc.close()]");
+ setCallState(2);
+ pc.close();
+}
+
+function test1() {
+ addStream();
+}
+
+function callHasStarted() {
+ setCallState(1);
+}
+
+// PeerConnection callbacks
+function onAddStream(e) {
+ var stream = pc.remoteStreams[0];
+ var url = webkitURL.createObjectURL(stream);
+ document.getElementById("remoteView").src = url;
+ trace("Started showing remote stream. url = " + url);
+ setTimeout(callHasStarted, 5000); // TODO(hta): workaround, to be removed
+}
+
+function onRemoveStream(e) {
+ // Currently if we get this callback, call has ended.
+ document.getElementById("remoteView").src = "";
+ trace("Stopped showing remote stream.");
+ if (call_state == 2) {
+ // Finalize close call. We assume here that this callback comes
+ // after onSignalingMessage.
+ pc = null;
+ setCallState(0);
+ document.getElementById("startcall").disabled = false;
+ }
+}
+
+function onSignalingMessage(msg) {
+ var peer_id = parseInt(document.getElementById("peer_id").value);
+ trace("Sending message to " + other_peers[peer_id] +
+ " (ID " + peer_id + "):\n" + msg);
+ sendToPeer(peer_id, msg);
+}
+
+// Server interaction
+
+function handleServerNotification(data) {
+ trace("Server notification: " + data);
+ notePeerAndMaybeCall(data);
+}
+
+function handlePeerMessage(peer_id, msg) {
+ trace("Received message from " + other_peers[peer_id] +
+ " (ID " + peer_id + "):\n" + msg);
+ // Assuming we have created a PeerConnection and that we receive the
+ // message from the peer want to communicate with
+ document.getElementById("peer_id").value =
+ peer_id; // Ensure we reply to the same peer
+
+ if (pc) {
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_warning("Signaling message error: " + e.description);
+ }
+ if (call_state == 1)
+ setCallState(3);
+ } else {
+ createPeerConnection();
+ pc.processSignalingMessage(msg);
+ }
+}
+
+function getIntHeader(r, name) {
+ var val = r.getResponseHeader(name);
+ trace("header value: " + val);
+ return val != null && val.length ? parseInt(val) : -1;
+}
+
+function hangingGetCallback() {
+ try {
+ if (hanging_get.readyState != 4)
+ return;
+ if (hanging_get.status != 200) {
+ trace("server error, status: " + hanging_get.status + ", text: " +
+ hanging_get.statusText);
+ disconnect();
+ } else {
+ var peer_id = getIntHeader(hanging_get, "Pragma");
+ if (peer_id == my_id) {
+ handleServerNotification(hanging_get.responseText);
+ } else {
+ handlePeerMessage(peer_id, hanging_get.responseText);
+ }
+ }
+
+ if (hanging_get) {
+ hanging_get.abort();
+ hanging_get = null;
+ }
+
+ if (my_id != -1)
+ window.setTimeout(startHangingGet, 0);
+ } catch (e) {
+ trace("Hanging get error: " + e.description);
+ }
+}
+
+function onHangingGetTimeout() {
+ trace("hanging get timeout. issuing again.");
+ hanging_get.abort();
+ hanging_get = null;
+ if (my_id != -1)
+ window.setTimeout(startHangingGet, 0);
+}
+
+function startHangingGet() {
+ try {
+ hanging_get = new XMLHttpRequest();
+ hanging_get.onreadystatechange = hangingGetCallback;
+ hanging_get.ontimeout = onHangingGetTimeout;
+ hanging_get.open("GET", server + "/wait?peer_id=" + my_id, true);
+ hanging_get.send();
+ } catch (e) {
+ trace("error" + e.description);
+ }
+}
+
+function sendToPeer(peer_id, data) {
+ if (my_id == -1) {
+ alert("Not connected");
+ return;
+ }
+ if (peer_id == my_id) {
+ alert("Can't send a message to oneself :)");
+ return;
+ }
+ var r = new XMLHttpRequest();
+ r.open("POST", server + "/message?peer_id=" + my_id + "&to=" +
+ peer_id, false);
+ r.setRequestHeader("Content-Type", "text/plain");
+ r.send(data);
+ r = null;
+}
+
+function notePeerAndMaybeCall(peerInfo) {
+ var parsed = peerInfo.split(",");
+ if (parseInt(parsed[2]) != 0) {
+ trace("Peer came on: " + peerInfo);
+ other_peers[parseInt(parsed[1])] = parsed[0];
+ // If we're not already in a call, call it.
+ if (!document.getElementById("startcall").disabled) {
+ document.getElementById("peer_id").value = parsed[1];
+ startCall();
+ }
+ } else {
+ trace("Peer went away: " + peerInfo);
+ other_peers[parseInt(parsed[1])] = null;
+ }
+}
+
+function signInCallback() {
+ try {
+ if (request.readyState == 4) {
+ if (request.status == 200) {
+ var peers = request.responseText.split("\n");
+ my_id = parseInt(peers[0].split(",")[1]);
+ trace("My id: " + my_id);
+ for (var i = 1; i < peers.length; ++i) {
+ if (peers[i].length > 0) {
+ trace("Peer " + i + ": " + peers[i]);
+ notePeerAndMaybeCall(peers[i]);
+ }
+ }
+ startHangingGet();
+ request = null;
+ }
+ }
+ } catch (e) {
+ trace("error: " + e.description);
+ }
+}
+
+function signIn() {
+ try {
+ request = new XMLHttpRequest();
+ request.onreadystatechange = signInCallback;
+ request.open("GET", server + "/sign_in?" + local_name, true);
+ request.send();
+ } catch (e) {
+ trace("error: " + e.description);
+ }
+}
+
+function connect() {
+ local_name = document.getElementById("local").value.toLowerCase();
+ server = document.getElementById("server").value.toLowerCase();
+ if (local_name.length == 0) {
+ alert("I need a name please.");
+ document.getElementById("local").focus();
+ } else {
+ document.getElementById("connect").disabled = true;
+ document.getElementById("disconnect").disabled = false;
+ trace("Connecting to server");
+ signIn();
+ }
+}
+
+function disconnect() {
+ if (request) {
+ request.abort();
+ request = null;
+ }
+
+ if (hanging_get) {
+ hanging_get.abort();
+ hanging_get = null;
+ }
+
+ if (my_id != -1) {
+ request = new XMLHttpRequest();
+ request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
+ request.send();
+ request = null;
+ my_id = -1;
+ }
+
+ document.getElementById("connect").disabled = false;
+ document.getElementById("disconnect").disabled = true;
+}
+
+function toggleLog()
+{
+ document.getElementById('debug').hidden =!
+ document.getElementById('debug').hidden;
+}
+
+window.onbeforeunload = disconnect;
+
+// TODO: Add audio here when available. Note: now it's hard coded,
+// audio will also be started.
+navigator.webkitGetUserMedia("video", gotStream, gotStreamFailed);
+
+</script>
+</head>
+
+<body>
+<h1>WebRTC autocaller</h1>
+This page should show your local camera, and connect to a remote browser.
+<table border="0">
+
+<tr>
+<th align="left">Local Preview</th>
+<th align="left">Remote Video</th>
+</tr>
+<tr>
+<td><video width="320" height="240" id="localView" autoplay="autoplay">
+ </video></td>
+<td><video width="640" height="480" id="remoteView" autoplay="autoplay">
+ </video></td>
+</tr>
+<tr>
+<td valign="top">
+<form>
+Target peer ID: <input type="text" id="peer_id" size="3" value="1"/><br>
+<button id="startcall" onclick="startCall();">(4) Start call</button><br>
+<button id="stopcall" onclick="stopCall();" disabled="true">(5) Stop call
+</button><br>
+<!--<input type="button" value="Test 1" onclick="test1()"/><br>-->
+</form>
+</td>
+<td valign="top">
+<button onclick="toggleExtraButtons();">Toggle extra buttons</button><br>
+<button id="createPcBtn" onclick="createPeerConnection();" hidden="true">
+ Create peer connection</button><br>
+<button id="addStreamBtn" onclick="addStream();" hidden="true">
+ Add stream</button><br>
+</td>
+</tr>
+
+<tr><td colspan="2">
+(1) Server: <input type="text" id="server" value="http://localhost:8888" />
+<br>
+(2) Your name: <input type="text" id="local" value="caller"/>
+<button id="connect" onclick="connect();" disabled="true">(3) Connect</button>
+<button id="disconnect" onclick="disconnect();" disabled="true">(6) Disconnect
+</button>
+<br>
+ <br>
+<button onclick="toggleLog()">Toggle view log</button>
+<button onclick="document.getElementById('debug').innerHTML='';">
+ Clear log</button>
+<pre id="debug">
+</pre>
+</td></tr>
+
+</table>
+</body>
+<script>
+// On init, we connect to the server.
+connect();
+</script>
+
+</html>
diff --git a/trunk/test/sanity_check/www/html/call_responder.html b/trunk/test/sanity_check/www/html/call_responder.html
new file mode 100644
index 0000000..4d7f3b4
--- /dev/null
+++ b/trunk/test/sanity_check/www/html/call_responder.html
@@ -0,0 +1,373 @@
+<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
+
+<html>
+
+<head>
+<title>WebRTC Autoresponder</title>
+This page should show a stream coming from a different browser.
+<script type="text/javascript">
+
+var local_name;
+var server;
+var my_id = -1;
+var other_peers = {};
+var request = null;
+var hanging_get = null;
+var pc = null;
+var local_stream = null;
+var call_state = 0;
+// 0 - Not started
+// 1 - Call ongoing
+// 2 - We have initiated closing the call
+// 3 - Other side has initiated closing the call
+
+// General
+function toggleExtraButtons() {
+ document.getElementById("createPcBtn").hidden =
+ !document.getElementById("createPcBtn").hidden;
+ document.getElementById("addStreamBtn").hidden =
+ !document.getElementById("addStreamBtn").hidden;
+}
+
+function trace(txt) {
+ var elem = document.getElementById("debug");
+ elem.innerHTML += txt + "<br>";
+}
+
+function trace_warning(txt) {
+ var wtxt = "<b>" + txt + "</b>";
+ trace(wtxt);
+}
+
+function setCallState(state) {
+ trace("Changing call state: " + call_state + " -> " + state);
+ call_state = state;
+}
+
+function checkPeerConnection() {
+ if (!pc) {
+ trace_warning("No PeerConnection object exists.");
+ return 0;
+ }
+ return 1;
+}
+
+// Local stream generation
+function gotStream(s) {
+ var url = webkitURL.createObjectURL(s);
+ document.getElementById("localView").src = url;
+ trace("User has granted access to webcam. url = " + url);
+ local_stream = s;
+}
+
+function gotStreamFailed(error) {
+ alert("Failed to get access to webcam. Error code was " + error.code);
+ trace("Failed to get access to webcam. Error code was " + error.code);
+}
+
+// Call control
+function createPeerConnection() {
+ if (pc) {
+ trace_warning("PeerConnection object already exists.");
+ }
+ trace("Creating PeerConnection object.");
+ pc = new webkitPeerConnection("STUN stun.l.google.com:19302",
+ onSignalingMessage);
+ pc.onaddstream = onAddStream;
+ pc.onremovestream = onRemoveStream;
+}
+
+function startCall() {
+ document.getElementById("startcall").disabled = true;
+ createPeerConnection();
+ addStream();
+}
+
+function addStream()
+{
+ trace("Starting call. [pc.addStream(local_stream)]");
+ pc.addStream(local_stream);
+ document.getElementById("stopcall").disabled = false;
+}
+
+function stopCall() {
+ document.getElementById("stopcall").disabled = true;
+ trace("Stopping call [pc.close()]");
+ setCallState(2);
+ pc.close();
+}
+
+function test1() {
+ addStream();
+}
+
+function callHasStarted() {
+ setCallState(1);
+}
+
+// PeerConnection callbacks
+function onAddStream(e) {
+ var stream = pc.remoteStreams[0];
+ var url = webkitURL.createObjectURL(stream);
+ document.getElementById("remoteView").src = url;
+ trace("Started showing remote stream. url = " + url);
+ setTimeout(callHasStarted, 5000); // TODO: Temp workaround, to be removed
+}
+
+function onRemoveStream(e) {
+ // Currently if we get this callback, call has ended.
+ document.getElementById("remoteView").src = "";
+ trace("Stopped showing remote stream.");
+ if (call_state == 2) {
+ // Finalize close call. We assume here that this
+ // callback comes after onSignalingMessage.
+ pc = null;
+ setCallState(0);
+ document.getElementById("startcall").disabled = false;
+ }
+}
+
+function onSignalingMessage(msg) {
+ var peer_id = parseInt(document.getElementById("peer_id").value);
+ trace("Sending message to " + other_peers[peer_id] +
+ " (ID " + peer_id + "):\n" + msg);
+ sendToPeer(peer_id, msg);
+}
+
+// Server interaction
+
+function handleServerNotification(data) {
+ trace("Server notification: " + data);
+ var parsed = data.split(",");
+ if (parseInt(parsed[2]) != 0)
+ other_peers[parseInt(parsed[1])] = parsed[0];
+}
+
+function handlePeerMessage(peer_id, msg) {
+ trace("Received message from " + other_peers[peer_id] +
+ " (ID " + peer_id + "):\n" + msg);
+ // Assuming we have created a PeerConnection and that we receive the message
+ // from the peer we want to communicate with
+ document.getElementById("peer_id").value = peer_id;
+ if (pc) {
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_warning("Signaling message error: " + e.description);
+ }
+ if (call_state == 1)
+ setCallState(3);
+ } else {
+ createPeerConnection();
+ pc.processSignalingMessage(msg);
+ }
+}
+
+function getIntHeader(r, name) {
+ var val = r.getResponseHeader(name);
+ trace("header value: " + val);
+ return val != null && val.length ? parseInt(val) : -1;
+}
+
+function hangingGetCallback() {
+ try {
+ if (hanging_get.readyState != 4)
+ return;
+ if (hanging_get.status != 200) {
+ trace("server error, status: " + hanging_get.status +
+ ", text: " + hanging_get.statusText);
+ disconnect();
+ } else {
+ var peer_id = getIntHeader(hanging_get, "Pragma");
+ if (peer_id == my_id) {
+ handleServerNotification(hanging_get.responseText);
+ } else {
+ handlePeerMessage(peer_id, hanging_get.responseText);
+ }
+ }
+
+ if (hanging_get) {
+ hanging_get.abort();
+ hanging_get = null;
+ }
+
+ if (my_id != -1)
+ window.setTimeout(startHangingGet, 0);
+ } catch (e) {
+ trace("Hanging get error: " + e.description);
+ }
+}
+
+function onHangingGetTimeout() {
+ trace("hanging get timeout. issuing again.");
+ hanging_get.abort();
+ hanging_get = null;
+ if (my_id != -1)
+ window.setTimeout(startHangingGet, 0);
+}
+
+function startHangingGet() {
+ try {
+ hanging_get = new XMLHttpRequest();
+ hanging_get.onreadystatechange = hangingGetCallback;
+ hanging_get.ontimeout = onHangingGetTimeout;
+ hanging_get.open("GET", server + "/wait?peer_id=" + my_id, true);
+ hanging_get.send();
+ } catch (e) {
+ trace("error" + e.description);
+ }
+}
+
+function sendToPeer(peer_id, data) {
+ if (my_id == -1) {
+ alert("Not connected");
+ return;
+ }
+ if (peer_id == my_id) {
+ alert("Can't send a message to oneself :)");
+ return;
+ }
+ var r = new XMLHttpRequest();
+ r.open("POST", server + "/message?peer_id=" + my_id + "&to=" + peer_id,
+ false);
+ r.setRequestHeader("Content-Type", "text/plain");
+ r.send(data);
+ r = null;
+}
+
+function signInCallback() {
+ try {
+ if (request.readyState == 4) {
+ if (request.status == 200) {
+ var peers = request.responseText.split("\n");
+ my_id = parseInt(peers[0].split(",")[1]);
+ trace("My id: " + my_id);
+ for (var i = 1; i < peers.length; ++i) {
+ if (peers[i].length > 0) {
+ trace("Peer " + i + ": " + peers[i]);
+ var parsed = peers[i].split(",");
+ other_peers[parseInt(parsed[1])] = parsed[0];
+ }
+ }
+ startHangingGet();
+ request = null;
+ }
+ }
+ } catch (e) {
+ trace("error: " + e.description);
+ }
+}
+
+function signIn() {
+ try {
+ request = new XMLHttpRequest();
+ request.onreadystatechange = signInCallback;
+ request.open("GET", server + "/sign_in?" + local_name, true);
+ request.send();
+ } catch (e) {
+ trace("error: " + e.description);
+ }
+}
+
+function connect() {
+ local_name = document.getElementById("local").value.toLowerCase();
+ server = document.getElementById("server").value.toLowerCase();
+ if (local_name.length == 0) {
+ alert("I need a name please.");
+ document.getElementById("local").focus();
+ } else {
+ document.getElementById("connect").disabled = true;
+ document.getElementById("disconnect").disabled = false;
+ signIn();
+ }
+}
+
+function disconnect() {
+ if (request) {
+ request.abort();
+ request = null;
+ }
+
+ if (hanging_get) {
+ hanging_get.abort();
+ hanging_get = null;
+ }
+
+ if (my_id != -1) {
+ request = new XMLHttpRequest();
+ request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
+ request.send();
+ request = null;
+ my_id = -1;
+ }
+
+ document.getElementById("connect").disabled = false;
+ document.getElementById("disconnect").disabled = true;
+}
+
+function toggleLog()
+{
+ document.getElementById('debug').hidden =!
+ document.getElementById('debug').hidden;
+}
+
+window.onbeforeunload = disconnect;
+
+</script>
+</head>
+
+<body>
+<table border="0">
+
+<tr>
+<th align="left">Local Preview</th>
+<th align="left">Remote Video</th>
+</tr>
+<tr>
+<td><video width="320" height="240" id="localView" autoplay="autoplay">
+ </video></td>
+<td><video width="640" height="480" id="remoteView" autoplay="autoplay">
+ </video></td>
+</tr>
+<tr>
+<td valign="top">
+<form>
+Target peer ID: <input type="text" id="peer_id" size="3" value="1"/><br>
+<button id="startcall" onclick="startCall();">(4) Start call</button><br>
+<button id="stopcall" onclick="stopCall();" disabled="true">(5) Stop call
+ </button><br>
+<!--<input type="button" value="Test 1" onclick="test1()"/><br>-->
+</form>
+</td>
+<td valign="top">
+<button onclick="toggleExtraButtons();">Toggle extra buttons</button><br>
+<button id="createPcBtn" onclick="createPeerConnection();" hidden="true">
+ Create peer connection</button><br>
+<button id="addStreamBtn" onclick="addStream();" hidden="true">Add stream
+ </button><br>
+</td>
+</tr>
+
+<tr><td colspan="2">
+(1) Server: <input type="text" id="server" value="http://localhost:8888" />
+<br>
+(2) Your name: <input type="text" id="local" value="responder"/>
+<button id="connect" onclick="connect();" disabled="true">(3) Connect</button>
+<button id="disconnect" onclick="disconnect();" disabled="true">(6) Disconnect
+ </button>
+<br>
+ <br>
+<button onclick="toggleLog()">Toggle view log</button>
+<button onclick="document.getElementById('debug').innerHTML='';">Clear log
+ </button>
+<pre id="debug">
+</pre>
+</td></tr>
+
+</table>
+</body>
+<script>
+// On init, we connect to the server.
+connect();
+</script>
+</html>
diff --git a/trunk/test/test.gyp b/trunk/test/test.gyp
new file mode 100644
index 0000000..b428bf3
--- /dev/null
+++ b/trunk/test/test.gyp
@@ -0,0 +1,79 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# TODO(andrew): consider moving test_support to src/base/test.
+{
+ 'includes': [
+ '../src/build/common.gypi',
+ ],
+ 'targets': [
+ {
+ 'target_name': 'test_support',
+ 'type': 'static_library',
+ 'include_dirs': [
+ '.',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '.', # Some includes are hierarchical
+ ],
+ },
+ 'dependencies': [
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ ],
+ 'all_dependent_settings': {
+ 'include_dirs': [
+ '.',
+ ],
+ },
+ 'sources': [
+ 'test_suite.cc',
+ 'test_suite.h',
+ 'testsupport/fileutils.h',
+ 'testsupport/fileutils.cc',
+ 'testsupport/frame_reader.h',
+ 'testsupport/frame_reader.cc',
+ 'testsupport/frame_writer.h',
+ 'testsupport/frame_writer.cc',
+ 'testsupport/gtest_prod_util.h',
+ 'testsupport/packet_reader.h',
+ 'testsupport/packet_reader.cc',
+ 'testsupport/mock/mock_frame_reader.h',
+ 'testsupport/mock/mock_frame_writer.h',
+ ],
+ },
+ {
+ # Depend on this target when you want to have test_support but also the
+ # main method needed for gtest to execute!
+ 'target_name': 'test_support_main',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'test_support',
+ ],
+ 'sources': [
+ 'run_all_unittests.cc',
+ ],
+ },
+ {
+ 'target_name': 'test_support_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'testsupport/unittest_utils.h',
+ 'testsupport/fileutils_unittest.cc',
+ 'testsupport/frame_reader_unittest.cc',
+ 'testsupport/frame_writer_unittest.cc',
+ 'testsupport/packet_reader_unittest.cc',
+ ],
+ },
+ ],
+}
diff --git a/trunk/test/test_suite.cc b/trunk/test/test_suite.cc
new file mode 100644
index 0000000..ac3f3a2
--- /dev/null
+++ b/trunk/test/test_suite.cc
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_suite.h"
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+
+namespace webrtc {
+namespace test {
+TestSuite::TestSuite(int argc, char** argv) {
+ testing::InitGoogleMock(&argc, argv); // Runs InitGoogleTest() internally.
+}
+
+TestSuite::~TestSuite() {
+}
+
+int TestSuite::Run() {
+ Initialize();
+ int result = RUN_ALL_TESTS();
+ Shutdown();
+ return result;
+}
+
+void TestSuite::Initialize() {
+ // TODO(andrew): initialize singletons here (e.g. Trace).
+}
+
+void TestSuite::Shutdown() {
+}
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/test_suite.h b/trunk/test/test_suite.h
new file mode 100644
index 0000000..f500daa
--- /dev/null
+++ b/trunk/test/test_suite.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TEST_SUITE_H_
+#define TEST_TEST_SUITE_H_
+
+// Derived from Chromium's src/base/test/test_suite.h.
+
+// Defines a basic test suite framework for running gtest based tests. You can
+// instantiate this class in your main function and call its Run method to run
+// any gtest based tests that are linked into your executable.
+
+#include "src/system_wrappers/interface/constructor_magic.h"
+
+namespace webrtc {
+namespace test {
+class TestSuite {
+ public:
+ TestSuite(int argc, char** argv);
+ virtual ~TestSuite();
+
+ int Run();
+
+ protected:
+ // Override these for custom initialization and shutdown handling. Use these
+ // instead of putting complex code in your constructor/destructor.
+ virtual void Initialize();
+ virtual void Shutdown();
+
+ DISALLOW_COPY_AND_ASSIGN(TestSuite);
+};
+} // namespace test
+} // namespace webrtc
+
+#endif // TEST_TEST_SUITE_H_
diff --git a/trunk/test/testsupport/fileutils.cc b/trunk/test/testsupport/fileutils.cc
new file mode 100644
index 0000000..1e6bbca
--- /dev/null
+++ b/trunk/test/testsupport/fileutils.cc
@@ -0,0 +1,167 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/fileutils.h"
+
+#ifdef WIN32
+#include <direct.h>
+#define GET_CURRENT_DIR _getcwd
+#else
+#include <unistd.h>
+#define GET_CURRENT_DIR getcwd
+#endif
+
+#include <sys/stat.h> // To check for directory existence.
+#ifndef S_ISDIR // Not defined in stat.h on Windows.
+#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
+#endif
+
+#include <cstdio>
+
+#include "typedefs.h" // For architecture defines
+
+namespace webrtc {
+namespace test {
+
+#ifdef WIN32
+static const char* kPathDelimiter = "\\";
+#else
+static const char* kPathDelimiter = "/";
+#endif
+// The file we're looking for to identify the project root dir.
+static const char* kProjectRootFileName = "DEPS";
+static const char* kOutputDirName = "out";
+static const char* kFallbackPath = "./";
+static const char* kResourcesDirName = "resources";
+const char* kCannotFindProjectRootDir = "ERROR_CANNOT_FIND_PROJECT_ROOT_DIR";
+
+std::string ProjectRootPath() {
+ std::string working_dir = WorkingDir();
+ if (working_dir == kFallbackPath) {
+ return kCannotFindProjectRootDir;
+ }
+ // Check for our file that verifies the root dir.
+ std::string current_path(working_dir);
+ FILE* file = NULL;
+ int path_delimiter_index = current_path.find_last_of(kPathDelimiter);
+ while (path_delimiter_index > -1) {
+ std::string root_filename = current_path + kPathDelimiter +
+ kProjectRootFileName;
+ file = fopen(root_filename.c_str(), "r");
+ if (file != NULL) {
+ fclose(file);
+ return current_path + kPathDelimiter;
+ }
+ // Move up one directory in the directory tree.
+ current_path = current_path.substr(0, path_delimiter_index);
+ path_delimiter_index = current_path.find_last_of(kPathDelimiter);
+ }
+ // Reached the root directory.
+ fprintf(stderr, "Cannot find project root directory!\n");
+ return kCannotFindProjectRootDir;
+}
+
+std::string OutputPath() {
+ std::string path = ProjectRootPath();
+ if (path == kCannotFindProjectRootDir) {
+ return kFallbackPath;
+ }
+ path += kOutputDirName;
+ if (!CreateDirectory(path)) {
+ return kFallbackPath;
+ }
+ return path + kPathDelimiter;
+}
+
+std::string WorkingDir() {
+ char path_buffer[FILENAME_MAX];
+ if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
+ fprintf(stderr, "Cannot get current directory!\n");
+ return kFallbackPath;
+ } else {
+ return std::string(path_buffer);
+ }
+}
+
+bool CreateDirectory(std::string directory_name) {
+ struct stat path_info = {0};
+ // Check if the path exists already:
+ if (stat(directory_name.c_str(), &path_info) == 0) {
+ if (!S_ISDIR(path_info.st_mode)) {
+ fprintf(stderr, "Path %s exists but is not a directory! Remove this "
+ "file and re-run to create the directory.\n",
+ directory_name.c_str());
+ return false;
+ }
+ } else {
+#ifdef WIN32
+ return _mkdir(directory_name.c_str()) == 0;
+#else
+ return mkdir(directory_name.c_str(), S_IRWXU | S_IRWXG | S_IRWXO) == 0;
+#endif
+ }
+ return true;
+}
+
+bool FileExists(std::string file_name) {
+ struct stat file_info = {0};
+ return stat(file_name.c_str(), &file_info) == 0;
+}
+
+std::string ResourcePath(std::string name, std::string extension) {
+ std::string platform = "win";
+#ifdef WEBRTC_LINUX
+ platform = "linux";
+#endif // WEBRTC_LINUX
+#ifdef WEBRTC_MAC
+ platform = "mac";
+#endif // WEBRTC_MAC
+
+#ifdef WEBRTC_ARCH_64_BITS
+ std::string architecture = "64";
+#else
+ std::string architecture = "32";
+#endif // WEBRTC_ARCH_64_BITS
+
+ std::string resources_path = ProjectRootPath() + kResourcesDirName +
+ kPathDelimiter;
+ std::string resource_file = resources_path + name + "_" + platform + "_" +
+ architecture + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Try without architecture.
+ resource_file = resources_path + name + "_" + platform + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Try without platform.
+ resource_file = resources_path + name + "_" + architecture + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Fall back on name without architecture or platform.
+ return resources_path + name + "." + extension;
+}
+
+size_t GetFileSize(std::string filename) {
+ FILE* f = fopen(filename.c_str(), "rb");
+ size_t size = 0;
+ if (f != NULL) {
+ if (fseek(f, 0, SEEK_END) == 0) {
+ size = ftell(f);
+ }
+ fclose(f);
+ }
+ return size;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/fileutils.h b/trunk/test/testsupport/fileutils.h
new file mode 100644
index 0000000..c89ac29
--- /dev/null
+++ b/trunk/test/testsupport/fileutils.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+
+// File utilities for testing purposes.
+//
+// The ProjectRootPath() method is a convenient way of getting an absolute
+// path to the project source tree root directory. Using this, it is easy to
+// refer to test resource files in a portable way.
+//
+// Notice that even if Windows platforms use backslash as path delimiter, it is
+// also supported to use slash, so there's no need for #ifdef checks in test
+// code for setting up the paths to the resource files.
+//
+// Example use:
+// Assume we have the following code being used in a test source file:
+// const std::string kInputFile = webrtc::test::ProjectRootPath() +
+// "test/data/voice_engine/audio_long16.wav";
+// // Use the kInputFile for the tests...
+//
+// Then here's some example outputs for different platforms:
+// Linux:
+// * Source tree located in /home/user/webrtc/trunk
+// * Test project located in /home/user/webrtc/trunk/src/testproject
+// * Test binary compiled as:
+// /home/user/webrtc/trunk/out/Debug/testproject_unittests
+// Then ProjectRootPath() will return /home/user/webrtc/trunk/ no matter if
+// the test binary is executed from standing in either of:
+// /home/user/webrtc/trunk
+// or
+// /home/user/webrtc/trunk/out/Debug
+// (or any other directory below the trunk for that matter).
+//
+// Windows:
+// * Source tree located in C:\Users\user\webrtc\trunk
+// * Test project located in C:\Users\user\webrtc\trunk\src\testproject
+// * Test binary compiled as:
+// C:\Users\user\webrtc\trunk\src\testproject\Debug\testproject_unittests.exe
+// Then ProjectRootPath() will return C:\Users\user\webrtc\trunk\ when the
+// test binary is executed from inside Visual Studio.
+// It will also return the same path if the test is executed from a command
+// prompt standing in C:\Users\user\webrtc\trunk\src\testproject\Debug
+//
+// Mac:
+// * Source tree located in /Users/user/webrtc/trunk
+// * Test project located in /Users/user/webrtc/trunk/src/testproject
+// * Test binary compiled as:
+// /Users/user/webrtc/trunk/xcodebuild/Debug/testproject_unittests
+// Then ProjectRootPath() will return /Users/user/webrtc/trunk/ no matter if
+// the test binary is executed from standing in either of:
+// /Users/user/webrtc/trunk
+// or
+// /Users/user/webrtc/trunk/out/Debug
+// (or any other directory below the trunk for that matter).
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
+#define WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
+
+#include <string>
+
+namespace webrtc {
+namespace test {
+
+// This is the "directory" returned if the ProjectPath() function fails
+// to find the project root.
+extern const char* kCannotFindProjectRootDir;
+
+// Finds the root dir of the project, to be able to set correct paths to
+// resource files used by tests.
+// The implementation is simple: it just looks for the file defined by
+// kProjectRootFileName, starting in the current directory (the working
+// directory) and then steps upward until it is found (or it is at the root of
+// the file system).
+// If the current working directory is above the project root dir, it will not
+// be found.
+//
+// If symbolic links occur in the path they will be resolved and the actual
+// directory will be returned.
+//
+// Returns the absolute path to the project root dir (usually the trunk dir)
+// WITH a trailing path delimiter.
+// If the project root is not found, the string specified by
+// kCannotFindProjectRootDir is returned.
+std::string ProjectRootPath();
+
+// Creates and returns the absolute path to the output directory where log files
+// and other test artifacts should be put. The output directory is always a
+// directory named "out" at the top-level of the project, i.e. a subfolder to
+// the path returned by ProjectRootPath().
+//
+// Details described for ProjectRootPath() apply here too.
+//
+// Returns the path WITH a trailing path delimiter. If the project root is not
+// found, the current working directory ("./") is returned as a fallback.
+std::string OutputPath();
+
+// Returns a path to a resource file for the currently executing platform.
+// Adapts to what filenames are currently present in the
+// [project-root]/resources/ dir.
+// Returns an absolute path according to this priority list (the directory
+// part of the path is left out for readability):
+// 1. [name]_[platform]_[architecture].[extension]
+// 2. [name]_[platform].[extension]
+// 3. [name]_[architecture].[extension]
+// 4. [name].[extension]
+// Where
+// * platform is either of "win", "mac" or "linux".
+// * architecture is either of "32" or "64".
+//
+// Arguments:
+// name - Name of the resource file. If a plain filename (no directory path)
+// is supplied, the file is assumed to be located in resources/
+// If a directory path is prepended to the filename, a subdirectory
+// hierarchy reflecting that path is assumed to be present.
+// extension - File extension, without the dot, i.e. "bmp" or "yuv".
+std::string ResourcePath(std::string name, std::string extension);
+
+// Gets the current working directory for the executing program.
+// Returns "./" if for some reason it is not possible to find the working
+// directory.
+std::string WorkingDir();
+
+// Creates a directory if it not already exists.
+// Returns true if successful. Will print an error message to stderr and return
+// false if a file with the same name already exists.
+bool CreateDirectory(std::string directory_name);
+
+// File size of the supplied file in bytes. Will return 0 if the file is
+// empty or if the file does not exist/is readable.
+size_t GetFileSize(std::string filename);
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
diff --git a/trunk/test/testsupport/fileutils_unittest.cc b/trunk/test/testsupport/fileutils_unittest.cc
new file mode 100644
index 0000000..1b76b3c
--- /dev/null
+++ b/trunk/test/testsupport/fileutils_unittest.cc
@@ -0,0 +1,192 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/fileutils.h"
+
+#include <cstdio>
+#include <list>
+#include <string>
+
+#include "gtest/gtest.h"
+
+#ifdef WIN32
+#define chdir _chdir
+static const char* kPathDelimiter = "\\";
+#else
+static const char* kPathDelimiter = "/";
+#endif
+
+static const std::string kDummyDir = "file_utils_unittest_dummy_dir";
+static const std::string kResourcesDir = "resources";
+static const std::string kTestName = "fileutils_unittest";
+static const std::string kExtension = "tmp";
+
+typedef std::list<std::string> FileList;
+
+namespace webrtc {
+
+// Test fixture to restore the working directory between each test, since some
+// of them change it with chdir during execution (not restored by the
+// gtest framework).
+class FileUtilsTest : public testing::Test {
+ protected:
+ FileUtilsTest() {
+ }
+ virtual ~FileUtilsTest() {}
+ // Runs before the first test
+ static void SetUpTestCase() {
+ original_working_dir_ = webrtc::test::WorkingDir();
+ std::string resources_path = original_working_dir_ + kPathDelimiter +
+ kResourcesDir + kPathDelimiter;
+ webrtc::test::CreateDirectory(resources_path);
+
+ files_.push_back(resources_path + kTestName + "." + kExtension);
+ files_.push_back(resources_path + kTestName + "_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win_64." + kExtension);
+
+ // Now that the resources dir exists, write some empty test files into it.
+ for (FileList::iterator file_it = files_.begin();
+ file_it != files_.end(); ++file_it) {
+ FILE* file = fopen(file_it->c_str(), "wb");
+ ASSERT_TRUE(file != NULL) << "Failed to write file: " << file_it->c_str();
+ ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0);
+ fclose(file);
+ }
+ // Create a dummy subdir that can be chdir'ed into for testing purposes.
+ empty_dummy_dir_ = original_working_dir_ + kPathDelimiter + kDummyDir;
+ webrtc::test::CreateDirectory(empty_dummy_dir_);
+ }
+ static void TearDownTestCase() {
+ // Clean up all resource files written
+ for (FileList::iterator file_it = files_.begin();
+ file_it != files_.end(); ++file_it) {
+ remove(file_it->c_str());
+ }
+ std::remove(empty_dummy_dir_.c_str());
+ }
+ void SetUp() {
+ ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
+ }
+ void TearDown() {
+ ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
+ }
+ protected:
+ static FileList files_;
+ static std::string empty_dummy_dir_;
+ private:
+ static std::string original_working_dir_;
+};
+
+FileList FileUtilsTest::files_;
+std::string FileUtilsTest::original_working_dir_ = "";
+std::string FileUtilsTest::empty_dummy_dir_ = "";
+
+// Tests that the project root path is returned for the default working
+// directory that is automatically set when the test executable is launched.
+// The test is not fully testing the implementation, since we cannot be sure
+// of where the executable was launched from.
+// The test will fail if the top level directory is not named "trunk".
+TEST_F(FileUtilsTest, ProjectRootPathFromUnchangedWorkingDir) {
+ std::string path = webrtc::test::ProjectRootPath();
+ std::string expected_end = "trunk";
+ expected_end = kPathDelimiter + expected_end + kPathDelimiter;
+ ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) {
+ std::string path = webrtc::test::OutputPath();
+ std::string expected_end = "out";
+ expected_end = kPathDelimiter + expected_end + kPathDelimiter;
+ ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
+}
+
+// Tests setting the current working directory to a directory three levels
+// deeper from the current one. Then testing that the project path returned
+// is still the same, when the function under test is called again.
+TEST_F(FileUtilsTest, ProjectRootPathFromDeeperWorkingDir) {
+ std::string path = webrtc::test::ProjectRootPath();
+ std::string original_working_dir = path; // This is the correct project root
+ // Change to a subdirectory path.
+ ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
+ ASSERT_EQ(original_working_dir, webrtc::test::ProjectRootPath());
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromDeeperWorkingDir) {
+ std::string path = webrtc::test::OutputPath();
+ std::string original_working_dir = path;
+ ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
+ ASSERT_EQ(original_working_dir, webrtc::test::OutputPath());
+}
+
+// Tests with current working directory set to a directory higher up in the
+// directory tree than the project root dir. This case shall return a specified
+// error string as a directory (which will be an invalid path).
+TEST_F(FileUtilsTest, ProjectRootPathFromRootWorkingDir) {
+ // Change current working dir to the root of the current file system
+ // (this will always be "above" our project root dir).
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ(webrtc::test::kCannotFindProjectRootDir,
+ webrtc::test::ProjectRootPath());
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) {
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ("./", webrtc::test::OutputPath());
+}
+
+// Only tests that the code executes
+TEST_F(FileUtilsTest, CreateDirectory) {
+ std::string directory = "fileutils-unittest-empty-dir";
+ // Make sure it's removed if a previous test has failed:
+ std::remove(directory.c_str());
+ ASSERT_TRUE(webrtc::test::CreateDirectory(directory));
+ std::remove(directory.c_str());
+}
+
+TEST_F(FileUtilsTest, WorkingDirReturnsValue) {
+ // Hard to cover all platforms. Just test that it returns something without
+ // crashing:
+ std::string working_dir = webrtc::test::WorkingDir();
+ ASSERT_GT(working_dir.length(), 0u);
+}
+
+// Due to multiple platforms, it is hard to make a complete test for
+// ResourcePath. Manual testing has been performed by removing files and
+// verified the result confirms with the specified documentation for the
+// function.
+TEST_F(FileUtilsTest, ResourcePathReturnsValue) {
+ std::string resource = webrtc::test::ResourcePath(kTestName, kExtension);
+ ASSERT_GT(resource.find(kTestName), 0u);
+ ASSERT_GT(resource.find(kExtension), 0u);
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ("./", webrtc::test::OutputPath());
+}
+
+TEST_F(FileUtilsTest, GetFileSizeExistingFile) {
+ ASSERT_GT(webrtc::test::GetFileSize(files_.front()), 0u);
+}
+
+TEST_F(FileUtilsTest, GetFileSizeNonExistingFile) {
+ ASSERT_EQ(0u, webrtc::test::GetFileSize("non-existing-file.tmp"));
+}
+
+} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_reader.cc b/trunk/test/testsupport/frame_reader.cc
new file mode 100644
index 0000000..b05ea58
--- /dev/null
+++ b/trunk/test/testsupport/frame_reader.cc
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_reader.h"
+
+#include <cassert>
+
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+FrameReaderImpl::FrameReaderImpl(std::string input_filename,
+ int frame_length_in_bytes)
+ : input_filename_(input_filename),
+ frame_length_in_bytes_(frame_length_in_bytes),
+ input_file_(NULL) {
+}
+
+FrameReaderImpl::~FrameReaderImpl() {
+ Close();
+}
+
+bool FrameReaderImpl::Init() {
+ if (frame_length_in_bytes_ <= 0) {
+ fprintf(stderr, "Frame length must be >0, was %d\n",
+ frame_length_in_bytes_);
+ return false;
+ }
+ input_file_ = fopen(input_filename_.c_str(), "rb");
+ if (input_file_ == NULL) {
+ fprintf(stderr, "Couldn't open input file for reading: %s\n",
+ input_filename_.c_str());
+ return false;
+ }
+ // Calculate total number of frames.
+ size_t source_file_size = GetFileSize(input_filename_);
+ if (source_file_size <= 0u) {
+ fprintf(stderr, "Found empty file: %s\n", input_filename_.c_str());
+ return false;
+ }
+ number_of_frames_ = source_file_size / frame_length_in_bytes_;
+ return true;
+}
+
+void FrameReaderImpl::Close() {
+ if (input_file_ != NULL) {
+ fclose(input_file_);
+ input_file_ = NULL;
+ }
+}
+
+bool FrameReaderImpl::ReadFrame(WebRtc_UWord8* source_buffer) {
+ assert(source_buffer);
+ if (input_file_ == NULL) {
+ fprintf(stderr, "FrameReader is not initialized (input file is NULL)\n");
+ return false;
+ }
+ size_t nbr_read = fread(source_buffer, 1, frame_length_in_bytes_,
+ input_file_);
+ if (nbr_read != static_cast<unsigned int>(frame_length_in_bytes_) &&
+ ferror(input_file_)) {
+ fprintf(stderr, "Error reading from input file: %s\n",
+ input_filename_.c_str());
+ return false;
+ }
+ if (feof(input_file_) != 0) {
+ return false; // No more frames to process.
+ }
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_reader.h b/trunk/test/testsupport/frame_reader.h
new file mode 100644
index 0000000..56d8fc4
--- /dev/null
+++ b/trunk/test/testsupport/frame_reader.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
+
+#include <cstdio>
+#include <string>
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles reading of frames from video files.
+class FrameReader {
+ public:
+ virtual ~FrameReader() {}
+
+ // Initializes the frame reader, i.e. opens the input file.
+ // This must be called before reading of frames has started.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Reads a frame into the supplied buffer, which must contain enough space
+ // for the frame size.
+ // Returns true if there are more frames to read, false if we've already
+ // read the last frame (in the previous call).
+ virtual bool ReadFrame(WebRtc_UWord8* source_buffer) = 0;
+
+ // Closes the input file if open. Essentially makes this class impossible
+ // to use anymore. Will also be invoked by the destructor.
+ virtual void Close() = 0;
+
+ // Frame length in bytes of a single frame image.
+ virtual int FrameLength() = 0;
+ // Total number of frames in the input video source.
+ virtual int NumberOfFrames() = 0;
+};
+
+class FrameReaderImpl : public FrameReader {
+ public:
+ // Creates a file handler. The input file is assumed to exist and be readable.
+ // Parameters:
+ // input_filename The file to read from.
+ // frame_length_in_bytes The size of each frame.
+ // For YUV this is 3 * width * height / 2
+ FrameReaderImpl(std::string input_filename, int frame_length_in_bytes);
+ virtual ~FrameReaderImpl();
+ bool Init();
+ bool ReadFrame(WebRtc_UWord8* source_buffer);
+ void Close();
+ int FrameLength() { return frame_length_in_bytes_; }
+ int NumberOfFrames() { return number_of_frames_; }
+
+ private:
+ std::string input_filename_;
+ int frame_length_in_bytes_;
+ int number_of_frames_;
+ FILE* input_file_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
diff --git a/trunk/test/testsupport/frame_reader_unittest.cc b/trunk/test/testsupport/frame_reader_unittest.cc
new file mode 100644
index 0000000..f1da5ce
--- /dev/null
+++ b/trunk/test/testsupport/frame_reader_unittest.cc
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_reader.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+const std::string kInputFilename = "temp_inputfile.tmp";
+const std::string kInputFileContents = "baz";
+// Setting the kFrameLength value to a value much larger than the
+// file to test causes the ReadFrame test to fail on Windows.
+const int kFrameLength = 1000;
+
+class FrameReaderTest: public testing::Test {
+ protected:
+ FrameReaderTest() {}
+ virtual ~FrameReaderTest() {}
+ void SetUp() {
+ // Cleanup any previous dummy input file.
+ std::remove(kInputFilename.c_str());
+
+ // Create a dummy input file.
+ FILE* dummy = fopen(kInputFilename.c_str(), "wb");
+ fprintf(dummy, "%s", kInputFileContents.c_str());
+ fclose(dummy);
+
+ frame_reader_ = new FrameReaderImpl(kInputFilename, kFrameLength);
+ ASSERT_TRUE(frame_reader_->Init());
+ }
+ void TearDown() {
+ delete frame_reader_;
+ // Cleanup the dummy input file.
+ std::remove(kInputFilename.c_str());
+ }
+ FrameReader* frame_reader_;
+};
+
+TEST_F(FrameReaderTest, InitSuccess) {
+ FrameReaderImpl frame_reader(kInputFilename, kFrameLength);
+ ASSERT_TRUE(frame_reader.Init());
+ ASSERT_EQ(kFrameLength, frame_reader.FrameLength());
+ ASSERT_EQ(0, frame_reader.NumberOfFrames());
+}
+
+TEST_F(FrameReaderTest, ReadFrame) {
+ WebRtc_UWord8 buffer[3];
+ bool result = frame_reader_->ReadFrame(buffer);
+ ASSERT_FALSE(result); // No more files to read.
+ ASSERT_EQ(kInputFileContents[0], buffer[0]);
+ ASSERT_EQ(kInputFileContents[1], buffer[1]);
+ ASSERT_EQ(kInputFileContents[2], buffer[2]);
+}
+
+TEST_F(FrameReaderTest, ReadFrameUninitialized) {
+ WebRtc_UWord8 buffer[3];
+ FrameReaderImpl file_reader(kInputFilename, kFrameLength);
+ ASSERT_FALSE(file_reader.ReadFrame(buffer));
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_writer.cc b/trunk/test/testsupport/frame_writer.cc
new file mode 100644
index 0000000..5f32539
--- /dev/null
+++ b/trunk/test/testsupport/frame_writer.cc
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_writer.h"
+
+#include <cassert>
+
+namespace webrtc {
+namespace test {
+
+FrameWriterImpl::FrameWriterImpl(std::string output_filename,
+ int frame_length_in_bytes)
+ : output_filename_(output_filename),
+ frame_length_in_bytes_(frame_length_in_bytes),
+ output_file_(NULL) {
+}
+
+FrameWriterImpl::~FrameWriterImpl() {
+ Close();
+}
+
+bool FrameWriterImpl::Init() {
+ if (frame_length_in_bytes_ <= 0) {
+ fprintf(stderr, "Frame length must be >0, was %d\n",
+ frame_length_in_bytes_);
+ return false;
+ }
+ output_file_ = fopen(output_filename_.c_str(), "wb");
+ if (output_file_ == NULL) {
+ fprintf(stderr, "Couldn't open output file for writing: %s\n",
+ output_filename_.c_str());
+ return false;
+ }
+ return true;
+}
+
+void FrameWriterImpl::Close() {
+ if (output_file_ != NULL) {
+ fclose(output_file_);
+ output_file_ = NULL;
+ }
+}
+
+bool FrameWriterImpl::WriteFrame(WebRtc_UWord8* frame_buffer) {
+ assert(frame_buffer);
+ if (output_file_ == NULL) {
+ fprintf(stderr, "FrameWriter is not initialized (output file is NULL)\n");
+ return false;
+ }
+ int bytes_written = fwrite(frame_buffer, 1, frame_length_in_bytes_,
+ output_file_);
+ if (bytes_written != frame_length_in_bytes_) {
+ fprintf(stderr, "Failed to write %d bytes to file %s\n",
+ frame_length_in_bytes_, output_filename_.c_str());
+ return false;
+ }
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_writer.h b/trunk/test/testsupport/frame_writer.h
new file mode 100644
index 0000000..abc5d35
--- /dev/null
+++ b/trunk/test/testsupport/frame_writer.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
+#define WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
+
+#include <cstdio>
+#include <string>
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles writing of video files.
+class FrameWriter {
+ public:
+ virtual ~FrameWriter() {}
+
+ // Initializes the file handler, i.e. opens the input and output files etc.
+ // This must be called before reading or writing frames has started.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Writes a frame of the configured frame length to the output file.
+ // Returns true if the write was successful, false otherwise.
+ virtual bool WriteFrame(WebRtc_UWord8* frame_buffer) = 0;
+
+ // Closes the output file if open. Essentially makes this class impossible
+ // to use anymore. Will also be invoked by the destructor.
+ virtual void Close() = 0;
+
+ // Frame length in bytes of a single frame image.
+ virtual int FrameLength() = 0;
+};
+
+class FrameWriterImpl : public FrameWriter {
+ public:
+ // Creates a file handler. The input file is assumed to exist and be readable
+ // and the output file must be writable.
+ // Parameters:
+ // output_filename The file to write. Will be overwritten if already
+ // existing.
+ // frame_length_in_bytes The size of each frame.
+ // For YUV: 3*width*height/2
+ FrameWriterImpl(std::string output_filename, int frame_length_in_bytes);
+ virtual ~FrameWriterImpl();
+ bool Init();
+ bool WriteFrame(WebRtc_UWord8* frame_buffer);
+ void Close();
+ int FrameLength() { return frame_length_in_bytes_; }
+
+ private:
+ std::string output_filename_;
+ int frame_length_in_bytes_;
+ int number_of_frames_;
+ FILE* output_file_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
diff --git a/trunk/test/testsupport/frame_writer_unittest.cc b/trunk/test/testsupport/frame_writer_unittest.cc
new file mode 100644
index 0000000..d25d1d2
--- /dev/null
+++ b/trunk/test/testsupport/frame_writer_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_writer.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+const std::string kOutputFilename = "temp_outputfile.tmp";
+const int kFrameLength = 1000;
+
+class FrameWriterTest: public testing::Test {
+ protected:
+ FrameWriterTest() {}
+ virtual ~FrameWriterTest() {}
+ void SetUp() {
+ // Cleanup any previous output file.
+ std::remove(kOutputFilename.c_str());
+ frame_writer_ = new FrameWriterImpl(kOutputFilename, kFrameLength);
+ ASSERT_TRUE(frame_writer_->Init());
+ }
+ void TearDown() {
+ delete frame_writer_;
+ // Cleanup the temporary file.
+ std::remove(kOutputFilename.c_str());
+ }
+ FrameWriter* frame_writer_;
+};
+
+TEST_F(FrameWriterTest, InitSuccess) {
+ FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
+ ASSERT_TRUE(frame_writer.Init());
+ ASSERT_EQ(kFrameLength, frame_writer.FrameLength());
+}
+
+TEST_F(FrameWriterTest, WriteFrame) {
+ WebRtc_UWord8 buffer[kFrameLength];
+ memset(buffer, 9, kFrameLength); // Write lots of 9s to the buffer
+ bool result = frame_writer_->WriteFrame(buffer);
+ ASSERT_TRUE(result); // success
+ // Close the file and verify the size.
+ frame_writer_->Close();
+ ASSERT_EQ(kFrameLength,
+ static_cast<int>(GetFileSize(kOutputFilename)));
+}
+
+TEST_F(FrameWriterTest, WriteFrameUninitialized) {
+ WebRtc_UWord8 buffer[3];
+ FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
+ ASSERT_FALSE(frame_writer.WriteFrame(buffer));
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/gtest_prod_util.h b/trunk/test/testsupport/gtest_prod_util.h
new file mode 100644
index 0000000..7d123a8
--- /dev/null
+++ b/trunk/test/testsupport/gtest_prod_util.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
+#define WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
+#pragma once
+
+#include "gtest/gtest_prod.h"
+
+// This file is a plain copy of Chromium's base/gtest_prod_util.h.
+//
+// This is a wrapper for gtest's FRIEND_TEST macro that friends
+// test with all possible prefixes. This is very helpful when changing the test
+// prefix, because the friend declarations don't need to be updated.
+//
+// Example usage:
+//
+// class MyClass {
+// private:
+// void MyMethod();
+// FRIEND_TEST_ALL_PREFIXES(MyClassTest, MyMethod);
+// };
+#define FRIEND_TEST_ALL_PREFIXES(test_case_name, test_name) \
+ FRIEND_TEST(test_case_name, test_name); \
+ FRIEND_TEST(test_case_name, DISABLED_##test_name); \
+ FRIEND_TEST(test_case_name, FLAKY_##test_name); \
+ FRIEND_TEST(test_case_name, FAILS_##test_name)
+
+#endif // WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
diff --git a/trunk/test/testsupport/metrics/video_metrics.cc b/trunk/test/testsupport/metrics/video_metrics.cc
new file mode 100644
index 0000000..438f31b
--- /dev/null
+++ b/trunk/test/testsupport/metrics/video_metrics.cc
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/metrics/video_metrics.h"
+
+#include <algorithm> // min_element, max_element
+#include <cassert>
+#include <cstdio>
+
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc {
+namespace test {
+
+// Used for calculating min and max values
+static bool LessForFrameResultValue (const FrameResult& s1,
+ const FrameResult& s2) {
+ return s1.value < s2.value;
+}
+
+enum VideoMetricsType { kPSNR, kSSIM, kBoth };
+
+// Calculates metrics for a frame and adds statistics to the result for it.
+void CalculateFrame(VideoMetricsType video_metrics_type,
+ uint8_t* ref,
+ uint8_t* test,
+ int width,
+ int height,
+ int frame_number,
+ QualityMetricsResult* result) {
+ FrameResult frame_result = {0, 0};
+ frame_result.frame_number = frame_number;
+ switch (video_metrics_type) {
+ case kPSNR:
+ frame_result.value = I420PSNR(ref, test, width, height);
+ break;
+ case kSSIM:
+ frame_result.value = I420SSIM(ref, test, width, height);
+ break;
+ default:
+ assert(false);
+ }
+ result->frames.push_back(frame_result);
+}
+
+// Calculates average, min and max values for the supplied struct, if non-NULL.
+void CalculateStats(QualityMetricsResult* result) {
+ if (result == NULL || result->frames.size() == 0) {
+ return;
+ }
+ // Calculate average
+ std::vector<FrameResult>::iterator iter;
+ double metrics_values_sum = 0.0;
+ for (iter = result->frames.begin(); iter != result->frames.end(); ++iter) {
+ metrics_values_sum += iter->value;
+ }
+ result->average = metrics_values_sum / result->frames.size();
+
+ // Calculate min/max statistics
+ iter = std::min_element(result->frames.begin(), result->frames.end(),
+ LessForFrameResultValue);
+ result->min = iter->value;
+ result->min_frame_number = iter->frame_number;
+ iter = std::max_element(result->frames.begin(), result->frames.end(),
+ LessForFrameResultValue);
+ result->max = iter->value;
+ result->max_frame_number = iter->frame_number;
+}
+
+// Single method that handles all combinations of video metrics calculation, to
+// minimize code duplication. Either psnr_result or ssim_result may be NULL,
+// depending on which VideoMetricsType is targeted.
+int CalculateMetrics(VideoMetricsType video_metrics_type,
+ const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result) {
+ assert(ref_filename != NULL);
+ assert(test_filename != NULL);
+ assert(width > 0);
+ assert(height > 0);
+
+ FILE* ref_fp = fopen(ref_filename, "rb");
+ if (ref_fp == NULL) {
+ // cannot open reference file
+ fprintf(stderr, "Cannot open file %s\n", ref_filename);
+ return -1;
+ }
+ FILE* test_fp = fopen(test_filename, "rb");
+ if (test_fp == NULL) {
+ // cannot open test file
+ fprintf(stderr, "Cannot open file %s\n", test_filename);
+ fclose(ref_fp);
+ return -2;
+ }
+ int frame_number = 0;
+
+ // Allocating size for one I420 frame.
+ const int frame_length = 3 * width * height >> 1;
+ uint8_t* ref = new uint8_t[frame_length];
+ uint8_t* test = new uint8_t[frame_length];
+
+ int ref_bytes = fread(ref, 1, frame_length, ref_fp);
+ int test_bytes = fread(test, 1, frame_length, test_fp);
+ while (ref_bytes == frame_length && test_bytes == frame_length) {
+ switch (video_metrics_type) {
+ case kPSNR:
+ CalculateFrame(kPSNR, ref, test, width, height, frame_number,
+ psnr_result);
+ break;
+ case kSSIM:
+ CalculateFrame(kSSIM, ref, test, width, height, frame_number,
+ ssim_result);
+ break;
+ case kBoth:
+ CalculateFrame(kPSNR, ref, test, width, height, frame_number,
+ psnr_result);
+ CalculateFrame(kSSIM, ref, test, width, height, frame_number,
+ ssim_result);
+ break;
+ }
+ frame_number++;
+ ref_bytes = fread(ref, 1, frame_length, ref_fp);
+ test_bytes = fread(test, 1, frame_length, test_fp);
+ }
+ int return_code = 0;
+ if (frame_number == 0) {
+ fprintf(stderr, "Tried to measure video metrics from empty files "
+ "(reference file: %s test file: %s)\n", ref_filename,
+ test_filename);
+ return_code = -3;
+ } else {
+ CalculateStats(psnr_result);
+ CalculateStats(ssim_result);
+ }
+ delete [] ref;
+ delete [] test;
+ fclose(ref_fp);
+ fclose(test_fp);
+ return return_code;
+}
+
+int I420MetricsFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result) {
+ assert(psnr_result != NULL);
+ assert(ssim_result != NULL);
+ return CalculateMetrics(kBoth, ref_filename, test_filename, width, height,
+ psnr_result, ssim_result);
+}
+
+int I420PSNRFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result) {
+ assert(result != NULL);
+ return CalculateMetrics(kPSNR, ref_filename, test_filename, width, height,
+ result, NULL);
+}
+
+int I420SSIMFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result) {
+ assert(result != NULL);
+ return CalculateMetrics(kSSIM, ref_filename, test_filename, width, height,
+ NULL, result);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/metrics/video_metrics.h b/trunk/test/testsupport/metrics/video_metrics.h
new file mode 100644
index 0000000..df11a49
--- /dev/null
+++ b/trunk/test/testsupport/metrics/video_metrics.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
+#define WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
+
+#include <limits>
+#include <vector>
+
+namespace webrtc {
+namespace test {
+
+// Contains video quality metrics result for a single frame.
+struct FrameResult {
+ int frame_number;
+ double value;
+};
+
+// Result from a PSNR/SSIM calculation operation.
+// The frames in this data structure are 0-indexed.
+struct QualityMetricsResult {
+ QualityMetricsResult() :
+ average(0.0),
+ min(std::numeric_limits<double>::max()),
+ max(std::numeric_limits<double>::min()),
+ min_frame_number(-1),
+ max_frame_number(-1)
+ {};
+ double average;
+ double min;
+ double max;
+ int min_frame_number;
+ int max_frame_number;
+ std::vector<FrameResult> frames;
+};
+
+// Calculates PSNR and SSIM values for the reference and test video files
+// (must be in I420 format). All calculated values are filled into the
+// QualityMetricsResult stucts.
+// PSNR values have the unit decibel (dB) where a high value means the test file
+// is similar to the reference file. The higher value, the more similar.
+// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
+// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
+// identical. For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420MetricsFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result);
+
+// Calculates PSNR values for the reference and test video files (must be in
+// I420 format). All calculated values are filled into the QualityMetricsResult
+// struct.
+// PSNR values have the unit decibel (dB) where a high value means the test file
+// is similar to the reference file. The higher value, the more similar.
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
+//
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420PSNRFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result);
+
+// Calculates SSIM values for the reference and test video files (must be in
+// I420 format). All calculated values are filled into the QualityMetricsResult
+// struct.
+// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
+// identical.
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
+//
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420SSIMFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result);
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
diff --git a/trunk/test/testsupport/metrics/video_metrics_unittest.cc b/trunk/test/testsupport/metrics/video_metrics_unittest.cc
new file mode 100644
index 0000000..e77dbff
--- /dev/null
+++ b/trunk/test/testsupport/metrics/video_metrics_unittest.cc
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/metrics/video_metrics.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+static const char* kEmptyFileName = "video_metrics_unittest_empty_file.tmp";
+static const char* kNonExistingFileName = "video_metrics_unittest_non_existing";
+static const int kWidth = 352;
+static const int kHeight = 288;
+
+static const int kMissingReferenceFileReturnCode = -1;
+static const int kMissingTestFileReturnCode = -2;
+static const int kEmptyFileReturnCode = -3;
+static const double kPsnrPerfectResult = 48.0;
+static const double kSsimPerfectResult = 1.0;
+
+class VideoMetricsTest: public testing::Test {
+ protected:
+ VideoMetricsTest() {
+ video_file_ = webrtc::test::ResourcePath("foreman_cif_short", "yuv");
+ }
+ virtual ~VideoMetricsTest() {}
+ void SetUp() {
+ // Create an empty file:
+ FILE* dummy = fopen(kEmptyFileName, "wb");
+ fclose(dummy);
+ }
+ void TearDown() {
+ std::remove(kEmptyFileName);
+ }
+ webrtc::test::QualityMetricsResult psnr_result_;
+ webrtc::test::QualityMetricsResult ssim_result_;
+ std::string video_file_;
+};
+
+// Tests that it is possible to run with the same reference as test file
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesPSNR) {
+ EXPECT_EQ(0, I420PSNRFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &psnr_result_));
+ EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
+}
+
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesSSIM) {
+ EXPECT_EQ(0, I420SSIMFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
+}
+
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesBothMetrics) {
+ EXPECT_EQ(0, I420MetricsFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &psnr_result_,
+ &ssim_result_));
+ EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
+ EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
+}
+
+// Tests that the right return code is given when the reference file is missing.
+TEST_F(VideoMetricsTest, MissingReferenceFilePSNR) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420PSNRFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingReferenceFileSSIM) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420SSIMFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingReferenceFileBothMetrics) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420MetricsFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+// Tests that the right return code is given when the test file is missing.
+TEST_F(VideoMetricsTest, MissingTestFilePSNR) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420PSNRFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingTestFileSSIM) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420SSIMFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingTestFileBothMetrics) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420MetricsFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+// Tests that the method can be executed with empty files.
+TEST_F(VideoMetricsTest, EmptyFilesPSNR) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420PSNRFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420PSNRFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, EmptyFilesSSIM) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420SSIMFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420SSIMFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, EmptyFilesBothMetrics) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420MetricsFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420MetricsFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+} // namespace webrtc
diff --git a/trunk/test/testsupport/mock/mock_frame_reader.h b/trunk/test/testsupport/mock/mock_frame_reader.h
new file mode 100644
index 0000000..ecfc13c
--- /dev/null
+++ b/trunk/test/testsupport/mock/mock_frame_reader.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+
+#include "testsupport/frame_reader.h"
+
+#include "gmock/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockFrameReader : public FrameReader {
+ public:
+ MOCK_METHOD0(Init, bool());
+ MOCK_METHOD1(ReadFrame, bool(WebRtc_UWord8* source_buffer));
+ MOCK_METHOD0(Close, void());
+ MOCK_METHOD0(FrameLength, int());
+ MOCK_METHOD0(NumberOfFrames, int());
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
diff --git a/trunk/test/testsupport/mock/mock_frame_writer.h b/trunk/test/testsupport/mock/mock_frame_writer.h
new file mode 100644
index 0000000..ba79184
--- /dev/null
+++ b/trunk/test/testsupport/mock/mock_frame_writer.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
+#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
+
+#include "testsupport/frame_writer.h"
+
+#include "gmock/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockFrameWriter : public FrameWriter {
+ public:
+ MOCK_METHOD0(Init, bool());
+ MOCK_METHOD1(WriteFrame, bool(WebRtc_UWord8* frame_buffer));
+ MOCK_METHOD0(Close, void());
+ MOCK_METHOD0(FrameLength, int());
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
diff --git a/trunk/test/testsupport/packet_reader.cc b/trunk/test/testsupport/packet_reader.cc
new file mode 100644
index 0000000..e8859d1
--- /dev/null
+++ b/trunk/test/testsupport/packet_reader.cc
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/packet_reader.h"
+
+#include <cassert>
+#include <cstdio>
+
+namespace webrtc {
+namespace test {
+
+PacketReader::PacketReader()
+ : initialized_(false) {}
+
+PacketReader::~PacketReader() {}
+
+void PacketReader::InitializeReading(WebRtc_UWord8* data,
+ int data_length_in_bytes,
+ int packet_size_in_bytes) {
+ assert(data);
+ assert(data_length_in_bytes >= 0);
+ assert(packet_size_in_bytes > 0);
+ data_ = data;
+ data_length_ = data_length_in_bytes;
+ packet_size_ = packet_size_in_bytes;
+ currentIndex_ = 0;
+ initialized_ = true;
+}
+
+int PacketReader::NextPacket(WebRtc_UWord8** packet_pointer) {
+ if (!initialized_) {
+ fprintf(stderr, "Attempting to use uninitialized PacketReader!\n");
+ return -1;
+ }
+ *packet_pointer = data_ + currentIndex_;
+ // Check if we're about to read the last packet:
+ if (data_length_ - currentIndex_ <= packet_size_) {
+ int size = data_length_ - currentIndex_;
+ currentIndex_ = data_length_;
+ assert(size >= 0);
+ return size;
+ }
+ currentIndex_ += packet_size_;
+ assert(packet_size_ >= 0);
+ return packet_size_;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/packet_reader.h b/trunk/test/testsupport/packet_reader.h
new file mode 100644
index 0000000..4cb0bb1
--- /dev/null
+++ b/trunk/test/testsupport/packet_reader.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Reads chunks of data to simulate network packets from a byte array.
+class PacketReader {
+ public:
+ PacketReader();
+ virtual ~PacketReader();
+
+ // Inizializes a new reading operation. Must be done before invoking the
+ // NextPacket method.
+ // * data_length_in_bytes is the length of the data byte array. Must be >= 0.
+ // 0 length will result in no packets are read.
+ // * packet_size_in_bytes is the number of bytes to read in each NextPacket
+ // method call. Must be > 0
+ virtual void InitializeReading(WebRtc_UWord8* data, int data_length_in_bytes,
+ int packet_size_in_bytes);
+
+ // Moves the supplied pointer to the beginning of the next packet.
+ // Returns:
+ // * The size of the packet ready to read (lower than the packet size for
+ // the last packet)
+ // * 0 if there are no more packets to read
+ // * -1 if InitializeReading has not been called (also prints to stderr).
+ virtual int NextPacket(WebRtc_UWord8** packet_pointer);
+
+ private:
+ WebRtc_UWord8* data_;
+ int data_length_;
+ int packet_size_;
+ int currentIndex_;
+ bool initialized_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
diff --git a/trunk/test/testsupport/packet_reader_unittest.cc b/trunk/test/testsupport/packet_reader_unittest.cc
new file mode 100644
index 0000000..6719e4c
--- /dev/null
+++ b/trunk/test/testsupport/packet_reader_unittest.cc
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/packet_reader.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/unittest_utils.h"
+
+namespace webrtc {
+namespace test {
+
+class PacketReaderTest: public PacketRelatedTest {
+ protected:
+ PacketReaderTest() {}
+ virtual ~PacketReaderTest() {}
+ void SetUp() {
+ reader_ = new PacketReader();
+ }
+ void TearDown() {
+ delete reader_;
+ }
+ void VerifyPacketData(int expected_length,
+ int actual_length,
+ WebRtc_UWord8* original_data_pointer,
+ WebRtc_UWord8* new_data_pointer) {
+ EXPECT_EQ(expected_length, actual_length);
+ EXPECT_EQ(*original_data_pointer, *new_data_pointer);
+ EXPECT_EQ(0, memcmp(original_data_pointer, new_data_pointer,
+ actual_length));
+ }
+ PacketReader* reader_;
+};
+
+// Test lack of initialization
+TEST_F(PacketReaderTest, Uninitialized) {
+ WebRtc_UWord8* data_pointer = NULL;
+ EXPECT_EQ(-1, reader_->NextPacket(&data_pointer));
+ EXPECT_EQ(NULL, data_pointer);
+}
+
+TEST_F(PacketReaderTest, InitializeZeroLengthArgument) {
+ reader_->InitializeReading(packet_data_, 0, kPacketSizeInBytes);
+ ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_));
+}
+
+// Test with something smaller than one packet
+TEST_F(PacketReaderTest, NormalSmallData) {
+ const int kDataLengthInBytes = 1499;
+ WebRtc_UWord8 data[kDataLengthInBytes];
+ WebRtc_UWord8* data_pointer = data;
+ memset(data, 1, kDataLengthInBytes);
+
+ reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
+ int length_to_read = reader_->NextPacket(&data_pointer);
+ VerifyPacketData(kDataLengthInBytes, length_to_read, data, data_pointer);
+ EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&data_pointer);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kDataLengthInBytes, data_pointer - data);
+}
+
+// Test with data length that exactly matches one packet
+TEST_F(PacketReaderTest, NormalOnePacketData) {
+ WebRtc_UWord8 data[kPacketSizeInBytes];
+ WebRtc_UWord8* data_pointer = data;
+ memset(data, 1, kPacketSizeInBytes);
+
+ reader_->InitializeReading(data, kPacketSizeInBytes, kPacketSizeInBytes);
+ int length_to_read = reader_->NextPacket(&data_pointer);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read, data, data_pointer);
+ EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&data_pointer);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kPacketSizeInBytes, data_pointer - data);
+}
+
+// Test with data length that will result in 3 packets
+TEST_F(PacketReaderTest, NormalLargeData) {
+ reader_->InitializeReading(packet_data_, kPacketDataLength,
+ kPacketSizeInBytes);
+
+ int length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read,
+ packet1_, packet_data_pointer_);
+
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read,
+ packet2_, packet_data_pointer_);
+
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(1u, length_to_read,
+ packet3_, packet_data_pointer_);
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kPacketDataLength, packet_data_pointer_ - packet_data_);
+}
+
+// Test with empty data.
+TEST_F(PacketReaderTest, EmptyData) {
+ const int kDataLengthInBytes = 0;
+ WebRtc_UWord8* data = new WebRtc_UWord8[kDataLengthInBytes];
+ reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
+ EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
+ // Do it again to make sure nothing changes
+ EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
+ delete[] data;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/trunk/test/testsupport/unittest_utils.h b/trunk/test/testsupport/unittest_utils.h
new file mode 100644
index 0000000..30464de
--- /dev/null
+++ b/trunk/test/testsupport/unittest_utils.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
+#define WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
+
+namespace webrtc {
+namespace test {
+
+const int kPacketSizeInBytes = 1500;
+const int kPacketDataLength = kPacketSizeInBytes * 2 + 1;
+const int kPacketDataNumberOfPackets = 3;
+
+// A base test fixture for packet related tests. Contains
+// two full prepared packets with 1s, 2s in their data and a third packet with
+// a single 3 in it (size=1).
+// A packet data structure is also available, that contains these three packets
+// in order.
+class PacketRelatedTest: public testing::Test {
+ protected:
+ // Tree packet byte arrays with data used for verification:
+ WebRtc_UWord8 packet1_[kPacketSizeInBytes];
+ WebRtc_UWord8 packet2_[kPacketSizeInBytes];
+ WebRtc_UWord8 packet3_[1];
+ // Construct a data structure containing these packets
+ WebRtc_UWord8 packet_data_[kPacketDataLength];
+ WebRtc_UWord8* packet_data_pointer_;
+
+ PacketRelatedTest() {
+ packet_data_pointer_ = packet_data_;
+
+ memset(packet1_, 1, kPacketSizeInBytes);
+ memset(packet2_, 2, kPacketSizeInBytes);
+ memset(packet3_, 3, 1);
+ // Fill the packet_data:
+ memcpy(packet_data_pointer_, packet1_, kPacketSizeInBytes);
+ memcpy(packet_data_pointer_ + kPacketSizeInBytes, packet2_,
+ kPacketSizeInBytes);
+ memcpy(packet_data_pointer_ + kPacketSizeInBytes * 2, packet3_, 1);
+ }
+ virtual ~PacketRelatedTest() {}
+ void SetUp() {}
+ void TearDown() {}
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
diff --git a/trunk/third_party/google-gflags/LICENSE b/trunk/third_party/google-gflags/LICENSE
new file mode 100644
index 0000000..d15b0c2
--- /dev/null
+++ b/trunk/third_party/google-gflags/LICENSE
@@ -0,0 +1,28 @@
+Copyright (c) 2006, Google Inc.
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+ * Redistributions of source code must retain the above copyright
+notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above
+copyright notice, this list of conditions and the following disclaimer
+in the documentation and/or other materials provided with the
+distribution.
+ * Neither the name of Google Inc. nor the names of its
+contributors may be used to endorse or promote products derived from
+this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/third_party/google-gflags/README.webrtc b/trunk/third_party/google-gflags/README.webrtc
new file mode 100644
index 0000000..9a993ce
--- /dev/null
+++ b/trunk/third_party/google-gflags/README.webrtc
@@ -0,0 +1,13 @@
+URL: http://code.google.com/p/google-gflags/
+Version: 1.5
+License: New BSD
+License File: LICENSE
+
+Description:
+The gflags package contains a library that implements commandline
+flags processing. As such it's a replacement for getopt(). It has
+increased flexibility, including built-in support for C++ types like
+string, and the ability to define flags in the source file in which
+they're used.
+
+Local Modifications: None
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h
new file mode 100644
index 0000000..0c2f997
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h
@@ -0,0 +1,585 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 1
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 1
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+namespace google {
+
+#if 1 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 1 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 0 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern void ShowUsageWithFlags(const char *argv0); // what --help does
+extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern const char* GetArgv(); // all of argv as a string
+extern const char* GetArgv0(); // only argv0
+extern uint32 GetArgvSum(); // simple checksum of argv
+extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern std::string SetCommandLineOption(const char* name, const char* value);
+extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} __attribute__ ((unused));
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern bool SaveCommandFlags(); // actually defined in google.cc !
+extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern bool BoolFromEnv(const char *varname, bool defval);
+extern int32 Int32FromEnv(const char *varname, int32 defval);
+extern int64 Int64FromEnv(const char *varname, int64 defval);
+extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern double DoubleFromEnv(const char *varname, double defval);
+extern const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used. Returns the index (into argv)
+// of the first non-flag argument.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing. If a flag is
+// defined more than once in the command line or flag file, the last
+// definition is used. Returns the index (into argv) of the first
+// non-flag argument. (If remove_flags is true, will always return 1.)
+extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized. Only flags
+// registered since the last parse will be recognized. Any flag value
+// must be provided as part of the argument using "=", not as a
+// separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+// Returns the index (into the original argv) of the first non-flag
+// argument. (If remove_flags is true, will always return 1.)
+extern uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ extern type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> double IsBoolFlag(const From& from);
+bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+// The weird 'using' + 'extern' inside the fLS namespace is to work around
+// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
+// http://code.google.com/p/google-gflags/issues/detail?id=20
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ extern clstring& FLAGS_##name; \
+ using fLS::FLAGS_##name; \
+ clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..9d9ce7a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h
@@ -0,0 +1,121 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+namespace google {
+
+void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h
new file mode 100644
index 0000000..98d8e1a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h
@@ -0,0 +1,110 @@
+/* src/config.h. Generated from config.h.in by configure. */
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#define GFLAGS_DLL_DECL /**/
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#define HAVE_DLFCN_H 1
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#define HAVE_FNMATCH_H 1
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#define HAVE_INTTYPES_H 1
+
+/* Define to 1 if you have the <memory.h> header file. */
+#define HAVE_MEMORY_H 1
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#define HAVE_PTHREAD 1
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#define HAVE_SETENV 1
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#define HAVE_STDINT_H 1
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#define HAVE_STRINGS_H 1
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#define HAVE_UNISTD_H 1
+
+/* define if your compiler has __attribute__ */
+#define HAVE___ATTRIBUTE__ 1
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#define LT_OBJDIR ".libs/"
+
+/* Name of package */
+#define PACKAGE "gflags"
+
+/* Define to the address where bug reports for this package should be sent. */
+#define PACKAGE_BUGREPORT "opensource@google.com"
+
+/* Define to the full name of this package. */
+#define PACKAGE_NAME "gflags"
+
+/* Define to the full name and version of this package. */
+#define PACKAGE_STRING "gflags 1.5"
+
+/* Define to the one symbol short name of this package. */
+#define PACKAGE_TARNAME "gflags"
+
+/* Define to the home page for this package. */
+#define PACKAGE_URL ""
+
+/* Define to the version of this package. */
+#define PACKAGE_VERSION "1.5"
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+/* #undef PTHREAD_CREATE_JOINABLE */
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#define VERSION "1.5"
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h
new file mode 100644
index 0000000..0c2f997
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h
@@ -0,0 +1,585 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 1
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 1
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+namespace google {
+
+#if 1 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 1 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 0 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern void ShowUsageWithFlags(const char *argv0); // what --help does
+extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern const char* GetArgv(); // all of argv as a string
+extern const char* GetArgv0(); // only argv0
+extern uint32 GetArgvSum(); // simple checksum of argv
+extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern std::string SetCommandLineOption(const char* name, const char* value);
+extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} __attribute__ ((unused));
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern bool SaveCommandFlags(); // actually defined in google.cc !
+extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern bool BoolFromEnv(const char *varname, bool defval);
+extern int32 Int32FromEnv(const char *varname, int32 defval);
+extern int64 Int64FromEnv(const char *varname, int64 defval);
+extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern double DoubleFromEnv(const char *varname, double defval);
+extern const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used. Returns the index (into argv)
+// of the first non-flag argument.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing. If a flag is
+// defined more than once in the command line or flag file, the last
+// definition is used. Returns the index (into argv) of the first
+// non-flag argument. (If remove_flags is true, will always return 1.)
+extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized. Only flags
+// registered since the last parse will be recognized. Any flag value
+// must be provided as part of the argument using "=", not as a
+// separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+// Returns the index (into the original argv) of the first non-flag
+// argument. (If remove_flags is true, will always return 1.)
+extern uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ extern type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> double IsBoolFlag(const From& from);
+bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+// The weird 'using' + 'extern' inside the fLS namespace is to work around
+// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
+// http://code.google.com/p/google-gflags/issues/detail?id=20
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ extern clstring& FLAGS_##name; \
+ using fLS::FLAGS_##name; \
+ clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..9d9ce7a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h
@@ -0,0 +1,121 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+namespace google {
+
+void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h
new file mode 100644
index 0000000..98d8e1a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h
@@ -0,0 +1,110 @@
+/* src/config.h. Generated from config.h.in by configure. */
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#define GFLAGS_DLL_DECL /**/
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#define HAVE_DLFCN_H 1
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#define HAVE_FNMATCH_H 1
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#define HAVE_INTTYPES_H 1
+
+/* Define to 1 if you have the <memory.h> header file. */
+#define HAVE_MEMORY_H 1
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#define HAVE_PTHREAD 1
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#define HAVE_SETENV 1
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#define HAVE_STDINT_H 1
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#define HAVE_STRINGS_H 1
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#define HAVE_UNISTD_H 1
+
+/* define if your compiler has __attribute__ */
+#define HAVE___ATTRIBUTE__ 1
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#define LT_OBJDIR ".libs/"
+
+/* Name of package */
+#define PACKAGE "gflags"
+
+/* Define to the address where bug reports for this package should be sent. */
+#define PACKAGE_BUGREPORT "opensource@google.com"
+
+/* Define to the full name of this package. */
+#define PACKAGE_NAME "gflags"
+
+/* Define to the full name and version of this package. */
+#define PACKAGE_STRING "gflags 1.5"
+
+/* Define to the one symbol short name of this package. */
+#define PACKAGE_TARNAME "gflags"
+
+/* Define to the home page for this package. */
+#define PACKAGE_URL ""
+
+/* Define to the version of this package. */
+#define PACKAGE_VERSION "1.5"
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+/* #undef PTHREAD_CREATE_JOINABLE */
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#define VERSION "1.5"
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h
new file mode 100644
index 0000000..0c2f997
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h
@@ -0,0 +1,585 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 1
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 1
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+namespace google {
+
+#if 1 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 1 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 0 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern void ShowUsageWithFlags(const char *argv0); // what --help does
+extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern const char* GetArgv(); // all of argv as a string
+extern const char* GetArgv0(); // only argv0
+extern uint32 GetArgvSum(); // simple checksum of argv
+extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern std::string SetCommandLineOption(const char* name, const char* value);
+extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} __attribute__ ((unused));
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern bool SaveCommandFlags(); // actually defined in google.cc !
+extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern bool BoolFromEnv(const char *varname, bool defval);
+extern int32 Int32FromEnv(const char *varname, int32 defval);
+extern int64 Int64FromEnv(const char *varname, int64 defval);
+extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern double DoubleFromEnv(const char *varname, double defval);
+extern const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used. Returns the index (into argv)
+// of the first non-flag argument.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing. If a flag is
+// defined more than once in the command line or flag file, the last
+// definition is used. Returns the index (into argv) of the first
+// non-flag argument. (If remove_flags is true, will always return 1.)
+extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized. Only flags
+// registered since the last parse will be recognized. Any flag value
+// must be provided as part of the argument using "=", not as a
+// separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+// Returns the index (into the original argv) of the first non-flag
+// argument. (If remove_flags is true, will always return 1.)
+extern uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ extern type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> double IsBoolFlag(const From& from);
+bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+// The weird 'using' + 'extern' inside the fLS namespace is to work around
+// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
+// http://code.google.com/p/google-gflags/issues/detail?id=20
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ extern clstring& FLAGS_##name; \
+ using fLS::FLAGS_##name; \
+ clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..9d9ce7a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h
@@ -0,0 +1,121 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+namespace google {
+
+void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h
new file mode 100644
index 0000000..98d8e1a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h
@@ -0,0 +1,110 @@
+/* src/config.h. Generated from config.h.in by configure. */
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#define GFLAGS_DLL_DECL /**/
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#define HAVE_DLFCN_H 1
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#define HAVE_FNMATCH_H 1
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#define HAVE_INTTYPES_H 1
+
+/* Define to 1 if you have the <memory.h> header file. */
+#define HAVE_MEMORY_H 1
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#define HAVE_PTHREAD 1
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#define HAVE_SETENV 1
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#define HAVE_STDINT_H 1
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#define HAVE_STRINGS_H 1
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#define HAVE_UNISTD_H 1
+
+/* define if your compiler has __attribute__ */
+#define HAVE___ATTRIBUTE__ 1
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#define LT_OBJDIR ".libs/"
+
+/* Name of package */
+#define PACKAGE "gflags"
+
+/* Define to the address where bug reports for this package should be sent. */
+#define PACKAGE_BUGREPORT "opensource@google.com"
+
+/* Define to the full name of this package. */
+#define PACKAGE_NAME "gflags"
+
+/* Define to the full name and version of this package. */
+#define PACKAGE_STRING "gflags 1.5"
+
+/* Define to the one symbol short name of this package. */
+#define PACKAGE_TARNAME "gflags"
+
+/* Define to the home page for this package. */
+#define PACKAGE_URL ""
+
+/* Define to the version of this package. */
+#define PACKAGE_VERSION "1.5"
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+/* #undef PTHREAD_CREATE_JOINABLE */
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#define VERSION "1.5"
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h
new file mode 100644
index 0000000..0c2f997
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h
@@ -0,0 +1,585 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 1
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 1
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+namespace google {
+
+#if 1 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 1 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 0 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern void ShowUsageWithFlags(const char *argv0); // what --help does
+extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern const char* GetArgv(); // all of argv as a string
+extern const char* GetArgv0(); // only argv0
+extern uint32 GetArgvSum(); // simple checksum of argv
+extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern std::string SetCommandLineOption(const char* name, const char* value);
+extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} __attribute__ ((unused));
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern bool SaveCommandFlags(); // actually defined in google.cc !
+extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern bool BoolFromEnv(const char *varname, bool defval);
+extern int32 Int32FromEnv(const char *varname, int32 defval);
+extern int64 Int64FromEnv(const char *varname, int64 defval);
+extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern double DoubleFromEnv(const char *varname, double defval);
+extern const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used. Returns the index (into argv)
+// of the first non-flag argument.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing. If a flag is
+// defined more than once in the command line or flag file, the last
+// definition is used. Returns the index (into argv) of the first
+// non-flag argument. (If remove_flags is true, will always return 1.)
+extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized. Only flags
+// registered since the last parse will be recognized. Any flag value
+// must be provided as part of the argument using "=", not as a
+// separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+// Returns the index (into the original argv) of the first non-flag
+// argument. (If remove_flags is true, will always return 1.)
+extern uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ extern type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> double IsBoolFlag(const From& from);
+bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+// The weird 'using' + 'extern' inside the fLS namespace is to work around
+// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
+// http://code.google.com/p/google-gflags/issues/detail?id=20
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ extern clstring& FLAGS_##name; \
+ using fLS::FLAGS_##name; \
+ clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..9d9ce7a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h
@@ -0,0 +1,121 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+namespace google {
+
+void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h
new file mode 100644
index 0000000..98d8e1a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h
@@ -0,0 +1,110 @@
+/* src/config.h. Generated from config.h.in by configure. */
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#define GFLAGS_DLL_DECL /**/
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#define HAVE_DLFCN_H 1
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#define HAVE_FNMATCH_H 1
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#define HAVE_INTTYPES_H 1
+
+/* Define to 1 if you have the <memory.h> header file. */
+#define HAVE_MEMORY_H 1
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#define HAVE_PTHREAD 1
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#define HAVE_SETENV 1
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#define HAVE_STDINT_H 1
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#define HAVE_STRINGS_H 1
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#define HAVE_UNISTD_H 1
+
+/* define if your compiler has __attribute__ */
+#define HAVE___ATTRIBUTE__ 1
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#define LT_OBJDIR ".libs/"
+
+/* Name of package */
+#define PACKAGE "gflags"
+
+/* Define to the address where bug reports for this package should be sent. */
+#define PACKAGE_BUGREPORT "opensource@google.com"
+
+/* Define to the full name of this package. */
+#define PACKAGE_NAME "gflags"
+
+/* Define to the full name and version of this package. */
+#define PACKAGE_STRING "gflags 1.5"
+
+/* Define to the one symbol short name of this package. */
+#define PACKAGE_TARNAME "gflags"
+
+/* Define to the home page for this package. */
+#define PACKAGE_URL ""
+
+/* Define to the version of this package. */
+#define PACKAGE_VERSION "1.5"
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+/* #undef PTHREAD_CREATE_JOINABLE */
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#define VERSION "1.5"
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h
new file mode 100644
index 0000000..fdafe2a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h
@@ -0,0 +1,601 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 0
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 0
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+// Annoying stuff for windows -- makes sure clients can import these functions
+#if defined(_WIN32)
+# ifndef GFLAGS_DLL_DECL
+# define GFLAGS_DLL_DECL __declspec(dllimport)
+# endif
+# ifndef GFLAGS_DLL_DECLARE_FLAG
+# define GFLAGS_DLL_DECLARE_FLAG __declspec(dllimport)
+# endif
+# ifndef GFLAGS_DLL_DEFINE_FLAG
+# define GFLAGS_DLL_DEFINE_FLAG __declspec(dllexport)
+# endif
+#else
+# ifndef GFLAGS_DLL_DECL
+# define GFLAGS_DLL_DECL
+# endif
+# ifndef GFLAGS_DLL_DECLARE_FLAG
+# define GFLAGS_DLL_DECLARE_FLAG
+# endif
+# ifndef GFLAGS_DLL_DEFINE_FLAG
+# define GFLAGS_DLL_DEFINE_FLAG
+# endif
+#endif
+
+namespace google {
+
+#if 0 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 0 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 1 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct GFLAGS_DLL_DECL CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern GFLAGS_DLL_DECL void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern GFLAGS_DLL_DECL void ShowUsageWithFlags(const char *argv0); // what --help does
+extern GFLAGS_DLL_DECL void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern GFLAGS_DLL_DECL std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern GFLAGS_DLL_DECL void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern GFLAGS_DLL_DECL const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern GFLAGS_DLL_DECL const char* GetArgv(); // all of argv as a string
+extern GFLAGS_DLL_DECL const char* GetArgv0(); // only argv0
+extern GFLAGS_DLL_DECL uint32 GetArgvSum(); // simple checksum of argv
+extern GFLAGS_DLL_DECL const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern GFLAGS_DLL_DECL const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern GFLAGS_DLL_DECL const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern GFLAGS_DLL_DECL bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern GFLAGS_DLL_DECL bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern GFLAGS_DLL_DECL CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum GFLAGS_DLL_DECL FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern GFLAGS_DLL_DECL std::string SetCommandLineOption(const char* name, const char* value);
+extern GFLAGS_DLL_DECL std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class GFLAGS_DLL_DECL FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} ;
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern GFLAGS_DLL_DECL std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern GFLAGS_DLL_DECL bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern GFLAGS_DLL_DECL bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern GFLAGS_DLL_DECL bool SaveCommandFlags(); // actually defined in google.cc !
+extern GFLAGS_DLL_DECL bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern GFLAGS_DLL_DECL bool BoolFromEnv(const char *varname, bool defval);
+extern GFLAGS_DLL_DECL int32 Int32FromEnv(const char *varname, int32 defval);
+extern GFLAGS_DLL_DECL int64 Int64FromEnv(const char *varname, int64 defval);
+extern GFLAGS_DLL_DECL uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern GFLAGS_DLL_DECL double DoubleFromEnv(const char *varname, double defval);
+extern GFLAGS_DLL_DECL const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern GFLAGS_DLL_DECL void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern GFLAGS_DLL_DECL uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used.
+extern GFLAGS_DLL_DECL uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern GFLAGS_DLL_DECL void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern GFLAGS_DLL_DECL void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized.
+// Only flags registered since the last parse will be recognized.
+// Any flag value must be provided as part of the argument using "=",
+// not as a separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+extern GFLAGS_DLL_DECL uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern GFLAGS_DLL_DECL void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class GFLAGS_DLL_DECL FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ /* We always want to export defined variables, dll or no */ \
+ GFLAGS_DLL_DEFINE_FLAG type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ /* We always want to import declared variables, dll or no */ \
+ extern GFLAGS_DLL_DECLARE_FLAG type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> GFLAGS_DLL_DECL double IsBoolFlag(const From& from);
+GFLAGS_DLL_DECL bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern GFLAGS_DLL_DECLARE_FLAG ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ GFLAGS_DLL_DEFINE_FLAG clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..e97de5b
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h
@@ -0,0 +1,130 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+// Annoying stuff for windows -- makes sure clients can import these functions
+#ifndef GFLAGS_DLL_DECL
+# ifdef _WIN32
+# define GFLAGS_DLL_DECL __declspec(dllimport)
+# else
+# define GFLAGS_DLL_DECL
+# endif
+#endif
+
+namespace google {
+
+GFLAGS_DLL_DECL void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h
new file mode 100644
index 0000000..dcca757
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h
@@ -0,0 +1,139 @@
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Sometimes we accidentally #include this config.h instead of the one
+ in .. -- this is particularly true for msys/mingw, which uses the
+ unix config.h but also runs code in the windows directory.
+ */
+#ifdef __MINGW32__
+#include "../config.h"
+#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+#endif
+
+#ifndef GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#ifndef GFLAGS_DLL_DECL
+# define GFLAGS_IS_A_DLL 1 /* not set if you're statically linking */
+# define GFLAGS_DLL_DECL __declspec(dllexport)
+# define GFLAGS_DLL_DECL_FOR_UNITTESTS __declspec(dllimport)
+#endif
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#undef HAVE_DLFCN_H
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#undef HAVE_FNMATCH_H
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#undef HAVE_INTTYPES_H
+
+/* Define to 1 if you have the <memory.h> header file. */
+#undef HAVE_MEMORY_H
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#undef HAVE_PTHREAD
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#undef HAVE_SETENV
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#undef HAVE_STDINT_H
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#undef HAVE_STRINGS_H
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#undef HAVE_UNISTD_H
+
+/* define if your compiler has __attribute__ */
+#undef HAVE___ATTRIBUTE__
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#undef LT_OBJDIR
+
+/* Name of package */
+#undef PACKAGE
+
+/* Define to the address where bug reports for this package should be sent. */
+#undef PACKAGE_BUGREPORT
+
+/* Define to the full name of this package. */
+#undef PACKAGE_NAME
+
+/* Define to the full name and version of this package. */
+#undef PACKAGE_STRING
+
+/* Define to the one symbol short name of this package. */
+#undef PACKAGE_TARNAME
+
+/* Define to the home page for this package. */
+#undef PACKAGE_URL
+
+/* Define to the version of this package. */
+#undef PACKAGE_VERSION
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+#undef PTHREAD_CREATE_JOINABLE
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#undef VERSION
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
+
+// ---------------------------------------------------------------------
+// Extra stuff not found in config.h.in
+
+// This must be defined before the windows.h is included. It's needed
+// for mutex.h, to give access to the TryLock method.
+#ifndef _WIN32_WINNT
+# define _WIN32_WINNT 0x0400
+#endif
+
+// TODO(csilvers): include windows/port.h in every relevant source file instead?
+#include "windows/port.h"
+
+#endif /* GOOGLE_GFLAGS_WINDOWS_CONFIG_H_ */
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h
new file mode 100644
index 0000000..fdafe2a
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h
@@ -0,0 +1,601 @@
+// Copyright (c) 2006, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+// ---
+// Author: Ray Sidney
+// Revamped and reorganized by Craig Silverstein
+//
+// This is the file that should be included by any file which declares
+// or defines a command line flag or wants to parse command line flags
+// or print a program usage message (which will include information about
+// flags). Executive summary, in the form of an example foo.cc file:
+//
+// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
+// #include "validators.h" // hypothetical file defining ValidateIsFile()
+//
+// DEFINE_int32(end, 1000, "The last record to read");
+//
+// DEFINE_string(filename, "my_file.txt", "The file to read");
+// // Crash if the specified file does not exist.
+// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
+// &ValidateIsFile);
+//
+// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
+//
+// void MyFunc() {
+// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
+// }
+//
+// Then, at the command-line:
+// ./foo --noverbose --start=5 --end=100
+//
+// For more details, see
+// doc/gflags.html
+//
+// --- A note about thread-safety:
+//
+// We describe many functions in this routine as being thread-hostile,
+// thread-compatible, or thread-safe. Here are the meanings we use:
+//
+// thread-safe: it is safe for multiple threads to call this routine
+// (or, when referring to a class, methods of this class)
+// concurrently.
+// thread-hostile: it is not safe for multiple threads to call this
+// routine (or methods of this class) concurrently. In gflags,
+// most thread-hostile routines are intended to be called early in,
+// or even before, main() -- that is, before threads are spawned.
+// thread-compatible: it is safe for multiple threads to read from
+// this variable (when applied to variables), or to call const
+// methods of this class (when applied to classes), as long as no
+// other thread is writing to the variable or calling non-const
+// methods of this class.
+
+#ifndef GOOGLE_GFLAGS_H_
+#define GOOGLE_GFLAGS_H_
+
+#include <string>
+#include <vector>
+
+// We care a lot about number of bits things take up. Unfortunately,
+// systems define their bit-specific ints in a lot of different ways.
+// We use our own way, and have a typedef to get there.
+// Note: these commands below may look like "#if 1" or "#if 0", but
+// that's because they were constructed that way at ./configure time.
+// Look at gflags.h.in to see how they're calculated (based on your config).
+#if 0
+#include <stdint.h> // the normal place uint16_t is defined
+#endif
+#if 1
+#include <sys/types.h> // the normal place u_int16_t is defined
+#endif
+#if 0
+#include <inttypes.h> // a third place for uint16_t or u_int16_t
+#endif
+
+// Annoying stuff for windows -- makes sure clients can import these functions
+#if defined(_WIN32)
+# ifndef GFLAGS_DLL_DECL
+# define GFLAGS_DLL_DECL __declspec(dllimport)
+# endif
+# ifndef GFLAGS_DLL_DECLARE_FLAG
+# define GFLAGS_DLL_DECLARE_FLAG __declspec(dllimport)
+# endif
+# ifndef GFLAGS_DLL_DEFINE_FLAG
+# define GFLAGS_DLL_DEFINE_FLAG __declspec(dllexport)
+# endif
+#else
+# ifndef GFLAGS_DLL_DECL
+# define GFLAGS_DLL_DECL
+# endif
+# ifndef GFLAGS_DLL_DECLARE_FLAG
+# define GFLAGS_DLL_DECLARE_FLAG
+# endif
+# ifndef GFLAGS_DLL_DEFINE_FLAG
+# define GFLAGS_DLL_DEFINE_FLAG
+# endif
+#endif
+
+namespace google {
+
+#if 0 // the C99 format
+typedef int32_t int32;
+typedef uint32_t uint32;
+typedef int64_t int64;
+typedef uint64_t uint64;
+#elif 0 // the BSD format
+typedef int32_t int32;
+typedef u_int32_t uint32;
+typedef int64_t int64;
+typedef u_int64_t uint64;
+#elif 1 // the windows (vc7) format
+typedef __int32 int32;
+typedef unsigned __int32 uint32;
+typedef __int64 int64;
+typedef unsigned __int64 uint64;
+#else
+#error Do not know how to define a 32-bit integer quantity on your system
+#endif
+
+// --------------------------------------------------------------------
+// To actually define a flag in a file, use DEFINE_bool,
+// DEFINE_string, etc. at the bottom of this file. You may also find
+// it useful to register a validator with the flag. This ensures that
+// when the flag is parsed from the commandline, or is later set via
+// SetCommandLineOption, we call the validation function. It is _not_
+// called when you assign the value to the flag directly using the = operator.
+//
+// The validation function should return true if the flag value is valid, and
+// false otherwise. If the function returns false for the new setting of the
+// flag, the flag will retain its current value. If it returns false for the
+// default value, ParseCommandLineFlags() will die.
+//
+// This function is safe to call at global construct time (as in the
+// example below).
+//
+// Example use:
+// static bool ValidatePort(const char* flagname, int32 value) {
+// if (value > 0 && value < 32768) // value is ok
+// return true;
+// printf("Invalid value for --%s: %d\n", flagname, (int)value);
+// return false;
+// }
+// DEFINE_int32(port, 0, "What port to listen on");
+// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
+
+// Returns true if successfully registered, false if not (because the
+// first argument doesn't point to a command-line flag, or because a
+// validator is already registered for this flag).
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const bool* flag,
+ bool (*validate_fn)(const char*, bool));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const int32* flag,
+ bool (*validate_fn)(const char*, int32));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const int64* flag,
+ bool (*validate_fn)(const char*, int64));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const uint64* flag,
+ bool (*validate_fn)(const char*, uint64));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const double* flag,
+ bool (*validate_fn)(const char*, double));
+GFLAGS_DLL_DECL bool RegisterFlagValidator(const std::string* flag,
+ bool (*validate_fn)(const char*, const std::string&));
+
+
+// --------------------------------------------------------------------
+// These methods are the best way to get access to info about the
+// list of commandline flags. Note that these routines are pretty slow.
+// GetAllFlags: mostly-complete info about the list, sorted by file.
+// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
+// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
+//
+// In addition to accessing flags, you can also access argv[0] (the program
+// name) and argv (the entire commandline), which we sock away a copy of.
+// These variables are static, so you should only set them once.
+
+struct GFLAGS_DLL_DECL CommandLineFlagInfo {
+ std::string name; // the name of the flag
+ std::string type; // the type of the flag: int32, etc
+ std::string description; // the "help text" associated with the flag
+ std::string current_value; // the current value, as a string
+ std::string default_value; // the default value, as a string
+ std::string filename; // 'cleaned' version of filename holding the flag
+ bool has_validator_fn; // true if RegisterFlagValidator called on flag
+ bool is_default; // true if the flag has the default value and
+ // has not been set explicitly from the cmdline
+ // or via SetCommandLineOption
+};
+
+// Using this inside of a validator is a recipe for a deadlock.
+// TODO(wojtekm) Fix locking when validators are running, to make it safe to
+// call validators during ParseAllFlags.
+// Also make sure then to uncomment the corresponding unit test in
+// commandlineflags_unittest.sh
+extern GFLAGS_DLL_DECL void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
+// These two are actually defined in commandlineflags_reporting.cc.
+extern GFLAGS_DLL_DECL void ShowUsageWithFlags(const char *argv0); // what --help does
+extern GFLAGS_DLL_DECL void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
+
+// Create a descriptive string for a flag.
+// Goes to some trouble to make pretty line breaks.
+extern GFLAGS_DLL_DECL std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
+
+// Thread-hostile; meant to be called before any threads are spawned.
+extern GFLAGS_DLL_DECL void SetArgv(int argc, const char** argv);
+// The following functions are thread-safe as long as SetArgv() is
+// only called before any threads start.
+extern GFLAGS_DLL_DECL const std::vector<std::string>& GetArgvs(); // all of argv as a vector
+extern GFLAGS_DLL_DECL const char* GetArgv(); // all of argv as a string
+extern GFLAGS_DLL_DECL const char* GetArgv0(); // only argv0
+extern GFLAGS_DLL_DECL uint32 GetArgvSum(); // simple checksum of argv
+extern GFLAGS_DLL_DECL const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
+extern GFLAGS_DLL_DECL const char* ProgramInvocationShortName(); // basename(argv0)
+// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
+// called before any threads start.
+extern GFLAGS_DLL_DECL const char* ProgramUsage(); // string set by SetUsageMessage()
+
+
+// --------------------------------------------------------------------
+// Normally you access commandline flags by just saying "if (FLAGS_foo)"
+// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
+// commonly, via the DEFINE_foo macro). But if you need a bit more
+// control, we have programmatic ways to get/set the flags as well.
+// These programmatic ways to access flags are thread-safe, but direct
+// access is only thread-compatible.
+
+// Return true iff the flagname was found.
+// OUTPUT is set to the flag's value, or unchanged if we return false.
+extern GFLAGS_DLL_DECL bool GetCommandLineOption(const char* name, std::string* OUTPUT);
+
+// Return true iff the flagname was found. OUTPUT is set to the flag's
+// CommandLineFlagInfo or unchanged if we return false.
+extern GFLAGS_DLL_DECL bool GetCommandLineFlagInfo(const char* name,
+ CommandLineFlagInfo* OUTPUT);
+
+// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
+// Example usage, to check if a flag's value is currently the default value:
+// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
+extern GFLAGS_DLL_DECL CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
+
+enum GFLAGS_DLL_DECL FlagSettingMode {
+ // update the flag's value (can call this multiple times).
+ SET_FLAGS_VALUE,
+ // update the flag's value, but *only if* it has not yet been updated
+ // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
+ SET_FLAG_IF_DEFAULT,
+ // set the flag's default value to this. If the flag has not yet updated
+ // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
+ // change the flag's current value to the new default value as well.
+ SET_FLAGS_DEFAULT
+};
+
+// Set a particular flag ("command line option"). Returns a string
+// describing the new value that the option has been set to. The
+// return value API is not well-specified, so basically just depend on
+// it to be empty if the setting failed for some reason -- the name is
+// not a valid flag name, or the value is not a valid value -- and
+// non-empty else.
+
+// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
+extern GFLAGS_DLL_DECL std::string SetCommandLineOption(const char* name, const char* value);
+extern GFLAGS_DLL_DECL std::string SetCommandLineOptionWithMode(const char* name, const char* value,
+ FlagSettingMode set_mode);
+
+
+// --------------------------------------------------------------------
+// Saves the states (value, default value, whether the user has set
+// the flag, registered validators, etc) of all flags, and restores
+// them when the FlagSaver is destroyed. This is very useful in
+// tests, say, when you want to let your tests change the flags, but
+// make sure that they get reverted to the original states when your
+// test is complete.
+//
+// Example usage:
+// void TestFoo() {
+// FlagSaver s1;
+// FLAG_foo = false;
+// FLAG_bar = "some value";
+//
+// // test happens here. You can return at any time
+// // without worrying about restoring the FLAG values.
+// }
+//
+// Note: This class is marked with __attribute__((unused)) because all the
+// work is done in the constructor and destructor, so in the standard
+// usage example above, the compiler would complain that it's an
+// unused variable.
+//
+// This class is thread-safe.
+
+class GFLAGS_DLL_DECL FlagSaver {
+ public:
+ FlagSaver();
+ ~FlagSaver();
+
+ private:
+ class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
+
+ FlagSaver(const FlagSaver&); // no copying!
+ void operator=(const FlagSaver&);
+} ;
+
+// --------------------------------------------------------------------
+// Some deprecated or hopefully-soon-to-be-deprecated functions.
+
+// This is often used for logging. TODO(csilvers): figure out a better way
+extern GFLAGS_DLL_DECL std::string CommandlineFlagsIntoString();
+// Usually where this is used, a FlagSaver should be used instead.
+extern GFLAGS_DLL_DECL bool ReadFlagsFromString(const std::string& flagfilecontents,
+ const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+// These let you manually implement --flagfile functionality.
+// DEPRECATED.
+extern GFLAGS_DLL_DECL bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
+extern GFLAGS_DLL_DECL bool SaveCommandFlags(); // actually defined in google.cc !
+extern GFLAGS_DLL_DECL bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
+ bool errors_are_fatal); // uses SET_FLAGS_VALUE
+
+
+// --------------------------------------------------------------------
+// Useful routines for initializing flags from the environment.
+// In each case, if 'varname' does not exist in the environment
+// return defval. If 'varname' does exist but is not valid
+// (e.g., not a number for an int32 flag), abort with an error.
+// Otherwise, return the value. NOTE: for booleans, for true use
+// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
+
+extern GFLAGS_DLL_DECL bool BoolFromEnv(const char *varname, bool defval);
+extern GFLAGS_DLL_DECL int32 Int32FromEnv(const char *varname, int32 defval);
+extern GFLAGS_DLL_DECL int64 Int64FromEnv(const char *varname, int64 defval);
+extern GFLAGS_DLL_DECL uint64 Uint64FromEnv(const char *varname, uint64 defval);
+extern GFLAGS_DLL_DECL double DoubleFromEnv(const char *varname, double defval);
+extern GFLAGS_DLL_DECL const char *StringFromEnv(const char *varname, const char *defval);
+
+
+// --------------------------------------------------------------------
+// The next two functions parse commandlineflags from main():
+
+// Set the "usage" message for this program. For example:
+// string usage("This program does nothing. Sample usage:\n");
+// usage += argv[0] + " <uselessarg1> <uselessarg2>";
+// SetUsageMessage(usage);
+// Do not include commandline flags in the usage: we do that for you!
+// Thread-hostile; meant to be called before any threads are spawned.
+extern GFLAGS_DLL_DECL void SetUsageMessage(const std::string& usage);
+
+// Looks for flags in argv and parses them. Rearranges argv to put
+// flags first, or removes them entirely if remove_flags is true.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used.
+// See top-of-file for more details on this function.
+#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
+extern GFLAGS_DLL_DECL uint32 ParseCommandLineFlags(int *argc, char*** argv,
+ bool remove_flags);
+#endif
+
+
+// Calls to ParseCommandLineNonHelpFlags and then to
+// HandleCommandLineHelpFlags can be used instead of a call to
+// ParseCommandLineFlags during initialization, in order to allow for
+// changing default values for some FLAGS (via
+// e.g. SetCommandLineOptionWithMode calls) between the time of
+// command line parsing and the time of dumping help information for
+// the flags as a result of command line parsing.
+// If a flag is defined more than once in the command line or flag
+// file, the last definition is used.
+extern GFLAGS_DLL_DECL uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
+ bool remove_flags);
+// This is actually defined in commandlineflags_reporting.cc.
+// This function is misnamed (it also handles --version, etc.), but
+// it's too late to change that now. :-(
+extern GFLAGS_DLL_DECL void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
+
+// Allow command line reparsing. Disables the error normally
+// generated when an unknown flag is found, since it may be found in a
+// later parse. Thread-hostile; meant to be called before any threads
+// are spawned.
+extern GFLAGS_DLL_DECL void AllowCommandLineReparsing();
+
+// Reparse the flags that have not yet been recognized.
+// Only flags registered since the last parse will be recognized.
+// Any flag value must be provided as part of the argument using "=",
+// not as a separate command line argument that follows the flag argument.
+// Intended for handling flags from dynamically loaded libraries,
+// since their flags are not registered until they are loaded.
+extern GFLAGS_DLL_DECL uint32 ReparseCommandLineNonHelpFlags();
+
+// Clean up memory allocated by flags. This is only needed to reduce
+// the quantity of "potentially leaked" reports emitted by memory
+// debugging tools such as valgrind. It is not required for normal
+// operation, or for the perftools heap-checker. It must only be called
+// when the process is about to exit, and all threads that might
+// access flags are quiescent. Referencing flags after this is called
+// will have unexpected consequences. This is not safe to run when
+// multiple threads might be running: the function is thread-hostile.
+extern GFLAGS_DLL_DECL void ShutDownCommandLineFlags();
+
+
+// --------------------------------------------------------------------
+// Now come the command line flag declaration/definition macros that
+// will actually be used. They're kind of hairy. A major reason
+// for this is initialization: we want people to be able to access
+// variables in global constructors and have that not crash, even if
+// their global constructor runs before the global constructor here.
+// (Obviously, we can't guarantee the flags will have the correct
+// default value in that case, but at least accessing them is safe.)
+// The only way to do that is have flags point to a static buffer.
+// So we make one, using a union to ensure proper alignment, and
+// then use placement-new to actually set up the flag with the
+// correct default value. In the same vein, we have to worry about
+// flag access in global destructors, so FlagRegisterer has to be
+// careful never to destroy the flag-values it constructs.
+//
+// Note that when we define a flag variable FLAGS_<name>, we also
+// preemptively define a junk variable, FLAGS_no<name>. This is to
+// cause a link-time error if someone tries to define 2 flags with
+// names like "logging" and "nologging". We do this because a bool
+// flag FLAG can be set from the command line to true with a "-FLAG"
+// argument, and to false with a "-noFLAG" argument, and so this can
+// potentially avert confusion.
+//
+// We also put flags into their own namespace. It is purposefully
+// named in an opaque way that people should have trouble typing
+// directly. The idea is that DEFINE puts the flag in the weird
+// namespace, and DECLARE imports the flag from there into the current
+// namespace. The net result is to force people to use DECLARE to get
+// access to a flag, rather than saying "extern bool FLAGS_whatever;"
+// or some such instead. We want this so we can put extra
+// functionality (like sanity-checking) in DECLARE if we want, and
+// make sure it is picked up everywhere.
+//
+// We also put the type of the variable in the namespace, so that
+// people can't DECLARE_int32 something that they DEFINE_bool'd
+// elsewhere.
+
+class GFLAGS_DLL_DECL FlagRegisterer {
+ public:
+ FlagRegisterer(const char* name, const char* type,
+ const char* help, const char* filename,
+ void* current_storage, void* defvalue_storage);
+};
+
+extern bool FlagsTypeWarn(const char *name);
+
+// If your application #defines STRIP_FLAG_HELP to a non-zero value
+// before #including this file, we remove the help message from the
+// binary file. This can reduce the size of the resulting binary
+// somewhat, and may also be useful for security reasons.
+
+extern const char kStrippedFlagHelp[];
+
+}
+
+#ifndef SWIG // In swig, ignore the main flag declarations
+
+#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
+// Need this construct to avoid the 'defined but not used' warning.
+#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : kStrippedFlagHelp)
+#else
+#define MAYBE_STRIPPED_HELP(txt) txt
+#endif
+
+// Each command-line flag has two variables associated with it: one
+// with the current value, and one with the default value. However,
+// we have a third variable, which is where value is assigned; it's a
+// constant. This guarantees that FLAG_##value is initialized at
+// static initialization time (e.g. before program-start) rather than
+// than global construction time (which is after program-start but
+// before main), at least when 'value' is a compile-time constant. We
+// use a small trick for the "default value" variable, and call it
+// FLAGS_no<name>. This serves the second purpose of assuring a
+// compile error if someone tries to define a flag named no<name>
+// which is illegal (--foo and --nofoo both affect the "foo" flag).
+#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
+ namespace fL##shorttype { \
+ static const type FLAGS_nono##name = value; \
+ /* We always want to export defined variables, dll or no */ \
+ GFLAGS_DLL_DEFINE_FLAG type FLAGS_##name = FLAGS_nono##name; \
+ type FLAGS_no##name = FLAGS_nono##name; \
+ static ::google::FlagRegisterer o_##name( \
+ #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
+ &FLAGS_##name, &FLAGS_no##name); \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+#define DECLARE_VARIABLE(type, shorttype, name) \
+ namespace fL##shorttype { \
+ /* We always want to import declared variables, dll or no */ \
+ extern GFLAGS_DLL_DECLARE_FLAG type FLAGS_##name; \
+ } \
+ using fL##shorttype::FLAGS_##name
+
+// For DEFINE_bool, we want to do the extra check that the passed-in
+// value is actually a bool, and not a string or something that can be
+// coerced to a bool. These declarations (no definition needed!) will
+// help us do that, and never evaluate From, which is important.
+// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
+// that the compiler have different sizes for bool & double. Since
+// this is not guaranteed by the standard, we check it with a
+// compile-time assert (msg[-1] will give a compile-time error).
+namespace fLB {
+struct CompileAssert {};
+typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
+ (sizeof(double) != sizeof(bool)) ? 1 : -1];
+template<typename From> GFLAGS_DLL_DECL double IsBoolFlag(const From& from);
+GFLAGS_DLL_DECL bool IsBoolFlag(bool from);
+} // namespace fLB
+
+#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
+#define DEFINE_bool(name, val, txt) \
+ namespace fLB { \
+ typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
+ (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
+ } \
+ DEFINE_VARIABLE(bool, B, name, val, txt)
+
+#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
+#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
+
+#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
+#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
+
+#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
+#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
+
+#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
+#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
+
+// Strings are trickier, because they're not a POD, so we can't
+// construct them at static-initialization time (instead they get
+// constructed at global-constructor time, which is much later). To
+// try to avoid crashes in that case, we use a char buffer to store
+// the string, which we can static-initialize, and then placement-new
+// into it later. It's not perfect, but the best we can do.
+
+namespace fLS {
+// The meaning of "string" might be different between now and when the
+// macros below get invoked (e.g., if someone is experimenting with
+// other string implementations that get defined after this file is
+// included). Save the current meaning now and use it in the macros.
+typedef std::string clstring;
+
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const char *value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ const clstring &value) {
+ return new(stringspot) clstring(value);
+}
+inline clstring* dont_pass0toDEFINE_string(char *stringspot,
+ int value);
+} // namespace fLS
+
+#define DECLARE_string(name) namespace fLS { extern GFLAGS_DLL_DECLARE_FLAG ::fLS::clstring& FLAGS_##name; } \
+ using fLS::FLAGS_##name
+
+// We need to define a var named FLAGS_no##name so people don't define
+// --string and --nostring. And we need a temporary place to put val
+// so we don't have to evaluate it twice. Two great needs that go
+// great together!
+#define DEFINE_string(name, val, txt) \
+ namespace fLS { \
+ using ::fLS::clstring; \
+ static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
+ clstring* const FLAGS_no##name = ::fLS:: \
+ dont_pass0toDEFINE_string(s_##name[0].s, \
+ val); \
+ static ::google::FlagRegisterer o_##name( \
+ #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
+ s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
+ GFLAGS_DLL_DEFINE_FLAG clstring& FLAGS_##name = *FLAGS_no##name; \
+ } \
+ using fLS::FLAGS_##name
+
+#endif // SWIG
+
+#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h
new file mode 100644
index 0000000..e97de5b
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h
@@ -0,0 +1,130 @@
+// Copyright (c) 2008, Google Inc.
+// All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+//
+// ---
+// Author: Dave Nicponski
+//
+// Implement helpful bash-style command line flag completions
+//
+// ** Functional API:
+// HandleCommandLineCompletions() should be called early during
+// program startup, but after command line flag code has been
+// initialized, such as the beginning of HandleCommandLineHelpFlags().
+// It checks the value of the flag --tab_completion_word. If this
+// flag is empty, nothing happens here. If it contains a string,
+// however, then HandleCommandLineCompletions() will hijack the
+// process, attempting to identify the intention behind this
+// completion. Regardless of the outcome of this deduction, the
+// process will be terminated, similar to --helpshort flag
+// handling.
+//
+// ** Overview of Bash completions:
+// Bash can be told to programatically determine completions for the
+// current 'cursor word'. It does this by (in this case) invoking a
+// command with some additional arguments identifying the command
+// being executed, the word being completed, and the previous word
+// (if any). Bash then expects a sequence of output lines to be
+// printed to stdout. If these lines all contain a common prefix
+// longer than the cursor word, bash will replace the cursor word
+// with that common prefix, and display nothing. If there isn't such
+// a common prefix, bash will display the lines in pages using 'more'.
+//
+// ** Strategy taken for command line completions:
+// If we can deduce either the exact flag intended, or a common flag
+// prefix, we'll output exactly that. Otherwise, if information
+// must be displayed to the user, we'll take the opportunity to add
+// some helpful information beyond just the flag name (specifically,
+// we'll include the default flag value and as much of the flag's
+// description as can fit on a single terminal line width, as specified
+// by the flag --tab_completion_columns). Furthermore, we'll try to
+// make bash order the output such that the most useful or relevent
+// flags are the most likely to be shown at the top.
+//
+// ** Additional features:
+// To assist in finding that one really useful flag, substring matching
+// was implemented. Before pressing a <TAB> to get completion for the
+// current word, you can append one or more '?' to the flag to do
+// substring matching. Here's the semantics:
+// --foo<TAB> Show me all flags with names prefixed by 'foo'
+// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
+// --foo??<TAB> Same as prior case, but also search in module
+// definition path for 'foo'
+// --foo???<TAB> Same as prior case, but also search in flag
+// descriptions for 'foo'
+// Finally, we'll trim the output to a relatively small number of
+// flags to keep bash quiet about the verbosity of output. If one
+// really wanted to see all possible matches, appending a '+' to the
+// search word will force the exhaustive list of matches to be printed.
+//
+// ** How to have bash accept completions from a binary:
+// Bash requires that it be informed about each command that programmatic
+// completion should be enabled for. Example addition to a .bashrc
+// file would be (your path to gflags_completions.sh file may differ):
+
+/*
+$ complete -o bashdefault -o default -o nospace -C \
+ '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
+ time env binary_name another_binary [...]
+*/
+
+// This would allow the following to work:
+// $ /path/to/binary_name --vmodule<TAB>
+// Or:
+// $ ./bin/path/another_binary --gfs_u<TAB>
+// (etc)
+//
+// Sadly, it appears that bash gives no easy way to force this behavior for
+// all commands. That's where the "time" in the above example comes in.
+// If you haven't specifically added a command to the list of completion
+// supported commands, you can still get completions by prefixing the
+// entire command with "env".
+// $ env /some/brand/new/binary --vmod<TAB>
+// Assuming that "binary" is a newly compiled binary, this should still
+// produce the expected completion output.
+
+
+#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
+#define GOOGLE_GFLAGS_COMPLETIONS_H_
+
+// Annoying stuff for windows -- makes sure clients can import these functions
+#ifndef GFLAGS_DLL_DECL
+# ifdef _WIN32
+# define GFLAGS_DLL_DECL __declspec(dllimport)
+# else
+# define GFLAGS_DLL_DECL
+# endif
+#endif
+
+namespace google {
+
+GFLAGS_DLL_DECL void HandleCommandLineCompletions(void);
+
+}
+
+#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h
new file mode 100644
index 0000000..dcca757
--- /dev/null
+++ b/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h
@@ -0,0 +1,139 @@
+/* src/config.h.in. Generated from configure.ac by autoheader. */
+
+/* Sometimes we accidentally #include this config.h instead of the one
+ in .. -- this is particularly true for msys/mingw, which uses the
+ unix config.h but also runs code in the windows directory.
+ */
+#ifdef __MINGW32__
+#include "../config.h"
+#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+#endif
+
+#ifndef GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
+
+/* Always the empty-string on non-windows systems. On windows, should be
+ "__declspec(dllexport)". This way, when we compile the dll, we export our
+ functions/classes. It's safe to define this here because config.h is only
+ used internally, to compile the DLL, and every DLL source file #includes
+ "config.h" before anything else. */
+#ifndef GFLAGS_DLL_DECL
+# define GFLAGS_IS_A_DLL 1 /* not set if you're statically linking */
+# define GFLAGS_DLL_DECL __declspec(dllexport)
+# define GFLAGS_DLL_DECL_FOR_UNITTESTS __declspec(dllimport)
+#endif
+
+/* Namespace for Google classes */
+#define GOOGLE_NAMESPACE ::google
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#undef HAVE_DLFCN_H
+
+/* Define to 1 if you have the <fnmatch.h> header file. */
+#undef HAVE_FNMATCH_H
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#undef HAVE_INTTYPES_H
+
+/* Define to 1 if you have the <memory.h> header file. */
+#undef HAVE_MEMORY_H
+
+/* define if the compiler implements namespaces */
+#define HAVE_NAMESPACES 1
+
+/* Define if you have POSIX threads libraries and header files. */
+#undef HAVE_PTHREAD
+
+/* Define to 1 if you have the `putenv' function. */
+#define HAVE_PUTENV 1
+
+/* Define to 1 if you have the `setenv' function. */
+#undef HAVE_SETENV
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#undef HAVE_STDINT_H
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#define HAVE_STDLIB_H 1
+
+/* Define to 1 if you have the <strings.h> header file. */
+#undef HAVE_STRINGS_H
+
+/* Define to 1 if you have the <string.h> header file. */
+#define HAVE_STRING_H 1
+
+/* Define to 1 if you have the `strtoll' function. */
+#define HAVE_STRTOLL 1
+
+/* Define to 1 if you have the `strtoq' function. */
+#define HAVE_STRTOQ 1
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#define HAVE_SYS_STAT_H 1
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#define HAVE_SYS_TYPES_H 1
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#undef HAVE_UNISTD_H
+
+/* define if your compiler has __attribute__ */
+#undef HAVE___ATTRIBUTE__
+
+/* Define to the sub-directory in which libtool stores uninstalled libraries.
+ */
+#undef LT_OBJDIR
+
+/* Name of package */
+#undef PACKAGE
+
+/* Define to the address where bug reports for this package should be sent. */
+#undef PACKAGE_BUGREPORT
+
+/* Define to the full name of this package. */
+#undef PACKAGE_NAME
+
+/* Define to the full name and version of this package. */
+#undef PACKAGE_STRING
+
+/* Define to the one symbol short name of this package. */
+#undef PACKAGE_TARNAME
+
+/* Define to the home page for this package. */
+#undef PACKAGE_URL
+
+/* Define to the version of this package. */
+#undef PACKAGE_VERSION
+
+/* Define to necessary symbol if this constant uses a non-standard name on
+ your system. */
+#undef PTHREAD_CREATE_JOINABLE
+
+/* Define to 1 if you have the ANSI C header files. */
+#define STDC_HEADERS 1
+
+/* the namespace where STL code like vector<> is defined */
+#define STL_NAMESPACE std
+
+/* Version number of package */
+#undef VERSION
+
+/* Stops putting the code inside the Google namespace */
+#define _END_GOOGLE_NAMESPACE_ }
+
+/* Puts following code inside the Google namespace */
+#define _START_GOOGLE_NAMESPACE_ namespace google {
+
+// ---------------------------------------------------------------------
+// Extra stuff not found in config.h.in
+
+// This must be defined before the windows.h is included. It's needed
+// for mutex.h, to give access to the TryLock method.
+#ifndef _WIN32_WINNT
+# define _WIN32_WINNT 0x0400
+#endif
+
+// TODO(csilvers): include windows/port.h in every relevant source file instead?
+#include "windows/port.h"
+
+#endif /* GOOGLE_GFLAGS_WINDOWS_CONFIG_H_ */
diff --git a/trunk/third_party/google-gflags/google-gflags.gyp b/trunk/third_party/google-gflags/google-gflags.gyp
new file mode 100644
index 0000000..bd7a3ed
--- /dev/null
+++ b/trunk/third_party/google-gflags/google-gflags.gyp
@@ -0,0 +1,62 @@
+# Copyright 2011 Google Inc.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+{
+ 'variables': {
+ 'gflags_root': '<(DEPTH)/third_party/google-gflags',
+ 'gflags_gen_arch_root': '<(gflags_root)/gen/arch/<(OS)/<(target_arch)',
+ },
+ 'targets': [
+ {
+ 'target_name': 'google-gflags',
+ 'type': '<(library)',
+ 'include_dirs': [
+ '<(gflags_gen_arch_root)/include/private', # For config.h
+ '<(gflags_gen_arch_root)/include', # For configured files.
+ '<(gflags_root)/src', # For everything else.
+ ],
+ 'defines': [
+ # These macros exist so flags and symbols are properly
+ # exported when building DLLs. Since we don't build DLLs, we
+ # need to disable them.
+ 'GFLAGS_DLL_DECL=',
+ 'GFLAGS_DLL_DECLARE_FLAG=',
+ 'GFLAGS_DLL_DEFINE_FLAG=',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '<(gflags_gen_arch_root)/include', # For configured files.
+ '<(gflags_root)/src', # For everything else.
+ ],
+ 'defines': [
+ 'GFLAGS_DLL_DECL=',
+ 'GFLAGS_DLL_DECLARE_FLAG=',
+ 'GFLAGS_DLL_DEFINE_FLAG=',
+ ],
+ },
+ 'sources': [
+ 'src/gflags.cc',
+ 'src/gflags_completions.cc',
+ 'src/gflags_reporting.cc',
+ ],
+ 'conditions': [
+ ['OS == "win"', {
+ 'sources': [
+ 'src/windows/port.cc',
+ ],
+ }],
+ ],
+ },
+ ],
+}
diff --git a/trunk/third_party_mods/ace/LICENSE b/trunk/third_party_mods/ace/LICENSE
new file mode 100644
index 0000000..9204394
--- /dev/null
+++ b/trunk/third_party_mods/ace/LICENSE
@@ -0,0 +1,66 @@
+Copyright and Licensing Information for ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM),
+and CoSMIC(TM)
+
+ACE(TM), TAO(TM), CIAO(TM), DAnCE>(TM), and CoSMIC(TM) (henceforth referred to
+as "DOC software") are copyrighted by Douglas C. Schmidt and his research
+group at Washington University, University of California, Irvine, and
+Vanderbilt University, Copyright (c) 1993-2009, all rights reserved. Since DOC
+software is open-source, freely available software, you are free to use,
+modify, copy, and distribute--perpetually and irrevocably--the DOC software
+source code and object code produced from the source, as well as copy and
+distribute modified versions of this software. You must, however, include this
+copyright statement along with any code built using DOC software that you
+release. No copyright statement needs to be provided if you just ship binary
+executables of your software products.
+You can use DOC software in commercial and/or binary software releases and are
+under no obligation to redistribute any of your source code that is built
+using DOC software. Note, however, that you may not misappropriate the DOC
+software code, such as copyrighting it yourself or claiming authorship of the
+DOC software code, in a way that will prevent DOC software from being
+distributed freely using an open-source development model. You needn't inform
+anyone that you're using DOC software in your software, though we encourage
+you to let us know so we can promote your project in the DOC software success
+stories.
+
+The ACE, TAO, CIAO, DAnCE, and CoSMIC web sites are maintained by the DOC
+Group at the Institute for Software Integrated Systems (ISIS) and the Center
+for Distributed Object Computing of Washington University, St. Louis for the
+development of open-source software as part of the open-source software
+community. Submissions are provided by the submitter ``as is'' with no
+warranties whatsoever, including any warranty of merchantability,
+noninfringement of third party intellectual property, or fitness for any
+particular purpose. In no event shall the submitter be liable for any direct,
+indirect, special, exemplary, punitive, or consequential damages, including
+without limitation, lost profits, even if advised of the possibility of such
+damages. Likewise, DOC software is provided as is with no warranties of any
+kind, including the warranties of design, merchantability, and fitness for a
+particular purpose, noninfringement, or arising from a course of dealing,
+usage or trade practice. Washington University, UC Irvine, Vanderbilt
+University, their employees, and students shall have no liability with respect
+to the infringement of copyrights, trade secrets or any patents by DOC
+software or any part thereof. Moreover, in no event will Washington
+University, UC Irvine, or Vanderbilt University, their employees, or students
+be liable for any lost revenue or profits or other special, indirect and
+consequential damages.
+
+DOC software is provided with no support and without any obligation on the
+part of Washington University, UC Irvine, Vanderbilt University, their
+employees, or students to assist in its use, correction, modification, or
+enhancement. A number of companies around the world provide commercial support
+for DOC software, however. DOC software is Y2K-compliant, as long as the
+underlying OS platform is Y2K-compliant. Likewise, DOC software is compliant
+with the new US daylight savings rule passed by Congress as "The Energy Policy
+Act of 2005," which established new daylight savings times (DST) rules for the
+United States that expand DST as of March 2007. Since DOC software obtains
+time/date and calendaring information from operating systems users will not be
+affected by the new DST rules as long as they upgrade their operating systems
+accordingly.
+
+The names ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), CoSMIC(TM), Washington
+University, UC Irvine, and Vanderbilt University, may not be used to endorse
+or promote products or services derived from this source without express
+written permission from Washington University, UC Irvine, or Vanderbilt
+University. This license grants no permission to call products or services
+derived from this source ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), or CoSMIC(TM),
+nor does it grant permission for the name Washington University, UC Irvine, or
+Vanderbilt University to appear in their names.
\ No newline at end of file
diff --git a/trunk/third_party_mods/chromium/LICENSE b/trunk/third_party_mods/chromium/LICENSE
new file mode 100644
index 0000000..8dc3504
--- /dev/null
+++ b/trunk/third_party_mods/chromium/LICENSE
@@ -0,0 +1,27 @@
+// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/third_party_mods/mslpl/LICENSE b/trunk/third_party_mods/mslpl/LICENSE
new file mode 100644
index 0000000..9a3d932
--- /dev/null
+++ b/trunk/third_party_mods/mslpl/LICENSE
@@ -0,0 +1,64 @@
+This license governs use of code marked as sample or example available on
+this web site without a license agreement, as provided under the section above
+titled NOTICE SPECIFIC TO SOFTWARE AVAILABLE ON THIS WEB SITE. If you use
+such code (the software), you accept this license. If you do not accept the
+license, do not use the software.
+
+1. Definitions
+
+The terms reproduce, reproduction, derivative works, and distribution
+have the same meaning here as under U.S. copyright law.
+
+A contribution is the original software, or any additions or changes to the
+software.
+
+A contributor is any person that distributes its contribution under this
+license.
+
+Licensed patents are a contributors patent claims that read directly on its
+contribution.
+
+2. Grant of Rights
+
+(A) Copyright Grant - Subject to the terms of this license, including the
+license conditions and limitations in section 3, each contributor grants you a
+non-exclusive, worldwide, royalty-free copyright license to reproduce its
+contribution, prepare derivative works of its contribution, and distribute its
+contribution or any derivative works that you create.
+
+(B) Patent Grant - Subject to the terms of this license, including the license
+conditions and limitations in section 3, each contributor grants you a
+non-exclusive, worldwide, royalty-free license under its licensed patents to
+make, have made, use, sell, offer for sale, import, and/or otherwise dispose
+of its contribution in the software or derivative works of the contribution in
+the software.
+
+3. Conditions and Limitations
+
+(A) No Trademark License- This license does not grant you rights to use any
+contributors name, logo, or trademarks.
+
+(B) If you bring a patent claim against any contributor over patents that you
+claim are infringed by the software, your patent license from such contributor
+to the software ends automatically.
+
+(C) If you distribute any portion of the software, you must retain all
+copyright, patent, trademark, and attribution notices that are present in the
+software.
+
+(D) If you distribute any portion of the software in source code form, you may
+do so only under this license by including a complete copy of this license
+with your distribution. If you distribute any portion of the software in
+compiled or object code form, you may only do so under a license that complies
+with this license.
+
+(E) The software is licensed as-is. You bear the risk of using it. The
+contributors give no express warranties, guarantees or conditions. You may
+have additional consumer rights under your local laws which this license
+cannot change. To the extent permitted under your local laws, the contributors
+exclude the implied warranties of merchantability, fitness for a particular
+purpose and non-infringement.
+
+(F) Platform Limitation - The licenses granted in sections 2(A) and 2(B)
+extend only to the software or derivative works that you create that run on a
+Microsoft Windows operating system product.
diff --git a/trunk/third_party_mods/sqrt_floor/LICENSE b/trunk/third_party_mods/sqrt_floor/LICENSE
new file mode 100644
index 0000000..e24dfe7
--- /dev/null
+++ b/trunk/third_party_mods/sqrt_floor/LICENSE
@@ -0,0 +1,26 @@
+The following email record is related to source files spl_sqrt_floor.c
+and spl_sqrt_floor.s in trunk/src/common_audio/signal_processing/.
+
+
+From: Wilco Dijkstra <Wilco.Dijkstra@ntlworld.com>
+Date: Fri, Jun 24, 2011 at 3:20 AM
+Subject: Re: sqrt routine
+To: Kevin Ma <kma@google.com>
+Hi Kevin,
+Thanks for asking. Those routines are public domain (originally posted to
+comp.sys.arm a long time ago), so you can use them freely for any purpose.
+Cheers,
+Wilco
+
+----- Original Message -----
+From: "Kevin Ma" <kma@google.com>
+To: <Wilco.Dijkstra@ntlworld.com>
+Sent: Thursday, June 23, 2011 11:44 PM
+Subject: Fwd: sqrt routine
+Hi Wilco,
+I saw your sqrt routine from several web sites, including
+http://www.finesse.demon.co.uk/steven/sqrt.html.
+Just wonder if there's any copyright information with your Successive
+approximation routines, or if I can freely use it for any purpose.
+Thanks.
+Kevin
diff --git a/trunk/tools/continuous_build/OWNERS b/trunk/tools/continuous_build/OWNERS
new file mode 100644
index 0000000..60e166c
--- /dev/null
+++ b/trunk/tools/continuous_build/OWNERS
@@ -0,0 +1,3 @@
+ivinnichenko@webrtc.org
+phoglund@webrtc.org
+kjellander@webrtc.org
diff --git a/trunk/tools/continuous_build/clean_old_files.py b/trunk/tools/continuous_build/clean_old_files.py
new file mode 100755
index 0000000..e87526e
--- /dev/null
+++ b/trunk/tools/continuous_build/clean_old_files.py
@@ -0,0 +1,134 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = "ivinnichenko@webrtc.org (Illya Vinnichenko)"
+
+"""This script will prune sufficiently old files and empty directories.
+
+ The algorithm is to look into the provided directory and delete any files
+ that is older than x days, recursively. Then all empty directories will be
+ deleted (we can't look at timestamps there since the act of deleting a file
+ will refresh the directory's timestamp).
+
+ Note: This script has only been tested on Linux.
+"""
+
+from optparse import OptionParser
+import os
+import sys
+import time
+
+# The path is considered whitelisted if any of these entries appear
+# at some point in the path
+WHITELIST = ["buildbot.tac", "master.cfg", "public_html", "changes.pck",
+ "webrtc_buildbot"]
+
+
+def is_whitelisted(path):
+ """Check if file is whitelisted.
+
+ Args:
+ path: file path.
+ """
+ for entry in WHITELIST:
+ if entry in path:
+ return True
+ return False
+
+
+def delete_directory(directory):
+ try:
+ os.rmdir(directory)
+ return True
+ except OSError as exception:
+ # The directory probably contains newer files.
+ print "Could not remove directory %s: reason %s." % (directory, exception)
+ return False
+
+
+def delete_file(file):
+ try:
+ os.remove(file)
+ except OSError as exception:
+ print "Unexpectedly failed to remove file %s: reason %s." % (file,
+ exception)
+
+
+def log_removal(file_or_directory, time_stamp, verbose):
+ if verbose:
+ str_stamp = time.strftime("%a, %d %b %Y %H:%M:%S +0000",
+ time.gmtime(time_stamp))
+ print "Removing [%s], stamped on %s" % (file_or_directory, str_stamp)
+
+
+def remove_old_files_and_directories(path, num_days, verbose, skip_dirs):
+ """Removes all files under path that are older than num_days days.
+ The algorithm also tried to delete all directories, except for those who
+ contain files that are sufficiently new.
+
+ Implementation note: it doesn't make sense to look at timestamps for
+ directories since their timestamps are updated when a file is deleted.
+
+ Args:
+ path: The starting point.
+ num_days: days limit for removal.
+ verbose: print every cmd?
+ """
+ current_time = time.time()
+ limit = 60 * 60 * 24 * num_days
+
+ # Walk bottom-up so directories are deleted in the right order.
+ for root, directories, files in os.walk(path, topdown=False):
+ for file in files:
+ current_file = os.path.join(root, file)
+ time_stamp = os.stat(current_file).st_mtime
+
+ if is_whitelisted(current_file):
+ continue
+
+ if (current_time - time_stamp) > limit:
+ delete_file(current_file)
+ log_removal(current_file, time_stamp, verbose)
+
+ if not skip_dirs:
+ for directory in directories:
+ current_directory = os.path.join(root, directory)
+ time_stamp = os.stat(current_directory).st_mtime
+ if delete_directory(current_directory):
+ log_removal(current_directory, time_stamp, verbose)
+
+
+def main():
+ usage = "usage: %prog -p <base path> -n <number of days> [-q] [-d]"
+ parser = OptionParser(usage)
+ parser.add_option("-p", "--path", dest="cleanup_path", help="base directory")
+ parser.add_option("-n", "--num_days", dest="num_days", help="number of days")
+ parser.add_option("-q", "--quiet",
+ action="store_false", dest="verbose", default=True,
+ help="don't print status messages to stdout")
+ parser.add_option("-d", "--delete-dirs-too",
+ action="store_false", dest="skip_dirs", default=True,
+ help="number of days")
+
+ options, args = parser.parse_args()
+ if not options.cleanup_path:
+ print "You must specify base directory"
+ sys.exit(2)
+ if not options.num_days:
+ print "You must specify number of days old"
+ sys.exit(2)
+
+ if options.verbose:
+ print "Cleaning up everything in %s older than %s days" % (
+ options.cleanup_path, options.num_days)
+ remove_old_files_and_directories(options.cleanup_path, int(options.num_days),
+ options.verbose, options.skip_dirs)
+
+if __name__ == "__main__":
+ main()
diff --git a/trunk/tools/continuous_build/clean_third_party_gcda.sh b/trunk/tools/continuous_build/clean_third_party_gcda.sh
new file mode 100755
index 0000000..38547d8
--- /dev/null
+++ b/trunk/tools/continuous_build/clean_third_party_gcda.sh
@@ -0,0 +1,13 @@
+#!/bin/sh
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+#
+
+# This script removes all .gcda files from third_party in order to work around
+# a bug in LCOV (this should also increase the bot speed).
+find . -name "*.gcda" -path "*/third_party/*" | xargs rm -f
diff --git a/trunk/tools/continuous_build/generate_coverage_html.sh b/trunk/tools/continuous_build/generate_coverage_html.sh
new file mode 100755
index 0000000..6b4c175
--- /dev/null
+++ b/trunk/tools/continuous_build/generate_coverage_html.sh
@@ -0,0 +1,18 @@
+#!/bin/sh
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+#
+
+# Generates a LCOV error report and makes the results readable to all.
+
+genhtml $1 --output-directory $2
+if [ "$?" -ne "0" ]; then
+ exit 1
+fi
+
+chmod -R 777 $2
diff --git a/trunk/tools/continuous_build/master.cfg b/trunk/tools/continuous_build/master.cfg
new file mode 100755
index 0000000..aaca7ff
--- /dev/null
+++ b/trunk/tools/continuous_build/master.cfg
@@ -0,0 +1,414 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'ivinnichenko@webrtc.org (Illya Vinnichenko)'
+
+# This is a buildmaster config file for WebRTC project. It must be installed as
+# 'master.cfg' in your buildmaster's base directory (although the filename
+# can be changed with the --basedir option to 'mktap buildbot master').
+
+# It has one job: define a dictionary named BuildmasterConfig. This
+# dictionary has a variety of keys to control different aspects of the
+# buildmaster. They are documented in docs/config.xhtml .
+
+# This is the dictionary that the buildmaster pays attention to. We also use
+# a shorter alias to save typing.
+c = BuildmasterConfig = {}
+
+####### BUILDSLAVES
+
+# the 'slaves' list defines the set of allowable buildslaves. Each element is
+# a BuildSlave object, which is created with bot-name, bot-password. These
+# correspond to values given to the buildslave's mktap invocation.
+
+from buildbot.buildslave import BuildSlave
+
+c['slaves'] = [BuildSlave('android', 'pass', max_builds=1),
+ BuildSlave('chromeos', 'pass', max_builds=1),
+ BuildSlave('linux-chrome', 'pass', max_builds=1),
+ BuildSlave('linux-clang', 'pass', max_builds=1),
+ BuildSlave('linux-slave-1', 'pass', max_builds=1),
+ BuildSlave('linux-slave-2', 'pass', max_builds=1),
+ BuildSlave('linux-slave-4', 'pass', max_builds=1),
+ BuildSlave('linux-slave-5', 'pass', max_builds=1),
+ BuildSlave('linux-slave-gcc-4.6', 'pass', max_builds=1),
+ BuildSlave('linux-valgrind', 'pass', max_builds=1),
+ BuildSlave('linux_video', 'pass', max_builds=1),
+ BuildSlave('mac-slave-3', 'pass', max_builds=1),
+ BuildSlave('mac-slave-2', 'pass', max_builds=1),
+ BuildSlave('win-slave-1', 'pass', max_builds=1),
+ BuildSlave('win-slave-2', 'pass', max_builds=1)]
+
+# 'slavePortnum' defines the TCP port to listen on. This must match the value
+# configured into the buildslaves (with their --master option)
+
+c['slavePortnum'] = 9989
+
+####### CHANGESOURCES
+
+# the 'change_source' setting tells the buildmaster how it should find out
+# about source code changes. Any class which implements IChangeSource can be
+# put here: there are several in buildbot/changes/*.py to choose from.
+
+from buildbot.changes.pb import PBChangeSource
+from buildbot.changes.svnpoller import SVNPoller
+
+source_code_svn_url = 'http://webrtc.googlecode.com/svn/trunk'
+svn_poller = SVNPoller(svnurl=source_code_svn_url, pollinterval=5*60,
+ histmax=10, svnbin='/usr/bin/svn')
+c['change_source'] = svn_poller
+
+####### SCHEDULERS
+
+## configure the Schedulers
+
+from buildbot.scheduler import Scheduler
+webrtc_scheduler = Scheduler(name='all', branch=None, treeStableTimer=5*60,
+ builderNames=['Win32Debug',
+ 'Win32Release',
+ 'MacOS32DBG',
+ 'MacOS32Release',
+ 'Linux32DBG',
+ 'Linux32Release',
+ 'Linux64DBG',
+ 'Linux64Release',
+ 'LinuxClang',
+ 'LinuxValgrind',
+ 'Linux64DBG-GCC4.6',
+ 'LinuxVideoTest',
+ 'Android',
+ 'ChromeOS'
+ ])
+chrome_scheduler = Scheduler(name='chrome', branch=None, treeStableTimer=60*60,
+ builderNames=['Chrome'])
+
+c['schedulers'] = [webrtc_scheduler, chrome_scheduler]
+
+####### TESTS
+# Tests to run on Virtual machine bots and our Mac hardware.
+# Defines the supported tests followed by a tuple defining if the tests are
+# enabled on Linux, Mac and/or Windows (in that order; defined in utils.py).
+
+from webrtc_buildbot import utils
+
+NORMAL_TESTS = {
+ # Test name Linux Mac Windows
+ 'audio_coding_module_test': (True, True, True),
+ 'audio_coding_unittests': (True, True, True),
+ 'audio_conference_mixer_unittests':(True, True, True),
+ 'audio_device_test_api': (True, False, False), # Issue 257
+ 'audioproc_unittest': (True, True, True),
+ 'cng_unittests': (True, True, True),
+ 'g711_unittests': (True, True, True),
+ 'g722_unittests': (True, True, True),
+ 'libyuv_unittests': (True, True, True),
+ 'jpeg_unittests': (True, True, True),
+ 'media_file_unittests': (True, True, True),
+ 'metrics_unittests': (True, True, True),
+ 'neteq_unittests': (True, True, True),
+ 'pcm16b_unittests': (True, True, True),
+ 'resampler_unittests': (True, True, True),
+ 'rtp_rtcp_unittests': (True, True, True),
+ 'signal_processing_unittests': (True, True, True),
+ 'system_wrappers_unittests': (True, True, True),
+ 'test_bwe': (True, True, True),
+ 'test_fec': (True, True, True),
+ 'test_support_unittests': (True, True, True),
+ 'udp_transport_unittests': (True, True, True),
+ 'vad_unittests': (True, True, True),
+ 'video_codecs_test_framework_integrationtests': (True, True, True),
+ 'video_codecs_test_framework_unittests': (True, True, True),
+ 'video_coding_unittests': (True, True, True),
+ 'video_engine_core_unittests': (True, True, True),
+ 'video_processing_unittests': (True, True, True),
+ 'voice_engine_unittests': (True, True, True),
+ 'vp8_unittests': (True, False, True), # Issue 273.
+ 'webrtc_utility_unittests': (True, True, False),
+}
+
+PHYSICAL_MACHINE_TESTS = {
+ # Test name Linux Mac Windows
+ 'video_render_module_test': (True, False, False),
+ 'vie_auto_test': (True, False, False),
+ 'voe_auto_test': (True, False, False),
+}
+
+VALGRIND_DISABLED_TESTS = [
+ 'audio_coding_module_test', # Issue 270
+ 'test_fec', # Too slow for Valgrind
+]
+
+linux_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Linux')
+mac_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Mac')
+windows_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Windows')
+
+linux_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
+ 'Linux')
+mac_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
+ 'Mac')
+windows_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
+ 'Windows')
+
+####### FACTORIES
+# Factories defines how the build is run and can be used in multiple instances
+# on multiple machines, depending on how many builders are defined.
+
+from buildbot.process import factory
+from buildbot.steps import shell
+
+############# Linux factories #######################################
+linux_factory_64_dbg = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_factory_64_dbg'))
+linux_factory_64_dbg.EnableBuild()
+linux_factory_64_dbg.EnableTests(linux_normal_tests)
+
+linux_factory_32_release = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_factory_32_release'))
+linux_factory_32_release.EnableBuild(release=True, build32=True)
+linux_factory_32_release.EnableTests(linux_normal_tests)
+
+linux_factory_64_release = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_factory_64_release'))
+linux_factory_64_release.EnableBuild(release=True)
+linux_factory_64_release.EnableTests(linux_normal_tests)
+
+linux_factory_32_dbg = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_factory_32_dbg'))
+linux_factory_32_dbg.EnableCoverage(
+ coverage_url='http://webrtc-cb-linux-slave-4.cbf.corp.google.com/coverage/')
+linux_factory_32_dbg.EnableBuild(build32=True)
+linux_factory_32_dbg.EnableTests(linux_normal_tests)
+
+linux_factory_video = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_factory_video'))
+linux_factory_video.EnableCoverage(
+ coverage_url='http://webrtc-build-bot-se.lul/coverage/')
+linux_factory_video.EnableBuild()
+linux_factory_video.EnableTests(linux_physical_machine_tests)
+
+chromeos_factory = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('chromeos_factory'))
+chromeos_factory.EnableBuild(chrome_os=True)
+chromeos_factory.EnableTests(linux_normal_tests)
+
+linux_chrome_factory = utils.WebRTCChromeFactory(
+ utils.BuildStatusOracle('linux_chrome'))
+linux_chrome_factory.EnableBuild()
+
+linux_clang = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_clang'))
+linux_clang.EnableBuild(clang=True)
+linux_clang.EnableTests(linux_normal_tests)
+
+linux_valgrind = utils.WebRTCLinuxFactory(
+ utils.BuildStatusOracle('linux_valgrind'), valgrind_enabled=True)
+linux_valgrind.EnableBuild(release=True)
+# Filter out disabled Valgrind tests:
+valgrind_tests = filter(lambda test: test not in VALGRIND_DISABLED_TESTS,
+ linux_normal_tests)
+linux_valgrind.EnableTests(valgrind_tests)
+
+android_factory = utils.WebRTCAndroidFactory(
+ utils.BuildStatusOracle('android_factory'))
+android_factory.EnableBuild(product='toro')
+
+############## Mac factories #######################################
+mac_factory_32_dbg = utils.WebRTCMacFactory(
+ utils.BuildStatusOracle('mac_factory_32_dbg'))
+mac_factory_32_dbg.EnableBuild(build_type='both')
+mac_factory_32_dbg.EnableTests(mac_normal_tests)
+
+mac_factory_32_release = utils.WebRTCMacFactory(
+ utils.BuildStatusOracle('mac_factory_32_release'))
+mac_factory_32_release.EnableBuild(build_type='both', release=True)
+mac_factory_32_release.EnableTests(mac_normal_tests)
+
+############# Windows factories #######################################
+win_factory_32_Debug = utils.WebRTCWinFactory(
+ utils.BuildStatusOracle('win_factory_32_debug'))
+win_factory_32_Debug.EnableBuild(configuration='Debug')
+win_factory_32_Debug.EnableTests(windows_normal_tests)
+
+win_factory_32_Release = utils.WebRTCWinFactory(
+ utils.BuildStatusOracle('mac_factory_32_release'))
+win_factory_32_Release.EnableBuild(configuration='Release')
+win_factory_32_Release.EnableTests(windows_normal_tests)
+
+####### BUILDERS
+
+# the 'builders' list defines the Builders. Each one is configured with a
+# dictionary, using the following keys:
+# name (required): the name used to describe this builder
+# slavename (required): which slave to use (must appear in c['bots'])
+# builddir (required): which subdirectory to run the builder in
+# factory (required): a BuildFactory to define how the build is run
+# periodicBuildTime (optional): if set, force a build every N seconds
+
+linux_builder_64_debug = {
+ 'name': 'Linux64DBG',
+ 'slavename': 'linux-slave-1',
+ 'builddir': 'linux-slave-1',
+ 'factory': linux_factory_64_dbg,
+ }
+linux_builder_32_release = {
+ 'name': 'Linux32Release',
+ 'slavename': 'linux-slave-2',
+ 'builddir': 'linux-slave-2',
+ 'factory': linux_factory_32_release,
+ }
+linux_builder_64_release = {
+ 'name': 'Linux64Release',
+ 'slavename': 'linux-slave-5',
+ 'builddir': 'linux-slave-5',
+ 'factory': linux_factory_64_release,
+ }
+linux_builder_32_debug = {
+ 'name': 'Linux32DBG',
+ 'slavename': 'linux-slave-4',
+ 'builddir': 'linux-slave-4',
+ 'factory': linux_factory_32_dbg,
+ }
+mac_builder_32_debug = {
+ 'name': 'MacOS32DBG',
+ 'slavename': 'mac-slave-3',
+ 'builddir': 'mac-slave-3',
+ 'factory': mac_factory_32_dbg,
+ }
+mac_builder_32_release = {
+ 'name': 'MacOS32Release',
+ 'slavename': 'mac-slave-2',
+ 'builddir': 'mac-slave-2',
+ 'factory': mac_factory_32_release,
+ }
+chromeos_builder = {
+ 'name': 'ChromeOS',
+ 'slavename': 'chromeos',
+ 'builddir': 'chromeos',
+ 'factory': chromeos_factory,
+ }
+win_builder_32_debug = {
+ 'name': 'Win32Debug',
+ 'slavename': 'win-slave-1',
+ 'builddir': 'win-32-dbg',
+ 'factory': win_factory_32_Debug,
+ }
+win_builder_32_release = {
+ 'name': 'Win32Release',
+ 'slavename': 'win-slave-2',
+ 'builddir': 'win-32-release',
+ 'factory': win_factory_32_Release,
+ }
+linux_builder_video = {
+ 'name': 'LinuxVideoTest',
+ 'slavename': 'linux_video',
+ 'builddir': 'video',
+ 'factory': linux_factory_video,
+ }
+linux_builder_chrome = {
+ 'name': "Chrome",
+ 'slavename': "linux-chrome",
+ 'builddir': "linux-chrome",
+ 'factory': linux_chrome_factory,
+ }
+linux_builder_clang = {
+ 'name': 'LinuxClang',
+ 'slavename': 'linux-clang',
+ 'builddir': 'linux-clang',
+ 'factory': linux_clang,
+ }
+linux_builder_valgrind = {
+ 'name': 'LinuxValgrind',
+ 'slavename': 'linux-valgrind',
+ 'builddir': 'linux-valgrind',
+ 'factory': linux_valgrind,
+ }
+android_builder_1 = {
+ 'name': 'Android',
+ 'slavename': 'android',
+ 'builddir': 'android',
+ 'factory': android_factory,
+ }
+linux_builder_gcc_4_6 = {
+ 'name': 'Linux64DBG-GCC4.6',
+ 'slavename': 'linux-slave-gcc-4.6',
+ 'builddir': 'linux-slave-gcc-4.6',
+ 'factory': linux_factory_64_dbg,
+ }
+c['builders'] = [
+ win_builder_32_debug,
+ win_builder_32_release,
+ mac_builder_32_debug,
+ mac_builder_32_release,
+ linux_builder_32_debug,
+ linux_builder_32_release,
+ linux_builder_64_release,
+ linux_builder_64_debug,
+ linux_builder_clang,
+ linux_builder_valgrind,
+ linux_builder_gcc_4_6,
+ linux_builder_video,
+ android_builder_1,
+ chromeos_builder,
+ linux_builder_chrome,
+ ]
+
+####### STATUS TARGETS
+
+# 'status' is a list of Status Targets. The results of each build will be
+# pushed to these targets. buildbot/status/*.py has a variety to choose from,
+# including web pages, email senders, and IRC bots.
+
+from buildbot.status import html
+from buildbot.status import mail
+
+web_page = html.WebStatus(http_port=8010, allowForce=True)
+email_notification = mail.MailNotifier(
+ fromaddr='webrtc-cb-watchlist@google.com',
+ extraRecipients=['webrtc-cb-watchlist@google.com'],
+ sendToInterestedUsers=True,
+ mode='failing')
+c['status'] = [web_page, email_notification]
+
+####### DEBUGGING OPTIONS
+
+# if you set 'debugPassword', then you can connect to the buildmaster with
+# the diagnostic tool in contrib/debugclient.py . From this tool, you can
+# manually force builds and inject changes, which may be useful for testing
+# your buildmaster without actually committing changes to your repository (or
+# before you have a functioning 'sources' set up). The debug tool uses the
+# same port number as the slaves do: 'slavePortnum'.
+
+#c['debugPassword'] = 'debugpassword'
+
+# if you set 'manhole', you can ssh into the buildmaster and get an
+# interactive python shell, which may be useful for debugging buildbot
+# internals. It is probably only useful for buildbot developers. You can also
+# use an authorized_keys file, or plain telnet.
+#from buildbot import manhole
+#c['manhole'] = manhole.PasswordManhole('tcp:9999:interface=127.0.0.1',
+# 'admin', 'password')
+
+
+####### PROJECT IDENTITY
+
+# the 'projectName' string will be used to describe the project that this
+# buildbot is working on. For example, it is used as the title of the
+# waterfall HTML page. The 'projectURL' string will be used to provide a link
+# from buildbot HTML pages to your project's home page.
+
+c['projectName'] = 'WebRTC'
+c['projectURL'] = 'http://www.webrtc.org'
+
+# the 'buildbotURL' string should point to the location where the buildbot's
+# internal web server (usually the html.Waterfall page) is visible. This
+# typically uses the port number set in the Waterfall 'status' entry, but
+# with an externally-visible host name which the buildbot cannot figure out
+# without some help.
+
+c['buildbotURL'] = 'http://webrtc-cb-linux-master.cbf.corp.google.com:8010/'
diff --git a/trunk/tools/continuous_build/public_html/README b/trunk/tools/continuous_build/public_html/README
new file mode 100644
index 0000000..5f59005
--- /dev/null
+++ b/trunk/tools/continuous_build/public_html/README
@@ -0,0 +1,18 @@
+Modify the look of the buildbot master's waterfall page
+-------------------------------------------------------
+
+To deploy these changes, follow these instructions:
+
+* Copy the favicon.ico file to master/public_html/ on the buildbot master
+ machine.
+
+* Edit the master/public_html/index.html and add the following to the <head>
+ element:
+ <link rel="shortcut icon" href="/favicon.ico" type="image/png" />
+ <link rel="icon" href="/favicon.ico" type="image/png" />
+
+* Edit the master/public_html/buildbot.css and change the section with
+ .LastBuild, .Activity {
+ width: 230px;
+
+ to be 160px instead. Save the file and exit.
diff --git a/trunk/tools/continuous_build/public_html/favicon.ico b/trunk/tools/continuous_build/public_html/favicon.ico
new file mode 100644
index 0000000..324a6ca
--- /dev/null
+++ b/trunk/tools/continuous_build/public_html/favicon.ico
Binary files differ
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/__init__.py b/trunk/tools/continuous_build/webrtc_buildbot/__init__.py
new file mode 100755
index 0000000..869bd84
--- /dev/null
+++ b/trunk/tools/continuous_build/webrtc_buildbot/__init__.py
@@ -0,0 +1,2 @@
+# -*- python -*-
+# ex: set syntax=python:
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/utils.py b/trunk/tools/continuous_build/webrtc_buildbot/utils.py
new file mode 100755
index 0000000..271096c
--- /dev/null
+++ b/trunk/tools/continuous_build/webrtc_buildbot/utils.py
@@ -0,0 +1,789 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'ivinnichenko@webrtc.org (Illya Vinnichenko)'
+
+import buildbot
+import os
+import sys
+import urlparse
+from buildbot.process import factory
+from buildbot.process import properties
+from buildbot.process.properties import WithProperties
+from buildbot.steps.shell import ShellCommand
+
+# Defines the order of the booleans of the supported platforms in the test
+# dictionaries in master.cfg.
+SUPPORTED_PLATFORMS = ('Linux', 'Mac', 'Windows')
+
+SVN_LOCATION = 'http://webrtc.googlecode.com/svn/trunk'
+VALGRIND_CMD = ['tools/valgrind-webrtc/webrtc_tests.sh', '-t', 'cmdline']
+
+DEFAULT_COVERAGE_DIR = '/var/www/coverage/'
+DEFAULT_MASTER_WORK_DIR = '.'
+GCLIENT_RETRIES = 3
+
+# Copied from trunk/tools/build/scripts/master/factory/chromium_factory.py
+# but converted to a list since we set defines instead of using an environment
+# variable.
+#
+# On valgrind bots, override the optimizer settings so we don't inline too
+# much and make the stacks harder to figure out. Use the same settings
+# on all buildbot masters to make it easier to move bots.
+MEMORY_TOOLS_GYP_DEFINES = [
+ # GCC flags
+ 'mac_debug_optimization=1 ',
+ 'mac_release_optimization=1 ',
+ 'release_optimize=1 ',
+ 'no_gc_sections=1 ',
+ 'debug_extra_cflags="-g -fno-inline -fno-omit-frame-pointer '
+ '-fno-builtin -fno-optimize-sibling-calls" ',
+ 'release_extra_cflags="-g -fno-inline -fno-omit-frame-pointer '
+ '-fno-builtin -fno-optimize-sibling-calls" ',
+ # MSVS flags
+ 'win_debug_RuntimeChecks=0 ',
+ 'win_debug_disable_iterator_debugging=1 ',
+ 'win_debug_Optimization=1 ',
+ 'win_debug_InlineFunctionExpansion=0 ',
+ 'win_release_InlineFunctionExpansion=0 ',
+ 'win_release_OmitFramePointers=0 ',
+
+ 'linux_use_tcmalloc=1 ',
+ 'release_valgrind_build=1 ',
+ 'werror= ',
+]
+
+
+class WebRTCFactory(factory.BuildFactory):
+ """Abstract superclass for all build factories.
+
+ A build factory defines a sequence of steps to take in a build process.
+ This class provides some helper methods and some abstract methods that
+ can be overridden to create customized build sequences.
+ """
+
+ def __init__(self, build_status_oracle):
+ """Creates the abstract factory.
+
+ Args:
+ build_status_oracle: An instance of BuildStatusOracle which is used to
+ keep track of our build state.
+ """
+ factory.BuildFactory.__init__(self)
+
+ self.build_status_oracle = build_status_oracle
+ self.properties = properties.Properties()
+ self.gyp_params = []
+ self.release = False
+
+ def EnableBuild(self):
+ """Adds steps for building WebRTC [must be overridden].
+
+ Implementations of this method must add clean and build steps so that
+ when all steps have been run, we have an up-to-date, complete and correct
+ build of WebRTC for the platform. It is up to the method how to do this.
+ """
+ pass
+
+ def EnableTests(self, tests):
+ """Adds test run steps for all tests in the list.
+
+ This method must be run after enabling the build.
+
+ Args:
+ tests: list of test to be run.
+ """
+ for test in tests:
+ self.EnableTest(test)
+
+ def AddCommonStep(self, cmd, descriptor='', workdir='build',
+ number_of_retries=0, halt_build_on_failure=True,
+ warn_on_failure=False):
+ """Adds a step which will run as a shell command on the slave.
+
+ NOTE: you are recommended to use this method to add new shell commands
+ instead of the base-class addStep method, since steps added here will
+ work with the smart-clean system (e.g. only do a full rebuild if the
+ previous build failed). Steps handled outside this method will not lead
+ to a full rebuild on the next build if they fail.
+
+ Args:
+ cmd: The command to run. This command follows the contract for
+ ShellCommand, so see that documentation for more details.
+ descriptor: A string, or a list of strings, describing what the step
+ does. The descriptor gets printed in the waterfall display.
+ workdir: The working directory to run the command in.
+ number_of_retries: Number of times to retry the command, if it fails.
+ halt_build_on_failure: Stops the build dead in its tracks if this step
+ fails. Use for critical steps. This option does not make sense with
+ warn_on_failure.
+ warn_on_failure: If true, this step isn't that important and will not
+ cause a failed build on failure.
+ """
+ flunk_on_failure = not warn_on_failure
+
+ if type(descriptor) is str:
+ descriptor = [descriptor]
+ # Add spaces to wrap long test names to make waterfall output more compact.
+ wrapped_text = self._WrapLongLines(descriptor)
+
+ self.addStep(MonitoredRetryingShellCommand(
+ build_status_oracle=self.build_status_oracle,
+ number_of_retries=number_of_retries,
+ command=cmd,
+ workdir=workdir,
+ description=wrapped_text + ['running...'],
+ descriptionDone=wrapped_text,
+ warnOnFailure=warn_on_failure,
+ flunkOnFailure=flunk_on_failure,
+ haltOnFailure=halt_build_on_failure,
+ name='_'.join(descriptor)))
+
+ def AddSmartCleanStep(self):
+ """Adds a smart clean step.
+
+ Smart clean only cleans the whole repository if the build status oracle
+ thinks the last build failed. Otherwise it cleans just the build output.
+ """
+ self.addStep(SmartClean(self.build_status_oracle))
+
+ def AddCommonTestRunStep(self, test, descriptor='', cmd=None,
+ workdir='build/trunk'):
+ """Adds a step for running a single test [must be overridden].
+
+ In general, failing tests should not halt the build and allow other tests
+ to execute. A failing test should fail, or 'flunk', the build though.
+
+ Implementations of this method must add new steps through AddCommonStep
+ and not by calling addStep.
+
+ Args:
+ test: The test binary name. The step will attempt to execute this
+ binary in the binary output folder, except if the cmd argument is
+ defined (in that case, we will run cmd instead and just use the
+ test name in the descriptor).
+ descriptor: This should either be a string or a list of strings. The
+ descriptor or descriptors are appended to the test name and
+ displayed in the waterfall.
+ cmd: If necessary, you can specify this argument to override the
+ default behavior, which is to just run the binary specified in
+ test without arguments.
+ workdir: The base working directory to run the command in. This
+ directory will map to the WebRTC project root, e.g. the trunk
+ directory. This method will make sure that the test binary is run
+ in the correct output directory for the platform.
+ """
+ pass
+
+ def EnableTest(self, test):
+ """Makes a test run in the build sequence. May be overridden.
+
+ Override to handle special cases for specific platforms, for instance if
+ a particular test binary requires command line arguments.
+
+ Args:
+ test: The test name to enable.
+ """
+ self.AddCommonTestRunStep(test)
+
+ def AddGclientSyncStep(self, force_sync):
+ """Helper method for invoking gclient sync. Will retry if the operation
+ fails.
+
+ Args:
+ force_sync: If the sync should be forced, i.e. update even for
+ unchanged modules (known to be required for Windows sometimes).
+ """
+ cmd = ['gclient', 'sync']
+ if force_sync:
+ cmd.append('--force')
+ self.AddCommonStep(cmd, descriptor='Sync',
+ number_of_retries=GCLIENT_RETRIES)
+
+ def AddCommonGYPStep(self, gyp_file, gyp_params=[], descriptor='gyp'):
+ """Helper method for invoking GYP on WebRTC.
+
+ GYP will generate makefiles or its equivalent in a platform-specific
+ manner. A failed GYP step will halt the build.
+
+ Implementations of this method must add new steps through AddCommonStep
+ and not by calling addStep.
+
+ Args:
+ gyp_file: The root GYP file to use.
+ gyp_params: Custom GYP parameters (same semantics as the GYP_PARAMS
+ environment variable).
+ descriptor: The descriptor to use for the step.
+ """
+ cmd = ['./build/gyp_chromium', '--depth=.', gyp_file]
+ cmd += gyp_params + self.gyp_params
+ self.AddCommonStep(cmd=cmd, workdir='build/trunk', descriptor=descriptor)
+
+ def _WrapLongLines(self, string_list, max_line_length=25, wrap_character='_'):
+ """ Creates a list with wrapped strings for lines that are too long.
+
+ This is done by inserting spaces to long lines with the wrap character
+ in. It's a simple way to make long test targets wrap nicer in the
+ waterfall display.
+
+ This method should only be used for lists that are displayed in the web
+ interface!
+
+ Args:
+ string_list: List of strings where each string represents one line.
+ max_line_length: Number of characters a line may have to avoid
+ getting wrapped.
+ wrap_character: The character we're looking for when inserting a
+ space if a string is larger than max_line_length. If no such
+ character is found, no space will be inserted.
+ Returns:
+ A new list of the same length as the input list, but with strings
+ that may contain extra spaces in them, if longer than the max
+ length.
+ """
+ result = []
+ for line in string_list:
+ if len(line) > max_line_length:
+ index = line.rfind(wrap_character)
+ if index != -1:
+ line = line[:index] + ' ' + line[index:]
+ result.append(line)
+ return result
+
+
+class BuildStatusOracle:
+ """Keeps track of a particular build's state.
+
+ The oracle uses files in the default master work directory to keep track
+ of whether a build has failed. It only keeps track of the most recent build
+ until told to forget it.
+ """
+
+ def __init__(self, builder_name):
+ """Creates the oracle.
+
+ Args:
+ builder_name: The name of the associated builder. This name is used
+ in the filename on disk. This name should be unique.
+ """
+ self.builder_name = builder_name
+ self.master_work_dir = DEFAULT_MASTER_WORK_DIR
+
+ def LastBuildFailed(self):
+ failure_file_path = self._GetFailureBuildPath()
+ return os.path.exists(failure_file_path)
+
+ def ForgetLastBuild(self):
+ if self.LastBuildFailed():
+ os.remove(self._GetFailureBuildPath())
+
+ def SetLastBuildAsFailed(self):
+ open(self._GetFailureBuildPath(), 'w').close()
+
+ def _GetFailureBuildPath(self):
+ return os.path.join(self.master_work_dir, self.builder_name + ".failed")
+
+
+class MonitoredRetryingShellCommand(ShellCommand):
+ """Wraps a shell command and notifies the oracle if the command fails.
+
+ If the command fails, there's an option to retry it a number of times.
+ Default behavior is to not retry."""
+
+ def __init__(self, build_status_oracle, number_of_retries=0, **kwargs):
+ ShellCommand.__init__(self, **kwargs)
+
+ self.addFactoryArguments(build_status_oracle=build_status_oracle,
+ number_of_retries=number_of_retries)
+ self.build_status_oracle = build_status_oracle
+ self.number_of_retries = number_of_retries
+
+ def finished(self, results):
+ if (results == buildbot.status.builder.FAILURE or
+ results == buildbot.status.builder.EXCEPTION):
+ if self.number_of_retries > 0:
+ self.number_of_retries -= 1
+ self.start()
+ return
+ else:
+ self.build_status_oracle.SetLastBuildAsFailed()
+ ShellCommand.finished(self, results)
+
+
+class SmartClean(ShellCommand):
+ """Cleans the repository fully or partially depending on the build state."""
+ def __init__(self, build_status_oracle, **kwargs):
+ ShellCommand.__init__(self, **kwargs)
+
+ self.addFactoryArguments(build_status_oracle=build_status_oracle)
+ self.haltOnFailure = True
+ self.build_status_oracle = build_status_oracle
+
+ def start(self):
+ if self.build_status_oracle.LastBuildFailed():
+ self.build_status_oracle.ForgetLastBuild()
+ self.description = ['Nuke Repository', '(Previous Failed)']
+ self.setCommand(['rm', '-rf', 'trunk'])
+ else:
+ self.description = ['Clean']
+ self.setCommand('rm -rf trunk/out && '
+ 'rm -rf trunk/xcodebuild &&'
+ 'rm -rf trunk/build/Debug &&'
+ 'rm -rf trunk/build/Release')
+ ShellCommand.start(self)
+
+
+class GenerateCodeCoverage(ShellCommand):
+ """This custom shell command generates coverage HTML using genhtml.
+
+ The command will dump the HTML output into coverage_dir, in a directory
+ whose name is generated from the build number and slave name. We will
+ expect that the coverage directory is somewhere under the web server root
+ (i.e. public html root) that corresponds to the web server URL. That is, if
+ we write Foo to the coverage directory we expect that directory to be
+ reachable from url/Foo.
+ """
+
+ def __init__(self, coverage_url, coverage_dir, coverage_file, **kwargs):
+ """Prepares the coverage command.
+
+ Args:
+ coverage_url: The base URL for the serving web server we will use
+ when we generate the link to the coverage. This will generally
+ be the slave's URL (something like http://slave-hostname/).
+ coverage_dir: Where to write coverage HTML.
+ coverage_file: The LCOV file to generate the coverage from.
+ """
+ ShellCommand.__init__(self, **kwargs)
+ self.addFactoryArguments(coverage_url=coverage_url,
+ coverage_dir=coverage_dir,
+ coverage_file=coverage_file)
+ self.setDefaultWorkdir('build/trunk')
+ self.coverage_url = coverage_url
+ self.coverage_dir = coverage_dir
+ self.coverage_file = coverage_file
+ self.description = ['Coverage Report']
+ self.warnOnFailure = True
+ self.flunkOnFailure = False
+ output_dir = os.path.join(coverage_dir,
+ '%(buildername)s_%(buildnumber)s')
+ self.setCommand(['./tools/continuous_build/generate_coverage_html.sh',
+ coverage_file, WithProperties(output_dir)])
+
+ def createSummary(self, log):
+ coverage_url = urlparse.urljoin(self.coverage_url,
+ '%s_%s' % (self.getProperty('buildername'),
+ self.getProperty('buildnumber')))
+ self.addURL('click here', coverage_url)
+
+ def start(self):
+ ShellCommand.start(self)
+
+
+class WebRTCAndroidFactory(WebRTCFactory):
+ """Sets up the Android build."""
+
+ def __init__(self, build_status_oracle):
+ WebRTCFactory.__init__(self, build_status_oracle)
+
+ def EnableBuild(self, product='toro'):
+ prefix = 'rm -rf out/target/product/%s/obj/' % product
+ cleanup_list = [
+ 'rm -rf external/webrtc',
+ prefix + 'STATIC_LIBRARIES/libwebrtc_*',
+ prefix + 'SHARE_LIBRARIES/libwebrtc_*',
+ prefix + 'EXECUTABLES/webrtc_*'
+ ]
+ cmd = ' ; '.join(cleanup_list)
+ self.AddCommonStep(cmd, descriptor='cleanup', workdir='build/trunk')
+
+ cmd = 'svn checkout %s external/webrtc' % SVN_LOCATION
+ self.AddCommonStep(cmd, descriptor='svn (checkout)', workdir='build/trunk')
+
+ cmd = ('source build/envsetup.sh && lunch full_%s-eng '
+ '&& mmm external/webrtc showcommands' % product)
+ self.AddCommonStep(cmd, descriptor='build', workdir='build/trunk')
+
+
+class WebRTCChromeFactory(WebRTCFactory):
+ """Sets up the Chrome OS build."""
+
+ def __init__(self, build_status_oracle):
+ WebRTCFactory.__init__(self, build_status_oracle)
+
+ def EnableBuild(self):
+ self.AddCommonStep(['rm', '-rf', 'src'], descriptor='Cleanup')
+ self.AddGclientSyncStep(force_sync=True)
+ self.AddCommonMakeStep('chrome')
+
+ def AddCommonMakeStep(self, target, make_extra=None):
+ descriptor = ['make ' + target]
+ cmd = ['make', target, '-j100']
+ if make_extra is not None:
+ cmd.append(make_extra)
+ self.AddCommonStep(cmd=cmd, descriptor=descriptor,
+ warn_on_failure=True, workdir='build/src')
+
+
+class WebRTCLinuxFactory(WebRTCFactory):
+ """Sets up the Linux build.
+
+ This factory is quite configurable and can run a variety of builds.
+ """
+
+ def __init__(self, build_status_oracle, valgrind_enabled=False):
+ WebRTCFactory.__init__(self, build_status_oracle)
+
+ self.build_enabled = False
+ self.coverage_enabled = False
+ self.valgrind_enabled = valgrind_enabled
+
+ def EnableCoverage(self, coverage_url, coverage_dir=DEFAULT_COVERAGE_DIR):
+ """Enables coverage measurements using LCOV/GCOV.
+
+ This method must be called before enabling build.
+
+ Args:
+ coverage_url: See the GenerateCodeCoverage command's contract for
+ this argument.
+ coverage_dir: See the GenerateCodeCoverage command's contract for
+ this argument.
+ """
+ assert self.build_enabled is False
+
+ self.coverage_enabled = True
+ self.coverage_url = coverage_url
+ self.coverage_dir = coverage_dir
+
+ def EnableBuild(self, release=False, build32=False, chrome_os=False,
+ clang=False):
+ if build32:
+ self.gyp_params.append('-Dtarget_arch=ia32')
+
+ self.build_enabled = True
+ self.release = release
+
+ self.AddSmartCleanStep()
+
+ # Valgrind bots need special GYP defines to enable memory profiling
+ # friendly compilation. They already has a custom .gclient configuration
+ # file created so they don't need one being generated like the other bots.
+ if self.valgrind_enabled:
+ for gyp_define in MEMORY_TOOLS_GYP_DEFINES:
+ self.gyp_params.append('-D' + gyp_define)
+ else:
+ self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
+ descriptor='gclient_config')
+ self.AddGclientSyncStep(force_sync=False)
+
+ if chrome_os:
+ self.gyp_params.append('-Dchromeos=1')
+
+ if clang:
+ self.gyp_params.append('-Dclang=1')
+
+ if self.coverage_enabled:
+ self.gyp_params.append('-Dcoverage=1')
+ self.AddCommonGYPStep('webrtc.gyp', descriptor='CommonGYP')
+
+ if clang:
+ self.AddCommonStep(['trunk/tools/clang/scripts/update.sh'],
+ descriptor='Update_Clang')
+
+ if self.release:
+ self.AddCommonMakeStep('all', make_extra='BUILDTYPE=Release')
+ else:
+ self.AddCommonMakeStep('all')
+
+ def AddCommonTestRunStep(self, test, extra_text=None, cmd=None,
+ workdir='build/trunk'):
+ descriptor = [test, extra_text] if extra_text else [test]
+ if cmd is None:
+ test_folder = 'Release' if self.release else 'Debug'
+ cmd = ['out/%s/%s' % (test_folder, test)]
+ if self.valgrind_enabled:
+ cmd = VALGRIND_CMD + cmd
+ self.AddCommonStep(cmd, descriptor=descriptor, workdir=workdir,
+ halt_build_on_failure=False)
+
+ def AddXvfbTestRunStep(self, test_name, test_binary, test_arguments=''):
+ """ Adds a test to be run inside a XVFB window manager."""
+ cmd = ('xvfb-run '
+ '--server-args="-screen 0 800x600x24 -extension Composite" '
+ '%s %s' % (test_binary, test_arguments))
+ self.AddCommonTestRunStep(test=test_name, cmd=cmd)
+
+ def AddCommonMakeStep(self, target, extra_text=None, make_extra=None):
+ descriptor = ['make ' + target, extra_text] if extra_text else ['make ' +
+ target]
+ cmd = ['make', target, '-j100']
+ if make_extra:
+ cmd.append(make_extra)
+ self.AddCommonStep(cmd=cmd, descriptor=descriptor, workdir='build/trunk')
+
+ def AddStepsToEstablishCoverageBaseline(self):
+ self.AddCommonStep(['lcov', '--directory', '.', '--capture', '-b',
+ '.', '--initial',
+ '--output-file', 'webrtc_base.info'],
+ workdir='build/trunk',
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ descriptor='LCOV (Baseline Capture)')
+ self.AddCommonStep(['lcov', '--extract', 'webrtc_base.info', '*/src/*',
+ '--output', 'filtered.info'],
+ workdir='build/trunk',
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ descriptor='LCOV (Baseline Extract)')
+ self.AddCommonStep(['lcov', '--remove', 'filtered.info', '*/usr/include/*',
+ '/third*', '/testing/*', '*/test/*', '*_unittest.*',
+ '*/mock/*', '--output',
+ 'webrtc_base_filtered_final.info'],
+ workdir='build/trunk',
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ descriptor='LCOV (Baseline Filter)')
+
+ def AddStepsToComputeCoverage(self):
+ """Enable coverage data."""
+
+ # Delete all third-party .gcda files to save time and work around a bug
+ # in lcov which tends to hang when capturing on libjpgturbo.
+ self.AddCommonStep(['./tools/continuous_build/clean_third_party_gcda.sh'],
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ workdir='build/trunk',
+ descriptor='LCOV (Delete 3rd party)')
+ self.AddCommonStep(['lcov', '--directory', '.', '--capture', '-b',
+ '.', '--output-file', 'webrtc.info'],
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ workdir='build/trunk',
+ descriptor='LCOV (Capture)')
+ self.AddCommonStep(['lcov', '--extract', 'webrtc.info', '*/src/*',
+ '--output', 'test.info'],
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ workdir='build/trunk',
+ descriptor='LCOV (Extract)')
+ self.AddCommonStep(['lcov', '--remove', 'test.info', '*/usr/include/*',
+ '/third*', '/testing/*', '*/test/*', '*_unittest.*',
+ '*/mock/*', '--output',
+ 'final.info'],
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ workdir='build/trunk',
+ descriptor='LCOV (Filter)')
+ self.AddCommonStep(['lcov', '-a', 'webrtc_base_filtered_final.info', '-a',
+ 'final.info', '-o', 'final.info'],
+ warn_on_failure=True,
+ halt_build_on_failure=False,
+ workdir='build/trunk',
+ descriptor='LCOV (Merge)')
+ self.addStep(GenerateCodeCoverage(coverage_url=self.coverage_url,
+ coverage_dir=self.coverage_dir,
+ coverage_file='final.info'))
+
+ def EnableTests(self, tests):
+ if self.coverage_enabled:
+ self.AddStepsToEstablishCoverageBaseline()
+
+ WebRTCFactory.EnableTests(self, tests)
+
+ if self.coverage_enabled:
+ self.AddStepsToComputeCoverage()
+
+ def EnableTest(self, test):
+ """Adds a step for running a test on Linux.
+
+ In general, this method will interpret the name as the name of a binary
+ in the default build output directory, except for a few special cases
+ which require custom command lines.
+
+ Args:
+ test: the test name as a string.
+ """
+ if test == 'audioproc_unittest':
+ self.AddCommonTestRunStep(test)
+ self.AddCommonGYPStep('webrtc.gyp', gyp_params=['-Dprefer_fixed_point=1'],
+ descriptor='GYP fixed point')
+ self.AddCommonMakeStep(test, extra_text='(fixed point)')
+ self.AddCommonTestRunStep(test, extra_text='(fixed point)')
+ elif test == 'vie_auto_test':
+ # TODO(phoglund): Enable the full stack test once it is completed and
+ # nonflaky.
+ binary = 'out/Debug/vie_auto_test'
+ args = (
+ '--automated --gtest_filter="'
+ '-ViEVideoVerificationTest.RunsFullStackWithoutErrors" '
+ '--capture_test_ensure_resolution_alignment_in_capture_device=false')
+ self.AddXvfbTestRunStep(test_name=test, test_binary=binary,
+ test_arguments=args)
+ elif test == 'video_render_module_test':
+ self.AddXvfbTestRunStep(test_name=test,
+ test_binary='out/Debug/video_render_module_test')
+ elif test == 'voe_auto_test':
+ cmd = 'out/Debug/voe_auto_test --automated'
+ self.AddCommonTestRunStep(test=test, cmd=cmd)
+ else:
+ self.AddCommonTestRunStep(test)
+
+
+class WebRTCMacFactory(WebRTCFactory):
+ """Sets up the Mac build, both for make and xcode."""
+
+ def __init__(self, build_status_oracle):
+ WebRTCFactory.__init__(self, build_status_oracle)
+ self.build_type = 'both'
+ self.allowed_build_types = ['both', 'xcode', 'make']
+
+ def EnableBuild(self, build_type='both', release=False):
+ self.release = release
+
+ if build_type not in self.allowed_build_types:
+ print '*** INCORRECT BUILD TYPE (%s)!!! ***' % build_type
+ sys.exit(0)
+ else:
+ self.build_type = build_type
+ self.AddSmartCleanStep()
+ self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
+ descriptor='gclient_config')
+ self.AddGclientSyncStep(force_sync=True)
+
+ if self.build_type == 'make' or self.build_type == 'both':
+ self.AddCommonGYPStep('webrtc.gyp', gyp_params=['-f', 'make'],
+ descriptor='EnableMake')
+ self.AddCommonMakeStep('all')
+
+ def AddCommonTestRunStep(self, test, extra_text=None, cmd=None,
+ workdir='build/trunk'):
+ descriptor = [test, extra_text] if extra_text else [test]
+ if cmd is None:
+ out_path = 'xcodebuild' if self.build_type == 'xcode' else 'out'
+ test_folder = 'Release' if self.release else 'Debug'
+ cmd = ['%s/%s/%s' % (out_path, test_folder, test)]
+
+ if self.build_type == 'xcode' or self.build_type == 'both':
+ self.AddCommonStep(cmd, descriptor=descriptor + ['(xcode)'],
+ halt_build_on_failure=False, workdir=workdir)
+ # Execute test only for 'make' build type.
+ # If 'both' is enabled we'll only execute the 'xcode' built ones.
+ if self.build_type == 'make':
+ self.AddCommonStep(cmd, descriptor=descriptor + ['(make)'],
+ halt_build_on_failure=False, workdir=workdir)
+
+ def AddCommonMakeStep(self, target, extra_text=None, make_extra=None):
+ descriptor = [target, extra_text] if extra_text else [target]
+ if self.build_type == 'make' or self.build_type == 'both':
+ cmd = ['make', target, '-j100']
+ if make_extra is not None:
+ cmd.append(make_extra)
+ if self.release:
+ cmd.append('BUILDTYPE=Release')
+ self.AddCommonStep(cmd, descriptor=descriptor + ['(make)'],
+ workdir='build/trunk')
+ if self.build_type == 'xcode' or self.build_type == 'both':
+ configuration = 'Release' if self.release else 'Debug'
+ cmd = ['xcodebuild', '-project', 'webrtc.xcodeproj', '-configuration',
+ configuration, '-target', 'All']
+ self.AddCommonStep(cmd, descriptor=descriptor + ['(xcode)'],
+ workdir='build/trunk')
+
+class WebRTCWinFactory(WebRTCFactory):
+ """Sets up the Windows build.
+
+ Allows building with Debug, Release or both in sequence.
+ """
+
+ def __init__(self, build_status_oracle):
+ WebRTCFactory.__init__(self, build_status_oracle)
+ self.configuration = 'Debug'
+ self.platform = 'x64'
+ self.allowed_platforms = ['x64', 'Win32']
+ self.allowed_configurations = ['Debug', 'Release', 'both']
+
+ def EnableBuild(self, platform='Win32', configuration='Debug'):
+ if platform not in self.allowed_platforms:
+ print '*** INCORRECT PLATFORM (%s)!!! ***' % platform
+ sys.exit(0)
+ else:
+ self.platform = platform
+ if configuration not in self.allowed_configurations:
+ print '*** INCORRECT CONFIGURATION (%s)!!! ***' % configuration
+ sys.exit(0)
+ else:
+ self.configuration = configuration
+
+ self.AddSmartCleanStep()
+ self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
+ descriptor='gclient_config')
+ self.AddGclientSyncStep(force_sync=True)
+
+ if self.configuration == 'Debug' or self.configuration == 'both':
+ cmd = ['msbuild', 'webrtc.sln', '/t:Clean',
+ '/p:Configuration=Debug;Platform=%s' % (self.platform)]
+ self.AddCommonStep(cmd, descriptor='Build(Clean)', workdir='build/trunk')
+ cmd = ['msbuild', 'webrtc.sln',
+ '/p:Configuration=Debug;Platform=%s' % (self.platform)]
+ self.AddCommonStep(cmd, descriptor='Build(Debug)', workdir='build/trunk')
+ if self.configuration == 'Release' or self.configuration == 'both':
+ cmd = ['msbuild', 'webrtc.sln', '/t:Clean',
+ '/p:Configuration=Release;Platform=%s' % (self.platform)]
+ self.AddCommonStep(cmd, descriptor='Build(Clean)', workdir='build/trunk')
+ cmd = ['msbuild', 'webrtc.sln',
+ '/p:Configuration=Release;Platform=%s' % (self.platform)]
+ self.AddCommonStep(cmd, descriptor='Build(Release)',
+ workdir='build/trunk')
+
+ def AddCommonTestRunStep(self, test, cmd=None, workdir='build/trunk'):
+ descriptor = [test]
+ if self.configuration == 'Debug' or self.configuration == 'both':
+ if cmd is None:
+ cmd = ['build\Debug\%s.exe' % test]
+ self.AddCommonStep(cmd, descriptor=descriptor,
+ halt_build_on_failure=False, workdir=workdir)
+ if self.configuration == 'Release' or self.configuration == 'both':
+ if cmd is None:
+ cmd = ['build\Release\%s.exe' % test]
+ self.AddCommonStep(cmd, descriptor=descriptor,
+ halt_build_on_failure=False, workdir=workdir)
+
+# Utility functions
+
+
+class UnsupportedPlatformError(Exception):
+ pass
+
+
+def GetEnabledTests(test_dict, platform):
+ """Returns a list of enabled test names for the provided platform.
+
+ Args:
+ test_dict: Dictionary mapping test names to tuples representing if the
+ test shall be enabled on each platform. Each tuple contains one
+ boolean for each platform. The platforms are in the order specified
+ by SUPPORTED_PLATFORMS.
+ platform: The platform we're looking to get the tests for.
+
+ Returns:
+ A list of test names, sorted alphabetically.
+
+ Raises:
+ UnsupportedPlatformError: if the platform supplied is not supported.
+ """
+ if platform not in SUPPORTED_PLATFORMS:
+ raise UnsupportedPlatformError('*** UNSUPPORTED PLATFORM (%s)!!! ***' %
+ platform)
+ result = []
+ platform_index = SUPPORTED_PLATFORMS.index(platform)
+ for test_name, enabled_platforms in test_dict.iteritems():
+ if enabled_platforms[platform_index]:
+ result.append(test_name)
+ result.sort()
+ return result
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py b/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py
new file mode 100644
index 0000000..7da58f6
--- /dev/null
+++ b/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py
@@ -0,0 +1,76 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Unit tests for helper functions in utils.py."""
+
+__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
+
+import unittest
+
+from webrtc_buildbot import utils
+
+
+class Test(unittest.TestCase):
+
+ def testGetEnabledTests(self):
+ tests = {
+ # Name Linux Mac Windows
+ 'test_1': (True, True, False),
+ 'test_2': (True, False, False),
+ }
+ result = utils.GetEnabledTests(tests, 'Linux')
+ self.assertEqual(2, len(result))
+ self.assertEqual('test_1', result[0])
+ self.assertEqual('test_2', result[1])
+
+ result = utils.GetEnabledTests(tests, 'Mac')
+ self.assertEqual(1, len(result))
+ self.assertEqual('test_1', result[0])
+
+ result = utils.GetEnabledTests(tests, 'Windows')
+ self.assertEqual(0, len(result))
+
+ self.assertRaises(utils.UnsupportedPlatformError,
+ utils.GetEnabledTests, tests, 'BeOS')
+
+ def testEmptyListExitQuietly(self):
+ factory = utils.WebRTCFactory()
+ self.assertEqual([], factory._WrapLongLines([]))
+
+ def testShortLinesShallNotWrap(self):
+ factory = utils.WebRTCFactory()
+ self.assertEqual(['a'], factory._WrapLongLines(['a']))
+
+ string_25_len = '12345678901234567890123_5'
+ result = factory._WrapLongLines([string_25_len, string_25_len])
+ self.assertEqual(string_25_len, result[0])
+ self.assertEqual(string_25_len, result[1])
+
+ def testLinesWithMoreThan25CharactersWithNoWrapCharacterIsUnchanged(self):
+ factory = utils.WebRTCFactory()
+ string_26_len = '12345678901234567890123456'
+ result = factory._WrapLongLines([string_26_len, string_26_len])
+ self.assertEqual(string_26_len, result[0])
+ self.assertEqual(string_26_len, result[1])
+
+ def testLinesWithMoreThan25CharactersShallWrapOnWrapCharacter(self):
+ factory = utils.WebRTCFactory()
+ string_26_len = '123456789012345678901234_6'
+ test_list = [string_26_len, string_26_len]
+ result = factory._WrapLongLines(test_list)
+ expected_result = '123456789012345678901234 _6'
+ self.assertEqual(expected_result, result[0])
+ self.assertEqual(expected_result, result[1])
+ # Verify the original test_list was not modified too.
+ self.assertEqual(string_26_len, test_list[0])
+ self.assertEqual(string_26_len, test_list[1])
+
+
+if __name__ == '__main__':
+ unittest.main()
diff --git a/trunk/tools/coverity/OWNERS b/trunk/tools/coverity/OWNERS
new file mode 100644
index 0000000..b44992b
--- /dev/null
+++ b/trunk/tools/coverity/OWNERS
@@ -0,0 +1,3 @@
+kjellander@webrtc.org
+phoglund@webrtc.org
+
diff --git a/trunk/tools/coverity/coverity.py b/trunk/tools/coverity/coverity.py
new file mode 100755
index 0000000..84b95bc
--- /dev/null
+++ b/trunk/tools/coverity/coverity.py
@@ -0,0 +1,324 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""
+Runs Coverity Static Analysis on a build of WebRTC.
+
+This script is a modified copy of Chromium's tools/coverity/coverity.py
+Changes made:
+ * Replaced deprecated switches for cov-commit-defects command:
+ * Using --host instead of --remote
+ * Using --stream instead of --product
+ * Removed --cxx (now default enabled)
+ * Changed cleaning of output path, since WebRTC's out dir is located directly
+ in trunk/
+ * Updated some default constants.
+
+The script runs on all WebRTC supported platforms.
+
+On Windows, this script should be run in a Visual Studio Command Prompt, so
+that the INCLUDE, LIB, and PATH environment variables are set properly for
+Visual Studio.
+
+Usage examples:
+ coverity.py
+ coverity.py --dry-run
+ coverity.py --target=debug
+ %comspec% /c ""C:\Program Files\Microsoft Visual Studio 8\VC\vcvarsall.bat"
+ x86 && C:\Python24\python.exe C:\coverity.py"
+
+For a full list of options, pass the '--help' switch.
+
+See http://support.microsoft.com/kb/308569 for running this script as a
+Scheduled Task on Windows XP.
+"""
+
+import optparse
+import os
+import os.path
+import shutil
+import subprocess
+import sys
+import time
+
+# These constants provide default values, but are exposed as command-line
+# flags. See the --help for more info. Note that for historical reasons
+# (the script started out as Windows-only and has legacy usages which pre-date
+# these switches), the constants are all tuned for Windows.
+# Usage of this script on Linux pretty much requires explicit
+# --source-dir, --coverity-bin-dir, --coverity-intermediate-dir, and
+# --coverity-target command line flags.
+
+WEBRTC_SOURCE_DIR = 'C:\\webrtc.latest'
+
+# Relative to WEBRTC_SOURCE_DIR. Only applies to Windows platform.
+WEBRTC_SOLUTION_FILE = 'webrtc.sln'
+
+# Relative to WEBRTC_SOURCE_DIR. Only applies to Windows platform.
+WEBRTC_SOLUTION_DIR = 'build'
+
+COVERITY_BIN_DIR = 'C:\\coverity-integrity-center\\static-analysis\\bin'
+
+COVERITY_INTERMEDIATE_DIR = 'C:\\coverity\\cvbuild\\cr_int'
+
+COVERITY_ANALYZE_OPTIONS = ('--security --concurrency '
+ '--enable ATOMICITY '
+ '--enable MISSING_LOCK '
+ '--enable DELETE_VOID '
+ '--checker-option PASS_BY_VALUE:size_threshold:16 '
+ '--checker-option '
+ 'USE_AFTER_FREE:allow_simple_use:false '
+ '--enable-constraint-fpp '
+ '--enable-callgraph-metrics')
+
+# Might need to be changed to FQDN
+COVERITY_REMOTE = 'localhost'
+
+COVERITY_PORT = '8080'
+
+COVERITY_STREAM = 'trunk'
+
+COVERITY_TARGET = 'Windows'
+
+COVERITY_USER = 'coverityanalyzer'
+# looking for a PASSWORD constant? Look at --coverity-password-file instead.
+
+# Relative to WEBRTC_SOURCE_DIR. Contains the pid of this script.
+LOCK_FILE = 'coverity.lock'
+
+
+def _ReadPassword(pwfilename):
+ """Reads the coverity password in from a file where it was stashed"""
+ pwfile = open(pwfilename, 'r')
+ password = pwfile.readline()
+ pwfile.close()
+ return password.rstrip()
+
+
+def _RunCommand(cmd, dry_run, shell=False, echo_cmd=True):
+ """Runs the command if dry_run is false, otherwise just prints the command."""
+ if echo_cmd:
+ print cmd
+ if not dry_run:
+ return subprocess.call(cmd, shell=shell)
+ else:
+ return 0
+
+
+def _ReleaseLock(lock_file, lock_filename):
+ """Removes the lockfile. Function-ized so we can bail from anywhere"""
+ os.close(lock_file)
+ os.remove(lock_filename)
+
+
+def run_coverity(options, args):
+ """Runs all the selected tests for the given build type and target."""
+ # Create the lock file to prevent another instance of this script from
+ # running.
+ lock_filename = os.path.join(options.source_dir, LOCK_FILE)
+ try:
+ lock_file = os.open(lock_filename,
+ os.O_CREAT | os.O_EXCL | os.O_TRUNC | os.O_RDWR)
+ except OSError, err:
+ print 'Failed to open lock file:\n ' + str(err)
+ return 1
+
+ # Write the pid of this script (the python.exe process) to the lock file.
+ os.write(lock_file, str(os.getpid()))
+
+ options.target = options.target.title()
+
+ start_time = time.time()
+
+ print 'Change directory to ' + options.source_dir
+ os.chdir(options.source_dir)
+
+ # The coverity-password filename may have been a relative path.
+ # If so, assume it's relative to the source directory, which means
+ # the time to read the password is after we do the chdir().
+ coverity_password = _ReadPassword(options.coverity_password_file)
+
+ cmd = 'gclient sync'
+ gclient_exit = _RunCommand(cmd, options.dry_run, shell=True)
+ if gclient_exit != 0:
+ print 'gclient aborted with status %s' % gclient_exit
+ _ReleaseLock(lock_file, lock_filename)
+ return 1
+
+ print 'Elapsed time: %ds' % (time.time() - start_time)
+
+ # Do a clean build. Remove the build output directory first.
+ if sys.platform.startswith('linux'):
+ rm_path = os.path.join(options.source_dir,'out',options.target)
+ elif sys.platform == 'win32':
+ rm_path = os.path.join(options.source_dir,options.solution_dir,
+ options.target)
+ elif sys.platform == 'darwin':
+ rm_path = os.path.join(options.source_dir,'xcodebuild')
+ else:
+ print 'Platform "%s" unrecognized, aborting' % sys.platform
+ _ReleaseLock(lock_file, lock_filename)
+ return 1
+
+ if options.dry_run:
+ print 'shutil.rmtree(%s)' % repr(rm_path)
+ else:
+ shutil.rmtree(rm_path,True)
+
+ if options.preserve_intermediate_dir:
+ print 'Preserving intermediate directory.'
+ else:
+ if options.dry_run:
+ print 'shutil.rmtree(%s)' % repr(options.coverity_intermediate_dir)
+ print 'os.mkdir(%s)' % repr(options.coverity_intermediate_dir)
+ else:
+ shutil.rmtree(options.coverity_intermediate_dir,True)
+ os.mkdir(options.coverity_intermediate_dir)
+
+ print 'Elapsed time: %ds' % (time.time() - start_time)
+
+ use_shell_during_make = False
+ if sys.platform.startswith('linux'):
+ use_shell_during_make = True
+ _RunCommand('pwd', options.dry_run, shell=True)
+ cmd = '%s/cov-build --dir %s make BUILDTYPE=%s All' % (
+ options.coverity_bin_dir, options.coverity_intermediate_dir,
+ options.target)
+ elif sys.platform == 'win32':
+ cmd = ('%s\\cov-build.exe --dir %s devenv.com %s\\%s /build %s '
+ '/project webrtc.vcproj') % (
+ options.coverity_bin_dir, options.coverity_intermediate_dir,
+ options.source_dir, options.solution_file, options.target)
+ elif sys.platform == 'darwin':
+ use_shell_during_make = True
+ _RunCommand('pwd', options.dry_run, shell=True)
+ cmd = ('%s/cov-build --dir %s xcodebuild -project webrtc.xcodeproj '
+ '-configuration %s -target All') % (
+ options.coverity_bin_dir, options.coverity_intermediate_dir,
+ options.target)
+
+
+ _RunCommand(cmd, options.dry_run, shell=use_shell_during_make)
+ print 'Elapsed time: %ds' % (time.time() - start_time)
+
+ cov_analyze_exe = os.path.join(options.coverity_bin_dir,'cov-analyze')
+ cmd = '%s --dir %s %s' % (cov_analyze_exe,
+ options.coverity_intermediate_dir,
+ options.coverity_analyze_options)
+ _RunCommand(cmd, options.dry_run, shell=use_shell_during_make)
+ print 'Elapsed time: %ds' % (time.time() - start_time)
+
+ cov_commit_exe = os.path.join(options.coverity_bin_dir,'cov-commit-defects')
+
+ # On Linux we have started using a Target with a space in it, so we want
+ # to quote it. On the other hand, Windows quoting doesn't work quite the
+ # same way. To be conservative, I'd like to avoid quoting an argument
+ # that doesn't need quoting and which we haven't historically been quoting
+ # on that platform. So, only quote the target if we have to.
+ coverity_target = options.coverity_target
+ if sys.platform != 'win32':
+ coverity_target = '"%s"' % coverity_target
+
+ cmd = ('%s --dir %s --host %s --port %s '
+ '--stream %s '
+ '--target %s '
+ '--user %s '
+ '--password %s') % (cov_commit_exe,
+ options.coverity_intermediate_dir,
+ options.coverity_dbhost,
+ options.coverity_port,
+ options.coverity_stream,
+ coverity_target,
+ options.coverity_user,
+ coverity_password)
+ # Avoid echoing the Commit command because it has a password in it
+ print 'Commiting defects to Coverity Integrity Manager server...'
+ _RunCommand(cmd, options.dry_run, shell=use_shell_during_make, echo_cmd=False)
+
+ print 'Completed! Total time: %ds' % (time.time() - start_time)
+
+ _ReleaseLock(lock_file, lock_filename)
+
+ return 0
+
+
+def main():
+ option_parser = optparse.OptionParser()
+ option_parser.add_option('', '--dry-run', action='store_true', default=False,
+ help='print but don\'t run the commands')
+
+ option_parser.add_option('', '--target', default='Release',
+ help='build target (Debug or Release)')
+
+ option_parser.add_option('', '--source-dir', dest='source_dir',
+ help='full path to directory ABOVE "src"',
+ default=WEBRTC_SOURCE_DIR)
+
+ option_parser.add_option('', '--solution-file', dest='solution_file',
+ help='filename of solution file to build (Win only)',
+ default=WEBRTC_SOLUTION_FILE)
+
+ option_parser.add_option('', '--solution-dir', dest='solution_dir',
+ help='build directory for the solution (Win only)',
+ default=WEBRTC_SOLUTION_DIR)
+
+ option_parser.add_option('', '--coverity-bin-dir', dest='coverity_bin_dir',
+ default=COVERITY_BIN_DIR)
+
+ option_parser.add_option('', '--coverity-intermediate-dir',
+ dest='coverity_intermediate_dir',
+ default=COVERITY_INTERMEDIATE_DIR)
+
+ option_parser.add_option('', '--coverity-analyze-options',
+ dest='coverity_analyze_options',
+ help=('all cov-analyze options, e.g. "%s"'
+ % COVERITY_ANALYZE_OPTIONS),
+ default=COVERITY_ANALYZE_OPTIONS)
+
+ option_parser.add_option('', '--coverity-db-host',
+ dest='coverity_dbhost',
+ help=('coverity defect db server hostname, e.g. %s'
+ % COVERITY_REMOTE),
+ default=COVERITY_REMOTE)
+
+ option_parser.add_option('', '--coverity-db-port', dest='coverity_port',
+ help=('port # of coverity web/db server, e.g. %s'
+ % COVERITY_PORT),
+ default=COVERITY_PORT)
+
+ option_parser.add_option('', '--coverity-stream', dest='coverity_stream',
+ help=('Name of stream reported to Coverity, e.g. %s'
+ % COVERITY_STREAM),
+ default=COVERITY_STREAM)
+
+ option_parser.add_option('', '--coverity-target', dest='coverity_target',
+ help='Platform Target reported to coverity',
+ default=COVERITY_TARGET)
+
+ option_parser.add_option('', '--coverity-user', dest='coverity_user',
+ help='Username used to log into coverity',
+ default=COVERITY_USER)
+
+ option_parser.add_option('', '--coverity-password-file',
+ dest='coverity_password_file',
+ help='file containing the coverity password',
+ default='coverity-password')
+
+ helpmsg = ('By default, the intermediate dir is emptied before analysis. '
+ 'This switch disables that behavior.')
+ option_parser.add_option('', '--preserve-intermediate-dir',
+ action='store_true', help=helpmsg,
+ default=False)
+
+ options, args = option_parser.parse_args()
+ return run_coverity(options, args)
+
+
+if '__main__' == __name__:
+ sys.exit(main())
diff --git a/trunk/tools/create_supplement_gypi.py b/trunk/tools/create_supplement_gypi.py
new file mode 100644
index 0000000..f5a08e6
--- /dev/null
+++ b/trunk/tools/create_supplement_gypi.py
@@ -0,0 +1,23 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+import sys
+
+supplement_gypi = """#!/usr/bin/env python
+# This file is generated by %s. Not for check-in.
+# Please see the WebRTC DEPS file for details.
+{
+ 'variables': {
+ 'build_with_chromium': 0,
+ 'inside_chromium_build': 0,
+ }
+}
+"""
+
+def main(argv):
+ open(argv[1], 'w').write(supplement_gypi % argv[0])
+
+if __name__ == '__main__':
+ sys.exit(main(sys.argv))
diff --git a/trunk/tools/matlab/maxUnwrap.m b/trunk/tools/matlab/maxUnwrap.m
new file mode 100644
index 0000000..276c952
--- /dev/null
+++ b/trunk/tools/matlab/maxUnwrap.m
@@ -0,0 +1,25 @@
+function sequence = maxUnwrap(sequence, max)
+%
+% sequence = maxUnwrap(sequence, max)
+% Unwraps when a wrap around is detected.
+%
+% Arguments
+%
+% sequence: The vector to unwrap.
+% max: The maximum value that the sequence can take,
+% and after which it will wrap over to 0.
+%
+% Return value
+%
+% sequence: The unwrapped vector.
+%
+
+% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+% Use of this source code is governed by a BSD-style license
+% that can be found in the LICENSE file in the root of the source
+% tree. An additional intellectual property rights grant can be found
+% in the file PATENTS. All contributing project authors may
+% be found in the AUTHORS file in the root of the source tree.
+
+sequence = round((unwrap(2 * pi * sequence / max) * max) / (2 * pi));
diff --git a/trunk/tools/matlab/parseLog.m b/trunk/tools/matlab/parseLog.m
new file mode 100644
index 0000000..5d4c3f7
--- /dev/null
+++ b/trunk/tools/matlab/parseLog.m
@@ -0,0 +1,54 @@
+function parsed = parseLog(filename)
+%
+% parsed = parseLog(filename)
+% Parses a DataLog text file, with the filename specified in the string
+% filename, into a struct with each column name as a field, and with the
+% column data stored as a vector in that field.
+%
+% Arguments
+%
+% filename: A string with the name of the file to parse.
+%
+% Return value
+%
+% parsed: A struct containing each column parsed from the input file
+% as a field and with the column data stored as a vector in that
+% field.
+%
+
+% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+% Use of this source code is governed by a BSD-style license
+% that can be found in the LICENSE file in the root of the source
+% tree. An additional intellectual property rights grant can be found
+% in the file PATENTS. All contributing project authors may
+% be found in the AUTHORS file in the root of the source tree.
+
+table = importdata(filename, ',', 1);
+if ~isstruct(table)
+ error('Malformed file, possibly empty or lacking data entries')
+end
+
+colheaders = table.textdata;
+if length(colheaders) == 1
+ colheaders = regexp(table.textdata{1}, ',', 'split');
+end
+
+parsed = struct;
+i = 1;
+while i <= length(colheaders)
+ % Checking for a multi-value column.
+ m = regexp(colheaders{i}, '([\w\t]+)\[(\d+)\]', 'tokens');
+ if ~isempty(m)
+ % Parse a multi-value column
+ n = str2double(m{1}{2}) - 1;
+ parsed.(strrep(m{1}{1}, ' ', '_')) = table.data(:, i:i+n);
+ i = i + n + 1;
+ elseif ~isempty(colheaders{i})
+ % Parse a single-value column
+ parsed.(strrep(colheaders{i}, ' ', '_')) = table.data(:, i);
+ i = i + 1;
+ else
+ error('Empty column');
+ end
+end
diff --git a/trunk/tools/python_charts/OWNERS b/trunk/tools/python_charts/OWNERS
new file mode 100644
index 0000000..0428a4a
--- /dev/null
+++ b/trunk/tools/python_charts/OWNERS
@@ -0,0 +1 @@
+kjellander@webrtc.org
diff --git a/trunk/tools/python_charts/README b/trunk/tools/python_charts/README
new file mode 100644
index 0000000..483c402
--- /dev/null
+++ b/trunk/tools/python_charts/README
@@ -0,0 +1,41 @@
+This file describes how to setup Eclipse and then the Python Charts project
+
+Setup Eclipse
+-------------
+These instructions were tested on Linux, but are very similar for Windows and
+Mac.
+1. Ensure you have Python 2.x installed
+2. Download and install Google App Engine SDK for Python from
+ http://code.google.com/appengine/downloads.html
+3. Note which location you put App Engine in, as this will be needed later on.
+4. Download Eclipse from http://www.eclipse.org. Any distribution will probably
+ do, but if you're going to do mainly web development, you might pick Eclipse
+ IDE for JavaScript Web Developers
+5. Install the PyDev plugin using the Eclipse update site mentioned at
+ http://pydev.org/download.html
+6. Install the Google Plugin for Eclipse: http://code.google.com/eclipse/
+
+Setup the project
+-----------------
+Generic instructions are available at
+http://code.google.com/appengine/docs/python/gettingstarted/ but the following
+should be enough:
+1. Launch Eclipse and create a workspace
+2. Create a new PyDev Project
+3. In the PyDev Project wizard, uncheck the "Use Default" checkbox for Project
+ contents and browse to your tools/python_charts directory.
+4. Enter a project name. We'll assume PythonCharts in the examples below.
+5. In the radio button of the lower part of the window, select
+ "Add project directory to the PYTHONPATH"
+6. Click Finish
+7. Select the Run > Run Configuration… menu item
+8. Create a new "Python Run" configuration
+9. Select your Python Charts project as project
+10. As Main Module, enter the path to your dev_appserver.py, which is a part
+ of your App Engine installation,
+ e.g. /usr/local/google_appengine/dev_appserver.py
+11. At the Arguments tab, enter the location of your project root.
+ Using Eclipse variables if your project name is PythonCharts:
+ ${workspace_loc:PythonCharts}
+12. Launch the development app server by clicking the Run button.
+13. Launch a browser and go to http://localhost:8080
diff --git a/trunk/tools/python_charts/app.yaml b/trunk/tools/python_charts/app.yaml
new file mode 100644
index 0000000..ace1b51
--- /dev/null
+++ b/trunk/tools/python_charts/app.yaml
@@ -0,0 +1,9 @@
+application: webrtc-python-charts
+version: 1
+runtime: python
+api_version: 1
+
+handlers:
+
+- url: /*
+ script: webrtc/main.py
\ No newline at end of file
diff --git a/trunk/tools/python_charts/data/vp8_hw.py b/trunk/tools/python_charts/data/vp8_hw.py
new file mode 100644
index 0000000..b8c6cc0
--- /dev/null
+++ b/trunk/tools/python_charts/data/vp8_hw.py
@@ -0,0 +1,49 @@
+# Sample output from the video_quality_measurment program, included only for
+# reference. Geneate your own by running with the --python flag and then change
+# the filenames in main.py
+test_configuration = [{'name': 'name', 'value': 'VP8 hardware test'},
+{'name': 'description', 'value': ''},
+{'name': 'test_number', 'value': '0'},
+{'name': 'input_filename', 'value': 'foreman_cif.yuv'},
+{'name': 'output_filename', 'value': 'foreman_cif_out.yuv'},
+{'name': 'output_dir', 'value': '.'},
+{'name': 'packet_size_in_bytes', 'value': '1500'},
+{'name': 'max_payload_size_in_bytes', 'value': '1440'},
+{'name': 'packet_loss_mode', 'value': 'Uniform'},
+{'name': 'packet_loss_probability', 'value': '0.000000'},
+{'name': 'packet_loss_burst_length', 'value': '1'},
+{'name': 'exclude_frame_types', 'value': 'ExcludeOnlyFirstKeyFrame'},
+{'name': 'frame_length_in_bytes', 'value': '152064'},
+{'name': 'use_single_core', 'value': 'False'},
+{'name': 'keyframe_interval;', 'value': '0'},
+{'name': 'video_codec_type', 'value': 'VP8'},
+{'name': 'width', 'value': '352'},
+{'name': 'height', 'value': '288'},
+{'name': 'bit_rate_in_kbps', 'value': '500'},
+]
+frame_data_types = {'frame_number': ('number', 'Frame number'),
+'encoding_successful': ('boolean', 'Encoding successful?'),
+'decoding_successful': ('boolean', 'Decoding successful?'),
+'encode_time': ('number', 'Encode time (us)'),
+'decode_time': ('number', 'Decode time (us)'),
+'encode_return_code': ('number', 'Encode return code'),
+'decode_return_code': ('number', 'Decode return code'),
+'bit_rate': ('number', 'Bit rate (kbps)'),
+'encoded_frame_length': ('number', 'Encoded frame length (bytes)'),
+'frame_type': ('string', 'Frame type'),
+'packets_dropped': ('number', 'Packets dropped'),
+'total_packets': ('number', 'Total packets'),
+'ssim': ('number', 'SSIM'),
+'psnr': ('number', 'PSNR (dB)'),
+}
+frame_data = [{'frame_number': 0, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 94676, 'decode_time': 37942, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 1098, 'encoded_frame_length': 4579, 'frame_type': 'Other', 'packets_dropped': 0, 'total_packets': 4, 'ssim': 0.910364, 'psnr': 35.067258},
+{'frame_number': 1, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 244007, 'decode_time': 39421, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 306, 'encoded_frame_length': 1277, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.911859, 'psnr': 35.115193},
+{'frame_number': 2, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 240508, 'decode_time': 38918, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 330, 'encoded_frame_length': 1379, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.913597, 'psnr': 35.181604},
+{'frame_number': 3, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 243449, 'decode_time': 39664, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 298, 'encoded_frame_length': 1242, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.912378, 'psnr': 35.164710},
+{'frame_number': 4, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 248024, 'decode_time': 39115, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 332, 'encoded_frame_length': 1385, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.911471, 'psnr': 35.109488},
+{'frame_number': 5, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 246910, 'decode_time': 39146, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 416, 'encoded_frame_length': 1734, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.915231, 'psnr': 35.392300},
+{'frame_number': 6, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 242953, 'decode_time': 38827, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 279, 'encoded_frame_length': 1165, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.916130, 'psnr': 35.452889},
+{'frame_number': 7, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 247343, 'decode_time': 41429, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 393, 'encoded_frame_length': 1639, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.919356, 'psnr': 35.647128},
+{'frame_number': 8, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 249529, 'decode_time': 40329, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 487, 'encoded_frame_length': 2033, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.924705, 'psnr': 36.179837},
+{'frame_number': 9, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 249408, 'decode_time': 41716, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 583, 'encoded_frame_length': 2433, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.928433, 'psnr': 36.589875},
+]
diff --git a/trunk/tools/python_charts/data/vp8_sw.py b/trunk/tools/python_charts/data/vp8_sw.py
new file mode 100644
index 0000000..0f29137
--- /dev/null
+++ b/trunk/tools/python_charts/data/vp8_sw.py
@@ -0,0 +1,49 @@
+# Sample output from the video_quality_measurment program, included only for
+# reference. Geneate your own by running with the --python flag and then change
+# the filenames in main.py
+test_configuration = [{'name': 'name', 'value': 'VP8 software test'},
+{'name': 'description', 'value': ''},
+{'name': 'test_number', 'value': '0'},
+{'name': 'input_filename', 'value': 'foreman_cif.yuv'},
+{'name': 'output_filename', 'value': 'foreman_cif_out.yuv'},
+{'name': 'output_dir', 'value': '.'},
+{'name': 'packet_size_in_bytes', 'value': '1500'},
+{'name': 'max_payload_size_in_bytes', 'value': '1440'},
+{'name': 'packet_loss_mode', 'value': 'Uniform'},
+{'name': 'packet_loss_probability', 'value': '0.000000'},
+{'name': 'packet_loss_burst_length', 'value': '1'},
+{'name': 'exclude_frame_types', 'value': 'ExcludeOnlyFirstKeyFrame'},
+{'name': 'frame_length_in_bytes', 'value': '152064'},
+{'name': 'use_single_core', 'value': 'False'},
+{'name': 'keyframe_interval;', 'value': '0'},
+{'name': 'video_codec_type', 'value': 'VP8'},
+{'name': 'width', 'value': '352'},
+{'name': 'height', 'value': '288'},
+{'name': 'bit_rate_in_kbps', 'value': '500'},
+]
+frame_data_types = {'frame_number': ('number', 'Frame number'),
+'encoding_successful': ('boolean', 'Encoding successful?'),
+'decoding_successful': ('boolean', 'Decoding successful?'),
+'encode_time': ('number', 'Encode time (us)'),
+'decode_time': ('number', 'Decode time (us)'),
+'encode_return_code': ('number', 'Encode return code'),
+'decode_return_code': ('number', 'Decode return code'),
+'bit_rate': ('number', 'Bit rate (kbps)'),
+'encoded_frame_length': ('number', 'Encoded frame length (bytes)'),
+'frame_type': ('string', 'Frame type'),
+'packets_dropped': ('number', 'Packets dropped'),
+'total_packets': ('number', 'Total packets'),
+'ssim': ('number', 'SSIM'),
+'psnr': ('number', 'PSNR (dB)'),
+}
+frame_data = [{'frame_number': 0, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 12427, 'decode_time': 4403, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 2270, 'encoded_frame_length': 9459, 'frame_type': 'Other', 'packets_dropped': 0, 'total_packets': 7, 'ssim': 0.947050, 'psnr': 38.332820},
+{'frame_number': 1, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 3292, 'decode_time': 821, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 88, 'encoded_frame_length': 368, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.927272, 'psnr': 35.883510},
+{'frame_number': 2, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4295, 'decode_time': 902, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 130, 'encoded_frame_length': 544, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.920539, 'psnr': 35.457107},
+{'frame_number': 3, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 3880, 'decode_time': 767, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 171, 'encoded_frame_length': 714, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.917434, 'psnr': 35.389298},
+{'frame_number': 4, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4471, 'decode_time': 909, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 248, 'encoded_frame_length': 1035, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.918892, 'psnr': 35.570229},
+{'frame_number': 5, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4447, 'decode_time': 976, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 269, 'encoded_frame_length': 1123, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.920609, 'psnr': 35.769663},
+{'frame_number': 6, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4432, 'decode_time': 891, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 271, 'encoded_frame_length': 1132, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.922672, 'psnr': 35.913519},
+{'frame_number': 7, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 5026, 'decode_time': 1068, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 366, 'encoded_frame_length': 1529, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.925505, 'psnr': 36.246713},
+{'frame_number': 8, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4877, 'decode_time': 1051, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 369, 'encoded_frame_length': 1538, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.926122, 'psnr': 36.305984},
+{'frame_number': 9, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4712, 'decode_time': 1087, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 406, 'encoded_frame_length': 1692, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.927183, 'psnr': 36.379735},
+]
diff --git a/trunk/tools/python_charts/gviz_api.py b/trunk/tools/python_charts/gviz_api.py
new file mode 120000
index 0000000..c9dca90
--- /dev/null
+++ b/trunk/tools/python_charts/gviz_api.py
@@ -0,0 +1 @@
+../../third_party/google-visualization-python/gviz_api.py
\ No newline at end of file
diff --git a/trunk/tools/python_charts/templates/chart_page_template.html b/trunk/tools/python_charts/templates/chart_page_template.html
new file mode 100644
index 0000000..1cb3951
--- /dev/null
+++ b/trunk/tools/python_charts/templates/chart_page_template.html
@@ -0,0 +1,90 @@
+<html>
+ <!--
+ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+
+ Use of this source code is governed by a BSD-style license
+ that can be found in the LICENSE file in the root of the source
+ tree. An additional intellectual property rights grant can be found
+ in the file PATENTS. All contributing project authors may
+ be found in the AUTHORS file in the root of the source tree.
+
+ Template file to be used to generate Charts for Video Quality Metrics.
+ -->
+ <head>
+ <link href="http://code.google.com/css/codesite.pack.04102009.css"
+ rel="stylesheet" type="text/css" />
+ </head>
+ <script src="https://www.google.com/jsapi" type="text/javascript"></script>
+ <script>
+ google.load('visualization', '1', {packages:['table', 'corechart']});
+
+ google.setOnLoadCallback(drawTable);
+ function drawTable() {
+ /* Build data tables and views */
+ var configurations_data_table =
+ new google.visualization.DataTable(%(json_configurations)s);
+ var ssim_data_table =
+ new google.visualization.DataTable(%(json_ssim_data)s);
+ var psnr_data_table =
+ new google.visualization.DataTable(%(json_psnr_data)s);
+ var packet_loss_data_table =
+ new google.visualization.DataTable(%(json_packet_loss_data)s);
+ var bit_rate_data_table =
+ new google.visualization.DataTable(%(json_bit_rate_data)s);
+
+ /* Display tables and charts */
+ var configurations_table = new google.visualization.Table(
+ document.getElementById('table_div_configurations'));
+ configurations_table.draw(configurations_data_table, {
+ height: 200
+ });
+
+ var ssim_chart = new google.visualization.LineChart(
+ document.getElementById('table_div_ssim'));
+ ssim_chart.draw(ssim_data_table, {
+ colors: ['blue', 'red', 'lightblue', 'pink'],
+ vAxis: {title: 'SSIM'},
+ hAxis: {title: 'Frame'},
+ width: 1200, height: 300,
+ });
+
+ var psnr_chart = new google.visualization.LineChart(
+ document.getElementById('table_div_psnr'));
+ psnr_chart.draw(psnr_data_table, {
+ colors: ['blue', 'red', 'lightblue', 'pink'],
+ vAxis: {title: 'PSNR (dB)'},
+ hAxis: {title: 'Frame'},
+ width: 1200, height: 300,
+ });
+
+ var packet_loss_chart = new google.visualization.LineChart(
+ document.getElementById('table_div_packet_loss'));
+ packet_loss_chart.draw(packet_loss_data_table, {
+ colors: ['blue', 'red', 'lightblue', 'pink'],
+ vAxis: {title: 'Packets dropped'},
+ hAxis: {title: 'Frame'},
+ width: 1200, height: 300,
+ });
+
+ var bit_rate_chart = new google.visualization.LineChart(
+ document.getElementById('table_div_bit_rate'));
+ bit_rate_chart.draw(bit_rate_data_table, {
+ colors: ['blue', 'red', 'lightblue', 'pink', 'green'],
+ vAxis: {title: 'Bit rate'},
+ hAxis: {title: 'Frame'},
+ width: 1200, height: 300,
+ });
+ }
+ </script>
+ <body>
+ <h3>Test Configurations:</h3>
+ <div id="table_div_configurations"></div>
+ <h3>Messages:</h3>
+ <pre>%(messages)s</pre>
+ <h3>Metrics measured per frame:</h3>
+ <div id="table_div_ssim"></div>
+ <div id="table_div_psnr"></div>
+ <div id="table_div_packet_loss"></div>
+ <div id="table_div_bit_rate"></div>
+ </body>
+</html>
diff --git a/trunk/tools/python_charts/webrtc/__init__.py b/trunk/tools/python_charts/webrtc/__init__.py
new file mode 100644
index 0000000..c1caaa2
--- /dev/null
+++ b/trunk/tools/python_charts/webrtc/__init__.py
@@ -0,0 +1,8 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
\ No newline at end of file
diff --git a/trunk/tools/python_charts/webrtc/data_helper.py b/trunk/tools/python_charts/webrtc/data_helper.py
new file mode 100644
index 0000000..fce949f
--- /dev/null
+++ b/trunk/tools/python_charts/webrtc/data_helper.py
@@ -0,0 +1,183 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
+
+class DataHelper(object):
+ """
+ Helper class for managing table data.
+ This class does not verify the consistency of the data tables sent into it.
+ """
+
+ def __init__(self, data_list, table_description, names_list, messages):
+ """ Initializes the DataHelper with data.
+
+ Args:
+ data_list: List of one or more data lists in the format that the
+ Google Visualization Python API expects (list of dictionaries, one
+ per row of data). See the gviz_api.DataTable documentation for more
+ info.
+ table_description: dictionary describing the data types of all
+ columns in the data lists, as defined in the gviz_api.DataTable
+ documentation.
+ names_list: List of strings of what we're going to name the data
+ columns after. Usually different runs of data collection.
+ messages: List of strings we might append error messages to.
+ """
+ self.data_list = data_list
+ self.table_description = table_description
+ self.names_list = names_list
+ self.messages = messages
+ self.number_of_datasets = len(data_list)
+ self.number_of_frames = len(data_list[0])
+
+ def CreateData(self, field_name, start_frame=0, end_frame=0):
+ """ Creates a data structure for a specified data field.
+
+ Creates a data structure (data type description dictionary and a list
+ of data dictionaries) to be used with the Google Visualization Python
+ API. The frame_number column is always present and one column per data
+ set is added and its field name is suffixed by _N where N is the number
+ of the data set (0, 1, 2...)
+
+ Args:
+ field_name: String name of the field, must be present in the data
+ structure this DataHelper was created with.
+ start_frame: Frame number to start at (zero indexed). Default: 0.
+ end_frame: Frame number to be the last frame. If zero all frames
+ will be included. Default: 0.
+
+ Returns:
+ A tuple containing:
+ - a dictionary describing the columns in the data result_data_table below.
+ This description uses the name for each data set specified by
+ names_list.
+
+ Example with two data sets named 'Foreman' and 'Crew':
+ {
+ 'frame_number': ('number', 'Frame number'),
+ 'ssim_0': ('number', 'Foreman'),
+ 'ssim_1': ('number', 'Crew'),
+ }
+ - a list containing dictionaries (one per row) with the frame_number
+ column and one column of the specified field_name column per data
+ set.
+
+ Example with two data sets named 'Foreman' and 'Crew':
+ [
+ {'frame_number': 0, 'ssim_0': 0.98, 'ssim_1': 0.77 },
+ {'frame_number': 1, 'ssim_0': 0.81, 'ssim_1': 0.53 },
+ ]
+ """
+
+ # Build dictionary that describes the data types
+ result_table_description = {'frame_number': ('string', 'Frame number')}
+ for dataset_index in range(self.number_of_datasets):
+ column_name = '%s_%s' % (field_name, dataset_index)
+ column_type = self.table_description[field_name][0]
+ column_description = self.names_list[dataset_index]
+ result_table_description[column_name] = (column_type, column_description)
+
+ # Build data table of all the data
+ result_data_table = []
+ # We're going to have one dictionary per row.
+ # Create that and copy frame_number values from the first data set
+ for source_row in self.data_list[0]:
+ row_dict = { 'frame_number': source_row['frame_number'] }
+ result_data_table.append(row_dict)
+
+ # Pick target field data points from the all data tables
+ if end_frame == 0: # Default to all frames
+ end_frame = self.number_of_frames
+
+ for dataset_index in range(self.number_of_datasets):
+ for row_number in range(start_frame, end_frame):
+ column_name = '%s_%s' % (field_name, dataset_index)
+ # Stop if any of the data sets are missing the frame
+ try:
+ result_data_table[row_number][column_name] = \
+ self.data_list[dataset_index][row_number][field_name]
+ except IndexError:
+ self.messages.append("Couldn't find frame data for row %d "
+ "for %s" % (row_number, self.names_list[dataset_index]))
+ break
+ return result_table_description, result_data_table
+
+ def GetOrdering(self, table_description):
+ """ Creates a list of column names, ordered alphabetically except for the
+ frame_number column which always will be the first column.
+
+ Args:
+ table_description: A dictionary of column definitions as defined by the
+ gviz_api.DataTable documentation.
+ Returns:
+ A list of column names, where frame_number is the first and the
+ remaining columns are sorted alphabetically.
+ """
+ # The JSON data representation generated from gviz_api.DataTable.ToJSon()
+ # must have frame_number as its first column in order for the chart to
+ # use it as it's X-axis value series.
+ # gviz_api.DataTable orders the columns by name by default, which will
+ # be incorrect if we have column names that are sorted before frame_number
+ # in our data table.
+ columns_ordering = ['frame_number']
+ # add all other columns:
+ for column in sorted(table_description.keys()):
+ if column != 'frame_number':
+ columns_ordering.append(column)
+ return columns_ordering
+
+ def CreateConfigurationTable(self, configurations):
+ """ Combines multiple test data configurations for display.
+
+ Args:
+ configurations: List of one ore more configurations. Each configuration
+ is required to be a list of dictionaries with two keys: 'name' and
+ 'value'.
+ Example of a single configuration:
+ [
+ {'name': 'name', 'value': 'VP8 software'},
+ {'name': 'test_number', 'value': '0'},
+ {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
+ ]
+ Returns:
+ A tuple containing:
+ - a dictionary describing the columns in the configuration table to be
+ displayed. All columns will have string as data type.
+ Example:
+ {
+ 'name': 'string',
+ 'test_number': 'string',
+ 'input_filename': 'string',
+ }
+ - a list containing dictionaries (one per configuration) with the
+ configuration column names mapped to the value for each test run:
+
+ Example matching the columns above:
+ [
+ {'name': 'VP8 software',
+ 'test_number': '12',
+ 'input_filename': 'foreman_cif.yuv' },
+ {'name': 'VP8 hardware',
+ 'test_number': '5',
+ 'input_filename': 'foreman_cif.yuv' },
+ ]
+ """
+ result_description = {}
+ result_data = []
+
+ for configuration in configurations:
+ data = {}
+ result_data.append(data)
+ for dict in configuration:
+ name = dict['name']
+ value = dict['value']
+ result_description[name] = 'string'
+ data[name] = value
+ return result_description, result_data
diff --git a/trunk/tools/python_charts/webrtc/data_helper_test.py b/trunk/tools/python_charts/webrtc/data_helper_test.py
new file mode 100644
index 0000000..6282f7b
--- /dev/null
+++ b/trunk/tools/python_charts/webrtc/data_helper_test.py
@@ -0,0 +1,115 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
+
+import unittest
+import webrtc.data_helper
+
+class Test(unittest.TestCase):
+
+ def setUp(self):
+ # Simulate frame data from two different test runs, with 2 frames each.
+ self.frame_data_0 = [{'frame_number': 0, 'ssim': 0.5, 'psnr': 30.5},
+ {'frame_number': 1, 'ssim': 0.55, 'psnr': 30.55}]
+ self.frame_data_1 = [{'frame_number': 0, 'ssim': 0.6, 'psnr': 30.6},
+ {'frame_number': 0, 'ssim': 0.66, 'psnr': 30.66}]
+ self.all_data = [ self.frame_data_0, self.frame_data_1 ]
+
+ # Test with frame_number column in a non-first position sice we need to
+ # support reordering that to be able to use the gviz_api as we want.
+ self.type_description = {
+ 'ssim': ('number', 'SSIM'),
+ 'frame_number': ('number', 'Frame number'),
+ 'psnr': ('number', 'PSRN'),
+ }
+ self.names = ["Test 0", "Test 1"]
+ self.configurations = [
+ [{'name': 'name', 'value': 'Test 0'},
+ {'name': 'test_number', 'value': '13'},
+ {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
+ ],
+ [{'name': 'name', 'value': 'Test 1'},
+ {'name': 'test_number', 'value': '5'},
+ {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
+ ],
+ ]
+
+ def testCreateData(self):
+ messages = []
+ helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
+ self.names, messages)
+ description, data_table = helper.CreateData('ssim')
+ self.assertEqual(3, len(description))
+ self.assertTrue('frame_number' in description)
+ self.assertTrue('ssim_0' in description)
+ self.assertTrue('number' in description['ssim_0'][0])
+ self.assertTrue('Test 0' in description['ssim_0'][1])
+ self.assertTrue('ssim_1' in description)
+ self.assertTrue('number' in description['ssim_1'][0])
+ self.assertTrue('Test 1' in description['ssim_1'][1])
+
+ self.assertEqual(0, len(messages))
+
+ self.assertEquals(2, len(data_table))
+ row = data_table[0]
+ self.assertEquals(0, row['frame_number'])
+ self.assertEquals(0.5, row['ssim_0'])
+ self.assertEquals(0.6, row['ssim_1'])
+ row = data_table[1]
+ self.assertEquals(1, row['frame_number'])
+ self.assertEquals(0.55, row['ssim_0'])
+ self.assertEquals(0.66, row['ssim_1'])
+
+ description, data_table = helper.CreateData('psnr')
+ self.assertEqual(3, len(description))
+ self.assertTrue('frame_number' in description)
+ self.assertTrue('psnr_0' in description)
+ self.assertTrue('psnr_1' in description)
+ self.assertEqual(0, len(messages))
+
+ self.assertEquals(2, len(data_table))
+ row = data_table[0]
+ self.assertEquals(0, row['frame_number'])
+ self.assertEquals(30.5, row['psnr_0'])
+ self.assertEquals(30.6, row['psnr_1'])
+ row = data_table[1]
+ self.assertEquals(1, row['frame_number'])
+ self.assertEquals(30.55, row['psnr_0'])
+ self.assertEquals(30.66, row['psnr_1'])
+
+ def testGetOrdering(self):
+ """ Tests that the ordering help method returns a list with frame_number
+ first and the rest sorted alphabetically """
+ messages = []
+ helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
+ self.names, messages)
+ description, data_table = helper.CreateData('ssim')
+ columns = helper.GetOrdering(description)
+ self.assertEqual(3, len(columns))
+ self.assertEqual(0, len(messages))
+ self.assertEqual('frame_number', columns[0])
+ self.assertEqual('ssim_0', columns[1])
+ self.assertEqual('ssim_1', columns[2])
+
+ def testCreateConfigurationTable(self):
+ messages = []
+ helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
+ self.names, messages)
+ description, data = helper.CreateConfigurationTable(self.configurations)
+ self.assertEqual(3, len(description)) # 3 columns
+ self.assertEqual(2, len(data)) # 2 data sets
+ self.assertTrue(description.has_key('name'))
+ self.assertTrue(description.has_key('test_number'))
+ self.assertTrue(description.has_key('input_filename'))
+ self.assertEquals('Test 0', data[0]['name'])
+ self.assertEquals('Test 1', data[1]['name'])
+
+if __name__ == "__main__":
+ unittest.main()
diff --git a/trunk/tools/python_charts/webrtc/main.py b/trunk/tools/python_charts/webrtc/main.py
new file mode 100644
index 0000000..82d8831
--- /dev/null
+++ b/trunk/tools/python_charts/webrtc/main.py
@@ -0,0 +1,157 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
+
+import os
+import gviz_api
+import webrtc.data_helper
+
+def main():
+ """
+ This Python script displays a web page with test created with the
+ video_quality_measurement program, which is a tool in WebRTC.
+
+ The script requires on two external files and one Python library:
+ - A HTML template file with layout and references to the json variables
+ defined in this script
+ - A data file in Python format, containing the following:
+ - test_configuration - a dictionary of test configuration names and values.
+ - frame_data_types - a dictionary that maps the different metrics to their
+ data types.
+ - frame_data - a list of dictionaries where each dictionary maps a metric to
+ it's value.
+ - The gviz_api.py of the Google Visualization Python API, available at
+ http://code.google.com/p/google-visualization-python/
+
+ The HTML file is shipped with the script, while the data file must be
+ generated by running video_quality_measurement with the --python flag
+ specified.
+ """
+ print 'Content-type: text/html\n' # the newline is required!
+
+ page_template_filename = '../templates/chart_page_template.html'
+ # The data files must be located in the project tree for app engine being
+ # able to access them.
+ data_filenames = [ '../data/vp8_sw.py', '../data/vp8_hw.py' ]
+ # Will contain info/error messages to be displayed on the resulting page.
+ messages = []
+ # Load the page HTML template.
+ try:
+ f = open(page_template_filename)
+ page_template = f.read()
+ f.close()
+ except IOError as e:
+ ShowErrorPage('Cannot open page template file: %s<br>Details: %s' %
+ (page_template_filename, e))
+ return
+
+ # Read data from external Python script files. First check that they exist.
+ for filename in data_filenames:
+ if not os.path.exists(filename):
+ messages.append('Cannot open data file: %s' % filename)
+ data_filenames.remove(filename)
+
+ # Read data from all existing input files.
+ data_list = []
+ test_configurations = []
+ names = []
+
+ for filename in data_filenames:
+ read_vars = {} # empty dictionary to load the data into.
+ execfile(filename, read_vars, read_vars)
+
+ test_configuration = read_vars['test_configuration']
+ table_description = read_vars['frame_data_types']
+ table_data = read_vars['frame_data']
+
+ # Verify the data in the file loaded properly.
+ if not table_description or not table_data:
+ messages.append('Invalid input file: %s. Missing description list or '
+ 'data dictionary variables.' % filename)
+ continue
+
+ # Frame numbers appear as number type in the data, but Chart API requires
+ # values of the X-axis to be of string type.
+ # Change the frame_number column data type:
+ table_description['frame_number'] = ('string', 'Frame number')
+ # Convert all the values to string types:
+ for row in table_data:
+ row['frame_number'] = str(row['frame_number'])
+
+ # Store the unique data from this file in the high level lists.
+ test_configurations.append(test_configuration)
+ data_list.append(table_data)
+ # Name of the test run must be present.
+ test_name = FindConfiguration(test_configuration, 'name')
+ if not test_name:
+ messages.append('Invalid input file: %s. Missing configuration key '
+ '"name"', filename)
+ continue
+ names.append(test_name)
+
+ # Create data helper and build data tables for each graph.
+ helper = webrtc.data_helper.DataHelper(data_list, table_description,
+ names, messages)
+
+ # Loading it into gviz_api.DataTable objects and create JSON strings.
+ description, data = helper.CreateConfigurationTable(test_configurations)
+ configurations = gviz_api.DataTable(description, data)
+ json_configurations = configurations.ToJSon()
+
+ description, data = helper.CreateData('ssim')
+ ssim = gviz_api.DataTable(description, data)
+ json_ssim_data = ssim.ToJSon(helper.GetOrdering(description))
+
+ description, data = helper.CreateData('psnr')
+ psnr = gviz_api.DataTable(description, data)
+ json_psnr_data = psnr.ToJSon(helper.GetOrdering(description))
+
+ description, data = helper.CreateData('packets_dropped')
+ packet_loss = gviz_api.DataTable(description, data)
+ json_packet_loss_data = packet_loss.ToJSon(helper.GetOrdering(description))
+
+ description, data = helper.CreateData('bit_rate')
+ # Add a column of data points for the desired bit rate to be plotted.
+ # (uses test configuration from the last data set, assuming it is the same
+ # for all of them)
+ desired_bit_rate = FindConfiguration(test_configuration, 'bit_rate_in_kbps')
+ if not desired_bit_rate:
+ ShowErrorPage('Cannot configuration field named "bit_rate_in_kbps"')
+ return
+ desired_bit_rate = int(desired_bit_rate)
+ # Add new column data type description.
+ description['desired_bit_rate'] = ('number', 'Desired bit rate (kbps)')
+ for row in data:
+ row['desired_bit_rate'] = desired_bit_rate
+ bit_rate = gviz_api.DataTable(description, data)
+ json_bit_rate_data = bit_rate.ToJSon(helper.GetOrdering(description))
+
+ # Format the messages list with newlines.
+ messages = '\n'.join(messages)
+
+ # Put the variables as JSon strings into the template.
+ print page_template % vars()
+
+def FindConfiguration(configuration, name):
+ """ Finds a configuration value using it's name.
+ Returns the first configuration with a matching name. Returns None if no
+ matching configuration is found. """
+ return_value = None
+ for row in configuration:
+ if row['name'] == name:
+ return_value = row['value']
+ break
+ return return_value
+
+def ShowErrorPage(error_message):
+ print '<html><body>%s</body></html>' % error_message
+
+if __name__ == '__main__':
+ main()
diff --git a/trunk/tools/quality_tracking/OWNERS b/trunk/tools/quality_tracking/OWNERS
new file mode 100644
index 0000000..323e8e7
--- /dev/null
+++ b/trunk/tools/quality_tracking/OWNERS
@@ -0,0 +1,2 @@
+phoglund@webrtc.org
+kjellander@webrtc.org
diff --git a/trunk/tools/quality_tracking/README b/trunk/tools/quality_tracking/README
new file mode 100644
index 0000000..faf3e7a
--- /dev/null
+++ b/trunk/tools/quality_tracking/README
@@ -0,0 +1,31 @@
+This file describes the coverage tracking script and the coverage dashboard.
+
+ABSTRACT:
+The intention of this small tracking system is to track code coverage data
+over time. Since code coverage is continuously recomputed on the build bots,
+the track_coverage.py script is intended to run on the build bot as a cron job
+and extract the data from there. The dashboard doesn't care how often this
+script runs, but running each hour should be more than enough.
+
+The track_coverage.py script uses OAuth to authenticate itself. In order to do
+this, it needs two files: consumer.secret and access.token. The consumer secret
+is known within the organization and is stored in a plain file on the bot
+running the scripts (we don't want to check in this secret in the code in the
+public repository). The consumer secret is a plain file with a single line
+containing the secret string.
+
+The access.token file is generated by request_oauth_permission.py. It does this
+by going through the three-legged OAuth authorization process. An administrator
+of the dashboard must approve the request from the script. Once that is done,
+access.token will be written and track_coverage.py will be able to report
+results.
+
+HOW TO RUN LOCALLY:
+Follow the following instructions:
+http://code.google.com/appengine/docs/python/gettingstartedpython27/devenvironment.html
+The dashboard can be started on 127.0.0.1:8080 using the dev_appserver.py script
+as described in the above URL (and in the following 'hello world' page).
+
+HOW TO DEPLOY:
+Follow the following instructions:
+http://code.google.com/appengine/docs/python/gettingstartedpython27/uploading.html
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/constants.py b/trunk/tools/quality_tracking/constants.py
new file mode 100644
index 0000000..c6aac74
--- /dev/null
+++ b/trunk/tools/quality_tracking/constants.py
@@ -0,0 +1,37 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Contains tweakable constants for quality dashboard utility scripts."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+# This identifies our application using the information we got when we
+# registered the application on Google appengine.
+DASHBOARD_SERVER = 'webrtc-dashboard.appspot.com'
+DASHBOARD_SERVER_HTTP = 'http://' + DASHBOARD_SERVER
+CONSUMER_KEY = DASHBOARD_SERVER
+CONSUMER_SECRET_FILE = 'consumer.secret'
+ACCESS_TOKEN_FILE = 'access.token'
+
+# OAuth URL:s.
+REQUEST_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthGetRequestToken'
+AUTHORIZE_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthAuthorizeToken'
+ACCESS_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthGetAccessToken'
+
+# The build master URL.
+BUILD_MASTER_SERVER = 'webrtc-cb-linux-master.cbf.corp.google.com:8010'
+BUILD_MASTER_TRANSPOSED_GRID_URL = '/tgrid'
+
+# Build bot constants.
+BUILD_BOT_COVERAGE_WWW_DIRECTORY = '/var/www/'
+
+# Dashboard data input URLs.
+ADD_COVERAGE_DATA_URL = DASHBOARD_SERVER_HTTP + '/add_coverage_data'
+ADD_BUILD_STATUS_DATA_URL = DASHBOARD_SERVER_HTTP + '/add_build_status_data'
diff --git a/trunk/tools/quality_tracking/dashboard/add_build_status_data.py b/trunk/tools/quality_tracking/dashboard/add_build_status_data.py
new file mode 100644
index 0000000..106d178
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/add_build_status_data.py
@@ -0,0 +1,171 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Implements a handler for adding build status data."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import datetime
+import logging
+
+from google.appengine.ext import db
+
+import oauth_post_request_handler
+
+VALID_STATUSES = ['OK', 'failed', 'building', 'warnings']
+
+
+class OrphanedBuildStatusesExistException(Exception):
+ pass
+
+
+class BuildStatusRoot(db.Model):
+ """Exists solely to be the root parent for all build status data and to keep
+ track of when the last update was made.
+
+ Since all build status data will refer to this as their parent,
+ we can run transactions on the build status data as a whole.
+ """
+ last_updated_at = db.DateTimeProperty()
+
+
+class BuildStatusData(db.Model):
+ """This represents one build status report from the build bot."""
+ bot_name = db.StringProperty(required=True)
+ revision = db.IntegerProperty(required=True)
+ build_number = db.IntegerProperty(required=True)
+ status = db.StringProperty(required=True)
+
+
+def _ensure_build_status_root_exists():
+ root = db.GqlQuery('SELECT * FROM BuildStatusRoot').get()
+ if not root:
+ # Create a new root, but ensure we don't have any orphaned build statuses
+ # (in that case, we would not have a single entity group as we desire).
+ orphans = db.GqlQuery('SELECT * FROM BuildStatusData').get()
+ if orphans:
+ raise OrphanedBuildStatusesExistException('Parent is gone and there are '
+ 'orphaned build statuses in '
+ 'the database!')
+ root = BuildStatusRoot()
+ root.put()
+
+ return root
+
+
+def _filter_oauth_parameters(post_keys):
+ return filter(lambda post_key: not post_key.startswith('oauth_'),
+ post_keys)
+
+
+def _parse_status(build_number_and_status):
+ parsed_status = build_number_and_status.split('--')
+ if len(parsed_status) != 2:
+ raise ValueError('Malformed status string %s.' % build_number_and_status)
+
+ parsed_build_number = int(parsed_status[0])
+ status = parsed_status[1]
+
+ if status not in VALID_STATUSES:
+ raise ValueError('Invalid status in %s.' % build_number_and_status)
+
+ return (parsed_build_number, status)
+
+
+def _parse_name(revision_and_bot_name):
+ parsed_name = revision_and_bot_name.split('--')
+ if len(parsed_name) != 2:
+ raise ValueError('Malformed name string %s.' % revision_and_bot_name)
+
+ revision = parsed_name[0]
+ bot_name = parsed_name[1]
+ if '\n' in bot_name:
+ raise ValueError('Bot name %s can not contain newlines.' % bot_name)
+
+ return (int(revision), bot_name)
+
+
+def _delete_all_with_revision(revision, build_status_root):
+ query_result = db.GqlQuery('SELECT * FROM BuildStatusData '
+ 'WHERE revision = :1 AND ANCESTOR IS :2',
+ revision, build_status_root)
+ for entry in query_result:
+ entry.delete()
+
+
+class AddBuildStatusData(oauth_post_request_handler.OAuthPostRequestHandler):
+ """Used to report build status data.
+
+ Build status data is reported as a POST request. The POST request, aside
+ from the required oauth_* parameters should contain name-value entries that
+ abide by the following rules:
+
+ 1) The name should be on the form <revision>--<bot name>, for instance
+ 1568--Win32Release.
+ 2) The value should be on the form <build number>--<status>, for instance
+ 553--OK, 554--building. The status is permitted to be failed, OK or
+ building.
+
+ Data is keyed by revision. This handler will delete all data from a revision
+ if data with that revision is present in the current update, since we
+ assume that more recent data is always better data. We also assume that
+ an update always has complete information on a revision (e.g. the status
+ for all the bots are reported in each update).
+
+ In particular the revision arrangement solves the problem when the latest
+ revision reports 'building' for a bot. Had we not deleted the old revision
+ we would first store a 'building' status for that bot and revision, and
+ later store a 'OK' or 'failed' status for that bot and revision. This is
+ undesirable since we don't want multiple statuses for one bot-revision
+ combination. Now we will effectively update the bot's status instead.
+ """
+
+ def _parse_and_store_data(self):
+ build_status_root = _ensure_build_status_root_exists()
+ build_status_data = _filter_oauth_parameters(self.request.arguments())
+
+ db.run_in_transaction(self._parse_and_store_data_in_transaction,
+ build_status_root, build_status_data)
+
+ def _parse_and_store_data_in_transaction(self, build_status_root,
+ build_status_data):
+
+ encountered_revisions = set()
+ for revision_and_bot_name in build_status_data:
+ build_number_and_status = self.request.get(revision_and_bot_name)
+
+ try:
+ (build_number, status) = _parse_status(build_number_and_status)
+ (revision, bot_name) = _parse_name(revision_and_bot_name)
+ except ValueError as error:
+ logger.warn('Invalid parameter in request: %s.' % error)
+ self.response.set_status(400)
+ return
+
+ if revision not in encountered_revisions:
+ # There's new data on this revision in this update, so clear all status
+ # entries with that revision. Only do this once when we first encounter
+ # the revision.
+ _delete_all_with_revision(revision, build_status_root)
+ encountered_revisions.add(revision)
+
+ # Finally, write the item.
+ item = BuildStatusData(parent=build_status_root,
+ bot_name=bot_name,
+ revision=revision,
+ build_number=build_number,
+ status=status)
+ item.put()
+
+ request_posix_timestamp = float(self.request.get('oauth_timestamp'))
+ request_datetime = datetime.datetime.fromtimestamp(request_posix_timestamp)
+ build_status_root.last_updated_at = request_datetime
+ build_status_root.put()
+
diff --git a/trunk/tools/quality_tracking/dashboard/add_coverage_data.py b/trunk/tools/quality_tracking/dashboard/add_coverage_data.py
new file mode 100644
index 0000000..1c9d9da
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/add_coverage_data.py
@@ -0,0 +1,66 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Implements a handler for adding coverage data."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import datetime
+import logging
+
+from google.appengine.ext import db
+
+import oauth_post_request_handler
+
+class CoverageData(db.Model):
+ """This represents one coverage report from the build bot."""
+ date = db.DateTimeProperty(required=True)
+ line_coverage = db.FloatProperty(required=True)
+ function_coverage = db.FloatProperty(required=True)
+
+
+def _parse_percentage(string_value):
+ percentage = float(string_value)
+ if percentage < 0.0 or percentage > 100.0:
+ raise ValueError('%s is not a valid percentage.' % string_value)
+ return percentage
+
+
+class AddCoverageData(oauth_post_request_handler.OAuthPostRequestHandler):
+ """Used to report coverage data.
+
+ Coverage data is reported as a POST request and should contain, aside from
+ the regular oauth_* parameters, these values:
+
+ date: The POSIX timestamp for when the coverage observation was made.
+ line_coverage: A float percentage in the interval 0-100.0.
+ function_coverage: A float percentage in the interval 0-100.0.
+ """
+
+ def _parse_and_store_data(self):
+ try:
+ posix_time = int(self.request.get('date'))
+ parsed_date = datetime.datetime.fromtimestamp(posix_time)
+
+ line_coverage_string = self.request.get('line_coverage')
+ line_coverage = _parse_percentage(line_coverage_string)
+ function_coverage_string = self.request.get('function_coverage')
+ function_coverage = _parse_percentage(function_coverage_string)
+
+ except ValueError as error:
+ logger.warn('Invalid parameter in request: %s.' % error)
+ self.response.set_status(400)
+ return
+
+ item = CoverageData(date=parsed_date,
+ line_coverage=line_coverage,
+ function_coverage=function_coverage)
+ item.put()
+
diff --git a/trunk/tools/quality_tracking/dashboard/app.yaml b/trunk/tools/quality_tracking/dashboard/app.yaml
new file mode 100644
index 0000000..d06a813
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/app.yaml
@@ -0,0 +1,24 @@
+application: webrtc-dashboard
+version: 1
+runtime: python27
+api_version: 1
+threadsafe: false
+
+handlers:
+# Serve stylesheets statically.
+- url: /stylesheets
+ static_dir: stylesheets
+# This magic file is here to prove to the Google Account Domain Management
+# that we own this domain. It needs to stay there so the domain management
+# doesn't get suspicious.
+- url: /google403c95edcde16425.html
+ static_files: static/google403c95edcde16425.html
+ upload: static/google403c95edcde16425.html
+
+# Note: tests should be disabled in production.
+# - url: /test.*
+# script: gaeunit.py
+
+# Redirect all other requests to our dynamic handlers.
+- url: /.*
+ script: dashboard.app
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/dashboard.py b/trunk/tools/quality_tracking/dashboard/dashboard.py
new file mode 100644
index 0000000..5cc5a61
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/dashboard.py
@@ -0,0 +1,57 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Implements the quality tracker dashboard and reporting facilities."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+from google.appengine.ext.webapp import template
+import webapp2
+
+import add_build_status_data
+import add_coverage_data
+import load_build_status
+import load_coverage
+
+
+class ShowDashboard(webapp2.RequestHandler):
+ """Shows the dashboard page.
+
+ The page is shown by grabbing data we have stored previously
+ in the App Engine database using the AddCoverageData handler.
+ """
+
+ def get(self):
+ build_status_loader = load_build_status.BuildStatusLoader()
+ build_status_data = build_status_loader.load_build_status_data()
+ last_updated_at = build_status_loader.load_last_modified_at()
+ if last_updated_at is None:
+ self._show_error_page("No data has yet been uploaded to the dashboard.")
+ return
+
+ last_updated_at = last_updated_at.strftime("%Y-%m-%d %H:%M")
+ lkgr = build_status_loader.compute_lkgr()
+
+ coverage_loader = load_coverage.CoverageDataLoader()
+ coverage_json_data = coverage_loader.load_coverage_json_data()
+
+ page_template_filename = 'templates/dashboard_template.html'
+ self.response.write(template.render(page_template_filename, vars()))
+
+ def _show_error_page(self, error_message):
+ self.response.write('<html><body>%s</body></html>' % error_message)
+
+
+app = webapp2.WSGIApplication([('/', ShowDashboard),
+ ('/add_coverage_data',
+ add_coverage_data.AddCoverageData),
+ ('/add_build_status_data',
+ add_build_status_data.AddBuildStatusData)],
+ debug=True)
diff --git a/trunk/tools/quality_tracking/dashboard/gaeunit.py b/trunk/tools/quality_tracking/dashboard/gaeunit.py
new file mode 120000
index 0000000..a93f6bd
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/gaeunit.py
@@ -0,0 +1 @@
+../../../third_party/gaeunit/gaeunit.py
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/gviz_api.py b/trunk/tools/quality_tracking/dashboard/gviz_api.py
new file mode 120000
index 0000000..f3a22fc
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/gviz_api.py
@@ -0,0 +1 @@
+../../../third_party/google-visualization-python/gviz_api.py
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/load_build_status.py b/trunk/tools/quality_tracking/dashboard/load_build_status.py
new file mode 100644
index 0000000..9a0eebb
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/load_build_status.py
@@ -0,0 +1,120 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Loads build status data for the dashboard."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+from google.appengine.ext import db
+
+
+def _status_not_ok(status):
+ return status not in ('OK', 'warnings')
+
+
+def _all_ok(statuses):
+ return filter(_status_not_ok, statuses) == []
+
+
+def _get_first_entry(iterable):
+ if not iterable:
+ return None
+ for item in iterable:
+ return item
+
+
+class BuildStatusLoader:
+ """ Loads various build status data from the database."""
+
+ def load_build_status_data(self):
+ """Returns the latest conclusive build status for each bot.
+
+ The statuses OK, failed and warnings are considered to be conclusive.
+
+ The two most recent revisions are considered. The set of bots returned
+ will therefore be the bots that were reported the two most recent
+ revisions. This script will therefore adapt automatically to any changes
+ in the set of available bots.
+
+ Returns:
+ A list of BuildStatusData entities with one entity per bot.
+ """
+
+ build_status_entries = db.GqlQuery('SELECT * '
+ 'FROM BuildStatusData '
+ 'ORDER BY revision DESC ')
+
+ bots_to_latest_conclusive_entry = dict()
+ for entry in build_status_entries:
+ if entry.status == 'building':
+ # The 'building' status it not conclusive, so discard this entry and
+ # pick up the entry for this bot on the next revision instead. That
+ # entry is guaranteed to have a status != 'building' since a bot cannot
+ # be building two revisions simultaneously.
+ continue
+ if bots_to_latest_conclusive_entry.has_key(entry.bot_name):
+ # We've already determined this bot's status.
+ continue
+
+ bots_to_latest_conclusive_entry[entry.bot_name] = entry
+
+ return bots_to_latest_conclusive_entry.values()
+
+ def load_last_modified_at(self):
+ build_status_root = db.GqlQuery('SELECT * '
+ 'FROM BuildStatusRoot').get()
+ if not build_status_root:
+ # Operating on completely empty database
+ return None
+
+ return build_status_root.last_updated_at
+
+ def compute_lkgr(self):
+ """ Finds the most recent revision for which all bots are green.
+
+ Returns:
+ The last known good revision (as an integer) or None if there
+ is no green revision in the database.
+
+ Implementation note: The data store fetches stuff as we go, so we won't
+ read in the whole status table unless the LKGR is right at the end or
+ we don't have a LKGR.
+ """
+ build_status_entries = db.GqlQuery('SELECT * '
+ 'FROM BuildStatusData '
+ 'ORDER BY revision DESC ')
+
+ first_entry = _get_first_entry(build_status_entries)
+ if first_entry is None:
+ # No entries => no LKGR
+ return None
+
+ current_lkgr = first_entry.revision
+ statuses_for_current_lkgr = [first_entry.status]
+
+ for entry in build_status_entries:
+ if current_lkgr == entry.revision:
+ statuses_for_current_lkgr.append(entry.status)
+ else:
+ # Starting on new revision, check previous revision.
+ if _all_ok(statuses_for_current_lkgr):
+ # All bots are green; LKGR found.
+ return current_lkgr
+ else:
+ # Not all bots are green, so start over on the next revision.
+ current_lkgr = entry.revision
+ statuses_for_current_lkgr = [entry.status]
+
+ if _all_ok(statuses_for_current_lkgr):
+ # There was only one revision and it was OK.
+ return current_lkgr
+
+ # There is no all-green revision in the database.
+ return None
diff --git a/trunk/tools/quality_tracking/dashboard/load_coverage.py b/trunk/tools/quality_tracking/dashboard/load_coverage.py
new file mode 100644
index 0000000..eafed3b
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/load_coverage.py
@@ -0,0 +1,39 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Loads coverage data from the database."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+from google.appengine.ext import db
+import gviz_api
+
+
+class CoverageDataLoader:
+ """ Loads coverage data from the database."""
+
+ def load_coverage_json_data(self):
+ coverage_entries = db.GqlQuery('SELECT * '
+ 'FROM CoverageData '
+ 'ORDER BY date ASC')
+ data = []
+ for coverage_entry in coverage_entries:
+ data.append({'date': coverage_entry.date,
+ 'line_coverage': coverage_entry.line_coverage,
+ 'function_coverage': coverage_entry.function_coverage,
+ })
+
+ description = {
+ 'date': ('datetime', 'Date'),
+ 'line_coverage': ('number', 'Line Coverage'),
+ 'function_coverage': ('number', 'Function Coverage')
+ }
+ coverage_data = gviz_api.DataTable(description, data)
+ return coverage_data.ToJSon(order_by='date')
diff --git a/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py b/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py
new file mode 100644
index 0000000..416e1b7
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py
@@ -0,0 +1,68 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Provides a OAuth request handler base class."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+from google.appengine.api import oauth
+import logging
+import webapp2
+
+
+class UserNotAuthenticatedException(Exception):
+ """Gets thrown if a user is not permitted to store data."""
+ pass
+
+
+class OAuthPostRequestHandler(webapp2.RequestHandler):
+ """Works like a normal request handler but adds OAuth authentication.
+
+ This handler will expect a proper OAuth request over POST. This abstract
+ class deals with the authentication but leaves user-defined data handling
+ to its subclasses. Subclasses should not implement the post() method but
+ the _parse_and_store_data() method. Otherwise they may act like regular
+ request handlers. Subclasses should NOT override the get() method.
+
+ The handler will accept an OAuth request if it is correctly formed and
+ the consumer is acting on behalf of an administrator for the dashboard.
+ """
+
+ def post(self):
+ try:
+ self._authenticate_user()
+ except UserNotAuthenticatedException as exception:
+ logging.warn('Failed to authenticate: %s.' % exception)
+ self.response.set_status(403)
+ return
+
+ # Do the actual work.
+ self._parse_and_store_data()
+
+ def _parse_and_store_data(self):
+ """Reads data from POST request and responds accordingly."""
+ raise NotImplementedError('You must override this method!')
+
+ def _authenticate_user(self):
+ try:
+ if oauth.is_current_user_admin():
+ # The user on whose behalf we are acting is indeed an administrator
+ # of this application, so we're good to go.
+ logging.info('Authenticated on behalf of user %s.' %
+ oauth.get_current_user())
+ return
+ else:
+ raise UserNotAuthenticatedException('We are acting on behalf of '
+ 'user %s, but that user is not '
+ 'an administrator.' %
+ oauth.get_current_user())
+ except oauth.OAuthRequestError as exception:
+ raise UserNotAuthenticatedException('Invalid OAuth request: %s' %
+ exception.__class__.__name__)
diff --git a/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html b/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html
new file mode 100644
index 0000000..95c7e2d
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html
@@ -0,0 +1 @@
+google-site-verification: google403c95edcde16425.html
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css b/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css
new file mode 100644
index 0000000..c6eb679
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css
@@ -0,0 +1,45 @@
+/********************************************************************
+*
+* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+*
+* Use of this source code is governed by a BSD-style license
+* that can be found in the LICENSE file in the root of the source
+* tree. An additional intellectual property rights grant can be found
+* in the file PATENTS. All contributing project authors may
+* be found in the AUTHORS file in the root of the source tree.
+*
+*********************************************************************/
+
+.status_OK {
+ color: #FFFFFF;
+ background-color: #8fdf5f;
+}
+
+.status_failed {
+ color: #FFFFFF;
+ background-color: #e98080;
+}
+
+.status_building {
+ color: #666666;
+ background-color: #fffc6c;
+}
+
+.status_warnings {
+ color: #000000;
+ background-color: #FFC343;
+}
+
+.last_known_good_revision {
+ font-size: 800%;
+}
+
+.status_cell {
+ width: 100px;
+ text-align: center;
+}
+
+body {
+ margin-left: 35px;
+ margin-top: 25px;
+}
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html b/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html
new file mode 100644
index 0000000..ab019d5
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01//EN"
+ "http://www.w3.org/TR/html4/strict.dtd">
+<html>
+ <!--
+ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+
+ Use of this source code is governed by a BSD-style license
+ that can be found in the LICENSE file in the root of the source
+ tree. An additional intellectual property rights grant can be found
+ in the file PATENTS. All contributing project authors may
+ be found in the AUTHORS file in the root of the source tree.
+
+ Template file to be used to generate the WebRTC dashboard.
+ -->
+ <head>
+ <title>WebRTC Coverage Dashboard</title>
+ <meta http-equiv="refresh" content="60">
+ <link href="http://code.google.com/css/codesite.pack.04102009.css"
+ rel="stylesheet" type="text/css">
+ <link href="stylesheets/stylesheet.css"
+ rel="stylesheet" type="text/css">
+
+ <script src="https://www.google.com/jsapi" type="text/javascript"></script>
+ <script type="text/javascript">
+ google.load('visualization', '1', {packages:['table', 'corechart']});
+
+ google.setOnLoadCallback(drawTable);
+ function drawTable() {
+ /* Build data tables and views */
+ {% comment %}
+ Disable Django auto-escaping here since that will mess up our
+ coverage table JSON data otherwise.
+ {% endcomment %}
+ {% autoescape off %}
+ var coverage_data_table =
+ new google.visualization.DataTable({{ coverage_json_data }});
+ {% endautoescape %}
+
+ /* Display tables and charts */
+ var coverage_chart = new google.visualization.LineChart(
+ document.getElementById('table_div_coverage'));
+ coverage_chart.draw(coverage_data_table, {
+ colors: ['blue', 'red'],
+ vAxis: {title: 'Coverage'},
+ hAxis: {title: 'Date'},
+ width: 1200, height: 300,
+ });
+ }
+ </script>
+ </head>
+ <body>
+
+ <h1>WebRTC Quality Dashboard</h1>
+ <h2>Current Build Status</h2>
+ <div>(as of {{ last_updated_at }} UTC)</div>
+ <table>
+ <tr>
+ {% for entry in build_status_data %}
+ <th class="status_cell">{{ entry.bot_name }}</th>
+ {% endfor %}
+ </tr>
+ <tr>
+ {% for entry in build_status_data %}
+ <td title="Last built revision {{ entry.revision }}"
+ class="status_cell status_{{entry.status}}">
+ {{entry.status}}
+ </td>
+ {% endfor %}
+ </tr>
+ </table>
+ <p></p>
+
+ <h2>Last Known Good Revision (LKGR)</h2>
+ <div class="last_known_good_revision">
+ {% if lkgr %}
+ <a href="http://code.google.com/p/webrtc/source/detail?r={{ lkgr }}">
+ {{ lkgr }}</a>
+ {% else %}
+ ????
+ {% endif %}
+ </div>
+
+ <h2>Code Coverage History</h2>
+ <div id="table_div_coverage"></div>
+ </body>
+</html>
diff --git a/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py b/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py
new file mode 100755
index 0000000..56c9379
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py
@@ -0,0 +1,114 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import unittest
+from google.appengine.ext import db
+from google.appengine.ext import testbed
+
+from add_build_status_data import BuildStatusData
+import load_build_status
+
+class LoadBuildStatusTest(unittest.TestCase):
+ def setUp(self):
+ # First, create an instance of the Testbed class.
+ self.testbed = testbed.Testbed()
+ # Then activate the testbed, which prepares the service stubs for use.
+ self.testbed.activate()
+ # Next, declare which service stubs you want to use.
+ self.testbed.init_datastore_v3_stub()
+
+ def test_returns_latest_nonbuilding_entries_when_loading_build_status(self):
+ BuildStatusData(bot_name="Bot1", revision=17,
+ build_number=499, status="OK").put()
+ BuildStatusData(bot_name="Bot2", revision=17,
+ build_number=505, status="OK").put()
+ BuildStatusData(bot_name="Bot3", revision=17,
+ build_number=344, status="failed").put()
+ BuildStatusData(bot_name="Bot1", revision=18,
+ build_number=499, status="building").put()
+ BuildStatusData(bot_name="Bot2", revision=18,
+ build_number=505, status="failed").put()
+ BuildStatusData(bot_name="Bot3", revision=18,
+ build_number=344, status="OK").put()
+
+ loader = load_build_status.BuildStatusLoader()
+ result = loader.load_build_status_data()
+
+ self.assertEqual(3, len(result))
+
+ # We make no guarantees on order, but we can use the fact that the testbed
+ # is deterministic to evaluate that the corrects bots were selected like so:
+ self.assertEqual("Bot1", result[0].bot_name)
+ self.assertEqual(17, result[0].revision)
+ self.assertEqual("OK", result[0].status)
+
+ self.assertEqual("Bot3", result[1].bot_name)
+ self.assertEqual(18, result[1].revision)
+ self.assertEqual("OK", result[1].status)
+
+ self.assertEqual("Bot2", result[2].bot_name)
+ self.assertEqual(18, result[2].revision)
+ self.assertEqual("failed", result[2].status)
+
+ def test_returns_lkgr_for_single_green_revision(self):
+ BuildStatusData(bot_name="Bot1", revision=17,
+ build_number=499, status="OK").put()
+ BuildStatusData(bot_name="Bot2", revision=17,
+ build_number=505, status="OK").put()
+ BuildStatusData(bot_name="Bot3", revision=17,
+ build_number=344, status="OK").put()
+
+ loader = load_build_status.BuildStatusLoader()
+ self.assertEqual(17, loader.compute_lkgr())
+
+ def test_returns_correct_lkgr_with_most_recent_revision_failed(self):
+ BuildStatusData(bot_name="Bot1", revision=17,
+ build_number=499, status="OK").put()
+ BuildStatusData(bot_name="Bot2", revision=17,
+ build_number=505, status="OK").put()
+ BuildStatusData(bot_name="Bot3", revision=17,
+ build_number=344, status="OK").put()
+ BuildStatusData(bot_name="Bot1", revision=18,
+ build_number=499, status="OK").put()
+ BuildStatusData(bot_name="Bot2", revision=18,
+ build_number=505, status="failed").put()
+ BuildStatusData(bot_name="Bot3", revision=18,
+ build_number=344, status="OK").put()
+
+ loader = load_build_status.BuildStatusLoader()
+ self.assertEqual(17, loader.compute_lkgr())
+
+ def test_returns_none_if_no_revisions(self):
+ loader = load_build_status.BuildStatusLoader()
+ self.assertEqual(None, loader.compute_lkgr())
+
+ def test_returns_none_if_no_green_revisions(self):
+ BuildStatusData(bot_name="Bot2", revision=18,
+ build_number=505, status="failed").put()
+
+ loader = load_build_status.BuildStatusLoader()
+ self.assertEqual(None, loader.compute_lkgr())
+
+ def test_skips_partially_building_revisions(self):
+ BuildStatusData(bot_name="Bot1", revision=18,
+ build_number=499, status="building").put()
+ BuildStatusData(bot_name="Bot2", revision=18,
+ build_number=505, status="OK").put()
+ BuildStatusData(bot_name="Bot1", revision=17,
+ build_number=344, status="OK").put()
+
+ loader = load_build_status.BuildStatusLoader()
+ self.assertEqual(17, loader.compute_lkgr())
+
+
+if __name__ == '__main__':
+ unittest.main()
diff --git a/trunk/tools/quality_tracking/dashboard_connection.py b/trunk/tools/quality_tracking/dashboard_connection.py
new file mode 100644
index 0000000..9a6e30f
--- /dev/null
+++ b/trunk/tools/quality_tracking/dashboard_connection.py
@@ -0,0 +1,134 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Contains utilities for communicating with the dashboard."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import httplib
+import shelve
+import urlparse
+import oauth.oauth as oauth
+
+import constants
+
+
+class FailedToReadRequiredInputFile(Exception):
+ pass
+
+
+class FailedToReportToDashboard(Exception):
+ pass
+
+
+class DashboardConnection:
+ """Helper class for pushing data to the dashboard.
+
+ This class deals with most of details for accessing protected resources
+ (i.e. data-writing operations) on the dashboard. Such operations are
+ authenticated using OAuth. This class requires a consumer secret and a
+ access token.
+
+ The access token and consumer secrets are stored as files on disk in the
+ working directory of the scripts. Both files are created by the
+ request_oauth_permission script.
+ """
+
+ def __init__(self, consumer_key):
+ self.consumer_key_ = consumer_key
+
+ def read_required_files(self, consumer_secret_file, access_token_file):
+ """Reads required data for making OAuth requests.
+
+ Args:
+ consumer_secret_file: A shelve file with an entry consumer_secret
+ containing the consumer secret in string form.
+ access_token_file: A shelve file with an entry access_token
+ containing the access token in string form.
+ """
+ self.access_token_string_ = self._read_access_token(access_token_file)
+ self.consumer_secret_ = self._read_consumer_secret(consumer_secret_file)
+
+ def send_post_request(self, url, parameters):
+ """Sends an OAuth request for a protected resource in the dashboard.
+
+ Use this when you want to report new data to the dashboard. You must have
+ called the read_required_files method prior to calling this method, since
+ that method will read in the consumer secret and access token we need to
+ make the OAuth request. These concepts are described in the class
+ description.
+
+ The server is expected to respond with HTTP status 200 and a completely
+ empty response if the call failed. The server may put diagnostic
+ information in the response.
+
+ Args:
+ url: An absolute url within the dashboard domain, for example
+ http://webrtc-dashboard.appspot.com/add_coverage_data.
+ parameters: A dict which maps from POST parameter names to values.
+
+ Raises:
+ FailedToReportToDashboard: If the dashboard didn't respond
+ with HTTP 200 to our request or if the response is non-empty.
+ """
+ consumer = oauth.OAuthConsumer(self.consumer_key_, self.consumer_secret_)
+ access_token = oauth.OAuthToken.from_string(self.access_token_string_)
+
+ oauth_request = oauth.OAuthRequest.from_consumer_and_token(
+ consumer,
+ token=access_token,
+ http_method='POST',
+ http_url=url,
+ parameters=parameters)
+
+ signature_method_hmac_sha1 = oauth.OAuthSignatureMethod_HMAC_SHA1()
+ oauth_request.sign_request(signature_method_hmac_sha1, consumer,
+ access_token)
+
+ connection = httplib.HTTPConnection(constants.DASHBOARD_SERVER)
+
+ headers = {'Content-Type': 'application/x-www-form-urlencoded'}
+ connection.request('POST', url, body=oauth_request.to_postdata(),
+ headers=headers)
+
+ response = connection.getresponse()
+ connection.close()
+
+ if response.status != 200:
+ message = ('Failed to report to %s: got response %d (%s)' %
+ (url, response.status, response.reason))
+ raise FailedToReportToDashboard(message)
+
+ # The response content should be empty on success, so check that:
+ response_content = response.read()
+ if response_content:
+ message = ('Dashboard reported the following error: %s.' %
+ response_content)
+ raise FailedToReportToDashboard(message)
+
+ def _read_access_token(self, filename):
+ return self._read_shelve(filename, 'access_token')
+
+ def _read_consumer_secret(self, filename):
+ return self._read_shelve(filename, 'consumer_secret')
+
+ def _read_shelve(self, filename, key):
+ input_file = shelve.open(filename)
+
+ if not input_file.has_key(key):
+ raise FailedToReadRequiredInputFile('Missing correct %s file in current '
+ 'directory. You may have to run '
+ 'request_oauth_permission.py.' %
+ filename)
+
+ result = input_file[key]
+ input_file.close()
+
+ return result
diff --git a/trunk/tools/quality_tracking/oauth2 b/trunk/tools/quality_tracking/oauth2
new file mode 120000
index 0000000..a4d1dde
--- /dev/null
+++ b/trunk/tools/quality_tracking/oauth2
@@ -0,0 +1 @@
+../../third_party/oauth2/oauth2/
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/request_oauth_permission.py b/trunk/tools/quality_tracking/request_oauth_permission.py
new file mode 100755
index 0000000..fb97738
--- /dev/null
+++ b/trunk/tools/quality_tracking/request_oauth_permission.py
@@ -0,0 +1,142 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""This script request an access token from the appengine running the dashboard.
+
+ The script is intended to be run manually whenever we wish to change which
+ dashboard administrator we act on behalf of when running the
+ track_coverage.py script. For example, this will be useful if the current
+ dashboard administrator leaves the project. This script can also be used to
+ launch a new dashboard if that is desired.
+
+ This script should be run on the build bot which runs the track_coverage.py
+ script. This script will present a link during its execution, which the new
+ administrator should follow and then click approve on the web page that
+ appears. The new administrator should have admin rights on the coverage
+ dashboard, otherwise the track_* scripts will not work.
+
+ If successful, this script will write the access token to a file access.token
+ in the current directory, which later can be read by the track_* scripts.
+ The token is stored in string form (as reported by the web server) using the
+ shelve module. The consumer secret passed in as an argument to this script
+ will also similarly be stored in a file consumer.secret. The shelve keys
+ will be 'access_token' and 'consumer_secret', respectively.
+"""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import shelve
+import sys
+import urlparse
+import oauth2 as oauth
+
+import constants
+
+
+class FailedToRequestPermissionException(Exception):
+ pass
+
+
+def _ensure_token_response_is_200(response, queried_url, token_type):
+ if response.status != 200:
+ raise FailedToRequestPermissionException('Failed to request %s from %s: '
+ 'received status %d, reason %s.' %
+ (token_type,
+ queried_url,
+ response.status,
+ response.reason))
+
+
+def _request_unauthorized_token(consumer, request_token_url):
+ """Requests the initial token from the dashboard service.
+
+ Given that the response from the server is correct, we will return a
+ dictionary containing oauth_token and oauth_token_secret mapped to the
+ token and secret value, respectively.
+ """
+ client = oauth.Client(consumer)
+
+ try:
+ response, content = client.request(request_token_url, 'POST')
+ except AttributeError as error:
+ # This catch handler is here since we'll get very confusing messages
+ # if the target server is down for some reason.
+ raise FailedToRequestPermissionException('Failed to request token: '
+ 'the dashboard is likely down.',
+ error)
+
+ _ensure_token_response_is_200(response, request_token_url,
+ 'unauthorized token')
+
+ return dict(urlparse.parse_qsl(content))
+
+
+def _ask_user_to_authorize_us(unauthorized_token):
+ """This function will block until the user enters y + newline."""
+ print 'Go to the following link in your browser:'
+ print '%s?oauth_token=%s' % (constants.AUTHORIZE_TOKEN_URL,
+ unauthorized_token['oauth_token'])
+
+ accepted = 'n'
+ while accepted.lower() != 'y':
+ accepted = raw_input('Have you authorized me yet? (y/n) ')
+
+
+def _request_access_token(consumer, unauthorized_token):
+ token = oauth.Token(unauthorized_token['oauth_token'],
+ unauthorized_token['oauth_token_secret'])
+ client = oauth.Client(consumer, token)
+ response, content = client.request(constants.ACCESS_TOKEN_URL, 'POST')
+
+ _ensure_token_response_is_200(response, constants.ACCESS_TOKEN_URL,
+ 'access token')
+
+ return content
+
+
+def _write_access_token_to_file(access_token, filename):
+ output = shelve.open(filename)
+ output['access_token'] = access_token
+ output.close()
+
+ print 'Wrote the access token to the file %s.' % filename
+
+
+def _write_consumer_secret_to_file(consumer_secret, filename):
+ output = shelve.open(filename)
+ output['consumer_secret'] = consumer_secret
+ output.close()
+
+ print 'Wrote the consumer secret to the file %s.' % filename
+
+
+def _main():
+ if len(sys.argv) != 2:
+ print ('Usage: %s <consumer secret>.\n\nThe consumer secret is an OAuth '
+ 'concept and is obtained from the Google Accounts domain dashboard.'
+ % sys.argv[0])
+ return
+
+ consumer_secret = sys.argv[1]
+ consumer = oauth.Consumer(constants.CONSUMER_KEY, consumer_secret)
+
+ unauthorized_token = _request_unauthorized_token(consumer,
+ constants.REQUEST_TOKEN_URL)
+
+ _ask_user_to_authorize_us(unauthorized_token)
+
+ access_token_string = _request_access_token(consumer, unauthorized_token)
+
+ _write_access_token_to_file(access_token_string, constants.ACCESS_TOKEN_FILE)
+ _write_consumer_secret_to_file(consumer_secret,
+ constants.CONSUMER_SECRET_FILE)
+
+if __name__ == '__main__':
+ _main()
diff --git a/trunk/tools/quality_tracking/tgrid_parser.py b/trunk/tools/quality_tracking/tgrid_parser.py
new file mode 100644
index 0000000..2b674d5
--- /dev/null
+++ b/trunk/tools/quality_tracking/tgrid_parser.py
@@ -0,0 +1,73 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Contains functions for parsing the build master's transposed grid page."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import re
+
+
+class FailedToParseBuildStatus(Exception):
+ pass
+
+
+def _parse_builds(revision, html):
+ """Parses the bot list, which is a sequence of <td></td> lines.
+
+ Example input:
+ <td class="build success"><a href="builders/Android/builds/119">OK</a></td>
+ The first regular expression group captures Android, second 119, third OK.
+ """
+ result = {}
+
+ for match in re.finditer('<td.*?>.*?<a href="builders/(.+?)/builds/(\d+)">'
+ '(OK|failed|building|warnings)</a>.*?</td>',
+ html, re.DOTALL):
+ revision_and_bot_name = revision + "--" + match.group(1)
+ build_number_and_status = match.group(2) + "--" + match.group(3)
+
+ result[revision_and_bot_name] = build_number_and_status
+
+ return result
+
+
+def parse_tgrid_page(html):
+ """Parses the build master's tgrid page.
+
+ Example input:
+ <tr>
+ <td valign="bottom" class="sourcestamp">1568</td>
+ LIST OF BOTS
+ </tr>
+ The first regular expression group captures 1568, second group captures
+ everything in LIST OF BOTS. The list of bots is then passed into a
+ separate function for parsing.
+
+ Args:
+ html: The raw HTML from the tgrid page.
+
+ Returns: A dictionary with <svn revision>--<bot name> mapped to
+ <bot build number>--<status>, where status is either OK, failed,
+ building or warnings.
+ """
+ result = {}
+
+ for match in re.finditer('<td.*?class="sourcestamp">(\d+)</td>(.*?)</tr>',
+ html, re.DOTALL):
+ revision = match.group(1)
+ builds_for_revision_html = match.group(2)
+ result.update(_parse_builds(revision, builds_for_revision_html))
+
+ if not result:
+ raise FailedToParseBuildStatus('Could not find any build statuses in %s.' %
+ html)
+
+ return result
diff --git a/trunk/tools/quality_tracking/tgrid_parser_test.py b/trunk/tools/quality_tracking/tgrid_parser_test.py
new file mode 100755
index 0000000..8b15c90
--- /dev/null
+++ b/trunk/tools/quality_tracking/tgrid_parser_test.py
@@ -0,0 +1,207 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Contains functions for parsing the build master's transposed grid page."""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import unittest
+
+import tgrid_parser
+
+
+SAMPLE_FILE = """
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN"
+ "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
+
+<html
+ xmlns="http://www.w3.org/1999/xhtml"
+ lang="en"
+ xml:lang="en">
+<head>
+ <title>Buildbot</title>
+ <link href="buildbot.css" rel="stylesheet" type="text/css" />
+</head>
+
+<body vlink="#800080">
+<table class="Grid" border="0" cellspacing="0">
+<tr>
+<td valign="bottom" class="sourcestamp">1570</td>
+<td class="build warnings"><a href="builders/Chrome/builds/109">warnings</a>
+<br />
+make chrome</td>
+<td class="build success">
+ <a href="builders/Android/builds/121">OK</a></td>
+<td class="build success">
+ <a href="builders/ChromeOS/builds/578">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux32bitDBG/builds/564">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux32bitRelease/builds/684">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitDBG/builds/680">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitDBG-GCC4.6/builds/5">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitRelease/builds/570">OK</a></td>
+<td class="build success">
+ <a href="builders/LinuxCLANG/builds/259">OK</a></td>
+<td class="build success">
+ <a href="builders/LinuxVideoTest/builds/345">OK</a></td>
+<td class="build success">
+ <a href="builders/MacOS/builds/670">OK</a></td>
+<td class="build success">
+ <a href="builders/Win32Debug/builds/432">OK</a></td>
+<td class="build success">
+ <a href="builders/Win32Release/builds/440">OK</a></td>
+</tr>
+<tr>
+<td valign="bottom" class="sourcestamp">1571</td>
+<td class="build warnings"><a href="builders/Chrome/builds/109">warnings</a>
+<br />
+make chrome</td>
+<td class="build success">
+ <a href="builders/Android/builds/122">OK</a></td>
+<td class="build success">
+ <a href="builders/ChromeOS/builds/579">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux32bitDBG/builds/565">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux32bitRelease/builds/685">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitDBG/builds/681">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitDBG-GCC4.6/builds/6">OK</a></td>
+<td class="build success">
+ <a href="builders/Linux64bitRelease/builds/571">OK</a></td>
+<td class="build success">
+ <a href="builders/LinuxCLANG/builds/260">OK</a></td>
+<td class="build failure">
+ <a href="builders/LinuxVideoTest/builds/346">failed</a><br />
+voe_auto_test</td>
+<td class="build success">
+ <a href="builders/MacOS/builds/671">OK</a></td>
+<td class="build running">
+ <a href="builders/Win32Debug/builds/441">building</a></td>
+<td class="build success">
+ <a href="builders/Win32Release/builds/441">OK</a></td>
+</tr>
+</table>
+</body>
+</html>
+"""
+
+MINIMAL_OK = """
+<tr>
+<td valign="bottom" class="sourcestamp">1570</td>
+<td class="build success">
+<a href="builders/Android/builds/121">OK</a></td>
+</tr>
+"""
+
+MINIMAL_FAIL = """
+<tr>
+<td valign="bottom" class="sourcestamp">1573</td>
+<td class="build failure">
+<a href="builders/LinuxVideoTest/builds/347">failed</a><br />
+voe_auto_test</td>
+</tr>
+"""
+
+MINIMAL_BUILDING = """
+<tr>
+<td valign="bottom" class="sourcestamp">1576</td>
+<td class="build running">
+<a href="builders/Win32Debug/builds/434">building</a></td>
+voe_auto_test</td>
+</tr>
+"""
+
+MINIMAL_WARNED = """
+<tr>
+<td valign="bottom" class="sourcestamp">1576</td>
+<td class="build warnings">
+<a href="builders/Chrome/builds/109">warnings</a><br />
+make chrome</td>
+</tr>
+"""
+
+class TGridParserTest(unittest.TestCase):
+ def test_parser_throws_exception_on_empty_html(self):
+ self.assertRaises(tgrid_parser.FailedToParseBuildStatus,
+ tgrid_parser.parse_tgrid_page, '');
+
+ def test_parser_finds_successful_bot(self):
+ result = tgrid_parser.parse_tgrid_page(MINIMAL_OK)
+
+ self.assertEqual(1, len(result), 'There is only one bot in the sample.')
+ first_mapping = result.items()[0]
+
+ self.assertEqual('1570--Android', first_mapping[0])
+ self.assertEqual('121--OK', first_mapping[1])
+
+ def test_parser_finds_failed_bot(self):
+ result = tgrid_parser.parse_tgrid_page(MINIMAL_FAIL)
+
+ self.assertEqual(1, len(result), 'There is only one bot in the sample.')
+ first_mapping = result.items()[0]
+
+ self.assertEqual('1573--LinuxVideoTest', first_mapping[0])
+ self.assertEqual('347--failed', first_mapping[1])
+
+ def test_parser_finds_building_bot(self):
+ result = tgrid_parser.parse_tgrid_page(MINIMAL_BUILDING)
+
+ self.assertEqual(1, len(result), 'There is only one bot in the sample.')
+ first_mapping = result.items()[0]
+
+ self.assertEqual('1576--Win32Debug', first_mapping[0])
+ self.assertEqual('434--building', first_mapping[1])
+
+ def test_parser_finds_warned_bot(self):
+ result = tgrid_parser.parse_tgrid_page(MINIMAL_WARNED)
+
+ self.assertEqual(1, len(result), 'There is only one bot in the sample.')
+ first_mapping = result.items()[0]
+
+ self.assertEqual('1576--Chrome', first_mapping[0])
+ self.assertEqual('109--warnings', first_mapping[1])
+
+ def test_parser_finds_all_bots_and_revisions(self):
+ result = tgrid_parser.parse_tgrid_page(SAMPLE_FILE)
+
+ # 2 * 13 = 26 bots in sample
+ self.assertEqual(26, len(result))
+
+ # Make some samples
+ self.assertTrue(result.has_key('1570--ChromeOS'))
+ self.assertEquals('578--OK', result['1570--ChromeOS'])
+
+ self.assertTrue(result.has_key('1570--Chrome'))
+ self.assertEquals('109--warnings', result['1570--Chrome'])
+
+ self.assertTrue(result.has_key('1570--LinuxCLANG'))
+ self.assertEquals('259--OK', result['1570--LinuxCLANG'])
+
+ self.assertTrue(result.has_key('1570--Win32Release'))
+ self.assertEquals('440--OK', result['1570--Win32Release'])
+
+ self.assertTrue(result.has_key('1571--ChromeOS'))
+ self.assertEquals('579--OK', result['1571--ChromeOS'])
+
+ self.assertTrue(result.has_key('1571--LinuxVideoTest'))
+ self.assertEquals('346--failed', result['1571--LinuxVideoTest'])
+
+ self.assertTrue(result.has_key('1571--Win32Debug'))
+ self.assertEquals('441--building', result['1571--Win32Debug'])
+
+
+if __name__ == '__main__':
+ unittest.main()
diff --git a/trunk/tools/quality_tracking/track_build_status.py b/trunk/tools/quality_tracking/track_build_status.py
new file mode 100755
index 0000000..1ddf140
--- /dev/null
+++ b/trunk/tools/quality_tracking/track_build_status.py
@@ -0,0 +1,57 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""This script checks the current build status on the master and submits
+ it to the dashboard. It is adapted to build bot version 0.7.12.
+"""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+
+import httplib
+
+import constants
+import dashboard_connection
+import tgrid_parser
+
+
+class FailedToGetStatusFromMaster(Exception):
+ pass
+
+
+def _download_and_parse_build_status():
+ connection = httplib.HTTPConnection(constants.BUILD_MASTER_SERVER)
+ connection.request('GET', constants.BUILD_MASTER_TRANSPOSED_GRID_URL)
+ response = connection.getresponse()
+
+ if response.status != 200:
+ raise FailedToGetStatusFromMaster(('Failed to get build status from master:'
+ ' got status %d, reason %s.' %
+ (response.status, response.reason)))
+
+ full_response = response.read()
+ connection.close()
+
+ return tgrid_parser.parse_tgrid_page(full_response)
+
+
+def _main():
+ dashboard = dashboard_connection.DashboardConnection(constants.CONSUMER_KEY)
+ dashboard.read_required_files(constants.CONSUMER_SECRET_FILE,
+ constants.ACCESS_TOKEN_FILE)
+
+ bot_to_status_mapping = _download_and_parse_build_status()
+
+ dashboard.send_post_request(constants.ADD_BUILD_STATUS_DATA_URL,
+ bot_to_status_mapping)
+
+
+if __name__ == '__main__':
+ _main()
diff --git a/trunk/tools/quality_tracking/track_coverage.py b/trunk/tools/quality_tracking/track_coverage.py
new file mode 100755
index 0000000..a46ddfc
--- /dev/null
+++ b/trunk/tools/quality_tracking/track_coverage.py
@@ -0,0 +1,119 @@
+#!/usr/bin/env python
+#-*- coding: utf-8 -*-
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""This script grabs and reports coverage information.
+
+ It grabs coverage information from the latest Linux 32-bit build and
+ pushes it to the coverage tracker, enabling us to track code coverage
+ over time. This script is intended to run on the 32-bit Linux slave.
+
+ This script requires an access.token file in the current directory, as
+ generated by the request_oauth_permission.py script. It also expects a file
+ customer.secret with a single line containing the customer secret. The
+ customer secret is an OAuth concept and is received when one registers the
+ application with the App Engine running the dashboard.
+
+ The script assumes that all coverage data is stored under
+ /home/<build bot user>/www.
+"""
+
+__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
+
+import os
+import re
+import time
+
+import constants
+import dashboard_connection
+
+
+class FailedToParseCoverageHtml(Exception):
+ pass
+
+
+class CouldNotFindCoverageDirectory(Exception):
+ pass
+
+
+def _find_latest_32bit_debug_build(www_directory_contents, coverage_www_dir):
+ """Finds the latest 32-bit coverage directory in the directory listing.
+
+ Coverage directories have the form Linux32bitDBG_<number>. There may be
+ other directories in the list though, for instance for other build
+ configurations.
+ """
+
+ # This sort ensures we will encounter the directory with the highest number
+ # first.
+ www_directory_contents.sort(reverse=True)
+
+ for entry in www_directory_contents:
+ match = re.match('Linux32DBG_\d+', entry)
+ if match is not None:
+ return entry
+
+ raise CouldNotFindCoverageDirectory('Error: Found no 32-bit '
+ 'debug build in directory %s.' %
+ coverage_www_dir)
+
+
+def _grab_coverage_percentage(label, index_html_contents):
+ """Extracts coverage from a LCOV coverage report.
+
+ Grabs coverage by assuming that the label in the coverage HTML report
+ is close to the actual number and that the number is followed by a space
+ and a percentage sign.
+ """
+ match = re.search('<td[^>]*>' + label + '</td>.*?(\d+\.\d) %',
+ index_html_contents, re.DOTALL)
+ if match is None:
+ raise FailedToParseCoverageHtml('Missing coverage at label "%s".' % label)
+
+ try:
+ return float(match.group(1))
+ except ValueError:
+ raise FailedToParseCoverageHtml('%s is not a float.' % match.group(1))
+
+
+def _report_coverage_to_dashboard(dashboard, now, line_coverage,
+ function_coverage):
+ parameters = {'date': '%d' % now,
+ 'line_coverage': '%f' % line_coverage,
+ 'function_coverage': '%f' % function_coverage
+ }
+
+ dashboard.send_post_request(constants.ADD_COVERAGE_DATA_URL, parameters)
+
+
+def _main():
+ dashboard = dashboard_connection.DashboardConnection(constants.CONSUMER_KEY)
+ dashboard.read_required_files(constants.CONSUMER_SECRET_FILE,
+ constants.ACCESS_TOKEN_FILE)
+
+ www_dir_contents = os.listdir(BUILD_BOT_COVERAGE_WWW_DIRECTORY)
+ latest_build_directory = _find_latest_32bit_debug_build(www_dir_contents,
+ coverage_www_dir)
+
+ index_html_path = os.path.join(coverage_www_dir, latest_build_directory,
+ 'index.html')
+ index_html_file = open(index_html_path)
+ whole_file = index_html_file.read()
+
+ line_coverage = _grab_coverage_percentage('Lines:', whole_file)
+ function_coverage = _grab_coverage_percentage('Functions:', whole_file)
+ now = int(time.time())
+
+ _report_coverage_to_dashboard(dashboard, now, line_coverage,
+ function_coverage)
+
+
+if __name__ == '__main__':
+ _main()
+
diff --git a/trunk/tools/refactoring/addfileheader.py b/trunk/tools/refactoring/addfileheader.py
new file mode 100644
index 0000000..01c8a8b
--- /dev/null
+++ b/trunk/tools/refactoring/addfileheader.py
@@ -0,0 +1,163 @@
+#!/usr/bin/env python
+
+import stringmanipulation
+import filemanagement
+import sys
+
+extensions = ['.h','.cc','.c','.cpp']
+
+ignore_these = ['my_ignore_header.h']
+
+if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
+ print 'parameters are: directory [--commit]'
+ quit()
+
+directory = sys.argv[1];
+if(not filemanagement.pathexist(directory)):
+ print 'path ' + directory + ' does not exist'
+ quit()
+
+if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
+ print 'parameters are: parent directory extension new extension [--commit]'
+ quit()
+
+commit = False
+if(len(sys.argv) == 3):
+ commit = True
+
+files_to_fix = []
+for extension in extensions:
+ files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
+ extension))
+
+# Just steal the header from the template
+def fileheaderasstring():
+ template_file_name = 'license_template.txt'
+ if (not filemanagement.fileexist(template_file_name)):
+ print 'File ' + template_file_name + ' not found!'
+ quit()
+ template_file = open(template_file_name,'r')
+ return_string = ''
+ for line in template_file:
+ return_string += line
+ return return_string
+
+# Just steal the header from the template
+def fileheaderasarray():
+ template_file_name = 'license_template.txt'
+ if (not filemanagement.fileexist(template_file_name)):
+ print 'File ' + template_file_name + ' not found!'
+ quit()
+ template_file = open(template_file_name,'r')
+ return_value = []
+ for line in template_file:
+ return_value.append(line)
+ return return_value
+
+
+def findheader(path, file_name):
+ full_file_name = path + file_name
+ if (not filemanagement.fileexist(full_file_name)):
+ print 'File ' + file_name + ' not found!'
+ print 'Unexpected error!'
+ quit()
+ file_handle = open(full_file_name)
+ template_file_content = fileheaderasarray()
+ compare_content = []
+ # load the same number of lines from file as the fileheader
+ for index in range(len(template_file_content)):
+ line = file_handle.readline()
+ if (line == ''):
+ return False
+ compare_content.append(line)
+
+ while (True):
+ found = True
+ for index in range(len(template_file_content)):
+ line1 = template_file_content[index]
+ line2 = compare_content[index]
+ if(line1 != line2):
+ found = False
+ break
+ if (found):
+ return True
+ compare_content = compare_content[1:len(compare_content)]
+ line = file_handle.readline()
+ if (line == ''):
+ return False
+ compare_content.append(line)
+ return False
+
+# Used to store temporary result before flushing to real file when finished
+def temporaryfilename(old_file_name):
+ return old_file_name + '.deleteme'
+
+def updatefile(path, old_file_name):
+ full_old_file_name = path + old_file_name
+ if (not filemanagement.fileexist(full_old_file_name)):
+ print 'File ' + full_old_file_name + ' is not found.'
+ print 'Should not happen! Ever!'
+ quit()
+
+ full_temporary_file_name = path + temporaryfilename(old_file_name)
+
+ # Make sure that the files are closed by putting them out of scope
+ old_file = open(full_old_file_name,'r')
+ temporary_file = open(full_temporary_file_name,'w')
+
+ temporary_file.writelines(fileheaderasstring())
+ remove_whitespaces = True
+ for line in old_file:
+ if (remove_whitespaces and (len(line.split()) == 0)):
+ continue
+ else:
+ remove_whitespaces = False
+ temporary_file.writelines(line)
+ old_file.close()
+ temporary_file.close()
+
+ filemanagement.copyfile(full_old_file_name,full_temporary_file_name)
+ filemanagement.deletefile(full_temporary_file_name)
+
+
+failed_files = []
+skipped_files = []
+for index in range(len(files_to_fix)):
+ if(commit):
+ print (100*index)/len(files_to_fix)
+ path_dir = files_to_fix[index][0]
+ filename = files_to_fix[index][1]
+ is_ignore = False
+ for ignore_names in ignore_these:
+ if(filename == ignore_names):
+ is_ignore = True
+ break
+ if(is_ignore):
+ continue
+
+# Let the word copyright be our sanity, i.e. make sure there is only one
+# copy right occurance or report that there will be no change
+ if(filemanagement.findstringinfile(path_dir,filename,'Copyright') or
+ filemanagement.findstringinfile(path_dir,filename,'copyright') or
+ filemanagement.findstringinfile(path_dir,filename,'COPYRIGHT')):
+ if(findheader(path_dir,filename)):
+ skipped_files.append(path_dir + filename)
+ else:
+ failed_files.append(path_dir + filename)
+ continue
+
+ if (not commit):
+ print 'File ' + path_dir + filename + ' will be updated'
+ continue
+ updatefile(path_dir,filename)
+
+tense = 'will be'
+if (commit):
+ tense = 'has been'
+if (len(skipped_files) > 0):
+ print str(len(skipped_files)) + ' file(s) ' + tense + ' skipped since they already have the correct header'
+
+if (len(failed_files) > 0):
+ print 'Following files seem to have an invalid file header:'
+for line in failed_files:
+ print line
diff --git a/trunk/tools/refactoring/filemanagement.py b/trunk/tools/refactoring/filemanagement.py
new file mode 100644
index 0000000..4ff64ce
--- /dev/null
+++ b/trunk/tools/refactoring/filemanagement.py
@@ -0,0 +1,72 @@
+import fnmatch
+import os
+import stringmanipulation
+
+def fileexist( file_name ):
+ return os.path.isfile(file_name)
+
+def pathexist( path ):
+ return os.path.exists(path)
+
+def fixpath( path ):
+ return_value = path
+ if( return_value[len(return_value) - 1] != '/'):
+ return_value = return_value + '/'
+ return return_value
+
+def listallfilesinfolder( path, extension ):
+ matches = []
+ signature = '*' + extension
+ for root, dirnames, filenames in os.walk(path):
+ for filename in fnmatch.filter(filenames, signature):
+ matches.append([fixpath(root), filename])
+ return matches
+
+def copyfile(to_file, from_file):
+ if(not fileexist(from_file)):
+ return
+ command = 'cp -f ' + from_file + ' ' + to_file
+ os.system(command)
+ #print command
+
+def deletefile(file_to_delete):
+ if(not fileexist(file_to_delete)):
+ return
+ os.system('rm ' + file_to_delete)
+
+# very ugly but works, so keep for now
+def findstringinfile(path,file_name,search_string):
+ command = 'grep \'' + search_string + '\' ' + path + file_name + ' > deleteme.txt'
+ return_value = os.system(command)
+# print command
+ return (return_value == 0)
+
+def replacestringinfolder( path, old_string, new_string, extension ):
+ if(not stringmanipulation.isextension(extension)):
+ print 'failed to search and replace'
+ return
+ if(len(old_string) == 0):
+ print 'failed to search and replace'
+ return
+ find_command = 'ls '+ path + '/*' + extension
+ sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
+ '/g\' *' + extension
+ command_string = find_command + ' | xargs ' + sed_command + ' 2> deleteme.txt'
+ os.system(command_string)
+ #print command_string
+
+#find ./ -name "*.h" -type f | xargs -P 0 sed -i 's/process_thread_wrapper.h/process_thread.h/g' *.h deleteme.txt
+def replacestringinallsubfolders( old_string, new_string, extension):
+ if(not stringmanipulation.isextension(extension)):
+ print 'failed to search and replace'
+ return
+ if(len(old_string) == 0):
+ print 'failed to search and replace'
+ return
+
+ find_command = 'find ./ -name \"*' + extension + '\" -type f'
+ sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
+ '/g\' *' + extension
+ command_string = find_command + ' | xargs -P 0 ' + sed_command + ' 2> deleteme.txt'
+ os.system(command_string)
+ #print command_string
diff --git a/trunk/tools/refactoring/fixincludeguards.py b/trunk/tools/refactoring/fixincludeguards.py
new file mode 100644
index 0000000..0b56355
--- /dev/null
+++ b/trunk/tools/refactoring/fixincludeguards.py
@@ -0,0 +1,145 @@
+#!/usr/bin/env python
+
+import stringmanipulation
+import filemanagement
+import sys
+
+extensions = ['.h']
+
+ignore_these = ['my_ignore_header.h']
+
+if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
+ print 'parameters are: directory [--commit]'
+ quit()
+
+directory = sys.argv[1];
+if(not filemanagement.pathexist(directory)):
+ print 'path ' + directory + ' does not exist'
+ quit()
+
+if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
+ print 'parameters are: parent directory extension new extension [--commit]'
+ quit()
+
+commit = False
+if(len(sys.argv) == 3):
+ commit = True
+
+for extension in extensions:
+ files_to_fix = filemanagement.listallfilesinfolder(directory,\
+ extension)
+
+def buildincludeguardname(path,filename):
+ full_file_name = 'WEBRTC_' + path + filename
+ full_file_name = full_file_name.upper()
+ full_file_name = stringmanipulation.replaceoccurances(full_file_name, '/', '_')
+ full_file_name = stringmanipulation.replaceoccurances(full_file_name, '\\', '_')
+ full_file_name = stringmanipulation.replaceoccurances(full_file_name, '.', '_')
+ full_file_name += '_'
+ return full_file_name
+
+def buildnewincludeguardset(path,filename):
+ include_guard_name = buildincludeguardname(path,filename)
+ if(include_guard_name == ''):
+ return []
+ return_value = []
+ return_value.append('#ifndef ' + include_guard_name)
+ return_value.append('#define ' + include_guard_name)
+ return_value.append(include_guard_name)
+ return return_value
+
+def printincludeguardset(include_guard_set):
+ print 'First line: ' + include_guard_set[0]
+ print 'Second line: ' + include_guard_set[1]
+ print 'Last line: ' + include_guard_set[2]
+ return
+
+include_guard_begin_identifier = ['#ifndef', '#if !defined']
+include_guard_second_identifier = ['#define']
+def findincludeguardidentifier(line):
+ for begin_identifier in include_guard_begin_identifier:
+ line = stringmanipulation.removealloccurances(line,begin_identifier)
+ for second_identifier in include_guard_begin_identifier:
+ line = stringmanipulation.removealloccurances(line,second_identifier)
+ removed_prefix = [True,'']
+ line = stringmanipulation.whitespacestoonespace(line)
+ while(removed_prefix[0]):
+ removed_prefix = stringmanipulation.removeprefix(line,' ')
+ line = removed_prefix[1]
+ line = stringmanipulation.removealloccurances(line,'(')
+ if(line == ''):
+ return ''
+ word_pos = stringmanipulation.getword(line,0)
+ return_value = line[0:word_pos[1]]
+ return_value = return_value.rstrip('\r\n')
+ return return_value
+
+def findoldincludeguardset(path,filename):
+ return_value = []
+ full_file_name = path + filename
+ file_pointer = open(full_file_name,'r')
+ include_guard_name = ''
+ for line in file_pointer:
+ if (include_guard_name == ''):
+ for compare_string in include_guard_begin_identifier:
+ if (stringmanipulation.issubstring(compare_string, line) != -1):
+ include_guard_name = findincludeguardidentifier(line)
+ if (include_guard_name == ''):
+ break
+ line = line.rstrip('\r\n')
+ return_value.append(line)
+ break
+ else:
+ for compare_string in include_guard_second_identifier:
+ if (stringmanipulation.issubstring(compare_string, line) != -1):
+ if (stringmanipulation.issubstring(include_guard_name, line) != -1):
+ line = line.rstrip('\r\n')
+ return_value.append(line)
+ return_value.append(include_guard_name)
+ return return_value
+ include_guard_name = ''
+ return_value = []
+ return []
+
+failed_files = []
+for index in range(len(files_to_fix)):
+ if(commit):
+ print (100*index)/len(files_to_fix)
+ path_dir = files_to_fix[index][0]
+ filename = files_to_fix[index][1]
+ is_ignore = False
+ for ignore_names in ignore_these:
+ if(filename == ignore_names):
+ is_ignore = True
+ break
+ if(is_ignore):
+ continue
+ old_include_guard_set = findoldincludeguardset(path_dir,filename)
+ if (len(old_include_guard_set) != 3) :
+ failed_files.append('unable to figure out the include guards for ' + filename)
+ continue
+
+ new_include_guard_set = buildnewincludeguardset(path_dir,filename)
+ if (len(new_include_guard_set) != 3) :
+ failed_files.append('unable to figure out new the include guards for ' + filename)
+ continue
+
+ if(not commit):
+ print 'old guard: ' + old_include_guard_set[2]
+ print 'new guard: ' + new_include_guard_set[2]
+ continue
+
+ for index in range(2):
+ # enough to only replace for file. However, no function for that
+ for extension in extensions:
+ filemanagement.replacestringinfolder(path_dir,old_include_guard_set[index],new_include_guard_set[index],extension)
+ # special case for last to avoid complications
+ for extension in extensions:
+ filemanagement.replacestringinfolder(path_dir,' ' + old_include_guard_set[2],' ' + new_include_guard_set[2],extension)
+ filemanagement.replacestringinfolder(path_dir,'\\/\\/' + old_include_guard_set[2],'\\/\\/ ' + new_include_guard_set[2],extension)
+
+
+if(len(failed_files) > 0):
+ print 'Following failures should be investigated manually:'
+for line in failed_files:
+ print line
diff --git a/trunk/tools/refactoring/fixnames.py b/trunk/tools/refactoring/fixnames.py
new file mode 100644
index 0000000..15381e3
--- /dev/null
+++ b/trunk/tools/refactoring/fixnames.py
@@ -0,0 +1,387 @@
+#!/usr/bin/env python
+
+import stringmanipulation
+import filemanagement
+import p4commands
+import sys
+
+name_space_to_ignore = 'GIPS::'
+#only allow one prefix to be removed since allowing multiple will complicate
+# things
+prefix_to_filter = 'gips'
+#words_to_filter = ['Module']
+# it might be dangerous to remove GIPS but keep it default
+words_to_filter = ['Module','GIPS']
+
+# This script finds all the words that should be replaced in an h-file. Once
+# all words that should be replaced are found it does a global search and
+# replace.
+
+extensions_to_edit = ['.cpp','.cc','.h']
+
+#line = ' ~hiGIPSCriticalSectionScoped()'
+#print line
+#position = stringmanipulation.getword(line,11)
+#old_word = line[position[0]:position[0]+position[1]]
+#result = stringmanipulation.removealloccurances(old_word,'gips')
+#new_word = result
+#print old_word
+#print position[0]
+#print position[0]+position[1]
+#print new_word
+#quit()
+
+# Ignore whole line if any item in this table is a substring of the line
+do_not_replace_line_table = []
+do_not_replace_line_table.append('namespace GIPS')
+
+# [old_string,new_string]
+# List of things to remove that are static:
+manual_replace_table = []
+#manual_replace_table.append(['using namespace GIPS;',''])
+#manual_replace_table.append(['CreateGipsEvent','CreateEvent'])
+#manual_replace_table.append(['CreateGIPSTrace','CreateTrace'])
+#manual_replace_table.append(['ReturnGIPSTrace','ReturnTrace'])
+#manual_replace_table.append(['CreateGIPSFile','CreateFile'])
+replace_table = manual_replace_table
+#replace_table.append(['GIPS::','webrtc::'])
+# List of things to not remove that are static, i.e. exceptions:
+# don't replace any of the GIPS_Words since that will affect all files
+# do that in a separate script!
+do_not_replace_table = []
+do_not_replace_table.append('GIPS_CipherTypes')
+do_not_replace_table.append('GIPS_AuthenticationTypes')
+do_not_replace_table.append('GIPS_SecurityLevels')
+do_not_replace_table.append('GIPS_encryption')
+do_not_replace_table.append('~GIPS_encryption')
+do_not_replace_table.append('GIPS_transport')
+do_not_replace_table.append('~GIPS_transport')
+do_not_replace_table.append('GIPSTraceCallback')
+do_not_replace_table.append('~GIPSTraceCallback')
+do_not_replace_table.append('GIPS_RTP_CSRC_SIZE')
+do_not_replace_table.append('GIPS_RTPDirections')
+do_not_replace_table.append('GIPS_RTP_INCOMING')
+do_not_replace_table.append('GIPS_RTP_OUTGOING')
+do_not_replace_table.append('GIPSFrameType')
+do_not_replace_table.append('GIPS_FRAME_EMPTY')
+do_not_replace_table.append('GIPS_AUDIO_FRAME_SPEECH')
+do_not_replace_table.append('GIPS_AUDIO_FRAME_CN')
+do_not_replace_table.append('GIPS_VIDEO_FRAME_KEY')
+do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA')
+do_not_replace_table.append('GIPS_VIDEO_FRAME_GOLDEN')
+do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA_KEY')
+do_not_replace_table.append('GIPS_PacketType')
+do_not_replace_table.append('GIPS_PACKET_TYPE_RTP')
+do_not_replace_table.append('GIPS_PACKET_TYPE_KEEP_ALIVE')
+do_not_replace_table.append('GIPS_AudioLayers')
+do_not_replace_table.append('GIPS_AUDIO_PLATFORM_DEFAULT')
+do_not_replace_table.append('GIPS_AUDIO_WINDOWS_WAVE')
+do_not_replace_table.append('GIPS_AUDIO_WINDOWS_CORE')
+do_not_replace_table.append('GIPS_AUDIO_LINUX_ALSA')
+do_not_replace_table.append('GIPS_AUDIO_LINUX_PULSE')
+do_not_replace_table.append('GIPS_AUDIO_FORMAT')
+do_not_replace_table.append('GIPS_PCM_16_16KHZ')
+do_not_replace_table.append('GIPS_PCM_16_8KHZ')
+do_not_replace_table.append('GIPS_G729')
+do_not_replace_table.append('GIPSAMRmode')
+do_not_replace_table.append('GIPS_RFC3267_BWEFFICIENT')
+do_not_replace_table.append('GIPS_RFC3267_OCTETALIGNED')
+do_not_replace_table.append('GIPS_RFC3267_FILESTORAGE')
+do_not_replace_table.append('GIPS_NCModes')
+do_not_replace_table.append('GIPS_NC_OFF')
+do_not_replace_table.append('GIPS_NC_MILD')
+do_not_replace_table.append('GIPS_NC_MODERATE')
+do_not_replace_table.append('GIPS_NC_AGGRESSIVE')
+do_not_replace_table.append('GIPS_NC_VERY_AGGRESSIVE')
+do_not_replace_table.append('GIPS_AGCModes')
+do_not_replace_table.append('GIPS_AGC_OFF')
+do_not_replace_table.append('GIPS_AGC_ANALOG')
+do_not_replace_table.append('GIPS_AGC_DIGITAL')
+do_not_replace_table.append('GIPS_AGC_STANDALONE_DIG')
+do_not_replace_table.append('GIPS_ECModes')
+do_not_replace_table.append('GIPS_EC_UNCHANGED')
+do_not_replace_table.append('GIPS_EC_DEFAULT')
+do_not_replace_table.append('GIPS_EC_CONFERENCE')
+do_not_replace_table.append('GIPS_EC_AEC')
+do_not_replace_table.append('GIPS_EC_AES')
+do_not_replace_table.append('GIPS_EC_AECM')
+do_not_replace_table.append('GIPS_EC_NEC_IAD')
+do_not_replace_table.append('GIPS_AESModes')
+do_not_replace_table.append('GIPS_AES_DEFAULT')
+do_not_replace_table.append('GIPS_AES_NORMAL')
+do_not_replace_table.append('GIPS_AES_HIGH')
+do_not_replace_table.append('GIPS_AES_ATTENUATE')
+do_not_replace_table.append('GIPS_AES_NORMAL_SOFT_TRANS')
+do_not_replace_table.append('GIPS_AES_HIGH_SOFT_TRANS')
+do_not_replace_table.append('GIPS_AES_ATTENUATE_SOFT_TRANS')
+do_not_replace_table.append('GIPS_AECMModes')
+do_not_replace_table.append('GIPS_AECM_QUIET_EARPIECE_OR_HEADSET')
+do_not_replace_table.append('GIPS_AECM_EARPIECE')
+do_not_replace_table.append('GIPS_AECM_LOUD_EARPIECE')
+do_not_replace_table.append('GIPS_AECM_SPEAKERPHONE')
+do_not_replace_table.append('GIPS_AECM_LOUD_SPEAKERPHONE')
+do_not_replace_table.append('AECM_LOUD_SPEAKERPHONE')
+do_not_replace_table.append('GIPS_VAD_CONVENTIONAL')
+do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_LOW')
+do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_MID')
+do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_HIGH')
+do_not_replace_table.append('GIPS_NetEQModes')
+do_not_replace_table.append('GIPS_NETEQ_DEFAULT')
+do_not_replace_table.append('GIPS_NETEQ_STREAMING')
+do_not_replace_table.append('GIPS_NETEQ_FAX')
+do_not_replace_table.append('GIPS_NetEQBGNModes')
+do_not_replace_table.append('GIPS_BGN_ON')
+do_not_replace_table.append('GIPS_BGN_FADE')
+do_not_replace_table.append('GIPS_BGN_OFF')
+do_not_replace_table.append('GIPS_OnHoldModes')
+do_not_replace_table.append('GIPS_HOLD_SEND_AND_PLAY')
+do_not_replace_table.append('GIPS_HOLD_SEND_ONLY')
+do_not_replace_table.append('GIPS_HOLD_PLAY_ONLY')
+do_not_replace_table.append('GIPS_PayloadFrequencies')
+do_not_replace_table.append('GIPS_FREQ_8000_HZ')
+do_not_replace_table.append('GIPS_FREQ_16000_HZ')
+do_not_replace_table.append('GIPS_FREQ_32000_HZ')
+do_not_replace_table.append('GIPS_TelephoneEventDetectionMethods')
+do_not_replace_table.append('GIPS_IN_BAND')
+do_not_replace_table.append('GIPS_OUT_OF_BAND')
+do_not_replace_table.append('GIPS_IN_AND_OUT_OF_BAND')
+do_not_replace_table.append('GIPS_ProcessingTypes')
+do_not_replace_table.append('GIPS_PLAYBACK_PER_CHANNEL')
+do_not_replace_table.append('GIPS_PLAYBACK_ALL_CHANNELS_MIXED')
+do_not_replace_table.append('GIPS_RECORDING_PER_CHANNEL')
+do_not_replace_table.append('GIPS_RECORDING_ALL_CHANNELS_MIXED')
+do_not_replace_table.append('GIPS_StereoChannel')
+do_not_replace_table.append('GIPS_StereoLeft')
+do_not_replace_table.append('GIPS_StereoRight')
+do_not_replace_table.append('GIPS_StereoBoth')
+do_not_replace_table.append('GIPS_stat_val')
+do_not_replace_table.append('GIPS_P56_statistics')
+do_not_replace_table.append('GIPS_echo_statistics')
+do_not_replace_table.append('GIPS_NetworkStatistics')
+do_not_replace_table.append('GIPS_JitterStatistics')
+do_not_replace_table.append('GIPSVideoRawType')
+do_not_replace_table.append('GIPS_VIDEO_I420')
+do_not_replace_table.append('GIPS_VIDEO_YV12')
+do_not_replace_table.append('GIPS_VIDEO_YUY2')
+do_not_replace_table.append('GIPS_VIDEO_UYVY')
+do_not_replace_table.append('GIPS_VIDEO_IYUV')
+do_not_replace_table.append('GIPS_VIDEO_ARGB')
+do_not_replace_table.append('GIPS_VIDEO_RGB24')
+do_not_replace_table.append('GIPS_VIDEO_RGB565')
+do_not_replace_table.append('GIPS_VIDEO_ARGB4444')
+do_not_replace_table.append('GIPS_VIDEO_ARGB1555')
+do_not_replace_table.append('GIPS_VIDEO_MJPG')
+do_not_replace_table.append('GIPS_VIDEO_NV12')
+do_not_replace_table.append('GIPS_VIDEO_NV21')
+do_not_replace_table.append('GIPS_VIDEO_Unknown')
+do_not_replace_table.append('GIPSVideoLayouts')
+do_not_replace_table.append('GIPS_LAYOUT_NONE')
+do_not_replace_table.append('GIPS_LAYOUT_DEFAULT')
+do_not_replace_table.append('GIPS_LAYOUT_ADVANCED1')
+do_not_replace_table.append('GIPS_LAYOUT_ADVANCED2')
+do_not_replace_table.append('GIPS_LAYOUT_ADVANCED3')
+do_not_replace_table.append('GIPS_LAYOUT_ADVANCED4')
+do_not_replace_table.append('GIPS_LAYOUT_FULL')
+do_not_replace_table.append('KGIPSConfigParameterSize')
+do_not_replace_table.append('KGIPSPayloadNameSize')
+do_not_replace_table.append('GIPSVideoCodecH263')
+do_not_replace_table.append('GIPSVideoH264Packetization')
+do_not_replace_table.append('GIPS_H264_SingleMode')
+do_not_replace_table.append('GIPS_H264_NonInterleavedMode')
+do_not_replace_table.append('GIPSVideoCodecComplexity')
+do_not_replace_table.append('GIPSVideoCodec_Complexity_Normal')
+do_not_replace_table.append('GIPSVideoCodec_Comlexity_High')
+do_not_replace_table.append('GIPSVideoCodec_Comlexity_Higher')
+do_not_replace_table.append('GIPSVideoCodec_Comlexity_Max')
+do_not_replace_table.append('GIPSVideoCodecH264')
+do_not_replace_table.append('GIPSVideoH264Packetization')
+do_not_replace_table.append('GIPSVideoCodecComplexity')
+do_not_replace_table.append('GIPSVideoCodecProfile')
+do_not_replace_table.append('KGIPSConfigParameterSize')
+do_not_replace_table.append('KGIPSMaxSVCLayers')
+do_not_replace_table.append('GIPSVideoH264LayerTypes')
+do_not_replace_table.append('GIPS_H264SVC_Base')
+do_not_replace_table.append('GIPS_H264SVC_Extend_2X2')
+do_not_replace_table.append('GIPS_H264SVC_Extend_1X1')
+do_not_replace_table.append('GIPS_H264SVC_Extend_MGS')
+do_not_replace_table.append('GIPS_H264SVC_Extend_1_5')
+do_not_replace_table.append('GIPS_H264SVC_Extend_Custom')
+do_not_replace_table.append('GIPSVideoH264LayersProperties')
+do_not_replace_table.append('GIPSVideoH264LayerTypes')
+do_not_replace_table.append('GIPSVideoH264Layers')
+do_not_replace_table.append('GIPSVideoH264LayersProperties')
+do_not_replace_table.append('GIPSVideoCodecH264SVC')
+do_not_replace_table.append('GIPSVideoCodecComplexity')
+do_not_replace_table.append('GIPSVideoCodecProfile')
+do_not_replace_table.append('GIPSVideoH264Layers')
+do_not_replace_table.append('GIPSVideoCodecVP8')
+do_not_replace_table.append('GIPSVideoCodecComplexity')
+do_not_replace_table.append('GIPSVideoCodecMPEG')
+do_not_replace_table.append('GIPSVideoCodecGeneric')
+do_not_replace_table.append('GIPSVideoCodecType')
+do_not_replace_table.append('GIPSVideoCodec_H263')
+do_not_replace_table.append('GIPSVideoCodec_H264')
+do_not_replace_table.append('GIPSVideoCodec_H264SVC')
+do_not_replace_table.append('GIPSVideoCodec_VP8')
+do_not_replace_table.append('GIPSVideoCodec_MPEG4')
+do_not_replace_table.append('GIPSVideoCodec_I420')
+do_not_replace_table.append('GIPSVideoCodec_RED')
+do_not_replace_table.append('GIPSVideoCodec_ULPFEC')
+do_not_replace_table.append('GIPSVideoCodec_Unknown')
+do_not_replace_table.append('GIPSVideoCodecUnion')
+do_not_replace_table.append('GIPSVideoCodecH263')
+do_not_replace_table.append('GIPSVideoCodecH264')
+do_not_replace_table.append('GIPSVideoCodecH264SVC')
+do_not_replace_table.append('GIPSVideoCodecVP8')
+do_not_replace_table.append('GIPSVideoCodecMPEG4')
+do_not_replace_table.append('GIPSVideoCodecGeneric')
+do_not_replace_table.append('GIPSVideoCodec')
+do_not_replace_table.append('GIPSVideoCodecType')
+do_not_replace_table.append('GIPSVideoCodecUnion')
+do_not_replace_table.append('GIPSAudioFrame')
+do_not_replace_table.append('GIPS_CodecInst')
+do_not_replace_table.append('GIPS_FileFormats')
+do_not_replace_table.append('GIPSTickTime')
+do_not_replace_table.append('GIPS_Word64')
+do_not_replace_table.append('GIPS_UWord64')
+do_not_replace_table.append('GIPS_Word32')
+do_not_replace_table.append('GIPS_UWord32')
+do_not_replace_table.append('GIPS_Word16')
+do_not_replace_table.append('GIPS_UWord16')
+do_not_replace_table.append('GIPS_Word8')
+do_not_replace_table.append('GIPS_UWord8')
+
+if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
+ print 'parameters are: parent directory [--commit]'
+ quit()
+
+if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
+ print 'parameters are: parent directory [--commit]'
+ quit()
+
+commit = (len(sys.argv) == 3)
+
+directory = sys.argv[1];
+if(not filemanagement.pathexist(directory)):
+ print 'path ' + directory + ' does not exist'
+ quit()
+
+# APIs are all in h-files
+extension = '.h'
+
+# All h-files
+files_to_modify = filemanagement.listallfilesinfolder(directory,\
+ extension)
+
+def isinmanualremovetable( compare_word ):
+ for old_word, new_word in manual_replace_table:
+ if(old_word == compare_word):
+ return True
+ return False
+
+# Begin
+# This function looks at each line and decides which words should be replaced
+# that is this is the only part of the script that you will ever want to change!
+def findstringstoreplace(line):
+ original_line = line
+# Dont replace compiler directives
+ if(line[0] == '#'):
+ return []
+# Dont allow global removal of namespace gips since it is very intrusive
+ for sub_string_compare in do_not_replace_line_table:
+ index = stringmanipulation.issubstring(line,sub_string_compare)
+ if(index != -1):
+ return []
+
+ return_value = []
+
+ line = stringmanipulation.removeccomment(line)
+ line = stringmanipulation.whitespacestoonespace(line)
+ if(len(line) == 0):
+ return []
+ if(line[0] == '*'):
+ return []
+ index = stringmanipulation.issubstring(line,prefix_to_filter)
+ while index >= 0:
+ dont_store_hit = False
+ word_position = stringmanipulation.getword(line, index)
+ start_of_word = word_position[0]
+ size_of_word = word_position[1]
+ end_of_word = start_of_word + size_of_word
+ old_word = line[start_of_word:end_of_word]
+ if(isinmanualremovetable(old_word)):
+ dont_store_hit = True
+ if((end_of_word + 2 < len(line)) and\
+ name_space_to_ignore == line[start_of_word:end_of_word+2]):
+ dont_store_hit = True
+
+ result = stringmanipulation.removeprefix(old_word,prefix_to_filter)
+ new_word = result[1]
+ for word_to_filter in words_to_filter:
+ new_word = stringmanipulation.removealloccurances(new_word,word_to_filter)
+ result = stringmanipulation.removeprefix(new_word,'_')
+ new_word = result[1]
+ new_word = stringmanipulation.fixabbreviations(new_word)
+ new_word = stringmanipulation.removealloccurances(new_word,'_')
+ if(not dont_store_hit):
+ return_value.append([old_word,new_word])
+# remove the word we found from the string so we dont find it again
+ line = line[0:start_of_word] + line[end_of_word:len(line)]
+ index = stringmanipulation.issubstring(line,'GIPS')
+
+ return return_value
+# End
+
+# loop through all files
+for path, file_name in files_to_modify:
+# if(file_name != 'GIPSTickUtil.h'):
+# continue
+ full_file_name = path + file_name
+ file_pointer = open(full_file_name,'r')
+# print file_name
+#loop through all lines
+ for line in file_pointer:
+# print line
+ local_replace_string = findstringstoreplace(line)
+ #print local_replace_string
+ if(len(local_replace_string) != 0):
+ replace_table.extend(local_replace_string)
+
+
+# we have built our replace table now
+replace_table = stringmanipulation.removeduplicates( replace_table )
+replace_table = stringmanipulation.ordertablesizefirst( replace_table )
+replace_table = stringmanipulation.complement(replace_table,\
+ do_not_replace_table)
+
+def replaceoriginal( path,my_table ):
+ size_of_table = len(my_table)
+ for index in range(len(my_table)):
+ old_name = my_table[index][0]
+ new_name = my_table[index][1]
+ filemanagement.replacestringinfolder(path, old_name, new_name,\
+ ".h")
+ print (100*index) / (size_of_table*2)
+
+def replaceall( my_table, extension_list ):
+ size_of_table = len(my_table)
+ for index in range(len(my_table)):
+ old_name = my_table[index][0]
+ new_name = my_table[index][1]
+ new_name = new_name
+ for extension in extensions_to_edit:
+ filemanagement.replacestringinallsubfolders(old_name, new_name,
+ extension)
+ print 100*(size_of_table + index) / (size_of_table*2)
+
+
+if(commit):
+ print 'commiting'
+ replace_table = stringmanipulation.removenochange(replace_table)
+ p4commands.checkoutallfiles()
+ replaceoriginal(directory,replace_table)
+ replaceall(replace_table,extensions_to_edit)
+ p4commands.revertunchangedfiles()
+else:
+ for old_name, new_name in replace_table:
+ print 'Going to replace [' + old_name + '] with [' + new_name + ']'
diff --git a/trunk/tools/refactoring/integratefiles.py b/trunk/tools/refactoring/integratefiles.py
new file mode 100644
index 0000000..c5cc892
--- /dev/null
+++ b/trunk/tools/refactoring/integratefiles.py
@@ -0,0 +1,100 @@
+#!/usr/bin/env python
+
+import stringmanipulation
+import filemanagement
+import p4commands
+import sys
+
+extensions = ['.h', '.cpp', '.cc', '.gyp']
+
+ignore_these = ['list_no_stl.h','map_no_stl.h','constructor_magic.h']
+
+exceptions = [
+['GIPSRWLock.h','rw_lock.h'],
+['GIPSCriticalsection.h','critical_section.h'],
+]
+
+if((len(sys.argv) != 4) and (len(sys.argv) != 5)):
+ print 'parameters are: parent directory extension new extension [--commit]'
+ quit()
+
+directory = sys.argv[1];
+if(not filemanagement.pathexist(directory)):
+ print 'path ' + directory + ' does not exist'
+ quit()
+
+old_extension = sys.argv[2]
+if(not stringmanipulation.isextension(old_extension)):
+ print old_extension + ' is not a valid extension'
+ quit()
+
+new_extension = sys.argv[3]
+if(not stringmanipulation.isextension(new_extension)):
+ print new_extension + ' is not a valid extension'
+ quit()
+
+if((len(sys.argv) == 5) and (sys.argv[4] != '--commit')):
+ print 'parameters are: parent directory extension new extension [--commit]'
+ quit()
+
+commit = False
+if(len(sys.argv) == 5):
+ commit = True
+
+files_to_integrate = filemanagement.listallfilesinfolder(directory,\
+ old_extension)
+
+if(commit):
+ p4commands.checkoutallfiles()
+for index in range(len(files_to_integrate)):
+ if(commit):
+ print (100*index)/len(files_to_integrate)
+ path_dir = files_to_integrate[index][0]
+ filename = files_to_integrate[index][1]
+ is_ignore = False
+ for ignore_names in ignore_these:
+ if(filename == ignore_names):
+ is_ignore = True
+ break
+ if(is_ignore):
+ continue
+
+ new_file_name = ''
+ is_exception = False
+ for exception_name,exception_name_new in exceptions:
+ if(filename == exception_name):
+ is_exception = True
+ new_file_name = exception_name_new
+ break
+
+ if(not is_exception):
+ new_file_name = filename
+
+ new_file_name = stringmanipulation.removeallprefix(new_file_name,\
+ 'gips')
+ new_file_name = stringmanipulation.removealloccurances(new_file_name,\
+ 'module')
+ new_file_name = stringmanipulation.changeextension(new_file_name,\
+ old_extension,\
+ new_extension)
+ new_file_name = stringmanipulation.fixabbreviations( new_file_name )
+ new_file_name = stringmanipulation.lowercasewithunderscore(new_file_name)
+ if(not commit):
+ print 'File ' + filename + ' will be replaced with ' + new_file_name
+ continue
+ full_new_file_name = path_dir + new_file_name
+ full_old_file_name = path_dir + filename
+ if(full_new_file_name != full_old_file_name):
+ p4commands.integratefile(full_old_file_name,full_new_file_name)
+ else:
+ print 'skipping ' + new_file_name + ' due to no change'
+ for extension in extensions:
+ print 'replacing ' + filename
+ if (extension == ".gyp"):
+ filemanagement.replacestringinallsubfolders(
+ filename,new_file_name,extension)
+ else:
+ filemanagement.replacestringinallsubfolders(
+ '\"' + filename + '\"', '\"' + new_file_name + '\"', extension)
+if(commit):
+ p4commands.revertunchangedfiles()
diff --git a/trunk/tools/refactoring/p4commands.py b/trunk/tools/refactoring/p4commands.py
new file mode 100644
index 0000000..71ac31b
--- /dev/null
+++ b/trunk/tools/refactoring/p4commands.py
@@ -0,0 +1,31 @@
+import os
+import filemanagement
+
+# checks out entire p4 repository
+def checkoutallfiles():
+ os.system('p4 edit //depotGoogle/...')
+ return
+
+# reverts all unchanged files, this is completely innoculus
+def revertunchangedfiles():
+ os.system('p4 revert -a //depotGoogle/...')
+ return
+
+def integratefile( old_name, new_name):
+ if(old_name == new_name):
+ return
+ if(not filemanagement.fileexist(old_name)):
+ return
+ integrate_command = 'p4 integrate -o -f ' +\
+ old_name +\
+ ' ' +\
+ new_name +\
+ ' > p4summary.txt 2> error.txt'
+ os.system(integrate_command)
+ #print integrate_command
+ delete_command = 'p4 delete -c default ' +\
+ old_name +\
+ ' > p4summary.txt 2> error.txt'
+ os.system(delete_command)
+ #print delete_command
+ return
diff --git a/trunk/tools/refactoring/removetrace.py b/trunk/tools/refactoring/removetrace.py
new file mode 100644
index 0000000..43c622d
--- /dev/null
+++ b/trunk/tools/refactoring/removetrace.py
@@ -0,0 +1,161 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# NOTE: This is a hack which disobeys a number of conventions and best
+# practices. It's here just to be easily shared. If it's to remain in the
+# repository it should be refactored.
+
+#!/usr/bin/env python
+
+import stringmanipulation
+import filemanagement
+import sys
+
+trace_remove_key_word = 'kTraceModuleCall'
+
+if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
+ print 'parameters are: parent directory [--commit]'
+ quit()
+
+if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
+ print 'parameters are: parent directory [--commit]'
+ quit()
+
+commit = (len(sys.argv) == 3)
+
+directory = sys.argv[1];
+occurances = []
+
+trace_identifier = 'WEBRTC_TRACE('
+extensions = ['.h','.cc','.c','.cpp']
+files_to_fix = []
+for extension in extensions:
+ files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
+ extension))
+
+# This function identifies the begining of a trace statement
+def istracebegining(line):
+ return stringmanipulation.issubstring(line, trace_identifier) != -1
+
+def endofstatement(line):
+ return stringmanipulation.issubstring(line, ';') != -1
+
+def removekeywordfound(line):
+ return stringmanipulation.issubstring(line, trace_remove_key_word) != -1
+
+# Used to store temporary result before flushing to real file when finished
+def temporaryfilename():
+ return 'deleteme.txt'
+
+
+def find_occurances(path, file_name):
+ full_filename = path + file_name
+ file_handle = open(full_filename,'r')
+ line_is_trace = False
+ last_trace_line = -1
+ for line_nr, line in enumerate(file_handle):
+ if(istracebegining(line)):
+ line_is_trace = True;
+ last_trace_line = line_nr
+
+ if(line_is_trace):
+ if(removekeywordfound(line)):
+ occurances.append(last_trace_line)
+
+ if(endofstatement(line)):
+ line_is_trace = False;
+
+def remove_occurances(path, file_name):
+ full_file_name = path + file_name
+ if (not filemanagement.fileexist(full_file_name)):
+ print 'File ' + full_file_name + ' is not found.'
+ print 'Should not happen! Ever!'
+ quit()
+
+ full_temporary_file_name = path + temporaryfilename()
+ temporary_file = open(full_temporary_file_name,'w')
+ original_file = open(full_file_name,'r')
+ next_occurance_id = 0;
+ removing_statement = False
+ if(len(occurances) == next_occurance_id):
+ return
+ next_occurance = occurances[next_occurance_id]
+ next_occurance_id += 1
+ for line_nr, line in enumerate(original_file):
+ if(line_nr == next_occurance):
+ removing_statement = True
+ if(len(occurances) == next_occurance_id):
+ next_occurance_id = -1
+ else:
+ next_occurance = occurances[next_occurance_id]
+ next_occurance_id += 1
+
+ if (not removing_statement):
+ temporary_file.writelines(line)
+
+ if(endofstatement(line)):
+ removing_statement = False;
+
+ temporary_file.close()
+ original_file.close()
+ filemanagement.copyfile(full_file_name,full_temporary_file_name)
+ filemanagement.deletefile(full_temporary_file_name)
+
+def nextoccurance():
+ if (len(occurances) == 0):
+ return -1
+ return_value = occurances[0]
+ occurances = occurances[1:len(occurances)]
+ return return_value
+
+def would_be_removed_occurances(path, file_name):
+ full_file_name = path + file_name
+ if (not filemanagement.fileexist(full_file_name)):
+ print 'File ' + full_file_name + ' is not found.'
+ print 'Should not happen! Ever!'
+ quit()
+
+ original_file = open(full_file_name,'r')
+ removing_statement = False
+ next_occurance_id = 0;
+ if(len(occurances) == next_occurance_id):
+ return
+ next_occurance = occurances[next_occurance_id]
+ next_occurance_id += 1
+ for line_nr, line in enumerate(original_file):
+ if(line_nr == next_occurance):
+ removing_statement = True
+ if(len(occurances) == next_occurance_id):
+ return
+ next_occurance = occurances[next_occurance_id]
+ next_occurance_id += 1
+
+ if (removing_statement):
+ print line_nr
+
+ if(endofstatement(line)):
+ removing_statement = False;
+ if(next_occurance == -1):
+ break
+ original_file.close()
+
+for index in range(len(files_to_fix)):
+ if(commit):
+ print (100*index)/len(files_to_fix)
+
+ path_dir = files_to_fix[index][0]
+ filename = files_to_fix[index][1]
+
+ #print path_dir + filename
+ occurances = []
+ find_occurances(path_dir, filename)
+
+ if(not commit):
+ would_be_removed_occurances(path_dir, filename)
+ continue
+ remove_occurances(path_dir, filename)
diff --git a/trunk/tools/refactoring/stringmanipulation.py b/trunk/tools/refactoring/stringmanipulation.py
new file mode 100644
index 0000000..0d9e0ff
--- /dev/null
+++ b/trunk/tools/refactoring/stringmanipulation.py
@@ -0,0 +1,303 @@
+import string
+
+# returns tuple, [success,updated_string] where the updated string has
+# has one less (the first) occurance of match string
+def removefirstoccurance( remove_string, match_string ):
+ lowercase_string = remove_string.lower()
+ lowercase_match_string = match_string.lower()
+ lowest_index = lowercase_string.find(lowercase_match_string)
+ if(lowest_index == -1):
+ return [False,remove_string]
+ past_match_index = lowest_index + len(lowercase_match_string)
+ highest_index = len(remove_string)
+ remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
+ return [True,remove_string]
+
+# returns a string with all occurances of match_string removed
+def removealloccurances( remove_string, match_string ):
+ return_value = [True, remove_string]
+ while(return_value[0]):
+ return_value = removefirstoccurance(return_value[1],match_string)
+ return return_value[1]
+
+# removes an occurance of match_string only if it's first in the string
+# returns tuple [succes, new_string]
+def removeprefix( remove_string, match_string ):
+ lowercase_string = remove_string.lower()
+ lowercase_match_string = match_string.lower()
+ lowest_index = lowercase_string.find(lowercase_match_string)
+ if(lowest_index == -1):
+ return [False,remove_string]
+ if(lowest_index != 0):
+ return [False,remove_string]
+ past_match_index = lowest_index + len(lowercase_match_string)
+ highest_index = len(remove_string)
+ remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
+# print lowest_index
+# print past_match_index
+ return [True,remove_string]
+
+# removes multiple occurances of match string as long as they are first in
+# the string
+def removeallprefix( remove_string, match_string ):
+ return_value = [True, remove_string]
+ while(return_value[0]):
+ return_value = removeprefix(return_value[1],match_string)
+ return return_value[1]
+
+# returns true if extensionstring is a correct extension
+def isextension( extensionstring ):
+ if(len(extensionstring) < 2):
+ return False
+ if(extensionstring[0] != '.'):
+ return False
+ if(extensionstring[1:len(extensionstring)-1].find('.') != -1):
+ return False
+ return True
+
+# returns the index of start of the last occurance of match_string
+def findlastoccurance( original_string, match_string ):
+ search_index = original_string.find(match_string)
+ found_index = search_index
+ last_index = len(original_string) - 1
+ while((search_index != -1) and (search_index < last_index)):
+ search_index = original_string[search_index+1:last_index].find(match_string)
+ if(search_index != -1):
+ found_index = search_index
+ return found_index
+
+# changes extension from original_extension to new_extension
+def changeextension( original_string, original_extension, new_extension):
+ if(not isextension(original_extension)):
+ return original_string
+ if(not isextension(new_extension)):
+ return original_string
+ index = findlastoccurance(original_string, original_extension)
+ if(index == -1):
+ return original_string
+ return_value = original_string[0:index] + new_extension
+ return return_value
+
+# wanted to do this with str.find however didnt seem to work so do it manually
+# returns the index of the first capital letter
+def findfirstcapitalletter( original_string ):
+ for index in range(len(original_string)):
+ if(original_string[index].lower() != original_string[index]):
+ return index
+ return -1
+
+
+# replaces capital letters with underscore and lower case letter (except very
+# first
+def lowercasewithunderscore( original_string ):
+# ignore the first letter since there should be no underscore in front of it
+ if(len(original_string) < 2):
+ return original_string
+ return_value = original_string[1:len(original_string)]
+ index = findfirstcapitalletter(return_value)
+ while(index != -1):
+ return_value = return_value[0:index] + \
+ '_' + \
+ return_value[index].lower() + \
+ return_value[index+1:len(return_value)]
+ index = findfirstcapitalletter(return_value)
+ return_value = original_string[0].lower() + return_value
+ return return_value
+
+# my table is a duplicate of strings
+def removeduplicates( my_table ):
+ new_table = []
+ for old_string1, new_string1 in my_table:
+ found = 0
+ for old_string2, new_string2 in new_table:
+ if(old_string1 == old_string2):
+ found += 1
+ if(new_string1 == new_string2):
+ if(new_string1 == ''):
+ found += found
+ else:
+ found += 1
+ if(found == 1):
+ print 'missmatching set, terminating program'
+ print old_string1
+ print new_string1
+ print old_string2
+ print new_string2
+ quit()
+ if(found == 2):
+ break
+ if(found == 0):
+ new_table.append([old_string1,new_string1])
+ return new_table
+
+def removenochange( my_table ):
+ new_table = []
+ for old_string, new_string in my_table:
+ if(old_string != new_string):
+ new_table.append([old_string,new_string])
+ return new_table
+
+# order table after size of the string (can be used to replace bigger strings
+# first which is useful since smaller strings can be inside the bigger string)
+# E.g. GIPS is a sub string of GIPSVE if we remove GIPS first GIPSVE will never
+# be removed. N is small so no need for fancy sort algorithm. Use selection sort
+def ordertablesizefirst( my_table ):
+ for current_index in range(len(my_table)):
+ biggest_string = 0
+ biggest_string_index = -1
+ for search_index in range(len(my_table)):
+ if(search_index < current_index):
+ continue
+ length_of_string = len(my_table[search_index][0])
+ if(length_of_string > biggest_string):
+ biggest_string = length_of_string
+ biggest_string_index = search_index
+ if(biggest_string_index == -1):
+ print 'sorting algorithm failed, program exit'
+ quit()
+ old_value = my_table[current_index]
+ my_table[current_index] = my_table[biggest_string_index]
+ my_table[biggest_string_index] = old_value
+ return my_table
+
+# returns true if string 1 or 2 is a substring of the other, assuming neither
+# has whitespaces
+def issubstring( string1, string2 ):
+ if(len(string1) == 0):
+ return -1
+ if(len(string2) == 0):
+ return -1
+ large_string = string1
+ small_string = string2
+ if(len(string1) < len(string2)):
+ large_string = string2
+ small_string = string1
+
+ for index in range(len(large_string)):
+ large_sub_string = large_string[index:index+len(small_string)].lower()
+ if(large_sub_string ==\
+ small_string.lower()):
+ return index
+ return -1
+
+#not_part_of_word_table = [' ','(',')','{','}',':','\t','*','&','/','[',']','.',',','\n']
+#def ispartofword( char ):
+# for item in not_part_of_word_table:
+# if(char == item):
+# return False
+# return True
+
+# must be numerical,_ or charachter
+def ispartofword( char ):
+ if(char.isalpha()):
+ return True
+ if(char.isalnum()):
+ return True
+ if(char == '_'):
+ return True
+ return False
+
+# returns the index of the first letter in the word that the current_index
+# is pointing to and the size of the word
+def getword( line, current_index):
+ if(current_index < 0):
+ return []
+ line = line.rstrip()
+ if(len(line) <= current_index):
+ return []
+ if(line[current_index] == ' '):
+ return []
+ start_pos = current_index
+ while start_pos >= 0:
+ if(not ispartofword(line[start_pos])):
+ start_pos += 1
+ break
+ start_pos -= 1
+ if(start_pos == -1):
+ start_pos = 0
+ end_pos = current_index
+ while end_pos < len(line):
+ if(not ispartofword(line[end_pos])):
+ break
+ end_pos += 1
+ return [start_pos,end_pos - start_pos]
+
+# my table is a tuple [string1,string2] complement_to_table is just a list
+# of strings to compare to string1
+def complement( my_table, complement_to_table ):
+ new_table = []
+ for index in range(len(my_table)):
+ found = False;
+ for compare_string in complement_to_table:
+ if(my_table[index][0].lower() == compare_string.lower()):
+ found = True
+ if(not found):
+ new_table.append(my_table[index])
+ return new_table
+
+def removestringfromhead( line, remove_string):
+ for index in range(len(line)):
+ if(line[index:index+len(remove_string)] != remove_string):
+ return line[index:index+len(line)]
+ return ''
+
+def removeccomment( line ):
+ comment_string = '//'
+ for index in range(len(line)):
+ if(line[index:index+len(comment_string)] == comment_string):
+ return line[0:index]
+ return line
+
+def whitespacestoonespace( line ):
+ return ' '.join(line.split())
+
+def fixabbreviations( original_string ):
+ previouswascapital = (original_string[0].upper() == original_string[0])
+ new_string = ''
+ for index in range(len(original_string)):
+ if(index == 0):
+ new_string += original_string[index]
+ continue
+ if(original_string[index] == '_'):
+ new_string += original_string[index]
+ previouswascapital = False
+ continue
+ if(original_string[index].isdigit()):
+ new_string += original_string[index]
+ previouswascapital = False
+ continue
+ currentiscapital = (original_string[index].upper() == original_string[index])
+ letter_to_add = original_string[index]
+ if(previouswascapital and currentiscapital):
+ letter_to_add = letter_to_add.lower()
+ if(previouswascapital and (not currentiscapital)):
+ old_letter = new_string[len(new_string)-1]
+ new_string = new_string[0:len(new_string)-1]
+ new_string += old_letter.upper()
+ previouswascapital = currentiscapital
+ new_string += letter_to_add
+ return new_string
+
+def replaceoccurances(old_string, replace_string, replace_with_string):
+ if (len(replace_string) == 0):
+ return old_string
+ if (len(old_string) < len(replace_string)):
+ return old_string
+ # Simple implementation, could proably be done smarter
+ new_string = ''
+ for index in range(len(old_string)):
+ #print new_string
+ if(len(replace_string) > (len(old_string) - index)):
+ new_string += old_string[index:index + len(old_string)]
+ break
+ match = (len(replace_string) > 0)
+ for replace_index in range(len(replace_string)):
+ if (replace_string[replace_index] != old_string[index + replace_index]):
+ match = False
+ break
+ if (match):
+ new_string += replace_with_string
+ index =+ len(replace_string)
+ else:
+ new_string += old_string[index]
+ return new_string
diff --git a/trunk/tools/refactoring/trim.py b/trunk/tools/refactoring/trim.py
new file mode 100644
index 0000000..5539f5f
--- /dev/null
+++ b/trunk/tools/refactoring/trim.py
@@ -0,0 +1,29 @@
+#!/usr/bin/env python
+
+import sys
+import fileinput
+
+# Defaults
+TABSIZE = 4
+
+usage = """
+Replaces all TAB characters with %(TABSIZE)d space characters.
+In addition, all trailing space characters are removed.
+usage: trim file ...
+file ... : files are changed in place without taking any backup.
+""" % vars()
+
+def main():
+
+ if len(sys.argv) == 1:
+ sys.stderr.write(usage)
+ sys.exit(2)
+
+ # Iterate over the lines of all files listed in sys.argv[1:]
+ for line in fileinput.input(sys.argv[1:], inplace=True):
+ line = line.replace('\t',' '*TABSIZE); # replace TABs
+ line = line.rstrip(None) # remove trailing whitespaces
+ print line # modify the file
+
+if __name__ == '__main__':
+ main()
diff --git a/trunk/tools/refactoring/trimall.py b/trunk/tools/refactoring/trimall.py
new file mode 100644
index 0000000..7a1c458
--- /dev/null
+++ b/trunk/tools/refactoring/trimall.py
@@ -0,0 +1,59 @@
+#!/usr/bin/env python
+
+import sys
+import fileinput
+import filemanagement
+import p4commands
+
+# Defaults
+TABSIZE = 4
+
+extensions = ['.h','.cc','.c','.cpp']
+
+ignore_these = ['my_ignore_header.h']
+
+usage = """
+Replaces all TAB characters with %(TABSIZE)d space characters.
+In addition, all trailing space characters are removed.
+usage: trim directory
+""" % vars()
+
+if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
+ sys.stderr.write(usage)
+ sys.exit(2)
+
+directory = sys.argv[1];
+if(not filemanagement.pathexist(directory)):
+ sys.stderr.write(usage)
+ sys.exit(2)
+
+if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
+ sys.stderr.write(usage)
+ sys.exit(2)
+
+commit = False
+if(len(sys.argv) == 3):
+ commit = True
+
+files_to_fix = []
+for extension in extensions:
+ files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
+ extension))
+
+def main():
+ if (commit):
+ p4commands.checkoutallfiles()
+ for path,file_name in files_to_fix:
+ full_file_name = path + file_name
+ if (not commit):
+ print full_file_name + ' will be edited'
+ continue
+ for line in fileinput.input(full_file_name, inplace=True):
+ line = line.replace('\t',' '*TABSIZE); # replace TABs
+ line = line.rstrip(None) # remove trailing whitespaces
+ print line # modify the file
+ if (commit):
+ p4commands.revertunchangedfiles()
+
+if __name__ == '__main__':
+ main()
diff --git a/trunk/tools/resources/update.py b/trunk/tools/resources/update.py
new file mode 100755
index 0000000..a07e2c1
--- /dev/null
+++ b/trunk/tools/resources/update.py
@@ -0,0 +1,131 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
+
+"""Downloads WebRTC resources files from a remote host."""
+
+from optparse import OptionParser
+from urlparse import urljoin
+import os
+import shutil
+import sys
+import tarfile
+import tempfile
+import urllib2
+
+
+def main():
+ """
+ Downloads WebRTC resources files from a remote host.
+
+ This script will download WebRTC resource files used for testing, like audio
+ and video files. It will check the current version in the DEPS file and
+ compare it with the one downloaded (kept in a text file in the download dir).
+ If the DEPS version is different than the one downloaded, the correct version
+ will be downloaded.
+ """
+ # Constants.
+ deps_key = 'webrtc_resources_revision'
+ remote_url_base = 'http://commondatastorage.googleapis.com/webrtc-resources/'
+ version_filename = 'webrtc-resources-version'
+ filename_prefix = 'webrtc-resources-'
+ extension = '.tgz'
+
+ # Variables used by the script.
+ project_root_dir = os.path.normpath(sys.path[0] + '/../../')
+ deps_file = os.path.join(project_root_dir, 'DEPS')
+ downloads_dir = os.path.join(project_root_dir, 'resources')
+ current_version_file = os.path.join(downloads_dir, version_filename)
+
+ # Ensure the downloads dir is created.
+ if not os.path.isdir(downloads_dir):
+ os.mkdir(downloads_dir)
+
+ # Define and parse arguments.
+ parser = OptionParser()
+ parser.add_option('-f', '--force', action='store_true', dest='force',
+ help='forces download and removes all existing resources.')
+ (options, unused_args) = parser.parse_args()
+
+ # Check if we have an existing version already downloaded.
+ current_version = 0
+ if os.path.isfile(current_version_file):
+ f = open(current_version_file)
+ current_version = int(f.read())
+ f.close()
+ print 'Found downloaded resources: version: %s' % current_version
+
+ # Check the DEPS file for the latest version number.
+ deps_vars = EvalDepsFile(deps_file)['vars']
+ latest_version = int(deps_vars[deps_key])
+ print 'Version in DEPS file: %d' % latest_version
+
+ # Download archive if forced or DEPS version is different than our current.
+ if latest_version != current_version or options.force:
+ temp_dir = tempfile.mkdtemp(prefix='webrtc-resources-')
+ archive_name = '%s%s%s' % (filename_prefix, latest_version, extension)
+ remote_archive_url = urljoin(remote_url_base, archive_name)
+ # Download into the temporary directory with display of progress, inspired
+ # by the Stack Overflow post at http://goo.gl/JIrbo
+ temp_file = os.path.join(temp_dir, archive_name)
+ print 'Downloading: %s' % remote_archive_url
+ u = urllib2.urlopen(remote_archive_url)
+ f = open(temp_file, 'wb')
+ meta = u.info()
+ file_size = int(meta.getheaders('Content-Length')[0])
+ print 'Progress: %s bytes: %s' % (archive_name, file_size)
+
+ file_size_dl = 0
+ block_size = 65536
+ while True:
+ file_buffer = u.read(block_size)
+ if not file_buffer:
+ break
+ file_size_dl += len(file_buffer)
+ f.write(file_buffer)
+ status = r'%10d [%3.2f%%]' % (file_size_dl,
+ file_size_dl * 100. / file_size)
+ status += chr(8) * (len(status) + 1)
+ print status,
+ print
+ f.close()
+
+ # Clean up the existing resources dir.
+ print 'Removing old resources in %s' % downloads_dir
+ shutil.rmtree(downloads_dir)
+ os.mkdir(downloads_dir)
+
+ # Write the downloaded version to a text file in the resources dir to avoid
+ # re-download of the same version in the future.
+ new_version_file = os.path.join(downloads_dir, version_filename)
+ f = open(new_version_file, 'w')
+ f.write('%d' % latest_version)
+ f.close()
+
+ # Extract the archive.
+ archive = tarfile.open(temp_file, 'r:gz')
+ archive.extractall(downloads_dir)
+ archive.close()
+ print 'Extracted resource files into %s' % downloads_dir
+ # Clean up the temp dir.
+ shutil.rmtree(temp_dir)
+ else:
+ print 'Already have correct version: %s' % current_version
+
+
+def EvalDepsFile(path):
+ scope = {'Var': lambda name: scope['vars'][name],
+ 'File': lambda name: name,
+ 'From': lambda deps, definition: deps}
+ execfile(path, {}, scope)
+ return scope
+
+if __name__ == '__main__':
+ main()
diff --git a/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt b/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt
new file mode 100644
index 0000000..1123017
--- /dev/null
+++ b/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt
@@ -0,0 +1,30 @@
+# There are four kinds of suppressions in this file.
+# 1. third party stuff we have no control over
+#
+# 2. intentional unit test errors, or stuff that is somehow a false positive
+# in our own code, or stuff that is so trivial it's not worth fixing
+#
+# 3. Suppressions for real WebRTC bugs that are not yet fixed.
+# These should all be in WebRTC's bug tracking system.
+# Periodically we should sweep this file and the bug tracker clean by
+# running overnight and removing outdated bugs/suppressions.
+#
+# 4. issues that happen only on Google workstations.
+#-----------------------------------------------------------------------
+
+
+
+# 1. third party stuff we have no control over
+
+#-----------------------------------------------------------------------
+# 2. intentional unit test errors, or stuff that is somehow a false positive
+# in our own code, or stuff that is so trivial it's not worth fixing
+
+#-----------------------------------------------------------------------
+# 3. Suppressions for real chromium bugs that are not yet fixed.
+# These should all be in WebRTC's bug tracking system.
+
+
+#-----------------------------------------------------------------------
+# 4. These only occur on our Google workstations
+
diff --git a/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt b/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt
new file mode 100644
index 0000000..a387a64
--- /dev/null
+++ b/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt
@@ -0,0 +1,323 @@
+# There are three kinds of suppressions in this file:
+# 1. Third party stuff we have no control over.
+#
+# 2. Intentional unit test errors, stuff that is somehow a false positive
+# in our own code, or stuff that is so trivial it's not worth fixing.
+#
+# 3. Suppressions for real WebRTC bugs that are not yet fixed.
+# These should all be in WebRTC's bug tracking system.
+# Periodically we should sweep this file and the bug tracker clean by
+# running overnight and removing outdated bugs/suppressions.
+#-----------------------------------------------------------------------
+
+# 1. Third party stuff we have no control over.
+{
+ FIXME mac kevent libevent probably needs valgrind hooks
+ Memcheck:Param
+ kevent(changelist)
+ fun:kevent
+ fun:event_base_new
+}
+{
+ # CoreAudio leak. See http://crbug.com/9351
+ bug_9351
+ Memcheck:Leak
+ ...
+ fun:_ZN12HALCADClient19AddPropertyListenerEmPK26AudioObjectPropertyAddressPFlmmS2_PvES3_
+ fun:_ZN16HALDefaultDevice22InstallServerListenersEv
+ fun:_ZN16HALDefaultDevice10InitializeEv
+ fun:_ZN9HALSystem16CheckOutInstanceEv
+}
+{
+ # Mac test_shell_tests. See http://crbug.com/11134
+ # Doesn't happen on bots, but happens like crazy on the smo
+ # test machine 'caliban'. Don't delete just because it
+ # doesn't happen on the bots.
+ bug_11134
+ Memcheck:Uninitialized
+ fun:vCMMVectorConvert8BitRGBToRGB
+ fun:_ZNK15CMMConvRGBToRGB7ConvertER8CMM8BitsP14CMMRuntimeInfomm
+}
+{
+ # Mac system library bug? See http://crbug.com/11327
+ bug_11327
+ Memcheck:Uninitialized
+ fun:_ZN19AudioConverterChain5ResetEv
+ fun:AudioConverterReset
+ obj:/System/Library/Components/CoreAudio.component/Contents/MacOS/CoreAudio
+}
+{
+ # Mac system library bug? See http://crbug.com/11327
+ bug_11327b
+ Memcheck:Uninitialized
+ fun:AUNetSendEntry
+ fun:AUNetSendEntry
+ obj:/System/Library/Components/CoreAudio.component/Contents/MacOS/CoreAudio
+}
+{
+ # Filed with Apple as rdar://6915060; see http://crbug.com/11270
+ bug_11270
+ Memcheck:Leak
+ fun:calloc
+ fun:CMSSetLabCLUT
+}
+{
+ # Mac leak in CMOpenOrNewAccess in unit_tests PlatformCanvas_SkLayer_Test,
+ # ToolbarControllerTest_FocusLocation_Test. See http://crbug.com/11333.
+ bug_11333
+ Memcheck:Leak
+ fun:malloc
+ fun:stdSmartNewPtr
+ fun:stdSmartNewHandle
+ fun:IOCreateAndOpen
+ fun:ScratchInit
+ fun:CMOpenOrNewAccess
+}
+{
+ # suddenly very common as of 6 aug 2009
+ bug_11333b
+ Memcheck:Leak
+ fun:malloc
+ fun:stdSmartNewPtr
+ fun:stdSmartNewHandle
+ fun:IOCreateAndOpen
+ fun:ScratchInit
+ fun:CMNewAccessFromAnother
+}
+{
+ # Tiny one-time leak, widely seen by valgind users; everyone suppresses this.
+ # See related discussion at http://gcc.gnu.org/bugzilla/show_bug.cgi?id=39366
+ plugin_bundle_global_leak
+ Memcheck:Leak
+ fun:malloc
+ fun:__cxa_get_globals
+ fun:__cxa_allocate_exception
+ fun:_ZN4dyld4loadEPKcRKNS_11LoadContextE
+ fun:dlopen
+ fun:dlopen
+ fun:_CFBundleDlfcnCheckLoaded
+}
+{
+ bug_18215
+ Memcheck:Uninitialized
+ fun:_DPSNextEvent
+ fun:-[NSApplication nextEventMatchingMask:untilDate:inMode:dequeue:]
+ fun:-[NSApplication run]
+}
+{
+ bug_18223
+ Memcheck:Uninitialized
+ fun:_ZNK8Security12UnixPlusPlus17StaticForkMonitorclEv
+ fun:_ZN12ocspdGlobals10serverPortEv
+}
+{
+ # Filed with Apple as rdar://7255382
+ bug_20459a
+ Memcheck:Leak
+ ...
+ fun:_CFRuntimeCreateInstance
+ fun:CFRunLoopSourceCreate
+ fun:CFMachPortCreateRunLoopSource
+ fun:_ZN8Security12MachPlusPlus10CFAutoPort6enableEv
+ fun:_ZN8Security14SecurityServer14ThreadNotifierC2Ev
+}
+{
+ # Also filed with Apple as rdar://7255382
+ bug_20459b
+ Memcheck:Leak
+ fun:malloc_zone_malloc
+ fun:_CFRuntimeCreateInstance
+ fun:__CFArrayInit
+ fun:CFArrayCreateMutableCopy
+ fun:_ZN8Security12KeychainCore5Trust8evaluateEv
+}
+# See description of bug_20653a/b in suppressions.txt.
+{
+ bug_20653a_mac
+ Memcheck:Param
+ write(buf)
+ fun:write$UNIX2003
+ fun:pager_write_pagelist
+}
+{
+ bug_20653b_mac
+ Memcheck:Param
+ write(buf)
+ fun:write$UNIX2003
+ ...
+ fun:pager_write
+}
+
+# See http://www.openradar.me/8287193
+{
+ Invalid redzone accesses in DKeyHas8Words
+ Memcheck:Unaddressable
+ fun:DKeyHas8Words
+}
+
+# See https://bugs.kde.org/show_bug.cgi?id=188572
+# This suppression is missing in Valgrind on Mac 10.6
+# TODO(glider): remove it once it arrives in the trunk.
+{
+ Unavoidable leak in setenv()
+ Memcheck:Leak
+ fun:malloc_zone_malloc
+ fun:__setenv
+ fun:setenv$UNIX2003
+}
+{
+ # Reported to Apple as rdar://6915429
+ bug_12525
+ Memcheck:Leak
+ ...
+ fun:-[CIContextImpl render:toBitmap:rowBytes:bounds:format:colorSpace:]
+}
+{
+ bug_69436
+ Memcheck:Leak
+ ...
+ fun:-[CIKernel initWithCString:noCopy:]
+ ...
+ fun:-[NSPopUpButtonCell _drawIndicatorWithFrame:inView:]
+}
+{
+ # Capturer on Mac uses OpenGL driver, which triggers several warnings.
+ # The check has to be quite generic, as different hardware graphics cards
+ # will cause different sets of warnings.
+ bug_75037
+ Memcheck:Uninitialized
+ ...
+ fun:_ZN8remoting*CapturerMac*
+}
+{
+ # See also http://openradar.appspot.com/radar?id=1235407
+ bug_77063
+ Memcheck:Free
+ fun:_ZdlPv
+ fun:_ZN15THFSPlusCatalogD2Ev
+ fun:_ZN5TNode10SetCatalogEP15THFSPlusCatalog
+ fun:_ZN15TMountPointList9AddVolumeEsb
+ fun:_ZN15TMountPointList4FindEsPN5TNode12StPopulatingE
+ fun:_ZN15TMountPointList20SupportsInvisibleBitEsPN5TNode12StPopulatingEb
+ fun:_ZNK21THFSPlusPropertyStore4OpenEbb
+ fun:_ZNK21THFSPlusPropertyStore13GetPropertiesEb
+ fun:_ZN16TFSCopyOperation22GetSourcePropertyStoreERK11THFSPlusRef
+ fun:_ZN16TFSCopyOperation13DoMoveToTrashERK11THFSPlusRef
+ fun:_ZN16TFSCopyOperation3RunEv
+ fun:_FSOperation
+ fun:_FSOperateOnObjectSync
+ fun:FSMoveObjectToTrashSync
+ fun:_Z9TrashFuncRK8FilePath
+}
+{
+ # See also http://openradar.appspot.com/radar?id=1169404
+ bug_79533a
+ Memcheck:Uninitialized
+ ...
+ fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
+ fun:CSSM_DL_DataAbortQuery
+ fun:_ZN11SSDLSession14DataAbortQueryEll
+ fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
+ fun:CSSM_DL_DataAbortQuery
+ fun:tpDbFindIssuerCrl
+ fun:tpVerifyCertGroupWithCrls
+}
+{
+ # See also http://openradar.appspot.com/radar?id=1169404
+ bug_79533b
+ Memcheck:Uninitialized
+ ...
+ fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
+ fun:CSSM_DL_DataAbortQuery
+ fun:_ZN11SSDLSession14DataAbortQueryEll
+ fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
+ fun:CSSM_DL_DataAbortQuery
+ fun:tpDbFindIssuerCrl
+ fun:tpVerifyCertGroupWithCrls
+}
+{
+ bug_85213_a
+ Memcheck:Leak
+ ...
+ fun:_CFBundleCopyDirectoryContentsAtPath
+}
+{
+ bug_85213_b
+ Memcheck:Leak
+ ...
+ fun:_CFBundleCopyInfoDictionaryInDirectoryWithVersion
+}
+{
+ bug_85213_c
+ Memcheck:Leak
+ ...
+ fun:_CFBundleURLLooksLikeBundleVersion
+}
+{
+ bug_85213_d
+ Memcheck:Leak
+ ...
+ fun:_CFBundleCreate
+ fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
+}
+{
+ bug_85213_e
+ Memcheck:Leak
+ ...
+ fun:CFBundlePreflightExecutable
+ fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
+}
+{
+ bug_85213_f
+ Memcheck:Leak
+ ...
+ fun:CFBundleGetPackageInfo
+ fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
+}
+{
+ bug_86927
+ Memcheck:Leak
+ fun:malloc
+ fun:CGSMapShmem
+ fun:CGSResolveShmemReference
+ fun:CGSScoreboard
+ fun:initCGDisplayState
+ fun:initCGDisplayMappings
+ fun:cgsInit
+ fun:pthread_once
+ fun:CGSInitialize
+ fun:CGSServerOperationState
+ fun:+[NSThemeFrame initialize]
+ fun:_class_initialize
+}
+{
+ # QTKit leak. See http://crbug.com/100772 and rdar://10319535.
+ bug_100772
+ Memcheck:Leak
+ fun:calloc
+ fun:QTMLCreateMutex
+ fun:WarholCreateGlobals
+ fun:INIT_QuickTimeLibInternal
+ fun:pthread_once
+ fun:INIT_QuickTimeLib
+ fun:EnterMovies_priv
+ fun:EnterMovies
+ fun:TundraUnitInputFromTSFileEntry
+ fun:TundraUnitVDIGInputEntry
+ fun:TundraUnitCreateFromDescription
+ fun:+[QTCaptureVDIGDevice _refreshDevices]
+ fun:+[QTCaptureVDIGDevice devicesWithIOType:]
+ fun:+[QTCaptureDevice devicesWithIOType:]
+ fun:+[QTCaptureDevice inputDevices]
+ fun:+[QTCaptureDevice inputDevicesWithMediaType:]
+ fun:+[VideoCaptureDeviceQTKit deviceNames]
+ fun:_ZN5media18VideoCaptureDevice14GetDeviceNamesEPSt4listINS0_4NameESaIS2_EE
+ fun:_ZN5media21VideoCaptureDeviceMac4InitEv
+ fun:_ZN5media18VideoCaptureDevice6CreateERKNS0_4NameE
+ fun:_ZN5media45VideoCaptureDeviceTest_OpenInvalidDevice_Test8TestBodyEv
+}
+
+# 2. Intentional unit test errors, stuff that is somehow a false positive
+# in our own code, or stuff that is so trivial it's not worth fixing.
+
+# 3. Suppressions for real WebRTC bugs that are not yet fixed.
diff --git a/trunk/tools/valgrind-webrtc/webrtc_tests.py b/trunk/tools/valgrind-webrtc/webrtc_tests.py
new file mode 100755
index 0000000..b532a1f
--- /dev/null
+++ b/trunk/tools/valgrind-webrtc/webrtc_tests.py
@@ -0,0 +1,156 @@
+#!/usr/bin/env python
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+''' Runs various WebRTC tests through valgrind_test.py.
+
+This script inherits the chrome_tests.py in Chrome, replacing its tests with
+our own in WebRTC instead.
+'''
+
+import optparse
+import sys
+
+import logging_utils
+
+import chrome_tests
+
+class WebRTCTests(chrome_tests.ChromeTests):
+ # WebRTC tests, similar functions for each tests as the Chrome tests in the
+ # parent class.
+ def TestSignalProcessing(self):
+ return self.SimpleTest("signal_processing", "signal_processing_unittests")
+
+ def TestResampler(self):
+ return self.SimpleTest("resampler", "resampler_unittests")
+
+ def TestVAD(self):
+ return self.SimpleTest("vad", "vad_unittests")
+
+ def TestCNG(self):
+ return self.SimpleTest("cng", "cng_unittests")
+
+ def TestG711(self):
+ return self.SimpleTest("g711", "g711_unittests")
+
+ def TestG722(self):
+ return self.SimpleTest("g722", "g722_unittests")
+
+ def TestPCM16B(self):
+ return self.SimpleTest("pcm16b", "pcm16b_unittests")
+
+ def TestNetEQ(self):
+ return self.SimpleTest("neteq", "neteq_unittests")
+
+ def TestAudioConferenceMixer(self):
+ return self.SimpleTest("audio_conference_mixer", "audio_conference_mixer_unittests")
+
+ def TestMediaFile(self):
+ return self.SimpleTest("media_file", "media_file_unittests")
+
+ def TestRTPRTCP(self):
+ return self.SimpleTest("rtp_rtcp", "rtp_rtcp_unittests")
+
+ def TestBWE(self):
+ return self.SimpleTest("test_bwe", "test_bwe")
+
+ def TestUDPTransport(self):
+ return self.SimpleTest("udp_transport", "udp_transport_unittests")
+
+ def TestWebRTCUtility(self):
+ return self.SimpleTest("webrtc_utility", "webrtc_utility_unittests")
+
+ def TestVP8(self):
+ return self.SimpleTest("vp8", "vp8_unittests")
+
+ def TestVideoCoding(self):
+ return self.SimpleTest("video_coding", "video_coding_unittests")
+
+ def TestVideoProcessing(self):
+ return self.SimpleTest("video_processing", "video_processing_unittests")
+
+ def TestSystemWrappers(self):
+ return self.SimpleTest("system_wrappers", "system_wrappers_unittests")
+
+ def TestTestSupport(self):
+ return self.SimpleTest("test_support", "test_support_unittests")
+
+def _main(_):
+ parser = optparse.OptionParser("usage: %prog -b <dir> -t <test> "
+ "[-t <test> ...]")
+ parser.disable_interspersed_args()
+ parser.add_option("-b", "--build_dir",
+ help="the location of the compiler output")
+ parser.add_option("-t", "--test", action="append", default=[],
+ help="which test to run, supports test:gtest_filter format "
+ "as well.")
+ parser.add_option("", "--baseline", action="store_true", default=False,
+ help="generate baseline data instead of validating")
+ parser.add_option("", "--gtest_filter",
+ help="additional arguments to --gtest_filter")
+ parser.add_option("", "--gtest_repeat",
+ help="argument for --gtest_repeat")
+ parser.add_option("-v", "--verbose", action="store_true", default=False,
+ help="verbose output - enable debug log messages")
+ parser.add_option("", "--tool", dest="valgrind_tool", default="memcheck",
+ help="specify a valgrind tool to run the tests under")
+ parser.add_option("", "--tool_flags", dest="valgrind_tool_flags", default="",
+ help="specify custom flags for the selected valgrind tool")
+ parser.add_option("", "--keep_logs", action="store_true", default=False,
+ help="store memory tool logs in the <tool>.logs directory "
+ "instead of /tmp.\nThis can be useful for tool "
+ "developers/maintainers.\nPlease note that the <tool>"
+ ".logs directory will be clobbered on tool startup.")
+ options, args = parser.parse_args()
+
+ if options.verbose:
+ logging_utils.config_root(logging.DEBUG)
+ else:
+ logging_utils.config_root()
+
+ if not options.test:
+ parser.error("--test not specified")
+
+ if len(options.test) != 1 and options.gtest_filter:
+ parser.error("--gtest_filter and multiple tests don't make sense together")
+
+ for t in options.test:
+ tests = WebRTCTests(options, args, t)
+ ret = tests.Run()
+ if ret: return ret
+ return 0
+
+if __name__ == "__main__":
+ # Overwrite the ChromeTests tests dictionary with our WebRTC tests.
+ # The cmdline option allows the user to pass any executable as parameter to
+ # the test script, which is useful when developing new tests that are not yet
+ # present in this script.
+ chrome_tests.ChromeTests._test_list = {
+ "cmdline": chrome_tests.ChromeTests.RunCmdLine,
+ "signal_processing": WebRTCTests.TestSignalProcessing,
+ "resampler": WebRTCTests.TestResampler,
+ "vad": WebRTCTests.TestVAD,
+ "cng": WebRTCTests.TestCNG,
+ "g711": WebRTCTests.TestG711,
+ "g722": WebRTCTests.TestG722,
+ "pcm16b": WebRTCTests.TestPCM16B,
+ "neteq": WebRTCTests.TestNetEQ,
+ "audio_conference_mixer": WebRTCTests.TestAudioConferenceMixer,
+ "media_file": WebRTCTests.TestMediaFile,
+ "rtp_rtcp": WebRTCTests.TestRTPRTCP,
+ "test_bwe": WebRTCTests.TestBWE,
+ "udp_transport": WebRTCTests.TestUDPTransport,
+ "webrtc_utility": WebRTCTests.TestWebRTCUtility,
+ "vp8": WebRTCTests.TestVP8,
+ "video_coding": WebRTCTests.TestVideoCoding,
+ "video_processing": WebRTCTests.TestVideoProcessing,
+ "system_wrappers": WebRTCTests.TestSystemWrappers,
+ "test_support": WebRTCTests.TestTestSupport,
+ }
+ ret = _main(sys.argv)
+ sys.exit(ret)
\ No newline at end of file
diff --git a/trunk/tools/valgrind-webrtc/webrtc_tests.sh b/trunk/tools/valgrind-webrtc/webrtc_tests.sh
new file mode 100755
index 0000000..e33b78c
--- /dev/null
+++ b/trunk/tools/valgrind-webrtc/webrtc_tests.sh
@@ -0,0 +1,102 @@
+#!/bin/bash
+
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Set up some paths and re-direct the arguments to webrtc_tests.py
+
+# This script is a copy of the chrome_tests.sh wrapper script with the following
+# changes:
+# - The locate_valgrind.sh of Chromium's Valgrind scripts dir is used to locate
+# the Valgrind framework install.
+# - webrtc_tests.py is invoked instead of chrome_tests.py.
+# - Chromium's Valgrind scripts directory is added to the PYTHONPATH to make it
+# possible to execute the Python scripts properly.
+
+export THISDIR=`dirname $0`
+ARGV_COPY="$@"
+
+# We need to set CHROME_VALGRIND iff using Memcheck or TSan-Valgrind:
+# tools/valgrind/chrome_tests.sh --tool memcheck
+# or
+# tools/valgrind/chrome_tests.sh --tool=memcheck
+# (same for "--tool=tsan")
+tool="memcheck" # Default to memcheck.
+while (( "$#" ))
+do
+ if [[ "$1" == "--tool" ]]
+ then
+ tool="$2"
+ shift
+ elif [[ "$1" =~ --tool=(.*) ]]
+ then
+ tool="${BASH_REMATCH[1]}"
+ fi
+ shift
+done
+
+NEEDS_VALGRIND=0
+NEEDS_DRMEMORY=0
+
+case "$tool" in
+ "memcheck")
+ NEEDS_VALGRIND=1
+ ;;
+ "tsan" | "tsan_rv")
+ NEEDS_VALGRIND=1
+ ;;
+ "drmemory" | "drmemory_light" | "drmemory_full")
+ NEEDS_DRMEMORY=1
+ ;;
+esac
+
+# For WebRTC, we'll use the locate_valgrind.sh script in Chromium's Valgrind
+# scripts dir to locate the Valgrind framework install
+CHROME_VALGRIND_SCRIPTS=$THISDIR/../valgrind
+
+if [ "$NEEDS_VALGRIND" == "1" ]
+then
+ CHROME_VALGRIND=`sh $CHROME_VALGRIND_SCRIPTS/locate_valgrind.sh`
+ if [ "$CHROME_VALGRIND" = "" ]
+ then
+ # locate_valgrind.sh failed
+ exit 1
+ fi
+ echo "Using valgrind binaries from ${CHROME_VALGRIND}"
+
+ PATH="${CHROME_VALGRIND}/bin:$PATH"
+ # We need to set these variables to override default lib paths hard-coded into
+ # Valgrind binary.
+ export VALGRIND_LIB="$CHROME_VALGRIND/lib/valgrind"
+ export VALGRIND_LIB_INNER="$CHROME_VALGRIND/lib/valgrind"
+fi
+
+if [ "$NEEDS_DRMEMORY" == "1" ]
+then
+ if [ -z "$DRMEMORY_COMMAND" ]
+ then
+ DRMEMORY_PATH="$THISDIR/../../third_party/drmemory"
+ DRMEMORY_SFX="$DRMEMORY_PATH/drmemory-windows-sfx.exe"
+ if [ ! -f "$DRMEMORY_SFX" ]
+ then
+ echo "Can't find Dr. Memory executables."
+ echo "See http://www.chromium.org/developers/how-tos/using-valgrind/dr-memory"
+ echo "for the instructions on how to get them."
+ exit 1
+ fi
+
+ chmod +x "$DRMEMORY_SFX" # Cygwin won't run it without +x.
+ "$DRMEMORY_SFX" -o"$DRMEMORY_PATH/unpacked" -y
+ export DRMEMORY_COMMAND="$DRMEMORY_PATH/unpacked/bin/drmemory.exe"
+ fi
+fi
+
+# Add Chrome's Valgrind scripts dir to the PYTHON_PATH since it contains
+# the scripts that are needed for this script to run
+PYTHONPATH=$THISDIR/../python/google:$CHROME_VALGRIND_SCRIPTS python \
+ "$THISDIR/webrtc_tests.py" $ARGV_COPY
diff --git a/trunk/webrtc.gyp b/trunk/webrtc.gyp
new file mode 100644
index 0000000..e18399b
--- /dev/null
+++ b/trunk/webrtc.gyp
@@ -0,0 +1,29 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ 'src/build/common.gypi', ],
+ 'targets': [
+ {
+ 'target_name': 'All',
+ 'type': 'none',
+ 'dependencies': [
+ 'src/common_audio/common_audio.gyp:*',
+ 'src/common_video/common_video.gyp:*',
+ 'src/modules/modules.gyp:*',
+ 'src/system_wrappers/source/system_wrappers.gyp:*',
+ 'src/video_engine/video_engine.gyp:*',
+ 'src/voice_engine/voice_engine.gyp:*',
+ 'test/metrics.gyp:*',
+ 'test/test.gyp:*',
+ ],
+ },
+ ],
+ 'conditions': [
+ ], # conditions
+}
